Delete erroneous WebRTC code drop r1804 as it doesn't match with any version of libjingle
Change-Id: Iab4611b13fbb06d925c3ba25cd777876b54a2096
diff --git a/.gitignore b/.gitignore
deleted file mode 100644
index 90ec22b..0000000
--- a/.gitignore
+++ /dev/null
@@ -1 +0,0 @@
-.svn
diff --git a/trunk/.gitignore b/trunk/.gitignore
deleted file mode 100644
index c27b166..0000000
--- a/trunk/.gitignore
+++ /dev/null
@@ -1,65 +0,0 @@
-*.target.mk
-*.Makefile
-*.ncb
-*.ninja
-*.props
-*.pyc
-*.rules
-*.scons
-*.sdf
-*.sln
-*.suo
-*.targets
-*.user
-*.vcproj
-*.vcxproj
-*.vcxproj.filters
-*.vpj
-*.vpw
-*.vpwhistu
-*.vtg
-*.xcodeproj
-*~
-.*.sw?
-.DS_Store
-.cproject
-.gdb_history
-.gdbinit
-.metadata
-.project
-.pydevproject
-.settings
-/build
-/chromium_deps
-/gyp-mac-tool
-/Makefile
-/out
-/resources
-/src/supplement.gypi
-/testing
-/third_party/asan
-/third_party/cygwin
-/third_party/expat
-/third_party/gaeunit
-/third_party/gold
-/third_party/google-gflags/src
-/third_party/google-visualization-python
-/third_party/jsoncpp
-/third_party/libjingle
-/third_party/libjpeg
-/third_party/libjpeg_turbo
-/third_party/libsrtp
-/third_party/libvpx
-/third_party/libyuv
-/third_party/llvm-build
-/third_party/oauth2
-/third_party/protobuf
-/third_party/valgrind
-/third_party/yasm
-/tools/clang
-/tools/gyp
-/tools/python
-/tools/valgrind
-/tools/win
-/x86-generic_out/
-/xcodebuild
diff --git a/trunk/AUTHORS b/trunk/AUTHORS
deleted file mode 100644
index c3d75d6..0000000
--- a/trunk/AUTHORS
+++ /dev/null
@@ -1,6 +0,0 @@
-# Names should be added to this file like so:
-# Name or Organization <email address>
-
-Google Inc.
-Mozilla Foundation
-Ben Strong <bstrong@gmail.com>
\ No newline at end of file
diff --git a/trunk/Android.mk b/trunk/Android.mk
deleted file mode 100644
index b7c28e6..0000000
--- a/trunk/Android.mk
+++ /dev/null
@@ -1,167 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-MY_WEBRTC_ROOT_PATH := $(call my-dir)
-
-# voice
-include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/resampler/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/signal_processing/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/common_audio/vad/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/neteq/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/cng/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/g711/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/g722/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/pcm16b/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/ilbc/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/iSAC/fix/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/codecs/iSAC/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_coding/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_conference_mixer/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_device/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/aec/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/aecm/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/agc/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/ns/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/audio_processing/utility/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/media_file/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/rtp_rtcp/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/udp_transport/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/utility/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/system_wrappers/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/voice_engine/main/source/Android.mk
-
-# video
-include $(MY_WEBRTC_ROOT_PATH)/src/common_video/jpeg/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/common_video/libyuv/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_capture/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_coding/codecs/i420/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_coding/codecs/vp8/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_coding/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_processing/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/modules/video_render/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/src/video_engine/Android.mk
-
-# third party
-include $(MY_WEBRTC_ROOT_PATH)/libvpx.mk
-
-# build .so
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-include $(LOCAL_PATH)/../../external/webrtc/android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE := libwebrtc_audio_preprocessing
-LOCAL_MODULE_TAGS := optional
-
-LOCAL_WHOLE_STATIC_LIBRARIES := \
- libwebrtc_spl \
- libwebrtc_resampler \
- libwebrtc_apm \
- libwebrtc_apm_utility \
- libwebrtc_vad \
- libwebrtc_ns \
- libwebrtc_agc \
- libwebrtc_aec \
- libwebrtc_aecm \
- libwebrtc_system_wrappers
-
-# Add Neon libraries.
-ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
-LOCAL_WHOLE_STATIC_LIBRARIES += \
- libwebrtc_aecm_neon \
- libwebrtc_ns_neon
-endif
-
-LOCAL_STATIC_LIBRARIES := \
- libprotobuf-cpp-2.3.0-lite
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-LOCAL_PRELINK_MODULE := false
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_SHARED_LIBRARY)
-
-###
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE := libwebrtc
-LOCAL_MODULE_TAGS := optional
-
-LOCAL_WHOLE_STATIC_LIBRARIES := \
- libwebrtc_system_wrappers \
- libwebrtc_audio_device \
- libwebrtc_pcm16b \
- libwebrtc_cng \
- libwebrtc_audio_coding \
- libwebrtc_rtp_rtcp \
- libwebrtc_media_file \
- libwebrtc_udp_transport \
- libwebrtc_utility \
- libwebrtc_neteq \
- libwebrtc_audio_conference_mixer \
- libwebrtc_isac \
- libwebrtc_ilbc \
- libwebrtc_isacfix \
- libwebrtc_g722 \
- libwebrtc_g711 \
- libwebrtc_voe_core \
- libwebrtc_video_render \
- libwebrtc_video_capture \
- libwebrtc_i420 \
- libwebrtc_video_coding \
- libwebrtc_video_processing \
- libwebrtc_vp8 \
- libwebrtc_vie_core \
- libwebrtc_yuv \
- libwebrtc_jpeg \
- libwebrtc_vpx
-
-# Add Neon libraries.
-ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
-LOCAL_WHOLE_STATIC_LIBRARIES += \
- libwebrtc_isacfix_neon
-endif
-
-LOCAL_STATIC_LIBRARIES := \
- libyuv_static
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport \
- libjpeg \
- libGLESv2 \
- libOpenSLES \
- libwebrtc_audio_preprocessing
-
-LOCAL_PRELINK_MODULE := false
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_SHARED_LIBRARY)
-
-# test apps, they're for test only; all these test apps have LOCAL_MODULE_TAGS:=tests
-# voice engine test apps
-include $(MY_WEBRTC_ROOT_PATH)/src/voice_engine/main/test/cmd_test/Android.mk
-#include $(MY_WEBRTC_ROOT_PATH)/src/voice_engine/main/test/auto_test/Android.mk
-# video engien test apps
-include $(MY_WEBRTC_ROOT_PATH)/src/video_engine/main/test/android_test/Android.mk
-#include $(MY_WEBRTC_ROOT_PATH)/src/video_engine/test/auto_test/android/Android.mk
diff --git a/trunk/DEPS b/trunk/DEPS
deleted file mode 100644
index 0c8892e..0000000
--- a/trunk/DEPS
+++ /dev/null
@@ -1,130 +0,0 @@
-vars = {
- # Use this googlecode_url variable only if there is an internal mirror for it.
- # If you do not know, use the full path while defining your new deps entry.
- "googlecode_url": "http://%s.googlecode.com/svn",
- "chromium_trunk" : "http://src.chromium.org/svn/trunk",
- "chromium_revision": "122775",
-
- # External resources like video and audio files used for testing purposes.
- # Downloaded on demand when needed.
- "webrtc_resources_revision": "8",
-}
-
-# NOTE: Prefer revision numbers to tags for svn deps. Use http rather than
-# https; the latter can cause problems for users behind proxies.
-deps = {
- "trunk/chromium_deps":
- File(Var("chromium_trunk") + "/src/DEPS@" + Var("chromium_revision")),
-
- "trunk/build":
- Var("chromium_trunk") + "/src/build@" + Var("chromium_revision"),
-
- "trunk/testing":
- Var("chromium_trunk") + "/src/testing@" + Var("chromium_revision"),
-
- "trunk/testing/gmock":
- From("trunk/chromium_deps", "src/testing/gmock"),
-
- "trunk/testing/gtest":
- From("trunk/chromium_deps", "src/testing/gtest"),
-
- "trunk/third_party/expat":
- Var("chromium_trunk") + "/src/third_party/expat@" + Var("chromium_revision"),
-
- # Used by tools/quality_tracking.
- "trunk/third_party/gaeunit":
- "http://code.google.com/p/gaeunit.git@e16d5bd4",
-
- "trunk/third_party/google-gflags/src":
- (Var("googlecode_url") % "google-gflags") + "/trunk/src@45",
-
- # Used by tools/quality_tracking/dashboard and tools/python_charts.
- "trunk/third_party/google-visualization-python":
- (Var("googlecode_url") % "google-visualization-python") + "/trunk@15",
-
- "trunk/third_party/libjpeg":
- Var("chromium_trunk") + "/src/third_party/libjpeg@" + Var("chromium_revision"),
-
- "trunk/third_party/libjpeg_turbo":
- From("trunk/chromium_deps", "src/third_party/libjpeg_turbo"),
-
- "trunk/third_party/libvpx/source/libvpx":
- "http://git.chromium.org/webm/libvpx.git@v1.0.0",
-
- "trunk/third_party/libyuv":
- (Var("googlecode_url") % "libyuv") + "/trunk@190",
-
- "trunk/third_party/protobuf":
- Var("chromium_trunk") + "/src/third_party/protobuf@" + Var("chromium_revision"),
-
- # Used by tools/quality_tracking.
- "trunk/third_party/oauth2":
- "http://github.com/simplegeo/python-oauth2.git@a83f4a29",
-
- "trunk/third_party/yasm":
- Var("chromium_trunk") + "/src/third_party/yasm@" + Var("chromium_revision"),
-
- "trunk/third_party/yasm/source/patched-yasm":
- From("trunk/chromium_deps", "src/third_party/yasm/source/patched-yasm"),
-
- "trunk/tools/clang":
- Var("chromium_trunk") + "/src/tools/clang@" + Var("chromium_revision"),
-
- "trunk/tools/gyp":
- From("trunk/chromium_deps", "src/tools/gyp"),
-
- "trunk/tools/python":
- Var("chromium_trunk") + "/src/tools/python@" + Var("chromium_revision"),
-
- "trunk/tools/valgrind":
- Var("chromium_trunk") + "/src/tools/valgrind@" + Var("chromium_revision"),
-
- # Needed by build/common.gypi.
- "trunk/tools/win/supalink":
- Var("chromium_trunk") + "/src/tools/win/supalink@" + Var("chromium_revision"),
-}
-
-deps_os = {
- "win": {
- "trunk/third_party/cygwin":
- Var("chromium_trunk") + "/deps/third_party/cygwin@66844",
-
- # Used by libjpeg-turbo.
- "trunk/third_party/yasm/binaries":
- From("trunk/chromium_deps", "src/third_party/yasm/binaries"),
- },
- "unix": {
- "trunk/third_party/gold":
- From("trunk/chromium_deps", "src/third_party/gold"),
- },
-}
-
-hooks = [
- {
- # Create a supplement.gypi file under trunk/. This file will be picked up
- # by gyp and we use it to set Chromium related variables (inside_chromium_build)
- # to 0 and enable the standalone build.
- "pattern": ".",
- "action": ["python", "trunk/tools/create_supplement_gypi.py", "trunk/src/supplement.gypi"],
- },
- {
- # Pull clang on mac. If nothing changed, or on non-mac platforms, this takes
- # zero seconds to run. If something changed, it downloads a prebuilt clang.
- "pattern": ".",
- "action": ["python", "trunk/tools/clang/scripts/update.py", "--mac-only"],
- },
- {
- # Download test resources, i.e. video and audio files. If the latest
- # version is already downloaded, this takes zero seconds to run.
- # If a newer version or no current download exists, it will download
- # the resources and extract them.
- "pattern": ".",
- "action": ["python", "trunk/tools/resources/update.py"],
- },
- {
- # A change to a .gyp, .gypi, or to GYP itself should run the generator.
- "pattern": ".",
- "action": ["python", "trunk/build/gyp_chromium", "--depth=trunk", "trunk/webrtc.gyp"],
- },
-]
-
diff --git a/trunk/LICENSE b/trunk/LICENSE
deleted file mode 100644
index dd4a345..0000000
--- a/trunk/LICENSE
+++ /dev/null
@@ -1 +0,0 @@
-Refer to src/LICENSE.
diff --git a/trunk/LICENSE_THIRD_PARTY b/trunk/LICENSE_THIRD_PARTY
deleted file mode 100644
index d47c055..0000000
--- a/trunk/LICENSE_THIRD_PARTY
+++ /dev/null
@@ -1 +0,0 @@
-Refer to src/LICENSE_THIRD_PARTY.
diff --git a/trunk/OWNERS b/trunk/OWNERS
deleted file mode 100644
index b110a52..0000000
--- a/trunk/OWNERS
+++ /dev/null
@@ -1,5 +0,0 @@
-henrika@webrtc.org
-niklas.enbom@webrtc.org
-andrew@webrtc.org
-tina.legrand@webrtc.org
-tommi@webrtc.org
\ No newline at end of file
diff --git a/trunk/PATENTS b/trunk/PATENTS
deleted file mode 100644
index 5cb83ec..0000000
--- a/trunk/PATENTS
+++ /dev/null
@@ -1 +0,0 @@
-Refer to src/PATENTS.
diff --git a/trunk/PRESUBMIT.py b/trunk/PRESUBMIT.py
deleted file mode 100644
index 56c8d5b..0000000
--- a/trunk/PRESUBMIT.py
+++ /dev/null
@@ -1,144 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-def _LicenseHeader(input_api):
- """Returns the license header regexp."""
- license_header = (
- r'.*? Copyright \(c\) %(year)s The WebRTC project authors\. '
- r'All Rights Reserved\.\n'
- r'.*?\n'
- r'.*? Use of this source code is governed by a BSD-style license\n'
- r'.*? that can be found in the LICENSE file in the root of the source\n'
- r'.*? tree\. An additional intellectual property rights grant can be '
- r'found\n'
- r'.*? in the file PATENTS\. All contributing project authors may\n'
- r'.*? be found in the AUTHORS file in the root of the source tree\.\n'
- ) % {
- 'year': input_api.time.strftime('%Y'),
- }
- return license_header
-
-def _CheckNoIOStreamInHeaders(input_api, output_api):
- """Checks to make sure no .h files include <iostream>."""
- files = []
- pattern = input_api.re.compile(r'^#include\s*<iostream>',
- input_api.re.MULTILINE)
- for f in input_api.AffectedSourceFiles(input_api.FilterSourceFile):
- if not f.LocalPath().endswith('.h'):
- continue
- contents = input_api.ReadFile(f)
- if pattern.search(contents):
- files.append(f)
-
- if len(files):
- return [ output_api.PresubmitError(
- 'Do not #include <iostream> in header files, since it inserts static ' +
- 'initialization into every file including the header. Instead, ' +
- '#include <ostream>. See http://crbug.com/94794',
- files) ]
- return []
-
-def _CheckNoFRIEND_TEST(input_api, output_api):
- """Make sure that gtest's FRIEND_TEST() macro is not used, the
- FRIEND_TEST_ALL_PREFIXES() macro from testsupport/gtest_prod_util.h should be
- used instead since that allows for FLAKY_, FAILS_ and DISABLED_ prefixes."""
- problems = []
-
- file_filter = lambda f: f.LocalPath().endswith(('.cc', '.h'))
- for f in input_api.AffectedFiles(file_filter=file_filter):
- for line_num, line in f.ChangedContents():
- if 'FRIEND_TEST(' in line:
- problems.append(' %s:%d' % (f.LocalPath(), line_num))
-
- if not problems:
- return []
- return [output_api.PresubmitPromptWarning('WebRTC\'s code should not use '
- 'gtest\'s FRIEND_TEST() macro. Include testsupport/gtest_prod_util.h and '
- 'use FRIEND_TEST_ALL_PREFIXES() instead.\n' + '\n'.join(problems))]
-
-def _CheckNewFilesLintClean(input_api, output_api, source_file_filter=None):
- """Checks that all NEW '.cc' and '.h' files pass cpplint.py.
- This check is based on _CheckChangeLintsClean in
- depot_tools/presubmit_canned_checks.py but has less filters and only checks
- added files."""
- result = []
-
- # Initialize cpplint.
- import cpplint
- # Access to a protected member _XX of a client class
- # pylint: disable=W0212
- cpplint._cpplint_state.ResetErrorCounts()
-
- # Justifications for each filter:
- #
- # - build/header_guard : WebRTC coding style says they should be prefixed
- # with WEBRTC_, which is not possible to configure in
- # cpplint.py.
- cpplint._SetFilters('-build/header_guard')
-
- # Use the strictest verbosity level for cpplint.py (level 1) which is the
- # default when running cpplint.py from command line.
- # To make it possible to work with not-yet-converted code, we're only applying
- # it to new (or moved/renamed) files.
- verbosity_level = 1
- files = []
- for f in input_api.AffectedSourceFiles(source_file_filter):
- # Note that moved/renamed files also count as added for svn.
- if (f.Action() == 'A'):
- files.append(f.AbsoluteLocalPath())
- for file_name in files:
- cpplint.ProcessFile(file_name, verbosity_level)
-
- if cpplint._cpplint_state.error_count > 0:
- if input_api.is_committing:
- # TODO(kjellander): Change back to PresubmitError below when we're
- # confident with the lint settings.
- res_type = output_api.PresubmitPromptWarning
- else:
- res_type = output_api.PresubmitPromptWarning
- result = [res_type('Changelist failed cpplint.py check.')]
-
- return result
-
-def _CommonChecks(input_api, output_api):
- """Checks common to both upload and commit."""
- # TODO(kjellander): Use presubmit_canned_checks.PanProjectChecks too.
- results = []
- results.extend(input_api.canned_checks.CheckLongLines(
- input_api, output_api))
- results.extend(input_api.canned_checks.CheckChangeHasNoTabs(
- input_api, output_api))
- results.extend(input_api.canned_checks.CheckChangeHasNoStrayWhitespace(
- input_api, output_api))
- results.extend(input_api.canned_checks.CheckChangeTodoHasOwner(
- input_api, output_api))
- results.extend(_CheckNewFilesLintClean(input_api, output_api))
- results.extend(input_api.canned_checks.CheckLicense(
- input_api, output_api, _LicenseHeader(input_api)))
- results.extend(_CheckNoIOStreamInHeaders(input_api, output_api))
- results.extend(_CheckNoFRIEND_TEST(input_api, output_api))
- return results
-
-def CheckChangeOnUpload(input_api, output_api):
- results = []
- results.extend(_CommonChecks(input_api, output_api))
- return results
-
-def CheckChangeOnCommit(input_api, output_api):
- results = []
- results.extend(_CommonChecks(input_api, output_api))
- results.extend(input_api.canned_checks.CheckOwners(input_api, output_api))
- results.extend(input_api.canned_checks.CheckChangeWasUploaded(
- input_api, output_api))
- results.extend(input_api.canned_checks.CheckChangeHasDescription(
- input_api, output_api))
- results.extend(input_api.canned_checks.CheckChangeHasBugField(
- input_api, output_api))
- results.extend(input_api.canned_checks.CheckChangeHasTestField(
- input_api, output_api))
- return results
diff --git a/trunk/WATCHLISTS b/trunk/WATCHLISTS
deleted file mode 100644
index 1b592f5..0000000
--- a/trunk/WATCHLISTS
+++ /dev/null
@@ -1,112 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-# Inspired by chromium.org:
-# http://dev.chromium.org/developers/contributing-code/watchlists
-
-{
- 'WATCHLIST_DEFINITIONS': {
- 'this_file': {
- 'filepath': '^WATCHLISTS$',
- },
- 'all_src': {
- 'filepath': 'src/.*',
- },
- 'root_files': {
- # src/build/ and non-recursive contents of ./ and src/
- 'filepath': '^[^/]*$|src/[^/]*$|src/build/.*',
- },
- 'documented_interfaces': {
- 'filepath': 'src/[^/]*\.h$|'\
- 'src/video_engine/main/interface/.*|'\
- 'src/voice_engine/main/interface/.*',
- },
- 'build_files': {
- 'filepath': '\.gyp$|\.gypi$|Android\.mk$',
- },
- 'java_files': {
- 'filepath': '\.java$|\.xml$',
- },
- 'video_engine': {
- 'filepath': 'src/video_engine/.*',
- },
- 'voice_engine': {
- 'filepath': 'src/voice_engine/.*',
- },
- 'common_audio': {
- 'filepath': 'src/common_audio/.*',
- },
- 'video_capture': {
- 'filepath': 'src/modules/video_capture/.*',
- },
- 'video_render': {
- 'filepath': 'src/modules/video_render/.*',
- },
- 'audio_device': {
- 'filepath': 'src/modules/audio_device/.*',
- },
- 'audio_coding': {
- 'filepath': 'src/modules/audio_coding/.*',
- },
- 'neteq': {
- 'filepath': 'src/modules/audio_coding/neteq/.*',
- },
- 'audio_processing': {
- 'filepath': 'src/modules/audio_processing/.*',
- },
- 'video_codecs': {
- 'filepath': 'src/modules/video_coding/codecs/.*',
- },
- 'video_coding': {
- 'filepath': 'src/modules/video_coding/.*',
- },
- 'rtp_rtcp': {
- 'filepath': 'src/modules/rtp_rtcp/.*'
- },
- 'system_wrappers': {
- 'filepath': 'src/system_wrappers/.*',
- },
- },
-
- 'WATCHLISTS': {
- 'this_file': [''],
- 'all_src': ['tterriberry@mozilla.com',
- 'giles@mozilla.com'],
- 'root_files': ['andrew@webrtc.org',
- 'niklas.enbom@webrtc.org'],
- 'documented_interfaces': ['interface-changes@webrtc.org',
- 'rwolff@gocast.it'],
- 'build_files': ['leozwang@webrtc.org'],
- 'java_files': ['leozwang@webrtc.org'],
- 'common_audio': ['bjornv@webrtc.org',
- 'andrew@webrtc.org'],
- 'video_engine': ['mflodman@webrtc.org',
- 'perkj@webrtc.org'],
- 'voice_engine': ['henrika@webrtc.org'],
- 'video_capture': ['mflodman@webrtc.org',
- 'perkj@webrtc.org',
- 'leozwang@webrtc.org'],
- 'video_render': ['mflodman@webrtc.org',
- 'perkj@webrtc.org',
- 'leozwang@webrtc.org'],
- 'audio_device': ['henrika@webrtc.org',
- 'leozwang@webrtc.org'],
- 'audio_coding': ['tina.legrand@webrtc.org'],
- 'neteq': ['henrik.lundin@webrtc.org'],
- 'audio_processing': ['andrew@webrtc.org',
- 'bjornv@webrtc.org'],
- 'video_codecs': ['henrik.lundin@webrtc.org',
- 'pwestin@webrtc.org'],
- 'video_coding': ['stefan@webrtc.org'],
- 'rtp_rtcp': ['mflodman@webrtc.org',
- 'pwestin@webrtc.org'],
- 'system_wrappers': ['mflodman@webrtc.org',
- 'henrika@webrtc.org',
- 'andrew@webrtc.org'],
- },
-}
diff --git a/trunk/android-webrtc.mk b/trunk/android-webrtc.mk
deleted file mode 100644
index cd495b0..0000000
--- a/trunk/android-webrtc.mk
+++ /dev/null
@@ -1,46 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-# These defines will apply to all source files
-# Think again before changing it
-MY_WEBRTC_COMMON_DEFS := \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_THREAD_RR' \
- '-DWEBRTC_CLOCK_TYPE_REALTIME' \
- '-DWEBRTC_ANDROID'
-# The following macros are used by modules,
-# we might need to re-organize them
-# '-DWEBRTC_ANDROID_OPENSLES' [module audio_device]
-# '-DNETEQ_VOICEENGINE_CODECS' [module audio_coding neteq]
-# '-DWEBRTC_MODULE_UTILITY_VIDEO' [module media_file] [module utility]
-ifeq ($(TARGET_ARCH),arm)
-MY_WEBRTC_COMMON_DEFS += \
- '-DWEBRTC_ARCH_ARM'
-# '-DWEBRTC_DETECT_ARM_NEON' # only used in a build configuration without Neon
-# TODO(kma): figure out if the above define could be moved to NDK build only.
-
-# TODO(kma): test if the code under next two macros works with generic GCC compilers
-ifeq ($(ARCH_ARM_HAVE_NEON),true)
-MY_WEBRTC_COMMON_DEFS += \
- '-DWEBRTC_ARCH_ARM_NEON'
-MY_ARM_CFLAGS_NEON := \
- -flax-vector-conversions
-endif
-
-ifneq (,$(filter '-DWEBRTC_DETECT_ARM_NEON' '-DWEBRTC_ARCH_ARM_NEON', \
- $(MY_WEBRTC_COMMON_DEFS)))
-WEBRTC_BUILD_NEON_LIBS := true
-endif
-
-ifeq ($(ARCH_ARM_HAVE_ARMV7A),true)
-MY_WEBRTC_COMMON_DEFS += \
- '-DWEBRTC_ARCH_ARM_V7A'
-endif
-
-endif # ifeq ($(TARGET_ARCH),arm)
\ No newline at end of file
diff --git a/trunk/codereview.settings b/trunk/codereview.settings
deleted file mode 100644
index b70057e..0000000
--- a/trunk/codereview.settings
+++ /dev/null
@@ -1,9 +0,0 @@
-# This file is used by gcl to get repository specific information.
-CODE_REVIEW_SERVER: webrtc-codereview.appspot.com
-#CC_LIST:
-#VIEW_VC:
-#STATUS:
-TRY_ON_UPLOAD: False
-#TRYSERVER_SVN_URL:
-#GITCL_PREUPLOAD:
-#GITCL_PREDCOMMIT:
diff --git a/trunk/libvpx.mk b/trunk/libvpx.mk
deleted file mode 100644
index 07c04dc..0000000
--- a/trunk/libvpx.mk
+++ /dev/null
@@ -1,107 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-include $(CLEAR_VARS)
-
-MY_LIBVPX_PATH = ../libvpx
-
-LOCAL_SRC_FILES = \
- $(MY_LIBVPX_PATH)/vp8/common/alloccommon.c \
- $(MY_LIBVPX_PATH)/vp8/common/blockd.c \
- $(MY_LIBVPX_PATH)/vp8/common/debugmodes.c \
- $(MY_LIBVPX_PATH)/vp8/common/entropy.c \
- $(MY_LIBVPX_PATH)/vp8/common/entropymode.c \
- $(MY_LIBVPX_PATH)/vp8/common/entropymv.c \
- $(MY_LIBVPX_PATH)/vp8/common/extend.c \
- $(MY_LIBVPX_PATH)/vp8/common/filter.c \
- $(MY_LIBVPX_PATH)/vp8/common/findnearmv.c \
- $(MY_LIBVPX_PATH)/vp8/common/generic/systemdependent.c \
- $(MY_LIBVPX_PATH)/vp8/common/idctllm.c \
- $(MY_LIBVPX_PATH)/vp8/common/invtrans.c \
- $(MY_LIBVPX_PATH)/vp8/common/loopfilter.c \
- $(MY_LIBVPX_PATH)/vp8/common/loopfilter_filters.c \
- $(MY_LIBVPX_PATH)/vp8/common/mbpitch.c \
- $(MY_LIBVPX_PATH)/vp8/common/modecont.c \
- $(MY_LIBVPX_PATH)/vp8/common/modecontext.c \
- $(MY_LIBVPX_PATH)/vp8/common/quant_common.c \
- $(MY_LIBVPX_PATH)/vp8/common/recon.c \
- $(MY_LIBVPX_PATH)/vp8/common/reconinter.c \
- $(MY_LIBVPX_PATH)/vp8/common/reconintra.c \
- $(MY_LIBVPX_PATH)/vp8/common/reconintra4x4.c \
- $(MY_LIBVPX_PATH)/vp8/common/setupintrarecon.c \
- $(MY_LIBVPX_PATH)/vp8/common/swapyv12buffer.c \
- $(MY_LIBVPX_PATH)/vp8/common/textblit.c \
- $(MY_LIBVPX_PATH)/vp8/common/treecoder.c \
- $(MY_LIBVPX_PATH)/vp8/vp8_cx_iface.c \
- $(MY_LIBVPX_PATH)/vp8/vp8_dx_iface.c \
- $(MY_LIBVPX_PATH)/vpx_config.c \
- $(MY_LIBVPX_PATH)/vpx/src/vpx_codec.c \
- $(MY_LIBVPX_PATH)/vpx/src/vpx_decoder.c \
- $(MY_LIBVPX_PATH)/vpx/src/vpx_image.c \
- $(MY_LIBVPX_PATH)/vpx_mem/vpx_mem.c \
- $(MY_LIBVPX_PATH)/vpx_scale/generic/vpxscale.c \
- $(MY_LIBVPX_PATH)/vpx_scale/generic/yv12config.c \
- $(MY_LIBVPX_PATH)/vpx_scale/generic/yv12extend.c \
- $(MY_LIBVPX_PATH)/vpx_scale/generic/gen_scalers.c \
- $(MY_LIBVPX_PATH)/vpx_scale/generic/scalesystemdependent.c \
- $(MY_LIBVPX_PATH)/vpx/src/vpx_encoder.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/bitstream.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/boolhuff.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/dct.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/encodeframe.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/encodeintra.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/encodemb.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/encodemv.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/ethreading.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/firstpass.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/generic/csystemdependent.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/mcomp.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/modecosts.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/pickinter.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/picklpf.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/psnr.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/quantize.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/ratectrl.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/rdopt.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/sad_c.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/segmentation.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/tokenize.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/treewriter.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/onyx_if.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/temporal_filter.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/variance_c.c \
- $(MY_LIBVPX_PATH)/vp8/decoder/dboolhuff.c \
- $(MY_LIBVPX_PATH)/vp8/decoder/decodemv.c \
- $(MY_LIBVPX_PATH)/vp8/decoder/decodframe.c \
- $(MY_LIBVPX_PATH)/vp8/decoder/dequantize.c \
- $(MY_LIBVPX_PATH)/vp8/decoder/detokenize.c \
- $(MY_LIBVPX_PATH)/vp8/decoder/generic/dsystemdependent.c \
- $(MY_LIBVPX_PATH)/vp8/decoder/onyxd_if.c \
- $(MY_LIBVPX_PATH)/vp8/decoder/reconintra_mt.c \
- $(MY_LIBVPX_PATH)/vp8/decoder/threading.c \
- $(MY_LIBVPX_PATH)/vp8/decoder/idct_blk.c \
- $(MY_LIBVPX_PATH)/vp8/common/arm/arm_systemdependent.c \
- $(MY_LIBVPX_PATH)/vp8/encoder/arm/arm_csystemdependent.c \
- $(MY_LIBVPX_PATH)/vp8/decoder/arm/arm_dsystemdependent.c \
-
-LOCAL_CFLAGS := \
- -DHAVE_CONFIG_H=vpx_config.h \
- -include $(LOCAL_PATH)/third_party/libvpx/source/config/android/vpx_config.h
-
-LOCAL_MODULE := libwebrtc_vpx
-
-LOCAL_C_INCLUDES := \
- external/libvpx \
- external/libvpx/vpx_ports \
- external/libvpx/vp8/common \
- external/libvpx/vp8/encoder \
- external/libvpx/vp8 \
- external/libvpx/vpx_codec
-
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/license_template.txt b/trunk/license_template.txt
deleted file mode 100644
index 5a3e653..0000000
--- a/trunk/license_template.txt
+++ /dev/null
@@ -1,10 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
diff --git a/trunk/peerconnection/DEPS b/trunk/peerconnection/DEPS
deleted file mode 100644
index 7f1e4fe..0000000
--- a/trunk/peerconnection/DEPS
+++ /dev/null
@@ -1,132 +0,0 @@
-vars = {
- # Use this googlecode_url variable only if there is an internal mirror for it.
- # If you do not know, use the full path while defining your new deps entry.
- "googlecode_url": "http://%s.googlecode.com/svn",
- "chromium_trunk" : "http://src.chromium.org/svn/trunk",
- "chromium_revision": "120526",
- "webrtc_revision": "1538",
- "libjingle_revision": "115",
-}
-
-# NOTE: Prefer revision numbers to tags for svn deps. Use http rather than
-# https; the latter can cause problems for users behind proxies.
-deps = {
- # WebRTC deps.
- "trunk/src":
- (Var("googlecode_url") % "webrtc") + "/trunk/src@" + Var("webrtc_revision"),
-
- "trunk/tools":
- (Var("googlecode_url") % "webrtc") + "/trunk/tools@" + Var("webrtc_revision"),
-
- "trunk/test":
- (Var("googlecode_url") % "webrtc") + "/trunk/test@" + Var("webrtc_revision"),
-
- "trunk/third_party/google-gflags":
- (Var("googlecode_url") % "webrtc") + "/trunk/third_party/google-gflags@" + Var("webrtc_revision"),
-
- "trunk/third_party/libvpx":
- (Var("googlecode_url") % "webrtc") + "/trunk/third_party/libvpx@" + Var("webrtc_revision"),
-
- "trunk/build":
- Var("chromium_trunk") + "/src/build@" + Var("chromium_revision"),
-
- "trunk/testing":
- Var("chromium_trunk") + "/src/testing@" + Var("chromium_revision"),
-
- "trunk/testing/gtest":
- (Var("googlecode_url") % "googletest") + "/trunk@573",
-
- "trunk/testing/gmock":
- (Var("googlecode_url") % "googlemock") + "/trunk@386",
-
- "trunk/tools/gyp":
- (Var("googlecode_url") % "gyp") + "/trunk@1187",
-
- # Needed by build/common.gypi.
- "trunk/tools/win/supalink":
- Var("chromium_trunk") + "/src/tools/win/supalink@" + Var("chromium_revision"),
-
- "trunk/tools/clang/scripts":
- Var("chromium_trunk") + "/src/tools/clang/scripts@" + Var("chromium_revision"),
-
- "trunk/tools/python":
- Var("chromium_trunk") + "/src/tools/python@" + Var("chromium_revision"),
-
- "trunk/tools/valgrind":
- Var("chromium_trunk") + "/src/tools/valgrind@" + Var("chromium_revision"),
-
- "trunk/third_party/protobuf/":
- Var("chromium_trunk") + "/src/third_party/protobuf@" + Var("chromium_revision"),
-
- "trunk/third_party/libvpx/source/libvpx":
- "http://git.chromium.org/webm/libvpx.git@e479379a",
-
- "trunk/third_party/libjpeg_turbo/":
- Var("chromium_trunk") + "/deps/third_party/libjpeg_turbo@119959",
-
- "trunk/third_party/libjpeg/":
- Var("chromium_trunk") + "/src/third_party/libjpeg@" + Var("chromium_revision"),
-
- "trunk/third_party/yasm/":
- Var("chromium_trunk") + "/src/third_party/yasm@" + Var("chromium_revision"),
-
- "trunk/third_party/expat/":
- Var("chromium_trunk") + "/src/third_party/expat@" + Var("chromium_revision"),
-
- "trunk/third_party/google-gflags/src":
- (Var("googlecode_url") % "google-gflags") + "/trunk/src@45",
-
- "trunk/third_party/yasm/source/patched-yasm":
- Var("chromium_trunk") + "/deps/third_party/yasm/patched-yasm@73761",
-
- # Used by libjpeg-turbo
- "trunk/third_party/yasm/binaries":
- Var("chromium_trunk") + "/deps/third_party/yasm/binaries@74228",
-
- # TODO(andrew): roll to 164 after fixing:
- # http://code.google.com/p/webrtc/issues/detail?id=267
- "trunk/third_party/libyuv":
- (Var("googlecode_url") % "libyuv") + "/trunk@121",
-
- # libjingle deps.
- "trunk/third_party/libjingle/source":
- (Var("googlecode_url") % "libjingle") + "/trunk/@" + Var("libjingle_revision"),
-
- "trunk/third_party/libsrtp/":
- Var("chromium_trunk") + "/deps/third_party/libsrtp@119742",
-
- "trunk/third_party/jsoncpp/":
- Var("chromium_trunk") + "/src/third_party/jsoncpp@" + Var("chromium_revision"),
-
- "trunk/third_party/jsoncpp/source":
- "http://jsoncpp.svn.sourceforge.net/svnroot/jsoncpp/trunk/jsoncpp@248",
-}
-
-deps_os = {
- "win": {
- "trunk/third_party/cygwin/":
- Var("chromium_trunk") + "/deps/third_party/cygwin@66844",
- }
-}
-
-hooks = [
- {
- # Create a supplement.gypi file under trunk/. This file will be picked up
- # by gyp and we use it to set Chromium related variables (inside_chromium_build)
- # to 0 and enable the standalone build.
- "pattern": ".",
- "action": ["python", "trunk/tools/create_supplement_gypi.py", "trunk/src/supplement.gypi"],
- },
- {
- # Pull clang on mac. If nothing changed, or on non-mac platforms, this takes
- # zero seconds to run. If something changed, it downloads a prebuilt clang.
- "pattern": ".",
- "action": ["python", "trunk/tools/clang/scripts/update.py", "--mac-only"],
- },
- {
- # A change to a .gyp, .gypi, or to GYP itself should run the generator.
- "pattern": ".",
- "action": ["python", "trunk/build/gyp_chromium", "--depth=trunk", "trunk/peerconnection.gyp"],
- },
-]
-
diff --git a/trunk/peerconnection/OWNERS b/trunk/peerconnection/OWNERS
deleted file mode 100644
index 1527445..0000000
--- a/trunk/peerconnection/OWNERS
+++ /dev/null
@@ -1,5 +0,0 @@
-henrike@webrtc.org
-mallinath@webrtc.org
-perkj@webrtc.org
-wu@webrtc.org
-tommi@webrtc.org
diff --git a/trunk/peerconnection/README b/trunk/peerconnection/README
deleted file mode 100644
index bfe1234..0000000
--- a/trunk/peerconnection/README
+++ /dev/null
@@ -1,10 +0,0 @@
-This folder can be used to pull together the chromium version of webrtc
-and libjingle, and build the peerconnection sample client and server.
-
-Follow the instructions for WebRTC here:
-http://www.webrtc.org/reference/getting-started
-
-but substitute this "gclient config" command:
-gclient config --name trunk
-http://webrtc.googlecode.com/svn/trunk/peerconnection
-
diff --git a/trunk/peerconnection/peerconnection.gyp b/trunk/peerconnection/peerconnection.gyp
deleted file mode 100644
index e928194..0000000
--- a/trunk/peerconnection/peerconnection.gyp
+++ /dev/null
@@ -1,113 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'includes': [ 'src/build/common.gypi', ],
- 'variables': {
- 'peerconnection_sample': 'third_party/libjingle/source/talk/examples/peerconnection',
- },
-
- 'targets': [
- {
- 'target_name': 'peerconnection_server',
- 'type': 'executable',
- 'sources': [
- '<(peerconnection_sample)/server/data_socket.cc',
- '<(peerconnection_sample)/server/data_socket.h',
- '<(peerconnection_sample)/server/main.cc',
- '<(peerconnection_sample)/server/peer_channel.cc',
- '<(peerconnection_sample)/server/peer_channel.h',
- '<(peerconnection_sample)/server/utils.cc',
- '<(peerconnection_sample)/server/utils.h',
- ],
- 'include_dirs': [
- 'third_party/libjingle/source',
- ],
- },
- ],
- 'conditions': [
- ['OS=="win"', {
- 'targets': [
- {
- 'target_name': 'peerconnection_client',
- 'type': 'executable',
- 'sources': [
- '<(peerconnection_sample)/client/conductor.cc',
- '<(peerconnection_sample)/client/conductor.h',
- '<(peerconnection_sample)/client/defaults.cc',
- '<(peerconnection_sample)/client/defaults.h',
- '<(peerconnection_sample)/client/main.cc',
- '<(peerconnection_sample)/client/main_wnd.cc',
- '<(peerconnection_sample)/client/main_wnd.h',
- '<(peerconnection_sample)/client/peer_connection_client.cc',
- '<(peerconnection_sample)/client/peer_connection_client.h',
- 'third_party/libjingle/source/talk/base/win32socketinit.cc',
- 'third_party/libjingle/source/talk/base/win32socketserver.cc',
- ],
- 'msvs_settings': {
- 'VCLinkerTool': {
- 'SubSystem': '2', # Windows
- },
- },
- 'dependencies': [
- 'third_party/libjingle/libjingle.gyp:libjingle_app',
- ],
- 'include_dirs': [
- 'src',
- 'src/modules/interface',
- 'third_party/libjingle/source',
- ],
- },
- ], # targets
- }, ], # OS="win"
- ['OS=="linux"', {
- 'targets': [
- {
- 'target_name': 'peerconnection_client',
- 'type': 'executable',
- 'sources': [
- '<(peerconnection_sample)/client/conductor.cc',
- '<(peerconnection_sample)/client/conductor.h',
- '<(peerconnection_sample)/client/defaults.cc',
- '<(peerconnection_sample)/client/defaults.h',
- '<(peerconnection_sample)/client/linux/main.cc',
- '<(peerconnection_sample)/client/linux/main_wnd.cc',
- '<(peerconnection_sample)/client/linux/main_wnd.h',
- '<(peerconnection_sample)/client/peer_connection_client.cc',
- '<(peerconnection_sample)/client/peer_connection_client.h',
- ],
- 'dependencies': [
- 'third_party/libjingle/libjingle.gyp:libjingle_app',
- # TODO(tommi): Switch to this and remove specific gtk dependency
- # sections below for cflags and link_settings.
- # '<(DEPTH)/build/linux/system.gyp:gtk',
- ],
- 'include_dirs': [
- 'src',
- 'src/modules/interface',
- 'third_party/libjingle/source',
- ],
- 'cflags': [
- '<!@(pkg-config --cflags gtk+-2.0)',
- ],
- 'link_settings': {
- 'ldflags': [
- '<!@(pkg-config --libs-only-L --libs-only-other gtk+-2.0 gthread-2.0)',
- ],
- 'libraries': [
- '<!@(pkg-config --libs-only-l gtk+-2.0 gthread-2.0)',
- '-lX11',
- '-lXext',
- ],
- },
- },
- ], # targets
- }, ], # OS="linux"
- ],
-
-}
diff --git a/trunk/peerconnection/third_party/libjingle/libjingle.gyp b/trunk/peerconnection/third_party/libjingle/libjingle.gyp
deleted file mode 100644
index 48939bf..0000000
--- a/trunk/peerconnection/third_party/libjingle/libjingle.gyp
+++ /dev/null
@@ -1,698 +0,0 @@
-# Copyright (c) 2012 The Chromium Authors. All rights reserved.
-# Use of this source code is governed by a BSD-style license that can be
-# found in the LICENSE file.
-
-{
- # TODO(wu): Use the libjingle.gyp from chromium and get rid of this file.
- 'variables': {
- 'no_libjingle_logging%': 0,
- 'conditions': [
- ['inside_chromium_build==1', {
- 'overrides': 'overrides',
- },{
- 'overrides': 'source',
- }],
- ],
- },
- 'target_defaults': {
- 'defines': [
- 'FEATURE_ENABLE_SSL',
- 'FEATURE_ENABLE_VOICEMAIL', # TODO(ncarter): Do we really need this?
- '_USE_32BIT_TIME_T',
- 'SAFE_TO_DEFINE_TALK_BASE_LOGGING_MACROS',
- 'EXPAT_RELATIVE_PATH',
- 'JSONCPP_RELATIVE_PATH',
- 'WEBRTC_RELATIVE_PATH',
- 'HAVE_WEBRTC',
- 'HAVE_WEBRTC_VIDEO',
- 'HAVE_WEBRTC_VOICE',
- 'HAVE_SRTP',
- 'SRTP_RELATIVE_PATH',
- ],
- 'configurations': {
- 'Debug': {
- 'defines': [
- # TODO(sergeyu): Fix libjingle to use NDEBUG instead of
- # _DEBUG and remove this define. See below as well.
- '_DEBUG',
- ],
- }
- },
- 'dependencies': [
- '../expat/expat.gyp:expat',
- ],
- 'export_dependent_settings': [
- '../expat/expat.gyp:expat',
- ],
- 'include_dirs': [
- '../../third_party/libyuv/include/',
- ],
- 'direct_dependent_settings': {
- 'defines': [
- 'FEATURE_ENABLE_SSL',
- 'FEATURE_ENABLE_VOICEMAIL',
- 'EXPAT_RELATIVE_PATH',
- 'WEBRTC_RELATIVE_PATH',
- ],
- 'conditions': [
- ['OS=="win"', {
- 'link_settings': {
- 'libraries': [
- '-lsecur32.lib',
- '-lcrypt32.lib',
- '-liphlpapi.lib',
- ],
- },
- }],
- ['OS=="win"', {
- 'include_dirs': [
- '../third_party/platformsdk_win7/files/Include',
- ],
- 'defines': [
- '_CRT_SECURE_NO_WARNINGS', # Suppres warnings about _vsnprinf
- ],
- }],
- ['OS=="linux"', {
- 'defines': [
- 'LINUX',
- ],
- }],
- ['OS=="mac"', {
- 'defines': [
- 'OSX',
- ],
- }],
- ['os_posix == 1', {
- 'defines': [
- 'POSIX',
- ],
- }],
- ['OS=="openbsd" or OS=="freebsd"', {
- 'defines': [
- 'BSD',
- ],
- }],
- ['no_libjingle_logging==1', {
- 'defines': [
- 'NO_LIBJINGLE_LOGGING',
- ],
- }],
- ],
- },
- 'all_dependent_settings': {
- 'configurations': {
- 'Debug': {
- 'defines': [
- # TODO(sergeyu): Fix libjingle to use NDEBUG instead of
- # _DEBUG and remove this define. See above as well.
- '_DEBUG',
- ],
- }
- },
- },
- 'conditions': [
- ['inside_chromium_build==1', {
- 'defines': [
- 'NO_SOUND_SYSTEM',
- ],
- 'include_dirs': [
- '<(overrides)',
- 'source',
- '../..', # the third_party folder for webrtc includes
- '../../third_party/expat/files',
- ],
- 'direct_dependent_settings': {
- 'defines': [
- 'NO_SOUND_SYSTEM',
- ],
- 'include_dirs': [
- '<(overrides)',
- 'source',
- '../../third_party/expat/files'
- ],
- },
- 'dependencies': [
- '../../base/base.gyp:base',
- '../../net/net.gyp:net',
- ],
- },{
- 'include_dirs': [
- # the third_party folder for webrtc/ includes (non-chromium).
- '../../src',
- 'source',
- '../../third_party/expat/files',
- ],
- }],
- ['OS=="win"', {
- 'include_dirs': [
- '../third_party/platformsdk_win7/files/Include',
- ],
- }],
- ['OS=="linux"', {
- 'defines': [
- 'LINUX',
- ],
- }],
- ['OS=="mac"', {
- 'defines': [
- 'OSX',
- ],
- }],
- ['os_posix == 1', {
- 'defines': [
- 'POSIX',
- ],
- }],
- ['OS=="openbsd" or OS=="freebsd"', {
- 'defines': [
- 'BSD',
- ],
- }],
- ],
- },
- 'targets': [
- {
- 'target_name': 'libjingle',
- 'type': 'static_library',
- 'sources': [
- '<(overrides)/talk/base/basictypes.h',
- '<(overrides)/talk/base/constructormagic.h',
-
- # Need to override logging.h because we need
- # SAFE_TO_DEFINE_TALK_BASE_LOGGING_MACROS to work.
- # TODO(sergeyu): push SAFE_TO_DEFINE_TALK_BASE_LOGGING_MACROS to
- # libjingle and remove this override.
- '<(overrides)/talk/base/logging.h',
-
- '<(overrides)/talk/base/scoped_ptr.h',
-
- # Libjingle's QName is not threadsafe, so we need to use our own version
- # here.
- # TODO(sergeyu): Fix QName in Libjingle.
- '<(overrides)/talk/xmllite/qname.cc',
- '<(overrides)/talk/xmllite/qname.h',
-
- 'source/talk/base/Equifax_Secure_Global_eBusiness_CA-1.h',
- 'source/talk/base/asyncfile.cc',
- 'source/talk/base/asyncfile.h',
- 'source/talk/base/asynchttprequest.cc',
- 'source/talk/base/asynchttprequest.h',
- 'source/talk/base/asyncpacketsocket.h',
- 'source/talk/base/asyncsocket.cc',
- 'source/talk/base/asyncsocket.h',
- 'source/talk/base/asynctcpsocket.cc',
- 'source/talk/base/asynctcpsocket.h',
- 'source/talk/base/asyncudpsocket.cc',
- 'source/talk/base/asyncudpsocket.h',
- 'source/talk/base/autodetectproxy.cc',
- 'source/talk/base/autodetectproxy.h',
- 'source/talk/base/base64.cc',
- 'source/talk/base/base64.h',
- 'source/talk/base/basicdefs.h',
- 'source/talk/base/basicpacketsocketfactory.cc',
- 'source/talk/base/basicpacketsocketfactory.h',
- 'source/talk/base/bytebuffer.cc',
- 'source/talk/base/bytebuffer.h',
- 'source/talk/base/byteorder.h',
- 'source/talk/base/checks.cc',
- 'source/talk/base/checks.h',
- 'source/talk/base/common.cc',
- 'source/talk/base/common.h',
- 'source/talk/base/criticalsection.h',
- 'source/talk/base/cryptstring.h',
- 'source/talk/base/diskcache.cc',
- 'source/talk/base/diskcache.h',
- 'source/talk/base/event.cc',
- 'source/talk/base/event.h',
- 'source/talk/base/fileutils.cc',
- 'source/talk/base/fileutils.h',
- 'source/talk/base/firewallsocketserver.cc',
- 'source/talk/base/firewallsocketserver.h',
- 'source/talk/base/flags.cc',
- 'source/talk/base/flags.h',
- 'source/talk/base/helpers.cc',
- 'source/talk/base/helpers.h',
- 'source/talk/base/host.cc',
- 'source/talk/base/host.h',
- 'source/talk/base/httpbase.cc',
- 'source/talk/base/httpbase.h',
- 'source/talk/base/httpclient.h',
- 'source/talk/base/httpclient.cc',
- 'source/talk/base/httpcommon-inl.h',
- 'source/talk/base/httpcommon.cc',
- 'source/talk/base/httpcommon.h',
- 'source/talk/base/httprequest.cc',
- 'source/talk/base/httprequest.h',
- 'source/talk/base/ipaddress.cc',
- 'source/talk/base/ipaddress.h',
- 'source/talk/base/json.cc',
- 'source/talk/base/json.h',
- 'source/talk/base/linked_ptr.h',
- 'source/talk/base/logging.cc',
- 'source/talk/base/md5.h',
- 'source/talk/base/md5c.c',
- 'source/talk/base/messagehandler.cc',
- 'source/talk/base/messagehandler.h',
- 'source/talk/base/messagequeue.cc',
- 'source/talk/base/messagequeue.h',
- 'source/talk/base/nethelpers.cc',
- 'source/talk/base/nethelpers.h',
- 'source/talk/base/network.cc',
- 'source/talk/base/network.h',
- 'source/talk/base/pathutils.cc',
- 'source/talk/base/pathutils.h',
- 'source/talk/base/physicalsocketserver.cc',
- 'source/talk/base/physicalsocketserver.h',
- 'source/talk/base/proxydetect.cc',
- 'source/talk/base/proxydetect.h',
- 'source/talk/base/proxyinfo.cc',
- 'source/talk/base/proxyinfo.h',
- 'source/talk/base/ratetracker.cc',
- 'source/talk/base/ratetracker.h',
- 'source/talk/base/sec_buffer.h',
- 'source/talk/base/signalthread.cc',
- 'source/talk/base/signalthread.h',
- 'source/talk/base/sigslot.h',
- 'source/talk/base/sigslotrepeater.h',
- 'source/talk/base/socket.h',
- 'source/talk/base/socketadapters.cc',
- 'source/talk/base/socketadapters.h',
- 'source/talk/base/socketaddress.cc',
- 'source/talk/base/socketaddress.h',
- 'source/talk/base/socketaddresspair.cc',
- 'source/talk/base/socketaddresspair.h',
- 'source/talk/base/socketfactory.h',
- 'source/talk/base/socketpool.cc',
- 'source/talk/base/socketpool.h',
- 'source/talk/base/socketserver.h',
- 'source/talk/base/socketstream.cc',
- 'source/talk/base/socketstream.h',
- 'source/talk/base/ssladapter.cc',
- 'source/talk/base/ssladapter.h',
- 'source/talk/base/sslsocketfactory.cc',
- 'source/talk/base/sslsocketfactory.h',
- 'source/talk/base/stream.cc',
- 'source/talk/base/stream.h',
- 'source/talk/base/stringdigest.cc',
- 'source/talk/base/stringdigest.h',
- 'source/talk/base/stringencode.cc',
- 'source/talk/base/stringencode.h',
- 'source/talk/base/stringutils.cc',
- 'source/talk/base/stringutils.h',
- 'source/talk/base/task.cc',
- 'source/talk/base/task.h',
- 'source/talk/base/taskparent.cc',
- 'source/talk/base/taskparent.h',
- 'source/talk/base/taskrunner.cc',
- 'source/talk/base/taskrunner.h',
- 'source/talk/base/thread.cc',
- 'source/talk/base/thread.h',
- 'source/talk/base/time.cc',
- 'source/talk/base/time.h',
- 'source/talk/base/urlencode.cc',
- 'source/talk/base/urlencode.h',
- 'source/talk/base/worker.cc',
- 'source/talk/base/worker.h',
- 'source/talk/xmllite/xmlbuilder.cc',
- 'source/talk/xmllite/xmlbuilder.h',
- 'source/talk/xmllite/xmlconstants.cc',
- 'source/talk/xmllite/xmlconstants.h',
- 'source/talk/xmllite/xmlelement.cc',
- 'source/talk/xmllite/xmlelement.h',
- 'source/talk/xmllite/xmlnsstack.cc',
- 'source/talk/xmllite/xmlnsstack.h',
- 'source/talk/xmllite/xmlparser.cc',
- 'source/talk/xmllite/xmlparser.h',
- 'source/talk/xmllite/xmlprinter.cc',
- 'source/talk/xmllite/xmlprinter.h',
- 'source/talk/xmpp/asyncsocket.h',
- 'source/talk/xmpp/constants.cc',
- 'source/talk/xmpp/constants.h',
- 'source/talk/xmpp/jid.cc',
- 'source/talk/xmpp/jid.h',
- 'source/talk/xmpp/plainsaslhandler.h',
- 'source/talk/xmpp/prexmppauth.h',
- 'source/talk/xmpp/ratelimitmanager.cc',
- 'source/talk/xmpp/ratelimitmanager.h',
- 'source/talk/xmpp/saslcookiemechanism.h',
- 'source/talk/xmpp/saslhandler.h',
- 'source/talk/xmpp/saslmechanism.cc',
- 'source/talk/xmpp/saslmechanism.h',
- 'source/talk/xmpp/saslplainmechanism.h',
- 'source/talk/xmpp/xmppclient.cc',
- 'source/talk/xmpp/xmppclient.h',
- 'source/talk/xmpp/xmppclientsettings.h',
- 'source/talk/xmpp/xmppengine.h',
- 'source/talk/xmpp/xmppengineimpl.cc',
- 'source/talk/xmpp/xmppengineimpl.h',
- 'source/talk/xmpp/xmppengineimpl_iq.cc',
- 'source/talk/xmpp/xmpplogintask.cc',
- 'source/talk/xmpp/xmpplogintask.h',
- 'source/talk/xmpp/xmppstanzaparser.cc',
- 'source/talk/xmpp/xmppstanzaparser.h',
- 'source/talk/xmpp/xmpptask.cc',
- 'source/talk/xmpp/xmpptask.h',
- ],
- 'conditions': [
- ['inside_chromium_build==0', {
- 'sources': [
- 'source/talk/sound/automaticallychosensoundsystem.h',
- 'source/talk/sound/platformsoundsystem.cc',
- 'source/talk/sound/platformsoundsystem.h',
- 'source/talk/sound/platformsoundsystemfactory.cc',
- 'source/talk/sound/platformsoundsystemfactory.h',
- 'source/talk/sound/sounddevicelocator.h',
- 'source/talk/sound/soundinputstreaminterface.h',
- 'source/talk/sound/soundoutputstreaminterface.h',
- 'source/talk/sound/soundsystemfactory.h',
- 'source/talk/sound/soundsysteminterface.cc',
- 'source/talk/sound/soundsysteminterface.h',
- 'source/talk/sound/soundsystemproxy.cc',
- 'source/talk/sound/soundsystemproxy.h',
- ],
- 'conditions' : [
- ['OS=="linux"', {
- 'sources': [
- 'source/talk/sound/alsasoundsystem.cc',
- 'source/talk/sound/alsasoundsystem.h',
- 'source/talk/sound/alsasymboltable.cc',
- 'source/talk/sound/alsasymboltable.h',
- 'source/talk/sound/linuxsoundsystem.cc',
- 'source/talk/sound/linuxsoundsystem.h',
- 'source/talk/sound/pulseaudiosoundsystem.cc',
- 'source/talk/sound/pulseaudiosoundsystem.h',
- 'source/talk/sound/pulseaudiosymboltable.cc',
- 'source/talk/sound/pulseaudiosymboltable.h',
- ],
- }],
- ],
- }],
- ['OS=="win"', {
- 'sources': [
- '<(overrides)/talk/base/win32socketinit.cc',
- 'source/talk/base/schanneladapter.cc',
- 'source/talk/base/schanneladapter.h',
- 'source/talk/base/win32.h',
- 'source/talk/base/win32.cc',
- 'source/talk/base/win32filesystem.cc',
- 'source/talk/base/win32filesystem.h',
- 'source/talk/base/win32window.h',
- 'source/talk/base/win32window.cc',
- 'source/talk/base/win32securityerrors.cc',
- 'source/talk/base/winfirewall.cc',
- 'source/talk/base/winfirewall.h',
- 'source/talk/base/winping.cc',
- 'source/talk/base/winping.h',
- ],
- }],
- ['os_posix == 1', {
- 'sources': [
- 'source/talk/base/latebindingsymboltable.cc',
- 'source/talk/base/latebindingsymboltable.h',
- 'source/talk/base/sslstreamadapter.cc',
- 'source/talk/base/sslstreamadapter.h',
- 'source/talk/base/unixfilesystem.cc',
- 'source/talk/base/unixfilesystem.h',
- ],
- }],
- ['OS=="linux"', {
- 'sources': [
- 'source/talk/base/linux.cc',
- 'source/talk/base/linux.h',
- ],
- }],
- ['OS=="mac"', {
- 'sources': [
- 'source/talk/base/macconversion.cc',
- 'source/talk/base/macconversion.h',
- 'source/talk/base/macutils.cc',
- 'source/talk/base/macutils.h',
- ],
- }],
- ],
- 'dependencies': [
- '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
- ],
- },
- # This has to be is a separate project due to a bug in MSVS:
- # https://connect.microsoft.com/VisualStudio/feedback/details/368272/duplicate-cpp-filename-in-c-project-visual-studio-2008
- # We have two files named "constants.cc" and MSVS doesn't handle this
- # properly.
- {
- 'target_name': 'libjingle_p2p',
- 'type': 'static_library',
- 'sources': [
- 'source/talk/p2p/base/candidate.h',
- 'source/talk/p2p/base/common.h',
- 'source/talk/p2p/base/constants.cc',
- 'source/talk/p2p/base/constants.h',
- 'source/talk/p2p/base/p2ptransport.cc',
- 'source/talk/p2p/base/p2ptransport.h',
- 'source/talk/p2p/base/p2ptransportchannel.cc',
- 'source/talk/p2p/base/p2ptransportchannel.h',
- 'source/talk/p2p/base/port.cc',
- 'source/talk/p2p/base/port.h',
- 'source/talk/p2p/base/portallocator.cc',
- 'source/talk/p2p/base/portallocator.h',
- 'source/talk/p2p/base/portallocatorsessionproxy.cc',
- 'source/talk/p2p/base/portallocatorsessionproxy.h',
- 'source/talk/p2p/base/portproxy.cc',
- 'source/talk/p2p/base/portproxy.h',
- 'source/talk/p2p/base/pseudotcp.cc',
- 'source/talk/p2p/base/pseudotcp.h',
- 'source/talk/p2p/base/rawtransport.cc',
- 'source/talk/p2p/base/rawtransport.h',
- 'source/talk/p2p/base/rawtransportchannel.cc',
- 'source/talk/p2p/base/rawtransportchannel.h',
- 'source/talk/p2p/base/relayport.cc',
- 'source/talk/p2p/base/relayport.h',
- 'source/talk/p2p/base/session.cc',
- 'source/talk/p2p/base/session.h',
- 'source/talk/p2p/base/sessionclient.h',
- 'source/talk/p2p/base/sessiondescription.cc',
- 'source/talk/p2p/base/sessiondescription.h',
- 'source/talk/p2p/base/sessionid.h',
- 'source/talk/p2p/base/sessionmanager.cc',
- 'source/talk/p2p/base/sessionmanager.h',
- 'source/talk/p2p/base/sessionmessages.cc',
- 'source/talk/p2p/base/sessionmessages.h',
- 'source/talk/p2p/base/parsing.cc',
- 'source/talk/p2p/base/parsing.h',
- 'source/talk/p2p/base/stun.cc',
- 'source/talk/p2p/base/stun.h',
- 'source/talk/p2p/base/stunport.cc',
- 'source/talk/p2p/base/stunport.h',
- 'source/talk/p2p/base/stunrequest.cc',
- 'source/talk/p2p/base/stunrequest.h',
- 'source/talk/p2p/base/tcpport.cc',
- 'source/talk/p2p/base/tcpport.h',
- 'source/talk/p2p/base/transport.cc',
- 'source/talk/p2p/base/transport.h',
- 'source/talk/p2p/base/transportchannel.cc',
- 'source/talk/p2p/base/transportchannel.h',
- 'source/talk/p2p/base/transportchannelimpl.h',
- 'source/talk/p2p/base/transportchannelproxy.cc',
- 'source/talk/p2p/base/transportchannelproxy.h',
- 'source/talk/p2p/base/udpport.cc',
- 'source/talk/p2p/base/udpport.h',
- 'source/talk/p2p/client/basicportallocator.cc',
- 'source/talk/p2p/client/basicportallocator.h',
- 'source/talk/p2p/client/httpportallocator.cc',
- 'source/talk/p2p/client/httpportallocator.h',
- 'source/talk/p2p/client/sessionmanagertask.h',
- 'source/talk/p2p/client/sessionsendtask.h',
- 'source/talk/p2p/client/socketmonitor.cc',
- 'source/talk/p2p/client/socketmonitor.h',
- 'source/talk/session/phone/audiomonitor.cc',
- 'source/talk/session/phone/audiomonitor.h',
- 'source/talk/session/phone/call.cc',
- 'source/talk/session/phone/call.h',
- 'source/talk/session/phone/channel.cc',
- 'source/talk/session/phone/channel.h',
- 'source/talk/session/phone/channelmanager.cc',
- 'source/talk/session/phone/channelmanager.h',
- 'source/talk/session/phone/codec.cc',
- 'source/talk/session/phone/codec.h',
- 'source/talk/session/phone/cryptoparams.h',
- 'source/talk/session/phone/currentspeakermonitor.cc',
- 'source/talk/session/phone/currentspeakermonitor.h',
- 'source/talk/session/phone/devicemanager.cc',
- 'source/talk/session/phone/devicemanager.h',
- 'source/talk/session/phone/filemediaengine.cc',
- 'source/talk/session/phone/filemediaengine.h',
- 'source/talk/session/phone/mediachannel.h',
- 'source/talk/session/phone/mediaengine.cc',
- 'source/talk/session/phone/mediaengine.h',
- 'source/talk/session/phone/mediamessages.cc',
- 'source/talk/session/phone/mediamessages.h',
- 'source/talk/session/phone/mediamonitor.cc',
- 'source/talk/session/phone/mediamonitor.h',
- 'source/talk/session/phone/mediasession.cc',
- 'source/talk/session/phone/mediasession.h',
- 'source/talk/session/phone/mediasessionclient.cc',
- 'source/talk/session/phone/mediasessionclient.h',
- 'source/talk/session/phone/mediasink.h',
- 'source/talk/session/phone/rtcpmuxfilter.cc',
- 'source/talk/session/phone/rtcpmuxfilter.h',
- 'source/talk/session/phone/rtpdump.cc',
- 'source/talk/session/phone/rtpdump.h',
- 'source/talk/session/phone/rtputils.cc',
- 'source/talk/session/phone/rtputils.h',
- 'source/talk/session/phone/soundclip.cc',
- 'source/talk/session/phone/soundclip.h',
- 'source/talk/session/phone/srtpfilter.cc',
- 'source/talk/session/phone/srtpfilter.h',
- 'source/talk/session/phone/ssrcmuxfilter.cc',
- 'source/talk/session/phone/ssrcmuxfilter.h',
- 'source/talk/session/phone/streamparams.cc',
- 'source/talk/session/phone/videocapturer.cc',
- 'source/talk/session/phone/videocapturer.h',
- 'source/talk/session/phone/videocommon.cc',
- 'source/talk/session/phone/videocommon.h',
- 'source/talk/session/phone/videoframe.cc',
- 'source/talk/session/phone/videoframe.h',
- 'source/talk/session/phone/voicechannel.h',
- 'source/talk/session/phone/webrtcpassthroughrender.cc',
- 'source/talk/session/phone/webrtccommon.h',
- 'source/talk/session/phone/webrtcvideocapturer.cc',
- 'source/talk/session/phone/webrtcvideocapturer.h',
- 'source/talk/session/phone/webrtcvideoengine.cc',
- 'source/talk/session/phone/webrtcvideoengine.h',
- 'source/talk/session/phone/webrtcvideoframe.cc',
- 'source/talk/session/phone/webrtcvideoframe.h',
- 'source/talk/session/phone/webrtcvie.h',
- 'source/talk/session/phone/webrtcvoe.h',
- 'source/talk/session/phone/webrtcvoiceengine.cc',
- 'source/talk/session/phone/webrtcvoiceengine.h',
- 'source/talk/session/tunnel/pseudotcpchannel.cc',
- 'source/talk/session/tunnel/pseudotcpchannel.h',
- 'source/talk/session/tunnel/tunnelsessionclient.cc',
- 'source/talk/session/tunnel/tunnelsessionclient.h',
- ],
- 'conditions': [
- ['OS=="win"', {
- 'sources': [
- 'source/talk/session/phone/gdivideorenderer.cc',
- 'source/talk/session/phone/gdivideorenderer.h',
- 'source/talk/session/phone/win32devicemanager.cc',
- 'source/talk/session/phone/win32devicemanager.h',
- ],
- }],
- ['OS=="linux"', {
- 'sources': [
- 'source/talk/session/phone/gtkvideorenderer.cc',
- 'source/talk/session/phone/gtkvideorenderer.cc',
- 'source/talk/session/phone/gtkvideorenderer.h',
- 'source/talk/session/phone/libudevsymboltable.cc',
- 'source/talk/session/phone/libudevsymboltable.h',
- 'source/talk/session/phone/linuxdevicemanager.cc',
- 'source/talk/session/phone/linuxdevicemanager.h',
- 'source/talk/session/phone/v4llookup.cc',
- 'source/talk/session/phone/v4llookup.h',
- ],
- 'include_dirs': [
- 'source/talk/third_party/libudev',
- ],
- 'cflags': [
- '<!@(pkg-config --cflags gtk+-2.0)',
- ],
- }],
- ['inside_chromium_build==1', {
- 'dependencies': [
- '../../third_party/webrtc/modules/modules.gyp:audio_device',
- '../../third_party/webrtc/modules/modules.gyp:video_capture_module',
- '../../third_party/webrtc/modules/modules.gyp:video_render_module',
- '../../third_party/webrtc/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '../../third_party/webrtc/video_engine/video_engine.gyp:video_engine_core',
- '../../third_party/webrtc/voice_engine/voice_engine.gyp:voice_engine_core',
- '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
- 'libjingle',
- ],
- }, {
- 'dependencies': [
- '../../src/modules/modules.gyp:audio_device',
- '../../src/modules/modules.gyp:video_capture_module',
- '../../src/modules/modules.gyp:video_render_module',
- '../../src/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '../../src/video_engine/video_engine.gyp:video_engine_core',
- '../../src/voice_engine/voice_engine.gyp:voice_engine_core',
- '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
- 'libjingle',
- ],
- } ], # inside_chromium_build
- ], # conditions
- },
- # seperate project for app
- {
- 'target_name': 'libjingle_app',
- 'type': '<(library)',
- 'sources': [
- 'source/talk/app/webrtc/audiotrackimpl.cc',
- 'source/talk/app/webrtc/audiotrackimpl.h',
- 'source/talk/app/webrtc/mediastream.h',
- 'source/talk/app/webrtc/mediastreamhandler.cc',
- 'source/talk/app/webrtc/mediastreamhandler.h',
- 'source/talk/app/webrtc/mediastreamimpl.cc',
- 'source/talk/app/webrtc/mediastreamimpl.h',
- 'source/talk/app/webrtc/mediastreamprovider.h',
- 'source/talk/app/webrtc/mediastreamproxy.cc',
- 'source/talk/app/webrtc/mediastreamproxy.h',
- 'source/talk/app/webrtc/mediastreamtrackproxy.cc',
- 'source/talk/app/webrtc/mediastreamtrackproxy.h',
- 'source/talk/app/webrtc/mediatrackimpl.h',
- 'source/talk/app/webrtc/notifierimpl.h',
- 'source/talk/app/webrtc/peerconnection.h',
- 'source/talk/app/webrtc/peerconnectionfactoryimpl.cc',
- 'source/talk/app/webrtc/peerconnectionfactoryimpl.h',
- 'source/talk/app/webrtc/peerconnectionimpl.cc',
- 'source/talk/app/webrtc/peerconnectionimpl.h',
- 'source/talk/app/webrtc/peerconnectionsignaling.cc',
- 'source/talk/app/webrtc/peerconnectionsignaling.h',
- 'source/talk/app/webrtc/portallocatorfactory.cc',
- 'source/talk/app/webrtc/portallocatorfactory.h',
- 'source/talk/app/webrtc/roaperrorcodes.h',
- 'source/talk/app/webrtc/roapmessages.cc',
- 'source/talk/app/webrtc/roapmessages.h',
- 'source/talk/app/webrtc/roapsession.cc',
- 'source/talk/app/webrtc/roapsession.h',
- 'source/talk/app/webrtc/sessiondescriptionprovider.h',
- 'source/talk/app/webrtc/streamcollectionimpl.h',
- 'source/talk/app/webrtc/videorendererimpl.cc',
- 'source/talk/app/webrtc/videotrackimpl.cc',
- 'source/talk/app/webrtc/videotrackimpl.h',
- 'source/talk/app/webrtc/webrtcjson.cc',
- 'source/talk/app/webrtc/webrtcjson.h',
- 'source/talk/app/webrtc/webrtcsdp.cc',
- 'source/talk/app/webrtc/webrtcsdp.h',
- 'source/talk/app/webrtc/webrtcsession.cc',
- 'source/talk/app/webrtc/webrtcsession.h',
- 'source/talk/app/webrtc/webrtcsessionobserver.h',
- ],
- 'dependencies': [
- '<(DEPTH)/third_party/jsoncpp/jsoncpp.gyp:jsoncpp',
- '<(DEPTH)/third_party/libsrtp/libsrtp.gyp:libsrtp',
- ],
- 'conditions': [
- ['inside_chromium_build==1', {
- 'dependencies': [
- '../../third_party/webrtc/modules/modules.gyp:video_capture_module',
- '../../third_party/webrtc/modules/modules.gyp:video_render_module',
- '../../third_party/webrtc/video_engine/video_engine.gyp:video_engine_core',
- '../../third_party/webrtc/voice_engine/voice_engine.gyp:voice_engine_core',
- '../../third_party/webrtc/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- 'libjingle_p2p',
- ],
- }, {
- 'dependencies': [
- '../../src/modules/modules.gyp:video_capture_module',
- '../../src/modules/modules.gyp:video_render_module',
- '../../src/video_engine/video_engine.gyp:video_engine_core',
- '../../src/voice_engine/voice_engine.gyp:voice_engine_core',
- '../../src/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- 'libjingle_p2p',
- ],
- } ], # inside_chromium_build
- ], # conditions
- },
- ],
-}
diff --git a/trunk/src/LICENSE b/trunk/src/LICENSE
deleted file mode 100644
index 4c41b7b..0000000
--- a/trunk/src/LICENSE
+++ /dev/null
@@ -1,29 +0,0 @@
-Copyright (c) 2011, The WebRTC project authors. All rights reserved.
-
-Redistribution and use in source and binary forms, with or without
-modification, are permitted provided that the following conditions are
-met:
-
- * Redistributions of source code must retain the above copyright
- notice, this list of conditions and the following disclaimer.
-
- * Redistributions in binary form must reproduce the above copyright
- notice, this list of conditions and the following disclaimer in
- the documentation and/or other materials provided with the
- distribution.
-
- * Neither the name of Google nor the names of its contributors may
- be used to endorse or promote products derived from this software
- without specific prior written permission.
-
-THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
-HOLDER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
-LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
-DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
-THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
-(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
-OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
diff --git a/trunk/src/LICENSE_THIRD_PARTY b/trunk/src/LICENSE_THIRD_PARTY
deleted file mode 100644
index e19708a..0000000
--- a/trunk/src/LICENSE_THIRD_PARTY
+++ /dev/null
@@ -1,27 +0,0 @@
-This source tree contains third party source code which is governed by third
-party licenses. This file contains references to files which are under other
-licenses than the one provided in the LICENSE file in the root of the source
-tree.
-
-Files governed by third party licenses:
-common_audio/signal_processing/spl_sqrt_floor.c
-common_audio/signal_processing/spl_sqrt_floor.s
-modules/audio_coding/codecs/G711/main/source/g711.h
-modules/audio_coding/codecs/G711/main/source/g711.c
-modules/audio_coding/codecs/G722/main/source/g722_decode.h
-modules/audio_coding/codecs/G722/main/source/g722_decode.c
-modules/audio_coding/codecs/G722/main/source/g722_encode.h
-modules/audio_coding/codecs/G722/main/source/g722_encode.c
-modules/audio_coding/codecs/iSAC/main/source/fft.c
-modules/audio_device/main/source/Mac/portaudio/pa_memorybarrier.h
-modules/audio_device/main/source/Mac/portaudio/pa_ringbuffer.h
-modules/audio_device/main/source/Mac/portaudio/pa_ringbuffer.c
-modules/audio_processing/utility/fft4g.c
-modules/audio_processing/aec/aec_rdft.c
-system_wrappers/interface/fix_interlocked_exchange_pointer_windows.h
-system_wrappers/interface/scoped_ptr.h
-system_wrappers/interface/scoped_refptr.h
-system_wrappers/source/condition_variable_windows.cc
-system_wrappers/source/spreadsortlib/constants.hpp
-system_wrappers/source/spreadsortlib/spreadsort.hpp
-system_wrappers/source/thread_windows_set_name.h
diff --git a/trunk/src/PATENTS b/trunk/src/PATENTS
deleted file mode 100644
index 190607a..0000000
--- a/trunk/src/PATENTS
+++ /dev/null
@@ -1,24 +0,0 @@
-Additional IP Rights Grant (Patents)
-
-"This implementation" means the copyrightable works distributed by
-Google as part of the WebRTC code package.
-
-Google hereby grants to you a perpetual, worldwide, non-exclusive,
-no-charge, irrevocable (except as stated in this section) patent
-license to make, have made, use, offer to sell, sell, import,
-transfer, and otherwise run, modify and propagate the contents of this
-implementation of the WebRTC code package, where such license applies
-only to those patent claims, both currently owned by Google and
-acquired in the future, licensable by Google that are necessarily
-infringed by this implementation of the WebRTC code package. This
-grant does not include claims that would be infringed only as a
-consequence of further modification of this implementation. If you or
-your agent or exclusive licensee institute or order or agree to the
-institution of patent litigation against any entity (including a
-cross-claim or counterclaim in a lawsuit) alleging that this
-implementation of the WebRTC code package or any code incorporated
-within this implementation of the WebRTC code package constitutes
-direct or contributory patent infringement, or inducement of patent
-infringement, then any patent rights granted to you under this License
-for this implementation of the WebRTC code package shall terminate as
-of the date such litigation is filed.
diff --git a/trunk/src/README.chromium b/trunk/src/README.chromium
deleted file mode 100644
index 246c13d..0000000
--- a/trunk/src/README.chromium
+++ /dev/null
@@ -1,13 +0,0 @@
-Name: WebRTC
-URL: http://www.webrtc.org
-Version: 90
-License: BSD
-License File: LICENSE
-
-Description:
-WebRTC provides real time voice and video processing
-functionality to enable the implementation of
-PeerConnection/MediaStream.
-
-Third party code used in this project is described
-in the file LICENSE_THIRD_PARTY.
diff --git a/trunk/src/build/common.gypi b/trunk/src/build/common.gypi
deleted file mode 100644
index 74b6003..0000000
--- a/trunk/src/build/common.gypi
+++ /dev/null
@@ -1,180 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-# This file contains common settings for building WebRTC components.
-
-{
- 'variables': {
- # These variables need to be nested in order to use them in a conditions
- # block to set other variables.
- 'variables': {
- # This will be set to zero in the supplement.gypi triggered by a gclient
- # hook in the standalone build.
- 'build_with_chromium%': 1,
- },
-
- 'build_with_chromium%': '<(build_with_chromium)',
-
- # The Chromium common.gypi we use treats all gyp files without
- # chromium_code==1 as third party code. This disables many of the
- # preferred warning settings.
- #
- # We can set this here to have WebRTC code treated as Chromium code. Our
- # third party code will still have the reduced warning settings.
- 'chromium_code': 1,
-
- # Adds video support to dependencies shared by voice and video engine.
- # This should normally be enabled; the intended use is to disable only
- # when building voice engine exclusively.
- 'enable_video%': 1,
-
- # Selects fixed-point code where possible.
- # TODO(andrew): we'd like to set this based on the target OS/architecture.
- 'prefer_fixed_point%': 0,
-
- # Enable data logging. Produces text files with data logged within engines
- # which can be easily parsed for offline processing.
- 'enable_data_logging%': 0,
-
- # Disable these to not build components which can be externally provided.
- 'build_libjpeg%': 1,
- 'build_libyuv%': 1,
-
- 'conditions': [
- ['OS=="win"', {
- # TODO(andrew, perkj): does this need to be here?
- # Path needed to build Direct Show base classes on Windows.
- # The code is included in the Windows SDK.
- 'direct_show_base_classes':
- 'C:/Program Files/Microsoft SDKs/Windows/v7.1/Samples/multimedia/directshow/baseclasses/',
- }],
- ['build_with_chromium==1', {
- # Exclude pulse audio on Chromium since its prerequisites don't require
- # pulse audio.
- 'include_pulse_audio%': 0,
-
- # Exclude internal ADM since Chromium uses its own IO handling.
- 'include_internal_audio_device%': 0,
-
- # Exclude internal VCM in Chromium build.
- 'include_internal_video_capture%': 0,
-
- # Exclude internal video render module in Chromium build.
- 'include_internal_video_render%': 0,
-
- # Disable the use of protocol buffers in production code.
- 'enable_protobuf%': 0,
-
- 'webrtc_root%': '<(DEPTH)/third_party/webrtc',
- }, {
- # Settings for the standalone (not-in-Chromium) build.
-
- 'include_pulse_audio%': 1,
-
- 'include_internal_audio_device%': 1,
-
- 'include_internal_video_capture%': 1,
-
- 'include_internal_video_render%': 1,
-
- 'enable_protobuf%': 1,
-
- 'webrtc_root%': '<(DEPTH)/src',
-
- 'conditions': [
- ['OS=="mac"', {
- # TODO(andrew): clang is the default on Mac. For now, disable the
- # Chrome plugins, which causes a flood of chromium-style warnings.
- # Investigate enabling the plugins:
- # http://code.google.com/p/webrtc/issues/detail?id=163
- 'clang_use_chrome_plugins%': 0,
- }],
- ],
- }],
- ], # conditions
- },
- 'target_defaults': {
- 'include_dirs': [
- '..','../..', # common_types.h, typedefs.h
- ],
- 'conditions': [
- ['build_with_chromium==1', {
- 'defines': [
- # Changes settings for Chromium build.
- 'WEBRTC_CHROMIUM_BUILD',
- ],
- }, {
- 'conditions': [
- ['os_posix==1', {
- 'cflags': [
- '-Wextra',
- # We need to repeat some flags from Chromium's common.gypi here
- # that get overridden by -Wextra.
- '-Wno-unused-parameter',
- '-Wno-missing-field-initializers',
- ],
- 'cflags_cc': [
- # This is enabled for clang; enable for gcc as well.
- '-Woverloaded-virtual',
- ],
- }],
- ],
- }],
- ['OS=="linux"', {
- 'defines': [
- 'WEBRTC_TARGET_PC',
- 'WEBRTC_LINUX',
- 'WEBRTC_THREAD_RR',
- # TODO(andrew): can we select this automatically?
- # Define this if the Linux system does not support CLOCK_MONOTONIC.
- #'WEBRTC_CLOCK_TYPE_REALTIME',
- ],
- }],
- ['OS=="mac"', {
- # TODO(andrew): what about PowerPC?
- # Setup for Intel
- 'defines': [
- 'WEBRTC_TARGET_MAC_INTEL',
- 'WEBRTC_MAC_INTEL',
- 'WEBRTC_MAC',
- 'WEBRTC_THREAD_RR',
- 'WEBRTC_CLOCK_TYPE_REALTIME',
- ],
- }],
- ['OS=="win"', {
- 'defines': [
- 'WEBRTC_TARGET_PC',
- ],
- # TODO(andrew): remove this block when possible.
- # 4389: Signed/unsigned mismatch.
- # 4373: MSVC legacy warning for ignoring const / volatile in
- # signatures. TODO(phoglund): get rid of 4373 supression when
- # http://code.google.com/p/webrtc/issues/detail?id=261 is solved.
- 'msvs_disabled_warnings': [4389, 4373],
- }],
- ['OS=="android"', {
- 'defines': [
- 'WEBRTC_TARGET_PC',
- 'WEBRTC_LINUX',
- # TODO(leozwang): Investigate CLOCK_REALTIME and CLOCK_MONOTONIC
- # support on Android. Keep WEBRTC_CLOCK_TYPE_REALTIME for now,
- # remove it after I verify that CLOCK_MONOTONIC is fully functional
- # with condition and event functions in system_wrappers.
- 'WEBRTC_CLOCK_TYPE_REALTIME',
- 'WEBRTC_THREAD_RR',
- 'WEBRTC_ANDROID',
- 'WEBRTC_ARM_INLINE_CALLS',
- # TODO(leozwang): move WEBRTC_ARCH_ARM to typedefs.h.
- 'WEBRTC_ARCH_ARM',
- 'WEBRTC_ANDROID_OPENSLES',
- ],
- }],
- ], # conditions
- }, # target_defaults
-}
-
diff --git a/trunk/src/build/merge_libs.gyp b/trunk/src/build/merge_libs.gyp
deleted file mode 100644
index d23bf11..0000000
--- a/trunk/src/build/merge_libs.gyp
+++ /dev/null
@@ -1,43 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'includes': [ 'common.gypi', ],
- 'targets': [
- {
- 'target_name': 'no_op',
- 'type': 'executable',
- 'dependencies': [
- '../video_engine/video_engine.gyp:video_engine_core',
- ],
- 'sources': [ 'no_op.cc', ],
- },
- {
- 'target_name': 'merged_lib',
- 'type': 'none',
- 'dependencies': [
- 'no_op',
- ],
- 'actions': [
- {
- 'variables': {
- 'output_lib_name': 'webrtc',
- 'output_lib': '<(PRODUCT_DIR)/<(STATIC_LIB_PREFIX)<(output_lib_name)_<(OS)_<(target_arch)_<(CONFIGURATION_NAME)<(STATIC_LIB_SUFFIX)',
- },
- 'action_name': 'merge_libs',
- 'inputs': ['<(PRODUCT_DIR)/<(EXECUTABLE_PREFIX)no_op<(EXECUTABLE_SUFFIX)'],
- 'outputs': ['<(output_lib)'],
- 'action': ['python',
- 'merge_libs.py',
- '<(PRODUCT_DIR)',
- '<(output_lib)',],
- },
- ],
- },
- ],
-}
diff --git a/trunk/src/build/merge_libs.py b/trunk/src/build/merge_libs.py
deleted file mode 100644
index 31c5efb..0000000
--- a/trunk/src/build/merge_libs.py
+++ /dev/null
@@ -1,58 +0,0 @@
-#!/usr/bin/env python
-
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-# Searches for libraries and/or object files on the specified path and
-# merges them into a single library.
-
-import subprocess
-import sys
-
-if __name__ == '__main__':
- if len(sys.argv) != 3:
- sys.stderr.write('Usage: ' + sys.argv[0] + ' <search_path> <output_lib>\n')
- sys.exit(2)
-
- search_path = sys.argv[1]
- output_lib = sys.argv[2]
-
- from subprocess import call, PIPE
- if sys.platform.startswith('linux'):
- call(["rm -f " + output_lib], shell=True)
- call(["rm -rf " + search_path + "/obj.target/*do_not_use*"], shell=True)
- call(["ar crs " + output_lib + " $(find " + search_path +
- "/obj.target -name *\.o)"], shell=True)
- call(["ar crs " + output_lib + " $(find " + search_path +
- "/obj/gen -name *\.o)"], shell=True)
-
- elif sys.platform == 'darwin':
- call(["rm -f " + output_lib], shell=True)
- call(["rm -f " + search_path + "/*do_not_use*"], shell=True)
- call(["libtool -static -v -o " + output_lib + " " + search_path + "/*.a"],
- shell=True)
-
- elif sys.platform == 'win32':
- # We need to execute a batch file to set some environment variables for the
- # lib command. VS 8 uses vsvars.bat and VS 9 uses vsvars32.bat. It's
- # required that at least one of them is in the system PATH. We try both and
- # suppress stderr and stdout to fail silently.
- call(["vsvars.bat"], stderr=PIPE, stdout=PIPE, shell=True)
- call(["vsvars32.bat"], stderr=PIPE, stdout=PIPE, shell=True)
- call(["del " + output_lib], shell=True)
- call(["del /F /S /Q " + search_path + "/lib/*do_not_use*"],
- shell=True)
- call(["lib /OUT:" + output_lib + " " + search_path + "/lib/*.lib"],
- shell=True)
-
- else:
- sys.stderr.write('Platform not supported: %r\n\n' % sys.platform)
- sys.exit(1)
-
- sys.exit(0)
-
diff --git a/trunk/src/build/no_op.cc b/trunk/src/build/no_op.cc
deleted file mode 100644
index 4cce7f6..0000000
--- a/trunk/src/build/no_op.cc
+++ /dev/null
@@ -1,3 +0,0 @@
-int main() {
- return 0;
-}
diff --git a/trunk/src/build/protoc.gypi b/trunk/src/build/protoc.gypi
deleted file mode 100644
index 70bf71e..0000000
--- a/trunk/src/build/protoc.gypi
+++ /dev/null
@@ -1,97 +0,0 @@
-# Copyright (c) 2011 The Chromium Authors. All rights reserved.
-# Use of this source code is governed by a BSD-style license that can be
-# found in the LICENSE file.
-
-# It was necessary to copy this file to WebRTC, because the path to
-# build/common.gypi is different for the standalone and Chromium builds. Gyp
-# doesn't permit conditional inclusion or variable expansion in include paths.
-# http://code.google.com/p/gyp/wiki/InputFormatReference#Including_Other_Files
-
-# This file is meant to be included into a target to provide a rule
-# to invoke protoc in a consistent manner.
-#
-# To use this, create a gyp target with the following form:
-# {
-# 'target_name': 'my_proto_lib',
-# 'type': 'static_library',
-# 'sources': [
-# 'foo.proto',
-# 'bar.proto',
-# ],
-# 'variables': {
-# # Optional, see below: 'proto_in_dir': '.'
-# 'proto_out_dir': 'dir/for/my_proto_lib'
-# },
-# 'includes': ['path/to/this/gypi/file'],
-# }
-# If necessary, you may add normal .cc files to the sources list or other gyp
-# dependencies. The proto headers are guaranteed to be generated before any
-# source files, even within this target, are compiled.
-#
-# The 'proto_in_dir' variable must be the relative path to the
-# directory containing the .proto files. If left out, it defaults to '.'.
-#
-# The 'proto_out_dir' variable specifies the path suffix that output
-# files are generated under. Targets that gyp-depend on my_proto_lib
-# will be able to include the resulting proto headers with an include
-# like:
-# #include "dir/for/my_proto_lib/foo.pb.h"
-#
-# Implementation notes:
-# A proto_out_dir of foo/bar produces
-# <(SHARED_INTERMEDIATE_DIR)/protoc_out/foo/bar/{file1,file2}.pb.{cc,h}
-# <(SHARED_INTERMEDIATE_DIR)/pyproto/foo/bar/{file1,file2}_pb2.py
-
-{
- 'variables': {
- 'protoc': '<(PRODUCT_DIR)/<(EXECUTABLE_PREFIX)protoc<(EXECUTABLE_SUFFIX)',
- 'cc_dir': '<(SHARED_INTERMEDIATE_DIR)/protoc_out/<(proto_out_dir)',
- 'py_dir': '<(PRODUCT_DIR)/pyproto/<(proto_out_dir)',
- 'proto_in_dir%': '.',
- },
- 'rules': [
- {
- 'rule_name': 'genproto',
- 'extension': 'proto',
- 'inputs': [
- '<(protoc)',
- ],
- 'outputs': [
- '<(py_dir)/<(RULE_INPUT_ROOT)_pb2.py',
- '<(cc_dir)/<(RULE_INPUT_ROOT).pb.cc',
- '<(cc_dir)/<(RULE_INPUT_ROOT).pb.h',
- ],
- 'action': [
- '<(protoc)',
- '--proto_path=<(proto_in_dir)',
- # Naively you'd use <(RULE_INPUT_PATH) here, but protoc requires
- # --proto_path is a strict prefix of the path given as an argument.
- '<(proto_in_dir)/<(RULE_INPUT_ROOT)<(RULE_INPUT_EXT)',
- '--cpp_out=<(cc_dir)',
- '--python_out=<(py_dir)',
- ],
- 'message': 'Generating C++ and Python code from <(RULE_INPUT_PATH)',
- 'process_outputs_as_sources': 1,
- },
- ],
- 'dependencies': [
- '<(DEPTH)/third_party/protobuf/protobuf.gyp:protoc#host',
- '<(DEPTH)/third_party/protobuf/protobuf.gyp:protobuf_lite',
- ],
- 'include_dirs': [
- '<(SHARED_INTERMEDIATE_DIR)/protoc_out',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '<(SHARED_INTERMEDIATE_DIR)/protoc_out',
- ]
- },
- 'export_dependent_settings': [
- # The generated headers reference headers within protobuf_lite,
- # so dependencies must be able to find those headers too.
- '<(DEPTH)/third_party/protobuf/protobuf.gyp:protobuf_lite',
- ],
- # This target exports a hard dependency because it generates header
- # files.
- 'hard_dependency': 1,
-}
diff --git a/trunk/src/common_audio/OWNERS b/trunk/src/common_audio/OWNERS
deleted file mode 100644
index 84582f2..0000000
--- a/trunk/src/common_audio/OWNERS
+++ /dev/null
@@ -1,4 +0,0 @@
-bjornv@webrtc.org
-tina.legrand@webrtc.org
-jan.skoglund@webrtc.org
-andrew@webrtc.org
diff --git a/trunk/src/common_audio/common_audio.gyp b/trunk/src/common_audio/common_audio.gyp
deleted file mode 100644
index 3d3da3f..0000000
--- a/trunk/src/common_audio/common_audio.gyp
+++ /dev/null
@@ -1,16 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'includes': [
- '../build/common.gypi',
- 'signal_processing/signal_processing.gypi',
- 'resampler/resampler.gypi',
- 'vad/vad.gypi',
- ],
-}
diff --git a/trunk/src/common_audio/resampler/Android.mk b/trunk/src/common_audio/resampler/Android.mk
deleted file mode 100644
index b1d630a..0000000
--- a/trunk/src/common_audio/resampler/Android.mk
+++ /dev/null
@@ -1,47 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_resampler
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := resampler.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include \
- $(LOCAL_PATH)/../.. \
- $(LOCAL_PATH)/../signal_processing/include
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
-LOCAL_LDLIBS += -ldl -lpthread
-endif
-
-ifneq ($(TARGET_SIMULATOR),true)
-LOCAL_SHARED_LIBRARIES += libdl
-endif
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/common_audio/resampler/include/resampler.h b/trunk/src/common_audio/resampler/include/resampler.h
deleted file mode 100644
index 38e6bd3..0000000
--- a/trunk/src/common_audio/resampler/include/resampler.h
+++ /dev/null
@@ -1,116 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * A wrapper for resampling a numerous amount of sampling combinations.
- */
-
-#ifndef WEBRTC_RESAMPLER_RESAMPLER_H_
-#define WEBRTC_RESAMPLER_RESAMPLER_H_
-
-#include "typedefs.h"
-
-namespace webrtc
-{
-
-// TODO(andrew): the implementation depends on the exact values of this enum.
-// It should be rewritten in a less fragile way.
-enum ResamplerType
-{
- // 4 MSB = Number of channels
- // 4 LSB = Synchronous or asynchronous
-
- kResamplerSynchronous = 0x10,
- kResamplerAsynchronous = 0x11,
- kResamplerSynchronousStereo = 0x20,
- kResamplerAsynchronousStereo = 0x21,
- kResamplerInvalid = 0xff
-};
-
-// TODO(andrew): doesn't need to be part of the interface.
-enum ResamplerMode
-{
- kResamplerMode1To1,
- kResamplerMode1To2,
- kResamplerMode1To3,
- kResamplerMode1To4,
- kResamplerMode1To6,
- kResamplerMode1To12,
- kResamplerMode2To3,
- kResamplerMode2To11,
- kResamplerMode4To11,
- kResamplerMode8To11,
- kResamplerMode11To16,
- kResamplerMode11To32,
- kResamplerMode2To1,
- kResamplerMode3To1,
- kResamplerMode4To1,
- kResamplerMode6To1,
- kResamplerMode12To1,
- kResamplerMode3To2,
- kResamplerMode11To2,
- kResamplerMode11To4,
- kResamplerMode11To8
-};
-
-class Resampler
-{
-
-public:
- Resampler();
- // TODO(andrew): use an init function instead.
- Resampler(int inFreq, int outFreq, ResamplerType type);
- ~Resampler();
-
- // Reset all states
- int Reset(int inFreq, int outFreq, ResamplerType type);
-
- // Reset all states if any parameter has changed
- int ResetIfNeeded(int inFreq, int outFreq, ResamplerType type);
-
- // Synchronous resampling, all output samples are written to samplesOut
- int Push(const WebRtc_Word16* samplesIn, int lengthIn, WebRtc_Word16* samplesOut,
- int maxLen, int &outLen);
-
- // Asynchronous resampling, input
- int Insert(WebRtc_Word16* samplesIn, int lengthIn);
-
- // Asynchronous resampling output, remaining samples are buffered
- int Pull(WebRtc_Word16* samplesOut, int desiredLen, int &outLen);
-
-private:
- // Generic pointers since we don't know what states we'll need
- void* state1_;
- void* state2_;
- void* state3_;
-
- // Storage if needed
- WebRtc_Word16* in_buffer_;
- WebRtc_Word16* out_buffer_;
- int in_buffer_size_;
- int out_buffer_size_;
- int in_buffer_size_max_;
- int out_buffer_size_max_;
-
- // State
- int my_in_frequency_khz_;
- int my_out_frequency_khz_;
- ResamplerMode my_mode_;
- ResamplerType my_type_;
-
- // Extra instance for stereo
- Resampler* slave_left_;
- Resampler* slave_right_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_RESAMPLER_RESAMPLER_H_
diff --git a/trunk/src/common_audio/resampler/resampler.cc b/trunk/src/common_audio/resampler/resampler.cc
deleted file mode 100644
index 2db27b1..0000000
--- a/trunk/src/common_audio/resampler/resampler.cc
+++ /dev/null
@@ -1,1084 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * A wrapper for resampling a numerous amount of sampling combinations.
- */
-
-#include <stdlib.h>
-#include <string.h>
-
-#include "signal_processing_library.h"
-#include "resampler.h"
-
-
-namespace webrtc
-{
-
-Resampler::Resampler()
-{
- state1_ = NULL;
- state2_ = NULL;
- state3_ = NULL;
- in_buffer_ = NULL;
- out_buffer_ = NULL;
- in_buffer_size_ = 0;
- out_buffer_size_ = 0;
- in_buffer_size_max_ = 0;
- out_buffer_size_max_ = 0;
- // we need a reset before we will work
- my_in_frequency_khz_ = 0;
- my_out_frequency_khz_ = 0;
- my_mode_ = kResamplerMode1To1;
- my_type_ = kResamplerInvalid;
- slave_left_ = NULL;
- slave_right_ = NULL;
-}
-
-Resampler::Resampler(int inFreq, int outFreq, ResamplerType type)
-{
- state1_ = NULL;
- state2_ = NULL;
- state3_ = NULL;
- in_buffer_ = NULL;
- out_buffer_ = NULL;
- in_buffer_size_ = 0;
- out_buffer_size_ = 0;
- in_buffer_size_max_ = 0;
- out_buffer_size_max_ = 0;
- // we need a reset before we will work
- my_in_frequency_khz_ = 0;
- my_out_frequency_khz_ = 0;
- my_mode_ = kResamplerMode1To1;
- my_type_ = kResamplerInvalid;
- slave_left_ = NULL;
- slave_right_ = NULL;
-
- Reset(inFreq, outFreq, type);
-}
-
-Resampler::~Resampler()
-{
- if (state1_)
- {
- free(state1_);
- }
- if (state2_)
- {
- free(state2_);
- }
- if (state3_)
- {
- free(state3_);
- }
- if (in_buffer_)
- {
- free(in_buffer_);
- }
- if (out_buffer_)
- {
- free(out_buffer_);
- }
- if (slave_left_)
- {
- delete slave_left_;
- }
- if (slave_right_)
- {
- delete slave_right_;
- }
-}
-
-int Resampler::ResetIfNeeded(int inFreq, int outFreq, ResamplerType type)
-{
- int tmpInFreq_kHz = inFreq / 1000;
- int tmpOutFreq_kHz = outFreq / 1000;
-
- if ((tmpInFreq_kHz != my_in_frequency_khz_) || (tmpOutFreq_kHz != my_out_frequency_khz_)
- || (type != my_type_))
- {
- return Reset(inFreq, outFreq, type);
- } else
- {
- return 0;
- }
-}
-
-int Resampler::Reset(int inFreq, int outFreq, ResamplerType type)
-{
-
- if (state1_)
- {
- free(state1_);
- state1_ = NULL;
- }
- if (state2_)
- {
- free(state2_);
- state2_ = NULL;
- }
- if (state3_)
- {
- free(state3_);
- state3_ = NULL;
- }
- if (in_buffer_)
- {
- free(in_buffer_);
- in_buffer_ = NULL;
- }
- if (out_buffer_)
- {
- free(out_buffer_);
- out_buffer_ = NULL;
- }
- if (slave_left_)
- {
- delete slave_left_;
- slave_left_ = NULL;
- }
- if (slave_right_)
- {
- delete slave_right_;
- slave_right_ = NULL;
- }
-
- in_buffer_size_ = 0;
- out_buffer_size_ = 0;
- in_buffer_size_max_ = 0;
- out_buffer_size_max_ = 0;
-
- // This might be overridden if parameters are not accepted.
- my_type_ = type;
-
- // Start with a math exercise, Euclid's algorithm to find the gcd:
-
- int a = inFreq;
- int b = outFreq;
- int c = a % b;
- while (c != 0)
- {
- a = b;
- b = c;
- c = a % b;
- }
- // b is now the gcd;
-
- // We need to track what domain we're in.
- my_in_frequency_khz_ = inFreq / 1000;
- my_out_frequency_khz_ = outFreq / 1000;
-
- // Scale with GCD
- inFreq = inFreq / b;
- outFreq = outFreq / b;
-
- // Do we need stereo?
- if ((my_type_ & 0xf0) == 0x20)
- {
- // Change type to mono
- type = static_cast<ResamplerType>(
- ((static_cast<int>(type) & 0x0f) + 0x10));
- slave_left_ = new Resampler(inFreq, outFreq, type);
- slave_right_ = new Resampler(inFreq, outFreq, type);
- }
-
- if (inFreq == outFreq)
- {
- my_mode_ = kResamplerMode1To1;
- } else if (inFreq == 1)
- {
- switch (outFreq)
- {
- case 2:
- my_mode_ = kResamplerMode1To2;
- break;
- case 3:
- my_mode_ = kResamplerMode1To3;
- break;
- case 4:
- my_mode_ = kResamplerMode1To4;
- break;
- case 6:
- my_mode_ = kResamplerMode1To6;
- break;
- case 12:
- my_mode_ = kResamplerMode1To12;
- break;
- default:
- my_type_ = kResamplerInvalid;
- return -1;
- }
- } else if (outFreq == 1)
- {
- switch (inFreq)
- {
- case 2:
- my_mode_ = kResamplerMode2To1;
- break;
- case 3:
- my_mode_ = kResamplerMode3To1;
- break;
- case 4:
- my_mode_ = kResamplerMode4To1;
- break;
- case 6:
- my_mode_ = kResamplerMode6To1;
- break;
- case 12:
- my_mode_ = kResamplerMode12To1;
- break;
- default:
- my_type_ = kResamplerInvalid;
- return -1;
- }
- } else if ((inFreq == 2) && (outFreq == 3))
- {
- my_mode_ = kResamplerMode2To3;
- } else if ((inFreq == 2) && (outFreq == 11))
- {
- my_mode_ = kResamplerMode2To11;
- } else if ((inFreq == 4) && (outFreq == 11))
- {
- my_mode_ = kResamplerMode4To11;
- } else if ((inFreq == 8) && (outFreq == 11))
- {
- my_mode_ = kResamplerMode8To11;
- } else if ((inFreq == 3) && (outFreq == 2))
- {
- my_mode_ = kResamplerMode3To2;
- } else if ((inFreq == 11) && (outFreq == 2))
- {
- my_mode_ = kResamplerMode11To2;
- } else if ((inFreq == 11) && (outFreq == 4))
- {
- my_mode_ = kResamplerMode11To4;
- } else if ((inFreq == 11) && (outFreq == 16))
- {
- my_mode_ = kResamplerMode11To16;
- } else if ((inFreq == 11) && (outFreq == 32))
- {
- my_mode_ = kResamplerMode11To32;
- } else if ((inFreq == 11) && (outFreq == 8))
- {
- my_mode_ = kResamplerMode11To8;
- } else
- {
- my_type_ = kResamplerInvalid;
- return -1;
- }
-
- // Now create the states we need
- switch (my_mode_)
- {
- case kResamplerMode1To1:
- // No state needed;
- break;
- case kResamplerMode1To2:
- state1_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
- break;
- case kResamplerMode1To3:
- state1_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
- WebRtcSpl_ResetResample16khzTo48khz((WebRtcSpl_State16khzTo48khz *)state1_);
- break;
- case kResamplerMode1To4:
- // 1:2
- state1_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
- // 2:4
- state2_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
- break;
- case kResamplerMode1To6:
- // 1:2
- state1_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
- // 2:6
- state2_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
- WebRtcSpl_ResetResample16khzTo48khz((WebRtcSpl_State16khzTo48khz *)state2_);
- break;
- case kResamplerMode1To12:
- // 1:2
- state1_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
- // 2:4
- state2_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
- // 4:12
- state3_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
- WebRtcSpl_ResetResample16khzTo48khz(
- (WebRtcSpl_State16khzTo48khz*) state3_);
- break;
- case kResamplerMode2To3:
- // 2:6
- state1_ = malloc(sizeof(WebRtcSpl_State16khzTo48khz));
- WebRtcSpl_ResetResample16khzTo48khz((WebRtcSpl_State16khzTo48khz *)state1_);
- // 6:3
- state2_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
- break;
- case kResamplerMode2To11:
- state1_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
-
- state2_ = malloc(sizeof(WebRtcSpl_State8khzTo22khz));
- WebRtcSpl_ResetResample8khzTo22khz((WebRtcSpl_State8khzTo22khz *)state2_);
- break;
- case kResamplerMode4To11:
- state1_ = malloc(sizeof(WebRtcSpl_State8khzTo22khz));
- WebRtcSpl_ResetResample8khzTo22khz((WebRtcSpl_State8khzTo22khz *)state1_);
- break;
- case kResamplerMode8To11:
- state1_ = malloc(sizeof(WebRtcSpl_State16khzTo22khz));
- WebRtcSpl_ResetResample16khzTo22khz((WebRtcSpl_State16khzTo22khz *)state1_);
- break;
- case kResamplerMode11To16:
- state1_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
-
- state2_ = malloc(sizeof(WebRtcSpl_State22khzTo16khz));
- WebRtcSpl_ResetResample22khzTo16khz((WebRtcSpl_State22khzTo16khz *)state2_);
- break;
- case kResamplerMode11To32:
- // 11 -> 22
- state1_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
-
- // 22 -> 16
- state2_ = malloc(sizeof(WebRtcSpl_State22khzTo16khz));
- WebRtcSpl_ResetResample22khzTo16khz((WebRtcSpl_State22khzTo16khz *)state2_);
-
- // 16 -> 32
- state3_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state3_, 0, 8 * sizeof(WebRtc_Word32));
-
- break;
- case kResamplerMode2To1:
- state1_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
- break;
- case kResamplerMode3To1:
- state1_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
- WebRtcSpl_ResetResample48khzTo16khz((WebRtcSpl_State48khzTo16khz *)state1_);
- break;
- case kResamplerMode4To1:
- // 4:2
- state1_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
- // 2:1
- state2_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
- break;
- case kResamplerMode6To1:
- // 6:2
- state1_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
- WebRtcSpl_ResetResample48khzTo16khz((WebRtcSpl_State48khzTo16khz *)state1_);
- // 2:1
- state2_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
- break;
- case kResamplerMode12To1:
- // 12:4
- state1_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
- WebRtcSpl_ResetResample48khzTo16khz(
- (WebRtcSpl_State48khzTo16khz*) state1_);
- // 4:2
- state2_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
- // 2:1
- state3_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state3_, 0, 8 * sizeof(WebRtc_Word32));
- break;
- case kResamplerMode3To2:
- // 3:6
- state1_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state1_, 0, 8 * sizeof(WebRtc_Word32));
- // 6:2
- state2_ = malloc(sizeof(WebRtcSpl_State48khzTo16khz));
- WebRtcSpl_ResetResample48khzTo16khz((WebRtcSpl_State48khzTo16khz *)state2_);
- break;
- case kResamplerMode11To2:
- state1_ = malloc(sizeof(WebRtcSpl_State22khzTo8khz));
- WebRtcSpl_ResetResample22khzTo8khz((WebRtcSpl_State22khzTo8khz *)state1_);
-
- state2_ = malloc(8 * sizeof(WebRtc_Word32));
- memset(state2_, 0, 8 * sizeof(WebRtc_Word32));
-
- break;
- case kResamplerMode11To4:
- state1_ = malloc(sizeof(WebRtcSpl_State22khzTo8khz));
- WebRtcSpl_ResetResample22khzTo8khz((WebRtcSpl_State22khzTo8khz *)state1_);
- break;
- case kResamplerMode11To8:
- state1_ = malloc(sizeof(WebRtcSpl_State22khzTo16khz));
- WebRtcSpl_ResetResample22khzTo16khz((WebRtcSpl_State22khzTo16khz *)state1_);
- break;
-
- }
-
- return 0;
-}
-
-// Synchronous resampling, all output samples are written to samplesOut
-int Resampler::Push(const WebRtc_Word16 * samplesIn, int lengthIn, WebRtc_Word16* samplesOut,
- int maxLen, int &outLen)
-{
- // Check that the resampler is not in asynchronous mode
- if (my_type_ & 0x0f)
- {
- return -1;
- }
-
- // Do we have a stereo signal?
- if ((my_type_ & 0xf0) == 0x20)
- {
-
- // Split up the signal and call the slave object for each channel
-
- WebRtc_Word16* left = (WebRtc_Word16*)malloc(lengthIn * sizeof(WebRtc_Word16) / 2);
- WebRtc_Word16* right = (WebRtc_Word16*)malloc(lengthIn * sizeof(WebRtc_Word16) / 2);
- WebRtc_Word16* out_left = (WebRtc_Word16*)malloc(maxLen / 2 * sizeof(WebRtc_Word16));
- WebRtc_Word16* out_right =
- (WebRtc_Word16*)malloc(maxLen / 2 * sizeof(WebRtc_Word16));
- int res = 0;
- for (int i = 0; i < lengthIn; i += 2)
- {
- left[i >> 1] = samplesIn[i];
- right[i >> 1] = samplesIn[i + 1];
- }
-
- // It's OK to overwrite the local parameter, since it's just a copy
- lengthIn = lengthIn / 2;
-
- int actualOutLen_left = 0;
- int actualOutLen_right = 0;
- // Do resampling for right channel
- res |= slave_left_->Push(left, lengthIn, out_left, maxLen / 2, actualOutLen_left);
- res |= slave_right_->Push(right, lengthIn, out_right, maxLen / 2, actualOutLen_right);
- if (res || (actualOutLen_left != actualOutLen_right))
- {
- free(left);
- free(right);
- free(out_left);
- free(out_right);
- return -1;
- }
-
- // Reassemble the signal
- for (int i = 0; i < actualOutLen_left; i++)
- {
- samplesOut[i * 2] = out_left[i];
- samplesOut[i * 2 + 1] = out_right[i];
- }
- outLen = 2 * actualOutLen_left;
-
- free(left);
- free(right);
- free(out_left);
- free(out_right);
-
- return 0;
- }
-
- // Containers for temp samples
- WebRtc_Word16* tmp;
- WebRtc_Word16* tmp_2;
- // tmp data for resampling routines
- WebRtc_Word32* tmp_mem;
-
- switch (my_mode_)
- {
- case kResamplerMode1To1:
- memcpy(samplesOut, samplesIn, lengthIn * sizeof(WebRtc_Word16));
- outLen = lengthIn;
- break;
- case kResamplerMode1To2:
- if (maxLen < (lengthIn * 2))
- {
- return -1;
- }
- WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, samplesOut, (WebRtc_Word32*)state1_);
- outLen = lengthIn * 2;
- return 0;
- case kResamplerMode1To3:
-
- // We can only handle blocks of 160 samples
- // Can be fixed, but I don't think it's needed
- if ((lengthIn % 160) != 0)
- {
- return -1;
- }
- if (maxLen < (lengthIn * 3))
- {
- return -1;
- }
- tmp_mem = (WebRtc_Word32*)malloc(336 * sizeof(WebRtc_Word32));
-
- for (int i = 0; i < lengthIn; i += 160)
- {
- WebRtcSpl_Resample16khzTo48khz(samplesIn + i, samplesOut + i * 3,
- (WebRtcSpl_State16khzTo48khz *)state1_,
- tmp_mem);
- }
- outLen = lengthIn * 3;
- free(tmp_mem);
- return 0;
- case kResamplerMode1To4:
- if (maxLen < (lengthIn * 4))
- {
- return -1;
- }
-
- tmp = (WebRtc_Word16*)malloc(sizeof(WebRtc_Word16) * 2 * lengthIn);
- // 1:2
- WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
- // 2:4
- WebRtcSpl_UpsampleBy2(tmp, lengthIn * 2, samplesOut, (WebRtc_Word32*)state2_);
- outLen = lengthIn * 4;
- free(tmp);
- return 0;
- case kResamplerMode1To6:
- // We can only handle blocks of 80 samples
- // Can be fixed, but I don't think it's needed
- if ((lengthIn % 80) != 0)
- {
- return -1;
- }
- if (maxLen < (lengthIn * 6))
- {
- return -1;
- }
-
- //1:2
-
- tmp_mem = (WebRtc_Word32*)malloc(336 * sizeof(WebRtc_Word32));
- tmp = (WebRtc_Word16*)malloc(sizeof(WebRtc_Word16) * 2 * lengthIn);
-
- WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
- outLen = lengthIn * 2;
-
- for (int i = 0; i < outLen; i += 160)
- {
- WebRtcSpl_Resample16khzTo48khz(tmp + i, samplesOut + i * 3,
- (WebRtcSpl_State16khzTo48khz *)state2_,
- tmp_mem);
- }
- outLen = outLen * 3;
- free(tmp_mem);
- free(tmp);
-
- return 0;
- case kResamplerMode1To12:
- // We can only handle blocks of 40 samples
- // Can be fixed, but I don't think it's needed
- if ((lengthIn % 40) != 0) {
- return -1;
- }
- if (maxLen < (lengthIn * 12)) {
- return -1;
- }
-
- tmp_mem = (WebRtc_Word32*) malloc(336 * sizeof(WebRtc_Word32));
- tmp = (WebRtc_Word16*) malloc(sizeof(WebRtc_Word16) * 4 * lengthIn);
- //1:2
- WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, samplesOut,
- (WebRtc_Word32*) state1_);
- outLen = lengthIn * 2;
- //2:4
- WebRtcSpl_UpsampleBy2(samplesOut, outLen, tmp, (WebRtc_Word32*) state2_);
- outLen = outLen * 2;
- // 4:12
- for (int i = 0; i < outLen; i += 160) {
- // WebRtcSpl_Resample16khzTo48khz() takes a block of 160 samples
- // as input and outputs a resampled block of 480 samples. The
- // data is now actually in 32 kHz sampling rate, despite the
- // function name, and with a resampling factor of three becomes
- // 96 kHz.
- WebRtcSpl_Resample16khzTo48khz(tmp + i, samplesOut + i * 3,
- (WebRtcSpl_State16khzTo48khz*) state3_,
- tmp_mem);
- }
- outLen = outLen * 3;
- free(tmp_mem);
- free(tmp);
-
- return 0;
- case kResamplerMode2To3:
- if (maxLen < (lengthIn * 3 / 2))
- {
- return -1;
- }
- // 2:6
- // We can only handle blocks of 160 samples
- // Can be fixed, but I don't think it's needed
- if ((lengthIn % 160) != 0)
- {
- return -1;
- }
- tmp = static_cast<WebRtc_Word16*> (malloc(sizeof(WebRtc_Word16) * lengthIn * 3));
- tmp_mem = (WebRtc_Word32*)malloc(336 * sizeof(WebRtc_Word32));
- for (int i = 0; i < lengthIn; i += 160)
- {
- WebRtcSpl_Resample16khzTo48khz(samplesIn + i, tmp + i * 3,
- (WebRtcSpl_State16khzTo48khz *)state1_,
- tmp_mem);
- }
- lengthIn = lengthIn * 3;
- // 6:3
- WebRtcSpl_DownsampleBy2(tmp, lengthIn, samplesOut, (WebRtc_Word32*)state2_);
- outLen = lengthIn / 2;
- free(tmp);
- free(tmp_mem);
- return 0;
- case kResamplerMode2To11:
-
- // We can only handle blocks of 80 samples
- // Can be fixed, but I don't think it's needed
- if ((lengthIn % 80) != 0)
- {
- return -1;
- }
- if (maxLen < ((lengthIn * 11) / 2))
- {
- return -1;
- }
- tmp = (WebRtc_Word16*)malloc(sizeof(WebRtc_Word16) * 2 * lengthIn);
- // 1:2
- WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
- lengthIn *= 2;
-
- tmp_mem = (WebRtc_Word32*)malloc(98 * sizeof(WebRtc_Word32));
-
- for (int i = 0; i < lengthIn; i += 80)
- {
- WebRtcSpl_Resample8khzTo22khz(tmp + i, samplesOut + (i * 11) / 4,
- (WebRtcSpl_State8khzTo22khz *)state2_,
- tmp_mem);
- }
- outLen = (lengthIn * 11) / 4;
- free(tmp_mem);
- free(tmp);
- return 0;
- case kResamplerMode4To11:
-
- // We can only handle blocks of 80 samples
- // Can be fixed, but I don't think it's needed
- if ((lengthIn % 80) != 0)
- {
- return -1;
- }
- if (maxLen < ((lengthIn * 11) / 4))
- {
- return -1;
- }
- tmp_mem = (WebRtc_Word32*)malloc(98 * sizeof(WebRtc_Word32));
-
- for (int i = 0; i < lengthIn; i += 80)
- {
- WebRtcSpl_Resample8khzTo22khz(samplesIn + i, samplesOut + (i * 11) / 4,
- (WebRtcSpl_State8khzTo22khz *)state1_,
- tmp_mem);
- }
- outLen = (lengthIn * 11) / 4;
- free(tmp_mem);
- return 0;
- case kResamplerMode8To11:
- // We can only handle blocks of 160 samples
- // Can be fixed, but I don't think it's needed
- if ((lengthIn % 160) != 0)
- {
- return -1;
- }
- if (maxLen < ((lengthIn * 11) / 8))
- {
- return -1;
- }
- tmp_mem = (WebRtc_Word32*)malloc(88 * sizeof(WebRtc_Word32));
-
- for (int i = 0; i < lengthIn; i += 160)
- {
- WebRtcSpl_Resample16khzTo22khz(samplesIn + i, samplesOut + (i * 11) / 8,
- (WebRtcSpl_State16khzTo22khz *)state1_,
- tmp_mem);
- }
- outLen = (lengthIn * 11) / 8;
- free(tmp_mem);
- return 0;
-
- case kResamplerMode11To16:
- // We can only handle blocks of 110 samples
- if ((lengthIn % 110) != 0)
- {
- return -1;
- }
- if (maxLen < ((lengthIn * 16) / 11))
- {
- return -1;
- }
-
- tmp_mem = (WebRtc_Word32*)malloc(104 * sizeof(WebRtc_Word32));
- tmp = (WebRtc_Word16*)malloc((sizeof(WebRtc_Word16) * lengthIn * 2));
-
- WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
-
- for (int i = 0; i < (lengthIn * 2); i += 220)
- {
- WebRtcSpl_Resample22khzTo16khz(tmp + i, samplesOut + (i / 220) * 160,
- (WebRtcSpl_State22khzTo16khz *)state2_,
- tmp_mem);
- }
-
- outLen = (lengthIn * 16) / 11;
-
- free(tmp_mem);
- free(tmp);
- return 0;
-
- case kResamplerMode11To32:
-
- // We can only handle blocks of 110 samples
- if ((lengthIn % 110) != 0)
- {
- return -1;
- }
- if (maxLen < ((lengthIn * 32) / 11))
- {
- return -1;
- }
-
- tmp_mem = (WebRtc_Word32*)malloc(104 * sizeof(WebRtc_Word32));
- tmp = (WebRtc_Word16*)malloc((sizeof(WebRtc_Word16) * lengthIn * 2));
-
- // 11 -> 22 kHz in samplesOut
- WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, samplesOut, (WebRtc_Word32*)state1_);
-
- // 22 -> 16 in tmp
- for (int i = 0; i < (lengthIn * 2); i += 220)
- {
- WebRtcSpl_Resample22khzTo16khz(samplesOut + i, tmp + (i / 220) * 160,
- (WebRtcSpl_State22khzTo16khz *)state2_,
- tmp_mem);
- }
-
- // 16 -> 32 in samplesOut
- WebRtcSpl_UpsampleBy2(tmp, (lengthIn * 16) / 11, samplesOut,
- (WebRtc_Word32*)state3_);
-
- outLen = (lengthIn * 32) / 11;
-
- free(tmp_mem);
- free(tmp);
- return 0;
-
- case kResamplerMode2To1:
- if (maxLen < (lengthIn / 2))
- {
- return -1;
- }
- WebRtcSpl_DownsampleBy2(samplesIn, lengthIn, samplesOut, (WebRtc_Word32*)state1_);
- outLen = lengthIn / 2;
- return 0;
- case kResamplerMode3To1:
- // We can only handle blocks of 480 samples
- // Can be fixed, but I don't think it's needed
- if ((lengthIn % 480) != 0)
- {
- return -1;
- }
- if (maxLen < (lengthIn / 3))
- {
- return -1;
- }
- tmp_mem = (WebRtc_Word32*)malloc(496 * sizeof(WebRtc_Word32));
-
- for (int i = 0; i < lengthIn; i += 480)
- {
- WebRtcSpl_Resample48khzTo16khz(samplesIn + i, samplesOut + i / 3,
- (WebRtcSpl_State48khzTo16khz *)state1_,
- tmp_mem);
- }
- outLen = lengthIn / 3;
- free(tmp_mem);
- return 0;
- case kResamplerMode4To1:
- if (maxLen < (lengthIn / 4))
- {
- return -1;
- }
- tmp = (WebRtc_Word16*)malloc(sizeof(WebRtc_Word16) * lengthIn / 2);
- // 4:2
- WebRtcSpl_DownsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
- // 2:1
- WebRtcSpl_DownsampleBy2(tmp, lengthIn / 2, samplesOut, (WebRtc_Word32*)state2_);
- outLen = lengthIn / 4;
- free(tmp);
- return 0;
-
- case kResamplerMode6To1:
- // We can only handle blocks of 480 samples
- // Can be fixed, but I don't think it's needed
- if ((lengthIn % 480) != 0)
- {
- return -1;
- }
- if (maxLen < (lengthIn / 6))
- {
- return -1;
- }
-
- tmp_mem = (WebRtc_Word32*)malloc(496 * sizeof(WebRtc_Word32));
- tmp = (WebRtc_Word16*)malloc((sizeof(WebRtc_Word16) * lengthIn) / 3);
-
- for (int i = 0; i < lengthIn; i += 480)
- {
- WebRtcSpl_Resample48khzTo16khz(samplesIn + i, tmp + i / 3,
- (WebRtcSpl_State48khzTo16khz *)state1_,
- tmp_mem);
- }
- outLen = lengthIn / 3;
- free(tmp_mem);
- WebRtcSpl_DownsampleBy2(tmp, outLen, samplesOut, (WebRtc_Word32*)state2_);
- free(tmp);
- outLen = outLen / 2;
- return 0;
- case kResamplerMode12To1:
- // We can only handle blocks of 480 samples
- // Can be fixed, but I don't think it's needed
- if ((lengthIn % 480) != 0) {
- return -1;
- }
- if (maxLen < (lengthIn / 12)) {
- return -1;
- }
-
- tmp_mem = (WebRtc_Word32*) malloc(496 * sizeof(WebRtc_Word32));
- tmp = (WebRtc_Word16*) malloc((sizeof(WebRtc_Word16) * lengthIn) / 3);
- tmp_2 = (WebRtc_Word16*) malloc((sizeof(WebRtc_Word16) * lengthIn) / 6);
- // 12:4
- for (int i = 0; i < lengthIn; i += 480) {
- // WebRtcSpl_Resample48khzTo16khz() takes a block of 480 samples
- // as input and outputs a resampled block of 160 samples. The
- // data is now actually in 96 kHz sampling rate, despite the
- // function name, and with a resampling factor of 1/3 becomes
- // 32 kHz.
- WebRtcSpl_Resample48khzTo16khz(samplesIn + i, tmp + i / 3,
- (WebRtcSpl_State48khzTo16khz*) state1_,
- tmp_mem);
- }
- outLen = lengthIn / 3;
- free(tmp_mem);
- // 4:2
- WebRtcSpl_DownsampleBy2(tmp, outLen, tmp_2,
- (WebRtc_Word32*) state2_);
- outLen = outLen / 2;
- free(tmp);
- // 2:1
- WebRtcSpl_DownsampleBy2(tmp_2, outLen, samplesOut,
- (WebRtc_Word32*) state3_);
- free(tmp_2);
- outLen = outLen / 2;
- return 0;
- case kResamplerMode3To2:
- if (maxLen < (lengthIn * 2 / 3))
- {
- return -1;
- }
- // 3:6
- tmp = static_cast<WebRtc_Word16*> (malloc(sizeof(WebRtc_Word16) * lengthIn * 2));
- WebRtcSpl_UpsampleBy2(samplesIn, lengthIn, tmp, (WebRtc_Word32*)state1_);
- lengthIn *= 2;
- // 6:2
- // We can only handle blocks of 480 samples
- // Can be fixed, but I don't think it's needed
- if ((lengthIn % 480) != 0)
- {
- free(tmp);
- return -1;
- }
- tmp_mem = (WebRtc_Word32*)malloc(496 * sizeof(WebRtc_Word32));
- for (int i = 0; i < lengthIn; i += 480)
- {
- WebRtcSpl_Resample48khzTo16khz(tmp + i, samplesOut + i / 3,
- (WebRtcSpl_State48khzTo16khz *)state2_,
- tmp_mem);
- }
- outLen = lengthIn / 3;
- free(tmp);
- free(tmp_mem);
- return 0;
- case kResamplerMode11To2:
- // We can only handle blocks of 220 samples
- // Can be fixed, but I don't think it's needed
- if ((lengthIn % 220) != 0)
- {
- return -1;
- }
- if (maxLen < ((lengthIn * 2) / 11))
- {
- return -1;
- }
- tmp_mem = (WebRtc_Word32*)malloc(126 * sizeof(WebRtc_Word32));
- tmp = (WebRtc_Word16*)malloc((lengthIn * 4) / 11 * sizeof(WebRtc_Word16));
-
- for (int i = 0; i < lengthIn; i += 220)
- {
- WebRtcSpl_Resample22khzTo8khz(samplesIn + i, tmp + (i * 4) / 11,
- (WebRtcSpl_State22khzTo8khz *)state1_,
- tmp_mem);
- }
- lengthIn = (lengthIn * 4) / 11;
-
- WebRtcSpl_DownsampleBy2(tmp, lengthIn, samplesOut, (WebRtc_Word32*)state2_);
- outLen = lengthIn / 2;
-
- free(tmp_mem);
- free(tmp);
- return 0;
- case kResamplerMode11To4:
- // We can only handle blocks of 220 samples
- // Can be fixed, but I don't think it's needed
- if ((lengthIn % 220) != 0)
- {
- return -1;
- }
- if (maxLen < ((lengthIn * 4) / 11))
- {
- return -1;
- }
- tmp_mem = (WebRtc_Word32*)malloc(126 * sizeof(WebRtc_Word32));
-
- for (int i = 0; i < lengthIn; i += 220)
- {
- WebRtcSpl_Resample22khzTo8khz(samplesIn + i, samplesOut + (i * 4) / 11,
- (WebRtcSpl_State22khzTo8khz *)state1_,
- tmp_mem);
- }
- outLen = (lengthIn * 4) / 11;
- free(tmp_mem);
- return 0;
- case kResamplerMode11To8:
- // We can only handle blocks of 160 samples
- // Can be fixed, but I don't think it's needed
- if ((lengthIn % 220) != 0)
- {
- return -1;
- }
- if (maxLen < ((lengthIn * 8) / 11))
- {
- return -1;
- }
- tmp_mem = (WebRtc_Word32*)malloc(104 * sizeof(WebRtc_Word32));
-
- for (int i = 0; i < lengthIn; i += 220)
- {
- WebRtcSpl_Resample22khzTo16khz(samplesIn + i, samplesOut + (i * 8) / 11,
- (WebRtcSpl_State22khzTo16khz *)state1_,
- tmp_mem);
- }
- outLen = (lengthIn * 8) / 11;
- free(tmp_mem);
- return 0;
- break;
-
- }
- return 0;
-}
-
-// Asynchronous resampling, input
-int Resampler::Insert(WebRtc_Word16 * samplesIn, int lengthIn)
-{
- if (my_type_ != kResamplerAsynchronous)
- {
- return -1;
- }
- int sizeNeeded, tenMsblock;
-
- // Determine need for size of outBuffer
- sizeNeeded = out_buffer_size_ + ((lengthIn + in_buffer_size_) * my_out_frequency_khz_)
- / my_in_frequency_khz_;
- if (sizeNeeded > out_buffer_size_max_)
- {
- // Round the value upwards to complete 10 ms blocks
- tenMsblock = my_out_frequency_khz_ * 10;
- sizeNeeded = (sizeNeeded / tenMsblock + 1) * tenMsblock;
- out_buffer_ = (WebRtc_Word16*)realloc(out_buffer_, sizeNeeded * sizeof(WebRtc_Word16));
- out_buffer_size_max_ = sizeNeeded;
- }
-
- // If we need to use inBuffer, make sure all input data fits there.
-
- tenMsblock = my_in_frequency_khz_ * 10;
- if (in_buffer_size_ || (lengthIn % tenMsblock))
- {
- // Check if input buffer size is enough
- if ((in_buffer_size_ + lengthIn) > in_buffer_size_max_)
- {
- // Round the value upwards to complete 10 ms blocks
- sizeNeeded = ((in_buffer_size_ + lengthIn) / tenMsblock + 1) * tenMsblock;
- in_buffer_ = (WebRtc_Word16*)realloc(in_buffer_,
- sizeNeeded * sizeof(WebRtc_Word16));
- in_buffer_size_max_ = sizeNeeded;
- }
- // Copy in data to input buffer
- memcpy(in_buffer_ + in_buffer_size_, samplesIn, lengthIn * sizeof(WebRtc_Word16));
-
- // Resample all available 10 ms blocks
- int lenOut;
- int dataLenToResample = (in_buffer_size_ / tenMsblock) * tenMsblock;
- Push(in_buffer_, dataLenToResample, out_buffer_ + out_buffer_size_,
- out_buffer_size_max_ - out_buffer_size_, lenOut);
- out_buffer_size_ += lenOut;
-
- // Save the rest
- memmove(in_buffer_, in_buffer_ + dataLenToResample,
- (in_buffer_size_ - dataLenToResample) * sizeof(WebRtc_Word16));
- in_buffer_size_ -= dataLenToResample;
- } else
- {
- // Just resample
- int lenOut;
- Push(in_buffer_, lengthIn, out_buffer_ + out_buffer_size_,
- out_buffer_size_max_ - out_buffer_size_, lenOut);
- out_buffer_size_ += lenOut;
- }
-
- return 0;
-}
-
-// Asynchronous resampling output, remaining samples are buffered
-int Resampler::Pull(WebRtc_Word16* samplesOut, int desiredLen, int &outLen)
-{
- if (my_type_ != kResamplerAsynchronous)
- {
- return -1;
- }
-
- // Check that we have enough data
- if (desiredLen <= out_buffer_size_)
- {
- // Give out the date
- memcpy(samplesOut, out_buffer_, desiredLen * sizeof(WebRtc_Word32));
-
- // Shuffle down remaining
- memmove(out_buffer_, out_buffer_ + desiredLen,
- (out_buffer_size_ - desiredLen) * sizeof(WebRtc_Word16));
-
- // Update remaining size
- out_buffer_size_ -= desiredLen;
-
- return 0;
- } else
- {
- return -1;
- }
-}
-
-} // namespace webrtc
diff --git a/trunk/src/common_audio/resampler/resampler.gypi b/trunk/src/common_audio/resampler/resampler.gypi
deleted file mode 100644
index 69f9b0e..0000000
--- a/trunk/src/common_audio/resampler/resampler.gypi
+++ /dev/null
@@ -1,55 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'resampler',
- 'type': '<(library)',
- 'dependencies': [
- 'signal_processing',
- ],
- 'include_dirs': [
- 'include',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- ],
- },
- 'sources': [
- 'include/resampler.h',
- 'resampler.cc',
- ],
- },
- ], # targets
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets' : [
- {
- 'target_name': 'resampler_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'resampler',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'sources': [
- 'resampler_unittest.cc',
- ],
- }, # resampler_unittests
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/common_audio/resampler/resampler_unittest.cc b/trunk/src/common_audio/resampler/resampler_unittest.cc
deleted file mode 100644
index 9b1061a..0000000
--- a/trunk/src/common_audio/resampler/resampler_unittest.cc
+++ /dev/null
@@ -1,143 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "gtest/gtest.h"
-
-#include "common_audio/resampler/include/resampler.h"
-
-// TODO(andrew): this is a work-in-progress. Many more tests are needed.
-
-namespace webrtc {
-namespace {
-const ResamplerType kTypes[] = {
- kResamplerSynchronous,
- kResamplerAsynchronous,
- kResamplerSynchronousStereo,
- kResamplerAsynchronousStereo
- // kResamplerInvalid excluded
-};
-const size_t kTypesSize = sizeof(kTypes) / sizeof(*kTypes);
-
-// Rates we must support.
-const int kMaxRate = 96000;
-const int kRates[] = {
- 8000,
- 16000,
- 32000,
- 44000,
- 48000,
- kMaxRate
-};
-const size_t kRatesSize = sizeof(kRates) / sizeof(*kRates);
-const int kMaxChannels = 2;
-const size_t kDataSize = static_cast<size_t> (kMaxChannels * kMaxRate / 100);
-
-// TODO(andrew): should we be supporting these combinations?
-bool ValidRates(int in_rate, int out_rate) {
- // Not the most compact notation, for clarity.
- if ((in_rate == 44000 && (out_rate == 48000 || out_rate == 96000)) ||
- (out_rate == 44000 && (in_rate == 48000 || in_rate == 96000))) {
- return false;
- }
-
- return true;
-}
-
-class ResamplerTest : public testing::Test {
- protected:
- ResamplerTest();
- virtual void SetUp();
- virtual void TearDown();
-
- Resampler rs_;
- int16_t data_in_[kDataSize];
- int16_t data_out_[kDataSize];
-};
-
-ResamplerTest::ResamplerTest() {}
-
-void ResamplerTest::SetUp() {
- // Initialize input data with anything. The tests are content independent.
- memset(data_in_, 1, sizeof(data_in_));
-}
-
-void ResamplerTest::TearDown() {}
-
-TEST_F(ResamplerTest, Reset) {
- // The only failure mode for the constructor is if Reset() fails. For the
- // time being then (until an Init function is added), we rely on Reset()
- // to test the constructor.
-
- // Check that all required combinations are supported.
- for (size_t i = 0; i < kRatesSize; ++i) {
- for (size_t j = 0; j < kRatesSize; ++j) {
- for (size_t k = 0; k < kTypesSize; ++k) {
- std::ostringstream ss;
- ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j]
- << ", type: " << kTypes[k];
- SCOPED_TRACE(ss.str());
- if (ValidRates(kRates[i], kRates[j]))
- EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kTypes[k]));
- else
- EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kTypes[k]));
- }
- }
- }
-}
-
-// TODO(tlegrand): Replace code inside the two tests below with a function
-// with number of channels and ResamplerType as input.
-TEST_F(ResamplerTest, Synchronous) {
- for (size_t i = 0; i < kRatesSize; ++i) {
- for (size_t j = 0; j < kRatesSize; ++j) {
- std::ostringstream ss;
- ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j];
- SCOPED_TRACE(ss.str());
-
- if (ValidRates(kRates[i], kRates[j])) {
- int in_length = kRates[i] / 100;
- int out_length = 0;
- EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j], kResamplerSynchronous));
- EXPECT_EQ(0, rs_.Push(data_in_, in_length, data_out_, kDataSize,
- out_length));
- EXPECT_EQ(kRates[j] / 100, out_length);
- } else {
- EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j], kResamplerSynchronous));
- }
- }
- }
-}
-
-TEST_F(ResamplerTest, SynchronousStereo) {
- // Number of channels is 2, stereo mode.
- const int kChannels = 2;
- for (size_t i = 0; i < kRatesSize; ++i) {
- for (size_t j = 0; j < kRatesSize; ++j) {
- std::ostringstream ss;
- ss << "Input rate: " << kRates[i] << ", output rate: " << kRates[j];
- SCOPED_TRACE(ss.str());
-
- if (ValidRates(kRates[i], kRates[j])) {
- int in_length = kChannels * kRates[i] / 100;
- int out_length = 0;
- EXPECT_EQ(0, rs_.Reset(kRates[i], kRates[j],
- kResamplerSynchronousStereo));
- EXPECT_EQ(0, rs_.Push(data_in_, in_length, data_out_, kDataSize,
- out_length));
- EXPECT_EQ(kChannels * kRates[j] / 100, out_length);
- } else {
- EXPECT_EQ(-1, rs_.Reset(kRates[i], kRates[j],
- kResamplerSynchronousStereo));
- }
- }
- }
-}
-} // namespace
-} // namespace webrtc
diff --git a/trunk/src/common_audio/signal_processing/Android.mk b/trunk/src/common_audio/signal_processing/Android.mk
deleted file mode 100644
index 3ff066c..0000000
--- a/trunk/src/common_audio/signal_processing/Android.mk
+++ /dev/null
@@ -1,100 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_spl
-LOCAL_MODULE_TAGS := optional
-LOCAL_SRC_FILES := \
- auto_corr_to_refl_coef.c \
- auto_correlation.c \
- complex_fft.c \
- copy_set_operations.c \
- division_operations.c \
- dot_product_with_scale.c \
- energy.c \
- filter_ar.c \
- filter_ma_fast_q12.c \
- get_hanning_window.c \
- get_scaling_square.c \
- ilbc_specific_functions.c \
- levinson_durbin.c \
- lpc_to_refl_coef.c \
- min_max_operations.c \
- randomization_functions.c \
- refl_coef_to_lpc.c \
- resample.c \
- resample_48khz.c \
- resample_by_2.c \
- resample_by_2_internal.c \
- resample_fractional.c \
- spl_sqrt.c \
- spl_version.c \
- splitting_filter.c \
- sqrt_of_one_minus_x_squared.c \
- vector_scaling_operations.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include \
- $(LOCAL_PATH)/../..
-
-ifeq ($(ARCH_ARM_HAVE_NEON),true)
-LOCAL_SRC_FILES += \
- min_max_operations_neon.c \
- cross_correlation_neon.s \
- downsample_fast_neon.s
-LOCAL_CFLAGS += \
- $(MY_ARM_CFLAGS_NEON)
-else
-LOCAL_SRC_FILES += \
- cross_correlation.c \
- downsample_fast.c
-endif
-
-ifeq ($(ARCH_ARM_HAVE_ARMV7A),true)
-LOCAL_SRC_FILES += \
- filter_ar_fast_q12_armv7.s
-else
-LOCAL_SRC_FILES += \
- filter_ar_fast_q12.c
-endif
-
-ifeq ($(TARGET_ARCH),arm)
-LOCAL_SRC_FILES += \
- complex_bit_reverse_arm.s \
- spl_sqrt_floor.s
-else
-LOCAL_SRC_FILES += \
- complex_bit_reverse.c \
- spl_sqrt_floor.c
-endif
-
-LOCAL_SHARED_LIBRARIES := libstlport
-
-ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
-LOCAL_LDLIBS += -ldl -lpthread
-endif
-
-ifneq ($(TARGET_SIMULATOR),true)
-LOCAL_SHARED_LIBRARIES += libdl
-endif
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/common_audio/signal_processing/auto_corr_to_refl_coef.c b/trunk/src/common_audio/signal_processing/auto_corr_to_refl_coef.c
deleted file mode 100644
index b7e8858..0000000
--- a/trunk/src/common_audio/signal_processing/auto_corr_to_refl_coef.c
+++ /dev/null
@@ -1,103 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_AutoCorrToReflCoef().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_AutoCorrToReflCoef(G_CONST WebRtc_Word32 *R, int use_order, WebRtc_Word16 *K)
-{
- int i, n;
- WebRtc_Word16 tmp;
- G_CONST WebRtc_Word32 *rptr;
- WebRtc_Word32 L_num, L_den;
- WebRtc_Word16 *acfptr, *pptr, *wptr, *p1ptr, *w1ptr, ACF[WEBRTC_SPL_MAX_LPC_ORDER],
- P[WEBRTC_SPL_MAX_LPC_ORDER], W[WEBRTC_SPL_MAX_LPC_ORDER];
-
- // Initialize loop and pointers.
- acfptr = ACF;
- rptr = R;
- pptr = P;
- p1ptr = &P[1];
- w1ptr = &W[1];
- wptr = w1ptr;
-
- // First loop; n=0. Determine shifting.
- tmp = WebRtcSpl_NormW32(*R);
- *acfptr = (WebRtc_Word16)((*rptr++ << tmp) >> 16);
- *pptr++ = *acfptr++;
-
- // Initialize ACF, P and W.
- for (i = 1; i <= use_order; i++)
- {
- *acfptr = (WebRtc_Word16)((*rptr++ << tmp) >> 16);
- *wptr++ = *acfptr;
- *pptr++ = *acfptr++;
- }
-
- // Compute reflection coefficients.
- for (n = 1; n <= use_order; n++, K++)
- {
- tmp = WEBRTC_SPL_ABS_W16(*p1ptr);
- if (*P < tmp)
- {
- for (i = n; i <= use_order; i++)
- *K++ = 0;
-
- return;
- }
-
- // Division: WebRtcSpl_div(tmp, *P)
- *K = 0;
- if (tmp != 0)
- {
- L_num = tmp;
- L_den = *P;
- i = 15;
- while (i--)
- {
- (*K) <<= 1;
- L_num <<= 1;
- if (L_num >= L_den)
- {
- L_num -= L_den;
- (*K)++;
- }
- }
- if (*p1ptr > 0)
- *K = -*K;
- }
-
- // Last iteration; don't do Schur recursion.
- if (n == use_order)
- return;
-
- // Schur recursion.
- pptr = P;
- wptr = w1ptr;
- tmp = (WebRtc_Word16)(((WebRtc_Word32)*p1ptr * (WebRtc_Word32)*K + 16384) >> 15);
- *pptr = WEBRTC_SPL_ADD_SAT_W16( *pptr, tmp );
- pptr++;
- for (i = 1; i <= use_order - n; i++)
- {
- tmp = (WebRtc_Word16)(((WebRtc_Word32)*wptr * (WebRtc_Word32)*K + 16384) >> 15);
- *pptr = WEBRTC_SPL_ADD_SAT_W16( *(pptr+1), tmp );
- pptr++;
- tmp = (WebRtc_Word16)(((WebRtc_Word32)*pptr * (WebRtc_Word32)*K + 16384) >> 15);
- *wptr = WEBRTC_SPL_ADD_SAT_W16( *wptr, tmp );
- wptr++;
- }
- }
-}
diff --git a/trunk/src/common_audio/signal_processing/auto_correlation.c b/trunk/src/common_audio/signal_processing/auto_correlation.c
deleted file mode 100644
index a00fde4..0000000
--- a/trunk/src/common_audio/signal_processing/auto_correlation.c
+++ /dev/null
@@ -1,141 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_AutoCorrelation().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-int WebRtcSpl_AutoCorrelation(G_CONST WebRtc_Word16* in_vector,
- int in_vector_length,
- int order,
- WebRtc_Word32* result,
- int* scale)
-{
- WebRtc_Word32 sum;
- int i, j;
- WebRtc_Word16 smax; // Sample max
- G_CONST WebRtc_Word16* xptr1;
- G_CONST WebRtc_Word16* xptr2;
- WebRtc_Word32* resultptr;
- int scaling = 0;
-
-#ifdef _ARM_OPT_
-#pragma message("NOTE: _ARM_OPT_ optimizations are used")
- WebRtc_Word16 loops4;
-#endif
-
- if (order < 0)
- order = in_vector_length;
-
- // Find the max. sample
- smax = WebRtcSpl_MaxAbsValueW16(in_vector, in_vector_length);
-
- // In order to avoid overflow when computing the sum we should scale the samples so that
- // (in_vector_length * smax * smax) will not overflow.
-
- if (smax == 0)
- {
- scaling = 0;
- } else
- {
- int nbits = WebRtcSpl_GetSizeInBits(in_vector_length); // # of bits in the sum loop
- int t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax)); // # of bits to normalize smax
-
- if (t > nbits)
- {
- scaling = 0;
- } else
- {
- scaling = nbits - t;
- }
-
- }
-
- resultptr = result;
-
- // Perform the actual correlation calculation
- for (i = 0; i < order + 1; i++)
- {
- int loops = (in_vector_length - i);
- sum = 0;
- xptr1 = in_vector;
- xptr2 = &in_vector[i];
-#ifndef _ARM_OPT_
- for (j = loops; j > 0; j--)
- {
- sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1++, *xptr2++, scaling);
- }
-#else
- loops4 = (loops >> 2) << 2;
-
- if (scaling == 0)
- {
- for (j = 0; j < loops4; j = j + 4)
- {
- sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
- xptr1++;
- xptr2++;
- sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
- xptr1++;
- xptr2++;
- sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
- xptr1++;
- xptr2++;
- sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
- xptr1++;
- xptr2++;
- }
-
- for (j = loops4; j < loops; j++)
- {
- sum += WEBRTC_SPL_MUL_16_16(*xptr1, *xptr2);
- xptr1++;
- xptr2++;
- }
- }
- else
- {
- for (j = 0; j < loops4; j = j + 4)
- {
- sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
- xptr1++;
- xptr2++;
- sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
- xptr1++;
- xptr2++;
- sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
- xptr1++;
- xptr2++;
- sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
- xptr1++;
- xptr2++;
- }
-
- for (j = loops4; j < loops; j++)
- {
- sum += WEBRTC_SPL_MUL_16_16_RSFT(*xptr1, *xptr2, scaling);
- xptr1++;
- xptr2++;
- }
- }
-
-#endif
- *resultptr++ = sum;
- }
-
- *scale = scaling;
-
- return order + 1;
-}
diff --git a/trunk/src/common_audio/signal_processing/complex_bit_reverse.c b/trunk/src/common_audio/signal_processing/complex_bit_reverse.c
deleted file mode 100644
index 02fde1e..0000000
--- a/trunk/src/common_audio/signal_processing/complex_bit_reverse.c
+++ /dev/null
@@ -1,109 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "signal_processing_library.h"
-
-/* Tables for data buffer indexes that are bit reversed and thus need to be
- * swapped. Note that, index_7[{0, 2, 4, ...}] are for the left side of the swap
- * operations, while index_7[{1, 3, 5, ...}] are for the right side of the
- * operation. Same for index_8.
- */
-
-/* Indexes for the case of stages == 7. */
-static const int16_t index_7[112] = {
- 1, 64, 2, 32, 3, 96, 4, 16, 5, 80, 6, 48, 7, 112, 9, 72, 10, 40, 11, 104,
- 12, 24, 13, 88, 14, 56, 15, 120, 17, 68, 18, 36, 19, 100, 21, 84, 22, 52,
- 23, 116, 25, 76, 26, 44, 27, 108, 29, 92, 30, 60, 31, 124, 33, 66, 35, 98,
- 37, 82, 38, 50, 39, 114, 41, 74, 43, 106, 45, 90, 46, 58, 47, 122, 49, 70,
- 51, 102, 53, 86, 55, 118, 57, 78, 59, 110, 61, 94, 63, 126, 67, 97, 69,
- 81, 71, 113, 75, 105, 77, 89, 79, 121, 83, 101, 87, 117, 91, 109, 95, 125,
- 103, 115, 111, 123
-};
-
-/* Indexes for the case of stages == 8. */
-static const int16_t index_8[240] = {
- 1, 128, 2, 64, 3, 192, 4, 32, 5, 160, 6, 96, 7, 224, 8, 16, 9, 144, 10, 80,
- 11, 208, 12, 48, 13, 176, 14, 112, 15, 240, 17, 136, 18, 72, 19, 200, 20,
- 40, 21, 168, 22, 104, 23, 232, 25, 152, 26, 88, 27, 216, 28, 56, 29, 184,
- 30, 120, 31, 248, 33, 132, 34, 68, 35, 196, 37, 164, 38, 100, 39, 228, 41,
- 148, 42, 84, 43, 212, 44, 52, 45, 180, 46, 116, 47, 244, 49, 140, 50, 76,
- 51, 204, 53, 172, 54, 108, 55, 236, 57, 156, 58, 92, 59, 220, 61, 188, 62,
- 124, 63, 252, 65, 130, 67, 194, 69, 162, 70, 98, 71, 226, 73, 146, 74, 82,
- 75, 210, 77, 178, 78, 114, 79, 242, 81, 138, 83, 202, 85, 170, 86, 106, 87,
- 234, 89, 154, 91, 218, 93, 186, 94, 122, 95, 250, 97, 134, 99, 198, 101,
- 166, 103, 230, 105, 150, 107, 214, 109, 182, 110, 118, 111, 246, 113, 142,
- 115, 206, 117, 174, 119, 238, 121, 158, 123, 222, 125, 190, 127, 254, 131,
- 193, 133, 161, 135, 225, 137, 145, 139, 209, 141, 177, 143, 241, 147, 201,
- 149, 169, 151, 233, 155, 217, 157, 185, 159, 249, 163, 197, 167, 229, 171,
- 213, 173, 181, 175, 245, 179, 205, 183, 237, 187, 221, 191, 253, 199, 227,
- 203, 211, 207, 243, 215, 235, 223, 251, 239, 247
-};
-
-void WebRtcSpl_ComplexBitReverse(int16_t* __restrict complex_data, int stages) {
- /* For any specific value of stages, we know exactly the indexes that are
- * bit reversed. Currently (Feb. 2012) in WebRTC the only possible values of
- * stages are 7 and 8, so we use tables to save unnecessary iterations and
- * calculations for these two cases.
- */
- if (stages == 7 || stages == 8) {
- int m = 0;
- int length = 112;
- const int16_t* index = index_7;
-
- if (stages == 8) {
- length = 240;
- index = index_8;
- }
-
- /* Decimation in time. Swap the elements with bit-reversed indexes. */
- for (m = 0; m < length; m += 2) {
- /* We declare a int32_t* type pointer, to load both the 16-bit real
- * and imaginary elements from complex_data in one instruction, reducing
- * complexity.
- */
- int32_t* complex_data_ptr = (int32_t*)complex_data;
- int32_t temp = 0;
-
- temp = complex_data_ptr[index[m]]; /* Real and imaginary */
- complex_data_ptr[index[m]] = complex_data_ptr[index[m + 1]];
- complex_data_ptr[index[m + 1]] = temp;
- }
- }
- else {
- int m = 0, mr = 0, l = 0;
- int n = 1 << stages;
- int nn = n - 1;
-
- /* Decimation in time - re-order data */
- for (m = 1; m <= nn; ++m) {
- int32_t* complex_data_ptr = (int32_t*)complex_data;
- int32_t temp = 0;
-
- /* Find out indexes that are bit-reversed. */
- l = n;
- do {
- l >>= 1;
- } while (l > nn - mr);
- mr = (mr & (l - 1)) + l;
-
- if (mr <= m) {
- continue;
- }
-
- /* Swap the elements with bit-reversed indexes.
- * This is similar to the loop in the stages == 7 or 8 cases.
- */
- temp = complex_data_ptr[m]; /* Real and imaginary */
- complex_data_ptr[m] = complex_data_ptr[mr];
- complex_data_ptr[mr] = temp;
- }
- }
-}
-
diff --git a/trunk/src/common_audio/signal_processing/complex_bit_reverse_arm.s b/trunk/src/common_audio/signal_processing/complex_bit_reverse_arm.s
deleted file mode 100644
index 4828077..0000000
--- a/trunk/src/common_audio/signal_processing/complex_bit_reverse_arm.s
+++ /dev/null
@@ -1,126 +0,0 @@
-@
-@ Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-@
-@ Use of this source code is governed by a BSD-style license
-@ that can be found in the LICENSE file in the root of the source
-@ tree. An additional intellectual property rights grant can be found
-@ in the file PATENTS. All contributing project authors may
-@ be found in the AUTHORS file in the root of the source tree.
-@
-
-@ This file contains the function WebRtcSpl_ComplexBitReverse(), optimized
-@ for ARMv5 platforms.
-@ Reference C code is in file complex_bit_reverse.c. Bit-exact.
-
-.arch armv5
-
-.global WebRtcSpl_ComplexBitReverse
-
-.align 2
-
-WebRtcSpl_ComplexBitReverse:
-.fnstart
-
- push {r4-r7}
-
- cmp r1, #7
- adr r3, index_7 @ Table pointer.
- mov r4, #112 @ Number of interations.
- beq PRE_LOOP_STAGES_7_OR_8
-
- cmp r1, #8
- adr r3, index_8 @ Table pointer.
- mov r4, #240 @ Number of interations.
- beq PRE_LOOP_STAGES_7_OR_8
-
- mov r3, #1 @ Initialize m.
- mov r1, r3, asl r1 @ n = 1 << stages;
- subs r6, r1, #1 @ nn = n - 1;
- ble END
-
- mov r5, r0 @ &complex_data
- mov r4, #0 @ ml
-
-LOOP_GENERIC:
- rsb r12, r4, r6 @ l > nn - mr
- mov r2, r1 @ n
-
-LOOP_SHIFT:
- asr r2, #1 @ l >>= 1;
- cmp r2, r12
- bgt LOOP_SHIFT
-
- sub r12, r2, #1
- and r4, r12, r4
- add r4, r2 @ mr = (mr & (l - 1)) + l;
- cmp r4, r3 @ mr <= m ?
- ble UPDATE_REGISTERS
-
- mov r12, r4, asl #2
- ldr r7, [r5, #4] @ complex_data[2 * m, 2 * m + 1].
- @ Offset 4 due to m incrementing from 1.
- ldr r2, [r0, r12] @ complex_data[2 * mr, 2 * mr + 1].
- str r7, [r0, r12]
- str r2, [r5, #4]
-
-UPDATE_REGISTERS:
- add r3, r3, #1
- add r5, #4
- cmp r3, r1
- bne LOOP_GENERIC
-
- b END
-
-PRE_LOOP_STAGES_7_OR_8:
- add r4, r3, r4, asl #1
-
-LOOP_STAGES_7_OR_8:
- ldrsh r2, [r3], #2 @ index[m]
- ldrsh r5, [r3], #2 @ index[m + 1]
- ldr r1, [r0, r2] @ complex_data[index[m], index[m] + 1]
- ldr r12, [r0, r5] @ complex_data[index[m + 1], index[m + 1] + 1]
- cmp r3, r4
- str r1, [r0, r5]
- str r12, [r0, r2]
- bne LOOP_STAGES_7_OR_8
-
-END:
- pop {r4-r7}
- bx lr
-
-.fnend
-
-
-@ The index tables. Note the values are doubles of the actual indexes for 16-bit
-@ elements, different from the generic C code. It actually provides byte offsets
-@ for the indexes.
-
-.align 2
-index_7: @ Indexes for stages == 7.
- .hword 4, 256, 8, 128, 12, 384, 16, 64, 20, 320, 24, 192, 28, 448, 36, 288
- .hword 40, 160, 44, 416, 48, 96, 52, 352, 56, 224, 60, 480, 68, 272, 72, 144
- .hword 76, 400, 84, 336, 88, 208, 92, 464, 100, 304, 104, 176, 108, 432, 116
- .hword 368, 120, 240, 124, 496, 132, 264, 140, 392, 148, 328, 152, 200, 156
- .hword 456, 164, 296, 172, 424, 180, 360, 184, 232, 188, 488, 196, 280, 204
- .hword 408, 212, 344, 220, 472, 228, 312, 236, 440, 244, 376, 252, 504, 268
- .hword 388, 276, 324, 284, 452, 300, 420, 308, 356, 316, 484, 332, 404, 348
- .hword 468, 364, 436, 380, 500, 412, 460, 444, 492
-
-index_8: @ Indexes for stages == 8.
- .hword 4, 512, 8, 256, 12, 768, 16, 128, 20, 640, 24, 384, 28, 896, 32, 64
- .hword 36, 576, 40, 320, 44, 832, 48, 192, 52, 704, 56, 448, 60, 960, 68, 544
- .hword 72, 288, 76, 800, 80, 160, 84, 672, 88, 416, 92, 928, 100, 608, 104
- .hword 352, 108, 864, 112, 224, 116, 736, 120, 480, 124, 992, 132, 528, 136
- .hword 272, 140, 784, 148, 656, 152, 400, 156, 912, 164, 592, 168, 336, 172
- .hword 848, 176, 208, 180, 720, 184, 464, 188, 976, 196, 560, 200, 304, 204
- .hword 816, 212, 688, 216, 432, 220, 944, 228, 624, 232, 368, 236, 880, 244
- .hword 752, 248, 496, 252, 1008, 260, 520, 268, 776, 276, 648, 280, 392, 284
- .hword 904, 292, 584, 296, 328, 300, 840, 308, 712, 312, 456, 316, 968, 324
- .hword 552, 332, 808, 340, 680, 344, 424, 348, 936, 356, 616, 364, 872, 372
- .hword 744, 376, 488, 380, 1000, 388, 536, 396, 792, 404, 664, 412, 920, 420
- .hword 600, 428, 856, 436, 728, 440, 472, 444, 984, 452, 568, 460, 824, 468
- .hword 696, 476, 952, 484, 632, 492, 888, 500, 760, 508, 1016, 524, 772, 532
- .hword 644, 540, 900, 548, 580, 556, 836, 564, 708, 572, 964, 588, 804, 596
- .hword 676, 604, 932, 620, 868, 628, 740, 636, 996, 652, 788, 668, 916, 684
- .hword 852, 692, 724, 700, 980, 716, 820, 732, 948, 748, 884, 764, 1012, 796
- .hword 908, 812, 844, 828, 972, 860, 940, 892, 1004, 956, 988
diff --git a/trunk/src/common_audio/signal_processing/complex_fft.c b/trunk/src/common_audio/signal_processing/complex_fft.c
deleted file mode 100644
index 1e8503c..0000000
--- a/trunk/src/common_audio/signal_processing/complex_fft.c
+++ /dev/null
@@ -1,425 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_ComplexFFT().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#define CFFTSFT 14
-#define CFFTRND 1
-#define CFFTRND2 16384
-
-#define CIFFTSFT 14
-#define CIFFTRND 1
-
-static const WebRtc_Word16 kSinTable1024[] = {
- 0, 201, 402, 603, 804, 1005, 1206, 1406,
- 1607, 1808, 2009, 2209, 2410, 2610, 2811, 3011,
- 3211, 3411, 3611, 3811, 4011, 4210, 4409, 4608,
- 4807, 5006, 5205, 5403, 5601, 5799, 5997, 6195,
- 6392, 6589, 6786, 6982, 7179, 7375, 7571, 7766,
- 7961, 8156, 8351, 8545, 8739, 8932, 9126, 9319,
- 9511, 9703, 9895, 10087, 10278, 10469, 10659, 10849,
- 11038, 11227, 11416, 11604, 11792, 11980, 12166, 12353,
- 12539, 12724, 12909, 13094, 13278, 13462, 13645, 13827,
- 14009, 14191, 14372, 14552, 14732, 14911, 15090, 15268,
- 15446, 15623, 15799, 15975, 16150, 16325, 16499, 16672,
- 16845, 17017, 17189, 17360, 17530, 17699, 17868, 18036,
- 18204, 18371, 18537, 18702, 18867, 19031, 19194, 19357,
- 19519, 19680, 19840, 20000, 20159, 20317, 20474, 20631,
- 20787, 20942, 21096, 21249, 21402, 21554, 21705, 21855,
- 22004, 22153, 22301, 22448, 22594, 22739, 22883, 23027,
- 23169, 23311, 23452, 23592, 23731, 23869, 24006, 24143,
- 24278, 24413, 24546, 24679, 24811, 24942, 25072, 25201,
- 25329, 25456, 25582, 25707, 25831, 25954, 26077, 26198,
- 26318, 26437, 26556, 26673, 26789, 26905, 27019, 27132,
- 27244, 27355, 27466, 27575, 27683, 27790, 27896, 28001,
- 28105, 28208, 28309, 28410, 28510, 28608, 28706, 28802,
- 28897, 28992, 29085, 29177, 29268, 29358, 29446, 29534,
- 29621, 29706, 29790, 29873, 29955, 30036, 30116, 30195,
- 30272, 30349, 30424, 30498, 30571, 30643, 30713, 30783,
- 30851, 30918, 30984, 31049,
- 31113, 31175, 31236, 31297,
- 31356, 31413, 31470, 31525, 31580, 31633, 31684, 31735,
- 31785, 31833, 31880, 31926, 31970, 32014, 32056, 32097,
- 32137, 32176, 32213, 32249, 32284, 32318, 32350, 32382,
- 32412, 32441, 32468, 32495, 32520, 32544, 32567, 32588,
- 32609, 32628, 32646, 32662, 32678, 32692, 32705, 32717,
- 32727, 32736, 32744, 32751, 32757, 32761, 32764, 32766,
- 32767, 32766, 32764, 32761, 32757, 32751, 32744, 32736,
- 32727, 32717, 32705, 32692, 32678, 32662, 32646, 32628,
- 32609, 32588, 32567, 32544, 32520, 32495, 32468, 32441,
- 32412, 32382, 32350, 32318, 32284, 32249, 32213, 32176,
- 32137, 32097, 32056, 32014, 31970, 31926, 31880, 31833,
- 31785, 31735, 31684, 31633, 31580, 31525, 31470, 31413,
- 31356, 31297, 31236, 31175, 31113, 31049, 30984, 30918,
- 30851, 30783, 30713, 30643, 30571, 30498, 30424, 30349,
- 30272, 30195, 30116, 30036, 29955, 29873, 29790, 29706,
- 29621, 29534, 29446, 29358, 29268, 29177, 29085, 28992,
- 28897, 28802, 28706, 28608, 28510, 28410, 28309, 28208,
- 28105, 28001, 27896, 27790, 27683, 27575, 27466, 27355,
- 27244, 27132, 27019, 26905, 26789, 26673, 26556, 26437,
- 26318, 26198, 26077, 25954, 25831, 25707, 25582, 25456,
- 25329, 25201, 25072, 24942, 24811, 24679, 24546, 24413,
- 24278, 24143, 24006, 23869, 23731, 23592, 23452, 23311,
- 23169, 23027, 22883, 22739, 22594, 22448, 22301, 22153,
- 22004, 21855, 21705, 21554, 21402, 21249, 21096, 20942,
- 20787, 20631, 20474, 20317, 20159, 20000, 19840, 19680,
- 19519, 19357, 19194, 19031, 18867, 18702, 18537, 18371,
- 18204, 18036, 17868, 17699, 17530, 17360, 17189, 17017,
- 16845, 16672, 16499, 16325, 16150, 15975, 15799, 15623,
- 15446, 15268, 15090, 14911, 14732, 14552, 14372, 14191,
- 14009, 13827, 13645, 13462, 13278, 13094, 12909, 12724,
- 12539, 12353, 12166, 11980, 11792, 11604, 11416, 11227,
- 11038, 10849, 10659, 10469, 10278, 10087, 9895, 9703,
- 9511, 9319, 9126, 8932, 8739, 8545, 8351, 8156,
- 7961, 7766, 7571, 7375, 7179, 6982, 6786, 6589,
- 6392, 6195, 5997, 5799, 5601, 5403, 5205, 5006,
- 4807, 4608, 4409, 4210, 4011, 3811, 3611, 3411,
- 3211, 3011, 2811, 2610, 2410, 2209, 2009, 1808,
- 1607, 1406, 1206, 1005, 804, 603, 402, 201,
- 0, -201, -402, -603, -804, -1005, -1206, -1406,
- -1607, -1808, -2009, -2209, -2410, -2610, -2811, -3011,
- -3211, -3411, -3611, -3811, -4011, -4210, -4409, -4608,
- -4807, -5006, -5205, -5403, -5601, -5799, -5997, -6195,
- -6392, -6589, -6786, -6982, -7179, -7375, -7571, -7766,
- -7961, -8156, -8351, -8545, -8739, -8932, -9126, -9319,
- -9511, -9703, -9895, -10087, -10278, -10469, -10659, -10849,
- -11038, -11227, -11416, -11604, -11792, -11980, -12166, -12353,
- -12539, -12724, -12909, -13094, -13278, -13462, -13645, -13827,
- -14009, -14191, -14372, -14552, -14732, -14911, -15090, -15268,
- -15446, -15623, -15799, -15975, -16150, -16325, -16499, -16672,
- -16845, -17017, -17189, -17360, -17530, -17699, -17868, -18036,
- -18204, -18371, -18537, -18702, -18867, -19031, -19194, -19357,
- -19519, -19680, -19840, -20000, -20159, -20317, -20474, -20631,
- -20787, -20942, -21096, -21249, -21402, -21554, -21705, -21855,
- -22004, -22153, -22301, -22448, -22594, -22739, -22883, -23027,
- -23169, -23311, -23452, -23592, -23731, -23869, -24006, -24143,
- -24278, -24413, -24546, -24679, -24811, -24942, -25072, -25201,
- -25329, -25456, -25582, -25707, -25831, -25954, -26077, -26198,
- -26318, -26437, -26556, -26673, -26789, -26905, -27019, -27132,
- -27244, -27355, -27466, -27575, -27683, -27790, -27896, -28001,
- -28105, -28208, -28309, -28410, -28510, -28608, -28706, -28802,
- -28897, -28992, -29085, -29177, -29268, -29358, -29446, -29534,
- -29621, -29706, -29790, -29873, -29955, -30036, -30116, -30195,
- -30272, -30349, -30424, -30498, -30571, -30643, -30713, -30783,
- -30851, -30918, -30984, -31049, -31113, -31175, -31236, -31297,
- -31356, -31413, -31470, -31525, -31580, -31633, -31684, -31735,
- -31785, -31833, -31880, -31926, -31970, -32014, -32056, -32097,
- -32137, -32176, -32213, -32249, -32284, -32318, -32350, -32382,
- -32412, -32441, -32468, -32495, -32520, -32544, -32567, -32588,
- -32609, -32628, -32646, -32662, -32678, -32692, -32705, -32717,
- -32727, -32736, -32744, -32751, -32757, -32761, -32764, -32766,
- -32767, -32766, -32764, -32761, -32757, -32751, -32744, -32736,
- -32727, -32717, -32705, -32692, -32678, -32662, -32646, -32628,
- -32609, -32588, -32567, -32544, -32520, -32495, -32468, -32441,
- -32412, -32382, -32350, -32318, -32284, -32249, -32213, -32176,
- -32137, -32097, -32056, -32014, -31970, -31926, -31880, -31833,
- -31785, -31735, -31684, -31633, -31580, -31525, -31470, -31413,
- -31356, -31297, -31236, -31175, -31113, -31049, -30984, -30918,
- -30851, -30783, -30713, -30643, -30571, -30498, -30424, -30349,
- -30272, -30195, -30116, -30036, -29955, -29873, -29790, -29706,
- -29621, -29534, -29446, -29358, -29268, -29177, -29085, -28992,
- -28897, -28802, -28706, -28608, -28510, -28410, -28309, -28208,
- -28105, -28001, -27896, -27790, -27683, -27575, -27466, -27355,
- -27244, -27132, -27019, -26905, -26789, -26673, -26556, -26437,
- -26318, -26198, -26077, -25954, -25831, -25707, -25582, -25456,
- -25329, -25201, -25072, -24942, -24811, -24679, -24546, -24413,
- -24278, -24143, -24006, -23869, -23731, -23592, -23452, -23311,
- -23169, -23027, -22883, -22739, -22594, -22448, -22301, -22153,
- -22004, -21855, -21705, -21554, -21402, -21249, -21096, -20942,
- -20787, -20631, -20474, -20317, -20159, -20000, -19840, -19680,
- -19519, -19357, -19194, -19031, -18867, -18702, -18537, -18371,
- -18204, -18036, -17868, -17699, -17530, -17360, -17189, -17017,
- -16845, -16672, -16499, -16325, -16150, -15975, -15799, -15623,
- -15446, -15268, -15090, -14911, -14732, -14552, -14372, -14191,
- -14009, -13827, -13645, -13462, -13278, -13094, -12909, -12724,
- -12539, -12353, -12166, -11980, -11792, -11604, -11416, -11227,
- -11038, -10849, -10659, -10469, -10278, -10087, -9895, -9703,
- -9511, -9319, -9126, -8932, -8739, -8545, -8351, -8156,
- -7961, -7766, -7571, -7375, -7179, -6982, -6786, -6589,
- -6392, -6195, -5997, -5799, -5601, -5403, -5205, -5006,
- -4807, -4608, -4409, -4210, -4011, -3811, -3611, -3411,
- -3211, -3011, -2811, -2610, -2410, -2209, -2009, -1808,
- -1607, -1406, -1206, -1005, -804, -603, -402, -201
-};
-
-int WebRtcSpl_ComplexFFT(WebRtc_Word16 frfi[], int stages, int mode)
-{
- int i, j, l, k, istep, n, m;
- WebRtc_Word16 wr, wi;
- WebRtc_Word32 tr32, ti32, qr32, qi32;
-
- /* The 1024-value is a constant given from the size of kSinTable1024[],
- * and should not be changed depending on the input parameter 'stages'
- */
- n = 1 << stages;
- if (n > 1024)
- return -1;
-
- l = 1;
- k = 10 - 1; /* Constant for given kSinTable1024[]. Do not change
- depending on the input parameter 'stages' */
-
- if (mode == 0)
- {
- // mode==0: Low-complexity and Low-accuracy mode
- while (l < n)
- {
- istep = l << 1;
-
- for (m = 0; m < l; ++m)
- {
- j = m << k;
-
- /* The 256-value is a constant given as 1/4 of the size of
- * kSinTable1024[], and should not be changed depending on the input
- * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
- */
- wr = kSinTable1024[j + 256];
- wi = -kSinTable1024[j];
-
- for (i = m; i < n; i += istep)
- {
- j = i + l;
-
- tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
- - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1])), 15);
-
- ti32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
- + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j])), 15);
-
- qr32 = (WebRtc_Word32)frfi[2 * i];
- qi32 = (WebRtc_Word32)frfi[2 * i + 1];
- frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 - tr32, 1);
- frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 - ti32, 1);
- frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 + tr32, 1);
- frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 + ti32, 1);
- }
- }
-
- --k;
- l = istep;
-
- }
-
- } else
- {
- // mode==1: High-complexity and High-accuracy mode
- while (l < n)
- {
- istep = l << 1;
-
- for (m = 0; m < l; ++m)
- {
- j = m << k;
-
- /* The 256-value is a constant given as 1/4 of the size of
- * kSinTable1024[], and should not be changed depending on the input
- * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
- */
- wr = kSinTable1024[j + 256];
- wi = -kSinTable1024[j];
-
-#ifdef WEBRTC_ARCH_ARM_V7A
- WebRtc_Word32 wri;
- WebRtc_Word32 frfi_r;
- __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(wri) :
- "r"((WebRtc_Word32)wr), "r"((WebRtc_Word32)wi));
-#endif
-
- for (i = m; i < n; i += istep)
- {
- j = i + l;
-
-#ifdef WEBRTC_ARCH_ARM_V7A
- __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(frfi_r) :
- "r"((WebRtc_Word32)frfi[2*j]), "r"((WebRtc_Word32)frfi[2*j +1]));
- __asm__("smlsd %0, %1, %2, %3" : "=r"(tr32) :
- "r"(wri), "r"(frfi_r), "r"(CFFTRND));
- __asm__("smladx %0, %1, %2, %3" : "=r"(ti32) :
- "r"(wri), "r"(frfi_r), "r"(CFFTRND));
-
-#else
- tr32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
- - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CFFTRND;
-
- ti32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
- + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CFFTRND;
-#endif
-
- tr32 = WEBRTC_SPL_RSHIFT_W32(tr32, 15 - CFFTSFT);
- ti32 = WEBRTC_SPL_RSHIFT_W32(ti32, 15 - CFFTSFT);
-
- qr32 = ((WebRtc_Word32)frfi[2 * i]) << CFFTSFT;
- qi32 = ((WebRtc_Word32)frfi[2 * i + 1]) << CFFTSFT;
-
- frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
- (qr32 - tr32 + CFFTRND2), 1 + CFFTSFT);
- frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
- (qi32 - ti32 + CFFTRND2), 1 + CFFTSFT);
- frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
- (qr32 + tr32 + CFFTRND2), 1 + CFFTSFT);
- frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
- (qi32 + ti32 + CFFTRND2), 1 + CFFTSFT);
- }
- }
-
- --k;
- l = istep;
- }
- }
- return 0;
-}
-
-int WebRtcSpl_ComplexIFFT(WebRtc_Word16 frfi[], int stages, int mode)
-{
- int i, j, l, k, istep, n, m, scale, shift;
- WebRtc_Word16 wr, wi;
- WebRtc_Word32 tr32, ti32, qr32, qi32;
- WebRtc_Word32 tmp32, round2;
-
- /* The 1024-value is a constant given from the size of kSinTable1024[],
- * and should not be changed depending on the input parameter 'stages'
- */
- n = 1 << stages;
- if (n > 1024)
- return -1;
-
- scale = 0;
-
- l = 1;
- k = 10 - 1; /* Constant for given kSinTable1024[]. Do not change
- depending on the input parameter 'stages' */
-
- while (l < n)
- {
- // variable scaling, depending upon data
- shift = 0;
- round2 = 8192;
-
- tmp32 = (WebRtc_Word32)WebRtcSpl_MaxAbsValueW16(frfi, 2 * n);
- if (tmp32 > 13573)
- {
- shift++;
- scale++;
- round2 <<= 1;
- }
- if (tmp32 > 27146)
- {
- shift++;
- scale++;
- round2 <<= 1;
- }
-
- istep = l << 1;
-
- if (mode == 0)
- {
- // mode==0: Low-complexity and Low-accuracy mode
- for (m = 0; m < l; ++m)
- {
- j = m << k;
-
- /* The 256-value is a constant given as 1/4 of the size of
- * kSinTable1024[], and should not be changed depending on the input
- * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
- */
- wr = kSinTable1024[j + 256];
- wi = kSinTable1024[j];
-
- for (i = m; i < n; i += istep)
- {
- j = i + l;
-
- tr32 = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j], 0)
- - WEBRTC_SPL_MUL_16_16_RSFT(wi, frfi[2 * j + 1], 0)), 15);
-
- ti32 = WEBRTC_SPL_RSHIFT_W32(
- (WEBRTC_SPL_MUL_16_16_RSFT(wr, frfi[2 * j + 1], 0)
- + WEBRTC_SPL_MUL_16_16_RSFT(wi,frfi[2*j],0)), 15);
-
- qr32 = (WebRtc_Word32)frfi[2 * i];
- qi32 = (WebRtc_Word32)frfi[2 * i + 1];
- frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 - tr32, shift);
- frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 - ti32, shift);
- frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qr32 + tr32, shift);
- frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(qi32 + ti32, shift);
- }
- }
- } else
- {
- // mode==1: High-complexity and High-accuracy mode
-
- for (m = 0; m < l; ++m)
- {
- j = m << k;
-
- /* The 256-value is a constant given as 1/4 of the size of
- * kSinTable1024[], and should not be changed depending on the input
- * parameter 'stages'. It will result in 0 <= j < N_SINE_WAVE/2
- */
- wr = kSinTable1024[j + 256];
- wi = kSinTable1024[j];
-
-#ifdef WEBRTC_ARCH_ARM_V7A
- WebRtc_Word32 wri;
- WebRtc_Word32 frfi_r;
- __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(wri) :
- "r"((WebRtc_Word32)wr), "r"((WebRtc_Word32)wi));
-#endif
-
- for (i = m; i < n; i += istep)
- {
- j = i + l;
-
-#ifdef WEBRTC_ARCH_ARM_V7A
- __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(frfi_r) :
- "r"((WebRtc_Word32)frfi[2*j]), "r"((WebRtc_Word32)frfi[2*j +1]));
- __asm__("smlsd %0, %1, %2, %3" : "=r"(tr32) :
- "r"(wri), "r"(frfi_r), "r"(CIFFTRND));
- __asm__("smladx %0, %1, %2, %3" : "=r"(ti32) :
- "r"(wri), "r"(frfi_r), "r"(CIFFTRND));
-#else
-
- tr32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j])
- - WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j + 1]) + CIFFTRND;
-
- ti32 = WEBRTC_SPL_MUL_16_16(wr, frfi[2 * j + 1])
- + WEBRTC_SPL_MUL_16_16(wi, frfi[2 * j]) + CIFFTRND;
-#endif
- tr32 = WEBRTC_SPL_RSHIFT_W32(tr32, 15 - CIFFTSFT);
- ti32 = WEBRTC_SPL_RSHIFT_W32(ti32, 15 - CIFFTSFT);
-
- qr32 = ((WebRtc_Word32)frfi[2 * i]) << CIFFTSFT;
- qi32 = ((WebRtc_Word32)frfi[2 * i + 1]) << CIFFTSFT;
-
- frfi[2 * j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((qr32 - tr32+round2),
- shift+CIFFTSFT);
- frfi[2 * j + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
- (qi32 - ti32 + round2), shift + CIFFTSFT);
- frfi[2 * i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((qr32 + tr32 + round2),
- shift + CIFFTSFT);
- frfi[2 * i + 1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
- (qi32 + ti32 + round2), shift + CIFFTSFT);
- }
- }
-
- }
- --k;
- l = istep;
- }
- return scale;
-}
diff --git a/trunk/src/common_audio/signal_processing/copy_set_operations.c b/trunk/src/common_audio/signal_processing/copy_set_operations.c
deleted file mode 100644
index 8247337..0000000
--- a/trunk/src/common_audio/signal_processing/copy_set_operations.c
+++ /dev/null
@@ -1,108 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the implementation of functions
- * WebRtcSpl_MemSetW16()
- * WebRtcSpl_MemSetW32()
- * WebRtcSpl_MemCpyReversedOrder()
- * WebRtcSpl_CopyFromEndW16()
- * WebRtcSpl_ZerosArrayW16()
- * WebRtcSpl_ZerosArrayW32()
- * WebRtcSpl_OnesArrayW16()
- * WebRtcSpl_OnesArrayW32()
- *
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include <string.h>
-#include "signal_processing_library.h"
-
-
-void WebRtcSpl_MemSetW16(WebRtc_Word16 *ptr, WebRtc_Word16 set_value, int length)
-{
- int j;
- WebRtc_Word16 *arrptr = ptr;
-
- for (j = length; j > 0; j--)
- {
- *arrptr++ = set_value;
- }
-}
-
-void WebRtcSpl_MemSetW32(WebRtc_Word32 *ptr, WebRtc_Word32 set_value, int length)
-{
- int j;
- WebRtc_Word32 *arrptr = ptr;
-
- for (j = length; j > 0; j--)
- {
- *arrptr++ = set_value;
- }
-}
-
-void WebRtcSpl_MemCpyReversedOrder(WebRtc_Word16* dest, WebRtc_Word16* source, int length)
-{
- int j;
- WebRtc_Word16* destPtr = dest;
- WebRtc_Word16* sourcePtr = source;
-
- for (j = 0; j < length; j++)
- {
- *destPtr-- = *sourcePtr++;
- }
-}
-
-WebRtc_Word16 WebRtcSpl_CopyFromEndW16(G_CONST WebRtc_Word16 *vector_in,
- WebRtc_Word16 length,
- WebRtc_Word16 samples,
- WebRtc_Word16 *vector_out)
-{
- // Copy the last <samples> of the input vector to vector_out
- WEBRTC_SPL_MEMCPY_W16(vector_out, &vector_in[length - samples], samples);
-
- return samples;
-}
-
-WebRtc_Word16 WebRtcSpl_ZerosArrayW16(WebRtc_Word16 *vector, WebRtc_Word16 length)
-{
- WebRtcSpl_MemSetW16(vector, 0, length);
- return length;
-}
-
-WebRtc_Word16 WebRtcSpl_ZerosArrayW32(WebRtc_Word32 *vector, WebRtc_Word16 length)
-{
- WebRtcSpl_MemSetW32(vector, 0, length);
- return length;
-}
-
-WebRtc_Word16 WebRtcSpl_OnesArrayW16(WebRtc_Word16 *vector, WebRtc_Word16 length)
-{
- WebRtc_Word16 i;
- WebRtc_Word16 *tmpvec = vector;
- for (i = 0; i < length; i++)
- {
- *tmpvec++ = 1;
- }
- return length;
-}
-
-WebRtc_Word16 WebRtcSpl_OnesArrayW32(WebRtc_Word32 *vector, WebRtc_Word16 length)
-{
- WebRtc_Word16 i;
- WebRtc_Word32 *tmpvec = vector;
- for (i = 0; i < length; i++)
- {
- *tmpvec++ = 1;
- }
- return length;
-}
diff --git a/trunk/src/common_audio/signal_processing/cross_correlation.c b/trunk/src/common_audio/signal_processing/cross_correlation.c
deleted file mode 100644
index 726a749..0000000
--- a/trunk/src/common_audio/signal_processing/cross_correlation.c
+++ /dev/null
@@ -1,271 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_CrossCorrelation().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-/* TODO(kma): Clean up the code in this file, and break it up for
- * various platforms (Xscale, ARM/Neon etc.).
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_CrossCorrelation(WebRtc_Word32* cross_correlation, WebRtc_Word16* seq1,
- WebRtc_Word16* seq2, WebRtc_Word16 dim_seq,
- WebRtc_Word16 dim_cross_correlation,
- WebRtc_Word16 right_shifts,
- WebRtc_Word16 step_seq2)
-{
- int i, j;
- WebRtc_Word16* seq1Ptr;
- WebRtc_Word16* seq2Ptr;
- WebRtc_Word32* CrossCorrPtr;
-
-#ifdef _XSCALE_OPT_
-
-#ifdef _WIN32
-#pragma message("NOTE: _XSCALE_OPT_ optimizations are used (overrides _ARM_OPT_ and requires /QRxscale compiler flag)")
-#endif
-
- __int64 macc40;
-
- int iseq1[250];
- int iseq2[250];
- int iseq3[250];
- int * iseq1Ptr;
- int * iseq2Ptr;
- int * iseq3Ptr;
- int len, i_len;
-
- seq1Ptr = seq1;
- iseq1Ptr = iseq1;
- for(i = 0; i < ((dim_seq + 1) >> 1); i++)
- {
- *iseq1Ptr = (unsigned short)*seq1Ptr++;
- *iseq1Ptr++ |= (WebRtc_Word32)*seq1Ptr++ << 16;
-
- }
-
- if(dim_seq%2)
- {
- *(iseq1Ptr-1) &= 0x0000ffff;
- }
- *iseq1Ptr = 0;
- iseq1Ptr++;
- *iseq1Ptr = 0;
- iseq1Ptr++;
- *iseq1Ptr = 0;
-
- if(step_seq2 < 0)
- {
- seq2Ptr = seq2 - dim_cross_correlation + 1;
- CrossCorrPtr = &cross_correlation[dim_cross_correlation - 1];
- }
- else
- {
- seq2Ptr = seq2;
- CrossCorrPtr = cross_correlation;
- }
-
- len = dim_seq + dim_cross_correlation - 1;
- i_len = (len + 1) >> 1;
- iseq2Ptr = iseq2;
-
- iseq3Ptr = iseq3;
- for(i = 0; i < i_len; i++)
- {
- *iseq2Ptr = (unsigned short)*seq2Ptr++;
- *iseq3Ptr = (unsigned short)*seq2Ptr;
- *iseq2Ptr++ |= (WebRtc_Word32)*seq2Ptr++ << 16;
- *iseq3Ptr++ |= (WebRtc_Word32)*seq2Ptr << 16;
- }
-
- if(len % 2)
- {
- iseq2[i_len - 1] &= 0x0000ffff;
- iseq3[i_len - 1] = 0;
- }
- else
- iseq3[i_len - 1] &= 0x0000ffff;
-
- iseq2[i_len] = 0;
- iseq3[i_len] = 0;
- iseq2[i_len + 1] = 0;
- iseq3[i_len + 1] = 0;
- iseq2[i_len + 2] = 0;
- iseq3[i_len + 2] = 0;
-
- // Set pointer to start value
- iseq2Ptr = iseq2;
- iseq3Ptr = iseq3;
-
- i_len = (dim_seq + 7) >> 3;
- for (i = 0; i < dim_cross_correlation; i++)
- {
-
- iseq1Ptr = iseq1;
-
- macc40 = 0;
-
- _WriteCoProcessor(macc40, 0);
-
- if((i & 1))
- {
- iseq3Ptr = iseq3 + (i >> 1);
- for (j = i_len; j > 0; j--)
- {
- _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
- _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
- _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
- _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq3Ptr++);
- }
- }
- else
- {
- iseq2Ptr = iseq2 + (i >> 1);
- for (j = i_len; j > 0; j--)
- {
- _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
- _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
- _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
- _SmulAddPack_2SW_ACC(*iseq1Ptr++, *iseq2Ptr++);
- }
-
- }
-
- macc40 = _ReadCoProcessor(0);
- *CrossCorrPtr = (WebRtc_Word32)(macc40 >> right_shifts);
- CrossCorrPtr += step_seq2;
- }
-#else // #ifdef _XSCALE_OPT_
-#ifdef _ARM_OPT_
- WebRtc_Word16 dim_seq8 = (dim_seq >> 3) << 3;
-#endif
-
- CrossCorrPtr = cross_correlation;
-
- for (i = 0; i < dim_cross_correlation; i++)
- {
- // Set the pointer to the static vector, set the pointer to the sliding vector
- // and initialize cross_correlation
- seq1Ptr = seq1;
- seq2Ptr = seq2 + (step_seq2 * i);
- (*CrossCorrPtr) = 0;
-
-#ifndef _ARM_OPT_
-#ifdef _WIN32
-#pragma message("NOTE: default implementation is used")
-#endif
- // Perform the cross correlation
- for (j = 0; j < dim_seq; j++)
- {
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr), right_shifts);
- seq1Ptr++;
- seq2Ptr++;
- }
-#else
-#ifdef _WIN32
-#pragma message("NOTE: _ARM_OPT_ optimizations are used")
-#endif
- if (right_shifts == 0)
- {
- // Perform the optimized cross correlation
- for (j = 0; j < dim_seq8; j = j + 8)
- {
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
- seq1Ptr++;
- seq2Ptr++;
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
- seq1Ptr++;
- seq2Ptr++;
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
- seq1Ptr++;
- seq2Ptr++;
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
- seq1Ptr++;
- seq2Ptr++;
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
- seq1Ptr++;
- seq2Ptr++;
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
- seq1Ptr++;
- seq2Ptr++;
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
- seq1Ptr++;
- seq2Ptr++;
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
- seq1Ptr++;
- seq2Ptr++;
- }
-
- for (j = dim_seq8; j < dim_seq; j++)
- {
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
- seq1Ptr++;
- seq2Ptr++;
- }
- }
- else // right_shifts != 0
-
- {
- // Perform the optimized cross correlation
- for (j = 0; j < dim_seq8; j = j + 8)
- {
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
- right_shifts);
- seq1Ptr++;
- seq2Ptr++;
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
- right_shifts);
- seq1Ptr++;
- seq2Ptr++;
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
- right_shifts);
- seq1Ptr++;
- seq2Ptr++;
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
- right_shifts);
- seq1Ptr++;
- seq2Ptr++;
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
- right_shifts);
- seq1Ptr++;
- seq2Ptr++;
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
- right_shifts);
- seq1Ptr++;
- seq2Ptr++;
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
- right_shifts);
- seq1Ptr++;
- seq2Ptr++;
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
- right_shifts);
- seq1Ptr++;
- seq2Ptr++;
- }
-
- for (j = dim_seq8; j < dim_seq; j++)
- {
- (*CrossCorrPtr) += WEBRTC_SPL_MUL_16_16_RSFT((*seq1Ptr), (*seq2Ptr),
- right_shifts);
- seq1Ptr++;
- seq2Ptr++;
- }
- }
-#endif
- CrossCorrPtr++;
- }
-#endif
-}
diff --git a/trunk/src/common_audio/signal_processing/cross_correlation_neon.s b/trunk/src/common_audio/signal_processing/cross_correlation_neon.s
deleted file mode 100644
index e9b1c69..0000000
--- a/trunk/src/common_audio/signal_processing/cross_correlation_neon.s
+++ /dev/null
@@ -1,168 +0,0 @@
-@
-@ Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-@
-@ Use of this source code is governed by a BSD-style license
-@ that can be found in the LICENSE file in the root of the source
-@ tree. An additional intellectual property rights grant can be found
-@ in the file PATENTS. All contributing project authors may
-@ be found in the AUTHORS file in the root of the source tree.
-@
-
-@ cross_correlation_neon.s
-@ This file contains the function WebRtcSpl_CrossCorrelation(),
-@ optimized for ARM Neon platform.
-@
-@ Reference Ccode at end of this file.
-@ Output is bit-exact with the reference C code, but not with the generic
-@ C code in file cross_correlation.c, due to reduction of shift operations
-@ from using Neon registers.
-
-@ Register usage:
-@
-@ r0: *cross_correlation (function argument)
-@ r1: *seq1 (function argument)
-@ r2: *seq2 (function argument)
-@ r3: dim_seq (function argument); then, total iteration of LOOP_DIM_SEQ
-@ r4: counter for LOOP_DIM_CROSS_CORRELATION
-@ r5: seq2_ptr
-@ r6: seq1_ptr
-@ r7: Total iteration of LOOP_DIM_SEQ_RESIDUAL
-@ r8, r9, r10, r11, r12: scratch
-
-.arch armv7-a
-.fpu neon
-
-.align 2
-.global WebRtcSpl_CrossCorrelation
-
-WebRtcSpl_CrossCorrelation:
-
-.fnstart
-
-.save {r4-r11}
- push {r4-r11}
-
- @ Put the shift value (-right_shifts) into a Neon register.
- ldrsh r10, [sp, #36]
- rsb r10, r10, #0
- mov r8, r10, asr #31
- vmov.32 d16, r10, r8
-
- @ Initialize loop counters.
- and r7, r3, #7 @ inner_loop_len2 = dim_seq % 8;
- asr r3, r3, #3 @ inner_loop_len1 = dim_seq / 8;
- ldrsh r4, [sp, #32] @ dim_cross_correlation
-
-LOOP_DIM_CROSS_CORRELATION:
- vmov.i32 q9, #0
- vmov.i32 q14, #0
- movs r8, r3 @ inner_loop_len1
- mov r6, r1 @ seq1_ptr
- mov r5, r2 @ seq2_ptr
- ble POST_LOOP_DIM_SEQ
-
-LOOP_DIM_SEQ:
- vld1.16 {d20, d21}, [r6]! @ seq1_ptr
- vld1.16 {d22, d23}, [r5]! @ seq2_ptr
- subs r8, r8, #1
- vmull.s16 q12, d20, d22
- vmull.s16 q13, d21, d23
- vpadal.s32 q9, q12
- vpadal.s32 q14, q13
- bgt LOOP_DIM_SEQ
-
-POST_LOOP_DIM_SEQ:
- movs r10, r7 @ Loop counter
- mov r12, #0
- mov r8, #0
- ble POST_LOOP_DIM_SEQ_RESIDUAL
-
-LOOP_DIM_SEQ_RESIDUAL:
- ldrh r11, [r6], #2
- ldrh r9, [r5], #2
- smulbb r11, r11, r9
- adds r8, r8, r11
- adc r12, r12, r11, asr #31
- subs r10, #1
- bgt LOOP_DIM_SEQ_RESIDUAL
-
-POST_LOOP_DIM_SEQ_RESIDUAL: @ Sum the results up and do the shift.
- vadd.i64 d18, d19
- vadd.i64 d28, d29
- vadd.i64 d18, d28
- vmov.32 d17[0], r8
- vmov.32 d17[1], r12
- vadd.i64 d17, d18
- vshl.s64 d17, d16
- vst1.32 d17[0], [r0]! @ Store the output
-
- ldr r8, [sp, #40] @ step_seq2
- add r2, r8, lsl #1 @ prepare for seq2_ptr(r5) in the next loop.
-
- subs r4, #1
- bgt LOOP_DIM_CROSS_CORRELATION
-
- pop {r4-r11}
- bx lr
-
-.fnend
-
-
-@ TODO(kma): Place this piece of reference code into a C code file.
-@ void WebRtcSpl_CrossCorrelation(WebRtc_Word32* cross_correlation,
-@ WebRtc_Word16* seq1,
-@ WebRtc_Word16* seq2,
-@ WebRtc_Word16 dim_seq,
-@ WebRtc_Word16 dim_cross_correlation,
-@ WebRtc_Word16 right_shifts,
-@ WebRtc_Word16 step_seq2) {
-@ int i = 0;
-@ int j = 0;
-@ int inner_loop_len1 = dim_seq >> 3;
-@ int inner_loop_len2 = dim_seq - (inner_loop_len1 << 3);
-@
-@ assert(dim_cross_correlation > 0);
-@ assert(dim_seq > 0);
-@
-@ for (i = 0; i < dim_cross_correlation; i++) {
-@ int16_t *seq1_ptr = seq1;
-@ int16_t *seq2_ptr = seq2 + (step_seq2 * i);
-@ int64_t sum = 0;
-@
-@ for (j = inner_loop_len1; j > 0; j -= 1) {
-@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
-@ seq1_ptr++;
-@ seq2_ptr++;
-@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
-@ seq1_ptr++;
-@ seq2_ptr++;
-@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
-@ seq1_ptr++;
-@ seq2_ptr++;
-@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
-@ seq1_ptr++;
-@ seq2_ptr++;
-@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
-@ seq1_ptr++;
-@ seq2_ptr++;
-@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
-@ seq1_ptr++;
-@ seq2_ptr++;
-@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
-@ seq1_ptr++;
-@ seq2_ptr++;
-@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
-@ seq1_ptr++;
-@ seq2_ptr++;
-@ }
-@
-@ // Calculate the rest of the samples.
-@ for (j = inner_loop_len2; j > 0; j -= 1) {
-@ sum += WEBRTC_SPL_MUL_16_16(*seq1_ptr, *seq2_ptr);
-@ seq1_ptr++;
-@ seq2_ptr++;
-@ }
-@
-@ *cross_correlation++ = (int32_t)(sum >> right_shifts);
-@ }
-@ }
diff --git a/trunk/src/common_audio/signal_processing/division_operations.c b/trunk/src/common_audio/signal_processing/division_operations.c
deleted file mode 100644
index b143373..0000000
--- a/trunk/src/common_audio/signal_processing/division_operations.c
+++ /dev/null
@@ -1,144 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains implementations of the divisions
- * WebRtcSpl_DivU32U16()
- * WebRtcSpl_DivW32W16()
- * WebRtcSpl_DivW32W16ResW16()
- * WebRtcSpl_DivResultInQ31()
- * WebRtcSpl_DivW32HiLow()
- *
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_UWord32 WebRtcSpl_DivU32U16(WebRtc_UWord32 num, WebRtc_UWord16 den)
-{
- // Guard against division with 0
- if (den != 0)
- {
- return (WebRtc_UWord32)(num / den);
- } else
- {
- return (WebRtc_UWord32)0xFFFFFFFF;
- }
-}
-
-WebRtc_Word32 WebRtcSpl_DivW32W16(WebRtc_Word32 num, WebRtc_Word16 den)
-{
- // Guard against division with 0
- if (den != 0)
- {
- return (WebRtc_Word32)(num / den);
- } else
- {
- return (WebRtc_Word32)0x7FFFFFFF;
- }
-}
-
-WebRtc_Word16 WebRtcSpl_DivW32W16ResW16(WebRtc_Word32 num, WebRtc_Word16 den)
-{
- // Guard against division with 0
- if (den != 0)
- {
- return (WebRtc_Word16)(num / den);
- } else
- {
- return (WebRtc_Word16)0x7FFF;
- }
-}
-
-WebRtc_Word32 WebRtcSpl_DivResultInQ31(WebRtc_Word32 num, WebRtc_Word32 den)
-{
- WebRtc_Word32 L_num = num;
- WebRtc_Word32 L_den = den;
- WebRtc_Word32 div = 0;
- int k = 31;
- int change_sign = 0;
-
- if (num == 0)
- return 0;
-
- if (num < 0)
- {
- change_sign++;
- L_num = -num;
- }
- if (den < 0)
- {
- change_sign++;
- L_den = -den;
- }
- while (k--)
- {
- div <<= 1;
- L_num <<= 1;
- if (L_num >= L_den)
- {
- L_num -= L_den;
- div++;
- }
- }
- if (change_sign == 1)
- {
- div = -div;
- }
- return div;
-}
-
-WebRtc_Word32 WebRtcSpl_DivW32HiLow(WebRtc_Word32 num, WebRtc_Word16 den_hi,
- WebRtc_Word16 den_low)
-{
- WebRtc_Word16 approx, tmp_hi, tmp_low, num_hi, num_low;
- WebRtc_Word32 tmpW32;
-
- approx = (WebRtc_Word16)WebRtcSpl_DivW32W16((WebRtc_Word32)0x1FFFFFFF, den_hi);
- // result in Q14 (Note: 3FFFFFFF = 0.5 in Q30)
-
- // tmpW32 = 1/den = approx * (2.0 - den * approx) (in Q30)
- tmpW32 = (WEBRTC_SPL_MUL_16_16(den_hi, approx) << 1)
- + ((WEBRTC_SPL_MUL_16_16(den_low, approx) >> 15) << 1);
- // tmpW32 = den * approx
-
- tmpW32 = (WebRtc_Word32)0x7fffffffL - tmpW32; // result in Q30 (tmpW32 = 2.0-(den*approx))
-
- // Store tmpW32 in hi and low format
- tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 16);
- tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((tmpW32
- - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
-
- // tmpW32 = 1/den in Q29
- tmpW32 = ((WEBRTC_SPL_MUL_16_16(tmp_hi, approx) + (WEBRTC_SPL_MUL_16_16(tmp_low, approx)
- >> 15)) << 1);
-
- // 1/den in hi and low format
- tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32, 16);
- tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((tmpW32
- - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
-
- // Store num in hi and low format
- num_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(num, 16);
- num_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((num
- - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)num_hi, 16)), 1);
-
- // num * (1/den) by 32 bit multiplication (result in Q28)
-
- tmpW32 = (WEBRTC_SPL_MUL_16_16(num_hi, tmp_hi) + (WEBRTC_SPL_MUL_16_16(num_hi, tmp_low)
- >> 15) + (WEBRTC_SPL_MUL_16_16(num_low, tmp_hi) >> 15));
-
- // Put result in Q31 (convert from Q28)
- tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);
-
- return tmpW32;
-}
diff --git a/trunk/src/common_audio/signal_processing/dot_product_with_scale.c b/trunk/src/common_audio/signal_processing/dot_product_with_scale.c
deleted file mode 100644
index 6e085fd..0000000
--- a/trunk/src/common_audio/signal_processing/dot_product_with_scale.c
+++ /dev/null
@@ -1,91 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_DotProductWithScale().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_Word32 WebRtcSpl_DotProductWithScale(WebRtc_Word16 *vector1, WebRtc_Word16 *vector2,
- int length, int scaling)
-{
- WebRtc_Word32 sum;
- int i;
-#ifdef _ARM_OPT_
-#pragma message("NOTE: _ARM_OPT_ optimizations are used")
- WebRtc_Word16 len4 = (length >> 2) << 2;
-#endif
-
- sum = 0;
-
-#ifndef _ARM_OPT_
- for (i = 0; i < length; i++)
- {
- sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1++, *vector2++, scaling);
- }
-#else
- if (scaling == 0)
- {
- for (i = 0; i < len4; i = i + 4)
- {
- sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
- vector1++;
- vector2++;
- sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
- vector1++;
- vector2++;
- sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
- vector1++;
- vector2++;
- sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
- vector1++;
- vector2++;
- }
-
- for (i = len4; i < length; i++)
- {
- sum += WEBRTC_SPL_MUL_16_16(*vector1, *vector2);
- vector1++;
- vector2++;
- }
- }
- else
- {
- for (i = 0; i < len4; i = i + 4)
- {
- sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
- vector1++;
- vector2++;
- sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
- vector1++;
- vector2++;
- sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
- vector1++;
- vector2++;
- sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
- vector1++;
- vector2++;
- }
-
- for (i = len4; i < length; i++)
- {
- sum += WEBRTC_SPL_MUL_16_16_RSFT(*vector1, *vector2, scaling);
- vector1++;
- vector2++;
- }
- }
-#endif
-
- return sum;
-}
diff --git a/trunk/src/common_audio/signal_processing/downsample_fast.c b/trunk/src/common_audio/signal_processing/downsample_fast.c
deleted file mode 100644
index 526cdca..0000000
--- a/trunk/src/common_audio/signal_processing/downsample_fast.c
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "signal_processing_library.h"
-
-// TODO(Bjornv): Change the function parameter order to WebRTC code style.
-int WebRtcSpl_DownsampleFast(const int16_t* data_in,
- int data_in_length,
- int16_t* data_out,
- int data_out_length,
- const int16_t* __restrict coefficients,
- int coefficients_length,
- int factor,
- int delay) {
- int i = 0;
- int j = 0;
- int32_t out_s32 = 0;
- int endpos = delay + factor * (data_out_length - 1) + 1;
-
- // Return error if any of the running conditions doesn't meet.
- if (data_out_length <= 0 || coefficients_length <= 0
- || data_in_length < endpos) {
- return -1;
- }
-
- for (i = delay; i < endpos; i += factor) {
- out_s32 = 2048; // Round value, 0.5 in Q12.
-
- for (j = 0; j < coefficients_length; j++) {
- out_s32 += coefficients[j] * data_in[i - j]; // Q12.
- }
-
- out_s32 >>= 12; // Q0.
-
- // Saturate and store the output.
- *data_out++ = WebRtcSpl_SatW32ToW16(out_s32);
- }
-
- return 0;
-}
diff --git a/trunk/src/common_audio/signal_processing/downsample_fast_neon.s b/trunk/src/common_audio/signal_processing/downsample_fast_neon.s
deleted file mode 100644
index 906b0a1..0000000
--- a/trunk/src/common_audio/signal_processing/downsample_fast_neon.s
+++ /dev/null
@@ -1,222 +0,0 @@
-@
-@ Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-@
-@ Use of this source code is governed by a BSD-style license
-@ that can be found in the LICENSE file in the root of the source
-@ tree. An additional intellectual property rights grant can be found
-@ in the file PATENTS. All contributing project authors may
-@ be found in the AUTHORS file in the root of the source tree.
-@
-
-@ This file contains the function WebRtcSpl_DownsampleFast(), optimized for
-@ ARM Neon platform. The description header can be found in
-@ signal_processing_library.h
-@
-@ The reference C code is in file downsample_fast.c. Bit-exact.
-
-.arch armv7-a
-.fpu neon
-
-.align 2
-.global WebRtcSpl_DownsampleFast
-
-WebRtcSpl_DownsampleFast:
-
-.fnstart
-
-.save {r4-r11}
- push {r4-r11}
-
- cmp r3, #0 @ data_out_length <= 0?
- movle r0, #-1
- ble END
-
- ldrsh r12, [sp, #44]
- ldr r5, [sp, #40] @ r5: factor
- add r4, r12, #1 @ r4: delay + 1
- sub r3, r3, #1 @ r3: data_out_length - 1
- smulbb r3, r5, r3
- ldr r8, [sp, #32] @ &coefficients[0]
- mov r9, r12 @ Iteration counter for outer loops.
- add r3, r4 @ delay + factor * (out_length-1) +1
-
- cmp r3, r1 @ data_in_length < endpos?
- movgt r0, #-1
- bgt END
-
- @ Initializations.
- sub r3, r5, asl #3
- add r11, r0, r12, asl #1 @ &data_in[delay]
- ldr r0, [sp, #36] @ coefficients_length
- add r3, r5 @ endpos - factor * 7
-
- cmp r0, #0 @ coefficients_length <= 0 ?
- movle r0, #-1
- ble END
-
- add r8, r0, asl #1 @ &coeffieient[coefficients_length]
- cmp r9, r3
- bge POST_LOOP_ENDPOS @ branch when Iteration < 8 times.
-
-@
-@ First part, unroll the loop 8 times, with 3 subcases (factor == 2, 4, others)
-@
- mov r4, #-2
-
- @ Direct program flow to the right channel.
-
- @ r10 is an offset to &data_in[] in the loop. After an iteration, we need to
- @ move the pointer back to original after advancing 16 bytes by a vld1, and
- @ then move 2 bytes forward to increment one more sample.
- cmp r5, #2
- moveq r10, #-14
- beq LOOP_ENDPOS_FACTOR2 @ Branch when factor == 2
-
- @ Similar here, for r10, we need to move the pointer back to original after
- @ advancing 32 bytes, then move 2 bytes forward to increment one sample.
- cmp r5, #4
- moveq r10, #-30
- beq LOOP_ENDPOS_FACTOR4 @ Branch when factor == 4
-
- @ For r10, we need to move the pointer back to original after advancing
- @ (factor * 7 * 2) bytes, then move 2 bytes forward to increment one sample.
- mov r10, r5, asl #4
- rsb r10, #2
- add r10, r5, asl #1
- lsl r5, #1 @ r5 = factor * sizeof(data_in)
-
-@ The general case (factor != 2 && factor != 4)
-LOOP_ENDPOS_GENERAL:
- @ Initializations.
- vmov.i32 q2, #2048
- vmov.i32 q3, #2048
- sub r7, r8, #2
- sub r12, r0, #1 @ coefficients_length - 1
- sub r1, r11, r12, asl #1 @ &data_in[i - j]
-
-LOOP_COEFF_LENGTH_GENERAL:
- vld1.16 {d2[], d3[]}, [r7], r4 @ coefficients[j]
- vld1.16 d0[0], [r1], r5 @ data_in[i - j]
- vld1.16 d0[1], [r1], r5 @ data_in[i + factor - j]
- vld1.16 d0[2], [r1], r5 @ data_in[i + factor * 2 - j]
- vld1.16 d0[3], [r1], r5 @ data_in[i + factor * 3 - j]
- vld1.16 d1[0], [r1], r5 @ data_in[i + factor * 4 - j]
- vld1.16 d1[1], [r1], r5 @ data_in[i + factor * 5 - j]
- vld1.16 d1[2], [r1], r5 @ data_in[i + factor * 6 - j]
- vld1.16 d1[3], [r1], r10 @ data_in[i + factor * 7 - j]
- subs r12, #1
- vmlal.s16 q2, d0, d2
- vmlal.s16 q3, d1, d3
- bge LOOP_COEFF_LENGTH_GENERAL
-
- @ Shift, saturate, and store the result.
- vqshrn.s32 d0, q2, #12
- vqshrn.s32 d1, q3, #12
- vst1.16 {d0, d1}, [r2]!
-
- add r11, r5, asl #3 @ r11 -> &data_in[i + factor * 8]
- add r9, r5, asl #2 @ Counter i = delay + factor * 8.
- cmp r9, r3 @ i < endpos - factor * 7 ?
- blt LOOP_ENDPOS_GENERAL
- asr r5, #1 @ Restore r5 to the value of factor.
- b POST_LOOP_ENDPOS
-
-@ The case for factor == 2.
-LOOP_ENDPOS_FACTOR2:
- @ Initializations.
- vmov.i32 q2, #2048
- vmov.i32 q3, #2048
- sub r7, r8, #2
- sub r12, r0, #1 @ coefficients_length - 1
- sub r1, r11, r12, asl #1 @ &data_in[i - j]
-
-LOOP_COEFF_LENGTH_FACTOR2:
- vld1.16 {d16[], d17[]}, [r7], r4 @ coefficients[j]
- vld2.16 {d0, d1}, [r1]! @ data_in[]
- vld2.16 {d2, d3}, [r1], r10 @ data_in[]
- subs r12, #1
- vmlal.s16 q2, d0, d16
- vmlal.s16 q3, d2, d17
- bge LOOP_COEFF_LENGTH_FACTOR2
-
- @ Shift, saturate, and store the result.
- vqshrn.s32 d0, q2, #12
- vqshrn.s32 d1, q3, #12
- vst1.16 {d0, d1}, [r2]!
-
- add r11, r5, asl #4 @ r11 -> &data_in[i + factor * 8]
- add r9, r5, asl #3 @ Counter i = delay + factor * 8.
- cmp r9, r3 @ i < endpos - factor * 7 ?
- blt LOOP_ENDPOS_FACTOR2
- b POST_LOOP_ENDPOS
-
-@ The case for factor == 4.
-LOOP_ENDPOS_FACTOR4:
- @ Initializations.
- vmov.i32 q2, #2048
- vmov.i32 q3, #2048
- sub r7, r8, #2
- sub r12, r0, #1 @ coefficients_length - 1
- sub r1, r11, r12, asl #1 @ &data_in[i - j]
-
-LOOP_COEFF_LENGTH_FACTOR4:
- vld1.16 {d16[], d17[]}, [r7], r4 @ coefficients[j]
- vld4.16 {d0, d1, d2, d3}, [r1]! @ data_in[]
- vld4.16 {d18, d19, d20, d21}, [r1], r10 @ data_in[]
- subs r12, #1
- vmlal.s16 q2, d0, d16
- vmlal.s16 q3, d18, d17
- bge LOOP_COEFF_LENGTH_FACTOR4
-
- @ Shift, saturate, and store the result.
- vqshrn.s32 d0, q2, #12
- vqshrn.s32 d1, q3, #12
- vst1.16 {d0, d1}, [r2]!
-
- add r11, r5, asl #4 @ r11 -> &data_in[i + factor * 8]
- add r9, r5, asl #3 @ Counter i = delay + factor * 8.
- cmp r9, r3 @ i < endpos - factor * 7 ?
- blt LOOP_ENDPOS_FACTOR4
-
-@
-@ Second part, do the rest iterations (if any).
-@
-
-POST_LOOP_ENDPOS:
- add r3, r5, asl #3
- sub r3, r5 @ Restore r3 to endpos.
- cmp r9, r3
- movge r0, #0
- bge END
-
-LOOP2_ENDPOS:
- @ Initializations.
- mov r7, r8
- sub r12, r0, #1 @ coefficients_length - 1
- sub r6, r11, r12, asl #1 @ &data_in[i - j]
-
- mov r1, #2048
-
-LOOP2_COEFF_LENGTH:
- ldrsh r4, [r7, #-2]! @ coefficients[j]
- ldrsh r10, [r6], #2 @ data_in[i - j]
- smlabb r1, r4, r10, r1
- subs r12, #1
- bge LOOP2_COEFF_LENGTH
-
- @ Shift, saturate, and store the result.
- ssat r1, #16, r1, asr #12
- strh r1, [r2], #2
-
- add r11, r5, asl #1 @ r11 -> &data_in[i + factor]
- add r9, r5 @ Counter i = delay + factor.
- cmp r9, r3 @ i < endpos?
- blt LOOP2_ENDPOS
-
- mov r0, #0
-
-END:
- pop {r4-r11}
- bx lr
-
-.fnend
diff --git a/trunk/src/common_audio/signal_processing/energy.c b/trunk/src/common_audio/signal_processing/energy.c
deleted file mode 100644
index e8fdf94..0000000
--- a/trunk/src/common_audio/signal_processing/energy.c
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_Energy().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_Word32 WebRtcSpl_Energy(WebRtc_Word16* vector, int vector_length, int* scale_factor)
-{
- WebRtc_Word32 en = 0;
- int i;
- int scaling = WebRtcSpl_GetScalingSquare(vector, vector_length, vector_length);
- int looptimes = vector_length;
- WebRtc_Word16 *vectorptr = vector;
-
- for (i = 0; i < looptimes; i++)
- {
- en += WEBRTC_SPL_MUL_16_16_RSFT(*vectorptr, *vectorptr, scaling);
- vectorptr++;
- }
- *scale_factor = scaling;
-
- return en;
-}
diff --git a/trunk/src/common_audio/signal_processing/filter_ar.c b/trunk/src/common_audio/signal_processing/filter_ar.c
deleted file mode 100644
index 24e83a6..0000000
--- a/trunk/src/common_audio/signal_processing/filter_ar.c
+++ /dev/null
@@ -1,89 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_FilterAR().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-int WebRtcSpl_FilterAR(G_CONST WebRtc_Word16* a,
- int a_length,
- G_CONST WebRtc_Word16* x,
- int x_length,
- WebRtc_Word16* state,
- int state_length,
- WebRtc_Word16* state_low,
- int state_low_length,
- WebRtc_Word16* filtered,
- WebRtc_Word16* filtered_low,
- int filtered_low_length)
-{
- WebRtc_Word32 o;
- WebRtc_Word32 oLOW;
- int i, j, stop;
- G_CONST WebRtc_Word16* x_ptr = &x[0];
- WebRtc_Word16* filteredFINAL_ptr = filtered;
- WebRtc_Word16* filteredFINAL_LOW_ptr = filtered_low;
-
- for (i = 0; i < x_length; i++)
- {
- // Calculate filtered[i] and filtered_low[i]
- G_CONST WebRtc_Word16* a_ptr = &a[1];
- WebRtc_Word16* filtered_ptr = &filtered[i - 1];
- WebRtc_Word16* filtered_low_ptr = &filtered_low[i - 1];
- WebRtc_Word16* state_ptr = &state[state_length - 1];
- WebRtc_Word16* state_low_ptr = &state_low[state_length - 1];
-
- o = (WebRtc_Word32)(*x_ptr++) << 12;
- oLOW = (WebRtc_Word32)0;
-
- stop = (i < a_length) ? i + 1 : a_length;
- for (j = 1; j < stop; j++)
- {
- o -= WEBRTC_SPL_MUL_16_16(*a_ptr, *filtered_ptr--);
- oLOW -= WEBRTC_SPL_MUL_16_16(*a_ptr++, *filtered_low_ptr--);
- }
- for (j = i + 1; j < a_length; j++)
- {
- o -= WEBRTC_SPL_MUL_16_16(*a_ptr, *state_ptr--);
- oLOW -= WEBRTC_SPL_MUL_16_16(*a_ptr++, *state_low_ptr--);
- }
-
- o += (oLOW >> 12);
- *filteredFINAL_ptr = (WebRtc_Word16)((o + (WebRtc_Word32)2048) >> 12);
- *filteredFINAL_LOW_ptr++ = (WebRtc_Word16)(o - ((WebRtc_Word32)(*filteredFINAL_ptr++)
- << 12));
- }
-
- // Save the filter state
- if (x_length >= state_length)
- {
- WebRtcSpl_CopyFromEndW16(filtered, x_length, a_length - 1, state);
- WebRtcSpl_CopyFromEndW16(filtered_low, x_length, a_length - 1, state_low);
- } else
- {
- for (i = 0; i < state_length - x_length; i++)
- {
- state[i] = state[i + x_length];
- state_low[i] = state_low[i + x_length];
- }
- for (i = 0; i < x_length; i++)
- {
- state[state_length - x_length + i] = filtered[i];
- state[state_length - x_length + i] = filtered_low[i];
- }
- }
-
- return x_length;
-}
diff --git a/trunk/src/common_audio/signal_processing/filter_ar_fast_q12.c b/trunk/src/common_audio/signal_processing/filter_ar_fast_q12.c
deleted file mode 100644
index 0402302..0000000
--- a/trunk/src/common_audio/signal_processing/filter_ar_fast_q12.c
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-#include <assert.h>
-
-#include "signal_processing_library.h"
-
-// TODO(bjornv): Change the return type to report errors.
-
-void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
- int16_t* data_out,
- const int16_t* __restrict coefficients,
- int coefficients_length,
- int data_length) {
- int i = 0;
- int j = 0;
-
- assert(data_length > 0);
- assert(coefficients_length > 1);
-
- for (i = 0; i < data_length; i++) {
- int32_t output = 0;
- int32_t sum = 0;
-
- for (j = coefficients_length - 1; j > 0; j--) {
- sum += coefficients[j] * data_out[i - j];
- }
-
- output = coefficients[0] * data_in[i];
- output -= sum;
-
- // Saturate and store the output.
- output = WEBRTC_SPL_SAT(134215679, output, -134217728);
- data_out[i] = (int16_t)((output + 2048) >> 12);
- }
-}
-
diff --git a/trunk/src/common_audio/signal_processing/filter_ar_fast_q12_armv7.s b/trunk/src/common_audio/signal_processing/filter_ar_fast_q12_armv7.s
deleted file mode 100644
index 5591bb8..0000000
--- a/trunk/src/common_audio/signal_processing/filter_ar_fast_q12_armv7.s
+++ /dev/null
@@ -1,223 +0,0 @@
-@
-@ Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-@
-@ Use of this source code is governed by a BSD-style license
-@ that can be found in the LICENSE file in the root of the source
-@ tree. An additional intellectual property rights grant can be found
-@ in the file PATENTS. All contributing project authors may
-@ be found in the AUTHORS file in the root of the source tree.
-@
-
-@ This file contains the function WebRtcSpl_FilterARFastQ12(), optimized for
-@ ARMv7 platform. The description header can be found in
-@ signal_processing_library.h
-@
-@ Output is bit-exact with the generic C code as in filter_ar_fast_q12.c, and
-@ the reference C code at end of this file.
-
-@ Assumptions:
-@ (1) data_length > 0
-@ (2) coefficients_length > 1
-
-@ Register usage:
-@
-@ r0: &data_in[i]
-@ r1: &data_out[i], for result ouput
-@ r2: &coefficients[0]
-@ r3: coefficients_length
-@ r4: Iteration counter for the outer loop.
-@ r5: data_out[j] as multiplication inputs
-@ r6: Calculated value for output data_out[]; interation counter for inner loop
-@ r7: Partial sum of a filtering multiplication results
-@ r8: Partial sum of a filtering multiplication results
-@ r9: &data_out[], for filtering input; data_in[i]
-@ r10: coefficients[j]
-@ r11: Scratch
-@ r12: &coefficients[j]
-
-.arch armv7-a
-
-.align 2
-.global WebRtcSpl_FilterARFastQ12
-
-WebRtcSpl_FilterARFastQ12:
-
-.fnstart
-
-.save {r4-r11}
- push {r4-r11}
-
- ldrsh r12, [sp, #32] @ data_length
- subs r4, r12, #1
- beq ODD_LENGTH @ jump if data_length == 1
-
-LOOP_LENGTH:
- add r12, r2, r3, lsl #1
- sub r12, #4 @ &coefficients[coefficients_length - 2]
- sub r9, r1, r3, lsl #1
- add r9, #2 @ &data_out[i - coefficients_length + 1]
- ldr r5, [r9], #4 @ data_out[i - coefficients_length + {1,2}]
-
- mov r7, #0 @ sum1
- mov r8, #0 @ sum2
- subs r6, r3, #3 @ Iteration counter for inner loop.
- beq ODD_A_LENGTH @ branch if coefficients_length == 3
- blt POST_LOOP_A_LENGTH @ branch if coefficients_length == 2
-
-LOOP_A_LENGTH:
- ldr r10, [r12], #-4 @ coefficients[j - 1], coefficients[j]
- subs r6, #2
- smlatt r8, r10, r5, r8 @ sum2 += coefficients[j] * data_out[i - j + 1];
- smlatb r7, r10, r5, r7 @ sum1 += coefficients[j] * data_out[i - j];
- smlabt r7, r10, r5, r7 @ coefficients[j - 1] * data_out[i - j + 1];
- ldr r5, [r9], #4 @ data_out[i - j + 2], data_out[i - j + 3]
- smlabb r8, r10, r5, r8 @ coefficients[j - 1] * data_out[i - j + 2];
- bgt LOOP_A_LENGTH
- blt POST_LOOP_A_LENGTH
-
-ODD_A_LENGTH:
- ldrsh r10, [r12, #2] @ Filter coefficients coefficients[2]
- sub r12, #2 @ &coefficients[0]
- smlabb r7, r10, r5, r7 @ sum1 += coefficients[2] * data_out[i - 2];
- smlabt r8, r10, r5, r8 @ sum2 += coefficients[2] * data_out[i - 1];
- ldr r5, [r9, #-2] @ data_out[i - 1], data_out[i]
-
-POST_LOOP_A_LENGTH:
- ldr r10, [r12] @ coefficients[0], coefficients[1]
- smlatb r7, r10, r5, r7 @ sum1 += coefficients[1] * data_out[i - 1];
-
- ldr r9, [r0], #4 @ data_in[i], data_in[i + 1]
- smulbb r6, r10, r9 @ output1 = coefficients[0] * data_in[i];
- sub r6, r7 @ output1 -= sum1;
-
- sbfx r11, r6, #12, #16
- ssat r7, #16, r6, asr #12
- cmp r7, r11
- addeq r6, r6, #2048
- ssat r6, #16, r6, asr #12
- strh r6, [r1], #2 @ Store data_out[i]
-
- smlatb r8, r10, r6, r8 @ sum2 += coefficients[1] * data_out[i];
- smulbt r6, r10, r9 @ output2 = coefficients[0] * data_in[i + 1];
- sub r6, r8 @ output1 -= sum1;
-
- sbfx r11, r6, #12, #16
- ssat r7, #16, r6, asr #12
- cmp r7, r11
- addeq r6, r6, #2048
- ssat r6, #16, r6, asr #12
- strh r6, [r1], #2 @ Store data_out[i + 1]
-
- subs r4, #2
- bgt LOOP_LENGTH
- blt END @ For even data_length, it's done. Jump to END.
-
-@ Process i = data_length -1, for the case of an odd length.
-ODD_LENGTH:
- add r12, r2, r3, lsl #1
- sub r12, #4 @ &coefficients[coefficients_length - 2]
- sub r9, r1, r3, lsl #1
- add r9, #2 @ &data_out[i - coefficients_length + 1]
- mov r7, #0 @ sum1
- mov r8, #0 @ sum1
- subs r6, r3, #2 @ inner loop counter
- beq EVEN_A_LENGTH @ branch if coefficients_length == 2
-
-LOOP2_A_LENGTH:
- ldr r10, [r12], #-4 @ coefficients[j - 1], coefficients[j]
- ldr r5, [r9], #4 @ data_out[i - j], data_out[i - j + 1]
- subs r6, #2
- smlatb r7, r10, r5, r7 @ sum1 += coefficients[j] * data_out[i - j];
- smlabt r8, r10, r5, r8 @ coefficients[j - 1] * data_out[i - j + 1];
- bgt LOOP2_A_LENGTH
- addlt r12, #2
- blt POST_LOOP2_A_LENGTH
-
-EVEN_A_LENGTH:
- ldrsh r10, [r12, #2] @ Filter coefficients coefficients[1]
- ldrsh r5, [r9] @ data_out[i - 1]
- smlabb r7, r10, r5, r7 @ sum1 += coefficients[1] * data_out[i - 1];
-
-POST_LOOP2_A_LENGTH:
- ldrsh r10, [r12] @ Filter coefficients coefficients[0]
- ldrsh r9, [r0] @ data_in[i]
- smulbb r6, r10, r9 @ output1 = coefficients[0] * data_in[i];
- sub r6, r7 @ output1 -= sum1;
- sub r6, r8 @ output1 -= sum1;
- sbfx r8, r6, #12, #16
- ssat r7, #16, r6, asr #12
- cmp r7, r8
- addeq r6, r6, #2048
- ssat r6, #16, r6, asr #12
- strh r6, [r1] @ Store the data_out[i]
-
-END:
- pop {r4-r11}
- bx lr
-
-.fnend
-
-
-@Reference C code:
-@
-@void WebRtcSpl_FilterARFastQ12(int16_t* data_in,
-@ int16_t* data_out,
-@ int16_t* __restrict coefficients,
-@ int coefficients_length,
-@ int data_length) {
-@ int i = 0;
-@ int j = 0;
-@
-@ for (i = 0; i < data_length - 1; i += 2) {
-@ int32_t output1 = 0;
-@ int32_t sum1 = 0;
-@ int32_t output2 = 0;
-@ int32_t sum2 = 0;
-@
-@ for (j = coefficients_length - 1; j > 2; j -= 2) {
-@ sum1 += coefficients[j] * data_out[i - j];
-@ sum1 += coefficients[j - 1] * data_out[i - j + 1];
-@ sum2 += coefficients[j] * data_out[i - j + 1];
-@ sum2 += coefficients[j - 1] * data_out[i - j + 2];
-@ }
-@
-@ if (j == 2) {
-@ sum1 += coefficients[2] * data_out[i - 2];
-@ sum2 += coefficients[2] * data_out[i - 1];
-@ }
-@
-@ sum1 += coefficients[1] * data_out[i - 1];
-@ output1 = coefficients[0] * data_in[i];
-@ output1 -= sum1;
-@ // Saturate and store the output.
-@ output1 = WEBRTC_SPL_SAT(134215679, output1, -134217728);
-@ data_out[i] = (int16_t)((output1 + 2048) >> 12);
-@
-@ sum2 += coefficients[1] * data_out[i];
-@ output2 = coefficients[0] * data_in[i + 1];
-@ output2 -= sum2;
-@ // Saturate and store the output.
-@ output2 = WEBRTC_SPL_SAT(134215679, output2, -134217728);
-@ data_out[i + 1] = (int16_t)((output2 + 2048) >> 12);
-@ }
-@
-@ if (i == data_length - 1) {
-@ int32_t output1 = 0;
-@ int32_t sum1 = 0;
-@
-@ for (j = coefficients_length - 1; j > 1; j -= 2) {
-@ sum1 += coefficients[j] * data_out[i - j];
-@ sum1 += coefficients[j - 1] * data_out[i - j + 1];
-@ }
-@
-@ if (j == 1) {
-@ sum1 += coefficients[1] * data_out[i - 1];
-@ }
-@
-@ output1 = coefficients[0] * data_in[i];
-@ output1 -= sum1;
-@ // Saturate and store the output.
-@ output1 = WEBRTC_SPL_SAT(134215679, output1, -134217728);
-@ data_out[i] = (int16_t)((output1 + 2048) >> 12);
-@ }
-@}
diff --git a/trunk/src/common_audio/signal_processing/filter_ma_fast_q12.c b/trunk/src/common_audio/signal_processing/filter_ma_fast_q12.c
deleted file mode 100644
index 19ad9b1..0000000
--- a/trunk/src/common_audio/signal_processing/filter_ma_fast_q12.c
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_FilterMAFastQ12().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_FilterMAFastQ12(WebRtc_Word16* in_ptr,
- WebRtc_Word16* out_ptr,
- WebRtc_Word16* B,
- WebRtc_Word16 B_length,
- WebRtc_Word16 length)
-{
- WebRtc_Word32 o;
- int i, j;
- for (i = 0; i < length; i++)
- {
- G_CONST WebRtc_Word16* b_ptr = &B[0];
- G_CONST WebRtc_Word16* x_ptr = &in_ptr[i];
-
- o = (WebRtc_Word32)0;
-
- for (j = 0; j < B_length; j++)
- {
- o += WEBRTC_SPL_MUL_16_16(*b_ptr++, *x_ptr--);
- }
-
- // If output is higher than 32768, saturate it. Same with negative side
- // 2^27 = 134217728, which corresponds to 32768 in Q12
-
- // Saturate the output
- o = WEBRTC_SPL_SAT((WebRtc_Word32)134215679, o, (WebRtc_Word32)-134217728);
-
- *out_ptr++ = (WebRtc_Word16)((o + (WebRtc_Word32)2048) >> 12);
- }
- return;
-}
diff --git a/trunk/src/common_audio/signal_processing/get_hanning_window.c b/trunk/src/common_audio/signal_processing/get_hanning_window.c
deleted file mode 100644
index 6d67e60..0000000
--- a/trunk/src/common_audio/signal_processing/get_hanning_window.c
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_GetHanningWindow().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-// Hanning table with 256 entries
-static const WebRtc_Word16 kHanningTable[] = {
- 1, 2, 6, 10, 15, 22, 30, 39,
- 50, 62, 75, 89, 104, 121, 138, 157,
- 178, 199, 222, 246, 271, 297, 324, 353,
- 383, 413, 446, 479, 513, 549, 586, 624,
- 663, 703, 744, 787, 830, 875, 920, 967,
- 1015, 1064, 1114, 1165, 1218, 1271, 1325, 1381,
- 1437, 1494, 1553, 1612, 1673, 1734, 1796, 1859,
- 1924, 1989, 2055, 2122, 2190, 2259, 2329, 2399,
- 2471, 2543, 2617, 2691, 2765, 2841, 2918, 2995,
- 3073, 3152, 3232, 3312, 3393, 3475, 3558, 3641,
- 3725, 3809, 3895, 3980, 4067, 4154, 4242, 4330,
- 4419, 4509, 4599, 4689, 4781, 4872, 4964, 5057,
- 5150, 5244, 5338, 5432, 5527, 5622, 5718, 5814,
- 5910, 6007, 6104, 6202, 6299, 6397, 6495, 6594,
- 6693, 6791, 6891, 6990, 7090, 7189, 7289, 7389,
- 7489, 7589, 7690, 7790, 7890, 7991, 8091, 8192,
- 8293, 8393, 8494, 8594, 8694, 8795, 8895, 8995,
- 9095, 9195, 9294, 9394, 9493, 9593, 9691, 9790,
- 9889, 9987, 10085, 10182, 10280, 10377, 10474, 10570,
-10666, 10762, 10857, 10952, 11046, 11140, 11234, 11327,
-11420, 11512, 11603, 11695, 11785, 11875, 11965, 12054,
-12142, 12230, 12317, 12404, 12489, 12575, 12659, 12743,
-12826, 12909, 12991, 13072, 13152, 13232, 13311, 13389,
-13466, 13543, 13619, 13693, 13767, 13841, 13913, 13985,
-14055, 14125, 14194, 14262, 14329, 14395, 14460, 14525,
-14588, 14650, 14711, 14772, 14831, 14890, 14947, 15003,
-15059, 15113, 15166, 15219, 15270, 15320, 15369, 15417,
-15464, 15509, 15554, 15597, 15640, 15681, 15721, 15760,
-15798, 15835, 15871, 15905, 15938, 15971, 16001, 16031,
-16060, 16087, 16113, 16138, 16162, 16185, 16206, 16227,
-16246, 16263, 16280, 16295, 16309, 16322, 16334, 16345,
-16354, 16362, 16369, 16374, 16378, 16382, 16383, 16384
-};
-
-void WebRtcSpl_GetHanningWindow(WebRtc_Word16 *v, WebRtc_Word16 size)
-{
- int jj;
- WebRtc_Word16 *vptr1;
-
- WebRtc_Word32 index;
- WebRtc_Word32 factor = ((WebRtc_Word32)0x40000000);
-
- factor = WebRtcSpl_DivW32W16(factor, size);
- if (size < 513)
- index = (WebRtc_Word32)-0x200000;
- else
- index = (WebRtc_Word32)-0x100000;
- vptr1 = v;
-
- for (jj = 0; jj < size; jj++)
- {
- index += factor;
- (*vptr1++) = kHanningTable[index >> 22];
- }
-
-}
diff --git a/trunk/src/common_audio/signal_processing/get_scaling_square.c b/trunk/src/common_audio/signal_processing/get_scaling_square.c
deleted file mode 100644
index dccbf33..0000000
--- a/trunk/src/common_audio/signal_processing/get_scaling_square.c
+++ /dev/null
@@ -1,44 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_GetScalingSquare().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-int WebRtcSpl_GetScalingSquare(WebRtc_Word16 *in_vector, int in_vector_length, int times)
-{
- int nbits = WebRtcSpl_GetSizeInBits(times);
- int i;
- WebRtc_Word16 smax = -1;
- WebRtc_Word16 sabs;
- WebRtc_Word16 *sptr = in_vector;
- int t;
- int looptimes = in_vector_length;
-
- for (i = looptimes; i > 0; i--)
- {
- sabs = (*sptr > 0 ? *sptr++ : -*sptr++);
- smax = (sabs > smax ? sabs : smax);
- }
- t = WebRtcSpl_NormW32(WEBRTC_SPL_MUL(smax, smax));
-
- if (smax == 0)
- {
- return 0; // Since norm(0) returns 0
- } else
- {
- return (t > nbits) ? 0 : nbits - t;
- }
-}
diff --git a/trunk/src/common_audio/signal_processing/ilbc_specific_functions.c b/trunk/src/common_audio/signal_processing/ilbc_specific_functions.c
deleted file mode 100644
index 5a9e577..0000000
--- a/trunk/src/common_audio/signal_processing/ilbc_specific_functions.c
+++ /dev/null
@@ -1,120 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains implementations of the iLBC specific functions
- * WebRtcSpl_ScaleAndAddVectorsWithRound()
- * WebRtcSpl_ReverseOrderMultArrayElements()
- * WebRtcSpl_ElementwiseVectorMult()
- * WebRtcSpl_AddVectorsAndShift()
- * WebRtcSpl_AddAffineVectorToVector()
- * WebRtcSpl_AffineTransformVector()
- *
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_ScaleAndAddVectorsWithRound(WebRtc_Word16 *vector1, WebRtc_Word16 scale1,
- WebRtc_Word16 *vector2, WebRtc_Word16 scale2,
- WebRtc_Word16 right_shifts, WebRtc_Word16 *out,
- WebRtc_Word16 vector_length)
-{
- int i;
- WebRtc_Word16 roundVal;
- roundVal = 1 << right_shifts;
- roundVal = roundVal >> 1;
- for (i = 0; i < vector_length; i++)
- {
- out[i] = (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16(vector1[i], scale1)
- + WEBRTC_SPL_MUL_16_16(vector2[i], scale2) + roundVal) >> right_shifts);
- }
-}
-
-void WebRtcSpl_ReverseOrderMultArrayElements(WebRtc_Word16 *out, G_CONST WebRtc_Word16 *in,
- G_CONST WebRtc_Word16 *win,
- WebRtc_Word16 vector_length,
- WebRtc_Word16 right_shifts)
-{
- int i;
- WebRtc_Word16 *outptr = out;
- G_CONST WebRtc_Word16 *inptr = in;
- G_CONST WebRtc_Word16 *winptr = win;
- for (i = 0; i < vector_length; i++)
- {
- (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++,
- *winptr--, right_shifts);
- }
-}
-
-void WebRtcSpl_ElementwiseVectorMult(WebRtc_Word16 *out, G_CONST WebRtc_Word16 *in,
- G_CONST WebRtc_Word16 *win, WebRtc_Word16 vector_length,
- WebRtc_Word16 right_shifts)
-{
- int i;
- WebRtc_Word16 *outptr = out;
- G_CONST WebRtc_Word16 *inptr = in;
- G_CONST WebRtc_Word16 *winptr = win;
- for (i = 0; i < vector_length; i++)
- {
- (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++,
- *winptr++, right_shifts);
- }
-}
-
-void WebRtcSpl_AddVectorsAndShift(WebRtc_Word16 *out, G_CONST WebRtc_Word16 *in1,
- G_CONST WebRtc_Word16 *in2, WebRtc_Word16 vector_length,
- WebRtc_Word16 right_shifts)
-{
- int i;
- WebRtc_Word16 *outptr = out;
- G_CONST WebRtc_Word16 *in1ptr = in1;
- G_CONST WebRtc_Word16 *in2ptr = in2;
- for (i = vector_length; i > 0; i--)
- {
- (*outptr++) = (WebRtc_Word16)(((*in1ptr++) + (*in2ptr++)) >> right_shifts);
- }
-}
-
-void WebRtcSpl_AddAffineVectorToVector(WebRtc_Word16 *out, WebRtc_Word16 *in,
- WebRtc_Word16 gain, WebRtc_Word32 add_constant,
- WebRtc_Word16 right_shifts, int vector_length)
-{
- WebRtc_Word16 *inPtr;
- WebRtc_Word16 *outPtr;
- int i;
-
- inPtr = in;
- outPtr = out;
- for (i = 0; i < vector_length; i++)
- {
- (*outPtr++) += (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
- + (WebRtc_Word32)add_constant) >> right_shifts);
- }
-}
-
-void WebRtcSpl_AffineTransformVector(WebRtc_Word16 *out, WebRtc_Word16 *in,
- WebRtc_Word16 gain, WebRtc_Word32 add_constant,
- WebRtc_Word16 right_shifts, int vector_length)
-{
- WebRtc_Word16 *inPtr;
- WebRtc_Word16 *outPtr;
- int i;
-
- inPtr = in;
- outPtr = out;
- for (i = 0; i < vector_length; i++)
- {
- (*outPtr++) = (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16((*inPtr++), gain)
- + (WebRtc_Word32)add_constant) >> right_shifts);
- }
-}
diff --git a/trunk/src/common_audio/signal_processing/include/signal_processing_library.h b/trunk/src/common_audio/signal_processing/include/signal_processing_library.h
deleted file mode 100644
index 87b8f29..0000000
--- a/trunk/src/common_audio/signal_processing/include/signal_processing_library.h
+++ /dev/null
@@ -1,1667 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes all of the fix point signal processing library (SPL) function
- * descriptions and declarations.
- * For specific function calls, see bottom of file.
- */
-
-#ifndef WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
-#define WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
-
-#include <string.h>
-#include "typedefs.h"
-
-#ifdef ARM_WINM
-#include <Armintr.h> // intrinsic file for windows mobile
-#endif
-
-// Macros specific for the fixed point implementation
-#define WEBRTC_SPL_WORD16_MAX 32767
-#define WEBRTC_SPL_WORD16_MIN -32768
-#define WEBRTC_SPL_WORD32_MAX (WebRtc_Word32)0x7fffffff
-#define WEBRTC_SPL_WORD32_MIN (WebRtc_Word32)0x80000000
-#define WEBRTC_SPL_MAX_LPC_ORDER 14
-#define WEBRTC_SPL_MAX_SEED_USED 0x80000000L
-#define WEBRTC_SPL_MIN(A, B) (A < B ? A : B) // Get min value
-#define WEBRTC_SPL_MAX(A, B) (A > B ? A : B) // Get max value
-#define WEBRTC_SPL_ABS_W16(a) \
- (((WebRtc_Word16)a >= 0) ? ((WebRtc_Word16)a) : -((WebRtc_Word16)a))
-#define WEBRTC_SPL_ABS_W32(a) \
- (((WebRtc_Word32)a >= 0) ? ((WebRtc_Word32)a) : -((WebRtc_Word32)a))
-
-#if (defined WEBRTC_TARGET_PC)||(defined __TARGET_XSCALE)
-#define WEBRTC_SPL_GET_BYTE(a, nr) (((WebRtc_Word8 *)a)[nr])
-#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
- (((WebRtc_Word8 *)d_ptr)[index] = (val))
-#elif defined WEBRTC_BIG_ENDIAN
-#define WEBRTC_SPL_GET_BYTE(a, nr) \
- ((((WebRtc_Word16 *)a)[nr >> 1]) >> (((nr + 1) & 0x1) * 8) & 0x00ff)
-#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
- ((WebRtc_Word16 *)d_ptr)[index >> 1] = \
- ((((WebRtc_Word16 *)d_ptr)[index >> 1]) \
- & (0x00ff << (8 * ((index) & 0x1)))) | (val << (8 * ((index + 1) & 0x1)))
-#else
-#define WEBRTC_SPL_GET_BYTE(a,nr) \
- ((((WebRtc_Word16 *)(a))[(nr) >> 1]) >> (((nr) & 0x1) * 8) & 0x00ff)
-#define WEBRTC_SPL_SET_BYTE(d_ptr, val, index) \
- ((WebRtc_Word16 *)(d_ptr))[(index) >> 1] = \
- ((((WebRtc_Word16 *)(d_ptr))[(index) >> 1]) \
- & (0x00ff << (8 * (((index) + 1) & 0x1)))) | \
- ((val) << (8 * ((index) & 0x1)))
-#endif
-
-#define WEBRTC_SPL_MUL(a, b) \
- ((WebRtc_Word32) ((WebRtc_Word32)(a) * (WebRtc_Word32)(b)))
-#define WEBRTC_SPL_UMUL(a, b) \
- ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord32)(b)))
-#define WEBRTC_SPL_UMUL_RSFT16(a, b) \
- ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord32)(b)) >> 16)
-#define WEBRTC_SPL_UMUL_16_16(a, b) \
- ((WebRtc_UWord32) (WebRtc_UWord16)(a) * (WebRtc_UWord16)(b))
-#define WEBRTC_SPL_UMUL_16_16_RSFT16(a, b) \
- (((WebRtc_UWord32) (WebRtc_UWord16)(a) * (WebRtc_UWord16)(b)) >> 16)
-#define WEBRTC_SPL_UMUL_32_16(a, b) \
- ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord16)(b)))
-#define WEBRTC_SPL_UMUL_32_16_RSFT16(a, b) \
- ((WebRtc_UWord32) ((WebRtc_UWord32)(a) * (WebRtc_UWord16)(b)) >> 16)
-#define WEBRTC_SPL_MUL_16_U16(a, b) \
- ((WebRtc_Word32)(WebRtc_Word16)(a) * (WebRtc_UWord16)(b))
-#define WEBRTC_SPL_DIV(a, b) \
- ((WebRtc_Word32) ((WebRtc_Word32)(a) / (WebRtc_Word32)(b)))
-#define WEBRTC_SPL_UDIV(a, b) \
- ((WebRtc_UWord32) ((WebRtc_UWord32)(a) / (WebRtc_UWord32)(b)))
-
-#ifndef WEBRTC_ARCH_ARM_V7A
-// For ARMv7 platforms, these are inline functions in spl_inl_armv7.h
-#define WEBRTC_SPL_MUL_16_16(a, b) \
- ((WebRtc_Word32) (((WebRtc_Word16)(a)) * ((WebRtc_Word16)(b))))
-#define WEBRTC_SPL_MUL_16_32_RSFT16(a, b) \
- (WEBRTC_SPL_MUL_16_16(a, b >> 16) \
- + ((WEBRTC_SPL_MUL_16_16(a, (b & 0xffff) >> 1) + 0x4000) >> 15))
-#define WEBRTC_SPL_MUL_32_32_RSFT32(a32a, a32b, b32) \
- ((WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT16(a32a, b32) \
- + (WEBRTC_SPL_MUL_16_32_RSFT16(a32b, b32) >> 16)))
-#define WEBRTC_SPL_MUL_32_32_RSFT32BI(a32, b32) \
- ((WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT16(( \
- (WebRtc_Word16)(a32 >> 16)), b32) + \
- (WEBRTC_SPL_MUL_16_32_RSFT16(( \
- (WebRtc_Word16)((a32 & 0x0000FFFF) >> 1)), b32) >> 15)))
-#endif
-
-#define WEBRTC_SPL_MUL_16_32_RSFT11(a, b) \
- ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 5) \
- + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x0200) >> 10))
-#define WEBRTC_SPL_MUL_16_32_RSFT14(a, b) \
- ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 2) \
- + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x1000) >> 13))
-#define WEBRTC_SPL_MUL_16_32_RSFT15(a, b) \
- ((WEBRTC_SPL_MUL_16_16(a, (b) >> 16) << 1) \
- + (((WEBRTC_SPL_MUL_16_U16(a, (WebRtc_UWord16)(b)) >> 1) + 0x2000) >> 14))
-
-#ifdef ARM_WINM
-#define WEBRTC_SPL_MUL_16_16(a, b) \
- _SmulLo_SW_SL((WebRtc_Word16)(a), (WebRtc_Word16)(b))
-#endif
-
-#define WEBRTC_SPL_MUL_16_16_RSFT(a, b, c) \
- (WEBRTC_SPL_MUL_16_16(a, b) >> (c))
-
-#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, c) \
- ((WEBRTC_SPL_MUL_16_16(a, b) + ((WebRtc_Word32) \
- (((WebRtc_Word32)1) << ((c) - 1)))) >> (c))
-#define WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b) \
- ((WEBRTC_SPL_MUL_16_16(a, b) + ((WebRtc_Word32) (1 << 14))) >> 15)
-
-// C + the 32 most significant bits of A * B
-#define WEBRTC_SPL_SCALEDIFF32(A, B, C) \
- (C + (B >> 16) * A + (((WebRtc_UWord32)(0x0000FFFF & B) * A) >> 16))
-
-#define WEBRTC_SPL_ADD_SAT_W32(a, b) WebRtcSpl_AddSatW32(a, b)
-#define WEBRTC_SPL_SAT(a, b, c) (b > a ? a : b < c ? c : b)
-#define WEBRTC_SPL_MUL_32_16(a, b) ((a) * (b))
-
-#define WEBRTC_SPL_SUB_SAT_W32(a, b) WebRtcSpl_SubSatW32(a, b)
-#define WEBRTC_SPL_ADD_SAT_W16(a, b) WebRtcSpl_AddSatW16(a, b)
-#define WEBRTC_SPL_SUB_SAT_W16(a, b) WebRtcSpl_SubSatW16(a, b)
-
-// We cannot do casting here due to signed/unsigned problem
-#define WEBRTC_SPL_IS_NEG(a) ((a) & 0x80000000)
-// Shifting with negative numbers allowed
-// Positive means left shift
-#define WEBRTC_SPL_SHIFT_W16(x, c) \
- (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c))))
-#define WEBRTC_SPL_SHIFT_W32(x, c) \
- (((c) >= 0) ? ((x) << (c)) : ((x) >> (-(c))))
-
-// Shifting with negative numbers not allowed
-// We cannot do casting here due to signed/unsigned problem
-#define WEBRTC_SPL_RSHIFT_W16(x, c) ((x) >> (c))
-#define WEBRTC_SPL_LSHIFT_W16(x, c) ((x) << (c))
-#define WEBRTC_SPL_RSHIFT_W32(x, c) ((x) >> (c))
-#define WEBRTC_SPL_LSHIFT_W32(x, c) ((x) << (c))
-
-#define WEBRTC_SPL_RSHIFT_U16(x, c) ((WebRtc_UWord16)(x) >> (c))
-#define WEBRTC_SPL_LSHIFT_U16(x, c) ((WebRtc_UWord16)(x) << (c))
-#define WEBRTC_SPL_RSHIFT_U32(x, c) ((WebRtc_UWord32)(x) >> (c))
-#define WEBRTC_SPL_LSHIFT_U32(x, c) ((WebRtc_UWord32)(x) << (c))
-
-#define WEBRTC_SPL_VNEW(t, n) (t *) malloc (sizeof (t) * (n))
-#define WEBRTC_SPL_FREE free
-
-#define WEBRTC_SPL_RAND(a) \
- ((WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT((a), 18816, 7) & 0x00007fff))
-
-#ifdef __cplusplus
-extern "C"
-{
-#endif
-
-#define WEBRTC_SPL_MEMCPY_W8(v1, v2, length) \
- memcpy(v1, v2, (length) * sizeof(char))
-#define WEBRTC_SPL_MEMCPY_W16(v1, v2, length) \
- memcpy(v1, v2, (length) * sizeof(WebRtc_Word16))
-
-#define WEBRTC_SPL_MEMMOVE_W16(v1, v2, length) \
- memmove(v1, v2, (length) * sizeof(WebRtc_Word16))
-
-// inline functions:
-#include "spl_inl.h"
-
-// Get SPL Version
-WebRtc_Word16 WebRtcSpl_get_version(char* version,
- WebRtc_Word16 length_in_bytes);
-
-int WebRtcSpl_GetScalingSquare(WebRtc_Word16* in_vector,
- int in_vector_length,
- int times);
-
-// Copy and set operations. Implementation in copy_set_operations.c.
-// Descriptions at bottom of file.
-void WebRtcSpl_MemSetW16(WebRtc_Word16* vector,
- WebRtc_Word16 set_value,
- int vector_length);
-void WebRtcSpl_MemSetW32(WebRtc_Word32* vector,
- WebRtc_Word32 set_value,
- int vector_length);
-void WebRtcSpl_MemCpyReversedOrder(WebRtc_Word16* out_vector,
- WebRtc_Word16* in_vector,
- int vector_length);
-WebRtc_Word16 WebRtcSpl_CopyFromEndW16(G_CONST WebRtc_Word16* in_vector,
- WebRtc_Word16 in_vector_length,
- WebRtc_Word16 samples,
- WebRtc_Word16* out_vector);
-WebRtc_Word16 WebRtcSpl_ZerosArrayW16(WebRtc_Word16* vector,
- WebRtc_Word16 vector_length);
-WebRtc_Word16 WebRtcSpl_ZerosArrayW32(WebRtc_Word32* vector,
- WebRtc_Word16 vector_length);
-WebRtc_Word16 WebRtcSpl_OnesArrayW16(WebRtc_Word16* vector,
- WebRtc_Word16 vector_length);
-WebRtc_Word16 WebRtcSpl_OnesArrayW32(WebRtc_Word32* vector,
- WebRtc_Word16 vector_length);
-// End: Copy and set operations.
-
-// Minimum and maximum operations. Implementation in min_max_operations.c.
-// Descriptions at bottom of file.
-WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(const WebRtc_Word16* vector,
- WebRtc_Word16 length);
-WebRtc_Word32 WebRtcSpl_MaxAbsValueW32(G_CONST WebRtc_Word32* vector,
- WebRtc_Word16 length);
-WebRtc_Word16 WebRtcSpl_MinValueW16(G_CONST WebRtc_Word16* vector,
- WebRtc_Word16 length);
-WebRtc_Word32 WebRtcSpl_MinValueW32(G_CONST WebRtc_Word32* vector,
- WebRtc_Word16 length);
-WebRtc_Word16 WebRtcSpl_MaxValueW16(G_CONST WebRtc_Word16* vector,
- WebRtc_Word16 length);
-
-WebRtc_Word16 WebRtcSpl_MaxAbsIndexW16(G_CONST WebRtc_Word16* vector,
- WebRtc_Word16 length);
-WebRtc_Word32 WebRtcSpl_MaxValueW32(G_CONST WebRtc_Word32* vector,
- WebRtc_Word16 length);
-WebRtc_Word16 WebRtcSpl_MinIndexW16(G_CONST WebRtc_Word16* vector,
- WebRtc_Word16 length);
-WebRtc_Word16 WebRtcSpl_MinIndexW32(G_CONST WebRtc_Word32* vector,
- WebRtc_Word16 length);
-WebRtc_Word16 WebRtcSpl_MaxIndexW16(G_CONST WebRtc_Word16* vector,
- WebRtc_Word16 length);
-WebRtc_Word16 WebRtcSpl_MaxIndexW32(G_CONST WebRtc_Word32* vector,
- WebRtc_Word16 length);
-// End: Minimum and maximum operations.
-
-// Vector scaling operations. Implementation in vector_scaling_operations.c.
-// Description at bottom of file.
-void WebRtcSpl_VectorBitShiftW16(WebRtc_Word16* out_vector,
- WebRtc_Word16 vector_length,
- G_CONST WebRtc_Word16* in_vector,
- WebRtc_Word16 right_shifts);
-void WebRtcSpl_VectorBitShiftW32(WebRtc_Word32* out_vector,
- WebRtc_Word16 vector_length,
- G_CONST WebRtc_Word32* in_vector,
- WebRtc_Word16 right_shifts);
-void WebRtcSpl_VectorBitShiftW32ToW16(WebRtc_Word16* out_vector,
- WebRtc_Word16 vector_length,
- G_CONST WebRtc_Word32* in_vector,
- WebRtc_Word16 right_shifts);
-
-void WebRtcSpl_ScaleVector(G_CONST WebRtc_Word16* in_vector,
- WebRtc_Word16* out_vector,
- WebRtc_Word16 gain,
- WebRtc_Word16 vector_length,
- WebRtc_Word16 right_shifts);
-void WebRtcSpl_ScaleVectorWithSat(G_CONST WebRtc_Word16* in_vector,
- WebRtc_Word16* out_vector,
- WebRtc_Word16 gain,
- WebRtc_Word16 vector_length,
- WebRtc_Word16 right_shifts);
-void WebRtcSpl_ScaleAndAddVectors(G_CONST WebRtc_Word16* in_vector1,
- WebRtc_Word16 gain1, int right_shifts1,
- G_CONST WebRtc_Word16* in_vector2,
- WebRtc_Word16 gain2, int right_shifts2,
- WebRtc_Word16* out_vector,
- int vector_length);
-// End: Vector scaling operations.
-
-// iLBC specific functions. Implementations in ilbc_specific_functions.c.
-// Description at bottom of file.
-void WebRtcSpl_ScaleAndAddVectorsWithRound(WebRtc_Word16* in_vector1,
- WebRtc_Word16 scale1,
- WebRtc_Word16* in_vector2,
- WebRtc_Word16 scale2,
- WebRtc_Word16 right_shifts,
- WebRtc_Word16* out_vector,
- WebRtc_Word16 vector_length);
-void WebRtcSpl_ReverseOrderMultArrayElements(WebRtc_Word16* out_vector,
- G_CONST WebRtc_Word16* in_vector,
- G_CONST WebRtc_Word16* window,
- WebRtc_Word16 vector_length,
- WebRtc_Word16 right_shifts);
-void WebRtcSpl_ElementwiseVectorMult(WebRtc_Word16* out_vector,
- G_CONST WebRtc_Word16* in_vector,
- G_CONST WebRtc_Word16* window,
- WebRtc_Word16 vector_length,
- WebRtc_Word16 right_shifts);
-void WebRtcSpl_AddVectorsAndShift(WebRtc_Word16* out_vector,
- G_CONST WebRtc_Word16* in_vector1,
- G_CONST WebRtc_Word16* in_vector2,
- WebRtc_Word16 vector_length,
- WebRtc_Word16 right_shifts);
-void WebRtcSpl_AddAffineVectorToVector(WebRtc_Word16* out_vector,
- WebRtc_Word16* in_vector,
- WebRtc_Word16 gain,
- WebRtc_Word32 add_constant,
- WebRtc_Word16 right_shifts,
- int vector_length);
-void WebRtcSpl_AffineTransformVector(WebRtc_Word16* out_vector,
- WebRtc_Word16* in_vector,
- WebRtc_Word16 gain,
- WebRtc_Word32 add_constant,
- WebRtc_Word16 right_shifts,
- int vector_length);
-// End: iLBC specific functions.
-
-// Signal processing operations. Descriptions at bottom of this file.
-int WebRtcSpl_AutoCorrelation(G_CONST WebRtc_Word16* vector,
- int vector_length, int order,
- WebRtc_Word32* result_vector,
- int* scale);
-WebRtc_Word16 WebRtcSpl_LevinsonDurbin(WebRtc_Word32* auto_corr,
- WebRtc_Word16* lpc_coef,
- WebRtc_Word16* refl_coef,
- WebRtc_Word16 order);
-void WebRtcSpl_ReflCoefToLpc(G_CONST WebRtc_Word16* refl_coef,
- int use_order,
- WebRtc_Word16* lpc_coef);
-void WebRtcSpl_LpcToReflCoef(WebRtc_Word16* lpc_coef,
- int use_order,
- WebRtc_Word16* refl_coef);
-void WebRtcSpl_AutoCorrToReflCoef(G_CONST WebRtc_Word32* auto_corr,
- int use_order,
- WebRtc_Word16* refl_coef);
-void WebRtcSpl_CrossCorrelation(WebRtc_Word32* cross_corr,
- WebRtc_Word16* vector1,
- WebRtc_Word16* vector2,
- WebRtc_Word16 dim_vector,
- WebRtc_Word16 dim_cross_corr,
- WebRtc_Word16 right_shifts,
- WebRtc_Word16 step_vector2);
-void WebRtcSpl_GetHanningWindow(WebRtc_Word16* window, WebRtc_Word16 size);
-void WebRtcSpl_SqrtOfOneMinusXSquared(WebRtc_Word16* in_vector,
- int vector_length,
- WebRtc_Word16* out_vector);
-// End: Signal processing operations.
-
-// Randomization functions. Implementations collected in randomization_functions.c and
-// descriptions at bottom of this file.
-WebRtc_UWord32 WebRtcSpl_IncreaseSeed(WebRtc_UWord32* seed);
-WebRtc_Word16 WebRtcSpl_RandU(WebRtc_UWord32* seed);
-WebRtc_Word16 WebRtcSpl_RandN(WebRtc_UWord32* seed);
-WebRtc_Word16 WebRtcSpl_RandUArray(WebRtc_Word16* vector,
- WebRtc_Word16 vector_length,
- WebRtc_UWord32* seed);
-// End: Randomization functions.
-
-// Math functions
-WebRtc_Word32 WebRtcSpl_Sqrt(WebRtc_Word32 value);
-WebRtc_Word32 WebRtcSpl_SqrtFloor(WebRtc_Word32 value);
-
-// Divisions. Implementations collected in division_operations.c and
-// descriptions at bottom of this file.
-WebRtc_UWord32 WebRtcSpl_DivU32U16(WebRtc_UWord32 num, WebRtc_UWord16 den);
-WebRtc_Word32 WebRtcSpl_DivW32W16(WebRtc_Word32 num, WebRtc_Word16 den);
-WebRtc_Word16 WebRtcSpl_DivW32W16ResW16(WebRtc_Word32 num, WebRtc_Word16 den);
-WebRtc_Word32 WebRtcSpl_DivResultInQ31(WebRtc_Word32 num, WebRtc_Word32 den);
-WebRtc_Word32 WebRtcSpl_DivW32HiLow(WebRtc_Word32 num, WebRtc_Word16 den_hi,
- WebRtc_Word16 den_low);
-// End: Divisions.
-
-WebRtc_Word32 WebRtcSpl_Energy(WebRtc_Word16* vector,
- int vector_length,
- int* scale_factor);
-
-WebRtc_Word32 WebRtcSpl_DotProductWithScale(WebRtc_Word16* vector1,
- WebRtc_Word16* vector2,
- int vector_length,
- int scaling);
-
-// Filter operations.
-int WebRtcSpl_FilterAR(G_CONST WebRtc_Word16* ar_coef, int ar_coef_length,
- G_CONST WebRtc_Word16* in_vector, int in_vector_length,
- WebRtc_Word16* filter_state, int filter_state_length,
- WebRtc_Word16* filter_state_low,
- int filter_state_low_length, WebRtc_Word16* out_vector,
- WebRtc_Word16* out_vector_low, int out_vector_low_length);
-
-void WebRtcSpl_FilterMAFastQ12(WebRtc_Word16* in_vector,
- WebRtc_Word16* out_vector,
- WebRtc_Word16* ma_coef,
- WebRtc_Word16 ma_coef_length,
- WebRtc_Word16 vector_length);
-
-// Performs a AR filtering on a vector in Q12
-// Input:
-// - data_in : Input samples
-// - data_out : State information in positions
-// data_out[-order] .. data_out[-1]
-// - coefficients : Filter coefficients (in Q12)
-// - coefficients_length: Number of coefficients (order+1)
-// - data_length : Number of samples to be filtered
-// Output:
-// - data_out : Filtered samples
-void WebRtcSpl_FilterARFastQ12(const int16_t* data_in,
- int16_t* data_out,
- const int16_t* __restrict coefficients,
- int coefficients_length,
- int data_length);
-
-// Performs a MA down sampling filter on a vector
-// Input:
-// - data_in : Input samples (state in positions
-// data_in[-order] .. data_in[-1])
-// - data_in_length : Number of samples in |data_in| to be filtered.
-// This must be at least
-// |delay| + |factor|*(|out_vector_length|-1) + 1)
-// - data_out_length : Number of down sampled samples desired
-// - coefficients : Filter coefficients (in Q12)
-// - coefficients_length: Number of coefficients (order+1)
-// - factor : Decimation factor
-// - delay : Delay of filter (compensated for in out_vector)
-// Output:
-// - data_out : Filtered samples
-// Return value : 0 if OK, -1 if |in_vector| is too short
-int WebRtcSpl_DownsampleFast(const int16_t* data_in,
- int data_in_length,
- int16_t* data_out,
- int data_out_length,
- const int16_t* __restrict coefficients,
- int coefficients_length,
- int factor,
- int delay);
-
-// End: Filter operations.
-
-// FFT operations
-
-int WebRtcSpl_ComplexFFT(WebRtc_Word16 vector[], int stages, int mode);
-int WebRtcSpl_ComplexIFFT(WebRtc_Word16 vector[], int stages, int mode);
-
-// Treat a 16-bit complex data buffer |complex_data| as an array of 32-bit
-// values, and swap elements whose indexes are bit-reverses of each other.
-//
-// Input:
-// - complex_data : Complex data buffer containing 2^|stages| real
-// elements interleaved with 2^|stages| imaginary
-// elements: [Re Im Re Im Re Im....]
-// - stages : Number of FFT stages. Must be at least 3 and at most
-// 10, since the table WebRtcSpl_kSinTable1024[] is 1024
-// elements long.
-//
-// Output:
-// - complex_data : The complex data buffer.
-
-void WebRtcSpl_ComplexBitReverse(int16_t* __restrict complex_data, int stages);
-
-// End: FFT operations
-
-/************************************************************
- *
- * RESAMPLING FUNCTIONS AND THEIR STRUCTS ARE DEFINED BELOW
- *
- ************************************************************/
-
-/*******************************************************************
- * resample.c
- *
- * Includes the following resampling combinations
- * 22 kHz -> 16 kHz
- * 16 kHz -> 22 kHz
- * 22 kHz -> 8 kHz
- * 8 kHz -> 22 kHz
- *
- ******************************************************************/
-
-// state structure for 22 -> 16 resampler
-typedef struct
-{
- WebRtc_Word32 S_22_44[8];
- WebRtc_Word32 S_44_32[8];
- WebRtc_Word32 S_32_16[8];
-} WebRtcSpl_State22khzTo16khz;
-
-void WebRtcSpl_Resample22khzTo16khz(const WebRtc_Word16* in,
- WebRtc_Word16* out,
- WebRtcSpl_State22khzTo16khz* state,
- WebRtc_Word32* tmpmem);
-
-void WebRtcSpl_ResetResample22khzTo16khz(WebRtcSpl_State22khzTo16khz* state);
-
-// state structure for 16 -> 22 resampler
-typedef struct
-{
- WebRtc_Word32 S_16_32[8];
- WebRtc_Word32 S_32_22[8];
-} WebRtcSpl_State16khzTo22khz;
-
-void WebRtcSpl_Resample16khzTo22khz(const WebRtc_Word16* in,
- WebRtc_Word16* out,
- WebRtcSpl_State16khzTo22khz* state,
- WebRtc_Word32* tmpmem);
-
-void WebRtcSpl_ResetResample16khzTo22khz(WebRtcSpl_State16khzTo22khz* state);
-
-// state structure for 22 -> 8 resampler
-typedef struct
-{
- WebRtc_Word32 S_22_22[16];
- WebRtc_Word32 S_22_16[8];
- WebRtc_Word32 S_16_8[8];
-} WebRtcSpl_State22khzTo8khz;
-
-void WebRtcSpl_Resample22khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
- WebRtcSpl_State22khzTo8khz* state,
- WebRtc_Word32* tmpmem);
-
-void WebRtcSpl_ResetResample22khzTo8khz(WebRtcSpl_State22khzTo8khz* state);
-
-// state structure for 8 -> 22 resampler
-typedef struct
-{
- WebRtc_Word32 S_8_16[8];
- WebRtc_Word32 S_16_11[8];
- WebRtc_Word32 S_11_22[8];
-} WebRtcSpl_State8khzTo22khz;
-
-void WebRtcSpl_Resample8khzTo22khz(const WebRtc_Word16* in, WebRtc_Word16* out,
- WebRtcSpl_State8khzTo22khz* state,
- WebRtc_Word32* tmpmem);
-
-void WebRtcSpl_ResetResample8khzTo22khz(WebRtcSpl_State8khzTo22khz* state);
-
-/*******************************************************************
- * resample_fractional.c
- * Functions for internal use in the other resample functions
- *
- * Includes the following resampling combinations
- * 48 kHz -> 32 kHz
- * 32 kHz -> 24 kHz
- * 44 kHz -> 32 kHz
- *
- ******************************************************************/
-
-void WebRtcSpl_Resample48khzTo32khz(const WebRtc_Word32* In, WebRtc_Word32* Out,
- const WebRtc_Word32 K);
-
-void WebRtcSpl_Resample32khzTo24khz(const WebRtc_Word32* In, WebRtc_Word32* Out,
- const WebRtc_Word32 K);
-
-void WebRtcSpl_Resample44khzTo32khz(const WebRtc_Word32* In, WebRtc_Word32* Out,
- const WebRtc_Word32 K);
-
-/*******************************************************************
- * resample_48khz.c
- *
- * Includes the following resampling combinations
- * 48 kHz -> 16 kHz
- * 16 kHz -> 48 kHz
- * 48 kHz -> 8 kHz
- * 8 kHz -> 48 kHz
- *
- ******************************************************************/
-
-typedef struct
-{
- WebRtc_Word32 S_48_48[16];
- WebRtc_Word32 S_48_32[8];
- WebRtc_Word32 S_32_16[8];
-} WebRtcSpl_State48khzTo16khz;
-
-void WebRtcSpl_Resample48khzTo16khz(const WebRtc_Word16* in, WebRtc_Word16* out,
- WebRtcSpl_State48khzTo16khz* state,
- WebRtc_Word32* tmpmem);
-
-void WebRtcSpl_ResetResample48khzTo16khz(WebRtcSpl_State48khzTo16khz* state);
-
-typedef struct
-{
- WebRtc_Word32 S_16_32[8];
- WebRtc_Word32 S_32_24[8];
- WebRtc_Word32 S_24_48[8];
-} WebRtcSpl_State16khzTo48khz;
-
-void WebRtcSpl_Resample16khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
- WebRtcSpl_State16khzTo48khz* state,
- WebRtc_Word32* tmpmem);
-
-void WebRtcSpl_ResetResample16khzTo48khz(WebRtcSpl_State16khzTo48khz* state);
-
-typedef struct
-{
- WebRtc_Word32 S_48_24[8];
- WebRtc_Word32 S_24_24[16];
- WebRtc_Word32 S_24_16[8];
- WebRtc_Word32 S_16_8[8];
-} WebRtcSpl_State48khzTo8khz;
-
-void WebRtcSpl_Resample48khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
- WebRtcSpl_State48khzTo8khz* state,
- WebRtc_Word32* tmpmem);
-
-void WebRtcSpl_ResetResample48khzTo8khz(WebRtcSpl_State48khzTo8khz* state);
-
-typedef struct
-{
- WebRtc_Word32 S_8_16[8];
- WebRtc_Word32 S_16_12[8];
- WebRtc_Word32 S_12_24[8];
- WebRtc_Word32 S_24_48[8];
-} WebRtcSpl_State8khzTo48khz;
-
-void WebRtcSpl_Resample8khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
- WebRtcSpl_State8khzTo48khz* state,
- WebRtc_Word32* tmpmem);
-
-void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state);
-
-/*******************************************************************
- * resample_by_2.c
- *
- * Includes down and up sampling by a factor of two.
- *
- ******************************************************************/
-
-void WebRtcSpl_DownsampleBy2(const WebRtc_Word16* in, const WebRtc_Word16 len,
- WebRtc_Word16* out, WebRtc_Word32* filtState);
-
-void WebRtcSpl_UpsampleBy2(const WebRtc_Word16* in, WebRtc_Word16 len, WebRtc_Word16* out,
- WebRtc_Word32* filtState);
-
-/************************************************************
- * END OF RESAMPLING FUNCTIONS
- ************************************************************/
-void WebRtcSpl_AnalysisQMF(const WebRtc_Word16* in_data,
- WebRtc_Word16* low_band,
- WebRtc_Word16* high_band,
- WebRtc_Word32* filter_state1,
- WebRtc_Word32* filter_state2);
-void WebRtcSpl_SynthesisQMF(const WebRtc_Word16* low_band,
- const WebRtc_Word16* high_band,
- WebRtc_Word16* out_data,
- WebRtc_Word32* filter_state1,
- WebRtc_Word32* filter_state2);
-
-#ifdef __cplusplus
-}
-#endif // __cplusplus
-#endif // WEBRTC_SPL_SIGNAL_PROCESSING_LIBRARY_H_
-
-//
-// WebRtcSpl_AddSatW16(...)
-// WebRtcSpl_AddSatW32(...)
-//
-// Returns the result of a saturated 16-bit, respectively 32-bit, addition of
-// the numbers specified by the |var1| and |var2| parameters.
-//
-// Input:
-// - var1 : Input variable 1
-// - var2 : Input variable 2
-//
-// Return value : Added and saturated value
-//
-
-//
-// WebRtcSpl_SubSatW16(...)
-// WebRtcSpl_SubSatW32(...)
-//
-// Returns the result of a saturated 16-bit, respectively 32-bit, subtraction
-// of the numbers specified by the |var1| and |var2| parameters.
-//
-// Input:
-// - var1 : Input variable 1
-// - var2 : Input variable 2
-//
-// Returned value : Subtracted and saturated value
-//
-
-//
-// WebRtcSpl_GetSizeInBits(...)
-//
-// Returns the # of bits that are needed at the most to represent the number
-// specified by the |value| parameter.
-//
-// Input:
-// - value : Input value
-//
-// Return value : Number of bits needed to represent |value|
-//
-
-//
-// WebRtcSpl_NormW32(...)
-//
-// Norm returns the # of left shifts required to 32-bit normalize the 32-bit
-// signed number specified by the |value| parameter.
-//
-// Input:
-// - value : Input value
-//
-// Return value : Number of bit shifts needed to 32-bit normalize |value|
-//
-
-//
-// WebRtcSpl_NormW16(...)
-//
-// Norm returns the # of left shifts required to 16-bit normalize the 16-bit
-// signed number specified by the |value| parameter.
-//
-// Input:
-// - value : Input value
-//
-// Return value : Number of bit shifts needed to 32-bit normalize |value|
-//
-
-//
-// WebRtcSpl_NormU32(...)
-//
-// Norm returns the # of left shifts required to 32-bit normalize the unsigned
-// 32-bit number specified by the |value| parameter.
-//
-// Input:
-// - value : Input value
-//
-// Return value : Number of bit shifts needed to 32-bit normalize |value|
-//
-
-//
-// WebRtcSpl_GetScalingSquare(...)
-//
-// Returns the # of bits required to scale the samples specified in the
-// |in_vector| parameter so that, if the squares of the samples are added the
-// # of times specified by the |times| parameter, the 32-bit addition will not
-// overflow (result in WebRtc_Word32).
-//
-// Input:
-// - in_vector : Input vector to check scaling on
-// - in_vector_length : Samples in |in_vector|
-// - times : Number of additions to be performed
-//
-// Return value : Number of right bit shifts needed to avoid
-// overflow in the addition calculation
-//
-
-//
-// WebRtcSpl_MemSetW16(...)
-//
-// Sets all the values in the WebRtc_Word16 vector |vector| of length
-// |vector_length| to the specified value |set_value|
-//
-// Input:
-// - vector : Pointer to the WebRtc_Word16 vector
-// - set_value : Value specified
-// - vector_length : Length of vector
-//
-
-//
-// WebRtcSpl_MemSetW32(...)
-//
-// Sets all the values in the WebRtc_Word32 vector |vector| of length
-// |vector_length| to the specified value |set_value|
-//
-// Input:
-// - vector : Pointer to the WebRtc_Word16 vector
-// - set_value : Value specified
-// - vector_length : Length of vector
-//
-
-//
-// WebRtcSpl_MemCpyReversedOrder(...)
-//
-// Copies all the values from the source WebRtc_Word16 vector |in_vector| to a
-// destination WebRtc_Word16 vector |out_vector|. It is done in reversed order,
-// meaning that the first sample of |in_vector| is copied to the last sample of
-// the |out_vector|. The procedure continues until the last sample of
-// |in_vector| has been copied to the first sample of |out_vector|. This
-// creates a reversed vector. Used in e.g. prediction in iLBC.
-//
-// Input:
-// - in_vector : Pointer to the first sample in a WebRtc_Word16 vector
-// of length |length|
-// - vector_length : Number of elements to copy
-//
-// Output:
-// - out_vector : Pointer to the last sample in a WebRtc_Word16 vector
-// of length |length|
-//
-
-//
-// WebRtcSpl_CopyFromEndW16(...)
-//
-// Copies the rightmost |samples| of |in_vector| (of length |in_vector_length|)
-// to the vector |out_vector|.
-//
-// Input:
-// - in_vector : Input vector
-// - in_vector_length : Number of samples in |in_vector|
-// - samples : Number of samples to extract (from right side)
-// from |in_vector|
-//
-// Output:
-// - out_vector : Vector with the requested samples
-//
-// Return value : Number of copied samples in |out_vector|
-//
-
-//
-// WebRtcSpl_ZerosArrayW16(...)
-// WebRtcSpl_ZerosArrayW32(...)
-//
-// Inserts the value "zero" in all positions of a w16 and a w32 vector
-// respectively.
-//
-// Input:
-// - vector_length : Number of samples in vector
-//
-// Output:
-// - vector : Vector containing all zeros
-//
-// Return value : Number of samples in vector
-//
-
-//
-// WebRtcSpl_OnesArrayW16(...)
-// WebRtcSpl_OnesArrayW32(...)
-//
-// Inserts the value "one" in all positions of a w16 and a w32 vector
-// respectively.
-//
-// Input:
-// - vector_length : Number of samples in vector
-//
-// Output:
-// - vector : Vector containing all ones
-//
-// Return value : Number of samples in vector
-//
-
-//
-// WebRtcSpl_MinValueW16(...)
-// WebRtcSpl_MinValueW32(...)
-//
-// Returns the minimum value of a vector
-//
-// Input:
-// - vector : Input vector
-// - vector_length : Number of samples in vector
-//
-// Return value : Minimum sample value in vector
-//
-
-//
-// WebRtcSpl_MaxValueW16(...)
-// WebRtcSpl_MaxValueW32(...)
-//
-// Returns the maximum value of a vector
-//
-// Input:
-// - vector : Input vector
-// - vector_length : Number of samples in vector
-//
-// Return value : Maximum sample value in vector
-//
-
-//
-// WebRtcSpl_MaxAbsValueW16(...)
-// WebRtcSpl_MaxAbsValueW32(...)
-//
-// Returns the largest absolute value of a vector
-//
-// Input:
-// - vector : Input vector
-// - vector_length : Number of samples in vector
-//
-// Return value : Maximum absolute value in vector
-//
-
-//
-// WebRtcSpl_MaxAbsIndexW16(...)
-//
-// Returns the vector index to the largest absolute value of a vector
-//
-// Input:
-// - vector : Input vector
-// - vector_length : Number of samples in vector
-//
-// Return value : Index to maximum absolute value in vector
-//
-
-//
-// WebRtcSpl_MinIndexW16(...)
-// WebRtcSpl_MinIndexW32(...)
-//
-// Returns the vector index to the minimum sample value of a vector
-//
-// Input:
-// - vector : Input vector
-// - vector_length : Number of samples in vector
-//
-// Return value : Index to minimum sample value in vector
-//
-
-//
-// WebRtcSpl_MaxIndexW16(...)
-// WebRtcSpl_MaxIndexW32(...)
-//
-// Returns the vector index to the maximum sample value of a vector
-//
-// Input:
-// - vector : Input vector
-// - vector_length : Number of samples in vector
-//
-// Return value : Index to maximum sample value in vector
-//
-
-//
-// WebRtcSpl_VectorBitShiftW16(...)
-// WebRtcSpl_VectorBitShiftW32(...)
-//
-// Bit shifts all the values in a vector up or downwards. Different calls for
-// WebRtc_Word16 and WebRtc_Word32 vectors respectively.
-//
-// Input:
-// - vector_length : Length of vector
-// - in_vector : Pointer to the vector that should be bit shifted
-// - right_shifts : Number of right bit shifts (negative value gives left
-// shifts)
-//
-// Output:
-// - out_vector : Pointer to the result vector (can be the same as
-// |in_vector|)
-//
-
-//
-// WebRtcSpl_VectorBitShiftW32ToW16(...)
-//
-// Bit shifts all the values in a WebRtc_Word32 vector up or downwards and
-// stores the result as a WebRtc_Word16 vector
-//
-// Input:
-// - vector_length : Length of vector
-// - in_vector : Pointer to the vector that should be bit shifted
-// - right_shifts : Number of right bit shifts (negative value gives left
-// shifts)
-//
-// Output:
-// - out_vector : Pointer to the result vector (can be the same as
-// |in_vector|)
-//
-
-//
-// WebRtcSpl_ScaleVector(...)
-//
-// Performs the vector operation:
-// out_vector[k] = (gain*in_vector[k])>>right_shifts
-//
-// Input:
-// - in_vector : Input vector
-// - gain : Scaling gain
-// - vector_length : Elements in the |in_vector|
-// - right_shifts : Number of right bit shifts applied
-//
-// Output:
-// - out_vector : Output vector (can be the same as |in_vector|)
-//
-
-//
-// WebRtcSpl_ScaleVectorWithSat(...)
-//
-// Performs the vector operation:
-// out_vector[k] = SATURATE( (gain*in_vector[k])>>right_shifts )
-//
-// Input:
-// - in_vector : Input vector
-// - gain : Scaling gain
-// - vector_length : Elements in the |in_vector|
-// - right_shifts : Number of right bit shifts applied
-//
-// Output:
-// - out_vector : Output vector (can be the same as |in_vector|)
-//
-
-//
-// WebRtcSpl_ScaleAndAddVectors(...)
-//
-// Performs the vector operation:
-// out_vector[k] = (gain1*in_vector1[k])>>right_shifts1
-// + (gain2*in_vector2[k])>>right_shifts2
-//
-// Input:
-// - in_vector1 : Input vector 1
-// - gain1 : Gain to be used for vector 1
-// - right_shifts1 : Right bit shift to be used for vector 1
-// - in_vector2 : Input vector 2
-// - gain2 : Gain to be used for vector 2
-// - right_shifts2 : Right bit shift to be used for vector 2
-// - vector_length : Elements in the input vectors
-//
-// Output:
-// - out_vector : Output vector
-//
-
-//
-// WebRtcSpl_ScaleAndAddVectorsWithRound(...)
-//
-// Performs the vector operation:
-//
-// out_vector[k] = ((scale1*in_vector1[k]) + (scale2*in_vector2[k])
-// + round_value) >> right_shifts
-//
-// where:
-//
-// round_value = (1<<right_shifts)>>1
-//
-// Input:
-// - in_vector1 : Input vector 1
-// - scale1 : Gain to be used for vector 1
-// - in_vector2 : Input vector 2
-// - scale2 : Gain to be used for vector 2
-// - right_shifts : Number of right bit shifts to be applied
-// - vector_length : Number of elements in the input vectors
-//
-// Output:
-// - out_vector : Output vector
-//
-
-//
-// WebRtcSpl_ReverseOrderMultArrayElements(...)
-//
-// Performs the vector operation:
-// out_vector[n] = (in_vector[n]*window[-n])>>right_shifts
-//
-// Input:
-// - in_vector : Input vector
-// - window : Window vector (should be reversed). The pointer
-// should be set to the last value in the vector
-// - right_shifts : Number of right bit shift to be applied after the
-// multiplication
-// - vector_length : Number of elements in |in_vector|
-//
-// Output:
-// - out_vector : Output vector (can be same as |in_vector|)
-//
-
-//
-// WebRtcSpl_ElementwiseVectorMult(...)
-//
-// Performs the vector operation:
-// out_vector[n] = (in_vector[n]*window[n])>>right_shifts
-//
-// Input:
-// - in_vector : Input vector
-// - window : Window vector.
-// - right_shifts : Number of right bit shift to be applied after the
-// multiplication
-// - vector_length : Number of elements in |in_vector|
-//
-// Output:
-// - out_vector : Output vector (can be same as |in_vector|)
-//
-
-//
-// WebRtcSpl_AddVectorsAndShift(...)
-//
-// Performs the vector operation:
-// out_vector[k] = (in_vector1[k] + in_vector2[k])>>right_shifts
-//
-// Input:
-// - in_vector1 : Input vector 1
-// - in_vector2 : Input vector 2
-// - right_shifts : Number of right bit shift to be applied after the
-// multiplication
-// - vector_length : Number of elements in |in_vector1| and |in_vector2|
-//
-// Output:
-// - out_vector : Output vector (can be same as |in_vector1|)
-//
-
-//
-// WebRtcSpl_AddAffineVectorToVector(...)
-//
-// Adds an affine transformed vector to another vector |out_vector|, i.e,
-// performs
-// out_vector[k] += (in_vector[k]*gain+add_constant)>>right_shifts
-//
-// Input:
-// - in_vector : Input vector
-// - gain : Gain value, used to multiply the in vector with
-// - add_constant : Constant value to add (usually 1<<(right_shifts-1),
-// but others can be used as well
-// - right_shifts : Number of right bit shifts (0-16)
-// - vector_length : Number of samples in |in_vector| and |out_vector|
-//
-// Output:
-// - out_vector : Vector with the output
-//
-
-//
-// WebRtcSpl_AffineTransformVector(...)
-//
-// Affine transforms a vector, i.e, performs
-// out_vector[k] = (in_vector[k]*gain+add_constant)>>right_shifts
-//
-// Input:
-// - in_vector : Input vector
-// - gain : Gain value, used to multiply the in vector with
-// - add_constant : Constant value to add (usually 1<<(right_shifts-1),
-// but others can be used as well
-// - right_shifts : Number of right bit shifts (0-16)
-// - vector_length : Number of samples in |in_vector| and |out_vector|
-//
-// Output:
-// - out_vector : Vector with the output
-//
-
-//
-// WebRtcSpl_AutoCorrelation(...)
-//
-// A 32-bit fix-point implementation of auto-correlation computation
-//
-// Input:
-// - vector : Vector to calculate autocorrelation upon
-// - vector_length : Length (in samples) of |vector|
-// - order : The order up to which the autocorrelation should be
-// calculated
-//
-// Output:
-// - result_vector : auto-correlation values (values should be seen
-// relative to each other since the absolute values
-// might have been down shifted to avoid overflow)
-//
-// - scale : The number of left shifts required to obtain the
-// auto-correlation in Q0
-//
-// Return value : Number of samples in |result_vector|, i.e., (order+1)
-//
-
-//
-// WebRtcSpl_LevinsonDurbin(...)
-//
-// A 32-bit fix-point implementation of the Levinson-Durbin algorithm that
-// does NOT use the 64 bit class
-//
-// Input:
-// - auto_corr : Vector with autocorrelation values of length >=
-// |use_order|+1
-// - use_order : The LPC filter order (support up to order 20)
-//
-// Output:
-// - lpc_coef : lpc_coef[0..use_order] LPC coefficients in Q12
-// - refl_coef : refl_coef[0...use_order-1]| Reflection coefficients in
-// Q15
-//
-// Return value : 1 for stable 0 for unstable
-//
-
-//
-// WebRtcSpl_ReflCoefToLpc(...)
-//
-// Converts reflection coefficients |refl_coef| to LPC coefficients |lpc_coef|.
-// This version is a 16 bit operation.
-//
-// NOTE: The 16 bit refl_coef -> lpc_coef conversion might result in a
-// "slightly unstable" filter (i.e., a pole just outside the unit circle) in
-// "rare" cases even if the reflection coefficients are stable.
-//
-// Input:
-// - refl_coef : Reflection coefficients in Q15 that should be converted
-// to LPC coefficients
-// - use_order : Number of coefficients in |refl_coef|
-//
-// Output:
-// - lpc_coef : LPC coefficients in Q12
-//
-
-//
-// WebRtcSpl_LpcToReflCoef(...)
-//
-// Converts LPC coefficients |lpc_coef| to reflection coefficients |refl_coef|.
-// This version is a 16 bit operation.
-// The conversion is implemented by the step-down algorithm.
-//
-// Input:
-// - lpc_coef : LPC coefficients in Q12, that should be converted to
-// reflection coefficients
-// - use_order : Number of coefficients in |lpc_coef|
-//
-// Output:
-// - refl_coef : Reflection coefficients in Q15.
-//
-
-//
-// WebRtcSpl_AutoCorrToReflCoef(...)
-//
-// Calculates reflection coefficients (16 bit) from auto-correlation values
-//
-// Input:
-// - auto_corr : Auto-correlation values
-// - use_order : Number of coefficients wanted be calculated
-//
-// Output:
-// - refl_coef : Reflection coefficients in Q15.
-//
-
-//
-// WebRtcSpl_CrossCorrelation(...)
-//
-// Calculates the cross-correlation between two sequences |vector1| and
-// |vector2|. |vector1| is fixed and |vector2| slides as the pointer is
-// increased with the amount |step_vector2|
-//
-// Input:
-// - vector1 : First sequence (fixed throughout the correlation)
-// - vector2 : Second sequence (slides |step_vector2| for each
-// new correlation)
-// - dim_vector : Number of samples to use in the cross-correlation
-// - dim_cross_corr : Number of cross-correlations to calculate (the
-// start position for |vector2| is updated for each
-// new one)
-// - right_shifts : Number of right bit shifts to use. This will
-// become the output Q-domain.
-// - step_vector2 : How many (positive or negative) steps the
-// |vector2| pointer should be updated for each new
-// cross-correlation value.
-//
-// Output:
-// - cross_corr : The cross-correlation in Q(-right_shifts)
-//
-
-//
-// WebRtcSpl_GetHanningWindow(...)
-//
-// Creates (the first half of) a Hanning window. Size must be at least 1 and
-// at most 512.
-//
-// Input:
-// - size : Length of the requested Hanning window (1 to 512)
-//
-// Output:
-// - window : Hanning vector in Q14.
-//
-
-//
-// WebRtcSpl_SqrtOfOneMinusXSquared(...)
-//
-// Calculates y[k] = sqrt(1 - x[k]^2) for each element of the input vector
-// |in_vector|. Input and output values are in Q15.
-//
-// Inputs:
-// - in_vector : Values to calculate sqrt(1 - x^2) of
-// - vector_length : Length of vector |in_vector|
-//
-// Output:
-// - out_vector : Output values in Q15
-//
-
-//
-// WebRtcSpl_IncreaseSeed(...)
-//
-// Increases the seed (and returns the new value)
-//
-// Input:
-// - seed : Seed for random calculation
-//
-// Output:
-// - seed : Updated seed value
-//
-// Return value : The new seed value
-//
-
-//
-// WebRtcSpl_RandU(...)
-//
-// Produces a uniformly distributed value in the WebRtc_Word16 range
-//
-// Input:
-// - seed : Seed for random calculation
-//
-// Output:
-// - seed : Updated seed value
-//
-// Return value : Uniformly distributed value in the range
-// [Word16_MIN...Word16_MAX]
-//
-
-//
-// WebRtcSpl_RandN(...)
-//
-// Produces a normal distributed value in the WebRtc_Word16 range
-//
-// Input:
-// - seed : Seed for random calculation
-//
-// Output:
-// - seed : Updated seed value
-//
-// Return value : N(0,1) value in the Q13 domain
-//
-
-//
-// WebRtcSpl_RandUArray(...)
-//
-// Produces a uniformly distributed vector with elements in the WebRtc_Word16
-// range
-//
-// Input:
-// - vector_length : Samples wanted in the vector
-// - seed : Seed for random calculation
-//
-// Output:
-// - vector : Vector with the uniform values
-// - seed : Updated seed value
-//
-// Return value : Number of samples in vector, i.e., |vector_length|
-//
-
-//
-// WebRtcSpl_Sqrt(...)
-//
-// Returns the square root of the input value |value|. The precision of this
-// function is integer precision, i.e., sqrt(8) gives 2 as answer.
-// If |value| is a negative number then 0 is returned.
-//
-// Algorithm:
-//
-// A sixth order Taylor Series expansion is used here to compute the square
-// root of a number y^0.5 = (1+x)^0.5
-// where
-// x = y-1
-// = 1+(x/2)-0.5*((x/2)^2+0.5*((x/2)^3-0.625*((x/2)^4+0.875*((x/2)^5)
-// 0.5 <= x < 1
-//
-// Input:
-// - value : Value to calculate sqrt of
-//
-// Return value : Result of the sqrt calculation
-//
-
-//
-// WebRtcSpl_SqrtFloor(...)
-//
-// Returns the square root of the input value |value|. The precision of this
-// function is rounding down integer precision, i.e., sqrt(8) gives 2 as answer.
-// If |value| is a negative number then 0 is returned.
-//
-// Algorithm:
-//
-// An iterative 4 cylce/bit routine
-//
-// Input:
-// - value : Value to calculate sqrt of
-//
-// Return value : Result of the sqrt calculation
-//
-
-//
-// WebRtcSpl_DivU32U16(...)
-//
-// Divides a WebRtc_UWord32 |num| by a WebRtc_UWord16 |den|.
-//
-// If |den|==0, (WebRtc_UWord32)0xFFFFFFFF is returned.
-//
-// Input:
-// - num : Numerator
-// - den : Denominator
-//
-// Return value : Result of the division (as a WebRtc_UWord32), i.e., the
-// integer part of num/den.
-//
-
-//
-// WebRtcSpl_DivW32W16(...)
-//
-// Divides a WebRtc_Word32 |num| by a WebRtc_Word16 |den|.
-//
-// If |den|==0, (WebRtc_Word32)0x7FFFFFFF is returned.
-//
-// Input:
-// - num : Numerator
-// - den : Denominator
-//
-// Return value : Result of the division (as a WebRtc_Word32), i.e., the
-// integer part of num/den.
-//
-
-//
-// WebRtcSpl_DivW32W16ResW16(...)
-//
-// Divides a WebRtc_Word32 |num| by a WebRtc_Word16 |den|, assuming that the
-// result is less than 32768, otherwise an unpredictable result will occur.
-//
-// If |den|==0, (WebRtc_Word16)0x7FFF is returned.
-//
-// Input:
-// - num : Numerator
-// - den : Denominator
-//
-// Return value : Result of the division (as a WebRtc_Word16), i.e., the
-// integer part of num/den.
-//
-
-//
-// WebRtcSpl_DivResultInQ31(...)
-//
-// Divides a WebRtc_Word32 |num| by a WebRtc_Word16 |den|, assuming that the
-// absolute value of the denominator is larger than the numerator, otherwise
-// an unpredictable result will occur.
-//
-// Input:
-// - num : Numerator
-// - den : Denominator
-//
-// Return value : Result of the division in Q31.
-//
-
-//
-// WebRtcSpl_DivW32HiLow(...)
-//
-// Divides a WebRtc_Word32 |num| by a denominator in hi, low format. The
-// absolute value of the denominator has to be larger (or equal to) the
-// numerator.
-//
-// Input:
-// - num : Numerator
-// - den_hi : High part of denominator
-// - den_low : Low part of denominator
-//
-// Return value : Divided value in Q31
-//
-
-//
-// WebRtcSpl_Energy(...)
-//
-// Calculates the energy of a vector
-//
-// Input:
-// - vector : Vector which the energy should be calculated on
-// - vector_length : Number of samples in vector
-//
-// Output:
-// - scale_factor : Number of left bit shifts needed to get the physical
-// energy value, i.e, to get the Q0 value
-//
-// Return value : Energy value in Q(-|scale_factor|)
-//
-
-//
-// WebRtcSpl_FilterAR(...)
-//
-// Performs a 32-bit AR filtering on a vector in Q12
-//
-// Input:
-// - ar_coef : AR-coefficient vector (values in Q12),
-// ar_coef[0] must be 4096.
-// - ar_coef_length : Number of coefficients in |ar_coef|.
-// - in_vector : Vector to be filtered.
-// - in_vector_length : Number of samples in |in_vector|.
-// - filter_state : Current state (higher part) of the filter.
-// - filter_state_length : Length (in samples) of |filter_state|.
-// - filter_state_low : Current state (lower part) of the filter.
-// - filter_state_low_length : Length (in samples) of |filter_state_low|.
-// - out_vector_low_length : Maximum length (in samples) of
-// |out_vector_low|.
-//
-// Output:
-// - filter_state : Updated state (upper part) vector.
-// - filter_state_low : Updated state (lower part) vector.
-// - out_vector : Vector containing the upper part of the
-// filtered values.
-// - out_vector_low : Vector containing the lower part of the
-// filtered values.
-//
-// Return value : Number of samples in the |out_vector|.
-//
-
-//
-// WebRtcSpl_FilterMAFastQ12(...)
-//
-// Performs a MA filtering on a vector in Q12
-//
-// Input:
-// - in_vector : Input samples (state in positions
-// in_vector[-order] .. in_vector[-1])
-// - ma_coef : Filter coefficients (in Q12)
-// - ma_coef_length : Number of B coefficients (order+1)
-// - vector_length : Number of samples to be filtered
-//
-// Output:
-// - out_vector : Filtered samples
-//
-
-
-//
-// WebRtcSpl_DotProductWithScale(...)
-//
-// Calculates the dot product between two (WebRtc_Word16) vectors
-//
-// Input:
-// - vector1 : Vector 1
-// - vector2 : Vector 2
-// - vector_length : Number of samples used in the dot product
-// - scaling : The number of right bit shifts to apply on each term
-// during calculation to avoid overflow, i.e., the
-// output will be in Q(-|scaling|)
-//
-// Return value : The dot product in Q(-scaling)
-//
-
-//
-// WebRtcSpl_ComplexIFFT(...)
-//
-// Complex Inverse FFT
-//
-// Computes an inverse complex 2^|stages|-point FFT on the input vector, which
-// is in bit-reversed order. The original content of the vector is destroyed in
-// the process, since the input is overwritten by the output, normal-ordered,
-// FFT vector. With X as the input complex vector, y as the output complex
-// vector and with M = 2^|stages|, the following is computed:
-//
-// M-1
-// y(k) = sum[X(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
-// i=0
-//
-// The implementations are optimized for speed, not for code size. It uses the
-// decimation-in-time algorithm with radix-2 butterfly technique.
-//
-// Input:
-// - vector : In pointer to complex vector containing 2^|stages|
-// real elements interleaved with 2^|stages| imaginary
-// elements.
-// [ReImReImReIm....]
-// The elements are in Q(-scale) domain, see more on Return
-// Value below.
-//
-// - stages : Number of FFT stages. Must be at least 3 and at most 10,
-// since the table WebRtcSpl_kSinTable1024[] is 1024
-// elements long.
-//
-// - mode : This parameter gives the user to choose how the FFT
-// should work.
-// mode==0: Low-complexity and Low-accuracy mode
-// mode==1: High-complexity and High-accuracy mode
-//
-// Output:
-// - vector : Out pointer to the FFT vector (the same as input).
-//
-// Return Value : The scale value that tells the number of left bit shifts
-// that the elements in the |vector| should be shifted with
-// in order to get Q0 values, i.e. the physically correct
-// values. The scale parameter is always 0 or positive,
-// except if N>1024 (|stages|>10), which returns a scale
-// value of -1, indicating error.
-//
-
-//
-// WebRtcSpl_ComplexFFT(...)
-//
-// Complex FFT
-//
-// Computes a complex 2^|stages|-point FFT on the input vector, which is in
-// bit-reversed order. The original content of the vector is destroyed in
-// the process, since the input is overwritten by the output, normal-ordered,
-// FFT vector. With x as the input complex vector, Y as the output complex
-// vector and with M = 2^|stages|, the following is computed:
-//
-// M-1
-// Y(k) = 1/M * sum[x(i)*[cos(2*pi*i*k/M) + j*sin(2*pi*i*k/M)]]
-// i=0
-//
-// The implementations are optimized for speed, not for code size. It uses the
-// decimation-in-time algorithm with radix-2 butterfly technique.
-//
-// This routine prevents overflow by scaling by 2 before each FFT stage. This is
-// a fixed scaling, for proper normalization - there will be log2(n) passes, so
-// this results in an overall factor of 1/n, distributed to maximize arithmetic
-// accuracy.
-//
-// Input:
-// - vector : In pointer to complex vector containing 2^|stages| real
-// elements interleaved with 2^|stages| imaginary elements.
-// [ReImReImReIm....]
-// The output is in the Q0 domain.
-//
-// - stages : Number of FFT stages. Must be at least 3 and at most 10,
-// since the table WebRtcSpl_kSinTable1024[] is 1024
-// elements long.
-//
-// - mode : This parameter gives the user to choose how the FFT
-// should work.
-// mode==0: Low-complexity and Low-accuracy mode
-// mode==1: High-complexity and High-accuracy mode
-//
-// Output:
-// - vector : The output FFT vector is in the Q0 domain.
-//
-// Return value : The scale parameter is always 0, except if N>1024,
-// which returns a scale value of -1, indicating error.
-//
-
-//
-// WebRtcSpl_AnalysisQMF(...)
-//
-// Splits a 0-2*F Hz signal into two sub bands: 0-F Hz and F-2*F Hz. The
-// current version has F = 8000, therefore, a super-wideband audio signal is
-// split to lower-band 0-8 kHz and upper-band 8-16 kHz.
-//
-// Input:
-// - in_data : Wide band speech signal, 320 samples (10 ms)
-//
-// Input & Output:
-// - filter_state1 : Filter state for first All-pass filter
-// - filter_state2 : Filter state for second All-pass filter
-//
-// Output:
-// - low_band : Lower-band signal 0-8 kHz band, 160 samples (10 ms)
-// - high_band : Upper-band signal 8-16 kHz band (flipped in frequency
-// domain), 160 samples (10 ms)
-//
-
-//
-// WebRtcSpl_SynthesisQMF(...)
-//
-// Combines the two sub bands (0-F and F-2*F Hz) into a signal of 0-2*F
-// Hz, (current version has F = 8000 Hz). So the filter combines lower-band
-// (0-8 kHz) and upper-band (8-16 kHz) channels to obtain super-wideband 0-16
-// kHz audio.
-//
-// Input:
-// - low_band : The signal with the 0-8 kHz band, 160 samples (10 ms)
-// - high_band : The signal with the 8-16 kHz band, 160 samples (10 ms)
-//
-// Input & Output:
-// - filter_state1 : Filter state for first All-pass filter
-// - filter_state2 : Filter state for second All-pass filter
-//
-// Output:
-// - out_data : Super-wideband speech signal, 0-16 kHz
-//
-
-// WebRtc_Word16 WebRtcSpl_SatW32ToW16(...)
-//
-// This function saturates a 32-bit word into a 16-bit word.
-//
-// Input:
-// - value32 : The value of a 32-bit word.
-//
-// Output:
-// - out16 : the saturated 16-bit word.
-//
-
-// int32_t WebRtc_MulAccumW16(...)
-//
-// This function multiply a 16-bit word by a 16-bit word, and accumulate this
-// value to a 32-bit integer.
-//
-// Input:
-// - a : The value of the first 16-bit word.
-// - b : The value of the second 16-bit word.
-// - c : The value of an 32-bit integer.
-//
-// Return Value: The value of a * b + c.
-//
-
-// WebRtc_Word16 WebRtcSpl_get_version(...)
-//
-// This function gives the version string of the Signal Processing Library.
-//
-// Input:
-// - length_in_bytes : The size of Allocated space (in Bytes) where
-// the version number is written to (in string format).
-//
-// Output:
-// - version : Pointer to a buffer where the version number is written to.
-//
diff --git a/trunk/src/common_audio/signal_processing/include/spl_inl.h b/trunk/src/common_audio/signal_processing/include/spl_inl.h
deleted file mode 100644
index 23b3209..0000000
--- a/trunk/src/common_audio/signal_processing/include/spl_inl.h
+++ /dev/null
@@ -1,159 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-// This header file includes the inline functions in
-// the fix point signal processing library.
-
-#ifndef WEBRTC_SPL_SPL_INL_H_
-#define WEBRTC_SPL_SPL_INL_H_
-
-#ifdef WEBRTC_ARCH_ARM_V7A
-#include "spl_inl_armv7.h"
-#else
-
-static __inline WebRtc_Word16 WebRtcSpl_SatW32ToW16(WebRtc_Word32 value32) {
- WebRtc_Word16 out16 = (WebRtc_Word16) value32;
-
- if (value32 > 32767)
- out16 = 32767;
- else if (value32 < -32768)
- out16 = -32768;
-
- return out16;
-}
-
-static __inline WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
- WebRtc_Word16 b) {
- return WebRtcSpl_SatW32ToW16((WebRtc_Word32) a + (WebRtc_Word32) b);
-}
-
-static __inline WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 l_var1,
- WebRtc_Word32 l_var2) {
- WebRtc_Word32 l_sum;
-
- // perform long addition
- l_sum = l_var1 + l_var2;
-
- // check for under or overflow
- if (WEBRTC_SPL_IS_NEG(l_var1)) {
- if (WEBRTC_SPL_IS_NEG(l_var2) && !WEBRTC_SPL_IS_NEG(l_sum)) {
- l_sum = (WebRtc_Word32)0x80000000;
- }
- } else {
- if (!WEBRTC_SPL_IS_NEG(l_var2) && WEBRTC_SPL_IS_NEG(l_sum)) {
- l_sum = (WebRtc_Word32)0x7FFFFFFF;
- }
- }
-
- return l_sum;
-}
-
-static __inline WebRtc_Word16 WebRtcSpl_SubSatW16(WebRtc_Word16 var1,
- WebRtc_Word16 var2) {
- return WebRtcSpl_SatW32ToW16((WebRtc_Word32) var1 - (WebRtc_Word32) var2);
-}
-
-static __inline WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 l_var1,
- WebRtc_Word32 l_var2) {
- WebRtc_Word32 l_diff;
-
- // perform subtraction
- l_diff = l_var1 - l_var2;
-
- // check for underflow
- if ((l_var1 < 0) && (l_var2 > 0) && (l_diff > 0))
- l_diff = (WebRtc_Word32)0x80000000;
- // check for overflow
- if ((l_var1 > 0) && (l_var2 < 0) && (l_diff < 0))
- l_diff = (WebRtc_Word32)0x7FFFFFFF;
-
- return l_diff;
-}
-
-static __inline WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 n) {
- int bits;
-
- if (0xFFFF0000 & n) {
- bits = 16;
- } else {
- bits = 0;
- }
- if (0x0000FF00 & (n >> bits)) bits += 8;
- if (0x000000F0 & (n >> bits)) bits += 4;
- if (0x0000000C & (n >> bits)) bits += 2;
- if (0x00000002 & (n >> bits)) bits += 1;
- if (0x00000001 & (n >> bits)) bits += 1;
-
- return bits;
-}
-
-static __inline int WebRtcSpl_NormW32(WebRtc_Word32 a) {
- int zeros;
-
- if (a <= 0) a ^= 0xFFFFFFFF;
-
- if (!(0xFFFF8000 & a)) {
- zeros = 16;
- } else {
- zeros = 0;
- }
- if (!(0xFF800000 & (a << zeros))) zeros += 8;
- if (!(0xF8000000 & (a << zeros))) zeros += 4;
- if (!(0xE0000000 & (a << zeros))) zeros += 2;
- if (!(0xC0000000 & (a << zeros))) zeros += 1;
-
- return zeros;
-}
-
-static __inline int WebRtcSpl_NormU32(WebRtc_UWord32 a) {
- int zeros;
-
- if (a == 0) return 0;
-
- if (!(0xFFFF0000 & a)) {
- zeros = 16;
- } else {
- zeros = 0;
- }
- if (!(0xFF000000 & (a << zeros))) zeros += 8;
- if (!(0xF0000000 & (a << zeros))) zeros += 4;
- if (!(0xC0000000 & (a << zeros))) zeros += 2;
- if (!(0x80000000 & (a << zeros))) zeros += 1;
-
- return zeros;
-}
-
-static __inline int WebRtcSpl_NormW16(WebRtc_Word16 a) {
- int zeros;
-
- if (a <= 0) a ^= 0xFFFF;
-
- if (!(0xFF80 & a)) {
- zeros = 8;
- } else {
- zeros = 0;
- }
- if (!(0xF800 & (a << zeros))) zeros += 4;
- if (!(0xE000 & (a << zeros))) zeros += 2;
- if (!(0xC000 & (a << zeros))) zeros += 1;
-
- return zeros;
-}
-
-static __inline int32_t WebRtc_MulAccumW16(int16_t a,
- int16_t b,
- int32_t c) {
- return (a * b + c);
-}
-
-#endif // WEBRTC_ARCH_ARM_V7A
-
-#endif // WEBRTC_SPL_SPL_INL_H_
diff --git a/trunk/src/common_audio/signal_processing/include/spl_inl_armv7.h b/trunk/src/common_audio/signal_processing/include/spl_inl_armv7.h
deleted file mode 100644
index 5b19c2c..0000000
--- a/trunk/src/common_audio/signal_processing/include/spl_inl_armv7.h
+++ /dev/null
@@ -1,140 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-// This header file includes the inline functions for ARM processors in
-// the fix point signal processing library.
-
-#ifndef WEBRTC_SPL_SPL_INL_ARMV7_H_
-#define WEBRTC_SPL_SPL_INL_ARMV7_H_
-
-// TODO(kma): Replace some assembly code with GCC intrinsics
-// (e.g. __builtin_clz).
-
-static __inline WebRtc_Word32 WEBRTC_SPL_MUL_16_32_RSFT16(WebRtc_Word16 a,
- WebRtc_Word32 b) {
- WebRtc_Word32 tmp;
- __asm__("smulwb %0, %1, %2":"=r"(tmp):"r"(b), "r"(a));
- return tmp;
-}
-
-static __inline WebRtc_Word32 WEBRTC_SPL_MUL_32_32_RSFT32(WebRtc_Word16 a,
- WebRtc_Word16 b,
- WebRtc_Word32 c) {
- WebRtc_Word32 tmp;
- __asm__("pkhbt %0, %1, %2, lsl #16" : "=r"(tmp) : "r"(b), "r"(a));
- __asm__("smmul %0, %1, %2":"=r"(tmp):"r"(tmp), "r"(c));
- return tmp;
-}
-
-static __inline WebRtc_Word32 WEBRTC_SPL_MUL_32_32_RSFT32BI(WebRtc_Word32 a,
- WebRtc_Word32 b) {
- WebRtc_Word32 tmp;
- __asm__("smmul %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
- return tmp;
-}
-
-static __inline WebRtc_Word32 WEBRTC_SPL_MUL_16_16(WebRtc_Word16 a,
- WebRtc_Word16 b) {
- WebRtc_Word32 tmp;
- __asm__("smulbb %0, %1, %2":"=r"(tmp):"r"(a), "r"(b));
- return tmp;
-}
-
-static __inline int32_t WebRtc_MulAccumW16(int16_t a,
- int16_t b,
- int32_t c) {
- int32_t tmp = 0;
- __asm__("smlabb %0, %1, %2, %3":"=r"(tmp):"r"(a), "r"(b), "r"(c));
- return tmp;
-}
-
-static __inline WebRtc_Word16 WebRtcSpl_AddSatW16(WebRtc_Word16 a,
- WebRtc_Word16 b) {
- WebRtc_Word32 s_sum;
-
- __asm__("qadd16 %0, %1, %2":"=r"(s_sum):"r"(a), "r"(b));
-
- return (WebRtc_Word16) s_sum;
-}
-
-static __inline WebRtc_Word32 WebRtcSpl_AddSatW32(WebRtc_Word32 l_var1,
- WebRtc_Word32 l_var2) {
- WebRtc_Word32 l_sum;
-
- __asm__("qadd %0, %1, %2":"=r"(l_sum):"r"(l_var1), "r"(l_var2));
-
- return l_sum;
-}
-
-static __inline WebRtc_Word16 WebRtcSpl_SubSatW16(WebRtc_Word16 var1,
- WebRtc_Word16 var2) {
- WebRtc_Word32 s_sub;
-
- __asm__("qsub16 %0, %1, %2":"=r"(s_sub):"r"(var1), "r"(var2));
-
- return (WebRtc_Word16)s_sub;
-}
-
-static __inline WebRtc_Word32 WebRtcSpl_SubSatW32(WebRtc_Word32 l_var1,
- WebRtc_Word32 l_var2) {
- WebRtc_Word32 l_sub;
-
- __asm__("qsub %0, %1, %2":"=r"(l_sub):"r"(l_var1), "r"(l_var2));
-
- return l_sub;
-}
-
-static __inline WebRtc_Word16 WebRtcSpl_GetSizeInBits(WebRtc_UWord32 n) {
- WebRtc_Word32 tmp;
-
- __asm__("clz %0, %1":"=r"(tmp):"r"(n));
-
- return (WebRtc_Word16)(32 - tmp);
-}
-
-static __inline int WebRtcSpl_NormW32(WebRtc_Word32 a) {
- WebRtc_Word32 tmp;
-
- if (a <= 0) a ^= 0xFFFFFFFF;
-
- __asm__("clz %0, %1":"=r"(tmp):"r"(a));
-
- return tmp - 1;
-}
-
-static __inline int WebRtcSpl_NormU32(WebRtc_UWord32 a) {
- int tmp;
-
- if (a == 0) return 0;
-
- __asm__("clz %0, %1":"=r"(tmp):"r"(a));
-
- return tmp;
-}
-
-static __inline int WebRtcSpl_NormW16(WebRtc_Word16 a) {
- WebRtc_Word32 tmp;
-
- if (a <= 0) a ^= 0xFFFFFFFF;
-
- __asm__("clz %0, %1":"=r"(tmp):"r"(a));
-
- return tmp - 17;
-}
-
-static __inline WebRtc_Word16 WebRtcSpl_SatW32ToW16(WebRtc_Word32 value32) {
- WebRtc_Word16 out16;
-
- __asm__("ssat %r0, #16, %r1" : "=r"(out16) : "r"(value32));
-
- return out16;
-}
-#endif // WEBRTC_SPL_SPL_INL_ARMV7_H_
diff --git a/trunk/src/common_audio/signal_processing/levinson_durbin.c b/trunk/src/common_audio/signal_processing/levinson_durbin.c
deleted file mode 100644
index 4e11cdb..0000000
--- a/trunk/src/common_audio/signal_processing/levinson_durbin.c
+++ /dev/null
@@ -1,259 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_LevinsonDurbin().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#define SPL_LEVINSON_MAXORDER 20
-
-WebRtc_Word16 WebRtcSpl_LevinsonDurbin(WebRtc_Word32 *R, WebRtc_Word16 *A, WebRtc_Word16 *K,
- WebRtc_Word16 order)
-{
- WebRtc_Word16 i, j;
- // Auto-correlation coefficients in high precision
- WebRtc_Word16 R_hi[SPL_LEVINSON_MAXORDER + 1], R_low[SPL_LEVINSON_MAXORDER + 1];
- // LPC coefficients in high precision
- WebRtc_Word16 A_hi[SPL_LEVINSON_MAXORDER + 1], A_low[SPL_LEVINSON_MAXORDER + 1];
- // LPC coefficients for next iteration
- WebRtc_Word16 A_upd_hi[SPL_LEVINSON_MAXORDER + 1], A_upd_low[SPL_LEVINSON_MAXORDER + 1];
- // Reflection coefficient in high precision
- WebRtc_Word16 K_hi, K_low;
- // Prediction gain Alpha in high precision and with scale factor
- WebRtc_Word16 Alpha_hi, Alpha_low, Alpha_exp;
- WebRtc_Word16 tmp_hi, tmp_low;
- WebRtc_Word32 temp1W32, temp2W32, temp3W32;
- WebRtc_Word16 norm;
-
- // Normalize the autocorrelation R[0]...R[order+1]
-
- norm = WebRtcSpl_NormW32(R[0]);
-
- for (i = order; i >= 0; i--)
- {
- temp1W32 = WEBRTC_SPL_LSHIFT_W32(R[i], norm);
- // Put R in hi and low format
- R_hi[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
- R_low[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
- - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16)), 1);
- }
-
- // K = A[1] = -R[1] / R[0]
-
- temp2W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[1],16)
- + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[1],1); // R[1] in Q31
- temp3W32 = WEBRTC_SPL_ABS_W32(temp2W32); // abs R[1]
- temp1W32 = WebRtcSpl_DivW32HiLow(temp3W32, R_hi[0], R_low[0]); // abs(R[1])/R[0] in Q31
- // Put back the sign on R[1]
- if (temp2W32 > 0)
- {
- temp1W32 = -temp1W32;
- }
-
- // Put K in hi and low format
- K_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
- K_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
- - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1);
-
- // Store first reflection coefficient
- K[0] = K_hi;
-
- temp1W32 = WEBRTC_SPL_RSHIFT_W32(temp1W32, 4); // A[1] in Q27
-
- // Put A[1] in hi and low format
- A_hi[1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
- A_low[1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
- - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[1], 16)), 1);
-
- // Alpha = R[0] * (1-K^2)
-
- temp1W32 = (((WEBRTC_SPL_MUL_16_16(K_hi, K_low) >> 14) + WEBRTC_SPL_MUL_16_16(K_hi, K_hi))
- << 1); // temp1W32 = k^2 in Q31
-
- temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
- temp1W32 = (WebRtc_Word32)0x7fffffffL - temp1W32; // temp1W32 = (1 - K[0]*K[0]) in Q31
-
- // Store temp1W32 = 1 - K[0]*K[0] on hi and low format
- tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
- tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
- - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
-
- // Calculate Alpha in Q31
- temp1W32 = ((WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_hi)
- + (WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_low) >> 15)
- + (WEBRTC_SPL_MUL_16_16(R_low[0], tmp_hi) >> 15)) << 1);
-
- // Normalize Alpha and put it in hi and low format
-
- Alpha_exp = WebRtcSpl_NormW32(temp1W32);
- temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, Alpha_exp);
- Alpha_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
- Alpha_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
- - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1);
-
- // Perform the iterative calculations in the Levinson-Durbin algorithm
-
- for (i = 2; i <= order; i++)
- {
- /* ----
- temp1W32 = R[i] + > R[j]*A[i-j]
- /
- ----
- j=1..i-1
- */
-
- temp1W32 = 0;
-
- for (j = 1; j < i; j++)
- {
- // temp1W32 is in Q31
- temp1W32 += ((WEBRTC_SPL_MUL_16_16(R_hi[j], A_hi[i-j]) << 1)
- + (((WEBRTC_SPL_MUL_16_16(R_hi[j], A_low[i-j]) >> 15)
- + (WEBRTC_SPL_MUL_16_16(R_low[j], A_hi[i-j]) >> 15)) << 1));
- }
-
- temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, 4);
- temp1W32 += (WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16)
- + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[i], 1));
-
- // K = -temp1W32 / Alpha
- temp2W32 = WEBRTC_SPL_ABS_W32(temp1W32); // abs(temp1W32)
- temp3W32 = WebRtcSpl_DivW32HiLow(temp2W32, Alpha_hi, Alpha_low); // abs(temp1W32)/Alpha
-
- // Put the sign of temp1W32 back again
- if (temp1W32 > 0)
- {
- temp3W32 = -temp3W32;
- }
-
- // Use the Alpha shifts from earlier to de-normalize
- norm = WebRtcSpl_NormW32(temp3W32);
- if ((Alpha_exp <= norm) || (temp3W32 == 0))
- {
- temp3W32 = WEBRTC_SPL_LSHIFT_W32(temp3W32, Alpha_exp);
- } else
- {
- if (temp3W32 > 0)
- {
- temp3W32 = (WebRtc_Word32)0x7fffffffL;
- } else
- {
- temp3W32 = (WebRtc_Word32)0x80000000L;
- }
- }
-
- // Put K on hi and low format
- K_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
- K_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp3W32
- - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1);
-
- // Store Reflection coefficient in Q15
- K[i - 1] = K_hi;
-
- // Test for unstable filter.
- // If unstable return 0 and let the user decide what to do in that case
-
- if ((WebRtc_Word32)WEBRTC_SPL_ABS_W16(K_hi) > (WebRtc_Word32)32750)
- {
- return 0; // Unstable filter
- }
-
- /*
- Compute updated LPC coefficient: Anew[i]
- Anew[j]= A[j] + K*A[i-j] for j=1..i-1
- Anew[i]= K
- */
-
- for (j = 1; j < i; j++)
- {
- // temp1W32 = A[j] in Q27
- temp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[j],16)
- + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_low[j],1);
-
- // temp1W32 += K*A[i-j] in Q27
- temp1W32 += ((WEBRTC_SPL_MUL_16_16(K_hi, A_hi[i-j])
- + (WEBRTC_SPL_MUL_16_16(K_hi, A_low[i-j]) >> 15)
- + (WEBRTC_SPL_MUL_16_16(K_low, A_hi[i-j]) >> 15)) << 1);
-
- // Put Anew in hi and low format
- A_upd_hi[j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
- A_upd_low[j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
- - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_upd_hi[j], 16)), 1);
- }
-
- // temp3W32 = K in Q27 (Convert from Q31 to Q27)
- temp3W32 = WEBRTC_SPL_RSHIFT_W32(temp3W32, 4);
-
- // Store Anew in hi and low format
- A_upd_hi[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
- A_upd_low[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp3W32
- - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_upd_hi[i], 16)), 1);
-
- // Alpha = Alpha * (1-K^2)
-
- temp1W32 = (((WEBRTC_SPL_MUL_16_16(K_hi, K_low) >> 14)
- + WEBRTC_SPL_MUL_16_16(K_hi, K_hi)) << 1); // K*K in Q31
-
- temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); // Guard against <0
- temp1W32 = (WebRtc_Word32)0x7fffffffL - temp1W32; // 1 - K*K in Q31
-
- // Convert 1- K^2 in hi and low format
- tmp_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
- tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
- - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
-
- // Calculate Alpha = Alpha * (1-K^2) in Q31
- temp1W32 = ((WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_hi)
- + (WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_low) >> 15)
- + (WEBRTC_SPL_MUL_16_16(Alpha_low, tmp_hi) >> 15)) << 1);
-
- // Normalize Alpha and store it on hi and low format
-
- norm = WebRtcSpl_NormW32(temp1W32);
- temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, norm);
-
- Alpha_hi = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
- Alpha_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32
- - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1);
-
- // Update the total normalization of Alpha
- Alpha_exp = Alpha_exp + norm;
-
- // Update A[]
-
- for (j = 1; j <= i; j++)
- {
- A_hi[j] = A_upd_hi[j];
- A_low[j] = A_upd_low[j];
- }
- }
-
- /*
- Set A[0] to 1.0 and store the A[i] i=1...order in Q12
- (Convert from Q27 and use rounding)
- */
-
- A[0] = 4096;
-
- for (i = 1; i <= order; i++)
- {
- // temp1W32 in Q27
- temp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[i], 16)
- + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_low[i], 1);
- // Round and store upper word
- A[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32<<1)+(WebRtc_Word32)32768, 16);
- }
- return 1; // Stable filters
-}
diff --git a/trunk/src/common_audio/signal_processing/lpc_to_refl_coef.c b/trunk/src/common_audio/signal_processing/lpc_to_refl_coef.c
deleted file mode 100644
index 2cb83c2..0000000
--- a/trunk/src/common_audio/signal_processing/lpc_to_refl_coef.c
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_LpcToReflCoef().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#define SPL_LPC_TO_REFL_COEF_MAX_AR_MODEL_ORDER 50
-
-void WebRtcSpl_LpcToReflCoef(WebRtc_Word16* a16, int use_order, WebRtc_Word16* k16)
-{
- int m, k;
- WebRtc_Word32 tmp32[SPL_LPC_TO_REFL_COEF_MAX_AR_MODEL_ORDER];
- WebRtc_Word32 tmp_inv_denom32;
- WebRtc_Word16 tmp_inv_denom16;
-
- k16[use_order - 1] = WEBRTC_SPL_LSHIFT_W16(a16[use_order], 3); //Q12<<3 => Q15
- for (m = use_order - 1; m > 0; m--)
- {
- // (1 - k^2) in Q30
- tmp_inv_denom32 = ((WebRtc_Word32)1073741823) - WEBRTC_SPL_MUL_16_16(k16[m], k16[m]);
- // (1 - k^2) in Q15
- tmp_inv_denom16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp_inv_denom32, 15);
-
- for (k = 1; k <= m; k++)
- {
- // tmp[k] = (a[k] - RC[m] * a[m-k+1]) / (1.0 - RC[m]*RC[m]);
-
- // [Q12<<16 - (Q15*Q12)<<1] = [Q28 - Q28] = Q28
- tmp32[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)a16[k], 16)
- - WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(k16[m], a16[m-k+1]), 1);
-
- tmp32[k] = WebRtcSpl_DivW32W16(tmp32[k], tmp_inv_denom16); //Q28/Q15 = Q13
- }
-
- for (k = 1; k < m; k++)
- {
- a16[k] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32[k], 1); //Q13>>1 => Q12
- }
-
- tmp32[m] = WEBRTC_SPL_SAT(8191, tmp32[m], -8191);
- k16[m - 1] = (WebRtc_Word16)WEBRTC_SPL_LSHIFT_W32(tmp32[m], 2); //Q13<<2 => Q15
- }
- return;
-}
diff --git a/trunk/src/common_audio/signal_processing/min_max_operations.c b/trunk/src/common_audio/signal_processing/min_max_operations.c
deleted file mode 100644
index 57eaff7..0000000
--- a/trunk/src/common_audio/signal_processing/min_max_operations.c
+++ /dev/null
@@ -1,265 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the implementation of functions
- * WebRtcSpl_MaxAbsValueW16()
- * WebRtcSpl_MaxAbsIndexW16()
- * WebRtcSpl_MaxAbsValueW32()
- * WebRtcSpl_MaxValueW16()
- * WebRtcSpl_MaxIndexW16()
- * WebRtcSpl_MaxValueW32()
- * WebRtcSpl_MaxIndexW32()
- * WebRtcSpl_MinValueW16()
- * WebRtcSpl_MinIndexW16()
- * WebRtcSpl_MinValueW32()
- * WebRtcSpl_MinIndexW32()
- *
- * The description header can be found in signal_processing_library.h.
- *
- */
-
-#include "signal_processing_library.h"
-
-#if !(defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM_NEON))
-
-// Maximum absolute value of word16 vector.
-WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(const WebRtc_Word16 *vector, WebRtc_Word16 length)
-{
- WebRtc_Word32 tempMax = 0;
- WebRtc_Word32 absVal;
- WebRtc_Word16 totMax;
- int i;
- G_CONST WebRtc_Word16 *tmpvector = vector;
-
- for (i = 0; i < length; i++)
- {
- absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
- if (absVal > tempMax)
- {
- tempMax = absVal;
- }
- tmpvector++;
- }
- totMax = (WebRtc_Word16)WEBRTC_SPL_MIN(tempMax, WEBRTC_SPL_WORD16_MAX);
- return totMax;
-}
-
-#endif
-
-// Index of maximum absolute value in a word16 vector.
-WebRtc_Word16 WebRtcSpl_MaxAbsIndexW16(G_CONST WebRtc_Word16* vector, WebRtc_Word16 length)
-{
- WebRtc_Word16 tempMax;
- WebRtc_Word16 absTemp;
- WebRtc_Word16 tempMaxIndex = 0;
- WebRtc_Word16 i = 0;
- G_CONST WebRtc_Word16 *tmpvector = vector;
-
- tempMax = WEBRTC_SPL_ABS_W16(*tmpvector);
- tmpvector++;
- for (i = 1; i < length; i++)
- {
- absTemp = WEBRTC_SPL_ABS_W16(*tmpvector);
- tmpvector++;
- if (absTemp > tempMax)
- {
- tempMax = absTemp;
- tempMaxIndex = i;
- }
- }
- return tempMaxIndex;
-}
-
-// Maximum absolute value of word32 vector.
-WebRtc_Word32 WebRtcSpl_MaxAbsValueW32(G_CONST WebRtc_Word32 *vector, WebRtc_Word16 length)
-{
- WebRtc_UWord32 tempMax = 0;
- WebRtc_UWord32 absVal;
- WebRtc_Word32 retval;
- int i;
- G_CONST WebRtc_Word32 *tmpvector = vector;
-
- for (i = 0; i < length; i++)
- {
- absVal = WEBRTC_SPL_ABS_W32((*tmpvector));
- if (absVal > tempMax)
- {
- tempMax = absVal;
- }
- tmpvector++;
- }
- retval = (WebRtc_Word32)(WEBRTC_SPL_MIN(tempMax, WEBRTC_SPL_WORD32_MAX));
- return retval;
-}
-
-// Maximum value of word16 vector.
-#ifndef XSCALE_OPT
-WebRtc_Word16 WebRtcSpl_MaxValueW16(G_CONST WebRtc_Word16* vector, WebRtc_Word16 length)
-{
- WebRtc_Word16 tempMax;
- WebRtc_Word16 i;
- G_CONST WebRtc_Word16 *tmpvector = vector;
-
- tempMax = *tmpvector++;
- for (i = 1; i < length; i++)
- {
- if (*tmpvector++ > tempMax)
- tempMax = vector[i];
- }
- return tempMax;
-}
-#else
-#pragma message(">> WebRtcSpl_MaxValueW16 is excluded from this build")
-#endif
-
-// Index of maximum value in a word16 vector.
-WebRtc_Word16 WebRtcSpl_MaxIndexW16(G_CONST WebRtc_Word16 *vector, WebRtc_Word16 length)
-{
- WebRtc_Word16 tempMax;
- WebRtc_Word16 tempMaxIndex = 0;
- WebRtc_Word16 i = 0;
- G_CONST WebRtc_Word16 *tmpvector = vector;
-
- tempMax = *tmpvector++;
- for (i = 1; i < length; i++)
- {
- if (*tmpvector++ > tempMax)
- {
- tempMax = vector[i];
- tempMaxIndex = i;
- }
- }
- return tempMaxIndex;
-}
-
-// Maximum value of word32 vector.
-#ifndef XSCALE_OPT
-WebRtc_Word32 WebRtcSpl_MaxValueW32(G_CONST WebRtc_Word32* vector, WebRtc_Word16 length)
-{
- WebRtc_Word32 tempMax;
- WebRtc_Word16 i;
- G_CONST WebRtc_Word32 *tmpvector = vector;
-
- tempMax = *tmpvector++;
- for (i = 1; i < length; i++)
- {
- if (*tmpvector++ > tempMax)
- tempMax = vector[i];
- }
- return tempMax;
-}
-#else
-#pragma message(">> WebRtcSpl_MaxValueW32 is excluded from this build")
-#endif
-
-// Index of maximum value in a word32 vector.
-WebRtc_Word16 WebRtcSpl_MaxIndexW32(G_CONST WebRtc_Word32* vector, WebRtc_Word16 length)
-{
- WebRtc_Word32 tempMax;
- WebRtc_Word16 tempMaxIndex = 0;
- WebRtc_Word16 i = 0;
- G_CONST WebRtc_Word32 *tmpvector = vector;
-
- tempMax = *tmpvector++;
- for (i = 1; i < length; i++)
- {
- if (*tmpvector++ > tempMax)
- {
- tempMax = vector[i];
- tempMaxIndex = i;
- }
- }
- return tempMaxIndex;
-}
-
-// Minimum value of word16 vector.
-WebRtc_Word16 WebRtcSpl_MinValueW16(G_CONST WebRtc_Word16 *vector, WebRtc_Word16 length)
-{
- WebRtc_Word16 tempMin;
- WebRtc_Word16 i;
- G_CONST WebRtc_Word16 *tmpvector = vector;
-
- // Find the minimum value
- tempMin = *tmpvector++;
- for (i = 1; i < length; i++)
- {
- if (*tmpvector++ < tempMin)
- tempMin = (vector[i]);
- }
- return tempMin;
-}
-
-// Index of minimum value in a word16 vector.
-#ifndef XSCALE_OPT
-WebRtc_Word16 WebRtcSpl_MinIndexW16(G_CONST WebRtc_Word16* vector, WebRtc_Word16 length)
-{
- WebRtc_Word16 tempMin;
- WebRtc_Word16 tempMinIndex = 0;
- WebRtc_Word16 i = 0;
- G_CONST WebRtc_Word16* tmpvector = vector;
-
- // Find index of smallest value
- tempMin = *tmpvector++;
- for (i = 1; i < length; i++)
- {
- if (*tmpvector++ < tempMin)
- {
- tempMin = vector[i];
- tempMinIndex = i;
- }
- }
- return tempMinIndex;
-}
-#else
-#pragma message(">> WebRtcSpl_MinIndexW16 is excluded from this build")
-#endif
-
-// Minimum value of word32 vector.
-WebRtc_Word32 WebRtcSpl_MinValueW32(G_CONST WebRtc_Word32 *vector, WebRtc_Word16 length)
-{
- WebRtc_Word32 tempMin;
- WebRtc_Word16 i;
- G_CONST WebRtc_Word32 *tmpvector = vector;
-
- // Find the minimum value
- tempMin = *tmpvector++;
- for (i = 1; i < length; i++)
- {
- if (*tmpvector++ < tempMin)
- tempMin = (vector[i]);
- }
- return tempMin;
-}
-
-// Index of minimum value in a word32 vector.
-#ifndef XSCALE_OPT
-WebRtc_Word16 WebRtcSpl_MinIndexW32(G_CONST WebRtc_Word32* vector, WebRtc_Word16 length)
-{
- WebRtc_Word32 tempMin;
- WebRtc_Word16 tempMinIndex = 0;
- WebRtc_Word16 i = 0;
- G_CONST WebRtc_Word32 *tmpvector = vector;
-
- // Find index of smallest value
- tempMin = *tmpvector++;
- for (i = 1; i < length; i++)
- {
- if (*tmpvector++ < tempMin)
- {
- tempMin = vector[i];
- tempMinIndex = i;
- }
- }
- return tempMinIndex;
-}
-#else
-#pragma message(">> WebRtcSpl_MinIndexW32 is excluded from this build")
-#endif
diff --git a/trunk/src/common_audio/signal_processing/min_max_operations_neon.c b/trunk/src/common_audio/signal_processing/min_max_operations_neon.c
deleted file mode 100644
index 158bcc1..0000000
--- a/trunk/src/common_audio/signal_processing/min_max_operations_neon.c
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#if (defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM_NEON))
-
-#include <arm_neon.h>
-
-#include "signal_processing_library.h"
-
-// Maximum absolute value of word16 vector.
-WebRtc_Word16 WebRtcSpl_MaxAbsValueW16(const WebRtc_Word16* vector,
- WebRtc_Word16 length) {
- WebRtc_Word32 temp_max = 0;
- WebRtc_Word32 abs_val;
- WebRtc_Word16 tot_max;
- int i;
-
- __asm__("vmov.i16 d25, #0" : : : "d25");
-
- for (i = 0; i < length - 7; i += 8) {
- __asm__("vld1.16 {d26, d27}, [%0]" : : "r"(&vector[i]) : "q13");
- __asm__("vabs.s16 q13, q13" : : : "q13");
- __asm__("vpmax.s16 d26, d27" : : : "q13");
- __asm__("vpmax.s16 d25, d26" : : : "d25", "d26");
- }
- __asm__("vpmax.s16 d25, d25" : : : "d25");
- __asm__("vpmax.s16 d25, d25" : : : "d25");
- __asm__("vmov.s16 %0, d25[0]" : "=r"(temp_max): : "d25");
-
- for (; i < length; i++) {
- abs_val = WEBRTC_SPL_ABS_W32((vector[i]));
- if (abs_val > temp_max) {
- temp_max = abs_val;
- }
- }
- tot_max = (WebRtc_Word16)WEBRTC_SPL_MIN(temp_max, WEBRTC_SPL_WORD16_MAX);
- return tot_max;
-}
-
-#endif
diff --git a/trunk/src/common_audio/signal_processing/randomization_functions.c b/trunk/src/common_audio/signal_processing/randomization_functions.c
deleted file mode 100644
index 04271ad..0000000
--- a/trunk/src/common_audio/signal_processing/randomization_functions.c
+++ /dev/null
@@ -1,119 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains implementations of the randomization functions
- * WebRtcSpl_IncreaseSeed()
- * WebRtcSpl_RandU()
- * WebRtcSpl_RandN()
- * WebRtcSpl_RandUArray()
- *
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-static const WebRtc_Word16 kRandNTable[] = {
- 9178, -7260, 40, 10189, 4894, -3531, -13779, 14764,
- -4008, -8884, -8990, 1008, 7368, 5184, 3251, -5817,
- -9786, 5963, 1770, 8066, -7135, 10772, -2298, 1361,
- 6484, 2241, -8633, 792, 199, -3344, 6553, -10079,
- -15040, 95, 11608, -12469, 14161, -4176, 2476, 6403,
- 13685, -16005, 6646, 2239, 10916, -3004, -602, -3141,
- 2142, 14144, -5829, 5305, 8209, 4713, 2697, -5112,
- 16092, -1210, -2891, -6631, -5360, -11878, -6781, -2739,
- -6392, 536, 10923, 10872, 5059, -4748, -7770, 5477,
- 38, -1025, -2892, 1638, 6304, 14375, -11028, 1553,
- -1565, 10762, -393, 4040, 5257, 12310, 6554, -4799,
- 4899, -6354, 1603, -1048, -2220, 8247, -186, -8944,
- -12004, 2332, 4801, -4933, 6371, 131, 8614, -5927,
- -8287, -22760, 4033, -15162, 3385, 3246, 3153, -5250,
- 3766, 784, 6494, -62, 3531, -1582, 15572, 662,
- -3952, -330, -3196, 669, 7236, -2678, -6569, 23319,
- -8645, -741, 14830, -15976, 4903, 315, -11342, 10311,
- 1858, -7777, 2145, 5436, 5677, -113, -10033, 826,
- -1353, 17210, 7768, 986, -1471, 8291, -4982, 8207,
- -14911, -6255, -2449, -11881, -7059, -11703, -4338, 8025,
- 7538, -2823, -12490, 9470, -1613, -2529, -10092, -7807,
- 9480, 6970, -12844, 5123, 3532, 4816, 4803, -8455,
- -5045, 14032, -4378, -1643, 5756, -11041, -2732, -16618,
- -6430, -18375, -3320, 6098, 5131, -4269, -8840, 2482,
- -7048, 1547, -21890, -6505, -7414, -424, -11722, 7955,
- 1653, -17299, 1823, 473, -9232, 3337, 1111, 873,
- 4018, -8982, 9889, 3531, -11763, -3799, 7373, -4539,
- 3231, 7054, -8537, 7616, 6244, 16635, 447, -2915,
- 13967, 705, -2669, -1520, -1771, -16188, 5956, 5117,
- 6371, -9936, -1448, 2480, 5128, 7550, -8130, 5236,
- 8213, -6443, 7707, -1950, -13811, 7218, 7031, -3883,
- 67, 5731, -2874, 13480, -3743, 9298, -3280, 3552,
- -4425, -18, -3785, -9988, -5357, 5477, -11794, 2117,
- 1416, -9935, 3376, 802, -5079, -8243, 12652, 66,
- 3653, -2368, 6781, -21895, -7227, 2487, 7839, -385,
- 6646, -7016, -4658, 5531, -1705, 834, 129, 3694,
- -1343, 2238, -22640, -6417, -11139, 11301, -2945, -3494,
- -5626, 185, -3615, -2041, -7972, -3106, -60, -23497,
- -1566, 17064, 3519, 2518, 304, -6805, -10269, 2105,
- 1936, -426, -736, -8122, -1467, 4238, -6939, -13309,
- 360, 7402, -7970, 12576, 3287, 12194, -6289, -16006,
- 9171, 4042, -9193, 9123, -2512, 6388, -4734, -8739,
- 1028, -5406, -1696, 5889, -666, -4736, 4971, 3565,
- 9362, -6292, 3876, -3652, -19666, 7523, -4061, 391,
- -11773, 7502, -3763, 4929, -9478, 13278, 2805, 4496,
- 7814, 16419, 12455, -14773, 2127, -2746, 3763, 4847,
- 3698, 6978, 4751, -6957, -3581, -45, 6252, 1513,
- -4797, -7925, 11270, 16188, -2359, -5269, 9376, -10777,
- 7262, 20031, -6515, -2208, -5353, 8085, -1341, -1303,
- 7333, 5576, 3625, 5763, -7931, 9833, -3371, -10305,
- 6534, -13539, -9971, 997, 8464, -4064, -1495, 1857,
- 13624, 5458, 9490, -11086, -4524, 12022, -550, -198,
- 408, -8455, -7068, 10289, 9712, -3366, 9028, -7621,
- -5243, 2362, 6909, 4672, -4933, -1799, 4709, -4563,
- -62, -566, 1624, -7010, 14730, -17791, -3697, -2344,
- -1741, 7099, -9509, -6855, -1989, 3495, -2289, 2031,
- 12784, 891, 14189, -3963, -5683, 421, -12575, 1724,
- -12682, -5970, -8169, 3143, -1824, -5488, -5130, 8536,
- 12799, 794, 5738, 3459, -11689, -258, -3738, -3775,
- -8742, 2333, 8312, -9383, 10331, 13119, 8398, 10644,
- -19433, -6446, -16277, -11793, 16284, 9345, 15222, 15834,
- 2009, -7349, 130, -14547, 338, -5998, 3337, 21492,
- 2406, 7703, -951, 11196, -564, 3406, 2217, 4806,
- 2374, -5797, 11839, 8940, -11874, 18213, 2855, 10492
-};
-
-WebRtc_UWord32 WebRtcSpl_IncreaseSeed(WebRtc_UWord32 *seed)
-{
- seed[0] = (seed[0] * ((WebRtc_Word32)69069) + 1) & (WEBRTC_SPL_MAX_SEED_USED - 1);
- return seed[0];
-}
-
-WebRtc_Word16 WebRtcSpl_RandU(WebRtc_UWord32 *seed)
-{
- return (WebRtc_Word16)(WebRtcSpl_IncreaseSeed(seed) >> 16);
-}
-
-WebRtc_Word16 WebRtcSpl_RandN(WebRtc_UWord32 *seed)
-{
- return kRandNTable[WebRtcSpl_IncreaseSeed(seed) >> 23];
-}
-
-// Creates an array of uniformly distributed variables
-WebRtc_Word16 WebRtcSpl_RandUArray(WebRtc_Word16* vector,
- WebRtc_Word16 vector_length,
- WebRtc_UWord32* seed)
-{
- int i;
- for (i = 0; i < vector_length; i++)
- {
- vector[i] = WebRtcSpl_RandU(seed);
- }
- return vector_length;
-}
diff --git a/trunk/src/common_audio/signal_processing/refl_coef_to_lpc.c b/trunk/src/common_audio/signal_processing/refl_coef_to_lpc.c
deleted file mode 100644
index d07804d..0000000
--- a/trunk/src/common_audio/signal_processing/refl_coef_to_lpc.c
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_ReflCoefToLpc().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_ReflCoefToLpc(G_CONST WebRtc_Word16 *k, int use_order, WebRtc_Word16 *a)
-{
- WebRtc_Word16 any[WEBRTC_SPL_MAX_LPC_ORDER + 1];
- WebRtc_Word16 *aptr, *aptr2, *anyptr;
- G_CONST WebRtc_Word16 *kptr;
- int m, i;
-
- kptr = k;
- *a = 4096; // i.e., (Word16_MAX >> 3)+1.
- *any = *a;
- a[1] = WEBRTC_SPL_RSHIFT_W16((*k), 3);
-
- for (m = 1; m < use_order; m++)
- {
- kptr++;
- aptr = a;
- aptr++;
- aptr2 = &a[m];
- anyptr = any;
- anyptr++;
-
- any[m + 1] = WEBRTC_SPL_RSHIFT_W16((*kptr), 3);
- for (i = 0; i < m; i++)
- {
- *anyptr = (*aptr)
- + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((*aptr2), (*kptr), 15);
- anyptr++;
- aptr++;
- aptr2--;
- }
-
- aptr = a;
- anyptr = any;
- for (i = 0; i < (m + 2); i++)
- {
- *aptr = *anyptr;
- aptr++;
- anyptr++;
- }
- }
-}
diff --git a/trunk/src/common_audio/signal_processing/resample.c b/trunk/src/common_audio/signal_processing/resample.c
deleted file mode 100644
index 19d1778..0000000
--- a/trunk/src/common_audio/signal_processing/resample.c
+++ /dev/null
@@ -1,505 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the resampling functions for 22 kHz.
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-#include "resample_by_2_internal.h"
-
-// Declaration of internally used functions
-static void WebRtcSpl_32khzTo22khzIntToShort(const WebRtc_Word32 *In, WebRtc_Word16 *Out,
- const WebRtc_Word32 K);
-
-void WebRtcSpl_32khzTo22khzIntToInt(const WebRtc_Word32 *In, WebRtc_Word32 *Out,
- const WebRtc_Word32 K);
-
-// interpolation coefficients
-static const WebRtc_Word16 kCoefficients32To22[5][9] = {
- {127, -712, 2359, -6333, 23456, 16775, -3695, 945, -154},
- {-39, 230, -830, 2785, 32366, -2324, 760, -218, 38},
- {117, -663, 2222, -6133, 26634, 13070, -3174, 831, -137},
- {-77, 457, -1677, 5958, 31175, -4136, 1405, -408, 71},
- { 98, -560, 1900, -5406, 29240, 9423, -2480, 663, -110}
-};
-
-//////////////////////
-// 22 kHz -> 16 kHz //
-//////////////////////
-
-// number of subblocks; options: 1, 2, 4, 5, 10
-#define SUB_BLOCKS_22_16 5
-
-// 22 -> 16 resampler
-void WebRtcSpl_Resample22khzTo16khz(const WebRtc_Word16* in, WebRtc_Word16* out,
- WebRtcSpl_State22khzTo16khz* state, WebRtc_Word32* tmpmem)
-{
- int k;
-
- // process two blocks of 10/SUB_BLOCKS_22_16 ms (to reduce temp buffer size)
- for (k = 0; k < SUB_BLOCKS_22_16; k++)
- {
- ///// 22 --> 44 /////
- // WebRtc_Word16 in[220/SUB_BLOCKS_22_16]
- // WebRtc_Word32 out[440/SUB_BLOCKS_22_16]
- /////
- WebRtcSpl_UpBy2ShortToInt(in, 220 / SUB_BLOCKS_22_16, tmpmem + 16, state->S_22_44);
-
- ///// 44 --> 32 /////
- // WebRtc_Word32 in[440/SUB_BLOCKS_22_16]
- // WebRtc_Word32 out[320/SUB_BLOCKS_22_16]
- /////
- // copy state to and from input array
- tmpmem[8] = state->S_44_32[0];
- tmpmem[9] = state->S_44_32[1];
- tmpmem[10] = state->S_44_32[2];
- tmpmem[11] = state->S_44_32[3];
- tmpmem[12] = state->S_44_32[4];
- tmpmem[13] = state->S_44_32[5];
- tmpmem[14] = state->S_44_32[6];
- tmpmem[15] = state->S_44_32[7];
- state->S_44_32[0] = tmpmem[440 / SUB_BLOCKS_22_16 + 8];
- state->S_44_32[1] = tmpmem[440 / SUB_BLOCKS_22_16 + 9];
- state->S_44_32[2] = tmpmem[440 / SUB_BLOCKS_22_16 + 10];
- state->S_44_32[3] = tmpmem[440 / SUB_BLOCKS_22_16 + 11];
- state->S_44_32[4] = tmpmem[440 / SUB_BLOCKS_22_16 + 12];
- state->S_44_32[5] = tmpmem[440 / SUB_BLOCKS_22_16 + 13];
- state->S_44_32[6] = tmpmem[440 / SUB_BLOCKS_22_16 + 14];
- state->S_44_32[7] = tmpmem[440 / SUB_BLOCKS_22_16 + 15];
-
- WebRtcSpl_Resample44khzTo32khz(tmpmem + 8, tmpmem, 40 / SUB_BLOCKS_22_16);
-
- ///// 32 --> 16 /////
- // WebRtc_Word32 in[320/SUB_BLOCKS_22_16]
- // WebRtc_Word32 out[160/SUB_BLOCKS_22_16]
- /////
- WebRtcSpl_DownBy2IntToShort(tmpmem, 320 / SUB_BLOCKS_22_16, out, state->S_32_16);
-
- // move input/output pointers 10/SUB_BLOCKS_22_16 ms seconds ahead
- in += 220 / SUB_BLOCKS_22_16;
- out += 160 / SUB_BLOCKS_22_16;
- }
-}
-
-// initialize state of 22 -> 16 resampler
-void WebRtcSpl_ResetResample22khzTo16khz(WebRtcSpl_State22khzTo16khz* state)
-{
- int k;
- for (k = 0; k < 8; k++)
- {
- state->S_22_44[k] = 0;
- state->S_44_32[k] = 0;
- state->S_32_16[k] = 0;
- }
-}
-
-//////////////////////
-// 16 kHz -> 22 kHz //
-//////////////////////
-
-// number of subblocks; options: 1, 2, 4, 5, 10
-#define SUB_BLOCKS_16_22 4
-
-// 16 -> 22 resampler
-void WebRtcSpl_Resample16khzTo22khz(const WebRtc_Word16* in, WebRtc_Word16* out,
- WebRtcSpl_State16khzTo22khz* state, WebRtc_Word32* tmpmem)
-{
- int k;
-
- // process two blocks of 10/SUB_BLOCKS_16_22 ms (to reduce temp buffer size)
- for (k = 0; k < SUB_BLOCKS_16_22; k++)
- {
- ///// 16 --> 32 /////
- // WebRtc_Word16 in[160/SUB_BLOCKS_16_22]
- // WebRtc_Word32 out[320/SUB_BLOCKS_16_22]
- /////
- WebRtcSpl_UpBy2ShortToInt(in, 160 / SUB_BLOCKS_16_22, tmpmem + 8, state->S_16_32);
-
- ///// 32 --> 22 /////
- // WebRtc_Word32 in[320/SUB_BLOCKS_16_22]
- // WebRtc_Word32 out[220/SUB_BLOCKS_16_22]
- /////
- // copy state to and from input array
- tmpmem[0] = state->S_32_22[0];
- tmpmem[1] = state->S_32_22[1];
- tmpmem[2] = state->S_32_22[2];
- tmpmem[3] = state->S_32_22[3];
- tmpmem[4] = state->S_32_22[4];
- tmpmem[5] = state->S_32_22[5];
- tmpmem[6] = state->S_32_22[6];
- tmpmem[7] = state->S_32_22[7];
- state->S_32_22[0] = tmpmem[320 / SUB_BLOCKS_16_22];
- state->S_32_22[1] = tmpmem[320 / SUB_BLOCKS_16_22 + 1];
- state->S_32_22[2] = tmpmem[320 / SUB_BLOCKS_16_22 + 2];
- state->S_32_22[3] = tmpmem[320 / SUB_BLOCKS_16_22 + 3];
- state->S_32_22[4] = tmpmem[320 / SUB_BLOCKS_16_22 + 4];
- state->S_32_22[5] = tmpmem[320 / SUB_BLOCKS_16_22 + 5];
- state->S_32_22[6] = tmpmem[320 / SUB_BLOCKS_16_22 + 6];
- state->S_32_22[7] = tmpmem[320 / SUB_BLOCKS_16_22 + 7];
-
- WebRtcSpl_32khzTo22khzIntToShort(tmpmem, out, 20 / SUB_BLOCKS_16_22);
-
- // move input/output pointers 10/SUB_BLOCKS_16_22 ms seconds ahead
- in += 160 / SUB_BLOCKS_16_22;
- out += 220 / SUB_BLOCKS_16_22;
- }
-}
-
-// initialize state of 16 -> 22 resampler
-void WebRtcSpl_ResetResample16khzTo22khz(WebRtcSpl_State16khzTo22khz* state)
-{
- int k;
- for (k = 0; k < 8; k++)
- {
- state->S_16_32[k] = 0;
- state->S_32_22[k] = 0;
- }
-}
-
-//////////////////////
-// 22 kHz -> 8 kHz //
-//////////////////////
-
-// number of subblocks; options: 1, 2, 5, 10
-#define SUB_BLOCKS_22_8 2
-
-// 22 -> 8 resampler
-void WebRtcSpl_Resample22khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
- WebRtcSpl_State22khzTo8khz* state, WebRtc_Word32* tmpmem)
-{
- int k;
-
- // process two blocks of 10/SUB_BLOCKS_22_8 ms (to reduce temp buffer size)
- for (k = 0; k < SUB_BLOCKS_22_8; k++)
- {
- ///// 22 --> 22 lowpass /////
- // WebRtc_Word16 in[220/SUB_BLOCKS_22_8]
- // WebRtc_Word32 out[220/SUB_BLOCKS_22_8]
- /////
- WebRtcSpl_LPBy2ShortToInt(in, 220 / SUB_BLOCKS_22_8, tmpmem + 16, state->S_22_22);
-
- ///// 22 --> 16 /////
- // WebRtc_Word32 in[220/SUB_BLOCKS_22_8]
- // WebRtc_Word32 out[160/SUB_BLOCKS_22_8]
- /////
- // copy state to and from input array
- tmpmem[8] = state->S_22_16[0];
- tmpmem[9] = state->S_22_16[1];
- tmpmem[10] = state->S_22_16[2];
- tmpmem[11] = state->S_22_16[3];
- tmpmem[12] = state->S_22_16[4];
- tmpmem[13] = state->S_22_16[5];
- tmpmem[14] = state->S_22_16[6];
- tmpmem[15] = state->S_22_16[7];
- state->S_22_16[0] = tmpmem[220 / SUB_BLOCKS_22_8 + 8];
- state->S_22_16[1] = tmpmem[220 / SUB_BLOCKS_22_8 + 9];
- state->S_22_16[2] = tmpmem[220 / SUB_BLOCKS_22_8 + 10];
- state->S_22_16[3] = tmpmem[220 / SUB_BLOCKS_22_8 + 11];
- state->S_22_16[4] = tmpmem[220 / SUB_BLOCKS_22_8 + 12];
- state->S_22_16[5] = tmpmem[220 / SUB_BLOCKS_22_8 + 13];
- state->S_22_16[6] = tmpmem[220 / SUB_BLOCKS_22_8 + 14];
- state->S_22_16[7] = tmpmem[220 / SUB_BLOCKS_22_8 + 15];
-
- WebRtcSpl_Resample44khzTo32khz(tmpmem + 8, tmpmem, 20 / SUB_BLOCKS_22_8);
-
- ///// 16 --> 8 /////
- // WebRtc_Word32 in[160/SUB_BLOCKS_22_8]
- // WebRtc_Word32 out[80/SUB_BLOCKS_22_8]
- /////
- WebRtcSpl_DownBy2IntToShort(tmpmem, 160 / SUB_BLOCKS_22_8, out, state->S_16_8);
-
- // move input/output pointers 10/SUB_BLOCKS_22_8 ms seconds ahead
- in += 220 / SUB_BLOCKS_22_8;
- out += 80 / SUB_BLOCKS_22_8;
- }
-}
-
-// initialize state of 22 -> 8 resampler
-void WebRtcSpl_ResetResample22khzTo8khz(WebRtcSpl_State22khzTo8khz* state)
-{
- int k;
- for (k = 0; k < 8; k++)
- {
- state->S_22_22[k] = 0;
- state->S_22_22[k + 8] = 0;
- state->S_22_16[k] = 0;
- state->S_16_8[k] = 0;
- }
-}
-
-//////////////////////
-// 8 kHz -> 22 kHz //
-//////////////////////
-
-// number of subblocks; options: 1, 2, 5, 10
-#define SUB_BLOCKS_8_22 2
-
-// 8 -> 22 resampler
-void WebRtcSpl_Resample8khzTo22khz(const WebRtc_Word16* in, WebRtc_Word16* out,
- WebRtcSpl_State8khzTo22khz* state, WebRtc_Word32* tmpmem)
-{
- int k;
-
- // process two blocks of 10/SUB_BLOCKS_8_22 ms (to reduce temp buffer size)
- for (k = 0; k < SUB_BLOCKS_8_22; k++)
- {
- ///// 8 --> 16 /////
- // WebRtc_Word16 in[80/SUB_BLOCKS_8_22]
- // WebRtc_Word32 out[160/SUB_BLOCKS_8_22]
- /////
- WebRtcSpl_UpBy2ShortToInt(in, 80 / SUB_BLOCKS_8_22, tmpmem + 18, state->S_8_16);
-
- ///// 16 --> 11 /////
- // WebRtc_Word32 in[160/SUB_BLOCKS_8_22]
- // WebRtc_Word32 out[110/SUB_BLOCKS_8_22]
- /////
- // copy state to and from input array
- tmpmem[10] = state->S_16_11[0];
- tmpmem[11] = state->S_16_11[1];
- tmpmem[12] = state->S_16_11[2];
- tmpmem[13] = state->S_16_11[3];
- tmpmem[14] = state->S_16_11[4];
- tmpmem[15] = state->S_16_11[5];
- tmpmem[16] = state->S_16_11[6];
- tmpmem[17] = state->S_16_11[7];
- state->S_16_11[0] = tmpmem[160 / SUB_BLOCKS_8_22 + 10];
- state->S_16_11[1] = tmpmem[160 / SUB_BLOCKS_8_22 + 11];
- state->S_16_11[2] = tmpmem[160 / SUB_BLOCKS_8_22 + 12];
- state->S_16_11[3] = tmpmem[160 / SUB_BLOCKS_8_22 + 13];
- state->S_16_11[4] = tmpmem[160 / SUB_BLOCKS_8_22 + 14];
- state->S_16_11[5] = tmpmem[160 / SUB_BLOCKS_8_22 + 15];
- state->S_16_11[6] = tmpmem[160 / SUB_BLOCKS_8_22 + 16];
- state->S_16_11[7] = tmpmem[160 / SUB_BLOCKS_8_22 + 17];
-
- WebRtcSpl_32khzTo22khzIntToInt(tmpmem + 10, tmpmem, 10 / SUB_BLOCKS_8_22);
-
- ///// 11 --> 22 /////
- // WebRtc_Word32 in[110/SUB_BLOCKS_8_22]
- // WebRtc_Word16 out[220/SUB_BLOCKS_8_22]
- /////
- WebRtcSpl_UpBy2IntToShort(tmpmem, 110 / SUB_BLOCKS_8_22, out, state->S_11_22);
-
- // move input/output pointers 10/SUB_BLOCKS_8_22 ms seconds ahead
- in += 80 / SUB_BLOCKS_8_22;
- out += 220 / SUB_BLOCKS_8_22;
- }
-}
-
-// initialize state of 8 -> 22 resampler
-void WebRtcSpl_ResetResample8khzTo22khz(WebRtcSpl_State8khzTo22khz* state)
-{
- int k;
- for (k = 0; k < 8; k++)
- {
- state->S_8_16[k] = 0;
- state->S_16_11[k] = 0;
- state->S_11_22[k] = 0;
- }
-}
-
-// compute two inner-products and store them to output array
-static void WebRtcSpl_DotProdIntToInt(const WebRtc_Word32* in1, const WebRtc_Word32* in2,
- const WebRtc_Word16* coef_ptr, WebRtc_Word32* out1,
- WebRtc_Word32* out2)
-{
- WebRtc_Word32 tmp1 = 16384;
- WebRtc_Word32 tmp2 = 16384;
- WebRtc_Word16 coef;
-
- coef = coef_ptr[0];
- tmp1 += coef * in1[0];
- tmp2 += coef * in2[-0];
-
- coef = coef_ptr[1];
- tmp1 += coef * in1[1];
- tmp2 += coef * in2[-1];
-
- coef = coef_ptr[2];
- tmp1 += coef * in1[2];
- tmp2 += coef * in2[-2];
-
- coef = coef_ptr[3];
- tmp1 += coef * in1[3];
- tmp2 += coef * in2[-3];
-
- coef = coef_ptr[4];
- tmp1 += coef * in1[4];
- tmp2 += coef * in2[-4];
-
- coef = coef_ptr[5];
- tmp1 += coef * in1[5];
- tmp2 += coef * in2[-5];
-
- coef = coef_ptr[6];
- tmp1 += coef * in1[6];
- tmp2 += coef * in2[-6];
-
- coef = coef_ptr[7];
- tmp1 += coef * in1[7];
- tmp2 += coef * in2[-7];
-
- coef = coef_ptr[8];
- *out1 = tmp1 + coef * in1[8];
- *out2 = tmp2 + coef * in2[-8];
-}
-
-// compute two inner-products and store them to output array
-static void WebRtcSpl_DotProdIntToShort(const WebRtc_Word32* in1, const WebRtc_Word32* in2,
- const WebRtc_Word16* coef_ptr, WebRtc_Word16* out1,
- WebRtc_Word16* out2)
-{
- WebRtc_Word32 tmp1 = 16384;
- WebRtc_Word32 tmp2 = 16384;
- WebRtc_Word16 coef;
-
- coef = coef_ptr[0];
- tmp1 += coef * in1[0];
- tmp2 += coef * in2[-0];
-
- coef = coef_ptr[1];
- tmp1 += coef * in1[1];
- tmp2 += coef * in2[-1];
-
- coef = coef_ptr[2];
- tmp1 += coef * in1[2];
- tmp2 += coef * in2[-2];
-
- coef = coef_ptr[3];
- tmp1 += coef * in1[3];
- tmp2 += coef * in2[-3];
-
- coef = coef_ptr[4];
- tmp1 += coef * in1[4];
- tmp2 += coef * in2[-4];
-
- coef = coef_ptr[5];
- tmp1 += coef * in1[5];
- tmp2 += coef * in2[-5];
-
- coef = coef_ptr[6];
- tmp1 += coef * in1[6];
- tmp2 += coef * in2[-6];
-
- coef = coef_ptr[7];
- tmp1 += coef * in1[7];
- tmp2 += coef * in2[-7];
-
- coef = coef_ptr[8];
- tmp1 += coef * in1[8];
- tmp2 += coef * in2[-8];
-
- // scale down, round and saturate
- tmp1 >>= 15;
- if (tmp1 > (WebRtc_Word32)0x00007FFF)
- tmp1 = 0x00007FFF;
- if (tmp1 < (WebRtc_Word32)0xFFFF8000)
- tmp1 = 0xFFFF8000;
- tmp2 >>= 15;
- if (tmp2 > (WebRtc_Word32)0x00007FFF)
- tmp2 = 0x00007FFF;
- if (tmp2 < (WebRtc_Word32)0xFFFF8000)
- tmp2 = 0xFFFF8000;
- *out1 = (WebRtc_Word16)tmp1;
- *out2 = (WebRtc_Word16)tmp2;
-}
-
-// Resampling ratio: 11/16
-// input: WebRtc_Word32 (normalized, not saturated) :: size 16 * K
-// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) :: size 11 * K
-// K: Number of blocks
-
-void WebRtcSpl_32khzTo22khzIntToInt(const WebRtc_Word32* In,
- WebRtc_Word32* Out,
- const WebRtc_Word32 K)
-{
- /////////////////////////////////////////////////////////////
- // Filter operation:
- //
- // Perform resampling (16 input samples -> 11 output samples);
- // process in sub blocks of size 16 samples.
- WebRtc_Word32 m;
-
- for (m = 0; m < K; m++)
- {
- // first output sample
- Out[0] = ((WebRtc_Word32)In[3] << 15) + (1 << 14);
-
- // sum and accumulate filter coefficients and input samples
- WebRtcSpl_DotProdIntToInt(&In[0], &In[22], kCoefficients32To22[0], &Out[1], &Out[10]);
-
- // sum and accumulate filter coefficients and input samples
- WebRtcSpl_DotProdIntToInt(&In[2], &In[20], kCoefficients32To22[1], &Out[2], &Out[9]);
-
- // sum and accumulate filter coefficients and input samples
- WebRtcSpl_DotProdIntToInt(&In[3], &In[19], kCoefficients32To22[2], &Out[3], &Out[8]);
-
- // sum and accumulate filter coefficients and input samples
- WebRtcSpl_DotProdIntToInt(&In[5], &In[17], kCoefficients32To22[3], &Out[4], &Out[7]);
-
- // sum and accumulate filter coefficients and input samples
- WebRtcSpl_DotProdIntToInt(&In[6], &In[16], kCoefficients32To22[4], &Out[5], &Out[6]);
-
- // update pointers
- In += 16;
- Out += 11;
- }
-}
-
-// Resampling ratio: 11/16
-// input: WebRtc_Word32 (normalized, not saturated) :: size 16 * K
-// output: WebRtc_Word16 (saturated) :: size 11 * K
-// K: Number of blocks
-
-void WebRtcSpl_32khzTo22khzIntToShort(const WebRtc_Word32 *In,
- WebRtc_Word16 *Out,
- const WebRtc_Word32 K)
-{
- /////////////////////////////////////////////////////////////
- // Filter operation:
- //
- // Perform resampling (16 input samples -> 11 output samples);
- // process in sub blocks of size 16 samples.
- WebRtc_Word32 tmp;
- WebRtc_Word32 m;
-
- for (m = 0; m < K; m++)
- {
- // first output sample
- tmp = In[3];
- if (tmp > (WebRtc_Word32)0x00007FFF)
- tmp = 0x00007FFF;
- if (tmp < (WebRtc_Word32)0xFFFF8000)
- tmp = 0xFFFF8000;
- Out[0] = (WebRtc_Word16)tmp;
-
- // sum and accumulate filter coefficients and input samples
- WebRtcSpl_DotProdIntToShort(&In[0], &In[22], kCoefficients32To22[0], &Out[1], &Out[10]);
-
- // sum and accumulate filter coefficients and input samples
- WebRtcSpl_DotProdIntToShort(&In[2], &In[20], kCoefficients32To22[1], &Out[2], &Out[9]);
-
- // sum and accumulate filter coefficients and input samples
- WebRtcSpl_DotProdIntToShort(&In[3], &In[19], kCoefficients32To22[2], &Out[3], &Out[8]);
-
- // sum and accumulate filter coefficients and input samples
- WebRtcSpl_DotProdIntToShort(&In[5], &In[17], kCoefficients32To22[3], &Out[4], &Out[7]);
-
- // sum and accumulate filter coefficients and input samples
- WebRtcSpl_DotProdIntToShort(&In[6], &In[16], kCoefficients32To22[4], &Out[5], &Out[6]);
-
- // update pointers
- In += 16;
- Out += 11;
- }
-}
diff --git a/trunk/src/common_audio/signal_processing/resample_48khz.c b/trunk/src/common_audio/signal_processing/resample_48khz.c
deleted file mode 100644
index 31cbe6b..0000000
--- a/trunk/src/common_audio/signal_processing/resample_48khz.c
+++ /dev/null
@@ -1,186 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains resampling functions between 48 kHz and nb/wb.
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include <string.h>
-#include "signal_processing_library.h"
-#include "resample_by_2_internal.h"
-
-////////////////////////////
-///// 48 kHz -> 16 kHz /////
-////////////////////////////
-
-// 48 -> 16 resampler
-void WebRtcSpl_Resample48khzTo16khz(const WebRtc_Word16* in, WebRtc_Word16* out,
- WebRtcSpl_State48khzTo16khz* state, WebRtc_Word32* tmpmem)
-{
- ///// 48 --> 48(LP) /////
- // WebRtc_Word16 in[480]
- // WebRtc_Word32 out[480]
- /////
- WebRtcSpl_LPBy2ShortToInt(in, 480, tmpmem + 16, state->S_48_48);
-
- ///// 48 --> 32 /////
- // WebRtc_Word32 in[480]
- // WebRtc_Word32 out[320]
- /////
- // copy state to and from input array
- memcpy(tmpmem + 8, state->S_48_32, 8 * sizeof(WebRtc_Word32));
- memcpy(state->S_48_32, tmpmem + 488, 8 * sizeof(WebRtc_Word32));
- WebRtcSpl_Resample48khzTo32khz(tmpmem + 8, tmpmem, 160);
-
- ///// 32 --> 16 /////
- // WebRtc_Word32 in[320]
- // WebRtc_Word16 out[160]
- /////
- WebRtcSpl_DownBy2IntToShort(tmpmem, 320, out, state->S_32_16);
-}
-
-// initialize state of 48 -> 16 resampler
-void WebRtcSpl_ResetResample48khzTo16khz(WebRtcSpl_State48khzTo16khz* state)
-{
- memset(state->S_48_48, 0, 16 * sizeof(WebRtc_Word32));
- memset(state->S_48_32, 0, 8 * sizeof(WebRtc_Word32));
- memset(state->S_32_16, 0, 8 * sizeof(WebRtc_Word32));
-}
-
-////////////////////////////
-///// 16 kHz -> 48 kHz /////
-////////////////////////////
-
-// 16 -> 48 resampler
-void WebRtcSpl_Resample16khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
- WebRtcSpl_State16khzTo48khz* state, WebRtc_Word32* tmpmem)
-{
- ///// 16 --> 32 /////
- // WebRtc_Word16 in[160]
- // WebRtc_Word32 out[320]
- /////
- WebRtcSpl_UpBy2ShortToInt(in, 160, tmpmem + 16, state->S_16_32);
-
- ///// 32 --> 24 /////
- // WebRtc_Word32 in[320]
- // WebRtc_Word32 out[240]
- // copy state to and from input array
- /////
- memcpy(tmpmem + 8, state->S_32_24, 8 * sizeof(WebRtc_Word32));
- memcpy(state->S_32_24, tmpmem + 328, 8 * sizeof(WebRtc_Word32));
- WebRtcSpl_Resample32khzTo24khz(tmpmem + 8, tmpmem, 80);
-
- ///// 24 --> 48 /////
- // WebRtc_Word32 in[240]
- // WebRtc_Word16 out[480]
- /////
- WebRtcSpl_UpBy2IntToShort(tmpmem, 240, out, state->S_24_48);
-}
-
-// initialize state of 16 -> 48 resampler
-void WebRtcSpl_ResetResample16khzTo48khz(WebRtcSpl_State16khzTo48khz* state)
-{
- memset(state->S_16_32, 0, 8 * sizeof(WebRtc_Word32));
- memset(state->S_32_24, 0, 8 * sizeof(WebRtc_Word32));
- memset(state->S_24_48, 0, 8 * sizeof(WebRtc_Word32));
-}
-
-////////////////////////////
-///// 48 kHz -> 8 kHz /////
-////////////////////////////
-
-// 48 -> 8 resampler
-void WebRtcSpl_Resample48khzTo8khz(const WebRtc_Word16* in, WebRtc_Word16* out,
- WebRtcSpl_State48khzTo8khz* state, WebRtc_Word32* tmpmem)
-{
- ///// 48 --> 24 /////
- // WebRtc_Word16 in[480]
- // WebRtc_Word32 out[240]
- /////
- WebRtcSpl_DownBy2ShortToInt(in, 480, tmpmem + 256, state->S_48_24);
-
- ///// 24 --> 24(LP) /////
- // WebRtc_Word32 in[240]
- // WebRtc_Word32 out[240]
- /////
- WebRtcSpl_LPBy2IntToInt(tmpmem + 256, 240, tmpmem + 16, state->S_24_24);
-
- ///// 24 --> 16 /////
- // WebRtc_Word32 in[240]
- // WebRtc_Word32 out[160]
- /////
- // copy state to and from input array
- memcpy(tmpmem + 8, state->S_24_16, 8 * sizeof(WebRtc_Word32));
- memcpy(state->S_24_16, tmpmem + 248, 8 * sizeof(WebRtc_Word32));
- WebRtcSpl_Resample48khzTo32khz(tmpmem + 8, tmpmem, 80);
-
- ///// 16 --> 8 /////
- // WebRtc_Word32 in[160]
- // WebRtc_Word16 out[80]
- /////
- WebRtcSpl_DownBy2IntToShort(tmpmem, 160, out, state->S_16_8);
-}
-
-// initialize state of 48 -> 8 resampler
-void WebRtcSpl_ResetResample48khzTo8khz(WebRtcSpl_State48khzTo8khz* state)
-{
- memset(state->S_48_24, 0, 8 * sizeof(WebRtc_Word32));
- memset(state->S_24_24, 0, 16 * sizeof(WebRtc_Word32));
- memset(state->S_24_16, 0, 8 * sizeof(WebRtc_Word32));
- memset(state->S_16_8, 0, 8 * sizeof(WebRtc_Word32));
-}
-
-////////////////////////////
-///// 8 kHz -> 48 kHz /////
-////////////////////////////
-
-// 8 -> 48 resampler
-void WebRtcSpl_Resample8khzTo48khz(const WebRtc_Word16* in, WebRtc_Word16* out,
- WebRtcSpl_State8khzTo48khz* state, WebRtc_Word32* tmpmem)
-{
- ///// 8 --> 16 /////
- // WebRtc_Word16 in[80]
- // WebRtc_Word32 out[160]
- /////
- WebRtcSpl_UpBy2ShortToInt(in, 80, tmpmem + 264, state->S_8_16);
-
- ///// 16 --> 12 /////
- // WebRtc_Word32 in[160]
- // WebRtc_Word32 out[120]
- /////
- // copy state to and from input array
- memcpy(tmpmem + 256, state->S_16_12, 8 * sizeof(WebRtc_Word32));
- memcpy(state->S_16_12, tmpmem + 416, 8 * sizeof(WebRtc_Word32));
- WebRtcSpl_Resample32khzTo24khz(tmpmem + 256, tmpmem + 240, 40);
-
- ///// 12 --> 24 /////
- // WebRtc_Word32 in[120]
- // WebRtc_Word16 out[240]
- /////
- WebRtcSpl_UpBy2IntToInt(tmpmem + 240, 120, tmpmem, state->S_12_24);
-
- ///// 24 --> 48 /////
- // WebRtc_Word32 in[240]
- // WebRtc_Word16 out[480]
- /////
- WebRtcSpl_UpBy2IntToShort(tmpmem, 240, out, state->S_24_48);
-}
-
-// initialize state of 8 -> 48 resampler
-void WebRtcSpl_ResetResample8khzTo48khz(WebRtcSpl_State8khzTo48khz* state)
-{
- memset(state->S_8_16, 0, 8 * sizeof(WebRtc_Word32));
- memset(state->S_16_12, 0, 8 * sizeof(WebRtc_Word32));
- memset(state->S_12_24, 0, 8 * sizeof(WebRtc_Word32));
- memset(state->S_24_48, 0, 8 * sizeof(WebRtc_Word32));
-}
diff --git a/trunk/src/common_audio/signal_processing/resample_by_2.c b/trunk/src/common_audio/signal_processing/resample_by_2.c
deleted file mode 100644
index e239db7..0000000
--- a/trunk/src/common_audio/signal_processing/resample_by_2.c
+++ /dev/null
@@ -1,181 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the resampling by two functions.
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-#ifdef WEBRTC_ARCH_ARM_V7A
-
-// allpass filter coefficients.
-static const WebRtc_UWord32 kResampleAllpass1[3] = {3284, 24441, 49528 << 15};
-static const WebRtc_UWord32 kResampleAllpass2[3] =
- {12199, 37471 << 15, 60255 << 15};
-
-// Multiply two 32-bit values and accumulate to another input value.
-// Return: state + ((diff * tbl_value) >> 16)
-
-static __inline WebRtc_Word32 MUL_ACCUM_1(WebRtc_Word32 tbl_value,
- WebRtc_Word32 diff,
- WebRtc_Word32 state) {
- WebRtc_Word32 result;
- __asm__("smlawb %r0, %r1, %r2, %r3": "=r"(result): "r"(diff),
- "r"(tbl_value), "r"(state));
- return result;
-}
-
-// Multiply two 32-bit values and accumulate to another input value.
-// Return: Return: state + (((diff << 1) * tbl_value) >> 32)
-//
-// The reason to introduce this function is that, in case we can't use smlawb
-// instruction (in MUL_ACCUM_1) due to input value range, we can still use
-// smmla to save some cycles.
-
-static __inline WebRtc_Word32 MUL_ACCUM_2(WebRtc_Word32 tbl_value,
- WebRtc_Word32 diff,
- WebRtc_Word32 state) {
- WebRtc_Word32 result;
- __asm__("smmla %r0, %r1, %r2, %r3": "=r"(result): "r"(diff << 1),
- "r"(tbl_value), "r"(state));
- return result;
-}
-
-#else
-
-// allpass filter coefficients.
-static const WebRtc_UWord16 kResampleAllpass1[3] = {3284, 24441, 49528};
-static const WebRtc_UWord16 kResampleAllpass2[3] = {12199, 37471, 60255};
-
-// Multiply a 32-bit value with a 16-bit value and accumulate to another input:
-#define MUL_ACCUM_1(a, b, c) WEBRTC_SPL_SCALEDIFF32(a, b, c)
-#define MUL_ACCUM_2(a, b, c) WEBRTC_SPL_SCALEDIFF32(a, b, c)
-
-#endif // WEBRTC_ARCH_ARM_V7A
-
-
-// decimator
-void WebRtcSpl_DownsampleBy2(const WebRtc_Word16* in, const WebRtc_Word16 len,
- WebRtc_Word16* out, WebRtc_Word32* filtState) {
- WebRtc_Word32 tmp1, tmp2, diff, in32, out32;
- WebRtc_Word16 i;
-
- register WebRtc_Word32 state0 = filtState[0];
- register WebRtc_Word32 state1 = filtState[1];
- register WebRtc_Word32 state2 = filtState[2];
- register WebRtc_Word32 state3 = filtState[3];
- register WebRtc_Word32 state4 = filtState[4];
- register WebRtc_Word32 state5 = filtState[5];
- register WebRtc_Word32 state6 = filtState[6];
- register WebRtc_Word32 state7 = filtState[7];
-
- for (i = (len >> 1); i > 0; i--) {
- // lower allpass filter
- in32 = (WebRtc_Word32)(*in++) << 10;
- diff = in32 - state1;
- tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state0);
- state0 = in32;
- diff = tmp1 - state2;
- tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state1);
- state1 = tmp1;
- diff = tmp2 - state3;
- state3 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state2);
- state2 = tmp2;
-
- // upper allpass filter
- in32 = (WebRtc_Word32)(*in++) << 10;
- diff = in32 - state5;
- tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state4);
- state4 = in32;
- diff = tmp1 - state6;
- tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state5);
- state5 = tmp1;
- diff = tmp2 - state7;
- state7 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state6);
- state6 = tmp2;
-
- // add two allpass outputs, divide by two and round
- out32 = (state3 + state7 + 1024) >> 11;
-
- // limit amplitude to prevent wrap-around, and write to output array
- *out++ = WebRtcSpl_SatW32ToW16(out32);
- }
-
- filtState[0] = state0;
- filtState[1] = state1;
- filtState[2] = state2;
- filtState[3] = state3;
- filtState[4] = state4;
- filtState[5] = state5;
- filtState[6] = state6;
- filtState[7] = state7;
-}
-
-
-void WebRtcSpl_UpsampleBy2(const WebRtc_Word16* in, WebRtc_Word16 len,
- WebRtc_Word16* out, WebRtc_Word32* filtState) {
- WebRtc_Word32 tmp1, tmp2, diff, in32, out32;
- WebRtc_Word16 i;
-
- register WebRtc_Word32 state0 = filtState[0];
- register WebRtc_Word32 state1 = filtState[1];
- register WebRtc_Word32 state2 = filtState[2];
- register WebRtc_Word32 state3 = filtState[3];
- register WebRtc_Word32 state4 = filtState[4];
- register WebRtc_Word32 state5 = filtState[5];
- register WebRtc_Word32 state6 = filtState[6];
- register WebRtc_Word32 state7 = filtState[7];
-
- for (i = len; i > 0; i--) {
- // lower allpass filter
- in32 = (WebRtc_Word32)(*in++) << 10;
- diff = in32 - state1;
- tmp1 = MUL_ACCUM_1(kResampleAllpass1[0], diff, state0);
- state0 = in32;
- diff = tmp1 - state2;
- tmp2 = MUL_ACCUM_1(kResampleAllpass1[1], diff, state1);
- state1 = tmp1;
- diff = tmp2 - state3;
- state3 = MUL_ACCUM_2(kResampleAllpass1[2], diff, state2);
- state2 = tmp2;
-
- // round; limit amplitude to prevent wrap-around; write to output array
- out32 = (state3 + 512) >> 10;
- *out++ = WebRtcSpl_SatW32ToW16(out32);
-
- // upper allpass filter
- diff = in32 - state5;
- tmp1 = MUL_ACCUM_1(kResampleAllpass2[0], diff, state4);
- state4 = in32;
- diff = tmp1 - state6;
- tmp2 = MUL_ACCUM_2(kResampleAllpass2[1], diff, state5);
- state5 = tmp1;
- diff = tmp2 - state7;
- state7 = MUL_ACCUM_2(kResampleAllpass2[2], diff, state6);
- state6 = tmp2;
-
- // round; limit amplitude to prevent wrap-around; write to output array
- out32 = (state7 + 512) >> 10;
- *out++ = WebRtcSpl_SatW32ToW16(out32);
- }
-
- filtState[0] = state0;
- filtState[1] = state1;
- filtState[2] = state2;
- filtState[3] = state3;
- filtState[4] = state4;
- filtState[5] = state5;
- filtState[6] = state6;
- filtState[7] = state7;
-}
diff --git a/trunk/src/common_audio/signal_processing/resample_by_2_internal.c b/trunk/src/common_audio/signal_processing/resample_by_2_internal.c
deleted file mode 100644
index cbd2395..0000000
--- a/trunk/src/common_audio/signal_processing/resample_by_2_internal.c
+++ /dev/null
@@ -1,679 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file contains some internal resampling functions.
- *
- */
-
-#include "resample_by_2_internal.h"
-
-// allpass filter coefficients.
-static const WebRtc_Word16 kResampleAllpass[2][3] = {
- {821, 6110, 12382},
- {3050, 9368, 15063}
-};
-
-//
-// decimator
-// input: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) OVERWRITTEN!
-// output: WebRtc_Word16 (saturated) (of length len/2)
-// state: filter state array; length = 8
-
-void WebRtcSpl_DownBy2IntToShort(WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word16 *out,
- WebRtc_Word32 *state)
-{
- WebRtc_Word32 tmp0, tmp1, diff;
- WebRtc_Word32 i;
-
- len >>= 1;
-
- // lower allpass filter (operates on even input samples)
- for (i = 0; i < len; i++)
- {
- tmp0 = in[i << 1];
- diff = tmp0 - state[1];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[0] + diff * kResampleAllpass[1][0];
- state[0] = tmp0;
- diff = tmp1 - state[2];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[1] + diff * kResampleAllpass[1][1];
- state[1] = tmp1;
- diff = tmp0 - state[3];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[3] = state[2] + diff * kResampleAllpass[1][2];
- state[2] = tmp0;
-
- // divide by two and store temporarily
- in[i << 1] = (state[3] >> 1);
- }
-
- in++;
-
- // upper allpass filter (operates on odd input samples)
- for (i = 0; i < len; i++)
- {
- tmp0 = in[i << 1];
- diff = tmp0 - state[5];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[4] + diff * kResampleAllpass[0][0];
- state[4] = tmp0;
- diff = tmp1 - state[6];
- // scale down and round
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[5] + diff * kResampleAllpass[0][1];
- state[5] = tmp1;
- diff = tmp0 - state[7];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[7] = state[6] + diff * kResampleAllpass[0][2];
- state[6] = tmp0;
-
- // divide by two and store temporarily
- in[i << 1] = (state[7] >> 1);
- }
-
- in--;
-
- // combine allpass outputs
- for (i = 0; i < len; i += 2)
- {
- // divide by two, add both allpass outputs and round
- tmp0 = (in[i << 1] + in[(i << 1) + 1]) >> 15;
- tmp1 = (in[(i << 1) + 2] + in[(i << 1) + 3]) >> 15;
- if (tmp0 > (WebRtc_Word32)0x00007FFF)
- tmp0 = 0x00007FFF;
- if (tmp0 < (WebRtc_Word32)0xFFFF8000)
- tmp0 = 0xFFFF8000;
- out[i] = (WebRtc_Word16)tmp0;
- if (tmp1 > (WebRtc_Word32)0x00007FFF)
- tmp1 = 0x00007FFF;
- if (tmp1 < (WebRtc_Word32)0xFFFF8000)
- tmp1 = 0xFFFF8000;
- out[i + 1] = (WebRtc_Word16)tmp1;
- }
-}
-
-//
-// decimator
-// input: WebRtc_Word16
-// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) (of length len/2)
-// state: filter state array; length = 8
-
-void WebRtcSpl_DownBy2ShortToInt(const WebRtc_Word16 *in,
- WebRtc_Word32 len,
- WebRtc_Word32 *out,
- WebRtc_Word32 *state)
-{
- WebRtc_Word32 tmp0, tmp1, diff;
- WebRtc_Word32 i;
-
- len >>= 1;
-
- // lower allpass filter (operates on even input samples)
- for (i = 0; i < len; i++)
- {
- tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
- diff = tmp0 - state[1];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[0] + diff * kResampleAllpass[1][0];
- state[0] = tmp0;
- diff = tmp1 - state[2];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[1] + diff * kResampleAllpass[1][1];
- state[1] = tmp1;
- diff = tmp0 - state[3];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[3] = state[2] + diff * kResampleAllpass[1][2];
- state[2] = tmp0;
-
- // divide by two and store temporarily
- out[i] = (state[3] >> 1);
- }
-
- in++;
-
- // upper allpass filter (operates on odd input samples)
- for (i = 0; i < len; i++)
- {
- tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
- diff = tmp0 - state[5];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[4] + diff * kResampleAllpass[0][0];
- state[4] = tmp0;
- diff = tmp1 - state[6];
- // scale down and round
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[5] + diff * kResampleAllpass[0][1];
- state[5] = tmp1;
- diff = tmp0 - state[7];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[7] = state[6] + diff * kResampleAllpass[0][2];
- state[6] = tmp0;
-
- // divide by two and store temporarily
- out[i] += (state[7] >> 1);
- }
-
- in--;
-}
-
-//
-// interpolator
-// input: WebRtc_Word16
-// output: WebRtc_Word32 (normalized, not saturated) (of length len*2)
-// state: filter state array; length = 8
-void WebRtcSpl_UpBy2ShortToInt(const WebRtc_Word16 *in, WebRtc_Word32 len, WebRtc_Word32 *out,
- WebRtc_Word32 *state)
-{
- WebRtc_Word32 tmp0, tmp1, diff;
- WebRtc_Word32 i;
-
- // upper allpass filter (generates odd output samples)
- for (i = 0; i < len; i++)
- {
- tmp0 = ((WebRtc_Word32)in[i] << 15) + (1 << 14);
- diff = tmp0 - state[5];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[4] + diff * kResampleAllpass[0][0];
- state[4] = tmp0;
- diff = tmp1 - state[6];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[5] + diff * kResampleAllpass[0][1];
- state[5] = tmp1;
- diff = tmp0 - state[7];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[7] = state[6] + diff * kResampleAllpass[0][2];
- state[6] = tmp0;
-
- // scale down, round and store
- out[i << 1] = state[7] >> 15;
- }
-
- out++;
-
- // lower allpass filter (generates even output samples)
- for (i = 0; i < len; i++)
- {
- tmp0 = ((WebRtc_Word32)in[i] << 15) + (1 << 14);
- diff = tmp0 - state[1];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[0] + diff * kResampleAllpass[1][0];
- state[0] = tmp0;
- diff = tmp1 - state[2];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[1] + diff * kResampleAllpass[1][1];
- state[1] = tmp1;
- diff = tmp0 - state[3];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[3] = state[2] + diff * kResampleAllpass[1][2];
- state[2] = tmp0;
-
- // scale down, round and store
- out[i << 1] = state[3] >> 15;
- }
-}
-
-//
-// interpolator
-// input: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384)
-// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) (of length len*2)
-// state: filter state array; length = 8
-void WebRtcSpl_UpBy2IntToInt(const WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word32 *out,
- WebRtc_Word32 *state)
-{
- WebRtc_Word32 tmp0, tmp1, diff;
- WebRtc_Word32 i;
-
- // upper allpass filter (generates odd output samples)
- for (i = 0; i < len; i++)
- {
- tmp0 = in[i];
- diff = tmp0 - state[5];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[4] + diff * kResampleAllpass[0][0];
- state[4] = tmp0;
- diff = tmp1 - state[6];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[5] + diff * kResampleAllpass[0][1];
- state[5] = tmp1;
- diff = tmp0 - state[7];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[7] = state[6] + diff * kResampleAllpass[0][2];
- state[6] = tmp0;
-
- // scale down, round and store
- out[i << 1] = state[7];
- }
-
- out++;
-
- // lower allpass filter (generates even output samples)
- for (i = 0; i < len; i++)
- {
- tmp0 = in[i];
- diff = tmp0 - state[1];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[0] + diff * kResampleAllpass[1][0];
- state[0] = tmp0;
- diff = tmp1 - state[2];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[1] + diff * kResampleAllpass[1][1];
- state[1] = tmp1;
- diff = tmp0 - state[3];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[3] = state[2] + diff * kResampleAllpass[1][2];
- state[2] = tmp0;
-
- // scale down, round and store
- out[i << 1] = state[3];
- }
-}
-
-//
-// interpolator
-// input: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384)
-// output: WebRtc_Word16 (saturated) (of length len*2)
-// state: filter state array; length = 8
-void WebRtcSpl_UpBy2IntToShort(const WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word16 *out,
- WebRtc_Word32 *state)
-{
- WebRtc_Word32 tmp0, tmp1, diff;
- WebRtc_Word32 i;
-
- // upper allpass filter (generates odd output samples)
- for (i = 0; i < len; i++)
- {
- tmp0 = in[i];
- diff = tmp0 - state[5];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[4] + diff * kResampleAllpass[0][0];
- state[4] = tmp0;
- diff = tmp1 - state[6];
- // scale down and round
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[5] + diff * kResampleAllpass[0][1];
- state[5] = tmp1;
- diff = tmp0 - state[7];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[7] = state[6] + diff * kResampleAllpass[0][2];
- state[6] = tmp0;
-
- // scale down, saturate and store
- tmp1 = state[7] >> 15;
- if (tmp1 > (WebRtc_Word32)0x00007FFF)
- tmp1 = 0x00007FFF;
- if (tmp1 < (WebRtc_Word32)0xFFFF8000)
- tmp1 = 0xFFFF8000;
- out[i << 1] = (WebRtc_Word16)tmp1;
- }
-
- out++;
-
- // lower allpass filter (generates even output samples)
- for (i = 0; i < len; i++)
- {
- tmp0 = in[i];
- diff = tmp0 - state[1];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[0] + diff * kResampleAllpass[1][0];
- state[0] = tmp0;
- diff = tmp1 - state[2];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[1] + diff * kResampleAllpass[1][1];
- state[1] = tmp1;
- diff = tmp0 - state[3];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[3] = state[2] + diff * kResampleAllpass[1][2];
- state[2] = tmp0;
-
- // scale down, saturate and store
- tmp1 = state[3] >> 15;
- if (tmp1 > (WebRtc_Word32)0x00007FFF)
- tmp1 = 0x00007FFF;
- if (tmp1 < (WebRtc_Word32)0xFFFF8000)
- tmp1 = 0xFFFF8000;
- out[i << 1] = (WebRtc_Word16)tmp1;
- }
-}
-
-// lowpass filter
-// input: WebRtc_Word16
-// output: WebRtc_Word32 (normalized, not saturated)
-// state: filter state array; length = 8
-void WebRtcSpl_LPBy2ShortToInt(const WebRtc_Word16* in, WebRtc_Word32 len, WebRtc_Word32* out,
- WebRtc_Word32* state)
-{
- WebRtc_Word32 tmp0, tmp1, diff;
- WebRtc_Word32 i;
-
- len >>= 1;
-
- // lower allpass filter: odd input -> even output samples
- in++;
- // initial state of polyphase delay element
- tmp0 = state[12];
- for (i = 0; i < len; i++)
- {
- diff = tmp0 - state[1];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[0] + diff * kResampleAllpass[1][0];
- state[0] = tmp0;
- diff = tmp1 - state[2];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[1] + diff * kResampleAllpass[1][1];
- state[1] = tmp1;
- diff = tmp0 - state[3];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[3] = state[2] + diff * kResampleAllpass[1][2];
- state[2] = tmp0;
-
- // scale down, round and store
- out[i << 1] = state[3] >> 1;
- tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
- }
- in--;
-
- // upper allpass filter: even input -> even output samples
- for (i = 0; i < len; i++)
- {
- tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
- diff = tmp0 - state[5];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[4] + diff * kResampleAllpass[0][0];
- state[4] = tmp0;
- diff = tmp1 - state[6];
- // scale down and round
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[5] + diff * kResampleAllpass[0][1];
- state[5] = tmp1;
- diff = tmp0 - state[7];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[7] = state[6] + diff * kResampleAllpass[0][2];
- state[6] = tmp0;
-
- // average the two allpass outputs, scale down and store
- out[i << 1] = (out[i << 1] + (state[7] >> 1)) >> 15;
- }
-
- // switch to odd output samples
- out++;
-
- // lower allpass filter: even input -> odd output samples
- for (i = 0; i < len; i++)
- {
- tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
- diff = tmp0 - state[9];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[8] + diff * kResampleAllpass[1][0];
- state[8] = tmp0;
- diff = tmp1 - state[10];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[9] + diff * kResampleAllpass[1][1];
- state[9] = tmp1;
- diff = tmp0 - state[11];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[11] = state[10] + diff * kResampleAllpass[1][2];
- state[10] = tmp0;
-
- // scale down, round and store
- out[i << 1] = state[11] >> 1;
- }
-
- // upper allpass filter: odd input -> odd output samples
- in++;
- for (i = 0; i < len; i++)
- {
- tmp0 = ((WebRtc_Word32)in[i << 1] << 15) + (1 << 14);
- diff = tmp0 - state[13];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[12] + diff * kResampleAllpass[0][0];
- state[12] = tmp0;
- diff = tmp1 - state[14];
- // scale down and round
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[13] + diff * kResampleAllpass[0][1];
- state[13] = tmp1;
- diff = tmp0 - state[15];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[15] = state[14] + diff * kResampleAllpass[0][2];
- state[14] = tmp0;
-
- // average the two allpass outputs, scale down and store
- out[i << 1] = (out[i << 1] + (state[15] >> 1)) >> 15;
- }
-}
-
-// lowpass filter
-// input: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384)
-// output: WebRtc_Word32 (normalized, not saturated)
-// state: filter state array; length = 8
-void WebRtcSpl_LPBy2IntToInt(const WebRtc_Word32* in, WebRtc_Word32 len, WebRtc_Word32* out,
- WebRtc_Word32* state)
-{
- WebRtc_Word32 tmp0, tmp1, diff;
- WebRtc_Word32 i;
-
- len >>= 1;
-
- // lower allpass filter: odd input -> even output samples
- in++;
- // initial state of polyphase delay element
- tmp0 = state[12];
- for (i = 0; i < len; i++)
- {
- diff = tmp0 - state[1];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[0] + diff * kResampleAllpass[1][0];
- state[0] = tmp0;
- diff = tmp1 - state[2];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[1] + diff * kResampleAllpass[1][1];
- state[1] = tmp1;
- diff = tmp0 - state[3];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[3] = state[2] + diff * kResampleAllpass[1][2];
- state[2] = tmp0;
-
- // scale down, round and store
- out[i << 1] = state[3] >> 1;
- tmp0 = in[i << 1];
- }
- in--;
-
- // upper allpass filter: even input -> even output samples
- for (i = 0; i < len; i++)
- {
- tmp0 = in[i << 1];
- diff = tmp0 - state[5];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[4] + diff * kResampleAllpass[0][0];
- state[4] = tmp0;
- diff = tmp1 - state[6];
- // scale down and round
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[5] + diff * kResampleAllpass[0][1];
- state[5] = tmp1;
- diff = tmp0 - state[7];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[7] = state[6] + diff * kResampleAllpass[0][2];
- state[6] = tmp0;
-
- // average the two allpass outputs, scale down and store
- out[i << 1] = (out[i << 1] + (state[7] >> 1)) >> 15;
- }
-
- // switch to odd output samples
- out++;
-
- // lower allpass filter: even input -> odd output samples
- for (i = 0; i < len; i++)
- {
- tmp0 = in[i << 1];
- diff = tmp0 - state[9];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[8] + diff * kResampleAllpass[1][0];
- state[8] = tmp0;
- diff = tmp1 - state[10];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[9] + diff * kResampleAllpass[1][1];
- state[9] = tmp1;
- diff = tmp0 - state[11];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[11] = state[10] + diff * kResampleAllpass[1][2];
- state[10] = tmp0;
-
- // scale down, round and store
- out[i << 1] = state[11] >> 1;
- }
-
- // upper allpass filter: odd input -> odd output samples
- in++;
- for (i = 0; i < len; i++)
- {
- tmp0 = in[i << 1];
- diff = tmp0 - state[13];
- // scale down and round
- diff = (diff + (1 << 13)) >> 14;
- tmp1 = state[12] + diff * kResampleAllpass[0][0];
- state[12] = tmp0;
- diff = tmp1 - state[14];
- // scale down and round
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- tmp0 = state[13] + diff * kResampleAllpass[0][1];
- state[13] = tmp1;
- diff = tmp0 - state[15];
- // scale down and truncate
- diff = diff >> 14;
- if (diff < 0)
- diff += 1;
- state[15] = state[14] + diff * kResampleAllpass[0][2];
- state[14] = tmp0;
-
- // average the two allpass outputs, scale down and store
- out[i << 1] = (out[i << 1] + (state[15] >> 1)) >> 15;
- }
-}
diff --git a/trunk/src/common_audio/signal_processing/resample_by_2_internal.h b/trunk/src/common_audio/signal_processing/resample_by_2_internal.h
deleted file mode 100644
index b6ac9f0..0000000
--- a/trunk/src/common_audio/signal_processing/resample_by_2_internal.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file contains some internal resampling functions.
- *
- */
-
-#ifndef WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
-#define WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
-
-#include "typedefs.h"
-
-/*******************************************************************
- * resample_by_2_fast.c
- * Functions for internal use in the other resample functions
- ******************************************************************/
-void WebRtcSpl_DownBy2IntToShort(WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word16 *out,
- WebRtc_Word32 *state);
-
-void WebRtcSpl_DownBy2ShortToInt(const WebRtc_Word16 *in, WebRtc_Word32 len,
- WebRtc_Word32 *out, WebRtc_Word32 *state);
-
-void WebRtcSpl_UpBy2ShortToInt(const WebRtc_Word16 *in, WebRtc_Word32 len,
- WebRtc_Word32 *out, WebRtc_Word32 *state);
-
-void WebRtcSpl_UpBy2IntToInt(const WebRtc_Word32 *in, WebRtc_Word32 len, WebRtc_Word32 *out,
- WebRtc_Word32 *state);
-
-void WebRtcSpl_UpBy2IntToShort(const WebRtc_Word32 *in, WebRtc_Word32 len,
- WebRtc_Word16 *out, WebRtc_Word32 *state);
-
-void WebRtcSpl_LPBy2ShortToInt(const WebRtc_Word16* in, WebRtc_Word32 len,
- WebRtc_Word32* out, WebRtc_Word32* state);
-
-void WebRtcSpl_LPBy2IntToInt(const WebRtc_Word32* in, WebRtc_Word32 len, WebRtc_Word32* out,
- WebRtc_Word32* state);
-
-#endif // WEBRTC_SPL_RESAMPLE_BY_2_INTERNAL_H_
diff --git a/trunk/src/common_audio/signal_processing/resample_fractional.c b/trunk/src/common_audio/signal_processing/resample_fractional.c
deleted file mode 100644
index 51003d4..0000000
--- a/trunk/src/common_audio/signal_processing/resample_fractional.c
+++ /dev/null
@@ -1,242 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the resampling functions between 48, 44, 32 and 24 kHz.
- * The description headers can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-// interpolation coefficients
-static const WebRtc_Word16 kCoefficients48To32[2][8] = {
- {778, -2050, 1087, 23285, 12903, -3783, 441, 222},
- {222, 441, -3783, 12903, 23285, 1087, -2050, 778}
-};
-
-static const WebRtc_Word16 kCoefficients32To24[3][8] = {
- {767, -2362, 2434, 24406, 10620, -3838, 721, 90},
- {386, -381, -2646, 19062, 19062, -2646, -381, 386},
- {90, 721, -3838, 10620, 24406, 2434, -2362, 767}
-};
-
-static const WebRtc_Word16 kCoefficients44To32[4][9] = {
- {117, -669, 2245, -6183, 26267, 13529, -3245, 845, -138},
- {-101, 612, -2283, 8532, 29790, -5138, 1789, -524, 91},
- {50, -292, 1016, -3064, 32010, 3933, -1147, 315, -53},
- {-156, 974, -3863, 18603, 21691, -6246, 2353, -712, 126}
-};
-
-// Resampling ratio: 2/3
-// input: WebRtc_Word32 (normalized, not saturated) :: size 3 * K
-// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) :: size 2 * K
-// K: number of blocks
-
-void WebRtcSpl_Resample48khzTo32khz(const WebRtc_Word32 *In, WebRtc_Word32 *Out,
- const WebRtc_Word32 K)
-{
- /////////////////////////////////////////////////////////////
- // Filter operation:
- //
- // Perform resampling (3 input samples -> 2 output samples);
- // process in sub blocks of size 3 samples.
- WebRtc_Word32 tmp;
- WebRtc_Word32 m;
-
- for (m = 0; m < K; m++)
- {
- tmp = 1 << 14;
- tmp += kCoefficients48To32[0][0] * In[0];
- tmp += kCoefficients48To32[0][1] * In[1];
- tmp += kCoefficients48To32[0][2] * In[2];
- tmp += kCoefficients48To32[0][3] * In[3];
- tmp += kCoefficients48To32[0][4] * In[4];
- tmp += kCoefficients48To32[0][5] * In[5];
- tmp += kCoefficients48To32[0][6] * In[6];
- tmp += kCoefficients48To32[0][7] * In[7];
- Out[0] = tmp;
-
- tmp = 1 << 14;
- tmp += kCoefficients48To32[1][0] * In[1];
- tmp += kCoefficients48To32[1][1] * In[2];
- tmp += kCoefficients48To32[1][2] * In[3];
- tmp += kCoefficients48To32[1][3] * In[4];
- tmp += kCoefficients48To32[1][4] * In[5];
- tmp += kCoefficients48To32[1][5] * In[6];
- tmp += kCoefficients48To32[1][6] * In[7];
- tmp += kCoefficients48To32[1][7] * In[8];
- Out[1] = tmp;
-
- // update pointers
- In += 3;
- Out += 2;
- }
-}
-
-// Resampling ratio: 3/4
-// input: WebRtc_Word32 (normalized, not saturated) :: size 4 * K
-// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) :: size 3 * K
-// K: number of blocks
-
-void WebRtcSpl_Resample32khzTo24khz(const WebRtc_Word32 *In, WebRtc_Word32 *Out,
- const WebRtc_Word32 K)
-{
- /////////////////////////////////////////////////////////////
- // Filter operation:
- //
- // Perform resampling (4 input samples -> 3 output samples);
- // process in sub blocks of size 4 samples.
- WebRtc_Word32 m;
- WebRtc_Word32 tmp;
-
- for (m = 0; m < K; m++)
- {
- tmp = 1 << 14;
- tmp += kCoefficients32To24[0][0] * In[0];
- tmp += kCoefficients32To24[0][1] * In[1];
- tmp += kCoefficients32To24[0][2] * In[2];
- tmp += kCoefficients32To24[0][3] * In[3];
- tmp += kCoefficients32To24[0][4] * In[4];
- tmp += kCoefficients32To24[0][5] * In[5];
- tmp += kCoefficients32To24[0][6] * In[6];
- tmp += kCoefficients32To24[0][7] * In[7];
- Out[0] = tmp;
-
- tmp = 1 << 14;
- tmp += kCoefficients32To24[1][0] * In[1];
- tmp += kCoefficients32To24[1][1] * In[2];
- tmp += kCoefficients32To24[1][2] * In[3];
- tmp += kCoefficients32To24[1][3] * In[4];
- tmp += kCoefficients32To24[1][4] * In[5];
- tmp += kCoefficients32To24[1][5] * In[6];
- tmp += kCoefficients32To24[1][6] * In[7];
- tmp += kCoefficients32To24[1][7] * In[8];
- Out[1] = tmp;
-
- tmp = 1 << 14;
- tmp += kCoefficients32To24[2][0] * In[2];
- tmp += kCoefficients32To24[2][1] * In[3];
- tmp += kCoefficients32To24[2][2] * In[4];
- tmp += kCoefficients32To24[2][3] * In[5];
- tmp += kCoefficients32To24[2][4] * In[6];
- tmp += kCoefficients32To24[2][5] * In[7];
- tmp += kCoefficients32To24[2][6] * In[8];
- tmp += kCoefficients32To24[2][7] * In[9];
- Out[2] = tmp;
-
- // update pointers
- In += 4;
- Out += 3;
- }
-}
-
-//
-// fractional resampling filters
-// Fout = 11/16 * Fin
-// Fout = 8/11 * Fin
-//
-
-// compute two inner-products and store them to output array
-static void WebRtcSpl_ResampDotProduct(const WebRtc_Word32 *in1, const WebRtc_Word32 *in2,
- const WebRtc_Word16 *coef_ptr, WebRtc_Word32 *out1,
- WebRtc_Word32 *out2)
-{
- WebRtc_Word32 tmp1 = 16384;
- WebRtc_Word32 tmp2 = 16384;
- WebRtc_Word16 coef;
-
- coef = coef_ptr[0];
- tmp1 += coef * in1[0];
- tmp2 += coef * in2[-0];
-
- coef = coef_ptr[1];
- tmp1 += coef * in1[1];
- tmp2 += coef * in2[-1];
-
- coef = coef_ptr[2];
- tmp1 += coef * in1[2];
- tmp2 += coef * in2[-2];
-
- coef = coef_ptr[3];
- tmp1 += coef * in1[3];
- tmp2 += coef * in2[-3];
-
- coef = coef_ptr[4];
- tmp1 += coef * in1[4];
- tmp2 += coef * in2[-4];
-
- coef = coef_ptr[5];
- tmp1 += coef * in1[5];
- tmp2 += coef * in2[-5];
-
- coef = coef_ptr[6];
- tmp1 += coef * in1[6];
- tmp2 += coef * in2[-6];
-
- coef = coef_ptr[7];
- tmp1 += coef * in1[7];
- tmp2 += coef * in2[-7];
-
- coef = coef_ptr[8];
- *out1 = tmp1 + coef * in1[8];
- *out2 = tmp2 + coef * in2[-8];
-}
-
-// Resampling ratio: 8/11
-// input: WebRtc_Word32 (normalized, not saturated) :: size 11 * K
-// output: WebRtc_Word32 (shifted 15 positions to the left, + offset 16384) :: size 8 * K
-// K: number of blocks
-
-void WebRtcSpl_Resample44khzTo32khz(const WebRtc_Word32 *In, WebRtc_Word32 *Out,
- const WebRtc_Word32 K)
-{
- /////////////////////////////////////////////////////////////
- // Filter operation:
- //
- // Perform resampling (11 input samples -> 8 output samples);
- // process in sub blocks of size 11 samples.
- WebRtc_Word32 tmp;
- WebRtc_Word32 m;
-
- for (m = 0; m < K; m++)
- {
- tmp = 1 << 14;
-
- // first output sample
- Out[0] = ((WebRtc_Word32)In[3] << 15) + tmp;
-
- // sum and accumulate filter coefficients and input samples
- tmp += kCoefficients44To32[3][0] * In[5];
- tmp += kCoefficients44To32[3][1] * In[6];
- tmp += kCoefficients44To32[3][2] * In[7];
- tmp += kCoefficients44To32[3][3] * In[8];
- tmp += kCoefficients44To32[3][4] * In[9];
- tmp += kCoefficients44To32[3][5] * In[10];
- tmp += kCoefficients44To32[3][6] * In[11];
- tmp += kCoefficients44To32[3][7] * In[12];
- tmp += kCoefficients44To32[3][8] * In[13];
- Out[4] = tmp;
-
- // sum and accumulate filter coefficients and input samples
- WebRtcSpl_ResampDotProduct(&In[0], &In[17], kCoefficients44To32[0], &Out[1], &Out[7]);
-
- // sum and accumulate filter coefficients and input samples
- WebRtcSpl_ResampDotProduct(&In[2], &In[15], kCoefficients44To32[1], &Out[2], &Out[6]);
-
- // sum and accumulate filter coefficients and input samples
- WebRtcSpl_ResampDotProduct(&In[3], &In[14], kCoefficients44To32[2], &Out[3], &Out[5]);
-
- // update pointers
- In += 11;
- Out += 8;
- }
-}
diff --git a/trunk/src/common_audio/signal_processing/signal_processing.gypi b/trunk/src/common_audio/signal_processing/signal_processing.gypi
deleted file mode 100644
index c67bf7c..0000000
--- a/trunk/src/common_audio/signal_processing/signal_processing.gypi
+++ /dev/null
@@ -1,85 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'signal_processing',
- 'type': '<(library)',
- 'include_dirs': [
- 'include',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- ],
- },
- 'sources': [
- 'include/signal_processing_library.h',
- 'include/spl_inl.h',
- 'auto_corr_to_refl_coef.c',
- 'auto_correlation.c',
- 'complex_fft.c',
- 'complex_bit_reverse.c',
- 'copy_set_operations.c',
- 'cross_correlation.c',
- 'division_operations.c',
- 'dot_product_with_scale.c',
- 'downsample_fast.c',
- 'energy.c',
- 'filter_ar.c',
- 'filter_ar_fast_q12.c',
- 'filter_ma_fast_q12.c',
- 'get_hanning_window.c',
- 'get_scaling_square.c',
- 'ilbc_specific_functions.c',
- 'levinson_durbin.c',
- 'lpc_to_refl_coef.c',
- 'min_max_operations.c',
- 'randomization_functions.c',
- 'refl_coef_to_lpc.c',
- 'resample.c',
- 'resample_48khz.c',
- 'resample_by_2.c',
- 'resample_by_2_internal.c',
- 'resample_by_2_internal.h',
- 'resample_fractional.c',
- 'spl_sqrt.c',
- 'spl_sqrt_floor.c',
- 'spl_version.c',
- 'splitting_filter.c',
- 'sqrt_of_one_minus_x_squared.c',
- 'vector_scaling_operations.c',
- ],
- }, # spl
- ], # targets
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'signal_processing_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'signal_processing',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'sources': [
- 'signal_processing_unittest.cc',
- ],
- }, # spl_unittests
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/common_audio/signal_processing/signal_processing_unittest.cc b/trunk/src/common_audio/signal_processing/signal_processing_unittest.cc
deleted file mode 100644
index f6d1123..0000000
--- a/trunk/src/common_audio/signal_processing/signal_processing_unittest.cc
+++ /dev/null
@@ -1,454 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "signal_processing_library.h"
-#include "gtest/gtest.h"
-
-class SplTest : public testing::Test {
- protected:
- virtual ~SplTest() {
- }
- void SetUp() {
- }
- void TearDown() {
- }
-};
-
-TEST_F(SplTest, MacroTest) {
- // Macros with inputs.
- int A = 10;
- int B = 21;
- int a = -3;
- int b = WEBRTC_SPL_WORD32_MAX;
- int nr = 2;
- int d_ptr2 = 0;
-
- EXPECT_EQ(10, WEBRTC_SPL_MIN(A, B));
- EXPECT_EQ(21, WEBRTC_SPL_MAX(A, B));
-
- EXPECT_EQ(3, WEBRTC_SPL_ABS_W16(a));
- EXPECT_EQ(3, WEBRTC_SPL_ABS_W32(a));
- EXPECT_EQ(0, WEBRTC_SPL_GET_BYTE(&B, nr));
- WEBRTC_SPL_SET_BYTE(&d_ptr2, 1, nr);
- EXPECT_EQ(65536, d_ptr2);
-
- EXPECT_EQ(-63, WEBRTC_SPL_MUL(a, B));
- EXPECT_EQ(-2147483645, WEBRTC_SPL_MUL(a, b));
- EXPECT_EQ(2147483651u, WEBRTC_SPL_UMUL(a, b));
- b = WEBRTC_SPL_WORD16_MAX >> 1;
- EXPECT_EQ(65535u, WEBRTC_SPL_UMUL_RSFT16(a, b));
- EXPECT_EQ(1073627139u, WEBRTC_SPL_UMUL_16_16(a, b));
- EXPECT_EQ(16382u, WEBRTC_SPL_UMUL_16_16_RSFT16(a, b));
- EXPECT_EQ(4294918147u, WEBRTC_SPL_UMUL_32_16(a, b));
- EXPECT_EQ(65535u, WEBRTC_SPL_UMUL_32_16_RSFT16(a, b));
- EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_U16(a, b));
-
- a = b;
- b = -3;
- EXPECT_EQ(-5461, WEBRTC_SPL_DIV(a, b));
- EXPECT_EQ(0u, WEBRTC_SPL_UDIV(a, b));
-
- EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT16(a, b));
- EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_32_RSFT15(a, b));
- EXPECT_EQ(-3, WEBRTC_SPL_MUL_16_32_RSFT14(a, b));
- EXPECT_EQ(-24, WEBRTC_SPL_MUL_16_32_RSFT11(a, b));
-
- int a32 = WEBRTC_SPL_WORD32_MAX;
- int a32a = (WEBRTC_SPL_WORD32_MAX >> 16);
- int a32b = (WEBRTC_SPL_WORD32_MAX & 0x0000ffff);
- EXPECT_EQ(5, WEBRTC_SPL_MUL_32_32_RSFT32(a32a, a32b, A));
- EXPECT_EQ(5, WEBRTC_SPL_MUL_32_32_RSFT32BI(a32, A));
-
- EXPECT_EQ(-49149, WEBRTC_SPL_MUL_16_16(a, b));
- EXPECT_EQ(-12288, WEBRTC_SPL_MUL_16_16_RSFT(a, b, 2));
-
- EXPECT_EQ(-12287, WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(a, b, 2));
- EXPECT_EQ(-1, WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(a, b));
-
- EXPECT_EQ(16380, WEBRTC_SPL_ADD_SAT_W32(a, b));
- EXPECT_EQ(21, WEBRTC_SPL_SAT(a, A, B));
- EXPECT_EQ(21, WEBRTC_SPL_SAT(a, B, A));
- EXPECT_EQ(-49149, WEBRTC_SPL_MUL_32_16(a, b));
-
- EXPECT_EQ(16386, WEBRTC_SPL_SUB_SAT_W32(a, b));
- EXPECT_EQ(16380, WEBRTC_SPL_ADD_SAT_W16(a, b));
- EXPECT_EQ(16386, WEBRTC_SPL_SUB_SAT_W16(a, b));
-
- EXPECT_TRUE(WEBRTC_SPL_IS_NEG(b));
-
- // Shifting with negative numbers allowed
- int shift_amount = 1; // Workaround compiler warning using variable here.
- // Positive means left shift
- EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W16(a, shift_amount));
- EXPECT_EQ(32766, WEBRTC_SPL_SHIFT_W32(a, shift_amount));
-
- // Shifting with negative numbers not allowed
- // We cannot do casting here due to signed/unsigned problem
- EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W16(a, 1));
- EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W16(a, 1));
- EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_W32(a, 1));
- EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_W32(a, 1));
-
- EXPECT_EQ(8191, WEBRTC_SPL_RSHIFT_U16(a, 1));
- EXPECT_EQ(32766, WEBRTC_SPL_LSHIFT_U16(a, 1));
- EXPECT_EQ(8191u, WEBRTC_SPL_RSHIFT_U32(a, 1));
- EXPECT_EQ(32766u, WEBRTC_SPL_LSHIFT_U32(a, 1));
-
- EXPECT_EQ(1470, WEBRTC_SPL_RAND(A));
-}
-
-TEST_F(SplTest, InlineTest) {
- WebRtc_Word16 a = 121;
- WebRtc_Word16 b = -17;
- WebRtc_Word32 A = 111121;
- WebRtc_Word32 B = -1711;
- char bVersion[8];
-
- EXPECT_EQ(104, WebRtcSpl_AddSatW16(a, b));
- EXPECT_EQ(138, WebRtcSpl_SubSatW16(a, b));
-
- EXPECT_EQ(109410, WebRtcSpl_AddSatW32(A, B));
- EXPECT_EQ(112832, WebRtcSpl_SubSatW32(A, B));
-
- EXPECT_EQ(17, WebRtcSpl_GetSizeInBits(A));
- EXPECT_EQ(14, WebRtcSpl_NormW32(A));
- EXPECT_EQ(4, WebRtcSpl_NormW16(B));
- EXPECT_EQ(15, WebRtcSpl_NormU32(A));
-
- EXPECT_EQ(0, WebRtcSpl_get_version(bVersion, 8));
-}
-
-TEST_F(SplTest, MathOperationsTest) {
- int A = 117;
- WebRtc_Word32 num = 117;
- WebRtc_Word32 den = -5;
- WebRtc_UWord16 denU = 5;
- EXPECT_EQ(10, WebRtcSpl_Sqrt(A));
- EXPECT_EQ(10, WebRtcSpl_SqrtFloor(A));
-
-
- EXPECT_EQ(-91772805, WebRtcSpl_DivResultInQ31(den, num));
- EXPECT_EQ(-23, WebRtcSpl_DivW32W16ResW16(num, (WebRtc_Word16)den));
- EXPECT_EQ(-23, WebRtcSpl_DivW32W16(num, (WebRtc_Word16)den));
- EXPECT_EQ(23u, WebRtcSpl_DivU32U16(num, denU));
- EXPECT_EQ(0, WebRtcSpl_DivW32HiLow(128, 0, 256));
-}
-
-TEST_F(SplTest, BasicArrayOperationsTest) {
- const int kVectorSize = 4;
- int B[] = {4, 12, 133, 1100};
- WebRtc_UWord8 b8[kVectorSize];
- WebRtc_Word16 b16[kVectorSize];
- WebRtc_Word32 b32[kVectorSize];
-
- WebRtc_UWord8 bTmp8[kVectorSize];
- WebRtc_Word16 bTmp16[kVectorSize];
- WebRtc_Word32 bTmp32[kVectorSize];
-
- WebRtcSpl_MemSetW16(b16, 3, kVectorSize);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ(3, b16[kk]);
- }
- EXPECT_EQ(kVectorSize, WebRtcSpl_ZerosArrayW16(b16, kVectorSize));
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ(0, b16[kk]);
- }
- EXPECT_EQ(kVectorSize, WebRtcSpl_OnesArrayW16(b16, kVectorSize));
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ(1, b16[kk]);
- }
- WebRtcSpl_MemSetW32(b32, 3, kVectorSize);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ(3, b32[kk]);
- }
- EXPECT_EQ(kVectorSize, WebRtcSpl_ZerosArrayW32(b32, kVectorSize));
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ(0, b32[kk]);
- }
- EXPECT_EQ(kVectorSize, WebRtcSpl_OnesArrayW32(b32, kVectorSize));
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ(1, b32[kk]);
- }
- for (int kk = 0; kk < kVectorSize; ++kk) {
- bTmp8[kk] = (WebRtc_Word8)kk;
- bTmp16[kk] = (WebRtc_Word16)kk;
- bTmp32[kk] = (WebRtc_Word32)kk;
- }
- WEBRTC_SPL_MEMCPY_W8(b8, bTmp8, kVectorSize);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ(b8[kk], bTmp8[kk]);
- }
- WEBRTC_SPL_MEMCPY_W16(b16, bTmp16, kVectorSize);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ(b16[kk], bTmp16[kk]);
- }
-// WEBRTC_SPL_MEMCPY_W32(b32, bTmp32, kVectorSize);
-// for (int kk = 0; kk < kVectorSize; ++kk) {
-// EXPECT_EQ(b32[kk], bTmp32[kk]);
-// }
- EXPECT_EQ(2, WebRtcSpl_CopyFromEndW16(b16, kVectorSize, 2, bTmp16));
- for (int kk = 0; kk < 2; ++kk) {
- EXPECT_EQ(kk+2, bTmp16[kk]);
- }
-
- for (int kk = 0; kk < kVectorSize; ++kk) {
- b32[kk] = B[kk];
- b16[kk] = (WebRtc_Word16)B[kk];
- }
- WebRtcSpl_VectorBitShiftW32ToW16(bTmp16, kVectorSize, b32, 1);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
- }
- WebRtcSpl_VectorBitShiftW16(bTmp16, kVectorSize, b16, 1);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ((B[kk]>>1), bTmp16[kk]);
- }
- WebRtcSpl_VectorBitShiftW32(bTmp32, kVectorSize, b32, 1);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ((B[kk]>>1), bTmp32[kk]);
- }
-
- WebRtcSpl_MemCpyReversedOrder(&bTmp16[3], b16, kVectorSize);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ(b16[3-kk], bTmp16[kk]);
- }
-}
-
-TEST_F(SplTest, MinMaxOperationsTest) {
- const int kVectorSize = 4;
- int B[] = {4, 12, 133, -1100};
- WebRtc_Word16 b16[kVectorSize];
- WebRtc_Word32 b32[kVectorSize];
-
- for (int kk = 0; kk < kVectorSize; ++kk) {
- b16[kk] = B[kk];
- b32[kk] = B[kk];
- }
-
- EXPECT_EQ(1100, WebRtcSpl_MaxAbsValueW16(b16, kVectorSize));
- EXPECT_EQ(1100, WebRtcSpl_MaxAbsValueW32(b32, kVectorSize));
- EXPECT_EQ(133, WebRtcSpl_MaxValueW16(b16, kVectorSize));
- EXPECT_EQ(133, WebRtcSpl_MaxValueW32(b32, kVectorSize));
- EXPECT_EQ(3, WebRtcSpl_MaxAbsIndexW16(b16, kVectorSize));
- EXPECT_EQ(2, WebRtcSpl_MaxIndexW16(b16, kVectorSize));
- EXPECT_EQ(2, WebRtcSpl_MaxIndexW32(b32, kVectorSize));
-
- EXPECT_EQ(-1100, WebRtcSpl_MinValueW16(b16, kVectorSize));
- EXPECT_EQ(-1100, WebRtcSpl_MinValueW32(b32, kVectorSize));
- EXPECT_EQ(3, WebRtcSpl_MinIndexW16(b16, kVectorSize));
- EXPECT_EQ(3, WebRtcSpl_MinIndexW32(b32, kVectorSize));
-
- EXPECT_EQ(0, WebRtcSpl_GetScalingSquare(b16, kVectorSize, 1));
-}
-
-TEST_F(SplTest, VectorOperationsTest) {
- const int kVectorSize = 4;
- int B[] = {4, 12, 133, 1100};
- WebRtc_Word16 a16[kVectorSize];
- WebRtc_Word16 b16[kVectorSize];
- WebRtc_Word32 b32[kVectorSize];
- WebRtc_Word16 bTmp16[kVectorSize];
-
- for (int kk = 0; kk < kVectorSize; ++kk) {
- a16[kk] = B[kk];
- b16[kk] = B[kk];
- }
-
- WebRtcSpl_AffineTransformVector(bTmp16, b16, 3, 7, 2, kVectorSize);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ((B[kk]*3+7)>>2, bTmp16[kk]);
- }
- WebRtcSpl_ScaleAndAddVectorsWithRound(b16, 3, b16, 2, 2, bTmp16, kVectorSize);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ((B[kk]*3+B[kk]*2+2)>>2, bTmp16[kk]);
- }
-
- WebRtcSpl_AddAffineVectorToVector(bTmp16, b16, 3, 7, 2, kVectorSize);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ(((B[kk]*3+B[kk]*2+2)>>2)+((b16[kk]*3+7)>>2), bTmp16[kk]);
- }
-
- WebRtcSpl_CrossCorrelation(b32, b16, bTmp16, kVectorSize, 2, 2, 0);
- for (int kk = 0; kk < 2; ++kk) {
- EXPECT_EQ(614236, b32[kk]);
- }
-// EXPECT_EQ(, WebRtcSpl_DotProduct(b16, bTmp16, 4));
- EXPECT_EQ(306962, WebRtcSpl_DotProductWithScale(b16, b16, kVectorSize, 2));
-
- WebRtcSpl_ScaleVector(b16, bTmp16, 13, kVectorSize, 2);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
- }
- WebRtcSpl_ScaleVectorWithSat(b16, bTmp16, 13, kVectorSize, 2);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ((b16[kk]*13)>>2, bTmp16[kk]);
- }
- WebRtcSpl_ScaleAndAddVectors(a16, 13, 2, b16, 7, 2, bTmp16, kVectorSize);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ(((a16[kk]*13)>>2)+((b16[kk]*7)>>2), bTmp16[kk]);
- }
-
- WebRtcSpl_AddVectorsAndShift(bTmp16, a16, b16, kVectorSize, 2);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ(B[kk] >> 1, bTmp16[kk]);
- }
- WebRtcSpl_ReverseOrderMultArrayElements(bTmp16, a16, &b16[3], kVectorSize, 2);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ((a16[kk]*b16[3-kk])>>2, bTmp16[kk]);
- }
- WebRtcSpl_ElementwiseVectorMult(bTmp16, a16, b16, kVectorSize, 6);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ((a16[kk]*b16[kk])>>6, bTmp16[kk]);
- }
-
- WebRtcSpl_SqrtOfOneMinusXSquared(b16, kVectorSize, bTmp16);
- for (int kk = 0; kk < kVectorSize - 1; ++kk) {
- EXPECT_EQ(32767, bTmp16[kk]);
- }
- EXPECT_EQ(32749, bTmp16[kVectorSize - 1]);
-}
-
-TEST_F(SplTest, EstimatorsTest) {
- const int kVectorSize = 4;
- int B[] = {4, 12, 133, 1100};
- WebRtc_Word16 b16[kVectorSize];
- WebRtc_Word32 b32[kVectorSize];
- WebRtc_Word16 bTmp16[kVectorSize];
-
- for (int kk = 0; kk < kVectorSize; ++kk) {
- b16[kk] = B[kk];
- b32[kk] = B[kk];
- }
-
- EXPECT_EQ(0, WebRtcSpl_LevinsonDurbin(b32, b16, bTmp16, 2));
-}
-
-TEST_F(SplTest, FilterTest) {
- const int kVectorSize = 4;
- const int kFilterOrder = 3;
- WebRtc_Word16 A[] = {1, 2, 33, 100};
- WebRtc_Word16 A5[] = {1, 2, 33, 100, -5};
- WebRtc_Word16 B[] = {4, 12, 133, 110};
- WebRtc_Word16 data_in[kVectorSize];
- WebRtc_Word16 data_out[kVectorSize];
- WebRtc_Word16 bTmp16Low[kVectorSize];
- WebRtc_Word16 bState[kVectorSize];
- WebRtc_Word16 bStateLow[kVectorSize];
-
- WebRtcSpl_ZerosArrayW16(bState, kVectorSize);
- WebRtcSpl_ZerosArrayW16(bStateLow, kVectorSize);
-
- for (int kk = 0; kk < kVectorSize; ++kk) {
- data_in[kk] = A[kk];
- data_out[kk] = 0;
- }
-
- // MA filters.
- // Note that the input data has |kFilterOrder| states before the actual
- // data (one sample).
- WebRtcSpl_FilterMAFastQ12(&data_in[kFilterOrder], data_out, B,
- kFilterOrder + 1, 1);
- EXPECT_EQ(0, data_out[0]);
- // AR filters.
- // Note that the output data has |kFilterOrder| states before the actual
- // data (one sample).
- WebRtcSpl_FilterARFastQ12(data_in, &data_out[kFilterOrder], A,
- kFilterOrder + 1, 1);
- EXPECT_EQ(0, data_out[kFilterOrder]);
-
- EXPECT_EQ(kVectorSize, WebRtcSpl_FilterAR(A5,
- 5,
- data_in,
- kVectorSize,
- bState,
- kVectorSize,
- bStateLow,
- kVectorSize,
- data_out,
- bTmp16Low,
- kVectorSize));
-}
-
-TEST_F(SplTest, RandTest) {
- const int kVectorSize = 4;
- WebRtc_Word16 BU[] = {3653, 12446, 8525, 30691};
- WebRtc_Word16 b16[kVectorSize];
- WebRtc_UWord32 bSeed = 100000;
-
- EXPECT_EQ(464449057u, WebRtcSpl_IncreaseSeed(&bSeed));
- EXPECT_EQ(31565, WebRtcSpl_RandU(&bSeed));
- EXPECT_EQ(-9786, WebRtcSpl_RandN(&bSeed));
- EXPECT_EQ(kVectorSize, WebRtcSpl_RandUArray(b16, kVectorSize, &bSeed));
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ(BU[kk], b16[kk]);
- }
-}
-
-TEST_F(SplTest, SignalProcessingTest) {
- const int kVectorSize = 4;
- int A[] = {1, 2, 33, 100};
- const WebRtc_Word16 kHanning[4] = { 2399, 8192, 13985, 16384 };
- WebRtc_Word16 b16[kVectorSize];
-
- WebRtc_Word16 bTmp16[kVectorSize];
- WebRtc_Word32 bTmp32[kVectorSize];
-
- int bScale = 0;
-
- for (int kk = 0; kk < kVectorSize; ++kk) {
- b16[kk] = A[kk];
- }
-
- EXPECT_EQ(2, WebRtcSpl_AutoCorrelation(b16, kVectorSize, 1, bTmp32, &bScale));
- // TODO(bjornv): Activate the Reflection Coefficient tests when refactoring.
-// WebRtcSpl_ReflCoefToLpc(b16, kVectorSize, bTmp16);
-//// for (int kk = 0; kk < kVectorSize; ++kk) {
-//// EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
-//// }
-// WebRtcSpl_LpcToReflCoef(bTmp16, kVectorSize, b16);
-//// for (int kk = 0; kk < kVectorSize; ++kk) {
-//// EXPECT_EQ(a16[kk], b16[kk]);
-//// }
-// WebRtcSpl_AutoCorrToReflCoef(b32, kVectorSize, bTmp16);
-//// for (int kk = 0; kk < kVectorSize; ++kk) {
-//// EXPECT_EQ(aTmp16[kk], bTmp16[kk]);
-//// }
-
- WebRtcSpl_GetHanningWindow(bTmp16, kVectorSize);
- for (int kk = 0; kk < kVectorSize; ++kk) {
- EXPECT_EQ(kHanning[kk], bTmp16[kk]);
- }
-
- for (int kk = 0; kk < kVectorSize; ++kk) {
- b16[kk] = A[kk];
- }
- EXPECT_EQ(11094 , WebRtcSpl_Energy(b16, kVectorSize, &bScale));
- EXPECT_EQ(0, bScale);
-}
-
-TEST_F(SplTest, FFTTest) {
- WebRtc_Word16 B[] = {1, 2, 33, 100,
- 2, 3, 34, 101,
- 3, 4, 35, 102,
- 4, 5, 36, 103};
-
- EXPECT_EQ(0, WebRtcSpl_ComplexFFT(B, 3, 1));
-// for (int kk = 0; kk < 16; ++kk) {
-// EXPECT_EQ(A[kk], B[kk]);
-// }
- EXPECT_EQ(0, WebRtcSpl_ComplexIFFT(B, 3, 1));
-// for (int kk = 0; kk < 16; ++kk) {
-// EXPECT_EQ(A[kk], B[kk]);
-// }
- WebRtcSpl_ComplexBitReverse(B, 3);
- for (int kk = 0; kk < 16; ++kk) {
- //EXPECT_EQ(A[kk], B[kk]);
- }
-}
diff --git a/trunk/src/common_audio/signal_processing/spl_sqrt.c b/trunk/src/common_audio/signal_processing/spl_sqrt.c
deleted file mode 100644
index cfe2cd3..0000000
--- a/trunk/src/common_audio/signal_processing/spl_sqrt.c
+++ /dev/null
@@ -1,184 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_Sqrt().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-WebRtc_Word32 WebRtcSpl_SqrtLocal(WebRtc_Word32 in);
-
-WebRtc_Word32 WebRtcSpl_SqrtLocal(WebRtc_Word32 in)
-{
-
- WebRtc_Word16 x_half, t16;
- WebRtc_Word32 A, B, x2;
-
- /* The following block performs:
- y=in/2
- x=y-2^30
- x_half=x/2^31
- t = 1 + (x_half) - 0.5*((x_half)^2) + 0.5*((x_half)^3) - 0.625*((x_half)^4)
- + 0.875*((x_half)^5)
- */
-
- B = in;
-
- B = WEBRTC_SPL_RSHIFT_W32(B, 1); // B = in/2
- B = B - ((WebRtc_Word32)0x40000000); // B = in/2 - 1/2
- x_half = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(B, 16);// x_half = x/2 = (in-1)/2
- B = B + ((WebRtc_Word32)0x40000000); // B = 1 + x/2
- B = B + ((WebRtc_Word32)0x40000000); // Add 0.5 twice (since 1.0 does not exist in Q31)
-
- x2 = ((WebRtc_Word32)x_half) * ((WebRtc_Word32)x_half) * 2; // A = (x/2)^2
- A = -x2; // A = -(x/2)^2
- B = B + (A >> 1); // B = 1 + x/2 - 0.5*(x/2)^2
-
- A = WEBRTC_SPL_RSHIFT_W32(A, 16);
- A = A * A * 2; // A = (x/2)^4
- t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16);
- B = B + WEBRTC_SPL_MUL_16_16(-20480, t16) * 2; // B = B - 0.625*A
- // After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4
-
- t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16);
- A = WEBRTC_SPL_MUL_16_16(x_half, t16) * 2; // A = (x/2)^5
- t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16);
- B = B + WEBRTC_SPL_MUL_16_16(28672, t16) * 2; // B = B + 0.875*A
- // After this, B = 1 + x/2 - 0.5*(x/2)^2 - 0.625*(x/2)^4 + 0.875*(x/2)^5
-
- t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(x2, 16);
- A = WEBRTC_SPL_MUL_16_16(x_half, t16) * 2; // A = x/2^3
-
- B = B + (A >> 1); // B = B + 0.5*A
- // After this, B = 1 + x/2 - 0.5*(x/2)^2 + 0.5*(x/2)^3 - 0.625*(x/2)^4 + 0.875*(x/2)^5
-
- B = B + ((WebRtc_Word32)32768); // Round off bit
-
- return B;
-}
-
-WebRtc_Word32 WebRtcSpl_Sqrt(WebRtc_Word32 value)
-{
- /*
- Algorithm:
-
- Six term Taylor Series is used here to compute the square root of a number
- y^0.5 = (1+x)^0.5 where x = y-1
- = 1+(x/2)-0.5*((x/2)^2+0.5*((x/2)^3-0.625*((x/2)^4+0.875*((x/2)^5)
- 0.5 <= x < 1
-
- Example of how the algorithm works, with ut=sqrt(in), and
- with in=73632 and ut=271 (even shift value case):
-
- in=73632
- y= in/131072
- x=y-1
- t = 1 + (x/2) - 0.5*((x/2)^2) + 0.5*((x/2)^3) - 0.625*((x/2)^4) + 0.875*((x/2)^5)
- ut=t*(1/sqrt(2))*512
-
- or:
-
- in=73632
- in2=73632*2^14
- y= in2/2^31
- x=y-1
- t = 1 + (x/2) - 0.5*((x/2)^2) + 0.5*((x/2)^3) - 0.625*((x/2)^4) + 0.875*((x/2)^5)
- ut=t*(1/sqrt(2))
- ut2=ut*2^9
-
- which gives:
-
- in = 73632
- in2 = 1206386688
- y = 0.56176757812500
- x = -0.43823242187500
- t = 0.74973506527313
- ut = 0.53014274874797
- ut2 = 2.714330873589594e+002
-
- or:
-
- in=73632
- in2=73632*2^14
- y=in2/2
- x=y-2^30
- x_half=x/2^31
- t = 1 + (x_half) - 0.5*((x_half)^2) + 0.5*((x_half)^3) - 0.625*((x_half)^4)
- + 0.875*((x_half)^5)
- ut=t*(1/sqrt(2))
- ut2=ut*2^9
-
- which gives:
-
- in = 73632
- in2 = 1206386688
- y = 603193344
- x = -470548480
- x_half = -0.21911621093750
- t = 0.74973506527313
- ut = 0.53014274874797
- ut2 = 2.714330873589594e+002
-
- */
-
- WebRtc_Word16 x_norm, nshift, t16, sh;
- WebRtc_Word32 A;
-
- WebRtc_Word16 k_sqrt_2 = 23170; // 1/sqrt2 (==5a82)
-
- A = value;
-
- if (A == 0)
- return (WebRtc_Word32)0; // sqrt(0) = 0
-
- sh = WebRtcSpl_NormW32(A); // # shifts to normalize A
- A = WEBRTC_SPL_LSHIFT_W32(A, sh); // Normalize A
- if (A < (WEBRTC_SPL_WORD32_MAX - 32767))
- {
- A = A + ((WebRtc_Word32)32768); // Round off bit
- } else
- {
- A = WEBRTC_SPL_WORD32_MAX;
- }
-
- x_norm = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16); // x_norm = AH
-
- nshift = WEBRTC_SPL_RSHIFT_W16(sh, 1); // nshift = sh>>1
- nshift = -nshift; // Negate the power for later de-normalization
-
- A = (WebRtc_Word32)WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)x_norm, 16);
- A = WEBRTC_SPL_ABS_W32(A); // A = abs(x_norm<<16)
- A = WebRtcSpl_SqrtLocal(A); // A = sqrt(A)
-
- if ((-2 * nshift) == sh)
- { // Even shift value case
-
- t16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(A, 16); // t16 = AH
-
- A = WEBRTC_SPL_MUL_16_16(k_sqrt_2, t16) * 2; // A = 1/sqrt(2)*t16
- A = A + ((WebRtc_Word32)32768); // Round off
- A = A & ((WebRtc_Word32)0x7fff0000); // Round off
-
- A = WEBRTC_SPL_RSHIFT_W32(A, 15); // A = A>>16
-
- } else
- {
- A = WEBRTC_SPL_RSHIFT_W32(A, 16); // A = A>>16
- }
-
- A = A & ((WebRtc_Word32)0x0000ffff);
- A = (WebRtc_Word32)WEBRTC_SPL_SHIFT_W32(A, nshift); // De-normalize the result
-
- return A;
-}
diff --git a/trunk/src/common_audio/signal_processing/spl_sqrt_floor.c b/trunk/src/common_audio/signal_processing/spl_sqrt_floor.c
deleted file mode 100644
index f0e8ae2..0000000
--- a/trunk/src/common_audio/signal_processing/spl_sqrt_floor.c
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- * Written by Wilco Dijkstra, 1996. Refer to file LICENSE under
- * trunk/third_party_mods/sqrt_floor.
- *
- * Minor modifications in code style for WebRTC, 2012.
- */
-
-#include "signal_processing_library.h"
-
-/*
- * Algorithm:
- * Successive approximation of the equation (root + delta) ^ 2 = N
- * until delta < 1. If delta < 1 we have the integer part of SQRT (N).
- * Use delta = 2^i for i = 15 .. 0.
- *
- * Output precision is 16 bits. Note for large input values (close to
- * 0x7FFFFFFF), bit 15 (the highest bit of the low 16-bit half word)
- * contains the MSB information (a non-sign value). Do with caution
- * if you need to cast the output to int16_t type.
- *
- * If the input value is negative, it returns 0.
- */
-
-#define WEBRTC_SPL_SQRT_ITER(N) \
- try1 = root + (1 << (N)); \
- if (value >= try1 << (N)) \
- { \
- value -= try1 << (N); \
- root |= 2 << (N); \
- }
-
-int32_t WebRtcSpl_SqrtFloor(int32_t value)
-{
- int32_t root = 0, try1;
-
- WEBRTC_SPL_SQRT_ITER (15);
- WEBRTC_SPL_SQRT_ITER (14);
- WEBRTC_SPL_SQRT_ITER (13);
- WEBRTC_SPL_SQRT_ITER (12);
- WEBRTC_SPL_SQRT_ITER (11);
- WEBRTC_SPL_SQRT_ITER (10);
- WEBRTC_SPL_SQRT_ITER ( 9);
- WEBRTC_SPL_SQRT_ITER ( 8);
- WEBRTC_SPL_SQRT_ITER ( 7);
- WEBRTC_SPL_SQRT_ITER ( 6);
- WEBRTC_SPL_SQRT_ITER ( 5);
- WEBRTC_SPL_SQRT_ITER ( 4);
- WEBRTC_SPL_SQRT_ITER ( 3);
- WEBRTC_SPL_SQRT_ITER ( 2);
- WEBRTC_SPL_SQRT_ITER ( 1);
- WEBRTC_SPL_SQRT_ITER ( 0);
-
- return root >> 1;
-}
diff --git a/trunk/src/common_audio/signal_processing/spl_sqrt_floor.s b/trunk/src/common_audio/signal_processing/spl_sqrt_floor.s
deleted file mode 100644
index cfd9ed0..0000000
--- a/trunk/src/common_audio/signal_processing/spl_sqrt_floor.s
+++ /dev/null
@@ -1,88 +0,0 @@
-@ Written by Wilco Dijkstra, 1996. Refer to file LICENSE under
-@ trunk/third_party_mods/sqrt_floor.
-@
-@ Minor modifications in code style for WebRTC, 2012.
-@ Output is bit-exact with the reference C code in spl_sqrt_floor.c.
-
-@ Input : r0 32 bit unsigned integer
-@ Output: r0 = INT (SQRT (r0)), precision is 16 bits
-@ Registers touched: r1, r2
-
-.global WebRtcSpl_SqrtFloor
-
-.align 2
-WebRtcSpl_SqrtFloor:
-.fnstart
- mov r1, #3 << 30
- mov r2, #1 << 30
-
- @ unroll for i = 0 .. 15
-
- cmp r0, r2, ror #2 * 0
- subhs r0, r0, r2, ror #2 * 0
- adc r2, r1, r2, lsl #1
-
- cmp r0, r2, ror #2 * 1
- subhs r0, r0, r2, ror #2 * 1
- adc r2, r1, r2, lsl #1
-
- cmp r0, r2, ror #2 * 2
- subhs r0, r0, r2, ror #2 * 2
- adc r2, r1, r2, lsl #1
-
- cmp r0, r2, ror #2 * 3
- subhs r0, r0, r2, ror #2 * 3
- adc r2, r1, r2, lsl #1
-
- cmp r0, r2, ror #2 * 4
- subhs r0, r0, r2, ror #2 * 4
- adc r2, r1, r2, lsl #1
-
- cmp r0, r2, ror #2 * 5
- subhs r0, r0, r2, ror #2 * 5
- adc r2, r1, r2, lsl #1
-
- cmp r0, r2, ror #2 * 6
- subhs r0, r0, r2, ror #2 * 6
- adc r2, r1, r2, lsl #1
-
- cmp r0, r2, ror #2 * 7
- subhs r0, r0, r2, ror #2 * 7
- adc r2, r1, r2, lsl #1
-
- cmp r0, r2, ror #2 * 8
- subhs r0, r0, r2, ror #2 * 8
- adc r2, r1, r2, lsl #1
-
- cmp r0, r2, ror #2 * 9
- subhs r0, r0, r2, ror #2 * 9
- adc r2, r1, r2, lsl #1
-
- cmp r0, r2, ror #2 * 10
- subhs r0, r0, r2, ror #2 * 10
- adc r2, r1, r2, lsl #1
-
- cmp r0, r2, ror #2 * 11
- subhs r0, r0, r2, ror #2 * 11
- adc r2, r1, r2, lsl #1
-
- cmp r0, r2, ror #2 * 12
- subhs r0, r0, r2, ror #2 * 12
- adc r2, r1, r2, lsl #1
-
- cmp r0, r2, ror #2 * 13
- subhs r0, r0, r2, ror #2 * 13
- adc r2, r1, r2, lsl #1
-
- cmp r0, r2, ror #2 * 14
- subhs r0, r0, r2, ror #2 * 14
- adc r2, r1, r2, lsl #1
-
- cmp r0, r2, ror #2 * 15
- subhs r0, r0, r2, ror #2 * 15
- adc r2, r1, r2, lsl #1
-
- bic r0, r2, #3 << 30 @ for rounding add: cmp r0, r2 adc r2, #1
- bx lr
-
-.fnend
diff --git a/trunk/src/common_audio/signal_processing/spl_version.c b/trunk/src/common_audio/signal_processing/spl_version.c
deleted file mode 100644
index 936925e..0000000
--- a/trunk/src/common_audio/signal_processing/spl_version.c
+++ /dev/null
@@ -1,25 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_get_version().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include <string.h>
-#include "signal_processing_library.h"
-
-WebRtc_Word16 WebRtcSpl_get_version(char* version, WebRtc_Word16 length_in_bytes)
-{
- strncpy(version, "1.2.0", length_in_bytes);
- return 0;
-}
diff --git a/trunk/src/common_audio/signal_processing/splitting_filter.c b/trunk/src/common_audio/signal_processing/splitting_filter.c
deleted file mode 100644
index f1acf67..0000000
--- a/trunk/src/common_audio/signal_processing/splitting_filter.c
+++ /dev/null
@@ -1,198 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the splitting filter functions.
- *
- */
-
-#include "signal_processing_library.h"
-
-// Number of samples in a low/high-band frame.
-enum
-{
- kBandFrameLength = 160
-};
-
-// QMF filter coefficients in Q16.
-static const WebRtc_UWord16 WebRtcSpl_kAllPassFilter1[3] = {6418, 36982, 57261};
-static const WebRtc_UWord16 WebRtcSpl_kAllPassFilter2[3] = {21333, 49062, 63010};
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcSpl_AllPassQMF(...)
-//
-// Allpass filter used by the analysis and synthesis parts of the QMF filter.
-//
-// Input:
-// - in_data : Input data sequence (Q10)
-// - data_length : Length of data sequence (>2)
-// - filter_coefficients : Filter coefficients (length 3, Q16)
-//
-// Input & Output:
-// - filter_state : Filter state (length 6, Q10).
-//
-// Output:
-// - out_data : Output data sequence (Q10), length equal to
-// |data_length|
-//
-
-void WebRtcSpl_AllPassQMF(WebRtc_Word32* in_data, const WebRtc_Word16 data_length,
- WebRtc_Word32* out_data, const WebRtc_UWord16* filter_coefficients,
- WebRtc_Word32* filter_state)
-{
- // The procedure is to filter the input with three first order all pass filters
- // (cascade operations).
- //
- // a_3 + q^-1 a_2 + q^-1 a_1 + q^-1
- // y[n] = ----------- ----------- ----------- x[n]
- // 1 + a_3q^-1 1 + a_2q^-1 1 + a_1q^-1
- //
- // The input vector |filter_coefficients| includes these three filter coefficients.
- // The filter state contains the in_data state, in_data[-1], followed by
- // the out_data state, out_data[-1]. This is repeated for each cascade.
- // The first cascade filter will filter the |in_data| and store the output in
- // |out_data|. The second will the take the |out_data| as input and make an
- // intermediate storage in |in_data|, to save memory. The third, and final, cascade
- // filter operation takes the |in_data| (which is the output from the previous cascade
- // filter) and store the output in |out_data|.
- // Note that the input vector values are changed during the process.
- WebRtc_Word16 k;
- WebRtc_Word32 diff;
- // First all-pass cascade; filter from in_data to out_data.
-
- // Let y_i[n] indicate the output of cascade filter i (with filter coefficient a_i) at
- // vector position n. Then the final output will be y[n] = y_3[n]
-
- // First loop, use the states stored in memory.
- // "diff" should be safe from wrap around since max values are 2^25
- diff = WEBRTC_SPL_SUB_SAT_W32(in_data[0], filter_state[1]); // = (x[0] - y_1[-1])
- // y_1[0] = x[-1] + a_1 * (x[0] - y_1[-1])
- out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, filter_state[0]);
-
- // For the remaining loops, use previous values.
- for (k = 1; k < data_length; k++)
- {
- diff = WEBRTC_SPL_SUB_SAT_W32(in_data[k], out_data[k - 1]); // = (x[n] - y_1[n-1])
- // y_1[n] = x[n-1] + a_1 * (x[n] - y_1[n-1])
- out_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[0], diff, in_data[k - 1]);
- }
-
- // Update states.
- filter_state[0] = in_data[data_length - 1]; // x[N-1], becomes x[-1] next time
- filter_state[1] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time
-
- // Second all-pass cascade; filter from out_data to in_data.
- diff = WEBRTC_SPL_SUB_SAT_W32(out_data[0], filter_state[3]); // = (y_1[0] - y_2[-1])
- // y_2[0] = y_1[-1] + a_2 * (y_1[0] - y_2[-1])
- in_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, filter_state[2]);
- for (k = 1; k < data_length; k++)
- {
- diff = WEBRTC_SPL_SUB_SAT_W32(out_data[k], in_data[k - 1]); // =(y_1[n] - y_2[n-1])
- // y_2[0] = y_1[-1] + a_2 * (y_1[0] - y_2[-1])
- in_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[1], diff, out_data[k-1]);
- }
-
- filter_state[2] = out_data[data_length - 1]; // y_1[N-1], becomes y_1[-1] next time
- filter_state[3] = in_data[data_length - 1]; // y_2[N-1], becomes y_2[-1] next time
-
- // Third all-pass cascade; filter from in_data to out_data.
- diff = WEBRTC_SPL_SUB_SAT_W32(in_data[0], filter_state[5]); // = (y_2[0] - y[-1])
- // y[0] = y_2[-1] + a_3 * (y_2[0] - y[-1])
- out_data[0] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[2], diff, filter_state[4]);
- for (k = 1; k < data_length; k++)
- {
- diff = WEBRTC_SPL_SUB_SAT_W32(in_data[k], out_data[k - 1]); // = (y_2[n] - y[n-1])
- // y[n] = y_2[n-1] + a_3 * (y_2[n] - y[n-1])
- out_data[k] = WEBRTC_SPL_SCALEDIFF32(filter_coefficients[2], diff, in_data[k-1]);
- }
- filter_state[4] = in_data[data_length - 1]; // y_2[N-1], becomes y_2[-1] next time
- filter_state[5] = out_data[data_length - 1]; // y[N-1], becomes y[-1] next time
-}
-
-void WebRtcSpl_AnalysisQMF(const WebRtc_Word16* in_data, WebRtc_Word16* low_band,
- WebRtc_Word16* high_band, WebRtc_Word32* filter_state1,
- WebRtc_Word32* filter_state2)
-{
- WebRtc_Word16 i;
- WebRtc_Word16 k;
- WebRtc_Word32 tmp;
- WebRtc_Word32 half_in1[kBandFrameLength];
- WebRtc_Word32 half_in2[kBandFrameLength];
- WebRtc_Word32 filter1[kBandFrameLength];
- WebRtc_Word32 filter2[kBandFrameLength];
-
- // Split even and odd samples. Also shift them to Q10.
- for (i = 0, k = 0; i < kBandFrameLength; i++, k += 2)
- {
- half_in2[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)in_data[k], 10);
- half_in1[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)in_data[k + 1], 10);
- }
-
- // All pass filter even and odd samples, independently.
- WebRtcSpl_AllPassQMF(half_in1, kBandFrameLength, filter1, WebRtcSpl_kAllPassFilter1,
- filter_state1);
- WebRtcSpl_AllPassQMF(half_in2, kBandFrameLength, filter2, WebRtcSpl_kAllPassFilter2,
- filter_state2);
-
- // Take the sum and difference of filtered version of odd and even
- // branches to get upper & lower band.
- for (i = 0; i < kBandFrameLength; i++)
- {
- tmp = filter1[i] + filter2[i] + 1024;
- tmp = WEBRTC_SPL_RSHIFT_W32(tmp, 11);
- low_band[i] = WebRtcSpl_SatW32ToW16(tmp);
-
- tmp = filter1[i] - filter2[i] + 1024;
- tmp = WEBRTC_SPL_RSHIFT_W32(tmp, 11);
- high_band[i] = WebRtcSpl_SatW32ToW16(tmp);
- }
-}
-
-void WebRtcSpl_SynthesisQMF(const WebRtc_Word16* low_band, const WebRtc_Word16* high_band,
- WebRtc_Word16* out_data, WebRtc_Word32* filter_state1,
- WebRtc_Word32* filter_state2)
-{
- WebRtc_Word32 tmp;
- WebRtc_Word32 half_in1[kBandFrameLength];
- WebRtc_Word32 half_in2[kBandFrameLength];
- WebRtc_Word32 filter1[kBandFrameLength];
- WebRtc_Word32 filter2[kBandFrameLength];
- WebRtc_Word16 i;
- WebRtc_Word16 k;
-
- // Obtain the sum and difference channels out of upper and lower-band channels.
- // Also shift to Q10 domain.
- for (i = 0; i < kBandFrameLength; i++)
- {
- tmp = (WebRtc_Word32)low_band[i] + (WebRtc_Word32)high_band[i];
- half_in1[i] = WEBRTC_SPL_LSHIFT_W32(tmp, 10);
- tmp = (WebRtc_Word32)low_band[i] - (WebRtc_Word32)high_band[i];
- half_in2[i] = WEBRTC_SPL_LSHIFT_W32(tmp, 10);
- }
-
- // all-pass filter the sum and difference channels
- WebRtcSpl_AllPassQMF(half_in1, kBandFrameLength, filter1, WebRtcSpl_kAllPassFilter2,
- filter_state1);
- WebRtcSpl_AllPassQMF(half_in2, kBandFrameLength, filter2, WebRtcSpl_kAllPassFilter1,
- filter_state2);
-
- // The filtered signals are even and odd samples of the output. Combine
- // them. The signals are Q10 should shift them back to Q0 and take care of
- // saturation.
- for (i = 0, k = 0; i < kBandFrameLength; i++)
- {
- tmp = WEBRTC_SPL_RSHIFT_W32(filter2[i] + 512, 10);
- out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);
-
- tmp = WEBRTC_SPL_RSHIFT_W32(filter1[i] + 512, 10);
- out_data[k++] = WebRtcSpl_SatW32ToW16(tmp);
- }
-
-}
diff --git a/trunk/src/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c b/trunk/src/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
deleted file mode 100644
index 9fb2c73..0000000
--- a/trunk/src/common_audio/signal_processing/sqrt_of_one_minus_x_squared.c
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the function WebRtcSpl_SqrtOfOneMinusXSquared().
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_SqrtOfOneMinusXSquared(WebRtc_Word16 *xQ15, int vector_length,
- WebRtc_Word16 *yQ15)
-{
- WebRtc_Word32 sq;
- int m;
- WebRtc_Word16 tmp;
-
- for (m = 0; m < vector_length; m++)
- {
- tmp = xQ15[m];
- sq = WEBRTC_SPL_MUL_16_16(tmp, tmp); // x^2 in Q30
- sq = 1073741823 - sq; // 1-x^2, where 1 ~= 0.99999999906 is 1073741823 in Q30
- sq = WebRtcSpl_Sqrt(sq); // sqrt(1-x^2) in Q15
- yQ15[m] = (WebRtc_Word16)sq;
- }
-}
diff --git a/trunk/src/common_audio/signal_processing/vector_scaling_operations.c b/trunk/src/common_audio/signal_processing/vector_scaling_operations.c
deleted file mode 100644
index 20d239c..0000000
--- a/trunk/src/common_audio/signal_processing/vector_scaling_operations.c
+++ /dev/null
@@ -1,151 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains implementations of the functions
- * WebRtcSpl_VectorBitShiftW16()
- * WebRtcSpl_VectorBitShiftW32()
- * WebRtcSpl_VectorBitShiftW32ToW16()
- * WebRtcSpl_ScaleVector()
- * WebRtcSpl_ScaleVectorWithSat()
- * WebRtcSpl_ScaleAndAddVectors()
- *
- * The description header can be found in signal_processing_library.h
- *
- */
-
-#include "signal_processing_library.h"
-
-void WebRtcSpl_VectorBitShiftW16(WebRtc_Word16 *res,
- WebRtc_Word16 length,
- G_CONST WebRtc_Word16 *in,
- WebRtc_Word16 right_shifts)
-{
- int i;
-
- if (right_shifts > 0)
- {
- for (i = length; i > 0; i--)
- {
- (*res++) = ((*in++) >> right_shifts);
- }
- } else
- {
- for (i = length; i > 0; i--)
- {
- (*res++) = ((*in++) << (-right_shifts));
- }
- }
-}
-
-void WebRtcSpl_VectorBitShiftW32(WebRtc_Word32 *out_vector,
- WebRtc_Word16 vector_length,
- G_CONST WebRtc_Word32 *in_vector,
- WebRtc_Word16 right_shifts)
-{
- int i;
-
- if (right_shifts > 0)
- {
- for (i = vector_length; i > 0; i--)
- {
- (*out_vector++) = ((*in_vector++) >> right_shifts);
- }
- } else
- {
- for (i = vector_length; i > 0; i--)
- {
- (*out_vector++) = ((*in_vector++) << (-right_shifts));
- }
- }
-}
-
-void WebRtcSpl_VectorBitShiftW32ToW16(WebRtc_Word16 *res,
- WebRtc_Word16 length,
- G_CONST WebRtc_Word32 *in,
- WebRtc_Word16 right_shifts)
-{
- int i;
-
- if (right_shifts >= 0)
- {
- for (i = length; i > 0; i--)
- {
- (*res++) = (WebRtc_Word16)((*in++) >> right_shifts);
- }
- } else
- {
- WebRtc_Word16 left_shifts = -right_shifts;
- for (i = length; i > 0; i--)
- {
- (*res++) = (WebRtc_Word16)((*in++) << left_shifts);
- }
- }
-}
-
-void WebRtcSpl_ScaleVector(G_CONST WebRtc_Word16 *in_vector, WebRtc_Word16 *out_vector,
- WebRtc_Word16 gain, WebRtc_Word16 in_vector_length,
- WebRtc_Word16 right_shifts)
-{
- // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
- int i;
- G_CONST WebRtc_Word16 *inptr;
- WebRtc_Word16 *outptr;
-
- inptr = in_vector;
- outptr = out_vector;
-
- for (i = 0; i < in_vector_length; i++)
- {
- (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts);
- }
-}
-
-void WebRtcSpl_ScaleVectorWithSat(G_CONST WebRtc_Word16 *in_vector, WebRtc_Word16 *out_vector,
- WebRtc_Word16 gain, WebRtc_Word16 in_vector_length,
- WebRtc_Word16 right_shifts)
-{
- // Performs vector operation: out_vector = (gain*in_vector)>>right_shifts
- int i;
- WebRtc_Word32 tmpW32;
- G_CONST WebRtc_Word16 *inptr;
- WebRtc_Word16 *outptr;
-
- inptr = in_vector;
- outptr = out_vector;
-
- for (i = 0; i < in_vector_length; i++)
- {
- tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(*inptr++, gain, right_shifts);
- (*outptr++) = WebRtcSpl_SatW32ToW16(tmpW32);
- }
-}
-
-void WebRtcSpl_ScaleAndAddVectors(G_CONST WebRtc_Word16 *in1, WebRtc_Word16 gain1, int shift1,
- G_CONST WebRtc_Word16 *in2, WebRtc_Word16 gain2, int shift2,
- WebRtc_Word16 *out, int vector_length)
-{
- // Performs vector operation: out = (gain1*in1)>>shift1 + (gain2*in2)>>shift2
- int i;
- G_CONST WebRtc_Word16 *in1ptr;
- G_CONST WebRtc_Word16 *in2ptr;
- WebRtc_Word16 *outptr;
-
- in1ptr = in1;
- in2ptr = in2;
- outptr = out;
-
- for (i = 0; i < vector_length; i++)
- {
- (*outptr++) = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gain1, *in1ptr++, shift1)
- + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gain2, *in2ptr++, shift2);
- }
-}
diff --git a/trunk/src/common_audio/signal_processing/webrtc_fft_t_1024_8.c b/trunk/src/common_audio/signal_processing/webrtc_fft_t_1024_8.c
deleted file mode 100644
index b587380..0000000
--- a/trunk/src/common_audio/signal_processing/webrtc_fft_t_1024_8.c
+++ /dev/null
@@ -1,704 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the Q14 radix-8 tables used in ARM9e optimizations.
- *
- */
-
-extern const int s_Q14S_8;
-const int s_Q14S_8 = 1024;
-extern const unsigned short t_Q14S_8[2032];
-const unsigned short t_Q14S_8[2032] = {
- 0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
- 0x22a3,0x187e ,0x3249,0x0c7c ,0x11a8,0x238e ,
- 0x0000,0x2d41 ,0x22a3,0x187e ,0xdd5d,0x3b21 ,
- 0xdd5d,0x3b21 ,0x11a8,0x238e ,0xb4be,0x3ec5 ,
- 0xc000,0x4000 ,0x0000,0x2d41 ,0xa57e,0x2d41 ,
- 0xac61,0x3b21 ,0xee58,0x3537 ,0xb4be,0x0c7c ,
- 0xa57e,0x2d41 ,0xdd5d,0x3b21 ,0xdd5d,0xe782 ,
- 0xac61,0x187e ,0xcdb7,0x3ec5 ,0x11a8,0xcac9 ,
- 0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
- 0x396b,0x0646 ,0x3cc8,0x0324 ,0x35eb,0x0964 ,
- 0x3249,0x0c7c ,0x396b,0x0646 ,0x2aaa,0x1294 ,
- 0x2aaa,0x1294 ,0x35eb,0x0964 ,0x1e7e,0x1b5d ,
- 0x22a3,0x187e ,0x3249,0x0c7c ,0x11a8,0x238e ,
- 0x1a46,0x1e2b ,0x2e88,0x0f8d ,0x0471,0x2afb ,
- 0x11a8,0x238e ,0x2aaa,0x1294 ,0xf721,0x3179 ,
- 0x08df,0x289a ,0x26b3,0x1590 ,0xea02,0x36e5 ,
- 0x0000,0x2d41 ,0x22a3,0x187e ,0xdd5d,0x3b21 ,
- 0xf721,0x3179 ,0x1e7e,0x1b5d ,0xd178,0x3e15 ,
- 0xee58,0x3537 ,0x1a46,0x1e2b ,0xc695,0x3fb1 ,
- 0xe5ba,0x3871 ,0x15fe,0x20e7 ,0xbcf0,0x3fec ,
- 0xdd5d,0x3b21 ,0x11a8,0x238e ,0xb4be,0x3ec5 ,
- 0xd556,0x3d3f ,0x0d48,0x2620 ,0xae2e,0x3c42 ,
- 0xcdb7,0x3ec5 ,0x08df,0x289a ,0xa963,0x3871 ,
- 0xc695,0x3fb1 ,0x0471,0x2afb ,0xa678,0x3368 ,
- 0xc000,0x4000 ,0x0000,0x2d41 ,0xa57e,0x2d41 ,
- 0xba09,0x3fb1 ,0xfb8f,0x2f6c ,0xa678,0x2620 ,
- 0xb4be,0x3ec5 ,0xf721,0x3179 ,0xa963,0x1e2b ,
- 0xb02d,0x3d3f ,0xf2b8,0x3368 ,0xae2e,0x1590 ,
- 0xac61,0x3b21 ,0xee58,0x3537 ,0xb4be,0x0c7c ,
- 0xa963,0x3871 ,0xea02,0x36e5 ,0xbcf0,0x0324 ,
- 0xa73b,0x3537 ,0xe5ba,0x3871 ,0xc695,0xf9ba ,
- 0xa5ed,0x3179 ,0xe182,0x39db ,0xd178,0xf073 ,
- 0xa57e,0x2d41 ,0xdd5d,0x3b21 ,0xdd5d,0xe782 ,
- 0xa5ed,0x289a ,0xd94d,0x3c42 ,0xea02,0xdf19 ,
- 0xa73b,0x238e ,0xd556,0x3d3f ,0xf721,0xd766 ,
- 0xa963,0x1e2b ,0xd178,0x3e15 ,0x0471,0xd094 ,
- 0xac61,0x187e ,0xcdb7,0x3ec5 ,0x11a8,0xcac9 ,
- 0xb02d,0x1294 ,0xca15,0x3f4f ,0x1e7e,0xc625 ,
- 0xb4be,0x0c7c ,0xc695,0x3fb1 ,0x2aaa,0xc2c1 ,
- 0xba09,0x0646 ,0xc338,0x3fec ,0x35eb,0xc0b1 ,
- 0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
- 0x3e69,0x0192 ,0x3f36,0x00c9 ,0x3d9a,0x025b ,
- 0x3cc8,0x0324 ,0x3e69,0x0192 ,0x3b1e,0x04b5 ,
- 0x3b1e,0x04b5 ,0x3d9a,0x025b ,0x388e,0x070e ,
- 0x396b,0x0646 ,0x3cc8,0x0324 ,0x35eb,0x0964 ,
- 0x37af,0x07d6 ,0x3bf4,0x03ed ,0x3334,0x0bb7 ,
- 0x35eb,0x0964 ,0x3b1e,0x04b5 ,0x306c,0x0e06 ,
- 0x341e,0x0af1 ,0x3a46,0x057e ,0x2d93,0x1050 ,
- 0x3249,0x0c7c ,0x396b,0x0646 ,0x2aaa,0x1294 ,
- 0x306c,0x0e06 ,0x388e,0x070e ,0x27b3,0x14d2 ,
- 0x2e88,0x0f8d ,0x37af,0x07d6 ,0x24ae,0x1709 ,
- 0x2c9d,0x1112 ,0x36ce,0x089d ,0x219c,0x1937 ,
- 0x2aaa,0x1294 ,0x35eb,0x0964 ,0x1e7e,0x1b5d ,
- 0x28b2,0x1413 ,0x3505,0x0a2b ,0x1b56,0x1d79 ,
- 0x26b3,0x1590 ,0x341e,0x0af1 ,0x1824,0x1f8c ,
- 0x24ae,0x1709 ,0x3334,0x0bb7 ,0x14ea,0x2193 ,
- 0x22a3,0x187e ,0x3249,0x0c7c ,0x11a8,0x238e ,
- 0x2093,0x19ef ,0x315b,0x0d41 ,0x0e61,0x257e ,
- 0x1e7e,0x1b5d ,0x306c,0x0e06 ,0x0b14,0x2760 ,
- 0x1c64,0x1cc6 ,0x2f7b,0x0eca ,0x07c4,0x2935 ,
- 0x1a46,0x1e2b ,0x2e88,0x0f8d ,0x0471,0x2afb ,
- 0x1824,0x1f8c ,0x2d93,0x1050 ,0x011c,0x2cb2 ,
- 0x15fe,0x20e7 ,0x2c9d,0x1112 ,0xfdc7,0x2e5a ,
- 0x13d5,0x223d ,0x2ba4,0x11d3 ,0xfa73,0x2ff2 ,
- 0x11a8,0x238e ,0x2aaa,0x1294 ,0xf721,0x3179 ,
- 0x0f79,0x24da ,0x29af,0x1354 ,0xf3d2,0x32ef ,
- 0x0d48,0x2620 ,0x28b2,0x1413 ,0xf087,0x3453 ,
- 0x0b14,0x2760 ,0x27b3,0x14d2 ,0xed41,0x35a5 ,
- 0x08df,0x289a ,0x26b3,0x1590 ,0xea02,0x36e5 ,
- 0x06a9,0x29ce ,0x25b1,0x164c ,0xe6cb,0x3812 ,
- 0x0471,0x2afb ,0x24ae,0x1709 ,0xe39c,0x392b ,
- 0x0239,0x2c21 ,0x23a9,0x17c4 ,0xe077,0x3a30 ,
- 0x0000,0x2d41 ,0x22a3,0x187e ,0xdd5d,0x3b21 ,
- 0xfdc7,0x2e5a ,0x219c,0x1937 ,0xda4f,0x3bfd ,
- 0xfb8f,0x2f6c ,0x2093,0x19ef ,0xd74e,0x3cc5 ,
- 0xf957,0x3076 ,0x1f89,0x1aa7 ,0xd45c,0x3d78 ,
- 0xf721,0x3179 ,0x1e7e,0x1b5d ,0xd178,0x3e15 ,
- 0xf4ec,0x3274 ,0x1d72,0x1c12 ,0xcea5,0x3e9d ,
- 0xf2b8,0x3368 ,0x1c64,0x1cc6 ,0xcbe2,0x3f0f ,
- 0xf087,0x3453 ,0x1b56,0x1d79 ,0xc932,0x3f6b ,
- 0xee58,0x3537 ,0x1a46,0x1e2b ,0xc695,0x3fb1 ,
- 0xec2b,0x3612 ,0x1935,0x1edc ,0xc40c,0x3fe1 ,
- 0xea02,0x36e5 ,0x1824,0x1f8c ,0xc197,0x3ffb ,
- 0xe7dc,0x37b0 ,0x1711,0x203a ,0xbf38,0x3fff ,
- 0xe5ba,0x3871 ,0x15fe,0x20e7 ,0xbcf0,0x3fec ,
- 0xe39c,0x392b ,0x14ea,0x2193 ,0xbabf,0x3fc4 ,
- 0xe182,0x39db ,0x13d5,0x223d ,0xb8a6,0x3f85 ,
- 0xdf6d,0x3a82 ,0x12bf,0x22e7 ,0xb6a5,0x3f30 ,
- 0xdd5d,0x3b21 ,0x11a8,0x238e ,0xb4be,0x3ec5 ,
- 0xdb52,0x3bb6 ,0x1091,0x2435 ,0xb2f2,0x3e45 ,
- 0xd94d,0x3c42 ,0x0f79,0x24da ,0xb140,0x3daf ,
- 0xd74e,0x3cc5 ,0x0e61,0x257e ,0xafa9,0x3d03 ,
- 0xd556,0x3d3f ,0x0d48,0x2620 ,0xae2e,0x3c42 ,
- 0xd363,0x3daf ,0x0c2e,0x26c1 ,0xacd0,0x3b6d ,
- 0xd178,0x3e15 ,0x0b14,0x2760 ,0xab8e,0x3a82 ,
- 0xcf94,0x3e72 ,0x09fa,0x27fe ,0xaa6a,0x3984 ,
- 0xcdb7,0x3ec5 ,0x08df,0x289a ,0xa963,0x3871 ,
- 0xcbe2,0x3f0f ,0x07c4,0x2935 ,0xa87b,0x374b ,
- 0xca15,0x3f4f ,0x06a9,0x29ce ,0xa7b1,0x3612 ,
- 0xc851,0x3f85 ,0x058d,0x2a65 ,0xa705,0x34c6 ,
- 0xc695,0x3fb1 ,0x0471,0x2afb ,0xa678,0x3368 ,
- 0xc4e2,0x3fd4 ,0x0355,0x2b8f ,0xa60b,0x31f8 ,
- 0xc338,0x3fec ,0x0239,0x2c21 ,0xa5bc,0x3076 ,
- 0xc197,0x3ffb ,0x011c,0x2cb2 ,0xa58d,0x2ee4 ,
- 0xc000,0x4000 ,0x0000,0x2d41 ,0xa57e,0x2d41 ,
- 0xbe73,0x3ffb ,0xfee4,0x2dcf ,0xa58d,0x2b8f ,
- 0xbcf0,0x3fec ,0xfdc7,0x2e5a ,0xa5bc,0x29ce ,
- 0xbb77,0x3fd4 ,0xfcab,0x2ee4 ,0xa60b,0x27fe ,
- 0xba09,0x3fb1 ,0xfb8f,0x2f6c ,0xa678,0x2620 ,
- 0xb8a6,0x3f85 ,0xfa73,0x2ff2 ,0xa705,0x2435 ,
- 0xb74d,0x3f4f ,0xf957,0x3076 ,0xa7b1,0x223d ,
- 0xb600,0x3f0f ,0xf83c,0x30f9 ,0xa87b,0x203a ,
- 0xb4be,0x3ec5 ,0xf721,0x3179 ,0xa963,0x1e2b ,
- 0xb388,0x3e72 ,0xf606,0x31f8 ,0xaa6a,0x1c12 ,
- 0xb25e,0x3e15 ,0xf4ec,0x3274 ,0xab8e,0x19ef ,
- 0xb140,0x3daf ,0xf3d2,0x32ef ,0xacd0,0x17c4 ,
- 0xb02d,0x3d3f ,0xf2b8,0x3368 ,0xae2e,0x1590 ,
- 0xaf28,0x3cc5 ,0xf19f,0x33df ,0xafa9,0x1354 ,
- 0xae2e,0x3c42 ,0xf087,0x3453 ,0xb140,0x1112 ,
- 0xad41,0x3bb6 ,0xef6f,0x34c6 ,0xb2f2,0x0eca ,
- 0xac61,0x3b21 ,0xee58,0x3537 ,0xb4be,0x0c7c ,
- 0xab8e,0x3a82 ,0xed41,0x35a5 ,0xb6a5,0x0a2b ,
- 0xaac8,0x39db ,0xec2b,0x3612 ,0xb8a6,0x07d6 ,
- 0xaa0f,0x392b ,0xeb16,0x367d ,0xbabf,0x057e ,
- 0xa963,0x3871 ,0xea02,0x36e5 ,0xbcf0,0x0324 ,
- 0xa8c5,0x37b0 ,0xe8ef,0x374b ,0xbf38,0x00c9 ,
- 0xa834,0x36e5 ,0xe7dc,0x37b0 ,0xc197,0xfe6e ,
- 0xa7b1,0x3612 ,0xe6cb,0x3812 ,0xc40c,0xfc13 ,
- 0xa73b,0x3537 ,0xe5ba,0x3871 ,0xc695,0xf9ba ,
- 0xa6d3,0x3453 ,0xe4aa,0x38cf ,0xc932,0xf763 ,
- 0xa678,0x3368 ,0xe39c,0x392b ,0xcbe2,0xf50f ,
- 0xa62c,0x3274 ,0xe28e,0x3984 ,0xcea5,0xf2bf ,
- 0xa5ed,0x3179 ,0xe182,0x39db ,0xd178,0xf073 ,
- 0xa5bc,0x3076 ,0xe077,0x3a30 ,0xd45c,0xee2d ,
- 0xa599,0x2f6c ,0xdf6d,0x3a82 ,0xd74e,0xebed ,
- 0xa585,0x2e5a ,0xde64,0x3ad3 ,0xda4f,0xe9b4 ,
- 0xa57e,0x2d41 ,0xdd5d,0x3b21 ,0xdd5d,0xe782 ,
- 0xa585,0x2c21 ,0xdc57,0x3b6d ,0xe077,0xe559 ,
- 0xa599,0x2afb ,0xdb52,0x3bb6 ,0xe39c,0xe33a ,
- 0xa5bc,0x29ce ,0xda4f,0x3bfd ,0xe6cb,0xe124 ,
- 0xa5ed,0x289a ,0xd94d,0x3c42 ,0xea02,0xdf19 ,
- 0xa62c,0x2760 ,0xd84d,0x3c85 ,0xed41,0xdd19 ,
- 0xa678,0x2620 ,0xd74e,0x3cc5 ,0xf087,0xdb26 ,
- 0xa6d3,0x24da ,0xd651,0x3d03 ,0xf3d2,0xd93f ,
- 0xa73b,0x238e ,0xd556,0x3d3f ,0xf721,0xd766 ,
- 0xa7b1,0x223d ,0xd45c,0x3d78 ,0xfa73,0xd59b ,
- 0xa834,0x20e7 ,0xd363,0x3daf ,0xfdc7,0xd3df ,
- 0xa8c5,0x1f8c ,0xd26d,0x3de3 ,0x011c,0xd231 ,
- 0xa963,0x1e2b ,0xd178,0x3e15 ,0x0471,0xd094 ,
- 0xaa0f,0x1cc6 ,0xd085,0x3e45 ,0x07c4,0xcf07 ,
- 0xaac8,0x1b5d ,0xcf94,0x3e72 ,0x0b14,0xcd8c ,
- 0xab8e,0x19ef ,0xcea5,0x3e9d ,0x0e61,0xcc21 ,
- 0xac61,0x187e ,0xcdb7,0x3ec5 ,0x11a8,0xcac9 ,
- 0xad41,0x1709 ,0xcccc,0x3eeb ,0x14ea,0xc983 ,
- 0xae2e,0x1590 ,0xcbe2,0x3f0f ,0x1824,0xc850 ,
- 0xaf28,0x1413 ,0xcafb,0x3f30 ,0x1b56,0xc731 ,
- 0xb02d,0x1294 ,0xca15,0x3f4f ,0x1e7e,0xc625 ,
- 0xb140,0x1112 ,0xc932,0x3f6b ,0x219c,0xc52d ,
- 0xb25e,0x0f8d ,0xc851,0x3f85 ,0x24ae,0xc44a ,
- 0xb388,0x0e06 ,0xc772,0x3f9c ,0x27b3,0xc37b ,
- 0xb4be,0x0c7c ,0xc695,0x3fb1 ,0x2aaa,0xc2c1 ,
- 0xb600,0x0af1 ,0xc5ba,0x3fc4 ,0x2d93,0xc21d ,
- 0xb74d,0x0964 ,0xc4e2,0x3fd4 ,0x306c,0xc18e ,
- 0xb8a6,0x07d6 ,0xc40c,0x3fe1 ,0x3334,0xc115 ,
- 0xba09,0x0646 ,0xc338,0x3fec ,0x35eb,0xc0b1 ,
- 0xbb77,0x04b5 ,0xc266,0x3ff5 ,0x388e,0xc064 ,
- 0xbcf0,0x0324 ,0xc197,0x3ffb ,0x3b1e,0xc02c ,
- 0xbe73,0x0192 ,0xc0ca,0x3fff ,0x3d9a,0xc00b ,
- 0x4000,0x0000 ,0x3f9b,0x0065 ,0x3f36,0x00c9 ,
- 0x3ed0,0x012e ,0x3e69,0x0192 ,0x3e02,0x01f7 ,
- 0x3d9a,0x025b ,0x3d31,0x02c0 ,0x3cc8,0x0324 ,
- 0x3c5f,0x0388 ,0x3bf4,0x03ed ,0x3b8a,0x0451 ,
- 0x3b1e,0x04b5 ,0x3ab2,0x051a ,0x3a46,0x057e ,
- 0x39d9,0x05e2 ,0x396b,0x0646 ,0x38fd,0x06aa ,
- 0x388e,0x070e ,0x381f,0x0772 ,0x37af,0x07d6 ,
- 0x373f,0x0839 ,0x36ce,0x089d ,0x365d,0x0901 ,
- 0x35eb,0x0964 ,0x3578,0x09c7 ,0x3505,0x0a2b ,
- 0x3492,0x0a8e ,0x341e,0x0af1 ,0x33a9,0x0b54 ,
- 0x3334,0x0bb7 ,0x32bf,0x0c1a ,0x3249,0x0c7c ,
- 0x31d2,0x0cdf ,0x315b,0x0d41 ,0x30e4,0x0da4 ,
- 0x306c,0x0e06 ,0x2ff4,0x0e68 ,0x2f7b,0x0eca ,
- 0x2f02,0x0f2b ,0x2e88,0x0f8d ,0x2e0e,0x0fee ,
- 0x2d93,0x1050 ,0x2d18,0x10b1 ,0x2c9d,0x1112 ,
- 0x2c21,0x1173 ,0x2ba4,0x11d3 ,0x2b28,0x1234 ,
- 0x2aaa,0x1294 ,0x2a2d,0x12f4 ,0x29af,0x1354 ,
- 0x2931,0x13b4 ,0x28b2,0x1413 ,0x2833,0x1473 ,
- 0x27b3,0x14d2 ,0x2733,0x1531 ,0x26b3,0x1590 ,
- 0x2632,0x15ee ,0x25b1,0x164c ,0x252f,0x16ab ,
- 0x24ae,0x1709 ,0x242b,0x1766 ,0x23a9,0x17c4 ,
- 0x2326,0x1821 ,0x22a3,0x187e ,0x221f,0x18db ,
- 0x219c,0x1937 ,0x2117,0x1993 ,0x2093,0x19ef ,
- 0x200e,0x1a4b ,0x1f89,0x1aa7 ,0x1f04,0x1b02 ,
- 0x1e7e,0x1b5d ,0x1df8,0x1bb8 ,0x1d72,0x1c12 ,
- 0x1ceb,0x1c6c ,0x1c64,0x1cc6 ,0x1bdd,0x1d20 ,
- 0x1b56,0x1d79 ,0x1ace,0x1dd3 ,0x1a46,0x1e2b ,
- 0x19be,0x1e84 ,0x1935,0x1edc ,0x18ad,0x1f34 ,
- 0x1824,0x1f8c ,0x179b,0x1fe3 ,0x1711,0x203a ,
- 0x1688,0x2091 ,0x15fe,0x20e7 ,0x1574,0x213d ,
- 0x14ea,0x2193 ,0x145f,0x21e8 ,0x13d5,0x223d ,
- 0x134a,0x2292 ,0x12bf,0x22e7 ,0x1234,0x233b ,
- 0x11a8,0x238e ,0x111d,0x23e2 ,0x1091,0x2435 ,
- 0x1005,0x2488 ,0x0f79,0x24da ,0x0eed,0x252c ,
- 0x0e61,0x257e ,0x0dd4,0x25cf ,0x0d48,0x2620 ,
- 0x0cbb,0x2671 ,0x0c2e,0x26c1 ,0x0ba1,0x2711 ,
- 0x0b14,0x2760 ,0x0a87,0x27af ,0x09fa,0x27fe ,
- 0x096d,0x284c ,0x08df,0x289a ,0x0852,0x28e7 ,
- 0x07c4,0x2935 ,0x0736,0x2981 ,0x06a9,0x29ce ,
- 0x061b,0x2a1a ,0x058d,0x2a65 ,0x04ff,0x2ab0 ,
- 0x0471,0x2afb ,0x03e3,0x2b45 ,0x0355,0x2b8f ,
- 0x02c7,0x2bd8 ,0x0239,0x2c21 ,0x01aa,0x2c6a ,
- 0x011c,0x2cb2 ,0x008e,0x2cfa ,0x0000,0x2d41 ,
- 0xff72,0x2d88 ,0xfee4,0x2dcf ,0xfe56,0x2e15 ,
- 0xfdc7,0x2e5a ,0xfd39,0x2e9f ,0xfcab,0x2ee4 ,
- 0xfc1d,0x2f28 ,0xfb8f,0x2f6c ,0xfb01,0x2faf ,
- 0xfa73,0x2ff2 ,0xf9e5,0x3034 ,0xf957,0x3076 ,
- 0xf8ca,0x30b8 ,0xf83c,0x30f9 ,0xf7ae,0x3139 ,
- 0xf721,0x3179 ,0xf693,0x31b9 ,0xf606,0x31f8 ,
- 0xf579,0x3236 ,0xf4ec,0x3274 ,0xf45f,0x32b2 ,
- 0xf3d2,0x32ef ,0xf345,0x332c ,0xf2b8,0x3368 ,
- 0xf22c,0x33a3 ,0xf19f,0x33df ,0xf113,0x3419 ,
- 0xf087,0x3453 ,0xeffb,0x348d ,0xef6f,0x34c6 ,
- 0xeee3,0x34ff ,0xee58,0x3537 ,0xedcc,0x356e ,
- 0xed41,0x35a5 ,0xecb6,0x35dc ,0xec2b,0x3612 ,
- 0xeba1,0x3648 ,0xeb16,0x367d ,0xea8c,0x36b1 ,
- 0xea02,0x36e5 ,0xe978,0x3718 ,0xe8ef,0x374b ,
- 0xe865,0x377e ,0xe7dc,0x37b0 ,0xe753,0x37e1 ,
- 0xe6cb,0x3812 ,0xe642,0x3842 ,0xe5ba,0x3871 ,
- 0xe532,0x38a1 ,0xe4aa,0x38cf ,0xe423,0x38fd ,
- 0xe39c,0x392b ,0xe315,0x3958 ,0xe28e,0x3984 ,
- 0xe208,0x39b0 ,0xe182,0x39db ,0xe0fc,0x3a06 ,
- 0xe077,0x3a30 ,0xdff2,0x3a59 ,0xdf6d,0x3a82 ,
- 0xdee9,0x3aab ,0xde64,0x3ad3 ,0xdde1,0x3afa ,
- 0xdd5d,0x3b21 ,0xdcda,0x3b47 ,0xdc57,0x3b6d ,
- 0xdbd5,0x3b92 ,0xdb52,0x3bb6 ,0xdad1,0x3bda ,
- 0xda4f,0x3bfd ,0xd9ce,0x3c20 ,0xd94d,0x3c42 ,
- 0xd8cd,0x3c64 ,0xd84d,0x3c85 ,0xd7cd,0x3ca5 ,
- 0xd74e,0x3cc5 ,0xd6cf,0x3ce4 ,0xd651,0x3d03 ,
- 0xd5d3,0x3d21 ,0xd556,0x3d3f ,0xd4d8,0x3d5b ,
- 0xd45c,0x3d78 ,0xd3df,0x3d93 ,0xd363,0x3daf ,
- 0xd2e8,0x3dc9 ,0xd26d,0x3de3 ,0xd1f2,0x3dfc ,
- 0xd178,0x3e15 ,0xd0fe,0x3e2d ,0xd085,0x3e45 ,
- 0xd00c,0x3e5c ,0xcf94,0x3e72 ,0xcf1c,0x3e88 ,
- 0xcea5,0x3e9d ,0xce2e,0x3eb1 ,0xcdb7,0x3ec5 ,
- 0xcd41,0x3ed8 ,0xcccc,0x3eeb ,0xcc57,0x3efd ,
- 0xcbe2,0x3f0f ,0xcb6e,0x3f20 ,0xcafb,0x3f30 ,
- 0xca88,0x3f40 ,0xca15,0x3f4f ,0xc9a3,0x3f5d ,
- 0xc932,0x3f6b ,0xc8c1,0x3f78 ,0xc851,0x3f85 ,
- 0xc7e1,0x3f91 ,0xc772,0x3f9c ,0xc703,0x3fa7 ,
- 0xc695,0x3fb1 ,0xc627,0x3fbb ,0xc5ba,0x3fc4 ,
- 0xc54e,0x3fcc ,0xc4e2,0x3fd4 ,0xc476,0x3fdb ,
- 0xc40c,0x3fe1 ,0xc3a1,0x3fe7 ,0xc338,0x3fec ,
- 0xc2cf,0x3ff1 ,0xc266,0x3ff5 ,0xc1fe,0x3ff8 ,
- 0xc197,0x3ffb ,0xc130,0x3ffd ,0xc0ca,0x3fff ,
- 0xc065,0x4000 ,0xc000,0x4000 ,0xbf9c,0x4000 ,
- 0xbf38,0x3fff ,0xbed5,0x3ffd ,0xbe73,0x3ffb ,
- 0xbe11,0x3ff8 ,0xbdb0,0x3ff5 ,0xbd50,0x3ff1 ,
- 0xbcf0,0x3fec ,0xbc91,0x3fe7 ,0xbc32,0x3fe1 ,
- 0xbbd4,0x3fdb ,0xbb77,0x3fd4 ,0xbb1b,0x3fcc ,
- 0xbabf,0x3fc4 ,0xba64,0x3fbb ,0xba09,0x3fb1 ,
- 0xb9af,0x3fa7 ,0xb956,0x3f9c ,0xb8fd,0x3f91 ,
- 0xb8a6,0x3f85 ,0xb84f,0x3f78 ,0xb7f8,0x3f6b ,
- 0xb7a2,0x3f5d ,0xb74d,0x3f4f ,0xb6f9,0x3f40 ,
- 0xb6a5,0x3f30 ,0xb652,0x3f20 ,0xb600,0x3f0f ,
- 0xb5af,0x3efd ,0xb55e,0x3eeb ,0xb50e,0x3ed8 ,
- 0xb4be,0x3ec5 ,0xb470,0x3eb1 ,0xb422,0x3e9d ,
- 0xb3d5,0x3e88 ,0xb388,0x3e72 ,0xb33d,0x3e5c ,
- 0xb2f2,0x3e45 ,0xb2a7,0x3e2d ,0xb25e,0x3e15 ,
- 0xb215,0x3dfc ,0xb1cd,0x3de3 ,0xb186,0x3dc9 ,
- 0xb140,0x3daf ,0xb0fa,0x3d93 ,0xb0b5,0x3d78 ,
- 0xb071,0x3d5b ,0xb02d,0x3d3f ,0xafeb,0x3d21 ,
- 0xafa9,0x3d03 ,0xaf68,0x3ce4 ,0xaf28,0x3cc5 ,
- 0xaee8,0x3ca5 ,0xaea9,0x3c85 ,0xae6b,0x3c64 ,
- 0xae2e,0x3c42 ,0xadf2,0x3c20 ,0xadb6,0x3bfd ,
- 0xad7b,0x3bda ,0xad41,0x3bb6 ,0xad08,0x3b92 ,
- 0xacd0,0x3b6d ,0xac98,0x3b47 ,0xac61,0x3b21 ,
- 0xac2b,0x3afa ,0xabf6,0x3ad3 ,0xabc2,0x3aab ,
- 0xab8e,0x3a82 ,0xab5b,0x3a59 ,0xab29,0x3a30 ,
- 0xaaf8,0x3a06 ,0xaac8,0x39db ,0xaa98,0x39b0 ,
- 0xaa6a,0x3984 ,0xaa3c,0x3958 ,0xaa0f,0x392b ,
- 0xa9e3,0x38fd ,0xa9b7,0x38cf ,0xa98d,0x38a1 ,
- 0xa963,0x3871 ,0xa93a,0x3842 ,0xa912,0x3812 ,
- 0xa8eb,0x37e1 ,0xa8c5,0x37b0 ,0xa89f,0x377e ,
- 0xa87b,0x374b ,0xa857,0x3718 ,0xa834,0x36e5 ,
- 0xa812,0x36b1 ,0xa7f1,0x367d ,0xa7d0,0x3648 ,
- 0xa7b1,0x3612 ,0xa792,0x35dc ,0xa774,0x35a5 ,
- 0xa757,0x356e ,0xa73b,0x3537 ,0xa71f,0x34ff ,
- 0xa705,0x34c6 ,0xa6eb,0x348d ,0xa6d3,0x3453 ,
- 0xa6bb,0x3419 ,0xa6a4,0x33df ,0xa68e,0x33a3 ,
- 0xa678,0x3368 ,0xa664,0x332c ,0xa650,0x32ef ,
- 0xa63e,0x32b2 ,0xa62c,0x3274 ,0xa61b,0x3236 ,
- 0xa60b,0x31f8 ,0xa5fb,0x31b9 ,0xa5ed,0x3179 ,
- 0xa5e0,0x3139 ,0xa5d3,0x30f9 ,0xa5c7,0x30b8 ,
- 0xa5bc,0x3076 ,0xa5b2,0x3034 ,0xa5a9,0x2ff2 ,
- 0xa5a1,0x2faf ,0xa599,0x2f6c ,0xa593,0x2f28 ,
- 0xa58d,0x2ee4 ,0xa588,0x2e9f ,0xa585,0x2e5a ,
- 0xa581,0x2e15 ,0xa57f,0x2dcf ,0xa57e,0x2d88 ,
- 0xa57e,0x2d41 ,0xa57e,0x2cfa ,0xa57f,0x2cb2 ,
- 0xa581,0x2c6a ,0xa585,0x2c21 ,0xa588,0x2bd8 ,
- 0xa58d,0x2b8f ,0xa593,0x2b45 ,0xa599,0x2afb ,
- 0xa5a1,0x2ab0 ,0xa5a9,0x2a65 ,0xa5b2,0x2a1a ,
- 0xa5bc,0x29ce ,0xa5c7,0x2981 ,0xa5d3,0x2935 ,
- 0xa5e0,0x28e7 ,0xa5ed,0x289a ,0xa5fb,0x284c ,
- 0xa60b,0x27fe ,0xa61b,0x27af ,0xa62c,0x2760 ,
- 0xa63e,0x2711 ,0xa650,0x26c1 ,0xa664,0x2671 ,
- 0xa678,0x2620 ,0xa68e,0x25cf ,0xa6a4,0x257e ,
- 0xa6bb,0x252c ,0xa6d3,0x24da ,0xa6eb,0x2488 ,
- 0xa705,0x2435 ,0xa71f,0x23e2 ,0xa73b,0x238e ,
- 0xa757,0x233b ,0xa774,0x22e7 ,0xa792,0x2292 ,
- 0xa7b1,0x223d ,0xa7d0,0x21e8 ,0xa7f1,0x2193 ,
- 0xa812,0x213d ,0xa834,0x20e7 ,0xa857,0x2091 ,
- 0xa87b,0x203a ,0xa89f,0x1fe3 ,0xa8c5,0x1f8c ,
- 0xa8eb,0x1f34 ,0xa912,0x1edc ,0xa93a,0x1e84 ,
- 0xa963,0x1e2b ,0xa98d,0x1dd3 ,0xa9b7,0x1d79 ,
- 0xa9e3,0x1d20 ,0xaa0f,0x1cc6 ,0xaa3c,0x1c6c ,
- 0xaa6a,0x1c12 ,0xaa98,0x1bb8 ,0xaac8,0x1b5d ,
- 0xaaf8,0x1b02 ,0xab29,0x1aa7 ,0xab5b,0x1a4b ,
- 0xab8e,0x19ef ,0xabc2,0x1993 ,0xabf6,0x1937 ,
- 0xac2b,0x18db ,0xac61,0x187e ,0xac98,0x1821 ,
- 0xacd0,0x17c4 ,0xad08,0x1766 ,0xad41,0x1709 ,
- 0xad7b,0x16ab ,0xadb6,0x164c ,0xadf2,0x15ee ,
- 0xae2e,0x1590 ,0xae6b,0x1531 ,0xaea9,0x14d2 ,
- 0xaee8,0x1473 ,0xaf28,0x1413 ,0xaf68,0x13b4 ,
- 0xafa9,0x1354 ,0xafeb,0x12f4 ,0xb02d,0x1294 ,
- 0xb071,0x1234 ,0xb0b5,0x11d3 ,0xb0fa,0x1173 ,
- 0xb140,0x1112 ,0xb186,0x10b1 ,0xb1cd,0x1050 ,
- 0xb215,0x0fee ,0xb25e,0x0f8d ,0xb2a7,0x0f2b ,
- 0xb2f2,0x0eca ,0xb33d,0x0e68 ,0xb388,0x0e06 ,
- 0xb3d5,0x0da4 ,0xb422,0x0d41 ,0xb470,0x0cdf ,
- 0xb4be,0x0c7c ,0xb50e,0x0c1a ,0xb55e,0x0bb7 ,
- 0xb5af,0x0b54 ,0xb600,0x0af1 ,0xb652,0x0a8e ,
- 0xb6a5,0x0a2b ,0xb6f9,0x09c7 ,0xb74d,0x0964 ,
- 0xb7a2,0x0901 ,0xb7f8,0x089d ,0xb84f,0x0839 ,
- 0xb8a6,0x07d6 ,0xb8fd,0x0772 ,0xb956,0x070e ,
- 0xb9af,0x06aa ,0xba09,0x0646 ,0xba64,0x05e2 ,
- 0xbabf,0x057e ,0xbb1b,0x051a ,0xbb77,0x04b5 ,
- 0xbbd4,0x0451 ,0xbc32,0x03ed ,0xbc91,0x0388 ,
- 0xbcf0,0x0324 ,0xbd50,0x02c0 ,0xbdb0,0x025b ,
- 0xbe11,0x01f7 ,0xbe73,0x0192 ,0xbed5,0x012e ,
- 0xbf38,0x00c9 ,0xbf9c,0x0065 };
-
-
-extern const int s_Q14R_8;
-const int s_Q14R_8 = 1024;
-extern const unsigned short t_Q14R_8[2032];
-const unsigned short t_Q14R_8[2032] = {
- 0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
- 0x3b21,0x187e ,0x3ec5,0x0c7c ,0x3537,0x238e ,
- 0x2d41,0x2d41 ,0x3b21,0x187e ,0x187e,0x3b21 ,
- 0x187e,0x3b21 ,0x3537,0x238e ,0xf384,0x3ec5 ,
- 0x0000,0x4000 ,0x2d41,0x2d41 ,0xd2bf,0x2d41 ,
- 0xe782,0x3b21 ,0x238e,0x3537 ,0xc13b,0x0c7c ,
- 0xd2bf,0x2d41 ,0x187e,0x3b21 ,0xc4df,0xe782 ,
- 0xc4df,0x187e ,0x0c7c,0x3ec5 ,0xdc72,0xcac9 ,
- 0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
- 0x3fb1,0x0646 ,0x3fec,0x0324 ,0x3f4f,0x0964 ,
- 0x3ec5,0x0c7c ,0x3fb1,0x0646 ,0x3d3f,0x1294 ,
- 0x3d3f,0x1294 ,0x3f4f,0x0964 ,0x39db,0x1b5d ,
- 0x3b21,0x187e ,0x3ec5,0x0c7c ,0x3537,0x238e ,
- 0x3871,0x1e2b ,0x3e15,0x0f8d ,0x2f6c,0x2afb ,
- 0x3537,0x238e ,0x3d3f,0x1294 ,0x289a,0x3179 ,
- 0x3179,0x289a ,0x3c42,0x1590 ,0x20e7,0x36e5 ,
- 0x2d41,0x2d41 ,0x3b21,0x187e ,0x187e,0x3b21 ,
- 0x289a,0x3179 ,0x39db,0x1b5d ,0x0f8d,0x3e15 ,
- 0x238e,0x3537 ,0x3871,0x1e2b ,0x0646,0x3fb1 ,
- 0x1e2b,0x3871 ,0x36e5,0x20e7 ,0xfcdc,0x3fec ,
- 0x187e,0x3b21 ,0x3537,0x238e ,0xf384,0x3ec5 ,
- 0x1294,0x3d3f ,0x3368,0x2620 ,0xea70,0x3c42 ,
- 0x0c7c,0x3ec5 ,0x3179,0x289a ,0xe1d5,0x3871 ,
- 0x0646,0x3fb1 ,0x2f6c,0x2afb ,0xd9e0,0x3368 ,
- 0x0000,0x4000 ,0x2d41,0x2d41 ,0xd2bf,0x2d41 ,
- 0xf9ba,0x3fb1 ,0x2afb,0x2f6c ,0xcc98,0x2620 ,
- 0xf384,0x3ec5 ,0x289a,0x3179 ,0xc78f,0x1e2b ,
- 0xed6c,0x3d3f ,0x2620,0x3368 ,0xc3be,0x1590 ,
- 0xe782,0x3b21 ,0x238e,0x3537 ,0xc13b,0x0c7c ,
- 0xe1d5,0x3871 ,0x20e7,0x36e5 ,0xc014,0x0324 ,
- 0xdc72,0x3537 ,0x1e2b,0x3871 ,0xc04f,0xf9ba ,
- 0xd766,0x3179 ,0x1b5d,0x39db ,0xc1eb,0xf073 ,
- 0xd2bf,0x2d41 ,0x187e,0x3b21 ,0xc4df,0xe782 ,
- 0xce87,0x289a ,0x1590,0x3c42 ,0xc91b,0xdf19 ,
- 0xcac9,0x238e ,0x1294,0x3d3f ,0xce87,0xd766 ,
- 0xc78f,0x1e2b ,0x0f8d,0x3e15 ,0xd505,0xd094 ,
- 0xc4df,0x187e ,0x0c7c,0x3ec5 ,0xdc72,0xcac9 ,
- 0xc2c1,0x1294 ,0x0964,0x3f4f ,0xe4a3,0xc625 ,
- 0xc13b,0x0c7c ,0x0646,0x3fb1 ,0xed6c,0xc2c1 ,
- 0xc04f,0x0646 ,0x0324,0x3fec ,0xf69c,0xc0b1 ,
- 0x4000,0x0000 ,0x4000,0x0000 ,0x4000,0x0000 ,
- 0x3ffb,0x0192 ,0x3fff,0x00c9 ,0x3ff5,0x025b ,
- 0x3fec,0x0324 ,0x3ffb,0x0192 ,0x3fd4,0x04b5 ,
- 0x3fd4,0x04b5 ,0x3ff5,0x025b ,0x3f9c,0x070e ,
- 0x3fb1,0x0646 ,0x3fec,0x0324 ,0x3f4f,0x0964 ,
- 0x3f85,0x07d6 ,0x3fe1,0x03ed ,0x3eeb,0x0bb7 ,
- 0x3f4f,0x0964 ,0x3fd4,0x04b5 ,0x3e72,0x0e06 ,
- 0x3f0f,0x0af1 ,0x3fc4,0x057e ,0x3de3,0x1050 ,
- 0x3ec5,0x0c7c ,0x3fb1,0x0646 ,0x3d3f,0x1294 ,
- 0x3e72,0x0e06 ,0x3f9c,0x070e ,0x3c85,0x14d2 ,
- 0x3e15,0x0f8d ,0x3f85,0x07d6 ,0x3bb6,0x1709 ,
- 0x3daf,0x1112 ,0x3f6b,0x089d ,0x3ad3,0x1937 ,
- 0x3d3f,0x1294 ,0x3f4f,0x0964 ,0x39db,0x1b5d ,
- 0x3cc5,0x1413 ,0x3f30,0x0a2b ,0x38cf,0x1d79 ,
- 0x3c42,0x1590 ,0x3f0f,0x0af1 ,0x37b0,0x1f8c ,
- 0x3bb6,0x1709 ,0x3eeb,0x0bb7 ,0x367d,0x2193 ,
- 0x3b21,0x187e ,0x3ec5,0x0c7c ,0x3537,0x238e ,
- 0x3a82,0x19ef ,0x3e9d,0x0d41 ,0x33df,0x257e ,
- 0x39db,0x1b5d ,0x3e72,0x0e06 ,0x3274,0x2760 ,
- 0x392b,0x1cc6 ,0x3e45,0x0eca ,0x30f9,0x2935 ,
- 0x3871,0x1e2b ,0x3e15,0x0f8d ,0x2f6c,0x2afb ,
- 0x37b0,0x1f8c ,0x3de3,0x1050 ,0x2dcf,0x2cb2 ,
- 0x36e5,0x20e7 ,0x3daf,0x1112 ,0x2c21,0x2e5a ,
- 0x3612,0x223d ,0x3d78,0x11d3 ,0x2a65,0x2ff2 ,
- 0x3537,0x238e ,0x3d3f,0x1294 ,0x289a,0x3179 ,
- 0x3453,0x24da ,0x3d03,0x1354 ,0x26c1,0x32ef ,
- 0x3368,0x2620 ,0x3cc5,0x1413 ,0x24da,0x3453 ,
- 0x3274,0x2760 ,0x3c85,0x14d2 ,0x22e7,0x35a5 ,
- 0x3179,0x289a ,0x3c42,0x1590 ,0x20e7,0x36e5 ,
- 0x3076,0x29ce ,0x3bfd,0x164c ,0x1edc,0x3812 ,
- 0x2f6c,0x2afb ,0x3bb6,0x1709 ,0x1cc6,0x392b ,
- 0x2e5a,0x2c21 ,0x3b6d,0x17c4 ,0x1aa7,0x3a30 ,
- 0x2d41,0x2d41 ,0x3b21,0x187e ,0x187e,0x3b21 ,
- 0x2c21,0x2e5a ,0x3ad3,0x1937 ,0x164c,0x3bfd ,
- 0x2afb,0x2f6c ,0x3a82,0x19ef ,0x1413,0x3cc5 ,
- 0x29ce,0x3076 ,0x3a30,0x1aa7 ,0x11d3,0x3d78 ,
- 0x289a,0x3179 ,0x39db,0x1b5d ,0x0f8d,0x3e15 ,
- 0x2760,0x3274 ,0x3984,0x1c12 ,0x0d41,0x3e9d ,
- 0x2620,0x3368 ,0x392b,0x1cc6 ,0x0af1,0x3f0f ,
- 0x24da,0x3453 ,0x38cf,0x1d79 ,0x089d,0x3f6b ,
- 0x238e,0x3537 ,0x3871,0x1e2b ,0x0646,0x3fb1 ,
- 0x223d,0x3612 ,0x3812,0x1edc ,0x03ed,0x3fe1 ,
- 0x20e7,0x36e5 ,0x37b0,0x1f8c ,0x0192,0x3ffb ,
- 0x1f8c,0x37b0 ,0x374b,0x203a ,0xff37,0x3fff ,
- 0x1e2b,0x3871 ,0x36e5,0x20e7 ,0xfcdc,0x3fec ,
- 0x1cc6,0x392b ,0x367d,0x2193 ,0xfa82,0x3fc4 ,
- 0x1b5d,0x39db ,0x3612,0x223d ,0xf82a,0x3f85 ,
- 0x19ef,0x3a82 ,0x35a5,0x22e7 ,0xf5d5,0x3f30 ,
- 0x187e,0x3b21 ,0x3537,0x238e ,0xf384,0x3ec5 ,
- 0x1709,0x3bb6 ,0x34c6,0x2435 ,0xf136,0x3e45 ,
- 0x1590,0x3c42 ,0x3453,0x24da ,0xeeee,0x3daf ,
- 0x1413,0x3cc5 ,0x33df,0x257e ,0xecac,0x3d03 ,
- 0x1294,0x3d3f ,0x3368,0x2620 ,0xea70,0x3c42 ,
- 0x1112,0x3daf ,0x32ef,0x26c1 ,0xe83c,0x3b6d ,
- 0x0f8d,0x3e15 ,0x3274,0x2760 ,0xe611,0x3a82 ,
- 0x0e06,0x3e72 ,0x31f8,0x27fe ,0xe3ee,0x3984 ,
- 0x0c7c,0x3ec5 ,0x3179,0x289a ,0xe1d5,0x3871 ,
- 0x0af1,0x3f0f ,0x30f9,0x2935 ,0xdfc6,0x374b ,
- 0x0964,0x3f4f ,0x3076,0x29ce ,0xddc3,0x3612 ,
- 0x07d6,0x3f85 ,0x2ff2,0x2a65 ,0xdbcb,0x34c6 ,
- 0x0646,0x3fb1 ,0x2f6c,0x2afb ,0xd9e0,0x3368 ,
- 0x04b5,0x3fd4 ,0x2ee4,0x2b8f ,0xd802,0x31f8 ,
- 0x0324,0x3fec ,0x2e5a,0x2c21 ,0xd632,0x3076 ,
- 0x0192,0x3ffb ,0x2dcf,0x2cb2 ,0xd471,0x2ee4 ,
- 0x0000,0x4000 ,0x2d41,0x2d41 ,0xd2bf,0x2d41 ,
- 0xfe6e,0x3ffb ,0x2cb2,0x2dcf ,0xd11c,0x2b8f ,
- 0xfcdc,0x3fec ,0x2c21,0x2e5a ,0xcf8a,0x29ce ,
- 0xfb4b,0x3fd4 ,0x2b8f,0x2ee4 ,0xce08,0x27fe ,
- 0xf9ba,0x3fb1 ,0x2afb,0x2f6c ,0xcc98,0x2620 ,
- 0xf82a,0x3f85 ,0x2a65,0x2ff2 ,0xcb3a,0x2435 ,
- 0xf69c,0x3f4f ,0x29ce,0x3076 ,0xc9ee,0x223d ,
- 0xf50f,0x3f0f ,0x2935,0x30f9 ,0xc8b5,0x203a ,
- 0xf384,0x3ec5 ,0x289a,0x3179 ,0xc78f,0x1e2b ,
- 0xf1fa,0x3e72 ,0x27fe,0x31f8 ,0xc67c,0x1c12 ,
- 0xf073,0x3e15 ,0x2760,0x3274 ,0xc57e,0x19ef ,
- 0xeeee,0x3daf ,0x26c1,0x32ef ,0xc493,0x17c4 ,
- 0xed6c,0x3d3f ,0x2620,0x3368 ,0xc3be,0x1590 ,
- 0xebed,0x3cc5 ,0x257e,0x33df ,0xc2fd,0x1354 ,
- 0xea70,0x3c42 ,0x24da,0x3453 ,0xc251,0x1112 ,
- 0xe8f7,0x3bb6 ,0x2435,0x34c6 ,0xc1bb,0x0eca ,
- 0xe782,0x3b21 ,0x238e,0x3537 ,0xc13b,0x0c7c ,
- 0xe611,0x3a82 ,0x22e7,0x35a5 ,0xc0d0,0x0a2b ,
- 0xe4a3,0x39db ,0x223d,0x3612 ,0xc07b,0x07d6 ,
- 0xe33a,0x392b ,0x2193,0x367d ,0xc03c,0x057e ,
- 0xe1d5,0x3871 ,0x20e7,0x36e5 ,0xc014,0x0324 ,
- 0xe074,0x37b0 ,0x203a,0x374b ,0xc001,0x00c9 ,
- 0xdf19,0x36e5 ,0x1f8c,0x37b0 ,0xc005,0xfe6e ,
- 0xddc3,0x3612 ,0x1edc,0x3812 ,0xc01f,0xfc13 ,
- 0xdc72,0x3537 ,0x1e2b,0x3871 ,0xc04f,0xf9ba ,
- 0xdb26,0x3453 ,0x1d79,0x38cf ,0xc095,0xf763 ,
- 0xd9e0,0x3368 ,0x1cc6,0x392b ,0xc0f1,0xf50f ,
- 0xd8a0,0x3274 ,0x1c12,0x3984 ,0xc163,0xf2bf ,
- 0xd766,0x3179 ,0x1b5d,0x39db ,0xc1eb,0xf073 ,
- 0xd632,0x3076 ,0x1aa7,0x3a30 ,0xc288,0xee2d ,
- 0xd505,0x2f6c ,0x19ef,0x3a82 ,0xc33b,0xebed ,
- 0xd3df,0x2e5a ,0x1937,0x3ad3 ,0xc403,0xe9b4 ,
- 0xd2bf,0x2d41 ,0x187e,0x3b21 ,0xc4df,0xe782 ,
- 0xd1a6,0x2c21 ,0x17c4,0x3b6d ,0xc5d0,0xe559 ,
- 0xd094,0x2afb ,0x1709,0x3bb6 ,0xc6d5,0xe33a ,
- 0xcf8a,0x29ce ,0x164c,0x3bfd ,0xc7ee,0xe124 ,
- 0xce87,0x289a ,0x1590,0x3c42 ,0xc91b,0xdf19 ,
- 0xcd8c,0x2760 ,0x14d2,0x3c85 ,0xca5b,0xdd19 ,
- 0xcc98,0x2620 ,0x1413,0x3cc5 ,0xcbad,0xdb26 ,
- 0xcbad,0x24da ,0x1354,0x3d03 ,0xcd11,0xd93f ,
- 0xcac9,0x238e ,0x1294,0x3d3f ,0xce87,0xd766 ,
- 0xc9ee,0x223d ,0x11d3,0x3d78 ,0xd00e,0xd59b ,
- 0xc91b,0x20e7 ,0x1112,0x3daf ,0xd1a6,0xd3df ,
- 0xc850,0x1f8c ,0x1050,0x3de3 ,0xd34e,0xd231 ,
- 0xc78f,0x1e2b ,0x0f8d,0x3e15 ,0xd505,0xd094 ,
- 0xc6d5,0x1cc6 ,0x0eca,0x3e45 ,0xd6cb,0xcf07 ,
- 0xc625,0x1b5d ,0x0e06,0x3e72 ,0xd8a0,0xcd8c ,
- 0xc57e,0x19ef ,0x0d41,0x3e9d ,0xda82,0xcc21 ,
- 0xc4df,0x187e ,0x0c7c,0x3ec5 ,0xdc72,0xcac9 ,
- 0xc44a,0x1709 ,0x0bb7,0x3eeb ,0xde6d,0xc983 ,
- 0xc3be,0x1590 ,0x0af1,0x3f0f ,0xe074,0xc850 ,
- 0xc33b,0x1413 ,0x0a2b,0x3f30 ,0xe287,0xc731 ,
- 0xc2c1,0x1294 ,0x0964,0x3f4f ,0xe4a3,0xc625 ,
- 0xc251,0x1112 ,0x089d,0x3f6b ,0xe6c9,0xc52d ,
- 0xc1eb,0x0f8d ,0x07d6,0x3f85 ,0xe8f7,0xc44a ,
- 0xc18e,0x0e06 ,0x070e,0x3f9c ,0xeb2e,0xc37b ,
- 0xc13b,0x0c7c ,0x0646,0x3fb1 ,0xed6c,0xc2c1 ,
- 0xc0f1,0x0af1 ,0x057e,0x3fc4 ,0xefb0,0xc21d ,
- 0xc0b1,0x0964 ,0x04b5,0x3fd4 ,0xf1fa,0xc18e ,
- 0xc07b,0x07d6 ,0x03ed,0x3fe1 ,0xf449,0xc115 ,
- 0xc04f,0x0646 ,0x0324,0x3fec ,0xf69c,0xc0b1 ,
- 0xc02c,0x04b5 ,0x025b,0x3ff5 ,0xf8f2,0xc064 ,
- 0xc014,0x0324 ,0x0192,0x3ffb ,0xfb4b,0xc02c ,
- 0xc005,0x0192 ,0x00c9,0x3fff ,0xfda5,0xc00b ,
- 0x4000,0x0000 ,0x4000,0x0065 ,0x3fff,0x00c9 ,
- 0x3ffd,0x012e ,0x3ffb,0x0192 ,0x3ff8,0x01f7 ,
- 0x3ff5,0x025b ,0x3ff1,0x02c0 ,0x3fec,0x0324 ,
- 0x3fe7,0x0388 ,0x3fe1,0x03ed ,0x3fdb,0x0451 ,
- 0x3fd4,0x04b5 ,0x3fcc,0x051a ,0x3fc4,0x057e ,
- 0x3fbb,0x05e2 ,0x3fb1,0x0646 ,0x3fa7,0x06aa ,
- 0x3f9c,0x070e ,0x3f91,0x0772 ,0x3f85,0x07d6 ,
- 0x3f78,0x0839 ,0x3f6b,0x089d ,0x3f5d,0x0901 ,
- 0x3f4f,0x0964 ,0x3f40,0x09c7 ,0x3f30,0x0a2b ,
- 0x3f20,0x0a8e ,0x3f0f,0x0af1 ,0x3efd,0x0b54 ,
- 0x3eeb,0x0bb7 ,0x3ed8,0x0c1a ,0x3ec5,0x0c7c ,
- 0x3eb1,0x0cdf ,0x3e9d,0x0d41 ,0x3e88,0x0da4 ,
- 0x3e72,0x0e06 ,0x3e5c,0x0e68 ,0x3e45,0x0eca ,
- 0x3e2d,0x0f2b ,0x3e15,0x0f8d ,0x3dfc,0x0fee ,
- 0x3de3,0x1050 ,0x3dc9,0x10b1 ,0x3daf,0x1112 ,
- 0x3d93,0x1173 ,0x3d78,0x11d3 ,0x3d5b,0x1234 ,
- 0x3d3f,0x1294 ,0x3d21,0x12f4 ,0x3d03,0x1354 ,
- 0x3ce4,0x13b4 ,0x3cc5,0x1413 ,0x3ca5,0x1473 ,
- 0x3c85,0x14d2 ,0x3c64,0x1531 ,0x3c42,0x1590 ,
- 0x3c20,0x15ee ,0x3bfd,0x164c ,0x3bda,0x16ab ,
- 0x3bb6,0x1709 ,0x3b92,0x1766 ,0x3b6d,0x17c4 ,
- 0x3b47,0x1821 ,0x3b21,0x187e ,0x3afa,0x18db ,
- 0x3ad3,0x1937 ,0x3aab,0x1993 ,0x3a82,0x19ef ,
- 0x3a59,0x1a4b ,0x3a30,0x1aa7 ,0x3a06,0x1b02 ,
- 0x39db,0x1b5d ,0x39b0,0x1bb8 ,0x3984,0x1c12 ,
- 0x3958,0x1c6c ,0x392b,0x1cc6 ,0x38fd,0x1d20 ,
- 0x38cf,0x1d79 ,0x38a1,0x1dd3 ,0x3871,0x1e2b ,
- 0x3842,0x1e84 ,0x3812,0x1edc ,0x37e1,0x1f34 ,
- 0x37b0,0x1f8c ,0x377e,0x1fe3 ,0x374b,0x203a ,
- 0x3718,0x2091 ,0x36e5,0x20e7 ,0x36b1,0x213d ,
- 0x367d,0x2193 ,0x3648,0x21e8 ,0x3612,0x223d ,
- 0x35dc,0x2292 ,0x35a5,0x22e7 ,0x356e,0x233b ,
- 0x3537,0x238e ,0x34ff,0x23e2 ,0x34c6,0x2435 ,
- 0x348d,0x2488 ,0x3453,0x24da ,0x3419,0x252c ,
- 0x33df,0x257e ,0x33a3,0x25cf ,0x3368,0x2620 ,
- 0x332c,0x2671 ,0x32ef,0x26c1 ,0x32b2,0x2711 ,
- 0x3274,0x2760 ,0x3236,0x27af ,0x31f8,0x27fe ,
- 0x31b9,0x284c ,0x3179,0x289a ,0x3139,0x28e7 ,
- 0x30f9,0x2935 ,0x30b8,0x2981 ,0x3076,0x29ce ,
- 0x3034,0x2a1a ,0x2ff2,0x2a65 ,0x2faf,0x2ab0 ,
- 0x2f6c,0x2afb ,0x2f28,0x2b45 ,0x2ee4,0x2b8f ,
- 0x2e9f,0x2bd8 ,0x2e5a,0x2c21 ,0x2e15,0x2c6a ,
- 0x2dcf,0x2cb2 ,0x2d88,0x2cfa ,0x2d41,0x2d41 ,
- 0x2cfa,0x2d88 ,0x2cb2,0x2dcf ,0x2c6a,0x2e15 ,
- 0x2c21,0x2e5a ,0x2bd8,0x2e9f ,0x2b8f,0x2ee4 ,
- 0x2b45,0x2f28 ,0x2afb,0x2f6c ,0x2ab0,0x2faf ,
- 0x2a65,0x2ff2 ,0x2a1a,0x3034 ,0x29ce,0x3076 ,
- 0x2981,0x30b8 ,0x2935,0x30f9 ,0x28e7,0x3139 ,
- 0x289a,0x3179 ,0x284c,0x31b9 ,0x27fe,0x31f8 ,
- 0x27af,0x3236 ,0x2760,0x3274 ,0x2711,0x32b2 ,
- 0x26c1,0x32ef ,0x2671,0x332c ,0x2620,0x3368 ,
- 0x25cf,0x33a3 ,0x257e,0x33df ,0x252c,0x3419 ,
- 0x24da,0x3453 ,0x2488,0x348d ,0x2435,0x34c6 ,
- 0x23e2,0x34ff ,0x238e,0x3537 ,0x233b,0x356e ,
- 0x22e7,0x35a5 ,0x2292,0x35dc ,0x223d,0x3612 ,
- 0x21e8,0x3648 ,0x2193,0x367d ,0x213d,0x36b1 ,
- 0x20e7,0x36e5 ,0x2091,0x3718 ,0x203a,0x374b ,
- 0x1fe3,0x377e ,0x1f8c,0x37b0 ,0x1f34,0x37e1 ,
- 0x1edc,0x3812 ,0x1e84,0x3842 ,0x1e2b,0x3871 ,
- 0x1dd3,0x38a1 ,0x1d79,0x38cf ,0x1d20,0x38fd ,
- 0x1cc6,0x392b ,0x1c6c,0x3958 ,0x1c12,0x3984 ,
- 0x1bb8,0x39b0 ,0x1b5d,0x39db ,0x1b02,0x3a06 ,
- 0x1aa7,0x3a30 ,0x1a4b,0x3a59 ,0x19ef,0x3a82 ,
- 0x1993,0x3aab ,0x1937,0x3ad3 ,0x18db,0x3afa ,
- 0x187e,0x3b21 ,0x1821,0x3b47 ,0x17c4,0x3b6d ,
- 0x1766,0x3b92 ,0x1709,0x3bb6 ,0x16ab,0x3bda ,
- 0x164c,0x3bfd ,0x15ee,0x3c20 ,0x1590,0x3c42 ,
- 0x1531,0x3c64 ,0x14d2,0x3c85 ,0x1473,0x3ca5 ,
- 0x1413,0x3cc5 ,0x13b4,0x3ce4 ,0x1354,0x3d03 ,
- 0x12f4,0x3d21 ,0x1294,0x3d3f ,0x1234,0x3d5b ,
- 0x11d3,0x3d78 ,0x1173,0x3d93 ,0x1112,0x3daf ,
- 0x10b1,0x3dc9 ,0x1050,0x3de3 ,0x0fee,0x3dfc ,
- 0x0f8d,0x3e15 ,0x0f2b,0x3e2d ,0x0eca,0x3e45 ,
- 0x0e68,0x3e5c ,0x0e06,0x3e72 ,0x0da4,0x3e88 ,
- 0x0d41,0x3e9d ,0x0cdf,0x3eb1 ,0x0c7c,0x3ec5 ,
- 0x0c1a,0x3ed8 ,0x0bb7,0x3eeb ,0x0b54,0x3efd ,
- 0x0af1,0x3f0f ,0x0a8e,0x3f20 ,0x0a2b,0x3f30 ,
- 0x09c7,0x3f40 ,0x0964,0x3f4f ,0x0901,0x3f5d ,
- 0x089d,0x3f6b ,0x0839,0x3f78 ,0x07d6,0x3f85 ,
- 0x0772,0x3f91 ,0x070e,0x3f9c ,0x06aa,0x3fa7 ,
- 0x0646,0x3fb1 ,0x05e2,0x3fbb ,0x057e,0x3fc4 ,
- 0x051a,0x3fcc ,0x04b5,0x3fd4 ,0x0451,0x3fdb ,
- 0x03ed,0x3fe1 ,0x0388,0x3fe7 ,0x0324,0x3fec ,
- 0x02c0,0x3ff1 ,0x025b,0x3ff5 ,0x01f7,0x3ff8 ,
- 0x0192,0x3ffb ,0x012e,0x3ffd ,0x00c9,0x3fff ,
- 0x0065,0x4000 ,0x0000,0x4000 ,0xff9b,0x4000 ,
- 0xff37,0x3fff ,0xfed2,0x3ffd ,0xfe6e,0x3ffb ,
- 0xfe09,0x3ff8 ,0xfda5,0x3ff5 ,0xfd40,0x3ff1 ,
- 0xfcdc,0x3fec ,0xfc78,0x3fe7 ,0xfc13,0x3fe1 ,
- 0xfbaf,0x3fdb ,0xfb4b,0x3fd4 ,0xfae6,0x3fcc ,
- 0xfa82,0x3fc4 ,0xfa1e,0x3fbb ,0xf9ba,0x3fb1 ,
- 0xf956,0x3fa7 ,0xf8f2,0x3f9c ,0xf88e,0x3f91 ,
- 0xf82a,0x3f85 ,0xf7c7,0x3f78 ,0xf763,0x3f6b ,
- 0xf6ff,0x3f5d ,0xf69c,0x3f4f ,0xf639,0x3f40 ,
- 0xf5d5,0x3f30 ,0xf572,0x3f20 ,0xf50f,0x3f0f ,
- 0xf4ac,0x3efd ,0xf449,0x3eeb ,0xf3e6,0x3ed8 ,
- 0xf384,0x3ec5 ,0xf321,0x3eb1 ,0xf2bf,0x3e9d ,
- 0xf25c,0x3e88 ,0xf1fa,0x3e72 ,0xf198,0x3e5c ,
- 0xf136,0x3e45 ,0xf0d5,0x3e2d ,0xf073,0x3e15 ,
- 0xf012,0x3dfc ,0xefb0,0x3de3 ,0xef4f,0x3dc9 ,
- 0xeeee,0x3daf ,0xee8d,0x3d93 ,0xee2d,0x3d78 ,
- 0xedcc,0x3d5b ,0xed6c,0x3d3f ,0xed0c,0x3d21 ,
- 0xecac,0x3d03 ,0xec4c,0x3ce4 ,0xebed,0x3cc5 ,
- 0xeb8d,0x3ca5 ,0xeb2e,0x3c85 ,0xeacf,0x3c64 ,
- 0xea70,0x3c42 ,0xea12,0x3c20 ,0xe9b4,0x3bfd ,
- 0xe955,0x3bda ,0xe8f7,0x3bb6 ,0xe89a,0x3b92 ,
- 0xe83c,0x3b6d ,0xe7df,0x3b47 ,0xe782,0x3b21 ,
- 0xe725,0x3afa ,0xe6c9,0x3ad3 ,0xe66d,0x3aab ,
- 0xe611,0x3a82 ,0xe5b5,0x3a59 ,0xe559,0x3a30 ,
- 0xe4fe,0x3a06 ,0xe4a3,0x39db ,0xe448,0x39b0 ,
- 0xe3ee,0x3984 ,0xe394,0x3958 ,0xe33a,0x392b ,
- 0xe2e0,0x38fd ,0xe287,0x38cf ,0xe22d,0x38a1 ,
- 0xe1d5,0x3871 ,0xe17c,0x3842 ,0xe124,0x3812 ,
- 0xe0cc,0x37e1 ,0xe074,0x37b0 ,0xe01d,0x377e ,
- 0xdfc6,0x374b ,0xdf6f,0x3718 ,0xdf19,0x36e5 ,
- 0xdec3,0x36b1 ,0xde6d,0x367d ,0xde18,0x3648 ,
- 0xddc3,0x3612 ,0xdd6e,0x35dc ,0xdd19,0x35a5 ,
- 0xdcc5,0x356e ,0xdc72,0x3537 ,0xdc1e,0x34ff ,
- 0xdbcb,0x34c6 ,0xdb78,0x348d ,0xdb26,0x3453 ,
- 0xdad4,0x3419 ,0xda82,0x33df ,0xda31,0x33a3 ,
- 0xd9e0,0x3368 ,0xd98f,0x332c ,0xd93f,0x32ef ,
- 0xd8ef,0x32b2 ,0xd8a0,0x3274 ,0xd851,0x3236 ,
- 0xd802,0x31f8 ,0xd7b4,0x31b9 ,0xd766,0x3179 ,
- 0xd719,0x3139 ,0xd6cb,0x30f9 ,0xd67f,0x30b8 ,
- 0xd632,0x3076 ,0xd5e6,0x3034 ,0xd59b,0x2ff2 ,
- 0xd550,0x2faf ,0xd505,0x2f6c ,0xd4bb,0x2f28 ,
- 0xd471,0x2ee4 ,0xd428,0x2e9f ,0xd3df,0x2e5a ,
- 0xd396,0x2e15 ,0xd34e,0x2dcf ,0xd306,0x2d88 ,
- 0xd2bf,0x2d41 ,0xd278,0x2cfa ,0xd231,0x2cb2 ,
- 0xd1eb,0x2c6a ,0xd1a6,0x2c21 ,0xd161,0x2bd8 ,
- 0xd11c,0x2b8f ,0xd0d8,0x2b45 ,0xd094,0x2afb ,
- 0xd051,0x2ab0 ,0xd00e,0x2a65 ,0xcfcc,0x2a1a ,
- 0xcf8a,0x29ce ,0xcf48,0x2981 ,0xcf07,0x2935 ,
- 0xcec7,0x28e7 ,0xce87,0x289a ,0xce47,0x284c ,
- 0xce08,0x27fe ,0xcdca,0x27af ,0xcd8c,0x2760 ,
- 0xcd4e,0x2711 ,0xcd11,0x26c1 ,0xccd4,0x2671 ,
- 0xcc98,0x2620 ,0xcc5d,0x25cf ,0xcc21,0x257e ,
- 0xcbe7,0x252c ,0xcbad,0x24da ,0xcb73,0x2488 ,
- 0xcb3a,0x2435 ,0xcb01,0x23e2 ,0xcac9,0x238e ,
- 0xca92,0x233b ,0xca5b,0x22e7 ,0xca24,0x2292 ,
- 0xc9ee,0x223d ,0xc9b8,0x21e8 ,0xc983,0x2193 ,
- 0xc94f,0x213d ,0xc91b,0x20e7 ,0xc8e8,0x2091 ,
- 0xc8b5,0x203a ,0xc882,0x1fe3 ,0xc850,0x1f8c ,
- 0xc81f,0x1f34 ,0xc7ee,0x1edc ,0xc7be,0x1e84 ,
- 0xc78f,0x1e2b ,0xc75f,0x1dd3 ,0xc731,0x1d79 ,
- 0xc703,0x1d20 ,0xc6d5,0x1cc6 ,0xc6a8,0x1c6c ,
- 0xc67c,0x1c12 ,0xc650,0x1bb8 ,0xc625,0x1b5d ,
- 0xc5fa,0x1b02 ,0xc5d0,0x1aa7 ,0xc5a7,0x1a4b ,
- 0xc57e,0x19ef ,0xc555,0x1993 ,0xc52d,0x1937 ,
- 0xc506,0x18db ,0xc4df,0x187e ,0xc4b9,0x1821 ,
- 0xc493,0x17c4 ,0xc46e,0x1766 ,0xc44a,0x1709 ,
- 0xc426,0x16ab ,0xc403,0x164c ,0xc3e0,0x15ee ,
- 0xc3be,0x1590 ,0xc39c,0x1531 ,0xc37b,0x14d2 ,
- 0xc35b,0x1473 ,0xc33b,0x1413 ,0xc31c,0x13b4 ,
- 0xc2fd,0x1354 ,0xc2df,0x12f4 ,0xc2c1,0x1294 ,
- 0xc2a5,0x1234 ,0xc288,0x11d3 ,0xc26d,0x1173 ,
- 0xc251,0x1112 ,0xc237,0x10b1 ,0xc21d,0x1050 ,
- 0xc204,0x0fee ,0xc1eb,0x0f8d ,0xc1d3,0x0f2b ,
- 0xc1bb,0x0eca ,0xc1a4,0x0e68 ,0xc18e,0x0e06 ,
- 0xc178,0x0da4 ,0xc163,0x0d41 ,0xc14f,0x0cdf ,
- 0xc13b,0x0c7c ,0xc128,0x0c1a ,0xc115,0x0bb7 ,
- 0xc103,0x0b54 ,0xc0f1,0x0af1 ,0xc0e0,0x0a8e ,
- 0xc0d0,0x0a2b ,0xc0c0,0x09c7 ,0xc0b1,0x0964 ,
- 0xc0a3,0x0901 ,0xc095,0x089d ,0xc088,0x0839 ,
- 0xc07b,0x07d6 ,0xc06f,0x0772 ,0xc064,0x070e ,
- 0xc059,0x06aa ,0xc04f,0x0646 ,0xc045,0x05e2 ,
- 0xc03c,0x057e ,0xc034,0x051a ,0xc02c,0x04b5 ,
- 0xc025,0x0451 ,0xc01f,0x03ed ,0xc019,0x0388 ,
- 0xc014,0x0324 ,0xc00f,0x02c0 ,0xc00b,0x025b ,
- 0xc008,0x01f7 ,0xc005,0x0192 ,0xc003,0x012e ,
- 0xc001,0x00c9 ,0xc000,0x0065 };
diff --git a/trunk/src/common_audio/signal_processing/webrtc_fft_t_rad.c b/trunk/src/common_audio/signal_processing/webrtc_fft_t_rad.c
deleted file mode 100644
index 13fbd9f..0000000
--- a/trunk/src/common_audio/signal_processing/webrtc_fft_t_rad.c
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file contains the Q14 radix-2 tables used in ARM9E optimization routines.
- *
- */
-
-extern const unsigned short t_Q14S_rad8[2];
-const unsigned short t_Q14S_rad8[2] = { 0x0000,0x2d41 };
-
-//extern const int t_Q30S_rad8[2];
-//const int t_Q30S_rad8[2] = { 0x00000000,0x2d413ccd };
-
-extern const unsigned short t_Q14R_rad8[2];
-const unsigned short t_Q14R_rad8[2] = { 0x2d41,0x2d41 };
-
-//extern const int t_Q30R_rad8[2];
-//const int t_Q30R_rad8[2] = { 0x2d413ccd,0x2d413ccd };
diff --git a/trunk/src/common_audio/vad/Android.mk b/trunk/src/common_audio/vad/Android.mk
deleted file mode 100644
index b7be3f0..0000000
--- a/trunk/src/common_audio/vad/Android.mk
+++ /dev/null
@@ -1,50 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_vad
-LOCAL_MODULE_TAGS := optional
-LOCAL_SRC_FILES := \
- webrtc_vad.c \
- vad_core.c \
- vad_filterbank.c \
- vad_gmm.c \
- vad_sp.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include \
- $(LOCAL_PATH)/../.. \
- $(LOCAL_PATH)/../signal_processing/include
-
-LOCAL_SHARED_LIBRARIES := \
- libdl \
- libstlport
-
-ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
-LOCAL_LDLIBS += -ldl -lpthread
-endif
-
-ifneq ($(TARGET_SIMULATOR),true)
-LOCAL_SHARED_LIBRARIES += libdl
-endif
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/common_audio/vad/include/webrtc_vad.h b/trunk/src/common_audio/vad/include/webrtc_vad.h
deleted file mode 100644
index 3f9e402..0000000
--- a/trunk/src/common_audio/vad/include/webrtc_vad.h
+++ /dev/null
@@ -1,130 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the VAD API calls. Specific function calls are given below.
- */
-
-#ifndef WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
-#define WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
-
-#include "typedefs.h"
-
-typedef struct WebRtcVadInst VadInst;
-
-#ifdef __cplusplus
-extern "C"
-{
-#endif
-
-/****************************************************************************
- * WebRtcVad_AssignSize(...)
- *
- * This functions get the size needed for storing the instance for encoder
- * and decoder, respectively
- *
- * Input/Output:
- * - size_in_bytes : Pointer to integer where the size is returned
- *
- * Return value : 0
- */
-WebRtc_Word16 WebRtcVad_AssignSize(int *size_in_bytes);
-
-/****************************************************************************
- * WebRtcVad_Assign(...)
- *
- * This functions Assigns memory for the instances.
- *
- * Input:
- * - vad_inst_addr : Address to where to assign memory
- * Output:
- * - vad_inst : Pointer to the instance that should be created
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-WebRtc_Word16 WebRtcVad_Assign(VadInst **vad_inst, void *vad_inst_addr);
-
-// Creates an instance to the VAD structure.
-//
-// - handle [o] : Pointer to the VAD instance that should be created.
-//
-// returns : 0 - (OK), -1 - (Error)
-int WebRtcVad_Create(VadInst** handle);
-
-// Frees the dynamic memory of a specified VAD instance.
-//
-// - handle [i] : Pointer to VAD instance that should be freed.
-//
-// returns : 0 - (OK), -1 - (NULL pointer in)
-int WebRtcVad_Free(VadInst* handle);
-
-/****************************************************************************
- * WebRtcVad_Init(...)
- *
- * This function initializes a VAD instance
- *
- * Input:
- * - vad_inst : Instance that should be initialized
- *
- * Output:
- * - vad_inst : Initialized instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcVad_Init(VadInst *vad_inst);
-
-/****************************************************************************
- * WebRtcVad_set_mode(...)
- *
- * This function initializes a VAD instance
- *
- * Input:
- * - vad_inst : VAD instance
- * - mode : Aggressiveness setting (0, 1, 2, or 3)
- *
- * Output:
- * - vad_inst : Initialized instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcVad_set_mode(VadInst *vad_inst, int mode);
-
-/****************************************************************************
- * WebRtcVad_Process(...)
- *
- * This functions does a VAD for the inserted speech frame
- *
- * Input
- * - vad_inst : VAD Instance. Needs to be initiated before call.
- * - fs : sampling frequency (Hz): 8000, 16000, or 32000
- * - speech_frame : Pointer to speech frame buffer
- * - frame_length : Length of speech frame buffer in number of samples
- *
- * Output:
- * - vad_inst : Updated VAD instance
- *
- * Return value : 1 - Active Voice
- * 0 - Non-active Voice
- * -1 - Error
- */
-WebRtc_Word16 WebRtcVad_Process(VadInst *vad_inst,
- WebRtc_Word16 fs,
- WebRtc_Word16 *speech_frame,
- WebRtc_Word16 frame_length);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif // WEBRTC_COMMON_AUDIO_VAD_INCLUDE_WEBRTC_VAD_H_
diff --git a/trunk/src/common_audio/vad/vad.gypi b/trunk/src/common_audio/vad/vad.gypi
deleted file mode 100644
index f15e651..0000000
--- a/trunk/src/common_audio/vad/vad.gypi
+++ /dev/null
@@ -1,69 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'vad',
- 'type': '<(library)',
- 'dependencies': [
- 'signal_processing',
- ],
- 'include_dirs': [
- 'include',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- ],
- },
- 'sources': [
- 'include/webrtc_vad.h',
- 'webrtc_vad.c',
- 'vad_core.c',
- 'vad_core.h',
- 'vad_defines.h',
- 'vad_filterbank.c',
- 'vad_filterbank.h',
- 'vad_gmm.c',
- 'vad_gmm.h',
- 'vad_sp.c',
- 'vad_sp.h',
- ],
- },
- ], # targets
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets' : [
- {
- 'target_name': 'vad_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'vad',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'sources': [
- 'vad_core_unittest.cc',
- 'vad_filterbank_unittest.cc',
- 'vad_gmm_unittest.cc',
- 'vad_sp_unittest.cc',
- 'vad_unittest.cc',
- 'vad_unittest.h',
- ],
- }, # vad_unittests
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/common_audio/vad/vad_core.c b/trunk/src/common_audio/vad/vad_core.c
deleted file mode 100644
index 2c6e6a9..0000000
--- a/trunk/src/common_audio/vad/vad_core.c
+++ /dev/null
@@ -1,663 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vad_core.h"
-
-#include "signal_processing_library.h"
-#include "typedefs.h"
-#include "vad_defines.h"
-#include "vad_filterbank.h"
-#include "vad_gmm.h"
-#include "vad_sp.h"
-
-// Spectrum Weighting
-static const WebRtc_Word16 kSpectrumWeight[6] = { 6, 8, 10, 12, 14, 16 };
-static const WebRtc_Word16 kNoiseUpdateConst = 655; // Q15
-static const WebRtc_Word16 kSpeechUpdateConst = 6554; // Q15
-static const WebRtc_Word16 kBackEta = 154; // Q8
-// Minimum difference between the two models, Q5
-static const WebRtc_Word16 kMinimumDifference[6] = {
- 544, 544, 576, 576, 576, 576 };
-// Upper limit of mean value for speech model, Q7
-static const WebRtc_Word16 kMaximumSpeech[6] = {
- 11392, 11392, 11520, 11520, 11520, 11520 };
-// Minimum value for mean value
-static const WebRtc_Word16 kMinimumMean[2] = { 640, 768 };
-// Upper limit of mean value for noise model, Q7
-static const WebRtc_Word16 kMaximumNoise[6] = {
- 9216, 9088, 8960, 8832, 8704, 8576 };
-// Start values for the Gaussian models, Q7
-// Weights for the two Gaussians for the six channels (noise)
-static const WebRtc_Word16 kNoiseDataWeights[12] = {
- 34, 62, 72, 66, 53, 25, 94, 66, 56, 62, 75, 103 };
-// Weights for the two Gaussians for the six channels (speech)
-static const WebRtc_Word16 kSpeechDataWeights[12] = {
- 48, 82, 45, 87, 50, 47, 80, 46, 83, 41, 78, 81 };
-// Means for the two Gaussians for the six channels (noise)
-static const WebRtc_Word16 kNoiseDataMeans[12] = {
- 6738, 4892, 7065, 6715, 6771, 3369, 7646, 3863, 7820, 7266, 5020, 4362 };
-// Means for the two Gaussians for the six channels (speech)
-static const WebRtc_Word16 kSpeechDataMeans[12] = {
- 8306, 10085, 10078, 11823, 11843, 6309, 9473, 9571, 10879, 7581, 8180, 7483
-};
-// Stds for the two Gaussians for the six channels (noise)
-static const WebRtc_Word16 kNoiseDataStds[12] = {
- 378, 1064, 493, 582, 688, 593, 474, 697, 475, 688, 421, 455 };
-// Stds for the two Gaussians for the six channels (speech)
-static const WebRtc_Word16 kSpeechDataStds[12] = {
- 555, 505, 567, 524, 585, 1231, 509, 828, 492, 1540, 1079, 850 };
-
-// Constants used in GmmProbability().
-//
-// Maximum number of counted speech (VAD = 1) frames in a row.
-static const int16_t kMaxSpeechFrames = 6;
-// Minimum standard deviation for both speech and noise.
-static const int16_t kMinStd = 384;
-
-// Constants in WebRtcVad_InitCore().
-// Default aggressiveness mode.
-static const short kDefaultMode = 0;
-static const int kInitCheck = 42;
-
-// Calculates the probabilities for both speech and background noise using
-// Gaussian Mixture Models. A hypothesis-test is performed to decide which type
-// of signal is most probable.
-//
-// - inst [i/o] : Pointer to VAD instance
-// - feature_vector [i] : Feature vector = log10(energy in frequency band)
-// - total_power [i] : Total power in audio frame.
-// - frame_length [i] : Number of input samples
-//
-// - returns : the VAD decision (0 - noise, 1 - speech).
-static int16_t GmmProbability(VadInstT *inst, WebRtc_Word16 *feature_vector,
- WebRtc_Word16 total_power, int frame_length)
-{
- int n, k;
- WebRtc_Word16 backval;
- WebRtc_Word16 h0, h1;
- WebRtc_Word16 ratvec, xval;
- WebRtc_Word16 vadflag;
- WebRtc_Word16 shifts0, shifts1;
- WebRtc_Word16 tmp16, tmp16_1, tmp16_2;
- WebRtc_Word16 diff, nr, pos;
- WebRtc_Word16 nmk, nmk2, nmk3, smk, smk2, nsk, ssk;
- WebRtc_Word16 delt, ndelt;
- WebRtc_Word16 maxspe, maxmu;
- WebRtc_Word16 deltaN[NUM_TABLE_VALUES], deltaS[NUM_TABLE_VALUES];
- WebRtc_Word16 ngprvec[NUM_TABLE_VALUES], sgprvec[NUM_TABLE_VALUES];
- WebRtc_Word32 h0test, h1test;
- WebRtc_Word32 tmp32_1, tmp32_2;
- WebRtc_Word32 dotVal;
- WebRtc_Word32 nmid, smid;
- WebRtc_Word32 probn[NUM_MODELS], probs[NUM_MODELS];
- WebRtc_Word16 *nmean1ptr, *nmean2ptr, *smean1ptr, *smean2ptr, *nstd1ptr, *nstd2ptr,
- *sstd1ptr, *sstd2ptr;
- WebRtc_Word16 overhead1, overhead2, individualTest, totalTest;
-
- // Set the thresholds to different values based on frame length
- if (frame_length == 80)
- {
- // 80 input samples
- overhead1 = inst->over_hang_max_1[0];
- overhead2 = inst->over_hang_max_2[0];
- individualTest = inst->individual[0];
- totalTest = inst->total[0];
- } else if (frame_length == 160)
- {
- // 160 input samples
- overhead1 = inst->over_hang_max_1[1];
- overhead2 = inst->over_hang_max_2[1];
- individualTest = inst->individual[1];
- totalTest = inst->total[1];
- } else
- {
- // 240 input samples
- overhead1 = inst->over_hang_max_1[2];
- overhead2 = inst->over_hang_max_2[2];
- individualTest = inst->individual[2];
- totalTest = inst->total[2];
- }
-
- if (total_power > MIN_ENERGY)
- { // If signal present at all
-
- // Set pointers to the gaussian parameters
- nmean1ptr = &inst->noise_means[0];
- nmean2ptr = &inst->noise_means[NUM_CHANNELS];
- smean1ptr = &inst->speech_means[0];
- smean2ptr = &inst->speech_means[NUM_CHANNELS];
- nstd1ptr = &inst->noise_stds[0];
- nstd2ptr = &inst->noise_stds[NUM_CHANNELS];
- sstd1ptr = &inst->speech_stds[0];
- sstd2ptr = &inst->speech_stds[NUM_CHANNELS];
-
- vadflag = 0;
- dotVal = 0;
- for (n = 0; n < NUM_CHANNELS; n++)
- { // For all channels
-
- pos = WEBRTC_SPL_LSHIFT_W16(n, 1);
- xval = feature_vector[n];
-
- // Probability for Noise, Q7 * Q20 = Q27
- tmp32_1 = WebRtcVad_GaussianProbability(xval, *nmean1ptr++, *nstd1ptr++,
- &deltaN[pos]);
- probn[0] = (WebRtc_Word32)(kNoiseDataWeights[n] * tmp32_1);
- tmp32_1 = WebRtcVad_GaussianProbability(xval, *nmean2ptr++, *nstd2ptr++,
- &deltaN[pos + 1]);
- probn[1] = (WebRtc_Word32)(kNoiseDataWeights[n + NUM_CHANNELS] * tmp32_1);
- h0test = probn[0] + probn[1]; // Q27
- h0 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(h0test, 12); // Q15
-
- // Probability for Speech
- tmp32_1 = WebRtcVad_GaussianProbability(xval, *smean1ptr++, *sstd1ptr++,
- &deltaS[pos]);
- probs[0] = (WebRtc_Word32)(kSpeechDataWeights[n] * tmp32_1);
- tmp32_1 = WebRtcVad_GaussianProbability(xval, *smean2ptr++, *sstd2ptr++,
- &deltaS[pos + 1]);
- probs[1] = (WebRtc_Word32)(kSpeechDataWeights[n + NUM_CHANNELS] * tmp32_1);
- h1test = probs[0] + probs[1]; // Q27
- h1 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(h1test, 12); // Q15
-
- // Get likelihood ratio. Approximate log2(H1/H0) with shifts0 - shifts1
- shifts0 = WebRtcSpl_NormW32(h0test);
- shifts1 = WebRtcSpl_NormW32(h1test);
-
- if ((h0test > 0) && (h1test > 0))
- {
- ratvec = shifts0 - shifts1;
- } else if (h1test > 0)
- {
- ratvec = 31 - shifts1;
- } else if (h0test > 0)
- {
- ratvec = shifts0 - 31;
- } else
- {
- ratvec = 0;
- }
-
- // VAD decision with spectrum weighting
- dotVal += WEBRTC_SPL_MUL_16_16(ratvec, kSpectrumWeight[n]);
-
- // Individual channel test
- if ((ratvec << 2) > individualTest)
- {
- vadflag = 1;
- }
-
- // Probabilities used when updating model
- if (h0 > 0)
- {
- tmp32_1 = probn[0] & 0xFFFFF000; // Q27
- tmp32_2 = WEBRTC_SPL_LSHIFT_W32(tmp32_1, 2); // Q29
- ngprvec[pos] = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_2, h0);
- ngprvec[pos + 1] = 16384 - ngprvec[pos];
- } else
- {
- ngprvec[pos] = 16384;
- ngprvec[pos + 1] = 0;
- }
-
- // Probabilities used when updating model
- if (h1 > 0)
- {
- tmp32_1 = probs[0] & 0xFFFFF000;
- tmp32_2 = WEBRTC_SPL_LSHIFT_W32(tmp32_1, 2);
- sgprvec[pos] = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_2, h1);
- sgprvec[pos + 1] = 16384 - sgprvec[pos];
- } else
- {
- sgprvec[pos] = 0;
- sgprvec[pos + 1] = 0;
- }
- }
-
- // Overall test
- if (dotVal >= totalTest)
- {
- vadflag |= 1;
- }
-
- // Set pointers to the means and standard deviations.
- nmean1ptr = &inst->noise_means[0];
- smean1ptr = &inst->speech_means[0];
- nstd1ptr = &inst->noise_stds[0];
- sstd1ptr = &inst->speech_stds[0];
-
- maxspe = 12800;
-
- // Update the model's parameters
- for (n = 0; n < NUM_CHANNELS; n++)
- {
-
- pos = WEBRTC_SPL_LSHIFT_W16(n, 1);
-
- // Get min value in past which is used for long term correction
- backval = WebRtcVad_FindMinimum(inst, feature_vector[n], n); // Q4
-
- // Compute the "global" mean, that is the sum of the two means weighted
- nmid = WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n], *nmean1ptr); // Q7 * Q7
- nmid += WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n+NUM_CHANNELS],
- *(nmean1ptr+NUM_CHANNELS));
- tmp16_1 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(nmid, 6); // Q8
-
- for (k = 0; k < NUM_MODELS; k++)
- {
-
- nr = pos + k;
-
- nmean2ptr = nmean1ptr + k * NUM_CHANNELS;
- smean2ptr = smean1ptr + k * NUM_CHANNELS;
- nstd2ptr = nstd1ptr + k * NUM_CHANNELS;
- sstd2ptr = sstd1ptr + k * NUM_CHANNELS;
- nmk = *nmean2ptr;
- smk = *smean2ptr;
- nsk = *nstd2ptr;
- ssk = *sstd2ptr;
-
- // Update noise mean vector if the frame consists of noise only
- nmk2 = nmk;
- if (!vadflag)
- {
- // deltaN = (x-mu)/sigma^2
- // ngprvec[k] = probn[k]/(probn[0] + probn[1])
-
- delt = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ngprvec[nr],
- deltaN[nr], 11); // Q14*Q11
- nmk2 = nmk + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(delt,
- kNoiseUpdateConst,
- 22); // Q7+(Q14*Q15>>22)
- }
-
- // Long term correction of the noise mean
- ndelt = WEBRTC_SPL_LSHIFT_W16(backval, 4);
- ndelt -= tmp16_1; // Q8 - Q8
- nmk3 = nmk2 + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ndelt,
- kBackEta,
- 9); // Q7+(Q8*Q8)>>9
-
- // Control that the noise mean does not drift to much
- tmp16 = WEBRTC_SPL_LSHIFT_W16(k+5, 7);
- if (nmk3 < tmp16)
- nmk3 = tmp16;
- tmp16 = WEBRTC_SPL_LSHIFT_W16(72+k-n, 7);
- if (nmk3 > tmp16)
- nmk3 = tmp16;
- *nmean2ptr = nmk3;
-
- if (vadflag)
- {
- // Update speech mean vector:
- // deltaS = (x-mu)/sigma^2
- // sgprvec[k] = probn[k]/(probn[0] + probn[1])
-
- delt = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sgprvec[nr],
- deltaS[nr],
- 11); // (Q14*Q11)>>11=Q14
- tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(delt,
- kSpeechUpdateConst,
- 21) + 1;
- smk2 = smk + (tmp16 >> 1); // Q7 + (Q14 * Q15 >> 22)
-
- // Control that the speech mean does not drift to much
- maxmu = maxspe + 640;
- if (smk2 < kMinimumMean[k])
- smk2 = kMinimumMean[k];
- if (smk2 > maxmu)
- smk2 = maxmu;
-
- *smean2ptr = smk2;
-
- // (Q7>>3) = Q4
- tmp16 = WEBRTC_SPL_RSHIFT_W16((smk + 4), 3);
-
- tmp16 = feature_vector[n] - tmp16; // Q4
- tmp32_1 = WEBRTC_SPL_MUL_16_16_RSFT(deltaS[nr], tmp16, 3);
- tmp32_2 = tmp32_1 - (WebRtc_Word32)4096; // Q12
- tmp16 = WEBRTC_SPL_RSHIFT_W16((sgprvec[nr]), 2);
- tmp32_1 = (WebRtc_Word32)(tmp16 * tmp32_2);// (Q15>>3)*(Q14>>2)=Q12*Q12=Q24
-
- tmp32_2 = WEBRTC_SPL_RSHIFT_W32(tmp32_1, 4); // Q20
-
- // 0.1 * Q20 / Q7 = Q13
- if (tmp32_2 > 0)
- tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_2, ssk * 10);
- else
- {
- tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(-tmp32_2, ssk * 10);
- tmp16 = -tmp16;
- }
- // divide by 4 giving an update factor of 0.025
- tmp16 += 128; // Rounding
- ssk += WEBRTC_SPL_RSHIFT_W16(tmp16, 8);
- // Division with 8 plus Q7
- if (ssk < kMinStd)
- ssk = kMinStd;
- *sstd2ptr = ssk;
- } else
- {
- // Update GMM variance vectors
- // deltaN * (feature_vector[n] - nmk) - 1, Q11 * Q4
- tmp16 = feature_vector[n] - WEBRTC_SPL_RSHIFT_W16(nmk, 3);
-
- // (Q15>>3) * (Q14>>2) = Q12 * Q12 = Q24
- tmp32_1 = WEBRTC_SPL_MUL_16_16_RSFT(deltaN[nr], tmp16, 3) - 4096;
- tmp16 = WEBRTC_SPL_RSHIFT_W16((ngprvec[nr]+2), 2);
- tmp32_2 = (WebRtc_Word32)(tmp16 * tmp32_1);
- tmp32_1 = WEBRTC_SPL_RSHIFT_W32(tmp32_2, 14);
- // Q20 * approx 0.001 (2^-10=0.0009766)
-
- // Q20 / Q7 = Q13
- tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_1, nsk);
- if (tmp32_1 > 0)
- tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32_1, nsk);
- else
- {
- tmp16 = (WebRtc_Word16)WebRtcSpl_DivW32W16(-tmp32_1, nsk);
- tmp16 = -tmp16;
- }
- tmp16 += 32; // Rounding
- nsk += WEBRTC_SPL_RSHIFT_W16(tmp16, 6);
-
- if (nsk < kMinStd)
- nsk = kMinStd;
-
- *nstd2ptr = nsk;
- }
- }
-
- // Separate models if they are too close - nmid in Q14
- nmid = WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n], *nmean1ptr);
- nmid += WEBRTC_SPL_MUL_16_16(kNoiseDataWeights[n+NUM_CHANNELS], *nmean2ptr);
-
- // smid in Q14
- smid = WEBRTC_SPL_MUL_16_16(kSpeechDataWeights[n], *smean1ptr);
- smid += WEBRTC_SPL_MUL_16_16(kSpeechDataWeights[n+NUM_CHANNELS], *smean2ptr);
-
- // diff = "global" speech mean - "global" noise mean
- diff = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(smid, 9);
- tmp16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(nmid, 9);
- diff -= tmp16;
-
- if (diff < kMinimumDifference[n])
- {
-
- tmp16 = kMinimumDifference[n] - diff; // Q5
-
- // tmp16_1 = ~0.8 * (kMinimumDifference - diff) in Q7
- // tmp16_2 = ~0.2 * (kMinimumDifference - diff) in Q7
- tmp16_1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(13, tmp16, 2);
- tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(3, tmp16, 2);
-
- // First Gauss, speech model
- tmp16 = tmp16_1 + *smean1ptr;
- *smean1ptr = tmp16;
- smid = WEBRTC_SPL_MUL_16_16(tmp16, kSpeechDataWeights[n]);
-
- // Second Gauss, speech model
- tmp16 = tmp16_1 + *smean2ptr;
- *smean2ptr = tmp16;
- smid += WEBRTC_SPL_MUL_16_16(tmp16, kSpeechDataWeights[n+NUM_CHANNELS]);
-
- // First Gauss, noise model
- tmp16 = *nmean1ptr - tmp16_2;
- *nmean1ptr = tmp16;
-
- nmid = WEBRTC_SPL_MUL_16_16(tmp16, kNoiseDataWeights[n]);
-
- // Second Gauss, noise model
- tmp16 = *nmean2ptr - tmp16_2;
- *nmean2ptr = tmp16;
- nmid += WEBRTC_SPL_MUL_16_16(tmp16, kNoiseDataWeights[n+NUM_CHANNELS]);
- }
-
- // Control that the speech & noise means do not drift to much
- maxspe = kMaximumSpeech[n];
- tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(smid, 7);
- if (tmp16_2 > maxspe)
- { // Upper limit of speech model
- tmp16_2 -= maxspe;
-
- *smean1ptr -= tmp16_2;
- *smean2ptr -= tmp16_2;
- }
-
- tmp16_2 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(nmid, 7);
- if (tmp16_2 > kMaximumNoise[n])
- {
- tmp16_2 -= kMaximumNoise[n];
-
- *nmean1ptr -= tmp16_2;
- *nmean2ptr -= tmp16_2;
- }
-
- nmean1ptr++;
- smean1ptr++;
- nstd1ptr++;
- sstd1ptr++;
- }
- inst->frame_counter++;
- } else
- {
- vadflag = 0;
- }
-
- // Hangover smoothing
- if (!vadflag)
- {
- if (inst->over_hang > 0)
- {
- vadflag = 2 + inst->over_hang;
- inst->over_hang = inst->over_hang - 1;
- }
- inst->num_of_speech = 0;
- } else
- {
- inst->num_of_speech = inst->num_of_speech + 1;
- if (inst->num_of_speech > kMaxSpeechFrames)
- {
- inst->num_of_speech = kMaxSpeechFrames;
- inst->over_hang = overhead2;
- } else
- inst->over_hang = overhead1;
- }
- return vadflag;
-}
-
-// Initialize the VAD. Set aggressiveness mode to default value.
-int WebRtcVad_InitCore(VadInstT* self) {
- int i;
-
- if (self == NULL) {
- return -1;
- }
-
- // Initialization of general struct variables.
- self->vad = 1; // Speech active (=1).
- self->frame_counter = 0;
- self->over_hang = 0;
- self->num_of_speech = 0;
-
- // Initialization of downsampling filter state.
- memset(self->downsampling_filter_states, 0,
- sizeof(self->downsampling_filter_states));
-
- // Read initial PDF parameters.
- for (i = 0; i < NUM_TABLE_VALUES; i++) {
- self->noise_means[i] = kNoiseDataMeans[i];
- self->speech_means[i] = kSpeechDataMeans[i];
- self->noise_stds[i] = kNoiseDataStds[i];
- self->speech_stds[i] = kSpeechDataStds[i];
- }
-
- // Initialize Index and Minimum value vectors.
- for (i = 0; i < 16 * NUM_CHANNELS; i++) {
- self->low_value_vector[i] = 10000;
- self->index_vector[i] = 0;
- }
-
- // Initialize splitting filter states.
- memset(self->upper_state, 0, sizeof(self->upper_state));
- memset(self->lower_state, 0, sizeof(self->lower_state));
-
- // Initialize high pass filter states.
- memset(self->hp_filter_state, 0, sizeof(self->hp_filter_state));
-
- // Initialize mean value memory, for WebRtcVad_FindMinimum().
- for (i = 0; i < NUM_CHANNELS; i++) {
- self->mean_value[i] = 1600;
- }
-
- // Set aggressiveness mode to default (=|kDefaultMode|).
- if (WebRtcVad_set_mode_core(self, kDefaultMode) != 0) {
- return -1;
- }
-
- self->init_flag = kInitCheck;
-
- return 0;
-}
-
-// Set aggressiveness mode
-int WebRtcVad_set_mode_core(VadInstT *inst, int mode)
-{
-
- if (mode == 0)
- {
- // Quality mode
- inst->over_hang_max_1[0] = OHMAX1_10MS_Q; // Overhang short speech burst
- inst->over_hang_max_1[1] = OHMAX1_20MS_Q; // Overhang short speech burst
- inst->over_hang_max_1[2] = OHMAX1_30MS_Q; // Overhang short speech burst
- inst->over_hang_max_2[0] = OHMAX2_10MS_Q; // Overhang long speech burst
- inst->over_hang_max_2[1] = OHMAX2_20MS_Q; // Overhang long speech burst
- inst->over_hang_max_2[2] = OHMAX2_30MS_Q; // Overhang long speech burst
-
- inst->individual[0] = INDIVIDUAL_10MS_Q;
- inst->individual[1] = INDIVIDUAL_20MS_Q;
- inst->individual[2] = INDIVIDUAL_30MS_Q;
-
- inst->total[0] = TOTAL_10MS_Q;
- inst->total[1] = TOTAL_20MS_Q;
- inst->total[2] = TOTAL_30MS_Q;
- } else if (mode == 1)
- {
- // Low bitrate mode
- inst->over_hang_max_1[0] = OHMAX1_10MS_LBR; // Overhang short speech burst
- inst->over_hang_max_1[1] = OHMAX1_20MS_LBR; // Overhang short speech burst
- inst->over_hang_max_1[2] = OHMAX1_30MS_LBR; // Overhang short speech burst
- inst->over_hang_max_2[0] = OHMAX2_10MS_LBR; // Overhang long speech burst
- inst->over_hang_max_2[1] = OHMAX2_20MS_LBR; // Overhang long speech burst
- inst->over_hang_max_2[2] = OHMAX2_30MS_LBR; // Overhang long speech burst
-
- inst->individual[0] = INDIVIDUAL_10MS_LBR;
- inst->individual[1] = INDIVIDUAL_20MS_LBR;
- inst->individual[2] = INDIVIDUAL_30MS_LBR;
-
- inst->total[0] = TOTAL_10MS_LBR;
- inst->total[1] = TOTAL_20MS_LBR;
- inst->total[2] = TOTAL_30MS_LBR;
- } else if (mode == 2)
- {
- // Aggressive mode
- inst->over_hang_max_1[0] = OHMAX1_10MS_AGG; // Overhang short speech burst
- inst->over_hang_max_1[1] = OHMAX1_20MS_AGG; // Overhang short speech burst
- inst->over_hang_max_1[2] = OHMAX1_30MS_AGG; // Overhang short speech burst
- inst->over_hang_max_2[0] = OHMAX2_10MS_AGG; // Overhang long speech burst
- inst->over_hang_max_2[1] = OHMAX2_20MS_AGG; // Overhang long speech burst
- inst->over_hang_max_2[2] = OHMAX2_30MS_AGG; // Overhang long speech burst
-
- inst->individual[0] = INDIVIDUAL_10MS_AGG;
- inst->individual[1] = INDIVIDUAL_20MS_AGG;
- inst->individual[2] = INDIVIDUAL_30MS_AGG;
-
- inst->total[0] = TOTAL_10MS_AGG;
- inst->total[1] = TOTAL_20MS_AGG;
- inst->total[2] = TOTAL_30MS_AGG;
- } else if (mode == 3)
- {
- // Very aggressive mode
- inst->over_hang_max_1[0] = OHMAX1_10MS_VAG; // Overhang short speech burst
- inst->over_hang_max_1[1] = OHMAX1_20MS_VAG; // Overhang short speech burst
- inst->over_hang_max_1[2] = OHMAX1_30MS_VAG; // Overhang short speech burst
- inst->over_hang_max_2[0] = OHMAX2_10MS_VAG; // Overhang long speech burst
- inst->over_hang_max_2[1] = OHMAX2_20MS_VAG; // Overhang long speech burst
- inst->over_hang_max_2[2] = OHMAX2_30MS_VAG; // Overhang long speech burst
-
- inst->individual[0] = INDIVIDUAL_10MS_VAG;
- inst->individual[1] = INDIVIDUAL_20MS_VAG;
- inst->individual[2] = INDIVIDUAL_30MS_VAG;
-
- inst->total[0] = TOTAL_10MS_VAG;
- inst->total[1] = TOTAL_20MS_VAG;
- inst->total[2] = TOTAL_30MS_VAG;
- } else
- {
- return -1;
- }
-
- return 0;
-}
-
-// Calculate VAD decision by first extracting feature values and then calculate
-// probability for both speech and background noise.
-
-WebRtc_Word16 WebRtcVad_CalcVad32khz(VadInstT *inst, WebRtc_Word16 *speech_frame,
- int frame_length)
-{
- WebRtc_Word16 len, vad;
- WebRtc_Word16 speechWB[480]; // Downsampled speech frame: 960 samples (30ms in SWB)
- WebRtc_Word16 speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
-
-
- // Downsample signal 32->16->8 before doing VAD
- WebRtcVad_Downsampling(speech_frame, speechWB, &(inst->downsampling_filter_states[2]),
- frame_length);
- len = WEBRTC_SPL_RSHIFT_W16(frame_length, 1);
-
- WebRtcVad_Downsampling(speechWB, speechNB, inst->downsampling_filter_states, len);
- len = WEBRTC_SPL_RSHIFT_W16(len, 1);
-
- // Do VAD on an 8 kHz signal
- vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
-
- return vad;
-}
-
-WebRtc_Word16 WebRtcVad_CalcVad16khz(VadInstT *inst, WebRtc_Word16 *speech_frame,
- int frame_length)
-{
- WebRtc_Word16 len, vad;
- WebRtc_Word16 speechNB[240]; // Downsampled speech frame: 480 samples (30ms in WB)
-
- // Wideband: Downsample signal before doing VAD
- WebRtcVad_Downsampling(speech_frame, speechNB, inst->downsampling_filter_states,
- frame_length);
-
- len = WEBRTC_SPL_RSHIFT_W16(frame_length, 1);
- vad = WebRtcVad_CalcVad8khz(inst, speechNB, len);
-
- return vad;
-}
-
-WebRtc_Word16 WebRtcVad_CalcVad8khz(VadInstT *inst, WebRtc_Word16 *speech_frame,
- int frame_length)
-{
- WebRtc_Word16 feature_vector[NUM_CHANNELS], total_power;
-
- // Get power in the bands
- total_power = WebRtcVad_CalculateFeatures(inst, speech_frame, frame_length,
- feature_vector);
-
- // Make a VAD
- inst->vad = GmmProbability(inst, feature_vector, total_power, frame_length);
-
- return inst->vad;
-}
diff --git a/trunk/src/common_audio/vad/vad_core.h b/trunk/src/common_audio/vad/vad_core.h
deleted file mode 100644
index c82fbce..0000000
--- a/trunk/src/common_audio/vad/vad_core.h
+++ /dev/null
@@ -1,106 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the descriptions of the core VAD calls.
- */
-
-#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_CORE_H_
-#define WEBRTC_COMMON_AUDIO_VAD_VAD_CORE_H_
-
-#include "typedefs.h"
-#include "vad_defines.h"
-
-typedef struct VadInstT_
-{
-
- WebRtc_Word16 vad;
- WebRtc_Word32 downsampling_filter_states[4];
- WebRtc_Word16 noise_means[NUM_TABLE_VALUES];
- WebRtc_Word16 speech_means[NUM_TABLE_VALUES];
- WebRtc_Word16 noise_stds[NUM_TABLE_VALUES];
- WebRtc_Word16 speech_stds[NUM_TABLE_VALUES];
- // TODO(bjornv): Change to |frame_count|.
- WebRtc_Word32 frame_counter;
- WebRtc_Word16 over_hang; // Over Hang
- WebRtc_Word16 num_of_speech;
- // TODO(bjornv): Change to |age_vector|.
- WebRtc_Word16 index_vector[16 * NUM_CHANNELS];
- WebRtc_Word16 low_value_vector[16 * NUM_CHANNELS];
- // TODO(bjornv): Change to |median|.
- WebRtc_Word16 mean_value[NUM_CHANNELS];
- WebRtc_Word16 upper_state[5];
- WebRtc_Word16 lower_state[5];
- WebRtc_Word16 hp_filter_state[4];
- WebRtc_Word16 over_hang_max_1[3];
- WebRtc_Word16 over_hang_max_2[3];
- WebRtc_Word16 individual[3];
- WebRtc_Word16 total[3];
-
- int init_flag;
-
-} VadInstT;
-
-// Initializes the core VAD component. The default aggressiveness mode is
-// controlled by |kDefaultMode| in vad_core.c.
-//
-// - self [i/o] : Instance that should be initialized
-//
-// returns : 0 (OK), -1 (NULL pointer in or if the default mode can't be
-// set)
-int WebRtcVad_InitCore(VadInstT* self);
-
-/****************************************************************************
- * WebRtcVad_set_mode_core(...)
- *
- * This function changes the VAD settings
- *
- * Input:
- * - inst : VAD instance
- * - mode : Aggressiveness degree
- * 0 (High quality) - 3 (Highly aggressive)
- *
- * Output:
- * - inst : Changed instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcVad_set_mode_core(VadInstT* inst, int mode);
-
-/****************************************************************************
- * WebRtcVad_CalcVad32khz(...)
- * WebRtcVad_CalcVad16khz(...)
- * WebRtcVad_CalcVad8khz(...)
- *
- * Calculate probability for active speech and make VAD decision.
- *
- * Input:
- * - inst : Instance that should be initialized
- * - speech_frame : Input speech frame
- * - frame_length : Number of input samples
- *
- * Output:
- * - inst : Updated filter states etc.
- *
- * Return value : VAD decision
- * 0 - No active speech
- * 1-6 - Active speech
- */
-WebRtc_Word16 WebRtcVad_CalcVad32khz(VadInstT* inst, WebRtc_Word16* speech_frame,
- int frame_length);
-WebRtc_Word16 WebRtcVad_CalcVad16khz(VadInstT* inst, WebRtc_Word16* speech_frame,
- int frame_length);
-WebRtc_Word16 WebRtcVad_CalcVad8khz(VadInstT* inst, WebRtc_Word16* speech_frame,
- int frame_length);
-
-#endif // WEBRTC_COMMON_AUDIO_VAD_VAD_CORE_H_
diff --git a/trunk/src/common_audio/vad/vad_core_unittest.cc b/trunk/src/common_audio/vad/vad_core_unittest.cc
deleted file mode 100644
index 141b796..0000000
--- a/trunk/src/common_audio/vad/vad_core_unittest.cc
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdlib.h>
-
-#include "gtest/gtest.h"
-#include "typedefs.h"
-#include "vad_unittest.h"
-
-extern "C" {
-#include "vad_core.h"
-}
-
-namespace {
-
-TEST_F(VadTest, InitCore) {
- // Test WebRtcVad_InitCore().
- VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
-
- // NULL pointer test.
- EXPECT_EQ(-1, WebRtcVad_InitCore(NULL));
-
- // Verify return = 0 for non-NULL pointer.
- EXPECT_EQ(0, WebRtcVad_InitCore(self));
- // Verify init_flag is set.
- EXPECT_EQ(42, self->init_flag);
-
- free(self);
-}
-
-TEST_F(VadTest, set_mode_core) {
- VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
-
- // TODO(bjornv): Add NULL pointer check if we take care of it in
- // vad_core.c
-
- ASSERT_EQ(0, WebRtcVad_InitCore(self));
- // Test WebRtcVad_set_mode_core().
- // Invalid modes should return -1.
- EXPECT_EQ(-1, WebRtcVad_set_mode_core(self, -1));
- EXPECT_EQ(-1, WebRtcVad_set_mode_core(self, 1000));
- // Valid modes should return 0.
- for (size_t j = 0; j < kModesSize; ++j) {
- EXPECT_EQ(0, WebRtcVad_set_mode_core(self, kModes[j]));
- }
-
- free(self);
-}
-
-TEST_F(VadTest, CalcVad) {
- VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
- int16_t speech[kMaxFrameLength];
-
- // TODO(bjornv): Add NULL pointer check if we take care of it in
- // vad_core.c
-
- // Test WebRtcVad_CalcVadXXkhz()
- // Verify that all zeros in gives VAD = 0 out.
- memset(speech, 0, sizeof(speech));
- ASSERT_EQ(0, WebRtcVad_InitCore(self));
- for (size_t j = 0; j < kFrameLengthsSize; ++j) {
- if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
- EXPECT_EQ(0, WebRtcVad_CalcVad8khz(self, speech, kFrameLengths[j]));
- }
- if (ValidRatesAndFrameLengths(16000, kFrameLengths[j])) {
- EXPECT_EQ(0, WebRtcVad_CalcVad16khz(self, speech, kFrameLengths[j]));
- }
- if (ValidRatesAndFrameLengths(32000, kFrameLengths[j])) {
- EXPECT_EQ(0, WebRtcVad_CalcVad32khz(self, speech, kFrameLengths[j]));
- }
- }
-
- // Construct a speech signal that will trigger the VAD in all modes. It is
- // known that (i * i) will wrap around, but that doesn't matter in this case.
- for (int16_t i = 0; i < kMaxFrameLength; ++i) {
- speech[i] = (i * i);
- }
- for (size_t j = 0; j < kFrameLengthsSize; ++j) {
- if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
- EXPECT_EQ(1, WebRtcVad_CalcVad8khz(self, speech, kFrameLengths[j]));
- }
- if (ValidRatesAndFrameLengths(16000, kFrameLengths[j])) {
- EXPECT_EQ(1, WebRtcVad_CalcVad16khz(self, speech, kFrameLengths[j]));
- }
- if (ValidRatesAndFrameLengths(32000, kFrameLengths[j])) {
- EXPECT_EQ(1, WebRtcVad_CalcVad32khz(self, speech, kFrameLengths[j]));
- }
- }
-
- free(self);
-}
-} // namespace
diff --git a/trunk/src/common_audio/vad/vad_defines.h b/trunk/src/common_audio/vad/vad_defines.h
deleted file mode 100644
index 5d1539d..0000000
--- a/trunk/src/common_audio/vad/vad_defines.h
+++ /dev/null
@@ -1,93 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This header file includes the macros used in VAD.
- */
-
-#ifndef WEBRTC_VAD_DEFINES_H_
-#define WEBRTC_VAD_DEFINES_H_
-
-#define NUM_CHANNELS 6 // Eight frequency bands
-#define NUM_MODELS 2 // Number of Gaussian models
-#define NUM_TABLE_VALUES NUM_CHANNELS * NUM_MODELS
-
-#define MIN_ENERGY 10
-#define ALPHA1 6553 // 0.2 in Q15
-#define ALPHA2 32439 // 0.99 in Q15
-// Mode 0, Quality thresholds - Different thresholds for the different frame lengths
-#define INDIVIDUAL_10MS_Q 24
-#define INDIVIDUAL_20MS_Q 21 // (log10(2)*66)<<2 ~=16
-#define INDIVIDUAL_30MS_Q 24
-
-#define TOTAL_10MS_Q 57
-#define TOTAL_20MS_Q 48
-#define TOTAL_30MS_Q 57
-
-#define OHMAX1_10MS_Q 8 // Max Overhang 1
-#define OHMAX2_10MS_Q 14 // Max Overhang 2
-#define OHMAX1_20MS_Q 4 // Max Overhang 1
-#define OHMAX2_20MS_Q 7 // Max Overhang 2
-#define OHMAX1_30MS_Q 3
-#define OHMAX2_30MS_Q 5
-
-// Mode 1, Low bitrate thresholds - Different thresholds for the different frame lengths
-#define INDIVIDUAL_10MS_LBR 37
-#define INDIVIDUAL_20MS_LBR 32
-#define INDIVIDUAL_30MS_LBR 37
-
-#define TOTAL_10MS_LBR 100
-#define TOTAL_20MS_LBR 80
-#define TOTAL_30MS_LBR 100
-
-#define OHMAX1_10MS_LBR 8 // Max Overhang 1
-#define OHMAX2_10MS_LBR 14 // Max Overhang 2
-#define OHMAX1_20MS_LBR 4
-#define OHMAX2_20MS_LBR 7
-
-#define OHMAX1_30MS_LBR 3
-#define OHMAX2_30MS_LBR 5
-
-// Mode 2, Very aggressive thresholds - Different thresholds for the different frame lengths
-#define INDIVIDUAL_10MS_AGG 82
-#define INDIVIDUAL_20MS_AGG 78
-#define INDIVIDUAL_30MS_AGG 82
-
-#define TOTAL_10MS_AGG 285 //580
-#define TOTAL_20MS_AGG 260
-#define TOTAL_30MS_AGG 285
-
-#define OHMAX1_10MS_AGG 6 // Max Overhang 1
-#define OHMAX2_10MS_AGG 9 // Max Overhang 2
-#define OHMAX1_20MS_AGG 3
-#define OHMAX2_20MS_AGG 5
-
-#define OHMAX1_30MS_AGG 2
-#define OHMAX2_30MS_AGG 3
-
-// Mode 3, Super aggressive thresholds - Different thresholds for the different frame lengths
-#define INDIVIDUAL_10MS_VAG 94
-#define INDIVIDUAL_20MS_VAG 94
-#define INDIVIDUAL_30MS_VAG 94
-
-#define TOTAL_10MS_VAG 1100 //1700
-#define TOTAL_20MS_VAG 1050
-#define TOTAL_30MS_VAG 1100
-
-#define OHMAX1_10MS_VAG 6 // Max Overhang 1
-#define OHMAX2_10MS_VAG 9 // Max Overhang 2
-#define OHMAX1_20MS_VAG 3
-#define OHMAX2_20MS_VAG 5
-
-#define OHMAX1_30MS_VAG 2
-#define OHMAX2_30MS_VAG 3
-
-#endif // WEBRTC_VAD_DEFINES_H_
diff --git a/trunk/src/common_audio/vad/vad_filterbank.c b/trunk/src/common_audio/vad/vad_filterbank.c
deleted file mode 100644
index 2f5db44..0000000
--- a/trunk/src/common_audio/vad/vad_filterbank.c
+++ /dev/null
@@ -1,335 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vad_filterbank.h"
-
-#include <assert.h>
-
-#include "signal_processing_library.h"
-#include "typedefs.h"
-#include "vad_defines.h"
-
-// Constants used in LogOfEnergy().
-static const int16_t kLogConst = 24660; // 160*log10(2) in Q9.
-static const int16_t kLogEnergyIntPart = 14336; // 14 in Q10
-
-// Coefficients used by HighPassFilter, Q14.
-static const int16_t kHpZeroCoefs[3] = { 6631, -13262, 6631 };
-static const int16_t kHpPoleCoefs[3] = { 16384, -7756, 5620 };
-
-// Allpass filter coefficients, upper and lower, in Q15.
-// Upper: 0.64, Lower: 0.17
-static const int16_t kAllPassCoefsQ15[2] = { 20972, 5571 };
-
-// Adjustment for division with two in SplitFilter.
-static const int16_t kOffsetVector[6] = { 368, 368, 272, 176, 176, 176 };
-
-// High pass filtering, with a cut-off frequency at 80 Hz, if the |data_in| is
-// sampled at 500 Hz.
-//
-// - data_in [i] : Input audio data sampled at 500 Hz.
-// - data_length [i] : Length of input and output data.
-// - filter_state [i/o] : State of the filter.
-// - data_out [o] : Output audio data in the frequency interval
-// 80 - 250 Hz.
-static void HighPassFilter(const int16_t* data_in, int data_length,
- int16_t* filter_state, int16_t* data_out) {
- int i;
- const int16_t* in_ptr = data_in;
- int16_t* out_ptr = data_out;
- int32_t tmp32 = 0;
-
-
- // The sum of the absolute values of the impulse response:
- // The zero/pole-filter has a max amplification of a single sample of: 1.4546
- // Impulse response: 0.4047 -0.6179 -0.0266 0.1993 0.1035 -0.0194
- // The all-zero section has a max amplification of a single sample of: 1.6189
- // Impulse response: 0.4047 -0.8094 0.4047 0 0 0
- // The all-pole section has a max amplification of a single sample of: 1.9931
- // Impulse response: 1.0000 0.4734 -0.1189 -0.2187 -0.0627 0.04532
-
- for (i = 0; i < data_length; i++) {
- // All-zero section (filter coefficients in Q14).
- tmp32 = WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[0], *in_ptr);
- tmp32 += WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[1], filter_state[0]);
- tmp32 += WEBRTC_SPL_MUL_16_16(kHpZeroCoefs[2], filter_state[1]);
- filter_state[1] = filter_state[0];
- filter_state[0] = *in_ptr++;
-
- // All-pole section (filter coefficients in Q14).
- tmp32 -= WEBRTC_SPL_MUL_16_16(kHpPoleCoefs[1], filter_state[2]);
- tmp32 -= WEBRTC_SPL_MUL_16_16(kHpPoleCoefs[2], filter_state[3]);
- filter_state[3] = filter_state[2];
- filter_state[2] = (int16_t) (tmp32 >> 14);
- *out_ptr++ = filter_state[2];
- }
-}
-
-// All pass filtering of |data_in|, used before splitting the signal into two
-// frequency bands (low pass vs high pass).
-// Note that |data_in| and |data_out| can NOT correspond to the same address.
-//
-// - data_in [i] : Input audio signal given in Q0.
-// - data_length [i] : Length of input and output data.
-// - filter_coefficient [i] : Given in Q15.
-// - filter_state [i/o] : State of the filter given in Q(-1).
-// - data_out [o] : Output audio signal given in Q(-1).
-static void AllPassFilter(const int16_t* data_in, int data_length,
- int16_t filter_coefficient, int16_t* filter_state,
- int16_t* data_out) {
- // The filter can only cause overflow (in the w16 output variable)
- // if more than 4 consecutive input numbers are of maximum value and
- // has the the same sign as the impulse responses first taps.
- // First 6 taps of the impulse response:
- // 0.6399 0.5905 -0.3779 0.2418 -0.1547 0.0990
-
- int i;
- int16_t tmp16 = 0;
- int32_t tmp32 = 0;
- int32_t state32 = ((int32_t) (*filter_state) << 16); // Q15
-
- for (i = 0; i < data_length; i++) {
- tmp32 = state32 + WEBRTC_SPL_MUL_16_16(filter_coefficient, *data_in);
- tmp16 = (int16_t) (tmp32 >> 16); // Q(-1)
- *data_out++ = tmp16;
- state32 = (((int32_t) (*data_in)) << 14); // Q14
- state32 -= WEBRTC_SPL_MUL_16_16(filter_coefficient, tmp16); // Q14
- state32 <<= 1; // Q15.
- data_in += 2;
- }
-
- *filter_state = (int16_t) (state32 >> 16); // Q(-1)
-}
-
-// Splits |data_in| into |hp_data_out| and |lp_data_out| corresponding to
-// an upper (high pass) part and a lower (low pass) part respectively.
-//
-// - data_in [i] : Input audio data to be split into two frequency bands.
-// - data_length [i] : Length of |data_in|.
-// - upper_state [i/o] : State of the upper filter, given in Q(-1).
-// - lower_state [i/o] : State of the lower filter, given in Q(-1).
-// - hp_data_out [o] : Output audio data of the upper half of the spectrum.
-// The length is |data_length| / 2.
-// - lp_data_out [o] : Output audio data of the lower half of the spectrum.
-// The length is |data_length| / 2.
-static void SplitFilter(const int16_t* data_in, int data_length,
- int16_t* upper_state, int16_t* lower_state,
- int16_t* hp_data_out, int16_t* lp_data_out) {
- int i;
- int half_length = data_length >> 1; // Downsampling by 2.
- int16_t tmp_out;
-
- // All-pass filtering upper branch.
- AllPassFilter(&data_in[0], half_length, kAllPassCoefsQ15[0], upper_state,
- hp_data_out);
-
- // All-pass filtering lower branch.
- AllPassFilter(&data_in[1], half_length, kAllPassCoefsQ15[1], lower_state,
- lp_data_out);
-
- // Make LP and HP signals.
- for (i = 0; i < half_length; i++) {
- tmp_out = *hp_data_out;
- *hp_data_out++ -= *lp_data_out;
- *lp_data_out++ += tmp_out;
- }
-}
-
-// Calculates the energy of |data_in| in dB, and also updates an overall
-// |total_energy| if necessary.
-//
-// - data_in [i] : Input audio data for energy calculation.
-// - data_length [i] : Length of input data.
-// - offset [i] : Offset value added to |log_energy|.
-// - total_energy [i/o] : An external energy updated with the energy of
-// |data_in|.
-// NOTE: |total_energy| is only updated if
-// |total_energy| <= MIN_ENERGY.
-// - log_energy [o] : 10 * log10("energy of |data_in|") given in Q4.
-static void LogOfEnergy(const int16_t* data_in, int data_length,
- int16_t offset, int16_t* total_energy,
- int16_t* log_energy) {
- // |tot_rshifts| accumulates the number of right shifts performed on |energy|.
- int tot_rshifts = 0;
- // The |energy| will be normalized to 15 bits. We use unsigned integer because
- // we eventually will mask out the fractional part.
- uint32_t energy = 0;
-
- assert(data_in != NULL);
- assert(data_length > 0);
-
- energy = (uint32_t) WebRtcSpl_Energy((int16_t*) data_in, data_length,
- &tot_rshifts);
-
- if (energy != 0) {
- // By construction, normalizing to 15 bits is equivalent with 17 leading
- // zeros of an unsigned 32 bit value.
- int normalizing_rshifts = 17 - WebRtcSpl_NormU32(energy);
- // In a 15 bit representation the leading bit is 2^14. log2(2^14) in Q10 is
- // (14 << 10), which is what we initialize |log2_energy| with. For a more
- // detailed derivations, see below.
- int16_t log2_energy = kLogEnergyIntPart;
-
- tot_rshifts += normalizing_rshifts;
- // Normalize |energy| to 15 bits.
- // |tot_rshifts| is now the total number of right shifts performed on
- // |energy| after normalization. This means that |energy| is in
- // Q(-tot_rshifts).
- if (normalizing_rshifts < 0) {
- energy <<= -normalizing_rshifts;
- } else {
- energy >>= normalizing_rshifts;
- }
-
- // Calculate the energy of |data_in| in dB, in Q4.
- //
- // 10 * log10("true energy") in Q4 = 2^4 * 10 * log10("true energy") =
- // 160 * log10(|energy| * 2^|tot_rshifts|) =
- // 160 * log10(2) * log2(|energy| * 2^|tot_rshifts|) =
- // 160 * log10(2) * (log2(|energy|) + log2(2^|tot_rshifts|)) =
- // (160 * log10(2)) * (log2(|energy|) + |tot_rshifts|) =
- // |kLogConst| * (|log2_energy| + |tot_rshifts|)
- //
- // We know by construction that |energy| is normalized to 15 bits. Hence,
- // |energy| = 2^14 + frac_Q15, where frac_Q15 is a fractional part in Q15.
- // Further, we'd like |log2_energy| in Q10
- // log2(|energy|) in Q10 = 2^10 * log2(2^14 + frac_Q15) =
- // 2^10 * log2(2^14 * (1 + frac_Q15 * 2^-14)) =
- // 2^10 * (14 + log2(1 + frac_Q15 * 2^-14)) ~=
- // (14 << 10) + 2^10 * (frac_Q15 * 2^-14) =
- // (14 << 10) + (frac_Q15 * 2^-4) = (14 << 10) + (frac_Q15 >> 4)
- //
- // Note that frac_Q15 = (|energy| & 0x00003FFF)
-
- // Calculate and add the fractional part to |log2_energy|.
- log2_energy += (int16_t) ((energy & 0x00003FFF) >> 4);
-
- // |kLogConst| is in Q9, |log2_energy| in Q10 and |tot_rshifts| in Q0.
- // Note that we in our derivation above have accounted for an output in Q4.
- *log_energy = (int16_t) (WEBRTC_SPL_MUL_16_16_RSFT(
- kLogConst, log2_energy, 19) +
- WEBRTC_SPL_MUL_16_16_RSFT(tot_rshifts, kLogConst, 9));
-
- if (*log_energy < 0) {
- *log_energy = 0;
- }
- } else {
- *log_energy = offset;
- return;
- }
-
- *log_energy += offset;
-
- // Update the approximate |total_energy| with the energy of |data_in|, if
- // |total_energy| has not exceeded MIN_ENERGY. |total_energy| is used as an
- // energy indicator in WebRtcVad_GmmProbability() in vad_core.c.
- if (*total_energy <= MIN_ENERGY) {
- if (tot_rshifts >= 0) {
- // We know by construction that the |energy| > MIN_ENERGY in Q0, so add an
- // arbitrary value such that |total_energy| exceeds MIN_ENERGY.
- *total_energy += MIN_ENERGY + 1;
- } else {
- // By construction |energy| is represented by 15 bits, hence any number of
- // right shifted |energy| will fit in an int16_t. In addition, adding the
- // value to |total_energy| is wrap around safe as long as
- // MIN_ENERGY < 8192.
- *total_energy += (int16_t) (energy >> -tot_rshifts); // Q0.
- }
- }
-}
-
-int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
- int data_length, int16_t* features) {
- int16_t total_energy = 0;
- // We expect |data_length| to be 80, 160 or 240 samples, which corresponds to
- // 10, 20 or 30 ms in 8 kHz. Therefore, the intermediate downsampled data will
- // have at most 120 samples after the first split and at most 60 samples after
- // the second split.
- int16_t hp_120[120], lp_120[120];
- int16_t hp_60[60], lp_60[60];
- const int half_data_length = data_length >> 1;
- int length = half_data_length; // |data_length| / 2, corresponds to
- // bandwidth = 2000 Hz after downsampling.
-
- // Initialize variables for the first SplitFilter().
- int frequency_band = 0;
- const int16_t* in_ptr = data_in; // [0 - 4000] Hz.
- int16_t* hp_out_ptr = hp_120; // [2000 - 4000] Hz.
- int16_t* lp_out_ptr = lp_120; // [0 - 2000] Hz.
-
- assert(data_length >= 0);
- assert(data_length <= 240);
- assert(4 < NUM_CHANNELS - 1); // Checking maximum |frequency_band|.
-
- // Split at 2000 Hz and downsample.
- SplitFilter(in_ptr, data_length, &self->upper_state[frequency_band],
- &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
-
- // For the upper band (2000 Hz - 4000 Hz) split at 3000 Hz and downsample.
- frequency_band = 1;
- in_ptr = hp_120; // [2000 - 4000] Hz.
- hp_out_ptr = hp_60; // [3000 - 4000] Hz.
- lp_out_ptr = lp_60; // [2000 - 3000] Hz.
- SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
- &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
-
- // Energy in 3000 Hz - 4000 Hz.
- length >>= 1; // |data_length| / 4 <=> bandwidth = 1000 Hz.
-
- LogOfEnergy(hp_60, length, kOffsetVector[5], &total_energy, &features[5]);
-
- // Energy in 2000 Hz - 3000 Hz.
- LogOfEnergy(lp_60, length, kOffsetVector[4], &total_energy, &features[4]);
-
- // For the lower band (0 Hz - 2000 Hz) split at 1000 Hz and downsample.
- frequency_band = 2;
- in_ptr = lp_120; // [0 - 2000] Hz.
- hp_out_ptr = hp_60; // [1000 - 2000] Hz.
- lp_out_ptr = lp_60; // [0 - 1000] Hz.
- length = half_data_length; // |data_length| / 2 <=> bandwidth = 2000 Hz.
- SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
- &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
-
- // Energy in 1000 Hz - 2000 Hz.
- length >>= 1; // |data_length| / 4 <=> bandwidth = 1000 Hz.
- LogOfEnergy(hp_60, length, kOffsetVector[3], &total_energy, &features[3]);
-
- // For the lower band (0 Hz - 1000 Hz) split at 500 Hz and downsample.
- frequency_band = 3;
- in_ptr = lp_60; // [0 - 1000] Hz.
- hp_out_ptr = hp_120; // [500 - 1000] Hz.
- lp_out_ptr = lp_120; // [0 - 500] Hz.
- SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
- &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
-
- // Energy in 500 Hz - 1000 Hz.
- length >>= 1; // |data_length| / 8 <=> bandwidth = 500 Hz.
- LogOfEnergy(hp_120, length, kOffsetVector[2], &total_energy, &features[2]);
-
- // For the lower band (0 Hz - 500 Hz) split at 250 Hz and downsample.
- frequency_band = 4;
- in_ptr = lp_120; // [0 - 500] Hz.
- hp_out_ptr = hp_60; // [250 - 500] Hz.
- lp_out_ptr = lp_60; // [0 - 250] Hz.
- SplitFilter(in_ptr, length, &self->upper_state[frequency_band],
- &self->lower_state[frequency_band], hp_out_ptr, lp_out_ptr);
-
- // Energy in 250 Hz - 500 Hz.
- length >>= 1; // |data_length| / 16 <=> bandwidth = 250 Hz.
- LogOfEnergy(hp_60, length, kOffsetVector[1], &total_energy, &features[1]);
-
- // Remove 0 Hz - 80 Hz, by high pass filtering the lower band.
- HighPassFilter(lp_60, length, self->hp_filter_state, hp_120);
-
- // Energy in 80 Hz - 250 Hz.
- LogOfEnergy(hp_120, length, kOffsetVector[0], &total_energy, &features[0]);
-
- return total_energy;
-}
diff --git a/trunk/src/common_audio/vad/vad_filterbank.h b/trunk/src/common_audio/vad/vad_filterbank.h
deleted file mode 100644
index 0c5c00c..0000000
--- a/trunk/src/common_audio/vad/vad_filterbank.h
+++ /dev/null
@@ -1,44 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file includes feature calculating functionality used in vad_core.c.
- */
-
-#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
-#define WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
-
-#include "typedefs.h"
-#include "vad_core.h"
-
-// Takes |data_length| samples of |data_in| and calculates the logarithm of the
-// energy of each of the |NUM_CHANNELS| = 6 frequency bands used by the VAD:
-// 80 Hz - 250 Hz
-// 250 Hz - 500 Hz
-// 500 Hz - 1000 Hz
-// 1000 Hz - 2000 Hz
-// 2000 Hz - 3000 Hz
-// 3000 Hz - 4000 Hz
-//
-// The values are given in Q4 and written to |features|. Further, an approximate
-// overall energy is returned. The return value is used in
-// WebRtcVad_GmmProbability() as a signal indicator, hence it is arbitrary above
-// the threshold MIN_ENERGY.
-//
-// - self [i/o] : State information of the VAD.
-// - data_in [i] : Input audio data, for feature extraction.
-// - data_length [i] : Audio data size, in number of samples.
-// - features [o] : 10 * log10(energy in each frequency band), Q4.
-// - returns : Total energy of the signal (NOTE! This value is not
-// exact. It is only used in a comparison.)
-int16_t WebRtcVad_CalculateFeatures(VadInstT* self, const int16_t* data_in,
- int data_length, int16_t* features);
-
-#endif // WEBRTC_COMMON_AUDIO_VAD_VAD_FILTERBANK_H_
diff --git a/trunk/src/common_audio/vad/vad_filterbank_unittest.cc b/trunk/src/common_audio/vad/vad_filterbank_unittest.cc
deleted file mode 100644
index 320fda9..0000000
--- a/trunk/src/common_audio/vad/vad_filterbank_unittest.cc
+++ /dev/null
@@ -1,93 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdlib.h>
-
-#include "gtest/gtest.h"
-#include "typedefs.h"
-#include "vad_unittest.h"
-
-extern "C" {
-#include "vad_core.h"
-#include "vad_defines.h"
-#include "vad_filterbank.h"
-}
-
-namespace {
-
-enum { kNumValidFrameLengths = 3 };
-
-TEST_F(VadTest, vad_filterbank) {
- VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
- static const int16_t kReference[kNumValidFrameLengths] = { 48, 11, 11 };
- static const int16_t kFeatures[kNumValidFrameLengths * NUM_CHANNELS] = {
- 1213, 759, 587, 462, 434, 272,
- 1479, 1385, 1291, 1200, 1103, 1099,
- 1732, 1692, 1681, 1629, 1436, 1436
- };
- static const int16_t kOffsetVector[NUM_CHANNELS] = {
- 368, 368, 272, 176, 176, 176 };
- int16_t features[NUM_CHANNELS];
-
- // Construct a speech signal that will trigger the VAD in all modes. It is
- // known that (i * i) will wrap around, but that doesn't matter in this case.
- int16_t speech[kMaxFrameLength];
- for (int16_t i = 0; i < kMaxFrameLength; ++i) {
- speech[i] = (i * i);
- }
-
- int frame_length_index = 0;
- ASSERT_EQ(0, WebRtcVad_InitCore(self));
- for (size_t j = 0; j < kFrameLengthsSize; ++j) {
- if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
- EXPECT_EQ(kReference[frame_length_index],
- WebRtcVad_CalculateFeatures(self, speech, kFrameLengths[j],
- features));
- for (int k = 0; k < NUM_CHANNELS; ++k) {
- EXPECT_EQ(kFeatures[k + frame_length_index * NUM_CHANNELS],
- features[k]);
- }
- frame_length_index++;
- }
- }
- EXPECT_EQ(kNumValidFrameLengths, frame_length_index);
-
- // Verify that all zeros in gives kOffsetVector out.
- memset(speech, 0, sizeof(speech));
- ASSERT_EQ(0, WebRtcVad_InitCore(self));
- for (size_t j = 0; j < kFrameLengthsSize; ++j) {
- if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
- EXPECT_EQ(0, WebRtcVad_CalculateFeatures(self, speech, kFrameLengths[j],
- features));
- for (int k = 0; k < NUM_CHANNELS; ++k) {
- EXPECT_EQ(kOffsetVector[k], features[k]);
- }
- }
- }
-
- // Verify that all ones in gives kOffsetVector out. Any other constant input
- // will have a small impact in the sub bands.
- for (int16_t i = 0; i < kMaxFrameLength; ++i) {
- speech[i] = 1;
- }
- for (size_t j = 0; j < kFrameLengthsSize; ++j) {
- if (ValidRatesAndFrameLengths(8000, kFrameLengths[j])) {
- ASSERT_EQ(0, WebRtcVad_InitCore(self));
- EXPECT_EQ(0, WebRtcVad_CalculateFeatures(self, speech, kFrameLengths[j],
- features));
- for (int k = 0; k < NUM_CHANNELS; ++k) {
- EXPECT_EQ(kOffsetVector[k], features[k]);
- }
- }
- }
-
- free(self);
-}
-} // namespace
diff --git a/trunk/src/common_audio/vad/vad_gmm.c b/trunk/src/common_audio/vad/vad_gmm.c
deleted file mode 100644
index 20a703a..0000000
--- a/trunk/src/common_audio/vad/vad_gmm.c
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vad_gmm.h"
-
-#include "signal_processing_library.h"
-#include "typedefs.h"
-
-static const int32_t kCompVar = 22005;
-static const int16_t kLog2Exp = 5909; // log2(exp(1)) in Q12.
-
-// For a normal distribution, the probability of |input| is calculated and
-// returned (in Q20). The formula for normal distributed probability is
-//
-// 1 / s * exp(-(x - m)^2 / (2 * s^2))
-//
-// where the parameters are given in the following Q domains:
-// m = |mean| (Q7)
-// s = |std| (Q7)
-// x = |input| (Q4)
-// in addition to the probability we output |delta| (in Q11) used when updating
-// the noise/speech model.
-int32_t WebRtcVad_GaussianProbability(int16_t input,
- int16_t mean,
- int16_t std,
- int16_t* delta) {
- int16_t tmp16, inv_std, inv_std2, exp_value = 0;
- int32_t tmp32;
-
- // Calculate |inv_std| = 1 / s, in Q10.
- // 131072 = 1 in Q17, and (|std| >> 1) is for rounding instead of truncation.
- // Q-domain: Q17 / Q7 = Q10.
- tmp32 = (int32_t) 131072 + (int32_t) (std >> 1);
- inv_std = (int16_t) WebRtcSpl_DivW32W16(tmp32, std);
-
- // Calculate |inv_std2| = 1 / s^2, in Q14.
- tmp16 = (inv_std >> 2); // Q10 -> Q8.
- // Q-domain: (Q8 * Q8) >> 2 = Q14.
- inv_std2 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(tmp16, tmp16, 2);
- // TODO(bjornv): Investigate if changing to
- // |inv_std2| = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(|inv_std|, |inv_std|, 6);
- // gives better accuracy.
-
- tmp16 = (input << 3); // Q4 -> Q7
- tmp16 = tmp16 - mean; // Q7 - Q7 = Q7
-
- // To be used later, when updating noise/speech model.
- // |delta| = (x - m) / s^2, in Q11.
- // Q-domain: (Q14 * Q7) >> 10 = Q11.
- *delta = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(inv_std2, tmp16, 10);
-
- // Calculate the exponent |tmp32| = (x - m)^2 / (2 * s^2), in Q10. Replacing
- // division by two with one shift.
- // Q-domain: (Q11 * Q7) >> 8 = Q10.
- tmp32 = WEBRTC_SPL_MUL_16_16_RSFT(*delta, tmp16, 9);
-
- // If the exponent is small enough to give a non-zero probability we calculate
- // |exp_value| ~= exp(-(x - m)^2 / (2 * s^2))
- // ~= exp2(-log2(exp(1)) * |tmp32|).
- if (tmp32 < kCompVar) {
- // Calculate |tmp16| = log2(exp(1)) * |tmp32|, in Q10.
- // Q-domain: (Q12 * Q10) >> 12 = Q10.
- tmp16 = (int16_t) WEBRTC_SPL_MUL_16_16_RSFT(kLog2Exp, (int16_t) tmp32, 12);
- tmp16 = -tmp16;
- exp_value = (0x0400 | (tmp16 & 0x03FF));
- tmp16 ^= 0xFFFF;
- tmp16 >>= 10;
- tmp16 += 1;
- // Get |exp_value| = exp(-|tmp32|) in Q10.
- exp_value >>= tmp16;
- }
-
- // Calculate and return (1 / s) * exp(-(x - m)^2 / (2 * s^2)), in Q20.
- // Q-domain: Q10 * Q10 = Q20.
- return WEBRTC_SPL_MUL_16_16(inv_std, exp_value);
-}
diff --git a/trunk/src/common_audio/vad/vad_gmm.h b/trunk/src/common_audio/vad/vad_gmm.h
deleted file mode 100644
index 2333af7..0000000
--- a/trunk/src/common_audio/vad/vad_gmm.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Gaussian probability calculations internally used in vad_core.c.
-
-#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_GMM_H_
-#define WEBRTC_COMMON_AUDIO_VAD_VAD_GMM_H_
-
-#include "typedefs.h"
-
-// Calculates the probability for |input|, given that |input| comes from a
-// normal distribution with mean and standard deviation (|mean|, |std|).
-//
-// Inputs:
-// - input : input sample in Q4.
-// - mean : mean input in the statistical model, Q7.
-// - std : standard deviation, Q7.
-//
-// Output:
-//
-// - delta : input used when updating the model, Q11.
-// |delta| = (|input| - |mean|) / |std|^2.
-//
-// Return:
-// (probability for |input|) =
-// 1 / |std| * exp(-(|input| - |mean|)^2 / (2 * |std|^2));
-int32_t WebRtcVad_GaussianProbability(int16_t input,
- int16_t mean,
- int16_t std,
- int16_t* delta);
-
-#endif // WEBRTC_COMMON_AUDIO_VAD_VAD_GMM_H_
diff --git a/trunk/src/common_audio/vad/vad_gmm_unittest.cc b/trunk/src/common_audio/vad/vad_gmm_unittest.cc
deleted file mode 100644
index 205435a..0000000
--- a/trunk/src/common_audio/vad/vad_gmm_unittest.cc
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "gtest/gtest.h"
-#include "typedefs.h"
-#include "vad_unittest.h"
-
-extern "C" {
-#include "vad_gmm.h"
-}
-
-namespace {
-
-TEST_F(VadTest, vad_gmm) {
- int16_t delta = 0;
- // Input value at mean.
- EXPECT_EQ(1048576, WebRtcVad_GaussianProbability(0, 0, 128, &delta));
- EXPECT_EQ(0, delta);
- EXPECT_EQ(1048576, WebRtcVad_GaussianProbability(16, 128, 128, &delta));
- EXPECT_EQ(0, delta);
- EXPECT_EQ(1048576, WebRtcVad_GaussianProbability(-16, -128, 128, &delta));
- EXPECT_EQ(0, delta);
-
- // Largest possible input to give non-zero probability.
- EXPECT_EQ(1024, WebRtcVad_GaussianProbability(59, 0, 128, &delta));
- EXPECT_EQ(7552, delta);
- EXPECT_EQ(1024, WebRtcVad_GaussianProbability(75, 128, 128, &delta));
- EXPECT_EQ(7552, delta);
- EXPECT_EQ(1024, WebRtcVad_GaussianProbability(-75, -128, 128, &delta));
- EXPECT_EQ(-7552, delta);
-
- // Too large input, should give zero probability.
- EXPECT_EQ(0, WebRtcVad_GaussianProbability(105, 0, 128, &delta));
- EXPECT_EQ(13440, delta);
-}
-} // namespace
diff --git a/trunk/src/common_audio/vad/vad_sp.c b/trunk/src/common_audio/vad/vad_sp.c
deleted file mode 100644
index 4fface3..0000000
--- a/trunk/src/common_audio/vad/vad_sp.c
+++ /dev/null
@@ -1,181 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vad_sp.h"
-
-#include <assert.h>
-
-#include "signal_processing_library.h"
-#include "typedefs.h"
-#include "vad_defines.h"
-
-// Allpass filter coefficients, upper and lower, in Q13.
-// Upper: 0.64, Lower: 0.17.
-static const int16_t kAllPassCoefsQ13[2] = { 5243, 1392 }; // Q13
-
-// TODO(bjornv): Move this function to vad_filterbank.c.
-// Downsampling filter based on splitting filter and allpass functions.
-void WebRtcVad_Downsampling(int16_t* signal_in,
- int16_t* signal_out,
- int32_t* filter_state,
- int in_length) {
- int16_t tmp16_1 = 0, tmp16_2 = 0;
- int32_t tmp32_1 = filter_state[0];
- int32_t tmp32_2 = filter_state[1];
- int n = 0;
- int half_length = (in_length >> 1); // Downsampling by 2 gives half length.
-
- // Filter coefficients in Q13, filter state in Q0.
- for (n = 0; n < half_length; n++) {
- // All-pass filtering upper branch.
- tmp16_1 = (int16_t) ((tmp32_1 >> 1) +
- WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[0], *signal_in, 14));
- *signal_out = tmp16_1;
- tmp32_1 = (int32_t) (*signal_in++) -
- WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[0], tmp16_1, 12);
-
- // All-pass filtering lower branch.
- tmp16_2 = (int16_t) ((tmp32_2 >> 1) +
- WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[1], *signal_in, 14));
- *signal_out++ += tmp16_2;
- tmp32_2 = (int32_t) (*signal_in++) -
- WEBRTC_SPL_MUL_16_16_RSFT(kAllPassCoefsQ13[1], tmp16_2, 12);
- }
- // Store the filter states.
- filter_state[0] = tmp32_1;
- filter_state[1] = tmp32_2;
-}
-
-// Inserts |feature_value| into |low_value_vector|, if it is one of the 16
-// smallest values the last 100 frames. Then calculates and returns the median
-// of the five smallest values.
-int16_t WebRtcVad_FindMinimum(VadInstT* self,
- int16_t feature_value,
- int channel) {
- int i = 0, j = 0;
- int position = -1;
- // Offset to beginning of the 16 minimum values in memory.
- int offset = (channel << 4);
- int16_t current_median = 1600;
- int16_t alpha = 0;
- int32_t tmp32 = 0;
- // Pointer to memory for the 16 minimum values and the age of each value of
- // the |channel|.
- int16_t* age_ptr = &self->index_vector[offset];
- int16_t* value_ptr = &self->low_value_vector[offset];
- int16_t *p1, *p2, *p3;
-
- assert(channel < NUM_CHANNELS);
-
- // Each value in |low_value_vector| is getting 1 loop older.
- // Update age of each value in |age_ptr|, and remove old values.
- for (i = 0; i < 16; i++) {
- p3 = age_ptr + i;
- if (*p3 != 100) {
- *p3 += 1;
- } else {
- p1 = value_ptr + i + 1;
- p2 = p3 + 1;
- for (j = i; j < 16; j++) {
- *(value_ptr + j) = *p1++;
- *(age_ptr + j) = *p2++;
- }
- *(age_ptr + 15) = 101;
- *(value_ptr + 15) = 10000;
- }
- }
-
- // Check if |feature_value| is smaller than any of the values in
- // |low_value_vector|. If so, find the |position| where to insert the new
- // value.
- if (feature_value < *(value_ptr + 7)) {
- if (feature_value < *(value_ptr + 3)) {
- if (feature_value < *(value_ptr + 1)) {
- if (feature_value < *value_ptr) {
- position = 0;
- } else {
- position = 1;
- }
- } else if (feature_value < *(value_ptr + 2)) {
- position = 2;
- } else {
- position = 3;
- }
- } else if (feature_value < *(value_ptr + 5)) {
- if (feature_value < *(value_ptr + 4)) {
- position = 4;
- } else {
- position = 5;
- }
- } else if (feature_value < *(value_ptr + 6)) {
- position = 6;
- } else {
- position = 7;
- }
- } else if (feature_value < *(value_ptr + 15)) {
- if (feature_value < *(value_ptr + 11)) {
- if (feature_value < *(value_ptr + 9)) {
- if (feature_value < *(value_ptr + 8)) {
- position = 8;
- } else {
- position = 9;
- }
- } else if (feature_value < *(value_ptr + 10)) {
- position = 10;
- } else {
- position = 11;
- }
- } else if (feature_value < *(value_ptr + 13)) {
- if (feature_value < *(value_ptr + 12)) {
- position = 12;
- } else {
- position = 13;
- }
- } else if (feature_value < *(value_ptr + 14)) {
- position = 14;
- } else {
- position = 15;
- }
- }
-
- // If we have a new small value, put it in the correct position and shift
- // larger values up.
- if (position > -1) {
- for (i = 15; i > position; i--) {
- j = i - 1;
- *(value_ptr + i) = *(value_ptr + j);
- *(age_ptr + i) = *(age_ptr + j);
- }
- *(value_ptr + position) = feature_value;
- *(age_ptr + position) = 1;
- }
-
- // Get |current_median|.
- if (self->frame_counter > 2) {
- current_median = *(value_ptr + 2);
- } else if (self->frame_counter > 0) {
- current_median = *value_ptr;
- }
-
- // Smooth the median value.
- if (self->frame_counter > 0) {
- if (current_median < self->mean_value[channel]) {
- alpha = (int16_t) ALPHA1; // 0.2 in Q15.
- } else {
- alpha = (int16_t) ALPHA2; // 0.99 in Q15.
- }
- }
- tmp32 = WEBRTC_SPL_MUL_16_16(alpha + 1, self->mean_value[channel]);
- tmp32 += WEBRTC_SPL_MUL_16_16(WEBRTC_SPL_WORD16_MAX - alpha, current_median);
- tmp32 += 16384;
- self->mean_value[channel] = (int16_t) (tmp32 >> 15);
-
- return self->mean_value[channel];
-}
diff --git a/trunk/src/common_audio/vad/vad_sp.h b/trunk/src/common_audio/vad/vad_sp.h
deleted file mode 100644
index 9e8b204..0000000
--- a/trunk/src/common_audio/vad/vad_sp.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-// This file includes specific signal processing tools used in vad_core.c.
-
-#ifndef WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_
-#define WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_
-
-#include "typedefs.h"
-#include "vad_core.h"
-
-// Downsamples the signal by a factor 2, eg. 32->16 or 16->8.
-//
-// Inputs:
-// - signal_in : Input signal.
-// - in_length : Length of input signal in samples.
-//
-// Input & Output:
-// - filter_state : Current filter states of the two all-pass filters. The
-// |filter_state| is updated after all samples have been
-// processed.
-//
-// Output:
-// - signal_out : Downsampled signal (of length |in_length| / 2).
-void WebRtcVad_Downsampling(int16_t* signal_in,
- int16_t* signal_out,
- int32_t* filter_state,
- int in_length);
-
-// Updates and returns the smoothed feature minimum. As minimum we use the
-// median of the five smallest feature values in a 100 frames long window.
-// As long as |handle->frame_counter| is zero, that is, we haven't received any
-// "valid" data, FindMinimum() outputs the default value of 1600.
-//
-// Inputs:
-// - feature_value : New feature value to update with.
-// - channel : Channel number.
-//
-// Input & Output:
-// - handle : State information of the VAD.
-//
-// Returns:
-// : Smoothed minimum value for a moving window.
-int16_t WebRtcVad_FindMinimum(VadInstT* handle,
- int16_t feature_value,
- int channel);
-
-#endif // WEBRTC_COMMON_AUDIO_VAD_VAD_SP_H_
diff --git a/trunk/src/common_audio/vad/vad_sp_unittest.cc b/trunk/src/common_audio/vad/vad_sp_unittest.cc
deleted file mode 100644
index 03c844e..0000000
--- a/trunk/src/common_audio/vad/vad_sp_unittest.cc
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdlib.h>
-
-#include "gtest/gtest.h"
-#include "typedefs.h"
-#include "vad_unittest.h"
-
-extern "C" {
-#include "vad_core.h"
-#include "vad_defines.h"
-#include "vad_sp.h"
-}
-
-namespace {
-
-TEST_F(VadTest, vad_sp) {
- VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
- int16_t zeros[kMaxFrameLength] = { 0 };
- int32_t state[2] = { 0 };
- int16_t data_in[kMaxFrameLength];
- int16_t data_out[kMaxFrameLength];
-
- // We expect the first value to be 1600 as long as |frame_counter| is zero,
- // which is true for the first iteration.
- static const int16_t kReferenceMin[32] = {
- 1600, 720, 509, 512, 532, 552, 570, 588,
- 606, 624, 642, 659, 675, 691, 707, 723,
- 1600, 544, 502, 522, 542, 561, 579, 597,
- 615, 633, 651, 667, 683, 699, 715, 731
- };
-
- // Construct a speech signal that will trigger the VAD in all modes. It is
- // known that (i * i) will wrap around, but that doesn't matter in this case.
- for (int16_t i = 0; i < kMaxFrameLength; ++i) {
- data_in[i] = (i * i);
- }
- // Input values all zeros, expect all zeros out.
- WebRtcVad_Downsampling(zeros, data_out, state,
- static_cast<int>(kMaxFrameLength));
- EXPECT_EQ(0, state[0]);
- EXPECT_EQ(0, state[1]);
- for (int16_t i = 0; i < kMaxFrameLength / 2; ++i) {
- EXPECT_EQ(0, data_out[i]);
- }
- // Make a simple non-zero data test.
- WebRtcVad_Downsampling(data_in, data_out, state,
- static_cast<int>(kMaxFrameLength));
- EXPECT_EQ(207, state[0]);
- EXPECT_EQ(2270, state[1]);
-
- ASSERT_EQ(0, WebRtcVad_InitCore(self));
- // TODO(bjornv): Replace this part of the test with taking values from an
- // array and calculate the reference value here. Make sure the values are not
- // ordered.
- for (int16_t i = 0; i < 16; ++i) {
- int16_t value = 500 * (i + 1);
- for (int j = 0; j < NUM_CHANNELS; ++j) {
- // Use values both above and below initialized value.
- EXPECT_EQ(kReferenceMin[i], WebRtcVad_FindMinimum(self, value, j));
- EXPECT_EQ(kReferenceMin[i + 16], WebRtcVad_FindMinimum(self, 12000, j));
- }
- self->frame_counter++;
- }
-
- free(self);
-}
-} // namespace
diff --git a/trunk/src/common_audio/vad/vad_unittest.cc b/trunk/src/common_audio/vad/vad_unittest.cc
deleted file mode 100644
index 60ee208..0000000
--- a/trunk/src/common_audio/vad/vad_unittest.cc
+++ /dev/null
@@ -1,131 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vad_unittest.h"
-
-#include <stdlib.h>
-
-#include "gtest/gtest.h"
-#include "typedefs.h"
-#include "webrtc_vad.h"
-
-VadTest::VadTest() {}
-
-void VadTest::SetUp() {}
-
-void VadTest::TearDown() {}
-
-// Returns true if the rate and frame length combination is valid.
-bool VadTest::ValidRatesAndFrameLengths(int16_t rate, int16_t frame_length) {
- if (rate == 8000) {
- if (frame_length == 80 || frame_length == 160 || frame_length == 240) {
- return true;
- }
- return false;
- } else if (rate == 16000) {
- if (frame_length == 160 || frame_length == 320 || frame_length == 480) {
- return true;
- }
- return false;
- }
- if (rate == 32000) {
- if (frame_length == 320 || frame_length == 640 || frame_length == 960) {
- return true;
- }
- return false;
- }
-
- return false;
-}
-
-namespace {
-
-TEST_F(VadTest, ApiTest) {
- // This API test runs through the APIs for all possible valid and invalid
- // combinations.
-
- VadInst* handle = NULL;
- int16_t zeros[kMaxFrameLength] = { 0 };
-
- // Construct a speech signal that will trigger the VAD in all modes. It is
- // known that (i * i) will wrap around, but that doesn't matter in this case.
- int16_t speech[kMaxFrameLength];
- for (int16_t i = 0; i < kMaxFrameLength; i++) {
- speech[i] = (i * i);
- }
-
- // Null instance tests
- EXPECT_EQ(-1, WebRtcVad_Create(NULL));
- EXPECT_EQ(-1, WebRtcVad_Init(NULL));
- EXPECT_EQ(-1, WebRtcVad_Assign(NULL, NULL));
- EXPECT_EQ(-1, WebRtcVad_Free(NULL));
- EXPECT_EQ(-1, WebRtcVad_set_mode(NULL, kModes[0]));
- EXPECT_EQ(-1, WebRtcVad_Process(NULL, kRates[0], speech, kFrameLengths[0]));
-
- // WebRtcVad_AssignSize tests
- int handle_size_bytes = 0;
- EXPECT_EQ(0, WebRtcVad_AssignSize(&handle_size_bytes));
- EXPECT_EQ(576, handle_size_bytes);
-
- // WebRtcVad_Assign tests
- void* tmp_handle = malloc(handle_size_bytes);
- EXPECT_EQ(-1, WebRtcVad_Assign(&handle, NULL));
- EXPECT_EQ(0, WebRtcVad_Assign(&handle, tmp_handle));
- EXPECT_EQ(handle, tmp_handle);
- free(tmp_handle);
-
- // WebRtcVad_Create()
- ASSERT_EQ(0, WebRtcVad_Create(&handle));
-
- // Not initialized tests
- EXPECT_EQ(-1, WebRtcVad_Process(handle, kRates[0], speech, kFrameLengths[0]));
- EXPECT_EQ(-1, WebRtcVad_set_mode(handle, kModes[0]));
-
- // WebRtcVad_Init() test
- ASSERT_EQ(0, WebRtcVad_Init(handle));
-
- // WebRtcVad_set_mode() invalid modes tests
- EXPECT_EQ(-1, WebRtcVad_set_mode(handle, kModes[0] - 1));
- EXPECT_EQ(-1, WebRtcVad_set_mode(handle, kModes[kModesSize - 1] + 1));
-
- // WebRtcVad_Process() tests
- // NULL speech pointer
- EXPECT_EQ(-1, WebRtcVad_Process(handle, kRates[0], NULL, kFrameLengths[0]));
- // Invalid sampling rate
- EXPECT_EQ(-1, WebRtcVad_Process(handle, 9999, speech, kFrameLengths[0]));
- // All zeros as input should work
- EXPECT_EQ(0, WebRtcVad_Process(handle, kRates[0], zeros, kFrameLengths[0]));
- for (size_t k = 0; k < kModesSize; k++) {
- // Test valid modes
- EXPECT_EQ(0, WebRtcVad_set_mode(handle, kModes[k]));
- // Loop through sampling rate and frame length combinations
- for (size_t i = 0; i < kRatesSize; i++) {
- for (size_t j = 0; j < kFrameLengthsSize; j++) {
- if (ValidRatesAndFrameLengths(kRates[i], kFrameLengths[j])) {
- EXPECT_EQ(1, WebRtcVad_Process(handle,
- kRates[i],
- speech,
- kFrameLengths[j]));
- } else {
- EXPECT_EQ(-1, WebRtcVad_Process(handle,
- kRates[i],
- speech,
- kFrameLengths[j]));
- }
- }
- }
- }
-
- EXPECT_EQ(0, WebRtcVad_Free(handle));
-}
-
-// TODO(bjornv): Add a process test, run on file.
-
-} // namespace
diff --git a/trunk/src/common_audio/vad/vad_unittest.h b/trunk/src/common_audio/vad/vad_unittest.h
deleted file mode 100644
index f68427c..0000000
--- a/trunk/src/common_audio/vad/vad_unittest.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_COMMONT_AUDIO_VAD_VAD_UNIT_TESTS_H
-#define WEBRTC_COMMONT_AUDIO_VAD_VAD_UNIT_TESTS_H
-
-#include <stddef.h> // size_t
-
-#include "gtest/gtest.h"
-#include "typedefs.h"
-
-namespace {
-
-// Modes we support
-const int kModes[] = { 0, 1, 2, 3 };
-const size_t kModesSize = sizeof(kModes) / sizeof(*kModes);
-
-// Rates we support.
-const int16_t kRates[] = { 8000, 12000, 16000, 24000, 32000 };
-const size_t kRatesSize = sizeof(kRates) / sizeof(*kRates);
-
-// Frame lengths we support.
-const int16_t kMaxFrameLength = 960;
-const int16_t kFrameLengths[] = { 80, 120, 160, 240, 320, 480, 640,
- kMaxFrameLength };
-const size_t kFrameLengthsSize = sizeof(kFrameLengths) / sizeof(*kFrameLengths);
-
-} // namespace
-
-class VadTest : public ::testing::Test {
- protected:
- VadTest();
- virtual void SetUp();
- virtual void TearDown();
-
- // Returns true if the rate and frame length combination is valid.
- bool ValidRatesAndFrameLengths(int16_t rate, int16_t frame_length);
-};
-
-#endif // WEBRTC_COMMONT_AUDIO_VAD_VAD_UNIT_TESTS_H
diff --git a/trunk/src/common_audio/vad/webrtc_vad.c b/trunk/src/common_audio/vad/webrtc_vad.c
deleted file mode 100644
index 26941e6..0000000
--- a/trunk/src/common_audio/vad/webrtc_vad.c
+++ /dev/null
@@ -1,165 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes the VAD API calls. For a specific function call description,
- * see webrtc_vad.h
- */
-
-#include <stdlib.h>
-#include <string.h>
-
-#include "webrtc_vad.h"
-#include "vad_core.h"
-
-static const int kInitCheck = 42;
-
-WebRtc_Word16 WebRtcVad_AssignSize(int *size_in_bytes)
-{
- *size_in_bytes = sizeof(VadInstT) * 2 / sizeof(WebRtc_Word16);
- return 0;
-}
-
-WebRtc_Word16 WebRtcVad_Assign(VadInst **vad_inst, void *vad_inst_addr)
-{
-
- if (vad_inst == NULL)
- {
- return -1;
- }
-
- if (vad_inst_addr != NULL)
- {
- *vad_inst = (VadInst*)vad_inst_addr;
- return 0;
- } else
- {
- return -1;
- }
-}
-
-int WebRtcVad_Create(VadInst** handle) {
- VadInstT* self = NULL;
-
- if (handle == NULL) {
- return -1;
- }
-
- *handle = NULL;
- self = (VadInstT*) malloc(sizeof(VadInstT));
- *handle = (VadInst*) self;
-
- if (self == NULL) {
- return -1;
- }
-
- self->init_flag = 0;
-
- return 0;
-}
-
-int WebRtcVad_Free(VadInst* handle) {
- if (handle == NULL) {
- return -1;
- }
-
- free(handle);
-
- return 0;
-}
-
-int WebRtcVad_Init(VadInst* handle) {
- // Initialize the core VAD component.
- return WebRtcVad_InitCore((VadInstT*) handle);
-}
-
-int WebRtcVad_set_mode(VadInst *vad_inst, int mode)
-{
- VadInstT* vad_ptr;
-
- if (vad_inst == NULL)
- {
- return -1;
- }
-
- vad_ptr = (VadInstT*)vad_inst;
- if (vad_ptr->init_flag != kInitCheck)
- {
- return -1;
- }
-
- return WebRtcVad_set_mode_core((VadInstT*)vad_inst, mode);
-}
-
-WebRtc_Word16 WebRtcVad_Process(VadInst *vad_inst,
- WebRtc_Word16 fs,
- WebRtc_Word16 *speech_frame,
- WebRtc_Word16 frame_length)
-{
- WebRtc_Word16 vad;
- VadInstT* vad_ptr;
-
- if (vad_inst == NULL)
- {
- return -1;
- }
-
- vad_ptr = (VadInstT*)vad_inst;
- if (vad_ptr->init_flag != kInitCheck)
- {
- return -1;
- }
-
- if (speech_frame == NULL)
- {
- return -1;
- }
-
- if (fs == 32000)
- {
- if ((frame_length != 320) && (frame_length != 640) && (frame_length != 960))
- {
- return -1;
- }
- vad = WebRtcVad_CalcVad32khz((VadInstT*)vad_inst, speech_frame, frame_length);
-
- } else if (fs == 16000)
- {
- if ((frame_length != 160) && (frame_length != 320) && (frame_length != 480))
- {
- return -1;
- }
- vad = WebRtcVad_CalcVad16khz((VadInstT*)vad_inst, speech_frame, frame_length);
-
- } else if (fs == 8000)
- {
- if ((frame_length != 80) && (frame_length != 160) && (frame_length != 240))
- {
- return -1;
- }
- vad = WebRtcVad_CalcVad8khz((VadInstT*)vad_inst, speech_frame, frame_length);
-
- } else
- {
- return -1; // Not a supported sampling frequency
- }
-
- if (vad > 0)
- {
- return 1;
- } else if (vad == 0)
- {
- return 0;
- } else
- {
- return -1;
- }
-}
diff --git a/trunk/src/common_types.h b/trunk/src/common_types.h
deleted file mode 100644
index 9b3a0ce..0000000
--- a/trunk/src/common_types.h
+++ /dev/null
@@ -1,573 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_COMMON_TYPES_H
-#define WEBRTC_COMMON_TYPES_H
-
-#include "typedefs.h"
-
-#ifdef WEBRTC_EXPORT
- #define WEBRTC_DLLEXPORT _declspec(dllexport)
-#elif WEBRTC_DLL
- #define WEBRTC_DLLEXPORT _declspec(dllimport)
-#else
- #define WEBRTC_DLLEXPORT
-#endif
-
-#ifndef NULL
- #define NULL 0
-#endif
-
-#define RTP_PAYLOAD_NAME_SIZE 32
-
-namespace webrtc {
-
-class InStream
-{
-public:
- virtual int Read(void *buf,int len) = 0;
- virtual int Rewind() {return -1;}
- virtual ~InStream() {}
-protected:
- InStream() {}
-};
-
-class OutStream
-{
-public:
- virtual bool Write(const void *buf,int len) = 0;
- virtual int Rewind() {return -1;}
- virtual ~OutStream() {}
-protected:
- OutStream() {}
-};
-
-enum TraceModule
-{
- // not a module, triggered from the engine code
- kTraceVoice = 0x0001,
- // not a module, triggered from the engine code
- kTraceVideo = 0x0002,
- // not a module, triggered from the utility code
- kTraceUtility = 0x0003,
- kTraceRtpRtcp = 0x0004,
- kTraceTransport = 0x0005,
- kTraceSrtp = 0x0006,
- kTraceAudioCoding = 0x0007,
- kTraceAudioMixerServer = 0x0008,
- kTraceAudioMixerClient = 0x0009,
- kTraceFile = 0x000a,
- kTraceAudioProcessing = 0x000b,
- kTraceVideoCoding = 0x0010,
- kTraceVideoMixer = 0x0011,
- kTraceAudioDevice = 0x0012,
- kTraceVideoRenderer = 0x0014,
- kTraceVideoCapture = 0x0015,
- kTraceVideoPreocessing = 0x0016
-};
-
-enum TraceLevel
-{
- kTraceNone = 0x0000, // no trace
- kTraceStateInfo = 0x0001,
- kTraceWarning = 0x0002,
- kTraceError = 0x0004,
- kTraceCritical = 0x0008,
- kTraceApiCall = 0x0010,
- kTraceDefault = 0x00ff,
-
- kTraceModuleCall = 0x0020,
- kTraceMemory = 0x0100, // memory info
- kTraceTimer = 0x0200, // timing info
- kTraceStream = 0x0400, // "continuous" stream of data
-
- // used for debug purposes
- kTraceDebug = 0x0800, // debug
- kTraceInfo = 0x1000, // debug info
-
- kTraceAll = 0xffff
-};
-
-// External Trace API
-class TraceCallback
-{
-public:
- virtual void Print(const TraceLevel level,
- const char *traceString,
- const int length) = 0;
-protected:
- virtual ~TraceCallback() {}
- TraceCallback() {}
-};
-
-
-enum FileFormats
-{
- kFileFormatWavFile = 1,
- kFileFormatCompressedFile = 2,
- kFileFormatAviFile = 3,
- kFileFormatPreencodedFile = 4,
- kFileFormatPcm16kHzFile = 7,
- kFileFormatPcm8kHzFile = 8,
- kFileFormatPcm32kHzFile = 9
-};
-
-
-enum ProcessingTypes
-{
- kPlaybackPerChannel = 0,
- kPlaybackAllChannelsMixed,
- kRecordingPerChannel,
- kRecordingAllChannelsMixed
-};
-
-// Encryption enums
-enum CipherTypes
-{
- kCipherNull = 0,
- kCipherAes128CounterMode = 1
-};
-
-enum AuthenticationTypes
-{
- kAuthNull = 0,
- kAuthHmacSha1 = 3
-};
-
-enum SecurityLevels
-{
- kNoProtection = 0,
- kEncryption = 1,
- kAuthentication = 2,
- kEncryptionAndAuthentication = 3
-};
-
-// Interface for encrypting and decrypting regular data and rtp/rtcp packets.
-// Implement this interface if you wish to provide an encryption scheme to
-// the voice or video engines.
-class Encryption
-{
-public:
- // Encrypt the given data.
- //
- // Args:
- // channel: The channel to encrypt data for.
- // in_data: The data to encrypt. This data is bytes_in bytes long.
- // out_data: The buffer to write the encrypted data to. You may write more
- // bytes of encrypted data than what you got as input, up to a maximum
- // of webrtc::kViEMaxMtu if you are encrypting in the video engine, or
- // webrtc::kVoiceEngineMaxIpPacketSizeBytes for the voice engine.
- // bytes_in: The number of bytes in the input buffer.
- // bytes_out: The number of bytes written in out_data.
- virtual void encrypt(
- int channel,
- unsigned char* in_data,
- unsigned char* out_data,
- int bytes_in,
- int* bytes_out) = 0;
-
- // Decrypts the given data. This should reverse the effects of encrypt().
- //
- // Args:
- // channel_no: The channel to decrypt data for.
- // in_data: The data to decrypt. This data is bytes_in bytes long.
- // out_data: The buffer to write the decrypted data to. You may write more
- // bytes of decrypted data than what you got as input, up to a maximum
- // of webrtc::kViEMaxMtu if you are encrypting in the video engine, or
- // webrtc::kVoiceEngineMaxIpPacketSizeBytes for the voice engine.
- // bytes_in: The number of bytes in the input buffer.
- // bytes_out: The number of bytes written in out_data.
- virtual void decrypt(
- int channel,
- unsigned char* in_data,
- unsigned char* out_data,
- int bytes_in,
- int* bytes_out) = 0;
-
- // Encrypts a RTCP packet. Otherwise, this method has the same contract as
- // encrypt().
- virtual void encrypt_rtcp(
- int channel,
- unsigned char* in_data,
- unsigned char* out_data,
- int bytes_in,
- int* bytes_out) = 0;
-
- // Decrypts a RTCP packet. Otherwise, this method has the same contract as
- // decrypt().
- virtual void decrypt_rtcp(
- int channel,
- unsigned char* in_data,
- unsigned char* out_data,
- int bytes_in,
- int* bytes_out) = 0;
-
-protected:
- virtual ~Encryption() {}
- Encryption() {}
-};
-
-// External transport callback interface
-class Transport
-{
-public:
- virtual int SendPacket(int channel, const void *data, int len) = 0;
- virtual int SendRTCPPacket(int channel, const void *data, int len) = 0;
-
-protected:
- virtual ~Transport() {}
- Transport() {}
-};
-
-// ==================================================================
-// Voice specific types
-// ==================================================================
-
-// Each codec supported can be described by this structure.
-struct CodecInst
-{
- int pltype;
- char plname[RTP_PAYLOAD_NAME_SIZE];
- int plfreq;
- int pacsize;
- int channels;
- int rate;
-};
-
-enum FrameType
-{
- kFrameEmpty = 0,
- kAudioFrameSpeech = 1,
- kAudioFrameCN = 2,
- kVideoFrameKey = 3, // independent frame
- kVideoFrameDelta = 4, // depends on the previus frame
- kVideoFrameGolden = 5, // depends on a old known previus frame
- kVideoFrameAltRef = 6
-};
-
-// RTP
-enum {kRtpCsrcSize = 15}; // RFC 3550 page 13
-
-enum RTPDirections
-{
- kRtpIncoming = 0,
- kRtpOutgoing
-};
-
-enum PayloadFrequencies
-{
- kFreq8000Hz = 8000,
- kFreq16000Hz = 16000,
- kFreq32000Hz = 32000
-};
-
-enum VadModes // degree of bandwidth reduction
-{
- kVadConventional = 0, // lowest reduction
- kVadAggressiveLow,
- kVadAggressiveMid,
- kVadAggressiveHigh // highest reduction
-};
-
-struct NetworkStatistics // NETEQ statistics
-{
- // current jitter buffer size in ms
- WebRtc_UWord16 currentBufferSize;
- // preferred (optimal) buffer size in ms
- WebRtc_UWord16 preferredBufferSize;
- // adding extra delay due to "peaky jitter"
- bool jitterPeaksFound;
- // loss rate (network + late) in percent (in Q14)
- WebRtc_UWord16 currentPacketLossRate;
- // late loss rate in percent (in Q14)
- WebRtc_UWord16 currentDiscardRate;
- // fraction (of original stream) of synthesized speech inserted through
- // expansion (in Q14)
- WebRtc_UWord16 currentExpandRate;
- // fraction of synthesized speech inserted through pre-emptive expansion
- // (in Q14)
- WebRtc_UWord16 currentPreemptiveRate;
- // fraction of data removed through acceleration (in Q14)
- WebRtc_UWord16 currentAccelerateRate;
- // clock-drift in parts-per-million (negative or positive)
- int32_t clockDriftPPM;
- // average packet waiting time in the jitter buffer (ms)
- int meanWaitingTimeMs;
- // median packet waiting time in the jitter buffer (ms)
- int medianWaitingTimeMs;
- // min packet waiting time in the jitter buffer (ms)
- int minWaitingTimeMs;
- // max packet waiting time in the jitter buffer (ms)
- int maxWaitingTimeMs;
-};
-
-typedef struct
-{
- int min; // minumum
- int max; // maximum
- int average; // average
-} StatVal;
-
-typedef struct // All levels are reported in dBm0
-{
- StatVal speech_rx; // long-term speech levels on receiving side
- StatVal speech_tx; // long-term speech levels on transmitting side
- StatVal noise_rx; // long-term noise/silence levels on receiving side
- StatVal noise_tx; // long-term noise/silence levels on transmitting side
-} LevelStatistics;
-
-typedef struct // All levels are reported in dB
-{
- StatVal erl; // Echo Return Loss
- StatVal erle; // Echo Return Loss Enhancement
- StatVal rerl; // RERL = ERL + ERLE
- // Echo suppression inside EC at the point just before its NLP
- StatVal a_nlp;
-} EchoStatistics;
-
-enum TelephoneEventDetectionMethods
-{
- kInBand = 0,
- kOutOfBand = 1,
- kInAndOutOfBand = 2
-};
-
-enum NsModes // type of Noise Suppression
-{
- kNsUnchanged = 0, // previously set mode
- kNsDefault, // platform default
- kNsConference, // conferencing default
- kNsLowSuppression, // lowest suppression
- kNsModerateSuppression,
- kNsHighSuppression,
- kNsVeryHighSuppression, // highest suppression
-};
-
-enum AgcModes // type of Automatic Gain Control
-{
- kAgcUnchanged = 0, // previously set mode
- kAgcDefault, // platform default
- // adaptive mode for use when analog volume control exists (e.g. for
- // PC softphone)
- kAgcAdaptiveAnalog,
- // scaling takes place in the digital domain (e.g. for conference servers
- // and embedded devices)
- kAgcAdaptiveDigital,
- // can be used on embedded devices where the the capture signal is level
- // is predictable
- kAgcFixedDigital
-};
-
-// EC modes
-enum EcModes // type of Echo Control
-{
- kEcUnchanged = 0, // previously set mode
- kEcDefault, // platform default
- kEcConference, // conferencing default (aggressive AEC)
- kEcAec, // Acoustic Echo Cancellation
- kEcAecm, // AEC mobile
-};
-
-// AECM modes
-enum AecmModes // mode of AECM
-{
- kAecmQuietEarpieceOrHeadset = 0,
- // Quiet earpiece or headset use
- kAecmEarpiece, // most earpiece use
- kAecmLoudEarpiece, // Loud earpiece or quiet speakerphone use
- kAecmSpeakerphone, // most speakerphone use (default)
- kAecmLoudSpeakerphone // Loud speakerphone
-};
-
-// AGC configuration
-typedef struct
-{
- unsigned short targetLeveldBOv;
- unsigned short digitalCompressionGaindB;
- bool limiterEnable;
-} AgcConfig; // AGC configuration parameters
-
-enum StereoChannel
-{
- kStereoLeft = 0,
- kStereoRight,
- kStereoBoth
-};
-
-// Audio device layers
-enum AudioLayers
-{
- kAudioPlatformDefault = 0,
- kAudioWindowsWave = 1,
- kAudioWindowsCore = 2,
- kAudioLinuxAlsa = 3,
- kAudioLinuxPulse = 4
-};
-
-enum NetEqModes // NetEQ playout configurations
-{
- // Optimized trade-off between low delay and jitter robustness for two-way
- // communication.
- kNetEqDefault = 0,
- // Improved jitter robustness at the cost of increased delay. Can be
- // used in one-way communication.
- kNetEqStreaming = 1,
- // Optimzed for decodability of fax signals rather than for perceived audio
- // quality.
- kNetEqFax = 2,
-};
-
-enum NetEqBgnModes // NetEQ Background Noise (BGN) configurations
-{
- // BGN is always on and will be generated when the incoming RTP stream
- // stops (default).
- kBgnOn = 0,
- // The BGN is faded to zero (complete silence) after a few seconds.
- kBgnFade = 1,
- // BGN is not used at all. Silence is produced after speech extrapolation
- // has faded.
- kBgnOff = 2,
-};
-
-enum OnHoldModes // On Hold direction
-{
- kHoldSendAndPlay = 0, // Put both sending and playing in on-hold state.
- kHoldSendOnly, // Put only sending in on-hold state.
- kHoldPlayOnly // Put only playing in on-hold state.
-};
-
-enum AmrMode
-{
- kRfc3267BwEfficient = 0,
- kRfc3267OctetAligned = 1,
- kRfc3267FileStorage = 2,
-};
-
-// ==================================================================
-// Video specific types
-// ==================================================================
-
-// Raw video types
-enum RawVideoType
-{
- kVideoI420 = 0,
- kVideoYV12 = 1,
- kVideoYUY2 = 2,
- kVideoUYVY = 3,
- kVideoIYUV = 4,
- kVideoARGB = 5,
- kVideoRGB24 = 6,
- kVideoRGB565 = 7,
- kVideoARGB4444 = 8,
- kVideoARGB1555 = 9,
- kVideoMJPEG = 10,
- kVideoNV12 = 11,
- kVideoNV21 = 12,
- kVideoBGRA = 13,
- kVideoUnknown = 99
-};
-
-// Video codec
-enum { kConfigParameterSize = 128};
-enum { kPayloadNameSize = 32};
-enum { kMaxSimulcastStreams = 4};
-enum { kMaxTemporalStreams = 4};
-
-enum VideoCodecComplexity
-{
- kComplexityNormal = 0,
- kComplexityHigh = 1,
- kComplexityHigher = 2,
- kComplexityMax = 3
-};
-
-enum VideoCodecProfile
-{
- kProfileBase = 0x00,
- kProfileMain = 0x01
-};
-
-enum VP8ResilienceMode {
- kResilienceOff, // The stream produced by the encoder requires a
- // recovery frame (typically a key frame) to be
- // decodable after a packet loss.
- kResilientStream, // A stream produced by the encoder is resilient to
- // packet losses, but packets within a frame subsequent
- // to a loss can't be decoded.
- kResilientFrames // Same as kResilientStream but with added resilience
- // within a frame.
-};
-
-// VP8 specific
-struct VideoCodecVP8
-{
- bool pictureLossIndicationOn;
- bool feedbackModeOn;
- VideoCodecComplexity complexity;
- VP8ResilienceMode resilience;
- unsigned char numberOfTemporalLayers;
-};
-
-// Unknown specific
-struct VideoCodecGeneric
-{
-};
-
-// Video codec types
-enum VideoCodecType
-{
- kVideoCodecVP8,
- kVideoCodecI420,
- kVideoCodecRED,
- kVideoCodecULPFEC,
- kVideoCodecUnknown
-};
-
-union VideoCodecUnion
-{
- VideoCodecVP8 VP8;
- VideoCodecGeneric Generic;
-};
-
-
-// Simulcast is when the same stream is encoded multiple times with different
-// settings such as resolution.
-struct SimulcastStream
-{
- unsigned short width;
- unsigned short height;
- unsigned char numberOfTemporalLayers;
- unsigned int maxBitrate;
- unsigned int qpMax; // minimum quality
-};
-
-// Common video codec properties
-struct VideoCodec
-{
- VideoCodecType codecType;
- char plName[kPayloadNameSize];
- unsigned char plType;
-
- unsigned short width;
- unsigned short height;
-
- unsigned int startBitrate;
- unsigned int maxBitrate;
- unsigned int minBitrate;
- unsigned char maxFramerate;
-
- VideoCodecUnion codecSpecific;
-
- unsigned int qpMax;
- unsigned char numberOfSimulcastStreams;
- SimulcastStream simulcastStream[kMaxSimulcastStreams];
-};
-} // namespace webrtc
-#endif // WEBRTC_COMMON_TYPES_H
diff --git a/trunk/src/common_video/OWNERS b/trunk/src/common_video/OWNERS
deleted file mode 100644
index 7183cf2..0000000
--- a/trunk/src/common_video/OWNERS
+++ /dev/null
@@ -1,4 +0,0 @@
-stefan@webrtc.org
-mikhal@webrtc.org
-marpan@webrtc.org
-henrik.lundin@webrtc.org
diff --git a/trunk/src/common_video/common_video.gyp b/trunk/src/common_video/common_video.gyp
deleted file mode 100644
index 3adb56c..0000000
--- a/trunk/src/common_video/common_video.gyp
+++ /dev/null
@@ -1,15 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'includes': [
- '../build/common.gypi',
- 'libyuv/libyuv.gypi',
- 'jpeg/jpeg.gypi',
- ],
-}
diff --git a/trunk/src/common_video/interface/video_image.h b/trunk/src/common_video/interface/video_image.h
deleted file mode 100644
index 82e7b15..0000000
--- a/trunk/src/common_video/interface/video_image.h
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef COMMON_VIDEO_INTERFACE_VIDEO_IMAGE_H
-#define COMMON_VIDEO_INTERFACE_VIDEO_IMAGE_H
-
-#include "typedefs.h"
-#include <stdlib.h>
-
-namespace webrtc
-{
-
-enum VideoFrameType
-{
- kKeyFrame = 0,
- kDeltaFrame = 1,
- kGoldenFrame = 2,
- kAltRefFrame = 3,
- kSkipFrame = 4
-};
-
-class RawImage
-{
-public:
- RawImage() : _width(0), _height(0), _timeStamp(0), _buffer(NULL),
- _length(0), _size(0) {}
-
- RawImage(WebRtc_UWord8* buffer, WebRtc_UWord32 length,
- WebRtc_UWord32 size) :
- _width(0), _height(0), _timeStamp(0),
- _buffer(buffer), _length(length), _size(size) {}
-
- WebRtc_UWord32 _width;
- WebRtc_UWord32 _height;
- WebRtc_UWord32 _timeStamp;
- WebRtc_UWord8* _buffer;
- WebRtc_UWord32 _length;
- WebRtc_UWord32 _size;
-};
-
-class EncodedImage
-{
-public:
- EncodedImage() :
- _encodedWidth(0), _encodedHeight(0), _timeStamp(0),
- _frameType(kDeltaFrame), _buffer(NULL), _length(0),
- _size(0), _completeFrame(false) {}
-
- EncodedImage(WebRtc_UWord8* buffer,
- WebRtc_UWord32 length,
- WebRtc_UWord32 size) :
- _encodedWidth(0), _encodedHeight(0), _timeStamp(0),
- _frameType(kDeltaFrame), _buffer(buffer), _length(length),
- _size(size), _completeFrame(false) {}
-
- WebRtc_UWord32 _encodedWidth;
- WebRtc_UWord32 _encodedHeight;
- WebRtc_UWord32 _timeStamp;
- VideoFrameType _frameType;
- WebRtc_UWord8* _buffer;
- WebRtc_UWord32 _length;
- WebRtc_UWord32 _size;
- bool _completeFrame;
-};
-
-} // namespace webrtc
-
-#endif // COMMON_VIDEO_INTERFACE_VIDEO_IMAGE_H
diff --git a/trunk/src/common_video/jpeg/Android.mk b/trunk/src/common_video/jpeg/Android.mk
deleted file mode 100644
index b5bc243..0000000
--- a/trunk/src/common_video/jpeg/Android.mk
+++ /dev/null
@@ -1,42 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../android-webrtc.mk
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_jpeg
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := \
- jpeg.cc \
- data_manager.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include \
- $(LOCAL_PATH)/../../ \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../../../ \
- external/jpeg
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/common_video/jpeg/data_manager.cc b/trunk/src/common_video/jpeg/data_manager.cc
deleted file mode 100644
index a5a7a48..0000000
--- a/trunk/src/common_video/jpeg/data_manager.cc
+++ /dev/null
@@ -1,86 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "common_video/jpeg/data_manager.h"
-
-namespace webrtc
-{
-
-typedef struct
-{
- jpeg_source_mgr mgr;
- JOCTET* next_input_byte;
- size_t bytes_in_buffer; /* # of byte spaces remaining in buffer */
-} DataSrcMgr;
-
-void
-jpegSetSrcBuffer(j_decompress_ptr cinfo, JOCTET* srcBuffer, size_t bufferSize)
-{
- DataSrcMgr* src;
- if (cinfo->src == NULL)
- { /* first time for this JPEG object? */
- cinfo->src = (struct jpeg_source_mgr *)
- (*cinfo->mem->alloc_small) ((j_common_ptr) cinfo,
- JPOOL_PERMANENT, sizeof(DataSrcMgr));
- }
-
- // Setting required functionality
- src = (DataSrcMgr*) cinfo->src;
- src->mgr.init_source = initSrc;;
- src->mgr.fill_input_buffer = fillInputBuffer;
- src->mgr.skip_input_data = skipInputData;
- src->mgr.resync_to_restart = jpeg_resync_to_restart; // use default
- src->mgr.term_source = termSource;
- // setting buffer/src
- src->bytes_in_buffer = bufferSize;
- src->next_input_byte = srcBuffer;
-
-}
-
-
-void
-initSrc(j_decompress_ptr cinfo)
-{
- DataSrcMgr *src = (DataSrcMgr*)cinfo->src;
- src->mgr.next_input_byte = src->next_input_byte;
- src->mgr.bytes_in_buffer = src->bytes_in_buffer;
-}
-
-boolean
-fillInputBuffer(j_decompress_ptr cinfo)
-{
- return false;
-}
-
-
-void
-skipInputData(j_decompress_ptr cinfo, long num_bytes)
-{
- DataSrcMgr* src = (DataSrcMgr*)cinfo->src;
- if (num_bytes > 0)
- {
- if ((unsigned long)num_bytes > src->mgr.bytes_in_buffer)
- src->mgr.bytes_in_buffer = 0;
- else
- {
- src->mgr.next_input_byte += num_bytes;
- src->mgr.bytes_in_buffer -= num_bytes;
- }
- }
-}
-
-
-void
-termSource (j_decompress_ptr cinfo)
-{
- //
-}
-
-} // end of namespace webrtc
diff --git a/trunk/src/common_video/jpeg/data_manager.h b/trunk/src/common_video/jpeg/data_manager.h
deleted file mode 100644
index 61609ec..0000000
--- a/trunk/src/common_video/jpeg/data_manager.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Jpeg source data manager
- */
-
-#ifndef WEBRTC_COMMON_VIDEO_JPEG_DATA_MANAGER
-#define WEBRTC_COMMON_VIDEO_JPEG_DATA_MANAGER
-
-#include <stdio.h>
-extern "C" {
-#if defined(USE_SYSTEM_LIBJPEG)
-#include <jpeglib.h>
-#else
-#include "jpeglib.h"
-#endif
-}
-
-namespace webrtc
-{
-
-// Source manager:
-
-
-// a general function that will set these values
-void
-jpegSetSrcBuffer(j_decompress_ptr cinfo, JOCTET* srcBuffer, size_t bufferSize);
-
-
-// Initialize source. This is called by jpeg_read_header() before any
-// data is actually read.
-
-void
-initSrc(j_decompress_ptr cinfo);
-
-
-// Fill input buffer
-// This is called whenever bytes_in_buffer has reached zero and more
-// data is wanted.
-
-boolean
-fillInputBuffer(j_decompress_ptr cinfo);
-
-// Skip input data
-// Skip num_bytes worth of data.
-
-void
-skipInputData(j_decompress_ptr cinfo, long num_bytes);
-
-
-
-
-// Terminate source
-void
-termSource (j_decompress_ptr cinfo);
-
-} // end of namespace webrtc
-
-
-#endif /* WEBRTC_COMMON_VIDEO_JPEG_DATA_MANAGER */
diff --git a/trunk/src/common_video/jpeg/include/jpeg.h b/trunk/src/common_video/jpeg/include/jpeg.h
deleted file mode 100644
index 05e759c..0000000
--- a/trunk/src/common_video/jpeg/include/jpeg.h
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_COMMON_VIDEO_JPEG
-#define WEBRTC_COMMON_VIDEO_JPEG
-
-#include "typedefs.h"
-#include "video_image.h"
-
-// jpeg forward declaration
-struct jpeg_compress_struct;
-struct jpeg_decompress_struct;
-
-namespace webrtc
-{
-
-class JpegEncoder
-{
-public:
- JpegEncoder();
- ~JpegEncoder();
-
-// SetFileName
-// Input:
-// - fileName - Pointer to input vector (should be less than 256) to which the
-// compressed file will be written to
-// Output:
-// - 0 : OK
-// - (-1) : Error
- WebRtc_Word32 SetFileName(const char* fileName);
-
-// Encode an I420 image. The encoded image is saved to a file
-//
-// Input:
-// - inputImage : Image to be encoded
-//
-// Output:
-// - 0 : OK
-// - (-1) : Error
- WebRtc_Word32 Encode(const RawImage& inputImage);
-
-private:
-
- jpeg_compress_struct* _cinfo;
- char _fileName[256];
-};
-
-class JpegDecoder
-{
- public:
- JpegDecoder();
- ~JpegDecoder();
-
-// Decodes a JPEG-stream
-// Supports 1 image component. 3 interleaved image components,
-// YCbCr sub-sampling 4:4:4, 4:2:2, 4:2:0.
-//
-// Input:
-// - inputImage : encoded image to be decoded.
-// - outputImage : RawImage to store decoded output
-//
-// Output:
-// - 0 : OK
-// - (-1) : Error
- WebRtc_Word32 Decode(const EncodedImage& inputImage,
- RawImage& outputImage);
- private:
- jpeg_decompress_struct* _cinfo;
-};
-
-
-}
-#endif /* WEBRTC_COMMON_VIDEO_JPEG */
diff --git a/trunk/src/common_video/jpeg/jpeg.cc b/trunk/src/common_video/jpeg/jpeg.cc
deleted file mode 100644
index 00586d4..0000000
--- a/trunk/src/common_video/jpeg/jpeg.cc
+++ /dev/null
@@ -1,375 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#if defined(WIN32)
- #include <basetsd.h>
-#endif
-#include <setjmp.h>
-#include <stdio.h>
-#include <string.h>
-
-#include "common_video/jpeg/include/jpeg.h"
-#include "common_video/jpeg/data_manager.h"
-
-extern "C" {
-#if defined(USE_SYSTEM_LIBJPEG)
-#include <jpeglib.h>
-#else
-#include "jpeglib.h"
-#endif
-}
-
-
-namespace webrtc
-{
-
-// Error handler
-struct myErrorMgr {
-
- struct jpeg_error_mgr pub;
- jmp_buf setjmp_buffer;
-};
-typedef struct myErrorMgr * myErrorPtr;
-
-METHODDEF(void)
-MyErrorExit (j_common_ptr cinfo)
-{
- myErrorPtr myerr = (myErrorPtr) cinfo->err;
-
- // Return control to the setjmp point
- longjmp(myerr->setjmp_buffer, 1);
-}
-
-JpegEncoder::JpegEncoder()
-{
- _cinfo = new jpeg_compress_struct;
- strcpy(_fileName, "Snapshot.jpg");
-}
-
-JpegEncoder::~JpegEncoder()
-{
- if (_cinfo != NULL)
- {
- delete _cinfo;
- _cinfo = NULL;
- }
-}
-
-
-WebRtc_Word32
-JpegEncoder::SetFileName(const char* fileName)
-{
- if (!fileName)
- {
- return -1;
- }
-
- if (fileName)
- {
- strncpy(_fileName, fileName, 256);
- }
- return 0;
-}
-
-
-WebRtc_Word32
-JpegEncoder::Encode(const RawImage& inputImage)
-{
- if (inputImage._buffer == NULL || inputImage._size == 0)
- {
- return -1;
- }
- if (inputImage._width < 1 || inputImage._height < 1)
- {
- return -1;
- }
-
- FILE* outFile = NULL;
-
- const WebRtc_UWord32 width = inputImage._width;
- const WebRtc_UWord32 height = inputImage._height;
-
- // Set error handler
- myErrorMgr jerr;
- _cinfo->err = jpeg_std_error(&jerr.pub);
- jerr.pub.error_exit = MyErrorExit;
- // Establish the setjmp return context
- if (setjmp(jerr.setjmp_buffer))
- {
- // If we get here, the JPEG code has signaled an error.
- jpeg_destroy_compress(_cinfo);
- if (outFile != NULL)
- {
- fclose(outFile);
- }
- return -1;
- }
-
- if ((outFile = fopen(_fileName, "wb")) == NULL)
- {
- return -2;
- }
- // Create a compression object
- jpeg_create_compress(_cinfo);
-
- // Setting destination file
- jpeg_stdio_dest(_cinfo, outFile);
-
- // Set parameters for compression
- _cinfo->in_color_space = JCS_YCbCr;
- jpeg_set_defaults(_cinfo);
-
- _cinfo->image_width = width;
- _cinfo->image_height = height;
- _cinfo->input_components = 3;
-
- _cinfo->comp_info[0].h_samp_factor = 2; // Y
- _cinfo->comp_info[0].v_samp_factor = 2;
- _cinfo->comp_info[1].h_samp_factor = 1; // U
- _cinfo->comp_info[1].v_samp_factor = 1;
- _cinfo->comp_info[2].h_samp_factor = 1; // V
- _cinfo->comp_info[2].v_samp_factor = 1;
- _cinfo->raw_data_in = TRUE;
-
- WebRtc_UWord32 height16 = (height + 15) & ~15;
- WebRtc_UWord8* imgPtr = inputImage._buffer;
- WebRtc_UWord8* origImagePtr = NULL;
- if (height16 != height)
- {
- // Copy image to an adequate size buffer
- WebRtc_UWord32 requiredSize = height16 * width * 3 >> 1;
- origImagePtr = new WebRtc_UWord8[requiredSize];
- memset(origImagePtr, 0, requiredSize);
- memcpy(origImagePtr, inputImage._buffer, inputImage._length);
- imgPtr = origImagePtr;
- }
-
- jpeg_start_compress(_cinfo, TRUE);
-
- JSAMPROW y[16],u[8],v[8];
- JSAMPARRAY data[3];
-
- data[0] = y;
- data[1] = u;
- data[2] = v;
-
- WebRtc_UWord32 i, j;
-
- for (j = 0; j < height; j += 16)
- {
- for (i = 0; i < 16; i++)
- {
- y[i] = (JSAMPLE*)imgPtr + width * (i + j);
-
- if (i % 2 == 0)
- {
- u[i / 2] = (JSAMPLE*) imgPtr + width * height +
- width / 2 * ((i + j) / 2);
- v[i / 2] = (JSAMPLE*) imgPtr + width * height +
- width * height / 4 + width / 2 * ((i + j) / 2);
- }
- }
- jpeg_write_raw_data(_cinfo, data, 16);
- }
-
- jpeg_finish_compress(_cinfo);
- jpeg_destroy_compress(_cinfo);
-
- fclose(outFile);
-
- if (origImagePtr != NULL)
- {
- delete [] origImagePtr;
- }
-
- return 0;
-}
-
-JpegDecoder::JpegDecoder()
-{
- _cinfo = new jpeg_decompress_struct;
-}
-
-JpegDecoder::~JpegDecoder()
-{
- if (_cinfo != NULL)
- {
- delete _cinfo;
- _cinfo = NULL;
- }
-}
-
-WebRtc_Word32
-JpegDecoder::Decode(const EncodedImage& inputImage,
- RawImage& outputImage)
-{
-
- WebRtc_UWord8* tmpBuffer = NULL;
- // Set error handler
- myErrorMgr jerr;
- _cinfo->err = jpeg_std_error(&jerr.pub);
- jerr.pub.error_exit = MyErrorExit;
-
- // Establish the setjmp return context
- if (setjmp(jerr.setjmp_buffer))
- {
- if (_cinfo->is_decompressor)
- {
- jpeg_destroy_decompress(_cinfo);
- }
- if (tmpBuffer != NULL)
- {
- delete [] tmpBuffer;
- }
- return -1;
- }
-
- _cinfo->out_color_space = JCS_YCbCr;
-
- // Create decompression object
- jpeg_create_decompress(_cinfo);
-
- // Specify data source
- jpegSetSrcBuffer(_cinfo, (JOCTET*) inputImage._buffer, inputImage._size);
-
- // Read header data
- jpeg_read_header(_cinfo, TRUE);
-
- _cinfo->raw_data_out = TRUE;
- jpeg_start_decompress(_cinfo);
-
- // Check header
- if (_cinfo->num_components == 4)
- {
- return -2; // not supported
- }
- if (_cinfo->progressive_mode == 1)
- {
- return -2; // not supported
- }
-
-
- WebRtc_UWord32 height = _cinfo->image_height;
- WebRtc_UWord32 width = _cinfo->image_width;
-
- // Making sure width and height are even
- if (height % 2)
- {
- height++;
- }
- if (width % 2)
- {
- width++;
- }
-
- WebRtc_UWord32 height16 = (height + 15) & ~15;
- WebRtc_UWord32 stride = (width + 15) & ~15;
- WebRtc_UWord32 uvStride = ((((stride + 1) >> 1) + 15) & ~15);
-
- WebRtc_UWord32 tmpRequiredSize = stride * height16 +
- 2 * (uvStride * ((height16 + 1) >> 1));
- WebRtc_UWord32 requiredSize = width * height * 3 >> 1;
-
- // verify sufficient buffer size
- if (outputImage._buffer && outputImage._size < requiredSize)
- {
- delete [] outputImage._buffer;
- outputImage._buffer = NULL;
- }
-
- if (outputImage._buffer == NULL)
- {
- outputImage._buffer = new WebRtc_UWord8[requiredSize];
- outputImage._size = requiredSize;
- }
-
- WebRtc_UWord8* outPtr = outputImage._buffer;
-
- if (tmpRequiredSize > requiredSize)
- {
- tmpBuffer = new WebRtc_UWord8[(int) (tmpRequiredSize)];
- outPtr = tmpBuffer;
- }
-
- JSAMPROW y[16],u[8],v[8];
- JSAMPARRAY data[3];
- data[0] = y;
- data[1] = u;
- data[2] = v;
-
- WebRtc_UWord32 hInd, i;
- WebRtc_UWord32 numScanLines = 16;
- WebRtc_UWord32 numLinesProcessed = 0;
-
- while (_cinfo->output_scanline < _cinfo->output_height)
- {
- hInd = _cinfo->output_scanline;
- for (i = 0; i < numScanLines; i++)
- {
- y[i] = outPtr + stride * (i + hInd);
-
- if (i % 2 == 0)
- {
- u[i / 2] = outPtr + stride * height16 +
- stride / 2 * ((i + hInd) / 2);
- v[i / 2] = outPtr + stride * height16 +
- stride * height16 / 4 +
- stride / 2 * ((i + hInd) / 2);
- }
- }
- // Processes exactly one iMCU row per call
- numLinesProcessed = jpeg_read_raw_data(_cinfo, data, numScanLines);
- // Error in read
- if (numLinesProcessed == 0)
- {
- jpeg_abort((j_common_ptr)_cinfo);
- return -1;
- }
- }
-
- if (tmpRequiredSize > requiredSize)
- {
- WebRtc_UWord8* dstFramePtr = outputImage._buffer;
- WebRtc_UWord8* tmpPtr = outPtr;
-
- for (WebRtc_UWord32 p = 0; p < 3; p++)
- {
- const WebRtc_UWord32 h = (p == 0) ? height : height >> 1;
- const WebRtc_UWord32 h16 = (p == 0) ? height16 : height16 >> 1;
- const WebRtc_UWord32 w = (p == 0) ? width : width >> 1;
- const WebRtc_UWord32 s = (p == 0) ? stride : stride >> 1;
-
- for (WebRtc_UWord32 i = 0; i < h; i++)
- {
- memcpy(dstFramePtr, tmpPtr, w);
- dstFramePtr += w;
- tmpPtr += s;
- }
- tmpPtr += (h16 - h) * s;
- }
- }
-
- if (tmpBuffer != NULL)
- {
- delete [] tmpBuffer;
- }
- // Setting output Image parameter
- outputImage._width = width;
- outputImage._height = height;
- outputImage._length = requiredSize;
- outputImage._timeStamp = inputImage._timeStamp;
-
- jpeg_finish_decompress(_cinfo);
- jpeg_destroy_decompress(_cinfo);
- return 0;
-}
-
-
-}
diff --git a/trunk/src/common_video/jpeg/jpeg.gypi b/trunk/src/common_video/jpeg/jpeg.gypi
deleted file mode 100644
index 1738a3d..0000000
--- a/trunk/src/common_video/jpeg/jpeg.gypi
+++ /dev/null
@@ -1,90 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'variables': {
- 'use_libjpeg_turbo%': '<(use_libjpeg_turbo)',
- },
- 'targets': [
- {
- 'target_name': 'webrtc_jpeg',
- 'type': '<(library)',
- 'dependencies': [
- 'webrtc_libyuv',
- ],
- 'include_dirs': [
- 'include',
- '<(webrtc_root)',
- '<(webrtc_root)/common_video/interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- '<(webrtc_root)/common_video/interface',
- ],
- },
- 'conditions': [
- ['build_libjpeg==1', {
- 'conditions': [
- ['build_with_chromium==1', {
- 'dependencies': [
- '<(libjpeg_gyp_path):libjpeg',
- ],
- }, {
- 'conditions': [
- ['use_libjpeg_turbo==1', {
- 'dependencies': [
- '<(DEPTH)/third_party/libjpeg_turbo/libjpeg.gyp:libjpeg',
- ],
- }, {
- 'dependencies': [
- '<(DEPTH)/third_party/libjpeg/libjpeg.gyp:libjpeg',
- ],
- }],
- ],
- }],
- ],
- }, {
- # Need to add a directory normally exported by libjpeg.gyp.
- 'include_dirs': [ '<(DEPTH)/third_party/libjpeg', ],
- }],
- ],
- 'sources': [
- 'include/jpeg.h',
- 'data_manager.cc',
- 'data_manager.h',
- 'jpeg.cc',
- ],
- },
- ], # targets
- # Exclude the test target when building with chromium.
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'jpeg_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'webrtc_jpeg',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- ],
- 'sources': [
- 'jpeg_unittest.cc',
- ],
- },
- ] # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/common_video/jpeg/jpeg_unittest.cc b/trunk/src/common_video/jpeg/jpeg_unittest.cc
deleted file mode 100644
index fdcbe34..0000000
--- a/trunk/src/common_video/jpeg/jpeg_unittest.cc
+++ /dev/null
@@ -1,123 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cstdio>
-#include <string>
-
-#include "common_video/jpeg/include/jpeg.h"
-#include "common_video/interface/video_image.h"
-#include "gtest/gtest.h"
-#include "testsupport/fileutils.h"
-
-namespace webrtc {
-
-const unsigned int kImageWidth = 640;
-const unsigned int kImageHeight = 480;
-
-class JpegTest: public testing::Test {
- protected:
- JpegTest()
- : input_filename_(webrtc::test::ProjectRootPath() +
- "test/data/common_video/jpeg/webrtc_logo.jpg"),
- decoded_filename_(webrtc::test::OutputPath() + "TestJpegDec.yuv"),
- encoded_filename_(webrtc::test::OutputPath() + "TestJpegEnc.jpg"),
- encoded_buffer_(NULL) {}
- virtual ~JpegTest() {}
-
- void SetUp() {
- encoder_ = new JpegEncoder();
- decoder_ = new JpegDecoder();
- }
-
- void TearDown() {
- if (encoded_buffer_ != NULL) {
- if (encoded_buffer_->_buffer != NULL) {
- delete [] encoded_buffer_->_buffer;
- }
- delete encoded_buffer_;
- }
- delete encoder_;
- delete decoder_;
- }
-
- // Reads an encoded image. Caller will have to deallocate the memory of this
- // object and it's _buffer byte array.
- EncodedImage* ReadEncodedImage(std::string input_filename) {
- FILE* open_file = fopen(input_filename.c_str(), "rb");
- assert(open_file != NULL);
- size_t length = webrtc::test::GetFileSize(input_filename);
- EncodedImage* encoded_buffer = new EncodedImage();
- encoded_buffer->_buffer = new WebRtc_UWord8[length];
- encoded_buffer->_size = length;
- encoded_buffer->_length = length;
- if (fread(encoded_buffer->_buffer, 1, length, open_file) != length) {
- ADD_FAILURE() << "Error reading file:" << input_filename;
- }
- fclose(open_file);
- return encoded_buffer;
- }
-
- std::string input_filename_;
- std::string decoded_filename_;
- std::string encoded_filename_;
- EncodedImage* encoded_buffer_;
- JpegEncoder* encoder_;
- JpegDecoder* decoder_;
-};
-
-TEST_F(JpegTest, Decode) {
- encoded_buffer_ = ReadEncodedImage(input_filename_);
- RawImage image_buffer;
- EXPECT_EQ(0, decoder_->Decode(*encoded_buffer_, image_buffer));
- EXPECT_GT(image_buffer._length, 0u);
- EXPECT_EQ(kImageWidth, image_buffer._width);
- EXPECT_EQ(kImageHeight, image_buffer._height);
- delete [] image_buffer._buffer;
-}
-
-TEST_F(JpegTest, EncodeInvalidInputs) {
- RawImage empty;
- empty._width = 164;
- empty._height = 164;
- EXPECT_EQ(-1, encoder_->SetFileName(0));
- EXPECT_EQ(-1, encoder_->Encode(empty));
-
- empty._buffer = new WebRtc_UWord8[10];
- empty._size = 0;
- EXPECT_EQ(-1, encoder_->Encode(empty));
-
- empty._size = 10;
- empty._height = 0;
- EXPECT_EQ(-1, encoder_->Encode(empty));
-
- empty._height = 164;
- empty._width = 0;
- EXPECT_EQ(-1, encoder_->Encode(empty));
- delete[] empty._buffer;
-}
-
-TEST_F(JpegTest, Encode) {
- // Decode our input image then encode it again to a new file:
- encoded_buffer_ = ReadEncodedImage(input_filename_);
- RawImage image_buffer;
- EXPECT_EQ(0, decoder_->Decode(*encoded_buffer_, image_buffer));
-
- EXPECT_EQ(0, encoder_->SetFileName(encoded_filename_.c_str()));
- EXPECT_EQ(0, encoder_->Encode(image_buffer));
-
- // Save decoded image to file.
- FILE* save_file = fopen(decoded_filename_.c_str(), "wb");
- fwrite(image_buffer._buffer, 1, image_buffer._length, save_file);
- fclose(save_file);
-
- delete[] image_buffer._buffer;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/common_video/libyuv/Android.mk b/trunk/src/common_video/libyuv/Android.mk
deleted file mode 100644
index 90f6250..0000000
--- a/trunk/src/common_video/libyuv/Android.mk
+++ /dev/null
@@ -1,42 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../android-webrtc.mk
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_yuv
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := \
- libyuv.cc \
- scaler.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../.. \
- $(LOCAL_PATH)/../../../.. \
- $(LOCAL_PATH)/../../../../libyuv/files/include
-
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
\ No newline at end of file
diff --git a/trunk/src/common_video/libyuv/include/libyuv.h b/trunk/src/common_video/libyuv/include/libyuv.h
deleted file mode 100644
index bf106f1..0000000
--- a/trunk/src/common_video/libyuv/include/libyuv.h
+++ /dev/null
@@ -1,194 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * WebRTC's Wrapper to libyuv.
- */
-
-#ifndef WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_LIBYUV_H_
-#define WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_LIBYUV_H_
-
-#include "common_types.h" // RawVideoTypes.
-#include "typedefs.h"
-
-namespace webrtc {
-
-// TODO(mikhal): 1. Sync libyuv and WebRtc meaning of stride.
-// 2. Reorder parameters for consistency.
-
-// Supported video types.
-enum VideoType {
- kUnknown,
- kI420,
- kIYUV,
- kRGB24,
- kABGR,
- kARGB,
- kARGB4444,
- kRGB565,
- kARGB1555,
- kYUY2,
- kYV12,
- kUYVY,
- kMJPG,
- kNV21,
- kNV12,
- kBGRA,
-};
-
-// Conversion between the RawVideoType and the LibYuv videoType.
-// TODO(wu): Consolidate types into one type throughout WebRtc.
-VideoType RawVideoTypeToCommonVideoVideoType(RawVideoType type);
-
-// Supported rotation
-// Direction of rotation - clockwise.
-enum VideoRotationMode {
- kRotateNone = 0,
- kRotate90 = 90,
- kRotate180 = 180,
- kRotate270 = 270,
-};
-
-// Calculate the required buffer size.
-// Input:
-// - type - The type of the designated video frame.
-// - width - frame width in pixels.
-// - height - frame height in pixels.
-// Return value: The required size in bytes to accommodate the specified
-// video frame or -1 in case of an error .
-int CalcBufferSize(VideoType type, int width, int height);
-
-// Compute required buffer size when converting from one type to another.
-// Input:
-// - src_video_type - Type of the existing video frame.
-// - dst_video_type - Type of the designated video frame.
-// - length - length in bytes of the data.
-// Return value: The required size in bytes to accommodate the specified
-// converted video frame or -1 in case of an error.
-int CalcBufferSize(VideoType src_video_type,
- VideoType dst_video_type,
- int length);
-// TODO (mikhal): Merge the two functions above.
-
-
-// Convert To I420
-// Input:
-// - src_video_type : Type of input video.
-// - src_frame : Pointer to a source frame.
-// - crop_x/crop_y : Starting positions for cropping (0 for no crop).
-// - src/dst_width : src/dst width in pixels.
-// - src/dst_height : src/dst height in pixels.
-// - sample_size : Required only for the parsing of MJPG (set to 0 else).
-// - dst_stride : Number of bytes in a row of the dst Y plane.
-// - rotate : Rotation mode of output image.
-// Output:
-// - dst_frame : Pointer to a destination frame.
-// Return value: 0 if OK, < 0 otherwise.
-
-int ConvertToI420(VideoType src_video_type,
- const uint8_t* src_frame,
- int crop_x, int crop_y,
- int src_width, int src_height,
- int sample_size,
- int dst_width, int dst_height, int dst_stride,
- VideoRotationMode rotation,
- uint8_t* dst_frame);
-
-// Convert From I420
-// Input:
-// - src_frame : Pointer to a source frame.
-// - src_stride : Number of bytes in a row of the src Y plane.
-// - dst_video_type : Type of output video.
-// - dst_sample_size : Required only for the parsing of MJPG.
-// - width : Width in pixels.
-// - height : Height in pixels.
-// - dst_frame : Pointer to a destination frame.
-// Return value: 0 if OK, < 0 otherwise.
-int ConvertFromI420(const uint8_t* src_frame, int src_stride,
- VideoType dst_video_type, int dst_sample_size,
- int width, int height,
- uint8_t* dst_frame);
-// ConvertFrom YV12.
-// Interface - same as above.
-int ConvertFromYV12(const uint8_t* src_frame, int src_stride,
- VideoType dst_video_type, int dst_sample_size,
- int width, int height,
- uint8_t* dst_frame);
-
-// The following list describes designated conversion functions which
-// are not covered by the previous general functions.
-// Input and output descriptions mostly match the above descriptions, and are
-// therefore omitted.
-// Possible additional input value - dst_stride - stride of the dst frame.
-
-int ConvertI420ToARGB4444(const uint8_t* src_frame,
- uint8_t* dst_frame,
- int width,
- int height,
- int dst_stride);
-int ConvertI420ToRGB565(const uint8_t* src_frame,
- uint8_t* dst_frame,
- int width,
- int height);
-int ConvertI420ToARGB1555(const uint8_t* src_frame,
- uint8_t* dst_frame,
- int width,
- int height,
- int dst_stride);
-int ConvertRGB24ToARGB(const uint8_t* src_frame,
- uint8_t* dst_frame,
- int width, int height,
- int dst_stride);
-int ConvertNV12ToRGB565(const uint8_t* src_frame,
- uint8_t* dst_frame,
- int width, int height);
-
-// Mirror functions
-// The following 2 functions perform mirroring on a given image
-// (LeftRight/UpDown).
-// Input:
-// - width : Image width in pixels.
-// - height : Image height in pixels.
-// - src_frame : Pointer to a source frame.
-// - dst_frame : Pointer to a destination frame.
-// Return value: 0 if OK, < 0 otherwise.
-int MirrorI420LeftRight(const uint8_t* src_frame,
- uint8_t* dst_frame,
- int width, int height);
-int MirrorI420UpDown(const uint8_t* src_frame,
- uint8_t* dst_frame,
- int width, int height);
-
-// Mirror functions + conversion
-// Input:
-// - src_frame : Pointer to source frame.
-// - dst_frame : Pointer to destination frame.
-// - src_width : Width of input buffer.
-// - src_height : Height of input buffer.
-// - src_color_space : Color space to convert from, I420 if no
-// conversion should be done.
-// Return value: 0 if OK, < 0 otherwise.
-int ConvertToI420AndMirrorUpDown(const uint8_t* src_frame,
- uint8_t* dst_frame,
- int src_width,
- int src_height,
- VideoType src_video_type);
-
-// Compute PSNR for an I420 frame (all planes).
-double I420PSNR(const uint8_t* ref_frame,
- const uint8_t* test_frame,
- int width, int height);
-// Compute SSIM for an I420 frame (all planes).
-double I420SSIM(const uint8_t* ref_frame,
- const uint8_t* test_frame,
- int width, int height);
-}
-
-#endif // WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_LIBYUV_H_
diff --git a/trunk/src/common_video/libyuv/include/scaler.h b/trunk/src/common_video/libyuv/include/scaler.h
deleted file mode 100644
index 8f432f3..0000000
--- a/trunk/src/common_video/libyuv/include/scaler.h
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Interface to the LibYuv scaling functionality
- */
-
-#ifndef WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_SCALER_H_
-#define WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_SCALER_H_
-
-#include "common_video/libyuv/include/libyuv.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-// Supported scaling types
-enum ScaleMethod {
- kScalePoint, // no interpolation
- kScaleBilinear,
- kScaleBox
-};
-
-// TODO (mikhal): Have set return the expected value of the dst_frame, such
-// that the user can allocate memory for Scale().
-class Scaler {
- public:
- Scaler();
- ~Scaler();
-
- // Set interpolation properties:
- //
- // Return value: 0 - OK
- // -1 - parameter error
- int Set(int src_width, int src_height,
- int dst_width, int dst_height,
- VideoType src_video_type, VideoType dst_video_type,
- ScaleMethod method);
-
- // Scale frame
- // Memory is allocated by user. If dst_frame is not of sufficient size,
- // the frame will be reallocated to the appropriate size.
- // Return value: 0 - OK,
- // -1 - parameter error
- // -2 - scaler not set
- int Scale(const uint8_t* src_frame,
- uint8_t*& dst_frame,
- int& dst_size);
-
- private:
- // Determine if the VideoTypes are currently supported.
- bool SupportedVideoType(VideoType src_video_type,
- VideoType dst_video_type);
-
- ScaleMethod method_;
- int src_width_;
- int src_height_;
- int dst_width_;
- int dst_height_;
- bool set_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_COMMON_VIDEO_LIBYUV_INCLUDE_SCALER_H_
diff --git a/trunk/src/common_video/libyuv/libyuv.cc b/trunk/src/common_video/libyuv/libyuv.cc
deleted file mode 100644
index d1ced7a..0000000
--- a/trunk/src/common_video/libyuv/libyuv.cc
+++ /dev/null
@@ -1,450 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "common_video/libyuv/include/libyuv.h"
-
-#include <assert.h>
-
-// LibYuv includes
-#ifdef WEBRTC_ANDROID
-#include "libyuv/files/include/libyuv.h"
-#else
-#include "third_party/libyuv/include/libyuv.h"
-#endif
-
-namespace webrtc {
-
-VideoType RawVideoTypeToCommonVideoVideoType(RawVideoType type) {
- switch (type) {
- case kVideoI420:
- return kI420;
- case kVideoIYUV:
- return kIYUV;
- case kVideoRGB24:
- return kRGB24;
- case kVideoARGB:
- return kARGB;
- case kVideoARGB4444:
- return kARGB4444;
- case kVideoRGB565:
- return kRGB565;
- case kVideoARGB1555:
- return kARGB1555;
- case kVideoYUY2:
- return kYUY2;
- case kVideoYV12:
- return kYV12;
- case kVideoUYVY:
- return kUYVY;
- case kVideoNV21:
- return kNV21;
- case kVideoNV12:
- return kNV12;
- case kVideoBGRA:
- return kBGRA;
- case kVideoMJPEG:
- return kMJPG;
- default:
- assert(false);
- }
- return kUnknown;
-}
-
-int CalcBufferSize(VideoType type, int width, int height) {
- int bits_per_pixel = 32;
- switch (type) {
- case kI420:
- case kNV12:
- case kNV21:
- case kIYUV:
- case kYV12:
- bits_per_pixel = 12;
- break;
- case kARGB4444:
- case kRGB565:
- case kARGB1555:
- case kYUY2:
- case kUYVY:
- bits_per_pixel = 16;
- break;
- case kRGB24:
- bits_per_pixel = 24;
- break;
- case kBGRA:
- case kARGB:
- bits_per_pixel = 32;
- break;
- default:
- assert(false);
- return -1;
- }
- return (width * height * bits_per_pixel) / 8; // bytes
-}
-
-int CalcBufferSize(VideoType src_video_type,
- VideoType dst_video_type,
- int length) {
- int src_bits_per_pixel = 32;
- switch (src_video_type) {
- case kI420:
- case kNV12:
- case kNV21:
- case kIYUV:
- case kYV12:
- src_bits_per_pixel = 12;
- break;
- case kARGB4444:
- case kRGB565:
- case kARGB1555:
- case kYUY2:
- case kUYVY:
- src_bits_per_pixel = 16;
- break;
- case kRGB24:
- src_bits_per_pixel = 24;
- break;
- case kBGRA:
- case kARGB:
- src_bits_per_pixel = 32;
- break;
- default:
- assert(false);
- return -1;
- }
-
- int dst_bits_per_pixel = 32;
- switch (dst_video_type) {
- case kI420:
- case kIYUV:
- case kYV12:
- dst_bits_per_pixel = 12;
- break;
- case kARGB4444:
- case kRGB565:
- case kARGB1555:
- case kYUY2:
- case kUYVY:
- dst_bits_per_pixel = 16;
- break;
- case kRGB24:
- dst_bits_per_pixel = 24;
- break;
- case kBGRA:
- case kARGB:
- dst_bits_per_pixel = 32;
- break;
- default:
- assert(false);
- return -1;
- }
- return (length * dst_bits_per_pixel) / src_bits_per_pixel;
-}
-
-int ConvertI420ToARGB4444(const uint8_t* src_frame,
- uint8_t* dst_frame,
- int width, int height,
- int dst_stride) {
- if (dst_stride == 0 || dst_stride == width)
- dst_stride = 2 * width;
- const uint8_t* yplane = src_frame;
- const uint8_t* uplane = src_frame + width * height;
- const uint8_t* vplane = uplane + (width * height / 4);
-
- return libyuv::I420ToARGB4444(yplane, width,
- uplane, width / 2,
- vplane, width / 2,
- dst_frame, dst_stride,
- width, height);
-}
-
-int ConvertI420ToRGB565(const uint8_t* src_frame,
- uint8_t* dst_frame,
- int width, int height) {
- int abs_height = (height < 0) ? -height : height;
- const uint8_t* yplane = src_frame;
- const uint8_t* uplane = src_frame + width * abs_height;
- const uint8_t* vplane = uplane + (width * abs_height / 4);
-
- return libyuv::I420ToRGB565(yplane, width,
- uplane, width / 2,
- vplane, width / 2,
- dst_frame, width,
- width, height);
-}
-
-int ConvertI420ToARGB1555(const uint8_t* src_frame,
- uint8_t* dst_frame,
- int width, int height,
- int dst_stride) {
- if (dst_stride == 0 || dst_stride == width)
- dst_stride = 2 * width;
- else if (dst_stride < 2 * width)
- return -1;
-
- const uint8_t* yplane = src_frame;
- const uint8_t* uplane = src_frame + width * height;
- const uint8_t* vplane = uplane + (width * height / 4);
-
- return libyuv::I420ToARGB1555(yplane, width,
- uplane, width / 2,
- vplane, width / 2,
- dst_frame, dst_stride,
- width, height);
-}
-
-int ConvertNV12ToRGB565(const uint8_t* src_frame,
- uint8_t* dst_frame,
- int width, int height) {
- const uint8_t* yplane = src_frame;
- const uint8_t* uvInterlaced = src_frame + (width * height);
-
- return libyuv::NV12ToRGB565(yplane, width,
- uvInterlaced, width / 2,
- dst_frame, width,
- width, height);
-}
-
-int ConvertRGB24ToARGB(const uint8_t* src_frame, uint8_t* dst_frame,
- int width, int height, int dst_stride) {
- if (dst_stride == 0 || dst_stride == width)
- dst_stride = width;
- // Stride - currently webrtc style
- return libyuv::RGB24ToARGB(src_frame, width,
- dst_frame, dst_stride,
- width, height);
-}
-
-libyuv::RotationMode ConvertRotationMode(VideoRotationMode rotation) {
- switch(rotation) {
- case kRotateNone:
- return libyuv::kRotate0;
- case kRotate90:
- return libyuv::kRotate90;
- case kRotate180:
- return libyuv::kRotate180;
- case kRotate270:
- return libyuv::kRotate270;
- }
- assert(false);
- return libyuv::kRotate0;
-}
-
-int ConvertVideoType(VideoType video_type) {
- switch(video_type) {
- case kUnknown:
- return libyuv::FOURCC_ANY;
- case kI420:
- return libyuv::FOURCC_I420;
- case kIYUV: // same as KYV12
- case kYV12:
- return libyuv::FOURCC_YV12;
- case kRGB24:
- return libyuv::FOURCC_24BG;
- case kABGR:
- return libyuv::FOURCC_ABGR;
- case kARGB4444:
- case kRGB565:
- case kARGB1555:
- // TODO(mikhal): Not supported;
- assert(false);
- return libyuv::FOURCC_ANY;
- case kYUY2:
- return libyuv::FOURCC_YUY2;
- case kUYVY:
- return libyuv::FOURCC_UYVY;
- case kMJPG:
- return libyuv::FOURCC_MJPG;
- case kNV21:
- return libyuv::FOURCC_NV21;
- case kNV12:
- return libyuv::FOURCC_NV12;
- case kARGB:
- return libyuv::FOURCC_ARGB;
- case kBGRA:
- return libyuv::FOURCC_BGRA;
- }
- assert(false);
- return libyuv::FOURCC_ANY;
-}
-
-int ConvertToI420(VideoType src_video_type,
- const uint8_t* src_frame,
- int crop_x, int crop_y,
- int src_width, int src_height,
- int sample_size,
- int dst_width, int dst_height, int dst_stride,
- VideoRotationMode rotation,
- uint8_t* dst_frame) {
- // All sanity tests are conducted within LibYuv.
- uint8_t* dst_yplane = dst_frame;
- uint8_t* dst_uplane = dst_yplane + dst_width * dst_height;
- uint8_t* dst_vplane = dst_uplane + (dst_width * dst_height / 4);
- return libyuv::ConvertToI420(src_frame, sample_size,
- dst_yplane, dst_stride,
- dst_uplane, (dst_stride + 1) / 2,
- dst_vplane, (dst_stride + 1) / 2,
- crop_x, crop_y,
- src_width, src_height,
- dst_width, dst_height,
- ConvertRotationMode(rotation),
- ConvertVideoType(src_video_type));
-}
-
-int ConvertFromI420(const uint8_t* src_frame, int src_stride,
- VideoType dst_video_type, int dst_sample_size,
- int width, int height,
- uint8_t* dst_frame) {
- const uint8_t* src_yplane = src_frame;
- const uint8_t* src_uplane = src_yplane + width * height;
- const uint8_t* src_vplane = src_uplane + (width * height / 4);
- return libyuv::ConvertFromI420(src_yplane, src_stride,
- src_uplane, (src_stride + 1) / 2,
- src_vplane, (src_stride + 1) / 2,
- dst_frame, dst_sample_size,
- width, height,
- ConvertVideoType(dst_video_type));
-}
-
-int ConvertFromYV12(const uint8_t* src_frame, int src_stride,
- VideoType dst_video_type, int dst_sample_size,
- int width, int height,
- uint8_t* dst_frame) {
- const uint8_t* src_yplane = src_frame;
- const uint8_t* src_uplane = src_yplane + width * height;
- const uint8_t* src_vplane = src_uplane + (width * height / 4);
- // YV12 = Y, V, U
- return libyuv::ConvertFromI420(src_yplane, src_stride,
- src_vplane, (src_stride + 1) / 2,
- src_uplane, (src_stride + 1) / 2,
- dst_frame, dst_sample_size,
- width, height,
- ConvertVideoType(dst_video_type));
-}
-
-int MirrorI420LeftRight(const uint8_t* src_frame,
- uint8_t* dst_frame,
- int width, int height) {
- const uint8_t* src_yplane = src_frame;
- const uint8_t* src_uplane = src_yplane + width * height;
- const uint8_t* src_vplane = src_uplane + (width * height / 4);
- uint8_t* dst_yplane = dst_frame;
- uint8_t* dst_uplane = dst_yplane + width * height;
- uint8_t* dst_vplane = dst_uplane + (width * height / 4);
- return libyuv::I420Mirror(src_yplane, width,
- src_uplane, width / 2,
- src_vplane, width / 2,
- dst_yplane, width,
- dst_uplane, width / 2,
- dst_vplane, width / 2,
- width, height);
-}
-
-int MirrorI420UpDown(const uint8_t* src_frame, uint8_t* dst_frame,
- int width, int height) {
- const uint8_t* src_yplane = src_frame;
- const uint8_t* src_uplane = src_frame + width * height;
- const uint8_t* src_vplane = src_uplane + (width * height / 4);
- uint8_t* dst_yplane = dst_frame;
- uint8_t* dst_uplane = dst_frame + width * height;
- uint8_t* dst_vplane = dst_uplane + (width * height / 4);
-
- // Inserting negative height flips the frame.
- return libyuv::I420Copy(src_yplane, width,
- src_uplane, width / 2,
- src_vplane, width / 2,
- dst_yplane, width,
- dst_uplane, width / 2,
- dst_vplane, width / 2,
- width, -height);
-}
-
-int ConvertToI420AndMirrorUpDown(const uint8_t* src_frame,
- uint8_t* dst_frame,
- int src_width, int src_height,
- VideoType src_video_type) {
- if (src_video_type != kI420 && src_video_type != kYV12)
- return -1;
- // TODO(mikhal): Use a more general convert function - with negative height
- // Inserting negative height flips the frame.
- // Using I420Copy with a negative height.
- const uint8_t* src_yplane = src_frame;
- const uint8_t* src_uplane = src_frame + src_width * src_height;
- const uint8_t* src_vplane = src_uplane + (src_width * src_height / 4);
- uint8_t* dst_yplane = dst_frame;
- uint8_t* dst_uplane = dst_frame + src_width * src_height;
- uint8_t* dst_vplane = dst_uplane + (src_width * src_height / 4);
- if (src_video_type == kYV12) {
- // Switch U and V
- dst_vplane = dst_frame + src_width * src_height;
- dst_uplane = dst_vplane + (src_width * src_height / 4);
- }
- // Inserting negative height flips the frame.
- return libyuv::I420Copy(src_yplane, src_width,
- src_uplane, src_width / 2,
- src_vplane, src_width / 2,
- dst_yplane, src_width,
- dst_uplane, src_width / 2,
- dst_vplane, src_width / 2,
- src_width, -src_height);
-}
-
-// Compute PSNR for an I420 frame (all planes)
-double I420PSNR(const uint8_t* ref_frame,
- const uint8_t* test_frame,
- int width, int height) {
- if (!ref_frame || !test_frame)
- return -1;
- else if (height < 0 || width < 0)
- return -1;
- const uint8_t* src_y_a = ref_frame;
- const uint8_t* src_u_a = src_y_a + width * height;
- const uint8_t* src_v_a = src_u_a + (width * height / 4);
- const uint8_t* src_y_b = test_frame;
- const uint8_t* src_u_b = src_y_b + width * height;
- const uint8_t* src_v_b = src_u_b + (width * height / 4);
- int stride_y = width;
- int stride_uv = (width + 1) / 2;
- double psnr = libyuv::I420Psnr(src_y_a, stride_y,
- src_u_a, stride_uv,
- src_v_a, stride_uv,
- src_y_b, stride_y,
- src_u_b, stride_uv,
- src_v_b, stride_uv,
- width, height);
- // LibYuv sets the max psnr value to 128, we restrict it to 48.
- // In case of 0 mse in one frame, 128 can skew the results significantly.
- return (psnr > 48.0) ? 48.0 : psnr;
-}
-// Compute SSIM for an I420 frame (all planes)
-double I420SSIM(const uint8_t* ref_frame,
- const uint8_t* test_frame,
- int width, int height) {
- if (!ref_frame || !test_frame)
- return -1;
- else if (height < 0 || width < 0)
- return -1;
- const uint8_t* src_y_a = ref_frame;
- const uint8_t* src_u_a = src_y_a + width * height;
- const uint8_t* src_v_a = src_u_a + (width * height / 4);
- const uint8_t* src_y_b = test_frame;
- const uint8_t* src_u_b = src_y_b + width * height;
- const uint8_t* src_v_b = src_u_b + (width * height / 4);
- int stride_y = width;
- int stride_uv = (width + 1) / 2;
- return libyuv::I420Ssim(src_y_a, stride_y,
- src_u_a, stride_uv,
- src_v_a, stride_uv,
- src_y_b, stride_y,
- src_u_b, stride_uv,
- src_v_b, stride_uv,
- width, height);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/common_video/libyuv/libyuv.gypi b/trunk/src/common_video/libyuv/libyuv.gypi
deleted file mode 100644
index bace5a2..0000000
--- a/trunk/src/common_video/libyuv/libyuv.gypi
+++ /dev/null
@@ -1,55 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'webrtc_libyuv',
- 'type': '<(library)',
- 'conditions': [
- ['build_libyuv==1', {
- 'dependencies': [
- '<(DEPTH)/third_party/libyuv/libyuv.gyp:libyuv'
- ],
- }, {
- # Need to add a directory normally exported by libyuv.gyp.
- 'include_dirs': [ '<(DEPTH)/third_party/libyuv/include', ],
- }],
- ],
- 'sources': [
- 'include/libyuv.h',
- 'include/scaler.h',
- 'libyuv.cc',
- 'scaler.cc',
- ],
- 'include_dirs': [
- '<(DEPTH)',
- ],
- },
- ], # targets
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'libyuv_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'webrtc_libyuv',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- ],
- 'sources': [
- 'libyuv_unittest.cc',
- 'scaler_unittest.cc',
- ],
- },
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
diff --git a/trunk/src/common_video/libyuv/libyuv_unittest.cc b/trunk/src/common_video/libyuv/libyuv_unittest.cc
deleted file mode 100644
index 07225ec..0000000
--- a/trunk/src/common_video/libyuv/libyuv_unittest.cc
+++ /dev/null
@@ -1,247 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <math.h>
-#include <string.h>
-
-#include "common_video/libyuv/include/libyuv.h"
-#include "gtest/gtest.h"
-#include "system_wrappers/interface/tick_util.h"
-#include "testsupport/fileutils.h"
-
-namespace webrtc {
-
-int PrintFrame(const uint8_t* frame, int width, int height) {
- if (frame == NULL)
- return -1;
- int k = 0;
- for (int i = 0; i < height; i++) {
- for (int j = 0; j < width; j++) {
- printf("%d ", frame[k++]);
- }
- printf(" \n");
- }
- printf(" \n");
- return 0;
-}
-
-int PrintFrame(const uint8_t* frame, int width,
- int height, const char* str) {
- if (frame == NULL)
- return -1;
- printf("%s %dx%d \n", str, width, height);
-
- const uint8_t* frame_y = frame;
- const uint8_t* frame_u = frame_y + width * height;
- const uint8_t* frame_v = frame_u + width * height / 4;
-
- int ret = 0;
- ret += PrintFrame(frame_y, width, height);
- ret += PrintFrame(frame_u, width / 2, height / 2);
- ret += PrintFrame(frame_v, width / 2, height / 2);
-
- return ret;
-}
-
-void CreateImage(int width, int height,
- uint8_t* frame, int offset,
- int height_factor, int width_factor) {
- if (frame == NULL)
- return;
- for (int i = 0; i < height; i++) {
- for (int j = 0; j < width; j++) {
- *frame = static_cast<uint8_t>((i + offset) * height_factor
- + j * width_factor);
- frame++;
- }
- }
-}
-
-class TestLibYuv : public ::testing::Test {
- protected:
- TestLibYuv();
- virtual void SetUp();
- virtual void TearDown();
-
- FILE* source_file_;
- const int width_;
- const int height_;
- const int frame_length_;
-};
-
-// TODO (mikhal): Use scoped_ptr when handling buffers.
-TestLibYuv::TestLibYuv()
- : source_file_(NULL),
- width_(352),
- height_(288),
- frame_length_(CalcBufferSize(kI420, 352, 288)) {
-}
-
-void TestLibYuv::SetUp() {
- const std::string input_file_name = webrtc::test::ProjectRootPath() +
- "resources/foreman_cif.yuv";
- source_file_ = fopen(input_file_name.c_str(), "rb");
- ASSERT_TRUE(source_file_ != NULL) << "Cannot read file: "<<
- input_file_name << "\n";
-}
-
-void TestLibYuv::TearDown() {
- if (source_file_ != NULL) {
- ASSERT_EQ(0, fclose(source_file_));
- }
- source_file_ = NULL;
-}
-
-TEST_F(TestLibYuv, ConvertSanityTest) {
- // TODO(mikhal)
-}
-
-TEST_F(TestLibYuv, ConvertTest) {
- // Reading YUV frame - testing on the first frame of the foreman sequence
- int j = 0;
- std::string output_file_name = webrtc::test::OutputPath() +
- "LibYuvTest_conversion.yuv";
- FILE* output_file = fopen(output_file_name.c_str(), "wb");
- ASSERT_TRUE(output_file != NULL);
-
- double psnr = 0;
-
- uint8_t* orig_buffer = new uint8_t[frame_length_];
- EXPECT_GT(fread(orig_buffer, 1, frame_length_, source_file_), 0U);
-
- // printf("\nConvert #%d I420 <-> RGB24\n", j);
- uint8_t* res_rgb_buffer2 = new uint8_t[width_ * height_ * 3];
- uint8_t* res_i420_buffer = new uint8_t[frame_length_];
-
- EXPECT_EQ(0, ConvertFromI420(orig_buffer, width_, kRGB24, 0,
- width_, height_, res_rgb_buffer2));
-
- EXPECT_EQ(0, ConvertToI420(kRGB24, res_rgb_buffer2, 0, 0, width_, height_,
- 0, width_, height_, width_, kRotateNone,
- res_i420_buffer));
-
- fwrite(res_i420_buffer, frame_length_, 1, output_file);
- psnr = I420PSNR(orig_buffer, res_i420_buffer, width_, height_);
- // Optimization Speed- quality trade-off => 45 dB only (platform dependant).
- EXPECT_GT(ceil(psnr), 44);
- j++;
- delete [] res_rgb_buffer2;
-
- // printf("\nConvert #%d I420 <-> UYVY\n", j);
- uint8_t* out_uyvy_buffer = new uint8_t[width_ * height_ * 2];
- EXPECT_EQ(0, ConvertFromI420(orig_buffer, width_,
- kUYVY, 0, width_, height_, out_uyvy_buffer));
- EXPECT_EQ(0, ConvertToI420(kUYVY, out_uyvy_buffer, 0, 0, width_, height_,
- 0, width_, height_, width_,kRotateNone, res_i420_buffer));
- psnr = I420PSNR(orig_buffer, res_i420_buffer, width_, height_);
- EXPECT_EQ(48.0, psnr);
- fwrite(res_i420_buffer, frame_length_, 1, output_file);
-
- j++;
- delete [] out_uyvy_buffer;
-
- // printf("\nConvert #%d I420 <-> I420 \n", j);
- uint8_t* out_i420_buffer = new uint8_t[width_ * height_ * 3 / 2 ];
- EXPECT_EQ(0, ConvertToI420(kI420, orig_buffer, 0, 0, width_, height_,
- 0, width_, height_, width_,
- kRotateNone, out_i420_buffer));
- EXPECT_EQ(0, ConvertFromI420(out_i420_buffer, width_, kI420, 0,
- width_, height_, res_i420_buffer));
- fwrite(res_i420_buffer, frame_length_, 1, output_file);
- psnr = I420PSNR(orig_buffer, res_i420_buffer, width_, height_);
- EXPECT_EQ(48.0, psnr);
- j++;
- delete [] out_i420_buffer;
-
- // printf("\nConvert #%d I420 <-> YV12\n", j);
- uint8_t* outYV120Buffer = new uint8_t[frame_length_];
-
- EXPECT_EQ(0, ConvertFromI420(orig_buffer, width_, kYV12, 0,
- width_, height_, outYV120Buffer));
- EXPECT_EQ(0, ConvertFromYV12(outYV120Buffer, width_,
- kI420, 0,
- width_, height_,
- res_i420_buffer));
- fwrite(res_i420_buffer, frame_length_, 1, output_file);
-
- psnr = I420PSNR(orig_buffer, res_i420_buffer, width_, height_);
- EXPECT_EQ(48.0, psnr);
- j++;
- delete [] outYV120Buffer;
-
- // printf("\nConvert #%d I420 <-> YUY2\n", j);
- uint8_t* out_yuy2_buffer = new uint8_t[width_ * height_ * 2];
- EXPECT_EQ(0, ConvertFromI420(orig_buffer, width_,
- kYUY2, 0, width_, height_, out_yuy2_buffer));
-
- EXPECT_EQ(0, ConvertToI420(kYUY2, out_yuy2_buffer, 0, 0, width_, height_,
- 0, width_, height_, width_,
- kRotateNone, res_i420_buffer));
-
- fwrite(res_i420_buffer, frame_length_, 1, output_file);
- psnr = I420PSNR(orig_buffer, res_i420_buffer, width_, height_);
- EXPECT_EQ(48.0, psnr);
- ASSERT_EQ(0, fclose(output_file));
- delete [] out_yuy2_buffer;
- delete [] res_i420_buffer;
- delete [] orig_buffer;
-}
-
-TEST_F(TestLibYuv, MirrorTest) {
- // TODO (mikhal): Add an automated test to confirm output.
- std::string str;
- int width = 16;
- int height = 8;
- int factor_y = 1;
- int factor_u = 1;
- int factor_v = 1;
- int start_buffer_offset = 10;
- int length = webrtc::CalcBufferSize(kI420, width, height);
-
- uint8_t* test_frame = new uint8_t[length];
- memset(test_frame, 255, length);
-
- // Create input frame
- uint8_t* in_frame = test_frame;
- uint8_t* in_frame_cb = in_frame + width * height;
- uint8_t* in_frame_cr = in_frame_cb + (width * height) / 4;
- CreateImage(width, height, in_frame, 10, factor_y, 1); // Y
- CreateImage(width / 2, height / 2, in_frame_cb, 100, factor_u, 1); // Cb
- CreateImage(width / 2, height / 2, in_frame_cr, 200, factor_v, 1); // Cr
- EXPECT_EQ(0, PrintFrame(test_frame, width, height, "InputFrame"));
-
- uint8_t* test_frame2 = new uint8_t[length + start_buffer_offset * 2];
- memset(test_frame2, 255, length + start_buffer_offset * 2);
- uint8_t* out_frame = test_frame2;
-
- // LeftRight
- std::cout << "Test Mirror function: LeftRight" << std::endl;
- EXPECT_EQ(0, MirrorI420LeftRight(in_frame, out_frame, width, height));
- EXPECT_EQ(0, PrintFrame(test_frame2, width, height, "OutputFrame"));
- EXPECT_EQ(0, MirrorI420LeftRight(out_frame, test_frame, width, height));
-
- EXPECT_EQ(0, memcmp(in_frame, test_frame, length));
-
- // UpDown
- std::cout << "Test Mirror function: UpDown" << std::endl;
- EXPECT_EQ(0, MirrorI420UpDown(in_frame, out_frame, width, height));
- EXPECT_EQ(0, PrintFrame(test_frame2, width, height, "OutputFrame"));
- EXPECT_EQ(0, MirrorI420UpDown(out_frame, test_frame, width, height));
-
- EXPECT_EQ(0, memcmp(in_frame, test_frame, length));
-
- // TODO(mikhal): Write to a file, and ask to look at the file.
-
- std::cout << "Do the mirrored frames look correct?" << std::endl;
- delete [] test_frame;
- delete [] test_frame2;
-}
-
-} // namespace
diff --git a/trunk/src/common_video/libyuv/scaler.cc b/trunk/src/common_video/libyuv/scaler.cc
deleted file mode 100644
index a12462c..0000000
--- a/trunk/src/common_video/libyuv/scaler.cc
+++ /dev/null
@@ -1,105 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "common_video/libyuv/include/scaler.h"
-
-// LibYuv
-#ifdef WEBRTC_ANDROID
-#include "libyuv/files/include/libyuv.h"
-#else
-#include "third_party/libyuv/include/libyuv.h"
-#endif
-
-namespace webrtc {
-
-Scaler::Scaler()
- : method_(kScaleBox),
- src_width_(0),
- src_height_(0),
- dst_width_(0),
- dst_height_(0),
- set_(false) {}
-
-Scaler::~Scaler() {}
-
-int Scaler::Set(int src_width, int src_height,
- int dst_width, int dst_height,
- VideoType src_video_type, VideoType dst_video_type,
- ScaleMethod method) {
- set_ = false;
- if (src_width < 1 || src_height < 1 || dst_width < 1 || dst_height < 1)
- return -1;
-
- if (!SupportedVideoType(src_video_type, dst_video_type))
- return -1;
-
- src_width_ = src_width;
- src_height_ = src_height;
- dst_width_ = dst_width;
- dst_height_ = dst_height;
- method_ = method;
- set_ = true;
- return 0;
-}
-
-int Scaler::Scale(const uint8_t* src_frame,
- uint8_t*& dst_frame,
- int& dst_size) {
- if (src_frame == NULL)
- return -1;
- if (!set_)
- return -2;
-
- // Making sure that destination frame is of sufficient size
- int required_dst_size = dst_width_ * dst_height_ * 3 / 2;
- if (dst_frame && required_dst_size > dst_size) {
- // allocated buffer is too small
- delete [] dst_frame;
- dst_frame = NULL;
- }
- if (dst_frame == NULL) {
- dst_frame = new uint8_t[required_dst_size];
- dst_size = required_dst_size;
- }
-
- // Converting to planes:
- const uint8_t* src_yplane = src_frame;
- const uint8_t* src_uplane = src_frame + src_width_ * src_height_;
- const uint8_t* src_vplane = src_uplane + src_width_ * src_height_ / 4;
-
- uint8_t* dst_yplane = dst_frame;
- uint8_t* dst_uplane = dst_frame + dst_width_ * dst_height_;
- uint8_t* dst_vplane = dst_uplane + dst_width_ * dst_height_ / 4;
-
- return libyuv::I420Scale(src_yplane, src_width_,
- src_uplane, src_width_ / 2,
- src_vplane, src_width_ / 2,
- src_width_, src_height_,
- dst_yplane, dst_width_,
- dst_uplane, dst_width_ / 2,
- dst_vplane, dst_width_ / 2,
- dst_width_, dst_height_,
- libyuv::FilterMode(method_));
-}
-
-// TODO(mikhal): Add support for more types.
-bool Scaler::SupportedVideoType(VideoType src_video_type,
- VideoType dst_video_type) {
- if (src_video_type != dst_video_type)
- return false;
-
- if ((src_video_type == kI420) || (src_video_type == kIYUV) ||
- (src_video_type == kYV12))
- return true;
-
- return false;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/common_video/libyuv/scaler_unittest.cc b/trunk/src/common_video/libyuv/scaler_unittest.cc
deleted file mode 100644
index 899ad89..0000000
--- a/trunk/src/common_video/libyuv/scaler_unittest.cc
+++ /dev/null
@@ -1,239 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <math.h>
-#include <string.h>
-
-#include "common_video/libyuv/include/scaler.h"
-#include "gtest/gtest.h"
-#include "system_wrappers/interface/tick_util.h"
-#include "testsupport/fileutils.h"
-
-namespace webrtc {
-
-class TestScaler : public ::testing::Test {
- protected:
- TestScaler();
- virtual void SetUp();
- virtual void TearDown();
-
- void ScaleSequence(ScaleMethod method,
- FILE* source_file, std::string out_name,
- int src_width, int src_height,
- int dst_width, int dst_height);
-
- Scaler test_scaler_;
- FILE* source_file_;
- uint8_t* test_buffer_;
- const int width_;
- const int height_;
- const int frame_length_;
-};
-
-
-// TODO (mikhal): Use scoped_ptr when handling buffers.
-TestScaler::TestScaler()
- : source_file_(NULL),
- width_(352),
- height_(288),
- frame_length_(CalcBufferSize(kI420, 352, 288)) {
-}
-
-void TestScaler::SetUp() {
- const std::string input_file_name =
- webrtc::test::ResourcePath("foreman_cif", "yuv");
- source_file_ = fopen(input_file_name.c_str(), "rb");
- ASSERT_TRUE(source_file_ != NULL) << "Cannot read file: "<<
- input_file_name << "\n";
- test_buffer_ = new uint8_t[frame_length_];
-}
-
-void TestScaler::TearDown() {
- if (source_file_ != NULL) {
- ASSERT_EQ(0, fclose(source_file_));
- }
- source_file_ = NULL;
- delete [] test_buffer_;
-}
-
-TEST_F(TestScaler, ScaleWithoutSettingValues) {
- int size = 100;
- EXPECT_EQ(-2, test_scaler_.Scale(test_buffer_, test_buffer_, size));
-}
-
-TEST_F(TestScaler, ScaleBadInitialValues) {
- EXPECT_EQ(-1, test_scaler_.Set(0, 288, 352, 288, kI420, kI420, kScalePoint));
- EXPECT_EQ(-1, test_scaler_.Set(704, 0, 352, 288, kI420, kI420, kScaleBox));
- EXPECT_EQ(-1, test_scaler_.Set(704, 576, 352, 0, kI420, kI420,
- kScaleBilinear));
- EXPECT_EQ(-1, test_scaler_.Set(704, 576, 0, 288, kI420, kI420, kScalePoint));
-}
-
-TEST_F(TestScaler, ScaleSendingNullSourcePointer) {
- int size = 0;
- EXPECT_EQ(-1, test_scaler_.Scale(NULL, test_buffer_, size));
-}
-
-TEST_F(TestScaler, ScaleSendingBufferTooSmall) {
- // Sending a buffer which is too small (should reallocate and update size)
- EXPECT_EQ(0, test_scaler_.Set(352, 288, 144, 288, kI420, kI420, kScalePoint));
- uint8_t* test_buffer2 = NULL;
- int size = 0;
- EXPECT_GT(fread(test_buffer_, 1, frame_length_, source_file_), 0U);
- EXPECT_EQ(0, test_scaler_.Scale(test_buffer_, test_buffer2, size));
- EXPECT_EQ(144 * 288 * 3 / 2, size);
- delete [] test_buffer2;
-}
-
-//TODO (mikhal): Converge the test into one function that accepts the method.
-TEST_F(TestScaler, PointScaleTest) {
- ScaleMethod method = kScalePoint;
- std::string out_name = webrtc::test::OutputPath() +
- "LibYuvTest_PointScale_176_144.yuv";
- ScaleSequence(method,
- source_file_, out_name,
- width_, height_,
- width_ / 2, height_ / 2);
- out_name = webrtc::test::OutputPath() + "LibYuvTest_PointScale_320_240.yuv";
- ScaleSequence(method,
- source_file_, out_name,
- width_, height_,
- 320, 240);
- out_name = webrtc::test::OutputPath() + "LibYuvTest_PointScale_704_576.yuv";
- ScaleSequence(method,
- source_file_, out_name,
- width_, height_,
- width_ * 2, height_ * 2);
- out_name = webrtc::test::OutputPath() + "LibYuvTest_PointScale_300_200.yuv";
- ScaleSequence(method,
- source_file_, out_name,
- width_, height_,
- 300, 200);
- out_name = webrtc::test::OutputPath() + "LibYuvTest_PointScale_400_300.yuv";
- ScaleSequence(method,
- source_file_, out_name,
- width_, height_,
- 400, 300);
-}
-
-TEST_F(TestScaler, BiLinearScaleTest) {
- ScaleMethod method = kScaleBilinear;
- std::string out_name = webrtc::test::OutputPath() +
- "LibYuvTest_BilinearScale_176_144.yuv";
- ScaleSequence(method,
- source_file_, out_name,
- width_, height_,
- width_ / 2, height_ / 2);
- out_name = webrtc::test::OutputPath() +
- "LibYuvTest_BilinearScale_320_240.yuv";
- ScaleSequence(method,
- source_file_, out_name,
- width_, height_,
- 320, 240);
- out_name = webrtc::test::OutputPath() +
- "LibYuvTest_BilinearScale_704_576.yuv";
- ScaleSequence(method,
- source_file_, out_name,
- width_, height_,
- width_ * 2, height_ * 2);
- out_name = webrtc::test::OutputPath() +
- "LibYuvTest_BilinearScale_300_200.yuv";
- ScaleSequence(method,
- source_file_, out_name,
- width_, height_,
- 300, 200);
- out_name = webrtc::test::OutputPath() +
- "LibYuvTest_BilinearScale_400_300.yuv";
- ScaleSequence(method,
- source_file_, out_name,
- width_, height_,
- 400, 300);
-}
-
-TEST_F(TestScaler, BoxScaleTest) {
- ScaleMethod method = kScaleBox;
- std::string out_name = webrtc::test::OutputPath() +
- "LibYuvTest_BoxScale_176_144.yuv";
- ScaleSequence(method,
- source_file_, out_name,
- width_, height_,
- width_ / 2, height_ / 2);
- out_name = webrtc::test::OutputPath() + "LibYuvTest_BoxScale_320_240.yuv";
- ScaleSequence(method,
- source_file_, out_name,
- width_, height_,
- 320, 240);
- out_name = webrtc::test::OutputPath() + "LibYuvTest_BoxScale_704_576.yuv";
- ScaleSequence(method,
- source_file_, out_name,
- width_, height_,
- width_ * 2, height_ * 2);
- out_name = webrtc::test::OutputPath() + "LibYuvTest_BoxScale_300_200.yuv";
- ScaleSequence(method,
- source_file_, out_name,
- width_, height_,
- 300, 200);
- out_name = webrtc::test::OutputPath() + "LibYuvTest_BoxScale_400_300.yuv";
- ScaleSequence(method,
- source_file_, out_name,
- width_, height_,
- 400, 300);
-}
-
-// TODO (mikhal): Move part to a separate scale test.
-void TestScaler::ScaleSequence(ScaleMethod method,
- FILE* source_file, std::string out_name,
- int src_width, int src_height,
- int dst_width, int dst_height) {
- FILE* output_file;
- EXPECT_EQ(0, test_scaler_.Set(src_width, src_height,
- dst_width, dst_height,
- kI420, kI420, method));
-
- output_file = fopen(out_name.c_str(), "wb");
- ASSERT_TRUE(output_file != NULL);
-
- rewind(source_file);
-
- int out_required_size = dst_width * dst_height * 3 / 2;
- int in_required_size = src_height * src_width * 3 / 2;
- uint8_t* input_buffer = new uint8_t[in_required_size];
- uint8_t* output_buffer = new uint8_t[out_required_size];
-
- int64_t start_clock, total_clock;
- total_clock = 0;
- int frame_count = 0;
-
- // Running through entire sequence
- while (feof(source_file) == 0) {
- if ((size_t)in_required_size !=
- fread(input_buffer, 1, in_required_size, source_file))
- break;
-
- start_clock = TickTime::MillisecondTimestamp();
- EXPECT_EQ(0, test_scaler_.Scale(input_buffer, output_buffer,
- out_required_size));
- total_clock += TickTime::MillisecondTimestamp() - start_clock;
- fwrite(output_buffer, out_required_size, 1, output_file);
- frame_count++;
- }
-
- if (frame_count) {
- printf("Scaling[%d %d] => [%d %d]: ",
- src_width, src_height, dst_width, dst_height);
- printf("Average time per frame[ms]: %.2lf\n",
- (static_cast<double>(total_clock) / frame_count));
- }
- ASSERT_EQ(0, fclose(output_file));
- delete [] input_buffer;
- delete [] output_buffer;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/engine_configurations.h b/trunk/src/engine_configurations.h
deleted file mode 100644
index 49a2f8c..0000000
--- a/trunk/src/engine_configurations.h
+++ /dev/null
@@ -1,149 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_ENGINE_CONFIGURATIONS_H_
-#define WEBRTC_ENGINE_CONFIGURATIONS_H_
-
-// ============================================================================
-// Voice and Video
-// ============================================================================
-
-// Don't link in socket support in Chrome
-#ifdef WEBRTC_CHROMIUM_BUILD
-#define WEBRTC_EXTERNAL_TRANSPORT
-#endif
-
-// Optional to enable stand-alone
-// #define WEBRTC_EXTERNAL_TRANSPORT
-
-// ----------------------------------------------------------------------------
-// [Voice] Codec settings
-// ----------------------------------------------------------------------------
-
-#define WEBRTC_CODEC_ISAC // floating-point iSAC implementation (default)
-// #define WEBRTC_CODEC_ISACFX // fix-point iSAC implementation
-#define WEBRTC_CODEC_AVT
-
-#ifndef WEBRTC_CHROMIUM_BUILD
-#define WEBRTC_CODEC_ILBC
-#define WEBRTC_CODEC_G722
-#define WEBRTC_CODEC_PCM16
-#define WEBRTC_CODEC_RED
-#endif
-
-// ----------------------------------------------------------------------------
-// [Video] Codec settings
-// ----------------------------------------------------------------------------
-
-#define VIDEOCODEC_I420
-#define VIDEOCODEC_VP8
-
-// ============================================================================
-// VoiceEngine
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// Settings for VoiceEngine
-// ----------------------------------------------------------------------------
-
-#define WEBRTC_VOICE_ENGINE_AGC // Near-end AGC
-#define WEBRTC_VOICE_ENGINE_ECHO // Near-end AEC
-#define WEBRTC_VOICE_ENGINE_NR // Near-end NS
-#define WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
-
-#ifndef WEBRTC_CHROMIUM_BUILD
-#define WEBRTC_VOICE_ENGINE_TYPING_DETECTION // Typing detection
-#endif
-
-// ----------------------------------------------------------------------------
-// VoiceEngine sub-APIs
-// ----------------------------------------------------------------------------
-
-#define WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
-#define WEBRTC_VOICE_ENGINE_CODEC_API
-#define WEBRTC_VOICE_ENGINE_DTMF_API
-#define WEBRTC_VOICE_ENGINE_HARDWARE_API
-#define WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
-#define WEBRTC_VOICE_ENGINE_NETWORK_API
-#define WEBRTC_VOICE_ENGINE_RTP_RTCP_API
-#define WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
-#define WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
-
-#ifndef WEBRTC_CHROMIUM_BUILD
-#define WEBRTC_VOICE_ENGINE_CALL_REPORT_API
-#define WEBRTC_VOICE_ENGINE_ENCRYPTION_API
-#define WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
-#define WEBRTC_VOICE_ENGINE_FILE_API
-#endif
-
-// ============================================================================
-// VideoEngine
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// Settings for special VideoEngine configurations
-// ----------------------------------------------------------------------------
-// ----------------------------------------------------------------------------
-// VideoEngine sub-API:s
-// ----------------------------------------------------------------------------
-
-#define WEBRTC_VIDEO_ENGINE_CAPTURE_API
-#define WEBRTC_VIDEO_ENGINE_CODEC_API
-#define WEBRTC_VIDEO_ENGINE_ENCRYPTION_API
-#define WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
-#define WEBRTC_VIDEO_ENGINE_NETWORK_API
-#define WEBRTC_VIDEO_ENGINE_RENDER_API
-#define WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
-// #define WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
-
-#ifndef WEBRTC_CHROMIUM_BUILD
-#define WEBRTC_VIDEO_ENGINE_FILE_API
-#endif
-
-// ============================================================================
-// Platform specific configurations
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// VideoEngine Windows
-// ----------------------------------------------------------------------------
-
-#if defined(_WIN32)
- // #define DIRECTDRAW_RENDERING
- #define DIRECT3D9_RENDERING // Requires DirectX 9.
-#endif
-
-// ----------------------------------------------------------------------------
-// VideoEngine MAC
-// ----------------------------------------------------------------------------
-
-#if defined(WEBRTC_MAC) && !defined(MAC_IPHONE)
- // #define CARBON_RENDERING
- #define COCOA_RENDERING
-#endif
-
-// ----------------------------------------------------------------------------
-// VideoEngine Mobile iPhone
-// ----------------------------------------------------------------------------
-
-#if defined(MAC_IPHONE)
- #define EAGL_RENDERING
-#endif
-
-// ----------------------------------------------------------------------------
-// Deprecated
-// ----------------------------------------------------------------------------
-
-// #define WEBRTC_CODEC_G729
-// #define WEBRTC_DTMF_DETECTION
-// #define WEBRTC_SRTP
-// #define WEBRTC_SRTP_ALLOW_ROC_ITERATION
-
-#endif // WEBRTC_ENGINE_CONFIGURATIONS_H_
diff --git a/trunk/src/modules/audio_coding/codecs/OWNERS b/trunk/src/modules/audio_coding/codecs/OWNERS
deleted file mode 100644
index e1e6256..0000000
--- a/trunk/src/modules/audio_coding/codecs/OWNERS
+++ /dev/null
@@ -1,3 +0,0 @@
-tina.legrand@webrtc.org
-turaj@webrtc.org
-jan.skoglund@webrtc.org
diff --git a/trunk/src/modules/audio_coding/codecs/cng/Android.mk b/trunk/src/modules/audio_coding/codecs/cng/Android.mk
deleted file mode 100644
index 32671cc..0000000
--- a/trunk/src/modules/audio_coding/codecs/cng/Android.mk
+++ /dev/null
@@ -1,40 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_cng
-LOCAL_MODULE_TAGS := optional
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := \
- webrtc_cng.c \
- cng_helpfuns.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include \
- $(LOCAL_PATH)/../../../.. \
- $(LOCAL_PATH)/../../../../common_audio/signal_processing/include
-
-LOCAL_SHARED_LIBRARIES := \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_coding/codecs/cng/cng.gypi b/trunk/src/modules/audio_coding/codecs/cng/cng.gypi
deleted file mode 100644
index 2903cc2..0000000
--- a/trunk/src/modules/audio_coding/codecs/cng/cng.gypi
+++ /dev/null
@@ -1,57 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'CNG',
- 'type': '<(library)',
- 'dependencies': [
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- ],
- 'include_dirs': [
- 'include',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- ],
- },
- 'sources': [
- 'include/webrtc_cng.h',
- 'webrtc_cng.c',
- 'cng_helpfuns.c',
- 'cng_helpfuns.h',
- ],
- },
- ], # targets
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'cng_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'CNG',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- ],
- 'sources': [
- 'cng_unittest.cc',
- ],
- }, # CNG_unittests
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/cng/cng_helpfuns.c b/trunk/src/modules/audio_coding/codecs/cng/cng_helpfuns.c
deleted file mode 100644
index 2e9029f..0000000
--- a/trunk/src/modules/audio_coding/codecs/cng/cng_helpfuns.c
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-#include "webrtc_cng.h"
-#include "signal_processing_library.h"
-#include "typedefs.h"
-#include "cng_helpfuns.h"
-
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-
-void WebRtcCng_K2a16(
- WebRtc_Word16 *k, /* Q15. */
- int useOrder,
- WebRtc_Word16 *a /* Q12. */
-)
-{
- WebRtc_Word16 any[WEBRTC_SPL_MAX_LPC_ORDER+1];
- WebRtc_Word16 *aptr, *aptr2, *anyptr;
- G_CONST WebRtc_Word16 *kptr;
- int m, i;
-
- kptr = k;
- *a = 4096; /* i.e., (Word16_MAX >> 3)+1 */
- *any = *a;
- a[1] = (*k+4) >> 3;
- for( m=1; m<useOrder; m++ )
- {
- kptr++;
- aptr = a;
- aptr++;
- aptr2 = &a[m];
- anyptr = any;
- anyptr++;
-
- any[m+1] = (*kptr+4) >> 3;
- for( i=0; i<m; i++ ) {
- *anyptr++ = (*aptr++) + (WebRtc_Word16)( (( (WebRtc_Word32)(*aptr2--) * (WebRtc_Word32)*kptr )+16384) >> 15);
- }
-
- aptr = a;
- anyptr = any;
- for( i=0; i<(m+2); i++ ){
- *aptr++ = *anyptr++;
- }
- }
-}
-
-
-#ifdef __cplusplus
-}
-#endif
-
diff --git a/trunk/src/modules/audio_coding/codecs/cng/cng_helpfuns.h b/trunk/src/modules/audio_coding/codecs/cng/cng_helpfuns.h
deleted file mode 100644
index fd8d6dc..0000000
--- a/trunk/src/modules/audio_coding/codecs/cng/cng_helpfuns.h
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_SOURCE_CNG_HELPFUNS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_SOURCE_CNG_HELPFUNS_H_
-
-extern WebRtc_Word32 lpc_lagwinTbl_fixw32[WEBRTC_CNG_MAX_LPC_ORDER + 1];
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-
-void WebRtcCng_K2a16(WebRtc_Word16 *k, int useOrder, WebRtc_Word16 *a);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_SOURCE_CNG_HELPFUNS_H_
diff --git a/trunk/src/modules/audio_coding/codecs/cng/cng_unittest.cc b/trunk/src/modules/audio_coding/codecs/cng/cng_unittest.cc
deleted file mode 100644
index 6a4edc0..0000000
--- a/trunk/src/modules/audio_coding/codecs/cng/cng_unittest.cc
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Empty test just to get code coverage metrics for this dir.
- */
-#include "webrtc_cng.h"
-#include "gtest/gtest.h"
-
-TEST(CngTest, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/modules/audio_coding/codecs/cng/include/webrtc_cng.h b/trunk/src/modules/audio_coding/codecs/cng/include/webrtc_cng.h
deleted file mode 100644
index d405e3a..0000000
--- a/trunk/src/modules/audio_coding/codecs/cng/include/webrtc_cng.h
+++ /dev/null
@@ -1,236 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_INTERFACE_WEBRTC_CNG_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_INTERFACE_WEBRTC_CNG_H_
-
-#include "typedefs.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-#define WEBRTC_CNG_MAX_LPC_ORDER 12
-#define WEBRTC_CNG_MAX_OUTSIZE_ORDER 640
-
-/* Define Error codes */
-
-/* 6100 Encoder */
-#define CNG_ENCODER_MEMORY_ALLOCATION_FAILED 6110
-#define CNG_ENCODER_NOT_INITIATED 6120
-#define CNG_DISALLOWED_LPC_ORDER 6130
-#define CNG_DISALLOWED_FRAME_SIZE 6140
-#define CNG_DISALLOWED_SAMPLING_FREQUENCY 6150
-/* 6200 Decoder */
-#define CNG_DECODER_MEMORY_ALLOCATION_FAILED 6210
-#define CNG_DECODER_NOT_INITIATED 6220
-
-
-typedef struct WebRtcCngEncInst CNG_enc_inst;
-typedef struct WebRtcCngDecInst CNG_dec_inst;
-
-
-/****************************************************************************
- * WebRtcCng_Version(...)
- *
- * These functions returns the version name (string must be at least
- * 500 characters long)
- *
- * Output:
- * - version : Pointer to character string
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcCng_Version(char *version);
-
-/****************************************************************************
- * WebRtcCng_AssignSizeEnc/Dec(...)
- *
- * These functions get the size needed for storing the instance for encoder
- * and decoder, respectively
- *
- * Input/Output:
- * - sizeinbytes : Pointer to integer where the size is returned
- *
- * Return value : 0
- */
-
-WebRtc_Word16 WebRtcCng_AssignSizeEnc(int *sizeinbytes);
-WebRtc_Word16 WebRtcCng_AssignSizeDec(int *sizeinbytes);
-
-
-/****************************************************************************
- * WebRtcCng_AssignEnc/Dec(...)
- *
- * These functions Assignes memory for the instances.
- *
- * Input:
- * - CNG_inst_Addr : Adress to where to assign memory
- * Output:
- * - inst : Pointer to the instance that should be created
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcCng_AssignEnc(CNG_enc_inst **inst, void *CNG_inst_Addr);
-WebRtc_Word16 WebRtcCng_AssignDec(CNG_dec_inst **inst, void *CNG_inst_Addr);
-
-
-/****************************************************************************
- * WebRtcCng_CreateEnc/Dec(...)
- *
- * These functions create an instance to the specified structure
- *
- * Input:
- * - XXX_inst : Pointer to created instance that should be created
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcCng_CreateEnc(CNG_enc_inst **cng_inst);
-WebRtc_Word16 WebRtcCng_CreateDec(CNG_dec_inst **cng_inst);
-
-
-/****************************************************************************
- * WebRtcCng_InitEnc/Dec(...)
- *
- * This function initializes a instance
- *
- * Input:
- * - cng_inst : Instance that should be initialized
- *
- * - fs : 8000 for narrowband and 16000 for wideband
- * - interval : generate SID data every interval ms
- * - quality : Number of refl. coefs, maximum allowed is 12
- *
- * Output:
- * - cng_inst : Initialized instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcCng_InitEnc(CNG_enc_inst *cng_inst,
- WebRtc_Word16 fs,
- WebRtc_Word16 interval,
- WebRtc_Word16 quality);
-WebRtc_Word16 WebRtcCng_InitDec(CNG_dec_inst *cng_dec_inst);
-
-
-/****************************************************************************
- * WebRtcCng_FreeEnc/Dec(...)
- *
- * These functions frees the dynamic memory of a specified instance
- *
- * Input:
- * - cng_inst : Pointer to created instance that should be freed
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-
-WebRtc_Word16 WebRtcCng_FreeEnc(CNG_enc_inst *cng_inst);
-WebRtc_Word16 WebRtcCng_FreeDec(CNG_dec_inst *cng_inst);
-
-
-
-/****************************************************************************
- * WebRtcCng_Encode(...)
- *
- * These functions analyzes background noise
- *
- * Input:
- * - cng_inst : Pointer to created instance
- * - speech : Signal to be analyzed
- * - nrOfSamples : Size of speech vector
- * - forceSID : not zero to force SID frame and reset
- *
- * Output:
- * - bytesOut : Nr of bytes to transmit, might be 0
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcCng_Encode(CNG_enc_inst *cng_inst,
- WebRtc_Word16 *speech,
- WebRtc_Word16 nrOfSamples,
- WebRtc_UWord8* SIDdata,
- WebRtc_Word16 *bytesOut,
- WebRtc_Word16 forceSID);
-
-
-/****************************************************************************
- * WebRtcCng_UpdateSid(...)
- *
- * These functions updates the CN state, when a new SID packet arrives
- *
- * Input:
- * - cng_inst : Pointer to created instance that should be freed
- * - SID : SID packet, all headers removed
- * - length : Length in bytes of SID packet
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-WebRtc_Word16 WebRtcCng_UpdateSid(CNG_dec_inst *cng_inst,
- WebRtc_UWord8 *SID,
- WebRtc_Word16 length);
-
-
-/****************************************************************************
- * WebRtcCng_Generate(...)
- *
- * These functions generates CN data when needed
- *
- * Input:
- * - cng_inst : Pointer to created instance that should be freed
- * - outData : pointer to area to write CN data
- * - nrOfSamples : How much data to generate
- * - new_period : >0 if a new period of CNG, will reset history
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-WebRtc_Word16 WebRtcCng_Generate(CNG_dec_inst *cng_inst,
- WebRtc_Word16 * outData,
- WebRtc_Word16 nrOfSamples,
- WebRtc_Word16 new_period);
-
-
-/*****************************************************************************
- * WebRtcCng_GetErrorCodeEnc/Dec(...)
- *
- * This functions can be used to check the error code of a CNG instance. When
- * a function returns -1 a error code will be set for that instance. The
- * function below extract the code of the last error that occurred in the
- * specified instance.
- *
- * Input:
- * - CNG_inst : CNG enc/dec instance
- *
- * Return value : Error code
- */
-
-WebRtc_Word16 WebRtcCng_GetErrorCodeEnc(CNG_enc_inst *cng_inst);
-WebRtc_Word16 WebRtcCng_GetErrorCodeDec(CNG_dec_inst *cng_inst);
-
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_INTERFACE_WEBRTC_CNG_H_
diff --git a/trunk/src/modules/audio_coding/codecs/cng/test/CNG.cc b/trunk/src/modules/audio_coding/codecs/cng/test/CNG.cc
deleted file mode 100644
index b247831..0000000
--- a/trunk/src/modules/audio_coding/codecs/cng/test/CNG.cc
+++ /dev/null
@@ -1,225 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * CNG.cpp : Defines the entry point for the console application.
- */
-
-#include <stdlib.h>
-#include <string.h>
-#include "stdafx.h"
-#include "webrtc_cng.h"
-#include "webrtc_vad.h"
-
-CNG_enc_inst *e_inst;
-CNG_dec_inst *d_inst;
-
-VadInst *vinst;
-//#define ASSIGN
-
-short anaSpeech[WEBRTC_CNG_MAX_OUTSIZE_ORDER], genSpeech[WEBRTC_CNG_MAX_OUTSIZE_ORDER], state[WEBRTC_CNG_MAX_OUTSIZE_ORDER];
-unsigned char SIDpkt[114];
-
-int main(int argc, char* argv[])
-{
- FILE * infile, *outfile, *statefile;
- short res=0,errtype;
- /*float time=0.0;*/
-
- WebRtcVad_Create(&vinst);
- WebRtcVad_Init(vinst);
-
- short size;
- int samps=0;
-
- if (argc < 5){
- printf("Usage:\n CNG.exe infile outfile samplingfreq(Hz) interval(ms) order\n\n");
- return(0);
- }
-
- infile=fopen(argv[1],"rb");
- if (infile==NULL){
- printf("file %s does not exist\n",argv[1]);
- return(0);
- }
- outfile=fopen(argv[2],"wb");
- statefile=fopen("CNGVAD.d","wb");
- if (outfile==NULL){
- printf("file %s could not be created\n",argv[2]);
- return(0);
- }
-
- unsigned int fs=16000;
- short frameLen=fs/50;
-
-#ifndef ASSIGN
- res=WebRtcCng_CreateEnc(&e_inst);
- if (res < 0) {
- /* exit if returned with error */
- errtype=WebRtcCng_GetErrorCodeEnc(e_inst);
- fprintf(stderr,"\n\n Error in initialization: %d.\n\n", errtype);
- exit(EXIT_FAILURE);
- }
- res=WebRtcCng_CreateDec(&d_inst);
- if (res < 0) {
- /* exit if returned with error */
- errtype=WebRtcCng_GetErrorCodeDec(d_inst);
- fprintf(stderr,"\n\n Error in initialization: %d.\n\n", errtype);
- exit(EXIT_FAILURE);
- }
-
-#else
-
- // Test the Assign-functions
- int Esize, Dsize;
- void *Eaddr, *Daddr;
-
- res=WebRtcCng_AssignSizeEnc(&Esize);
- res=WebRtcCng_AssignSizeDec(&Dsize);
- Eaddr=malloc(Esize);
- Daddr=malloc(Dsize);
-
- res=WebRtcCng_AssignEnc(&e_inst, Eaddr);
- if (res < 0) {
- /* exit if returned with error */
- errtype=WebRtcCng_GetErrorCodeEnc(e_inst);
- fprintf(stderr,"\n\n Error in initialization: %d.\n\n", errtype);
- exit(EXIT_FAILURE);
- }
-
- res=WebRtcCng_AssignDec(&d_inst, Daddr);
- if (res < 0) {
- /* exit if returned with error */
- errtype=WebRtcCng_GetErrorCodeDec(d_inst);
- fprintf(stderr,"\n\n Error in initialization: %d.\n\n", errtype);
- exit(EXIT_FAILURE);
- }
-
-#endif
-
- res=WebRtcCng_InitEnc(e_inst,atoi(argv[3]),atoi(argv[4]),atoi(argv[5]));
- if (res < 0) {
- /* exit if returned with error */
- errtype=WebRtcCng_GetErrorCodeEnc(e_inst);
- fprintf(stderr,"\n\n Error in initialization: %d.\n\n", errtype);
- exit(EXIT_FAILURE);
- }
-
- res=WebRtcCng_InitDec(d_inst);
- if (res < 0) {
- /* exit if returned with error */
- errtype=WebRtcCng_GetErrorCodeDec(d_inst);
- fprintf(stderr,"\n\n Error in initialization: %d.\n\n", errtype);
- exit(EXIT_FAILURE);
- }
-
-
- static bool firstSilent=true;
-
- int numSamp=0;
- int speech=0;
- int silent=0;
- long cnt=0;
-
- while(fread(anaSpeech,2,frameLen,infile)==frameLen){
-
- cnt++;
- if (cnt==60){
- cnt=60;
- }
- /* time+=(float)frameLen/fs;
- numSamp+=frameLen;
- float temp[640];
- for(unsigned int j=0;j<frameLen;j++)
- temp[j]=(float)anaSpeech[j]; */
-
- // if(!WebRtcVad_Process(vinst, fs, anaSpeech, frameLen)){
-
-
- if(1){ // Do CNG coding of entire file
-
- // if(!((anaSpeech[0]==0)&&(anaSpeech[1]==0)&&(anaSpeech[2]==0))){
- if(firstSilent){
- res = WebRtcCng_Encode(e_inst, anaSpeech, frameLen/2, SIDpkt,&size,1);
- if (res < 0) {
- /* exit if returned with error */
- errtype=WebRtcCng_GetErrorCodeEnc(e_inst);
- fprintf(stderr,"\n\n Error in encoder: %d.\n\n", errtype);
- exit(EXIT_FAILURE);
- }
-
-
- firstSilent=false;
-
- res=WebRtcCng_Encode(e_inst, &anaSpeech[frameLen/2], frameLen/2, SIDpkt,&size,1);
- if (res < 0) {
- /* exit if returned with error */
- errtype=WebRtcCng_GetErrorCodeEnc(e_inst);
- fprintf(stderr,"\n\n Error in encoder: %d.\n\n", errtype);
- exit(EXIT_FAILURE);
- }
-
- }
- else{
- res=WebRtcCng_Encode(e_inst, anaSpeech, frameLen/2, SIDpkt,&size,0);
- if (res < 0) {
- /* exit if returned with error */
- errtype=WebRtcCng_GetErrorCodeEnc(e_inst);
- fprintf(stderr,"\n\n Error in encoder: %d.\n\n", errtype);
- exit(EXIT_FAILURE);
- }
- res=WebRtcCng_Encode(e_inst, &anaSpeech[frameLen/2], frameLen/2, SIDpkt,&size,0);
- if (res < 0) {
- /* exit if returned with error */
- errtype=WebRtcCng_GetErrorCodeEnc(e_inst);
- fprintf(stderr,"\n\n Error in encoder: %d.\n\n", errtype);
- exit(EXIT_FAILURE);
- }
- }
-
- if(size>0){
- res=WebRtcCng_UpdateSid(d_inst,SIDpkt, size);
- if (res < 0) {
- /* exit if returned with error */
- errtype=WebRtcCng_GetErrorCodeDec(d_inst);
- fprintf(stderr,"\n\n Error in decoder: %d.\n\n", errtype);
- exit(EXIT_FAILURE);
- }
- }
- res=WebRtcCng_Generate(d_inst,genSpeech, frameLen,0);
- if (res < 0) {
- /* exit if returned with error */
- errtype=WebRtcCng_GetErrorCodeDec(d_inst);
- fprintf(stderr,"\n\n Error in decoder: %d.\n\n", errtype);
- exit(EXIT_FAILURE);
- }
- memcpy(state,anaSpeech,2*frameLen);
- }
- else{
- firstSilent=true;
- memcpy(genSpeech,anaSpeech,2*frameLen);
-
- memset(anaSpeech,0,frameLen*2);
- memset(state,0,frameLen*2);
-
- }
- fwrite(genSpeech,2,frameLen,outfile);
- fwrite(state,2,frameLen,statefile);
-
- }
-
- fclose(infile);
- fclose(outfile);
- fclose(statefile);
- return 0;
-}
-
-
-
diff --git a/trunk/src/modules/audio_coding/codecs/cng/test/StdAfx.cc b/trunk/src/modules/audio_coding/codecs/cng/test/StdAfx.cc
deleted file mode 100644
index 995e510..0000000
--- a/trunk/src/modules/audio_coding/codecs/cng/test/StdAfx.cc
+++ /dev/null
@@ -1,18 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// stdafx.cpp : source file that includes just the standard includes
-// CNG.pch will be the pre-compiled header
-// stdafx.obj will contain the pre-compiled type information
-
-#include "stdafx.h"
-
-// TODO: reference any additional headers you need in STDAFX.H
-// and not in this file
diff --git a/trunk/src/modules/audio_coding/codecs/cng/test/StdAfx.h b/trunk/src/modules/audio_coding/codecs/cng/test/StdAfx.h
deleted file mode 100644
index dd6c445..0000000
--- a/trunk/src/modules/audio_coding/codecs/cng/test/StdAfx.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// stdafx.h : include file for standard system include files,
-// or project specific include files that are used frequently, but
-// are changed infrequently
-//
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_TEST_STDAFX_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_MAIN_TEST_STDAFX_H_
-
-#if _MSC_VER > 1000
-#pragma once
-#endif // _MSC_VER > 1000
-
-#define WIN32_LEAN_AND_MEAN // Exclude rarely-used stuff from Windows headers
-
-#include <stdio.h>
-
-// TODO: reference additional headers your program requires here
-
-//{{AFX_INSERT_LOCATION}}
-// Microsoft Visual C++ will insert additional declarations immediately before the previous line.
-
-#endif // !defined(AFX_STDAFX_H__DE2097A7_569B_42A0_A615_41BF352D6FFB__INCLUDED_)
diff --git a/trunk/src/modules/audio_coding/codecs/cng/webrtc_cng.c b/trunk/src/modules/audio_coding/codecs/cng/webrtc_cng.c
deleted file mode 100644
index 65f6672..0000000
--- a/trunk/src/modules/audio_coding/codecs/cng/webrtc_cng.c
+++ /dev/null
@@ -1,732 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-#include <string.h>
-#include <stdlib.h>
-
-#include "webrtc_cng.h"
-#include "signal_processing_library.h"
-#include "cng_helpfuns.h"
-#include "stdio.h"
-
-
-typedef struct WebRtcCngDecInst_t_ {
-
- WebRtc_UWord32 dec_seed;
- WebRtc_Word32 dec_target_energy;
- WebRtc_Word32 dec_used_energy;
- WebRtc_Word16 dec_target_reflCoefs[WEBRTC_CNG_MAX_LPC_ORDER+1];
- WebRtc_Word16 dec_used_reflCoefs[WEBRTC_CNG_MAX_LPC_ORDER+1];
- WebRtc_Word16 dec_filtstate[WEBRTC_CNG_MAX_LPC_ORDER+1];
- WebRtc_Word16 dec_filtstateLow[WEBRTC_CNG_MAX_LPC_ORDER+1];
- WebRtc_Word16 dec_Efiltstate[WEBRTC_CNG_MAX_LPC_ORDER+1];
- WebRtc_Word16 dec_EfiltstateLow[WEBRTC_CNG_MAX_LPC_ORDER+1];
- WebRtc_Word16 dec_order;
- WebRtc_Word16 dec_target_scale_factor; /*Q29*/
- WebRtc_Word16 dec_used_scale_factor; /*Q29*/
- WebRtc_Word16 target_scale_factor; /* Q13 */
- WebRtc_Word16 errorcode;
- WebRtc_Word16 initflag;
-
-} WebRtcCngDecInst_t;
-
-
-typedef struct WebRtcCngEncInst_t_ {
-
- WebRtc_Word16 enc_nrOfCoefs;
- WebRtc_Word16 enc_sampfreq;
- WebRtc_Word16 enc_interval;
- WebRtc_Word16 enc_msSinceSID;
- WebRtc_Word32 enc_Energy;
- WebRtc_Word16 enc_reflCoefs[WEBRTC_CNG_MAX_LPC_ORDER+1];
- WebRtc_Word32 enc_corrVector[WEBRTC_CNG_MAX_LPC_ORDER+1];
- WebRtc_Word16 enc_filtstate[WEBRTC_CNG_MAX_LPC_ORDER+1];
- WebRtc_Word16 enc_filtstateLow[WEBRTC_CNG_MAX_LPC_ORDER+1];
- WebRtc_UWord32 enc_seed;
- WebRtc_Word16 errorcode;
- WebRtc_Word16 initflag;
-
-} WebRtcCngEncInst_t;
-
-const WebRtc_Word32 WebRtcCng_kDbov[94]={
- 1081109975, 858756178, 682134279, 541838517, 430397633, 341876992,
- 271562548, 215709799, 171344384, 136103682, 108110997, 85875618,
- 68213428, 54183852, 43039763, 34187699, 27156255, 21570980,
- 17134438, 13610368, 10811100, 8587562, 6821343, 5418385,
- 4303976, 3418770, 2715625, 2157098, 1713444, 1361037,
- 1081110, 858756, 682134, 541839, 430398, 341877,
- 271563, 215710, 171344, 136104, 108111, 85876,
- 68213, 54184, 43040, 34188, 27156, 21571,
- 17134, 13610, 10811, 8588, 6821, 5418,
- 4304, 3419, 2716, 2157, 1713, 1361,
- 1081, 859, 682, 542, 430, 342,
- 272, 216, 171, 136, 108, 86,
- 68, 54, 43, 34, 27, 22,
- 17, 14, 11, 9, 7, 5,
- 4, 3, 3, 2, 2, 1,
- 1, 1, 1, 1
-};
-const WebRtc_Word16 WebRtcCng_kCorrWindow[WEBRTC_CNG_MAX_LPC_ORDER] = {
- 32702, 32636, 32570, 32505, 32439, 32374,
- 32309, 32244, 32179, 32114, 32049, 31985
-};
-
-/****************************************************************************
- * WebRtcCng_Version(...)
- *
- * These functions returns the version name (string must be at least
- * 500 characters long)
- *
- * Output:
- * - version : Pointer to character string
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcCng_Version(char *version)
-{
- strcpy((char*)version,(const char*)"1.2.0\n");
- return(0);
-}
-
-
-/****************************************************************************
- * WebRtcCng_AssignSizeEnc/Dec(...)
- *
- * These functions get the size needed for storing the instance for encoder
- * and decoder, respectively
- *
- * Input/Output:
- * - sizeinbytes : Pointer to integer where the size is returned
- *
- * Return value : 0
- */
-
-WebRtc_Word16 WebRtcCng_AssignSizeEnc(int *sizeinbytes)
-{
- *sizeinbytes=sizeof(WebRtcCngEncInst_t)*2/sizeof(WebRtc_Word16);
- return(0);
-}
-
-WebRtc_Word16 WebRtcCng_AssignSizeDec(int *sizeinbytes)
-{
- *sizeinbytes=sizeof(WebRtcCngDecInst_t)*2/sizeof(WebRtc_Word16);
- return(0);
-}
-
-
-/****************************************************************************
- * WebRtcCng_AssignEnc/Dec(...)
- *
- * These functions Assignes memory for the instances.
- *
- * Input:
- * - CNG_inst_Addr : Adress to where to assign memory
- * Output:
- * - inst : Pointer to the instance that should be created
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcCng_AssignEnc(CNG_enc_inst **inst, void *CNG_inst_Addr)
-{
- if (CNG_inst_Addr!=NULL) {
- *inst = (CNG_enc_inst*)CNG_inst_Addr;
- (*(WebRtcCngEncInst_t**) inst)->errorcode = 0;
- (*(WebRtcCngEncInst_t**) inst)->initflag = 0;
- return(0);
- } else {
- /* The memory could not be allocated */
- return(-1);
- }
-}
-
-WebRtc_Word16 WebRtcCng_AssignDec(CNG_dec_inst **inst, void *CNG_inst_Addr)
-{
- if (CNG_inst_Addr!=NULL) {
- *inst = (CNG_dec_inst*)CNG_inst_Addr;
- (*(WebRtcCngDecInst_t**) inst)->errorcode = 0;
- (*(WebRtcCngDecInst_t**) inst)->initflag = 0;
- return(0);
- } else {
- /* The memory could not be allocated */
- return(-1);
- }
-}
-
-
-/****************************************************************************
- * WebRtcCng_CreateEnc/Dec(...)
- *
- * These functions create an instance to the specified structure
- *
- * Input:
- * - XXX_inst : Pointer to created instance that should be created
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcCng_CreateEnc(CNG_enc_inst **cng_inst)
-{
- *cng_inst=(CNG_enc_inst*)malloc(sizeof(WebRtcCngEncInst_t));
- if(cng_inst!=NULL) {
- (*(WebRtcCngEncInst_t**) cng_inst)->errorcode = 0;
- (*(WebRtcCngEncInst_t**) cng_inst)->initflag = 0;
- return(0);
- }
- else {
- /* The memory could not be allocated */
- return(-1);
- }
-}
-
-WebRtc_Word16 WebRtcCng_CreateDec(CNG_dec_inst **cng_inst)
-{
- *cng_inst=(CNG_dec_inst*)malloc(sizeof(WebRtcCngDecInst_t));
- if(cng_inst!=NULL) {
- (*(WebRtcCngDecInst_t**) cng_inst)->errorcode = 0;
- (*(WebRtcCngDecInst_t**) cng_inst)->initflag = 0;
- return(0);
- }
- else {
- /* The memory could not be allocated */
- return(-1);
- }
-}
-
-
-/****************************************************************************
- * WebRtcCng_InitEnc/Dec(...)
- *
- * This function initializes a instance
- *
- * Input:
- * - cng_inst : Instance that should be initialized
- *
- * - fs : 8000 for narrowband and 16000 for wideband
- * - interval : generate SID data every interval ms
- * - quality : TBD
- *
- * Output:
- * - cng_inst : Initialized instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-
-WebRtc_Word16 WebRtcCng_InitEnc(CNG_enc_inst *cng_inst,
- WebRtc_Word16 fs,
- WebRtc_Word16 interval,
- WebRtc_Word16 quality)
-{
- int i;
-
- WebRtcCngEncInst_t* inst=(WebRtcCngEncInst_t*)cng_inst;
-
- memset(inst, 0, sizeof(WebRtcCngEncInst_t));
-
- /* Check LPC order */
-
- if (quality>WEBRTC_CNG_MAX_LPC_ORDER) {
- inst->errorcode = CNG_DISALLOWED_LPC_ORDER;
- return (-1);
- }
-
- if (fs<=0) {
- inst->errorcode = CNG_DISALLOWED_SAMPLING_FREQUENCY;
- return (-1);
- }
-
- inst->enc_sampfreq=fs;
- inst->enc_interval=interval;
- inst->enc_nrOfCoefs=quality;
- inst->enc_msSinceSID=0;
- inst->enc_seed=7777; /*For debugging only*/
- inst->enc_Energy=0;
- for(i=0;i<(WEBRTC_CNG_MAX_LPC_ORDER+1);i++){
- inst->enc_reflCoefs[i]=0;
- inst->enc_corrVector[i]=0;
- }
- inst->initflag=1;
-
- return(0);
-}
-
-WebRtc_Word16 WebRtcCng_InitDec(CNG_dec_inst *cng_inst)
-{
- int i;
-
- WebRtcCngDecInst_t* inst=(WebRtcCngDecInst_t*)cng_inst;
-
- memset(inst, 0, sizeof(WebRtcCngDecInst_t));
- inst->dec_seed=7777; /*For debugging only*/
- inst->dec_order=5;
- inst->dec_target_scale_factor=0;
- inst->dec_used_scale_factor=0;
- for(i=0;i<(WEBRTC_CNG_MAX_LPC_ORDER+1);i++){
- inst->dec_filtstate[i]=0;
- inst->dec_target_reflCoefs[i]=0;
- inst->dec_used_reflCoefs[i]=0;
- }
- inst->dec_target_reflCoefs[0]=0;
- inst->dec_used_reflCoefs[0]=0;
- inst ->dec_used_energy=0;
- inst->initflag=1;
-
- return(0);
-}
-
-/****************************************************************************
- * WebRtcCng_FreeEnc/Dec(...)
- *
- * These functions frees the dynamic memory of a specified instance
- *
- * Input:
- * - cng_inst : Pointer to created instance that should be freed
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-
-WebRtc_Word16 WebRtcCng_FreeEnc(CNG_enc_inst *cng_inst)
-{
- free(cng_inst);
- return(0);
-}
-
-WebRtc_Word16 WebRtcCng_FreeDec(CNG_dec_inst *cng_inst)
-{
- free(cng_inst);
- return(0);
-}
-
-
-
-/****************************************************************************
- * WebRtcCng_Encode(...)
- *
- * These functions analyzes background noise
- *
- * Input:
- * - cng_inst : Pointer to created instance
- * - speech : Signal (noise) to be analyzed
- * - nrOfSamples : Size of speech vector
- * - bytesOut : Nr of bytes to transmit, might be 0
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-WebRtc_Word16 WebRtcCng_Encode(CNG_enc_inst *cng_inst,
- WebRtc_Word16 *speech,
- WebRtc_Word16 nrOfSamples,
- WebRtc_UWord8* SIDdata,
- WebRtc_Word16* bytesOut,
- WebRtc_Word16 forceSID)
-{
- WebRtcCngEncInst_t* inst=(WebRtcCngEncInst_t*)cng_inst;
-
- WebRtc_Word16 arCoefs[WEBRTC_CNG_MAX_LPC_ORDER+1];
- WebRtc_Word32 corrVector[WEBRTC_CNG_MAX_LPC_ORDER+1];
- WebRtc_Word16 refCs[WEBRTC_CNG_MAX_LPC_ORDER+1];
- WebRtc_Word16 hanningW[WEBRTC_CNG_MAX_OUTSIZE_ORDER];
- WebRtc_Word16 ReflBeta=19661; /*0.6 in q15*/
- WebRtc_Word16 ReflBetaComp=13107; /*0.4 in q15*/
- WebRtc_Word32 outEnergy;
- int outShifts;
- int i, stab;
- int acorrScale;
- int index;
- WebRtc_Word16 ind,factor;
- WebRtc_Word32 *bptr, blo, bhi;
- WebRtc_Word16 negate;
- const WebRtc_Word16 *aptr;
-
- WebRtc_Word16 speechBuf[WEBRTC_CNG_MAX_OUTSIZE_ORDER];
-
-
- /* check if encoder initiated */
- if (inst->initflag != 1) {
- inst->errorcode = CNG_ENCODER_NOT_INITIATED;
- return (-1);
- }
-
-
- /* check framesize */
- if (nrOfSamples>WEBRTC_CNG_MAX_OUTSIZE_ORDER) {
- inst->errorcode = CNG_DISALLOWED_FRAME_SIZE;
- return (-1);
- }
-
-
- for(i=0;i<nrOfSamples;i++){
- speechBuf[i]=speech[i];
- }
-
- factor=nrOfSamples;
-
- /* Calculate energy and a coefficients */
- outEnergy =WebRtcSpl_Energy(speechBuf, nrOfSamples, &outShifts);
- while(outShifts>0){
- if(outShifts>5){ /*We can only do 5 shifts without destroying accuracy in division factor*/
- outEnergy<<=(outShifts-5);
- outShifts=5;
- }
- else{
- factor/=2;
- outShifts--;
- }
- }
- outEnergy=WebRtcSpl_DivW32W16(outEnergy,factor);
-
- if (outEnergy > 1){
- /* Create Hanning Window */
- WebRtcSpl_GetHanningWindow(hanningW, nrOfSamples/2);
- for( i=0;i<(nrOfSamples/2);i++ )
- hanningW[nrOfSamples-i-1]=hanningW[i];
-
- WebRtcSpl_ElementwiseVectorMult(speechBuf, hanningW, speechBuf, nrOfSamples, 14);
-
- WebRtcSpl_AutoCorrelation( speechBuf, nrOfSamples, inst->enc_nrOfCoefs, corrVector, &acorrScale );
-
- if( *corrVector==0 )
- *corrVector = WEBRTC_SPL_WORD16_MAX;
-
- /* Adds the bandwidth expansion */
- aptr = WebRtcCng_kCorrWindow;
- bptr = corrVector;
-
- // (zzz) lpc16_1 = 17+1+820+2+2 = 842 (ordo2=700)
- for( ind=0; ind<inst->enc_nrOfCoefs; ind++ )
- {
- // The below code multiplies the 16 b corrWindow values (Q15) with
- // the 32 b corrvector (Q0) and shifts the result down 15 steps.
-
- negate = *bptr<0;
- if( negate )
- *bptr = -*bptr;
-
- blo = (WebRtc_Word32)*aptr * (*bptr & 0xffff);
- bhi = ((blo >> 16) & 0xffff) + ((WebRtc_Word32)(*aptr++) * ((*bptr >> 16) & 0xffff));
- blo = (blo & 0xffff) | ((bhi & 0xffff) << 16);
-
- *bptr = (( (bhi>>16) & 0x7fff) << 17) | ((WebRtc_UWord32)blo >> 15);
- if( negate )
- *bptr = -*bptr;
- bptr++;
- }
-
- // end of bandwidth expansion
-
- stab=WebRtcSpl_LevinsonDurbin(corrVector, arCoefs, refCs, inst->enc_nrOfCoefs);
-
- if(!stab){
- // disregard from this frame
- *bytesOut=0;
- return(0);
- }
-
- }
- else {
- for(i=0;i<inst->enc_nrOfCoefs; i++)
- refCs[i]=0;
- }
-
- if(forceSID){
- /*Read instantaneous values instead of averaged*/
- for(i=0;i<inst->enc_nrOfCoefs;i++)
- inst->enc_reflCoefs[i]=refCs[i];
- inst->enc_Energy=outEnergy;
- }
- else{
- /*Average history with new values*/
- for(i=0;i<(inst->enc_nrOfCoefs);i++){
- inst->enc_reflCoefs[i]=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->enc_reflCoefs[i],ReflBeta,15);
- inst->enc_reflCoefs[i]+=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(refCs[i],ReflBetaComp,15);
- }
- inst->enc_Energy=(outEnergy>>2)+(inst->enc_Energy>>1)+(inst->enc_Energy>>2);
- }
-
-
- if(inst->enc_Energy<1){
- inst->enc_Energy=1;
- }
-
- if((inst->enc_msSinceSID>(inst->enc_interval-1))||forceSID){
-
- /* Search for best dbov value */
- index=0;
- for(i=1;i<93;i++){
- /* Always round downwards */
- if((inst->enc_Energy-WebRtcCng_kDbov[i])>0){
- index=i;
- break;
- }
- }
- if((i==93)&&(index==0))
- index=94;
- SIDdata[0]=index;
-
-
- /* Quantize coefs with tweak for WebRtc implementation of RFC3389 */
- if(inst->enc_nrOfCoefs==WEBRTC_CNG_MAX_LPC_ORDER){
- for(i=0;i<inst->enc_nrOfCoefs;i++){
- SIDdata[i+1]=((inst->enc_reflCoefs[i]+128)>>8); /* Q15 to Q7*/ /* +127 */
- }
- }else{
- for(i=0;i<inst->enc_nrOfCoefs;i++){
- SIDdata[i+1]=(127+((inst->enc_reflCoefs[i]+128)>>8)); /* Q15 to Q7*/ /* +127 */
- }
- }
-
- inst->enc_msSinceSID=0;
- *bytesOut=inst->enc_nrOfCoefs+1;
-
- inst->enc_msSinceSID+=(1000*nrOfSamples)/inst->enc_sampfreq;
- return(inst->enc_nrOfCoefs+1);
- }else{
- inst->enc_msSinceSID+=(1000*nrOfSamples)/inst->enc_sampfreq;
- *bytesOut=0;
- return(0);
- }
-}
-
-
-/****************************************************************************
- * WebRtcCng_UpdateSid(...)
- *
- * These functions updates the CN state, when a new SID packet arrives
- *
- * Input:
- * - cng_inst : Pointer to created instance that should be freed
- * - SID : SID packet, all headers removed
- * - length : Length in bytes of SID packet
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcCng_UpdateSid(CNG_dec_inst *cng_inst,
- WebRtc_UWord8 *SID,
- WebRtc_Word16 length)
-{
-
- WebRtcCngDecInst_t* inst=(WebRtcCngDecInst_t*)cng_inst;
- WebRtc_Word16 refCs[WEBRTC_CNG_MAX_LPC_ORDER];
- WebRtc_Word32 targetEnergy;
- int i;
-
- if (inst->initflag != 1) {
- inst->errorcode = CNG_DECODER_NOT_INITIATED;
- return (-1);
- }
-
- /*Throw away reflection coefficients of higher order than we can handle*/
- if(length> (WEBRTC_CNG_MAX_LPC_ORDER+1))
- length=WEBRTC_CNG_MAX_LPC_ORDER+1;
-
- inst->dec_order=length-1;
-
- if(SID[0]>93)
- SID[0]=93;
- targetEnergy=WebRtcCng_kDbov[SID[0]];
- /* Take down target energy to 75% */
- targetEnergy=targetEnergy>>1;
- targetEnergy+=targetEnergy>>2;
-
- inst->dec_target_energy=targetEnergy;
-
- /* Reconstruct coeffs with tweak for WebRtc implementation of RFC3389 */
- if(inst->dec_order==WEBRTC_CNG_MAX_LPC_ORDER){
- for(i=0;i<(inst->dec_order);i++){
- refCs[i]=SID[i+1]<<8; /* Q7 to Q15*/
- inst->dec_target_reflCoefs[i]=refCs[i];
- }
- }else{
- for(i=0;i<(inst->dec_order);i++){
- refCs[i]=(SID[i+1]-127)<<8; /* Q7 to Q15*/
- inst->dec_target_reflCoefs[i]=refCs[i];
- }
- }
-
- for(i=(inst->dec_order);i<WEBRTC_CNG_MAX_LPC_ORDER;i++){
- refCs[i]=0;
- inst->dec_target_reflCoefs[i]=refCs[i];
- }
-
- return(0);
-}
-
-
-/****************************************************************************
- * WebRtcCng_Generate(...)
- *
- * These functions generates CN data when needed
- *
- * Input:
- * - cng_inst : Pointer to created instance that should be freed
- * - outData : pointer to area to write CN data
- * - nrOfSamples : How much data to generate
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-WebRtc_Word16 WebRtcCng_Generate(CNG_dec_inst *cng_inst,
- WebRtc_Word16 *outData,
- WebRtc_Word16 nrOfSamples,
- WebRtc_Word16 new_period)
-{
- WebRtcCngDecInst_t* inst=(WebRtcCngDecInst_t*)cng_inst;
-
- int i;
- WebRtc_Word16 excitation[WEBRTC_CNG_MAX_OUTSIZE_ORDER];
- WebRtc_Word16 low[WEBRTC_CNG_MAX_OUTSIZE_ORDER];
- WebRtc_Word16 lpPoly[WEBRTC_CNG_MAX_LPC_ORDER+1];
- WebRtc_Word16 ReflBetaStd=26214; /*0.8 in q15*/
- WebRtc_Word16 ReflBetaCompStd=6553; /*0.2in q15*/
- WebRtc_Word16 ReflBetaNewP=19661; /*0.6 in q15*/
- WebRtc_Word16 ReflBetaCompNewP=13107; /*0.4 in q15*/
- WebRtc_Word16 Beta,BetaC, tmp1, tmp2, tmp3;
- WebRtc_Word32 targetEnergy;
- WebRtc_Word16 En;
- WebRtc_Word16 temp16;
-
- if (nrOfSamples>WEBRTC_CNG_MAX_OUTSIZE_ORDER) {
- inst->errorcode = CNG_DISALLOWED_FRAME_SIZE;
- return (-1);
- }
-
-
- if (new_period) {
- inst->dec_used_scale_factor=inst->dec_target_scale_factor;
- Beta=ReflBetaNewP;
- BetaC=ReflBetaCompNewP;
- } else {
- Beta=ReflBetaStd;
- BetaC=ReflBetaCompStd;
- }
-
- /*Here we use a 0.5 weighting, should possibly be modified to 0.6*/
- tmp1=inst->dec_used_scale_factor<<2; /* Q13->Q15 */
- tmp2=inst->dec_target_scale_factor<<2; /* Q13->Q15 */
- tmp3=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp1,Beta,15);
- tmp3+=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp2,BetaC,15);
- inst->dec_used_scale_factor=tmp3>>2; /* Q15->Q13 */
-
- inst->dec_used_energy=inst->dec_used_energy>>1;
- inst->dec_used_energy+=inst->dec_target_energy>>1;
-
-
- /* Do the same for the reflection coeffs */
- for (i=0;i<WEBRTC_CNG_MAX_LPC_ORDER;i++) {
- inst->dec_used_reflCoefs[i]=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->dec_used_reflCoefs[i],Beta,15);
- inst->dec_used_reflCoefs[i]+=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->dec_target_reflCoefs[i],BetaC,15);
- }
-
- /* Compute the polynomial coefficients */
- WebRtcCng_K2a16(inst->dec_used_reflCoefs, WEBRTC_CNG_MAX_LPC_ORDER, lpPoly);
-
- /***/
-
- targetEnergy=inst->dec_used_energy;
-
- // Calculate scaling factor based on filter energy
- En=8192; //1.0 in Q13
- for (i=0; i<(WEBRTC_CNG_MAX_LPC_ORDER); i++) {
-
- // Floating point value for reference
- // E*=1.0-((float)inst->dec_used_reflCoefs[i]/32768.0)*((float)inst->dec_used_reflCoefs[i]/32768.0);
-
- // Same in fixed point
- temp16=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->dec_used_reflCoefs[i],inst->dec_used_reflCoefs[i],15); // K(i).^2 in Q15
- temp16=0x7fff - temp16; // 1 - K(i).^2 in Q15
- En=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(En,temp16,15);
-
- }
-
- //float scaling= sqrt(E*inst->dec_target_energy/((1<<24)));
-
- //Calculate sqrt(En*target_energy/exctiation energy)
-
- targetEnergy=WebRtcSpl_Sqrt(inst->dec_used_energy);
-
- En=(WebRtc_Word16)WebRtcSpl_Sqrt(En)<<6; //We are missing a factor sqrt(2) here
- En=(En*3)>>1; //1.5 estimates sqrt(2)
-
- inst->dec_used_scale_factor=(WebRtc_Word16)((En*targetEnergy)>>12);
-
-
- /***/
-
- /*Generate excitation*/
- /*Excitation energy per sample is 2.^24 - Q13 N(0,1) */
- for(i=0;i<nrOfSamples;i++){
- excitation[i]=WebRtcSpl_RandN(&inst->dec_seed)>>1;
- }
-
- /*Scale to correct energy*/
- WebRtcSpl_ScaleVector(excitation, excitation, inst->dec_used_scale_factor, nrOfSamples, 13);
-
- WebRtcSpl_FilterAR(
- lpPoly, /* Coefficients in Q12 */
- WEBRTC_CNG_MAX_LPC_ORDER+1,
- excitation, /* Speech samples */
- nrOfSamples,
- inst->dec_filtstate, /* State preservation */
- WEBRTC_CNG_MAX_LPC_ORDER,
- inst->dec_filtstateLow, /* State preservation */
- WEBRTC_CNG_MAX_LPC_ORDER,
- outData, /* Filtered speech samples */
- low,
- nrOfSamples
- );
-
- return(0);
-
-}
-
-
-
-/****************************************************************************
- * WebRtcCng_GetErrorCodeEnc/Dec(...)
- *
- * This functions can be used to check the error code of a CNG instance. When
- * a function returns -1 a error code will be set for that instance. The
- * function below extract the code of the last error that occured in the
- * specified instance.
- *
- * Input:
- * - CNG_inst : CNG enc/dec instance
- *
- * Return value : Error code
- */
-
-WebRtc_Word16 WebRtcCng_GetErrorCodeEnc(CNG_enc_inst *cng_inst)
-{
-
- /* typecast pointer to real structure */
- WebRtcCngEncInst_t* inst=(WebRtcCngEncInst_t*)cng_inst;
-
- return inst->errorcode;
-}
-
-WebRtc_Word16 WebRtcCng_GetErrorCodeDec(CNG_dec_inst *cng_inst)
-{
-
- /* typecast pointer to real structure */
- WebRtcCngDecInst_t* inst=(WebRtcCngDecInst_t*)cng_inst;
-
- return inst->errorcode;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/g711/Android.mk b/trunk/src/modules/audio_coding/codecs/g711/Android.mk
deleted file mode 100644
index 779bb7c..0000000
--- a/trunk/src/modules/audio_coding/codecs/g711/Android.mk
+++ /dev/null
@@ -1,40 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_g711
-LOCAL_MODULE_TAGS := optional
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := \
- g711_interface.c \
- g711.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include \
- $(LOCAL_PATH)/../../../..
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_coding/codecs/g711/g711.c b/trunk/src/modules/audio_coding/codecs/g711/g711.c
deleted file mode 100644
index 954f377..0000000
--- a/trunk/src/modules/audio_coding/codecs/g711/g711.c
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- * SpanDSP - a series of DSP components for telephony
- *
- * g711.c - A-law and u-law transcoding routines
- *
- * Written by Steve Underwood <steveu@coppice.org>
- *
- * Copyright (C) 2006 Steve Underwood
- *
- * Despite my general liking of the GPL, I place this code in the
- * public domain for the benefit of all mankind - even the slimy
- * ones who might try to proprietize my work and use it to my
- * detriment.
- *
- * $Id: g711.c,v 1.1 2006/06/07 15:46:39 steveu Exp $
- *
- * Modifications for WebRtc, 2011/04/28, by tlegrand:
- * -Removed unused include files
- * -Changed to use WebRtc types
- * -Added option to run encoder bitexact with ITU-T reference implementation
- */
-
-/*! \file */
-
-#include "g711.h"
-#include "typedefs.h"
-
-/* Copied from the CCITT G.711 specification */
-static const WebRtc_UWord8 ulaw_to_alaw_table[256] =
-{
- 42, 43, 40, 41, 46, 47, 44, 45, 34, 35, 32, 33, 38, 39, 36, 37,
- 58, 59, 56, 57, 62, 63, 60, 61, 50, 51, 48, 49, 54, 55, 52, 53,
- 10, 11, 8, 9, 14, 15, 12, 13, 2, 3, 0, 1, 6, 7, 4, 26,
- 27, 24, 25, 30, 31, 28, 29, 18, 19, 16, 17, 22, 23, 20, 21, 106,
- 104, 105, 110, 111, 108, 109, 98, 99, 96, 97, 102, 103, 100, 101, 122, 120,
- 126, 127, 124, 125, 114, 115, 112, 113, 118, 119, 116, 117, 75, 73, 79, 77,
- 66, 67, 64, 65, 70, 71, 68, 69, 90, 91, 88, 89, 94, 95, 92, 93,
- 82, 82, 83, 83, 80, 80, 81, 81, 86, 86, 87, 87, 84, 84, 85, 85,
- 170, 171, 168, 169, 174, 175, 172, 173, 162, 163, 160, 161, 166, 167, 164, 165,
- 186, 187, 184, 185, 190, 191, 188, 189, 178, 179, 176, 177, 182, 183, 180, 181,
- 138, 139, 136, 137, 142, 143, 140, 141, 130, 131, 128, 129, 134, 135, 132, 154,
- 155, 152, 153, 158, 159, 156, 157, 146, 147, 144, 145, 150, 151, 148, 149, 234,
- 232, 233, 238, 239, 236, 237, 226, 227, 224, 225, 230, 231, 228, 229, 250, 248,
- 254, 255, 252, 253, 242, 243, 240, 241, 246, 247, 244, 245, 203, 201, 207, 205,
- 194, 195, 192, 193, 198, 199, 196, 197, 218, 219, 216, 217, 222, 223, 220, 221,
- 210, 210, 211, 211, 208, 208, 209, 209, 214, 214, 215, 215, 212, 212, 213, 213
-};
-
-/* These transcoding tables are copied from the CCITT G.711 specification. To achieve
- optimal results, do not change them. */
-
-static const WebRtc_UWord8 alaw_to_ulaw_table[256] =
-{
- 42, 43, 40, 41, 46, 47, 44, 45, 34, 35, 32, 33, 38, 39, 36, 37,
- 57, 58, 55, 56, 61, 62, 59, 60, 49, 50, 47, 48, 53, 54, 51, 52,
- 10, 11, 8, 9, 14, 15, 12, 13, 2, 3, 0, 1, 6, 7, 4, 5,
- 26, 27, 24, 25, 30, 31, 28, 29, 18, 19, 16, 17, 22, 23, 20, 21,
- 98, 99, 96, 97, 102, 103, 100, 101, 93, 93, 92, 92, 95, 95, 94, 94,
- 116, 118, 112, 114, 124, 126, 120, 122, 106, 107, 104, 105, 110, 111, 108, 109,
- 72, 73, 70, 71, 76, 77, 74, 75, 64, 65, 63, 63, 68, 69, 66, 67,
- 86, 87, 84, 85, 90, 91, 88, 89, 79, 79, 78, 78, 82, 83, 80, 81,
- 170, 171, 168, 169, 174, 175, 172, 173, 162, 163, 160, 161, 166, 167, 164, 165,
- 185, 186, 183, 184, 189, 190, 187, 188, 177, 178, 175, 176, 181, 182, 179, 180,
- 138, 139, 136, 137, 142, 143, 140, 141, 130, 131, 128, 129, 134, 135, 132, 133,
- 154, 155, 152, 153, 158, 159, 156, 157, 146, 147, 144, 145, 150, 151, 148, 149,
- 226, 227, 224, 225, 230, 231, 228, 229, 221, 221, 220, 220, 223, 223, 222, 222,
- 244, 246, 240, 242, 252, 254, 248, 250, 234, 235, 232, 233, 238, 239, 236, 237,
- 200, 201, 198, 199, 204, 205, 202, 203, 192, 193, 191, 191, 196, 197, 194, 195,
- 214, 215, 212, 213, 218, 219, 216, 217, 207, 207, 206, 206, 210, 211, 208, 209
-};
-
-WebRtc_UWord8 alaw_to_ulaw(WebRtc_UWord8 alaw)
-{
- return alaw_to_ulaw_table[alaw];
-}
-/*- End of function --------------------------------------------------------*/
-
-WebRtc_UWord8 ulaw_to_alaw(WebRtc_UWord8 ulaw)
-{
- return ulaw_to_alaw_table[ulaw];
-}
-/*- End of function --------------------------------------------------------*/
-/*- End of file ------------------------------------------------------------*/
diff --git a/trunk/src/modules/audio_coding/codecs/g711/g711.gypi b/trunk/src/modules/audio_coding/codecs/g711/g711.gypi
deleted file mode 100644
index 763e6db..0000000
--- a/trunk/src/modules/audio_coding/codecs/g711/g711.gypi
+++ /dev/null
@@ -1,65 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'G711',
- 'type': '<(library)',
- 'include_dirs': [
- 'include',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- ],
- },
- 'sources': [
- 'include/g711_interface.h',
- 'g711_interface.c',
- 'g711.c',
- 'g711.h',
- ],
- },
- ], # targets
- # Exclude the test target when building with chromium.
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'g711_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'G711',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'sources': [
- 'g711_unittest.cc',
- ],
- },
- {
- 'target_name': 'g711_test',
- 'type': 'executable',
- 'dependencies': [
- 'G711',
- ],
- 'sources': [
- 'test/testG711.cc',
- ],
- },
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/g711/g711.h b/trunk/src/modules/audio_coding/codecs/g711/g711.h
deleted file mode 100644
index cd5e3d7..0000000
--- a/trunk/src/modules/audio_coding/codecs/g711/g711.h
+++ /dev/null
@@ -1,382 +0,0 @@
-/*
- * SpanDSP - a series of DSP components for telephony
- *
- * g711.h - In line A-law and u-law conversion routines
- *
- * Written by Steve Underwood <steveu@coppice.org>
- *
- * Copyright (C) 2001 Steve Underwood
- *
- * Despite my general liking of the GPL, I place this code in the
- * public domain for the benefit of all mankind - even the slimy
- * ones who might try to proprietize my work and use it to my
- * detriment.
- *
- * $Id: g711.h,v 1.1 2006/06/07 15:46:39 steveu Exp $
- *
- * Modifications for WebRtc, 2011/04/28, by tlegrand:
- * -Changed to use WebRtc types
- * -Changed __inline__ to __inline
- * -Two changes to make implementation bitexact with ITU-T reference implementation
- */
-
-/*! \file */
-
-/*! \page g711_page A-law and mu-law handling
-Lookup tables for A-law and u-law look attractive, until you consider the impact
-on the CPU cache. If it causes a substantial area of your processor cache to get
-hit too often, cache sloshing will severely slow things down. The main reason
-these routines are slow in C, is the lack of direct access to the CPU's "find
-the first 1" instruction. A little in-line assembler fixes that, and the
-conversion routines can be faster than lookup tables, in most real world usage.
-A "find the first 1" instruction is available on most modern CPUs, and is a
-much underused feature.
-
-If an assembly language method of bit searching is not available, these routines
-revert to a method that can be a little slow, so the cache thrashing might not
-seem so bad :(
-
-Feel free to submit patches to add fast "find the first 1" support for your own
-favourite processor.
-
-Look up tables are used for transcoding between A-law and u-law, since it is
-difficult to achieve the precise transcoding procedure laid down in the G.711
-specification by other means.
-*/
-
-#if !defined(_G711_H_)
-#define _G711_H_
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-#include "typedefs.h"
-
-#if defined(__i386__)
-/*! \brief Find the bit position of the highest set bit in a word
- \param bits The word to be searched
- \return The bit number of the highest set bit, or -1 if the word is zero. */
-static __inline__ int top_bit(unsigned int bits)
-{
- int res;
-
- __asm__ __volatile__(" movl $-1,%%edx;\n"
- " bsrl %%eax,%%edx;\n"
- : "=d" (res)
- : "a" (bits));
- return res;
-}
-/*- End of function --------------------------------------------------------*/
-
-/*! \brief Find the bit position of the lowest set bit in a word
- \param bits The word to be searched
- \return The bit number of the lowest set bit, or -1 if the word is zero. */
-static __inline__ int bottom_bit(unsigned int bits)
-{
- int res;
-
- __asm__ __volatile__(" movl $-1,%%edx;\n"
- " bsfl %%eax,%%edx;\n"
- : "=d" (res)
- : "a" (bits));
- return res;
-}
-/*- End of function --------------------------------------------------------*/
-#elif defined(__x86_64__)
-static __inline__ int top_bit(unsigned int bits)
-{
- int res;
-
- __asm__ __volatile__(" movq $-1,%%rdx;\n"
- " bsrq %%rax,%%rdx;\n"
- : "=d" (res)
- : "a" (bits));
- return res;
-}
-/*- End of function --------------------------------------------------------*/
-
-static __inline__ int bottom_bit(unsigned int bits)
-{
- int res;
-
- __asm__ __volatile__(" movq $-1,%%rdx;\n"
- " bsfq %%rax,%%rdx;\n"
- : "=d" (res)
- : "a" (bits));
- return res;
-}
-/*- End of function --------------------------------------------------------*/
-#else
-static __inline int top_bit(unsigned int bits)
-{
- int i;
-
- if (bits == 0)
- return -1;
- i = 0;
- if (bits & 0xFFFF0000)
- {
- bits &= 0xFFFF0000;
- i += 16;
- }
- if (bits & 0xFF00FF00)
- {
- bits &= 0xFF00FF00;
- i += 8;
- }
- if (bits & 0xF0F0F0F0)
- {
- bits &= 0xF0F0F0F0;
- i += 4;
- }
- if (bits & 0xCCCCCCCC)
- {
- bits &= 0xCCCCCCCC;
- i += 2;
- }
- if (bits & 0xAAAAAAAA)
- {
- bits &= 0xAAAAAAAA;
- i += 1;
- }
- return i;
-}
-/*- End of function --------------------------------------------------------*/
-
-static __inline int bottom_bit(unsigned int bits)
-{
- int i;
-
- if (bits == 0)
- return -1;
- i = 32;
- if (bits & 0x0000FFFF)
- {
- bits &= 0x0000FFFF;
- i -= 16;
- }
- if (bits & 0x00FF00FF)
- {
- bits &= 0x00FF00FF;
- i -= 8;
- }
- if (bits & 0x0F0F0F0F)
- {
- bits &= 0x0F0F0F0F;
- i -= 4;
- }
- if (bits & 0x33333333)
- {
- bits &= 0x33333333;
- i -= 2;
- }
- if (bits & 0x55555555)
- {
- bits &= 0x55555555;
- i -= 1;
- }
- return i;
-}
-/*- End of function --------------------------------------------------------*/
-#endif
-
-/* N.B. It is tempting to use look-up tables for A-law and u-law conversion.
- * However, you should consider the cache footprint.
- *
- * A 64K byte table for linear to x-law and a 512 byte table for x-law to
- * linear sound like peanuts these days, and shouldn't an array lookup be
- * real fast? No! When the cache sloshes as badly as this one will, a tight
- * calculation may be better. The messiest part is normally finding the
- * segment, but a little inline assembly can fix that on an i386, x86_64 and
- * many other modern processors.
- */
-
-/*
- * Mu-law is basically as follows:
- *
- * Biased Linear Input Code Compressed Code
- * ------------------------ ---------------
- * 00000001wxyza 000wxyz
- * 0000001wxyzab 001wxyz
- * 000001wxyzabc 010wxyz
- * 00001wxyzabcd 011wxyz
- * 0001wxyzabcde 100wxyz
- * 001wxyzabcdef 101wxyz
- * 01wxyzabcdefg 110wxyz
- * 1wxyzabcdefgh 111wxyz
- *
- * Each biased linear code has a leading 1 which identifies the segment
- * number. The value of the segment number is equal to 7 minus the number
- * of leading 0's. The quantization interval is directly available as the
- * four bits wxyz. * The trailing bits (a - h) are ignored.
- *
- * Ordinarily the complement of the resulting code word is used for
- * transmission, and so the code word is complemented before it is returned.
- *
- * For further information see John C. Bellamy's Digital Telephony, 1982,
- * John Wiley & Sons, pps 98-111 and 472-476.
- */
-
-//#define ULAW_ZEROTRAP /* turn on the trap as per the MIL-STD */
-#define ULAW_BIAS 0x84 /* Bias for linear code. */
-
-/*! \brief Encode a linear sample to u-law
- \param linear The sample to encode.
- \return The u-law value.
-*/
-static __inline WebRtc_UWord8 linear_to_ulaw(int linear)
-{
- WebRtc_UWord8 u_val;
- int mask;
- int seg;
-
- /* Get the sign and the magnitude of the value. */
- if (linear < 0)
- {
- /* WebRtc, tlegrand: -1 added to get bitexact to reference implementation */
- linear = ULAW_BIAS - linear - 1;
- mask = 0x7F;
- }
- else
- {
- linear = ULAW_BIAS + linear;
- mask = 0xFF;
- }
-
- seg = top_bit(linear | 0xFF) - 7;
-
- /*
- * Combine the sign, segment, quantization bits,
- * and complement the code word.
- */
- if (seg >= 8)
- u_val = (WebRtc_UWord8) (0x7F ^ mask);
- else
- u_val = (WebRtc_UWord8) (((seg << 4) | ((linear >> (seg + 3)) & 0xF)) ^ mask);
-#ifdef ULAW_ZEROTRAP
- /* Optional ITU trap */
- if (u_val == 0)
- u_val = 0x02;
-#endif
- return u_val;
-}
-/*- End of function --------------------------------------------------------*/
-
-/*! \brief Decode an u-law sample to a linear value.
- \param ulaw The u-law sample to decode.
- \return The linear value.
-*/
-static __inline WebRtc_Word16 ulaw_to_linear(WebRtc_UWord8 ulaw)
-{
- int t;
-
- /* Complement to obtain normal u-law value. */
- ulaw = ~ulaw;
- /*
- * Extract and bias the quantization bits. Then
- * shift up by the segment number and subtract out the bias.
- */
- t = (((ulaw & 0x0F) << 3) + ULAW_BIAS) << (((int) ulaw & 0x70) >> 4);
- return (WebRtc_Word16) ((ulaw & 0x80) ? (ULAW_BIAS - t) : (t - ULAW_BIAS));
-}
-/*- End of function --------------------------------------------------------*/
-
-/*
- * A-law is basically as follows:
- *
- * Linear Input Code Compressed Code
- * ----------------- ---------------
- * 0000000wxyza 000wxyz
- * 0000001wxyza 001wxyz
- * 000001wxyzab 010wxyz
- * 00001wxyzabc 011wxyz
- * 0001wxyzabcd 100wxyz
- * 001wxyzabcde 101wxyz
- * 01wxyzabcdef 110wxyz
- * 1wxyzabcdefg 111wxyz
- *
- * For further information see John C. Bellamy's Digital Telephony, 1982,
- * John Wiley & Sons, pps 98-111 and 472-476.
- */
-
-#define ALAW_AMI_MASK 0x55
-
-/*! \brief Encode a linear sample to A-law
- \param linear The sample to encode.
- \return The A-law value.
-*/
-static __inline WebRtc_UWord8 linear_to_alaw(int linear)
-{
- int mask;
- int seg;
-
- if (linear >= 0)
- {
- /* Sign (bit 7) bit = 1 */
- mask = ALAW_AMI_MASK | 0x80;
- }
- else
- {
- /* Sign (bit 7) bit = 0 */
- mask = ALAW_AMI_MASK;
- /* WebRtc, tlegrand: Changed from -8 to -1 to get bitexact to reference
- * implementation */
- linear = -linear - 1;
- }
-
- /* Convert the scaled magnitude to segment number. */
- seg = top_bit(linear | 0xFF) - 7;
- if (seg >= 8)
- {
- if (linear >= 0)
- {
- /* Out of range. Return maximum value. */
- return (WebRtc_UWord8) (0x7F ^ mask);
- }
- /* We must be just a tiny step below zero */
- return (WebRtc_UWord8) (0x00 ^ mask);
- }
- /* Combine the sign, segment, and quantization bits. */
- return (WebRtc_UWord8) (((seg << 4) | ((linear >> ((seg) ? (seg + 3) : 4)) & 0x0F)) ^ mask);
-}
-/*- End of function --------------------------------------------------------*/
-
-/*! \brief Decode an A-law sample to a linear value.
- \param alaw The A-law sample to decode.
- \return The linear value.
-*/
-static __inline WebRtc_Word16 alaw_to_linear(WebRtc_UWord8 alaw)
-{
- int i;
- int seg;
-
- alaw ^= ALAW_AMI_MASK;
- i = ((alaw & 0x0F) << 4);
- seg = (((int) alaw & 0x70) >> 4);
- if (seg)
- i = (i + 0x108) << (seg - 1);
- else
- i += 8;
- return (WebRtc_Word16) ((alaw & 0x80) ? i : -i);
-}
-/*- End of function --------------------------------------------------------*/
-
-/*! \brief Transcode from A-law to u-law, using the procedure defined in G.711.
- \param alaw The A-law sample to transcode.
- \return The best matching u-law value.
-*/
-WebRtc_UWord8 alaw_to_ulaw(WebRtc_UWord8 alaw);
-
-/*! \brief Transcode from u-law to A-law, using the procedure defined in G.711.
- \param alaw The u-law sample to transcode.
- \return The best matching A-law value.
-*/
-WebRtc_UWord8 ulaw_to_alaw(WebRtc_UWord8 ulaw);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif
-/*- End of file ------------------------------------------------------------*/
diff --git a/trunk/src/modules/audio_coding/codecs/g711/g711_interface.c b/trunk/src/modules/audio_coding/codecs/g711/g711_interface.c
deleted file mode 100644
index a49abdb..0000000
--- a/trunk/src/modules/audio_coding/codecs/g711/g711_interface.c
+++ /dev/null
@@ -1,171 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-#include <string.h>
-#include "g711.h"
-#include "g711_interface.h"
-#include "typedefs.h"
-
-WebRtc_Word16 WebRtcG711_EncodeA(void *state,
- WebRtc_Word16 *speechIn,
- WebRtc_Word16 len,
- WebRtc_Word16 *encoded)
-{
- int n;
- WebRtc_UWord16 tempVal, tempVal2;
-
- // Set and discard to avoid getting warnings
- (void)(state = NULL);
-
- // Sanity check of input length
- if (len < 0) {
- return (-1);
- }
-
- // Loop over all samples
- for (n = 0; n < len; n++) {
- tempVal = (WebRtc_UWord16)linear_to_alaw(speechIn[n]);
-
-#ifdef WEBRTC_BIG_ENDIAN
- if ((n & 0x1) == 1) {
- encoded[n>>1]|=((WebRtc_UWord16)tempVal);
- } else {
- encoded[n>>1]=((WebRtc_UWord16)tempVal)<<8;
- }
-#else
- if ((n & 0x1) == 1) {
- tempVal2 |= ((WebRtc_UWord16) tempVal) << 8;
- encoded[n >> 1] |= ((WebRtc_UWord16) tempVal) << 8;
- } else {
- tempVal2 = ((WebRtc_UWord16) tempVal);
- encoded[n >> 1] = ((WebRtc_UWord16) tempVal);
- }
-#endif
- }
- return (len);
-}
-
-WebRtc_Word16 WebRtcG711_EncodeU(void *state,
- WebRtc_Word16 *speechIn,
- WebRtc_Word16 len,
- WebRtc_Word16 *encoded)
-{
- int n;
- WebRtc_UWord16 tempVal;
-
- // Set and discard to avoid getting warnings
- (void)(state = NULL);
-
- // Sanity check of input length
- if (len < 0) {
- return (-1);
- }
-
- // Loop over all samples
- for (n = 0; n < len; n++) {
- tempVal = (WebRtc_UWord16)linear_to_ulaw(speechIn[n]);
-
- #ifdef WEBRTC_BIG_ENDIAN
- if ((n & 0x1) == 1) {
- encoded[n>>1]|=((WebRtc_UWord16)tempVal);
- } else {
- encoded[n>>1]=((WebRtc_UWord16)tempVal)<<8;
- }
- #else
- if ((n & 0x1) == 1) {
- encoded[n >> 1] |= ((WebRtc_UWord16) tempVal) << 8;
- } else {
- encoded[n >> 1] = ((WebRtc_UWord16) tempVal);
- }
- #endif
- }
- return (len);
-}
-
-WebRtc_Word16 WebRtcG711_DecodeA(void *state,
- WebRtc_Word16 *encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType)
-{
- int n;
- WebRtc_UWord16 tempVal;
-
- // Set and discard to avoid getting warnings
- (void)(state = NULL);
-
- // Sanity check of input length
- if (len < 0) {
- return (-1);
- }
-
- for (n = 0; n < len; n++) {
- #ifdef WEBRTC_BIG_ENDIAN
- if ((n & 0x1) == 1) {
- tempVal=((WebRtc_UWord16)encoded[n>>1] & 0xFF);
- } else {
- tempVal=((WebRtc_UWord16)encoded[n>>1] >> 8);
- }
- #else
- if ((n & 0x1) == 1) {
- tempVal = (encoded[n >> 1] >> 8);
- } else {
- tempVal = (encoded[n >> 1] & 0xFF);
- }
- #endif
- decoded[n] = (WebRtc_Word16) alaw_to_linear(tempVal);
- }
-
- *speechType = 1;
- return (len);
-}
-
-WebRtc_Word16 WebRtcG711_DecodeU(void *state,
- WebRtc_Word16 *encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType)
-{
- int n;
- WebRtc_UWord16 tempVal;
-
- // Set and discard to avoid getting warnings
- (void)(state = NULL);
-
- // Sanity check of input length
- if (len < 0) {
- return (-1);
- }
-
- for (n = 0; n < len; n++) {
- #ifdef WEBRTC_BIG_ENDIAN
- if ((n & 0x1) == 1) {
- tempVal=((WebRtc_UWord16)encoded[n>>1] & 0xFF);
- } else {
- tempVal=((WebRtc_UWord16)encoded[n>>1] >> 8);
- }
- #else
- if ((n & 0x1) == 1) {
- tempVal = (encoded[n >> 1] >> 8);
- } else {
- tempVal = (encoded[n >> 1] & 0xFF);
- }
- #endif
- decoded[n] = (WebRtc_Word16) ulaw_to_linear(tempVal);
- }
-
- *speechType = 1;
- return (len);
-}
-
-WebRtc_Word16 WebRtcG711_Version(char* version, WebRtc_Word16 lenBytes)
-{
- strncpy(version, "2.0.0", lenBytes);
- return 0;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/g711/g711_unittest.cc b/trunk/src/modules/audio_coding/codecs/g711/g711_unittest.cc
deleted file mode 100644
index c903bed..0000000
--- a/trunk/src/modules/audio_coding/codecs/g711/g711_unittest.cc
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Empty test just to get code coverage metrics for this dir.
- */
-#include "g711_interface.h"
-#include "gtest/gtest.h"
-
-TEST(G711Test, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/modules/audio_coding/codecs/g711/include/g711_interface.h b/trunk/src/modules/audio_coding/codecs/g711/include/g711_interface.h
deleted file mode 100644
index 25a9903..0000000
--- a/trunk/src/modules/audio_coding/codecs/g711/include/g711_interface.h
+++ /dev/null
@@ -1,148 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef MODULES_AUDIO_CODING_CODECS_G711_MAIN_INTERFACE_G711_INTERFACE_H_
-#define MODULES_AUDIO_CODING_CODECS_G711_MAIN_INTERFACE_G711_INTERFACE_H_
-
-#include "typedefs.h"
-
-// Comfort noise constants
-#define G711_WEBRTC_SPEECH 1
-#define G711_WEBRTC_CNG 2
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/****************************************************************************
- * WebRtcG711_EncodeA(...)
- *
- * This function encodes a G711 A-law frame and inserts it into a packet.
- * Input speech length has be of any length.
- *
- * Input:
- * - state : Dummy state to make this codec look more like
- * other codecs
- * - speechIn : Input speech vector
- * - len : Samples in speechIn
- *
- * Output:
- * - encoded : The encoded data vector
- *
- * Return value : >0 - Length (in bytes) of coded data
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcG711_EncodeA(void *state,
- WebRtc_Word16 *speechIn,
- WebRtc_Word16 len,
- WebRtc_Word16 *encoded);
-
-/****************************************************************************
- * WebRtcG711_EncodeU(...)
- *
- * This function encodes a G711 U-law frame and inserts it into a packet.
- * Input speech length has be of any length.
- *
- * Input:
- * - state : Dummy state to make this codec look more like
- * other codecs
- * - speechIn : Input speech vector
- * - len : Samples in speechIn
- *
- * Output:
- * - encoded : The encoded data vector
- *
- * Return value : >0 - Length (in bytes) of coded data
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcG711_EncodeU(void *state,
- WebRtc_Word16 *speechIn,
- WebRtc_Word16 len,
- WebRtc_Word16 *encoded);
-
-/****************************************************************************
- * WebRtcG711_DecodeA(...)
- *
- * This function decodes a packet G711 A-law frame.
- *
- * Input:
- * - state : Dummy state to make this codec look more like
- * other codecs
- * - encoded : Encoded data
- * - len : Bytes in encoded vector
- *
- * Output:
- * - decoded : The decoded vector
- * - speechType : 1 normal, 2 CNG (for G711 it should
- * always return 1 since G711 does not have a
- * built-in DTX/CNG scheme)
- *
- * Return value : >0 - Samples in decoded vector
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcG711_DecodeA(void *state,
- WebRtc_Word16 *encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType);
-
-/****************************************************************************
- * WebRtcG711_DecodeU(...)
- *
- * This function decodes a packet G711 U-law frame.
- *
- * Input:
- * - state : Dummy state to make this codec look more like
- * other codecs
- * - encoded : Encoded data
- * - len : Bytes in encoded vector
- *
- * Output:
- * - decoded : The decoded vector
- * - speechType : 1 normal, 2 CNG (for G711 it should
- * always return 1 since G711 does not have a
- * built-in DTX/CNG scheme)
- *
- * Return value : >0 - Samples in decoded vector
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcG711_DecodeU(void *state,
- WebRtc_Word16 *encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType);
-
-/**********************************************************************
-* WebRtcG711_Version(...)
-*
-* This function gives the version string of the G.711 codec.
-*
-* Input:
-* - lenBytes: the size of Allocated space (in Bytes) where
-* the version number is written to (in string format).
-*
-* Output:
-* - version: Pointer to a buffer where the version number is
-* written to.
-*
-*/
-
-WebRtc_Word16 WebRtcG711_Version(char* version, WebRtc_Word16 lenBytes);
-
-#ifdef __cplusplus
-}
-#endif
-
-
-#endif /* MODULES_AUDIO_CODING_CODECS_G711_MAIN_INTERFACE_G711_INTERFACE_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/g711/test/testG711.cc b/trunk/src/modules/audio_coding/codecs/g711/test/testG711.cc
deleted file mode 100644
index 747f922..0000000
--- a/trunk/src/modules/audio_coding/codecs/g711/test/testG711.cc
+++ /dev/null
@@ -1,169 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * testG711.cpp : Defines the entry point for the console application.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-/* include API */
-#include "g711_interface.h"
-
-/* Runtime statistics */
-#include <time.h>
-#define CLOCKS_PER_SEC_G711 1000
-
-
-/* function for reading audio data from PCM file */
-int readframe(WebRtc_Word16 *data, FILE *inp, int length) {
-
- short k, rlen, status = 0;
-
- rlen = (short)fread(data, sizeof(WebRtc_Word16), length, inp);
- if (rlen < length) {
- for (k = rlen; k < length; k++)
- data[k] = 0;
- status = 1;
- }
-
- return status;
-}
-
-int main(int argc, char* argv[])
-{
- char inname[80], outname[40], bitname[40];
- FILE *inp, *outp, *bitp = NULL;
- int framecnt, endfile;
-
- WebRtc_Word16 framelength = 80;
-
- int err;
-
- /* Runtime statistics */
- double starttime;
- double runtime;
- double length_file;
-
- WebRtc_Word16 stream_len = 0;
- WebRtc_Word16 shortdata[480];
- WebRtc_Word16 decoded[480];
- WebRtc_Word16 streamdata[500];
- WebRtc_Word16 speechType[1];
- char law[2];
- char versionNumber[40];
-
- /* handling wrong input arguments in the command line */
- if ((argc!=5) && (argc!=6)) {
- printf("\n\nWrong number of arguments or flag values.\n\n");
-
- printf("\n");
- printf("\nG.711 test application\n\n");
- printf("Usage:\n\n");
- printf("./testG711.exe framelength law infile outfile \n\n");
- printf("framelength: Framelength in samples.\n");
- printf("law : Coding law, A och u.\n");
- printf("infile : Normal speech input file\n");
- printf("outfile : Speech output file\n\n");
- printf("outbits : Output bitstream file [optional]\n\n");
- exit(0);
-
- }
-
- /* Get version and print */
- WebRtcG711_Version(versionNumber, 40);
-
- printf("-----------------------------------\n");
- printf("G.711 version: %s\n\n", versionNumber);
- /* Get frame length */
- framelength = atoi(argv[1]);
-
- /* Get compression law */
- strcpy(law, argv[2]);
-
- /* Get Input and Output files */
- sscanf(argv[3], "%s", inname);
- sscanf(argv[4], "%s", outname);
- if (argc==6) {
- sscanf(argv[5], "%s", bitname);
- if ((bitp = fopen(bitname,"wb")) == NULL) {
- printf(" G.711: Cannot read file %s.\n", bitname);
- exit(1);
- }
- }
-
- if ((inp = fopen(inname,"rb")) == NULL) {
- printf(" G.711: Cannot read file %s.\n", inname);
- exit(1);
- }
- if ((outp = fopen(outname,"wb")) == NULL) {
- printf(" G.711: Cannot write file %s.\n", outname);
- exit(1);
- }
- printf("\nInput: %s\nOutput: %s\n", inname, outname);
- if (argc==6) {
- printf("\nBitfile: %s\n", bitname);
- }
-
- starttime = clock()/(double)CLOCKS_PER_SEC_G711; /* Runtime statistics */
-
- /* Initialize encoder and decoder */
- framecnt= 0;
- endfile = 0;
- while (endfile == 0) {
- framecnt++;
- /* Read speech block */
- endfile = readframe(shortdata, inp, framelength);
-
- /* G.711 encoding */
- if (!strcmp(law,"A")) {
- /* A-law encoding */
- stream_len = WebRtcG711_EncodeA(NULL, shortdata, framelength, streamdata);
- if (argc==6){
- /* Write bits to file */
- fwrite(streamdata,sizeof(unsigned char),stream_len,bitp);
- }
- err = WebRtcG711_DecodeA(NULL, streamdata, stream_len, decoded, speechType);
- } else if (!strcmp(law,"u")){
- /* u-law encoding */
- stream_len = WebRtcG711_EncodeU(NULL, shortdata, framelength, streamdata);
- if (argc==6){
- /* Write bits to file */
- fwrite(streamdata,sizeof(unsigned char),stream_len,bitp);
- }
- err = WebRtcG711_DecodeU(NULL, streamdata, stream_len, decoded, speechType);
- } else {
- printf("Wrong law mode\n");
- exit (1);
- }
- if (stream_len < 0 || err < 0) {
- /* exit if returned with error */
- printf("Error in encoder/decoder\n");
- } else {
- /* Write coded speech to file */
- fwrite(decoded,sizeof(short),framelength,outp);
- }
- }
-
-
- runtime = (double)(clock()/(double)CLOCKS_PER_SEC_G711-starttime);
- length_file = ((double)framecnt*(double)framelength/8000);
- printf("\n\nLength of speech file: %.1f s\n", length_file);
- printf("Time to run G.711: %.2f s (%.2f %% of realtime)\n\n", runtime, (100*runtime/length_file));
- printf("---------------------END----------------------\n");
-
- fclose(inp);
- fclose(outp);
-
-
- return 0;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/g722/Android.mk b/trunk/src/modules/audio_coding/codecs/g722/Android.mk
deleted file mode 100644
index 39dea9e..0000000
--- a/trunk/src/modules/audio_coding/codecs/g722/Android.mk
+++ /dev/null
@@ -1,40 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_g722
-LOCAL_MODULE_TAGS := optional
-LOCAL_SRC_FILES := \
- g722_interface.c \
- g722_encode.c \
- g722_decode.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include \
- $(LOCAL_PATH)/../../../..
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_coding/codecs/g722/g722.gypi b/trunk/src/modules/audio_coding/codecs/g722/g722.gypi
deleted file mode 100644
index c1e55d9..0000000
--- a/trunk/src/modules/audio_coding/codecs/g722/g722.gypi
+++ /dev/null
@@ -1,65 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-{
- 'targets': [
- {
- 'target_name': 'G722',
- 'type': '<(library)',
- 'include_dirs': [
- 'include',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- ],
- },
- 'sources': [
- 'include/g722_interface.h',
- 'g722_interface.c',
- 'g722_encode.c',
- 'g722_decode.c',
- 'g722_enc_dec.h',
- ],
- },
- ], # targets
- # Exclude the test target when building with chromium.
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'g722_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'G722',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'sources': [
- 'g722_unittest.cc',
- ],
- },
- {
- 'target_name': 'G722Test',
- 'type': 'executable',
- 'dependencies': [
- 'G722',
- ],
- 'sources': [
- 'test/testG722.cc',
- ],
- },
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/g722/g722_decode.c b/trunk/src/modules/audio_coding/codecs/g722/g722_decode.c
deleted file mode 100644
index 499cc8f..0000000
--- a/trunk/src/modules/audio_coding/codecs/g722/g722_decode.c
+++ /dev/null
@@ -1,410 +0,0 @@
-/*
- * SpanDSP - a series of DSP components for telephony
- *
- * g722_decode.c - The ITU G.722 codec, decode part.
- *
- * Written by Steve Underwood <steveu@coppice.org>
- *
- * Copyright (C) 2005 Steve Underwood
- *
- * Despite my general liking of the GPL, I place my own contributions
- * to this code in the public domain for the benefit of all mankind -
- * even the slimy ones who might try to proprietize my work and use it
- * to my detriment.
- *
- * Based in part on a single channel G.722 codec which is:
- *
- * Copyright (c) CMU 1993
- * Computer Science, Speech Group
- * Chengxiang Lu and Alex Hauptmann
- *
- * $Id: g722_decode.c,v 1.15 2006/07/07 16:37:49 steveu Exp $
- *
- * Modifications for WebRtc, 2011/04/28, by tlegrand:
- * -Removed usage of inttypes.h and tgmath.h
- * -Changed to use WebRtc types
- * -Changed __inline__ to __inline
- * -Added saturation check on output
- */
-
-/*! \file */
-
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-#include <stdio.h>
-#include <memory.h>
-#include <stdlib.h>
-
-#include "typedefs.h"
-#include "g722_enc_dec.h"
-
-
-#if !defined(FALSE)
-#define FALSE 0
-#endif
-#if !defined(TRUE)
-#define TRUE (!FALSE)
-#endif
-
-static __inline WebRtc_Word16 saturate(WebRtc_Word32 amp)
-{
- WebRtc_Word16 amp16;
-
- /* Hopefully this is optimised for the common case - not clipping */
- amp16 = (WebRtc_Word16) amp;
- if (amp == amp16)
- return amp16;
- if (amp > WEBRTC_INT16_MAX)
- return WEBRTC_INT16_MAX;
- return WEBRTC_INT16_MIN;
-}
-/*- End of function --------------------------------------------------------*/
-
-static void block4(g722_decode_state_t *s, int band, int d);
-
-static void block4(g722_decode_state_t *s, int band, int d)
-{
- int wd1;
- int wd2;
- int wd3;
- int i;
-
- /* Block 4, RECONS */
- s->band[band].d[0] = d;
- s->band[band].r[0] = saturate(s->band[band].s + d);
-
- /* Block 4, PARREC */
- s->band[band].p[0] = saturate(s->band[band].sz + d);
-
- /* Block 4, UPPOL2 */
- for (i = 0; i < 3; i++)
- s->band[band].sg[i] = s->band[band].p[i] >> 15;
- wd1 = saturate(s->band[band].a[1] << 2);
-
- wd2 = (s->band[band].sg[0] == s->band[band].sg[1]) ? -wd1 : wd1;
- if (wd2 > 32767)
- wd2 = 32767;
- wd3 = (s->band[band].sg[0] == s->band[band].sg[2]) ? 128 : -128;
- wd3 += (wd2 >> 7);
- wd3 += (s->band[band].a[2]*32512) >> 15;
- if (wd3 > 12288)
- wd3 = 12288;
- else if (wd3 < -12288)
- wd3 = -12288;
- s->band[band].ap[2] = wd3;
-
- /* Block 4, UPPOL1 */
- s->band[band].sg[0] = s->band[band].p[0] >> 15;
- s->band[band].sg[1] = s->band[band].p[1] >> 15;
- wd1 = (s->band[band].sg[0] == s->band[band].sg[1]) ? 192 : -192;
- wd2 = (s->band[band].a[1]*32640) >> 15;
-
- s->band[band].ap[1] = saturate(wd1 + wd2);
- wd3 = saturate(15360 - s->band[band].ap[2]);
- if (s->band[band].ap[1] > wd3)
- s->band[band].ap[1] = wd3;
- else if (s->band[band].ap[1] < -wd3)
- s->band[band].ap[1] = -wd3;
-
- /* Block 4, UPZERO */
- wd1 = (d == 0) ? 0 : 128;
- s->band[band].sg[0] = d >> 15;
- for (i = 1; i < 7; i++)
- {
- s->band[band].sg[i] = s->band[band].d[i] >> 15;
- wd2 = (s->band[band].sg[i] == s->band[band].sg[0]) ? wd1 : -wd1;
- wd3 = (s->band[band].b[i]*32640) >> 15;
- s->band[band].bp[i] = saturate(wd2 + wd3);
- }
-
- /* Block 4, DELAYA */
- for (i = 6; i > 0; i--)
- {
- s->band[band].d[i] = s->band[band].d[i - 1];
- s->band[band].b[i] = s->band[band].bp[i];
- }
-
- for (i = 2; i > 0; i--)
- {
- s->band[band].r[i] = s->band[band].r[i - 1];
- s->band[band].p[i] = s->band[band].p[i - 1];
- s->band[band].a[i] = s->band[band].ap[i];
- }
-
- /* Block 4, FILTEP */
- wd1 = saturate(s->band[band].r[1] + s->band[band].r[1]);
- wd1 = (s->band[band].a[1]*wd1) >> 15;
- wd2 = saturate(s->band[band].r[2] + s->band[band].r[2]);
- wd2 = (s->band[band].a[2]*wd2) >> 15;
- s->band[band].sp = saturate(wd1 + wd2);
-
- /* Block 4, FILTEZ */
- s->band[band].sz = 0;
- for (i = 6; i > 0; i--)
- {
- wd1 = saturate(s->band[band].d[i] + s->band[band].d[i]);
- s->band[band].sz += (s->band[band].b[i]*wd1) >> 15;
- }
- s->band[band].sz = saturate(s->band[band].sz);
-
- /* Block 4, PREDIC */
- s->band[band].s = saturate(s->band[band].sp + s->band[band].sz);
-}
-/*- End of function --------------------------------------------------------*/
-
-g722_decode_state_t *WebRtc_g722_decode_init(g722_decode_state_t *s,
- int rate,
- int options)
-{
- if (s == NULL)
- {
- if ((s = (g722_decode_state_t *) malloc(sizeof(*s))) == NULL)
- return NULL;
- }
- memset(s, 0, sizeof(*s));
- if (rate == 48000)
- s->bits_per_sample = 6;
- else if (rate == 56000)
- s->bits_per_sample = 7;
- else
- s->bits_per_sample = 8;
- if ((options & G722_SAMPLE_RATE_8000))
- s->eight_k = TRUE;
- if ((options & G722_PACKED) && s->bits_per_sample != 8)
- s->packed = TRUE;
- else
- s->packed = FALSE;
- s->band[0].det = 32;
- s->band[1].det = 8;
- return s;
-}
-/*- End of function --------------------------------------------------------*/
-
-int WebRtc_g722_decode_release(g722_decode_state_t *s)
-{
- free(s);
- return 0;
-}
-/*- End of function --------------------------------------------------------*/
-
-int WebRtc_g722_decode(g722_decode_state_t *s, WebRtc_Word16 amp[],
- const WebRtc_UWord8 g722_data[], int len)
-{
- static const int wl[8] = {-60, -30, 58, 172, 334, 538, 1198, 3042 };
- static const int rl42[16] = {0, 7, 6, 5, 4, 3, 2, 1,
- 7, 6, 5, 4, 3, 2, 1, 0 };
- static const int ilb[32] =
- {
- 2048, 2093, 2139, 2186, 2233, 2282, 2332,
- 2383, 2435, 2489, 2543, 2599, 2656, 2714,
- 2774, 2834, 2896, 2960, 3025, 3091, 3158,
- 3228, 3298, 3371, 3444, 3520, 3597, 3676,
- 3756, 3838, 3922, 4008
- };
- static const int wh[3] = {0, -214, 798};
- static const int rh2[4] = {2, 1, 2, 1};
- static const int qm2[4] = {-7408, -1616, 7408, 1616};
- static const int qm4[16] =
- {
- 0, -20456, -12896, -8968,
- -6288, -4240, -2584, -1200,
- 20456, 12896, 8968, 6288,
- 4240, 2584, 1200, 0
- };
- static const int qm5[32] =
- {
- -280, -280, -23352, -17560,
- -14120, -11664, -9752, -8184,
- -6864, -5712, -4696, -3784,
- -2960, -2208, -1520, -880,
- 23352, 17560, 14120, 11664,
- 9752, 8184, 6864, 5712,
- 4696, 3784, 2960, 2208,
- 1520, 880, 280, -280
- };
- static const int qm6[64] =
- {
- -136, -136, -136, -136,
- -24808, -21904, -19008, -16704,
- -14984, -13512, -12280, -11192,
- -10232, -9360, -8576, -7856,
- -7192, -6576, -6000, -5456,
- -4944, -4464, -4008, -3576,
- -3168, -2776, -2400, -2032,
- -1688, -1360, -1040, -728,
- 24808, 21904, 19008, 16704,
- 14984, 13512, 12280, 11192,
- 10232, 9360, 8576, 7856,
- 7192, 6576, 6000, 5456,
- 4944, 4464, 4008, 3576,
- 3168, 2776, 2400, 2032,
- 1688, 1360, 1040, 728,
- 432, 136, -432, -136
- };
- static const int qmf_coeffs[12] =
- {
- 3, -11, 12, 32, -210, 951, 3876, -805, 362, -156, 53, -11,
- };
-
- int dlowt;
- int rlow;
- int ihigh;
- int dhigh;
- int rhigh;
- int xout1;
- int xout2;
- int wd1;
- int wd2;
- int wd3;
- int code;
- int outlen;
- int i;
- int j;
-
- outlen = 0;
- rhigh = 0;
- for (j = 0; j < len; )
- {
- if (s->packed)
- {
- /* Unpack the code bits */
- if (s->in_bits < s->bits_per_sample)
- {
- s->in_buffer |= (g722_data[j++] << s->in_bits);
- s->in_bits += 8;
- }
- code = s->in_buffer & ((1 << s->bits_per_sample) - 1);
- s->in_buffer >>= s->bits_per_sample;
- s->in_bits -= s->bits_per_sample;
- }
- else
- {
- code = g722_data[j++];
- }
-
- switch (s->bits_per_sample)
- {
- default:
- case 8:
- wd1 = code & 0x3F;
- ihigh = (code >> 6) & 0x03;
- wd2 = qm6[wd1];
- wd1 >>= 2;
- break;
- case 7:
- wd1 = code & 0x1F;
- ihigh = (code >> 5) & 0x03;
- wd2 = qm5[wd1];
- wd1 >>= 1;
- break;
- case 6:
- wd1 = code & 0x0F;
- ihigh = (code >> 4) & 0x03;
- wd2 = qm4[wd1];
- break;
- }
- /* Block 5L, LOW BAND INVQBL */
- wd2 = (s->band[0].det*wd2) >> 15;
- /* Block 5L, RECONS */
- rlow = s->band[0].s + wd2;
- /* Block 6L, LIMIT */
- if (rlow > 16383)
- rlow = 16383;
- else if (rlow < -16384)
- rlow = -16384;
-
- /* Block 2L, INVQAL */
- wd2 = qm4[wd1];
- dlowt = (s->band[0].det*wd2) >> 15;
-
- /* Block 3L, LOGSCL */
- wd2 = rl42[wd1];
- wd1 = (s->band[0].nb*127) >> 7;
- wd1 += wl[wd2];
- if (wd1 < 0)
- wd1 = 0;
- else if (wd1 > 18432)
- wd1 = 18432;
- s->band[0].nb = wd1;
-
- /* Block 3L, SCALEL */
- wd1 = (s->band[0].nb >> 6) & 31;
- wd2 = 8 - (s->band[0].nb >> 11);
- wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
- s->band[0].det = wd3 << 2;
-
- block4(s, 0, dlowt);
-
- if (!s->eight_k)
- {
- /* Block 2H, INVQAH */
- wd2 = qm2[ihigh];
- dhigh = (s->band[1].det*wd2) >> 15;
- /* Block 5H, RECONS */
- rhigh = dhigh + s->band[1].s;
- /* Block 6H, LIMIT */
- if (rhigh > 16383)
- rhigh = 16383;
- else if (rhigh < -16384)
- rhigh = -16384;
-
- /* Block 2H, INVQAH */
- wd2 = rh2[ihigh];
- wd1 = (s->band[1].nb*127) >> 7;
- wd1 += wh[wd2];
- if (wd1 < 0)
- wd1 = 0;
- else if (wd1 > 22528)
- wd1 = 22528;
- s->band[1].nb = wd1;
-
- /* Block 3H, SCALEH */
- wd1 = (s->band[1].nb >> 6) & 31;
- wd2 = 10 - (s->band[1].nb >> 11);
- wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
- s->band[1].det = wd3 << 2;
-
- block4(s, 1, dhigh);
- }
-
- if (s->itu_test_mode)
- {
- amp[outlen++] = (WebRtc_Word16) (rlow << 1);
- amp[outlen++] = (WebRtc_Word16) (rhigh << 1);
- }
- else
- {
- if (s->eight_k)
- {
- amp[outlen++] = (WebRtc_Word16) (rlow << 1);
- }
- else
- {
- /* Apply the receive QMF */
- for (i = 0; i < 22; i++)
- s->x[i] = s->x[i + 2];
- s->x[22] = rlow + rhigh;
- s->x[23] = rlow - rhigh;
-
- xout1 = 0;
- xout2 = 0;
- for (i = 0; i < 12; i++)
- {
- xout2 += s->x[2*i]*qmf_coeffs[i];
- xout1 += s->x[2*i + 1]*qmf_coeffs[11 - i];
- }
- /* We shift by 12 to allow for the QMF filters (DC gain = 4096), less 1
- to allow for the 15 bit input to the G.722 algorithm. */
- /* WebRtc, tlegrand: added saturation */
- amp[outlen++] = saturate(xout1 >> 11);
- amp[outlen++] = saturate(xout2 >> 11);
- }
- }
- }
- return outlen;
-}
-/*- End of function --------------------------------------------------------*/
-/*- End of file ------------------------------------------------------------*/
diff --git a/trunk/src/modules/audio_coding/codecs/g722/g722_enc_dec.h b/trunk/src/modules/audio_coding/codecs/g722/g722_enc_dec.h
deleted file mode 100644
index d2d19b0..0000000
--- a/trunk/src/modules/audio_coding/codecs/g722/g722_enc_dec.h
+++ /dev/null
@@ -1,158 +0,0 @@
-/*
- * SpanDSP - a series of DSP components for telephony
- *
- * g722.h - The ITU G.722 codec.
- *
- * Written by Steve Underwood <steveu@coppice.org>
- *
- * Copyright (C) 2005 Steve Underwood
- *
- * Despite my general liking of the GPL, I place my own contributions
- * to this code in the public domain for the benefit of all mankind -
- * even the slimy ones who might try to proprietize my work and use it
- * to my detriment.
- *
- * Based on a single channel G.722 codec which is:
- *
- ***** Copyright (c) CMU 1993 *****
- * Computer Science, Speech Group
- * Chengxiang Lu and Alex Hauptmann
- *
- * $Id: g722.h,v 1.10 2006/06/16 12:45:53 steveu Exp $
- *
- * Modifications for WebRtc, 2011/04/28, by tlegrand:
- * -Changed to use WebRtc types
- * -Added new defines for minimum and maximum values of short int
- */
-
-
-/*! \file */
-
-#if !defined(_G722_ENC_DEC_H_)
-#define _G722_ENC_DEC_H_
-
-/*! \page g722_page G.722 encoding and decoding
-\section g722_page_sec_1 What does it do?
-The G.722 module is a bit exact implementation of the ITU G.722 specification for all three
-specified bit rates - 64000bps, 56000bps and 48000bps. It passes the ITU tests.
-
-To allow fast and flexible interworking with narrow band telephony, the encoder and decoder
-support an option for the linear audio to be an 8k samples/second stream. In this mode the
-codec is considerably faster, and still fully compatible with wideband terminals using G.722.
-
-\section g722_page_sec_2 How does it work?
-???.
-*/
-
-#define WEBRTC_INT16_MAX 32767
-#define WEBRTC_INT16_MIN -32768
-
-enum
-{
- G722_SAMPLE_RATE_8000 = 0x0001,
- G722_PACKED = 0x0002
-};
-
-typedef struct
-{
- /*! TRUE if the operating in the special ITU test mode, with the band split filters
- disabled. */
- int itu_test_mode;
- /*! TRUE if the G.722 data is packed */
- int packed;
- /*! TRUE if encode from 8k samples/second */
- int eight_k;
- /*! 6 for 48000kbps, 7 for 56000kbps, or 8 for 64000kbps. */
- int bits_per_sample;
-
- /*! Signal history for the QMF */
- int x[24];
-
- struct
- {
- int s;
- int sp;
- int sz;
- int r[3];
- int a[3];
- int ap[3];
- int p[3];
- int d[7];
- int b[7];
- int bp[7];
- int sg[7];
- int nb;
- int det;
- } band[2];
-
- unsigned int in_buffer;
- int in_bits;
- unsigned int out_buffer;
- int out_bits;
-} g722_encode_state_t;
-
-typedef struct
-{
- /*! TRUE if the operating in the special ITU test mode, with the band split filters
- disabled. */
- int itu_test_mode;
- /*! TRUE if the G.722 data is packed */
- int packed;
- /*! TRUE if decode to 8k samples/second */
- int eight_k;
- /*! 6 for 48000kbps, 7 for 56000kbps, or 8 for 64000kbps. */
- int bits_per_sample;
-
- /*! Signal history for the QMF */
- int x[24];
-
- struct
- {
- int s;
- int sp;
- int sz;
- int r[3];
- int a[3];
- int ap[3];
- int p[3];
- int d[7];
- int b[7];
- int bp[7];
- int sg[7];
- int nb;
- int det;
- } band[2];
-
- unsigned int in_buffer;
- int in_bits;
- unsigned int out_buffer;
- int out_bits;
-} g722_decode_state_t;
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-g722_encode_state_t *WebRtc_g722_encode_init(g722_encode_state_t *s,
- int rate,
- int options);
-int WebRtc_g722_encode_release(g722_encode_state_t *s);
-int WebRtc_g722_encode(g722_encode_state_t *s,
- WebRtc_UWord8 g722_data[],
- const WebRtc_Word16 amp[],
- int len);
-
-g722_decode_state_t *WebRtc_g722_decode_init(g722_decode_state_t *s,
- int rate,
- int options);
-int WebRtc_g722_decode_release(g722_decode_state_t *s);
-int WebRtc_g722_decode(g722_decode_state_t *s,
- WebRtc_Word16 amp[],
- const WebRtc_UWord8 g722_data[],
- int len);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/g722/g722_encode.c b/trunk/src/modules/audio_coding/codecs/g722/g722_encode.c
deleted file mode 100644
index 7487b64..0000000
--- a/trunk/src/modules/audio_coding/codecs/g722/g722_encode.c
+++ /dev/null
@@ -1,434 +0,0 @@
-/*
- * SpanDSP - a series of DSP components for telephony
- *
- * g722_encode.c - The ITU G.722 codec, encode part.
- *
- * Written by Steve Underwood <steveu@coppice.org>
- *
- * Copyright (C) 2005 Steve Underwood
- *
- * All rights reserved.
- *
- * Despite my general liking of the GPL, I place my own contributions
- * to this code in the public domain for the benefit of all mankind -
- * even the slimy ones who might try to proprietize my work and use it
- * to my detriment.
- *
- * Based on a single channel 64kbps only G.722 codec which is:
- *
- ***** Copyright (c) CMU 1993 *****
- * Computer Science, Speech Group
- * Chengxiang Lu and Alex Hauptmann
- *
- * $Id: g722_encode.c,v 1.14 2006/07/07 16:37:49 steveu Exp $
- *
- * Modifications for WebRtc, 2011/04/28, by tlegrand:
- * -Removed usage of inttypes.h and tgmath.h
- * -Changed to use WebRtc types
- * -Added option to run encoder bitexact with ITU-T reference implementation
- */
-
-/*! \file */
-
-#ifdef HAVE_CONFIG_H
-#include <config.h>
-#endif
-
-#include <stdio.h>
-#include <memory.h>
-#include <stdlib.h>
-
-#include "typedefs.h"
-#include "g722_enc_dec.h"
-
-#if !defined(FALSE)
-#define FALSE 0
-#endif
-#if !defined(TRUE)
-#define TRUE (!FALSE)
-#endif
-
-static __inline WebRtc_Word16 saturate(WebRtc_Word32 amp)
-{
- WebRtc_Word16 amp16;
-
- /* Hopefully this is optimised for the common case - not clipping */
- amp16 = (WebRtc_Word16) amp;
- if (amp == amp16)
- return amp16;
- if (amp > WEBRTC_INT16_MAX)
- return WEBRTC_INT16_MAX;
- return WEBRTC_INT16_MIN;
-}
-/*- End of function --------------------------------------------------------*/
-
-static void block4(g722_encode_state_t *s, int band, int d)
-{
- int wd1;
- int wd2;
- int wd3;
- int i;
-
- /* Block 4, RECONS */
- s->band[band].d[0] = d;
- s->band[band].r[0] = saturate(s->band[band].s + d);
-
- /* Block 4, PARREC */
- s->band[band].p[0] = saturate(s->band[band].sz + d);
-
- /* Block 4, UPPOL2 */
- for (i = 0; i < 3; i++)
- s->band[band].sg[i] = s->band[band].p[i] >> 15;
- wd1 = saturate(s->band[band].a[1] << 2);
-
- wd2 = (s->band[band].sg[0] == s->band[band].sg[1]) ? -wd1 : wd1;
- if (wd2 > 32767)
- wd2 = 32767;
- wd3 = (wd2 >> 7) + ((s->band[band].sg[0] == s->band[band].sg[2]) ? 128 : -128);
- wd3 += (s->band[band].a[2]*32512) >> 15;
- if (wd3 > 12288)
- wd3 = 12288;
- else if (wd3 < -12288)
- wd3 = -12288;
- s->band[band].ap[2] = wd3;
-
- /* Block 4, UPPOL1 */
- s->band[band].sg[0] = s->band[band].p[0] >> 15;
- s->band[band].sg[1] = s->band[band].p[1] >> 15;
- wd1 = (s->band[band].sg[0] == s->band[band].sg[1]) ? 192 : -192;
- wd2 = (s->band[band].a[1]*32640) >> 15;
-
- s->band[band].ap[1] = saturate(wd1 + wd2);
- wd3 = saturate(15360 - s->band[band].ap[2]);
- if (s->band[band].ap[1] > wd3)
- s->band[band].ap[1] = wd3;
- else if (s->band[band].ap[1] < -wd3)
- s->band[band].ap[1] = -wd3;
-
- /* Block 4, UPZERO */
- wd1 = (d == 0) ? 0 : 128;
- s->band[band].sg[0] = d >> 15;
- for (i = 1; i < 7; i++)
- {
- s->band[band].sg[i] = s->band[band].d[i] >> 15;
- wd2 = (s->band[band].sg[i] == s->band[band].sg[0]) ? wd1 : -wd1;
- wd3 = (s->band[band].b[i]*32640) >> 15;
- s->band[band].bp[i] = saturate(wd2 + wd3);
- }
-
- /* Block 4, DELAYA */
- for (i = 6; i > 0; i--)
- {
- s->band[band].d[i] = s->band[band].d[i - 1];
- s->band[band].b[i] = s->band[band].bp[i];
- }
-
- for (i = 2; i > 0; i--)
- {
- s->band[band].r[i] = s->band[band].r[i - 1];
- s->band[band].p[i] = s->band[band].p[i - 1];
- s->band[band].a[i] = s->band[band].ap[i];
- }
-
- /* Block 4, FILTEP */
- wd1 = saturate(s->band[band].r[1] + s->band[band].r[1]);
- wd1 = (s->band[band].a[1]*wd1) >> 15;
- wd2 = saturate(s->band[band].r[2] + s->band[band].r[2]);
- wd2 = (s->band[band].a[2]*wd2) >> 15;
- s->band[band].sp = saturate(wd1 + wd2);
-
- /* Block 4, FILTEZ */
- s->band[band].sz = 0;
- for (i = 6; i > 0; i--)
- {
- wd1 = saturate(s->band[band].d[i] + s->band[band].d[i]);
- s->band[band].sz += (s->band[band].b[i]*wd1) >> 15;
- }
- s->band[band].sz = saturate(s->band[band].sz);
-
- /* Block 4, PREDIC */
- s->band[band].s = saturate(s->band[band].sp + s->band[band].sz);
-}
-/*- End of function --------------------------------------------------------*/
-
-g722_encode_state_t *WebRtc_g722_encode_init(g722_encode_state_t *s,
- int rate, int options)
-{
- if (s == NULL)
- {
- if ((s = (g722_encode_state_t *) malloc(sizeof(*s))) == NULL)
- return NULL;
- }
- memset(s, 0, sizeof(*s));
- if (rate == 48000)
- s->bits_per_sample = 6;
- else if (rate == 56000)
- s->bits_per_sample = 7;
- else
- s->bits_per_sample = 8;
- if ((options & G722_SAMPLE_RATE_8000))
- s->eight_k = TRUE;
- if ((options & G722_PACKED) && s->bits_per_sample != 8)
- s->packed = TRUE;
- else
- s->packed = FALSE;
- s->band[0].det = 32;
- s->band[1].det = 8;
- return s;
-}
-/*- End of function --------------------------------------------------------*/
-
-int WebRtc_g722_encode_release(g722_encode_state_t *s)
-{
- free(s);
- return 0;
-}
-/*- End of function --------------------------------------------------------*/
-
-/* WebRtc, tlegrand:
- * Only define the following if bit-exactness with reference implementation
- * is needed. Will only have any effect if input signal is saturated.
- */
-//#define RUN_LIKE_REFERENCE_G722
-#ifdef RUN_LIKE_REFERENCE_G722
-WebRtc_Word16 limitValues (WebRtc_Word16 rl)
-{
-
- WebRtc_Word16 yl;
-
- yl = (rl > 16383) ? 16383 : ((rl < -16384) ? -16384 : rl);
-
- return (yl);
-}
-#endif
-
-int WebRtc_g722_encode(g722_encode_state_t *s, WebRtc_UWord8 g722_data[],
- const WebRtc_Word16 amp[], int len)
-{
- static const int q6[32] =
- {
- 0, 35, 72, 110, 150, 190, 233, 276,
- 323, 370, 422, 473, 530, 587, 650, 714,
- 786, 858, 940, 1023, 1121, 1219, 1339, 1458,
- 1612, 1765, 1980, 2195, 2557, 2919, 0, 0
- };
- static const int iln[32] =
- {
- 0, 63, 62, 31, 30, 29, 28, 27,
- 26, 25, 24, 23, 22, 21, 20, 19,
- 18, 17, 16, 15, 14, 13, 12, 11,
- 10, 9, 8, 7, 6, 5, 4, 0
- };
- static const int ilp[32] =
- {
- 0, 61, 60, 59, 58, 57, 56, 55,
- 54, 53, 52, 51, 50, 49, 48, 47,
- 46, 45, 44, 43, 42, 41, 40, 39,
- 38, 37, 36, 35, 34, 33, 32, 0
- };
- static const int wl[8] =
- {
- -60, -30, 58, 172, 334, 538, 1198, 3042
- };
- static const int rl42[16] =
- {
- 0, 7, 6, 5, 4, 3, 2, 1, 7, 6, 5, 4, 3, 2, 1, 0
- };
- static const int ilb[32] =
- {
- 2048, 2093, 2139, 2186, 2233, 2282, 2332,
- 2383, 2435, 2489, 2543, 2599, 2656, 2714,
- 2774, 2834, 2896, 2960, 3025, 3091, 3158,
- 3228, 3298, 3371, 3444, 3520, 3597, 3676,
- 3756, 3838, 3922, 4008
- };
- static const int qm4[16] =
- {
- 0, -20456, -12896, -8968,
- -6288, -4240, -2584, -1200,
- 20456, 12896, 8968, 6288,
- 4240, 2584, 1200, 0
- };
- static const int qm2[4] =
- {
- -7408, -1616, 7408, 1616
- };
- static const int qmf_coeffs[12] =
- {
- 3, -11, 12, 32, -210, 951, 3876, -805, 362, -156, 53, -11,
- };
- static const int ihn[3] = {0, 1, 0};
- static const int ihp[3] = {0, 3, 2};
- static const int wh[3] = {0, -214, 798};
- static const int rh2[4] = {2, 1, 2, 1};
-
- int dlow;
- int dhigh;
- int el;
- int wd;
- int wd1;
- int ril;
- int wd2;
- int il4;
- int ih2;
- int wd3;
- int eh;
- int mih;
- int i;
- int j;
- /* Low and high band PCM from the QMF */
- int xlow;
- int xhigh;
- int g722_bytes;
- /* Even and odd tap accumulators */
- int sumeven;
- int sumodd;
- int ihigh;
- int ilow;
- int code;
-
- g722_bytes = 0;
- xhigh = 0;
- for (j = 0; j < len; )
- {
- if (s->itu_test_mode)
- {
- xlow =
- xhigh = amp[j++] >> 1;
- }
- else
- {
- if (s->eight_k)
- {
- /* We shift by 1 to allow for the 15 bit input to the G.722 algorithm. */
- xlow = amp[j++] >> 1;
- }
- else
- {
- /* Apply the transmit QMF */
- /* Shuffle the buffer down */
- for (i = 0; i < 22; i++)
- s->x[i] = s->x[i + 2];
- s->x[22] = amp[j++];
- s->x[23] = amp[j++];
-
- /* Discard every other QMF output */
- sumeven = 0;
- sumodd = 0;
- for (i = 0; i < 12; i++)
- {
- sumodd += s->x[2*i]*qmf_coeffs[i];
- sumeven += s->x[2*i + 1]*qmf_coeffs[11 - i];
- }
- /* We shift by 12 to allow for the QMF filters (DC gain = 4096), plus 1
- to allow for us summing two filters, plus 1 to allow for the 15 bit
- input to the G.722 algorithm. */
- xlow = (sumeven + sumodd) >> 14;
- xhigh = (sumeven - sumodd) >> 14;
-
-#ifdef RUN_LIKE_REFERENCE_G722
- /* The following lines are only used to verify bit-exactness
- * with reference implementation of G.722. Higher precision
- * is achieved without limiting the values.
- */
- xlow = limitValues(xlow);
- xhigh = limitValues(xhigh);
-#endif
- }
- }
- /* Block 1L, SUBTRA */
- el = saturate(xlow - s->band[0].s);
-
- /* Block 1L, QUANTL */
- wd = (el >= 0) ? el : -(el + 1);
-
- for (i = 1; i < 30; i++)
- {
- wd1 = (q6[i]*s->band[0].det) >> 12;
- if (wd < wd1)
- break;
- }
- ilow = (el < 0) ? iln[i] : ilp[i];
-
- /* Block 2L, INVQAL */
- ril = ilow >> 2;
- wd2 = qm4[ril];
- dlow = (s->band[0].det*wd2) >> 15;
-
- /* Block 3L, LOGSCL */
- il4 = rl42[ril];
- wd = (s->band[0].nb*127) >> 7;
- s->band[0].nb = wd + wl[il4];
- if (s->band[0].nb < 0)
- s->band[0].nb = 0;
- else if (s->band[0].nb > 18432)
- s->band[0].nb = 18432;
-
- /* Block 3L, SCALEL */
- wd1 = (s->band[0].nb >> 6) & 31;
- wd2 = 8 - (s->band[0].nb >> 11);
- wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
- s->band[0].det = wd3 << 2;
-
- block4(s, 0, dlow);
-
- if (s->eight_k)
- {
- /* Just leave the high bits as zero */
- code = (0xC0 | ilow) >> (8 - s->bits_per_sample);
- }
- else
- {
- /* Block 1H, SUBTRA */
- eh = saturate(xhigh - s->band[1].s);
-
- /* Block 1H, QUANTH */
- wd = (eh >= 0) ? eh : -(eh + 1);
- wd1 = (564*s->band[1].det) >> 12;
- mih = (wd >= wd1) ? 2 : 1;
- ihigh = (eh < 0) ? ihn[mih] : ihp[mih];
-
- /* Block 2H, INVQAH */
- wd2 = qm2[ihigh];
- dhigh = (s->band[1].det*wd2) >> 15;
-
- /* Block 3H, LOGSCH */
- ih2 = rh2[ihigh];
- wd = (s->band[1].nb*127) >> 7;
- s->band[1].nb = wd + wh[ih2];
- if (s->band[1].nb < 0)
- s->band[1].nb = 0;
- else if (s->band[1].nb > 22528)
- s->band[1].nb = 22528;
-
- /* Block 3H, SCALEH */
- wd1 = (s->band[1].nb >> 6) & 31;
- wd2 = 10 - (s->band[1].nb >> 11);
- wd3 = (wd2 < 0) ? (ilb[wd1] << -wd2) : (ilb[wd1] >> wd2);
- s->band[1].det = wd3 << 2;
-
- block4(s, 1, dhigh);
- code = ((ihigh << 6) | ilow) >> (8 - s->bits_per_sample);
- }
-
- if (s->packed)
- {
- /* Pack the code bits */
- s->out_buffer |= (code << s->out_bits);
- s->out_bits += s->bits_per_sample;
- if (s->out_bits >= 8)
- {
- g722_data[g722_bytes++] = (WebRtc_UWord8) (s->out_buffer & 0xFF);
- s->out_bits -= 8;
- s->out_buffer >>= 8;
- }
- }
- else
- {
- g722_data[g722_bytes++] = (WebRtc_UWord8) code;
- }
- }
- return g722_bytes;
-}
-/*- End of function --------------------------------------------------------*/
-/*- End of file ------------------------------------------------------------*/
diff --git a/trunk/src/modules/audio_coding/codecs/g722/g722_interface.c b/trunk/src/modules/audio_coding/codecs/g722/g722_interface.c
deleted file mode 100644
index d559014..0000000
--- a/trunk/src/modules/audio_coding/codecs/g722/g722_interface.c
+++ /dev/null
@@ -1,115 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-
-#include <stdlib.h>
-#include <string.h>
-#include "g722_interface.h"
-#include "g722_enc_dec.h"
-#include "typedefs.h"
-
-
-WebRtc_Word16 WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst)
-{
- *G722enc_inst=(G722EncInst*)malloc(sizeof(g722_encode_state_t));
- if (*G722enc_inst!=NULL) {
- return(0);
- } else {
- return(-1);
- }
-}
-
-WebRtc_Word16 WebRtcG722_EncoderInit(G722EncInst *G722enc_inst)
-{
- // Create and/or reset the G.722 encoder
- // Bitrate 64 kbps and wideband mode (2)
- G722enc_inst = (G722EncInst *) WebRtc_g722_encode_init(
- (g722_encode_state_t*) G722enc_inst, 64000, 2);
- if (G722enc_inst == NULL) {
- return -1;
- } else {
- return 0;
- }
-}
-
-WebRtc_Word16 WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst)
-{
- // Free encoder memory
- return WebRtc_g722_encode_release((g722_encode_state_t*) G722enc_inst);
-}
-
-WebRtc_Word16 WebRtcG722_Encode(G722EncInst *G722enc_inst,
- WebRtc_Word16 *speechIn,
- WebRtc_Word16 len,
- WebRtc_Word16 *encoded)
-{
- unsigned char *codechar = (unsigned char*) encoded;
- // Encode the input speech vector
- return WebRtc_g722_encode((g722_encode_state_t*) G722enc_inst,
- codechar, speechIn, len);
-}
-
-WebRtc_Word16 WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst)
-{
- *G722dec_inst=(G722DecInst*)malloc(sizeof(g722_decode_state_t));
- if (*G722dec_inst!=NULL) {
- return(0);
- } else {
- return(-1);
- }
-}
-
-WebRtc_Word16 WebRtcG722_DecoderInit(G722DecInst *G722dec_inst)
-{
- // Create and/or reset the G.722 decoder
- // Bitrate 64 kbps and wideband mode (2)
- G722dec_inst = (G722DecInst *) WebRtc_g722_decode_init(
- (g722_decode_state_t*) G722dec_inst, 64000, 2);
- if (G722dec_inst == NULL) {
- return -1;
- } else {
- return 0;
- }
-}
-
-WebRtc_Word16 WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst)
-{
- // Free encoder memory
- return WebRtc_g722_decode_release((g722_decode_state_t*) G722dec_inst);
-}
-
-WebRtc_Word16 WebRtcG722_Decode(G722DecInst *G722dec_inst,
- WebRtc_Word16 *encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType)
-{
- // Decode the G.722 encoder stream
- *speechType=G722_WEBRTC_SPEECH;
- return WebRtc_g722_decode((g722_decode_state_t*) G722dec_inst,
- decoded, (WebRtc_UWord8*) encoded, len);
-}
-
-WebRtc_Word16 WebRtcG722_Version(char *versionStr, short len)
-{
- // Get version string
- char version[30] = "2.0.0\n";
- if (strlen(version) < (unsigned int)len)
- {
- strcpy(versionStr, version);
- return 0;
- }
- else
- {
- return -1;
- }
-}
-
diff --git a/trunk/src/modules/audio_coding/codecs/g722/g722_unittest.cc b/trunk/src/modules/audio_coding/codecs/g722/g722_unittest.cc
deleted file mode 100644
index a828edd..0000000
--- a/trunk/src/modules/audio_coding/codecs/g722/g722_unittest.cc
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Empty test just to get code coverage metrics for this dir.
- */
-#include "g722_interface.h"
-#include "gtest/gtest.h"
-
-TEST(G722Test, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/modules/audio_coding/codecs/g722/include/g722_interface.h b/trunk/src/modules/audio_coding/codecs/g722/include/g722_interface.h
deleted file mode 100644
index e50d66f..0000000
--- a/trunk/src/modules/audio_coding/codecs/g722/include/g722_interface.h
+++ /dev/null
@@ -1,190 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef MODULES_AUDIO_CODING_CODECS_G722_MAIN_INTERFACE_G722_INTERFACE_H_
-#define MODULES_AUDIO_CODING_CODECS_G722_MAIN_INTERFACE_G722_INTERFACE_H_
-
-#include "typedefs.h"
-
-/*
- * Solution to support multiple instances
- */
-
-typedef struct WebRtcG722EncInst G722EncInst;
-typedef struct WebRtcG722DecInst G722DecInst;
-
-/*
- * Comfort noise constants
- */
-
-#define G722_WEBRTC_SPEECH 1
-#define G722_WEBRTC_CNG 2
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-
-/****************************************************************************
- * WebRtcG722_CreateEncoder(...)
- *
- * Create memory used for G722 encoder
- *
- * Input:
- * - G722enc_inst : G722 instance for encoder
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-WebRtc_Word16 WebRtcG722_CreateEncoder(G722EncInst **G722enc_inst);
-
-
-/****************************************************************************
- * WebRtcG722_EncoderInit(...)
- *
- * This function initializes a G722 instance
- *
- * Input:
- * - G722enc_inst : G722 instance, i.e. the user that should receive
- * be initialized
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcG722_EncoderInit(G722EncInst *G722enc_inst);
-
-
-/****************************************************************************
- * WebRtcG722_FreeEncoder(...)
- *
- * Free the memory used for G722 encoder
- *
- * Input:
- * - G722enc_inst : G722 instance for encoder
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-WebRtc_Word16 WebRtcG722_FreeEncoder(G722EncInst *G722enc_inst);
-
-
-
-/****************************************************************************
- * WebRtcG722_Encode(...)
- *
- * This function encodes G722 encoded data.
- *
- * Input:
- * - G722enc_inst : G722 instance, i.e. the user that should encode
- * a packet
- * - speechIn : Input speech vector
- * - len : Samples in speechIn
- *
- * Output:
- * - encoded : The encoded data vector
- *
- * Return value : >0 - Length (in bytes) of coded data
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcG722_Encode(G722EncInst *G722enc_inst,
- WebRtc_Word16 *speechIn,
- WebRtc_Word16 len,
- WebRtc_Word16 *encoded);
-
-
-/****************************************************************************
- * WebRtcG722_CreateDecoder(...)
- *
- * Create memory used for G722 encoder
- *
- * Input:
- * - G722dec_inst : G722 instance for decoder
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-WebRtc_Word16 WebRtcG722_CreateDecoder(G722DecInst **G722dec_inst);
-
-
-/****************************************************************************
- * WebRtcG722_DecoderInit(...)
- *
- * This function initializes a G729 instance
- *
- * Input:
- * - G729_decinst_t : G729 instance, i.e. the user that should receive
- * be initialized
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcG722_DecoderInit(G722DecInst *G722dec_inst);
-
-
-/****************************************************************************
- * WebRtcG722_FreeDecoder(...)
- *
- * Free the memory used for G722 decoder
- *
- * Input:
- * - G722dec_inst : G722 instance for decoder
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcG722_FreeDecoder(G722DecInst *G722dec_inst);
-
-
-/****************************************************************************
- * WebRtcG722_Decode(...)
- *
- * This function decodes a packet with G729 frame(s). Output speech length
- * will be a multiple of 80 samples (80*frames/packet).
- *
- * Input:
- * - G722dec_inst : G722 instance, i.e. the user that should decode
- * a packet
- * - encoded : Encoded G722 frame(s)
- * - len : Bytes in encoded vector
- *
- * Output:
- * - decoded : The decoded vector
- * - speechType : 1 normal, 2 CNG (Since G722 does not have its own
- * DTX/CNG scheme it should always return 1)
- *
- * Return value : >0 - Samples in decoded vector
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcG722_Decode(G722DecInst *G722dec_inst,
- WebRtc_Word16 *encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType);
-
-/****************************************************************************
- * WebRtcG722_Version(...)
- *
- * Get a string with the current version of the codec
- */
-
-WebRtc_Word16 WebRtcG722_Version(char *versionStr, short len);
-
-
-#ifdef __cplusplus
-}
-#endif
-
-
-#endif /* MODULES_AUDIO_CODING_CODECS_G722_MAIN_INTERFACE_G722_INTERFACE_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/g722/test/testG722.cc b/trunk/src/modules/audio_coding/codecs/g722/test/testG722.cc
deleted file mode 100644
index 9ef8f2d..0000000
--- a/trunk/src/modules/audio_coding/codecs/g722/test/testG722.cc
+++ /dev/null
@@ -1,157 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * testG722.cpp : Defines the entry point for the console application.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include "typedefs.h"
-
-/* include API */
-#include "g722_interface.h"
-
-/* Runtime statistics */
-#include <time.h>
-#define CLOCKS_PER_SEC_G722 100000
-
-// Forward declaration
-typedef struct WebRtcG722EncInst G722EncInst;
-typedef struct WebRtcG722DecInst G722DecInst;
-
-/* function for reading audio data from PCM file */
-int readframe(WebRtc_Word16 *data, FILE *inp, int length)
-{
- short k, rlen, status = 0;
-
- rlen = (short)fread(data, sizeof(WebRtc_Word16), length, inp);
- if (rlen < length) {
- for (k = rlen; k < length; k++)
- data[k] = 0;
- status = 1;
- }
-
- return status;
-}
-
-int main(int argc, char* argv[])
-{
- char inname[60], outbit[40], outname[40];
- FILE *inp, *outbitp, *outp;
-
- int framecnt, endfile;
- WebRtc_Word16 framelength = 160;
- G722EncInst *G722enc_inst;
- G722DecInst *G722dec_inst;
- int err;
-
- /* Runtime statistics */
- double starttime;
- double runtime = 0;
- double length_file;
-
- WebRtc_Word16 stream_len = 0;
- WebRtc_Word16 shortdata[960];
- WebRtc_Word16 decoded[960];
- WebRtc_Word16 streamdata[80*3];
- WebRtc_Word16 speechType[1];
-
- /* handling wrong input arguments in the command line */
- if (argc!=5) {
- printf("\n\nWrong number of arguments or flag values.\n\n");
-
- printf("\n");
- printf("Usage:\n\n");
- printf("./testG722.exe framelength infile outbitfile outspeechfile \n\n");
- printf("with:\n");
- printf("framelength : Framelength in samples.\n\n");
- printf("infile : Normal speech input file\n\n");
- printf("outbitfile : Bitstream output file\n\n");
- printf("outspeechfile: Speech output file\n\n");
- exit(0);
-
- }
-
- /* Get frame length */
- framelength = atoi(argv[1]);
-
- /* Get Input and Output files */
- sscanf(argv[2], "%s", inname);
- sscanf(argv[3], "%s", outbit);
- sscanf(argv[4], "%s", outname);
-
- if ((inp = fopen(inname,"rb")) == NULL) {
- printf(" G.722: Cannot read file %s.\n", inname);
- exit(1);
- }
- if ((outbitp = fopen(outbit,"wb")) == NULL) {
- printf(" G.722: Cannot write file %s.\n", outbit);
- exit(1);
- }
- if ((outp = fopen(outname,"wb")) == NULL) {
- printf(" G.722: Cannot write file %s.\n", outname);
- exit(1);
- }
- printf("\nInput:%s\nOutput bitstream:%s\nOutput:%s\n", inname, outbit, outname);
-
- /* Create and init */
- WebRtcG722_CreateEncoder((G722EncInst **)&G722enc_inst);
- WebRtcG722_CreateDecoder((G722DecInst **)&G722dec_inst);
- WebRtcG722_EncoderInit((G722EncInst *)G722enc_inst);
- WebRtcG722_DecoderInit((G722DecInst *)G722dec_inst);
-
-
- /* Initialize encoder and decoder */
- framecnt = 0;
- endfile = 0;
- while (endfile == 0) {
- framecnt++;
-
- /* Read speech block */
- endfile = readframe(shortdata, inp, framelength);
-
- /* Start clock before call to encoder and decoder */
- starttime = clock()/(double)CLOCKS_PER_SEC_G722;
-
- /* G.722 encoding + decoding */
- stream_len = WebRtcG722_Encode((G722EncInst *)G722enc_inst, shortdata, framelength, streamdata);
- err = WebRtcG722_Decode((G722DecInst *)G722dec_inst, streamdata, stream_len, decoded, speechType);
-
- /* Stop clock after call to encoder and decoder */
- runtime += (double)((clock()/(double)CLOCKS_PER_SEC_G722)-starttime);
-
- if (stream_len < 0 || err < 0) {
- /* exit if returned with error */
- printf("Error in encoder/decoder\n");
- } else {
- /* Write coded bits to file */
- fwrite(streamdata,sizeof(short),stream_len/2,outbitp);
- /* Write coded speech to file */
- fwrite(decoded,sizeof(short),framelength,outp);
- }
- }
-
- WebRtcG722_FreeEncoder((G722EncInst *)G722enc_inst);
- WebRtcG722_FreeDecoder((G722DecInst *)G722dec_inst);
-
- length_file = ((double)framecnt*(double)framelength/16000);
- printf("\n\nLength of speech file: %.1f s\n", length_file);
- printf("Time to run G.722: %.2f s (%.2f %% of realtime)\n\n", runtime, (100*runtime/length_file));
- printf("---------------------END----------------------\n");
-
- fclose(inp);
- fclose(outbitp);
- fclose(outp);
-
- return 0;
-}
-
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/interface/isacfix.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/interface/isacfix.h
deleted file mode 100644
index 28e9429..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/interface/isacfix.h
+++ /dev/null
@@ -1,633 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_ISACFIX_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_ISACFIX_H_
-
-/*
- * Define the fixpoint numeric formats
- */
-#include "typedefs.h"
-
-
-typedef struct {
- void *dummy;
-} ISACFIX_MainStruct;
-
-
-#if defined(__cplusplus)
-extern "C" {
-#endif
-
-
- /**************************************************************************
- * WebRtcIsacfix_AssignSize(...)
- *
- * Functions used when malloc is not allowed
- * Output the number of bytes needed to allocate for iSAC struct.
- *
- */
-
- WebRtc_Word16 WebRtcIsacfix_AssignSize(int *sizeinbytes);
-
- /**************************************************************************
- * WebRtcIsacfix_Assign(...)
- *
- * Functions used when malloc is not allowed, it
- * places a struct at the given address.
- *
- * Input:
- * - *ISAC_main_inst : a pointer to the coder instance.
- * - ISACFIX_inst_Addr : address of the memory where a space is
- * for iSAC structure.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsacfix_Assign(ISACFIX_MainStruct **inst,
- void *ISACFIX_inst_Addr);
-
- /****************************************************************************
- * WebRtcIsacfix_Create(...)
- *
- * This function creates an ISAC instance, which will contain the state
- * information for one coding/decoding channel.
- *
- * Input:
- * - *ISAC_main_inst : a pointer to the coder instance.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsacfix_Create(ISACFIX_MainStruct **ISAC_main_inst);
-
-
- /****************************************************************************
- * WebRtcIsacfix_Free(...)
- *
- * This function frees the ISAC instance created at the beginning.
- *
- * Input:
- * - ISAC_main_inst : a ISAC instance.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsacfix_Free(ISACFIX_MainStruct *ISAC_main_inst);
-
-
- /****************************************************************************
- * WebRtcIsacfix_EncoderInit(...)
- *
- * This function initializes an ISAC instance prior to the encoder calls.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - CodingMode : 0 - Bit rate and frame length are automatically
- * adjusted to available bandwidth on
- * transmission channel.
- * 1 - User sets a frame length and a target bit
- * rate which is taken as the maximum short-term
- * average bit rate.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst,
- WebRtc_Word16 CodingMode);
-
-
- /****************************************************************************
- * WebRtcIsacfix_Encode(...)
- *
- * This function encodes 10ms frame(s) and inserts it into a package.
- * Input speech length has to be 160 samples (10ms). The encoder buffers those
- * 10ms frames until it reaches the chosen Framesize (480 or 960 samples
- * corresponding to 30 or 60 ms frames), and then proceeds to the encoding.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - speechIn : input speech vector.
- *
- * Output:
- * - encoded : the encoded data vector
- *
- * Return value : >0 - Length (in bytes) of coded data
- * 0 - The buffer didn't reach the chosen framesize
- * so it keeps buffering speech samples.
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
- const WebRtc_Word16 *speechIn,
- WebRtc_Word16 *encoded);
-
-
-
- /****************************************************************************
- * WebRtcIsacfix_EncodeNb(...)
- *
- * This function encodes 10ms narrow band (8 kHz sampling) frame(s) and inserts
- * it into a package. Input speech length has to be 80 samples (10ms). The encoder
- * interpolates into wide-band (16 kHz sampling) buffers those
- * 10ms frames until it reaches the chosen Framesize (480 or 960 wide-band samples
- * corresponding to 30 or 60 ms frames), and then proceeds to the encoding.
- *
- * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - speechIn : input speech vector.
- *
- * Output:
- * - encoded : the encoded data vector
- *
- * Return value : >0 - Length (in bytes) of coded data
- * 0 - The buffer didn't reach the chosen framesize
- * so it keeps buffering speech samples.
- * -1 - Error
- */
-
-
-#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
- WebRtc_Word16 WebRtcIsacfix_EncodeNb(ISACFIX_MainStruct *ISAC_main_inst,
- const WebRtc_Word16 *speechIn,
- WebRtc_Word16 *encoded);
-#endif // WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-
-
-
- /****************************************************************************
- * WebRtcIsacfix_DecoderInit(...)
- *
- * This function initializes an ISAC instance prior to the decoder calls.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- *
- * Return value
- * : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst);
-
-
- /****************************************************************************
- * WebRtcIsacfix_UpdateBwEstimate1(...)
- *
- * This function updates the estimate of the bandwidth.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - encoded : encoded ISAC frame(s).
- * - packet_size : size of the packet.
- * - rtp_seq_number : the RTP number of the packet.
- * - arr_ts : the arrival time of the packet (from NetEq)
- * in samples.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst,
- const WebRtc_UWord16 *encoded,
- WebRtc_Word32 packet_size,
- WebRtc_UWord16 rtp_seq_number,
- WebRtc_UWord32 arr_ts);
-
- /****************************************************************************
- * WebRtcIsacfix_UpdateBwEstimate(...)
- *
- * This function updates the estimate of the bandwidth.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - encoded : encoded ISAC frame(s).
- * - packet_size : size of the packet.
- * - rtp_seq_number : the RTP number of the packet.
- * - send_ts : the send time of the packet from RTP header,
- * in samples.
- * - arr_ts : the arrival time of the packet (from NetEq)
- * in samples.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
- const WebRtc_UWord16 *encoded,
- WebRtc_Word32 packet_size,
- WebRtc_UWord16 rtp_seq_number,
- WebRtc_UWord32 send_ts,
- WebRtc_UWord32 arr_ts);
-
- /****************************************************************************
- * WebRtcIsacfix_Decode(...)
- *
- * This function decodes an ISAC frame. Output speech length
- * will be a multiple of 480 samples: 480 or 960 samples,
- * depending on the framesize (30 or 60 ms).
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - encoded : encoded ISAC frame(s)
- * - len : bytes in encoded vector
- *
- * Output:
- * - decoded : The decoded vector
- *
- * Return value : >0 - number of samples in decoded vector
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
- const WebRtc_UWord16 *encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType);
-
-
- /****************************************************************************
- * WebRtcIsacfix_DecodeNb(...)
- *
- * This function decodes a ISAC frame in narrow-band (8 kHz sampling).
- * Output speech length will be a multiple of 240 samples: 240 or 480 samples,
- * depending on the framesize (30 or 60 ms).
- *
- * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - encoded : encoded ISAC frame(s)
- * - len : bytes in encoded vector
- *
- * Output:
- * - decoded : The decoded vector
- *
- * Return value : >0 - number of samples in decoded vector
- * -1 - Error
- */
-
-#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
- WebRtc_Word16 WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
- const WebRtc_UWord16 *encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType);
-#endif // WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-
-
- /****************************************************************************
- * WebRtcIsacfix_DecodePlcNb(...)
- *
- * This function conducts PLC for ISAC frame(s) in narrow-band (8kHz sampling).
- * Output speech length will be "240*noOfLostFrames" samples
- * that equevalent of "30*noOfLostFrames" millisecond.
- *
- * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - noOfLostFrames : Number of PLC frames (240 sample=30ms) to produce
- * NOTE! Maximum number is 2 (480 samples = 60ms)
- *
- * Output:
- * - decoded : The decoded vector
- *
- * Return value : >0 - number of samples in decoded PLC vector
- * -1 - Error
- */
-
-#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
- WebRtc_Word16 WebRtcIsacfix_DecodePlcNb(ISACFIX_MainStruct *ISAC_main_inst,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 noOfLostFrames );
-#endif // WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-
-
-
-
- /****************************************************************************
- * WebRtcIsacfix_DecodePlc(...)
- *
- * This function conducts PLC for ISAC frame(s) in wide-band (16kHz sampling).
- * Output speech length will be "480*noOfLostFrames" samples
- * that is equevalent of "30*noOfLostFrames" millisecond.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - noOfLostFrames : Number of PLC frames (480sample = 30ms)
- * to produce
- * NOTE! Maximum number is 2 (960 samples = 60ms)
- *
- * Output:
- * - decoded : The decoded vector
- *
- * Return value : >0 - number of samples in decoded PLC vector
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct *ISAC_main_inst,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 noOfLostFrames );
-
-
- /****************************************************************************
- * WebRtcIsacfix_ReadFrameLen(...)
- *
- * This function returns the length of the frame represented in the packet.
- *
- * Input:
- * - encoded : Encoded bitstream
- *
- * Output:
- * - frameLength : Length of frame in packet (in samples)
- *
- */
-
- WebRtc_Word16 WebRtcIsacfix_ReadFrameLen(const WebRtc_Word16* encoded,
- WebRtc_Word16* frameLength);
-
- /****************************************************************************
- * WebRtcIsacfix_Control(...)
- *
- * This function sets the limit on the short-term average bit rate and the
- * frame length. Should be used only in Instantaneous mode.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - rate : limit on the short-term average bit rate,
- * in bits/second (between 10000 and 32000)
- * - framesize : number of milliseconds per frame (30 or 60)
- *
- * Return value : 0 - ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
- WebRtc_Word16 rate,
- WebRtc_Word16 framesize);
-
-
-
- /****************************************************************************
- * WebRtcIsacfix_ControlBwe(...)
- *
- * This function sets the initial values of bottleneck and frame-size if
- * iSAC is used in channel-adaptive mode. Through this API, users can
- * enforce a frame-size for all values of bottleneck. Then iSAC will not
- * automatically change the frame-size.
- *
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - rateBPS : initial value of bottleneck in bits/second
- * 10000 <= rateBPS <= 32000 is accepted
- * - frameSizeMs : number of milliseconds per frame (30 or 60)
- * - enforceFrameSize : 1 to enforce the given frame-size through out
- * the adaptation process, 0 to let iSAC change
- * the frame-size if required.
- *
- * Return value : 0 - ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
- WebRtc_Word16 rateBPS,
- WebRtc_Word16 frameSizeMs,
- WebRtc_Word16 enforceFrameSize);
-
-
-
- /****************************************************************************
- * WebRtcIsacfix_version(...)
- *
- * This function returns the version number.
- *
- * Output:
- * - version : Pointer to character string
- *
- */
-
- void WebRtcIsacfix_version(char *version);
-
-
- /****************************************************************************
- * WebRtcIsacfix_GetErrorCode(...)
- *
- * This function can be used to check the error code of an iSAC instance. When
- * a function returns -1 a error code will be set for that instance. The
- * function below extract the code of the last error that occured in the
- * specified instance.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance
- *
- * Return value : Error code
- */
-
- WebRtc_Word16 WebRtcIsacfix_GetErrorCode(ISACFIX_MainStruct *ISAC_main_inst);
-
-
- /****************************************************************************
- * WebRtcIsacfix_GetUplinkBw(...)
- *
- * This function return iSAC send bitrate
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- *
- * Return value : <0 Error code
- * else bitrate
- */
-
- WebRtc_Word32 WebRtcIsacfix_GetUplinkBw(ISACFIX_MainStruct *ISAC_main_inst);
-
-
- /****************************************************************************
- * WebRtcIsacfix_SetMaxPayloadSize(...)
- *
- * This function sets a limit for the maximum payload size of iSAC. The same
- * value is used both for 30 and 60 msec packets.
- * The absolute max will be valid until next time the function is called.
- * NOTE! This function may override the function WebRtcIsacfix_SetMaxRate()
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- * - maxPayloadBytes : maximum size of the payload in bytes
- * valid values are between 100 and 400 bytes
- *
- *
- * Return value : 0 if sucessful
- * -1 if error happens
- */
-
- WebRtc_Word16 WebRtcIsacfix_SetMaxPayloadSize(ISACFIX_MainStruct *ISAC_main_inst,
- WebRtc_Word16 maxPayloadBytes);
-
-
- /****************************************************************************
- * WebRtcIsacfix_SetMaxRate(...)
- *
- * This function sets the maximum rate which the codec may not exceed for a
- * singel packet. The maximum rate is set in bits per second.
- * The codec has an absolute maximum rate of 53400 bits per second (200 bytes
- * per 30 msec).
- * It is possible to set a maximum rate between 32000 and 53400 bits per second.
- *
- * The rate limit is valid until next time the function is called.
- *
- * NOTE! Packet size will never go above the value set if calling
- * WebRtcIsacfix_SetMaxPayloadSize() (default max packet size is 400 bytes).
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- * - maxRateInBytes : maximum rate in bits per second,
- * valid values are 32000 to 53400 bits
- *
- * Return value : 0 if sucessful
- * -1 if error happens
- */
-
- WebRtc_Word16 WebRtcIsacfix_SetMaxRate(ISACFIX_MainStruct *ISAC_main_inst,
- WebRtc_Word32 maxRate);
-
- /****************************************************************************
- * WebRtcIsacfix_CreateInternal(...)
- *
- * This function creates the memory that is used to store data in the encoder
- *
- * Input:
- * - *ISAC_main_inst : a pointer to the coder instance.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsacfix_CreateInternal(ISACFIX_MainStruct *ISAC_main_inst);
-
-
- /****************************************************************************
- * WebRtcIsacfix_FreeInternal(...)
- *
- * This function frees the internal memory for storing encoder data.
- *
- * Input:
- * - ISAC_main_inst : an ISAC instance.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsacfix_FreeInternal(ISACFIX_MainStruct *ISAC_main_inst);
-
-
- /****************************************************************************
- * WebRtcIsacfix_GetNewBitStream(...)
- *
- * This function returns encoded data, with the recieved bwe-index in the
- * stream. It should always return a complete packet, i.e. only called once
- * even for 60 msec frames
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - bweIndex : index of bandwidth estimate to put in new bitstream
- * - scale : factor for rate change (0.4 ~=> half the rate, 1 no change).
- *
- * Output:
- * - encoded : the encoded data vector
- *
- * Return value : >0 - Length (in bytes) of coded data
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
- WebRtc_Word16 bweIndex,
- float scale,
- WebRtc_Word16 *encoded);
-
-
- /****************************************************************************
- * WebRtcIsacfix_GetDownLinkBwIndex(...)
- *
- * This function returns index representing the Bandwidth estimate from
- * other side to this side.
- *
- * Input:
- * - ISAC_main_inst : iSAC struct
- *
- * Output:
- * - rateIndex : Bandwidth estimate to transmit to other side.
- *
- */
-
- WebRtc_Word16 WebRtcIsacfix_GetDownLinkBwIndex(ISACFIX_MainStruct* ISAC_main_inst,
- WebRtc_Word16* rateIndex);
-
-
- /****************************************************************************
- * WebRtcIsacfix_UpdateUplinkBw(...)
- *
- * This function takes an index representing the Bandwidth estimate from
- * this side to other side and updates BWE.
- *
- * Input:
- * - ISAC_main_inst : iSAC struct
- * - rateIndex : Bandwidth estimate from other side.
- *
- */
-
- WebRtc_Word16 WebRtcIsacfix_UpdateUplinkBw(ISACFIX_MainStruct* ISAC_main_inst,
- WebRtc_Word16 rateIndex);
-
-
- /****************************************************************************
- * WebRtcIsacfix_ReadBwIndex(...)
- *
- * This function returns the index of the Bandwidth estimate from the bitstream.
- *
- * Input:
- * - encoded : Encoded bitstream
- *
- * Output:
- * - rateIndex : Bandwidth estimate in bitstream
- *
- */
-
- WebRtc_Word16 WebRtcIsacfix_ReadBwIndex(const WebRtc_Word16* encoded,
- WebRtc_Word16* rateIndex);
-
-
- /****************************************************************************
- * WebRtcIsacfix_GetNewFrameLen(...)
- *
- * This function return the next frame length (in samples) of iSAC.
- *
- * Input:
- * -ISAC_main_inst : iSAC instance
- *
- * Return value : frame lenght in samples
- */
-
- WebRtc_Word16 WebRtcIsacfix_GetNewFrameLen(ISACFIX_MainStruct *ISAC_main_inst);
-
-
-#if defined(__cplusplus)
-}
-#endif
-
-
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_INTERFACE_ISACFIX_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/Android.mk b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/Android.mk
deleted file mode 100644
index c81d13a..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/Android.mk
+++ /dev/null
@@ -1,143 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-#############################
-# Build the non-neon library.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_isacfix
-LOCAL_MODULE_TAGS := optional
-LOCAL_SRC_FILES := \
- arith_routines.c \
- arith_routines_hist.c \
- arith_routines_logist.c \
- bandwidth_estimator.c \
- decode.c \
- decode_bwe.c \
- decode_plc.c \
- encode.c \
- entropy_coding.c \
- fft.c \
- filterbank_tables.c \
- filterbanks.c \
- filters.c \
- initialize.c \
- isacfix.c \
- lattice.c \
- lpc_masking_model.c \
- lpc_tables.c \
- pitch_estimator.c \
- pitch_filter.c \
- pitch_gain_tables.c \
- pitch_lag_tables.c \
- spectrum_ar_model_tables.c \
- transform.c
-
-ifeq ($(ARCH_ARM_HAVE_ARMV7A),true)
-LOCAL_SRC_FILES += \
- lattice_armv7.S
-else
-LOCAL_SRC_FILES += \
- lattice_c.c
-endif
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../../../../.. \
- $(LOCAL_PATH)/../../../../../../common_audio/signal_processing/include
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
-
-#########################
-# Build the neon library.
-ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_isacfix_neon
-LOCAL_MODULE_TAGS := optional
-LOCAL_SRC_FILES := \
- filters_neon.c \
- lattice_neon.S #.S extention is for including a header file in assembly.
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS) \
- -mfpu=neon \
- -mfloat-abi=softfp \
- -flax-vector-conversions
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../../../../.. \
- $(LOCAL_PATH)/../../../../../../common_audio/signal_processing/include
-
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
-
-endif # ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
-
-###########################
-# isac test app
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_TAGS := tests
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES:= ../test/kenny.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../../../../..
-
-LOCAL_STATIC_LIBRARIES := \
- libwebrtc_isacfix \
- libwebrtc_spl
-
-ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
-LOCAL_STATIC_LIBRARIES += \
- libwebrtc_isacfix_neon
-endif
-
-LOCAL_SHARED_LIBRARIES := \
- libutils
-
-LOCAL_MODULE:= webrtc_isac_test
-
-ifdef NDK_ROOT
-include $(BUILD_EXECUTABLE)
-else
-include $(BUILD_NATIVE_TEST)
-endif
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines.c
deleted file mode 100644
index ee62bad..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines.c
+++ /dev/null
@@ -1,124 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * arith_routins.c
- *
- * This C file contains a function for finalizing the bitstream
- * after arithmetic coding.
- *
- */
-
-#include "arith_routins.h"
-
-
-/****************************************************************************
- * WebRtcIsacfix_EncTerminate(...)
- *
- * Final call to the arithmetic coder for an encoder call. This function
- * terminates and return byte stream.
- *
- * Input:
- * - streamData : in-/output struct containing bitstream
- *
- * Return value : number of bytes in the stream
- */
-WebRtc_Word16 WebRtcIsacfix_EncTerminate(Bitstr_enc *streamData)
-{
- WebRtc_UWord16 *streamPtr;
- WebRtc_UWord16 negCarry;
-
- /* point to the right place in the stream buffer */
- streamPtr = streamData->stream + streamData->stream_index;
-
- /* find minimum length (determined by current interval width) */
- if ( streamData->W_upper > 0x01FFFFFF )
- {
- streamData->streamval += 0x01000000;
-
- /* if result is less than the added value we must take care of the carry */
- if (streamData->streamval < 0x01000000)
- {
- /* propagate carry */
- if (streamData->full == 0) {
- /* Add value to current value */
- negCarry = *streamPtr;
- negCarry += 0x0100;
- *streamPtr = negCarry;
-
- /* if value is too big, propagate carry to next byte, and so on */
- while (!(negCarry))
- {
- negCarry = *--streamPtr;
- negCarry++;
- *streamPtr = negCarry;
- }
- } else {
- /* propagate carry by adding one to the previous byte in the
- * stream if that byte is 0xFFFF we need to propagate the carry
- * furhter back in the stream */
- while ( !(++(*--streamPtr)) );
- }
-
- /* put pointer back to the old value */
- streamPtr = streamData->stream + streamData->stream_index;
- }
- /* write remaining data to bitstream, if "full == 0" first byte has data */
- if (streamData->full == 0) {
- *streamPtr++ += (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24);
- streamData->full = 1;
- } else {
- *streamPtr = (WebRtc_UWord16) WEBRTC_SPL_LSHIFT_W32(
- WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24), 8);
- streamData->full = 0;
- }
- }
- else
- {
- streamData->streamval += 0x00010000;
-
- /* if result is less than the added value we must take care of the carry */
- if (streamData->streamval < 0x00010000)
- {
- /* propagate carry */
- if (streamData->full == 0) {
- /* Add value to current value */
- negCarry = *streamPtr;
- negCarry += 0x0100;
- *streamPtr = negCarry;
-
- /* if value to big, propagate carry to next byte, and so on */
- while (!(negCarry))
- {
- negCarry = *--streamPtr;
- negCarry++;
- *streamPtr = negCarry;
- }
- } else {
- /* Add carry to previous byte */
- while ( !(++(*--streamPtr)) );
- }
-
- /* put pointer back to the old value */
- streamPtr = streamData->stream + streamData->stream_index;
- }
- /* write remaining data (2 bytes) to bitstream */
- if (streamData->full) {
- *streamPtr++ = (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 16);
- } else {
- *streamPtr++ |= (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24);
- *streamPtr = (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 8)
- & 0xFF00;
- }
- }
-
- /* calculate stream length in bytes */
- return (((streamPtr - streamData->stream)<<1) + !(streamData->full));
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines_hist.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines_hist.c
deleted file mode 100644
index 14f1add..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines_hist.c
+++ /dev/null
@@ -1,404 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * arith_routinshist.c
- *
- * This C file contains arithmetic encoding and decoding.
- *
- */
-
-#include "arith_routins.h"
-
-
-/****************************************************************************
- * WebRtcIsacfix_EncHistMulti(...)
- *
- * Encode the histogram interval
- *
- * Input:
- * - streamData : in-/output struct containing bitstream
- * - data : data vector
- * - cdf : array of cdf arrays
- * - lenData : data vector length
- *
- * Return value : 0 if ok
- * <0 if error detected
- */
-int WebRtcIsacfix_EncHistMulti(Bitstr_enc *streamData,
- const WebRtc_Word16 *data,
- const WebRtc_UWord16 **cdf,
- const WebRtc_Word16 lenData)
-{
- WebRtc_UWord32 W_lower;
- WebRtc_UWord32 W_upper;
- WebRtc_UWord32 W_upper_LSB;
- WebRtc_UWord32 W_upper_MSB;
- WebRtc_UWord16 *streamPtr;
- WebRtc_UWord16 negCarry;
- WebRtc_UWord16 *maxStreamPtr;
- WebRtc_UWord16 *streamPtrCarry;
- WebRtc_UWord32 cdfLo;
- WebRtc_UWord32 cdfHi;
- int k;
-
-
- /* point to beginning of stream buffer
- * and set maximum streamPtr value */
- streamPtr = streamData->stream + streamData->stream_index;
- maxStreamPtr = streamData->stream + STREAM_MAXW16_60MS - 1;
-
- W_upper = streamData->W_upper;
-
- for (k = lenData; k > 0; k--)
- {
- /* fetch cdf_lower and cdf_upper from cdf tables */
- cdfLo = (WebRtc_UWord32) *(*cdf + (WebRtc_UWord32)*data);
- cdfHi = (WebRtc_UWord32) *(*cdf++ + (WebRtc_UWord32)*data++ + 1);
-
- /* update interval */
- W_upper_LSB = W_upper & 0x0000FFFF;
- W_upper_MSB = WEBRTC_SPL_RSHIFT_W32(W_upper, 16);
- W_lower = WEBRTC_SPL_UMUL(W_upper_MSB, cdfLo);
- W_lower += WEBRTC_SPL_UMUL_RSFT16(W_upper_LSB, cdfLo);
- W_upper = WEBRTC_SPL_UMUL(W_upper_MSB, cdfHi);
- W_upper += WEBRTC_SPL_UMUL_RSFT16(W_upper_LSB, cdfHi);
-
- /* shift interval such that it begins at zero */
- W_upper -= ++W_lower;
-
- /* add integer to bitstream */
- streamData->streamval += W_lower;
-
- /* handle carry */
- if (streamData->streamval < W_lower)
- {
- /* propagate carry */
- streamPtrCarry = streamPtr;
- if (streamData->full == 0) {
- negCarry = *streamPtrCarry;
- negCarry += 0x0100;
- *streamPtrCarry = negCarry;
- while (!(negCarry))
- {
- negCarry = *--streamPtrCarry;
- negCarry++;
- *streamPtrCarry = negCarry;
- }
- } else {
- while ( !(++(*--streamPtrCarry)) );
- }
- }
-
- /* renormalize interval, store most significant byte of streamval and update streamval
- * W_upper < 2^24 */
- while ( !(W_upper & 0xFF000000) )
- {
- W_upper = WEBRTC_SPL_LSHIFT_W32(W_upper, 8);
- if (streamData->full == 0) {
- *streamPtr++ += (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24);
- streamData->full = 1;
- } else {
- *streamPtr = (WebRtc_UWord16) WEBRTC_SPL_LSHIFT_W32(
- WEBRTC_SPL_RSHIFT_W32(streamData->streamval, 24), 8);
- streamData->full = 0;
- }
-
- if( streamPtr > maxStreamPtr ) {
- return -ISAC_DISALLOWED_BITSTREAM_LENGTH;
- }
- streamData->streamval = WEBRTC_SPL_LSHIFT_W32(streamData->streamval, 8);
- }
- }
-
- /* calculate new stream_index */
- streamData->stream_index = streamPtr - streamData->stream;
- streamData->W_upper = W_upper;
-
- return 0;
-}
-
-
-/****************************************************************************
- * WebRtcIsacfix_DecHistBisectMulti(...)
- *
- * Function to decode more symbols from the arithmetic bytestream, using
- * method of bisection cdf tables should be of size 2^k-1 (which corresponds
- * to an alphabet size of 2^k-2)
- *
- * Input:
- * - streamData : in-/output struct containing bitstream
- * - cdf : array of cdf arrays
- * - cdfSize : array of cdf table sizes+1 (power of two: 2^k)
- * - lenData : data vector length
- *
- * Output:
- * - data : data vector
- *
- * Return value : number of bytes in the stream
- * <0 if error detected
- */
-WebRtc_Word16 WebRtcIsacfix_DecHistBisectMulti(WebRtc_Word16 *data,
- Bitstr_dec *streamData,
- const WebRtc_UWord16 **cdf,
- const WebRtc_UWord16 *cdfSize,
- const WebRtc_Word16 lenData)
-{
- WebRtc_UWord32 W_lower = 0;
- WebRtc_UWord32 W_upper;
- WebRtc_UWord32 W_tmp;
- WebRtc_UWord32 W_upper_LSB;
- WebRtc_UWord32 W_upper_MSB;
- WebRtc_UWord32 streamval;
- const WebRtc_UWord16 *streamPtr;
- const WebRtc_UWord16 *cdfPtr;
- WebRtc_Word16 sizeTmp;
- int k;
-
-
- streamPtr = streamData->stream + streamData->stream_index;
- W_upper = streamData->W_upper;
-
- /* Error check: should not be possible in normal operation */
- if (W_upper == 0) {
- return -2;
- }
-
- /* first time decoder is called for this stream */
- if (streamData->stream_index == 0)
- {
- /* read first word from bytestream */
- streamval = WEBRTC_SPL_LSHIFT_W32((WebRtc_UWord32)*streamPtr++, 16);
- streamval |= *streamPtr++;
- } else {
- streamval = streamData->streamval;
- }
-
- for (k = lenData; k > 0; k--)
- {
- /* find the integer *data for which streamval lies in [W_lower+1, W_upper] */
- W_upper_LSB = W_upper & 0x0000FFFF;
- W_upper_MSB = WEBRTC_SPL_RSHIFT_W32(W_upper, 16);
-
- /* start halfway the cdf range */
- sizeTmp = WEBRTC_SPL_RSHIFT_W16(*cdfSize++, 1);
- cdfPtr = *cdf + (sizeTmp - 1);
-
- /* method of bisection */
- for ( ;; )
- {
- W_tmp = WEBRTC_SPL_UMUL_32_16(W_upper_MSB, *cdfPtr);
- W_tmp += WEBRTC_SPL_UMUL_32_16_RSFT16(W_upper_LSB, *cdfPtr);
- sizeTmp = WEBRTC_SPL_RSHIFT_W16(sizeTmp, 1);
- if (sizeTmp == 0) {
- break;
- }
-
- if (streamval > W_tmp)
- {
- W_lower = W_tmp;
- cdfPtr += sizeTmp;
- } else {
- W_upper = W_tmp;
- cdfPtr -= sizeTmp;
- }
- }
- if (streamval > W_tmp)
- {
- W_lower = W_tmp;
- *data++ = cdfPtr - *cdf++;
- } else {
- W_upper = W_tmp;
- *data++ = cdfPtr - *cdf++ - 1;
- }
-
- /* shift interval to start at zero */
- W_upper -= ++W_lower;
-
- /* add integer to bitstream */
- streamval -= W_lower;
-
- /* renormalize interval and update streamval */
- /* W_upper < 2^24 */
- while ( !(W_upper & 0xFF000000) )
- {
- /* read next byte from stream */
- if (streamData->full == 0) {
- streamval = WEBRTC_SPL_LSHIFT_W32(streamval, 8) |
- (*streamPtr++ & 0x00FF);
- streamData->full = 1;
- } else {
- streamval = WEBRTC_SPL_LSHIFT_W32(streamval, 8) |
- WEBRTC_SPL_RSHIFT_W16(*streamPtr, 8);
- streamData->full = 0;
- }
- W_upper = WEBRTC_SPL_LSHIFT_W32(W_upper, 8);
- }
-
-
- /* Error check: should not be possible in normal operation */
- if (W_upper == 0) {
- return -2;
- }
-
- }
-
- streamData->stream_index = streamPtr - streamData->stream;
- streamData->W_upper = W_upper;
- streamData->streamval = streamval;
-
- if ( W_upper > 0x01FFFFFF ) {
- return (streamData->stream_index*2 - 3 + !streamData->full);
- } else {
- return (streamData->stream_index*2 - 2 + !streamData->full);
- }
-}
-
-
-/****************************************************************************
- * WebRtcIsacfix_DecHistOneStepMulti(...)
- *
- * Function to decode more symbols from the arithmetic bytestream, taking
- * single step up or down at a time.
- * cdf tables can be of arbitrary size, but large tables may take a lot of
- * iterations.
- *
- * Input:
- * - streamData : in-/output struct containing bitstream
- * - cdf : array of cdf arrays
- * - initIndex : vector of initial cdf table search entries
- * - lenData : data vector length
- *
- * Output:
- * - data : data vector
- *
- * Return value : number of bytes in original stream
- * <0 if error detected
- */
-WebRtc_Word16 WebRtcIsacfix_DecHistOneStepMulti(WebRtc_Word16 *data,
- Bitstr_dec *streamData,
- const WebRtc_UWord16 **cdf,
- const WebRtc_UWord16 *initIndex,
- const WebRtc_Word16 lenData)
-{
- WebRtc_UWord32 W_lower;
- WebRtc_UWord32 W_upper;
- WebRtc_UWord32 W_tmp;
- WebRtc_UWord32 W_upper_LSB;
- WebRtc_UWord32 W_upper_MSB;
- WebRtc_UWord32 streamval;
- const WebRtc_UWord16 *streamPtr;
- const WebRtc_UWord16 *cdfPtr;
- int k;
-
-
- streamPtr = streamData->stream + streamData->stream_index;
- W_upper = streamData->W_upper;
- /* Error check: Should not be possible in normal operation */
- if (W_upper == 0) {
- return -2;
- }
-
- /* Check if it is the first time decoder is called for this stream */
- if (streamData->stream_index == 0)
- {
- /* read first word from bytestream */
- streamval = WEBRTC_SPL_LSHIFT_U32(*streamPtr++, 16);
- streamval |= *streamPtr++;
- } else {
- streamval = streamData->streamval;
- }
-
- for (k = lenData; k > 0; k--)
- {
- /* find the integer *data for which streamval lies in [W_lower+1, W_upper] */
- W_upper_LSB = W_upper & 0x0000FFFF;
- W_upper_MSB = WEBRTC_SPL_RSHIFT_U32(W_upper, 16);
-
- /* start at the specified table entry */
- cdfPtr = *cdf + (*initIndex++);
- W_tmp = WEBRTC_SPL_UMUL_32_16(W_upper_MSB, *cdfPtr);
- W_tmp += WEBRTC_SPL_UMUL_32_16_RSFT16(W_upper_LSB, *cdfPtr);
-
- if (streamval > W_tmp)
- {
- for ( ;; )
- {
- W_lower = W_tmp;
-
- /* range check */
- if (cdfPtr[0] == 65535) {
- return -3;
- }
-
- W_tmp = WEBRTC_SPL_UMUL_32_16(W_upper_MSB, *++cdfPtr);
- W_tmp += WEBRTC_SPL_UMUL_32_16_RSFT16(W_upper_LSB, *cdfPtr);
-
- if (streamval <= W_tmp) {
- break;
- }
- }
- W_upper = W_tmp;
- *data++ = cdfPtr - *cdf++ - 1;
- } else {
- for ( ;; )
- {
- W_upper = W_tmp;
- --cdfPtr;
-
- /* range check */
- if (cdfPtr < *cdf) {
- return -3;
- }
-
- W_tmp = WEBRTC_SPL_UMUL_32_16(W_upper_MSB, *cdfPtr);
- W_tmp += WEBRTC_SPL_UMUL_32_16_RSFT16(W_upper_LSB, *cdfPtr);
-
- if (streamval > W_tmp) {
- break;
- }
- }
- W_lower = W_tmp;
- *data++ = cdfPtr - *cdf++;
- }
-
- /* shift interval to start at zero */
- W_upper -= ++W_lower;
-
- /* add integer to bitstream */
- streamval -= W_lower;
-
- /* renormalize interval and update streamval */
- /* W_upper < 2^24 */
- while ( !(W_upper & 0xFF000000) )
- {
- /* read next byte from stream */
- if (streamData->full == 0) {
- streamval = WEBRTC_SPL_LSHIFT_W32(streamval, 8) | (*streamPtr++ & 0x00FF);
- streamData->full = 1;
- } else {
- streamval = WEBRTC_SPL_LSHIFT_W32(streamval, 8) | (*streamPtr >> 8);
- streamData->full = 0;
- }
- W_upper = WEBRTC_SPL_LSHIFT_W32(W_upper, 8);
- }
- }
-
- streamData->stream_index = streamPtr - streamData->stream;
- streamData->W_upper = W_upper;
- streamData->streamval = streamval;
-
- /* find number of bytes in original stream (determined by current interval width) */
- if ( W_upper > 0x01FFFFFF ) {
- return (streamData->stream_index*2 - 3 + !streamData->full);
- } else {
- return (streamData->stream_index*2 - 2 + !streamData->full);
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines_logist.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines_logist.c
deleted file mode 100644
index 39c437e..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routines_logist.c
+++ /dev/null
@@ -1,404 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * arith_routinslogist.c
- *
- * This C file contains arithmetic encode and decode logistic
- *
- */
-
-#include "arith_routins.h"
-
-
-/* Tables for piecewise linear cdf functions: y = k*x */
-
-/* x Points for function piecewise() in Q15 */
-static const WebRtc_Word32 kHistEdges[51] = {
- -327680, -314573, -301466, -288359, -275252, -262144, -249037, -235930, -222823, -209716,
- -196608, -183501, -170394, -157287, -144180, -131072, -117965, -104858, -91751, -78644,
- -65536, -52429, -39322, -26215, -13108, 0, 13107, 26214, 39321, 52428,
- 65536, 78643, 91750, 104857, 117964, 131072, 144179, 157286, 170393, 183500,
- 196608, 209715, 222822, 235929, 249036, 262144, 275251, 288358, 301465, 314572,
- 327680
-};
-
-
-/* k Points for function piecewise() in Q0 */
-static const WebRtc_UWord16 kCdfSlope[51] = {
- 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
- 5, 5, 13, 23, 47, 87, 154, 315, 700, 1088,
- 2471, 6064, 14221, 21463, 36634, 36924, 19750, 13270, 5806, 2312,
- 1095, 660, 316, 145, 86, 41, 32, 5, 5, 5,
- 5, 5, 5, 5, 5, 5, 5, 5, 5, 2,
- 0
-};
-
-/* y Points for function piecewise() in Q0 */
-static const WebRtc_UWord16 kCdfLogistic[51] = {
- 0, 2, 4, 6, 8, 10, 12, 14, 16, 18,
- 20, 22, 24, 29, 38, 57, 92, 153, 279, 559,
- 994, 1983, 4408, 10097, 18682, 33336, 48105, 56005, 61313, 63636,
- 64560, 64998, 65262, 65389, 65447, 65481, 65497, 65510, 65512, 65514,
- 65516, 65518, 65520, 65522, 65524, 65526, 65528, 65530, 65532, 65534,
- 65535
-};
-
-
-/****************************************************************************
- * WebRtcIsacfix_Piecewise(...)
- *
- * Piecewise linear function
- *
- * Input:
- * - xinQ15 : input value x in Q15
- *
- * Return value : korresponding y-value in Q0
- */
-
-
-static __inline WebRtc_UWord16 WebRtcIsacfix_Piecewise(WebRtc_Word32 xinQ15) {
- WebRtc_Word32 ind;
- WebRtc_Word32 qtmp1;
- WebRtc_UWord16 qtmp2;
-
- /* Find index for x-value */
- qtmp1 = WEBRTC_SPL_SAT(kHistEdges[50],xinQ15,kHistEdges[0]);
- ind = WEBRTC_SPL_MUL(5, qtmp1 - kHistEdges[0]);
- ind = WEBRTC_SPL_RSHIFT_W32(ind, 16);
-
- /* Calculate corresponding y-value ans return*/
- qtmp1 = qtmp1 - kHistEdges[ind];
- qtmp2 = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_U32(
- WEBRTC_SPL_UMUL_32_16(qtmp1,kCdfSlope[ind]), 15);
- return (kCdfLogistic[ind] + qtmp2);
-}
-
-/****************************************************************************
- * WebRtcIsacfix_EncLogisticMulti2(...)
- *
- * Arithmetic coding of spectrum.
- *
- * Input:
- * - streamData : in-/output struct containing bitstream
- * - dataQ7 : data vector in Q7
- * - envQ8 : side info vector defining the width of the pdf
- * in Q8
- * - lenData : data vector length
- *
- * Return value : 0 if ok,
- * <0 otherwise.
- */
-int WebRtcIsacfix_EncLogisticMulti2(Bitstr_enc *streamData,
- WebRtc_Word16 *dataQ7,
- const WebRtc_UWord16 *envQ8,
- const WebRtc_Word16 lenData)
-{
- WebRtc_UWord32 W_lower;
- WebRtc_UWord32 W_upper;
- WebRtc_UWord16 W_upper_LSB;
- WebRtc_UWord16 W_upper_MSB;
- WebRtc_UWord16 *streamPtr;
- WebRtc_UWord16 *maxStreamPtr;
- WebRtc_UWord16 *streamPtrCarry;
- WebRtc_UWord16 negcarry;
- WebRtc_UWord32 cdfLo;
- WebRtc_UWord32 cdfHi;
- int k;
-
- /* point to beginning of stream buffer
- * and set maximum streamPtr value */
- streamPtr = streamData->stream + streamData->stream_index;
- maxStreamPtr = streamData->stream + STREAM_MAXW16_60MS - 1;
- W_upper = streamData->W_upper;
-
- for (k = 0; k < lenData; k++)
- {
- /* compute cdf_lower and cdf_upper by evaluating the
- * WebRtcIsacfix_Piecewise linear cdf */
- cdfLo = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(*dataQ7 - 64, *envQ8));
- cdfHi = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(*dataQ7 + 64, *envQ8));
-
- /* test and clip if probability gets too small */
- while ((cdfLo + 1) >= cdfHi) {
- /* clip */
- if (*dataQ7 > 0) {
- *dataQ7 -= 128;
- cdfHi = cdfLo;
- cdfLo = WebRtcIsacfix_Piecewise(
- WEBRTC_SPL_MUL_16_U16(*dataQ7 - 64, *envQ8));
- } else {
- *dataQ7 += 128;
- cdfLo = cdfHi;
- cdfHi = WebRtcIsacfix_Piecewise(
- WEBRTC_SPL_MUL_16_U16(*dataQ7 + 64, *envQ8));
- }
- }
-
- dataQ7++;
- /* increment only once per 4 iterations */
- envQ8 += (k & 1) & (k >> 1);
-
-
- /* update interval */
- W_upper_LSB = (WebRtc_UWord16)W_upper;
- W_upper_MSB = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_U32(W_upper, 16);
- W_lower = WEBRTC_SPL_UMUL_32_16(cdfLo, W_upper_MSB);
- W_lower += WEBRTC_SPL_UMUL_32_16_RSFT16(cdfLo, W_upper_LSB);
- W_upper = WEBRTC_SPL_UMUL_32_16(cdfHi, W_upper_MSB);
- W_upper += WEBRTC_SPL_UMUL_32_16_RSFT16(cdfHi, W_upper_LSB);
-
- /* shift interval such that it begins at zero */
- W_upper -= ++W_lower;
-
- /* add integer to bitstream */
- streamData->streamval += W_lower;
-
- /* handle carry */
- if (streamData->streamval < W_lower)
- {
- /* propagate carry */
- streamPtrCarry = streamPtr;
- if (streamData->full == 0) {
- negcarry = *streamPtrCarry;
- negcarry += 0x0100;
- *streamPtrCarry = negcarry;
- while (!(negcarry))
- {
- negcarry = *--streamPtrCarry;
- negcarry++;
- *streamPtrCarry = negcarry;
- }
- } else {
- while (!(++(*--streamPtrCarry)));
- }
- }
-
- /* renormalize interval, store most significant byte of streamval and update streamval
- * W_upper < 2^24 */
- while ( !(W_upper & 0xFF000000) )
- {
- W_upper = WEBRTC_SPL_LSHIFT_U32(W_upper, 8);
- if (streamData->full == 0) {
- *streamPtr++ += (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_U32(
- streamData->streamval, 24);
- streamData->full = 1;
- } else {
- *streamPtr = (WebRtc_UWord16) WEBRTC_SPL_LSHIFT_U32(
- WEBRTC_SPL_RSHIFT_U32(streamData->streamval, 24), 8);
- streamData->full = 0;
- }
-
- if( streamPtr > maxStreamPtr )
- return -ISAC_DISALLOWED_BITSTREAM_LENGTH;
-
- streamData->streamval = WEBRTC_SPL_LSHIFT_U32(streamData->streamval, 8);
- }
- }
-
- /* calculate new stream_index */
- streamData->stream_index = streamPtr - streamData->stream;
- streamData->W_upper = W_upper;
-
- return 0;
-}
-
-
-/****************************************************************************
- * WebRtcIsacfix_DecLogisticMulti2(...)
- *
- * Arithmetic decoding of spectrum.
- *
- * Input:
- * - streamData : in-/output struct containing bitstream
- * - envQ8 : side info vector defining the width of the pdf
- * in Q8
- * - lenData : data vector length
- *
- * Input/Output:
- * - dataQ7 : input: dither vector, output: data vector
- *
- * Return value : number of bytes in the stream so far
- * -1 if error detected
- */
-WebRtc_Word16 WebRtcIsacfix_DecLogisticMulti2(WebRtc_Word16 *dataQ7,
- Bitstr_dec *streamData,
- const WebRtc_Word32 *envQ8,
- const WebRtc_Word16 lenData)
-{
- WebRtc_UWord32 W_lower;
- WebRtc_UWord32 W_upper;
- WebRtc_UWord32 W_tmp;
- WebRtc_UWord16 W_upper_LSB;
- WebRtc_UWord16 W_upper_MSB;
- WebRtc_UWord32 streamVal;
- WebRtc_UWord16 cdfTmp;
- WebRtc_Word32 res;
- WebRtc_Word32 inSqrt;
- WebRtc_Word32 newRes;
- const WebRtc_UWord16 *streamPtr;
- WebRtc_Word16 candQ7;
- WebRtc_Word16 envCount;
- WebRtc_UWord16 tmpARSpecQ8 = 0;
- int k, i;
-
-
- /* point to beginning of stream buffer */
- streamPtr = streamData->stream + streamData->stream_index;
- W_upper = streamData->W_upper;
-
- /* Check if it is first time decoder is called for this stream */
- if (streamData->stream_index == 0)
- {
- /* read first word from bytestream */
- streamVal = WEBRTC_SPL_LSHIFT_U32(*streamPtr++, 16);
- streamVal |= *streamPtr++;
-
- } else {
- streamVal = streamData->streamval;
- }
-
-
- res = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1,
- WEBRTC_SPL_RSHIFT_W16(WebRtcSpl_GetSizeInBits(envQ8[0]), 1));
- envCount = 0;
-
- /* code assumes lenData%4 == 0 */
- for (k = 0; k < lenData; k += 4)
- {
- int k4;
-
- /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */
- inSqrt = envQ8[envCount];
- i = 10;
-
- /* For safty reasons */
- if (inSqrt < 0)
- inSqrt=-inSqrt;
-
- newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(inSqrt, res) + res, 1);
- do
- {
- res = newRes;
- newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(inSqrt, res) + res, 1);
- } while (newRes != res && i-- > 0);
-
- tmpARSpecQ8 = (WebRtc_UWord16)newRes;
-
- for(k4 = 0; k4 < 4; k4++)
- {
- /* find the integer *data for which streamVal lies in [W_lower+1, W_upper] */
- W_upper_LSB = (WebRtc_UWord16) (W_upper & 0x0000FFFF);
- W_upper_MSB = (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_U32(W_upper, 16);
-
- /* find first candidate by inverting the logistic cdf
- * Input dither value collected from io-stream */
- candQ7 = - *dataQ7 + 64;
- cdfTmp = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8));
-
- W_tmp = WEBRTC_SPL_UMUL_16_16(cdfTmp, W_upper_MSB);
- W_tmp += WEBRTC_SPL_UMUL_16_16_RSFT16(cdfTmp, W_upper_LSB);
-
- if (streamVal > W_tmp)
- {
- W_lower = W_tmp;
- candQ7 += 128;
- cdfTmp = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8));
-
- W_tmp = WEBRTC_SPL_UMUL_16_16(cdfTmp, W_upper_MSB);
- W_tmp += WEBRTC_SPL_UMUL_16_16_RSFT16(cdfTmp, W_upper_LSB);
-
- while (streamVal > W_tmp)
- {
- W_lower = W_tmp;
- candQ7 += 128;
- cdfTmp = WebRtcIsacfix_Piecewise(
- WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8));
-
- W_tmp = WEBRTC_SPL_UMUL_16_16(cdfTmp, W_upper_MSB);
- W_tmp += WEBRTC_SPL_UMUL_16_16_RSFT16(cdfTmp, W_upper_LSB);
-
- /* error check */
- if (W_lower == W_tmp) {
- return -1;
- }
- }
- W_upper = W_tmp;
-
- /* Output value put in dataQ7: another sample decoded */
- *dataQ7 = candQ7 - 64;
- }
- else
- {
- W_upper = W_tmp;
- candQ7 -= 128;
- cdfTmp = WebRtcIsacfix_Piecewise(WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8));
-
- W_tmp = WEBRTC_SPL_UMUL_16_16(cdfTmp, W_upper_MSB);
- W_tmp += WEBRTC_SPL_UMUL_16_16_RSFT16(cdfTmp, W_upper_LSB);
-
- while ( !(streamVal > W_tmp) )
- {
- W_upper = W_tmp;
- candQ7 -= 128;
- cdfTmp = WebRtcIsacfix_Piecewise(
- WEBRTC_SPL_MUL_16_U16(candQ7, tmpARSpecQ8));
-
- W_tmp = WEBRTC_SPL_UMUL_16_16(cdfTmp, W_upper_MSB);
- W_tmp += WEBRTC_SPL_UMUL_16_16_RSFT16(cdfTmp, W_upper_LSB);
-
- /* error check */
- if (W_upper == W_tmp){
- return -1;
- }
- }
- W_lower = W_tmp;
-
- /* Output value put in dataQ7: another sample decoded */
- *dataQ7 = candQ7 + 64;
- }
-
- dataQ7++;
-
- /* shift interval to start at zero */
- W_upper -= ++W_lower;
-
- /* add integer to bitstream */
- streamVal -= W_lower;
-
- /* renormalize interval and update streamVal
- * W_upper < 2^24 */
- while ( !(W_upper & 0xFF000000) )
- {
- /* read next byte from stream */
- if (streamData->full == 0) {
- streamVal = WEBRTC_SPL_LSHIFT_W32(streamVal, 8) | (*streamPtr++ & 0x00FF);
- streamData->full = 1;
- } else {
- streamVal = WEBRTC_SPL_LSHIFT_W32(streamVal, 8) |
- WEBRTC_SPL_RSHIFT_U16(*streamPtr, 8);
- streamData->full = 0;
- }
- W_upper = WEBRTC_SPL_LSHIFT_W32(W_upper, 8);
- }
- }
- envCount++;
- }
-
- streamData->stream_index = streamPtr - streamData->stream;
- streamData->W_upper = W_upper;
- streamData->streamval = streamVal;
-
- /* find number of bytes in original stream (determined by current interval width) */
- if ( W_upper > 0x01FFFFFF )
- return (streamData->stream_index*2 - 3 + !streamData->full);
- else
- return (streamData->stream_index*2 - 2 + !streamData->full);
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routins.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routins.h
deleted file mode 100644
index 9aa49da..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/arith_routins.h
+++ /dev/null
@@ -1,160 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * arith_routins.h
- *
- * Functions for arithmetic coding.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_
-
-#include "structs.h"
-
-
-/****************************************************************************
- * WebRtcIsacfix_EncLogisticMulti2(...)
- *
- * Arithmetic coding of spectrum.
- *
- * Input:
- * - streamData : in-/output struct containing bitstream
- * - dataQ7 : data vector in Q7
- * - envQ8 : side info vector defining the width of the pdf
- * in Q8
- * - lenData : data vector length
- *
- * Return value : 0 if ok,
- * <0 otherwise.
- */
-int WebRtcIsacfix_EncLogisticMulti2(
- Bitstr_enc *streamData,
- WebRtc_Word16 *dataQ7,
- const WebRtc_UWord16 *env,
- const WebRtc_Word16 lenData);
-
-
-/****************************************************************************
- * WebRtcIsacfix_EncTerminate(...)
- *
- * Final call to the arithmetic coder for an encoder call. This function
- * terminates and return byte stream.
- *
- * Input:
- * - streamData : in-/output struct containing bitstream
- *
- * Return value : number of bytes in the stream
- */
-WebRtc_Word16 WebRtcIsacfix_EncTerminate(Bitstr_enc *streamData);
-
-
-/****************************************************************************
- * WebRtcIsacfix_DecLogisticMulti2(...)
- *
- * Arithmetic decoding of spectrum.
- *
- * Input:
- * - streamData : in-/output struct containing bitstream
- * - envQ8 : side info vector defining the width of the pdf
- * in Q8
- * - lenData : data vector length
- *
- * Input/Output:
- * - dataQ7 : input: dither vector, output: data vector, in Q7
- *
- * Return value : number of bytes in the stream so far
- * <0 if error detected
- */
-WebRtc_Word16 WebRtcIsacfix_DecLogisticMulti2(
- WebRtc_Word16 *data,
- Bitstr_dec *streamData,
- const WebRtc_Word32 *env,
- const WebRtc_Word16 lenData);
-
-
-/****************************************************************************
- * WebRtcIsacfix_EncHistMulti(...)
- *
- * Encode the histogram interval
- *
- * Input:
- * - streamData : in-/output struct containing bitstream
- * - data : data vector
- * - cdf : array of cdf arrays
- * - lenData : data vector length
- *
- * Return value : 0 if ok
- * <0 if error detected
- */
-int WebRtcIsacfix_EncHistMulti(
- Bitstr_enc *streamData,
- const WebRtc_Word16 *data,
- const WebRtc_UWord16 **cdf,
- const WebRtc_Word16 lenData);
-
-
-/****************************************************************************
- * WebRtcIsacfix_DecHistBisectMulti(...)
- *
- * Function to decode more symbols from the arithmetic bytestream, using
- * method of bisection.
- * C df tables should be of size 2^k-1 (which corresponds to an
- * alphabet size of 2^k-2)
- *
- * Input:
- * - streamData : in-/output struct containing bitstream
- * - cdf : array of cdf arrays
- * - cdfSize : array of cdf table sizes+1 (power of two: 2^k)
- * - lenData : data vector length
- *
- * Output:
- * - data : data vector
- *
- * Return value : number of bytes in the stream
- * <0 if error detected
- */
-WebRtc_Word16 WebRtcIsacfix_DecHistBisectMulti(
- WebRtc_Word16 *data,
- Bitstr_dec *streamData,
- const WebRtc_UWord16 **cdf,
- const WebRtc_UWord16 *cdfSize,
- const WebRtc_Word16 lenData);
-
-
-/****************************************************************************
- * WebRtcIsacfix_DecHistOneStepMulti(...)
- *
- * Function to decode more symbols from the arithmetic bytestream, taking
- * single step up or down at a time.
- * cdf tables can be of arbitrary size, but large tables may take a lot of
- * iterations.
- *
- * Input:
- * - streamData : in-/output struct containing bitstream
- * - cdf : array of cdf arrays
- * - initIndex : vector of initial cdf table search entries
- * - lenData : data vector length
- *
- * Output:
- * - data : data vector
- *
- * Return value : number of bytes in original stream
- * <0 if error detected
- */
-WebRtc_Word16 WebRtcIsacfix_DecHistOneStepMulti(
- WebRtc_Word16 *data,
- Bitstr_dec *streamData,
- const WebRtc_UWord16 **cdf,
- const WebRtc_UWord16 *initIndex,
- const WebRtc_Word16 lenData);
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ARITH_ROUTINS_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/bandwidth_estimator.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/bandwidth_estimator.c
deleted file mode 100644
index a274b66..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/bandwidth_estimator.c
+++ /dev/null
@@ -1,1022 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * bandwidth_estimator.c
- *
- * This file contains the code for the Bandwidth Estimator designed
- * for iSAC.
- *
- * NOTE! Castings needed for C55, do not remove!
- *
- */
-
-#include "bandwidth_estimator.h"
-#include "settings.h"
-
-
-/* array of quantization levels for bottle neck info; Matlab code: */
-/* sprintf('%4.1ff, ', logspace(log10(5000), log10(40000), 12)) */
-static const WebRtc_Word16 kQRateTable[12] = {
- 10000, 11115, 12355, 13733, 15265, 16967,
- 18860, 20963, 23301, 25900, 28789, 32000
-};
-
-/* 0.1 times the values in the table kQRateTable */
-/* values are in Q16 */
-static const WebRtc_Word32 KQRate01[12] = {
- 65536000, 72843264, 80969728, 90000589, 100040704, 111194931,
- 123600896, 137383117, 152705434, 169738240, 188671590, 209715200
-};
-
-/* Bits per Bytes Seconds
- * 8 bits/byte * 1000 msec/sec * 1/framelength (in msec)->bits/byte*sec
- * frame length will either be 30 or 60 msec. 8738 is 1/60 in Q19 and 1/30 in Q18
- * The following number is either in Q15 or Q14 depending on the current frame length */
-static const WebRtc_Word32 kBitsByteSec = 4369000;
-
-/* Received header rate. First value is for 30 ms packets and second for 60 ms */
-static const WebRtc_Word16 kRecHeaderRate[2] = {
- 9333, 4666
-};
-
-/* Inverted minimum and maximum bandwidth in Q30.
- minBwInv 30 ms, maxBwInv 30 ms,
- minBwInv 60 ms, maxBwInv 69 ms
-*/
-static const WebRtc_Word32 kInvBandwidth[4] = {
- 55539, 25978,
- 73213, 29284
-};
-
-/* Number of samples in 25 msec */
-static const WebRtc_Word32 kSamplesIn25msec = 400;
-
-
-/****************************************************************************
- * WebRtcIsacfix_InitBandwidthEstimator(...)
- *
- * This function initializes the struct for the bandwidth estimator
- *
- * Input/Output:
- * - bweStr : Struct containing bandwidth information.
- *
- * Return value : 0
- */
-WebRtc_Word32 WebRtcIsacfix_InitBandwidthEstimator(BwEstimatorstr *bweStr)
-{
- bweStr->prevFrameSizeMs = INIT_FRAME_LEN;
- bweStr->prevRtpNumber = 0;
- bweStr->prevSendTime = 0;
- bweStr->prevArrivalTime = 0;
- bweStr->prevRtpRate = 1;
- bweStr->lastUpdate = 0;
- bweStr->lastReduction = 0;
- bweStr->countUpdates = -9;
-
- /* INIT_BN_EST = 20000
- * INIT_BN_EST_Q7 = 2560000
- * INIT_HDR_RATE = 4666
- * INIT_REC_BN_EST_Q5 = 789312
- *
- * recBwInv = 1/(INIT_BN_EST + INIT_HDR_RATE) in Q30
- * recBwAvg = INIT_BN_EST + INIT_HDR_RATE in Q5
- */
- bweStr->recBwInv = 43531;
- bweStr->recBw = INIT_BN_EST;
- bweStr->recBwAvgQ = INIT_BN_EST_Q7;
- bweStr->recBwAvg = INIT_REC_BN_EST_Q5;
- bweStr->recJitter = (WebRtc_Word32) 327680; /* 10 in Q15 */
- bweStr->recJitterShortTerm = 0;
- bweStr->recJitterShortTermAbs = (WebRtc_Word32) 40960; /* 5 in Q13 */
- bweStr->recMaxDelay = (WebRtc_Word32) 10;
- bweStr->recMaxDelayAvgQ = (WebRtc_Word32) 5120; /* 10 in Q9 */
- bweStr->recHeaderRate = INIT_HDR_RATE;
- bweStr->countRecPkts = 0;
- bweStr->sendBwAvg = INIT_BN_EST_Q7;
- bweStr->sendMaxDelayAvg = (WebRtc_Word32) 5120; /* 10 in Q9 */
-
- bweStr->countHighSpeedRec = 0;
- bweStr->highSpeedRec = 0;
- bweStr->countHighSpeedSent = 0;
- bweStr->highSpeedSend = 0;
- bweStr->inWaitPeriod = 0;
-
- /* Find the inverse of the max bw and min bw in Q30
- * (1 / (MAX_ISAC_BW + INIT_HDR_RATE) in Q30
- * (1 / (MIN_ISAC_BW + INIT_HDR_RATE) in Q30
- */
- bweStr->maxBwInv = kInvBandwidth[3];
- bweStr->minBwInv = kInvBandwidth[2];
-
- return 0;
-}
-
-/****************************************************************************
- * WebRtcIsacfix_UpdateUplinkBwImpl(...)
- *
- * This function updates bottle neck rate received from other side in payload
- * and calculates a new bottle neck to send to the other side.
- *
- * Input/Output:
- * - bweStr : struct containing bandwidth information.
- * - rtpNumber : value from RTP packet, from NetEq
- * - frameSize : length of signal frame in ms, from iSAC decoder
- * - sendTime : value in RTP header giving send time in samples
- * - arrivalTime : value given by timeGetTime() time of arrival in
- * samples of packet from NetEq
- * - pksize : size of packet in bytes, from NetEq
- * - Index : integer (range 0...23) indicating bottle neck &
- * jitter as estimated by other side
- *
- * Return value : 0 if everything went fine,
- * -1 otherwise
- */
-WebRtc_Word32 WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bweStr,
- const WebRtc_UWord16 rtpNumber,
- const WebRtc_Word16 frameSize,
- const WebRtc_UWord32 sendTime,
- const WebRtc_UWord32 arrivalTime,
- const WebRtc_Word16 pksize,
- const WebRtc_UWord16 Index)
-{
- WebRtc_UWord16 weight = 0;
- WebRtc_UWord32 currBwInv = 0;
- WebRtc_UWord16 recRtpRate;
- WebRtc_UWord32 arrTimeProj;
- WebRtc_Word32 arrTimeDiff;
- WebRtc_Word32 arrTimeNoise;
- WebRtc_Word32 arrTimeNoiseAbs;
- WebRtc_Word32 sendTimeDiff;
-
- WebRtc_Word32 delayCorrFactor = DELAY_CORRECTION_MED;
- WebRtc_Word32 lateDiff = 0;
- WebRtc_Word16 immediateSet = 0;
- WebRtc_Word32 frameSizeSampl;
-
- WebRtc_Word32 temp;
- WebRtc_Word32 msec;
- WebRtc_UWord32 exponent;
- WebRtc_UWord32 reductionFactor;
- WebRtc_UWord32 numBytesInv;
- WebRtc_Word32 sign;
-
- WebRtc_UWord32 byteSecondsPerBit;
- WebRtc_UWord32 tempLower;
- WebRtc_UWord32 tempUpper;
- WebRtc_Word32 recBwAvgInv;
- WebRtc_Word32 numPktsExpected;
-
- WebRtc_Word16 errCode;
-
- /* UPDATE ESTIMATES FROM OTHER SIDE */
-
- /* The function also checks if Index has a valid value */
- errCode = WebRtcIsacfix_UpdateUplinkBwRec(bweStr, Index);
- if (errCode <0) {
- return(errCode);
- }
-
-
- /* UPDATE ESTIMATES ON THIS SIDE */
-
- /* Bits per second per byte * 1/30 or 1/60 */
- if (frameSize == 60) {
- /* If frameSize changed since last call, from 30 to 60, recalculate some values */
- if ( (frameSize != bweStr->prevFrameSizeMs) && (bweStr->countUpdates > 0)) {
- bweStr->countUpdates = 10;
- bweStr->recHeaderRate = kRecHeaderRate[1];
-
- bweStr->maxBwInv = kInvBandwidth[3];
- bweStr->minBwInv = kInvBandwidth[2];
- bweStr->recBwInv = WEBRTC_SPL_UDIV(1073741824, (bweStr->recBw + bweStr->recHeaderRate));
- }
-
- /* kBitsByteSec is in Q15 */
- recRtpRate = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(kBitsByteSec,
- (WebRtc_Word32)pksize), 15) + bweStr->recHeaderRate;
-
- } else {
- /* If frameSize changed since last call, from 60 to 30, recalculate some values */
- if ( (frameSize != bweStr->prevFrameSizeMs) && (bweStr->countUpdates > 0)) {
- bweStr->countUpdates = 10;
- bweStr->recHeaderRate = kRecHeaderRate[0];
-
- bweStr->maxBwInv = kInvBandwidth[1];
- bweStr->minBwInv = kInvBandwidth[0];
- bweStr->recBwInv = WEBRTC_SPL_UDIV(1073741824, (bweStr->recBw + bweStr->recHeaderRate));
- }
-
- /* kBitsByteSec is in Q14 */
- recRtpRate = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(kBitsByteSec,
- (WebRtc_Word32)pksize), 14) + bweStr->recHeaderRate;
- }
-
-
- /* Check for timer wrap-around */
- if (arrivalTime < bweStr->prevArrivalTime) {
- bweStr->prevArrivalTime = arrivalTime;
- bweStr->lastUpdate = arrivalTime;
- bweStr->lastReduction = arrivalTime + FS3;
-
- bweStr->countRecPkts = 0;
-
- /* store frame size */
- bweStr->prevFrameSizeMs = frameSize;
-
- /* store far-side transmission rate */
- bweStr->prevRtpRate = recRtpRate;
-
- /* store far-side RTP time stamp */
- bweStr->prevRtpNumber = rtpNumber;
-
- return 0;
- }
-
- bweStr->countRecPkts++;
-
- /* Calculate framesize in msec */
- frameSizeSampl = WEBRTC_SPL_MUL_16_16((WebRtc_Word16)SAMPLES_PER_MSEC, frameSize);
-
- /* Check that it's not one of the first 9 packets */
- if ( bweStr->countUpdates > 0 ) {
-
- /* Stay in Wait Period for 1.5 seconds (no updates in wait period) */
- if(bweStr->inWaitPeriod) {
- if ((arrivalTime - bweStr->startWaitPeriod)> FS_1_HALF) {
- bweStr->inWaitPeriod = 0;
- }
- }
-
- /* If not been updated for a long time, reduce the BN estimate */
-
- /* Check send time difference between this packet and previous received */
- sendTimeDiff = sendTime - bweStr->prevSendTime;
- if (sendTimeDiff <= WEBRTC_SPL_LSHIFT_W32(frameSizeSampl, 1)) {
-
- /* Only update if 3 seconds has past since last update */
- if ((arrivalTime - bweStr->lastUpdate) > FS3) {
-
- /* Calculate expected number of received packets since last update */
- numPktsExpected = WEBRTC_SPL_UDIV(arrivalTime - bweStr->lastUpdate, frameSizeSampl);
-
- /* If received number of packets is more than 90% of expected (922 = 0.9 in Q10): */
- /* do the update, else not */
- if(WEBRTC_SPL_LSHIFT_W32(bweStr->countRecPkts, 10) > WEBRTC_SPL_MUL_16_16(922, numPktsExpected)) {
- /* Q4 chosen to approx dividing by 16 */
- msec = (arrivalTime - bweStr->lastReduction);
-
- /* the number below represents 13 seconds, highly unlikely
- but to insure no overflow when reduction factor is multiplied by recBw inverse */
- if (msec > 208000) {
- msec = 208000;
- }
-
- /* Q20 2^(negative number: - 76/1048576) = .99995
- product is Q24 */
- exponent = WEBRTC_SPL_UMUL(0x0000004C, msec);
-
- /* do the approx with positive exponent so that value is actually rf^-1
- and multiply by bw inverse */
- reductionFactor = WEBRTC_SPL_RSHIFT_U32(0x01000000 | (exponent & 0x00FFFFFF),
- WEBRTC_SPL_RSHIFT_U32(exponent, 24));
-
- /* reductionFactor in Q13 */
- reductionFactor = WEBRTC_SPL_RSHIFT_U32(reductionFactor, 11);
-
- if ( reductionFactor != 0 ) {
- bweStr->recBwInv = WEBRTC_SPL_MUL((WebRtc_Word32)bweStr->recBwInv, (WebRtc_Word32)reductionFactor);
- bweStr->recBwInv = WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)bweStr->recBwInv, 13);
-
- } else {
- /* recBwInv = 1 / (INIT_BN_EST + INIT_HDR_RATE) in Q26 (Q30??)*/
- bweStr->recBwInv = WEBRTC_SPL_DIV((1073741824 +
- WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)INIT_BN_EST + INIT_HDR_RATE), 1)), INIT_BN_EST + INIT_HDR_RATE);
- }
-
- /* reset time-since-update counter */
- bweStr->lastReduction = arrivalTime;
- } else {
- /* Delay last reduction with 3 seconds */
- bweStr->lastReduction = arrivalTime + FS3;
- bweStr->lastUpdate = arrivalTime;
- bweStr->countRecPkts = 0;
- }
- }
- } else {
- bweStr->lastReduction = arrivalTime + FS3;
- bweStr->lastUpdate = arrivalTime;
- bweStr->countRecPkts = 0;
- }
-
-
- /* update only if previous packet was not lost */
- if ( rtpNumber == bweStr->prevRtpNumber + 1 ) {
- arrTimeDiff = arrivalTime - bweStr->prevArrivalTime;
-
- if (!(bweStr->highSpeedSend && bweStr->highSpeedRec)) {
- if (arrTimeDiff > frameSizeSampl) {
- if (sendTimeDiff > 0) {
- lateDiff = arrTimeDiff - sendTimeDiff -
- WEBRTC_SPL_LSHIFT_W32(frameSizeSampl, 1);
- } else {
- lateDiff = arrTimeDiff - frameSizeSampl;
- }
-
- /* 8000 is 1/2 second (in samples at FS) */
- if (lateDiff > 8000) {
- delayCorrFactor = (WebRtc_Word32) DELAY_CORRECTION_MAX;
- bweStr->inWaitPeriod = 1;
- bweStr->startWaitPeriod = arrivalTime;
- immediateSet = 1;
- } else if (lateDiff > 5120) {
- delayCorrFactor = (WebRtc_Word32) DELAY_CORRECTION_MED;
- immediateSet = 1;
- bweStr->inWaitPeriod = 1;
- bweStr->startWaitPeriod = arrivalTime;
- }
- }
- }
-
- if ((bweStr->prevRtpRate > WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32) bweStr->recBwAvg, 5)) &&
- (recRtpRate > WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)bweStr->recBwAvg, 5)) &&
- !bweStr->inWaitPeriod) {
-
- /* test if still in initiation period and increment counter */
- if (bweStr->countUpdates++ > 99) {
- /* constant weight after initiation part, 0.01 in Q13 */
- weight = (WebRtc_UWord16) 82;
- } else {
- /* weight decreases with number of updates, 1/countUpdates in Q13 */
- weight = (WebRtc_UWord16) WebRtcSpl_DivW32W16(
- (WebRtc_Word32)(8192 + WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32) bweStr->countUpdates, 1)),
- (WebRtc_Word16)bweStr->countUpdates);
- }
-
- /* Bottle Neck Estimation */
-
- /* limit outliers, if more than 25 ms too much */
- if (arrTimeDiff > frameSizeSampl + kSamplesIn25msec) {
- arrTimeDiff = frameSizeSampl + kSamplesIn25msec;
- }
-
- /* don't allow it to be less than frame rate - 10 ms */
- if (arrTimeDiff < frameSizeSampl - FRAMESAMPLES_10ms) {
- arrTimeDiff = frameSizeSampl - FRAMESAMPLES_10ms;
- }
-
- /* compute inverse receiving rate for last packet, in Q19 */
- numBytesInv = (WebRtc_UWord16) WebRtcSpl_DivW32W16(
- (WebRtc_Word32)(524288 + WEBRTC_SPL_RSHIFT_W32(((WebRtc_Word32)pksize + HEADER_SIZE), 1)),
- (WebRtc_Word16)(pksize + HEADER_SIZE));
-
- /* 8389 is ~ 1/128000 in Q30 */
- byteSecondsPerBit = WEBRTC_SPL_MUL_16_16(arrTimeDiff, 8389);
-
- /* get upper N bits */
- tempUpper = WEBRTC_SPL_RSHIFT_U32(byteSecondsPerBit, 15);
-
- /* get lower 15 bits */
- tempLower = byteSecondsPerBit & 0x00007FFF;
-
- tempUpper = WEBRTC_SPL_MUL(tempUpper, numBytesInv);
- tempLower = WEBRTC_SPL_MUL(tempLower, numBytesInv);
- tempLower = WEBRTC_SPL_RSHIFT_U32(tempLower, 15);
-
- currBwInv = tempUpper + tempLower;
- currBwInv = WEBRTC_SPL_RSHIFT_U32(currBwInv, 4);
-
- /* Limit inv rate. Note that minBwInv > maxBwInv! */
- if(currBwInv < bweStr->maxBwInv) {
- currBwInv = bweStr->maxBwInv;
- } else if(currBwInv > bweStr->minBwInv) {
- currBwInv = bweStr->minBwInv;
- }
-
- /* update bottle neck rate estimate */
- bweStr->recBwInv = WEBRTC_SPL_UMUL(weight, currBwInv) +
- WEBRTC_SPL_UMUL((WebRtc_UWord32) 8192 - weight, bweStr->recBwInv);
-
- /* Shift back to Q30 from Q40 (actual used bits shouldn't be more than 27 based on minBwInv)
- up to 30 bits used with Q13 weight */
- bweStr->recBwInv = WEBRTC_SPL_RSHIFT_U32(bweStr->recBwInv, 13);
-
- /* reset time-since-update counter */
- bweStr->lastUpdate = arrivalTime;
- bweStr->lastReduction = arrivalTime + FS3;
- bweStr->countRecPkts = 0;
-
- /* to save resolution compute the inverse of recBwAvg in Q26 by left shifting numerator to 2^31
- and NOT right shifting recBwAvg 5 bits to an integer
- At max 13 bits are used
- shift to Q5 */
- recBwAvgInv = WEBRTC_SPL_UDIV((WebRtc_UWord32)(0x80000000 + WEBRTC_SPL_RSHIFT_U32(bweStr->recBwAvg, 1)),
- bweStr->recBwAvg);
-
- /* Calculate Projected arrival time difference */
-
- /* The numerator of the quotient can be 22 bits so right shift inv by 4 to avoid overflow
- result in Q22 */
- arrTimeProj = WEBRTC_SPL_MUL((WebRtc_Word32)8000, recBwAvgInv);
- /* shift to Q22 */
- arrTimeProj = WEBRTC_SPL_RSHIFT_U32(arrTimeProj, 4);
- /* complete calulation */
- arrTimeProj = WEBRTC_SPL_MUL(((WebRtc_Word32)pksize + HEADER_SIZE), arrTimeProj);
- /* shift to Q10 */
- arrTimeProj = WEBRTC_SPL_RSHIFT_U32(arrTimeProj, 12);
-
- /* difference between projected and actual arrival time differences */
- /* Q9 (only shift arrTimeDiff by 5 to simulate divide by 16 (need to revisit if change sampling rate) DH */
- if (WEBRTC_SPL_LSHIFT_W32(arrTimeDiff, 6) > (WebRtc_Word32)arrTimeProj) {
- arrTimeNoise = WEBRTC_SPL_LSHIFT_W32(arrTimeDiff, 6) - arrTimeProj;
- sign = 1;
- } else {
- arrTimeNoise = arrTimeProj - WEBRTC_SPL_LSHIFT_W32(arrTimeDiff, 6);
- sign = -1;
- }
-
- /* Q9 */
- arrTimeNoiseAbs = arrTimeNoise;
-
- /* long term averaged absolute jitter, Q15 */
- weight = WEBRTC_SPL_RSHIFT_W32(weight, 3);
- bweStr->recJitter = WEBRTC_SPL_MUL(weight, WEBRTC_SPL_LSHIFT_W32(arrTimeNoiseAbs, 5))
- + WEBRTC_SPL_MUL(1024 - weight, bweStr->recJitter);
-
- /* remove the fractional portion */
- bweStr->recJitter = WEBRTC_SPL_RSHIFT_W32(bweStr->recJitter, 10);
-
- /* Maximum jitter is 10 msec in Q15 */
- if (bweStr->recJitter > (WebRtc_Word32)327680) {
- bweStr->recJitter = (WebRtc_Word32)327680;
- }
-
- /* short term averaged absolute jitter */
- /* Calculation in Q13 products in Q23 */
- bweStr->recJitterShortTermAbs = WEBRTC_SPL_MUL(51, WEBRTC_SPL_LSHIFT_W32(arrTimeNoiseAbs, 3)) +
- WEBRTC_SPL_MUL(973, bweStr->recJitterShortTermAbs);
- bweStr->recJitterShortTermAbs = WEBRTC_SPL_RSHIFT_W32(bweStr->recJitterShortTermAbs , 10);
-
- /* short term averaged jitter */
- /* Calculation in Q13 products in Q23 */
- bweStr->recJitterShortTerm = WEBRTC_SPL_MUL(205, WEBRTC_SPL_LSHIFT_W32(arrTimeNoise, 3)) * sign +
- WEBRTC_SPL_MUL(3891, bweStr->recJitterShortTerm);
-
- if (bweStr->recJitterShortTerm < 0) {
- temp = -bweStr->recJitterShortTerm;
- temp = WEBRTC_SPL_RSHIFT_W32(temp, 12);
- bweStr->recJitterShortTerm = -temp;
- } else {
- bweStr->recJitterShortTerm = WEBRTC_SPL_RSHIFT_W32(bweStr->recJitterShortTerm, 12);
- }
- }
- }
- } else {
- /* reset time-since-update counter when receiving the first 9 packets */
- bweStr->lastUpdate = arrivalTime;
- bweStr->lastReduction = arrivalTime + FS3;
- bweStr->countRecPkts = 0;
- bweStr->countUpdates++;
- }
-
- /* Limit to minimum or maximum bottle neck rate (in Q30) */
- if (bweStr->recBwInv > bweStr->minBwInv) {
- bweStr->recBwInv = bweStr->minBwInv;
- } else if (bweStr->recBwInv < bweStr->maxBwInv) {
- bweStr->recBwInv = bweStr->maxBwInv;
- }
-
-
- /* store frame length */
- bweStr->prevFrameSizeMs = frameSize;
-
- /* store far-side transmission rate */
- bweStr->prevRtpRate = recRtpRate;
-
- /* store far-side RTP time stamp */
- bweStr->prevRtpNumber = rtpNumber;
-
- /* Replace bweStr->recMaxDelay by the new value (atomic operation) */
- if (bweStr->prevArrivalTime != 0xffffffff) {
- bweStr->recMaxDelay = WEBRTC_SPL_MUL(3, bweStr->recJitter);
- }
-
- /* store arrival time stamp */
- bweStr->prevArrivalTime = arrivalTime;
- bweStr->prevSendTime = sendTime;
-
- /* Replace bweStr->recBw by the new value */
- bweStr->recBw = WEBRTC_SPL_UDIV(1073741824, bweStr->recBwInv) - bweStr->recHeaderRate;
-
- if (immediateSet) {
- /* delay correction factor is in Q10 */
- bweStr->recBw = WEBRTC_SPL_UMUL(delayCorrFactor, bweStr->recBw);
- bweStr->recBw = WEBRTC_SPL_RSHIFT_U32(bweStr->recBw, 10);
-
- if (bweStr->recBw < (WebRtc_Word32) MIN_ISAC_BW) {
- bweStr->recBw = (WebRtc_Word32) MIN_ISAC_BW;
- }
-
- bweStr->recBwAvg = WEBRTC_SPL_LSHIFT_U32(bweStr->recBw + bweStr->recHeaderRate, 5);
-
- bweStr->recBwAvgQ = WEBRTC_SPL_LSHIFT_U32(bweStr->recBw, 7);
-
- bweStr->recJitterShortTerm = 0;
-
- bweStr->recBwInv = WEBRTC_SPL_UDIV(1073741824, bweStr->recBw + bweStr->recHeaderRate);
-
- immediateSet = 0;
- }
-
-
- return 0;
-}
-
-/* This function updates the send bottle neck rate */
-/* Index - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
-/* returns 0 if everything went fine, -1 otherwise */
-WebRtc_Word16 WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr *bweStr,
- const WebRtc_Word16 Index)
-{
- WebRtc_UWord16 RateInd;
-
- if ( (Index < 0) || (Index > 23) ) {
- return -ISAC_RANGE_ERROR_BW_ESTIMATOR;
- }
-
- /* UPDATE ESTIMATES FROM OTHER SIDE */
-
- if ( Index > 11 ) {
- RateInd = Index - 12;
- /* compute the jitter estimate as decoded on the other side in Q9 */
- /* sendMaxDelayAvg = 0.9 * sendMaxDelayAvg + 0.1 * MAX_ISAC_MD */
- bweStr->sendMaxDelayAvg = WEBRTC_SPL_MUL(461, bweStr->sendMaxDelayAvg) +
- WEBRTC_SPL_MUL(51, WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)MAX_ISAC_MD, 9));
- bweStr->sendMaxDelayAvg = WEBRTC_SPL_RSHIFT_W32(bweStr->sendMaxDelayAvg, 9);
-
- } else {
- RateInd = Index;
- /* compute the jitter estimate as decoded on the other side in Q9 */
- /* sendMaxDelayAvg = 0.9 * sendMaxDelayAvg + 0.1 * MIN_ISAC_MD */
- bweStr->sendMaxDelayAvg = WEBRTC_SPL_MUL(461, bweStr->sendMaxDelayAvg) +
- WEBRTC_SPL_MUL(51, WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)MIN_ISAC_MD,9));
- bweStr->sendMaxDelayAvg = WEBRTC_SPL_RSHIFT_W32(bweStr->sendMaxDelayAvg, 9);
-
- }
-
-
- /* compute the BN estimate as decoded on the other side */
- /* sendBwAvg = 0.9 * sendBwAvg + 0.1 * kQRateTable[RateInd]; */
- bweStr->sendBwAvg = WEBRTC_SPL_UMUL(461, bweStr->sendBwAvg) +
- WEBRTC_SPL_UMUL(51, WEBRTC_SPL_LSHIFT_U32(kQRateTable[RateInd], 7));
- bweStr->sendBwAvg = WEBRTC_SPL_RSHIFT_U32(bweStr->sendBwAvg, 9);
-
-
- if (WEBRTC_SPL_RSHIFT_U32(bweStr->sendBwAvg, 7) > 28000 && !bweStr->highSpeedSend) {
- bweStr->countHighSpeedSent++;
-
- /* approx 2 seconds with 30ms frames */
- if (bweStr->countHighSpeedSent >= 66) {
- bweStr->highSpeedSend = 1;
- }
- } else if (!bweStr->highSpeedSend) {
- bweStr->countHighSpeedSent = 0;
- }
-
- return 0;
-}
-
-/****************************************************************************
- * WebRtcIsacfix_GetDownlinkBwIndexImpl(...)
- *
- * This function calculates and returns the bandwidth/jitter estimation code
- * (integer 0...23) to put in the sending iSAC payload.
- *
- * Input:
- * - bweStr : BWE struct
- *
- * Return:
- * bandwith and jitter index (0..23)
- */
-WebRtc_UWord16 WebRtcIsacfix_GetDownlinkBwIndexImpl(BwEstimatorstr *bweStr)
-{
- WebRtc_Word32 rate;
- WebRtc_Word32 maxDelay;
- WebRtc_UWord16 rateInd;
- WebRtc_UWord16 maxDelayBit;
- WebRtc_Word32 tempTerm1;
- WebRtc_Word32 tempTerm2;
- WebRtc_Word32 tempTermX;
- WebRtc_Word32 tempTermY;
- WebRtc_Word32 tempMin;
- WebRtc_Word32 tempMax;
-
- /* Get Rate Index */
-
- /* Get unquantized rate. Always returns 10000 <= rate <= 32000 */
- rate = WebRtcIsacfix_GetDownlinkBandwidth(bweStr);
-
- /* Compute the averaged BN estimate on this side */
-
- /* recBwAvg = 0.9 * recBwAvg + 0.1 * (rate + bweStr->recHeaderRate), 0.9 and 0.1 in Q9 */
- bweStr->recBwAvg = WEBRTC_SPL_UMUL(922, bweStr->recBwAvg) +
- WEBRTC_SPL_UMUL(102, WEBRTC_SPL_LSHIFT_U32((WebRtc_UWord32)rate + bweStr->recHeaderRate, 5));
- bweStr->recBwAvg = WEBRTC_SPL_RSHIFT_U32(bweStr->recBwAvg, 10);
-
- /* find quantization index that gives the closest rate after averaging */
- for (rateInd = 1; rateInd < 12; rateInd++) {
- if (rate <= kQRateTable[rateInd]){
- break;
- }
- }
-
- /* find closest quantization index, and update quantized average by taking: */
- /* 0.9*recBwAvgQ + 0.1*kQRateTable[rateInd] */
-
- /* 0.9 times recBwAvgQ in Q16 */
- /* 461/512 - 25/65536 =0.900009 */
- tempTerm1 = WEBRTC_SPL_MUL(bweStr->recBwAvgQ, 25);
- tempTerm1 = WEBRTC_SPL_RSHIFT_W32(tempTerm1, 7);
- tempTermX = WEBRTC_SPL_UMUL(461, bweStr->recBwAvgQ) - tempTerm1;
-
- /* rate in Q16 */
- tempTermY = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)rate, 16);
-
- /* 0.1 * kQRateTable[rateInd] = KQRate01[rateInd] */
- tempTerm1 = tempTermX + KQRate01[rateInd] - tempTermY;
- tempTerm2 = tempTermY - tempTermX - KQRate01[rateInd-1];
-
- /* Compare (0.9 * recBwAvgQ + 0.1 * kQRateTable[rateInd] - rate) >
- (rate - 0.9 * recBwAvgQ - 0.1 * kQRateTable[rateInd-1]) */
- if (tempTerm1 > tempTerm2) {
- rateInd--;
- }
-
- /* Update quantized average by taking: */
- /* 0.9*recBwAvgQ + 0.1*kQRateTable[rateInd] */
-
- /* Add 0.1 times kQRateTable[rateInd], in Q16 */
- tempTermX += KQRate01[rateInd];
-
- /* Shift back to Q7 */
- bweStr->recBwAvgQ = WEBRTC_SPL_RSHIFT_W32(tempTermX, 9);
-
- /* Count consecutive received bandwidth above 28000 kbps (28000 in Q7 = 3584000) */
- /* If 66 high estimates in a row, set highSpeedRec to one */
- /* 66 corresponds to ~2 seconds in 30 msec mode */
- if ((bweStr->recBwAvgQ > 3584000) && !bweStr->highSpeedRec) {
- bweStr->countHighSpeedRec++;
- if (bweStr->countHighSpeedRec >= 66) {
- bweStr->highSpeedRec = 1;
- }
- } else if (!bweStr->highSpeedRec) {
- bweStr->countHighSpeedRec = 0;
- }
-
- /* Get Max Delay Bit */
-
- /* get unquantized max delay */
- maxDelay = WebRtcIsacfix_GetDownlinkMaxDelay(bweStr);
-
- /* Update quantized max delay average */
- tempMax = 652800; /* MAX_ISAC_MD * 0.1 in Q18 */
- tempMin = 130560; /* MIN_ISAC_MD * 0.1 in Q18 */
- tempTermX = WEBRTC_SPL_MUL((WebRtc_Word32)bweStr->recMaxDelayAvgQ, (WebRtc_Word32)461);
- tempTermY = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)maxDelay, 18);
-
- tempTerm1 = tempTermX + tempMax - tempTermY;
- tempTerm2 = tempTermY - tempTermX - tempMin;
-
- if ( tempTerm1 > tempTerm2) {
- maxDelayBit = 0;
- tempTerm1 = tempTermX + tempMin;
-
- /* update quantized average, shift back to Q9 */
- bweStr->recMaxDelayAvgQ = WEBRTC_SPL_RSHIFT_W32(tempTerm1, 9);
- } else {
- maxDelayBit = 12;
- tempTerm1 = tempTermX + tempMax;
-
- /* update quantized average, shift back to Q9 */
- bweStr->recMaxDelayAvgQ = WEBRTC_SPL_RSHIFT_W32(tempTerm1, 9);
- }
-
- /* Return bandwitdh and jitter index (0..23) */
- return (WebRtc_UWord16)(rateInd + maxDelayBit);
-}
-
-/* get the bottle neck rate from far side to here, as estimated on this side */
-WebRtc_UWord16 WebRtcIsacfix_GetDownlinkBandwidth(const BwEstimatorstr *bweStr)
-{
- WebRtc_UWord32 recBw;
- WebRtc_Word32 jitter_sign; /* Q8 */
- WebRtc_Word32 bw_adjust; /* Q16 */
- WebRtc_Word32 rec_jitter_short_term_abs_inv; /* Q18 */
- WebRtc_Word32 temp;
-
- /* Q18 rec jitter short term abs is in Q13, multiply it by 2^13 to save precision
- 2^18 then needs to be shifted 13 bits to 2^31 */
- rec_jitter_short_term_abs_inv = WEBRTC_SPL_UDIV(0x80000000, bweStr->recJitterShortTermAbs);
-
- /* Q27 = 9 + 18 */
- jitter_sign = WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(bweStr->recJitterShortTerm, 4), (WebRtc_Word32)rec_jitter_short_term_abs_inv);
-
- if (jitter_sign < 0) {
- temp = -jitter_sign;
- temp = WEBRTC_SPL_RSHIFT_W32(temp, 19);
- jitter_sign = -temp;
- } else {
- jitter_sign = WEBRTC_SPL_RSHIFT_W32(jitter_sign, 19);
- }
-
- /* adjust bw proportionally to negative average jitter sign */
- //bw_adjust = 1.0f - jitter_sign * (0.15f + 0.15f * jitter_sign * jitter_sign);
- //Q8 -> Q16 .15 +.15 * jitter^2 first term is .15 in Q16 latter term is Q8*Q8*Q8
- //38 in Q8 ~.15 9830 in Q16 ~.15
- temp = 9830 + WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL(38, WEBRTC_SPL_MUL(jitter_sign, jitter_sign))), 8);
-
- if (jitter_sign < 0) {
- temp = WEBRTC_SPL_MUL(jitter_sign, temp);
- temp = -temp;
- temp = WEBRTC_SPL_RSHIFT_W32(temp, 8);
- bw_adjust = (WebRtc_UWord32)65536 + temp; /* (1 << 16) + temp; */
- } else {
- bw_adjust = (WebRtc_UWord32)65536 - WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(jitter_sign, temp), 8);/* (1 << 16) - ((jitter_sign * temp) >> 8); */
- }
-
- //make sure following multiplication won't overflow
- //bw adjust now Q14
- bw_adjust = WEBRTC_SPL_RSHIFT_W32(bw_adjust, 2);//see if good resolution is maintained
-
- /* adjust Rate if jitter sign is mostly constant */
- recBw = WEBRTC_SPL_UMUL(bweStr->recBw, bw_adjust);
-
- recBw = WEBRTC_SPL_RSHIFT_W32(recBw, 14);
-
- /* limit range of bottle neck rate */
- if (recBw < MIN_ISAC_BW) {
- recBw = MIN_ISAC_BW;
- } else if (recBw > MAX_ISAC_BW) {
- recBw = MAX_ISAC_BW;
- }
-
- return (WebRtc_UWord16) recBw;
-}
-
-/* Returns the mmax delay (in ms) */
-WebRtc_Word16 WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr *bweStr)
-{
- WebRtc_Word16 recMaxDelay;
-
- recMaxDelay = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(bweStr->recMaxDelay, 15);
-
- /* limit range of jitter estimate */
- if (recMaxDelay < MIN_ISAC_MD) {
- recMaxDelay = MIN_ISAC_MD;
- } else if (recMaxDelay > MAX_ISAC_MD) {
- recMaxDelay = MAX_ISAC_MD;
- }
-
- return recMaxDelay;
-}
-
-/* get the bottle neck rate from here to far side, as estimated by far side */
-WebRtc_Word16 WebRtcIsacfix_GetUplinkBandwidth(const BwEstimatorstr *bweStr)
-{
- WebRtc_Word16 send_bw;
-
- send_bw = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_U32(bweStr->sendBwAvg, 7);
-
- /* limit range of bottle neck rate */
- if (send_bw < MIN_ISAC_BW) {
- send_bw = MIN_ISAC_BW;
- } else if (send_bw > MAX_ISAC_BW) {
- send_bw = MAX_ISAC_BW;
- }
-
- return send_bw;
-}
-
-
-
-/* Returns the max delay value from the other side in ms */
-WebRtc_Word16 WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr *bweStr)
-{
- WebRtc_Word16 send_max_delay;
-
- send_max_delay = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(bweStr->sendMaxDelayAvg, 9);
-
- /* limit range of jitter estimate */
- if (send_max_delay < MIN_ISAC_MD) {
- send_max_delay = MIN_ISAC_MD;
- } else if (send_max_delay > MAX_ISAC_MD) {
- send_max_delay = MAX_ISAC_MD;
- }
-
- return send_max_delay;
-}
-
-
-
-
-/*
- * update long-term average bitrate and amount of data in buffer
- * returns minimum payload size (bytes)
- */
-WebRtc_UWord16 WebRtcIsacfix_GetMinBytes(RateModel *State,
- WebRtc_Word16 StreamSize, /* bytes in bitstream */
- const WebRtc_Word16 FrameSamples, /* samples per frame */
- const WebRtc_Word16 BottleNeck, /* bottle neck rate; excl headers (bps) */
- const WebRtc_Word16 DelayBuildUp) /* max delay from bottle neck buffering (ms) */
-{
- WebRtc_Word32 MinRate = 0;
- WebRtc_UWord16 MinBytes;
- WebRtc_Word16 TransmissionTime;
- WebRtc_Word32 inv_Q12;
- WebRtc_Word32 den;
-
-
- /* first 10 packets @ low rate, then INIT_BURST_LEN packets @ fixed rate of INIT_RATE bps */
- if (State->InitCounter > 0) {
- if (State->InitCounter-- <= INIT_BURST_LEN) {
- MinRate = INIT_RATE;
- } else {
- MinRate = 0;
- }
- } else {
- /* handle burst */
- if (State->BurstCounter) {
- if (State->StillBuffered < WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL((512 - WEBRTC_SPL_DIV(512, BURST_LEN)), DelayBuildUp), 9)) {
- /* max bps derived from BottleNeck and DelayBuildUp values */
- inv_Q12 = WEBRTC_SPL_DIV(4096, WEBRTC_SPL_MUL(BURST_LEN, FrameSamples));
- MinRate = WEBRTC_SPL_MUL(512 + WEBRTC_SPL_MUL(SAMPLES_PER_MSEC, WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(DelayBuildUp, inv_Q12), 3)), BottleNeck);
- } else {
- /* max bps derived from StillBuffered and DelayBuildUp values */
- inv_Q12 = WEBRTC_SPL_DIV(4096, FrameSamples);
- if (DelayBuildUp > State->StillBuffered) {
- MinRate = WEBRTC_SPL_MUL(512 + WEBRTC_SPL_MUL(SAMPLES_PER_MSEC, WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(DelayBuildUp - State->StillBuffered, inv_Q12), 3)), BottleNeck);
- } else if ((den = WEBRTC_SPL_MUL(SAMPLES_PER_MSEC, (State->StillBuffered - DelayBuildUp))) >= FrameSamples) {
- /* MinRate will be negative here */
- MinRate = 0;
- } else {
- MinRate = WEBRTC_SPL_MUL((512 - WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(den, inv_Q12), 3)), BottleNeck);
- }
- //if (MinRate < 1.04 * BottleNeck)
- // MinRate = 1.04 * BottleNeck;
- //Q9
- if (MinRate < WEBRTC_SPL_MUL(532, BottleNeck)) {
- MinRate += WEBRTC_SPL_MUL(22, BottleNeck);
- }
- }
-
- State->BurstCounter--;
- }
- }
-
-
- /* convert rate from bits/second to bytes/packet */
- //round and shift before conversion
- MinRate += 256;
- MinRate = WEBRTC_SPL_RSHIFT_W32(MinRate, 9);
- MinBytes = (WebRtc_UWord16)WEBRTC_SPL_UDIV(WEBRTC_SPL_MUL(MinRate, FrameSamples), FS8);
-
- /* StreamSize will be adjusted if less than MinBytes */
- if (StreamSize < MinBytes) {
- StreamSize = MinBytes;
- }
-
- /* keep track of when bottle neck was last exceeded by at least 1% */
- //517/512 ~ 1.01
- if (WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(StreamSize, FS8), FrameSamples) > (WEBRTC_SPL_MUL(517, BottleNeck) >> 9)) {
- if (State->PrevExceed) {
- /* bottle_neck exceded twice in a row, decrease ExceedAgo */
- State->ExceedAgo -= WEBRTC_SPL_DIV(BURST_INTERVAL, BURST_LEN - 1);
- if (State->ExceedAgo < 0) {
- State->ExceedAgo = 0;
- }
- } else {
- State->ExceedAgo += (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4); /* ms */
- State->PrevExceed = 1;
- }
- } else {
- State->PrevExceed = 0;
- State->ExceedAgo += (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4); /* ms */
- }
-
- /* set burst flag if bottle neck not exceeded for long time */
- if ((State->ExceedAgo > BURST_INTERVAL) && (State->BurstCounter == 0)) {
- if (State->PrevExceed) {
- State->BurstCounter = BURST_LEN - 1;
- } else {
- State->BurstCounter = BURST_LEN;
- }
- }
-
-
- /* Update buffer delay */
- TransmissionTime = (WebRtc_Word16)WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(StreamSize, 8000), BottleNeck); /* ms */
- State->StillBuffered += TransmissionTime;
- State->StillBuffered -= (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4); //>>4 = SAMPLES_PER_MSEC /* ms */
- if (State->StillBuffered < 0) {
- State->StillBuffered = 0;
- }
-
- if (State->StillBuffered > 2000) {
- State->StillBuffered = 2000;
- }
-
- return MinBytes;
-}
-
-
-/*
- * update long-term average bitrate and amount of data in buffer
- */
-void WebRtcIsacfix_UpdateRateModel(RateModel *State,
- WebRtc_Word16 StreamSize, /* bytes in bitstream */
- const WebRtc_Word16 FrameSamples, /* samples per frame */
- const WebRtc_Word16 BottleNeck) /* bottle neck rate; excl headers (bps) */
-{
- WebRtc_Word16 TransmissionTime;
-
- /* avoid the initial "high-rate" burst */
- State->InitCounter = 0;
-
- /* Update buffer delay */
- TransmissionTime = (WebRtc_Word16)WEBRTC_SPL_DIV(WEBRTC_SPL_MUL(WEBRTC_SPL_MUL(StreamSize, 8), 1000), BottleNeck); /* ms */
- State->StillBuffered += TransmissionTime;
- State->StillBuffered -= (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(FrameSamples, 4); /* ms */
- if (State->StillBuffered < 0) {
- State->StillBuffered = 0;
- }
-
-}
-
-
-void WebRtcIsacfix_InitRateModel(RateModel *State)
-{
- State->PrevExceed = 0; /* boolean */
- State->ExceedAgo = 0; /* ms */
- State->BurstCounter = 0; /* packets */
- State->InitCounter = INIT_BURST_LEN + 10; /* packets */
- State->StillBuffered = 1; /* ms */
-}
-
-
-
-
-
-WebRtc_Word16 WebRtcIsacfix_GetNewFrameLength(WebRtc_Word16 bottle_neck, WebRtc_Word16 current_framesamples)
-{
- WebRtc_Word16 new_framesamples;
-
- new_framesamples = current_framesamples;
-
- /* find new framelength */
- switch(current_framesamples) {
- case 480:
- if (bottle_neck < Thld_30_60) {
- new_framesamples = 960;
- }
- break;
- case 960:
- if (bottle_neck >= Thld_60_30) {
- new_framesamples = 480;
- }
- break;
- default:
- new_framesamples = -1; /* Error */
- }
-
- return new_framesamples;
-}
-
-WebRtc_Word16 WebRtcIsacfix_GetSnr(WebRtc_Word16 bottle_neck, WebRtc_Word16 framesamples)
-{
- WebRtc_Word16 s2nr = 0;
-
- /* find new SNR value */
- //consider BottleNeck to be in Q10 ( * 1 in Q10)
- switch(framesamples) {
- case 480:
- /*s2nr = -1*(a_30 << 10) + ((b_30 * bottle_neck) >> 10);*/
- s2nr = -22500 + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(500, bottle_neck, 10); //* 0.001; //+ c_30 * bottle_neck * bottle_neck * 0.000001;
- break;
- case 960:
- /*s2nr = -1*(a_60 << 10) + ((b_60 * bottle_neck) >> 10);*/
- s2nr = -22500 + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(500, bottle_neck, 10); //* 0.001; //+ c_30 * bottle_neck * bottle_neck * 0.000001;
- break;
- default:
- s2nr = -1; /* Error */
- }
-
- return s2nr; //return in Q10
-
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/bandwidth_estimator.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/bandwidth_estimator.h
deleted file mode 100644
index 76a50f8..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/bandwidth_estimator.h
+++ /dev/null
@@ -1,127 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * bandwidth_estimator.h
- *
- * This header file contains the API for the Bandwidth Estimator
- * designed for iSAC.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_
-
-#include "structs.h"
-
-
-/****************************************************************************
- * WebRtcIsacfix_InitBandwidthEstimator(...)
- *
- * This function initializes the struct for the bandwidth estimator
- *
- * Input/Output:
- * - bwest_str : Struct containing bandwidth information.
- *
- * Return value : 0
- */
-
-WebRtc_Word32 WebRtcIsacfix_InitBandwidthEstimator(BwEstimatorstr *bwest_str);
-
-
-/****************************************************************************
- * WebRtcIsacfix_UpdateUplinkBwImpl(...)
- *
- * This function updates bottle neck rate received from other side in payload
- * and calculates a new bottle neck to send to the other side.
- *
- * Input/Output:
- * - bweStr : struct containing bandwidth information.
- * - rtpNumber : value from RTP packet, from NetEq
- * - frameSize : length of signal frame in ms, from iSAC decoder
- * - sendTime : value in RTP header giving send time in samples
- * - arrivalTime : value given by timeGetTime() time of arrival in
- * samples of packet from NetEq
- * - pksize : size of packet in bytes, from NetEq
- * - Index : integer (range 0...23) indicating bottle neck &
- * jitter as estimated by other side
- *
- * Return value : 0 if everything went fine,
- * -1 otherwise
- */
-
-WebRtc_Word32 WebRtcIsacfix_UpdateUplinkBwImpl(BwEstimatorstr *bwest_str,
- const WebRtc_UWord16 rtp_number,
- const WebRtc_Word16 frameSize,
- const WebRtc_UWord32 send_ts,
- const WebRtc_UWord32 arr_ts,
- const WebRtc_Word16 pksize,
- const WebRtc_UWord16 Index);
-
-/* Update receiving estimates. Used when we only receive BWE index, no iSAC data packet. */
-WebRtc_Word16 WebRtcIsacfix_UpdateUplinkBwRec(BwEstimatorstr *bwest_str,
- const WebRtc_Word16 Index);
-
-/****************************************************************************
- * WebRtcIsacfix_GetDownlinkBwIndexImpl(...)
- *
- * This function calculates and returns the bandwidth/jitter estimation code
- * (integer 0...23) to put in the sending iSAC payload.
- *
- * Input:
- * - bweStr : BWE struct
- *
- * Return:
- * bandwith and jitter index (0..23)
- */
-WebRtc_UWord16 WebRtcIsacfix_GetDownlinkBwIndexImpl(BwEstimatorstr *bwest_str);
-
-/* Returns the bandwidth estimation (in bps) */
-WebRtc_UWord16 WebRtcIsacfix_GetDownlinkBandwidth(const BwEstimatorstr *bwest_str);
-
-/* Returns the bandwidth that iSAC should send with in bps */
-WebRtc_Word16 WebRtcIsacfix_GetUplinkBandwidth(const BwEstimatorstr *bwest_str);
-
-/* Returns the max delay (in ms) */
-WebRtc_Word16 WebRtcIsacfix_GetDownlinkMaxDelay(const BwEstimatorstr *bwest_str);
-
-/* Returns the max delay value from the other side in ms */
-WebRtc_Word16 WebRtcIsacfix_GetUplinkMaxDelay(const BwEstimatorstr *bwest_str);
-
-/*
- * update amount of data in bottle neck buffer and burst handling
- * returns minimum payload size (bytes)
- */
-WebRtc_UWord16 WebRtcIsacfix_GetMinBytes(RateModel *State,
- WebRtc_Word16 StreamSize, /* bytes in bitstream */
- const WebRtc_Word16 FrameLen, /* ms per frame */
- const WebRtc_Word16 BottleNeck, /* bottle neck rate; excl headers (bps) */
- const WebRtc_Word16 DelayBuildUp); /* max delay from bottle neck buffering (ms) */
-
-/*
- * update long-term average bitrate and amount of data in buffer
- */
-void WebRtcIsacfix_UpdateRateModel(RateModel *State,
- WebRtc_Word16 StreamSize, /* bytes in bitstream */
- const WebRtc_Word16 FrameSamples, /* samples per frame */
- const WebRtc_Word16 BottleNeck); /* bottle neck rate; excl headers (bps) */
-
-
-void WebRtcIsacfix_InitRateModel(RateModel *State);
-
-/* Returns the new framelength value (input argument: bottle_neck) */
-WebRtc_Word16 WebRtcIsacfix_GetNewFrameLength(WebRtc_Word16 bottle_neck, WebRtc_Word16 current_framelength);
-
-/* Returns the new SNR value (input argument: bottle_neck) */
-//returns snr in Q10
-WebRtc_Word16 WebRtcIsacfix_GetSnr(WebRtc_Word16 bottle_neck, WebRtc_Word16 framesamples);
-
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_BANDWIDTH_ESTIMATOR_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/codec.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/codec.h
deleted file mode 100644
index 154f2e9..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/codec.h
+++ /dev/null
@@ -1,183 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * codec.h
- *
- * This header file contains the calls to the internal encoder
- * and decoder functions.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_CODEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_CODEC_H_
-
-#include "structs.h"
-
-
-int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr *bwest_str,
- Bitstr_dec *streamdata,
- WebRtc_Word32 packet_size,
- WebRtc_UWord16 rtp_seq_number,
- WebRtc_UWord32 send_ts,
- WebRtc_UWord32 arr_ts);
-
-WebRtc_Word16 WebRtcIsacfix_DecodeImpl(WebRtc_Word16 *signal_out16,
- ISACFIX_DecInst_t *ISACdec_obj,
- WebRtc_Word16 *current_framesamples);
-
-WebRtc_Word16 WebRtcIsacfix_DecodePlcImpl(WebRtc_Word16 *decoded,
- ISACFIX_DecInst_t *ISACdec_obj,
- WebRtc_Word16 *current_framesample );
-
-int WebRtcIsacfix_EncodeImpl(WebRtc_Word16 *in,
- ISACFIX_EncInst_t *ISACenc_obj,
- BwEstimatorstr *bw_estimatordata,
- WebRtc_Word16 CodingMode);
-
-int WebRtcIsacfix_EncodeStoredData(ISACFIX_EncInst_t *ISACenc_obj,
- int BWnumber,
- float scale);
-
-/************************** initialization functions *************************/
-
-void WebRtcIsacfix_InitMaskingEnc(MaskFiltstr_enc *maskdata);
-void WebRtcIsacfix_InitMaskingDec(MaskFiltstr_dec *maskdata);
-
-void WebRtcIsacfix_InitPreFilterbank(PreFiltBankstr *prefiltdata);
-
-void WebRtcIsacfix_InitPostFilterbank(PostFiltBankstr *postfiltdata);
-
-void WebRtcIsacfix_InitPitchFilter(PitchFiltstr *pitchfiltdata);
-
-void WebRtcIsacfix_InitPitchAnalysis(PitchAnalysisStruct *State);
-
-void WebRtcIsacfix_InitPlc( PLCstr *State );
-
-
-/**************************** transform functions ****************************/
-
-void WebRtcIsacfix_InitTransform();
-
-
-void WebRtcIsacfix_Time2Spec(WebRtc_Word16 *inre1Q9,
- WebRtc_Word16 *inre2Q9,
- WebRtc_Word16 *outre,
- WebRtc_Word16 *outim);
-
-
-
-void WebRtcIsacfix_Spec2Time(WebRtc_Word16 *inreQ7,
- WebRtc_Word16 *inimQ7,
- WebRtc_Word32 *outre1Q16,
- WebRtc_Word32 *outre2Q16);
-
-
-
-
-/***************************** filterbank functions **************************/
-
-
-
-void WebRtcIsacfix_SplitAndFilter1(WebRtc_Word16 *in,
- WebRtc_Word16 *LP16,
- WebRtc_Word16 *HP16,
- PreFiltBankstr *prefiltdata);
-
-void WebRtcIsacfix_FilterAndCombine1(WebRtc_Word16 *tempin_ch1,
- WebRtc_Word16 *tempin_ch2,
- WebRtc_Word16 *out16,
- PostFiltBankstr *postfiltdata);
-
-#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-
-void WebRtcIsacfix_SplitAndFilter2(WebRtc_Word16 *in,
- WebRtc_Word16 *LP16,
- WebRtc_Word16 *HP16,
- PreFiltBankstr *prefiltdata);
-
-void WebRtcIsacfix_FilterAndCombine2(WebRtc_Word16 *tempin_ch1,
- WebRtc_Word16 *tempin_ch2,
- WebRtc_Word16 *out16,
- PostFiltBankstr *postfiltdata,
- WebRtc_Word16 len);
-
-#endif
-
-/************************* normalized lattice filters ************************/
-
-
-void WebRtcIsacfix_NormLatticeFilterMa(WebRtc_Word16 orderCoef,
- WebRtc_Word32 *stateGQ15,
- WebRtc_Word16 *lat_inQ0,
- WebRtc_Word16 *filt_coefQ15,
- WebRtc_Word32 *gain_lo_hiQ17,
- WebRtc_Word16 lo_hi,
- WebRtc_Word16 *lat_outQ9);
-
-void WebRtcIsacfix_NormLatticeFilterAr(WebRtc_Word16 orderCoef,
- WebRtc_Word16 *stateGQ0,
- WebRtc_Word32 *lat_inQ25,
- WebRtc_Word16 *filt_coefQ15,
- WebRtc_Word32 *gain_lo_hiQ17,
- WebRtc_Word16 lo_hi,
- WebRtc_Word16 *lat_outQ0);
-
-int WebRtcIsacfix_AutocorrC(WebRtc_Word32* __restrict r,
- const WebRtc_Word16* __restrict x,
- WebRtc_Word16 N,
- WebRtc_Word16 order,
- WebRtc_Word16* __restrict scale);
-
-void WebRtcIsacfix_FilterMaLoopC(int16_t input0,
- int16_t input1,
- int32_t input2,
- int32_t* ptr0,
- int32_t* ptr1,
- int32_t* ptr2);
-
-// Functions for ARM-Neon platforms, in place of the above two generic C ones.
-#if (defined(WEBRTC_ANDROID) && defined(WEBRTC_ARCH_ARM_NEON))
-int WebRtcIsacfix_AutocorrNeon(WebRtc_Word32* __restrict r,
- const WebRtc_Word16* __restrict x,
- WebRtc_Word16 N,
- WebRtc_Word16 order,
- WebRtc_Word16* __restrict scale);
-
-void WebRtcIsacfix_FilterMaLoopNeon(int16_t input0,
- int16_t input1,
- int32_t input2,
- int32_t* ptr0,
- int32_t* ptr1,
- int32_t* ptr2);
-#endif
-
-/**** Function pointers associated with
- **** WebRtcIsacfix_AutocorrC() / WebRtcIsacfix_AutocorrNeon()
- **** and WebRtcIsacfix_FilterMaLoopC() / WebRtcIsacfix_FilterMaLoopNeon().
- ****/
-
-typedef int (*AutocorrFix)(WebRtc_Word32* __restrict r,
- const WebRtc_Word16* __restrict x,
- WebRtc_Word16 N,
- WebRtc_Word16 order,
- WebRtc_Word16* __restrict scale);
-extern AutocorrFix WebRtcIsacfix_AutocorrFix;
-
-typedef void (*FilterMaLoopFix)(int16_t input0,
- int16_t input1,
- int32_t input2,
- int32_t* ptr0,
- int32_t* ptr1,
- int32_t* ptr2);
-extern FilterMaLoopFix WebRtcIsacfix_FilterMaLoopFix;
-
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_CODEC_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode.c
deleted file mode 100644
index 2e15e7a..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode.c
+++ /dev/null
@@ -1,217 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * decode.c
- *
- * This C file contains the internal decoding function.
- *
- */
-
-#include <string.h>
-
-#include "bandwidth_estimator.h"
-#include "codec.h"
-#include "entropy_coding.h"
-#include "pitch_estimator.h"
-#include "settings.h"
-#include "structs.h"
-
-
-
-
-WebRtc_Word16 WebRtcIsacfix_DecodeImpl(WebRtc_Word16 *signal_out16,
- ISACFIX_DecInst_t *ISACdec_obj,
- WebRtc_Word16 *current_framesamples)
-{
- int k;
- int err;
- WebRtc_Word16 BWno;
- WebRtc_Word16 len = 0;
-
- WebRtc_Word16 model;
-
-
- WebRtc_Word16 Vector_Word16_1[FRAMESAMPLES/2];
- WebRtc_Word16 Vector_Word16_2[FRAMESAMPLES/2];
-
- WebRtc_Word32 Vector_Word32_1[FRAMESAMPLES/2];
- WebRtc_Word32 Vector_Word32_2[FRAMESAMPLES/2];
-
- WebRtc_Word16 lofilt_coefQ15[ORDERLO*SUBFRAMES]; //refl. coeffs
- WebRtc_Word16 hifilt_coefQ15[ORDERHI*SUBFRAMES]; //refl. coeffs
- WebRtc_Word32 gain_lo_hiQ17[2*SUBFRAMES];
-
- WebRtc_Word16 PitchLags_Q7[PITCH_SUBFRAMES];
- WebRtc_Word16 PitchGains_Q12[PITCH_SUBFRAMES];
- WebRtc_Word16 AvgPitchGain_Q12;
-
- WebRtc_Word16 tmp_1, tmp_2;
- WebRtc_Word32 tmp32a, tmp32b;
- WebRtc_Word16 gainQ13;
-
-
- WebRtc_Word16 frame_nb; /* counter */
- WebRtc_Word16 frame_mode; /* 0 for 20ms and 30ms, 1 for 60ms */
- WebRtc_Word16 processed_samples;
-
- /* PLC */
- WebRtc_Word16 overlapWin[ 240 ];
-
- (ISACdec_obj->bitstr_obj).W_upper = 0xFFFFFFFF;
- (ISACdec_obj->bitstr_obj).streamval = 0;
- (ISACdec_obj->bitstr_obj).stream_index = 0;
- (ISACdec_obj->bitstr_obj).full = 1;
-
-
- /* decode framelength and BW estimation - not used, only for stream pointer*/
- err = WebRtcIsacfix_DecodeFrameLen(&ISACdec_obj->bitstr_obj, current_framesamples);
- if (err<0) // error check
- return err;
-
- frame_mode = (WebRtc_Word16)WEBRTC_SPL_DIV(*current_framesamples, MAX_FRAMESAMPLES); /* 0, or 1 */
- processed_samples = (WebRtc_Word16)WEBRTC_SPL_DIV(*current_framesamples, frame_mode+1); /* either 320 (20ms) or 480 (30, 60 ms) */
-
- err = WebRtcIsacfix_DecodeSendBandwidth(&ISACdec_obj->bitstr_obj, &BWno);
- if (err<0) // error check
- return err;
-
- /* one loop if it's one frame (20 or 30ms), 2 loops if 2 frames bundled together (60ms) */
- for (frame_nb = 0; frame_nb <= frame_mode; frame_nb++) {
-
- /* decode & dequantize pitch parameters */
- err = WebRtcIsacfix_DecodePitchGain(&(ISACdec_obj->bitstr_obj), PitchGains_Q12);
- if (err<0) // error check
- return err;
-
- err = WebRtcIsacfix_DecodePitchLag(&ISACdec_obj->bitstr_obj, PitchGains_Q12, PitchLags_Q7);
- if (err<0) // error check
- return err;
-
- AvgPitchGain_Q12 = (WebRtc_Word16)(((WebRtc_Word32)PitchGains_Q12[0] + PitchGains_Q12[1] + PitchGains_Q12[2] + PitchGains_Q12[3])>>2);
-
- /* decode & dequantize FiltCoef */
- err = WebRtcIsacfix_DecodeLpc(gain_lo_hiQ17, lofilt_coefQ15, hifilt_coefQ15,
- &ISACdec_obj->bitstr_obj, &model);
-
- if (err<0) // error check
- return err;
-
- /* decode & dequantize spectrum */
- len = WebRtcIsacfix_DecodeSpec(&ISACdec_obj->bitstr_obj, Vector_Word16_1, Vector_Word16_2, AvgPitchGain_Q12);
- if (len < 0) // error check
- return len;
-
- // Why does this need Q16 in and out? /JS
- WebRtcIsacfix_Spec2Time(Vector_Word16_1, Vector_Word16_2, Vector_Word32_1, Vector_Word32_2);
-
- for (k=0; k<FRAMESAMPLES/2; k++) {
- Vector_Word16_1[k] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(Vector_Word32_1[k]+64, 7); //Q16 -> Q9
- }
-
- /* ---- If this is recovery frame ---- */
- if( (ISACdec_obj->plcstr_obj).used == PLC_WAS_USED )
- {
- (ISACdec_obj->plcstr_obj).used = PLC_NOT_USED;
- if( (ISACdec_obj->plcstr_obj).B < 1000 )
- {
- (ISACdec_obj->plcstr_obj).decayCoeffPriodic = 4000;
- }
-
- ISACdec_obj->plcstr_obj.decayCoeffPriodic = WEBRTC_SPL_WORD16_MAX; /* DECAY_RATE is in Q15 */
- ISACdec_obj->plcstr_obj.decayCoeffNoise = WEBRTC_SPL_WORD16_MAX; /* DECAY_RATE is in Q15 */
- ISACdec_obj->plcstr_obj.pitchCycles = 0;
-
- PitchGains_Q12[0] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(PitchGains_Q12[0], 700, 10 );
-
- /* ---- Add-overlap ---- */
- WebRtcSpl_GetHanningWindow( overlapWin, RECOVERY_OVERLAP );
- for( k = 0; k < RECOVERY_OVERLAP; k++ )
- Vector_Word16_1[k] = WEBRTC_SPL_ADD_SAT_W16(
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT( (ISACdec_obj->plcstr_obj).overlapLP[k], overlapWin[RECOVERY_OVERLAP - k - 1], 14),
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT( Vector_Word16_1[k], overlapWin[k], 14) );
-
-
-
- }
-
- /* --- Store side info --- */
- if( frame_nb == frame_mode )
- {
- /* --- LPC info */
- WEBRTC_SPL_MEMCPY_W16( (ISACdec_obj->plcstr_obj).lofilt_coefQ15, &lofilt_coefQ15[(SUBFRAMES-1)*ORDERLO], ORDERLO );
- WEBRTC_SPL_MEMCPY_W16( (ISACdec_obj->plcstr_obj).hifilt_coefQ15, &hifilt_coefQ15[(SUBFRAMES-1)*ORDERHI], ORDERHI );
- (ISACdec_obj->plcstr_obj).gain_lo_hiQ17[0] = gain_lo_hiQ17[(SUBFRAMES-1) * 2];
- (ISACdec_obj->plcstr_obj).gain_lo_hiQ17[1] = gain_lo_hiQ17[(SUBFRAMES-1) * 2 + 1];
-
- /* --- LTP info */
- (ISACdec_obj->plcstr_obj).AvgPitchGain_Q12 = PitchGains_Q12[3];
- (ISACdec_obj->plcstr_obj).lastPitchGain_Q12 = PitchGains_Q12[3];
- (ISACdec_obj->plcstr_obj).lastPitchLag_Q7 = PitchLags_Q7[3];
-
- if( PitchLags_Q7[3] < 3000 )
- (ISACdec_obj->plcstr_obj).lastPitchLag_Q7 += PitchLags_Q7[3];
-
- WEBRTC_SPL_MEMCPY_W16( (ISACdec_obj->plcstr_obj).prevPitchInvIn, Vector_Word16_1, FRAMESAMPLES/2 );
-
- }
- /* ~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~~ */
-
- /* inverse pitch filter */
- WebRtcIsacfix_PitchFilter(Vector_Word16_1, Vector_Word16_2, &ISACdec_obj->pitchfiltstr_obj, PitchLags_Q7, PitchGains_Q12, 4);
-
- if( frame_nb == frame_mode )
- {
- WEBRTC_SPL_MEMCPY_W16( (ISACdec_obj->plcstr_obj).prevPitchInvOut, &(Vector_Word16_2[FRAMESAMPLES/2 - (PITCH_MAX_LAG + 10)]), PITCH_MAX_LAG );
- }
-
-
- /* reduce gain to compensate for pitch enhancer */
- /* gain = 1.0f - 0.45f * AvgPitchGain; */
- tmp32a = WEBRTC_SPL_MUL_16_16_RSFT(AvgPitchGain_Q12, 29, 0); // Q18
- tmp32b = 262144 - tmp32a; // Q18
- gainQ13 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5); // Q13
-
- for (k = 0; k < FRAMESAMPLES/2; k++)
- {
- Vector_Word32_1[k] = (WebRtc_Word32) WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(Vector_Word16_2[k], gainQ13), 3); // Q25
- }
-
-
- /* perceptual post-filtering (using normalized lattice filter) */
- WebRtcIsacfix_NormLatticeFilterAr(ORDERLO, (ISACdec_obj->maskfiltstr_obj).PostStateLoGQ0,
- Vector_Word32_1, lofilt_coefQ15, gain_lo_hiQ17, 0, Vector_Word16_1);
-
- /* --- Store Highpass Residual --- */
- for (k = 0; k < FRAMESAMPLES/2; k++)
- Vector_Word32_1[k] = WEBRTC_SPL_LSHIFT_W32(Vector_Word32_2[k], 9); // Q16 -> Q25
-
- for( k = 0; k < PITCH_MAX_LAG + 10; k++ )
- (ISACdec_obj->plcstr_obj).prevHP[k] = Vector_Word32_1[FRAMESAMPLES/2 - (PITCH_MAX_LAG + 10) + k];
-
-
- WebRtcIsacfix_NormLatticeFilterAr(ORDERHI, (ISACdec_obj->maskfiltstr_obj).PostStateHiGQ0,
- Vector_Word32_1, hifilt_coefQ15, gain_lo_hiQ17, 1, Vector_Word16_2);
-
- /* recombine the 2 bands */
-
- /* Form the polyphase signals, and compensate for DC offset */
- for (k=0;k<FRAMESAMPLES/2;k++) {
- tmp_1 = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(((WebRtc_Word32)Vector_Word16_1[k]+Vector_Word16_2[k] + 1)); /* Construct a new upper channel signal*/
- tmp_2 = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(((WebRtc_Word32)Vector_Word16_1[k]-Vector_Word16_2[k])); /* Construct a new lower channel signal*/
- Vector_Word16_1[k] = tmp_1;
- Vector_Word16_2[k] = tmp_2;
- }
-
- WebRtcIsacfix_FilterAndCombine1(Vector_Word16_1, Vector_Word16_2, signal_out16 + frame_nb * processed_samples, &ISACdec_obj->postfiltbankstr_obj);
-
- }
- return len;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode_bwe.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode_bwe.c
deleted file mode 100644
index 68c6003..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode_bwe.c
+++ /dev/null
@@ -1,69 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * decode_bwe.c
- *
- * This C file contains the internal decode bandwidth estimate function.
- *
- */
-
-
-#include "bandwidth_estimator.h"
-#include "codec.h"
-#include "entropy_coding.h"
-#include "structs.h"
-
-
-
-
-int WebRtcIsacfix_EstimateBandwidth(BwEstimatorstr *bwest_str,
- Bitstr_dec *streamdata,
- WebRtc_Word32 packet_size,
- WebRtc_UWord16 rtp_seq_number,
- WebRtc_UWord32 send_ts,
- WebRtc_UWord32 arr_ts)
-{
- WebRtc_Word16 index;
- WebRtc_Word16 frame_samples;
- int err;
-
- /* decode framelength */
- err = WebRtcIsacfix_DecodeFrameLen(streamdata, &frame_samples);
- /* error check */
- if (err<0) {
- return err;
- }
-
- /* decode BW estimation */
- err = WebRtcIsacfix_DecodeSendBandwidth(streamdata, &index);
- /* error check */
- if (err<0) {
- return err;
- }
-
- /* Update BWE with received data */
- err = WebRtcIsacfix_UpdateUplinkBwImpl(
- bwest_str,
- rtp_seq_number,
- (WebRtc_UWord16)WEBRTC_SPL_UDIV(WEBRTC_SPL_UMUL(frame_samples,1000), FS),
- send_ts,
- arr_ts,
- (WebRtc_Word16) packet_size, /* in bytes */
- index);
-
- /* error check */
- if (err<0) {
- return err;
- }
-
- /* Succesful */
- return 0;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode_plc.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode_plc.c
deleted file mode 100644
index de51658..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/decode_plc.c
+++ /dev/null
@@ -1,830 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * decode_plc.c
- *
- * Packet Loss Concealment.
- *
- */
-
-#include <string.h>
-
-#include "settings.h"
-#include "entropy_coding.h"
-#include "pitch_estimator.h"
-#include "bandwidth_estimator.h"
-#include "structs.h"
-#include "codec.h"
-
-
-#define NO_OF_PRIMES 8
-#define NOISE_FILTER_LEN 30
-
-/*
- * function to decode the bitstream
- * returns the total number of bytes in the stream
- */
-
-static WebRtc_Word16 plc_filterma_Fast(
- WebRtc_Word16 *In, /* (i) Vector to be filtered. InOut[-orderCoef+1]
- to InOut[-1] contains state */
- WebRtc_Word16 *Out, /* (o) Filtered vector */
- WebRtc_Word16 *B, /* (i) The filter coefficients (in Q0) */
- WebRtc_Word16 Blen, /* (i) Number of B coefficients */
- WebRtc_Word16 len, /* (i) Number of samples to be filtered */
- WebRtc_Word16 reduceDecay,
- WebRtc_Word16 decay,
- WebRtc_Word16 rshift )
-{
- int i, j;
- WebRtc_Word32 o;
- WebRtc_Word32 lim;
-
- lim = WEBRTC_SPL_LSHIFT_W32( (WebRtc_Word32)1, 15 + rshift )-1;
-
- for (i = 0; i < len; i++)
- {
- G_CONST WebRtc_Word16 *b_ptr = &B[0];
- G_CONST WebRtc_Word16 *x_ptr = &In[i];
-
- o = (WebRtc_Word32)0;
-
- for (j = 0;j < Blen; j++)
- {
- o = WEBRTC_SPL_ADD_SAT_W32( o, WEBRTC_SPL_MUL_16_16( *b_ptr, *x_ptr) );
- b_ptr++;
- x_ptr--;
- }
-
- /* to round off correctly */
- o = WEBRTC_SPL_ADD_SAT_W32( o, WEBRTC_SPL_LSHIFT_W32( 1, (rshift-1) ) );
-
- /* saturate according to the domain of the filter coefficients */
- o = WEBRTC_SPL_SAT((WebRtc_Word32)lim, o, (WebRtc_Word32)-lim);
-
- /* o should be in the range of WebRtc_Word16 */
- o = WEBRTC_SPL_RSHIFT_W32( o, rshift );
-
- /* decay the output signal; this is specific to plc */
- *Out++ = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT( (WebRtc_Word16)o, decay, 15); // ((o + (WebRtc_Word32)2048) >> 12);
-
- /* change the decay */
- decay -= reduceDecay;
- if( decay < 0 )
- decay = 0;
- }
- return( decay );
-}
-
-
-
-
-
-
-
-
-static __inline WebRtc_Word32 log2_Q8_T( WebRtc_UWord32 x ) {
-
- WebRtc_Word32 zeros, lg2;
- WebRtc_Word16 frac;
-
- zeros=WebRtcSpl_NormU32(x);
- frac=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(((WebRtc_UWord32)WEBRTC_SPL_LSHIFT_W32(x, zeros)&0x7FFFFFFF), 23);
- /* log2(magn(i)) */
-
- lg2= (WEBRTC_SPL_LSHIFT_W16((31-zeros), 8)+frac);
- return lg2;
-
-}
-
-static __inline WebRtc_Word16 exp2_Q10_T(WebRtc_Word16 x) { // Both in and out in Q10
-
- WebRtc_Word16 tmp16_1, tmp16_2;
-
- tmp16_2=(WebRtc_Word16)(0x0400|(x&0x03FF));
- tmp16_1=-(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(x,10);
- if(tmp16_1>0)
- return (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
- else
- return (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
-
-}
-
-
-/*
- This is a fixed-point version of the above code with limLow = 700 and limHigh = 5000,
- hard-coded. The values 700 and 5000 were experimentally obtained.
-
- The function implements membership values for two sets. The mebership functions are
- of second orders corresponding to half-bell-shapped pulses.
-*/
-static void MemshipValQ15( WebRtc_Word16 in, WebRtc_Word16 *A, WebRtc_Word16 *B )
-{
- WebRtc_Word16 x;
-
- in -= 700; /* translate the lowLim to 0, limHigh = 5000 - 700, M = 2150 */
-
- if( in <= 2150 )
- {
- if( in > 0 )
- {
- /* b = in^2 / (2 * M^2), a = 1 - b in Q0.
- We have to compute in Q15 */
-
- /* x = in / 2150 {in Q15} = x * 15.2409 {in Q15} =
- x*15 + (x*983)/(2^12); note that 983/2^12 = 0.23999 */
-
- /* we are sure that x is in the range of WebRtc_Word16 */
- x = (WebRtc_Word16)( WEBRTC_SPL_MUL_16_16( in, 15 ) +
- WEBRTC_SPL_MUL_16_16_RSFT( in, 983, 12) );
- /* b = x^2 / 2 {in Q15} so a shift of 16 is required to
- be in correct domain and one more for the division by 2 */
- *B = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32( WEBRTC_SPL_MUL_16_16( x, x ) + 0x00010000, 17 );
- *A = WEBRTC_SPL_WORD16_MAX - *B;
- }
- else
- {
- *B = 0;
- *A = WEBRTC_SPL_WORD16_MAX;
- }
- }
- else
- {
- if( in < 4300 )
- {
- /* This is a mirror case of the above */
- in = 4300 - in;
- x = (WebRtc_Word16)( WEBRTC_SPL_MUL_16_16( in, 15 ) +
- WEBRTC_SPL_MUL_16_16_RSFT( in, 983, 12) );
- /* b = x^2 / 2 {in Q15} so a shift of 16 is required to
- be in correct domain and one more for the division by 2 */
- *A = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32( WEBRTC_SPL_MUL_16_16( x, x ) + 0x00010000, 17 );
- *B = WEBRTC_SPL_WORD16_MAX - *A;
-
- }
- else
- {
- *A = 0;
- *B = WEBRTC_SPL_WORD16_MAX;
- }
- }
-}
-
-
-
-
-static void LinearResampler( WebRtc_Word16 *in, WebRtc_Word16 *out, WebRtc_Word16 lenIn, WebRtc_Word16 lenOut )
-{
- WebRtc_Word32 n;
- WebRtc_Word16 resOut, i, j, relativePos, diff; /* */
- WebRtc_UWord16 udiff;
-
- if( lenIn == lenOut )
- {
- WEBRTC_SPL_MEMCPY_W16( out, in, lenIn );
- return;
- }
-
- n = WEBRTC_SPL_MUL_16_16( (WebRtc_Word16)(lenIn-1), RESAMP_RES );
- resOut = WebRtcSpl_DivW32W16ResW16( n, (WebRtc_Word16)(lenOut-1) );
-
- out[0] = in[0];
- for( i = 1, j = 0, relativePos = 0; i < lenOut; i++ )
- {
-
- relativePos += resOut;
- while( relativePos > RESAMP_RES )
- {
- j++;
- relativePos -= RESAMP_RES;
- }
-
-
- /* an overflow may happen and the differce in sample values may
- * require more than 16 bits. We like to avoid 32 bit arithmatic
- * as much as possible */
-
- if( (in[ j ] > 0) && (in[j + 1] < 0) )
- {
- udiff = (WebRtc_UWord16)(in[ j ] - in[j + 1]);
- out[ i ] = in[ j ] - (WebRtc_UWord16)( ((WebRtc_Word32)( udiff * relativePos )) >> RESAMP_RES_BIT);
- }
- else
- {
- if( (in[j] < 0) && (in[j+1] > 0) )
- {
- udiff = (WebRtc_UWord16)( in[j + 1] - in[ j ] );
- out[ i ] = in[ j ] + (WebRtc_UWord16)( ((WebRtc_Word32)( udiff * relativePos )) >> RESAMP_RES_BIT);
- }
- else
- {
- diff = in[ j + 1 ] - in[ j ];
- out[ i ] = in[ j ] + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT( diff, relativePos, RESAMP_RES_BIT );
- }
- }
- }
-}
-
-
-
-
-
-WebRtc_Word16 WebRtcIsacfix_DecodePlcImpl(WebRtc_Word16 *signal_out16,
- ISACFIX_DecInst_t *ISACdec_obj,
- WebRtc_Word16 *current_framesamples )
-{
- int subframecnt;
- WebRtc_Word16 len = 0;
-
- WebRtc_Word16* Vector_Word16_1;
- WebRtc_Word16 Vector_Word16_Extended_1[FRAMESAMPLES_HALF + NOISE_FILTER_LEN];
- WebRtc_Word16* Vector_Word16_2;
- WebRtc_Word16 Vector_Word16_Extended_2[FRAMESAMPLES_HALF + NOISE_FILTER_LEN];
-
- WebRtc_Word32 Vector_Word32_1[FRAMESAMPLES_HALF];
- WebRtc_Word32 Vector_Word32_2[FRAMESAMPLES_HALF];
-
- WebRtc_Word16 lofilt_coefQ15[ORDERLO*SUBFRAMES]; //refl. coeffs
- WebRtc_Word16 hifilt_coefQ15[ORDERHI*SUBFRAMES]; //refl. coeffs
-
- WebRtc_Word16 pitchLags_Q7[PITCH_SUBFRAMES];
- WebRtc_Word16 pitchGains_Q12[PITCH_SUBFRAMES];
-
- WebRtc_Word16 tmp_1, tmp_2;
- WebRtc_Word32 tmp32a, tmp32b;
- WebRtc_Word16 gainQ13;
-
- WebRtc_Word16 myDecayRate;
-
- /* ---------- PLC variables ------------ */
- WebRtc_Word16 lag0, i, k, noiseIndex;
- WebRtc_Word16 stretchPitchLP[PITCH_MAX_LAG + 10], stretchPitchLP1[PITCH_MAX_LAG + 10];
-
- WebRtc_Word32 gain_lo_hiQ17[2*SUBFRAMES];
-
- WebRtc_Word16 nLP, pLP, wNoisyLP, wPriodicLP, tmp16, minIdx;
- WebRtc_Word32 nHP, pHP, wNoisyHP, wPriodicHP, corr, minCorr, maxCoeff;
- WebRtc_Word16 noise1, rshift;
-
-
- WebRtc_Word16 ltpGain, pitchGain, myVoiceIndicator, myAbs, maxAbs;
- WebRtc_Word32 varIn, varOut, logVarIn, logVarOut, Q, logMaxAbs;
- int rightShiftIn, rightShiftOut;
-
-
- /* ------------------------------------- */
-
-
- myDecayRate = (DECAY_RATE);
- Vector_Word16_1 = &Vector_Word16_Extended_1[NOISE_FILTER_LEN];
- Vector_Word16_2 = &Vector_Word16_Extended_2[NOISE_FILTER_LEN];
-
-
- /* ----- Simply Copy Previous LPC parameters ------ */
- for( subframecnt = 0; subframecnt < SUBFRAMES; subframecnt++ )
- {
- /* lower Band */
- WEBRTC_SPL_MEMCPY_W16(&lofilt_coefQ15[ subframecnt * ORDERLO ],
- (ISACdec_obj->plcstr_obj).lofilt_coefQ15, ORDERLO);
- gain_lo_hiQ17[2*subframecnt] = (ISACdec_obj->plcstr_obj).gain_lo_hiQ17[0];
-
- /* Upper Band */
- WEBRTC_SPL_MEMCPY_W16(&hifilt_coefQ15[ subframecnt * ORDERHI ],
- (ISACdec_obj->plcstr_obj).hifilt_coefQ15, ORDERHI);
- gain_lo_hiQ17[2*subframecnt + 1] = (ISACdec_obj->plcstr_obj).gain_lo_hiQ17[1];
- }
-
-
-
-
- lag0 = WEBRTC_SPL_RSHIFT_W16(
- (ISACdec_obj->plcstr_obj).lastPitchLag_Q7 + 64, 7 ) + 1;
-
-
- if( (ISACdec_obj->plcstr_obj).used != PLC_WAS_USED )
- {
- (ISACdec_obj->plcstr_obj).pitchCycles = 0;
-
- (ISACdec_obj->plcstr_obj).lastPitchLP =
- &((ISACdec_obj->plcstr_obj).prevPitchInvIn[FRAMESAMPLES_HALF - lag0]);
- minCorr = WEBRTC_SPL_WORD32_MAX;
-
- if ( (FRAMESAMPLES_HALF - 2*lag0 - 10) > 0 )
- {
- minIdx = 11;
- for( i = 0; i < 21; i++ )
- {
- corr = 0;
- for( k = 0; k < lag0; k++ )
- {
- corr = WEBRTC_SPL_ADD_SAT_W32( corr, WEBRTC_SPL_ABS_W32(
- WEBRTC_SPL_SUB_SAT_W16(
- (ISACdec_obj->plcstr_obj).lastPitchLP[k],
- (ISACdec_obj->plcstr_obj).prevPitchInvIn[
- FRAMESAMPLES_HALF - 2*lag0 - 10 + i + k ] ) ) );
- }
- if( corr < minCorr )
- {
- minCorr = corr;
- minIdx = i;
- }
- }
- (ISACdec_obj->plcstr_obj).prevPitchLP =
- &( (ISACdec_obj->plcstr_obj).prevPitchInvIn[
- FRAMESAMPLES_HALF - lag0*2 - 10 + minIdx] );
- }
- else
- {
- (ISACdec_obj->plcstr_obj).prevPitchLP =
- (ISACdec_obj->plcstr_obj).lastPitchLP;
- }
- pitchGain = (ISACdec_obj->plcstr_obj).lastPitchGain_Q12;
-
- WebRtcSpl_AutoCorrelation(
- &(ISACdec_obj->plcstr_obj).prevPitchInvIn[FRAMESAMPLES_HALF - lag0],
- lag0, 0, &varIn, &rightShiftIn);
- WebRtcSpl_AutoCorrelation(
- &(ISACdec_obj->plcstr_obj).prevPitchInvOut[PITCH_MAX_LAG + 10 - lag0],
- lag0, 0, &varOut, &rightShiftOut);
-
- maxAbs = 0;
- for( i = 0; i< lag0; i++)
- {
- myAbs = WEBRTC_SPL_ABS_W16(
- (ISACdec_obj->plcstr_obj).prevPitchInvOut[
- PITCH_MAX_LAG + 10 - lag0 + i] );
- maxAbs = (myAbs > maxAbs)? myAbs:maxAbs;
- }
- logVarIn = log2_Q8_T( (WebRtc_UWord32)( varIn ) ) +
- (WebRtc_Word32)(rightShiftIn << 8);
- logVarOut = log2_Q8_T( (WebRtc_UWord32)( varOut ) ) +
- (WebRtc_Word32)(rightShiftOut << 8);
- logMaxAbs = log2_Q8_T( (WebRtc_UWord32)( maxAbs ) );
-
- ltpGain = (WebRtc_Word16)(logVarOut - logVarIn);
- Q = 2 * logMaxAbs - ( logVarOut - 1512 );
-
- /*
- * ---
- * We are computing sqrt( (VarIn/lag0) / var( noise ) )
- * var( noise ) is almost 256. we have already computed log2( VarIn ) in Q8
- * so we actually compute 2^( 0.5*(log2( VarIn ) - log2( lag0 ) - log2( var(noise ) ) ).
- * Note that put log function is in Q8 but the exponential function is in Q10.
- * --
- */
-
- logVarIn -= log2_Q8_T( (WebRtc_UWord32)( lag0 ) );
- tmp16 = (WebRtc_Word16)((logVarIn<<1) - (4<<10) );
- rightShiftIn = 0;
- if( tmp16 > 4096 )
- {
- tmp16 -= 4096;
- tmp16 = exp2_Q10_T( tmp16 );
- tmp16 >>= 6;
- }
- else
- tmp16 = exp2_Q10_T( tmp16 )>>10;
-
- (ISACdec_obj->plcstr_obj).std = tmp16 - 4;
-
- if( (ltpGain < 110) || (ltpGain > 230) )
- {
- if( ltpGain < 100 && (pitchGain < 1800) )
- {
- (ISACdec_obj->plcstr_obj).A = WEBRTC_SPL_WORD16_MAX;
- }
- else
- {
- (ISACdec_obj->plcstr_obj).A = ((ltpGain < 110) && (Q < 800)
- )? WEBRTC_SPL_WORD16_MAX:0;
- }
- (ISACdec_obj->plcstr_obj).B = WEBRTC_SPL_WORD16_MAX -
- (ISACdec_obj->plcstr_obj).A;
- }
- else
- {
- if( (pitchGain < 450) || (pitchGain > 1600) )
- {
- (ISACdec_obj->plcstr_obj).A = ((pitchGain < 450)
- )? WEBRTC_SPL_WORD16_MAX:0;
- (ISACdec_obj->plcstr_obj).B = WEBRTC_SPL_WORD16_MAX -
- (ISACdec_obj->plcstr_obj).A;
- }
- else
- {
- myVoiceIndicator = ltpGain * 2 + pitchGain;
- MemshipValQ15( myVoiceIndicator,
- &(ISACdec_obj->plcstr_obj).A, &(ISACdec_obj->plcstr_obj).B );
- }
- }
-
-
-
- myVoiceIndicator = ltpGain * 16 + pitchGain * 2 + (pitchGain >> 8);
- MemshipValQ15( myVoiceIndicator,
- &(ISACdec_obj->plcstr_obj).A, &(ISACdec_obj->plcstr_obj).B );
-
-
-
- (ISACdec_obj->plcstr_obj).stretchLag = lag0;
- (ISACdec_obj->plcstr_obj).pitchIndex = 0;
-
- }
- else
- {
- myDecayRate = (DECAY_RATE<<2);
- }
-
- if( (ISACdec_obj->plcstr_obj).B < 1000 )
- {
- myDecayRate += (DECAY_RATE<<3);
- }
-
- /* ------------ reconstructing the residual signal ------------------ */
-
- LinearResampler( (ISACdec_obj->plcstr_obj).lastPitchLP,
- stretchPitchLP, lag0, (ISACdec_obj->plcstr_obj).stretchLag );
- /* inverse pitch filter */
-
- pitchLags_Q7[0] = pitchLags_Q7[1] = pitchLags_Q7[2] = pitchLags_Q7[3] =
- ((ISACdec_obj->plcstr_obj).stretchLag<<7);
- pitchGains_Q12[3] = ( (ISACdec_obj->plcstr_obj).lastPitchGain_Q12);
- pitchGains_Q12[2] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
- pitchGains_Q12[3], 1010, 10 );
- pitchGains_Q12[1] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
- pitchGains_Q12[2], 1010, 10 );
- pitchGains_Q12[0] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
- pitchGains_Q12[1], 1010, 10 );
-
-
- /* most of the time either B or A are zero so seperating */
- if( (ISACdec_obj->plcstr_obj).B == 0 )
- {
- for( i = 0; i < FRAMESAMPLES_HALF; i++ )
- {
- /* --- Low Pass */
- (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND(
- (ISACdec_obj->plcstr_obj).seed );
- Vector_Word16_1[i] = WEBRTC_SPL_RSHIFT_W16(
- (ISACdec_obj->plcstr_obj).seed, 10 ) - 16;
-
- /* --- Highpass */
- (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND(
- (ISACdec_obj->plcstr_obj).seed );
- Vector_Word16_2[i] = WEBRTC_SPL_RSHIFT_W16(
- (ISACdec_obj->plcstr_obj).seed, 10 ) - 16;
-
- }
- for( i = 1; i < NOISE_FILTER_LEN; i++ )
- {
- (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND(
- (ISACdec_obj->plcstr_obj).seed );
- Vector_Word16_Extended_1[ i ] = WEBRTC_SPL_RSHIFT_W16(
- (ISACdec_obj->plcstr_obj).seed, 10 ) - 16;
-
- (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND(
- (ISACdec_obj->plcstr_obj).seed );
- Vector_Word16_Extended_2[ i ] = WEBRTC_SPL_RSHIFT_W16(
- (ISACdec_obj->plcstr_obj).seed, 10 ) - 16;
- }
- plc_filterma_Fast(Vector_Word16_1, Vector_Word16_Extended_1,
- &(ISACdec_obj->plcstr_obj).prevPitchInvIn[FRAMESAMPLES_HALF -
- NOISE_FILTER_LEN], (WebRtc_Word16) NOISE_FILTER_LEN,
- (WebRtc_Word16) FRAMESAMPLES_HALF, (WebRtc_Word16)(5),
- (ISACdec_obj->plcstr_obj).decayCoeffNoise, (WebRtc_Word16)(6));
-
- maxCoeff = WebRtcSpl_MaxAbsValueW32(
- &(ISACdec_obj->plcstr_obj).prevHP[
- PITCH_MAX_LAG + 10 - NOISE_FILTER_LEN], NOISE_FILTER_LEN );
-
- rshift = 0;
- while( maxCoeff > WEBRTC_SPL_WORD16_MAX )
- {
- maxCoeff = WEBRTC_SPL_RSHIFT_W32(maxCoeff, 1);
- rshift++;
- }
- for( i = 0; i < NOISE_FILTER_LEN; i++ ) {
- Vector_Word16_1[ FRAMESAMPLES_HALF - NOISE_FILTER_LEN + i] =
- (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
- (ISACdec_obj->plcstr_obj).prevHP[
- PITCH_MAX_LAG + 10 - NOISE_FILTER_LEN + i], rshift);
- }
- (ISACdec_obj->plcstr_obj).decayCoeffNoise = plc_filterma_Fast(
- Vector_Word16_2,
- Vector_Word16_Extended_2,
- &Vector_Word16_1[FRAMESAMPLES_HALF - NOISE_FILTER_LEN],
- (WebRtc_Word16) NOISE_FILTER_LEN,
- (WebRtc_Word16) FRAMESAMPLES_HALF,
- (WebRtc_Word16) (5),
- (ISACdec_obj->plcstr_obj).decayCoeffNoise,
- (WebRtc_Word16) (7) );
-
- for( i = 0; i < FRAMESAMPLES_HALF; i++ )
- Vector_Word32_2[i] = WEBRTC_SPL_LSHIFT_W32(
- (WebRtc_Word32)Vector_Word16_Extended_2[i], rshift );
-
- Vector_Word16_1 = Vector_Word16_Extended_1;
- }
- else
- {
- if( (ISACdec_obj->plcstr_obj).A == 0 )
- {
- /* ------ Periodic Vector --- */
- for( i = 0, noiseIndex = 0; i < FRAMESAMPLES_HALF; i++, noiseIndex++ )
- {
- /* --- Lowpass */
- pLP = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
- stretchPitchLP[(ISACdec_obj->plcstr_obj).pitchIndex],
- (ISACdec_obj->plcstr_obj).decayCoeffPriodic, 15 );
-
- /* --- Highpass */
- pHP = (WebRtc_Word32)WEBRTC_SPL_MUL_16_32_RSFT15(
- (ISACdec_obj->plcstr_obj).decayCoeffPriodic,
- (ISACdec_obj->plcstr_obj).prevHP[PITCH_MAX_LAG + 10 -
- (ISACdec_obj->plcstr_obj).stretchLag +
- (ISACdec_obj->plcstr_obj).pitchIndex] );
-
- /* --- lower the muliplier (more decay at next sample) --- */
- (ISACdec_obj->plcstr_obj).decayCoeffPriodic -= (myDecayRate);
- if( (ISACdec_obj->plcstr_obj).decayCoeffPriodic < 0 )
- (ISACdec_obj->plcstr_obj).decayCoeffPriodic = 0;
-
- (ISACdec_obj->plcstr_obj).pitchIndex++;
-
- if( (ISACdec_obj->plcstr_obj).pitchIndex ==
- (ISACdec_obj->plcstr_obj).stretchLag )
- {
- (ISACdec_obj->plcstr_obj).pitchIndex = 0;
- (ISACdec_obj->plcstr_obj).pitchCycles++;
-
- if( (ISACdec_obj->plcstr_obj).stretchLag != (lag0 + 1) )
- {
- (ISACdec_obj->plcstr_obj).stretchLag = lag0 + 1;
- }
- else
- {
- (ISACdec_obj->plcstr_obj).stretchLag = lag0;
- }
-
- (ISACdec_obj->plcstr_obj).stretchLag = (
- (ISACdec_obj->plcstr_obj).stretchLag > PITCH_MAX_LAG
- )? (PITCH_MAX_LAG):(ISACdec_obj->plcstr_obj).stretchLag;
-
- LinearResampler( (ISACdec_obj->plcstr_obj).lastPitchLP,
- stretchPitchLP, lag0, (ISACdec_obj->plcstr_obj).stretchLag );
-
- LinearResampler( (ISACdec_obj->plcstr_obj).prevPitchLP,
- stretchPitchLP1, lag0, (ISACdec_obj->plcstr_obj).stretchLag );
-
- switch( (ISACdec_obj->plcstr_obj).pitchCycles )
- {
- case 1:
- {
- for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
- {
- stretchPitchLP[k] = (WebRtc_Word16)((
- (WebRtc_Word32)stretchPitchLP[k]* 3 +
- (WebRtc_Word32)stretchPitchLP1[k])>>2);
- }
- break;
- }
- case 2:
- {
- for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
- {
- stretchPitchLP[k] = (WebRtc_Word16)((
- (WebRtc_Word32)stretchPitchLP[k] +
- (WebRtc_Word32)stretchPitchLP1[k] )>>1);
- }
- break;
- }
- case 3:
- {
- for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
- {
- stretchPitchLP[k] = (WebRtc_Word16)((stretchPitchLP[k] +
- (WebRtc_Word32)stretchPitchLP1[k]*3 )>>2);
- }
- break;
- }
- }
-
- if( (ISACdec_obj->plcstr_obj).pitchCycles == 3 )
- {
- myDecayRate += 35; //(myDecayRate>>1);
- (ISACdec_obj->plcstr_obj).pitchCycles = 0;
- }
-
- }
-
- /* ------ Sum the noisy and periodic signals ------ */
- Vector_Word16_1[i] = pLP;
- Vector_Word32_2[i] = pHP;
- }
- }
- else
- {
- for( i = 0, noiseIndex = 0; i < FRAMESAMPLES_HALF; i++, noiseIndex++ )
- {
-
- (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND(
- (ISACdec_obj->plcstr_obj).seed );
-
- noise1 = WEBRTC_SPL_RSHIFT_W16(
- (ISACdec_obj->plcstr_obj).seed, 10 ) - 16;
-
- nLP = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
- (WebRtc_Word16)((noise1)*(ISACdec_obj->plcstr_obj).std),
- (ISACdec_obj->plcstr_obj).decayCoeffNoise, 15 );
-
- /* --- Highpass */
- (ISACdec_obj->plcstr_obj).seed = WEBRTC_SPL_RAND(
- (ISACdec_obj->plcstr_obj).seed );
- noise1 = WEBRTC_SPL_RSHIFT_W16(
- (ISACdec_obj->plcstr_obj).seed, 11 ) - 8;
-
- nHP = (WebRtc_Word32)WEBRTC_SPL_MUL_16_32_RSFT15(
- (ISACdec_obj->plcstr_obj).decayCoeffNoise,
- (WebRtc_Word32)(noise1*(ISACdec_obj->plcstr_obj).std) );
-
- /* --- lower the muliplier (more decay at next sample) --- */
- (ISACdec_obj->plcstr_obj).decayCoeffNoise -= (myDecayRate);
- if( (ISACdec_obj->plcstr_obj).decayCoeffNoise < 0 )
- (ISACdec_obj->plcstr_obj).decayCoeffNoise = 0;
-
- /* ------ Periodic Vector --- */
- /* --- Lowpass */
- pLP = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
- stretchPitchLP[(ISACdec_obj->plcstr_obj).pitchIndex],
- (ISACdec_obj->plcstr_obj).decayCoeffPriodic, 15 );
-
- /* --- Highpass */
- pHP = (WebRtc_Word32)WEBRTC_SPL_MUL_16_32_RSFT15(
- (ISACdec_obj->plcstr_obj).decayCoeffPriodic,
- (ISACdec_obj->plcstr_obj).prevHP[PITCH_MAX_LAG + 10 -
- (ISACdec_obj->plcstr_obj).stretchLag +
- (ISACdec_obj->plcstr_obj).pitchIndex] );
-
- /* --- lower the muliplier (more decay at next sample) --- */
- (ISACdec_obj->plcstr_obj).decayCoeffPriodic -= (myDecayRate);
- if( (ISACdec_obj->plcstr_obj).decayCoeffPriodic < 0 )
- {
- (ISACdec_obj->plcstr_obj).decayCoeffPriodic = 0;
- }
-
- /* ------ Weighting the noisy and periodic vectors ------- */
- wNoisyLP = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT(
- (ISACdec_obj->plcstr_obj).A, nLP, 15 ) );
- wNoisyHP = (WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT15(
- (ISACdec_obj->plcstr_obj).A, (nHP) ) );
-
- wPriodicLP = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT(
- (ISACdec_obj->plcstr_obj).B, pLP, 15));
- wPriodicHP = (WebRtc_Word32)(WEBRTC_SPL_MUL_16_32_RSFT15(
- (ISACdec_obj->plcstr_obj).B, pHP));
-
- (ISACdec_obj->plcstr_obj).pitchIndex++;
-
- if((ISACdec_obj->plcstr_obj).pitchIndex ==
- (ISACdec_obj->plcstr_obj).stretchLag)
- {
- (ISACdec_obj->plcstr_obj).pitchIndex = 0;
- (ISACdec_obj->plcstr_obj).pitchCycles++;
-
- if( (ISACdec_obj->plcstr_obj).stretchLag != (lag0 + 1) )
- (ISACdec_obj->plcstr_obj).stretchLag = lag0 + 1;
- else
- (ISACdec_obj->plcstr_obj).stretchLag = lag0;
-
- (ISACdec_obj->plcstr_obj).stretchLag = (
- (ISACdec_obj->plcstr_obj).stretchLag > PITCH_MAX_LAG
- )? (PITCH_MAX_LAG):(ISACdec_obj->plcstr_obj).stretchLag;
- LinearResampler(
- (ISACdec_obj->plcstr_obj).lastPitchLP,
- stretchPitchLP, lag0, (ISACdec_obj->plcstr_obj).stretchLag );
-
- LinearResampler((ISACdec_obj->plcstr_obj).prevPitchLP,
- stretchPitchLP1, lag0, (ISACdec_obj->plcstr_obj).stretchLag );
-
- switch((ISACdec_obj->plcstr_obj).pitchCycles)
- {
- case 1:
- {
- for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
- {
- stretchPitchLP[k] = (WebRtc_Word16)((
- (WebRtc_Word32)stretchPitchLP[k]* 3 +
- (WebRtc_Word32)stretchPitchLP1[k] )>>2);
- }
- break;
- }
- case 2:
- {
- for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
- {
- stretchPitchLP[k] = (WebRtc_Word16)((
- (WebRtc_Word32)stretchPitchLP[k] +
- (WebRtc_Word32)stretchPitchLP1[k])>>1);
- }
- break;
- }
- case 3:
- {
- for( k=0; k<(ISACdec_obj->plcstr_obj).stretchLag; k++ )
- {
- stretchPitchLP[k] = (WebRtc_Word16)(
- (stretchPitchLP[k] +
- (WebRtc_Word32)stretchPitchLP1[k]*3 )>>2);
- }
- break;
- }
- }
-
- if( (ISACdec_obj->plcstr_obj).pitchCycles == 3 )
- {
- myDecayRate += 55; //(myDecayRate>>1);
- (ISACdec_obj->plcstr_obj).pitchCycles = 0;
- }
- }
-
- /* ------ Sum the noisy and periodic signals ------ */
- Vector_Word16_1[i] = (WebRtc_Word16)WEBRTC_SPL_ADD_SAT_W16(
- wNoisyLP, wPriodicLP );
- Vector_Word32_2[i] = (WebRtc_Word32)WEBRTC_SPL_ADD_SAT_W32(
- wNoisyHP, wPriodicHP );
- }
- }
- }
- /* ----------------- residual signal is reconstructed ------------------ */
-
- k = (ISACdec_obj->plcstr_obj).pitchIndex;
- /* --- Write one pitch cycle for recovery block --- */
-
- for( i = 0; i < RECOVERY_OVERLAP; i++ )
- {
- (ISACdec_obj->plcstr_obj).overlapLP[i] = (WebRtc_Word16)(
- WEBRTC_SPL_MUL_16_16_RSFT(stretchPitchLP[k],
- (ISACdec_obj->plcstr_obj).decayCoeffPriodic, 15) );
- k = ( k < ((ISACdec_obj->plcstr_obj).stretchLag - 1) )? (k+1):0;
- }
-
- (ISACdec_obj->plcstr_obj).lastPitchLag_Q7 = (ISACdec_obj->plcstr_obj).stretchLag << 7;
-
-
- /* --- Inverse Pitch Filter --- */
- WebRtcIsacfix_PitchFilter(Vector_Word16_1, Vector_Word16_2,
- &ISACdec_obj->pitchfiltstr_obj, pitchLags_Q7, pitchGains_Q12, 4);
-
- /* reduce gain to compensate for pitch enhancer */
- /* gain = 1.0f - 0.45f * AvgPitchGain; */
- tmp32a = WEBRTC_SPL_MUL_16_16_RSFT((ISACdec_obj->plcstr_obj).AvgPitchGain_Q12,
- 29, 0); // Q18
- tmp32b = 262144 - tmp32a; // Q18
- gainQ13 = (WebRtc_Word16) (tmp32b >> 5); // Q13
-
- /* perceptual post-filtering (using normalized lattice filter) */
- for (k = 0; k < FRAMESAMPLES_HALF; k++)
- Vector_Word32_1[k] = (WebRtc_Word32) WEBRTC_SPL_MUL_16_16(
- Vector_Word16_2[k], gainQ13) << 3; // Q25
-
-
- WebRtcIsacfix_NormLatticeFilterAr(ORDERLO,
- (ISACdec_obj->maskfiltstr_obj).PostStateLoGQ0,
- Vector_Word32_1, lofilt_coefQ15, gain_lo_hiQ17, 0, Vector_Word16_1);
-
- WebRtcIsacfix_NormLatticeFilterAr(ORDERHI,
- (ISACdec_obj->maskfiltstr_obj).PostStateHiGQ0,
- Vector_Word32_2, hifilt_coefQ15, gain_lo_hiQ17, 1, Vector_Word16_2);
-
- /* recombine the 2 bands */
-
- /* Form the polyphase signals, and compensate for DC offset */
- for (k=0;k<FRAMESAMPLES_HALF;k++)
- {
- /* Construct a new upper channel signal*/
- tmp_1 = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(
- ((WebRtc_Word32)Vector_Word16_1[k]+Vector_Word16_2[k] + 1));
- /* Construct a new lower channel signal*/
- tmp_2 = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(
- ((WebRtc_Word32)Vector_Word16_1[k]-Vector_Word16_2[k]));
- Vector_Word16_1[k] = tmp_1;
- Vector_Word16_2[k] = tmp_2;
- }
-
-
- WebRtcIsacfix_FilterAndCombine1(Vector_Word16_1,
- Vector_Word16_2, signal_out16, &ISACdec_obj->postfiltbankstr_obj);
-
- (ISACdec_obj->plcstr_obj).used = PLC_WAS_USED;
- *current_framesamples = 480;
-
- return len;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/encode.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/encode.c
deleted file mode 100644
index cb531e5..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/encode.c
+++ /dev/null
@@ -1,626 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * encode.c
- *
- * Encoding function for the iSAC coder.
- *
- */
-
-#include "arith_routins.h"
-#include "bandwidth_estimator.h"
-#include "codec.h"
-#include "pitch_gain_tables.h"
-#include "pitch_lag_tables.h"
-#include "entropy_coding.h"
-#include "lpc_tables.h"
-#include "lpc_masking_model.h"
-#include "pitch_estimator.h"
-#include "structs.h"
-#include <stdio.h>
-
-
-int WebRtcIsacfix_EncodeImpl(WebRtc_Word16 *in,
- ISACFIX_EncInst_t *ISACenc_obj,
- BwEstimatorstr *bw_estimatordata,
- WebRtc_Word16 CodingMode)
-{
- WebRtc_Word16 stream_length = 0;
- WebRtc_Word16 usefulstr_len = 0;
- int k;
- WebRtc_Word16 BWno;
-
- WebRtc_Word16 lofilt_coefQ15[(ORDERLO)*SUBFRAMES];
- WebRtc_Word16 hifilt_coefQ15[(ORDERHI)*SUBFRAMES];
- WebRtc_Word32 gain_lo_hiQ17[2*SUBFRAMES];
-
- WebRtc_Word16 LPandHP[FRAMESAMPLES/2 + QLOOKAHEAD];
- WebRtc_Word16 LP16a[FRAMESAMPLES/2 + QLOOKAHEAD];
- WebRtc_Word16 HP16a[FRAMESAMPLES/2 + QLOOKAHEAD];
-
- WebRtc_Word16 PitchLags_Q7[PITCH_SUBFRAMES];
- WebRtc_Word16 PitchGains_Q12[PITCH_SUBFRAMES];
- WebRtc_Word16 AvgPitchGain_Q12;
-
- WebRtc_Word16 frame_mode; /* 0 for 30ms, 1 for 60ms */
- WebRtc_Word16 processed_samples;
- int status;
-
- WebRtc_Word32 bits_gainsQ11;
- WebRtc_Word16 MinBytes;
- WebRtc_Word16 bmodel;
-
- transcode_obj transcodingParam;
- WebRtc_Word16 payloadLimitBytes;
- WebRtc_Word16 arithLenBeforeEncodingDFT;
- WebRtc_Word16 iterCntr;
-
- /* copy new frame length and bottle neck rate only for the first 10 ms data */
- if (ISACenc_obj->buffer_index == 0) {
- /* set the framelength for the next packet */
- ISACenc_obj->current_framesamples = ISACenc_obj->new_framelength;
- }
-
- frame_mode = ISACenc_obj->current_framesamples/MAX_FRAMESAMPLES; /* 0 (30 ms) or 1 (60 ms) */
- processed_samples = ISACenc_obj->current_framesamples/(frame_mode+1); /* 480 (30, 60 ms) */
-
- /* buffer speech samples (by 10ms packet) until the framelength is reached (30 or 60 ms) */
- /**************************************************************************************/
- /* fill the buffer with 10ms input data */
- for(k=0; k<FRAMESAMPLES_10ms; k++) {
- ISACenc_obj->data_buffer_fix[k + ISACenc_obj->buffer_index] = in[k];
- }
- /* if buffersize is not equal to current framesize, and end of file is not reached yet, */
- /* increase index and go back to main to get more speech samples */
- if (ISACenc_obj->buffer_index + FRAMESAMPLES_10ms != processed_samples) {
- ISACenc_obj->buffer_index = ISACenc_obj->buffer_index + FRAMESAMPLES_10ms;
- return 0;
- }
- /* if buffer reached the right size, reset index and continue with encoding the frame */
- ISACenc_obj->buffer_index = 0;
-
- /* end of buffer function */
- /**************************/
-
- /* encoding */
- /************/
-
- if (frame_mode == 0 || ISACenc_obj->frame_nb == 0 )
- {
- /* reset bitstream */
- ISACenc_obj->bitstr_obj.W_upper = 0xFFFFFFFF;
- ISACenc_obj->bitstr_obj.streamval = 0;
- ISACenc_obj->bitstr_obj.stream_index = 0;
- ISACenc_obj->bitstr_obj.full = 1;
-
- if (CodingMode == 0) {
- ISACenc_obj->BottleNeck = WebRtcIsacfix_GetUplinkBandwidth(bw_estimatordata);
- ISACenc_obj->MaxDelay = WebRtcIsacfix_GetUplinkMaxDelay(bw_estimatordata);
- }
- if (CodingMode == 0 && frame_mode == 0 && (ISACenc_obj->enforceFrameSize == 0)) {
- ISACenc_obj->new_framelength = WebRtcIsacfix_GetNewFrameLength(ISACenc_obj->BottleNeck,
- ISACenc_obj->current_framesamples);
- }
-
- // multiply the bottleneck by 0.88 before computing SNR, 0.88 is tuned by experimenting on TIMIT
- // 901/1024 is 0.87988281250000
- ISACenc_obj->s2nr = WebRtcIsacfix_GetSnr((WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ISACenc_obj->BottleNeck, 901, 10),
- ISACenc_obj->current_framesamples);
-
- /* encode frame length */
- status = WebRtcIsacfix_EncodeFrameLen(ISACenc_obj->current_framesamples, &ISACenc_obj->bitstr_obj);
- if (status < 0)
- {
- /* Wrong frame size */
- if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
- {
- // If this is the second 30ms of a 60ms frame reset this such that in the next call
- // encoder starts fresh.
- ISACenc_obj->frame_nb = 0;
- }
- return status;
- }
-
- /* Save framelength for multiple packets memory */
- if (ISACenc_obj->SaveEnc_ptr != NULL) {
- (ISACenc_obj->SaveEnc_ptr)->framelength=ISACenc_obj->current_framesamples;
- }
-
- /* bandwidth estimation and coding */
- BWno = WebRtcIsacfix_GetDownlinkBwIndexImpl(bw_estimatordata);
- status = WebRtcIsacfix_EncodeReceiveBandwidth(&BWno, &ISACenc_obj->bitstr_obj);
- if (status < 0)
- {
- if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
- {
- // If this is the second 30ms of a 60ms frame reset this such that in the next call
- // encoder starts fresh.
- ISACenc_obj->frame_nb = 0;
- }
- return status;
- }
- }
-
- /* split signal in two bands */
- WebRtcIsacfix_SplitAndFilter1(ISACenc_obj->data_buffer_fix, LP16a, HP16a, &ISACenc_obj->prefiltbankstr_obj );
-
- /* estimate pitch parameters and pitch-filter lookahead signal */
- WebRtcIsacfix_PitchAnalysis(LP16a+QLOOKAHEAD, LPandHP,
- &ISACenc_obj->pitchanalysisstr_obj, PitchLags_Q7, PitchGains_Q12); /* LPandHP = LP_lookahead_pfQ0, */
-
- /* Set where to store data in multiple packets memory */
- if (ISACenc_obj->SaveEnc_ptr != NULL) {
- if (frame_mode == 0 || ISACenc_obj->frame_nb == 0)
- {
- (ISACenc_obj->SaveEnc_ptr)->startIdx = 0;
- }
- else
- {
- (ISACenc_obj->SaveEnc_ptr)->startIdx = 1;
- }
- }
-
- /* quantize & encode pitch parameters */
- status = WebRtcIsacfix_EncodePitchGain(PitchGains_Q12, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr);
- if (status < 0)
- {
- if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
- {
- // If this is the second 30ms of a 60ms frame reset this such that in the next call
- // encoder starts fresh.
- ISACenc_obj->frame_nb = 0;
- }
- return status;
- }
- status = WebRtcIsacfix_EncodePitchLag(PitchLags_Q7 , PitchGains_Q12, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr);
- if (status < 0)
- {
- if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
- {
- // If this is the second 30ms of a 60ms frame reset this such that in the next call
- // encoder starts fresh.
- ISACenc_obj->frame_nb = 0;
- }
- return status;
- }
- AvgPitchGain_Q12 = WEBRTC_SPL_RSHIFT_W32(PitchGains_Q12[0] + PitchGains_Q12[1] + PitchGains_Q12[2] + PitchGains_Q12[3], 2);
-
- /* find coefficients for perceptual pre-filters */
- WebRtcIsacfix_GetLpcCoef(LPandHP, HP16a+QLOOKAHEAD, &ISACenc_obj->maskfiltstr_obj,
- ISACenc_obj->s2nr, PitchGains_Q12,
- gain_lo_hiQ17, lofilt_coefQ15, hifilt_coefQ15); /*LPandHP = LP_lookahead_pfQ0*/
-
- // record LPC Gains for possible bit-rate reduction
- for(k = 0; k < KLT_ORDER_GAIN; k++)
- {
- transcodingParam.lpcGains[k] = gain_lo_hiQ17[k];
- }
-
- /* code LPC model and shape - gains not quantized yet */
- status = WebRtcIsacfix_EncodeLpc(gain_lo_hiQ17, lofilt_coefQ15, hifilt_coefQ15,
- &bmodel, &bits_gainsQ11, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr, &transcodingParam);
- if (status < 0)
- {
- if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
- {
- // If this is the second 30ms of a 60ms frame reset this such that in the next call
- // encoder starts fresh.
- ISACenc_obj->frame_nb = 0;
- }
- return status;
- }
- arithLenBeforeEncodingDFT = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full);
-
- /* low-band filtering */
- WebRtcIsacfix_NormLatticeFilterMa(ORDERLO, ISACenc_obj->maskfiltstr_obj.PreStateLoGQ15,
- LP16a, lofilt_coefQ15, gain_lo_hiQ17, 0, LPandHP);/* LPandHP = LP16b */
-
- /* pitch filter */
- WebRtcIsacfix_PitchFilter(LPandHP, LP16a, &ISACenc_obj->pitchfiltstr_obj, PitchLags_Q7, PitchGains_Q12, 1);/* LPandHP = LP16b */
-
- /* high-band filtering */
- WebRtcIsacfix_NormLatticeFilterMa(ORDERHI, ISACenc_obj->maskfiltstr_obj.PreStateHiGQ15,
- HP16a, hifilt_coefQ15, gain_lo_hiQ17, 1, LPandHP);/*LPandHP = HP16b*/
-
- /* transform */
- WebRtcIsacfix_Time2Spec(LP16a, LPandHP, LP16a, LPandHP); /*LPandHP = HP16b*/
-
- /* Save data for multiple packets memory */
- if (ISACenc_obj->SaveEnc_ptr != NULL) {
- for (k = 0; k < FRAMESAMPLES_HALF; k++) {
- (ISACenc_obj->SaveEnc_ptr)->fre[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LP16a[k];
- (ISACenc_obj->SaveEnc_ptr)->fim[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LPandHP[k];
- }
- (ISACenc_obj->SaveEnc_ptr)->AvgPitchGain[(ISACenc_obj->SaveEnc_ptr)->startIdx] = AvgPitchGain_Q12;
- }
-
- /* quantization and lossless coding */
- status = WebRtcIsacfix_EncodeSpec(LP16a, LPandHP, &ISACenc_obj->bitstr_obj, AvgPitchGain_Q12);
- if((status <= -1) && (status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) /*LPandHP = HP16b*/
- {
- if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
- {
- // If this is the second 30ms of a 60ms frame reset this such that in the next call
- // encoder starts fresh.
- ISACenc_obj->frame_nb = 0;
- }
- return status;
- }
-
- if((frame_mode == 1) && (ISACenc_obj->frame_nb == 0))
- {
- // it is a 60ms and we are in the first 30ms
- // then the limit at this point should be half of the assigned value
- payloadLimitBytes = ISACenc_obj->payloadLimitBytes60 >> 1;
- }
- else if (frame_mode == 0)
- {
- // it is a 30ms frame
- payloadLimitBytes = (ISACenc_obj->payloadLimitBytes30) - 3;
- }
- else
- {
- // this is the second half of a 60ms frame.
- payloadLimitBytes = ISACenc_obj->payloadLimitBytes60 - 3; // subract 3 because termination process may add 3 bytes
- }
-
- iterCntr = 0;
- while((((ISACenc_obj->bitstr_obj.stream_index) << 1) > payloadLimitBytes) ||
- (status == -ISAC_DISALLOWED_BITSTREAM_LENGTH))
- {
- WebRtc_Word16 arithLenDFTByte;
- WebRtc_Word16 bytesLeftQ5;
- WebRtc_Word16 ratioQ5[8] = {0, 6, 9, 12, 16, 19, 22, 25};
-
- // According to experiments on TIMIT the following is proper for audio, but it is not agressive enough for tonal inputs
- // such as DTMF, sweep-sine, ...
- //
- // (0.55 - (0.8 - ratio[i]/32) * 5 / 6) * 2^14
- // WebRtc_Word16 scaleQ14[8] = {0, 648, 1928, 3208, 4915, 6195, 7475, 8755};
-
-
- // This is a supper-agressive scaling passed the tests (tonal inputs) tone with one iteration for payload limit
- // of 120 (32kbps bottleneck), number of frames needed a rate-reduction was 58403
- //
- WebRtc_Word16 scaleQ14[8] = {0, 348, 828, 1408, 2015, 3195, 3500, 3500};
- WebRtc_Word16 idx;
-
- if(iterCntr >= MAX_PAYLOAD_LIMIT_ITERATION)
- {
- // We were not able to limit the payload size
-
- if((frame_mode == 1) && (ISACenc_obj->frame_nb == 0))
- {
- // This was the first 30ms of a 60ms frame. Although the payload is larger than it
- // should be but we let the second 30ms be encoded. Maybe togetehr we won't exceed
- // the limit.
- ISACenc_obj->frame_nb = 1;
- return 0;
- }
- else if((frame_mode == 1) && (ISACenc_obj->frame_nb == 1))
- {
- ISACenc_obj->frame_nb = 0;
- }
-
- if(status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)
- {
- return -ISAC_PAYLOAD_LARGER_THAN_LIMIT;
- }
- else
- {
- return status;
- }
- }
- if(status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)
- {
- arithLenDFTByte = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full) - arithLenBeforeEncodingDFT;
- bytesLeftQ5 = (payloadLimitBytes - arithLenBeforeEncodingDFT) << 5;
-
- // bytesLeft / arithLenDFTBytes indicates how much scaling is required a rough estimate (agressive)
- // scale = 0.55 - (0.8 - bytesLeft / arithLenDFTBytes) * 5 / 6
- // bytesLeft / arithLenDFTBytes below 0.2 will have a scale of zero and above 0.8 are treated as 0.8
- // to avoid division we do more simplification.
- //
- // values of (bytesLeft / arithLenDFTBytes)*32 between ratioQ5[i] and ratioQ5[i+1] are rounded to ratioQ5[i]
- // and the corresponding scale is chosen
-
- // we compare bytesLeftQ5 with ratioQ5[]*arithLenDFTByte;
- idx = 4;
- idx += (bytesLeftQ5 >= WEBRTC_SPL_MUL_16_16(ratioQ5[idx], arithLenDFTByte))? 2:-2;
- idx += (bytesLeftQ5 >= WEBRTC_SPL_MUL_16_16(ratioQ5[idx], arithLenDFTByte))? 1:-1;
- idx += (bytesLeftQ5 >= WEBRTC_SPL_MUL_16_16(ratioQ5[idx], arithLenDFTByte))? 0:-1;
- }
- else
- {
- // we are here because the bit-stream did not fit into the buffer, in this case, the stream_index is not
- // trustable, especially if the is the first 30ms of a packet. Thereforem, we will go for the most agressive
- // case.
- idx = 0;
- }
- // scale FFT coefficients to reduce the bit-rate
- for(k = 0; k < FRAMESAMPLES_HALF; k++)
- {
- LP16a[k] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(LP16a[k], scaleQ14[idx], 14);
- LPandHP[k] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(LPandHP[k], scaleQ14[idx], 14);
- }
-
- // Save data for multiple packets memory
- if (ISACenc_obj->SaveEnc_ptr != NULL)
- {
- for(k = 0; k < FRAMESAMPLES_HALF; k++)
- {
- (ISACenc_obj->SaveEnc_ptr)->fre[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LP16a[k];
- (ISACenc_obj->SaveEnc_ptr)->fim[k + (ISACenc_obj->SaveEnc_ptr)->startIdx*FRAMESAMPLES_HALF] = LPandHP[k];
- }
- }
-
- // scale the unquantized LPC gains and save the scaled version for the future use
- for(k = 0; k < KLT_ORDER_GAIN; k++)
- {
- gain_lo_hiQ17[k] = WEBRTC_SPL_MUL_16_32_RSFT14(scaleQ14[idx], transcodingParam.lpcGains[k]);//transcodingParam.lpcGains[k]; //
- transcodingParam.lpcGains[k] = gain_lo_hiQ17[k];
- }
-
- // reset the bit-stream object to the state which it had before encoding LPC Gains
- ISACenc_obj->bitstr_obj.full = transcodingParam.full;
- ISACenc_obj->bitstr_obj.stream_index = transcodingParam.stream_index;
- ISACenc_obj->bitstr_obj.streamval = transcodingParam.streamval;
- ISACenc_obj->bitstr_obj.W_upper = transcodingParam.W_upper;
- ISACenc_obj->bitstr_obj.stream[transcodingParam.stream_index-1] = transcodingParam.beforeLastWord;
- ISACenc_obj->bitstr_obj.stream[transcodingParam.stream_index] = transcodingParam.lastWord;
-
-
- // quantize and encode LPC gain
- WebRtcIsacfix_EstCodeLpcGain(gain_lo_hiQ17, &ISACenc_obj->bitstr_obj, ISACenc_obj->SaveEnc_ptr);
- arithLenBeforeEncodingDFT = (ISACenc_obj->bitstr_obj.stream_index << 1) + (1-ISACenc_obj->bitstr_obj.full);
- status = WebRtcIsacfix_EncodeSpec(LP16a, LPandHP, &ISACenc_obj->bitstr_obj, AvgPitchGain_Q12);
- if((status <= -1) && (status != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) /*LPandHP = HP16b*/
- {
- if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
- {
- // If this is the second 30ms of a 60ms frame reset this such that in the next call
- // encoder starts fresh.
- ISACenc_obj->frame_nb = 0;
- }
- return status;
- }
- iterCntr++;
- }
-
- if (frame_mode == 1 && ISACenc_obj->frame_nb == 0)
- /* i.e. 60 ms framesize and just processed the first 30ms, */
- /* go back to main function to buffer the other 30ms speech frame */
- {
- ISACenc_obj->frame_nb = 1;
- return 0;
- }
- else if (frame_mode == 1 && ISACenc_obj->frame_nb == 1)
- {
- ISACenc_obj->frame_nb = 0;
- /* also update the framelength for next packet, in Adaptive mode only */
- if (CodingMode == 0 && (ISACenc_obj->enforceFrameSize == 0)) {
- ISACenc_obj->new_framelength = WebRtcIsacfix_GetNewFrameLength(ISACenc_obj->BottleNeck,
- ISACenc_obj->current_framesamples);
- }
- }
-
-
- /* complete arithmetic coding */
- stream_length = WebRtcIsacfix_EncTerminate(&ISACenc_obj->bitstr_obj);
- /* can this be negative? */
-
- if(CodingMode == 0)
- {
-
- /* update rate model and get minimum number of bytes in this packet */
- MinBytes = WebRtcIsacfix_GetMinBytes(&ISACenc_obj->rate_data_obj, (WebRtc_Word16) stream_length,
- ISACenc_obj->current_framesamples, ISACenc_obj->BottleNeck, ISACenc_obj->MaxDelay);
-
- /* if bitstream is too short, add garbage at the end */
-
- /* Store length of coded data */
- usefulstr_len = stream_length;
-
- /* Make sure MinBytes does not exceed packet size limit */
- if ((ISACenc_obj->frame_nb == 0) && (MinBytes > ISACenc_obj->payloadLimitBytes30)) {
- MinBytes = ISACenc_obj->payloadLimitBytes30;
- } else if ((ISACenc_obj->frame_nb == 1) && (MinBytes > ISACenc_obj->payloadLimitBytes60)) {
- MinBytes = ISACenc_obj->payloadLimitBytes60;
- }
-
- /* Make sure we don't allow more than 255 bytes of garbage data.
- We store the length of the garbage data in 8 bits in the bitstream,
- 255 is the max garbage lenght we can signal using 8 bits. */
- if( MinBytes > usefulstr_len + 255 ) {
- MinBytes = usefulstr_len + 255;
- }
-
- /* Save data for creation of multiple bitstreams */
- if (ISACenc_obj->SaveEnc_ptr != NULL) {
- (ISACenc_obj->SaveEnc_ptr)->minBytes = MinBytes;
- }
-
- while (stream_length < MinBytes)
- {
- if (stream_length & 0x0001){
- ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed );
- ISACenc_obj->bitstr_obj.stream[ WEBRTC_SPL_RSHIFT_W16(stream_length, 1) ] |= (WebRtc_UWord16)(ISACenc_obj->bitstr_seed & 0xFF);
- } else {
- ISACenc_obj->bitstr_seed = WEBRTC_SPL_RAND( ISACenc_obj->bitstr_seed );
- ISACenc_obj->bitstr_obj.stream[ WEBRTC_SPL_RSHIFT_W16(stream_length, 1) ] = WEBRTC_SPL_LSHIFT_U16(ISACenc_obj->bitstr_seed, 8);
- }
- stream_length++;
- }
-
- /* to get the real stream_length, without garbage */
- if (usefulstr_len & 0x0001) {
- ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] &= 0xFF00;
- ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] += (MinBytes - usefulstr_len) & 0x00FF;
- }
- else {
- ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] &= 0x00FF;
- ISACenc_obj->bitstr_obj.stream[usefulstr_len>>1] += WEBRTC_SPL_LSHIFT_U16((MinBytes - usefulstr_len) & 0x00FF, 8);
- }
- }
- else
- {
- /* update rate model */
- WebRtcIsacfix_UpdateRateModel(&ISACenc_obj->rate_data_obj, (WebRtc_Word16) stream_length,
- ISACenc_obj->current_framesamples, ISACenc_obj->BottleNeck);
- }
- return stream_length;
-}
-
-/* This function is used to create a new bitstream with new BWE.
- The same data as previously encoded with the fucntion WebRtcIsacfix_EncodeImpl()
- is used. The data needed is taken from the struct, where it was stored
- when calling the encoder. */
-int WebRtcIsacfix_EncodeStoredData(ISACFIX_EncInst_t *ISACenc_obj,
- int BWnumber,
- float scale)
-{
- int ii;
- int status;
- WebRtc_Word16 BWno = BWnumber;
- int stream_length = 0;
-
- WebRtc_Word16 model;
- const WebRtc_UWord16 *Q_PitchGain_cdf_ptr[1];
- const WebRtc_UWord16 **cdf;
- const ISAC_SaveEncData_t *SaveEnc_str;
- WebRtc_Word32 tmpLPCcoeffs_g[KLT_ORDER_GAIN<<1];
- WebRtc_Word16 tmpLPCindex_g[KLT_ORDER_GAIN<<1];
- WebRtc_Word16 tmp_fre[FRAMESAMPLES];
- WebRtc_Word16 tmp_fim[FRAMESAMPLES];
-
- SaveEnc_str = ISACenc_obj->SaveEnc_ptr;
-
- /* Check if SaveEnc memory exists */
- if (SaveEnc_str == NULL) {
- return (-1);
- }
-
- /* Sanity Check - possible values for BWnumber is 0 - 23 */
- if ((BWnumber < 0) || (BWnumber > 23)) {
- return -ISAC_RANGE_ERROR_BW_ESTIMATOR;
- }
-
- /* reset bitstream */
- ISACenc_obj->bitstr_obj.W_upper = 0xFFFFFFFF;
- ISACenc_obj->bitstr_obj.streamval = 0;
- ISACenc_obj->bitstr_obj.stream_index = 0;
- ISACenc_obj->bitstr_obj.full = 1;
-
- /* encode frame length */
- status = WebRtcIsacfix_EncodeFrameLen(SaveEnc_str->framelength, &ISACenc_obj->bitstr_obj);
- if (status < 0) {
- /* Wrong frame size */
- return status;
- }
-
- /* encode bandwidth estimate */
- status = WebRtcIsacfix_EncodeReceiveBandwidth(&BWno, &ISACenc_obj->bitstr_obj);
- if (status < 0) {
- return status;
- }
-
- /* Transcoding */
- /* If scale < 1, rescale data to produce lower bitrate signal */
- if ((0.0 < scale) && (scale < 1.0)) {
- /* Compensate LPC gain */
- for (ii = 0; ii < (KLT_ORDER_GAIN*(1+SaveEnc_str->startIdx)); ii++) {
- tmpLPCcoeffs_g[ii] = (WebRtc_Word32) ((scale) * (float) SaveEnc_str->LPCcoeffs_g[ii]);
- }
-
- /* Scale DFT */
- for (ii = 0; ii < (FRAMESAMPLES_HALF*(1+SaveEnc_str->startIdx)); ii++) {
- tmp_fre[ii] = (WebRtc_Word16) ((scale) * (float) SaveEnc_str->fre[ii]) ;
- tmp_fim[ii] = (WebRtc_Word16) ((scale) * (float) SaveEnc_str->fim[ii]) ;
- }
- } else {
- for (ii = 0; ii < (KLT_ORDER_GAIN*(1+SaveEnc_str->startIdx)); ii++) {
- tmpLPCindex_g[ii] = SaveEnc_str->LPCindex_g[ii];
- }
-
- for (ii = 0; ii < (FRAMESAMPLES_HALF*(1+SaveEnc_str->startIdx)); ii++) {
- tmp_fre[ii] = SaveEnc_str->fre[ii];
- tmp_fim[ii] = SaveEnc_str->fim[ii];
- }
- }
-
- /* Loop over number of 30 msec */
- for (ii = 0; ii <= SaveEnc_str->startIdx; ii++)
- {
-
- /* encode pitch gains */
- *Q_PitchGain_cdf_ptr = WebRtcIsacfix_kPitchGainCdf;
- status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &SaveEnc_str->pitchGain_index[ii],
- Q_PitchGain_cdf_ptr, 1);
- if (status < 0) {
- return status;
- }
-
- /* entropy coding of quantization pitch lags */
- /* voicing classificiation */
- if (SaveEnc_str->meanGain[ii] <= 819) {
- cdf = WebRtcIsacfix_kPitchLagPtrLo;
- } else if (SaveEnc_str->meanGain[ii] <= 1638) {
- cdf = WebRtcIsacfix_kPitchLagPtrMid;
- } else {
- cdf = WebRtcIsacfix_kPitchLagPtrHi;
- }
- status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj,
- &SaveEnc_str->pitchIndex[PITCH_SUBFRAMES*ii], cdf, PITCH_SUBFRAMES);
- if (status < 0) {
- return status;
- }
-
- /* LPC */
- /* entropy coding of model number */
- model = 0;
- status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &model,
- WebRtcIsacfix_kModelCdfPtr, 1);
- if (status < 0) {
- return status;
- }
-
- /* entropy coding of quantization indices - LPC shape only */
- status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &SaveEnc_str->LPCindex_s[KLT_ORDER_SHAPE*ii],
- WebRtcIsacfix_kCdfShapePtr[0], KLT_ORDER_SHAPE);
- if (status < 0) {
- return status;
- }
-
- /* If transcoding, get new LPC gain indices */
- if (scale < 1.0) {
- WebRtcIsacfix_TranscodeLpcCoef(&tmpLPCcoeffs_g[KLT_ORDER_GAIN*ii], &tmpLPCindex_g[KLT_ORDER_GAIN*ii]);
- }
-
- /* entropy coding of quantization indices - LPC gain */
- status = WebRtcIsacfix_EncHistMulti(&ISACenc_obj->bitstr_obj, &tmpLPCindex_g[KLT_ORDER_GAIN*ii],
- WebRtcIsacfix_kCdfGainPtr[0], KLT_ORDER_GAIN);
- if (status < 0) {
- return status;
- }
-
- /* quantization and lossless coding */
- status = WebRtcIsacfix_EncodeSpec(&tmp_fre[ii*FRAMESAMPLES_HALF], &tmp_fim[ii*FRAMESAMPLES_HALF],
- &ISACenc_obj->bitstr_obj, SaveEnc_str->AvgPitchGain[ii]);
- if (status < 0) {
- return status;
- }
- }
-
- /* complete arithmetic coding */
- stream_length = WebRtcIsacfix_EncTerminate(&ISACenc_obj->bitstr_obj);
-
- return stream_length;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/entropy_coding.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/entropy_coding.c
deleted file mode 100644
index 0b64d83..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/entropy_coding.c
+++ /dev/null
@@ -1,2072 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * entropy_coding.c
- *
- * This file contains all functions used to arithmetically
- * encode the iSAC bistream.
- *
- */
-
-#include <stddef.h>
-
-#include "arith_routins.h"
-#include "spectrum_ar_model_tables.h"
-#include "pitch_gain_tables.h"
-#include "pitch_lag_tables.h"
-#include "entropy_coding.h"
-#include "lpc_tables.h"
-#include "settings.h"
-#include "signal_processing_library.h"
-
-
-/*
- This function implements the fix-point correspondant function to lrint.
-
- FLP: (WebRtc_Word32)floor(flt+.499999999999)
- FIP: (fixVal+roundVal)>>qDomain
-
- where roundVal = 2^(qDomain-1) = 1<<(qDomain-1)
-
-*/
-static __inline WebRtc_Word32 CalcLrIntQ(WebRtc_Word32 fixVal, WebRtc_Word16 qDomain) {
- WebRtc_Word32 intgr;
- WebRtc_Word32 roundVal;
-
- roundVal = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, qDomain-1);
- intgr = WEBRTC_SPL_RSHIFT_W32(fixVal+roundVal, qDomain);
-
- return intgr;
-}
-
-/*
- __inline WebRtc_UWord32 stepwise(WebRtc_Word32 dinQ10) {
-
- WebRtc_Word32 ind, diQ10, dtQ10;
-
- diQ10 = dinQ10;
- if (diQ10 < DPMIN_Q10)
- diQ10 = DPMIN_Q10;
- if (diQ10 >= DPMAX_Q10)
- diQ10 = DPMAX_Q10 - 1;
-
- dtQ10 = diQ10 - DPMIN_Q10;*/ /* Q10 + Q10 = Q10 */
-/* ind = (dtQ10 * 5) >> 10; */ /* 2^10 / 5 = 0.2 in Q10 */
-/* Q10 -> Q0 */
-
-/* return rpointsFIX_Q10[ind];
-
- }
-*/
-
-/* logN(x) = logN(2)*log2(x) = 0.6931*log2(x). Output in Q8. */
-/* The input argument X to logN(X) is 2^17 times higher than the
- input floating point argument Y to log(Y), since the X value
- is a Q17 value. This can be compensated for after the call, by
- subraction a value Z for each Q-step. One Q-step means that
- X gets 2 thimes higher, i.e. Z = logN(2)*256 = 0.693147180559*256 =
- 177.445678 should be subtracted (since logN() returns a Q8 value).
- For a X value in Q17, the value 177.445678*17 = 3017 should be
- subtracted */
-static WebRtc_Word16 CalcLogN(WebRtc_Word32 arg) {
- WebRtc_Word16 zeros, log2, frac, logN;
-
- zeros=WebRtcSpl_NormU32(arg);
- frac=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(WEBRTC_SPL_LSHIFT_W32(arg, zeros)&0x7FFFFFFF, 23);
- log2=(WebRtc_Word16)(WEBRTC_SPL_LSHIFT_W32(31-zeros, 8)+frac); // log2(x) in Q8
- logN=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(log2,22713,15); //Q8*Q15 log(2) = 0.693147 = 22713 in Q15
- logN=logN+11; //Scalar compensation which minimizes the (log(x)-logN(x))^2 error over all x.
-
- return logN;
-}
-
-
-/*
- expN(x) = 2^(a*x), where a = log2(e) ~= 1.442695
-
- Input: Q8 (WebRtc_Word16)
- Output: Q17 (WebRtc_Word32)
-
- a = log2(e) = log2(exp(1)) ~= 1.442695 ==> a = 23637 in Q14 (1.442688)
- To this value, 700 is added or subtracted in order to get an average error
- nearer zero, instead of always same-sign.
-*/
-
-static WebRtc_Word32 CalcExpN(WebRtc_Word16 x) {
- WebRtc_Word16 ax, axINT, axFRAC;
- WebRtc_Word16 exp16;
- WebRtc_Word32 exp;
-
- if (x>=0) {
- // ax=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637-700, 14); //Q8
- ax=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637, 14); //Q8
- axINT = WEBRTC_SPL_RSHIFT_W16(ax, 8); //Q0
- axFRAC = ax&0x00FF;
- exp16 = WEBRTC_SPL_LSHIFT_W32(1, axINT); //Q0
- axFRAC = axFRAC+256; //Q8
- exp = WEBRTC_SPL_MUL_16_16(exp16, axFRAC); // Q0*Q8 = Q8
- exp = WEBRTC_SPL_LSHIFT_W32(exp, 9); //Q17
- } else {
- // ax=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637+700, 14); //Q8
- ax=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(x, 23637, 14); //Q8
- ax = -ax;
- axINT = 1 + WEBRTC_SPL_RSHIFT_W16(ax, 8); //Q0
- axFRAC = 0x00FF - (ax&0x00FF);
- exp16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(32768, axINT); //Q15
- axFRAC = axFRAC+256; //Q8
- exp = WEBRTC_SPL_MUL_16_16(exp16, axFRAC); // Q15*Q8 = Q23
- exp = WEBRTC_SPL_RSHIFT_W32(exp, 6); //Q17
- }
-
- return exp;
-}
-
-
-/* compute correlation from power spectrum */
-static void CalcCorrelation(WebRtc_Word32 *PSpecQ12, WebRtc_Word32 *CorrQ7)
-{
- WebRtc_Word32 summ[FRAMESAMPLES/8];
- WebRtc_Word32 diff[FRAMESAMPLES/8];
- WebRtc_Word32 sum;
- int k, n;
-
- for (k = 0; k < FRAMESAMPLES/8; k++) {
- summ[k] = WEBRTC_SPL_RSHIFT_W32(PSpecQ12[k] + PSpecQ12[FRAMESAMPLES/4-1 - k] + 16, 5);
- diff[k] = WEBRTC_SPL_RSHIFT_W32(PSpecQ12[k] - PSpecQ12[FRAMESAMPLES/4-1 - k] + 16, 5);
- }
-
- sum = 2;
- for (n = 0; n < FRAMESAMPLES/8; n++)
- sum += summ[n];
- CorrQ7[0] = sum;
-
- for (k = 0; k < AR_ORDER; k += 2) {
- sum = 0;
- for (n = 0; n < FRAMESAMPLES/8; n++)
- sum += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(WebRtcIsacfix_kCos[k][n], diff[n]) + 256, 9);
- CorrQ7[k+1] = sum;
- }
-
- for (k=1; k<AR_ORDER; k+=2) {
- sum = 0;
- for (n = 0; n < FRAMESAMPLES/8; n++)
- sum += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(WebRtcIsacfix_kCos[k][n], summ[n]) + 256, 9);
- CorrQ7[k+1] = sum;
- }
-}
-
-
-/* compute inverse AR power spectrum */
-static void CalcInvArSpec(const WebRtc_Word16 *ARCoefQ12,
- const WebRtc_Word32 gainQ10,
- WebRtc_Word32 *CurveQ16)
-{
- WebRtc_Word32 CorrQ11[AR_ORDER+1];
- WebRtc_Word32 sum, tmpGain;
- WebRtc_Word32 diffQ16[FRAMESAMPLES/8];
- const WebRtc_Word16 *CS_ptrQ9;
- int k, n;
- WebRtc_Word16 round, shftVal = 0, sh;
-
- sum = 0;
- for (n = 0; n < AR_ORDER+1; n++)
- sum += WEBRTC_SPL_MUL(ARCoefQ12[n], ARCoefQ12[n]); /* Q24 */
- sum = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(sum, 6), 65) + 32768, 16); /* result in Q8 */
- CorrQ11[0] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, gainQ10) + 256, 9);
-
- /* To avoid overflow, we shift down gainQ10 if it is large. We will not lose any precision */
- if(gainQ10>400000){
- tmpGain = WEBRTC_SPL_RSHIFT_W32(gainQ10, 3);
- round = 32;
- shftVal = 6;
- } else {
- tmpGain = gainQ10;
- round = 256;
- shftVal = 9;
- }
-
- for (k = 1; k < AR_ORDER+1; k++) {
- sum = 16384;
- for (n = k; n < AR_ORDER+1; n++)
- sum += WEBRTC_SPL_MUL(ARCoefQ12[n-k], ARCoefQ12[n]); /* Q24 */
- sum = WEBRTC_SPL_RSHIFT_W32(sum, 15);
- CorrQ11[k] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, tmpGain) + round, shftVal);
- }
- sum = WEBRTC_SPL_LSHIFT_W32(CorrQ11[0], 7);
- for (n = 0; n < FRAMESAMPLES/8; n++)
- CurveQ16[n] = sum;
-
- for (k = 1; k < AR_ORDER; k += 2) {
- for (n = 0; n < FRAMESAMPLES/8; n++)
- CurveQ16[n] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(WebRtcIsacfix_kCos[k][n], CorrQ11[k+1]) + 2, 2);
- }
-
- CS_ptrQ9 = WebRtcIsacfix_kCos[0];
-
- /* If CorrQ11[1] too large we avoid getting overflow in the calculation by shifting */
- sh=WebRtcSpl_NormW32(CorrQ11[1]);
- if (CorrQ11[1]==0) /* Use next correlation */
- sh=WebRtcSpl_NormW32(CorrQ11[2]);
-
- if (sh<9)
- shftVal = 9 - sh;
- else
- shftVal = 0;
-
- for (n = 0; n < FRAMESAMPLES/8; n++)
- diffQ16[n] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CS_ptrQ9[n], WEBRTC_SPL_RSHIFT_W32(CorrQ11[1], shftVal)) + 2, 2);
- for (k = 2; k < AR_ORDER; k += 2) {
- CS_ptrQ9 = WebRtcIsacfix_kCos[k];
- for (n = 0; n < FRAMESAMPLES/8; n++)
- diffQ16[n] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CS_ptrQ9[n], WEBRTC_SPL_RSHIFT_W32(CorrQ11[k+1], shftVal)) + 2, 2);
- }
-
- for (k=0; k<FRAMESAMPLES/8; k++) {
- CurveQ16[FRAMESAMPLES/4-1 - k] = CurveQ16[k] - WEBRTC_SPL_LSHIFT_W32(diffQ16[k], shftVal);
- CurveQ16[k] += WEBRTC_SPL_LSHIFT_W32(diffQ16[k], shftVal);
- }
-}
-
-static void CalcRootInvArSpec(const WebRtc_Word16 *ARCoefQ12,
- const WebRtc_Word32 gainQ10,
- WebRtc_UWord16 *CurveQ8)
-{
- WebRtc_Word32 CorrQ11[AR_ORDER+1];
- WebRtc_Word32 sum, tmpGain;
- WebRtc_Word32 summQ16[FRAMESAMPLES/8];
- WebRtc_Word32 diffQ16[FRAMESAMPLES/8];
-
- const WebRtc_Word16 *CS_ptrQ9;
- int k, n, i;
- WebRtc_Word16 round, shftVal = 0, sh;
- WebRtc_Word32 res, in_sqrt, newRes;
-
- sum = 0;
- for (n = 0; n < AR_ORDER+1; n++)
- sum += WEBRTC_SPL_MUL(ARCoefQ12[n], ARCoefQ12[n]); /* Q24 */
- sum = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(sum, 6), 65) + 32768, 16); /* result in Q8 */
- CorrQ11[0] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, gainQ10) + 256, 9);
-
- /* To avoid overflow, we shift down gainQ10 if it is large. We will not lose any precision */
- if(gainQ10>400000){
- tmpGain = WEBRTC_SPL_RSHIFT_W32(gainQ10, 3);
- round = 32;
- shftVal = 6;
- } else {
- tmpGain = gainQ10;
- round = 256;
- shftVal = 9;
- }
-
- for (k = 1; k < AR_ORDER+1; k++) {
- sum = 16384;
- for (n = k; n < AR_ORDER+1; n++)
- sum += WEBRTC_SPL_MUL(ARCoefQ12[n-k], ARCoefQ12[n]); /* Q24 */
- sum = WEBRTC_SPL_RSHIFT_W32(sum, 15);
- CorrQ11[k] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, tmpGain) + round, shftVal);
- }
- sum = WEBRTC_SPL_LSHIFT_W32(CorrQ11[0], 7);
- for (n = 0; n < FRAMESAMPLES/8; n++)
- summQ16[n] = sum;
-
- for (k = 1; k < (AR_ORDER); k += 2) {
- for (n = 0; n < FRAMESAMPLES/8; n++)
- summQ16[n] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(CorrQ11[k+1],WebRtcIsacfix_kCos[k][n]) + 2, 2);
- }
-
- CS_ptrQ9 = WebRtcIsacfix_kCos[0];
-
- /* If CorrQ11[1] too large we avoid getting overflow in the calculation by shifting */
- sh=WebRtcSpl_NormW32(CorrQ11[1]);
- if (CorrQ11[1]==0) /* Use next correlation */
- sh=WebRtcSpl_NormW32(CorrQ11[2]);
-
- if (sh<9)
- shftVal = 9 - sh;
- else
- shftVal = 0;
-
- for (n = 0; n < FRAMESAMPLES/8; n++)
- diffQ16[n] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CS_ptrQ9[n], WEBRTC_SPL_RSHIFT_W32(CorrQ11[1], shftVal)) + 2, 2);
- for (k = 2; k < AR_ORDER; k += 2) {
- CS_ptrQ9 = WebRtcIsacfix_kCos[k];
- for (n = 0; n < FRAMESAMPLES/8; n++)
- diffQ16[n] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CS_ptrQ9[n], WEBRTC_SPL_RSHIFT_W32(CorrQ11[k+1], shftVal)) + 2, 2);
- }
-
- in_sqrt = summQ16[0] + WEBRTC_SPL_LSHIFT_W32(diffQ16[0], shftVal);
-
- /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */
- res = WEBRTC_SPL_LSHIFT_W32(1, WEBRTC_SPL_RSHIFT_W16(WebRtcSpl_GetSizeInBits(in_sqrt), 1));
-
- for (k = 0; k < FRAMESAMPLES/8; k++)
- {
- in_sqrt = summQ16[k] + WEBRTC_SPL_LSHIFT_W32(diffQ16[k], shftVal);
- i = 10;
-
- /* make in_sqrt positive to prohibit sqrt of negative values */
- if(in_sqrt<0)
- in_sqrt=-in_sqrt;
-
- newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
- do
- {
- res = newRes;
- newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
- } while (newRes != res && i-- > 0);
-
- CurveQ8[k] = (WebRtc_Word16)newRes;
- }
- for (k = FRAMESAMPLES/8; k < FRAMESAMPLES/4; k++) {
-
- in_sqrt = summQ16[FRAMESAMPLES/4-1 - k] - WEBRTC_SPL_LSHIFT_W32(diffQ16[FRAMESAMPLES/4-1 - k], shftVal);
- i = 10;
-
- /* make in_sqrt positive to prohibit sqrt of negative values */
- if(in_sqrt<0)
- in_sqrt=-in_sqrt;
-
- newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
- do
- {
- res = newRes;
- newRes = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_DIV(in_sqrt, res) + res, 1);
- } while (newRes != res && i-- > 0);
-
- CurveQ8[k] = (WebRtc_Word16)newRes;
- }
-
-}
-
-
-
-/* generate array of dither samples in Q7 */
-static void GenerateDitherQ7(WebRtc_Word16 *bufQ7,
- WebRtc_UWord32 seed,
- WebRtc_Word16 length,
- WebRtc_Word16 AvgPitchGain_Q12)
-{
- int k;
- WebRtc_Word16 dither1_Q7, dither2_Q7, dither_gain_Q14, shft;
-
- if (AvgPitchGain_Q12 < 614) /* this threshold should be equal to that in decode_spec() */
- {
- for (k = 0; k < length-2; k += 3)
- {
- /* new random unsigned WebRtc_Word32 */
- seed = WEBRTC_SPL_UMUL(seed, 196314165) + 907633515;
-
- /* fixed-point dither sample between -64 and 64 (Q7) */
- dither1_Q7 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)seed + 16777216, 25); // * 128/4294967295
-
- /* new random unsigned WebRtc_Word32 */
- seed = WEBRTC_SPL_UMUL(seed, 196314165) + 907633515;
-
- /* fixed-point dither sample between -64 and 64 */
- dither2_Q7 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(seed + 16777216, 25);
-
- shft = (WebRtc_Word16)(WEBRTC_SPL_RSHIFT_U32(seed, 25) & 15);
- if (shft < 5)
- {
- bufQ7[k] = dither1_Q7;
- bufQ7[k+1] = dither2_Q7;
- bufQ7[k+2] = 0;
- }
- else if (shft < 10)
- {
- bufQ7[k] = dither1_Q7;
- bufQ7[k+1] = 0;
- bufQ7[k+2] = dither2_Q7;
- }
- else
- {
- bufQ7[k] = 0;
- bufQ7[k+1] = dither1_Q7;
- bufQ7[k+2] = dither2_Q7;
- }
- }
- }
- else
- {
- dither_gain_Q14 = (WebRtc_Word16)(22528 - WEBRTC_SPL_MUL(10, AvgPitchGain_Q12));
-
- /* dither on half of the coefficients */
- for (k = 0; k < length-1; k += 2)
- {
- /* new random unsigned WebRtc_Word32 */
- seed = WEBRTC_SPL_UMUL(seed, 196314165) + 907633515;
-
- /* fixed-point dither sample between -64 and 64 */
- dither1_Q7 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)seed + 16777216, 25);
-
- /* dither sample is placed in either even or odd index */
- shft = (WebRtc_Word16)(WEBRTC_SPL_RSHIFT_U32(seed, 25) & 1); /* either 0 or 1 */
-
- bufQ7[k + shft] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(dither_gain_Q14, dither1_Q7) + 8192, 14);
- bufQ7[k + 1 - shft] = 0;
- }
- }
-}
-
-
-
-
-/*
- * function to decode the complex spectrum from the bitstream
- * returns the total number of bytes in the stream
- */
-WebRtc_Word16 WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
- WebRtc_Word16 *frQ7,
- WebRtc_Word16 *fiQ7,
- WebRtc_Word16 AvgPitchGain_Q12)
-{
- WebRtc_Word16 data[FRAMESAMPLES];
- WebRtc_Word32 invARSpec2_Q16[FRAMESAMPLES/4];
- WebRtc_Word16 ARCoefQ12[AR_ORDER+1];
- WebRtc_Word16 RCQ15[AR_ORDER];
- WebRtc_Word16 gainQ10;
- WebRtc_Word32 gain2_Q10;
- WebRtc_Word16 len;
- int k;
-
- /* create dither signal */
- GenerateDitherQ7(data, streamdata->W_upper, FRAMESAMPLES, AvgPitchGain_Q12); /* Dither is output in vector 'Data' */
-
- /* decode model parameters */
- if (WebRtcIsacfix_DecodeRcCoef(streamdata, RCQ15) < 0)
- return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
-
-
- WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12);
-
- if (WebRtcIsacfix_DecodeGain2(streamdata, &gain2_Q10) < 0)
- return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
-
- /* compute inverse AR power spectrum */
- CalcInvArSpec(ARCoefQ12, gain2_Q10, invARSpec2_Q16);
-
- /* arithmetic decoding of spectrum */
- /* 'data' input and output. Input = Dither */
- len = WebRtcIsacfix_DecLogisticMulti2(data, streamdata, invARSpec2_Q16, (WebRtc_Word16)FRAMESAMPLES);
-
- if (len<1)
- return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
-
- /* subtract dither and scale down spectral samples with low SNR */
- if (AvgPitchGain_Q12 <= 614)
- {
- for (k = 0; k < FRAMESAMPLES; k += 4)
- {
- gainQ10 = WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)30, 10),
- (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(invARSpec2_Q16[k>>2] + (WebRtc_UWord32)2195456, 16));
- *frQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[ k ], gainQ10) + 512, 10);
- *fiQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+1], gainQ10) + 512, 10);
- *frQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+2], gainQ10) + 512, 10);
- *fiQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+3], gainQ10) + 512, 10);
- }
- }
- else
- {
- for (k = 0; k < FRAMESAMPLES; k += 4)
- {
- gainQ10 = WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)36, 10),
- (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(invARSpec2_Q16[k>>2] + (WebRtc_UWord32)2654208, 16));
- *frQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[ k ], gainQ10) + 512, 10);
- *fiQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+1], gainQ10) + 512, 10);
- *frQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+2], gainQ10) + 512, 10);
- *fiQ7++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(data[k+3], gainQ10) + 512, 10);
- }
- }
-
- return len;
-}
-
-
-int WebRtcIsacfix_EncodeSpec(const WebRtc_Word16 *fr,
- const WebRtc_Word16 *fi,
- Bitstr_enc *streamdata,
- WebRtc_Word16 AvgPitchGain_Q12)
-{
- WebRtc_Word16 dataQ7[FRAMESAMPLES];
- WebRtc_Word32 PSpec[FRAMESAMPLES/4];
- WebRtc_UWord16 invARSpecQ8[FRAMESAMPLES/4];
- WebRtc_Word32 CorrQ7[AR_ORDER+1];
- WebRtc_Word32 CorrQ7_norm[AR_ORDER+1];
- WebRtc_Word16 RCQ15[AR_ORDER];
- WebRtc_Word16 ARCoefQ12[AR_ORDER+1];
- WebRtc_Word32 gain2_Q10;
- WebRtc_Word16 val;
- WebRtc_Word32 nrg;
- WebRtc_UWord32 sum;
- WebRtc_Word16 lft_shft;
- WebRtc_Word16 status;
- int k, n, j;
-
-
- /* create dither_float signal */
- GenerateDitherQ7(dataQ7, streamdata->W_upper, FRAMESAMPLES, AvgPitchGain_Q12);
-
- /* add dither and quantize, and compute power spectrum */
- /* Vector dataQ7 contains Dither in Q7 */
- for (k = 0; k < FRAMESAMPLES; k += 4)
- {
- val = ((*fr++ + dataQ7[k] + 64) & 0xFF80) - dataQ7[k]; /* Data = Dither */
- dataQ7[k] = val; /* New value in Data */
- sum = WEBRTC_SPL_UMUL(val, val);
-
- val = ((*fi++ + dataQ7[k+1] + 64) & 0xFF80) - dataQ7[k+1]; /* Data = Dither */
- dataQ7[k+1] = val; /* New value in Data */
- sum += WEBRTC_SPL_UMUL(val, val);
-
- val = ((*fr++ + dataQ7[k+2] + 64) & 0xFF80) - dataQ7[k+2]; /* Data = Dither */
- dataQ7[k+2] = val; /* New value in Data */
- sum += WEBRTC_SPL_UMUL(val, val);
-
- val = ((*fi++ + dataQ7[k+3] + 64) & 0xFF80) - dataQ7[k+3]; /* Data = Dither */
- dataQ7[k+3] = val; /* New value in Data */
- sum += WEBRTC_SPL_UMUL(val, val);
-
- PSpec[k>>2] = WEBRTC_SPL_RSHIFT_U32(sum, 2);
- }
-
- /* compute correlation from power spectrum */
- CalcCorrelation(PSpec, CorrQ7);
-
-
- /* find AR coefficients */
- /* number of bit shifts to 14-bit normalize CorrQ7[0] (leaving room for sign) */
- lft_shft = WebRtcSpl_NormW32(CorrQ7[0]) - 18;
-
- if (lft_shft > 0) {
- for (k=0; k<AR_ORDER+1; k++)
- CorrQ7_norm[k] = WEBRTC_SPL_LSHIFT_W32(CorrQ7[k], lft_shft);
- } else {
- for (k=0; k<AR_ORDER+1; k++)
- CorrQ7_norm[k] = WEBRTC_SPL_RSHIFT_W32(CorrQ7[k], -lft_shft);
- }
-
- /* find RC coefficients */
- WebRtcSpl_AutoCorrToReflCoef(CorrQ7_norm, AR_ORDER, RCQ15);
-
- /* quantize & code RC Coef */
- status = WebRtcIsacfix_EncodeRcCoef(RCQ15, streamdata);
- if (status < 0) {
- return status;
- }
-
- /* RC -> AR coefficients */
- WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12);
-
- /* compute ARCoef' * Corr * ARCoef in Q19 */
- nrg = 0;
- for (j = 0; j <= AR_ORDER; j++) {
- for (n = 0; n <= j; n++)
- nrg += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(ARCoefQ12[j], WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CorrQ7_norm[j-n], ARCoefQ12[n]) + 256, 9)) + 4, 3);
- for (n = j+1; n <= AR_ORDER; n++)
- nrg += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(ARCoefQ12[j], WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CorrQ7_norm[n-j], ARCoefQ12[n]) + 256, 9)) + 4, 3);
- }
-
- if (lft_shft > 0)
- nrg = WEBRTC_SPL_RSHIFT_W32(nrg, lft_shft);
- else
- nrg = WEBRTC_SPL_LSHIFT_W32(nrg, -lft_shft);
-
- if(nrg>131072)
- gain2_Q10 = WebRtcSpl_DivResultInQ31(FRAMESAMPLES >> 2, nrg); /* also shifts 31 bits to the left! */
- else
- gain2_Q10 = WEBRTC_SPL_RSHIFT_W32(FRAMESAMPLES, 2);
-
- /* quantize & code gain2_Q10 */
- if (WebRtcIsacfix_EncodeGain2(&gain2_Q10, streamdata))
- return -1;
-
- /* compute inverse AR magnitude spectrum */
- CalcRootInvArSpec(ARCoefQ12, gain2_Q10, invARSpecQ8);
-
-
- /* arithmetic coding of spectrum */
- status = WebRtcIsacfix_EncLogisticMulti2(streamdata, dataQ7, invARSpecQ8, (WebRtc_Word16)FRAMESAMPLES);
- if ( status )
- return( status );
-
- return 0;
-}
-
-
-/* Matlab's LAR definition */
-static void Rc2LarFix(const WebRtc_Word16 *rcQ15, WebRtc_Word32 *larQ17, WebRtc_Word16 order) {
-
- /*
-
- This is a piece-wise implemenetation of a rc2lar-function (all values in the comment
- are Q15 values and are based on [0 24956/32768 30000/32768 32500/32768], i.e.
- [0.76159667968750 0.91552734375000 0.99182128906250]
-
- x0 x1 a k x0(again) b
- ==================================================================================
- 0.00 0.76: 0 2.625997508581 0 0
- 0.76 0.91: 2.000012018559 7.284502668663 0.761596679688 -3.547841027073
- 0.91 0.99: 3.121320351712 31.115835041229 0.915527343750 -25.366077452148
- 0.99 1.00: 5.495270168700 686.663805654056 0.991821289063 -675.552510708011
-
- The implementation is y(x)= a + (x-x0)*k, but this can be simplified to
-
- y(x) = a-x0*k + x*k = b + x*k, where b = a-x0*k
-
- akx=[0 2.625997508581 0
- 2.000012018559 7.284502668663 0.761596679688
- 3.121320351712 31.115835041229 0.915527343750
- 5.495270168700 686.663805654056 0.991821289063];
-
- b = akx(:,1) - akx(:,3).*akx(:,2)
-
- [ 0.0
- -3.547841027073
- -25.366077452148
- -675.552510708011]
-
- */
-
- int k;
- WebRtc_Word16 rc;
- WebRtc_Word32 larAbsQ17;
-
- for (k = 0; k < order; k++) {
-
- rc = WEBRTC_SPL_ABS_W16(rcQ15[k]); //Q15
-
- /* Calculate larAbsQ17 in Q17 from rc in Q15 */
-
- if (rc<24956) { //0.7615966 in Q15
- // (Q15*Q13)>>11 = Q17
- larAbsQ17 = WEBRTC_SPL_MUL_16_16_RSFT(rc, 21512, 11);
- } else if (rc<30000) { //0.91552734375 in Q15
- // Q17 + (Q15*Q12)>>10 = Q17
- larAbsQ17 = -465024 + WEBRTC_SPL_MUL_16_16_RSFT(rc, 29837, 10);
- } else if (rc<32500) { //0.99182128906250 in Q15
- // Q17 + (Q15*Q10)>>8 = Q17
- larAbsQ17 = -3324784 + WEBRTC_SPL_MUL_16_16_RSFT(rc, 31863, 8);
- } else {
- // Q17 + (Q15*Q5)>>3 = Q17
- larAbsQ17 = -88546020 + WEBRTC_SPL_MUL_16_16_RSFT(rc, 21973, 3);
- }
-
- if (rcQ15[k]>0) {
- larQ17[k] = larAbsQ17;
- } else {
- larQ17[k] = -larAbsQ17;
- }
- }
-}
-
-
-static void Lar2RcFix(const WebRtc_Word32 *larQ17, WebRtc_Word16 *rcQ15, WebRtc_Word16 order) {
-
- /*
- This is a piece-wise implemenetation of a lar2rc-function
- See comment in Rc2LarFix() about details.
- */
-
- int k;
- WebRtc_Word16 larAbsQ11;
- WebRtc_Word32 rc;
-
- for (k = 0; k < order; k++) {
-
- larAbsQ11 = (WebRtc_Word16) WEBRTC_SPL_ABS_W32(WEBRTC_SPL_RSHIFT_W32(larQ17[k]+32,6)); //Q11
-
- if (larAbsQ11<4097) { //2.000012018559 in Q11
- // Q11*Q16>>12 = Q15
- rc = WEBRTC_SPL_MUL_16_16_RSFT(larAbsQ11, 24957, 12);
- } else if (larAbsQ11<6393) { //3.121320351712 in Q11
- // (Q11*Q17 + Q13)>>13 = Q15
- rc = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(larAbsQ11, 17993) + 130738688), 13);
- } else if (larAbsQ11<11255) { //5.495270168700 in Q11
- // (Q11*Q19 + Q30)>>15 = Q15
- rc = WEBRTC_SPL_RSHIFT_W32((WEBRTC_SPL_MUL_16_16(larAbsQ11, 16850) + 875329820), 15);
- } else {
- // (Q11*Q24>>16 + Q19)>>4 = Q15
- rc = WEBRTC_SPL_RSHIFT_W32(((WEBRTC_SPL_MUL_16_16_RSFT(larAbsQ11, 24433, 16)) + 515804), 4);
- }
-
- if (larQ17[k]<=0) {
- rc = -rc;
- }
-
- rcQ15[k] = (WebRtc_Word16) rc; // Q15
- }
-}
-
-static void Poly2LarFix(WebRtc_Word16 *lowbandQ15,
- WebRtc_Word16 orderLo,
- WebRtc_Word16 *hibandQ15,
- WebRtc_Word16 orderHi,
- WebRtc_Word16 Nsub,
- WebRtc_Word32 *larsQ17) {
-
- int k, n;
- WebRtc_Word32 *outpQ17;
- WebRtc_Word16 orderTot;
- WebRtc_Word32 larQ17[MAX_ORDER]; // Size 7+6 is enough
-
- orderTot = (orderLo + orderHi);
- outpQ17 = larsQ17;
- for (k = 0; k < Nsub; k++) {
-
- Rc2LarFix(lowbandQ15, larQ17, orderLo);
-
- for (n = 0; n < orderLo; n++)
- outpQ17[n] = larQ17[n]; //Q17
-
- Rc2LarFix(hibandQ15, larQ17, orderHi);
-
- for (n = 0; n < orderHi; n++)
- outpQ17[n + orderLo] = larQ17[n]; //Q17;
-
- outpQ17 += orderTot;
- lowbandQ15 += orderLo;
- hibandQ15 += orderHi;
- }
-}
-
-
-static void Lar2polyFix(WebRtc_Word32 *larsQ17,
- WebRtc_Word16 *lowbandQ15,
- WebRtc_Word16 orderLo,
- WebRtc_Word16 *hibandQ15,
- WebRtc_Word16 orderHi,
- WebRtc_Word16 Nsub) {
-
- int k, n;
- WebRtc_Word16 orderTot;
- WebRtc_Word16 *outplQ15, *outphQ15;
- WebRtc_Word32 *inpQ17;
- WebRtc_Word16 rcQ15[7+6];
-
- orderTot = (orderLo + orderHi);
- outplQ15 = lowbandQ15;
- outphQ15 = hibandQ15;
- inpQ17 = larsQ17;
- for (k = 0; k < Nsub; k++) {
-
- /* gains not handled here as in the FLP version */
-
- /* Low band */
- Lar2RcFix(&inpQ17[0], rcQ15, orderLo);
- for (n = 0; n < orderLo; n++)
- outplQ15[n] = rcQ15[n]; // Refl. coeffs
-
- /* High band */
- Lar2RcFix(&inpQ17[orderLo], rcQ15, orderHi);
- for (n = 0; n < orderHi; n++)
- outphQ15[n] = rcQ15[n]; // Refl. coeffs
-
- inpQ17 += orderTot;
- outplQ15 += orderLo;
- outphQ15 += orderHi;
- }
-}
-
-int WebRtcIsacfix_DecodeLpc(WebRtc_Word32 *gain_lo_hiQ17,
- WebRtc_Word16 *LPCCoef_loQ15,
- WebRtc_Word16 *LPCCoef_hiQ15,
- Bitstr_dec *streamdata,
- WebRtc_Word16 *outmodel) {
-
- WebRtc_Word32 larsQ17[KLT_ORDER_SHAPE]; // KLT_ORDER_GAIN+KLT_ORDER_SHAPE == (ORDERLO+ORDERHI)*SUBFRAMES
- int err;
-
- err = WebRtcIsacfix_DecodeLpcCoef(streamdata, larsQ17, gain_lo_hiQ17, outmodel);
- if (err<0) // error check
- return -ISAC_RANGE_ERROR_DECODE_LPC;
-
- Lar2polyFix(larsQ17, LPCCoef_loQ15, ORDERLO, LPCCoef_hiQ15, ORDERHI, SUBFRAMES);
-
- return 0;
-}
-
-/* decode & dequantize LPC Coef */
-int WebRtcIsacfix_DecodeLpcCoef(Bitstr_dec *streamdata,
- WebRtc_Word32 *LPCCoefQ17,
- WebRtc_Word32 *gain_lo_hiQ17,
- WebRtc_Word16 *outmodel)
-{
- int j, k, n;
- int err;
- WebRtc_Word16 pos, pos2, posg, poss, offsg, offss, offs2;
- WebRtc_Word16 gainpos;
- WebRtc_Word16 model;
- WebRtc_Word16 index_QQ[KLT_ORDER_SHAPE];
- WebRtc_Word32 tmpcoeffs_gQ17[KLT_ORDER_GAIN];
- WebRtc_Word32 tmpcoeffs2_gQ21[KLT_ORDER_GAIN];
- WebRtc_Word16 tmpcoeffs_sQ10[KLT_ORDER_SHAPE];
- WebRtc_Word32 tmpcoeffs_sQ17[KLT_ORDER_SHAPE];
- WebRtc_Word32 tmpcoeffs2_sQ18[KLT_ORDER_SHAPE];
- WebRtc_Word32 sumQQ;
- WebRtc_Word16 sumQQ16;
- WebRtc_Word32 tmp32;
-
-
-
- /* entropy decoding of model number */
- err = WebRtcIsacfix_DecHistOneStepMulti(&model, streamdata, WebRtcIsacfix_kModelCdfPtr, WebRtcIsacfix_kModelInitIndex, 1);
- if (err<0) // error check
- return err;
-
- /* entropy decoding of quantization indices */
- err = WebRtcIsacfix_DecHistOneStepMulti(index_QQ, streamdata, WebRtcIsacfix_kCdfShapePtr[model], WebRtcIsacfix_kInitIndexShape[model], KLT_ORDER_SHAPE);
- if (err<0) // error check
- return err;
- /* find quantization levels for coefficients */
- for (k=0; k<KLT_ORDER_SHAPE; k++) {
- tmpcoeffs_sQ10[WebRtcIsacfix_kSelIndShape[k]] = WebRtcIsacfix_kLevelsShapeQ10[WebRtcIsacfix_kOfLevelsShape[model]+WebRtcIsacfix_kOffsetShape[model][k] + index_QQ[k]];
- }
-
- err = WebRtcIsacfix_DecHistOneStepMulti(index_QQ, streamdata, WebRtcIsacfix_kCdfGainPtr[model], WebRtcIsacfix_kInitIndexGain[model], KLT_ORDER_GAIN);
- if (err<0) // error check
- return err;
- /* find quantization levels for coefficients */
- for (k=0; k<KLT_ORDER_GAIN; k++) {
- tmpcoeffs_gQ17[WebRtcIsacfix_kSelIndGain[k]] = WebRtcIsacfix_kLevelsGainQ17[WebRtcIsacfix_kOfLevelsGain[model]+ WebRtcIsacfix_kOffsetGain[model][k] + index_QQ[k]];
- }
-
-
- /* inverse KLT */
-
- /* left transform */ // Transpose matrix!
- offsg = 0;
- offss = 0;
- posg = 0;
- poss = 0;
- for (j=0; j<SUBFRAMES; j++) {
- offs2 = 0;
- for (k=0; k<2; k++) {
- sumQQ = 0;
- pos = offsg;
- pos2 = offs2;
- for (n=0; n<2; n++) {
- sumQQ += (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT1GainQ15[model][pos2], tmpcoeffs_gQ17[pos]<<5)); // (Q15*Q17)>>(16-5) = Q21
- pos++;
- pos2++;
- }
- tmpcoeffs2_gQ21[posg] = sumQQ; //Q21
- posg++;
- offs2 += 2;
- }
- offs2 = 0;
-
- for (k=0; k<LPC_SHAPE_ORDER; k++) {
- sumQQ = 0;
- pos = offss;
- pos2 = offs2;
- for (n=0; n<LPC_SHAPE_ORDER; n++) {
- sumQQ += WEBRTC_SPL_MUL_16_16_RSFT(tmpcoeffs_sQ10[pos], WebRtcIsacfix_kT1ShapeQ15[model][pos2], 7); // (Q10*Q15)>>7 = Q18
- pos++;
- pos2++;
- }
- tmpcoeffs2_sQ18[poss] = sumQQ; //Q18
- poss++;
- offs2 += LPC_SHAPE_ORDER;
- }
- offsg += 2;
- offss += LPC_SHAPE_ORDER;
- }
-
- /* right transform */ // Transpose matrix
- offsg = 0;
- offss = 0;
- posg = 0;
- poss = 0;
- for (j=0; j<SUBFRAMES; j++) {
- posg = offsg;
- for (k=0; k<2; k++) {
- sumQQ = 0;
- pos = k;
- pos2 = j;
- for (n=0; n<SUBFRAMES; n++) {
- sumQQ += WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT2GainQ15[model][pos2], tmpcoeffs2_gQ21[pos]), 1); // (Q15*Q21)>>(16-1) = Q21
- pos += 2;
- pos2 += SUBFRAMES;
-
- }
- tmpcoeffs_gQ17[posg] = WEBRTC_SPL_RSHIFT_W32(sumQQ, 4);
- posg++;
- }
- poss = offss;
- for (k=0; k<LPC_SHAPE_ORDER; k++) {
- sumQQ = 0;
- pos = k;
- pos2 = j;
- for (n=0; n<SUBFRAMES; n++) {
- sumQQ += (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT2ShapeQ15[model][pos2], tmpcoeffs2_sQ18[pos])); // (Q15*Q18)>>16 = Q17
- pos += LPC_SHAPE_ORDER;
- pos2 += SUBFRAMES;
- }
- tmpcoeffs_sQ17[poss] = sumQQ;
- poss++;
- }
- offsg += 2;
- offss += LPC_SHAPE_ORDER;
- }
-
- /* scaling, mean addition, and gain restoration */
- gainpos = 0;
- posg = 0;poss = 0;pos=0;
- for (k=0; k<SUBFRAMES; k++) {
-
- /* log gains */
- sumQQ16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmpcoeffs_gQ17[posg], 2+9); //Divide by 4 and get Q17 to Q8, i.e. shift 2+9
- sumQQ16 += WebRtcIsacfix_kMeansGainQ8[model][posg];
- sumQQ = CalcExpN(sumQQ16); // Q8 in and Q17 out
- gain_lo_hiQ17[gainpos] = sumQQ; //Q17
- gainpos++;
- posg++;
-
- sumQQ16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmpcoeffs_gQ17[posg], 2+9); //Divide by 4 and get Q17 to Q8, i.e. shift 2+9
- sumQQ16 += WebRtcIsacfix_kMeansGainQ8[model][posg];
- sumQQ = CalcExpN(sumQQ16); // Q8 in and Q17 out
- gain_lo_hiQ17[gainpos] = sumQQ; //Q17
- gainpos++;
- posg++;
-
- /* lo band LAR coeffs */
- for (n=0; n<ORDERLO; n++, pos++, poss++) {
- tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(31208, tmpcoeffs_sQ17[poss]); // (Q16*Q17)>>16 = Q17, with 1/2.1 = 0.47619047619 ~= 31208 in Q16
- tmp32 = tmp32 + WebRtcIsacfix_kMeansShapeQ17[model][poss]; // Q17+Q17 = Q17
- LPCCoefQ17[pos] = tmp32;
- }
-
- /* hi band LAR coeffs */
- for (n=0; n<ORDERHI; n++, pos++, poss++) {
- tmp32 = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(18204, tmpcoeffs_sQ17[poss]), 3); // ((Q13*Q17)>>16)<<3 = Q17, with 1/0.45 = 2.222222222222 ~= 18204 in Q13
- tmp32 = tmp32 + WebRtcIsacfix_kMeansShapeQ17[model][poss]; // Q17+Q17 = Q17
- LPCCoefQ17[pos] = tmp32;
- }
- }
-
-
- *outmodel=model;
-
- return 0;
-}
-
-/* estimate codel length of LPC Coef */
-static int EstCodeLpcCoef(WebRtc_Word32 *LPCCoefQ17,
- WebRtc_Word32 *gain_lo_hiQ17,
- WebRtc_Word16 *model,
- WebRtc_Word32 *sizeQ11,
- Bitstr_enc *streamdata,
- ISAC_SaveEncData_t* encData,
- transcode_obj *transcodingParam) {
- int j, k, n;
- WebRtc_Word16 posQQ, pos2QQ, gainpos;
- WebRtc_Word16 pos, pos2, poss, posg, offsg, offss, offs2;
- WebRtc_Word16 index_gQQ[KLT_ORDER_GAIN], index_sQQ[KLT_ORDER_SHAPE];
- WebRtc_Word16 index_ovr_gQQ[KLT_ORDER_GAIN], index_ovr_sQQ[KLT_ORDER_SHAPE];
- WebRtc_Word32 BitsQQ;
-
- WebRtc_Word16 tmpcoeffs_gQ6[KLT_ORDER_GAIN];
- WebRtc_Word32 tmpcoeffs_gQ17[KLT_ORDER_GAIN];
- WebRtc_Word32 tmpcoeffs_sQ17[KLT_ORDER_SHAPE];
- WebRtc_Word32 tmpcoeffs2_gQ21[KLT_ORDER_GAIN];
- WebRtc_Word32 tmpcoeffs2_sQ17[KLT_ORDER_SHAPE];
- WebRtc_Word32 sumQQ;
- WebRtc_Word32 tmp32;
- WebRtc_Word16 sumQQ16;
- int status = 0;
-
- /* write LAR coefficients to statistics file */
- /* Save data for creation of multiple bitstreams (and transcoding) */
- if (encData != NULL) {
- for (k=0; k<KLT_ORDER_GAIN; k++) {
- encData->LPCcoeffs_g[KLT_ORDER_GAIN*encData->startIdx + k] = gain_lo_hiQ17[k];
- }
- }
-
- /* log gains, mean removal and scaling */
- posg = 0;poss = 0;pos=0; gainpos=0;
-
- for (k=0; k<SUBFRAMES; k++) {
- /* log gains */
-
- /* The input argument X to logN(X) is 2^17 times higher than the
- input floating point argument Y to log(Y), since the X value
- is a Q17 value. This can be compensated for after the call, by
- subraction a value Z for each Q-step. One Q-step means that
- X gets 2 times higher, i.e. Z = logN(2)*256 = 0.693147180559*256 =
- 177.445678 should be subtracted (since logN() returns a Q8 value).
- For a X value in Q17, the value 177.445678*17 = 3017 should be
- subtracted */
- tmpcoeffs_gQ6[posg] = CalcLogN(gain_lo_hiQ17[gainpos])-3017; //Q8
- tmpcoeffs_gQ6[posg] -= WebRtcIsacfix_kMeansGainQ8[0][posg]; //Q8, but Q6 after not-needed mult. by 4
- posg++; gainpos++;
-
- tmpcoeffs_gQ6[posg] = CalcLogN(gain_lo_hiQ17[gainpos])-3017; //Q8
- tmpcoeffs_gQ6[posg] -= WebRtcIsacfix_kMeansGainQ8[0][posg]; //Q8, but Q6 after not-needed mult. by 4
- posg++; gainpos++;
-
- /* lo band LAR coeffs */
- for (n=0; n<ORDERLO; n++, poss++, pos++) {
- tmp32 = LPCCoefQ17[pos] - WebRtcIsacfix_kMeansShapeQ17[0][poss]; //Q17
- tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(17203, tmp32<<3); // tmp32 = 2.1*tmp32
- tmpcoeffs_sQ17[poss] = tmp32; //Q17
- }
-
- /* hi band LAR coeffs */
- for (n=0; n<ORDERHI; n++, poss++, pos++) {
- tmp32 = LPCCoefQ17[pos] - WebRtcIsacfix_kMeansShapeQ17[0][poss]; //Q17
- tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(14746, tmp32<<1); // tmp32 = 0.45*tmp32
- tmpcoeffs_sQ17[poss] = tmp32; //Q17
- }
-
- }
-
-
- /* KLT */
-
- /* left transform */
- offsg = 0;
- offss = 0;
- for (j=0; j<SUBFRAMES; j++) {
- posg = offsg;
- for (k=0; k<2; k++) {
- sumQQ = 0;
- pos = offsg;
- pos2 = k;
- for (n=0; n<2; n++) {
- sumQQ += WEBRTC_SPL_MUL_16_16(tmpcoeffs_gQ6[pos], WebRtcIsacfix_kT1GainQ15[0][pos2]); //Q21 = Q6*Q15
- pos++;
- pos2 += 2;
- }
- tmpcoeffs2_gQ21[posg] = sumQQ;
- posg++;
- }
- poss = offss;
- for (k=0; k<LPC_SHAPE_ORDER; k++) {
- sumQQ = 0;
- pos = offss;
- pos2 = k;
- for (n=0; n<LPC_SHAPE_ORDER; n++) {
- sumQQ += (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT1ShapeQ15[0][pos2], tmpcoeffs_sQ17[pos]<<1)); // (Q15*Q17)>>(16-1) = Q17
- pos++;
- pos2 += LPC_SHAPE_ORDER;
- }
- tmpcoeffs2_sQ17[poss] = sumQQ; //Q17
- poss++;
- }
- offsg += 2;
- offss += LPC_SHAPE_ORDER;
- }
-
- /* right transform */
- offsg = 0;
- offss = 0;
- offs2 = 0;
- for (j=0; j<SUBFRAMES; j++) {
- posg = offsg;
- for (k=0; k<2; k++) {
- sumQQ = 0;
- pos = k;
- pos2 = offs2;
- for (n=0; n<SUBFRAMES; n++) {
- sumQQ += WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT2GainQ15[0][pos2], tmpcoeffs2_gQ21[pos]), 1); // (Q15*Q21)>>(16-1) = Q21
- pos += 2;
- pos2++;
- }
- tmpcoeffs_gQ17[posg] = WEBRTC_SPL_RSHIFT_W32(sumQQ, 4);
- posg++;
- }
- poss = offss;
- for (k=0; k<LPC_SHAPE_ORDER; k++) {
- sumQQ = 0;
- pos = k;
- pos2 = offs2;
- for (n=0; n<SUBFRAMES; n++) {
- sumQQ += (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT2ShapeQ15[0][pos2], tmpcoeffs2_sQ17[pos]<<1)); // (Q15*Q17)>>(16-1) = Q17
- pos += LPC_SHAPE_ORDER;
- pos2++;
- }
- tmpcoeffs_sQ17[poss] = sumQQ;
- poss++;
- }
- offs2 += SUBFRAMES;
- offsg += 2;
- offss += LPC_SHAPE_ORDER;
- }
-
- /* quantize coefficients */
-
- BitsQQ = 0;
- for (k=0; k<KLT_ORDER_GAIN; k++) //ATTN: ok?
- {
- posQQ = WebRtcIsacfix_kSelIndGain[k];
- pos2QQ= (WebRtc_Word16)CalcLrIntQ(tmpcoeffs_gQ17[posQQ], 17);
-
- index_gQQ[k] = pos2QQ + WebRtcIsacfix_kQuantMinGain[k]; //ATTN: ok?
- if (index_gQQ[k] < 0) {
- index_gQQ[k] = 0;
- }
- else if (index_gQQ[k] > WebRtcIsacfix_kMaxIndGain[k]) {
- index_gQQ[k] = WebRtcIsacfix_kMaxIndGain[k];
- }
- index_ovr_gQQ[k] = WebRtcIsacfix_kOffsetGain[0][k]+index_gQQ[k];
- posQQ = WebRtcIsacfix_kOfLevelsGain[0] + index_ovr_gQQ[k];
-
- /* Save data for creation of multiple bitstreams */
- if (encData != NULL) {
- encData->LPCindex_g[KLT_ORDER_GAIN*encData->startIdx + k] = index_gQQ[k];
- }
-
- /* determine number of bits */
- sumQQ = WebRtcIsacfix_kCodeLenGainQ11[posQQ]; //Q11
- BitsQQ += sumQQ;
- }
-
- for (k=0; k<KLT_ORDER_SHAPE; k++) //ATTN: ok?
- {
- index_sQQ[k] = (WebRtc_Word16)(CalcLrIntQ(tmpcoeffs_sQ17[WebRtcIsacfix_kSelIndShape[k]], 17) + WebRtcIsacfix_kQuantMinShape[k]); //ATTN: ok?
-
- if (index_sQQ[k] < 0)
- index_sQQ[k] = 0;
- else if (index_sQQ[k] > WebRtcIsacfix_kMaxIndShape[k])
- index_sQQ[k] = WebRtcIsacfix_kMaxIndShape[k];
- index_ovr_sQQ[k] = WebRtcIsacfix_kOffsetShape[0][k]+index_sQQ[k];
-
- posQQ = WebRtcIsacfix_kOfLevelsShape[0] + index_ovr_sQQ[k];
- sumQQ = WebRtcIsacfix_kCodeLenShapeQ11[posQQ]; //Q11
- BitsQQ += sumQQ;
- }
-
-
-
- *model = 0;
- *sizeQ11=BitsQQ;
-
- /* entropy coding of model number */
- status = WebRtcIsacfix_EncHistMulti(streamdata, model, WebRtcIsacfix_kModelCdfPtr, 1);
- if (status < 0) {
- return status;
- }
-
- /* entropy coding of quantization indices - shape only */
- status = WebRtcIsacfix_EncHistMulti(streamdata, index_sQQ, WebRtcIsacfix_kCdfShapePtr[0], KLT_ORDER_SHAPE);
- if (status < 0) {
- return status;
- }
-
- /* Save data for creation of multiple bitstreams */
- if (encData != NULL) {
- for (k=0; k<KLT_ORDER_SHAPE; k++)
- {
- encData->LPCindex_s[KLT_ORDER_SHAPE*encData->startIdx + k] = index_sQQ[k];
- }
- }
- /* save the state of the bitstream object 'streamdata' for the possible bit-rate reduction */
- transcodingParam->full = streamdata->full;
- transcodingParam->stream_index = streamdata->stream_index;
- transcodingParam->streamval = streamdata->streamval;
- transcodingParam->W_upper = streamdata->W_upper;
- transcodingParam->beforeLastWord = streamdata->stream[streamdata->stream_index-1];
- transcodingParam->lastWord = streamdata->stream[streamdata->stream_index];
-
- /* entropy coding of index */
- status = WebRtcIsacfix_EncHistMulti(streamdata, index_gQQ, WebRtcIsacfix_kCdfGainPtr[0], KLT_ORDER_GAIN);
- if (status < 0) {
- return status;
- }
-
- /* find quantization levels for shape coefficients */
- for (k=0; k<KLT_ORDER_SHAPE; k++) {
- tmpcoeffs_sQ17[WebRtcIsacfix_kSelIndShape[k]] = WEBRTC_SPL_MUL(128, WebRtcIsacfix_kLevelsShapeQ10[WebRtcIsacfix_kOfLevelsShape[0]+index_ovr_sQQ[k]]);
-
- }
- /* inverse KLT */
-
- /* left transform */ // Transpose matrix!
- offss = 0;
- poss = 0;
- for (j=0; j<SUBFRAMES; j++) {
- offs2 = 0;
- for (k=0; k<LPC_SHAPE_ORDER; k++) {
- sumQQ = 0;
- pos = offss;
- pos2 = offs2;
- for (n=0; n<LPC_SHAPE_ORDER; n++) {
- sumQQ += (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT1ShapeQ15[0][pos2], tmpcoeffs_sQ17[pos]<<1)); // (Q15*Q17)>>(16-1) = Q17
- pos++;
- pos2++;
- }
- tmpcoeffs2_sQ17[poss] = sumQQ;
-
- poss++;
- offs2 += LPC_SHAPE_ORDER;
- }
- offss += LPC_SHAPE_ORDER;
- }
-
-
- /* right transform */ // Transpose matrix
- offss = 0;
- poss = 0;
- for (j=0; j<SUBFRAMES; j++) {
- poss = offss;
- for (k=0; k<LPC_SHAPE_ORDER; k++) {
- sumQQ = 0;
- pos = k;
- pos2 = j;
- for (n=0; n<SUBFRAMES; n++) {
- sumQQ += (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT2ShapeQ15[0][pos2], tmpcoeffs2_sQ17[pos]<<1)); // (Q15*Q17)>>(16-1) = Q17
- pos += LPC_SHAPE_ORDER;
- pos2 += SUBFRAMES;
- }
- tmpcoeffs_sQ17[poss] = sumQQ;
- poss++;
- }
- offss += LPC_SHAPE_ORDER;
- }
-
- /* scaling, mean addition, and gain restoration */
- poss = 0;pos=0;
- for (k=0; k<SUBFRAMES; k++) {
-
- /* lo band LAR coeffs */
- for (n=0; n<ORDERLO; n++, pos++, poss++) {
- tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(31208, tmpcoeffs_sQ17[poss]); // (Q16*Q17)>>16 = Q17, with 1/2.1 = 0.47619047619 ~= 31208 in Q16
- tmp32 = tmp32 + WebRtcIsacfix_kMeansShapeQ17[0][poss]; // Q17+Q17 = Q17
- LPCCoefQ17[pos] = tmp32;
- }
-
- /* hi band LAR coeffs */
- for (n=0; n<ORDERHI; n++, pos++, poss++) {
- tmp32 = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(18204, tmpcoeffs_sQ17[poss]), 3); // ((Q13*Q17)>>16)<<3 = Q17, with 1/0.45 = 2.222222222222 ~= 18204 in Q13
- tmp32 = tmp32 + WebRtcIsacfix_kMeansShapeQ17[0][poss]; // Q17+Q17 = Q17
- LPCCoefQ17[pos] = tmp32;
- }
-
- }
-
- //to update tmpcoeffs_gQ17 to the proper state
- for (k=0; k<KLT_ORDER_GAIN; k++) {
- tmpcoeffs_gQ17[WebRtcIsacfix_kSelIndGain[k]] = WebRtcIsacfix_kLevelsGainQ17[WebRtcIsacfix_kOfLevelsGain[0]+index_ovr_gQQ[k]];
- }
-
-
-
- /* find quantization levels for coefficients */
-
- /* left transform */
- offsg = 0;
- posg = 0;
- for (j=0; j<SUBFRAMES; j++) {
- offs2 = 0;
- for (k=0; k<2; k++) {
- sumQQ = 0;
- pos = offsg;
- pos2 = offs2;
- for (n=0; n<2; n++) {
- sumQQ += (WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT1GainQ15[0][pos2], tmpcoeffs_gQ17[pos])<<1); // (Q15*Q17)>>(16-1) = Q17
- pos++;
- pos2++;
- }
- tmpcoeffs2_gQ21[posg] = WEBRTC_SPL_LSHIFT_W32(sumQQ, 4); //Q17<<4 = Q21
- posg++;
- offs2 += 2;
- }
- offsg += 2;
- }
-
- /* right transform */ // Transpose matrix
- offsg = 0;
- posg = 0;
- for (j=0; j<SUBFRAMES; j++) {
- posg = offsg;
- for (k=0; k<2; k++) {
- sumQQ = 0;
- pos = k;
- pos2 = j;
- for (n=0; n<SUBFRAMES; n++) {
- sumQQ += WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT2GainQ15[0][pos2], tmpcoeffs2_gQ21[pos]), 1); // (Q15*Q21)>>(16-1) = Q21
- pos += 2;
- pos2 += SUBFRAMES;
- }
- tmpcoeffs_gQ17[posg] = WEBRTC_SPL_RSHIFT_W32(sumQQ, 4);
- posg++;
- }
- offsg += 2;
- }
-
- /* scaling, mean addition, and gain restoration */
- posg = 0;
- gainpos = 0;
- for (k=0; k<2*SUBFRAMES; k++) {
-
- sumQQ16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmpcoeffs_gQ17[posg], 2+9); //Divide by 4 and get Q17 to Q8, i.e. shift 2+9
- sumQQ16 += WebRtcIsacfix_kMeansGainQ8[0][posg];
- sumQQ = CalcExpN(sumQQ16); // Q8 in and Q17 out
- gain_lo_hiQ17[gainpos] = sumQQ; //Q17
-
- gainpos++;
- pos++;posg++;
- }
-
- return 0;
-}
-
-int WebRtcIsacfix_EstCodeLpcGain(WebRtc_Word32 *gain_lo_hiQ17,
- Bitstr_enc *streamdata,
- ISAC_SaveEncData_t* encData) {
- int j, k, n;
- WebRtc_Word16 posQQ, pos2QQ, gainpos;
- WebRtc_Word16 pos, pos2, posg, offsg, offs2;
- WebRtc_Word16 index_gQQ[KLT_ORDER_GAIN];
-
- WebRtc_Word16 tmpcoeffs_gQ6[KLT_ORDER_GAIN];
- WebRtc_Word32 tmpcoeffs_gQ17[KLT_ORDER_GAIN];
- WebRtc_Word32 tmpcoeffs2_gQ21[KLT_ORDER_GAIN];
- WebRtc_Word32 sumQQ;
- int status = 0;
-
- /* write LAR coefficients to statistics file */
- /* Save data for creation of multiple bitstreams (and transcoding) */
- if (encData != NULL) {
- for (k=0; k<KLT_ORDER_GAIN; k++) {
- encData->LPCcoeffs_g[KLT_ORDER_GAIN*encData->startIdx + k] = gain_lo_hiQ17[k];
- }
- }
-
- /* log gains, mean removal and scaling */
- posg = 0; pos = 0; gainpos = 0;
-
- for (k=0; k<SUBFRAMES; k++) {
- /* log gains */
-
- /* The input argument X to logN(X) is 2^17 times higher than the
- input floating point argument Y to log(Y), since the X value
- is a Q17 value. This can be compensated for after the call, by
- subraction a value Z for each Q-step. One Q-step means that
- X gets 2 times higher, i.e. Z = logN(2)*256 = 0.693147180559*256 =
- 177.445678 should be subtracted (since logN() returns a Q8 value).
- For a X value in Q17, the value 177.445678*17 = 3017 should be
- subtracted */
- tmpcoeffs_gQ6[posg] = CalcLogN(gain_lo_hiQ17[gainpos])-3017; //Q8
- tmpcoeffs_gQ6[posg] -= WebRtcIsacfix_kMeansGainQ8[0][posg]; //Q8, but Q6 after not-needed mult. by 4
- posg++; gainpos++;
-
- tmpcoeffs_gQ6[posg] = CalcLogN(gain_lo_hiQ17[gainpos])-3017; //Q8
- tmpcoeffs_gQ6[posg] -= WebRtcIsacfix_kMeansGainQ8[0][posg]; //Q8, but Q6 after not-needed mult. by 4
- posg++; gainpos++;
- }
-
-
- /* KLT */
-
- /* left transform */
- offsg = 0;
- for (j=0; j<SUBFRAMES; j++) {
- posg = offsg;
- for (k=0; k<2; k++) {
- sumQQ = 0;
- pos = offsg;
- pos2 = k;
- for (n=0; n<2; n++) {
- sumQQ += WEBRTC_SPL_MUL_16_16(tmpcoeffs_gQ6[pos], WebRtcIsacfix_kT1GainQ15[0][pos2]); //Q21 = Q6*Q15
- pos++;
- pos2 += 2;
- }
- tmpcoeffs2_gQ21[posg] = sumQQ;
- posg++;
- }
- offsg += 2;
- }
-
- /* right transform */
- offsg = 0;
- offs2 = 0;
- for (j=0; j<SUBFRAMES; j++) {
- posg = offsg;
- for (k=0; k<2; k++) {
- sumQQ = 0;
- pos = k;
- pos2 = offs2;
- for (n=0; n<SUBFRAMES; n++) {
- sumQQ += WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT2GainQ15[0][pos2], tmpcoeffs2_gQ21[pos]), 1); // (Q15*Q21)>>(16-1) = Q21
- pos += 2;
- pos2++;
- }
- tmpcoeffs_gQ17[posg] = WEBRTC_SPL_RSHIFT_W32(sumQQ, 4);
- posg++;
- }
- offsg += 2;
- offs2 += SUBFRAMES;
- }
-
- /* quantize coefficients */
-
- for (k=0; k<KLT_ORDER_GAIN; k++) //ATTN: ok?
- {
- posQQ = WebRtcIsacfix_kSelIndGain[k];
- pos2QQ= (WebRtc_Word16)CalcLrIntQ(tmpcoeffs_gQ17[posQQ], 17);
-
- index_gQQ[k] = pos2QQ + WebRtcIsacfix_kQuantMinGain[k]; //ATTN: ok?
- if (index_gQQ[k] < 0) {
- index_gQQ[k] = 0;
- }
- else if (index_gQQ[k] > WebRtcIsacfix_kMaxIndGain[k]) {
- index_gQQ[k] = WebRtcIsacfix_kMaxIndGain[k];
- }
-
- /* Save data for creation of multiple bitstreams */
- if (encData != NULL) {
- encData->LPCindex_g[KLT_ORDER_GAIN*encData->startIdx + k] = index_gQQ[k];
- }
- }
-
- /* entropy coding of index */
- status = WebRtcIsacfix_EncHistMulti(streamdata, index_gQQ, WebRtcIsacfix_kCdfGainPtr[0], KLT_ORDER_GAIN);
- if (status < 0) {
- return status;
- }
-
- return 0;
-}
-
-
-int WebRtcIsacfix_EncodeLpc(WebRtc_Word32 *gain_lo_hiQ17,
- WebRtc_Word16 *LPCCoef_loQ15,
- WebRtc_Word16 *LPCCoef_hiQ15,
- WebRtc_Word16 *model,
- WebRtc_Word32 *sizeQ11,
- Bitstr_enc *streamdata,
- ISAC_SaveEncData_t* encData,
- transcode_obj *transcodeParam)
-{
- int status = 0;
- WebRtc_Word32 larsQ17[KLT_ORDER_SHAPE]; // KLT_ORDER_SHAPE == (ORDERLO+ORDERHI)*SUBFRAMES
- // = (6+12)*6 == 108
-
- Poly2LarFix(LPCCoef_loQ15, ORDERLO, LPCCoef_hiQ15, ORDERHI, SUBFRAMES, larsQ17);
-
- status = EstCodeLpcCoef(larsQ17, gain_lo_hiQ17, model, sizeQ11, streamdata, encData, transcodeParam);
- if (status < 0) {
- return (status);
- }
-
- Lar2polyFix(larsQ17, LPCCoef_loQ15, ORDERLO, LPCCoef_hiQ15, ORDERHI, SUBFRAMES);
-
- return 0;
-}
-
-
-/* decode & dequantize RC */
-int WebRtcIsacfix_DecodeRcCoef(Bitstr_dec *streamdata, WebRtc_Word16 *RCQ15)
-{
- int k, err;
- WebRtc_Word16 index[AR_ORDER];
-
- /* entropy decoding of quantization indices */
- err = WebRtcIsacfix_DecHistOneStepMulti(index, streamdata, WebRtcIsacfix_kRcCdfPtr, WebRtcIsacfix_kRcInitInd, AR_ORDER);
- if (err<0) // error check
- return err;
-
- /* find quantization levels for reflection coefficients */
- for (k=0; k<AR_ORDER; k++)
- {
- RCQ15[k] = *(WebRtcIsacfix_kRcLevPtr[k] + index[k]);
- }
-
- return 0;
-}
-
-
-
-/* quantize & code RC */
-int WebRtcIsacfix_EncodeRcCoef(WebRtc_Word16 *RCQ15, Bitstr_enc *streamdata)
-{
- int k;
- WebRtc_Word16 index[AR_ORDER];
- int status;
-
- /* quantize reflection coefficients (add noise feedback?) */
- for (k=0; k<AR_ORDER; k++)
- {
- index[k] = WebRtcIsacfix_kRcInitInd[k];
-
- if (RCQ15[k] > WebRtcIsacfix_kRcBound[index[k]])
- {
- while (RCQ15[k] > WebRtcIsacfix_kRcBound[index[k] + 1])
- index[k]++;
- }
- else
- {
- while (RCQ15[k] < WebRtcIsacfix_kRcBound[--index[k]]) ;
- }
-
- RCQ15[k] = *(WebRtcIsacfix_kRcLevPtr[k] + index[k]);
- }
-
-
- /* entropy coding of quantization indices */
- status = WebRtcIsacfix_EncHistMulti(streamdata, index, WebRtcIsacfix_kRcCdfPtr, AR_ORDER);
-
- /* If error in WebRtcIsacfix_EncHistMulti(), status will be negative, otherwise 0 */
- return status;
-}
-
-
-/* decode & dequantize squared Gain */
-int WebRtcIsacfix_DecodeGain2(Bitstr_dec *streamdata, WebRtc_Word32 *gainQ10)
-{
- int err;
- WebRtc_Word16 index;
-
- /* entropy decoding of quantization index */
- err = WebRtcIsacfix_DecHistOneStepMulti(
- &index,
- streamdata,
- WebRtcIsacfix_kGainPtr,
- WebRtcIsacfix_kGainInitInd,
- 1);
- /* error check */
- if (err<0) {
- return err;
- }
-
- /* find quantization level */
- *gainQ10 = WebRtcIsacfix_kGain2Lev[index];
-
- return 0;
-}
-
-
-
-/* quantize & code squared Gain */
-int WebRtcIsacfix_EncodeGain2(WebRtc_Word32 *gainQ10, Bitstr_enc *streamdata)
-{
- WebRtc_Word16 index;
- int status = 0;
-
- /* find quantization index */
- index = WebRtcIsacfix_kGainInitInd[0];
- if (*gainQ10 > WebRtcIsacfix_kGain2Bound[index])
- {
- while (*gainQ10 > WebRtcIsacfix_kGain2Bound[index + 1])
- index++;
- }
- else
- {
- while (*gainQ10 < WebRtcIsacfix_kGain2Bound[--index]) ;
- }
-
- /* dequantize */
- *gainQ10 = WebRtcIsacfix_kGain2Lev[index];
-
- /* entropy coding of quantization index */
- status = WebRtcIsacfix_EncHistMulti(streamdata, &index, WebRtcIsacfix_kGainPtr, 1);
-
- /* If error in WebRtcIsacfix_EncHistMulti(), status will be negative, otherwise 0 */
- return status;
-}
-
-
-/* code and decode Pitch Gains and Lags functions */
-
-/* decode & dequantize Pitch Gains */
-int WebRtcIsacfix_DecodePitchGain(Bitstr_dec *streamdata, WebRtc_Word16 *PitchGains_Q12)
-{
- int err;
- WebRtc_Word16 index_comb;
- const WebRtc_UWord16 *pitch_gain_cdf_ptr[1];
-
- /* entropy decoding of quantization indices */
- *pitch_gain_cdf_ptr = WebRtcIsacfix_kPitchGainCdf;
- err = WebRtcIsacfix_DecHistBisectMulti(&index_comb, streamdata, pitch_gain_cdf_ptr, WebRtcIsacfix_kCdfTableSizeGain, 1);
- /* error check, Q_mean_Gain.. tables are of size 144 */
- if ((err<0) || (index_comb<0) || (index_comb>144))
- return -ISAC_RANGE_ERROR_DECODE_PITCH_GAIN;
-
- /* unquantize back to pitch gains by table look-up */
- PitchGains_Q12[0] = WebRtcIsacfix_kPitchGain1[index_comb];
- PitchGains_Q12[1] = WebRtcIsacfix_kPitchGain2[index_comb];
- PitchGains_Q12[2] = WebRtcIsacfix_kPitchGain3[index_comb];
- PitchGains_Q12[3] = WebRtcIsacfix_kPitchGain4[index_comb];
-
- return 0;
-}
-
-
-/* quantize & code Pitch Gains */
-int WebRtcIsacfix_EncodePitchGain(WebRtc_Word16 *PitchGains_Q12, Bitstr_enc *streamdata, ISAC_SaveEncData_t* encData)
-{
- int k,j;
- WebRtc_Word16 SQ15[PITCH_SUBFRAMES];
- WebRtc_Word16 index[3];
- WebRtc_Word16 index_comb;
- const WebRtc_UWord16 *pitch_gain_cdf_ptr[1];
- WebRtc_Word32 CQ17;
- int status = 0;
-
-
- /* get the approximate arcsine (almost linear)*/
- for (k=0; k<PITCH_SUBFRAMES; k++)
- SQ15[k] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(PitchGains_Q12[k],33,2); //Q15
-
-
- /* find quantization index; only for the first three transform coefficients */
- for (k=0; k<3; k++)
- {
- /* transform */
- CQ17=0;
- for (j=0; j<PITCH_SUBFRAMES; j++) {
- CQ17 += WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIsacfix_kTransform[k][j], SQ15[j],10); // Q17
- }
-
- index[k] = (WebRtc_Word16)((CQ17 + 8192)>>14); // Rounding and scaling with stepsize (=1/0.125=8)
-
- /* check that the index is not outside the boundaries of the table */
- if (index[k] < WebRtcIsacfix_kLowerlimiGain[k]) index[k] = WebRtcIsacfix_kLowerlimiGain[k];
- else if (index[k] > WebRtcIsacfix_kUpperlimitGain[k]) index[k] = WebRtcIsacfix_kUpperlimitGain[k];
- index[k] -= WebRtcIsacfix_kLowerlimiGain[k];
- }
-
- /* calculate unique overall index */
- index_comb = (WebRtc_Word16)(WEBRTC_SPL_MUL(WebRtcIsacfix_kMultsGain[0], index[0]) +
- WEBRTC_SPL_MUL(WebRtcIsacfix_kMultsGain[1], index[1]) + index[2]);
-
- /* unquantize back to pitch gains by table look-up */
- // (Y)
- PitchGains_Q12[0] = WebRtcIsacfix_kPitchGain1[index_comb];
- PitchGains_Q12[1] = WebRtcIsacfix_kPitchGain2[index_comb];
- PitchGains_Q12[2] = WebRtcIsacfix_kPitchGain3[index_comb];
- PitchGains_Q12[3] = WebRtcIsacfix_kPitchGain4[index_comb];
-
-
- /* entropy coding of quantization pitch gains */
- *pitch_gain_cdf_ptr = WebRtcIsacfix_kPitchGainCdf;
- status = WebRtcIsacfix_EncHistMulti(streamdata, &index_comb, pitch_gain_cdf_ptr, 1);
- if (status < 0) {
- return status;
- }
-
- /* Save data for creation of multiple bitstreams */
- if (encData != NULL) {
- encData->pitchGain_index[encData->startIdx] = index_comb;
- }
-
- return 0;
-}
-
-
-
-/* Pitch LAG */
-
-
-/* decode & dequantize Pitch Lags */
-int WebRtcIsacfix_DecodePitchLag(Bitstr_dec *streamdata,
- WebRtc_Word16 *PitchGain_Q12,
- WebRtc_Word16 *PitchLags_Q7)
-{
- int k, err;
- WebRtc_Word16 index[PITCH_SUBFRAMES];
- const WebRtc_Word16 *mean_val2Q10, *mean_val4Q10;
-
- const WebRtc_Word16 *lower_limit;
- const WebRtc_UWord16 *init_index;
- const WebRtc_UWord16 *cdf_size;
- const WebRtc_UWord16 **cdf;
-
- WebRtc_Word32 meangainQ12;
- WebRtc_Word32 CQ11, CQ10,tmp32a,tmp32b;
- WebRtc_Word16 shft,tmp16a,tmp16c;
-
- meangainQ12=0;
- for (k = 0; k < 4; k++)
- meangainQ12 += PitchGain_Q12[k];
-
- meangainQ12 = WEBRTC_SPL_RSHIFT_W32(meangainQ12, 2); // Get average
-
- /* voicing classificiation */
- if (meangainQ12 <= 819) { // mean_gain < 0.2
- shft = -1; // StepSize=2.0;
- cdf = WebRtcIsacfix_kPitchLagPtrLo;
- cdf_size = WebRtcIsacfix_kPitchLagSizeLo;
- mean_val2Q10 = WebRtcIsacfix_kMeanLag2Lo;
- mean_val4Q10 = WebRtcIsacfix_kMeanLag4Lo;
- lower_limit = WebRtcIsacfix_kLowerLimitLo;
- init_index = WebRtcIsacfix_kInitIndLo;
- } else if (meangainQ12 <= 1638) { // mean_gain < 0.4
- shft = 0; // StepSize=1.0;
- cdf = WebRtcIsacfix_kPitchLagPtrMid;
- cdf_size = WebRtcIsacfix_kPitchLagSizeMid;
- mean_val2Q10 = WebRtcIsacfix_kMeanLag2Mid;
- mean_val4Q10 = WebRtcIsacfix_kMeanLag4Mid;
- lower_limit = WebRtcIsacfix_kLowerLimitMid;
- init_index = WebRtcIsacfix_kInitIndMid;
- } else {
- shft = 1; // StepSize=0.5;
- cdf = WebRtcIsacfix_kPitchLagPtrHi;
- cdf_size = WebRtcIsacfix_kPitchLagSizeHi;
- mean_val2Q10 = WebRtcIsacfix_kMeanLag2Hi;
- mean_val4Q10 = WebRtcIsacfix_kMeanLag4Hi;
- lower_limit = WebRtcIsacfix_kLowerLimitHi;
- init_index = WebRtcIsacfix_kInitIndHi;
- }
-
- /* entropy decoding of quantization indices */
- err = WebRtcIsacfix_DecHistBisectMulti(index, streamdata, cdf, cdf_size, 1);
- if ((err<0) || (index[0]<0)) // error check
- return -ISAC_RANGE_ERROR_DECODE_PITCH_LAG;
-
- err = WebRtcIsacfix_DecHistOneStepMulti(index+1, streamdata, cdf+1, init_index, 3);
- if (err<0) // error check
- return -ISAC_RANGE_ERROR_DECODE_PITCH_LAG;
-
-
- /* unquantize back to transform coefficients and do the inverse transform: S = T'*C */
- CQ11 = ((WebRtc_Word32)index[0] + lower_limit[0]); // Q0
- CQ11 = WEBRTC_SPL_SHIFT_W32(CQ11,11-shft); // Scale with StepSize, Q11
- for (k=0; k<PITCH_SUBFRAMES; k++) {
- tmp32a = WEBRTC_SPL_MUL_16_32_RSFT11(WebRtcIsacfix_kTransform[0][k], CQ11);
- tmp16a = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32a, 5);
- PitchLags_Q7[k] = tmp16a;
- }
-
- CQ10 = mean_val2Q10[index[1]];
- for (k=0; k<PITCH_SUBFRAMES; k++) {
- tmp32b = (WebRtc_Word32) WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) WebRtcIsacfix_kTransform[1][k], (WebRtc_Word16) CQ10,10);
- tmp16c = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5);
- PitchLags_Q7[k] += tmp16c;
- }
-
- CQ10 = mean_val4Q10[index[3]];
- for (k=0; k<PITCH_SUBFRAMES; k++) {
- tmp32b = (WebRtc_Word32) WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) WebRtcIsacfix_kTransform[3][k], (WebRtc_Word16) CQ10,10);
- tmp16c = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5);
- PitchLags_Q7[k] += tmp16c;
- }
-
- return 0;
-}
-
-
-
-/* quantize & code Pitch Lags */
-int WebRtcIsacfix_EncodePitchLag(WebRtc_Word16 *PitchLagsQ7,WebRtc_Word16 *PitchGain_Q12,
- Bitstr_enc *streamdata, ISAC_SaveEncData_t* encData)
-{
- int k, j;
- WebRtc_Word16 index[PITCH_SUBFRAMES];
- WebRtc_Word32 meangainQ12, CQ17;
- WebRtc_Word32 CQ11, CQ10,tmp32a;
-
- const WebRtc_Word16 *mean_val2Q10,*mean_val4Q10;
- const WebRtc_Word16 *lower_limit, *upper_limit;
- const WebRtc_UWord16 **cdf;
- WebRtc_Word16 shft, tmp16a, tmp16b, tmp16c;
- WebRtc_Word32 tmp32b;
- int status = 0;
-
- /* compute mean pitch gain */
- meangainQ12=0;
- for (k = 0; k < 4; k++)
- meangainQ12 += PitchGain_Q12[k];
-
- meangainQ12 = WEBRTC_SPL_RSHIFT_W32(meangainQ12, 2);
-
- /* Save data for creation of multiple bitstreams */
- if (encData != NULL) {
- encData->meanGain[encData->startIdx] = meangainQ12;
- }
-
- /* voicing classificiation */
- if (meangainQ12 <= 819) { // mean_gain < 0.2
- shft = -1; // StepSize=2.0;
- cdf = WebRtcIsacfix_kPitchLagPtrLo;
- mean_val2Q10 = WebRtcIsacfix_kMeanLag2Lo;
- mean_val4Q10 = WebRtcIsacfix_kMeanLag4Lo;
- lower_limit = WebRtcIsacfix_kLowerLimitLo;
- upper_limit = WebRtcIsacfix_kUpperLimitLo;
- } else if (meangainQ12 <= 1638) { // mean_gain < 0.4
- shft = 0; // StepSize=1.0;
- cdf = WebRtcIsacfix_kPitchLagPtrMid;
- mean_val2Q10 = WebRtcIsacfix_kMeanLag2Mid;
- mean_val4Q10 = WebRtcIsacfix_kMeanLag4Mid;
- lower_limit = WebRtcIsacfix_kLowerLimitMid;
- upper_limit = WebRtcIsacfix_kUpperLimitMid;
- } else {
- shft = 1; // StepSize=0.5;
- cdf = WebRtcIsacfix_kPitchLagPtrHi;
- mean_val2Q10 = WebRtcIsacfix_kMeanLag2Hi;
- mean_val4Q10 = WebRtcIsacfix_kMeanLag4Hi;
- lower_limit = WebRtcIsacfix_kLowerLimitHi;
- upper_limit = WebRtcIsacfix_kUpperLimitHi;
- }
-
- /* find quantization index */
- for (k=0; k<4; k++)
- {
- /* transform */
- CQ17=0;
- for (j=0; j<PITCH_SUBFRAMES; j++)
- CQ17 += WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIsacfix_kTransform[k][j], PitchLagsQ7[j],2); // Q17
-
- CQ17 = WEBRTC_SPL_SHIFT_W32(CQ17,shft); // Scale with StepSize
-
- /* quantize */
- tmp16b = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(CQ17 + 65536, 17 );
- index[k] = tmp16b;
-
- /* check that the index is not outside the boundaries of the table */
- if (index[k] < lower_limit[k]) index[k] = lower_limit[k];
- else if (index[k] > upper_limit[k]) index[k] = upper_limit[k];
- index[k] -= lower_limit[k];
-
- /* Save data for creation of multiple bitstreams */
- if(encData != NULL) {
- encData->pitchIndex[PITCH_SUBFRAMES*encData->startIdx + k] = index[k];
- }
- }
-
- /* unquantize back to transform coefficients and do the inverse transform: S = T'*C */
- CQ11 = (index[0] + lower_limit[0]); // Q0
- CQ11 = WEBRTC_SPL_SHIFT_W32(CQ11,11-shft); // Scale with StepSize, Q11
-
- for (k=0; k<PITCH_SUBFRAMES; k++) {
- tmp32a = WEBRTC_SPL_MUL_16_32_RSFT11(WebRtcIsacfix_kTransform[0][k], CQ11); // Q12
- tmp16a = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32a, 5);// Q7
- PitchLagsQ7[k] = tmp16a;
- }
-
- CQ10 = mean_val2Q10[index[1]];
- for (k=0; k<PITCH_SUBFRAMES; k++) {
- tmp32b = (WebRtc_Word32) WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) WebRtcIsacfix_kTransform[1][k], (WebRtc_Word16) CQ10,10);
- tmp16c = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5); // Q7
- PitchLagsQ7[k] += tmp16c;
- }
-
- CQ10 = mean_val4Q10[index[3]];
- for (k=0; k<PITCH_SUBFRAMES; k++) {
- tmp32b = (WebRtc_Word32) WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) WebRtcIsacfix_kTransform[3][k], (WebRtc_Word16) CQ10,10);
- tmp16c = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32b, 5); // Q7
- PitchLagsQ7[k] += tmp16c;
- }
-
- /* entropy coding of quantization pitch lags */
- status = WebRtcIsacfix_EncHistMulti(streamdata, index, cdf, PITCH_SUBFRAMES);
-
- /* If error in WebRtcIsacfix_EncHistMulti(), status will be negative, otherwise 0 */
- return status;
-}
-
-
-
-/* Routines for inband signaling of bandwitdh estimation */
-/* Histograms based on uniform distribution of indices */
-/* Move global variables later! */
-
-
-/* cdf array for frame length indicator */
-const WebRtc_UWord16 kFrameLenCdf[4] = {
- 0, 21845, 43690, 65535};
-
-/* pointer to cdf array for frame length indicator */
-const WebRtc_UWord16 *kFrameLenCdfPtr[1] = {kFrameLenCdf};
-
-/* initial cdf index for decoder of frame length indicator */
-const WebRtc_UWord16 kFrameLenInitIndex[1] = {1};
-
-
-int WebRtcIsacfix_DecodeFrameLen(Bitstr_dec *streamdata,
- WebRtc_Word16 *framesamples)
-{
-
- int err;
- WebRtc_Word16 frame_mode;
-
- err = 0;
- /* entropy decoding of frame length [1:30ms,2:60ms] */
- err = WebRtcIsacfix_DecHistOneStepMulti(&frame_mode, streamdata, kFrameLenCdfPtr, kFrameLenInitIndex, 1);
- if (err<0) // error check
- return -ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH;
-
- switch(frame_mode) {
- case 1:
- *framesamples = 480; /* 30ms */
- break;
- case 2:
- *framesamples = 960; /* 60ms */
- break;
- default:
- err = -ISAC_DISALLOWED_FRAME_MODE_DECODER;
- }
-
- return err;
-}
-
-
-int WebRtcIsacfix_EncodeFrameLen(WebRtc_Word16 framesamples, Bitstr_enc *streamdata) {
-
- int status;
- WebRtc_Word16 frame_mode;
-
- status = 0;
- frame_mode = 0;
- /* entropy coding of frame length [1:480 samples,2:960 samples] */
- switch(framesamples) {
- case 480:
- frame_mode = 1;
- break;
- case 960:
- frame_mode = 2;
- break;
- default:
- status = - ISAC_DISALLOWED_FRAME_MODE_ENCODER;
- }
-
- if (status < 0)
- return status;
-
- status = WebRtcIsacfix_EncHistMulti(streamdata, &frame_mode, kFrameLenCdfPtr, 1);
-
- return status;
-}
-
-/* cdf array for estimated bandwidth */
-const WebRtc_UWord16 kBwCdf[25] = {
- 0, 2731, 5461, 8192, 10923, 13653, 16384, 19114, 21845, 24576, 27306, 30037,
- 32768, 35498, 38229, 40959, 43690, 46421, 49151, 51882, 54613, 57343, 60074,
- 62804, 65535};
-
-/* pointer to cdf array for estimated bandwidth */
-const WebRtc_UWord16 *kBwCdfPtr[1] = {kBwCdf};
-
-/* initial cdf index for decoder of estimated bandwidth*/
-const WebRtc_UWord16 kBwInitIndex[1] = {7};
-
-
-int WebRtcIsacfix_DecodeSendBandwidth(Bitstr_dec *streamdata, WebRtc_Word16 *BWno) {
-
- int err;
- WebRtc_Word16 BWno32;
-
- /* entropy decoding of sender's BW estimation [0..23] */
- err = WebRtcIsacfix_DecHistOneStepMulti(&BWno32, streamdata, kBwCdfPtr, kBwInitIndex, 1);
- if (err<0) // error check
- return -ISAC_RANGE_ERROR_DECODE_BANDWIDTH;
- *BWno = (WebRtc_Word16)BWno32;
- return err;
-
-}
-
-
-int WebRtcIsacfix_EncodeReceiveBandwidth(WebRtc_Word16 *BWno, Bitstr_enc *streamdata)
-{
- int status = 0;
- /* entropy encoding of receiver's BW estimation [0..23] */
- status = WebRtcIsacfix_EncHistMulti(streamdata, BWno, kBwCdfPtr, 1);
-
- return status;
-}
-
-/* estimate codel length of LPC Coef */
-void WebRtcIsacfix_TranscodeLpcCoef(WebRtc_Word32 *gain_lo_hiQ17,
- WebRtc_Word16 *index_gQQ) {
- int j, k, n;
- WebRtc_Word16 posQQ, pos2QQ;
- WebRtc_Word16 pos, pos2, posg, offsg, offs2, gainpos;
- WebRtc_Word32 tmpcoeffs_gQ6[KLT_ORDER_GAIN];
- WebRtc_Word32 tmpcoeffs_gQ17[KLT_ORDER_GAIN];
- WebRtc_Word32 tmpcoeffs2_gQ21[KLT_ORDER_GAIN];
- WebRtc_Word32 sumQQ;
-
-
- /* log gains, mean removal and scaling */
- posg = 0;pos=0; gainpos=0;
-
- for (k=0; k<SUBFRAMES; k++) {
- /* log gains */
-
- /* The input argument X to logN(X) is 2^17 times higher than the
- input floating point argument Y to log(Y), since the X value
- is a Q17 value. This can be compensated for after the call, by
- subraction a value Z for each Q-step. One Q-step means that
- X gets 2 times higher, i.e. Z = logN(2)*256 = 0.693147180559*256 =
- 177.445678 should be subtracted (since logN() returns a Q8 value).
- For a X value in Q17, the value 177.445678*17 = 3017 should be
- subtracted */
- tmpcoeffs_gQ6[posg] = CalcLogN(gain_lo_hiQ17[gainpos])-3017; //Q8
- tmpcoeffs_gQ6[posg] -= WebRtcIsacfix_kMeansGainQ8[0][posg]; //Q8, but Q6 after not-needed mult. by 4
- posg++; gainpos++;
-
- tmpcoeffs_gQ6[posg] = CalcLogN(gain_lo_hiQ17[gainpos])-3017; //Q8
- tmpcoeffs_gQ6[posg] -= WebRtcIsacfix_kMeansGainQ8[0][posg]; //Q8, but Q6 after not-needed mult. by 4
- posg++; gainpos++;
-
- }
-
-
- /* KLT */
-
- /* left transform */
- offsg = 0;
- for (j=0; j<SUBFRAMES; j++) {
- posg = offsg;
- for (k=0; k<2; k++) {
- sumQQ = 0;
- pos = offsg;
- pos2 = k;
- for (n=0; n<2; n++) {
- sumQQ += WEBRTC_SPL_MUL_16_16(tmpcoeffs_gQ6[pos], WebRtcIsacfix_kT1GainQ15[0][pos2]); //Q21 = Q6*Q15
- pos++;
- pos2 += 2;
- }
- tmpcoeffs2_gQ21[posg] = sumQQ;
- posg++;
- }
-
- offsg += 2;
- }
-
- /* right transform */
- offsg = 0;
- offs2 = 0;
- for (j=0; j<SUBFRAMES; j++) {
- posg = offsg;
- for (k=0; k<2; k++) {
- sumQQ = 0;
- pos = k;
- pos2 = offs2;
- for (n=0; n<SUBFRAMES; n++) {
- sumQQ += WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(WebRtcIsacfix_kT2GainQ15[0][pos2], tmpcoeffs2_gQ21[pos]), 1); // (Q15*Q21)>>(16-1) = Q21
- pos += 2;
- pos2++;
- }
- tmpcoeffs_gQ17[posg] = WEBRTC_SPL_RSHIFT_W32(sumQQ, 4);
- posg++;
- }
- offsg += 2;
- offs2 += SUBFRAMES;
- }
-
- /* quantize coefficients */
- for (k=0; k<KLT_ORDER_GAIN; k++) //ATTN: ok?
- {
- posQQ = WebRtcIsacfix_kSelIndGain[k];
- pos2QQ= (WebRtc_Word16)CalcLrIntQ(tmpcoeffs_gQ17[posQQ], 17);
-
- index_gQQ[k] = pos2QQ + WebRtcIsacfix_kQuantMinGain[k]; //ATTN: ok?
- if (index_gQQ[k] < 0) {
- index_gQQ[k] = 0;
- }
- else if (index_gQQ[k] > WebRtcIsacfix_kMaxIndGain[k]) {
- index_gQQ[k] = WebRtcIsacfix_kMaxIndGain[k];
- }
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/entropy_coding.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/entropy_coding.h
deleted file mode 100644
index 298ea22..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/entropy_coding.h
+++ /dev/null
@@ -1,111 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * entropy_coding.h
- *
- * This header file contains all of the functions used to arithmetically
- * encode the iSAC bistream
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_
-
-#include "structs.h"
-
-/* decode complex spectrum (return number of bytes in stream) */
-WebRtc_Word16 WebRtcIsacfix_DecodeSpec(Bitstr_dec *streamdata,
- WebRtc_Word16 *frQ7,
- WebRtc_Word16 *fiQ7,
- WebRtc_Word16 AvgPitchGain_Q12);
-
-/* encode complex spectrum */
-int WebRtcIsacfix_EncodeSpec(const WebRtc_Word16 *fr,
- const WebRtc_Word16 *fi,
- Bitstr_enc *streamdata,
- WebRtc_Word16 AvgPitchGain_Q12);
-
-
-/* decode & dequantize LPC Coef */
-int WebRtcIsacfix_DecodeLpcCoef(Bitstr_dec *streamdata,
- WebRtc_Word32 *LPCCoefQ17,
- WebRtc_Word32 *gain_lo_hiQ17,
- WebRtc_Word16 *outmodel);
-
-int WebRtcIsacfix_DecodeLpc(WebRtc_Word32 *gain_lo_hiQ17,
- WebRtc_Word16 *LPCCoef_loQ15,
- WebRtc_Word16 *LPCCoef_hiQ15,
- Bitstr_dec *streamdata,
- WebRtc_Word16 *outmodel);
-
-/* quantize & code LPC Coef */
-int WebRtcIsacfix_EncodeLpc(WebRtc_Word32 *gain_lo_hiQ17,
- WebRtc_Word16 *LPCCoef_loQ15,
- WebRtc_Word16 *LPCCoef_hiQ15,
- WebRtc_Word16 *model,
- WebRtc_Word32 *sizeQ11,
- Bitstr_enc *streamdata,
- ISAC_SaveEncData_t* encData,
- transcode_obj *transcodeParam);
-
-int WebRtcIsacfix_EstCodeLpcGain(WebRtc_Word32 *gain_lo_hiQ17,
- Bitstr_enc *streamdata,
- ISAC_SaveEncData_t* encData);
-/* decode & dequantize RC */
-int WebRtcIsacfix_DecodeRcCoef(Bitstr_dec *streamdata,
- WebRtc_Word16 *RCQ15);
-
-/* quantize & code RC */
-int WebRtcIsacfix_EncodeRcCoef(WebRtc_Word16 *RCQ15,
- Bitstr_enc *streamdata);
-
-/* decode & dequantize squared Gain */
-int WebRtcIsacfix_DecodeGain2(Bitstr_dec *streamdata,
- WebRtc_Word32 *Gain2);
-
-/* quantize & code squared Gain (input is squared gain) */
-int WebRtcIsacfix_EncodeGain2(WebRtc_Word32 *gain2,
- Bitstr_enc *streamdata);
-
-int WebRtcIsacfix_EncodePitchGain(WebRtc_Word16 *PitchGains_Q12,
- Bitstr_enc *streamdata,
- ISAC_SaveEncData_t* encData);
-
-int WebRtcIsacfix_EncodePitchLag(WebRtc_Word16 *PitchLagQ7,
- WebRtc_Word16 *PitchGain_Q12,
- Bitstr_enc *streamdata,
- ISAC_SaveEncData_t* encData);
-
-int WebRtcIsacfix_DecodePitchGain(Bitstr_dec *streamdata,
- WebRtc_Word16 *PitchGain_Q12);
-
-int WebRtcIsacfix_DecodePitchLag(Bitstr_dec *streamdata,
- WebRtc_Word16 *PitchGain_Q12,
- WebRtc_Word16 *PitchLagQ7);
-
-int WebRtcIsacfix_DecodeFrameLen(Bitstr_dec *streamdata,
- WebRtc_Word16 *framelength);
-
-
-int WebRtcIsacfix_EncodeFrameLen(WebRtc_Word16 framelength,
- Bitstr_enc *streamdata);
-
-int WebRtcIsacfix_DecodeSendBandwidth(Bitstr_dec *streamdata,
- WebRtc_Word16 *BWno);
-
-
-int WebRtcIsacfix_EncodeReceiveBandwidth(WebRtc_Word16 *BWno,
- Bitstr_enc *streamdata);
-
-void WebRtcIsacfix_TranscodeLpcCoef(WebRtc_Word32 *tmpcoeffs_gQ6,
- WebRtc_Word16 *index_gQQ);
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_ENTROPY_CODING_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/fft.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/fft.c
deleted file mode 100644
index fff35c4..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/fft.c
+++ /dev/null
@@ -1,415 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * fft.c
- *
- * Fast Fourier Transform
- *
- */
-
-
-#include "fft.h"
-
-const WebRtc_Word16 kSortTabFft[240] = {
- 0, 60, 120, 180, 20, 80, 140, 200, 40, 100, 160, 220,
- 4, 64, 124, 184, 24, 84, 144, 204, 44, 104, 164, 224,
- 8, 68, 128, 188, 28, 88, 148, 208, 48, 108, 168, 228,
- 12, 72, 132, 192, 32, 92, 152, 212, 52, 112, 172, 232,
- 16, 76, 136, 196, 36, 96, 156, 216, 56, 116, 176, 236,
- 1, 61, 121, 181, 21, 81, 141, 201, 41, 101, 161, 221,
- 5, 65, 125, 185, 25, 85, 145, 205, 45, 105, 165, 225,
- 9, 69, 129, 189, 29, 89, 149, 209, 49, 109, 169, 229,
- 13, 73, 133, 193, 33, 93, 153, 213, 53, 113, 173, 233,
- 17, 77, 137, 197, 37, 97, 157, 217, 57, 117, 177, 237,
- 2, 62, 122, 182, 22, 82, 142, 202, 42, 102, 162, 222,
- 6, 66, 126, 186, 26, 86, 146, 206, 46, 106, 166, 226,
- 10, 70, 130, 190, 30, 90, 150, 210, 50, 110, 170, 230,
- 14, 74, 134, 194, 34, 94, 154, 214, 54, 114, 174, 234,
- 18, 78, 138, 198, 38, 98, 158, 218, 58, 118, 178, 238,
- 3, 63, 123, 183, 23, 83, 143, 203, 43, 103, 163, 223,
- 7, 67, 127, 187, 27, 87, 147, 207, 47, 107, 167, 227,
- 11, 71, 131, 191, 31, 91, 151, 211, 51, 111, 171, 231,
- 15, 75, 135, 195, 35, 95, 155, 215, 55, 115, 175, 235,
- 19, 79, 139, 199, 39, 99, 159, 219, 59, 119, 179, 239
-};
-
-/* Cosine table in Q14 */
-const WebRtc_Word16 kCosTabFfftQ14[240] = {
- 16384, 16378, 16362, 16333, 16294, 16244, 16182, 16110, 16026, 15931, 15826, 15709,
- 15582, 15444, 15296, 15137, 14968, 14788, 14598, 14399, 14189, 13970, 13741, 13502,
- 13255, 12998, 12733, 12458, 12176, 11885, 11585, 11278, 10963, 10641, 10311, 9974,
- 9630, 9280, 8923, 8561, 8192, 7818, 7438, 7053, 6664, 6270, 5872, 5469,
- 5063, 4653, 4240, 3825, 3406, 2986, 2563, 2139, 1713, 1285, 857, 429,
- 0, -429, -857, -1285, -1713, -2139, -2563, -2986, -3406, -3825, -4240, -4653,
- -5063, -5469, -5872, -6270, -6664, -7053, -7438, -7818, -8192, -8561, -8923, -9280,
- -9630, -9974, -10311, -10641, -10963, -11278, -11585, -11885, -12176, -12458, -12733, -12998,
- -13255, -13502, -13741, -13970, -14189, -14399, -14598, -14788, -14968, -15137, -15296, -15444,
- -15582, -15709, -15826, -15931, -16026, -16110, -16182, -16244, -16294, -16333, -16362, -16378,
- -16384, -16378, -16362, -16333, -16294, -16244, -16182, -16110, -16026, -15931, -15826, -15709,
- -15582, -15444, -15296, -15137, -14968, -14788, -14598, -14399, -14189, -13970, -13741, -13502,
- -13255, -12998, -12733, -12458, -12176, -11885, -11585, -11278, -10963, -10641, -10311, -9974,
- -9630, -9280, -8923, -8561, -8192, -7818, -7438, -7053, -6664, -6270, -5872, -5469,
- -5063, -4653, -4240, -3825, -3406, -2986, -2563, -2139, -1713, -1285, -857, -429,
- 0, 429, 857, 1285, 1713, 2139, 2563, 2986, 3406, 3825, 4240, 4653,
- 5063, 5469, 5872, 6270, 6664, 7053, 7438, 7818, 8192, 8561, 8923, 9280,
- 9630, 9974, 10311, 10641, 10963, 11278, 11585, 11885, 12176, 12458, 12733, 12998,
- 13255, 13502, 13741, 13970, 14189, 14399, 14598, 14788, 14968, 15137, 15296, 15444,
- 15582, 15709, 15826, 15931, 16026, 16110, 16182, 16244, 16294, 16333, 16362, 16378
-};
-
-
-
-/* Uses 16x16 mul, without rounding, which is faster. Uses WEBRTC_SPL_MUL_16_16_RSFT */
-WebRtc_Word16 WebRtcIsacfix_FftRadix16Fastest(WebRtc_Word16 RexQx[], WebRtc_Word16 ImxQx[], WebRtc_Word16 iSign) {
-
- WebRtc_Word16 dd, ee, ff, gg, hh, ii;
- WebRtc_Word16 k0, k1, k2, k3, k4, kk;
- WebRtc_Word16 tmp116, tmp216;
-
- WebRtc_Word16 ccc1Q14, ccc2Q14, ccc3Q14, sss1Q14, sss2Q14, sss3Q14;
- WebRtc_Word16 sss60Q14, ccc72Q14, sss72Q14;
- WebRtc_Word16 aaQx, ajQx, akQx, ajmQx, ajpQx, akmQx, akpQx;
- WebRtc_Word16 bbQx, bjQx, bkQx, bjmQx, bjpQx, bkmQx, bkpQx;
-
- WebRtc_Word16 ReDATAQx[240], ImDATAQx[240];
-
- sss60Q14 = kCosTabFfftQ14[20];
- ccc72Q14 = kCosTabFfftQ14[48];
- sss72Q14 = kCosTabFfftQ14[12];
-
- if (iSign < 0) {
- sss72Q14 = -sss72Q14;
- sss60Q14 = -sss60Q14;
- }
- /* Complexity is: 10 cycles */
-
- /* compute fourier transform */
-
- // transform for factor of 4
- for (kk=0; kk<60; kk++) {
- k0 = kk;
- k1 = k0 + 60;
- k2 = k1 + 60;
- k3 = k2 + 60;
-
- akpQx = RexQx[k0] + RexQx[k2];
- akmQx = RexQx[k0] - RexQx[k2];
- ajpQx = RexQx[k1] + RexQx[k3];
- ajmQx = RexQx[k1] - RexQx[k3];
- bkpQx = ImxQx[k0] + ImxQx[k2];
- bkmQx = ImxQx[k0] - ImxQx[k2];
- bjpQx = ImxQx[k1] + ImxQx[k3];
- bjmQx = ImxQx[k1] - ImxQx[k3];
-
- RexQx[k0] = akpQx + ajpQx;
- ImxQx[k0] = bkpQx + bjpQx;
- ajpQx = akpQx - ajpQx;
- bjpQx = bkpQx - bjpQx;
- if (iSign < 0) {
- akpQx = akmQx + bjmQx;
- bkpQx = bkmQx - ajmQx;
- akmQx -= bjmQx;
- bkmQx += ajmQx;
- } else {
- akpQx = akmQx - bjmQx;
- bkpQx = bkmQx + ajmQx;
- akmQx += bjmQx;
- bkmQx -= ajmQx;
- }
-
- ccc1Q14 = kCosTabFfftQ14[kk];
- ccc2Q14 = kCosTabFfftQ14[WEBRTC_SPL_MUL_16_16(2, kk)];
- ccc3Q14 = kCosTabFfftQ14[WEBRTC_SPL_MUL_16_16(3, kk)];
- sss1Q14 = kCosTabFfftQ14[kk+60];
- sss2Q14 = kCosTabFfftQ14[WEBRTC_SPL_MUL_16_16(2, kk)+60];
- sss3Q14 = kCosTabFfftQ14[WEBRTC_SPL_MUL_16_16(3, kk)+60];
- if (iSign==1) {
- sss1Q14 = -sss1Q14;
- sss2Q14 = -sss2Q14;
- sss3Q14 = -sss3Q14;
- }
-
- //Do several multiplications like Q14*Q16>>14 = Q16
- // RexQ16[k1] = akpQ16 * ccc1Q14 - bkpQ16 * sss1Q14;
- // RexQ16[k2] = ajpQ16 * ccc2Q14 - bjpQ16 * sss2Q14;
- // RexQ16[k3] = akmQ16 * ccc3Q14 - bkmQ16 * sss3Q14;
- // ImxQ16[k1] = akpQ16 * sss1Q14 + bkpQ16 * ccc1Q14;
- // ImxQ16[k2] = ajpQ16 * sss2Q14 + bjpQ16 * ccc2Q14;
- // ImxQ16[k3] = akmQ16 * sss3Q14 + bkmQ16 * ccc3Q14;
-
- RexQx[k1] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc1Q14, akpQx, 14) -
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss1Q14, bkpQx, 14); // 6 non-mul + 2 mul cycles, i.e. 8 cycles (6+2*7=20 cycles if 16x32mul)
- RexQx[k2] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, ajpQx, 14) -
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bjpQx, 14);
- RexQx[k3] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc3Q14, akmQx, 14) -
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss3Q14, bkmQx, 14);
- ImxQx[k1] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss1Q14, akpQx, 14) +
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc1Q14, bkpQx, 14);
- ImxQx[k2] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, ajpQx, 14) +
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bjpQx, 14);
- ImxQx[k3] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss3Q14, akmQx, 14) +
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc3Q14, bkmQx, 14);
- //This mul segment needs 6*8 = 48 cycles for 16x16 muls, but 6*20 = 120 cycles for 16x32 muls
-
-
- }
- /* Complexity is: 51+48 = 99 cycles for 16x16 muls, but 51+120 = 171 cycles for 16x32 muls*/
-
- // transform for factor of 3
- kk=0;
- k1=20;
- k2=40;
-
- for (hh=0; hh<4; hh++) {
- for (ii=0; ii<20; ii++) {
- akQx = RexQx[kk];
- bkQx = ImxQx[kk];
- ajQx = RexQx[k1] + RexQx[k2];
- bjQx = ImxQx[k1] + ImxQx[k2];
- RexQx[kk] = akQx + ajQx;
- ImxQx[kk] = bkQx + bjQx;
- tmp116 = WEBRTC_SPL_RSHIFT_W16(ajQx, 1);
- tmp216 = WEBRTC_SPL_RSHIFT_W16(bjQx, 1);
- akQx = akQx - tmp116;
- bkQx = bkQx - tmp216;
- tmp116 = RexQx[k1] - RexQx[k2];
- tmp216 = ImxQx[k1] - ImxQx[k2];
-
- ajQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss60Q14, tmp116, 14); // Q14*Qx>>14 = Qx
- bjQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss60Q14, tmp216, 14); // Q14*Qx>>14 = Qx
- RexQx[k1] = akQx - bjQx;
- RexQx[k2] = akQx + bjQx;
- ImxQx[k1] = bkQx + ajQx;
- ImxQx[k2] = bkQx - ajQx;
-
- kk++;
- k1++;
- k2++;
- }
- /* Complexity : (31+6)*20 = 740 cycles for 16x16 muls, but (31+18)*20 = 980 cycles for 16x32 muls*/
- kk=kk+40;
- k1=k1+40;
- k2=k2+40;
- }
- /* Complexity : 4*(740+3) = 2972 cycles for 16x16 muls, but 4*(980+3) = 3932 cycles for 16x32 muls*/
-
- /* multiply by rotation factor for odd factor 3 or 5 (not for 4)
- Same code (duplicated) for both ii=2 and ii=3 */
- kk = 1;
- ee = 0;
- ff = 0;
-
- for (gg=0; gg<19; gg++) {
- kk += 20;
- ff = ff+4;
- for (hh=0; hh<2; hh++) {
- ee = ff + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(hh, ff);
- dd = ee + 60;
- ccc2Q14 = kCosTabFfftQ14[ee];
- sss2Q14 = kCosTabFfftQ14[dd];
- if (iSign==1) {
- sss2Q14 = -sss2Q14;
- }
- for (ii=0; ii<4; ii++) {
- akQx = RexQx[kk];
- bkQx = ImxQx[kk];
- RexQx[kk] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, akQx, 14) - // Q14*Qx>>14 = Qx
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bkQx, 14);
- ImxQx[kk] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, akQx, 14) + // Q14*Qx>>14 = Qx
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bkQx, 14);
-
-
- kk += 60;
- }
- kk = kk - 220;
- }
- // Complexity: 2*(13+5+4*13+2) = 144 for 16x16 muls, but 2*(13+5+4*33+2) = 304 cycles for 16x32 muls
- kk = kk - 59;
- }
- // Complexity: 19*144 = 2736 for 16x16 muls, but 19*304 = 5776 cycles for 16x32 muls
-
- // transform for factor of 5
- kk = 0;
- ccc2Q14 = kCosTabFfftQ14[96];
- sss2Q14 = kCosTabFfftQ14[84];
- if (iSign==1) {
- sss2Q14 = -sss2Q14;
- }
-
- for (hh=0; hh<4; hh++) {
- for (ii=0; ii<12; ii++) {
- k1 = kk + 4;
- k2 = k1 + 4;
- k3 = k2 + 4;
- k4 = k3 + 4;
-
- akpQx = RexQx[k1] + RexQx[k4];
- akmQx = RexQx[k1] - RexQx[k4];
- bkpQx = ImxQx[k1] + ImxQx[k4];
- bkmQx = ImxQx[k1] - ImxQx[k4];
- ajpQx = RexQx[k2] + RexQx[k3];
- ajmQx = RexQx[k2] - RexQx[k3];
- bjpQx = ImxQx[k2] + ImxQx[k3];
- bjmQx = ImxQx[k2] - ImxQx[k3];
- aaQx = RexQx[kk];
- bbQx = ImxQx[kk];
- RexQx[kk] = aaQx + akpQx + ajpQx;
- ImxQx[kk] = bbQx + bkpQx + bjpQx;
-
- akQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, akpQx, 14) +
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, ajpQx, 14) + aaQx;
- bkQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, bkpQx, 14) +
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bjpQx, 14) + bbQx;
- ajQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, akmQx, 14) +
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, ajmQx, 14);
- bjQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, bkmQx, 14) +
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bjmQx, 14);
- // 32+4*8=64 or 32+4*20=112
-
- RexQx[k1] = akQx - bjQx;
- RexQx[k4] = akQx + bjQx;
- ImxQx[k1] = bkQx + ajQx;
- ImxQx[k4] = bkQx - ajQx;
-
- akQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, akpQx, 14) +
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, ajpQx, 14) + aaQx;
- bkQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bkpQx, 14) +
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc72Q14, bjpQx, 14) + bbQx;
- ajQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, akmQx, 14) -
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, ajmQx, 14);
- bjQx = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bkmQx, 14) -
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss72Q14, bjmQx, 14);
- // 8+4*8=40 or 8+4*20=88
-
- RexQx[k2] = akQx - bjQx;
- RexQx[k3] = akQx + bjQx;
- ImxQx[k2] = bkQx + ajQx;
- ImxQx[k3] = bkQx - ajQx;
-
- kk = k4 + 4;
- }
- // Complexity: 12*(64+40+10) = 1368 for 16x16 muls, but 12*(112+88+10) = 2520 cycles for 16x32 muls
- kk -= 239;
- }
- // Complexity: 4*1368 = 5472 for 16x16 muls, but 4*2520 = 10080 cycles for 16x32 muls
-
- /* multiply by rotation factor for odd factor 3 or 5 (not for 4)
- Same code (duplicated) for both ii=2 and ii=3 */
- kk = 1;
- ee=0;
-
- for (gg=0; gg<3; gg++) {
- kk += 4;
- dd = 12 + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(12, gg);
- ff = 0;
- for (hh=0; hh<4; hh++) {
- ff = ff+dd;
- ee = ff+60;
- for (ii=0; ii<12; ii++) {
- akQx = RexQx[kk];
- bkQx = ImxQx[kk];
-
- ccc2Q14 = kCosTabFfftQ14[ff];
- sss2Q14 = kCosTabFfftQ14[ee];
-
- if (iSign==1) {
- sss2Q14 = -sss2Q14;
- }
-
- RexQx[kk] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, akQx, 14) -
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, bkQx, 14);
- ImxQx[kk] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(sss2Q14, akQx, 14) +
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(ccc2Q14, bkQx, 14);
-
- kk += 20;
- }
- kk = kk - 236;
- // Complexity: 12*(12+12) = 288 for 16x16 muls, but 12*(12+32) = 528 cycles for 16x32 muls
- }
- kk = kk - 19;
- // Complexity: 4*288+6 for 16x16 muls, but 4*528+6 cycles for 16x32 muls
- }
- // Complexity: 3*4*288+6 = 3462 for 16x16 muls, but 3*4*528+6 = 6342 cycles for 16x32 muls
-
-
- // last transform for factor of 4 */
- for (kk=0; kk<240; kk=kk+4) {
- k1 = kk + 1;
- k2 = k1 + 1;
- k3 = k2 + 1;
-
- akpQx = RexQx[kk] + RexQx[k2];
- akmQx = RexQx[kk] - RexQx[k2];
- ajpQx = RexQx[k1] + RexQx[k3];
- ajmQx = RexQx[k1] - RexQx[k3];
- bkpQx = ImxQx[kk] + ImxQx[k2];
- bkmQx = ImxQx[kk] - ImxQx[k2];
- bjpQx = ImxQx[k1] + ImxQx[k3];
- bjmQx = ImxQx[k1] - ImxQx[k3];
- RexQx[kk] = akpQx + ajpQx;
- ImxQx[kk] = bkpQx + bjpQx;
- ajpQx = akpQx - ajpQx;
- bjpQx = bkpQx - bjpQx;
- if (iSign < 0) {
- akpQx = akmQx + bjmQx;
- bkpQx = bkmQx - ajmQx;
- akmQx -= bjmQx;
- bkmQx += ajmQx;
- } else {
- akpQx = akmQx - bjmQx;
- bkpQx = bkmQx + ajmQx;
- akmQx += bjmQx;
- bkmQx -= ajmQx;
- }
- RexQx[k1] = akpQx;
- RexQx[k2] = ajpQx;
- RexQx[k3] = akmQx;
- ImxQx[k1] = bkpQx;
- ImxQx[k2] = bjpQx;
- ImxQx[k3] = bkmQx;
- }
- // Complexity: 60*45 = 2700 for 16x16 muls, but 60*45 = 2700 cycles for 16x32 muls
-
- /* permute the results to normal order */
- for (ii=0; ii<240; ii++) {
- ReDATAQx[ii]=RexQx[ii];
- ImDATAQx[ii]=ImxQx[ii];
- }
- // Complexity: 240*2=480 cycles
-
- for (ii=0; ii<240; ii++) {
- RexQx[ii]=ReDATAQx[kSortTabFft[ii]];
- ImxQx[ii]=ImDATAQx[kSortTabFft[ii]];
- }
- // Complexity: 240*2*2=960 cycles
-
- // Total complexity:
- // 16x16 16x32
- // Complexity: 10 10
- // Complexity: 99 171
- // Complexity: 2972 3932
- // Complexity: 2736 5776
- // Complexity: 5472 10080
- // Complexity: 3462 6342
- // Complexity: 2700 2700
- // Complexity: 480 480
- // Complexity: 960 960
- // =======================
- // 18891 30451
- //
- // If this FFT is called 2 time each frame, i.e. 67 times per second, it will correspond to
- // a C54 complexity of 67*18891/1000000 = 1.27 MIPS with 16x16-muls, and 67*30451/1000000 =
- // = 2.04 MIPS with 16x32-muls. Note that this routine somtimes is called 6 times during the
- // encoding of a frame, i.e. the max complexity would be 7/2*1.27 = 4.4 MIPS for the 16x16 mul case.
-
-
- return 0;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/fft.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/fft.h
deleted file mode 100644
index efa116e..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/fft.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*--------------------------------*-C-*---------------------------------*
- * File:
- * fft.h
- * ---------------------------------------------------------------------*
- * Re[]: real value array
- * Im[]: imaginary value array
- * nTotal: total number of complex values
- * nPass: number of elements involved in this pass of transform
- * nSpan: nspan/nPass = number of bytes to increment pointer
- * in Re[] and Im[]
- * isign: exponent: +1 = forward -1 = reverse
- * scaling: normalizing constant by which the final result is *divided*
- * scaling == -1, normalize by total dimension of the transform
- * scaling < -1, normalize by the square-root of the total dimension
- *
- * ----------------------------------------------------------------------
- * See the comments in the code for correct usage!
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_
-
-
-#include "structs.h"
-
-
-WebRtc_Word16 WebRtcIsacfix_FftRadix16Fastest(WebRtc_Word16 RexQx[], WebRtc_Word16 ImxQx[], WebRtc_Word16 iSign);
-
-
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FFT_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbank_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbank_tables.c
deleted file mode 100644
index 87c62aa..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbank_tables.c
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * filterbank_tables.c
- *
- * This file contains variables that are used in
- * filterbanks.c
- *
- */
-
-#include "filterbank_tables.h"
-#include "settings.h"
-
-
-/* HPstcoeff_in_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2};
- * In float, they are:
- * {-1.94895953203325f, 0.94984516000000f, -0.05101826139794f, 0.05015484000000f};
- */
-const WebRtc_Word16 WebRtcIsacfix_kHpStCoeffInQ30[8] = {
- -31932, 16189, /* Q30 hi/lo pair */
- 15562, 17243, /* Q30 hi/lo pair */
- -26748, -17186, /* Q35 hi/lo pair */
- 26296, -27476 /* Q35 hi/lo pair */
-};
-
-/* HPstcoeff_out_1_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2};
- * In float, they are:
- * {-1.99701049409000f, 0.99714204490000f, 0.01701049409000f, -0.01704204490000f};
- */
-const WebRtc_Word16 WebRtcIsacfix_kHPStCoeffOut1Q30[8] = {
- -32719, -1306, /* Q30 hi/lo pair */
- 16337, 11486, /* Q30 hi/lo pair */
- 8918, 26078, /* Q35 hi/lo pair */
- -8935, 3956 /* Q35 hi/lo pair */
-};
-
-/* HPstcoeff_out_2_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2};
- * In float, they are:
- * {-1.98645294509837f, 0.98672435560000f, 0.00645294509837f, -0.00662435560000f};
- */
-const WebRtc_Word16 WebRtcIsacfix_kHPStCoeffOut2Q30[8] = {
- -32546, -2953, /* Q30 hi/lo pair */
- 16166, 32233, /* Q30 hi/lo pair */
- 3383, 13217, /* Q35 hi/lo pair */
- -3473, -4597 /* Q35 hi/lo pair */
-};
-
-/* The upper channel all-pass filter factors */
-const WebRtc_Word16 WebRtcIsacfix_kUpperApFactorsQ15[2] = {
- 1137, 12537
-};
-
-/* The lower channel all-pass filter factors */
-const WebRtc_Word16 WebRtcIsacfix_kLowerApFactorsQ15[2] = {
- 5059, 24379
-};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbank_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbank_tables.h
deleted file mode 100644
index b6be4f0..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbank_tables.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * filterbank_tables.h
- *
- * Header file for variables that are defined in
- * filterbank_tables.c.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_TABLES_H_
-
-#include "typedefs.h"
-
-/********************* Coefficient Tables ************************/
-
-/* HPstcoeff_in_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */
-extern const WebRtc_Word16 WebRtcIsacfix_kHpStCoeffInQ30[8]; /* [Q30hi Q30lo Q30hi Q30lo Q35hi Q35lo Q35hi Q35lo] */
-
-/* HPstcoeff_out_1_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */
-extern const WebRtc_Word16 WebRtcIsacfix_kHPStCoeffOut1Q30[8]; /* [Q30hi Q30lo Q30hi Q30lo Q35hi Q35lo Q35hi Q35lo] */
-
-/* HPstcoeff_out_2_Q14 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */
-extern const WebRtc_Word16 WebRtcIsacfix_kHPStCoeffOut2Q30[8]; /* [Q30hi Q30lo Q30hi Q30lo Q35hi Q35lo Q35hi Q35lo] */
-
-/* The upper channel all-pass filter factors */
-extern const WebRtc_Word16 WebRtcIsacfix_kUpperApFactorsQ15[2];
-
-/* The lower channel all-pass filter factors */
-extern const WebRtc_Word16 WebRtcIsacfix_kLowerApFactorsQ15[2];
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_FILTERBANK_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbanks.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbanks.c
deleted file mode 100644
index a53aa66..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filterbanks.c
+++ /dev/null
@@ -1,326 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * filterbanks.c
- *
- * This file contains function
- * WebRtcIsacfix_SplitAndFilter, and WebRtcIsacfix_FilterAndCombine
- * which implement filterbanks that produce decimated lowpass and
- * highpass versions of a signal, and performs reconstruction.
- *
- */
-
-#include "codec.h"
-#include "filterbank_tables.h"
-#include "settings.h"
-
-
-static void AllpassFilter2FixDec16(WebRtc_Word16 *InOut16, //Q0
- const WebRtc_Word16 *APSectionFactors, //Q15
- WebRtc_Word16 lengthInOut,
- WebRtc_Word16 NumberOfSections,
- WebRtc_Word32 *FilterState) //Q16
-{
- int n, j;
- WebRtc_Word32 a, b;
-
- for (j=0; j<NumberOfSections; j++) {
- for (n=0;n<lengthInOut;n++) {
-
-
- a = WEBRTC_SPL_MUL_16_16(APSectionFactors[j], InOut16[n]); //Q15*Q0=Q15
- a = WEBRTC_SPL_LSHIFT_W32(a, 1); // Q15 -> Q16
- b = WEBRTC_SPL_ADD_SAT_W32(a, FilterState[j]); //Q16+Q16=Q16
- a = WEBRTC_SPL_MUL_16_16_RSFT(-APSectionFactors[j], (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(b, 16), 0); //Q15*Q0=Q15
- FilterState[j] = WEBRTC_SPL_ADD_SAT_W32(WEBRTC_SPL_LSHIFT_W32(a,1), WEBRTC_SPL_LSHIFT_W32((WebRtc_UWord32)InOut16[n],16)); // Q15<<1 + Q0<<16 = Q16 + Q16 = Q16
- InOut16[n] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(b, 16); //Save as Q0
-
- }
- }
-
-}
-
-
-static void HighpassFilterFixDec32(
- WebRtc_Word16 *io, /* Q0:input Q0: Output */
- WebRtc_Word16 len, /* length of input, Input */
- const WebRtc_Word16 *coeff, /* Coeff: [Q30hi Q30lo Q30hi Q30lo Q35hi Q35lo Q35hi Q35lo] */
- WebRtc_Word32 *state) /* Q4:filter state Input/Output */
-{
- int k;
- WebRtc_Word32 a, b, c, in;
-
-
-
- for (k=0; k<len; k++) {
- in = (WebRtc_Word32)io[k];
- /* Q35 * Q4 = Q39 ; shift 32 bit => Q7 */
- a = WEBRTC_SPL_MUL_32_32_RSFT32(coeff[2*2], coeff[2*2+1], state[0]);
- b = WEBRTC_SPL_MUL_32_32_RSFT32(coeff[2*3], coeff[2*3+1], state[1]);
-
- c = ((WebRtc_Word32)in) + WEBRTC_SPL_RSHIFT_W32(a+b, 7); // Q0
- //c = WEBRTC_SPL_RSHIFT_W32(c, 1); // Q-1
- io[k] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(c); // Write output as Q0
-
- /* Q30 * Q4 = Q34 ; shift 32 bit => Q2 */
- a = WEBRTC_SPL_MUL_32_32_RSFT32(coeff[2*0], coeff[2*0+1], state[0]);
- b = WEBRTC_SPL_MUL_32_32_RSFT32(coeff[2*1], coeff[2*1+1], state[1]);
-
- c = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)in, 2) - a - b; // New state in Q2
- c= (WebRtc_Word32)WEBRTC_SPL_SAT((WebRtc_Word32)536870911, c, (WebRtc_Word32)-536870912); // Check for wrap-around
-
- state[1] = state[0];
- state[0] = WEBRTC_SPL_LSHIFT_W32(c, 2); // Write state as Q4
-
- }
-}
-
-
-void WebRtcIsacfix_SplitAndFilter1(WebRtc_Word16 *pin,
- WebRtc_Word16 *LP16,
- WebRtc_Word16 *HP16,
- PreFiltBankstr *prefiltdata)
-{
- /* Function WebRtcIsacfix_SplitAndFilter */
- /* This function creates low-pass and high-pass decimated versions of part of
- the input signal, and part of the signal in the input 'lookahead buffer'. */
-
- int k;
-
- WebRtc_Word16 tempin_ch1[FRAMESAMPLES/2 + QLOOKAHEAD];
- WebRtc_Word16 tempin_ch2[FRAMESAMPLES/2 + QLOOKAHEAD];
- WebRtc_Word32 tmpState[WEBRTC_SPL_MUL_16_16(2,(QORDER-1))]; /* 4 */
-
-
- /* High pass filter */
- HighpassFilterFixDec32(pin, FRAMESAMPLES, WebRtcIsacfix_kHpStCoeffInQ30, prefiltdata->HPstates_fix);
-
-
- /* First Channel */
- for (k=0;k<FRAMESAMPLES/2;k++) {
- tempin_ch1[QLOOKAHEAD + k] = pin[1+WEBRTC_SPL_MUL_16_16(2, k)];
- }
- for (k=0;k<QLOOKAHEAD;k++) {
- tempin_ch1[k]=prefiltdata->INLABUF1_fix[k];
- prefiltdata->INLABUF1_fix[k]=pin[FRAMESAMPLES+1-WEBRTC_SPL_MUL_16_16(2, QLOOKAHEAD)+WEBRTC_SPL_MUL_16_16(2, k)];
- }
-
- /* Second Channel. This is exactly like the first channel, except that the
- even samples are now filtered instead (lower channel). */
- for (k=0;k<FRAMESAMPLES/2;k++) {
- tempin_ch2[QLOOKAHEAD+k] = pin[WEBRTC_SPL_MUL_16_16(2, k)];
- }
- for (k=0;k<QLOOKAHEAD;k++) {
- tempin_ch2[k]=prefiltdata->INLABUF2_fix[k];
- prefiltdata->INLABUF2_fix[k]=pin[FRAMESAMPLES-WEBRTC_SPL_MUL_16_16(2, QLOOKAHEAD)+WEBRTC_SPL_MUL_16_16(2, k)];
- }
-
-
- /*obtain polyphase components by forward all-pass filtering through each channel */
- /* The all pass filtering automatically updates the filter states which are exported in the
- prefiltdata structure */
- AllpassFilter2FixDec16(tempin_ch1,WebRtcIsacfix_kUpperApFactorsQ15, FRAMESAMPLES/2 , NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTAT1_fix);
- AllpassFilter2FixDec16(tempin_ch2,WebRtcIsacfix_kLowerApFactorsQ15, FRAMESAMPLES/2 , NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTAT2_fix);
-
- for (k=0;k<WEBRTC_SPL_MUL_16_16(2, (QORDER-1));k++)
- tmpState[k] = prefiltdata->INSTAT1_fix[k];
- AllpassFilter2FixDec16(tempin_ch1 + FRAMESAMPLES/2,WebRtcIsacfix_kUpperApFactorsQ15, QLOOKAHEAD , NUMBEROFCHANNELAPSECTIONS, tmpState);
- for (k=0;k<WEBRTC_SPL_MUL_16_16(2, (QORDER-1));k++)
- tmpState[k] = prefiltdata->INSTAT2_fix[k];
- AllpassFilter2FixDec16(tempin_ch2 + FRAMESAMPLES/2,WebRtcIsacfix_kLowerApFactorsQ15, QLOOKAHEAD , NUMBEROFCHANNELAPSECTIONS, tmpState);
-
-
- /* Now Construct low-pass and high-pass signals as combinations of polyphase components */
- for (k=0; k<FRAMESAMPLES/2 + QLOOKAHEAD; k++) {
- WebRtc_Word32 tmp1, tmp2, tmp3;
- tmp1 = (WebRtc_Word32)tempin_ch1[k]; // Q0 -> Q0
- tmp2 = (WebRtc_Word32)tempin_ch2[k]; // Q0 -> Q0
- tmp3 = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_W32((tmp1 + tmp2), 1);/* low pass signal*/
- LP16[k] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp3); /*low pass */
- tmp3 = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_W32((tmp1 - tmp2), 1);/* high pass signal*/
- HP16[k] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp3); /*high pass */
- }
-
-}/*end of WebRtcIsacfix_SplitAndFilter */
-
-
-#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-
-/* Without lookahead */
-void WebRtcIsacfix_SplitAndFilter2(WebRtc_Word16 *pin,
- WebRtc_Word16 *LP16,
- WebRtc_Word16 *HP16,
- PreFiltBankstr *prefiltdata)
-{
- /* Function WebRtcIsacfix_SplitAndFilter2 */
- /* This function creates low-pass and high-pass decimated versions of part of
- the input signal. */
-
- int k;
-
- WebRtc_Word16 tempin_ch1[FRAMESAMPLES/2];
- WebRtc_Word16 tempin_ch2[FRAMESAMPLES/2];
-
-
- /* High pass filter */
- HighpassFilterFixDec32(pin, FRAMESAMPLES, WebRtcIsacfix_kHpStCoeffInQ30, prefiltdata->HPstates_fix);
-
-
- /* First Channel */
- for (k=0;k<FRAMESAMPLES/2;k++) {
- tempin_ch1[k] = pin[1+WEBRTC_SPL_MUL_16_16(2, k)];
- }
-
- /* Second Channel. This is exactly like the first channel, except that the
- even samples are now filtered instead (lower channel). */
- for (k=0;k<FRAMESAMPLES/2;k++) {
- tempin_ch2[k] = pin[WEBRTC_SPL_MUL_16_16(2, k)];
- }
-
-
- /*obtain polyphase components by forward all-pass filtering through each channel */
- /* The all pass filtering automatically updates the filter states which are exported in the
- prefiltdata structure */
- AllpassFilter2FixDec16(tempin_ch1,WebRtcIsacfix_kUpperApFactorsQ15, FRAMESAMPLES/2 , NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTAT1_fix);
- AllpassFilter2FixDec16(tempin_ch2,WebRtcIsacfix_kLowerApFactorsQ15, FRAMESAMPLES/2 , NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTAT2_fix);
-
-
- /* Now Construct low-pass and high-pass signals as combinations of polyphase components */
- for (k=0; k<FRAMESAMPLES/2; k++) {
- WebRtc_Word32 tmp1, tmp2, tmp3;
- tmp1 = (WebRtc_Word32)tempin_ch1[k]; // Q0 -> Q0
- tmp2 = (WebRtc_Word32)tempin_ch2[k]; // Q0 -> Q0
- tmp3 = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_W32((tmp1 + tmp2), 1);/* low pass signal*/
- LP16[k] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp3); /*low pass */
- tmp3 = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_W32((tmp1 - tmp2), 1);/* high pass signal*/
- HP16[k] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp3); /*high pass */
- }
-
-}/*end of WebRtcIsacfix_SplitAndFilter */
-
-#endif
-
-
-
-//////////////////////////////////////////////////////////
-////////// Combining
-/* Function WebRtcIsacfix_FilterAndCombine */
-/* This is a decoder function that takes the decimated
- length FRAMESAMPLES/2 input low-pass and
- high-pass signals and creates a reconstructed fullband
- output signal of length FRAMESAMPLES. WebRtcIsacfix_FilterAndCombine
- is the sibling function of WebRtcIsacfix_SplitAndFilter */
-/* INPUTS:
- inLP: a length FRAMESAMPLES/2 array of input low-pass
- samples.
- inHP: a length FRAMESAMPLES/2 array of input high-pass
- samples.
- postfiltdata: input data structure containing the filterbank
- states from the previous decoding iteration.
- OUTPUTS:
- Out: a length FRAMESAMPLES array of output reconstructed
- samples (fullband) based on the input low-pass and
- high-pass signals.
- postfiltdata: the input data structure containing the filterbank
- states is updated for the next decoding iteration */
-void WebRtcIsacfix_FilterAndCombine1(WebRtc_Word16 *tempin_ch1,
- WebRtc_Word16 *tempin_ch2,
- WebRtc_Word16 *out16,
- PostFiltBankstr *postfiltdata)
-{
- int k;
- WebRtc_Word16 in[FRAMESAMPLES];
-
- /* all-pass filter the new upper channel signal. HOWEVER, use the all-pass filter factors
- that were used as a lower channel at the encoding side. So at the decoder, the
- corresponding all-pass filter factors for each channel are swapped.*/
-
- AllpassFilter2FixDec16(tempin_ch1, WebRtcIsacfix_kLowerApFactorsQ15, FRAMESAMPLES/2, NUMBEROFCHANNELAPSECTIONS,postfiltdata->STATE_0_UPPER_fix);
-
- /* Now, all-pass filter the new lower channel signal. But since all-pass filter factors
- at the decoder are swapped from the ones at the encoder, the 'upper' channel
- all-pass filter factors (kUpperApFactors) are used to filter this new lower channel signal */
-
- AllpassFilter2FixDec16(tempin_ch2, WebRtcIsacfix_kUpperApFactorsQ15, FRAMESAMPLES/2, NUMBEROFCHANNELAPSECTIONS,postfiltdata->STATE_0_LOWER_fix);
-
- /* Merge outputs to form the full length output signal.*/
- for (k=0;k<FRAMESAMPLES/2;k++) {
- in[WEBRTC_SPL_MUL_16_16(2, k)]=tempin_ch2[k];
- in[WEBRTC_SPL_MUL_16_16(2, k)+1]=tempin_ch1[k];
- }
-
- /* High pass filter */
- HighpassFilterFixDec32(in, FRAMESAMPLES, WebRtcIsacfix_kHPStCoeffOut1Q30, postfiltdata->HPstates1_fix);
- HighpassFilterFixDec32(in, FRAMESAMPLES, WebRtcIsacfix_kHPStCoeffOut2Q30, postfiltdata->HPstates2_fix);
-
- for (k=0;k<FRAMESAMPLES;k++) {
- out16[k] = in[k];
- }
-}
-
-
-#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-/* Function WebRtcIsacfix_FilterAndCombine */
-/* This is a decoder function that takes the decimated
- length len/2 input low-pass and
- high-pass signals and creates a reconstructed fullband
- output signal of length len. WebRtcIsacfix_FilterAndCombine
- is the sibling function of WebRtcIsacfix_SplitAndFilter */
-/* INPUTS:
- inLP: a length len/2 array of input low-pass
- samples.
- inHP: a length len/2 array of input high-pass
- samples.
- postfiltdata: input data structure containing the filterbank
- states from the previous decoding iteration.
- OUTPUTS:
- Out: a length len array of output reconstructed
- samples (fullband) based on the input low-pass and
- high-pass signals.
- postfiltdata: the input data structure containing the filterbank
- states is updated for the next decoding iteration */
-void WebRtcIsacfix_FilterAndCombine2(WebRtc_Word16 *tempin_ch1,
- WebRtc_Word16 *tempin_ch2,
- WebRtc_Word16 *out16,
- PostFiltBankstr *postfiltdata,
- WebRtc_Word16 len)
-{
- int k;
- WebRtc_Word16 in[FRAMESAMPLES];
-
- /* all-pass filter the new upper channel signal. HOWEVER, use the all-pass filter factors
- that were used as a lower channel at the encoding side. So at the decoder, the
- corresponding all-pass filter factors for each channel are swapped.*/
-
- AllpassFilter2FixDec16(tempin_ch1, WebRtcIsacfix_kLowerApFactorsQ15,(WebRtc_Word16) (len/2), NUMBEROFCHANNELAPSECTIONS,postfiltdata->STATE_0_UPPER_fix);
-
- /* Now, all-pass filter the new lower channel signal. But since all-pass filter factors
- at the decoder are swapped from the ones at the encoder, the 'upper' channel
- all-pass filter factors (kUpperApFactors) are used to filter this new lower channel signal */
-
- AllpassFilter2FixDec16(tempin_ch2, WebRtcIsacfix_kUpperApFactorsQ15, (WebRtc_Word16) (len/2), NUMBEROFCHANNELAPSECTIONS,postfiltdata->STATE_0_LOWER_fix);
-
- /* Merge outputs to form the full length output signal.*/
- for (k=0;k<len/2;k++) {
- in[WEBRTC_SPL_MUL_16_16(2, k)]=tempin_ch2[k];
- in[WEBRTC_SPL_MUL_16_16(2, k)+1]=tempin_ch1[k];
- }
-
- /* High pass filter */
- HighpassFilterFixDec32(in, len, WebRtcIsacfix_kHPStCoeffOut1Q30, postfiltdata->HPstates1_fix);
- HighpassFilterFixDec32(in, len, WebRtcIsacfix_kHPStCoeffOut2Q30, postfiltdata->HPstates2_fix);
-
- for (k=0;k<len;k++) {
- out16[k] = in[k];
- }
-}
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filters.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filters.c
deleted file mode 100644
index 6ee0477..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filters.c
+++ /dev/null
@@ -1,122 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * filters.c
- *
- * This file contains function WebRtcIsacfix_AutocorrC,
- * AllpassFilterForDec32, and WebRtcIsacfix_DecimateAllpass32
- *
- */
-
-#include <string.h>
-
-#include "pitch_estimator.h"
-#include "lpc_masking_model.h"
-#include "codec.h"
-
-// Autocorrelation function in fixed point.
-// NOTE! Different from SPLIB-version in how it scales the signal.
-int WebRtcIsacfix_AutocorrC(WebRtc_Word32* __restrict r,
- const WebRtc_Word16* __restrict x,
- WebRtc_Word16 N,
- WebRtc_Word16 order,
- WebRtc_Word16* __restrict scale) {
- int i = 0;
- int j = 0;
- int16_t scaling = 0;
- int32_t sum = 0;
- uint32_t temp = 0;
- int64_t prod = 0;
-
- // Calculate r[0].
- for (i = 0; i < N; i++) {
- prod += WEBRTC_SPL_MUL_16_16(x[i], x[i]);
- }
-
- // Calculate scaling (the value of shifting).
- temp = (uint32_t)(prod >> 31);
- if(temp == 0) {
- scaling = 0;
- } else {
- scaling = 32 - WebRtcSpl_NormU32(temp);
- }
- r[0] = (int32_t)(prod >> scaling);
-
- // Perform the actual correlation calculation.
- for (i = 1; i < order + 1; i++) {
- prod = 0;
- for (j = 0; j < N - i; j++) {
- prod += WEBRTC_SPL_MUL_16_16(x[j], x[i + j]);
- }
- sum = (int32_t)(prod >> scaling);
- r[i] = sum;
- }
-
- *scale = scaling;
-
- return(order + 1);
-}
-
-static const WebRtc_Word32 kApUpperQ15[ALLPASSSECTIONS] = { 1137, 12537 };
-static const WebRtc_Word32 kApLowerQ15[ALLPASSSECTIONS] = { 5059, 24379 };
-
-
-static void AllpassFilterForDec32(WebRtc_Word16 *InOut16, //Q0
- const WebRtc_Word32 *APSectionFactors, //Q15
- WebRtc_Word16 lengthInOut,
- WebRtc_Word32 *FilterState) //Q16
-{
- int n, j;
- WebRtc_Word32 a, b;
-
- for (j=0; j<ALLPASSSECTIONS; j++) {
- for (n=0;n<lengthInOut;n+=2){
- a = WEBRTC_SPL_MUL_16_32_RSFT16(InOut16[n], APSectionFactors[j]); //Q0*Q31=Q31 shifted 16 gives Q15
- a = WEBRTC_SPL_LSHIFT_W32(a, 1); // Q15 -> Q16
- b = WEBRTC_SPL_ADD_SAT_W32(a, FilterState[j]); //Q16+Q16=Q16
- a = WEBRTC_SPL_MUL_16_32_RSFT16(
- (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(b, 16),
- -APSectionFactors[j]); //Q0*Q31=Q31 shifted 16 gives Q15
- FilterState[j] = WEBRTC_SPL_ADD_SAT_W32(
- WEBRTC_SPL_LSHIFT_W32(a,1),
- WEBRTC_SPL_LSHIFT_W32((WebRtc_UWord32)InOut16[n], 16)); // Q15<<1 + Q0<<16 = Q16 + Q16 = Q16
- InOut16[n] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(b, 16); //Save as Q0
- }
- }
-}
-
-
-
-
-void WebRtcIsacfix_DecimateAllpass32(const WebRtc_Word16 *in,
- WebRtc_Word32 *state_in, /* array of size: 2*ALLPASSSECTIONS+1 */
- WebRtc_Word16 N, /* number of input samples */
- WebRtc_Word16 *out) /* array of size N/2 */
-{
- int n;
- WebRtc_Word16 data_vec[PITCH_FRAME_LEN];
-
- /* copy input */
- memcpy(data_vec+1, in, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), (N-1)));
-
-
- data_vec[0] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(state_in[WEBRTC_SPL_MUL_16_16(2, ALLPASSSECTIONS)],16); //the z^(-1) state
- state_in[WEBRTC_SPL_MUL_16_16(2, ALLPASSSECTIONS)] = WEBRTC_SPL_LSHIFT_W32((WebRtc_UWord32)in[N-1],16);
-
-
-
- AllpassFilterForDec32(data_vec+1, kApUpperQ15, N, state_in);
- AllpassFilterForDec32(data_vec, kApLowerQ15, N, state_in+ALLPASSSECTIONS);
-
- for (n=0;n<N/2;n++) {
- out[n]=WEBRTC_SPL_ADD_SAT_W16(data_vec[WEBRTC_SPL_MUL_16_16(2, n)], data_vec[WEBRTC_SPL_MUL_16_16(2, n)+1]);
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filters_neon.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filters_neon.c
deleted file mode 100644
index 8270359..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/filters_neon.c
+++ /dev/null
@@ -1,167 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * filters_neon.c
- *
- * This file contains function WebRtcIsacfix_AutocorrNeon, optimized for
- * ARM Neon platform.
- *
- */
-
-#include <arm_neon.h>
-#include <assert.h>
-
-#include "codec.h"
-
-// Autocorrelation function in fixed point.
-// NOTE! Different from SPLIB-version in how it scales the signal.
-int WebRtcIsacfix_AutocorrNeon(
- WebRtc_Word32* __restrict r,
- const WebRtc_Word16* __restrict x,
- WebRtc_Word16 N,
- WebRtc_Word16 order,
- WebRtc_Word16* __restrict scale) {
-
- // The 1st for loop assumed N % 4 == 0.
- assert(N % 4 == 0);
-
- int i = 0;
- int zeros_low = 0;
- int zeros_high = 0;
- int16_t scaling = 0;
- int32_t sum = 0;
-
- // Step 1, calculate r[0] and how much scaling is needed.
-
- int16x4_t reg16x4;
- int64x1_t reg64x1a;
- int64x1_t reg64x1b;
- int32x4_t reg32x4;
- int64x2_t reg64x2 = vdupq_n_s64(0); // zeros
-
- // Loop over the samples and do:
- // sum += WEBRTC_SPL_MUL_16_16(x[i], x[i]);
- for (i = 0; i < N; i += 4) {
- reg16x4 = vld1_s16(&x[i]);
- reg32x4 = vmull_s16(reg16x4, reg16x4);
- reg64x2 = vpadalq_s32(reg64x2, reg32x4);
- }
- reg64x1a = vget_low_s64(reg64x2);
- reg64x1b = vget_high_s64(reg64x2);
- reg64x1a = vadd_s64(reg64x1a, reg64x1b);
-
- // Calculate the value of shifting (scaling).
- __asm__ __volatile__(
- "vmov %[z_l], %[z_h], %P[reg]\n\t"
- "clz %[z_l], %[z_l]\n\t"
- "clz %[z_h], %[z_h]\n\t"
- :[z_l]"+r"(zeros_low),
- [z_h]"+r"(zeros_high)
- :[reg]"w"(reg64x1a)
- );
- if (zeros_high != 32) {
- scaling = (32 - zeros_high + 1);
- } else if (zeros_low == 0) {
- scaling = 1;
- }
- reg64x1b = -scaling;
- reg64x1a = vshl_s64(reg64x1a, reg64x1b);
-
- // Record the result.
- r[0] = (int32_t)vget_lane_s64(reg64x1a, 0);
-
-
- // Step 2, perform the actual correlation calculation.
-
- /* Original C code (for the rest of the function):
- for (i = 1; i < order + 1; i++) {
- prod = 0;
- for (j = 0; j < N - i; j++) {
- prod += WEBRTC_SPL_MUL_16_16(x[j], x[i + j]);
- }
- sum = (int32_t)(prod >> scaling);
- r[i] = sum;
- }
- */
-
- for (i = 1; i < order + 1; i++) {
- int32_t prod_lower = 0;
- int32_t prod_upper = 0;
- int16_t* ptr0 = &x[0];
- int16_t* ptr1 = &x[i];
- int32_t tmp = 0;
-
- // Initialize the sum (q9) to zero.
- __asm__ __volatile__("vmov.i32 q9, #0\n\t":::"q9");
-
- // Calculate the major block of the samples (a multiple of 8).
- for (; ptr0 < &x[N - i - 7];) {
- __asm__ __volatile__(
- "vld1.16 {d20, d21}, [%[ptr0]]!\n\t"
- "vld1.16 {d22, d23}, [%[ptr1]]!\n\t"
- "vmull.s16 q12, d20, d22\n\t"
- "vmull.s16 q13, d21, d23\n\t"
- "vpadal.s32 q9, q12\n\t"
- "vpadal.s32 q9, q13\n\t"
-
- // Specify constraints.
- :[ptr0]"+r"(ptr0),
- [ptr1]"+r"(ptr1)
- :
- :"d18", "d19", "d20", "d21", "d22", "d23", "d24", "d25", "d26", "d27"
- );
- }
-
- // Calculate the rest of the samples.
- for (; ptr0 < &x[N - i]; ptr0++, ptr1++) {
- __asm__ __volatile__(
- "smulbb %[tmp], %[ptr0], %[ptr1]\n\t"
- "adds %[prod_lower], %[prod_lower], %[tmp]\n\t"
- "adc %[prod_upper], %[prod_upper], %[tmp], asr #31\n\t"
-
- // Specify constraints.
- :[prod_lower]"+r"(prod_lower),
- [prod_upper]"+r"(prod_upper),
- [tmp]"+r"(tmp)
- :[ptr0]"r"(*ptr0),
- [ptr1]"r"(*ptr1)
- );
- }
-
- // Sum the results up, and do shift.
- __asm__ __volatile__(
- "vadd.i64 d18, d19\n\t"
- "vmov.32 d17[0], %[prod_lower]\n\t"
- "vmov.32 d17[1], %[prod_upper]\n\t"
- "vadd.i64 d17, d18\n\t"
- "mov %[tmp], %[scaling], asr #31\n\t"
- "vmov.32 d16, %[scaling], %[tmp]\n\t"
- "vshl.s64 d17, d16\n\t"
- "vmov.32 %[sum], d17[0]\n\t"
-
- // Specify constraints.
- :[sum]"=r"(sum),
- [tmp]"+r"(tmp)
- :[prod_upper]"r"(prod_upper),
- [prod_lower]"r"(prod_lower),
- [scaling]"r"(-scaling)
- :"d16", "d17", "d18", "d19"
- );
-
- // Record the result.
- r[i] = sum;
- }
-
- // Record the result.
- *scale = scaling;
-
- return(order + 1);
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/initialize.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/initialize.c
deleted file mode 100644
index 4d11af5..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/initialize.c
+++ /dev/null
@@ -1,175 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * initialize.c
- *
- * Internal initfunctions
- *
- */
-
-#include "codec.h"
-#include "structs.h"
-#include "pitch_estimator.h"
-
-
-void WebRtcIsacfix_InitMaskingEnc(MaskFiltstr_enc *maskdata) {
-
- int k;
-
- for (k = 0; k < WINLEN; k++) {
- maskdata->DataBufferLoQ0[k] = (WebRtc_Word16) 0;
- maskdata->DataBufferHiQ0[k] = (WebRtc_Word16) 0;
- }
- for (k = 0; k < ORDERLO+1; k++) {
- maskdata->CorrBufLoQQ[k] = (WebRtc_Word32) 0;
- maskdata->CorrBufLoQdom[k] = 0;
-
- maskdata->PreStateLoGQ15[k] = 0;
-
- }
- for (k = 0; k < ORDERHI+1; k++) {
- maskdata->CorrBufHiQQ[k] = (WebRtc_Word32) 0;
- maskdata->CorrBufHiQdom[k] = 0;
- maskdata->PreStateHiGQ15[k] = 0;
- }
-
- maskdata->OldEnergy = 10;
-
- return;
-}
-
-void WebRtcIsacfix_InitMaskingDec(MaskFiltstr_dec *maskdata) {
-
- int k;
-
- for (k = 0; k < ORDERLO+1; k++)
- {
- maskdata->PostStateLoGQ0[k] = 0;
- }
- for (k = 0; k < ORDERHI+1; k++)
- {
- maskdata->PostStateHiGQ0[k] = 0;
- }
-
- maskdata->OldEnergy = 10;
-
- return;
-}
-
-
-
-
-
-
-
-void WebRtcIsacfix_InitPreFilterbank(PreFiltBankstr *prefiltdata)
-{
- int k;
-
- for (k = 0; k < QLOOKAHEAD; k++) {
- prefiltdata->INLABUF1_fix[k] = 0;
- prefiltdata->INLABUF2_fix[k] = 0;
- }
- for (k = 0; k < WEBRTC_SPL_MUL_16_16(2,(QORDER-1)); k++) {
-
- prefiltdata->INSTAT1_fix[k] = 0;
- prefiltdata->INSTAT2_fix[k] = 0;
- }
-
- /* High pass filter states */
- prefiltdata->HPstates_fix[0] = 0;
- prefiltdata->HPstates_fix[1] = 0;
-
- return;
-}
-
-void WebRtcIsacfix_InitPostFilterbank(PostFiltBankstr *postfiltdata)
-{
- int k;
-
- for (k = 0; k < WEBRTC_SPL_MUL_16_16(2, POSTQORDER); k++) {
-
- postfiltdata->STATE_0_LOWER_fix[k] = 0;
- postfiltdata->STATE_0_UPPER_fix[k] = 0;
- }
-
- /* High pass filter states */
-
- postfiltdata->HPstates1_fix[0] = 0;
- postfiltdata->HPstates1_fix[1] = 0;
-
- postfiltdata->HPstates2_fix[0] = 0;
- postfiltdata->HPstates2_fix[1] = 0;
-
- return;
-}
-
-
-void WebRtcIsacfix_InitPitchFilter(PitchFiltstr *pitchfiltdata)
-{
- int k;
-
- for (k = 0; k < PITCH_BUFFSIZE; k++)
- pitchfiltdata->ubufQQ[k] = 0;
- for (k = 0; k < (PITCH_DAMPORDER); k++)
- pitchfiltdata->ystateQQ[k] = 0;
-
- pitchfiltdata->oldlagQ7 = 6400; /* 50.0 in Q7 */
- pitchfiltdata->oldgainQ12 = 0;
-}
-
-void WebRtcIsacfix_InitPitchAnalysis(PitchAnalysisStruct *State)
-{
- int k;
-
- for (k = 0; k < PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2; k++) {
- State->dec_buffer16[k] = 0;
- }
- for (k = 0; k < WEBRTC_SPL_MUL_16_16(2, ALLPASSSECTIONS)+1; k++) {
- State->decimator_state32[k] = 0;
- }
-
- for (k = 0; k < QLOOKAHEAD; k++)
- State->inbuf[k] = 0;
-
- WebRtcIsacfix_InitPitchFilter(&(State->PFstr_wght));
-
- WebRtcIsacfix_InitPitchFilter(&(State->PFstr));
-}
-
-
-void WebRtcIsacfix_InitPlc( PLCstr *State )
-{
- State->decayCoeffPriodic = WEBRTC_SPL_WORD16_MAX;
- State->decayCoeffNoise = WEBRTC_SPL_WORD16_MAX;
-
- State->used = PLC_WAS_USED;
-
- WebRtcSpl_ZerosArrayW16(State->overlapLP, RECOVERY_OVERLAP);
- WebRtcSpl_ZerosArrayW16(State->lofilt_coefQ15, ORDERLO);
- WebRtcSpl_ZerosArrayW16(State->hifilt_coefQ15, ORDERHI );
-
- State->AvgPitchGain_Q12 = 0;
- State->lastPitchGain_Q12 = 0;
- State->lastPitchLag_Q7 = 0;
- State->gain_lo_hiQ17[0]=State->gain_lo_hiQ17[1] = 0;
- WebRtcSpl_ZerosArrayW16(State->prevPitchInvIn, FRAMESAMPLES/2);
- WebRtcSpl_ZerosArrayW16(State->prevPitchInvOut, PITCH_MAX_LAG + 10 );
- WebRtcSpl_ZerosArrayW32(State->prevHP, PITCH_MAX_LAG + 10 );
- State->pitchCycles = 0;
- State->A = 0;
- State->B = 0;
- State->pitchIndex = 0;
- State->stretchLag = 240;
- State->seed = 4447;
-
-
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/isacfix.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/isacfix.c
deleted file mode 100644
index 3a37785..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/isacfix.c
+++ /dev/null
@@ -1,1529 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * isacfix.c
- *
- * This C file contains the functions for the ISAC API
- *
- */
-
-#include <stdlib.h>
-#include <string.h>
-
-#include "isacfix.h"
-#include "bandwidth_estimator.h"
-#include "codec.h"
-#include "entropy_coding.h"
-#include "structs.h"
-
-
-/**************************************************************************
- * WebRtcIsacfix_AssignSize(...)
- *
- * Functions used when malloc is not allowed
- * Returns number of bytes needed to allocate for iSAC struct.
- *
- */
-
-WebRtc_Word16 WebRtcIsacfix_AssignSize(int *sizeinbytes) {
- *sizeinbytes=sizeof(ISACFIX_SubStruct)*2/sizeof(WebRtc_Word16);
- return(0);
-}
-
-/***************************************************************************
- * WebRtcIsacfix_Assign(...)
- *
- * Functions used when malloc is not allowed
- * Place struct at given address
- *
- * If successful, Return 0, else Return -1
- */
-
-WebRtc_Word16 WebRtcIsacfix_Assign(ISACFIX_MainStruct **inst, void *ISACFIX_inst_Addr) {
- if (ISACFIX_inst_Addr!=NULL) {
- *inst = (ISACFIX_MainStruct*)ISACFIX_inst_Addr;
- (*(ISACFIX_SubStruct**)inst)->errorcode = 0;
- (*(ISACFIX_SubStruct**)inst)->initflag = 0;
- (*(ISACFIX_SubStruct**)inst)->ISACenc_obj.SaveEnc_ptr = NULL;
- return(0);
- } else {
- return(-1);
- }
-}
-
-
-#ifndef ISACFIX_NO_DYNAMIC_MEM
-
-/****************************************************************************
- * WebRtcIsacfix_Create(...)
- *
- * This function creates a ISAC instance, which will contain the state
- * information for one coding/decoding channel.
- *
- * Input:
- * - *ISAC_main_inst : a pointer to the coder instance.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcIsacfix_Create(ISACFIX_MainStruct **ISAC_main_inst)
-{
- ISACFIX_SubStruct *tempo;
- tempo = malloc(1 * sizeof(ISACFIX_SubStruct));
- *ISAC_main_inst = (ISACFIX_MainStruct *)tempo;
- if (*ISAC_main_inst!=NULL) {
- (*(ISACFIX_SubStruct**)ISAC_main_inst)->errorcode = 0;
- (*(ISACFIX_SubStruct**)ISAC_main_inst)->initflag = 0;
- (*(ISACFIX_SubStruct**)ISAC_main_inst)->ISACenc_obj.SaveEnc_ptr = NULL;
- return(0);
- } else {
- return(-1);
- }
-}
-
-
-/****************************************************************************
- * WebRtcIsacfix_CreateInternal(...)
- *
- * This function creates the memory that is used to store data in the encoder
- *
- * Input:
- * - *ISAC_main_inst : a pointer to the coder instance.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcIsacfix_CreateInternal(ISACFIX_MainStruct *ISAC_main_inst)
-{
- ISACFIX_SubStruct *ISAC_inst;
-
- /* typecast pointer to real structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
- /* Allocate memory for storing encoder data */
- ISAC_inst->ISACenc_obj.SaveEnc_ptr = malloc(1 * sizeof(ISAC_SaveEncData_t));
-
- if (ISAC_inst->ISACenc_obj.SaveEnc_ptr!=NULL) {
- return(0);
- } else {
- return(-1);
- }
-}
-
-
-#endif
-
-
-
-/****************************************************************************
- * WebRtcIsacfix_Free(...)
- *
- * This function frees the ISAC instance created at the beginning.
- *
- * Input:
- * - ISAC_main_inst : a ISAC instance.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcIsacfix_Free(ISACFIX_MainStruct *ISAC_main_inst)
-{
- free(ISAC_main_inst);
- return(0);
-}
-
-/****************************************************************************
- * WebRtcIsacfix_FreeInternal(...)
- *
- * This function frees the internal memory for storing encoder data.
- *
- * Input:
- * - ISAC_main_inst : a ISAC instance.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcIsacfix_FreeInternal(ISACFIX_MainStruct *ISAC_main_inst)
-{
- ISACFIX_SubStruct *ISAC_inst;
-
- /* typecast pointer to real structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
- /* Release memory */
- free(ISAC_inst->ISACenc_obj.SaveEnc_ptr);
-
- return(0);
-}
-
-/****************************************************************************
- * WebRtcIsacfix_EncoderInit(...)
- *
- * This function initializes a ISAC instance prior to the encoder calls.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - CodingMode : 0 -> Bit rate and frame length are automatically
- * adjusted to available bandwidth on
- * transmission channel.
- * 1 -> User sets a frame length and a target bit
- * rate which is taken as the maximum short-term
- * average bit rate.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcIsacfix_EncoderInit(ISACFIX_MainStruct *ISAC_main_inst,
- WebRtc_Word16 CodingMode)
-{
- int k;
- WebRtc_Word16 statusInit;
- ISACFIX_SubStruct *ISAC_inst;
-
- statusInit = 0;
- /* typecast pointer to rela structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
- /* flag encoder init */
- ISAC_inst->initflag |= 2;
-
- if (CodingMode == 0)
- /* Adaptive mode */
- ISAC_inst->ISACenc_obj.new_framelength = INITIAL_FRAMESAMPLES;
- else if (CodingMode == 1)
- /* Instantaneous mode */
- ISAC_inst->ISACenc_obj.new_framelength = 480; /* default for I-mode */
- else {
- ISAC_inst->errorcode = ISAC_DISALLOWED_CODING_MODE;
- statusInit = -1;
- }
-
- ISAC_inst->CodingMode = CodingMode;
-
- WebRtcIsacfix_InitMaskingEnc(&ISAC_inst->ISACenc_obj.maskfiltstr_obj);
- WebRtcIsacfix_InitPreFilterbank(&ISAC_inst->ISACenc_obj.prefiltbankstr_obj);
- WebRtcIsacfix_InitPitchFilter(&ISAC_inst->ISACenc_obj.pitchfiltstr_obj);
- WebRtcIsacfix_InitPitchAnalysis(&ISAC_inst->ISACenc_obj.pitchanalysisstr_obj);
-
-
- WebRtcIsacfix_InitBandwidthEstimator(&ISAC_inst->bwestimator_obj);
- WebRtcIsacfix_InitRateModel(&ISAC_inst->ISACenc_obj.rate_data_obj);
-
-
- ISAC_inst->ISACenc_obj.buffer_index = 0;
- ISAC_inst->ISACenc_obj.frame_nb = 0;
- ISAC_inst->ISACenc_obj.BottleNeck = 32000; /* default for I-mode */
- ISAC_inst->ISACenc_obj.MaxDelay = 10; /* default for I-mode */
- ISAC_inst->ISACenc_obj.current_framesamples = 0;
- ISAC_inst->ISACenc_obj.s2nr = 0;
- ISAC_inst->ISACenc_obj.MaxBits = 0;
- ISAC_inst->ISACenc_obj.bitstr_seed = 4447;
- ISAC_inst->ISACenc_obj.payloadLimitBytes30 = STREAM_MAXW16_30MS << 1;
- ISAC_inst->ISACenc_obj.payloadLimitBytes60 = STREAM_MAXW16_60MS << 1;
- ISAC_inst->ISACenc_obj.maxPayloadBytes = STREAM_MAXW16_60MS << 1;
- ISAC_inst->ISACenc_obj.maxRateInBytes = STREAM_MAXW16_30MS << 1;
- ISAC_inst->ISACenc_obj.enforceFrameSize = 0;
-
- /* Init the bistream data area to zero */
- for (k=0; k<STREAM_MAXW16_60MS; k++){
- ISAC_inst->ISACenc_obj.bitstr_obj.stream[k] = 0;
- }
-
-#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
- WebRtcIsacfix_InitPostFilterbank(&ISAC_inst->ISACenc_obj.interpolatorstr_obj);
-#endif
-
- // Initiaze function pointers.
- WebRtcIsacfix_AutocorrFix = WebRtcIsacfix_AutocorrC;
- WebRtcIsacfix_FilterMaLoopFix = WebRtcIsacfix_FilterMaLoopC;
-
-#ifdef WEBRTC_ARCH_ARM_NEON
- WebRtcIsacfix_AutocorrFix = WebRtcIsacfix_AutocorrNeon;
- WebRtcIsacfix_FilterMaLoopFix = WebRtcIsacfix_FilterMaLoopNeon;
-#endif
-
- return statusInit;
-}
-
-/****************************************************************************
- * WebRtcIsacfix_Encode(...)
- *
- * This function encodes 10ms frame(s) and inserts it into a package.
- * Input speech length has to be 160 samples (10ms). The encoder buffers those
- * 10ms frames until it reaches the chosen Framesize (480 or 960 samples
- * corresponding to 30 or 60 ms frames), and then proceeds to the encoding.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - speechIn : input speech vector.
- *
- * Output:
- * - encoded : the encoded data vector
- *
- * Return value:
- * : >0 - Length (in bytes) of coded data
- * : 0 - The buffer didn't reach the chosen framesize
- * so it keeps buffering speech samples.
- * : -1 - Error
- */
-
-WebRtc_Word16 WebRtcIsacfix_Encode(ISACFIX_MainStruct *ISAC_main_inst,
- const WebRtc_Word16 *speechIn,
- WebRtc_Word16 *encoded)
-{
- ISACFIX_SubStruct *ISAC_inst;
- WebRtc_Word16 stream_len;
-#ifndef WEBRTC_BIG_ENDIAN
- int k;
-#endif
-
- /* typecast pointer to rela structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
-
- /* check if encoder initiated */
- if ((ISAC_inst->initflag & 2) != 2) {
- ISAC_inst->errorcode = ISAC_ENCODER_NOT_INITIATED;
- return (-1);
- }
-
- stream_len = WebRtcIsacfix_EncodeImpl((WebRtc_Word16*)speechIn,
- &ISAC_inst->ISACenc_obj,
- &ISAC_inst->bwestimator_obj,
- ISAC_inst->CodingMode);
- if (stream_len<0) {
- ISAC_inst->errorcode = - stream_len;
- return -1;
- }
-
-
- /* convert from bytes to WebRtc_Word16 */
-#ifndef WEBRTC_BIG_ENDIAN
- for (k=0;k<(stream_len+1)>>1;k++) {
- encoded[k] = (WebRtc_Word16)( ( (WebRtc_UWord16)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8 )
- | (((ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] & 0x00FF) << 8));
- }
-
-#else
- WEBRTC_SPL_MEMCPY_W16(encoded, (ISAC_inst->ISACenc_obj.bitstr_obj).stream, (stream_len + 1)>>1);
-#endif
-
-
-
- return stream_len;
-
-}
-
-
-
-
-/****************************************************************************
- * WebRtcIsacfix_EncodeNb(...)
- *
- * This function encodes 10ms narrow band (8 kHz sampling) frame(s) and inserts
- * it into a package. Input speech length has to be 80 samples (10ms). The encoder
- * interpolates into wide-band (16 kHz sampling) buffers those
- * 10ms frames until it reaches the chosen Framesize (480 or 960 wide-band samples
- * corresponding to 30 or 60 ms frames), and then proceeds to the encoding.
- *
- * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - speechIn : input speech vector.
- *
- * Output:
- * - encoded : the encoded data vector
- *
- * Return value:
- * : >0 - Length (in bytes) of coded data
- * : 0 - The buffer didn't reach the chosen framesize
- * so it keeps buffering speech samples.
- * : -1 - Error
- */
-#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-WebRtc_Word16 WebRtcIsacfix_EncodeNb(ISACFIX_MainStruct *ISAC_main_inst,
- const WebRtc_Word16 *speechIn,
- WebRtc_Word16 *encoded)
-{
- ISACFIX_SubStruct *ISAC_inst;
- WebRtc_Word16 stream_len;
- WebRtc_Word16 speechInWB[FRAMESAMPLES_10ms];
- WebRtc_Word16 Vector_Word16_1[FRAMESAMPLES_10ms/2];
- WebRtc_Word16 Vector_Word16_2[FRAMESAMPLES_10ms/2];
-
- int k;
-
-
- /* typecast pointer to rela structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
-
- /* check if encoder initiated */
- if ((ISAC_inst->initflag & 2) != 2) {
- ISAC_inst->errorcode = ISAC_ENCODER_NOT_INITIATED;
- return (-1);
- }
-
-
- /* Oversample to WB */
-
- /* Form polyphase signals, and compensate for DC offset */
- for (k=0;k<FRAMESAMPLES_10ms/2;k++) {
- Vector_Word16_1[k] = speechIn[k] + 1;
- Vector_Word16_2[k] = speechIn[k];
- }
- WebRtcIsacfix_FilterAndCombine2(Vector_Word16_1, Vector_Word16_2, speechInWB, &ISAC_inst->ISACenc_obj.interpolatorstr_obj, FRAMESAMPLES_10ms);
-
-
- /* Encode WB signal */
- stream_len = WebRtcIsacfix_EncodeImpl((WebRtc_Word16*)speechInWB,
- &ISAC_inst->ISACenc_obj,
- &ISAC_inst->bwestimator_obj,
- ISAC_inst->CodingMode);
- if (stream_len<0) {
- ISAC_inst->errorcode = - stream_len;
- return -1;
- }
-
-
- /* convert from bytes to WebRtc_Word16 */
-#ifndef WEBRTC_BIG_ENDIAN
- for (k=0;k<(stream_len+1)>>1;k++) {
- encoded[k] = (WebRtc_Word16)(((WebRtc_UWord16)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8)
- | (((ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] & 0x00FF) << 8));
- }
-
-#else
- WEBRTC_SPL_MEMCPY_W16(encoded, (ISAC_inst->ISACenc_obj.bitstr_obj).stream, (stream_len + 1)>>1);
-#endif
-
-
-
- return stream_len;
-}
-#endif /* WEBRTC_ISAC_FIX_NB_CALLS_ENABLED */
-
-
-/****************************************************************************
- * WebRtcIsacfix_GetNewBitStream(...)
- *
- * This function returns encoded data, with the recieved bwe-index in the
- * stream. It should always return a complete packet, i.e. only called once
- * even for 60 msec frames
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - bweIndex : index of bandwidth estimate to put in new bitstream
- *
- * Output:
- * - encoded : the encoded data vector
- *
- * Return value:
- * : >0 - Length (in bytes) of coded data
- * : -1 - Error
- */
-
-WebRtc_Word16 WebRtcIsacfix_GetNewBitStream(ISACFIX_MainStruct *ISAC_main_inst,
- WebRtc_Word16 bweIndex,
- float scale,
- WebRtc_Word16 *encoded)
-{
- ISACFIX_SubStruct *ISAC_inst;
- WebRtc_Word16 stream_len;
-#ifndef WEBRTC_BIG_ENDIAN
- int k;
-#endif
-
- /* typecast pointer to rela structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
-
- /* check if encoder initiated */
- if ((ISAC_inst->initflag & 2) != 2) {
- ISAC_inst->errorcode = ISAC_ENCODER_NOT_INITIATED;
- return (-1);
- }
-
- stream_len = WebRtcIsacfix_EncodeStoredData(&ISAC_inst->ISACenc_obj,
- bweIndex,
- scale);
- if (stream_len<0) {
- ISAC_inst->errorcode = - stream_len;
- return -1;
- }
-
-#ifndef WEBRTC_BIG_ENDIAN
- for (k=0;k<(stream_len+1)>>1;k++) {
- encoded[k] = (WebRtc_Word16)( ( (WebRtc_UWord16)(ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] >> 8 )
- | (((ISAC_inst->ISACenc_obj.bitstr_obj).stream[k] & 0x00FF) << 8));
- }
-
-#else
- WEBRTC_SPL_MEMCPY_W16(encoded, (ISAC_inst->ISACenc_obj.bitstr_obj).stream, (stream_len + 1)>>1);
-#endif
-
- return stream_len;
-
-}
-
-
-
-/****************************************************************************
- * WebRtcIsacfix_DecoderInit(...)
- *
- * This function initializes a ISAC instance prior to the decoder calls.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- *
- * Return value
- * : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcIsacfix_DecoderInit(ISACFIX_MainStruct *ISAC_main_inst)
-{
- ISACFIX_SubStruct *ISAC_inst;
-
- /* typecast pointer to real structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
- /* flag decoder init */
- ISAC_inst->initflag |= 1;
-
-
- WebRtcIsacfix_InitMaskingDec(&ISAC_inst->ISACdec_obj.maskfiltstr_obj);
- WebRtcIsacfix_InitPostFilterbank(&ISAC_inst->ISACdec_obj.postfiltbankstr_obj);
- WebRtcIsacfix_InitPitchFilter(&ISAC_inst->ISACdec_obj.pitchfiltstr_obj);
-
- /* TS */
- WebRtcIsacfix_InitPlc( &ISAC_inst->ISACdec_obj.plcstr_obj );
-
-
-#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
- WebRtcIsacfix_InitPreFilterbank(&ISAC_inst->ISACdec_obj.decimatorstr_obj);
-#endif
-
- return 0;
-}
-
-
-/****************************************************************************
- * WebRtcIsacfix_UpdateBwEstimate1(...)
- *
- * This function updates the estimate of the bandwidth.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - encoded : encoded ISAC frame(s).
- * - packet_size : size of the packet.
- * - rtp_seq_number : the RTP number of the packet.
- * - arr_ts : the arrival time of the packet (from NetEq)
- * in samples.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_MainStruct *ISAC_main_inst,
- const WebRtc_UWord16 *encoded,
- WebRtc_Word32 packet_size,
- WebRtc_UWord16 rtp_seq_number,
- WebRtc_UWord32 arr_ts)
-{
- ISACFIX_SubStruct *ISAC_inst;
- Bitstr_dec streamdata;
- WebRtc_UWord16 partOfStream[5];
-#ifndef WEBRTC_BIG_ENDIAN
- int k;
-#endif
- WebRtc_Word16 err;
-
- /* Set stream pointer to point at partOfStream */
- streamdata.stream = (WebRtc_UWord16 *)partOfStream;
-
- /* typecast pointer to real structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
- /* Sanity check of packet length */
- if (packet_size <= 0) {
- /* return error code if the packet length is null or less */
- ISAC_inst->errorcode = ISAC_EMPTY_PACKET;
- return -1;
- } else if (packet_size > (STREAM_MAXW16<<1)) {
- /* return error code if length of stream is too long */
- ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
- return -1;
- }
-
- /* check if decoder initiated */
- if ((ISAC_inst->initflag & 1) != 1) {
- ISAC_inst->errorcode = ISAC_DECODER_NOT_INITIATED;
- return (-1);
- }
-
- streamdata.W_upper = 0xFFFFFFFF;
- streamdata.streamval = 0;
- streamdata.stream_index = 0;
- streamdata.full = 1;
-
-#ifndef WEBRTC_BIG_ENDIAN
- for (k=0; k<5; k++) {
- streamdata.stream[k] = (WebRtc_UWord16) (((WebRtc_UWord16)encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
- }
-#else
- memcpy(streamdata.stream, encoded, 5);
-#endif
-
- if (packet_size == 0)
- {
- /* return error code if the packet length is null */
- ISAC_inst->errorcode = ISAC_EMPTY_PACKET;
- return -1;
- }
-
- err = WebRtcIsacfix_EstimateBandwidth(&ISAC_inst->bwestimator_obj,
- &streamdata,
- packet_size,
- rtp_seq_number,
- 0,
- arr_ts);
-
-
- if (err < 0)
- {
- /* return error code if something went wrong */
- ISAC_inst->errorcode = -err;
- return -1;
- }
-
-
- return 0;
-}
-
-/****************************************************************************
- * WebRtcIsacfix_UpdateBwEstimate(...)
- *
- * This function updates the estimate of the bandwidth.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - encoded : encoded ISAC frame(s).
- * - packet_size : size of the packet.
- * - rtp_seq_number : the RTP number of the packet.
- * - send_ts : Send Time Stamp from RTP header
- * - arr_ts : the arrival time of the packet (from NetEq)
- * in samples.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcIsacfix_UpdateBwEstimate(ISACFIX_MainStruct *ISAC_main_inst,
- const WebRtc_UWord16 *encoded,
- WebRtc_Word32 packet_size,
- WebRtc_UWord16 rtp_seq_number,
- WebRtc_UWord32 send_ts,
- WebRtc_UWord32 arr_ts)
-{
- ISACFIX_SubStruct *ISAC_inst;
- Bitstr_dec streamdata;
- WebRtc_UWord16 partOfStream[5];
-#ifndef WEBRTC_BIG_ENDIAN
- int k;
-#endif
- WebRtc_Word16 err;
-
- /* Set stream pointer to point at partOfStream */
- streamdata.stream = (WebRtc_UWord16 *)partOfStream;
-
- /* typecast pointer to real structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
- /* Sanity check of packet length */
- if (packet_size <= 0) {
- /* return error code if the packet length is null or less */
- ISAC_inst->errorcode = ISAC_EMPTY_PACKET;
- return -1;
- } else if (packet_size > (STREAM_MAXW16<<1)) {
- /* return error code if length of stream is too long */
- ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
- return -1;
- }
-
- /* check if decoder initiated */
- if ((ISAC_inst->initflag & 1) != 1) {
- ISAC_inst->errorcode = ISAC_DECODER_NOT_INITIATED;
- return (-1);
- }
-
- streamdata.W_upper = 0xFFFFFFFF;
- streamdata.streamval = 0;
- streamdata.stream_index = 0;
- streamdata.full = 1;
-
-#ifndef WEBRTC_BIG_ENDIAN
- for (k=0; k<5; k++) {
- streamdata.stream[k] = (WebRtc_UWord16) ((encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
- }
-#else
- memcpy(streamdata.stream, encoded, 5);
-#endif
-
- if (packet_size == 0)
- {
- /* return error code if the packet length is null */
- ISAC_inst->errorcode = ISAC_EMPTY_PACKET;
- return -1;
- }
-
- err = WebRtcIsacfix_EstimateBandwidth(&ISAC_inst->bwestimator_obj,
- &streamdata,
- packet_size,
- rtp_seq_number,
- send_ts,
- arr_ts);
-
- if (err < 0)
- {
- /* return error code if something went wrong */
- ISAC_inst->errorcode = -err;
- return -1;
- }
-
-
- return 0;
-}
-
-/****************************************************************************
- * WebRtcIsacfix_Decode(...)
- *
- * This function decodes a ISAC frame. Output speech length
- * will be a multiple of 480 samples: 480 or 960 samples,
- * depending on the framesize (30 or 60 ms).
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - encoded : encoded ISAC frame(s)
- * - len : bytes in encoded vector
- *
- * Output:
- * - decoded : The decoded vector
- *
- * Return value : >0 - number of samples in decoded vector
- * -1 - Error
- */
-
-
-WebRtc_Word16 WebRtcIsacfix_Decode(ISACFIX_MainStruct *ISAC_main_inst,
- const WebRtc_UWord16 *encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType)
-{
- ISACFIX_SubStruct *ISAC_inst;
- /* number of samples (480 or 960), output from decoder */
- /* that were actually used in the encoder/decoder (determined on the fly) */
- WebRtc_Word16 number_of_samples;
-#ifndef WEBRTC_BIG_ENDIAN
- int k;
-#endif
- WebRtc_Word16 declen = 0;
-
- /* typecast pointer to real structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
- /* check if decoder initiated */
- if ((ISAC_inst->initflag & 1) != 1) {
- ISAC_inst->errorcode = ISAC_DECODER_NOT_INITIATED;
- return (-1);
- }
-
- /* Sanity check of packet length */
- if (len <= 0) {
- /* return error code if the packet length is null or less */
- ISAC_inst->errorcode = ISAC_EMPTY_PACKET;
- return -1;
- } else if (len > (STREAM_MAXW16<<1)) {
- /* return error code if length of stream is too long */
- ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
- return -1;
- }
-
- (ISAC_inst->ISACdec_obj.bitstr_obj).stream = (WebRtc_UWord16 *)encoded;
-
- /* convert bitstream from WebRtc_Word16 to bytes */
-#ifndef WEBRTC_BIG_ENDIAN
- for (k=0; k<(len>>1); k++) {
- (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (WebRtc_UWord16) ((encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
- }
- if (len & 0x0001)
- (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (WebRtc_UWord16) ((encoded[k] & 0xFF)<<8);
-#endif
-
- /* added for NetEq purposes (VAD/DTX related) */
- *speechType=1;
-
- declen = WebRtcIsacfix_DecodeImpl(decoded,&ISAC_inst->ISACdec_obj, &number_of_samples);
-
- if (declen < 0) {
- /* Some error inside the decoder */
- ISAC_inst->errorcode = -declen;
- memset(decoded, 0, sizeof(WebRtc_Word16) * MAX_FRAMESAMPLES);
- return -1;
- }
-
- /* error check */
-
- if (declen & 0x0001) {
- if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) & 0x00FF) ) {
- ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
- memset(decoded, 0, sizeof(WebRtc_Word16) * number_of_samples);
- return -1;
- }
- } else {
- if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) >> 8) ) {
- ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
- memset(decoded, 0, sizeof(WebRtc_Word16) * number_of_samples);
- return -1;
- }
- }
-
- return number_of_samples;
-}
-
-
-
-
-
-/****************************************************************************
- * WebRtcIsacfix_DecodeNb(...)
- *
- * This function decodes a ISAC frame in narrow-band (8 kHz sampling).
- * Output speech length will be a multiple of 240 samples: 240 or 480 samples,
- * depending on the framesize (30 or 60 ms).
- *
- * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - encoded : encoded ISAC frame(s)
- * - len : bytes in encoded vector
- *
- * Output:
- * - decoded : The decoded vector
- *
- * Return value : >0 - number of samples in decoded vector
- * -1 - Error
- */
-
-#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-WebRtc_Word16 WebRtcIsacfix_DecodeNb(ISACFIX_MainStruct *ISAC_main_inst,
- const WebRtc_UWord16 *encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType)
-{
- ISACFIX_SubStruct *ISAC_inst;
- /* twice the number of samples (480 or 960), output from decoder */
- /* that were actually used in the encoder/decoder (determined on the fly) */
- WebRtc_Word16 number_of_samples;
-#ifndef WEBRTC_BIG_ENDIAN
- int k;
-#endif
- WebRtc_Word16 declen = 0;
- WebRtc_Word16 dummy[FRAMESAMPLES/2];
-
-
- /* typecast pointer to real structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
- /* check if decoder initiated */
- if ((ISAC_inst->initflag & 1) != 1) {
- ISAC_inst->errorcode = ISAC_DECODER_NOT_INITIATED;
- return (-1);
- }
-
- if (len == 0)
- { /* return error code if the packet length is null */
-
- ISAC_inst->errorcode = ISAC_EMPTY_PACKET;
- return -1;
- }
-
- (ISAC_inst->ISACdec_obj.bitstr_obj).stream = (WebRtc_UWord16 *)encoded;
-
- /* convert bitstream from WebRtc_Word16 to bytes */
-#ifndef WEBRTC_BIG_ENDIAN
- for (k=0; k<(len>>1); k++) {
- (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (WebRtc_UWord16) ((encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
- }
- if (len & 0x0001)
- (ISAC_inst->ISACdec_obj.bitstr_obj).stream[k] = (WebRtc_UWord16) ((encoded[k] & 0xFF)<<8);
-#endif
-
- /* added for NetEq purposes (VAD/DTX related) */
- *speechType=1;
-
- declen = WebRtcIsacfix_DecodeImpl(decoded,&ISAC_inst->ISACdec_obj, &number_of_samples);
-
- if (declen < 0) {
- /* Some error inside the decoder */
- ISAC_inst->errorcode = -declen;
- memset(decoded, 0, sizeof(WebRtc_Word16) * FRAMESAMPLES);
- return -1;
- }
-
- /* error check */
-
- if (declen & 0x0001) {
- if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) & 0x00FF) ) {
- ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
- memset(decoded, 0, sizeof(WebRtc_Word16) * number_of_samples);
- return -1;
- }
- } else {
- if (len != declen && len != declen + (((ISAC_inst->ISACdec_obj.bitstr_obj).stream[declen>>1]) >> 8) ) {
- ISAC_inst->errorcode = ISAC_LENGTH_MISMATCH;
- memset(decoded, 0, sizeof(WebRtc_Word16) * number_of_samples);
- return -1;
- }
- }
-
- WebRtcIsacfix_SplitAndFilter2(decoded, decoded, dummy, &ISAC_inst->ISACdec_obj.decimatorstr_obj);
-
- if (number_of_samples>FRAMESAMPLES) {
- WebRtcIsacfix_SplitAndFilter2(decoded + FRAMESAMPLES, decoded + FRAMESAMPLES/2,
- dummy, &ISAC_inst->ISACdec_obj.decimatorstr_obj);
- }
-
- return number_of_samples/2;
-}
-#endif /* WEBRTC_ISAC_FIX_NB_CALLS_ENABLED */
-
-
-/****************************************************************************
- * WebRtcIsacfix_DecodePlcNb(...)
- *
- * This function conducts PLC for ISAC frame(s) in narrow-band (8kHz sampling).
- * Output speech length will be "240*noOfLostFrames" samples
- * that is equevalent of "30*noOfLostFrames" millisecond.
- *
- * The function is enabled if WEBRTC_ISAC_FIX_NB_CALLS_ENABLED is defined
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - noOfLostFrames : Number of PLC frames (240 sample=30ms) to produce
- *
- * Output:
- * - decoded : The decoded vector
- *
- * Return value : >0 - number of samples in decoded PLC vector
- * -1 - Error
- */
-
-#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
-WebRtc_Word16 WebRtcIsacfix_DecodePlcNb(ISACFIX_MainStruct *ISAC_main_inst,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 noOfLostFrames )
-{
- WebRtc_Word16 no_of_samples, declen, k, ok;
- WebRtc_Word16 outframeNB[FRAMESAMPLES];
- WebRtc_Word16 outframeWB[FRAMESAMPLES];
- WebRtc_Word16 dummy[FRAMESAMPLES/2];
-
-
- ISACFIX_SubStruct *ISAC_inst;
- /* typecast pointer to real structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
- /* Limit number of frames to two = 60 msec. Otherwise we exceed data vectors */
- if (noOfLostFrames > 2){
- noOfLostFrames = 2;
- }
-
- k = 0;
- declen = 0;
- while( noOfLostFrames > 0 )
- {
- ok = WebRtcIsacfix_DecodePlcImpl( outframeWB, &ISAC_inst->ISACdec_obj, &no_of_samples );
- if(ok)
- return -1;
-
- WebRtcIsacfix_SplitAndFilter2(outframeWB, &(outframeNB[k*240]), dummy, &ISAC_inst->ISACdec_obj.decimatorstr_obj);
-
- declen += no_of_samples;
- noOfLostFrames--;
- k++;
- }
-
- declen>>=1;
-
- for (k=0;k<declen;k++) {
- decoded[k] = outframeNB[k];
- }
-
- return declen;
-}
-#endif /* WEBRTC_ISAC_FIX_NB_CALLS_ENABLED */
-
-
-
-
-/****************************************************************************
- * WebRtcIsacfix_DecodePlc(...)
- *
- * This function conducts PLC for ISAC frame(s) in wide-band (16kHz sampling).
- * Output speech length will be "480*noOfLostFrames" samples
- * that is equevalent of "30*noOfLostFrames" millisecond.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - noOfLostFrames : Number of PLC frames (480sample = 30ms)
- * to produce
- *
- * Output:
- * - decoded : The decoded vector
- *
- * Return value : >0 - number of samples in decoded PLC vector
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcIsacfix_DecodePlc(ISACFIX_MainStruct *ISAC_main_inst,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 noOfLostFrames)
-{
-
- WebRtc_Word16 no_of_samples, declen, k, ok;
- WebRtc_Word16 outframe16[MAX_FRAMESAMPLES];
-
- ISACFIX_SubStruct *ISAC_inst;
- /* typecast pointer to real structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
- /* Limit number of frames to two = 60 msec. Otherwise we exceed data vectors */
- if (noOfLostFrames > 2) {
- noOfLostFrames = 2;
- }
- k = 0;
- declen = 0;
- while( noOfLostFrames > 0 )
- {
- ok = WebRtcIsacfix_DecodePlcImpl( &(outframe16[k*480]), &ISAC_inst->ISACdec_obj, &no_of_samples );
- if(ok)
- return -1;
- declen += no_of_samples;
- noOfLostFrames--;
- k++;
- }
-
- for (k=0;k<declen;k++) {
- decoded[k] = outframe16[k];
- }
-
- return declen;
-}
-
-
-/****************************************************************************
- * WebRtcIsacfix_Control(...)
- *
- * This function sets the limit on the short-term average bit rate and the
- * frame length. Should be used only in Instantaneous mode.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - rate : limit on the short-term average bit rate,
- * in bits/second (between 10000 and 32000)
- * - framesize : number of milliseconds per frame (30 or 60)
- *
- * Return value : 0 - ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcIsacfix_Control(ISACFIX_MainStruct *ISAC_main_inst,
- WebRtc_Word16 rate,
- WebRtc_Word16 framesize)
-{
- ISACFIX_SubStruct *ISAC_inst;
- /* typecast pointer to real structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
- if (ISAC_inst->CodingMode == 0)
- {
- /* in adaptive mode */
- ISAC_inst->errorcode = ISAC_MODE_MISMATCH;
- return -1;
- }
-
-
- if (rate >= 10000 && rate <= 32000)
- ISAC_inst->ISACenc_obj.BottleNeck = rate;
- else {
- ISAC_inst->errorcode = ISAC_DISALLOWED_BOTTLENECK;
- return -1;
- }
-
-
-
- if (framesize == 30 || framesize == 60)
- ISAC_inst->ISACenc_obj.new_framelength = (FS/1000) * framesize;
- else {
- ISAC_inst->errorcode = ISAC_DISALLOWED_FRAME_LENGTH;
- return -1;
- }
-
- return 0;
-}
-
-
-/****************************************************************************
- * WebRtcIsacfix_ControlBwe(...)
- *
- * This function sets the initial values of bottleneck and frame-size if
- * iSAC is used in channel-adaptive mode. Through this API, users can
- * enforce a frame-size for all values of bottleneck. Then iSAC will not
- * automatically change the frame-size.
- *
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - rateBPS : initial value of bottleneck in bits/second
- * 10000 <= rateBPS <= 32000 is accepted
- * For default bottleneck set rateBPS = 0
- * - frameSizeMs : number of milliseconds per frame (30 or 60)
- * - enforceFrameSize : 1 to enforce the given frame-size through out
- * the adaptation process, 0 to let iSAC change
- * the frame-size if required.
- *
- * Return value : 0 - ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcIsacfix_ControlBwe(ISACFIX_MainStruct *ISAC_main_inst,
- WebRtc_Word16 rateBPS,
- WebRtc_Word16 frameSizeMs,
- WebRtc_Word16 enforceFrameSize)
-{
- ISACFIX_SubStruct *ISAC_inst;
- /* Typecast pointer to real structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
- /* check if encoder initiated */
- if ((ISAC_inst->initflag & 2) != 2) {
- ISAC_inst->errorcode = ISAC_ENCODER_NOT_INITIATED;
- return (-1);
- }
-
- /* Check that we are in channel-adaptive mode, otherwise, return -1 */
- if (ISAC_inst->CodingMode != 0) {
- ISAC_inst->errorcode = ISAC_MODE_MISMATCH;
- return (-1);
- }
-
- /* Set struct variable if enforceFrameSize is set. ISAC will then keep the */
- /* chosen frame size. */
- ISAC_inst->ISACenc_obj.enforceFrameSize = (enforceFrameSize != 0)? 1:0;
-
- /* Set initial rate, if value between 10000 and 32000, */
- /* if rateBPS is 0, keep the default initial bottleneck value (15000) */
- if ((rateBPS >= 10000) && (rateBPS <= 32000)) {
- ISAC_inst->bwestimator_obj.sendBwAvg = (((WebRtc_UWord32)rateBPS) << 7);
- } else if (rateBPS != 0) {
- ISAC_inst->errorcode = ISAC_DISALLOWED_BOTTLENECK;
- return -1;
- }
-
- /* Set initial framesize. If enforceFrameSize is set the frame size will not change */
- if ((frameSizeMs == 30) || (frameSizeMs == 60)) {
- ISAC_inst->ISACenc_obj.new_framelength = (FS/1000) * frameSizeMs;
- } else {
- ISAC_inst->errorcode = ISAC_DISALLOWED_FRAME_LENGTH;
- return -1;
- }
-
- return 0;
-}
-
-
-
-
-
-/****************************************************************************
- * WebRtcIsacfix_GetDownLinkBwIndex(...)
- *
- * This function returns index representing the Bandwidth estimate from
- * other side to this side.
- *
- * Input:
- * - ISAC_main_inst: iSAC struct
- *
- * Output:
- * - rateIndex : Bandwidth estimate to transmit to other side.
- *
- */
-
-WebRtc_Word16 WebRtcIsacfix_GetDownLinkBwIndex(ISACFIX_MainStruct* ISAC_main_inst,
- WebRtc_Word16* rateIndex)
-{
- ISACFIX_SubStruct *ISAC_inst;
-
- /* typecast pointer to real structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
- /* Call function to get Bandwidth Estimate */
- *rateIndex = WebRtcIsacfix_GetDownlinkBwIndexImpl(&ISAC_inst->bwestimator_obj);
-
- return 0;
-}
-
-
-/****************************************************************************
- * WebRtcIsacfix_UpdateUplinkBw(...)
- *
- * This function takes an index representing the Bandwidth estimate from
- * this side to other side and updates BWE.
- *
- * Input:
- * - ISAC_main_inst: iSAC struct
- * - rateIndex : Bandwidth estimate from other side.
- *
- */
-
-WebRtc_Word16 WebRtcIsacfix_UpdateUplinkBw(ISACFIX_MainStruct* ISAC_main_inst,
- WebRtc_Word16 rateIndex)
-{
- WebRtc_Word16 err = 0;
- ISACFIX_SubStruct *ISAC_inst;
-
- /* typecast pointer to real structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
- /* Call function to update BWE with received Bandwidth Estimate */
- err = WebRtcIsacfix_UpdateUplinkBwRec(&ISAC_inst->bwestimator_obj, rateIndex);
- if (err < 0) {
- ISAC_inst->errorcode = -err;
- return (-1);
- }
-
- return 0;
-}
-
-/****************************************************************************
- * WebRtcIsacfix_ReadFrameLen(...)
- *
- * This function returns the length of the frame represented in the packet.
- *
- * Input:
- * - encoded : Encoded bitstream
- *
- * Output:
- * - frameLength : Length of frame in packet (in samples)
- *
- */
-
-WebRtc_Word16 WebRtcIsacfix_ReadFrameLen(const WebRtc_Word16* encoded,
- WebRtc_Word16* frameLength)
-{
- Bitstr_dec streamdata;
- WebRtc_UWord16 partOfStream[5];
-#ifndef WEBRTC_BIG_ENDIAN
- int k;
-#endif
- WebRtc_Word16 err;
-
- /* Set stream pointer to point at partOfStream */
- streamdata.stream = (WebRtc_UWord16 *)partOfStream;
-
- streamdata.W_upper = 0xFFFFFFFF;
- streamdata.streamval = 0;
- streamdata.stream_index = 0;
- streamdata.full = 1;
-
-#ifndef WEBRTC_BIG_ENDIAN
- for (k=0; k<5; k++) {
- streamdata.stream[k] = (WebRtc_UWord16) (((WebRtc_UWord16)encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
- }
-#else
- memcpy(streamdata.stream, encoded, 5);
-#endif
-
- /* decode frame length */
- err = WebRtcIsacfix_DecodeFrameLen(&streamdata, frameLength);
- if (err<0) // error check
- return err;
-
- return 0;
-}
-
-
-/****************************************************************************
- * WebRtcIsacfix_ReadBwIndex(...)
- *
- * This function returns the index of the Bandwidth estimate from the bitstream.
- *
- * Input:
- * - encoded : Encoded bitstream
- *
- * Output:
- * - frameLength : Length of frame in packet (in samples)
- * - rateIndex : Bandwidth estimate in bitstream
- *
- */
-
-WebRtc_Word16 WebRtcIsacfix_ReadBwIndex(const WebRtc_Word16* encoded,
- WebRtc_Word16* rateIndex)
-{
- Bitstr_dec streamdata;
- WebRtc_UWord16 partOfStream[5];
-#ifndef WEBRTC_BIG_ENDIAN
- int k;
-#endif
- WebRtc_Word16 err;
-
- /* Set stream pointer to point at partOfStream */
- streamdata.stream = (WebRtc_UWord16 *)partOfStream;
-
- streamdata.W_upper = 0xFFFFFFFF;
- streamdata.streamval = 0;
- streamdata.stream_index = 0;
- streamdata.full = 1;
-
-#ifndef WEBRTC_BIG_ENDIAN
- for (k=0; k<5; k++) {
- streamdata.stream[k] = (WebRtc_UWord16) (((WebRtc_UWord16)encoded[k] >> 8)|((encoded[k] & 0xFF)<<8));
- }
-#else
- memcpy(streamdata.stream, encoded, 5);
-#endif
-
- /* decode frame length, needed to get to the rateIndex in the bitstream */
- err = WebRtcIsacfix_DecodeFrameLen(&streamdata, rateIndex);
- if (err<0) // error check
- return err;
-
- /* decode BW estimation */
- err = WebRtcIsacfix_DecodeSendBandwidth(&streamdata, rateIndex);
- if (err<0) // error check
- return err;
-
- return 0;
-}
-
-
-
-
-/****************************************************************************
- * WebRtcIsacfix_GetErrorCode(...)
- *
- * This function can be used to check the error code of an iSAC instance. When
- * a function returns -1 a error code will be set for that instance. The
- * function below extract the code of the last error that occured in the
- * specified instance.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance
- *
- * Return value : Error code
- */
-
-WebRtc_Word16 WebRtcIsacfix_GetErrorCode(ISACFIX_MainStruct *ISAC_main_inst)
-{
- ISACFIX_SubStruct *ISAC_inst;
- /* typecast pointer to real structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
- return ISAC_inst->errorcode;
-}
-
-
-
-/****************************************************************************
- * WebRtcIsacfix_GetUplinkBw(...)
- *
- * This function returns the inst quantized iSAC send bitrate
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- *
- * Return value : bitrate
- */
-
-WebRtc_Word32 WebRtcIsacfix_GetUplinkBw(ISACFIX_MainStruct *ISAC_main_inst)
-{
- ISACFIX_SubStruct *ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
- BwEstimatorstr * bw = (BwEstimatorstr*)&(ISAC_inst->bwestimator_obj);
-
- return (WebRtc_Word32) WebRtcIsacfix_GetUplinkBandwidth(bw);
-}
-
-/****************************************************************************
- * WebRtcIsacfix_GetNewFrameLen(...)
- *
- * This function return the next frame length (in samples) of iSAC.
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- *
- * Return value : frame lenght in samples
- */
-
-WebRtc_Word16 WebRtcIsacfix_GetNewFrameLen(ISACFIX_MainStruct *ISAC_main_inst)
-{
- ISACFIX_SubStruct *ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
- return ISAC_inst->ISACenc_obj.new_framelength;
-}
-
-
-/****************************************************************************
- * WebRtcIsacfix_SetMaxPayloadSize(...)
- *
- * This function sets a limit for the maximum payload size of iSAC. The same
- * value is used both for 30 and 60 msec packets.
- * The absolute max will be valid until next time the function is called.
- * NOTE! This function may override the function WebRtcIsacfix_SetMaxRate()
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- * - maxPayloadBytes : maximum size of the payload in bytes
- * valid values are between 100 and 400 bytes
- *
- *
- * Return value : 0 if sucessful
- * -1 if error happens
- */
-
-WebRtc_Word16 WebRtcIsacfix_SetMaxPayloadSize(ISACFIX_MainStruct *ISAC_main_inst,
- WebRtc_Word16 maxPayloadBytes)
-{
- ISACFIX_SubStruct *ISAC_inst;
-
- /* typecast pointer to real structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
- if((maxPayloadBytes < 100) || (maxPayloadBytes > 400))
- {
- /* maxPayloadBytes is out of valid range */
- return -1;
- }
- else
- {
- /* Set new absolute max, which will not change unless this function
- is called again with a new value */
- ISAC_inst->ISACenc_obj.maxPayloadBytes = maxPayloadBytes;
-
- /* Set new maximum values for 30 and 60 msec packets */
- if (maxPayloadBytes < ISAC_inst->ISACenc_obj.maxRateInBytes) {
- ISAC_inst->ISACenc_obj.payloadLimitBytes30 = maxPayloadBytes;
- } else {
- ISAC_inst->ISACenc_obj.payloadLimitBytes30 = ISAC_inst->ISACenc_obj.maxRateInBytes;
- }
-
- if ( maxPayloadBytes < (ISAC_inst->ISACenc_obj.maxRateInBytes << 1)) {
- ISAC_inst->ISACenc_obj.payloadLimitBytes60 = maxPayloadBytes;
- } else {
- ISAC_inst->ISACenc_obj.payloadLimitBytes60 = (ISAC_inst->ISACenc_obj.maxRateInBytes << 1);
- }
- }
- return 0;
-}
-
-
-/****************************************************************************
- * WebRtcIsacfix_SetMaxRate(...)
- *
- * This function sets the maximum rate which the codec may not exceed for a
- * singel packet. The maximum rate is set in bits per second.
- * The codec has an absolute maximum rate of 53400 bits per second (200 bytes
- * per 30 msec).
- * It is possible to set a maximum rate between 32000 and 53400 bits per second.
- *
- * The rate limit is valid until next time the function is called.
- *
- * NOTE! Packet size will never go above the value set if calling
- * WebRtcIsacfix_SetMaxPayloadSize() (default max packet size is 400 bytes).
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- * - maxRateInBytes : maximum rate in bits per second,
- * valid values are 32000 to 53400 bits
- *
- * Return value : 0 if sucessful
- * -1 if error happens
- */
-
-WebRtc_Word16 WebRtcIsacfix_SetMaxRate(ISACFIX_MainStruct *ISAC_main_inst,
- WebRtc_Word32 maxRate)
-{
- ISACFIX_SubStruct *ISAC_inst;
- WebRtc_Word16 maxRateInBytes;
-
- /* typecast pointer to real structure */
- ISAC_inst = (ISACFIX_SubStruct *)ISAC_main_inst;
-
- if((maxRate < 32000) || (maxRate > 53400))
- {
- /* maxRate is out of valid range */
- return -1;
- }
- else
- {
- /* Calculate maximum number of bytes per 30 msec packets for the given
- maximum rate. Multiply with 30/1000 to get number of bits per 30 msec,
- divide by 8 to get number of bytes per 30 msec:
- maxRateInBytes = floor((maxRate * 30/1000) / 8); */
- maxRateInBytes = (WebRtc_Word16)( WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_MUL(maxRate, 3), 800) );
-
- /* Store the value for usage in the WebRtcIsacfix_SetMaxPayloadSize-function */
- ISAC_inst->ISACenc_obj.maxRateInBytes = maxRateInBytes;
-
- /* For 30 msec packets: if the new limit is below the maximum
- payload size, set a new limit */
- if (maxRateInBytes < ISAC_inst->ISACenc_obj.maxPayloadBytes) {
- ISAC_inst->ISACenc_obj.payloadLimitBytes30 = maxRateInBytes;
- } else {
- ISAC_inst->ISACenc_obj.payloadLimitBytes30 = ISAC_inst->ISACenc_obj.maxPayloadBytes;
- }
-
- /* For 60 msec packets: if the new limit (times 2) is below the
- maximum payload size, set a new limit */
- if ( (maxRateInBytes << 1) < ISAC_inst->ISACenc_obj.maxPayloadBytes) {
- ISAC_inst->ISACenc_obj.payloadLimitBytes60 = (maxRateInBytes << 1);
- } else {
- ISAC_inst->ISACenc_obj.payloadLimitBytes60 = ISAC_inst->ISACenc_obj.maxPayloadBytes;
- }
- }
-
- return 0;
-}
-
-
-
-/****************************************************************************
- * WebRtcIsacfix_version(...)
- *
- * This function returns the version number.
- *
- * Output:
- * - version : Pointer to character string
- *
- */
-
-void WebRtcIsacfix_version(char *version)
-{
- strcpy(version, "3.6.0");
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/isacfix.gypi b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/isacfix.gypi
deleted file mode 100644
index a13d1f5..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/isacfix.gypi
+++ /dev/null
@@ -1,82 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'iSACFix',
- 'type': '<(library)',
- 'dependencies': [
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- ],
- 'include_dirs': [
- '../interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- ],
- },
- 'sources': [
- '../interface/isacfix.h',
- 'arith_routines.c',
- 'arith_routines_hist.c',
- 'arith_routines_logist.c',
- 'bandwidth_estimator.c',
- 'decode.c',
- 'decode_bwe.c',
- 'decode_plc.c',
- 'encode.c',
- 'entropy_coding.c',
- 'fft.c',
- 'filterbank_tables.c',
- 'filterbanks.c',
- 'filters.c',
- 'initialize.c',
- 'isacfix.c',
- 'lattice.c',
- 'lattice_c.c',
- 'lpc_masking_model.c',
- 'lpc_tables.c',
- 'pitch_estimator.c',
- 'pitch_filter.c',
- 'pitch_gain_tables.c',
- 'pitch_lag_tables.c',
- 'spectrum_ar_model_tables.c',
- 'transform.c',
- 'arith_routins.h',
- 'bandwidth_estimator.h',
- 'codec.h',
- 'entropy_coding.h',
- 'fft.h',
- 'filterbank_tables.h',
- 'lpc_masking_model.h',
- 'lpc_tables.h',
- 'pitch_estimator.h',
- 'pitch_gain_tables.h',
- 'pitch_lag_tables.h',
- 'settings.h',
- 'spectrum_ar_model_tables.h',
- 'structs.h',
- ],
- 'conditions': [
- ['OS!="win"', {
- 'defines': [
- 'WEBRTC_LINUX',
- ],
- }],
- ],
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice.c
deleted file mode 100644
index 8822c6e..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice.c
+++ /dev/null
@@ -1,313 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * lattice.c
- *
- * Contains the normalized lattice filter routines (MA and AR) for iSAC codec
- *
- */
-
-#include "codec.h"
-#include "settings.h"
-
-#define LATTICE_MUL_32_32_RSFT16(a32a, a32b, b32) \
- ((WebRtc_Word32)(WEBRTC_SPL_MUL(a32a, b32) + (WEBRTC_SPL_MUL_16_32_RSFT16(a32b, b32))))
-/* This macro is FORBIDDEN to use elsewhere than in a function in this file and
- its corresponding neon version. It might give unpredictable results, since a
- general WebRtc_Word32*WebRtc_Word32 multiplication results in a 64 bit value.
- The result is then shifted just 16 steps to the right, giving need for 48
- bits, i.e. in the generel case, it will NOT fit in a WebRtc_Word32. In the
- cases used in here, the WebRtc_Word32 will be enough, since (for a good
- reason) the involved multiplicands aren't big enough to overflow a
- WebRtc_Word32 after shifting right 16 bits. I have compared the result of a
- multiplication between t32 and tmp32, done in two ways:
- 1) Using (WebRtc_Word32) (((float)(tmp32))*((float)(tmp32b))/65536.0);
- 2) Using LATTICE_MUL_32_32_RSFT16(t16a, t16b, tmp32b);
- By running 25 files, I haven't found any bigger diff than 64 - this was in the
- case when method 1) gave 650235648 and 2) gave 650235712.
-*/
-
-/* Function prototype: filtering ar_g_Q0[] and ar_f_Q0[] through an AR filter
- with coefficients cth_Q15[] and sth_Q15[].
- Implemented for both generic and ARMv7 platforms.
- */
-void WebRtcIsacfix_FilterArLoop(int16_t* ar_g_Q0,
- int16_t* ar_f_Q0,
- int16_t* cth_Q15,
- int16_t* sth_Q15,
- int16_t order_coef);
-
-/* Inner loop used for function WebRtcIsacfix_NormLatticeFilterMa().
- It does:
- for 0 <= n < HALF_SUBFRAMELEN - 1:
- *ptr2 = input2 * (*ptr2) + input0 * (*ptr0));
- *ptr1 = input1 * (*ptr0) + input0 * (*ptr2);
-*/
-void WebRtcIsacfix_FilterMaLoopC(int16_t input0, // Filter coefficient
- int16_t input1, // Filter coefficient
- int32_t input2, // Inverse coeff. (1/input1)
- int32_t* ptr0, // Sample buffer
- int32_t* ptr1, // Sample buffer
- int32_t* ptr2) { // Sample buffer
- int n = 0;
-
- // Separate the 32-bit variable input2 into two 16-bit integers (high 16 and
- // low 16 bits), for using LATTICE_MUL_32_32_RSFT16 in the loop.
- int16_t t16a = (int16_t)(input2 >> 16);
- int16_t t16b = (int16_t)input2;
- if (t16b < 0) t16a++;
-
- // The loop filtering the samples *ptr0, *ptr1, *ptr2 with filter coefficients
- // input0, input1, and input2.
- for(n = 0; n < HALF_SUBFRAMELEN - 1; n++, ptr0++, ptr1++, ptr2++) {
- int32_t tmp32a = 0;
- int32_t tmp32b = 0;
-
- // Calculate *ptr2 = input2 * (*ptr2 + input0 * (*ptr0));
- tmp32a = WEBRTC_SPL_MUL_16_32_RSFT15(input0, *ptr0); // Q15 * Q15 >> 15 = Q15
- tmp32b = *ptr2 + tmp32a; // Q15 + Q15 = Q15
- *ptr2 = LATTICE_MUL_32_32_RSFT16(t16a, t16b, tmp32b);
-
- // Calculate *ptr1 = input1 * (*ptr0) + input0 * (*ptr2);
- tmp32a = WEBRTC_SPL_MUL_16_32_RSFT15(input1, *ptr0); // Q15*Q15>>15 = Q15
- tmp32b = WEBRTC_SPL_MUL_16_32_RSFT15(input0, *ptr2); // Q15*Q15>>15 = Q15
- *ptr1 = tmp32a + tmp32b; // Q15 + Q15 = Q15
- }
-}
-
-// Declare a function pointer.
-FilterMaLoopFix WebRtcIsacfix_FilterMaLoopFix;
-
-/* filter the signal using normalized lattice filter */
-/* MA filter */
-void WebRtcIsacfix_NormLatticeFilterMa(WebRtc_Word16 orderCoef,
- WebRtc_Word32 *stateGQ15,
- WebRtc_Word16 *lat_inQ0,
- WebRtc_Word16 *filt_coefQ15,
- WebRtc_Word32 *gain_lo_hiQ17,
- WebRtc_Word16 lo_hi,
- WebRtc_Word16 *lat_outQ9)
-{
- WebRtc_Word16 sthQ15[MAX_AR_MODEL_ORDER];
- WebRtc_Word16 cthQ15[MAX_AR_MODEL_ORDER];
-
- int u, i, k, n;
- WebRtc_Word16 temp2,temp3;
- WebRtc_Word16 ord_1 = orderCoef+1;
- WebRtc_Word32 inv_cthQ16[MAX_AR_MODEL_ORDER];
-
- WebRtc_Word32 gain32, fQtmp;
- WebRtc_Word16 gain16;
- WebRtc_Word16 gain_sh;
-
- WebRtc_Word32 tmp32, tmp32b;
- WebRtc_Word32 fQ15vec[HALF_SUBFRAMELEN];
- WebRtc_Word32 gQ15[MAX_AR_MODEL_ORDER+1][HALF_SUBFRAMELEN];
- WebRtc_Word16 sh;
- WebRtc_Word16 t16a;
- WebRtc_Word16 t16b;
-
- for (u=0;u<SUBFRAMES;u++)
- {
- int32_t temp1 = WEBRTC_SPL_MUL_16_16(u, HALF_SUBFRAMELEN);
-
- /* set the Direct Form coefficients */
- temp2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(u, orderCoef);
- temp3 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(2, u)+lo_hi;
-
- /* compute lattice filter coefficients */
- memcpy(sthQ15, &filt_coefQ15[temp2], orderCoef * sizeof(WebRtc_Word16));
-
- WebRtcSpl_SqrtOfOneMinusXSquared(sthQ15, orderCoef, cthQ15);
-
- /* compute the gain */
- gain32 = gain_lo_hiQ17[temp3];
- gain_sh = WebRtcSpl_NormW32(gain32);
- gain32 = WEBRTC_SPL_LSHIFT_W32(gain32, gain_sh); //Q(17+gain_sh)
-
- for (k=0;k<orderCoef;k++)
- {
- gain32 = WEBRTC_SPL_MUL_16_32_RSFT15(cthQ15[k], gain32); //Q15*Q(17+gain_sh)>>15 = Q(17+gain_sh)
- inv_cthQ16[k] = WebRtcSpl_DivW32W16((WebRtc_Word32)2147483647, cthQ15[k]); // 1/cth[k] in Q31/Q15 = Q16
- }
- gain16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(gain32, 16); //Q(1+gain_sh)
-
- /* normalized lattice filter */
- /*****************************/
-
- /* initial conditions */
- for (i=0;i<HALF_SUBFRAMELEN;i++)
- {
- fQ15vec[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)lat_inQ0[i + temp1], 15); //Q15
- gQ15[0][i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)lat_inQ0[i + temp1], 15); //Q15
- }
-
-
- fQtmp = fQ15vec[0];
-
- /* get the state of f&g for the first input, for all orders */
- for (i=1;i<ord_1;i++)
- {
- // Calculate f[i][0] = inv_cth[i-1]*(f[i-1][0] + sth[i-1]*stateG[i-1]);
- tmp32 = WEBRTC_SPL_MUL_16_32_RSFT15(sthQ15[i-1], stateGQ15[i-1]);//Q15*Q15>>15 = Q15
- tmp32b= fQtmp + tmp32; //Q15+Q15=Q15
- tmp32 = inv_cthQ16[i-1]; //Q16
- t16a = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
- t16b = (WebRtc_Word16) (tmp32-WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)t16a), 16));
- if (t16b<0) t16a++;
- tmp32 = LATTICE_MUL_32_32_RSFT16(t16a, t16b, tmp32b);
- fQtmp = tmp32; // Q15
-
- // Calculate g[i][0] = cth[i-1]*stateG[i-1] + sth[i-1]* f[i][0];
- tmp32 = WEBRTC_SPL_MUL_16_32_RSFT15(cthQ15[i-1], stateGQ15[i-1]); //Q15*Q15>>15 = Q15
- tmp32b = WEBRTC_SPL_MUL_16_32_RSFT15(sthQ15[i-1], fQtmp); //Q15*Q15>>15 = Q15
- tmp32 = tmp32 + tmp32b;//Q15+Q15 = Q15
- gQ15[i][0] = tmp32; // Q15
- }
-
- /* filtering */
- /* save the states */
- for(k=0;k<orderCoef;k++)
- {
- // for 0 <= n < HALF_SUBFRAMELEN - 1:
- // f[k+1][n+1] = inv_cth[k]*(f[k][n+1] + sth[k]*g[k][n]);
- // g[k+1][n+1] = cth[k]*g[k][n] + sth[k]* f[k+1][n+1];
- WebRtcIsacfix_FilterMaLoopFix(sthQ15[k], cthQ15[k], inv_cthQ16[k],
- &gQ15[k][0], &gQ15[k+1][1], &fQ15vec[1]);
- }
-
- fQ15vec[0] = fQtmp;
-
- for(n=0;n<HALF_SUBFRAMELEN;n++)
- {
- //gain32 = WEBRTC_SPL_RSHIFT_W32(gain32, gain_sh); // Q(17+gain_sh) -> Q17
- tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(gain16, fQ15vec[n]); //Q(1+gain_sh)*Q15>>16 = Q(gain_sh)
- sh = 9-gain_sh; //number of needed shifts to reach Q9
- t16a = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(tmp32, sh);
- lat_outQ9[n + temp1] = t16a;
- }
-
- /* save the states */
- for (i=0;i<ord_1;i++)
- {
- stateGQ15[i] = gQ15[i][HALF_SUBFRAMELEN-1];
- }
- //process next frame
- }
-
- return;
-}
-
-
-
-
-
-/* ----------------AR filter-------------------------*/
-/* filter the signal using normalized lattice filter */
-void WebRtcIsacfix_NormLatticeFilterAr(WebRtc_Word16 orderCoef,
- WebRtc_Word16 *stateGQ0,
- WebRtc_Word32 *lat_inQ25,
- WebRtc_Word16 *filt_coefQ15,
- WebRtc_Word32 *gain_lo_hiQ17,
- WebRtc_Word16 lo_hi,
- WebRtc_Word16 *lat_outQ0)
-{
- int ii,n,k,i,u;
- WebRtc_Word16 sthQ15[MAX_AR_MODEL_ORDER];
- WebRtc_Word16 cthQ15[MAX_AR_MODEL_ORDER];
- WebRtc_Word32 tmp32;
-
-
- WebRtc_Word16 tmpAR;
- WebRtc_Word16 ARfQ0vec[HALF_SUBFRAMELEN];
- WebRtc_Word16 ARgQ0vec[MAX_AR_MODEL_ORDER+1];
-
- WebRtc_Word32 inv_gain32;
- WebRtc_Word16 inv_gain16;
- WebRtc_Word16 den16;
- WebRtc_Word16 sh;
-
- WebRtc_Word16 temp2,temp3;
- WebRtc_Word16 ord_1 = orderCoef+1;
-
- for (u=0;u<SUBFRAMES;u++)
- {
- int32_t temp1 = WEBRTC_SPL_MUL_16_16(u, HALF_SUBFRAMELEN);
-
- //set the denominator and numerator of the Direct Form
- temp2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(u, orderCoef);
- temp3 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(2, u) + lo_hi;
-
- for (ii=0; ii<orderCoef; ii++) {
- sthQ15[ii] = filt_coefQ15[temp2+ii];
- }
-
- WebRtcSpl_SqrtOfOneMinusXSquared(sthQ15, orderCoef, cthQ15);
-
- /* Simulation of the 25 files shows that maximum value in
- the vector gain_lo_hiQ17[] is 441344, which means that
- it is log2((2^31)/441344) = 12.2 shifting bits from
- saturation. Therefore, it should be safe to use Q27 instead
- of Q17. */
-
- tmp32 = WEBRTC_SPL_LSHIFT_W32(gain_lo_hiQ17[temp3], 10); // Q27
-
- for (k=0;k<orderCoef;k++) {
- tmp32 = WEBRTC_SPL_MUL_16_32_RSFT15(cthQ15[k], tmp32); // Q15*Q27>>15 = Q27
- }
-
- sh = WebRtcSpl_NormW32(tmp32); // tmp32 is the gain
- den16 = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(tmp32, sh-16); //Q(27+sh-16) = Q(sh+11) (all 16 bits are value bits)
- inv_gain32 = WebRtcSpl_DivW32W16((WebRtc_Word32)2147483647, den16); // 1/gain in Q31/Q(sh+11) = Q(20-sh)
-
- //initial conditions
- inv_gain16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(inv_gain32, 2); // 1/gain in Q(20-sh-2) = Q(18-sh)
-
- for (i=0;i<HALF_SUBFRAMELEN;i++)
- {
-
- tmp32 = WEBRTC_SPL_LSHIFT_W32(lat_inQ25[i + temp1], 1); //Q25->Q26
- tmp32 = WEBRTC_SPL_MUL_16_32_RSFT16(inv_gain16, tmp32); //lat_in[]*inv_gain in (Q(18-sh)*Q26)>>16 = Q(28-sh)
- tmp32 = WEBRTC_SPL_SHIFT_W32(tmp32, -(28-sh)); // lat_in[]*inv_gain in Q0
-
- ARfQ0vec[i] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp32); // Q0
- }
-
- for (i=orderCoef-1;i>=0;i--) //get the state of f&g for the first input, for all orders
- {
- tmp32 = WEBRTC_SPL_RSHIFT_W32(((WEBRTC_SPL_MUL_16_16(cthQ15[i],ARfQ0vec[0])) - (WEBRTC_SPL_MUL_16_16(sthQ15[i],stateGQ0[i])) + 16384), 15);
- tmpAR = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp32); // Q0
-
- tmp32 = WEBRTC_SPL_RSHIFT_W32(((WEBRTC_SPL_MUL_16_16(sthQ15[i],ARfQ0vec[0])) + (WEBRTC_SPL_MUL_16_16(cthQ15[i], stateGQ0[i])) + 16384), 15);
- ARgQ0vec[i+1] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp32); // Q0
- ARfQ0vec[0] = tmpAR;
- }
- ARgQ0vec[0] = ARfQ0vec[0];
-
- // Filter ARgQ0vec[] and ARfQ0vec[] through coefficients cthQ15[] and sthQ15[].
- WebRtcIsacfix_FilterArLoop(ARgQ0vec, ARfQ0vec, cthQ15, sthQ15, orderCoef);
-
- for(n=0;n<HALF_SUBFRAMELEN;n++)
- {
- lat_outQ0[n + temp1] = ARfQ0vec[n];
- }
-
-
- /* cannot use memcpy in the following */
-
- for (i=0;i<ord_1;i++)
- {
- stateGQ0[i] = ARgQ0vec[i];
- }
- }
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_armv7.S b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_armv7.S
deleted file mode 100644
index 1cd3a76..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_armv7.S
+++ /dev/null
@@ -1,82 +0,0 @@
-@
-@ Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-@
-@ Use of this source code is governed by a BSD-style license
-@ that can be found in the LICENSE file in the root of the source
-@ tree. An additional intellectual property rights grant can be found
-@ in the file PATENTS. All contributing project authors may
-@ be found in the AUTHORS file in the root of the source tree.
-@
-
-@ Contains a function for the core loop in the normalized lattice AR
-@ filter routine for iSAC codec, optimized for ARMv7 platforms.
-@
-@ Output is bit-exact with the reference C code in lattic_c.c
-@
-@ Register usage:
-@
-@ r0: &ar_g_Q0
-@ r1: &ar_f_Q0
-@ r2: &cth_Q15
-@ r3: &sth_Q15
-@ r4: out loop counter
-@ r5: tmpAR
-@ r9: inner loop counter
-@ r12: constant #16384
-@ r6, r7, r8, r10, r11: scratch
-
-#include "settings.h"
-
-.arch armv7-a
-.global WebRtcIsacfix_FilterArLoop
-.align 2
-
-WebRtcIsacfix_FilterArLoop:
-.fnstart
-
-.save {r4-r11}
- push {r4-r11}
-
- add r1, #2 @ &ar_f_Q0[1]
- mov r12, #16384
- mov r4, #HALF_SUBFRAMELEN
- sub r4, #1 @ Outer loop counter = HALF_SUBFRAMELEN - 1
-
-HALF_SUBFRAME_LOOP: @ for(n = 0; n < HALF_SUBFRAMELEN - 1; n++)
-
- ldr r9, [sp, #32] @ Restore the inner loop counter to order_coef
- ldrh r5, [r1] @ tmpAR = ar_f_Q0[n+1]
- add r0, r9, asl #1 @ Restore r0 to &ar_g_Q0[order_coef]
- add r2, r9, asl #1 @ Restore r2 to &cth_Q15[order_coef]
- add r3, r9, asl #1 @ Restore r3 to &sth_Q15[order_coef]
-
-ORDER_COEF_LOOP: @ for(k = order_coef - 1 ; k >= 0; k--)
-
- ldrh r7, [r3, #-2]! @ sth_Q15[k]
- ldrh r6, [r2, #-2]! @ cth_Q15[k]
-
- ldrh r8, [r0, #-2] @ ar_g_Q0[k]
- smlabb r11, r7, r5, r12 @ sth_Q15[k] * tmpAR + 16384
- smlabb r10, r6, r5, r12 @ cth_Q15[k] * tmpAR + 16384
- smulbb r7, r7, r8 @ sth_Q15[k] * ar_g_Q0[k]
- smlabb r11, r6, r8, r11 @ cth_Q15[k]*ar_g_Q0[k]+(sth_Q15[k]*tmpAR+16384)
-
- sub r10, r10, r7 @ cth_Q15[k]*tmpAR+16384-(sth_Q15[k]*ar_g_Q0[k])
- ssat r11, #16, r11, asr #15
- ssat r5, #16, r10, asr #15
- strh r11, [r0], #-2 @ Output: ar_g_Q0[k+1]
-
- subs r9, #1
- bgt ORDER_COEF_LOOP
-
- strh r5, [r0] @ Output: ar_g_Q0[0] = tmpAR;
- strh r5, [r1], #2 @ Output: ar_f_Q0[n+1] = tmpAR;
-
- subs r4, #1
- bne HALF_SUBFRAME_LOOP
-
- pop {r4-r11}
- bx lr
-
-.fnend
-
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_c.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_c.c
deleted file mode 100644
index 80ccf39..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_c.c
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Contains the core loop function for the lattice filter AR routine
- * for iSAC codec.
- *
- */
-
-#include "settings.h"
-#include "signal_processing_library.h"
-#include "typedefs.h"
-
-/* Filter ar_g_Q0[] and ar_f_Q0[] through an AR filter with coefficients
- * cth_Q15[] and sth_Q15[].
- */
-void WebRtcIsacfix_FilterArLoop(int16_t* ar_g_Q0, // Input samples
- int16_t* ar_f_Q0, // Input samples
- int16_t* cth_Q15, // Filter coefficients
- int16_t* sth_Q15, // Filter coefficients
- int16_t order_coef) { // order of the filter
- int n = 0;
-
- for (n = 0; n < HALF_SUBFRAMELEN - 1; n++) {
- int k = 0;
- int16_t tmpAR = 0;
- int32_t tmp32 = 0;
- int32_t tmp32_2 = 0;
-
- tmpAR = ar_f_Q0[n + 1];
- for (k = order_coef - 1; k >= 0; k--) {
- tmp32 = WEBRTC_SPL_RSHIFT_W32(((WEBRTC_SPL_MUL_16_16(cth_Q15[k], tmpAR))
- - (WEBRTC_SPL_MUL_16_16(sth_Q15[k], ar_g_Q0[k])) + 16384), 15);
- tmp32_2 = WEBRTC_SPL_RSHIFT_W32(((WEBRTC_SPL_MUL_16_16(sth_Q15[k], tmpAR))
- + (WEBRTC_SPL_MUL_16_16(cth_Q15[k], ar_g_Q0[k])) + 16384), 15);
- tmpAR = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp32);
- ar_g_Q0[k + 1] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmp32_2);
- }
- ar_f_Q0[n + 1] = tmpAR;
- ar_g_Q0[0] = tmpAR;
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_neon.S b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_neon.S
deleted file mode 100644
index a59b6e3..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lattice_neon.S
+++ /dev/null
@@ -1,155 +0,0 @@
-@
-@ Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-@
-@ Use of this source code is governed by a BSD-style license
-@ that can be found in the LICENSE file in the root of the source
-@ tree. An additional intellectual property rights grant can be found
-@ in the file PATENTS. All contributing project authors may
-@ be found in the AUTHORS file in the root of the source tree.
-@
-
-@ lattice_neon.s
-@
-@ Contains a function for the core loop in the normalized lattice MA
-@ filter routine for iSAC codec, optimized for ARM Neon platform.
-@ void WebRtcIsacfix_FilterMaLoopNeon(int16_t input0,
-@ int16_t input1,
-@ int32_t input2,
-@ int32_t* ptr0,
-@ int32_t* ptr1,
-@ int32_t* __restrict ptr2);
-@ It calculates
-@ *ptr2 = input2 * (*ptr2) + input0 * (*ptr0));
-@ *ptr1 = input1 * (*ptr0) + input0 * (*ptr2);
-@ in Q15 domain.
-@
-@ Reference code in lattice.c.
-@ Output is not bit-exact with the reference C code, due to the replacement
-@ of WEBRTC_SPL_MUL_16_32_RSFT15 and LATTICE_MUL_32_32_RSFT16 with Neon
-@ instructions, smulwb, and smull. Speech quality was not degraded by
-@ testing speech and tone vectors.
-
-.arch armv7-a
-.fpu neon
-
-#include "settings.h"
-
-.global WebRtcIsacfix_FilterMaLoopNeon
-
-.align 2
-
-WebRtcIsacfix_FilterMaLoopNeon:
-.fnstart
-
-.save {r4-r8}
- push {r4-r8}
-
- vdup.32 d28, r0 @ Initialize Neon register with input0
- vdup.32 d29, r1 @ Initialize Neon register with input1
- vdup.32 d30, r2 @ Initialize Neon register with input2
- ldr r4, [sp, #20] @ ptr1
- ldr r12, [sp, #24] @ ptr2
-
- @ Number of loop iterations after unrolling: r5 = (HALF_SUBFRAMELEN - 1) >> 2
- @ Leftover samples after the loop, in r6:
- @ r6 = (HALF_SUBFRAMELEN - 1) - (HALF_SUBFRAMELEN - 1) >> 2 << 2
- mov r6, #HALF_SUBFRAMELEN
- sub r6, #1
- lsr r5, r6, #2
- sub r6, r5, lsl #2
-
- @ First r5 iterations in a loop.
-
-LOOP:
- vld1.32 {d0, d1}, [r3]! @ *ptr0
-
- vmull.s32 q10, d0, d28 @ tmp32a = input0 * (*ptr0)
- vmull.s32 q11, d1, d28 @ tmp32a = input0 * (*ptr0)
- vmull.s32 q12, d0, d29 @ input1 * (*ptr0)
- vmull.s32 q13, d1, d29 @ input1 * (*ptr0)
-
- vrshrn.i64 d4, q10, #15
- vrshrn.i64 d5, q11, #15
-
- vld1.32 {d2, d3}, [r12] @ *ptr2
- vadd.i32 q3, q2, q1 @ tmp32b = *ptr2 + tmp32a
-
- vrshrn.i64 d0, q12, #15
-
- vmull.s32 q10, d6, d30 @ input2 * (*ptr2 + tmp32b)
- vmull.s32 q11, d7, d30 @ input2 * (*ptr2 + tmp32b)
-
- vrshrn.i64 d16, q10, #16
- vrshrn.i64 d17, q11, #16
-
- vmull.s32 q10, d16, d28 @ input0 * (*ptr2)
- vmull.s32 q11, d17, d28 @ input0 * (*ptr2)
-
- vrshrn.i64 d1, q13, #15
- vrshrn.i64 d18, q10, #15
- vrshrn.i64 d19, q11, #15
-
- vst1.32 {d16, d17}, [r12]! @ *ptr2
-
- vadd.i32 q9, q0, q9
- subs r5, #1
- vst1.32 {d18, d19}, [r4]! @ *ptr1
-
- bgt LOOP
-
- @ Check how many samples still need to be processed.
- subs r6, #2
- blt LAST_SAMPLE
-
- @ Process two more samples:
- vld1.32 d0, [r3]! @ *ptr0
-
- vmull.s32 q11, d0, d28 @ tmp32a = input0 * (*ptr0)
- vmull.s32 q13, d0, d29 @ input1 * (*ptr0)
-
- vld1.32 d18, [r12] @ *ptr2
- vrshrn.i64 d4, q11, #15
-
- vadd.i32 d7, d4, d18 @ tmp32b = *ptr2 + tmp32a
- vmull.s32 q11, d7, d30 @ input2 * (*ptr2 + tmp32b)
- vrshrn.i64 d16, q11, #16
-
- vmull.s32 q11, d16, d28 @ input0 * (*ptr2)
- vst1.32 d16, [r12]! @ *ptr2
-
- vrshrn.i64 d0, q13, #15
- vrshrn.i64 d19, q11, #15
- vadd.i32 d19, d0, d19
-
- vst1.32 d19, [r4]! @ *ptr1
-
- @ If there's still one more sample, process it here.
-LAST_SAMPLE:
- cmp r6, #1
- bne END
-
- @ *ptr2 = input2 * (*ptr2 + input0 * (*ptr0));
-
- ldr r7, [r3] @ *ptr0
- ldr r8, [r12] @ *ptr2
-
- smulwb r5, r7, r0 @ tmp32a = *ptr0 * input0 >> 16
- add r8, r8, r5, lsl #1 @ tmp32b = *ptr2 + (tmp32a << 1)
- smull r5, r6, r8, r2 @ tmp32b * input2, in 64 bits
- lsl r6, #16
- add r6, r5, lsr #16 @ Only take the middle 32 bits
- str r6, [r12] @ Output (*ptr2, as 32 bits)
-
- @ *ptr1 = input1 * (*ptr0) + input0 * (*ptr2);
-
- smulwb r5, r7, r1 @ tmp32a = *ptr0 * input1 >> 16
- smulwb r6, r6, r0 @ tmp32b = *ptr2 * input0 >> 16
- lsl r5, r5, #1
- add r5, r6, lsl #1
- str r5, [r4] @ Output (*ptr1)
-
-END:
- pop {r4-r8}
- bx lr
-
-.fnend
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_masking_model.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_masking_model.c
deleted file mode 100644
index 4fa8ebb..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_masking_model.c
+++ /dev/null
@@ -1,1035 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * lpc_masking_model.c
- *
- * LPC analysis and filtering functions
- *
- */
-
-#include "codec.h"
-#include "entropy_coding.h"
-#include "settings.h"
-
-
-/* The conversion is implemented by the step-down algorithm */
-void WebRtcSpl_AToK_JSK(
- WebRtc_Word16 *a16, /* Q11 */
- WebRtc_Word16 useOrder,
- WebRtc_Word16 *k16 /* Q15 */
- )
-{
- int m, k;
- WebRtc_Word32 tmp32[MAX_AR_MODEL_ORDER];
- WebRtc_Word32 tmp32b;
- WebRtc_Word32 tmp_inv_denum32;
- WebRtc_Word16 tmp_inv_denum16;
-
- k16[useOrder-1]= WEBRTC_SPL_LSHIFT_W16(a16[useOrder], 4); //Q11<<4 => Q15
-
- for (m=useOrder-1; m>0; m--) {
- tmp_inv_denum32 = ((WebRtc_Word32) 1073741823) - WEBRTC_SPL_MUL_16_16(k16[m], k16[m]); // (1 - k^2) in Q30
- tmp_inv_denum16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp_inv_denum32, 15); // (1 - k^2) in Q15
-
- for (k=1; k<=m; k++) {
- tmp32b = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)a16[k], 16) -
- WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(k16[m], a16[m-k+1]), 1);
-
- tmp32[k] = WebRtcSpl_DivW32W16(tmp32b, tmp_inv_denum16); //Q27/Q15 = Q12
- }
-
- for (k=1; k<m; k++) {
- a16[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32[k], 1); //Q12>>1 => Q11
- }
-
- tmp32[m] = WEBRTC_SPL_SAT(4092, tmp32[m], -4092);
- k16[m-1] = (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W32(tmp32[m], 3); //Q12<<3 => Q15
- }
-
- return;
-}
-
-
-
-
-
-WebRtc_Word16 WebRtcSpl_LevinsonW32_JSK(
- WebRtc_Word32 *R, /* (i) Autocorrelation of length >= order+1 */
- WebRtc_Word16 *A, /* (o) A[0..order] LPC coefficients (Q11) */
- WebRtc_Word16 *K, /* (o) K[0...order-1] Reflection coefficients (Q15) */
- WebRtc_Word16 order /* (i) filter order */
- ) {
- WebRtc_Word16 i, j;
- WebRtc_Word16 R_hi[LEVINSON_MAX_ORDER+1], R_low[LEVINSON_MAX_ORDER+1];
- /* Aurocorr coefficients in high precision */
- WebRtc_Word16 A_hi[LEVINSON_MAX_ORDER+1], A_low[LEVINSON_MAX_ORDER+1];
- /* LPC coefficients in high precicion */
- WebRtc_Word16 A_upd_hi[LEVINSON_MAX_ORDER+1], A_upd_low[LEVINSON_MAX_ORDER+1];
- /* LPC coefficients for next iteration */
- WebRtc_Word16 K_hi, K_low; /* reflection coefficient in high precision */
- WebRtc_Word16 Alpha_hi, Alpha_low, Alpha_exp; /* Prediction gain Alpha in high precision
- and with scale factor */
- WebRtc_Word16 tmp_hi, tmp_low;
- WebRtc_Word32 temp1W32, temp2W32, temp3W32;
- WebRtc_Word16 norm;
-
- /* Normalize the autocorrelation R[0]...R[order+1] */
-
- norm = WebRtcSpl_NormW32(R[0]);
-
- for (i=order;i>=0;i--) {
- temp1W32 = WEBRTC_SPL_LSHIFT_W32(R[i], norm);
- /* Put R in hi and low format */
- R_hi[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
- R_low[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16)), 1);
- }
-
- /* K = A[1] = -R[1] / R[0] */
-
- temp2W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[1],16) +
- WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[1],1); /* R[1] in Q31 */
- temp3W32 = WEBRTC_SPL_ABS_W32(temp2W32); /* abs R[1] */
- temp1W32 = WebRtcSpl_DivW32HiLow(temp3W32, R_hi[0], R_low[0]); /* abs(R[1])/R[0] in Q31 */
- /* Put back the sign on R[1] */
- if (temp2W32 > 0) {
- temp1W32 = -temp1W32;
- }
-
- /* Put K in hi and low format */
- K_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
- K_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1);
-
- /* Store first reflection coefficient */
- K[0] = K_hi;
-
- temp1W32 = WEBRTC_SPL_RSHIFT_W32(temp1W32, 4); /* A[1] in Q27 */
-
- /* Put A[1] in hi and low format */
- A_hi[1] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
- A_low[1] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[1], 16)), 1);
-
- /* Alpha = R[0] * (1-K^2) */
-
- temp1W32 = WEBRTC_SPL_LSHIFT_W32((WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(K_hi, K_low), 14) +
- WEBRTC_SPL_MUL_16_16(K_hi, K_hi)), 1); /* temp1W32 = k^2 in Q31 */
-
- temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); /* Guard against <0 */
- temp1W32 = (WebRtc_Word32)0x7fffffffL - temp1W32; /* temp1W32 = (1 - K[0]*K[0]) in Q31 */
-
- /* Store temp1W32 = 1 - K[0]*K[0] on hi and low format */
- tmp_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
- tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
-
- /* Calculate Alpha in Q31 */
- temp1W32 = WEBRTC_SPL_LSHIFT_W32((WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_hi) +
- WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(R_hi[0], tmp_low), 15) +
- WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(R_low[0], tmp_hi), 15) ), 1);
-
- /* Normalize Alpha and put it in hi and low format */
-
- Alpha_exp = WebRtcSpl_NormW32(temp1W32);
- temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, Alpha_exp);
- Alpha_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
- Alpha_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1);
-
- /* Perform the iterative calculations in the
- Levinson Durbin algorithm */
-
- for (i=2; i<=order; i++)
- {
-
- /* ----
- \
- temp1W32 = R[i] + > R[j]*A[i-j]
- /
- ----
- j=1..i-1
- */
-
- temp1W32 = 0;
-
- for(j=1; j<i; j++) {
- /* temp1W32 is in Q31 */
- temp1W32 += (WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(R_hi[j], A_hi[i-j]), 1) +
- WEBRTC_SPL_LSHIFT_W32(( WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(R_hi[j], A_low[i-j]), 15) +
- WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(R_low[j], A_hi[i-j]), 15) ), 1));
- }
-
- temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, 4);
- temp1W32 += (WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_hi[i], 16) +
- WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)R_low[i], 1));
-
- /* K = -temp1W32 / Alpha */
- temp2W32 = WEBRTC_SPL_ABS_W32(temp1W32); /* abs(temp1W32) */
- temp3W32 = WebRtcSpl_DivW32HiLow(temp2W32, Alpha_hi, Alpha_low); /* abs(temp1W32)/Alpha */
-
- /* Put the sign of temp1W32 back again */
- if (temp1W32 > 0) {
- temp3W32 = -temp3W32;
- }
-
- /* Use the Alpha shifts from earlier to denormalize */
- norm = WebRtcSpl_NormW32(temp3W32);
- if ((Alpha_exp <= norm)||(temp3W32==0)) {
- temp3W32 = WEBRTC_SPL_LSHIFT_W32(temp3W32, Alpha_exp);
- } else {
- if (temp3W32 > 0)
- {
- temp3W32 = (WebRtc_Word32)0x7fffffffL;
- } else
- {
- temp3W32 = (WebRtc_Word32)0x80000000L;
- }
- }
-
- /* Put K on hi and low format */
- K_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
- K_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp3W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)K_hi, 16)), 1);
-
- /* Store Reflection coefficient in Q15 */
- K[i-1] = K_hi;
-
- /* Test for unstable filter. If unstable return 0 and let the
- user decide what to do in that case
- */
-
- if ((WebRtc_Word32)WEBRTC_SPL_ABS_W16(K_hi) > (WebRtc_Word32)32740) {
- return(-i); /* Unstable filter */
- }
-
- /*
- Compute updated LPC coefficient: Anew[i]
- Anew[j]= A[j] + K*A[i-j] for j=1..i-1
- Anew[i]= K
- */
-
- for(j=1; j<i; j++)
- {
- temp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[j],16) +
- WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_low[j],1); /* temp1W32 = A[j] in Q27 */
-
- temp1W32 += WEBRTC_SPL_LSHIFT_W32(( WEBRTC_SPL_MUL_16_16(K_hi, A_hi[i-j]) +
- WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(K_hi, A_low[i-j]), 15) +
- WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(K_low, A_hi[i-j]), 15) ), 1); /* temp1W32 += K*A[i-j] in Q27 */
-
- /* Put Anew in hi and low format */
- A_upd_hi[j] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
- A_upd_low[j] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_upd_hi[j], 16)), 1);
- }
-
- temp3W32 = WEBRTC_SPL_RSHIFT_W32(temp3W32, 4); /* temp3W32 = K in Q27 (Convert from Q31 to Q27) */
-
- /* Store Anew in hi and low format */
- A_upd_hi[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp3W32, 16);
- A_upd_low[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp3W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_upd_hi[i], 16)), 1);
-
- /* Alpha = Alpha * (1-K^2) */
-
- temp1W32 = WEBRTC_SPL_LSHIFT_W32((WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(K_hi, K_low), 14) +
- WEBRTC_SPL_MUL_16_16(K_hi, K_hi)), 1); /* K*K in Q31 */
-
- temp1W32 = WEBRTC_SPL_ABS_W32(temp1W32); /* Guard against <0 */
- temp1W32 = (WebRtc_Word32)0x7fffffffL - temp1W32; /* 1 - K*K in Q31 */
-
- /* Convert 1- K^2 in hi and low format */
- tmp_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
- tmp_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tmp_hi, 16)), 1);
-
- /* Calculate Alpha = Alpha * (1-K^2) in Q31 */
- temp1W32 = WEBRTC_SPL_LSHIFT_W32(( WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_hi) +
- WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(Alpha_hi, tmp_low), 15) +
- WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(Alpha_low, tmp_hi), 15)), 1);
-
- /* Normalize Alpha and store it on hi and low format */
-
- norm = WebRtcSpl_NormW32(temp1W32);
- temp1W32 = WEBRTC_SPL_LSHIFT_W32(temp1W32, norm);
-
- Alpha_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(temp1W32, 16);
- Alpha_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((temp1W32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)Alpha_hi, 16)), 1);
-
- /* Update the total nomalization of Alpha */
- Alpha_exp = Alpha_exp + norm;
-
- /* Update A[] */
-
- for(j=1; j<=i; j++)
- {
- A_hi[j] =A_upd_hi[j];
- A_low[j] =A_upd_low[j];
- }
- }
-
- /*
- Set A[0] to 1.0 and store the A[i] i=1...order in Q12
- (Convert from Q27 and use rounding)
- */
-
- A[0] = 2048;
-
- for(i=1; i<=order; i++) {
- /* temp1W32 in Q27 */
- temp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_hi[i], 16) +
- WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)A_low[i], 1);
- /* Round and store upper word */
- A[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(temp1W32+(WebRtc_Word32)32768, 16);
- }
- return(1); /* Stable filters */
-}
-
-
-
-
-
-/* window */
-/* Matlab generation of floating point code:
- * t = (1:256)/257; r = 1-(1-t).^.45; w = sin(r*pi).^3; w = w/sum(w); plot((1:256)/8, w); grid;
- * for k=1:16, fprintf(1, '%.8f, ', w(k*16 + (-15:0))); fprintf(1, '\n'); end
- * All values are multiplyed with 2^21 in fixed point code.
- */
-static const WebRtc_Word16 kWindowAutocorr[WINLEN] = {
- 0, 0, 0, 0, 0, 1, 1, 2, 2, 3, 5, 6,
- 8, 10, 12, 14, 17, 20, 24, 28, 33, 38, 43, 49,
- 56, 63, 71, 79, 88, 98, 108, 119, 131, 143, 157, 171,
- 186, 202, 219, 237, 256, 275, 296, 318, 341, 365, 390, 416,
- 444, 472, 502, 533, 566, 600, 635, 671, 709, 748, 789, 831,
- 875, 920, 967, 1015, 1065, 1116, 1170, 1224, 1281, 1339, 1399, 1461,
- 1525, 1590, 1657, 1726, 1797, 1870, 1945, 2021, 2100, 2181, 2263, 2348,
- 2434, 2523, 2614, 2706, 2801, 2898, 2997, 3099, 3202, 3307, 3415, 3525,
- 3637, 3751, 3867, 3986, 4106, 4229, 4354, 4481, 4611, 4742, 4876, 5012,
- 5150, 5291, 5433, 5578, 5725, 5874, 6025, 6178, 6333, 6490, 6650, 6811,
- 6974, 7140, 7307, 7476, 7647, 7820, 7995, 8171, 8349, 8529, 8711, 8894,
- 9079, 9265, 9453, 9642, 9833, 10024, 10217, 10412, 10607, 10803, 11000, 11199,
- 11398, 11597, 11797, 11998, 12200, 12401, 12603, 12805, 13008, 13210, 13412, 13614,
- 13815, 14016, 14216, 14416, 14615, 14813, 15009, 15205, 15399, 15591, 15782, 15971,
- 16157, 16342, 16524, 16704, 16881, 17056, 17227, 17395, 17559, 17720, 17877, 18030,
- 18179, 18323, 18462, 18597, 18727, 18851, 18970, 19082, 19189, 19290, 19384, 19471,
- 19551, 19623, 19689, 19746, 19795, 19835, 19867, 19890, 19904, 19908, 19902, 19886,
- 19860, 19823, 19775, 19715, 19644, 19561, 19465, 19357, 19237, 19102, 18955, 18793,
- 18618, 18428, 18223, 18004, 17769, 17518, 17252, 16970, 16672, 16357, 16025, 15677,
- 15311, 14929, 14529, 14111, 13677, 13225, 12755, 12268, 11764, 11243, 10706, 10152,
- 9583, 8998, 8399, 7787, 7162, 6527, 5883, 5231, 4576, 3919, 3265, 2620,
- 1990, 1386, 825, 333
-};
-
-
-/* By using a hearing threshold level in dB of -28 dB (higher value gives more noise),
- the H_T_H (in float) can be calculated as:
- H_T_H = pow(10.0, 0.05 * (-28.0)) = 0.039810717055350
- In Q19, H_T_H becomes round(0.039810717055350*2^19) ~= 20872, i.e.
- H_T_H = 20872/524288.0, and H_T_HQ19 = 20872;
-*/
-
-
-/* The bandwidth expansion vectors are created from:
- kPolyVecLo=[0.900000,0.810000,0.729000,0.656100,0.590490,0.531441,0.478297,0.430467,0.387420,0.348678,0.313811,0.282430];
- kPolyVecHi=[0.800000,0.640000,0.512000,0.409600,0.327680,0.262144];
- round(kPolyVecLo*32768)
- round(kPolyVecHi*32768)
-*/
-static const WebRtc_Word16 kPolyVecLo[12] = {
- 29491, 26542, 23888, 21499, 19349, 17414, 15673, 14106, 12695, 11425, 10283, 9255
-};
-static const WebRtc_Word16 kPolyVecHi[6] = {
- 26214, 20972, 16777, 13422, 10737, 8590
-};
-
-static __inline WebRtc_Word32 log2_Q8_LPC( WebRtc_UWord32 x ) {
-
- WebRtc_Word32 zeros, lg2;
- WebRtc_Word16 frac;
-
- zeros=WebRtcSpl_NormU32(x);
- frac=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(((WebRtc_UWord32)WEBRTC_SPL_LSHIFT_W32(x, zeros)&0x7FFFFFFF), 23);
-
- /* log2(x) */
-
- lg2= (WEBRTC_SPL_LSHIFT_W16((31-zeros), 8)+frac);
- return lg2;
-
-}
-
-static const WebRtc_Word16 kMulPitchGain = -25; /* 200/256 in Q5 */
-static const WebRtc_Word16 kChngFactor = 3523; /* log10(2)*10/4*0.4/1.4=log10(2)/1.4= 0.2150 in Q14 */
-static const WebRtc_Word16 kExp2 = 11819; /* 1/log(2) */
-
-void WebRtcIsacfix_GetVars(const WebRtc_Word16 *input, const WebRtc_Word16 *pitchGains_Q12,
- WebRtc_UWord32 *oldEnergy, WebRtc_Word16 *varscale)
-{
- int k;
- WebRtc_UWord32 nrgQ[4];
- WebRtc_Word16 nrgQlog[4];
- WebRtc_Word16 tmp16, chng1, chng2, chng3, chng4, tmp, chngQ, oldNrgQlog, pgQ, pg3;
- WebRtc_Word32 expPg32;
- WebRtc_Word16 expPg, divVal;
- WebRtc_Word16 tmp16_1, tmp16_2;
-
- /* Calculate energies of first and second frame halfs */
- nrgQ[0]=0;
- for (k = QLOOKAHEAD/2; k < (FRAMESAMPLES/4 + QLOOKAHEAD) / 2; k++) {
- nrgQ[0] +=WEBRTC_SPL_MUL_16_16(input[k],input[k]);
- }
- nrgQ[1]=0;
- for ( ; k < (FRAMESAMPLES/2 + QLOOKAHEAD) / 2; k++) {
- nrgQ[1] +=WEBRTC_SPL_MUL_16_16(input[k],input[k]);
- }
- nrgQ[2]=0;
- for ( ; k < (WEBRTC_SPL_MUL_16_16(FRAMESAMPLES, 3)/4 + QLOOKAHEAD) / 2; k++) {
- nrgQ[2] +=WEBRTC_SPL_MUL_16_16(input[k],input[k]);
- }
- nrgQ[3]=0;
- for ( ; k < (FRAMESAMPLES + QLOOKAHEAD) / 2; k++) {
- nrgQ[3] +=WEBRTC_SPL_MUL_16_16(input[k],input[k]);
- }
-
- for ( k=0; k<4; k++) {
- nrgQlog[k] = (WebRtc_Word16)log2_Q8_LPC(nrgQ[k]); /* log2(nrgQ) */
- }
- oldNrgQlog = (WebRtc_Word16)log2_Q8_LPC(*oldEnergy);
-
- /* Calculate average level change */
- chng1 = WEBRTC_SPL_ABS_W16(nrgQlog[3]-nrgQlog[2]);
- chng2 = WEBRTC_SPL_ABS_W16(nrgQlog[2]-nrgQlog[1]);
- chng3 = WEBRTC_SPL_ABS_W16(nrgQlog[1]-nrgQlog[0]);
- chng4 = WEBRTC_SPL_ABS_W16(nrgQlog[0]-oldNrgQlog);
- tmp = chng1+chng2+chng3+chng4;
- chngQ = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp, kChngFactor, 10); /* Q12 */
- chngQ += 2926; /* + 1.0/1.4 in Q12 */
-
- /* Find average pitch gain */
- pgQ = 0;
- for (k=0; k<4; k++)
- {
- pgQ += pitchGains_Q12[k];
- }
-
- pg3 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(pgQ, pgQ,11); /* pgQ in Q(12+2)=Q14. Q14*Q14>>11 => Q17 */
- pg3 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(pgQ, pg3,13); /* Q17*Q14>>13 =>Q18 */
- pg3 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(pg3, kMulPitchGain ,5); /* Q10 kMulPitchGain = -25 = -200 in Q-3. */
-
- tmp16=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(kExp2,pg3,13);/* Q13*Q10>>13 => Q10*/
- if (tmp16<0) {
- tmp16_2 = (0x0400 | (tmp16 & 0x03FF));
- tmp16_1 = (WEBRTC_SPL_RSHIFT_W16((WebRtc_UWord16)(tmp16 ^ 0xFFFF), 10)-3); /* Gives result in Q14 */
- if (tmp16_1<0)
- expPg=(WebRtc_Word16) -WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
- else
- expPg=(WebRtc_Word16) -WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
- } else
- expPg = (WebRtc_Word16) -16384; /* 1 in Q14, since 2^0=1 */
-
- expPg32 = (WebRtc_Word32)WEBRTC_SPL_LSHIFT_W16((WebRtc_Word32)expPg, 8); /* Q22 */
- divVal = WebRtcSpl_DivW32W16ResW16(expPg32, chngQ); /* Q22/Q12=Q10 */
-
- tmp16=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(kExp2,divVal,13);/* Q13*Q10>>13 => Q10*/
- if (tmp16<0) {
- tmp16_2 = (0x0400 | (tmp16 & 0x03FF));
- tmp16_1 = (WEBRTC_SPL_RSHIFT_W16((WebRtc_UWord16)(tmp16 ^ 0xFFFF), 10)-3); /* Gives result in Q14 */
- if (tmp16_1<0)
- expPg=(WebRtc_Word16) WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
- else
- expPg=(WebRtc_Word16) WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
- } else
- expPg = (WebRtc_Word16) 16384; /* 1 in Q14, since 2^0=1 */
-
- *varscale = expPg-1;
- *oldEnergy = nrgQ[3];
-}
-
-
-
-static __inline WebRtc_Word16 exp2_Q10_T(WebRtc_Word16 x) { // Both in and out in Q10
-
- WebRtc_Word16 tmp16_1, tmp16_2;
-
- tmp16_2=(WebRtc_Word16)(0x0400|(x&0x03FF));
- tmp16_1=-(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(x,10);
- if(tmp16_1>0)
- return (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
- else
- return (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
-
-}
-
-
-// Declare a function pointer.
-AutocorrFix WebRtcIsacfix_AutocorrFix;
-
-void WebRtcIsacfix_GetLpcCoef(WebRtc_Word16 *inLoQ0,
- WebRtc_Word16 *inHiQ0,
- MaskFiltstr_enc *maskdata,
- WebRtc_Word16 snrQ10,
- const WebRtc_Word16 *pitchGains_Q12,
- WebRtc_Word32 *gain_lo_hiQ17,
- WebRtc_Word16 *lo_coeffQ15,
- WebRtc_Word16 *hi_coeffQ15)
-{
- int k, n, j, ii;
- WebRtc_Word16 pos1, pos2;
- WebRtc_Word16 sh_lo, sh_hi, sh, ssh, shMem;
- WebRtc_Word16 varscaleQ14;
-
- WebRtc_Word16 tmpQQlo, tmpQQhi;
- WebRtc_Word32 tmp32;
- WebRtc_Word16 tmp16,tmp16b;
-
- WebRtc_Word16 polyHI[ORDERHI+1];
- WebRtc_Word16 rcQ15_lo[ORDERLO], rcQ15_hi[ORDERHI];
-
-
- WebRtc_Word16 DataLoQ6[WINLEN], DataHiQ6[WINLEN];
- WebRtc_Word32 corrloQQ[ORDERLO+2];
- WebRtc_Word32 corrhiQQ[ORDERHI+1];
- WebRtc_Word32 corrlo2QQ[ORDERLO+1];
- WebRtc_Word16 scale;
- WebRtc_Word16 QdomLO, QdomHI, newQdomHI, newQdomLO;
-
- WebRtc_Word32 round;
- WebRtc_Word32 res_nrgQQ;
- WebRtc_Word32 sqrt_nrg;
-
- WebRtc_Word32 aSQR32;
-
- /* less-noise-at-low-frequencies factor */
- WebRtc_Word16 aaQ14;
-
- /* Multiplication with 1/sqrt(12) ~= 0.28901734104046 can be done by convertion to
- Q15, i.e. round(0.28901734104046*32768) = 9471, and use 9471/32768.0 ~= 0.289032
- */
- WebRtc_Word16 snrq, shft;
-
- WebRtc_Word16 tmp16a;
- WebRtc_Word32 tmp32a, tmp32b, tmp32c;
-
- WebRtc_Word16 a_LOQ11[ORDERLO+1];
- WebRtc_Word16 k_vecloQ15[ORDERLO];
- WebRtc_Word16 a_HIQ12[ORDERHI+1];
- WebRtc_Word16 k_vechiQ15[ORDERHI];
-
- WebRtc_Word16 stab;
-
- snrq=snrQ10;
-
- /* SNR= C * 2 ^ (D * snrq) ; C=0.289, D=0.05*log2(10)=0.166 (~=172 in Q10)*/
- tmp16 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(snrq, 172, 10); // Q10
- tmp16b = exp2_Q10_T(tmp16); // Q10
- snrq = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(tmp16b, 285, 10); // Q10
-
- /* change quallevel depending on pitch gains and level fluctuations */
- WebRtcIsacfix_GetVars(inLoQ0, pitchGains_Q12, &(maskdata->OldEnergy), &varscaleQ14);
-
- /* less-noise-at-low-frequencies factor */
- /* Calculation of 0.35 * (0.5 + 0.5 * varscale) in fixpoint:
- With 0.35 in Q16 (0.35 ~= 22938/65536.0 = 0.3500061) and varscaleQ14 in Q14,
- we get Q16*Q14>>16 = Q14
- */
- aaQ14 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
- (WEBRTC_SPL_MUL_16_16(22938, (8192 + WEBRTC_SPL_RSHIFT_W32(varscaleQ14, 1)))
- + ((WebRtc_Word32)32768)), 16);
-
- /* Calculate tmp = (1.0 + aa*aa); in Q12 */
- tmp16 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(aaQ14, aaQ14, 15); //Q14*Q14>>15 = Q13
- tmpQQlo = 4096 + WEBRTC_SPL_RSHIFT_W16(tmp16, 1); // Q12 + Q13>>1 = Q12
-
- /* Calculate tmp = (1.0+aa) * (1.0+aa); */
- tmp16 = 8192 + WEBRTC_SPL_RSHIFT_W16(aaQ14, 1); // 1+a in Q13
- tmpQQhi = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(tmp16, tmp16, 14); //Q13*Q13>>14 = Q12
-
- /* replace data in buffer by new look-ahead data */
- for (pos1 = 0; pos1 < QLOOKAHEAD; pos1++) {
- maskdata->DataBufferLoQ0[pos1 + WINLEN - QLOOKAHEAD] = inLoQ0[pos1];
- }
-
- for (k = 0; k < SUBFRAMES; k++) {
-
- /* Update input buffer and multiply signal with window */
- for (pos1 = 0; pos1 < WINLEN - UPDATE/2; pos1++) {
- maskdata->DataBufferLoQ0[pos1] = maskdata->DataBufferLoQ0[pos1 + UPDATE/2];
- maskdata->DataBufferHiQ0[pos1] = maskdata->DataBufferHiQ0[pos1 + UPDATE/2];
- DataLoQ6[pos1] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(
- maskdata->DataBufferLoQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
- DataHiQ6[pos1] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(
- maskdata->DataBufferHiQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
- }
- pos2 = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16(k, UPDATE)/2);
- for (n = 0; n < UPDATE/2; n++, pos1++) {
- maskdata->DataBufferLoQ0[pos1] = inLoQ0[QLOOKAHEAD + pos2];
- maskdata->DataBufferHiQ0[pos1] = inHiQ0[pos2++];
- DataLoQ6[pos1] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(
- maskdata->DataBufferLoQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
- DataHiQ6[pos1] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(
- maskdata->DataBufferHiQ0[pos1], kWindowAutocorr[pos1], 15); // Q0*Q21>>15 = Q6
- }
-
- /* Get correlation coefficients */
- /* The highest absolute value measured inside DataLo in the test set
- For DataHi, corresponding value was 160.
-
- This means that it should be possible to represent the input values
- to WebRtcSpl_AutoCorrelation() as Q6 values (since 307*2^6 =
- 19648). Of course, Q0 will also work, but due to the low energy in
- DataLo and DataHi, the outputted autocorrelation will be more accurate
- and mimic the floating point code better, by being in an high as possible
- Q-domain.
- */
-
- WebRtcIsacfix_AutocorrFix(corrloQQ,DataLoQ6,WINLEN, ORDERLO+1, &scale);
- QdomLO = 12-scale; // QdomLO is the Q-domain of corrloQQ
- sh_lo = WebRtcSpl_NormW32(corrloQQ[0]);
- QdomLO += sh_lo;
- for (ii=0; ii<ORDERLO+2; ii++) {
- corrloQQ[ii] = WEBRTC_SPL_LSHIFT_W32(corrloQQ[ii], sh_lo);
- }
- /* It is investigated whether it was possible to use 16 bits for the
- 32-bit vector corrloQQ, but it didn't work. */
-
- WebRtcIsacfix_AutocorrFix(corrhiQQ,DataHiQ6,WINLEN, ORDERHI, &scale);
-
- QdomHI = 12-scale; // QdomHI is the Q-domain of corrhiQQ
- sh_hi = WebRtcSpl_NormW32(corrhiQQ[0]);
- QdomHI += sh_hi;
- for (ii=0; ii<ORDERHI+1; ii++) {
- corrhiQQ[ii] = WEBRTC_SPL_LSHIFT_W32(corrhiQQ[ii], sh_hi);
- }
-
- /* less noise for lower frequencies, by filtering/scaling autocorrelation sequences */
-
- /* Calculate corrlo2[0] = tmpQQlo * corrlo[0] - 2.0*tmpQQlo * corrlo[1];*/
- corrlo2QQ[0] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(tmpQQlo, corrloQQ[0]), 1)- // Q(12+QdomLO-16)>>1 = Q(QdomLO-5)
- WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(aaQ14, corrloQQ[1]), 2); // 2*Q(14+QdomLO-16)>>3 = Q(QdomLO-2)>>2 = Q(QdomLO-5)
-
- /* Calculate corrlo2[n] = tmpQQlo * corrlo[n] - tmpQQlo * (corrlo[n-1] + corrlo[n+1]);*/
- for (n = 1; n <= ORDERLO; n++) {
-
- tmp32 = WEBRTC_SPL_RSHIFT_W32(corrloQQ[n-1], 1) + WEBRTC_SPL_RSHIFT_W32(corrloQQ[n+1], 1); // Q(QdomLO-1)
- corrlo2QQ[n] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(tmpQQlo, corrloQQ[n]), 1)- // Q(12+QdomLO-16)>>1 = Q(QdomLO-5)
- WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(aaQ14, tmp32), 2); // Q(14+QdomLO-1-16)>>2 = Q(QdomLO-3)>>2 = Q(QdomLO-5)
-
- }
- QdomLO -= 5;
-
- /* Calculate corrhi[n] = tmpQQhi * corrhi[n]; */
- for (n = 0; n <= ORDERHI; n++) {
- corrhiQQ[n] = WEBRTC_SPL_MUL_16_32_RSFT16(tmpQQhi, corrhiQQ[n]); // Q(12+QdomHI-16) = Q(QdomHI-4)
- }
- QdomHI -= 4;
-
- /* add white noise floor */
- /* corrlo2QQ is in Q(QdomLO) and corrhiQQ is in Q(QdomHI) */
- /* Calculate corrlo2[0] += 9.5367431640625e-7; and
- corrhi[0] += 9.5367431640625e-7, where the constant is 1/2^20 */
-
- tmp32 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32) 1, QdomLO-20);
- corrlo2QQ[0] += tmp32;
- tmp32 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32) 1, QdomHI-20);
- corrhiQQ[0] += tmp32;
-
- /* corrlo2QQ is in Q(QdomLO) and corrhiQQ is in Q(QdomHI) before the following
- code segment, where we want to make sure we get a 1-bit margin */
- for (n = 0; n <= ORDERLO; n++) {
- corrlo2QQ[n] = WEBRTC_SPL_RSHIFT_W32(corrlo2QQ[n], 1); // Make sure we have a 1-bit margin
- }
- QdomLO -= 1; // Now, corrlo2QQ is in Q(QdomLO), with a 1-bit margin
-
- for (n = 0; n <= ORDERHI; n++) {
- corrhiQQ[n] = WEBRTC_SPL_RSHIFT_W32(corrhiQQ[n], 1); // Make sure we have a 1-bit margin
- }
- QdomHI -= 1; // Now, corrhiQQ is in Q(QdomHI), with a 1-bit margin
-
-
- newQdomLO = QdomLO;
-
- for (n = 0; n <= ORDERLO; n++) {
- WebRtc_Word32 tmp, tmpB, tmpCorr;
- WebRtc_Word16 alpha=328; //0.01 in Q15
- WebRtc_Word16 beta=324; //(1-0.01)*0.01=0.0099 in Q15
- WebRtc_Word16 gamma=32440; //(1-0.01)=0.99 in Q15
-
- if (maskdata->CorrBufLoQQ[n] != 0) {
- shMem=WebRtcSpl_NormW32(maskdata->CorrBufLoQQ[n]);
- sh = QdomLO - maskdata->CorrBufLoQdom[n];
- if (sh<=shMem) {
- tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufLoQQ[n], sh); // Get CorrBufLoQQ to same domain as corrlo2
- tmp = WEBRTC_SPL_MUL_16_32_RSFT15(alpha, tmp);
- } else if ((sh-shMem)<7){
- tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufLoQQ[n], shMem); // Shift up CorrBufLoQQ as much as possible
- tmp = WEBRTC_SPL_MUL_16_32_RSFT15(WEBRTC_SPL_LSHIFT_W16(alpha, (sh-shMem)), tmp); // Shift alpha the number of times required to get tmp in QdomLO
- } else {
- tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufLoQQ[n], shMem); // Shift up CorrBufHiQQ as much as possible
- tmp = WEBRTC_SPL_MUL_16_32_RSFT15(WEBRTC_SPL_LSHIFT_W16(alpha, 6), tmp); // Shift alpha as much as possible without overflow the number of times required to get tmp in QdomHI
- tmpCorr = WEBRTC_SPL_RSHIFT_W32(corrloQQ[n], sh-shMem-6);
- tmp = tmp + tmpCorr;
- maskdata->CorrBufLoQQ[n] = tmp;
- newQdomLO = QdomLO-(sh-shMem-6);
- maskdata->CorrBufLoQdom[n] = newQdomLO;
- }
- } else
- tmp = 0;
-
- tmp = tmp + corrlo2QQ[n];
-
- maskdata->CorrBufLoQQ[n] = tmp;
- maskdata->CorrBufLoQdom[n] = QdomLO;
-
- tmp=WEBRTC_SPL_MUL_16_32_RSFT15(beta, tmp);
- tmpB=WEBRTC_SPL_MUL_16_32_RSFT15(gamma, corrlo2QQ[n]);
- corrlo2QQ[n] = tmp + tmpB;
- }
- if( newQdomLO!=QdomLO) {
- for (n = 0; n <= ORDERLO; n++) {
- if (maskdata->CorrBufLoQdom[n] != newQdomLO)
- corrloQQ[n] = WEBRTC_SPL_RSHIFT_W32(corrloQQ[n], maskdata->CorrBufLoQdom[n]-newQdomLO);
- }
- QdomLO = newQdomLO;
- }
-
-
- newQdomHI = QdomHI;
-
- for (n = 0; n <= ORDERHI; n++) {
- WebRtc_Word32 tmp, tmpB, tmpCorr;
- WebRtc_Word16 alpha=328; //0.01 in Q15
- WebRtc_Word16 beta=324; //(1-0.01)*0.01=0.0099 in Q15
- WebRtc_Word16 gamma=32440; //(1-0.01)=0.99 in Q1
- if (maskdata->CorrBufHiQQ[n] != 0) {
- shMem=WebRtcSpl_NormW32(maskdata->CorrBufHiQQ[n]);
- sh = QdomHI - maskdata->CorrBufHiQdom[n];
- if (sh<=shMem) {
- tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufHiQQ[n], sh); // Get CorrBufHiQQ to same domain as corrhi
- tmp = WEBRTC_SPL_MUL_16_32_RSFT15(alpha, tmp);
- tmpCorr = corrhiQQ[n];
- tmp = tmp + tmpCorr;
- maskdata->CorrBufHiQQ[n] = tmp;
- maskdata->CorrBufHiQdom[n] = QdomHI;
- } else if ((sh-shMem)<7) {
- tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufHiQQ[n], shMem); // Shift up CorrBufHiQQ as much as possible
- tmp = WEBRTC_SPL_MUL_16_32_RSFT15(WEBRTC_SPL_LSHIFT_W16(alpha, (sh-shMem)), tmp); // Shift alpha the number of times required to get tmp in QdomHI
- tmpCorr = corrhiQQ[n];
- tmp = tmp + tmpCorr;
- maskdata->CorrBufHiQQ[n] = tmp;
- maskdata->CorrBufHiQdom[n] = QdomHI;
- } else {
- tmp = WEBRTC_SPL_SHIFT_W32(maskdata->CorrBufHiQQ[n], shMem); // Shift up CorrBufHiQQ as much as possible
- tmp = WEBRTC_SPL_MUL_16_32_RSFT15(WEBRTC_SPL_LSHIFT_W16(alpha, 6), tmp); // Shift alpha as much as possible without overflow the number of times required to get tmp in QdomHI
- tmpCorr = WEBRTC_SPL_RSHIFT_W32(corrhiQQ[n], sh-shMem-6);
- tmp = tmp + tmpCorr;
- maskdata->CorrBufHiQQ[n] = tmp;
- newQdomHI = QdomHI-(sh-shMem-6);
- maskdata->CorrBufHiQdom[n] = newQdomHI;
- }
- } else {
- tmp = corrhiQQ[n];
- tmpCorr = tmp;
- maskdata->CorrBufHiQQ[n] = tmp;
- maskdata->CorrBufHiQdom[n] = QdomHI;
- }
-
- tmp=WEBRTC_SPL_MUL_16_32_RSFT15(beta, tmp);
- tmpB=WEBRTC_SPL_MUL_16_32_RSFT15(gamma, tmpCorr);
- corrhiQQ[n] = tmp + tmpB;
- }
-
- if( newQdomHI!=QdomHI) {
- for (n = 0; n <= ORDERHI; n++) {
- if (maskdata->CorrBufHiQdom[n] != newQdomHI)
- corrhiQQ[n] = WEBRTC_SPL_RSHIFT_W32(corrhiQQ[n], maskdata->CorrBufHiQdom[n]-newQdomHI);
- }
- QdomHI = newQdomHI;
- }
-
- stab=WebRtcSpl_LevinsonW32_JSK(corrlo2QQ, a_LOQ11, k_vecloQ15, ORDERLO);
-
- if (stab<0) { // If unstable use lower order
- a_LOQ11[0]=2048;
- for (n = 1; n <= ORDERLO; n++) {
- a_LOQ11[n]=0;
- }
-
- stab=WebRtcSpl_LevinsonW32_JSK(corrlo2QQ, a_LOQ11, k_vecloQ15, 8);
- }
-
-
- WebRtcSpl_LevinsonDurbin(corrhiQQ, a_HIQ12, k_vechiQ15, ORDERHI);
-
- /* bandwidth expansion */
- for (n = 1; n <= ORDERLO; n++) {
- a_LOQ11[n] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(kPolyVecLo[n-1], a_LOQ11[n]);
- }
-
-
- polyHI[0] = a_HIQ12[0];
- for (n = 1; n <= ORDERHI; n++) {
- a_HIQ12[n] = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT_WITH_FIXROUND(kPolyVecHi[n-1], a_HIQ12[n]);
- polyHI[n] = a_HIQ12[n];
- }
-
- /* Normalize the corrlo2 vector */
- sh = WebRtcSpl_NormW32(corrlo2QQ[0]);
- for (n = 0; n <= ORDERLO; n++) {
- corrlo2QQ[n] = WEBRTC_SPL_LSHIFT_W32(corrlo2QQ[n], sh);
- }
- QdomLO += sh; /* Now, corrlo2QQ is still in Q(QdomLO) */
-
-
- /* residual energy */
-
- sh_lo = 31;
- res_nrgQQ = 0;
- for (j = 0; j <= ORDERLO; j++)
- {
- for (n = 0; n < j; n++)
- {
- WebRtc_Word16 index, diff, sh_corr;
-
- index = j - n; //WEBRTC_SPL_ABS_W16(j-n);
-
- /* Calculation of res_nrg += a_LO[j] * corrlo2[j-n] * a_LO[n]; */
- /* corrlo2QQ is in Q(QdomLO) */
- tmp32 = ((WebRtc_Word32) WEBRTC_SPL_MUL_16_16(a_LOQ11[j], a_LOQ11[n])); // Q11*Q11 = Q22
- // multiply by 2 as loop only on half of the matrix. a_LOQ11 gone through bandwidth
- // expation so the following shift is safe.
- tmp32 = WEBRTC_SPL_LSHIFT_W32(tmp32, 1);
- sh = WebRtcSpl_NormW32(tmp32);
- aSQR32 = WEBRTC_SPL_LSHIFT_W32(tmp32, sh); // Q(22+sh)
- sh_corr = WebRtcSpl_NormW32(corrlo2QQ[index]);
- tmp32 = WEBRTC_SPL_LSHIFT_W32(corrlo2QQ[index], sh_corr);
- tmp32 = (WebRtc_Word32) WEBRTC_SPL_MUL_32_32_RSFT32BI(aSQR32, tmp32); // Q(22+sh)*Q(QdomLO+sh_corr)>>32 = Q(22+sh+QdomLO+sh_corr-32) = Q(sh+QdomLO+sh_corr-10)
- sh = sh+QdomLO+sh_corr-10;
-
- diff = sh_lo-sh;
-
- round = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, (WEBRTC_SPL_ABS_W32(diff)-1));
- if (diff==0)
- round = 0;
- if (diff>=31) {
- res_nrgQQ = tmp32;
- sh_lo = sh;
- } else if (diff>0) {
- res_nrgQQ = WEBRTC_SPL_RSHIFT_W32((res_nrgQQ+round), (diff+1)) + WEBRTC_SPL_RSHIFT_W32(tmp32, 1);
- sh_lo = sh-1;
- } else if (diff>-31){
- res_nrgQQ = WEBRTC_SPL_RSHIFT_W32(res_nrgQQ, 1) + WEBRTC_SPL_SHIFT_W32((tmp32+round), -(-diff+1));
- sh_lo = sh_lo-1;
- }
- sh = WebRtcSpl_NormW32(res_nrgQQ);
- res_nrgQQ = WEBRTC_SPL_LSHIFT_W32(res_nrgQQ, sh);
- sh_lo += sh;
- }
- n = j;
- {
- WebRtc_Word16 index, diff, sh_corr;
-
- index = 0; //WEBRTC_SPL_ABS_W16(j-n);
-
- /* Calculation of res_nrg += a_LO[j] * corrlo2[j-n] * a_LO[n]; */
- /* corrlo2QQ is in Q(QdomLO) */
- tmp32 = (WebRtc_Word32) WEBRTC_SPL_MUL_16_16(a_LOQ11[j], a_LOQ11[n]); // Q11*Q11 = Q22
- sh = WebRtcSpl_NormW32(tmp32);
- aSQR32 = WEBRTC_SPL_LSHIFT_W32(tmp32, sh); // Q(22+sh)
- sh_corr = WebRtcSpl_NormW32(corrlo2QQ[index]);
- tmp32 = WEBRTC_SPL_LSHIFT_W32(corrlo2QQ[index], sh_corr);
- tmp32 = (WebRtc_Word32) WEBRTC_SPL_MUL_32_32_RSFT32BI(aSQR32, tmp32); // Q(22+sh)*Q(QdomLO+sh_corr)>>32 = Q(22+sh+QdomLO+sh_corr-32) = Q(sh+QdomLO+sh_corr-10)
- sh = sh+QdomLO+sh_corr-10;
- diff = sh_lo-sh;
-
- round = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, (WEBRTC_SPL_ABS_W32(diff)-1));
- if (diff==0)
- round = 0;
- if (diff>=31) {
- res_nrgQQ = tmp32;
- sh_lo = sh;
- } else if (diff>0) {
- res_nrgQQ = WEBRTC_SPL_RSHIFT_W32((res_nrgQQ+round), (diff+1)) + WEBRTC_SPL_RSHIFT_W32(tmp32, 1);
- sh_lo = sh-1;
- } else if (diff>-31){
- res_nrgQQ = WEBRTC_SPL_RSHIFT_W32(res_nrgQQ, 1) + WEBRTC_SPL_SHIFT_W32((tmp32+round), -(-diff+1));
- sh_lo = sh_lo-1;
- }
- sh = WebRtcSpl_NormW32(res_nrgQQ);
- res_nrgQQ = WEBRTC_SPL_LSHIFT_W32(res_nrgQQ, sh);
- sh_lo += sh;
- }
- }
- /* Convert to reflection coefficients */
-
-
- WebRtcSpl_AToK_JSK(a_LOQ11, ORDERLO, rcQ15_lo);
-
- if (sh_lo & 0x0001) {
- res_nrgQQ=WEBRTC_SPL_RSHIFT_W32(res_nrgQQ, 1);
- sh_lo-=1;
- }
-
-
- if( res_nrgQQ > 0 )
- {
- sqrt_nrg=WebRtcSpl_Sqrt(res_nrgQQ);
-
- /* add hearing threshold and compute the gain */
- /* lo_coeff = varscale * S_N_R / (sqrt_nrg + varscale * H_T_H); */
-
-
- //tmp32a=WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, H_T_HQ19, 17); // Q14
- tmp32a=WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32) varscaleQ14,1); // H_T_HQ19=65536 (16-17=-1) ssh= WEBRTC_SPL_RSHIFT_W16(sh_lo, 1); // sqrt_nrg is in Qssh
- ssh= WEBRTC_SPL_RSHIFT_W16(sh_lo, 1); // sqrt_nrg is in Qssh
- sh = ssh - 14;
- tmp32b = WEBRTC_SPL_SHIFT_W32(tmp32a, sh); // Q14->Qssh
- tmp32c = sqrt_nrg + tmp32b; // Qssh (denominator)
- tmp32a = WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, snrq, 0); //Q24 (numerator)
-
- sh = WebRtcSpl_NormW32(tmp32c);
- shft = 16 - sh;
- tmp16a = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(tmp32c, -shft); // Q(ssh-shft) (denominator)
-
- tmp32b = WebRtcSpl_DivW32W16(tmp32a, tmp16a); // Q(24-ssh+shft)
- sh = ssh-shft-7;
- *gain_lo_hiQ17 = WEBRTC_SPL_SHIFT_W32(tmp32b, sh); // Gains in Q17
- }
- else
- {
- *gain_lo_hiQ17 = 100; //(WebRtc_Word32)WEBRTC_SPL_LSHIFT_W32( (WebRtc_Word32)1, 17); // Gains in Q17
- }
- gain_lo_hiQ17++;
-
- /* copy coefficients to output array */
- for (n = 0; n < ORDERLO; n++) {
- *lo_coeffQ15 = (WebRtc_Word16) (rcQ15_lo[n]);
- lo_coeffQ15++;
- }
- /* residual energy */
- res_nrgQQ = 0;
- sh_hi = 31;
-
-
- for (j = 0; j <= ORDERHI; j++)
- {
- for (n = 0; n < j; n++)
- {
- WebRtc_Word16 index, diff, sh_corr;
-
- index = j-n; //WEBRTC_SPL_ABS_W16(j-n);
-
- /* Calculation of res_nrg += a_HI[j] * corrhi[j-n] * a_HI[n] * 2; for j != n */
- /* corrhiQQ is in Q(QdomHI) */
- tmp32 = ((WebRtc_Word32) WEBRTC_SPL_MUL_16_16(a_HIQ12[j], a_HIQ12[n])); // Q12*Q12 = Q24
- tmp32 = WEBRTC_SPL_LSHIFT_W32(tmp32, 1);
- sh = WebRtcSpl_NormW32(tmp32);
- aSQR32 = WEBRTC_SPL_LSHIFT_W32(tmp32, sh); // Q(24+sh)
- sh_corr = WebRtcSpl_NormW32(corrhiQQ[index]);
- tmp32 = WEBRTC_SPL_LSHIFT_W32(corrhiQQ[index],sh_corr);
- tmp32 = (WebRtc_Word32) WEBRTC_SPL_MUL_32_32_RSFT32BI(aSQR32, tmp32); // Q(24+sh)*Q(QdomHI+sh_corr)>>32 = Q(24+sh+QdomHI+sh_corr-32) = Q(sh+QdomHI+sh_corr-8)
- sh = sh+QdomHI+sh_corr-8;
- diff = sh_hi-sh;
-
- round = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, (WEBRTC_SPL_ABS_W32(diff)-1));
- if (diff==0)
- round = 0;
- if (diff>=31) {
- res_nrgQQ = tmp32;
- sh_hi = sh;
- } else if (diff>0) {
- res_nrgQQ = WEBRTC_SPL_RSHIFT_W32((res_nrgQQ+round), (diff+1)) + WEBRTC_SPL_RSHIFT_W32(tmp32, 1);
- sh_hi = sh-1;
- } else if (diff>-31){
- res_nrgQQ = WEBRTC_SPL_RSHIFT_W32(res_nrgQQ, 1) + WEBRTC_SPL_SHIFT_W32((tmp32+round), -(-diff+1));
- sh_hi = sh_hi-1;
- }
-
- sh = WebRtcSpl_NormW32(res_nrgQQ);
- res_nrgQQ = WEBRTC_SPL_LSHIFT_W32(res_nrgQQ, sh);
- sh_hi += sh;
- }
-
- n = j;
- {
- WebRtc_Word16 index, diff, sh_corr;
-
- index = 0; //n-j; //WEBRTC_SPL_ABS_W16(j-n);
-
- /* Calculation of res_nrg += a_HI[j] * corrhi[j-n] * a_HI[n];*/
- /* corrhiQQ is in Q(QdomHI) */
- tmp32 = ((WebRtc_Word32) WEBRTC_SPL_MUL_16_16(a_HIQ12[j], a_HIQ12[n])); // Q12*Q12 = Q24
- sh = WebRtcSpl_NormW32(tmp32);
- aSQR32 = WEBRTC_SPL_LSHIFT_W32(tmp32, sh); // Q(24+sh)
- sh_corr = WebRtcSpl_NormW32(corrhiQQ[index]);
- tmp32 = WEBRTC_SPL_LSHIFT_W32(corrhiQQ[index],sh_corr);
- tmp32 = (WebRtc_Word32) WEBRTC_SPL_MUL_32_32_RSFT32BI(aSQR32, tmp32); // Q(24+sh)*Q(QdomHI+sh_corr)>>32 = Q(24+sh+QdomHI+sh_corr-32) = Q(sh+QdomHI+sh_corr-8)
- sh = sh+QdomHI+sh_corr-8;
- diff = sh_hi-sh;
-
- round = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, (WEBRTC_SPL_ABS_W32(diff)-1));
- if (diff==0)
- round = 0;
- if (diff>=31) {
- res_nrgQQ = tmp32;
- sh_hi = sh;
- } else if (diff>0) {
- res_nrgQQ = WEBRTC_SPL_RSHIFT_W32((res_nrgQQ+round), (diff+1)) + WEBRTC_SPL_RSHIFT_W32(tmp32, 1);
- sh_hi = sh-1;
- } else if (diff>-31){
- res_nrgQQ = WEBRTC_SPL_RSHIFT_W32(res_nrgQQ, 1) + WEBRTC_SPL_SHIFT_W32((tmp32+round), -(-diff+1));
- sh_hi = sh_hi-1;
- }
-
- sh = WebRtcSpl_NormW32(res_nrgQQ);
- res_nrgQQ = WEBRTC_SPL_LSHIFT_W32(res_nrgQQ, sh);
- sh_hi += sh;
- }
- }
-
- /* Convert to reflection coefficients */
- WebRtcSpl_LpcToReflCoef(polyHI, ORDERHI, rcQ15_hi);
-
- if (sh_hi & 0x0001) {
- res_nrgQQ=WEBRTC_SPL_RSHIFT_W32(res_nrgQQ, 1);
- sh_hi-=1;
- }
-
-
- if( res_nrgQQ > 0 )
- {
- sqrt_nrg=WebRtcSpl_Sqrt(res_nrgQQ);
-
-
- /* add hearing threshold and compute the gain */
- /* hi_coeff = varscale * S_N_R / (sqrt_nrg + varscale * H_T_H); */
-
- //tmp32a=WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, H_T_HQ19, 17); // Q14
- tmp32a=WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32) varscaleQ14,1); // H_T_HQ19=65536 (16-17=-1)
-
- ssh= WEBRTC_SPL_RSHIFT_W32(sh_hi, 1); // sqrt_nrg is in Qssh
- sh = ssh - 14;
- tmp32b = WEBRTC_SPL_SHIFT_W32(tmp32a, sh); // Q14->Qssh
- tmp32c = sqrt_nrg + tmp32b; // Qssh (denominator)
- tmp32a = WEBRTC_SPL_MUL_16_16_RSFT(varscaleQ14, snrq, 0); //Q24 (numerator)
-
- sh = WebRtcSpl_NormW32(tmp32c);
- shft = 16 - sh;
- tmp16a = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(tmp32c, -shft); // Q(ssh-shft) (denominator)
-
- tmp32b = WebRtcSpl_DivW32W16(tmp32a, tmp16a); // Q(24-ssh+shft)
- sh = ssh-shft-7;
- *gain_lo_hiQ17 = WEBRTC_SPL_SHIFT_W32(tmp32b, sh); // Gains in Q17
- }
- else
- {
- *gain_lo_hiQ17 = 100; //(WebRtc_Word32)WEBRTC_SPL_LSHIFT_W32( (WebRtc_Word32)1, 17); // Gains in Q17
- }
- gain_lo_hiQ17++;
-
-
- /* copy coefficients to output array */
- for (n = 0; n < ORDERHI; n++) {
- *hi_coeffQ15 = rcQ15_hi[n];
- hi_coeffQ15++;
- }
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_masking_model.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_masking_model.h
deleted file mode 100644
index 9a64844..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_masking_model.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * lpc_masking_model.h
- *
- * LPC functions
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_MASKING_MODEL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_MASKING_MODEL_H_
-
-#include "structs.h"
-
-void WebRtcIsacfix_GetVars(const WebRtc_Word16 *input,
- const WebRtc_Word16 *pitchGains_Q12,
- WebRtc_UWord32 *oldEnergy,
- WebRtc_Word16 *varscale);
-
-void WebRtcIsacfix_GetLpcCoef(WebRtc_Word16 *inLoQ0,
- WebRtc_Word16 *inHiQ0,
- MaskFiltstr_enc *maskdata,
- WebRtc_Word16 snrQ10,
- const WebRtc_Word16 *pitchGains_Q12,
- WebRtc_Word32 *gain_lo_hiQ17,
- WebRtc_Word16 *lo_coeffQ15,
- WebRtc_Word16 *hi_coeffQ15);
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_MASKING_MODEL_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_tables.c
deleted file mode 100644
index 90cc9af..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_tables.c
+++ /dev/null
@@ -1,1280 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * lpc_tables.c
- *
- * Coding tables for the KLT coefficients
- *
- */
-
-
-#include "settings.h"
-#include "lpc_tables.h"
-
-/* indices of KLT coefficients used */
-const WebRtc_UWord16 WebRtcIsacfix_kSelIndGain[12] = {
- 0, 1, 2, 3, 4, 5, 6, 7, 8, 9,
- 10, 11};
-
-const WebRtc_UWord16 WebRtcIsacfix_kSelIndShape[108] = {
- 0, 1, 2, 3, 4, 5, 6, 7, 8, 9,
- 10, 11, 12, 13, 14, 15, 16, 17, 18, 19,
- 20, 21, 22, 23, 24, 25, 26, 27, 28, 29,
- 30, 31, 32, 33, 34, 35, 36, 37, 38, 39,
- 40, 41, 42, 43, 44, 45, 46, 47, 48, 49,
- 50, 51, 52, 53, 54, 55, 56, 57, 58, 59,
- 60, 61, 62, 63, 64, 65, 66, 67, 68, 69,
- 70, 71, 72, 73, 74, 75, 76, 77, 78, 79,
- 80, 81, 82, 83, 84, 85, 86, 87, 88, 89,
- 90, 91, 92, 93, 94, 95, 96, 97, 98, 99,
- 100, 101, 102, 103, 104, 105, 106, 107
-};
-
-/* cdf array for model indicator */
-const WebRtc_UWord16 WebRtcIsacfix_kModelCdf[4] = {
- 0, 15434, 37548, 65535
-};
-
-/* pointer to cdf array for model indicator */
-const WebRtc_UWord16 *WebRtcIsacfix_kModelCdfPtr[1] = {
- WebRtcIsacfix_kModelCdf
-};
-
-/* initial cdf index for decoder of model indicator */
-const WebRtc_UWord16 WebRtcIsacfix_kModelInitIndex[1] = {
- 1
-};
-
-/* offset to go from rounded value to quantization index */
-const WebRtc_Word16 WebRtcIsacfix_kQuantMinGain[12] ={
- 3, 6, 4, 6, 6, 9, 5, 16, 11, 34, 32, 47
-};
-
-const WebRtc_Word16 WebRtcIsacfix_kQuantMinShape[108] = {
- 0, 0, 0, 0, 0, 0, 0, 0, 0, 1,
- 1, 1, 1, 1, 2, 2, 2, 3, 0, 0,
- 0, 0, 1, 0, 0, 0, 0, 1, 1, 1,
- 1, 1, 1, 2, 2, 3, 0, 0, 0, 0,
- 1, 0, 1, 1, 1, 1, 1, 1, 1, 2,
- 2, 4, 3, 5, 0, 0, 0, 0, 1, 1,
- 1, 1, 1, 1, 2, 1, 2, 2, 3, 4,
- 4, 7, 0, 0, 1, 1, 1, 1, 1, 1,
- 1, 2, 3, 2, 3, 4, 4, 5, 7, 13,
- 0, 1, 1, 2, 3, 2, 2, 2, 4, 4,
- 5, 6, 7, 11, 9, 13, 12, 26
-};
-
-/* maximum quantization index */
-const WebRtc_UWord16 WebRtcIsacfix_kMaxIndGain[12] = {
- 6, 12, 8, 14, 10, 19, 12, 31, 22, 56, 52, 138
-};
-
-const WebRtc_UWord16 WebRtcIsacfix_kMaxIndShape[108] = {
- 0, 0, 0, 0, 0, 0, 0, 0, 0, 1,
- 2, 2, 2, 2, 4, 4, 5, 6, 0, 0,
- 0, 0, 1, 0, 0, 0, 0, 1, 2, 2,
- 2, 2, 3, 4, 5, 7, 0, 0, 0, 0,
- 2, 0, 2, 2, 2, 2, 3, 2, 2, 4,
- 4, 6, 6, 9, 0, 0, 0, 0, 2, 2,
- 2, 2, 2, 2, 3, 2, 4, 4, 7, 7,
- 9, 13, 0, 0, 2, 2, 2, 2, 2, 2,
- 3, 4, 5, 4, 6, 8, 8, 10, 16, 25,
- 0, 2, 2, 4, 5, 4, 4, 4, 7, 8,
- 9, 10, 13, 19, 17, 23, 25, 49
-};
-
-/* index offset */
-const WebRtc_UWord16 WebRtcIsacfix_kOffsetGain[3][12] = {
- { 0, 7, 20, 29, 44, 55, 75, 88, 120, 143, 200, 253},
- { 0, 7, 19, 27, 42, 53, 73, 86, 117, 140, 197, 249},
- { 0, 7, 20, 28, 44, 55, 75, 89, 121, 145, 202, 257}
-};
-
-const WebRtc_UWord16 WebRtcIsacfix_kOffsetShape[3][108] = {
- {
- 0, 1, 2, 3, 4, 5, 6, 7, 8, 9,
- 11, 14, 17, 20, 23, 28, 33, 39, 46, 47,
- 48, 49, 50, 52, 53, 54, 55, 56, 58, 61,
- 64, 67, 70, 74, 79, 85, 93, 94, 95, 96,
- 97, 100, 101, 104, 107, 110, 113, 117, 120, 123,
- 128, 133, 140, 147, 157, 158, 159, 160, 161, 164,
- 167, 170, 173, 176, 179, 183, 186, 191, 196, 204,
- 212, 222, 236, 237, 238, 241, 244, 247, 250, 253,
- 256, 260, 265, 271, 276, 283, 292, 301, 312, 329,
- 355, 356, 359, 362, 367, 373, 378, 383, 388, 396,
- 405, 415, 426, 440, 460, 478, 502, 528
- },
- {
- 0, 1, 2, 3, 4, 6, 7, 8, 9, 11,
- 13, 16, 19, 22, 26, 29, 34, 39, 45, 46,
- 47, 48, 49, 50, 51, 52, 53, 55, 57, 60,
- 63, 66, 70, 73, 78, 84, 91, 92, 93, 94,
- 95, 96, 97, 99, 102, 105, 108, 111, 114, 118,
- 123, 128, 134, 141, 151, 152, 153, 154, 156, 159,
- 162, 165, 168, 171, 174, 177, 181, 186, 194, 200,
- 208, 218, 233, 234, 235, 236, 239, 242, 245, 248,
- 251, 254, 258, 263, 270, 277, 288, 297, 308, 324,
- 349, 351, 354, 357, 361, 366, 372, 378, 383, 390,
- 398, 407, 420, 431, 450, 472, 496, 524
- },
- {
- 0, 1, 2, 3, 4, 5, 6, 7, 8, 11,
- 14, 17, 20, 23, 26, 29, 34, 40, 47, 48,
- 49, 50, 51, 52, 53, 54, 55, 58, 61, 64,
- 67, 70, 73, 77, 82, 88, 96, 97, 98, 99,
- 101, 102, 104, 107, 110, 113, 116, 119, 122, 125,
- 129, 134, 141, 150, 160, 161, 162, 163, 166, 168,
- 171, 174, 177, 180, 183, 186, 190, 195, 201, 208,
- 216, 226, 243, 244, 245, 248, 251, 254, 257, 260,
- 263, 268, 273, 278, 284, 291, 299, 310, 323, 340,
- 366, 368, 371, 374, 379, 383, 389, 394, 399, 406,
- 414, 422, 433, 445, 461, 480, 505, 533
- }
-};
-
-/* initial cdf index for KLT coefficients */
-const WebRtc_UWord16 WebRtcIsacfix_kInitIndexGain[3][12] = {
- { 3, 6, 4, 7, 5, 10, 6, 16, 11, 28, 26, 69},
- { 3, 6, 4, 7, 5, 10, 6, 15, 11, 28, 26, 69},
- { 3, 6, 4, 8, 5, 10, 7, 16, 12, 28, 27, 70}
-};
-
-const WebRtc_UWord16 WebRtcIsacfix_kInitIndexShape[3][108] = {
- {
- 0, 0, 0, 0, 0, 0, 0, 0, 0, 1,
- 1, 1, 1, 1, 2, 2, 3, 3, 0, 0,
- 0, 0, 1, 0, 0, 0, 0, 1, 1, 1,
- 1, 1, 2, 2, 3, 4, 0, 0, 0, 0,
- 1, 0, 1, 1, 1, 1, 2, 1, 1, 2,
- 2, 3, 3, 5, 0, 0, 0, 0, 1, 1,
- 1, 1, 1, 1, 2, 1, 2, 2, 4, 4,
- 5, 7, 0, 0, 1, 1, 1, 1, 1, 1,
- 2, 2, 3, 2, 3, 4, 4, 5, 8, 13,
- 0, 1, 1, 2, 3, 2, 2, 2, 4, 4,
- 5, 5, 7, 10, 9, 12, 13, 25
- },
- {
- 0, 0, 0, 0, 1, 0, 0, 0, 1, 1,
- 1, 1, 1, 2, 1, 2, 2, 3, 0, 0,
- 0, 0, 0, 0, 0, 0, 1, 1, 1, 1,
- 1, 2, 1, 2, 3, 3, 0, 0, 0, 0,
- 0, 0, 1, 1, 1, 1, 1, 1, 2, 2,
- 2, 3, 3, 5, 0, 0, 0, 1, 1, 1,
- 1, 1, 1, 1, 1, 2, 2, 4, 3, 4,
- 5, 7, 0, 0, 0, 1, 1, 1, 1, 1,
- 1, 2, 2, 3, 3, 5, 4, 5, 8, 12,
- 1, 1, 1, 2, 2, 3, 3, 2, 3, 4,
- 4, 6, 5, 9, 11, 12, 14, 25
- },
- {
- 0, 0, 0, 0, 0, 0, 0, 0, 1, 1,
- 1, 1, 1, 1, 1, 2, 3, 3, 0, 0,
- 0, 0, 0, 0, 0, 0, 1, 1, 1, 1,
- 1, 1, 2, 2, 3, 4, 0, 0, 0, 1,
- 0, 1, 1, 1, 1, 1, 1, 1, 1, 2,
- 2, 3, 4, 5, 0, 0, 0, 1, 1, 1,
- 1, 1, 1, 1, 1, 2, 2, 3, 3, 4,
- 5, 8, 0, 0, 1, 1, 1, 1, 1, 1,
- 2, 2, 2, 3, 3, 4, 5, 6, 8, 13,
- 1, 1, 1, 2, 2, 3, 2, 2, 3, 4,
- 4, 5, 6, 8, 9, 12, 14, 25
- }
-};
-
-/* offsets for quantizer representation levels*/
-const WebRtc_UWord16 WebRtcIsacfix_kOfLevelsGain[3] = {
- 0, 392, 779
-};
-
-const WebRtc_UWord16 WebRtcIsacfix_kOfLevelsShape[3] = {
- 0, 578, 1152
-};
-
-/* quantizer representation levels */
-
-
-
-const WebRtc_Word32 WebRtcIsacfix_kLevelsGainQ17[1176] = {
- -364547,-231664,-102123,-573,104807,238257,368823,-758583,-640135,-510291
- ,-377662,-252785,-113177,2627,112906,248601,389461,522691,644517,763974
- ,-538963,-368179,-245823,-106095,-890,104299,241111,350730,493190,-800763
- ,-646230,-510239,-382115,-248623,-111829,-2983,113852,251105,388114,519757
- ,644048,774712,896334,1057931,-770009,-635310,-503690,-375087,-248106,-108525
- ,-105,108259,243472,377948,519271,-1160885,-1032044,-914636,-777593,-647891
- ,-518408,-388028,-254321,-115293,-598,117849,251296,385367,515510,652727
- ,777432,920363,1038625,1153637,1316836,-632287,-505430,-379430,-248458,-118013
- ,-888,118762,250266,381650,513327,652169,766471,932113,-2107480,-1971030
- ,-1827020,-1698263,-1558670,-1436314,-1305377,-1172252,-1047355,-914202,-779651,-651001
- ,-520999,-390394,-255761,-123490,-1893,126839,256703,385493,518607,651760
- ,782750,908693,1044131,1163865,1311066,1424177,1582628,1709823,1831740,1955391
- ,-1423044,-1288917,-1181281,-1043222,-911770,-780354,-646799,-522664,-386721,-258266
- ,-128060,-1101,128233,259996,390336,519590,649290,778701,908010,1040796
- ,1161235,1306889,1441882,-4446002,-4301031,-4194304,-4080591,-3947740,-3808975,-3686530
- ,-3567839,-3383251,-3287089,-3136577,-3017405,-2869860,-2751321,-2619984,-2482932,-2354790
- ,-2223147,-2090669,-1964135,-1831208,-1706697,-1570817,-1446008,-1305386,-1175773,-1046066
- ,-915356,-785120,-653614,-524331,-393767,-260442,-130187,-799,128841,261466
- ,393616,520542,652117,784613,914159,1045399,1181072,1308971,1442502,1570346
- ,1693912,1843986,1966014,2090474,2224869,2364593,2475934,2628403,2752512,2856640
- ,-4192441,-4063232,-3917821,-3799195,-3666233,-3519199,-3411021,-3269192,-3135684,-3008826
- ,-2880875,-2747342,-2620981,-2494872,-2354979,-2229718,-2098939,-1964971,-1835399,-1703452
- ,-1572806,-1440482,-1311794,-1179338,-1046521,-919823,-785914,-655335,-523416,-395507
- ,-264833,-132184,-2546,131698,256217,391372,522688,651248,789964,909618
- ,1035305,1179145,1313824,1436934,1552353,1693722,1815508,1972826,2096328,2228224
- ,2359296,2490368,2598848,-6160384,-6029312,-5881382,-5767168,-5636096,-5505024,-5373952
- ,-5228418,-5110384,-4954923,-4880576,-4710990,-4587364,-4471340,-4333905,-4211513,-4051293
- ,-3907927,-3800105,-3675961,-3538640,-3413663,-3271148,-3152105,-3019103,-2869647,-2744015
- ,-2620639,-2479385,-2364211,-2227611,-2095427,-1974497,-1834168,-1703561,-1568961,-1439826
- ,-1309192,-1174050,-1050191,-917836,-786015,-656943,-518934,-394831,-257708,-128041
- ,1610,128991,264442,393977,521383,653849,788164,918641,1049122,1181971
- ,1308934,1439505,1571808,1706305,1836318,1966235,2097269,2228990,2357005,2490292
- ,2617400,2749680,2881234,3014880,3145637,3276467,3409099,3536637,3671493,3802918
- ,3929740,4065036,4194143,4325999,4456126,4586857,4717194,4843923,4978676,5110913
- ,5245281,5371394,5499780,5633779,5762611,5897682,6028688,6167546,6296465,6421682
- ,6548882,6682074,6809432,6941956,7078143,7204509,7334296,7475137,7609896,7732044
- ,7861604,8002039,8131670,8259222,8390299,8522399,8650037,8782348,8908402,9037815
- ,9164594,9300338,9434679,9574500,9699702,9833934,9948152,10083972,10244937,10332822
- ,10485760,10600122,10760754,10892964,11010048,11111004,11272192,11403264,11525091,11624984
- ,11796480,11915146,-393216,-262144,-101702,-740,100568,262144,393216,-786432
- ,-655360,-524288,-383907,-243301,-94956,-156,95547,269629,416691,524288
- ,655360,-393216,-262144,-88448,-37,87318,262144,393216,524288,-917504
- ,-786432,-655360,-495894,-373308,-267503,-93211,4119,91308,250895,393216
- ,526138,655360,786432,917504,-786432,-655360,-524288,-393216,-262144,-83497
- ,222,86893,240922,393216,524288,-1048576,-917504,-790472,-655360,-508639
- ,-383609,-262016,-95550,-3775,96692,256797,364847,534906,655360,786432
- ,889679,1048576,1179648,1310720,1441792,-655360,-524288,-377684,-248408,-93690
- ,1261,95441,227519,393216,524288,655360,786432,917504,-2097152,-1966080
- ,-1809470,-1703936,-1572864,-1441792,-1314289,-1195149,-1056205,-917504,-809951,-657769
- ,-521072,-383788,-248747,-106350,-2944,105550,243408,388548,521064,628732
- ,786432,885456,1064548,1179648,1310720,1441792,1572864,1703936,1835008,-1441792
- ,-1310720,-1179648,-1037570,-888492,-767774,-646634,-519935,-373458,-248029,-111915
- ,760,111232,247735,379432,507672,672699,786432,917504,1048576,1179648
- ,1310720,1441792,-4456448,-4325376,-4194304,-4063232,-3932160,-3801088,-3670016,-3538944
- ,-3407872,-3276800,-3145728,-3014656,-2883584,-2752512,-2647002,-2490368,-2359296,-2228224
- ,-2097152,-1951753,-1835008,-1703936,-1594177,-1462001,-1289150,-1160774,-1025917,-924928
- ,-782509,-641294,-516191,-386630,-251910,-118886,5210,121226,253949,386008
- ,517973,649374,780064,917783,1052462,1183856,1290593,1419389,1556641,1699884
- ,1835008,1988314,2090470,2228224,2359296,2490368,2621440,2752512,2883584,-3801088
- ,-3643514,-3539937,-3409931,-3263294,-3145658,-3012952,-2879230,-2752359,-2622556,-2483471
- ,-2357556,-2226500,-2093112,-1965892,-1833664,-1701035,-1567767,-1440320,-1310556,-1178339
- ,-1049625,-916812,-786477,-655277,-525050,-393773,-264828,-130696,-480,132126
- ,260116,394197,527846,652294,785563,917183,1049511,1175958,1308161,1438759
- ,1572253,1698835,1828535,1967072,2089391,2212798,2348901,2461547,2621440,2752512
- ,2883584,-7309870,-7203780,-7062699,-6939106,-6790819,-6672036,-6553600,-6422317,-6288422
- ,-6164694,-6026456,-5901410,-5754168,-5621459,-5502710,-5369686,-5240454,-5120712,-4976140
- ,-4847970,-4723070,-4589083,-4450923,-4324680,-4189892,-4065551,-3931803,-3800209,-3668539
- ,-3539395,-3404801,-3277470,-3141389,-3016710,-2885724,-2752612,-2618541,-2486762,-2354153
- ,-2225059,-2094984,-1968194,-1830895,-1699508,-1575743,-1444516,-1308683,-1179714,-1053088
- ,-917981,-783707,-653900,-524980,-395409,-260309,-131948,-3452,132113,263241
- ,392185,522597,654134,788288,919810,1045795,1179210,1314201,1444235,1574447
- ,1705193,1834009,1967332,2098102,2229019,2359147,2489859,2619878,2754966,2879671
- ,3014438,3146143,3276733,3405958,3542196,3667493,3798815,3932961,4062458,4187125
- ,4322346,4454875,4587752,4716809,4848274,4975027,5111957,5242215,5373085,5501158
- ,5640140,5762918,5895358,6024008,6157906,6290628,6422713,6546339,6675888,6815606
- ,6955288,7077501,7211630,7337893,7473635,7607175,7728310,7866475,7999658,8127888
- ,8241758,8386483,8522550,8641582,8771915,8922139,9038632,9179385,9313426,9437184
- ,9568256,9699328,9830400,9952933,10120004,10223616,10354688,10474645,10616832,-393216
- ,-262144,-85425,-121,82533,262144,393216,-786432,-655360,-524288,-379928
- ,-222821,-95200,287,95541,227093,393216,493567,655360,786432,-393216
- ,-262144,-86805,510,86722,262144,393216,524288,-1048576,-917504,-786432
- ,-624456,-529951,-395071,-241627,-101168,81,99975,241605,393216,524288
- ,655360,786432,917504,-786432,-655360,-524288,-393216,-230359,-95619,-137
- ,94425,226222,393216,524288,-1179648,-1048576,-917504,-773841,-655360,-492258
- ,-379715,-244707,-103621,-434,104523,242680,381575,523659,650565,786432
- ,917504,1048576,1179648,1310720,-786432,-629344,-524288,-376757,-242858,-101932
- ,-2715,107155,239212,366480,514943,655360,786432,917504,-2228224,-2097152
- ,-1966080,-1835008,-1703936,-1572864,-1441792,-1284584,-1179648,-1048819,-934658,-777181
- ,-626371,-515660,-377493,-248975,-113036,436,113584,248354,379718,512475
- ,653932,796494,917504,1048576,1179648,1310720,1441792,1572864,1703936,1835008
- ,-1572864,-1441792,-1297608,-1161159,-1032316,-917092,-779770,-647384,-515529,-384269
- ,-250003,-119252,1053,118111,249512,380545,512039,648101,770656,907003
- ,1021725,1178082,1310720,1441792,-4587520,-4456448,-4325376,-4194304,-4063232,-3932160
- ,-3801088,-3670016,-3538944,-3407872,-3276800,-3145728,-2999335,-2883584,-2752512,-2621440
- ,-2490368,-2359296,-2228224,-2112691,-1966080,-1848781,-1709830,-1566109,-1438427,-1303530
- ,-1176124,-1040936,-913876,-784585,-652025,-518361,-385267,-256342,-127297,-2733
- ,125422,257792,389363,519911,651106,783805,909407,1044143,1174156,1309267
- ,1436173,1553771,1708958,1814083,1967036,2095386,2255169,2359296,2478303,2621440
- ,2752512,-4456448,-4325376,-4194304,-4063232,-3932160,-3797524,-3670016,-3560250,-3413217
- ,-3257719,-3166416,-2986626,-2878000,-2781144,-2625383,-2495465,-2346792,-2230930,-2077063
- ,-1949225,-1819274,-1697261,-1568664,-1443074,-1304302,-1175289,-1043794,-913423,-785561
- ,-652104,-522835,-392667,-260517,-130088,-2,129509,260990,391931,522470
- ,655770,784902,917093,1046445,1176951,1303121,1441362,1565401,1702022,1822856
- ,1952852,2090384,2214607,2338436,2457483,2621440,-8781824,-8650752,-8519680,-8388608
- ,-8260828,-8126464,-8003337,-7859030,-7750057,-7602176,-7471104,-7340032,-7193045,-7090588
- ,-6946816,-6843344,-6676635,-6557575,-6447804,-6277614,-6159736,-6035729,-5884723,-5739567
- ,-5634818,-5489867,-5372864,-5243300,-5098939,-4988639,-4856258,-4728494,-4591717,-4447428
- ,-4322409,-4192918,-4062638,-3934141,-3797545,-3673373,-3531587,-3407391,-3277404,-3147797
- ,-3013578,-2886548,-2749811,-2616428,-2490949,-2361301,-2228482,-2096883,-1964343,-1831754
- ,-1702201,-1572495,-1442012,-1309242,-1182451,-1048996,-916905,-786510,-657079,-524730
- ,-393672,-261313,-128743,166,130678,261334,393287,524155,655570,786839
- ,917353,1052167,1179013,1309360,1442634,1571153,1703961,1832027,1965014,2097912
- ,2224861,2355341,2490455,2623051,2753484,2877015,3015783,3144157,3273705,3405255
- ,3542006,3669580,3802417,3935413,4065088,4190896,4333521,4456355,4579781,4713832
- ,4845707,4978625,5113278,5243817,5382318,5500592,5638135,5761179,5900822,6029270
- ,6186398,6297816,6436435,6559163,6666389,6806548,6950461,7086078,7195777,7350973
- ,7480132,7614852,7743514,7847288,8014762,8126464,8257536,8388608,8519680,8650752
- ,8781824,8912896,9043968,9175040,9306112,9437184
-};
-
-
-
-const WebRtc_Word16 WebRtcIsacfix_kLevelsShapeQ10[1735] = {
- 0, 0, -1, 0, 0, 1, 0, 1, 0, -821
- , 1, -763, -1, 656, -620, 0, 633, -636, 4, 615
- , -630, 1, 649, -1773, -670, 5, 678, 1810, -1876, -676
- , 0, 691, 1843, -1806, -743, -1, 749, 1795, 2920, -2872
- , -1761, -772, -3, 790, 1763, 2942, 0, 0, 0, 0
- , -792, 2, 0, 0, 1, 0, -854, 0, -702, -1
- , 662, -624, -5, 638, -611, -6, 638, -647, 0, 651
- , -685, -4, 679, 2123, -1814, -693, 0, 664, 1791, -1735
- , -737, 0, 771, 1854, 2873, -2867, -1842, -793, -1, 821
- , 1826, 2805, 3922, 0, 0, 0, -1, -779, 1, 786
- , 1, -708, 0, 789, -799, 1, 797, -663, 2, 646
- , -600, 3, 609, -600, 1, 658, 1807, -627, -3, 612
- , -625, 3, 632, -1732, -674, 1, 672, 2048, -1768, -715
- , 0, 724, 1784, -3881, -3072, -1774, -719, -1, 730, 1811
- , -2963, -1829, -806, -1, 816, 1795, 3050, -5389, -3784, -2942
- , -1893, -865, -12, 867, 1885, 2945, 3928, -2, 1, 4
- , 0, -694, 2, 665, -598, 5, 587, -599, -1, 661
- , -656, -7, 611, -607, 5, 603, -618, -4, 620, -1794
- , -645, -2, 654, -655, -1, 658, -1801, -700, 5, 707
- , 1927, -1752, -745, -8, 752, 1843, -2838, -1781, -801, 11
- , 796, 1811, 2942, 3866, -3849, -3026, -1848, -819, 2, 827
- , 1825, 2963, -3873, -2904, -1869, -910, -6, 903, 1902, 2885
- , 3978, 5286, -7168, -6081, -4989, -3968, -2963, -1970, -943, -2
- , 953, 1951, 2968, 3974, 5009, 6032, -2, 3, -1024, 2
- , 1024, -637, 1, 669, -613, -7, 630, -603, 4, 612
- , -612, 0, 590, -645, -11, 627, -657, -2, 671, 1849
- , -1853, -694, 2, 702, 1838, -3304, -1780, -736, -8, 732
- , 1772, -1709, -755, -6, 760, 1780, -2994, -1780, -800, 8
- , 819, 1830, 2816, -4096, -2822, -1881, -851, -4, 855, 1872
- , 2840, 3899, -3908, -2904, -1878, -887, 6, 897, 1872, 2942
- , 4008, -4992, -3881, -2933, -1915, -928, 1, 937, 1919, 2900
- , 4009, 4881, -6848, -6157, -5065, -3981, -2983, -1972, -978, -1
- , 968, 1979, 2988, 4008, 5007, 6108, 7003, 8051, 9027,-13272
- ,-12012,-11228,-10213, -9261, -8084, -7133, -6075, -5052, -4050, -3036
- , -2014, -996, -4, 1007, 2031, 3038, 4049, 5074, 6134, 7069
- , 8094, 9069, 10212, 11049, 12104, 51, -1024, -13, 1024, -609
- , -107, 613, -2048, -687, -95, 667, 2048, -3072, -1724, -785
- , -34, 732, 1819, -2048, -703, -26, 681, 2048, -2048, -686
- , -9, 665, 2048, -2048, -702, 37, 748, 1723, -4096, -2786
- , -1844, -837, 37, 811, 1742, 3072, -4096, -2783, -1848, -881
- , 39, 898, 1843, 2792, 3764, -5120, -4096, -2923, -1833, -852
- , -14, 862, 1824, 2834, 4096, -6144, -5120, -3914, -2842, -1870
- , -886, -27, 888, 1929, 2931, 4051, -7168, -6144, -5120, -3866
- , -2933, -1915, -927, 64, 933, 1902, 2929, 3912, 5063, 6144
- ,-11264,-10240, -9216, -8192, -7086, -6144, -5039, -3972, -2943, -1929
- , -941, 3, 938, 1942, 2959, 3933, 4905, 6088, 6983, 8192
- , -9216, -8192, -7202, -6088, -4983, -4019, -2955, -1975, -966, 17
- , 997, 1981, 2967, 3990, 4948, 6022, 6967, 8192,-13312,-12288
- ,-11264,-10240, -9216, -8049, -6997, -6040, -5026, -4043, -3029, -2034
- , -1015, -23, 984, 1997, 3010, 4038, 5002, 6015, 6946, 8061
- , 9216, 10240,-12381,-11264,-10240, -9060, -8058, -7153, -6085, -5075
- , -4051, -3042, -2037, -1017, -5, 1007, 2028, 3035, 4050, 5088
- , 6111, 7160, 8156, 9215, 10095, 11229, 12202, 13016,-26624,-25600
- ,-24582,-23671,-22674,-21400,-20355,-19508,-18315,-17269,-16361,-15299
- ,-14363,-13294,-12262,-11237,-10203, -9227, -8165, -7156, -6116, -5122
- , -4076, -3056, -2043, -1020, -8, 1027, 2047, 3065, 4110, 5130
- , 6125, 7168, 8195, 9206, 10230, 11227, 12256, 13304, 14281, 15316
- , 16374, 17382, 18428, 19388, 20361, 21468, 22448, 23781, 0, 0
- , -1, 0, -2, 1024, 0, 0, 0, -1, 1024, -1024
- , 1, -1024, 4, 1024, -1024, 2, 1024, -1024, 2, 1024
- , -2048, -1024, -4, 1024, -1024, 2, 1024, -2048, -1024, -3
- , 1024, 2048, -2048, -1024, 4, 1024, 2048, -3072, -2048, -1024
- , -1, 662, 2048, 0, 1, 0, 0, 1, -2, -2
- , 0, 2, 1024, -1, 1024, -1024, 4, 1024, -1024, 1
- , 1024, -1024, 1, 1024, -2048, -781, -4, 844, -807, -5
- , 866, -2048, -726, -13, 777, 2048, -2048, -643, -4, 617
- , 2048, 3072, -3072, -2048, -629, 1, 630, 2048, 3072, 0
- , -1, 1, -2, 2, 1, -1024, 5, -1024, 6, 1024
- , -1024, 4, 1024, -1024, 1, 1024, -1024, -9, 1024, -673
- , -7, 655, -2048, -665, -15, 716, -2048, -647, 4, 640
- , 2048, -2048, -615, -1, 635, 2048, -2048, -613, 10, 637
- , 2048, 3072, -3072, -2048, -647, -3, 641, 2048, 3072, -5120
- , -4096, -3072, -2048, -681, 6, 685, 2048, 3072, 4096, 1
- , 1, 0, -1, 1024, -1024, -3, 1024, -1024, 6, 1024
- , -1024, -1, 769, -733, 0, 1024, -876, -2, 653, -1024
- , -4, 786, -596, -13, 595, -634, -2, 638, 2048, -2048
- , -620, -5, 620, 2048, -4096, -3072, -2048, -639, 11, 655
- , 2048, 3072, -3072, -2048, -659, 5, 663, 2048, -3072, -1823
- , -687, 22, 695, 2048, 3072, 4096, -4096, -3072, -1848, -715
- , -3, 727, 1816, 3072, 4096, 5120, -8192, -7168, -6144, -5120
- , -4096, -2884, -1771, -756, -14, 775, 1844, 3072, 4096, 5120
- , 6144, -1, 1, 0, -1024, 2, 815, -768, 2, 708
- , -1024, -3, 693, -661, -7, 607, -643, -5, 609, -624
- , 3, 631, -682, -3, 691, 2048, -2048, -640, 5, 650
- , 2048, -3072, -2048, -701, 9, 704, 2048, 3072, -3072, -2048
- , -670, 10, 674, 2048, 3072, -5120, -4096, -3072, -1749, -738
- , 0, 733, 1811, 3072, 4096, 5120, -4096, -3072, -1873, -753
- , 0, 756, 1874, 3072, 4096, -5120, -4096, -2900, -1838, -793
- , -6, 793, 1868, 2837, 4096, 5120, -7168, -6144, -5120, -4096
- , -2832, -1891, -828, 1, 828, 1901, 2823, 3912, 5120, 6144
- , 7168, 8192,-13312,-12288,-11264,-10240, -9216, -8192, -7168, -6144
- , -5120, -3976, -3004, -1911, -869, 7, 869, 1932, 3024, 3992
- , 5009, 6144, 7168, 8192, 9216, 10240, 11264, -4, 1024, -629
- , -22, 609, -623, 9, 640, -2048, -768, 1, 682, -2048
- , -741, 49, 722, 2048, -3072, -1706, -808, -20, 768, 1750
- , -1684, -727, -29, 788, 1840, 3033, -1758, -784, 0, 801
- , 1702, -3072, -1813, -814, 38, 820, 1884, 2927, -4096, -3241
- , -1839, -922, 25, 882, 1886, 2812, -4096, -2982, -1923, -894
- , 84, 912, 1869, 2778, 4096, -4928, -3965, -2902, -1920, -883
- , 3, 917, 1953, 2921, 3957, 4922, 6144, 7168, -5120, -3916
- , -2897, -1949, -930, 31, 959, 1934, 2901, 3851, 5120, -9216
- , -8192, -7046, -6029, -5030, -4034, -2980, -1969, -1013, -76, 963
- , 1963, 2901, 3929, 4893, 6270, 7168, 8192, 9216,-12288,-11264
- ,-10240, -9216, -8192, -6846, -6123, -5108, -4008, -3000, -1963, -954
- , -6, 958, 1992, 3009, 4020, 5085, 6097, 7168, 8192, 9216
- ,-11264,-10139, -9194, -8127, -7156, -6102, -5053, -4049, -3036, -2025
- , -1009, -34, 974, 1984, 3034, 4028, 5138, 6000, 7057, 8166
- , 9070, 10033, 11360, 12288,-13312,-12288,-10932,-10190, -9120, -8123
- , -7128, -6103, -5074, -4081, -3053, -2029, -989, -4, 1010, 2028
- , 3051, 4073, 5071, 6099, 7132, 8147, 9295, 10159, 11023, 12263
- , 13312, 14336,-25600,-24576,-23552,-22529,-21504,-20480,-19456,-18637
- ,-17425,-16165,-15316,-14327,-13606,-12135,-11182,-10107, -9153, -8144
- , -7146, -6160, -5129, -4095, -3064, -2038, -1025, 1, 1031, 2072
- , 3074, 4088, 5123, 6149, 7157, 8173, 9198, 10244, 11250, 12268
- , 13263, 14289, 15351, 16370, 17402, 18413, 19474, 20337, 21386, 22521
- , 23367, 24350, 0, 0, 0, 0, 0, 0, 0, 0
- , -1024, 0, 1024, -1024, 0, 1024, -1024, 0, 1024, -1024
- , 0, 1024, -1024, 0, 1024, -773, 0, 1024, -674, 0
- , 645, -2048, -745, 0, 628, 2048, -2048, -712, 0, 681
- , 2048, 3072, -3072, -2048, -673, 0, 682, 1964, 3257, 0
- , 0, 0, 0, 0, 0, 0, 0, -1024, 0, 1024
- , -1024, 0, 1024, -1024, 0, 1024, -705, 0, 623, -771
- , 0, 1024, -786, 0, 688, -631, 0, 652, 2048, -2048
- , -627, -1, 666, 2048, -3072, -1756, -694, 0, 674, 2048
- , -3098, -1879, -720, 5, 694, 1886, 2958, 4096, 0, 0
- , 0, 0, 1024, 0, 0, 1024, -769, 0, 1024, -1024
- , 0, 1024, -1024, 0, 1024, -817, 0, 734, -786, 0
- , 651, -638, 0, 637, -623, 0, 671, -652, 0, 619
- , 2048, -2048, -670, -1, 663, 2048, -1908, -680, 1, 686
- , 2048, 3072, 4096, -4096, -3072, -1833, -711, 0, 727, 1747
- , 3072, 4096, -4096, -2971, -1826, -762, 2, 766, 1832, 2852
- , 3928, 5079, 0, 0, 0, -1024, 0, 1024, -1024, 0
- , -656, 0, 1024, -599, 0, 620, -1024, 0, 1024, -603
- , 0, 622, -643, 0, 660, -599, 0, 611, -641, -1
- , 651, 2048, -2048, -648, -2, 647, 1798, -3072, -2048, -672
- , 2, 670, 2048, -3072, -1780, -694, -1, 706, 1751, 3072
- , -3072, -1862, -757, 7, 739, 1798, 3072, 4096, -5120, -4096
- , -3253, -1811, -787, 3, 782, 1887, 3123, 4096, -7252, -6144
- , -5354, -4060, -2864, -1863, -820, -11, 847, 1903, 2970, 3851
- , 4921, 5957, 7168, 8192, 9306, 0, 0, -1024, 0, 1024
- , -726, 0, 706, -692, 0, 593, -598, 0, 616, -624
- , 0, 616, -605, 0, 613, -2048, -652, 1, 635, 2048
- , -2048, -647, -1, 660, 2048, -1811, -668, -2, 685, 2048
- , -1796, -731, -2, 730, 1702, 3072, -3072, -1766, -747, -4
- , 756, 1770, 3072, -4096, -3024, -1762, -783, 4, 771, 1781
- , 3072, -5120, -4057, -2807, -1832, -822, 0, 816, 1804, 2851
- , 3949, 5120, -6144, -4899, -3927, -2920, -1893, -874, -2, 868
- , 1881, 2905, 3960, 4912, 6144, -9216, -8192, -7168, -6225, -4963
- , -3943, -2956, -1890, -902, 0, 897, 1914, 2916, 3984, 4990
- , 6050, 7168,-11264,-10217, -9114, -8132, -7035, -5988, -4984, -4000
- , -2980, -1962, -927, 7, 931, 1956, 2981, 4031, 4972, 6213
- , 7227, 8192, 9216, 10240, 11170, 12288, 13312, 14336, 0, 1024
- , -557, 1, 571, -606, -4, 612, -1676, -707, 10, 673
- , 2048, -2048, -727, 5, 686, -3072, -1772, -755, 12, 716
- , 1877, -1856, -786, 2, 786, 1712, -1685, -818, -16, 863
- , 1729, -3072, -1762, -857, 3, 866, 1838, 2841, -3862, -2816
- , -1864, -925, -2, 923, 1897, 2779, -2782, -1838, -920, -28
- , 931, 1951, 2835, 3804, -4815, -4001, -2940, -1934, -959, -22
- , 975, 1957, 2904, 3971, 4835, -5148, -3892, -2944, -1953, -986
- , -11, 989, 1968, 2939, 3949, 4947, 5902, -9216, -8192, -6915
- , -6004, -4965, -4013, -3009, -1977, -987, -1, 982, 1972, 3000
- , 3960, 4939, 5814, -8976, -7888, -7084, -5955, -5043, -4009, -2991
- , -2002, -1000, -8, 993, 2011, 3023, 4026, 5028, 6023, 7052
- , 8014, 9216,-11240,-10036, -9125, -8118, -7105, -6062, -5048, -4047
- , -3044, -2025, -1009, -1, 1011, 2023, 3042, 4074, 5085, 6108
- , 7119, 8142, 9152, 10114, 11141, 12250, 13307,-15360,-14099,-13284
- ,-12291,-11223,-10221, -9152, -8147, -7128, -6104, -5077, -4072, -3062
- , -2033, -1020, 7, 1018, 2038, 3059, 4081, 5084, 6109, 7102
- , 8128, 9134, 10125, 11239, 12080,-23552,-22528,-21504,-20480,-19456
- ,-18159,-17240,-16291,-15364,-14285,-13305,-12271,-11233,-10217, -9198
- , -8175, -7157, -6134, -5122, -4089, -3071, -2047, -1018, 3, 1026
- , 2041, 3077, 4090, 5108, 6131, 7150, 8172, 9175, 10196, 11272
- , 12303, 13273, 14328, 15332, 16334, 17381, 18409, 19423, 20423, 21451
- , 22679, 23391, 24568, 25600, 26589
-};
-
-/* cdf tables for quantizer indices */
-const WebRtc_UWord16 WebRtcIsacfix_kCdfGain[1212] = {
- 0, 13, 301, 3730, 61784, 65167, 65489, 65535, 0, 17,
- 142, 314, 929, 2466, 7678, 56450, 63463, 64740, 65204, 65426,
- 65527, 65535, 0, 8, 100, 724, 6301, 60105, 65125, 65510,
- 65531, 65535, 0, 13, 117, 368, 1068, 3010, 11928, 53603,
- 61177, 63404, 64505, 65108, 65422, 65502, 65531, 65535, 0, 4,
- 17, 96, 410, 1859, 12125, 54361, 64103, 65305, 65497, 65535,
- 0, 4, 88, 230, 469, 950, 1746, 3228, 6092, 16592,
- 44756, 56848, 61256, 63308, 64325, 64920, 65309, 65460, 65502, 65522,
- 65535, 0, 88, 352, 1675, 6339, 20749, 46686, 59284, 63525,
- 64949, 65359, 65502, 65527, 65535, 0, 13, 38, 63, 117,
- 234, 381, 641, 929, 1407, 2043, 2809, 4032, 5753, 8792,
- 14407, 24308, 38941, 48947, 55403, 59293, 61411, 62688, 63630, 64329,
- 64840, 65188, 65376, 65472, 65506, 65527, 65531, 65535, 0, 8,
- 29, 75, 222, 615, 1327, 2801, 5623, 9931, 16094, 24966,
- 34419, 43458, 50676, 56186, 60055, 62500, 63936, 64765, 65225, 65435,
- 65514, 65535, 0, 8, 13, 15, 17, 21, 33, 59,
- 71, 92, 151, 243, 360, 456, 674, 934, 1223, 1583,
- 1989, 2504, 3031, 3617, 4354, 5154, 6163, 7411, 8780, 10747,
- 12874, 15591, 18974, 23027, 27436, 32020, 36948, 41830, 46205, 49797,
- 53042, 56094, 58418, 60360, 61763, 62818, 63559, 64103, 64509, 64798,
- 65045, 65162, 65288, 65363, 65447, 65506, 65522, 65531, 65533, 65535,
- 0, 4, 6, 25, 38, 71, 138, 264, 519, 808,
- 1227, 1825, 2516, 3408, 4279, 5560, 7092, 9197, 11420, 14108,
- 16947, 20300, 23926, 27459, 31164, 34827, 38575, 42178, 45540, 48747,
- 51444, 54090, 56426, 58460, 60080, 61595, 62734, 63668, 64275, 64673,
- 64936, 65112, 65217, 65334, 65426, 65464, 65477, 65489, 65518, 65527,
- 65529, 65531, 65533, 65535, 0, 2, 4, 8, 10, 12,
- 14, 16, 21, 33, 50, 71, 84, 92, 105, 138,
- 180, 255, 318, 377, 435, 473, 511, 590, 682, 758,
- 913, 1097, 1256, 1449, 1671, 1884, 2169, 2445, 2772, 3157,
- 3563, 3944, 4375, 4848, 5334, 5820, 6448, 7101, 7716, 8378,
- 9102, 9956, 10752, 11648, 12707, 13670, 14758, 15910, 17187, 18472,
- 19627, 20649, 21951, 23169, 24283, 25552, 26862, 28227, 29391, 30764,
- 31882, 33213, 34432, 35600, 36910, 38116, 39464, 40729, 41872, 43144,
- 44371, 45514, 46762, 47813, 48968, 50069, 51032, 51974, 52908, 53737,
- 54603, 55445, 56282, 56990, 57572, 58191, 58840, 59410, 59887, 60264,
- 60607, 60946, 61269, 61516, 61771, 61960, 62198, 62408, 62558, 62776,
- 62985, 63207, 63408, 63546, 63739, 63906, 64070, 64237, 64371, 64551,
- 64677, 64836, 64999, 65095, 65213, 65284, 65338, 65380, 65426, 65447,
- 65472, 65485, 65487, 65489, 65502, 65510, 65512, 65514, 65516, 65518,
- 65522, 65531, 65533, 65535, 0, 2, 4, 6, 65528, 65531,
- 65533, 65535, 0, 2, 4, 6, 8, 10, 222, 65321,
- 65513, 65528, 65531, 65533, 65535, 0, 2, 4, 50, 65476,
- 65529, 65531, 65533, 65535, 0, 2, 4, 6, 8, 12,
- 38, 544, 64936, 65509, 65523, 65525, 65529, 65531, 65533, 65535,
- 0, 2, 4, 6, 8, 10, 1055, 64508, 65528, 65531,
- 65533, 65535, 0, 2, 4, 6, 8, 10, 12, 123,
- 3956, 62999, 65372, 65495, 65515, 65521, 65523, 65525, 65527, 65529,
- 65531, 65533, 65535, 0, 2, 4, 12, 53, 4707, 59445,
- 65467, 65525, 65527, 65529, 65531, 65533, 65535, 0, 2, 4,
- 6, 8, 10, 12, 14, 16, 38, 40, 50, 67,
- 96, 234, 929, 14345, 55750, 64866, 65389, 65462, 65514, 65517,
- 65519, 65521, 65523, 65525, 65527, 65529, 65531, 65533, 65535, 0,
- 2, 4, 6, 8, 10, 15, 35, 91, 377, 1946,
- 13618, 52565, 63714, 65184, 65465, 65520, 65523, 65525, 65527, 65529,
- 65531, 65533, 65535, 0, 2, 4, 6, 8, 10, 12,
- 14, 16, 18, 20, 22, 24, 26, 28, 30, 32,
- 34, 36, 38, 40, 42, 44, 46, 48, 50, 52,
- 54, 82, 149, 362, 751, 1701, 4239, 12893, 38627, 55072,
- 60875, 63071, 64158, 64702, 65096, 65283, 65412, 65473, 65494, 65505,
- 65508, 65517, 65519, 65521, 65523, 65525, 65527, 65529, 65531, 65533,
- 65535, 0, 2, 15, 23, 53, 143, 260, 418, 698,
- 988, 1353, 1812, 2411, 3144, 4015, 5143, 6401, 7611, 8999,
- 10653, 12512, 14636, 16865, 19404, 22154, 24798, 27521, 30326, 33102,
- 35790, 38603, 41415, 43968, 46771, 49435, 52152, 54715, 57143, 59481,
- 61178, 62507, 63603, 64489, 64997, 65257, 65427, 65473, 65503, 65520,
- 65529, 65531, 65533, 65535, 0, 3, 6, 9, 26, 32,
- 44, 46, 64, 94, 111, 164, 205, 254, 327, 409,
- 506, 608, 733, 885, 1093, 1292, 1482, 1742, 1993, 2329,
- 2615, 3029, 3374, 3798, 4257, 4870, 5405, 5992, 6618, 7225,
- 7816, 8418, 9051, 9761, 10532, 11380, 12113, 13010, 13788, 14594,
- 15455, 16361, 17182, 18088, 18997, 20046, 20951, 21968, 22947, 24124,
- 25296, 26547, 27712, 28775, 29807, 30835, 31709, 32469, 33201, 34014,
- 34876, 35773, 36696, 37620, 38558, 39547, 40406, 41277, 42367, 43290,
- 44445, 45443, 46510, 47684, 48973, 50157, 51187, 52242, 53209, 54083,
- 55006, 55871, 56618, 57293, 57965, 58556, 59222, 59722, 60180, 60554,
- 60902, 61250, 61554, 61837, 62100, 62372, 62631, 62856, 63078, 63324,
- 63557, 63768, 63961, 64089, 64235, 64352, 64501, 64633, 64770, 64887,
- 65001, 65059, 65121, 65188, 65246, 65302, 65346, 65390, 65428, 65463,
- 65477, 65506, 65515, 65517, 65519, 65521, 65523, 65525, 65527, 65529,
- 65531, 65533, 65535, 0, 2, 4, 109, 65332, 65531, 65533,
- 65535, 0, 2, 4, 6, 8, 25, 1817, 63874, 65511,
- 65527, 65529, 65531, 65533, 65535, 0, 2, 4, 907, 65014,
- 65529, 65531, 65533, 65535, 0, 2, 4, 6, 8, 10,
- 12, 132, 2743, 62708, 65430, 65525, 65527, 65529, 65531, 65533,
- 65535, 0, 2, 4, 6, 8, 35, 3743, 61666, 65485,
- 65531, 65533, 65535, 0, 2, 4, 6, 8, 10, 23,
- 109, 683, 6905, 58417, 64911, 65398, 65497, 65518, 65525, 65527,
- 65529, 65531, 65533, 65535, 0, 2, 4, 6, 53, 510,
- 10209, 55212, 64573, 65441, 65522, 65529, 65531, 65533, 65535, 0,
- 2, 4, 6, 8, 10, 12, 14, 16, 18, 20,
- 22, 32, 90, 266, 1037, 3349, 14468, 50488, 62394, 64685,
- 65341, 65480, 65514, 65519, 65521, 65523, 65525, 65527, 65529, 65531,
- 65533, 65535, 0, 2, 4, 6, 9, 16, 37, 106,
- 296, 748, 1868, 5733, 18897, 45553, 60165, 63949, 64926, 65314,
- 65441, 65508, 65524, 65529, 65531, 65533, 65535, 0, 2, 4,
- 6, 8, 10, 12, 14, 16, 18, 20, 22, 24,
- 26, 28, 30, 32, 34, 36, 38, 40, 42, 44,
- 46, 48, 50, 83, 175, 344, 667, 1293, 2337, 4357,
- 8033, 14988, 28600, 43244, 52011, 57042, 59980, 61779, 63065, 63869,
- 64390, 64753, 64988, 65164, 65326, 65422, 65462, 65492, 65506, 65522,
- 65524, 65526, 65531, 65533, 65535, 0, 2, 4, 6, 8,
- 10, 12, 14, 16, 25, 39, 48, 55, 62, 65,
- 85, 106, 139, 169, 194, 252, 323, 485, 688, 1074,
- 1600, 2544, 3863, 5733, 8303, 11397, 15529, 20273, 25734, 31455,
- 36853, 41891, 46410, 50306, 53702, 56503, 58673, 60479, 61880, 62989,
- 63748, 64404, 64852, 65124, 65309, 65424, 65480, 65524, 65528, 65533,
- 65535, 0, 2, 4, 6, 8, 10, 12, 14, 21,
- 23, 25, 27, 29, 31, 39, 41, 43, 48, 60,
- 72, 79, 106, 136, 166, 187, 224, 252, 323, 381,
- 427, 478, 568, 660, 783, 912, 1046, 1175, 1365, 1567,
- 1768, 2024, 2347, 2659, 3049, 3529, 4033, 4623, 5281, 5925,
- 6726, 7526, 8417, 9468, 10783, 12141, 13571, 15222, 16916, 18659,
- 20350, 22020, 23725, 25497, 27201, 29026, 30867, 32632, 34323, 36062,
- 37829, 39466, 41144, 42654, 43981, 45343, 46579, 47759, 49013, 50171,
- 51249, 52283, 53245, 54148, 54938, 55669, 56421, 57109, 57791, 58464,
- 59092, 59674, 60105, 60653, 61083, 61407, 61757, 62095, 62388, 62649,
- 62873, 63157, 63358, 63540, 63725, 63884, 64046, 64155, 64278, 64426,
- 64548, 64654, 64806, 64906, 64994, 65077, 65137, 65215, 65277, 65324,
- 65354, 65409, 65437, 65455, 65462, 65490, 65495, 65499, 65508, 65511,
- 65513, 65515, 65517, 65519, 65521, 65523, 65525, 65527, 65529, 65531,
- 65533, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsacfix_kCdfShape[2059] = {
- 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535,
- 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 4,
- 65535, 0, 8, 65514, 65535, 0, 29, 65481, 65535, 0,
- 121, 65439, 65535, 0, 239, 65284, 65535, 0, 8, 779,
- 64999, 65527, 65535, 0, 8, 888, 64693, 65522, 65535, 0,
- 29, 2604, 62843, 65497, 65531, 65535, 0, 25, 176, 4576,
- 61164, 65275, 65527, 65535, 0, 65535, 0, 65535, 0, 65535,
- 0, 65535, 0, 4, 65535, 0, 65535, 0, 65535, 0,
- 65535, 0, 65535, 0, 4, 65535, 0, 33, 65502, 65535,
- 0, 54, 65481, 65535, 0, 251, 65309, 65535, 0, 611,
- 65074, 65535, 0, 1273, 64292, 65527, 65535, 0, 4, 1809,
- 63940, 65518, 65535, 0, 88, 4392, 60603, 65426, 65531, 65535,
- 0, 25, 419, 7046, 57756, 64961, 65514, 65531, 65535, 0,
- 65535, 0, 65535, 0, 65535, 0, 65535, 0, 4, 65531,
- 65535, 0, 65535, 0, 8, 65531, 65535, 0, 4, 65527,
- 65535, 0, 17, 65510, 65535, 0, 42, 65481, 65535, 0,
- 197, 65342, 65531, 65535, 0, 385, 65154, 65535, 0, 1005,
- 64522, 65535, 0, 8, 1985, 63469, 65533, 65535, 0, 38,
- 3119, 61884, 65514, 65535, 0, 4, 6, 67, 4961, 60804,
- 65472, 65535, 0, 17, 565, 9182, 56538, 65087, 65514, 65535,
- 0, 8, 63, 327, 2118, 14490, 52774, 63839, 65376, 65522,
- 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
- 17, 65522, 65535, 0, 59, 65489, 65535, 0, 50, 65522,
- 65535, 0, 54, 65489, 65535, 0, 310, 65179, 65535, 0,
- 615, 64836, 65535, 0, 4, 1503, 63965, 65535, 0, 2780,
- 63383, 65535, 0, 21, 3919, 61051, 65527, 65535, 0, 84,
- 6674, 59929, 65435, 65535, 0, 4, 255, 7976, 55784, 65150,
- 65518, 65531, 65535, 0, 4, 8, 582, 10726, 53465, 64949,
- 65518, 65535, 0, 29, 339, 3006, 17555, 49517, 62956, 65200,
- 65497, 65531, 65535, 0, 2, 33, 138, 565, 2324, 7670,
- 22089, 45966, 58949, 63479, 64966, 65380, 65518, 65535, 0, 65535,
- 0, 65535, 0, 2, 65533, 65535, 0, 46, 65514, 65535,
- 0, 414, 65091, 65535, 0, 540, 64911, 65535, 0, 419,
- 65162, 65535, 0, 976, 64790, 65535, 0, 2977, 62495, 65531,
- 65535, 0, 4, 3852, 61034, 65527, 65535, 0, 4, 29,
- 6021, 60243, 65468, 65535, 0, 84, 6711, 58066, 65418, 65535,
- 0, 13, 281, 9550, 54917, 65125, 65506, 65535, 0, 2,
- 63, 984, 12108, 52644, 64342, 65435, 65527, 65535, 0, 29,
- 251, 2014, 14871, 47553, 62881, 65229, 65518, 65535, 0, 13,
- 142, 749, 4220, 18497, 45200, 60913, 64823, 65426, 65527, 65535,
- 0, 13, 71, 264, 1176, 3789, 10500, 24480, 43488, 56324,
- 62315, 64493, 65242, 65464, 65514, 65522, 65531, 65535, 0, 4,
- 13, 38, 109, 205, 448, 850, 1708, 3429, 6276, 11371,
- 19221, 29734, 40955, 49391, 55411, 59460, 62102, 63793, 64656, 65150,
- 65401, 65485, 65522, 65531, 65535, 0, 65535, 0, 2, 65533,
- 65535, 0, 1160, 65476, 65535, 0, 2, 6640, 64763, 65533,
- 65535, 0, 2, 38, 9923, 61009, 65527, 65535, 0, 2,
- 4949, 63092, 65533, 65535, 0, 2, 3090, 63398, 65533, 65535,
- 0, 2, 2520, 58744, 65510, 65535, 0, 2, 13, 544,
- 8784, 51403, 65148, 65533, 65535, 0, 2, 25, 1017, 10412,
- 43550, 63651, 65489, 65527, 65535, 0, 2, 4, 29, 783,
- 13377, 52462, 64524, 65495, 65533, 65535, 0, 2, 4, 6,
- 100, 1817, 18451, 52590, 63559, 65376, 65531, 65535, 0, 2,
- 4, 6, 46, 385, 2562, 11225, 37416, 60488, 65026, 65487,
- 65529, 65533, 65535, 0, 2, 4, 6, 8, 10, 12,
- 42, 222, 971, 5221, 19811, 45048, 60312, 64486, 65294, 65474,
- 65525, 65529, 65533, 65535, 0, 2, 4, 8, 71, 167,
- 666, 2533, 7875, 19622, 38082, 54359, 62108, 64633, 65290, 65495,
- 65529, 65533, 65535, 0, 2, 4, 6, 8, 10, 13,
- 109, 586, 1930, 4949, 11600, 22641, 36125, 48312, 56899, 61495,
- 63927, 64932, 65389, 65489, 65518, 65531, 65533, 65535, 0, 4,
- 6, 8, 67, 209, 712, 1838, 4195, 8432, 14432, 22834,
- 31723, 40523, 48139, 53929, 57865, 60657, 62403, 63584, 64363, 64907,
- 65167, 65372, 65472, 65514, 65535, 0, 2, 4, 13, 25,
- 42, 46, 50, 75, 113, 147, 281, 448, 657, 909,
- 1185, 1591, 1976, 2600, 3676, 5317, 7398, 9914, 12941, 16169,
- 19477, 22885, 26464, 29851, 33360, 37228, 41139, 44802, 48654, 52058,
- 55181, 57676, 59581, 61022, 62190, 63107, 63676, 64199, 64547, 64924,
- 65158, 65313, 65430, 65481, 65518, 65535, 0, 65535, 0, 65535,
- 0, 65535, 0, 65535, 0, 65533, 65535, 0, 65535, 0,
- 65535, 0, 65535, 0, 65533, 65535, 0, 2, 65535, 0,
- 2, 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533,
- 65535, 0, 2, 4, 65533, 65535, 0, 2, 65533, 65535,
- 0, 2, 4, 65531, 65533, 65535, 0, 2, 4, 65531,
- 65533, 65535, 0, 2, 4, 6, 65524, 65533, 65535, 0,
- 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
- 65535, 0, 65535, 0, 65535, 0, 65533, 65535, 0, 65533,
- 65535, 0, 2, 65533, 65535, 0, 2, 65533, 65535, 0,
- 2, 65533, 65535, 0, 2, 4, 65532, 65535, 0, 6,
- 65523, 65535, 0, 2, 15, 65530, 65533, 65535, 0, 2,
- 35, 65493, 65531, 65533, 65535, 0, 2, 4, 158, 65382,
- 65531, 65533, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
- 65535, 0, 65535, 0, 65535, 0, 2, 65535, 0, 2,
- 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533, 65535,
- 0, 2, 65533, 65535, 0, 9, 65512, 65535, 0, 2,
- 12, 65529, 65535, 0, 2, 73, 65434, 65533, 65535, 0,
- 2, 240, 65343, 65533, 65535, 0, 2, 476, 65017, 65531,
- 65533, 65535, 0, 2, 4, 1046, 64686, 65531, 65533, 65535,
- 0, 2, 4, 6, 8, 1870, 63898, 65529, 65531, 65533,
- 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65533, 65535,
- 0, 2, 65533, 65535, 0, 2, 65533, 65535, 0, 2,
- 65532, 65535, 0, 6, 65533, 65535, 0, 6, 65523, 65535,
- 0, 2, 65532, 65535, 0, 137, 65439, 65535, 0, 576,
- 64899, 65533, 65535, 0, 2, 289, 65299, 65533, 65535, 0,
- 2, 4, 6, 880, 64134, 65531, 65533, 65535, 0, 2,
- 4, 1853, 63347, 65533, 65535, 0, 2, 6, 2516, 61762,
- 65529, 65531, 65533, 65535, 0, 2, 4, 9, 3980, 61380,
- 65503, 65529, 65531, 65533, 65535, 0, 2, 4, 6, 8,
- 10, 12, 61, 6393, 59859, 65466, 65527, 65529, 65531, 65533,
- 65535, 0, 65535, 0, 65535, 0, 65535, 0, 2, 65532,
- 65535, 0, 3, 65529, 65535, 0, 2, 65529, 65535, 0,
- 61, 65453, 65535, 0, 234, 65313, 65535, 0, 503, 65138,
- 65535, 0, 155, 65402, 65533, 65535, 0, 2, 1058, 64554,
- 65533, 65535, 0, 2, 4, 3138, 62109, 65531, 65533, 65535,
- 0, 2, 4, 2031, 63339, 65531, 65533, 65535, 0, 2,
- 4, 6, 9, 4155, 60778, 65523, 65529, 65531, 65533, 65535,
- 0, 2, 4, 41, 6189, 59269, 65490, 65531, 65533, 65535,
- 0, 2, 4, 6, 210, 8789, 57043, 65400, 65528, 65531,
- 65533, 65535, 0, 2, 4, 6, 8, 26, 453, 10086,
- 55499, 64948, 65483, 65524, 65527, 65529, 65531, 65533, 65535, 0,
- 2, 4, 6, 8, 10, 12, 14, 16, 18, 20,
- 114, 1014, 11202, 52670, 64226, 65356, 65503, 65514, 65523, 65525,
- 65527, 65529, 65531, 65533, 65535, 0, 65533, 65535, 0, 15,
- 65301, 65535, 0, 152, 64807, 65535, 0, 2, 3328, 63308,
- 65535, 0, 2, 4050, 59730, 65533, 65535, 0, 2, 164,
- 10564, 61894, 65529, 65535, 0, 15, 6712, 59831, 65076, 65532,
- 65535, 0, 32, 7712, 57449, 65459, 65535, 0, 2, 210,
- 7849, 53110, 65021, 65523, 65535, 0, 2, 12, 1081, 13883,
- 48262, 62870, 65477, 65535, 0, 2, 88, 847, 6145, 37852,
- 62012, 65454, 65533, 65535, 0, 9, 47, 207, 1823, 14522,
- 45521, 61069, 64891, 65481, 65528, 65531, 65533, 65535, 0, 2,
- 9, 488, 2881, 12758, 38703, 58412, 64420, 65410, 65533, 65535,
- 0, 2, 4, 6, 61, 333, 1891, 6486, 19720, 43188,
- 57547, 62472, 64796, 65421, 65497, 65523, 65529, 65531, 65533, 65535,
- 0, 2, 4, 6, 8, 10, 12, 29, 117, 447,
- 1528, 6138, 21242, 43133, 56495, 62432, 64746, 65362, 65500, 65529,
- 65531, 65533, 65535, 0, 2, 18, 105, 301, 760, 1490,
- 3472, 7568, 15002, 26424, 40330, 53029, 60048, 62964, 64274, 64890,
- 65337, 65445, 65489, 65513, 65527, 65530, 65533, 65535, 0, 2,
- 4, 6, 41, 102, 409, 853, 2031, 4316, 7302, 11328,
- 16869, 24825, 34926, 43481, 50877, 56126, 59874, 62103, 63281, 63857,
- 64166, 64675, 65382, 65522, 65531, 65533, 65535, 0, 2, 4,
- 6, 8, 10, 12, 14, 16, 18, 29, 38, 53,
- 58, 96, 181, 503, 1183, 2849, 5590, 8600, 11379, 13942,
- 16478, 19453, 22638, 26039, 29411, 32921, 37596, 41433, 44998, 48560,
- 51979, 55106, 57666, 59892, 61485, 62616, 63484, 64018, 64375, 64685,
- 64924, 65076, 65278, 65395, 65471, 65509, 65529, 65535, 0, 65535,
- 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535,
- 0, 65535, 0, 65535, 0, 2, 65533, 65535, 0, 2,
- 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533, 65535,
- 0, 2, 65533, 65535, 0, 2, 65533, 65535, 0, 7,
- 65519, 65535, 0, 2, 14, 65491, 65533, 65535, 0, 2,
- 81, 65427, 65531, 65533, 65535, 0, 2, 4, 312, 65293,
- 65528, 65533, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
- 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
- 2, 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533,
- 65535, 0, 5, 65523, 65535, 0, 2, 65533, 65535, 0,
- 7, 65526, 65535, 0, 46, 65464, 65533, 65535, 0, 2,
- 120, 65309, 65533, 65535, 0, 2, 5, 362, 65097, 65533,
- 65535, 0, 2, 18, 1164, 64785, 65528, 65531, 65533, 65535,
- 0, 65535, 0, 65535, 0, 65535, 0, 65533, 65535, 0,
- 65535, 0, 65533, 65535, 0, 2, 65533, 65535, 0, 2,
- 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65530, 65535,
- 0, 2, 65523, 65535, 0, 69, 65477, 65535, 0, 141,
- 65459, 65535, 0, 194, 65325, 65533, 65535, 0, 2, 543,
- 64912, 65533, 65535, 0, 5, 1270, 64301, 65529, 65531, 65533,
- 65535, 0, 2, 4, 12, 2055, 63538, 65508, 65531, 65533,
- 65535, 0, 2, 7, 102, 3775, 61970, 65429, 65526, 65528,
- 65533, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 2,
- 65533, 65535, 0, 2, 65535, 0, 9, 65533, 65535, 0,
- 25, 65512, 65535, 0, 2, 65533, 65535, 0, 44, 65480,
- 65535, 0, 48, 65475, 65535, 0, 162, 65373, 65535, 0,
- 637, 64806, 65533, 65535, 0, 2, 935, 64445, 65533, 65535,
- 0, 2, 4, 1662, 64083, 65533, 65535, 0, 2, 12,
- 3036, 62469, 65521, 65533, 65535, 0, 2, 120, 5405, 60468,
- 65469, 65531, 65533, 65535, 0, 2, 4, 18, 254, 6663,
- 58999, 65272, 65528, 65533, 65535, 0, 2, 4, 9, 12,
- 67, 591, 8981, 56781, 64564, 65365, 65508, 65524, 65526, 65529,
- 65531, 65533, 65535, 0, 65535, 0, 65535, 0, 2, 65533,
- 65535, 0, 9, 65526, 65535, 0, 14, 65503, 65535, 0,
- 127, 65390, 65535, 0, 517, 64990, 65535, 0, 178, 65330,
- 65535, 0, 2, 1055, 64533, 65533, 65535, 0, 2, 1558,
- 63942, 65533, 65535, 0, 2, 2205, 63173, 65533, 65535, 0,
- 25, 4493, 60862, 65505, 65533, 65535, 0, 2, 48, 5890,
- 59442, 65482, 65533, 65535, 0, 2, 4, 127, 7532, 58191,
- 65394, 65533, 65535, 0, 2, 5, 32, 550, 10388, 54924,
- 65046, 65510, 65531, 65533, 65535, 0, 2, 4, 30, 150,
- 1685, 14340, 51375, 63619, 65288, 65503, 65528, 65533, 65535, 0,
- 2, 4, 6, 8, 28, 97, 473, 2692, 15407, 50020,
- 62880, 65064, 65445, 65508, 65531, 65533, 65535, 0, 2, 4,
- 12, 32, 79, 150, 372, 907, 2184, 5868, 18207, 45431,
- 59856, 64031, 65096, 65401, 65481, 65507, 65521, 65523, 65525, 65527,
- 65529, 65531, 65533, 65535, 0, 65533, 65535, 0, 182, 65491,
- 65535, 0, 877, 64286, 65535, 0, 9, 2708, 63612, 65533,
- 65535, 0, 2, 6038, 59532, 65535, 0, 2, 92, 5500,
- 60539, 65533, 65535, 0, 268, 8908, 56512, 65385, 65535, 0,
- 129, 13110, 52742, 65036, 65535, 0, 2, 806, 14003, 51929,
- 64732, 65523, 65535, 0, 7, 92, 2667, 18159, 47678, 62610,
- 65355, 65535, 0, 32, 1836, 19676, 48237, 61677, 64960, 65526,
- 65535, 0, 21, 159, 967, 5668, 22782, 44709, 58317, 64020,
- 65406, 65528, 65535, 0, 7, 162, 1838, 8328, 23929, 43014,
- 56394, 63374, 65216, 65484, 65521, 65535, 0, 2, 4, 6,
- 28, 268, 1120, 3613, 10688, 24185, 40989, 54917, 61684, 64510,
- 65403, 65530, 65535, 0, 2, 16, 44, 139, 492, 1739,
- 5313, 13558, 26766, 41566, 52446, 58937, 62815, 64480, 65201, 65454,
- 65524, 65533, 65535, 0, 7, 25, 76, 263, 612, 1466,
- 3325, 6832, 12366, 20152, 29466, 39255, 47360, 53506, 57740, 60726,
- 62845, 64131, 64882, 65260, 65459, 65521, 65528, 65530, 65535, 0,
- 2, 4, 14, 48, 136, 312, 653, 1240, 2369, 4327,
- 7028, 10759, 15449, 21235, 28027, 35386, 42938, 49562, 54990, 59119,
- 62086, 63916, 64863, 65249, 65445, 65493, 65523, 65535, 0, 2,
- 4, 6, 8, 10, 12, 21, 83, 208, 409, 723,
- 1152, 1868, 2951, 4463, 6460, 8979, 11831, 15195, 18863, 22657,
- 26762, 30881, 34963, 39098, 43054, 47069, 50620, 53871, 56821, 59386,
- 61340, 62670, 63512, 64023, 64429, 64750, 64944, 65126, 65279, 65366,
- 65413, 65445, 65473, 65505, 65510, 65521, 65528, 65530, 65535
-};
-
-/* pointers to cdf tables for quantizer indices */
-const WebRtc_UWord16 *WebRtcIsacfix_kCdfGainPtr[3][12] = {
- { WebRtcIsacfix_kCdfGain +0 +0, WebRtcIsacfix_kCdfGain +0 +8, WebRtcIsacfix_kCdfGain +0 +22,
- WebRtcIsacfix_kCdfGain +0 +32, WebRtcIsacfix_kCdfGain +0 +48, WebRtcIsacfix_kCdfGain +0 +60,
- WebRtcIsacfix_kCdfGain +0 +81, WebRtcIsacfix_kCdfGain +0 +95, WebRtcIsacfix_kCdfGain +0 +128,
- WebRtcIsacfix_kCdfGain +0 +152, WebRtcIsacfix_kCdfGain +0 +210, WebRtcIsacfix_kCdfGain +0 +264
- },
- { WebRtcIsacfix_kCdfGain +404 +0, WebRtcIsacfix_kCdfGain +404 +8, WebRtcIsacfix_kCdfGain +404 +21,
- WebRtcIsacfix_kCdfGain +404 +30, WebRtcIsacfix_kCdfGain +404 +46, WebRtcIsacfix_kCdfGain +404 +58,
- WebRtcIsacfix_kCdfGain +404 +79, WebRtcIsacfix_kCdfGain +404 +93, WebRtcIsacfix_kCdfGain +404 +125,
- WebRtcIsacfix_kCdfGain +404 +149, WebRtcIsacfix_kCdfGain +404 +207, WebRtcIsacfix_kCdfGain +404 +260
- },
- { WebRtcIsacfix_kCdfGain +803 +0, WebRtcIsacfix_kCdfGain +803 +8, WebRtcIsacfix_kCdfGain +803 +22,
- WebRtcIsacfix_kCdfGain +803 +31, WebRtcIsacfix_kCdfGain +803 +48, WebRtcIsacfix_kCdfGain +803 +60,
- WebRtcIsacfix_kCdfGain +803 +81, WebRtcIsacfix_kCdfGain +803 +96, WebRtcIsacfix_kCdfGain +803 +129,
- WebRtcIsacfix_kCdfGain +803 +154, WebRtcIsacfix_kCdfGain +803 +212, WebRtcIsacfix_kCdfGain +803 +268
- }
-};
-
-const WebRtc_UWord16 *WebRtcIsacfix_kCdfShapePtr[3][108] = {
- { WebRtcIsacfix_kCdfShape +0 +0, WebRtcIsacfix_kCdfShape +0 +2, WebRtcIsacfix_kCdfShape +0 +4,
- WebRtcIsacfix_kCdfShape +0 +6, WebRtcIsacfix_kCdfShape +0 +8, WebRtcIsacfix_kCdfShape +0 +10,
- WebRtcIsacfix_kCdfShape +0 +12, WebRtcIsacfix_kCdfShape +0 +14, WebRtcIsacfix_kCdfShape +0 +16,
- WebRtcIsacfix_kCdfShape +0 +18, WebRtcIsacfix_kCdfShape +0 +21, WebRtcIsacfix_kCdfShape +0 +25,
- WebRtcIsacfix_kCdfShape +0 +29, WebRtcIsacfix_kCdfShape +0 +33, WebRtcIsacfix_kCdfShape +0 +37,
- WebRtcIsacfix_kCdfShape +0 +43, WebRtcIsacfix_kCdfShape +0 +49, WebRtcIsacfix_kCdfShape +0 +56,
- WebRtcIsacfix_kCdfShape +0 +64, WebRtcIsacfix_kCdfShape +0 +66, WebRtcIsacfix_kCdfShape +0 +68,
- WebRtcIsacfix_kCdfShape +0 +70, WebRtcIsacfix_kCdfShape +0 +72, WebRtcIsacfix_kCdfShape +0 +75,
- WebRtcIsacfix_kCdfShape +0 +77, WebRtcIsacfix_kCdfShape +0 +79, WebRtcIsacfix_kCdfShape +0 +81,
- WebRtcIsacfix_kCdfShape +0 +83, WebRtcIsacfix_kCdfShape +0 +86, WebRtcIsacfix_kCdfShape +0 +90,
- WebRtcIsacfix_kCdfShape +0 +94, WebRtcIsacfix_kCdfShape +0 +98, WebRtcIsacfix_kCdfShape +0 +102,
- WebRtcIsacfix_kCdfShape +0 +107, WebRtcIsacfix_kCdfShape +0 +113, WebRtcIsacfix_kCdfShape +0 +120,
- WebRtcIsacfix_kCdfShape +0 +129, WebRtcIsacfix_kCdfShape +0 +131, WebRtcIsacfix_kCdfShape +0 +133,
- WebRtcIsacfix_kCdfShape +0 +135, WebRtcIsacfix_kCdfShape +0 +137, WebRtcIsacfix_kCdfShape +0 +141,
- WebRtcIsacfix_kCdfShape +0 +143, WebRtcIsacfix_kCdfShape +0 +147, WebRtcIsacfix_kCdfShape +0 +151,
- WebRtcIsacfix_kCdfShape +0 +155, WebRtcIsacfix_kCdfShape +0 +159, WebRtcIsacfix_kCdfShape +0 +164,
- WebRtcIsacfix_kCdfShape +0 +168, WebRtcIsacfix_kCdfShape +0 +172, WebRtcIsacfix_kCdfShape +0 +178,
- WebRtcIsacfix_kCdfShape +0 +184, WebRtcIsacfix_kCdfShape +0 +192, WebRtcIsacfix_kCdfShape +0 +200,
- WebRtcIsacfix_kCdfShape +0 +211, WebRtcIsacfix_kCdfShape +0 +213, WebRtcIsacfix_kCdfShape +0 +215,
- WebRtcIsacfix_kCdfShape +0 +217, WebRtcIsacfix_kCdfShape +0 +219, WebRtcIsacfix_kCdfShape +0 +223,
- WebRtcIsacfix_kCdfShape +0 +227, WebRtcIsacfix_kCdfShape +0 +231, WebRtcIsacfix_kCdfShape +0 +235,
- WebRtcIsacfix_kCdfShape +0 +239, WebRtcIsacfix_kCdfShape +0 +243, WebRtcIsacfix_kCdfShape +0 +248,
- WebRtcIsacfix_kCdfShape +0 +252, WebRtcIsacfix_kCdfShape +0 +258, WebRtcIsacfix_kCdfShape +0 +264,
- WebRtcIsacfix_kCdfShape +0 +273, WebRtcIsacfix_kCdfShape +0 +282, WebRtcIsacfix_kCdfShape +0 +293,
- WebRtcIsacfix_kCdfShape +0 +308, WebRtcIsacfix_kCdfShape +0 +310, WebRtcIsacfix_kCdfShape +0 +312,
- WebRtcIsacfix_kCdfShape +0 +316, WebRtcIsacfix_kCdfShape +0 +320, WebRtcIsacfix_kCdfShape +0 +324,
- WebRtcIsacfix_kCdfShape +0 +328, WebRtcIsacfix_kCdfShape +0 +332, WebRtcIsacfix_kCdfShape +0 +336,
- WebRtcIsacfix_kCdfShape +0 +341, WebRtcIsacfix_kCdfShape +0 +347, WebRtcIsacfix_kCdfShape +0 +354,
- WebRtcIsacfix_kCdfShape +0 +360, WebRtcIsacfix_kCdfShape +0 +368, WebRtcIsacfix_kCdfShape +0 +378,
- WebRtcIsacfix_kCdfShape +0 +388, WebRtcIsacfix_kCdfShape +0 +400, WebRtcIsacfix_kCdfShape +0 +418,
- WebRtcIsacfix_kCdfShape +0 +445, WebRtcIsacfix_kCdfShape +0 +447, WebRtcIsacfix_kCdfShape +0 +451,
- WebRtcIsacfix_kCdfShape +0 +455, WebRtcIsacfix_kCdfShape +0 +461, WebRtcIsacfix_kCdfShape +0 +468,
- WebRtcIsacfix_kCdfShape +0 +474, WebRtcIsacfix_kCdfShape +0 +480, WebRtcIsacfix_kCdfShape +0 +486,
- WebRtcIsacfix_kCdfShape +0 +495, WebRtcIsacfix_kCdfShape +0 +505, WebRtcIsacfix_kCdfShape +0 +516,
- WebRtcIsacfix_kCdfShape +0 +528, WebRtcIsacfix_kCdfShape +0 +543, WebRtcIsacfix_kCdfShape +0 +564,
- WebRtcIsacfix_kCdfShape +0 +583, WebRtcIsacfix_kCdfShape +0 +608, WebRtcIsacfix_kCdfShape +0 +635
- },
- { WebRtcIsacfix_kCdfShape +686 +0, WebRtcIsacfix_kCdfShape +686 +2, WebRtcIsacfix_kCdfShape +686 +4,
- WebRtcIsacfix_kCdfShape +686 +6, WebRtcIsacfix_kCdfShape +686 +8, WebRtcIsacfix_kCdfShape +686 +11,
- WebRtcIsacfix_kCdfShape +686 +13, WebRtcIsacfix_kCdfShape +686 +15, WebRtcIsacfix_kCdfShape +686 +17,
- WebRtcIsacfix_kCdfShape +686 +20, WebRtcIsacfix_kCdfShape +686 +23, WebRtcIsacfix_kCdfShape +686 +27,
- WebRtcIsacfix_kCdfShape +686 +31, WebRtcIsacfix_kCdfShape +686 +35, WebRtcIsacfix_kCdfShape +686 +40,
- WebRtcIsacfix_kCdfShape +686 +44, WebRtcIsacfix_kCdfShape +686 +50, WebRtcIsacfix_kCdfShape +686 +56,
- WebRtcIsacfix_kCdfShape +686 +63, WebRtcIsacfix_kCdfShape +686 +65, WebRtcIsacfix_kCdfShape +686 +67,
- WebRtcIsacfix_kCdfShape +686 +69, WebRtcIsacfix_kCdfShape +686 +71, WebRtcIsacfix_kCdfShape +686 +73,
- WebRtcIsacfix_kCdfShape +686 +75, WebRtcIsacfix_kCdfShape +686 +77, WebRtcIsacfix_kCdfShape +686 +79,
- WebRtcIsacfix_kCdfShape +686 +82, WebRtcIsacfix_kCdfShape +686 +85, WebRtcIsacfix_kCdfShape +686 +89,
- WebRtcIsacfix_kCdfShape +686 +93, WebRtcIsacfix_kCdfShape +686 +97, WebRtcIsacfix_kCdfShape +686 +102,
- WebRtcIsacfix_kCdfShape +686 +106, WebRtcIsacfix_kCdfShape +686 +112, WebRtcIsacfix_kCdfShape +686 +119,
- WebRtcIsacfix_kCdfShape +686 +127, WebRtcIsacfix_kCdfShape +686 +129, WebRtcIsacfix_kCdfShape +686 +131,
- WebRtcIsacfix_kCdfShape +686 +133, WebRtcIsacfix_kCdfShape +686 +135, WebRtcIsacfix_kCdfShape +686 +137,
- WebRtcIsacfix_kCdfShape +686 +139, WebRtcIsacfix_kCdfShape +686 +142, WebRtcIsacfix_kCdfShape +686 +146,
- WebRtcIsacfix_kCdfShape +686 +150, WebRtcIsacfix_kCdfShape +686 +154, WebRtcIsacfix_kCdfShape +686 +158,
- WebRtcIsacfix_kCdfShape +686 +162, WebRtcIsacfix_kCdfShape +686 +167, WebRtcIsacfix_kCdfShape +686 +173,
- WebRtcIsacfix_kCdfShape +686 +179, WebRtcIsacfix_kCdfShape +686 +186, WebRtcIsacfix_kCdfShape +686 +194,
- WebRtcIsacfix_kCdfShape +686 +205, WebRtcIsacfix_kCdfShape +686 +207, WebRtcIsacfix_kCdfShape +686 +209,
- WebRtcIsacfix_kCdfShape +686 +211, WebRtcIsacfix_kCdfShape +686 +214, WebRtcIsacfix_kCdfShape +686 +218,
- WebRtcIsacfix_kCdfShape +686 +222, WebRtcIsacfix_kCdfShape +686 +226, WebRtcIsacfix_kCdfShape +686 +230,
- WebRtcIsacfix_kCdfShape +686 +234, WebRtcIsacfix_kCdfShape +686 +238, WebRtcIsacfix_kCdfShape +686 +242,
- WebRtcIsacfix_kCdfShape +686 +247, WebRtcIsacfix_kCdfShape +686 +253, WebRtcIsacfix_kCdfShape +686 +262,
- WebRtcIsacfix_kCdfShape +686 +269, WebRtcIsacfix_kCdfShape +686 +278, WebRtcIsacfix_kCdfShape +686 +289,
- WebRtcIsacfix_kCdfShape +686 +305, WebRtcIsacfix_kCdfShape +686 +307, WebRtcIsacfix_kCdfShape +686 +309,
- WebRtcIsacfix_kCdfShape +686 +311, WebRtcIsacfix_kCdfShape +686 +315, WebRtcIsacfix_kCdfShape +686 +319,
- WebRtcIsacfix_kCdfShape +686 +323, WebRtcIsacfix_kCdfShape +686 +327, WebRtcIsacfix_kCdfShape +686 +331,
- WebRtcIsacfix_kCdfShape +686 +335, WebRtcIsacfix_kCdfShape +686 +340, WebRtcIsacfix_kCdfShape +686 +346,
- WebRtcIsacfix_kCdfShape +686 +354, WebRtcIsacfix_kCdfShape +686 +362, WebRtcIsacfix_kCdfShape +686 +374,
- WebRtcIsacfix_kCdfShape +686 +384, WebRtcIsacfix_kCdfShape +686 +396, WebRtcIsacfix_kCdfShape +686 +413,
- WebRtcIsacfix_kCdfShape +686 +439, WebRtcIsacfix_kCdfShape +686 +442, WebRtcIsacfix_kCdfShape +686 +446,
- WebRtcIsacfix_kCdfShape +686 +450, WebRtcIsacfix_kCdfShape +686 +455, WebRtcIsacfix_kCdfShape +686 +461,
- WebRtcIsacfix_kCdfShape +686 +468, WebRtcIsacfix_kCdfShape +686 +475, WebRtcIsacfix_kCdfShape +686 +481,
- WebRtcIsacfix_kCdfShape +686 +489, WebRtcIsacfix_kCdfShape +686 +498, WebRtcIsacfix_kCdfShape +686 +508,
- WebRtcIsacfix_kCdfShape +686 +522, WebRtcIsacfix_kCdfShape +686 +534, WebRtcIsacfix_kCdfShape +686 +554,
- WebRtcIsacfix_kCdfShape +686 +577, WebRtcIsacfix_kCdfShape +686 +602, WebRtcIsacfix_kCdfShape +686 +631
- },
- { WebRtcIsacfix_kCdfShape +1368 +0, WebRtcIsacfix_kCdfShape +1368 +2, WebRtcIsacfix_kCdfShape +1368 +4,
- WebRtcIsacfix_kCdfShape +1368 +6, WebRtcIsacfix_kCdfShape +1368 +8, WebRtcIsacfix_kCdfShape +1368 +10,
- WebRtcIsacfix_kCdfShape +1368 +12, WebRtcIsacfix_kCdfShape +1368 +14, WebRtcIsacfix_kCdfShape +1368 +16,
- WebRtcIsacfix_kCdfShape +1368 +20, WebRtcIsacfix_kCdfShape +1368 +24, WebRtcIsacfix_kCdfShape +1368 +28,
- WebRtcIsacfix_kCdfShape +1368 +32, WebRtcIsacfix_kCdfShape +1368 +36, WebRtcIsacfix_kCdfShape +1368 +40,
- WebRtcIsacfix_kCdfShape +1368 +44, WebRtcIsacfix_kCdfShape +1368 +50, WebRtcIsacfix_kCdfShape +1368 +57,
- WebRtcIsacfix_kCdfShape +1368 +65, WebRtcIsacfix_kCdfShape +1368 +67, WebRtcIsacfix_kCdfShape +1368 +69,
- WebRtcIsacfix_kCdfShape +1368 +71, WebRtcIsacfix_kCdfShape +1368 +73, WebRtcIsacfix_kCdfShape +1368 +75,
- WebRtcIsacfix_kCdfShape +1368 +77, WebRtcIsacfix_kCdfShape +1368 +79, WebRtcIsacfix_kCdfShape +1368 +81,
- WebRtcIsacfix_kCdfShape +1368 +85, WebRtcIsacfix_kCdfShape +1368 +89, WebRtcIsacfix_kCdfShape +1368 +93,
- WebRtcIsacfix_kCdfShape +1368 +97, WebRtcIsacfix_kCdfShape +1368 +101, WebRtcIsacfix_kCdfShape +1368 +105,
- WebRtcIsacfix_kCdfShape +1368 +110, WebRtcIsacfix_kCdfShape +1368 +116, WebRtcIsacfix_kCdfShape +1368 +123,
- WebRtcIsacfix_kCdfShape +1368 +132, WebRtcIsacfix_kCdfShape +1368 +134, WebRtcIsacfix_kCdfShape +1368 +136,
- WebRtcIsacfix_kCdfShape +1368 +138, WebRtcIsacfix_kCdfShape +1368 +141, WebRtcIsacfix_kCdfShape +1368 +143,
- WebRtcIsacfix_kCdfShape +1368 +146, WebRtcIsacfix_kCdfShape +1368 +150, WebRtcIsacfix_kCdfShape +1368 +154,
- WebRtcIsacfix_kCdfShape +1368 +158, WebRtcIsacfix_kCdfShape +1368 +162, WebRtcIsacfix_kCdfShape +1368 +166,
- WebRtcIsacfix_kCdfShape +1368 +170, WebRtcIsacfix_kCdfShape +1368 +174, WebRtcIsacfix_kCdfShape +1368 +179,
- WebRtcIsacfix_kCdfShape +1368 +185, WebRtcIsacfix_kCdfShape +1368 +193, WebRtcIsacfix_kCdfShape +1368 +203,
- WebRtcIsacfix_kCdfShape +1368 +214, WebRtcIsacfix_kCdfShape +1368 +216, WebRtcIsacfix_kCdfShape +1368 +218,
- WebRtcIsacfix_kCdfShape +1368 +220, WebRtcIsacfix_kCdfShape +1368 +224, WebRtcIsacfix_kCdfShape +1368 +227,
- WebRtcIsacfix_kCdfShape +1368 +231, WebRtcIsacfix_kCdfShape +1368 +235, WebRtcIsacfix_kCdfShape +1368 +239,
- WebRtcIsacfix_kCdfShape +1368 +243, WebRtcIsacfix_kCdfShape +1368 +247, WebRtcIsacfix_kCdfShape +1368 +251,
- WebRtcIsacfix_kCdfShape +1368 +256, WebRtcIsacfix_kCdfShape +1368 +262, WebRtcIsacfix_kCdfShape +1368 +269,
- WebRtcIsacfix_kCdfShape +1368 +277, WebRtcIsacfix_kCdfShape +1368 +286, WebRtcIsacfix_kCdfShape +1368 +297,
- WebRtcIsacfix_kCdfShape +1368 +315, WebRtcIsacfix_kCdfShape +1368 +317, WebRtcIsacfix_kCdfShape +1368 +319,
- WebRtcIsacfix_kCdfShape +1368 +323, WebRtcIsacfix_kCdfShape +1368 +327, WebRtcIsacfix_kCdfShape +1368 +331,
- WebRtcIsacfix_kCdfShape +1368 +335, WebRtcIsacfix_kCdfShape +1368 +339, WebRtcIsacfix_kCdfShape +1368 +343,
- WebRtcIsacfix_kCdfShape +1368 +349, WebRtcIsacfix_kCdfShape +1368 +355, WebRtcIsacfix_kCdfShape +1368 +361,
- WebRtcIsacfix_kCdfShape +1368 +368, WebRtcIsacfix_kCdfShape +1368 +376, WebRtcIsacfix_kCdfShape +1368 +385,
- WebRtcIsacfix_kCdfShape +1368 +397, WebRtcIsacfix_kCdfShape +1368 +411, WebRtcIsacfix_kCdfShape +1368 +429,
- WebRtcIsacfix_kCdfShape +1368 +456, WebRtcIsacfix_kCdfShape +1368 +459, WebRtcIsacfix_kCdfShape +1368 +463,
- WebRtcIsacfix_kCdfShape +1368 +467, WebRtcIsacfix_kCdfShape +1368 +473, WebRtcIsacfix_kCdfShape +1368 +478,
- WebRtcIsacfix_kCdfShape +1368 +485, WebRtcIsacfix_kCdfShape +1368 +491, WebRtcIsacfix_kCdfShape +1368 +497,
- WebRtcIsacfix_kCdfShape +1368 +505, WebRtcIsacfix_kCdfShape +1368 +514, WebRtcIsacfix_kCdfShape +1368 +523,
- WebRtcIsacfix_kCdfShape +1368 +535, WebRtcIsacfix_kCdfShape +1368 +548, WebRtcIsacfix_kCdfShape +1368 +565,
- WebRtcIsacfix_kCdfShape +1368 +585, WebRtcIsacfix_kCdfShape +1368 +611, WebRtcIsacfix_kCdfShape +1368 +640
- }
-};
-
-/* code length for all coefficients using different models */
-
-const WebRtc_Word16 WebRtcIsacfix_kCodeLenGainQ11[392] = {
- 25189, 16036, 8717, 358, 8757, 15706, 21456, 24397, 18502, 17559
- , 13794, 11088, 7480, 873, 6603, 11636, 14627, 16805, 19132, 26624
- , 26624, 19408, 13751, 7280, 583, 7591, 15178, 23773, 28672, 25189
- , 19045, 16442, 13412, 10397, 5893, 1338, 6376, 9992, 12074, 13853
- , 15781, 19821, 22819, 28672, 28672, 25189, 19858, 15781, 11262, 5477
- , 1298, 5632, 11814, 17234, 22020, 28672, 19677, 18125, 16587, 14521
- , 13032, 11196, 9249, 5411, 2495, 4994, 7975, 10234, 12308, 13892
- , 15148, 17944, 21725, 23917, 25189, 19539, 16293, 11531, 7808, 4475
- , 2739, 4872, 8089, 11314, 14992, 18105, 23257, 26624, 25189, 23257
- , 23257, 20982, 18697, 18023, 16338, 16036, 14539, 13695, 13146, 11763
- , 10754, 9074, 7260, 5584, 4430, 5553, 6848, 8344, 10141, 11636
- , 12535, 13416, 14342, 15477, 17296, 19282, 22349, 23773, 28672, 28672
- , 26624, 23773, 21456, 18023, 15118, 13362, 11212, 9293, 8043, 6985
- , 5908, 5721, 5853, 6518, 7316, 8360, 9716, 11289, 12912, 14652
- , 16969, 19858, 23773, 26624, 28013, 30720, 30720, 28672, 25426, 23141
- , 25426, 23773, 20720, 19408, 18697, 19282, 16859, 16338, 16026, 15377
- , 15021, 14319, 14251, 13937, 13260, 13017, 12332, 11703, 11430, 10359
- , 10128, 9405, 8757, 8223, 7974, 7859, 7646, 7673, 7997, 8580
- , 8880, 9061, 9866, 10397, 11358, 12200, 13244, 14157, 15021, 16026
- , 16490, 18697, 18479, 20011, 19677, 20720, 24576, 26276, 30720, 30720
- , 28672, 30720, 24068, 25189, 22437, 20345, 18479, 16396, 16026, 14928
- , 13877, 13450, 12696, 12766, 11626, 11098, 10159, 9998, 9437, 9275
- , 8783, 8552, 8629, 8488, 8522, 8454, 8571, 8775, 8915, 9427
- , 9483, 9851, 10260, 10933, 11131, 11974, 12560, 13833, 15080, 16304
- , 17491, 19017, 18697, 19408, 22020, 25189, 25426, 22819, 26276, 30720
- , 30720, 30720, 30720, 30720, 30720, 28672, 30720, 30720, 30720, 30720
- , 28013, 25426, 24397, 23773, 25189, 26624, 25189, 22437, 21725, 20011
- , 20527, 20720, 20771, 22020, 22020, 19858, 19408, 19972, 17866, 17360
- , 17791, 17219, 16805, 16927, 16067, 16162, 15661, 15178, 15021, 15209
- , 14845, 14570, 14490, 14490, 13733, 13617, 13794, 13577, 13312, 12824
- , 13032, 12683, 12189, 12469, 12109, 11940, 11636, 11617, 11932, 12294
- , 11578, 11775, 12039, 11654, 11560, 11439, 11909, 11421, 12029, 11513
- , 11773, 11899, 11560, 11805, 11476, 11664, 11963, 11647, 11754, 11963
- , 11703, 12211, 11932, 12074, 12469, 12535, 12560, 12912, 12783, 12866
- , 12884, 13378, 13957, 13775, 13635, 14019, 14545, 15240, 15520, 15554
- , 15697, 16490, 16396, 17281, 16599, 16969, 17963, 16859, 16983, 16805
- , 17099, 18210, 17219, 17646, 17700, 17646, 18297, 17425, 18479, 17791
- , 17718, 19282, 18672, 20173, 20982, 21725, 21456, 23773, 23257, 25189
- , 30720, 30720, 25189, 26624, 30720, 30720, 30720, 30720, 28672, 26276
- , 30720, 30720
-};
-
-const WebRtc_Word16 WebRtcIsacfix_kCodeLenShapeQ11[577] = {
- 0, 0, 0, 0, 0, 0, 0, 0, 0, 28672
- , 0, 26624, 1, 23773, 22819, 4, 20982, 18598, 10, 19282
- , 16587, 22, 16442, 26624, 13126, 60, 14245, 26624, 26624, 12736
- , 79, 12912, 25189, 22819, 9563, 249, 9474, 22349, 28672, 23257
- , 17944, 7980, 434, 8181, 16431, 26624, 0, 0, 0, 0
- , 28672, 0, 0, 0, 0, 0, 28672, 0, 22437, 3
- , 22437, 20982, 5, 20982, 16442, 22, 16752, 13814, 49, 14646
- , 11645, 116, 11734, 26624, 28672, 10613, 158, 11010, 24397, 19539
- , 8046, 453, 7709, 19017, 28672, 23257, 15110, 6770, 758, 6523
- , 14108, 24397, 28672, 0, 0, 0, 0, 28672, 0, 28672
- , 0, 26624, 1, 28672, 28672, 1, 26624, 24397, 2, 23257
- , 21725, 4, 20982, 17158, 18, 17281, 28672, 15178, 35, 15209
- , 12343, 92, 12320, 26624, 10344, 189, 10217, 30720, 22020, 9033
- , 322, 8549, 23773, 28672, 30720, 20622, 7666, 473, 7806, 20527
- , 24397, 14135, 5995, 960, 6018, 14872, 23773, 26624, 20928, 16293
- , 10636, 4926, 1588, 5256, 11088, 18043, 25189, 0, 0, 0
- , 0, 24397, 1, 25189, 20720, 5, 21456, 21209, 3, 25189
- , 20982, 5, 21456, 15818, 30, 15410, 13794, 60, 13416, 28672
- , 11162, 142, 11025, 9337, 231, 10094, 23773, 8338, 405, 7930
- , 26624, 19677, 6787, 613, 7318, 19161, 28672, 16442, 6319, 932
- , 5748, 15312, 25189, 28672, 28672, 28672, 13998, 5513, 1263, 5146
- , 14024, 24397, 22819, 15818, 9460, 4447, 2122, 4681, 9970, 15945
- , 22349, 28672, 30720, 22622, 19017, 14872, 10689, 7405, 4473, 2983
- , 4783, 7894, 11186, 14964, 18210, 24397, 0, 0, 30720, 0
- , 30720, 21456, 3, 23773, 14964, 39, 14757, 14179, 53, 13751
- , 14928, 36, 15272, 12430, 79, 13228, 9135, 285, 9077, 28672
- , 28672, 8377, 403, 7919, 26624, 28672, 23257, 7068, 560, 7473
- , 20345, 19677, 6770, 720, 6464, 18697, 25189, 16249, 5779, 1087
- , 5494, 15209, 22819, 30720, 20622, 12601, 5240, 1419, 5091, 12095
- , 19408, 26624, 22819, 16805, 10683, 4812, 2056, 4293, 9836, 16026
- , 24397, 25189, 18409, 13833, 8681, 4503, 2653, 4220, 8329, 13853
- , 19132, 26624, 25189, 20771, 17219, 12630, 9520, 6733, 4565, 3657
- , 4817, 7069, 10058, 13212, 16805, 21209, 26624, 26276, 28672, 28672
- , 26276, 23257, 20173, 19282, 16538, 15051, 12811, 10754, 9267, 7547
- , 6270, 5407, 5214, 6057, 7054, 8226, 9488, 10806, 12793, 14442
- , 16442, 19677, 22099, 26276, 28672, 0, 30720, 0, 30720, 11920
- , 56, 20720, 30720, 6766, 355, 13130, 30720, 30720, 22180, 5589
- , 736, 7902, 26624, 30720, 7634, 354, 9721, 30720, 30720, 9027
- , 246, 10117, 30720, 30720, 9630, 453, 6709, 23257, 30720, 25683
- , 14228, 6127, 1271, 4615, 15178, 30720, 30720, 23504, 12382, 5739
- , 2015, 3492, 10560, 22020, 26624, 30720, 30720, 23257, 13192, 4873
- , 1527, 5001, 12445, 22020, 30720, 30720, 30720, 30720, 19344, 10761
- , 4051, 1927, 5281, 10594, 17866, 28672, 30720, 30720, 30720, 21869
- , 15554, 10060, 5979, 2710, 3085, 7889, 14646, 21725, 28672, 30720
- , 30720, 30720, 30720, 30720, 30720, 30720, 22719, 17425, 13212, 8083
- , 4439, 2820, 4305, 8136, 12988, 17425, 21151, 28672, 28672, 30720
- , 30720, 30720, 28672, 20527, 19282, 14412, 10513, 7407, 5079, 3744
- , 4115, 6308, 9621, 13599, 17040, 22349, 28672, 30720, 30720, 30720
- , 30720, 30720, 30720, 29522, 19282, 14545, 11485, 9093, 6760, 5262
- , 4672, 4970, 6005, 7852, 9732, 12343, 14672, 19161, 22819, 25189
- , 30720, 30720, 28672, 30720, 30720, 20720, 18125, 14388, 12007, 9825
- , 8092, 7064, 6069, 5903, 5932, 6359, 7169, 8310, 9324, 10711
- , 11867, 13096, 14157, 16338, 17040, 19161, 21725, 23773, 30720, 30720
- , 26276, 25426, 24397, 28672, 28672, 23257, 22020, 22349, 18297, 17646
- , 16983, 16431, 16162, 15021, 15178, 13751, 12142, 10895, 10193, 9632
- , 9086, 8896, 8823, 8735, 8591, 8754, 8649, 8361, 8329, 8522
- , 8373, 8739, 8993, 9657, 10454, 11279, 11899, 12614, 14024, 14273
- , 15477, 15240, 16649, 17866, 18697, 21151, 22099
-};
-
-/* left KLT transforms */
-const WebRtc_Word16 WebRtcIsacfix_kT1GainQ15[3][4] = {
- { -26130, 19773, 19773, 26130 },
- { -26664, 19046, 19046, 26664 },
- { -23538, 22797, 22797, 23538 }
-};
-
-
-
-const WebRtc_Word16 WebRtcIsacfix_kT1ShapeQ15[3][324] = {
- { 52,16,168,7,439,-138,-89,306,671,882,
- 157,1301,291,1598,-3571,-1943,-1119,32404,96,-12,
- 379,-64,-307,345,-836,539,1045,2541,-2865,-992,
- 1683,-4717,5808,7427,30599,2319,183,-73,451,481,
- 933,-198,781,-397,1244,-777,3690,-2414,149,-1356,
- -2593,-31140,8289,-1737,-202,-14,-214,360,501,450,
- -245,-7,797,3638,-2804,3042,-337,22137,-22103,2264,
- 6838,-3381,305,172,263,-195,-355,351,179,513,
- 2234,3343,5509,7531,19075,-17740,-16836,2244,-629,-1505,
- -153,108,124,-324,2694,-124,1492,-850,5347,4285,
- 7439,-10229,-22822,-12467,-12891,3645,822,-232,131,13,
- 374,565,536,4681,1294,-1935,1926,-5734,-10643,26462,
- -12480,-5589,-1038,-2468,964,-704,-247,-106,186,-558,
- -4050,3760,2972,2141,-7393,6294,26740,11991,-3251,5461,
- 5341,1574,2208,-51,-552,-297,-753,-154,2068,-5371,
- 3578,4106,28043,-10533,8041,2353,2389,4609,3410,1906,
- 351,-249,18,-15,1117,539,2870,9084,17585,-24528,
- -366,-6490,2009,-3170,2942,1116,-232,1672,1065,606,
- -399,-388,-518,38,3728,28948,-11936,4543,4104,-4441,
- 1545,-4044,1485,622,-68,186,-473,135,-280,125,
- -546,-1813,6989,6606,23711,19376,-2636,2870,-4553,-1687,
- 878,-375,205,-208,-409,-108,-200,-45,-1670,-337,
- 8213,-5524,-2334,5240,-12939,-26205,5937,-1582,-592,-959,
- -5374,2449,3400,559,349,-492,668,12379,-27684,3419,
- 5117,4415,-297,-8270,-1252,-3490,-1272,-1199,-3159,191,
- 630,488,-797,-3071,12912,-27783,-10249,1047,647,619,
- 111,-3722,-915,-1055,-502,5,-1384,-306,221,68,
- 5219,13173,-26474,-11663,-5626,927,806,-1127,236,-589,
- -522,-230,-312,-315,-428,-573,426,192,-11830,-26883,
- -14121,-2785,-1429,-109,410,-832,-302,539,-459,104,
- 1,-530,-202,-289,153,116,30082,-12944,-671,20,
- 649,98,103,215,234,0,280,-51,-169,298,
- 31,230,-73,-51
- },
- { -154,-7,-192,61,-739,-389,-947,-162,-60,94,
- 511,-716,1520,-1428,4168,-2214,1816,32270,-123,-77,
- -199,-99,-42,-588,203,-240,-930,-35,1580,234,
- 3206,-5507,-1495,-10946,30000,-2667,-136,-176,-240,-175,
- -204,-661,-1796,-1039,-1271,498,3143,734,2663,2699,
- -8127,29333,10495,2356,-72,113,-91,118,-2840,-723,
- -1733,-1158,-389,-2116,-3054,-3,-5179,8071,29546,6308,
- 5657,-3178,-186,-294,-473,-635,1213,-983,-1437,-1715,
- -1094,1280,-92,-9573,948,29576,-7060,-5921,2954,1349,
- -337,-108,-1099,962,418,-413,-1149,-334,1241,3975,
- -6825,26725,-14377,7051,-4772,-1707,2335,2008,-150,570,
- 1371,42,-1649,-619,2039,3369,-1225,1583,-2755,-15207,
- -27504,-4855,-4304,1495,2733,1324,15,-448,403,353,
- 3016,-1242,2338,2673,2064,-7496,-30447,-3686,5833,-1301,
- -2455,2122,1519,608,43,-653,773,-3072,912,-1537,
- 4505,10284,30237,1549,3200,-691,205,1702,658,1014,
- 1499,148,79,-322,-1162,-4639,-813,7536,3204,29109,
- -10747,-26,1611,2286,2114,2561,1022,372,348,207,
- 1062,-1088,-443,-9849,2381,5671,29097,-7612,-2927,3853,
- 194,1155,275,1438,1438,1312,581,888,-784,906,
- 112,-11103,25104,14438,-9311,-3068,1210,368,370,-940,
- -2434,-1148,1925,392,657,258,-526,1475,-2281,-4265,
- -1880,1534,2185,-1472,959,-30934,6306,3114,-4109,1768,
- -2612,-703,45,644,2185,2033,5670,7211,19114,-22427,
- 6432,5150,-4090,-2694,3860,1245,-596,293,1829,369,
- -319,229,-3256,2170,-6374,-26216,-4570,-16053,-5766,-262,
- -2006,2873,-1477,147,378,-1544,-344,-544,-985,-481,
- 4210,4542,30757,-7291,-4863,1529,-2079,-628,-603,-783,
- -408,1646,697,808,-620,-292,181,158,-13313,-29173,
- 5984,-1262,859,-1776,-558,-24,-883,-1421,739,210,
- -531,-285,131,-160,-246,-56,29345,-13706,-2859,-2966,
- -300,-970,-2382,-268,-103,-636,-12,-62,-691,-253,
- -147,-127,27,66
- },
- { 55,-212,-198,489,-274,81,682,399,328,-934,
- -389,-37,1357,-3632,5276,6581,-9493,-29921,29,-45,
- 2,190,172,-15,311,-130,-1085,-25,324,-684,
- 3223,-6580,4485,-5280,-29521,9933,82,-320,-530,229,
- -705,-533,-414,848,-1842,-4473,1390,-857,6717,-6692,
- 4648,29397,576,8339,-68,-85,238,-330,264,-1012,
- -381,-203,-3384,-3329,3906,6810,3790,-6250,28312,-8078,
- 8089,1565,160,-569,-612,-613,-1063,-1928,-1125,3421,
- -7481,-7484,4942,-6984,4330,-25591,-10574,-6982,5682,-1781,
- -308,89,178,-1715,-420,-3530,-5776,1219,-8617,-7137,
- 7015,4981,24875,12657,-5408,-3356,-785,-1972,326,-858,
- -506,-3382,-986,-6258,-2259,4015,-8374,-10482,3127,23826,
- -14126,-514,-5417,2178,-2912,-17,-587,80,67,-5881,
- -1702,-5351,-4481,398,-10156,-225,20727,-15460,-11603,7752,
- 3660,1714,-2001,-359,499,-527,-1225,-7820,-1297,-6326,
- -8526,7900,-18328,13311,-17488,-2926,-196,-17,2281,873,
- 480,-160,-624,471,780,-8729,1707,-14262,-20647,1721,
- 18590,-2206,-1214,-1066,312,-2602,783,-412,-113,49,
- -119,1305,-2371,-15132,-1833,-18252,20295,-8316,2227,341,
- -2074,-702,3082,-262,-465,-198,430,30,-70,-788,
- 2342,-25132,-4863,19783,-484,2137,2811,-1906,799,1586,
- 962,-734,-191,-30,-129,-93,-1126,1729,5860,-2030,
- 8953,603,-3338,-10869,-1144,22070,12130,10513,3191,-6881,
- -3514,2090,711,-666,1843,-5997,-5681,2921,-17641,-2801,
- 4969,18590,7169,12214,8587,4405,3008,-1074,-371,-77,
- 253,331,-5611,5014,13152,-1985,18483,-1696,8043,20463,
- 2381,-393,1688,-1205,618,1220,457,248,-83,176,
- 7920,-13676,-22139,-3038,17402,2036,844,3258,994,719,
- 2087,-44,426,494,12,-91,46,5,-14204,22912,
- -18156,-361,442,2298,-829,2229,386,1433,1335,1323,
- 55,-592,-139,49,-12,-57,27783,17134,350,-282,
- 552,158,142,2488,465,329,1087,118,143,10,
- 56,65,-15,-31
- }
-};
-
-/* right KLT transforms */
-const WebRtc_Word16 WebRtcIsacfix_kT2GainQ15[3][36] = {
- { 4775, -14892, 20313, -17104, 10533, -3613, -6782, 16044, -8889,
- -11019, 21330, -10720, 13193, -15678, -11101, 14461, 12250, -13096,
- -16951, 2167, 16066, 15569, -702, -16754, -19195, -12823, -4321,
- 5128, 13348, 17825, 13232, 13404, 13494, 13490, 13383, 13261
- },
- { -3725, 11408, -18493, 20031, -13097, 3865, 9344, -19294, 10740,
- 8856, -18432, 8982, 13975, -14444, -11930, 11774, 14285, -13594,
- -16323, -4, 16340, 15609, 359, -17220, -18401, -13471, -4643,
- 5225, 13375, 18053, 13124, 13463, 13621, 13583, 13393, 13072
- },
- { -3513, 11402, -17883, 19504, -14399, 4885, 8702, -19513, 12046,
- 8533, -18110, 8447, 12778, -14838, -12444, 13177, 14107, -12759,
- -17268, 914, 15822, 15661, 838, -16686, -18907, -12936, -4820,
- 4175, 12398, 18830, 12913, 13215, 13433, 13572, 13601, 13518
- }
-};
-
-const WebRtc_Word16 WebRtcIsacfix_kT2ShapeQ15[3][36] = {
- { 4400, -11512, 17205, -19470, 14770, -5345, 9784, -19222, 11228,
- 6842, -18371, 9909, 14191, -13496, -11563, 14015, 11827, -14839,
- -15439, 948, 17802, 14827, -2053, -17132, 18723, 14516, 4135,
- -6822, -13869, -16016, 12975, 13341, 13563, 13603, 13478, 13296
- },
- { 5420, -14215, 19060, -18073, 11709, -3911, 9645, -18335, 7717,
- 10842, -19283, 9777, 14898, -12555, -13661, 11668, 13520, -13733,
- -15936, -1358, 15671, 16728, 328, -17100, 17527, 13973, 5587,
- -5194, -14165, -17677, 12970, 13446, 13693, 13660, 13462, 13015
- },
- { 4386, -12426, 18019, -18895, 13894, -5034, 9713, -19270, 10283,
- 8692, -18439, 9317, 13992, -13454, -13241, 12850, 13366, -13336,
- -16334, -498, 15976, 16213, -114, -16987, 18191, 13659, 4958,
- -5116, -13444, -18021, 12911, 13424, 13718, 13674, 13464, 13054
- }
-};
-
-/* means of log gains and LAR coefficients*/
-const WebRtc_Word16 WebRtcIsacfix_kMeansGainQ8[3][12] = {
- { -1758, -1370, -1758, -1373, -1757, -1375,
- -1758, -1374, -1758, -1373, -1755, -1370
- },
- { -1569, -1224, -1569, -1225, -1569, -1227,
- -1569, -1226, -1567, -1225, -1565, -1224
- },
- { -1452, -957, -1447, -951, -1438, -944,
- -1431, -938, -1419, -931, -1406, -926
- }
-};
-
-
-const WebRtc_Word32 WebRtcIsacfix_kMeansShapeQ17[3][108] = {
- { -119581, 34418, -44193, 11112, -4428, 18906, 9222, 8068, 1953, 5425,
- 1871, 1689, 109933, 33751, 10471, -2566, 1090, 2320, -119219, 33728,
- -43759, 11450, -4870, 19117, 9174, 8037, 1972, 5331, 1872, 1843,
- 109899, 34301, 10629, -2316, 1272, 2562, -118608, 32318, -44012, 11591,
- -4914, 18932, 9456, 8088, 1900, 5419, 1723, 1853, 109963, 35059,
- 10745, -2335, 1161, 2520, -119174, 32107, -44462, 11635, -4694, 18611,
- 9757, 8108, 1969, 5486, 1673, 1777, 109636, 34907, 10643, -2406,
- 1034, 2420, -118597, 32320, -44590, 10854, -4569, 18821, 9701, 7866,
- 2003, 5577, 1732, 1626, 109913, 34448, 10714, -2752, 990, 2228,
- -118138, 32996, -44352, 10334, -3772, 18488, 9464, 7865, 2208, 5540,
- 1745, 1664, 109880, 33381, 10640, -2779, 980, 2054
- },
- { -146328, 46370, 1047, 26431, 10035, 13933, 6415, 14359, -2368, 6661,
- 2269, 1764, 96623, 7802, 4163, 10742, 1643, 2954, -146871, 46561, 1127,
- 26225, 10113, 14096, 6771, 14323, -2037, 6788, 2297, 1761, 96324, 8382,
- 4309, 10450, 1695, 3016, -146502, 46475, 1580, 26118, 10487, 14179, 6622,
- 14439, -2034, 6757, 2342, 1761, 95869, 8966, 4347, 10358, 1999, 2855,
- -146958, 47717, 826, 25952, 10263, 14061, 5266, 13681, -2417, 6582, 2047,
- 1608, 96257, 9107, 4452, 10301, 1792, 2676, -146992, 47123, 446, 25822,
- 10405, 14292, 5140, 13804, -2403, 6496, 1834, 1735, 97489, 9253, 4414,
- 10684, 1549, 2721, -145811, 46182, 901, 26482, 10241, 14524, 6075, 14514,
- -2147, 6691, 2196, 1899, 97011, 8178, 4102, 10758, 1638, 2869
- },
- { -166617, 46969, -43908, 17726, 6330, 25615, 6913, 5450, -2301, 1984,
- 507, 2883, 149998, 28709, 19333, 16703, 11093, 8965, -168254, 46604,
- -44315, 17862, 6474, 25746, 7018, 5373, -2343, 1930, 513, 2819, 150391,
- 28627, 19194, 16678, 10998, 8929, -169093, 46084, -44767, 17427, 6401,
- 25674, 7147, 5472, -2336, 1820, 491, 2802, 149860, 28430, 19064, 16524,
- 10898, 8875, -170205, 46189, -44877, 17403, 6190, 25209, 7035, 5673, -2173,
- 1894, 574, 2756, 148830, 28230, 18819, 16418, 10789, 8811, -171263, 45045,
- -44834, 16858, 6103, 24726, 7014, 5713, -2103, 1877, 518, 2729, 147073,
- 27744, 18629, 16277, 10690, 8703, -171720, 44153, -45062, 15951, 5872,
- 24429, 7044, 5585, -2082, 1807, 519, 2769, 144791, 27402, 18490, 16126,
- 10548, 8635
- }
-};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_tables.h
deleted file mode 100644
index 4f2e0e7..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/lpc_tables.h
+++ /dev/null
@@ -1,98 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * lpc_tables.h
- *
- * header file for coding tables for the LPC coefficients
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_
-
-#include "typedefs.h"
-
-
-/* indices of KLT coefficients used */
-extern const WebRtc_UWord16 WebRtcIsacfix_kSelIndGain[12];
-
-extern const WebRtc_UWord16 WebRtcIsacfix_kSelIndShape[108];
-
-/* cdf array for model indicator */
-extern const WebRtc_UWord16 WebRtcIsacfix_kModelCdf[KLT_NUM_MODELS+1];
-
-/* pointer to cdf array for model indicator */
-extern const WebRtc_UWord16 *WebRtcIsacfix_kModelCdfPtr[1];
-
-/* initial cdf index for decoder of model indicator */
-extern const WebRtc_UWord16 WebRtcIsacfix_kModelInitIndex[1];
-
-/* offset to go from rounded value to quantization index */
-extern const WebRtc_Word16 WebRtcIsacfix_kQuantMinGain[12];
-
-extern const WebRtc_Word16 WebRtcIsacfix_kQuantMinShape[108];
-
-/* maximum quantization index */
-extern const WebRtc_UWord16 WebRtcIsacfix_kMaxIndGain[12];
-
-extern const WebRtc_UWord16 WebRtcIsacfix_kMaxIndShape[108];
-
-/* index offset */
-extern const WebRtc_UWord16 WebRtcIsacfix_kOffsetGain[KLT_NUM_MODELS][12];
-
-extern const WebRtc_UWord16 WebRtcIsacfix_kOffsetShape[KLT_NUM_MODELS][108];
-
-/* initial cdf index for KLT coefficients */
-extern const WebRtc_UWord16 WebRtcIsacfix_kInitIndexGain[KLT_NUM_MODELS][12];
-
-extern const WebRtc_UWord16 WebRtcIsacfix_kInitIndexShape[KLT_NUM_MODELS][108];
-
-/* offsets for quantizer representation levels */
-extern const WebRtc_UWord16 WebRtcIsacfix_kOfLevelsGain[3];
-
-extern const WebRtc_UWord16 WebRtcIsacfix_kOfLevelsShape[3];
-
-/* quantizer representation levels */
-extern const WebRtc_Word32 WebRtcIsacfix_kLevelsGainQ17[1176];
-
-extern const WebRtc_Word16 WebRtcIsacfix_kLevelsShapeQ10[1735];
-
-/* cdf tables for quantizer indices */
-extern const WebRtc_UWord16 WebRtcIsacfix_kCdfGain[1212];
-
-extern const WebRtc_UWord16 WebRtcIsacfix_kCdfShape[2059];
-
-/* pointers to cdf tables for quantizer indices */
-extern const WebRtc_UWord16 *WebRtcIsacfix_kCdfGainPtr[KLT_NUM_MODELS][12];
-
-extern const WebRtc_UWord16 *WebRtcIsacfix_kCdfShapePtr[KLT_NUM_MODELS][108];
-
-/* code length for all coefficients using different models */
-extern const WebRtc_Word16 WebRtcIsacfix_kCodeLenGainQ11[392];
-
-extern const WebRtc_Word16 WebRtcIsacfix_kCodeLenShapeQ11[577];
-
-/* left KLT transforms */
-extern const WebRtc_Word16 WebRtcIsacfix_kT1GainQ15[KLT_NUM_MODELS][4];
-
-extern const WebRtc_Word16 WebRtcIsacfix_kT1ShapeQ15[KLT_NUM_MODELS][324];
-
-/* right KLT transforms */
-extern const WebRtc_Word16 WebRtcIsacfix_kT2GainQ15[KLT_NUM_MODELS][36];
-
-extern const WebRtc_Word16 WebRtcIsacfix_kT2ShapeQ15[KLT_NUM_MODELS][36];
-
-/* means of log gains and LAR coefficients */
-extern const WebRtc_Word16 WebRtcIsacfix_kMeansGainQ8[KLT_NUM_MODELS][12];
-
-extern const WebRtc_Word32 WebRtcIsacfix_kMeansShapeQ17[3][108];
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_LPC_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_estimator.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_estimator.c
deleted file mode 100644
index 1702098..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_estimator.c
+++ /dev/null
@@ -1,519 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * pitch_estimator.c
- *
- * Pitch filter functions
- *
- */
-
-#include <string.h>
-
-#include "signal_processing_library.h"
-#include "pitch_estimator.h"
-
-/* log2[0.2, 0.5, 0.98] in Q8 */
-static const WebRtc_Word16 kLogLagWinQ8[3] = {
- -594, -256, -7
-};
-
-/* [1 -0.75 0.25] in Q12 */
-static const WebRtc_Word16 kACoefQ12[3] = {
- 4096, -3072, 1024
-};
-
-
-
-static __inline WebRtc_Word32 Log2Q8( WebRtc_UWord32 x ) {
-
- WebRtc_Word32 zeros, lg2;
- WebRtc_Word16 frac;
-
- zeros=WebRtcSpl_NormU32(x);
- frac=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(((WebRtc_UWord32)(WEBRTC_SPL_LSHIFT_W32(x, zeros))&0x7FFFFFFF), 23);
- /* log2(magn(i)) */
-
- lg2= (WEBRTC_SPL_LSHIFT_W32((31-zeros), 8)+frac);
- return lg2;
-
-}
-
-static __inline WebRtc_Word16 Exp2Q10(WebRtc_Word16 x) { // Both in and out in Q10
-
- WebRtc_Word16 tmp16_1, tmp16_2;
-
- tmp16_2=(WebRtc_Word16)(0x0400|(x&0x03FF));
- tmp16_1=-(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(x,10);
- if(tmp16_1>0)
- return (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W16(tmp16_2, tmp16_1);
- else
- return (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W16(tmp16_2, -tmp16_1);
-
-}
-
-
-
-/* 1D parabolic interpolation . All input and output values are in Q8 */
-static __inline void Intrp1DQ8(WebRtc_Word32 *x, WebRtc_Word32 *fx, WebRtc_Word32 *y, WebRtc_Word32 *fy) {
-
- WebRtc_Word16 sign1=1, sign2=1;
- WebRtc_Word32 r32, q32, t32, nom32, den32;
- WebRtc_Word16 t16, tmp16, tmp16_1;
-
- if ((fx[0]>0) && (fx[2]>0)) {
- r32=fx[1]-fx[2];
- q32=fx[0]-fx[1];
- nom32=q32+r32;
- den32=WEBRTC_SPL_MUL_32_16((q32-r32), 2);
- if (nom32<0)
- sign1=-1;
- if (den32<0)
- sign2=-1;
-
- /* t = (q32+r32)/(2*(q32-r32)) = (fx[0]-fx[1] + fx[1]-fx[2])/(2 * fx[0]-fx[1] - (fx[1]-fx[2]))*/
- /* (Signs are removed because WebRtcSpl_DivResultInQ31 can't handle negative numbers) */
- t32=WebRtcSpl_DivResultInQ31(WEBRTC_SPL_MUL_32_16(nom32, sign1),WEBRTC_SPL_MUL_32_16(den32, sign2)); /* t in Q31, without signs */
-
- t16=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(t32, 23); /* Q8 */
- t16=t16*sign1*sign2; /* t in Q8 with signs */
-
- *y = x[0]+t16; /* Q8 */
- // *y = x[1]+t16; /* Q8 */
-
- /* The following code calculates fy in three steps */
- /* fy = 0.5 * t * (t-1) * fx[0] + (1-t*t) * fx[1] + 0.5 * t * (t+1) * fx[2]; */
-
- /* Part I: 0.5 * t * (t-1) * fx[0] */
- tmp16_1=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(t16,t16); /* Q8*Q8=Q16 */
- tmp16_1 = WEBRTC_SPL_RSHIFT_W16(tmp16_1,2); /* Q16>>2 = Q14 */
- t16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(t16, 64); /* Q8<<6 = Q14 */
- tmp16 = tmp16_1-t16;
- *fy = WEBRTC_SPL_MUL_16_32_RSFT15(tmp16, fx[0]); /* (Q14 * Q8 >>15)/2 = Q8 */
-
- /* Part II: (1-t*t) * fx[1] */
- tmp16 = 16384-tmp16_1; /* 1 in Q14 - Q14 */
- *fy += WEBRTC_SPL_MUL_16_32_RSFT14(tmp16, fx[1]);/* Q14 * Q8 >> 14 = Q8 */
-
- /* Part III: 0.5 * t * (t+1) * fx[2] */
- tmp16 = tmp16_1+t16;
- *fy += WEBRTC_SPL_MUL_16_32_RSFT15(tmp16, fx[2]);/* (Q14 * Q8 >>15)/2 = Q8 */
- } else {
- *y = x[0];
- *fy= fx[1];
- }
-}
-
-
-static void FindFour32(WebRtc_Word32 *in, WebRtc_Word16 length, WebRtc_Word16 *bestind)
-{
- WebRtc_Word32 best[4]= {-100, -100, -100, -100};
- WebRtc_Word16 k;
-
- for (k=0; k<length; k++) {
- if (in[k] > best[3]) {
- if (in[k] > best[2]) {
- if (in[k] > best[1]) {
- if (in[k] > best[0]) { // The Best
- best[3] = best[2];
- bestind[3] = bestind[2];
- best[2] = best[1];
- bestind[2] = bestind[1];
- best[1] = best[0];
- bestind[1] = bestind[0];
- best[0] = in[k];
- bestind[0] = k;
- } else { // 2nd best
- best[3] = best[2];
- bestind[3] = bestind[2];
- best[2] = best[1];
- bestind[2] = bestind[1];
- best[1] = in[k];
- bestind[1] = k;
- }
- } else { // 3rd best
- best[3] = best[2];
- bestind[3] = bestind[2];
- best[2] = in[k];
- bestind[2] = k;
- }
- } else { // 4th best
- best[3] = in[k];
- bestind[3] = k;
- }
- }
- }
-}
-
-
-
-
-
-static void PCorr2Q32(const WebRtc_Word16 *in, WebRtc_Word32 *logcorQ8)
-{
- WebRtc_Word16 scaling,n,k;
- WebRtc_Word32 ysum32,csum32, lys, lcs;
- WebRtc_Word32 prod32, oneQ8;
-
-
- const WebRtc_Word16 *x, *inptr;
-
- oneQ8 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, 8); // 1.00 in Q8
-
- x = in + PITCH_MAX_LAG/2 + 2;
- scaling = WebRtcSpl_GetScalingSquare ((WebRtc_Word16 *) in, PITCH_CORR_LEN2, PITCH_CORR_LEN2);
- ysum32 = 1;
- csum32 = 0;
- x = in + PITCH_MAX_LAG/2 + 2;
- for (n = 0; n < PITCH_CORR_LEN2; n++) {
- ysum32 += WEBRTC_SPL_MUL_16_16_RSFT( (WebRtc_Word16) in[n],(WebRtc_Word16) in[n], scaling); // Q0
- csum32 += WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) x[n],(WebRtc_Word16) in[n], scaling); // Q0
- }
-
- logcorQ8 += PITCH_LAG_SPAN2 - 1;
-
- lys=Log2Q8((WebRtc_UWord32) ysum32); // Q8
- lys=WEBRTC_SPL_RSHIFT_W32(lys, 1); //sqrt(ysum);
-
- if (csum32>0) {
-
- lcs=Log2Q8((WebRtc_UWord32) csum32); // 2log(csum) in Q8
-
- if (lcs>(lys + oneQ8) ){ // csum/sqrt(ysum) > 2 in Q8
- *logcorQ8 = lcs - lys; // log2(csum/sqrt(ysum))
- } else {
- *logcorQ8 = oneQ8; // 1.00
- }
-
- } else {
- *logcorQ8 = 0;
- }
-
-
- for (k = 1; k < PITCH_LAG_SPAN2; k++) {
- inptr = &in[k];
- ysum32 -= WEBRTC_SPL_MUL_16_16_RSFT( (WebRtc_Word16) in[k-1],(WebRtc_Word16) in[k-1], scaling);
- ysum32 += WEBRTC_SPL_MUL_16_16_RSFT( (WebRtc_Word16) in[PITCH_CORR_LEN2 + k - 1],(WebRtc_Word16) in[PITCH_CORR_LEN2 + k - 1], scaling);
- csum32 = 0;
- prod32 = WEBRTC_SPL_MUL_16_16_RSFT( (WebRtc_Word16) x[0],(WebRtc_Word16) inptr[0], scaling);
-
- for (n = 1; n < PITCH_CORR_LEN2; n++) {
- csum32 += prod32;
- prod32 = WEBRTC_SPL_MUL_16_16_RSFT( (WebRtc_Word16) x[n],(WebRtc_Word16) inptr[n], scaling);
- }
-
- csum32 += prod32;
- logcorQ8--;
-
- lys=Log2Q8((WebRtc_UWord32)ysum32); // Q8
- lys=WEBRTC_SPL_RSHIFT_W32(lys, 1); //sqrt(ysum);
-
- if (csum32>0) {
-
- lcs=Log2Q8((WebRtc_UWord32) csum32); // 2log(csum) in Q8
-
- if (lcs>(lys + oneQ8) ){ // csum/sqrt(ysum) > 2
- *logcorQ8 = lcs - lys; // log2(csum/sqrt(ysum))
- } else {
- *logcorQ8 = oneQ8; // 1.00
- }
-
- } else {
- *logcorQ8 = 0;
- }
- }
-}
-
-
-
-void WebRtcIsacfix_InitialPitch(const WebRtc_Word16 *in, /* Q0 */
- PitchAnalysisStruct *State,
- WebRtc_Word16 *lagsQ7 /* Q7 */
- )
-{
- WebRtc_Word16 buf_dec16[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2];
- WebRtc_Word32 *crrvecQ8_1,*crrvecQ8_2;
- WebRtc_Word32 cv1q[PITCH_LAG_SPAN2+2],cv2q[PITCH_LAG_SPAN2+2], peakvq[PITCH_LAG_SPAN2+2];
- int k;
- WebRtc_Word16 peaks_indq;
- WebRtc_Word16 peakiq[PITCH_LAG_SPAN2];
- WebRtc_Word32 corr;
- WebRtc_Word32 corr32, corr_max32, corr_max_o32;
- WebRtc_Word16 npkq;
- WebRtc_Word16 best4q[4]={0,0,0,0};
- WebRtc_Word32 xq[3],yq[1],fyq[1];
- WebRtc_Word32 *fxq;
- WebRtc_Word32 best_lag1q, best_lag2q;
- WebRtc_Word32 tmp32a,tmp32b,lag32,ratq;
- WebRtc_Word16 start;
- WebRtc_Word16 oldgQ12, tmp16a, tmp16b, gain_bias16,tmp16c, tmp16d, bias16;
- WebRtc_Word32 tmp32c,tmp32d, tmp32e;
- WebRtc_Word16 old_lagQ;
- WebRtc_Word32 old_lagQ8;
- WebRtc_Word32 lagsQ8[4];
-
- old_lagQ = State->PFstr_wght.oldlagQ7; // Q7
- old_lagQ8= WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)old_lagQ,1); //Q8
-
- oldgQ12= State->PFstr_wght.oldgainQ12;
-
- crrvecQ8_1=&cv1q[1];
- crrvecQ8_2=&cv2q[1];
-
-
- /* copy old values from state buffer */
- memcpy(buf_dec16, State->dec_buffer16, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2)));
-
- /* decimation; put result after the old values */
- WebRtcIsacfix_DecimateAllpass32(in, State->decimator_state32, PITCH_FRAME_LEN,
- &buf_dec16[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2]);
-
- /* low-pass filtering */
- start= PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2;
- WebRtcSpl_FilterARFastQ12(&buf_dec16[start],&buf_dec16[start],(WebRtc_Word16*)kACoefQ12,3, PITCH_FRAME_LEN/2);
-
- /* copy end part back into state buffer */
- for (k = 0; k < (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2); k++)
- State->dec_buffer16[k] = buf_dec16[k+PITCH_FRAME_LEN/2];
-
-
- /* compute correlation for first and second half of the frame */
- PCorr2Q32(buf_dec16, crrvecQ8_1);
- PCorr2Q32(buf_dec16 + PITCH_CORR_STEP2, crrvecQ8_2);
-
-
- /* bias towards pitch lag of previous frame */
- tmp32a = Log2Q8((WebRtc_UWord32) old_lagQ8) - 2304; // log2(0.5*oldlag) in Q8
- tmp32b = WEBRTC_SPL_MUL_16_16_RSFT(oldgQ12,oldgQ12, 10); //Q12 & * 4.0;
- gain_bias16 = (WebRtc_Word16) tmp32b; //Q12
- if (gain_bias16 > 3276) gain_bias16 = 3276; // 0.8 in Q12
-
-
- for (k = 0; k < PITCH_LAG_SPAN2; k++)
- {
- if (crrvecQ8_1[k]>0) {
- tmp32b = Log2Q8((WebRtc_UWord32) (k + (PITCH_MIN_LAG/2-2)));
- tmp16a = (WebRtc_Word16) (tmp32b - tmp32a); // Q8 & fabs(ratio)<4
- tmp32c = WEBRTC_SPL_MUL_16_16_RSFT(tmp16a,tmp16a, 6); //Q10
- tmp16b = (WebRtc_Word16) tmp32c; // Q10 & <8
- tmp32d = WEBRTC_SPL_MUL_16_16_RSFT(tmp16b, 177 , 8); // mult with ln2 in Q8
- tmp16c = (WebRtc_Word16) tmp32d; // Q10 & <4
- tmp16d = Exp2Q10((WebRtc_Word16) -tmp16c); //Q10
- tmp32c = WEBRTC_SPL_MUL_16_16_RSFT(gain_bias16,tmp16d,13); // Q10 & * 0.5
- bias16 = (WebRtc_Word16) (1024 + tmp32c); // Q10
- tmp32b = Log2Q8((WebRtc_UWord32) bias16) - 2560; // Q10 in -> Q8 out with 10*2^8 offset
- crrvecQ8_1[k] += tmp32b ; // -10*2^8 offset
- }
- }
-
- /* taper correlation functions */
- for (k = 0; k < 3; k++) {
- crrvecQ8_1[k] += kLogLagWinQ8[k];
- crrvecQ8_2[k] += kLogLagWinQ8[k];
-
- crrvecQ8_1[PITCH_LAG_SPAN2-1-k] += kLogLagWinQ8[k];
- crrvecQ8_2[PITCH_LAG_SPAN2-1-k] += kLogLagWinQ8[k];
- }
-
-
- /* Make zeropadded corr vectors */
- cv1q[0]=0;
- cv2q[0]=0;
- cv1q[PITCH_LAG_SPAN2+1]=0;
- cv2q[PITCH_LAG_SPAN2+1]=0;
- corr_max32 = 0;
-
- for (k = 1; k <= PITCH_LAG_SPAN2; k++)
- {
-
-
- corr32=crrvecQ8_1[k-1];
- if (corr32 > corr_max32)
- corr_max32 = corr32;
-
- corr32=crrvecQ8_2[k-1];
- corr32 += -4; // Compensate for later (log2(0.99))
-
- if (corr32 > corr_max32)
- corr_max32 = corr32;
-
- }
-
- /* threshold value to qualify as a peak */
- // corr_max32 += -726; // log(0.14)/log(2.0) in Q8
- corr_max32 += -1000; // log(0.14)/log(2.0) in Q8
- corr_max_o32 = corr_max32;
-
-
- /* find peaks in corr1 */
- peaks_indq = 0;
- for (k = 1; k <= PITCH_LAG_SPAN2; k++)
- {
- corr32=cv1q[k];
- if (corr32>corr_max32) { // Disregard small peaks
- if ((corr32>=cv1q[k-1]) && (corr32>cv1q[k+1])) { // Peak?
- peakvq[peaks_indq] = corr32;
- peakiq[peaks_indq++] = k;
- }
- }
- }
-
-
- /* find highest interpolated peak */
- corr_max32=0;
- best_lag1q =0;
- if (peaks_indq > 0) {
- FindFour32(peakvq, (WebRtc_Word16) peaks_indq, best4q);
- npkq = WEBRTC_SPL_MIN(peaks_indq, 4);
-
- for (k=0;k<npkq;k++) {
-
- lag32 = peakiq[best4q[k]];
- fxq = &cv1q[peakiq[best4q[k]]-1];
- xq[0]= lag32;
- xq[0] = WEBRTC_SPL_LSHIFT_W32(xq[0], 8);
- Intrp1DQ8(xq, fxq, yq, fyq);
-
- tmp32a= Log2Q8((WebRtc_UWord32) *yq) - 2048; // offset 8*2^8
- /* Bias towards short lags */
- /* log(pow(0.8, log(2.0 * *y )))/log(2.0) */
- tmp32b= WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) tmp32a, -42, 8);
- tmp32c= tmp32b + 256;
- *fyq += tmp32c;
- if (*fyq > corr_max32) {
- corr_max32 = *fyq;
- best_lag1q = *yq;
- }
- }
- tmp32a = best_lag1q - OFFSET_Q8;
- tmp32b = WEBRTC_SPL_LSHIFT_W32(tmp32a, 1);
- lagsQ8[0] = tmp32b + PITCH_MIN_LAG_Q8;
- lagsQ8[1] = lagsQ8[0];
- } else {
- lagsQ8[0] = old_lagQ8;
- lagsQ8[1] = lagsQ8[0];
- }
-
- /* Bias towards constant pitch */
- tmp32a = lagsQ8[0] - PITCH_MIN_LAG_Q8;
- ratq = WEBRTC_SPL_RSHIFT_W32(tmp32a, 1) + OFFSET_Q8;
-
- for (k = 1; k <= PITCH_LAG_SPAN2; k++)
- {
- tmp32a = WEBRTC_SPL_LSHIFT_W32(k, 7); // 0.5*k Q8
- tmp32b = (WebRtc_Word32) (WEBRTC_SPL_LSHIFT_W32(tmp32a, 1)) - ratq; // Q8
- tmp32c = WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) tmp32b, (WebRtc_Word16) tmp32b, 8); // Q8
-
- tmp32b = (WebRtc_Word32) tmp32c + (WebRtc_Word32) WEBRTC_SPL_RSHIFT_W32(ratq, 1); // (k-r)^2 + 0.5 * r Q8
- tmp32c = Log2Q8((WebRtc_UWord32) tmp32a) - 2048; // offset 8*2^8 , log2(0.5*k) Q8
- tmp32d = Log2Q8((WebRtc_UWord32) tmp32b) - 2048; // offset 8*2^8 , log2(0.5*k) Q8
- tmp32e = tmp32c -tmp32d;
-
- cv2q[k] += WEBRTC_SPL_RSHIFT_W32(tmp32e, 1);
-
- }
-
- /* find peaks in corr2 */
- corr_max32 = corr_max_o32;
- peaks_indq = 0;
-
- for (k = 1; k <= PITCH_LAG_SPAN2; k++)
- {
- corr=cv2q[k];
- if (corr>corr_max32) { // Disregard small peaks
- if ((corr>=cv2q[k-1]) && (corr>cv2q[k+1])) { // Peak?
- peakvq[peaks_indq] = corr;
- peakiq[peaks_indq++] = k;
- }
- }
- }
-
-
-
- /* find highest interpolated peak */
- corr_max32 = 0;
- best_lag2q =0;
- if (peaks_indq > 0) {
-
- FindFour32(peakvq, (WebRtc_Word16) peaks_indq, best4q);
- npkq = WEBRTC_SPL_MIN(peaks_indq, 4);
- for (k=0;k<npkq;k++) {
-
- lag32 = peakiq[best4q[k]];
- fxq = &cv2q[peakiq[best4q[k]]-1];
-
- xq[0]= lag32;
- xq[0] = WEBRTC_SPL_LSHIFT_W32(xq[0], 8);
- Intrp1DQ8(xq, fxq, yq, fyq);
-
- /* Bias towards short lags */
- /* log(pow(0.8, log(2.0f * *y )))/log(2.0f) */
- tmp32a= Log2Q8((WebRtc_UWord32) *yq) - 2048; // offset 8*2^8
- tmp32b= WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16) tmp32a, -82, 8);
- tmp32c= tmp32b + 256;
- *fyq += tmp32c;
- if (*fyq > corr_max32) {
- corr_max32 = *fyq;
- best_lag2q = *yq;
- }
- }
-
- tmp32a = best_lag2q - OFFSET_Q8;
- tmp32b = WEBRTC_SPL_LSHIFT_W32(tmp32a, 1);
- lagsQ8[2] = tmp32b + PITCH_MIN_LAG_Q8;
- lagsQ8[3] = lagsQ8[2];
- } else {
- lagsQ8[2] = lagsQ8[0];
- lagsQ8[3] = lagsQ8[0];
- }
-
- lagsQ7[0]=(WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(lagsQ8[0], 1);
- lagsQ7[1]=(WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(lagsQ8[1], 1);
- lagsQ7[2]=(WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(lagsQ8[2], 1);
- lagsQ7[3]=(WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(lagsQ8[3], 1);
-
-
-}
-
-
-
-void WebRtcIsacfix_PitchAnalysis(const WebRtc_Word16 *inn, /* PITCH_FRAME_LEN samples */
- WebRtc_Word16 *outQ0, /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
- PitchAnalysisStruct *State,
- WebRtc_Word16 *PitchLags_Q7,
- WebRtc_Word16 *PitchGains_Q12)
-{
- WebRtc_Word16 inbufQ0[PITCH_FRAME_LEN + QLOOKAHEAD];
- WebRtc_Word16 k;
-
- /* inital pitch estimate */
- WebRtcIsacfix_InitialPitch(inn, State, PitchLags_Q7);
-
-
- /* Calculate gain */
- WebRtcIsacfix_PitchFilterGains(inn, &(State->PFstr_wght), PitchLags_Q7, PitchGains_Q12);
-
- /* concatenate previous input's end and current input */
- for (k = 0; k < QLOOKAHEAD; k++) {
- inbufQ0[k] = State->inbuf[k];
- }
- for (k = 0; k < PITCH_FRAME_LEN; k++) {
- inbufQ0[k+QLOOKAHEAD] = (WebRtc_Word16) inn[k];
- }
-
- /* lookahead pitch filtering for masking analysis */
- WebRtcIsacfix_PitchFilter(inbufQ0, outQ0, &(State->PFstr), PitchLags_Q7,PitchGains_Q12, 2);
-
-
- /* store last part of input */
- for (k = 0; k < QLOOKAHEAD; k++) {
- State->inbuf[k] = inbufQ0[k + PITCH_FRAME_LEN];
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_estimator.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_estimator.h
deleted file mode 100644
index afdc978..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_estimator.h
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * pitch_estimator.h
- *
- * Pitch functions
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_
-
-#include "structs.h"
-
-
-
-void WebRtcIsacfix_PitchAnalysis(const WebRtc_Word16 *in, /* PITCH_FRAME_LEN samples */
- WebRtc_Word16 *outQ0, /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
- PitchAnalysisStruct *State,
- WebRtc_Word16 *lagsQ7,
- WebRtc_Word16 *PitchGains_Q12);
-
-
-void WebRtcIsacfix_InitialPitch(const WebRtc_Word16 *in,
- PitchAnalysisStruct *State,
- WebRtc_Word16 *qlags);
-
-void WebRtcIsacfix_PitchFilter(WebRtc_Word16 *indatFix,
- WebRtc_Word16 *outdatQQ,
- PitchFiltstr *pfp,
- WebRtc_Word16 *lagsQ7,
- WebRtc_Word16 *gainsQ12,
- WebRtc_Word16 type);
-
-void WebRtcIsacfix_PitchFilterGains(const WebRtc_Word16 *indatQ0,
- PitchFiltstr *pfp,
- WebRtc_Word16 *lagsQ7,
- WebRtc_Word16 *gainsQ12);
-
-
-
-void WebRtcIsacfix_DecimateAllpass32(const WebRtc_Word16 *in,
- WebRtc_Word32 *state_in, /* array of size: 2*ALLPASSSECTIONS+1 */
- WebRtc_Word16 N, /* number of input samples */
- WebRtc_Word16 *out); /* array of size N/2 */
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_ESTIMATOR_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_filter.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_filter.c
deleted file mode 100644
index 87af153..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_filter.c
+++ /dev/null
@@ -1,331 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * pitch_estimatorfilter.c
- *
- * Pitch filter functions
- *
- */
-
-#include <string.h>
-
-#include "pitch_estimator.h"
-
-
-/* Filter coefficicients in Q15 */
-static const WebRtc_Word16 kDampFilter[PITCH_DAMPORDER] = {
- -2294, 8192, 20972, 8192, -2294
-};
-
-/* Interpolation coefficients; generated by design_pitch_filter.m.
- * Coefficients are stored in Q14.
- */
-static const WebRtc_Word16 kIntrpCoef[PITCH_FRACS][PITCH_FRACORDER] = {
- {-367, 1090, -2706, 9945, 10596, -3318, 1626, -781, 287},
- {-325, 953, -2292, 7301, 12963, -3320, 1570, -743, 271},
- {-240, 693, -1622, 4634, 14809, -2782, 1262, -587, 212},
- {-125, 358, -817, 2144, 15982, -1668, 721, -329, 118},
- { 0, 0, -1, 1, 16380, 1, -1, 0, 0},
- { 118, -329, 721, -1668, 15982, 2144, -817, 358, -125},
- { 212, -587, 1262, -2782, 14809, 4634, -1622, 693, -240},
- { 271, -743, 1570, -3320, 12963, 7301, -2292, 953, -325}
-};
-
-
-
-
-static __inline WebRtc_Word32 CalcLrIntQ(WebRtc_Word32 fixVal, WebRtc_Word16 qDomain) {
- WebRtc_Word32 intgr;
- WebRtc_Word32 roundVal;
-
- roundVal = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, qDomain-1);
- intgr = WEBRTC_SPL_RSHIFT_W32(fixVal+roundVal, qDomain);
-
- return intgr;
-}
-
-void WebRtcIsacfix_PitchFilter(WebRtc_Word16 *indatQQ, /* Q10 if type is 1 or 4, Q0 if type is 2 */
- WebRtc_Word16 *outdatQQ,
- PitchFiltstr *pfp,
- WebRtc_Word16 *lagsQ7,
- WebRtc_Word16 *gainsQ12,
- WebRtc_Word16 type)
-{
- int k, n, m, ind;
- WebRtc_Word16 sign = 1;
- WebRtc_Word16 inystateQQ[PITCH_DAMPORDER];
- WebRtc_Word16 ubufQQ[PITCH_INTBUFFSIZE+QLOOKAHEAD];
- WebRtc_Word16 Gain = 21299; /* 1.3 in Q14 */
- WebRtc_Word16 DivFactor = 6553; /* 0.2 in Q15 */
- WebRtc_Word16 oldLagQ7, oldGainQ12,
- lagdeltaQ7, curLagQ7,
- gaindeltaQ12, curGainQ12;
- WebRtc_Word16 tmpW16, indW16=0, frcQQ, cnt=0, pos, pos2;
- const WebRtc_Word16 *fracoeffQQ=NULL;
- WebRtc_Word32 tmpW32;
-
- if (type==4)
- sign = -1;
-
- /* Set up buffer and states */
- memcpy(ubufQQ, pfp->ubufQQ, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), PITCH_BUFFSIZE));
- memcpy(inystateQQ, pfp->ystateQQ, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), PITCH_DAMPORDER));
-
- /* Get old lag and gain value from memory */
- oldLagQ7 = pfp->oldlagQ7;
- oldGainQ12 = pfp->oldgainQ12;
-
- if (type==4) {
- /* make output more periodic */
- /* Fixed 1.3 = 21299 in Q14 */
- for (k=0;k<PITCH_SUBFRAMES;k++)
- gainsQ12[k] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gainsQ12[k], Gain, 14);
- }
-
- /* No interpolation if pitch lag step is big */
- if ((WEBRTC_SPL_RSHIFT_W16(WEBRTC_SPL_MUL_16_16(lagsQ7[0], 3), 1) < oldLagQ7) ||
- (lagsQ7[0] > WEBRTC_SPL_RSHIFT_W16(WEBRTC_SPL_MUL_16_16(oldLagQ7, 3), 1))) {
- oldLagQ7 = lagsQ7[0];
- oldGainQ12 = gainsQ12[0];
- }
-
- ind=0;
- for (k=0;k<PITCH_SUBFRAMES;k++) {
-
- /* Calculate interpolation steps */
- lagdeltaQ7 = lagsQ7[k]-oldLagQ7;
- lagdeltaQ7 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(lagdeltaQ7,DivFactor,15);
- curLagQ7 = oldLagQ7;
- gaindeltaQ12 = gainsQ12[k]-oldGainQ12;
- gaindeltaQ12 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gaindeltaQ12,DivFactor,15);
-
- curGainQ12 = oldGainQ12;
- oldLagQ7 = lagsQ7[k];
- oldGainQ12 = gainsQ12[k];
-
-
- for (n=0;n<PITCH_SUBFRAME_LEN;n++) {
-
- if (cnt == 0) { /* Update parameters */
-
- curGainQ12 += gaindeltaQ12;
- curLagQ7 += lagdeltaQ7;
- indW16 = (WebRtc_Word16)CalcLrIntQ(curLagQ7,7);
- tmpW16 = WEBRTC_SPL_LSHIFT_W16(indW16,7);
- tmpW16 -= curLagQ7;
- frcQQ = WEBRTC_SPL_RSHIFT_W16(tmpW16,4);
- frcQQ += 4;
-
- if(frcQQ==PITCH_FRACS)
- frcQQ=0;
- fracoeffQQ = kIntrpCoef[frcQQ];
-
- cnt=12;
- }
-
- /* shift low pass filter state */
- for (m=PITCH_DAMPORDER-1;m>0;m--)
- inystateQQ[m] = inystateQQ[m-1];
-
- /* Filter to get fractional pitch */
- pos = ind + PITCH_BUFFSIZE;
- pos2 = pos - (indW16 + 2);
-
- tmpW32=0;
- for (m=0;m<PITCH_FRACORDER;m++)
- tmpW32 += WEBRTC_SPL_MUL_16_16(ubufQQ[pos2+m], fracoeffQQ[m]);
-
- /* Saturate to avoid overflow in tmpW16 */
- tmpW32 = WEBRTC_SPL_SAT(536862719, tmpW32, -536879104);
- tmpW32 += 8192;
- tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32,14);
-
- inystateQQ[0] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(curGainQ12, tmpW16,12); /* Multiply with gain */
-
- /* Low pass filter */
- tmpW32=0;
- for (m=0;m<PITCH_DAMPORDER;m++)
- tmpW32 += WEBRTC_SPL_MUL_16_16(inystateQQ[m], kDampFilter[m]);
-
- /* Saturate to avoid overflow in tmpW16 */
- tmpW32 = WEBRTC_SPL_SAT(1073725439, tmpW32, -1073758208);
- tmpW32 += 16384;
- tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32,15);
-
- /* Subtract from input and update buffer */
- tmpW32 = indatQQ[ind] - WEBRTC_SPL_MUL_16_16(sign, tmpW16);
- outdatQQ[ind] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmpW32);
- tmpW32 = indatQQ[ind] + (WebRtc_Word32)outdatQQ[ind];
- ubufQQ[pos] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmpW32);
-
- ind++;
- cnt--;
- }
- }
-
-
- /* Export buffer and states */
- memcpy(pfp->ubufQQ, ubufQQ+PITCH_FRAME_LEN, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), PITCH_BUFFSIZE));
- memcpy(pfp->ystateQQ, inystateQQ, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), PITCH_DAMPORDER));
-
- pfp->oldlagQ7 = oldLagQ7;
- pfp->oldgainQ12 = oldGainQ12;
-
- if (type==2) {
- /* Filter look-ahead segment */
- for (n=0;n<QLOOKAHEAD;n++) {
- /* shift low pass filter state */
- for (m=PITCH_DAMPORDER-1;m>0;m--)
- inystateQQ[m] = inystateQQ[m-1];
-
- /* Filter to get fractional pitch */
- pos = ind + PITCH_BUFFSIZE;
- pos2= pos - (indW16 + 2);
-
- tmpW32=0;
- for (m=0;m<PITCH_FRACORDER;m++)
- tmpW32 += WEBRTC_SPL_MUL_16_16(ubufQQ[pos2+m], fracoeffQQ[m]);
-
- if (tmpW32<536862720) {//536870912)
- tmpW32 += 8192;
- tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32,14);
- } else
- tmpW16= 32767;
- inystateQQ[0] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(curGainQ12, tmpW16,12); /* Multiply with gain */
-
- /* Low pass filter */
- tmpW32=0;
- for (m=0;m<PITCH_DAMPORDER;m++)
- tmpW32 += WEBRTC_SPL_MUL_16_16(inystateQQ[m], kDampFilter[m]);
- if (tmpW32<1073725440) { //1073741824)
- tmpW32 += 16384;
- tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32,15);
- } else
- tmpW16 = 32767;
-
- /* Subtract from input and update buffer */
- tmpW32 = indatQQ[ind] - (WebRtc_Word32)tmpW16;
- outdatQQ[ind] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmpW32);
- tmpW32 = indatQQ[ind] + (WebRtc_Word32)outdatQQ[ind];
- ubufQQ[pos] = (WebRtc_Word16)WebRtcSpl_SatW32ToW16(tmpW32);
-
- ind++;
- }
-
- }
-
-
-}
-
-
-void WebRtcIsacfix_PitchFilterGains(const WebRtc_Word16 *indatQ0,
- PitchFiltstr *pfp,
- WebRtc_Word16 *lagsQ7,
- WebRtc_Word16 *gainsQ12)
-{
- int k, n, m, ind;
-
- WebRtc_Word16 ubufQQ[PITCH_INTBUFFSIZE];
- WebRtc_Word16 oldLagQ7,lagdeltaQ7, curLagQ7;
- WebRtc_Word16 DivFactor = 6553;
- const WebRtc_Word16 *fracoeffQQ = NULL;
- WebRtc_Word16 scale;
- WebRtc_Word16 cnt=0, pos, pos3QQ, frcQQ, indW16 = 0, tmpW16;
- WebRtc_Word32 tmpW32, tmp2W32, csum1QQ, esumxQQ;
-
- /* Set up buffer and states */
- memcpy(ubufQQ, pfp->ubufQQ, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), PITCH_BUFFSIZE));
- oldLagQ7 = pfp->oldlagQ7;
-
- /* No interpolation if pitch lag step is big */
- if ((WEBRTC_SPL_RSHIFT_W16(WEBRTC_SPL_MUL_16_16(lagsQ7[0], 3), 1) < oldLagQ7) ||
- (lagsQ7[0] > WEBRTC_SPL_RSHIFT_W16(WEBRTC_SPL_MUL_16_16(oldLagQ7, 3), 1))) {
- oldLagQ7 = lagsQ7[0];
- }
-
- ind=0;
- scale=0;
- for (k=0;k<PITCH_SUBFRAMES;k++) {
-
- /* Calculate interpolation steps */
- lagdeltaQ7 = lagsQ7[k]-oldLagQ7;
- lagdeltaQ7 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(lagdeltaQ7,DivFactor,15);
- curLagQ7 = oldLagQ7;
- oldLagQ7 = lagsQ7[k];
-
- csum1QQ=1;
- esumxQQ=1;
-
- for (n=0;n<PITCH_SUBFRAME_LEN;n++) {
-
- if (cnt == 0) { /* Update parameters */
- curLagQ7 += lagdeltaQ7;
- indW16 = (WebRtc_Word16)CalcLrIntQ(curLagQ7,7);
- tmpW16 = WEBRTC_SPL_LSHIFT_W16(indW16,7);
- tmpW16 -= curLagQ7;
- frcQQ = WEBRTC_SPL_RSHIFT_W16(tmpW16,4);
- frcQQ += 4;
-
- if(frcQQ==PITCH_FRACS)
- frcQQ=0;
- fracoeffQQ = kIntrpCoef[frcQQ];
-
- cnt=12;
- }
-
- /* Filter to get fractional pitch */
- pos = ind + PITCH_BUFFSIZE;
- pos3QQ = pos - (indW16 + 4);
-
- tmpW32=0;
- for (m=0;m<PITCH_FRACORDER;m++){
- tmpW32 += WEBRTC_SPL_MUL_16_16(ubufQQ[pos3QQ+m], fracoeffQQ[m]);
- }
-
- /* Subtract from input and update buffer */
- ubufQQ[pos] = indatQ0[ind];
-
- tmp2W32 = WEBRTC_SPL_MUL_16_32_RSFT14(indatQ0[ind], tmpW32);
- tmpW32 += 8192;
- tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32,14);
- tmpW32 = WEBRTC_SPL_MUL_16_16(tmpW16, tmpW16);
-
- if ((tmp2W32>1073700000) || (csum1QQ>1073700000) || (tmpW32>1073700000) || (esumxQQ>1073700000)) {//2^30
- scale++;
- csum1QQ = WEBRTC_SPL_RSHIFT_W32(csum1QQ,1);
- esumxQQ = WEBRTC_SPL_RSHIFT_W32(esumxQQ,1);
- }
- tmp2W32 = WEBRTC_SPL_RSHIFT_W32(tmp2W32,scale);
- csum1QQ += tmp2W32;
- tmpW32 = WEBRTC_SPL_RSHIFT_W32(tmpW32,scale);
- esumxQQ += tmpW32;
-
- ind++;
- cnt--;
- }
-
- if (csum1QQ<esumxQQ) {
- tmp2W32=WebRtcSpl_DivResultInQ31(csum1QQ,esumxQQ);
-
- /* Gain should be half the correlation */
- tmpW32=WEBRTC_SPL_RSHIFT_W32(tmp2W32,20);
- } else
- tmpW32=4096;
- gainsQ12[k]=(WebRtc_Word16)WEBRTC_SPL_SAT(PITCH_MAX_GAIN_Q12, tmpW32, 0);
-
- }
-
- /* Export buffer and states */
- memcpy(pfp->ubufQQ, ubufQQ+PITCH_FRAME_LEN, WEBRTC_SPL_MUL_16_16(sizeof(WebRtc_Word16), PITCH_BUFFSIZE));
- pfp->oldlagQ7 = lagsQ7[PITCH_SUBFRAMES-1];
- pfp->oldgainQ12 = gainsQ12[PITCH_SUBFRAMES-1];
-
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_gain_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_gain_tables.c
deleted file mode 100644
index 50ea658..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_gain_tables.c
+++ /dev/null
@@ -1,149 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * pitch_gain_tables.c
- *
- * This file contains tables for the pitch filter side-info in the entropy coder.
- *
- */
-
-#include "pitch_gain_tables.h"
-
-
-/********************* Pitch Filter Gain Coefficient Tables ************************/
-
-/* cdf for quantized pitch filter gains */
-const WebRtc_UWord16 WebRtcIsacfix_kPitchGainCdf[255] = {
- 0, 2, 4, 6, 64, 901, 903, 905, 16954, 16956,
- 16961, 17360, 17362, 17364, 17366, 17368, 17370, 17372, 17374, 17411,
- 17514, 17516, 17583, 18790, 18796, 18802, 20760, 20777, 20782, 21722,
- 21724, 21728, 21738, 21740, 21742, 21744, 21746, 21748, 22224, 22227,
- 22230, 23214, 23229, 23239, 25086, 25108, 25120, 26088, 26094, 26098,
- 26175, 26177, 26179, 26181, 26183, 26185, 26484, 26507, 26522, 27705,
- 27731, 27750, 29767, 29799, 29817, 30866, 30883, 30885, 31025, 31029,
- 31031, 31033, 31035, 31037, 31114, 31126, 31134, 32687, 32722, 32767,
- 35718, 35742, 35757, 36943, 36952, 36954, 37115, 37128, 37130, 37132,
- 37134, 37136, 37143, 37145, 37152, 38843, 38863, 38897, 47458, 47467,
- 47474, 49040, 49061, 49063, 49145, 49157, 49159, 49161, 49163, 49165,
- 49167, 49169, 49171, 49757, 49770, 49782, 61333, 61344, 61346, 62860,
- 62883, 62885, 62887, 62889, 62891, 62893, 62895, 62897, 62899, 62901,
- 62903, 62905, 62907, 62909, 65496, 65498, 65500, 65521, 65523, 65525,
- 65527, 65529, 65531, 65533, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535
-};
-
-/* index limits and ranges */
-const WebRtc_Word16 WebRtcIsacfix_kLowerlimiGain[3] = {
- -7, -2, -1
-};
-
-const WebRtc_Word16 WebRtcIsacfix_kUpperlimitGain[3] = {
- 0, 3, 1
-};
-
-const WebRtc_UWord16 WebRtcIsacfix_kMultsGain[2] = {
- 18, 3
-};
-
-/* size of cdf table */
-const WebRtc_UWord16 WebRtcIsacfix_kCdfTableSizeGain[1] = {
- 256
-};
-
-/* mean values of pitch filter gains in FIXED point Q12 */
-const WebRtc_Word16 WebRtcIsacfix_kPitchGain1[144] = {
- 843, 1092, 1336, 1222, 1405, 1656, 1500, 1815, 1843, 1838, 1839,
- 1843, 1843, 1843, 1843, 1843, 1843, 1843, 814, 846, 1092, 1013,
- 1174, 1383, 1391, 1511, 1584, 1734, 1753, 1843, 1843, 1843, 1843,
- 1843, 1843, 1843, 524, 689, 777, 845, 947, 1069, 1090, 1263,
- 1380, 1447, 1559, 1676, 1645, 1749, 1843, 1843, 1843, 1843, 81,
- 477, 563, 611, 706, 806, 849, 1012, 1192, 1128, 1330, 1489,
- 1425, 1576, 1826, 1741, 1843, 1843, 0, 290, 305, 356, 488,
- 575, 602, 741, 890, 835, 1079, 1196, 1182, 1376, 1519, 1506,
- 1680, 1843, 0, 47, 97, 69, 289, 381, 385, 474, 617,
- 664, 803, 1079, 935, 1160, 1269, 1265, 1506, 1741, 0, 0,
- 0, 0, 112, 120, 190, 283, 442, 343, 526, 809, 684,
- 935, 1134, 1020, 1265, 1506, 0, 0, 0, 0, 0, 0,
- 0, 111, 256, 87, 373, 597, 430, 684, 935, 770, 1020,
- 1265
-};
-
-const WebRtc_Word16 WebRtcIsacfix_kPitchGain2[144] = {
- 1760, 1525, 1285, 1747, 1671, 1393, 1843, 1826, 1555, 1843, 1784,
- 1606, 1843, 1843, 1711, 1843, 1843, 1814, 1389, 1275, 1040, 1564,
- 1414, 1252, 1610, 1495, 1343, 1753, 1592, 1405, 1804, 1720, 1475,
- 1843, 1814, 1581, 1208, 1061, 856, 1349, 1148, 994, 1390, 1253,
- 1111, 1495, 1343, 1178, 1770, 1465, 1234, 1814, 1581, 1342, 1040,
- 793, 713, 1053, 895, 737, 1128, 1003, 861, 1277, 1094, 981,
- 1475, 1192, 1019, 1581, 1342, 1098, 855, 570, 483, 833, 648,
- 540, 948, 744, 572, 1009, 844, 636, 1234, 934, 685, 1342,
- 1217, 984, 537, 318, 124, 603, 423, 350, 687, 479, 322,
- 791, 581, 430, 987, 671, 488, 1098, 849, 597, 283, 27,
- 0, 397, 222, 38, 513, 271, 124, 624, 325, 157, 737,
- 484, 233, 849, 597, 343, 27, 0, 0, 141, 0, 0,
- 256, 69, 0, 370, 87, 0, 484, 229, 0, 597, 343,
- 87
-};
-
-const WebRtc_Word16 WebRtcIsacfix_kPitchGain3[144] = {
- 1843, 1843, 1711, 1843, 1818, 1606, 1843, 1827, 1511, 1814, 1639,
- 1393, 1760, 1525, 1285, 1656, 1419, 1176, 1835, 1718, 1475, 1841,
- 1650, 1387, 1648, 1498, 1287, 1600, 1411, 1176, 1522, 1299, 1040,
- 1419, 1176, 928, 1773, 1461, 1128, 1532, 1355, 1202, 1429, 1260,
- 1115, 1398, 1151, 1025, 1172, 1080, 790, 1176, 928, 677, 1475,
- 1147, 1019, 1276, 1096, 922, 1214, 1010, 901, 1057, 893, 800,
- 1040, 796, 734, 928, 677, 424, 1137, 897, 753, 1120, 830,
- 710, 875, 751, 601, 795, 642, 583, 790, 544, 475, 677,
- 474, 140, 987, 750, 482, 697, 573, 450, 691, 487, 303,
- 661, 394, 332, 537, 303, 220, 424, 168, 0, 737, 484,
- 229, 624, 348, 153, 441, 261, 136, 397, 166, 51, 283,
- 27, 0, 168, 0, 0, 484, 229, 0, 370, 57, 0,
- 256, 43, 0, 141, 0, 0, 27, 0, 0, 0, 0,
- 0
-};
-
-
-const WebRtc_Word16 WebRtcIsacfix_kPitchGain4[144] = {
- 1843, 1843, 1843, 1843, 1841, 1843, 1500, 1821, 1843, 1222, 1434,
- 1656, 843, 1092, 1336, 504, 757, 1007, 1843, 1843, 1843, 1838,
- 1791, 1843, 1265, 1505, 1599, 965, 1219, 1425, 730, 821, 1092,
- 249, 504, 757, 1783, 1819, 1843, 1351, 1567, 1727, 1096, 1268,
- 1409, 805, 961, 1131, 444, 670, 843, 0, 249, 504, 1425,
- 1655, 1743, 1096, 1324, 1448, 822, 1019, 1199, 490, 704, 867,
- 81, 450, 555, 0, 0, 249, 1247, 1428, 1530, 881, 1073,
- 1283, 610, 759, 939, 278, 464, 645, 0, 200, 270, 0,
- 0, 0, 935, 1163, 1410, 528, 790, 1068, 377, 499, 717,
- 173, 240, 274, 0, 43, 62, 0, 0, 0, 684, 935,
- 1182, 343, 551, 735, 161, 262, 423, 0, 55, 27, 0,
- 0, 0, 0, 0, 0, 430, 684, 935, 87, 377, 597,
- 0, 46, 256, 0, 0, 0, 0, 0, 0, 0, 0,
- 0
-};
-
-
-
-/* transform matrix in Q12*/
-const WebRtc_Word16 WebRtcIsacfix_kTransform[4][4] = {
- { -2048, -2048, -2048, -2048 },
- { 2748, 916, -916, -2748 },
- { 2048, -2048, -2048, 2048 },
- { 916, -2748, 2748, -916 }
-};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_gain_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_gain_tables.h
deleted file mode 100644
index 788e553..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_gain_tables.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * pitch_gain_tables.h
- *
- * This file contains tables for the pitch filter side-info in the entropy coder.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_
-
-#include "typedefs.h"
-
-
-/********************* Pitch Filter Gain Coefficient Tables ************************/
-/* cdf for quantized pitch filter gains */
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchGainCdf[255];
-
-/* index limits and ranges */
-extern const WebRtc_Word16 WebRtcIsacfix_kLowerlimiGain[3];
-extern const WebRtc_Word16 WebRtcIsacfix_kUpperlimitGain[3];
-extern const WebRtc_UWord16 WebRtcIsacfix_kMultsGain[2];
-
-/* mean values of pitch filter gains in Q12*/
-extern const WebRtc_Word16 WebRtcIsacfix_kPitchGain1[144];
-extern const WebRtc_Word16 WebRtcIsacfix_kPitchGain2[144];
-extern const WebRtc_Word16 WebRtcIsacfix_kPitchGain3[144];
-extern const WebRtc_Word16 WebRtcIsacfix_kPitchGain4[144];
-
-/* size of cdf table */
-extern const WebRtc_UWord16 WebRtcIsacfix_kCdfTableSizeGain[1];
-
-/* transform matrix */
-extern const WebRtc_Word16 WebRtcIsacfix_kTransform[4][4];
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_GAIN_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_lag_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_lag_tables.c
deleted file mode 100644
index 81700e4..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_lag_tables.c
+++ /dev/null
@@ -1,306 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * pitch_lag_tables.c
- *
- * This file contains tables for the pitch filter side-info in the entropy coder.
- *
- */
-
-#include "settings.h"
-#include "pitch_lag_tables.h"
-
-
-/********************* Pitch Filter Gain Coefficient Tables ************************/
-
-/* tables for use with small pitch gain */
-
-/* cdf for quantized pitch filter lags */
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Lo[127] = {
- 0, 134, 336, 549, 778, 998, 1264, 1512, 1777, 2070,
- 2423, 2794, 3051, 3361, 3708, 3979, 4315, 4610, 4933, 5269,
- 5575, 5896, 6155, 6480, 6816, 7129, 7477, 7764, 8061, 8358,
- 8718, 9020, 9390, 9783, 10177, 10543, 10885, 11342, 11795, 12213,
- 12680, 13096, 13524, 13919, 14436, 14903, 15349, 15795, 16267, 16734,
- 17266, 17697, 18130, 18632, 19080, 19447, 19884, 20315, 20735, 21288,
- 21764, 22264, 22723, 23193, 23680, 24111, 24557, 25022, 25537, 26082,
- 26543, 27090, 27620, 28139, 28652, 29149, 29634, 30175, 30692, 31273,
- 31866, 32506, 33059, 33650, 34296, 34955, 35629, 36295, 36967, 37726,
- 38559, 39458, 40364, 41293, 42256, 43215, 44231, 45253, 46274, 47359,
- 48482, 49678, 50810, 51853, 53016, 54148, 55235, 56263, 57282, 58363,
- 59288, 60179, 61076, 61806, 62474, 63129, 63656, 64160, 64533, 64856,
- 65152, 65535, 65535, 65535, 65535, 65535, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Lo[20] = {
- 0, 429, 3558, 5861, 8558, 11639, 15210, 19502, 24773, 31983,
- 42602, 48567, 52601, 55676, 58160, 60172, 61889, 63235, 65383, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Lo[2] = {
- 0, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Lo[10] = {
- 0, 2966, 6368, 11182, 19431, 37793, 48532, 55353, 60626, 65535
-};
-
-const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrLo[4] = {
- WebRtcIsacfix_kPitchLagCdf1Lo,
- WebRtcIsacfix_kPitchLagCdf2Lo,
- WebRtcIsacfix_kPitchLagCdf3Lo,
- WebRtcIsacfix_kPitchLagCdf4Lo
-};
-
-/* size of first cdf table */
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeLo[1] = {
- 128
-};
-
-/* index limits and ranges */
-const WebRtc_Word16 WebRtcIsacfix_kLowerLimitLo[4] = {
- -140, -9, 0, -4
-};
-
-const WebRtc_Word16 WebRtcIsacfix_kUpperLimitLo[4] = {
- -20, 9, 0, 4
-};
-
-/* initial index for arithmetic decoder */
-const WebRtc_UWord16 WebRtcIsacfix_kInitIndLo[3] = {
- 10, 1, 5
-};
-
-/* mean values of pitch filter lags in Q10 */
-
-const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Lo[19] = {
- -17627, -16207, -14409, -12319, -10253, -8200, -6054, -3986, -1948, -19,
- 1937, 3974, 6064, 8155, 10229, 12270, 14296, 16127, 17520
-};
-
-const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Lo[9] = {
- -7949, -6063, -4036, -1941, 38, 1977, 4060, 6059
-};
-
-
-
-/* tables for use with medium pitch gain */
-
-/* cdf for quantized pitch filter lags */
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Mid[255] = {
- 0, 28, 61, 88, 121, 149, 233, 331, 475, 559,
- 624, 661, 689, 712, 745, 791, 815, 843, 866, 922,
- 959, 1024, 1061, 1117, 1178, 1238, 1280, 1350, 1453, 1513,
- 1564, 1625, 1671, 1741, 1788, 1904, 2072, 2421, 2626, 2770,
- 2840, 2900, 2942, 3012, 3068, 3115, 3147, 3194, 3254, 3319,
- 3366, 3520, 3678, 3780, 3850, 3911, 3957, 4032, 4106, 4185,
- 4292, 4474, 4683, 4842, 5019, 5191, 5321, 5428, 5540, 5675,
- 5763, 5847, 5959, 6127, 6304, 6564, 6839, 7090, 7263, 7421,
- 7556, 7728, 7872, 7984, 8142, 8361, 8580, 8743, 8938, 9227,
- 9409, 9539, 9674, 9795, 9930, 10060, 10177, 10382, 10614, 10861,
- 11038, 11271, 11415, 11629, 11792, 12044, 12193, 12416, 12574, 12821,
- 13007, 13235, 13445, 13654, 13901, 14134, 14488, 15000, 15703, 16285,
- 16504, 16797, 17086, 17328, 17579, 17807, 17998, 18268, 18538, 18836,
- 19087, 19274, 19474, 19716, 19935, 20270, 20833, 21303, 21532, 21741,
- 21978, 22207, 22523, 22770, 23054, 23613, 23943, 24204, 24399, 24651,
- 24832, 25074, 25270, 25549, 25759, 26015, 26150, 26424, 26713, 27048,
- 27342, 27504, 27681, 27854, 28021, 28207, 28412, 28664, 28859, 29064,
- 29278, 29548, 29748, 30107, 30377, 30656, 30856, 31164, 31452, 31755,
- 32011, 32328, 32626, 32919, 33319, 33789, 34329, 34925, 35396, 35973,
- 36443, 36964, 37551, 38156, 38724, 39357, 40023, 40908, 41587, 42602,
- 43924, 45037, 45810, 46597, 47421, 48291, 49092, 50051, 51448, 52719,
- 53440, 54241, 54944, 55977, 56676, 57299, 57872, 58389, 59059, 59688,
- 60237, 60782, 61094, 61573, 61890, 62290, 62658, 63030, 63217, 63454,
- 63622, 63882, 64003, 64273, 64427, 64529, 64581, 64697, 64758, 64902,
- 65414, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Mid[36] = {
- 0, 71, 335, 581, 836, 1039, 1323, 1795, 2258, 2608,
- 3005, 3591, 4243, 5344, 7163, 10583, 16848, 28078, 49448, 57007,
- 60357, 61850, 62837, 63437, 63872, 64188, 64377, 64614, 64774, 64949,
- 65039, 65115, 65223, 65360, 65474, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Mid[2] = {
- 0, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Mid[20] = {
- 0, 28, 246, 459, 667, 1045, 1523, 2337, 4337, 11347,
- 44231, 56709, 60781, 62243, 63161, 63969, 64608, 65062, 65502, 65535
-};
-
-const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrMid[4] = {
- WebRtcIsacfix_kPitchLagCdf1Mid,
- WebRtcIsacfix_kPitchLagCdf2Mid,
- WebRtcIsacfix_kPitchLagCdf3Mid,
- WebRtcIsacfix_kPitchLagCdf4Mid
-};
-
-/* size of first cdf table */
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeMid[1] = {
- 256
-};
-
-/* index limits and ranges */
-const WebRtc_Word16 WebRtcIsacfix_kLowerLimitMid[4] = {
- -280, -17, 0, -9
-};
-
-const WebRtc_Word16 WebRtcIsacfix_kUpperLimitMid[4] = {
- -40, 17, 0, 9
-};
-
-/* initial index for arithmetic decoder */
-const WebRtc_UWord16 WebRtcIsacfix_kInitIndMid[3] = {
- 18, 1, 10
-};
-
-/* mean values of pitch filter lags in Q10 */
-
-const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Mid[35] = {
- -17297, -16250, -15416, -14343, -13341, -12363, -11270,
- -10355, -9122, -8217, -7172, -6083, -5102, -4004, -3060,
- -1982, -952, -18, 935, 1976, 3040, 4032,
- 5082, 6065, 7257, 8202, 9264, 10225, 11242,
- 12234, 13337, 14336, 15374, 16187, 17347
-};
-
-
-const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Mid[19] = {
- -8811, -8081, -7203, -6003, -5057, -4025, -2983, -1964,
- -891, 29, 921, 1920, 2988, 4064, 5187, 6079, 7173, 8074, 8849
-};
-
-
-/* tables for use with large pitch gain */
-
-/* cdf for quantized pitch filter lags */
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Hi[511] = {
- 0, 7, 18, 33, 69, 105, 156, 228, 315, 612,
- 680, 691, 709, 724, 735, 738, 742, 746, 749, 753,
- 756, 760, 764, 774, 782, 785, 789, 796, 800, 803,
- 807, 814, 818, 822, 829, 832, 847, 854, 858, 869,
- 876, 883, 898, 908, 934, 977, 1010, 1050, 1060, 1064,
- 1075, 1078, 1086, 1089, 1093, 1104, 1111, 1122, 1133, 1136,
- 1151, 1162, 1183, 1209, 1252, 1281, 1339, 1364, 1386, 1401,
- 1411, 1415, 1426, 1430, 1433, 1440, 1448, 1455, 1462, 1477,
- 1487, 1495, 1502, 1506, 1509, 1516, 1524, 1531, 1535, 1542,
- 1553, 1556, 1578, 1589, 1611, 1625, 1639, 1643, 1654, 1665,
- 1672, 1687, 1694, 1705, 1708, 1719, 1730, 1744, 1752, 1759,
- 1791, 1795, 1820, 1867, 1886, 1915, 1936, 1943, 1965, 1987,
- 2041, 2099, 2161, 2175, 2200, 2211, 2226, 2233, 2244, 2251,
- 2266, 2280, 2287, 2298, 2309, 2316, 2331, 2342, 2356, 2378,
- 2403, 2418, 2447, 2497, 2544, 2602, 2863, 2895, 2903, 2935,
- 2950, 2971, 3004, 3011, 3018, 3029, 3040, 3062, 3087, 3127,
- 3152, 3170, 3199, 3243, 3293, 3322, 3340, 3377, 3402, 3427,
- 3474, 3518, 3543, 3579, 3601, 3637, 3659, 3706, 3731, 3760,
- 3818, 3847, 3869, 3901, 3920, 3952, 4068, 4169, 4220, 4271,
- 4524, 4571, 4604, 4632, 4672, 4730, 4777, 4806, 4857, 4904,
- 4951, 5002, 5031, 5060, 5107, 5150, 5212, 5266, 5331, 5382,
- 5432, 5490, 5544, 5610, 5700, 5762, 5812, 5874, 5972, 6022,
- 6091, 6163, 6232, 6305, 6402, 6540, 6685, 6880, 7090, 7271,
- 7379, 7452, 7542, 7625, 7687, 7770, 7843, 7911, 7966, 8024,
- 8096, 8190, 8252, 8320, 8411, 8501, 8585, 8639, 8751, 8842,
- 8918, 8986, 9066, 9127, 9203, 9269, 9345, 9406, 9464, 9536,
- 9612, 9667, 9735, 9844, 9931, 10036, 10119, 10199, 10260, 10358,
- 10441, 10514, 10666, 10734, 10872, 10951, 11053, 11125, 11223, 11324,
- 11516, 11664, 11737, 11816, 11892, 12008, 12120, 12200, 12280, 12392,
- 12490, 12576, 12685, 12812, 12917, 13003, 13108, 13210, 13300, 13384,
- 13470, 13579, 13673, 13771, 13879, 13999, 14136, 14201, 14368, 14614,
- 14759, 14867, 14958, 15030, 15121, 15189, 15280, 15385, 15461, 15555,
- 15653, 15768, 15884, 15971, 16069, 16145, 16210, 16279, 16380, 16463,
- 16539, 16615, 16688, 16818, 16919, 17017, 18041, 18338, 18523, 18649,
- 18790, 18917, 19047, 19167, 19315, 19460, 19601, 19731, 19858, 20068,
- 20173, 20318, 20466, 20625, 20741, 20911, 21045, 21201, 21396, 21588,
- 21816, 22022, 22305, 22547, 22786, 23072, 23322, 23600, 23879, 24168,
- 24433, 24769, 25120, 25511, 25895, 26289, 26792, 27219, 27683, 28077,
- 28566, 29094, 29546, 29977, 30491, 30991, 31573, 32105, 32594, 33173,
- 33788, 34497, 35181, 35833, 36488, 37255, 37921, 38645, 39275, 39894,
- 40505, 41167, 41790, 42431, 43096, 43723, 44385, 45134, 45858, 46607,
- 47349, 48091, 48768, 49405, 49955, 50555, 51167, 51985, 52611, 53078,
- 53494, 53965, 54435, 54996, 55601, 56125, 56563, 56838, 57244, 57566,
- 57967, 58297, 58771, 59093, 59419, 59647, 59886, 60143, 60461, 60693,
- 60917, 61170, 61416, 61634, 61891, 62122, 62310, 62455, 62632, 62839,
- 63103, 63436, 63639, 63805, 63906, 64015, 64192, 64355, 64475, 64558,
- 64663, 64742, 64811, 64865, 64916, 64956, 64981, 65025, 65068, 65115,
- 65195, 65314, 65419, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535
-};
-
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Hi[68] = {
- 0, 7, 11, 22, 37, 52, 56, 59, 81, 85,
- 89, 96, 115, 130, 137, 152, 170, 181, 193, 200,
- 207, 233, 237, 259, 289, 318, 363, 433, 592, 992,
- 1607, 3062, 6149, 12206, 25522, 48368, 58223, 61918, 63640, 64584,
- 64943, 65098, 65206, 65268, 65294, 65335, 65350, 65372, 65387, 65402,
- 65413, 65420, 65428, 65435, 65439, 65450, 65454, 65468, 65472, 65476,
- 65483, 65491, 65498, 65505, 65516, 65520, 65528, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Hi[2] = {
- 0, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Hi[35] = {
- 0, 7, 19, 30, 41, 48, 63, 74, 82, 96,
- 122, 152, 215, 330, 701, 2611, 10931, 48106, 61177, 64341,
- 65112, 65238, 65309, 65338, 65364, 65379, 65401, 65427, 65453,
- 65465, 65476, 65490, 65509, 65528, 65535
-};
-
-const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrHi[4] = {
- WebRtcIsacfix_kPitchLagCdf1Hi,
- WebRtcIsacfix_kPitchLagCdf2Hi,
- WebRtcIsacfix_kPitchLagCdf3Hi,
- WebRtcIsacfix_kPitchLagCdf4Hi
-};
-
-/* size of first cdf table */
-const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeHi[1] = {
- 512
-};
-
-/* index limits and ranges */
-const WebRtc_Word16 WebRtcIsacfix_kLowerLimitHi[4] = {
- -552, -34, 0, -16
-};
-
-const WebRtc_Word16 WebRtcIsacfix_kUpperLimitHi[4] = {
- -80, 32, 0, 17
-};
-
-/* initial index for arithmetic decoder */
-const WebRtc_UWord16 WebRtcIsacfix_kInitIndHi[3] = {
- 34, 1, 18
-};
-
-/* mean values of pitch filter lags */
-
-const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Hi[67] = {
- -17482, -16896, -16220, -15929, -15329, -14848, -14336, -13807, -13312, -12800, -12218, -11720,
- -11307, -10649, -10396, -9742, -9148, -8668, -8297, -7718, -7155, -6656, -6231, -5600, -5129,
- -4610, -4110, -3521, -3040, -2525, -2016, -1506, -995, -477, -5, 469, 991, 1510, 2025, 2526, 3079,
- 3555, 4124, 4601, 5131, 5613, 6194, 6671, 7140, 7645, 8207, 8601, 9132, 9728, 10359, 10752, 11302,
- 11776, 12288, 12687, 13204, 13759, 14295, 14810, 15360, 15764, 16350
-};
-
-
-const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Hi[34] = {
- -8175, -7659, -7205, -6684, -6215, -5651, -5180, -4566, -4087, -3536, -3096,
- -2532, -1990, -1482, -959, -440, 11, 451, 954, 1492, 2020, 2562, 3059,
- 3577, 4113, 4618, 5134, 5724, 6060, 6758, 7015, 7716, 8066, 8741
-};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_lag_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_lag_tables.h
deleted file mode 100644
index 9517c29..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/pitch_lag_tables.h
+++ /dev/null
@@ -1,103 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * pitch_lag_tables.h
- *
- * This file contains tables for the pitch filter side-info in the entropy coder.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_
-
-
-#include "typedefs.h"
-
-
-/********************* Pitch Filter Lag Coefficient Tables ************************/
-
-/* tables for use with small pitch gain */
-
-/* cdfs for quantized pitch lags */
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Lo[127];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Lo[20];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Lo[2];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Lo[10];
-
-extern const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrLo[4];
-
-/* size of first cdf table */
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeLo[1];
-
-/* index limits and ranges */
-extern const WebRtc_Word16 WebRtcIsacfix_kLowerLimitLo[4];
-extern const WebRtc_Word16 WebRtcIsacfix_kUpperLimitLo[4];
-
-/* initial index for arithmetic decoder */
-extern const WebRtc_UWord16 WebRtcIsacfix_kInitIndLo[3];
-
-/* mean values of pitch filter lags */
-extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Lo[19];
-extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Lo[9];
-
-
-
-/* tables for use with medium pitch gain */
-
-/* cdfs for quantized pitch lags */
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Mid[255];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Mid[36];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Mid[2];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Mid[20];
-
-extern const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrMid[4];
-
-/* size of first cdf table */
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeMid[1];
-
-/* index limits and ranges */
-extern const WebRtc_Word16 WebRtcIsacfix_kLowerLimitMid[4];
-extern const WebRtc_Word16 WebRtcIsacfix_kUpperLimitMid[4];
-
-/* initial index for arithmetic decoder */
-extern const WebRtc_UWord16 WebRtcIsacfix_kInitIndMid[3];
-
-/* mean values of pitch filter lags */
-extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Mid[35];
-extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Mid[19];
-
-
-/* tables for use with large pitch gain */
-
-/* cdfs for quantized pitch lags */
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf1Hi[511];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf2Hi[68];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf3Hi[2];
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagCdf4Hi[35];
-
-extern const WebRtc_UWord16 *WebRtcIsacfix_kPitchLagPtrHi[4];
-
-/* size of first cdf table */
-extern const WebRtc_UWord16 WebRtcIsacfix_kPitchLagSizeHi[1];
-
-/* index limits and ranges */
-extern const WebRtc_Word16 WebRtcIsacfix_kLowerLimitHi[4];
-extern const WebRtc_Word16 WebRtcIsacfix_kUpperLimitHi[4];
-
-/* initial index for arithmetic decoder */
-extern const WebRtc_UWord16 WebRtcIsacfix_kInitIndHi[3];
-
-/* mean values of pitch filter lags */
-extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag2Hi[67];
-extern const WebRtc_Word16 WebRtcIsacfix_kMeanLag4Hi[34];
-
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_PITCH_LAG_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/settings.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/settings.h
deleted file mode 100644
index da88ba2..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/settings.h
+++ /dev/null
@@ -1,205 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * settings.h
- *
- * Declaration of #defines used in the iSAC codec
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_
-
-
-/* sampling frequency (Hz) */
-#define FS 16000
-/* 1.5 times Sampling frequency */
-#define FS_1_HALF (WebRtc_UWord32) 24000
-/* Three times Sampling frequency */
-#define FS3 (WebRtc_UWord32) 48000
-/* Eight times Sampling frequency */
-#define FS8 (WebRtc_UWord32) 128000
-
-/* number of samples per frame (either 480 (30ms) or 960 (60ms)) */
-#define INITIAL_FRAMESAMPLES 960
-
-/* miliseconds */
-#define FRAMESIZE 30
-/* number of samples per frame processed in the encoder (30ms) */
-#define FRAMESAMPLES 480 /* ((FRAMESIZE*FS)/1000) */
-#define FRAMESAMPLES_HALF 240
-/* max number of samples per frame (= 60 ms frame) */
-#define MAX_FRAMESAMPLES 960
-/* number of samples per 10ms frame */
-#define FRAMESAMPLES_10ms 160 /* ((10*FS)/1000) */
-/* Number of samples per 1 ms */
-#define SAMPLES_PER_MSEC 16
-/* number of subframes */
-#define SUBFRAMES 6
-/* length of a subframe */
-#define UPDATE 80
-/* length of half a subframe (low/high band) */
-#define HALF_SUBFRAMELEN 40 /* (UPDATE/2) */
-/* samples of look ahead (in a half-band, so actually half the samples of look ahead @ FS) */
-#define QLOOKAHEAD 24 /* 3 ms */
-
-/* order of AR model in spectral entropy coder */
-#define AR_ORDER 6
-#define MAX_ORDER 13
-#define LEVINSON_MAX_ORDER 12
-
-/* window length (masking analysis) */
-#define WINLEN 256
-/* order of low-band pole filter used to approximate masking curve */
-#define ORDERLO 12
-/* order of hi-band pole filter used to approximate masking curve */
-#define ORDERHI 6
-
-#define KLT_NUM_AVG_GAIN 0
-#define KLT_NUM_AVG_SHAPE 0
-#define KLT_NUM_MODELS 3
-#define LPC_SHAPE_ORDER 18 /* (ORDERLO + ORDERHI) */
-
-#define KLT_ORDER_GAIN 12 /* (2 * SUBFRAMES) */
-#define KLT_ORDER_SHAPE 108 /* (LPC_SHAPE_ORDER * SUBFRAMES) */
-
-
-
-/* order for post_filter_bank */
-#define POSTQORDER 3
-/* order for pre-filterbank */
-#define QORDER 3
-/* for decimator */
-#define ALLPASSSECTIONS 2
-/* The number of composite all-pass filter factors */
-#define NUMBEROFCOMPOSITEAPSECTIONS 4
-
-/* The number of all-pass filter factors in an upper or lower channel*/
-#define NUMBEROFCHANNELAPSECTIONS 2
-
-
-
-#define DPMIN_Q10 -10240 /* -10.00 in Q10 */
-#define DPMAX_Q10 10240 /* 10.00 in Q10 */
-#define MINBITS_Q10 10240 /* 10.0 in Q10 */
-
-
-/* array size for byte stream in number of Word16. */
-#define STREAM_MAXW16 300 /* The old maximum size still needed for the decoding */
-#define STREAM_MAXW16_30MS 100 /* 100 Word16 = 200 bytes = 53.4 kbit/s @ 30 ms.framelength */
-#define STREAM_MAXW16_60MS 200 /* 200 Word16 = 400 bytes = 53.4 kbit/s @ 60 ms.framelength */
-
-
-/* storage size for bit counts */
-//#define BIT_COUNTER_SIZE 30
-/* maximum order of any AR model or filter */
-#define MAX_AR_MODEL_ORDER 12
-
-/* Maximum number of iterations allowed to limit payload size */
-#define MAX_PAYLOAD_LIMIT_ITERATION 1
-
-/* Bandwidth estimator */
-
-#define MIN_ISAC_BW 10000 /* Minimum bandwidth in bits per sec */
-#define MAX_ISAC_BW 32000 /* Maxmum bandwidth in bits per sec */
-#define MIN_ISAC_MD 5 /* Minimum Max Delay in ?? */
-#define MAX_ISAC_MD 25 /* Maxmum Max Delay in ?? */
-#define DELAY_CORRECTION_MAX 717
-#define DELAY_CORRECTION_MED 819
-#define Thld_30_60 18000
-#define Thld_60_30 27000
-
-/* assumed header size; we don't know the exact number (header compression may be used) */
-#define HEADER_SIZE 35 /* bytes */
-#define INIT_FRAME_LEN 60
-#define INIT_BN_EST 20000
-#define INIT_BN_EST_Q7 2560000 /* 20 kbps in Q7 */
-#define INIT_REC_BN_EST_Q5 789312 /* INIT_BN_EST + INIT_HDR_RATE in Q5 */
-
-/* 8738 in Q18 is ~ 1/30 */
-/* #define INIT_HDR_RATE (((HEADER_SIZE * 8 * 1000) * 8738) >> NUM_BITS_TO_SHIFT (INIT_FRAME_LEN)) */
-#define INIT_HDR_RATE 4666
-/* number of packets in a row for a high rate burst */
-#define BURST_LEN 3
-/* ms, max time between two full bursts */
-#define BURST_INTERVAL 800
-/* number of packets in a row for initial high rate burst */
-#define INIT_BURST_LEN 5
-/* bits/s, rate for the first BURST_LEN packets */
-#define INIT_RATE 10240000 /* INIT_BN_EST in Q9 */
-
-
-/* For pitch analysis */
-#define PITCH_FRAME_LEN 240 /* (FRAMESAMPLES/2) 30 ms */
-#define PITCH_MAX_LAG 140 /* 57 Hz */
-#define PITCH_MIN_LAG 20 /* 400 Hz */
-#define PITCH_MIN_LAG_Q8 5120 /* 256 * PITCH_MIN_LAG */
-#define OFFSET_Q8 768 /* 256 * 3 */
-
-#define PITCH_MAX_GAIN_Q12 1843 /* 0.45 */
-#define PITCH_LAG_SPAN2 65 /* (PITCH_MAX_LAG/2-PITCH_MIN_LAG/2+5) */
-#define PITCH_CORR_LEN2 60 /* 15 ms */
-#define PITCH_CORR_STEP2 60 /* (PITCH_FRAME_LEN/4) */
-#define PITCH_SUBFRAMES 4
-#define PITCH_SUBFRAME_LEN 60 /* (PITCH_FRAME_LEN/PITCH_SUBFRAMES) */
-
-/* For pitch filter */
-#define PITCH_BUFFSIZE 190 /* (PITCH_MAX_LAG + 50) Extra 50 for fraction and LP filters */
-#define PITCH_INTBUFFSIZE 430 /* (PITCH_FRAME_LEN+PITCH_BUFFSIZE) */
-#define PITCH_FRACS 8
-#define PITCH_FRACORDER 9
-#define PITCH_DAMPORDER 5
-
-
-/* Order of high pass filter */
-#define HPORDER 2
-
-
-/* PLC */
-#define DECAY_RATE 10 /* Q15, 20% of decay every lost frame apllied linearly sample by sample*/
-#define PLC_WAS_USED 1
-#define PLC_NOT_USED 3
-#define RECOVERY_OVERLAP 80
-#define RESAMP_RES 256
-#define RESAMP_RES_BIT 8
-
-
-
-/* Define Error codes */
-/* 6000 General */
-#define ISAC_MEMORY_ALLOCATION_FAILED 6010
-#define ISAC_MODE_MISMATCH 6020
-#define ISAC_DISALLOWED_BOTTLENECK 6030
-#define ISAC_DISALLOWED_FRAME_LENGTH 6040
-/* 6200 Bandwidth estimator */
-#define ISAC_RANGE_ERROR_BW_ESTIMATOR 6240
-/* 6400 Encoder */
-#define ISAC_ENCODER_NOT_INITIATED 6410
-#define ISAC_DISALLOWED_CODING_MODE 6420
-#define ISAC_DISALLOWED_FRAME_MODE_ENCODER 6430
-#define ISAC_DISALLOWED_BITSTREAM_LENGTH 6440
-#define ISAC_PAYLOAD_LARGER_THAN_LIMIT 6450
-/* 6600 Decoder */
-#define ISAC_DECODER_NOT_INITIATED 6610
-#define ISAC_EMPTY_PACKET 6620
-#define ISAC_DISALLOWED_FRAME_MODE_DECODER 6630
-#define ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH 6640
-#define ISAC_RANGE_ERROR_DECODE_BANDWIDTH 6650
-#define ISAC_RANGE_ERROR_DECODE_PITCH_GAIN 6660
-#define ISAC_RANGE_ERROR_DECODE_PITCH_LAG 6670
-#define ISAC_RANGE_ERROR_DECODE_LPC 6680
-#define ISAC_RANGE_ERROR_DECODE_SPECTRUM 6690
-#define ISAC_LENGTH_MISMATCH 6730
-/* 6800 Call setup formats */
-#define ISAC_INCOMPATIBLE_FORMATS 6810
-
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SETTINGS_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/spectrum_ar_model_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/spectrum_ar_model_tables.c
deleted file mode 100644
index 81b932f..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/spectrum_ar_model_tables.c
+++ /dev/null
@@ -1,193 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * spectrum_ar_model_tables.c
- *
- * This file contains tables with AR coefficients, Gain coefficients
- * and cosine tables.
- *
- */
-
-#include "spectrum_ar_model_tables.h"
-#include "settings.h"
-
-/********************* AR Coefficient Tables ************************/
-
-/* cdf for quantized reflection coefficient 1 */
-const WebRtc_UWord16 WebRtcIsacfix_kRc1Cdf[12] = {
- 0, 2, 4, 129, 7707, 57485, 65495, 65527, 65529, 65531,
- 65533, 65535
-};
-
-/* cdf for quantized reflection coefficient 2 */
-const WebRtc_UWord16 WebRtcIsacfix_kRc2Cdf[12] = {
- 0, 2, 4, 7, 531, 25298, 64525, 65526, 65529, 65531,
- 65533, 65535
-};
-
-/* cdf for quantized reflection coefficient 3 */
-const WebRtc_UWord16 WebRtcIsacfix_kRc3Cdf[12] = {
- 0, 2, 4, 6, 620, 22898, 64843, 65527, 65529, 65531,
- 65533, 65535
-};
-
-/* cdf for quantized reflection coefficient 4 */
-const WebRtc_UWord16 WebRtcIsacfix_kRc4Cdf[12] = {
- 0, 2, 4, 6, 35, 10034, 60733, 65506, 65529, 65531,
- 65533, 65535
-};
-
-/* cdf for quantized reflection coefficient 5 */
-const WebRtc_UWord16 WebRtcIsacfix_kRc5Cdf[12] = {
- 0, 2, 4, 6, 36, 7567, 56727, 65385, 65529, 65531,
- 65533, 65535
-};
-
-/* cdf for quantized reflection coefficient 6 */
-const WebRtc_UWord16 WebRtcIsacfix_kRc6Cdf[12] = {
- 0, 2, 4, 6, 14, 6579, 57360, 65409, 65529, 65531,
- 65533, 65535
-};
-
-/* representation levels for quantized reflection coefficient 1 */
-const WebRtc_Word16 WebRtcIsacfix_kRc1Levels[11] = {
- -32104, -29007, -23202, -15496, -9279, -2577, 5934, 17535, 24512, 29503, 32104
-};
-
-/* representation levels for quantized reflection coefficient 2 */
-const WebRtc_Word16 WebRtcIsacfix_kRc2Levels[11] = {
- -32104, -29503, -23494, -15261, -7309, -1399, 6158, 16381, 24512, 29503, 32104
-};
-
-/* representation levels for quantized reflection coefficient 3 */
-const WebRtc_Word16 WebRtcIsacfix_kRc3Levels[11] = {
- -32104, -29503, -23157, -15186, -7347, -1359, 5829, 17535, 24512, 29503, 32104
-};
-
-/* representation levels for quantized reflection coefficient 4 */
-const WebRtc_Word16 WebRtcIsacfix_kRc4Levels[11] = {
- -32104, -29503, -24512, -15362, -6665, -342, 6596, 14585, 24512, 29503, 32104
-};
-
-/* representation levels for quantized reflection coefficient 5 */
-const WebRtc_Word16 WebRtcIsacfix_kRc5Levels[11] = {
- -32104, -29503, -24512, -15005, -6564, -106, 7123, 14920, 24512, 29503, 32104
-};
-
-/* representation levels for quantized reflection coefficient 6 */
-const WebRtc_Word16 WebRtcIsacfix_kRc6Levels[11] = {
- -32104, -29503, -24512, -15096, -6656, -37, 7036, 14847, 24512, 29503, 32104
-};
-
-/* quantization boundary levels for reflection coefficients */
-const WebRtc_Word16 WebRtcIsacfix_kRcBound[12] = {
- -32768, -31441, -27566, -21458, -13612, -4663,
- 4663, 13612, 21458, 27566, 31441, 32767
-};
-
-/* initial index for AR reflection coefficient quantizer and cdf table search */
-const WebRtc_UWord16 WebRtcIsacfix_kRcInitInd[6] = {
- 5, 5, 5, 5, 5, 5
-};
-
-/* pointers to AR cdf tables */
-const WebRtc_UWord16 *WebRtcIsacfix_kRcCdfPtr[AR_ORDER] = {
- WebRtcIsacfix_kRc1Cdf,
- WebRtcIsacfix_kRc2Cdf,
- WebRtcIsacfix_kRc3Cdf,
- WebRtcIsacfix_kRc4Cdf,
- WebRtcIsacfix_kRc5Cdf,
- WebRtcIsacfix_kRc6Cdf
-};
-
-/* pointers to AR representation levels tables */
-const WebRtc_Word16 *WebRtcIsacfix_kRcLevPtr[AR_ORDER] = {
- WebRtcIsacfix_kRc1Levels,
- WebRtcIsacfix_kRc2Levels,
- WebRtcIsacfix_kRc3Levels,
- WebRtcIsacfix_kRc4Levels,
- WebRtcIsacfix_kRc5Levels,
- WebRtcIsacfix_kRc6Levels
-};
-
-
-/******************** GAIN Coefficient Tables ***********************/
-
-/* cdf for Gain coefficient */
-const WebRtc_UWord16 WebRtcIsacfix_kGainCdf[19] = {
- 0, 2, 4, 6, 8, 10, 12, 14, 16, 1172,
- 11119, 29411, 51699, 64445, 65527, 65529, 65531, 65533, 65535
-};
-
-/* representation levels for quantized squared Gain coefficient */
-const WebRtc_Word32 WebRtcIsacfix_kGain2Lev[18] = {
- 128, 128, 128, 128, 128, 215, 364, 709, 1268,
- 1960, 3405, 6078, 11286, 17827, 51918, 134498, 487432, 2048000
-};
-
-/* quantization boundary levels for squared Gain coefficient */
-const WebRtc_Word32 WebRtcIsacfix_kGain2Bound[19] = {
- 0, 21, 35, 59, 99, 166, 280, 475, 815, 1414,
- 2495, 4505, 8397, 16405, 34431, 81359, 240497, 921600, 0x7FFFFFFF
-};
-
-/* pointers to Gain cdf table */
-const WebRtc_UWord16 *WebRtcIsacfix_kGainPtr[1] = {
- WebRtcIsacfix_kGainCdf
-};
-
-/* gain initial index for gain quantizer and cdf table search */
-const WebRtc_UWord16 WebRtcIsacfix_kGainInitInd[1] = {
- 11
-};
-
-
-/************************* Cosine Tables ****************************/
-
-/* cosine table */
-const WebRtc_Word16 WebRtcIsacfix_kCos[6][60] = {
- { 512, 512, 511, 510, 508, 507, 505, 502, 499, 496,
- 493, 489, 485, 480, 476, 470, 465, 459, 453, 447,
- 440, 433, 426, 418, 410, 402, 394, 385, 376, 367,
- 357, 348, 338, 327, 317, 306, 295, 284, 273, 262,
- 250, 238, 226, 214, 202, 190, 177, 165, 152, 139,
- 126, 113, 100, 87, 73, 60, 47, 33, 20, 7 },
- { 512, 510, 508, 503, 498, 491, 483, 473, 462, 450,
- 437, 422, 406, 389, 371, 352, 333, 312, 290, 268,
- 244, 220, 196, 171, 145, 120, 93, 67, 40, 13,
- -13, -40, -67, -93, -120, -145, -171, -196, -220, -244,
- -268, -290, -312, -333, -352, -371, -389, -406, -422, -437,
- -450, -462, -473, -483, -491, -498, -503, -508, -510, -512 },
- { 512, 508, 502, 493, 480, 465, 447, 426, 402, 376,
- 348, 317, 284, 250, 214, 177, 139, 100, 60, 20,
- -20, -60, -100, -139, -177, -214, -250, -284, -317, -348,
- -376, -402, -426, -447, -465, -480, -493, -502, -508, -512,
- -512, -508, -502, -493, -480, -465, -447, -426, -402, -376,
- -348, -317, -284, -250, -214, -177, -139, -100, -60, -20 },
- { 511, 506, 495, 478, 456, 429, 398, 362, 322, 279,
- 232, 183, 133, 80, 27, -27, -80, -133, -183, -232,
- -279, -322, -362, -398, -429, -456, -478, -495, -506, -511,
- -511, -506, -495, -478, -456, -429, -398, -362, -322, -279,
- -232, -183, -133, -80, -27, 27, 80, 133, 183, 232,
- 279, 322, 362, 398, 429, 456, 478, 495, 506, 511 },
- { 511, 502, 485, 459, 426, 385, 338, 284, 226, 165,
- 100, 33, -33, -100, -165, -226, -284, -338, -385, -426,
- -459, -485, -502, -511, -511, -502, -485, -459, -426, -385,
- -338, -284, -226, -165, -100, -33, 33, 100, 165, 226,
- 284, 338, 385, 426, 459, 485, 502, 511, 511, 502,
- 485, 459, 426, 385, 338, 284, 226, 165, 100, 33 },
- { 510, 498, 473, 437, 389, 333, 268, 196, 120, 40,
- -40, -120, -196, -268, -333, -389, -437, -473, -498, -510,
- -510, -498, -473, -437, -389, -333, -268, -196, -120, -40,
- 40, 120, 196, 268, 333, 389, 437, 473, 498, 510,
- 510, 498, 473, 437, 389, 333, 268, 196, 120, 40,
- -40, -120, -196, -268, -333, -389, -437, -473, -498, -510 }
-};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/spectrum_ar_model_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/spectrum_ar_model_tables.h
deleted file mode 100644
index b506d0e..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/spectrum_ar_model_tables.h
+++ /dev/null
@@ -1,96 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * spectrum_ar_model_tables.h
- *
- * This file contains definitions of tables with AR coefficients,
- * Gain coefficients and cosine tables.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
-
-#include "typedefs.h"
-#include "settings.h"
-
-
-/********************* AR Coefficient Tables ************************/
-/* cdf for quantized reflection coefficient 1 */
-extern const WebRtc_UWord16 WebRtcIsacfix_kRc1Cdf[12];
-
-/* cdf for quantized reflection coefficient 2 */
-extern const WebRtc_UWord16 WebRtcIsacfix_kRc2Cdf[12];
-
-/* cdf for quantized reflection coefficient 3 */
-extern const WebRtc_UWord16 WebRtcIsacfix_kRc3Cdf[12];
-
-/* cdf for quantized reflection coefficient 4 */
-extern const WebRtc_UWord16 WebRtcIsacfix_kRc4Cdf[12];
-
-/* cdf for quantized reflection coefficient 5 */
-extern const WebRtc_UWord16 WebRtcIsacfix_kRc5Cdf[12];
-
-/* cdf for quantized reflection coefficient 6 */
-extern const WebRtc_UWord16 WebRtcIsacfix_kRc6Cdf[12];
-
-/* representation levels for quantized reflection coefficient 1 */
-extern const WebRtc_Word16 WebRtcIsacfix_kRc1Levels[11];
-
-/* representation levels for quantized reflection coefficient 2 */
-extern const WebRtc_Word16 WebRtcIsacfix_kRc2Levels[11];
-
-/* representation levels for quantized reflection coefficient 3 */
-extern const WebRtc_Word16 WebRtcIsacfix_kRc3Levels[11];
-
-/* representation levels for quantized reflection coefficient 4 */
-extern const WebRtc_Word16 WebRtcIsacfix_kRc4Levels[11];
-
-/* representation levels for quantized reflection coefficient 5 */
-extern const WebRtc_Word16 WebRtcIsacfix_kRc5Levels[11];
-
-/* representation levels for quantized reflection coefficient 6 */
-extern const WebRtc_Word16 WebRtcIsacfix_kRc6Levels[11];
-
-/* quantization boundary levels for reflection coefficients */
-extern const WebRtc_Word16 WebRtcIsacfix_kRcBound[12];
-
-/* initial indices for AR reflection coefficient quantizer and cdf table search */
-extern const WebRtc_UWord16 WebRtcIsacfix_kRcInitInd[AR_ORDER];
-
-/* pointers to AR cdf tables */
-extern const WebRtc_UWord16 *WebRtcIsacfix_kRcCdfPtr[AR_ORDER];
-
-/* pointers to AR representation levels tables */
-extern const WebRtc_Word16 *WebRtcIsacfix_kRcLevPtr[AR_ORDER];
-
-
-/******************** GAIN Coefficient Tables ***********************/
-/* cdf for Gain coefficient */
-extern const WebRtc_UWord16 WebRtcIsacfix_kGainCdf[19];
-
-/* representation levels for quantized Gain coefficient */
-extern const WebRtc_Word32 WebRtcIsacfix_kGain2Lev[18];
-
-/* squared quantization boundary levels for Gain coefficient */
-extern const WebRtc_Word32 WebRtcIsacfix_kGain2Bound[19];
-
-/* pointer to Gain cdf table */
-extern const WebRtc_UWord16 *WebRtcIsacfix_kGainPtr[1];
-
-/* Gain initial index for gain quantizer and cdf table search */
-extern const WebRtc_UWord16 WebRtcIsacfix_kGainInitInd[1];
-
-/************************* Cosine Tables ****************************/
-/* Cosine table */
-extern const WebRtc_Word16 WebRtcIsacfix_kCos[6][60];
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/structs.h b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/structs.h
deleted file mode 100644
index 54dffa9..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/structs.h
+++ /dev/null
@@ -1,382 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * structs.h
- *
- * This header file contains all the structs used in the ISAC codec
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_
-
-
-#include "typedefs.h"
-#include "signal_processing_library.h"
-#include "settings.h"
-
-/* Bitstream struct for decoder */
-typedef struct Bitstreamstruct_dec {
-
- WebRtc_UWord16 *stream; /* Pointer to bytestream to decode */
- WebRtc_UWord32 W_upper; /* Upper boundary of interval W */
- WebRtc_UWord32 streamval;
- WebRtc_UWord16 stream_index; /* Index to the current position in bytestream */
- WebRtc_Word16 full; /* 0 - first byte in memory filled, second empty*/
- /* 1 - both bytes are empty (we just filled the previous memory */
-
-} Bitstr_dec;
-
-/* Bitstream struct for encoder */
-typedef struct Bitstreamstruct_enc {
-
- WebRtc_UWord16 stream[STREAM_MAXW16_60MS]; /* Vector for adding encoded bytestream */
- WebRtc_UWord32 W_upper; /* Upper boundary of interval W */
- WebRtc_UWord32 streamval;
- WebRtc_UWord16 stream_index; /* Index to the current position in bytestream */
- WebRtc_Word16 full; /* 0 - first byte in memory filled, second empty*/
- /* 1 - both bytes are empty (we just filled the previous memory */
-
-} Bitstr_enc;
-
-
-typedef struct {
-
- WebRtc_Word16 DataBufferLoQ0[WINLEN];
- WebRtc_Word16 DataBufferHiQ0[WINLEN];
-
- WebRtc_Word32 CorrBufLoQQ[ORDERLO+1];
- WebRtc_Word32 CorrBufHiQQ[ORDERHI+1];
-
- WebRtc_Word16 CorrBufLoQdom[ORDERLO+1];
- WebRtc_Word16 CorrBufHiQdom[ORDERHI+1];
-
- WebRtc_Word32 PreStateLoGQ15[ORDERLO+1];
- WebRtc_Word32 PreStateHiGQ15[ORDERHI+1];
-
- WebRtc_UWord32 OldEnergy;
-
-} MaskFiltstr_enc;
-
-
-
-typedef struct {
-
- WebRtc_Word16 PostStateLoGQ0[ORDERLO+1];
- WebRtc_Word16 PostStateHiGQ0[ORDERHI+1];
-
- WebRtc_UWord32 OldEnergy;
-
-} MaskFiltstr_dec;
-
-
-
-
-
-
-
-
-typedef struct {
-
- //state vectors for each of the two analysis filters
-
- WebRtc_Word32 INSTAT1_fix[2*(QORDER-1)];
- WebRtc_Word32 INSTAT2_fix[2*(QORDER-1)];
- WebRtc_Word16 INLABUF1_fix[QLOOKAHEAD];
- WebRtc_Word16 INLABUF2_fix[QLOOKAHEAD];
-
- /* High pass filter */
- WebRtc_Word32 HPstates_fix[HPORDER];
-
-} PreFiltBankstr;
-
-
-typedef struct {
-
- //state vectors for each of the two analysis filters
- WebRtc_Word32 STATE_0_LOWER_fix[2*POSTQORDER];
- WebRtc_Word32 STATE_0_UPPER_fix[2*POSTQORDER];
-
- /* High pass filter */
-
- WebRtc_Word32 HPstates1_fix[HPORDER];
- WebRtc_Word32 HPstates2_fix[HPORDER];
-
-} PostFiltBankstr;
-
-typedef struct {
-
-
- /* data buffer for pitch filter */
- WebRtc_Word16 ubufQQ[PITCH_BUFFSIZE];
-
- /* low pass state vector */
- WebRtc_Word16 ystateQQ[PITCH_DAMPORDER];
-
- /* old lag and gain */
- WebRtc_Word16 oldlagQ7;
- WebRtc_Word16 oldgainQ12;
-
-} PitchFiltstr;
-
-
-
-typedef struct {
-
- //for inital estimator
- WebRtc_Word16 dec_buffer16[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2];
- WebRtc_Word32 decimator_state32[2*ALLPASSSECTIONS+1];
- WebRtc_Word16 inbuf[QLOOKAHEAD];
-
- PitchFiltstr PFstr_wght;
- PitchFiltstr PFstr;
-
-
-} PitchAnalysisStruct;
-
-
-typedef struct {
- /* Parameters used in PLC to avoid re-computation */
-
- /* --- residual signals --- */
- WebRtc_Word16 prevPitchInvIn[FRAMESAMPLES/2];
- WebRtc_Word16 prevPitchInvOut[PITCH_MAX_LAG + 10]; // [FRAMESAMPLES/2]; save 90
- WebRtc_Word32 prevHP[PITCH_MAX_LAG + 10]; // [FRAMESAMPLES/2]; save 90
-
-
- WebRtc_Word16 decayCoeffPriodic; /* how much to supress a sample */
- WebRtc_Word16 decayCoeffNoise;
- WebRtc_Word16 used; /* if PLC is used */
-
-
- WebRtc_Word16 *lastPitchLP; // [FRAMESAMPLES/2]; saved 240;
-
-
- /* --- LPC side info --- */
- WebRtc_Word16 lofilt_coefQ15[ ORDERLO ];
- WebRtc_Word16 hifilt_coefQ15[ ORDERHI ];
- WebRtc_Word32 gain_lo_hiQ17[2];
-
- /* --- LTP side info --- */
- WebRtc_Word16 AvgPitchGain_Q12;
- WebRtc_Word16 lastPitchGain_Q12;
- WebRtc_Word16 lastPitchLag_Q7;
-
- /* --- Add-overlap in recovery packet --- */
- WebRtc_Word16 overlapLP[ RECOVERY_OVERLAP ]; // [FRAMESAMPLES/2]; saved 160
-
- WebRtc_Word16 pitchCycles;
- WebRtc_Word16 A;
- WebRtc_Word16 B;
- WebRtc_Word16 pitchIndex;
- WebRtc_Word16 stretchLag;
- WebRtc_Word16 *prevPitchLP; // [ FRAMESAMPLES/2 ]; saved 240
- WebRtc_Word16 seed;
-
- WebRtc_Word16 std;
-} PLCstr;
-
-
-
-/* Have instance of struct together with other iSAC structs */
-typedef struct {
-
- WebRtc_Word16 prevFrameSizeMs; /* Previous frame size (in ms) */
- WebRtc_UWord16 prevRtpNumber; /* Previous RTP timestamp from received packet */
- /* (in samples relative beginning) */
- WebRtc_UWord32 prevSendTime; /* Send time for previous packet, from RTP header */
- WebRtc_UWord32 prevArrivalTime; /* Arrival time for previous packet (in ms using timeGetTime()) */
- WebRtc_UWord16 prevRtpRate; /* rate of previous packet, derived from RTP timestamps (in bits/s) */
- WebRtc_UWord32 lastUpdate; /* Time since the last update of the Bottle Neck estimate (in samples) */
- WebRtc_UWord32 lastReduction; /* Time sinse the last reduction (in samples) */
- WebRtc_Word32 countUpdates; /* How many times the estimate was update in the beginning */
-
- /* The estimated bottle neck rate from there to here (in bits/s) */
- WebRtc_UWord32 recBw;
- WebRtc_UWord32 recBwInv;
- WebRtc_UWord32 recBwAvg;
- WebRtc_UWord32 recBwAvgQ;
-
- WebRtc_UWord32 minBwInv;
- WebRtc_UWord32 maxBwInv;
-
- /* The estimated mean absolute jitter value, as seen on this side (in ms) */
- WebRtc_Word32 recJitter;
- WebRtc_Word32 recJitterShortTerm;
- WebRtc_Word32 recJitterShortTermAbs;
- WebRtc_Word32 recMaxDelay;
- WebRtc_Word32 recMaxDelayAvgQ;
-
-
- WebRtc_Word16 recHeaderRate; /* (assumed) bitrate for headers (bps) */
-
- WebRtc_UWord32 sendBwAvg; /* The estimated bottle neck rate from here to there (in bits/s) */
- WebRtc_Word32 sendMaxDelayAvg; /* The estimated mean absolute jitter value, as seen on the other siee (in ms) */
-
-
- WebRtc_Word16 countRecPkts; /* number of packets received since last update */
- WebRtc_Word16 highSpeedRec; /* flag for marking that a high speed network has been detected downstream */
-
- /* number of consecutive pkts sent during which the bwe estimate has
- remained at a value greater than the downstream threshold for determining highspeed network */
- WebRtc_Word16 countHighSpeedRec;
-
- /* flag indicating bwe should not adjust down immediately for very late pckts */
- WebRtc_Word16 inWaitPeriod;
-
- /* variable holding the time of the start of a window of time when
- bwe should not adjust down immediately for very late pckts */
- WebRtc_UWord32 startWaitPeriod;
-
- /* number of consecutive pkts sent during which the bwe estimate has
- remained at a value greater than the upstream threshold for determining highspeed network */
- WebRtc_Word16 countHighSpeedSent;
-
- /* flag indicated the desired number of packets over threshold rate have been sent and
- bwe will assume the connection is over broadband network */
- WebRtc_Word16 highSpeedSend;
-
-
-
-
-} BwEstimatorstr;
-
-
-typedef struct {
-
- /* boolean, flags if previous packet exceeded B.N. */
- WebRtc_Word16 PrevExceed;
- /* ms */
- WebRtc_Word16 ExceedAgo;
- /* packets left to send in current burst */
- WebRtc_Word16 BurstCounter;
- /* packets */
- WebRtc_Word16 InitCounter;
- /* ms remaining in buffer when next packet will be sent */
- WebRtc_Word16 StillBuffered;
-
-} RateModel;
-
-/* The following strutc is used to store data from encoding, to make it
- fast and easy to construct a new bitstream with a different Bandwidth
- estimate. All values (except framelength and minBytes) is double size to
- handle 60 ms of data.
-*/
-typedef struct {
-
- /* Used to keep track of if it is first or second part of 60 msec packet */
- int startIdx;
-
- /* Frame length in samples */
- WebRtc_Word16 framelength;
-
- /* Pitch Gain */
- WebRtc_Word16 pitchGain_index[2];
-
- /* Pitch Lag */
- WebRtc_Word32 meanGain[2];
- WebRtc_Word16 pitchIndex[PITCH_SUBFRAMES*2];
-
- /* LPC */
- WebRtc_Word32 LPCcoeffs_g[12*2]; /* KLT_ORDER_GAIN = 12 */
- WebRtc_Word16 LPCindex_s[108*2]; /* KLT_ORDER_SHAPE = 108 */
- WebRtc_Word16 LPCindex_g[12*2]; /* KLT_ORDER_GAIN = 12 */
-
- /* Encode Spec */
- WebRtc_Word16 fre[FRAMESAMPLES];
- WebRtc_Word16 fim[FRAMESAMPLES];
- WebRtc_Word16 AvgPitchGain[2];
-
- /* Used in adaptive mode only */
- int minBytes;
-
-} ISAC_SaveEncData_t;
-
-typedef struct {
-
- Bitstr_enc bitstr_obj;
- MaskFiltstr_enc maskfiltstr_obj;
- PreFiltBankstr prefiltbankstr_obj;
- PitchFiltstr pitchfiltstr_obj;
- PitchAnalysisStruct pitchanalysisstr_obj;
- RateModel rate_data_obj;
-
- WebRtc_Word16 buffer_index;
- WebRtc_Word16 current_framesamples;
-
- WebRtc_Word16 data_buffer_fix[FRAMESAMPLES]; // the size was MAX_FRAMESAMPLES
-
- WebRtc_Word16 frame_nb;
- WebRtc_Word16 BottleNeck;
- WebRtc_Word16 MaxDelay;
- WebRtc_Word16 new_framelength;
- WebRtc_Word16 s2nr;
- WebRtc_UWord16 MaxBits;
-
- WebRtc_Word16 bitstr_seed;
-#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
- PostFiltBankstr interpolatorstr_obj;
-#endif
-
- ISAC_SaveEncData_t *SaveEnc_ptr;
- WebRtc_Word16 payloadLimitBytes30; /* Maximum allowed number of bits for a 30 msec packet */
- WebRtc_Word16 payloadLimitBytes60; /* Maximum allowed number of bits for a 30 msec packet */
- WebRtc_Word16 maxPayloadBytes; /* Maximum allowed number of bits for both 30 and 60 msec packet */
- WebRtc_Word16 maxRateInBytes; /* Maximum allowed rate in bytes per 30 msec packet */
- WebRtc_Word16 enforceFrameSize; /* If set iSAC will never change packet size */
-
-} ISACFIX_EncInst_t;
-
-
-typedef struct {
-
- Bitstr_dec bitstr_obj;
- MaskFiltstr_dec maskfiltstr_obj;
- PostFiltBankstr postfiltbankstr_obj;
- PitchFiltstr pitchfiltstr_obj;
- PLCstr plcstr_obj; /* TS; for packet loss concealment */
-
-#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
- PreFiltBankstr decimatorstr_obj;
-#endif
-
-} ISACFIX_DecInst_t;
-
-
-
-typedef struct {
-
- ISACFIX_EncInst_t ISACenc_obj;
- ISACFIX_DecInst_t ISACdec_obj;
- BwEstimatorstr bwestimator_obj;
- WebRtc_Word16 CodingMode; /* 0 = adaptive; 1 = instantaneous */
- WebRtc_Word16 errorcode;
- WebRtc_Word16 initflag; /* 0 = nothing initiated; 1 = encoder or decoder */
- /* not initiated; 2 = all initiated */
-} ISACFIX_SubStruct;
-
-
-typedef struct {
- WebRtc_Word32 lpcGains[12]; /* 6 lower-band & 6 upper-band we may need to double it for 60*/
- /* */
- WebRtc_UWord32 W_upper; /* Upper boundary of interval W */
- WebRtc_UWord32 streamval;
- WebRtc_UWord16 stream_index; /* Index to the current position in bytestream */
- WebRtc_Word16 full; /* 0 - first byte in memory filled, second empty*/
- /* 1 - both bytes are empty (we just filled the previous memory */
- WebRtc_UWord16 beforeLastWord;
- WebRtc_UWord16 lastWord;
-} transcode_obj;
-
-
-//Bitstr_enc myBitStr;
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_FIX_SOURCE_STRUCTS_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/transform.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/transform.c
deleted file mode 100644
index 56ef9f2..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/source/transform.c
+++ /dev/null
@@ -1,296 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * WebRtcIsacfix_kTransform.c
- *
- * Transform functions
- *
- */
-
-#include "fft.h"
-#include "codec.h"
-#include "settings.h"
-
-
-/* Cosine table 1 in Q14 */
-static const WebRtc_Word16 kCosTab1[FRAMESAMPLES/2] = {
- 16384, 16383, 16378, 16371, 16362, 16349, 16333, 16315, 16294, 16270,
- 16244, 16214, 16182, 16147, 16110, 16069, 16026, 15980, 15931, 15880,
- 15826, 15769, 15709, 15647, 15582, 15515, 15444, 15371, 15296, 15218,
- 15137, 15053, 14968, 14879, 14788, 14694, 14598, 14500, 14399, 14295,
- 14189, 14081, 13970, 13856, 13741, 13623, 13502, 13380, 13255, 13128,
- 12998, 12867, 12733, 12597, 12458, 12318, 12176, 12031, 11885, 11736,
- 11585, 11433, 11278, 11121, 10963, 10803, 10641, 10477, 10311, 10143,
- 9974, 9803, 9630, 9456, 9280, 9102, 8923, 8743, 8561, 8377,
- 8192, 8006, 7818, 7629, 7438, 7246, 7053, 6859, 6664, 6467,
- 6270, 6071, 5872, 5671, 5469, 5266, 5063, 4859, 4653, 4447,
- 4240, 4033, 3825, 3616, 3406, 3196, 2986, 2775, 2563, 2351,
- 2139, 1926, 1713, 1499, 1285, 1072, 857, 643, 429, 214,
- 0, -214, -429, -643, -857, -1072, -1285, -1499, -1713, -1926,
- -2139, -2351, -2563, -2775, -2986, -3196, -3406, -3616, -3825, -4033,
- -4240, -4447, -4653, -4859, -5063, -5266, -5469, -5671, -5872, -6071,
- -6270, -6467, -6664, -6859, -7053, -7246, -7438, -7629, -7818, -8006,
- -8192, -8377, -8561, -8743, -8923, -9102, -9280, -9456, -9630, -9803,
- -9974, -10143, -10311, -10477, -10641, -10803, -10963, -11121, -11278, -11433,
- -11585, -11736, -11885, -12031, -12176, -12318, -12458, -12597, -12733, -12867,
- -12998, -13128, -13255, -13380, -13502, -13623, -13741, -13856, -13970, -14081,
- -14189, -14295, -14399, -14500, -14598, -14694, -14788, -14879, -14968, -15053,
- -15137, -15218, -15296, -15371, -15444, -15515, -15582, -15647, -15709, -15769,
- -15826, -15880, -15931, -15980, -16026, -16069, -16110, -16147, -16182, -16214,
- -16244, -16270, -16294, -16315, -16333, -16349, -16362, -16371, -16378, -16383
-};
-
-
-/* Sine table 1 in Q14 */
-static const WebRtc_Word16 kSinTab1[FRAMESAMPLES/2] = {
- 0, 214, 429, 643, 857, 1072, 1285, 1499, 1713, 1926,
- 2139, 2351, 2563, 2775, 2986, 3196, 3406, 3616, 3825, 4033,
- 4240, 4447, 4653, 4859, 5063, 5266, 5469, 5671, 5872, 6071,
- 6270, 6467, 6664, 6859, 7053, 7246, 7438, 7629, 7818, 8006,
- 8192, 8377, 8561, 8743, 8923, 9102, 9280, 9456, 9630, 9803,
- 9974, 10143, 10311, 10477, 10641, 10803, 10963, 11121, 11278, 11433,
- 11585, 11736, 11885, 12031, 12176, 12318, 12458, 12597, 12733, 12867,
- 12998, 13128, 13255, 13380, 13502, 13623, 13741, 13856, 13970, 14081,
- 14189, 14295, 14399, 14500, 14598, 14694, 14788, 14879, 14968, 15053,
- 15137, 15218, 15296, 15371, 15444, 15515, 15582, 15647, 15709, 15769,
- 15826, 15880, 15931, 15980, 16026, 16069, 16110, 16147, 16182, 16214,
- 16244, 16270, 16294, 16315, 16333, 16349, 16362, 16371, 16378, 16383,
- 16384, 16383, 16378, 16371, 16362, 16349, 16333, 16315, 16294, 16270,
- 16244, 16214, 16182, 16147, 16110, 16069, 16026, 15980, 15931, 15880,
- 15826, 15769, 15709, 15647, 15582, 15515, 15444, 15371, 15296, 15218,
- 15137, 15053, 14968, 14879, 14788, 14694, 14598, 14500, 14399, 14295,
- 14189, 14081, 13970, 13856, 13741, 13623, 13502, 13380, 13255, 13128,
- 12998, 12867, 12733, 12597, 12458, 12318, 12176, 12031, 11885, 11736,
- 11585, 11433, 11278, 11121, 10963, 10803, 10641, 10477, 10311, 10143,
- 9974, 9803, 9630, 9456, 9280, 9102, 8923, 8743, 8561, 8377,
- 8192, 8006, 7818, 7629, 7438, 7246, 7053, 6859, 6664, 6467,
- 6270, 6071, 5872, 5671, 5469, 5266, 5063, 4859, 4653, 4447,
- 4240, 4033, 3825, 3616, 3406, 3196, 2986, 2775, 2563, 2351,
- 2139, 1926, 1713, 1499, 1285, 1072, 857, 643, 429, 214
-};
-
-
-/* Cosine table 2 in Q14 */
-static const WebRtc_Word16 kCosTab2[FRAMESAMPLES/4] = {
- 107, -322, 536, -750, 965, -1179, 1392, -1606, 1819, -2032,
- 2245, -2457, 2669, -2880, 3091, -3301, 3511, -3720, 3929, -4137,
- 4344, -4550, 4756, -4961, 5165, -5368, 5570, -5771, 5971, -6171,
- 6369, -6566, 6762, -6957, 7150, -7342, 7534, -7723, 7912, -8099,
- 8285, -8469, 8652, -8833, 9013, -9191, 9368, -9543, 9717, -9889,
- 10059, -10227, 10394, -10559, 10722, -10883, 11042, -11200, 11356, -11509,
- 11661, -11810, 11958, -12104, 12247, -12389, 12528, -12665, 12800, -12933,
- 13063, -13192, 13318, -13441, 13563, -13682, 13799, -13913, 14025, -14135,
- 14242, -14347, 14449, -14549, 14647, -14741, 14834, -14924, 15011, -15095,
- 15178, -15257, 15334, -15408, 15480, -15549, 15615, -15679, 15739, -15798,
- 15853, -15906, 15956, -16003, 16048, -16090, 16129, -16165, 16199, -16229,
- 16257, -16283, 16305, -16325, 16342, -16356, 16367, -16375, 16381, -16384
-};
-
-
-/* Sine table 2 in Q14 */
-static const WebRtc_Word16 kSinTab2[FRAMESAMPLES/4] = {
- 16384, -16381, 16375, -16367, 16356, -16342, 16325, -16305, 16283, -16257,
- 16229, -16199, 16165, -16129, 16090, -16048, 16003, -15956, 15906, -15853,
- 15798, -15739, 15679, -15615, 15549, -15480, 15408, -15334, 15257, -15178,
- 15095, -15011, 14924, -14834, 14741, -14647, 14549, -14449, 14347, -14242,
- 14135, -14025, 13913, -13799, 13682, -13563, 13441, -13318, 13192, -13063,
- 12933, -12800, 12665, -12528, 12389, -12247, 12104, -11958, 11810, -11661,
- 11509, -11356, 11200, -11042, 10883, -10722, 10559, -10394, 10227, -10059,
- 9889, -9717, 9543, -9368, 9191, -9013, 8833, -8652, 8469, -8285,
- 8099, -7912, 7723, -7534, 7342, -7150, 6957, -6762, 6566, -6369,
- 6171, -5971, 5771, -5570, 5368, -5165, 4961, -4756, 4550, -4344,
- 4137, -3929, 3720, -3511, 3301, -3091, 2880, -2669, 2457, -2245,
- 2032, -1819, 1606, -1392, 1179, -965, 750, -536, 322, -107
-};
-
-
-
-void WebRtcIsacfix_Time2Spec(WebRtc_Word16 *inre1Q9,
- WebRtc_Word16 *inre2Q9,
- WebRtc_Word16 *outreQ7,
- WebRtc_Word16 *outimQ7)
-{
-
- int k;
- WebRtc_Word32 tmpreQ16[FRAMESAMPLES/2], tmpimQ16[FRAMESAMPLES/2];
- WebRtc_Word16 tmp1rQ14, tmp1iQ14;
- WebRtc_Word32 xrQ16, xiQ16, yrQ16, yiQ16;
- WebRtc_Word32 v1Q16, v2Q16;
- WebRtc_Word16 factQ19, sh;
-
- /* Multiply with complex exponentials and combine into one complex vector */
- factQ19 = 16921; // 0.5/sqrt(240) in Q19 is round(.5/sqrt(240)*(2^19)) = 16921
- for (k = 0; k < FRAMESAMPLES/2; k++) {
- tmp1rQ14 = kCosTab1[k];
- tmp1iQ14 = kSinTab1[k];
- xrQ16 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(tmp1rQ14, inre1Q9[k]) + WEBRTC_SPL_MUL_16_16(tmp1iQ14, inre2Q9[k]), 7);
- xiQ16 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(tmp1rQ14, inre2Q9[k]) - WEBRTC_SPL_MUL_16_16(tmp1iQ14, inre1Q9[k]), 7);
- tmpreQ16[k] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(factQ19, xrQ16)+4, 3); // (Q16*Q19>>16)>>3 = Q16
- tmpimQ16[k] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_32_RSFT16(factQ19, xiQ16)+4, 3); // (Q16*Q19>>16)>>3 = Q16
- }
-
-
- xrQ16 = WebRtcSpl_MaxAbsValueW32(tmpreQ16, FRAMESAMPLES/2);
- yrQ16 = WebRtcSpl_MaxAbsValueW32(tmpimQ16, FRAMESAMPLES/2);
- if (yrQ16>xrQ16) {
- xrQ16 = yrQ16;
- }
-
- sh = WebRtcSpl_NormW32(xrQ16);
- sh = sh-24; //if sh becomes >=0, then we should shift sh steps to the left, and the domain will become Q(16+sh)
- //if sh becomes <0, then we should shift -sh steps to the right, and the domain will become Q(16+sh)
-
- //"Fastest" vectors
- if (sh>=0) {
- for (k=0; k<FRAMESAMPLES/2; k++) {
- inre1Q9[k] = (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W32(tmpreQ16[k], sh); //Q(16+sh)
- inre2Q9[k] = (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W32(tmpimQ16[k], sh); //Q(16+sh)
- }
- } else {
- WebRtc_Word32 round = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, -sh-1);
- for (k=0; k<FRAMESAMPLES/2; k++) {
- inre1Q9[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmpreQ16[k]+round, -sh); //Q(16+sh)
- inre2Q9[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmpimQ16[k]+round, -sh); //Q(16+sh)
- }
- }
-
- /* Get DFT */
- WebRtcIsacfix_FftRadix16Fastest(inre1Q9, inre2Q9, -1); // real call
-
- //"Fastest" vectors
- if (sh>=0) {
- for (k=0; k<FRAMESAMPLES/2; k++) {
- tmpreQ16[k] = WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)inre1Q9[k], sh); //Q(16+sh) -> Q16
- tmpimQ16[k] = WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)inre2Q9[k], sh); //Q(16+sh) -> Q16
- }
- } else {
- for (k=0; k<FRAMESAMPLES/2; k++) {
- tmpreQ16[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inre1Q9[k], -sh); //Q(16+sh) -> Q16
- tmpimQ16[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inre2Q9[k], -sh); //Q(16+sh) -> Q16
- }
- }
-
-
- /* Use symmetry to separate into two complex vectors and center frames in time around zero */
- for (k = 0; k < FRAMESAMPLES/4; k++) {
- xrQ16 = tmpreQ16[k] + tmpreQ16[FRAMESAMPLES/2 - 1 - k];
- yiQ16 = -tmpreQ16[k] + tmpreQ16[FRAMESAMPLES/2 - 1 - k];
- xiQ16 = tmpimQ16[k] - tmpimQ16[FRAMESAMPLES/2 - 1 - k];
- yrQ16 = tmpimQ16[k] + tmpimQ16[FRAMESAMPLES/2 - 1 - k];
- tmp1rQ14 = kCosTab2[k];
- tmp1iQ14 = kSinTab2[k];
- v1Q16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, xrQ16) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, xiQ16);
- v2Q16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, xrQ16) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, xiQ16);
- outreQ7[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(v1Q16, 9);
- outimQ7[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(v2Q16, 9);
- v1Q16 = -WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, yrQ16) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, yiQ16);
- v2Q16 = -WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, yrQ16) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, yiQ16);
- outreQ7[FRAMESAMPLES/2 - 1 - k] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(v1Q16, 9); //CalcLrIntQ(v1Q16, 9);
- outimQ7[FRAMESAMPLES/2 - 1 - k] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(v2Q16, 9); //CalcLrIntQ(v2Q16, 9);
-
- }
-}
-
-
-void WebRtcIsacfix_Spec2Time(WebRtc_Word16 *inreQ7, WebRtc_Word16 *inimQ7, WebRtc_Word32 *outre1Q16, WebRtc_Word32 *outre2Q16)
-{
-
- int k;
- WebRtc_Word16 tmp1rQ14, tmp1iQ14;
- WebRtc_Word32 xrQ16, xiQ16, yrQ16, yiQ16;
- WebRtc_Word32 tmpInRe, tmpInIm, tmpInRe2, tmpInIm2;
- WebRtc_Word16 factQ11;
- WebRtc_Word16 sh;
-
- for (k = 0; k < FRAMESAMPLES/4; k++) {
- /* Move zero in time to beginning of frames */
- tmp1rQ14 = kCosTab2[k];
- tmp1iQ14 = kSinTab2[k];
-
- tmpInRe = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) inreQ7[k], 9); // Q7 -> Q16
- tmpInIm = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) inimQ7[k], 9); // Q7 -> Q16
- tmpInRe2 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) inreQ7[FRAMESAMPLES/2 - 1 - k], 9); // Q7 -> Q16
- tmpInIm2 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) inimQ7[FRAMESAMPLES/2 - 1 - k], 9); // Q7 -> Q16
-
- xrQ16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, tmpInRe) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, tmpInIm);
- xiQ16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, tmpInIm) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, tmpInRe);
- yrQ16 = -WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, tmpInIm2) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, tmpInRe2);
- yiQ16 = -WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, tmpInRe2) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, tmpInIm2);
-
- /* Combine into one vector, z = x + j * y */
- outre1Q16[k] = xrQ16 - yiQ16;
- outre1Q16[FRAMESAMPLES/2 - 1 - k] = xrQ16 + yiQ16;
- outre2Q16[k] = xiQ16 + yrQ16;
- outre2Q16[FRAMESAMPLES/2 - 1 - k] = -xiQ16 + yrQ16;
- }
-
- /* Get IDFT */
- tmpInRe = WebRtcSpl_MaxAbsValueW32(outre1Q16, 240);
- tmpInIm = WebRtcSpl_MaxAbsValueW32(outre2Q16, 240);
- if (tmpInIm>tmpInRe) {
- tmpInRe = tmpInIm;
- }
-
- sh = WebRtcSpl_NormW32(tmpInRe);
- sh = sh-24; //if sh becomes >=0, then we should shift sh steps to the left, and the domain will become Q(16+sh)
- //if sh becomes <0, then we should shift -sh steps to the right, and the domain will become Q(16+sh)
-
- //"Fastest" vectors
- if (sh>=0) {
- for (k=0; k<240; k++) {
- inreQ7[k] = (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W32(outre1Q16[k], sh); //Q(16+sh)
- inimQ7[k] = (WebRtc_Word16) WEBRTC_SPL_LSHIFT_W32(outre2Q16[k], sh); //Q(16+sh)
- }
- } else {
- WebRtc_Word32 round = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)1, -sh-1);
- for (k=0; k<240; k++) {
- inreQ7[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(outre1Q16[k]+round, -sh); //Q(16+sh)
- inimQ7[k] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(outre2Q16[k]+round, -sh); //Q(16+sh)
- }
- }
-
- WebRtcIsacfix_FftRadix16Fastest(inreQ7, inimQ7, 1); // real call
-
- //"Fastest" vectors
- if (sh>=0) {
- for (k=0; k<240; k++) {
- outre1Q16[k] = WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)inreQ7[k], sh); //Q(16+sh) -> Q16
- outre2Q16[k] = WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)inimQ7[k], sh); //Q(16+sh) -> Q16
- }
- } else {
- for (k=0; k<240; k++) {
- outre1Q16[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inreQ7[k], -sh); //Q(16+sh) -> Q16
- outre2Q16[k] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inimQ7[k], -sh); //Q(16+sh) -> Q16
- }
- }
-
- /* Divide through by the normalizing constant: */
- /* scale all values with 1/240, i.e. with 273 in Q16 */
- /* 273/65536 ~= 0.0041656 */
- /* 1/240 ~= 0.0041666 */
- for (k=0; k<240; k++) {
- outre1Q16[k] = WEBRTC_SPL_MUL_16_32_RSFT16(273, outre1Q16[k]);
- outre2Q16[k] = WEBRTC_SPL_MUL_16_32_RSFT16(273, outre2Q16[k]);
- }
-
- /* Demodulate and separate */
- factQ11 = 31727; // sqrt(240) in Q11 is round(15.49193338482967*2048) = 31727
- for (k = 0; k < FRAMESAMPLES/2; k++) {
- tmp1rQ14 = kCosTab1[k];
- tmp1iQ14 = kSinTab1[k];
- xrQ16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, outre1Q16[k]) - WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, outre2Q16[k]);
- xiQ16 = WEBRTC_SPL_MUL_16_32_RSFT14(tmp1rQ14, outre2Q16[k]) + WEBRTC_SPL_MUL_16_32_RSFT14(tmp1iQ14, outre1Q16[k]);
- xrQ16 = WEBRTC_SPL_MUL_16_32_RSFT11(factQ11, xrQ16);
- xiQ16 = WEBRTC_SPL_MUL_16_32_RSFT11(factQ11, xiQ16);
- outre2Q16[k] = xiQ16;
- outre1Q16[k] = xrQ16;
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/ISACHist.cc b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/ISACHist.cc
deleted file mode 100644
index 753acd7..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/ISACHist.cc
+++ /dev/null
@@ -1,173 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <ctype.h>
-
-//#include "isac_codec.h"
-//#include "isac_structs.h"
-#include "isacfix.h"
-
-
-#define NUM_CODECS 1
-
-int main(int argc, char* argv[])
-{
- FILE *inFileList;
- FILE *audioFile;
- FILE *outFile;
- char audioFileName[501];
- short audioBuff[960];
- short encoded[600];
- short startAudio;
- short encodedLen;
- ISACFIX_MainStruct *isac_struct;
- unsigned long int hist[601];
-
- // reset the histogram
- for(short n=0; n < 601; n++)
- {
- hist[n] = 0;
- }
-
-
- inFileList = fopen(argv[1], "r");
- if(inFileList == NULL)
- {
- printf("Could not open the input file.\n");
- getchar();
- exit(-1);
- }
- outFile = fopen(argv[2], "w");
- if(outFile == NULL)
- {
- printf("Could not open the histogram file.\n");
- getchar();
- exit(-1);
- }
-
- short frameSizeMsec = 30;
- if(argc > 3)
- {
- frameSizeMsec = atoi(argv[3]);
- }
-
- short audioOffset = 0;
- if(argc > 4)
- {
- audioOffset = atoi(argv[4]);
- }
- int ok;
- ok = WebRtcIsacfix_Create(&isac_struct);
- // instantaneous mode
- ok |= WebRtcIsacfix_EncoderInit(isac_struct, 1);
- // is not used but initialize
- ok |= WebRtcIsacfix_DecoderInit(isac_struct);
- ok |= WebRtcIsacfix_Control(isac_struct, 32000, frameSizeMsec);
-
- if(ok != 0)
- {
- printf("\nProblem in seting up iSAC\n");
- exit(-1);
- }
-
- while( fgets(audioFileName, 500, inFileList) != NULL )
- {
- // remove trailing white-spaces and any Cntrl character
- if(strlen(audioFileName) == 0)
- {
- continue;
- }
- short n = strlen(audioFileName) - 1;
- while(isspace(audioFileName[n]) || iscntrl(audioFileName[n]))
- {
- audioFileName[n] = '\0';
- n--;
- if(n < 0)
- {
- break;
- }
- }
-
- // remove leading spaces
- if(strlen(audioFileName) == 0)
- {
- continue;
- }
- n = 0;
- while((isspace(audioFileName[n]) || iscntrl(audioFileName[n])) &&
- (audioFileName[n] != '\0'))
- {
- n++;
- }
- memmove(audioFileName, &audioFileName[n], 500 - n);
- if(strlen(audioFileName) == 0)
- {
- continue;
- }
- audioFile = fopen(audioFileName, "rb");
- if(audioFile == NULL)
- {
- printf("\nCannot open %s!!!!!\n", audioFileName);
- exit(0);
- }
-
- if(audioOffset > 0)
- {
- fseek(audioFile, (audioOffset<<1), SEEK_SET);
- }
-
- while(fread(audioBuff, sizeof(short), (480*frameSizeMsec/30), audioFile) >= (480*frameSizeMsec/30))
- {
- startAudio = 0;
- do
- {
- encodedLen = WebRtcIsacfix_Encode(isac_struct,
- &audioBuff[startAudio], encoded);
- startAudio += 160;
- } while(encodedLen == 0);
-
- if(encodedLen < 0)
- {
- printf("\nEncoding Error!!!\n");
- exit(0);
- }
- hist[encodedLen]++;
- }
- fclose(audioFile);
- }
- fclose(inFileList);
- unsigned long totalFrames = 0;
- for(short n=0; n < 601; n++)
- {
- totalFrames += hist[n];
- fprintf(outFile, "%10lu\n", hist[n]);
- }
- fclose(outFile);
-
- short topTenCntr = 0;
- printf("\nTotal number of Frames %lu\n\n", totalFrames);
- printf("Payload Len # occurences\n");
- printf("=========== ============\n");
-
- for(short n = 600; (n >= 0) && (topTenCntr < 10); n--)
- {
- if(hist[n] > 0)
- {
- topTenCntr++;
- printf(" %3d %3d\n", n, hist[n]);
- }
- }
- WebRtcIsacfix_Free(isac_struct);
- return 0;
-}
-
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/Isac_test.cc b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/Isac_test.cc
deleted file mode 100644
index 2791db4..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/Isac_test.cc
+++ /dev/null
@@ -1,260 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
- Stand Alone test application for ISACFIX and ISAC LC
-
-******************************************************************/
-
-#include <string.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include "typedefs.h"
-
-#include "isacfix.h"
-ISACFIX_MainStruct *ISACfix_inst;
-
-#define FS 16000
-
-
-typedef struct {
- WebRtc_UWord32 arrival_time; /* samples */
- WebRtc_UWord32 sample_count; /* samples */
- WebRtc_UWord16 rtp_number;
-} BottleNeckModel;
-
-void get_arrival_time(int current_framesamples, /* samples */
- int packet_size, /* bytes */
- int bottleneck, /* excluding headers; bits/s */
- BottleNeckModel *BN_data)
-{
- const int HeaderSize = 35;
- int HeaderRate;
-
- HeaderRate = HeaderSize * 8 * FS / current_framesamples; /* bits/s */
-
- /* everything in samples */
- BN_data->sample_count = BN_data->sample_count + current_framesamples;
-
- BN_data->arrival_time += ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate);
-
- if (BN_data->arrival_time < BN_data->sample_count)
- BN_data->arrival_time = BN_data->sample_count;
-
- BN_data->rtp_number++;
-}
-
-/*
-#ifdef __cplusplus
-extern "C" {
-#endif
-*/
-int main(int argc, char* argv[]){
-
- /* Parameters */
- FILE *pInFile, *pOutFile, *pChcFile;
- WebRtc_Word8 inFile[40];
- WebRtc_Word8 outFile[40];
- WebRtc_Word8 chcFile[40];
- WebRtc_Word8 codec[10];
- WebRtc_Word16 bitrt, spType, size;
- WebRtc_UWord16 frameLen;
- WebRtc_Word16 sigOut[1000], sigIn[1000];
- WebRtc_UWord16 bitStream[500]; /* double to 32 kbps for 60 ms */
-
- WebRtc_Word16 chc, ok;
- int noOfCalls, cdlen;
- WebRtc_Word16 noOfLostFrames;
- int err, errtype;
-
- BottleNeckModel BN_data;
-
- int totalbits =0;
- int totalsmpls =0;
-
- /*End Parameters*/
-
-
- if ((argc==6)||(argc==7) ){
-
- strcpy(codec,argv[5]);
-
- if(argc==7){
- if (!_stricmp("isac",codec)){
- bitrt = atoi(argv[6]);
- if ( (bitrt<10000)&&(bitrt>32000)){
- printf("Error: Supported bit rate in the range 10000-32000 bps!\n");
- exit(-1);
- }
-
- }else{
- printf("Error: Codec not recognized. Check spelling!\n");
- exit(-1);
- }
-
- } else {
- printf("Error: Codec not recognized. Check spelling!\n");
- exit(-1);
- }
- } else {
- printf("Error: Wrong number of input parameter!\n\n");
- exit(-1);
- }
-
- frameLen = atoi(argv[4]);
- strcpy(chcFile,argv[3]);
- strcpy(outFile,argv[2]);
- strcpy(inFile,argv[1]);
-
- /* Open file streams */
- if( (pInFile = fopen(inFile,"rb")) == NULL ) {
- printf( "Error: Did not find input file!\n" );
- exit(-1);
- }
- strcat(outFile,"_");
- strcat(outFile, argv[4]);
- strcat(outFile,"_");
- strcat(outFile, codec);
-
- if (argc==7){
- strcat(outFile,"_");
- strcat(outFile, argv[6]);
- }
- if (_stricmp("none", chcFile)){
- strcat(outFile,"_");
- strcat(outFile, "plc");
- }
-
- strcat(outFile, ".otp");
-
- if (_stricmp("none", chcFile)){
- if( (pChcFile = fopen(chcFile,"rb")) == NULL ) {
- printf( "Error: Did not find channel file!\n" );
- exit(-1);
- }
- }
- /******************************************************************/
- if (!_stricmp("isac", codec)){ /* ISAC */
- if ((frameLen!=480)&&(frameLen!=960)) {
- printf("Error: ISAC only supports 480 and 960 samples per frame (not %d)\n", frameLen);
- exit(-1);
- }
- if( (pOutFile = fopen(outFile,"wb")) == NULL ) {
- printf( "Could not open output file!\n" );
- exit(-1);
- }
- ok=WebRtcIsacfix_Create(&ISACfix_inst);
- if (ok!=0) {
- printf("Couldn't allocate memory for iSAC fix instance\n");
- exit(-1);
- }
-
- BN_data.arrival_time = 0;
- BN_data.sample_count = 0;
- BN_data.rtp_number = 0;
-
- WebRtcIsacfix_EncoderInit(ISACfix_inst,1);
- WebRtcIsacfix_DecoderInit(ISACfix_inst);
- err = WebRtcIsacfix_Control(ISACfix_inst, bitrt, (frameLen>>4));
- if (err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISACfix_inst);
- printf("\n\n Error in initialization: %d.\n\n", errtype);
- exit(EXIT_FAILURE);
- }
- /* loop over frame */
- while (fread(sigIn,sizeof(WebRtc_Word16),frameLen,pInFile) == frameLen) {
-
- noOfCalls=0;
- cdlen=0;
- while (cdlen<=0) {
- cdlen=WebRtcIsacfix_Encode(ISACfix_inst,&sigIn[noOfCalls*160],(WebRtc_Word16*)bitStream);
- if(cdlen==-1){
- errtype=WebRtcIsacfix_GetErrorCode(ISACfix_inst);
- printf("\n\nError in encoder: %d.\n\n", errtype);
- exit(-1);
- }
- noOfCalls++;
- }
-
-
- if(_stricmp("none", chcFile)){
- if (fread(&chc,sizeof(WebRtc_Word16),1,pChcFile)!=1) /* packet may be lost */
- break;
- } else {
- chc = 1; /* packets never lost */
- }
-
- /* simulate packet handling through NetEq and the modem */
- get_arrival_time(frameLen, cdlen, bitrt, &BN_data);
-
- if (chc){ /* decode */
-
- err = WebRtcIsacfix_UpdateBwEstimate1(ISACfix_inst,
- bitStream,
- cdlen,
- BN_data.rtp_number,
- BN_data.arrival_time);
-
- if (err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISACfix_inst);
- printf("\n\nError in decoder: %d.\n\n", errtype);
- exit(EXIT_FAILURE);
- }
- size = WebRtcIsacfix_Decode(ISACfix_inst, bitStream, cdlen, sigOut, &spType);
- if(size<=0){
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISACfix_inst);
- printf("\n\nError in decoder: %d.\n\n", errtype);
- exit(-1);
- }
- } else { /* PLC */
- if (frameLen == 480){
- noOfLostFrames = 1;
- } else{
- noOfLostFrames = 2;
- }
- size = WebRtcIsacfix_DecodePlc(ISACfix_inst, sigOut, noOfLostFrames );
- if(size<=0){
- errtype=WebRtcIsacfix_GetErrorCode(ISACfix_inst);
- printf("\n\nError in decoder: %d.\n\n", errtype);
- exit(-1);
- }
- }
-
- /* Write decoded speech to file */
- fwrite(sigOut,sizeof(short),size,pOutFile);
-
- totalbits += 8 * cdlen;
- totalsmpls += size;
-
- }
- /******************************************************************/
- }
-
-// printf("\n\ntotal bits = %d bits", totalbits);
- printf("\nmeasured average bitrate = %0.3f kbits/s", (double)totalbits * 16 / totalsmpls);
- printf("\n");
-
-
- fclose(pInFile);
- fclose(pOutFile);
- if (_stricmp("none", chcFile)){
- fclose(pChcFile);
- }
-
- if (!_stricmp("isac", codec)){
- WebRtcIsacfix_Free(ISACfix_inst);
- }
-
- return 0;
-
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/ChannelFiles.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/ChannelFiles.txt
deleted file mode 100644
index 05f7410..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/ChannelFiles.txt
+++ /dev/null
@@ -1,3 +0,0 @@
-bottlenecks.txt
-lowrates.txt
-tworates.txt
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/InputFiles.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/InputFiles.txt
deleted file mode 100644
index f26b7af..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/InputFiles.txt
+++ /dev/null
@@ -1,31 +0,0 @@
-DTMF_16kHz_long.pcm
-DTMF_16kHz_short.pcm
-F00.INP
-F01.INP
-F02.INP
-F03.INP
-F04.INP
-F05.INP
-F06.INP
-longtest.pcm
-ltest_speech_clean.pcm
-ltest_music.pcm
-ltest_speech_noisy.pcm
-misc2.pcm
-purenb.pcm
-sawsweep_380_60.pcm
-sinesweep.pcm
-sinesweep_half.pcm
-speechmusic.pcm
-speechmusic_nb.pcm
-speechoffice0dB.pcm
-speech_and_misc_NB.pcm
-speech_and_misc_WB.pcm
-testM4.pcm
-testM4D_rev.pcm
-testM4D.pcm
-testfile.pcm
-tone_cisco.pcm
-tone_cisco_long.pcm
-wb_contspeech.pcm
-wb_speech_office25db.pcm
\ No newline at end of file
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/InputFilesFew.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/InputFilesFew.txt
deleted file mode 100644
index 08bbde3..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/InputFilesFew.txt
+++ /dev/null
@@ -1,6 +0,0 @@
-DTMF_16kHz_short.pcm
-ltest_speech_noisy.pcm
-misc2.pcm
-sinesweep.pcm
-speechmusic.pcm
-tone_cisco.pcm
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/ListOfTestCases.xls b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/ListOfTestCases.xls
deleted file mode 100644
index f0889ef..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/ListOfTestCases.xls
+++ /dev/null
Binary files differ
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/diffiSAC.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/diffiSAC.txt
deleted file mode 100644
index 96b87c0..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/diffiSAC.txt
+++ /dev/null
@@ -1,481 +0,0 @@
-#!/bin/bash
-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
-
-diff ../dataqa350/i30_1DTMF_16kHz_long.pcm ../dataqa351/i30_1DTMF_16kHz_long.pcm
-diff ../dataqa350/i60_1DTMF_16kHz_long.pcm ../dataqa351/i60_1DTMF_16kHz_long.pcm
-diff ../dataqa350/i30_2DTMF_16kHz_long.pcm ../dataqa351/i30_2DTMF_16kHz_long.pcm
-diff ../dataqa350/i60_2DTMF_16kHz_long.pcm ../dataqa351/i60_2DTMF_16kHz_long.pcm
-diff ../dataqa350/i30_3DTMF_16kHz_long.pcm ../dataqa351/i30_3DTMF_16kHz_long.pcm
-diff ../dataqa350/i60_3DTMF_16kHz_long.pcm ../dataqa351/i60_3DTMF_16kHz_long.pcm
-diff ../dataqa350/i30_4DTMF_16kHz_long.pcm ../dataqa351/i30_4DTMF_16kHz_long.pcm
-diff ../dataqa350/i60_4DTMF_16kHz_long.pcm ../dataqa351/i60_4DTMF_16kHz_long.pcm
-diff ../dataqa350/i30_5DTMF_16kHz_long.pcm ../dataqa351/i30_5DTMF_16kHz_long.pcm
-diff ../dataqa350/i60_5DTMF_16kHz_long.pcm ../dataqa351/i60_5DTMF_16kHz_long.pcm
-diff ../dataqa350/i30_6DTMF_16kHz_long.pcm ../dataqa351/i30_6DTMF_16kHz_long.pcm
-diff ../dataqa350/i60_6DTMF_16kHz_long.pcm ../dataqa351/i60_6DTMF_16kHz_long.pcm
-diff ../dataqa350/a1DTMF_16kHz_long.pcm ../dataqa351/a1DTMF_16kHz_long.pcm
-diff ../dataqa350/a2DTMF_16kHz_long.pcm ../dataqa351/a2DTMF_16kHz_long.pcm
-diff ../dataqa350/a3DTMF_16kHz_long.pcm ../dataqa351/a3DTMF_16kHz_long.pcm
-diff ../dataqa350/i30_7DTMF_16kHz_short.pcm ../dataqa351/i30_7DTMF_16kHz_short.pcm
-diff ../dataqa350/i60_7DTMF_16kHz_short.pcm ../dataqa351/i60_7DTMF_16kHz_short.pcm
-diff ../dataqa350/i30_8DTMF_16kHz_short.pcm ../dataqa351/i30_8DTMF_16kHz_short.pcm
-diff ../dataqa350/i60_8DTMF_16kHz_short.pcm ../dataqa351/i60_8DTMF_16kHz_short.pcm
-diff ../dataqa350/i30_9DTMF_16kHz_short.pcm ../dataqa351/i30_9DTMF_16kHz_short.pcm
-diff ../dataqa350/i60_9DTMF_16kHz_short.pcm ../dataqa351/i60_9DTMF_16kHz_short.pcm
-diff ../dataqa350/i30_10DTMF_16kHz_short.pcm ../dataqa351/i30_10DTMF_16kHz_short.pcm
-diff ../dataqa350/i60_10DTMF_16kHz_short.pcm ../dataqa351/i60_10DTMF_16kHz_short.pcm
-diff ../dataqa350/i30_11DTMF_16kHz_short.pcm ../dataqa351/i30_11DTMF_16kHz_short.pcm
-diff ../dataqa350/i60_11DTMF_16kHz_short.pcm ../dataqa351/i60_11DTMF_16kHz_short.pcm
-diff ../dataqa350/i30_12DTMF_16kHz_short.pcm ../dataqa351/i30_12DTMF_16kHz_short.pcm
-diff ../dataqa350/i60_12DTMF_16kHz_short.pcm ../dataqa351/i60_12DTMF_16kHz_short.pcm
-diff ../dataqa350/a4DTMF_16kHz_short.pcm ../dataqa351/a4DTMF_16kHz_short.pcm
-diff ../dataqa350/a5DTMF_16kHz_short.pcm ../dataqa351/a5DTMF_16kHz_short.pcm
-diff ../dataqa350/a6DTMF_16kHz_short.pcm ../dataqa351/a6DTMF_16kHz_short.pcm
-diff ../dataqa350/i30_13F00.INP ../dataqa350/i30_13F00.INP
-diff ../dataqa350/i60_13F00.INP ../dataqa350/i60_13F00.INP
-diff ../dataqa350/i30_14F00.INP ../dataqa350/i30_14F00.INP
-diff ../dataqa350/i60_14F00.INP ../dataqa350/i60_14F00.INP
-diff ../dataqa350/i30_15F00.INP ../dataqa350/i30_15F00.INP
-diff ../dataqa350/i60_15F00.INP ../dataqa350/i60_15F00.INP
-diff ../dataqa350/i30_16F00.INP ../dataqa350/i30_16F00.INP
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-diff ../dataqa350/i30_17F00.INP ../dataqa350/i30_17F00.INP
-diff ../dataqa350/i60_17F00.INP ../dataqa350/i60_17F00.INP
-diff ../dataqa350/i30_18F00.INP ../dataqa350/i30_18F00.INP
-diff ../dataqa350/i60_18F00.INP ../dataqa350/i60_18F00.INP
-diff ../dataqa350/a7F00.INP ../dataqa350/a7F00.INP
-diff ../dataqa350/a8F00.INP ../dataqa350/a8F00.INP
-diff ../dataqa350/a9F00.INP ../dataqa350/a9F00.INP
-diff ../dataqa350/i30_19F01.INP ../dataqa350/i30_19F01.INP
-diff ../dataqa350/i60_19F01.INP ../dataqa350/i60_19F01.INP
-diff ../dataqa350/i30_20F01.INP ../dataqa350/i30_20F01.INP
-diff ../dataqa350/i60_20F01.INP ../dataqa350/i60_20F01.INP
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-diff ../dataqa350/a10F01.INP ../dataqa350/a10F01.INP
-diff ../dataqa350/a11F01.INP ../dataqa350/a11F01.INP
-diff ../dataqa350/a12F01.INP ../dataqa350/a12F01.INP
-diff ../dataqa350/i30_25F02.INP ../dataqa350/i30_25F02.INP
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-diff ../dataqa350/i30_26F02.INP ../dataqa350/i30_26F02.INP
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-diff ../dataqa350/i30_29F02.INP ../dataqa350/i30_29F02.INP
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-diff ../dataqa350/i30_30F02.INP ../dataqa350/i30_30F02.INP
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-diff ../dataqa350/a13F02.INP ../dataqa350/a13F02.INP
-diff ../dataqa350/a14F02.INP ../dataqa350/a14F02.INP
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-diff ../dataqa350/i30_31F03.INP ../dataqa350/i30_31F03.INP
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-diff ../dataqa350/i30_37F04.INP ../dataqa350/i30_37F04.INP
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-diff ../dataqa350/i30_38F04.INP ../dataqa350/i30_38F04.INP
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-diff ../dataqa350/i30_40F04.INP ../dataqa350/i30_40F04.INP
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-diff ../dataqa350/i30_41F04.INP ../dataqa350/i30_41F04.INP
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-diff ../dataqa350/a19F04.INP ../dataqa350/a19F04.INP
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-diff ../dataqa350/a21F04.INP ../dataqa350/a21F04.INP
-diff ../dataqa350/i30_43F05.INP ../dataqa350/i30_43F05.INP
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-diff ../dataqa350/a22F05.INP ../dataqa350/a22F05.INP
-diff ../dataqa350/a23F05.INP ../dataqa350/a23F05.INP
-diff ../dataqa350/a24F05.INP ../dataqa350/a24F05.INP
-diff ../dataqa350/i30_49F06.INP ../dataqa350/i30_49F06.INP
-diff ../dataqa350/i60_49F06.INP ../dataqa350/i60_49F06.INP
-diff ../dataqa350/i30_50F06.INP ../dataqa350/i30_50F06.INP
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-diff ../dataqa350/i30_54F06.INP ../dataqa350/i30_54F06.INP
-diff ../dataqa350/i60_54F06.INP ../dataqa350/i60_54F06.INP
-diff ../dataqa350/a25F06.INP ../dataqa350/a25F06.INP
-diff ../dataqa350/a26F06.INP ../dataqa350/a26F06.INP
-diff ../dataqa350/a27F06.INP ../dataqa350/a27F06.INP
-diff ../dataqa350/i30_55longtest.pcm ../dataqa351/i30_55longtest.pcm
-diff ../dataqa350/i60_55longtest.pcm ../dataqa351/i60_55longtest.pcm
-diff ../dataqa350/i30_56longtest.pcm ../dataqa351/i30_56longtest.pcm
-diff ../dataqa350/i60_56longtest.pcm ../dataqa351/i60_56longtest.pcm
-diff ../dataqa350/i30_57longtest.pcm ../dataqa351/i30_57longtest.pcm
-diff ../dataqa350/i60_57longtest.pcm ../dataqa351/i60_57longtest.pcm
-diff ../dataqa350/i30_58longtest.pcm ../dataqa351/i30_58longtest.pcm
-diff ../dataqa350/i60_58longtest.pcm ../dataqa351/i60_58longtest.pcm
-diff ../dataqa350/i30_59longtest.pcm ../dataqa351/i30_59longtest.pcm
-diff ../dataqa350/i60_59longtest.pcm ../dataqa351/i60_59longtest.pcm
-diff ../dataqa350/i30_60longtest.pcm ../dataqa351/i30_60longtest.pcm
-diff ../dataqa350/i60_60longtest.pcm ../dataqa351/i60_60longtest.pcm
-diff ../dataqa350/a28longtest.pcm ../dataqa351/a28longtest.pcm
-diff ../dataqa350/a29longtest.pcm ../dataqa351/a29longtest.pcm
-diff ../dataqa350/a30longtest.pcm ../dataqa351/a30longtest.pcm
-diff ../dataqa350/i30_61ltest_speech_clean.pcm ../dataqa351/i30_61ltest_speech_clean.pcm
-diff ../dataqa350/i60_61ltest_speech_clean.pcm ../dataqa351/i60_61ltest_speech_clean.pcm
-diff ../dataqa350/i30_62ltest_speech_clean.pcm ../dataqa351/i30_62ltest_speech_clean.pcm
-diff ../dataqa350/i60_62ltest_speech_clean.pcm ../dataqa351/i60_62ltest_speech_clean.pcm
-diff ../dataqa350/i30_63ltest_speech_clean.pcm ../dataqa351/i30_63ltest_speech_clean.pcm
-diff ../dataqa350/i60_63ltest_speech_clean.pcm ../dataqa351/i60_63ltest_speech_clean.pcm
-diff ../dataqa350/i30_64ltest_speech_clean.pcm ../dataqa351/i30_64ltest_speech_clean.pcm
-diff ../dataqa350/i60_64ltest_speech_clean.pcm ../dataqa351/i60_64ltest_speech_clean.pcm
-diff ../dataqa350/i30_65ltest_speech_clean.pcm ../dataqa351/i30_65ltest_speech_clean.pcm
-diff ../dataqa350/i60_65ltest_speech_clean.pcm ../dataqa351/i60_65ltest_speech_clean.pcm
-diff ../dataqa350/i30_66ltest_speech_clean.pcm ../dataqa351/i30_66ltest_speech_clean.pcm
-diff ../dataqa350/i60_66ltest_speech_clean.pcm ../dataqa351/i60_66ltest_speech_clean.pcm
-diff ../dataqa350/a31ltest_speech_clean.pcm ../dataqa351/a31ltest_speech_clean.pcm
-diff ../dataqa350/a32ltest_speech_clean.pcm ../dataqa351/a32ltest_speech_clean.pcm
-diff ../dataqa350/a33ltest_speech_clean.pcm ../dataqa351/a33ltest_speech_clean.pcm
-diff ../dataqa350/i30_67ltest_music.pcm ../dataqa351/i30_67ltest_music.pcm
-diff ../dataqa350/i60_67ltest_music.pcm ../dataqa351/i60_67ltest_music.pcm
-diff ../dataqa350/i30_68ltest_music.pcm ../dataqa351/i30_68ltest_music.pcm
-diff ../dataqa350/i60_68ltest_music.pcm ../dataqa351/i60_68ltest_music.pcm
-diff ../dataqa350/i30_69ltest_music.pcm ../dataqa351/i30_69ltest_music.pcm
-diff ../dataqa350/i60_69ltest_music.pcm ../dataqa351/i60_69ltest_music.pcm
-diff ../dataqa350/i30_70ltest_music.pcm ../dataqa351/i30_70ltest_music.pcm
-diff ../dataqa350/i60_70ltest_music.pcm ../dataqa351/i60_70ltest_music.pcm
-diff ../dataqa350/i30_71ltest_music.pcm ../dataqa351/i30_71ltest_music.pcm
-diff ../dataqa350/i60_71ltest_music.pcm ../dataqa351/i60_71ltest_music.pcm
-diff ../dataqa350/i30_72ltest_music.pcm ../dataqa351/i30_72ltest_music.pcm
-diff ../dataqa350/i60_72ltest_music.pcm ../dataqa351/i60_72ltest_music.pcm
-diff ../dataqa350/a34ltest_music.pcm ../dataqa351/a34ltest_music.pcm
-diff ../dataqa350/a35ltest_music.pcm ../dataqa351/a35ltest_music.pcm
-diff ../dataqa350/a36ltest_music.pcm ../dataqa351/a36ltest_music.pcm
-diff ../dataqa350/i30_73ltest_speech_noisy.pcm ../dataqa351/i30_73ltest_speech_noisy.pcm
-diff ../dataqa350/i60_73ltest_speech_noisy.pcm ../dataqa351/i60_73ltest_speech_noisy.pcm
-diff ../dataqa350/i30_74ltest_speech_noisy.pcm ../dataqa351/i30_74ltest_speech_noisy.pcm
-diff ../dataqa350/i60_74ltest_speech_noisy.pcm ../dataqa351/i60_74ltest_speech_noisy.pcm
-diff ../dataqa350/i30_75ltest_speech_noisy.pcm ../dataqa351/i30_75ltest_speech_noisy.pcm
-diff ../dataqa350/i60_75ltest_speech_noisy.pcm ../dataqa351/i60_75ltest_speech_noisy.pcm
-diff ../dataqa350/i30_76ltest_speech_noisy.pcm ../dataqa351/i30_76ltest_speech_noisy.pcm
-diff ../dataqa350/i60_76ltest_speech_noisy.pcm ../dataqa351/i60_76ltest_speech_noisy.pcm
-diff ../dataqa350/i30_77ltest_speech_noisy.pcm ../dataqa351/i30_77ltest_speech_noisy.pcm
-diff ../dataqa350/i60_77ltest_speech_noisy.pcm ../dataqa351/i60_77ltest_speech_noisy.pcm
-diff ../dataqa350/i30_78ltest_speech_noisy.pcm ../dataqa351/i30_78ltest_speech_noisy.pcm
-diff ../dataqa350/i60_78ltest_speech_noisy.pcm ../dataqa351/i60_78ltest_speech_noisy.pcm
-diff ../dataqa350/a37ltest_speech_noisy.pcm ../dataqa351/a37ltest_speech_noisy.pcm
-diff ../dataqa350/a38ltest_speech_noisy.pcm ../dataqa351/a38ltest_speech_noisy.pcm
-diff ../dataqa350/a39ltest_speech_noisy.pcm ../dataqa351/a39ltest_speech_noisy.pcm
-diff ../dataqa350/i30_79misc2.pcm ../dataqa351/i30_79misc2.pcm
-diff ../dataqa350/i60_79misc2.pcm ../dataqa351/i60_79misc2.pcm
-diff ../dataqa350/i30_80misc2.pcm ../dataqa351/i30_80misc2.pcm
-diff ../dataqa350/i60_80misc2.pcm ../dataqa351/i60_80misc2.pcm
-diff ../dataqa350/i30_81misc2.pcm ../dataqa351/i30_81misc2.pcm
-diff ../dataqa350/i60_81misc2.pcm ../dataqa351/i60_81misc2.pcm
-diff ../dataqa350/i30_82misc2.pcm ../dataqa351/i30_82misc2.pcm
-diff ../dataqa350/i60_82misc2.pcm ../dataqa351/i60_82misc2.pcm
-diff ../dataqa350/i30_83misc2.pcm ../dataqa351/i30_83misc2.pcm
-diff ../dataqa350/i60_83misc2.pcm ../dataqa351/i60_83misc2.pcm
-diff ../dataqa350/i30_84misc2.pcm ../dataqa351/i30_84misc2.pcm
-diff ../dataqa350/i60_84misc2.pcm ../dataqa351/i60_84misc2.pcm
-diff ../dataqa350/a40misc2.pcm ../dataqa351/a40misc2.pcm
-diff ../dataqa350/a41misc2.pcm ../dataqa351/a41misc2.pcm
-diff ../dataqa350/a42misc2.pcm ../dataqa351/a42misc2.pcm
-diff ../dataqa350/i30_85purenb.pcm ../dataqa351/i30_85purenb.pcm
-diff ../dataqa350/i60_85purenb.pcm ../dataqa351/i60_85purenb.pcm
-diff ../dataqa350/i30_86purenb.pcm ../dataqa351/i30_86purenb.pcm
-diff ../dataqa350/i60_86purenb.pcm ../dataqa351/i60_86purenb.pcm
-diff ../dataqa350/i30_87purenb.pcm ../dataqa351/i30_87purenb.pcm
-diff ../dataqa350/i60_87purenb.pcm ../dataqa351/i60_87purenb.pcm
-diff ../dataqa350/i30_88purenb.pcm ../dataqa351/i30_88purenb.pcm
-diff ../dataqa350/i60_88purenb.pcm ../dataqa351/i60_88purenb.pcm
-diff ../dataqa350/i30_89purenb.pcm ../dataqa351/i30_89purenb.pcm
-diff ../dataqa350/i60_89purenb.pcm ../dataqa351/i60_89purenb.pcm
-diff ../dataqa350/i30_90purenb.pcm ../dataqa351/i30_90purenb.pcm
-diff ../dataqa350/i60_90purenb.pcm ../dataqa351/i60_90purenb.pcm
-diff ../dataqa350/a43purenb.pcm ../dataqa351/a43purenb.pcm
-diff ../dataqa350/a44purenb.pcm ../dataqa351/a44purenb.pcm
-diff ../dataqa350/a45purenb.pcm ../dataqa351/a45purenb.pcm
-diff ../dataqa350/i30_91sawsweep_380_60.pcm ../dataqa351/i30_91sawsweep_380_60.pcm
-diff ../dataqa350/i60_91sawsweep_380_60.pcm ../dataqa351/i60_91sawsweep_380_60.pcm
-diff ../dataqa350/i30_92sawsweep_380_60.pcm ../dataqa351/i30_92sawsweep_380_60.pcm
-diff ../dataqa350/i60_92sawsweep_380_60.pcm ../dataqa351/i60_92sawsweep_380_60.pcm
-diff ../dataqa350/i30_93sawsweep_380_60.pcm ../dataqa351/i30_93sawsweep_380_60.pcm
-diff ../dataqa350/i60_93sawsweep_380_60.pcm ../dataqa351/i60_93sawsweep_380_60.pcm
-diff ../dataqa350/i30_94sawsweep_380_60.pcm ../dataqa351/i30_94sawsweep_380_60.pcm
-diff ../dataqa350/i60_94sawsweep_380_60.pcm ../dataqa351/i60_94sawsweep_380_60.pcm
-diff ../dataqa350/i30_95sawsweep_380_60.pcm ../dataqa351/i30_95sawsweep_380_60.pcm
-diff ../dataqa350/i60_95sawsweep_380_60.pcm ../dataqa351/i60_95sawsweep_380_60.pcm
-diff ../dataqa350/i30_96sawsweep_380_60.pcm ../dataqa351/i30_96sawsweep_380_60.pcm
-diff ../dataqa350/i60_96sawsweep_380_60.pcm ../dataqa351/i60_96sawsweep_380_60.pcm
-diff ../dataqa350/a46sawsweep_380_60.pcm ../dataqa351/a46sawsweep_380_60.pcm
-diff ../dataqa350/a47sawsweep_380_60.pcm ../dataqa351/a47sawsweep_380_60.pcm
-diff ../dataqa350/a48sawsweep_380_60.pcm ../dataqa351/a48sawsweep_380_60.pcm
-diff ../dataqa350/i30_97sinesweep.pcm ../dataqa351/i30_97sinesweep.pcm
-diff ../dataqa350/i60_97sinesweep.pcm ../dataqa351/i60_97sinesweep.pcm
-diff ../dataqa350/i30_98sinesweep.pcm ../dataqa351/i30_98sinesweep.pcm
-diff ../dataqa350/i60_98sinesweep.pcm ../dataqa351/i60_98sinesweep.pcm
-diff ../dataqa350/i30_99sinesweep.pcm ../dataqa351/i30_99sinesweep.pcm
-diff ../dataqa350/i60_99sinesweep.pcm ../dataqa351/i60_99sinesweep.pcm
-diff ../dataqa350/i30_100sinesweep.pcm ../dataqa351/i30_100sinesweep.pcm
-diff ../dataqa350/i60_100sinesweep.pcm ../dataqa351/i60_100sinesweep.pcm
-diff ../dataqa350/i30_101sinesweep.pcm ../dataqa351/i30_101sinesweep.pcm
-diff ../dataqa350/i60_101sinesweep.pcm ../dataqa351/i60_101sinesweep.pcm
-diff ../dataqa350/i30_102sinesweep.pcm ../dataqa351/i30_102sinesweep.pcm
-diff ../dataqa350/i60_102sinesweep.pcm ../dataqa351/i60_102sinesweep.pcm
-diff ../dataqa350/a49sinesweep.pcm ../dataqa351/a49sinesweep.pcm
-diff ../dataqa350/a50sinesweep.pcm ../dataqa351/a50sinesweep.pcm
-diff ../dataqa350/a51sinesweep.pcm ../dataqa351/a51sinesweep.pcm
-diff ../dataqa350/i30_103sinesweep_half.pcm ../dataqa351/i30_103sinesweep_half.pcm
-diff ../dataqa350/i60_103sinesweep_half.pcm ../dataqa351/i60_103sinesweep_half.pcm
-diff ../dataqa350/i30_104sinesweep_half.pcm ../dataqa351/i30_104sinesweep_half.pcm
-diff ../dataqa350/i60_104sinesweep_half.pcm ../dataqa351/i60_104sinesweep_half.pcm
-diff ../dataqa350/i30_105sinesweep_half.pcm ../dataqa351/i30_105sinesweep_half.pcm
-diff ../dataqa350/i60_105sinesweep_half.pcm ../dataqa351/i60_105sinesweep_half.pcm
-diff ../dataqa350/i30_106sinesweep_half.pcm ../dataqa351/i30_106sinesweep_half.pcm
-diff ../dataqa350/i60_106sinesweep_half.pcm ../dataqa351/i60_106sinesweep_half.pcm
-diff ../dataqa350/i30_107sinesweep_half.pcm ../dataqa351/i30_107sinesweep_half.pcm
-diff ../dataqa350/i60_107sinesweep_half.pcm ../dataqa351/i60_107sinesweep_half.pcm
-diff ../dataqa350/i30_108sinesweep_half.pcm ../dataqa351/i30_108sinesweep_half.pcm
-diff ../dataqa350/i60_108sinesweep_half.pcm ../dataqa351/i60_108sinesweep_half.pcm
-diff ../dataqa350/a52sinesweep_half.pcm ../dataqa351/a52sinesweep_half.pcm
-diff ../dataqa350/a53sinesweep_half.pcm ../dataqa351/a53sinesweep_half.pcm
-diff ../dataqa350/a54sinesweep_half.pcm ../dataqa351/a54sinesweep_half.pcm
-diff ../dataqa350/i30_109speechmusic.pcm ../dataqa351/i30_109speechmusic.pcm
-diff ../dataqa350/i60_109speechmusic.pcm ../dataqa351/i60_109speechmusic.pcm
-diff ../dataqa350/i30_110speechmusic.pcm ../dataqa351/i30_110speechmusic.pcm
-diff ../dataqa350/i60_110speechmusic.pcm ../dataqa351/i60_110speechmusic.pcm
-diff ../dataqa350/i30_111speechmusic.pcm ../dataqa351/i30_111speechmusic.pcm
-diff ../dataqa350/i60_111speechmusic.pcm ../dataqa351/i60_111speechmusic.pcm
-diff ../dataqa350/i30_112speechmusic.pcm ../dataqa351/i30_112speechmusic.pcm
-diff ../dataqa350/i60_112speechmusic.pcm ../dataqa351/i60_112speechmusic.pcm
-diff ../dataqa350/i30_113speechmusic.pcm ../dataqa351/i30_113speechmusic.pcm
-diff ../dataqa350/i60_113speechmusic.pcm ../dataqa351/i60_113speechmusic.pcm
-diff ../dataqa350/i30_114speechmusic.pcm ../dataqa351/i30_114speechmusic.pcm
-diff ../dataqa350/i60_114speechmusic.pcm ../dataqa351/i60_114speechmusic.pcm
-diff ../dataqa350/a55speechmusic.pcm ../dataqa351/a55speechmusic.pcm
-diff ../dataqa350/a56speechmusic.pcm ../dataqa351/a56speechmusic.pcm
-diff ../dataqa350/a57speechmusic.pcm ../dataqa351/a57speechmusic.pcm
-diff ../dataqa350/i30_115speechmusic_nb.pcm ../dataqa351/i30_115speechmusic_nb.pcm
-diff ../dataqa350/i60_115speechmusic_nb.pcm ../dataqa351/i60_115speechmusic_nb.pcm
-diff ../dataqa350/i30_116speechmusic_nb.pcm ../dataqa351/i30_116speechmusic_nb.pcm
-diff ../dataqa350/i60_116speechmusic_nb.pcm ../dataqa351/i60_116speechmusic_nb.pcm
-diff ../dataqa350/i30_117speechmusic_nb.pcm ../dataqa351/i30_117speechmusic_nb.pcm
-diff ../dataqa350/i60_117speechmusic_nb.pcm ../dataqa351/i60_117speechmusic_nb.pcm
-diff ../dataqa350/i30_118speechmusic_nb.pcm ../dataqa351/i30_118speechmusic_nb.pcm
-diff ../dataqa350/i60_118speechmusic_nb.pcm ../dataqa351/i60_118speechmusic_nb.pcm
-diff ../dataqa350/i30_119speechmusic_nb.pcm ../dataqa351/i30_119speechmusic_nb.pcm
-diff ../dataqa350/i60_119speechmusic_nb.pcm ../dataqa351/i60_119speechmusic_nb.pcm
-diff ../dataqa350/i30_120speechmusic_nb.pcm ../dataqa351/i30_120speechmusic_nb.pcm
-diff ../dataqa350/i60_120speechmusic_nb.pcm ../dataqa351/i60_120speechmusic_nb.pcm
-diff ../dataqa350/a58speechmusic_nb.pcm ../dataqa351/a58speechmusic_nb.pcm
-diff ../dataqa350/a59speechmusic_nb.pcm ../dataqa351/a59speechmusic_nb.pcm
-diff ../dataqa350/a60speechmusic_nb.pcm ../dataqa351/a60speechmusic_nb.pcm
-diff ../dataqa350/i30_121speechoffice0dB.pcm ../dataqa351/i30_121speechoffice0dB.pcm
-diff ../dataqa350/i60_121speechoffice0dB.pcm ../dataqa351/i60_121speechoffice0dB.pcm
-diff ../dataqa350/i30_122speechoffice0dB.pcm ../dataqa351/i30_122speechoffice0dB.pcm
-diff ../dataqa350/i60_122speechoffice0dB.pcm ../dataqa351/i60_122speechoffice0dB.pcm
-diff ../dataqa350/i30_123speechoffice0dB.pcm ../dataqa351/i30_123speechoffice0dB.pcm
-diff ../dataqa350/i60_123speechoffice0dB.pcm ../dataqa351/i60_123speechoffice0dB.pcm
-diff ../dataqa350/i30_124speechoffice0dB.pcm ../dataqa351/i30_124speechoffice0dB.pcm
-diff ../dataqa350/i60_124speechoffice0dB.pcm ../dataqa351/i60_124speechoffice0dB.pcm
-diff ../dataqa350/i30_125speechoffice0dB.pcm ../dataqa351/i30_125speechoffice0dB.pcm
-diff ../dataqa350/i60_125speechoffice0dB.pcm ../dataqa351/i60_125speechoffice0dB.pcm
-diff ../dataqa350/i30_126speechoffice0dB.pcm ../dataqa351/i30_126speechoffice0dB.pcm
-diff ../dataqa350/i60_126speechoffice0dB.pcm ../dataqa351/i60_126speechoffice0dB.pcm
-diff ../dataqa350/a61speechoffice0dB.pcm ../dataqa351/a61speechoffice0dB.pcm
-diff ../dataqa350/a62speechoffice0dB.pcm ../dataqa351/a62speechoffice0dB.pcm
-diff ../dataqa350/a63speechoffice0dB.pcm ../dataqa351/a63speechoffice0dB.pcm
-diff ../dataqa350/i30_127speech_and_misc_NB.pcm ../dataqa351/i30_127speech_and_misc_NB.pcm
-diff ../dataqa350/i60_127speech_and_misc_NB.pcm ../dataqa351/i60_127speech_and_misc_NB.pcm
-diff ../dataqa350/i30_128speech_and_misc_NB.pcm ../dataqa351/i30_128speech_and_misc_NB.pcm
-diff ../dataqa350/i60_128speech_and_misc_NB.pcm ../dataqa351/i60_128speech_and_misc_NB.pcm
-diff ../dataqa350/i30_129speech_and_misc_NB.pcm ../dataqa351/i30_129speech_and_misc_NB.pcm
-diff ../dataqa350/i60_129speech_and_misc_NB.pcm ../dataqa351/i60_129speech_and_misc_NB.pcm
-diff ../dataqa350/i30_130speech_and_misc_NB.pcm ../dataqa351/i30_130speech_and_misc_NB.pcm
-diff ../dataqa350/i60_130speech_and_misc_NB.pcm ../dataqa351/i60_130speech_and_misc_NB.pcm
-diff ../dataqa350/i30_131speech_and_misc_NB.pcm ../dataqa351/i30_131speech_and_misc_NB.pcm
-diff ../dataqa350/i60_131speech_and_misc_NB.pcm ../dataqa351/i60_131speech_and_misc_NB.pcm
-diff ../dataqa350/i30_132speech_and_misc_NB.pcm ../dataqa351/i30_132speech_and_misc_NB.pcm
-diff ../dataqa350/i60_132speech_and_misc_NB.pcm ../dataqa351/i60_132speech_and_misc_NB.pcm
-diff ../dataqa350/a64speech_and_misc_NB.pcm ../dataqa351/a64speech_and_misc_NB.pcm
-diff ../dataqa350/a65speech_and_misc_NB.pcm ../dataqa351/a65speech_and_misc_NB.pcm
-diff ../dataqa350/a66speech_and_misc_NB.pcm ../dataqa351/a66speech_and_misc_NB.pcm
-diff ../dataqa350/i30_133speech_and_misc_WB.pcm ../dataqa351/i30_133speech_and_misc_WB.pcm
-diff ../dataqa350/i60_133speech_and_misc_WB.pcm ../dataqa351/i60_133speech_and_misc_WB.pcm
-diff ../dataqa350/i30_134speech_and_misc_WB.pcm ../dataqa351/i30_134speech_and_misc_WB.pcm
-diff ../dataqa350/i60_134speech_and_misc_WB.pcm ../dataqa351/i60_134speech_and_misc_WB.pcm
-diff ../dataqa350/i30_135speech_and_misc_WB.pcm ../dataqa351/i30_135speech_and_misc_WB.pcm
-diff ../dataqa350/i60_135speech_and_misc_WB.pcm ../dataqa351/i60_135speech_and_misc_WB.pcm
-diff ../dataqa350/i30_136speech_and_misc_WB.pcm ../dataqa351/i30_136speech_and_misc_WB.pcm
-diff ../dataqa350/i60_136speech_and_misc_WB.pcm ../dataqa351/i60_136speech_and_misc_WB.pcm
-diff ../dataqa350/i30_137speech_and_misc_WB.pcm ../dataqa351/i30_137speech_and_misc_WB.pcm
-diff ../dataqa350/i60_137speech_and_misc_WB.pcm ../dataqa351/i60_137speech_and_misc_WB.pcm
-diff ../dataqa350/i30_138speech_and_misc_WB.pcm ../dataqa351/i30_138speech_and_misc_WB.pcm
-diff ../dataqa350/i60_138speech_and_misc_WB.pcm ../dataqa351/i60_138speech_and_misc_WB.pcm
-diff ../dataqa350/a67speech_and_misc_WB.pcm ../dataqa351/a67speech_and_misc_WB.pcm
-diff ../dataqa350/a68speech_and_misc_WB.pcm ../dataqa351/a68speech_and_misc_WB.pcm
-diff ../dataqa350/a69speech_and_misc_WB.pcm ../dataqa351/a69speech_and_misc_WB.pcm
-diff ../dataqa350/i30_139testM4.pcm ../dataqa351/i30_139testM4.pcm
-diff ../dataqa350/i60_139testM4.pcm ../dataqa351/i60_139testM4.pcm
-diff ../dataqa350/i30_140testM4.pcm ../dataqa351/i30_140testM4.pcm
-diff ../dataqa350/i60_140testM4.pcm ../dataqa351/i60_140testM4.pcm
-diff ../dataqa350/i30_141testM4.pcm ../dataqa351/i30_141testM4.pcm
-diff ../dataqa350/i60_141testM4.pcm ../dataqa351/i60_141testM4.pcm
-diff ../dataqa350/i30_142testM4.pcm ../dataqa351/i30_142testM4.pcm
-diff ../dataqa350/i60_142testM4.pcm ../dataqa351/i60_142testM4.pcm
-diff ../dataqa350/i30_143testM4.pcm ../dataqa351/i30_143testM4.pcm
-diff ../dataqa350/i60_143testM4.pcm ../dataqa351/i60_143testM4.pcm
-diff ../dataqa350/i30_144testM4.pcm ../dataqa351/i30_144testM4.pcm
-diff ../dataqa350/i60_144testM4.pcm ../dataqa351/i60_144testM4.pcm
-diff ../dataqa350/a70testM4.pcm ../dataqa351/a70testM4.pcm
-diff ../dataqa350/a71testM4.pcm ../dataqa351/a71testM4.pcm
-diff ../dataqa350/a72testM4.pcm ../dataqa351/a72testM4.pcm
-diff ../dataqa350/i30_145testM4D_rev.pcm ../dataqa351/i30_145testM4D_rev.pcm
-diff ../dataqa350/i60_145testM4D_rev.pcm ../dataqa351/i60_145testM4D_rev.pcm
-diff ../dataqa350/i30_146testM4D_rev.pcm ../dataqa351/i30_146testM4D_rev.pcm
-diff ../dataqa350/i60_146testM4D_rev.pcm ../dataqa351/i60_146testM4D_rev.pcm
-diff ../dataqa350/i30_147testM4D_rev.pcm ../dataqa351/i30_147testM4D_rev.pcm
-diff ../dataqa350/i60_147testM4D_rev.pcm ../dataqa351/i60_147testM4D_rev.pcm
-diff ../dataqa350/i30_148testM4D_rev.pcm ../dataqa351/i30_148testM4D_rev.pcm
-diff ../dataqa350/i60_148testM4D_rev.pcm ../dataqa351/i60_148testM4D_rev.pcm
-diff ../dataqa350/i30_149testM4D_rev.pcm ../dataqa351/i30_149testM4D_rev.pcm
-diff ../dataqa350/i60_149testM4D_rev.pcm ../dataqa351/i60_149testM4D_rev.pcm
-diff ../dataqa350/i30_150testM4D_rev.pcm ../dataqa351/i30_150testM4D_rev.pcm
-diff ../dataqa350/i60_150testM4D_rev.pcm ../dataqa351/i60_150testM4D_rev.pcm
-diff ../dataqa350/a73testM4D_rev.pcm ../dataqa351/a73testM4D_rev.pcm
-diff ../dataqa350/a74testM4D_rev.pcm ../dataqa351/a74testM4D_rev.pcm
-diff ../dataqa350/a75testM4D_rev.pcm ../dataqa351/a75testM4D_rev.pcm
-diff ../dataqa350/i30_151testM4D.pcm ../dataqa351/i30_151testM4D.pcm
-diff ../dataqa350/i60_151testM4D.pcm ../dataqa351/i60_151testM4D.pcm
-diff ../dataqa350/i30_152testM4D.pcm ../dataqa351/i30_152testM4D.pcm
-diff ../dataqa350/i60_152testM4D.pcm ../dataqa351/i60_152testM4D.pcm
-diff ../dataqa350/i30_153testM4D.pcm ../dataqa351/i30_153testM4D.pcm
-diff ../dataqa350/i60_153testM4D.pcm ../dataqa351/i60_153testM4D.pcm
-diff ../dataqa350/i30_154testM4D.pcm ../dataqa351/i30_154testM4D.pcm
-diff ../dataqa350/i60_154testM4D.pcm ../dataqa351/i60_154testM4D.pcm
-diff ../dataqa350/i30_155testM4D.pcm ../dataqa351/i30_155testM4D.pcm
-diff ../dataqa350/i60_155testM4D.pcm ../dataqa351/i60_155testM4D.pcm
-diff ../dataqa350/i30_156testM4D.pcm ../dataqa351/i30_156testM4D.pcm
-diff ../dataqa350/i60_156testM4D.pcm ../dataqa351/i60_156testM4D.pcm
-diff ../dataqa350/a76testM4D.pcm ../dataqa351/a76testM4D.pcm
-diff ../dataqa350/a77testM4D.pcm ../dataqa351/a77testM4D.pcm
-diff ../dataqa350/a78testM4D.pcm ../dataqa351/a78testM4D.pcm
-diff ../dataqa350/i30_157testfile.pcm ../dataqa351/i30_157testfile.pcm
-diff ../dataqa350/i60_157testfile.pcm ../dataqa351/i60_157testfile.pcm
-diff ../dataqa350/i30_158testfile.pcm ../dataqa351/i30_158testfile.pcm
-diff ../dataqa350/i60_158testfile.pcm ../dataqa351/i60_158testfile.pcm
-diff ../dataqa350/i30_159testfile.pcm ../dataqa351/i30_159testfile.pcm
-diff ../dataqa350/i60_159testfile.pcm ../dataqa351/i60_159testfile.pcm
-diff ../dataqa350/i30_160testfile.pcm ../dataqa351/i30_160testfile.pcm
-diff ../dataqa350/i60_160testfile.pcm ../dataqa351/i60_160testfile.pcm
-diff ../dataqa350/i30_161testfile.pcm ../dataqa351/i30_161testfile.pcm
-diff ../dataqa350/i60_161testfile.pcm ../dataqa351/i60_161testfile.pcm
-diff ../dataqa350/i30_162testfile.pcm ../dataqa351/i30_162testfile.pcm
-diff ../dataqa350/i60_162testfile.pcm ../dataqa351/i60_162testfile.pcm
-diff ../dataqa350/a79testfile.pcm ../dataqa351/a79testfile.pcm
-diff ../dataqa350/a80testfile.pcm ../dataqa351/a80testfile.pcm
-diff ../dataqa350/a81testfile.pcm ../dataqa351/a81testfile.pcm
-diff ../dataqa350/i30_163tone_cisco.pcm ../dataqa351/i30_163tone_cisco.pcm
-diff ../dataqa350/i60_163tone_cisco.pcm ../dataqa351/i60_163tone_cisco.pcm
-diff ../dataqa350/i30_164tone_cisco.pcm ../dataqa351/i30_164tone_cisco.pcm
-diff ../dataqa350/i60_164tone_cisco.pcm ../dataqa351/i60_164tone_cisco.pcm
-diff ../dataqa350/i30_165tone_cisco.pcm ../dataqa351/i30_165tone_cisco.pcm
-diff ../dataqa350/i60_165tone_cisco.pcm ../dataqa351/i60_165tone_cisco.pcm
-diff ../dataqa350/i30_166tone_cisco.pcm ../dataqa351/i30_166tone_cisco.pcm
-diff ../dataqa350/i60_166tone_cisco.pcm ../dataqa351/i60_166tone_cisco.pcm
-diff ../dataqa350/i30_167tone_cisco.pcm ../dataqa351/i30_167tone_cisco.pcm
-diff ../dataqa350/i60_167tone_cisco.pcm ../dataqa351/i60_167tone_cisco.pcm
-diff ../dataqa350/i30_168tone_cisco.pcm ../dataqa351/i30_168tone_cisco.pcm
-diff ../dataqa350/i60_168tone_cisco.pcm ../dataqa351/i60_168tone_cisco.pcm
-diff ../dataqa350/a82tone_cisco.pcm ../dataqa351/a82tone_cisco.pcm
-diff ../dataqa350/a83tone_cisco.pcm ../dataqa351/a83tone_cisco.pcm
-diff ../dataqa350/a84tone_cisco.pcm ../dataqa351/a84tone_cisco.pcm
-diff ../dataqa350/i30_169tone_cisco_long.pcm ../dataqa351/i30_169tone_cisco_long.pcm
-diff ../dataqa350/i60_169tone_cisco_long.pcm ../dataqa351/i60_169tone_cisco_long.pcm
-diff ../dataqa350/i30_170tone_cisco_long.pcm ../dataqa351/i30_170tone_cisco_long.pcm
-diff ../dataqa350/i60_170tone_cisco_long.pcm ../dataqa351/i60_170tone_cisco_long.pcm
-diff ../dataqa350/i30_171tone_cisco_long.pcm ../dataqa351/i30_171tone_cisco_long.pcm
-diff ../dataqa350/i60_171tone_cisco_long.pcm ../dataqa351/i60_171tone_cisco_long.pcm
-diff ../dataqa350/i30_172tone_cisco_long.pcm ../dataqa351/i30_172tone_cisco_long.pcm
-diff ../dataqa350/i60_172tone_cisco_long.pcm ../dataqa351/i60_172tone_cisco_long.pcm
-diff ../dataqa350/i30_173tone_cisco_long.pcm ../dataqa351/i30_173tone_cisco_long.pcm
-diff ../dataqa350/i60_173tone_cisco_long.pcm ../dataqa351/i60_173tone_cisco_long.pcm
-diff ../dataqa350/i30_174tone_cisco_long.pcm ../dataqa351/i30_174tone_cisco_long.pcm
-diff ../dataqa350/i60_174tone_cisco_long.pcm ../dataqa351/i60_174tone_cisco_long.pcm
-diff ../dataqa350/a85tone_cisco_long.pcm ../dataqa351/a85tone_cisco_long.pcm
-diff ../dataqa350/a86tone_cisco_long.pcm ../dataqa351/a86tone_cisco_long.pcm
-diff ../dataqa350/a87tone_cisco_long.pcm ../dataqa351/a87tone_cisco_long.pcm
-diff ../dataqa350/i30_175wb_contspeech.pcm ../dataqa351/i30_175wb_contspeech.pcm
-diff ../dataqa350/i60_175wb_contspeech.pcm ../dataqa351/i60_175wb_contspeech.pcm
-diff ../dataqa350/i30_176wb_contspeech.pcm ../dataqa351/i30_176wb_contspeech.pcm
-diff ../dataqa350/i60_176wb_contspeech.pcm ../dataqa351/i60_176wb_contspeech.pcm
-diff ../dataqa350/i30_177wb_contspeech.pcm ../dataqa351/i30_177wb_contspeech.pcm
-diff ../dataqa350/i60_177wb_contspeech.pcm ../dataqa351/i60_177wb_contspeech.pcm
-diff ../dataqa350/i30_178wb_contspeech.pcm ../dataqa351/i30_178wb_contspeech.pcm
-diff ../dataqa350/i60_178wb_contspeech.pcm ../dataqa351/i60_178wb_contspeech.pcm
-diff ../dataqa350/i30_179wb_contspeech.pcm ../dataqa351/i30_179wb_contspeech.pcm
-diff ../dataqa350/i60_179wb_contspeech.pcm ../dataqa351/i60_179wb_contspeech.pcm
-diff ../dataqa350/i30_180wb_contspeech.pcm ../dataqa351/i30_180wb_contspeech.pcm
-diff ../dataqa350/i60_180wb_contspeech.pcm ../dataqa351/i60_180wb_contspeech.pcm
-diff ../dataqa350/a88wb_contspeech.pcm ../dataqa351/a88wb_contspeech.pcm
-diff ../dataqa350/a89wb_contspeech.pcm ../dataqa351/a89wb_contspeech.pcm
-diff ../dataqa350/a90wb_contspeech.pcm ../dataqa351/a90wb_contspeech.pcm
-diff ../dataqa350/i30_181wb_speech_office25db.pcm ../dataqa351/i30_181wb_speech_office25db.pcm
-diff ../dataqa350/i60_181wb_speech_office25db.pcm ../dataqa351/i60_181wb_speech_office25db.pcm
-diff ../dataqa350/i30_182wb_speech_office25db.pcm ../dataqa351/i30_182wb_speech_office25db.pcm
-diff ../dataqa350/i60_182wb_speech_office25db.pcm ../dataqa351/i60_182wb_speech_office25db.pcm
-diff ../dataqa350/i30_183wb_speech_office25db.pcm ../dataqa351/i30_183wb_speech_office25db.pcm
-diff ../dataqa350/i60_183wb_speech_office25db.pcm ../dataqa351/i60_183wb_speech_office25db.pcm
-diff ../dataqa350/i30_184wb_speech_office25db.pcm ../dataqa351/i30_184wb_speech_office25db.pcm
-diff ../dataqa350/i60_184wb_speech_office25db.pcm ../dataqa351/i60_184wb_speech_office25db.pcm
-diff ../dataqa350/i30_185wb_speech_office25db.pcm ../dataqa351/i30_185wb_speech_office25db.pcm
-diff ../dataqa350/i60_185wb_speech_office25db.pcm ../dataqa351/i60_185wb_speech_office25db.pcm
-diff ../dataqa350/i30_186wb_speech_office25db.pcm ../dataqa351/i30_186wb_speech_office25db.pcm
-diff ../dataqa350/i60_186wb_speech_office25db.pcm ../dataqa351/i60_186wb_speech_office25db.pcm
-diff ../dataqa350/a91wb_speech_office25db.pcm ../dataqa351/a91wb_speech_office25db.pcm
-diff ../dataqa350/a92wb_speech_office25db.pcm ../dataqa351/a92wb_speech_office25db.pcm
-diff ../dataqa350/a93wb_speech_office25db.pcm ../dataqa351/a93wb_speech_office25db.pcm
-diff ../dataqa350/a30_1DTMF_16kHz_short.pcm ../dataqa351/a30_1DTMF_16kHz_short.pcm
-diff ../dataqa350/a60_1DTMF_16kHz_short.pcm ../dataqa351/a60_1DTMF_16kHz_short.pcm
-diff ../dataqa350/a30_2ltest_speech_noisy.pcm ../dataqa351/a30_2ltest_speech_noisy.pcm
-diff ../dataqa350/a60_2ltest_speech_noisy.pcm ../dataqa351/a60_2ltest_speech_noisy.pcm
-diff ../dataqa350/a30_3misc2.pcm ../dataqa351/a30_3misc2.pcm
-diff ../dataqa350/a60_3misc2.pcm ../dataqa351/a60_3misc2.pcm
-diff ../dataqa350/a30_4sinesweep.pcm ../dataqa351/a30_4sinesweep.pcm
-diff ../dataqa350/a60_4sinesweep.pcm ../dataqa351/a60_4sinesweep.pcm
-diff ../dataqa350/a30_5speechmusic.pcm ../dataqa351/a30_5speechmusic.pcm
-diff ../dataqa350/a60_5speechmusic.pcm ../dataqa351/a60_5speechmusic.pcm
-diff ../dataqa350/a30_6tone_cisco.pcm ../dataqa351/a30_6tone_cisco.pcm
-diff ../dataqa350/a60_6tone_cisco.pcm ../dataqa351/a60_6tone_cisco.pcm
-diff ../dataqa350/a60_7tone_cisco.pcm ../dataqa351/a60_7tone_cisco.pcm
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/diffiSACPLC.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/diffiSACPLC.txt
deleted file mode 100644
index 9e3629b..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/diffiSACPLC.txt
+++ /dev/null
@@ -1,20 +0,0 @@
-#!/bin/bash
-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
-
-LOGFILE=logplc.txt
-echo "START PLC TEST" > $LOGFILE
-
-OUTDIR1=../dataqaplc_0
-OUTDIR2=../dataqaplc_1
-
-diff $OUTDIR1/outplc1.pcm $OUTDIR2/outplc1.pcm
-diff $OUTDIR1/outplc2.pcm $OUTDIR2/outplc2.pcm
-diff $OUTDIR1/outplc3.pcm $OUTDIR2/outplc3.pcm
-diff $OUTDIR1/outplc4.pcm $OUTDIR2/outplc4.pcm
-diff $OUTDIR1/outplc5.pcm $OUTDIR2/outplc5.pcm
-diff $OUTDIR1/outplc6.pcm $OUTDIR2/outplc6.pcm
-
-echo DONE!
-
-
-
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACLongtest.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACLongtest.txt
deleted file mode 100644
index eeffc0c..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACLongtest.txt
+++ /dev/null
@@ -1,61 +0,0 @@
-#!/bin/bash
-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
-
-LOGFILE=logNormal.txt
-echo "START ISAC TEST" > $LOGFILE
-echo >> $LOGFILE
-
-ISAC=../Release/kenny.exe
-ISACFIXFLOAT=../Release/testFixFloat.exe
-
-INFILES=$(cat InputFiles.txt)
-SUBSET=$(cat InputFilesFew.txt)
-CHANNELFILES=$(cat ChannelFiles.txt)
-CHANNELLIST=($(cat ChannelFiles.txt))
-INDIR=../data/orig
-OUTDIR=../dataqa
-mkdir -p $OUTDIR
-
-TARGETRATE=(10000 15000 20000 25000 30000 32000)
-#echo ${CHANNELFILES[1]}
-
-index1=0
-index2=0
-
-for file in $INFILES # loop over all input files
- do
-
- for rate in ${TARGETRATE[*]}
- do
- let "index1=index1+1"
- $ISAC -I $rate -FL 30 $INDIR/"$file" $OUTDIR/i30_$index1"$file" >> $LOGFILE
- $ISAC -I $rate -FL 60 $INDIR/"$file" $OUTDIR/i60_$index1"$file" >> $LOGFILE
- done
- for channel in $CHANNELFILES
- do
- let "index2=index2+1"
- $ISAC $INDIR/$channel $INDIR/"$file" $OUTDIR/a$index2"$file" >> $LOGFILE
- done
-
-done
-
-index1=0
-
-for file in $SUBSET # loop over the subset of input files
- do
- let "index1=index1+1"
- $ISAC $INDIR/${CHANNELLIST[0]} -FL 30 -FIXED_FL $INDIR/"$file" $OUTDIR/a30_$index1"$file" >> $LOGFILE
- $ISAC $INDIR/${CHANNELLIST[0]} -FL 60 -FIXED_FL $INDIR/"$file" $OUTDIR/a60_$index1"$file" >> $LOGFILE
-done
-
-let "index1=index1+1"
- $ISAC $INDIR/${CHANNELLIST[0]} -INITRATE 25000 -FL 30 $INDIR/"$file" $OUTDIR/a60_$index1"$file" >> $LOGFILE
-
-# Run fault test
-
-#./runiSACfault.txt
-
-echo DONE!
-
-
-
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACNB.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACNB.txt
deleted file mode 100644
index 605595c..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACNB.txt
+++ /dev/null
@@ -1,45 +0,0 @@
-#!/bin/bash
-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
-
-LOGFILE=logNB.txt
-echo "START NARROWBAND TEST" > $LOGFILE
-echo >> $LOGFILE
-
-ISAC=../Release/kenny.exe
-ISACFIXFLOAT=../Release/testFixFloat.exe
-
-INFILES=$(cat InputFiles.txt)
-SUBSET=$(cat InputFilesFew.txt)
-CHANNELFILES=$(cat ChannelFiles.txt)
-CHANNELLIST=($(cat ChannelFiles.txt))
-INDIR=../data/orig
-OUTDIR=../dataqaNB
-mkdir -p $OUTDIR
-
-TARGETRATE=(10000 15000 20000 25000 30000 32000)
-#echo ${CHANNELFILES[1]}
-
-index1=0
-index2=0
-
-# Narrowband Interfaces
-
-for file in $SUBSET # loop over all input files
- do
- for rate in ${TARGETRATE[*]}
- do
- let "index1=index1+1"
- $ISAC $rate -FL 30 -NB 1 $INDIR/"$file" $OUTDIR/nb130_$index1"$file" >> $LOGFILE
- $ISAC $rate -FL 60 -NB 1 $INDIR/"$file" $OUTDIR/nb160_$index1"$file" >> $LOGFILE
- $ISAC $rate -FL 30 -NB 2 $INDIR/"$file" $OUTDIR/nb230_$index1"$file" >> $LOGFILE
- $ISAC $rate -FL 60 -NB 2 $INDIR/"$file" $OUTDIR/nb260_$index1"$file" >> $LOGFILE
- $ISAC $rate -FL 30 -NB 2 -PL 10 $INDIR/"$file" $OUTDIR/nb2plc30_$index1"$file" >> $LOGFILE
- $ISAC $rate -FL 60 -NB 2 -PL 10 $INDIR/"$file" $OUTDIR/nb2plc60_$index1"$file" >> $LOGFILE
- done
-
-done
-
-echo DONE!
-
-
-
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACPLC.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACPLC.txt
deleted file mode 100644
index 6bee6f7..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACPLC.txt
+++ /dev/null
@@ -1,23 +0,0 @@
-#!/bin/bash
-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
-
-LOGFILE=logplc.txt
-echo "START PLC TEST" > $LOGFILE
-
-ISAC=../Release/kenny.exe
-
-INDIR=../data/orig
-OUTDIR=../dataqaplc_0
-mkdir -p $OUTDIR
-
-$ISAC 12000 -PL 15 $INDIR/speechmusic.pcm $OUTDIR/outplc1.pcm
-$ISAC 20000 -PL 15 $INDIR/speechmusic.pcm $OUTDIR/outplc2.pcm
-$ISAC 32000 -PL 15 $INDIR/speechmusic.pcm $OUTDIR/outplc3.pcm
-$ISAC 12000 -PL 15 $INDIR/tone_cisco.pcm $OUTDIR/outplc4.pcm
-$ISAC 20000 -PL 15 $INDIR/tone_cisco.pcm $OUTDIR/outplc5.pcm
-$ISAC 32000 -PL 15 $INDIR/tone_cisco.pcm $OUTDIR/outplc6.pcm
-
-echo DONE!
-
-
-
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACRate.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACRate.txt
deleted file mode 100644
index d8403e0..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACRate.txt
+++ /dev/null
@@ -1,23 +0,0 @@
-#!/bin/bash
-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
-
-LOGG=loggRate.txt
-OUTDIR=../dataqaRate
-mkdir -p $OUTDIR
-
-../Release/kenny.exe 13000 -FIXED_FL -FL 30 -MAX 100 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_1.pcm > $LOGG
-../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 30 -MAXRATE 32000 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_2.pcm >> $LOGG
-../Release/kenny.exe 13000 -FIXED_FL -FL 30 -MAX 100 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_3.pcm >> $LOGG
-../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 30 -MAXRATE 32000 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_4.pcm >> $LOGG
-../Release/kenny.exe 13000 -FIXED_FL -FL 60 -MAX 100 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_5.pcm >> $LOGG
-../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 60 -MAXRATE 32000 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_6.pcm >> $LOGG
-../Release/kenny.exe 13000 -INIT_RATE 32000 -FIXED_FL -FL 60 -MAX 100 ../data/orig/sawsweep_380_60.pcm $OUTDIR/out_napi_7.pcm >> $LOGG
-
-../Release/kenny.exe 13000 -FIXED_FL -FL 30 -MAX 100 ../data/orig/longspeech.pcm $OUTDIR/out_napi_11.pcm >> $LOGG
-../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 30 -MAXRATE 32000 ../data/orig/longspeech.pcm $OUTDIR/out_napi_12.pcm >> $LOGG
-../Release/kenny.exe 13000 -FIXED_FL -FL 30 -MAX 100 ../data/orig/longspeech.pcm $OUTDIR/out_napi_13.pcm >> $LOGG
-../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 30 -MAXRATE 32000 ../data/orig/longspeech.pcm $OUTDIR/out_napi_14.pcm >> $LOGG
-../Release/kenny.exe 13000 -FIXED_FL -FL 60 -MAX 100 ../data/orig/longspeech.pcm $OUTDIR/out_napi_15.pcm >> $LOGG
-../Release/kenny.exe ../data/orig/bottlenecks.txt -FIXED_FL -FL 60 -MAXRATE 32000 ../data/orig/longspeech.pcm $OUTDIR/out_napi_16.pcm >> $LOGG
-../Release/kenny.exe 13000 -INIT_RATE 32000 -FIXED_FL -FL 60 -MAX 100 ../data/orig/longspeech.pcm $OUTDIR/out_napi_17.pcm >> $LOGG
-
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACfault.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACfault.txt
deleted file mode 100644
index f4d9478..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACfault.txt
+++ /dev/null
@@ -1,40 +0,0 @@
-#!/bin/bash
-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
-
-LOGFILE=logfault.txt
-echo "START FAULT TEST" > $LOGFILE
-
-ISAC=../Release/kenny.exe
-ISACFIXFLOAT=../Release/testFixFloat.exe
-
-INFILES=$(cat InputFiles.txt)
-SUBSET=$(cat InputFilesFew.txt)
-CHANNELFILES=$(cat ChannelFiles.txt)
-CHANNELLIST=($(cat ChannelFiles.txt))
-INDIR=../data/orig
-OUTDIR=../dataqaft
-mkdir -p $OUTDIR
-
-TARGETRATE=(10000 15000 20000 25000 30000 32000)
-FAULTTEST=(1 2 3 4 5 6 7 9)
-
-index1=0
-
-file=wb_contspeech.pcm
-
-# Fault test
-for testnr in ${FAULTTEST[*]}
- do
- $ISAC 32000 -F $testnr $INDIR/"$file" $OUTDIR/ft$testnr"$file" >> $LOGFILE
-done
-
-# Fault test number 10, error in bitstream
- $ISAC 32000 -F 10 $INDIR/"$file" $OUTDIR/ft10_"$file" >> $LOGFILE
- $ISAC 32000 -F 10 -PL 10 $INDIR/"$file" $OUTDIR/ft10plc_"$file" >> $LOGFILE
- $ISAC 32000 -F 10 -NB 1 $INDIR/"$file" $OUTDIR/ft10nb1_"$file" >> $LOGFILE
- $ISAC 32000 -F 10 -NB 2 -PL 10 $INDIR/"$file" $OUTDIR/ft10nb2_"$file" >> $LOGFILE
-
-echo DONE!
-
-
-
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACfixfloat.txt b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACfixfloat.txt
deleted file mode 100644
index c9e02df..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/QA/runiSACfixfloat.txt
+++ /dev/null
@@ -1,47 +0,0 @@
-#!/bin/bash
-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
-
-LOGFILE=logfxfl.txt
-echo "START FIX-FLOAT TEST" > $LOGFILE
-
-
-ISACFIXFLOAT=../testFixFloat.exe
-
-INFILES=$(cat InputFiles.txt)
-SUBSET=$(cat InputFilesFew.txt)
-CHANNELFILES=$(cat ChannelFiles.txt)
-CHANNELLIST=($(cat ChannelFiles.txt))
-INDIR=../data/orig
-OUTDIR=../dataqafxfl
-mkdir -p $OUTDIR
-
-index1=0
-
-for file in $INFILES # loop over all input files
- do
-
- for channel in $CHANNELFILES
- do
- let "index1=index1+1"
-
- $ISACFIXFLOAT $INDIR/$channel -m 1 -PLC $INDIR/"$file" $OUTDIR/flfx$index1"$file" >> $LOGFILE
- $ISACFIXFLOAT $INDIR/$channel -m 2 -PLC $INDIR/"$file" $OUTDIR/fxfl$index1"$file" >> $LOGFILE
- done
-
-done
-
-index1=0
-
-for file in $SUBSET # loop over the subset of input files
- do
- let "index1=index1+1"
- $ISACFIXFLOAT $INDIR/$channel -m 1 -NB 1 $INDIR/"$file" $OUTDIR/flfxnb1_$index1"$file" >> $LOGFILE
- $ISACFIXFLOAT $INDIR/$channel -m 2 -NB 1 $INDIR/"$file" $OUTDIR/fxflnb1_$index1"$file" >> $LOGFILE
- $ISACFIXFLOAT $INDIR/$channel -m 1 -NB 2 -PLC $INDIR/"$file" $OUTDIR/flfxnb2_$index1"$file" >> $LOGFILE
- $ISACFIXFLOAT $INDIR/$channel -m 2 -NB 2 -PLC $INDIR/"$file" $OUTDIR/fxflnb2_$index1"$file" >> $LOGFILE
-done
-
-echo DONE!
-
-
-
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/kenny.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/kenny.c
deleted file mode 100644
index 1b9c44c..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/kenny.c
+++ /dev/null
@@ -1,847 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/* kenny.c - Main function for the iSAC coder */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <time.h>
-#include <ctype.h>
-
-#include "isacfix.h"
-
-
-/* Defines */
-#define SEED_FILE "randseed.txt" /* Used when running decoder on garbage data */
-#define MAX_FRAMESAMPLES 960 /* max number of samples per frame (= 60 ms frame) */
-#define FRAMESAMPLES_10ms 160 /* number of samples per 10ms frame */
-#define FS 16000 /* sampling frequency (Hz) */
-
-/* Function for reading audio data from PCM file */
-int readframe(WebRtc_Word16 *data, FILE *inp, int length) {
-
- short k, rlen, status = 0;
-
- rlen = fread(data, sizeof(WebRtc_Word16), length, inp);
- if (rlen < length) {
- for (k = rlen; k < length; k++)
- data[k] = 0;
- status = 1;
- }
-
- return status;
-}
-
-/* Struct for bottleneck model */
-typedef struct {
- WebRtc_UWord32 send_time; /* samples */
- WebRtc_UWord32 arrival_time; /* samples */
- WebRtc_UWord32 sample_count; /* samples */
- WebRtc_UWord16 rtp_number;
-} BottleNeckModel;
-
-void get_arrival_time(int current_framesamples, /* samples */
- int packet_size, /* bytes */
- int bottleneck, /* excluding headers; bits/s */
- BottleNeckModel *BN_data)
-{
- const int HeaderSize = 35;
- int HeaderRate;
-
- HeaderRate = HeaderSize * 8 * FS / current_framesamples; /* bits/s */
-
- /* everything in samples */
- BN_data->sample_count = BN_data->sample_count + current_framesamples;
-
- BN_data->arrival_time += ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate);
- BN_data->send_time += current_framesamples;
-
- if (BN_data->arrival_time < BN_data->sample_count)
- BN_data->arrival_time = BN_data->sample_count;
-
- BN_data->rtp_number++;
-}
-
-void get_arrival_time2(int current_framesamples,
- int current_delay,
- BottleNeckModel *BN_data)
-{
- if (current_delay == -1)
- //dropped packet
- {
- BN_data->arrival_time += current_framesamples;
- }
- else if (current_delay != -2)
- {
- //
- BN_data->arrival_time += (current_framesamples + ((FS/1000) * current_delay));
- }
- //else
- //current packet has same timestamp as previous packet
-
- BN_data->rtp_number++;
-}
-
-int main(int argc, char* argv[])
-{
-
- char inname[100], outname[100], outbitsname[100], bottleneck_file[100];
- FILE *inp, *outp, *f_bn, *outbits;
- int endfile;
-
- int i, errtype, h = 0, k, packetLossPercent = 0;
- WebRtc_Word16 CodingMode;
- WebRtc_Word16 bottleneck;
- WebRtc_Word16 framesize = 30; /* ms */
- int cur_framesmpls, err = 0, lostPackets = 0;
-
- /* Runtime statistics */
- double starttime, runtime, length_file;
-
- WebRtc_Word16 stream_len = 0;
- WebRtc_Word16 framecnt, declen = 0;
- WebRtc_Word16 shortdata[FRAMESAMPLES_10ms];
- WebRtc_Word16 decoded[MAX_FRAMESAMPLES];
- WebRtc_UWord16 streamdata[500];
- WebRtc_Word16 speechType[1];
- WebRtc_Word16 prevFrameSize = 1;
- WebRtc_Word16 rateBPS = 0;
- WebRtc_Word16 fixedFL = 0;
- WebRtc_Word16 payloadSize = 0;
- WebRtc_Word32 payloadRate = 0;
- int setControlBWE = 0;
- int readLoss;
- FILE *plFile = NULL;
-
- char version_number[20];
- char tmpBit[5] = ".bit";
-
- double kbps;
- int totalbits =0;
- int totalsmpls =0;
-#ifdef _DEBUG
- FILE *fy;
-#endif
- WebRtc_Word16 testNum, testCE;
-
- FILE *fp_gns = NULL;
- int gns = 0;
- int cur_delay = 0;
- char gns_file[100];
-
- int nbTest = 0;
- WebRtc_Word16 lostFrame;
- float scale = (float)0.7;
- /* only one structure used for ISAC encoder */
- ISACFIX_MainStruct *ISAC_main_inst;
-
- /* For fault test 10, garbage data */
- FILE *seedfile;
- unsigned int random_seed = (unsigned int) time(NULL);//1196764538
-
- BottleNeckModel BN_data;
- f_bn = NULL;
-
-#ifdef _DEBUG
- fy = fopen("bit_rate.dat", "w");
- fclose(fy);
- fy = fopen("bytes_frames.dat", "w");
- fclose(fy);
-#endif
-
- readLoss = 0;
- packetLossPercent = 0;
-
- /* Handling wrong input arguments in the command line */
- if ((argc<3) || (argc>21)) {
- printf("\n\nWrong number of arguments or flag values.\n\n");
-
- printf("\n");
- WebRtcIsacfix_version(version_number);
- printf("iSAC version %s \n\n", version_number);
-
- printf("Usage:\n\n");
- printf("./kenny.exe [-F num][-I] bottleneck_value infile outfile \n\n");
- printf("with:\n");
- printf("[-I] :if -I option is specified, the coder will use\n");
- printf(" an instantaneous Bottleneck value. If not, it\n");
- printf(" will be an adaptive Bottleneck value.\n\n");
- printf("bottleneck_value :the value of the bottleneck provided either\n");
- printf(" as a fixed value (e.g. 25000) or\n");
- printf(" read from a file (e.g. bottleneck.txt)\n\n");
- printf("[-INITRATE num] :Set a new value for initial rate. Note! Only used"
- " in adaptive mode.\n\n");
- printf("[-FL num] :Set (initial) frame length in msec. Valid length"
- " are 30 and 60 msec.\n\n");
- printf("[-FIXED_FL] :Frame length will be fixed to initial value.\n\n");
- printf("[-MAX num] :Set the limit for the payload size of iSAC"
- " in bytes. \n");
- printf(" Minimum 100, maximum 400.\n\n");
- printf("[-MAXRATE num] :Set the maxrate for iSAC in bits per second. \n");
- printf(" Minimum 32000, maximum 53400.\n\n");
- printf("[-F num] :if -F option is specified, the test function\n");
- printf(" will run the iSAC API fault scenario specified"
- " by the\n");
- printf(" supplied number.\n");
- printf(" F 1 - Call encoder prior to init encoder call\n");
- printf(" F 2 - Call decoder prior to init decoder call\n");
- printf(" F 3 - Call decoder prior to encoder call\n");
- printf(" F 4 - Call decoder with a too short coded"
- " sequence\n");
- printf(" F 5 - Call decoder with a too long coded"
- " sequence\n");
- printf(" F 6 - Call decoder with random bit stream\n");
- printf(" F 7 - Call init encoder/decoder at random"
- " during a call\n");
- printf(" F 8 - Call encoder/decoder without having"
- " allocated memory for \n");
- printf(" encoder/decoder instance\n");
- printf(" F 9 - Call decodeB without calling decodeA\n");
- printf(" F 10 - Call decodeB with garbage data\n");
- printf("[-PL num] : if -PL option is specified 0<num<100 will "
- "specify the\n");
- printf(" percentage of packet loss\n\n");
- printf("[-G file] : if -G option is specified the file given is"
- " a .gns file\n");
- printf(" that represents a network profile\n\n");
- printf("[-NB num] : if -NB option, use the narrowband interfaces\n");
- printf(" num=1 => encode with narrowband encoder"
- " (infile is narrowband)\n");
- printf(" num=2 => decode with narrowband decoder"
- " (outfile is narrowband)\n\n");
- printf("[-CE num] : Test of APIs used by Conference Engine.\n");
- printf(" CE 1 - createInternal, freeInternal,"
- " getNewBitstream \n");
- printf(" CE 2 - transcode, getBWE \n");
- printf(" CE 3 - getSendBWE, setSendBWE. \n\n");
- printf("[-RTP_INIT num] : if -RTP_INIT option is specified num will be"
- " the initial\n");
- printf(" value of the rtp sequence number.\n\n");
- printf("infile : Normal speech input file\n\n");
- printf("outfile : Speech output file\n\n");
- printf("Example usage : \n\n");
- printf("./kenny.exe -I bottleneck.txt speechIn.pcm speechOut.pcm\n\n");
- exit(0);
-
- }
-
- /* Print version number */
- WebRtcIsacfix_version(version_number);
- printf("iSAC version %s \n\n", version_number);
-
- /* Loop over all command line arguments */
- CodingMode = 0;
- testNum = 0;
- testCE = 0;
- for (i = 1; i < argc-2;i++) {
- /* Instantaneous mode */
- if (!strcmp ("-I", argv[i])) {
- printf("\nInstantaneous BottleNeck\n");
- CodingMode = 1;
- i++;
- }
-
- /* Set (initial) bottleneck value */
- if (!strcmp ("-INITRATE", argv[i])) {
- rateBPS = atoi(argv[i + 1]);
- setControlBWE = 1;
- if ((rateBPS < 10000) || (rateBPS > 32000)) {
- printf("\n%d is not a initial rate. "
- "Valid values are in the range 10000 to 32000.\n", rateBPS);
- exit(0);
- }
- printf("\nNew initial rate: %d\n", rateBPS);
- i++;
- }
-
- /* Set (initial) framelength */
- if (!strcmp ("-FL", argv[i])) {
- framesize = atoi(argv[i + 1]);
- if ((framesize != 30) && (framesize != 60)) {
- printf("\n%d is not a valid frame length. "
- "Valid length are 30 and 60 msec.\n", framesize);
- exit(0);
- }
- printf("\nFrame Length: %d\n", framesize);
- i++;
- }
-
- /* Fixed frame length */
- if (!strcmp ("-FIXED_FL", argv[i])) {
- fixedFL = 1;
- setControlBWE = 1;
- }
-
- /* Set maximum allowed payload size in bytes */
- if (!strcmp ("-MAX", argv[i])) {
- payloadSize = atoi(argv[i + 1]);
- printf("Maximum Payload Size: %d\n", payloadSize);
- i++;
- }
-
- /* Set maximum rate in bytes */
- if (!strcmp ("-MAXRATE", argv[i])) {
- payloadRate = atoi(argv[i + 1]);
- printf("Maximum Rate in kbps: %d\n", payloadRate);
- i++;
- }
-
- /* Test of fault scenarious */
- if (!strcmp ("-F", argv[i])) {
- testNum = atoi(argv[i + 1]);
- printf("\nFault test: %d\n", testNum);
- if (testNum < 1 || testNum > 10) {
- printf("\n%d is not a valid Fault Scenario number."
- " Valid Fault Scenarios are numbered 1-10.\n", testNum);
- exit(0);
- }
- i++;
- }
-
- /* Packet loss test */
- if (!strcmp ("-PL", argv[i])) {
- if( isdigit( *argv[i+1] ) ) {
- packetLossPercent = atoi( argv[i+1] );
- if( (packetLossPercent < 0) | (packetLossPercent > 100) ) {
- printf( "\nInvalid packet loss perentage \n" );
- exit( 0 );
- }
- if( packetLossPercent > 0 ) {
- printf( "\nSimulating %d %% of independent packet loss\n",
- packetLossPercent );
- } else {
- printf( "\nNo Packet Loss Is Simulated \n" );
- }
- readLoss = 0;
- } else {
- readLoss = 1;
- plFile = fopen( argv[i+1], "rb" );
- if( plFile == NULL ) {
- printf( "\n couldn't open the frameloss file: %s\n", argv[i+1] );
- exit( 0 );
- }
- printf( "\nSimulating packet loss through the given "
- "channel file: %s\n", argv[i+1] );
- }
- i++;
- }
-
- /* Random packetlosses */
- if (!strcmp ("-rnd", argv[i])) {
- srand(time(NULL) );
- printf( "\n Random pattern in lossed packets \n" );
- }
-
- /* Use gns file */
- if (!strcmp ("-G", argv[i])) {
- sscanf(argv[i + 1], "%s", gns_file);
- fp_gns = fopen(gns_file, "rb");
- if (fp_gns == NULL) {
- printf("Cannot read file %s.\n", gns_file);
- exit(0);
- }
- gns = 1;
- i++;
- }
-
- /* Run Narrowband interfaces (either encoder or decoder) */
- if (!strcmp ("-NB", argv[i])) {
- nbTest = atoi(argv[i + 1]);
- i++;
- }
-
- /* Run Conference Engine APIs */
- if (!strcmp ("-CE", argv[i])) {
- testCE = atoi(argv[i + 1]);
- if (testCE==1 || testCE==2) {
- i++;
- scale = (float)atof( argv[i+1] );
- } else if (testCE < 1 || testCE > 3) {
- printf("\n%d is not a valid CE-test number, valid Fault "
- "Scenarios are numbered 1-3\n", testCE);
- exit(0);
- }
- i++;
- }
-
- /* Set initial RTP number */
- if (!strcmp ("-RTP_INIT", argv[i])) {
- i++;
- }
- }
-
- /* Get Bottleneck value */
- /* Gns files and bottleneck should not and can not be used simultaneously */
- bottleneck = atoi(argv[CodingMode+1]);
- if (bottleneck == 0 && gns == 0) {
- sscanf(argv[CodingMode+1], "%s", bottleneck_file);
- f_bn = fopen(bottleneck_file, "rb");
- if (f_bn == NULL) {
- printf("No value provided for BottleNeck and cannot read file %s\n", bottleneck_file);
- exit(0);
- } else {
- int aux_var;
- printf("reading bottleneck rates from file %s\n\n",bottleneck_file);
- if (fscanf(f_bn, "%d", &aux_var) == EOF) {
- /* Set pointer to beginning of file */
- fseek(f_bn, 0L, SEEK_SET);
- if (fscanf(f_bn, "%d", &aux_var) == EOF) {
- exit(0);
- }
- }
- bottleneck = (WebRtc_Word16)aux_var;
- /* Bottleneck is a cosine function
- * Matlab code for writing the bottleneck file:
- * BottleNeck_10ms = 20e3 + 10e3 * cos((0:5999)/5999*2*pi);
- * fid = fopen('bottleneck.txt', 'wb');
- * fprintf(fid, '%d\n', BottleNeck_10ms); fclose(fid);
- */
- }
- } else {
- f_bn = NULL;
- printf("\nfixed bottleneck rate of %d bits/s\n\n", bottleneck);
- }
-
- if (CodingMode == 0) {
- printf("\nAdaptive BottleNeck\n");
- }
-
- /* Get Input and Output files */
- sscanf(argv[argc-2], "%s", inname);
- sscanf(argv[argc-1], "%s", outname);
-
- /* Add '.bit' to output bitstream file */
- while ((int)outname[h] != 0) {
- outbitsname[h] = outname[h];
- h++;
- }
- for (k=0; k<5; k++) {
- outbitsname[h] = tmpBit[k];
- h++;
- }
- if ((inp = fopen(inname,"rb")) == NULL) {
- printf(" iSAC: Cannot read file %s\n", inname);
- exit(1);
- }
- if ((outp = fopen(outname,"wb")) == NULL) {
- printf(" iSAC: Cannot write file %s\n", outname);
- exit(1);
- }
-
- if ((outbits = fopen(outbitsname,"wb")) == NULL) {
- printf(" iSAC: Cannot write file %s\n", outbitsname);
- exit(1);
- }
- printf("\nInput:%s\nOutput:%s\n\n", inname, outname);
-
- /* Error test number 10, garbage data */
- if (testNum == 10) {
- /* Test to run decoder with garbage data */
- srand(random_seed);
-
- if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
- printf("Error: Could not open file %s\n", SEED_FILE);
- }
- else {
- fprintf(seedfile, "%u\n", random_seed);
- fclose(seedfile);
- }
- }
-
- /* Runtime statistics */
- starttime = clock()/(double)CLOCKS_PER_SEC;
-
- /* Initialize the ISAC and BN structs */
- if (testNum != 8)
- {
- if(1){
- err =WebRtcIsacfix_Create(&ISAC_main_inst);
- }else{
- /* Test the Assign functions */
- int sss;
- void *ppp;
- err =WebRtcIsacfix_AssignSize(&sss);
- ppp=malloc(sss);
- err =WebRtcIsacfix_Assign(&ISAC_main_inst,ppp);
- }
- /* Error check */
- if (err < 0) {
- printf("\n\n Error in create.\n\n");
- }
- if (testCE == 1) {
- err = WebRtcIsacfix_CreateInternal(ISAC_main_inst);
- /* Error check */
- if (err < 0) {
- printf("\n\n Error in createInternal.\n\n");
- }
- }
- }
-
- /* Init of bandwidth data */
- BN_data.send_time = 0;
- BN_data.arrival_time = 0;
- BN_data.sample_count = 0;
- BN_data.rtp_number = 0;
-
- /* Initialize encoder and decoder */
- framecnt= 0;
- endfile = 0;
- if (testNum != 1) {
- WebRtcIsacfix_EncoderInit(ISAC_main_inst, CodingMode);
- }
- if (testNum != 2) {
- WebRtcIsacfix_DecoderInit(ISAC_main_inst);
- }
-
- if (CodingMode == 1) {
- err = WebRtcIsacfix_Control(ISAC_main_inst, bottleneck, framesize);
- if (err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
- printf("\n\n Error in control: %d.\n\n", errtype);
- }
- } else if(setControlBWE == 1) {
- err = WebRtcIsacfix_ControlBwe(ISAC_main_inst, rateBPS, framesize, fixedFL);
- }
-
- if (payloadSize != 0) {
- err = WebRtcIsacfix_SetMaxPayloadSize(ISAC_main_inst, payloadSize);
- if (err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
- printf("\n\n Error in SetMaxPayloadSize: %d.\n\n", errtype);
- exit(EXIT_FAILURE);
- }
- }
- if (payloadRate != 0) {
- err = WebRtcIsacfix_SetMaxRate(ISAC_main_inst, payloadRate);
- if (err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
- printf("\n\n Error in SetMaxRateInBytes: %d.\n\n", errtype);
- exit(EXIT_FAILURE);
- }
- }
-
- *speechType = 1;
-
-
- while (endfile == 0) {
-
- if(testNum == 7 && (rand()%2 == 0)) {
- err = WebRtcIsacfix_EncoderInit(ISAC_main_inst, CodingMode);
- /* Error check */
- if (err < 0) {
- errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
- printf("\n\n Error in encoderinit: %d.\n\n", errtype);
- }
-
- err = WebRtcIsacfix_DecoderInit(ISAC_main_inst);
- /* Error check */
- if (err < 0) {
- errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
- printf("\n\n Error in decoderinit: %d.\n\n", errtype);
- }
- }
-
-
- cur_framesmpls = 0;
- while (1) {
- /* Read 10 ms speech block */
- if (nbTest != 1) {
- endfile = readframe(shortdata, inp, FRAMESAMPLES_10ms);
- } else {
- endfile = readframe(shortdata, inp, (FRAMESAMPLES_10ms/2));
- }
-
- if (testNum == 7) {
- srand(time(NULL));
- }
-
- /* iSAC encoding */
- if (!(testNum == 3 && framecnt == 0)) {
- if (nbTest != 1) {
- short bwe;
-
- /* Encode */
- stream_len = WebRtcIsacfix_Encode(ISAC_main_inst,
- shortdata,
- (WebRtc_Word16*)streamdata);
-
- /* If packet is ready, and CE testing, call the different API functions
- from the internal API. */
- if (stream_len>0) {
- if (testCE == 1) {
- err = WebRtcIsacfix_ReadBwIndex((WebRtc_Word16*)streamdata, &bwe);
- stream_len = WebRtcIsacfix_GetNewBitStream(
- ISAC_main_inst,
- bwe,
- scale,
- (WebRtc_Word16*)streamdata);
- } else if (testCE == 2) {
- /* transcode function not supported */
- } else if (testCE == 3) {
- /* Only for Function testing. The functions should normally
- not be used in this way */
-
- err = WebRtcIsacfix_GetDownLinkBwIndex(ISAC_main_inst, &bwe);
- /* Error Check */
- if (err < 0) {
- errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
- printf("\nError in getSendBWE: %d.\n", errtype);
- }
-
- err = WebRtcIsacfix_UpdateUplinkBw(ISAC_main_inst, bwe);
- /* Error Check */
- if (err < 0) {
- errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
- printf("\nError in setBWE: %d.\n", errtype);
- }
-
- }
- }
- } else {
-#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
- stream_len = WebRtcIsacfix_EncodeNb(ISAC_main_inst,
- shortdata,
- streamdata);
-#else
- stream_len = -1;
-#endif
- }
- }
- else
- {
- break;
- }
-
- if (stream_len < 0 || err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
- printf("\nError in encoder: %d.\n", errtype);
- } else {
- fwrite(streamdata, sizeof(char), stream_len, outbits);
- }
-
- cur_framesmpls += FRAMESAMPLES_10ms;
-
- /* read next bottleneck rate */
- if (f_bn != NULL) {
- int aux_var;
- if (fscanf(f_bn, "%d", &aux_var) == EOF) {
- /* Set pointer to beginning of file */
- fseek(f_bn, 0L, SEEK_SET);
- if (fscanf(f_bn, "%d", &aux_var) == EOF) {
- exit(0);
- }
- }
- bottleneck = (WebRtc_Word16)aux_var;
- if (CodingMode == 1) {
- WebRtcIsacfix_Control(ISAC_main_inst, bottleneck, framesize);
- }
- }
-
- /* exit encoder loop if the encoder returned a bitstream */
- if (stream_len != 0) break;
- }
-
- /* make coded sequence to short be inreasing */
- /* the length the decoder expects */
- if (testNum == 4) {
- stream_len += 10;
- }
-
- /* make coded sequence to long be decreasing */
- /* the length the decoder expects */
- if (testNum == 5) {
- stream_len -= 10;
- }
-
- if (testNum == 6) {
- srand(time(NULL));
- for (i = 0; i < stream_len; i++ ) {
- streamdata[i] = rand();
- }
- }
-
- /* set pointer to beginning of file */
- if (fp_gns != NULL) {
- if (fscanf(fp_gns, "%d", &cur_delay) == EOF) {
- fseek(fp_gns, 0L, SEEK_SET);
- if (fscanf(fp_gns, "%d", &cur_delay) == EOF) {
- exit(0);
- }
- }
- }
-
- /* simulate packet handling through NetEq and the modem */
- if (!(testNum == 3 && framecnt == 0)) {
- if (gns == 0) {
- get_arrival_time(cur_framesmpls, stream_len, bottleneck,
- &BN_data);
- } else {
- get_arrival_time2(cur_framesmpls, cur_delay, &BN_data);
- }
- }
-
- /* packet not dropped */
- if (cur_delay != -1) {
-
- /* Error test number 10, garbage data */
- if (testNum == 10) {
- for ( i = 0; i < stream_len; i++) {
- streamdata[i] = (short) (streamdata[i] + (short) rand());
- }
- }
-
- if (testNum != 9) {
- err = WebRtcIsacfix_UpdateBwEstimate(ISAC_main_inst,
- streamdata,
- stream_len,
- BN_data.rtp_number,
- BN_data.send_time,
- BN_data.arrival_time);
-
- if (err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
- printf("\nError in decoder: %d.\n", errtype);
- }
- }
-#ifdef _DEBUG
- fprintf(stderr," \rframe = %7d", framecnt);
-#endif
-
- if( readLoss == 1 ) {
- if( fread( &lostFrame, sizeof(WebRtc_Word16), 1, plFile ) != 1 ) {
- rewind( plFile );
- }
- lostFrame = !lostFrame;
- } else {
- lostFrame = (rand()%100 < packetLossPercent);
- }
-
-
-
- /* iSAC decoding */
- if( lostFrame && framecnt > 0) {
- if (nbTest !=2) {
- declen = WebRtcIsacfix_DecodePlc(ISAC_main_inst,
- decoded, prevFrameSize );
- } else {
-#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
- declen = WebRtcIsacfix_DecodePlcNb(ISAC_main_inst, decoded,
- prevFrameSize );
-#else
- declen = -1;
-#endif
- }
- lostPackets++;
- } else {
- if (nbTest !=2 ) {
- short FL;
- /* Call getFramelen, only used here for function test */
- err = WebRtcIsacfix_ReadFrameLen((WebRtc_Word16*)streamdata, &FL);
- declen = WebRtcIsacfix_Decode( ISAC_main_inst, streamdata, stream_len,
- decoded, speechType );
- /* Error check */
- if (err<0 || declen<0 || FL!=declen) {
- errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
- printf("\nError in decode_B/or getFrameLen: %d.\n", errtype);
- }
- prevFrameSize = declen/480;
-
- } else {
-#ifdef WEBRTC_ISAC_FIX_NB_CALLS_ENABLED
- declen = WebRtcIsacfix_DecodeNb( ISAC_main_inst, streamdata,
- stream_len, decoded, speechType );
-#else
- declen = -1;
-#endif
- prevFrameSize = declen/240;
- }
- }
-
- if (declen <= 0) {
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISAC_main_inst);
- printf("\nError in decoder: %d.\n", errtype);
- }
-
- /* Write decoded speech frame to file */
- fwrite(decoded, sizeof(WebRtc_Word16), declen, outp);
- // fprintf( ratefile, "%f \n", stream_len / ( ((double)declen)/
- // ((double)FS) ) * 8 );
- } else {
- lostPackets++;
- }
- framecnt++;
-
- totalsmpls += declen;
- totalbits += 8 * stream_len;
- kbps = ((double) FS) / ((double) cur_framesmpls) * 8.0 *
- stream_len / 1000.0;// kbits/s
-
- /* Error test number 10, garbage data */
- if (testNum == 10) {
- if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
- printf( "Error: Could not open file %s\n", SEED_FILE);
- }
- else {
- fprintf(seedfile, "ok\n\n");
- fclose(seedfile);
- }
- }
-
-#ifdef _DEBUG
-
- fy = fopen("bit_rate.dat", "a");
- fprintf(fy, "Frame %i = %0.14f\n", framecnt, kbps);
- fclose(fy);
-
-#endif /* _DEBUG */
-
- }
- printf("\nLost Frames %d ~ %4.1f%%\n", lostPackets,
- (double)lostPackets/(double)framecnt*100.0 );
- printf("\n\ntotal bits = %d bits", totalbits);
- printf("\nmeasured average bitrate = %0.3f kbits/s",
- (double)totalbits *(FS/1000) / totalsmpls);
- printf("\n");
-
-#ifdef _DEBUG
- /* fprintf(stderr,"\n\ntotal bits = %d bits", totalbits);
- fprintf(stderr,"\nmeasured average bitrate = %0.3f kbits/s",
- (double)totalbits *(FS/1000) / totalsmpls);
- fprintf(stderr,"\n");
- */
-#endif /* _DEBUG */
-
- /* Runtime statistics */
-
-
- runtime = (double)(((double)clock()/(double)CLOCKS_PER_SEC)-starttime);
- length_file = ((double)framecnt*(double)declen/FS);
- printf("\n\nLength of speech file: %.1f s\n", length_file);
- printf("Time to run iSAC: %.2f s (%.2f %% of realtime)\n\n",
- runtime, (100*runtime/length_file));
- printf("\n\n_______________________________________________\n");
-
- fclose(inp);
- fclose(outp);
- fclose(outbits);
-
- if ( testCE == 1) {
- WebRtcIsacfix_FreeInternal(ISAC_main_inst);
- }
- WebRtcIsacfix_Free(ISAC_main_inst);
- return 0;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/test_iSACfixfloat.c b/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/test_iSACfixfloat.c
deleted file mode 100644
index 57c30ca..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/fix/test/test_iSACfixfloat.c
+++ /dev/null
@@ -1,693 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * test_iSACfixfloat.c
- *
- * Test compatibility and quality between floating- and fixed-point code
- * */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-/* include API */
-#include "isac.h"
-#include "isacfix.h"
-
-
-/* max number of samples per frame (= 60 ms frame) */
-#define MAX_FRAMESAMPLES 960
-/* number of samples per 10ms frame */
-#define FRAMESAMPLES_10ms 160
-/* sampling frequency (Hz) */
-#define FS 16000
-
-
-
-/* Runtime statistics */
-#include <time.h>
-#define CLOCKS_PER_SEC 1000
-
-
-
-// FILE *histfile, *ratefile;
-
-
-/* function for reading audio data from PCM file */
-int readframe(WebRtc_Word16 *data, FILE *inp, int length) {
-
- short k, rlen, status = 0;
-
- rlen = fread(data, sizeof(WebRtc_Word16), length, inp);
- if (rlen < length) {
- for (k = rlen; k < length; k++)
- data[k] = 0;
- status = 1;
- }
-
- return status;
-}
-
-typedef struct {
- WebRtc_UWord32 send_time; /* samples */
- WebRtc_UWord32 arrival_time; /* samples */
- WebRtc_UWord32 sample_count; /* samples */
- WebRtc_UWord16 rtp_number;
-} BottleNeckModel;
-
-void get_arrival_time(int current_framesamples, /* samples */
- int packet_size, /* bytes */
- int bottleneck, /* excluding headers; bits/s */
- BottleNeckModel *BN_data)
-{
- const int HeaderSize = 35;
- int HeaderRate;
-
- HeaderRate = HeaderSize * 8 * FS / current_framesamples; /* bits/s */
-
- /* everything in samples */
- BN_data->sample_count = BN_data->sample_count + current_framesamples;
-
- BN_data->arrival_time += ((packet_size + HeaderSize) * 8 * FS) / (bottleneck + HeaderRate);
- BN_data->send_time += current_framesamples;
-
- if (BN_data->arrival_time < BN_data->sample_count)
- BN_data->arrival_time = BN_data->sample_count;
-
- BN_data->rtp_number++;
-}
-
-
-
-int main(int argc, char* argv[])
-{
-
- char inname[50], outname[50], bottleneck_file[50], bitfilename[60], bitending[10]="_bits.pcm";
- FILE *inp, *outp, *f_bn, *bitsp;
- int framecnt, endfile;
-
-
- int i,j,errtype, plc=0;
- WebRtc_Word16 CodingMode;
- WebRtc_Word16 bottleneck;
-
- WebRtc_Word16 framesize = 30; /* ms */
- //WebRtc_Word16 framesize = 60; /* To invoke cisco complexity case at frame 2252 */
-
- int cur_framesmpls, err;
-
- /* Runtime statistics */
- double starttime;
- double runtime;
- double length_file;
-
- WebRtc_Word16 stream_len = 0;
- WebRtc_Word16 declen;
-
- WebRtc_Word16 shortdata[FRAMESAMPLES_10ms];
- WebRtc_Word16 decoded[MAX_FRAMESAMPLES];
- WebRtc_UWord16 streamdata[600];
- WebRtc_Word16 speechType[1];
-
-// WebRtc_Word16 *iSACstruct;
-
- char version_number[20];
- int mode=-1, tmp, nbTest=0; /*,sss;*/
-
-#ifdef _DEBUG
- FILE *fy;
- double kbps;
- int totalbits =0;
- int totalsmpls =0;
-#endif /* _DEBUG */
-
-
-
-
- /* only one structure used for ISAC encoder */
- ISAC_MainStruct *ISAC_main_inst;
- ISACFIX_MainStruct *ISACFIX_main_inst;
-
- BottleNeckModel BN_data;
- f_bn = NULL;
-
-#ifdef _DEBUG
- fy = fopen("bit_rate.dat", "w");
- fclose(fy);
- fy = fopen("bytes_frames.dat", "w");
- fclose(fy);
-#endif /* _DEBUG */
-
-
-//histfile = fopen("histo.dat", "ab");
-//ratefile = fopen("rates.dat", "ab");
-
- /* handling wrong input arguments in the command line */
- if ((argc<6) || (argc>10)) {
- printf("\n\nWrong number of arguments or flag values.\n\n");
-
- printf("\n");
- WebRtcIsacfix_version(version_number);
- printf("iSAC version %s \n\n", version_number);
-
- printf("Usage:\n\n");
- printf("./kenny.exe [-I] bottleneck_value infile outfile \n\n");
- printf("with:\n");
-
- printf("[-I] : if -I option is specified, the coder will use\n");
- printf(" an instantaneous Bottleneck value. If not, it\n");
- printf(" will be an adaptive Bottleneck value.\n\n");
- printf("bottleneck_value : the value of the bottleneck provided either\n");
- printf(" as a fixed value (e.g. 25000) or\n");
- printf(" read from a file (e.g. bottleneck.txt)\n\n");
- printf("[-m] mode : Mode (encoder - decoder):\n");
- printf(" : 0 - float - float \n");
- printf(" : 1 - float - fix \n");
- printf(" : 2 - fix - float \n");
- printf(" : 3 - fix - fix \n");
- printf("[-PLC] : Test PLC packetlosses\n");
- printf("[-NB] num : Test NB interfaces, num=1 encNB, num=2 decNB\n");
- printf("infile : Normal speech input file\n\n");
- printf("outfile : Speech output file\n\n");
- printf("Example usage:\n\n");
- printf("./kenny.exe -I bottleneck.txt -m 1 speechIn.pcm speechOut.pcm\n\n");
- exit(0);
-
- }
-
-
- printf("--------------------START---------------------\n\n");
- WebRtcIsac_version(version_number);
- printf("iSAC FLOAT version %s \n", version_number);
- WebRtcIsacfix_version(version_number);
- printf("iSAC FIX version %s \n\n", version_number);
-
- CodingMode = 0;
- tmp=1;
- for (i = 1; i < argc;i++)
- {
- if (!strcmp ("-I", argv[i]))
- {
- printf("\nInstantaneous BottleNeck\n");
- CodingMode = 1;
- i++;
- tmp=0;
- }
-
- if (!strcmp ("-m", argv[i])) {
- mode=atoi(argv[i+1]);
- i++;
- }
-
- if (!strcmp ("-PLC", argv[i]))
- {
- plc=1;
- }
-
- if (!strcmp ("-NB", argv[i]))
- {
- nbTest = atoi(argv[i + 1]);
- i++;
- }
-
- }
-
- if(mode<0) {
- printf("\nError! Mode must be set: -m 0 \n");
- exit(0);
- }
-
- if (CodingMode == 0)
- {
- printf("\nAdaptive BottleNeck\n");
- }
-
-
-
- /* Get Bottleneck value */
- bottleneck = atoi(argv[2-tmp]);
- if (bottleneck == 0)
- {
- sscanf(argv[2-tmp], "%s", bottleneck_file);
- f_bn = fopen(bottleneck_file, "rb");
- if (f_bn == NULL)
- {
- printf("No value provided for BottleNeck and cannot read file %s.\n", bottleneck_file);
- exit(0);
- }
- else {
- printf("reading bottleneck rates from file %s\n\n",bottleneck_file);
- if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
- /* Set pointer to beginning of file */
- fseek(f_bn, 0L, SEEK_SET);
- fscanf(f_bn, "%d", &bottleneck);
- }
-
- /* Bottleneck is a cosine function
- * Matlab code for writing the bottleneck file:
- * BottleNeck_10ms = 20e3 + 10e3 * cos((0:5999)/5999*2*pi);
- * fid = fopen('bottleneck.txt', 'wb');
- * fprintf(fid, '%d\n', BottleNeck_10ms); fclose(fid);
- */
- }
- }
- else
- {
- printf("\nfixed bottleneck rate of %d bits/s\n\n", bottleneck);
- }
-
-
-
- /* Get Input and Output files */
- sscanf(argv[argc-2], "%s", inname);
- sscanf(argv[argc-1], "%s", outname);
-
- if ((inp = fopen(inname,"rb")) == NULL) {
- printf(" iSAC: Cannot read file %s.\n", inname);
- exit(1);
- }
- if ((outp = fopen(outname,"wb")) == NULL) {
- printf(" iSAC: Cannot write file %s.\n", outname);
- exit(1);
- }
- printf("\nInput:%s\nOutput:%s\n", inname, outname);
-
- i=0;
- while (outname[i]!='\0') {
- bitfilename[i]=outname[i];
- i++;
- }
- i-=4;
- for (j=0;j<9;j++, i++)
- bitfilename[i]=bitending[j];
- bitfilename[i]='\0';
- if ((bitsp = fopen(bitfilename,"wb")) == NULL) {
- printf(" iSAC: Cannot read file %s.\n", bitfilename);
- exit(1);
- }
- printf("Bitstream:%s\n\n", bitfilename);
-
-
-
- starttime = clock()/(double)CLOCKS_PER_SEC; /* Runtime statistics */
-
-
- /* Initialize the ISAC and BN structs */
- WebRtcIsac_create(&ISAC_main_inst);
-/* WebRtcIsacfix_AssignSize(&sss);
- iSACstruct=malloc(sss);
- WebRtcIsacfix_Assign(&ISACFIX_main_inst,iSACstruct);*/
- WebRtcIsacfix_Create(&ISACFIX_main_inst);
-
- BN_data.send_time = 0;
- BN_data.arrival_time = 0;
- BN_data.sample_count = 0;
- BN_data.rtp_number = 0;
-
- /* Initialize encoder and decoder */
- framecnt= 0;
- endfile = 0;
-
- if (mode==0) { /* Encode using FLOAT, decode using FLOAT */
-
- printf("Coding mode: Encode using FLOAT, decode using FLOAT \n\n");
-
- /* Init iSAC FLOAT */
- WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode);
- WebRtcIsac_DecoderInit(ISAC_main_inst);
- if (CodingMode == 1) {
- err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
- if (err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- printf("\n\n Error in initialization: %d.\n\n", errtype);
- // exit(EXIT_FAILURE);
- }
- }
-
- } else if (mode==1) { /* Encode using FLOAT, decode using FIX */
-
- printf("Coding mode: Encode using FLOAT, decode using FIX \n\n");
-
- /* Init iSAC FLOAT */
- WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode);
- WebRtcIsac_DecoderInit(ISAC_main_inst);
- if (CodingMode == 1) {
- err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
- if (err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- printf("\n\n Error in initialization: %d.\n\n", errtype);
- // exit(EXIT_FAILURE);
- }
- }
-
- /* Init iSAC FIX */
- WebRtcIsacfix_EncoderInit(ISACFIX_main_inst, CodingMode);
- WebRtcIsacfix_DecoderInit(ISACFIX_main_inst);
- if (CodingMode == 1) {
- err = WebRtcIsacfix_Control(ISACFIX_main_inst, bottleneck, framesize);
- if (err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
- printf("\n\n Error in initialization: %d.\n\n", errtype);
- //exit(EXIT_FAILURE);
- }
- }
- } else if (mode==2) { /* Encode using FIX, decode using FLOAT */
-
- printf("Coding mode: Encode using FIX, decode using FLOAT \n\n");
-
- /* Init iSAC FLOAT */
- WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode);
- WebRtcIsac_DecoderInit(ISAC_main_inst);
- if (CodingMode == 1) {
- err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
- if (err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- printf("\n\n Error in initialization: %d.\n\n", errtype);
- //exit(EXIT_FAILURE);
- }
- }
-
- /* Init iSAC FIX */
- WebRtcIsacfix_EncoderInit(ISACFIX_main_inst, CodingMode);
- WebRtcIsacfix_DecoderInit(ISACFIX_main_inst);
- if (CodingMode == 1) {
- err = WebRtcIsacfix_Control(ISACFIX_main_inst, bottleneck, framesize);
- if (err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
- printf("\n\n Error in initialization: %d.\n\n", errtype);
- //exit(EXIT_FAILURE);
- }
- }
- } else if (mode==3) {
-
- printf("Coding mode: Encode using FIX, decode using FIX \n\n");
-
- WebRtcIsacfix_EncoderInit(ISACFIX_main_inst, CodingMode);
- WebRtcIsacfix_DecoderInit(ISACFIX_main_inst);
- if (CodingMode == 1) {
- err = WebRtcIsacfix_Control(ISACFIX_main_inst, bottleneck, framesize);
- if (err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
- printf("\n\n Error in initialization: %d.\n\n", errtype);
- //exit(EXIT_FAILURE);
- }
- }
-
- } else
- printf("Mode must be value between 0 and 3\n");
- *speechType = 1;
-
-//#define BI_TEST 1
-#ifdef BI_TEST
- err = WebRtcIsacfix_SetMaxPayloadSize(ISACFIX_main_inst, 300);
- if (err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
- printf("\n\n Error in setMaxPayloadSize: %d.\n\n", errtype);
- fclose(inp);
- fclose(outp);
- fclose(bitsp);
- return(EXIT_FAILURE);
- }
-#endif
-
-
- while (endfile == 0) {
-
- cur_framesmpls = 0;
- while (1) {
- /* Read 10 ms speech block */
- if (nbTest != 1)
- endfile = readframe(shortdata, inp, FRAMESAMPLES_10ms);
- else
- endfile = readframe(shortdata, inp, (FRAMESAMPLES_10ms/2));
-
- /* iSAC encoding */
-
- if (mode==0 || mode ==1) {
- stream_len = WebRtcIsac_Encode(ISAC_main_inst, shortdata, streamdata);
- if (stream_len < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- printf("\n\nError in encoder: %d.\n\n", errtype);
- // exit(EXIT_FAILURE);
- }
- } else if (mode==2 || mode==3) {
- /* iSAC encoding */
- if (nbTest != 1)
- stream_len = WebRtcIsacfix_Encode(ISACFIX_main_inst, shortdata, streamdata);
- else
- stream_len = WebRtcIsacfix_EncodeNb(ISACFIX_main_inst, shortdata, streamdata);
-
- if (stream_len < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
- printf("\n\nError in encoder: %d.\n\n", errtype);
- // exit(EXIT_FAILURE);
- }
- }
-
- cur_framesmpls += FRAMESAMPLES_10ms;
-
- /* read next bottleneck rate */
- if (f_bn != NULL) {
- if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
- /* Set pointer to beginning of file */
- fseek(f_bn, 0L, SEEK_SET);
- fscanf(f_bn, "%d", &bottleneck);
- }
- if (CodingMode == 1) {
- if (mode==0 || mode==1)
- WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
- else if (mode==2 || mode==3)
- WebRtcIsacfix_Control(ISACFIX_main_inst, bottleneck, framesize);
- }
- }
-
- /* exit encoder loop if the encoder returned a bitstream */
- if (stream_len != 0) break;
- }
-
- fwrite(streamdata, 1, stream_len, bitsp); /* NOTE! Writes bytes to file */
-
- /* simulate packet handling through NetEq and the modem */
- get_arrival_time(cur_framesmpls, stream_len, bottleneck,
- &BN_data);
-//*****************************
- if (1){
- if (mode==0) {
- err = WebRtcIsac_UpdateBwEstimate(ISAC_main_inst,
- streamdata,
- stream_len,
- BN_data.rtp_number,
- BN_data.arrival_time);
-
- if (err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- printf("\n\nError in decoder: %d.\n\n", errtype);
- //exit(EXIT_FAILURE);
- }
- /* iSAC decoding */
- declen = WebRtcIsac_Decode(ISAC_main_inst,
- streamdata,
- stream_len,
- decoded,
- speechType);
- if (declen <= 0) {
- /* exit if returned with error */
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- printf("\n\nError in decoder: %d.\n\n", errtype);
- //exit(EXIT_FAILURE);
- }
- } else if (mode==1) {
-
- err = WebRtcIsac_UpdateBwEstimate(ISAC_main_inst,
- streamdata,
- stream_len,
- BN_data.rtp_number,
- BN_data.arrival_time);
- err = WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_main_inst,
- streamdata,
- stream_len,
- BN_data.rtp_number,
- BN_data.arrival_time);
- if (err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
- printf("\n\nError in decoder: %d.\n\n", errtype);
- //exit(EXIT_FAILURE);
- }
-
- declen = WebRtcIsac_Decode(ISAC_main_inst,
- streamdata,
- stream_len,
- decoded,
- speechType);
-
- /* iSAC decoding */
- if (plc && (framecnt+1)%10 == 0) {
- if (nbTest !=2 )
- declen = WebRtcIsacfix_DecodePlc( ISACFIX_main_inst, decoded, 1 );
- else
- declen = WebRtcIsacfix_DecodePlcNb( ISACFIX_main_inst, decoded, 1 );
- } else {
- if (nbTest !=2 )
- declen = WebRtcIsacfix_Decode(ISACFIX_main_inst,
- streamdata,
- stream_len,
- decoded,
- speechType);
- else
- declen = WebRtcIsacfix_DecodeNb(ISACFIX_main_inst,
- streamdata,
- stream_len,
- decoded,
- speechType);
- }
-
- if (declen <= 0) {
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
- printf("\n\nError in decoder: %d.\n\n", errtype);
- //exit(EXIT_FAILURE);
- }
- } else if (mode==2) {
- err = WebRtcIsacfix_UpdateBwEstimate1(ISACFIX_main_inst,
- streamdata,
- stream_len,
- BN_data.rtp_number,
- BN_data.arrival_time);
-
- err = WebRtcIsac_UpdateBwEstimate(ISAC_main_inst,
- streamdata,
- stream_len,
- BN_data.rtp_number,
- BN_data.arrival_time);
-
- if (err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- printf("\n\nError in decoder: %d.\n\n", errtype);
- //exit(EXIT_FAILURE);
- }
- /* iSAC decoding */
- declen = WebRtcIsac_Decode(ISAC_main_inst,
- streamdata,
- stream_len,
- decoded,
- speechType);
- if (declen <= 0) {
- /* exit if returned with error */
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- printf("\n\nError in decoder: %d.\n\n", errtype);
- //exit(EXIT_FAILURE);
- }
- } else if (mode==3) {
- err = WebRtcIsacfix_UpdateBwEstimate(ISACFIX_main_inst,
- streamdata,
- stream_len,
- BN_data.rtp_number,
- BN_data.send_time,
- BN_data.arrival_time);
-
- if (err < 0) {
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
- printf("\n\nError in decoder: %d.\n\n", errtype);
- //exit(EXIT_FAILURE);
- }
- /* iSAC decoding */
-
- if (plc && (framecnt+1)%10 == 0) {
- if (nbTest !=2 )
- declen = WebRtcIsacfix_DecodePlc( ISACFIX_main_inst, decoded, 1 );
- else
- declen = WebRtcIsacfix_DecodePlcNb( ISACFIX_main_inst, decoded, 1 );
- } else {
- if (nbTest !=2 )
- declen = WebRtcIsacfix_Decode(ISACFIX_main_inst,
- streamdata,
- stream_len,
- decoded,
- speechType);
- else
- declen = WebRtcIsacfix_DecodeNb(ISACFIX_main_inst,
- streamdata,
- stream_len,
- decoded,
- speechType);
- }
- if (declen <= 0) {
- /* exit if returned with error */
- errtype=WebRtcIsacfix_GetErrorCode(ISACFIX_main_inst);
- printf("\n\nError in decoder: %d.\n\n", errtype);
- //exit(EXIT_FAILURE);
- }
- }
-
- /* Write decoded speech frame to file */
- fwrite(decoded, sizeof(WebRtc_Word16), declen, outp);
- }
-
- fprintf(stderr," \rframe = %d", framecnt);
- framecnt++;
-
-
-
-#ifdef _DEBUG
-
- totalsmpls += declen;
- totalbits += 8 * stream_len;
- kbps = ((double) FS) / ((double) cur_framesmpls) * 8.0 * stream_len / 1000.0;// kbits/s
- fy = fopen("bit_rate.dat", "a");
- fprintf(fy, "Frame %i = %0.14f\n", framecnt, kbps);
- fclose(fy);
-
-#endif /* _DEBUG */
-
- }
-
-#ifdef _DEBUG
- printf("\n\ntotal bits = %d bits", totalbits);
- printf("\nmeasured average bitrate = %0.3f kbits/s", (double)totalbits *(FS/1000) / totalsmpls);
- printf("\n");
-#endif /* _DEBUG */
-
- /* Runtime statistics */
- runtime = (double)(clock()/(double)CLOCKS_PER_SEC-starttime);
- length_file = ((double)framecnt*(double)declen/FS);
- printf("\n\nLength of speech file: %.1f s\n", length_file);
- printf("Time to run iSAC: %.2f s (%.2f %% of realtime)\n\n", runtime, (100*runtime/length_file));
- printf("---------------------END----------------------\n");
-
- fclose(inp);
- fclose(outp);
-
- WebRtcIsac_Free(ISAC_main_inst);
- WebRtcIsacfix_Free(ISACFIX_main_inst);
-
-
-
-// fclose(histfile);
-// fclose(ratefile);
-
- return 0;
-
-}
-
-
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/isac_test.gypi b/trunk/src/modules/audio_coding/codecs/iSAC/isac_test.gypi
deleted file mode 100644
index abe2454..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/isac_test.gypi
+++ /dev/null
@@ -1,71 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- # simple kenny
- {
- 'target_name': 'iSACtest',
- 'type': 'executable',
- 'dependencies': [
- 'iSAC',
- ],
- 'include_dirs': [
- './main/test',
- './main/interface',
- './main/util',
- ],
- 'sources': [
- './main/test/simpleKenny.c',
- './main/util/utility.c',
- ],
- },
- # ReleaseTest-API
- {
- 'target_name': 'iSACAPITest',
- 'type': 'executable',
- 'dependencies': [
- 'iSAC',
- ],
- 'include_dirs': [
- './main/test',
- './main/interface',
- './main/util',
- ],
- 'sources': [
- './main/test/ReleaseTest-API/ReleaseTest-API.cc',
- './main/util/utility.c',
- ],
- },
- # SwitchingSampRate
- {
- 'target_name': 'iSACSwitchSampRateTest',
- 'type': 'executable',
- 'dependencies': [
- 'iSAC',
- ],
- 'include_dirs': [
- './main/test',
- './main/interface',
- '../../../../common_audio/signal_processing/include',
- './main/util',
- ],
- 'sources': [
- './main/test/SwitchingSampRate/SwitchingSampRate.cc',
- './main/util/utility.c',
- ],
- },
-
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/isacfix_test.gypi b/trunk/src/modules/audio_coding/codecs/iSAC/isacfix_test.gypi
deleted file mode 100644
index 627342f..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/isacfix_test.gypi
+++ /dev/null
@@ -1,33 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- # kenny
- {
- 'target_name': 'iSACFixtest',
- 'type': 'executable',
- 'dependencies': [
- 'iSACFix',
- ],
- 'include_dirs': [
- './fix/test',
- './fix/interface',
- ],
- 'sources': [
- './fix/test/kenny.c',
- ],
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/interface/isac.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/interface/isac.h
deleted file mode 100644
index 03c260b..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/interface/isac.h
+++ /dev/null
@@ -1,729 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_
-
-/*
- * Define the fixed-point numeric formats
- */
-#include "typedefs.h"
-
-typedef struct WebRtcISACStruct ISACStruct;
-
-enum IsacSamplingRate {kIsacWideband = 16, kIsacSuperWideband = 32};
-
-
-#if defined(__cplusplus)
-extern "C" {
-#endif
-
- /******************************************************************************
- * WebRtcIsac_AssignSize(...)
- *
- * This function returns the size of the ISAC instance, so that the instance
- * can be created outside iSAC.
- *
- * Input:
- * - samplingRate : sampling rate of the input/output audio.
- *
- * Output:
- * - sizeinbytes : number of bytes needed to allocate for the
- * instance.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsac_AssignSize(
- int* sizeinbytes);
-
-
- /******************************************************************************
- * WebRtcIsac_Assign(...)
- *
- * This function assignes the memory already created to the ISAC instance.
- *
- * Input:
- * - *ISAC_main_inst : a pointer to the coder instance.
- * - samplingRate : sampling rate of the input/output audio.
- * - ISAC_inst_Addr : the already allocated memory, where we put the
- * iSAC structure.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsac_Assign(
- ISACStruct** ISAC_main_inst,
- void* ISAC_inst_Addr);
-
-
- /******************************************************************************
- * WebRtcIsac_Create(...)
- *
- * This function creates an ISAC instance, which will contain the state
- * information for one coding/decoding channel.
- *
- * Input:
- * - *ISAC_main_inst : a pointer to the coder instance.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsac_Create(
- ISACStruct** ISAC_main_inst);
-
-
- /******************************************************************************
- * WebRtcIsac_Free(...)
- *
- * This function frees the ISAC instance created at the beginning.
- *
- * Input:
- * - ISAC_main_inst : an ISAC instance.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsac_Free(
- ISACStruct* ISAC_main_inst);
-
-
- /******************************************************************************
- * WebRtcIsac_EncoderInit(...)
- *
- * This function initializes an ISAC instance prior to the encoder calls.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - CodingMode : 0 -> Bit rate and frame length are
- * automatically adjusted to available bandwidth
- * on transmission channel, just valid if codec
- * is created to work in wideband mode.
- * 1 -> User sets a frame length and a target bit
- * rate which is taken as the maximum
- * short-term average bit rate.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsac_EncoderInit(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word16 CodingMode);
-
-
- /******************************************************************************
- * WebRtcIsac_Encode(...)
- *
- * This function encodes 10ms audio blocks and inserts it into a package.
- * Input speech length has 160 samples if operating at 16 kHz sampling
- * rate, or 320 if operating at 32 kHz sampling rate. The encoder buffers the
- * input audio until the whole frame is buffered then proceeds with encoding.
- *
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - speechIn : input speech vector.
- *
- * Output:
- * - encoded : the encoded data vector
- *
- * Return value:
- * : >0 - Length (in bytes) of coded data
- * : 0 - The buffer didn't reach the chosen
- * frame-size so it keeps buffering speech
- * samples.
- * : -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsac_Encode(
- ISACStruct* ISAC_main_inst,
- const WebRtc_Word16* speechIn,
- WebRtc_Word16* encoded);
-
-
- /******************************************************************************
- * WebRtcIsac_DecoderInit(...)
- *
- * This function initializes an ISAC instance prior to the decoder calls.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- *
- * Return value
- * : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsac_DecoderInit(
- ISACStruct* ISAC_main_inst);
-
-
- /******************************************************************************
- * WebRtcIsac_UpdateBwEstimate(...)
- *
- * This function updates the estimate of the bandwidth.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - encoded : encoded ISAC frame(s).
- * - packet_size : size of the packet.
- * - rtp_seq_number : the RTP number of the packet.
- * - send_ts : the RTP send timestamp, given in samples
- * - arr_ts : the arrival time of the packet (from NetEq)
- * in samples.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsac_UpdateBwEstimate(
- ISACStruct* ISAC_main_inst,
- const WebRtc_UWord16* encoded,
- WebRtc_Word32 packet_size,
- WebRtc_UWord16 rtp_seq_number,
- WebRtc_UWord32 send_ts,
- WebRtc_UWord32 arr_ts);
-
-
- /******************************************************************************
- * WebRtcIsac_Decode(...)
- *
- * This function decodes an ISAC frame. At 16 kHz sampling rate, the length
- * of the output audio could be either 480 or 960 samples, equivalent to
- * 30 or 60 ms respectively. At 32 kHz sampling rate, the length of the
- * output audio is 960 samples, which is 30 ms.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - encoded : encoded ISAC frame(s).
- * - len : bytes in encoded vector.
- *
- * Output:
- * - decoded : The decoded vector.
- *
- * Return value : >0 - number of samples in decoded vector.
- * -1 - Error.
- */
-
- WebRtc_Word16 WebRtcIsac_Decode(
- ISACStruct* ISAC_main_inst,
- const WebRtc_UWord16* encoded,
- WebRtc_Word16 len,
- WebRtc_Word16* decoded,
- WebRtc_Word16* speechType);
-
-
- /******************************************************************************
- * WebRtcIsac_DecodePlc(...)
- *
- * This function conducts PLC for ISAC frame(s). Output speech length
- * will be a multiple of frames, i.e. multiples of 30 ms audio. Therefore,
- * the output is multiple of 480 samples if operating at 16 kHz and multiple
- * of 960 if operating at 32 kHz.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - noOfLostFrames : Number of PLC frames to produce.
- *
- * Output:
- * - decoded : The decoded vector.
- *
- * Return value : >0 - number of samples in decoded PLC vector
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsac_DecodePlc(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word16* decoded,
- WebRtc_Word16 noOfLostFrames);
-
-
- /******************************************************************************
- * WebRtcIsac_Control(...)
- *
- * This function sets the limit on the short-term average bit-rate and the
- * frame length. Should be used only in Instantaneous mode. At 16 kHz sampling
- * rate, an average bit-rate between 10000 to 32000 bps is valid and a
- * frame-size of 30 or 60 ms is acceptable. At 32 kHz, an average bit-rate
- * between 10000 to 56000 is acceptable, and the valid frame-size is 30 ms.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - rate : limit on the short-term average bit rate,
- * in bits/second.
- * - framesize : frame-size in millisecond.
- *
- * Return value : 0 - ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsac_Control(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word32 rate,
- WebRtc_Word16 framesize);
-
-
- /******************************************************************************
- * WebRtcIsac_ControlBwe(...)
- *
- * This function sets the initial values of bottleneck and frame-size if
- * iSAC is used in channel-adaptive mode. Therefore, this API is not
- * applicable if the codec is created to operate in super-wideband mode.
- *
- * Through this API, users can enforce a frame-size for all values of
- * bottleneck. Then iSAC will not automatically change the frame-size.
- *
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - rateBPS : initial value of bottleneck in bits/second
- * 10000 <= rateBPS <= 56000 is accepted
- * For default bottleneck set rateBPS = 0
- * - frameSizeMs : number of milliseconds per frame (30 or 60)
- * - enforceFrameSize : 1 to enforce the given frame-size through
- * out the adaptation process, 0 to let iSAC
- * change the frame-size if required.
- *
- * Return value : 0 - ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIsac_ControlBwe(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word32 rateBPS,
- WebRtc_Word16 frameSizeMs,
- WebRtc_Word16 enforceFrameSize);
-
-
- /******************************************************************************
- * WebRtcIsac_ReadFrameLen(...)
- *
- * This function returns the length of the frame represented in the packet.
- *
- * Input:
- * - encoded : Encoded bit-stream
- *
- * Output:
- * - frameLength : Length of frame in packet (in samples)
- *
- */
-
- WebRtc_Word16 WebRtcIsac_ReadFrameLen(
- ISACStruct* ISAC_main_inst,
- const WebRtc_Word16* encoded,
- WebRtc_Word16* frameLength);
-
-
- /******************************************************************************
- * WebRtcIsac_version(...)
- *
- * This function returns the version number.
- *
- * Output:
- * - version : Pointer to character string
- *
- */
-
- void WebRtcIsac_version(
- char *version);
-
-
- /******************************************************************************
- * WebRtcIsac_GetErrorCode(...)
- *
- * This function can be used to check the error code of an iSAC instance. When
- * a function returns -1 a error code will be set for that instance. The
- * function below extract the code of the last error that occurred in the
- * specified instance.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance
- *
- * Return value : Error code
- */
-
- WebRtc_Word16 WebRtcIsac_GetErrorCode(
- ISACStruct* ISAC_main_inst);
-
-
- /****************************************************************************
- * WebRtcIsac_GetUplinkBw(...)
- *
- * This function outputs the target bottleneck of the codec. In
- * channel-adaptive mode, the target bottleneck is specified through in-band
- * signalling retreived by bandwidth estimator.
- * In channel-independent, also called instantaneous mode, the target
- * bottleneck is provided to the encoder by calling xxx_control(...). If
- * xxx_control is never called the default values is returned. The default
- * value for bottleneck at 16 kHz encoder sampling rate is 32000 bits/sec,
- * and it is 56000 bits/sec for 32 kHz sampling rate.
- * Note that the output is the iSAC internal operating bottleneck which might
- * differ slightly from the one provided through xxx_control().
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- *
- * Output:
- * - *bottleneck : bottleneck in bits/sec
- *
- * Return value : -1 if error happens
- * 0 bit-rates computed correctly.
- */
-
- WebRtc_Word16 WebRtcIsac_GetUplinkBw(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word32* bottleneck);
-
-
- /******************************************************************************
- * WebRtcIsac_SetMaxPayloadSize(...)
- *
- * This function sets a limit for the maximum payload size of iSAC. The same
- * value is used both for 30 and 60 ms packets. If the encoder sampling rate
- * is 16 kHz the maximum payload size is between 120 and 400 bytes. If the
- * encoder sampling rate is 32 kHz the maximum payload size is between 120
- * and 600 bytes.
- *
- * If an out of range limit is used, the function returns -1, but the closest
- * valid value will be applied.
- *
- * ---------------
- * IMPORTANT NOTES
- * ---------------
- * The size of a packet is limited to the minimum of 'max-payload-size' and
- * 'max-rate.' For instance, let's assume the max-payload-size is set to
- * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps
- * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms
- * frame-size. Then a packet with a frame-size of 30 ms is limited to 150,
- * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to
- * 170 bytes, i.e. min(170, 300).
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- * - maxPayloadBytes : maximum size of the payload in bytes
- * valid values are between 120 and 400 bytes
- * if encoder sampling rate is 16 kHz. For
- * 32 kHz encoder sampling rate valid values
- * are between 120 and 600 bytes.
- *
- * Return value : 0 if successful
- * -1 if error happens
- */
-
- WebRtc_Word16 WebRtcIsac_SetMaxPayloadSize(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word16 maxPayloadBytes);
-
-
- /******************************************************************************
- * WebRtcIsac_SetMaxRate(...)
- *
- * This function sets the maximum rate which the codec may not exceed for
- * any signal packet. The maximum rate is defined and payload-size per
- * frame-size in bits per second.
- *
- * The codec has a maximum rate of 53400 bits per second (200 bytes per 30
- * ms) if the encoder sampling rate is 16kHz, and 160 kbps (600 bytes/30 ms)
- * if the encoder sampling rate is 32 kHz.
- *
- * It is possible to set a maximum rate between 32000 and 53400 bits/sec
- * in wideband mode, and 32000 to 160000 bits/sec in super-wideband mode.
- *
- * If an out of range limit is used, the function returns -1, but the closest
- * valid value will be applied.
- *
- * ---------------
- * IMPORTANT NOTES
- * ---------------
- * The size of a packet is limited to the minimum of 'max-payload-size' and
- * 'max-rate.' For instance, let's assume the max-payload-size is set to
- * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps
- * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms
- * frame-size. Then a packet with a frame-size of 30 ms is limited to 150,
- * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to
- * 170 bytes, min(170, 300).
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- * - maxRate : maximum rate in bits per second,
- * valid values are 32000 to 53400 bits/sec in
- * wideband mode, and 32000 to 160000 bits/sec in
- * super-wideband mode.
- *
- * Return value : 0 if successful
- * -1 if error happens
- */
-
- WebRtc_Word16 WebRtcIsac_SetMaxRate(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word32 maxRate);
-
-
- /******************************************************************************
- * WebRtcIsac_DecSampRate()
- * Return the sampling rate of the decoded audio.
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- *
- * Return value : enumerator representing sampling frequency
- * associated with the decoder, i.e. the
- * sampling rate of the decoded audio.
- *
- */
-
- enum IsacSamplingRate WebRtcIsac_DecSampRate(
- ISACStruct* ISAC_main_inst);
-
-
- /******************************************************************************
- * WebRtcIsac_EncSampRate()
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- *
- * Return value : enumerator representing sampling frequency
- * associated with the encoder, the input audio
- * is expected to be sampled at this rate.
- *
- */
-
- enum IsacSamplingRate WebRtcIsac_EncSampRate(
- ISACStruct* ISAC_main_inst);
-
-
- /******************************************************************************
- * WebRtcIsac_SetDecSampRate()
- * Set the sampling rate of the decoder. Initialization of the decoder WILL
- * NOT overwrite the sampling rate of the encoder. The default value is 16 kHz
- * which is set when the instance is created.
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- * - sampRate : enumerator specifying the sampling rate.
- *
- * Return value : 0 if successful
- * -1 if failed.
- */
-
- WebRtc_Word16 WebRtcIsac_SetDecSampRate(
- ISACStruct* ISAC_main_inst,
- enum IsacSamplingRate sampRate);
-
-
- /******************************************************************************
- * WebRtcIsac_SetEncSampRate()
- * Set the sampling rate of the encoder. Initialization of the encoder WILL
- * NOT overwrite the sampling rate of the encoder. The default value is 16 kHz
- * which is set when the instance is created. The encoding-mode and the
- * bottleneck remain unchanged by this call, however, the maximum rate and
- * maximum payload-size will reset to their default value.
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- * - sampRate : enumerator specifying the sampling rate.
- *
- * Return value : 0 if successful
- * -1 if failed.
- */
-
- WebRtc_Word16 WebRtcIsac_SetEncSampRate(
- ISACStruct* ISAC_main_inst,
- enum IsacSamplingRate sampRate);
-
-
-
- /******************************************************************************
- * WebRtcIsac_GetNewBitStream(...)
- *
- * This function returns encoded data, with the recieved bwe-index in the
- * stream. If the rate is set to a value less than bottleneck of codec
- * the new bistream will be re-encoded with the given target rate.
- * It should always return a complete packet, i.e. only called once
- * even for 60 msec frames.
- *
- * NOTE 1! This function does not write in the ISACStruct, it is not allowed.
- * NOTE 2! Currently not implemented for SWB mode.
- * NOTE 3! Rates larger than the bottleneck of the codec will be limited
- * to the current bottleneck.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - bweIndex : Index of bandwidth estimate to put in new
- * bitstream
- * - rate : target rate of the transcoder is bits/sec.
- * Valid values are the accepted rate in iSAC,
- * i.e. 10000 to 56000.
- * - isRCU : if the new bit-stream is an RCU stream.
- * Note that the rate parameter always indicates
- * the target rate of the main paylaod, regardless
- * of 'isRCU' value.
- *
- * Output:
- * - encoded : The encoded data vector
- *
- * Return value : >0 - Length (in bytes) of coded data
- * -1 - Error or called in SWB mode
- * NOTE! No error code is written to
- * the struct since it is only allowed to read
- * the struct.
- */
- WebRtc_Word16 WebRtcIsac_GetNewBitStream(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word16 bweIndex,
- WebRtc_Word16 jitterInfo,
- WebRtc_Word32 rate,
- WebRtc_Word16* encoded,
- WebRtc_Word16 isRCU);
-
-
-
- /****************************************************************************
- * WebRtcIsac_GetDownLinkBwIndex(...)
- *
- * This function returns index representing the Bandwidth estimate from
- * other side to this side.
- *
- * Input:
- * - ISAC_main_inst : iSAC struct
- *
- * Output:
- * - bweIndex : Bandwidth estimate to transmit to other side.
- *
- */
-
- WebRtc_Word16 WebRtcIsac_GetDownLinkBwIndex(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word16* bweIndex,
- WebRtc_Word16* jitterInfo);
-
-
- /****************************************************************************
- * WebRtcIsac_UpdateUplinkBw(...)
- *
- * This function takes an index representing the Bandwidth estimate from
- * this side to other side and updates BWE.
- *
- * Input:
- * - ISAC_main_inst : iSAC struct
- * - bweIndex : Bandwidth estimate from other side.
- *
- */
-
- WebRtc_Word16 WebRtcIsac_UpdateUplinkBw(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word16 bweIndex);
-
-
- /****************************************************************************
- * WebRtcIsac_ReadBwIndex(...)
- *
- * This function returns the index of the Bandwidth estimate from the bitstream.
- *
- * Input:
- * - encoded : Encoded bitstream
- *
- * Output:
- * - frameLength : Length of frame in packet (in samples)
- * - bweIndex : Bandwidth estimate in bitstream
- *
- */
-
- WebRtc_Word16 WebRtcIsac_ReadBwIndex(
- const WebRtc_Word16* encoded,
- WebRtc_Word16* bweIndex);
-
-
-
- /*******************************************************************************
- * WebRtcIsac_GetNewFrameLen(...)
- *
- * returns the frame lenght (in samples) of the next packet. In the case of channel-adaptive
- * mode, iSAC decides on its frame lenght based on the estimated bottleneck
- * this allows a user to prepare for the next packet (at the encoder)
- *
- * The primary usage is in CE to make the iSAC works in channel-adaptive mode
- *
- * Input:
- * - ISAC_main_inst : iSAC struct
- *
- * Return Value : frame lenght in samples
- *
- */
-
- WebRtc_Word16 WebRtcIsac_GetNewFrameLen(
- ISACStruct* ISAC_main_inst);
-
-
- /****************************************************************************
- * WebRtcIsac_GetRedPayload(...)
- *
- * Populates "encoded" with the redundant payload of the recently encoded
- * frame. This function has to be called once that WebRtcIsac_Encode(...)
- * returns a positive value. Regardless of the frame-size this function will
- * be called only once after encoding is completed.
- *
- * Input:
- * - ISAC_main_inst : iSAC struct
- *
- * Output:
- * - encoded : the encoded data vector
- *
- *
- * Return value:
- * : >0 - Length (in bytes) of coded data
- * : -1 - Error
- *
- *
- */
- WebRtc_Word16 WebRtcIsac_GetRedPayload(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word16* encoded);
-
-
- /****************************************************************************
- * WebRtcIsac_DecodeRcu(...)
- *
- * This function decodes a redundant (RCU) iSAC frame. Function is called in
- * NetEq with a stored RCU payload i case of packet loss. Output speech length
- * will be a multiple of 480 samples: 480 or 960 samples,
- * depending on the framesize (30 or 60 ms).
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - encoded : encoded ISAC RCU frame(s)
- * - len : bytes in encoded vector
- *
- * Output:
- * - decoded : The decoded vector
- *
- * Return value : >0 - number of samples in decoded vector
- * -1 - Error
- */
- WebRtc_Word16 WebRtcIsac_DecodeRcu(
- ISACStruct* ISAC_main_inst,
- const WebRtc_UWord16* encoded,
- WebRtc_Word16 len,
- WebRtc_Word16* decoded,
- WebRtc_Word16* speechType);
-
-
-#if defined(__cplusplus)
-}
-#endif
-
-
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_INTERFACE_ISAC_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/Android.mk b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/Android.mk
deleted file mode 100644
index 07b2a31..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/Android.mk
+++ /dev/null
@@ -1,66 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_isac
-LOCAL_MODULE_TAGS := optional
-LOCAL_SRC_FILES := \
- arith_routines.c \
- arith_routines_hist.c \
- arith_routines_logist.c \
- bandwidth_estimator.c \
- crc.c \
- decode.c \
- decode_bwe.c \
- encode.c \
- encode_lpc_swb.c \
- entropy_coding.c \
- fft.c \
- filter_functions.c \
- filterbank_tables.c \
- intialize.c \
- isac.c \
- filterbanks.c \
- pitch_lag_tables.c \
- lattice.c \
- lpc_gain_swb_tables.c \
- lpc_analysis.c \
- lpc_shape_swb12_tables.c \
- lpc_shape_swb16_tables.c \
- lpc_tables.c \
- pitch_estimator.c \
- pitch_filter.c \
- pitch_gain_tables.c \
- spectrum_ar_model_tables.c \
- transform.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../../../../.. \
- $(LOCAL_PATH)/../../../../../../common_audio/signal_processing/include
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines.c
deleted file mode 100644
index 31c441a..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines.c
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "arith_routines.h"
-#include "settings.h"
-
-
-/*
- * terminate and return byte stream;
- * returns the number of bytes in the stream
- */
-int WebRtcIsac_EncTerminate(Bitstr *streamdata) /* in-/output struct containing bitstream */
-{
- WebRtc_UWord8 *stream_ptr;
-
-
- /* point to the right place in the stream buffer */
- stream_ptr = streamdata->stream + streamdata->stream_index;
-
- /* find minimum length (determined by current interval width) */
- if ( streamdata->W_upper > 0x01FFFFFF )
- {
- streamdata->streamval += 0x01000000;
- /* add carry to buffer */
- if (streamdata->streamval < 0x01000000)
- {
- /* propagate carry */
- while ( !(++(*--stream_ptr)) );
- /* put pointer back to the old value */
- stream_ptr = streamdata->stream + streamdata->stream_index;
- }
- /* write remaining data to bitstream */
- *stream_ptr++ = (WebRtc_UWord8) (streamdata->streamval >> 24);
- }
- else
- {
- streamdata->streamval += 0x00010000;
- /* add carry to buffer */
- if (streamdata->streamval < 0x00010000)
- {
- /* propagate carry */
- while ( !(++(*--stream_ptr)) );
- /* put pointer back to the old value */
- stream_ptr = streamdata->stream + streamdata->stream_index;
- }
- /* write remaining data to bitstream */
- *stream_ptr++ = (WebRtc_UWord8) (streamdata->streamval >> 24);
- *stream_ptr++ = (WebRtc_UWord8) ((streamdata->streamval >> 16) & 0x00FF);
- }
-
- /* calculate stream length */
- return (int)(stream_ptr - streamdata->stream);
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines.h
deleted file mode 100644
index 8e5f496..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines.h
+++ /dev/null
@@ -1,63 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * arith_routines.h
- *
- * Functions for arithmetic coding.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_
-
-#include "structs.h"
-
-
-int WebRtcIsac_EncLogisticMulti2(
- Bitstr *streamdata, /* in-/output struct containing bitstream */
- WebRtc_Word16 *dataQ7, /* input: data vector */
- const WebRtc_UWord16 *env, /* input: side info vector defining the width of the pdf */
- const int N, /* input: data vector length */
- const WebRtc_Word16 isSWB12kHz); /* if the codec is working in 12kHz bandwidth */
-
-/* returns the number of bytes in the stream */
-int WebRtcIsac_EncTerminate(Bitstr *streamdata); /* in-/output struct containing bitstream */
-
-/* returns the number of bytes in the stream so far */
-int WebRtcIsac_DecLogisticMulti2(
- WebRtc_Word16 *data, /* output: data vector */
- Bitstr *streamdata, /* in-/output struct containing bitstream */
- const WebRtc_UWord16 *env, /* input: side info vector defining the width of the pdf */
- const WebRtc_Word16 *dither, /* input: dither vector */
- const int N, /* input: data vector length */
- const WebRtc_Word16 isSWB12kHz); /* if the codec is working in 12kHz bandwidth */
-
-void WebRtcIsac_EncHistMulti(
- Bitstr *streamdata, /* in-/output struct containing bitstream */
- const int *data, /* input: data vector */
- const WebRtc_UWord16 **cdf, /* input: array of cdf arrays */
- const int N); /* input: data vector length */
-
-int WebRtcIsac_DecHistBisectMulti(
- int *data, /* output: data vector */
- Bitstr *streamdata, /* in-/output struct containing bitstream */
- const WebRtc_UWord16 **cdf, /* input: array of cdf arrays */
- const WebRtc_UWord16 *cdf_size, /* input: array of cdf table sizes+1 (power of two: 2^k) */
- const int N); /* input: data vector length */
-
-int WebRtcIsac_DecHistOneStepMulti(
- int *data, /* output: data vector */
- Bitstr *streamdata, /* in-/output struct containing bitstream */
- const WebRtc_UWord16 **cdf, /* input: array of cdf arrays */
- const WebRtc_UWord16 *init_index,/* input: vector of initial cdf table search entries */
- const int N); /* input: data vector length */
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ARITH_ROUTINES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines_hist.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines_hist.c
deleted file mode 100644
index f4a13d6..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines_hist.c
+++ /dev/null
@@ -1,291 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "settings.h"
-#include "arith_routines.h"
-
-
-/*
- * code symbols into arithmetic bytestream
- */
-void WebRtcIsac_EncHistMulti(Bitstr *streamdata, /* in-/output struct containing bitstream */
- const int *data, /* input: data vector */
- const WebRtc_UWord16 **cdf, /* input: array of cdf arrays */
- const int N) /* input: data vector length */
-{
- WebRtc_UWord32 W_lower, W_upper;
- WebRtc_UWord32 W_upper_LSB, W_upper_MSB;
- WebRtc_UWord8 *stream_ptr;
- WebRtc_UWord8 *stream_ptr_carry;
- WebRtc_UWord32 cdf_lo, cdf_hi;
- int k;
-
-
- /* point to beginning of stream buffer */
- stream_ptr = streamdata->stream + streamdata->stream_index;
- W_upper = streamdata->W_upper;
-
- for (k=N; k>0; k--)
- {
- /* fetch cdf_lower and cdf_upper from cdf tables */
- cdf_lo = (WebRtc_UWord32) *(*cdf + *data);
- cdf_hi = (WebRtc_UWord32) *(*cdf++ + *data++ + 1);
-
- /* update interval */
- W_upper_LSB = W_upper & 0x0000FFFF;
- W_upper_MSB = W_upper >> 16;
- W_lower = W_upper_MSB * cdf_lo;
- W_lower += (W_upper_LSB * cdf_lo) >> 16;
- W_upper = W_upper_MSB * cdf_hi;
- W_upper += (W_upper_LSB * cdf_hi) >> 16;
-
- /* shift interval such that it begins at zero */
- W_upper -= ++W_lower;
-
- /* add integer to bitstream */
- streamdata->streamval += W_lower;
-
- /* handle carry */
- if (streamdata->streamval < W_lower)
- {
- /* propagate carry */
- stream_ptr_carry = stream_ptr;
- while (!(++(*--stream_ptr_carry)));
- }
-
- /* renormalize interval, store most significant byte of streamval and update streamval */
- while ( !(W_upper & 0xFF000000) ) /* W_upper < 2^24 */
- {
- W_upper <<= 8;
- *stream_ptr++ = (WebRtc_UWord8) (streamdata->streamval >> 24);
- streamdata->streamval <<= 8;
- }
- }
-
- /* calculate new stream_index */
- streamdata->stream_index = (int)(stream_ptr - streamdata->stream);
- streamdata->W_upper = W_upper;
-
- return;
-}
-
-
-
-/*
- * function to decode more symbols from the arithmetic bytestream, using method of bisection
- * cdf tables should be of size 2^k-1 (which corresponds to an alphabet size of 2^k-2)
- */
-int WebRtcIsac_DecHistBisectMulti(int *data, /* output: data vector */
- Bitstr *streamdata, /* in-/output struct containing bitstream */
- const WebRtc_UWord16 **cdf, /* input: array of cdf arrays */
- const WebRtc_UWord16 *cdf_size, /* input: array of cdf table sizes+1 (power of two: 2^k) */
- const int N) /* input: data vector length */
-{
- WebRtc_UWord32 W_lower, W_upper;
- WebRtc_UWord32 W_tmp;
- WebRtc_UWord32 W_upper_LSB, W_upper_MSB;
- WebRtc_UWord32 streamval;
- const WebRtc_UWord8 *stream_ptr;
- const WebRtc_UWord16 *cdf_ptr;
- int size_tmp;
- int k;
-
- W_lower = 0; //to remove warning -DH
- stream_ptr = streamdata->stream + streamdata->stream_index;
- W_upper = streamdata->W_upper;
- if (W_upper == 0)
- /* Should not be possible in normal operation */
- return -2;
-
- if (streamdata->stream_index == 0) /* first time decoder is called for this stream */
- {
- /* read first word from bytestream */
- streamval = *stream_ptr << 24;
- streamval |= *++stream_ptr << 16;
- streamval |= *++stream_ptr << 8;
- streamval |= *++stream_ptr;
- } else {
- streamval = streamdata->streamval;
- }
-
- for (k=N; k>0; k--)
- {
- /* find the integer *data for which streamval lies in [W_lower+1, W_upper] */
- W_upper_LSB = W_upper & 0x0000FFFF;
- W_upper_MSB = W_upper >> 16;
-
- /* start halfway the cdf range */
- size_tmp = *cdf_size++ >> 1;
- cdf_ptr = *cdf + (size_tmp - 1);
-
- /* method of bisection */
- for ( ;; )
- {
- W_tmp = W_upper_MSB * *cdf_ptr;
- W_tmp += (W_upper_LSB * *cdf_ptr) >> 16;
- size_tmp >>= 1;
- if (size_tmp == 0) break;
- if (streamval > W_tmp)
- {
- W_lower = W_tmp;
- cdf_ptr += size_tmp;
- } else {
- W_upper = W_tmp;
- cdf_ptr -= size_tmp;
- }
- }
- if (streamval > W_tmp)
- {
- W_lower = W_tmp;
- *data++ = (int)(cdf_ptr - *cdf++);
- } else {
- W_upper = W_tmp;
- *data++ = (int)(cdf_ptr - *cdf++ - 1);
- }
-
- /* shift interval to start at zero */
- W_upper -= ++W_lower;
-
- /* add integer to bitstream */
- streamval -= W_lower;
-
- /* renormalize interval and update streamval */
- while ( !(W_upper & 0xFF000000) ) /* W_upper < 2^24 */
- {
- /* read next byte from stream */
- streamval = (streamval << 8) | *++stream_ptr;
- W_upper <<= 8;
- }
-
- if (W_upper == 0)
- /* Should not be possible in normal operation */
- return -2;
-
-
- }
-
- streamdata->stream_index = (int)(stream_ptr - streamdata->stream);
- streamdata->W_upper = W_upper;
- streamdata->streamval = streamval;
-
-
- /* find number of bytes in original stream (determined by current interval width) */
- if ( W_upper > 0x01FFFFFF )
- return streamdata->stream_index - 2;
- else
- return streamdata->stream_index - 1;
-}
-
-
-
-/*
- * function to decode more symbols from the arithmetic bytestream, taking single step up or
- * down at a time
- * cdf tables can be of arbitrary size, but large tables may take a lot of iterations
- */
-int WebRtcIsac_DecHistOneStepMulti(int *data, /* output: data vector */
- Bitstr *streamdata, /* in-/output struct containing bitstream */
- const WebRtc_UWord16 **cdf, /* input: array of cdf arrays */
- const WebRtc_UWord16 *init_index, /* input: vector of initial cdf table search entries */
- const int N) /* input: data vector length */
-{
- WebRtc_UWord32 W_lower, W_upper;
- WebRtc_UWord32 W_tmp;
- WebRtc_UWord32 W_upper_LSB, W_upper_MSB;
- WebRtc_UWord32 streamval;
- const WebRtc_UWord8 *stream_ptr;
- const WebRtc_UWord16 *cdf_ptr;
- int k;
-
-
- stream_ptr = streamdata->stream + streamdata->stream_index;
- W_upper = streamdata->W_upper;
- if (W_upper == 0)
- /* Should not be possible in normal operation */
- return -2;
-
- if (streamdata->stream_index == 0) /* first time decoder is called for this stream */
- {
- /* read first word from bytestream */
- streamval = *stream_ptr << 24;
- streamval |= *++stream_ptr << 16;
- streamval |= *++stream_ptr << 8;
- streamval |= *++stream_ptr;
- } else {
- streamval = streamdata->streamval;
- }
-
-
- for (k=N; k>0; k--)
- {
- /* find the integer *data for which streamval lies in [W_lower+1, W_upper] */
- W_upper_LSB = W_upper & 0x0000FFFF;
- W_upper_MSB = W_upper >> 16;
-
- /* start at the specified table entry */
- cdf_ptr = *cdf + (*init_index++);
- W_tmp = W_upper_MSB * *cdf_ptr;
- W_tmp += (W_upper_LSB * *cdf_ptr) >> 16;
- if (streamval > W_tmp)
- {
- for ( ;; )
- {
- W_lower = W_tmp;
- if (cdf_ptr[0]==65535)
- /* range check */
- return -3;
- W_tmp = W_upper_MSB * *++cdf_ptr;
- W_tmp += (W_upper_LSB * *cdf_ptr) >> 16;
- if (streamval <= W_tmp) break;
- }
- W_upper = W_tmp;
- *data++ = (int)(cdf_ptr - *cdf++ - 1);
- } else {
- for ( ;; )
- {
- W_upper = W_tmp;
- --cdf_ptr;
- if (cdf_ptr<*cdf) {
- /* range check */
- return -3;
- }
- W_tmp = W_upper_MSB * *cdf_ptr;
- W_tmp += (W_upper_LSB * *cdf_ptr) >> 16;
- if (streamval > W_tmp) break;
- }
- W_lower = W_tmp;
- *data++ = (int)(cdf_ptr - *cdf++);
- }
-
- /* shift interval to start at zero */
- W_upper -= ++W_lower;
- /* add integer to bitstream */
- streamval -= W_lower;
-
- /* renormalize interval and update streamval */
- while ( !(W_upper & 0xFF000000) ) /* W_upper < 2^24 */
- {
- /* read next byte from stream */
- streamval = (streamval << 8) | *++stream_ptr;
- W_upper <<= 8;
- }
- }
-
- streamdata->stream_index = (int)(stream_ptr - streamdata->stream);
- streamdata->W_upper = W_upper;
- streamdata->streamval = streamval;
-
-
- /* find number of bytes in original stream (determined by current interval width) */
- if ( W_upper > 0x01FFFFFF )
- return streamdata->stream_index - 2;
- else
- return streamdata->stream_index - 1;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines_logist.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines_logist.c
deleted file mode 100644
index 422855a..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/arith_routines_logist.c
+++ /dev/null
@@ -1,294 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * arith_routines.h
- *
- * This file contains functions for arithmatically encoding and
- * decoding DFT coefficients.
- *
- */
-
-
-#include "arith_routines.h"
-
-
-
-static const WebRtc_Word32 kHistEdgesQ15[51] = {
- -327680, -314573, -301466, -288359, -275252, -262144, -249037, -235930, -222823, -209716,
- -196608, -183501, -170394, -157287, -144180, -131072, -117965, -104858, -91751, -78644,
- -65536, -52429, -39322, -26215, -13108, 0, 13107, 26214, 39321, 52428,
- 65536, 78643, 91750, 104857, 117964, 131072, 144179, 157286, 170393, 183500,
- 196608, 209715, 222822, 235929, 249036, 262144, 275251, 288358, 301465, 314572,
- 327680};
-
-
-static const int kCdfSlopeQ0[51] = { /* Q0 */
- 5, 5, 5, 5, 5, 5, 5, 5, 5, 5,
- 5, 5, 13, 23, 47, 87, 154, 315, 700, 1088,
- 2471, 6064, 14221, 21463, 36634, 36924, 19750, 13270, 5806, 2312,
- 1095, 660, 316, 145, 86, 41, 32, 5, 5, 5,
- 5, 5, 5, 5, 5, 5, 5, 5, 5, 2, 0};
-
-
-static const int kCdfQ16[51] = { /* Q16 */
- 0, 2, 4, 6, 8, 10, 12, 14, 16, 18,
- 20, 22, 24, 29, 38, 57, 92, 153, 279, 559,
- 994, 1983, 4408, 10097, 18682, 33336, 48105, 56005, 61313, 63636,
- 64560, 64998, 65262, 65389, 65447, 65481, 65497, 65510, 65512, 65514,
- 65516, 65518, 65520, 65522, 65524, 65526, 65528, 65530, 65532, 65534,
- 65535};
-
-
-
-/* function to be converted to fixed point */
-static __inline WebRtc_UWord32 piecewise(WebRtc_Word32 xinQ15) {
-
- WebRtc_Word32 ind, qtmp1, qtmp2, qtmp3;
- WebRtc_UWord32 tmpUW32;
-
-
- qtmp2 = xinQ15;
-
- if (qtmp2 < kHistEdgesQ15[0]) {
- qtmp2 = kHistEdgesQ15[0];
- }
- if (qtmp2 > kHistEdgesQ15[50]) {
- qtmp2 = kHistEdgesQ15[50];
- }
-
- qtmp1 = qtmp2 - kHistEdgesQ15[0]; /* Q15 - Q15 = Q15 */
- ind = (qtmp1 * 5) >> 16; /* 2^16 / 5 = 0.4 in Q15 */
- /* Q15 -> Q0 */
- qtmp1 = qtmp2 - kHistEdgesQ15[ind]; /* Q15 - Q15 = Q15 */
- qtmp2 = kCdfSlopeQ0[ind] * qtmp1; /* Q0 * Q15 = Q15 */
- qtmp3 = qtmp2>>15; /* Q15 -> Q0 */
-
- tmpUW32 = kCdfQ16[ind] + qtmp3; /* Q0 + Q0 = Q0 */
- return tmpUW32;
-}
-
-
-
-int WebRtcIsac_EncLogisticMulti2(
- Bitstr *streamdata, /* in-/output struct containing bitstream */
- WebRtc_Word16 *dataQ7, /* input: data vector */
- const WebRtc_UWord16 *envQ8, /* input: side info vector defining the width of the pdf */
- const int N, /* input: data vector length / 2 */
- const WebRtc_Word16 isSWB12kHz)
-{
- WebRtc_UWord32 W_lower, W_upper;
- WebRtc_UWord32 W_upper_LSB, W_upper_MSB;
- WebRtc_UWord8 *stream_ptr;
- WebRtc_UWord8 *maxStreamPtr;
- WebRtc_UWord8 *stream_ptr_carry;
- WebRtc_UWord32 cdf_lo, cdf_hi;
- int k;
-
- /* point to beginning of stream buffer */
- stream_ptr = streamdata->stream + streamdata->stream_index;
- W_upper = streamdata->W_upper;
-
- maxStreamPtr = streamdata->stream + STREAM_SIZE_MAX_60 - 1;
- for (k = 0; k < N; k++)
- {
- /* compute cdf_lower and cdf_upper by evaluating the piecewise linear cdf */
- cdf_lo = piecewise((*dataQ7 - 64) * *envQ8);
- cdf_hi = piecewise((*dataQ7 + 64) * *envQ8);
-
- /* test and clip if probability gets too small */
- while (cdf_lo+1 >= cdf_hi) {
- /* clip */
- if (*dataQ7 > 0) {
- *dataQ7 -= 128;
- cdf_hi = cdf_lo;
- cdf_lo = piecewise((*dataQ7 - 64) * *envQ8);
- } else {
- *dataQ7 += 128;
- cdf_lo = cdf_hi;
- cdf_hi = piecewise((*dataQ7 + 64) * *envQ8);
- }
- }
-
- dataQ7++;
- // increment only once per 4 iterations for SWB-16kHz or WB
- // increment only once per 2 iterations for SWB-12kHz
- envQ8 += (isSWB12kHz)? (k & 1):((k & 1) & (k >> 1));
-
-
- /* update interval */
- W_upper_LSB = W_upper & 0x0000FFFF;
- W_upper_MSB = W_upper >> 16;
- W_lower = W_upper_MSB * cdf_lo;
- W_lower += (W_upper_LSB * cdf_lo) >> 16;
- W_upper = W_upper_MSB * cdf_hi;
- W_upper += (W_upper_LSB * cdf_hi) >> 16;
-
- /* shift interval such that it begins at zero */
- W_upper -= ++W_lower;
-
- /* add integer to bitstream */
- streamdata->streamval += W_lower;
-
- /* handle carry */
- if (streamdata->streamval < W_lower)
- {
- /* propagate carry */
- stream_ptr_carry = stream_ptr;
- while (!(++(*--stream_ptr_carry)));
- }
-
- /* renormalize interval, store most significant byte of streamval and update streamval */
- while ( !(W_upper & 0xFF000000) ) /* W_upper < 2^24 */
- {
- W_upper <<= 8;
- *stream_ptr++ = (WebRtc_UWord8) (streamdata->streamval >> 24);
-
- if(stream_ptr > maxStreamPtr)
- {
- return -ISAC_DISALLOWED_BITSTREAM_LENGTH;
- }
- streamdata->streamval <<= 8;
- }
- }
-
- /* calculate new stream_index */
- streamdata->stream_index = (int)(stream_ptr - streamdata->stream);
- streamdata->W_upper = W_upper;
-
- return 0;
-}
-
-
-
-int WebRtcIsac_DecLogisticMulti2(
- WebRtc_Word16 *dataQ7, /* output: data vector */
- Bitstr *streamdata, /* in-/output struct containing bitstream */
- const WebRtc_UWord16 *envQ8, /* input: side info vector defining the width of the pdf */
- const WebRtc_Word16 *ditherQ7,/* input: dither vector */
- const int N, /* input: data vector length */
- const WebRtc_Word16 isSWB12kHz)
-{
- WebRtc_UWord32 W_lower, W_upper;
- WebRtc_UWord32 W_tmp;
- WebRtc_UWord32 W_upper_LSB, W_upper_MSB;
- WebRtc_UWord32 streamval;
- const WebRtc_UWord8 *stream_ptr;
- WebRtc_UWord32 cdf_tmp;
- WebRtc_Word16 candQ7;
- int k;
-
- stream_ptr = streamdata->stream + streamdata->stream_index;
- W_upper = streamdata->W_upper;
- if (streamdata->stream_index == 0) /* first time decoder is called for this stream */
- {
- /* read first word from bytestream */
- streamval = *stream_ptr << 24;
- streamval |= *++stream_ptr << 16;
- streamval |= *++stream_ptr << 8;
- streamval |= *++stream_ptr;
- } else {
- streamval = streamdata->streamval;
- }
-
-
- for (k = 0; k < N; k++)
- {
- /* find the integer *data for which streamval lies in [W_lower+1, W_upper] */
- W_upper_LSB = W_upper & 0x0000FFFF;
- W_upper_MSB = W_upper >> 16;
-
- /* find first candidate by inverting the logistic cdf */
- candQ7 = - *ditherQ7 + 64;
- cdf_tmp = piecewise(candQ7 * *envQ8);
-
- W_tmp = W_upper_MSB * cdf_tmp;
- W_tmp += (W_upper_LSB * cdf_tmp) >> 16;
- if (streamval > W_tmp)
- {
- W_lower = W_tmp;
- candQ7 += 128;
- cdf_tmp = piecewise(candQ7 * *envQ8);
-
- W_tmp = W_upper_MSB * cdf_tmp;
- W_tmp += (W_upper_LSB * cdf_tmp) >> 16;
- while (streamval > W_tmp)
- {
- W_lower = W_tmp;
- candQ7 += 128;
- cdf_tmp = piecewise(candQ7 * *envQ8);
-
- W_tmp = W_upper_MSB * cdf_tmp;
- W_tmp += (W_upper_LSB * cdf_tmp) >> 16;
-
- /* error check */
- if (W_lower == W_tmp) return -1;
- }
- W_upper = W_tmp;
-
- /* another sample decoded */
- *dataQ7 = candQ7 - 64;
- }
- else
- {
- W_upper = W_tmp;
- candQ7 -= 128;
- cdf_tmp = piecewise(candQ7 * *envQ8);
-
- W_tmp = W_upper_MSB * cdf_tmp;
- W_tmp += (W_upper_LSB * cdf_tmp) >> 16;
- while ( !(streamval > W_tmp) )
- {
- W_upper = W_tmp;
- candQ7 -= 128;
- cdf_tmp = piecewise(candQ7 * *envQ8);
-
- W_tmp = W_upper_MSB * cdf_tmp;
- W_tmp += (W_upper_LSB * cdf_tmp) >> 16;
-
- /* error check */
- if (W_upper == W_tmp) return -1;
- }
- W_lower = W_tmp;
-
- /* another sample decoded */
- *dataQ7 = candQ7 + 64;
- }
- ditherQ7++;
- dataQ7++;
- // increment only once per 4 iterations for SWB-16kHz or WB
- // increment only once per 2 iterations for SWB-12kHz
- envQ8 += (isSWB12kHz)? (k & 1):((k & 1) & (k >> 1));
-
- /* shift interval to start at zero */
- W_upper -= ++W_lower;
-
- /* add integer to bitstream */
- streamval -= W_lower;
-
- /* renormalize interval and update streamval */
- while ( !(W_upper & 0xFF000000) ) /* W_upper < 2^24 */
- {
- /* read next byte from stream */
- streamval = (streamval << 8) | *++stream_ptr;
- W_upper <<= 8;
- }
- }
-
- streamdata->stream_index = (int)(stream_ptr - streamdata->stream);
- streamdata->W_upper = W_upper;
- streamdata->streamval = streamval;
-
- /* find number of bytes in original stream (determined by current interval width) */
- if ( W_upper > 0x01FFFFFF )
- return streamdata->stream_index - 2;
- else
- return streamdata->stream_index - 1;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/bandwidth_estimator.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/bandwidth_estimator.c
deleted file mode 100644
index d0a50c5..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/bandwidth_estimator.c
+++ /dev/null
@@ -1,1020 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * BwEstimator.c
- *
- * This file contains the code for the Bandwidth Estimator designed
- * for iSAC.
- *
- */
-
-#include "bandwidth_estimator.h"
-#include "settings.h"
-#include "isac.h"
-
-#include <math.h>
-
-/* array of quantization levels for bottle neck info; Matlab code: */
-/* sprintf('%4.1ff, ', logspace(log10(5000), log10(40000), 12)) */
-static const float kQRateTableWb[12] =
-{
- 10000.0f, 11115.3f, 12355.1f, 13733.1f, 15264.8f, 16967.3f,
- 18859.8f, 20963.3f, 23301.4f, 25900.3f, 28789.0f, 32000.0f};
-
-
-static const float kQRateTableSwb[24] =
-{
- 10000.0f, 11115.3f, 12355.1f, 13733.1f, 15264.8f, 16967.3f,
- 18859.8f, 20963.3f, 23153.1f, 25342.9f, 27532.7f, 29722.5f,
- 31912.3f, 34102.1f, 36291.9f, 38481.7f, 40671.4f, 42861.2f,
- 45051.0f, 47240.8f, 49430.6f, 51620.4f, 53810.2f, 56000.0f,
-};
-
-
-
-
-WebRtc_Word32 WebRtcIsac_InitBandwidthEstimator(
- BwEstimatorstr* bwest_str,
- enum IsacSamplingRate encoderSampRate,
- enum IsacSamplingRate decoderSampRate)
-{
- switch(encoderSampRate)
- {
- case kIsacWideband:
- {
- bwest_str->send_bw_avg = INIT_BN_EST_WB;
- break;
- }
- case kIsacSuperWideband:
- {
- bwest_str->send_bw_avg = INIT_BN_EST_SWB;
- break;
- }
- }
-
- switch(decoderSampRate)
- {
- case kIsacWideband:
- {
- bwest_str->prev_frame_length = INIT_FRAME_LEN_WB;
- bwest_str->rec_bw_inv = 1.0f /
- (INIT_BN_EST_WB + INIT_HDR_RATE_WB);
- bwest_str->rec_bw = (WebRtc_Word32)INIT_BN_EST_WB;
- bwest_str->rec_bw_avg_Q = INIT_BN_EST_WB;
- bwest_str->rec_bw_avg = INIT_BN_EST_WB + INIT_HDR_RATE_WB;
- bwest_str->rec_header_rate = INIT_HDR_RATE_WB;
- break;
- }
- case kIsacSuperWideband:
- {
- bwest_str->prev_frame_length = INIT_FRAME_LEN_SWB;
- bwest_str->rec_bw_inv = 1.0f /
- (INIT_BN_EST_SWB + INIT_HDR_RATE_SWB);
- bwest_str->rec_bw = (WebRtc_Word32)INIT_BN_EST_SWB;
- bwest_str->rec_bw_avg_Q = INIT_BN_EST_SWB;
- bwest_str->rec_bw_avg = INIT_BN_EST_SWB + INIT_HDR_RATE_SWB;
- bwest_str->rec_header_rate = INIT_HDR_RATE_SWB;
- break;
- }
- }
-
- bwest_str->prev_rec_rtp_number = 0;
- bwest_str->prev_rec_arr_ts = 0;
- bwest_str->prev_rec_send_ts = 0;
- bwest_str->prev_rec_rtp_rate = 1.0f;
- bwest_str->last_update_ts = 0;
- bwest_str->last_reduction_ts = 0;
- bwest_str->count_tot_updates_rec = -9;
- bwest_str->rec_jitter = 10.0f;
- bwest_str->rec_jitter_short_term = 0.0f;
- bwest_str->rec_jitter_short_term_abs = 5.0f;
- bwest_str->rec_max_delay = 10.0f;
- bwest_str->rec_max_delay_avg_Q = 10.0f;
- bwest_str->num_pkts_rec = 0;
-
- bwest_str->send_max_delay_avg = 10.0f;
-
- bwest_str->hsn_detect_rec = 0;
-
- bwest_str->num_consec_rec_pkts_over_30k = 0;
-
- bwest_str->hsn_detect_snd = 0;
-
- bwest_str->num_consec_snt_pkts_over_30k = 0;
-
- bwest_str->in_wait_period = 0;
-
- bwest_str->change_to_WB = 0;
-
- bwest_str->numConsecLatePkts = 0;
- bwest_str->consecLatency = 0;
- bwest_str->inWaitLatePkts = 0;
- bwest_str->senderTimestamp = 0;
- bwest_str->receiverTimestamp = 0;
- return 0;
-}
-
-/* This function updates both bottle neck rates */
-/* Parameters: */
-/* rtp_number - value from RTP packet, from NetEq */
-/* frame length - length of signal frame in ms, from iSAC decoder */
-/* send_ts - value in RTP header giving send time in samples */
-/* arr_ts - value given by timeGetTime() time of arrival in samples of packet from NetEq */
-/* pksize - size of packet in bytes, from NetEq */
-/* Index - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
-/* returns 0 if everything went fine, -1 otherwise */
-WebRtc_Word16 WebRtcIsac_UpdateBandwidthEstimator(
- BwEstimatorstr *bwest_str,
- const WebRtc_UWord16 rtp_number,
- const WebRtc_Word32 frame_length,
- const WebRtc_UWord32 send_ts,
- const WebRtc_UWord32 arr_ts,
- const WebRtc_Word32 pksize
- /*, const WebRtc_UWord16 Index*/)
-{
- float weight = 0.0f;
- float curr_bw_inv = 0.0f;
- float rec_rtp_rate;
- float t_diff_proj;
- float arr_ts_diff;
- float send_ts_diff;
- float arr_time_noise;
- float arr_time_noise_abs;
-
- float delay_correction_factor = 1;
- float late_diff = 0.0f;
- int immediate_set = 0;
- int num_pkts_expected;
-
-
- // We have to adjust the header-rate if the first packet has a
- // frame-size different than the initialized value.
- if ( frame_length != bwest_str->prev_frame_length )
- {
- bwest_str->rec_header_rate = (float)HEADER_SIZE * 8.0f *
- 1000.0f / (float)frame_length; /* bits/s */
- }
-
- /* UPDATE ESTIMATES ON THIS SIDE */
- /* compute far-side transmission rate */
- rec_rtp_rate = ((float)pksize * 8.0f * 1000.0f / (float)frame_length) +
- bwest_str->rec_header_rate;
- // rec_rtp_rate packet bits/s + header bits/s
-
- /* check for timer wrap-around */
- if (arr_ts < bwest_str->prev_rec_arr_ts)
- {
- bwest_str->prev_rec_arr_ts = arr_ts;
- bwest_str->last_update_ts = arr_ts;
- bwest_str->last_reduction_ts = arr_ts + 3*FS;
- bwest_str->num_pkts_rec = 0;
-
- /* store frame length */
- bwest_str->prev_frame_length = frame_length;
-
- /* store far-side transmission rate */
- bwest_str->prev_rec_rtp_rate = rec_rtp_rate;
-
- /* store far-side RTP time stamp */
- bwest_str->prev_rec_rtp_number = rtp_number;
-
- return 0;
- }
-
- bwest_str->num_pkts_rec++;
-
- /* check that it's not one of the first 9 packets */
- if ( bwest_str->count_tot_updates_rec > 0 )
- {
- if(bwest_str->in_wait_period > 0 )
- {
- bwest_str->in_wait_period--;
- }
-
- bwest_str->inWaitLatePkts -= ((bwest_str->inWaitLatePkts > 0)? 1:0);
- send_ts_diff = (float)(send_ts - bwest_str->prev_rec_send_ts);
-
- if (send_ts_diff <= (16 * frame_length)*2)
- //doesn't allow for a dropped packet, not sure necessary to be
- // that strict -DH
- {
- /* if not been updated for a long time, reduce the BN estimate */
- if((WebRtc_UWord32)(arr_ts - bwest_str->last_update_ts) *
- 1000.0f / FS > 3000)
- {
- //how many frames should have been received since the last
- // update if too many have been dropped or there have been
- // big delays won't allow this reduction may no longer need
- // the send_ts_diff here
- num_pkts_expected = (int)(((float)(arr_ts -
- bwest_str->last_update_ts) * 1000.0f /(float) FS) /
- (float)frame_length);
-
- if(((float)bwest_str->num_pkts_rec/(float)num_pkts_expected) >
- 0.9)
- {
- float inv_bitrate = (float) pow( 0.99995,
- (double)((WebRtc_UWord32)(arr_ts -
- bwest_str->last_reduction_ts)*1000.0f/FS) );
-
- if ( inv_bitrate )
- {
- bwest_str->rec_bw_inv /= inv_bitrate;
-
- //precautionary, likely never necessary
- if (bwest_str->hsn_detect_snd &&
- bwest_str->hsn_detect_rec)
- {
- if (bwest_str->rec_bw_inv > 0.000066f)
- {
- bwest_str->rec_bw_inv = 0.000066f;
- }
- }
- }
- else
- {
- bwest_str->rec_bw_inv = 1.0f /
- (INIT_BN_EST_WB + INIT_HDR_RATE_WB);
- }
- /* reset time-since-update counter */
- bwest_str->last_reduction_ts = arr_ts;
- }
- else
- //reset here?
- {
- bwest_str->last_reduction_ts = arr_ts + 3*FS;
- bwest_str->last_update_ts = arr_ts;
- bwest_str->num_pkts_rec = 0;
- }
- }
- }
- else
- {
- bwest_str->last_reduction_ts = arr_ts + 3*FS;
- bwest_str->last_update_ts = arr_ts;
- bwest_str->num_pkts_rec = 0;
- }
-
-
- /* temporarily speed up adaptation if frame length has changed */
- if ( frame_length != bwest_str->prev_frame_length )
- {
- bwest_str->count_tot_updates_rec = 10;
- bwest_str->rec_header_rate = (float)HEADER_SIZE * 8.0f *
- 1000.0f / (float)frame_length; /* bits/s */
-
- bwest_str->rec_bw_inv = 1.0f /((float)bwest_str->rec_bw +
- bwest_str->rec_header_rate);
- }
-
- ////////////////////////
- arr_ts_diff = (float)(arr_ts - bwest_str->prev_rec_arr_ts);
-
- if (send_ts_diff > 0 )
- {
- late_diff = arr_ts_diff - send_ts_diff;
- }
- else
- {
- late_diff = arr_ts_diff - (float)(16 * frame_length);
- }
-
- if((late_diff > 0) && !bwest_str->inWaitLatePkts)
- {
- bwest_str->numConsecLatePkts++;
- bwest_str->consecLatency += late_diff;
- }
- else
- {
- bwest_str->numConsecLatePkts = 0;
- bwest_str->consecLatency = 0;
- }
- if(bwest_str->numConsecLatePkts > 50)
- {
- float latencyMs = bwest_str->consecLatency/(FS/1000);
- float averageLatencyMs = latencyMs / bwest_str->numConsecLatePkts;
- delay_correction_factor = frame_length / (frame_length + averageLatencyMs);
- immediate_set = 1;
- bwest_str->inWaitLatePkts = (WebRtc_Word16)((bwest_str->consecLatency/(FS/1000)) / 30);// + 150;
- bwest_str->start_wait_period = arr_ts;
- }
- ///////////////////////////////////////////////
-
-
-
- /* update only if previous packet was not lost */
- if ( rtp_number == bwest_str->prev_rec_rtp_number + 1 )
- {
-
-
- if (!(bwest_str->hsn_detect_snd && bwest_str->hsn_detect_rec))
- {
- if ((arr_ts_diff > (float)(16 * frame_length)))
- {
- //1/2 second
- if ((late_diff > 8000.0f) && !bwest_str->in_wait_period)
- {
- delay_correction_factor = 0.7f;
- bwest_str->in_wait_period = 55;
- bwest_str->start_wait_period = arr_ts;
- immediate_set = 1;
- }
- //320 ms
- else if (late_diff > 5120.0f && !bwest_str->in_wait_period)
- {
- delay_correction_factor = 0.8f;
- immediate_set = 1;
- bwest_str->in_wait_period = 44;
- bwest_str->start_wait_period = arr_ts;
- }
- }
- }
-
-
- if ((bwest_str->prev_rec_rtp_rate > bwest_str->rec_bw_avg) &&
- (rec_rtp_rate > bwest_str->rec_bw_avg) &&
- !bwest_str->in_wait_period)
- {
- /* test if still in initiation period and increment counter */
- if (bwest_str->count_tot_updates_rec++ > 99)
- {
- /* constant weight after initiation part */
- weight = 0.01f;
- }
- else
- {
- /* weight decreases with number of updates */
- weight = 1.0f / (float) bwest_str->count_tot_updates_rec;
- }
- /* Bottle Neck Estimation */
-
- /* limit outliers */
- /* if more than 25 ms too much */
- if (arr_ts_diff > frame_length * FS/1000 + 400.0f)
- {
- // in samples, why 25ms??
- arr_ts_diff = frame_length * FS/1000 + 400.0f;
- }
- if(arr_ts_diff < (frame_length * FS/1000) - 160.0f)
- {
- /* don't allow it to be less than frame rate - 10 ms */
- arr_ts_diff = (float)frame_length * FS/1000 - 160.0f;
- }
-
- /* compute inverse receiving rate for last packet */
- curr_bw_inv = arr_ts_diff / ((float)(pksize + HEADER_SIZE) *
- 8.0f * FS); // (180+35)*8*16000 = 27.5 Mbit....
-
-
- if(curr_bw_inv <
- (1.0f / (MAX_ISAC_BW + bwest_str->rec_header_rate)))
- {
- // don't allow inv rate to be larger than MAX
- curr_bw_inv = (1.0f /
- (MAX_ISAC_BW + bwest_str->rec_header_rate));
- }
-
- /* update bottle neck rate estimate */
- bwest_str->rec_bw_inv = weight * curr_bw_inv +
- (1.0f - weight) * bwest_str->rec_bw_inv;
-
- /* reset time-since-update counter */
- bwest_str->last_update_ts = arr_ts;
- bwest_str->last_reduction_ts = arr_ts + 3 * FS;
- bwest_str->num_pkts_rec = 0;
-
- /* Jitter Estimation */
- /* projected difference between arrival times */
- t_diff_proj = ((float)(pksize + HEADER_SIZE) * 8.0f *
- 1000.0f) / bwest_str->rec_bw_avg;
-
-
- // difference between projected and actual
- // arrival time differences
- arr_time_noise = (float)(arr_ts_diff*1000.0f/FS) -
- t_diff_proj;
- arr_time_noise_abs = (float) fabs( arr_time_noise );
-
- /* long term averaged absolute jitter */
- bwest_str->rec_jitter = weight * arr_time_noise_abs +
- (1.0f - weight) * bwest_str->rec_jitter;
- if (bwest_str->rec_jitter > 10.0f)
- {
- bwest_str->rec_jitter = 10.0f;
- }
- /* short term averaged absolute jitter */
- bwest_str->rec_jitter_short_term_abs = 0.05f *
- arr_time_noise_abs + 0.95f *
- bwest_str->rec_jitter_short_term_abs;
-
- /* short term averaged jitter */
- bwest_str->rec_jitter_short_term = 0.05f * arr_time_noise +
- 0.95f * bwest_str->rec_jitter_short_term;
- }
- }
- }
- else
- {
- // reset time-since-update counter when
- // receiving the first 9 packets
- bwest_str->last_update_ts = arr_ts;
- bwest_str->last_reduction_ts = arr_ts + 3*FS;
- bwest_str->num_pkts_rec = 0;
-
- bwest_str->count_tot_updates_rec++;
- }
-
- /* limit minimum bottle neck rate */
- if (bwest_str->rec_bw_inv > 1.0f / ((float)MIN_ISAC_BW +
- bwest_str->rec_header_rate))
- {
- bwest_str->rec_bw_inv = 1.0f / ((float)MIN_ISAC_BW +
- bwest_str->rec_header_rate);
- }
-
- // limit maximum bitrate
- if (bwest_str->rec_bw_inv < 1.0f / ((float)MAX_ISAC_BW +
- bwest_str->rec_header_rate))
- {
- bwest_str->rec_bw_inv = 1.0f / ((float)MAX_ISAC_BW +
- bwest_str->rec_header_rate);
- }
-
- /* store frame length */
- bwest_str->prev_frame_length = frame_length;
-
- /* store far-side transmission rate */
- bwest_str->prev_rec_rtp_rate = rec_rtp_rate;
-
- /* store far-side RTP time stamp */
- bwest_str->prev_rec_rtp_number = rtp_number;
-
- // Replace bwest_str->rec_max_delay by the new
- // value (atomic operation)
- bwest_str->rec_max_delay = 3.0f * bwest_str->rec_jitter;
-
- /* store send and arrival time stamp */
- bwest_str->prev_rec_arr_ts = arr_ts ;
- bwest_str->prev_rec_send_ts = send_ts;
-
- /* Replace bwest_str->rec_bw by the new value (atomic operation) */
- bwest_str->rec_bw = (WebRtc_Word32)(1.0f / bwest_str->rec_bw_inv -
- bwest_str->rec_header_rate);
-
- if (immediate_set)
- {
- bwest_str->rec_bw = (WebRtc_Word32) (delay_correction_factor *
- (float) bwest_str->rec_bw);
-
- if (bwest_str->rec_bw < (WebRtc_Word32) MIN_ISAC_BW)
- {
- bwest_str->rec_bw = (WebRtc_Word32) MIN_ISAC_BW;
- }
-
- bwest_str->rec_bw_avg = bwest_str->rec_bw +
- bwest_str->rec_header_rate;
-
- bwest_str->rec_bw_avg_Q = (float) bwest_str->rec_bw;
-
- bwest_str->rec_jitter_short_term = 0.0f;
-
- bwest_str->rec_bw_inv = 1.0f / (bwest_str->rec_bw +
- bwest_str->rec_header_rate);
-
- bwest_str->count_tot_updates_rec = 1;
-
- immediate_set = 0;
- bwest_str->consecLatency = 0;
- bwest_str->numConsecLatePkts = 0;
- }
-
- return 0;
-}
-
-
-/* This function updates the send bottle neck rate */
-/* Index - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
-/* returns 0 if everything went fine, -1 otherwise */
-WebRtc_Word16 WebRtcIsac_UpdateUplinkBwImpl(
- BwEstimatorstr* bwest_str,
- WebRtc_Word16 index,
- enum IsacSamplingRate encoderSamplingFreq)
-{
- if((index < 0) || (index > 23))
- {
- return -ISAC_RANGE_ERROR_BW_ESTIMATOR;
- }
-
- /* UPDATE ESTIMATES FROM OTHER SIDE */
- if(encoderSamplingFreq == kIsacWideband)
- {
- if(index > 11)
- {
- index -= 12;
- /* compute the jitter estimate as decoded on the other side */
- bwest_str->send_max_delay_avg = 0.9f * bwest_str->send_max_delay_avg +
- 0.1f * (float)MAX_ISAC_MD;
- }
- else
- {
- /* compute the jitter estimate as decoded on the other side */
- bwest_str->send_max_delay_avg = 0.9f * bwest_str->send_max_delay_avg +
- 0.1f * (float)MIN_ISAC_MD;
- }
-
- /* compute the BN estimate as decoded on the other side */
- bwest_str->send_bw_avg = 0.9f * bwest_str->send_bw_avg +
- 0.1f * kQRateTableWb[index];
- }
- else
- {
- /* compute the BN estimate as decoded on the other side */
- bwest_str->send_bw_avg = 0.9f * bwest_str->send_bw_avg +
- 0.1f * kQRateTableSwb[index];
- }
-
- if (bwest_str->send_bw_avg > (float) 28000 && !bwest_str->hsn_detect_snd)
- {
- bwest_str->num_consec_snt_pkts_over_30k++;
-
- if (bwest_str->num_consec_snt_pkts_over_30k >= 66)
- {
- //approx 2 seconds with 30ms frames
- bwest_str->hsn_detect_snd = 1;
- }
- }
- else if (!bwest_str->hsn_detect_snd)
- {
- bwest_str->num_consec_snt_pkts_over_30k = 0;
- }
- return 0;
-}
-
-// called when there is upper-band bit-stream to update jitter
-// statistics.
-WebRtc_Word16 WebRtcIsac_UpdateUplinkJitter(
- BwEstimatorstr* bwest_str,
- WebRtc_Word32 index)
-{
- if((index < 0) || (index > 23))
- {
- return -ISAC_RANGE_ERROR_BW_ESTIMATOR;
- }
-
- if(index > 0)
- {
- /* compute the jitter estimate as decoded on the other side */
- bwest_str->send_max_delay_avg = 0.9f * bwest_str->send_max_delay_avg +
- 0.1f * (float)MAX_ISAC_MD;
- }
- else
- {
- /* compute the jitter estimate as decoded on the other side */
- bwest_str->send_max_delay_avg = 0.9f * bwest_str->send_max_delay_avg +
- 0.1f * (float)MIN_ISAC_MD;
- }
-
- return 0;
-}
-
-
-
-// Returns the bandwidth/jitter estimation code (integer 0...23)
-// to put in the sending iSAC payload
-WebRtc_UWord16
-WebRtcIsac_GetDownlinkBwJitIndexImpl(
- BwEstimatorstr* bwest_str,
- WebRtc_Word16* bottleneckIndex,
- WebRtc_Word16* jitterInfo,
- enum IsacSamplingRate decoderSamplingFreq)
-{
- float MaxDelay;
- //WebRtc_UWord16 MaxDelayBit;
-
- float rate;
- float r;
- float e1, e2;
- const float weight = 0.1f;
- const float* ptrQuantizationTable;
- WebRtc_Word16 addJitterInfo;
- WebRtc_Word16 minInd;
- WebRtc_Word16 maxInd;
- WebRtc_Word16 midInd;
-
- /* Get Max Delay Bit */
- /* get unquantized max delay */
- MaxDelay = (float)WebRtcIsac_GetDownlinkMaxDelay(bwest_str);
-
- if ( ((1.f - weight) * bwest_str->rec_max_delay_avg_Q + weight *
- MAX_ISAC_MD - MaxDelay) > (MaxDelay - (1.f-weight) *
- bwest_str->rec_max_delay_avg_Q - weight * MIN_ISAC_MD) )
- {
- jitterInfo[0] = 0;
- /* update quantized average */
- bwest_str->rec_max_delay_avg_Q =
- (1.f - weight) * bwest_str->rec_max_delay_avg_Q + weight *
- (float)MIN_ISAC_MD;
- }
- else
- {
- jitterInfo[0] = 1;
- /* update quantized average */
- bwest_str->rec_max_delay_avg_Q =
- (1.f-weight) * bwest_str->rec_max_delay_avg_Q + weight *
- (float)MAX_ISAC_MD;
- }
-
- // Get unquantized rate.
- rate = (float)WebRtcIsac_GetDownlinkBandwidth(bwest_str);
-
- /* Get Rate Index */
- if(decoderSamplingFreq == kIsacWideband)
- {
- ptrQuantizationTable = kQRateTableWb;
- addJitterInfo = 1;
- maxInd = 11;
- }
- else
- {
- ptrQuantizationTable = kQRateTableSwb;
- addJitterInfo = 0;
- maxInd = 23;
- }
-
- minInd = 0;
- while(maxInd > minInd + 1)
- {
- midInd = (maxInd + minInd) >> 1;
- if(rate > ptrQuantizationTable[midInd])
- {
- minInd = midInd;
- }
- else
- {
- maxInd = midInd;
- }
- }
- // Chose the index which gives results an average which is closest
- // to rate
- r = (1 - weight) * bwest_str->rec_bw_avg_Q - rate;
- e1 = weight * ptrQuantizationTable[minInd] + r;
- e2 = weight * ptrQuantizationTable[maxInd] + r;
- e1 = (e1 > 0)? e1:-e1;
- e2 = (e2 > 0)? e2:-e2;
- if(e1 < e2)
- {
- bottleneckIndex[0] = minInd;
- }
- else
- {
- bottleneckIndex[0] = maxInd;
- }
-
- bwest_str->rec_bw_avg_Q = (1 - weight) * bwest_str->rec_bw_avg_Q +
- weight * ptrQuantizationTable[bottleneckIndex[0]];
- bottleneckIndex[0] += jitterInfo[0] * 12 * addJitterInfo;
-
- bwest_str->rec_bw_avg = (1 - weight) * bwest_str->rec_bw_avg + weight *
- (rate + bwest_str->rec_header_rate);
-
- return 0;
-}
-
-
-
-/* get the bottle neck rate from far side to here, as estimated on this side */
-WebRtc_Word32 WebRtcIsac_GetDownlinkBandwidth( const BwEstimatorstr *bwest_str)
-{
- WebRtc_Word32 rec_bw;
- float jitter_sign;
- float bw_adjust;
-
- /* create a value between -1.0 and 1.0 indicating "average sign" of jitter */
- jitter_sign = bwest_str->rec_jitter_short_term /
- bwest_str->rec_jitter_short_term_abs;
-
- /* adjust bw proportionally to negative average jitter sign */
- bw_adjust = 1.0f - jitter_sign * (0.15f + 0.15f * jitter_sign * jitter_sign);
-
- /* adjust Rate if jitter sign is mostly constant */
- rec_bw = (WebRtc_Word32)(bwest_str->rec_bw * bw_adjust);
-
- /* limit range of bottle neck rate */
- if (rec_bw < MIN_ISAC_BW)
- {
- rec_bw = MIN_ISAC_BW;
- }
- else if (rec_bw > MAX_ISAC_BW)
- {
- rec_bw = MAX_ISAC_BW;
- }
- return rec_bw;
-}
-
-/* Returns the max delay (in ms) */
-WebRtc_Word32
-WebRtcIsac_GetDownlinkMaxDelay(const BwEstimatorstr *bwest_str)
-{
- WebRtc_Word32 rec_max_delay;
-
- rec_max_delay = (WebRtc_Word32)(bwest_str->rec_max_delay);
-
- /* limit range of jitter estimate */
- if (rec_max_delay < MIN_ISAC_MD)
- {
- rec_max_delay = MIN_ISAC_MD;
- }
- else if (rec_max_delay > MAX_ISAC_MD)
- {
- rec_max_delay = MAX_ISAC_MD;
- }
- return rec_max_delay;
-}
-
-/* get the bottle neck rate from here to far side, as estimated by far side */
-void
-WebRtcIsac_GetUplinkBandwidth(
- const BwEstimatorstr* bwest_str,
- WebRtc_Word32* bitRate)
-{
- /* limit range of bottle neck rate */
- if (bwest_str->send_bw_avg < MIN_ISAC_BW)
- {
- *bitRate = MIN_ISAC_BW;
- }
- else if (bwest_str->send_bw_avg > MAX_ISAC_BW)
- {
- *bitRate = MAX_ISAC_BW;
- }
- else
- {
- *bitRate = (WebRtc_Word32)(bwest_str->send_bw_avg);
- }
- return;
-}
-
-/* Returns the max delay value from the other side in ms */
-WebRtc_Word32
-WebRtcIsac_GetUplinkMaxDelay(const BwEstimatorstr *bwest_str)
-{
- WebRtc_Word32 send_max_delay;
-
- send_max_delay = (WebRtc_Word32)(bwest_str->send_max_delay_avg);
-
- /* limit range of jitter estimate */
- if (send_max_delay < MIN_ISAC_MD)
- {
- send_max_delay = MIN_ISAC_MD;
- }
- else if (send_max_delay > MAX_ISAC_MD)
- {
- send_max_delay = MAX_ISAC_MD;
- }
- return send_max_delay;
-}
-
-
-/*
- * update long-term average bitrate and amount of data in buffer
- * returns minimum payload size (bytes)
- */
-int WebRtcIsac_GetMinBytes(
- RateModel* State,
- int StreamSize, /* bytes in bitstream */
- const int FrameSamples, /* samples per frame */
- const double BottleNeck, /* bottle neck rate; excl headers (bps) */
- const double DelayBuildUp, /* max delay from bottleneck buffering (ms) */
- enum ISACBandwidth bandwidth
- /*,WebRtc_Word16 frequentLargePackets*/)
-{
- double MinRate = 0.0;
- int MinBytes;
- double TransmissionTime;
- int burstInterval = BURST_INTERVAL;
-
- // first 10 packets @ low rate, then INIT_BURST_LEN packets @
- // fixed rate of INIT_RATE bps
- if (State->InitCounter > 0)
- {
- if (State->InitCounter-- <= INIT_BURST_LEN)
- {
- if(bandwidth == isac8kHz)
- {
- MinRate = INIT_RATE_WB;
- }
- else
- {
- MinRate = INIT_RATE_SWB;
- }
- }
- else
- {
- MinRate = 0;
- }
- }
- else
- {
- /* handle burst */
- if (State->BurstCounter)
- {
- if (State->StillBuffered < (1.0 - 1.0/BURST_LEN) * DelayBuildUp)
- {
- /* max bps derived from BottleNeck and DelayBuildUp values */
- MinRate = (1.0 + (FS/1000) * DelayBuildUp /
- (double)(BURST_LEN * FrameSamples)) * BottleNeck;
- }
- else
- {
- // max bps derived from StillBuffered and DelayBuildUp
- // values
- MinRate = (1.0 + (FS/1000) * (DelayBuildUp -
- State->StillBuffered) / (double)FrameSamples) * BottleNeck;
- if (MinRate < 1.04 * BottleNeck)
- {
- MinRate = 1.04 * BottleNeck;
- }
- }
- State->BurstCounter--;
- }
- }
-
-
- /* convert rate from bits/second to bytes/packet */
- MinBytes = (int) (MinRate * FrameSamples / (8.0 * FS));
-
- /* StreamSize will be adjusted if less than MinBytes */
- if (StreamSize < MinBytes)
- {
- StreamSize = MinBytes;
- }
-
- /* keep track of when bottle neck was last exceeded by at least 1% */
- if (StreamSize * 8.0 * FS / FrameSamples > 1.01 * BottleNeck) {
- if (State->PrevExceed) {
- /* bottle_neck exceded twice in a row, decrease ExceedAgo */
- State->ExceedAgo -= /*BURST_INTERVAL*/ burstInterval / (BURST_LEN - 1);
- if (State->ExceedAgo < 0)
- State->ExceedAgo = 0;
- }
- else
- {
- State->ExceedAgo += (FrameSamples * 1000) / FS; /* ms */
- State->PrevExceed = 1;
- }
- }
- else
- {
- State->PrevExceed = 0;
- State->ExceedAgo += (FrameSamples * 1000) / FS; /* ms */
- }
-
- /* set burst flag if bottle neck not exceeded for long time */
- if ((State->ExceedAgo > burstInterval) &&
- (State->BurstCounter == 0))
- {
- if (State->PrevExceed)
- {
- State->BurstCounter = BURST_LEN - 1;
- }
- else
- {
- State->BurstCounter = BURST_LEN;
- }
- }
-
-
- /* Update buffer delay */
- TransmissionTime = StreamSize * 8.0 * 1000.0 / BottleNeck; /* ms */
- State->StillBuffered += TransmissionTime;
- State->StillBuffered -= (FrameSamples * 1000) / FS; /* ms */
- if (State->StillBuffered < 0.0)
- {
- State->StillBuffered = 0.0;
- }
-
- return MinBytes;
-}
-
-
-/*
- * update long-term average bitrate and amount of data in buffer
- */
-void WebRtcIsac_UpdateRateModel(
- RateModel *State,
- int StreamSize, /* bytes in bitstream */
- const int FrameSamples, /* samples per frame */
- const double BottleNeck) /* bottle neck rate; excl headers (bps) */
-{
- double TransmissionTime;
-
- /* avoid the initial "high-rate" burst */
- State->InitCounter = 0;
-
- /* Update buffer delay */
- TransmissionTime = StreamSize * 8.0 * 1000.0 / BottleNeck; /* ms */
- State->StillBuffered += TransmissionTime;
- State->StillBuffered -= (FrameSamples * 1000) / FS; /* ms */
- if (State->StillBuffered < 0.0)
- State->StillBuffered = 0.0;
-
-}
-
-
-void WebRtcIsac_InitRateModel(
- RateModel *State)
-{
- State->PrevExceed = 0; /* boolean */
- State->ExceedAgo = 0; /* ms */
- State->BurstCounter = 0; /* packets */
- State->InitCounter = INIT_BURST_LEN + 10; /* packets */
- State->StillBuffered = 1.0; /* ms */
-}
-
-int WebRtcIsac_GetNewFrameLength(
- double bottle_neck,
- int current_framesamples)
-{
- int new_framesamples;
-
- const int Thld_20_30 = 20000;
-
- //const int Thld_30_20 = 30000;
- const int Thld_30_20 = 1000000; // disable 20 ms frames
-
- const int Thld_30_60 = 18000;
- //const int Thld_30_60 = 0; // disable 60 ms frames
-
- const int Thld_60_30 = 27000;
-
-
- new_framesamples = current_framesamples;
-
- /* find new framelength */
- switch(current_framesamples) {
- case 320:
- if (bottle_neck < Thld_20_30)
- new_framesamples = 480;
- break;
- case 480:
- if (bottle_neck < Thld_30_60)
- new_framesamples = 960;
- else if (bottle_neck > Thld_30_20)
- new_framesamples = 320;
- break;
- case 960:
- if (bottle_neck >= Thld_60_30)
- new_framesamples = 480;
- break;
- }
-
- return new_framesamples;
-}
-
-double WebRtcIsac_GetSnr(
- double bottle_neck,
- int framesamples)
-{
- double s2nr;
-
- const double a_20 = -30.0;
- const double b_20 = 0.8;
- const double c_20 = 0.0;
-
- const double a_30 = -23.0;
- const double b_30 = 0.48;
- const double c_30 = 0.0;
-
- const double a_60 = -23.0;
- const double b_60 = 0.53;
- const double c_60 = 0.0;
-
-
- /* find new SNR value */
- switch(framesamples) {
- case 320:
- s2nr = a_20 + b_20 * bottle_neck * 0.001 + c_20 * bottle_neck *
- bottle_neck * 0.000001;
- break;
- case 480:
- s2nr = a_30 + b_30 * bottle_neck * 0.001 + c_30 * bottle_neck *
- bottle_neck * 0.000001;
- break;
- case 960:
- s2nr = a_60 + b_60 * bottle_neck * 0.001 + c_60 * bottle_neck *
- bottle_neck * 0.000001;
- break;
- default:
- s2nr = 0;
- }
-
- return s2nr;
-
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/bandwidth_estimator.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/bandwidth_estimator.h
deleted file mode 100644
index 5604d7b..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/bandwidth_estimator.h
+++ /dev/null
@@ -1,177 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * bandwidth_estimator.h
- *
- * This header file contains the API for the Bandwidth Estimator
- * designed for iSAC.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_
-
-#include "structs.h"
-#include "settings.h"
-
-
-#define MIN_ISAC_BW 10000
-#define MIN_ISAC_BW_LB 10000
-#define MIN_ISAC_BW_UB 25000
-
-#define MAX_ISAC_BW 56000
-#define MAX_ISAC_BW_UB 32000
-#define MAX_ISAC_BW_LB 32000
-
-#define MIN_ISAC_MD 5
-#define MAX_ISAC_MD 25
-
-// assumed header size, in bytes; we don't know the exact number
-// (header compression may be used)
-#define HEADER_SIZE 35
-
-// Initial Frame-Size, in ms, for Wideband & Super-Wideband Mode
-#define INIT_FRAME_LEN_WB 60
-#define INIT_FRAME_LEN_SWB 30
-
-// Initial Bottleneck Estimate, in bits/sec, for
-// Wideband & Super-wideband mode
-#define INIT_BN_EST_WB 20e3f
-#define INIT_BN_EST_SWB 56e3f
-
-// Initial Header rate (header rate depends on frame-size),
-// in bits/sec, for Wideband & Super-Wideband mode.
-#define INIT_HDR_RATE_WB \
- ((float)HEADER_SIZE * 8.0f * 1000.0f / (float)INIT_FRAME_LEN_WB)
-#define INIT_HDR_RATE_SWB \
- ((float)HEADER_SIZE * 8.0f * 1000.0f / (float)INIT_FRAME_LEN_SWB)
-
-// number of packets in a row for a high rate burst
-#define BURST_LEN 3
-
-// ms, max time between two full bursts
-#define BURST_INTERVAL 500
-
-// number of packets in a row for initial high rate burst
-#define INIT_BURST_LEN 5
-
-// bits/s, rate for the first BURST_LEN packets
-#define INIT_RATE_WB INIT_BN_EST_WB
-#define INIT_RATE_SWB INIT_BN_EST_SWB
-
-
-#if defined(__cplusplus)
-extern "C" {
-#endif
-
- /* This function initializes the struct */
- /* to be called before using the struct for anything else */
- /* returns 0 if everything went fine, -1 otherwise */
- WebRtc_Word32 WebRtcIsac_InitBandwidthEstimator(
- BwEstimatorstr* bwest_str,
- enum IsacSamplingRate encoderSampRate,
- enum IsacSamplingRate decoderSampRate);
-
- /* This function updates the receiving estimate */
- /* Parameters: */
- /* rtp_number - value from RTP packet, from NetEq */
- /* frame length - length of signal frame in ms, from iSAC decoder */
- /* send_ts - value in RTP header giving send time in samples */
- /* arr_ts - value given by timeGetTime() time of arrival in samples of packet from NetEq */
- /* pksize - size of packet in bytes, from NetEq */
- /* Index - integer (range 0...23) indicating bottle neck & jitter as estimated by other side */
- /* returns 0 if everything went fine, -1 otherwise */
- WebRtc_Word16 WebRtcIsac_UpdateBandwidthEstimator(
- BwEstimatorstr* bwest_str,
- const WebRtc_UWord16 rtp_number,
- const WebRtc_Word32 frame_length,
- const WebRtc_UWord32 send_ts,
- const WebRtc_UWord32 arr_ts,
- const WebRtc_Word32 pksize);
-
- /* Update receiving estimates. Used when we only receive BWE index, no iSAC data packet. */
- WebRtc_Word16 WebRtcIsac_UpdateUplinkBwImpl(
- BwEstimatorstr* bwest_str,
- WebRtc_Word16 Index,
- enum IsacSamplingRate encoderSamplingFreq);
-
- /* Returns the bandwidth/jitter estimation code (integer 0...23) to put in the sending iSAC payload */
- WebRtc_UWord16 WebRtcIsac_GetDownlinkBwJitIndexImpl(
- BwEstimatorstr* bwest_str,
- WebRtc_Word16* bottleneckIndex,
- WebRtc_Word16* jitterInfo,
- enum IsacSamplingRate decoderSamplingFreq);
-
- /* Returns the bandwidth estimation (in bps) */
- WebRtc_Word32 WebRtcIsac_GetDownlinkBandwidth(
- const BwEstimatorstr *bwest_str);
-
- /* Returns the max delay (in ms) */
- WebRtc_Word32 WebRtcIsac_GetDownlinkMaxDelay(
- const BwEstimatorstr *bwest_str);
-
- /* Returns the bandwidth that iSAC should send with in bps */
- void WebRtcIsac_GetUplinkBandwidth(
- const BwEstimatorstr* bwest_str,
- WebRtc_Word32* bitRate);
-
- /* Returns the max delay value from the other side in ms */
- WebRtc_Word32 WebRtcIsac_GetUplinkMaxDelay(
- const BwEstimatorstr *bwest_str);
-
-
- /*
- * update amount of data in bottle neck buffer and burst handling
- * returns minimum payload size (bytes)
- */
- int WebRtcIsac_GetMinBytes(
- RateModel* State,
- int StreamSize, /* bytes in bitstream */
- const int FrameLen, /* ms per frame */
- const double BottleNeck, /* bottle neck rate; excl headers (bps) */
- const double DelayBuildUp, /* max delay from bottleneck buffering (ms) */
- enum ISACBandwidth bandwidth
- /*,WebRtc_Word16 frequentLargePackets*/);
-
- /*
- * update long-term average bitrate and amount of data in buffer
- */
- void WebRtcIsac_UpdateRateModel(
- RateModel* State,
- int StreamSize, /* bytes in bitstream */
- const int FrameSamples, /* samples per frame */
- const double BottleNeck); /* bottle neck rate; excl headers (bps) */
-
-
- void WebRtcIsac_InitRateModel(
- RateModel *State);
-
- /* Returns the new framelength value (input argument: bottle_neck) */
- int WebRtcIsac_GetNewFrameLength(
- double bottle_neck,
- int current_framelength);
-
- /* Returns the new SNR value (input argument: bottle_neck) */
- double WebRtcIsac_GetSnr(
- double bottle_neck,
- int new_framelength);
-
-
- WebRtc_Word16 WebRtcIsac_UpdateUplinkJitter(
- BwEstimatorstr* bwest_str,
- WebRtc_Word32 index);
-
-#if defined(__cplusplus)
-}
-#endif
-
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_BANDWIDTH_ESTIMATOR_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/codec.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/codec.h
deleted file mode 100644
index 6af27ea..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/codec.h
+++ /dev/null
@@ -1,292 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * codec.h
- *
- * This header file contains the calls to the internal encoder
- * and decoder functions.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_
-
-#include "structs.h"
-
-int WebRtcIsac_EstimateBandwidth(
- BwEstimatorstr* bwest_str,
- Bitstr* streamdata,
- WebRtc_Word32 packet_size,
- WebRtc_UWord16 rtp_seq_number,
- WebRtc_UWord32 send_ts,
- WebRtc_UWord32 arr_ts,
- enum IsacSamplingRate encoderSampRate,
- enum IsacSamplingRate decoderSampRate);
-
-int WebRtcIsac_DecodeLb(
- float* signal_out,
- ISACLBDecStruct* ISACdec_obj,
- WebRtc_Word16* current_framesamples,
- WebRtc_Word16 isRCUPayload);
-
-int WebRtcIsac_DecodeRcuLb(
- float* signal_out,
- ISACLBDecStruct* ISACdec_obj,
- WebRtc_Word16* current_framesamples);
-
-int WebRtcIsac_EncodeLb(
- float* in,
- ISACLBEncStruct* ISACencLB_obj,
- WebRtc_Word16 codingMode,
- WebRtc_Word16 bottleneckIndex);
-
-int WebRtcIsac_EncodeStoredDataLb(
- const ISAC_SaveEncData_t* ISACSavedEnc_obj,
- Bitstr* ISACBitStr_obj,
- int BWnumber,
- float scale);
-
-
-int WebRtcIsac_EncodeStoredDataUb12(
- const ISACUBSaveEncDataStruct* ISACSavedEnc_obj,
- Bitstr* bitStream,
- WebRtc_Word32 jitterInfo,
- float scale);
-
-int WebRtcIsac_EncodeStoredDataUb16(
- const ISACUBSaveEncDataStruct* ISACSavedEnc_obj,
- Bitstr* bitStream,
- WebRtc_Word32 jitterInfo,
- float scale);
-
-
-WebRtc_Word16 WebRtcIsac_GetRedPayloadUb(
- const ISACUBSaveEncDataStruct* ISACSavedEncObj,
- Bitstr* bitStreamObj,
- enum ISACBandwidth bandwidth);
-/******************************************************************************
- * WebRtcIsac_RateAllocation()
- * Internal function to perform a rate-allocation for upper and lower-band,
- * given a total rate.
- *
- * Input:
- * - inRateBitPerSec : a total bit-rate in bits/sec.
- *
- * Output:
- * - rateLBBitPerSec : a bit-rate allocated to the lower-band
- * in bits/sec.
- * - rateUBBitPerSec : a bit-rate allocated to the upper-band
- * in bits/sec.
- *
- * Return value : 0 if rate allocation has been successful.
- * -1 if failed to allocate rates.
- */
-
-WebRtc_Word16
-WebRtcIsac_RateAllocation(
- WebRtc_Word32 inRateBitPerSec,
- double* rateLBBitPerSec,
- double* rateUBBitPerSec,
- enum ISACBandwidth* bandwidthKHz);
-
-
-/******************************************************************************
- * WebRtcIsac_DecodeUb16()
- *
- * Decode the upper-band if the codec is in 0-16 kHz mode.
- *
- * Input/Output:
- * -ISACdec_obj : pointer to the upper-band decoder object. The
- * bit-stream is stored inside the decoder object.
- *
- * Output:
- * -signal_out : decoded audio, 480 samples 30 ms.
- *
- * Return value : >0 number of decoded bytes.
- * <0 if an error occurred.
- */
-int WebRtcIsac_DecodeUb16(
- float* signal_out,
- ISACUBDecStruct* ISACdec_obj,
- WebRtc_Word16 isRCUPayload);
-
-
-/******************************************************************************
- * WebRtcIsac_DecodeUb12()
- *
- * Decode the upper-band if the codec is in 0-12 kHz mode.
- *
- * Input/Output:
- * -ISACdec_obj : pointer to the upper-band decoder object. The
- * bit-stream is stored inside the decoder object.
- *
- * Output:
- * -signal_out : decoded audio, 480 samples 30 ms.
- *
- * Return value : >0 number of decoded bytes.
- * <0 if an error occurred.
- */
-int WebRtcIsac_DecodeUb12(
- float* signal_out,
- ISACUBDecStruct* ISACdec_obj,
- WebRtc_Word16 isRCUPayload);
-
-
-/******************************************************************************
- * WebRtcIsac_EncodeUb16()
- *
- * Encode the upper-band if the codec is in 0-16 kHz mode.
- *
- * Input:
- * -in : upper-band audio, 160 samples (10 ms).
- *
- * Input/Output:
- * -ISACdec_obj : pointer to the upper-band encoder object. The
- * bit-stream is stored inside the encoder object.
- *
- * Return value : >0 number of encoded bytes.
- * <0 if an error occurred.
- */
-int WebRtcIsac_EncodeUb16(
- float* in,
- ISACUBEncStruct* ISACenc_obj,
- WebRtc_Word32 jitterInfo);
-
-
-/******************************************************************************
- * WebRtcIsac_EncodeUb12()
- *
- * Encode the upper-band if the codec is in 0-12 kHz mode.
- *
- * Input:
- * -in : upper-band audio, 160 samples (10 ms).
- *
- * Input/Output:
- * -ISACdec_obj : pointer to the upper-band encoder object. The
- * bit-stream is stored inside the encoder object.
- *
- * Return value : >0 number of encoded bytes.
- * <0 if an error occurred.
- */
-int WebRtcIsac_EncodeUb12(
- float* in,
- ISACUBEncStruct* ISACenc_obj,
- WebRtc_Word32 jitterInfo);
-
-/************************** initialization functions *************************/
-
-void WebRtcIsac_InitMasking(MaskFiltstr *maskdata);
-
-void WebRtcIsac_InitPreFilterbank(PreFiltBankstr *prefiltdata);
-
-void WebRtcIsac_InitPostFilterbank(PostFiltBankstr *postfiltdata);
-
-void WebRtcIsac_InitPitchFilter(PitchFiltstr *pitchfiltdata);
-
-void WebRtcIsac_InitPitchAnalysis(PitchAnalysisStruct *State);
-
-
-/**************************** transform functions ****************************/
-
-void WebRtcIsac_InitTransform();
-
-void WebRtcIsac_Time2Spec(double *inre1,
- double *inre2,
- WebRtc_Word16 *outre,
- WebRtc_Word16 *outim,
- FFTstr *fftstr_obj);
-
-void WebRtcIsac_Spec2time(double *inre,
- double *inim,
- double *outre1,
- double *outre2,
- FFTstr *fftstr_obj);
-
-
-/******************************* filter functions ****************************/
-
-void WebRtcIsac_AllPoleFilter(double *InOut,
- double *Coef,
- int lengthInOut,
- int orderCoef);
-
-void WebRtcIsac_AllZeroFilter(double *In,
- double *Coef,
- int lengthInOut,
- int orderCoef,
- double *Out);
-
-void WebRtcIsac_ZeroPoleFilter(double *In,
- double *ZeroCoef,
- double *PoleCoef,
- int lengthInOut,
- int orderCoef,
- double *Out);
-
-
-/***************************** filterbank functions **************************/
-
-void WebRtcIsac_SplitAndFilter(double *in,
- double *LP,
- double *HP,
- double *LP_la,
- double *HP_la,
- PreFiltBankstr *prefiltdata);
-
-
-void WebRtcIsac_FilterAndCombine(double *InLP,
- double *InHP,
- double *Out,
- PostFiltBankstr *postfiltdata);
-
-
-
-void WebRtcIsac_SplitAndFilterFloat(float *in,
- float *LP,
- float *HP,
- double *LP_la,
- double *HP_la,
- PreFiltBankstr *prefiltdata);
-
-
-void WebRtcIsac_FilterAndCombineFloat(float *InLP,
- float *InHP,
- float *Out,
- PostFiltBankstr *postfiltdata);
-
-
-/************************* normalized lattice filters ************************/
-
-void WebRtcIsac_NormLatticeFilterMa(int orderCoef,
- float *stateF,
- float *stateG,
- float *lat_in,
- double *filtcoeflo,
- double *lat_out);
-
-void WebRtcIsac_NormLatticeFilterAr(int orderCoef,
- float *stateF,
- float *stateG,
- double *lat_in,
- double *lo_filt_coef,
- float *lat_out);
-
-void WebRtcIsac_Dir2Lat(double *a,
- int orderCoef,
- float *sth,
- float *cth);
-
-void WebRtcIsac_AutoCorr(double *r,
- const double *x,
- int N,
- int order);
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CODEC_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/crc.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/crc.c
deleted file mode 100644
index 098e4b7..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/crc.c
+++ /dev/null
@@ -1,110 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "crc.h"
-#include <stdlib.h>
-#include "signal_processing_library.h"
-
-#define POLYNOMIAL 0x04c11db7L
-
-
-static const WebRtc_UWord32 kCrcTable[256] = {
- 0, 0x4c11db7, 0x9823b6e, 0xd4326d9, 0x130476dc, 0x17c56b6b,
- 0x1a864db2, 0x1e475005, 0x2608edb8, 0x22c9f00f, 0x2f8ad6d6, 0x2b4bcb61,
- 0x350c9b64, 0x31cd86d3, 0x3c8ea00a, 0x384fbdbd, 0x4c11db70, 0x48d0c6c7,
- 0x4593e01e, 0x4152fda9, 0x5f15adac, 0x5bd4b01b, 0x569796c2, 0x52568b75,
- 0x6a1936c8, 0x6ed82b7f, 0x639b0da6, 0x675a1011, 0x791d4014, 0x7ddc5da3,
- 0x709f7b7a, 0x745e66cd, 0x9823b6e0, 0x9ce2ab57, 0x91a18d8e, 0x95609039,
- 0x8b27c03c, 0x8fe6dd8b, 0x82a5fb52, 0x8664e6e5, 0xbe2b5b58, 0xbaea46ef,
- 0xb7a96036, 0xb3687d81, 0xad2f2d84, 0xa9ee3033, 0xa4ad16ea, 0xa06c0b5d,
- 0xd4326d90, 0xd0f37027, 0xddb056fe, 0xd9714b49, 0xc7361b4c, 0xc3f706fb,
- 0xceb42022, 0xca753d95, 0xf23a8028, 0xf6fb9d9f, 0xfbb8bb46, 0xff79a6f1,
- 0xe13ef6f4, 0xe5ffeb43, 0xe8bccd9a, 0xec7dd02d, 0x34867077, 0x30476dc0,
- 0x3d044b19, 0x39c556ae, 0x278206ab, 0x23431b1c, 0x2e003dc5, 0x2ac12072,
- 0x128e9dcf, 0x164f8078, 0x1b0ca6a1, 0x1fcdbb16, 0x18aeb13, 0x54bf6a4,
- 0x808d07d, 0xcc9cdca, 0x7897ab07, 0x7c56b6b0, 0x71159069, 0x75d48dde,
- 0x6b93dddb, 0x6f52c06c, 0x6211e6b5, 0x66d0fb02, 0x5e9f46bf, 0x5a5e5b08,
- 0x571d7dd1, 0x53dc6066, 0x4d9b3063, 0x495a2dd4, 0x44190b0d, 0x40d816ba,
- 0xaca5c697, 0xa864db20, 0xa527fdf9, 0xa1e6e04e, 0xbfa1b04b, 0xbb60adfc,
- 0xb6238b25, 0xb2e29692, 0x8aad2b2f, 0x8e6c3698, 0x832f1041, 0x87ee0df6,
- 0x99a95df3, 0x9d684044, 0x902b669d, 0x94ea7b2a, 0xe0b41de7, 0xe4750050,
- 0xe9362689, 0xedf73b3e, 0xf3b06b3b, 0xf771768c, 0xfa325055, 0xfef34de2,
- 0xc6bcf05f, 0xc27dede8, 0xcf3ecb31, 0xcbffd686, 0xd5b88683, 0xd1799b34,
- 0xdc3abded, 0xd8fba05a, 0x690ce0ee, 0x6dcdfd59, 0x608edb80, 0x644fc637,
- 0x7a089632, 0x7ec98b85, 0x738aad5c, 0x774bb0eb, 0x4f040d56, 0x4bc510e1,
- 0x46863638, 0x42472b8f, 0x5c007b8a, 0x58c1663d, 0x558240e4, 0x51435d53,
- 0x251d3b9e, 0x21dc2629, 0x2c9f00f0, 0x285e1d47, 0x36194d42, 0x32d850f5,
- 0x3f9b762c, 0x3b5a6b9b, 0x315d626, 0x7d4cb91, 0xa97ed48, 0xe56f0ff,
- 0x1011a0fa, 0x14d0bd4d, 0x19939b94, 0x1d528623, 0xf12f560e, 0xf5ee4bb9,
- 0xf8ad6d60, 0xfc6c70d7, 0xe22b20d2, 0xe6ea3d65, 0xeba91bbc, 0xef68060b,
- 0xd727bbb6, 0xd3e6a601, 0xdea580d8, 0xda649d6f, 0xc423cd6a, 0xc0e2d0dd,
- 0xcda1f604, 0xc960ebb3, 0xbd3e8d7e, 0xb9ff90c9, 0xb4bcb610, 0xb07daba7,
- 0xae3afba2, 0xaafbe615, 0xa7b8c0cc, 0xa379dd7b, 0x9b3660c6, 0x9ff77d71,
- 0x92b45ba8, 0x9675461f, 0x8832161a, 0x8cf30bad, 0x81b02d74, 0x857130c3,
- 0x5d8a9099, 0x594b8d2e, 0x5408abf7, 0x50c9b640, 0x4e8ee645, 0x4a4ffbf2,
- 0x470cdd2b, 0x43cdc09c, 0x7b827d21, 0x7f436096, 0x7200464f, 0x76c15bf8,
- 0x68860bfd, 0x6c47164a, 0x61043093, 0x65c52d24, 0x119b4be9, 0x155a565e,
- 0x18197087, 0x1cd86d30, 0x29f3d35, 0x65e2082, 0xb1d065b, 0xfdc1bec,
- 0x3793a651, 0x3352bbe6, 0x3e119d3f, 0x3ad08088, 0x2497d08d, 0x2056cd3a,
- 0x2d15ebe3, 0x29d4f654, 0xc5a92679, 0xc1683bce, 0xcc2b1d17, 0xc8ea00a0,
- 0xd6ad50a5, 0xd26c4d12, 0xdf2f6bcb, 0xdbee767c, 0xe3a1cbc1, 0xe760d676,
- 0xea23f0af, 0xeee2ed18, 0xf0a5bd1d, 0xf464a0aa, 0xf9278673, 0xfde69bc4,
- 0x89b8fd09, 0x8d79e0be, 0x803ac667, 0x84fbdbd0, 0x9abc8bd5, 0x9e7d9662,
- 0x933eb0bb, 0x97ffad0c, 0xafb010b1, 0xab710d06, 0xa6322bdf, 0xa2f33668,
- 0xbcb4666d, 0xb8757bda, 0xb5365d03, 0xb1f740b4
-};
-
-
-
-
-/****************************************************************************
- * WebRtcIsac_GetCrc(...)
- *
- * This function returns a 32 bit CRC checksum of a bit stream
- *
- * Input:
- * - bitstream : payload bitstream
- * - len_bitstream_in_bytes : number of 8-bit words in the bit stream
- *
- * Output:
- * - crc : checksum
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcIsac_GetCrc(const WebRtc_Word16* bitstream,
- WebRtc_Word16 len_bitstream_in_bytes,
- WebRtc_UWord32* crc)
-{
- WebRtc_UWord8* bitstream_ptr_uw8;
- WebRtc_UWord32 crc_state;
- int byte_cntr;
- int crc_tbl_indx;
-
- /* Sanity Check. */
- if (bitstream == NULL) {
- return -1;
- }
- /* cast to UWord8 pointer */
- bitstream_ptr_uw8 = (WebRtc_UWord8 *)bitstream;
-
- /* initialize */
- crc_state = 0xFFFFFFFF;
-
- for (byte_cntr = 0; byte_cntr < len_bitstream_in_bytes; byte_cntr++) {
- crc_tbl_indx = (WEBRTC_SPL_RSHIFT_U32(crc_state, 24) ^
- bitstream_ptr_uw8[byte_cntr]) & 0xFF;
- crc_state = WEBRTC_SPL_LSHIFT_U32(crc_state, 8) ^ kCrcTable[crc_tbl_indx];
- }
-
- *crc = ~crc_state;
- return 0;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/crc.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/crc.h
deleted file mode 100644
index 0151278..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/crc.h
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * crc.h
- *
- * Checksum functions
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_
-
-#include "typedefs.h"
-
-/****************************************************************************
- * WebRtcIsac_GetCrc(...)
- *
- * This function returns a 32 bit CRC checksum of a bit stream
- *
- * Input:
- * - encoded : payload bit stream
- * - no_of_word8s : number of 8-bit words in the bit stream
- *
- * Output:
- * - crc : checksum
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcIsac_GetCrc(
- const WebRtc_Word16* encoded,
- WebRtc_Word16 no_of_word8s,
- WebRtc_UWord32* crc);
-
-
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_CRC_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/decode.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/decode.c
deleted file mode 100644
index 25634b0..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/decode.c
+++ /dev/null
@@ -1,330 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * decode_B.c
- *
- * This file contains definition of funtions for decoding.
- * Decoding of lower-band, including normal-decoding and RCU decoding.
- * Decoding of upper-band, including 8-12 kHz, when the bandwidth is
- * 0-12 kHz, and 8-16 kHz, when the bandwidth is 0-16 kHz.
- *
- */
-
-
-#include "codec.h"
-#include "entropy_coding.h"
-#include "pitch_estimator.h"
-#include "bandwidth_estimator.h"
-#include "structs.h"
-#include "settings.h"
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-
-/*
- * function to decode the bitstream
- * returns the total number of bytes in the stream
- */
-int
-WebRtcIsac_DecodeLb(
- float* signal_out,
- ISACLBDecStruct* ISACdecLB_obj,
- WebRtc_Word16* current_framesamples,
- WebRtc_Word16 isRCUPayload)
-{
- int k, model;
- int len, err;
- WebRtc_Word16 bandwidthInd;
-
- float LP_dec_float[FRAMESAMPLES_HALF];
- float HP_dec_float[FRAMESAMPLES_HALF];
-
- double LPw[FRAMESAMPLES_HALF];
- double HPw[FRAMESAMPLES_HALF];
- double LPw_pf[FRAMESAMPLES_HALF];
-
- double lo_filt_coef[(ORDERLO+1)*SUBFRAMES];
- double hi_filt_coef[(ORDERHI+1)*SUBFRAMES];
-
- double real_f[FRAMESAMPLES_HALF];
- double imag_f[FRAMESAMPLES_HALF];
-
- double PitchLags[4];
- double PitchGains[4];
- double AvgPitchGain;
- WebRtc_Word16 PitchGains_Q12[4];
- WebRtc_Word16 AvgPitchGain_Q12;
-
- float gain;
-
- int frame_nb; /* counter */
- int frame_mode; /* 0 for 20ms and 30ms, 1 for 60ms */
- int processed_samples;
-
- (ISACdecLB_obj->bitstr_obj).W_upper = 0xFFFFFFFF;
- (ISACdecLB_obj->bitstr_obj).streamval = 0;
- (ISACdecLB_obj->bitstr_obj).stream_index = 0;
-
- len = 0;
-
- /* decode framelength and BW estimation - not used,
- only for stream pointer*/
- err = WebRtcIsac_DecodeFrameLen(&ISACdecLB_obj->bitstr_obj,
- current_framesamples);
- if (err < 0) { // error check
- return err;
- }
-
- /* frame_mode: 0, or 1 */
- frame_mode = *current_framesamples/MAX_FRAMESAMPLES;
- /* processed_samples: either 320 (20ms) or 480 (30, 60 ms) */
- processed_samples = *current_framesamples/(frame_mode+1);
-
- err = WebRtcIsac_DecodeSendBW(&ISACdecLB_obj->bitstr_obj, &bandwidthInd);
- if (err < 0) { // error check
- return err;
- }
-
- /* one loop if it's one frame (20 or 30ms), 2 loops if 2 frames
- bundled together (60ms) */
- for (frame_nb = 0; frame_nb <= frame_mode; frame_nb++) {
- /* decode & dequantize pitch parameters */
- err = WebRtcIsac_DecodePitchGain(&(ISACdecLB_obj->bitstr_obj),
- PitchGains_Q12);
- if (err < 0) { // error check
- return err;
- }
-
- err = WebRtcIsac_DecodePitchLag(&ISACdecLB_obj->bitstr_obj,
- PitchGains_Q12, PitchLags);
- if (err < 0) { // error check
- return err;
- }
-
- AvgPitchGain_Q12 = (PitchGains_Q12[0] + PitchGains_Q12[1] +
- PitchGains_Q12[2] + PitchGains_Q12[3])>>2;
-
- /* decode & dequantize FiltCoef */
- err = WebRtcIsac_DecodeLpc(&ISACdecLB_obj->bitstr_obj,
- lo_filt_coef,hi_filt_coef, &model);
- if (err < 0) { // error check
- return err;
- }
- /* decode & dequantize spectrum */
- len = WebRtcIsac_DecodeSpecLb(&ISACdecLB_obj->bitstr_obj,
- real_f, imag_f, AvgPitchGain_Q12);
- if (len < 0) { // error check
- return len;
- }
-
- /* inverse transform */
- WebRtcIsac_Spec2time(real_f, imag_f, LPw, HPw,
- &ISACdecLB_obj->fftstr_obj);
-
- /* convert PitchGains back to FLOAT for pitchfilter_post */
- for (k = 0; k < 4; k++) {
- PitchGains[k] = ((float)PitchGains_Q12[k])/4096;
- }
-
- if(isRCUPayload)
- {
- for (k = 0; k < 240; k++) {
- LPw[k] *= RCU_TRANSCODING_SCALE_INVERSE;
- HPw[k] *= RCU_TRANSCODING_SCALE_INVERSE;
- }
- }
-
- /* inverse pitch filter */
- WebRtcIsac_PitchfilterPost(LPw, LPw_pf,
- &ISACdecLB_obj->pitchfiltstr_obj, PitchLags, PitchGains);
- /* convert AvgPitchGain back to FLOAT for computation of gain */
- AvgPitchGain = ((float)AvgPitchGain_Q12)/4096;
- gain = 1.0f - 0.45f * (float)AvgPitchGain;
-
- for (k = 0; k < FRAMESAMPLES_HALF; k++) {
- /* reduce gain to compensate for pitch enhancer */
- LPw_pf[ k ] *= gain;
- }
-
- if(isRCUPayload)
- {
- for (k = 0; k < FRAMESAMPLES_HALF; k++) {
- /* compensation for transcoding gain changes*/
- LPw_pf[k] *= RCU_TRANSCODING_SCALE;
- HPw[k] *= RCU_TRANSCODING_SCALE;
- }
- }
-
- /* perceptual post-filtering (using normalized lattice filter) */
- WebRtcIsac_NormLatticeFilterAr(ORDERLO,
- ISACdecLB_obj->maskfiltstr_obj.PostStateLoF,
- (ISACdecLB_obj->maskfiltstr_obj).PostStateLoG,
- LPw_pf, lo_filt_coef, LP_dec_float);
- WebRtcIsac_NormLatticeFilterAr(ORDERHI,
- ISACdecLB_obj->maskfiltstr_obj.PostStateHiF,
- (ISACdecLB_obj->maskfiltstr_obj).PostStateHiG,
- HPw, hi_filt_coef, HP_dec_float);
-
- /* recombine the 2 bands */
- WebRtcIsac_FilterAndCombineFloat( LP_dec_float, HP_dec_float,
- signal_out + frame_nb * processed_samples,
- &ISACdecLB_obj->postfiltbankstr_obj);
- }
-
- return len;
-}
-
-
-/*
- * This decode function is called when the codec is operating in 16 kHz
- * bandwidth to decode the upperband, i.e. 8-16 kHz.
- *
- * Contrary to lower-band, the upper-band (8-16 kHz) is not split in
- * frequency, but split to 12 sub-frames, i.e. twice as lower-band.
- */
-int
-WebRtcIsac_DecodeUb16(
- float* signal_out,
- ISACUBDecStruct* ISACdecUB_obj,
- WebRtc_Word16 isRCUPayload)
-{
- int len, err;
-
- double halfFrameFirst[FRAMESAMPLES_HALF];
- double halfFrameSecond[FRAMESAMPLES_HALF];
-
- double percepFilterParam[(UB_LPC_ORDER+1) * (SUBFRAMES<<1) +
- (UB_LPC_ORDER+1)];
-
- double real_f[FRAMESAMPLES_HALF];
- double imag_f[FRAMESAMPLES_HALF];
-
- len = 0;
-
- /* decode & dequantize FiltCoef */
- memset(percepFilterParam, 0, sizeof(percepFilterParam));
- err = WebRtcIsac_DecodeInterpolLpcUb(&ISACdecUB_obj->bitstr_obj,
- percepFilterParam, isac16kHz);
- if (err < 0) { // error check
- return err;
- }
-
- /* decode & dequantize spectrum */
- len = WebRtcIsac_DecodeSpecUB16(&ISACdecUB_obj->bitstr_obj, real_f,
- imag_f);
- if (len < 0) { // error check
- return len;
- }
- if(isRCUPayload)
- {
- int n;
- for(n = 0; n < 240; n++)
- {
- real_f[n] *= RCU_TRANSCODING_SCALE_UB_INVERSE;
- imag_f[n] *= RCU_TRANSCODING_SCALE_UB_INVERSE;
- }
- }
-
- /* inverse transform */
- WebRtcIsac_Spec2time(real_f, imag_f, halfFrameFirst, halfFrameSecond,
- &ISACdecUB_obj->fftstr_obj);
-
- /* perceptual post-filtering (using normalized lattice filter) */
- WebRtcIsac_NormLatticeFilterAr(UB_LPC_ORDER,
- ISACdecUB_obj->maskfiltstr_obj.PostStateLoF,
- (ISACdecUB_obj->maskfiltstr_obj).PostStateLoG, halfFrameFirst,
- &percepFilterParam[(UB_LPC_ORDER+1)], signal_out);
-
- WebRtcIsac_NormLatticeFilterAr(UB_LPC_ORDER,
- ISACdecUB_obj->maskfiltstr_obj.PostStateLoF,
- (ISACdecUB_obj->maskfiltstr_obj).PostStateLoG, halfFrameSecond,
- &percepFilterParam[(UB_LPC_ORDER + 1) * SUBFRAMES + (UB_LPC_ORDER+1)],
- &signal_out[FRAMESAMPLES_HALF]);
-
- return len;
-}
-
-/*
- * This decode function is called when the codec operates at 0-12 kHz
- * bandwidth to decode the upperband, i.e. 8-12 kHz.
- *
- * At the encoder the upper-band is split into two band, 8-12 kHz & 12-16
- * kHz, and only 8-12 kHz is encoded. At the decoder, 8-12 kHz band is
- * reconstructed and 12-16 kHz replaced with zeros. Then two bands
- * are combined, to reconstruct the upperband 8-16 kHz.
- */
-int
-WebRtcIsac_DecodeUb12(
- float* signal_out,
- ISACUBDecStruct* ISACdecUB_obj,
- WebRtc_Word16 isRCUPayload)
-{
- int len, err;
-
- float LP_dec_float[FRAMESAMPLES_HALF];
- float HP_dec_float[FRAMESAMPLES_HALF];
-
- double LPw[FRAMESAMPLES_HALF];
- double HPw[FRAMESAMPLES_HALF];
-
- double percepFilterParam[(UB_LPC_ORDER+1)*SUBFRAMES];
-
- double real_f[FRAMESAMPLES_HALF];
- double imag_f[FRAMESAMPLES_HALF];
-
- len = 0;
-
- /* decode & dequantize FiltCoef */
- err = WebRtcIsac_DecodeInterpolLpcUb(&ISACdecUB_obj->bitstr_obj,
- percepFilterParam, isac12kHz);
- if(err < 0) { // error check
- return err;
- }
-
- /* decode & dequantize spectrum */
- len = WebRtcIsac_DecodeSpecUB12(&ISACdecUB_obj->bitstr_obj,
- real_f, imag_f);
- if(len < 0) { // error check
- return len;
- }
-
- if(isRCUPayload)
- {
- int n;
- for(n = 0; n < 240; n++)
- {
- real_f[n] *= RCU_TRANSCODING_SCALE_UB_INVERSE;
- imag_f[n] *= RCU_TRANSCODING_SCALE_UB_INVERSE;
- }
- }
-
- /* inverse transform */
- WebRtcIsac_Spec2time(real_f, imag_f, LPw, HPw, &ISACdecUB_obj->fftstr_obj);
-
- /* perceptual post-filtering (using normalized lattice filter) */
- WebRtcIsac_NormLatticeFilterAr(UB_LPC_ORDER,
- ISACdecUB_obj->maskfiltstr_obj.PostStateLoF,
- (ISACdecUB_obj->maskfiltstr_obj).PostStateLoG, LPw,
- percepFilterParam, LP_dec_float);
-
- /* Zerro for upper-band */
- memset(HP_dec_float, 0, sizeof(float) * (FRAMESAMPLES_HALF));
-
- /* recombine the 2 bands */
- WebRtcIsac_FilterAndCombineFloat(HP_dec_float, LP_dec_float, signal_out,
- &ISACdecUB_obj->postfiltbankstr_obj);
-
-
-
- return len;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/decode_bwe.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/decode_bwe.c
deleted file mode 100644
index cdac7fa..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/decode_bwe.c
+++ /dev/null
@@ -1,88 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "structs.h"
-#include "bandwidth_estimator.h"
-#include "entropy_coding.h"
-#include "codec.h"
-
-
-int
-WebRtcIsac_EstimateBandwidth(
- BwEstimatorstr* bwest_str,
- Bitstr* streamdata,
- WebRtc_Word32 packet_size,
- WebRtc_UWord16 rtp_seq_number,
- WebRtc_UWord32 send_ts,
- WebRtc_UWord32 arr_ts,
- enum IsacSamplingRate encoderSampRate,
- enum IsacSamplingRate decoderSampRate)
-{
- WebRtc_Word16 index;
- WebRtc_Word16 frame_samples;
- WebRtc_UWord32 sendTimestampIn16kHz;
- WebRtc_UWord32 arrivalTimestampIn16kHz;
- WebRtc_UWord32 diffSendTime;
- WebRtc_UWord32 diffArrivalTime;
- int err;
-
- /* decode framelength and BW estimation */
- err = WebRtcIsac_DecodeFrameLen(streamdata, &frame_samples);
- if(err < 0) // error check
- {
- return err;
- }
- err = WebRtcIsac_DecodeSendBW(streamdata, &index);
- if(err < 0) // error check
- {
- return err;
- }
-
- /* UPDATE ESTIMATES FROM OTHER SIDE */
- err = WebRtcIsac_UpdateUplinkBwImpl(bwest_str, index, encoderSampRate);
- if(err < 0)
- {
- return err;
- }
-
- // We like BWE to work at 16 kHz sampling rate,
- // therefore, we have to change the timestamps accordingly.
- // translate the send timestamp if required
- diffSendTime = (WebRtc_UWord32)((WebRtc_UWord32)send_ts -
- (WebRtc_UWord32)bwest_str->senderTimestamp);
- bwest_str->senderTimestamp = send_ts;
-
- diffArrivalTime = (WebRtc_UWord32)((WebRtc_UWord32)arr_ts -
- (WebRtc_UWord32)bwest_str->receiverTimestamp);
- bwest_str->receiverTimestamp = arr_ts;
-
- if(decoderSampRate == kIsacSuperWideband)
- {
- diffArrivalTime = (WebRtc_UWord32)diffArrivalTime >> 1;
- diffSendTime = (WebRtc_UWord32)diffSendTime >> 1;
- }
- // arrival timestamp in 16 kHz
- arrivalTimestampIn16kHz = (WebRtc_UWord32)((WebRtc_UWord32)
- bwest_str->prev_rec_arr_ts + (WebRtc_UWord32)diffArrivalTime);
- // send timestamp in 16 kHz
- sendTimestampIn16kHz = (WebRtc_UWord32)((WebRtc_UWord32)
- bwest_str->prev_rec_send_ts + (WebRtc_UWord32)diffSendTime);
-
- err = WebRtcIsac_UpdateBandwidthEstimator(bwest_str, rtp_seq_number,
- (frame_samples * 1000) / FS, sendTimestampIn16kHz,
- arrivalTimestampIn16kHz, packet_size);
- // error check
- if(err < 0)
- {
- return err;
- }
-
- return 0;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode.c
deleted file mode 100644
index 75cd726..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode.c
+++ /dev/null
@@ -1,1451 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * encode.c
- *
- * This file contains definition of funtions for encoding.
- * Decoding of upper-band, including 8-12 kHz, when the bandwidth is
- * 0-12 kHz, and 8-16 kHz, when the bandwidth is 0-16 kHz.
- *
- */
-
-#include <stdlib.h>
-#include <string.h>
-
-#include "structs.h"
-#include "codec.h"
-#include "pitch_estimator.h"
-#include "entropy_coding.h"
-#include "arith_routines.h"
-#include "pitch_gain_tables.h"
-#include "pitch_lag_tables.h"
-#include "spectrum_ar_model_tables.h"
-#include "lpc_tables.h"
-#include "lpc_analysis.h"
-#include "bandwidth_estimator.h"
-#include "lpc_shape_swb12_tables.h"
-#include "lpc_shape_swb16_tables.h"
-#include "lpc_gain_swb_tables.h"
-
-
-#define UB_LOOKAHEAD 24
-
-/*
- Rate allocation tables of lower and upper-band bottleneck for
- 12kHz & 16kHz bandwidth.
-
- 12 kHz bandwidth
- -----------------
- The overall bottleneck of the coder is between 38 kbps and 45 kbps. We have
- considered 7 enteries, uniformly distributed in this interval, i.e. 38,
- 39.17, 40.33, 41.5, 42.67, 43.83 and 45. For every entery, the lower-band
- and the upper-band bottlenecks are specified in
- 'kLowerBandBitRate12' and 'kUpperBandBitRate12'
- tables, respectively. E.g. the overall rate of 41.5 kbps corresponts to a
- bottleneck of 31 kbps for lower-band and 27 kbps for upper-band. Given an
- overall bottleneck of the codec, we use linear interpolation to get
- lower-band and upper-band bottlenecks.
-
- 16 kHz bandwidth
- -----------------
- The overall bottleneck of the coder is between 50 kbps and 56 kbps. We have
- considered 7 enteries, uniformly distributed in this interval, i.e. 50, 51.2,
- 52.4, 53.6, 54.8 and 56. For every entery, the lower-band and the upper-band
- bottlenecks are specified in 'kLowerBandBitRate16' and
- 'kUpperBandBitRate16' tables, respectively. E.g. the overall rate
- of 53.6 kbps corresponts to a bottleneck of 32 kbps for lower-band and 30
- kbps for upper-band. Given an overall bottleneck of the codec, we use linear
- interpolation to get lower-band and upper-band bottlenecks.
-
-*/
-
-// 38 39.17 40.33 41.5 42.67 43.83 45
-static const WebRtc_Word16 kLowerBandBitRate12[7] = {
- 29000, 30000, 30000, 31000, 31000, 32000, 32000};
-static const WebRtc_Word16 kUpperBandBitRate12[7] = {
- 25000, 25000, 27000, 27000, 29000, 29000, 32000};
-
-// 50 51.2 52.4 53.6 54.8 56
-static const WebRtc_Word16 kLowerBandBitRate16[6] = {
- 31000, 31000, 32000, 32000, 32000, 32000};
-static const WebRtc_Word16 kUpperBandBitRate16[6] = {
- 28000, 29000, 29000, 30000, 31000, 32000};
-
-/******************************************************************************
- * WebRtcIsac_RateAllocation()
- * Internal function to perform a rate-allocation for upper and lower-band,
- * given a total rate.
- *
- * Input:
- * - inRateBitPerSec : a total bottleneck in bits/sec.
- *
- * Output:
- * - rateLBBitPerSec : a bottleneck allocated to the lower-band
- * in bits/sec.
- * - rateUBBitPerSec : a bottleneck allocated to the upper-band
- * in bits/sec.
- *
- * Return value : 0 if rate allocation has been successful.
- * -1 if failed to allocate rates.
- */
-
-WebRtc_Word16
-WebRtcIsac_RateAllocation(
- WebRtc_Word32 inRateBitPerSec,
- double* rateLBBitPerSec,
- double* rateUBBitPerSec,
- enum ISACBandwidth* bandwidthKHz)
-{
- WebRtc_Word16 idx;
- double idxD;
- double idxErr;
- if(inRateBitPerSec < 38000)
- {
- // If the given overall bottleneck is less than 38000 then
- // then codec has to operate in wideband mode, i.e. 8 kHz
- // bandwidth.
- *rateLBBitPerSec = (WebRtc_Word16)((inRateBitPerSec > 32000)?
- 32000:inRateBitPerSec);
- *rateUBBitPerSec = 0;
- *bandwidthKHz = isac8kHz;
- }
- else if((inRateBitPerSec >= 38000) && (inRateBitPerSec < 50000))
- {
- // At a bottleneck between 38 and 50 kbps the codec is operating
- // at 12 kHz bandwidth. Using xxxBandBitRate12[] to calculates
- // upper/lower bottleneck
-
- // find the bottlenecks by linear interpolation
- // step is (45000 - 38000)/6.0 we use the inverse of it.
- const double stepSizeInv = 8.5714286e-4;
- idxD = (inRateBitPerSec - 38000) * stepSizeInv;
- idx = (idxD >= 6)? 6:((WebRtc_Word16)idxD);
- idxErr = idxD - idx;
- *rateLBBitPerSec = kLowerBandBitRate12[idx];
- *rateUBBitPerSec = kUpperBandBitRate12[idx];
-
- if(idx < 6)
- {
- *rateLBBitPerSec += (WebRtc_Word16)(idxErr *
- (kLowerBandBitRate12[idx + 1] -
- kLowerBandBitRate12[idx]));
- *rateUBBitPerSec += (WebRtc_Word16)(idxErr *
- (kUpperBandBitRate12[idx + 1] -
- kUpperBandBitRate12[idx]));
- }
-
- *bandwidthKHz = isac12kHz;
- }
- else if((inRateBitPerSec >= 50000) && (inRateBitPerSec <= 56000))
- {
- // A bottleneck between 50 and 56 kbps corresponds to bandwidth
- // of 16 kHz. Using xxxBandBitRate16[] to calculates
- // upper/lower bottleneck
-
- // find the bottlenecks by linear interpolation
- // step is (56000 - 50000)/5 we use the inverse of it
- const double stepSizeInv = 8.3333333e-4;
- idxD = (inRateBitPerSec - 50000) * stepSizeInv;
- idx = (idxD >= 5)? 5:((WebRtc_Word16)idxD);
- idxErr = idxD - idx;
- *rateLBBitPerSec = kLowerBandBitRate16[idx];
- *rateUBBitPerSec = kUpperBandBitRate16[idx];
-
- if(idx < 5)
- {
- *rateLBBitPerSec += (WebRtc_Word16)(idxErr *
- (kLowerBandBitRate16[idx + 1] -
- kLowerBandBitRate16[idx]));
-
- *rateUBBitPerSec += (WebRtc_Word16)(idxErr *
- (kUpperBandBitRate16[idx + 1] -
- kUpperBandBitRate16[idx]));
- }
-
- *bandwidthKHz = isac16kHz;
- }
- else
- {
- // Out-of-range botlteneck value.
- return -1;
- }
-
- // limit the values.
- *rateLBBitPerSec = (*rateLBBitPerSec > 32000)? 32000:*rateLBBitPerSec;
- *rateUBBitPerSec = (*rateUBBitPerSec > 32000)? 32000:*rateUBBitPerSec;
-
- return 0;
-}
-
-
-
-int
-WebRtcIsac_EncodeLb(
- float* in,
- ISACLBEncStruct* ISACencLB_obj,
- WebRtc_Word16 codingMode,
- WebRtc_Word16 bottleneckIndex)
-{
- int stream_length = 0;
- int err;
- int k;
- int iterCntr;
-
- double lofilt_coef[(ORDERLO+1)*SUBFRAMES];
- double hifilt_coef[(ORDERHI+1)*SUBFRAMES];
- float LP[FRAMESAMPLES_HALF];
- float HP[FRAMESAMPLES_HALF];
-
- double LP_lookahead[FRAMESAMPLES_HALF];
- double HP_lookahead[FRAMESAMPLES_HALF];
- double LP_lookahead_pf[FRAMESAMPLES_HALF + QLOOKAHEAD];
- double LPw[FRAMESAMPLES_HALF];
-
- double HPw[FRAMESAMPLES_HALF];
- double LPw_pf[FRAMESAMPLES_HALF];
- WebRtc_Word16 fre[FRAMESAMPLES_HALF]; /* Q7 */
- WebRtc_Word16 fim[FRAMESAMPLES_HALF]; /* Q7 */
-
- double PitchLags[4];
- double PitchGains[4];
- WebRtc_Word16 PitchGains_Q12[4];
- WebRtc_Word16 AvgPitchGain_Q12;
-
- int frame_mode; /* 0 for 30ms, 1 for 60ms */
- int processed_samples, status = 0;
-
- double bits_gains;
- int bmodel;
-
- transcode_obj transcodingParam;
- double bytesLeftSpecCoding;
- WebRtc_UWord16 payloadLimitBytes;
-
- /* copy new frame length and bottle neck rate only for the first
- 10 ms data */
- if (ISACencLB_obj->buffer_index == 0) {
- /* set the framelength for the next packet */
- ISACencLB_obj->current_framesamples = ISACencLB_obj->new_framelength;
- }
- /* frame_mode is 0 (30 ms) or 1 (60 ms) */
- frame_mode = ISACencLB_obj->current_framesamples/MAX_FRAMESAMPLES;
- /* processed_samples: 480 (30, 60 ms) */
- processed_samples = ISACencLB_obj->current_framesamples/(frame_mode+1);
-
- /* buffer speech samples (by 10ms packet) until the framelength */
- /* is reached (30 or 60 ms) */
- /****************************************************************/
-
- /* fill the buffer with 10ms input data */
- for (k = 0; k < FRAMESAMPLES_10ms; k++) {
- ISACencLB_obj->data_buffer_float[k + ISACencLB_obj->buffer_index] =
- in[k];
- }
-
- /* if buffersize is not equal to current framesize then increase index
- and return. We do no encoding untill we have enough audio. */
- if (ISACencLB_obj->buffer_index + FRAMESAMPLES_10ms != processed_samples) {
- ISACencLB_obj->buffer_index += FRAMESAMPLES_10ms;
- return 0;
- }
- /* if buffer reached the right size, reset index and continue with
- encoding the frame */
- ISACencLB_obj->buffer_index = 0;
-
- /* end of buffer function */
- /**************************/
-
- /* encoding */
- /************/
-
- if (frame_mode == 0 || ISACencLB_obj->frame_nb == 0 ) {
- // This is to avoid Linux warnings until we change 'int' to 'Word32'
- // at all places.
- int intVar;
- /* reset bitstream */
- ISACencLB_obj->bitstr_obj.W_upper = 0xFFFFFFFF;
- ISACencLB_obj->bitstr_obj.streamval = 0;
- ISACencLB_obj->bitstr_obj.stream_index = 0;
-
- if((codingMode == 0) && (frame_mode == 0) &&
- (ISACencLB_obj->enforceFrameSize == 0)) {
- ISACencLB_obj->new_framelength =
- WebRtcIsac_GetNewFrameLength(ISACencLB_obj->bottleneck,
- ISACencLB_obj->current_framesamples);
- }
-
- ISACencLB_obj->s2nr = WebRtcIsac_GetSnr(
- ISACencLB_obj->bottleneck, ISACencLB_obj->current_framesamples);
-
- /* encode frame length */
- status = WebRtcIsac_EncodeFrameLen(
- ISACencLB_obj->current_framesamples, &ISACencLB_obj->bitstr_obj);
- if (status < 0) {
- /* Wrong frame size */
- return status;
- }
- /* Save framelength for multiple packets memory */
- ISACencLB_obj->SaveEnc_obj.framelength =
- ISACencLB_obj->current_framesamples;
-
- /* To be used for Redundant Coding */
- ISACencLB_obj->lastBWIdx = bottleneckIndex;
- intVar = (int)bottleneckIndex;
- WebRtcIsac_EncodeReceiveBw(&intVar, &ISACencLB_obj->bitstr_obj);
- }
-
- /* split signal in two bands */
- WebRtcIsac_SplitAndFilterFloat(ISACencLB_obj->data_buffer_float, LP, HP,
- LP_lookahead, HP_lookahead, &ISACencLB_obj->prefiltbankstr_obj );
-
- /* estimate pitch parameters and pitch-filter lookahead signal */
- WebRtcIsac_PitchAnalysis(LP_lookahead, LP_lookahead_pf,
- &ISACencLB_obj->pitchanalysisstr_obj, PitchLags, PitchGains);
-
- /* encode in FIX Q12 */
-
- /* convert PitchGain to Fixed point */
- for (k=0;k<PITCH_SUBFRAMES;k++) {
- PitchGains_Q12[k] = (WebRtc_Word16)(PitchGains[k] * 4096.0);
- }
-
- /* Set where to store data in multiple packets memory */
- if (frame_mode == 0 || ISACencLB_obj->frame_nb == 0)
- {
- ISACencLB_obj->SaveEnc_obj.startIdx = 0;
- } else {
- ISACencLB_obj->SaveEnc_obj.startIdx = 1;
- }
-
- /* quantize & encode pitch parameters */
- WebRtcIsac_EncodePitchGain(PitchGains_Q12, &ISACencLB_obj->bitstr_obj,
- &ISACencLB_obj->SaveEnc_obj);
- WebRtcIsac_EncodePitchLag(PitchLags, PitchGains_Q12,
- &ISACencLB_obj->bitstr_obj, &ISACencLB_obj->SaveEnc_obj);
-
- AvgPitchGain_Q12 = (PitchGains_Q12[0] + PitchGains_Q12[1] +
- PitchGains_Q12[2] + PitchGains_Q12[3])>>2;
-
- /* find coefficients for perceptual pre-filters */
- WebRtcIsac_GetLpcCoefLb(LP_lookahead_pf, HP_lookahead,
- &ISACencLB_obj->maskfiltstr_obj, ISACencLB_obj->s2nr,
- PitchGains_Q12, lofilt_coef, hifilt_coef);
-
- /* code LPC model and shape - gains not quantized yet */
- WebRtcIsac_EncodeLpcLb(lofilt_coef, hifilt_coef, &bmodel, &bits_gains,
- &ISACencLB_obj->bitstr_obj, &ISACencLB_obj->SaveEnc_obj);
-
- /* convert PitchGains back to FLOAT for pitchfilter_pre */
- for (k = 0; k < 4; k++) {
- PitchGains[k] = ((float)PitchGains_Q12[k])/4096;
- }
-
- /* Store the state of arithmetic coder before coding LPC gains */
- transcodingParam.W_upper = ISACencLB_obj->bitstr_obj.W_upper;
- transcodingParam.stream_index = ISACencLB_obj->bitstr_obj.stream_index;
- transcodingParam.streamval = ISACencLB_obj->bitstr_obj.streamval;
- transcodingParam.stream[0] = ISACencLB_obj->bitstr_obj.stream[
- ISACencLB_obj->bitstr_obj.stream_index - 2];
- transcodingParam.stream[1] = ISACencLB_obj->bitstr_obj.stream[
- ISACencLB_obj->bitstr_obj.stream_index - 1];
- transcodingParam.stream[2] = ISACencLB_obj->bitstr_obj.stream[
- ISACencLB_obj->bitstr_obj.stream_index];
-
- /* Store LPC Gains before encoding them */
- for(k = 0; k < SUBFRAMES; k++) {
- transcodingParam.loFiltGain[k] = lofilt_coef[(LPC_LOBAND_ORDER+1)*k];
- transcodingParam.hiFiltGain[k] = hifilt_coef[(LPC_HIBAND_ORDER+1)*k];
- }
-
- /* Code gains */
- WebRtcIsac_EncodeLpcGainLb(lofilt_coef, hifilt_coef, bmodel,
- &ISACencLB_obj->bitstr_obj, &ISACencLB_obj->SaveEnc_obj);
-
- /* Get the correct value for the payload limit and calculate the
- number of bytes left for coding the spectrum.*/
- if((frame_mode == 1) && (ISACencLB_obj->frame_nb == 0)) {
- /* It is a 60ms and we are in the first 30ms then the limit at
- this point should be half of the assigned value */
- payloadLimitBytes = ISACencLB_obj->payloadLimitBytes60 >> 1;
- }
- else if (frame_mode == 0) {
- /* It is a 30ms frame */
- /* Subract 3 because termination process may add 3 bytes */
- payloadLimitBytes = ISACencLB_obj->payloadLimitBytes30 - 3;
- } else {
- /* This is the second half of a 60ms frame. */
- /* Subract 3 because termination process may add 3 bytes */
- payloadLimitBytes = ISACencLB_obj->payloadLimitBytes60 - 3;
- }
- bytesLeftSpecCoding = payloadLimitBytes - transcodingParam.stream_index;
-
- /* perceptual pre-filtering (using normalized lattice filter) */
- /* low-band filtering */
- WebRtcIsac_NormLatticeFilterMa(ORDERLO,
- ISACencLB_obj->maskfiltstr_obj.PreStateLoF,
- ISACencLB_obj->maskfiltstr_obj.PreStateLoG, LP, lofilt_coef, LPw);
- /* high-band filtering */
- WebRtcIsac_NormLatticeFilterMa(ORDERHI,
- ISACencLB_obj->maskfiltstr_obj.PreStateHiF,
- ISACencLB_obj->maskfiltstr_obj.PreStateHiG, HP, hifilt_coef, HPw);
-
-
- /* pitch filter */
- WebRtcIsac_PitchfilterPre(LPw, LPw_pf, &ISACencLB_obj->pitchfiltstr_obj,
- PitchLags, PitchGains);
-
- /* transform */
- WebRtcIsac_Time2Spec(LPw_pf, HPw, fre, fim, &ISACencLB_obj->fftstr_obj);
-
-
- /* Save data for multiple packets memory */
- for (k = 0; k < FRAMESAMPLES_HALF; k++) {
- ISACencLB_obj->SaveEnc_obj.fre[k +
- ISACencLB_obj->SaveEnc_obj.startIdx*FRAMESAMPLES_HALF] = fre[k];
- ISACencLB_obj->SaveEnc_obj.fim[k +
- ISACencLB_obj->SaveEnc_obj.startIdx*FRAMESAMPLES_HALF] = fim[k];
- }
- ISACencLB_obj->SaveEnc_obj.AvgPitchGain[
- ISACencLB_obj->SaveEnc_obj.startIdx] = AvgPitchGain_Q12;
-
- /* quantization and lossless coding */
- err = WebRtcIsac_EncodeSpecLb(fre, fim, &ISACencLB_obj->bitstr_obj,
- AvgPitchGain_Q12);
- if ((err < 0) && (err != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
- /* There has been an error but it was not too large payload
- (we can cure too large payload) */
- if (frame_mode == 1 && ISACencLB_obj->frame_nb == 1) {
- /* If this is the second 30ms of a 60ms frame reset
- this such that in the next call encoder starts fresh. */
- ISACencLB_obj->frame_nb = 0;
- }
- return err;
- }
- iterCntr = 0;
- while((ISACencLB_obj->bitstr_obj.stream_index > payloadLimitBytes) ||
- (err == -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
- double bytesSpecCoderUsed;
- double transcodeScale;
-
- if(iterCntr >= MAX_PAYLOAD_LIMIT_ITERATION) {
- /* We were not able to limit the payload size */
- if((frame_mode == 1) && (ISACencLB_obj->frame_nb == 0)) {
- /* This was the first 30ms of a 60ms frame. Although
- the payload is larger than it should be but we let
- the second 30ms be encoded. Maybe together we
- won't exceed the limit. */
- ISACencLB_obj->frame_nb = 1;
- return 0;
- } else if((frame_mode == 1) && (ISACencLB_obj->frame_nb == 1)) {
- ISACencLB_obj->frame_nb = 0;
- }
-
- if(err != -ISAC_DISALLOWED_BITSTREAM_LENGTH) {
- return -ISAC_PAYLOAD_LARGER_THAN_LIMIT;
- } else {
- return status;
- }
- }
-
- if(err == -ISAC_DISALLOWED_BITSTREAM_LENGTH) {
- bytesSpecCoderUsed = STREAM_SIZE_MAX;
- // being coservative
- transcodeScale = bytesLeftSpecCoding / bytesSpecCoderUsed * 0.5;
- } else {
- bytesSpecCoderUsed = ISACencLB_obj->bitstr_obj.stream_index -
- transcodingParam.stream_index;
- transcodeScale = bytesLeftSpecCoding / bytesSpecCoderUsed;
- }
-
- /* To be safe, we reduce the scale depending on
- the number of iterations. */
- transcodeScale *= (1.0 - (0.9 * (double)iterCntr /
- (double)MAX_PAYLOAD_LIMIT_ITERATION));
-
- /* Scale the LPC Gains */
- for (k = 0; k < SUBFRAMES; k++) {
- lofilt_coef[(LPC_LOBAND_ORDER+1) * k] =
- transcodingParam.loFiltGain[k] * transcodeScale;
- hifilt_coef[(LPC_HIBAND_ORDER+1) * k] =
- transcodingParam.hiFiltGain[k] * transcodeScale;
- transcodingParam.loFiltGain[k] =
- lofilt_coef[(LPC_LOBAND_ORDER+1) * k];
- transcodingParam.hiFiltGain[k] =
- hifilt_coef[(LPC_HIBAND_ORDER+1) * k];
- }
-
- /* Scale DFT coefficients */
- for (k = 0; k < FRAMESAMPLES_HALF; k++) {
- fre[k] = (WebRtc_Word16)(fre[k] * transcodeScale);
- fim[k] = (WebRtc_Word16)(fim[k] * transcodeScale);
- }
-
- /* Save data for multiple packets memory */
- for (k = 0; k < FRAMESAMPLES_HALF; k++) {
- ISACencLB_obj->SaveEnc_obj.fre[k +
- ISACencLB_obj->SaveEnc_obj.startIdx * FRAMESAMPLES_HALF] =
- fre[k];
- ISACencLB_obj->SaveEnc_obj.fim[k +
- ISACencLB_obj->SaveEnc_obj.startIdx * FRAMESAMPLES_HALF] =
- fim[k];
- }
-
- /* Re-store the state of arithmetic coder before coding LPC gains */
- ISACencLB_obj->bitstr_obj.W_upper = transcodingParam.W_upper;
- ISACencLB_obj->bitstr_obj.stream_index = transcodingParam.stream_index;
- ISACencLB_obj->bitstr_obj.streamval = transcodingParam.streamval;
- ISACencLB_obj->bitstr_obj.stream[transcodingParam.stream_index - 2] =
- transcodingParam.stream[0];
- ISACencLB_obj->bitstr_obj.stream[transcodingParam.stream_index - 1] =
- transcodingParam.stream[1];
- ISACencLB_obj->bitstr_obj.stream[transcodingParam.stream_index] =
- transcodingParam.stream[2];
-
- /* Code gains */
- WebRtcIsac_EncodeLpcGainLb(lofilt_coef, hifilt_coef, bmodel,
- &ISACencLB_obj->bitstr_obj, &ISACencLB_obj->SaveEnc_obj);
-
- /* Update the number of bytes left for encoding the spectrum */
- bytesLeftSpecCoding = payloadLimitBytes -
- transcodingParam.stream_index;
-
- /* Encode the spectrum */
- err = WebRtcIsac_EncodeSpecLb(fre, fim, &ISACencLB_obj->bitstr_obj,
- AvgPitchGain_Q12);
- if((err < 0) && (err != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
- /* There has been an error but it was not too large
- payload (we can cure too large payload) */
- if (frame_mode == 1 && ISACencLB_obj->frame_nb == 1) {
- /* If this is the second 30ms of a 60ms frame reset
- this such that in the next call encoder starts fresh. */
- ISACencLB_obj->frame_nb = 0;
- }
- return err;
- }
- iterCntr++;
- }
-
- /* i.e. 60 ms framesize and just processed the first 30ms, */
- /* go back to main function to buffer the other 30ms speech frame */
- if (frame_mode == 1)
- {
- if(ISACencLB_obj->frame_nb == 0)
- {
- ISACencLB_obj->frame_nb = 1;
- return 0;
- }
- else if(ISACencLB_obj->frame_nb == 1)
- {
- ISACencLB_obj->frame_nb = 0;
- /* also update the framelength for next packet,
- in Adaptive mode only */
- if (codingMode == 0 && (ISACencLB_obj->enforceFrameSize == 0))
- {
- ISACencLB_obj->new_framelength =
- WebRtcIsac_GetNewFrameLength(ISACencLB_obj->bottleneck,
- ISACencLB_obj->current_framesamples);
- }
- }
- }
- else
- {
- ISACencLB_obj->frame_nb = 0;
- }
-
- /* complete arithmetic coding */
- stream_length = WebRtcIsac_EncTerminate(&ISACencLB_obj->bitstr_obj);
-
- return stream_length;
-}
-
-int
-WebRtcIsac_EncodeUb16(
- float* in,
- ISACUBEncStruct* ISACencUB_obj,
- WebRtc_Word32 jitterInfo)
-{
- int err;
- int k;
-
- double lpcVecs[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
- double percepFilterParams[(1 + UB_LPC_ORDER) * (SUBFRAMES<<1) +
- (1 + UB_LPC_ORDER)];
-
- double LP_lookahead[FRAMESAMPLES];
- WebRtc_Word16 fre[FRAMESAMPLES_HALF]; /* Q7 */
- WebRtc_Word16 fim[FRAMESAMPLES_HALF]; /* Q7 */
-
- int status = 0;
-
- double varscale[2];
- double corr[SUBFRAMES<<1][UB_LPC_ORDER + 1];
- double lpcGains[SUBFRAMES<<1];
- transcode_obj transcodingParam;
- double bytesLeftSpecCoding;
- WebRtc_UWord16 payloadLimitBytes;
- WebRtc_UWord16 iterCntr;
- double s2nr;
-
- /* buffer speech samples (by 10ms packet) until the framelength is */
- /* reached (30 or 60 ms) */
- /*********************************************************************/
-
- /* fill the buffer with 10ms input data */
- for (k = 0; k < FRAMESAMPLES_10ms; k++) {
- ISACencUB_obj->data_buffer_float[k + ISACencUB_obj->buffer_index] =
- in[k];
- }
-
- /* if buffersize is not equal to current framesize, and end of file is
- not reached yet, we don't do encoding unless we have the whole frame */
- if (ISACencUB_obj->buffer_index + FRAMESAMPLES_10ms < FRAMESAMPLES) {
- ISACencUB_obj->buffer_index += FRAMESAMPLES_10ms;
- return 0;
- }
-
- /* end of buffer function */
- /**************************/
-
- /* encoding */
- /************/
-
- /* reset bitstream */
- ISACencUB_obj->bitstr_obj.W_upper = 0xFFFFFFFF;
- ISACencUB_obj->bitstr_obj.streamval = 0;
- ISACencUB_obj->bitstr_obj.stream_index = 0;
-
- /* bandwidth estimation and coding */
- /* To be used for Redundant Coding */
- WebRtcIsac_EncodeJitterInfo(jitterInfo, &ISACencUB_obj->bitstr_obj);
-
- status = WebRtcIsac_EncodeBandwidth(isac16kHz,
- &ISACencUB_obj->bitstr_obj);
- if (status < 0) {
- return status;
- }
-
- s2nr = WebRtcIsac_GetSnr(ISACencUB_obj->bottleneck,
- FRAMESAMPLES);
-
- memcpy(lpcVecs, ISACencUB_obj->lastLPCVec, UB_LPC_ORDER * sizeof(double));
-
- for (k = 0; k < FRAMESAMPLES; k++) {
- LP_lookahead[k] = ISACencUB_obj->data_buffer_float[UB_LOOKAHEAD + k];
- }
-
- /* find coefficients for perceptual pre-filters */
- WebRtcIsac_GetLpcCoefUb(LP_lookahead, &ISACencUB_obj->maskfiltstr_obj,
- &lpcVecs[UB_LPC_ORDER], corr, varscale, isac16kHz);
-
- memcpy(ISACencUB_obj->lastLPCVec,
- &lpcVecs[(UB16_LPC_VEC_PER_FRAME - 1) * (UB_LPC_ORDER)],
- sizeof(double) * UB_LPC_ORDER);
-
- /* code LPC model and shape - gains not quantized yet */
- WebRtcIsac_EncodeLpcUB(lpcVecs, &ISACencUB_obj->bitstr_obj,
- percepFilterParams, isac16kHz, &ISACencUB_obj->SaveEnc_obj);
-
-
- // the first set of lpc parameters are from the last sub-frame of
- // the previous frame. so we don't care about them
- WebRtcIsac_GetLpcGain(s2nr, &percepFilterParams[UB_LPC_ORDER + 1],
- (SUBFRAMES<<1), lpcGains, corr, varscale);
-
- /* Store the state of arithmetic coder before coding LPC gains */
- transcodingParam.stream_index = ISACencUB_obj->bitstr_obj.stream_index;
- transcodingParam.W_upper = ISACencUB_obj->bitstr_obj.W_upper;
- transcodingParam.streamval = ISACencUB_obj->bitstr_obj.streamval;
- transcodingParam.stream[0] = ISACencUB_obj->bitstr_obj.stream[
- ISACencUB_obj->bitstr_obj.stream_index - 2];
- transcodingParam.stream[1] = ISACencUB_obj->bitstr_obj.stream[
- ISACencUB_obj->bitstr_obj.stream_index - 1];
- transcodingParam.stream[2] = ISACencUB_obj->bitstr_obj.stream[
- ISACencUB_obj->bitstr_obj.stream_index];
-
- /* Store LPC Gains before encoding them */
- for(k = 0; k < SUBFRAMES; k++) {
- transcodingParam.loFiltGain[k] = lpcGains[k];
- transcodingParam.hiFiltGain[k] = lpcGains[SUBFRAMES + k];
- }
-
- // Store the gains for multiple encoding
- memcpy(ISACencUB_obj->SaveEnc_obj.lpcGain, lpcGains, (SUBFRAMES << 1) * sizeof(double));
-
- WebRtcIsac_EncodeLpcGainUb(lpcGains, &ISACencUB_obj->bitstr_obj,
- ISACencUB_obj->SaveEnc_obj.lpcGainIndex);
- WebRtcIsac_EncodeLpcGainUb(&lpcGains[SUBFRAMES], &ISACencUB_obj->bitstr_obj,
- &ISACencUB_obj->SaveEnc_obj.lpcGainIndex[SUBFRAMES]);
-
- /* Get the correct value for the payload limit and calculate the number of
- bytes left for coding the spectrum. It is a 30ms frame
- Subract 3 because termination process may add 3 bytes */
- payloadLimitBytes = ISACencUB_obj->maxPayloadSizeBytes -
- ISACencUB_obj->numBytesUsed - 3;
- bytesLeftSpecCoding = payloadLimitBytes -
- ISACencUB_obj->bitstr_obj.stream_index;
-
- for (k = 0; k < (SUBFRAMES<<1); k++) {
- percepFilterParams[k*(UB_LPC_ORDER + 1) + (UB_LPC_ORDER + 1)] =
- lpcGains[k];
- }
-
- /* perceptual pre-filtering (using normalized lattice filter) */
- /* first half-frame filtering */
- WebRtcIsac_NormLatticeFilterMa(UB_LPC_ORDER,
- ISACencUB_obj->maskfiltstr_obj.PreStateLoF,
- ISACencUB_obj->maskfiltstr_obj.PreStateLoG,
- &ISACencUB_obj->data_buffer_float[0],
- &percepFilterParams[UB_LPC_ORDER + 1],
- &LP_lookahead[0]);
-
- /* Second half-frame filtering */
- WebRtcIsac_NormLatticeFilterMa(UB_LPC_ORDER,
- ISACencUB_obj->maskfiltstr_obj.PreStateLoF,
- ISACencUB_obj->maskfiltstr_obj.PreStateLoG,
- &ISACencUB_obj->data_buffer_float[FRAMESAMPLES_HALF],
- &percepFilterParams[(UB_LPC_ORDER + 1) + SUBFRAMES *
- (UB_LPC_ORDER + 1)], &LP_lookahead[FRAMESAMPLES_HALF]);
-
- WebRtcIsac_Time2Spec(&LP_lookahead[0], &LP_lookahead[FRAMESAMPLES_HALF],
- fre, fim, &ISACencUB_obj->fftstr_obj);
-
- //Store FFT coefficients for multiple encoding
- memcpy(&ISACencUB_obj->SaveEnc_obj.realFFT, fre,
- FRAMESAMPLES_HALF * sizeof(WebRtc_Word16));
-
- memcpy(&ISACencUB_obj->SaveEnc_obj.imagFFT, fim,
- FRAMESAMPLES_HALF * sizeof(WebRtc_Word16));
-
- // Prepare the audio buffer for the next packet
- // move the last 3 ms to the beginning of the buffer
- memcpy(ISACencUB_obj->data_buffer_float,
- &ISACencUB_obj->data_buffer_float[FRAMESAMPLES],
- LB_TOTAL_DELAY_SAMPLES * sizeof(float));
- // start writing with 3 ms delay to compensate for the delay
- // of the lower-band.
- ISACencUB_obj->buffer_index = LB_TOTAL_DELAY_SAMPLES;
-
- // Save the bit-stream object at this point for FEC.
- memcpy(&ISACencUB_obj->SaveEnc_obj.bitStreamObj,
- &ISACencUB_obj->bitstr_obj, sizeof(Bitstr));
-
- /* quantization and lossless coding */
- err = WebRtcIsac_EncodeSpecUB16(fre, fim, &ISACencUB_obj->bitstr_obj);
- if ((err < 0) && (err != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
- return err;
- }
-
- iterCntr = 0;
- while((ISACencUB_obj->bitstr_obj.stream_index > payloadLimitBytes) ||
- (err == -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
- double bytesSpecCoderUsed;
- double transcodeScale;
-
- if (iterCntr >= MAX_PAYLOAD_LIMIT_ITERATION) {
- /* We were not able to limit the payload size */
- return -ISAC_PAYLOAD_LARGER_THAN_LIMIT;
- }
-
- if (err == -ISAC_DISALLOWED_BITSTREAM_LENGTH) {
- bytesSpecCoderUsed = STREAM_SIZE_MAX;
- // being conservative
- transcodeScale = bytesLeftSpecCoding / bytesSpecCoderUsed * 0.5;
- } else {
- bytesSpecCoderUsed = ISACencUB_obj->bitstr_obj.stream_index -
- transcodingParam.stream_index;
- transcodeScale = bytesLeftSpecCoding / bytesSpecCoderUsed;
- }
-
- /* To be safe, we reduce the scale depending on the
- number of iterations. */
- transcodeScale *= (1.0 - (0.9 * (double)iterCntr/
- (double)MAX_PAYLOAD_LIMIT_ITERATION));
-
- /* Scale the LPC Gains */
- for (k = 0; k < SUBFRAMES; k++) {
- transcodingParam.loFiltGain[k] *= transcodeScale;
- transcodingParam.hiFiltGain[k] *= transcodeScale;
- }
-
- /* Scale DFT coefficients */
- for (k = 0; k < FRAMESAMPLES_HALF; k++) {
- fre[k] = (WebRtc_Word16)(fre[k] * transcodeScale + 0.5);
- fim[k] = (WebRtc_Word16)(fim[k] * transcodeScale + 0.5);
- }
-
- //Store FFT coefficients for multiple encoding
- memcpy(&ISACencUB_obj->SaveEnc_obj.realFFT, fre,
- FRAMESAMPLES_HALF * sizeof(WebRtc_Word16));
-
- memcpy(&ISACencUB_obj->SaveEnc_obj.imagFFT, fim,
- FRAMESAMPLES_HALF * sizeof(WebRtc_Word16));
-
-
- /* Store the state of arithmetic coder before coding LPC gains */
- ISACencUB_obj->bitstr_obj.W_upper = transcodingParam.W_upper;
-
- ISACencUB_obj->bitstr_obj.stream_index = transcodingParam.stream_index;
-
- ISACencUB_obj->bitstr_obj.streamval = transcodingParam.streamval;
-
- ISACencUB_obj->bitstr_obj.stream[transcodingParam.stream_index - 2] =
- transcodingParam.stream[0];
-
- ISACencUB_obj->bitstr_obj.stream[transcodingParam.stream_index - 1] =
- transcodingParam.stream[1];
-
- ISACencUB_obj->bitstr_obj.stream[transcodingParam.stream_index] =
- transcodingParam.stream[2];
-
- // Store the gains for multiple encoding
- memcpy(ISACencUB_obj->SaveEnc_obj.lpcGain, lpcGains,
- (SUBFRAMES << 1) * sizeof(double));
-
- WebRtcIsac_EncodeLpcGainUb(transcodingParam.loFiltGain,
- &ISACencUB_obj->bitstr_obj,
- ISACencUB_obj->SaveEnc_obj.lpcGainIndex);
- WebRtcIsac_EncodeLpcGainUb(transcodingParam.hiFiltGain,
- &ISACencUB_obj->bitstr_obj,
- &ISACencUB_obj->SaveEnc_obj.lpcGainIndex[SUBFRAMES]);
-
- /* Update the number of bytes left for encoding the spectrum */
- bytesLeftSpecCoding = payloadLimitBytes -
- ISACencUB_obj->bitstr_obj.stream_index;
-
- // Save the bit-stream object at this point for FEC.
- memcpy(&ISACencUB_obj->SaveEnc_obj.bitStreamObj,
- &ISACencUB_obj->bitstr_obj, sizeof(Bitstr));
-
- /* Encode the spectrum */
- err = WebRtcIsac_EncodeSpecUB16(fre, fim, &ISACencUB_obj->bitstr_obj);
- if ((err < 0) && (err != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
- /* There has been an error but it was not too large payload
- (we can cure too large payload) */
- return err;
- }
- iterCntr++;
- }
-
- /* complete arithmetic coding */
- return WebRtcIsac_EncTerminate(&ISACencUB_obj->bitstr_obj);
-}
-
-
-int
-WebRtcIsac_EncodeUb12(
- float* in,
- ISACUBEncStruct* ISACencUB_obj,
- WebRtc_Word32 jitterInfo)
-{
- int err;
- int k;
- int iterCntr;
-
- double lpcVecs[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
-
- double percepFilterParams[(1 + UB_LPC_ORDER) * SUBFRAMES];
- float LP[FRAMESAMPLES_HALF];
- float HP[FRAMESAMPLES_HALF];
-
- double LP_lookahead[FRAMESAMPLES_HALF];
- double HP_lookahead[FRAMESAMPLES_HALF];
- double LPw[FRAMESAMPLES_HALF];
-
- double HPw[FRAMESAMPLES_HALF];
- WebRtc_Word16 fre[FRAMESAMPLES_HALF]; /* Q7 */
- WebRtc_Word16 fim[FRAMESAMPLES_HALF]; /* Q7 */
-
- int status = 0;
-
- double varscale[1];
-
- double corr[UB_LPC_GAIN_DIM][UB_LPC_ORDER + 1];
- double lpcGains[SUBFRAMES];
- transcode_obj transcodingParam;
- double bytesLeftSpecCoding;
- WebRtc_UWord16 payloadLimitBytes;
- double s2nr;
-
- /* buffer speech samples (by 10ms packet) until the framelength is */
- /* reached (30 or 60 ms) */
- /********************************************************************/
-
- /* fill the buffer with 10ms input data */
- for (k=0; k<FRAMESAMPLES_10ms; k++) {
- ISACencUB_obj->data_buffer_float[k + ISACencUB_obj->buffer_index] =
- in[k];
- }
-
- /* if buffer-size is not equal to current frame-size then increase the
- index and return. We do the encoding when we have enough audio. */
- if (ISACencUB_obj->buffer_index + FRAMESAMPLES_10ms < FRAMESAMPLES) {
- ISACencUB_obj->buffer_index += FRAMESAMPLES_10ms;
- return 0;
- }
- /* if buffer reached the right size, reset index and continue
- with encoding the frame */
- ISACencUB_obj->buffer_index = 0;
-
- /* end of buffer function */
- /**************************/
-
- /* encoding */
- /************/
-
- /* reset bitstream */
- ISACencUB_obj->bitstr_obj.W_upper = 0xFFFFFFFF;
- ISACencUB_obj->bitstr_obj.streamval = 0;
- ISACencUB_obj->bitstr_obj.stream_index = 0;
-
- /* bandwidth estimation and coding */
- /* To be used for Redundant Coding */
- WebRtcIsac_EncodeJitterInfo(jitterInfo, &ISACencUB_obj->bitstr_obj);
-
- status = WebRtcIsac_EncodeBandwidth(isac12kHz,
- &ISACencUB_obj->bitstr_obj);
- if (status < 0) {
- return status;
- }
-
-
- s2nr = WebRtcIsac_GetSnr(ISACencUB_obj->bottleneck,
- FRAMESAMPLES);
-
- /* split signal in two bands */
- WebRtcIsac_SplitAndFilterFloat(ISACencUB_obj->data_buffer_float, HP, LP,
- HP_lookahead, LP_lookahead, &ISACencUB_obj->prefiltbankstr_obj);
-
- /* find coefficients for perceptual pre-filters */
- WebRtcIsac_GetLpcCoefUb(LP_lookahead, &ISACencUB_obj->maskfiltstr_obj,
- lpcVecs, corr, varscale, isac12kHz);
-
- /* code LPC model and shape - gains not quantized yet */
- WebRtcIsac_EncodeLpcUB(lpcVecs, &ISACencUB_obj->bitstr_obj,
- percepFilterParams, isac12kHz, &ISACencUB_obj->SaveEnc_obj);
-
- WebRtcIsac_GetLpcGain(s2nr, percepFilterParams, SUBFRAMES, lpcGains,
- corr, varscale);
-
- /* Store the state of arithmetic coder before coding LPC gains */
- transcodingParam.W_upper = ISACencUB_obj->bitstr_obj.W_upper;
-
- transcodingParam.stream_index = ISACencUB_obj->bitstr_obj.stream_index;
-
- transcodingParam.streamval = ISACencUB_obj->bitstr_obj.streamval;
-
- transcodingParam.stream[0] = ISACencUB_obj->bitstr_obj.stream[
- ISACencUB_obj->bitstr_obj.stream_index - 2];
-
- transcodingParam.stream[1] = ISACencUB_obj->bitstr_obj.stream[
- ISACencUB_obj->bitstr_obj.stream_index - 1];
-
- transcodingParam.stream[2] = ISACencUB_obj->bitstr_obj.stream[
- ISACencUB_obj->bitstr_obj.stream_index];
-
- /* Store LPC Gains before encoding them */
- for(k = 0; k < SUBFRAMES; k++) {
- transcodingParam.loFiltGain[k] = lpcGains[k];
- }
-
- // Store the gains for multiple encoding
- memcpy(ISACencUB_obj->SaveEnc_obj.lpcGain, lpcGains, SUBFRAMES *
- sizeof(double));
-
- WebRtcIsac_EncodeLpcGainUb(lpcGains, &ISACencUB_obj->bitstr_obj,
- ISACencUB_obj->SaveEnc_obj.lpcGainIndex);
-
- for(k = 0; k < SUBFRAMES; k++) {
- percepFilterParams[k*(UB_LPC_ORDER + 1)] = lpcGains[k];
- }
-
- /* perceptual pre-filtering (using normalized lattice filter) */
- /* low-band filtering */
- WebRtcIsac_NormLatticeFilterMa(UB_LPC_ORDER,
- ISACencUB_obj->maskfiltstr_obj.PreStateLoF,
- ISACencUB_obj->maskfiltstr_obj.PreStateLoG, LP, percepFilterParams,
- LPw);
-
- /* Get the correct value for the payload limit and calculate the number
- of bytes left for coding the spectrum. It is a 30ms frame Subract 3
- because termination process may add 3 bytes */
- payloadLimitBytes = ISACencUB_obj->maxPayloadSizeBytes -
- ISACencUB_obj->numBytesUsed - 3;
- bytesLeftSpecCoding = payloadLimitBytes -
- ISACencUB_obj->bitstr_obj.stream_index;
-
- memset(HPw, 0, sizeof(double) * FRAMESAMPLES_HALF);
-
- /* transform */
- WebRtcIsac_Time2Spec(LPw, HPw, fre, fim, &ISACencUB_obj->fftstr_obj);
-
- //Store real FFT coefficients for multiple encoding
- memcpy(&ISACencUB_obj->SaveEnc_obj.realFFT, fre,
- FRAMESAMPLES_HALF * sizeof(WebRtc_Word16));
-
- //Store imaginary FFT coefficients for multiple encoding
- memcpy(&ISACencUB_obj->SaveEnc_obj.imagFFT, fim,
- FRAMESAMPLES_HALF * sizeof(WebRtc_Word16));
-
- // Save the bit-stream object at this point for FEC.
- memcpy(&ISACencUB_obj->SaveEnc_obj.bitStreamObj,
- &ISACencUB_obj->bitstr_obj, sizeof(Bitstr));
-
- /* quantization and lossless coding */
- err = WebRtcIsac_EncodeSpecUB12(fre, fim, &ISACencUB_obj->bitstr_obj);
- if ((err < 0) && (err != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
- /* There has been an error but it was not too large
- payload (we can cure too large payload) */
- return err;
- }
- iterCntr = 0;
- while((ISACencUB_obj->bitstr_obj.stream_index > payloadLimitBytes) ||
- (err == -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
- double bytesSpecCoderUsed;
- double transcodeScale;
-
- if (iterCntr >= MAX_PAYLOAD_LIMIT_ITERATION) {
- /* We were not able to limit the payload size */
- return -ISAC_PAYLOAD_LARGER_THAN_LIMIT;
- }
-
- if (err == -ISAC_DISALLOWED_BITSTREAM_LENGTH) {
- bytesSpecCoderUsed = STREAM_SIZE_MAX;
- // being coservative
- transcodeScale = bytesLeftSpecCoding / bytesSpecCoderUsed * 0.5;
- } else {
- bytesSpecCoderUsed = ISACencUB_obj->bitstr_obj.stream_index -
- transcodingParam.stream_index;
- transcodeScale = bytesLeftSpecCoding / bytesSpecCoderUsed;
- }
-
- /* To be safe, we reduce the scale depending on the
- number of iterations. */
- transcodeScale *= (1.0 - (0.9 * (double)iterCntr/
- (double)MAX_PAYLOAD_LIMIT_ITERATION));
-
- /* Scale the LPC Gains */
- for (k = 0; k < SUBFRAMES; k++) {
- transcodingParam.loFiltGain[k] *= transcodeScale;
- }
-
- /* Scale DFT coefficients */
- for (k = 0; k < FRAMESAMPLES_HALF; k++) {
- fre[k] = (WebRtc_Word16)(fre[k] * transcodeScale + 0.5);
- fim[k] = (WebRtc_Word16)(fim[k] * transcodeScale + 0.5);
- }
-
- //Store real FFT coefficients for multiple encoding
- memcpy(&ISACencUB_obj->SaveEnc_obj.realFFT, fre,
- FRAMESAMPLES_HALF * sizeof(WebRtc_Word16));
-
- //Store imaginary FFT coefficients for multiple encoding
- memcpy(&ISACencUB_obj->SaveEnc_obj.imagFFT, fim,
- FRAMESAMPLES_HALF * sizeof(WebRtc_Word16));
-
-
- /* Re-store the state of arithmetic coder before coding LPC gains */
- ISACencUB_obj->bitstr_obj.W_upper = transcodingParam.W_upper;
-
- ISACencUB_obj->bitstr_obj.stream_index = transcodingParam.stream_index;
-
- ISACencUB_obj->bitstr_obj.streamval = transcodingParam.streamval;
-
- ISACencUB_obj->bitstr_obj.stream[transcodingParam.stream_index - 2] =
- transcodingParam.stream[0];
-
- ISACencUB_obj->bitstr_obj.stream[transcodingParam.stream_index - 1] =
- transcodingParam.stream[1];
-
- ISACencUB_obj->bitstr_obj.stream[transcodingParam.stream_index] =
- transcodingParam.stream[2];
-
- // Store the gains for multiple encoding
- memcpy(&ISACencUB_obj->SaveEnc_obj.lpcGain, lpcGains, SUBFRAMES *
- sizeof(double));
-
- // encode LPC gain and store quantization indices. HAving quantization
- // indices reduces transcoding complexity if 'scale factor' is 1.
- WebRtcIsac_EncodeLpcGainUb(transcodingParam.loFiltGain,
- &ISACencUB_obj->bitstr_obj,
- ISACencUB_obj->SaveEnc_obj.lpcGainIndex);
-
- // Save the bit-stream object at this point for FEC.
- memcpy(&ISACencUB_obj->SaveEnc_obj.bitStreamObj,
- &ISACencUB_obj->bitstr_obj, sizeof(Bitstr));
-
- /* Update the number of bytes left for encoding the spectrum */
- bytesLeftSpecCoding = payloadLimitBytes -
- ISACencUB_obj->bitstr_obj.stream_index;
-
- /* Encode the spectrum */
- err = WebRtcIsac_EncodeSpecUB12(fre, fim,
- &ISACencUB_obj->bitstr_obj);
- if ((err < 0) && (err != -ISAC_DISALLOWED_BITSTREAM_LENGTH)) {
- /* There has been an error but it was not too large payload
- (we can cure too large payload) */
- return err;
- }
- iterCntr++;
- }
-
- /* complete arithmetic coding */
- return WebRtcIsac_EncTerminate(&ISACencUB_obj->bitstr_obj);
-}
-
-
-
-
-
-
-/* This function is used to create a new bitstream with new BWE.
- The same data as previously encoded with the function WebRtcIsac_Encoder().
- The data needed is taken from the struct, where it was stored
- when calling the encoder. */
-
-int WebRtcIsac_EncodeStoredDataLb(
- const ISAC_SaveEncData_t* ISACSavedEnc_obj,
- Bitstr* ISACBitStr_obj,
- int BWnumber,
- float scale)
-{
- int ii;
- int status;
- int BWno = BWnumber;
-
- const WebRtc_UWord16 *WebRtcIsac_kQPitchGainCdf_ptr[1];
- const WebRtc_UWord16 **cdf;
-
- double tmpLPCcoeffs_lo[(ORDERLO+1)*SUBFRAMES*2];
- double tmpLPCcoeffs_hi[(ORDERHI+1)*SUBFRAMES*2];
- int tmpLPCindex_g[12*2];
- WebRtc_Word16 tmp_fre[FRAMESAMPLES], tmp_fim[FRAMESAMPLES];
-
- /* Sanity Check - possible values for BWnumber is 0 - 23 */
- if ((BWnumber < 0) || (BWnumber > 23)) {
- return -ISAC_RANGE_ERROR_BW_ESTIMATOR;
- }
-
- /* reset bitstream */
- ISACBitStr_obj->W_upper = 0xFFFFFFFF;
- ISACBitStr_obj->streamval = 0;
- ISACBitStr_obj->stream_index = 0;
-
- /* encode frame length */
- status = WebRtcIsac_EncodeFrameLen(ISACSavedEnc_obj->framelength,
- ISACBitStr_obj);
- if (status < 0) {
- /* Wrong frame size */
- return status;
- }
-
- /* Transcoding */
- if ((scale > 0.0) && (scale < 1.0)) {
- /* Compensate LPC gain */
- for (ii = 0;
- ii < ((ORDERLO + 1)* SUBFRAMES * (1 + ISACSavedEnc_obj->startIdx));
- ii++) {
- tmpLPCcoeffs_lo[ii] = scale * ISACSavedEnc_obj->LPCcoeffs_lo[ii];
- }
- for (ii = 0;
- ii < ((ORDERHI + 1) * SUBFRAMES *(1 + ISACSavedEnc_obj->startIdx));
- ii++) {
- tmpLPCcoeffs_hi[ii] = scale * ISACSavedEnc_obj->LPCcoeffs_hi[ii];
- }
- /* Scale DFT */
- for (ii = 0;
- ii < (FRAMESAMPLES_HALF * (1 + ISACSavedEnc_obj->startIdx));
- ii++) {
- tmp_fre[ii] = (WebRtc_Word16)((scale) *
- (float)ISACSavedEnc_obj->fre[ii]) ;
- tmp_fim[ii] = (WebRtc_Word16)((scale) *
- (float)ISACSavedEnc_obj->fim[ii]) ;
- }
- } else {
- for (ii = 0;
- ii < (KLT_ORDER_GAIN * (1 + ISACSavedEnc_obj->startIdx));
- ii++) {
- tmpLPCindex_g[ii] = ISACSavedEnc_obj->LPCindex_g[ii];
- }
- for (ii = 0;
- ii < (FRAMESAMPLES_HALF * (1 + ISACSavedEnc_obj->startIdx));
- ii++) {
- tmp_fre[ii] = ISACSavedEnc_obj->fre[ii];
- tmp_fim[ii] = ISACSavedEnc_obj->fim[ii];
- }
- }
-
- /* encode bandwidth estimate */
- WebRtcIsac_EncodeReceiveBw(&BWno, ISACBitStr_obj);
-
- /* Loop over number of 30 msec */
- for (ii = 0; ii <= ISACSavedEnc_obj->startIdx; ii++) {
- /* encode pitch gains */
- *WebRtcIsac_kQPitchGainCdf_ptr = WebRtcIsac_kQPitchGainCdf;
- WebRtcIsac_EncHistMulti(ISACBitStr_obj,
- &ISACSavedEnc_obj->pitchGain_index[ii], WebRtcIsac_kQPitchGainCdf_ptr, 1);
-
- /* entropy coding of quantization pitch lags */
- /* voicing classificiation */
- if (ISACSavedEnc_obj->meanGain[ii] < 0.2) {
- cdf = WebRtcIsac_kQPitchLagCdfPtrLo;
- } else if (ISACSavedEnc_obj->meanGain[ii] < 0.4) {
- cdf = WebRtcIsac_kQPitchLagCdfPtrMid;
- } else {
- cdf = WebRtcIsac_kQPitchLagCdfPtrHi;
- }
- WebRtcIsac_EncHistMulti(ISACBitStr_obj,
- &ISACSavedEnc_obj->pitchIndex[PITCH_SUBFRAMES*ii], cdf,
- PITCH_SUBFRAMES);
-
- /* LPC */
- /* entropy coding of model number */
- WebRtcIsac_EncHistMulti(ISACBitStr_obj,
- &ISACSavedEnc_obj->LPCmodel[ii], WebRtcIsac_kQKltModelCdfPtr, 1);
-
- /* entropy coding of quantization indices - LPC shape only */
- WebRtcIsac_EncHistMulti(ISACBitStr_obj,
- &ISACSavedEnc_obj->LPCindex_s[KLT_ORDER_SHAPE*ii],
- WebRtcIsac_kQKltCdfPtrShape[ISACSavedEnc_obj->LPCmodel[ii]],
- KLT_ORDER_SHAPE);
-
- /* If transcoding, get new LPC gain indices */
- if (scale < 1.0) {
- WebRtcIsac_TranscodeLPCCoef(&tmpLPCcoeffs_lo[(ORDERLO+1) *
- SUBFRAMES*ii], &tmpLPCcoeffs_hi[(ORDERHI+1)*SUBFRAMES*ii],
- ISACSavedEnc_obj->LPCmodel[ii],
- &tmpLPCindex_g[KLT_ORDER_GAIN * ii]);
- }
-
- /* entropy coding of quantization indices - LPC gain */
- WebRtcIsac_EncHistMulti(ISACBitStr_obj,
- &tmpLPCindex_g[KLT_ORDER_GAIN*ii], WebRtcIsac_kQKltCdfPtrGain[
- ISACSavedEnc_obj->LPCmodel[ii]], KLT_ORDER_GAIN);
-
- /* quantization and lossless coding */
- status = WebRtcIsac_EncodeSpecLb(&tmp_fre[ii*FRAMESAMPLES_HALF],
- &tmp_fim[ii*FRAMESAMPLES_HALF], ISACBitStr_obj,
- ISACSavedEnc_obj->AvgPitchGain[ii]);
- if (status < 0) {
- return status;
- }
- }
-
- /* complete arithmetic coding */
- return WebRtcIsac_EncTerminate(ISACBitStr_obj);
-}
-
-
-
-
-int WebRtcIsac_EncodeStoredDataUb12(
- const ISACUBSaveEncDataStruct* ISACSavedEnc_obj,
- Bitstr* bitStream,
- WebRtc_Word32 jitterInfo,
- float scale)
-{
- int n;
- int err;
- double lpcGain[SUBFRAMES];
- WebRtc_Word16 realFFT[FRAMESAMPLES_HALF];
- WebRtc_Word16 imagFFT[FRAMESAMPLES_HALF];
-
- /* reset bitstream */
- bitStream->W_upper = 0xFFFFFFFF;
- bitStream->streamval = 0;
- bitStream->stream_index = 0;
-
- // Encode jitter index
- WebRtcIsac_EncodeJitterInfo(jitterInfo, bitStream);
-
- err = WebRtcIsac_EncodeBandwidth(isac12kHz, bitStream);
- if(err < 0)
- {
- return err;
- }
-
- // Encode LPC-shape
- WebRtcIsac_EncHistMulti(bitStream, ISACSavedEnc_obj->indexLPCShape,
- WebRtcIsac_kLpcShapeCdfMatUb12, UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME);
-
-
- // we only consider scales between zero and one.
- if((scale <= 0.0) || (scale > 1.0))
- {
- scale = 1.0f;
- }
-
- if(scale == 1.0f)
- {
- //memcpy(lpcGain, ISACSavedEnc_obj->lpcGain, SUBFRAMES * sizeof(double));
- WebRtcIsac_EncHistMulti(bitStream, ISACSavedEnc_obj->lpcGainIndex,
- WebRtcIsac_kLpcGainCdfMat, UB_LPC_GAIN_DIM);
- // store FFT coefficients
- err = WebRtcIsac_EncodeSpecUB12(ISACSavedEnc_obj->realFFT,
- ISACSavedEnc_obj->imagFFT, bitStream);
- }
- else
- {
- /* scale lpc gain and FFT coefficients */
- for(n = 0; n < SUBFRAMES; n++)
- {
- lpcGain[n] = scale * ISACSavedEnc_obj->lpcGain[n];
- }
- // store lpc gain
- WebRtcIsac_StoreLpcGainUb(lpcGain, bitStream);
- for(n = 0; n < FRAMESAMPLES_HALF; n++)
- {
- realFFT[n] = (WebRtc_Word16)(scale * (float)ISACSavedEnc_obj->realFFT[n] + 0.5f);
- imagFFT[n] = (WebRtc_Word16)(scale * (float)ISACSavedEnc_obj->imagFFT[n] + 0.5f);
- }
- // store FFT coefficients
- err = WebRtcIsac_EncodeSpecUB12(realFFT, imagFFT, bitStream);
- }
- if(err < 0)
- {
- // error happened while encoding FFT coefficients.
- return err;
- }
-
- /* complete arithmetic coding */
- return WebRtcIsac_EncTerminate(bitStream);
-}
-
-
-int
-WebRtcIsac_EncodeStoredDataUb16(
- const ISACUBSaveEncDataStruct* ISACSavedEnc_obj,
- Bitstr* bitStream,
- WebRtc_Word32 jitterInfo,
- float scale)
-{
- int n;
- int err;
- double lpcGain[SUBFRAMES << 1];
- WebRtc_Word16 realFFT[FRAMESAMPLES_HALF];
- WebRtc_Word16 imagFFT[FRAMESAMPLES_HALF];
-
- /* reset bitstream */
- bitStream->W_upper = 0xFFFFFFFF;
- bitStream->streamval = 0;
- bitStream->stream_index = 0;
-
- // Encode jitter index
- WebRtcIsac_EncodeJitterInfo(jitterInfo, bitStream);
-
- err = WebRtcIsac_EncodeBandwidth(isac16kHz, bitStream);
- if(err < 0)
- {
- return err;
- }
-
- WebRtcIsac_EncHistMulti(bitStream, ISACSavedEnc_obj->indexLPCShape,
- WebRtcIsac_kLpcShapeCdfMatUb16, UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME);
-
- // we only consider scales between zero and one.
- if((scale <= 0.0) || (scale > 1.0))
- {
- scale = 1.0f;
- }
-
- if(scale == 1.0f)
- {
- // store gains
- WebRtcIsac_EncHistMulti(bitStream, ISACSavedEnc_obj->lpcGainIndex,
- WebRtcIsac_kLpcGainCdfMat, UB_LPC_GAIN_DIM);
- WebRtcIsac_EncHistMulti(bitStream, &ISACSavedEnc_obj->lpcGainIndex[SUBFRAMES],
- WebRtcIsac_kLpcGainCdfMat, UB_LPC_GAIN_DIM);
- // store FFT coefficients
- err = WebRtcIsac_EncodeSpecUB16(ISACSavedEnc_obj->realFFT,
- ISACSavedEnc_obj->imagFFT, bitStream);
-
- }
- else
- {
- /* Scale Gain */
- for(n = 0; n < SUBFRAMES; n++)
- {
- lpcGain[n] = scale * ISACSavedEnc_obj->lpcGain[n];
- lpcGain[n + SUBFRAMES] = scale * ISACSavedEnc_obj->lpcGain[n + SUBFRAMES];
- }
- // store lpc gain
- WebRtcIsac_StoreLpcGainUb(lpcGain, bitStream);
- WebRtcIsac_StoreLpcGainUb(&lpcGain[SUBFRAMES], bitStream);
- /* scale FFT coefficients */
- for(n = 0; n < FRAMESAMPLES_HALF; n++)
- {
- realFFT[n] = (WebRtc_Word16)(scale * (float)ISACSavedEnc_obj->realFFT[n] + 0.5f);
- imagFFT[n] = (WebRtc_Word16)(scale * (float)ISACSavedEnc_obj->imagFFT[n] + 0.5f);
- }
- // store FFT coefficients
- err = WebRtcIsac_EncodeSpecUB16(realFFT, imagFFT, bitStream);
- }
-
- if(err < 0)
- {
- // error happened while encoding FFT coefficients.
- return err;
- }
-
- /* complete arithmetic coding */
- return WebRtcIsac_EncTerminate(bitStream);
-}
-
-
-WebRtc_Word16
-WebRtcIsac_GetRedPayloadUb(
- const ISACUBSaveEncDataStruct* ISACSavedEncObj,
- Bitstr* bitStreamObj,
- enum ISACBandwidth bandwidth)
-{
- int n;
- WebRtc_Word16 status;
- WebRtc_Word16 realFFT[FRAMESAMPLES_HALF];
- WebRtc_Word16 imagFFT[FRAMESAMPLES_HALF];
-
- // store bit-stream object.
- memcpy(bitStreamObj, &ISACSavedEncObj->bitStreamObj, sizeof(Bitstr));
-
- // Scale FFT coefficients.
- for(n = 0; n < FRAMESAMPLES_HALF; n++)
- {
- realFFT[n] = (WebRtc_Word16)((float)ISACSavedEncObj->realFFT[n] *
- RCU_TRANSCODING_SCALE_UB + 0.5);
- imagFFT[n] = (WebRtc_Word16)((float)ISACSavedEncObj->imagFFT[n] *
- RCU_TRANSCODING_SCALE_UB + 0.5);
- }
-
- switch(bandwidth)
- {
- case isac12kHz:
- {
- status = WebRtcIsac_EncodeSpecUB12(realFFT, imagFFT, bitStreamObj);
- break;
- }
- case isac16kHz:
- {
- status = WebRtcIsac_EncodeSpecUB16(realFFT, imagFFT, bitStreamObj);
- break;
- }
- default:
- return -1;
- }
-
- if(status < 0)
- {
- // error happened
- return status;
- }
- else
- {
- // terminate entropy coding
- return WebRtcIsac_EncTerminate(bitStreamObj);
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode_lpc_swb.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode_lpc_swb.c
deleted file mode 100644
index 2bf4c36..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode_lpc_swb.c
+++ /dev/null
@@ -1,708 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * code_LPC_UB.c
- *
- * This file contains definition of functions used to
- * encode LPC parameters (Shape & gain) of the upper band.
- *
- */
-
-#include "encode_lpc_swb.h"
-#include "typedefs.h"
-#include "settings.h"
-
-#include "lpc_shape_swb12_tables.h"
-#include "lpc_shape_swb16_tables.h"
-#include "lpc_gain_swb_tables.h"
-
-#include <stdio.h>
-#include <string.h>
-#include <math.h>
-
-/******************************************************************************
- * WebRtcIsac_RemoveLarMean()
- *
- * Remove the means from LAR coefficients.
- *
- * Input:
- * -lar : pointer to lar vectors. LAR vectors are
- * concatenated.
- * -bandwidth : indicates if the given LAR vectors belong
- * to SWB-12kHz or SWB-16kHz.
- *
- * Output:
- * -lar : pointer to mean-removed LAR:s.
- *
- *
- */
-WebRtc_Word16
-WebRtcIsac_RemoveLarMean(
- double* lar,
- WebRtc_Word16 bandwidth)
-{
- WebRtc_Word16 coeffCntr;
- WebRtc_Word16 vecCntr;
- WebRtc_Word16 numVec;
- const double* meanLAR;
- switch(bandwidth)
- {
- case isac12kHz:
- {
- numVec = UB_LPC_VEC_PER_FRAME;
- meanLAR = WebRtcIsac_kMeanLarUb12;
- break;
- }
- case isac16kHz:
- {
- numVec = UB16_LPC_VEC_PER_FRAME;
- meanLAR = WebRtcIsac_kMeanLarUb16;
- break;
- }
- default:
- return -1;
- }
-
- for(vecCntr = 0; vecCntr < numVec; vecCntr++)
- {
- for(coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++)
- {
- // REMOVE MEAN
- *lar++ -= meanLAR[coeffCntr];
- }
- }
- return 0;
-}
-
-/******************************************************************************
- * WebRtcIsac_DecorrelateIntraVec()
- *
- * Remove the correlation amonge the components of LAR vectors. If LAR vectors
- * of one frame are put in a matrix where each column is a LAR vector of a
- * sub-frame, then this is equivalent to multiplying the LAR matrix with
- * a decorrelting mtrix from left.
- *
- * Input:
- * -inLar : pointer to mean-removed LAR vecrtors.
- * -bandwidth : indicates if the given LAR vectors belong
- * to SWB-12kHz or SWB-16kHz.
- *
- * Output:
- * -out : decorrelated LAR vectors.
- */
-WebRtc_Word16
-WebRtcIsac_DecorrelateIntraVec(
- const double* data,
- double* out,
- WebRtc_Word16 bandwidth)
-{
- const double* ptrData;
- const double* ptrRow;
- WebRtc_Word16 rowCntr;
- WebRtc_Word16 colCntr;
- WebRtc_Word16 larVecCntr;
- WebRtc_Word16 numVec;
- const double* decorrMat;
- switch(bandwidth)
- {
- case isac12kHz:
- {
- decorrMat = &WebRtcIsac_kIntraVecDecorrMatUb12[0][0];
- numVec = UB_LPC_VEC_PER_FRAME;
- break;
- }
- case isac16kHz:
- {
- decorrMat = &WebRtcIsac_kIintraVecDecorrMatUb16[0][0];
- numVec = UB16_LPC_VEC_PER_FRAME;
- break;
- }
- default:
- return -1;
- }
-
- //
- // decorrMat * data
- //
- // data is assumed to contain 'numVec' of LAR
- // vectors (mean removed) each of dimension 'UB_LPC_ORDER'
- // concatenated one after the other.
- //
-
- ptrData = data;
- for(larVecCntr = 0; larVecCntr < numVec; larVecCntr++)
- {
- for(rowCntr = 0; rowCntr < UB_LPC_ORDER; rowCntr++)
- {
- ptrRow = &decorrMat[rowCntr * UB_LPC_ORDER];
- *out = 0;
- for(colCntr = 0; colCntr < UB_LPC_ORDER; colCntr++)
- {
- *out += ptrData[colCntr] * ptrRow[colCntr];
- }
- out++;
- }
- ptrData += UB_LPC_ORDER;
- }
- return 0;
-}
-
-/******************************************************************************
- * WebRtcIsac_DecorrelateInterVec()
- *
- * Remover the correlation among mean-removed LAR vectors. If LAR vectors
- * of one frame are put in a matrix where each column is a LAR vector of a
- * sub-frame, then this is equivalent to multiplying the LAR matrix with
- * a decorrelting mtrix from right.
- *
- * Input:
- * -data : pointer to matrix of LAR vectors. The matrix
- * is stored column-wise.
- * -bandwidth : indicates if the given LAR vectors belong
- * to SWB-12kHz or SWB-16kHz.
- *
- * Output:
- * -out : decorrelated LAR vectors.
- */
-WebRtc_Word16
-WebRtcIsac_DecorrelateInterVec(
- const double* data,
- double* out,
- WebRtc_Word16 bandwidth)
-{
- WebRtc_Word16 coeffCntr;
- WebRtc_Word16 rowCntr;
- WebRtc_Word16 colCntr;
- const double* decorrMat;
- WebRtc_Word16 interVecDim;
-
- switch(bandwidth)
- {
- case isac12kHz:
- {
- decorrMat = &WebRtcIsac_kInterVecDecorrMatUb12[0][0];
- interVecDim = UB_LPC_VEC_PER_FRAME;
- break;
- }
- case isac16kHz:
- {
- decorrMat = &WebRtcIsac_kInterVecDecorrMatUb16[0][0];
- interVecDim = UB16_LPC_VEC_PER_FRAME;
- break;
- }
- default:
- return -1;
- }
-
- //
- // data * decorrMat
- //
- // data is of size 'interVecDim' * 'UB_LPC_ORDER'
- // That is 'interVecDim' of LAR vectors (mean removed)
- // in columns each of dimension 'UB_LPC_ORDER'.
- // matrix is stored column-wise.
- //
-
- for(coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++)
- {
- for(colCntr = 0; colCntr < interVecDim; colCntr++)
- {
- out[coeffCntr + colCntr * UB_LPC_ORDER] = 0;
- for(rowCntr = 0; rowCntr < interVecDim; rowCntr++)
- {
- out[coeffCntr + colCntr * UB_LPC_ORDER] +=
- data[coeffCntr + rowCntr * UB_LPC_ORDER] *
- decorrMat[rowCntr * interVecDim + colCntr];
- }
- }
- }
- return 0;
-}
-
-/******************************************************************************
- * WebRtcIsac_QuantizeUncorrLar()
- *
- * Quantize the uncorrelated parameters.
- *
- * Input:
- * -data : uncorrelated LAR vectors.
- * -bandwidth : indicates if the given LAR vectors belong
- * to SWB-12kHz or SWB-16kHz.
- *
- * Output:
- * -data : quantized version of the input.
- * -idx : pointer to quantization indices.
- */
-double
-WebRtcIsac_QuantizeUncorrLar(
- double* data,
- int* recIdx,
- WebRtc_Word16 bandwidth)
-{
- WebRtc_Word16 cntr;
- WebRtc_Word32 idx;
- WebRtc_Word16 interVecDim;
- const double* leftRecPoint;
- double quantizationStepSize;
- const WebRtc_Word16* numQuantCell;
- switch(bandwidth)
- {
- case isac12kHz:
- {
- leftRecPoint = WebRtcIsac_kLpcShapeLeftRecPointUb12;
- quantizationStepSize = WebRtcIsac_kLpcShapeQStepSizeUb12;
- numQuantCell = WebRtcIsac_kLpcShapeNumRecPointUb12;
- interVecDim = UB_LPC_VEC_PER_FRAME;
- break;
- }
- case isac16kHz:
- {
- leftRecPoint = WebRtcIsac_kLpcShapeLeftRecPointUb16;
- quantizationStepSize = WebRtcIsac_kLpcShapeQStepSizeUb16;
- numQuantCell = WebRtcIsac_kLpcShapeNumRecPointUb16;
- interVecDim = UB16_LPC_VEC_PER_FRAME;
- break;
- }
- default:
- return -1;
- }
-
- //
- // Quantize the parametrs.
- //
- for(cntr = 0; cntr < UB_LPC_ORDER * interVecDim; cntr++)
- {
- idx = (WebRtc_Word32)floor((*data - leftRecPoint[cntr]) /
- quantizationStepSize + 0.5);
- if(idx < 0)
- {
- idx = 0;
- }
- else if(idx >= numQuantCell[cntr])
- {
- idx = numQuantCell[cntr] - 1;
- }
-
- *data++ = leftRecPoint[cntr] + idx * quantizationStepSize;
- *recIdx++ = idx;
- }
- return 0;
-}
-
-
-/******************************************************************************
- * WebRtcIsac_DequantizeLpcParam()
- *
- * Get the quantized value of uncorrelated LARs given the quantization indices.
- *
- * Input:
- * -idx : pointer to quantiztion indices.
- * -bandwidth : indicates if the given LAR vectors belong
- * to SWB-12kHz or SWB-16kHz.
- *
- * Output:
- * -out : pointer to quantized values.
- */
-WebRtc_Word16
-WebRtcIsac_DequantizeLpcParam(
- const int* idx,
- double* out,
- WebRtc_Word16 bandwidth)
-{
- WebRtc_Word16 cntr;
- WebRtc_Word16 interVecDim;
- const double* leftRecPoint;
- double quantizationStepSize;
-
- switch(bandwidth)
- {
- case isac12kHz:
- {
- leftRecPoint = WebRtcIsac_kLpcShapeLeftRecPointUb12;
- quantizationStepSize = WebRtcIsac_kLpcShapeQStepSizeUb12;
- interVecDim = UB_LPC_VEC_PER_FRAME;
- break;
- }
- case isac16kHz:
- {
- leftRecPoint = WebRtcIsac_kLpcShapeLeftRecPointUb16;
- quantizationStepSize = WebRtcIsac_kLpcShapeQStepSizeUb16;
- interVecDim = UB16_LPC_VEC_PER_FRAME;
- break;
- }
- default:
- return -1;
- }
-
- //
- // Dequantize given the quantization indices
- //
-
- for(cntr = 0; cntr < UB_LPC_ORDER * interVecDim; cntr++)
- {
- *out++ = leftRecPoint[cntr] + *idx++ * quantizationStepSize;
- }
- return 0;
-}
-
-
-/******************************************************************************
- * WebRtcIsac_CorrelateIntraVec()
- *
- * This is the inverse of WebRtcIsac_DecorrelateIntraVec().
- *
- * Input:
- * -data : uncorrelated parameters.
- * -bandwidth : indicates if the given LAR vectors belong
- * to SWB-12kHz or SWB-16kHz.
- *
- * Output:
- * -out : correlated parametrs.
- */
-WebRtc_Word16
-WebRtcIsac_CorrelateIntraVec(
- const double* data,
- double* out,
- WebRtc_Word16 bandwidth)
-{
- WebRtc_Word16 vecCntr;
- WebRtc_Word16 rowCntr;
- WebRtc_Word16 colCntr;
- WebRtc_Word16 numVec;
- const double* ptrData;
- const double* intraVecDecorrMat;
-
- switch(bandwidth)
- {
- case isac12kHz:
- {
- numVec = UB_LPC_VEC_PER_FRAME;
- intraVecDecorrMat = &WebRtcIsac_kIntraVecDecorrMatUb12[0][0];
- break;
- }
- case isac16kHz:
- {
- numVec = UB16_LPC_VEC_PER_FRAME;
- intraVecDecorrMat = &WebRtcIsac_kIintraVecDecorrMatUb16[0][0];
- break;
- }
- default:
- return -1;
- }
-
-
- ptrData = data;
- for(vecCntr = 0; vecCntr < numVec; vecCntr++)
- {
- for(colCntr = 0; colCntr < UB_LPC_ORDER; colCntr++)
- {
- *out = 0;
- for(rowCntr = 0; rowCntr < UB_LPC_ORDER; rowCntr++)
- {
- *out += ptrData[rowCntr] *
- intraVecDecorrMat[rowCntr * UB_LPC_ORDER + colCntr];
- }
- out++;
- }
- ptrData += UB_LPC_ORDER;
- }
- return 0;
-}
-
-/******************************************************************************
- * WebRtcIsac_CorrelateInterVec()
- *
- * This is the inverse of WebRtcIsac_DecorrelateInterVec().
- *
- * Input:
- * -data
- * -bandwidth : indicates if the given LAR vectors belong
- * to SWB-12kHz or SWB-16kHz.
- *
- * Output:
- * -out : correlated parametrs.
- */
-WebRtc_Word16
-WebRtcIsac_CorrelateInterVec(
- const double* data,
- double* out,
- WebRtc_Word16 bandwidth)
-{
- WebRtc_Word16 coeffCntr;
- WebRtc_Word16 rowCntr;
- WebRtc_Word16 colCntr;
- WebRtc_Word16 interVecDim;
- double myVec[UB16_LPC_VEC_PER_FRAME];
- const double* interVecDecorrMat;
-
- switch(bandwidth)
- {
- case isac12kHz:
- {
- interVecDim = UB_LPC_VEC_PER_FRAME;
- interVecDecorrMat = &WebRtcIsac_kInterVecDecorrMatUb12[0][0];
- break;
- }
- case isac16kHz:
- {
- interVecDim = UB16_LPC_VEC_PER_FRAME;
- interVecDecorrMat = &WebRtcIsac_kInterVecDecorrMatUb16[0][0];
- break;
- }
- default:
- return -1;
- }
-
- for(coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++)
- {
- for(rowCntr = 0; rowCntr < interVecDim; rowCntr++)
- {
- myVec[rowCntr] = 0;
- for(colCntr = 0; colCntr < interVecDim; colCntr++)
- {
- myVec[rowCntr] += data[coeffCntr + colCntr * UB_LPC_ORDER] * //*ptrData *
- interVecDecorrMat[rowCntr * interVecDim + colCntr];
- //ptrData += UB_LPC_ORDER;
- }
- }
-
- for(rowCntr = 0; rowCntr < interVecDim; rowCntr++)
- {
- out[coeffCntr + rowCntr * UB_LPC_ORDER] = myVec[rowCntr];
- }
- }
- return 0;
-}
-
-/******************************************************************************
- * WebRtcIsac_AddLarMean()
- *
- * This is the inverse of WebRtcIsac_RemoveLarMean()
- *
- * Input:
- * -data : pointer to mean-removed LAR:s.
- * -bandwidth : indicates if the given LAR vectors belong
- * to SWB-12kHz or SWB-16kHz.
- *
- * Output:
- * -data : pointer to LARs.
- */
-WebRtc_Word16
-WebRtcIsac_AddLarMean(
- double* data,
- WebRtc_Word16 bandwidth)
-{
- WebRtc_Word16 coeffCntr;
- WebRtc_Word16 vecCntr;
- WebRtc_Word16 numVec;
- const double* meanLAR;
-
- switch(bandwidth)
- {
- case isac12kHz:
- {
- numVec = UB_LPC_VEC_PER_FRAME;
- meanLAR = WebRtcIsac_kMeanLarUb12;
- break;
- }
- case isac16kHz:
- {
- numVec = UB16_LPC_VEC_PER_FRAME;
- meanLAR = WebRtcIsac_kMeanLarUb16;
- break;
- }
- default:
- return -1;
- }
-
- for(vecCntr = 0; vecCntr < numVec; vecCntr++)
- {
- for(coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++)
- {
- *data++ += meanLAR[coeffCntr];
- }
- }
- return 0;
-}
-
-/******************************************************************************
- * WebRtcIsac_ToLogDomainRemoveMean()
- *
- * Transform the LPC gain to log domain then remove the mean value.
- *
- * Input:
- * -lpcGain : pointer to LPC Gain, expecting 6 LPC gains
- *
- * Output:
- * -lpcGain : mean-removed in log domain.
- */
-WebRtc_Word16
-WebRtcIsac_ToLogDomainRemoveMean(
- double* data)
-{
- WebRtc_Word16 coeffCntr;
- for(coeffCntr = 0; coeffCntr < UB_LPC_GAIN_DIM; coeffCntr++)
- {
- data[coeffCntr] = log(data[coeffCntr]) - WebRtcIsac_kMeanLpcGain;
- }
- return 0;
-}
-
-
-/******************************************************************************
- * WebRtcIsac_DecorrelateLPGain()
- *
- * Decorrelate LPC gains. There are 6 LPC Gains per frame. This is like
- * multiplying gain vector with decorrelating matrix.
- *
- * Input:
- * -data : LPC gain in log-domain with mean removed.
- *
- * Output:
- * -out : decorrelated parameters.
- */
-WebRtc_Word16 WebRtcIsac_DecorrelateLPGain(
- const double* data,
- double* out)
-{
- WebRtc_Word16 rowCntr;
- WebRtc_Word16 colCntr;
-
- for(colCntr = 0; colCntr < UB_LPC_GAIN_DIM; colCntr++)
- {
- *out = 0;
- for(rowCntr = 0; rowCntr < UB_LPC_GAIN_DIM; rowCntr++)
- {
- *out += data[rowCntr] * WebRtcIsac_kLpcGainDecorrMat[rowCntr][colCntr];
- }
- out++;
- }
- return 0;
-}
-
-/******************************************************************************
- * WebRtcIsac_QuantizeLpcGain()
- *
- * Quantize the decorrelated log-domain gains.
- *
- * Input:
- * -lpcGain : uncorrelated LPC gains.
- *
- * Output:
- * -idx : quantization indices
- * -lpcGain : quantized value of the inpt.
- */
-double WebRtcIsac_QuantizeLpcGain(
- double* data,
- int* idx)
-{
- WebRtc_Word16 coeffCntr;
- for(coeffCntr = 0; coeffCntr < UB_LPC_GAIN_DIM; coeffCntr++)
- {
- *idx = (int)floor((*data - WebRtcIsac_kLeftRecPointLpcGain[coeffCntr]) /
- WebRtcIsac_kQSizeLpcGain + 0.5);
-
- if(*idx < 0)
- {
- *idx = 0;
- }
- else if(*idx >= WebRtcIsac_kNumQCellLpcGain[coeffCntr])
- {
- *idx = WebRtcIsac_kNumQCellLpcGain[coeffCntr] - 1;
- }
- *data = WebRtcIsac_kLeftRecPointLpcGain[coeffCntr] + *idx *
- WebRtcIsac_kQSizeLpcGain;
-
- data++;
- idx++;
- }
- return 0;
-}
-
-/******************************************************************************
- * WebRtcIsac_DequantizeLpcGain()
- *
- * Get the quantized values given the quantization indices.
- *
- * Input:
- * -idx : pointer to quantization indices.
- *
- * Output:
- * -lpcGains : quantized values of the given parametes.
- */
-WebRtc_Word16 WebRtcIsac_DequantizeLpcGain(
- const int* idx,
- double* out)
-{
- WebRtc_Word16 coeffCntr;
- for(coeffCntr = 0; coeffCntr < UB_LPC_GAIN_DIM; coeffCntr++)
- {
- *out = WebRtcIsac_kLeftRecPointLpcGain[coeffCntr] + *idx *
- WebRtcIsac_kQSizeLpcGain;
- out++;
- idx++;
- }
- return 0;
-}
-
-/******************************************************************************
- * WebRtcIsac_CorrelateLpcGain()
- *
- * This is the inverse of WebRtcIsac_DecorrelateLPGain().
- *
- * Input:
- * -data : decorrelated parameters.
- *
- * Output:
- * -out : correlated parameters.
- */
-WebRtc_Word16 WebRtcIsac_CorrelateLpcGain(
- const double* data,
- double* out)
-{
- WebRtc_Word16 rowCntr;
- WebRtc_Word16 colCntr;
-
- for(rowCntr = 0; rowCntr < UB_LPC_GAIN_DIM; rowCntr++)
- {
- *out = 0;
- for(colCntr = 0; colCntr < UB_LPC_GAIN_DIM; colCntr++)
- {
- *out += WebRtcIsac_kLpcGainDecorrMat[rowCntr][colCntr] * data[colCntr];
- }
- out++;
- }
-
- return 0;
-}
-
-
-/******************************************************************************
- * WebRtcIsac_AddMeanToLinearDomain()
- *
- * This is the inverse of WebRtcIsac_ToLogDomainRemoveMean().
- *
- * Input:
- * -lpcGain : LPC gain in log-domain & mean removed
- *
- * Output:
- * -lpcGain : LPC gain in normal domain.
- */
-WebRtc_Word16 WebRtcIsac_AddMeanToLinearDomain(
- double* lpcGains)
-{
- WebRtc_Word16 coeffCntr;
- for(coeffCntr = 0; coeffCntr < UB_LPC_GAIN_DIM; coeffCntr++)
- {
- lpcGains[coeffCntr] = exp(lpcGains[coeffCntr] + WebRtcIsac_kMeanLpcGain);
- }
- return 0;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode_lpc_swb.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode_lpc_swb.h
deleted file mode 100644
index e7f1a76..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/encode_lpc_swb.h
+++ /dev/null
@@ -1,283 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * encode_lpc_swb.h
- *
- * This file contains declaration of functions used to
- * encode LPC parameters (Shape & gain) of the upper band.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_
-
-#include "typedefs.h"
-#include "settings.h"
-#include "structs.h"
-
-
-/******************************************************************************
- * WebRtcIsac_RemoveLarMean()
- *
- * Remove the means from LAR coefficients.
- *
- * Input:
- * -lar : pointer to lar vectors. LAR vectors are
- * concatenated.
- * -bandwidth : indicates if the given LAR vectors belong
- * to SWB-12kHz or SWB-16kHz.
- *
- * Output:
- * -lar : pointer to mean-removed LAR:s.
- *
- *
- */
-WebRtc_Word16 WebRtcIsac_RemoveLarMean(
- double* lar,
- WebRtc_Word16 bandwidth);
-
-/******************************************************************************
- * WebRtcIsac_DecorrelateIntraVec()
- *
- * Remove the correlation amonge the components of LAR vectors. If LAR vectors
- * of one frame are put in a matrix where each column is a LAR vector of a
- * sub-frame, then this is equivalent to multiplying the LAR matrix with
- * a decorrelting mtrix from left.
- *
- * Input:
- * -inLar : pointer to mean-removed LAR vecrtors.
- * -bandwidth : indicates if the given LAR vectors belong
- * to SWB-12kHz or SWB-16kHz.
- *
- * Output:
- * -out : decorrelated LAR vectors.
- */
-WebRtc_Word16 WebRtcIsac_DecorrelateIntraVec(
- const double* inLAR,
- double* out,
- WebRtc_Word16 bandwidth);
-
-
-/******************************************************************************
- * WebRtcIsac_DecorrelateInterVec()
- *
- * Remover the correlation among mean-removed LAR vectors. If LAR vectors
- * of one frame are put in a matrix where each column is a LAR vector of a
- * sub-frame, then this is equivalent to multiplying the LAR matrix with
- * a decorrelting mtrix from right.
- *
- * Input:
- * -data : pointer to matrix of LAR vectors. The matrix
- * is stored column-wise.
- * -bandwidth : indicates if the given LAR vectors belong
- * to SWB-12kHz or SWB-16kHz.
- *
- * Output:
- * -out : decorrelated LAR vectors.
- */
-WebRtc_Word16 WebRtcIsac_DecorrelateInterVec(
- const double* data,
- double* out,
- WebRtc_Word16 bandwidth);
-
-
-/******************************************************************************
- * WebRtcIsac_QuantizeUncorrLar()
- *
- * Quantize the uncorrelated parameters.
- *
- * Input:
- * -data : uncorrelated LAR vectors.
- * -bandwidth : indicates if the given LAR vectors belong
- * to SWB-12kHz or SWB-16kHz.
- *
- * Output:
- * -data : quantized version of the input.
- * -idx : pointer to quantization indices.
- */
-double WebRtcIsac_QuantizeUncorrLar(
- double* data,
- int* idx,
- WebRtc_Word16 bandwidth);
-
-
-/******************************************************************************
- * WebRtcIsac_CorrelateIntraVec()
- *
- * This is the inverse of WebRtcIsac_DecorrelateIntraVec().
- *
- * Input:
- * -data : uncorrelated parameters.
- * -bandwidth : indicates if the given LAR vectors belong
- * to SWB-12kHz or SWB-16kHz.
- *
- * Output:
- * -out : correlated parametrs.
- */
-WebRtc_Word16 WebRtcIsac_CorrelateIntraVec(
- const double* data,
- double* out,
- WebRtc_Word16 bandwidth);
-
-
-/******************************************************************************
- * WebRtcIsac_CorrelateInterVec()
- *
- * This is the inverse of WebRtcIsac_DecorrelateInterVec().
- *
- * Input:
- * -data
- * -bandwidth : indicates if the given LAR vectors belong
- * to SWB-12kHz or SWB-16kHz.
- *
- * Output:
- * -out : correlated parametrs.
- */
-WebRtc_Word16 WebRtcIsac_CorrelateInterVec(
- const double* data,
- double* out,
- WebRtc_Word16 bandwidth);
-
-
-/******************************************************************************
- * WebRtcIsac_AddLarMean()
- *
- * This is the inverse of WebRtcIsac_RemoveLarMean()
- *
- * Input:
- * -data : pointer to mean-removed LAR:s.
- * -bandwidth : indicates if the given LAR vectors belong
- * to SWB-12kHz or SWB-16kHz.
- *
- * Output:
- * -data : pointer to LARs.
- */
-WebRtc_Word16 WebRtcIsac_AddLarMean(
- double* data,
- WebRtc_Word16 bandwidth);
-
-
-/******************************************************************************
- * WebRtcIsac_DequantizeLpcParam()
- *
- * Get the quantized value of uncorrelated LARs given the quantization indices.
- *
- * Input:
- * -idx : pointer to quantiztion indices.
- * -bandwidth : indicates if the given LAR vectors belong
- * to SWB-12kHz or SWB-16kHz.
- *
- * Output:
- * -out : pointer to quantized values.
- */
-WebRtc_Word16 WebRtcIsac_DequantizeLpcParam(
- const int* idx,
- double* out,
- WebRtc_Word16 bandwidth);
-
-
-/******************************************************************************
- * WebRtcIsac_ToLogDomainRemoveMean()
- *
- * Transform the LPC gain to log domain then remove the mean value.
- *
- * Input:
- * -lpcGain : pointer to LPC Gain, expecting 6 LPC gains
- *
- * Output:
- * -lpcGain : mean-removed in log domain.
- */
-WebRtc_Word16 WebRtcIsac_ToLogDomainRemoveMean(
- double* lpGains);
-
-
-/******************************************************************************
- * WebRtcIsac_DecorrelateLPGain()
- *
- * Decorrelate LPC gains. There are 6 LPC Gains per frame. This is like
- * multiplying gain vector with decorrelating matrix.
- *
- * Input:
- * -data : LPC gain in log-domain with mean removed.
- *
- * Output:
- * -out : decorrelated parameters.
- */
-WebRtc_Word16 WebRtcIsac_DecorrelateLPGain(
- const double* data,
- double* out);
-
-
-/******************************************************************************
- * WebRtcIsac_QuantizeLpcGain()
- *
- * Quantize the decorrelated log-domain gains.
- *
- * Input:
- * -lpcGain : uncorrelated LPC gains.
- *
- * Output:
- * -idx : quantization indices
- * -lpcGain : quantized value of the inpt.
- */
-double WebRtcIsac_QuantizeLpcGain(
- double* lpGains,
- int* idx);
-
-
-/******************************************************************************
- * WebRtcIsac_DequantizeLpcGain()
- *
- * Get the quantized values given the quantization indices.
- *
- * Input:
- * -idx : pointer to quantization indices.
- *
- * Output:
- * -lpcGains : quantized values of the given parametes.
- */
-WebRtc_Word16 WebRtcIsac_DequantizeLpcGain(
- const int* idx,
- double* lpGains);
-
-
-/******************************************************************************
- * WebRtcIsac_CorrelateLpcGain()
- *
- * This is the inverse of WebRtcIsac_DecorrelateLPGain().
- *
- * Input:
- * -data : decorrelated parameters.
- *
- * Output:
- * -out : correlated parameters.
- */
-WebRtc_Word16 WebRtcIsac_CorrelateLpcGain(
- const double* data,
- double* out);
-
-
-/******************************************************************************
- * WebRtcIsac_AddMeanToLinearDomain()
- *
- * This is the inverse of WebRtcIsac_ToLogDomainRemoveMean().
- *
- * Input:
- * -lpcGain : LPC gain in log-domain & mean removed
- *
- * Output:
- * -lpcGain : LPC gain in normal domain.
- */
-WebRtc_Word16 WebRtcIsac_AddMeanToLinearDomain(
- double* lpcGains);
-
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENCODE_LPC_SWB_H_
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/entropy_coding.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/entropy_coding.c
deleted file mode 100644
index a729944..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/entropy_coding.c
+++ /dev/null
@@ -1,2748 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * entropy_coding.c
- *
- * This header file defines all of the functions used to arithmetically
- * encode the iSAC bistream
- *
- */
-
-
-#include "entropy_coding.h"
-#include "settings.h"
-#include "arith_routines.h"
-#include "signal_processing_library.h"
-#include "spectrum_ar_model_tables.h"
-#include "lpc_tables.h"
-#include "pitch_gain_tables.h"
-#include "pitch_lag_tables.h"
-#include "encode_lpc_swb.h"
-#include "lpc_shape_swb12_tables.h"
-#include "lpc_shape_swb16_tables.h"
-#include "lpc_gain_swb_tables.h"
-#include "os_specific_inline.h"
-
-#include <math.h>
-#include <string.h>
-
-static const WebRtc_UWord16 kLpcVecPerSegmentUb12 = 5;
-static const WebRtc_UWord16 kLpcVecPerSegmentUb16 = 4;
-
-/* coefficients for the stepwise rate estimation */
-static const WebRtc_Word32 kRPointsQ10[100] = {
- 14495, 14295, 14112, 13944, 13788, 13643, 13459, 13276, 13195, 13239,
- 13243, 13191, 13133, 13216, 13263, 13330, 13316, 13242, 13191, 13106,
- 12942, 12669, 12291, 11840, 11361, 10795, 10192, 9561, 8934, 8335,
- 7750, 7161, 6589, 6062, 5570, 5048, 4548, 4069, 3587, 3143,
- 2717, 2305, 1915, 1557, 1235, 963, 720, 541, 423, 366,
- 369, 435, 561, 750, 1001, 1304, 1626, 1989, 2381, 2793,
- 3219, 3656, 4134, 4612, 5106, 5629, 6122, 6644, 7216, 7801,
- 8386, 8987, 9630, 10255, 10897, 11490, 11950, 12397, 12752, 12999,
- 13175, 13258, 13323, 13290, 13296, 13335, 13113, 13255, 13347, 13355,
- 13298, 13247, 13313, 13155, 13267, 13313, 13374, 13446, 13525, 13609};
-
-
-/* cdf array for encoder bandwidth (12 vs 16 kHz) indicator */
-static const WebRtc_UWord16 kOneBitEqualProbCdf[3] = {
- 0, 32768, 65535 };
-
-/* pointer to cdf array for encoder bandwidth (12 vs 16 kHz) indicator */
-static const WebRtc_UWord16 *kOneBitEqualProbCdf_ptr[1] = {
- kOneBitEqualProbCdf };
-
-/* initial cdf index for decoder of encoded bandwidth (12 vs 16 kHz) indicator */
-static const WebRtc_UWord16 kOneBitEqualProbInitIndex[1] = {1};
-
-
-/* coefficients for the stepwise rate estimation */
-
-
-static const WebRtc_Word32 acnQ10 = 426;
-static const WebRtc_Word32 bcnQ10 = -581224;
-static const WebRtc_Word32 ccnQ10 = 722631;
-static const WebRtc_Word32 lbcnQ10 = -402874;
-#define DPMIN_Q10 -10240 // -10.00 in Q10
-#define DPMAX_Q10 10240 // 10.00 in Q10
-#define MINBITS_Q10 10240 /* 10.0 in Q10 */
-#define IS_SWB_12KHZ 1
-
-__inline WebRtc_UWord32 stepwise(WebRtc_Word32 dinQ10) {
-
- WebRtc_Word32 ind, diQ10, dtQ10;
-
- diQ10 = dinQ10;
- if (diQ10 < DPMIN_Q10)
- diQ10 = DPMIN_Q10;
- if (diQ10 >= DPMAX_Q10)
- diQ10 = DPMAX_Q10 - 1;
-
- dtQ10 = diQ10 - DPMIN_Q10; /* Q10 + Q10 = Q10 */
- ind = (dtQ10 * 5) >> 10; /* 2^10 / 5 = 0.2 in Q10 */
- /* Q10 -> Q0 */
-
- return kRPointsQ10[ind];
-}
-
-
-__inline short log2_Q10_B( int x )
-{
- int zeros;
- short frac;
-
- zeros = WebRtcSpl_NormU32( x );
- frac = ((unsigned int)(x << zeros) & 0x7FFFFFFF) >> 21;
- return (short) (((31 - zeros) << 10) + frac);
-}
-
-
-
-/* compute correlation from power spectrum */
-static void WebRtcIsac_FindCorrelation(WebRtc_Word32 *PSpecQ12, WebRtc_Word32 *CorrQ7)
-{
- WebRtc_Word32 summ[FRAMESAMPLES/8];
- WebRtc_Word32 diff[FRAMESAMPLES/8];
- const WebRtc_Word16 *CS_ptrQ9;
- WebRtc_Word32 sum;
- int k, n;
-
- for (k = 0; k < FRAMESAMPLES/8; k++) {
- summ[k] = (PSpecQ12[k] + PSpecQ12[FRAMESAMPLES_QUARTER-1 - k] + 16) >> 5;
- diff[k] = (PSpecQ12[k] - PSpecQ12[FRAMESAMPLES_QUARTER-1 - k] + 16) >> 5;
- }
-
- sum = 2;
- for (n = 0; n < FRAMESAMPLES/8; n++)
- sum += summ[n];
- CorrQ7[0] = sum;
-
- for (k = 0; k < AR_ORDER; k += 2) {
- sum = 0;
- CS_ptrQ9 = WebRtcIsac_kCos[k];
- for (n = 0; n < FRAMESAMPLES/8; n++)
- sum += (CS_ptrQ9[n] * diff[n] + 256) >> 9;
- CorrQ7[k+1] = sum;
- }
-
- for (k=1; k<AR_ORDER; k+=2) {
- sum = 0;
- CS_ptrQ9 = WebRtcIsac_kCos[k];
- for (n = 0; n < FRAMESAMPLES/8; n++)
- sum += (CS_ptrQ9[n] * summ[n] + 256) >> 9;
- CorrQ7[k+1] = sum;
- }
-}
-
-/* compute inverse AR power spectrum */
-/* Changed to the function used in iSAC FIX for compatibility reasons */
-static void WebRtcIsac_FindInvArSpec(const WebRtc_Word16 *ARCoefQ12,
- const WebRtc_Word32 gainQ10,
- WebRtc_Word32 *CurveQ16)
-{
- WebRtc_Word32 CorrQ11[AR_ORDER+1];
- WebRtc_Word32 sum, tmpGain;
- WebRtc_Word32 diffQ16[FRAMESAMPLES/8];
- const WebRtc_Word16 *CS_ptrQ9;
- int k, n;
- WebRtc_Word16 round, shftVal = 0, sh;
-
- sum = 0;
- for (n = 0; n < AR_ORDER+1; n++)
- sum += WEBRTC_SPL_MUL(ARCoefQ12[n], ARCoefQ12[n]); /* Q24 */
- sum = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(sum, 6), 65) + 32768, 16); /* result in Q8 */
- CorrQ11[0] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, gainQ10) + 256, 9);
-
- /* To avoid overflow, we shift down gainQ10 if it is large. We will not lose any precision */
- if(gainQ10>400000){
- tmpGain = WEBRTC_SPL_RSHIFT_W32(gainQ10, 3);
- round = 32;
- shftVal = 6;
- } else {
- tmpGain = gainQ10;
- round = 256;
- shftVal = 9;
- }
-
- for (k = 1; k < AR_ORDER+1; k++) {
- sum = 16384;
- for (n = k; n < AR_ORDER+1; n++)
- sum += WEBRTC_SPL_MUL(ARCoefQ12[n-k], ARCoefQ12[n]); /* Q24 */
- sum = WEBRTC_SPL_RSHIFT_W32(sum, 15);
- CorrQ11[k] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(sum, tmpGain) + round, shftVal);
- }
- sum = WEBRTC_SPL_LSHIFT_W32(CorrQ11[0], 7);
- for (n = 0; n < FRAMESAMPLES/8; n++)
- CurveQ16[n] = sum;
-
- for (k = 1; k < AR_ORDER; k += 2) {
- //CS_ptrQ9 = WebRtcIsac_kCos[k];
- for (n = 0; n < FRAMESAMPLES/8; n++)
- CurveQ16[n] += WEBRTC_SPL_RSHIFT_W32(
- WEBRTC_SPL_MUL(WebRtcIsac_kCos[k][n], CorrQ11[k+1]) + 2, 2);
- }
-
- CS_ptrQ9 = WebRtcIsac_kCos[0];
-
- /* If CorrQ11[1] too large we avoid getting overflow in the calculation by shifting */
- sh=WebRtcSpl_NormW32(CorrQ11[1]);
- if (CorrQ11[1]==0) /* Use next correlation */
- sh=WebRtcSpl_NormW32(CorrQ11[2]);
-
- if (sh<9)
- shftVal = 9 - sh;
- else
- shftVal = 0;
-
- for (n = 0; n < FRAMESAMPLES/8; n++)
- diffQ16[n] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CS_ptrQ9[n], WEBRTC_SPL_RSHIFT_W32(CorrQ11[1], shftVal)) + 2, 2);
- for (k = 2; k < AR_ORDER; k += 2) {
- CS_ptrQ9 = WebRtcIsac_kCos[k];
- for (n = 0; n < FRAMESAMPLES/8; n++)
- diffQ16[n] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(CS_ptrQ9[n], WEBRTC_SPL_RSHIFT_W32(CorrQ11[k+1], shftVal)) + 2, 2);
- }
-
- for (k=0; k<FRAMESAMPLES/8; k++) {
- CurveQ16[FRAMESAMPLES_QUARTER-1 - k] = CurveQ16[k] - WEBRTC_SPL_LSHIFT_W32(diffQ16[k], shftVal);
- CurveQ16[k] += WEBRTC_SPL_LSHIFT_W32(diffQ16[k], shftVal);
- }
-}
-
-/* generate array of dither samples in Q7 */
-static void GenerateDitherQ7Lb(WebRtc_Word16 *bufQ7,
- WebRtc_UWord32 seed,
- int length,
- WebRtc_Word16 AvgPitchGain_Q12)
-{
- int k, shft;
- WebRtc_Word16 dither1_Q7, dither2_Q7, dither_gain_Q14;
-
- if (AvgPitchGain_Q12 < 614) /* this threshold should be equal to that in decode_spec() */
- {
- for (k = 0; k < length-2; k += 3)
- {
- /* new random unsigned int */
- seed = (seed * 196314165) + 907633515;
-
- /* fixed-point dither sample between -64 and 64 (Q7) */
- dither1_Q7 = (WebRtc_Word16)(((int)seed + 16777216)>>25); // * 128/4294967295
-
- /* new random unsigned int */
- seed = (seed * 196314165) + 907633515;
-
- /* fixed-point dither sample between -64 and 64 */
- dither2_Q7 = (WebRtc_Word16)(((int)seed + 16777216)>>25);
-
- shft = (seed >> 25) & 15;
- if (shft < 5)
- {
- bufQ7[k] = dither1_Q7;
- bufQ7[k+1] = dither2_Q7;
- bufQ7[k+2] = 0;
- }
- else if (shft < 10)
- {
- bufQ7[k] = dither1_Q7;
- bufQ7[k+1] = 0;
- bufQ7[k+2] = dither2_Q7;
- }
- else
- {
- bufQ7[k] = 0;
- bufQ7[k+1] = dither1_Q7;
- bufQ7[k+2] = dither2_Q7;
- }
- }
- }
- else
- {
- dither_gain_Q14 = (WebRtc_Word16)(22528 - 10 * AvgPitchGain_Q12);
-
- /* dither on half of the coefficients */
- for (k = 0; k < length-1; k += 2)
- {
- /* new random unsigned int */
- seed = (seed * 196314165) + 907633515;
-
- /* fixed-point dither sample between -64 and 64 */
- dither1_Q7 = (WebRtc_Word16)(((int)seed + 16777216)>>25);
-
- /* dither sample is placed in either even or odd index */
- shft = (seed >> 25) & 1; /* either 0 or 1 */
-
- bufQ7[k + shft] = (((dither_gain_Q14 * dither1_Q7) + 8192)>>14);
- bufQ7[k + 1 - shft] = 0;
- }
- }
-}
-
-
-
-/******************************************************************************
- * GenerateDitherQ7LbUB()
- *
- * generate array of dither samples in Q7 There are less zeros in dither
- * vector compared to GenerateDitherQ7Lb.
- *
- * A uniform random number generator with the range of [-64 64] is employed
- * but the generated dithers are scaled by 0.35, a heuristic scaling.
- *
- * Input:
- * -seed : the initial seed for the random number generator.
- * -length : the number of dither values to be generated.
- *
- * Output:
- * -bufQ7 : pointer to a buffer where dithers are written to.
- */
-static void GenerateDitherQ7LbUB(
- WebRtc_Word16 *bufQ7,
- WebRtc_UWord32 seed,
- int length)
-{
- int k;
- for (k = 0; k < length; k++) {
- /* new random unsigned int */
- seed = (seed * 196314165) + 907633515;
-
- /* fixed-point dither sample between -64 and 64 (Q7) */
- // * 128/4294967295
- bufQ7[k] = (WebRtc_Word16)(((int)seed + 16777216)>>25);
-
- // scale by 0.35
- bufQ7[k] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(bufQ7[k],
- 2048, 13);
- }
-}
-
-
-
-/*
- * function to decode the complex spectrum from the bit stream
- * returns the total number of bytes in the stream
- */
-int WebRtcIsac_DecodeSpecLb(Bitstr *streamdata,
- double *fr,
- double *fi,
- WebRtc_Word16 AvgPitchGain_Q12)
-{
- WebRtc_Word16 DitherQ7[FRAMESAMPLES];
- WebRtc_Word16 data[FRAMESAMPLES];
- WebRtc_Word32 invARSpec2_Q16[FRAMESAMPLES_QUARTER];
- WebRtc_UWord16 invARSpecQ8[FRAMESAMPLES_QUARTER];
- WebRtc_Word16 ARCoefQ12[AR_ORDER+1];
- WebRtc_Word16 RCQ15[AR_ORDER];
- WebRtc_Word16 gainQ10;
- WebRtc_Word32 gain2_Q10, res;
- WebRtc_Word32 in_sqrt;
- WebRtc_Word32 newRes;
- int k, len, i;
-
- /* create dither signal */
- GenerateDitherQ7Lb(DitherQ7, streamdata->W_upper, FRAMESAMPLES, AvgPitchGain_Q12);
-
- /* decode model parameters */
- if (WebRtcIsac_DecodeRc(streamdata, RCQ15) < 0)
- return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
-
- WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12);
-
- if (WebRtcIsac_DecodeGain2(streamdata, &gain2_Q10) < 0)
- return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
-
- /* compute inverse AR power spectrum */
- WebRtcIsac_FindInvArSpec(ARCoefQ12, gain2_Q10, invARSpec2_Q16);
-
- /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */
- res = 1 << (WebRtcSpl_GetSizeInBits(invARSpec2_Q16[0]) >> 1);
- for (k = 0; k < FRAMESAMPLES_QUARTER; k++)
- {
- in_sqrt = invARSpec2_Q16[k];
- i = 10;
-
- /* Negative values make no sense for a real sqrt-function. */
- if (in_sqrt<0)
- in_sqrt=-in_sqrt;
-
- newRes = (in_sqrt / res + res) >> 1;
- do
- {
- res = newRes;
- newRes = (in_sqrt / res + res) >> 1;
- } while (newRes != res && i-- > 0);
-
- invARSpecQ8[k] = (WebRtc_Word16)newRes;
- }
-
- /* arithmetic decoding of spectrum */
- if ((len = WebRtcIsac_DecLogisticMulti2(data, streamdata, invARSpecQ8, DitherQ7,
- FRAMESAMPLES, !IS_SWB_12KHZ)) <1)
- return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
-
- /* subtract dither and scale down spectral samples with low SNR */
- if (AvgPitchGain_Q12 <= 614)
- {
- for (k = 0; k < FRAMESAMPLES; k += 4)
- {
- gainQ10 = WebRtcSpl_DivW32W16ResW16(30 << 10,
- (WebRtc_Word16)((invARSpec2_Q16[k>>2] + (32768 + (33 << 16))) >> 16));
- *fr++ = (double)((data[ k ] * gainQ10 + 512) >> 10) / 128.0;
- *fi++ = (double)((data[k+1] * gainQ10 + 512) >> 10) / 128.0;
- *fr++ = (double)((data[k+2] * gainQ10 + 512) >> 10) / 128.0;
- *fi++ = (double)((data[k+3] * gainQ10 + 512) >> 10) / 128.0;
- }
- }
- else
- {
- for (k = 0; k < FRAMESAMPLES; k += 4)
- {
- gainQ10 = WebRtcSpl_DivW32W16ResW16(36 << 10,
- (WebRtc_Word16)((invARSpec2_Q16[k>>2] + (32768 + (40 << 16))) >> 16));
- *fr++ = (double)((data[ k ] * gainQ10 + 512) >> 10) / 128.0;
- *fi++ = (double)((data[k+1] * gainQ10 + 512) >> 10) / 128.0;
- *fr++ = (double)((data[k+2] * gainQ10 + 512) >> 10) / 128.0;
- *fi++ = (double)((data[k+3] * gainQ10 + 512) >> 10) / 128.0;
- }
- }
-
- return len;
-}
-
-/******************************************************************************
- * WebRtcIsac_DecodeSpecUB16()
- * Decode real and imaginary part of the DFT coefficients, given a bit-stream.
- * This function is called when the codec is in 0-16 kHz bandwidth.
- * The decoded DFT coefficient can be transformed to time domain by
- * WebRtcIsac_Time2Spec().
- *
- * Input:
- * - streamdata : pointer to a stucture containg the encoded
- * data and theparameters needed for entropy
- * coding.
- *
- * Output:
- * -*fr : pointer to a buffer where the real part of DFT
- * coefficients are written to.
- * -*fi : pointer to a buffer where the imaginary part
- * of DFT coefficients are written to.
- *
- * Return value : < 0 if an error occures
- * 0 if succeeded.
- */
-int WebRtcIsac_DecodeSpecUB16(
- Bitstr* streamdata,
- double* fr,
- double* fi)
-{
- WebRtc_Word16 DitherQ7[FRAMESAMPLES];
- WebRtc_Word16 data[FRAMESAMPLES];
- WebRtc_Word32 invARSpec2_Q16[FRAMESAMPLES_QUARTER];
- WebRtc_UWord16 invARSpecQ8[FRAMESAMPLES_QUARTER];
- WebRtc_Word16 ARCoefQ12[AR_ORDER+1];
- WebRtc_Word16 RCQ15[AR_ORDER];
- WebRtc_Word32 gain2_Q10, res;
- WebRtc_Word32 in_sqrt;
- WebRtc_Word32 newRes;
- int k, len, i, j;
-
- /* create dither signal */
- GenerateDitherQ7LbUB(DitherQ7, streamdata->W_upper, FRAMESAMPLES);
-
- /* decode model parameters */
- if (WebRtcIsac_DecodeRc(streamdata, RCQ15) < 0)
- return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
-
- WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12);
-
- if (WebRtcIsac_DecodeGain2(streamdata, &gain2_Q10) < 0)
- return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
-
- /* compute inverse AR power spectrum */
- WebRtcIsac_FindInvArSpec(ARCoefQ12, gain2_Q10, invARSpec2_Q16);
-
- /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */
- res = 1 << (WebRtcSpl_GetSizeInBits(invARSpec2_Q16[0]) >> 1);
- for (k = 0; k < FRAMESAMPLES_QUARTER; k++)
- {
- in_sqrt = invARSpec2_Q16[k];
- i = 10;
-
- /* Negative values make no sense for a real sqrt-function. */
- if (in_sqrt<0)
- in_sqrt=-in_sqrt;
-
- newRes = (in_sqrt / res + res) >> 1;
- do
- {
- res = newRes;
- newRes = (in_sqrt / res + res) >> 1;
- } while (newRes != res && i-- > 0);
-
- invARSpecQ8[k] = (WebRtc_Word16)newRes;
- }
-
- /* arithmetic decoding of spectrum */
- if ((len = WebRtcIsac_DecLogisticMulti2(data, streamdata, invARSpecQ8,
- DitherQ7, FRAMESAMPLES, !IS_SWB_12KHZ)) <1)
- return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
-
- /* re-arrange DFT coefficients and scale down */
- for (j = 0, k = 0; k < FRAMESAMPLES; k += 4, j++)
- {
- fr[j] = (double)data[ k ] / 128.0;
- fi[j] = (double)data[k+1] / 128.0;
- fr[(FRAMESAMPLES_HALF) - 1 - j] = (double)data[k+2] / 128.0;
- fi[(FRAMESAMPLES_HALF) - 1 - j] = (double)data[k+3] / 128.0;
-
- }
- return len;
-}
-
-
-
-
-/******************************************************************************
- * WebRtcIsac_DecodeSpecUB12()
- * Decode real and imaginary part of the DFT coefficients, given a bit-stream.
- * This function is called when the codec is in 0-12 kHz bandwidth.
- * The decoded DFT coefficient can be transformed to time domain by
- * WebRtcIsac_Time2Spec().
- *
- * Input:
- * - streamdata : pointer to a stucture containg the encoded
- * data and theparameters needed for entropy
- * coding.
- *
- * Output:
- * -*fr : pointer to a buffer where the real part of DFT
- * coefficients are written to.
- * -*fi : pointer to a buffer where the imaginary part
- * of DFT coefficients are written to.
- *
- * Return value : < 0 if an error occures
- * 0 if succeeded.
- */
-int WebRtcIsac_DecodeSpecUB12(
- Bitstr *streamdata,
- double *fr,
- double *fi)
-{
- WebRtc_Word16 DitherQ7[FRAMESAMPLES];
- WebRtc_Word16 data[FRAMESAMPLES];
- WebRtc_Word32 invARSpec2_Q16[FRAMESAMPLES_QUARTER];
- WebRtc_UWord16 invARSpecQ8[FRAMESAMPLES_QUARTER];
- WebRtc_Word16 ARCoefQ12[AR_ORDER+1];
- WebRtc_Word16 RCQ15[AR_ORDER];
- WebRtc_Word32 gain2_Q10;
- WebRtc_Word32 res;
- WebRtc_Word32 in_sqrt;
- WebRtc_Word32 newRes;
- int k, len, i;
-
- /* create dither signal */
- GenerateDitherQ7LbUB(DitherQ7, streamdata->W_upper, FRAMESAMPLES);
-
- /* decode model parameters */
- if (WebRtcIsac_DecodeRc(streamdata, RCQ15) < 0)
- return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
-
- WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12);
-
- if (WebRtcIsac_DecodeGain2(streamdata, &gain2_Q10) < 0)
- return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
-
-
- /* compute inverse AR power spectrum */
- WebRtcIsac_FindInvArSpec(ARCoefQ12, gain2_Q10, invARSpec2_Q16);
-
- /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */
- res = 1 << (WebRtcSpl_GetSizeInBits(invARSpec2_Q16[0]) >> 1);
- for (k = 0; k < FRAMESAMPLES_QUARTER; k++)
- {
- in_sqrt = invARSpec2_Q16[k];
- i = 10;
-
- /* Negative values make no sense for a real sqrt-function. */
- if (in_sqrt<0)
- in_sqrt=-in_sqrt;
-
- newRes = (in_sqrt / res + res) >> 1;
- do
- {
- res = newRes;
- newRes = (in_sqrt / res + res) >> 1;
- } while (newRes != res && i-- > 0);
-
- invARSpecQ8[k] = (WebRtc_Word16)newRes;
- }
-
- /* arithmetic decoding of spectrum */
- if ((len = WebRtcIsac_DecLogisticMulti2(data, streamdata,
- invARSpecQ8, DitherQ7, (FRAMESAMPLES_HALF), IS_SWB_12KHZ)) < 1)
- {
- return -ISAC_RANGE_ERROR_DECODE_SPECTRUM;
- }
-
- for (k = 0, i = 0; k < FRAMESAMPLES_HALF; k += 4)
- {
- fr[i] = (double)data[ k ] / 128.0;
- fi[i] = (double)data[k+1] / 128.0;
- i++;
- fr[i] = (double)data[k+2] / 128.0;
- fi[i] = (double)data[k+3] / 128.0;
- i++;
- }
-
- // The second half of real and imaginary coefficients is zero. This is
- // due to using the old FFT module which requires two signals as input
- // while in 0-12 kHz mode we only have 8-12 kHz band, and the second signal
- // is set to zero
- memset(&fr[FRAMESAMPLES_QUARTER], 0, FRAMESAMPLES_QUARTER * sizeof(double));
- memset(&fi[FRAMESAMPLES_QUARTER], 0, FRAMESAMPLES_QUARTER * sizeof(double));
-
- return len;
-}
-
-
-
-
-
-int WebRtcIsac_EncodeSpecLb(const WebRtc_Word16 *fr,
- const WebRtc_Word16 *fi,
- Bitstr *streamdata,
- WebRtc_Word16 AvgPitchGain_Q12)
-{
- WebRtc_Word16 ditherQ7[FRAMESAMPLES];
- WebRtc_Word16 dataQ7[FRAMESAMPLES];
- WebRtc_Word32 PSpec[FRAMESAMPLES_QUARTER];
- WebRtc_Word32 invARSpec2_Q16[FRAMESAMPLES_QUARTER];
- WebRtc_UWord16 invARSpecQ8[FRAMESAMPLES_QUARTER];
- WebRtc_Word32 CorrQ7[AR_ORDER+1];
- WebRtc_Word32 CorrQ7_norm[AR_ORDER+1];
- WebRtc_Word16 RCQ15[AR_ORDER];
- WebRtc_Word16 ARCoefQ12[AR_ORDER+1];
- WebRtc_Word32 gain2_Q10;
- WebRtc_Word16 val;
- WebRtc_Word32 nrg, res;
- WebRtc_UWord32 sum;
- WebRtc_Word32 in_sqrt;
- WebRtc_Word32 newRes;
- WebRtc_Word16 err;
- WebRtc_UWord32 nrg_u32;
- int shift_var;
- int k, n, j, i;
-
-
- /* create dither_float signal */
- GenerateDitherQ7Lb(ditherQ7, streamdata->W_upper, FRAMESAMPLES, AvgPitchGain_Q12);
-
- /* add dither and quantize, and compute power spectrum */
- for (k = 0; k < FRAMESAMPLES; k += 4)
- {
- val = ((*fr++ + ditherQ7[k] + 64) & 0xFF80) - ditherQ7[k];
- dataQ7[k] = val;
- sum = val * val;
-
- val = ((*fi++ + ditherQ7[k+1] + 64) & 0xFF80) - ditherQ7[k+1];
- dataQ7[k+1] = val;
- sum += val * val;
-
- val = ((*fr++ + ditherQ7[k+2] + 64) & 0xFF80) - ditherQ7[k+2];
- dataQ7[k+2] = val;
- sum += val * val;
-
- val = ((*fi++ + ditherQ7[k+3] + 64) & 0xFF80) - ditherQ7[k+3];
- dataQ7[k+3] = val;
- sum += val * val;
-
- PSpec[k>>2] = sum >> 2;
- }
-
- /* compute correlation from power spectrum */
- WebRtcIsac_FindCorrelation(PSpec, CorrQ7);
-
-
- /* find AR coefficients */
- /* number of bit shifts to 14-bit normalize CorrQ7[0] (leaving room for sign) */
- shift_var = WebRtcSpl_NormW32(CorrQ7[0]) - 18;
-
- if (shift_var > 0) {
- for (k=0; k<AR_ORDER+1; k++) {
- CorrQ7_norm[k] = CorrQ7[k] << shift_var;
- }
- } else {
- for (k=0; k<AR_ORDER+1; k++) {
- CorrQ7_norm[k] = CorrQ7[k] >> (-shift_var);
- }
- }
-
- /* find RC coefficients */
- WebRtcSpl_AutoCorrToReflCoef(CorrQ7_norm, AR_ORDER, RCQ15);
-
- /* quantize & code RC Coefficient */
- WebRtcIsac_EncodeRc(RCQ15, streamdata);
-
- /* RC -> AR coefficients */
- WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12);
-
- /* compute ARCoef' * Corr * ARCoef in Q19 */
- nrg = 0;
- for (j = 0; j <= AR_ORDER; j++) {
- for (n = 0; n <= j; n++) {
- nrg += ( ARCoefQ12[j] * ((CorrQ7_norm[j-n] * ARCoefQ12[n] + 256) >> 9) + 4 ) >> 3;
- }
- for (n = j+1; n <= AR_ORDER; n++) {
- nrg += ( ARCoefQ12[j] * ((CorrQ7_norm[n-j] * ARCoefQ12[n] + 256) >> 9) + 4 ) >> 3;
- }
- }
-
- nrg_u32 = (WebRtc_UWord32)nrg;
- if (shift_var > 0) {
- nrg_u32 = nrg_u32 >> shift_var;
- } else {
- nrg_u32 = nrg_u32 << (-shift_var);
- }
-
- if (nrg_u32 > 0x7FFFFFFF)
- nrg = 0x7FFFFFFF;
- else
- nrg = (WebRtc_Word32)nrg_u32;
-
- gain2_Q10 = WebRtcSpl_DivResultInQ31(FRAMESAMPLES_QUARTER, nrg); /* also shifts 31 bits to the left! */
-
- /* quantize & code gain2_Q10 */
- if (WebRtcIsac_EncodeGain2(&gain2_Q10, streamdata)) {
- return -1;
- }
-
- /* compute inverse AR power spectrum */
- WebRtcIsac_FindInvArSpec(ARCoefQ12, gain2_Q10, invARSpec2_Q16);
-
- /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */
- res = 1 << (WebRtcSpl_GetSizeInBits(invARSpec2_Q16[0]) >> 1);
- for (k = 0; k < FRAMESAMPLES_QUARTER; k++)
- {
- in_sqrt = invARSpec2_Q16[k];
- i = 10;
-
- /* Negative values make no sense for a real sqrt-function. */
- if (in_sqrt<0)
- in_sqrt=-in_sqrt;
-
- newRes = (in_sqrt / res + res) >> 1;
- do
- {
- res = newRes;
- newRes = (in_sqrt / res + res) >> 1;
- } while (newRes != res && i-- > 0);
-
- invARSpecQ8[k] = (WebRtc_Word16)newRes;
- }
-
- /* arithmetic coding of spectrum */
- err = WebRtcIsac_EncLogisticMulti2(streamdata, dataQ7, invARSpecQ8,
- FRAMESAMPLES, !IS_SWB_12KHZ);
- if (err < 0)
- {
- return (err);
- }
-
- return 0;
-}
-
-
-/******************************************************************************
- * WebRtcIsac_EncodeSpecUB16()
- * Quantize and encode real and imaginary part of the DFT coefficients.
- * This function is called when the codec is in 0-16 kHz bandwidth.
- * The real and imaginary part are computed by calling WebRtcIsac_Time2Spec().
- *
- *
- * Input:
- * -*fr : pointer to a buffer where the real part of DFT
- * coefficients are stored.
- * -*fi : pointer to a buffer where the imaginary part
- * of DFT coefficients are stored.
- *
- * Output:
- * - streamdata : pointer to a stucture containg the encoded
- * data and theparameters needed for entropy
- * coding.
- *
- * Return value : < 0 if an error occures
- * 0 if succeeded.
- */
-int WebRtcIsac_EncodeSpecUB16(
- const WebRtc_Word16* fr,
- const WebRtc_Word16* fi,
- Bitstr* streamdata)
-{
- WebRtc_Word16 ditherQ7[FRAMESAMPLES];
- WebRtc_Word16 dataQ7[FRAMESAMPLES];
- WebRtc_Word32 PSpec[FRAMESAMPLES_QUARTER];
- WebRtc_Word32 invARSpec2_Q16[FRAMESAMPLES_QUARTER];
- WebRtc_UWord16 invARSpecQ8[FRAMESAMPLES_QUARTER];
- WebRtc_Word32 CorrQ7[AR_ORDER+1];
- WebRtc_Word32 CorrQ7_norm[AR_ORDER+1];
- WebRtc_Word16 RCQ15[AR_ORDER];
- WebRtc_Word16 ARCoefQ12[AR_ORDER+1];
- WebRtc_Word32 gain2_Q10;
- WebRtc_Word16 val;
- WebRtc_Word32 nrg, res;
- WebRtc_UWord32 sum;
- WebRtc_Word32 in_sqrt;
- WebRtc_Word32 newRes;
- WebRtc_Word16 err;
- WebRtc_UWord32 nrg_u32;
- int shift_var;
- int k, n, j, i;
-
- /* create dither_float signal */
- GenerateDitherQ7LbUB(ditherQ7, streamdata->W_upper, FRAMESAMPLES);
-
- /* add dither and quantize, and compute power spectrum */
- for (j = 0, k = 0; k < FRAMESAMPLES; k += 4, j++)
- {
- val = ((fr[j] + ditherQ7[k] + 64) & 0xFF80) - ditherQ7[k];
- dataQ7[k] = val;
- sum = val * val;
-
- val = ((fi[j] + ditherQ7[k+1] + 64) & 0xFF80) - ditherQ7[k+1];
- dataQ7[k+1] = val;
- sum += val * val;
-
- val = ((fr[(FRAMESAMPLES_HALF) - 1 - j] + ditherQ7[k+2] + 64) &
- 0xFF80) - ditherQ7[k+2];
- dataQ7[k+2] = val;
- sum += val * val;
-
- val = ((fi[(FRAMESAMPLES_HALF) - 1 - j] + ditherQ7[k+3] + 64) &
- 0xFF80) - ditherQ7[k+3];
- dataQ7[k+3] = val;
- sum += val * val;
-
- PSpec[k>>2] = sum >> 2;
- }
-
- /* compute correlation from power spectrum */
- WebRtcIsac_FindCorrelation(PSpec, CorrQ7);
-
-
- /* find AR coefficients
- number of bit shifts to 14-bit normalize CorrQ7[0]
- (leaving room for sign) */
- shift_var = WebRtcSpl_NormW32(CorrQ7[0]) - 18;
-
- if (shift_var > 0) {
- for (k=0; k<AR_ORDER+1; k++) {
- CorrQ7_norm[k] = CorrQ7[k] << shift_var;
- }
- } else {
- for (k=0; k<AR_ORDER+1; k++) {
- CorrQ7_norm[k] = CorrQ7[k] >> (-shift_var);
- }
- }
-
- /* find RC coefficients */
- WebRtcSpl_AutoCorrToReflCoef(CorrQ7_norm, AR_ORDER, RCQ15);
-
- /* quantize & code RC Coef */
- WebRtcIsac_EncodeRc(RCQ15, streamdata);
-
- /* RC -> AR coefficients */
- WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12);
-
- /* compute ARCoef' * Corr * ARCoef in Q19 */
- nrg = 0;
- for (j = 0; j <= AR_ORDER; j++) {
- for (n = 0; n <= j; n++) {
- nrg += ( ARCoefQ12[j] * ((CorrQ7_norm[j-n] * ARCoefQ12[n] +
- 256) >> 9) + 4 ) >> 3;
- }
- for (n = j+1; n <= AR_ORDER; n++) {
- nrg += ( ARCoefQ12[j] * ((CorrQ7_norm[n-j] * ARCoefQ12[n] +
- 256) >> 9) + 4 ) >> 3;
- }
- }
- nrg_u32 = (WebRtc_UWord32)nrg;
- if (shift_var > 0) {
- nrg_u32 = nrg_u32 >> shift_var;
- } else {
- nrg_u32 = nrg_u32 << (-shift_var);
- }
-
- if (nrg_u32 > 0x7FFFFFFF)
- nrg = 0x7FFFFFFF;
- else
- nrg = (WebRtc_Word32)nrg_u32;
-
- gain2_Q10 = WebRtcSpl_DivResultInQ31(FRAMESAMPLES_QUARTER, nrg); /* also shifts 31 bits to the left! */
-
- /* quantize & code gain2_Q10 */
- if (WebRtcIsac_EncodeGain2(&gain2_Q10, streamdata)) {
- return -1;
- }
-
- /* compute inverse AR power spectrum */
- WebRtcIsac_FindInvArSpec(ARCoefQ12, gain2_Q10, invARSpec2_Q16);
-
- /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */
- res = 1 << (WebRtcSpl_GetSizeInBits(invARSpec2_Q16[0]) >> 1);
- for (k = 0; k < FRAMESAMPLES_QUARTER; k++)
- {
- in_sqrt = invARSpec2_Q16[k];
- i = 10;
-
- /* Negative values make no sense for a real sqrt-function. */
- if (in_sqrt<0)
- in_sqrt=-in_sqrt;
-
- newRes = (in_sqrt / res + res) >> 1;
- do
- {
- res = newRes;
- newRes = (in_sqrt / res + res) >> 1;
- } while (newRes != res && i-- > 0);
-
- invARSpecQ8[k] = (WebRtc_Word16)newRes;
- }
-
- /* arithmetic coding of spectrum */
- err = WebRtcIsac_EncLogisticMulti2(streamdata, dataQ7, invARSpecQ8,
- FRAMESAMPLES, !IS_SWB_12KHZ);
- if (err < 0)
- {
- return (err);
- }
-
- return 0;
-}
-
-
-
-
-int WebRtcIsac_EncodeSpecUB12(const WebRtc_Word16 *fr,
- const WebRtc_Word16 *fi,
- Bitstr *streamdata)
-{
- WebRtc_Word16 ditherQ7[FRAMESAMPLES];
- WebRtc_Word16 dataQ7[FRAMESAMPLES];
- WebRtc_Word32 PSpec[FRAMESAMPLES_QUARTER];
- WebRtc_Word32 invARSpec2_Q16[FRAMESAMPLES_QUARTER];
- WebRtc_UWord16 invARSpecQ8[FRAMESAMPLES_QUARTER];
- WebRtc_Word32 CorrQ7[AR_ORDER+1];
- WebRtc_Word32 CorrQ7_norm[AR_ORDER+1];
- WebRtc_Word16 RCQ15[AR_ORDER];
- WebRtc_Word16 ARCoefQ12[AR_ORDER+1];
- WebRtc_Word32 gain2_Q10;
- WebRtc_Word16 val;
- WebRtc_Word32 nrg, res;
- WebRtc_UWord32 sum;
- WebRtc_Word32 in_sqrt;
- WebRtc_Word32 newRes;
- WebRtc_Word16 err;
- int shift_var;
- int k, n, j, i;
- WebRtc_UWord32 nrg_u32;
-
- /* create dither_float signal */
- GenerateDitherQ7LbUB(ditherQ7, streamdata->W_upper, FRAMESAMPLES);
-
- /* add dither and quantize, and compute power spectrum */
- for (k = 0, j = 0; k < (FRAMESAMPLES_HALF); k += 4)
- {
- val = ((*fr++ + ditherQ7[k] + 64) & 0xFF80) - ditherQ7[k];
- dataQ7[k] = val;
- sum = (val) * (val);
-
- val = ((*fi++ + ditherQ7[k+1] + 64) & 0xFF80) - ditherQ7[k+1];
- dataQ7[k+1] = val;
- sum += (val) * (val);
-
- if(j < FRAMESAMPLES_QUARTER)
- {
- PSpec[j] = sum >> 1;
- j++;
- }
-
- val = ((*fr++ + ditherQ7[k+2] + 64) & 0xFF80) - ditherQ7[k+2];
- dataQ7[k+2] = val;
- sum = (val) * (val);
-
- val = ((*fi++ + ditherQ7[k+3] + 64) & 0xFF80) - ditherQ7[k+3];
- dataQ7[k+3] = val;
- sum += (val) * (val);
-
- if(j < FRAMESAMPLES_QUARTER)
- {
- PSpec[j] = sum >> 1;
- j++;
- }
- }
- /* compute correlation from power spectrum */
- WebRtcIsac_FindCorrelation(PSpec, CorrQ7);
-
-
- /* find AR coefficients */
- /* number of bit shifts to 14-bit normalize CorrQ7[0] (leaving room for sign) */
- shift_var = WebRtcSpl_NormW32(CorrQ7[0]) - 18;
-
- if (shift_var > 0) {
- for (k=0; k<AR_ORDER+1; k++) {
- CorrQ7_norm[k] = CorrQ7[k] << shift_var;
- }
- } else {
- for (k=0; k<AR_ORDER+1; k++) {
- CorrQ7_norm[k] = CorrQ7[k] >> (-shift_var);
- }
- }
-
- /* find RC coefficients */
- WebRtcSpl_AutoCorrToReflCoef(CorrQ7_norm, AR_ORDER, RCQ15);
-
- /* quantize & code RC Coef */
- WebRtcIsac_EncodeRc(RCQ15, streamdata);
-
-
- /* RC -> AR coefficients */
- WebRtcSpl_ReflCoefToLpc(RCQ15, AR_ORDER, ARCoefQ12);
-
-
- /* compute ARCoef' * Corr * ARCoef in Q19 */
- nrg = 0;
- for (j = 0; j <= AR_ORDER; j++) {
- for (n = 0; n <= j; n++) {
- nrg += ( ARCoefQ12[j] * ((CorrQ7_norm[j-n] * ARCoefQ12[n] + 256) >> 9) + 4 ) >> 3;
- }
- for (n = j+1; n <= AR_ORDER; n++) {
- nrg += ( ARCoefQ12[j] * ((CorrQ7_norm[n-j] * ARCoefQ12[n] + 256) >> 9) + 4 ) >> 3;
- }
- }
-
- nrg_u32 = (WebRtc_UWord32)nrg;
- if (shift_var > 0) {
- nrg_u32 = nrg_u32 >> shift_var;
- } else {
- nrg_u32 = nrg_u32 << (-shift_var);
- }
-
- if (nrg_u32 > 0x7FFFFFFF) {
- nrg = 0x7FFFFFFF;
- } else {
- nrg = (WebRtc_Word32)nrg_u32;
- }
-
- gain2_Q10 = WebRtcSpl_DivResultInQ31(FRAMESAMPLES_QUARTER, nrg); /* also shifts 31 bits to the left! */
-
- /* quantize & code gain2_Q10 */
- if (WebRtcIsac_EncodeGain2(&gain2_Q10, streamdata)) {
- return -1;
- }
-
- /* compute inverse AR power spectrum */
- WebRtcIsac_FindInvArSpec(ARCoefQ12, gain2_Q10, invARSpec2_Q16);
-
- /* convert to magnitude spectrum, by doing square-roots (modified from SPLIB) */
- res = 1 << (WebRtcSpl_GetSizeInBits(invARSpec2_Q16[0]) >> 1);
- for (k = 0; k < FRAMESAMPLES_QUARTER; k++)
- {
- in_sqrt = invARSpec2_Q16[k];
- i = 10;
-
- /* Negative values make no sense for a real sqrt-function. */
- if (in_sqrt<0)
- in_sqrt=-in_sqrt;
-
- newRes = (in_sqrt / res + res) >> 1;
- do
- {
- res = newRes;
- newRes = (in_sqrt / res + res) >> 1;
- } while (newRes != res && i-- > 0);
-
- invARSpecQ8[k] = (WebRtc_Word16)newRes;
- }
-
- /* arithmetic coding of spectrum */
- err = WebRtcIsac_EncLogisticMulti2(streamdata, dataQ7, invARSpecQ8,
- (FRAMESAMPLES_HALF), IS_SWB_12KHZ);
- if (err < 0)
- {
- return (err);
- }
-
- return 0;
-}
-
-
-
-/* step-up */
-void WebRtcIsac_Rc2Poly(double *RC, int N, double *a)
-{
- int m, k;
- double tmp[MAX_AR_MODEL_ORDER];
-
- a[0] = 1.0;
- tmp[0] = 1.0;
- for (m=1; m<=N; m++) {
- /* copy */
- for (k=1; k<m; k++)
- tmp[k] = a[k];
-
- a[m] = RC[m-1];
- for (k=1; k<m; k++)
- a[k] += RC[m-1] * tmp[m-k];
- }
- return;
-}
-
-/* step-down */
-void WebRtcIsac_Poly2Rc(double *a, int N, double *RC)
-{
- int m, k;
- double tmp[MAX_AR_MODEL_ORDER];
- double tmp_inv;
-
- RC[N-1] = a[N];
- for (m=N-1; m>0; m--) {
- tmp_inv = 1.0 / (1.0 - RC[m]*RC[m]);
- for (k=1; k<=m; k++)
- tmp[k] = (a[k] - RC[m] * a[m-k+1]) * tmp_inv;
-
- for (k=1; k<m; k++)
- a[k] = tmp[k];
-
- RC[m-1] = tmp[m];
- }
- return;
-}
-
-
-#define MAX_ORDER 100
-
-
-void WebRtcIsac_Rc2Lar(const double *refc, double *lar, int order) { /* Matlab's LAR definition */
-
- int k;
-
- for (k = 0; k < order; k++) {
- lar[k] = log((1 + refc[k]) / (1 - refc[k]));
- }
-
-}
-
-
-void WebRtcIsac_Lar2Rc(const double *lar, double *refc, int order) {
-
- int k;
- double tmp;
-
- for (k = 0; k < order; k++) {
- tmp = exp(lar[k]);
- refc[k] = (tmp - 1) / (tmp + 1);
- }
-}
-
-void WebRtcIsac_Poly2Lar(double *lowband, int orderLo, double *hiband, int orderHi, int Nsub, double *lars) {
-
- int k, n, orderTot;
- double poly[MAX_ORDER], lar[MAX_ORDER], rc[MAX_ORDER], *inpl, *inph, *outp;
-
- orderTot = (orderLo + orderHi + 2);
- inpl = lowband;
- inph = hiband;
- outp = lars;
- poly[0] = 1.0;
- for (k = 0; k < Nsub; k++) {
- /* gains */
- outp[0] = inpl[0];
- outp[1] = inph[0];
-
- /* Low band */
- for (n = 1; n <= orderLo; n++)
- poly[n] = inpl[n];
- WebRtcIsac_Poly2Rc(poly, orderLo, rc);
- WebRtcIsac_Rc2Lar(rc, lar, orderLo);
- for (n = 0; n < orderLo; n++)
- outp[n + 2] = lar[n];
-
- /* High band */
- for (n = 1; n <= orderHi; n++)
- poly[n] = inph[n];
- WebRtcIsac_Poly2Rc(poly, orderHi, rc);
- WebRtcIsac_Rc2Lar(rc, lar, orderHi);
- for (n = 0; n < orderHi; n++)
- outp[n + orderLo + 2] = lar[n];
-
- inpl += orderLo + 1;
- inph += orderHi + 1;
- outp += orderTot;
- }
-}
-
-
-
-WebRtc_Word16
-WebRtcIsac_Poly2LarUB(
- double* lpcVecs,
- WebRtc_Word16 bandwidth)
-{
- double poly[MAX_ORDER];
- double rc[MAX_ORDER];
- double* ptrIO;
- WebRtc_Word16 vecCntr;
- WebRtc_Word16 vecSize;
- WebRtc_Word16 numVec;
-
- vecSize = UB_LPC_ORDER;
- switch(bandwidth)
- {
- case isac12kHz:
- {
- numVec = UB_LPC_VEC_PER_FRAME;
- break;
- }
- case isac16kHz:
- {
- numVec = UB16_LPC_VEC_PER_FRAME;
- break;
- }
- default:
- return -1;
- }
-
- ptrIO = lpcVecs;
- poly[0] = 1.0;
- for(vecCntr = 0; vecCntr < numVec; vecCntr++)
- {
- memcpy(&poly[1], ptrIO, sizeof(double) * vecSize);
- WebRtcIsac_Poly2Rc(poly, vecSize, rc);
- WebRtcIsac_Rc2Lar(rc, ptrIO, vecSize);
- ptrIO += vecSize;
- }
- return 0;
-}
-
-
-
-void WebRtcIsac_Lar2Poly(double *lars, double *lowband, int orderLo, double *hiband, int orderHi, int Nsub) {
-
- int k, n, orderTot;
- double poly[MAX_ORDER], lar[MAX_ORDER], rc[MAX_ORDER], *outpl, *outph, *inp;
-
- orderTot = (orderLo + orderHi + 2);
- outpl = lowband;
- outph = hiband;
- inp = lars;
- for (k = 0; k < Nsub; k++) {
- /* gains */
- outpl[0] = inp[0];
- outph[0] = inp[1];
-
- /* Low band */
- for (n = 0; n < orderLo; n++)
- lar[n] = inp[n + 2];
- WebRtcIsac_Lar2Rc(lar, rc, orderLo);
- WebRtcIsac_Rc2Poly(rc, orderLo, poly);
- for (n = 1; n <= orderLo; n++)
- outpl[n] = poly[n];
-
- /* High band */
- for (n = 0; n < orderHi; n++)
- lar[n] = inp[n + orderLo + 2];
- WebRtcIsac_Lar2Rc(lar, rc, orderHi);
- WebRtcIsac_Rc2Poly(rc, orderHi, poly);
- for (n = 1; n <= orderHi; n++)
- outph[n] = poly[n];
-
- outpl += orderLo + 1;
- outph += orderHi + 1;
- inp += orderTot;
- }
-}
-
-// assumes 2 LAR vectors interpolates to 'numPolyVec' A-polynomials
-void
-WebRtcIsac_Lar2PolyInterpolUB(
- double* larVecs,
- double* percepFilterParams,
- int numPolyVecs) // includes the first and the last point of the interval
-{
-
- int polyCntr, coeffCntr;
- double larInterpol[UB_LPC_ORDER];
- double rc[UB_LPC_ORDER];
- double delta[UB_LPC_ORDER];
-
- // calculate the step-size for linear interpolation coefficients
- for(coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++)
- {
- delta[coeffCntr] = (larVecs[UB_LPC_ORDER + coeffCntr] -
- larVecs[coeffCntr]) / (numPolyVecs - 1);
- }
-
- for(polyCntr = 0; polyCntr < numPolyVecs; polyCntr++)
- {
- for(coeffCntr = 0; coeffCntr < UB_LPC_ORDER; coeffCntr++)
- {
- larInterpol[coeffCntr] = larVecs[coeffCntr] +
- delta[coeffCntr] * polyCntr;
- }
- WebRtcIsac_Lar2Rc(larInterpol, rc, UB_LPC_ORDER);
-
- // convert to A-polynomial, the following function returns A[0] = 1;
- // which is written where gains had to be written. Then we write the
- // gain (outside this function). This way we say a memcpy
- WebRtcIsac_Rc2Poly(rc, UB_LPC_ORDER, percepFilterParams);
- percepFilterParams += (UB_LPC_ORDER + 1);
- }
-}
-
-int WebRtcIsac_DecodeLpc(Bitstr *streamdata, double *LPCCoef_lo, double *LPCCoef_hi, int *outmodel) {
-
- double lars[KLT_ORDER_GAIN + KLT_ORDER_SHAPE];
- int err;
-
- err = WebRtcIsac_DecodeLpcCoef(streamdata, lars, outmodel);
- if (err<0) // error check
- return -ISAC_RANGE_ERROR_DECODE_LPC;
-
- WebRtcIsac_Lar2Poly(lars, LPCCoef_lo, ORDERLO, LPCCoef_hi, ORDERHI, SUBFRAMES);
-
- return 0;
-}
-
-WebRtc_Word16
-WebRtcIsac_DecodeInterpolLpcUb(
- Bitstr* streamdata,
- double* percepFilterParams,
- WebRtc_Word16 bandwidth)
-{
-
- double lpcCoeff[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
- int err;
- int interpolCntr;
- int subframeCntr;
- WebRtc_Word16 numSegments;
- WebRtc_Word16 numVecPerSegment;
- WebRtc_Word16 numGains;
-
- double percepFilterGains[SUBFRAMES<<1];
- double* ptrOutParam = percepFilterParams;
-
- err = WebRtcIsac_DecodeLpcCoefUB(streamdata, lpcCoeff, percepFilterGains,
- bandwidth);
-
- // error check
- if (err<0)
- {
- return -ISAC_RANGE_ERROR_DECODE_LPC;
- }
-
- switch(bandwidth)
- {
- case isac12kHz:
- {
- numGains = SUBFRAMES;
- numSegments = UB_LPC_VEC_PER_FRAME - 1;
- numVecPerSegment = kLpcVecPerSegmentUb12;
- break;
- }
- case isac16kHz:
- {
- numGains = SUBFRAMES << 1;
- numSegments = UB16_LPC_VEC_PER_FRAME - 1;
- numVecPerSegment = kLpcVecPerSegmentUb16;
- break;
- }
- default:
- return -1;
- }
-
-
-
- for(interpolCntr = 0; interpolCntr < numSegments; interpolCntr++)
- {
- WebRtcIsac_Lar2PolyInterpolUB(
- &lpcCoeff[interpolCntr * UB_LPC_ORDER], ptrOutParam,
- numVecPerSegment + 1);
-
- ptrOutParam += ((numVecPerSegment) *
- (UB_LPC_ORDER + 1));
- }
-
- ptrOutParam = percepFilterParams;
-
- if(bandwidth == isac16kHz)
- {
- ptrOutParam += (1 + UB_LPC_ORDER);
- }
-
- for(subframeCntr = 0; subframeCntr < numGains; subframeCntr++)
- {
- *ptrOutParam = percepFilterGains[subframeCntr];
- ptrOutParam += (1 + UB_LPC_ORDER);
- }
-
- return 0;
-}
-
-
-/* decode & dequantize LPC Coef */
-int WebRtcIsac_DecodeLpcCoef(Bitstr *streamdata, double *LPCCoef, int *outmodel)
-{
- int j, k, n, model, pos, pos2, posg, poss, offsg, offss, offs2;
- int index_g[KLT_ORDER_GAIN], index_s[KLT_ORDER_SHAPE];
- double tmpcoeffs_g[KLT_ORDER_GAIN],tmpcoeffs_s[KLT_ORDER_SHAPE];
- double tmpcoeffs2_g[KLT_ORDER_GAIN], tmpcoeffs2_s[KLT_ORDER_SHAPE];
- double sum;
- int err;
-
-
- /* entropy decoding of model number */
- err = WebRtcIsac_DecHistOneStepMulti(&model, streamdata, WebRtcIsac_kQKltModelCdfPtr, WebRtcIsac_kQKltModelInitIndex, 1);
- if (err<0) // error check
- return err;
-
- /* entropy decoding of quantization indices */
- err = WebRtcIsac_DecHistOneStepMulti(index_s, streamdata, WebRtcIsac_kQKltCdfPtrShape[model], WebRtcIsac_kQKltInitIndexShape[model], KLT_ORDER_SHAPE);
- if (err<0) // error check
- return err;
- err = WebRtcIsac_DecHistOneStepMulti(index_g, streamdata, WebRtcIsac_kQKltCdfPtrGain[model], WebRtcIsac_kQKltInitIndexGain[model], KLT_ORDER_GAIN);
- if (err<0) // error check
- return err;
-
-
- /* find quantization levels for coefficients */
- for (k=0; k<KLT_ORDER_SHAPE; k++) {
- tmpcoeffs_s[WebRtcIsac_kQKltSelIndShape[k]] = WebRtcIsac_kQKltLevelsShape[WebRtcIsac_kQKltOfLevelsShape[model]+WebRtcIsac_kQKltOffsetShape[model][k] + index_s[k]];
- }
- for (k=0; k<KLT_ORDER_GAIN; k++) {
- tmpcoeffs_g[WebRtcIsac_kQKltSelIndGain[k]] = WebRtcIsac_kQKltLevelsGain[WebRtcIsac_kQKltOfLevelsGain[model]+ WebRtcIsac_kQKltOffsetGain[model][k] + index_g[k]];
- }
-
-
- /* inverse KLT */
-
- /* left transform */ // Transpose matrix!
- offsg = 0;
- offss = 0;
- posg = 0;
- poss = 0;
- for (j=0; j<SUBFRAMES; j++) {
- offs2 = 0;
- for (k=0; k<LPC_GAIN_ORDER; k++) {
- sum = 0;
- pos = offsg;
- pos2 = offs2;
- for (n=0; n<LPC_GAIN_ORDER; n++)
- sum += tmpcoeffs_g[pos++] * WebRtcIsac_kKltT1Gain[model][pos2++];
- tmpcoeffs2_g[posg++] = sum;
- offs2 += LPC_GAIN_ORDER;
- }
- offs2 = 0;
- for (k=0; k<LPC_SHAPE_ORDER; k++) {
- sum = 0;
- pos = offss;
- pos2 = offs2;
- for (n=0; n<LPC_SHAPE_ORDER; n++)
- sum += tmpcoeffs_s[pos++] * WebRtcIsac_kKltT1Shape[model][pos2++];
- tmpcoeffs2_s[poss++] = sum;
- offs2 += LPC_SHAPE_ORDER;
- }
- offsg += LPC_GAIN_ORDER;
- offss += LPC_SHAPE_ORDER;
- }
-
-
- /* right transform */ // Transpose matrix
- offsg = 0;
- offss = 0;
- posg = 0;
- poss = 0;
- for (j=0; j<SUBFRAMES; j++) {
- posg = offsg;
- for (k=0; k<LPC_GAIN_ORDER; k++) {
- sum = 0;
- pos = k;
- pos2 = j;
- for (n=0; n<SUBFRAMES; n++) {
- sum += tmpcoeffs2_g[pos] * WebRtcIsac_kKltT2Gain[model][pos2];
- pos += LPC_GAIN_ORDER;
- pos2 += SUBFRAMES;
-
- }
- tmpcoeffs_g[posg++] = sum;
- }
- poss = offss;
- for (k=0; k<LPC_SHAPE_ORDER; k++) {
- sum = 0;
- pos = k;
- pos2 = j;
- for (n=0; n<SUBFRAMES; n++) {
- sum += tmpcoeffs2_s[pos] * WebRtcIsac_kKltT2Shape[model][pos2];
- pos += LPC_SHAPE_ORDER;
- pos2 += SUBFRAMES;
- }
- tmpcoeffs_s[poss++] = sum;
- }
- offsg += LPC_GAIN_ORDER;
- offss += LPC_SHAPE_ORDER;
- }
-
-
- /* scaling, mean addition, and gain restoration */
- posg = 0;poss = 0;pos=0;
- for (k=0; k<SUBFRAMES; k++) {
-
- /* log gains */
- LPCCoef[pos] = tmpcoeffs_g[posg] / LPC_GAIN_SCALE;
- LPCCoef[pos] += WebRtcIsac_kLpcMeansGain[model][posg];
- LPCCoef[pos] = exp(LPCCoef[pos]);
- pos++;posg++;
- LPCCoef[pos] = tmpcoeffs_g[posg] / LPC_GAIN_SCALE;
- LPCCoef[pos] += WebRtcIsac_kLpcMeansGain[model][posg];
- LPCCoef[pos] = exp(LPCCoef[pos]);
- pos++;posg++;
-
- /* lo band LAR coeffs */
- for (n=0; n<LPC_LOBAND_ORDER; n++, pos++, poss++) {
- LPCCoef[pos] = tmpcoeffs_s[poss] / LPC_LOBAND_SCALE;
- LPCCoef[pos] += WebRtcIsac_kLpcMeansShape[model][poss];
- }
-
- /* hi band LAR coeffs */
- for (n=0; n<LPC_HIBAND_ORDER; n++, pos++, poss++) {
- LPCCoef[pos] = tmpcoeffs_s[poss] / LPC_HIBAND_SCALE;
- LPCCoef[pos] += WebRtcIsac_kLpcMeansShape[model][poss];
- }
- }
-
-
- *outmodel=model;
-
- return 0;
-}
-
-/* estimate codel length of LPC Coef */
-void WebRtcIsac_EncodeLar(double *LPCCoef, int *model, double *size, Bitstr *streamdata, ISAC_SaveEncData_t* encData) {
- int j, k, n, bmodel, pos, pos2, poss, posg, offsg, offss, offs2;
- int index_g[KLT_ORDER_GAIN], index_s[KLT_ORDER_SHAPE];
- int index_ovr_g[KLT_ORDER_GAIN], index_ovr_s[KLT_ORDER_SHAPE];
- double Bits;
- double tmpcoeffs_g[KLT_ORDER_GAIN], tmpcoeffs_s[KLT_ORDER_SHAPE];
- double tmpcoeffs2_g[KLT_ORDER_GAIN], tmpcoeffs2_s[KLT_ORDER_SHAPE];
- double sum;
-
- /* Only one LPC model remains in iSAC. Tables for other models are saved for compatibility reasons. */
- bmodel = 0;
-
- /* log gains, mean removal and scaling */
- posg = 0;poss = 0;pos=0;
-
- for (k=0; k<SUBFRAMES; k++) {
- /* log gains */
- tmpcoeffs_g[posg] = log(LPCCoef[pos]);
- tmpcoeffs_g[posg] -= WebRtcIsac_kLpcMeansGain[bmodel][posg];
- tmpcoeffs_g[posg] *= LPC_GAIN_SCALE;
- posg++;pos++;
-
- tmpcoeffs_g[posg] = log(LPCCoef[pos]);
- tmpcoeffs_g[posg] -= WebRtcIsac_kLpcMeansGain[bmodel][posg];
- tmpcoeffs_g[posg] *= LPC_GAIN_SCALE;
- posg++;pos++;
-
- /* lo band LAR coeffs */
- for (n=0; n<LPC_LOBAND_ORDER; n++, poss++, pos++) {
- tmpcoeffs_s[poss] = LPCCoef[pos] - WebRtcIsac_kLpcMeansShape[bmodel][poss];
- tmpcoeffs_s[poss] *= LPC_LOBAND_SCALE;
- }
-
- /* hi band LAR coeffs */
- for (n=0; n<LPC_HIBAND_ORDER; n++, poss++, pos++) {
- tmpcoeffs_s[poss] = LPCCoef[pos] - WebRtcIsac_kLpcMeansShape[bmodel][poss];
- tmpcoeffs_s[poss] *= LPC_HIBAND_SCALE;
- }
- }
-
- /* KLT */
-
- /* left transform */
- offsg = 0;
- offss = 0;
- for (j=0; j<SUBFRAMES; j++) {
- posg = offsg;
- for (k=0; k<LPC_GAIN_ORDER; k++) {
- sum = 0;
- pos = offsg;
- pos2 = k;
- for (n=0; n<LPC_GAIN_ORDER; n++) {
- sum += tmpcoeffs_g[pos++] * WebRtcIsac_kKltT1Gain[bmodel][pos2];
- pos2 += LPC_GAIN_ORDER;
- }
- tmpcoeffs2_g[posg++] = sum;
- }
- poss = offss;
- for (k=0; k<LPC_SHAPE_ORDER; k++) {
- sum = 0;
- pos = offss;
- pos2 = k;
- for (n=0; n<LPC_SHAPE_ORDER; n++) {
- sum += tmpcoeffs_s[pos++] * WebRtcIsac_kKltT1Shape[bmodel][pos2];
- pos2 += LPC_SHAPE_ORDER;
- }
- tmpcoeffs2_s[poss++] = sum;
- }
- offsg += LPC_GAIN_ORDER;
- offss += LPC_SHAPE_ORDER;
- }
-
- /* right transform */
- offsg = 0;
- offss = 0;
- offs2 = 0;
- for (j=0; j<SUBFRAMES; j++) {
- posg = offsg;
- for (k=0; k<LPC_GAIN_ORDER; k++) {
- sum = 0;
- pos = k;
- pos2 = offs2;
- for (n=0; n<SUBFRAMES; n++) {
- sum += tmpcoeffs2_g[pos] * WebRtcIsac_kKltT2Gain[bmodel][pos2++];
- pos += LPC_GAIN_ORDER;
- }
- tmpcoeffs_g[posg++] = sum;
- }
- poss = offss;
- for (k=0; k<LPC_SHAPE_ORDER; k++) {
- sum = 0;
- pos = k;
- pos2 = offs2;
- for (n=0; n<SUBFRAMES; n++) {
- sum += tmpcoeffs2_s[pos] * WebRtcIsac_kKltT2Shape[bmodel][pos2++];
- pos += LPC_SHAPE_ORDER;
- }
- tmpcoeffs_s[poss++] = sum;
- }
- offs2 += SUBFRAMES;
- offsg += LPC_GAIN_ORDER;
- offss += LPC_SHAPE_ORDER;
- }
-
- /* quantize coefficients */
-
- Bits = 0.0;
- for (k=0; k<KLT_ORDER_GAIN; k++) //ATTN: ok?
- {
- pos = WebRtcIsac_kQKltSelIndGain[k];
- pos2= WebRtcIsac_lrint(tmpcoeffs_g[pos] / KLT_STEPSIZE);
- index_g[k] = (pos2) + WebRtcIsac_kQKltQuantMinGain[k]; //ATTN: ok?
- if (index_g[k] < 0) {
- index_g[k] = 0;
- }
- else if (index_g[k] > WebRtcIsac_kQKltMaxIndGain[k])
- index_g[k] = WebRtcIsac_kQKltMaxIndGain[k];
- index_ovr_g[k] = WebRtcIsac_kQKltOffsetGain[bmodel][k]+index_g[k];
- pos = WebRtcIsac_kQKltOfLevelsGain[bmodel] + index_ovr_g[k];
-
- /* determine number of bits */
- sum = WebRtcIsac_kQKltCodeLenGain[pos];
- Bits += sum;
- }
-
- for (k=0; k<KLT_ORDER_SHAPE; k++) //ATTN: ok?
- {
- index_s[k] = (WebRtcIsac_lrint(tmpcoeffs_s[WebRtcIsac_kQKltSelIndShape[k]] / KLT_STEPSIZE)) + WebRtcIsac_kQKltQuantMinShape[k]; //ATTN: ok?
- if (index_s[k] < 0)
- index_s[k] = 0;
- else if (index_s[k] > WebRtcIsac_kQKltMaxIndShape[k])
- index_s[k] = WebRtcIsac_kQKltMaxIndShape[k];
- index_ovr_s[k] = WebRtcIsac_kQKltOffsetShape[bmodel][k]+index_s[k];
- pos = WebRtcIsac_kQKltOfLevelsShape[bmodel] + index_ovr_s[k];
- sum = WebRtcIsac_kQKltCodeLenShape[pos];
- Bits += sum;
- }
-
-
- /* Only one model remains in this version of the code, model = 0 */
- *model=bmodel;
- *size=Bits;
-
- /* entropy coding of model number */
- WebRtcIsac_EncHistMulti(streamdata, model, WebRtcIsac_kQKltModelCdfPtr, 1);
-
- /* entropy coding of quantization indices - shape only */
- WebRtcIsac_EncHistMulti(streamdata, index_s, WebRtcIsac_kQKltCdfPtrShape[bmodel], KLT_ORDER_SHAPE);
-
- /* Save data for creation of multiple bit streams */
- encData->LPCmodel[encData->startIdx] = 0;
- for (k=0; k<KLT_ORDER_SHAPE; k++)
- {
- encData->LPCindex_s[KLT_ORDER_SHAPE*encData->startIdx + k] = index_s[k];
- }
-
- /* find quantization levels for shape coefficients */
- for (k=0; k<KLT_ORDER_SHAPE; k++) {
- tmpcoeffs_s[WebRtcIsac_kQKltSelIndShape[k]] = WebRtcIsac_kQKltLevelsShape[WebRtcIsac_kQKltOfLevelsShape[bmodel]+index_ovr_s[k]];
- }
- /* inverse KLT */
- /* left transform */ // Transpose matrix!
- offss = 0;
- poss = 0;
- for (j=0; j<SUBFRAMES; j++) {
- offs2 = 0;
- for (k=0; k<LPC_SHAPE_ORDER; k++) {
- sum = 0;
- pos = offss;
- pos2 = offs2;
- for (n=0; n<LPC_SHAPE_ORDER; n++)
- sum += tmpcoeffs_s[pos++] * WebRtcIsac_kKltT1Shape[bmodel][pos2++];
- tmpcoeffs2_s[poss++] = sum;
- offs2 += LPC_SHAPE_ORDER;
- }
- offss += LPC_SHAPE_ORDER;
- }
-
-
- /* right transform */ // Transpose matrix
- offss = 0;
- poss = 0;
- for (j=0; j<SUBFRAMES; j++) {
- poss = offss;
- for (k=0; k<LPC_SHAPE_ORDER; k++) {
- sum = 0;
- pos = k;
- pos2 = j;
- for (n=0; n<SUBFRAMES; n++) {
- sum += tmpcoeffs2_s[pos] * WebRtcIsac_kKltT2Shape[bmodel][pos2];
- pos += LPC_SHAPE_ORDER;
- pos2 += SUBFRAMES;
- }
- tmpcoeffs_s[poss++] = sum;
- }
- offss += LPC_SHAPE_ORDER;
- }
-
- /* scaling, mean addition, and gain restoration */
- poss = 0;pos=0;
- for (k=0; k<SUBFRAMES; k++) {
-
- /* log gains */
- pos+=2;
-
- /* lo band LAR coeffs */
- for (n=0; n<LPC_LOBAND_ORDER; n++, pos++, poss++) {
- LPCCoef[pos] = tmpcoeffs_s[poss] / LPC_LOBAND_SCALE;
- LPCCoef[pos] += WebRtcIsac_kLpcMeansShape[bmodel][poss];
- }
-
- /* hi band LAR coeffs */
- for (n=0; n<LPC_HIBAND_ORDER; n++, pos++, poss++) {
- LPCCoef[pos] = tmpcoeffs_s[poss] / LPC_HIBAND_SCALE;
- LPCCoef[pos] += WebRtcIsac_kLpcMeansShape[bmodel][poss];
- }
-
- }
-
-}
-
-
-void WebRtcIsac_EncodeLpcLb(double *LPCCoef_lo, double *LPCCoef_hi, int *model,
- double *size, Bitstr *streamdata, ISAC_SaveEncData_t* encData) {
-
- double lars[KLT_ORDER_GAIN+KLT_ORDER_SHAPE];
- int k;
-
- WebRtcIsac_Poly2Lar(LPCCoef_lo, ORDERLO, LPCCoef_hi, ORDERHI, SUBFRAMES, lars);
- WebRtcIsac_EncodeLar(lars, model, size, streamdata, encData);
- WebRtcIsac_Lar2Poly(lars, LPCCoef_lo, ORDERLO, LPCCoef_hi, ORDERHI, SUBFRAMES);
- /* Save data for creation of multiple bit streams (and transcoding) */
- for (k=0; k<(ORDERLO+1)*SUBFRAMES; k++) {
- encData->LPCcoeffs_lo[(ORDERLO+1)*SUBFRAMES*encData->startIdx + k] = LPCCoef_lo[k];
- }
- for (k=0; k<(ORDERHI+1)*SUBFRAMES; k++) {
- encData->LPCcoeffs_hi[(ORDERHI+1)*SUBFRAMES*encData->startIdx + k] = LPCCoef_hi[k];
- }
-}
-
-
-WebRtc_Word16
-WebRtcIsac_EncodeLpcUB(
- double* lpcVecs,
- Bitstr* streamdata,
- double* interpolLPCCoeff,
- WebRtc_Word16 bandwidth,
- ISACUBSaveEncDataStruct* encData)
-{
-
- double U[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
- int idx[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
- int interpolCntr;
-
- WebRtcIsac_Poly2LarUB(lpcVecs, bandwidth);
- WebRtcIsac_RemoveLarMean(lpcVecs, bandwidth);
- WebRtcIsac_DecorrelateIntraVec(lpcVecs, U, bandwidth);
- WebRtcIsac_DecorrelateInterVec(U, lpcVecs, bandwidth);
- WebRtcIsac_QuantizeUncorrLar(lpcVecs, idx, bandwidth);
-
- WebRtcIsac_CorrelateInterVec(lpcVecs, U, bandwidth);
- WebRtcIsac_CorrelateIntraVec(U, lpcVecs, bandwidth);
- WebRtcIsac_AddLarMean(lpcVecs, bandwidth);
-
- switch(bandwidth)
- {
- case isac12kHz:
- {
- // Stor the indices to be used for multiple encoding.
- memcpy(encData->indexLPCShape, idx, UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME *
- sizeof(int));
- WebRtcIsac_EncHistMulti(streamdata, idx, WebRtcIsac_kLpcShapeCdfMatUb12,
- UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME);
- for(interpolCntr = 0; interpolCntr < UB_INTERPOL_SEGMENTS; interpolCntr++)
- {
- WebRtcIsac_Lar2PolyInterpolUB(lpcVecs,
- interpolLPCCoeff, kLpcVecPerSegmentUb12 + 1);
- lpcVecs += UB_LPC_ORDER;
- interpolLPCCoeff += (kLpcVecPerSegmentUb12 * (UB_LPC_ORDER + 1));
- }
- break;
- }
- case isac16kHz:
- {
- // Stor the indices to be used for multiple encoding.
- memcpy(encData->indexLPCShape, idx, UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME *
- sizeof(int));
- WebRtcIsac_EncHistMulti(streamdata, idx, WebRtcIsac_kLpcShapeCdfMatUb16,
- UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME);
- for(interpolCntr = 0; interpolCntr < UB16_INTERPOL_SEGMENTS; interpolCntr++)
- {
- WebRtcIsac_Lar2PolyInterpolUB(lpcVecs,
- interpolLPCCoeff, kLpcVecPerSegmentUb16 + 1);
- lpcVecs += UB_LPC_ORDER;
- interpolLPCCoeff += (kLpcVecPerSegmentUb16 * (UB_LPC_ORDER + 1));
- }
- break;
- }
- default:
- return -1;
- }
- return 0;
-}
-
-void WebRtcIsac_EncodeLpcGainLb(double *LPCCoef_lo, double *LPCCoef_hi, int model, Bitstr *streamdata, ISAC_SaveEncData_t* encData) {
-
- int j, k, n, pos, pos2, posg, offsg, offs2;
- int index_g[KLT_ORDER_GAIN];
- int index_ovr_g[KLT_ORDER_GAIN];
- double tmpcoeffs_g[KLT_ORDER_GAIN];
- double tmpcoeffs2_g[KLT_ORDER_GAIN];
- double sum;
-
- /* log gains, mean removal and scaling */
- posg = 0;
- for (k=0; k<SUBFRAMES; k++) {
- tmpcoeffs_g[posg] = log(LPCCoef_lo[(LPC_LOBAND_ORDER+1)*k]);
- tmpcoeffs_g[posg] -= WebRtcIsac_kLpcMeansGain[model][posg];
- tmpcoeffs_g[posg] *= LPC_GAIN_SCALE;
- posg++;
- tmpcoeffs_g[posg] = log(LPCCoef_hi[(LPC_HIBAND_ORDER+1)*k]);
- tmpcoeffs_g[posg] -= WebRtcIsac_kLpcMeansGain[model][posg];
- tmpcoeffs_g[posg] *= LPC_GAIN_SCALE;
- posg++;
- }
-
- /* KLT */
-
- /* left transform */
- offsg = 0;
- for (j=0; j<SUBFRAMES; j++) {
- posg = offsg;
- for (k=0; k<LPC_GAIN_ORDER; k++) {
- sum = 0;
- pos = offsg;
- pos2 = k;
- for (n=0; n<LPC_GAIN_ORDER; n++) {
- sum += tmpcoeffs_g[pos++] * WebRtcIsac_kKltT1Gain[model][pos2];
- pos2 += LPC_GAIN_ORDER;
- }
- tmpcoeffs2_g[posg++] = sum;
- }
- offsg += LPC_GAIN_ORDER;
- }
-
- /* right transform */
- offsg = 0;
- offs2 = 0;
- for (j=0; j<SUBFRAMES; j++) {
- posg = offsg;
- for (k=0; k<LPC_GAIN_ORDER; k++) {
- sum = 0;
- pos = k;
- pos2 = offs2;
- for (n=0; n<SUBFRAMES; n++) {
- sum += tmpcoeffs2_g[pos] * WebRtcIsac_kKltT2Gain[model][pos2++];
- pos += LPC_GAIN_ORDER;
- }
- tmpcoeffs_g[posg++] = sum;
- }
- offs2 += SUBFRAMES;
- offsg += LPC_GAIN_ORDER;
- }
-
-
- /* quantize coefficients */
- for (k=0; k<KLT_ORDER_GAIN; k++) {
-
- /* get index */
- pos = WebRtcIsac_kQKltSelIndGain[k];
- pos2= WebRtcIsac_lrint(tmpcoeffs_g[pos] / KLT_STEPSIZE);
- index_g[k] = (pos2) + WebRtcIsac_kQKltQuantMinGain[k];
- if (index_g[k] < 0) {
- index_g[k] = 0;
- }
- else if (index_g[k] > WebRtcIsac_kQKltMaxIndGain[k]) {
- index_g[k] = WebRtcIsac_kQKltMaxIndGain[k];
- }
- index_ovr_g[k] = WebRtcIsac_kQKltOffsetGain[model][k]+index_g[k];
-
- /* find quantization levels for coefficients */
- tmpcoeffs_g[WebRtcIsac_kQKltSelIndGain[k]] = WebRtcIsac_kQKltLevelsGain[WebRtcIsac_kQKltOfLevelsGain[model]+index_ovr_g[k]];
-
- /* Save data for creation of multiple bit streams */
- encData->LPCindex_g[KLT_ORDER_GAIN*encData->startIdx + k] = index_g[k];
- }
-
-
- /* entropy coding of quantization indices - gain */
- WebRtcIsac_EncHistMulti(streamdata, index_g, WebRtcIsac_kQKltCdfPtrGain[model], KLT_ORDER_GAIN);
-
- /* find quantization levels for coefficients */
-
- /* left transform */
- offsg = 0;
- posg = 0;
- for (j=0; j<SUBFRAMES; j++) {
- offs2 = 0;
- for (k=0; k<LPC_GAIN_ORDER; k++) {
- sum = 0;
- pos = offsg;
- pos2 = offs2;
- for (n=0; n<LPC_GAIN_ORDER; n++)
- sum += tmpcoeffs_g[pos++] * WebRtcIsac_kKltT1Gain[model][pos2++];
- tmpcoeffs2_g[posg++] = sum;
- offs2 += LPC_GAIN_ORDER;
- }
- offsg += LPC_GAIN_ORDER;
- }
-
- /* right transform */ // Transpose matrix
- offsg = 0;
- posg = 0;
- for (j=0; j<SUBFRAMES; j++) {
- posg = offsg;
- for (k=0; k<LPC_GAIN_ORDER; k++) {
- sum = 0;
- pos = k;
- pos2 = j;
- for (n=0; n<SUBFRAMES; n++) {
- sum += tmpcoeffs2_g[pos] * WebRtcIsac_kKltT2Gain[model][pos2];
- pos += LPC_GAIN_ORDER;
- pos2 += SUBFRAMES;
- }
- tmpcoeffs_g[posg++] = sum;
- }
- offsg += LPC_GAIN_ORDER;
- }
-
-
- /* scaling, mean addition, and gain restoration */
- posg = 0;
- for (k=0; k<SUBFRAMES; k++) {
- sum = tmpcoeffs_g[posg] / LPC_GAIN_SCALE;
- sum += WebRtcIsac_kLpcMeansGain[model][posg];
- LPCCoef_lo[k*(LPC_LOBAND_ORDER+1)] = exp(sum);
- pos++;posg++;
- sum = tmpcoeffs_g[posg] / LPC_GAIN_SCALE;
- sum += WebRtcIsac_kLpcMeansGain[model][posg];
- LPCCoef_hi[k*(LPC_HIBAND_ORDER+1)] = exp(sum);
- pos++;posg++;
- }
-
-}
-
-void
-WebRtcIsac_EncodeLpcGainUb(
- double* lpGains,
- Bitstr* streamdata,
- int* lpcGainIndex)
-{
- double U[UB_LPC_GAIN_DIM];
- int idx[UB_LPC_GAIN_DIM];
- WebRtcIsac_ToLogDomainRemoveMean(lpGains);
- WebRtcIsac_DecorrelateLPGain(lpGains, U);
- WebRtcIsac_QuantizeLpcGain(U, idx);
- // Store the index for re-encoding for FEC.
- memcpy(lpcGainIndex, idx, UB_LPC_GAIN_DIM * sizeof(int));
- WebRtcIsac_CorrelateLpcGain(U, lpGains);
- WebRtcIsac_AddMeanToLinearDomain(lpGains);
- WebRtcIsac_EncHistMulti(streamdata, idx, WebRtcIsac_kLpcGainCdfMat, UB_LPC_GAIN_DIM);
-}
-
-
-void
-WebRtcIsac_StoreLpcGainUb(
- double* lpGains,
- Bitstr* streamdata)
-{
- double U[UB_LPC_GAIN_DIM];
- int idx[UB_LPC_GAIN_DIM];
- WebRtcIsac_ToLogDomainRemoveMean(lpGains);
- WebRtcIsac_DecorrelateLPGain(lpGains, U);
- WebRtcIsac_QuantizeLpcGain(U, idx);
- WebRtcIsac_EncHistMulti(streamdata, idx, WebRtcIsac_kLpcGainCdfMat, UB_LPC_GAIN_DIM);
-}
-
-
-
-WebRtc_Word16
-WebRtcIsac_DecodeLpcGainUb(
- double* lpGains,
- Bitstr* streamdata)
-{
- double U[UB_LPC_GAIN_DIM];
- int idx[UB_LPC_GAIN_DIM];
- int err;
- err = WebRtcIsac_DecHistOneStepMulti(idx, streamdata,
- WebRtcIsac_kLpcGainCdfMat, WebRtcIsac_kLpcGainEntropySearch,
- UB_LPC_GAIN_DIM);
- if(err < 0)
- {
- return -1;
- }
- WebRtcIsac_DequantizeLpcGain(idx, U);
- WebRtcIsac_CorrelateLpcGain(U, lpGains);
- WebRtcIsac_AddMeanToLinearDomain(lpGains);
- return 0;
-}
-
-
-
-/* decode & dequantize RC */
-int WebRtcIsac_DecodeRc(Bitstr *streamdata, WebRtc_Word16 *RCQ15)
-{
- int k, err;
- int index[AR_ORDER];
-
- /* entropy decoding of quantization indices */
- err = WebRtcIsac_DecHistOneStepMulti(index, streamdata, WebRtcIsac_kQArRcCdfPtr,
- WebRtcIsac_kQArRcInitIndex, AR_ORDER);
- if (err<0) // error check
- return err;
-
- /* find quantization levels for reflection coefficients */
- for (k=0; k<AR_ORDER; k++)
- {
- RCQ15[k] = *(WebRtcIsac_kQArRcLevelsPtr[k] + index[k]);
- }
-
- return 0;
-}
-
-
-
-/* quantize & code RC */
-void WebRtcIsac_EncodeRc(WebRtc_Word16 *RCQ15, Bitstr *streamdata)
-{
- int k;
- int index[AR_ORDER];
-
- /* quantize reflection coefficients (add noise feedback?) */
- for (k=0; k<AR_ORDER; k++)
- {
- index[k] = WebRtcIsac_kQArRcInitIndex[k];
-
- if (RCQ15[k] > WebRtcIsac_kQArBoundaryLevels[index[k]])
- {
- while (RCQ15[k] > WebRtcIsac_kQArBoundaryLevels[index[k] + 1])
- index[k]++;
- }
- else
- {
- while (RCQ15[k] < WebRtcIsac_kQArBoundaryLevels[--index[k]]) ;
- }
-
- RCQ15[k] = *(WebRtcIsac_kQArRcLevelsPtr[k] + index[k]);
- }
-
-
- /* entropy coding of quantization indices */
- WebRtcIsac_EncHistMulti(streamdata, index, WebRtcIsac_kQArRcCdfPtr, AR_ORDER);
-}
-
-
-/* decode & dequantize squared Gain */
-int WebRtcIsac_DecodeGain2(Bitstr *streamdata, WebRtc_Word32 *gainQ10)
-{
- int index, err;
-
- /* entropy decoding of quantization index */
- err = WebRtcIsac_DecHistOneStepMulti(&index, streamdata, WebRtcIsac_kQGainCdf_ptr,
- WebRtcIsac_kQGainInitIndex, 1);
- if (err<0) // error check
- return err;
-
- /* find quantization level */
- *gainQ10 = WebRtcIsac_kQGain2Levels[index];
-
- return 0;
-}
-
-
-
-/* quantize & code squared Gain */
-int WebRtcIsac_EncodeGain2(WebRtc_Word32 *gainQ10, Bitstr *streamdata)
-{
- int index;
-
-
- /* find quantization index */
- index = WebRtcIsac_kQGainInitIndex[0];
- if (*gainQ10 > WebRtcIsac_kQGain2BoundaryLevels[index])
- {
- while (*gainQ10 > WebRtcIsac_kQGain2BoundaryLevels[index + 1])
- index++;
- }
- else
- {
- while (*gainQ10 < WebRtcIsac_kQGain2BoundaryLevels[--index]) ;
- }
-
- /* dequantize */
- *gainQ10 = WebRtcIsac_kQGain2Levels[index];
-
-
- /* entropy coding of quantization index */
- WebRtcIsac_EncHistMulti(streamdata, &index, WebRtcIsac_kQGainCdf_ptr, 1);
-
- return 0;
-}
-
-
-/* code and decode Pitch Gains and Lags functions */
-
-/* decode & dequantize Pitch Gains */
-int WebRtcIsac_DecodePitchGain(Bitstr *streamdata, WebRtc_Word16 *PitchGains_Q12)
-{
- int index_comb, err;
- const WebRtc_UWord16 *WebRtcIsac_kQPitchGainCdf_ptr[1];
-
- /* entropy decoding of quantization indices */
- *WebRtcIsac_kQPitchGainCdf_ptr = WebRtcIsac_kQPitchGainCdf;
- err = WebRtcIsac_DecHistBisectMulti(&index_comb, streamdata, WebRtcIsac_kQPitchGainCdf_ptr, WebRtcIsac_kQCdfTableSizeGain, 1);
- /* error check, Q_mean_Gain.. tables are of size 144 */
- if ((err<0) || (index_comb<0) || (index_comb>144))
- return -ISAC_RANGE_ERROR_DECODE_PITCH_GAIN;
-
- /* unquantize back to pitch gains by table look-up */
- PitchGains_Q12[0] = WebRtcIsac_kQMeanGain1Q12[index_comb];
- PitchGains_Q12[1] = WebRtcIsac_kQMeanGain2Q12[index_comb];
- PitchGains_Q12[2] = WebRtcIsac_kQMeanGain3Q12[index_comb];
- PitchGains_Q12[3] = WebRtcIsac_kQMeanGain4Q12[index_comb];
-
- return 0;
-}
-
-
-/* quantize & code Pitch Gains */
-void WebRtcIsac_EncodePitchGain(WebRtc_Word16 *PitchGains_Q12, Bitstr *streamdata, ISAC_SaveEncData_t* encData)
-{
- int k,j;
- double C;
- double S[PITCH_SUBFRAMES];
- int index[3];
- int index_comb;
- const WebRtc_UWord16 *WebRtcIsac_kQPitchGainCdf_ptr[1];
- double PitchGains[PITCH_SUBFRAMES] = {0,0,0,0};
-
- /* take the asin */
- for (k=0; k<PITCH_SUBFRAMES; k++)
- {
- PitchGains[k] = ((float)PitchGains_Q12[k])/4096;
- S[k] = asin(PitchGains[k]);
- }
-
-
- /* find quantization index; only for the first three transform coefficients */
- for (k=0; k<3; k++)
- {
- /* transform */
- C = 0.0;
- for (j=0; j<PITCH_SUBFRAMES; j++)
- C += WebRtcIsac_kTransform[k][j] * S[j];
-
- /* quantize */
- index[k] = WebRtcIsac_lrint(C / PITCH_GAIN_STEPSIZE);
-
- /* check that the index is not outside the boundaries of the table */
- if (index[k] < WebRtcIsac_kIndexLowerLimitGain[k]) index[k] = WebRtcIsac_kIndexLowerLimitGain[k];
- else if (index[k] > WebRtcIsac_kIndexUpperLimitGain[k]) index[k] = WebRtcIsac_kIndexUpperLimitGain[k];
- index[k] -= WebRtcIsac_kIndexLowerLimitGain[k];
- }
-
- /* calculate unique overall index */
- index_comb = WebRtcIsac_kIndexMultsGain[0] * index[0] + WebRtcIsac_kIndexMultsGain[1] * index[1] + index[2];
-
- /* unquantize back to pitch gains by table look-up */
- PitchGains_Q12[0] = WebRtcIsac_kQMeanGain1Q12[index_comb];
- PitchGains_Q12[1] = WebRtcIsac_kQMeanGain2Q12[index_comb];
- PitchGains_Q12[2] = WebRtcIsac_kQMeanGain3Q12[index_comb];
- PitchGains_Q12[3] = WebRtcIsac_kQMeanGain4Q12[index_comb];
-
- /* entropy coding of quantization pitch gains */
- *WebRtcIsac_kQPitchGainCdf_ptr = WebRtcIsac_kQPitchGainCdf;
- WebRtcIsac_EncHistMulti(streamdata, &index_comb, WebRtcIsac_kQPitchGainCdf_ptr, 1);
- encData->pitchGain_index[encData->startIdx] = index_comb;
-
-}
-
-
-
-/* Pitch LAG */
-
-
-/* decode & dequantize Pitch Lags */
-int WebRtcIsac_DecodePitchLag(Bitstr *streamdata, WebRtc_Word16 *PitchGain_Q12, double *PitchLags)
-{
- int k, err;
- double StepSize;
- double C;
- int index[PITCH_SUBFRAMES];
- double mean_gain;
- const double *mean_val2, *mean_val3, *mean_val4;
- const WebRtc_Word16 *lower_limit;
- const WebRtc_UWord16 *init_index;
- const WebRtc_UWord16 *cdf_size;
- const WebRtc_UWord16 **cdf;
-
- //(Y)
- double PitchGain[4]={0,0,0,0};
- //
-
- /* compute mean pitch gain */
- mean_gain = 0.0;
- for (k = 0; k < 4; k++)
- {
- //(Y)
- PitchGain[k] = ((float)PitchGain_Q12[k])/4096;
- //(Y)
- mean_gain += PitchGain[k];
- }
- mean_gain /= 4.0;
-
- /* voicing classificiation */
- if (mean_gain < 0.2) {
- StepSize = WebRtcIsac_kQPitchLagStepsizeLo;
- cdf = WebRtcIsac_kQPitchLagCdfPtrLo;
- cdf_size = WebRtcIsac_kQPitchLagCdfSizeLo;
- mean_val2 = WebRtcIsac_kQMeanLag2Lo;
- mean_val3 = WebRtcIsac_kQMeanLag3Lo;
- mean_val4 = WebRtcIsac_kQMeanLag4Lo;
- lower_limit = WebRtcIsac_kQIndexLowerLimitLagLo;
- init_index = WebRtcIsac_kQInitIndexLagLo;
- } else if (mean_gain < 0.4) {
- StepSize = WebRtcIsac_kQPitchLagStepsizeMid;
- cdf = WebRtcIsac_kQPitchLagCdfPtrMid;
- cdf_size = WebRtcIsac_kQPitchLagCdfSizeMid;
- mean_val2 = WebRtcIsac_kQMeanLag2Mid;
- mean_val3 = WebRtcIsac_kQMeanLag3Mid;
- mean_val4 = WebRtcIsac_kQMeanLag4Mid;
- lower_limit = WebRtcIsac_kQIndexLowerLimitLagMid;
- init_index = WebRtcIsac_kQInitIndexLagMid;
- } else {
- StepSize = WebRtcIsac_kQPitchLagStepsizeHi;
- cdf = WebRtcIsac_kQPitchLagCdfPtrHi;
- cdf_size = WebRtcIsac_kQPitchLagCdfSizeHi;
- mean_val2 = WebRtcIsac_kQMeanLag2Hi;
- mean_val3 = WebRtcIsac_kQMeanLag3Hi;
- mean_val4 = WebRtcIsac_kQMeanLag4Hi;
- lower_limit = WebRtcIsac_kQindexLowerLimitLagHi;
- init_index = WebRtcIsac_kQInitIndexLagHi;
- }
-
- /* entropy decoding of quantization indices */
- err = WebRtcIsac_DecHistBisectMulti(index, streamdata, cdf, cdf_size, 1);
- if ((err<0) || (index[0]<0)) // error check
- return -ISAC_RANGE_ERROR_DECODE_PITCH_LAG;
-
- err = WebRtcIsac_DecHistOneStepMulti(index+1, streamdata, cdf+1, init_index, 3);
- if (err<0) // error check
- return -ISAC_RANGE_ERROR_DECODE_PITCH_LAG;
-
-
- /* unquantize back to transform coefficients and do the inverse transform: S = T'*C */
- C = (index[0] + lower_limit[0]) * StepSize;
- for (k=0; k<PITCH_SUBFRAMES; k++)
- PitchLags[k] = WebRtcIsac_kTransformTranspose[k][0] * C;
- C = mean_val2[index[1]];
- for (k=0; k<PITCH_SUBFRAMES; k++)
- PitchLags[k] += WebRtcIsac_kTransformTranspose[k][1] * C;
- C = mean_val3[index[2]];
- for (k=0; k<PITCH_SUBFRAMES; k++)
- PitchLags[k] += WebRtcIsac_kTransformTranspose[k][2] * C;
- C = mean_val4[index[3]];
- for (k=0; k<PITCH_SUBFRAMES; k++)
- PitchLags[k] += WebRtcIsac_kTransformTranspose[k][3] * C;
-
- return 0;
-}
-
-
-
-/* quantize & code Pitch Lags */
-void WebRtcIsac_EncodePitchLag(double* PitchLags, WebRtc_Word16* PitchGain_Q12, Bitstr* streamdata, ISAC_SaveEncData_t* encData)
-{
- int k, j;
- double StepSize;
- double C;
- int index[PITCH_SUBFRAMES];
- double mean_gain;
- const double *mean_val2, *mean_val3, *mean_val4;
- const WebRtc_Word16 *lower_limit, *upper_limit;
- const WebRtc_UWord16 **cdf;
-
- //(Y)
- double PitchGain[4]={0,0,0,0};
- //
-
- /* compute mean pitch gain */
- mean_gain = 0.0;
- for (k = 0; k < 4; k++)
- {
- //(Y)
- PitchGain[k] = ((float)PitchGain_Q12[k])/4096;
- //(Y)
- mean_gain += PitchGain[k];
- }
- mean_gain /= 4.0;
-
- /* Save data for creation of multiple bit streams */
- encData->meanGain[encData->startIdx] = mean_gain;
-
- /* voicing classification */
- if (mean_gain < 0.2) {
- StepSize = WebRtcIsac_kQPitchLagStepsizeLo;
- cdf = WebRtcIsac_kQPitchLagCdfPtrLo;
- mean_val2 = WebRtcIsac_kQMeanLag2Lo;
- mean_val3 = WebRtcIsac_kQMeanLag3Lo;
- mean_val4 = WebRtcIsac_kQMeanLag4Lo;
- lower_limit = WebRtcIsac_kQIndexLowerLimitLagLo;
- upper_limit = WebRtcIsac_kQIndexUpperLimitLagLo;
- } else if (mean_gain < 0.4) {
- StepSize = WebRtcIsac_kQPitchLagStepsizeMid;
- cdf = WebRtcIsac_kQPitchLagCdfPtrMid;
- mean_val2 = WebRtcIsac_kQMeanLag2Mid;
- mean_val3 = WebRtcIsac_kQMeanLag3Mid;
- mean_val4 = WebRtcIsac_kQMeanLag4Mid;
- lower_limit = WebRtcIsac_kQIndexLowerLimitLagMid;
- upper_limit = WebRtcIsac_kQIndexUpperLimitLagMid;
- } else {
- StepSize = WebRtcIsac_kQPitchLagStepsizeHi;
- cdf = WebRtcIsac_kQPitchLagCdfPtrHi;
- mean_val2 = WebRtcIsac_kQMeanLag2Hi;
- mean_val3 = WebRtcIsac_kQMeanLag3Hi;
- mean_val4 = WebRtcIsac_kQMeanLag4Hi;
- lower_limit = WebRtcIsac_kQindexLowerLimitLagHi;
- upper_limit = WebRtcIsac_kQindexUpperLimitLagHi;
- }
-
-
- /* find quantization index */
- for (k=0; k<4; k++)
- {
- /* transform */
- C = 0.0;
- for (j=0; j<PITCH_SUBFRAMES; j++)
- C += WebRtcIsac_kTransform[k][j] * PitchLags[j];
-
- /* quantize */
- index[k] = WebRtcIsac_lrint(C / StepSize);
-
- /* check that the index is not outside the boundaries of the table */
- if (index[k] < lower_limit[k]) index[k] = lower_limit[k];
- else if (index[k] > upper_limit[k]) index[k] = upper_limit[k];
- index[k] -= lower_limit[k];
-
- /* Save data for creation of multiple bit streams */
- encData->pitchIndex[PITCH_SUBFRAMES*encData->startIdx + k] = index[k];
- }
-
- /* unquantize back to transform coefficients and do the inverse transform: S = T'*C */
- C = (index[0] + lower_limit[0]) * StepSize;
- for (k=0; k<PITCH_SUBFRAMES; k++)
- PitchLags[k] = WebRtcIsac_kTransformTranspose[k][0] * C;
- C = mean_val2[index[1]];
- for (k=0; k<PITCH_SUBFRAMES; k++)
- PitchLags[k] += WebRtcIsac_kTransformTranspose[k][1] * C;
- C = mean_val3[index[2]];
- for (k=0; k<PITCH_SUBFRAMES; k++)
- PitchLags[k] += WebRtcIsac_kTransformTranspose[k][2] * C;
- C = mean_val4[index[3]];
- for (k=0; k<PITCH_SUBFRAMES; k++)
- PitchLags[k] += WebRtcIsac_kTransformTranspose[k][3] * C;
-
-
- /* entropy coding of quantization pitch lags */
- WebRtcIsac_EncHistMulti(streamdata, index, cdf, PITCH_SUBFRAMES);
-
-}
-
-
-
-/* Routines for in-band signaling of bandwidth estimation */
-/* Histograms based on uniform distribution of indices */
-/* Move global variables later! */
-
-
-/* cdf array for frame length indicator */
-const WebRtc_UWord16 WebRtcIsac_kFrameLengthCdf[4] = {
- 0, 21845, 43690, 65535};
-
-/* pointer to cdf array for frame length indicator */
-const WebRtc_UWord16 *WebRtcIsac_kFrameLengthCdf_ptr[1] = {WebRtcIsac_kFrameLengthCdf};
-
-/* initial cdf index for decoder of frame length indicator */
-const WebRtc_UWord16 WebRtcIsac_kFrameLengthInitIndex[1] = {1};
-
-
-int WebRtcIsac_DecodeFrameLen(Bitstr *streamdata,
- WebRtc_Word16 *framesamples)
-{
-
- int frame_mode, err;
-
- err = 0;
- /* entropy decoding of frame length [1:30ms,2:60ms] */
- err = WebRtcIsac_DecHistOneStepMulti(&frame_mode, streamdata, WebRtcIsac_kFrameLengthCdf_ptr, WebRtcIsac_kFrameLengthInitIndex, 1);
- if (err<0) // error check
- return -ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH;
-
- switch(frame_mode) {
- case 1:
- *framesamples = 480; /* 30ms */
- break;
- case 2:
- *framesamples = 960; /* 60ms */
- break;
- default:
- err = -ISAC_DISALLOWED_FRAME_MODE_DECODER;
- }
-
- return err;
-}
-
-int WebRtcIsac_EncodeFrameLen(WebRtc_Word16 framesamples, Bitstr *streamdata) {
-
- int frame_mode, status;
-
- status = 0;
- frame_mode = 0;
- /* entropy coding of frame length [1:480 samples,2:960 samples] */
- switch(framesamples) {
- case 480:
- frame_mode = 1;
- break;
- case 960:
- frame_mode = 2;
- break;
- default:
- status = - ISAC_DISALLOWED_FRAME_MODE_ENCODER;
- }
-
- if (status < 0)
- return status;
-
- WebRtcIsac_EncHistMulti(streamdata, &frame_mode, WebRtcIsac_kFrameLengthCdf_ptr, 1);
-
- return status;
-}
-
-/* cdf array for estimated bandwidth */
-static const WebRtc_UWord16 kBwCdf[25] = {
- 0, 2731, 5461, 8192, 10923, 13653, 16384, 19114, 21845, 24576, 27306, 30037,
- 32768, 35498, 38229, 40959, 43690, 46421, 49151, 51882, 54613, 57343, 60074,
- 62804, 65535};
-
-/* pointer to cdf array for estimated bandwidth */
-static const WebRtc_UWord16 *kBwCdfPtr[1] = { kBwCdf };
-
-/* initial cdf index for decoder of estimated bandwidth*/
-static const WebRtc_UWord16 kBwInitIndex[1] = { 7 };
-
-
-int WebRtcIsac_DecodeSendBW(Bitstr *streamdata, WebRtc_Word16 *BWno) {
-
- int BWno32, err;
-
- /* entropy decoding of sender's BW estimation [0..23] */
- err = WebRtcIsac_DecHistOneStepMulti(&BWno32, streamdata, kBwCdfPtr, kBwInitIndex, 1);
- if (err<0) // error check
- return -ISAC_RANGE_ERROR_DECODE_BANDWIDTH;
- *BWno = (WebRtc_Word16)BWno32;
- return err;
-
-}
-
-void WebRtcIsac_EncodeReceiveBw(int *BWno, Bitstr *streamdata) {
-
- /* entropy encoding of receiver's BW estimation [0..23] */
- WebRtcIsac_EncHistMulti(streamdata, BWno, kBwCdfPtr, 1);
-
-}
-
-
-/* estimate code length of LPC Coef */
-void WebRtcIsac_TranscodeLPCCoef(double *LPCCoef_lo, double *LPCCoef_hi, int model,
- int *index_g) {
-
- int j, k, n, pos, pos2, posg, offsg, offs2;
- int index_ovr_g[KLT_ORDER_GAIN];
- double tmpcoeffs_g[KLT_ORDER_GAIN];
- double tmpcoeffs2_g[KLT_ORDER_GAIN];
- double sum;
-
- /* log gains, mean removal and scaling */
- posg = 0;
- for (k=0; k<SUBFRAMES; k++) {
- tmpcoeffs_g[posg] = log(LPCCoef_lo[(LPC_LOBAND_ORDER+1)*k]);
- tmpcoeffs_g[posg] -= WebRtcIsac_kLpcMeansGain[model][posg];
- tmpcoeffs_g[posg] *= LPC_GAIN_SCALE;
- posg++;
- tmpcoeffs_g[posg] = log(LPCCoef_hi[(LPC_HIBAND_ORDER+1)*k]);
- tmpcoeffs_g[posg] -= WebRtcIsac_kLpcMeansGain[model][posg];
- tmpcoeffs_g[posg] *= LPC_GAIN_SCALE;
- posg++;
- }
-
- /* KLT */
-
- /* left transform */
- offsg = 0;
- for (j=0; j<SUBFRAMES; j++) {
- posg = offsg;
- for (k=0; k<LPC_GAIN_ORDER; k++) {
- sum = 0;
- pos = offsg;
- pos2 = k;
- for (n=0; n<LPC_GAIN_ORDER; n++) {
- sum += tmpcoeffs_g[pos++] * WebRtcIsac_kKltT1Gain[model][pos2];
- pos2 += LPC_GAIN_ORDER;
- }
- tmpcoeffs2_g[posg++] = sum;
- }
- offsg += LPC_GAIN_ORDER;
- }
-
- /* right transform */
- offsg = 0;
- offs2 = 0;
- for (j=0; j<SUBFRAMES; j++) {
- posg = offsg;
- for (k=0; k<LPC_GAIN_ORDER; k++) {
- sum = 0;
- pos = k;
- pos2 = offs2;
- for (n=0; n<SUBFRAMES; n++) {
- sum += tmpcoeffs2_g[pos] * WebRtcIsac_kKltT2Gain[model][pos2++];
- pos += LPC_GAIN_ORDER;
- }
- tmpcoeffs_g[posg++] = sum;
- }
- offs2 += SUBFRAMES;
- offsg += LPC_GAIN_ORDER;
- }
-
-
- /* quantize coefficients */
- for (k=0; k<KLT_ORDER_GAIN; k++) {
-
- /* get index */
- pos = WebRtcIsac_kQKltSelIndGain[k];
- pos2= WebRtcIsac_lrint(tmpcoeffs_g[pos] / KLT_STEPSIZE);
- index_g[k] = (pos2) + WebRtcIsac_kQKltQuantMinGain[k];
- if (index_g[k] < 0) {
- index_g[k] = 0;
- }
- else if (index_g[k] > WebRtcIsac_kQKltMaxIndGain[k]) {
- index_g[k] = WebRtcIsac_kQKltMaxIndGain[k];
- }
- index_ovr_g[k] = WebRtcIsac_kQKltOffsetGain[model][k]+index_g[k];
-
- /* find quantization levels for coefficients */
- tmpcoeffs_g[WebRtcIsac_kQKltSelIndGain[k]] = WebRtcIsac_kQKltLevelsGain[WebRtcIsac_kQKltOfLevelsGain[model]+index_ovr_g[k]];
- }
-}
-
-
-/* decode & dequantize LPC Coef */
-int
-WebRtcIsac_DecodeLpcCoefUB(
- Bitstr* streamdata,
- double* lpcVecs,
- double* percepFilterGains,
- WebRtc_Word16 bandwidth)
-{
- int index_s[KLT_ORDER_SHAPE];
-
- double U[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
- int err;
-
- /* entropy decoding of quantization indices */
- switch(bandwidth)
- {
- case isac12kHz:
- {
- err = WebRtcIsac_DecHistOneStepMulti(index_s, streamdata,
- WebRtcIsac_kLpcShapeCdfMatUb12, WebRtcIsac_kLpcShapeEntropySearchUb12,
- UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME);
- break;
- }
- case isac16kHz:
- {
- err = WebRtcIsac_DecHistOneStepMulti(index_s, streamdata,
- WebRtcIsac_kLpcShapeCdfMatUb16, WebRtcIsac_kLpcShapeEntropySearchUb16,
- UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME);
- break;
- }
- default:
- return -1;
- }
-
- if (err<0) // error check
- {
- return err;
- }
-
- WebRtcIsac_DequantizeLpcParam(index_s, lpcVecs, bandwidth);
- WebRtcIsac_CorrelateInterVec(lpcVecs, U, bandwidth);
- WebRtcIsac_CorrelateIntraVec(U, lpcVecs, bandwidth);
- WebRtcIsac_AddLarMean(lpcVecs, bandwidth);
-
-
- WebRtcIsac_DecodeLpcGainUb(percepFilterGains, streamdata);
-
- if(bandwidth == isac16kHz)
- {
- // decode another set of Gains
- WebRtcIsac_DecodeLpcGainUb(&percepFilterGains[SUBFRAMES], streamdata);
- }
-
- return 0;
-}
-
-WebRtc_Word16
-WebRtcIsac_EncodeBandwidth(
- enum ISACBandwidth bandwidth,
- Bitstr* streamData)
-{
- int bandwidthMode;
- switch(bandwidth)
- {
- case isac12kHz:
- {
- bandwidthMode = 0;
- break;
- }
- case isac16kHz:
- {
- bandwidthMode = 1;
- break;
- }
- default:
- return -ISAC_DISALLOWED_ENCODER_BANDWIDTH;
- }
-
- WebRtcIsac_EncHistMulti(streamData, &bandwidthMode,
- kOneBitEqualProbCdf_ptr, 1);
- return 0;
-}
-
-WebRtc_Word16
-WebRtcIsac_DecodeBandwidth(
- Bitstr* streamData,
- enum ISACBandwidth* bandwidth)
-{
- int bandwidthMode;
-
- if(WebRtcIsac_DecHistOneStepMulti(&bandwidthMode, streamData,
- kOneBitEqualProbCdf_ptr,
- kOneBitEqualProbInitIndex, 1) < 0)
- {
- // error check
- return -ISAC_RANGE_ERROR_DECODE_BANDWITH;
- }
-
- switch(bandwidthMode)
- {
- case 0:
- {
- *bandwidth = isac12kHz;
- break;
- }
- case 1:
- {
- *bandwidth = isac16kHz;
- break;
- }
- default:
- return -ISAC_DISALLOWED_BANDWIDTH_MODE_DECODER;
- }
- return 0;
-}
-
-WebRtc_Word16
-WebRtcIsac_EncodeJitterInfo(
- WebRtc_Word32 jitterIndex,
- Bitstr* streamData)
-{
- // This is to avoid LINUX warning until we change 'int' to
- // 'Word32'
- int intVar;
-
- if((jitterIndex < 0) || (jitterIndex > 1))
- {
- return -1;
- }
- intVar = (int)(jitterIndex);
- // Use the same CDF table as for bandwidth
- // both take two values with equal probability
- WebRtcIsac_EncHistMulti(streamData, &intVar,
- kOneBitEqualProbCdf_ptr, 1);
- return 0;
-
-}
-
-WebRtc_Word16
-WebRtcIsac_DecodeJitterInfo(
- Bitstr* streamData,
- WebRtc_Word32* jitterInfo)
-{
- int intVar;
-
- // Use the same CDF table as for bandwidth
- // both take two values with equal probability
- if(WebRtcIsac_DecHistOneStepMulti(&intVar, streamData,
- kOneBitEqualProbCdf_ptr,
- kOneBitEqualProbInitIndex, 1) < 0)
- {
- // error check
- return -ISAC_RANGE_ERROR_DECODE_BANDWITH;
- }
- *jitterInfo = (WebRtc_Word16)(intVar);
- return 0;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/entropy_coding.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/entropy_coding.h
deleted file mode 100644
index 8446bcf..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/entropy_coding.h
+++ /dev/null
@@ -1,412 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * entropy_coding.h
- *
- * This header file declares all of the functions used to arithmetically
- * encode the iSAC bistream
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_
-
-#include "structs.h"
-
-/* decode complex spectrum (return number of bytes in stream) */
-int WebRtcIsac_DecodeSpecLb(Bitstr *streamdata,
- double *fr,
- double *fi,
- WebRtc_Word16 AvgPitchGain_Q12);
-
-/******************************************************************************
- * WebRtcIsac_DecodeSpecUB16()
- * Decode real and imaginary part of the DFT coefficients, given a bit-stream.
- * This function is called when the codec is in 0-16 kHz bandwidth.
- * The decoded DFT coefficient can be transformed to time domain by
- * WebRtcIsac_Time2Spec().
- *
- * Input:
- * - streamdata : pointer to a stucture containg the encoded
- * data and theparameters needed for entropy
- * coding.
- *
- * Output:
- * -*fr : pointer to a buffer where the real part of DFT
- * coefficients are written to.
- * -*fi : pointer to a buffer where the imaginary part
- * of DFT coefficients are written to.
- *
- * Return value : < 0 if an error occures
- * 0 if succeeded.
- */
-int WebRtcIsac_DecodeSpecUB16(
- Bitstr* streamdata,
- double* fr,
- double* fi);
-
-
-/******************************************************************************
- * WebRtcIsac_DecodeSpecUB12()
- * Decode real and imaginary part of the DFT coefficients, given a bit-stream.
- * This function is called when the codec is in 0-12 kHz bandwidth.
- * The decoded DFT coefficient can be transformed to time domain by
- * WebRtcIsac_Time2Spec().
- *
- * Input:
- * - streamdata : pointer to a stucture containg the encoded
- * data and theparameters needed for entropy
- * coding.
- *
- * Output:
- * -*fr : pointer to a buffer where the real part of DFT
- * coefficients are written to.
- * -*fi : pointer to a buffer where the imaginary part
- * of DFT coefficients are written to.
- *
- * Return value : < 0 if an error occures
- * 0 if succeeded.
- */
-int WebRtcIsac_DecodeSpecUB12(
- Bitstr* streamdata,
- double* fr,
- double* fi);
-
-
-/* encode complex spectrum */
-int WebRtcIsac_EncodeSpecLb(const WebRtc_Word16* fr,
- const WebRtc_Word16* fi,
- Bitstr* streamdata,
- WebRtc_Word16 AvgPitchGain_Q12);
-
-
-/******************************************************************************
- * WebRtcIsac_EncodeSpecUB16()
- * Quantize and encode real and imaginary part of the DFT coefficients.
- * This function is called when the codec is in 0-16 kHz bandwidth.
- * The real and imaginary part are computed by calling WebRtcIsac_Time2Spec().
- *
- *
- * Input:
- * -*fr : pointer to a buffer where the real part of DFT
- * coefficients are stored.
- * -*fi : pointer to a buffer where the imaginary part
- * of DFT coefficients are stored.
- *
- * Output:
- * - streamdata : pointer to a stucture containg the encoded
- * data and theparameters needed for entropy
- * coding.
- *
- * Return value : < 0 if an error occures
- * 0 if succeeded.
- */
-int WebRtcIsac_EncodeSpecUB16(
- const WebRtc_Word16* fr,
- const WebRtc_Word16* fi,
- Bitstr* streamdata);
-
-
-/******************************************************************************
- * WebRtcIsac_EncodeSpecUB12()
- * Quantize and encode real and imaginary part of the DFT coefficients.
- * This function is called when the codec is in 0-12 kHz bandwidth.
- * The real and imaginary part are computed by calling WebRtcIsac_Time2Spec().
- *
- *
- * Input:
- * -*fr : pointer to a buffer where the real part of DFT
- * coefficients are stored.
- * -*fi : pointer to a buffer where the imaginary part
- * of DFT coefficients are stored.
- *
- * Output:
- * - streamdata : pointer to a stucture containg the encoded
- * data and theparameters needed for entropy
- * coding.
- *
- * Return value : < 0 if an error occures
- * 0 if succeeded.
- */
-int WebRtcIsac_EncodeSpecUB12(
- const WebRtc_Word16* fr,
- const WebRtc_Word16* fi,
- Bitstr* streamdata);
-
-
-/* decode & dequantize LPC Coef */
-int WebRtcIsac_DecodeLpcCoef(Bitstr *streamdata, double *LPCCoef, int *outmodel);
-int WebRtcIsac_DecodeLpcCoefUB(
- Bitstr* streamdata,
- double* lpcVecs,
- double* percepFilterGains,
- WebRtc_Word16 bandwidth);
-
-int WebRtcIsac_DecodeLpc(Bitstr *streamdata, double *LPCCoef_lo, double *LPCCoef_hi, int *outmodel);
-
-/* quantize & code LPC Coef */
-void WebRtcIsac_EncodeLpcLb(double *LPCCoef_lo, double *LPCCoef_hi, int *model, double *size, Bitstr *streamdata, ISAC_SaveEncData_t* encData);
-void WebRtcIsac_EncodeLpcGainLb(double *LPCCoef_lo, double *LPCCoef_hi, int model, Bitstr *streamdata, ISAC_SaveEncData_t* encData);
-
-/******************************************************************************
- * WebRtcIsac_EncodeLpcUB()
- * Encode LPC parameters, given as A-polynomial, of upper-band. The encoding
- * is performed in LAR domain.
- * For the upper-band, we compute and encode LPC of some sub-frames, LPC of
- * other sub-frames are computed by linear interpolation, in LAR domain. This
- * function performs the interpolation and returns the LPC of all sub-frames.
- *
- * Inputs:
- * - lpcCoef : a buffer containing A-polynomials of sub-frames
- * (excluding first coefficient that is 1).
- * - bandwidth : specifies if the codec is operating at 0-12 kHz
- * or 0-16 kHz mode.
- *
- * Input/output:
- * - streamdata : pointer to a stucture containg the encoded
- * data and theparameters needed for entropy
- * coding.
- *
- * Output:
- * - interpolLPCCoeff : Decoded and interpolated LPC (A-polynomial)
- * of all sub-frames.
- * If LP analysis is of order K, and there are N
- * sub-frames then this is a buffer of size
- * (k + 1) * N, each vector starts with the LPC gain
- * of the corresponding sub-frame. The LPC gains
- * are encoded and inserted after this function is
- * called. The first A-coefficient which is 1 is not
- * included.
- *
- * Return value : 0 if encoding is successful,
- * <0 if failed to encode.
- */
-WebRtc_Word16 WebRtcIsac_EncodeLpcUB(
- double* lpcCoeff,
- Bitstr* streamdata,
- double* interpolLPCCoeff,
- WebRtc_Word16 bandwidth,
- ISACUBSaveEncDataStruct* encData);
-
-/******************************************************************************
- * WebRtcIsac_DecodeInterpolLpcUb()
- * Decode LPC coefficients and interpolate to get the coefficients fo all
- * sub-frmaes.
- *
- * Inputs:
- * - bandwidth : spepecifies if the codec is in 0-12 kHz or
- * 0-16 kHz mode.
- *
- * Input/output:
- * - streamdata : pointer to a stucture containg the encoded
- * data and theparameters needed for entropy
- * coding.
- *
- * Output:
- * - percepFilterParam : Decoded and interpolated LPC (A-polynomial) of
- * all sub-frames.
- * If LP analysis is of order K, and there are N
- * sub-frames then this is a buffer of size
- * (k + 1) * N, each vector starts with the LPC gain
- * of the corresponding sub-frame. The LPC gains
- * are encoded and inserted after this function is
- * called. The first A-coefficient which is 1 is not
- * included.
- *
- * Return value : 0 if encoding is successful,
- * <0 if failed to encode.
- */
-WebRtc_Word16 WebRtcIsac_DecodeInterpolLpcUb(
- Bitstr* streamdata,
- double* percepFilterParam,
- WebRtc_Word16 bandwidth);
-
-/* decode & dequantize RC */
-int WebRtcIsac_DecodeRc(Bitstr *streamdata, WebRtc_Word16 *RCQ15);
-
-/* quantize & code RC */
-void WebRtcIsac_EncodeRc(WebRtc_Word16 *RCQ15, Bitstr *streamdata);
-
-/* decode & dequantize squared Gain */
-int WebRtcIsac_DecodeGain2(Bitstr *streamdata, WebRtc_Word32 *Gain2);
-
-/* quantize & code squared Gain (input is squared gain) */
-int WebRtcIsac_EncodeGain2(WebRtc_Word32 *gain2, Bitstr *streamdata);
-
-void WebRtcIsac_EncodePitchGain(WebRtc_Word16* PitchGains_Q12, Bitstr* streamdata, ISAC_SaveEncData_t* encData);
-
-void WebRtcIsac_EncodePitchLag(double* PitchLags, WebRtc_Word16* PitchGain_Q12, Bitstr* streamdata, ISAC_SaveEncData_t* encData);
-
-int WebRtcIsac_DecodePitchGain(Bitstr *streamdata, WebRtc_Word16 *PitchGain_Q12);
-int WebRtcIsac_DecodePitchLag(Bitstr *streamdata, WebRtc_Word16 *PitchGain_Q12, double *PitchLag);
-
-int WebRtcIsac_DecodeFrameLen(Bitstr *streamdata, WebRtc_Word16 *framelength);
-int WebRtcIsac_EncodeFrameLen(WebRtc_Word16 framelength, Bitstr *streamdata);
-int WebRtcIsac_DecodeSendBW(Bitstr *streamdata, WebRtc_Word16 *BWno);
-void WebRtcIsac_EncodeReceiveBw(int *BWno, Bitstr *streamdata);
-
-/* step-down */
-void WebRtcIsac_Poly2Rc(double *a, int N, double *RC);
-
-/* step-up */
-void WebRtcIsac_Rc2Poly(double *RC, int N, double *a);
-
-void WebRtcIsac_TranscodeLPCCoef(double *LPCCoef_lo, double *LPCCoef_hi, int model,
- int *index_g);
-
-
-/******************************************************************************
- * WebRtcIsac_EncodeLpcGainUb()
- * Encode LPC gains of sub-Frames.
- *
- * Input/outputs:
- * - lpGains : a buffer which contains 'SUBFRAME' number of
- * LP gains to be encoded. The input values are
- * overwritten by the quantized values.
- * - streamdata : pointer to a stucture containg the encoded
- * data and theparameters needed for entropy
- * coding.
- *
- * Output:
- * - lpcGainIndex : quantization indices for lpc gains, these will
- * be stored to be used for FEC.
- */
-void WebRtcIsac_EncodeLpcGainUb(
- double* lpGains,
- Bitstr* streamdata,
- int* lpcGainIndex);
-
-
-/******************************************************************************
- * WebRtcIsac_EncodeLpcGainUb()
- * Store LPC gains of sub-Frames in 'streamdata'.
- *
- * Input:
- * - lpGains : a buffer which contains 'SUBFRAME' number of
- * LP gains to be encoded.
- * Input/outputs:
- * - streamdata : pointer to a stucture containg the encoded
- * data and theparameters needed for entropy
- * coding.
- *
- */
-void WebRtcIsac_StoreLpcGainUb(
- double* lpGains,
- Bitstr* streamdata);
-
-
-/******************************************************************************
- * WebRtcIsac_DecodeLpcGainUb()
- * Decode the LPC gain of sub-frames.
- *
- * Input/output:
- * - streamdata : pointer to a stucture containg the encoded
- * data and theparameters needed for entropy
- * coding.
- *
- * Output:
- * - lpGains : a buffer where decoded LPC gians will be stored.
- *
- * Return value : 0 if succeeded.
- * <0 if failed.
- */
-WebRtc_Word16 WebRtcIsac_DecodeLpcGainUb(
- double* lpGains,
- Bitstr* streamdata);
-
-
-/******************************************************************************
- * WebRtcIsac_EncodeBandwidth()
- * Encode if the bandwidth of encoded audio is 0-12 kHz or 0-16 kHz.
- *
- * Input:
- * - bandwidth : an enumerator specifying if the codec in is
- * 0-12 kHz or 0-16 kHz mode.
- *
- * Input/output:
- * - streamdata : pointer to a stucture containg the encoded
- * data and theparameters needed for entropy
- * coding.
- *
- * Return value : 0 if succeeded.
- * <0 if failed.
- */
-WebRtc_Word16 WebRtcIsac_EncodeBandwidth(
- enum ISACBandwidth bandwidth,
- Bitstr* streamData);
-
-
-/******************************************************************************
- * WebRtcIsac_DecodeBandwidth()
- * Decode the bandwidth of the encoded audio, i.e. if the bandwidth is 0-12 kHz
- * or 0-16 kHz.
- *
- * Input/output:
- * - streamdata : pointer to a stucture containg the encoded
- * data and theparameters needed for entropy
- * coding.
- *
- * Output:
- * - bandwidth : an enumerator specifying if the codec is in
- * 0-12 kHz or 0-16 kHz mode.
- *
- * Return value : 0 if succeeded.
- * <0 if failed.
- */
-WebRtc_Word16 WebRtcIsac_DecodeBandwidth(
- Bitstr* streamData,
- enum ISACBandwidth* bandwidth);
-
-
-/******************************************************************************
- * WebRtcIsac_EncodeJitterInfo()
- * Decode the jitter information.
- *
- * Input/output:
- * - streamdata : pointer to a stucture containg the encoded
- * data and theparameters needed for entropy
- * coding.
- *
- * Input:
- * - jitterInfo : one bit of info specifying if the channel is
- * in high/low jitter. Zero indicates low jitter
- * and one indicates high jitter.
- *
- * Return value : 0 if succeeded.
- * <0 if failed.
- */
-WebRtc_Word16 WebRtcIsac_EncodeJitterInfo(
- WebRtc_Word32 jitterIndex,
- Bitstr* streamData);
-
-
-/******************************************************************************
- * WebRtcIsac_DecodeJitterInfo()
- * Decode the jitter information.
- *
- * Input/output:
- * - streamdata : pointer to a stucture containg the encoded
- * data and theparameters needed for entropy
- * coding.
- *
- * Output:
- * - jitterInfo : one bit of info specifying if the channel is
- * in high/low jitter. Zero indicates low jitter
- * and one indicates high jitter.
- *
- * Return value : 0 if succeeded.
- * <0 if failed.
- */
-WebRtc_Word16 WebRtcIsac_DecodeJitterInfo(
- Bitstr* streamData,
- WebRtc_Word32* jitterInfo);
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_ENTROPY_CODING_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/fft.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/fft.c
deleted file mode 100644
index c824798..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/fft.c
+++ /dev/null
@@ -1,947 +0,0 @@
-/*
- * Copyright(c)1995,97 Mark Olesen <olesen@me.QueensU.CA>
- * Queen's Univ at Kingston (Canada)
- *
- * Permission to use, copy, modify, and distribute this software for
- * any purpose without fee is hereby granted, provided that this
- * entire notice is included in all copies of any software which is
- * or includes a copy or modification of this software and in all
- * copies of the supporting documentation for such software.
- *
- * THIS SOFTWARE IS BEING PROVIDED "AS IS", WITHOUT ANY EXPRESS OR
- * IMPLIED WARRANTY. IN PARTICULAR, NEITHER THE AUTHOR NOR QUEEN'S
- * UNIVERSITY AT KINGSTON MAKES ANY REPRESENTATION OR WARRANTY OF ANY
- * KIND CONCERNING THE MERCHANTABILITY OF THIS SOFTWARE OR ITS
- * FITNESS FOR ANY PARTICULAR PURPOSE.
- *
- * All of which is to say that you can do what you like with this
- * source code provided you don't try to sell it as your own and you
- * include an unaltered copy of this message (including the
- * copyright).
- *
- * It is also implicitly understood that bug fixes and improvements
- * should make their way back to the general Internet community so
- * that everyone benefits.
- *
- * Changes:
- * Trivial type modifications by the WebRTC authors.
- */
-
-
-/*
- * File:
- * WebRtcIsac_Fftn.c
- *
- * Public:
- * WebRtcIsac_Fftn / fftnf ();
- *
- * Private:
- * WebRtcIsac_Fftradix / fftradixf ();
- *
- * Descript:
- * multivariate complex Fourier transform, computed in place
- * using mixed-radix Fast Fourier Transform algorithm.
- *
- * Fortran code by:
- * RC Singleton, Stanford Research Institute, Sept. 1968
- *
- * translated by f2c (version 19950721).
- *
- * int WebRtcIsac_Fftn (int ndim, const int dims[], REAL Re[], REAL Im[],
- * int iSign, double scaling);
- *
- * NDIM = the total number dimensions
- * DIMS = a vector of array sizes
- * if NDIM is zero then DIMS must be zero-terminated
- *
- * RE and IM hold the real and imaginary components of the data, and return
- * the resulting real and imaginary Fourier coefficients. Multidimensional
- * data *must* be allocated contiguously. There is no limit on the number
- * of dimensions.
- *
- * ISIGN = the sign of the complex exponential (ie, forward or inverse FFT)
- * the magnitude of ISIGN (normally 1) is used to determine the
- * correct indexing increment (see below).
- *
- * SCALING = normalizing constant by which the final result is *divided*
- * if SCALING == -1, normalize by total dimension of the transform
- * if SCALING < -1, normalize by the square-root of the total dimension
- *
- * example:
- * tri-variate transform with Re[n1][n2][n3], Im[n1][n2][n3]
- *
- * int dims[3] = {n1,n2,n3}
- * WebRtcIsac_Fftn (3, dims, Re, Im, 1, scaling);
- *
- *-----------------------------------------------------------------------*
- * int WebRtcIsac_Fftradix (REAL Re[], REAL Im[], size_t nTotal, size_t nPass,
- * size_t nSpan, int iSign, size_t max_factors,
- * size_t max_perm);
- *
- * RE, IM - see above documentation
- *
- * Although there is no limit on the number of dimensions, WebRtcIsac_Fftradix() must
- * be called once for each dimension, but the calls may be in any order.
- *
- * NTOTAL = the total number of complex data values
- * NPASS = the dimension of the current variable
- * NSPAN/NPASS = the spacing of consecutive data values while indexing the
- * current variable
- * ISIGN - see above documentation
- *
- * example:
- * tri-variate transform with Re[n1][n2][n3], Im[n1][n2][n3]
- *
- * WebRtcIsac_Fftradix (Re, Im, n1*n2*n3, n1, n1, 1, maxf, maxp);
- * WebRtcIsac_Fftradix (Re, Im, n1*n2*n3, n2, n1*n2, 1, maxf, maxp);
- * WebRtcIsac_Fftradix (Re, Im, n1*n2*n3, n3, n1*n2*n3, 1, maxf, maxp);
- *
- * single-variate transform,
- * NTOTAL = N = NSPAN = (number of complex data values),
- *
- * WebRtcIsac_Fftradix (Re, Im, n, n, n, 1, maxf, maxp);
- *
- * The data can also be stored in a single array with alternating real and
- * imaginary parts, the magnitude of ISIGN is changed to 2 to give correct
- * indexing increment, and data [0] and data [1] used to pass the initial
- * addresses for the sequences of real and imaginary values,
- *
- * example:
- * REAL data [2*NTOTAL];
- * WebRtcIsac_Fftradix ( &data[0], &data[1], NTOTAL, nPass, nSpan, 2, maxf, maxp);
- *
- * for temporary allocation:
- *
- * MAX_FACTORS >= the maximum prime factor of NPASS
- * MAX_PERM >= the number of prime factors of NPASS. In addition,
- * if the square-free portion K of NPASS has two or more prime
- * factors, then MAX_PERM >= (K-1)
- *
- * storage in FACTOR for a maximum of 15 prime factors of NPASS. if NPASS
- * has more than one square-free factor, the product of the square-free
- * factors must be <= 210 array storage for maximum prime factor of 23 the
- * following two constants should agree with the array dimensions.
- *
- *----------------------------------------------------------------------*/
-#include "fft.h"
-
-#include <stdlib.h>
-#include <math.h>
-
-
-
-/* double precision routine */
-static int
-WebRtcIsac_Fftradix (double Re[], double Im[],
- size_t nTotal, size_t nPass, size_t nSpan, int isign,
- int max_factors, unsigned int max_perm,
- FFTstr *fftstate);
-
-
-
-#ifndef M_PI
-# define M_PI 3.14159265358979323846264338327950288
-#endif
-
-#ifndef SIN60
-# define SIN60 0.86602540378443865 /* sin(60 deg) */
-# define COS72 0.30901699437494742 /* cos(72 deg) */
-# define SIN72 0.95105651629515357 /* sin(72 deg) */
-#endif
-
-# define REAL double
-# define FFTN WebRtcIsac_Fftn
-# define FFTNS "fftn"
-# define FFTRADIX WebRtcIsac_Fftradix
-# define FFTRADIXS "fftradix"
-
-
-int WebRtcIsac_Fftns(unsigned int ndim, const int dims[],
- double Re[],
- double Im[],
- int iSign,
- double scaling,
- FFTstr *fftstate)
-{
-
- size_t nSpan, nPass, nTotal;
- unsigned int i;
- int ret, max_factors, max_perm;
-
- /*
- * tally the number of elements in the data array
- * and determine the number of dimensions
- */
- nTotal = 1;
- if (ndim && dims [0])
- {
- for (i = 0; i < ndim; i++)
- {
- if (dims [i] <= 0)
- {
- return -1;
- }
- nTotal *= dims [i];
- }
- }
- else
- {
- ndim = 0;
- for (i = 0; dims [i]; i++)
- {
- if (dims [i] <= 0)
- {
- return -1;
- }
- nTotal *= dims [i];
- ndim++;
- }
- }
-
- /* determine maximum number of factors and permuations */
-#if 1
- /*
- * follow John Beale's example, just use the largest dimension and don't
- * worry about excess allocation. May be someone else will do it?
- */
- max_factors = max_perm = 1;
- for (i = 0; i < ndim; i++)
- {
- nSpan = dims [i];
- if ((int)nSpan > max_factors)
- {
- max_factors = (int)nSpan;
- }
- if ((int)nSpan > max_perm)
- {
- max_perm = (int)nSpan;
- }
- }
-#else
- /* use the constants used in the original Fortran code */
- max_factors = 23;
- max_perm = 209;
-#endif
- /* loop over the dimensions: */
- nPass = 1;
- for (i = 0; i < ndim; i++)
- {
- nSpan = dims [i];
- nPass *= nSpan;
- ret = FFTRADIX (Re, Im, nTotal, nSpan, nPass, iSign,
- max_factors, max_perm, fftstate);
- /* exit, clean-up already done */
- if (ret)
- return ret;
- }
-
- /* Divide through by the normalizing constant: */
- if (scaling && scaling != 1.0)
- {
- if (iSign < 0) iSign = -iSign;
- if (scaling < 0.0)
- {
- scaling = (double)nTotal;
- if (scaling < -1.0)
- scaling = sqrt (scaling);
- }
- scaling = 1.0 / scaling; /* multiply is often faster */
- for (i = 0; i < nTotal; i += iSign)
- {
- Re [i] *= scaling;
- Im [i] *= scaling;
- }
- }
- return 0;
-}
-
-/*
- * singleton's mixed radix routine
- *
- * could move allocation out to WebRtcIsac_Fftn(), but leave it here so that it's
- * possible to make this a standalone function
- */
-
-static int FFTRADIX (REAL Re[],
- REAL Im[],
- size_t nTotal,
- size_t nPass,
- size_t nSpan,
- int iSign,
- int max_factors,
- unsigned int max_perm,
- FFTstr *fftstate)
-{
- int ii, mfactor, kspan, ispan, inc;
- int j, jc, jf, jj, k, k1, k2, k3, k4, kk, kt, nn, ns, nt;
-
-
- REAL radf;
- REAL c1, c2, c3, cd, aa, aj, ak, ajm, ajp, akm, akp;
- REAL s1, s2, s3, sd, bb, bj, bk, bjm, bjp, bkm, bkp;
-
- REAL *Rtmp = NULL; /* temp space for real part*/
- REAL *Itmp = NULL; /* temp space for imaginary part */
- REAL *Cos = NULL; /* Cosine values */
- REAL *Sin = NULL; /* Sine values */
-
- REAL s60 = SIN60; /* sin(60 deg) */
- REAL c72 = COS72; /* cos(72 deg) */
- REAL s72 = SIN72; /* sin(72 deg) */
- REAL pi2 = M_PI; /* use PI first, 2 PI later */
-
-
- fftstate->SpaceAlloced = 0;
- fftstate->MaxPermAlloced = 0;
-
-
- // initialize to avoid warnings
- k3 = c2 = c3 = s2 = s3 = 0.0;
-
- if (nPass < 2)
- return 0;
-
- /* allocate storage */
- if (fftstate->SpaceAlloced < max_factors * sizeof (REAL))
- {
-#ifdef SUN_BROKEN_REALLOC
- if (!fftstate->SpaceAlloced) /* first time */
- {
- fftstate->SpaceAlloced = max_factors * sizeof (REAL);
- }
- else
- {
-#endif
- fftstate->SpaceAlloced = max_factors * sizeof (REAL);
-#ifdef SUN_BROKEN_REALLOC
- }
-#endif
- }
- else
- {
- /* allow full use of alloc'd space */
- max_factors = fftstate->SpaceAlloced / sizeof (REAL);
- }
- if (fftstate->MaxPermAlloced < max_perm)
- {
-#ifdef SUN_BROKEN_REALLOC
- if (!fftstate->MaxPermAlloced) /* first time */
- else
-#endif
- fftstate->MaxPermAlloced = max_perm;
- }
- else
- {
- /* allow full use of alloc'd space */
- max_perm = fftstate->MaxPermAlloced;
- }
- if (fftstate->Tmp0 == NULL || fftstate->Tmp1 == NULL || fftstate->Tmp2 == NULL || fftstate->Tmp3 == NULL
- || fftstate->Perm == NULL) {
- return -1;
- }
-
- /* assign pointers */
- Rtmp = (REAL *) fftstate->Tmp0;
- Itmp = (REAL *) fftstate->Tmp1;
- Cos = (REAL *) fftstate->Tmp2;
- Sin = (REAL *) fftstate->Tmp3;
-
- /*
- * Function Body
- */
- inc = iSign;
- if (iSign < 0) {
- s72 = -s72;
- s60 = -s60;
- pi2 = -pi2;
- inc = -inc; /* absolute value */
- }
-
- /* adjust for strange increments */
- nt = inc * (int)nTotal;
- ns = inc * (int)nSpan;
- kspan = ns;
-
- nn = nt - inc;
- jc = ns / (int)nPass;
- radf = pi2 * (double) jc;
- pi2 *= 2.0; /* use 2 PI from here on */
-
- ii = 0;
- jf = 0;
- /* determine the factors of n */
- mfactor = 0;
- k = (int)nPass;
- while (k % 16 == 0) {
- mfactor++;
- fftstate->factor [mfactor - 1] = 4;
- k /= 16;
- }
- j = 3;
- jj = 9;
- do {
- while (k % jj == 0) {
- mfactor++;
- fftstate->factor [mfactor - 1] = j;
- k /= jj;
- }
- j += 2;
- jj = j * j;
- } while (jj <= k);
- if (k <= 4) {
- kt = mfactor;
- fftstate->factor [mfactor] = k;
- if (k != 1)
- mfactor++;
- } else {
- if (k - (k / 4 << 2) == 0) {
- mfactor++;
- fftstate->factor [mfactor - 1] = 2;
- k /= 4;
- }
- kt = mfactor;
- j = 2;
- do {
- if (k % j == 0) {
- mfactor++;
- fftstate->factor [mfactor - 1] = j;
- k /= j;
- }
- j = ((j + 1) / 2 << 1) + 1;
- } while (j <= k);
- }
- if (kt) {
- j = kt;
- do {
- mfactor++;
- fftstate->factor [mfactor - 1] = fftstate->factor [j - 1];
- j--;
- } while (j);
- }
-
- /* test that mfactors is in range */
- if (mfactor > NFACTOR)
- {
- return -1;
- }
-
- /* compute fourier transform */
- for (;;) {
- sd = radf / (double) kspan;
- cd = sin(sd);
- cd = 2.0 * cd * cd;
- sd = sin(sd + sd);
- kk = 0;
- ii++;
-
- switch (fftstate->factor [ii - 1]) {
- case 2:
- /* transform for factor of 2 (including rotation factor) */
- kspan /= 2;
- k1 = kspan + 2;
- do {
- do {
- k2 = kk + kspan;
- ak = Re [k2];
- bk = Im [k2];
- Re [k2] = Re [kk] - ak;
- Im [k2] = Im [kk] - bk;
- Re [kk] += ak;
- Im [kk] += bk;
- kk = k2 + kspan;
- } while (kk < nn);
- kk -= nn;
- } while (kk < jc);
- if (kk >= kspan)
- goto Permute_Results_Label; /* exit infinite loop */
- do {
- c1 = 1.0 - cd;
- s1 = sd;
- do {
- do {
- do {
- k2 = kk + kspan;
- ak = Re [kk] - Re [k2];
- bk = Im [kk] - Im [k2];
- Re [kk] += Re [k2];
- Im [kk] += Im [k2];
- Re [k2] = c1 * ak - s1 * bk;
- Im [k2] = s1 * ak + c1 * bk;
- kk = k2 + kspan;
- } while (kk < (nt-1));
- k2 = kk - nt;
- c1 = -c1;
- kk = k1 - k2;
- } while (kk > k2);
- ak = c1 - (cd * c1 + sd * s1);
- s1 = sd * c1 - cd * s1 + s1;
- c1 = 2.0 - (ak * ak + s1 * s1);
- s1 *= c1;
- c1 *= ak;
- kk += jc;
- } while (kk < k2);
- k1 += inc + inc;
- kk = (k1 - kspan + 1) / 2 + jc - 1;
- } while (kk < (jc + jc));
- break;
-
- case 4: /* transform for factor of 4 */
- ispan = kspan;
- kspan /= 4;
-
- do {
- c1 = 1.0;
- s1 = 0.0;
- do {
- do {
- k1 = kk + kspan;
- k2 = k1 + kspan;
- k3 = k2 + kspan;
- akp = Re [kk] + Re [k2];
- akm = Re [kk] - Re [k2];
- ajp = Re [k1] + Re [k3];
- ajm = Re [k1] - Re [k3];
- bkp = Im [kk] + Im [k2];
- bkm = Im [kk] - Im [k2];
- bjp = Im [k1] + Im [k3];
- bjm = Im [k1] - Im [k3];
- Re [kk] = akp + ajp;
- Im [kk] = bkp + bjp;
- ajp = akp - ajp;
- bjp = bkp - bjp;
- if (iSign < 0) {
- akp = akm + bjm;
- bkp = bkm - ajm;
- akm -= bjm;
- bkm += ajm;
- } else {
- akp = akm - bjm;
- bkp = bkm + ajm;
- akm += bjm;
- bkm -= ajm;
- }
- /* avoid useless multiplies */
- if (s1 == 0.0) {
- Re [k1] = akp;
- Re [k2] = ajp;
- Re [k3] = akm;
- Im [k1] = bkp;
- Im [k2] = bjp;
- Im [k3] = bkm;
- } else {
- Re [k1] = akp * c1 - bkp * s1;
- Re [k2] = ajp * c2 - bjp * s2;
- Re [k3] = akm * c3 - bkm * s3;
- Im [k1] = akp * s1 + bkp * c1;
- Im [k2] = ajp * s2 + bjp * c2;
- Im [k3] = akm * s3 + bkm * c3;
- }
- kk = k3 + kspan;
- } while (kk < nt);
-
- c2 = c1 - (cd * c1 + sd * s1);
- s1 = sd * c1 - cd * s1 + s1;
- c1 = 2.0 - (c2 * c2 + s1 * s1);
- s1 *= c1;
- c1 *= c2;
- /* values of c2, c3, s2, s3 that will get used next time */
- c2 = c1 * c1 - s1 * s1;
- s2 = 2.0 * c1 * s1;
- c3 = c2 * c1 - s2 * s1;
- s3 = c2 * s1 + s2 * c1;
- kk = kk - nt + jc;
- } while (kk < kspan);
- kk = kk - kspan + inc;
- } while (kk < jc);
- if (kspan == jc)
- goto Permute_Results_Label; /* exit infinite loop */
- break;
-
- default:
- /* transform for odd factors */
-#ifdef FFT_RADIX4
- return -1;
- break;
-#else /* FFT_RADIX4 */
- k = fftstate->factor [ii - 1];
- ispan = kspan;
- kspan /= k;
-
- switch (k) {
- case 3: /* transform for factor of 3 (optional code) */
- do {
- do {
- k1 = kk + kspan;
- k2 = k1 + kspan;
- ak = Re [kk];
- bk = Im [kk];
- aj = Re [k1] + Re [k2];
- bj = Im [k1] + Im [k2];
- Re [kk] = ak + aj;
- Im [kk] = bk + bj;
- ak -= 0.5 * aj;
- bk -= 0.5 * bj;
- aj = (Re [k1] - Re [k2]) * s60;
- bj = (Im [k1] - Im [k2]) * s60;
- Re [k1] = ak - bj;
- Re [k2] = ak + bj;
- Im [k1] = bk + aj;
- Im [k2] = bk - aj;
- kk = k2 + kspan;
- } while (kk < (nn - 1));
- kk -= nn;
- } while (kk < kspan);
- break;
-
- case 5: /* transform for factor of 5 (optional code) */
- c2 = c72 * c72 - s72 * s72;
- s2 = 2.0 * c72 * s72;
- do {
- do {
- k1 = kk + kspan;
- k2 = k1 + kspan;
- k3 = k2 + kspan;
- k4 = k3 + kspan;
- akp = Re [k1] + Re [k4];
- akm = Re [k1] - Re [k4];
- bkp = Im [k1] + Im [k4];
- bkm = Im [k1] - Im [k4];
- ajp = Re [k2] + Re [k3];
- ajm = Re [k2] - Re [k3];
- bjp = Im [k2] + Im [k3];
- bjm = Im [k2] - Im [k3];
- aa = Re [kk];
- bb = Im [kk];
- Re [kk] = aa + akp + ajp;
- Im [kk] = bb + bkp + bjp;
- ak = akp * c72 + ajp * c2 + aa;
- bk = bkp * c72 + bjp * c2 + bb;
- aj = akm * s72 + ajm * s2;
- bj = bkm * s72 + bjm * s2;
- Re [k1] = ak - bj;
- Re [k4] = ak + bj;
- Im [k1] = bk + aj;
- Im [k4] = bk - aj;
- ak = akp * c2 + ajp * c72 + aa;
- bk = bkp * c2 + bjp * c72 + bb;
- aj = akm * s2 - ajm * s72;
- bj = bkm * s2 - bjm * s72;
- Re [k2] = ak - bj;
- Re [k3] = ak + bj;
- Im [k2] = bk + aj;
- Im [k3] = bk - aj;
- kk = k4 + kspan;
- } while (kk < (nn-1));
- kk -= nn;
- } while (kk < kspan);
- break;
-
- default:
- if (k != jf) {
- jf = k;
- s1 = pi2 / (double) k;
- c1 = cos(s1);
- s1 = sin(s1);
- if (jf > max_factors){
- return -1;
- }
- Cos [jf - 1] = 1.0;
- Sin [jf - 1] = 0.0;
- j = 1;
- do {
- Cos [j - 1] = Cos [k - 1] * c1 + Sin [k - 1] * s1;
- Sin [j - 1] = Cos [k - 1] * s1 - Sin [k - 1] * c1;
- k--;
- Cos [k - 1] = Cos [j - 1];
- Sin [k - 1] = -Sin [j - 1];
- j++;
- } while (j < k);
- }
- do {
- do {
- k1 = kk;
- k2 = kk + ispan;
- ak = aa = Re [kk];
- bk = bb = Im [kk];
- j = 1;
- k1 += kspan;
- do {
- k2 -= kspan;
- j++;
- Rtmp [j - 1] = Re [k1] + Re [k2];
- ak += Rtmp [j - 1];
- Itmp [j - 1] = Im [k1] + Im [k2];
- bk += Itmp [j - 1];
- j++;
- Rtmp [j - 1] = Re [k1] - Re [k2];
- Itmp [j - 1] = Im [k1] - Im [k2];
- k1 += kspan;
- } while (k1 < k2);
- Re [kk] = ak;
- Im [kk] = bk;
- k1 = kk;
- k2 = kk + ispan;
- j = 1;
- do {
- k1 += kspan;
- k2 -= kspan;
- jj = j;
- ak = aa;
- bk = bb;
- aj = 0.0;
- bj = 0.0;
- k = 1;
- do {
- k++;
- ak += Rtmp [k - 1] * Cos [jj - 1];
- bk += Itmp [k - 1] * Cos [jj - 1];
- k++;
- aj += Rtmp [k - 1] * Sin [jj - 1];
- bj += Itmp [k - 1] * Sin [jj - 1];
- jj += j;
- if (jj > jf) {
- jj -= jf;
- }
- } while (k < jf);
- k = jf - j;
- Re [k1] = ak - bj;
- Im [k1] = bk + aj;
- Re [k2] = ak + bj;
- Im [k2] = bk - aj;
- j++;
- } while (j < k);
- kk += ispan;
- } while (kk < nn);
- kk -= nn;
- } while (kk < kspan);
- break;
- }
-
- /* multiply by rotation factor (except for factors of 2 and 4) */
- if (ii == mfactor)
- goto Permute_Results_Label; /* exit infinite loop */
- kk = jc;
- do {
- c2 = 1.0 - cd;
- s1 = sd;
- do {
- c1 = c2;
- s2 = s1;
- kk += kspan;
- do {
- do {
- ak = Re [kk];
- Re [kk] = c2 * ak - s2 * Im [kk];
- Im [kk] = s2 * ak + c2 * Im [kk];
- kk += ispan;
- } while (kk < nt);
- ak = s1 * s2;
- s2 = s1 * c2 + c1 * s2;
- c2 = c1 * c2 - ak;
- kk = kk - nt + kspan;
- } while (kk < ispan);
- c2 = c1 - (cd * c1 + sd * s1);
- s1 += sd * c1 - cd * s1;
- c1 = 2.0 - (c2 * c2 + s1 * s1);
- s1 *= c1;
- c2 *= c1;
- kk = kk - ispan + jc;
- } while (kk < kspan);
- kk = kk - kspan + jc + inc;
- } while (kk < (jc + jc));
- break;
-#endif /* FFT_RADIX4 */
- }
- }
-
- /* permute the results to normal order---done in two stages */
- /* permutation for square factors of n */
-Permute_Results_Label:
- fftstate->Perm [0] = ns;
- if (kt) {
- k = kt + kt + 1;
- if (mfactor < k)
- k--;
- j = 1;
- fftstate->Perm [k] = jc;
- do {
- fftstate->Perm [j] = fftstate->Perm [j - 1] / fftstate->factor [j - 1];
- fftstate->Perm [k - 1] = fftstate->Perm [k] * fftstate->factor [j - 1];
- j++;
- k--;
- } while (j < k);
- k3 = fftstate->Perm [k];
- kspan = fftstate->Perm [1];
- kk = jc;
- k2 = kspan;
- j = 1;
- if (nPass != nTotal) {
- /* permutation for multivariate transform */
- Permute_Multi_Label:
- do {
- do {
- k = kk + jc;
- do {
- /* swap Re [kk] <> Re [k2], Im [kk] <> Im [k2] */
- ak = Re [kk]; Re [kk] = Re [k2]; Re [k2] = ak;
- bk = Im [kk]; Im [kk] = Im [k2]; Im [k2] = bk;
- kk += inc;
- k2 += inc;
- } while (kk < (k-1));
- kk += ns - jc;
- k2 += ns - jc;
- } while (kk < (nt-1));
- k2 = k2 - nt + kspan;
- kk = kk - nt + jc;
- } while (k2 < (ns-1));
- do {
- do {
- k2 -= fftstate->Perm [j - 1];
- j++;
- k2 = fftstate->Perm [j] + k2;
- } while (k2 > fftstate->Perm [j - 1]);
- j = 1;
- do {
- if (kk < (k2-1))
- goto Permute_Multi_Label;
- kk += jc;
- k2 += kspan;
- } while (k2 < (ns-1));
- } while (kk < (ns-1));
- } else {
- /* permutation for single-variate transform (optional code) */
- Permute_Single_Label:
- do {
- /* swap Re [kk] <> Re [k2], Im [kk] <> Im [k2] */
- ak = Re [kk]; Re [kk] = Re [k2]; Re [k2] = ak;
- bk = Im [kk]; Im [kk] = Im [k2]; Im [k2] = bk;
- kk += inc;
- k2 += kspan;
- } while (k2 < (ns-1));
- do {
- do {
- k2 -= fftstate->Perm [j - 1];
- j++;
- k2 = fftstate->Perm [j] + k2;
- } while (k2 >= fftstate->Perm [j - 1]);
- j = 1;
- do {
- if (kk < k2)
- goto Permute_Single_Label;
- kk += inc;
- k2 += kspan;
- } while (k2 < (ns-1));
- } while (kk < (ns-1));
- }
- jc = k3;
- }
-
- if ((kt << 1) + 1 >= mfactor)
- return 0;
- ispan = fftstate->Perm [kt];
- /* permutation for square-free factors of n */
- j = mfactor - kt;
- fftstate->factor [j] = 1;
- do {
- fftstate->factor [j - 1] *= fftstate->factor [j];
- j--;
- } while (j != kt);
- kt++;
- nn = fftstate->factor [kt - 1] - 1;
- if (nn > (int) max_perm) {
- return -1;
- }
- j = jj = 0;
- for (;;) {
- k = kt + 1;
- k2 = fftstate->factor [kt - 1];
- kk = fftstate->factor [k - 1];
- j++;
- if (j > nn)
- break; /* exit infinite loop */
- jj += kk;
- while (jj >= k2) {
- jj -= k2;
- k2 = kk;
- k++;
- kk = fftstate->factor [k - 1];
- jj += kk;
- }
- fftstate->Perm [j - 1] = jj;
- }
- /* determine the permutation cycles of length greater than 1 */
- j = 0;
- for (;;) {
- do {
- j++;
- kk = fftstate->Perm [j - 1];
- } while (kk < 0);
- if (kk != j) {
- do {
- k = kk;
- kk = fftstate->Perm [k - 1];
- fftstate->Perm [k - 1] = -kk;
- } while (kk != j);
- k3 = kk;
- } else {
- fftstate->Perm [j - 1] = -j;
- if (j == nn)
- break; /* exit infinite loop */
- }
- }
- max_factors *= inc;
- /* reorder a and b, following the permutation cycles */
- for (;;) {
- j = k3 + 1;
- nt -= ispan;
- ii = nt - inc + 1;
- if (nt < 0)
- break; /* exit infinite loop */
- do {
- do {
- j--;
- } while (fftstate->Perm [j - 1] < 0);
- jj = jc;
- do {
- kspan = jj;
- if (jj > max_factors) {
- kspan = max_factors;
- }
- jj -= kspan;
- k = fftstate->Perm [j - 1];
- kk = jc * k + ii + jj;
- k1 = kk + kspan - 1;
- k2 = 0;
- do {
- k2++;
- Rtmp [k2 - 1] = Re [k1];
- Itmp [k2 - 1] = Im [k1];
- k1 -= inc;
- } while (k1 != (kk-1));
- do {
- k1 = kk + kspan - 1;
- k2 = k1 - jc * (k + fftstate->Perm [k - 1]);
- k = -fftstate->Perm [k - 1];
- do {
- Re [k1] = Re [k2];
- Im [k1] = Im [k2];
- k1 -= inc;
- k2 -= inc;
- } while (k1 != (kk-1));
- kk = k2 + 1;
- } while (k != j);
- k1 = kk + kspan - 1;
- k2 = 0;
- do {
- k2++;
- Re [k1] = Rtmp [k2 - 1];
- Im [k1] = Itmp [k2 - 1];
- k1 -= inc;
- } while (k1 != (kk-1));
- } while (jj);
- } while (j != 1);
- }
- return 0; /* exit point here */
-}
-/* ---------------------- end-of-file (c source) ---------------------- */
-
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/fft.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/fft.h
deleted file mode 100644
index a42f57b..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/fft.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*--------------------------------*-C-*---------------------------------*
- * File:
- * fftn.h
- * ---------------------------------------------------------------------*
- * Re[]: real value array
- * Im[]: imaginary value array
- * nTotal: total number of complex values
- * nPass: number of elements involved in this pass of transform
- * nSpan: nspan/nPass = number of bytes to increment pointer
- * in Re[] and Im[]
- * isign: exponent: +1 = forward -1 = reverse
- * scaling: normalizing constant by which the final result is *divided*
- * scaling == -1, normalize by total dimension of the transform
- * scaling < -1, normalize by the square-root of the total dimension
- *
- * ----------------------------------------------------------------------
- * See the comments in the code for correct usage!
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FFT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FFT_H_
-
-
-#include "structs.h"
-
-
-/* double precision routine */
-
-
-int WebRtcIsac_Fftns (unsigned int ndim, const int dims[], double Re[], double Im[],
- int isign, double scaling, FFTstr *fftstate);
-
-
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FFT_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filter_functions.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filter_functions.c
deleted file mode 100644
index 33024a3..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filter_functions.c
+++ /dev/null
@@ -1,271 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <memory.h>
-#ifdef WEBRTC_ANDROID
-#include <stdlib.h>
-#endif
-#include "pitch_estimator.h"
-#include "lpc_analysis.h"
-#include "codec.h"
-
-
-
-void WebRtcIsac_AllPoleFilter(double *InOut, double *Coef, int lengthInOut, int orderCoef){
-
- /* the state of filter is assumed to be in InOut[-1] to InOut[-orderCoef] */
- double scal;
- double sum;
- int n,k;
-
- //if (fabs(Coef[0]-1.0)<0.001) {
- if ( (Coef[0] > 0.9999) && (Coef[0] < 1.0001) )
- {
- for(n = 0; n < lengthInOut; n++)
- {
- sum = Coef[1] * InOut[-1];
- for(k = 2; k <= orderCoef; k++){
- sum += Coef[k] * InOut[-k];
- }
- *InOut++ -= sum;
- }
- }
- else
- {
- scal = 1.0 / Coef[0];
- for(n=0;n<lengthInOut;n++)
- {
- *InOut *= scal;
- for(k=1;k<=orderCoef;k++){
- *InOut -= scal*Coef[k]*InOut[-k];
- }
- InOut++;
- }
- }
-}
-
-
-void WebRtcIsac_AllZeroFilter(double *In, double *Coef, int lengthInOut, int orderCoef, double *Out){
-
- /* the state of filter is assumed to be in In[-1] to In[-orderCoef] */
-
- int n, k;
- double tmp;
-
- for(n = 0; n < lengthInOut; n++)
- {
- tmp = In[0] * Coef[0];
-
- for(k = 1; k <= orderCoef; k++){
- tmp += Coef[k] * In[-k];
- }
-
- *Out++ = tmp;
- In++;
- }
-}
-
-
-
-void WebRtcIsac_ZeroPoleFilter(double *In, double *ZeroCoef, double *PoleCoef, int lengthInOut, int orderCoef, double *Out){
-
- /* the state of the zero section is assumed to be in In[-1] to In[-orderCoef] */
- /* the state of the pole section is assumed to be in Out[-1] to Out[-orderCoef] */
-
- WebRtcIsac_AllZeroFilter(In,ZeroCoef,lengthInOut,orderCoef,Out);
- WebRtcIsac_AllPoleFilter(Out,PoleCoef,lengthInOut,orderCoef);
-}
-
-
-void WebRtcIsac_AutoCorr(
- double *r,
- const double *x,
- int N,
- int order
- )
-{
- int lag, n;
- double sum, prod;
- const double *x_lag;
-
- for (lag = 0; lag <= order; lag++)
- {
- sum = 0.0f;
- x_lag = &x[lag];
- prod = x[0] * x_lag[0];
- for (n = 1; n < N - lag; n++) {
- sum += prod;
- prod = x[n] * x_lag[n];
- }
- sum += prod;
- r[lag] = sum;
- }
-
-}
-
-
-void WebRtcIsac_BwExpand(double *out, double *in, double coef, short length) {
- int i;
- double chirp;
-
- chirp = coef;
-
- out[0] = in[0];
- for (i = 1; i < length; i++) {
- out[i] = chirp * in[i];
- chirp *= coef;
- }
-}
-
-void WebRtcIsac_WeightingFilter(const double *in, double *weiout, double *whiout, WeightFiltstr *wfdata) {
-
- double tmpbuffer[PITCH_FRAME_LEN + PITCH_WLPCBUFLEN];
- double corr[PITCH_WLPCORDER+1], rc[PITCH_WLPCORDER+1];
- double apol[PITCH_WLPCORDER+1], apolr[PITCH_WLPCORDER+1];
- double rho=0.9, *inp, *dp, *dp2;
- double whoutbuf[PITCH_WLPCBUFLEN + PITCH_WLPCORDER];
- double weoutbuf[PITCH_WLPCBUFLEN + PITCH_WLPCORDER];
- double *weo, *who, opol[PITCH_WLPCORDER+1], ext[PITCH_WLPCWINLEN];
- int k, n, endpos, start;
-
- /* Set up buffer and states */
- memcpy(tmpbuffer, wfdata->buffer, sizeof(double) * PITCH_WLPCBUFLEN);
- memcpy(tmpbuffer+PITCH_WLPCBUFLEN, in, sizeof(double) * PITCH_FRAME_LEN);
- memcpy(wfdata->buffer, tmpbuffer+PITCH_FRAME_LEN, sizeof(double) * PITCH_WLPCBUFLEN);
-
- dp=weoutbuf;
- dp2=whoutbuf;
- for (k=0;k<PITCH_WLPCORDER;k++) {
- *dp++ = wfdata->weostate[k];
- *dp2++ = wfdata->whostate[k];
- opol[k]=0.0;
- }
- opol[0]=1.0;
- opol[PITCH_WLPCORDER]=0.0;
- weo=dp;
- who=dp2;
-
- endpos=PITCH_WLPCBUFLEN + PITCH_SUBFRAME_LEN;
- inp=tmpbuffer + PITCH_WLPCBUFLEN;
-
- for (n=0; n<PITCH_SUBFRAMES; n++) {
- /* Windowing */
- start=endpos-PITCH_WLPCWINLEN;
- for (k=0; k<PITCH_WLPCWINLEN; k++) {
- ext[k]=wfdata->window[k]*tmpbuffer[start+k];
- }
-
- /* Get LPC polynomial */
- WebRtcIsac_AutoCorr(corr, ext, PITCH_WLPCWINLEN, PITCH_WLPCORDER);
- corr[0]=1.01*corr[0]+1.0; /* White noise correction */
- WebRtcIsac_LevDurb(apol, rc, corr, PITCH_WLPCORDER);
- WebRtcIsac_BwExpand(apolr, apol, rho, PITCH_WLPCORDER+1);
-
- /* Filtering */
- WebRtcIsac_ZeroPoleFilter(inp, apol, apolr, PITCH_SUBFRAME_LEN, PITCH_WLPCORDER, weo);
- WebRtcIsac_ZeroPoleFilter(inp, apolr, opol, PITCH_SUBFRAME_LEN, PITCH_WLPCORDER, who);
-
- inp+=PITCH_SUBFRAME_LEN;
- endpos+=PITCH_SUBFRAME_LEN;
- weo+=PITCH_SUBFRAME_LEN;
- who+=PITCH_SUBFRAME_LEN;
- }
-
- /* Export filter states */
- for (k=0;k<PITCH_WLPCORDER;k++) {
- wfdata->weostate[k]=weoutbuf[PITCH_FRAME_LEN+k];
- wfdata->whostate[k]=whoutbuf[PITCH_FRAME_LEN+k];
- }
-
- /* Export output data */
- memcpy(weiout, weoutbuf+PITCH_WLPCORDER, sizeof(double) * PITCH_FRAME_LEN);
- memcpy(whiout, whoutbuf+PITCH_WLPCORDER, sizeof(double) * PITCH_FRAME_LEN);
-}
-
-
-static const double APupper[ALLPASSSECTIONS] = {0.0347, 0.3826};
-static const double APlower[ALLPASSSECTIONS] = {0.1544, 0.744};
-
-
-
-void WebRtcIsac_AllpassFilterForDec(double *InOut,
- const double *APSectionFactors,
- int lengthInOut,
- double *FilterState)
-{
- //This performs all-pass filtering--a series of first order all-pass sections are used
- //to filter the input in a cascade manner.
- int n,j;
- double temp;
- for (j=0; j<ALLPASSSECTIONS; j++){
- for (n=0;n<lengthInOut;n+=2){
- temp = InOut[n]; //store input
- InOut[n] = FilterState[j] + APSectionFactors[j]*temp;
- FilterState[j] = -APSectionFactors[j]*InOut[n] + temp;
- }
- }
-}
-
-void WebRtcIsac_DecimateAllpass(const double *in,
- double *state_in, /* array of size: 2*ALLPASSSECTIONS+1 */
- int N, /* number of input samples */
- double *out) /* array of size N/2 */
-{
- int n;
- double data_vec[PITCH_FRAME_LEN];
-
- /* copy input */
- memcpy(data_vec+1, in, sizeof(double) * (N-1));
-
- data_vec[0] = state_in[2*ALLPASSSECTIONS]; //the z^(-1) state
- state_in[2*ALLPASSSECTIONS] = in[N-1];
-
- WebRtcIsac_AllpassFilterForDec(data_vec+1, APupper, N, state_in);
- WebRtcIsac_AllpassFilterForDec(data_vec, APlower, N, state_in+ALLPASSSECTIONS);
-
- for (n=0;n<N/2;n++)
- out[n] = data_vec[2*n] + data_vec[2*n+1];
-
-}
-
-
-
-/* create high-pass filter ocefficients
- * z = 0.998 * exp(j*2*pi*35/8000);
- * p = 0.94 * exp(j*2*pi*140/8000);
- * HP_b = [1, -2*real(z), abs(z)^2];
- * HP_a = [1, -2*real(p), abs(p)^2]; */
-static const double a_coef[2] = { 1.86864659625574, -0.88360000000000};
-static const double b_coef[2] = {-1.99524591718270, 0.99600400000000};
-static const float a_coef_float[2] = { 1.86864659625574f, -0.88360000000000f};
-static const float b_coef_float[2] = {-1.99524591718270f, 0.99600400000000f};
-
-/* second order high-pass filter */
-void WebRtcIsac_Highpass(const double *in, double *out, double *state, int N)
-{
- int k;
-
- for (k=0; k<N; k++) {
- *out = *in + state[1];
- state[1] = state[0] + b_coef[0] * *in + a_coef[0] * *out;
- state[0] = b_coef[1] * *in++ + a_coef[1] * *out++;
- }
-}
-
-void WebRtcIsac_Highpass_float(const float *in, double *out, double *state, int N)
-{
- int k;
-
- for (k=0; k<N; k++) {
- *out = (double)*in + state[1];
- state[1] = state[0] + b_coef_float[0] * *in + a_coef_float[0] * *out;
- state[0] = b_coef_float[1] * (double)*in++ + a_coef_float[1] * *out++;
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbank_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbank_tables.c
deleted file mode 100644
index 0f844af..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbank_tables.c
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/* filterbank_tables.c*/
-/* This file contains variables that are used in filterbanks.c*/
-
-#include "filterbank_tables.h"
-#include "settings.h"
-
-/* The composite all-pass filter factors */
-const float WebRtcIsac_kCompositeApFactorsFloat[4] = {
- 0.03470000000000f, 0.15440000000000f, 0.38260000000000f, 0.74400000000000f};
-
-/* The upper channel all-pass filter factors */
-const float WebRtcIsac_kUpperApFactorsFloat[2] = {
- 0.03470000000000f, 0.38260000000000f};
-
-/* The lower channel all-pass filter factors */
-const float WebRtcIsac_kLowerApFactorsFloat[2] = {
- 0.15440000000000f, 0.74400000000000f};
-
-/* The matrix for transforming the backward composite state to upper channel state */
-const float WebRtcIsac_kTransform1Float[8] = {
- -0.00158678506084f, 0.00127157815343f, -0.00104805672709f, 0.00084837248079f,
- 0.00134467983258f, -0.00107756549387f, 0.00088814793277f, -0.00071893072525f};
-
-/* The matrix for transforming the backward composite state to lower channel state */
-const float WebRtcIsac_kTransform2Float[8] = {
- -0.00170686041697f, 0.00136780109829f, -0.00112736532350f, 0.00091257055385f,
- 0.00103094281812f, -0.00082615076557f, 0.00068092756088f, -0.00055119165484f};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbank_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbank_tables.h
deleted file mode 100644
index e8fda5e..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbank_tables.h
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * filterbank_tables.h
- *
- * Header file for variables that are defined in
- * filterbank_tables.c.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTERBANK_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTERBANK_TABLES_H_
-
-#include "structs.h"
-
-/********************* Coefficient Tables ************************/
-/* The number of composite all-pass filter factors */
-#define NUMBEROFCOMPOSITEAPSECTIONS 4
-
-/* The number of all-pass filter factors in an upper or lower channel*/
-#define NUMBEROFCHANNELAPSECTIONS 2
-
-/* The composite all-pass filter factors */
-extern const float WebRtcIsac_kCompositeApFactorsFloat[4];
-
-/* The upper channel all-pass filter factors */
-extern const float WebRtcIsac_kUpperApFactorsFloat[2];
-
-/* The lower channel all-pass filter factors */
-extern const float WebRtcIsac_kLowerApFactorsFloat[2];
-
-/* The matrix for transforming the backward composite state to upper channel state */
-extern const float WebRtcIsac_kTransform1Float[8];
-
-/* The matrix for transforming the backward composite state to lower channel state */
-extern const float WebRtcIsac_kTransform2Float[8];
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_FILTERBANK_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbanks.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbanks.c
deleted file mode 100644
index 671fd32..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/filterbanks.c
+++ /dev/null
@@ -1,346 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * filterbanks.c
- *
- * This file contains function WebRtcIsac_AllPassFilter2Float,
- * WebRtcIsac_SplitAndFilter, and WebRtcIsac_FilterAndCombine
- * which implement filterbanks that produce decimated lowpass and
- * highpass versions of a signal, and performs reconstruction.
- *
- */
-
-#include "settings.h"
-#include "filterbank_tables.h"
-#include "codec.h"
-
-/* This function performs all-pass filtering--a series of first order all-pass
- * sections are used to filter the input in a cascade manner.
- * The input is overwritten!!
- */
-static void WebRtcIsac_AllPassFilter2Float(float *InOut, const float *APSectionFactors,
- int lengthInOut, int NumberOfSections,
- float *FilterState)
-{
- int n, j;
- float temp;
- for (j=0; j<NumberOfSections; j++){
- for (n=0;n<lengthInOut;n++){
- temp = FilterState[j] + APSectionFactors[j] * InOut[n];
- FilterState[j] = -APSectionFactors[j] * temp + InOut[n];
- InOut[n] = temp;
- }
- }
-}
-
-/* HPstcoeff_in = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */
-static const float kHpStCoefInFloat[4] =
-{-1.94895953203325f, 0.94984516000000f, -0.05101826139794f, 0.05015484000000f};
-
-/* Function WebRtcIsac_SplitAndFilter
- * This function creates low-pass and high-pass decimated versions of part of
- the input signal, and part of the signal in the input 'lookahead buffer'.
-
- INPUTS:
- in: a length FRAMESAMPLES array of input samples
- prefiltdata: input data structure containing the filterbank states
- and lookahead samples from the previous encoding
- iteration.
- OUTPUTS:
- LP: a FRAMESAMPLES_HALF array of low-pass filtered samples that
- have been phase equalized. The first QLOOKAHEAD samples are
- based on the samples in the two prefiltdata->INLABUFx arrays
- each of length QLOOKAHEAD.
- The remaining FRAMESAMPLES_HALF-QLOOKAHEAD samples are based
- on the first FRAMESAMPLES_HALF-QLOOKAHEAD samples of the input
- array in[].
- HP: a FRAMESAMPLES_HALF array of high-pass filtered samples that
- have been phase equalized. The first QLOOKAHEAD samples are
- based on the samples in the two prefiltdata->INLABUFx arrays
- each of length QLOOKAHEAD.
- The remaining FRAMESAMPLES_HALF-QLOOKAHEAD samples are based
- on the first FRAMESAMPLES_HALF-QLOOKAHEAD samples of the input
- array in[].
-
- LP_la: a FRAMESAMPLES_HALF array of low-pass filtered samples.
- These samples are not phase equalized. They are computed
- from the samples in the in[] array.
- HP_la: a FRAMESAMPLES_HALF array of high-pass filtered samples
- that are not phase equalized. They are computed from
- the in[] vector.
- prefiltdata: this input data structure's filterbank state and
- lookahead sample buffers are updated for the next
- encoding iteration.
-*/
-void WebRtcIsac_SplitAndFilterFloat(float *pin, float *LP, float *HP,
- double *LP_la, double *HP_la,
- PreFiltBankstr *prefiltdata)
-{
- int k,n;
- float CompositeAPFilterState[NUMBEROFCOMPOSITEAPSECTIONS];
- float ForTransform_CompositeAPFilterState[NUMBEROFCOMPOSITEAPSECTIONS];
- float ForTransform_CompositeAPFilterState2[NUMBEROFCOMPOSITEAPSECTIONS];
- float tempinoutvec[FRAMESAMPLES+MAX_AR_MODEL_ORDER];
- float tempin_ch1[FRAMESAMPLES+MAX_AR_MODEL_ORDER];
- float tempin_ch2[FRAMESAMPLES+MAX_AR_MODEL_ORDER];
- float in[FRAMESAMPLES];
- float ftmp;
-
-
- /* High pass filter */
-
- for (k=0;k<FRAMESAMPLES;k++) {
- in[k] = pin[k] + kHpStCoefInFloat[2] * prefiltdata->HPstates_float[0] +
- kHpStCoefInFloat[3] * prefiltdata->HPstates_float[1];
- ftmp = pin[k] - kHpStCoefInFloat[0] * prefiltdata->HPstates_float[0] -
- kHpStCoefInFloat[1] * prefiltdata->HPstates_float[1];
- prefiltdata->HPstates_float[1] = prefiltdata->HPstates_float[0];
- prefiltdata->HPstates_float[0] = ftmp;
- }
-
- /*
- % backwards all-pass filtering to obtain zero-phase
- [tmp1(N2+LA:-1:LA+1, 1), state1] = filter(Q.coef, Q.coef(end:-1:1), in(N:-2:2));
- tmp1(LA:-1:1) = filter(Q.coef, Q.coef(end:-1:1), Q.LookAheadBuf1, state1);
- Q.LookAheadBuf1 = in(N:-2:N-2*LA+2);
- */
- /*Backwards all-pass filter the odd samples of the input (upper channel)
- to eventually obtain zero phase. The composite all-pass filter (comprised of both
- the upper and lower channel all-pass filsters in series) is used for the
- filtering. */
-
- /* First Channel */
-
- /*initial state of composite filter is zero */
- for (k=0;k<NUMBEROFCOMPOSITEAPSECTIONS;k++){
- CompositeAPFilterState[k] = 0.0;
- }
- /* put every other sample of input into a temporary vector in reverse (backward) order*/
- for (k=0;k<FRAMESAMPLES_HALF;k++) {
- tempinoutvec[k] = in[FRAMESAMPLES-1-2*k];
- }
-
- /* now all-pass filter the backwards vector. Output values overwrite the input vector. */
- WebRtcIsac_AllPassFilter2Float(tempinoutvec, WebRtcIsac_kCompositeApFactorsFloat,
- FRAMESAMPLES_HALF, NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
-
- /* save the backwards filtered output for later forward filtering,
- but write it in forward order*/
- for (k=0;k<FRAMESAMPLES_HALF;k++) {
- tempin_ch1[FRAMESAMPLES_HALF+QLOOKAHEAD-1-k] = tempinoutvec[k];
- }
-
- /* save the backwards filter state becaue it will be transformed
- later into a forward state */
- for (k=0; k<NUMBEROFCOMPOSITEAPSECTIONS; k++) {
- ForTransform_CompositeAPFilterState[k] = CompositeAPFilterState[k];
- }
-
- /* now backwards filter the samples in the lookahead buffer. The samples were
- placed there in the encoding of the previous frame. The output samples
- overwrite the input samples */
- WebRtcIsac_AllPassFilter2Float(prefiltdata->INLABUF1_float,
- WebRtcIsac_kCompositeApFactorsFloat, QLOOKAHEAD,
- NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
-
- /* save the output, but write it in forward order */
- /* write the lookahead samples for the next encoding iteration. Every other
- sample at the end of the input frame is written in reverse order for the
- lookahead length. Exported in the prefiltdata structure. */
- for (k=0;k<QLOOKAHEAD;k++) {
- tempin_ch1[QLOOKAHEAD-1-k]=prefiltdata->INLABUF1_float[k];
- prefiltdata->INLABUF1_float[k]=in[FRAMESAMPLES-1-2*k];
- }
-
- /* Second Channel. This is exactly like the first channel, except that the
- even samples are now filtered instead (lower channel). */
- for (k=0;k<NUMBEROFCOMPOSITEAPSECTIONS;k++){
- CompositeAPFilterState[k] = 0.0;
- }
-
- for (k=0;k<FRAMESAMPLES_HALF;k++) {
- tempinoutvec[k] = in[FRAMESAMPLES-2-2*k];
- }
-
- WebRtcIsac_AllPassFilter2Float(tempinoutvec, WebRtcIsac_kCompositeApFactorsFloat,
- FRAMESAMPLES_HALF, NUMBEROFCOMPOSITEAPSECTIONS, CompositeAPFilterState);
-
- for (k=0;k<FRAMESAMPLES_HALF;k++) {
- tempin_ch2[FRAMESAMPLES_HALF+QLOOKAHEAD-1-k] = tempinoutvec[k];
- }
-
- for (k=0; k<NUMBEROFCOMPOSITEAPSECTIONS; k++) {
- ForTransform_CompositeAPFilterState2[k] = CompositeAPFilterState[k];
- }
-
-
- WebRtcIsac_AllPassFilter2Float(prefiltdata->INLABUF2_float,
- WebRtcIsac_kCompositeApFactorsFloat, QLOOKAHEAD,NUMBEROFCOMPOSITEAPSECTIONS,
- CompositeAPFilterState);
-
- for (k=0;k<QLOOKAHEAD;k++) {
- tempin_ch2[QLOOKAHEAD-1-k]=prefiltdata->INLABUF2_float[k];
- prefiltdata->INLABUF2_float[k]=in[FRAMESAMPLES-2-2*k];
- }
-
- /* Transform filter states from backward to forward */
- /*At this point, each of the states of the backwards composite filters for the
- two channels are transformed into forward filtering states for the corresponding
- forward channel filters. Each channel's forward filtering state from the previous
- encoding iteration is added to the transformed state to get a proper forward state */
-
- /* So the existing NUMBEROFCOMPOSITEAPSECTIONS x 1 (4x1) state vector is multiplied by a
- NUMBEROFCHANNELAPSECTIONSxNUMBEROFCOMPOSITEAPSECTIONS (2x4) transform matrix to get the
- new state that is added to the previous 2x1 input state */
-
- for (k=0;k<NUMBEROFCHANNELAPSECTIONS;k++){ /* k is row variable */
- for (n=0; n<NUMBEROFCOMPOSITEAPSECTIONS;n++){/* n is column variable */
- prefiltdata->INSTAT1_float[k] += ForTransform_CompositeAPFilterState[n]*
- WebRtcIsac_kTransform1Float[k*NUMBEROFCHANNELAPSECTIONS+n];
- prefiltdata->INSTAT2_float[k] += ForTransform_CompositeAPFilterState2[n]*
- WebRtcIsac_kTransform2Float[k*NUMBEROFCHANNELAPSECTIONS+n];
- }
- }
-
- /*obtain polyphase components by forward all-pass filtering through each channel */
- /* the backward filtered samples are now forward filtered with the corresponding channel filters */
- /* The all pass filtering automatically updates the filter states which are exported in the
- prefiltdata structure */
- WebRtcIsac_AllPassFilter2Float(tempin_ch1,WebRtcIsac_kUpperApFactorsFloat,
- FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTAT1_float);
- WebRtcIsac_AllPassFilter2Float(tempin_ch2,WebRtcIsac_kLowerApFactorsFloat,
- FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTAT2_float);
-
- /* Now Construct low-pass and high-pass signals as combinations of polyphase components */
- for (k=0; k<FRAMESAMPLES_HALF; k++) {
- LP[k] = 0.5f*(tempin_ch1[k] + tempin_ch2[k]);/* low pass signal*/
- HP[k] = 0.5f*(tempin_ch1[k] - tempin_ch2[k]);/* high pass signal*/
- }
-
- /* Lookahead LP and HP signals */
- /* now create low pass and high pass signals of the input vector. However, no
- backwards filtering is performed, and hence no phase equalization is involved.
- Also, the input contains some samples that are lookahead samples. The high pass
- and low pass signals that are created are used outside this function for analysis
- (not encoding) purposes */
-
- /* set up input */
- for (k=0; k<FRAMESAMPLES_HALF; k++) {
- tempin_ch1[k]=in[2*k+1];
- tempin_ch2[k]=in[2*k];
- }
-
- /* the input filter states are passed in and updated by the all-pass filtering routine and
- exported in the prefiltdata structure*/
- WebRtcIsac_AllPassFilter2Float(tempin_ch1,WebRtcIsac_kUpperApFactorsFloat,
- FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTATLA1_float);
- WebRtcIsac_AllPassFilter2Float(tempin_ch2,WebRtcIsac_kLowerApFactorsFloat,
- FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS, prefiltdata->INSTATLA2_float);
-
- for (k=0; k<FRAMESAMPLES_HALF; k++) {
- LP_la[k] = (float)(0.5f*(tempin_ch1[k] + tempin_ch2[k])); /*low pass */
- HP_la[k] = (double)(0.5f*(tempin_ch1[k] - tempin_ch2[k])); /* high pass */
- }
-
-
-}/*end of WebRtcIsac_SplitAndFilter */
-
-
-/* Combining */
-
-/* HPstcoeff_out_1 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */
-static const float kHpStCoefOut1Float[4] =
-{-1.99701049409000f, 0.99714204490000f, 0.01701049409000f, -0.01704204490000f};
-
-/* HPstcoeff_out_2 = {a1, a2, b1 - b0 * a1, b2 - b0 * a2}; */
-static const float kHpStCoefOut2Float[4] =
-{-1.98645294509837f, 0.98672435560000f, 0.00645294509837f, -0.00662435560000f};
-
-
-/* Function WebRtcIsac_FilterAndCombine */
-/* This is a decoder function that takes the decimated
- length FRAMESAMPLES_HALF input low-pass and
- high-pass signals and creates a reconstructed fullband
- output signal of length FRAMESAMPLES. WebRtcIsac_FilterAndCombine
- is the sibling function of WebRtcIsac_SplitAndFilter */
-/* INPUTS:
- inLP: a length FRAMESAMPLES_HALF array of input low-pass
- samples.
- inHP: a length FRAMESAMPLES_HALF array of input high-pass
- samples.
- postfiltdata: input data structure containing the filterbank
- states from the previous decoding iteration.
- OUTPUTS:
- Out: a length FRAMESAMPLES array of output reconstructed
- samples (fullband) based on the input low-pass and
- high-pass signals.
- postfiltdata: the input data structure containing the filterbank
- states is updated for the next decoding iteration */
-void WebRtcIsac_FilterAndCombineFloat(float *InLP,
- float *InHP,
- float *Out,
- PostFiltBankstr *postfiltdata)
-{
- int k;
- float tempin_ch1[FRAMESAMPLES+MAX_AR_MODEL_ORDER];
- float tempin_ch2[FRAMESAMPLES+MAX_AR_MODEL_ORDER];
- float ftmp, ftmp2;
-
- /* Form the polyphase signals*/
- for (k=0;k<FRAMESAMPLES_HALF;k++) {
- tempin_ch1[k]=InLP[k]+InHP[k]; /* Construct a new upper channel signal*/
- tempin_ch2[k]=InLP[k]-InHP[k]; /* Construct a new lower channel signal*/
- }
-
-
- /* all-pass filter the new upper channel signal. HOWEVER, use the all-pass filter factors
- that were used as a lower channel at the encoding side. So at the decoder, the
- corresponding all-pass filter factors for each channel are swapped.*/
- WebRtcIsac_AllPassFilter2Float(tempin_ch1, WebRtcIsac_kLowerApFactorsFloat,
- FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,postfiltdata->STATE_0_UPPER_float);
-
- /* Now, all-pass filter the new lower channel signal. But since all-pass filter factors
- at the decoder are swapped from the ones at the encoder, the 'upper' channel
- all-pass filter factors (WebRtcIsac_kUpperApFactorsFloat) are used to filter this new
- lower channel signal */
- WebRtcIsac_AllPassFilter2Float(tempin_ch2, WebRtcIsac_kUpperApFactorsFloat,
- FRAMESAMPLES_HALF, NUMBEROFCHANNELAPSECTIONS,postfiltdata->STATE_0_LOWER_float);
-
-
- /* Merge outputs to form the full length output signal.*/
- for (k=0;k<FRAMESAMPLES_HALF;k++) {
- Out[2*k]=tempin_ch2[k];
- Out[2*k+1]=tempin_ch1[k];
- }
-
-
- /* High pass filter */
-
- for (k=0;k<FRAMESAMPLES;k++) {
- ftmp2 = Out[k] + kHpStCoefOut1Float[2] * postfiltdata->HPstates1_float[0] +
- kHpStCoefOut1Float[3] * postfiltdata->HPstates1_float[1];
- ftmp = Out[k] - kHpStCoefOut1Float[0] * postfiltdata->HPstates1_float[0] -
- kHpStCoefOut1Float[1] * postfiltdata->HPstates1_float[1];
- postfiltdata->HPstates1_float[1] = postfiltdata->HPstates1_float[0];
- postfiltdata->HPstates1_float[0] = ftmp;
- Out[k] = ftmp2;
- }
-
- for (k=0;k<FRAMESAMPLES;k++) {
- ftmp2 = Out[k] + kHpStCoefOut2Float[2] * postfiltdata->HPstates2_float[0] +
- kHpStCoefOut2Float[3] * postfiltdata->HPstates2_float[1];
- ftmp = Out[k] - kHpStCoefOut2Float[0] * postfiltdata->HPstates2_float[0] -
- kHpStCoefOut2Float[1] * postfiltdata->HPstates2_float[1];
- postfiltdata->HPstates2_float[1] = postfiltdata->HPstates2_float[0];
- postfiltdata->HPstates2_float[0] = ftmp;
- Out[k] = ftmp2;
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/intialize.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/intialize.c
deleted file mode 100644
index 6df034d..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/intialize.c
+++ /dev/null
@@ -1,175 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/* encode.c - Encoding function for the iSAC coder */
-
-#include "structs.h"
-#include "codec.h"
-#include "pitch_estimator.h"
-
-#include <math.h>
-
-void WebRtcIsac_InitMasking(MaskFiltstr *maskdata) {
-
- int k;
-
- for (k = 0; k < WINLEN; k++) {
- maskdata->DataBufferLo[k] = 0.0;
- maskdata->DataBufferHi[k] = 0.0;
- }
- for (k = 0; k < ORDERLO+1; k++) {
- maskdata->CorrBufLo[k] = 0.0;
- maskdata->PreStateLoF[k] = 0.0;
- maskdata->PreStateLoG[k] = 0.0;
- maskdata->PostStateLoF[k] = 0.0;
- maskdata->PostStateLoG[k] = 0.0;
- }
- for (k = 0; k < ORDERHI+1; k++) {
- maskdata->CorrBufHi[k] = 0.0;
- maskdata->PreStateHiF[k] = 0.0;
- maskdata->PreStateHiG[k] = 0.0;
- maskdata->PostStateHiF[k] = 0.0;
- maskdata->PostStateHiG[k] = 0.0;
- }
-
- maskdata->OldEnergy = 10.0;
-
- /* fill tables for transforms */
- WebRtcIsac_InitTransform();
-
- return;
-}
-
-void WebRtcIsac_InitPreFilterbank(PreFiltBankstr *prefiltdata)
-{
- int k;
-
- for (k = 0; k < QLOOKAHEAD; k++) {
- prefiltdata->INLABUF1[k] = 0;
- prefiltdata->INLABUF2[k] = 0;
-
- prefiltdata->INLABUF1_float[k] = 0;
- prefiltdata->INLABUF2_float[k] = 0;
- }
- for (k = 0; k < 2*(QORDER-1); k++) {
- prefiltdata->INSTAT1[k] = 0;
- prefiltdata->INSTAT2[k] = 0;
- prefiltdata->INSTATLA1[k] = 0;
- prefiltdata->INSTATLA2[k] = 0;
-
- prefiltdata->INSTAT1_float[k] = 0;
- prefiltdata->INSTAT2_float[k] = 0;
- prefiltdata->INSTATLA1_float[k] = 0;
- prefiltdata->INSTATLA2_float[k] = 0;
- }
-
- /* High pass filter states */
- prefiltdata->HPstates[0] = 0.0;
- prefiltdata->HPstates[1] = 0.0;
-
- prefiltdata->HPstates_float[0] = 0.0f;
- prefiltdata->HPstates_float[1] = 0.0f;
-
- return;
-}
-
-void WebRtcIsac_InitPostFilterbank(PostFiltBankstr *postfiltdata)
-{
- int k;
-
- for (k = 0; k < 2*POSTQORDER; k++) {
- postfiltdata->STATE_0_LOWER[k] = 0;
- postfiltdata->STATE_0_UPPER[k] = 0;
-
- postfiltdata->STATE_0_LOWER_float[k] = 0;
- postfiltdata->STATE_0_UPPER_float[k] = 0;
- }
-
- /* High pass filter states */
- postfiltdata->HPstates1[0] = 0.0;
- postfiltdata->HPstates1[1] = 0.0;
-
- postfiltdata->HPstates2[0] = 0.0;
- postfiltdata->HPstates2[1] = 0.0;
-
- postfiltdata->HPstates1_float[0] = 0.0f;
- postfiltdata->HPstates1_float[1] = 0.0f;
-
- postfiltdata->HPstates2_float[0] = 0.0f;
- postfiltdata->HPstates2_float[1] = 0.0f;
-
- return;
-}
-
-
-void WebRtcIsac_InitPitchFilter(PitchFiltstr *pitchfiltdata)
-{
- int k;
-
- for (k = 0; k < PITCH_BUFFSIZE; k++) {
- pitchfiltdata->ubuf[k] = 0.0;
- }
- pitchfiltdata->ystate[0] = 0.0;
- for (k = 1; k < (PITCH_DAMPORDER); k++) {
- pitchfiltdata->ystate[k] = 0.0;
- }
- pitchfiltdata->oldlagp[0] = 50.0;
- pitchfiltdata->oldgainp[0] = 0.0;
-}
-
-void WebRtcIsac_InitWeightingFilter(WeightFiltstr *wfdata)
-{
- int k;
- double t, dtmp, dtmp2, denum, denum2;
-
- for (k=0;k<PITCH_WLPCBUFLEN;k++)
- wfdata->buffer[k]=0.0;
-
- for (k=0;k<PITCH_WLPCORDER;k++) {
- wfdata->istate[k]=0.0;
- wfdata->weostate[k]=0.0;
- wfdata->whostate[k]=0.0;
- }
-
- /* next part should be in Matlab, writing to a global table */
- t = 0.5;
- denum = 1.0 / ((double) PITCH_WLPCWINLEN);
- denum2 = denum * denum;
- for (k=0;k<PITCH_WLPCWINLEN;k++) {
- dtmp = PITCH_WLPCASYM * t * denum + (1-PITCH_WLPCASYM) * t * t * denum2;
- dtmp *= 3.14159265;
- dtmp2 = sin(dtmp);
- wfdata->window[k] = dtmp2 * dtmp2;
- t++;
- }
-}
-
-/* clear all buffers */
-void WebRtcIsac_InitPitchAnalysis(PitchAnalysisStruct *State)
-{
- int k;
-
- for (k = 0; k < PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2; k++)
- State->dec_buffer[k] = 0.0;
- for (k = 0; k < 2*ALLPASSSECTIONS+1; k++)
- State->decimator_state[k] = 0.0;
- for (k = 0; k < 2; k++)
- State->hp_state[k] = 0.0;
- for (k = 0; k < QLOOKAHEAD; k++)
- State->whitened_buf[k] = 0.0;
- for (k = 0; k < QLOOKAHEAD; k++)
- State->inbuf[k] = 0.0;
-
- WebRtcIsac_InitPitchFilter(&(State->PFstr_wght));
-
- WebRtcIsac_InitPitchFilter(&(State->PFstr));
-
- WebRtcIsac_InitWeightingFilter(&(State->Wghtstr));
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/isac.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/isac.c
deleted file mode 100644
index 67db1e4..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/isac.c
+++ /dev/null
@@ -1,2793 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * isac.c
- *
- * This C file contains the functions for the ISAC API
- *
- */
-
-#include "isac.h"
-#include "bandwidth_estimator.h"
-#include "crc.h"
-#include "entropy_coding.h"
-#include "codec.h"
-#include "structs.h"
-#include "signal_processing_library.h"
-#include "lpc_shape_swb16_tables.h"
-#include "os_specific_inline.h"
-
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <math.h>
-
-#define BIT_MASK_DEC_INIT 0x0001
-#define BIT_MASK_ENC_INIT 0x0002
-
-#define LEN_CHECK_SUM_WORD8 4
-#define MAX_NUM_LAYERS 10
-
-
-/****************************************************************************
- * UpdatePayloadSizeLimit()
- *
- * Call this function to update the limit on the payload size. The limit on
- * payload size might change i) if a user ''directly changes the limit by
- * calling xxx_setMaxPayloadSize() or xxx_setMaxRate(), or ii) indirectly
- * when bandwidth is changing. The latter might be the result of bandwidth
- * adaptation, or direct change of the bottleneck in instantaneous mode.
- *
- * This function takes the current overall limit on payload, and translate it
- * to the limits on lower and upper-band. If the codec is in wideband mode
- * then the overall limit and the limit on the lower-band is the same.
- * Otherwise, a fraction of the limit should be allocated to lower-band
- * leaving some room for the upper-band bit-stream. That is why an update
- * of limit is required every time that the bandwidth is changing.
- *
- */
-static void UpdatePayloadSizeLimit(
- ISACMainStruct *instISAC)
-{
- WebRtc_Word16 lim30MsPayloadBytes;
- WebRtc_Word16 lim60MsPayloadBytes;
-
- lim30MsPayloadBytes = WEBRTC_SPL_MIN(
- (instISAC->maxPayloadSizeBytes),
- (instISAC->maxRateBytesPer30Ms));
-
- lim60MsPayloadBytes = WEBRTC_SPL_MIN(
- (instISAC->maxPayloadSizeBytes),
- (instISAC->maxRateBytesPer30Ms << 1));
-
- // The only time that iSAC will have 60 ms
- // frame-size is when operating in wideband so
- // there is no upper-band bit-stream
-
- if(instISAC->bandwidthKHz == isac8kHz)
- {
- // at 8 kHz there is no upper-band bit-stream
- // therefore the lower-band limit is as the overall
- // limit.
- instISAC->instLB.ISACencLB_obj.payloadLimitBytes60 =
- lim60MsPayloadBytes;
- instISAC->instLB.ISACencLB_obj.payloadLimitBytes30 =
- lim30MsPayloadBytes;
- }
- else
- {
- // when in super-wideband, we only have 30 ms frames
- // Do a rate allocation for the given limit.
- if(lim30MsPayloadBytes > 250)
- {
- // 4/5 to lower-band the rest for upper-band
- instISAC->instLB.ISACencLB_obj.payloadLimitBytes30 =
- (lim30MsPayloadBytes << 2) / 5;
- }
- else if(lim30MsPayloadBytes > 200)
- {
- // for the interval of 200 to 250 the share of
- // upper-band linearly grows from 20 to 50;
- instISAC->instLB.ISACencLB_obj.payloadLimitBytes30 =
- (lim30MsPayloadBytes << 1) / 5 + 100;
- }
- else
- {
- // allocate only 20 for upper-band
- instISAC->instLB.ISACencLB_obj.payloadLimitBytes30 =
- lim30MsPayloadBytes - 20;
- }
- instISAC->instUB.ISACencUB_obj.maxPayloadSizeBytes =
- lim30MsPayloadBytes;
- }
-}
-
-
-/****************************************************************************
- * UpdateBottleneck()
- *
- * This function updates the bottleneck only if the codec is operating in
- * channel-adaptive mode. Furthermore, as the update of bottleneck might
- * result in an update of bandwidth, therefore, the bottlenech should be
- * updated just right before the first 10ms of a frame is pushed into encoder.
- *
- */
-static void UpdateBottleneck(
- ISACMainStruct *instISAC)
-{
- // read the bottleneck from bandwidth estimator for the
- // first 10 ms audio. This way, if there is a change
- // in bandwidth upper and lower-band will be in sync.
- if((instISAC->codingMode == 0) &&
- (instISAC->instLB.ISACencLB_obj.buffer_index == 0) &&
- (instISAC->instLB.ISACencLB_obj.frame_nb == 0))
- {
- WebRtc_Word32 bottleneck;
- WebRtcIsac_GetUplinkBandwidth(&(instISAC->bwestimator_obj),
- &bottleneck);
-
- // Adding hysteresis when increasing signal bandwidth
- if((instISAC->bandwidthKHz == isac8kHz)
- && (bottleneck > 37000)
- && (bottleneck < 41000))
- {
- bottleneck = 37000;
- }
-
- // switching from 12 kHz to 16 kHz is not allowed at this revision
- // If we let this happen, we have to take care of buffer_index and
- // the last LPC vector.
- if((instISAC->bandwidthKHz != isac16kHz) &&
- (bottleneck > 46000))
- {
- bottleneck = 46000;
- }
-
- // we might need a rate allocation.
- if(instISAC->encoderSamplingRateKHz == kIsacWideband)
- {
- // wideband is the only choise we have here.
- instISAC->instLB.ISACencLB_obj.bottleneck =
- (bottleneck > 32000)? 32000:bottleneck;
- instISAC->bandwidthKHz = isac8kHz;
- }
- else
- {
- // do the rate-allosation and get the new bandwidth.
- enum ISACBandwidth bandwidth;
- WebRtcIsac_RateAllocation(bottleneck,
- &(instISAC->instLB.ISACencLB_obj.bottleneck),
- &(instISAC->instUB.ISACencUB_obj.bottleneck),
- &bandwidth);
- if(bandwidth != isac8kHz)
- {
- instISAC->instLB.ISACencLB_obj.new_framelength = 480;
- }
- if(bandwidth != instISAC->bandwidthKHz)
- {
- // bandwidth is changing.
- instISAC->bandwidthKHz = bandwidth;
- UpdatePayloadSizeLimit(instISAC);
- if(bandwidth == isac12kHz)
- {
- instISAC->instLB.ISACencLB_obj.buffer_index = 0;
- }
- // currently we don't let the bandwidth to switch to 16 kHz
- // if in adaptive mode. If we let this happen, we have to take
- // car of buffer_index and the last LPC vector.
- }
- }
- }
-}
-
-
-/****************************************************************************
- * GetSendBandwidthInfo()
- *
- * This is called to get the bandwidth info. This info is the bandwidth and
- * and the jitter of 'there-to-here' channel, estimated 'here.' These info
- * is signaled in an in-band fashion to the otherside.
- *
- * The call to the bandwidth estimator trigers a recursive averaging which
- * has to be synchronized between encoder & decoder, therefore. The call to
- * BWE should be once per packet. As the BWE info is inserted into bit-stream
- * we need a valid info right before the encodeLB function is going to
- * generating a bit-stream. That is when lower-band buffer has already 20ms
- * of audio, and the 3rd block of 10ms is going to be injected into encoder.
- *
- * Inputs:
- * - instISAC : iSAC instance.
- *
- * Outputs:
- * - bandwidthIndex : an index which has to be encoded in
- * lower-band bit-stream, indicating the
- * bandwidth of there-to-here channel.
- * - jitterInfo : this indicates if the jitter is high
- * or low and it is encoded in upper-band
- * bit-stream.
- *
- */
-static void GetSendBandwidthInfo(
- ISACMainStruct* instISAC,
- WebRtc_Word16* bandwidthIndex,
- WebRtc_Word16* jitterInfo)
-{
- if((instISAC->instLB.ISACencLB_obj.buffer_index ==
- (FRAMESAMPLES_10ms << 1)) &&
- (instISAC->instLB.ISACencLB_obj.frame_nb == 0))
- {
- /* bandwidth estimation and coding */
- WebRtcIsac_GetDownlinkBwJitIndexImpl(&(instISAC->bwestimator_obj),
- bandwidthIndex, jitterInfo, instISAC->decoderSamplingRateKHz);
- }
-}
-
-
-/****************************************************************************
- * WebRtcIsac_AssignSize(...)
- *
- * This function returns the size of the ISAC instance, so that the instance
- * can be created out side iSAC.
- *
- * Output:
- * - sizeinbytes : number of bytes needed to allocate for the
- * instance.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-WebRtc_Word16 WebRtcIsac_AssignSize(
- int *sizeInBytes)
-{
- *sizeInBytes = sizeof(ISACMainStruct) * 2 / sizeof(WebRtc_Word16);
- return 0;
-}
-
-
-/****************************************************************************
- * WebRtcIsac_Assign(...)
- *
- * This function assignes the memory already created to the ISAC instance.
- *
- * Input:
- * - ISAC_main_inst : address of the pointer to the coder instance.
- * - instISAC_Addr : the already allocaded memeory, where we put the
- * iSAC struct
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-WebRtc_Word16 WebRtcIsac_Assign(
- ISACStruct** ISAC_main_inst,
- void* instISAC_Addr)
-{
- if(instISAC_Addr != NULL)
- {
- ISACMainStruct* instISAC = (ISACMainStruct*)instISAC_Addr;
- instISAC->errorCode = 0;
- instISAC->initFlag = 0;
-
- // Assign the address
- *ISAC_main_inst = (ISACStruct*)instISAC_Addr;
-
- // Default is wideband.
- instISAC->encoderSamplingRateKHz = kIsacWideband;
- instISAC->decoderSamplingRateKHz = kIsacWideband;
- instISAC->bandwidthKHz = isac8kHz;
- return 0;
- }
- else
- {
- return -1;
- }
-}
-
-
-/****************************************************************************
- * WebRtcIsac_Create(...)
- *
- * This function creates an ISAC instance, which will contain the state
- * information for one coding/decoding channel.
- *
- * Input:
- * - ISAC_main_inst : address of the pointer to the coder instance.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-WebRtc_Word16 WebRtcIsac_Create(
- ISACStruct** ISAC_main_inst)
-{
- ISACMainStruct* instISAC;
-
- instISAC = (ISACMainStruct*)WEBRTC_SPL_VNEW(ISACMainStruct, 1);
- *ISAC_main_inst = (ISACStruct*)instISAC;
- if(*ISAC_main_inst != NULL)
- {
- instISAC->errorCode = 0;
- instISAC->initFlag = 0;
- // Default is wideband
- instISAC->bandwidthKHz = isac8kHz;
- instISAC->encoderSamplingRateKHz = kIsacWideband;
- instISAC->decoderSamplingRateKHz = kIsacWideband;
- return 0;
- }
- else
- {
- return -1;
- }
-}
-
-
-/****************************************************************************
- * WebRtcIsac_Free(...)
- *
- * This function frees the ISAC instance created at the beginning.
- *
- * Input:
- * - ISAC_main_inst : a ISAC instance.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-WebRtc_Word16 WebRtcIsac_Free(
- ISACStruct* ISAC_main_inst)
-{
- ISACMainStruct* instISAC;
-
- instISAC = (ISACMainStruct*)ISAC_main_inst;
- WEBRTC_SPL_FREE(instISAC);
- return 0;
-}
-
-
-/****************************************************************************
- * EncoderInitLb(...) - internal function for initialization of
- * Lower Band
- * EncoderInitUb(...) - internal function for initialization of
- * Upper Band
- * WebRtcIsac_EncoderInit(...) - API function
- *
- * This function initializes a ISAC instance prior to the encoder calls.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - CodingMode : 0 -> Bit rate and frame length are automatically
- * adjusted to available bandwidth on
- * transmission channel, applicable just to
- * wideband mode.
- * 1 -> User sets a frame length and a target bit
- * rate which is taken as the maximum
- * short-term average bit rate.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-static WebRtc_Word16 EncoderInitLb(
- ISACLBStruct* instLB,
- WebRtc_Word16 codingMode,
- enum IsacSamplingRate sampRate)
-{
- WebRtc_Word16 statusInit = 0;
- int k;
-
- /* Init stream vector to zero */
- for(k=0; k < STREAM_SIZE_MAX_60; k++)
- {
- instLB->ISACencLB_obj.bitstr_obj.stream[k] = 0;
- }
-
- if((codingMode == 1) || (sampRate == kIsacSuperWideband))
- {
- // 30 ms frame-size if either in super-wideband or
- // instanteneous mode (I-mode)
- instLB->ISACencLB_obj.new_framelength = 480;
- }
- else
- {
- instLB->ISACencLB_obj.new_framelength = INITIAL_FRAMESAMPLES;
- }
-
- WebRtcIsac_InitMasking(&instLB->ISACencLB_obj.maskfiltstr_obj);
- WebRtcIsac_InitPreFilterbank(&instLB->ISACencLB_obj.prefiltbankstr_obj);
- WebRtcIsac_InitPitchFilter(&instLB->ISACencLB_obj.pitchfiltstr_obj);
- WebRtcIsac_InitPitchAnalysis(
- &instLB->ISACencLB_obj.pitchanalysisstr_obj);
-
-
- instLB->ISACencLB_obj.buffer_index = 0;
- instLB->ISACencLB_obj.frame_nb = 0;
- /* default for I-mode */
- instLB->ISACencLB_obj.bottleneck = 32000;
- instLB->ISACencLB_obj.current_framesamples = 0;
- instLB->ISACencLB_obj.s2nr = 0;
- instLB->ISACencLB_obj.payloadLimitBytes30 = STREAM_SIZE_MAX_30;
- instLB->ISACencLB_obj.payloadLimitBytes60 = STREAM_SIZE_MAX_60;
- instLB->ISACencLB_obj.maxPayloadBytes = STREAM_SIZE_MAX_60;
- instLB->ISACencLB_obj.maxRateInBytes = STREAM_SIZE_MAX_30;
- instLB->ISACencLB_obj.enforceFrameSize = 0;
- /* invalid value prevents getRedPayload to
- run before encoder is called */
- instLB->ISACencLB_obj.lastBWIdx = -1;
- return statusInit;
-}
-
-static WebRtc_Word16 EncoderInitUb(
- ISACUBStruct* instUB,
- WebRtc_Word16 bandwidth)
-{
- WebRtc_Word16 statusInit = 0;
- int k;
-
- /* Init stream vector to zero */
- for(k = 0; k < STREAM_SIZE_MAX_60; k++)
- {
- instUB->ISACencUB_obj.bitstr_obj.stream[k] = 0;
- }
-
- WebRtcIsac_InitMasking(&instUB->ISACencUB_obj.maskfiltstr_obj);
- WebRtcIsac_InitPreFilterbank(&instUB->ISACencUB_obj.prefiltbankstr_obj);
-
- if(bandwidth == isac16kHz)
- {
- instUB->ISACencUB_obj.buffer_index = LB_TOTAL_DELAY_SAMPLES;
- }
- else
- {
- instUB->ISACencUB_obj.buffer_index = 0;
- }
- /* default for I-mode */
- instUB->ISACencUB_obj.bottleneck = 32000;
- // These store the limits for the wideband + super-wideband bit-stream.
- instUB->ISACencUB_obj.maxPayloadSizeBytes = STREAM_SIZE_MAX_30 << 1;
- // This has to be updated after each lower-band encoding to guarantee
- // a correct payload-limitation.
- instUB->ISACencUB_obj.numBytesUsed = 0;
- memset(instUB->ISACencUB_obj.data_buffer_float, 0,
- (MAX_FRAMESAMPLES + LB_TOTAL_DELAY_SAMPLES) * sizeof(float));
-
- memcpy(&(instUB->ISACencUB_obj.lastLPCVec),
- WebRtcIsac_kMeanLarUb16, sizeof(double) * UB_LPC_ORDER);
-
- return statusInit;
-}
-
-
-WebRtc_Word16 WebRtcIsac_EncoderInit(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word16 codingMode)
-{
- ISACMainStruct *instISAC;
- WebRtc_Word16 status;
-
- instISAC = (ISACMainStruct*)ISAC_main_inst;
-
- if((codingMode != 0) && (codingMode != 1))
- {
- instISAC->errorCode = ISAC_DISALLOWED_CODING_MODE;
- return -1;
- }
- // default bottleneck
- instISAC->bottleneck = MAX_ISAC_BW;
-
- if(instISAC->encoderSamplingRateKHz == kIsacWideband)
- {
- instISAC->bandwidthKHz = isac8kHz;
- instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX_60;
- instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX_30;
- }
- else
- {
- instISAC->bandwidthKHz = isac16kHz;
- instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX;
- instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX;
- }
-
- // Channel-adaptive = 0; Instantaneous (Channel-independent) = 1;
- instISAC->codingMode = codingMode;
-
- WebRtcIsac_InitBandwidthEstimator(&instISAC->bwestimator_obj,
- instISAC->encoderSamplingRateKHz,
- instISAC->decoderSamplingRateKHz);
-
- WebRtcIsac_InitRateModel(&instISAC->rate_data_obj);
- /* default for I-mode */
- instISAC->MaxDelay = 10.0;
-
- status = EncoderInitLb(&instISAC->instLB, codingMode,
- instISAC->encoderSamplingRateKHz);
- if(status < 0)
- {
- instISAC->errorCode = -status;
- return -1;
- }
-
- if(instISAC->encoderSamplingRateKHz == kIsacSuperWideband)
- {
- // Initialize encoder filter-bank.
- memset(instISAC->analysisFBState1, 0,
- FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
- memset(instISAC->analysisFBState2, 0,
- FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
-
- status = EncoderInitUb(&(instISAC->instUB),
- instISAC->bandwidthKHz);
- if(status < 0)
- {
- instISAC->errorCode = -status;
- return -1;
- }
- }
- // Initializtion is successful, set the flag
- instISAC->initFlag |= BIT_MASK_ENC_INIT;
- return 0;
-}
-
-
-/****************************************************************************
- * WebRtcIsac_Encode(...)
- *
- * This function encodes 10ms frame(s) and inserts it into a package.
- * Input speech length has to be 160 samples (10ms). The encoder buffers those
- * 10ms frames until it reaches the chosen Framesize (480 or 960 samples
- * corresponding to 30 or 60 ms frames), and then proceeds to the encoding.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - speechIn : input speech vector.
- *
- * Output:
- * - encoded : the encoded data vector
- *
- * Return value:
- * : >0 - Length (in bytes) of coded data
- * : 0 - The buffer didn't reach the chosen
- * frameSize so it keeps buffering speech
- * samples.
- * : -1 - Error
- */
-WebRtc_Word16 WebRtcIsac_Encode(
- ISACStruct* ISAC_main_inst,
- const WebRtc_Word16* speechIn,
- WebRtc_Word16* encoded)
-{
- ISACMainStruct* instISAC;
- ISACLBStruct* instLB;
- ISACUBStruct* instUB;
-
- float inFrame[FRAMESAMPLES_10ms];
- WebRtc_Word16 speechInLB[FRAMESAMPLES_10ms];
- WebRtc_Word16 speechInUB[FRAMESAMPLES_10ms];
- WebRtc_Word16 streamLenLB = 0;
- WebRtc_Word16 streamLenUB = 0;
- WebRtc_Word16 streamLen = 0;
- WebRtc_Word16 k = 0;
- WebRtc_UWord8* ptrEncodedUW8 = (WebRtc_UWord8*)encoded;
- int garbageLen = 0;
- WebRtc_Word32 bottleneck = 0;
- WebRtc_Word16 bottleneckIdx = 0;
- WebRtc_Word16 jitterInfo = 0;
-
- instISAC = (ISACMainStruct*)ISAC_main_inst;
- instLB = &(instISAC->instLB);
- instUB = &(instISAC->instUB);
-
- /* check if encoder initiated */
- if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
- BIT_MASK_ENC_INIT)
- {
- instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
- return -1;
- }
-
- if(instISAC->encoderSamplingRateKHz == kIsacSuperWideband)
- {
- WebRtcSpl_AnalysisQMF(speechIn, speechInLB, speechInUB,
- instISAC->analysisFBState1, instISAC->analysisFBState2);
-
- /* convert from fixed to floating point */
- for(k = 0; k < FRAMESAMPLES_10ms; k++)
- {
- inFrame[k] = (float)speechInLB[k];
- }
- }
- else
- {
- for(k = 0; k < FRAMESAMPLES_10ms; k++)
- {
- inFrame[k] = (float) speechIn[k];
- }
- }
-
- /* add some noise to avoid denormal numbers */
- inFrame[0] += (float)1.23455334e-3;
- inFrame[1] -= (float)2.04324239e-3;
- inFrame[2] += (float)1.90854954e-3;
- inFrame[9] += (float)1.84854878e-3;
-
-
- // This function will update the bottleneck if required
- UpdateBottleneck(instISAC);
-
- // Get the bandwith information which has to be sent to the other side
- GetSendBandwidthInfo(instISAC, &bottleneckIdx, &jitterInfo);
-
- //
- // ENCODE LOWER-BAND
- //
- streamLenLB = WebRtcIsac_EncodeLb(inFrame, &instLB->ISACencLB_obj,
- instISAC->codingMode, bottleneckIdx);
-
- if(streamLenLB < 0)
- {
- return -1;
- }
-
- if(instISAC->encoderSamplingRateKHz == kIsacSuperWideband)
- {
- instUB = &(instISAC->instUB);
-
- // convert to float
- for(k = 0; k < FRAMESAMPLES_10ms; k++)
- {
- inFrame[k] = (float) speechInUB[k];
- }
-
- /* add some noise to avoid denormal numbers */
- inFrame[0] += (float)1.23455334e-3;
- inFrame[1] -= (float)2.04324239e-3;
- inFrame[2] += (float)1.90854954e-3;
- inFrame[9] += (float)1.84854878e-3;
-
- // Tell to upper-band the number of bytes used so far.
- // This is for payload limitation.
- instUB->ISACencUB_obj.numBytesUsed = streamLenLB + 1 +
- LEN_CHECK_SUM_WORD8;
-
- //
- // ENCODE UPPER-BAND
- //
- switch(instISAC->bandwidthKHz)
- {
- case isac12kHz:
- {
- streamLenUB = WebRtcIsac_EncodeUb12(inFrame,
- &instUB->ISACencUB_obj,
- jitterInfo);
- break;
- }
- case isac16kHz:
- {
- streamLenUB = WebRtcIsac_EncodeUb16(inFrame,
- &instUB->ISACencUB_obj,
- jitterInfo);
- break;
- }
- case isac8kHz:
- {
- streamLenUB = 0;
- break;
- }
- }
-
- if((streamLenUB < 0) &&
- (streamLenUB != -ISAC_PAYLOAD_LARGER_THAN_LIMIT))
- {
- // an error has happened but this is not the error due to a
- // bit-stream larger than the limit
- return -1;
- }
-
- if(streamLenLB == 0)
- {
- return 0;
- }
-
- // One bite is allocated for the length. According to older decoders
- // so the length bit-stream plus one byte for size and
- // LEN_CHECK_SUM_WORD8 for the checksum should be less than or equal
- // to 255.
- if((streamLenUB > (255 - (LEN_CHECK_SUM_WORD8 + 1))) ||
- (streamLenUB == -ISAC_PAYLOAD_LARGER_THAN_LIMIT))
- {
- // we have got a too long bit-stream we skip the upper-band
- // bit-stream for this frame.
- streamLenUB = 0;
- }
-
- memcpy(ptrEncodedUW8, instLB->ISACencLB_obj.bitstr_obj.stream,
- streamLenLB);
- streamLen = streamLenLB;
- if(streamLenUB > 0)
- {
- ptrEncodedUW8[streamLenLB] = (WebRtc_UWord8)(streamLenUB + 1 +
- LEN_CHECK_SUM_WORD8);
- memcpy(&ptrEncodedUW8[streamLenLB + 1],
- instUB->ISACencUB_obj.bitstr_obj.stream, streamLenUB);
- streamLen += ptrEncodedUW8[streamLenLB];
- }
- else
- {
- ptrEncodedUW8[streamLenLB] = 0;
- }
- }
- else
- {
- if(streamLenLB == 0)
- {
- return 0;
- }
- memcpy(ptrEncodedUW8, instLB->ISACencLB_obj.bitstr_obj.stream,
- streamLenLB);
- streamLenUB = 0;
- streamLen = streamLenLB;
- }
-
- // Add Garbage if required.
- WebRtcIsac_GetUplinkBandwidth(&instISAC->bwestimator_obj, &bottleneck);
- if(instISAC->codingMode == 0)
- {
- int minBytes;
- int limit;
- WebRtc_UWord8* ptrGarbage;
-
- instISAC->MaxDelay = (double)WebRtcIsac_GetUplinkMaxDelay(
- &instISAC->bwestimator_obj);
-
- /* update rate model and get minimum number of bytes in this packet */
- minBytes = WebRtcIsac_GetMinBytes(&(instISAC->rate_data_obj),
- streamLen, instISAC->instLB.ISACencLB_obj.current_framesamples,
- bottleneck, instISAC->MaxDelay, instISAC->bandwidthKHz);
-
- /* Make sure MinBytes does not exceed packet size limit */
- if(instISAC->bandwidthKHz == isac8kHz)
- {
- if(instLB->ISACencLB_obj.current_framesamples == FRAMESAMPLES)
- {
- limit = instLB->ISACencLB_obj.payloadLimitBytes30;
- }
- else
- {
- limit = instLB->ISACencLB_obj.payloadLimitBytes60;
- }
- }
- else
- {
- limit = instUB->ISACencUB_obj.maxPayloadSizeBytes;
- }
- minBytes = (minBytes > limit)? limit:minBytes;
-
- /* Make sure we don't allow more than 255 bytes of garbage data.
- We store the length of the garbage data in 8 bits in the bitstream,
- 255 is the max garbage length we can signal using 8 bits. */
- if((instISAC->bandwidthKHz == isac8kHz) ||
- (streamLenUB == 0))
- {
- ptrGarbage = &ptrEncodedUW8[streamLenLB];
- limit = streamLen + 255;
- }
- else
- {
- ptrGarbage = &ptrEncodedUW8[streamLenLB + 1 + streamLenUB];
- limit = streamLen + (255 - ptrEncodedUW8[streamLenLB]);
- }
- minBytes = (minBytes > limit)? limit:minBytes;
-
- garbageLen = (minBytes > streamLen)? (minBytes - streamLen):0;
-
- /* Save data for creation of multiple bitstreams */
- //ISACencLB_obj->SaveEnc_obj.minBytes = MinBytes;
-
- /* if bitstream is too short, add garbage at the end */
- if(garbageLen > 0)
- {
- for(k = 0; k < garbageLen; k++)
- {
- ptrGarbage[k] = (WebRtc_UWord8)(rand() & 0xFF);
- }
-
- // for a correct length of the upper-band bit-stream together
- // with the garbage. Garbage is embeded in upper-band bit-stream.
- // That is the only way to preserve backward compatibility.
- if((instISAC->bandwidthKHz == isac8kHz) ||
- (streamLenUB == 0))
- {
- ptrEncodedUW8[streamLenLB] = (WebRtc_UWord8)garbageLen;
- }
- else
- {
- ptrEncodedUW8[streamLenLB] += (WebRtc_UWord8)garbageLen;
- // write the length of the garbage at the end of the upper-band
- // bit-stream, if exists. This helps for sanity check.
- ptrEncodedUW8[streamLenLB + 1 + streamLenUB] = (WebRtc_UWord8)garbageLen;
-
- }
-
- streamLen += garbageLen;
- }
- }
- else
- {
- /* update rate model */
- WebRtcIsac_UpdateRateModel(&instISAC->rate_data_obj, streamLen,
- instISAC->instLB.ISACencLB_obj.current_framesamples, bottleneck);
- garbageLen = 0;
- }
-
- // Generate CRC if required.
- if((instISAC->bandwidthKHz != isac8kHz) &&
- (streamLenUB > 0))
- {
- WebRtc_UWord32 crc;
-
- WebRtcIsac_GetCrc((WebRtc_Word16*)(&(ptrEncodedUW8[streamLenLB + 1])),
- streamLenUB + garbageLen, &crc);
-#ifndef WEBRTC_BIG_ENDIAN
- for(k = 0; k < LEN_CHECK_SUM_WORD8; k++)
- {
- ptrEncodedUW8[streamLen - LEN_CHECK_SUM_WORD8 + k] =
- (WebRtc_UWord8)((crc >> (24 - k * 8)) & 0xFF);
- }
-#else
- memcpy(&ptrEncodedUW8[streamLenLB + streamLenUB + 1], &crc,
- LEN_CHECK_SUM_WORD8);
-#endif
- }
-
- return streamLen;
-}
-
-
-/******************************************************************************
- * WebRtcIsac_GetNewBitStream(...)
- *
- * This function returns encoded data, with the recieved bwe-index in the
- * stream. If the rate is set to a value less than bottleneck of codec
- * the new bistream will be re-encoded with the given target rate.
- * It should always return a complete packet, i.e. only called once
- * even for 60 msec frames.
- *
- * NOTE 1! This function does not write in the ISACStruct, it is not allowed.
- * NOTE 3! Rates larger than the bottleneck of the codec will be limited
- * to the current bottleneck.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - bweIndex : Index of bandwidth estimate to put in new
- * bitstream
- * - rate : target rate of the transcoder is bits/sec.
- * Valid values are the accepted rate in iSAC,
- * i.e. 10000 to 56000.
- *
- * Output:
- * - encoded : The encoded data vector
- *
- * Return value : >0 - Length (in bytes) of coded data
- * -1 - Error or called in SWB mode
- * NOTE! No error code is written to
- * the struct since it is only allowed to read
- * the struct.
- */
-WebRtc_Word16 WebRtcIsac_GetNewBitStream(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word16 bweIndex,
- WebRtc_Word16 jitterInfo,
- WebRtc_Word32 rate,
- WebRtc_Word16* encoded,
- WebRtc_Word16 isRCU)
-{
- Bitstr iSACBitStreamInst; /* Local struct for bitstream handling */
- WebRtc_Word16 streamLenLB;
- WebRtc_Word16 streamLenUB;
- WebRtc_Word16 totalStreamLen;
- double gain2;
- double gain1;
- float scale;
- enum ISACBandwidth bandwidthKHz;
- double rateLB;
- double rateUB;
- WebRtc_Word32 currentBN;
- ISACMainStruct* instISAC;
- WebRtc_UWord8* encodedPtrUW8 = (WebRtc_UWord8*)encoded;
- WebRtc_UWord32 crc;
-#ifndef WEBRTC_BIG_ENDIAN
- WebRtc_Word16 k;
-#endif
-
- instISAC = (ISACMainStruct*)ISAC_main_inst;
-
- if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
- BIT_MASK_ENC_INIT)
- {
- return -1;
- }
-
- // Get the bottleneck of this iSAC and limit the
- // given rate to the current bottleneck.
- WebRtcIsac_GetUplinkBw(ISAC_main_inst, ¤tBN);
- if(rate > currentBN)
- {
- rate = currentBN;
- }
-
- if(WebRtcIsac_RateAllocation(rate, &rateLB, &rateUB, &bandwidthKHz) < 0)
- {
- return -1;
- }
-
- // Cannot transcode from 16 kHz to 12 kHz
- if((bandwidthKHz == isac12kHz) &&
- (instISAC->bandwidthKHz == isac16kHz))
- {
- return -1;
- }
-
- // These gains are in dB
- // gain for the given rate.
- gain1 = WebRtcIsac_GetSnr(rateLB,
- instISAC->instLB.ISACencLB_obj.current_framesamples);
- // gain of this iSAC
- gain2 = WebRtcIsac_GetSnr(
- instISAC->instLB.ISACencLB_obj.bottleneck,
- instISAC->instLB.ISACencLB_obj.current_framesamples);
-
- // scale is the ratio of two gains in normal domain.
- scale = (float)pow(10, (gain1 - gain2) / 20.0);
- // change the scale if this is a RCU bit-stream.
- scale = (isRCU)? (scale * RCU_TRANSCODING_SCALE):scale;
-
- streamLenLB = WebRtcIsac_EncodeStoredDataLb(
- &instISAC->instLB.ISACencLB_obj.SaveEnc_obj, &iSACBitStreamInst,
- bweIndex, scale);
-
- if(streamLenLB < 0)
- {
- return -1;
- }
-
- /* convert from bytes to WebRtc_Word16 */
- memcpy(encoded, iSACBitStreamInst.stream, streamLenLB);
-
- if(bandwidthKHz == isac8kHz)
- {
- return streamLenLB;
- }
-
- totalStreamLen = streamLenLB;
- // super-wideband is always at 30ms.
- // These gains are in dB
- // gain for the given rate.
- gain1 = WebRtcIsac_GetSnr(rateUB, FRAMESAMPLES);
- // gain of this iSAC
- gain2 = WebRtcIsac_GetSnr(
- instISAC->instUB.ISACencUB_obj.bottleneck, FRAMESAMPLES);
-
- // scale is the ratio of two gains in normal domain.
- scale = (float)pow(10, (gain1 - gain2) / 20.0);
-
- // change the scale if this is a RCU bit-stream.
- scale = (isRCU)? (scale * RCU_TRANSCODING_SCALE_UB):scale;
-
- switch(instISAC->bandwidthKHz)
- {
- case isac12kHz:
- {
- streamLenUB = WebRtcIsac_EncodeStoredDataUb12(
- &(instISAC->instUB.ISACencUB_obj.SaveEnc_obj),
- &iSACBitStreamInst, jitterInfo, scale);
- break;
- }
- case isac16kHz:
- {
- streamLenUB = WebRtcIsac_EncodeStoredDataUb16(
- &(instISAC->instUB.ISACencUB_obj.SaveEnc_obj),
- &iSACBitStreamInst, jitterInfo, scale);
- break;
- }
- default:
- return -1;
- }
-
- if(streamLenUB < 0)
- {
- return -1;
- }
-
- if(streamLenUB + 1 + LEN_CHECK_SUM_WORD8 > 255)
- {
- return streamLenLB;
- }
-
- totalStreamLen = streamLenLB + streamLenUB + 1 + LEN_CHECK_SUM_WORD8;
- encodedPtrUW8[streamLenLB] = streamLenUB + 1 + LEN_CHECK_SUM_WORD8;
-
- memcpy(&encodedPtrUW8[streamLenLB+1], iSACBitStreamInst.stream,
- streamLenUB);
-
- WebRtcIsac_GetCrc((WebRtc_Word16*)(&(encodedPtrUW8[streamLenLB + 1])),
- streamLenUB, &crc);
-#ifndef WEBRTC_BIG_ENDIAN
- for(k = 0; k < LEN_CHECK_SUM_WORD8; k++)
- {
- encodedPtrUW8[totalStreamLen - LEN_CHECK_SUM_WORD8 + k] =
- (WebRtc_UWord8)((crc >> (24 - k * 8)) & 0xFF);
- }
-#else
- memcpy(&encodedPtrUW8[streamLenLB + streamLenUB + 1], &crc,
- LEN_CHECK_SUM_WORD8);
-#endif
-
-
- return totalStreamLen;
-}
-
-
-/****************************************************************************
- * DecoderInitLb(...) - internal function for initialization of
- * Lower Band
- * DecoderInitUb(...) - internal function for initialization of
- * Upper Band
- * WebRtcIsac_DecoderInit(...) - API function
- *
- * This function initializes a ISAC instance prior to the decoder calls.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- *
- * Return value
- * : 0 - Ok
- * -1 - Error
- */
-static WebRtc_Word16 DecoderInitLb(
- ISACLBStruct* instISAC)
-{
- int i;
- /* Init stream vector to zero */
- for (i=0; i<STREAM_SIZE_MAX_60; i++)
- {
- instISAC->ISACdecLB_obj.bitstr_obj.stream[i] = 0;
- }
-
- WebRtcIsac_InitMasking(&instISAC->ISACdecLB_obj.maskfiltstr_obj);
- WebRtcIsac_InitPostFilterbank(
- &instISAC->ISACdecLB_obj.postfiltbankstr_obj);
- WebRtcIsac_InitPitchFilter(&instISAC->ISACdecLB_obj.pitchfiltstr_obj);
-
- return (0);
-}
-
-static WebRtc_Word16 DecoderInitUb(
- ISACUBStruct* instISAC)
-{
- int i;
- /* Init stream vector to zero */
- for (i = 0; i < STREAM_SIZE_MAX_60; i++)
- {
- instISAC->ISACdecUB_obj.bitstr_obj.stream[i] = 0;
- }
-
- WebRtcIsac_InitMasking(&instISAC->ISACdecUB_obj.maskfiltstr_obj);
- WebRtcIsac_InitPostFilterbank(
- &instISAC->ISACdecUB_obj.postfiltbankstr_obj);
- return (0);
-}
-
-WebRtc_Word16 WebRtcIsac_DecoderInit(
- ISACStruct *ISAC_main_inst)
-{
- ISACMainStruct* instISAC;
-
- instISAC = (ISACMainStruct*)ISAC_main_inst;
-
- if(DecoderInitLb(&instISAC->instLB) < 0)
- {
- return -1;
- }
-
- if(instISAC->decoderSamplingRateKHz == kIsacSuperWideband)
- {
- memset(instISAC->synthesisFBState1, 0,
- FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
- memset(instISAC->synthesisFBState2, 0,
- FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
-
- if(DecoderInitUb(&(instISAC->instUB)) < 0)
- {
- return -1;
- }
- }
-
- if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
- BIT_MASK_ENC_INIT)
- {
- WebRtcIsac_InitBandwidthEstimator(&instISAC->bwestimator_obj,
- instISAC->encoderSamplingRateKHz,
- instISAC->decoderSamplingRateKHz);
- }
-
- instISAC->initFlag |= BIT_MASK_DEC_INIT;
-
- instISAC->resetFlag_8kHz = 0;
-
- return 0;
-}
-
-
-/****************************************************************************
- * WebRtcIsac_UpdateBwEstimate(...)
- *
- * This function updates the estimate of the bandwidth.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - encoded : encoded ISAC frame(s).
- * - packet_size : size of the packet.
- * - rtp_seq_number : the RTP number of the packet.
- * - arr_ts : the arrival time of the packet (from NetEq)
- * in samples.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-WebRtc_Word16 WebRtcIsac_UpdateBwEstimate(
- ISACStruct* ISAC_main_inst,
- const WebRtc_UWord16* encoded,
- WebRtc_Word32 packet_size,
- WebRtc_UWord16 rtp_seq_number,
- WebRtc_UWord32 send_ts,
- WebRtc_UWord32 arr_ts)
-{
- ISACMainStruct *instISAC;
- Bitstr streamdata;
-#ifndef WEBRTC_BIG_ENDIAN
- int k;
-#endif
- WebRtc_Word16 err;
-
- /* typecast pointer to real structure */
- instISAC = (ISACMainStruct *)ISAC_main_inst;
-
- /* check if decoder initiated */
- if((instISAC->initFlag & BIT_MASK_DEC_INIT) !=
- BIT_MASK_DEC_INIT)
- {
- instISAC->errorCode = ISAC_DECODER_NOT_INITIATED;
- return -1;
- }
-
- if(packet_size <= 0)
- {
- /* return error code if the packet length is null */
- instISAC->errorCode = ISAC_EMPTY_PACKET;
- return -1;
- }
-
- streamdata.W_upper = 0xFFFFFFFF;
- streamdata.streamval = 0;
- streamdata.stream_index = 0;
-
-#ifndef WEBRTC_BIG_ENDIAN
- for(k = 0; k < 10; k++)
- {
- streamdata.stream[k] = (WebRtc_UWord8) ((encoded[k>>1] >>
- ((k&1) << 3)) & 0xFF);
- }
-#else
- memcpy(streamdata.stream, encoded, 10);
-#endif
-
- err = WebRtcIsac_EstimateBandwidth(&instISAC->bwestimator_obj, &streamdata,
- packet_size, rtp_seq_number, send_ts, arr_ts,
- instISAC->encoderSamplingRateKHz,
- instISAC->decoderSamplingRateKHz);
-
- if(err < 0)
- {
- /* return error code if something went wrong */
- instISAC->errorCode = -err;
- return -1;
- }
-
- return 0;
-}
-
-static WebRtc_Word16 Decode(
- ISACStruct* ISAC_main_inst,
- const WebRtc_UWord16* encoded,
- WebRtc_Word16 lenEncodedBytes,
- WebRtc_Word16* decoded,
- WebRtc_Word16* speechType,
- WebRtc_Word16 isRCUPayload)
-{
- /* number of samples (480 or 960), output from decoder
- that were actually used in the encoder/decoder
- (determined on the fly) */
- ISACMainStruct* instISAC;
- ISACUBDecStruct* decInstUB;
- ISACLBDecStruct* decInstLB;
-
- WebRtc_Word16 numSamplesLB;
- WebRtc_Word16 numSamplesUB;
- WebRtc_Word16 speechIdx;
- float outFrame[MAX_FRAMESAMPLES];
- WebRtc_Word16 outFrameLB[MAX_FRAMESAMPLES];
- WebRtc_Word16 outFrameUB[MAX_FRAMESAMPLES];
- WebRtc_Word16 numDecodedBytesLB;
- WebRtc_Word16 numDecodedBytesUB;
- WebRtc_Word16 lenEncodedLBBytes;
- WebRtc_Word16 validChecksum = 1;
- WebRtc_Word16 k;
- WebRtc_UWord8* ptrEncodedUW8 = (WebRtc_UWord8*)encoded;
- WebRtc_UWord16 numLayer;
- WebRtc_Word16 totSizeBytes;
- WebRtc_Word16 err;
-
- instISAC = (ISACMainStruct*)ISAC_main_inst;
- decInstUB = &(instISAC->instUB.ISACdecUB_obj);
- decInstLB = &(instISAC->instLB.ISACdecLB_obj);
-
- /* check if decoder initiated */
- if((instISAC->initFlag & BIT_MASK_DEC_INIT) !=
- BIT_MASK_DEC_INIT)
- {
- instISAC->errorCode = ISAC_DECODER_NOT_INITIATED;
- return -1;
- }
-
- if(lenEncodedBytes <= 0)
- {
- /* return error code if the packet length is null */
- instISAC->errorCode = ISAC_EMPTY_PACKET;
- return -1;
- }
-
- // the size of the rncoded lower-band is bounded by
- // STREAM_SIZE_MAX,
- // If a payload with the size larger than STREAM_SIZE_MAX
- // is received, it is not considered erroneous.
- lenEncodedLBBytes = (lenEncodedBytes > STREAM_SIZE_MAX)
- ? STREAM_SIZE_MAX:lenEncodedBytes;
-
- // Copy to lower-band bit-stream structure
- memcpy(instISAC->instLB.ISACdecLB_obj.bitstr_obj.stream, ptrEncodedUW8,
- lenEncodedLBBytes);
-
- // Regardless of that the current codec is setup to work in
- // wideband or super-wideband, the decoding of the lower-band
- // has to be performed.
- numDecodedBytesLB = WebRtcIsac_DecodeLb(outFrame, decInstLB,
- &numSamplesLB, isRCUPayload);
-
- // Check for error
- if((numDecodedBytesLB < 0) ||
- (numDecodedBytesLB > lenEncodedLBBytes) ||
- (numSamplesLB > MAX_FRAMESAMPLES))
- {
- instISAC->errorCode = ISAC_LENGTH_MISMATCH;
- return -1;
- }
-
- // Error Check, we accept multi-layer bit-stream
- // This will limit number of iterations of the
- // while loop. Even withouut this the number of iterations
- // is limited.
- numLayer = 1;
- totSizeBytes = numDecodedBytesLB;
- while(totSizeBytes != lenEncodedBytes)
- {
- if((totSizeBytes > lenEncodedBytes) ||
- (ptrEncodedUW8[totSizeBytes] == 0) ||
- (numLayer > MAX_NUM_LAYERS))
- {
- instISAC->errorCode = ISAC_LENGTH_MISMATCH;
- return -1;
- }
- totSizeBytes += ptrEncodedUW8[totSizeBytes];
- numLayer++;
- }
-
- if(instISAC->decoderSamplingRateKHz == kIsacWideband)
- {
- for(k = 0; k < numSamplesLB; k++)
- {
- if(outFrame[k] > 32767)
- {
- decoded[k] = 32767;
- }
- else if(outFrame[k] < -32768)
- {
- decoded[k] = -32768;
- }
- else
- {
- decoded[k] = (WebRtc_Word16)WebRtcIsac_lrint(outFrame[k]);
- }
- }
- numSamplesUB = 0;
- }
- else
- {
- WebRtc_UWord32 crc;
- // We don't accept larger than 30ms (480 samples at lower-band)
- // frame-size.
- for(k = 0; k < numSamplesLB; k++)
- {
- if(outFrame[k] > 32767)
- {
- outFrameLB[k] = 32767;
- }
- else if(outFrame[k] < -32768)
- {
- outFrameLB[k] = -32768;
- }
- else
- {
- outFrameLB[k] = (WebRtc_Word16)WebRtcIsac_lrint(outFrame[k]);
- }
- }
-
- //numSamplesUB = numSamplesLB;
-
- // Check for possible error, and if upper-band stream exist.
- if(numDecodedBytesLB == lenEncodedBytes)
- {
- // Decoding was successful. No super-wideband bitstream
- // exists.
- numSamplesUB = numSamplesLB;
- memset(outFrameUB, 0, sizeof(WebRtc_Word16) * numSamplesUB);
-
- // Prepare for the potential increase of signal bandwidth
- instISAC->resetFlag_8kHz = 2;
- }
- else
- {
- // this includes the check sum and the bytes that stores the
- // length
- WebRtc_Word16 lenNextStream = ptrEncodedUW8[numDecodedBytesLB];
-
- // Is this garbage or valid super-wideband bit-stream?
- // Check if checksum is valid
- if(lenNextStream <= (LEN_CHECK_SUM_WORD8 + 1))
- {
- // such a small second layer cannot be super-wideband layer.
- // It must be a short garbage.
- validChecksum = 0;
- }
- else
- {
- // Run CRC to see if the checksum match.
- WebRtcIsac_GetCrc((WebRtc_Word16*)(
- &ptrEncodedUW8[numDecodedBytesLB + 1]),
- lenNextStream - LEN_CHECK_SUM_WORD8 - 1, &crc);
-
- validChecksum = 1;
- for(k = 0; k < LEN_CHECK_SUM_WORD8; k++)
- {
- validChecksum &= (((crc >> (24 - k * 8)) & 0xFF) ==
- ptrEncodedUW8[numDecodedBytesLB + lenNextStream -
- LEN_CHECK_SUM_WORD8 + k]);
- }
- }
-
- if(!validChecksum)
- {
- // this is a garbage, we have received a wideband
- // bit-stream with garbage
- numSamplesUB = numSamplesLB;
- memset(outFrameUB, 0, sizeof(WebRtc_Word16) * numSamplesUB);
- }
- else
- {
- // A valid super-wideband biststream exists.
- enum ISACBandwidth bandwidthKHz;
- WebRtc_Word32 maxDelayBit;
-
- //instISAC->bwestimator_obj.incomingStreamSampFreq =
- // kIsacSuperWideband;
- // If we have super-wideband bit-stream, we cannot
- // have 60 ms frame-size.
- if(numSamplesLB > FRAMESAMPLES)
- {
- instISAC->errorCode = ISAC_LENGTH_MISMATCH;
- return -1;
- }
-
- // the rest of the bit-stream contains the upper-band
- // bit-stream curently this is the only thing there,
- // however, we might add more layers.
-
- // Have to exclude one byte where the length is stored
- // and last 'LEN_CHECK_SUM_WORD8' bytes where the
- // checksum is stored.
- lenNextStream -= (LEN_CHECK_SUM_WORD8 + 1);
-
- memcpy(decInstUB->bitstr_obj.stream,
- &ptrEncodedUW8[numDecodedBytesLB + 1], lenNextStream);
-
- // THIS IS THE FIRST DECODING
- decInstUB->bitstr_obj.W_upper = 0xFFFFFFFF;
- decInstUB->bitstr_obj.streamval = 0;
- decInstUB->bitstr_obj.stream_index = 0;
-
- // Decode jitter infotmation
- err = WebRtcIsac_DecodeJitterInfo(&decInstUB->bitstr_obj,
- &maxDelayBit);
- // error check
- if(err < 0)
- {
- instISAC->errorCode = -err;
- return -1;
- }
-
- // Update jitter info which is in the upper-band bit-stream
- // only if the encoder is in super-wideband. Otherwise,
- // the jitter info is already embeded in bandwidth index
- // and has been updated.
- if(instISAC->encoderSamplingRateKHz == kIsacSuperWideband)
- {
- err = WebRtcIsac_UpdateUplinkJitter(
- &(instISAC->bwestimator_obj), maxDelayBit);
- if(err < 0)
- {
- instISAC->errorCode = -err;
- return -1;
- }
- }
-
- // decode bandwidth information
- err = WebRtcIsac_DecodeBandwidth(&decInstUB->bitstr_obj,
- &bandwidthKHz);
- if(err < 0)
- {
- instISAC->errorCode = -err;
- return -1;
- }
-
- switch(bandwidthKHz)
- {
- case isac12kHz:
- {
- numDecodedBytesUB = WebRtcIsac_DecodeUb12(outFrame,
- decInstUB, isRCUPayload);
-
- // Hang-over for transient alleviation -
- // wait two frames to add the upper band going up from 8 kHz
- if (instISAC->resetFlag_8kHz > 0)
- {
- if (instISAC->resetFlag_8kHz == 2)
- {
- // Silence first and a half frame
- memset(outFrame, 0, MAX_FRAMESAMPLES *
- sizeof(float));
- }
- else
- {
- const float rampStep = 2.0f / MAX_FRAMESAMPLES;
- float rampVal = 0;
- memset(outFrame, 0, (MAX_FRAMESAMPLES>>1) *
- sizeof(float));
-
- // Ramp up second half of second frame
- for(k = MAX_FRAMESAMPLES/2; k < MAX_FRAMESAMPLES; k++)
- {
- outFrame[k] *= rampVal;
- rampVal += rampStep;
- }
- }
- instISAC->resetFlag_8kHz -= 1;
- }
-
- break;
- }
- case isac16kHz:
- {
- numDecodedBytesUB = WebRtcIsac_DecodeUb16(outFrame,
- decInstUB, isRCUPayload);
- break;
- }
- default:
- return -1;
- }
-
- // it might be less due to garbage.
- if((numDecodedBytesUB != lenNextStream) &&
- (numDecodedBytesUB != (lenNextStream - ptrEncodedUW8[
- numDecodedBytesLB + 1 + numDecodedBytesUB])))
- {
- instISAC->errorCode = ISAC_LENGTH_MISMATCH;
- return -1;
- }
-
- // If there is no error Upper-band always decodes
- // 30 ms (480 samples)
- numSamplesUB = FRAMESAMPLES;
-
- // Convert to W16
- for(k = 0; k < numSamplesUB; k++)
- {
- if(outFrame[k] > 32767)
- {
- outFrameUB[k] = 32767;
- }
- else if(outFrame[k] < -32768)
- {
- outFrameUB[k] = -32768;
- }
- else
- {
- outFrameUB[k] = (WebRtc_Word16)WebRtcIsac_lrint(
- outFrame[k]);
- }
- }
- }
- }
-
- speechIdx = 0;
- while(speechIdx < numSamplesLB)
- {
- WebRtcSpl_SynthesisQMF(&outFrameLB[speechIdx],
- &outFrameUB[speechIdx], &decoded[(speechIdx<<1)],
- instISAC->synthesisFBState1, instISAC->synthesisFBState2);
-
- speechIdx += FRAMESAMPLES_10ms;
- }
- }
- *speechType = 0;
- return (numSamplesLB + numSamplesUB);
-}
-
-
-
-
-
-
-
-/****************************************************************************
- * WebRtcIsac_Decode(...)
- *
- * This function decodes a ISAC frame. Output speech length
- * will be a multiple of 480 samples: 480 or 960 samples,
- * depending on the frameSize (30 or 60 ms).
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - encoded : encoded ISAC frame(s)
- * - len : bytes in encoded vector
- *
- * Output:
- * - decoded : The decoded vector
- *
- * Return value : >0 - number of samples in decoded vector
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcIsac_Decode(
- ISACStruct* ISAC_main_inst,
- const WebRtc_UWord16* encoded,
- WebRtc_Word16 lenEncodedBytes,
- WebRtc_Word16* decoded,
- WebRtc_Word16* speechType)
-{
- WebRtc_Word16 isRCUPayload = 0;
- return Decode(ISAC_main_inst, encoded, lenEncodedBytes, decoded,
- speechType, isRCUPayload);
-}
-
-/****************************************************************************
- * WebRtcIsac_DecodeRcu(...)
- *
- * This function decodes a redundant (RCU) iSAC frame. Function is called in
- * NetEq with a stored RCU payload i case of packet loss. Output speech length
- * will be a multiple of 480 samples: 480 or 960 samples,
- * depending on the framesize (30 or 60 ms).
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - encoded : encoded ISAC RCU frame(s)
- * - len : bytes in encoded vector
- *
- * Output:
- * - decoded : The decoded vector
- *
- * Return value : >0 - number of samples in decoded vector
- * -1 - Error
- */
-
-
-
-WebRtc_Word16 WebRtcIsac_DecodeRcu(
- ISACStruct* ISAC_main_inst,
- const WebRtc_UWord16* encoded,
- WebRtc_Word16 lenEncodedBytes,
- WebRtc_Word16* decoded,
- WebRtc_Word16* speechType)
-{
- WebRtc_Word16 isRCUPayload = 1;
- return Decode(ISAC_main_inst, encoded, lenEncodedBytes, decoded,
- speechType, isRCUPayload);
-}
-
-
-/****************************************************************************
- * WebRtcIsac_DecodePlc(...)
- *
- * This function conducts PLC for ISAC frame(s). Output speech length
- * will be a multiple of 480 samples: 480 or 960 samples,
- * depending on the frameSize (30 or 60 ms).
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - noOfLostFrames : Number of PLC frames to produce
- *
- * Output:
- * - decoded : The decoded vector
- *
- * Return value : >0 - number of samples in decoded PLC vector
- * -1 - Error
- */
-WebRtc_Word16 WebRtcIsac_DecodePlc(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word16* decoded,
- WebRtc_Word16 noOfLostFrames)
-{
- WebRtc_Word16 numSamples = 0;
- ISACMainStruct* instISAC;
-
-
- /* typecast pointer to real structure */
- instISAC = (ISACMainStruct*)ISAC_main_inst;
-
- /* Limit number of frames to two = 60 msec. Otherwise we exceed data vectors */
- if(noOfLostFrames > 2)
- {
- noOfLostFrames = 2;
- }
-
- /* Get the number of samples per frame */
- switch(instISAC->decoderSamplingRateKHz)
- {
- case kIsacWideband:
- {
- numSamples = 480 * noOfLostFrames;
- break;
- }
- case kIsacSuperWideband:
- {
- numSamples = 960 * noOfLostFrames;
- break;
- }
- }
-
- /* Set output samples to zero */
- memset(decoded, 0, numSamples * sizeof(WebRtc_Word16));
- return numSamples;
-}
-
-
-/****************************************************************************
- * ControlLb(...) - Internal function for controling Lower Band
- * ControlUb(...) - Internal function for controling Upper Band
- * WebRtcIsac_Control(...) - API function
- *
- * This function sets the limit on the short-term average bit rate and the
- * frame length. Should be used only in Instantaneous mode.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - rate : limit on the short-term average bit rate,
- * in bits/second (between 10000 and 32000)
- * - frameSize : number of milliseconds per frame (30 or 60)
- *
- * Return value : 0 - ok
- * -1 - Error
- */
-static WebRtc_Word16 ControlLb(
- ISACLBStruct* instISAC,
- double rate,
- WebRtc_Word16 frameSize)
-{
- if((rate >= 10000) && (rate <= 32000))
- {
- instISAC->ISACencLB_obj.bottleneck = rate;
- }
- else
- {
- return -ISAC_DISALLOWED_BOTTLENECK;
- }
-
- if((frameSize == 30) || (frameSize == 60))
- {
- instISAC->ISACencLB_obj.new_framelength = (FS/1000) * frameSize;
- }
- else
- {
- return -ISAC_DISALLOWED_FRAME_LENGTH;
- }
-
- return 0;
-}
-
-static WebRtc_Word16 ControlUb(
- ISACUBStruct* instISAC,
- double rate)
-{
- if((rate >= 10000) && (rate <= 32000))
- {
- instISAC->ISACencUB_obj.bottleneck = rate;
- }
- else
- {
- return -ISAC_DISALLOWED_BOTTLENECK;
- }
- return 0;
-}
-
-WebRtc_Word16 WebRtcIsac_Control(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word32 bottleneckBPS,
- WebRtc_Word16 frameSize)
-{
- ISACMainStruct *instISAC;
- WebRtc_Word16 status;
- double rateLB;
- double rateUB;
- enum ISACBandwidth bandwidthKHz;
-
-
- /* Typecast pointer to real structure */
- instISAC = (ISACMainStruct*)ISAC_main_inst;
-
- if(instISAC->codingMode == 0)
- {
- /* in adaptive mode */
- instISAC->errorCode = ISAC_MODE_MISMATCH;
- return -1;
- }
-
- /* check if encoder initiated */
- if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
- BIT_MASK_ENC_INIT)
- {
- instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
- return -1;
- }
-
- if(instISAC->encoderSamplingRateKHz == kIsacWideband)
- {
- // if the sampling rate is 16kHz then bandwith should be 8kHz,
- // regardless of bottleneck.
- bandwidthKHz = isac8kHz;
- rateLB = (bottleneckBPS > 32000)? 32000:bottleneckBPS;
- rateUB = 0;
- }
- else
- {
- if(WebRtcIsac_RateAllocation(bottleneckBPS, &rateLB, &rateUB,
- &bandwidthKHz) < 0)
- {
- return -1;
- }
- }
-
- if((instISAC->encoderSamplingRateKHz == kIsacSuperWideband) &&
- (frameSize != 30) &&
- (bandwidthKHz != isac8kHz))
- {
- // Cannot have 60 ms in super-wideband
- instISAC->errorCode = ISAC_DISALLOWED_FRAME_LENGTH;
- return -1;
- }
-
- status = ControlLb(&instISAC->instLB, rateLB, frameSize);
- if(status < 0)
- {
- instISAC->errorCode = -status;
- return -1;
- }
- if(bandwidthKHz != isac8kHz)
- {
- status = ControlUb(&(instISAC->instUB), rateUB);
- if(status < 0)
- {
- instISAC->errorCode = -status;
- return -1;
- }
- }
-
- //
- // Check if bandwidth is changing from wideband to super-wideband
- // then we have to synch data buffer of lower & upper-band. also
- // clean up the upper-band data buffer.
- //
- if((instISAC->bandwidthKHz == isac8kHz) &&
- (bandwidthKHz != isac8kHz))
- {
- memset(instISAC->instUB.ISACencUB_obj.data_buffer_float, 0,
- sizeof(float) * (MAX_FRAMESAMPLES + LB_TOTAL_DELAY_SAMPLES));
-
- if(bandwidthKHz == isac12kHz)
- {
- instISAC->instUB.ISACencUB_obj.buffer_index =
- instISAC->instLB.ISACencLB_obj.buffer_index;
- }
- else
- {
- instISAC->instUB.ISACencUB_obj.buffer_index = LB_TOTAL_DELAY_SAMPLES +
- instISAC->instLB.ISACencLB_obj.buffer_index;
-
- memcpy(&(instISAC->instUB.ISACencUB_obj.lastLPCVec),
- WebRtcIsac_kMeanLarUb16, sizeof(double) * UB_LPC_ORDER);
- }
- }
-
- // update the payload limit it the bandwidth is changing.
- if(instISAC->bandwidthKHz != bandwidthKHz)
- {
- instISAC->bandwidthKHz = bandwidthKHz;
- UpdatePayloadSizeLimit(instISAC);
- }
- instISAC->bottleneck = bottleneckBPS;
- return 0;
-}
-
-
-/****************************************************************************
- * WebRtcIsac_ControlBwe(...)
- *
- * This function sets the initial values of bottleneck and frame-size if
- * iSAC is used in channel-adaptive mode. Through this API, users can
- * enforce a frame-size for all values of bottleneck. Then iSAC will not
- * automatically change the frame-size.
- *
- *
- * Input:
- * - ISAC_main_inst : ISAC instance.
- * - rateBPS : initial value of bottleneck in bits/second
- * 10000 <= rateBPS <= 32000 is accepted
- * For default bottleneck set rateBPS = 0
- * - frameSizeMs : number of milliseconds per frame (30 or 60)
- * - enforceFrameSize : 1 to enforce the given frame-size through out
- * the adaptation process, 0 to let iSAC change
- * the frame-size if required.
- *
- * Return value : 0 - ok
- * -1 - Error
- */
-WebRtc_Word16 WebRtcIsac_ControlBwe(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word32 bottleneckBPS,
- WebRtc_Word16 frameSizeMs,
- WebRtc_Word16 enforceFrameSize)
-{
- ISACMainStruct *instISAC;
- enum ISACBandwidth bandwidth;
-
- /* Typecast pointer to real structure */
- instISAC = (ISACMainStruct *)ISAC_main_inst;
-
- /* check if encoder initiated */
- if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
- BIT_MASK_ENC_INIT)
- {
- instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
- return -1;
- }
-
- /* Check that we are in channel-adaptive mode, otherwise, return (-1) */
- if(instISAC->codingMode != 0)
- {
- instISAC->errorCode = ISAC_MODE_MISMATCH;
- return -1;
- }
- if((frameSizeMs != 30) &&
- (instISAC->encoderSamplingRateKHz == kIsacSuperWideband))
- {
- return -1;
- }
-
- /* Set struct variable if enforceFrameSize is set. ISAC will then */
- /* keep the chosen frame size. */
- if((enforceFrameSize != 0) /*||
- (instISAC->samplingRateKHz == kIsacSuperWideband)*/)
- {
- instISAC->instLB.ISACencLB_obj.enforceFrameSize = 1;
- }
- else
- {
- instISAC->instLB.ISACencLB_obj.enforceFrameSize = 0;
- }
-
- /* Set initial rate, if value between 10000 and 32000, */
- /* if rateBPS is 0, keep the default initial bottleneck value (15000) */
- if(bottleneckBPS != 0)
- {
- double rateLB;
- double rateUB;
- if(WebRtcIsac_RateAllocation(bottleneckBPS, &rateLB, &rateUB, &bandwidth) < 0)
- {
- return -1;
- }
- instISAC->bwestimator_obj.send_bw_avg = (float)bottleneckBPS;
- instISAC->bandwidthKHz = bandwidth;
- }
-
- /* Set initial frameSize. If enforceFrameSize is set the frame size will
- not change */
- if(frameSizeMs != 0)
- {
- if((frameSizeMs == 30) || (frameSizeMs == 60))
- {
- instISAC->instLB.ISACencLB_obj.new_framelength = (FS/1000) *
- frameSizeMs;
- //instISAC->bwestimator_obj.rec_header_rate = ((float)HEADER_SIZE *
- // 8.0f * 1000.0f / (float)frameSizeMs);
- }
- else
- {
- instISAC->errorCode = ISAC_DISALLOWED_FRAME_LENGTH;
- return -1;
- }
- }
- return 0;
-}
-
-
-/****************************************************************************
- * WebRtcIsac_GetDownLinkBwIndex(...)
- *
- * This function returns index representing the Bandwidth estimate from
- * other side to this side.
- *
- * Input:
- * - ISAC_main_inst : iSAC struct
- *
- * Output:
- * - bweIndex : Bandwidth estimate to transmit to other side.
- *
- */
-WebRtc_Word16 WebRtcIsac_GetDownLinkBwIndex(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word16* bweIndex,
- WebRtc_Word16* jitterInfo)
-{
- ISACMainStruct *instISAC;
-
- /* typecast pointer to real structure */
- instISAC = (ISACMainStruct*)ISAC_main_inst;
-
- /* check if encoder initiated */
- if((instISAC->initFlag & BIT_MASK_DEC_INIT) !=
- BIT_MASK_DEC_INIT)
- {
- instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
- return -1;
- }
-
- /* Call function to get Bandwidth Estimate */
- WebRtcIsac_GetDownlinkBwJitIndexImpl(&(instISAC->bwestimator_obj),
- bweIndex, jitterInfo, instISAC->decoderSamplingRateKHz);
- return 0;
-}
-
-
-/****************************************************************************
- * WebRtcIsac_UpdateUplinkBw(...)
- *
- * This function takes an index representing the Bandwidth estimate from
- * this side to other side and updates BWE.
- *
- * Input:
- * - ISAC_main_inst : iSAC struct
- * - rateIndex : Bandwidth estimate from other side.
- *
- * Return value : 0 - ok
- * -1 - index out of range
- */
-WebRtc_Word16 WebRtcIsac_UpdateUplinkBw(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word16 bweIndex)
-{
- ISACMainStruct *instISAC;
- WebRtc_Word16 returnVal;
-
- /* typecast pointer to real structure */
- instISAC = (ISACMainStruct *)ISAC_main_inst;
-
- /* check if encoder initiated */
- if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
- BIT_MASK_ENC_INIT)
- {
- instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
- return -1;
- }
-
- /* Call function to get Bandwidth Estimate */
- returnVal = WebRtcIsac_UpdateUplinkBwImpl(
- &(instISAC->bwestimator_obj), bweIndex,
- instISAC->encoderSamplingRateKHz);
-
- if(returnVal < 0)
- {
- instISAC->errorCode = -returnVal;
- return -1;
- }
- else
- {
- return 0;
- }
-}
-
-
-/****************************************************************************
- * WebRtcIsac_ReadBwIndex(...)
- *
- * This function returns the index of the Bandwidth estimate from the
- * bitstream.
- *
- * Input:
- * - encoded : Encoded bitstream
- *
- * Output:
- * - frameLength : Length of frame in packet (in samples)
- * - bweIndex : Bandwidth estimate in bitstream
- *
- */
-WebRtc_Word16 WebRtcIsac_ReadBwIndex(
- const WebRtc_Word16* encoded,
- WebRtc_Word16* bweIndex)
-{
- Bitstr streamdata;
-#ifndef WEBRTC_BIG_ENDIAN
- int k;
-#endif
- WebRtc_Word16 err;
-
- streamdata.W_upper = 0xFFFFFFFF;
- streamdata.streamval = 0;
- streamdata.stream_index = 0;
-
-#ifndef WEBRTC_BIG_ENDIAN
- for(k = 0; k < 10; k++)
- {
- streamdata.stream[k] = (WebRtc_UWord8) ((encoded[k>>1] >>
- ((k&1) << 3)) & 0xFF);
- }
-#else
- memcpy(streamdata.stream, encoded, 10);
-#endif
-
- /* decode frame length */
- err = WebRtcIsac_DecodeFrameLen(&streamdata, bweIndex);
- if(err < 0)
- {
- return err;
- }
-
- /* decode BW estimation */
- err = WebRtcIsac_DecodeSendBW(&streamdata, bweIndex);
- if(err < 0)
- {
- return err;
- }
-
- return 0;
-}
-
-
-/****************************************************************************
- * WebRtcIsac_ReadFrameLen(...)
- *
- * This function returns the length of the frame represented in the packet.
- *
- * Input:
- * - encoded : Encoded bitstream
- *
- * Output:
- * - frameLength : Length of frame in packet (in samples)
- *
- */
-WebRtc_Word16 WebRtcIsac_ReadFrameLen(
- ISACStruct* ISAC_main_inst,
- const WebRtc_Word16* encoded,
- WebRtc_Word16* frameLength)
-{
- Bitstr streamdata;
-#ifndef WEBRTC_BIG_ENDIAN
- int k;
-#endif
- WebRtc_Word16 err;
- ISACMainStruct* instISAC;
-
- streamdata.W_upper = 0xFFFFFFFF;
- streamdata.streamval = 0;
- streamdata.stream_index = 0;
-
-#ifndef WEBRTC_BIG_ENDIAN
- for (k=0; k<10; k++) {
- streamdata.stream[k] = (WebRtc_UWord8) ((encoded[k>>1] >>
- ((k&1) << 3)) & 0xFF);
- }
-#else
- memcpy(streamdata.stream, encoded, 10);
-#endif
-
- /* decode frame length */
- err = WebRtcIsac_DecodeFrameLen(&streamdata, frameLength);
- if(err < 0) {
- return -1;
- }
- instISAC = (ISACMainStruct*)ISAC_main_inst;
-
- if(instISAC->decoderSamplingRateKHz == kIsacSuperWideband)
- {
- // the decoded frame length indicates the number of samples in
- // lower-band in this case, multiply by 2 to get the total number
- // of samples.
- *frameLength <<= 1;
- }
-
- return 0;
-}
-
-
-/*******************************************************************************
- * WebRtcIsac_GetNewFrameLen(...)
- *
- * returns the frame lenght (in samples) of the next packet. In the case of
- * channel-adaptive mode, iSAC decides on its frame lenght based on the
- * estimated bottleneck this allows a user to prepare for the next packet
- * (at the encoder).
- *
- * The primary usage is in CE to make the iSAC works in channel-adaptive mode
- *
- * Input:
- * - ISAC_main_inst : iSAC struct
- *
- * Return Value : frame lenght in samples
- *
- */
-WebRtc_Word16 WebRtcIsac_GetNewFrameLen(
- ISACStruct *ISAC_main_inst)
-{
- ISACMainStruct *instISAC;
-
- /* typecast pointer to real structure */
- instISAC = (ISACMainStruct *)ISAC_main_inst;
-
- /* Return new frame length */
- if(instISAC->encoderSamplingRateKHz == kIsacWideband)
- {
- return (instISAC->instLB.ISACencLB_obj.new_framelength);
- }
- else
- {
- return ((instISAC->instLB.ISACencLB_obj.new_framelength) << 1);
- }
-}
-
-
-/****************************************************************************
- * WebRtcIsac_GetErrorCode(...)
- *
- * This function can be used to check the error code of an iSAC instance.
- * When a function returns -1 a error code will be set for that instance.
- * The function below extract the code of the last error that occured in
- * the specified instance.
- *
- * Input:
- * - ISAC_main_inst : ISAC instance
- *
- * Return value : Error code
- */
-WebRtc_Word16 WebRtcIsac_GetErrorCode(
- ISACStruct *ISAC_main_inst)
-{
- ISACMainStruct *instISAC;
- /* typecast pointer to real structure */
- instISAC = (ISACMainStruct *)ISAC_main_inst;
-
- return (instISAC->errorCode);
-}
-
-
-/****************************************************************************
- * WebRtcIsac_GetUplinkBw(...)
- *
- * This function outputs the target bottleneck of the codec. In
- * channel-adaptive mode, the target bottleneck is specified through in-band
- * signalling retreived by bandwidth estimator.
- * In channel-independent, also called instantaneous mode, the target
- * bottleneck is provided to the encoder by calling xxx_control(...) (if
- * xxx_control is never called the default values is).
- * Note that the output is the iSAC internal operating bottleneck whch might
- * differ slightly from the one provided through xxx_control().
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- *
- * Output:
- * - *bottleneck : bottleneck in bits/sec
- *
- * Return value : -1 if error happens
- * 0 bit-rates computed correctly.
- */
-WebRtc_Word16 WebRtcIsac_GetUplinkBw(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word32* bottleneck)
-{
- ISACMainStruct* instISAC = (ISACMainStruct *)ISAC_main_inst;
-
- if(instISAC->codingMode == 0)
- {
- // we are in adaptive mode then get the bottleneck from BWE
- *bottleneck = (WebRtc_Word32)instISAC->bwestimator_obj.send_bw_avg;
- }
- else
- {
- *bottleneck = instISAC->bottleneck;
- }
-
- if((*bottleneck > 32000) && (*bottleneck < 38000))
- {
- *bottleneck = 32000;
- }
- else if((*bottleneck > 45000) && (*bottleneck < 50000))
- {
- *bottleneck = 45000;
- }
- else if(*bottleneck > 56000)
- {
- *bottleneck = 56000;
- }
-
- return 0;
-}
-
-
-/******************************************************************************
- * WebRtcIsac_SetMaxPayloadSize(...)
- *
- * This function sets a limit for the maximum payload size of iSAC. The same
- * value is used both for 30 and 60 ms packets. If the encoder sampling rate
- * is 16 kHz the maximum payload size is between 120 and 400 bytes. If the
- * encoder sampling rate is 32 kHz the maximum payload size is between 120
- * and 600 bytes.
- *
- * ---------------
- * IMPORTANT NOTES
- * ---------------
- * The size of a packet is limited to the minimum of 'max-payload-size' and
- * 'max-rate.' For instance, let's assume the max-payload-size is set to
- * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps
- * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms
- * frame-size. Then a packet with a frame-size of 30 ms is limited to 150,
- * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to
- * 170 bytes, i.e. min(170, 300).
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- * - maxPayloadBytes : maximum size of the payload in bytes
- * valid values are between 100 and 400 bytes
- * if encoder sampling rate is 16 kHz. For
- * 32 kHz encoder sampling rate valid values
- * are between 100 and 600 bytes.
- *
- * Return value : 0 if successful
- * -1 if error happens
- */
-WebRtc_Word16 WebRtcIsac_SetMaxPayloadSize(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word16 maxPayloadBytes)
-{
- ISACMainStruct *instISAC;
- WebRtc_Word16 status = 0;
-
- /* typecast pointer to real structure */
- instISAC = (ISACMainStruct *)ISAC_main_inst;
-
- /* check if encoder initiated */
- if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
- BIT_MASK_ENC_INIT)
- {
- instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
- return -1;
- }
-
- if(instISAC->encoderSamplingRateKHz == kIsacSuperWideband)
- {
- // sanity check
- if(maxPayloadBytes < 120)
- {
- // maxRate is out of valid range
- // set to the acceptable value and return -1.
- maxPayloadBytes = 120;
- status = -1;
- }
-
- /* sanity check */
- if(maxPayloadBytes > STREAM_SIZE_MAX)
- {
- // maxRate is out of valid range
- // set to the acceptable value and return -1.
- maxPayloadBytes = STREAM_SIZE_MAX;
- status = -1;
- }
- }
- else
- {
- if(maxPayloadBytes < 120)
- {
- // max payload-size is out of valid range
- // set to the acceptable value and return -1.
- maxPayloadBytes = 120;
- status = -1;
- }
- if(maxPayloadBytes > STREAM_SIZE_MAX_60)
- {
- // max payload-size is out of valid range
- // set to the acceptable value and return -1.
- maxPayloadBytes = STREAM_SIZE_MAX_60;
- status = -1;
- }
- }
- instISAC->maxPayloadSizeBytes = maxPayloadBytes;
- UpdatePayloadSizeLimit(instISAC);
- return status;
-}
-
-
-/******************************************************************************
- * WebRtcIsac_SetMaxRate(...)
- *
- * This function sets the maximum rate which the codec may not exceed for
- * any signal packet. The maximum rate is defined and payload-size per
- * frame-size in bits per second.
- *
- * The codec has a maximum rate of 53400 bits per second (200 bytes per 30
- * ms) if the encoder sampling rate is 16kHz, and 160 kbps (600 bytes/30 ms)
- * if the encoder sampling rate is 32 kHz.
- *
- * It is possible to set a maximum rate between 32000 and 53400 bits/sec
- * in wideband mode, and 32000 to 160000 bits/sec in super-wideband mode.
- *
- * ---------------
- * IMPORTANT NOTES
- * ---------------
- * The size of a packet is limited to the minimum of 'max-payload-size' and
- * 'max-rate.' For instance, let's assume the max-payload-size is set to
- * 170 bytes, and max-rate is set to 40 kbps. Note that a limit of 40 kbps
- * translates to 150 bytes for 30ms frame-size & 300 bytes for 60ms
- * frame-size. Then a packet with a frame-size of 30 ms is limited to 150,
- * i.e. min(170, 150), and a packet with 60 ms frame-size is limited to
- * 170 bytes, min(170, 300).
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- * - maxRate : maximum rate in bits per second,
- * valid values are 32000 to 53400 bits/sec in
- * wideband mode, and 32000 to 160000 bits/sec in
- * super-wideband mode.
- *
- * Return value : 0 if successful
- * -1 if error happens
- */
-WebRtc_Word16 WebRtcIsac_SetMaxRate(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word32 maxRate)
-{
- ISACMainStruct *instISAC;
- WebRtc_Word16 maxRateInBytesPer30Ms;
- WebRtc_Word16 status = 0;
-
- /* typecast pointer to real structure */
- instISAC = (ISACMainStruct *)ISAC_main_inst;
-
- /* check if encoder initiated */
- if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
- BIT_MASK_ENC_INIT)
- {
- instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
- return -1;
- }
- /*
- Calculate maximum number of bytes per 30 msec packets for the
- given maximum rate. Multiply with 30/1000 to get number of
- bits per 30 ms, divide by 8 to get number of bytes per 30 ms:
- maxRateInBytes = floor((maxRate * 30/1000) / 8);
- */
- maxRateInBytesPer30Ms = (WebRtc_Word16)(maxRate*3/800);
-
- if(instISAC->encoderSamplingRateKHz == kIsacWideband)
- {
- if(maxRate < 32000)
- {
- // max rate is out of valid range
- // set to the acceptable value and return -1.
- maxRateInBytesPer30Ms = 120;
- status = -1;
- }
-
- if(maxRate > 53400)
- {
- // max rate is out of valid range
- // set to the acceptable value and return -1.
- maxRateInBytesPer30Ms = 200;
- status = -1;
- }
- }
- else
- {
- if(maxRateInBytesPer30Ms < 120)
- {
- // maxRate is out of valid range
- // set to the acceptable value and return -1.
- maxRateInBytesPer30Ms = 120;
- status = -1;
- }
-
- if(maxRateInBytesPer30Ms > STREAM_SIZE_MAX)
- {
- // maxRate is out of valid range
- // set to the acceptable value and return -1.
- maxRateInBytesPer30Ms = STREAM_SIZE_MAX;
- status = -1;
- }
- }
- instISAC->maxRateBytesPer30Ms = maxRateInBytesPer30Ms;
- UpdatePayloadSizeLimit(instISAC);
- return status;
-}
-
-
-/****************************************************************************
- * WebRtcIsac_GetRedPayload(...)
- *
- * Populates "encoded" with the redundant payload of the recently encoded
- * frame. This function has to be called once that WebRtcIsac_Encode(...)
- * returns a positive value. Regardless of the frame-size this function will
- * be called only once after encoding is completed. The bit-stream is
- * targeted for 16000 bit/sec.
- *
- * Input:
- * - ISAC_main_inst : iSAC struct
- *
- * Output:
- * - encoded : the encoded data vector
- *
- *
- * Return value : >0 - Length (in bytes) of coded data
- * : -1 - Error
- *
- *
- */
-WebRtc_Word16 WebRtcIsac_GetRedPayload(
- ISACStruct* ISAC_main_inst,
- WebRtc_Word16* encoded)
-{
- ISACMainStruct* instISAC;
- Bitstr iSACBitStreamInst;
- WebRtc_Word16 streamLenLB;
- WebRtc_Word16 streamLenUB;
- WebRtc_Word16 streamLen;
- WebRtc_Word16 totalLenUB;
- WebRtc_UWord8* ptrEncodedUW8 = (WebRtc_UWord8*)encoded;
-#ifndef WEBRTC_BIG_ENDIAN
- int k;
-#endif
-
- /* typecast pointer to real structure */
- instISAC = (ISACMainStruct*)ISAC_main_inst;
-
-
- if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
- BIT_MASK_ENC_INIT)
- {
- instISAC->errorCode = ISAC_ENCODER_NOT_INITIATED;
- }
-
-
- iSACBitStreamInst.W_upper = 0xFFFFFFFF;
- iSACBitStreamInst.streamval = 0;
- iSACBitStreamInst.stream_index = 0;
-
-
- streamLenLB = WebRtcIsac_EncodeStoredDataLb(
- &instISAC->instLB.ISACencLB_obj.SaveEnc_obj,
- &iSACBitStreamInst,
- instISAC->instLB.ISACencLB_obj.lastBWIdx,
- RCU_TRANSCODING_SCALE);
-
- if(streamLenLB < 0)
- {
- return -1;
- }
-
- /* convert from bytes to WebRtc_Word16 */
- memcpy(ptrEncodedUW8, iSACBitStreamInst.stream, streamLenLB);
-
- streamLen = streamLenLB;
-
- if(instISAC->bandwidthKHz == isac8kHz)
- {
- return streamLenLB;
- }
-
- streamLenUB = WebRtcIsac_GetRedPayloadUb(
- &instISAC->instUB.ISACencUB_obj.SaveEnc_obj,
- &iSACBitStreamInst, instISAC->bandwidthKHz);
-
- if(streamLenUB < 0)
- {
- // an error has happened but this is not the error due to a
- // bit-stream larger than the limit
- return -1;
- }
-
- // We have one byte to write the total length of the upper band
- // the length include the bitstream length, check-sum and the
- // single byte where the length is written to. This is according to
- // iSAC wideband and how the "garbage" is dealt.
- totalLenUB = streamLenUB + 1 + LEN_CHECK_SUM_WORD8;
- if(totalLenUB > 255)
- {
- streamLenUB = 0;
- }
-
- // Generate CRC if required.
- if((instISAC->bandwidthKHz != isac8kHz) &&
- (streamLenUB > 0))
- {
- WebRtc_UWord32 crc;
- streamLen += totalLenUB;
- ptrEncodedUW8[streamLenLB] = (WebRtc_UWord8)totalLenUB;
- memcpy(&ptrEncodedUW8[streamLenLB+1], iSACBitStreamInst.stream, streamLenUB);
-
- WebRtcIsac_GetCrc((WebRtc_Word16*)(&(ptrEncodedUW8[streamLenLB + 1])),
- streamLenUB, &crc);
-#ifndef WEBRTC_BIG_ENDIAN
- for(k = 0; k < LEN_CHECK_SUM_WORD8; k++)
- {
- ptrEncodedUW8[streamLen - LEN_CHECK_SUM_WORD8 + k] =
- (WebRtc_UWord8)((crc >> (24 - k * 8)) & 0xFF);
- }
-#else
- memcpy(&ptrEncodedUW8[streamLenLB + streamLenUB + 1], &crc,
- LEN_CHECK_SUM_WORD8);
-#endif
- }
-
-
- return streamLen;
-}
-
-
-/****************************************************************************
- * WebRtcIsac_version(...)
- *
- * This function returns the version number.
- *
- * Output:
- * - version : Pointer to character string
- *
- */
-void WebRtcIsac_version(char *version)
-{
- strcpy(version, "4.3.0");
-}
-
-
-/******************************************************************************
- * WebRtcIsac_SetEncSampRate()
- * Set the sampling rate of the encoder. Initialization of the encoder WILL
- * NOT overwrite the sampling rate of the encoder. The default value is 16 kHz
- * which is set when the instance is created. The encoding-mode and the
- * bottleneck remain unchanged by this call, however, the maximum rate and
- * maximum payload-size will reset to their default value.
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- * - sampRate : enumerator specifying the sampling rate.
- *
- * Return value : 0 if successful
- * -1 if failed.
- */
-WebRtc_Word16 WebRtcIsac_SetEncSampRate(
- ISACStruct* ISAC_main_inst,
- enum IsacSamplingRate sampRate)
-{
- ISACMainStruct* instISAC;
-
- instISAC = (ISACMainStruct*)ISAC_main_inst;
-
- if((sampRate != kIsacWideband) &&
- (sampRate != kIsacSuperWideband))
- {
- // Sampling Frequency is not supported
- instISAC->errorCode = ISAC_UNSUPPORTED_SAMPLING_FREQUENCY;
- return -1;
- }
- else if((instISAC->initFlag & BIT_MASK_ENC_INIT) !=
- BIT_MASK_ENC_INIT)
- {
- if(sampRate == kIsacWideband)
- {
- instISAC->bandwidthKHz = isac8kHz;
- }
- else
- {
- instISAC->bandwidthKHz = isac16kHz;
- }
- instISAC->encoderSamplingRateKHz = sampRate;
- return 0;
- }
- else
- {
- ISACUBStruct* instUB = &(instISAC->instUB);
- ISACLBStruct* instLB = &(instISAC->instLB);
- double bottleneckLB;
- double bottleneckUB;
- WebRtc_Word32 bottleneck = instISAC->bottleneck;
- WebRtc_Word16 codingMode = instISAC->codingMode;
- WebRtc_Word16 frameSizeMs = instLB->ISACencLB_obj.new_framelength / (FS / 1000);
-
- if((sampRate == kIsacWideband) &&
- (instISAC->encoderSamplingRateKHz == kIsacSuperWideband))
- {
- // changing from super-wideband to wideband.
- // we don't need to re-initialize the encoder of the
- // lower-band.
- instISAC->bandwidthKHz = isac8kHz;
- if(codingMode == 1)
- {
- ControlLb(instLB,
- (bottleneck > 32000)? 32000:bottleneck, FRAMESIZE);
- }
- instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX_60;
- instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX_30;
- }
- else if((sampRate == kIsacSuperWideband) &&
- (instISAC->encoderSamplingRateKHz == kIsacWideband))
- {
- if(codingMode == 1)
- {
- WebRtcIsac_RateAllocation(bottleneck, &bottleneckLB, &bottleneckUB,
- &(instISAC->bandwidthKHz));
- }
-
- instISAC->bandwidthKHz = isac16kHz;
- instISAC->maxPayloadSizeBytes = STREAM_SIZE_MAX;
- instISAC->maxRateBytesPer30Ms = STREAM_SIZE_MAX;
-
- EncoderInitLb(instLB, codingMode, sampRate);
- EncoderInitUb(instUB, instISAC->bandwidthKHz);
-
- memset(instISAC->analysisFBState1, 0,
- FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
- memset(instISAC->analysisFBState2, 0,
- FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
-
- if(codingMode == 1)
- {
- instISAC->bottleneck = bottleneck;
- ControlLb(instLB, bottleneckLB,
- (instISAC->bandwidthKHz == isac8kHz)? frameSizeMs:FRAMESIZE);
- if(instISAC->bandwidthKHz > isac8kHz)
- {
- ControlUb(instUB, bottleneckUB);
- }
- }
- else
- {
- instLB->ISACencLB_obj.enforceFrameSize = 0;
- instLB->ISACencLB_obj.new_framelength = FRAMESAMPLES;
- }
- }
- instISAC->encoderSamplingRateKHz = sampRate;
- return 0;
- }
-}
-
-
-/******************************************************************************
- * WebRtcIsac_SetDecSampRate()
- * Set the sampling rate of the decoder. Initialization of the decoder WILL
- * NOT overwrite the sampling rate of the encoder. The default value is 16 kHz
- * which is set when the instance is created.
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- * - sampRate : enumerator specifying the sampling rate.
- *
- * Return value : 0 if successful
- * -1 if failed.
- */
-WebRtc_Word16 WebRtcIsac_SetDecSampRate(
- ISACStruct* ISAC_main_inst,
- enum IsacSamplingRate sampRate)
-{
- ISACMainStruct* instISAC;
-
- instISAC = (ISACMainStruct*)ISAC_main_inst;
-
- if((sampRate != kIsacWideband) &&
- (sampRate != kIsacSuperWideband))
- {
- // Sampling Frequency is not supported
- instISAC->errorCode = ISAC_UNSUPPORTED_SAMPLING_FREQUENCY;
- return -1;
- }
- else
- {
- if((instISAC->decoderSamplingRateKHz == kIsacWideband) &&
- (sampRate == kIsacSuperWideband))
- {
- // switching from wideband to super-wideband at the decoder
- // we need to reset the filter-bank and initialize
- // upper-band decoder.
- memset(instISAC->synthesisFBState1, 0,
- FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
- memset(instISAC->synthesisFBState2, 0,
- FB_STATE_SIZE_WORD32 * sizeof(WebRtc_Word32));
-
- if(DecoderInitUb(&(instISAC->instUB)) < 0)
- {
- return -1;
- }
- }
- instISAC->decoderSamplingRateKHz = sampRate;
- return 0;
- }
-}
-
-
-/******************************************************************************
- * WebRtcIsac_EncSampRate()
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- *
- * Return value : enumerator representing sampling frequency
- * associated with the encoder, the input audio
- * is expected to be sampled at this rate.
- *
- */
-enum IsacSamplingRate WebRtcIsac_EncSampRate(
- ISACStruct* ISAC_main_inst)
-{
- ISACMainStruct* instISAC;
-
- instISAC = (ISACMainStruct*)ISAC_main_inst;
-
- return instISAC->encoderSamplingRateKHz;
-}
-
-
-/******************************************************************************
- * WebRtcIsac_DecSampRate()
- * Return the sampling rate of the decoded audio.
- *
- * Input:
- * - ISAC_main_inst : iSAC instance
- *
- * Return value : enumerator representing sampling frequency
- * associated with the decoder, i.e. the
- * sampling rate of the decoded audio.
- *
- */
-enum IsacSamplingRate WebRtcIsac_DecSampRate(
- ISACStruct* ISAC_main_inst)
-{
- ISACMainStruct* instISAC;
-
- instISAC = (ISACMainStruct*)ISAC_main_inst;
-
- return instISAC->decoderSamplingRateKHz;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/isac.gypi b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/isac.gypi
deleted file mode 100644
index d30be55..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/isac.gypi
+++ /dev/null
@@ -1,91 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'iSAC',
- 'type': '<(library)',
- 'dependencies': [
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- ],
- 'include_dirs': [
- '../interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- ],
- },
- 'sources': [
- '../interface/isac.h',
- 'arith_routines.c',
- 'arith_routines_hist.c',
- 'arith_routines_logist.c',
- 'bandwidth_estimator.c',
- 'crc.c',
- 'decode.c',
- 'decode_bwe.c',
- 'encode.c',
- 'encode_lpc_swb.c',
- 'entropy_coding.c',
- 'fft.c',
- 'filter_functions.c',
- 'filterbank_tables.c',
- 'intialize.c',
- 'isac.c',
- 'filterbanks.c',
- 'pitch_lag_tables.c',
- 'lattice.c',
- 'lpc_gain_swb_tables.c',
- 'lpc_analysis.c',
- 'lpc_shape_swb12_tables.c',
- 'lpc_shape_swb16_tables.c',
- 'lpc_tables.c',
- 'pitch_estimator.c',
- 'pitch_filter.c',
- 'pitch_gain_tables.c',
- 'spectrum_ar_model_tables.c',
- 'transform.c',
- 'arith_routines.h',
- 'bandwidth_estimator.h',
- 'codec.h',
- 'crc.h',
- 'encode_lpc_swb.h',
- 'entropy_coding.h',
- 'fft.h',
- 'filterbank_tables.h',
- 'lpc_gain_swb_tables.h',
- 'lpc_analysis.h',
- 'lpc_shape_swb12_tables.h',
- 'lpc_shape_swb16_tables.h',
- 'lpc_tables.h',
- 'pitch_estimator.h',
- 'pitch_gain_tables.h',
- 'pitch_lag_tables.h',
- 'settings.h',
- 'spectrum_ar_model_tables.h',
- 'structs.h',
- 'os_specific_inline.h',
- ],
- 'conditions': [
- ['OS!="win"', {
- 'defines': [
- 'WEBRTC_LINUX',
- ],
- }],
- ],
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lattice.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lattice.c
deleted file mode 100644
index a46135a..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lattice.c
+++ /dev/null
@@ -1,217 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * lattice.c
- *
- * contains the normalized lattice filter routines (MA and AR) for iSAC codec
- *
- */
-#include "settings.h"
-#include "codec.h"
-
-#include <math.h>
-#include <memory.h>
-#ifdef WEBRTC_ANDROID
-#include <stdlib.h>
-#endif
-
-/* filter the signal using normalized lattice filter */
-/* MA filter */
-void WebRtcIsac_NormLatticeFilterMa(int orderCoef,
- float *stateF,
- float *stateG,
- float *lat_in,
- double *filtcoeflo,
- double *lat_out)
-{
- int n,k,i,u,temp1;
- int ord_1 = orderCoef+1;
- float sth[MAX_AR_MODEL_ORDER];
- float cth[MAX_AR_MODEL_ORDER];
- float inv_cth[MAX_AR_MODEL_ORDER];
- double a[MAX_AR_MODEL_ORDER+1];
- float f[MAX_AR_MODEL_ORDER+1][HALF_SUBFRAMELEN], g[MAX_AR_MODEL_ORDER+1][HALF_SUBFRAMELEN];
- float gain1;
-
- for (u=0;u<SUBFRAMES;u++)
- {
- /* set the Direct Form coefficients */
- temp1 = u*ord_1;
- a[0] = 1;
- memcpy(a+1, filtcoeflo+temp1+1, sizeof(double) * (ord_1-1));
-
- /* compute lattice filter coefficients */
- WebRtcIsac_Dir2Lat(a,orderCoef,sth,cth);
-
- /* compute the gain */
- gain1 = (float)filtcoeflo[temp1];
- for (k=0;k<orderCoef;k++)
- {
- gain1 *= cth[k];
- inv_cth[k] = 1/cth[k];
- }
-
- /* normalized lattice filter */
- /*****************************/
-
- /* initial conditions */
- for (i=0;i<HALF_SUBFRAMELEN;i++)
- {
- f[0][i] = lat_in[i + u * HALF_SUBFRAMELEN];
- g[0][i] = lat_in[i + u * HALF_SUBFRAMELEN];
- }
-
- /* get the state of f&g for the first input, for all orders */
- for (i=1;i<ord_1;i++)
- {
- f[i][0] = inv_cth[i-1]*(f[i-1][0] + sth[i-1]*stateG[i-1]);
- g[i][0] = cth[i-1]*stateG[i-1] + sth[i-1]* f[i][0];
- }
-
- /* filtering */
- for(k=0;k<orderCoef;k++)
- {
- for(n=0;n<(HALF_SUBFRAMELEN-1);n++)
- {
- f[k+1][n+1] = inv_cth[k]*(f[k][n+1] + sth[k]*g[k][n]);
- g[k+1][n+1] = cth[k]*g[k][n] + sth[k]* f[k+1][n+1];
- }
- }
-
- for(n=0;n<HALF_SUBFRAMELEN;n++)
- {
- lat_out[n + u * HALF_SUBFRAMELEN] = gain1 * f[orderCoef][n];
- }
-
- /* save the states */
- for (i=0;i<ord_1;i++)
- {
- stateF[i] = f[i][HALF_SUBFRAMELEN-1];
- stateG[i] = g[i][HALF_SUBFRAMELEN-1];
- }
- /* process next frame */
- }
-
- return;
-}
-
-
-/*///////////////////AR filter ///////////////////////////////*/
-/* filter the signal using normalized lattice filter */
-void WebRtcIsac_NormLatticeFilterAr(int orderCoef,
- float *stateF,
- float *stateG,
- double *lat_in,
- double *lo_filt_coef,
- float *lat_out)
-{
- int n,k,i,u,temp1;
- int ord_1 = orderCoef+1;
- float sth[MAX_AR_MODEL_ORDER];
- float cth[MAX_AR_MODEL_ORDER];
- double a[MAX_AR_MODEL_ORDER+1];
- float ARf[MAX_AR_MODEL_ORDER+1][HALF_SUBFRAMELEN], ARg[MAX_AR_MODEL_ORDER+1][HALF_SUBFRAMELEN];
- float gain1,inv_gain1;
-
- for (u=0;u<SUBFRAMES;u++)
- {
- /* set the denominator and numerator of the Direct Form */
- temp1 = u*ord_1;
- a[0] = 1;
-
- memcpy(a+1, lo_filt_coef+temp1+1, sizeof(double) * (ord_1-1));
-
- WebRtcIsac_Dir2Lat(a,orderCoef,sth,cth);
-
- gain1 = (float)lo_filt_coef[temp1];
- for (k=0;k<orderCoef;k++)
- {
- gain1 = cth[k]*gain1;
- }
-
- /* initial conditions */
- inv_gain1 = 1/gain1;
- for (i=0;i<HALF_SUBFRAMELEN;i++)
- {
- ARf[orderCoef][i] = (float)lat_in[i + u * HALF_SUBFRAMELEN]*inv_gain1;
- }
-
-
- for (i=orderCoef-1;i>=0;i--) //get the state of f&g for the first input, for all orders
- {
- ARf[i][0] = cth[i]*ARf[i+1][0] - sth[i]*stateG[i];
- ARg[i+1][0] = sth[i]*ARf[i+1][0] + cth[i]* stateG[i];
- }
- ARg[0][0] = ARf[0][0];
-
- for(n=0;n<(HALF_SUBFRAMELEN-1);n++)
- {
- for(k=orderCoef-1;k>=0;k--)
- {
- ARf[k][n+1] = cth[k]*ARf[k+1][n+1] - sth[k]*ARg[k][n];
- ARg[k+1][n+1] = sth[k]*ARf[k+1][n+1] + cth[k]* ARg[k][n];
- }
- ARg[0][n+1] = ARf[0][n+1];
- }
-
- memcpy(lat_out+u * HALF_SUBFRAMELEN, &(ARf[0][0]), sizeof(float) * HALF_SUBFRAMELEN);
-
- /* cannot use memcpy in the following */
- for (i=0;i<ord_1;i++)
- {
- stateF[i] = ARf[i][HALF_SUBFRAMELEN-1];
- stateG[i] = ARg[i][HALF_SUBFRAMELEN-1];
- }
-
- }
-
- return;
-}
-
-
-/* compute the reflection coefficients using the step-down procedure*/
-/* converts the direct form parameters to lattice form.*/
-/* a and b are vectors which contain the direct form coefficients,
- according to
- A(z) = a(1) + a(2)*z + a(3)*z^2 + ... + a(M+1)*z^M
- B(z) = b(1) + b(2)*z + b(3)*z^2 + ... + b(M+1)*z^M
-*/
-
-void WebRtcIsac_Dir2Lat(double *a,
- int orderCoef,
- float *sth,
- float *cth)
-{
- int m, k;
- float tmp[MAX_AR_MODEL_ORDER];
- float tmp_inv, cth2;
-
- sth[orderCoef-1] = (float)a[orderCoef];
- cth2 = 1.0f - sth[orderCoef-1] * sth[orderCoef-1];
- cth[orderCoef-1] = (float)sqrt(cth2);
- for (m=orderCoef-1; m>0; m--)
- {
- tmp_inv = 1.0f / cth2;
- for (k=1; k<=m; k++)
- {
- tmp[k] = ((float)a[k] - sth[m] * (float)a[m-k+1]) * tmp_inv;
- }
-
- for (k=1; k<m; k++)
- {
- a[k] = tmp[k];
- }
-
- sth[m-1] = tmp[m];
- cth2 = 1 - sth[m-1] * sth[m-1];
- cth[m-1] = (float)sqrt(cth2);
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_analysis.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_analysis.c
deleted file mode 100644
index 854b2d7..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_analysis.c
+++ /dev/null
@@ -1,535 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "lpc_analysis.h"
-#include "settings.h"
-#include "codec.h"
-#include "entropy_coding.h"
-
-#include <math.h>
-#include <string.h>
-
-#define LEVINSON_EPS 1.0e-10
-
-
-/* window */
-/* Matlab generation code:
- * t = (1:256)/257; r = 1-(1-t).^.45; w = sin(r*pi).^3; w = w/sum(w); plot((1:256)/8, w); grid;
- * for k=1:16, fprintf(1, '%.8f, ', w(k*16 + (-15:0))); fprintf(1, '\n'); end
- */
-static const double kLpcCorrWindow[WINLEN] = {
- 0.00000000, 0.00000001, 0.00000004, 0.00000010, 0.00000020,
- 0.00000035, 0.00000055, 0.00000083, 0.00000118, 0.00000163,
- 0.00000218, 0.00000283, 0.00000361, 0.00000453, 0.00000558, 0.00000679,
- 0.00000817, 0.00000973, 0.00001147, 0.00001342, 0.00001558,
- 0.00001796, 0.00002058, 0.00002344, 0.00002657, 0.00002997,
- 0.00003365, 0.00003762, 0.00004190, 0.00004651, 0.00005144, 0.00005673,
- 0.00006236, 0.00006837, 0.00007476, 0.00008155, 0.00008875,
- 0.00009636, 0.00010441, 0.00011290, 0.00012186, 0.00013128,
- 0.00014119, 0.00015160, 0.00016252, 0.00017396, 0.00018594, 0.00019846,
- 0.00021155, 0.00022521, 0.00023946, 0.00025432, 0.00026978,
- 0.00028587, 0.00030260, 0.00031998, 0.00033802, 0.00035674,
- 0.00037615, 0.00039626, 0.00041708, 0.00043863, 0.00046092, 0.00048396,
- 0.00050775, 0.00053233, 0.00055768, 0.00058384, 0.00061080,
- 0.00063858, 0.00066720, 0.00069665, 0.00072696, 0.00075813,
- 0.00079017, 0.00082310, 0.00085692, 0.00089164, 0.00092728, 0.00096384,
- 0.00100133, 0.00103976, 0.00107914, 0.00111947, 0.00116077,
- 0.00120304, 0.00124630, 0.00129053, 0.00133577, 0.00138200,
- 0.00142924, 0.00147749, 0.00152676, 0.00157705, 0.00162836, 0.00168070,
- 0.00173408, 0.00178850, 0.00184395, 0.00190045, 0.00195799,
- 0.00201658, 0.00207621, 0.00213688, 0.00219860, 0.00226137,
- 0.00232518, 0.00239003, 0.00245591, 0.00252284, 0.00259079, 0.00265977,
- 0.00272977, 0.00280078, 0.00287280, 0.00294582, 0.00301984,
- 0.00309484, 0.00317081, 0.00324774, 0.00332563, 0.00340446,
- 0.00348421, 0.00356488, 0.00364644, 0.00372889, 0.00381220, 0.00389636,
- 0.00398135, 0.00406715, 0.00415374, 0.00424109, 0.00432920,
- 0.00441802, 0.00450754, 0.00459773, 0.00468857, 0.00478001,
- 0.00487205, 0.00496464, 0.00505775, 0.00515136, 0.00524542, 0.00533990,
- 0.00543476, 0.00552997, 0.00562548, 0.00572125, 0.00581725,
- 0.00591342, 0.00600973, 0.00610612, 0.00620254, 0.00629895,
- 0.00639530, 0.00649153, 0.00658758, 0.00668341, 0.00677894, 0.00687413,
- 0.00696891, 0.00706322, 0.00715699, 0.00725016, 0.00734266,
- 0.00743441, 0.00752535, 0.00761540, 0.00770449, 0.00779254,
- 0.00787947, 0.00796519, 0.00804963, 0.00813270, 0.00821431, 0.00829437,
- 0.00837280, 0.00844949, 0.00852436, 0.00859730, 0.00866822,
- 0.00873701, 0.00880358, 0.00886781, 0.00892960, 0.00898884,
- 0.00904542, 0.00909923, 0.00915014, 0.00919805, 0.00924283, 0.00928436,
- 0.00932252, 0.00935718, 0.00938821, 0.00941550, 0.00943890,
- 0.00945828, 0.00947351, 0.00948446, 0.00949098, 0.00949294,
- 0.00949020, 0.00948262, 0.00947005, 0.00945235, 0.00942938, 0.00940099,
- 0.00936704, 0.00932738, 0.00928186, 0.00923034, 0.00917268,
- 0.00910872, 0.00903832, 0.00896134, 0.00887763, 0.00878706,
- 0.00868949, 0.00858478, 0.00847280, 0.00835343, 0.00822653, 0.00809199,
- 0.00794970, 0.00779956, 0.00764145, 0.00747530, 0.00730103,
- 0.00711857, 0.00692787, 0.00672888, 0.00652158, 0.00630597,
- 0.00608208, 0.00584994, 0.00560962, 0.00536124, 0.00510493, 0.00484089,
- 0.00456935, 0.00429062, 0.00400505, 0.00371310, 0.00341532,
- 0.00311238, 0.00280511, 0.00249452, 0.00218184, 0.00186864,
- 0.00155690, 0.00124918, 0.00094895, 0.00066112, 0.00039320, 0.00015881
-};
-
-double WebRtcIsac_LevDurb(double *a, double *k, double *r, int order)
-{
-
- double sum, alpha;
- int m, m_h, i;
- alpha = 0; //warning -DH
- a[0] = 1.0;
- if (r[0] < LEVINSON_EPS) { /* if r[0] <= 0, set LPC coeff. to zero */
- for (i = 0; i < order; i++) {
- k[i] = 0;
- a[i+1] = 0;
- }
- } else {
- a[1] = k[0] = -r[1]/r[0];
- alpha = r[0] + r[1] * k[0];
- for (m = 1; m < order; m++){
- sum = r[m + 1];
- for (i = 0; i < m; i++){
- sum += a[i+1] * r[m - i];
- }
- k[m] = -sum / alpha;
- alpha += k[m] * sum;
- m_h = (m + 1) >> 1;
- for (i = 0; i < m_h; i++){
- sum = a[i+1] + k[m] * a[m - i];
- a[m - i] += k[m] * a[i+1];
- a[i+1] = sum;
- }
- a[m+1] = k[m];
- }
- }
- return alpha;
-}
-
-
-//was static before, but didn't work with MEX file
-void WebRtcIsac_GetVars(const double *input, const WebRtc_Word16 *pitchGains_Q12,
- double *oldEnergy, double *varscale)
-{
- double nrg[4], chng, pg;
- int k;
-
- double pitchGains[4]={0,0,0,0};;
-
- /* Calculate energies of first and second frame halfs */
- nrg[0] = 0.0001;
- for (k = QLOOKAHEAD/2; k < (FRAMESAMPLES_QUARTER + QLOOKAHEAD) / 2; k++) {
- nrg[0] += input[k]*input[k];
- }
- nrg[1] = 0.0001;
- for ( ; k < (FRAMESAMPLES_HALF + QLOOKAHEAD) / 2; k++) {
- nrg[1] += input[k]*input[k];
- }
- nrg[2] = 0.0001;
- for ( ; k < (FRAMESAMPLES*3/4 + QLOOKAHEAD) / 2; k++) {
- nrg[2] += input[k]*input[k];
- }
- nrg[3] = 0.0001;
- for ( ; k < (FRAMESAMPLES + QLOOKAHEAD) / 2; k++) {
- nrg[3] += input[k]*input[k];
- }
-
- /* Calculate average level change */
- chng = 0.25 * (fabs(10.0 * log10(nrg[3] / nrg[2])) +
- fabs(10.0 * log10(nrg[2] / nrg[1])) +
- fabs(10.0 * log10(nrg[1] / nrg[0])) +
- fabs(10.0 * log10(nrg[0] / *oldEnergy)));
-
-
- /* Find average pitch gain */
- pg = 0.0;
- for (k=0; k<4; k++)
- {
- pitchGains[k] = ((float)pitchGains_Q12[k])/4096;
- pg += pitchGains[k];
- }
- pg *= 0.25;
-
- /* If pitch gain is low and energy constant - increase noise level*/
- /* Matlab code:
- pg = 0:.01:.45; plot(pg, 0.0 + 1.0 * exp( -1.0 * exp(-200.0 * pg.*pg.*pg) / (1.0 + 0.4 * 0) ))
- */
- *varscale = 0.0 + 1.0 * exp( -1.4 * exp(-200.0 * pg*pg*pg) / (1.0 + 0.4 * chng) );
-
- *oldEnergy = nrg[3];
-}
-
-void
-WebRtcIsac_GetVarsUB(
- const double* input,
- double* oldEnergy,
- double* varscale)
-{
- double nrg[4], chng;
- int k;
-
- /* Calculate energies of first and second frame halfs */
- nrg[0] = 0.0001;
- for (k = 0; k < (FRAMESAMPLES_QUARTER) / 2; k++) {
- nrg[0] += input[k]*input[k];
- }
- nrg[1] = 0.0001;
- for ( ; k < (FRAMESAMPLES_HALF) / 2; k++) {
- nrg[1] += input[k]*input[k];
- }
- nrg[2] = 0.0001;
- for ( ; k < (FRAMESAMPLES*3/4) / 2; k++) {
- nrg[2] += input[k]*input[k];
- }
- nrg[3] = 0.0001;
- for ( ; k < (FRAMESAMPLES) / 2; k++) {
- nrg[3] += input[k]*input[k];
- }
-
- /* Calculate average level change */
- chng = 0.25 * (fabs(10.0 * log10(nrg[3] / nrg[2])) +
- fabs(10.0 * log10(nrg[2] / nrg[1])) +
- fabs(10.0 * log10(nrg[1] / nrg[0])) +
- fabs(10.0 * log10(nrg[0] / *oldEnergy)));
-
-
- /* If pitch gain is low and energy constant - increase noise level*/
- /* Matlab code:
- pg = 0:.01:.45; plot(pg, 0.0 + 1.0 * exp( -1.0 * exp(-200.0 * pg.*pg.*pg) / (1.0 + 0.4 * 0) ))
- */
- *varscale = exp( -1.4 / (1.0 + 0.4 * chng) );
-
- *oldEnergy = nrg[3];
-}
-
-void WebRtcIsac_GetLpcCoefLb(double *inLo, double *inHi, MaskFiltstr *maskdata,
- double signal_noise_ratio, const WebRtc_Word16 *pitchGains_Q12,
- double *lo_coeff, double *hi_coeff)
-{
- int k, n, j, pos1, pos2;
- double varscale;
-
- double DataLo[WINLEN], DataHi[WINLEN];
- double corrlo[ORDERLO+2], corrlo2[ORDERLO+1];
- double corrhi[ORDERHI+1];
- double k_veclo[ORDERLO], k_vechi[ORDERHI];
-
- double a_LO[ORDERLO+1], a_HI[ORDERHI+1];
- double tmp, res_nrg;
-
- double FwdA, FwdB;
-
- /* hearing threshold level in dB; higher value gives more noise */
- const double HearThresOffset = -28.0;
-
- /* bandwdith expansion factors for low- and high band */
- const double gammaLo = 0.9;
- const double gammaHi = 0.8;
-
- /* less-noise-at-low-frequencies factor */
- double aa;
-
-
- /* convert from dB to signal level */
- const double H_T_H = pow(10.0, 0.05 * HearThresOffset);
- double S_N_R = pow(10.0, 0.05 * signal_noise_ratio) / 3.46; /* divide by sqrt(12) */
-
- /* change quallevel depending on pitch gains and level fluctuations */
- WebRtcIsac_GetVars(inLo, pitchGains_Q12, &(maskdata->OldEnergy), &varscale);
-
- /* less-noise-at-low-frequencies factor */
- aa = 0.35 * (0.5 + 0.5 * varscale);
-
- /* replace data in buffer by new look-ahead data */
- for (pos1 = 0; pos1 < QLOOKAHEAD; pos1++)
- maskdata->DataBufferLo[pos1 + WINLEN - QLOOKAHEAD] = inLo[pos1];
-
- for (k = 0; k < SUBFRAMES; k++) {
-
- /* Update input buffer and multiply signal with window */
- for (pos1 = 0; pos1 < WINLEN - UPDATE/2; pos1++) {
- maskdata->DataBufferLo[pos1] = maskdata->DataBufferLo[pos1 + UPDATE/2];
- maskdata->DataBufferHi[pos1] = maskdata->DataBufferHi[pos1 + UPDATE/2];
- DataLo[pos1] = maskdata->DataBufferLo[pos1] * kLpcCorrWindow[pos1];
- DataHi[pos1] = maskdata->DataBufferHi[pos1] * kLpcCorrWindow[pos1];
- }
- pos2 = k * UPDATE/2;
- for (n = 0; n < UPDATE/2; n++, pos1++) {
- maskdata->DataBufferLo[pos1] = inLo[QLOOKAHEAD + pos2];
- maskdata->DataBufferHi[pos1] = inHi[pos2++];
- DataLo[pos1] = maskdata->DataBufferLo[pos1] * kLpcCorrWindow[pos1];
- DataHi[pos1] = maskdata->DataBufferHi[pos1] * kLpcCorrWindow[pos1];
- }
-
- /* Get correlation coefficients */
- WebRtcIsac_AutoCorr(corrlo, DataLo, WINLEN, ORDERLO+1); /* computing autocorrelation */
- WebRtcIsac_AutoCorr(corrhi, DataHi, WINLEN, ORDERHI);
-
-
- /* less noise for lower frequencies, by filtering/scaling autocorrelation sequences */
- corrlo2[0] = (1.0+aa*aa) * corrlo[0] - 2.0*aa * corrlo[1];
- tmp = (1.0 + aa*aa);
- for (n = 1; n <= ORDERLO; n++) {
- corrlo2[n] = tmp * corrlo[n] - aa * (corrlo[n-1] + corrlo[n+1]);
- }
- tmp = (1.0+aa) * (1.0+aa);
- for (n = 0; n <= ORDERHI; n++) {
- corrhi[n] = tmp * corrhi[n];
- }
-
- /* add white noise floor */
- corrlo2[0] += 1e-6;
- corrhi[0] += 1e-6;
-
-
- FwdA = 0.01;
- FwdB = 0.01;
-
- /* recursive filtering of correlation over subframes */
- for (n = 0; n <= ORDERLO; n++) {
- maskdata->CorrBufLo[n] = FwdA * maskdata->CorrBufLo[n] + corrlo2[n];
- corrlo2[n] = ((1.0-FwdA)*FwdB) * maskdata->CorrBufLo[n] + (1.0-FwdB) * corrlo2[n];
- }
- for (n = 0; n <= ORDERHI; n++) {
- maskdata->CorrBufHi[n] = FwdA * maskdata->CorrBufHi[n] + corrhi[n];
- corrhi[n] = ((1.0-FwdA)*FwdB) * maskdata->CorrBufHi[n] + (1.0-FwdB) * corrhi[n];
- }
-
- /* compute prediction coefficients */
- WebRtcIsac_LevDurb(a_LO, k_veclo, corrlo2, ORDERLO);
- WebRtcIsac_LevDurb(a_HI, k_vechi, corrhi, ORDERHI);
-
- /* bandwidth expansion */
- tmp = gammaLo;
- for (n = 1; n <= ORDERLO; n++) {
- a_LO[n] *= tmp;
- tmp *= gammaLo;
- }
-
- /* residual energy */
- res_nrg = 0.0;
- for (j = 0; j <= ORDERLO; j++) {
- for (n = 0; n <= j; n++) {
- res_nrg += a_LO[j] * corrlo2[j-n] * a_LO[n];
- }
- for (n = j+1; n <= ORDERLO; n++) {
- res_nrg += a_LO[j] * corrlo2[n-j] * a_LO[n];
- }
- }
-
- /* add hearing threshold and compute the gain */
- *lo_coeff++ = S_N_R / (sqrt(res_nrg) / varscale + H_T_H);
-
- /* copy coefficients to output array */
- for (n = 1; n <= ORDERLO; n++) {
- *lo_coeff++ = a_LO[n];
- }
-
-
- /* bandwidth expansion */
- tmp = gammaHi;
- for (n = 1; n <= ORDERHI; n++) {
- a_HI[n] *= tmp;
- tmp *= gammaHi;
- }
-
- /* residual energy */
- res_nrg = 0.0;
- for (j = 0; j <= ORDERHI; j++) {
- for (n = 0; n <= j; n++) {
- res_nrg += a_HI[j] * corrhi[j-n] * a_HI[n];
- }
- for (n = j+1; n <= ORDERHI; n++) {
- res_nrg += a_HI[j] * corrhi[n-j] * a_HI[n];
- }
- }
-
- /* add hearing threshold and compute of the gain */
- *hi_coeff++ = S_N_R / (sqrt(res_nrg) / varscale + H_T_H);
-
- /* copy coefficients to output array */
- for (n = 1; n <= ORDERHI; n++) {
- *hi_coeff++ = a_HI[n];
- }
- }
-}
-
-
-
-/******************************************************************************
- * WebRtcIsac_GetLpcCoefUb()
- *
- * Compute LP coefficients and correlation coefficients. At 12 kHz LP
- * coefficients of the first and the last sub-frame is computed. At 16 kHz
- * LP coefficients of 4th, 8th and 12th sub-frames are computed. We always
- * compute correlation coefficients of all sub-frames.
- *
- * Inputs:
- * -inSignal : Input signal
- * -maskdata : a structure keeping signal from previous frame.
- * -bandwidth : specifies if the codec is in 0-16 kHz mode or
- * 0-12 kHz mode.
- *
- * Outputs:
- * -lpCoeff : pointer to a buffer where A-polynomials are
- * written to (first coeff is 1 and it is not
- * written)
- * -corrMat : a matrix where correlation coefficients of each
- * sub-frame are written to one row.
- * -varscale : a scale used to compute LPC gains.
- */
-void
-WebRtcIsac_GetLpcCoefUb(
- double* inSignal,
- MaskFiltstr* maskdata,
- double* lpCoeff,
- double corrMat[][UB_LPC_ORDER + 1],
- double* varscale,
- WebRtc_Word16 bandwidth)
-{
- int frameCntr, activeFrameCntr, n, pos1, pos2;
- WebRtc_Word16 criterion1;
- WebRtc_Word16 criterion2;
- WebRtc_Word16 numSubFrames = SUBFRAMES * (1 + (bandwidth == isac16kHz));
- double data[WINLEN];
- double corrSubFrame[UB_LPC_ORDER+2];
- double reflecCoeff[UB_LPC_ORDER];
-
- double aPolynom[UB_LPC_ORDER+1];
- double tmp;
-
- /* bandwdith expansion factors */
- const double gamma = 0.9;
-
- /* change quallevel depending on pitch gains and level fluctuations */
- WebRtcIsac_GetVarsUB(inSignal, &(maskdata->OldEnergy), varscale);
-
- /* replace data in buffer by new look-ahead data */
- for(frameCntr = 0, activeFrameCntr = 0; frameCntr < numSubFrames;
- frameCntr++)
- {
- if(frameCntr == SUBFRAMES)
- {
- // we are in 16 kHz
- varscale++;
- WebRtcIsac_GetVarsUB(&inSignal[FRAMESAMPLES_HALF],
- &(maskdata->OldEnergy), varscale);
- }
- /* Update input buffer and multiply signal with window */
- for(pos1 = 0; pos1 < WINLEN - UPDATE/2; pos1++)
- {
- maskdata->DataBufferLo[pos1] = maskdata->DataBufferLo[pos1 +
- UPDATE/2];
- data[pos1] = maskdata->DataBufferLo[pos1] * kLpcCorrWindow[pos1];
- }
- pos2 = frameCntr * UPDATE/2;
- for(n = 0; n < UPDATE/2; n++, pos1++, pos2++)
- {
- maskdata->DataBufferLo[pos1] = inSignal[pos2];
- data[pos1] = maskdata->DataBufferLo[pos1] * kLpcCorrWindow[pos1];
- }
-
- /* Get correlation coefficients */
- /* computing autocorrelation */
- WebRtcIsac_AutoCorr(corrSubFrame, data, WINLEN, UB_LPC_ORDER+1);
- memcpy(corrMat[frameCntr], corrSubFrame,
- (UB_LPC_ORDER+1)*sizeof(double));
-
- criterion1 = ((frameCntr == 0) || (frameCntr == (SUBFRAMES - 1))) &&
- (bandwidth == isac12kHz);
- criterion2 = (((frameCntr+1) % 4) == 0) &&
- (bandwidth == isac16kHz);
- if(criterion1 || criterion2)
- {
- /* add noise */
- corrSubFrame[0] += 1e-6;
- /* compute prediction coefficients */
- WebRtcIsac_LevDurb(aPolynom, reflecCoeff, corrSubFrame,
- UB_LPC_ORDER);
-
- /* bandwidth expansion */
- tmp = gamma;
- for (n = 1; n <= UB_LPC_ORDER; n++)
- {
- *lpCoeff++ = aPolynom[n] * tmp;
- tmp *= gamma;
- }
- activeFrameCntr++;
- }
- }
-}
-
-
-
-/******************************************************************************
- * WebRtcIsac_GetLpcGain()
- *
- * Compute the LPC gains for each sub-frame, given the LPC of each sub-frame
- * and the corresponding correlation coefficients.
- *
- * Inputs:
- * -signal_noise_ratio : the desired SNR in dB.
- * -numVecs : number of sub-frames
- * -corrMat : a matrix of correlation coefficients where
- * each row is a set of correlation coefficients of
- * one sub-frame.
- * -varscale : a scale computed when WebRtcIsac_GetLpcCoefUb()
- * is called.
- *
- * Outputs:
- * -gain : pointer to a buffer where LP gains are written.
- *
- */
-void
-WebRtcIsac_GetLpcGain(
- double signal_noise_ratio,
- const double* filtCoeffVecs,
- int numVecs,
- double* gain,
- double corrMat[][UB_LPC_ORDER + 1],
- const double* varscale)
-{
- WebRtc_Word16 j, n;
- WebRtc_Word16 subFrameCntr;
- double aPolynom[ORDERLO + 1];
- double res_nrg;
-
- const double HearThresOffset = -28.0;
- const double H_T_H = pow(10.0, 0.05 * HearThresOffset);
- /* divide by sqrt(12) = 3.46 */
- const double S_N_R = pow(10.0, 0.05 * signal_noise_ratio) / 3.46;
-
- aPolynom[0] = 1;
- for(subFrameCntr = 0; subFrameCntr < numVecs; subFrameCntr++)
- {
- if(subFrameCntr == SUBFRAMES)
- {
- // we are in second half of a SWB frame. use new varscale
- varscale++;
- }
- memcpy(&aPolynom[1], &filtCoeffVecs[(subFrameCntr * (UB_LPC_ORDER + 1)) +
- 1], sizeof(double) * UB_LPC_ORDER);
-
- /* residual energy */
- res_nrg = 0.0;
- for(j = 0; j <= UB_LPC_ORDER; j++)
- {
- for(n = 0; n <= j; n++)
- {
- res_nrg += aPolynom[j] * corrMat[subFrameCntr][j-n] *
- aPolynom[n];
- }
- for(n = j+1; n <= UB_LPC_ORDER; n++)
- {
- res_nrg += aPolynom[j] * corrMat[subFrameCntr][n-j] *
- aPolynom[n];
- }
- }
-
- /* add hearing threshold and compute the gain */
- gain[subFrameCntr] = S_N_R / (sqrt(res_nrg) / *varscale + H_T_H);
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_analysis.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_analysis.h
deleted file mode 100644
index 4eafeac..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_analysis.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * lpc_analysis.h
- *
- * LPC functions
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYSIS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYSIS_H_
-
-#include "settings.h"
-#include "structs.h"
-
-double WebRtcIsac_LevDurb(double *a, double *k, double *r, int order);
-
-void WebRtcIsac_GetVars(const double *input, const WebRtc_Word16 *pitchGains_Q12,
- double *oldEnergy, double *varscale);
-
-void WebRtcIsac_GetLpcCoefLb(double *inLo, double *inHi, MaskFiltstr *maskdata,
- double signal_noise_ratio, const WebRtc_Word16 *pitchGains_Q12,
- double *lo_coeff, double *hi_coeff);
-
-
-void WebRtcIsac_GetLpcGain(
- double signal_noise_ratio,
- const double* filtCoeffVecs,
- int numVecs,
- double* gain,
- double corrLo[][UB_LPC_ORDER + 1],
- const double* varscale);
-
-void WebRtcIsac_GetLpcCoefUb(
- double* inSignal,
- MaskFiltstr* maskdata,
- double* lpCoeff,
- double corr[][UB_LPC_ORDER + 1],
- double* varscale,
- WebRtc_Word16 bandwidth);
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_ANALYIS_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_gain_swb_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_gain_swb_tables.c
deleted file mode 100644
index 25c69cb..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_gain_swb_tables.c
+++ /dev/null
@@ -1,137 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * SWB_KLT_Tables_LPCGain.c
- *
- * This file defines tables used for entropy coding of LPC Gain
- * of upper-band.
- *
- */
-
-#include "lpc_gain_swb_tables.h"
-#include "settings.h"
-#include "typedefs.h"
-
-const double WebRtcIsac_kQSizeLpcGain = 0.100000;
-
-const double WebRtcIsac_kMeanLpcGain = -3.3822;
-
-/*
-* The smallest reconstruction points for quantiztion of
-* LPC gains.
-*/
-const double WebRtcIsac_kLeftRecPointLpcGain[SUBFRAMES] =
-{
- -0.800000, -1.000000, -1.200000, -2.200000, -3.000000, -12.700000
-};
-
-/*
-* Number of reconstruction points of quantizers for LPC Gains.
-*/
-const WebRtc_Word16 WebRtcIsac_kNumQCellLpcGain[SUBFRAMES] =
-{
- 17, 20, 25, 45, 77, 170
-};
-/*
-* Starting index for entropy decoder to search for the right interval,
-* one entry per LAR coefficient
-*/
-const WebRtc_UWord16 WebRtcIsac_kLpcGainEntropySearch[SUBFRAMES] =
-{
- 8, 10, 12, 22, 38, 85
-};
-
-/*
-* The following 6 vectors define CDF of 6 decorrelated LPC
-* gains.
-*/
-const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec0[18] =
-{
- 0, 10, 27, 83, 234, 568, 1601, 4683, 16830, 57534, 63437,
- 64767, 65229, 65408, 65483, 65514, 65527, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec1[21] =
-{
- 0, 15, 33, 84, 185, 385, 807, 1619, 3529, 7850, 19488,
- 51365, 62437, 64548, 65088, 65304, 65409, 65484, 65507, 65522, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec2[26] =
-{
- 0, 15, 29, 54, 89, 145, 228, 380, 652, 1493, 4260,
- 12359, 34133, 50749, 57224, 60814, 62927, 64078, 64742, 65103, 65311, 65418,
- 65473, 65509, 65521, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec3[46] =
-{
- 0, 8, 12, 16, 26, 42, 56, 76, 111, 164, 247,
- 366, 508, 693, 1000, 1442, 2155, 3188, 4854, 7387, 11249, 17617,
- 30079, 46711, 56291, 60127, 62140, 63258, 63954, 64384, 64690, 64891, 65031,
- 65139, 65227, 65293, 65351, 65399, 65438, 65467, 65492, 65504, 65510, 65518,
- 65523, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec4[78] =
-{
- 0, 17, 29, 39, 51, 70, 104, 154, 234, 324, 443,
- 590, 760, 971, 1202, 1494, 1845, 2274, 2797, 3366, 4088, 4905,
- 5899, 7142, 8683, 10625, 12983, 16095, 20637, 28216, 38859, 47237, 51537,
- 54150, 56066, 57583, 58756, 59685, 60458, 61103, 61659, 62144, 62550, 62886,
- 63186, 63480, 63743, 63954, 64148, 64320, 64467, 64600, 64719, 64837, 64939,
- 65014, 65098, 65160, 65211, 65250, 65290, 65325, 65344, 65366, 65391, 65410,
- 65430, 65447, 65460, 65474, 65487, 65494, 65501, 65509, 65513, 65518, 65520,
- 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec5[171] =
-{
- 0, 10, 12, 14, 16, 18, 23, 29, 35, 42, 51,
- 58, 65, 72, 78, 87, 96, 103, 111, 122, 134, 150,
- 167, 184, 202, 223, 244, 265, 289, 315, 346, 379, 414,
- 450, 491, 532, 572, 613, 656, 700, 751, 802, 853, 905,
- 957, 1021, 1098, 1174, 1250, 1331, 1413, 1490, 1565, 1647, 1730,
- 1821, 1913, 2004, 2100, 2207, 2314, 2420, 2532, 2652, 2783, 2921,
- 3056, 3189, 3327, 3468, 3640, 3817, 3993, 4171, 4362, 4554, 4751,
- 4948, 5142, 5346, 5566, 5799, 6044, 6301, 6565, 6852, 7150, 7470,
- 7797, 8143, 8492, 8835, 9181, 9547, 9919, 10315, 10718, 11136, 11566,
- 12015, 12482, 12967, 13458, 13953, 14432, 14903, 15416, 15936, 16452, 16967,
- 17492, 18024, 18600, 19173, 19736, 20311, 20911, 21490, 22041, 22597, 23157,
- 23768, 24405, 25034, 25660, 26280, 26899, 27614, 28331, 29015, 29702, 30403,
- 31107, 31817, 32566, 33381, 34224, 35099, 36112, 37222, 38375, 39549, 40801,
- 42074, 43350, 44626, 45982, 47354, 48860, 50361, 51845, 53312, 54739, 56026,
- 57116, 58104, 58996, 59842, 60658, 61488, 62324, 63057, 63769, 64285, 64779,
- 65076, 65344, 65430, 65500, 65517, 65535
-};
-
-/*
-* An array of pointers to CDFs of decorrelated LPC Gains
-*/
-const WebRtc_UWord16* WebRtcIsac_kLpcGainCdfMat[SUBFRAMES] =
-{
- WebRtcIsac_kLpcGainCdfVec0, WebRtcIsac_kLpcGainCdfVec1,
- WebRtcIsac_kLpcGainCdfVec2, WebRtcIsac_kLpcGainCdfVec3,
- WebRtcIsac_kLpcGainCdfVec4, WebRtcIsac_kLpcGainCdfVec5
-};
-
-/*
-* A matrix to decorrellate LPC gains of subframes.
-*/
-const double WebRtcIsac_kLpcGainDecorrMat[SUBFRAMES][SUBFRAMES] =
-{
- {-0.150860, 0.327872, 0.367220, 0.504613, 0.559270, 0.409234},
- { 0.457128, -0.613591, -0.289283, -0.029734, 0.393760, 0.418240},
- {-0.626043, 0.136489, -0.439118, -0.448323, 0.135987, 0.420869},
- { 0.526617, 0.480187, 0.242552, -0.488754, -0.158713, 0.411331},
- {-0.302587, -0.494953, 0.588112, -0.063035, -0.404290, 0.387510},
- { 0.086378, 0.147714, -0.428875, 0.548300, -0.570121, 0.401391}
-};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_gain_swb_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_gain_swb_tables.h
deleted file mode 100644
index 1eba97c..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_gain_swb_tables.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * SWB_KLT_Tables_LPCGain.h
- *
- * This file declares tables used for entropy coding of LPC Gain
- * of upper-band.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_
-
-#include "settings.h"
-#include "typedefs.h"
-
-extern const double WebRtcIsac_kQSizeLpcGain;
-
-extern const double WebRtcIsac_kLeftRecPointLpcGain[SUBFRAMES];
-
-extern const WebRtc_Word16 WebRtcIsac_kNumQCellLpcGain[SUBFRAMES];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcGainEntropySearch[SUBFRAMES];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec0[18];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec1[21];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec2[26];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec3[46];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec4[78];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcGainCdfVec5[171];
-
-extern const WebRtc_UWord16* WebRtcIsac_kLpcGainCdfMat[SUBFRAMES];
-
-extern const double WebRtcIsac_kLpcGainDecorrMat[SUBFRAMES][SUBFRAMES];
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_GAIN_SWB_TABLES_H_
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb12_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb12_tables.c
deleted file mode 100644
index 695d583..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb12_tables.c
+++ /dev/null
@@ -1,159 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * SWB_KLT_Tables.c
- *
- * This file defines tables used for entropy coding of LPC shape of
- * upper-band signal if the bandwidth is 12 kHz.
- *
- */
-
-#include "lpc_shape_swb12_tables.h"
-#include "settings.h"
-#include "typedefs.h"
-
-/*
-* Mean value of LAR
-*/
-const double WebRtcIsac_kMeanLarUb12[UB_LPC_ORDER] =
-{
- 0.03748928306641, 0.09453441192543, -0.01112522344398, 0.03800237516842
-};
-
-/*
-* A rotation matrix to decorrelate intra-vector correlation,
-* i.e. correlation among components of LAR vector.
-*/
-const double WebRtcIsac_kIntraVecDecorrMatUb12[UB_LPC_ORDER][UB_LPC_ORDER] =
-{
- {-0.00075365493856, -0.05809964887743, -0.23397966154116, 0.97050367376411},
- { 0.00625021257734, -0.17299965610679, 0.95977735920651, 0.22104179375008},
- { 0.20543384258374, -0.96202143495696, -0.15301870801552, -0.09432375099565},
- {-0.97865075648479, -0.20300322280841, -0.02581111653779, -0.01913568980258}
-};
-
-/*
-* A rotation matrix to remove correlation among LAR coefficients
-* of different LAR vectors. One might guess that decorrelation matrix
-* for the first component should differ from the second component
-* but we haven't observed a significant benefit of having different
-* decorrelation matrices for different components.
-*/
-const double WebRtcIsac_kInterVecDecorrMatUb12
-[UB_LPC_VEC_PER_FRAME][UB_LPC_VEC_PER_FRAME] =
-{
- { 0.70650597970460, -0.70770707262373},
- {-0.70770707262373, -0.70650597970460}
-};
-
-/*
-* LAR quantization step-size.
-*/
-const double WebRtcIsac_kLpcShapeQStepSizeUb12 = 0.150000;
-
-/*
-* The smallest reconstruction points for quantiztion of LAR coefficients.
-*/
-const double WebRtcIsac_kLpcShapeLeftRecPointUb12
-[UB_LPC_ORDER*UB_LPC_VEC_PER_FRAME] =
-{
- -0.900000, -1.050000, -1.350000, -1.800000, -1.350000, -1.650000,
- -2.250000, -3.450000
-};
-
-/*
-* Number of reconstruction points of quantizers for LAR coefficients.
-*/
-const WebRtc_Word16 WebRtcIsac_kLpcShapeNumRecPointUb12
-[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME] =
-{
- 13, 15, 19, 27, 19, 24, 32, 48
-};
-
-/*
-* Starting index for entropy decoder to search for the right interval,
-* one entry per LAR coefficient
-*/
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeEntropySearchUb12
-[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME] =
-{
- 6, 7, 9, 13, 9, 12, 16, 24
-};
-
-/*
-* The following 8 vectors define CDF of 8 decorrelated LAR
-* coefficients.
-*/
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec0Ub12[14] =
-{
- 0, 13, 95, 418, 1687, 6498, 21317, 44200, 59029, 63849, 65147,
- 65449, 65525, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec1Ub12[16] =
-{
- 0, 10, 59, 255, 858, 2667, 8200, 22609, 42988, 57202, 62947,
- 64743, 65308, 65476, 65522, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec2Ub12[20] =
-{
- 0, 18, 40, 118, 332, 857, 2017, 4822, 11321, 24330, 41279,
- 54342, 60637, 63394, 64659, 65184, 65398, 65482, 65518, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec3Ub12[28] =
-{
- 0, 21, 38, 90, 196, 398, 770, 1400, 2589, 4650, 8211,
- 14933, 26044, 39592, 50814, 57452, 60971, 62884, 63995, 64621, 65019, 65273,
- 65410, 65480, 65514, 65522, 65531, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec4Ub12[20] =
-{
- 0, 7, 46, 141, 403, 969, 2132, 4649, 10633, 24902, 43254,
- 54665, 59928, 62674, 64173, 64938, 65293, 65464, 65523, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec5Ub12[25] =
-{
- 0, 7, 22, 72, 174, 411, 854, 1737, 3545, 6774, 13165,
- 25221, 40980, 52821, 58714, 61706, 63472, 64437, 64989, 65287, 65430, 65503,
- 65525, 65529, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec6Ub12[33] =
-{
- 0, 11, 21, 36, 65, 128, 228, 401, 707, 1241, 2126,
- 3589, 6060, 10517, 18853, 31114, 42477, 49770, 54271, 57467, 59838, 61569,
- 62831, 63772, 64433, 64833, 65123, 65306, 65419, 65466, 65499, 65519, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec7Ub12[49] =
-{
- 0, 14, 34, 67, 107, 167, 245, 326, 449, 645, 861,
- 1155, 1508, 2003, 2669, 3544, 4592, 5961, 7583, 9887, 13256, 18765,
- 26519, 34077, 40034, 44349, 47795, 50663, 53262, 55473, 57458, 59122, 60592,
- 61742, 62690, 63391, 63997, 64463, 64794, 65045, 65207, 65309, 65394, 65443,
- 65478, 65504, 65514, 65523, 65535
-};
-
-/*
-* An array of pointers to CDFs of decorrelated LARs
-*/
-const WebRtc_UWord16* WebRtcIsac_kLpcShapeCdfMatUb12
-[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME] =
-{
- WebRtcIsac_kLpcShapeCdfVec0Ub12, WebRtcIsac_kLpcShapeCdfVec1Ub12,
- WebRtcIsac_kLpcShapeCdfVec2Ub12, WebRtcIsac_kLpcShapeCdfVec3Ub12,
- WebRtcIsac_kLpcShapeCdfVec4Ub12, WebRtcIsac_kLpcShapeCdfVec5Ub12,
- WebRtcIsac_kLpcShapeCdfVec6Ub12, WebRtcIsac_kLpcShapeCdfVec7Ub12
-};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb12_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb12_tables.h
deleted file mode 100644
index 1e93847..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb12_tables.h
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * lpc_shape_swb12_tables.h
- *
- * This file declares tables used for entropy coding of LPC shape of
- * upper-band signal if the bandwidth is 12 kHz.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
-
-#include "settings.h"
-#include "typedefs.h"
-
-extern const double WebRtcIsac_kMeanLarUb12[UB_LPC_ORDER];
-
-extern const double WebRtcIsac_kMeanLpcGain;
-
-extern const double WebRtcIsac_kIntraVecDecorrMatUb12[UB_LPC_ORDER][UB_LPC_ORDER];
-
-extern const double WebRtcIsac_kInterVecDecorrMatUb12
-[UB_LPC_VEC_PER_FRAME][UB_LPC_VEC_PER_FRAME];
-
-extern const double WebRtcIsac_kLpcShapeQStepSizeUb12;
-
-extern const double WebRtcIsac_kLpcShapeLeftRecPointUb12
-[UB_LPC_ORDER*UB_LPC_VEC_PER_FRAME];
-
-
-extern const WebRtc_Word16 WebRtcIsac_kLpcShapeNumRecPointUb12
-[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeEntropySearchUb12
-[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec0Ub12[14];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec1Ub12[16];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec2Ub12[20];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec3Ub12[28];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec4Ub12[20];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec5Ub12[25];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec6Ub12[33];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec7Ub12[49];
-
-extern const WebRtc_UWord16* WebRtcIsac_kLpcShapeCdfMatUb12
-[UB_LPC_ORDER * UB_LPC_VEC_PER_FRAME];
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB12_TABLES_H_
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb16_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb16_tables.c
deleted file mode 100644
index 89f4523..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb16_tables.c
+++ /dev/null
@@ -1,248 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * SWB16_KLT_Tables.c
- *
- * This file defines tables used for entropy coding of LPC shape of
- * upper-band signal if the bandwidth is 16 kHz.
- *
- */
-
-#include "lpc_shape_swb16_tables.h"
-#include "settings.h"
-#include "typedefs.h"
-
-/*
-* Mean value of LAR
-*/
-const double WebRtcIsac_kMeanLarUb16[UB_LPC_ORDER] =
-{
-0.454978, 0.364747, 0.102999, 0.104523
-};
-
-/*
-* A rotation matrix to decorrelate intra-vector correlation,
-* i.e. correlation among components of LAR vector.
-*/
-const double WebRtcIsac_kIintraVecDecorrMatUb16[UB_LPC_ORDER][UB_LPC_ORDER] =
-{
- {-0.020528, -0.085858, -0.002431, 0.996093},
- {-0.033155, 0.036102, 0.998786, 0.004866},
- { 0.202627, 0.974853, -0.028940, 0.088132},
- {-0.978479, 0.202454, -0.039785, -0.002811}
-};
-
-/*
-* A rotation matrix to remove correlation among LAR coefficients
-* of different LAR vectors. One might guess that decorrelation matrix
-* for the first component should differ from the second component
-* but we haven't observed a significant benefit of having different
-* decorrelation matrices for different components.
-*/
-const double WebRtcIsac_kInterVecDecorrMatUb16
-[UB16_LPC_VEC_PER_FRAME][UB16_LPC_VEC_PER_FRAME] =
-{
- { 0.291675, -0.515786, 0.644927, 0.482658},
- {-0.647220, 0.479712, 0.289556, 0.516856},
- { 0.643084, 0.485489, -0.289307, 0.516763},
- {-0.287185, -0.517823, -0.645389, 0.482553}
-};
-
-/*
-* The following 16 vectors define CDF of 16 decorrelated LAR
-* coefficients.
-*/
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub16[14] =
-{
- 0, 2, 20, 159, 1034, 5688, 20892, 44653,
- 59849, 64485, 65383, 65518, 65534, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec1Ub16[16] =
-{
- 0, 1, 7, 43, 276, 1496, 6681, 21653,
- 43891, 58859, 64022, 65248, 65489, 65529, 65534, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec2Ub16[18] =
-{
- 0, 1, 9, 54, 238, 933, 3192, 9461,
- 23226, 42146, 56138, 62413, 64623, 65300, 65473, 65521,
- 65533, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec3Ub16[30] =
-{
- 0, 2, 4, 8, 17, 36, 75, 155,
- 329, 683, 1376, 2662, 5047, 9508, 17526, 29027,
- 40363, 48997, 55096, 59180, 61789, 63407, 64400, 64967,
- 65273, 65429, 65497, 65526, 65534, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec4Ub16[16] =
-{
- 0, 1, 10, 63, 361, 1785, 7407, 22242,
- 43337, 58125, 63729, 65181, 65472, 65527, 65534, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec5Ub16[17] =
-{
- 0, 1, 7, 29, 134, 599, 2443, 8590,
- 22962, 42635, 56911, 63060, 64940, 65408, 65513, 65531,
- 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec6Ub16[21] =
-{
- 0, 1, 5, 16, 57, 191, 611, 1808,
- 4847, 11755, 24612, 40910, 53789, 60698, 63729, 64924,
- 65346, 65486, 65523, 65532, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec7Ub16[36] =
-{
- 0, 1, 4, 12, 25, 55, 104, 184,
- 314, 539, 926, 1550, 2479, 3861, 5892, 8845,
- 13281, 20018, 29019, 38029, 45581, 51557, 56057, 59284,
- 61517, 63047, 64030, 64648, 65031, 65261, 65402, 65480,
- 65518, 65530, 65534, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec8Ub16[21] =
-{
- 0, 1, 2, 7, 26, 103, 351, 1149,
- 3583, 10204, 23846, 41711, 55361, 61917, 64382, 65186,
- 65433, 65506, 65528, 65534, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub160[21] =
-{
- 0, 6, 19, 63, 205, 638, 1799, 4784,
- 11721, 24494, 40803, 53805, 60886, 63822, 64931, 65333,
- 65472, 65517, 65530, 65533, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub161[28] =
-{
- 0, 1, 3, 11, 31, 86, 221, 506,
- 1101, 2296, 4486, 8477, 15356, 26079, 38941, 49952,
- 57165, 61257, 63426, 64549, 65097, 65351, 65463, 65510,
- 65526, 65532, 65534, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub162[55] =
-{
- 0, 3, 12, 23, 42, 65, 89, 115,
- 150, 195, 248, 327, 430, 580, 784, 1099,
- 1586, 2358, 3651, 5899, 9568, 14312, 19158, 23776,
- 28267, 32663, 36991, 41153, 45098, 48680, 51870, 54729,
- 57141, 59158, 60772, 62029, 63000, 63761, 64322, 64728,
- 65000, 65192, 65321, 65411, 65463, 65496, 65514, 65523,
- 65527, 65529, 65531, 65532, 65533, 65534, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub163[26] =
-{
- 0, 2, 4, 10, 21, 48, 114, 280,
- 701, 1765, 4555, 11270, 24267, 41213, 54285, 61003,
- 63767, 64840, 65254, 65421, 65489, 65514, 65526, 65532,
- 65534, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub164[28] =
-{
- 0, 1, 3, 6, 15, 36, 82, 196,
- 453, 1087, 2557, 5923, 13016, 25366, 40449, 52582,
- 59539, 62896, 64389, 65033, 65316, 65442, 65494, 65519,
- 65529, 65533, 65534, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub165[34] =
-{
- 0, 2, 4, 8, 18, 35, 73, 146,
- 279, 524, 980, 1789, 3235, 5784, 10040, 16998,
- 27070, 38543, 48499, 55421, 59712, 62257, 63748, 64591,
- 65041, 65278, 65410, 65474, 65508, 65522, 65530, 65533,
- 65534, 65535
-};
-
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub166[71] =
-{
- 0, 1, 2, 6, 13, 26, 55, 92,
- 141, 191, 242, 296, 355, 429, 522, 636,
- 777, 947, 1162, 1428, 1753, 2137, 2605, 3140,
- 3743, 4409, 5164, 6016, 6982, 8118, 9451, 10993,
- 12754, 14810, 17130, 19780, 22864, 26424, 30547, 35222,
- 40140, 44716, 48698, 52056, 54850, 57162, 59068, 60643,
- 61877, 62827, 63561, 64113, 64519, 64807, 65019, 65167,
- 65272, 65343, 65399, 65440, 65471, 65487, 65500, 65509,
- 65518, 65524, 65527, 65531, 65533, 65534, 65535
-};
-
-/*
-* An array of pointers to CDFs of decorrelated LARs
-*/
-const WebRtc_UWord16* WebRtcIsac_kLpcShapeCdfMatUb16
-[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME] = {
- WebRtcIsac_kLpcShapeCdfVec01Ub16,
- WebRtcIsac_kLpcShapeCdfVec1Ub16,
- WebRtcIsac_kLpcShapeCdfVec2Ub16,
- WebRtcIsac_kLpcShapeCdfVec3Ub16,
- WebRtcIsac_kLpcShapeCdfVec4Ub16,
- WebRtcIsac_kLpcShapeCdfVec5Ub16,
- WebRtcIsac_kLpcShapeCdfVec6Ub16,
- WebRtcIsac_kLpcShapeCdfVec7Ub16,
- WebRtcIsac_kLpcShapeCdfVec8Ub16,
- WebRtcIsac_kLpcShapeCdfVec01Ub160,
- WebRtcIsac_kLpcShapeCdfVec01Ub161,
- WebRtcIsac_kLpcShapeCdfVec01Ub162,
- WebRtcIsac_kLpcShapeCdfVec01Ub163,
- WebRtcIsac_kLpcShapeCdfVec01Ub164,
- WebRtcIsac_kLpcShapeCdfVec01Ub165,
- WebRtcIsac_kLpcShapeCdfVec01Ub166
-};
-
-/*
-* The smallest reconstruction points for quantiztion of LAR coefficients.
-*/
-const double WebRtcIsac_kLpcShapeLeftRecPointUb16
-[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME] =
-{
- -0.8250, -0.9750, -1.1250, -2.1750, -0.9750, -1.1250, -1.4250,
- -2.6250, -1.4250, -1.2750, -1.8750, -3.6750, -1.7250, -1.8750,
- -2.3250, -5.4750
-};
-
-/*
-* Number of reconstruction points of quantizers for LAR coefficients.
-*/
-const WebRtc_Word16 WebRtcIsac_kLpcShapeNumRecPointUb16
-[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME] =
-{
- 13, 15, 17, 29, 15, 16, 20, 35, 20,
- 20, 27, 54, 25, 27, 33, 70
-};
-
-/*
-* Starting index for entropy decoder to search for the right interval,
-* one entry per LAR coefficient
-*/
-const WebRtc_UWord16 WebRtcIsac_kLpcShapeEntropySearchUb16
-[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME] =
-{
- 6, 7, 8, 14, 7, 8, 10, 17, 10,
- 10, 13, 27, 12, 13, 16, 35
-};
-
-/*
-* LAR quantization step-size.
-*/
-const double WebRtcIsac_kLpcShapeQStepSizeUb16 = 0.150000;
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb16_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb16_tables.h
deleted file mode 100644
index 68d08b2..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_shape_swb16_tables.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * lpc_shape_swb16_tables.h
- *
- * This file declares tables used for entropy coding of LPC shape of
- * upper-band signal if the bandwidth is 16 kHz.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_
-
-#include "settings.h"
-#include "typedefs.h"
-
-
-extern const double WebRtcIsac_kMeanLarUb16[UB_LPC_ORDER];
-
-extern const double WebRtcIsac_kIintraVecDecorrMatUb16[UB_LPC_ORDER][UB_LPC_ORDER];
-
-extern const double WebRtcIsac_kInterVecDecorrMatUb16
-[UB16_LPC_VEC_PER_FRAME][UB16_LPC_VEC_PER_FRAME];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub16[14];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec1Ub16[16];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec2Ub16[18];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec3Ub16[30];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec4Ub16[16];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec5Ub16[17];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec6Ub16[21];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec7Ub16[36];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec8Ub16[21];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub160[21];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub161[28];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub162[55];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub163[26];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub164[28];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub165[34];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeCdfVec01Ub166[71];
-
-extern const WebRtc_UWord16* WebRtcIsac_kLpcShapeCdfMatUb16
-[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
-
-extern const double WebRtcIsac_kLpcShapeLeftRecPointUb16
-[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
-
-extern const WebRtc_Word16 WebRtcIsac_kLpcShapeNumRecPointUb16
-[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
-
-extern const WebRtc_UWord16 WebRtcIsac_kLpcShapeEntropySearchUb16
-[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
-
-extern const double WebRtcIsac_kLpcShapeQStepSizeUb16;
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_SHAPE_SWB16_TABLES_H_
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_tables.c
deleted file mode 100644
index 7df6121..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_tables.c
+++ /dev/null
@@ -1,1129 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/* coding tables for the KLT coefficients */
-
-#include "lpc_tables.h"
-#include "settings.h"
-
-/* indices of KLT coefficients used */
-const WebRtc_UWord16 WebRtcIsac_kQKltSelIndGain[12] = {
- 0, 1, 2, 3, 4, 5, 6, 7, 8, 9,
- 10, 11};
-
-const WebRtc_UWord16 WebRtcIsac_kQKltSelIndShape[108] = {
- 0, 1, 2, 3, 4, 5, 6, 7, 8, 9,
- 10, 11, 12, 13, 14, 15, 16, 17, 18, 19,
- 20, 21, 22, 23, 24, 25, 26, 27, 28, 29,
- 30, 31, 32, 33, 34, 35, 36, 37, 38, 39,
- 40, 41, 42, 43, 44, 45, 46, 47, 48, 49,
- 50, 51, 52, 53, 54, 55, 56, 57, 58, 59,
- 60, 61, 62, 63, 64, 65, 66, 67, 68, 69,
- 70, 71, 72, 73, 74, 75, 76, 77, 78, 79,
- 80, 81, 82, 83, 84, 85, 86, 87, 88, 89,
- 90, 91, 92, 93, 94, 95, 96, 97, 98, 99,
- 100, 101, 102, 103, 104, 105, 106, 107};
-
-/* cdf array for model indicator */
-const WebRtc_UWord16 WebRtcIsac_kQKltModelCdf[4] = {
- 0, 15434, 37548, 65535};
-
-/* pointer to cdf array for model indicator */
-const WebRtc_UWord16 *WebRtcIsac_kQKltModelCdfPtr[1] = {WebRtcIsac_kQKltModelCdf};
-
-/* initial cdf index for decoder of model indicator */
-const WebRtc_UWord16 WebRtcIsac_kQKltModelInitIndex[1] = {1};
-
-/* offset to go from rounded value to quantization index */
-const short WebRtcIsac_kQKltQuantMinGain[12] = {
- 3, 6, 4, 6, 6, 9, 5, 16, 11, 34, 32, 47};
-
-
-const short WebRtcIsac_kQKltQuantMinShape[108] = {
- 0, 0, 0, 0, 0, 0, 0, 0, 0, 1,
- 1, 1, 1, 1, 2, 2, 2, 3, 0, 0,
- 0, 0, 1, 0, 0, 0, 0, 1, 1, 1,
- 1, 1, 1, 2, 2, 3, 0, 0, 0, 0,
- 1, 0, 1, 1, 1, 1, 1, 1, 1, 2,
- 2, 4, 3, 5, 0, 0, 0, 0, 1, 1,
- 1, 1, 1, 1, 2, 1, 2, 2, 3, 4,
- 4, 7, 0, 0, 1, 1, 1, 1, 1, 1,
- 1, 2, 3, 2, 3, 4, 4, 5, 7, 13,
- 0, 1, 1, 2, 3, 2, 2, 2, 4, 4,
- 5, 6, 7, 11, 9, 13, 12, 26};
-
-/* maximum quantization index */
-const WebRtc_UWord16 WebRtcIsac_kQKltMaxIndGain[12] = {
- 6, 12, 8, 14, 10, 19, 12, 31, 22, 56, 52, 138};
-
-const WebRtc_UWord16 WebRtcIsac_kQKltMaxIndShape[108] = {
- 0, 0, 0, 0, 0, 0, 0, 0, 0, 1,
- 2, 2, 2, 2, 4, 4, 5, 6, 0, 0,
- 0, 0, 1, 0, 0, 0, 0, 1, 2, 2,
- 2, 2, 3, 4, 5, 7, 0, 0, 0, 0,
- 2, 0, 2, 2, 2, 2, 3, 2, 2, 4,
- 4, 6, 6, 9, 0, 0, 0, 0, 2, 2,
- 2, 2, 2, 2, 3, 2, 4, 4, 7, 7,
- 9, 13, 0, 0, 2, 2, 2, 2, 2, 2,
- 3, 4, 5, 4, 6, 8, 8, 10, 16, 25,
- 0, 2, 2, 4, 5, 4, 4, 4, 7, 8,
- 9, 10, 13, 19, 17, 23, 25, 49};
-
-/* index offset */
-const WebRtc_UWord16 WebRtcIsac_kQKltOffsetGain[3][12] = {
-{ 0, 7, 20, 29, 44, 55, 75, 88, 120, 143,
- 200, 253},
-{ 0, 7, 19, 27, 42, 53, 73, 86, 117, 140,
- 197, 249},
-{ 0, 7, 20, 28, 44, 55, 75, 89, 121, 145,
- 202, 257}};
-
-const WebRtc_UWord16 WebRtcIsac_kQKltOffsetShape[3][108] = {
-{ 0, 1, 2, 3, 4, 5, 6, 7, 8, 9,
- 11, 14, 17, 20, 23, 28, 33, 39, 46, 47,
- 48, 49, 50, 52, 53, 54, 55, 56, 58, 61,
- 64, 67, 70, 74, 79, 85, 93, 94, 95, 96,
- 97, 100, 101, 104, 107, 110, 113, 117, 120, 123,
- 128, 133, 140, 147, 157, 158, 159, 160, 161, 164,
- 167, 170, 173, 176, 179, 183, 186, 191, 196, 204,
- 212, 222, 236, 237, 238, 241, 244, 247, 250, 253,
- 256, 260, 265, 271, 276, 283, 292, 301, 312, 329,
- 355, 356, 359, 362, 367, 373, 378, 383, 388, 396,
- 405, 415, 426, 440, 460, 478, 502, 528},
-{ 0, 1, 2, 3, 4, 6, 7, 8, 9, 11,
- 13, 16, 19, 22, 26, 29, 34, 39, 45, 46,
- 47, 48, 49, 50, 51, 52, 53, 55, 57, 60,
- 63, 66, 70, 73, 78, 84, 91, 92, 93, 94,
- 95, 96, 97, 99, 102, 105, 108, 111, 114, 118,
- 123, 128, 134, 141, 151, 152, 153, 154, 156, 159,
- 162, 165, 168, 171, 174, 177, 181, 186, 194, 200,
- 208, 218, 233, 234, 235, 236, 239, 242, 245, 248,
- 251, 254, 258, 263, 270, 277, 288, 297, 308, 324,
- 349, 351, 354, 357, 361, 366, 372, 378, 383, 390,
- 398, 407, 420, 431, 450, 472, 496, 524},
-{ 0, 1, 2, 3, 4, 5, 6, 7, 8, 11,
- 14, 17, 20, 23, 26, 29, 34, 40, 47, 48,
- 49, 50, 51, 52, 53, 54, 55, 58, 61, 64,
- 67, 70, 73, 77, 82, 88, 96, 97, 98, 99,
- 101, 102, 104, 107, 110, 113, 116, 119, 122, 125,
- 129, 134, 141, 150, 160, 161, 162, 163, 166, 168,
- 171, 174, 177, 180, 183, 186, 190, 195, 201, 208,
- 216, 226, 243, 244, 245, 248, 251, 254, 257, 260,
- 263, 268, 273, 278, 284, 291, 299, 310, 323, 340,
- 366, 368, 371, 374, 379, 383, 389, 394, 399, 406,
- 414, 422, 433, 445, 461, 480, 505, 533}};
-
-/* initial cdf index for KLT coefficients */
-const WebRtc_UWord16 WebRtcIsac_kQKltInitIndexGain[3][12] = {
-{ 3, 6, 4, 7, 5, 10, 6, 16, 11, 28,
- 26, 69},
-{ 3, 6, 4, 7, 5, 10, 6, 15, 11, 28,
- 26, 69},
-{ 3, 6, 4, 8, 5, 10, 7, 16, 12, 28,
- 27, 70}};
-
-const WebRtc_UWord16 WebRtcIsac_kQKltInitIndexShape[3][108] = {
-{ 0, 0, 0, 0, 0, 0, 0, 0, 0, 1,
- 1, 1, 1, 1, 2, 2, 3, 3, 0, 0,
- 0, 0, 1, 0, 0, 0, 0, 1, 1, 1,
- 1, 1, 2, 2, 3, 4, 0, 0, 0, 0,
- 1, 0, 1, 1, 1, 1, 2, 1, 1, 2,
- 2, 3, 3, 5, 0, 0, 0, 0, 1, 1,
- 1, 1, 1, 1, 2, 1, 2, 2, 4, 4,
- 5, 7, 0, 0, 1, 1, 1, 1, 1, 1,
- 2, 2, 3, 2, 3, 4, 4, 5, 8, 13,
- 0, 1, 1, 2, 3, 2, 2, 2, 4, 4,
- 5, 5, 7, 10, 9, 12, 13, 25},
-{ 0, 0, 0, 0, 1, 0, 0, 0, 1, 1,
- 1, 1, 1, 2, 1, 2, 2, 3, 0, 0,
- 0, 0, 0, 0, 0, 0, 1, 1, 1, 1,
- 1, 2, 1, 2, 3, 3, 0, 0, 0, 0,
- 0, 0, 1, 1, 1, 1, 1, 1, 2, 2,
- 2, 3, 3, 5, 0, 0, 0, 1, 1, 1,
- 1, 1, 1, 1, 1, 2, 2, 4, 3, 4,
- 5, 7, 0, 0, 0, 1, 1, 1, 1, 1,
- 1, 2, 2, 3, 3, 5, 4, 5, 8, 12,
- 1, 1, 1, 2, 2, 3, 3, 2, 3, 4,
- 4, 6, 5, 9, 11, 12, 14, 25},
-{ 0, 0, 0, 0, 0, 0, 0, 0, 1, 1,
- 1, 1, 1, 1, 1, 2, 3, 3, 0, 0,
- 0, 0, 0, 0, 0, 0, 1, 1, 1, 1,
- 1, 1, 2, 2, 3, 4, 0, 0, 0, 1,
- 0, 1, 1, 1, 1, 1, 1, 1, 1, 2,
- 2, 3, 4, 5, 0, 0, 0, 1, 1, 1,
- 1, 1, 1, 1, 1, 2, 2, 3, 3, 4,
- 5, 8, 0, 0, 1, 1, 1, 1, 1, 1,
- 2, 2, 2, 3, 3, 4, 5, 6, 8, 13,
- 1, 1, 1, 2, 2, 3, 2, 2, 3, 4,
- 4, 5, 6, 8, 9, 12, 14, 25}};
-
-/* offsets for quantizer representation levels*/
-const WebRtc_UWord16 WebRtcIsac_kQKltOfLevelsGain[3] = {
- 0, 392, 779};
-
-const WebRtc_UWord16 WebRtcIsac_kQKltOfLevelsShape[3] = {
- 0, 578, 1152};
-
-/* quantizer representation levels */
-const double WebRtcIsac_kQKltLevelsGain[1176] = {
--2.78127126, -1.76745590, -0.77913790, -0.00437329, 0.79961206, 1.81775776, 2.81389782, -5.78753143, -4.88384084, -3.89320940,
--2.88133610, -1.92859977, -0.86347396, 0.02003888, 0.86140400, 1.89667156, 2.97134967, 3.98781964, 4.91727277, 5.82865898,
--4.11195874, -2.80898424, -1.87547977, -0.80943825, -0.00679084, 0.79573851, 1.83953397, 2.67586037, 3.76274082, -6.10933968,
--4.93034581, -3.89281296, -2.91530625, -1.89684163, -0.85319130, -0.02275767, 0.86862017, 1.91578276, 2.96107339, 3.96543056,
- 4.91369908, 5.91058154, 6.83848343, 8.07136925, -5.87470395, -4.84703049, -3.84284597, -2.86168446, -1.89290192, -0.82798145,
--0.00080013, 0.82594974, 1.85754329, 2.88351798, 3.96172628, -8.85684885, -7.87387461, -6.97811862, -5.93256270, -4.94301439,
--3.95513701, -2.96041544, -1.94031192, -0.87961478, -0.00456201, 0.89911505, 1.91723376, 2.94011511, 3.93302540, 4.97990967,
- 5.93133404, 7.02181199, 7.92407762, 8.80155440, 10.04665814, -4.82396678, -3.85612158, -2.89482244, -1.89558408, -0.90036978,
--0.00677823, 0.90607989, 1.90937981, 2.91175777, 3.91637730, 4.97565723, 5.84771228, 7.11145863, -16.07879840, -15.03776309,
--13.93905670, -12.95671800, -11.89171202, -10.95820934, -9.95923714, -8.94357334, -7.99068299, -6.97481009, -5.94826231, -4.96673988,
--3.97490466, -2.97846970, -1.95130435, -0.94215262, -0.01444043, 0.96770704, 1.95848598, 2.94107862, 3.95666119, 4.97253085,
- 5.97191122, 6.93277360, 7.96608727, 8.87958779, 10.00264269, 10.86560820, 12.07449071, 13.04491775, 13.97507061, 14.91845261,
--10.85696295, -9.83365357, -9.01245635, -7.95915145, -6.95625003, -5.95362618, -4.93468444, -3.98760978, -2.95044407, -1.97041277,
--0.97701799, -0.00840234, 0.97834289, 1.98361415, 2.97802439, 3.96415871, 4.95369042, 5.94101770, 6.92756798, 7.94063998,
- 8.85951828, 9.97077022, 11.00068503, -33.92030406, -32.81426422, -32.00000000, -31.13243639, -30.11886909, -29.06017570, -28.12598824,
--27.22045482, -25.81215858, -25.07849962, -23.93018013, -23.02097643, -21.89529725, -20.99091085, -19.98889048, -18.94327044, -17.96562071,
--16.96126218, -15.95054062, -14.98516200, -13.97101012, -13.02106500, -11.98438006, -11.03216748, -9.95930286, -8.97043946, -7.98085082,
--6.98360995, -5.98998802, -4.98668173, -4.00032906, -3.00420619, -1.98701132, -0.99324682, -0.00609324, 0.98297834, 1.99483076,
- 3.00305044, 3.97142097, 4.97525759, 5.98612258, 6.97448236, 7.97575900, 9.01086211, 9.98665542, 11.00541438, 11.98078628,
- 12.92352471, 14.06849675, 14.99949430, 15.94904834, 16.97440321, 18.04040916, 18.88987609, 20.05312391, 21.00000000, 21.79443341,
--31.98578825, -31.00000000, -29.89060567, -28.98555686, -27.97114102, -26.84935410, -26.02402230, -24.94195278, -23.92336849, -22.95552382,
--21.97932836, -20.96055470, -19.99649553, -19.03436122, -17.96706525, -17.01139515, -16.01363516, -14.99154248, -14.00298333, -12.99630613,
--11.99955519, -10.99000421, -10.00819092, -8.99763648, -7.98431793, -7.01769025, -5.99604690, -4.99980697, -3.99334671, -3.01748192,
--2.02051217, -1.00848371, -0.01942358, 1.00477757, 1.95477872, 2.98593031, 3.98779079, 4.96862849, 6.02694771, 6.93983733,
- 7.89874717, 8.99615862, 10.02367921, 10.96293452, 11.84351528, 12.92207187, 13.85122329, 15.05146877, 15.99371264, 17.00000000,
- 18.00000000, 19.00000000, 19.82763573, -47.00000000, -46.00000000, -44.87138498, -44.00000000, -43.00000000, -42.00000000, -41.00000000,
--39.88966612, -38.98913239, -37.80306486, -37.23584325, -35.94200288, -34.99881301, -34.11361858, -33.06507360, -32.13129135, -30.90891364,
--29.81511907, -28.99250380, -28.04535391, -26.99767800, -26.04418164, -24.95687851, -24.04865595, -23.03392645, -21.89366707, -20.93517364,
--19.99388660, -18.91620943, -18.03749683, -16.99532379, -15.98683813, -15.06421479, -13.99359211, -12.99714098, -11.97022520, -10.98500279,
--9.98834422, -8.95729330, -8.01232284, -7.00253661, -5.99681626, -5.01207817, -3.95914904, -3.01232178, -1.96615919, -0.97687670,
- 0.01228030, 0.98412288, 2.01753544, 3.00580570, 3.97783510, 4.98846894, 6.01321400, 7.00867732, 8.00416375, 9.01771966,
- 9.98637729, 10.98255180, 11.99194163, 13.01807333, 14.00999545, 15.00118556, 16.00089224, 17.00584148, 17.98251763, 18.99942091,
- 19.96917690, 20.97839265, 21.98207297, 23.00171271, 23.99930737, 24.99746061, 26.00936304, 26.98240132, 28.01126868, 29.01395915,
- 29.98153507, 31.01376711, 31.99876818, 33.00475317, 33.99753994, 34.99493913, 35.98933585, 36.95620160, 37.98428461, 38.99317544,
- 40.01832073, 40.98048133, 41.95999283, 42.98232091, 43.96523612, 44.99574268, 45.99524194, 47.05464025, 48.03821548, 48.99354366,
- 49.96400411, 50.98017973, 51.95184408, 52.96291806, 54.00194392, 54.96603783, 55.95623778, 57.03076595, 58.05889901, 58.99081551,
- 59.97928121, 61.05071612, 62.03971580, 63.01286038, 64.01290338, 65.02074503, 65.99454594, 67.00399425, 67.96571257, 68.95305727,
- 69.92030664, 70.95594862, 71.98088567, 73.04764124, 74.00285480, 75.02696330, 75.89837673, 76.93459997, 78.16266309, 78.83317543,
- 80.00000000, 80.87251574, 82.09803524, 83.10671664, 84.00000000, 84.77023523, 86.00000000, 87.00000000, 87.92946897, 88.69159118,
- 90.00000000, 90.90535270, -3.00000000, -2.00000000, -0.77592424, -0.00564307, 0.76727305, 2.00000000, 3.00000000, -6.00000000,
--5.00000000, -4.00000000, -2.92897924, -1.85623684, -0.72445303, -0.00119184, 0.72896652, 2.05710416, 3.17909894, 4.00000000,
- 5.00000000, -3.00000000, -2.00000000, -0.67480586, -0.00028016, 0.66618169, 2.00000000, 3.00000000, 4.00000000, -7.00000000,
--6.00000000, -5.00000000, -3.78336783, -2.84811556, -2.04088844, -0.71114371, 0.03142493, 0.69662772, 1.91417930, 3.00000000,
- 4.01411062, 5.00000000, 6.00000000, 7.00000000, -6.00000000, -5.00000000, -4.00000000, -3.00000000, -2.00000000, -0.63703469,
- 0.00169604, 0.66294191, 1.83808563, 3.00000000, 4.00000000, -8.00000000, -7.00000000, -6.03082300, -5.00000000, -3.88061019,
--2.92670084, -1.99902336, -0.72898996, -0.02880170, 0.73769927, 1.95920233, 2.78356263, 4.08100921, 5.00000000, 6.00000000,
- 6.78771437, 8.00000000, 9.00000000, 10.00000000, 11.00000000, -5.00000000, -4.00000000, -2.88150384, -1.89520024, -0.71479482,
- 0.00962397, 0.72816030, 1.73583550, 3.00000000, 4.00000000, 5.00000000, 6.00000000, 7.00000000, -16.00000000, -15.00000000,
--13.80516401, -13.00000000, -12.00000000, -11.00000000, -10.02723144, -9.11825995, -8.05820112, -7.00000000, -6.17943541, -5.01837980,
--3.97546169, -2.92806857, -1.89778775, -0.81138893, -0.02246016, 0.80528415, 1.85705214, 2.96438524, 3.97540151, 4.79684246,
- 6.00000000, 6.75549513, 8.12185828, 9.00000000, 10.00000000, 11.00000000, 12.00000000, 13.00000000, 14.00000000, -11.00000000,
--10.00000000, -9.00000000, -7.91603344, -6.77865892, -5.85765006, -4.93342332, -3.96679157, -2.84925552, -1.89230732, -0.85384229,
- 0.00579591, 0.84863246, 1.89006713, 2.89483818, 3.87322971, 5.13228411, 6.00000000, 7.00000000, 8.00000000, 9.00000000,
- 10.00000000, 11.00000000, -34.00000000, -33.00000000, -32.00000000, -31.00000000, -30.00000000, -29.00000000, -28.00000000, -27.00000000,
--26.00000000, -25.00000000, -24.00000000, -23.00000000, -22.00000000, -21.00000000, -20.19501953, -19.00000000, -18.00000000, -17.00000000,
--16.00000000, -14.89069633, -14.00000000, -13.00000000, -12.16260304, -11.15418282, -9.83543570, -8.85600407, -7.82712677, -7.05664308,
--5.97007352, -4.89268438, -3.93822771, -2.94975269, -1.92192127, -0.90702480, 0.03974847, 0.92488359, 1.93747579, 2.94500522,
- 3.95181797, 4.95433087, 5.95141808, 7.00212920, 8.02964757, 9.03210585, 9.84644504, 10.82907720, 11.87622530, 12.96908371,
- 14.00000000, 15.16963413, 15.94902025, 17.00000000, 18.00000000, 19.00000000, 20.00000000, 21.00000000, 22.00000000, -29.00000000,
--27.79780781, -27.00757888, -26.01571026, -24.89695568, -23.99946491, -22.98699614, -21.96678139, -20.99883532, -20.00851529, -18.94738054,
--17.98672566, -16.98684787, -15.96917397, -14.99856852, -13.98974852, -12.97786927, -11.96110939, -10.98877093, -9.99875257, -8.99001359,
--8.00799989, -6.99471760, -6.00034670, -4.99936372, -4.00581479, -3.00424577, -2.02047620, -0.99713266, -0.00366397, 1.00803955,
- 1.98452687, 3.00748501, 4.02714611, 4.97661026, 5.99337271, 6.99754716, 8.00713602, 8.97184974, 9.98047901, 10.97685939,
- 11.99533975, 12.96107876, 13.95061478, 15.00756776, 15.94078690, 16.88231059, 17.92069248, 18.78011047, 20.00000000, 21.00000000,
- 22.00000000, -55.76988333, -54.96048193, -53.88411581, -52.94117980, -51.80983449, -50.90359699, -50.00000000, -48.99838741, -47.97685542,
--47.03288597, -45.97820919, -45.02418374, -43.90081897, -42.88832512, -41.98234549, -40.96745512, -39.98148729, -39.06792854, -37.96493755,
--36.98707870, -36.03416079, -35.01192444, -33.95785029, -32.99469087, -31.96633807, -31.01769053, -29.99727691, -28.99329690, -27.98873019,
--27.00344273, -25.97657141, -25.00511074, -23.96689479, -23.01566842, -22.01632643, -21.00076343, -19.97788007, -18.97248680, -17.96076284,
--16.97585453, -15.98345587, -15.01612745, -13.96862118, -12.96622055, -12.02196641, -11.02078103, -9.98445656, -9.00050060, -8.03442387,
--7.00363761, -5.97921358, -4.98886269, -4.00528221, -3.01672947, -1.98599795, -1.00668518, -0.02633490, 1.00794139, 2.00837138,
- 2.99213287, 3.98710216, 4.99064334, 6.01416391, 7.01759708, 7.97878151, 8.99665730, 10.02656114, 11.01863887, 12.01207901,
- 13.00958725, 13.99237829, 15.00954971, 16.00724653, 17.00606559, 17.99886292, 18.99611967, 19.98808171, 21.01871930, 21.97014763,
- 22.99833843, 24.00316842, 24.99949142, 25.98539601, 27.02480733, 27.98075377, 28.98266019, 30.00611445, 30.99409128, 31.94523141,
- 32.97688339, 33.98800206, 35.00177074, 35.98639997, 36.98939428, 37.95644255, 39.00114054, 39.99492439, 40.99338254, 41.97050844,
- 43.03085663, 43.96757668, 44.97800970, 45.95953358, 46.98109551, 47.99368477, 49.00141209, 49.94459923, 50.93298108, 51.99894661,
- 53.06463883, 53.99704669, 55.02037199, 55.98368047, 57.01930954, 58.03813852, 58.96232502, 60.01644186, 61.03254711, 62.01086576,
- 62.87962247, 63.98378413, 65.02189831, 65.93003954, 66.92439900, 68.07051633, 68.95928756, 70.03315022, 71.05579859, 72.00000000,
- 73.00000000, 74.00000000, 75.00000000, 75.93485291, 77.20950456, 78.00000000, 79.00000000, 79.91519960, 81.00000000, -3.00000000,
--2.00000000, -0.65174074, -0.00092112, 0.62967387, 2.00000000, 3.00000000, -6.00000000, -5.00000000, -4.00000000, -2.89861729,
--1.69999061, -0.72632201, 0.00219241, 0.72891750, 1.73257865, 3.00000000, 3.76561508, 5.00000000, 6.00000000, -3.00000000,
--2.00000000, -0.66227013, 0.00389373, 0.66163500, 2.00000000, 3.00000000, 4.00000000, -8.00000000, -7.00000000, -6.00000000,
--4.76421796, -4.04320264, -3.01415201, -1.84346485, -0.77185048, 0.00061977, 0.76274524, 1.84330156, 3.00000000, 4.00000000,
- 5.00000000, 6.00000000, 7.00000000, -6.00000000, -5.00000000, -4.00000000, -3.00000000, -1.75749611, -0.72951347, -0.00104394,
- 0.72040315, 1.72594036, 3.00000000, 4.00000000, -9.00000000, -8.00000000, -7.00000000, -5.90394062, -5.00000000, -3.75562807,
--2.89699407, -1.86696610, -0.79056636, -0.00330943, 0.79744554, 1.85149941, 2.91118681, 3.99520311, 4.96341987, 6.00000000,
- 7.00000000, 8.00000000, 9.00000000, 10.00000000, -6.00000000, -4.80151529, -4.00000000, -2.87442856, -1.85285815, -0.77767592,
--0.02071301, 0.81752572, 1.82503940, 2.79602150, 3.92870203, 5.00000000, 6.00000000, 7.00000000, -17.00000000, -16.00000000,
--15.00000000, -14.00000000, -13.00000000, -12.00000000, -11.00000000, -9.80059874, -9.00000000, -8.00185204, -7.13087808, -5.92942149,
--4.77883243, -3.93417708, -2.88004618, -1.89952522, -0.86239337, 0.00332274, 0.86657548, 1.89479279, 2.89701813, 3.90987417,
- 4.98910145, 6.07676766, 7.00000000, 8.00000000, 9.00000000, 10.00000000, 11.00000000, 12.00000000, 13.00000000, 14.00000000,
--12.00000000, -11.00000000, -9.89996262, -8.85894205, -7.87594823, -6.99685317, -5.94917589, -4.93914916, -3.93317670, -2.93174244,
--1.90737478, -0.90982242, 0.00803316, 0.90111563, 1.90362879, 2.90332432, 3.90654662, 4.94461954, 5.87963665, 6.91988113,
- 7.79514004, 8.98805413, 10.00000000, 11.00000000, -35.00000000, -34.00000000, -33.00000000, -32.00000000, -31.00000000, -30.00000000,
--29.00000000, -28.00000000, -27.00000000, -26.00000000, -25.00000000, -24.00000000, -22.88310970, -22.00000000, -21.00000000, -20.00000000,
--19.00000000, -18.00000000, -17.00000000, -16.11854974, -15.00000000, -14.10507667, -13.04497040, -11.94846700, -10.97432494, -9.94514368,
--8.97311414, -7.94171496, -6.97232122, -5.98590548, -4.97455572, -3.95477903, -2.93935454, -1.95573532, -0.97120273, -0.02084826,
- 0.95689153, 1.96679781, 2.97060165, 3.96660892, 4.96754331, 5.97996089, 6.93822411, 7.96618014, 8.95809791, 9.98891474,
- 10.95713402, 11.85433084, 13.03831696, 13.84035295, 15.00729606, 15.98652872, 17.20557599, 18.00000000, 18.90794805, 20.00000000,
- 21.00000000, -34.00000000, -33.00000000, -32.00000000, -31.00000000, -30.00000000, -28.97280602, -28.00000000, -27.16255057, -26.04078092,
--24.85442050, -24.15783484, -22.78614956, -21.95739865, -21.21844626, -20.03008104, -19.03888543, -17.90460490, -17.02064693, -15.84673652,
--14.87140709, -13.87996048, -12.94907251, -11.96795995, -11.00977925, -9.95103238, -8.96674655, -7.96351667, -6.96886200, -5.99335494,
--4.97515534, -3.98891694, -2.99581150, -1.98758360, -0.99249128, -0.00001403, 0.98807868, 1.99119869, 2.99019366, 3.98612953,
- 5.00312941, 5.98833080, 6.99686651, 7.98373889, 8.97942222, 9.94202752, 10.99671622, 11.94306164, 12.98539825, 13.90728690,
- 14.89907642, 15.94836675, 16.89611342, 17.84084949, 18.74910958, 20.00000000, -67.00000000, -66.00000000, -65.00000000, -64.00000000,
--63.02511977, -62.00000000, -61.06061493, -59.95964043, -59.12824439, -58.00000000, -57.00000000, -56.00000000, -54.87857996, -54.09689334,
--53.00000000, -52.21057366, -50.93867921, -50.03032952, -49.19283867, -47.89439051, -46.99505692, -46.04895543, -44.89687413, -43.78942208,
--42.99025156, -41.88436155, -40.99169704, -40.00320429, -38.90181498, -38.06029271, -37.05030818, -36.07554573, -35.03202233, -33.93117946,
--32.97736655, -31.98942819, -30.99546798, -30.01511004, -28.97296525, -28.02561164, -26.94386985, -25.99632704, -25.00461143, -24.01578192,
--22.99177609, -22.02261094, -20.97939001, -19.96176066, -19.00442980, -18.01529434, -17.00196902, -15.99794828, -14.98675055, -13.97517657,
--12.98676283, -11.99718760, -11.00167809, -9.98872268, -9.02138474, -8.00320338, -6.99542797, -6.00059136, -5.01311763, -4.00336943,
--3.00348281, -1.99365875, -0.98223019, 0.00126343, 0.99699237, 1.99381968, 3.00054436, 3.99898305, 5.00160508, 6.00310399,
- 6.99885096, 8.02740039, 8.99515550, 9.98962151, 11.00642302, 11.98694516, 13.00018933, 13.97726018, 14.99186645, 16.00580131,
- 16.97434224, 17.96982658, 19.00066438, 20.01228749, 21.00741822, 21.94988312, 23.00860212, 23.98801542, 24.97638417, 25.98003521,
- 27.02336188, 27.99667029, 29.01014125, 30.02481912, 31.01415797, 31.97399854, 33.06214485, 33.99929330, 34.94095386, 35.96368372,
- 36.96980925, 37.98389244, 39.01121235, 40.00715026, 41.06382894, 41.96618280, 43.01555590, 43.95430436, 45.01970038, 45.99967821,
- 47.19847394, 48.04852502, 49.10609965, 50.04244122, 50.86051406, 51.92983796, 53.02781107, 54.06248545, 54.89942009, 56.08347165,
- 57.06887956, 58.09671115, 59.07832400, 59.87005277, 61.14778499, 62.00000000, 63.00000000, 64.00000000, 65.00000000, 66.00000000,
- 67.00000000, 68.00000000, 69.00000000, 70.00000000, 71.00000000, 72.00000000};
-
-const double WebRtcIsac_kQKltLevelsShape[1735] = {
- 0.00032397, 0.00008053, -0.00061202, -0.00012620, 0.00030437, 0.00054764, -0.00027902, 0.00069360, 0.00029449, -0.80219239,
- 0.00091089, -0.74514927, -0.00094283, 0.64030631, -0.60509119, 0.00035575, 0.61851665, -0.62129957, 0.00375219, 0.60054900,
--0.61554359, 0.00054977, 0.63362016, -1.73118727, -0.65422341, 0.00524568, 0.66165298, 1.76785515, -1.83182018, -0.65997434,
--0.00011887, 0.67524299, 1.79933938, -1.76344480, -0.72547708, -0.00133017, 0.73104704, 1.75305377, 2.85164534, -2.80423916,
--1.71959639, -0.75419722, -0.00329945, 0.77196760, 1.72211069, 2.87339653, 0.00031089, -0.00015311, 0.00018201, -0.00035035,
--0.77357251, 0.00154647, -0.00047625, -0.00045299, 0.00086590, 0.00044762, -0.83383829, 0.00024787, -0.68526258, -0.00122472,
- 0.64643255, -0.60904942, -0.00448987, 0.62309184, -0.59626442, -0.00574132, 0.62296546, -0.63222115, 0.00013441, 0.63609545,
--0.66911055, -0.00369971, 0.66346095, 2.07281301, -1.77184694, -0.67640425, -0.00010145, 0.64818392, 1.74948973, -1.69420224,
--0.71943894, -0.00004680, 0.75303493, 1.81075983, 2.80610041, -2.80005755, -1.79866753, -0.77409777, -0.00084220, 0.80141293,
- 1.78291081, 2.73954236, 3.82994169, 0.00015140, -0.00012766, -0.00034241, -0.00119125, -0.76113497, 0.00069246, 0.76722027,
- 0.00132862, -0.69107530, 0.00010656, 0.77061578, -0.78012970, 0.00095947, 0.77828502, -0.64787758, 0.00217168, 0.63050167,
--0.58601125, 0.00306596, 0.59466308, -0.58603410, 0.00059779, 0.64257970, 1.76512766, -0.61193600, -0.00259517, 0.59767574,
--0.61026273, 0.00315811, 0.61725479, -1.69169719, -0.65816029, 0.00067575, 0.65576890, 2.00000000, -1.72689193, -0.69780808,
--0.00040990, 0.70668487, 1.74198458, -3.79028154, -3.00000000, -1.73194459, -0.70179341, -0.00106695, 0.71302629, 1.76849782,
--2.89332364, -1.78585007, -0.78731491, -0.00132610, 0.79692976, 1.75247009, 2.97828682, -5.26238694, -3.69559829, -2.87286122,
--1.84908818, -0.84434577, -0.01167975, 0.84641753, 1.84087672, 2.87628156, 3.83556679, -0.00190204, 0.00092642, 0.00354385,
--0.00012982, -0.67742785, 0.00229509, 0.64935672, -0.58444751, 0.00470733, 0.57299534, -0.58456202, -0.00097715, 0.64593607,
--0.64060330, -0.00638534, 0.59680157, -0.59287537, 0.00490772, 0.58919707, -0.60306173, -0.00417464, 0.60562100, -1.75218757,
--0.63018569, -0.00225922, 0.63863300, -0.63949939, -0.00126421, 0.64268914, -1.75851182, -0.68318060, 0.00510418, 0.69049211,
- 1.88178506, -1.71136148, -0.72710534, -0.00815559, 0.73412917, 1.79996711, -2.77111145, -1.73940498, -0.78212945, 0.01074476,
- 0.77688916, 1.76873972, 2.87281379, 3.77554698, -3.75832725, -2.95463235, -1.80451491, -0.80017226, 0.00149902, 0.80729206,
- 1.78265046, 2.89391793, -3.78236148, -2.83640598, -1.82532067, -0.88844327, -0.00620952, 0.88208030, 1.85757631, 2.81712391,
- 3.88430176, 5.16179367, -7.00000000, -5.93805408, -4.87172597, -3.87524433, -2.89399744, -1.92359563, -0.92136341, -0.00172725,
- 0.93087018, 1.90528280, 2.89809686, 3.88085708, 4.89147740, 5.89078692, -0.00239502, 0.00312564, -1.00000000, 0.00178325,
- 1.00000000, -0.62198029, 0.00143254, 0.65344051, -0.59851220, -0.00676987, 0.61510140, -0.58894151, 0.00385055, 0.59794203,
--0.59808568, -0.00038214, 0.57625703, -0.63009713, -0.01107985, 0.61278758, -0.64206758, -0.00154369, 0.65480598, 1.80604162,
--1.80909286, -0.67810514, 0.00205762, 0.68571097, 1.79453891, -3.22682422, -1.73808453, -0.71870305, -0.00738594, 0.71486172,
- 1.73005326, -1.66891897, -0.73689615, -0.00616203, 0.74262409, 1.73807899, -2.92417482, -1.73866741, -0.78133871, 0.00764425,
- 0.80027264, 1.78668732, 2.74992588, -4.00000000, -2.75578740, -1.83697516, -0.83117035, -0.00355191, 0.83527172, 1.82814700,
- 2.77377675, 3.80718693, -3.81667698, -2.83575471, -1.83372350, -0.86579471, 0.00547578, 0.87582281, 1.82858793, 2.87265007,
- 3.91405377, -4.87521600, -3.78999094, -2.86437014, -1.86964365, -0.90618018, 0.00128243, 0.91497811, 1.87374952, 2.83199819,
- 3.91519130, 4.76632822, -6.68713448, -6.01252467, -4.94587936, -3.88795368, -2.91299088, -1.92592211, -0.95504570, -0.00089980,
- 0.94565200, 1.93239633, 2.91832808, 3.91363475, 4.88920034, 5.96471415, 6.83905252, 7.86195009, 8.81571018, -12.96141759,
--11.73039516, -10.96459719, -9.97382433, -9.04414433, -7.89460619, -6.96628608, -5.93236595, -4.93337924, -3.95479990, -2.96451499,
--1.96635876, -0.97271229, -0.00402238, 0.98343930, 1.98348291, 2.96641164, 3.95456471, 4.95517089, 5.98975714, 6.90322073,
- 7.90468849, 8.85639467, 9.97255498, 10.79006309, 11.81988596, 0.04950500, -1.00000000, -0.01226628, 1.00000000, -0.59479469,
--0.10438305, 0.59822144, -2.00000000, -0.67109149, -0.09256692, 0.65171621, 2.00000000, -3.00000000, -1.68391999, -0.76681039,
--0.03354151, 0.71509146, 1.77615472, -2.00000000, -0.68661511, -0.02497881, 0.66478398, 2.00000000, -2.00000000, -0.67032784,
--0.00920582, 0.64892756, 2.00000000, -2.00000000, -0.68561894, 0.03641869, 0.73021611, 1.68293863, -4.00000000, -2.72024184,
--1.80096059, -0.81696185, 0.03604685, 0.79232033, 1.70070730, 3.00000000, -4.00000000, -2.71795670, -1.80482986, -0.86001162,
- 0.03764903, 0.87723968, 1.79970771, 2.72685932, 3.67589143, -5.00000000, -4.00000000, -2.85492548, -1.78996365, -0.83250358,
--0.01376828, 0.84195506, 1.78161105, 2.76754458, 4.00000000, -6.00000000, -5.00000000, -3.82268811, -2.77563624, -1.82608163,
--0.86486114, -0.02671886, 0.86693165, 1.88422879, 2.86248347, 3.95632216, -7.00000000, -6.00000000, -5.00000000, -3.77533988,
--2.86391432, -1.87052039, -0.90513658, 0.06271236, 0.91083620, 1.85734756, 2.86031688, 3.82019418, 4.94420394, 6.00000000,
--11.00000000, -10.00000000, -9.00000000, -8.00000000, -6.91952415, -6.00000000, -4.92044374, -3.87845165, -2.87392362, -1.88413020,
--0.91915740, 0.00318517, 0.91602800, 1.89664838, 2.88925058, 3.84123856, 4.78988651, 5.94526812, 6.81953917, 8.00000000,
--9.00000000, -8.00000000, -7.03319143, -5.94530963, -4.86669720, -3.92438007, -2.88620396, -1.92848070, -0.94365985, 0.01671855,
- 0.97349410, 1.93419878, 2.89740109, 3.89662823, 4.83235583, 5.88106535, 6.80328232, 8.00000000, -13.00000000, -12.00000000,
--11.00000000, -10.00000000, -9.00000000, -7.86033489, -6.83344055, -5.89844215, -4.90811454, -3.94841298, -2.95820490, -1.98627966,
--0.99161468, -0.02286136, 0.96055651, 1.95052433, 2.93969396, 3.94304346, 4.88522624, 5.87434241, 6.78309433, 7.87244101,
- 9.00000000, 10.00000000, -12.09117356, -11.00000000, -10.00000000, -8.84766108, -7.86934236, -6.98544896, -5.94233429, -4.95583292,
--3.95575986, -2.97085529, -1.98955811, -0.99359873, -0.00485413, 0.98298870, 1.98093258, 2.96430203, 3.95540216, 4.96915010,
- 5.96775124, 6.99236918, 7.96503302, 8.99864542, 9.85857723, 10.96541926, 11.91647197, 12.71060069, -26.00000000, -25.00000000,
--24.00585596, -23.11642573, -22.14271284, -20.89800711, -19.87815799, -19.05036354, -17.88555651, -16.86471209, -15.97711073, -14.94012359,
--14.02661226, -12.98243228, -11.97489256, -10.97402777, -9.96425624, -9.01085220, -7.97372506, -6.98795002, -5.97271328, -5.00191694,
--3.98055849, -2.98458048, -1.99470442, -0.99656768, -0.00825666, 1.00272004, 1.99922218, 2.99357669, 4.01407905, 5.01003897,
- 5.98115528, 7.00018958, 8.00338125, 8.98981046, 9.98990318, 10.96341479, 11.96866930, 12.99175139, 13.94580443, 14.95745083,
- 15.98992869, 16.97484646, 17.99630043, 18.93396897, 19.88347741, 20.96532482, 21.92191032, 23.22314702, 0.00006846, 0.00014352,
--0.00056203, 0.00027588, -0.00147678, 1.00000000, 0.00003823, 0.00001975, -0.00033710, -0.00096712, 1.00000000, -1.00000000,
- 0.00067511, -1.00000000, 0.00342065, 1.00000000, -1.00000000, 0.00196254, 1.00000000, -1.00000000, 0.00201173, 1.00000000,
--2.00000000, -1.00000000, -0.00381686, 1.00000000, -1.00000000, 0.00178037, 1.00000000, -2.00000000, -1.00000000, -0.00320274,
- 1.00000000, 2.00000000, -2.00000000, -1.00000000, 0.00426519, 1.00000000, 2.00000000, -3.00000000, -2.00000000, -1.00000000,
--0.00074072, 0.64654602, 2.00000000, 0.00031217, 0.00063348, 0.00020247, 0.00047891, 0.00122893, -0.00150669, -0.00148276,
- 0.00016848, 0.00147085, 1.00000000, -0.00088160, 1.00000000, -1.00000000, 0.00381641, 1.00000000, -1.00000000, 0.00129816,
- 1.00000000, -1.00000000, 0.00074903, 1.00000000, -2.00000000, -0.76230566, -0.00370764, 0.82467977, -0.78769346, -0.00492670,
- 0.84532630, -2.00000000, -0.70943195, -0.01257613, 0.75905385, 2.00000000, -2.00000000, -0.62780445, -0.00408633, 0.60272506,
- 2.00000000, 3.00000000, -3.00000000, -2.00000000, -0.61412985, 0.00102833, 0.61527589, 2.00000000, 3.00000000, 0.00012115,
--0.00080909, 0.00071061, -0.00227957, 0.00179794, 0.00103827, -1.00000000, 0.00444757, -1.00000000, 0.00604068, 1.00000000,
--1.00000000, 0.00427327, 1.00000000, -1.00000000, 0.00086662, 1.00000000, -1.00000000, -0.00837492, 1.00000000, -0.65715934,
--0.00645342, 0.64004630, -2.00000000, -0.64987682, -0.01449567, 0.69893373, -2.00000000, -0.63221961, 0.00421765, 0.62452105,
- 2.00000000, -2.00000000, -0.60027006, -0.00110630, 0.62033821, 2.00000000, -2.00000000, -0.59823932, 0.00928313, 0.62188520,
- 2.00000000, 3.00000000, -3.00000000, -2.00000000, -0.63230286, -0.00248555, 0.62632575, 2.00000000, 3.00000000, -5.00000000,
--4.00000000, -3.00000000, -2.00000000, -0.66521143, 0.00544305, 0.66930486, 2.00000000, 3.00000000, 4.00000000, 0.00077008,
- 0.00061140, -0.00009317, -0.00049643, 1.00000000, -1.00000000, -0.00285084, 1.00000000, -1.00000000, 0.00601784, 1.00000000,
--1.00000000, -0.00091887, 0.75122772, -0.71579859, -0.00043545, 1.00000000, -0.85571363, -0.00227654, 0.63816873, -1.00000000,
--0.00393484, 0.76748004, -0.58223659, -0.01229777, 0.58080322, -0.61945902, -0.00232238, 0.62277938, 2.00000000, -2.00000000,
--0.60595489, -0.00535702, 0.60547736, 2.00000000, -4.00000000, -3.00000000, -2.00000000, -0.62368122, 0.01112097, 0.63997294,
- 2.00000000, 3.00000000, -3.00000000, -2.00000000, -0.64318217, 0.00515139, 0.64781184, 2.00000000, -3.00000000, -1.78031579,
--0.67122588, 0.02153711, 0.67899877, 2.00000000, 3.00000000, 4.00000000, -4.00000000, -3.00000000, -1.80503233, -0.69835727,
--0.00270770, 0.70999554, 1.77332849, 3.00000000, 4.00000000, 5.00000000, -8.00000000, -7.00000000, -6.00000000, -5.00000000,
--4.00000000, -2.81600693, -1.72970368, -0.73779413, -0.01384841, 0.75694606, 1.80042618, 3.00000000, 4.00000000, 5.00000000,
- 6.00000000, -0.00051787, 0.00059593, -0.00023319, -1.00000000, 0.00191861, 0.79547197, -0.75020995, 0.00217840, 0.69165833,
--1.00000000, -0.00304964, 0.67698951, -0.64516943, -0.00657667, 0.59260129, -0.62819301, -0.00456626, 0.59426260, -0.60909519,
- 0.00256476, 0.61660408, -0.66560131, -0.00293463, 0.67477566, 2.00000000, -2.00000000, -0.62484067, 0.00505116, 0.63491494,
- 2.00000000, -3.00000000, -2.00000000, -0.68427246, 0.00924353, 0.68755774, 2.00000000, 3.00000000, -3.00000000, -2.00000000,
--0.65390928, 0.01008025, 0.65849449, 2.00000000, 3.00000000, -5.00000000, -4.00000000, -3.00000000, -1.70848232, -0.72079538,
--0.00007674, 0.71556176, 1.76815351, 3.00000000, 4.00000000, 5.00000000, -4.00000000, -3.00000000, -1.82887466, -0.73529886,
- 0.00033458, 0.73847588, 1.83009515, 3.00000000, 4.00000000, -5.00000000, -4.00000000, -2.83203553, -1.79500085, -0.77452749,
--0.00614320, 0.77416943, 1.82469471, 2.77034612, 4.00000000, 5.00000000, -7.00000000, -6.00000000, -5.00000000, -4.00000000,
--2.76574798, -1.84700836, -0.80822297, 0.00054165, 0.80901445, 1.85687331, 2.75680191, 3.81986695, 5.00000000, 6.00000000,
- 7.00000000, 8.00000000, -13.00000000, -12.00000000, -11.00000000, -10.00000000, -9.00000000, -8.00000000, -7.00000000, -6.00000000,
--5.00000000, -3.88304817, -2.93396067, -1.86645989, -0.84825410, 0.00666207, 0.84853252, 1.88634684, 2.95282618, 3.89813287,
- 4.89189079, 6.00000000, 7.00000000, 8.00000000, 9.00000000, 10.00000000, 11.00000000, -0.00344877, 1.00000000, -0.61413659,
--0.02115630, 0.59438887, -0.60873054, 0.00844993, 0.62510557, -2.00000000, -0.75002947, 0.00120913, 0.66616051, -2.00000000,
--0.72324691, 0.04760499, 0.70532533, 2.00000000, -3.00000000, -1.66577589, -0.78941380, -0.01909714, 0.74993685, 1.70945570,
--1.64422308, -0.70992006, -0.02795108, 0.76990363, 1.79682243, 2.96233315, -1.71686461, -0.76572785, -0.00041846, 0.78174132,
- 1.66217596, -3.00000000, -1.77033369, -0.79475091, 0.03709740, 0.80097076, 1.83947400, 2.85879773, -4.00000000, -3.16528651,
--1.79564411, -0.90078981, 0.02403102, 0.86138856, 1.84207433, 2.74584048, -4.00000000, -2.91249347, -1.87804769, -0.87323549,
- 0.08164382, 0.89037056, 1.82505263, 2.71336163, 4.00000000, -4.81262228, -3.87173565, -2.83424209, -1.87517938, -0.86199960,
- 0.00268598, 0.89547657, 1.90713511, 2.85219071, 3.86417171, 4.80711781, 6.00000000, 7.00000000, -5.00000000, -3.82388480,
--2.82875808, -1.90350457, -0.90795818, 0.03047007, 0.93676836, 1.88844957, 2.83269711, 3.76109686, 5.00000000, -9.00000000,
--8.00000000, -6.88037957, -5.88776398, -4.91209139, -3.93902541, -2.90989221, -1.92281230, -0.98960535, -0.07440511, 0.94023957,
- 1.91666262, 2.83340828, 3.83651295, 4.77839424, 6.12284019, 7.00000000, 8.00000000, 9.00000000, -12.00000000, -11.00000000,
--10.00000000, -9.00000000, -8.00000000, -6.68554513, -5.97994708, -4.98789075, -3.91383581, -2.92952795, -1.91727195, -0.93148075,
--0.00568870, 0.93515148, 1.94580068, 2.93838956, 3.92567644, 4.96573603, 5.95402763, 7.00000000, 8.00000000, 9.00000000,
--11.00000000, -9.90096030, -8.97868124, -7.93663988, -6.98806055, -5.95937864, -4.93473664, -3.95454756, -2.96518446, -1.97711766,
--0.98552111, -0.03317271, 0.95115775, 1.93785086, 2.96310779, 3.93322450, 5.01716212, 5.85909823, 6.89163669, 7.97492693,
- 8.85698897, 9.79802946, 11.09373957, 12.00000000, -13.00000000, -12.00000000, -10.67579109, -9.95079100, -8.90576592, -7.93254656,
--6.96112672, -5.96015798, -4.95493809, -3.98556269, -2.98182856, -1.98150255, -0.96551153, -0.00399791, 0.98644875, 1.98043830,
- 2.97969033, 3.97728257, 4.95173541, 5.95649050, 6.96447378, 7.95591513, 9.07680954, 9.92093070, 10.76496555, 11.97525735,
- 13.00000000, 14.00000000, -25.00000000, -24.00000000, -23.00000000, -22.00072357, -21.00000000, -20.00000000, -19.00000000, -18.20003462,
--17.01648407, -15.78651996, -14.95660266, -13.99167850, -13.28722978, -11.85013840, -10.92025302, -9.87055810, -8.93841040, -7.95329867,
--6.97819441, -6.01593394, -5.00905213, -3.99905285, -2.99171810, -1.99062796, -1.00112466, 0.00140492, 1.00701091, 2.02327185,
- 3.00194633, 3.99188294, 5.00313145, 6.00448038, 6.98904951, 7.98158293, 8.98212774, 10.00363404, 10.98641678, 11.98034311,
- 12.95176779, 13.95383703, 14.99084578, 15.98600642, 16.99406826, 17.98134623, 19.01793961, 19.86072639, 20.88465474, 21.99287082,
- 22.81916620, 23.77946383, 0.00000234, 0.00000298, 0.00000048, 0.00002408, -0.00000165, -0.00001831, -0.00005703, -0.00000184,
--1.00000000, 0.00001977, 1.00000000, -1.00000000, 0.00000010, 1.00000000, -1.00000000, -0.00001152, 1.00000000, -1.00000000,
- 0.00000840, 1.00000000, -1.00000000, 0.00002353, 1.00000000, -0.75455603, -0.00001433, 1.00000000, -0.65859705, -0.00000703,
- 0.62995860, -2.00000000, -0.72724652, -0.00033969, 0.61359174, 2.00000000, -2.00000000, -0.69510998, -0.00031410, 0.66467605,
- 2.00000000, 3.00000000, -3.00000000, -2.00000000, -0.65738683, 0.00039019, 0.66554720, 1.91774106, 3.18089124, 0.00000070,
- 0.00001152, -0.00000795, -0.00000058, -0.00003502, -0.00001508, -0.00004225, -0.00002165, -1.00000000, 0.00004391, 1.00000000,
--1.00000000, 0.00001784, 1.00000000, -1.00000000, -0.00003678, 1.00000000, -0.68878314, -0.00013166, 0.60880149, -0.75291978,
- 0.00006493, 1.00000000, -0.76757316, 0.00003057, 0.67140524, -0.61602267, -0.00014495, 0.63625803, 2.00000000, -2.00000000,
--0.61253314, -0.00116483, 0.65071851, 2.00000000, -3.00000000, -1.71451667, -0.67799909, -0.00048294, 0.65846019, 2.00000000,
--3.02497593, -1.83515395, -0.70317981, 0.00519701, 0.67780009, 1.84218153, 2.88846262, 4.00000000, 0.00001124, 0.00000588,
--0.00000172, 0.00002835, 1.00000000, 0.00001012, -0.00008644, 1.00000000, -0.75115901, 0.00004347, 1.00000000, -1.00000000,
- 0.00002800, 1.00000000, -1.00000000, -0.00006039, 1.00000000, -0.79763258, -0.00011907, 0.71713616, -0.76791870, -0.00007113,
- 0.63583609, -0.62337806, 0.00012891, 0.62242094, -0.60837055, 0.00043216, 0.65515705, -0.63637782, -0.00019749, 0.60423967,
- 2.00000000, -2.00000000, -0.65404827, -0.00089304, 0.64706660, 2.00000000, -1.86334076, -0.66410366, 0.00063219, 0.66968004,
- 2.00000000, 3.00000000, 4.00000000, -4.00000000, -3.00000000, -1.79048834, -0.69451890, 0.00030677, 0.71009333, 1.70591343,
- 3.00000000, 4.00000000, -4.00000000, -2.90176499, -1.78368781, -0.74425178, 0.00234068, 0.74847325, 1.78886822, 2.78478854,
- 3.83608985, 4.95996151, 0.00002170, 0.00001281, 0.00002162, -1.00000000, -0.00007266, 1.00000000, -1.00000000, -0.00003250,
--0.64088804, 0.00015239, 1.00000000, -0.58450370, -0.00008410, 0.60567186, -1.00000000, -0.00010752, 1.00000000, -0.58922508,
--0.00017378, 0.60755779, -0.62797206, -0.00001016, 0.64432847, -0.58497934, -0.00001851, 0.59716791, -0.62642499, -0.00097386,
- 0.63568558, 2.00000000, -2.00000000, -0.63236390, -0.00173361, 0.63142762, 1.75629192, -3.00000000, -2.00000000, -0.65596684,
- 0.00209364, 0.65419742, 2.00000000, -3.00000000, -1.73856625, -0.67767521, -0.00119512, 0.68973603, 1.70985573, 3.00000000,
--3.00000000, -1.81820220, -0.73974134, 0.00695869, 0.72216179, 1.75624461, 3.00000000, 4.00000000, -5.00000000, -4.00000000,
--3.17718593, -1.76857567, -0.76822322, 0.00267400, 0.76414602, 1.84309221, 3.04940652, 4.00000000, -7.08189123, -6.00000000,
--5.22882249, -3.96477958, -2.79653492, -1.81923435, -0.80050253, -0.01086663, 0.82708565, 1.85804900, 2.89996354, 3.76028554,
- 4.80518081, 5.81738096, 7.00000000, 8.00000000, 9.08816091, -0.00002979, -0.00000333, -1.00000000, -0.00011532, 1.00000000,
--0.70921122, -0.00005325, 0.68933188, -0.67581263, -0.00023107, 0.57868212, -0.58388312, -0.00020850, 0.60149012, -0.60912457,
- 0.00001567, 0.60180554, -0.59130091, -0.00038863, 0.59908653, -2.00000000, -0.63697707, 0.00083913, 0.62040514, 2.00000000,
--2.00000000, -0.63216238, -0.00081100, 0.64411071, 2.00000000, -1.76856259, -0.65266989, -0.00243486, 0.66888899, 2.00000000,
--1.75427214, -0.71415385, -0.00226376, 0.71296778, 1.66182947, 3.00000000, -3.00000000, -1.72505821, -0.72920134, -0.00360424,
- 0.73800767, 1.72848281, 3.00000000, -4.00000000, -2.95284408, -1.72025758, -0.76503859, 0.00418761, 0.75297139, 1.73959808,
- 3.00000000, -5.00000000, -3.96232791, -2.74080544, -1.78897123, -0.80233505, -0.00002050, 0.79693417, 1.76182598, 2.78434458,
- 3.85693287, 5.00000000, -6.00000000, -4.78439284, -3.83501790, -2.85203629, -1.84909573, -0.85382658, -0.00181019, 0.84735145,
- 1.83676575, 2.83656843, 3.86722376, 4.79702431, 6.00000000, -9.00000000, -8.00000000, -7.00000000, -6.07957292, -4.84677515,
--3.85093972, -2.88683139, -1.84596391, -0.88058034, -0.00008692, 0.87554746, 1.86933183, 2.84729990, 3.89029797, 4.87311773,
- 5.90844023, 7.00000000, -11.00000000, -9.97745420, -8.90015761, -7.94187517, -6.86987726, -5.84795335, -4.86693435, -3.90601819,
--2.91031804, -1.91620096, -0.90497055, 0.00659199, 0.90926869, 1.90980821, 2.91070850, 3.93685967, 4.85581177, 6.06727337,
- 7.05801043, 8.00000000, 9.00000000, 10.00000000, 10.90825787, 12.00000000, 13.00000000, 14.00000000, -0.00008918, 1.00000000,
--0.54405938, 0.00120348, 0.55781920, -0.59227786, -0.00349602, 0.59777231, -1.63717598, -0.69048065, 0.00999281, 0.65770558,
- 2.00000000, -2.00000000, -0.71013571, 0.00454518, 0.66991065, -3.00000000, -1.73004867, -0.73743921, 0.01162454, 0.69964842,
- 1.83319587, -1.81225491, -0.76806000, 0.00164742, 0.76780397, 1.67168896, -1.64564794, -0.79903361, -0.01522880, 0.84277926,
- 1.68873752, -3.00000000, -1.72063244, -0.83687428, 0.00246724, 0.84618697, 1.79464483, 2.77447025, -3.77118426, -2.75025539,
--1.82050448, -0.90373722, -0.00187780, 0.90102245, 1.85249394, 2.71364180, -2.71720889, -1.79466125, -0.89860801, -0.02725825,
- 0.90877329, 1.90542096, 2.76847902, 3.71496428, -4.70257302, -3.90746659, -2.87078421, -1.88858709, -0.93608993, -0.02157425,
- 0.95181182, 1.91155682, 2.83614575, 3.87820801, 4.72172277, -5.02764544, -3.80066801, -2.87484378, -1.90707477, -0.96326017,
--0.01060091, 0.96558851, 1.92191548, 2.86970759, 3.85655474, 4.83135970, 5.76387469, -9.00000000, -8.00000000, -6.75261776,
--5.86333393, -4.84846871, -3.91871758, -2.93827286, -1.93050320, -0.96359634, -0.00141931, 0.95926312, 1.92541870, 2.93009411,
- 3.86699087, 4.82315929, 5.67815206, -8.76594345, -7.70350451, -6.91784020, -5.81539490, -4.92526872, -3.91513203, -2.92134949,
--1.95465646, -0.97638102, -0.00742564, 0.96948714, 1.96401112, 2.95256722, 3.93146353, 4.90991357, 5.88139022, 6.88640588,
- 7.82610489, 9.00000000, -10.97611369, -9.80036760, -8.91109518, -7.92809404, -6.93865353, -5.91965899, -4.92957669, -3.95206224,
--2.97308718, -1.97778214, -0.98552568, -0.00063212, 0.98686014, 1.97511867, 2.97114218, 3.97854244, 4.96578513, 5.96457765,
- 6.95180187, 7.95163483, 8.93760897, 9.87666900, 10.88024562, 11.96270158, 12.99519291, -15.00000000, -13.76826291, -12.97229116,
--12.00334834, -10.95980884, -9.98190891, -8.93798503, -7.95621309, -6.96109479, -5.96056649, -4.95843419, -3.97688640, -2.98989576,
--1.98533395, -0.99580972, 0.00694370, 0.99421120, 1.99033132, 2.98751217, 3.98549580, 4.96482394, 5.96623233, 6.93564626,
- 7.93772467, 8.92015276, 9.88785129, 10.97606096, 11.79686057, -23.00000000, -22.00000000, -21.00000000, -20.00000000, -19.00000000,
--17.73310977, -16.83574096, -15.90889480, -15.00437366, -13.95007272, -12.99296117, -11.98334751, -10.96970820, -9.97775151, -8.98193840,
--7.98378966, -6.98887770, -5.99059477, -5.00228769, -3.99355850, -2.99947486, -1.99897483, -0.99375857, 0.00324880, 1.00215912,
- 1.99277083, 3.00503747, 3.99390482, 4.98854283, 5.98753219, 6.98245347, 7.98089893, 8.95960522, 9.95663648, 11.00810285,
- 12.01421617, 12.96208687, 13.99227766, 14.97230040, 15.95114804, 16.97347393, 17.97794884, 18.96777118, 19.94446034, 20.94799029,
- 22.14740083, 22.84288347, 23.99212109, 25.00000000, 25.96562658};
-
-/* cdf tables for quantizer indices */
-const WebRtc_UWord16 WebRtcIsac_kQKltCdfGain[1212] = {
- 0, 13, 301, 3730, 61784, 65167, 65489, 65535, 0, 17,
- 142, 314, 929, 2466, 7678, 56450, 63463, 64740, 65204, 65426,
- 65527, 65535, 0, 8, 100, 724, 6301, 60105, 65125, 65510,
- 65531, 65535, 0, 13, 117, 368, 1068, 3010, 11928, 53603,
- 61177, 63404, 64505, 65108, 65422, 65502, 65531, 65535, 0, 4,
- 17, 96, 410, 1859, 12125, 54361, 64103, 65305, 65497, 65535,
- 0, 4, 88, 230, 469, 950, 1746, 3228, 6092, 16592,
- 44756, 56848, 61256, 63308, 64325, 64920, 65309, 65460, 65502, 65522,
- 65535, 0, 88, 352, 1675, 6339, 20749, 46686, 59284, 63525,
- 64949, 65359, 65502, 65527, 65535, 0, 13, 38, 63, 117,
- 234, 381, 641, 929, 1407, 2043, 2809, 4032, 5753, 8792,
- 14407, 24308, 38941, 48947, 55403, 59293, 61411, 62688, 63630, 64329,
- 64840, 65188, 65376, 65472, 65506, 65527, 65531, 65535, 0, 8,
- 29, 75, 222, 615, 1327, 2801, 5623, 9931, 16094, 24966,
- 34419, 43458, 50676, 56186, 60055, 62500, 63936, 64765, 65225, 65435,
- 65514, 65535, 0, 8, 13, 15, 17, 21, 33, 59,
- 71, 92, 151, 243, 360, 456, 674, 934, 1223, 1583,
- 1989, 2504, 3031, 3617, 4354, 5154, 6163, 7411, 8780, 10747,
- 12874, 15591, 18974, 23027, 27436, 32020, 36948, 41830, 46205, 49797,
- 53042, 56094, 58418, 60360, 61763, 62818, 63559, 64103, 64509, 64798,
- 65045, 65162, 65288, 65363, 65447, 65506, 65522, 65531, 65533, 65535,
- 0, 4, 6, 25, 38, 71, 138, 264, 519, 808,
- 1227, 1825, 2516, 3408, 4279, 5560, 7092, 9197, 11420, 14108,
- 16947, 20300, 23926, 27459, 31164, 34827, 38575, 42178, 45540, 48747,
- 51444, 54090, 56426, 58460, 60080, 61595, 62734, 63668, 64275, 64673,
- 64936, 65112, 65217, 65334, 65426, 65464, 65477, 65489, 65518, 65527,
- 65529, 65531, 65533, 65535, 0, 2, 4, 8, 10, 12,
- 14, 16, 21, 33, 50, 71, 84, 92, 105, 138,
- 180, 255, 318, 377, 435, 473, 511, 590, 682, 758,
- 913, 1097, 1256, 1449, 1671, 1884, 2169, 2445, 2772, 3157,
- 3563, 3944, 4375, 4848, 5334, 5820, 6448, 7101, 7716, 8378,
- 9102, 9956, 10752, 11648, 12707, 13670, 14758, 15910, 17187, 18472,
- 19627, 20649, 21951, 23169, 24283, 25552, 26862, 28227, 29391, 30764,
- 31882, 33213, 34432, 35600, 36910, 38116, 39464, 40729, 41872, 43144,
- 44371, 45514, 46762, 47813, 48968, 50069, 51032, 51974, 52908, 53737,
- 54603, 55445, 56282, 56990, 57572, 58191, 58840, 59410, 59887, 60264,
- 60607, 60946, 61269, 61516, 61771, 61960, 62198, 62408, 62558, 62776,
- 62985, 63207, 63408, 63546, 63739, 63906, 64070, 64237, 64371, 64551,
- 64677, 64836, 64999, 65095, 65213, 65284, 65338, 65380, 65426, 65447,
- 65472, 65485, 65487, 65489, 65502, 65510, 65512, 65514, 65516, 65518,
- 65522, 65531, 65533, 65535, 0, 2, 4, 6, 65528, 65531,
- 65533, 65535, 0, 2, 4, 6, 8, 10, 222, 65321,
- 65513, 65528, 65531, 65533, 65535, 0, 2, 4, 50, 65476,
- 65529, 65531, 65533, 65535, 0, 2, 4, 6, 8, 12,
- 38, 544, 64936, 65509, 65523, 65525, 65529, 65531, 65533, 65535,
- 0, 2, 4, 6, 8, 10, 1055, 64508, 65528, 65531,
- 65533, 65535, 0, 2, 4, 6, 8, 10, 12, 123,
- 3956, 62999, 65372, 65495, 65515, 65521, 65523, 65525, 65527, 65529,
- 65531, 65533, 65535, 0, 2, 4, 12, 53, 4707, 59445,
- 65467, 65525, 65527, 65529, 65531, 65533, 65535, 0, 2, 4,
- 6, 8, 10, 12, 14, 16, 38, 40, 50, 67,
- 96, 234, 929, 14345, 55750, 64866, 65389, 65462, 65514, 65517,
- 65519, 65521, 65523, 65525, 65527, 65529, 65531, 65533, 65535, 0,
- 2, 4, 6, 8, 10, 15, 35, 91, 377, 1946,
- 13618, 52565, 63714, 65184, 65465, 65520, 65523, 65525, 65527, 65529,
- 65531, 65533, 65535, 0, 2, 4, 6, 8, 10, 12,
- 14, 16, 18, 20, 22, 24, 26, 28, 30, 32,
- 34, 36, 38, 40, 42, 44, 46, 48, 50, 52,
- 54, 82, 149, 362, 751, 1701, 4239, 12893, 38627, 55072,
- 60875, 63071, 64158, 64702, 65096, 65283, 65412, 65473, 65494, 65505,
- 65508, 65517, 65519, 65521, 65523, 65525, 65527, 65529, 65531, 65533,
- 65535, 0, 2, 15, 23, 53, 143, 260, 418, 698,
- 988, 1353, 1812, 2411, 3144, 4015, 5143, 6401, 7611, 8999,
- 10653, 12512, 14636, 16865, 19404, 22154, 24798, 27521, 30326, 33102,
- 35790, 38603, 41415, 43968, 46771, 49435, 52152, 54715, 57143, 59481,
- 61178, 62507, 63603, 64489, 64997, 65257, 65427, 65473, 65503, 65520,
- 65529, 65531, 65533, 65535, 0, 3, 6, 9, 26, 32,
- 44, 46, 64, 94, 111, 164, 205, 254, 327, 409,
- 506, 608, 733, 885, 1093, 1292, 1482, 1742, 1993, 2329,
- 2615, 3029, 3374, 3798, 4257, 4870, 5405, 5992, 6618, 7225,
- 7816, 8418, 9051, 9761, 10532, 11380, 12113, 13010, 13788, 14594,
- 15455, 16361, 17182, 18088, 18997, 20046, 20951, 21968, 22947, 24124,
- 25296, 26547, 27712, 28775, 29807, 30835, 31709, 32469, 33201, 34014,
- 34876, 35773, 36696, 37620, 38558, 39547, 40406, 41277, 42367, 43290,
- 44445, 45443, 46510, 47684, 48973, 50157, 51187, 52242, 53209, 54083,
- 55006, 55871, 56618, 57293, 57965, 58556, 59222, 59722, 60180, 60554,
- 60902, 61250, 61554, 61837, 62100, 62372, 62631, 62856, 63078, 63324,
- 63557, 63768, 63961, 64089, 64235, 64352, 64501, 64633, 64770, 64887,
- 65001, 65059, 65121, 65188, 65246, 65302, 65346, 65390, 65428, 65463,
- 65477, 65506, 65515, 65517, 65519, 65521, 65523, 65525, 65527, 65529,
- 65531, 65533, 65535, 0, 2, 4, 109, 65332, 65531, 65533,
- 65535, 0, 2, 4, 6, 8, 25, 1817, 63874, 65511,
- 65527, 65529, 65531, 65533, 65535, 0, 2, 4, 907, 65014,
- 65529, 65531, 65533, 65535, 0, 2, 4, 6, 8, 10,
- 12, 132, 2743, 62708, 65430, 65525, 65527, 65529, 65531, 65533,
- 65535, 0, 2, 4, 6, 8, 35, 3743, 61666, 65485,
- 65531, 65533, 65535, 0, 2, 4, 6, 8, 10, 23,
- 109, 683, 6905, 58417, 64911, 65398, 65497, 65518, 65525, 65527,
- 65529, 65531, 65533, 65535, 0, 2, 4, 6, 53, 510,
- 10209, 55212, 64573, 65441, 65522, 65529, 65531, 65533, 65535, 0,
- 2, 4, 6, 8, 10, 12, 14, 16, 18, 20,
- 22, 32, 90, 266, 1037, 3349, 14468, 50488, 62394, 64685,
- 65341, 65480, 65514, 65519, 65521, 65523, 65525, 65527, 65529, 65531,
- 65533, 65535, 0, 2, 4, 6, 9, 16, 37, 106,
- 296, 748, 1868, 5733, 18897, 45553, 60165, 63949, 64926, 65314,
- 65441, 65508, 65524, 65529, 65531, 65533, 65535, 0, 2, 4,
- 6, 8, 10, 12, 14, 16, 18, 20, 22, 24,
- 26, 28, 30, 32, 34, 36, 38, 40, 42, 44,
- 46, 48, 50, 83, 175, 344, 667, 1293, 2337, 4357,
- 8033, 14988, 28600, 43244, 52011, 57042, 59980, 61779, 63065, 63869,
- 64390, 64753, 64988, 65164, 65326, 65422, 65462, 65492, 65506, 65522,
- 65524, 65526, 65531, 65533, 65535, 0, 2, 4, 6, 8,
- 10, 12, 14, 16, 25, 39, 48, 55, 62, 65,
- 85, 106, 139, 169, 194, 252, 323, 485, 688, 1074,
- 1600, 2544, 3863, 5733, 8303, 11397, 15529, 20273, 25734, 31455,
- 36853, 41891, 46410, 50306, 53702, 56503, 58673, 60479, 61880, 62989,
- 63748, 64404, 64852, 65124, 65309, 65424, 65480, 65524, 65528, 65533,
- 65535, 0, 2, 4, 6, 8, 10, 12, 14, 21,
- 23, 25, 27, 29, 31, 39, 41, 43, 48, 60,
- 72, 79, 106, 136, 166, 187, 224, 252, 323, 381,
- 427, 478, 568, 660, 783, 912, 1046, 1175, 1365, 1567,
- 1768, 2024, 2347, 2659, 3049, 3529, 4033, 4623, 5281, 5925,
- 6726, 7526, 8417, 9468, 10783, 12141, 13571, 15222, 16916, 18659,
- 20350, 22020, 23725, 25497, 27201, 29026, 30867, 32632, 34323, 36062,
- 37829, 39466, 41144, 42654, 43981, 45343, 46579, 47759, 49013, 50171,
- 51249, 52283, 53245, 54148, 54938, 55669, 56421, 57109, 57791, 58464,
- 59092, 59674, 60105, 60653, 61083, 61407, 61757, 62095, 62388, 62649,
- 62873, 63157, 63358, 63540, 63725, 63884, 64046, 64155, 64278, 64426,
- 64548, 64654, 64806, 64906, 64994, 65077, 65137, 65215, 65277, 65324,
- 65354, 65409, 65437, 65455, 65462, 65490, 65495, 65499, 65508, 65511,
- 65513, 65515, 65517, 65519, 65521, 65523, 65525, 65527, 65529, 65531,
- 65533, 65535};
-
-const WebRtc_UWord16 WebRtcIsac_kQKltCdfShape[2059] = {
- 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535,
- 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 4,
- 65535, 0, 8, 65514, 65535, 0, 29, 65481, 65535, 0,
- 121, 65439, 65535, 0, 239, 65284, 65535, 0, 8, 779,
- 64999, 65527, 65535, 0, 8, 888, 64693, 65522, 65535, 0,
- 29, 2604, 62843, 65497, 65531, 65535, 0, 25, 176, 4576,
- 61164, 65275, 65527, 65535, 0, 65535, 0, 65535, 0, 65535,
- 0, 65535, 0, 4, 65535, 0, 65535, 0, 65535, 0,
- 65535, 0, 65535, 0, 4, 65535, 0, 33, 65502, 65535,
- 0, 54, 65481, 65535, 0, 251, 65309, 65535, 0, 611,
- 65074, 65535, 0, 1273, 64292, 65527, 65535, 0, 4, 1809,
- 63940, 65518, 65535, 0, 88, 4392, 60603, 65426, 65531, 65535,
- 0, 25, 419, 7046, 57756, 64961, 65514, 65531, 65535, 0,
- 65535, 0, 65535, 0, 65535, 0, 65535, 0, 4, 65531,
- 65535, 0, 65535, 0, 8, 65531, 65535, 0, 4, 65527,
- 65535, 0, 17, 65510, 65535, 0, 42, 65481, 65535, 0,
- 197, 65342, 65531, 65535, 0, 385, 65154, 65535, 0, 1005,
- 64522, 65535, 0, 8, 1985, 63469, 65533, 65535, 0, 38,
- 3119, 61884, 65514, 65535, 0, 4, 6, 67, 4961, 60804,
- 65472, 65535, 0, 17, 565, 9182, 56538, 65087, 65514, 65535,
- 0, 8, 63, 327, 2118, 14490, 52774, 63839, 65376, 65522,
- 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
- 17, 65522, 65535, 0, 59, 65489, 65535, 0, 50, 65522,
- 65535, 0, 54, 65489, 65535, 0, 310, 65179, 65535, 0,
- 615, 64836, 65535, 0, 4, 1503, 63965, 65535, 0, 2780,
- 63383, 65535, 0, 21, 3919, 61051, 65527, 65535, 0, 84,
- 6674, 59929, 65435, 65535, 0, 4, 255, 7976, 55784, 65150,
- 65518, 65531, 65535, 0, 4, 8, 582, 10726, 53465, 64949,
- 65518, 65535, 0, 29, 339, 3006, 17555, 49517, 62956, 65200,
- 65497, 65531, 65535, 0, 2, 33, 138, 565, 2324, 7670,
- 22089, 45966, 58949, 63479, 64966, 65380, 65518, 65535, 0, 65535,
- 0, 65535, 0, 2, 65533, 65535, 0, 46, 65514, 65535,
- 0, 414, 65091, 65535, 0, 540, 64911, 65535, 0, 419,
- 65162, 65535, 0, 976, 64790, 65535, 0, 2977, 62495, 65531,
- 65535, 0, 4, 3852, 61034, 65527, 65535, 0, 4, 29,
- 6021, 60243, 65468, 65535, 0, 84, 6711, 58066, 65418, 65535,
- 0, 13, 281, 9550, 54917, 65125, 65506, 65535, 0, 2,
- 63, 984, 12108, 52644, 64342, 65435, 65527, 65535, 0, 29,
- 251, 2014, 14871, 47553, 62881, 65229, 65518, 65535, 0, 13,
- 142, 749, 4220, 18497, 45200, 60913, 64823, 65426, 65527, 65535,
- 0, 13, 71, 264, 1176, 3789, 10500, 24480, 43488, 56324,
- 62315, 64493, 65242, 65464, 65514, 65522, 65531, 65535, 0, 4,
- 13, 38, 109, 205, 448, 850, 1708, 3429, 6276, 11371,
- 19221, 29734, 40955, 49391, 55411, 59460, 62102, 63793, 64656, 65150,
- 65401, 65485, 65522, 65531, 65535, 0, 65535, 0, 2, 65533,
- 65535, 0, 1160, 65476, 65535, 0, 2, 6640, 64763, 65533,
- 65535, 0, 2, 38, 9923, 61009, 65527, 65535, 0, 2,
- 4949, 63092, 65533, 65535, 0, 2, 3090, 63398, 65533, 65535,
- 0, 2, 2520, 58744, 65510, 65535, 0, 2, 13, 544,
- 8784, 51403, 65148, 65533, 65535, 0, 2, 25, 1017, 10412,
- 43550, 63651, 65489, 65527, 65535, 0, 2, 4, 29, 783,
- 13377, 52462, 64524, 65495, 65533, 65535, 0, 2, 4, 6,
- 100, 1817, 18451, 52590, 63559, 65376, 65531, 65535, 0, 2,
- 4, 6, 46, 385, 2562, 11225, 37416, 60488, 65026, 65487,
- 65529, 65533, 65535, 0, 2, 4, 6, 8, 10, 12,
- 42, 222, 971, 5221, 19811, 45048, 60312, 64486, 65294, 65474,
- 65525, 65529, 65533, 65535, 0, 2, 4, 8, 71, 167,
- 666, 2533, 7875, 19622, 38082, 54359, 62108, 64633, 65290, 65495,
- 65529, 65533, 65535, 0, 2, 4, 6, 8, 10, 13,
- 109, 586, 1930, 4949, 11600, 22641, 36125, 48312, 56899, 61495,
- 63927, 64932, 65389, 65489, 65518, 65531, 65533, 65535, 0, 4,
- 6, 8, 67, 209, 712, 1838, 4195, 8432, 14432, 22834,
- 31723, 40523, 48139, 53929, 57865, 60657, 62403, 63584, 64363, 64907,
- 65167, 65372, 65472, 65514, 65535, 0, 2, 4, 13, 25,
- 42, 46, 50, 75, 113, 147, 281, 448, 657, 909,
- 1185, 1591, 1976, 2600, 3676, 5317, 7398, 9914, 12941, 16169,
- 19477, 22885, 26464, 29851, 33360, 37228, 41139, 44802, 48654, 52058,
- 55181, 57676, 59581, 61022, 62190, 63107, 63676, 64199, 64547, 64924,
- 65158, 65313, 65430, 65481, 65518, 65535, 0, 65535, 0, 65535,
- 0, 65535, 0, 65535, 0, 65533, 65535, 0, 65535, 0,
- 65535, 0, 65535, 0, 65533, 65535, 0, 2, 65535, 0,
- 2, 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533,
- 65535, 0, 2, 4, 65533, 65535, 0, 2, 65533, 65535,
- 0, 2, 4, 65531, 65533, 65535, 0, 2, 4, 65531,
- 65533, 65535, 0, 2, 4, 6, 65524, 65533, 65535, 0,
- 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
- 65535, 0, 65535, 0, 65535, 0, 65533, 65535, 0, 65533,
- 65535, 0, 2, 65533, 65535, 0, 2, 65533, 65535, 0,
- 2, 65533, 65535, 0, 2, 4, 65532, 65535, 0, 6,
- 65523, 65535, 0, 2, 15, 65530, 65533, 65535, 0, 2,
- 35, 65493, 65531, 65533, 65535, 0, 2, 4, 158, 65382,
- 65531, 65533, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
- 65535, 0, 65535, 0, 65535, 0, 2, 65535, 0, 2,
- 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533, 65535,
- 0, 2, 65533, 65535, 0, 9, 65512, 65535, 0, 2,
- 12, 65529, 65535, 0, 2, 73, 65434, 65533, 65535, 0,
- 2, 240, 65343, 65533, 65535, 0, 2, 476, 65017, 65531,
- 65533, 65535, 0, 2, 4, 1046, 64686, 65531, 65533, 65535,
- 0, 2, 4, 6, 8, 1870, 63898, 65529, 65531, 65533,
- 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65533, 65535,
- 0, 2, 65533, 65535, 0, 2, 65533, 65535, 0, 2,
- 65532, 65535, 0, 6, 65533, 65535, 0, 6, 65523, 65535,
- 0, 2, 65532, 65535, 0, 137, 65439, 65535, 0, 576,
- 64899, 65533, 65535, 0, 2, 289, 65299, 65533, 65535, 0,
- 2, 4, 6, 880, 64134, 65531, 65533, 65535, 0, 2,
- 4, 1853, 63347, 65533, 65535, 0, 2, 6, 2516, 61762,
- 65529, 65531, 65533, 65535, 0, 2, 4, 9, 3980, 61380,
- 65503, 65529, 65531, 65533, 65535, 0, 2, 4, 6, 8,
- 10, 12, 61, 6393, 59859, 65466, 65527, 65529, 65531, 65533,
- 65535, 0, 65535, 0, 65535, 0, 65535, 0, 2, 65532,
- 65535, 0, 3, 65529, 65535, 0, 2, 65529, 65535, 0,
- 61, 65453, 65535, 0, 234, 65313, 65535, 0, 503, 65138,
- 65535, 0, 155, 65402, 65533, 65535, 0, 2, 1058, 64554,
- 65533, 65535, 0, 2, 4, 3138, 62109, 65531, 65533, 65535,
- 0, 2, 4, 2031, 63339, 65531, 65533, 65535, 0, 2,
- 4, 6, 9, 4155, 60778, 65523, 65529, 65531, 65533, 65535,
- 0, 2, 4, 41, 6189, 59269, 65490, 65531, 65533, 65535,
- 0, 2, 4, 6, 210, 8789, 57043, 65400, 65528, 65531,
- 65533, 65535, 0, 2, 4, 6, 8, 26, 453, 10086,
- 55499, 64948, 65483, 65524, 65527, 65529, 65531, 65533, 65535, 0,
- 2, 4, 6, 8, 10, 12, 14, 16, 18, 20,
- 114, 1014, 11202, 52670, 64226, 65356, 65503, 65514, 65523, 65525,
- 65527, 65529, 65531, 65533, 65535, 0, 65533, 65535, 0, 15,
- 65301, 65535, 0, 152, 64807, 65535, 0, 2, 3328, 63308,
- 65535, 0, 2, 4050, 59730, 65533, 65535, 0, 2, 164,
- 10564, 61894, 65529, 65535, 0, 15, 6712, 59831, 65076, 65532,
- 65535, 0, 32, 7712, 57449, 65459, 65535, 0, 2, 210,
- 7849, 53110, 65021, 65523, 65535, 0, 2, 12, 1081, 13883,
- 48262, 62870, 65477, 65535, 0, 2, 88, 847, 6145, 37852,
- 62012, 65454, 65533, 65535, 0, 9, 47, 207, 1823, 14522,
- 45521, 61069, 64891, 65481, 65528, 65531, 65533, 65535, 0, 2,
- 9, 488, 2881, 12758, 38703, 58412, 64420, 65410, 65533, 65535,
- 0, 2, 4, 6, 61, 333, 1891, 6486, 19720, 43188,
- 57547, 62472, 64796, 65421, 65497, 65523, 65529, 65531, 65533, 65535,
- 0, 2, 4, 6, 8, 10, 12, 29, 117, 447,
- 1528, 6138, 21242, 43133, 56495, 62432, 64746, 65362, 65500, 65529,
- 65531, 65533, 65535, 0, 2, 18, 105, 301, 760, 1490,
- 3472, 7568, 15002, 26424, 40330, 53029, 60048, 62964, 64274, 64890,
- 65337, 65445, 65489, 65513, 65527, 65530, 65533, 65535, 0, 2,
- 4, 6, 41, 102, 409, 853, 2031, 4316, 7302, 11328,
- 16869, 24825, 34926, 43481, 50877, 56126, 59874, 62103, 63281, 63857,
- 64166, 64675, 65382, 65522, 65531, 65533, 65535, 0, 2, 4,
- 6, 8, 10, 12, 14, 16, 18, 29, 38, 53,
- 58, 96, 181, 503, 1183, 2849, 5590, 8600, 11379, 13942,
- 16478, 19453, 22638, 26039, 29411, 32921, 37596, 41433, 44998, 48560,
- 51979, 55106, 57666, 59892, 61485, 62616, 63484, 64018, 64375, 64685,
- 64924, 65076, 65278, 65395, 65471, 65509, 65529, 65535, 0, 65535,
- 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535,
- 0, 65535, 0, 65535, 0, 2, 65533, 65535, 0, 2,
- 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533, 65535,
- 0, 2, 65533, 65535, 0, 2, 65533, 65535, 0, 7,
- 65519, 65535, 0, 2, 14, 65491, 65533, 65535, 0, 2,
- 81, 65427, 65531, 65533, 65535, 0, 2, 4, 312, 65293,
- 65528, 65533, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
- 65535, 0, 65535, 0, 65535, 0, 65535, 0, 65535, 0,
- 2, 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65533,
- 65535, 0, 5, 65523, 65535, 0, 2, 65533, 65535, 0,
- 7, 65526, 65535, 0, 46, 65464, 65533, 65535, 0, 2,
- 120, 65309, 65533, 65535, 0, 2, 5, 362, 65097, 65533,
- 65535, 0, 2, 18, 1164, 64785, 65528, 65531, 65533, 65535,
- 0, 65535, 0, 65535, 0, 65535, 0, 65533, 65535, 0,
- 65535, 0, 65533, 65535, 0, 2, 65533, 65535, 0, 2,
- 65533, 65535, 0, 2, 65533, 65535, 0, 2, 65530, 65535,
- 0, 2, 65523, 65535, 0, 69, 65477, 65535, 0, 141,
- 65459, 65535, 0, 194, 65325, 65533, 65535, 0, 2, 543,
- 64912, 65533, 65535, 0, 5, 1270, 64301, 65529, 65531, 65533,
- 65535, 0, 2, 4, 12, 2055, 63538, 65508, 65531, 65533,
- 65535, 0, 2, 7, 102, 3775, 61970, 65429, 65526, 65528,
- 65533, 65535, 0, 65535, 0, 65535, 0, 65535, 0, 2,
- 65533, 65535, 0, 2, 65535, 0, 9, 65533, 65535, 0,
- 25, 65512, 65535, 0, 2, 65533, 65535, 0, 44, 65480,
- 65535, 0, 48, 65475, 65535, 0, 162, 65373, 65535, 0,
- 637, 64806, 65533, 65535, 0, 2, 935, 64445, 65533, 65535,
- 0, 2, 4, 1662, 64083, 65533, 65535, 0, 2, 12,
- 3036, 62469, 65521, 65533, 65535, 0, 2, 120, 5405, 60468,
- 65469, 65531, 65533, 65535, 0, 2, 4, 18, 254, 6663,
- 58999, 65272, 65528, 65533, 65535, 0, 2, 4, 9, 12,
- 67, 591, 8981, 56781, 64564, 65365, 65508, 65524, 65526, 65529,
- 65531, 65533, 65535, 0, 65535, 0, 65535, 0, 2, 65533,
- 65535, 0, 9, 65526, 65535, 0, 14, 65503, 65535, 0,
- 127, 65390, 65535, 0, 517, 64990, 65535, 0, 178, 65330,
- 65535, 0, 2, 1055, 64533, 65533, 65535, 0, 2, 1558,
- 63942, 65533, 65535, 0, 2, 2205, 63173, 65533, 65535, 0,
- 25, 4493, 60862, 65505, 65533, 65535, 0, 2, 48, 5890,
- 59442, 65482, 65533, 65535, 0, 2, 4, 127, 7532, 58191,
- 65394, 65533, 65535, 0, 2, 5, 32, 550, 10388, 54924,
- 65046, 65510, 65531, 65533, 65535, 0, 2, 4, 30, 150,
- 1685, 14340, 51375, 63619, 65288, 65503, 65528, 65533, 65535, 0,
- 2, 4, 6, 8, 28, 97, 473, 2692, 15407, 50020,
- 62880, 65064, 65445, 65508, 65531, 65533, 65535, 0, 2, 4,
- 12, 32, 79, 150, 372, 907, 2184, 5868, 18207, 45431,
- 59856, 64031, 65096, 65401, 65481, 65507, 65521, 65523, 65525, 65527,
- 65529, 65531, 65533, 65535, 0, 65533, 65535, 0, 182, 65491,
- 65535, 0, 877, 64286, 65535, 0, 9, 2708, 63612, 65533,
- 65535, 0, 2, 6038, 59532, 65535, 0, 2, 92, 5500,
- 60539, 65533, 65535, 0, 268, 8908, 56512, 65385, 65535, 0,
- 129, 13110, 52742, 65036, 65535, 0, 2, 806, 14003, 51929,
- 64732, 65523, 65535, 0, 7, 92, 2667, 18159, 47678, 62610,
- 65355, 65535, 0, 32, 1836, 19676, 48237, 61677, 64960, 65526,
- 65535, 0, 21, 159, 967, 5668, 22782, 44709, 58317, 64020,
- 65406, 65528, 65535, 0, 7, 162, 1838, 8328, 23929, 43014,
- 56394, 63374, 65216, 65484, 65521, 65535, 0, 2, 4, 6,
- 28, 268, 1120, 3613, 10688, 24185, 40989, 54917, 61684, 64510,
- 65403, 65530, 65535, 0, 2, 16, 44, 139, 492, 1739,
- 5313, 13558, 26766, 41566, 52446, 58937, 62815, 64480, 65201, 65454,
- 65524, 65533, 65535, 0, 7, 25, 76, 263, 612, 1466,
- 3325, 6832, 12366, 20152, 29466, 39255, 47360, 53506, 57740, 60726,
- 62845, 64131, 64882, 65260, 65459, 65521, 65528, 65530, 65535, 0,
- 2, 4, 14, 48, 136, 312, 653, 1240, 2369, 4327,
- 7028, 10759, 15449, 21235, 28027, 35386, 42938, 49562, 54990, 59119,
- 62086, 63916, 64863, 65249, 65445, 65493, 65523, 65535, 0, 2,
- 4, 6, 8, 10, 12, 21, 83, 208, 409, 723,
- 1152, 1868, 2951, 4463, 6460, 8979, 11831, 15195, 18863, 22657,
- 26762, 30881, 34963, 39098, 43054, 47069, 50620, 53871, 56821, 59386,
- 61340, 62670, 63512, 64023, 64429, 64750, 64944, 65126, 65279, 65366,
- 65413, 65445, 65473, 65505, 65510, 65521, 65528, 65530, 65535};
-
-/* pointers to cdf tables for quantizer indices */
-const WebRtc_UWord16 *WebRtcIsac_kQKltCdfPtrGain[3][12] = {
-{WebRtcIsac_kQKltCdfGain +0 +0, WebRtcIsac_kQKltCdfGain +0 +8, WebRtcIsac_kQKltCdfGain +0 +22, WebRtcIsac_kQKltCdfGain +0 +32, WebRtcIsac_kQKltCdfGain +0 +48, WebRtcIsac_kQKltCdfGain +0 +60, WebRtcIsac_kQKltCdfGain +0 +81, WebRtcIsac_kQKltCdfGain +0 +95, WebRtcIsac_kQKltCdfGain +0 +128, WebRtcIsac_kQKltCdfGain +0 +152,
-WebRtcIsac_kQKltCdfGain +0 +210, WebRtcIsac_kQKltCdfGain +0 +264},
-{WebRtcIsac_kQKltCdfGain +404 +0, WebRtcIsac_kQKltCdfGain +404 +8, WebRtcIsac_kQKltCdfGain +404 +21, WebRtcIsac_kQKltCdfGain +404 +30, WebRtcIsac_kQKltCdfGain +404 +46, WebRtcIsac_kQKltCdfGain +404 +58, WebRtcIsac_kQKltCdfGain +404 +79, WebRtcIsac_kQKltCdfGain +404 +93, WebRtcIsac_kQKltCdfGain +404 +125, WebRtcIsac_kQKltCdfGain +404 +149,
-WebRtcIsac_kQKltCdfGain +404 +207, WebRtcIsac_kQKltCdfGain +404 +260},
-{WebRtcIsac_kQKltCdfGain +803 +0, WebRtcIsac_kQKltCdfGain +803 +8, WebRtcIsac_kQKltCdfGain +803 +22, WebRtcIsac_kQKltCdfGain +803 +31, WebRtcIsac_kQKltCdfGain +803 +48, WebRtcIsac_kQKltCdfGain +803 +60, WebRtcIsac_kQKltCdfGain +803 +81, WebRtcIsac_kQKltCdfGain +803 +96, WebRtcIsac_kQKltCdfGain +803 +129, WebRtcIsac_kQKltCdfGain +803 +154,
-WebRtcIsac_kQKltCdfGain +803 +212, WebRtcIsac_kQKltCdfGain +803 +268}};
-
-const WebRtc_UWord16 *WebRtcIsac_kQKltCdfPtrShape[3][108] = {
-{WebRtcIsac_kQKltCdfShape +0 +0, WebRtcIsac_kQKltCdfShape +0 +2, WebRtcIsac_kQKltCdfShape +0 +4, WebRtcIsac_kQKltCdfShape +0 +6, WebRtcIsac_kQKltCdfShape +0 +8, WebRtcIsac_kQKltCdfShape +0 +10, WebRtcIsac_kQKltCdfShape +0 +12, WebRtcIsac_kQKltCdfShape +0 +14, WebRtcIsac_kQKltCdfShape +0 +16, WebRtcIsac_kQKltCdfShape +0 +18,
-WebRtcIsac_kQKltCdfShape +0 +21, WebRtcIsac_kQKltCdfShape +0 +25, WebRtcIsac_kQKltCdfShape +0 +29, WebRtcIsac_kQKltCdfShape +0 +33, WebRtcIsac_kQKltCdfShape +0 +37, WebRtcIsac_kQKltCdfShape +0 +43, WebRtcIsac_kQKltCdfShape +0 +49, WebRtcIsac_kQKltCdfShape +0 +56, WebRtcIsac_kQKltCdfShape +0 +64, WebRtcIsac_kQKltCdfShape +0 +66,
-WebRtcIsac_kQKltCdfShape +0 +68, WebRtcIsac_kQKltCdfShape +0 +70, WebRtcIsac_kQKltCdfShape +0 +72, WebRtcIsac_kQKltCdfShape +0 +75, WebRtcIsac_kQKltCdfShape +0 +77, WebRtcIsac_kQKltCdfShape +0 +79, WebRtcIsac_kQKltCdfShape +0 +81, WebRtcIsac_kQKltCdfShape +0 +83, WebRtcIsac_kQKltCdfShape +0 +86, WebRtcIsac_kQKltCdfShape +0 +90,
-WebRtcIsac_kQKltCdfShape +0 +94, WebRtcIsac_kQKltCdfShape +0 +98, WebRtcIsac_kQKltCdfShape +0 +102, WebRtcIsac_kQKltCdfShape +0 +107, WebRtcIsac_kQKltCdfShape +0 +113, WebRtcIsac_kQKltCdfShape +0 +120, WebRtcIsac_kQKltCdfShape +0 +129, WebRtcIsac_kQKltCdfShape +0 +131, WebRtcIsac_kQKltCdfShape +0 +133, WebRtcIsac_kQKltCdfShape +0 +135,
-WebRtcIsac_kQKltCdfShape +0 +137, WebRtcIsac_kQKltCdfShape +0 +141, WebRtcIsac_kQKltCdfShape +0 +143, WebRtcIsac_kQKltCdfShape +0 +147, WebRtcIsac_kQKltCdfShape +0 +151, WebRtcIsac_kQKltCdfShape +0 +155, WebRtcIsac_kQKltCdfShape +0 +159, WebRtcIsac_kQKltCdfShape +0 +164, WebRtcIsac_kQKltCdfShape +0 +168, WebRtcIsac_kQKltCdfShape +0 +172,
-WebRtcIsac_kQKltCdfShape +0 +178, WebRtcIsac_kQKltCdfShape +0 +184, WebRtcIsac_kQKltCdfShape +0 +192, WebRtcIsac_kQKltCdfShape +0 +200, WebRtcIsac_kQKltCdfShape +0 +211, WebRtcIsac_kQKltCdfShape +0 +213, WebRtcIsac_kQKltCdfShape +0 +215, WebRtcIsac_kQKltCdfShape +0 +217, WebRtcIsac_kQKltCdfShape +0 +219, WebRtcIsac_kQKltCdfShape +0 +223,
-WebRtcIsac_kQKltCdfShape +0 +227, WebRtcIsac_kQKltCdfShape +0 +231, WebRtcIsac_kQKltCdfShape +0 +235, WebRtcIsac_kQKltCdfShape +0 +239, WebRtcIsac_kQKltCdfShape +0 +243, WebRtcIsac_kQKltCdfShape +0 +248, WebRtcIsac_kQKltCdfShape +0 +252, WebRtcIsac_kQKltCdfShape +0 +258, WebRtcIsac_kQKltCdfShape +0 +264, WebRtcIsac_kQKltCdfShape +0 +273,
-WebRtcIsac_kQKltCdfShape +0 +282, WebRtcIsac_kQKltCdfShape +0 +293, WebRtcIsac_kQKltCdfShape +0 +308, WebRtcIsac_kQKltCdfShape +0 +310, WebRtcIsac_kQKltCdfShape +0 +312, WebRtcIsac_kQKltCdfShape +0 +316, WebRtcIsac_kQKltCdfShape +0 +320, WebRtcIsac_kQKltCdfShape +0 +324, WebRtcIsac_kQKltCdfShape +0 +328, WebRtcIsac_kQKltCdfShape +0 +332,
-WebRtcIsac_kQKltCdfShape +0 +336, WebRtcIsac_kQKltCdfShape +0 +341, WebRtcIsac_kQKltCdfShape +0 +347, WebRtcIsac_kQKltCdfShape +0 +354, WebRtcIsac_kQKltCdfShape +0 +360, WebRtcIsac_kQKltCdfShape +0 +368, WebRtcIsac_kQKltCdfShape +0 +378, WebRtcIsac_kQKltCdfShape +0 +388, WebRtcIsac_kQKltCdfShape +0 +400, WebRtcIsac_kQKltCdfShape +0 +418,
-WebRtcIsac_kQKltCdfShape +0 +445, WebRtcIsac_kQKltCdfShape +0 +447, WebRtcIsac_kQKltCdfShape +0 +451, WebRtcIsac_kQKltCdfShape +0 +455, WebRtcIsac_kQKltCdfShape +0 +461, WebRtcIsac_kQKltCdfShape +0 +468, WebRtcIsac_kQKltCdfShape +0 +474, WebRtcIsac_kQKltCdfShape +0 +480, WebRtcIsac_kQKltCdfShape +0 +486, WebRtcIsac_kQKltCdfShape +0 +495,
-WebRtcIsac_kQKltCdfShape +0 +505, WebRtcIsac_kQKltCdfShape +0 +516, WebRtcIsac_kQKltCdfShape +0 +528, WebRtcIsac_kQKltCdfShape +0 +543, WebRtcIsac_kQKltCdfShape +0 +564, WebRtcIsac_kQKltCdfShape +0 +583, WebRtcIsac_kQKltCdfShape +0 +608, WebRtcIsac_kQKltCdfShape +0 +635},
-{WebRtcIsac_kQKltCdfShape +686 +0, WebRtcIsac_kQKltCdfShape +686 +2, WebRtcIsac_kQKltCdfShape +686 +4, WebRtcIsac_kQKltCdfShape +686 +6, WebRtcIsac_kQKltCdfShape +686 +8, WebRtcIsac_kQKltCdfShape +686 +11, WebRtcIsac_kQKltCdfShape +686 +13, WebRtcIsac_kQKltCdfShape +686 +15, WebRtcIsac_kQKltCdfShape +686 +17, WebRtcIsac_kQKltCdfShape +686 +20,
-WebRtcIsac_kQKltCdfShape +686 +23, WebRtcIsac_kQKltCdfShape +686 +27, WebRtcIsac_kQKltCdfShape +686 +31, WebRtcIsac_kQKltCdfShape +686 +35, WebRtcIsac_kQKltCdfShape +686 +40, WebRtcIsac_kQKltCdfShape +686 +44, WebRtcIsac_kQKltCdfShape +686 +50, WebRtcIsac_kQKltCdfShape +686 +56, WebRtcIsac_kQKltCdfShape +686 +63, WebRtcIsac_kQKltCdfShape +686 +65,
-WebRtcIsac_kQKltCdfShape +686 +67, WebRtcIsac_kQKltCdfShape +686 +69, WebRtcIsac_kQKltCdfShape +686 +71, WebRtcIsac_kQKltCdfShape +686 +73, WebRtcIsac_kQKltCdfShape +686 +75, WebRtcIsac_kQKltCdfShape +686 +77, WebRtcIsac_kQKltCdfShape +686 +79, WebRtcIsac_kQKltCdfShape +686 +82, WebRtcIsac_kQKltCdfShape +686 +85, WebRtcIsac_kQKltCdfShape +686 +89,
-WebRtcIsac_kQKltCdfShape +686 +93, WebRtcIsac_kQKltCdfShape +686 +97, WebRtcIsac_kQKltCdfShape +686 +102, WebRtcIsac_kQKltCdfShape +686 +106, WebRtcIsac_kQKltCdfShape +686 +112, WebRtcIsac_kQKltCdfShape +686 +119, WebRtcIsac_kQKltCdfShape +686 +127, WebRtcIsac_kQKltCdfShape +686 +129, WebRtcIsac_kQKltCdfShape +686 +131, WebRtcIsac_kQKltCdfShape +686 +133,
-WebRtcIsac_kQKltCdfShape +686 +135, WebRtcIsac_kQKltCdfShape +686 +137, WebRtcIsac_kQKltCdfShape +686 +139, WebRtcIsac_kQKltCdfShape +686 +142, WebRtcIsac_kQKltCdfShape +686 +146, WebRtcIsac_kQKltCdfShape +686 +150, WebRtcIsac_kQKltCdfShape +686 +154, WebRtcIsac_kQKltCdfShape +686 +158, WebRtcIsac_kQKltCdfShape +686 +162, WebRtcIsac_kQKltCdfShape +686 +167,
-WebRtcIsac_kQKltCdfShape +686 +173, WebRtcIsac_kQKltCdfShape +686 +179, WebRtcIsac_kQKltCdfShape +686 +186, WebRtcIsac_kQKltCdfShape +686 +194, WebRtcIsac_kQKltCdfShape +686 +205, WebRtcIsac_kQKltCdfShape +686 +207, WebRtcIsac_kQKltCdfShape +686 +209, WebRtcIsac_kQKltCdfShape +686 +211, WebRtcIsac_kQKltCdfShape +686 +214, WebRtcIsac_kQKltCdfShape +686 +218,
-WebRtcIsac_kQKltCdfShape +686 +222, WebRtcIsac_kQKltCdfShape +686 +226, WebRtcIsac_kQKltCdfShape +686 +230, WebRtcIsac_kQKltCdfShape +686 +234, WebRtcIsac_kQKltCdfShape +686 +238, WebRtcIsac_kQKltCdfShape +686 +242, WebRtcIsac_kQKltCdfShape +686 +247, WebRtcIsac_kQKltCdfShape +686 +253, WebRtcIsac_kQKltCdfShape +686 +262, WebRtcIsac_kQKltCdfShape +686 +269,
-WebRtcIsac_kQKltCdfShape +686 +278, WebRtcIsac_kQKltCdfShape +686 +289, WebRtcIsac_kQKltCdfShape +686 +305, WebRtcIsac_kQKltCdfShape +686 +307, WebRtcIsac_kQKltCdfShape +686 +309, WebRtcIsac_kQKltCdfShape +686 +311, WebRtcIsac_kQKltCdfShape +686 +315, WebRtcIsac_kQKltCdfShape +686 +319, WebRtcIsac_kQKltCdfShape +686 +323, WebRtcIsac_kQKltCdfShape +686 +327,
-WebRtcIsac_kQKltCdfShape +686 +331, WebRtcIsac_kQKltCdfShape +686 +335, WebRtcIsac_kQKltCdfShape +686 +340, WebRtcIsac_kQKltCdfShape +686 +346, WebRtcIsac_kQKltCdfShape +686 +354, WebRtcIsac_kQKltCdfShape +686 +362, WebRtcIsac_kQKltCdfShape +686 +374, WebRtcIsac_kQKltCdfShape +686 +384, WebRtcIsac_kQKltCdfShape +686 +396, WebRtcIsac_kQKltCdfShape +686 +413,
-WebRtcIsac_kQKltCdfShape +686 +439, WebRtcIsac_kQKltCdfShape +686 +442, WebRtcIsac_kQKltCdfShape +686 +446, WebRtcIsac_kQKltCdfShape +686 +450, WebRtcIsac_kQKltCdfShape +686 +455, WebRtcIsac_kQKltCdfShape +686 +461, WebRtcIsac_kQKltCdfShape +686 +468, WebRtcIsac_kQKltCdfShape +686 +475, WebRtcIsac_kQKltCdfShape +686 +481, WebRtcIsac_kQKltCdfShape +686 +489,
-WebRtcIsac_kQKltCdfShape +686 +498, WebRtcIsac_kQKltCdfShape +686 +508, WebRtcIsac_kQKltCdfShape +686 +522, WebRtcIsac_kQKltCdfShape +686 +534, WebRtcIsac_kQKltCdfShape +686 +554, WebRtcIsac_kQKltCdfShape +686 +577, WebRtcIsac_kQKltCdfShape +686 +602, WebRtcIsac_kQKltCdfShape +686 +631},
-{WebRtcIsac_kQKltCdfShape +1368 +0, WebRtcIsac_kQKltCdfShape +1368 +2, WebRtcIsac_kQKltCdfShape +1368 +4, WebRtcIsac_kQKltCdfShape +1368 +6, WebRtcIsac_kQKltCdfShape +1368 +8, WebRtcIsac_kQKltCdfShape +1368 +10, WebRtcIsac_kQKltCdfShape +1368 +12, WebRtcIsac_kQKltCdfShape +1368 +14, WebRtcIsac_kQKltCdfShape +1368 +16, WebRtcIsac_kQKltCdfShape +1368 +20,
-WebRtcIsac_kQKltCdfShape +1368 +24, WebRtcIsac_kQKltCdfShape +1368 +28, WebRtcIsac_kQKltCdfShape +1368 +32, WebRtcIsac_kQKltCdfShape +1368 +36, WebRtcIsac_kQKltCdfShape +1368 +40, WebRtcIsac_kQKltCdfShape +1368 +44, WebRtcIsac_kQKltCdfShape +1368 +50, WebRtcIsac_kQKltCdfShape +1368 +57, WebRtcIsac_kQKltCdfShape +1368 +65, WebRtcIsac_kQKltCdfShape +1368 +67,
-WebRtcIsac_kQKltCdfShape +1368 +69, WebRtcIsac_kQKltCdfShape +1368 +71, WebRtcIsac_kQKltCdfShape +1368 +73, WebRtcIsac_kQKltCdfShape +1368 +75, WebRtcIsac_kQKltCdfShape +1368 +77, WebRtcIsac_kQKltCdfShape +1368 +79, WebRtcIsac_kQKltCdfShape +1368 +81, WebRtcIsac_kQKltCdfShape +1368 +85, WebRtcIsac_kQKltCdfShape +1368 +89, WebRtcIsac_kQKltCdfShape +1368 +93,
-WebRtcIsac_kQKltCdfShape +1368 +97, WebRtcIsac_kQKltCdfShape +1368 +101, WebRtcIsac_kQKltCdfShape +1368 +105, WebRtcIsac_kQKltCdfShape +1368 +110, WebRtcIsac_kQKltCdfShape +1368 +116, WebRtcIsac_kQKltCdfShape +1368 +123, WebRtcIsac_kQKltCdfShape +1368 +132, WebRtcIsac_kQKltCdfShape +1368 +134, WebRtcIsac_kQKltCdfShape +1368 +136, WebRtcIsac_kQKltCdfShape +1368 +138,
-WebRtcIsac_kQKltCdfShape +1368 +141, WebRtcIsac_kQKltCdfShape +1368 +143, WebRtcIsac_kQKltCdfShape +1368 +146, WebRtcIsac_kQKltCdfShape +1368 +150, WebRtcIsac_kQKltCdfShape +1368 +154, WebRtcIsac_kQKltCdfShape +1368 +158, WebRtcIsac_kQKltCdfShape +1368 +162, WebRtcIsac_kQKltCdfShape +1368 +166, WebRtcIsac_kQKltCdfShape +1368 +170, WebRtcIsac_kQKltCdfShape +1368 +174,
-WebRtcIsac_kQKltCdfShape +1368 +179, WebRtcIsac_kQKltCdfShape +1368 +185, WebRtcIsac_kQKltCdfShape +1368 +193, WebRtcIsac_kQKltCdfShape +1368 +203, WebRtcIsac_kQKltCdfShape +1368 +214, WebRtcIsac_kQKltCdfShape +1368 +216, WebRtcIsac_kQKltCdfShape +1368 +218, WebRtcIsac_kQKltCdfShape +1368 +220, WebRtcIsac_kQKltCdfShape +1368 +224, WebRtcIsac_kQKltCdfShape +1368 +227,
-WebRtcIsac_kQKltCdfShape +1368 +231, WebRtcIsac_kQKltCdfShape +1368 +235, WebRtcIsac_kQKltCdfShape +1368 +239, WebRtcIsac_kQKltCdfShape +1368 +243, WebRtcIsac_kQKltCdfShape +1368 +247, WebRtcIsac_kQKltCdfShape +1368 +251, WebRtcIsac_kQKltCdfShape +1368 +256, WebRtcIsac_kQKltCdfShape +1368 +262, WebRtcIsac_kQKltCdfShape +1368 +269, WebRtcIsac_kQKltCdfShape +1368 +277,
-WebRtcIsac_kQKltCdfShape +1368 +286, WebRtcIsac_kQKltCdfShape +1368 +297, WebRtcIsac_kQKltCdfShape +1368 +315, WebRtcIsac_kQKltCdfShape +1368 +317, WebRtcIsac_kQKltCdfShape +1368 +319, WebRtcIsac_kQKltCdfShape +1368 +323, WebRtcIsac_kQKltCdfShape +1368 +327, WebRtcIsac_kQKltCdfShape +1368 +331, WebRtcIsac_kQKltCdfShape +1368 +335, WebRtcIsac_kQKltCdfShape +1368 +339,
-WebRtcIsac_kQKltCdfShape +1368 +343, WebRtcIsac_kQKltCdfShape +1368 +349, WebRtcIsac_kQKltCdfShape +1368 +355, WebRtcIsac_kQKltCdfShape +1368 +361, WebRtcIsac_kQKltCdfShape +1368 +368, WebRtcIsac_kQKltCdfShape +1368 +376, WebRtcIsac_kQKltCdfShape +1368 +385, WebRtcIsac_kQKltCdfShape +1368 +397, WebRtcIsac_kQKltCdfShape +1368 +411, WebRtcIsac_kQKltCdfShape +1368 +429,
-WebRtcIsac_kQKltCdfShape +1368 +456, WebRtcIsac_kQKltCdfShape +1368 +459, WebRtcIsac_kQKltCdfShape +1368 +463, WebRtcIsac_kQKltCdfShape +1368 +467, WebRtcIsac_kQKltCdfShape +1368 +473, WebRtcIsac_kQKltCdfShape +1368 +478, WebRtcIsac_kQKltCdfShape +1368 +485, WebRtcIsac_kQKltCdfShape +1368 +491, WebRtcIsac_kQKltCdfShape +1368 +497, WebRtcIsac_kQKltCdfShape +1368 +505,
-WebRtcIsac_kQKltCdfShape +1368 +514, WebRtcIsac_kQKltCdfShape +1368 +523, WebRtcIsac_kQKltCdfShape +1368 +535, WebRtcIsac_kQKltCdfShape +1368 +548, WebRtcIsac_kQKltCdfShape +1368 +565, WebRtcIsac_kQKltCdfShape +1368 +585, WebRtcIsac_kQKltCdfShape +1368 +611, WebRtcIsac_kQKltCdfShape +1368 +640}};
-
-/* code length for all coefficients using different models */
-const double WebRtcIsac_kQKltCodeLenGain[392] = {
- 12.29956028, 7.83007500, 4.25642781, 0.17489215, 4.27591254, 7.66908312, 10.47643804, 11.91253716, 9.03421572, 8.57373525,
- 6.73555740, 5.41409855, 3.65237863, 0.42623449, 3.22418399, 5.68145719, 7.14201900, 8.20558413, 9.34178852, 13.00000000,
- 13.00000000, 9.47643804, 6.71459778, 3.55472644, 0.28457419, 3.70652835, 7.41128536, 11.60768258, 14.00000000, 12.29956028,
- 9.29956028, 8.02845645, 6.54878889, 5.07667251, 2.87749552, 0.65310542, 3.11316029, 4.87911416, 5.89540125, 6.76398581,
- 7.70537925, 9.67807191, 11.14201900, 14.00000000, 14.00000000, 12.29956028, 9.69621925, 7.70537925, 5.49915812, 2.67441345,
- 0.63381441, 2.74999773, 5.76877882, 8.41503750, 10.75207249, 14.00000000, 9.60768258, 8.85025288, 8.09913319, 7.09010692,
- 6.36337538, 5.46667027, 4.51618422, 2.64189829, 1.21843537, 2.43823474, 3.89409149, 4.99718498, 6.00989604, 6.78325414,
- 7.39637366, 8.76159526, 10.60768258, 11.67807191, 12.29956028, 9.54056838, 7.95560588, 5.63040265, 3.81264793, 2.18521728,
- 1.33727600, 2.37909290, 3.94981123, 5.52426657, 7.32051990, 8.84012866, 11.35614381, 13.00000000, 12.29956028, 11.35614381,
- 11.35614381, 10.24511250, 9.12963528, 8.80032766, 7.97763219, 7.83007500, 7.09913319, 6.68711704, 6.41879942, 5.74379131,
- 5.25096862, 4.43061904, 3.54492969, 2.72664147, 2.16306204, 2.71142226, 3.34357514, 4.07444556, 4.95151313, 5.68145719,
- 6.12041675, 6.55085135, 7.00282052, 7.55705650, 8.44541115, 9.41503750, 10.91253716, 11.60768258, 14.00000000, 14.00000000,
- 13.00000000, 11.60768258, 10.47643804, 8.80032766, 7.38161450, 6.52426657, 5.47447919, 4.53749773, 3.92719747, 3.41058292,
- 2.88495635, 2.79344346, 2.85805254, 3.18261657, 3.57216340, 4.08225499, 4.74438125, 5.51215997, 6.30477171, 7.15450995,
- 8.28575448, 9.69621925, 11.60768258, 13.00000000, 13.67807191, 15.00000000, 15.00000000, 14.00000000, 12.41503750, 11.29956028,
- 12.41503750, 11.60768258, 10.11735695, 9.47643804, 9.12963528, 9.41503750, 8.23181568, 7.97763219, 7.82507432, 7.50814690,
- 7.33466408, 6.99157138, 6.95834085, 6.80524315, 6.47447919, 6.35614381, 6.02128954, 5.71459778, 5.58109327, 5.05821876,
- 4.94539568, 4.59220115, 4.27591254, 4.01522554, 3.89376424, 3.83760867, 3.73321346, 3.74674342, 3.90493270, 4.18942837,
- 4.33599724, 4.42446075, 4.81760565, 5.07667251, 5.54570071, 5.95697272, 6.46667027, 6.91253716, 7.33466408, 7.82507432,
- 8.05163277, 9.12963528, 9.02272008, 9.77118131, 9.60768258, 10.11735695, 12.00000000, 12.83007500, 15.00000000, 15.00000000,
- 14.00000000, 15.00000000, 11.75207249, 12.29956028, 10.95560588, 9.93391081, 9.02272008, 8.00564656, 7.82507432, 7.28919357,
- 6.77599833, 6.56745810, 6.19910010, 6.23347109, 5.67694524, 5.41879942, 4.96039548, 4.88170777, 4.60768258, 4.52883287,
- 4.28876323, 4.17583679, 4.21332197, 4.14474217, 4.16119001, 4.12809476, 4.18501706, 4.28489599, 4.35299136, 4.60286019,
- 4.63040265, 4.81017544, 5.00989604, 5.33822190, 5.43489792, 5.84644797, 6.13272126, 6.75444729, 7.36337538, 7.96108101,
- 8.54056838, 9.28575448, 9.12963528, 9.47643804, 10.75207249, 12.29956028, 12.41503750, 11.14201900, 12.83007500, 15.00000000,
- 15.00000000, 15.00000000, 15.00000000, 15.00000000, 15.00000000, 14.00000000, 15.00000000, 15.00000000, 15.00000000, 15.00000000,
- 13.67807191, 12.41503750, 11.91253716, 11.60768258, 12.29956028, 13.00000000, 12.29956028, 10.95560588, 10.60768258, 9.77118131,
- 10.02272008, 10.11735695, 10.14201900, 10.75207249, 10.75207249, 9.69621925, 9.47643804, 9.75207249, 8.72387559, 8.47643804,
- 8.68711704, 8.40754296, 8.20558413, 8.26529038, 7.84518189, 7.89147554, 7.64685317, 7.41128536, 7.33466408, 7.42635281,
- 7.24845594, 7.11430363, 7.07518750, 7.07518750, 6.70537925, 6.64906082, 6.73555740, 6.62931259, 6.50015411, 6.26190774,
- 6.36337538, 6.19264508, 5.95151313, 6.08860801, 5.91253716, 5.83007500, 5.68145719, 5.67244736, 5.82632286, 6.00282052,
- 5.65348627, 5.74970158, 5.87846648, 5.69052365, 5.64464890, 5.58531476, 5.81512466, 5.57688409, 5.87329553, 5.62170514,
- 5.74851759, 5.81017544, 5.64464890, 5.76398581, 5.60339522, 5.69507833, 5.84139031, 5.68711704, 5.73908047, 5.84139031,
- 5.71459778, 5.96245305, 5.82632286, 5.89540125, 6.08860801, 6.12041675, 6.13272126, 6.30477171, 6.24177679, 6.28232358,
- 6.29091619, 6.53239445, 6.81512466, 6.72620440, 6.65792533, 6.84518189, 7.10215454, 7.44157929, 7.57793523, 7.59485854,
- 7.66460965, 8.05163277, 8.00564656, 8.43775758, 8.10518224, 8.28575448, 8.77118131, 8.23181568, 8.29264087, 8.20558413,
- 8.34894831, 8.89147554, 8.40754296, 8.61629571, 8.64244800, 8.61629571, 8.93391081, 8.50814690, 9.02272008, 8.68711704,
- 8.65127185, 9.41503750, 9.11735695, 9.85025288, 10.24511250, 10.60768258, 10.47643804, 11.60768258, 11.35614381, 12.29956028,
- 15.00000000, 15.00000000, 12.29956028, 13.00000000, 15.00000000, 15.00000000, 15.00000000, 15.00000000, 14.00000000, 12.83007500,
- 15.00000000, 15.00000000};
-
-const double WebRtcIsac_kQKltCodeLenShape[578] = {
- 0.00002201, 0.00002201, 0.00002201, 0.00002201, 0.00002201, 0.00002201, 0.00002201, 0.00002201, 0.00002201, 14.00000000,
- 0.00011007, 13.00000000, 0.00066056, 11.60768258, 11.14201900, 0.00185034, 10.24511250, 9.08113676, 0.00480700, 9.41503750,
- 8.09913319, 0.01084946, 8.02845645, 13.00000000, 6.40941295, 0.02926496, 6.95560588, 13.00000000, 13.00000000, 6.21864029,
- 0.03861814, 6.30477171, 12.29956028, 11.14201900, 4.66964328, 0.12158980, 4.62604734, 10.91253716, 14.00000000, 11.35614381,
- 8.76159526, 3.89671219, 0.21179147, 3.99472634, 8.02272008, 13.00000000, 0.00002201, 0.00002201, 0.00002201, 0.00002201,
- 14.00000000, 0.00011007, 0.00002201, 0.00002201, 0.00002201, 0.00002201, 14.00000000, 0.00011007, 10.95560588, 0.00147568,
- 10.95560588, 10.24511250, 0.00240150, 10.24511250, 8.02845645, 0.01056115, 8.17982104, 6.74497143, 0.02381629, 7.15137706,
- 5.68598330, 0.05650076, 5.72970467, 13.00000000, 14.00000000, 5.18221688, 0.07697435, 5.37611851, 11.91253716, 9.54056838,
- 3.92853764, 0.22143514, 3.76428491, 9.28575448, 14.00000000, 11.35614381, 7.37794818, 3.30585980, 0.37001735, 3.18521728,
- 6.88886433, 11.91253716, 14.00000000, 0.00002201, 0.00002201, 0.00002201, 0.00002201, 14.00000000, 0.00019814, 14.00000000,
- 0.00002201, 13.00000000, 0.00028621, 14.00000000, 14.00000000, 0.00028621, 13.00000000, 11.91253716, 0.00094690, 11.35614381,
- 10.60768258, 0.00213692, 10.24511250, 8.37794818, 0.00863317, 8.43775758, 14.00000000, 7.41128536, 0.01698415, 7.42635281,
- 6.02702021, 0.04514485, 6.01558154, 13.00000000, 5.05090284, 0.09207659, 4.98877274, 15.00000000, 10.75207249, 4.41081703,
- 0.15733047, 4.17424617, 11.60768258, 14.00000000, 15.00000000, 10.06926266, 3.74320161, 0.23091117, 3.81141115, 10.02272008,
- 11.91253716, 6.90196792, 2.92703003, 0.46874039, 2.93846004, 7.26190774, 11.60768258, 13.00000000, 10.21864029, 7.95560588,
- 5.19345038, 2.40520888, 0.77554605, 2.56628417, 5.41409855, 8.81017544, 12.29956028, 0.00002201, 0.00002201, 0.00002201,
- 0.00002201, 11.91253716, 0.00068259, 12.29956028, 10.11735695, 0.00233535, 10.47643804, 10.35614381, 0.00140957, 12.29956028,
- 10.24511250, 0.00222511, 10.47643804, 7.72387559, 0.01475842, 7.52426657, 6.73555740, 0.02924249, 6.55085135, 14.00000000,
- 5.45021533, 0.06930886, 5.38345116, 4.55913083, 0.11289841, 4.92853764, 11.60768258, 4.07148162, 0.19798859, 3.87200568,
- 13.00000000, 9.60768258, 3.31393725, 0.29937064, 3.57321111, 9.35614381, 14.00000000, 8.02845645, 3.08542800, 0.45503557,
- 2.80678268, 7.47643804, 12.29956028, 14.00000000, 14.00000000, 14.00000000, 6.83509307, 2.69166097, 0.61673447, 2.51266238,
- 6.84771516, 11.91253716, 11.14201900, 7.72387559, 4.61899789, 2.17136763, 1.03592993, 2.28586183, 4.86814304, 7.78568088,
- 10.91253716, 14.00000000, 15.00000000, 11.04580369, 9.28575448, 7.26190774, 5.21946023, 3.61575588, 2.18431651, 1.45666604,
- 2.33566383, 3.85470467, 5.46181107, 7.30651304, 8.89147554, 11.91253716, 0.00002201, 0.00002201, 15.00000000, 0.00011007,
- 15.00000000, 10.47643804, 0.00149771, 11.60768258, 7.30651304, 0.01903486, 7.20558413, 6.92318440, 0.02587674, 6.71459778,
- 7.28919357, 0.01756340, 7.45696818, 6.06926266, 0.03841465, 6.45890338, 4.46035649, 0.13896157, 4.43204392, 14.00000000,
- 14.00000000, 4.09010692, 0.19672654, 3.86653665, 13.00000000, 14.00000000, 11.35614381, 3.45117809, 0.27340929, 3.64878468,
- 9.93391081, 9.60768258, 3.30585980, 0.35178287, 3.15607895, 9.12963528, 12.29956028, 7.93391081, 2.82180202, 0.53064436,
- 2.68258739, 7.42635281, 11.14201900, 15.00000000, 10.06926266, 6.15294265, 2.55861197, 0.69308389, 2.48602573, 5.90592231,
- 9.47643804, 13.00000000, 11.14201900, 8.20558413, 5.21618324, 2.34973357, 1.00379135, 2.09611815, 4.80278331, 7.82507432,
- 11.91253716, 12.29956028, 8.98877274, 6.75444729, 4.23886435, 2.19859476, 1.29528579, 2.06032897, 4.06704711, 6.76398581,
- 9.34178852, 13.00000000, 12.29956028, 10.14201900, 8.40754296, 6.16710999, 4.64850859, 3.28768796, 2.22892326, 1.78568088,
- 2.35209193, 3.45141888, 4.91121176, 6.45117809, 8.20558413, 10.35614381, 13.00000000, 12.83007500, 14.00000000, 14.00000000,
- 12.83007500, 11.35614381, 9.85025288, 9.41503750, 8.07518750, 7.34894831, 6.25516616, 5.25096862, 4.52477322, 3.68513357,
- 3.06152306, 2.64011320, 2.54608637, 2.95765662, 3.44445223, 4.01665007, 4.63258525, 5.27633906, 6.24678325, 7.05163277,
- 8.02845645, 9.60768258, 10.79054663, 12.83007500, 14.00000000, 0.00002201, 15.00000000, 0.00011007, 15.00000000, 5.82009091,
- 0.02710994, 10.11735695, 15.00000000, 3.30346709, 0.17317845, 6.41128536, 15.00000000, 15.00000000, 10.83007500, 2.72897475,
- 0.35935964, 3.85853144, 13.00000000, 15.00000000, 3.72766182, 0.17268211, 4.74674342, 15.00000000, 15.00000000, 4.40754296,
- 0.11993823, 4.93997965, 15.00000000, 15.00000000, 4.70193743, 0.22110152, 3.27591254, 11.35614381, 15.00000000, 12.54056838,
- 6.94743195, 2.99157138, 0.62079088, 2.25338071, 7.41128536, 15.00000000, 15.00000000, 11.47643804, 6.04580369, 2.80232255,
- 0.98380109, 1.70502034, 5.15607895, 10.75207249, 13.00000000, 15.00000000, 15.00000000, 11.35614381, 6.44157929, 2.37955105,
- 0.74567258, 2.44181848, 6.07667251, 10.75207249, 15.00000000, 15.00000000, 15.00000000, 15.00000000, 9.44541115, 5.25432568,
- 1.97815248, 0.94086682, 2.57885561, 5.17265730, 8.72387559, 14.00000000, 15.00000000, 15.00000000, 15.00000000, 10.67807191,
- 7.59485854, 4.91187431, 2.91934900, 1.32321648, 1.50614455, 3.85215911, 7.15137706, 10.60768258, 14.00000000, 15.00000000,
- 15.00000000, 15.00000000, 15.00000000, 15.00000000, 15.00000000, 15.00000000, 11.09310940, 8.50814690, 6.45117809, 3.94675287,
- 2.16730774, 1.37674720, 2.10215454, 3.97278511, 6.34178852, 8.50814690, 10.32757466, 14.00000000, 14.00000000, 15.00000000,
- 15.00000000, 15.00000000, 14.00000000, 10.02272008, 9.41503750, 7.03710399, 5.13349379, 3.61683574, 2.47999526, 1.82788507,
- 2.00945280, 3.08020557, 4.69793233, 6.64025044, 8.32051990, 10.91253716, 14.00000000, 15.00000000, 15.00000000, 15.00000000,
- 15.00000000, 15.00000000, 15.00000000, 14.41503750, 9.41503750, 7.10215454, 5.60768258, 4.44014496, 3.30064444, 2.56941678,
- 2.28103909, 2.42694459, 2.93206152, 3.83383692, 4.75207249, 6.02702021, 7.16394964, 9.35614381, 11.14201900, 12.29956028,
- 15.00000000, 15.00000000, 14.00000000, 15.00000000, 15.00000000, 10.11735695, 8.85025288, 7.02558541, 5.86300889, 4.79726396,
- 3.95117259, 3.44925321, 2.96348293, 2.88219459, 2.89671219, 3.10518224, 3.50065237, 4.05748549, 4.55291677, 5.23016216,
- 5.79420675, 6.39452048, 6.91253716, 7.97763219, 8.32051990, 9.35614381, 10.60768258, 11.60768258, 15.00000000, 15.00000000,
- 12.83007500, 12.41503750, 11.91253716, 14.00000000, 14.00000000, 11.35614381, 10.75207249, 10.91253716, 8.93391081, 8.61629571,
- 8.29264087, 8.02272008, 7.89147554, 7.33466408, 7.41128536, 6.71459778, 5.92853764, 5.31964048, 4.97693875, 4.70308379,
- 4.43632704, 4.34357514, 4.30825648, 4.26529038, 4.19465917, 4.27420773, 4.22315577, 4.08262792, 4.06667818, 4.16119001,
- 4.08860801, 4.26698468, 4.39128315, 4.71517590, 5.10442472, 5.50714538, 5.81017544, 6.15922208};
-
-/* left KLT transforms */
-const double WebRtcIsac_kKltT1Gain[3][4] = {
-{-0.79742827, 0.60341375, 0.60341375, 0.79742827},
-{-0.81372390, 0.58125159, 0.58125159, 0.81372390},
-{-0.71832547, 0.69570721, 0.69570721, 0.71832547}};
-
-const double WebRtcIsac_kKltT1Shape[3][324] = {
-{ 0.00159597, 0.00049320, 0.00513821, 0.00021066, 0.01338581, -0.00422367, -0.00272072, 0.00935107, 0.02047622, 0.02691189,
- 0.00478236, 0.03969702, 0.00886698, 0.04877604, -0.10898362, -0.05930891, -0.03415047, 0.98889721, 0.00293558, -0.00035282,
- 0.01156321, -0.00195341, -0.00937631, 0.01052213, -0.02551163, 0.01644059, 0.03189927, 0.07754773, -0.08742313, -0.03026338,
- 0.05136248, -0.14395974, 0.17725040, 0.22664856, 0.93380230, 0.07076411, 0.00557890, -0.00222834, 0.01377569, 0.01466808,
- 0.02847361, -0.00603178, 0.02382480, -0.01210452, 0.03797267, -0.02371480, 0.11260335, -0.07366682, 0.00453436, -0.04136941,
--0.07912843, -0.95031418, 0.25295337, -0.05302216, -0.00617554, -0.00044040, -0.00653778, 0.01097838, 0.01529174, 0.01374431,
--0.00748512, -0.00020034, 0.02432713, 0.11101570, -0.08556891, 0.09282249, -0.01029446, 0.67556443, -0.67454300, 0.06910063,
- 0.20866865, -0.10318050, 0.00932175, 0.00524058, 0.00803610, -0.00594676, -0.01082578, 0.01069906, 0.00546768, 0.01565291,
- 0.06816200, 0.10201227, 0.16812734, 0.22984074, 0.58213170, -0.54138651, -0.51379962, 0.06847390, -0.01920037, -0.04592324,
--0.00467394, 0.00328858, 0.00377424, -0.00987448, 0.08222096, -0.00377301, 0.04551941, -0.02592517, 0.16317082, 0.13077530,
- 0.22702921, -0.31215289, -0.69645962, -0.38047101, -0.39339411, 0.11124777, 0.02508035, -0.00708074, 0.00400344, 0.00040331,
- 0.01142402, 0.01725406, 0.01635170, 0.14285366, 0.03949233, -0.05905676, 0.05877154, -0.17497577, -0.32479440, 0.80754464,
--0.38085603, -0.17055430, -0.03168622, -0.07531451, 0.02942002, -0.02148095, -0.00754114, -0.00322372, 0.00567812, -0.01701521,
--0.12358320, 0.11473564, 0.09070136, 0.06533068, -0.22560802, 0.19209022, 0.81605094, 0.36592275, -0.09919829, 0.16667122,
- 0.16300725, 0.04803807, 0.06739263, -0.00156752, -0.01685302, -0.00905240, -0.02297836, -0.00469939, 0.06310613, -0.16391930,
- 0.10919511, 0.12529293, 0.85581322, -0.32145522, 0.24539076, 0.07181839, 0.07289591, 0.14066759, 0.10406711, 0.05815518,
- 0.01072680, -0.00759339, 0.00053486, -0.00044865, 0.03407361, 0.01645348, 0.08758579, 0.27722240, 0.53665485, -0.74853376,
--0.01118192, -0.19805430, 0.06130619, -0.09675299, 0.08978480, 0.03405255, -0.00706867, 0.05102045, 0.03250746, 0.01849966,
--0.01216314, -0.01184187, -0.01579288, 0.00114807, 0.11376166, 0.88342114, -0.36425379, 0.13863190, 0.12524180, -0.13553892,
- 0.04715856, -0.12341103, 0.04531568, 0.01899360, -0.00206897, 0.00567768, -0.01444163, 0.00411946, -0.00855896, 0.00381663,
--0.01664861, -0.05534280, 0.21328278, 0.20161162, 0.72360394, 0.59130708, -0.08043791, 0.08757349, -0.13893918, -0.05147377,
- 0.02680690, -0.01144070, 0.00625162, -0.00634215, -0.01248947, -0.00329455, -0.00609625, -0.00136305, -0.05097048, -0.01029851,
- 0.25065384, -0.16856837, -0.07123372, 0.15992623, -0.39487617, -0.79972301, 0.18118185, -0.04826639, -0.01805578, -0.02927253,
--0.16400618, 0.07472763, 0.10376449, 0.01705406, 0.01065801, -0.01500498, 0.02039914, 0.37776349, -0.84484186, 0.10434286,
- 0.15616990, 0.13474456, -0.00906238, -0.25238368, -0.03820885, -0.10650905, -0.03880833, -0.03660028, -0.09640894, 0.00583314,
- 0.01922097, 0.01489911, -0.02431117, -0.09372217, 0.39404721, -0.84786223, -0.31277121, 0.03193850, 0.01974060, 0.01887901,
- 0.00337911, -0.11359599, -0.02792521, -0.03220184, -0.01533311, 0.00015962, -0.04225043, -0.00933965, 0.00675311, 0.00206060,
- 0.15926771, 0.40199829, -0.80792558, -0.35591604, -0.17169764, 0.02830436, 0.02459982, -0.03438589, 0.00718705, -0.01798329,
--0.01594508, -0.00702430, -0.00952419, -0.00962701, -0.01307212, -0.01749740, 0.01299602, 0.00587270, -0.36103108, -0.82039266,
--0.43092844, -0.08500097, -0.04361674, -0.00333482, 0.01250434, -0.02538295, -0.00921797, 0.01645071, -0.01400872, 0.00317607,
- 0.00003277, -0.01617646, -0.00616863, -0.00882661, 0.00466157, 0.00353237, 0.91803104, -0.39503305, -0.02048964, 0.00060125,
- 0.01980634, 0.00300109, 0.00313880, 0.00657337, 0.00715163, 0.00000261, 0.00854276, -0.00154825, -0.00516128, 0.00909527,
- 0.00095609, 0.00701196, -0.00221867, -0.00156741},
-{-0.00469582, -0.00020403, -0.00587134, 0.00185153, -0.02256479, -0.01185761, -0.02891481, -0.00493792, -0.00182344, 0.00285962,
- 0.01558059, -0.02185140, 0.04639438, -0.04357142, 0.12718613, -0.06756136, 0.05542227, 0.98480184, -0.00374376, -0.00236433,
--0.00607169, -0.00303290, -0.00127243, -0.01794845, 0.00620033, -0.00732704, -0.02837749, -0.00107164, 0.04820548, 0.00713300,
- 0.09784244, -0.16806261, -0.04563341, -0.33406041, 0.91554083, -0.08139655, -0.00415851, -0.00538193, -0.00731198, -0.00534534,
--0.00623075, -0.02016943, -0.05480133, -0.03172290, -0.03879603, 0.01518441, 0.09591688, 0.02238470, 0.08126640, 0.08236821,
--0.24802119, 0.89516402, 0.32029647, 0.07188887, -0.00220366, 0.00344025, -0.00277284, 0.00358963, -0.08668007, -0.02205910,
--0.05289669, -0.03535201, -0.01188017, -0.06456872, -0.09321006, -0.00009617, -0.15804070, 0.24632041, 0.90166119, 0.19250690,
- 0.17264619, -0.09699155, -0.00567329, -0.00897700, -0.01442565, -0.01939390, 0.03702127, -0.02999862, -0.04385696, -0.05232394,
--0.03339177, 0.03905964, -0.00281424, -0.29213275, 0.02892968, 0.90257613, -0.21546058, -0.18070946, 0.09014567, 0.04117230,
--0.01029696, -0.00329116, -0.03354346, 0.02937079, 0.01274208, -0.01260649, -0.03505571, -0.01020645, 0.03787209, 0.12132165,
--0.20826840, 0.81556933, -0.43874351, 0.21518682, -0.14564290, -0.05210031, 0.07124563, 0.06127983, -0.00457321, 0.01740496,
- 0.04185176, 0.00128036, -0.05033693, -0.01890046, 0.06221734, 0.10280078, -0.03738531, 0.04830209, -0.08408293, -0.46409009,
--0.83936263, -0.14817619, -0.13135927, 0.04563506, 0.08340661, 0.04040200, 0.00044396, -0.01365972, 0.01228951, 0.01078273,
- 0.09205406, -0.03791500, 0.07135889, 0.08158339, 0.06298278, -0.22875755, -0.92917558, -0.11248260, 0.17801883, -0.03971674,
--0.07491915, 0.06477287, 0.04635713, 0.01856159, 0.00130895, -0.01991604, 0.02358176, -0.09376056, 0.02782280, -0.04691559,
- 0.13749249, 0.31383132, 0.92274602, 0.04727419, 0.09765196, -0.02108945, 0.00626005, 0.05193322, 0.02009133, 0.03094066,
- 0.04573470, 0.00451733, 0.00240169, -0.00982355, -0.03546208, -0.14156875, -0.02480689, 0.22997442, 0.09778317, 0.88834235,
--0.32797611, -0.00079977, 0.04917079, 0.06977359, 0.06451185, 0.07816204, 0.03119314, 0.01136506, 0.01062006, 0.00632783,
- 0.03241828, -0.03318847, -0.01350502, -0.30055361, 0.07265375, 0.17308022, 0.88795796, -0.23231020, -0.08932700, 0.11759604,
- 0.00590705, 0.03525351, 0.00840466, 0.04389942, 0.04387629, 0.04003275, 0.01772966, 0.02709780, -0.02393282, 0.02766178,
- 0.00342983, -0.33882220, 0.76612668, 0.44061716, -0.28414784, -0.09364014, 0.03694060, 0.01124120, 0.01130268, -0.02869682,
--0.07428963, -0.03504754, 0.05874942, 0.01196795, 0.02003875, 0.00787152, -0.01605561, 0.04501257, -0.06959958, -0.13015784,
--0.05738065, 0.04681625, 0.06668700, -0.04492094, 0.02927765, -0.94404277, 0.19243952, 0.09504337, -0.12540826, 0.05394317,
--0.07972638, -0.02145188, 0.00136427, 0.01964678, 0.06667373, 0.06204535, 0.17302394, 0.22005905, 0.58329964, -0.68440447,
- 0.19628796, 0.15718011, -0.12481840, -0.08222507, 0.11780870, 0.03798206, -0.01818866, 0.00892766, 0.05582263, 0.01126832,
--0.00973589, 0.00697442, -0.09937902, 0.06621185, -0.19452202, -0.80004569, -0.13946094, -0.48990700, -0.17595191, -0.00798873,
--0.06121856, 0.08768040, -0.04507631, 0.00448896, 0.01153941, -0.04711652, -0.01050749, -0.01660047, -0.03007159, -0.01468906,
- 0.12848053, 0.13859838, 0.93863771, -0.22250065, -0.14841278, 0.04666032, -0.06344813, -0.01915105, -0.01840150, -0.02389410,
--0.01245496, 0.05023402, 0.02125840, 0.02467318, -0.01893022, -0.00889647, 0.00551817, 0.00481915, -0.40626968, -0.89028236,
- 0.18261687, -0.03852330, 0.02621926, -0.05420122, -0.01704117, -0.00072893, -0.02694170, -0.04335124, 0.02256467, 0.00642301,
--0.01619484, -0.00871160, 0.00400065, -0.00488820, -0.00752173, -0.00170603, 0.89554989, -0.41825934, -0.08725803, -0.09051404,
--0.00916236, -0.02959065, -0.07268075, -0.00816626, -0.00314215, -0.01941078, -0.00036782, -0.00188655, -0.02107724, -0.00771657,
--0.00448194, -0.00387517, 0.00082998, 0.00202471},
-{ 0.00167296, -0.00647772, -0.00604394, 0.01490810, -0.00837664, 0.00246438, 0.02082153, 0.01216715, 0.01001396, -0.02850860,
--0.01187868, -0.00113289, 0.04140237, -0.11084998, 0.16102260, 0.20084170, -0.28969446, -0.91312256, 0.00087788, -0.00136895,
- 0.00004622, 0.00578894, 0.00524119, -0.00044550, 0.00948906, -0.00396910, -0.03312197, -0.00075487, 0.00987494, -0.02088734,
- 0.09835550, -0.20080342, 0.13687782, -0.16111863, -0.90089988, 0.30312999, 0.00248784, -0.00975419, -0.01617200, 0.00699371,
--0.02151635, -0.01625774, -0.01262800, 0.02588781, -0.05620764, -0.13651454, 0.04242442, -0.02615307, 0.20497288, -0.20422909,
- 0.14184406, 0.89712919, 0.01758042, 0.25447787, -0.00207668, -0.00260329, 0.00724812, -0.01007749, 0.00806242, -0.03089729,
--0.01161934, -0.00618676, -0.10327342, -0.10160272, 0.11919283, 0.20781533, 0.11564869, -0.19072476, 0.86402008, -0.24650846,
- 0.24684161, 0.04775750, 0.00486888, -0.01735569, -0.01868000, -0.01870386, -0.03243262, -0.05883701, -0.03433371, 0.10441236,
--0.22831067, -0.22837988, 0.15082544, -0.21313767, 0.13215611, -0.78096079, -0.32270595, -0.21307018, 0.17339271, -0.05435742,
--0.00940813, 0.00272520, 0.00542917, -0.05232991, -0.01280809, -0.10773627, -0.17626479, 0.03719285, -0.26297104, -0.21780618,
- 0.21406665, 0.15202177, 0.75911044, 0.38627481, -0.16504189, -0.10242997, -0.02394939, -0.06018959, 0.00994733, -0.02617197,
--0.01543723, -0.10320051, -0.03010481, -0.19098072, -0.06893233, 0.12253174, -0.25556092, -0.31989059, 0.09542655, 0.72712041,
--0.43108921, -0.01568072, -0.16532685, 0.06646835, -0.08885408, -0.00050364, -0.01791050, 0.00245405, 0.00204794, -0.17948691,
--0.05193881, -0.16329387, -0.13676259, 0.01214133, -0.30994612, -0.00687734, 0.63254090, -0.47180795, -0.35409214, 0.23658315,
- 0.11170294, 0.05229887, -0.06107035, -0.01094212, 0.01523854, -0.01608284, -0.03739206, -0.23864328, -0.03958494, -0.19305719,
--0.26019058, 0.24108257, -0.55933566, 0.40623396, -0.53367968, -0.08930957, -0.00599383, -0.00050845, 0.06960811, 0.02664961,
- 0.01464197, -0.00486781, -0.01905736, 0.01437578, 0.02379930, -0.26639588, 0.05208876, -0.43525002, -0.63009424, 0.05251889,
- 0.56732782, -0.06731164, -0.03705909, -0.03253946, 0.00950673, -0.07941760, 0.02388267, -0.01258409, -0.00343524, 0.00148711,
--0.00362107, 0.03981813, -0.07235214, -0.46180041, -0.05595288, -0.55699317, 0.61935853, -0.25379716, 0.06796783, 0.01039267,
--0.06329171, -0.02143024, 0.09406929, -0.00799203, -0.01419805, -0.00603024, 0.01313145, 0.00091161, -0.00212107, -0.02405340,
- 0.07146405, -0.76695326, -0.14841817, 0.60372663, -0.01478424, 0.06522462, 0.08580016, -0.05817981, 0.02438942, 0.04840904,
- 0.02934363, -0.02239678, -0.00582247, -0.00091312, -0.00394148, -0.00285276, -0.03435745, 0.05277435, 0.17882781, -0.06194164,
- 0.27321118, 0.01840179, -0.10188148, -0.33168524, -0.03491221, 0.67351789, 0.37017376, 0.32083717, 0.09737800, -0.20998084,
--0.10725041, 0.06379186, 0.02169903, -0.02031584, 0.05623799, -0.18300962, -0.17337803, 0.08915172, -0.53835537, -0.08547263,
- 0.15163321, 0.56732906, 0.21878115, 0.37274266, 0.26206918, 0.13443927, 0.09178695, -0.03276324, -0.01131664, -0.00236369,
- 0.00772568, 0.01008805, -0.17122615, 0.15301569, 0.40135484, -0.06058913, 0.56405128, -0.05176853, 0.24544337, 0.62448073,
- 0.07265009, -0.01198695, 0.05151774, -0.03678498, 0.01886154, 0.03724094, 0.01393667, 0.00758055, -0.00254297, 0.00537118,
- 0.24169707, -0.41735970, -0.67564355, -0.09270478, 0.53106033, 0.06214579, 0.02574404, 0.09943837, 0.03032542, 0.02194476,
- 0.06369772, -0.00133741, 0.01301113, 0.01508494, 0.00036111, -0.00278870, 0.00139205, 0.00015792, -0.43347887, 0.69923146,
--0.55406563, -0.01102231, 0.01347767, 0.07012139, -0.02530164, 0.06803192, 0.01177196, 0.04374491, 0.04073027, 0.04037438,
- 0.00167330, -0.01807065, -0.00425562, 0.00149653, -0.00035119, -0.00172888, 0.84785495, 0.52289580, 0.01067734, -0.00859194,
- 0.01685964, 0.00481442, 0.00434738, 0.07592695, 0.01419942, 0.01005336, 0.03316937, 0.00360465, 0.00435039, 0.00029122,
- 0.00171268, 0.00198919, -0.00046889, -0.00094176}};
-
-/* right KLT transforms */
-const double WebRtcIsac_kKltT2Gain[3][36] = {
-{ 0.14572837, -0.45446306, 0.61990621, -0.52197033, 0.32145074, -0.11026900, -0.20698282, 0.48962182, -0.27127933, -0.33627476,
- 0.65094037, -0.32715751, 0.40262573, -0.47844405, -0.33876075, 0.44130653, 0.37383966, -0.39964662, -0.51730480, 0.06611973,
- 0.49030187, 0.47512886, -0.02141226, -0.51129451, -0.58578569, -0.39132064, -0.13187771, 0.15649421, 0.40735596, 0.54396897,
- 0.40381276, 0.40904942, 0.41179766, 0.41167576, 0.40840251, 0.40468132},
-{-0.11368135, 0.34815515, -0.56434996, 0.61130763, -0.39970336, 0.11795708, 0.28514257, -0.58879243, 0.32775812, 0.27024886,
--0.56251299, 0.27411037, 0.42649186, -0.44080232, -0.36408215, 0.35932457, 0.43592895, -0.41484213, -0.49813030, -0.00012592,
- 0.49865688, 0.47634953, 0.01094246, -0.52552726, -0.56154082, -0.41110686, -0.14170764, 0.15946614, 0.40818082, 0.55094554,
- 0.40051601, 0.41084781, 0.41567800, 0.41450700, 0.40871872, 0.39891823},
-{-0.10719481, 0.34796287, -0.54573957, 0.59521001, -0.43943367, 0.14907223, 0.26554957, -0.59549939, 0.36760692, 0.26040652,
--0.55268701, 0.25778784, 0.38994096, -0.45282773, -0.37975656, 0.40213055, 0.43052647, -0.38937904, -0.52698359, 0.02788094,
- 0.48284286, 0.47792474, 0.02557759, -0.50922240, -0.57699826, -0.39476779, -0.14708238, 0.12742149, 0.37835245, 0.57464021,
- 0.39408127, 0.40327462, 0.40993655, 0.41419345, 0.41506301, 0.41253853}};
-
-const double WebRtcIsac_kKltT2Shape[3][36] = {
-{ 0.13427386, -0.35132558, 0.52506528, -0.59419077, 0.45075085, -0.16312057, 0.29857439, -0.58660147, 0.34265431, 0.20879510,
--0.56063262, 0.30238345, 0.43308283, -0.41186999, -0.35288681, 0.42768996, 0.36094634, -0.45284910, -0.47116680, 0.02893449,
- 0.54326135, 0.45249040, -0.06264420, -0.52283830, 0.57137758, 0.44298139, 0.12617554, -0.20819946, -0.42324603, -0.48876443,
- 0.39597050, 0.40713935, 0.41389880, 0.41512486, 0.41130400, 0.40575001},
-{ 0.16540737, -0.43379435, 0.58165221, -0.55154773, 0.35734028, -0.11935912, 0.29434254, -0.55954817, 0.23549804, 0.33087258,
--0.58848503, 0.29835834, 0.45464789, -0.38316155, -0.41689708, 0.35607296, 0.41260747, -0.41910198, -0.48633899, -0.04144955,
- 0.47824583, 0.51050942, 0.01000345, -0.52184032, 0.53488229, 0.42641051, 0.17049774, -0.15849613, -0.43229355, -0.53945045,
- 0.39582002, 0.41033103, 0.41788713, 0.41688080, 0.41081697, 0.39719658},
-{ 0.13386268, -0.37919915, 0.54989123, -0.57663572, 0.42402636, -0.15362720, 0.29641577, -0.58806770, 0.31381040, 0.26524954,
--0.56271012, 0.28431868, 0.42699898, -0.41058922, -0.40408270, 0.39215865, 0.40788513, -0.40699735, -0.49846482, -0.01521208,
- 0.48756040, 0.49479418, -0.00347672, -0.51841384, 0.55513106, 0.41683793, 0.15131217, -0.15613621, -0.41029341, -0.54996461,
- 0.39402116, 0.40965305, 0.41862791, 0.41730770, 0.41089648, 0.39837262}};
-
-/* means of log gains and LAR coefficients*/
-const double WebRtcIsac_kLpcMeansGain[3][12] = {
-{-6.86881911, -5.35075273, -6.86792680, -5.36200897, -6.86401538, -5.36921533, -6.86802969, -5.36893966, -6.86538097, -5.36315063,
--6.85535304, -5.35155315},
-{-6.12914600, -4.78070092, -6.12971780, -4.78382183, -6.12858525, -4.79362198, -6.12926491, -4.79017481, -6.12102401, -4.78346122,
--6.11441152, -4.78019228},
-{-5.67273484, -3.73876311, -5.65246094, -3.71407895, -5.61716443, -3.68814580, -5.58804560, -3.66334094, -5.54189577, -3.63845640,
--5.49293185, -3.61760203}};
-
-const double WebRtcIsac_kLpcMeansShape[3][108] = {
-{-0.91232981, 0.26258634, -0.33716701, 0.08477430, -0.03378426, 0.14423909, 0.07036185, 0.06155019, 0.01490385, 0.04138740,
- 0.01427317, 0.01288970, 0.83872106, 0.25750199, 0.07988929, -0.01957923, 0.00831390, 0.01770300, -0.90957164, 0.25732216,
--0.33385344, 0.08735740, -0.03715332, 0.14584917, 0.06998990, 0.06131968, 0.01504379, 0.04067339, 0.01428039, 0.01406460,
- 0.83846243, 0.26169862, 0.08109025, -0.01767055, 0.00970539, 0.01954310, -0.90490803, 0.24656405, -0.33578607, 0.08843286,
--0.03749139, 0.14443959, 0.07214669, 0.06170993, 0.01449947, 0.04134309, 0.01314762, 0.01413471, 0.83895203, 0.26748062,
- 0.08197507, -0.01781298, 0.00885967, 0.01922394, -0.90922472, 0.24495889, -0.33921540, 0.08877169, -0.03581332, 0.14199172,
- 0.07444032, 0.06185940, 0.01502054, 0.04185113, 0.01276579, 0.01355457, 0.83645358, 0.26631720, 0.08119697, -0.01835449,
- 0.00788512, 0.01846446, -0.90482253, 0.24658310, -0.34019734, 0.08281090, -0.03486038, 0.14359248, 0.07401336, 0.06001471,
- 0.01528421, 0.04254560, 0.01321472, 0.01240799, 0.83857127, 0.26281654, 0.08174380, -0.02099842, 0.00755176, 0.01699448,
--0.90132307, 0.25174308, -0.33838268, 0.07883863, -0.02877906, 0.14105407, 0.07220290, 0.06000352, 0.01684879, 0.04226844,
- 0.01331331, 0.01269244, 0.83832138, 0.25467485, 0.08118028, -0.02120528, 0.00747832, 0.01567212},
-{-1.11639718, 0.35377266, 0.00798929, 0.20165280, 0.07656104, 0.10629964, 0.04894160, 0.10955305, -0.01806405, 0.05082282,
- 0.01730794, 0.01345957, 0.73717782, 0.05952284, 0.03176204, 0.08195122, 0.01253148, 0.02253385, -1.12053537, 0.35523538,
- 0.00859646, 0.20007706, 0.07715852, 0.10754596, 0.05165976, 0.10927703, -0.01554395, 0.05178866, 0.01752534, 0.01343468,
- 0.73489046, 0.06395167, 0.03287798, 0.07972374, 0.01293550, 0.02300929, -1.11772179, 0.35457623, 0.01205524, 0.19926481,
- 0.08000866, 0.10817921, 0.05052481, 0.11016167, -0.01552091, 0.05155510, 0.01787163, 0.01343778, 0.73142568, 0.06840830,
- 0.03316828, 0.07902608, 0.01525042, 0.02178127, -1.12120164, 0.36405233, 0.00630305, 0.19799738, 0.07829690, 0.10727588,
- 0.04017317, 0.10437949, -0.01844109, 0.05021700, 0.01561726, 0.01226571, 0.73438044, 0.06947982, 0.03396317, 0.07858683,
- 0.01367105, 0.02041955, -1.12146187, 0.35952226, 0.00340090, 0.19700813, 0.07938222, 0.10904137, 0.03921216, 0.10531403,
--0.01833415, 0.04956231, 0.01399539, 0.01323582, 0.74378099, 0.07059589, 0.03367692, 0.08151462, 0.01182040, 0.02075577,
--1.11245254, 0.35234230, 0.00687490, 0.20204252, 0.07813186, 0.11081259, 0.04634665, 0.11073238, -0.01637954, 0.05104577,
- 0.01675122, 0.01448696, 0.74013627, 0.06239059, 0.03129412, 0.08207461, 0.01249475, 0.02189238},
-{-1.27118948, 0.35834331, -0.33499347, 0.13524073, 0.04829079, 0.19542773, 0.05273835, 0.04157974, -0.01755227, 0.01513442,
- 0.00386630, 0.02199463, 1.14439142, 0.21903073, 0.14750213, 0.12743356, 0.08463334, 0.06839691, -1.28367777, 0.35556287,
--0.33809405, 0.13627881, 0.04939309, 0.19642571, 0.05354373, 0.04099247, -0.01787481, 0.01472425, 0.00391474, 0.02150716,
- 1.14739079, 0.21840872, 0.14643624, 0.12724347, 0.08390642, 0.06811938, -1.29007667, 0.35159558, -0.34154267, 0.13295849,
- 0.04883602, 0.19587595, 0.05452759, 0.04174703, -0.01782110, 0.01388270, 0.00374754, 0.02138105, 1.14333767, 0.21690116,
- 0.14544599, 0.12606728, 0.08314168, 0.06771389, -1.29856471, 0.35239315, -0.34238732, 0.13277553, 0.04722712, 0.19233156,
- 0.05366901, 0.04328110, -0.01657749, 0.01444736, 0.00438108, 0.02102563, 1.13548397, 0.21537812, 0.14357377, 0.12525845,
- 0.08230994, 0.06722511, -1.30663540, 0.34366563, -0.34205544, 0.12861679, 0.04655851, 0.18864359, 0.05351285, 0.04358693,
--0.01604498, 0.01431907, 0.00395326, 0.02082299, 1.12207794, 0.21167325, 0.14212491, 0.12418671, 0.08155467, 0.06639789,
--1.31011673, 0.33686271, -0.34379843, 0.12169569, 0.04480323, 0.18637557, 0.05374078, 0.04260827, -0.01588226, 0.01378294,
- 0.00396009, 0.02112406, 1.10466984, 0.20905894, 0.14107033, 0.12303074, 0.08047136, 0.06588031}};
-
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_tables.h
deleted file mode 100644
index 604d963..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/lpc_tables.h
+++ /dev/null
@@ -1,114 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * lpc_tables.h
- *
- * header file for coding tables for the LPC coefficients
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_
-
-#include "structs.h"
-
-#include "settings.h"
-
-#define KLT_STEPSIZE 1.00000000
-#define KLT_NUM_AVG_GAIN 0
-#define KLT_NUM_AVG_SHAPE 0
-#define KLT_NUM_MODELS 3
-#define LPC_GAIN_SCALE 4.000f
-#define LPC_LOBAND_SCALE 2.100f
-#define LPC_LOBAND_ORDER ORDERLO
-#define LPC_HIBAND_SCALE 0.450f
-#define LPC_HIBAND_ORDER ORDERHI
-#define LPC_GAIN_ORDER 2
-
-#define LPC_SHAPE_ORDER (LPC_LOBAND_ORDER + LPC_HIBAND_ORDER)
-
-#define KLT_ORDER_GAIN (LPC_GAIN_ORDER * SUBFRAMES)
-#define KLT_ORDER_SHAPE (LPC_SHAPE_ORDER * SUBFRAMES)
-/* indices of KLT coefficients used */
-extern const WebRtc_UWord16 WebRtcIsac_kQKltSelIndGain[12];
-
-extern const WebRtc_UWord16 WebRtcIsac_kQKltSelIndShape[108];
-
-/* cdf array for model indicator */
-extern const WebRtc_UWord16 WebRtcIsac_kQKltModelCdf[KLT_NUM_MODELS+1];
-
-/* pointer to cdf array for model indicator */
-extern const WebRtc_UWord16 *WebRtcIsac_kQKltModelCdfPtr[1];
-
-/* initial cdf index for decoder of model indicator */
-extern const WebRtc_UWord16 WebRtcIsac_kQKltModelInitIndex[1];
-
-/* offset to go from rounded value to quantization index */
-extern const short WebRtcIsac_kQKltQuantMinGain[12];
-
-extern const short WebRtcIsac_kQKltQuantMinShape[108];
-
-/* maximum quantization index */
-extern const WebRtc_UWord16 WebRtcIsac_kQKltMaxIndGain[12];
-
-extern const WebRtc_UWord16 WebRtcIsac_kQKltMaxIndShape[108];
-
-/* index offset */
-extern const WebRtc_UWord16 WebRtcIsac_kQKltOffsetGain[KLT_NUM_MODELS][12];
-
-extern const WebRtc_UWord16 WebRtcIsac_kQKltOffsetShape[KLT_NUM_MODELS][108];
-
-/* initial cdf index for KLT coefficients */
-extern const WebRtc_UWord16 WebRtcIsac_kQKltInitIndexGain[KLT_NUM_MODELS][12];
-
-extern const WebRtc_UWord16 WebRtcIsac_kQKltInitIndexShape[KLT_NUM_MODELS][108];
-
-/* offsets for quantizer representation levels */
-extern const WebRtc_UWord16 WebRtcIsac_kQKltOfLevelsGain[3];
-
-extern const WebRtc_UWord16 WebRtcIsac_kQKltOfLevelsShape[3];
-
-/* quantizer representation levels */
-extern const double WebRtcIsac_kQKltLevelsGain[1176];
-
-extern const double WebRtcIsac_kQKltLevelsShape[1735];
-
-/* cdf tables for quantizer indices */
-extern const WebRtc_UWord16 WebRtcIsac_kQKltCdfGain[1212];
-
-extern const WebRtc_UWord16 WebRtcIsac_kQKltCdfShape[2059];
-
-/* pointers to cdf tables for quantizer indices */
-extern const WebRtc_UWord16 *WebRtcIsac_kQKltCdfPtrGain[KLT_NUM_MODELS][12];
-
-extern const WebRtc_UWord16 *WebRtcIsac_kQKltCdfPtrShape[KLT_NUM_MODELS][108];
-
-/* code length for all coefficients using different models */
-extern const double WebRtcIsac_kQKltCodeLenGain[392];
-
-extern const double WebRtcIsac_kQKltCodeLenShape[578];
-
-/* left KLT transforms */
-extern const double WebRtcIsac_kKltT1Gain[KLT_NUM_MODELS][4];
-
-extern const double WebRtcIsac_kKltT1Shape[KLT_NUM_MODELS][324];
-
-/* right KLT transforms */
-extern const double WebRtcIsac_kKltT2Gain[KLT_NUM_MODELS][36];
-
-extern const double WebRtcIsac_kKltT2Shape[KLT_NUM_MODELS][36];
-
-/* means of log gains and LAR coefficients */
-extern const double WebRtcIsac_kLpcMeansGain[KLT_NUM_MODELS][12];
-
-extern const double WebRtcIsac_kLpcMeansShape[KLT_NUM_MODELS][108];
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_LPC_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/os_specific_inline.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/os_specific_inline.h
deleted file mode 100644
index c469c2e..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/os_specific_inline.h
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
-
-#include <math.h>
-#include "typedefs.h"
-
-// TODO(turaj): switch to WEBRTC_POSIX when available
-#if defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
-#define WebRtcIsac_lrint lrint
-#elif (defined(WEBRTC_ARCH_X86) && defined(WIN32))
-static __inline long int WebRtcIsac_lrint(double x_dbl) {
- long int x_int;
-
- __asm {
- fld x_dbl
- fistp x_int
- };
-
- return x_int;
-}
-#else // Do a slow but correct implementation of lrint
-
-static __inline long int WebRtcIsac_lrint(double x_dbl) {
- long int x_int;
- x_int = (long int)floor(x_dbl + 0.499999999999);
- return x_int;
-}
-
-#endif
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_OS_SPECIFIC_INLINE_H_
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_estimator.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_estimator.c
deleted file mode 100644
index 75525f6..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_estimator.c
+++ /dev/null
@@ -1,622 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "pitch_estimator.h"
-
-#include <math.h>
-#include <memory.h>
-#ifdef WEBRTC_ANDROID
-#include <stdlib.h>
-#endif
-
-static const double kInterpolWin[8] = {-0.00067556028640, 0.02184247643159, -0.12203175715679, 0.60086484101160,
- 0.60086484101160, -0.12203175715679, 0.02184247643159, -0.00067556028640};
-
-/* interpolation filter */
-__inline static void IntrepolFilter(double *data_ptr, double *intrp)
-{
- *intrp = kInterpolWin[0] * data_ptr[-3];
- *intrp += kInterpolWin[1] * data_ptr[-2];
- *intrp += kInterpolWin[2] * data_ptr[-1];
- *intrp += kInterpolWin[3] * data_ptr[0];
- *intrp += kInterpolWin[4] * data_ptr[1];
- *intrp += kInterpolWin[5] * data_ptr[2];
- *intrp += kInterpolWin[6] * data_ptr[3];
- *intrp += kInterpolWin[7] * data_ptr[4];
-}
-
-
-/* 2D parabolic interpolation */
-/* probably some 0.5 factors can be eliminated, and the square-roots can be removed from the Cholesky fact. */
-__inline static void Intrpol2D(double T[3][3], double *x, double *y, double *peak_val)
-{
- double c, b[2], A[2][2];
- double t1, t2, d;
- double delta1, delta2;
-
-
- // double T[3][3] = {{-1.25, -.25,-.25}, {-.25, .75, .75}, {-.25, .75, .75}};
- // should result in: delta1 = 0.5; delta2 = 0.0; peak_val = 1.0
-
- c = T[1][1];
- b[0] = 0.5 * (T[1][2] + T[2][1] - T[0][1] - T[1][0]);
- b[1] = 0.5 * (T[1][0] + T[2][1] - T[0][1] - T[1][2]);
- A[0][1] = -0.5 * (T[0][1] + T[2][1] - T[1][0] - T[1][2]);
- t1 = 0.5 * (T[0][0] + T[2][2]) - c;
- t2 = 0.5 * (T[2][0] + T[0][2]) - c;
- d = (T[0][1] + T[1][2] + T[1][0] + T[2][1]) - 4.0 * c - t1 - t2;
- A[0][0] = -t1 - 0.5 * d;
- A[1][1] = -t2 - 0.5 * d;
-
- /* deal with singularities or ill-conditioned cases */
- if ( (A[0][0] < 1e-7) || ((A[0][0] * A[1][1] - A[0][1] * A[0][1]) < 1e-7) ) {
- *peak_val = T[1][1];
- return;
- }
-
- /* Cholesky decomposition: replace A by upper-triangular factor */
- A[0][0] = sqrt(A[0][0]);
- A[0][1] = A[0][1] / A[0][0];
- A[1][1] = sqrt(A[1][1] - A[0][1] * A[0][1]);
-
- /* compute [x; y] = -0.5 * inv(A) * b */
- t1 = b[0] / A[0][0];
- t2 = (b[1] - t1 * A[0][1]) / A[1][1];
- delta2 = t2 / A[1][1];
- delta1 = 0.5 * (t1 - delta2 * A[0][1]) / A[0][0];
- delta2 *= 0.5;
-
- /* limit norm */
- t1 = delta1 * delta1 + delta2 * delta2;
- if (t1 > 1.0) {
- delta1 /= t1;
- delta2 /= t1;
- }
-
- *peak_val = 0.5 * (b[0] * delta1 + b[1] * delta2) + c;
-
- *x += delta1;
- *y += delta2;
-}
-
-
-static void PCorr(const double *in, double *outcorr)
-{
- double sum, ysum, prod;
- const double *x, *inptr;
- int k, n;
-
- //ysum = 1e-6; /* use this with float (i.s.o. double)! */
- ysum = 1e-13;
- sum = 0.0;
- x = in + PITCH_MAX_LAG/2 + 2;
- for (n = 0; n < PITCH_CORR_LEN2; n++) {
- ysum += in[n] * in[n];
- sum += x[n] * in[n];
- }
-
- outcorr += PITCH_LAG_SPAN2 - 1; /* index of last element in array */
- *outcorr = sum / sqrt(ysum);
-
- for (k = 1; k < PITCH_LAG_SPAN2; k++) {
- ysum -= in[k-1] * in[k-1];
- ysum += in[PITCH_CORR_LEN2 + k - 1] * in[PITCH_CORR_LEN2 + k - 1];
- sum = 0.0;
- inptr = &in[k];
- prod = x[0] * inptr[0];
- for (n = 1; n < PITCH_CORR_LEN2; n++) {
- sum += prod;
- prod = x[n] * inptr[n];
- }
- sum += prod;
- outcorr--;
- *outcorr = sum / sqrt(ysum);
- }
-}
-
-
-void WebRtcIsac_InitializePitch(const double *in,
- const double old_lag,
- const double old_gain,
- PitchAnalysisStruct *State,
- double *lags)
-{
- double buf_dec[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2];
- double ratio, log_lag, gain_bias;
- double bias;
- double corrvec1[PITCH_LAG_SPAN2];
- double corrvec2[PITCH_LAG_SPAN2];
- int m, k;
- // Allocating 10 extra entries at the begining of the CorrSurf
- double corrSurfBuff[10 + (2*PITCH_BW+3)*(PITCH_LAG_SPAN2+4)];
- double* CorrSurf[2*PITCH_BW+3];
- double *CorrSurfPtr1, *CorrSurfPtr2;
- double LagWin[3] = {0.2, 0.5, 0.98};
- int ind1, ind2, peaks_ind, peak, max_ind;
- int peaks[PITCH_MAX_NUM_PEAKS];
- double adj, gain_tmp;
- double corr, corr_max;
- double intrp_a, intrp_b, intrp_c, intrp_d;
- double peak_vals[PITCH_MAX_NUM_PEAKS];
- double lags1[PITCH_MAX_NUM_PEAKS];
- double lags2[PITCH_MAX_NUM_PEAKS];
- double T[3][3];
- int row;
-
- for(k = 0; k < 2*PITCH_BW+3; k++)
- {
- CorrSurf[k] = &corrSurfBuff[10 + k * (PITCH_LAG_SPAN2+4)];
- }
- /* reset CorrSurf matrix */
- memset(corrSurfBuff, 0, sizeof(double) * (10 + (2*PITCH_BW+3) * (PITCH_LAG_SPAN2+4)));
-
- //warnings -DH
- max_ind = 0;
- peak = 0;
-
- /* copy old values from state buffer */
- memcpy(buf_dec, State->dec_buffer, sizeof(double) * (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2));
-
- /* decimation; put result after the old values */
- WebRtcIsac_DecimateAllpass(in, State->decimator_state, PITCH_FRAME_LEN,
- &buf_dec[PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2]);
-
- /* low-pass filtering */
- for (k = PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2; k < PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2+2; k++)
- buf_dec[k] += 0.75 * buf_dec[k-1] - 0.25 * buf_dec[k-2];
-
- /* copy end part back into state buffer */
- memcpy(State->dec_buffer, buf_dec+PITCH_FRAME_LEN/2, sizeof(double) * (PITCH_CORR_LEN2+PITCH_CORR_STEP2+PITCH_MAX_LAG/2-PITCH_FRAME_LEN/2+2));
-
- /* compute correlation for first and second half of the frame */
- PCorr(buf_dec, corrvec1);
- PCorr(buf_dec + PITCH_CORR_STEP2, corrvec2);
-
- /* bias towards pitch lag of previous frame */
- log_lag = log(0.5 * old_lag);
- gain_bias = 4.0 * old_gain * old_gain;
- if (gain_bias > 0.8) gain_bias = 0.8;
- for (k = 0; k < PITCH_LAG_SPAN2; k++)
- {
- ratio = log((double) (k + (PITCH_MIN_LAG/2-2))) - log_lag;
- bias = 1.0 + gain_bias * exp(-5.0 * ratio * ratio);
- corrvec1[k] *= bias;
- }
-
- /* taper correlation functions */
- for (k = 0; k < 3; k++) {
- gain_tmp = LagWin[k];
- corrvec1[k] *= gain_tmp;
- corrvec2[k] *= gain_tmp;
- corrvec1[PITCH_LAG_SPAN2-1-k] *= gain_tmp;
- corrvec2[PITCH_LAG_SPAN2-1-k] *= gain_tmp;
- }
-
- corr_max = 0.0;
- /* fill middle row of correlation surface */
- ind1 = 0;
- ind2 = 0;
- CorrSurfPtr1 = &CorrSurf[PITCH_BW][2];
- for (k = 0; k < PITCH_LAG_SPAN2; k++) {
- corr = corrvec1[ind1++] + corrvec2[ind2++];
- CorrSurfPtr1[k] = corr;
- if (corr > corr_max) {
- corr_max = corr; /* update maximum */
- max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
- }
- }
- /* fill first and last rows of correlation surface */
- ind1 = 0;
- ind2 = PITCH_BW;
- CorrSurfPtr1 = &CorrSurf[0][2];
- CorrSurfPtr2 = &CorrSurf[2*PITCH_BW][PITCH_BW+2];
- for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW; k++) {
- ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
- adj = 0.2 * ratio * (2.0 - ratio); /* adjustment factor; inverse parabola as a function of ratio */
- corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
- CorrSurfPtr1[k] = corr;
- if (corr > corr_max) {
- corr_max = corr; /* update maximum */
- max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
- }
- corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
- CorrSurfPtr2[k] = corr;
- if (corr > corr_max) {
- corr_max = corr; /* update maximum */
- max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
- }
- }
- /* fill second and next to last rows of correlation surface */
- ind1 = 0;
- ind2 = PITCH_BW-1;
- CorrSurfPtr1 = &CorrSurf[1][2];
- CorrSurfPtr2 = &CorrSurf[2*PITCH_BW-1][PITCH_BW+1];
- for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW+1; k++) {
- ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
- adj = 0.9 * ratio * (2.0 - ratio); /* adjustment factor; inverse parabola as a function of ratio */
- corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
- CorrSurfPtr1[k] = corr;
- if (corr > corr_max) {
- corr_max = corr; /* update maximum */
- max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
- }
- corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
- CorrSurfPtr2[k] = corr;
- if (corr > corr_max) {
- corr_max = corr; /* update maximum */
- max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
- }
- }
- /* fill remainder of correlation surface */
- for (m = 2; m < PITCH_BW; m++) {
- ind1 = 0;
- ind2 = PITCH_BW - m; /* always larger than ind1 */
- CorrSurfPtr1 = &CorrSurf[m][2];
- CorrSurfPtr2 = &CorrSurf[2*PITCH_BW-m][PITCH_BW+2-m];
- for (k = 0; k < PITCH_LAG_SPAN2-PITCH_BW+m; k++) {
- ratio = ((double) (ind1 + 12)) / ((double) (ind2 + 12));
- adj = ratio * (2.0 - ratio); /* adjustment factor; inverse parabola as a function of ratio */
- corr = adj * (corrvec1[ind1] + corrvec2[ind2]);
- CorrSurfPtr1[k] = corr;
- if (corr > corr_max) {
- corr_max = corr; /* update maximum */
- max_ind = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
- }
- corr = adj * (corrvec1[ind2++] + corrvec2[ind1++]);
- CorrSurfPtr2[k] = corr;
- if (corr > corr_max) {
- corr_max = corr; /* update maximum */
- max_ind = (int)(&CorrSurfPtr2[k] - &CorrSurf[0][0]);
- }
- }
- }
-
- /* threshold value to qualify as a peak */
- corr_max *= 0.6;
-
- peaks_ind = 0;
- /* find peaks */
- for (m = 1; m < PITCH_BW+1; m++) {
- if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
- CorrSurfPtr1 = &CorrSurf[m][2];
- for (k = 2; k < PITCH_LAG_SPAN2-PITCH_BW-2+m; k++) {
- corr = CorrSurfPtr1[k];
- if (corr > corr_max) {
- if ( (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+5)]) && (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+4)]) ) {
- if ( (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+4)]) && (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+5)]) ) {
- /* found a peak; store index into matrix */
- peaks[peaks_ind++] = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
- if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
- }
- }
- }
- }
- }
- for (m = PITCH_BW+1; m < 2*PITCH_BW; m++) {
- if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
- CorrSurfPtr1 = &CorrSurf[m][2];
- for (k = 2+m-PITCH_BW; k < PITCH_LAG_SPAN2-2; k++) {
- corr = CorrSurfPtr1[k];
- if (corr > corr_max) {
- if ( (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+5)]) && (corr > CorrSurfPtr1[k - (PITCH_LAG_SPAN2+4)]) ) {
- if ( (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+4)]) && (corr > CorrSurfPtr1[k + (PITCH_LAG_SPAN2+5)]) ) {
- /* found a peak; store index into matrix */
- peaks[peaks_ind++] = (int)(&CorrSurfPtr1[k] - &CorrSurf[0][0]);
- if (peaks_ind == PITCH_MAX_NUM_PEAKS) break;
- }
- }
- }
- }
- }
-
- if (peaks_ind > 0) {
- /* examine each peak */
- CorrSurfPtr1 = &CorrSurf[0][0];
- for (k = 0; k < peaks_ind; k++) {
- peak = peaks[k];
-
- /* compute four interpolated values around current peak */
- IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)], &intrp_a);
- IntrepolFilter(&CorrSurfPtr1[peak - 1 ], &intrp_b);
- IntrepolFilter(&CorrSurfPtr1[peak ], &intrp_c);
- IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)], &intrp_d);
-
- /* determine maximum of the interpolated values */
- corr = CorrSurfPtr1[peak];
- corr_max = intrp_a;
- if (intrp_b > corr_max) corr_max = intrp_b;
- if (intrp_c > corr_max) corr_max = intrp_c;
- if (intrp_d > corr_max) corr_max = intrp_d;
-
- /* determine where the peak sits and fill a 3x3 matrix around it */
- row = peak / (PITCH_LAG_SPAN2+4);
- lags1[k] = (double) ((peak - row * (PITCH_LAG_SPAN2+4)) + PITCH_MIN_LAG/2 - 4);
- lags2[k] = (double) (lags1[k] + PITCH_BW - row);
- if ( corr > corr_max ) {
- T[0][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
- T[2][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
- T[1][1] = corr;
- T[0][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
- T[2][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
- T[1][0] = intrp_a;
- T[0][1] = intrp_b;
- T[2][1] = intrp_c;
- T[1][2] = intrp_d;
- } else {
- if (intrp_a == corr_max) {
- lags1[k] -= 0.5;
- lags2[k] += 0.5;
- IntrepolFilter(&CorrSurfPtr1[peak - 2*(PITCH_LAG_SPAN2+5)], &T[0][0]);
- IntrepolFilter(&CorrSurfPtr1[peak - (2*PITCH_LAG_SPAN2+9)], &T[2][0]);
- T[1][1] = intrp_a;
- T[0][2] = intrp_b;
- T[2][2] = intrp_c;
- T[1][0] = CorrSurfPtr1[peak - (2*PITCH_LAG_SPAN2+9)];
- T[0][1] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
- T[2][1] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
- T[1][2] = corr;
- } else if (intrp_b == corr_max) {
- lags1[k] -= 0.5;
- lags2[k] -= 0.5;
- IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+6)], &T[0][0]);
- T[2][0] = intrp_a;
- T[1][1] = intrp_b;
- IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+3)], &T[0][2]);
- T[2][2] = intrp_d;
- T[1][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+5)];
- T[0][1] = CorrSurfPtr1[peak - 1];
- T[2][1] = corr;
- T[1][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
- } else if (intrp_c == corr_max) {
- lags1[k] += 0.5;
- lags2[k] += 0.5;
- T[0][0] = intrp_a;
- IntrepolFilter(&CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)], &T[2][0]);
- T[1][1] = intrp_c;
- T[0][2] = intrp_d;
- IntrepolFilter(&CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)], &T[2][2]);
- T[1][0] = CorrSurfPtr1[peak - (PITCH_LAG_SPAN2+4)];
- T[0][1] = corr;
- T[2][1] = CorrSurfPtr1[peak + 1];
- T[1][2] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
- } else {
- lags1[k] += 0.5;
- lags2[k] -= 0.5;
- T[0][0] = intrp_b;
- T[2][0] = intrp_c;
- T[1][1] = intrp_d;
- IntrepolFilter(&CorrSurfPtr1[peak + 2*(PITCH_LAG_SPAN2+4)], &T[0][2]);
- IntrepolFilter(&CorrSurfPtr1[peak + (2*PITCH_LAG_SPAN2+9)], &T[2][2]);
- T[1][0] = corr;
- T[0][1] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+4)];
- T[2][1] = CorrSurfPtr1[peak + (PITCH_LAG_SPAN2+5)];
- T[1][2] = CorrSurfPtr1[peak + (2*PITCH_LAG_SPAN2+9)];
- }
- }
-
- /* 2D parabolic interpolation gives more accurate lags and peak value */
- Intrpol2D(T, &lags1[k], &lags2[k], &peak_vals[k]);
- }
-
- /* determine the highest peak, after applying a bias towards short lags */
- corr_max = 0.0;
- for (k = 0; k < peaks_ind; k++) {
- corr = peak_vals[k] * pow(PITCH_PEAK_DECAY, log(lags1[k] + lags2[k]));
- if (corr > corr_max) {
- corr_max = corr;
- peak = k;
- }
- }
-
- lags1[peak] *= 2.0;
- lags2[peak] *= 2.0;
-
- if (lags1[peak] < (double) PITCH_MIN_LAG) lags1[peak] = (double) PITCH_MIN_LAG;
- if (lags2[peak] < (double) PITCH_MIN_LAG) lags2[peak] = (double) PITCH_MIN_LAG;
- if (lags1[peak] > (double) PITCH_MAX_LAG) lags1[peak] = (double) PITCH_MAX_LAG;
- if (lags2[peak] > (double) PITCH_MAX_LAG) lags2[peak] = (double) PITCH_MAX_LAG;
-
- /* store lags of highest peak in output array */
- lags[0] = lags1[peak];
- lags[1] = lags1[peak];
- lags[2] = lags2[peak];
- lags[3] = lags2[peak];
- }
- else
- {
- row = max_ind / (PITCH_LAG_SPAN2+4);
- lags1[0] = (double) ((max_ind - row * (PITCH_LAG_SPAN2+4)) + PITCH_MIN_LAG/2 - 4);
- lags2[0] = (double) (lags1[0] + PITCH_BW - row);
-
- if (lags1[0] < (double) PITCH_MIN_LAG) lags1[0] = (double) PITCH_MIN_LAG;
- if (lags2[0] < (double) PITCH_MIN_LAG) lags2[0] = (double) PITCH_MIN_LAG;
- if (lags1[0] > (double) PITCH_MAX_LAG) lags1[0] = (double) PITCH_MAX_LAG;
- if (lags2[0] > (double) PITCH_MAX_LAG) lags2[0] = (double) PITCH_MAX_LAG;
-
- /* store lags of highest peak in output array */
- lags[0] = lags1[0];
- lags[1] = lags1[0];
- lags[2] = lags2[0];
- lags[3] = lags2[0];
- }
-}
-
-
-
-/* create weighting matrix by orthogonalizing a basis of polynomials of increasing order
- * t = (0:4)';
- * A = [t.^0, t.^1, t.^2, t.^3, t.^4];
- * [Q, dummy] = qr(A);
- * P.Weight = Q * diag([0, .1, .5, 1, 1]) * Q'; */
-static const double kWeight[5][5] = {
- { 0.29714285714286, -0.30857142857143, -0.05714285714286, 0.05142857142857, 0.01714285714286},
- {-0.30857142857143, 0.67428571428571, -0.27142857142857, -0.14571428571429, 0.05142857142857},
- {-0.05714285714286, -0.27142857142857, 0.65714285714286, -0.27142857142857, -0.05714285714286},
- { 0.05142857142857, -0.14571428571429, -0.27142857142857, 0.67428571428571, -0.30857142857143},
- { 0.01714285714286, 0.05142857142857, -0.05714285714286, -0.30857142857143, 0.29714285714286}
-};
-
-
-void WebRtcIsac_PitchAnalysis(const double *in, /* PITCH_FRAME_LEN samples */
- double *out, /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
- PitchAnalysisStruct *State,
- double *lags,
- double *gains)
-{
- double HPin[PITCH_FRAME_LEN];
- double Weighted[PITCH_FRAME_LEN];
- double Whitened[PITCH_FRAME_LEN + QLOOKAHEAD];
- double inbuf[PITCH_FRAME_LEN + QLOOKAHEAD];
- double out_G[PITCH_FRAME_LEN + QLOOKAHEAD]; // could be removed by using out instead
- double out_dG[4][PITCH_FRAME_LEN + QLOOKAHEAD];
- double old_lag, old_gain;
- double nrg_wht, tmp;
- double Wnrg, Wfluct, Wgain;
- double H[4][4];
- double grad[4];
- double dG[4];
- int k, m, n, iter;
-
- /* high pass filtering using second order pole-zero filter */
- WebRtcIsac_Highpass(in, HPin, State->hp_state, PITCH_FRAME_LEN);
-
- /* copy from state into buffer */
- memcpy(Whitened, State->whitened_buf, sizeof(double) * QLOOKAHEAD);
-
- /* compute weighted and whitened signals */
- WebRtcIsac_WeightingFilter(HPin, &Weighted[0], &Whitened[QLOOKAHEAD], &(State->Wghtstr));
-
- /* copy from buffer into state */
- memcpy(State->whitened_buf, Whitened+PITCH_FRAME_LEN, sizeof(double) * QLOOKAHEAD);
-
- old_lag = State->PFstr_wght.oldlagp[0];
- old_gain = State->PFstr_wght.oldgainp[0];
-
- /* inital pitch estimate */
- WebRtcIsac_InitializePitch(Weighted, old_lag, old_gain, State, lags);
-
-
- /* Iterative optimization of lags - to be done */
-
- /* compute energy of whitened signal */
- nrg_wht = 0.0;
- for (k = 0; k < PITCH_FRAME_LEN + QLOOKAHEAD; k++)
- nrg_wht += Whitened[k] * Whitened[k];
-
-
- /* Iterative optimization of gains */
-
- /* set weights for energy, gain fluctiation, and spectral gain penalty functions */
- Wnrg = 1.0 / nrg_wht;
- Wgain = 0.005;
- Wfluct = 3.0;
-
- /* set initial gains */
- for (k = 0; k < 4; k++)
- gains[k] = PITCH_MAX_GAIN_06;
-
- /* two iterations should be enough */
- for (iter = 0; iter < 2; iter++) {
- /* compute Jacobian of pre-filter output towards gains */
- WebRtcIsac_PitchfilterPre_gains(Whitened, out_G, out_dG, &(State->PFstr_wght), lags, gains);
-
- /* gradient and approximate Hessian (lower triangle) for minimizing the filter's output power */
- for (k = 0; k < 4; k++) {
- tmp = 0.0;
- for (n = 0; n < PITCH_FRAME_LEN + QLOOKAHEAD; n++)
- tmp += out_G[n] * out_dG[k][n];
- grad[k] = tmp * Wnrg;
- }
- for (k = 0; k < 4; k++) {
- for (m = 0; m <= k; m++) {
- tmp = 0.0;
- for (n = 0; n < PITCH_FRAME_LEN + QLOOKAHEAD; n++)
- tmp += out_dG[m][n] * out_dG[k][n];
- H[k][m] = tmp * Wnrg;
- }
- }
-
- /* add gradient and Hessian (lower triangle) for dampening fast gain changes */
- for (k = 0; k < 4; k++) {
- tmp = kWeight[k+1][0] * old_gain;
- for (m = 0; m < 4; m++)
- tmp += kWeight[k+1][m+1] * gains[m];
- grad[k] += tmp * Wfluct;
- }
- for (k = 0; k < 4; k++) {
- for (m = 0; m <= k; m++) {
- H[k][m] += kWeight[k+1][m+1] * Wfluct;
- }
- }
-
- /* add gradient and Hessian for dampening gain */
- for (k = 0; k < 3; k++) {
- tmp = 1.0 / (1 - gains[k]);
- grad[k] += tmp * tmp * Wgain;
- H[k][k] += 2.0 * tmp * (tmp * tmp * Wgain);
- }
- tmp = 1.0 / (1 - gains[3]);
- grad[3] += 1.33 * (tmp * tmp * Wgain);
- H[3][3] += 2.66 * tmp * (tmp * tmp * Wgain);
-
-
- /* compute Cholesky factorization of Hessian
- * by overwritting the upper triangle; scale factors on diagonal
- * (for non pc-platforms store the inverse of the diagonals seperately to minimize divisions) */
- H[0][1] = H[1][0] / H[0][0];
- H[0][2] = H[2][0] / H[0][0];
- H[0][3] = H[3][0] / H[0][0];
- H[1][1] -= H[0][0] * H[0][1] * H[0][1];
- H[1][2] = (H[2][1] - H[0][1] * H[2][0]) / H[1][1];
- H[1][3] = (H[3][1] - H[0][1] * H[3][0]) / H[1][1];
- H[2][2] -= H[0][0] * H[0][2] * H[0][2] + H[1][1] * H[1][2] * H[1][2];
- H[2][3] = (H[3][2] - H[0][2] * H[3][0] - H[1][2] * H[1][1] * H[1][3]) / H[2][2];
- H[3][3] -= H[0][0] * H[0][3] * H[0][3] + H[1][1] * H[1][3] * H[1][3] + H[2][2] * H[2][3] * H[2][3];
-
- /* Compute update as delta_gains = -inv(H) * grad */
- /* copy and negate */
- for (k = 0; k < 4; k++)
- dG[k] = -grad[k];
- /* back substitution */
- dG[1] -= dG[0] * H[0][1];
- dG[2] -= dG[0] * H[0][2] + dG[1] * H[1][2];
- dG[3] -= dG[0] * H[0][3] + dG[1] * H[1][3] + dG[2] * H[2][3];
- /* scale */
- for (k = 0; k < 4; k++)
- dG[k] /= H[k][k];
- /* back substitution */
- dG[2] -= dG[3] * H[2][3];
- dG[1] -= dG[3] * H[1][3] + dG[2] * H[1][2];
- dG[0] -= dG[3] * H[0][3] + dG[2] * H[0][2] + dG[1] * H[0][1];
-
- /* update gains and check range */
- for (k = 0; k < 4; k++) {
- gains[k] += dG[k];
- if (gains[k] > PITCH_MAX_GAIN)
- gains[k] = PITCH_MAX_GAIN;
- else if (gains[k] < 0.0)
- gains[k] = 0.0;
- }
- }
-
- /* update state for next frame */
- WebRtcIsac_PitchfilterPre(Whitened, out, &(State->PFstr_wght), lags, gains);
-
- /* concatenate previous input's end and current input */
- memcpy(inbuf, State->inbuf, sizeof(double) * QLOOKAHEAD);
- memcpy(inbuf+QLOOKAHEAD, in, sizeof(double) * PITCH_FRAME_LEN);
-
- /* lookahead pitch filtering for masking analysis */
- WebRtcIsac_PitchfilterPre_la(inbuf, out, &(State->PFstr), lags, gains);
-
- /* store last part of input */
- for (k = 0; k < QLOOKAHEAD; k++)
- State->inbuf[k] = inbuf[k + PITCH_FRAME_LEN];
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_estimator.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_estimator.h
deleted file mode 100644
index f5d9356..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_estimator.h
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * pitch_estimator.h
- *
- * Pitch functions
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_
-
-#include "structs.h"
-
-
-
-void WebRtcIsac_PitchAnalysis(const double *in, /* PITCH_FRAME_LEN samples */
- double *out, /* PITCH_FRAME_LEN+QLOOKAHEAD samples */
- PitchAnalysisStruct *State,
- double *lags,
- double *gains);
-
-void WebRtcIsac_InitializePitch(const double *in,
- const double old_lag,
- const double old_gain,
- PitchAnalysisStruct *State,
- double *lags);
-
-void WebRtcIsac_PitchfilterPre(double *indat,
- double *outdat,
- PitchFiltstr *pfp,
- double *lags,
- double *gains);
-
-void WebRtcIsac_PitchfilterPost(double *indat,
- double *outdat,
- PitchFiltstr *pfp,
- double *lags,
- double *gains);
-
-void WebRtcIsac_PitchfilterPre_la(double *indat,
- double *outdat,
- PitchFiltstr *pfp,
- double *lags,
- double *gains);
-
-void WebRtcIsac_PitchfilterPre_gains(double *indat,
- double *outdat,
- double out_dG[][PITCH_FRAME_LEN + QLOOKAHEAD],
- PitchFiltstr *pfp,
- double *lags,
- double *gains);
-
-void WebRtcIsac_WeightingFilter(const double *in, double *weiout, double *whiout, WeightFiltstr *wfdata);
-
-void WebRtcIsac_Highpass(const double *in, double *out, double *state, int N);
-
-void WebRtcIsac_DecimateAllpass(const double *in,
- double *state_in, /* array of size: 2*ALLPASSSECTIONS+1 */
- int N, /* number of input samples */
- double *out); /* array of size N/2 */
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_ESTIMATOR_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_filter.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_filter.c
deleted file mode 100644
index ccc8d21..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_filter.c
+++ /dev/null
@@ -1,469 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "pitch_estimator.h"
-#include "os_specific_inline.h"
-
-#include <stdlib.h>
-#include <memory.h>
-#include <math.h>
-
-static const double kDampFilter[PITCH_DAMPORDER] = {-0.07, 0.25, 0.64, 0.25, -0.07};
-
-/* interpolation coefficients; generated by design_pitch_filter.m */
-static const double kIntrpCoef[PITCH_FRACS][PITCH_FRACORDER] = {
- {-0.02239172458614, 0.06653315052934, -0.16515880017569, 0.60701333734125, 0.64671399919202, -0.20249000396417, 0.09926548334755, -0.04765933793109, 0.01754159521746},
- {-0.01985640750434, 0.05816126837866, -0.13991265473714, 0.44560418147643, 0.79117042386876, -0.20266133815188, 0.09585268418555, -0.04533310458084, 0.01654127246314},
- {-0.01463300534216, 0.04229888475060, -0.09897034715253, 0.28284326017787, 0.90385267956632, -0.16976950138649, 0.07704272393639, -0.03584218578311, 0.01295781500709},
- {-0.00764851320885, 0.02184035544377, -0.04985561057281, 0.13083306574393, 0.97545011664662, -0.10177807997561, 0.04400901776474, -0.02010737175166, 0.00719783432422},
- {-0.00000000000000, 0.00000000000000, -0.00000000000001, 0.00000000000001, 0.99999999999999, 0.00000000000001, -0.00000000000001, 0.00000000000000, -0.00000000000000},
- { 0.00719783432422, -0.02010737175166, 0.04400901776474, -0.10177807997562, 0.97545011664663, 0.13083306574393, -0.04985561057280, 0.02184035544377, -0.00764851320885},
- { 0.01295781500710, -0.03584218578312, 0.07704272393640, -0.16976950138650, 0.90385267956634, 0.28284326017785, -0.09897034715252, 0.04229888475059, -0.01463300534216},
- { 0.01654127246315, -0.04533310458085, 0.09585268418557, -0.20266133815190, 0.79117042386878, 0.44560418147640, -0.13991265473712, 0.05816126837865, -0.01985640750433}
-};
-
-
-void WebRtcIsac_PitchfilterPre(double *indat,
- double *outdat,
- PitchFiltstr *pfp,
- double *lags,
- double *gains)
-{
-
- double ubuf[PITCH_INTBUFFSIZE];
- const double *fracoeff = NULL;
- double curgain, curlag, gaindelta, lagdelta;
- double sum, inystate[PITCH_DAMPORDER];
- double ftmp, oldlag, oldgain;
- int k, n, m, pos, ind, pos2, Li, frc;
-
- Li = 0;
- /* Set up buffer and states */
- memcpy(ubuf, pfp->ubuf, sizeof(double) * PITCH_BUFFSIZE);
- memcpy(inystate, pfp->ystate, sizeof(double) * PITCH_DAMPORDER);
-
- oldlag = *pfp->oldlagp;
- oldgain = *pfp->oldgainp;
-
- /* No interpolation if pitch lag step is big */
- if ((lags[0] > (PITCH_UPSTEP * oldlag)) || (lags[0] < (PITCH_DOWNSTEP * oldlag))) {
- oldlag = lags[0];
- oldgain = gains[0];
- }
-
- ind=0;
- for (k=0;k<PITCH_SUBFRAMES;k++) {
-
- /* Calculate interpolation steps */
- lagdelta=(lags[k]-oldlag) / PITCH_GRAN_PER_SUBFRAME;
- curlag=oldlag ;
- gaindelta=(gains[k]-oldgain) / PITCH_GRAN_PER_SUBFRAME;
- curgain=oldgain ;
- oldlag=lags[k];
- oldgain=gains[k];
-
- for (n=0;n<PITCH_SUBFRAME_LEN;n++) {
- if ((ind % PITCH_UPDATE) == 0) { /* Update parameters */
- curgain += gaindelta;
- curlag += lagdelta;
- Li = WebRtcIsac_lrint(curlag+PITCH_FILTDELAY + 0.5);
- ftmp = Li - (curlag+PITCH_FILTDELAY);
- frc = WebRtcIsac_lrint(PITCH_FRACS * ftmp - 0.5);
- fracoeff = kIntrpCoef[frc];
- }
-
- /* shift low pass filter state */
- for (m=PITCH_DAMPORDER-1;m>0;m--)
- inystate[m] = inystate[m-1];
-
- /* Filter to get fractional pitch */
- pos = ind + PITCH_BUFFSIZE;
- pos2 = pos - Li;
- sum=0;
- for (m=0;m<PITCH_FRACORDER;m++)
- sum += ubuf[pos2+m] * fracoeff[m];
- inystate[0] = curgain * sum; /* Multiply with gain */
-
- /* Low pass filter */
- sum=0;
- for (m=0;m<PITCH_DAMPORDER;m++)
- sum += inystate[m] * kDampFilter[m];
-
- /* Subtract from input and update buffer */
- outdat[ind] = indat[ind] - sum;
- ubuf[pos] = indat[ind] + outdat[ind];
- ind++;
- }
- }
-
- /* Export buffer and states */
- memcpy(pfp->ubuf, ubuf+PITCH_FRAME_LEN, sizeof(double) * PITCH_BUFFSIZE);
- memcpy(pfp->ystate, inystate, sizeof(double) * PITCH_DAMPORDER);
-
- *pfp->oldlagp = oldlag;
- *pfp->oldgainp = oldgain;
-
-}
-
-
-void WebRtcIsac_PitchfilterPre_la(double *indat,
- double *outdat,
- PitchFiltstr *pfp,
- double *lags,
- double *gains)
-{
- double ubuf[PITCH_INTBUFFSIZE+QLOOKAHEAD];
- const double *fracoeff = NULL;
- double curgain, curlag, gaindelta, lagdelta;
- double sum, inystate[PITCH_DAMPORDER];
- double ftmp;
- double oldlag, oldgain;
- int k, n, m, pos, ind, pos2, Li, frc;
-
- Li = 0;
- /* Set up buffer and states */
- memcpy(ubuf, pfp->ubuf, sizeof(double) * PITCH_BUFFSIZE);
- memcpy(inystate, pfp->ystate, sizeof(double) * PITCH_DAMPORDER);
-
- oldlag = *pfp->oldlagp;
- oldgain = *pfp->oldgainp;
-
- /* No interpolation if pitch lag step is big */
- if ((lags[0] > (PITCH_UPSTEP * oldlag)) || (lags[0] < (PITCH_DOWNSTEP * oldlag))) {
- oldlag = lags[0];
- oldgain = gains[0];
- }
-
-
- ind=0;
- for (k=0;k<PITCH_SUBFRAMES;k++) {
-
- /* Calculate interpolation steps */
- lagdelta=(lags[k]-oldlag) / PITCH_GRAN_PER_SUBFRAME;
- curlag=oldlag ;
- gaindelta=(gains[k]-oldgain) / PITCH_GRAN_PER_SUBFRAME;
- curgain=oldgain ;
- oldlag=lags[k];
- oldgain=gains[k];
-
- for (n=0;n<PITCH_SUBFRAME_LEN;n++) {
- if ((ind % PITCH_UPDATE) == 0) { /* Update parameters */
- curgain += gaindelta;
- curlag += lagdelta;
- Li = WebRtcIsac_lrint(curlag+PITCH_FILTDELAY + 0.5);
- ftmp = Li - (curlag+PITCH_FILTDELAY);
- frc = WebRtcIsac_lrint(PITCH_FRACS * ftmp - 0.5);
- fracoeff = kIntrpCoef[frc];
- }
-
- /* shift low pass filter state */
- for (m=PITCH_DAMPORDER-1;m>0;m--)
- inystate[m] = inystate[m-1];
-
- /* Filter to get fractional pitch */
- pos = ind + PITCH_BUFFSIZE;
- pos2 = pos - Li;
- sum=0.0;
- for (m=0;m<PITCH_FRACORDER;m++)
- sum += ubuf[pos2+m] * fracoeff[m];
- inystate[0] = curgain * sum; /* Multiply with gain */
-
- /* Low pass filter */
- sum=0.0;
- for (m=0;m<PITCH_DAMPORDER;m++)
- sum += inystate[m] * kDampFilter[m];
-
- /* Subtract from input and update buffer */
- outdat[ind] = indat[ind] - sum;
- ubuf[pos] = indat[ind] + outdat[ind];
- ind++;
- }
- }
-
- /* Export buffer and states */
- memcpy(pfp->ubuf, ubuf+PITCH_FRAME_LEN, sizeof(double) * PITCH_BUFFSIZE);
- memcpy(pfp->ystate, inystate, sizeof(double) * PITCH_DAMPORDER);
-
- *pfp->oldlagp = oldlag;
- *pfp->oldgainp = oldgain;
-
-
- /* Filter look-ahead segment */
- for (n=0;n<QLOOKAHEAD;n++) {
- /* shift low pass filter state */
- for (m=PITCH_DAMPORDER-1;m>0;m--)
- inystate[m] = inystate[m-1];
-
- /* Filter to get fractional pitch */
- pos = ind + PITCH_BUFFSIZE;
- pos2 = pos - Li;
- sum=0.0;
- for (m=0;m<PITCH_FRACORDER;m++)
- sum += ubuf[pos2+m] * fracoeff[m];
- inystate[0] = curgain * sum; /* Multiply with gain */
-
- /* Low pass filter */
- sum=0.0;
- for (m=0;m<PITCH_DAMPORDER;m++)
- sum += inystate[m] * kDampFilter[m];
-
- /* Subtract from input and update buffer */
- outdat[ind] = indat[ind] - sum;
- ubuf[pos] = indat[ind] + outdat[ind];
- ind++;
- }
-}
-
-
-void WebRtcIsac_PitchfilterPre_gains(double *indat,
- double *outdat,
- double out_dG[][PITCH_FRAME_LEN + QLOOKAHEAD],
- PitchFiltstr *pfp,
- double *lags,
- double *gains)
-{
- double ubuf[PITCH_INTBUFFSIZE+QLOOKAHEAD];
- double inystate_dG[4][PITCH_DAMPORDER];
- double gain_mult[4];
- const double *fracoeff = NULL;
- double curgain, curlag, gaindelta, lagdelta;
- double sum, sum2, inystate[PITCH_DAMPORDER];
- double ftmp, oldlag, oldgain;
- int k, n, m, m_tmp, j, pos, ind, pos2, Li, frc;
-
- Li = 0;
-
- /* Set up buffer and states */
- memcpy(ubuf, pfp->ubuf, sizeof(double) * PITCH_BUFFSIZE);
- memcpy(inystate, pfp->ystate, sizeof(double) * PITCH_DAMPORDER);
-
- /* clear some buffers */
- for (k = 0; k < 4; k++) {
- gain_mult[k] = 0.0;
- for (n = 0; n < PITCH_DAMPORDER; n++)
- inystate_dG[k][n] = 0.0;
- }
-
- oldlag = *pfp->oldlagp;
- oldgain = *pfp->oldgainp;
-
- /* No interpolation if pitch lag step is big */
- if ((lags[0] > (PITCH_UPSTEP * oldlag)) || (lags[0] < (PITCH_DOWNSTEP * oldlag))) {
- oldlag = lags[0];
- oldgain = gains[0];
- gain_mult[0] = 1.0;
- }
-
-
- ind=0;
- for (k=0;k<PITCH_SUBFRAMES;k++) {
-
- /* Calculate interpolation steps */
- lagdelta=(lags[k]-oldlag) / PITCH_GRAN_PER_SUBFRAME;
- curlag=oldlag ;
- gaindelta=(gains[k]-oldgain) / PITCH_GRAN_PER_SUBFRAME;
- curgain=oldgain ;
- oldlag=lags[k];
- oldgain=gains[k];
-
- for (n=0;n<PITCH_SUBFRAME_LEN;n++) {
- if ((ind % PITCH_UPDATE) == 0) { /* Update parameters */
- curgain += gaindelta;
- curlag += lagdelta;
- Li = WebRtcIsac_lrint(curlag+PITCH_FILTDELAY + 0.5);
- ftmp = Li - (curlag+PITCH_FILTDELAY);
- frc = WebRtcIsac_lrint(PITCH_FRACS * ftmp - 0.5);
- fracoeff = kIntrpCoef[frc];
- gain_mult[k] += 0.2;
- if (gain_mult[k] > 1.0) gain_mult[k] = 1.0;
- if (k > 0) gain_mult[k-1] -= 0.2;
- }
-
- /* shift low pass filter states */
- for (m=PITCH_DAMPORDER-1;m>0;m--) {
- inystate[m] = inystate[m-1];
- for (j = 0; j < 4; j++)
- inystate_dG[j][m] = inystate_dG[j][m-1];
- }
-
- pos = ind + PITCH_BUFFSIZE;
- pos2 = pos - Li;
-
- /* Filter to get fractional pitch */
- sum=0.0;
- for (m=0;m<PITCH_FRACORDER;m++)
- sum += ubuf[pos2+m] * fracoeff[m];
- inystate[0] = curgain * sum; /* Multiply with gain */
- m_tmp = (Li-ind > 0) ? Li-ind : 0;
- for (j = 0; j < k+1; j++) {
- /* filter */
- sum2 = 0.0;
- for (m = PITCH_FRACORDER-1; m >= m_tmp; m--)
- sum2 += out_dG[j][ind-Li + m] * fracoeff[m];
- inystate_dG[j][0] = gain_mult[j] * sum + curgain * sum2;
- }
-
- /* Low pass filter */
- sum=0.0;
- for (m=0;m<PITCH_DAMPORDER;m++)
- sum += inystate[m] * kDampFilter[m];
-
- /* Subtract from input and update buffer */
- outdat[ind] = indat[ind] - sum;
- ubuf[pos] = indat[ind] + outdat[ind];
-
- for (j = 0; j < k+1; j++) {
- sum = 0.0;
- for (m=0;m<PITCH_DAMPORDER;m++)
- sum -= inystate_dG[j][m] * kDampFilter[m];
- out_dG[j][ind] = sum;
- }
- for (j = k+1; j < 4; j++)
- out_dG[j][ind] = 0.0;
-
-
- ind++;
- }
- }
-
- /* Filter look-ahead segment */
- for (n=0;n<QLOOKAHEAD;n++) {
- /* shift low pass filter states */
- for (m=PITCH_DAMPORDER-1;m>0;m--) {
- inystate[m] = inystate[m-1];
- for (j = 0; j < 4; j++)
- inystate_dG[j][m] = inystate_dG[j][m-1];
- }
-
- pos = ind + PITCH_BUFFSIZE;
- pos2 = pos - Li;
-
- /* Filter to get fractional pitch */
- sum=0.0;
- for (m=0;m<PITCH_FRACORDER;m++)
- sum += ubuf[pos2+m] * fracoeff[m];
- inystate[0] = curgain * sum; /* Multiply with gain */
- m_tmp = (Li-ind > 0) ? Li-ind : 0;
- for (j = 0; (j<k+1)&&(j<4); j++) {
- /* filter */
- sum2 = 0.0;
- for (m = PITCH_FRACORDER-1; m >= m_tmp; m--)
- sum2 += out_dG[j][ind-Li + m] * fracoeff[m];
- inystate_dG[j][0] = gain_mult[j] * sum + curgain * sum2;
- }
-
- /* Low pass filter */
- sum=0.0;
- for (m=0;m<PITCH_DAMPORDER;m++)
- sum += inystate[m] * kDampFilter[m];
-
- /* Subtract from input and update buffer */
- outdat[ind] = indat[ind] - sum;
- ubuf[pos] = indat[ind] + outdat[ind];
-
- for (j = 0; (j<k+1)&&(j<4); j++) {
- sum = 0.0;
- for (m=0;m<PITCH_DAMPORDER;m++)
- sum -= inystate_dG[j][m] * kDampFilter[m];
- out_dG[j][ind] = sum;
- }
-
- ind++;
- }
-}
-
-
-void WebRtcIsac_PitchfilterPost(double *indat,
- double *outdat,
- PitchFiltstr *pfp,
- double *lags,
- double *gains)
-{
-
- double ubuf[PITCH_INTBUFFSIZE];
- const double *fracoeff = NULL;
- double curgain, curlag, gaindelta, lagdelta;
- double sum, inystate[PITCH_DAMPORDER];
- double ftmp, oldlag, oldgain;
- int k, n, m, pos, ind, pos2, Li, frc;
-
- Li = 0;
-
- /* Set up buffer and states */
- memcpy(ubuf, pfp->ubuf, sizeof(double) * PITCH_BUFFSIZE);
- memcpy(inystate, pfp->ystate, sizeof(double) * PITCH_DAMPORDER);
-
- oldlag = *pfp->oldlagp;
- oldgain = *pfp->oldgainp;
-
- /* make output more periodic */
- for (k=0;k<PITCH_SUBFRAMES;k++)
- gains[k] *= 1.3;
-
- /* No interpolation if pitch lag step is big */
- if ((lags[0] > (PITCH_UPSTEP * oldlag)) || (lags[0] < (PITCH_DOWNSTEP * oldlag))) {
- oldlag = lags[0];
- oldgain = gains[0];
- }
-
-
- ind=0;
- for (k=0;k<PITCH_SUBFRAMES;k++) {
-
- /* Calculate interpolation steps */
- lagdelta=(lags[k]-oldlag) / PITCH_GRAN_PER_SUBFRAME;
- curlag=oldlag ;
- gaindelta=(gains[k]-oldgain) / PITCH_GRAN_PER_SUBFRAME;
- curgain=oldgain ;
- oldlag=lags[k];
- oldgain=gains[k];
-
- for (n=0;n<PITCH_SUBFRAME_LEN;n++) {
- if ((ind % PITCH_UPDATE) == 0) { /* Update parameters */
- curgain += gaindelta;
- curlag += lagdelta;
- Li = WebRtcIsac_lrint(curlag+PITCH_FILTDELAY + 0.5);
- ftmp = Li - (curlag+PITCH_FILTDELAY);
- frc = WebRtcIsac_lrint(PITCH_FRACS * ftmp - 0.5);
- fracoeff = kIntrpCoef[frc];
- }
-
- /* shift low pass filter state */
- for (m=PITCH_DAMPORDER-1;m>0;m--)
- inystate[m] = inystate[m-1];
-
- /* Filter to get fractional pitch */
- pos = ind + PITCH_BUFFSIZE;
- pos2 = pos - Li;
- sum=0.0;
- for (m=0;m<PITCH_FRACORDER;m++)
- sum += ubuf[pos2+m] * fracoeff[m];
- inystate[0] = curgain * sum; /* Multiply with gain */
-
- /* Low pass filter */
- sum=0.0;
- for (m=0;m<PITCH_DAMPORDER;m++)
- sum += inystate[m] * kDampFilter[m];
-
- /* Add to input and update buffer */
- outdat[ind] = indat[ind] + sum;
- ubuf[pos] = indat[ind] + outdat[ind];
- ind++;
- }
- }
-
- /* Export buffer and states */
- memcpy(pfp->ubuf, ubuf+PITCH_FRAME_LEN, sizeof(double) * PITCH_BUFFSIZE);
- memcpy(pfp->ystate, inystate, sizeof(double) * PITCH_DAMPORDER);
-
- *pfp->oldlagp = oldlag;
- *pfp->oldgainp = oldgain;
-
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_gain_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_gain_tables.c
deleted file mode 100644
index 5d998a2..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_gain_tables.c
+++ /dev/null
@@ -1,105 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "pitch_gain_tables.h"
-
-#include "settings.h"
-
-/* header file for coding tables for the pitch filter side-info in the entropy coder */
-/********************* Pitch Filter Gain Coefficient Tables ************************/
-/* cdf for quantized pitch filter gains */
-const WebRtc_UWord16 WebRtcIsac_kQPitchGainCdf[255] = {
- 0, 2, 4, 6, 64, 901, 903, 905, 16954, 16956,
- 16961, 17360, 17362, 17364, 17366, 17368, 17370, 17372, 17374, 17411,
- 17514, 17516, 17583, 18790, 18796, 18802, 20760, 20777, 20782, 21722,
- 21724, 21728, 21738, 21740, 21742, 21744, 21746, 21748, 22224, 22227,
- 22230, 23214, 23229, 23239, 25086, 25108, 25120, 26088, 26094, 26098,
- 26175, 26177, 26179, 26181, 26183, 26185, 26484, 26507, 26522, 27705,
- 27731, 27750, 29767, 29799, 29817, 30866, 30883, 30885, 31025, 31029,
- 31031, 31033, 31035, 31037, 31114, 31126, 31134, 32687, 32722, 32767,
- 35718, 35742, 35757, 36943, 36952, 36954, 37115, 37128, 37130, 37132,
- 37134, 37136, 37143, 37145, 37152, 38843, 38863, 38897, 47458, 47467,
- 47474, 49040, 49061, 49063, 49145, 49157, 49159, 49161, 49163, 49165,
- 49167, 49169, 49171, 49757, 49770, 49782, 61333, 61344, 61346, 62860,
- 62883, 62885, 62887, 62889, 62891, 62893, 62895, 62897, 62899, 62901,
- 62903, 62905, 62907, 62909, 65496, 65498, 65500, 65521, 65523, 65525,
- 65527, 65529, 65531, 65533, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535};
-
-/* index limits and ranges */
-const WebRtc_Word16 WebRtcIsac_kIndexLowerLimitGain[3] = {
- -7, -2, -1};
-
-const WebRtc_Word16 WebRtcIsac_kIndexUpperLimitGain[3] = {
- 0, 3, 1};
-
-const WebRtc_UWord16 WebRtcIsac_kIndexMultsGain[2] = {
- 18, 3};
-
-/* size of cdf table */
-const WebRtc_UWord16 WebRtcIsac_kQCdfTableSizeGain[1] = {
- 256};
-
-///////////////////////////FIXED POINT
-/* mean values of pitch filter gains in FIXED point */
-const WebRtc_Word16 WebRtcIsac_kQMeanGain1Q12[144] = {
- 843, 1092, 1336, 1222, 1405, 1656, 1500, 1815, 1843, 1838, 1839, 1843, 1843, 1843, 1843, 1843,
- 1843, 1843, 814, 846, 1092, 1013, 1174, 1383, 1391, 1511, 1584, 1734, 1753, 1843, 1843, 1843,
- 1843, 1843, 1843, 1843, 524, 689, 777, 845, 947, 1069, 1090, 1263, 1380, 1447, 1559, 1676,
- 1645, 1749, 1843, 1843, 1843, 1843, 81, 477, 563, 611, 706, 806, 849, 1012, 1192, 1128,
- 1330, 1489, 1425, 1576, 1826, 1741, 1843, 1843, 0, 290, 305, 356, 488, 575, 602, 741,
- 890, 835, 1079, 1196, 1182, 1376, 1519, 1506, 1680, 1843, 0, 47, 97, 69, 289, 381,
- 385, 474, 617, 664, 803, 1079, 935, 1160, 1269, 1265, 1506, 1741, 0, 0, 0, 0,
- 112, 120, 190, 283, 442, 343, 526, 809, 684, 935, 1134, 1020, 1265, 1506, 0, 0,
- 0, 0, 0, 0, 0, 111, 256, 87, 373, 597, 430, 684, 935, 770, 1020, 1265};
-
-const WebRtc_Word16 WebRtcIsac_kQMeanGain2Q12[144] = {
- 1760, 1525, 1285, 1747, 1671, 1393, 1843, 1826, 1555, 1843, 1784, 1606, 1843, 1843, 1711, 1843,
- 1843, 1814, 1389, 1275, 1040, 1564, 1414, 1252, 1610, 1495, 1343, 1753, 1592, 1405, 1804, 1720,
- 1475, 1843, 1814, 1581, 1208, 1061, 856, 1349, 1148, 994, 1390, 1253, 1111, 1495, 1343, 1178,
- 1770, 1465, 1234, 1814, 1581, 1342, 1040, 793, 713, 1053, 895, 737, 1128, 1003, 861, 1277,
- 1094, 981, 1475, 1192, 1019, 1581, 1342, 1098, 855, 570, 483, 833, 648, 540, 948, 744,
- 572, 1009, 844, 636, 1234, 934, 685, 1342, 1217, 984, 537, 318, 124, 603, 423, 350,
- 687, 479, 322, 791, 581, 430, 987, 671, 488, 1098, 849, 597, 283, 27, 0, 397,
- 222, 38, 513, 271, 124, 624, 325, 157, 737, 484, 233, 849, 597, 343, 27, 0,
- 0, 141, 0, 0, 256, 69, 0, 370, 87, 0, 484, 229, 0, 597, 343, 87};
-
-const WebRtc_Word16 WebRtcIsac_kQMeanGain3Q12[144] = {
- 1843, 1843, 1711, 1843, 1818, 1606, 1843, 1827, 1511, 1814, 1639, 1393, 1760, 1525, 1285, 1656,
- 1419, 1176, 1835, 1718, 1475, 1841, 1650, 1387, 1648, 1498, 1287, 1600, 1411, 1176, 1522, 1299,
- 1040, 1419, 1176, 928, 1773, 1461, 1128, 1532, 1355, 1202, 1429, 1260, 1115, 1398, 1151, 1025,
- 1172, 1080, 790, 1176, 928, 677, 1475, 1147, 1019, 1276, 1096, 922, 1214, 1010, 901, 1057,
- 893, 800, 1040, 796, 734, 928, 677, 424, 1137, 897, 753, 1120, 830, 710, 875, 751,
- 601, 795, 642, 583, 790, 544, 475, 677, 474, 140, 987, 750, 482, 697, 573, 450,
- 691, 487, 303, 661, 394, 332, 537, 303, 220, 424, 168, 0, 737, 484, 229, 624,
- 348, 153, 441, 261, 136, 397, 166, 51, 283, 27, 0, 168, 0, 0, 484, 229,
- 0, 370, 57, 0, 256, 43, 0, 141, 0, 0, 27, 0, 0, 0, 0, 0};
-
-
-const WebRtc_Word16 WebRtcIsac_kQMeanGain4Q12[144] = {
- 1843, 1843, 1843, 1843, 1841, 1843, 1500, 1821, 1843, 1222, 1434, 1656, 843, 1092, 1336, 504,
- 757, 1007, 1843, 1843, 1843, 1838, 1791, 1843, 1265, 1505, 1599, 965, 1219, 1425, 730, 821,
- 1092, 249, 504, 757, 1783, 1819, 1843, 1351, 1567, 1727, 1096, 1268, 1409, 805, 961, 1131,
- 444, 670, 843, 0, 249, 504, 1425, 1655, 1743, 1096, 1324, 1448, 822, 1019, 1199, 490,
- 704, 867, 81, 450, 555, 0, 0, 249, 1247, 1428, 1530, 881, 1073, 1283, 610, 759,
- 939, 278, 464, 645, 0, 200, 270, 0, 0, 0, 935, 1163, 1410, 528, 790, 1068,
- 377, 499, 717, 173, 240, 274, 0, 43, 62, 0, 0, 0, 684, 935, 1182, 343,
- 551, 735, 161, 262, 423, 0, 55, 27, 0, 0, 0, 0, 0, 0, 430, 684,
- 935, 87, 377, 597, 0, 46, 256, 0, 0, 0, 0, 0, 0, 0, 0, 0};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_gain_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_gain_tables.h
deleted file mode 100644
index f958f5d..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_gain_tables.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * pitch_gain_tables.h
- *
- * This file contains tables for the pitch filter side-info in the entropy coder.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_
-
-#include "typedefs.h"
-
-/* header file for coding tables for the pitch filter side-info in the entropy coder */
-/********************* Pitch Filter Gain Coefficient Tables ************************/
-/* cdf for quantized pitch filter gains */
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchGainCdf[255];
-
-/* index limits and ranges */
-extern const WebRtc_Word16 WebRtcIsac_kIndexLowerLimitGain[3];
-
-extern const WebRtc_Word16 WebRtcIsac_kIndexUpperLimitGain[3];
-extern const WebRtc_UWord16 WebRtcIsac_kIndexMultsGain[2];
-
-/* mean values of pitch filter gains */
-//(Y)
-extern const WebRtc_Word16 WebRtcIsac_kQMeanGain1Q12[144];
-extern const WebRtc_Word16 WebRtcIsac_kQMeanGain2Q12[144];
-extern const WebRtc_Word16 WebRtcIsac_kQMeanGain3Q12[144];
-extern const WebRtc_Word16 WebRtcIsac_kQMeanGain4Q12[144];
-//(Y)
-
-/* size of cdf table */
-extern const WebRtc_UWord16 WebRtcIsac_kQCdfTableSizeGain[1];
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_GAIN_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_lag_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_lag_tables.c
deleted file mode 100644
index 72a031e..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_lag_tables.c
+++ /dev/null
@@ -1,277 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "pitch_lag_tables.h"
-#include "settings.h"
-
-/* header file for coding tables for the pitch filter side-info in the entropy coder */
-/********************* Pitch Filter Gain Coefficient Tables ************************/
-
-/* tables for use with small pitch gain */
-
-/* cdf for quantized pitch filter lags */
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Lo[127] = {
- 0, 134, 336, 549, 778, 998, 1264, 1512, 1777, 2070,
- 2423, 2794, 3051, 3361, 3708, 3979, 4315, 4610, 4933, 5269,
- 5575, 5896, 6155, 6480, 6816, 7129, 7477, 7764, 8061, 8358,
- 8718, 9020, 9390, 9783, 10177, 10543, 10885, 11342, 11795, 12213,
- 12680, 13096, 13524, 13919, 14436, 14903, 15349, 15795, 16267, 16734,
- 17266, 17697, 18130, 18632, 19080, 19447, 19884, 20315, 20735, 21288,
- 21764, 22264, 22723, 23193, 23680, 24111, 24557, 25022, 25537, 26082,
- 26543, 27090, 27620, 28139, 28652, 29149, 29634, 30175, 30692, 31273,
- 31866, 32506, 33059, 33650, 34296, 34955, 35629, 36295, 36967, 37726,
- 38559, 39458, 40364, 41293, 42256, 43215, 44231, 45253, 46274, 47359,
- 48482, 49678, 50810, 51853, 53016, 54148, 55235, 56263, 57282, 58363,
- 59288, 60179, 61076, 61806, 62474, 63129, 63656, 64160, 64533, 64856,
- 65152, 65535, 65535, 65535, 65535, 65535, 65535};
-
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Lo[20] = {
- 0, 429, 3558, 5861, 8558, 11639, 15210, 19502, 24773, 31983,
- 42602, 48567, 52601, 55676, 58160, 60172, 61889, 63235, 65383, 65535};
-
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Lo[2] = {
- 0, 65535};
-
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Lo[10] = {
- 0, 2966, 6368, 11182, 19431, 37793, 48532, 55353, 60626, 65535};
-
-const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrLo[4] = {WebRtcIsac_kQPitchLagCdf1Lo, WebRtcIsac_kQPitchLagCdf2Lo, WebRtcIsac_kQPitchLagCdf3Lo, WebRtcIsac_kQPitchLagCdf4Lo};
-
-/* size of first cdf table */
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeLo[1] = {128};
-
-/* index limits and ranges */
-const WebRtc_Word16 WebRtcIsac_kQIndexLowerLimitLagLo[4] = {
--140, -9, 0, -4};
-
-const WebRtc_Word16 WebRtcIsac_kQIndexUpperLimitLagLo[4] = {
--20, 9, 0, 4};
-
-/* initial index for arithmetic decoder */
-const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagLo[3] = {
- 10, 1, 5};
-
-/* mean values of pitch filter lags */
-const double WebRtcIsac_kQMeanLag2Lo[19] = {
--17.21385070, -15.82678944, -14.07123081, -12.03003877, -10.01311864, -8.00794627, -5.91162987, -3.89231876, -1.90220980, -0.01879275,
- 1.89144232, 3.88123171, 5.92146992, 7.96435361, 9.98923648, 11.98266347, 13.96101002, 15.74855713, 17.10976611};
-
-const double WebRtcIsac_kQMeanLag3Lo[1] = {
- 0.00000000};
-
-const double WebRtcIsac_kQMeanLag4Lo[9] = {
--7.76246496, -5.92083980, -3.94095226, -1.89502305, 0.03724681, 1.93054221, 3.96443467, 5.91726366, 7.78434291};
-
-const double WebRtcIsac_kQPitchLagStepsizeLo = 2.000000;
-
-
-/* tables for use with medium pitch gain */
-
-/* cdf for quantized pitch filter lags */
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Mid[255] = {
- 0, 28, 61, 88, 121, 149, 233, 331, 475, 559,
- 624, 661, 689, 712, 745, 791, 815, 843, 866, 922,
- 959, 1024, 1061, 1117, 1178, 1238, 1280, 1350, 1453, 1513,
- 1564, 1625, 1671, 1741, 1788, 1904, 2072, 2421, 2626, 2770,
- 2840, 2900, 2942, 3012, 3068, 3115, 3147, 3194, 3254, 3319,
- 3366, 3520, 3678, 3780, 3850, 3911, 3957, 4032, 4106, 4185,
- 4292, 4474, 4683, 4842, 5019, 5191, 5321, 5428, 5540, 5675,
- 5763, 5847, 5959, 6127, 6304, 6564, 6839, 7090, 7263, 7421,
- 7556, 7728, 7872, 7984, 8142, 8361, 8580, 8743, 8938, 9227,
- 9409, 9539, 9674, 9795, 9930, 10060, 10177, 10382, 10614, 10861,
- 11038, 11271, 11415, 11629, 11792, 12044, 12193, 12416, 12574, 12821,
- 13007, 13235, 13445, 13654, 13901, 14134, 14488, 15000, 15703, 16285,
- 16504, 16797, 17086, 17328, 17579, 17807, 17998, 18268, 18538, 18836,
- 19087, 19274, 19474, 19716, 19935, 20270, 20833, 21303, 21532, 21741,
- 21978, 22207, 22523, 22770, 23054, 23613, 23943, 24204, 24399, 24651,
- 24832, 25074, 25270, 25549, 25759, 26015, 26150, 26424, 26713, 27048,
- 27342, 27504, 27681, 27854, 28021, 28207, 28412, 28664, 28859, 29064,
- 29278, 29548, 29748, 30107, 30377, 30656, 30856, 31164, 31452, 31755,
- 32011, 32328, 32626, 32919, 33319, 33789, 34329, 34925, 35396, 35973,
- 36443, 36964, 37551, 38156, 38724, 39357, 40023, 40908, 41587, 42602,
- 43924, 45037, 45810, 46597, 47421, 48291, 49092, 50051, 51448, 52719,
- 53440, 54241, 54944, 55977, 56676, 57299, 57872, 58389, 59059, 59688,
- 60237, 60782, 61094, 61573, 61890, 62290, 62658, 63030, 63217, 63454,
- 63622, 63882, 64003, 64273, 64427, 64529, 64581, 64697, 64758, 64902,
- 65414, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535};
-
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Mid[36] = {
- 0, 71, 335, 581, 836, 1039, 1323, 1795, 2258, 2608,
- 3005, 3591, 4243, 5344, 7163, 10583, 16848, 28078, 49448, 57007,
- 60357, 61850, 62837, 63437, 63872, 64188, 64377, 64614, 64774, 64949,
- 65039, 65115, 65223, 65360, 65474, 65535};
-
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Mid[2] = {
- 0, 65535};
-
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Mid[20] = {
- 0, 28, 246, 459, 667, 1045, 1523, 2337, 4337, 11347,
- 44231, 56709, 60781, 62243, 63161, 63969, 64608, 65062, 65502, 65535};
-
-const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrMid[4] = {WebRtcIsac_kQPitchLagCdf1Mid, WebRtcIsac_kQPitchLagCdf2Mid, WebRtcIsac_kQPitchLagCdf3Mid, WebRtcIsac_kQPitchLagCdf4Mid};
-
-/* size of first cdf table */
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeMid[1] = {256};
-
-/* index limits and ranges */
-const WebRtc_Word16 WebRtcIsac_kQIndexLowerLimitLagMid[4] = {
--280, -17, 0, -9};
-
-const WebRtc_Word16 WebRtcIsac_kQIndexUpperLimitLagMid[4] = {
--40, 17, 0, 9};
-
-/* initial index for arithmetic decoder */
-const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagMid[3] = {
- 18, 1, 10};
-
-/* mean values of pitch filter lags */
-const double WebRtcIsac_kQMeanLag2Mid[35] = {
--16.89183900, -15.86949778, -15.05476653, -14.00664348, -13.02793036, -12.07324237, -11.00542532, -10.11250602, -8.90792971, -8.02474753,
--7.00426767, -5.94055287, -4.98251338, -3.91053158, -2.98820425, -1.93524245, -0.92978085, -0.01722509, 0.91317387, 1.92973955,
- 2.96908851, 3.93728974, 4.96308471, 5.92244151, 7.08673497, 8.00993708, 9.04656316, 9.98538742, 10.97851694, 11.94772884,
- 13.02426166, 14.00039951, 15.01347042, 15.80758023, 16.94086895};
-
-const double WebRtcIsac_kQMeanLag3Mid[1] = {
- 0.00000000};
-
-const double WebRtcIsac_kQMeanLag4Mid[19] = {
--8.60409403, -7.89198395, -7.03450280, -5.86260421, -4.93822322, -3.93078706, -2.91302322, -1.91824007, -0.87003282, 0.02822649,
- 0.89951758, 1.87495484, 2.91802604, 3.96874074, 5.06571703, 5.93618227, 7.00520185, 7.88497726, 8.64160364};
-
-const double WebRtcIsac_kQPitchLagStepsizeMid = 1.000000;
-
-
-/* tables for use with large pitch gain */
-
-/* cdf for quantized pitch filter lags */
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Hi[511] = {
- 0, 7, 18, 33, 69, 105, 156, 228, 315, 612,
- 680, 691, 709, 724, 735, 738, 742, 746, 749, 753,
- 756, 760, 764, 774, 782, 785, 789, 796, 800, 803,
- 807, 814, 818, 822, 829, 832, 847, 854, 858, 869,
- 876, 883, 898, 908, 934, 977, 1010, 1050, 1060, 1064,
- 1075, 1078, 1086, 1089, 1093, 1104, 1111, 1122, 1133, 1136,
- 1151, 1162, 1183, 1209, 1252, 1281, 1339, 1364, 1386, 1401,
- 1411, 1415, 1426, 1430, 1433, 1440, 1448, 1455, 1462, 1477,
- 1487, 1495, 1502, 1506, 1509, 1516, 1524, 1531, 1535, 1542,
- 1553, 1556, 1578, 1589, 1611, 1625, 1639, 1643, 1654, 1665,
- 1672, 1687, 1694, 1705, 1708, 1719, 1730, 1744, 1752, 1759,
- 1791, 1795, 1820, 1867, 1886, 1915, 1936, 1943, 1965, 1987,
- 2041, 2099, 2161, 2175, 2200, 2211, 2226, 2233, 2244, 2251,
- 2266, 2280, 2287, 2298, 2309, 2316, 2331, 2342, 2356, 2378,
- 2403, 2418, 2447, 2497, 2544, 2602, 2863, 2895, 2903, 2935,
- 2950, 2971, 3004, 3011, 3018, 3029, 3040, 3062, 3087, 3127,
- 3152, 3170, 3199, 3243, 3293, 3322, 3340, 3377, 3402, 3427,
- 3474, 3518, 3543, 3579, 3601, 3637, 3659, 3706, 3731, 3760,
- 3818, 3847, 3869, 3901, 3920, 3952, 4068, 4169, 4220, 4271,
- 4524, 4571, 4604, 4632, 4672, 4730, 4777, 4806, 4857, 4904,
- 4951, 5002, 5031, 5060, 5107, 5150, 5212, 5266, 5331, 5382,
- 5432, 5490, 5544, 5610, 5700, 5762, 5812, 5874, 5972, 6022,
- 6091, 6163, 6232, 6305, 6402, 6540, 6685, 6880, 7090, 7271,
- 7379, 7452, 7542, 7625, 7687, 7770, 7843, 7911, 7966, 8024,
- 8096, 8190, 8252, 8320, 8411, 8501, 8585, 8639, 8751, 8842,
- 8918, 8986, 9066, 9127, 9203, 9269, 9345, 9406, 9464, 9536,
- 9612, 9667, 9735, 9844, 9931, 10036, 10119, 10199, 10260, 10358,
- 10441, 10514, 10666, 10734, 10872, 10951, 11053, 11125, 11223, 11324,
- 11516, 11664, 11737, 11816, 11892, 12008, 12120, 12200, 12280, 12392,
- 12490, 12576, 12685, 12812, 12917, 13003, 13108, 13210, 13300, 13384,
- 13470, 13579, 13673, 13771, 13879, 13999, 14136, 14201, 14368, 14614,
- 14759, 14867, 14958, 15030, 15121, 15189, 15280, 15385, 15461, 15555,
- 15653, 15768, 15884, 15971, 16069, 16145, 16210, 16279, 16380, 16463,
- 16539, 16615, 16688, 16818, 16919, 17017, 18041, 18338, 18523, 18649,
- 18790, 18917, 19047, 19167, 19315, 19460, 19601, 19731, 19858, 20068,
- 20173, 20318, 20466, 20625, 20741, 20911, 21045, 21201, 21396, 21588,
- 21816, 22022, 22305, 22547, 22786, 23072, 23322, 23600, 23879, 24168,
- 24433, 24769, 25120, 25511, 25895, 26289, 26792, 27219, 27683, 28077,
- 28566, 29094, 29546, 29977, 30491, 30991, 31573, 32105, 32594, 33173,
- 33788, 34497, 35181, 35833, 36488, 37255, 37921, 38645, 39275, 39894,
- 40505, 41167, 41790, 42431, 43096, 43723, 44385, 45134, 45858, 46607,
- 47349, 48091, 48768, 49405, 49955, 50555, 51167, 51985, 52611, 53078,
- 53494, 53965, 54435, 54996, 55601, 56125, 56563, 56838, 57244, 57566,
- 57967, 58297, 58771, 59093, 59419, 59647, 59886, 60143, 60461, 60693,
- 60917, 61170, 61416, 61634, 61891, 62122, 62310, 62455, 62632, 62839,
- 63103, 63436, 63639, 63805, 63906, 64015, 64192, 64355, 64475, 64558,
- 64663, 64742, 64811, 64865, 64916, 64956, 64981, 65025, 65068, 65115,
- 65195, 65314, 65419, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535, 65535,
- 65535};
-
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Hi[68] = {
- 0, 7, 11, 22, 37, 52, 56, 59, 81, 85,
- 89, 96, 115, 130, 137, 152, 170, 181, 193, 200,
- 207, 233, 237, 259, 289, 318, 363, 433, 592, 992,
- 1607, 3062, 6149, 12206, 25522, 48368, 58223, 61918, 63640, 64584,
- 64943, 65098, 65206, 65268, 65294, 65335, 65350, 65372, 65387, 65402,
- 65413, 65420, 65428, 65435, 65439, 65450, 65454, 65468, 65472, 65476,
- 65483, 65491, 65498, 65505, 65516, 65520, 65528, 65535};
-
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Hi[2] = {
- 0, 65535};
-
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Hi[35] = {
- 0, 7, 19, 30, 41, 48, 63, 74, 82, 96,
- 122, 152, 215, 330, 701, 2611, 10931, 48106, 61177, 64341,
- 65112, 65238, 65309, 65338, 65364, 65379, 65401, 65427, 65453, 65465,
- 65476, 65490, 65509, 65528, 65535};
-
-const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrHi[4] = {WebRtcIsac_kQPitchLagCdf1Hi, WebRtcIsac_kQPitchLagCdf2Hi, WebRtcIsac_kQPitchLagCdf3Hi, WebRtcIsac_kQPitchLagCdf4Hi};
-
-/* size of first cdf table */
-const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeHi[1] = {512};
-
-/* index limits and ranges */
-const WebRtc_Word16 WebRtcIsac_kQindexLowerLimitLagHi[4] = {
--552, -34, 0, -16};
-
-const WebRtc_Word16 WebRtcIsac_kQindexUpperLimitLagHi[4] = {
--80, 32, 0, 17};
-
-/* initial index for arithmetic decoder */
-const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagHi[3] = {
- 34, 1, 18};
-
-/* mean values of pitch filter lags */
-const double WebRtcIsac_kQMeanLag2Hi[67] = {
--17.07263295, -16.50000000, -15.83966081, -15.55613708, -14.96948007, -14.50000000, -14.00000000, -13.48377986, -13.00000000, -12.50000000,
--11.93199636, -11.44530414, -11.04197641, -10.39910301, -10.15202337, -9.51322461, -8.93357741, -8.46456632, -8.10270672, -7.53751847,
--6.98686404, -6.50000000, -6.08463150, -5.46872991, -5.00864717, -4.50163760, -4.01382410, -3.43856708, -2.96898001, -2.46554810,
--1.96861004, -1.47106701, -0.97197237, -0.46561654, -0.00531409, 0.45767857, 0.96777907, 1.47507903, 1.97740425, 2.46695420,
- 3.00695774, 3.47167185, 4.02712538, 4.49280007, 5.01087640, 5.48191963, 6.04916550, 6.51511058, 6.97297819, 7.46565499,
- 8.01489405, 8.39912001, 8.91819757, 9.50000000, 10.11654065, 10.50000000, 11.03712583, 11.50000000, 12.00000000, 12.38964346,
- 12.89466127, 13.43657881, 13.96013840, 14.46279912, 15.00000000, 15.39412269, 15.96662441};
-
-const double WebRtcIsac_kQMeanLag3Hi[1] = {
- 0.00000000};
-
-const double WebRtcIsac_kQMeanLag4Hi[34] = {
--7.98331221, -7.47988769, -7.03626557, -6.52708003, -6.06982173, -5.51856292, -5.05827033, -4.45909878, -3.99125864, -3.45308135,
--3.02328139, -2.47297273, -1.94341995, -1.44699056, -0.93612243, -0.43012406, 0.01120357, 0.44054812, 0.93199883, 1.45669587,
- 1.97218322, 2.50187419, 2.98748690, 3.49343202, 4.01660147, 4.50984306, 5.01402683, 5.58936797, 5.91787793, 6.59998900,
- 6.85034315, 7.53503316, 7.87711194, 8.53631648};
-
-const double WebRtcIsac_kQPitchLagStepsizeHi = 0.500000;
-
-/* transform matrix */
-const double WebRtcIsac_kTransform[4][4] = {
-{-0.50000000, -0.50000000, -0.50000000, -0.50000000},
-{ 0.67082039, 0.22360680, -0.22360680, -0.67082039},
-{ 0.50000000, -0.50000000, -0.50000000, 0.50000000},
-{ 0.22360680, -0.67082039, 0.67082039, -0.22360680}};
-
-/* transpose transform matrix */
-const double WebRtcIsac_kTransformTranspose[4][4] = {
-{-0.50000000, 0.67082039, 0.50000000, 0.22360680},
-{-0.50000000, 0.22360680, -0.50000000, -0.67082039},
-{-0.50000000, -0.22360680, -0.50000000, 0.67082039},
-{-0.50000000, -0.67082039, 0.50000000, -0.22360680}};
-
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_lag_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_lag_tables.h
deleted file mode 100644
index 67b02e5..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/pitch_lag_tables.h
+++ /dev/null
@@ -1,114 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * pitch_lag_tables.h
- *
- * This file contains tables for the pitch filter side-info in the entropy coder.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_
-
-#include "typedefs.h"
-/* header file for coding tables for the pitch filter side-info in the entropy coder */
-/********************* Pitch Filter Lag Coefficient Tables ************************/
-
-/* tables for use with small pitch gain */
-
-/* cdfs for quantized pitch lags */
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Lo[127];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Lo[20];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Lo[2];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Lo[10];
-
-extern const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrLo[4];
-
-/* size of first cdf table */
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeLo[1];
-
-/* index limits and ranges */
-extern const WebRtc_Word16 WebRtcIsac_kQIndexLowerLimitLagLo[4];
-extern const WebRtc_Word16 WebRtcIsac_kQIndexUpperLimitLagLo[4];
-
-/* initial index for arithmetic decoder */
-extern const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagLo[3];
-
-/* mean values of pitch filter lags */
-extern const double WebRtcIsac_kQMeanLag2Lo[19];
-extern const double WebRtcIsac_kQMeanLag3Lo[1];
-extern const double WebRtcIsac_kQMeanLag4Lo[9];
-
-extern const double WebRtcIsac_kQPitchLagStepsizeLo;
-
-
-/* tables for use with medium pitch gain */
-
-/* cdfs for quantized pitch lags */
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Mid[255];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Mid[36];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Mid[2];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Mid[20];
-
-extern const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrMid[4];
-
-/* size of first cdf table */
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeMid[1];
-
-/* index limits and ranges */
-extern const WebRtc_Word16 WebRtcIsac_kQIndexLowerLimitLagMid[4];
-extern const WebRtc_Word16 WebRtcIsac_kQIndexUpperLimitLagMid[4];
-
-/* initial index for arithmetic decoder */
-extern const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagMid[3];
-
-/* mean values of pitch filter lags */
-extern const double WebRtcIsac_kQMeanLag2Mid[35];
-extern const double WebRtcIsac_kQMeanLag3Mid[1];
-extern const double WebRtcIsac_kQMeanLag4Mid[19];
-
-extern const double WebRtcIsac_kQPitchLagStepsizeMid;
-
-
-/* tables for use with large pitch gain */
-
-/* cdfs for quantized pitch lags */
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf1Hi[511];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf2Hi[68];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf3Hi[2];
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdf4Hi[35];
-
-extern const WebRtc_UWord16 *WebRtcIsac_kQPitchLagCdfPtrHi[4];
-
-/* size of first cdf table */
-extern const WebRtc_UWord16 WebRtcIsac_kQPitchLagCdfSizeHi[1];
-
-/* index limits and ranges */
-extern const WebRtc_Word16 WebRtcIsac_kQindexLowerLimitLagHi[4];
-extern const WebRtc_Word16 WebRtcIsac_kQindexUpperLimitLagHi[4];
-
-/* initial index for arithmetic decoder */
-extern const WebRtc_UWord16 WebRtcIsac_kQInitIndexLagHi[3];
-
-/* mean values of pitch filter lags */
-extern const double WebRtcIsac_kQMeanLag2Hi[67];
-extern const double WebRtcIsac_kQMeanLag3Hi[1];
-extern const double WebRtcIsac_kQMeanLag4Hi[34];
-
-extern const double WebRtcIsac_kQPitchLagStepsizeHi;
-
-/* transform matrix */
-extern const double WebRtcIsac_kTransform[4][4];
-
-/* transpose transform matrix */
-extern const double WebRtcIsac_kTransformTranspose[4][4];
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_PITCH_LAG_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/settings.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/settings.h
deleted file mode 100644
index b7aed77..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/settings.h
+++ /dev/null
@@ -1,199 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * settings.h
- *
- * Declaration of #defines used in the iSAC codec
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_
-
-/* sampling frequency (Hz) */
-#define FS 16000
-
-/* number of samples per frame (either 320 (20ms), 480 (30ms) or 960 (60ms)) */
-#define INITIAL_FRAMESAMPLES 960
-
-
-#define MAXFFTSIZE 2048
-#define NFACTOR 11
-
-
-
-/* do not modify the following; this will have to be modified if we have a 20ms framesize option */
-/*************************************************************************************************/
-/* miliseconds */
-#define FRAMESIZE 30
-/* number of samples per frame processed in the encoder, 480 */
-#define FRAMESAMPLES 480 /* ((FRAMESIZE*FS)/1000) */
-#define FRAMESAMPLES_HALF 240
-#define FRAMESAMPLES_QUARTER 120
-/*************************************************************************************************/
-
-
-
-/* max number of samples per frame (= 60 ms frame) */
-#define MAX_FRAMESAMPLES 960
-#define MAX_SWBFRAMESAMPLES (MAX_FRAMESAMPLES * 2)
-/* number of samples per 10ms frame */
-#define FRAMESAMPLES_10ms ((10*FS)/1000)
-#define SWBFRAMESAMPLES_10ms (FRAMESAMPLES_10ms * 2)
-/* number of samples in 30 ms frame */
-#define FRAMESAMPLES_30ms 480
-/* number of subframes */
-#define SUBFRAMES 6
-/* length of a subframe */
-#define UPDATE 80
-/* length of half a subframe (low/high band) */
-#define HALF_SUBFRAMELEN (UPDATE/2)
-/* samples of look ahead (in a half-band, so actually half the samples of look ahead @ FS) */
-#define QLOOKAHEAD 24 /* 3 ms */
-/* order of AR model in spectral entropy coder */
-#define AR_ORDER 6
-/* order of LP model in spectral entropy coder */
-#define LP_ORDER 0
-
-/* window length (masking analysis) */
-#define WINLEN 256
-/* order of low-band pole filter used to approximate masking curve */
-#define ORDERLO 12
-/* order of hi-band pole filter used to approximate masking curve */
-#define ORDERHI 6
-
-#define UB_LPC_ORDER 4
-#define UB_LPC_VEC_PER_FRAME 2
-#define UB16_LPC_VEC_PER_FRAME 4
-#define UB_ACTIVE_SUBFRAMES 2
-#define UB_MAX_LPC_ORDER 6
-#define UB_INTERPOL_SEGMENTS 1
-#define UB16_INTERPOL_SEGMENTS 3
-#define LB_TOTAL_DELAY_SAMPLES 48
-enum ISACBandwidth {isac8kHz = 8, isac12kHz = 12, isac16kHz = 16};
-enum ISACBand{isacLowerBand = 0, isacUpperBand = 1};
-#define UB_LPC_GAIN_DIM SUBFRAMES
-#define FB_STATE_SIZE_WORD32 6
-
-
-/* order for post_filter_bank */
-#define POSTQORDER 3
-/* order for pre-filterbank */
-#define QORDER 3
-/* another order */
-#define QORDER_ALL (POSTQORDER+QORDER-1)
-/* for decimator */
-#define ALLPASSSECTIONS 2
-
-
-/* array size for byte stream in number of bytes. */
-#define STREAM_SIZE_MAX 600 /* The old maximum size still needed for the decoding */
-#define STREAM_SIZE_MAX_30 200 /* 200 bytes = 53.4 kbit/s @ 30 ms.framelength */
-#define STREAM_SIZE_MAX_60 400 /* 400 bytes = 53.4 kbit/s @ 60 ms.framelength */
-
-/* storage size for bit counts */
-#define BIT_COUNTER_SIZE 30
-/* maximum order of any AR model or filter */
-#define MAX_AR_MODEL_ORDER 12//50
-
-
-/* For pitch analysis */
-#define PITCH_FRAME_LEN (FRAMESAMPLES_HALF) /* 30 ms */
-#define PITCH_MAX_LAG 140 /* 57 Hz */
-#define PITCH_MIN_LAG 20 /* 400 Hz */
-#define PITCH_MAX_GAIN 0.45
-#define PITCH_MAX_GAIN_06 0.27 /* PITCH_MAX_GAIN*0.6 */
-#define PITCH_MAX_GAIN_Q12 1843
-#define PITCH_LAG_SPAN2 (PITCH_MAX_LAG/2-PITCH_MIN_LAG/2+5)
-#define PITCH_CORR_LEN2 60 /* 15 ms */
-#define PITCH_CORR_STEP2 (PITCH_FRAME_LEN/4)
-#define PITCH_BW 11 /* half the band width of correlation surface */
-#define PITCH_SUBFRAMES 4
-#define PITCH_GRAN_PER_SUBFRAME 5
-#define PITCH_SUBFRAME_LEN (PITCH_FRAME_LEN/PITCH_SUBFRAMES)
-#define PITCH_UPDATE (PITCH_SUBFRAME_LEN/PITCH_GRAN_PER_SUBFRAME)
-/* maximum number of peaks to be examined in correlation surface */
-#define PITCH_MAX_NUM_PEAKS 10
-#define PITCH_PEAK_DECAY 0.85
-/* For weighting filter */
-#define PITCH_WLPCORDER 6
-#define PITCH_WLPCWINLEN PITCH_FRAME_LEN
-#define PITCH_WLPCASYM 0.3 /* asymmetry parameter */
-#define PITCH_WLPCBUFLEN PITCH_WLPCWINLEN
-/* For pitch filter */
-#define PITCH_BUFFSIZE (PITCH_MAX_LAG + 50) /* Extra 50 for fraction and LP filters */
-#define PITCH_INTBUFFSIZE (PITCH_FRAME_LEN+PITCH_BUFFSIZE)
-/* Max rel. step for interpolation */
-#define PITCH_UPSTEP 1.5
-/* Max rel. step for interpolation */
-#define PITCH_DOWNSTEP 0.67
-#define PITCH_FRACS 8
-#define PITCH_FRACORDER 9
-#define PITCH_DAMPORDER 5
-#define PITCH_FILTDELAY 1.5f
-/* stepsize for quantization of the pitch Gain */
-#define PITCH_GAIN_STEPSIZE 0.125
-
-
-
-/* Order of high pass filter */
-#define HPORDER 2
-
-/* some mathematical constants */
-#define LOG2EXP 1.44269504088896 /* log2(exp) */
-#define PI 3.14159265358979
-
-/* Maximum number of iterations allowed to limit payload size */
-#define MAX_PAYLOAD_LIMIT_ITERATION 5
-
-/* Redundant Coding */
-#define RCU_BOTTLENECK_BPS 16000
-#define RCU_TRANSCODING_SCALE 0.40f
-#define RCU_TRANSCODING_SCALE_INVERSE 2.5f
-
-#define RCU_TRANSCODING_SCALE_UB 0.50f
-#define RCU_TRANSCODING_SCALE_UB_INVERSE 2.0f
-
-
-/* Define Error codes */
-/* 6000 General */
-#define ISAC_MEMORY_ALLOCATION_FAILED 6010
-#define ISAC_MODE_MISMATCH 6020
-#define ISAC_DISALLOWED_BOTTLENECK 6030
-#define ISAC_DISALLOWED_FRAME_LENGTH 6040
-#define ISAC_UNSUPPORTED_SAMPLING_FREQUENCY 6050
-
-/* 6200 Bandwidth estimator */
-#define ISAC_RANGE_ERROR_BW_ESTIMATOR 6240
-/* 6400 Encoder */
-#define ISAC_ENCODER_NOT_INITIATED 6410
-#define ISAC_DISALLOWED_CODING_MODE 6420
-#define ISAC_DISALLOWED_FRAME_MODE_ENCODER 6430
-#define ISAC_DISALLOWED_BITSTREAM_LENGTH 6440
-#define ISAC_PAYLOAD_LARGER_THAN_LIMIT 6450
-#define ISAC_DISALLOWED_ENCODER_BANDWIDTH 6460
-/* 6600 Decoder */
-#define ISAC_DECODER_NOT_INITIATED 6610
-#define ISAC_EMPTY_PACKET 6620
-#define ISAC_DISALLOWED_FRAME_MODE_DECODER 6630
-#define ISAC_RANGE_ERROR_DECODE_FRAME_LENGTH 6640
-#define ISAC_RANGE_ERROR_DECODE_BANDWIDTH 6650
-#define ISAC_RANGE_ERROR_DECODE_PITCH_GAIN 6660
-#define ISAC_RANGE_ERROR_DECODE_PITCH_LAG 6670
-#define ISAC_RANGE_ERROR_DECODE_LPC 6680
-#define ISAC_RANGE_ERROR_DECODE_SPECTRUM 6690
-#define ISAC_LENGTH_MISMATCH 6730
-#define ISAC_RANGE_ERROR_DECODE_BANDWITH 6740
-#define ISAC_DISALLOWED_BANDWIDTH_MODE_DECODER 6750
-/* 6800 Call setup formats */
-#define ISAC_INCOMPATIBLE_FORMATS 6810
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SETTINGS_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/spectrum_ar_model_tables.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/spectrum_ar_model_tables.c
deleted file mode 100644
index 92b9c4d..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/spectrum_ar_model_tables.c
+++ /dev/null
@@ -1,138 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "spectrum_ar_model_tables.h"
-#include "settings.h"
-
-/********************* AR Coefficient Tables ************************/
-/* cdf for quantized reflection coefficient 1 */
-const WebRtc_UWord16 WebRtcIsac_kQArRc1Cdf[12] = {
- 0, 2, 4, 129, 7707, 57485, 65495, 65527, 65529, 65531,
- 65533, 65535};
-
-/* cdf for quantized reflection coefficient 2 */
-const WebRtc_UWord16 WebRtcIsac_kQArRc2Cdf[12] = {
- 0, 2, 4, 7, 531, 25298, 64525, 65526, 65529, 65531,
- 65533, 65535};
-
-/* cdf for quantized reflection coefficient 3 */
-const WebRtc_UWord16 WebRtcIsac_kQArRc3Cdf[12] = {
- 0, 2, 4, 6, 620, 22898, 64843, 65527, 65529, 65531,
- 65533, 65535};
-
-/* cdf for quantized reflection coefficient 4 */
-const WebRtc_UWord16 WebRtcIsac_kQArRc4Cdf[12] = {
- 0, 2, 4, 6, 35, 10034, 60733, 65506, 65529, 65531,
- 65533, 65535};
-
-/* cdf for quantized reflection coefficient 5 */
-const WebRtc_UWord16 WebRtcIsac_kQArRc5Cdf[12] = {
- 0, 2, 4, 6, 36, 7567, 56727, 65385, 65529, 65531,
- 65533, 65535};
-
-/* cdf for quantized reflection coefficient 6 */
-const WebRtc_UWord16 WebRtcIsac_kQArRc6Cdf[12] = {
- 0, 2, 4, 6, 14, 6579, 57360, 65409, 65529, 65531,
- 65533, 65535};
-
-/* representation levels for quantized reflection coefficient 1 */
-const WebRtc_Word16 WebRtcIsac_kQArRc1Levels[11] = {
- -32104, -29007, -23202, -15496, -9279, -2577, 5934, 17535, 24512, 29503, 32104
-};
-
-/* representation levels for quantized reflection coefficient 2 */
-const WebRtc_Word16 WebRtcIsac_kQArRc2Levels[11] = {
- -32104, -29503, -23494, -15261, -7309, -1399, 6158, 16381, 24512, 29503, 32104
-};
-
-/* representation levels for quantized reflection coefficient 3 */
-const WebRtc_Word16 WebRtcIsac_kQArRc3Levels[11] = {
--32104, -29503, -23157, -15186, -7347, -1359, 5829, 17535, 24512, 29503, 32104
-};
-
-/* representation levels for quantized reflection coefficient 4 */
-const WebRtc_Word16 WebRtcIsac_kQArRc4Levels[11] = {
--32104, -29503, -24512, -15362, -6665, -342, 6596, 14585, 24512, 29503, 32104
-};
-
-/* representation levels for quantized reflection coefficient 5 */
-const WebRtc_Word16 WebRtcIsac_kQArRc5Levels[11] = {
--32104, -29503, -24512, -15005, -6564, -106, 7123, 14920, 24512, 29503, 32104
-};
-
-/* representation levels for quantized reflection coefficient 6 */
-const WebRtc_Word16 WebRtcIsac_kQArRc6Levels[11] = {
--32104, -29503, -24512, -15096, -6656, -37, 7036, 14847, 24512, 29503, 32104
-};
-
-/* quantization boundary levels for reflection coefficients */
-const WebRtc_Word16 WebRtcIsac_kQArBoundaryLevels[12] = {
--32768, -31441, -27566, -21458, -13612, -4663, 4663, 13612, 21458, 27566, 31441, 32767
-};
-
-/* initial index for AR reflection coefficient quantizer and cdf table search */
-const WebRtc_UWord16 WebRtcIsac_kQArRcInitIndex[6] = {
- 5, 5, 5, 5, 5, 5};
-
-/* pointers to AR cdf tables */
-const WebRtc_UWord16 *WebRtcIsac_kQArRcCdfPtr[AR_ORDER] = {
- WebRtcIsac_kQArRc1Cdf, WebRtcIsac_kQArRc2Cdf, WebRtcIsac_kQArRc3Cdf,
- WebRtcIsac_kQArRc4Cdf, WebRtcIsac_kQArRc5Cdf, WebRtcIsac_kQArRc6Cdf
-};
-
-/* pointers to AR representation levels tables */
-const WebRtc_Word16 *WebRtcIsac_kQArRcLevelsPtr[AR_ORDER] = {
- WebRtcIsac_kQArRc1Levels, WebRtcIsac_kQArRc2Levels, WebRtcIsac_kQArRc3Levels,
- WebRtcIsac_kQArRc4Levels, WebRtcIsac_kQArRc5Levels, WebRtcIsac_kQArRc6Levels
-};
-
-
-/******************** GAIN Coefficient Tables ***********************/
-/* cdf for Gain coefficient */
-const WebRtc_UWord16 WebRtcIsac_kQGainCdf[19] = {
- 0, 2, 4, 6, 8, 10, 12, 14, 16, 1172,
- 11119, 29411, 51699, 64445, 65527, 65529, 65531, 65533, 65535};
-
-/* representation levels for quantized squared Gain coefficient */
-const WebRtc_Word32 WebRtcIsac_kQGain2Levels[18] = {
-// 17, 28, 46, 76, 128, 215, 364, 709, 1268, 1960, 3405, 6078, 11286, 17827, 51918, 134498, 487432, 2048000};
- 128, 128, 128, 128, 128, 215, 364, 709, 1268, 1960, 3405, 6078, 11286, 17827, 51918, 134498, 487432, 2048000};
-/* quantization boundary levels for squared Gain coefficient */
-const WebRtc_Word32 WebRtcIsac_kQGain2BoundaryLevels[19] = {
-0, 21, 35, 59, 99, 166, 280, 475, 815, 1414, 2495, 4505, 8397, 16405, 34431, 81359, 240497, 921600, 0x7FFFFFFF};
-
-/* pointers to Gain cdf table */
-const WebRtc_UWord16 *WebRtcIsac_kQGainCdf_ptr[1] = {WebRtcIsac_kQGainCdf};
-
-/* Gain initial index for gain quantizer and cdf table search */
-const WebRtc_UWord16 WebRtcIsac_kQGainInitIndex[1] = {11};
-
-/************************* Cosine Tables ****************************/
-/* Cosine table */
-const WebRtc_Word16 WebRtcIsac_kCos[6][60] = {
-{512, 512, 511, 510, 508, 507, 505, 502, 499, 496, 493, 489, 485, 480, 476, 470, 465, 459, 453, 447,
-440, 433, 426, 418, 410, 402, 394, 385, 376, 367, 357, 348, 338, 327, 317, 306, 295, 284, 273, 262,
-250, 238, 226, 214, 202, 190, 177, 165, 152, 139, 126, 113, 100, 87, 73, 60, 47, 33, 20, 7},
-{512, 510, 508, 503, 498, 491, 483, 473, 462, 450, 437, 422, 406, 389, 371, 352, 333, 312, 290, 268,
-244, 220, 196, 171, 145, 120, 93, 67, 40, 13, -13, -40, -67, -93, -120, -145, -171, -196, -220, -244,
--268, -290, -312, -333, -352, -371, -389, -406, -422, -437, -450, -462, -473, -483, -491, -498, -503, -508, -510, -512},
-{512, 508, 502, 493, 480, 465, 447, 426, 402, 376, 348, 317, 284, 250, 214, 177, 139, 100, 60, 20,
--20, -60, -100, -139, -177, -214, -250, -284, -317, -348, -376, -402, -426, -447, -465, -480, -493, -502, -508, -512,
--512, -508, -502, -493, -480, -465, -447, -426, -402, -376, -348, -317, -284, -250, -214, -177, -139, -100, -60, -20},
-{511, 506, 495, 478, 456, 429, 398, 362, 322, 279, 232, 183, 133, 80, 27, -27, -80, -133, -183, -232,
--279, -322, -362, -398, -429, -456, -478, -495, -506, -511, -511, -506, -495, -478, -456, -429, -398, -362, -322, -279,
--232, -183, -133, -80, -27, 27, 80, 133, 183, 232, 279, 322, 362, 398, 429, 456, 478, 495, 506, 511},
-{511, 502, 485, 459, 426, 385, 338, 284, 226, 165, 100, 33, -33, -100, -165, -226, -284, -338, -385, -426,
--459, -485, -502, -511, -511, -502, -485, -459, -426, -385, -338, -284, -226, -165, -100, -33, 33, 100, 165, 226,
-284, 338, 385, 426, 459, 485, 502, 511, 511, 502, 485, 459, 426, 385, 338, 284, 226, 165, 100, 33},
-{510, 498, 473, 437, 389, 333, 268, 196, 120, 40, -40, -120, -196, -268, -333, -389, -437, -473, -498, -510,
--510, -498, -473, -437, -389, -333, -268, -196, -120, -40, 40, 120, 196, 268, 333, 389, 437, 473, 498, 510,
-510, 498, 473, 437, 389, 333, 268, 196, 120, 40, -40, -120, -196, -268, -333, -389, -437, -473, -498, -510}
-};
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/spectrum_ar_model_tables.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/spectrum_ar_model_tables.h
deleted file mode 100644
index 159245b..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/spectrum_ar_model_tables.h
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * spectrum_ar_model_tables.h
- *
- * This file contains definitions of tables with AR coefficients,
- * Gain coefficients and cosine tables.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_
-
-#include "structs.h"
-
-/********************* AR Coefficient Tables ************************/
-/* cdf for quantized reflection coefficient 1 */
-extern const WebRtc_UWord16 WebRtcIsac_kQArRc1Cdf[12];
-
-/* cdf for quantized reflection coefficient 2 */
-extern const WebRtc_UWord16 WebRtcIsac_kQArRc2Cdf[12];
-
-/* cdf for quantized reflection coefficient 3 */
-extern const WebRtc_UWord16 WebRtcIsac_kQArRc3Cdf[12];
-
-/* cdf for quantized reflection coefficient 4 */
-extern const WebRtc_UWord16 WebRtcIsac_kQArRc4Cdf[12];
-
-/* cdf for quantized reflection coefficient 5 */
-extern const WebRtc_UWord16 WebRtcIsac_kQArRc5Cdf[12];
-
-/* cdf for quantized reflection coefficient 6 */
-extern const WebRtc_UWord16 WebRtcIsac_kQArRc6Cdf[12];
-
-/* quantization boundary levels for reflection coefficients */
-extern const WebRtc_Word16 WebRtcIsac_kQArBoundaryLevels[12];
-
-/* initial indices for AR reflection coefficient quantizer and cdf table search */
-extern const WebRtc_UWord16 WebRtcIsac_kQArRcInitIndex[AR_ORDER];
-
-/* pointers to AR cdf tables */
-extern const WebRtc_UWord16 *WebRtcIsac_kQArRcCdfPtr[AR_ORDER];
-
-/* pointers to AR representation levels tables */
-extern const WebRtc_Word16 *WebRtcIsac_kQArRcLevelsPtr[AR_ORDER];
-
-
-/******************** GAIN Coefficient Tables ***********************/
-/* cdf for Gain coefficient */
-extern const WebRtc_UWord16 WebRtcIsac_kQGainCdf[19];
-
-/* representation levels for quantized Gain coefficient */
-extern const WebRtc_Word32 WebRtcIsac_kQGain2Levels[18];
-
-/* squared quantization boundary levels for Gain coefficient */
-extern const WebRtc_Word32 WebRtcIsac_kQGain2BoundaryLevels[19];
-
-/* pointer to Gain cdf table */
-extern const WebRtc_UWord16 *WebRtcIsac_kQGainCdf_ptr[1];
-
-/* Gain initial index for gain quantizer and cdf table search */
-extern const WebRtc_UWord16 WebRtcIsac_kQGainInitIndex[1];
-
-/************************* Cosine Tables ****************************/
-/* Cosine table */
-extern const WebRtc_Word16 WebRtcIsac_kCos[6][60];
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_SPECTRUM_AR_MODEL_TABLES_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/structs.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/structs.h
deleted file mode 100644
index 7523ad6..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/structs.h
+++ /dev/null
@@ -1,478 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * structs.h
- *
- * This header file contains all the structs used in the ISAC codec
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_
-
-
-#include "typedefs.h"
-#include "settings.h"
-#include "isac.h"
-
-typedef struct Bitstreamstruct {
-
- WebRtc_UWord8 stream[STREAM_SIZE_MAX];
- WebRtc_UWord32 W_upper;
- WebRtc_UWord32 streamval;
- WebRtc_UWord32 stream_index;
-
-} Bitstr;
-
-typedef struct {
-
- double DataBufferLo[WINLEN];
- double DataBufferHi[WINLEN];
-
- double CorrBufLo[ORDERLO+1];
- double CorrBufHi[ORDERHI+1];
-
- float PreStateLoF[ORDERLO+1];
- float PreStateLoG[ORDERLO+1];
- float PreStateHiF[ORDERHI+1];
- float PreStateHiG[ORDERHI+1];
- float PostStateLoF[ORDERLO+1];
- float PostStateLoG[ORDERLO+1];
- float PostStateHiF[ORDERHI+1];
- float PostStateHiG[ORDERHI+1];
-
- double OldEnergy;
-
-} MaskFiltstr;
-
-
-typedef struct {
-
- //state vectors for each of the two analysis filters
- double INSTAT1[2*(QORDER-1)];
- double INSTAT2[2*(QORDER-1)];
- double INSTATLA1[2*(QORDER-1)];
- double INSTATLA2[2*(QORDER-1)];
- double INLABUF1[QLOOKAHEAD];
- double INLABUF2[QLOOKAHEAD];
-
- float INSTAT1_float[2*(QORDER-1)];
- float INSTAT2_float[2*(QORDER-1)];
- float INSTATLA1_float[2*(QORDER-1)];
- float INSTATLA2_float[2*(QORDER-1)];
- float INLABUF1_float[QLOOKAHEAD];
- float INLABUF2_float[QLOOKAHEAD];
-
- /* High pass filter */
- double HPstates[HPORDER];
- float HPstates_float[HPORDER];
-
-} PreFiltBankstr;
-
-
-typedef struct {
-
- //state vectors for each of the two analysis filters
- double STATE_0_LOWER[2*POSTQORDER];
- double STATE_0_UPPER[2*POSTQORDER];
-
- /* High pass filter */
- double HPstates1[HPORDER];
- double HPstates2[HPORDER];
-
- float STATE_0_LOWER_float[2*POSTQORDER];
- float STATE_0_UPPER_float[2*POSTQORDER];
-
- float HPstates1_float[HPORDER];
- float HPstates2_float[HPORDER];
-
-} PostFiltBankstr;
-
-typedef struct {
-
- //data buffer for pitch filter
- double ubuf[PITCH_BUFFSIZE];
-
- //low pass state vector
- double ystate[PITCH_DAMPORDER];
-
- //old lag and gain
- double oldlagp[1];
- double oldgainp[1];
-
-} PitchFiltstr;
-
-typedef struct {
-
- //data buffer
- double buffer[PITCH_WLPCBUFLEN];
-
- //state vectors
- double istate[PITCH_WLPCORDER];
- double weostate[PITCH_WLPCORDER];
- double whostate[PITCH_WLPCORDER];
-
- //LPC window -> should be a global array because constant
- double window[PITCH_WLPCWINLEN];
-
-} WeightFiltstr;
-
-typedef struct {
-
- //for inital estimator
- double dec_buffer[PITCH_CORR_LEN2 + PITCH_CORR_STEP2 +
- PITCH_MAX_LAG/2 - PITCH_FRAME_LEN/2+2];
- double decimator_state[2*ALLPASSSECTIONS+1];
- double hp_state[2];
-
- double whitened_buf[QLOOKAHEAD];
-
- double inbuf[QLOOKAHEAD];
-
- PitchFiltstr PFstr_wght;
- PitchFiltstr PFstr;
- WeightFiltstr Wghtstr;
-
-} PitchAnalysisStruct;
-
-
-
-/* Have instance of struct together with other iSAC structs */
-typedef struct {
-
- /* Previous frame length (in ms) */
- WebRtc_Word32 prev_frame_length;
-
- /* Previous RTP timestamp from received
- packet (in samples relative beginning) */
- WebRtc_Word32 prev_rec_rtp_number;
-
- /* Send timestamp for previous packet (in ms using timeGetTime()) */
- WebRtc_UWord32 prev_rec_send_ts;
-
- /* Arrival time for previous packet (in ms using timeGetTime()) */
- WebRtc_UWord32 prev_rec_arr_ts;
-
- /* rate of previous packet, derived from RTP timestamps (in bits/s) */
- float prev_rec_rtp_rate;
-
- /* Time sinse the last update of the BN estimate (in ms) */
- WebRtc_UWord32 last_update_ts;
-
- /* Time sinse the last reduction (in ms) */
- WebRtc_UWord32 last_reduction_ts;
-
- /* How many times the estimate was update in the beginning */
- WebRtc_Word32 count_tot_updates_rec;
-
- /* The estimated bottle neck rate from there to here (in bits/s) */
- WebRtc_Word32 rec_bw;
- float rec_bw_inv;
- float rec_bw_avg;
- float rec_bw_avg_Q;
-
- /* The estimated mean absolute jitter value,
- as seen on this side (in ms) */
- float rec_jitter;
- float rec_jitter_short_term;
- float rec_jitter_short_term_abs;
- float rec_max_delay;
- float rec_max_delay_avg_Q;
-
- /* (assumed) bitrate for headers (bps) */
- float rec_header_rate;
-
- /* The estimated bottle neck rate from here to there (in bits/s) */
- float send_bw_avg;
-
- /* The estimated mean absolute jitter value, as seen on
- the other siee (in ms) */
- float send_max_delay_avg;
-
- // number of packets received since last update
- int num_pkts_rec;
-
- int num_consec_rec_pkts_over_30k;
-
- // flag for marking that a high speed network has been
- // detected downstream
- int hsn_detect_rec;
-
- int num_consec_snt_pkts_over_30k;
-
- // flag for marking that a high speed network has
- // been detected upstream
- int hsn_detect_snd;
-
- WebRtc_UWord32 start_wait_period;
-
- int in_wait_period;
-
- int change_to_WB;
-
- WebRtc_UWord32 senderTimestamp;
- WebRtc_UWord32 receiverTimestamp;
- //enum IsacSamplingRate incomingStreamSampFreq;
- WebRtc_UWord16 numConsecLatePkts;
- float consecLatency;
- WebRtc_Word16 inWaitLatePkts;
-} BwEstimatorstr;
-
-
-typedef struct {
-
- /* boolean, flags if previous packet exceeded B.N. */
- int PrevExceed;
- /* ms */
- int ExceedAgo;
- /* packets left to send in current burst */
- int BurstCounter;
- /* packets */
- int InitCounter;
- /* ms remaining in buffer when next packet will be sent */
- double StillBuffered;
-
-} RateModel;
-
-
-typedef struct {
-
- unsigned int SpaceAlloced;
- unsigned int MaxPermAlloced;
- double Tmp0[MAXFFTSIZE];
- double Tmp1[MAXFFTSIZE];
- double Tmp2[MAXFFTSIZE];
- double Tmp3[MAXFFTSIZE];
- int Perm[MAXFFTSIZE];
- int factor [NFACTOR];
-
-} FFTstr;
-
-
-/* The following strutc is used to store data from encoding, to make it
- fast and easy to construct a new bitstream with a different Bandwidth
- estimate. All values (except framelength and minBytes) is double size to
- handle 60 ms of data.
-*/
-typedef struct {
-
- /* Used to keep track of if it is first or second part of 60 msec packet */
- int startIdx;
-
- /* Frame length in samples */
- WebRtc_Word16 framelength;
-
- /* Pitch Gain */
- int pitchGain_index[2];
-
- /* Pitch Lag */
- double meanGain[2];
- int pitchIndex[PITCH_SUBFRAMES*2];
-
- /* LPC */
- int LPCmodel[2];
- int LPCindex_s[108*2]; /* KLT_ORDER_SHAPE = 108 */
- int LPCindex_g[12*2]; /* KLT_ORDER_GAIN = 12 */
- double LPCcoeffs_lo[(ORDERLO+1)*SUBFRAMES*2];
- double LPCcoeffs_hi[(ORDERHI+1)*SUBFRAMES*2];
-
- /* Encode Spec */
- WebRtc_Word16 fre[FRAMESAMPLES];
- WebRtc_Word16 fim[FRAMESAMPLES];
- WebRtc_Word16 AvgPitchGain[2];
-
- /* Used in adaptive mode only */
- int minBytes;
-
-} ISAC_SaveEncData_t;
-
-
-typedef struct {
-
- int indexLPCShape[UB_LPC_ORDER * UB16_LPC_VEC_PER_FRAME];
- double lpcGain[SUBFRAMES<<1];
- int lpcGainIndex[SUBFRAMES<<1];
-
- Bitstr bitStreamObj;
-
- WebRtc_Word16 realFFT[FRAMESAMPLES_HALF];
- WebRtc_Word16 imagFFT[FRAMESAMPLES_HALF];
-} ISACUBSaveEncDataStruct;
-
-
-
-typedef struct {
-
- Bitstr bitstr_obj;
- MaskFiltstr maskfiltstr_obj;
- PreFiltBankstr prefiltbankstr_obj;
- PitchFiltstr pitchfiltstr_obj;
- PitchAnalysisStruct pitchanalysisstr_obj;
- FFTstr fftstr_obj;
- ISAC_SaveEncData_t SaveEnc_obj;
-
- int buffer_index;
- WebRtc_Word16 current_framesamples;
-
- float data_buffer_float[FRAMESAMPLES_30ms];
-
- int frame_nb;
- double bottleneck;
- WebRtc_Word16 new_framelength;
- double s2nr;
-
- /* Maximum allowed number of bits for a 30 msec packet */
- WebRtc_Word16 payloadLimitBytes30;
- /* Maximum allowed number of bits for a 30 msec packet */
- WebRtc_Word16 payloadLimitBytes60;
- /* Maximum allowed number of bits for both 30 and 60 msec packet */
- WebRtc_Word16 maxPayloadBytes;
- /* Maximum allowed rate in bytes per 30 msec packet */
- WebRtc_Word16 maxRateInBytes;
-
- /*---
- If set to 1 iSAC will not addapt the frame-size, if used in
- channel-adaptive mode. The initial value will be used for all rates.
- ---*/
- WebRtc_Word16 enforceFrameSize;
-
- /*-----
- This records the BWE index the encoder injected into the bit-stream.
- It will be used in RCU. The same BWE index of main paylaod will be in
- the redundant payload. We can not retrive it from BWE because it is
- a recursive procedure (WebRtcIsac_GetDownlinkBwJitIndexImpl) and has to be
- called only once per each encode.
- -----*/
- WebRtc_Word16 lastBWIdx;
-} ISACLBEncStruct;
-
-typedef struct {
-
- Bitstr bitstr_obj;
- MaskFiltstr maskfiltstr_obj;
- PreFiltBankstr prefiltbankstr_obj;
- FFTstr fftstr_obj;
- ISACUBSaveEncDataStruct SaveEnc_obj;
-
- int buffer_index;
- float data_buffer_float[MAX_FRAMESAMPLES +
- LB_TOTAL_DELAY_SAMPLES];
- double bottleneck;
- /* Maximum allowed number of bits for a 30 msec packet */
- //WebRtc_Word16 payloadLimitBytes30;
- /* Maximum allowed number of bits for both 30 and 60 msec packet */
- //WebRtc_Word16 maxPayloadBytes;
- WebRtc_Word16 maxPayloadSizeBytes;
-
- double lastLPCVec[UB_LPC_ORDER];
- WebRtc_Word16 numBytesUsed;
- WebRtc_Word16 lastJitterInfo;
-} ISACUBEncStruct;
-
-
-
-typedef struct {
-
- Bitstr bitstr_obj;
- MaskFiltstr maskfiltstr_obj;
- PostFiltBankstr postfiltbankstr_obj;
- PitchFiltstr pitchfiltstr_obj;
- FFTstr fftstr_obj;
-
-} ISACLBDecStruct;
-
-typedef struct {
-
- Bitstr bitstr_obj;
- MaskFiltstr maskfiltstr_obj;
- PostFiltBankstr postfiltbankstr_obj;
- FFTstr fftstr_obj;
-
-} ISACUBDecStruct;
-
-
-
-typedef struct {
-
- ISACLBEncStruct ISACencLB_obj;
- ISACLBDecStruct ISACdecLB_obj;
-} ISACLBStruct;
-
-
-typedef struct {
-
- ISACUBEncStruct ISACencUB_obj;
- ISACUBDecStruct ISACdecUB_obj;
-} ISACUBStruct;
-
-/*
- This struct is used to take a snapshot of the entropy coder and LPC gains
- right before encoding LPC gains. This allows us to go back to that state
- if we like to limit the payload size.
-*/
-typedef struct {
- /* 6 lower-band & 6 upper-band */
- double loFiltGain[SUBFRAMES];
- double hiFiltGain[SUBFRAMES];
- /* Upper boundary of interval W */
- WebRtc_UWord32 W_upper;
- WebRtc_UWord32 streamval;
- /* Index to the current position in bytestream */
- WebRtc_UWord32 stream_index;
- WebRtc_UWord8 stream[3];
-} transcode_obj;
-
-
-typedef struct {
- // lower-band codec instance
- ISACLBStruct instLB;
- // upper-band codec instance
- ISACUBStruct instUB;
-
- // Bandwidth Estimator and model for the rate.
- BwEstimatorstr bwestimator_obj;
- RateModel rate_data_obj;
- double MaxDelay;
-
- /* 0 = adaptive; 1 = instantaneous */
- WebRtc_Word16 codingMode;
-
- // overall bottleneck of the codec
- WebRtc_Word32 bottleneck;
-
- // QMF Filter state
- WebRtc_Word32 analysisFBState1[FB_STATE_SIZE_WORD32];
- WebRtc_Word32 analysisFBState2[FB_STATE_SIZE_WORD32];
- WebRtc_Word32 synthesisFBState1[FB_STATE_SIZE_WORD32];
- WebRtc_Word32 synthesisFBState2[FB_STATE_SIZE_WORD32];
-
- // Error Code
- WebRtc_Word16 errorCode;
-
- // bandwidth of the encoded audio 8, 12 or 16 kHz
- enum ISACBandwidth bandwidthKHz;
- // Sampling rate of audio, encoder and decode, 8 or 16 kHz
- enum IsacSamplingRate encoderSamplingRateKHz;
- enum IsacSamplingRate decoderSamplingRateKHz;
- // Flag to keep track of initializations, lower & upper-band
- // encoder and decoder.
- WebRtc_Word16 initFlag;
-
- // Flag to to indicate signal bandwidth switch
- WebRtc_Word16 resetFlag_8kHz;
-
- // Maximum allowed rate, measured in Bytes per 30 ms.
- WebRtc_Word16 maxRateBytesPer30Ms;
- // Maximum allowed payload-size, measured in Bytes.
- WebRtc_Word16 maxPayloadSizeBytes;
-} ISACMainStruct;
-
-#endif /* WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_SOURCE_STRUCTS_H_ */
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/transform.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/source/transform.c
deleted file mode 100644
index 97b801a..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/source/transform.c
+++ /dev/null
@@ -1,131 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "settings.h"
-#include "fft.h"
-#include "codec.h"
-#include "os_specific_inline.h"
-#include <math.h>
-
-static double costab1[FRAMESAMPLES_HALF];
-static double sintab1[FRAMESAMPLES_HALF];
-static double costab2[FRAMESAMPLES_QUARTER];
-static double sintab2[FRAMESAMPLES_QUARTER];
-
-void WebRtcIsac_InitTransform()
-{
- int k;
- double fact, phase;
-
- fact = PI / (FRAMESAMPLES_HALF);
- phase = 0.0;
- for (k = 0; k < FRAMESAMPLES_HALF; k++) {
- costab1[k] = cos(phase);
- sintab1[k] = sin(phase);
- phase += fact;
- }
-
- fact = PI * ((double) (FRAMESAMPLES_HALF - 1)) / ((double) FRAMESAMPLES_HALF);
- phase = 0.5 * fact;
- for (k = 0; k < FRAMESAMPLES_QUARTER; k++) {
- costab2[k] = cos(phase);
- sintab2[k] = sin(phase);
- phase += fact;
- }
-}
-
-
-void WebRtcIsac_Time2Spec(double *inre1,
- double *inre2,
- WebRtc_Word16 *outreQ7,
- WebRtc_Word16 *outimQ7,
- FFTstr *fftstr_obj)
-{
-
- int k;
- int dims[1];
- double tmp1r, tmp1i, xr, xi, yr, yi, fact;
- double tmpre[FRAMESAMPLES_HALF], tmpim[FRAMESAMPLES_HALF];
-
-
- dims[0] = FRAMESAMPLES_HALF;
-
-
- /* Multiply with complex exponentials and combine into one complex vector */
- fact = 0.5 / sqrt(FRAMESAMPLES_HALF);
- for (k = 0; k < FRAMESAMPLES_HALF; k++) {
- tmp1r = costab1[k];
- tmp1i = sintab1[k];
- tmpre[k] = (inre1[k] * tmp1r + inre2[k] * tmp1i) * fact;
- tmpim[k] = (inre2[k] * tmp1r - inre1[k] * tmp1i) * fact;
- }
-
-
- /* Get DFT */
- WebRtcIsac_Fftns(1, dims, tmpre, tmpim, -1, 1.0, fftstr_obj);
-
- /* Use symmetry to separate into two complex vectors and center frames in time around zero */
- for (k = 0; k < FRAMESAMPLES_QUARTER; k++) {
- xr = tmpre[k] + tmpre[FRAMESAMPLES_HALF - 1 - k];
- yi = -tmpre[k] + tmpre[FRAMESAMPLES_HALF - 1 - k];
- xi = tmpim[k] - tmpim[FRAMESAMPLES_HALF - 1 - k];
- yr = tmpim[k] + tmpim[FRAMESAMPLES_HALF - 1 - k];
-
- tmp1r = costab2[k];
- tmp1i = sintab2[k];
- outreQ7[k] = (WebRtc_Word16)WebRtcIsac_lrint((xr * tmp1r - xi * tmp1i) * 128.0);
- outimQ7[k] = (WebRtc_Word16)WebRtcIsac_lrint((xr * tmp1i + xi * tmp1r) * 128.0);
- outreQ7[FRAMESAMPLES_HALF - 1 - k] = (WebRtc_Word16)WebRtcIsac_lrint((-yr * tmp1i - yi * tmp1r) * 128.0);
- outimQ7[FRAMESAMPLES_HALF - 1 - k] = (WebRtc_Word16)WebRtcIsac_lrint((-yr * tmp1r + yi * tmp1i) * 128.0);
- }
-}
-
-
-void WebRtcIsac_Spec2time(double *inre, double *inim, double *outre1, double *outre2, FFTstr *fftstr_obj)
-{
-
- int k;
- double tmp1r, tmp1i, xr, xi, yr, yi, fact;
-
- int dims;
-
- dims = FRAMESAMPLES_HALF;
-
- for (k = 0; k < FRAMESAMPLES_QUARTER; k++) {
- /* Move zero in time to beginning of frames */
- tmp1r = costab2[k];
- tmp1i = sintab2[k];
- xr = inre[k] * tmp1r + inim[k] * tmp1i;
- xi = inim[k] * tmp1r - inre[k] * tmp1i;
- yr = -inim[FRAMESAMPLES_HALF - 1 - k] * tmp1r - inre[FRAMESAMPLES_HALF - 1 - k] * tmp1i;
- yi = -inre[FRAMESAMPLES_HALF - 1 - k] * tmp1r + inim[FRAMESAMPLES_HALF - 1 - k] * tmp1i;
-
- /* Combine into one vector, z = x + j * y */
- outre1[k] = xr - yi;
- outre1[FRAMESAMPLES_HALF - 1 - k] = xr + yi;
- outre2[k] = xi + yr;
- outre2[FRAMESAMPLES_HALF - 1 - k] = -xi + yr;
- }
-
-
- /* Get IDFT */
- WebRtcIsac_Fftns(1, &dims, outre1, outre2, 1, FRAMESAMPLES_HALF, fftstr_obj);
-
-
- /* Demodulate and separate */
- fact = sqrt(FRAMESAMPLES_HALF);
- for (k = 0; k < FRAMESAMPLES_HALF; k++) {
- tmp1r = costab1[k];
- tmp1i = sintab1[k];
- xr = (outre1[k] * tmp1r - outre2[k] * tmp1i) * fact;
- outre2[k] = (outre2[k] * tmp1r + outre1[k] * tmp1i) * fact;
- outre1[k] = xr;
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACLongtest.txt b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACLongtest.txt
deleted file mode 100644
index 3f05224..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACLongtest.txt
+++ /dev/null
@@ -1,433 +0,0 @@
-#!/bin/bash
-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
-
-
-
-if [ "$1" = "x64" ] || [ "$2" = "x64" ] || [ "$#" -eq 0 ]
- then
- PLATFORM=_X64
- ISAC=../x64/Release/ReleaseTest-API_2005.exe
-elif [ "$1" = "LINUX" ] || [ "$2" = "LINUX" ]
- then
- PLATFORM=_linux
- ISAC=../ReleaseTest-API/isacswtest
-else
- PLATFORM=_2005
- ISAC=../win32/Release/ReleaseTest-API_2005.exe
-fi
-
-if [ "$#" -eq 0 ] || [ "$1" = "all" ] || [ "$1" = "wb" ]
- then
- LOGFILE=logNormal"$PLATFORM".txt
- echo "START ISAC WB TEST" > $LOGFILE
- echo >> $LOGFILE
-
- INFILES=$(cat InputFiles.txt)
- SUBSET=$(cat InputFilesFew.txt)
- CHANNELFILES=$(cat ChannelFiles.txt)
- CHANNELLIST=($(cat ChannelFiles.txt))
- INDIR=../data/orig
- OUTDIR=../dataqa"$PLATFORM"
- mkdir -p $OUTDIR
- rm -f $OUTDIR/*
-
- idx=0
- RATE=10000
- FRAMESIZE=30
-
-
- for file in $INFILES # loop over all input files
- do
-
- echo "Input file: " $file
- echo "-----------------------------------"
- echo "Instantaneous with RATE " $RATE ", and Frame-size " $FRAMESIZE
- $ISAC -I -B $RATE -FL $FRAMESIZE -FS 16 $INDIR/"$file" $OUTDIR/i_"$FRAMESIZE"_"$RATE"_"$file" >> $LOGFILE
- echo
-
- name="${CHANNELLIST[$idx]}"
- echo "Adaptive with channel file: " $name
-
- $ISAC -B $INDIR/${CHANNELLIST[$idx]} -FS 16 $INDIR/"$file" $OUTDIR/a_${name%.*}_"$file" >> $LOGFILE
-
- echo
- echo
-
-# alternate between 30 & 60 ms.
- if [ $FRAMESIZE -eq 30 ]
- then
- FRAMESIZE=60
- else
- FRAMESIZE=30
- fi
-
-# rate between 10000 to 32000 bits/sec
- if [ $RATE -le 30000 ]
- then
- let "RATE=RATE+2000"
- else
- let "RATE=10000"
- fi
-
-# there are only three channel file
- if [ $idx -ge 2 ]; then
- idx=0
- else
- let "idx=idx+1"
- fi
-
- done
-
- idx=0
-
-# loop over the subset of input files
- for file in $SUBSET
- do
-
- if [ $idx -eq 0 ]; then
- $ISAC -B $INDIR/${CHANNELLIST[0]} -FL 30 -FIXED_FL -FS 16 $INDIR/"$file" $OUTDIR/a30_"$file" >> $LOGFILE
- idx=1
- else
- $ISAC -B $INDIR/${CHANNELLIST[0]} -FL 60 -FIXED_FL -FS 16 $INDIR/"$file" $OUTDIR/a60_"$file" >> $LOGFILE
- idx=0
- fi
- done
-
- $ISAC -B $INDIR/${CHANNELLIST[0]} -INITRATE 25000 -FL 30 -FS 16 $INDIR/"$file" $OUTDIR/a60_Init25kbps_"$file" >> $LOGFILE
-
- echo
- echo WIDEBAND DONE!
- echo
- echo
-fi
-
-if [ "$#" -eq 0 ] || [ "$1" = "all" ] || [ "$1" = "swb" ]
- then
-
- LOGFILE=logNormal_SWB"$PLATFORM".txt
- echo "START ISAC SWB TEST" > $LOGFILE
- echo >> $LOGFILE
-
- echo STARTING TO TEST SUPER-WIDEBAND
-
- INFILES=$(cat InputFilesSWB.txt)
- INDIR=../data/origswb
- OUTDIR=../dataqaswb"$PLATFORM"
- mkdir -p $OUTDIR
- rm -f $OUTDIR/*
-
- for file in $INFILES
- do
- echo
- echo "Input file: " $file
- echo "--------------------------------"
- for RATE in 12000 20000 32000 38000 45000 50000 56000
- do
-
- echo "Rate " $RATE
- $ISAC -I -B $RATE -FL 30 -FS 32 $INDIR/"$file" $OUTDIR/swb_"$RATE"_"$file" >> $LOGFILE
- echo
-
- done
-
- done
-fi
-
-if [ "$#" -eq 0 ] || [ "$1" = "all" ] || [ "$1" = "API" ]
- then
-
- LOGFILE_API=logNormal_API"$PLATFORM".txt
- echo
- echo
- echo "START ISAC API TEST" > $LOGFILE_API
- echo >> $LOGFILE_API
- idx=1
- echo " Test Enforcement of frame-size"
- echo "========================================================================================"
- mkdir -p ../FrameSizeLim"$PLATFORM"
- rm -f ../FrameSizeLim"$PLATFORM"/*
- echo
- echo "-- No enforcement; BN 10000"
- echo
- $ISAC -B 10000 -FS 16 ../data/orig/speech_and_misc_WB.pcm \
- ../FrameSizeLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
- echo "-- Now Enforce 30 ms frame size with the same bottleneck"
- echo "There should not be any 60 ms frame"
- echo
- $ISAC -B 10000 -FL 30 -FIXED_FL -FS 16 ../data/orig/speech_and_misc_WB.pcm \
- ../FrameSizeLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
- echo "-- No enforcement; BN 32000"
- echo
- $ISAC -B 32000 -FS 16 ../data/orig/speech_and_misc_WB.pcm \
- ../FrameSizeLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
- echo "-- Now Enforce 60 ms frame size with the same bottleneck"
- echo "There should not be any 30 ms frame"
- echo
- $ISAC -B 32000 -FL 60 -FIXED_FL -FS 16 ../data/orig/speech_and_misc_WB.pcm \
- ../FrameSizeLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
- echo
- echo
- echo
-
- echo " Test Limiting of Payload Size and Rate"
- echo "========================================================================================"
- mkdir -p ../PayloadLim"$PLATFORM"
- rm -f ../PayloadLim"$PLATFORM"/*
- echo
- echo
- echo "-- No Limit, frame-size 60 ms, WIDEBAND"
- echo
- $ISAC -I -B 32000 -FL 60 -FS 16 ../data/orig/speech_and_misc_WB.pcm \
- ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
- echo "-- Payload-size limit of 250, frame-size 60 ms, WIDEBAND"
- echo
- $ISAC -I -B 32000 -FL 60 -FS 16 -MAX 250 ../data/orig/speech_and_misc_WB.pcm \
- ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
- echo "-- Rate limit of 33 kbps for 60 ms frame-size"
- echo
- $ISAC -I -B 32000 -FL 60 -FS 16 -MAXRATE 33000 ../data/orig/speech_and_misc_WB.pcm \
- ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo "________________________________________________________"
- echo
- echo
-
- echo "-- No Limit, frame-size 30 ms, WIDEBAND"
- echo
- $ISAC -I -B 32000 -FL 30 -FS 16 ../data/orig/speech_and_misc_WB.pcm \
- ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
- echo "-- Payload-size limit of 130, frame-size 30 ms, WIDEBAND"
- echo
- $ISAC -I -B 32000 -FL 30 -FS 16 -MAX 130 ../data/orig/speech_and_misc_WB.pcm \
- ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
- echo "-- Rate limit of 33 kbps for 30 ms frame-size, wideband"
- echo
- $ISAC -I -B 32000 -FL 30 -FS 16 -MAXRATE 33000 ../data/orig/speech_and_misc_WB.pcm \
- ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo "________________________________________________________"
- echo
- echo
-
- echo "-- No limit for 32 kbps, 30 ms, SUPER-WIDEBAND"
- echo
- $ISAC -I -B 32000 -FL 30 -FS 32 ../data/origswb/jstest_32.pcm \
- ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
- echo "-- Payload limit of 130 bytes for 32 kbps, 30 ms, SUPER-WIDEBAND"
- echo
- $ISAC -I -B 32000 -FL 30 -FS 32 -MAX 130 ../data/origswb/jstest_32.pcm \
- ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo "________________________________________________________"
- echo
- echo
-
- echo "-- No limit, Rate 45 kbps, 30 ms, SUPER-WIDEBAND, 12 kHz"
- echo
- $ISAC -I -B 45000 -FL 30 -FS 32 ../data/origswb/jstest_32.pcm \
- ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
- echo "-- Rate limit of 46 kbps for 42 kbps, 30 ms, SUPER-WIDEBAND, 12 kHz"
- echo
- $ISAC -I -B 45000 -FL 30 -FS 32 -MAXRATE 46000 ../data/origswb/jstest_32.pcm \
- ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
- echo "-- Payload limit of 170 bytes for 45 kbps, 30 ms, SUPER-WIDEBAND, 12 kHz"
- echo
- $ISAC -I -B 45000 -FL 30 -FS 32 -MAX 170 ../data/origswb/jstest_32.pcm \
- ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo "________________________________________________________"
- echo
- echo
-
- echo "-- No limit for 56 kbps, 30 ms, SUPER-WIDEBAND, 16 kHz"
- echo
- $ISAC -I -B 56000 -FL 30 -FS 32 ../data/origswb/jstest_32.pcm \
- ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
- echo "-- Payload limit of 200 bytes for 56 kbps 30 ms, SUPER-WIDEBAND, 16 kHz"
- echo
- $ISAC -I -B 56000 -FL 30 -FS 32 -MAX 200 ../data/origswb/jstest_32.pcm \
- ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
- echo "-- Rate limit of 57 kbps for 56 kbps 30 ms, SUPER-WIDEBAND, 16 kHz"
- echo
- $ISAC -I -B 56000 -FL 30 -FS 32 -MAXRATE 57000 ../data/origswb/jstest_32.pcm \
- ../PayloadLim"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo "________________________________________________________"
- echo
- echo
- echo
- echo
- echo
-
- echo " Test Trans-Coding"
- echo "========================================================================================"
- mkdir -p ../Transcoding"$PLATFORM"
- rm -f ../Transcoding"$PLATFORM"/*
- echo
- echo
- echo "-- 20 kbps, 30 ms, WIDEBAND"
- echo
- $ISAC -I -B 20000 -FL 30 -FS 16 ../data/orig/speech_and_misc_WB.pcm \
- ../Transcoding"$PLATFORM"/APITest_refTrans20WB.pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
-
- echo "-- 32 kbps trans-coding to 20 kbps, 30 ms, WIDEBAND"
- echo
- $ISAC -I -B 32000 -FL 30 -FS 16 -T 20000 ../Transcoding"$PLATFORM"/APITest_32T20.pcm \
- ../data/orig/speech_and_misc_WB.pcm ../Transcoding"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo "________________________________________________________"
-
- echo
- echo
- echo "-- 38 kbps, 30 ms, SUPER-WIDEBAND"
- echo
- $ISAC -I -B 38000 -FL 30 -FS 32 ../data/origswb/jstest_32.pcm \
- ../Transcoding"$PLATFORM"/APITest_refTrans38.pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
-
- echo "-- 45 kbps trans-coding to 38 kbps, 30 ms, SUPER-WIDEBAND"
- echo
- $ISAC -I -B 45000 -FL 30 -FS 32 -T 38000 ../Transcoding"$PLATFORM"/APITest_45T38.pcm \
- ../data/origswb/jstest_32.pcm ../Transcoding"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
-
- echo
- echo
- echo "-- 20 kbps, 30 ms, SUPER-WIDEBAND"
- echo
- $ISAC -I -B 20000 -FL 30 -FS 32 ../data/origswb/jstest_32.pcm \
- ../Transcoding"$PLATFORM"/APITest_refTrans20SWB.pcm >> $LOGFILE_API
- let "idx=idx+1"
-
- echo
- echo
-
- echo "-- 45 kbps trans-coding to 20 kbps, 30 ms, SUPER-WIDEBAND"
- echo
- $ISAC -I -B 45000 -FL 30 -FS 32 -T 20000 ../Transcoding"$PLATFORM"/APITest_45T20.pcm \
- ../data/origswb/jstest_32.pcm ../Transcoding"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo "________________________________________________________"
- echo
- echo
- echo "-- 50 kbps, 30 ms, SUPER-WIDEBAND"
- echo
- $ISAC -I -B 50000 -FL 30 -FS 32 ../data/origswb/jstest_32.pcm \
- ../Transcoding"$PLATFORM"/APITest_refTrans50.pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
-
- echo "-- 56 kbps trans-coding to 50 kbps, 30 ms, SUPER-WIDEBAND"
- echo
- $ISAC -I -B 56000 -FL 30 -FS 32 -T 50000 ../Transcoding"$PLATFORM"/APITest_56T50.pcm \
- ../data/origswb/jstest_32.pcm ../Transcoding"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
-
- echo "-- 56 kbps trans-coding to 20 kbps, 30 ms, SUPER-WIDEBAND"
- echo
- $ISAC -I -B 56000 -FL 30 -FS 32 -T 20000 ../Transcoding"$PLATFORM"/APITest_56T20.pcm \
- ../data/origswb/jstest_32.pcm ../Transcoding"$PLATFORM"/APITest_"$idx".pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo "________________________________________________________"
- echo
- echo
- echo
- echo
- echo
-
- echo " Test FEC"
- echo "========================================================================================"
- mkdir -p ../FEC"$PLATFORM"
- rm -f ../FEC"$PLATFORM"/*
- echo
- echo
- echo "-- 32 kbps with transcoding to 20kbps, 30 ms, WIDEBAND, 10% packet loss"
- $ISAC -I -B 32000 -FL 30 -FS 16 -PL 10 -T 20000 ../FEC"$PLATFORM"/APITest_PL10_WB30_T20.pcm \
- ../data/orig/speech_and_misc_WB.pcm ../FEC"$PLATFORM"/APITest_PL10_WB30.pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
-
- echo "-- 32 kbps, 60 ms, WIDEBAND, 10% packet loss"
- $ISAC -I -B 32000 -FL 60 -FS 16 -PL 10 ../data/orig/speech_and_misc_WB.pcm \
- ../FEC"$PLATFORM"/APITest_PL10_WB60.pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
-
- echo "-- 32 kbps with transcoding to 20 kbps, 30 ms, SUPER-WIDEBAND, 10% packet loss"
- $ISAC -I -B 32000 -FL 30 -FS 32 -PL 10 -T 20000 ../FEC"$PLATFORM"/APITest_PL10_SWB_8kHz_T20.pcm \
- ../data/origswb/jstest_32.pcm ../FEC"$PLATFORM"/APITest_PL10_SWB_8kHz.pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
-
- echo "-- 45 kbps with Trascoding to 38 kbps, 30 ms, SUPER-WIDEBAND, 10% packet loss"
- $ISAC -I -B 45000 -FL 30 -FS 32 -PL 10 -T 38000 ../FEC"$PLATFORM"/APITest_PL10_SWB_12kHz_T38.pcm \
- ../data/origswb/jstest_32.pcm ../FEC"$PLATFORM"/APITest_PL10_SWB_12kHz.pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
-
- echo "-- 56 kbps with transcoding to 50 kbps, 30 ms, SUPER-WIDEBAND, 10% packet loss"
- $ISAC -I -B 56000 -FL 30 -FS 32 -PL 10 -T 50000 ../FEC"$PLATFORM"/APITest_PL10_SWB_16kHz_T50.pcm \
- ../data/origswb/jstest_32.pcm ../FEC"$PLATFORM"/APITest_PL10_SWB_16kHz.pcm >> $LOGFILE_API
- let "idx=idx+1"
- echo
- echo
-fi
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACfault.txt b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACfault.txt
deleted file mode 100644
index 63829a4..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACfault.txt
+++ /dev/null
@@ -1,80 +0,0 @@
-#!/bin/bash
-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character????? ??????
-if [ "$1" = "x64" ] || [ "$#" -eq 0 ]
- then
- PLATFORM=_X64
- ISAC=../x64/Release/ReleaseTest-API_2005.exe
-elif [ "$1" = "2005" ]
- then
- PLATFORM=_2005
- ISAC=../win32/Release/ReleaseTest-API_2005.exe
-elif [ "$1" == "LINUX" ]
- then
- PLATFORM=_linux
- ISAC=../ReleaseTest-API/isacswtest
-else
- echo Unknown Platform
- exit 2
-fi
-
-LOGFILE=logfault$PLATFORM.txt
-echo "START FAULT TEST" > $LOGFILE
-
-
-INFILES=$(cat InputFiles.txt)
-SUBSET=$(cat InputFilesFew.txt)
-CHANNELFILES=$(cat ChannelFiles.txt)
-CHANNELLIST=($(cat ChannelFiles.txt))
-INDIR=../data/orig
-INDIRSWB=../data/origswb
-OUTDIR=../dataqaft$PLATFORM
-mkdir -p $OUTDIR
-
-#maximum Target rate for different bandwidth
-TARGETRATE=( 32000 32000 44000 56000 )
-SAMPFREQ=( 16 32 32 32 )
-FAULTTEST=(1 2 3 4 5 6 7 9)
-
-index1=0
-
-file_wb=../data/orig/16kHz.pcm
-file_swb=../data/origswb/32kHz.pcm
-
-for idx in 0 1 2 3
- do
-# Fault test
- echo
- echo "Sampling Frequency " ${SAMPFREQ[idx]} "kHz, Rate " ${TARGETRATE[idx]} "bps."
- echo "---------------------------------------------------"
- if [ ${SAMPFREQ[idx]} -eq 16 ]; then
- file=$file_wb
- else
- file=$file_swb
- fi
-
- for testnr in ${FAULTTEST[*]}
- do
- echo "Running Fault Test " $testnr
- $ISAC -I -B "${TARGETRATE[idx]}" -F $testnr -FS "${SAMPFREQ[idx]}" "$file" \
- $OUTDIR/ft"$testnr"_"${TARGETRATE[idx]}"_"${SAMPFREQ[idx]}".pcm >> LOGFILE
- echo
-
- done
-
-# Fault test number 10, error in bitstream
- echo "Running Fault Test 10"
- $ISAC -I -B "${TARGETRATE[idx]}" -F 10 -FS "${SAMPFREQ[idx]}" "$file" \
- $OUTDIR/ft10_"${TARGETRATE[idx]}"_"${SAMPFREQ[idx]}".pcm >> LOGFILE
- echo
- echo "Running Fault Test 10 with packetloss"
- $ISAC -I -B "${TARGETRATE[idx]}" -F 10 -PL 10 -FS "${SAMPFREQ[idx]}" "$file" \
- $OUTDIR/ft10plc_"${TARGETRATE[idx]}"_"${SAMPFREQ[idx]}".pcm >> LOGFILE
- echo
-done
-
-echo
-echo
-echo DONE!
-
-
-
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACfixfloat.txt b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACfixfloat.txt
deleted file mode 100644
index 4cda78e..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/QA/runiSACfixfloat.txt
+++ /dev/null
@@ -1,47 +0,0 @@
-#!/bin/bash
-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
-
-LOGFILE=logfxfl.txt
-echo "START FIX-FLOAT TEST" > $LOGFILE
-
-
-ISACFIXFLOAT=../../../fix/test/testFixFloat.exe
-
-INFILES=$(cat InputFiles.txt)
-SUBSET=$(cat InputFilesFew.txt)
-CHANNELFILES=$(cat ChannelFiles.txt)
-CHANNELLIST=($(cat ChannelFiles.txt))
-INDIR=../data/orig
-OUTDIR=../dataqafxfl
-mkdir -p $OUTDIR
-
-index1=0
-
-for file in $INFILES # loop over all input files
- do
-
- for channel in $CHANNELFILES
- do
- let "index1=index1+1"
-
- $ISACFIXFLOAT $INDIR/$channel -m 1 -PLC $INDIR/"$file" $OUTDIR/flfx$index1"$file" >> $LOGFILE
- $ISACFIXFLOAT $INDIR/$channel -m 2 -PLC $INDIR/"$file" $OUTDIR/fxfl$index1"$file" >> $LOGFILE
- done
-
-done
-
-index1=0
-
-for file in $SUBSET # loop over the subset of input files
- do
- let "index1=index1+1"
- $ISACFIXFLOAT $INDIR/$channel -m 1 -NB 1 $INDIR/"$file" $OUTDIR/flfxnb1_$index1"$file" >> $LOGFILE
- $ISACFIXFLOAT $INDIR/$channel -m 2 -NB 1 $INDIR/"$file" $OUTDIR/fxflnb1_$index1"$file" >> $LOGFILE
- $ISACFIXFLOAT $INDIR/$channel -m 1 -NB 2 -PLC $INDIR/"$file" $OUTDIR/flfxnb2_$index1"$file" >> $LOGFILE
- $ISACFIXFLOAT $INDIR/$channel -m 2 -NB 2 -PLC $INDIR/"$file" $OUTDIR/fxflnb2_$index1"$file" >> $LOGFILE
-done
-
-echo DONE!
-
-
-
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/ReleaseTest-API/ReleaseTest-API.cc b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/ReleaseTest-API/ReleaseTest-API.cc
deleted file mode 100644
index 19cff9e..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/ReleaseTest-API/ReleaseTest-API.cc
+++ /dev/null
@@ -1,1050 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// ReleaseTest-API.cpp : Defines the entry point for the console application.
-//
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <time.h>
-#include <ctype.h>
-#include <iostream>
-
-/* include API */
-#include "isac.h"
-#include "utility.h"
-
-/* Defines */
-#define SEED_FILE "randseed.txt" /* Used when running decoder on garbage data */
-#define MAX_FRAMESAMPLES 960 /* max number of samples per frame
- (= 60 ms frame & 16 kHz) or
- (= 30 ms frame & 32 kHz) */
-#define FRAMESAMPLES_10ms 160 /* number of samples per 10ms frame */
-#define SWBFRAMESAMPLES_10ms 320
-//#define FS 16000 /* sampling frequency (Hz) */
-
-#ifdef WIN32
-#define CLOCKS_PER_SEC 1000 /* Runtime statistics */
-#endif
-
-
-
-
-using namespace std;
-
-int main(int argc, char* argv[])
-{
-
- char inname[100], outname[100], bottleneck_file[100], vadfile[100];
- FILE *inp, *outp, *f_bn=NULL, *vadp, *bandwidthp;
- int framecnt, endfile;
-
- int i, errtype, VADusage = 0, packetLossPercent = 0;
- WebRtc_Word16 CodingMode;
- WebRtc_Word32 bottleneck;
- WebRtc_Word16 framesize = 30; /* ms */
- int cur_framesmpls, err;
-
- /* Runtime statistics */
- double starttime, runtime, length_file;
-
- WebRtc_Word16 stream_len = 0;
- WebRtc_Word16 declen, lostFrame = 0, declenTC = 0;
-
- WebRtc_Word16 shortdata[SWBFRAMESAMPLES_10ms];
- WebRtc_Word16 vaddata[SWBFRAMESAMPLES_10ms*3];
- WebRtc_Word16 decoded[MAX_FRAMESAMPLES << 1];
- WebRtc_Word16 decodedTC[MAX_FRAMESAMPLES << 1];
- WebRtc_UWord16 streamdata[500];
- WebRtc_Word16 speechType[1];
- WebRtc_Word16 rateBPS = 0;
- WebRtc_Word16 fixedFL = 0;
- WebRtc_Word16 payloadSize = 0;
- WebRtc_Word32 payloadRate = 0;
- int setControlBWE = 0;
- short FL, testNum;
- char version_number[20];
- FILE *plFile;
- WebRtc_Word32 sendBN;
-
-#ifdef _DEBUG
- FILE *fy;
- double kbps;
-#endif /* _DEBUG */
- int totalbits =0;
- int totalsmpls =0;
-
- /* If use GNS file */
- FILE *fp_gns = NULL;
- char gns_file[100];
- short maxStreamLen30 = 0;
- short maxStreamLen60 = 0;
- short sampFreqKHz = 32;
- short samplesIn10Ms;
- short useAssign = 0;
- //FILE logFile;
- bool doTransCoding = false;
- WebRtc_Word32 rateTransCoding = 0;
- WebRtc_UWord16 streamDataTransCoding[600];
- WebRtc_Word16 streamLenTransCoding = 0;
- FILE* transCodingFile = NULL;
- FILE* transcodingBitstream = NULL;
- WebRtc_UWord32 numTransCodingBytes = 0;
-
- /* only one structure used for ISAC encoder */
- ISACStruct* ISAC_main_inst;
- ISACStruct* decoderTransCoding;
-
- BottleNeckModel BN_data;
-
-#ifdef _DEBUG
- fy = fopen("bit_rate.dat", "w");
- fclose(fy);
- fy = fopen("bytes_frames.dat", "w");
- fclose(fy);
-#endif /* _DEBUG */
-
- /* Handling wrong input arguments in the command line */
- if((argc<3) || (argc>17)) {
- printf("\n\nWrong number of arguments or flag values.\n\n");
-
- printf("\n");
- WebRtcIsac_version(version_number);
- printf("iSAC-swb version %s \n\n", version_number);
-
- printf("Usage:\n\n");
- printf("./kenny.exe [-I] bottleneck_value infile outfile \n\n");
- printf("with:\n");
- printf("[-FS num] : sampling frequency in kHz, valid values are 16 & 32,\n");
- printf(" with 16 as default.\n");
- printf("[-I] : if -I option is specified, the coder will use\n");
- printf(" an instantaneous Bottleneck value. If not, it\n");
- printf(" will be an adaptive Bottleneck value.\n\n");
- printf("[-assign] : Use Assign API.\n");
- printf("[-B num] : the value of the bottleneck provided either\n");
- printf(" as a fixed value in bits/sec (e.g. 25000) or\n");
- printf(" read from a file (e.g. bottleneck.txt)\n\n");
- printf("[-INITRATE num] : Set a new value for initial rate. Note! Only used in \n");
- printf(" adaptive mode.\n\n");
- printf("[-FL num] : Set (initial) frame length in msec. Valid length are \n");
- printf(" 30 and 60 msec.\n\n");
- printf("[-FIXED_FL] : Frame length will be fixed to initial value.\n\n");
- printf("[-MAX num] : Set the limit for the payload size of iSAC in bytes. \n");
- printf(" Minimum 100 maximum 400.\n\n");
- printf("[-MAXRATE num] : Set the maxrate for iSAC in bits per second. \n");
- printf(" Minimum 32000, maximum 53400.\n\n");
- printf("[-F num] : if -F option is specified, the test function\n");
- printf(" will run the iSAC API fault scenario specified by the\n");
- printf(" supplied number.\n");
- printf(" F 1 - Call encoder prior to init encoder call\n");
- printf(" F 2 - Call decoder prior to init decoder call\n");
- printf(" F 3 - Call decoder prior to encoder call\n");
- printf(" F 4 - Call decoder with a too short coded sequence\n");
- printf(" F 5 - Call decoder with a too long coded sequence\n");
- printf(" F 6 - Call decoder with random bit stream\n");
- printf(" F 7 - Call init encoder/decoder at random during a call\n");
- printf(" F 8 - Call encoder/decoder without having allocated memory \n");
- printf(" for encoder/decoder instance\n");
- printf(" F 9 - Call decodeB without calling decodeA\n");
- printf(" F 10 - Call decodeB with garbage data\n");
- printf("[-PL num] : if -PL option is specified \n");
- printf("[-T rate file] : test trans-coding with target bottleneck 'rate' bits/sec\n");
- printf(" the output file is written to 'file'\n");
- printf("[-LOOP num] : number of times to repeat coding the input file for stress testing\n");
- //printf("[-CE num] : Test of APIs used by Conference Engine.\n");
- //printf(" CE 1 - getNewBitstream, getBWE \n");
- //printf(" (CE 2 - RESERVED for transcoding)\n");
- //printf(" CE 3 - getSendBWE, setSendBWE. \n\n");
- //printf("-L filename : write the logging info into file (appending)\n");
- printf("infile : Normal speech input file\n\n");
- printf("outfile : Speech output file\n\n");
- exit(0);
- }
-
- /* Print version number */
- printf("-------------------------------------------------\n");
- WebRtcIsac_version(version_number);
- printf("iSAC version %s \n\n", version_number);
-
- /* Loop over all command line arguments */
- CodingMode = 0;
- testNum = 0;
- useAssign = 0;
- //logFile = NULL;
- char transCodingFileName[500];
- WebRtc_Word16 totFileLoop = 0;
- WebRtc_Word16 numFileLoop = 0;
- for (i = 1; i < argc-2;i++)
- {
- if(!strcmp("-LOOP", argv[i]))
- {
- i++;
- totFileLoop = (WebRtc_Word16)atol(argv[i]);
- if(totFileLoop <= 0)
- {
- fprintf(stderr, "Invalid number of runs for the given input file, %d.", totFileLoop);
- exit(0);
- }
- }
-
- if(!strcmp("-T", argv[i]))
- {
- doTransCoding = true;
- i++;
- rateTransCoding = atoi(argv[i]);
- i++;
- strcpy(transCodingFileName, argv[i]);
- }
-
- /*Should we use assign API*/
- if(!strcmp("-assign", argv[i]))
- {
- useAssign = 1;
- }
-
- /* Set Sampling Rate */
- if(!strcmp("-FS", argv[i]))
- {
- i++;
- sampFreqKHz = atoi(argv[i]);
- }
-
- /* Instantaneous mode */
- if(!strcmp ("-I", argv[i]))
- {
- printf("Instantaneous BottleNeck\n");
- CodingMode = 1;
- }
-
- /* Set (initial) bottleneck value */
- if(!strcmp ("-INITRATE", argv[i])) {
- rateBPS = atoi(argv[i + 1]);
- setControlBWE = 1;
- if((rateBPS < 10000) || (rateBPS > 32000))
- {
- printf("\n%d is not a initial rate. Valid values are in the range 10000 to 32000.\n", rateBPS);
- exit(0);
- }
- printf("New initial rate: %d\n", rateBPS);
- i++;
- }
-
- /* Set (initial) framelength */
- if(!strcmp ("-FL", argv[i])) {
- framesize = atoi(argv[i + 1]);
- if((framesize != 30) && (framesize != 60))
- {
- printf("\n%d is not a valid frame length. Valid length are 30 and 60 msec.\n", framesize);
- exit(0);
- }
- setControlBWE = 1;
- printf("Frame Length: %d\n", framesize);
- i++;
- }
-
- /* Fixed frame length */
- if(!strcmp ("-FIXED_FL", argv[i]))
- {
- fixedFL = 1;
- setControlBWE = 1;
- printf("Fixed Frame Length\n");
- }
-
- /* Set maximum allowed payload size in bytes */
- if(!strcmp ("-MAX", argv[i])) {
- payloadSize = atoi(argv[i + 1]);
- printf("Maximum Payload Size: %d\n", payloadSize);
- i++;
- }
-
- /* Set maximum rate in bytes */
- if(!strcmp ("-MAXRATE", argv[i])) {
- payloadRate = atoi(argv[i + 1]);
- printf("Maximum Rate in kbps: %d\n", payloadRate);
- i++;
- }
-
- /* Test of fault scenarious */
- if(!strcmp ("-F", argv[i]))
- {
- testNum = atoi(argv[i + 1]);
- printf("Fault test: %d\n", testNum);
- if(testNum < 1 || testNum > 10)
- {
- printf("\n%d is not a valid Fault Scenario number. Valid Fault Scenarios are numbered 1-10.\n", testNum);
- exit(0);
- }
- i++;
- }
-
- /* Packet loss test */
- if(!strcmp ("-PL", argv[i]))
- {
- if( isdigit( *argv[i+1] ) )
- {
- packetLossPercent = atoi( argv[i+1] );
- if( (packetLossPercent < 0) | (packetLossPercent > 100) )
- {
- printf( "\nInvalid packet loss perentage \n" );
- exit( 0 );
- }
- if( packetLossPercent > 0 )
- {
- printf( "Simulating %d %% of independent packet loss\n", packetLossPercent );
- }
- else
- {
- printf( "\nNo Packet Loss Is Simulated \n" );
- }
- }
- else
- {
- plFile = fopen( argv[i+1], "rb" );
- if( plFile == NULL )
- {
- printf( "\n couldn't open the frameloss file: %s\n", argv[i+1] );
- exit( 0 );
- }
- printf( "Simulating packet loss through the given channel file: %s\n", argv[i+1] );
- }
- i++;
- }
-
- /* Random packetlosses */
- if(!strcmp ("-rnd", argv[i]))
- {
- srand((unsigned int)time(NULL) );
- printf( "Random pattern in lossed packets \n" );
- }
-
- /* Use gns file */
- if(!strcmp ("-G", argv[i]))
- {
- sscanf(argv[i + 1], "%s", gns_file);
- fp_gns = fopen(gns_file, "rb");
- if(fp_gns == NULL)
- {
- printf("Cannot read file %s.\n", gns_file);
- exit(0);
- }
- i++;
- }
-
-
- // make it with '-B'
- /* Get Bottleneck value */
- if(!strcmp("-B", argv[i]))
- {
- i++;
- bottleneck = atoi(argv[i]);
- if(bottleneck == 0)
- {
- sscanf(argv[i], "%s", bottleneck_file);
- f_bn = fopen(bottleneck_file, "rb");
- if(f_bn == NULL)
- {
- printf("Error No value provided for BottleNeck and cannot read file %s.\n", bottleneck_file);
- exit(0);
- }
- else
- {
- printf("reading bottleneck rates from file %s\n\n",bottleneck_file);
- if(fscanf(f_bn, "%d", &bottleneck) == EOF)
- {
- /* Set pointer to beginning of file */
- fseek(f_bn, 0L, SEEK_SET);
- if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
- exit(0);
- }
- }
-
- /* Bottleneck is a cosine function
- * Matlab code for writing the bottleneck file:
- * BottleNeck_10ms = 20e3 + 10e3 * cos((0:5999)/5999*2*pi);
- * fid = fopen('bottleneck.txt', 'wb');
- * fprintf(fid, '%d\n', BottleNeck_10ms); fclose(fid);
- */
- }
- }
- else
- {
- printf("\nfixed bottleneck rate of %d bits/s\n\n", bottleneck);
- }
- }
- /* Run Conference Engine APIs */
- // Do not test it in the first release
- //
- // if(!strcmp ("-CE", argv[i]))
- // {
- // testCE = atoi(argv[i + 1]);
- // if(testCE==1)
- // {
- // i++;
- // scale = (float)atof( argv[i+1] );
- // }
- // else if(testCE == 2)
- // {
- // printf("\nCE-test 2 (transcoding) not implemented.\n");
- // exit(0);
- // }
- // else if(testCE < 1 || testCE > 3)
- // {
- // printf("\n%d is not a valid CE-test number. Valid CE tests are 1-3.\n", testCE);
- // exit(0);
- // }
- // printf("CE-test number: %d\n", testCE);
- // i++;
- // }
- }
-
- if(CodingMode == 0)
- {
- printf("\nAdaptive BottleNeck\n");
- }
-
- switch(sampFreqKHz)
- {
- case 16:
- {
- printf("iSAC Wideband.\n");
- samplesIn10Ms = FRAMESAMPLES_10ms;
- break;
- }
- case 32:
- {
- printf("iSAC Supper-Wideband.\n");
- samplesIn10Ms = SWBFRAMESAMPLES_10ms;
- break;
- }
- default:
- printf("Unsupported sampling frequency %d kHz", sampFreqKHz);
- exit(0);
- }
-
-
-
-
- /* Get Input and Output files */
- sscanf(argv[argc-2], "%s", inname);
- sscanf(argv[argc-1], "%s", outname);
- printf("\nInput file: %s\n", inname);
- printf("Output file: %s\n\n", outname);
- if((inp = fopen(inname,"rb")) == NULL)
- {
- printf(" Error iSAC Cannot read file %s.\n", inname);
- cout << flush;
- exit(1);
- }
-
- if((outp = fopen(outname,"wb")) == NULL)
- {
- printf(" Error iSAC Cannot write file %s.\n", outname);
- cout << flush;
- getchar();
- exit(1);
- }
- if(VADusage)
- {
- if((vadp = fopen(vadfile,"rb")) == NULL)
- {
- printf(" Error iSAC Cannot read file %s.\n", vadfile);
- cout << flush;
- exit(1);
- }
- }
-
- if((bandwidthp = fopen("bwe.pcm","wb")) == NULL)
- {
- printf(" Error iSAC Cannot read file %s.\n", "bwe.pcm");
- cout << flush;
- exit(1);
- }
-
-
- starttime = clock()/(double)CLOCKS_PER_SEC; /* Runtime statistics */
-
- /* Initialize the ISAC and BN structs */
- if(testNum != 8)
- {
- if(!useAssign)
- {
- err =WebRtcIsac_Create(&ISAC_main_inst);
- WebRtcIsac_SetEncSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
- WebRtcIsac_SetDecSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
- }
- else
- {
- /* Test the Assign functions */
- int sss;
- void *ppp;
- err = WebRtcIsac_AssignSize(&sss);
- ppp = malloc(sss);
- err = WebRtcIsac_Assign(&ISAC_main_inst, ppp);
- WebRtcIsac_SetEncSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
- WebRtcIsac_SetDecSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
- }
- /* Error check */
- if(err < 0)
- {
- printf("\n\n Error in create.\n\n");
- cout << flush;
- exit(EXIT_FAILURE);
- }
- }
- BN_data.arrival_time = 0;
- BN_data.sample_count = 0;
- BN_data.rtp_number = 0;
-
- /* Initialize encoder and decoder */
- framecnt= 0;
- endfile = 0;
-
- if(doTransCoding)
- {
- WebRtcIsac_Create(&decoderTransCoding);
- WebRtcIsac_SetEncSampRate(decoderTransCoding, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
- WebRtcIsac_SetDecSampRate(decoderTransCoding, (sampFreqKHz == 16)? kIsacWideband:kIsacSuperWideband);
- WebRtcIsac_DecoderInit(decoderTransCoding);
- transCodingFile = fopen(transCodingFileName, "wb");
- if(transCodingFile == NULL)
- {
- printf("Could not open %s to output trans-coding.\n", transCodingFileName);
- exit(0);
- }
- strcat(transCodingFileName, ".bit");
- transcodingBitstream = fopen(transCodingFileName, "wb");
- if(transcodingBitstream == NULL)
- {
- printf("Could not open %s to write the bit-stream of transcoder.\n", transCodingFileName);
- exit(0);
- }
- }
-
- if(testNum != 1)
- {
- if(WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode) < 0)
- {
- printf("Error could not initialize the encoder \n");
- cout << flush;
- return 0;
- }
- }
- if(testNum != 2)
- {
- if(WebRtcIsac_DecoderInit(ISAC_main_inst) < 0)
- {
- printf("Error could not initialize the decoder \n");
- cout << flush;
- return 0;
- }
- }
- if(CodingMode == 1)
- {
- err = WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
- if(err < 0)
- {
- /* exit if returned with error */
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- printf("\n\n Error in initialization (control): %d.\n\n", errtype);
- cout << flush;
- if(testNum == 0)
- {
- exit(EXIT_FAILURE);
- }
- }
- }
-
- if((setControlBWE) && (CodingMode == 0))
- {
- err = WebRtcIsac_ControlBwe(ISAC_main_inst, rateBPS, framesize, fixedFL);
- if(err < 0)
- {
- /* exit if returned with error */
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
-
- printf("\n\n Error in Control BWE: %d.\n\n", errtype);
- cout << flush;
- exit(EXIT_FAILURE);
- }
- }
-
- if(payloadSize != 0)
- {
- err = WebRtcIsac_SetMaxPayloadSize(ISAC_main_inst, payloadSize);
- if(err < 0)
- {
- /* exit if returned with error */
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- printf("\n\n Error in SetMaxPayloadSize: %d.\n\n", errtype);
- cout << flush;
- exit(EXIT_FAILURE);
- }
- }
- if(payloadRate != 0)
- {
- err = WebRtcIsac_SetMaxRate(ISAC_main_inst, payloadRate);
- if(err < 0)
- {
- /* exit if returned with error */
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- printf("\n\n Error in SetMaxRateInBytes: %d.\n\n", errtype);
- cout << flush;
- exit(EXIT_FAILURE);
- }
- }
-
- *speechType = 1;
-
- cout << "\n" << flush;
-
- length_file = 0;
- WebRtc_Word16 bnIdxTC;
- WebRtc_Word16 jitterInfoTC;
- while (endfile == 0)
- {
- /* Call init functions at random, fault test number 7 */
- if(testNum == 7 && (rand()%2 == 0))
- {
- err = WebRtcIsac_EncoderInit(ISAC_main_inst, CodingMode);
- /* Error check */
- if(err < 0)
- {
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- printf("\n\n Error in encoderinit: %d.\n\n", errtype);
- cout << flush;
- }
-
- err = WebRtcIsac_DecoderInit(ISAC_main_inst);
- /* Error check */
- if(err < 0)
- {
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- printf("\n\n Error in decoderinit: %d.\n\n", errtype);
- cout << flush;
- }
- }
-
- cur_framesmpls = 0;
- while (1)
- {
- /* Read 10 ms speech block */
- endfile = readframe(shortdata, inp, samplesIn10Ms);
-
- if(endfile)
- {
- numFileLoop++;
- if(numFileLoop < totFileLoop)
- {
- rewind(inp);
- framecnt = 0;
- fprintf(stderr, "\n");
- endfile = readframe(shortdata, inp, samplesIn10Ms);
- }
- }
-
- if(testNum == 7)
- {
- srand((unsigned int)time(NULL));
- }
-
- /* iSAC encoding */
- if(!(testNum == 3 && framecnt == 0))
- {
- stream_len = WebRtcIsac_Encode(ISAC_main_inst,
- shortdata,
- (WebRtc_Word16*)streamdata);
- if((payloadSize != 0) && (stream_len > payloadSize))
- {
- if(testNum == 0)
- {
- printf("\n\n");
- }
-
- printf("\nError: Streamsize out of range %d\n", stream_len - payloadSize);
- cout << flush;
- }
-
- WebRtcIsac_GetUplinkBw(ISAC_main_inst, &sendBN);
-
- if(stream_len>0)
- {
- if(doTransCoding)
- {
- WebRtc_Word16 indexStream;
- WebRtc_UWord8 auxUW8;
-
- /************************* Main Transcoding stream *******************************/
- WebRtcIsac_GetDownLinkBwIndex(ISAC_main_inst, &bnIdxTC, &jitterInfoTC);
- streamLenTransCoding = WebRtcIsac_GetNewBitStream(
- ISAC_main_inst, bnIdxTC, jitterInfoTC, rateTransCoding,
- (WebRtc_Word16*)streamDataTransCoding, false);
- if(streamLenTransCoding < 0)
- {
- fprintf(stderr, "Error in trans-coding\n");
- exit(0);
- }
- auxUW8 = (WebRtc_UWord8)(((streamLenTransCoding & 0xFF00) >> 8) & 0x00FF);
- fwrite(&auxUW8, sizeof(WebRtc_UWord8),
- 1, transcodingBitstream);
-
- auxUW8 = (WebRtc_UWord8)(streamLenTransCoding & 0x00FF);
- fwrite(&auxUW8, sizeof(WebRtc_UWord8),
- 1, transcodingBitstream);
-
- fwrite((WebRtc_UWord8*)streamDataTransCoding, sizeof(WebRtc_UWord8),
- streamLenTransCoding, transcodingBitstream);
-
- WebRtcIsac_ReadBwIndex((WebRtc_Word16*)streamDataTransCoding, &indexStream);
- if(indexStream != bnIdxTC)
- {
- fprintf(stderr, "Error in inserting Bandwidth index into transcoding stream.\n");
- exit(0);
- }
- numTransCodingBytes += streamLenTransCoding;
- }
- }
- }
- else
- {
- break;
- }
-
- if(stream_len < 0)
- {
- /* exit if returned with error */
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- printf("\n\nError in encoder: %d.\n\n", errtype);
- cout << flush;
- }
- cur_framesmpls += samplesIn10Ms;
- /* exit encoder loop if the encoder returned a bitstream */
- if(stream_len != 0) break;
- }
-
- /* read next bottleneck rate */
- if(f_bn != NULL)
- {
- if(fscanf(f_bn, "%d", &bottleneck) == EOF)
- {
- /* Set pointer to beginning of file */
- fseek(f_bn, 0L, SEEK_SET);
- if (fscanf(f_bn, "%d", &bottleneck) == EOF) {
- exit(0);
- }
- }
- if(CodingMode == 1)
- {
- WebRtcIsac_Control(ISAC_main_inst, bottleneck, framesize);
- }
- }
-
- length_file += cur_framesmpls;
- if(cur_framesmpls == (3 * samplesIn10Ms))
- {
- maxStreamLen30 = (stream_len > maxStreamLen30)? stream_len:maxStreamLen30;
- }
- else
- {
- maxStreamLen60 = (stream_len > maxStreamLen60)? stream_len:maxStreamLen60;
- }
-
- if(!lostFrame)
- {
- lostFrame = ((rand()%100) < packetLossPercent);
- }
- else
- {
- lostFrame = 0;
- }
-
- // RED.
- if(lostFrame)
- {
- stream_len = WebRtcIsac_GetRedPayload(ISAC_main_inst,
- (WebRtc_Word16*)streamdata);
-
- if(doTransCoding)
- {
- streamLenTransCoding = WebRtcIsac_GetNewBitStream(
- ISAC_main_inst, bnIdxTC, jitterInfoTC, rateTransCoding,
- (WebRtc_Word16*)streamDataTransCoding, true);
- if(streamLenTransCoding < 0)
- {
- fprintf(stderr, "Error in RED trans-coding\n");
- exit(0);
- }
- }
- }
-
- /* make coded sequence to short be inreasing */
- /* the length the decoder expects */
- if(testNum == 4)
- {
- stream_len += 10;
- }
-
- /* make coded sequence to long be decreasing */
- /* the length the decoder expects */
- if(testNum == 5)
- {
- stream_len -= 10;
- }
-
- if(testNum == 6)
- {
- srand((unsigned int)time(NULL));
- for(i = 0; i < stream_len; i++)
- {
- streamdata[i] = rand();
- }
- }
-
- if(VADusage){
- readframe(vaddata, vadp, samplesIn10Ms*3);
- }
-
- /* simulate packet handling through NetEq and the modem */
- if(!(testNum == 3 && framecnt == 0))
- {
- get_arrival_time(cur_framesmpls, stream_len, bottleneck, &BN_data,
- sampFreqKHz*1000, sampFreqKHz*1000);
- }
-
- if(VADusage && (framecnt>10 && vaddata[0]==0))
- {
- BN_data.rtp_number--;
- }
- else
- {
- /* Error test number 10, garbage data */
- if(testNum == 10)
- {
- /* Test to run decoder with garbage data */
- for(i = 0; i < stream_len; i++)
- {
- streamdata[i] = (short) (streamdata[i]) + (short) rand();
- }
- }
-
- if(testNum != 9)
- {
- err = WebRtcIsac_UpdateBwEstimate(ISAC_main_inst, streamdata,
- stream_len, BN_data.rtp_number, BN_data.sample_count,
- BN_data.arrival_time);
-
- if(err < 0)
- {
- /* exit if returned with error */
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- if(testNum == 0)
- {
- printf("\n\n");
- }
-
- printf("Error: in decoder: %d.", errtype);
- cout << flush;
- if(testNum == 0)
- {
- printf("\n\n");
- }
-
- }
- }
-
- /* Call getFramelen, only used here for function test */
- err = WebRtcIsac_ReadFrameLen(ISAC_main_inst,
- (WebRtc_Word16*)streamdata, &FL);
- if(err < 0)
- {
- /* exit if returned with error */
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- if(testNum == 0)
- {
- printf("\n\n");
- }
- printf(" Error: in getFrameLen %d.", errtype);
- cout << flush;
- if(testNum == 0)
- {
- printf("\n\n");
- }
- }
-
- // iSAC decoding
-
- if(lostFrame)
- {
- declen = WebRtcIsac_DecodeRcu(ISAC_main_inst, streamdata,
- stream_len, decoded, speechType);
-
- if(doTransCoding)
- {
- declenTC = WebRtcIsac_DecodeRcu(decoderTransCoding,
- streamDataTransCoding, streamLenTransCoding,
- decodedTC, speechType);
- }
- }
- else
- {
- declen = WebRtcIsac_Decode(ISAC_main_inst, streamdata,
- stream_len, decoded, speechType);
-
- if(doTransCoding)
- {
- declenTC = WebRtcIsac_Decode(decoderTransCoding,
- streamDataTransCoding, streamLenTransCoding,
- decodedTC, speechType);
- }
- }
-
- if(declen < 0)
- {
- /* exit if returned with error */
- errtype=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- if(testNum == 0)
- {
- printf("\n\n");
- }
- printf(" Error: in decoder %d.", errtype);
- cout << flush;
- if(testNum == 0)
- {
- printf("\n\n");
- }
- }
-
- if(declenTC < 0)
- {
- if(testNum == 0)
- {
- printf("\n\n");
- }
- printf(" Error: in decoding the transcoded stream");
- cout << flush;
- if(testNum == 0)
- {
- printf("\n\n");
- }
-
- }
- }
- /* Write decoded speech frame to file */
- if((declen > 0) && (numFileLoop == 0))
- {
- fwrite(decoded, sizeof(WebRtc_Word16), declen, outp);
- }
-
- if((declenTC > 0) && (numFileLoop == 0))
- {
- fwrite(decodedTC, sizeof(WebRtc_Word16), declen, transCodingFile);
- }
-
-
- fprintf(stderr, "\rframe = %5d ", framecnt);
- fflush(stderr);
- framecnt++;
-
- /* Error test number 10, garbage data */
- //if(testNum == 10)
- //{
- // /* Test to run decoder with garbage data */
- // if( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL )
- // {
- // fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
- // }
- // else
- // {
- // fprintf(seedfile, "ok\n\n");
- // fclose(seedfile);
- // }
- //}
- /* Error test number 10, garbage data */
- //if(testNum == 10)
- //{
- // /* Test to run decoder with garbage data */
- // for ( i = 0; i < stream_len; i++)
- // {
- // streamdata[i] = (short) (streamdata[i] + (short) rand());
- // }
- //}
-
-
- totalsmpls += declen;
- totalbits += 8 * stream_len;
-#ifdef _DEBUG
- kbps = ((double) sampFreqKHz * 1000.) / ((double) cur_framesmpls) * 8.0 * stream_len / 1000.0;// kbits/s
- fy = fopen("bit_rate.dat", "a");
- fprintf(fy, "Frame %i = %0.14f\n", framecnt, kbps);
- fclose(fy);
-
-#endif /* _DEBUG */
-
- }
- printf("\n");
- printf("total bits = %d bits\n", totalbits);
- printf("measured average bitrate = %0.3f kbits/s\n",
- (double)totalbits *(sampFreqKHz) / totalsmpls);
- if(doTransCoding)
- {
- printf("Transcoding average bit-rate = %0.3f kbps\n",
- (double)numTransCodingBytes * 8.0 *(sampFreqKHz) / totalsmpls);
- fclose(transCodingFile);
- }
- printf("\n");
-
- /* Runtime statistics */
- runtime = (double)(clock()/(double)CLOCKS_PER_SEC-starttime);
- length_file = length_file /(sampFreqKHz * 1000.);
-
- printf("\n\nLength of speech file: %.1f s\n", length_file);
- printf("Time to run iSAC: %.2f s (%.2f %% of realtime)\n\n", runtime, (100*runtime/length_file));
-
- if(maxStreamLen30 != 0)
- {
- printf("Maximum payload size 30ms Frames %d bytes (%0.3f kbps)\n",
- maxStreamLen30,
- maxStreamLen30 * 8 / 30.);
- }
- if(maxStreamLen60 != 0)
- {
- printf("Maximum payload size 60ms Frames %d bytes (%0.3f kbps)\n",
- maxStreamLen60,
- maxStreamLen60 * 8 / 60.);
- }
- //fprintf(stderr, "\n");
-
- fprintf(stderr, " %.1f s", length_file);
- fprintf(stderr, " %0.1f kbps", (double)totalbits *(sampFreqKHz) / totalsmpls);
- if(maxStreamLen30 != 0)
- {
- fprintf(stderr, " plmax-30ms %d bytes (%0.0f kbps)",
- maxStreamLen30,
- maxStreamLen30 * 8 / 30.);
- }
- if(maxStreamLen60 != 0)
- {
- fprintf(stderr, " plmax-60ms %d bytes (%0.0f kbps)",
- maxStreamLen60,
- maxStreamLen60 * 8 / 60.);
- }
- if(doTransCoding)
- {
- fprintf(stderr, " transcoding rate %.0f kbps",
- (double)numTransCodingBytes * 8.0 *(sampFreqKHz) / totalsmpls);
- }
-
- fclose(inp);
- fclose(outp);
- WebRtcIsac_Free(ISAC_main_inst);
-
-
- exit(0);
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/SwitchingSampRate/SwitchingSampRate.cc b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/SwitchingSampRate/SwitchingSampRate.cc
deleted file mode 100644
index e0eed12..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/SwitchingSampRate/SwitchingSampRate.cc
+++ /dev/null
@@ -1,443 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// SwitchingSampRate.cpp : Defines the entry point for the console
-// application.
-//
-
-#include <iostream>
-#include "isac.h"
-#include "utility.h"
-#include "signal_processing_library.h"
-
-#define MAX_FILE_NAME 500
-#define MAX_NUM_CLIENTS 2
-
-
-#define NUM_CLIENTS 2
-
-using namespace std;
-
-int main(int argc, char* argv[])
-{
- char fileNameWB[MAX_FILE_NAME];
- char fileNameSWB[MAX_FILE_NAME];
-
- char outFileName[MAX_NUM_CLIENTS][MAX_FILE_NAME];
-
- FILE* inFile[MAX_NUM_CLIENTS];
- FILE* outFile[MAX_NUM_CLIENTS];
-
- ISACStruct* codecInstance[MAX_NUM_CLIENTS];
- WebRtc_Word32 resamplerState[MAX_NUM_CLIENTS][8];
-
- enum IsacSamplingRate encoderSampRate[MAX_NUM_CLIENTS];
-
- int minBn = 16000;
- int maxBn = 56000;
-
- int bnWB = 32000;
- int bnSWB = 56000;
-
- strcpy(outFileName[0], "switchSampRate_out1.pcm");
- strcpy(outFileName[1], "switchSampRate_out2.pcm");
-
- short clientCntr;
-
- unsigned int lenEncodedInBytes[MAX_NUM_CLIENTS];
- unsigned int lenAudioIn10ms[MAX_NUM_CLIENTS];
- unsigned int lenEncodedInBytesTmp[MAX_NUM_CLIENTS];
- unsigned int lenAudioIn10msTmp[MAX_NUM_CLIENTS];
- BottleNeckModel* packetData[MAX_NUM_CLIENTS];
-
- char versionNumber[100];
- short samplesIn10ms[MAX_NUM_CLIENTS];
- int bottleneck[MAX_NUM_CLIENTS];
-
- printf("\n\n");
- printf("____________________________________________\n\n");
- WebRtcIsac_version(versionNumber);
- printf(" iSAC-swb version %s\n", versionNumber);
- printf("____________________________________________\n");
-
-
- fileNameWB[0] = '\0';
- fileNameSWB[0] = '\0';
-
- char myFlag[20];
- strcpy(myFlag, "-wb");
- // READ THE WIDEBAND AND SUPER-WIDEBAND FILE NAMES
- if(readParamString(argc, argv, myFlag, fileNameWB, MAX_FILE_NAME) <= 0)
- {
- printf("No wideband file is specified");
- }
-
- strcpy(myFlag, "-swb");
- if(readParamString(argc, argv, myFlag, fileNameSWB, MAX_FILE_NAME) <= 0)
- {
- printf("No super-wideband file is specified");
- }
-
- // THE FIRST CLIENT STARTS IN WIDEBAND
- encoderSampRate[0] = kIsacWideband;
- OPEN_FILE_RB(inFile[0], fileNameWB);
-
- // THE SECOND CLIENT STARTS IN SUPER-WIDEBAND
- encoderSampRate[1] = kIsacSuperWideband;
- OPEN_FILE_RB(inFile[1], fileNameSWB);
-
- strcpy(myFlag, "-I");
- short codingMode = readSwitch(argc, argv, myFlag);
-
- for(clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++)
- {
- codecInstance[clientCntr] = NULL;
-
- printf("\n");
- printf("Client %d\n", clientCntr + 1);
- printf("---------\n");
- printf("Starting %s",
- (encoderSampRate[clientCntr] == kIsacWideband)
- ? "wideband":"super-wideband");
-
- // Open output File Name
- OPEN_FILE_WB(outFile[clientCntr], outFileName[clientCntr]);
- printf("Output File...................... %s\n", outFileName[clientCntr]);
-
- samplesIn10ms[clientCntr] = encoderSampRate[clientCntr] * 10;
-
- if(codingMode == 1)
- {
- bottleneck[clientCntr] = (clientCntr)? bnSWB:bnWB;
- }
- else
- {
- bottleneck[clientCntr] = (clientCntr)? minBn:maxBn;
- }
-
- printf("Bottleneck....................... %0.3f kbits/sec \n",
- bottleneck[clientCntr] / 1000.0);
-
- // coding-mode
- printf("Encoding Mode.................... %s\n",
- (codingMode == 1)? "Channel-Independent (Instantaneous)":"Adaptive");
-
- lenEncodedInBytes[clientCntr] = 0;
- lenAudioIn10ms[clientCntr] = 0;
- lenEncodedInBytesTmp[clientCntr] = 0;
- lenAudioIn10msTmp[clientCntr] = 0;
-
- packetData[clientCntr] = (BottleNeckModel*)new(BottleNeckModel);
- if(packetData[clientCntr] == NULL)
- {
- printf("Could not allocate memory for packetData \n");
- return -1;
- }
- memset(packetData[clientCntr], 0, sizeof(BottleNeckModel));
- memset(resamplerState[clientCntr], 0, sizeof(WebRtc_Word32) * 8);
- }
-
- for(clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++)
- {
- // Create
- if(WebRtcIsac_Create(&codecInstance[clientCntr]))
- {
- printf("Could not creat client %d\n", clientCntr + 1);
- return -1;
- }
-
- WebRtcIsac_SetEncSampRate(codecInstance[clientCntr], encoderSampRate[clientCntr]);
-
- WebRtcIsac_SetDecSampRate(codecInstance[clientCntr],
- encoderSampRate[clientCntr + (1 - ((clientCntr & 1)<<1))]);
-
- // Initialize Encoder
- if(WebRtcIsac_EncoderInit(codecInstance[clientCntr],
- codingMode) < 0)
- {
- printf("Could not initialize client, %d\n", clientCntr + 1);
- return -1;
- }
-
- // Initialize Decoder
- if(WebRtcIsac_DecoderInit(codecInstance[clientCntr]) < 0)
- {
- printf("Could not initialize decoder of client %d\n",
- clientCntr + 1);
- return -1;
- }
-
- // setup Rate if in Instantaneous mode
- if(codingMode != 0)
- {
- // ONLY Clients who are not in Adaptive mode
- if(WebRtcIsac_Control(codecInstance[clientCntr],
- bottleneck[clientCntr], 30) < 0)
- {
- printf("Could not setup bottleneck and frame-size for client %d\n",
- clientCntr + 1);
- return -1;
- }
- }
- }
-
-
- short streamLen;
- short numSamplesRead;
- short lenDecodedAudio;
- short senderIdx;
- short receiverIdx;
-
- printf("\n");
- short num10ms[MAX_NUM_CLIENTS];
- memset(num10ms, 0, sizeof(short)*MAX_NUM_CLIENTS);
- FILE* arrivalTimeFile1 = fopen("arrivalTime1.dat", "wb");
- FILE* arrivalTimeFile2 = fopen("arrivalTime2.dat", "wb");
- short numPrint[MAX_NUM_CLIENTS];
- memset(numPrint, 0, sizeof(short) * MAX_NUM_CLIENTS);
-
- // Audio Buffers
- short silence10ms[10 * 32];
- memset(silence10ms, 0, 320 * sizeof(short));
- short audioBuff10ms[10 * 32];
- short audioBuff60ms[60 * 32];
- short resampledAudio60ms[60 * 32];
-
- unsigned short bitStream[600+600];
- short speechType[1];
-
- short numSampFreqChanged = 0;
- while(numSampFreqChanged < 10)
- {
- for(clientCntr = 0; clientCntr < NUM_CLIENTS; clientCntr++)
- {
- // Encoding/decoding for this pair of clients, if there is
- // audio for any of them
- //if(audioLeft[clientCntr] || audioLeft[clientCntr + 1])
- //{
- //for(pairCntr = 0; pairCntr < 2; pairCntr++)
- //{
- senderIdx = clientCntr; // + pairCntr;
- receiverIdx = 1 - clientCntr;// + (1 - pairCntr);
-
- //if(num10ms[senderIdx] > 6)
- //{
- // printf("Too many frames read for client %d",
- // senderIdx + 1);
- // return -1;
- //}
-
- numSamplesRead = (short)fread(audioBuff10ms, sizeof(short),
- samplesIn10ms[senderIdx], inFile[senderIdx]);
- if(numSamplesRead != samplesIn10ms[senderIdx])
- {
- // file finished switch encoder sampling frequency.
- printf("Changing Encoder Sampling frequency in client %d to ", senderIdx+1);
- fclose(inFile[senderIdx]);
- numSampFreqChanged++;
- if(encoderSampRate[senderIdx] == kIsacWideband)
- {
- printf("super-wideband.\n");
- OPEN_FILE_RB(inFile[senderIdx], fileNameSWB);
- encoderSampRate[senderIdx] = kIsacSuperWideband;
- }
- else
- {
- printf("wideband.\n");
- OPEN_FILE_RB(inFile[senderIdx], fileNameWB);
- encoderSampRate[senderIdx] = kIsacWideband;
- }
- WebRtcIsac_SetEncSampRate(codecInstance[senderIdx], encoderSampRate[senderIdx]);
- WebRtcIsac_SetDecSampRate(codecInstance[receiverIdx], encoderSampRate[senderIdx]);
-
- samplesIn10ms[clientCntr] = encoderSampRate[clientCntr] * 10;
-
- numSamplesRead = (short)fread(audioBuff10ms, sizeof(short),
- samplesIn10ms[senderIdx], inFile[senderIdx]);
- if(numSamplesRead != samplesIn10ms[senderIdx])
- {
- printf(" File %s for client %d has not enough audio\n",
- (encoderSampRate[senderIdx]==kIsacWideband)? "wideband":"super-wideband",
- senderIdx + 1);
- return -1;
- }
- }
- num10ms[senderIdx]++;
-
- // sanity check
- //if(num10ms[senderIdx] > 6)
- //{
- // printf("Client %d has got more than 60 ms audio and encoded no packet.\n",
- // senderIdx);
- // return -1;
- //}
-
- // Encode
-
-
- streamLen = WebRtcIsac_Encode(codecInstance[senderIdx],
- audioBuff10ms, (short*)bitStream);
- WebRtc_Word16 ggg;
- if (streamLen > 0) {
- if(( WebRtcIsac_ReadFrameLen(codecInstance[receiverIdx],
- (short *) bitStream, &ggg))<0)
- printf("ERROR\n");
- }
-
- // Sanity check
- if(streamLen < 0)
- {
- printf(" Encoder error in client %d \n", senderIdx + 1);
- return -1;
- }
-
-
- if(streamLen > 0)
- {
- // Packet generated; model sending through a channel, do bandwidth
- // estimation at the receiver and decode.
- lenEncodedInBytes[senderIdx] += streamLen;
- lenAudioIn10ms[senderIdx] += (unsigned int)num10ms[senderIdx];
- lenEncodedInBytesTmp[senderIdx] += streamLen;
- lenAudioIn10msTmp[senderIdx] += (unsigned int)num10ms[senderIdx];
-
- // Print after ~5 sec.
- if(lenAudioIn10msTmp[senderIdx] >= 100)
- {
- numPrint[senderIdx]++;
- printf(" %d, %6.3f => %6.3f ", senderIdx+1,
- bottleneck[senderIdx] / 1000.0,
- lenEncodedInBytesTmp[senderIdx] * 0.8 /
- lenAudioIn10msTmp[senderIdx]);
-
- if(codingMode == 0)
- {
- WebRtc_Word32 bn;
- WebRtcIsac_GetUplinkBw(codecInstance[senderIdx], &bn);
- printf("[%d] ", bn);
- }
- //WebRtc_Word16 rateIndexLB;
- //WebRtc_Word16 rateIndexUB;
- //WebRtcIsac_GetDownLinkBwIndex(codecInstance[receiverIdx],
- // &rateIndexLB, &rateIndexUB);
- //printf(" (%2d, %2d) ", rateIndexLB, rateIndexUB);
-
- cout << flush;
- lenEncodedInBytesTmp[senderIdx] = 0;
- lenAudioIn10msTmp[senderIdx] = 0;
- //if(senderIdx == (NUM_CLIENTS - 1))
- //{
- printf(" %0.1f \n", lenAudioIn10ms[senderIdx] * 10. /1000);
- //}
-
- // After ~20 sec change the bottleneck.
- // if((numPrint[senderIdx] == 4) && (codingMode == 0))
- // {
- // numPrint[senderIdx] = 0;
- // if(codingMode == 0)
- // {
- // int newBottleneck = bottleneck[senderIdx] +
- // (bottleneckChange[senderIdx] * 1000);
-
- // if(bottleneckChange[senderIdx] > 0)
- // {
- // if(newBottleneck >maxBn)
- // {
- // bottleneckChange[senderIdx] = -1;
- // newBottleneck = bottleneck[senderIdx] +
- // (bottleneckChange[senderIdx] * 1000);
- // if(newBottleneck > minBn)
- // {
- // bottleneck[senderIdx] = newBottleneck;
- // }
- // }
- // else
- // {
- // bottleneck[senderIdx] = newBottleneck;
- // }
- // }
- // else
- // {
- // if(newBottleneck < minBn)
- // {
- // bottleneckChange[senderIdx] = 1;
- // newBottleneck = bottleneck[senderIdx] +
- // (bottleneckChange[senderIdx] * 1000);
- // if(newBottleneck < maxBn)
- // {
- // bottleneck[senderIdx] = newBottleneck;
- // }
- // }
- // else
- // {
- // bottleneck[senderIdx] = newBottleneck;
- // }
- // }
- // }
- // }
- }
-
- // model a channel of given bottleneck, to get the receive timestamp
- get_arrival_time(num10ms[senderIdx] * samplesIn10ms[senderIdx],
- streamLen, bottleneck[senderIdx], packetData[senderIdx],
- encoderSampRate[senderIdx]*1000, encoderSampRate[senderIdx]*1000);
-
- // Write the arrival time.
- if(senderIdx == 0)
- {
- fwrite(&(packetData[senderIdx]->arrival_time), sizeof(unsigned int), 1,
- arrivalTimeFile1);
- }
- else
- {
- fwrite(&(packetData[senderIdx]->arrival_time), sizeof(unsigned int), 1,
- arrivalTimeFile2);
- }
-
- // BWE
- if(WebRtcIsac_UpdateBwEstimate(codecInstance[receiverIdx],
- bitStream, streamLen, packetData[senderIdx]->rtp_number,
- packetData[senderIdx]->sample_count,
- packetData[senderIdx]->arrival_time) < 0)
- {
- printf(" BWE Error at client %d \n", receiverIdx + 1);
- return -1;
- }
- /**/
- // Decode
- lenDecodedAudio = WebRtcIsac_Decode(
- codecInstance[receiverIdx], bitStream, streamLen,
- audioBuff60ms, speechType);
- if(lenDecodedAudio < 0)
- {
- printf(" Decoder error in client %d \n", receiverIdx + 1);
- return -1;
- }
-
-
- if(encoderSampRate[senderIdx] == kIsacWideband)
- {
- WebRtcSpl_UpsampleBy2(audioBuff60ms, lenDecodedAudio, resampledAudio60ms,
- resamplerState[receiverIdx]);
- fwrite(resampledAudio60ms, sizeof(short), lenDecodedAudio << 1,
- outFile[receiverIdx]);
- }
- else
- {
- fwrite(audioBuff60ms, sizeof(short), lenDecodedAudio,
- outFile[receiverIdx]);
- }
- num10ms[senderIdx] = 0;
- }
- //}
- //}
- }
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/debugUtility.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/debugUtility.h
deleted file mode 100644
index d708ad1..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/debugUtility.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_TEST_DEBUGUTILITY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_TEST_DEBUGUTILITY_H_
-
-#include <stdio.h>
-#include <string.h>
-#include "utility.h"
-
-typedef struct
-{
- FILE* res0to4FilePtr;
- FILE* res4to8FilePtr;
- FILE* res8to12FilePtr;
- FILE* res8to16FilePtr;
-
- FILE* res0to4DecFilePtr;
- FILE* res4to8DecFilePtr;
- FILE* res8to12DecFilePtr;
- FILE* res8to16DecFilePtr;
-
- FILE* in0to4FilePtr;
- FILE* in4to8FilePtr;
- FILE* in8to12FilePtr;
- FILE* in8to16FilePtr;
-
- FILE* out0to4FilePtr;
- FILE* out4to8FilePtr;
- FILE* out8to12FilePtr;
- FILE* out8to16FilePtr;
-
- FILE* fftFilePtr;
- FILE* fftDecFilePtr;
-
- FILE* arrivalTime;
-
- float lastArrivalTime;
-
- int prevPacketLost;
- int currPacketLost;
- int nextPacketLost;
-
- //double residualSignal4kHZ[240];
- int packetLossPercent;
-
- int maxPayloadLB;
- int maxPayloadUB;
- int lbBytes;
- int ubBytes;
-
-
-}debugStruct;
-
-
-#define PRINT_ENTROPY_INFO(obj) \
- do \
- { \
- printf("%10u, %u; ", \
- obj->bitstr_obj.streamval, obj->bitstr_obj.stream_index); \
- } while(0)
-
-int setupDebugStruct(debugStruct* str);
-
-#endif
\ No newline at end of file
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/simpleKenny.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/test/simpleKenny.c
deleted file mode 100644
index d92ab89..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/test/simpleKenny.c
+++ /dev/null
@@ -1,634 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/* kenny.c - Main function for the iSAC coder */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <time.h>
-
-#ifdef WIN32
-#include "windows.h"
-#define CLOCKS_PER_SEC 1000
-#endif
-
-#include <ctype.h>
-#include <math.h>
-
-/* include API */
-#include "isac.h"
-#include "utility.h"
-//#include "commonDefs.h"
-
-/* max number of samples per frame (= 60 ms frame) */
-#define MAX_FRAMESAMPLES_SWB 1920
-/* number of samples per 10ms frame */
-#define FRAMESAMPLES_SWB_10ms 320
-#define FRAMESAMPLES_WB_10ms 160
-
-/* sampling frequency (Hz) */
-#define FS_SWB 32000
-#define FS_WB 16000
-
-//#define CHANGE_OUTPUT_NAME
-
-#ifdef HAVE_DEBUG_INFO
- #include "debugUtility.h"
- debugStruct debugInfo;
-#endif
-
-unsigned long framecnt = 0;
-
-int main(int argc, char* argv[])
-{
- //--- File IO ----
- FILE* inp;
- FILE* outp;
- char inname[500];
- char outname[500];
-
- /* Runtime statistics */
- double rate;
- double rateRCU;
- unsigned long totalbits = 0;
- unsigned long totalBitsRCU = 0;
- unsigned long totalsmpls =0;
-
- WebRtc_Word32 bottleneck = 39;
- WebRtc_Word16 frameSize = 30; /* ms */
- WebRtc_Word16 codingMode = 1;
- WebRtc_Word16 shortdata[FRAMESAMPLES_SWB_10ms];
- WebRtc_Word16 decoded[MAX_FRAMESAMPLES_SWB];
- //WebRtc_UWord16 streamdata[1000];
- WebRtc_Word16 speechType[1];
- WebRtc_Word16 payloadLimit;
- WebRtc_Word32 rateLimit;
- ISACStruct* ISAC_main_inst;
-
- WebRtc_Word16 stream_len = 0;
- WebRtc_Word16 declen;
- WebRtc_Word16 err;
- WebRtc_Word16 cur_framesmpls;
- int endfile;
-#ifdef WIN32
- double length_file;
- double runtime;
- char outDrive[10];
- char outPath[500];
- char outPrefix[500];
- char outSuffix[500];
- char bitrateFileName[500];
- FILE* bitrateFile;
- double starttime;
- double rateLB = 0;
- double rateUB = 0;
-#endif
- FILE* histFile;
- FILE* averageFile;
- int sampFreqKHz;
- int samplesIn10Ms;
- WebRtc_Word16 maxStreamLen = 0;
- char histFileName[500];
- char averageFileName[500];
- unsigned int hist[600];
- unsigned int tmpSumStreamLen = 0;
- unsigned int packetCntr = 0;
- unsigned int lostPacketCntr = 0;
- WebRtc_UWord16 payload[600];
- WebRtc_UWord16 payloadRCU[600];
- WebRtc_UWord16 packetLossPercent = 0;
- WebRtc_Word16 rcuStreamLen = 0;
- int onlyEncode;
- int onlyDecode;
-
-
- BottleNeckModel packetData;
- packetData.arrival_time = 0;
- packetData.sample_count = 0;
- packetData.rtp_number = 0;
- memset(hist, 0, sizeof(hist));
-
- /* handling wrong input arguments in the command line */
- if(argc < 5)
- {
- int size;
- WebRtcIsac_AssignSize(&size);
-
- printf("\n\nWrong number of arguments or flag values.\n\n");
-
- printf("Usage:\n\n");
- printf("%s infile outfile -bn bottelneck [options] \n\n", argv[0]);
- printf("with:\n");
- printf("-I................... indicates encoding in instantaneous mode.\n");
- printf("-bn bottleneck....... the value of the bottleneck in bit/sec, e.g. 39742,\n");
- printf(" in instantaneous (channel-independent) mode.\n\n");
- printf("infile............... Normal speech input file\n\n");
- printf("outfile.............. Speech output file\n\n");
- printf("OPTIONS\n");
- printf("-------\n");
- printf("-fs sampFreq......... sampling frequency of codec 16 or 32 (default) kHz.\n");
- printf("-plim payloadLim..... payload limit in bytes,\n");
- printf(" default is the maximum possible.\n");
- printf("-rlim rateLim........ rate limit in bits/sec, \n");
- printf(" default is the maimum possible.\n");
- printf("-h file.............. record histogram and *append* to 'file'.\n");
- printf("-ave file............ record average rate of 3 sec intervales and *append* to 'file'.\n");
- printf("-ploss............... packet-loss percentage.\n");
- printf("-enc................. do only encoding and store the bit-stream\n");
- printf("-dec................. the input file is a bit-stream, decode it.\n");
-
- printf("\n");
- printf("Example usage:\n\n");
- printf("%s speechIn.pcm speechOut.pcm -B 40000 -fs 32 \n\n", argv[0]);
-
- printf("structure size %d bytes\n", size);
-
- exit(0);
- }
-
-
-
- /* Get Bottleneck value */
- bottleneck = readParamInt(argc, argv, "-bn", 50000);
- fprintf(stderr,"\nfixed bottleneck rate of %d bits/s\n\n", bottleneck);
-
- /* Get Input and Output files */
- sscanf(argv[1], "%s", inname);
- sscanf(argv[2], "%s", outname);
- codingMode = readSwitch(argc, argv, "-I");
- sampFreqKHz = (WebRtc_Word16)readParamInt(argc, argv, "-fs", 32);
- if(readParamString(argc, argv, "-h", histFileName, 500) > 0)
- {
- histFile = fopen(histFileName, "a");
- if(histFile == NULL)
- {
- printf("cannot open hist file %s", histFileName);
- exit(0);
- }
- }
- else
- {
- // NO recording of hitstogram
- histFile = NULL;
- }
-
-
- packetLossPercent = readParamInt(argc, argv, "-ploss", 0);
-
- if(readParamString(argc, argv, "-ave", averageFileName, 500) > 0)
- {
- averageFile = fopen(averageFileName, "a");
- if(averageFile == NULL)
- {
- printf("cannot open file to write rate %s", averageFileName);
- exit(0);
- }
- }
- else
- {
- averageFile = NULL;
- }
-
- onlyEncode = readSwitch(argc, argv, "-enc");
- onlyDecode = readSwitch(argc, argv, "-dec");
-
-
- switch(sampFreqKHz)
- {
- case 16:
- {
- samplesIn10Ms = 160;
- break;
- }
- case 32:
- {
- samplesIn10Ms = 320;
- break;
- }
- default:
- printf("A sampling frequency of %d kHz is not supported,\
-valid values are 8 and 16.\n", sampFreqKHz);
- exit(-1);
- }
- payloadLimit = (WebRtc_Word16)readParamInt(argc, argv, "-plim", 400);
- rateLimit = readParamInt(argc, argv, "-rlim", 106800);
-
- if ((inp = fopen(inname,"rb")) == NULL) {
- printf(" iSAC: Cannot read file %s.\n", inname);
- exit(1);
- }
- if ((outp = fopen(outname,"wb")) == NULL) {
- printf(" iSAC: Cannot write file %s.\n", outname);
- exit(1);
- }
-
-#ifdef WIN32
- _splitpath(outname, outDrive, outPath, outPrefix, outSuffix);
- _makepath(bitrateFileName, outDrive, outPath, "bitrate", ".txt");
-
- bitrateFile = fopen(bitrateFileName, "a");
- fprintf(bitrateFile, "% %%s \n", inname);
-#endif
-
- printf("\n");
- printf("Input.................... %s\n", inname);
- printf("Output................... %s\n", outname);
- printf("Encoding Mode............ %s\n",
- (codingMode == 1)? "Channel-Independent":"Channel-Adaptive");
- printf("Bottleneck............... %d bits/sec\n", bottleneck);
- printf("Packet-loss Percentage... %d\n", packetLossPercent);
- printf("\n");
-
-#ifdef WIN32
- starttime = clock()/(double)CLOCKS_PER_SEC; /* Runtime statistics */
-#endif
-
- /* Initialize the ISAC and BN structs */
- err = WebRtcIsac_Create(&ISAC_main_inst);
-
- WebRtcIsac_SetEncSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband: kIsacSuperWideband);
- WebRtcIsac_SetDecSampRate(ISAC_main_inst, (sampFreqKHz == 16)? kIsacWideband: kIsacSuperWideband);
- /* Error check */
- if (err < 0) {
- fprintf(stderr,"\n\n Error in create.\n\n");
- exit(EXIT_FAILURE);
- }
-
- framecnt = 0;
- endfile = 0;
-
- /* Initialize encoder and decoder */
- if(WebRtcIsac_EncoderInit(ISAC_main_inst, codingMode) < 0)
- {
- printf("cannot initialize encoder\n");
- return -1;
- }
- if(WebRtcIsac_DecoderInit(ISAC_main_inst) < 0)
- {
- printf("cannot initialize decoder\n");
- return -1;
- }
-
- //{
- // WebRtc_Word32 b1, b2;
- // FILE* fileID = fopen("GetBNTest.txt", "w");
- // b2 = 32100;
- // while(b2 <= 52000)
- // {
- // WebRtcIsac_Control(ISAC_main_inst, b2, frameSize);
- // WebRtcIsac_GetUplinkBw(ISAC_main_inst, &b1);
- // fprintf(fileID, "%5d %5d\n", b2, b1);
- // b2 += 10;
- // }
- //}
-
- if(codingMode == 1)
- {
- if(WebRtcIsac_Control(ISAC_main_inst, bottleneck, frameSize) < 0)
- {
- printf("cannot set bottleneck\n");
- return -1;
- }
- }
- else
- {
- if(WebRtcIsac_ControlBwe(ISAC_main_inst, 15000, 30, 1) < 0)
- {
- printf("cannot configure BWE\n");
- return -1;
- }
- }
-
- if(WebRtcIsac_SetMaxPayloadSize(ISAC_main_inst, payloadLimit) < 0)
- {
- printf("cannot set maximum payload size %d.\n", payloadLimit);
- return -1;
- }
-
- if (rateLimit < 106800) {
- if(WebRtcIsac_SetMaxRate(ISAC_main_inst, rateLimit) < 0)
- {
- printf("cannot set the maximum rate %d.\n", rateLimit);
- return -1;
- }
- }
-
- //=====================================
-//#ifdef HAVE_DEBUG_INFO
-// if(setupDebugStruct(&debugInfo) < 0)
-// {
-// exit(1);
-// }
-//#endif
-
- while (endfile == 0)
- {
- fprintf(stderr," \rframe = %7li", framecnt);
-
- //============== Readind from the file and encoding =================
- cur_framesmpls = 0;
- stream_len = 0;
-
-
- if(onlyDecode)
- {
- WebRtc_UWord8 auxUW8;
- size_t auxSizet;
- if(fread(&auxUW8, sizeof(WebRtc_UWord8), 1, inp) < 1)
- {
- break;
- }
- stream_len = ((WebRtc_UWord8)auxUW8) << 8;
- if(fread(&auxUW8, sizeof(WebRtc_UWord8), 1, inp) < 1)
- {
- break;
- }
- stream_len |= (WebRtc_UWord16)auxUW8;
- auxSizet = (size_t)stream_len;
- if(fread(payload, 1, auxSizet, inp) < auxSizet)
- {
- printf("last payload is corrupted\n");
- break;
- }
- }
- else
- {
- while(stream_len == 0)
- {
- // Read 10 ms speech block
- endfile = readframe(shortdata, inp, samplesIn10Ms);
- if(endfile)
- {
- break;
- }
- cur_framesmpls += samplesIn10Ms;
-
- //-------- iSAC encoding ---------
- stream_len = WebRtcIsac_Encode(ISAC_main_inst, shortdata,
- (WebRtc_Word16*)payload);
-
- if(stream_len < 0)
- {
- // exit if returned with error
- //errType=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- fprintf(stderr,"\nError in encoder\n");
- getchar();
- exit(EXIT_FAILURE);
- }
-
-
- }
- //===================================================================
- if(endfile)
- {
- break;
- }
-
- rcuStreamLen = WebRtcIsac_GetRedPayload(ISAC_main_inst, (WebRtc_Word16*)payloadRCU);
-
- get_arrival_time(cur_framesmpls, stream_len, bottleneck, &packetData,
- sampFreqKHz * 1000, sampFreqKHz * 1000);
- if(WebRtcIsac_UpdateBwEstimate(ISAC_main_inst,
- payload, stream_len, packetData.rtp_number,
- packetData.sample_count,
- packetData.arrival_time) < 0)
- {
- printf(" BWE Error at client\n");
- return -1;
- }
- }
-
- if(endfile)
- {
- break;
- }
-
- maxStreamLen = (stream_len > maxStreamLen)? stream_len:maxStreamLen;
- packetCntr++;
-
- hist[stream_len]++;
- if(averageFile != NULL)
- {
- tmpSumStreamLen += stream_len;
- if(packetCntr == 100)
- {
- // kbps
- fprintf(averageFile, "%8.3f ", (double)tmpSumStreamLen * 8.0 / (30.0 * packetCntr));
- packetCntr = 0;
- tmpSumStreamLen = 0;
- }
- }
-
- if(onlyEncode)
- {
- WebRtc_UWord8 auxUW8;
- auxUW8 = (WebRtc_UWord8)(((stream_len & 0x7F00) >> 8) & 0xFF);
- fwrite(&auxUW8, sizeof(WebRtc_UWord8), 1, outp);
-
- auxUW8 = (WebRtc_UWord8)(stream_len & 0xFF);
- fwrite(&auxUW8, sizeof(WebRtc_UWord8), 1, outp);
- fwrite(payload, 1, stream_len, outp);
- }
- else
- {
-
- //======================= iSAC decoding ===========================
-
- if((rand() % 100) < packetLossPercent)
- {
- declen = WebRtcIsac_DecodeRcu(ISAC_main_inst, payloadRCU,
- rcuStreamLen, decoded, speechType);
- lostPacketCntr++;
- }
- else
- {
- declen = WebRtcIsac_Decode(ISAC_main_inst, payload,
- stream_len, decoded, speechType);
- }
- if(declen <= 0)
- {
- //errType=WebRtcIsac_GetErrorCode(ISAC_main_inst);
- fprintf(stderr,"\nError in decoder.\n");
- getchar();
- exit(1);
- }
-
- // Write decoded speech frame to file
- fwrite(decoded, sizeof(WebRtc_Word16), declen, outp);
- cur_framesmpls = declen;
- }
- // Update Statistics
- framecnt++;
- totalsmpls += cur_framesmpls;
- if(stream_len > 0)
- {
- totalbits += 8 * stream_len;
- }
- if(rcuStreamLen > 0)
- {
- totalBitsRCU += 8 * rcuStreamLen;
- }
- }
-
- rate = ((double)totalbits * (sampFreqKHz)) / (double)totalsmpls;
- rateRCU = ((double)totalBitsRCU * (sampFreqKHz)) / (double)totalsmpls;
-
- printf("\n\n");
- printf("Sampling Rate......................... %d kHz\n", sampFreqKHz);
- printf("Payload Limit......................... %d bytes \n", payloadLimit);
- printf("Rate Limit............................ %d bits/sec \n", rateLimit);
-
-#ifdef WIN32
-#ifdef HAVE_DEBUG_INFO
- rateLB = ((double)debugInfo.lbBytes * 8. *
- (sampFreqKHz)) / (double)totalsmpls;
- rateUB = ((double)debugInfo.ubBytes * 8. *
- (sampFreqKHz)) / (double)totalsmpls;
-#endif
-
- fprintf(bitrateFile, "%d %10u %d %6.3f %6.3f %6.3f\n",
- sampFreqKHz,
- framecnt,
- bottleneck,
- rateLB,
- rateUB,
- rate);
- fclose(bitrateFile);
-#endif // WIN32
-
- printf("\n");
- printf("Measured bit-rate..................... %0.3f kbps\n", rate);
- printf("Measured RCU bit-ratre................ %0.3f kbps\n", rateRCU);
- printf("Maximum bit-rate/payloadsize.......... %0.3f / %d\n",
- maxStreamLen * 8 / 0.03, maxStreamLen);
- printf("Measured packet-loss.................. %0.1f%% \n",
- 100.0f * (float)lostPacketCntr / (float)packetCntr);
-
-//#ifdef HAVE_DEBUG_INFO
-// printf("Measured lower-band bit-rate.......... %0.3f kbps (%.0f%%)\n",
-// rateLB, (double)(rateLB) * 100. /(double)(rate));
-// printf("Measured upper-band bit-rate.......... %0.3f kbps (%.0f%%)\n",
-// rateUB, (double)(rateUB) * 100. /(double)(rate));
-//
-// printf("Maximum payload lower-band............ %d bytes (%0.3f kbps)\n",
-// debugInfo.maxPayloadLB, debugInfo.maxPayloadLB * 8.0 / 0.03);
-// printf("Maximum payload upper-band............ %d bytes (%0.3f kbps)\n",
-// debugInfo.maxPayloadUB, debugInfo.maxPayloadUB * 8.0 / 0.03);
-//#endif
-
- printf("\n");
-
- /* Runtime statistics */
-#ifdef WIN32
- runtime = (double)(clock()/(double)CLOCKS_PER_SEC-starttime);
- length_file = ((double)framecnt*(double)declen/(sampFreqKHz*1000));
- printf("Length of speech file................ %.1f s\n", length_file);
- printf("Time to run iSAC..................... %.2f s (%.2f %% of realtime)\n\n",
- runtime, (100*runtime/length_file));
-#endif
- printf("\n\n_______________________________________________\n");
-
- if(histFile != NULL)
- {
- int n;
- for(n = 0; n < 600; n++)
- {
- fprintf(histFile, "%6d ", hist[n]);
- }
- fprintf(histFile, "\n");
- fclose(histFile);
- }
- if(averageFile != NULL)
- {
- if(packetCntr > 0)
- {
- fprintf(averageFile, "%8.3f ", (double)tmpSumStreamLen * 8.0 / (30.0 * packetCntr));
- }
- fprintf(averageFile, "\n");
- fclose(averageFile);
- }
-
- fclose(inp);
- fclose(outp);
-
- WebRtcIsac_Free(ISAC_main_inst);
-
-
-#ifdef CHANGE_OUTPUT_NAME
- {
- char* p;
- char myExt[50];
- char bitRateStr[10];
- char newOutName[500];
- strcpy(newOutName, outname);
-
- myExt[0] = '\0';
- p = strchr(newOutName, '.');
- if(p != NULL)
- {
- strcpy(myExt, p);
- *p = '_';
- p++;
- *p = '\0';
- }
- else
- {
- strcat(newOutName, "_");
- }
- sprintf(bitRateStr, "%0.0fkbps", rate);
- strcat(newOutName, bitRateStr);
- strcat(newOutName, myExt);
- rename(outname, newOutName);
- }
-#endif
- exit(0);
-}
-
-
-#ifdef HAVE_DEBUG_INFO
-int setupDebugStruct(debugStruct* str)
-{
- str->prevPacketLost = 0;
- str->currPacketLost = 0;
-
- OPEN_FILE_WB(str->res0to4FilePtr, "Res0to4.dat");
- OPEN_FILE_WB(str->res4to8FilePtr, "Res4to8.dat");
- OPEN_FILE_WB(str->res8to12FilePtr, "Res8to12.dat");
- OPEN_FILE_WB(str->res8to16FilePtr, "Res8to16.dat");
-
- OPEN_FILE_WB(str->res0to4DecFilePtr, "Res0to4Dec.dat");
- OPEN_FILE_WB(str->res4to8DecFilePtr, "Res4to8Dec.dat");
- OPEN_FILE_WB(str->res8to12DecFilePtr, "Res8to12Dec.dat");
- OPEN_FILE_WB(str->res8to16DecFilePtr, "Res8to16Dec.dat");
-
- OPEN_FILE_WB(str->in0to4FilePtr, "in0to4.dat");
- OPEN_FILE_WB(str->in4to8FilePtr, "in4to8.dat");
- OPEN_FILE_WB(str->in8to12FilePtr, "in8to12.dat");
- OPEN_FILE_WB(str->in8to16FilePtr, "in8to16.dat");
-
- OPEN_FILE_WB(str->out0to4FilePtr, "out0to4.dat");
- OPEN_FILE_WB(str->out4to8FilePtr, "out4to8.dat");
- OPEN_FILE_WB(str->out8to12FilePtr, "out8to12.dat");
- OPEN_FILE_WB(str->out8to16FilePtr, "out8to16.dat");
- OPEN_FILE_WB(str->fftFilePtr, "riFFT.dat");
- OPEN_FILE_WB(str->fftDecFilePtr, "riFFTDec.dat");
-
- OPEN_FILE_WB(str->arrivalTime, NULL/*"ArivalTime.dat"*/);
- str->lastArrivalTime = 0;
-
- str->maxPayloadLB = 0;
- str->maxPayloadUB = 0;
- str->lbBytes = 0;
- str->ubBytes = 0;
-
- return 0;
-};
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/util/utility.c b/trunk/src/modules/audio_coding/codecs/iSAC/main/util/utility.c
deleted file mode 100644
index 0a2256a..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/util/utility.c
+++ /dev/null
@@ -1,178 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdlib.h>
-#include <stdio.h>
-#include <string.h>
-#include <math.h>
-#include "utility.h"
-
-/* function for reading audio data from PCM file */
-int
-readframe(
- short* data,
- FILE* inp,
- int length)
-{
- short k, rlen, status = 0;
- unsigned char* ptrUChar;
- ptrUChar = (unsigned char*)data;
-
- rlen = (short)fread(data, sizeof(short), length, inp);
- if (rlen < length) {
- for (k = rlen; k < length; k++)
- data[k] = 0;
- status = 1;
- }
-
- // Assuming that our PCM files are written in Intel machines
- for(k = 0; k < length; k++)
- {
- data[k] = (short)ptrUChar[k<<1] | ((((short)ptrUChar[(k<<1) + 1]) << 8) & 0xFF00);
- }
-
- return status;
-}
-
-short
-readSwitch(
- int argc,
- char* argv[],
- char* strID)
-{
- short n;
- for(n = 0; n < argc; n++)
- {
- if(strcmp(argv[n], strID) == 0)
- {
- return 1;
- }
- }
- return 0;
-}
-
-double
-readParamDouble(
- int argc,
- char* argv[],
- char* strID,
- double defaultVal)
-{
- double returnVal = defaultVal;
- short n;
- for(n = 0; n < argc; n++)
- {
- if(strcmp(argv[n], strID) == 0)
- {
- n++;
- if(n < argc)
- {
- returnVal = atof(argv[n]);
- }
- break;
- }
- }
- return returnVal;
-}
-
-int
-readParamInt(
- int argc,
- char* argv[],
- char* strID,
- int defaultVal)
-{
- int returnVal = defaultVal;
- short n;
- for(n = 0; n < argc; n++)
- {
- if(strcmp(argv[n], strID) == 0)
- {
- n++;
- if(n < argc)
- {
- returnVal = atoi(argv[n]);
- }
- break;
- }
- }
- return returnVal;
-}
-
-int
-readParamString(
- int argc,
- char* argv[],
- char* strID,
- char* stringParam,
- int maxSize)
-{
- int paramLenght = 0;
- short n;
- for(n = 0; n < argc; n++)
- {
- if(strcmp(argv[n], strID) == 0)
- {
- n++;
- if(n < argc)
- {
- strncpy(stringParam, argv[n], maxSize);
- paramLenght = (int)strlen(argv[n]);
- }
- break;
- }
- }
- return paramLenght;
-}
-
-void
-get_arrival_time(
- int current_framesamples, /* samples */
- int packet_size, /* bytes */
- int bottleneck, /* excluding headers; bits/s */
- BottleNeckModel* BN_data,
- short senderSampFreqHz,
- short receiverSampFreqHz)
-{
- unsigned int travelTimeMs;
- const int headerSizeByte = 35;
-
- int headerRate;
-
- BN_data->whenPackGeneratedMs += (current_framesamples / (senderSampFreqHz / 1000));
-
- headerRate = headerSizeByte * 8 * senderSampFreqHz / current_framesamples; /* bits/s */
-
- /* everything in samples */
- BN_data->sample_count = BN_data->sample_count + current_framesamples;
-
- //travelTimeMs = ((packet_size + HeaderSize) * 8 * sampFreqHz) /
- // (bottleneck + HeaderRate)
- travelTimeMs = (unsigned int)floor((double)((packet_size + headerSizeByte) * 8 * 1000)
- / (double)(bottleneck + headerRate) + 0.5);
-
- if(BN_data->whenPrevPackLeftMs > BN_data->whenPackGeneratedMs)
- {
- BN_data->whenPrevPackLeftMs += travelTimeMs;
- }
- else
- {
- BN_data->whenPrevPackLeftMs = BN_data->whenPackGeneratedMs +
- travelTimeMs;
- }
-
- BN_data->arrival_time = (BN_data->whenPrevPackLeftMs *
- (receiverSampFreqHz / 1000));
-
-// if (BN_data->arrival_time < BN_data->sample_count)
-// BN_data->arrival_time = BN_data->sample_count;
-
- BN_data->rtp_number++;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/iSAC/main/util/utility.h b/trunk/src/modules/audio_coding/codecs/iSAC/main/util/utility.h
deleted file mode 100644
index f9fba94..0000000
--- a/trunk/src/modules/audio_coding/codecs/iSAC/main/util/utility.h
+++ /dev/null
@@ -1,144 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_UTIL_UTILITY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_MAIN_UTIL_UTILITY_H_
-
-#include <stdlib.h>
-#include <stdio.h>
-
-#if defined(__cplusplus)
-extern "C" {
-#endif
-
-#define OPEN_FILE_WB(filePtr, fullPath) \
- do \
- { \
- if(fullPath != NULL) \
- { \
- filePtr = fopen(fullPath, "wb"); \
- if(filePtr == NULL) \
- { \
- printf("could not open %s to write to.", fullPath); \
- return -1; \
- } \
- } \
- else \
- { \
- filePtr = NULL; \
- } \
- }while(0)
-
-#define OPEN_FILE_AB(filePtr, fullPath) \
- do \
- { \
- if(fullPath != NULL) \
- { \
- filePtr = fopen(fullPath, "ab"); \
- if(filePtr == NULL) \
- { \
- printf("could not open %s to write to.", fullPath); \
- return -1; \
- } \
- } \
- else \
- { \
- filePtr = NULL; \
- } \
- }while(0)
-
-#define OPEN_FILE_RB(filePtr, fullPath) \
- do \
- { \
- if(fullPath != NULL) \
- { \
- filePtr = fopen(fullPath, "rb"); \
- if(filePtr == NULL) \
- { \
- printf("could not open %s to read from.", fullPath); \
- return -1; \
- } \
- } \
- else \
- { \
- filePtr = NULL; \
- } \
- }while(0)
-
-#define WRITE_FILE_D(bufferPtr, len, filePtr) \
- do \
- { \
- if(filePtr != NULL) \
- { \
- double dummy[1000]; \
- int cntr; \
- for(cntr = 0; cntr < (len); cntr++) \
- { \
- dummy[cntr] = (double)bufferPtr[cntr]; \
- } \
- fwrite(dummy, sizeof(double), len, filePtr); \
- fflush(filePtr); \
- } \
- } while(0)
-
- typedef struct {
- unsigned int whenPackGeneratedMs;
- unsigned int whenPrevPackLeftMs;
- unsigned int sendTimeMs ; /* milisecond */
- unsigned int arrival_time; /* samples */
- unsigned int sample_count; /* samples, also used as "send time stamp" */
- unsigned int rtp_number;
- } BottleNeckModel;
-
- void get_arrival_time(
- int current_framesamples, /* samples */
- int packet_size, /* bytes */
- int bottleneck, /* excluding headers; bits/s */
- BottleNeckModel* BN_data,
- short senderSampFreqHz,
- short receiverSampFreqHz);
-
- /* function for reading audio data from PCM file */
- int readframe(
- short* data,
- FILE* inp,
- int length);
-
- short readSwitch(
- int argc,
- char* argv[],
- char* strID);
-
- double readParamDouble(
- int argc,
- char* argv[],
- char* strID,
- double defaultVal);
-
- int readParamInt(
- int argc,
- char* argv[],
- char* strID,
- int defaultVal);
-
- int readParamString(
- int argc,
- char* argv[],
- char* strID,
- char* stringParam,
- int maxSize);
-
-#if defined(__cplusplus)
-}
-#endif
-
-
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/Android.mk b/trunk/src/modules/audio_coding/codecs/ilbc/Android.mk
deleted file mode 100644
index cbadcab..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/Android.mk
+++ /dev/null
@@ -1,165 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../android-webrtc.mk
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_ilbc
-LOCAL_MODULE_TAGS := optional
-LOCAL_SRC_FILES := \
- abs_quant.c \
- abs_quant_loop.c \
- augmented_cb_corr.c \
- bw_expand.c \
- cb_construct.c \
- cb_mem_energy.c \
- cb_mem_energy_augmentation.c \
- cb_mem_energy_calc.c \
- cb_search.c \
- cb_search_core.c \
- cb_update_best_index.c \
- chebyshev.c \
- comp_corr.c \
- constants.c \
- create_augmented_vec.c \
- decode.c \
- decode_residual.c \
- decoder_interpolate_lsf.c \
- do_plc.c \
- encode.c \
- energy_inverse.c \
- enh_upsample.c \
- enhancer.c \
- enhancer_interface.c \
- filtered_cb_vecs.c \
- frame_classify.c \
- gain_dequant.c \
- gain_quant.c \
- get_cd_vec.c \
- get_lsp_poly.c \
- get_sync_seq.c \
- hp_input.c \
- hp_output.c \
- ilbc.c \
- index_conv_dec.c \
- index_conv_enc.c \
- init_decode.c \
- init_encode.c \
- interpolate.c \
- interpolate_samples.c \
- lpc_encode.c \
- lsf_check.c \
- lsf_interpolate_to_poly_dec.c \
- lsf_interpolate_to_poly_enc.c \
- lsf_to_lsp.c \
- lsf_to_poly.c \
- lsp_to_lsf.c \
- my_corr.c \
- nearest_neighbor.c \
- pack_bits.c \
- poly_to_lsf.c \
- poly_to_lsp.c \
- refiner.c \
- simple_interpolate_lsf.c \
- simple_lpc_analysis.c \
- simple_lsf_dequant.c \
- simple_lsf_quant.c \
- smooth.c \
- smooth_out_data.c \
- sort_sq.c \
- split_vq.c \
- state_construct.c \
- state_search.c \
- swap_bytes.c \
- unpack_bits.c \
- vq3.c \
- vq4.c \
- window32_w32.c \
- xcorr_coef.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/interface \
- $(LOCAL_PATH)/../../../.. \
- $(LOCAL_PATH)/../../../../common_audio/signal_processing/include
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
-
-
-# iLBC test app
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_TAGS := tests
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES:= test/iLBC_test.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/interface \
- $(LOCAL_PATH)/../../../..
-
-LOCAL_STATIC_LIBRARIES := \
- libwebrtc_ilbc \
- libwebrtc_spl
-
-LOCAL_SHARED_LIBRARIES := \
- libutils
-
-LOCAL_MODULE:= webrtc_ilbc_test
-
-ifdef NDK_ROOT
-include $(BUILD_EXECUTABLE)
-else
-include $(BUILD_NATIVE_TEST)
-endif
-
-# iLBC_testLib test app
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_TAGS := tests
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES:= test/iLBC_testLib.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/interface \
- $(LOCAL_PATH)/../../../..
-
-LOCAL_STATIC_LIBRARIES := \
- libwebrtc_ilbc \
- libwebrtc_spl
-
-LOCAL_SHARED_LIBRARIES := \
- libutils
-
-LOCAL_MODULE:= webrtc_ilbc_testLib
-
-ifdef NDK_ROOT
-include $(BUILD_EXECUTABLE)
-else
-include $(BUILD_NATIVE_TEST)
-endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant.c b/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant.c
deleted file mode 100644
index 4a70c8b..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant.c
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_AbsQuant.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-#include "abs_quant_loop.h"
-
-
-/*----------------------------------------------------------------*
- * predictive noise shaping encoding of scaled start state
- * (subrutine for WebRtcIlbcfix_StateSearch)
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_AbsQuant(
- iLBC_Enc_Inst_t *iLBCenc_inst,
- /* (i) Encoder instance */
- iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
- and idxVec, uses state_first as
- input) */
- WebRtc_Word16 *in, /* (i) vector to encode */
- WebRtc_Word16 *weightDenum /* (i) denominator of synthesis filter */
- ) {
- WebRtc_Word16 *syntOut;
- WebRtc_Word16 quantLen[2];
-
- /* Stack based */
- WebRtc_Word16 syntOutBuf[LPC_FILTERORDER+STATE_SHORT_LEN_30MS];
- WebRtc_Word16 in_weightedVec[STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
- WebRtc_Word16 *in_weighted = &in_weightedVec[LPC_FILTERORDER];
-
- /* Initialize the buffers */
- WebRtcSpl_MemSetW16(syntOutBuf, 0, LPC_FILTERORDER+STATE_SHORT_LEN_30MS);
- syntOut = &syntOutBuf[LPC_FILTERORDER];
- /* Start with zero state */
- WebRtcSpl_MemSetW16(in_weightedVec, 0, LPC_FILTERORDER);
-
- /* Perform the quantization loop in two sections of length quantLen[i],
- where the perceptual weighting filter is updated at the subframe
- border */
-
- if (iLBC_encbits->state_first) {
- quantLen[0]=SUBL;
- quantLen[1]=iLBCenc_inst->state_short_len-SUBL;
- } else {
- quantLen[0]=iLBCenc_inst->state_short_len-SUBL;
- quantLen[1]=SUBL;
- }
-
- /* Calculate the weighted residual, switch perceptual weighting
- filter at the subframe border */
- WebRtcSpl_FilterARFastQ12(
- in, in_weighted,
- weightDenum, LPC_FILTERORDER+1, quantLen[0]);
- WebRtcSpl_FilterARFastQ12(
- &in[quantLen[0]], &in_weighted[quantLen[0]],
- &weightDenum[LPC_FILTERORDER+1], LPC_FILTERORDER+1, quantLen[1]);
-
- WebRtcIlbcfix_AbsQuantLoop(
- syntOut,
- in_weighted,
- weightDenum,
- quantLen,
- iLBC_encbits->idxVec);
-
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant.h b/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant.h
deleted file mode 100644
index fa59593..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_AbsQuant.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * predictive noise shaping encoding of scaled start state
- * (subrutine for WebRtcIlbcfix_StateSearch)
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_AbsQuant(
- iLBC_Enc_Inst_t *iLBCenc_inst,
- /* (i) Encoder instance */
- iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits (outputs idxForMax
- and idxVec, uses state_first as
- input) */
- WebRtc_Word16 *in, /* (i) vector to encode */
- WebRtc_Word16 *weightDenum /* (i) denominator of synthesis filter */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant_loop.c b/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
deleted file mode 100644
index 4eebc3e..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant_loop.c
+++ /dev/null
@@ -1,95 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_AbsQuantLoop.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-#include "sort_sq.h"
-
-void WebRtcIlbcfix_AbsQuantLoop(
- WebRtc_Word16 *syntOutIN,
- WebRtc_Word16 *in_weightedIN,
- WebRtc_Word16 *weightDenumIN,
- WebRtc_Word16 *quantLenIN,
- WebRtc_Word16 *idxVecIN
- )
-{
- int n, k1, k2;
- WebRtc_Word16 index;
- WebRtc_Word32 toQW32;
- WebRtc_Word32 toQ32;
- WebRtc_Word16 tmp16a;
- WebRtc_Word16 xq;
-
- WebRtc_Word16 *syntOut = syntOutIN;
- WebRtc_Word16 *in_weighted = in_weightedIN;
- WebRtc_Word16 *weightDenum = weightDenumIN;
- WebRtc_Word16 *quantLen = quantLenIN;
- WebRtc_Word16 *idxVec = idxVecIN;
-
- n=0;
-
- for(k1=0;k1<2;k1++) {
- for(k2=0;k2<quantLen[k1];k2++){
-
- /* Filter to get the predicted value */
- WebRtcSpl_FilterARFastQ12(
- syntOut, syntOut,
- weightDenum, LPC_FILTERORDER+1, 1);
-
- /* the quantizer */
- toQW32 = (WebRtc_Word32)(*in_weighted) - (WebRtc_Word32)(*syntOut);
-
- toQ32 = (((WebRtc_Word32)toQW32)<<2);
-
- if (toQ32 > 32767) {
- toQ32 = (WebRtc_Word32) 32767;
- } else if (toQ32 < -32768) {
- toQ32 = (WebRtc_Word32) -32768;
- }
-
- /* Quantize the state */
- if (toQW32<(-7577)) {
- /* To prevent negative overflow */
- index=0;
- } else if (toQW32>8151) {
- /* To prevent positive overflow */
- index=7;
- } else {
- /* Find the best quantization index
- (state_sq3Tbl is in Q13 and toQ is in Q11)
- */
- WebRtcIlbcfix_SortSq(&xq, &index,
- (WebRtc_Word16)toQ32,
- WebRtcIlbcfix_kStateSq3, 8);
- }
-
- /* Store selected index */
- (*idxVec++) = index;
-
- /* Compute decoded sample and update of the prediction filter */
- tmp16a = ((WebRtcIlbcfix_kStateSq3[index] + 2 ) >> 2);
-
- *syntOut = (WebRtc_Word16) (tmp16a + (WebRtc_Word32)(*in_weighted) - toQW32);
-
- n++;
- syntOut++; in_weighted++;
- }
- /* Update perceptual weighting filter at subframe border */
- weightDenum += 11;
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant_loop.h b/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
deleted file mode 100644
index f506e8e..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/abs_quant_loop.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_AbsQuantLoop.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ABS_QUANT_LOOP_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * predictive noise shaping encoding of scaled start state
- * (subrutine for WebRtcIlbcfix_StateSearch)
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_AbsQuantLoop(
- WebRtc_Word16 *syntOutIN,
- WebRtc_Word16 *in_weightedIN,
- WebRtc_Word16 *weightDenumIN,
- WebRtc_Word16 *quantLenIN,
- WebRtc_Word16 *idxVecIN
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c b/trunk/src/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
deleted file mode 100644
index 6011e92..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/augmented_cb_corr.c
+++ /dev/null
@@ -1,63 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_AugmentedCbCorr.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-#include "augmented_cb_corr.h"
-
-void WebRtcIlbcfix_AugmentedCbCorr(
- WebRtc_Word16 *target, /* (i) Target vector */
- WebRtc_Word16 *buffer, /* (i) Memory buffer */
- WebRtc_Word16 *interpSamples, /* (i) buffer with
- interpolated samples */
- WebRtc_Word32 *crossDot, /* (o) The cross correlation between
- the target and the Augmented
- vector */
- WebRtc_Word16 low, /* (i) Lag to start from (typically
- 20) */
- WebRtc_Word16 high, /* (i) Lag to end at (typically 39) */
- WebRtc_Word16 scale) /* (i) Scale factor to use for
- the crossDot */
-{
- int lagcount;
- WebRtc_Word16 ilow;
- WebRtc_Word16 *targetPtr;
- WebRtc_Word32 *crossDotPtr;
- WebRtc_Word16 *iSPtr=interpSamples;
-
- /* Calculate the correlation between the target and the
- interpolated codebook. The correlation is calculated in
- 3 sections with the interpolated part in the middle */
- crossDotPtr=crossDot;
- for (lagcount=low; lagcount<=high; lagcount++) {
-
- ilow = (WebRtc_Word16) (lagcount-4);
-
- /* Compute dot product for the first (lagcount-4) samples */
- (*crossDotPtr) = WebRtcSpl_DotProductWithScale(target, buffer-lagcount, ilow, scale);
-
- /* Compute dot product on the interpolated samples */
- (*crossDotPtr) += WebRtcSpl_DotProductWithScale(target+ilow, iSPtr, 4, scale);
- targetPtr = target + lagcount;
- iSPtr += lagcount-ilow;
-
- /* Compute dot product for the remaining samples */
- (*crossDotPtr) += WebRtcSpl_DotProductWithScale(targetPtr, buffer-lagcount, SUBL-lagcount, scale);
- crossDotPtr++;
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h b/trunk/src/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
deleted file mode 100644
index 8e097fe..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/augmented_cb_corr.h
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_AugmentedCbCorr.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_AUGMENTED_CB_CORR_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Calculate correlation between target and Augmented codebooks
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_AugmentedCbCorr(
- WebRtc_Word16 *target, /* (i) Target vector */
- WebRtc_Word16 *buffer, /* (i) Memory buffer */
- WebRtc_Word16 *interpSamples, /* (i) buffer with
- interpolated samples */
- WebRtc_Word32 *crossDot, /* (o) The cross correlation between
- the target and the Augmented
- vector */
- WebRtc_Word16 low, /* (i) Lag to start from (typically
- 20) */
- WebRtc_Word16 high, /* (i) Lag to end at (typically 39 */
- WebRtc_Word16 scale); /* (i) Scale factor to use for
- the crossDot */
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/bw_expand.c b/trunk/src/modules/audio_coding/codecs/ilbc/bw_expand.c
deleted file mode 100644
index a2287aa..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/bw_expand.c
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_BwExpand.c
-
-******************************************************************/
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * lpc bandwidth expansion
- *---------------------------------------------------------------*/
-
-/* The output is in the same domain as the input */
-void WebRtcIlbcfix_BwExpand(
- WebRtc_Word16 *out, /* (o) the bandwidth expanded lpc coefficients */
- WebRtc_Word16 *in, /* (i) the lpc coefficients before bandwidth
- expansion */
- WebRtc_Word16 *coef, /* (i) the bandwidth expansion factor Q15 */
- WebRtc_Word16 length /* (i) the length of lpc coefficient vectors */
- ) {
- int i;
-
- out[0] = in[0];
- for (i = 1; i < length; i++) {
- /* out[i] = coef[i] * in[i] with rounding.
- in[] and out[] are in Q12 and coef[] is in Q15
- */
- out[i] = (WebRtc_Word16)((WEBRTC_SPL_MUL_16_16(coef[i], in[i])+16384)>>15);
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/bw_expand.h b/trunk/src/modules/audio_coding/codecs/ilbc/bw_expand.h
deleted file mode 100644
index c9f3fab..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/bw_expand.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_BwExpand.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_BW_EXPAND_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * lpc bandwidth expansion
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_BwExpand(
- WebRtc_Word16 *out, /* (o) the bandwidth expanded lpc coefficients */
- WebRtc_Word16 *in, /* (i) the lpc coefficients before bandwidth
- expansion */
- WebRtc_Word16 *coef, /* (i) the bandwidth expansion factor Q15 */
- WebRtc_Word16 length /* (i) the length of lpc coefficient vectors */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_construct.c b/trunk/src/modules/audio_coding/codecs/ilbc/cb_construct.c
deleted file mode 100644
index 094a7e4..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/cb_construct.c
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CbConstruct.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "gain_dequant.h"
-#include "get_cd_vec.h"
-
-/*----------------------------------------------------------------*
- * Construct decoded vector from codebook and gains.
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_CbConstruct(
- WebRtc_Word16 *decvector, /* (o) Decoded vector */
- WebRtc_Word16 *index, /* (i) Codebook indices */
- WebRtc_Word16 *gain_index, /* (i) Gain quantization indices */
- WebRtc_Word16 *mem, /* (i) Buffer for codevector construction */
- WebRtc_Word16 lMem, /* (i) Length of buffer */
- WebRtc_Word16 veclen /* (i) Length of vector */
- ){
- int j;
- WebRtc_Word16 gain[CB_NSTAGES];
- /* Stack based */
- WebRtc_Word16 cbvec0[SUBL];
- WebRtc_Word16 cbvec1[SUBL];
- WebRtc_Word16 cbvec2[SUBL];
- WebRtc_Word32 a32;
- WebRtc_Word16 *gainPtr;
-
- /* gain de-quantization */
-
- gain[0] = WebRtcIlbcfix_GainDequant(gain_index[0], 16384, 0);
- gain[1] = WebRtcIlbcfix_GainDequant(gain_index[1], gain[0], 1);
- gain[2] = WebRtcIlbcfix_GainDequant(gain_index[2], gain[1], 2);
-
- /* codebook vector construction and construction of total vector */
-
- /* Stack based */
- WebRtcIlbcfix_GetCbVec(cbvec0, mem, index[0], lMem, veclen);
- WebRtcIlbcfix_GetCbVec(cbvec1, mem, index[1], lMem, veclen);
- WebRtcIlbcfix_GetCbVec(cbvec2, mem, index[2], lMem, veclen);
-
- gainPtr = &gain[0];
- for (j=0;j<veclen;j++) {
- a32 = WEBRTC_SPL_MUL_16_16(*gainPtr++, cbvec0[j]);
- a32 += WEBRTC_SPL_MUL_16_16(*gainPtr++, cbvec1[j]);
- a32 += WEBRTC_SPL_MUL_16_16(*gainPtr, cbvec2[j]);
- gainPtr -= 2;
- decvector[j] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(a32 + 8192, 14);
- }
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_construct.h b/trunk/src/modules/audio_coding/codecs/ilbc/cb_construct.h
deleted file mode 100644
index bec759f..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/cb_construct.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CbConstruct.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_CONSTRUCT_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Construct decoded vector from codebook and gains.
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_CbConstruct(
- WebRtc_Word16 *decvector, /* (o) Decoded vector */
- WebRtc_Word16 *index, /* (i) Codebook indices */
- WebRtc_Word16 *gain_index, /* (i) Gain quantization indices */
- WebRtc_Word16 *mem, /* (i) Buffer for codevector construction */
- WebRtc_Word16 lMem, /* (i) Length of buffer */
- WebRtc_Word16 veclen /* (i) Length of vector */
- );
-
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy.c b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
deleted file mode 100644
index 8613fa2..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy.c
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CbMemEnergy.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-#include "cb_mem_energy_calc.h"
-
-/*----------------------------------------------------------------*
- * Function WebRtcIlbcfix_CbMemEnergy computes the energy of all
- * the vectors in the codebook memory that will be used in the
- * following search for the best match.
- *----------------------------------------------------------------*/
-
-void WebRtcIlbcfix_CbMemEnergy(
- WebRtc_Word16 range,
- WebRtc_Word16 *CB, /* (i) The CB memory (1:st section) */
- WebRtc_Word16 *filteredCB, /* (i) The filtered CB memory (2:nd section) */
- WebRtc_Word16 lMem, /* (i) Length of the CB memory */
- WebRtc_Word16 lTarget, /* (i) Length of the target vector */
- WebRtc_Word16 *energyW16, /* (o) Energy in the CB vectors */
- WebRtc_Word16 *energyShifts, /* (o) Shift value of the energy */
- WebRtc_Word16 scale, /* (i) The scaling of all energy values */
- WebRtc_Word16 base_size /* (i) Index to where the energy values should be stored */
- ) {
- WebRtc_Word16 *ppi, *ppo, *pp;
- WebRtc_Word32 energy, tmp32;
-
- /* Compute the energy and store it in a vector. Also the
- * corresponding shift values are stored. The energy values
- * are reused in all three stages. */
-
- /* Calculate the energy in the first block of 'lTarget' sampels. */
- ppi = CB+lMem-lTarget-1;
- ppo = CB+lMem-1;
-
- pp=CB+lMem-lTarget;
- energy = WebRtcSpl_DotProductWithScale( pp, pp, lTarget, scale);
-
- /* Normalize the energy and store the number of shifts */
- energyShifts[0] = (WebRtc_Word16)WebRtcSpl_NormW32(energy);
- tmp32 = WEBRTC_SPL_LSHIFT_W32(energy, energyShifts[0]);
- energyW16[0] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
-
- /* Compute the energy of the rest of the cb memory
- * by step wise adding and subtracting the next
- * sample and the last sample respectively. */
- WebRtcIlbcfix_CbMemEnergyCalc(energy, range, ppi, ppo, energyW16, energyShifts, scale, 0);
-
- /* Next, precompute the energy values for the filtered cb section */
- energy=0;
- pp=filteredCB+lMem-lTarget;
-
- energy = WebRtcSpl_DotProductWithScale( pp, pp, lTarget, scale);
-
- /* Normalize the energy and store the number of shifts */
- energyShifts[base_size] = (WebRtc_Word16)WebRtcSpl_NormW32(energy);
- tmp32 = WEBRTC_SPL_LSHIFT_W32(energy, energyShifts[base_size]);
- energyW16[base_size] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
-
- ppi = filteredCB + lMem - 1 - lTarget;
- ppo = filteredCB + lMem - 1;
-
- WebRtcIlbcfix_CbMemEnergyCalc(energy, range, ppi, ppo, energyW16, energyShifts, scale, base_size);
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy.h b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
deleted file mode 100644
index 1aa2b7b..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CbMemEnergy.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_H_
-
-void WebRtcIlbcfix_CbMemEnergy(
- WebRtc_Word16 range,
- WebRtc_Word16 *CB, /* (i) The CB memory (1:st section) */
- WebRtc_Word16 *filteredCB, /* (i) The filtered CB memory (2:nd section) */
- WebRtc_Word16 lMem, /* (i) Length of the CB memory */
- WebRtc_Word16 lTarget, /* (i) Length of the target vector */
- WebRtc_Word16 *energyW16, /* (o) Energy in the CB vectors */
- WebRtc_Word16 *energyShifts, /* (o) Shift value of the energy */
- WebRtc_Word16 scale, /* (i) The scaling of all energy values */
- WebRtc_Word16 base_size /* (i) Index to where the energy values should be stored */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
deleted file mode 100644
index 0c6f479..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.c
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CbMemEnergyAugmentation.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-void WebRtcIlbcfix_CbMemEnergyAugmentation(
- WebRtc_Word16 *interpSamples, /* (i) The interpolated samples */
- WebRtc_Word16 *CBmem, /* (i) The CB memory */
- WebRtc_Word16 scale, /* (i) The scaling of all energy values */
- WebRtc_Word16 base_size, /* (i) Index to where the energy values should be stored */
- WebRtc_Word16 *energyW16, /* (o) Energy in the CB vectors */
- WebRtc_Word16 *energyShifts /* (o) Shift value of the energy */
- ){
- WebRtc_Word32 energy, tmp32;
- WebRtc_Word16 *ppe, *pp, *interpSamplesPtr;
- WebRtc_Word16 *CBmemPtr, lagcount;
- WebRtc_Word16 *enPtr=&energyW16[base_size-20];
- WebRtc_Word16 *enShPtr=&energyShifts[base_size-20];
- WebRtc_Word32 nrjRecursive;
-
- CBmemPtr = CBmem+147;
- interpSamplesPtr = interpSamples;
-
- /* Compute the energy for the first (low-5) noninterpolated samples */
- nrjRecursive = WebRtcSpl_DotProductWithScale( CBmemPtr-19, CBmemPtr-19, 15, scale);
- ppe = CBmemPtr - 20;
-
- for (lagcount=20; lagcount<=39; lagcount++) {
-
- /* Update the energy recursively to save complexity */
- nrjRecursive = nrjRecursive +
- WEBRTC_SPL_MUL_16_16_RSFT(*ppe, *ppe, scale);
- ppe--;
- energy = nrjRecursive;
-
- /* interpolation */
- energy += WebRtcSpl_DotProductWithScale(interpSamplesPtr, interpSamplesPtr, 4, scale);
- interpSamplesPtr += 4;
-
- /* Compute energy for the remaining samples */
- pp = CBmemPtr - lagcount;
- energy += WebRtcSpl_DotProductWithScale(pp, pp, SUBL-lagcount, scale);
-
- /* Normalize the energy and store the number of shifts */
- (*enShPtr) = (WebRtc_Word16)WebRtcSpl_NormW32(energy);
- tmp32 = WEBRTC_SPL_LSHIFT_W32(energy, (*enShPtr));
- (*enPtr) = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 16);
- enShPtr++;
- enPtr++;
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
deleted file mode 100644
index 938b87e..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_augmentation.h
+++ /dev/null
@@ -1,31 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CbMemEnergyAugmentation.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_AUGMENTATION_H_
-
-void WebRtcIlbcfix_CbMemEnergyAugmentation(
- WebRtc_Word16 *interpSamples, /* (i) The interpolated samples */
- WebRtc_Word16 *CBmem, /* (i) The CB memory */
- WebRtc_Word16 scale, /* (i) The scaling of all energy values */
- WebRtc_Word16 base_size, /* (i) Index to where the energy values should be stored */
- WebRtc_Word16 *energyW16, /* (o) Energy in the CB vectors */
- WebRtc_Word16 *energyShifts /* (o) Shift value of the energy */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
deleted file mode 100644
index 40bb708..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.c
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CbMemEnergyCalc.c
-
-******************************************************************/
-
-#include "defines.h"
-
-/* Compute the energy of the rest of the cb memory
- * by step wise adding and subtracting the next
- * sample and the last sample respectively */
-void WebRtcIlbcfix_CbMemEnergyCalc(
- WebRtc_Word32 energy, /* (i) input start energy */
- WebRtc_Word16 range, /* (i) number of iterations */
- WebRtc_Word16 *ppi, /* (i) input pointer 1 */
- WebRtc_Word16 *ppo, /* (i) input pointer 2 */
- WebRtc_Word16 *energyW16, /* (o) Energy in the CB vectors */
- WebRtc_Word16 *energyShifts, /* (o) Shift value of the energy */
- WebRtc_Word16 scale, /* (i) The scaling of all energy values */
- WebRtc_Word16 base_size /* (i) Index to where the energy values should be stored */
- )
-{
- WebRtc_Word16 j,shft;
- WebRtc_Word32 tmp;
- WebRtc_Word16 *eSh_ptr;
- WebRtc_Word16 *eW16_ptr;
-
-
- eSh_ptr = &energyShifts[1+base_size];
- eW16_ptr = &energyW16[1+base_size];
-
- for(j=0;j<range-1;j++) {
-
- /* Calculate next energy by a +/-
- operation on the edge samples */
- tmp = WEBRTC_SPL_MUL_16_16(*ppi, *ppi);
- tmp -= WEBRTC_SPL_MUL_16_16(*ppo, *ppo);
- energy += WEBRTC_SPL_RSHIFT_W32(tmp, scale);
- energy = WEBRTC_SPL_MAX(energy, 0);
-
- ppi--;
- ppo--;
-
- /* Normalize the energy into a WebRtc_Word16 and store
- the number of shifts */
-
- shft = (WebRtc_Word16)WebRtcSpl_NormW32(energy);
- *eSh_ptr++ = shft;
-
- tmp = WEBRTC_SPL_LSHIFT_W32(energy, shft);
- *eW16_ptr++ = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp, 16);
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h b/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
deleted file mode 100644
index ee2e285..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/cb_mem_energy_calc.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CbMemEnergyCalc.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_MEM_ENERGY_CALC_H_
-
-void WebRtcIlbcfix_CbMemEnergyCalc(
- WebRtc_Word32 energy, /* (i) input start energy */
- WebRtc_Word16 range, /* (i) number of iterations */
- WebRtc_Word16 *ppi, /* (i) input pointer 1 */
- WebRtc_Word16 *ppo, /* (i) input pointer 2 */
- WebRtc_Word16 *energyW16, /* (o) Energy in the CB vectors */
- WebRtc_Word16 *energyShifts, /* (o) Shift value of the energy */
- WebRtc_Word16 scale, /* (i) The scaling of all energy values */
- WebRtc_Word16 base_size /* (i) Index to where the energy values should be stored */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_search.c b/trunk/src/modules/audio_coding/codecs/ilbc/cb_search.c
deleted file mode 100644
index c51ccf7..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/cb_search.c
+++ /dev/null
@@ -1,396 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CbSearch.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "gain_quant.h"
-#include "filtered_cb_vecs.h"
-#include "constants.h"
-#include "cb_mem_energy.h"
-#include "interpolate_samples.h"
-#include "cb_mem_energy_augmentation.h"
-#include "cb_search_core.h"
-#include "energy_inverse.h"
-#include "augmented_cb_corr.h"
-#include "cb_update_best_index.h"
-#include "create_augmented_vec.h"
-
-/*----------------------------------------------------------------*
- * Search routine for codebook encoding and gain quantization.
- *----------------------------------------------------------------*/
-
-void WebRtcIlbcfix_CbSearch(
- iLBC_Enc_Inst_t *iLBCenc_inst,
- /* (i) the encoder state structure */
- WebRtc_Word16 *index, /* (o) Codebook indices */
- WebRtc_Word16 *gain_index, /* (o) Gain quantization indices */
- WebRtc_Word16 *intarget, /* (i) Target vector for encoding */
- WebRtc_Word16 *decResidual,/* (i) Decoded residual for codebook construction */
- WebRtc_Word16 lMem, /* (i) Length of buffer */
- WebRtc_Word16 lTarget, /* (i) Length of vector */
- WebRtc_Word16 *weightDenum,/* (i) weighting filter coefficients in Q12 */
- WebRtc_Word16 block /* (i) the subblock number */
- ) {
- WebRtc_Word16 i, j, stage, range;
- WebRtc_Word16 *pp, scale, tmp;
- WebRtc_Word16 bits, temp1, temp2;
- WebRtc_Word16 base_size;
- WebRtc_Word32 codedEner, targetEner;
- WebRtc_Word16 gains[CB_NSTAGES+1];
- WebRtc_Word16 *cb_vecPtr;
- WebRtc_Word16 indexOffset, sInd, eInd;
- WebRtc_Word32 CritMax=0;
- WebRtc_Word16 shTotMax=WEBRTC_SPL_WORD16_MIN;
- WebRtc_Word16 bestIndex=0;
- WebRtc_Word16 bestGain=0;
- WebRtc_Word16 indexNew, CritNewSh;
- WebRtc_Word32 CritNew;
- WebRtc_Word32 *cDotPtr;
- WebRtc_Word16 noOfZeros;
- WebRtc_Word16 *gainPtr;
- WebRtc_Word32 t32, tmpW32;
- WebRtc_Word16 *WebRtcIlbcfix_kGainSq5_ptr;
- /* Stack based */
- WebRtc_Word16 CBbuf[CB_MEML+LPC_FILTERORDER+CB_HALFFILTERLEN];
- WebRtc_Word32 cDot[128];
- WebRtc_Word32 Crit[128];
- WebRtc_Word16 targetVec[SUBL+LPC_FILTERORDER];
- WebRtc_Word16 cbvectors[CB_MEML];
- WebRtc_Word16 codedVec[SUBL];
- WebRtc_Word16 interpSamples[20*4];
- WebRtc_Word16 interpSamplesFilt[20*4];
- WebRtc_Word16 energyW16[CB_EXPAND*128];
- WebRtc_Word16 energyShifts[CB_EXPAND*128];
- WebRtc_Word16 *inverseEnergy=energyW16; /* Reuse memory */
- WebRtc_Word16 *inverseEnergyShifts=energyShifts; /* Reuse memory */
- WebRtc_Word16 *buf = &CBbuf[LPC_FILTERORDER];
- WebRtc_Word16 *target = &targetVec[LPC_FILTERORDER];
- WebRtc_Word16 *aug_vec = (WebRtc_Word16*)cDot; /* length [SUBL], reuse memory */
-
- /* Determine size of codebook sections */
-
- base_size=lMem-lTarget+1;
- if (lTarget==SUBL) {
- base_size=lMem-19;
- }
-
- /* weighting of the CB memory */
- noOfZeros=lMem-WebRtcIlbcfix_kFilterRange[block];
- WebRtcSpl_MemSetW16(&buf[-LPC_FILTERORDER], 0, noOfZeros+LPC_FILTERORDER);
- WebRtcSpl_FilterARFastQ12(
- decResidual+noOfZeros, buf+noOfZeros,
- weightDenum, LPC_FILTERORDER+1, WebRtcIlbcfix_kFilterRange[block]);
-
- /* weighting of the target vector */
- WEBRTC_SPL_MEMCPY_W16(&target[-LPC_FILTERORDER], buf+noOfZeros+WebRtcIlbcfix_kFilterRange[block]-LPC_FILTERORDER, LPC_FILTERORDER);
- WebRtcSpl_FilterARFastQ12(
- intarget, target,
- weightDenum, LPC_FILTERORDER+1, lTarget);
-
- /* Store target, towards the end codedVec is calculated as
- the initial target minus the remaining target */
- WEBRTC_SPL_MEMCPY_W16(codedVec, target, lTarget);
-
- /* Find the highest absolute value to calculate proper
- vector scale factor (so that it uses 12 bits) */
- temp1 = WebRtcSpl_MaxAbsValueW16(buf, (WebRtc_Word16)lMem);
- temp2 = WebRtcSpl_MaxAbsValueW16(target, (WebRtc_Word16)lTarget);
-
- if ((temp1>0)&&(temp2>0)) {
- temp1 = WEBRTC_SPL_MAX(temp1, temp2);
- scale = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_MUL_16_16(temp1, temp1));
- } else {
- /* temp1 or temp2 is negative (maximum was -32768) */
- scale = 30;
- }
-
- /* Scale to so that a mul-add 40 times does not overflow */
- scale = scale - 25;
- scale = WEBRTC_SPL_MAX(0, scale);
-
- /* Compute energy of the original target */
- targetEner = WebRtcSpl_DotProductWithScale(target, target, lTarget, scale);
-
- /* Prepare search over one more codebook section. This section
- is created by filtering the original buffer with a filter. */
- WebRtcIlbcfix_FilteredCbVecs(cbvectors, buf, lMem, WebRtcIlbcfix_kFilterRange[block]);
-
- range = WebRtcIlbcfix_kSearchRange[block][0];
-
- if(lTarget == SUBL) {
- /* Create the interpolated samples and store them for use in all stages */
-
- /* First section, non-filtered half of the cb */
- WebRtcIlbcfix_InterpolateSamples(interpSamples, buf, lMem);
-
- /* Second section, filtered half of the cb */
- WebRtcIlbcfix_InterpolateSamples(interpSamplesFilt, cbvectors, lMem);
-
- /* Compute the CB vectors' energies for the first cb section (non-filtered) */
- WebRtcIlbcfix_CbMemEnergyAugmentation(interpSamples, buf,
- scale, 20, energyW16, energyShifts);
-
- /* Compute the CB vectors' energies for the second cb section (filtered cb) */
- WebRtcIlbcfix_CbMemEnergyAugmentation(interpSamplesFilt, cbvectors,
- scale, (WebRtc_Word16)(base_size+20), energyW16, energyShifts);
-
- /* Compute the CB vectors' energies and store them in the vector
- * energyW16. Also the corresponding shift values are stored. The
- * energy values are used in all three stages. */
- WebRtcIlbcfix_CbMemEnergy(range, buf, cbvectors, lMem,
- lTarget, energyW16+20, energyShifts+20, scale, base_size);
-
- } else {
- /* Compute the CB vectors' energies and store them in the vector
- * energyW16. Also the corresponding shift values are stored. The
- * energy values are used in all three stages. */
- WebRtcIlbcfix_CbMemEnergy(range, buf, cbvectors, lMem,
- lTarget, energyW16, energyShifts, scale, base_size);
-
- /* Set the energy positions 58-63 and 122-127 to zero
- (otherwise they are uninitialized) */
- WebRtcSpl_MemSetW16(energyW16+range, 0, (base_size-range));
- WebRtcSpl_MemSetW16(energyW16+range+base_size, 0, (base_size-range));
- }
-
- /* Calculate Inverse Energy (energyW16 is already normalized
- and will contain the inverse energy in Q29 after this call */
- WebRtcIlbcfix_EnergyInverse(energyW16, base_size*CB_EXPAND);
-
- /* The gain value computed in the previous stage is used
- * as an upper limit to what the next stage gain value
- * is allowed to be. In stage 0, 16384 (1.0 in Q14) is used as
- * the upper limit. */
- gains[0] = 16384;
-
- for (stage=0; stage<CB_NSTAGES; stage++) {
-
- /* Set up memories */
- range = WebRtcIlbcfix_kSearchRange[block][stage];
-
- /* initialize search measures */
- CritMax=0;
- shTotMax=-100;
- bestIndex=0;
- bestGain=0;
-
- /* loop over lags 40+ in the first codebook section, full search */
- cb_vecPtr = buf+lMem-lTarget;
-
- /* Calculate all the cross correlations (augmented part of CB) */
- if (lTarget==SUBL) {
- WebRtcIlbcfix_AugmentedCbCorr(target, buf+lMem,
- interpSamples, cDot,
- 20, 39, scale);
- cDotPtr=&cDot[20];
- } else {
- cDotPtr=cDot;
- }
- /* Calculate all the cross correlations (main part of CB) */
- WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, range, scale, -1);
-
- /* Adjust the search range for the augmented vectors */
- if (lTarget==SUBL) {
- range=WebRtcIlbcfix_kSearchRange[block][stage]+20;
- } else {
- range=WebRtcIlbcfix_kSearchRange[block][stage];
- }
-
- indexOffset=0;
-
- /* Search for best index in this part of the vector */
- WebRtcIlbcfix_CbSearchCore(
- cDot, range, stage, inverseEnergy,
- inverseEnergyShifts, Crit,
- &indexNew, &CritNew, &CritNewSh);
-
- /* Update the global best index and the corresponding gain */
- WebRtcIlbcfix_CbUpdateBestIndex(
- CritNew, CritNewSh, (WebRtc_Word16)(indexNew+indexOffset), cDot[indexNew+indexOffset],
- inverseEnergy[indexNew+indexOffset], inverseEnergyShifts[indexNew+indexOffset],
- &CritMax, &shTotMax, &bestIndex, &bestGain);
-
- sInd=bestIndex-(WebRtc_Word16)(CB_RESRANGE>>1);
- eInd=sInd+CB_RESRANGE;
- if (sInd<0) {
- eInd-=sInd;
- sInd=0;
- }
- if (eInd>=range) {
- eInd=range-1;
- sInd=eInd-CB_RESRANGE;
- }
-
- range = WebRtcIlbcfix_kSearchRange[block][stage];
-
- if (lTarget==SUBL) {
- i=sInd;
- if (sInd<20) {
- WebRtcIlbcfix_AugmentedCbCorr(target, cbvectors+lMem,
- interpSamplesFilt, cDot,
- (WebRtc_Word16)(sInd+20), (WebRtc_Word16)(WEBRTC_SPL_MIN(39, (eInd+20))), scale);
- i=20;
- }
-
- cDotPtr=&cDot[WEBRTC_SPL_MAX(0,(20-sInd))];
- cb_vecPtr = cbvectors+lMem-20-i;
-
- /* Calculate the cross correlations (main part of the filtered CB) */
- WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, (WebRtc_Word16)(eInd-i+1), scale, -1);
-
- } else {
- cDotPtr = cDot;
- cb_vecPtr = cbvectors+lMem-lTarget-sInd;
-
- /* Calculate the cross correlations (main part of the filtered CB) */
- WebRtcSpl_CrossCorrelation(cDotPtr, target, cb_vecPtr, lTarget, (WebRtc_Word16)(eInd-sInd+1), scale, -1);
-
- }
-
- /* Adjust the search range for the augmented vectors */
- indexOffset=base_size+sInd;
-
- /* Search for best index in this part of the vector */
- WebRtcIlbcfix_CbSearchCore(
- cDot, (WebRtc_Word16)(eInd-sInd+1), stage, inverseEnergy+indexOffset,
- inverseEnergyShifts+indexOffset, Crit,
- &indexNew, &CritNew, &CritNewSh);
-
- /* Update the global best index and the corresponding gain */
- WebRtcIlbcfix_CbUpdateBestIndex(
- CritNew, CritNewSh, (WebRtc_Word16)(indexNew+indexOffset), cDot[indexNew],
- inverseEnergy[indexNew+indexOffset], inverseEnergyShifts[indexNew+indexOffset],
- &CritMax, &shTotMax, &bestIndex, &bestGain);
-
- index[stage] = bestIndex;
-
-
- bestGain = WebRtcIlbcfix_GainQuant(bestGain,
- (WebRtc_Word16)WEBRTC_SPL_ABS_W16(gains[stage]), stage, &gain_index[stage]);
-
- /* Extract the best (according to measure) codebook vector
- Also adjust the index, so that the augmented vectors are last.
- Above these vectors were first...
- */
-
- if(lTarget==(STATE_LEN-iLBCenc_inst->state_short_len)) {
-
- if(index[stage]<base_size) {
- pp=buf+lMem-lTarget-index[stage];
- } else {
- pp=cbvectors+lMem-lTarget-
- index[stage]+base_size;
- }
-
- } else {
-
- if (index[stage]<base_size) {
- if (index[stage]>=20) {
- /* Adjust index and extract vector */
- index[stage]-=20;
- pp=buf+lMem-lTarget-index[stage];
- } else {
- /* Adjust index and extract vector */
- index[stage]+=(base_size-20);
-
- WebRtcIlbcfix_CreateAugmentedVec((WebRtc_Word16)(index[stage]-base_size+40),
- buf+lMem, aug_vec);
- pp = aug_vec;
-
- }
- } else {
-
- if ((index[stage] - base_size) >= 20) {
- /* Adjust index and extract vector */
- index[stage]-=20;
- pp=cbvectors+lMem-lTarget-
- index[stage]+base_size;
- } else {
- /* Adjust index and extract vector */
- index[stage]+=(base_size-20);
- WebRtcIlbcfix_CreateAugmentedVec((WebRtc_Word16)(index[stage]-2*base_size+40),
- cbvectors+lMem, aug_vec);
- pp = aug_vec;
- }
- }
- }
-
- /* Subtract the best codebook vector, according
- to measure, from the target vector */
-
- WebRtcSpl_AddAffineVectorToVector(target, pp, (WebRtc_Word16)(-bestGain), (WebRtc_Word32)8192, (WebRtc_Word16)14, (int)lTarget);
-
- /* record quantized gain */
- gains[stage+1] = bestGain;
-
- } /* end of Main Loop. for (stage=0;... */
-
- /* Calculte the coded vector (original target - what's left) */
- for (i=0;i<lTarget;i++) {
- codedVec[i]-=target[i];
- }
-
- /* Gain adjustment for energy matching */
- codedEner = WebRtcSpl_DotProductWithScale(codedVec, codedVec, lTarget, scale);
-
- j=gain_index[0];
-
- temp1 = (WebRtc_Word16)WebRtcSpl_NormW32(codedEner);
- temp2 = (WebRtc_Word16)WebRtcSpl_NormW32(targetEner);
-
- if(temp1 < temp2) {
- bits = 16 - temp1;
- } else {
- bits = 16 - temp2;
- }
-
- tmp = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(gains[1],gains[1], 14);
-
- targetEner = WEBRTC_SPL_MUL_16_16(
- WEBRTC_SPL_SHIFT_W32(targetEner, -bits), tmp);
-
- tmpW32 = ((WebRtc_Word32)(gains[1]-1))<<1;
-
- /* Pointer to the table that contains
- gain_sq5TblFIX * gain_sq5TblFIX in Q14 */
- gainPtr=(WebRtc_Word16*)WebRtcIlbcfix_kGainSq5Sq+gain_index[0];
- temp1 = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(codedEner, -bits);
-
- WebRtcIlbcfix_kGainSq5_ptr = (WebRtc_Word16*)&WebRtcIlbcfix_kGainSq5[j];
-
- /* targetEner and codedEner are in Q(-2*scale) */
- for (i=gain_index[0];i<32;i++) {
-
- /* Change the index if
- (codedEnergy*gainTbl[i]*gainTbl[i])<(targetEn*gain[0]*gain[0]) AND
- gainTbl[i] < 2*gain[0]
- */
-
- t32 = WEBRTC_SPL_MUL_16_16(temp1, (*gainPtr));
- t32 = t32 - targetEner;
- if (t32 < 0) {
- if ((*WebRtcIlbcfix_kGainSq5_ptr) < tmpW32) {
- j=i;
- WebRtcIlbcfix_kGainSq5_ptr = (WebRtc_Word16*)&WebRtcIlbcfix_kGainSq5[i];
- }
- }
- gainPtr++;
- }
- gain_index[0]=j;
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_search.h b/trunk/src/modules/audio_coding/codecs/ilbc/cb_search.h
deleted file mode 100644
index e4ad4b5..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/cb_search.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CbSearch.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_H_
-
-void WebRtcIlbcfix_CbSearch(
- iLBC_Enc_Inst_t *iLBCenc_inst,
- /* (i) the encoder state structure */
- WebRtc_Word16 *index, /* (o) Codebook indices */
- WebRtc_Word16 *gain_index, /* (o) Gain quantization indices */
- WebRtc_Word16 *intarget, /* (i) Target vector for encoding */
- WebRtc_Word16 *decResidual,/* (i) Decoded residual for codebook construction */
- WebRtc_Word16 lMem, /* (i) Length of buffer */
- WebRtc_Word16 lTarget, /* (i) Length of vector */
- WebRtc_Word16 *weightDenum,/* (i) weighting filter coefficients in Q12 */
- WebRtc_Word16 block /* (i) the subblock number */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_search_core.c b/trunk/src/modules/audio_coding/codecs/ilbc/cb_search_core.c
deleted file mode 100644
index 711e2df..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/cb_search_core.c
+++ /dev/null
@@ -1,113 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CbSearchCore.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-void WebRtcIlbcfix_CbSearchCore(
- WebRtc_Word32 *cDot, /* (i) Cross Correlation */
- WebRtc_Word16 range, /* (i) Search range */
- WebRtc_Word16 stage, /* (i) Stage of this search */
- WebRtc_Word16 *inverseEnergy, /* (i) Inversed energy */
- WebRtc_Word16 *inverseEnergyShift, /* (i) Shifts of inversed energy
- with the offset 2*16-29 */
- WebRtc_Word32 *Crit, /* (o) The criteria */
- WebRtc_Word16 *bestIndex, /* (o) Index that corresponds to
- maximum criteria (in this
- vector) */
- WebRtc_Word32 *bestCrit, /* (o) Value of critera for the
- chosen index */
- WebRtc_Word16 *bestCritSh) /* (o) The domain of the chosen
- criteria */
-{
- WebRtc_Word32 maxW32, tmp32;
- WebRtc_Word16 max, sh, tmp16;
- int i;
- WebRtc_Word32 *cDotPtr;
- WebRtc_Word16 cDotSqW16;
- WebRtc_Word16 *inverseEnergyPtr;
- WebRtc_Word32 *critPtr;
- WebRtc_Word16 *inverseEnergyShiftPtr;
-
- /* Don't allow negative values for stage 0 */
- if (stage==0) {
- cDotPtr=cDot;
- for (i=0;i<range;i++) {
- *cDotPtr=WEBRTC_SPL_MAX(0, (*cDotPtr));
- cDotPtr++;
- }
- }
-
- /* Normalize cDot to WebRtc_Word16, calculate the square of cDot and store the upper WebRtc_Word16 */
- maxW32 = WebRtcSpl_MaxAbsValueW32(cDot, range);
-
- sh = (WebRtc_Word16)WebRtcSpl_NormW32(maxW32);
- cDotPtr = cDot;
- inverseEnergyPtr = inverseEnergy;
- critPtr = Crit;
- inverseEnergyShiftPtr=inverseEnergyShift;
- max=WEBRTC_SPL_WORD16_MIN;
-
- for (i=0;i<range;i++) {
- /* Calculate cDot*cDot and put the result in a WebRtc_Word16 */
- tmp32 = WEBRTC_SPL_LSHIFT_W32(*cDotPtr,sh);
- tmp16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32,16);
- cDotSqW16 = (WebRtc_Word16)(((WebRtc_Word32)(tmp16)*(tmp16))>>16);
-
- /* Calculate the criteria (cDot*cDot/energy) */
- *critPtr=WEBRTC_SPL_MUL_16_16(cDotSqW16, (*inverseEnergyPtr));
-
- /* Extract the maximum shift value under the constraint
- that the criteria is not zero */
- if ((*critPtr)!=0) {
- max = WEBRTC_SPL_MAX((*inverseEnergyShiftPtr), max);
- }
-
- inverseEnergyPtr++;
- inverseEnergyShiftPtr++;
- critPtr++;
- cDotPtr++;
- }
-
- /* If no max shifts still at initialization value, set shift to zero */
- if (max==WEBRTC_SPL_WORD16_MIN) {
- max = 0;
- }
-
- /* Modify the criterias, so that all of them use the same Q domain */
- critPtr=Crit;
- inverseEnergyShiftPtr=inverseEnergyShift;
- for (i=0;i<range;i++) {
- /* Guarantee that the shift value is less than 16
- in order to simplify for DSP's (and guard against >31) */
- tmp16 = WEBRTC_SPL_MIN(16, max-(*inverseEnergyShiftPtr));
-
- (*critPtr)=WEBRTC_SPL_SHIFT_W32((*critPtr),-tmp16);
- critPtr++;
- inverseEnergyShiftPtr++;
- }
-
- /* Find the index of the best value */
- *bestIndex = WebRtcSpl_MaxIndexW32(Crit, range);
- *bestCrit = Crit[*bestIndex];
-
- /* Calculate total shifts of this criteria */
- *bestCritSh = 32 - 2*sh + max;
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_search_core.h b/trunk/src/modules/audio_coding/codecs/ilbc/cb_search_core.h
deleted file mode 100644
index e074c52..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/cb_search_core.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CbSearchCore.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_SEARCH_CORE_H_
-
-#include "defines.h"
-
-void WebRtcIlbcfix_CbSearchCore(
- WebRtc_Word32 *cDot, /* (i) Cross Correlation */
- WebRtc_Word16 range, /* (i) Search range */
- WebRtc_Word16 stage, /* (i) Stage of this search */
- WebRtc_Word16 *inverseEnergy, /* (i) Inversed energy */
- WebRtc_Word16 *inverseEnergyShift, /* (i) Shifts of inversed energy
- with the offset 2*16-29 */
- WebRtc_Word32 *Crit, /* (o) The criteria */
- WebRtc_Word16 *bestIndex, /* (o) Index that corresponds to
- maximum criteria (in this
- vector) */
- WebRtc_Word32 *bestCrit, /* (o) Value of critera for the
- chosen index */
- WebRtc_Word16 *bestCritSh); /* (o) The domain of the chosen
- criteria */
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_update_best_index.c b/trunk/src/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
deleted file mode 100644
index bf85408..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/cb_update_best_index.c
+++ /dev/null
@@ -1,89 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CbUpdateBestIndex.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "cb_update_best_index.h"
-#include "constants.h"
-
-void WebRtcIlbcfix_CbUpdateBestIndex(
- WebRtc_Word32 CritNew, /* (i) New Potentially best Criteria */
- WebRtc_Word16 CritNewSh, /* (i) Shift value of above Criteria */
- WebRtc_Word16 IndexNew, /* (i) Index of new Criteria */
- WebRtc_Word32 cDotNew, /* (i) Cross dot of new index */
- WebRtc_Word16 invEnergyNew, /* (i) Inversed energy new index */
- WebRtc_Word16 energyShiftNew, /* (i) Energy shifts of new index */
- WebRtc_Word32 *CritMax, /* (i/o) Maximum Criteria (so far) */
- WebRtc_Word16 *shTotMax, /* (i/o) Shifts of maximum criteria */
- WebRtc_Word16 *bestIndex, /* (i/o) Index that corresponds to
- maximum criteria */
- WebRtc_Word16 *bestGain) /* (i/o) Gain in Q14 that corresponds
- to maximum criteria */
-{
- WebRtc_Word16 shOld, shNew, tmp16;
- WebRtc_Word16 scaleTmp;
- WebRtc_Word32 gainW32;
-
- /* Normalize the new and old Criteria to the same domain */
- if (CritNewSh>(*shTotMax)) {
- shOld=WEBRTC_SPL_MIN(31,CritNewSh-(*shTotMax));
- shNew=0;
- } else {
- shOld=0;
- shNew=WEBRTC_SPL_MIN(31,(*shTotMax)-CritNewSh);
- }
-
- /* Compare the two criterias. If the new one is better,
- calculate the gain and store this index as the new best one
- */
-
- if (WEBRTC_SPL_RSHIFT_W32(CritNew, shNew)>
- WEBRTC_SPL_RSHIFT_W32((*CritMax),shOld)) {
-
- tmp16 = (WebRtc_Word16)WebRtcSpl_NormW32(cDotNew);
- tmp16 = 16 - tmp16;
-
- /* Calculate the gain in Q14
- Compensate for inverseEnergyshift in Q29 and that the energy
- value was stored in a WebRtc_Word16 (shifted down 16 steps)
- => 29-14+16 = 31 */
-
- scaleTmp = -energyShiftNew-tmp16+31;
- scaleTmp = WEBRTC_SPL_MIN(31, scaleTmp);
-
- gainW32 = WEBRTC_SPL_MUL_16_16_RSFT(
- ((WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(cDotNew, -tmp16)), invEnergyNew, scaleTmp);
-
- /* Check if criteria satisfies Gain criteria (max 1.3)
- if it is larger set the gain to 1.3
- (slightly different from FLP version)
- */
- if (gainW32>21299) {
- *bestGain=21299;
- } else if (gainW32<-21299) {
- *bestGain=-21299;
- } else {
- *bestGain=(WebRtc_Word16)gainW32;
- }
-
- *CritMax=CritNew;
- *shTotMax=CritNewSh;
- *bestIndex = IndexNew;
- }
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/cb_update_best_index.h b/trunk/src/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
deleted file mode 100644
index 9015187..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/cb_update_best_index.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CbUpdateBestIndex.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CB_UPDATE_BEST_INDEX_H_
-
-#include "defines.h"
-
-void WebRtcIlbcfix_CbUpdateBestIndex(
- WebRtc_Word32 CritNew, /* (i) New Potentially best Criteria */
- WebRtc_Word16 CritNewSh, /* (i) Shift value of above Criteria */
- WebRtc_Word16 IndexNew, /* (i) Index of new Criteria */
- WebRtc_Word32 cDotNew, /* (i) Cross dot of new index */
- WebRtc_Word16 invEnergyNew, /* (i) Inversed energy new index */
- WebRtc_Word16 energyShiftNew, /* (i) Energy shifts of new index */
- WebRtc_Word32 *CritMax, /* (i/o) Maximum Criteria (so far) */
- WebRtc_Word16 *shTotMax, /* (i/o) Shifts of maximum criteria */
- WebRtc_Word16 *bestIndex, /* (i/o) Index that corresponds to
- maximum criteria */
- WebRtc_Word16 *bestGain); /* (i/o) Gain in Q14 that corresponds
- to maximum criteria */
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/chebyshev.c b/trunk/src/modules/audio_coding/codecs/ilbc/chebyshev.c
deleted file mode 100644
index 90108ff..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/chebyshev.c
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Chebyshev.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-/*------------------------------------------------------------------*
- * Calculate the Chevyshev polynomial series
- * F(w) = 2*exp(-j5w)*C(x)
- * C(x) = (T_0(x) + f(1)T_1(x) + ... + f(4)T_1(x) + f(5)/2)
- * T_i(x) is the i:th order Chebyshev polynomial
- *------------------------------------------------------------------*/
-
-WebRtc_Word16 WebRtcIlbcfix_Chebyshev(
- /* (o) Result of C(x) */
- WebRtc_Word16 x, /* (i) Value to the Chevyshev polynomial */
- WebRtc_Word16 *f /* (i) The coefficients in the polynomial */
- ) {
- WebRtc_Word16 b1_high, b1_low; /* Use the high, low format to increase the accuracy */
- WebRtc_Word32 b2;
- WebRtc_Word32 tmp1W32;
- WebRtc_Word32 tmp2W32;
- int i;
-
- b2 = (WebRtc_Word32)0x1000000; /* b2 = 1.0 (Q23) */
- /* Calculate b1 = 2*x + f[1] */
- tmp1W32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)x, 10);
- tmp1W32 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)f[1], 14);
-
- for (i = 2; i < 5; i++) {
- tmp2W32 = tmp1W32;
-
- /* Split b1 (in tmp1W32) into a high and low part */
- b1_high = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp1W32, 16);
- b1_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp1W32-WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)b1_high),16), 1);
-
- /* Calculate 2*x*b1-b2+f[i] */
- tmp1W32 = WEBRTC_SPL_LSHIFT_W32( (WEBRTC_SPL_MUL_16_16(b1_high, x) +
- WEBRTC_SPL_MUL_16_16_RSFT(b1_low, x, 15)), 2);
-
- tmp1W32 -= b2;
- tmp1W32 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)f[i], 14);
-
- /* Update b2 for next round */
- b2 = tmp2W32;
- }
-
- /* Split b1 (in tmp1W32) into a high and low part */
- b1_high = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp1W32, 16);
- b1_low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp1W32-WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)b1_high),16), 1);
-
- /* tmp1W32 = x*b1 - b2 + f[i]/2 */
- tmp1W32 = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(b1_high, x), 1) +
- WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16_RSFT(b1_low, x, 15), 1);
-
- tmp1W32 -= b2;
- tmp1W32 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)f[i], 13);
-
- /* Handle overflows and set to maximum or minimum WebRtc_Word16 instead */
- if (tmp1W32>((WebRtc_Word32)33553408)) {
- return(WEBRTC_SPL_WORD16_MAX);
- } else if (tmp1W32<((WebRtc_Word32)-33554432)) {
- return(WEBRTC_SPL_WORD16_MIN);
- } else {
- return((WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp1W32, 10));
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/chebyshev.h b/trunk/src/modules/audio_coding/codecs/ilbc/chebyshev.h
deleted file mode 100644
index 57aab99..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/chebyshev.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Chebyshev.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CHEBYSHEV_H_
-
-#include "defines.h"
-
-/*------------------------------------------------------------------*
- * Calculate the Chevyshev polynomial series
- * F(w) = 2*exp(-j5w)*C(x)
- * C(x) = (T_0(x) + f(1)T_1(x) + ... + f(4)T_1(x) + f(5)/2)
- * T_i(x) is the i:th order Chebyshev polynomial
- *------------------------------------------------------------------*/
-
-WebRtc_Word16 WebRtcIlbcfix_Chebyshev(
- /* (o) Result of C(x) */
- WebRtc_Word16 x, /* (i) Value to the Chevyshev polynomial */
- WebRtc_Word16 *f /* (i) The coefficients in the polynomial */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/comp_corr.c b/trunk/src/modules/audio_coding/codecs/ilbc/comp_corr.c
deleted file mode 100644
index 3d7f93e..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/comp_corr.c
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CompCorr.c
-
-******************************************************************/
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Compute cross correlation and pitch gain for pitch prediction
- * of last subframe at given lag.
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_CompCorr(
- WebRtc_Word32 *corr, /* (o) cross correlation */
- WebRtc_Word32 *ener, /* (o) energy */
- WebRtc_Word16 *buffer, /* (i) signal buffer */
- WebRtc_Word16 lag, /* (i) pitch lag */
- WebRtc_Word16 bLen, /* (i) length of buffer */
- WebRtc_Word16 sRange, /* (i) correlation search length */
- WebRtc_Word16 scale /* (i) number of rightshifts to use */
- ){
- WebRtc_Word16 *w16ptr;
-
- w16ptr=&buffer[bLen-sRange-lag];
-
- /* Calculate correlation and energy */
- (*corr)=WebRtcSpl_DotProductWithScale(&buffer[bLen-sRange], w16ptr, sRange, scale);
- (*ener)=WebRtcSpl_DotProductWithScale(w16ptr, w16ptr, sRange, scale);
-
- /* For zero energy set the energy to 0 in order to avoid potential
- problems for coming divisions */
- if (*ener == 0) {
- *corr = 0;
- *ener = 1;
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/comp_corr.h b/trunk/src/modules/audio_coding/codecs/ilbc/comp_corr.h
deleted file mode 100644
index cd46532..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/comp_corr.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CompCorr.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_COMP_CORR_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Compute cross correlation and pitch gain for pitch prediction
- * of last subframe at given lag.
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_CompCorr(
- WebRtc_Word32 *corr, /* (o) cross correlation */
- WebRtc_Word32 *ener, /* (o) energy */
- WebRtc_Word16 *buffer, /* (i) signal buffer */
- WebRtc_Word16 lag, /* (i) pitch lag */
- WebRtc_Word16 bLen, /* (i) length of buffer */
- WebRtc_Word16 sRange, /* (i) correlation search length */
- WebRtc_Word16 scale /* (i) number of rightshifts to use */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/complexityMeasures.m b/trunk/src/modules/audio_coding/codecs/ilbc/complexityMeasures.m
deleted file mode 100644
index f768194..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/complexityMeasures.m
+++ /dev/null
@@ -1,49 +0,0 @@
-clear;
-pack;
-%
-% Enter the path to YOUR executable and remember to define the perprocessor
-% variable PRINT_MIPS te get the instructions printed to the screen.
-%
-command = '!iLBCtest.exe 30 speechAndBGnoise.pcm out1.bit out1.pcm tlm10_30ms.dat';
-cout=' > st.txt'; %saves to matlab variable 'st'
-eval(strcat(command,cout));
-if(length(cout)>3)
- load st.txt
-else
- disp('No cout file to load')
-end
-
-% initialize vector to zero
-index = find(st(1:end,1)==-1);
-indexnonzero = find(st(1:end,1)>0);
-frames = length(index)-indexnonzero(1)+1;
-start = indexnonzero(1) - 1;
-functionOrder=max(st(:,2));
-new=zeros(frames,functionOrder);
-
-for i = 1:frames,
- for j = index(start-1+i)+1:(index(start+i)-1),
- new(i,st(j,2)) = new(i,st(j,2)) + st(j,1);
- end
-end
-
-result=zeros(functionOrder,3);
-for i=1:functionOrder
- nonzeroelements = find(new(1:end,i)>0);
- result(i,1)=i;
-
- % Compute each function's mean complexity
- % result(i,2)=(sum(new(nonzeroelements,i))/(length(nonzeroelements)*0.03))/1000000;
-
- % Compute each function's maximum complexity in encoding
- % and decoding respectively and then add it together:
- % result(i,3)=(max(new(1:end,i))/0.03)/1000000;
- result(i,3)=(max(new(1:size(new,1)/2,i))/0.03)/1000000 + (max(new(size(new,1)/2+1:end,i))/0.03)/1000000;
-end
-
-result
-
-% Compute maximum complexity for a single frame (enc/dec separately and together)
-maxEncComplexityInAFrame = (max(sum(new(1:size(new,1)/2,:),2))/0.03)/1000000
-maxDecComplexityInAFrame = (max(sum(new(size(new,1)/2+1:end,:),2))/0.03)/1000000
-totalComplexity = maxEncComplexityInAFrame + maxDecComplexityInAFrame
\ No newline at end of file
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/constants.c b/trunk/src/modules/audio_coding/codecs/ilbc/constants.c
deleted file mode 100644
index 5ebe9be..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/constants.c
+++ /dev/null
@@ -1,666 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- constants.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-/* HP Filters {b[0] b[1] b[2] -a[1] -a[2]} */
-
-const WebRtc_Word16 WebRtcIlbcfix_kHpInCoefs[5] = {3798, -7596, 3798, 7807, -3733};
-const WebRtc_Word16 WebRtcIlbcfix_kHpOutCoefs[5] = {3849, -7699, 3849, 7918, -3833};
-
-/* Window in Q11 to window the energies of the 5 choises (3 for 20ms) in the choise for
- the 80 sample start state
-*/
-const WebRtc_Word16 WebRtcIlbcfix_kStartSequenceEnrgWin[NSUB_MAX-1]= {
- 1638, 1843, 2048, 1843, 1638
-};
-
-/* LP Filter coeffs used for downsampling */
-const WebRtc_Word16 WebRtcIlbcfix_kLpFiltCoefs[FILTERORDER_DS_PLUS1]= {
- -273, 512, 1297, 1696, 1297, 512, -273
-};
-
-/* Constants used in the LPC calculations */
-
-/* Hanning LPC window (in Q15) */
-const WebRtc_Word16 WebRtcIlbcfix_kLpcWin[BLOCKL_MAX] = {
- 6, 22, 50, 89, 139, 200, 272, 355, 449, 554, 669, 795,
- 932, 1079, 1237, 1405, 1583, 1771, 1969, 2177, 2395, 2622, 2858, 3104,
- 3359, 3622, 3894, 4175, 4464, 4761, 5066, 5379, 5699, 6026, 6361, 6702,
- 7050, 7404, 7764, 8130, 8502, 8879, 9262, 9649, 10040, 10436, 10836, 11240,
- 11647, 12058, 12471, 12887, 13306, 13726, 14148, 14572, 14997, 15423, 15850, 16277,
- 16704, 17131, 17558, 17983, 18408, 18831, 19252, 19672, 20089, 20504, 20916, 21325,
- 21730, 22132, 22530, 22924, 23314, 23698, 24078, 24452, 24821, 25185, 25542, 25893,
- 26238, 26575, 26906, 27230, 27547, 27855, 28156, 28450, 28734, 29011, 29279, 29538,
- 29788, 30029, 30261, 30483, 30696, 30899, 31092, 31275, 31448, 31611, 31764, 31906,
- 32037, 32158, 32268, 32367, 32456, 32533, 32600, 32655, 32700, 32733, 32755, 32767,
- 32767, 32755, 32733, 32700, 32655, 32600, 32533, 32456, 32367, 32268, 32158, 32037,
- 31906, 31764, 31611, 31448, 31275, 31092, 30899, 30696, 30483, 30261, 30029, 29788,
- 29538, 29279, 29011, 28734, 28450, 28156, 27855, 27547, 27230, 26906, 26575, 26238,
- 25893, 25542, 25185, 24821, 24452, 24078, 23698, 23314, 22924, 22530, 22132, 21730,
- 21325, 20916, 20504, 20089, 19672, 19252, 18831, 18408, 17983, 17558, 17131, 16704,
- 16277, 15850, 15423, 14997, 14572, 14148, 13726, 13306, 12887, 12471, 12058, 11647,
- 11240, 10836, 10436, 10040, 9649, 9262, 8879, 8502, 8130, 7764, 7404, 7050,
- 6702, 6361, 6026, 5699, 5379, 5066, 4761, 4464, 4175, 3894, 3622, 3359,
- 3104, 2858, 2622, 2395, 2177, 1969, 1771, 1583, 1405, 1237, 1079, 932,
- 795, 669, 554, 449, 355, 272, 200, 139, 89, 50, 22, 6
-};
-
-/* Asymmetric LPC window (in Q15)*/
-const WebRtc_Word16 WebRtcIlbcfix_kLpcAsymWin[BLOCKL_MAX] = {
- 2, 7, 15, 27, 42, 60, 81, 106, 135, 166, 201, 239,
- 280, 325, 373, 424, 478, 536, 597, 661, 728, 798, 872, 949,
- 1028, 1111, 1197, 1287, 1379, 1474, 1572, 1674, 1778, 1885, 1995, 2108,
- 2224, 2343, 2465, 2589, 2717, 2847, 2980, 3115, 3254, 3395, 3538, 3684,
- 3833, 3984, 4138, 4295, 4453, 4615, 4778, 4944, 5112, 5283, 5456, 5631,
- 5808, 5987, 6169, 6352, 6538, 6725, 6915, 7106, 7300, 7495, 7692, 7891,
- 8091, 8293, 8497, 8702, 8909, 9118, 9328, 9539, 9752, 9966, 10182, 10398,
- 10616, 10835, 11055, 11277, 11499, 11722, 11947, 12172, 12398, 12625, 12852, 13080,
- 13309, 13539, 13769, 14000, 14231, 14463, 14695, 14927, 15160, 15393, 15626, 15859,
- 16092, 16326, 16559, 16792, 17026, 17259, 17492, 17725, 17957, 18189, 18421, 18653,
- 18884, 19114, 19344, 19573, 19802, 20030, 20257, 20483, 20709, 20934, 21157, 21380,
- 21602, 21823, 22042, 22261, 22478, 22694, 22909, 23123, 23335, 23545, 23755, 23962,
- 24168, 24373, 24576, 24777, 24977, 25175, 25371, 25565, 25758, 25948, 26137, 26323,
- 26508, 26690, 26871, 27049, 27225, 27399, 27571, 27740, 27907, 28072, 28234, 28394,
- 28552, 28707, 28860, 29010, 29157, 29302, 29444, 29584, 29721, 29855, 29987, 30115,
- 30241, 30364, 30485, 30602, 30717, 30828, 30937, 31043, 31145, 31245, 31342, 31436,
- 31526, 31614, 31699, 31780, 31858, 31933, 32005, 32074, 32140, 32202, 32261, 32317,
- 32370, 32420, 32466, 32509, 32549, 32585, 32618, 32648, 32675, 32698, 32718, 32734,
- 32748, 32758, 32764, 32767, 32767, 32667, 32365, 31863, 31164, 30274, 29197, 27939,
- 26510, 24917, 23170, 21281, 19261, 17121, 14876, 12540, 10126, 7650, 5126, 2571
-};
-
-/* Lag window for LPC (Q31) */
-const WebRtc_Word32 WebRtcIlbcfix_kLpcLagWin[LPC_FILTERORDER + 1]={
- 2147483647, 2144885453, 2137754373, 2125918626, 2109459810,
- 2088483140, 2063130336, 2033564590, 1999977009, 1962580174,
- 1921610283};
-
-/* WebRtcIlbcfix_kLpcChirpSyntDenum vector in Q15 corresponding
- * floating point vector {1 0.9025 0.9025^2 0.9025^3 ...}
- */
-const WebRtc_Word16 WebRtcIlbcfix_kLpcChirpSyntDenum[LPC_FILTERORDER + 1] = {
- 32767, 29573, 26690, 24087,
- 21739, 19619, 17707, 15980,
- 14422, 13016, 11747};
-
-/* WebRtcIlbcfix_kLpcChirpWeightDenum in Q15 corresponding to
- * floating point vector {1 0.4222 0.4222^2... }
- */
-const WebRtc_Word16 WebRtcIlbcfix_kLpcChirpWeightDenum[LPC_FILTERORDER + 1] = {
- 32767, 13835, 5841, 2466, 1041, 440,
- 186, 78, 33, 14, 6};
-
-/* LSF quantization Q13 domain */
-const WebRtc_Word16 WebRtcIlbcfix_kLsfCb[64 * 3 + 128 * 3 + 128 * 4] = {
- 1273, 2238, 3696,
- 3199, 5309, 8209,
- 3606, 5671, 7829,
- 2815, 5262, 8778,
- 2608, 4027, 5493,
- 1582, 3076, 5945,
- 2983, 4181, 5396,
- 2437, 4322, 6902,
- 1861, 2998, 4613,
- 2007, 3250, 5214,
- 1388, 2459, 4262,
- 2563, 3805, 5269,
- 2036, 3522, 5129,
- 1935, 4025, 6694,
- 2744, 5121, 7338,
- 2810, 4248, 5723,
- 3054, 5405, 7745,
- 1449, 2593, 4763,
- 3411, 5128, 6596,
- 2484, 4659, 7496,
- 1668, 2879, 4818,
- 1812, 3072, 5036,
- 1638, 2649, 3900,
- 2464, 3550, 4644,
- 1853, 2900, 4158,
- 2458, 4163, 5830,
- 2556, 4036, 6254,
- 2703, 4432, 6519,
- 3062, 4953, 7609,
- 1725, 3703, 6187,
- 2221, 3877, 5427,
- 2339, 3579, 5197,
- 2021, 4633, 7037,
- 2216, 3328, 4535,
- 2961, 4739, 6667,
- 2807, 3955, 5099,
- 2788, 4501, 6088,
- 1642, 2755, 4431,
- 3341, 5282, 7333,
- 2414, 3726, 5727,
- 1582, 2822, 5269,
- 2259, 3447, 4905,
- 3117, 4986, 7054,
- 1825, 3491, 5542,
- 3338, 5736, 8627,
- 1789, 3090, 5488,
- 2566, 3720, 4923,
- 2846, 4682, 7161,
- 1950, 3321, 5976,
- 1834, 3383, 6734,
- 3238, 4769, 6094,
- 2031, 3978, 5903,
- 1877, 4068, 7436,
- 2131, 4644, 8296,
- 2764, 5010, 8013,
- 2194, 3667, 6302,
- 2053, 3127, 4342,
- 3523, 6595, 10010,
- 3134, 4457, 5748,
- 3142, 5819, 9414,
- 2223, 4334, 6353,
- 2022, 3224, 4822,
- 2186, 3458, 5544,
- 2552, 4757, 6870,
- 10905, 12917, 14578,
- 9503, 11485, 14485,
- 9518, 12494, 14052,
- 6222, 7487, 9174,
- 7759, 9186, 10506,
- 8315, 12755, 14786,
- 9609, 11486, 13866,
- 8909, 12077, 13643,
- 7369, 9054, 11520,
- 9408, 12163, 14715,
- 6436, 9911, 12843,
- 7109, 9556, 11884,
- 7557, 10075, 11640,
- 6482, 9202, 11547,
- 6463, 7914, 10980,
- 8611, 10427, 12752,
- 7101, 9676, 12606,
- 7428, 11252, 13172,
- 10197, 12955, 15842,
- 7487, 10955, 12613,
- 5575, 7858, 13621,
- 7268, 11719, 14752,
- 7476, 11744, 13795,
- 7049, 8686, 11922,
- 8234, 11314, 13983,
- 6560, 11173, 14984,
- 6405, 9211, 12337,
- 8222, 12054, 13801,
- 8039, 10728, 13255,
- 10066, 12733, 14389,
- 6016, 7338, 10040,
- 6896, 8648, 10234,
- 7538, 9170, 12175,
- 7327, 12608, 14983,
- 10516, 12643, 15223,
- 5538, 7644, 12213,
- 6728, 12221, 14253,
- 7563, 9377, 12948,
- 8661, 11023, 13401,
- 7280, 8806, 11085,
- 7723, 9793, 12333,
- 12225, 14648, 16709,
- 8768, 13389, 15245,
- 10267, 12197, 13812,
- 5301, 7078, 11484,
- 7100, 10280, 11906,
- 8716, 12555, 14183,
- 9567, 12464, 15434,
- 7832, 12305, 14300,
- 7608, 10556, 12121,
- 8913, 11311, 12868,
- 7414, 9722, 11239,
- 8666, 11641, 13250,
- 9079, 10752, 12300,
- 8024, 11608, 13306,
- 10453, 13607, 16449,
- 8135, 9573, 10909,
- 6375, 7741, 10125,
- 10025, 12217, 14874,
- 6985, 11063, 14109,
- 9296, 13051, 14642,
- 8613, 10975, 12542,
- 6583, 10414, 13534,
- 6191, 9368, 13430,
- 5742, 6859, 9260,
- 7723, 9813, 13679,
- 8137, 11291, 12833,
- 6562, 8973, 10641,
- 6062, 8462, 11335,
- 6928, 8784, 12647,
- 7501, 8784, 10031,
- 8372, 10045, 12135,
- 8191, 9864, 12746,
- 5917, 7487, 10979,
- 5516, 6848, 10318,
- 6819, 9899, 11421,
- 7882, 12912, 15670,
- 9558, 11230, 12753,
- 7752, 9327, 11472,
- 8479, 9980, 11358,
- 11418, 14072, 16386,
- 7968, 10330, 14423,
- 8423, 10555, 12162,
- 6337, 10306, 14391,
- 8850, 10879, 14276,
- 6750, 11885, 15710,
- 7037, 8328, 9764,
- 6914, 9266, 13476,
- 9746, 13949, 15519,
- 11032, 14444, 16925,
- 8032, 10271, 11810,
- 10962, 13451, 15833,
- 10021, 11667, 13324,
- 6273, 8226, 12936,
- 8543, 10397, 13496,
- 7936, 10302, 12745,
- 6769, 8138, 10446,
- 6081, 7786, 11719,
- 8637, 11795, 14975,
- 8790, 10336, 11812,
- 7040, 8490, 10771,
- 7338, 10381, 13153,
- 6598, 7888, 9358,
- 6518, 8237, 12030,
- 9055, 10763, 12983,
- 6490, 10009, 12007,
- 9589, 12023, 13632,
- 6867, 9447, 10995,
- 7930, 9816, 11397,
- 10241, 13300, 14939,
- 5830, 8670, 12387,
- 9870, 11915, 14247,
- 9318, 11647, 13272,
- 6721, 10836, 12929,
- 6543, 8233, 9944,
- 8034, 10854, 12394,
- 9112, 11787, 14218,
- 9302, 11114, 13400,
- 9022, 11366, 13816,
- 6962, 10461, 12480,
- 11288, 13333, 15222,
- 7249, 8974, 10547,
- 10566, 12336, 14390,
- 6697, 11339, 13521,
- 11851, 13944, 15826,
- 6847, 8381, 11349,
- 7509, 9331, 10939,
- 8029, 9618, 11909,
- 13973, 17644, 19647, 22474,
- 14722, 16522, 20035, 22134,
- 16305, 18179, 21106, 23048,
- 15150, 17948, 21394, 23225,
- 13582, 15191, 17687, 22333,
- 11778, 15546, 18458, 21753,
- 16619, 18410, 20827, 23559,
- 14229, 15746, 17907, 22474,
- 12465, 15327, 20700, 22831,
- 15085, 16799, 20182, 23410,
- 13026, 16935, 19890, 22892,
- 14310, 16854, 19007, 22944,
- 14210, 15897, 18891, 23154,
- 14633, 18059, 20132, 22899,
- 15246, 17781, 19780, 22640,
- 16396, 18904, 20912, 23035,
- 14618, 17401, 19510, 21672,
- 15473, 17497, 19813, 23439,
- 18851, 20736, 22323, 23864,
- 15055, 16804, 18530, 20916,
- 16490, 18196, 19990, 21939,
- 11711, 15223, 21154, 23312,
- 13294, 15546, 19393, 21472,
- 12956, 16060, 20610, 22417,
- 11628, 15843, 19617, 22501,
- 14106, 16872, 19839, 22689,
- 15655, 18192, 20161, 22452,
- 12953, 15244, 20619, 23549,
- 15322, 17193, 19926, 21762,
- 16873, 18676, 20444, 22359,
- 14874, 17871, 20083, 21959,
- 11534, 14486, 19194, 21857,
- 17766, 19617, 21338, 23178,
- 13404, 15284, 19080, 23136,
- 15392, 17527, 19470, 21953,
- 14462, 16153, 17985, 21192,
- 17734, 19750, 21903, 23783,
- 16973, 19096, 21675, 23815,
- 16597, 18936, 21257, 23461,
- 15966, 17865, 20602, 22920,
- 15416, 17456, 20301, 22972,
- 18335, 20093, 21732, 23497,
- 15548, 17217, 20679, 23594,
- 15208, 16995, 20816, 22870,
- 13890, 18015, 20531, 22468,
- 13211, 15377, 19951, 22388,
- 12852, 14635, 17978, 22680,
- 16002, 17732, 20373, 23544,
- 11373, 14134, 19534, 22707,
- 17329, 19151, 21241, 23462,
- 15612, 17296, 19362, 22850,
- 15422, 19104, 21285, 23164,
- 13792, 17111, 19349, 21370,
- 15352, 17876, 20776, 22667,
- 15253, 16961, 18921, 22123,
- 14108, 17264, 20294, 23246,
- 15785, 17897, 20010, 21822,
- 17399, 19147, 20915, 22753,
- 13010, 15659, 18127, 20840,
- 16826, 19422, 22218, 24084,
- 18108, 20641, 22695, 24237,
- 18018, 20273, 22268, 23920,
- 16057, 17821, 21365, 23665,
- 16005, 17901, 19892, 23016,
- 13232, 16683, 21107, 23221,
- 13280, 16615, 19915, 21829,
- 14950, 18575, 20599, 22511,
- 16337, 18261, 20277, 23216,
- 14306, 16477, 21203, 23158,
- 12803, 17498, 20248, 22014,
- 14327, 17068, 20160, 22006,
- 14402, 17461, 21599, 23688,
- 16968, 18834, 20896, 23055,
- 15070, 17157, 20451, 22315,
- 15419, 17107, 21601, 23946,
- 16039, 17639, 19533, 21424,
- 16326, 19261, 21745, 23673,
- 16489, 18534, 21658, 23782,
- 16594, 18471, 20549, 22807,
- 18973, 21212, 22890, 24278,
- 14264, 18674, 21123, 23071,
- 15117, 16841, 19239, 23118,
- 13762, 15782, 20478, 23230,
- 14111, 15949, 20058, 22354,
- 14990, 16738, 21139, 23492,
- 13735, 16971, 19026, 22158,
- 14676, 17314, 20232, 22807,
- 16196, 18146, 20459, 22339,
- 14747, 17258, 19315, 22437,
- 14973, 17778, 20692, 23367,
- 15715, 17472, 20385, 22349,
- 15702, 18228, 20829, 23410,
- 14428, 16188, 20541, 23630,
- 16824, 19394, 21365, 23246,
- 13069, 16392, 18900, 21121,
- 12047, 16640, 19463, 21689,
- 14757, 17433, 19659, 23125,
- 15185, 16930, 19900, 22540,
- 16026, 17725, 19618, 22399,
- 16086, 18643, 21179, 23472,
- 15462, 17248, 19102, 21196,
- 17368, 20016, 22396, 24096,
- 12340, 14475, 19665, 23362,
- 13636, 16229, 19462, 22728,
- 14096, 16211, 19591, 21635,
- 12152, 14867, 19943, 22301,
- 14492, 17503, 21002, 22728,
- 14834, 16788, 19447, 21411,
- 14650, 16433, 19326, 22308,
- 14624, 16328, 19659, 23204,
- 13888, 16572, 20665, 22488,
- 12977, 16102, 18841, 22246,
- 15523, 18431, 21757, 23738,
- 14095, 16349, 18837, 20947,
- 13266, 17809, 21088, 22839,
- 15427, 18190, 20270, 23143,
- 11859, 16753, 20935, 22486,
- 12310, 17667, 21736, 23319,
- 14021, 15926, 18702, 22002,
- 12286, 15299, 19178, 21126,
- 15703, 17491, 21039, 23151,
- 12272, 14018, 18213, 22570,
- 14817, 16364, 18485, 22598,
- 17109, 19683, 21851, 23677,
- 12657, 14903, 19039, 22061,
- 14713, 16487, 20527, 22814,
- 14635, 16726, 18763, 21715,
- 15878, 18550, 20718, 22906
-};
-
-const WebRtc_Word16 WebRtcIlbcfix_kLsfDimCb[LSF_NSPLIT] = {3, 3, 4};
-const WebRtc_Word16 WebRtcIlbcfix_kLsfSizeCb[LSF_NSPLIT] = {64,128,128};
-
-const WebRtc_Word16 WebRtcIlbcfix_kLsfMean[LPC_FILTERORDER] = {
- 2308, 3652, 5434, 7885,
- 10255, 12559, 15160, 17513,
- 20328, 22752};
-
-const WebRtc_Word16 WebRtcIlbcfix_kLspMean[LPC_FILTERORDER] = {
- 31476, 29565, 25819, 18725, 10276,
- 1236, -9049, -17600, -25884, -30618
-};
-
-/* Q14 */
-const WebRtc_Word16 WebRtcIlbcfix_kLsfWeight20ms[4] = {12288, 8192, 4096, 0};
-const WebRtc_Word16 WebRtcIlbcfix_kLsfWeight30ms[6] = {8192, 16384, 10923, 5461, 0, 0};
-
-/*
- cos(x) in Q15
- WebRtcIlbcfix_kCos[i] = cos(pi*i/64.0)
- used in WebRtcIlbcfix_Lsp2Lsf()
-*/
-
-const WebRtc_Word16 WebRtcIlbcfix_kCos[64] = {
- 32767, 32729, 32610, 32413, 32138, 31786, 31357, 30853,
- 30274, 29622, 28899, 28106, 27246, 26320, 25330, 24279,
- 23170, 22006, 20788, 19520, 18205, 16846, 15447, 14010,
- 12540, 11039, 9512, 7962, 6393, 4808, 3212, 1608,
- 0, -1608, -3212, -4808, -6393, -7962, -9512, -11039,
- -12540, -14010, -15447, -16846, -18205, -19520, -20788, -22006,
- -23170, -24279, -25330, -26320, -27246, -28106, -28899, -29622,
- -30274, -30853, -31357, -31786, -32138, -32413, -32610, -32729
-};
-
-/*
- Derivative in Q19, used to interpolate between the
- WebRtcIlbcfix_kCos[] values to get a more exact y = cos(x)
-*/
-const WebRtc_Word16 WebRtcIlbcfix_kCosDerivative[64] = {
- -632, -1893, -3150, -4399, -5638, -6863, -8072, -9261,
- -10428, -11570, -12684, -13767, -14817, -15832, -16808, -17744,
- -18637, -19486, -20287, -21039, -21741, -22390, -22986, -23526,
- -24009, -24435, -24801, -25108, -25354, -25540, -25664, -25726,
- -25726, -25664, -25540, -25354, -25108, -24801, -24435, -24009,
- -23526, -22986, -22390, -21741, -21039, -20287, -19486, -18637,
- -17744, -16808, -15832, -14817, -13767, -12684, -11570, -10428,
- -9261, -8072, -6863, -5638, -4399, -3150, -1893, -632};
-
-/*
- Table in Q15, used for a2lsf conversion
- WebRtcIlbcfix_kCosGrid[i] = cos((2*pi*i)/(float)(2*COS_GRID_POINTS));
-*/
-
-const WebRtc_Word16 WebRtcIlbcfix_kCosGrid[COS_GRID_POINTS + 1] = {
- 32760, 32723, 32588, 32364, 32051, 31651, 31164, 30591,
- 29935, 29196, 28377, 27481, 26509, 25465, 24351, 23170,
- 21926, 20621, 19260, 17846, 16384, 14876, 13327, 11743,
- 10125, 8480, 6812, 5126, 3425, 1714, 0, -1714, -3425,
- -5126, -6812, -8480, -10125, -11743, -13327, -14876,
- -16384, -17846, -19260, -20621, -21926, -23170, -24351,
- -25465, -26509, -27481, -28377, -29196, -29935, -30591,
- -31164, -31651, -32051, -32364, -32588, -32723, -32760
-};
-
-/*
- Derivative of y = acos(x) in Q12
- used in WebRtcIlbcfix_Lsp2Lsf()
-*/
-
-const WebRtc_Word16 WebRtcIlbcfix_kAcosDerivative[64] = {
- -26887, -8812, -5323, -3813, -2979, -2444, -2081, -1811,
- -1608, -1450, -1322, -1219, -1132, -1059, -998, -946,
- -901, -861, -827, -797, -772, -750, -730, -713,
- -699, -687, -677, -668, -662, -657, -654, -652,
- -652, -654, -657, -662, -668, -677, -687, -699,
- -713, -730, -750, -772, -797, -827, -861, -901,
- -946, -998, -1059, -1132, -1219, -1322, -1450, -1608,
- -1811, -2081, -2444, -2979, -3813, -5323, -8812, -26887
-};
-
-
-/* Tables for quantization of start state */
-
-/* State quantization tables */
-const WebRtc_Word16 WebRtcIlbcfix_kStateSq3[8] = { /* Values in Q13 */
- -30473, -17838, -9257, -2537,
- 3639, 10893, 19958, 32636
-};
-
-/* This table defines the limits for the selection of the freqg
- less or equal than value 0 => index = 0
- less or equal than value k => index = k
-*/
-const WebRtc_Word32 WebRtcIlbcfix_kChooseFrgQuant[64] = {
- 118, 163, 222, 305, 425, 604,
- 851, 1174, 1617, 2222, 3080, 4191,
- 5525, 7215, 9193, 11540, 14397, 17604,
- 21204, 25209, 29863, 35720, 42531, 50375,
- 59162, 68845, 80108, 93754, 110326, 129488,
- 150654, 174328, 201962, 233195, 267843, 308239,
- 354503, 405988, 464251, 531550, 608652, 697516,
- 802526, 928793, 1080145, 1258120, 1481106, 1760881,
- 2111111, 2546619, 3078825, 3748642, 4563142, 5573115,
- 6887601, 8582108, 10797296, 14014513, 18625760, 25529599,
- 37302935, 58819185, 109782723, WEBRTC_SPL_WORD32_MAX
-};
-
-const WebRtc_Word16 WebRtcIlbcfix_kScale[64] = {
- /* Values in Q16 */
- 29485, 25003, 21345, 18316, 15578, 13128, 10973, 9310, 7955,
- 6762, 5789, 4877, 4255, 3699, 3258, 2904, 2595, 2328,
- 2123, 1932, 1785, 1631, 1493, 1370, 1260, 1167, 1083,
- /* Values in Q21 */
- 32081, 29611, 27262, 25229, 23432, 21803, 20226, 18883, 17609,
- 16408, 15311, 14327, 13390, 12513, 11693, 10919, 10163, 9435,
- 8739, 8100, 7424, 6813, 6192, 5648, 5122, 4639, 4207, 3798,
- 3404, 3048, 2706, 2348, 2036, 1713, 1393, 1087, 747
-};
-
-/*frgq in fixpoint, but already computed like this:
- for(i=0; i<64; i++){
- a = (pow(10,frgq[i])/4.5);
- WebRtcIlbcfix_kFrgQuantMod[i] = round(a);
- }
-
- Value 0 :36 in Q8
- 37:58 in Q5
- 59:63 in Q3
-*/
-const WebRtc_Word16 WebRtcIlbcfix_kFrgQuantMod[64] = {
- /* First 37 values in Q8 */
- 569, 671, 786, 916, 1077, 1278,
- 1529, 1802, 2109, 2481, 2898, 3440,
- 3943, 4535, 5149, 5778, 6464, 7208,
- 7904, 8682, 9397, 10285, 11240, 12246,
- 13313, 14382, 15492, 16735, 18131, 19693,
- 21280, 22912, 24624, 26544, 28432, 30488,
- 32720,
- /* 22 values in Q5 */
- 4383, 4684, 5012, 5363, 5739, 6146,
- 6603, 7113, 7679, 8285, 9040, 9850,
- 10838, 11882, 13103, 14467, 15950, 17669,
- 19712, 22016, 24800, 28576,
- /* 5 values in Q3 */
- 8240, 9792, 12040, 15440, 22472
-};
-
-/* Constants for codebook search and creation */
-
-/* Expansion filter to get additional cb section.
- * Q12 and reversed compared to flp
- */
-const WebRtc_Word16 WebRtcIlbcfix_kCbFiltersRev[CB_FILTERLEN]={
- -140, 446, -755, 3302, 2922, -590, 343, -138};
-
-/* Weighting coefficients for short lags.
- * [0.2 0.4 0.6 0.8] in Q15 */
-const WebRtc_Word16 WebRtcIlbcfix_kAlpha[4]={
- 6554, 13107, 19661, 26214};
-
-/* Ranges for search and filters at different subframes */
-
-const WebRtc_Word16 WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES]={
- {58,58,58}, {108,44,44}, {108,108,108}, {108,108,108}, {108,108,108}};
-
-const WebRtc_Word16 WebRtcIlbcfix_kFilterRange[5]={63, 85, 125, 147, 147};
-
-/* Gain Quantization for the codebook gains of the 3 stages */
-
-/* Q14 (one extra value (max WebRtc_Word16) to simplify for the search) */
-const WebRtc_Word16 WebRtcIlbcfix_kGainSq3[9]={
- -16384, -10813, -5407, 0, 4096, 8192,
- 12288, 16384, 32767};
-
-/* Q14 (one extra value (max WebRtc_Word16) to simplify for the search) */
-const WebRtc_Word16 WebRtcIlbcfix_kGainSq4[17]={
- -17203, -14746, -12288, -9830, -7373, -4915,
- -2458, 0, 2458, 4915, 7373, 9830,
- 12288, 14746, 17203, 19661, 32767};
-
-/* Q14 (one extra value (max WebRtc_Word16) to simplify for the search) */
-const WebRtc_Word16 WebRtcIlbcfix_kGainSq5[33]={
- 614, 1229, 1843, 2458, 3072, 3686,
- 4301, 4915, 5530, 6144, 6758, 7373,
- 7987, 8602, 9216, 9830, 10445, 11059,
- 11674, 12288, 12902, 13517, 14131, 14746,
- 15360, 15974, 16589, 17203, 17818, 18432,
- 19046, 19661, 32767};
-
-/* Q14 gain_sq5Tbl squared in Q14 */
-const WebRtc_Word16 WebRtcIlbcfix_kGainSq5Sq[32] = {
- 23, 92, 207, 368, 576, 829,
- 1129, 1474, 1866, 2304, 2787, 3317,
- 3893, 4516, 5184, 5897, 6658, 7464,
- 8318, 9216, 10160, 11151, 12187, 13271,
- 14400, 15574, 16796, 18062, 19377, 20736,
- 22140, 23593
-};
-
-const WebRtc_Word16* const WebRtcIlbcfix_kGain[3] =
-{WebRtcIlbcfix_kGainSq5, WebRtcIlbcfix_kGainSq4, WebRtcIlbcfix_kGainSq3};
-
-
-/* Tables for the Enhancer, using upsamling factor 4 (ENH_UPS0 = 4) */
-
-const WebRtc_Word16 WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0][ENH_FLO_MULT2_PLUS1]={
- {0, 0, 0, 4096, 0, 0, 0},
- {64, -315, 1181, 3531, -436, 77, -64},
- {97, -509, 2464, 2464, -509, 97, -97},
- {77, -436, 3531, 1181, -315, 64, -77}
-};
-
-const WebRtc_Word16 WebRtcIlbcfix_kEnhWt[3] = {
- 4800, 16384, 27968 /* Q16 */
-};
-
-const WebRtc_Word16 WebRtcIlbcfix_kEnhPlocs[ENH_NBLOCKS_TOT] = {
- 160, 480, 800, 1120, 1440, 1760, 2080, 2400 /* Q(-2) */
-};
-
-/* PLC table */
-
-const WebRtc_Word16 WebRtcIlbcfix_kPlcPerSqr[6] = { /* Grid points for square of periodiciy in Q15 */
- 839, 1343, 2048, 2998, 4247, 5849
-};
-
-const WebRtc_Word16 WebRtcIlbcfix_kPlcPitchFact[6] = { /* Value of y=(x^4-0.4)/(0.7-0.4) in grid points in Q15 */
- 0, 5462, 10922, 16384, 21846, 27306
-};
-
-const WebRtc_Word16 WebRtcIlbcfix_kPlcPfSlope[6] = { /* Slope of y=(x^4-0.4)/(0.7-0.4) in Q11 */
- 26667, 18729, 13653, 10258, 7901, 6214
-};
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/constants.h b/trunk/src/modules/audio_coding/codecs/ilbc/constants.h
deleted file mode 100644
index f787f74..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/constants.h
+++ /dev/null
@@ -1,92 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- constants.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CONSTANTS_H_
-
-#include "defines.h"
-#include "typedefs.h"
-
-/* high pass filters */
-
-extern const WebRtc_Word16 WebRtcIlbcfix_kHpInCoefs[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kHpOutCoefs[];
-
-/* Window for start state decision */
-extern const WebRtc_Word16 WebRtcIlbcfix_kStartSequenceEnrgWin[];
-
-/* low pass filter used for downsampling */
-extern const WebRtc_Word16 WebRtcIlbcfix_kLpFiltCoefs[];
-
-/* LPC analysis and quantization */
-
-extern const WebRtc_Word16 WebRtcIlbcfix_kLpcWin[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLpcAsymWin[];
-extern const WebRtc_Word32 WebRtcIlbcfix_kLpcLagWin[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLpcChirpSyntDenum[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLpcChirpWeightDenum[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLsfDimCb[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLsfSizeCb[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLsfCb[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLsfWeight20ms[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLsfWeight30ms[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLsfMean[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kLspMean[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kCos[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kCosDerivative[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kCosGrid[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kAcosDerivative[];
-
-/* state quantization tables */
-
-extern const WebRtc_Word16 WebRtcIlbcfix_kStateSq3[];
-extern const WebRtc_Word32 WebRtcIlbcfix_kChooseFrgQuant[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kScale[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kFrgQuantMod[];
-
-/* Ranges for search and filters at different subframes */
-
-extern const WebRtc_Word16 WebRtcIlbcfix_kSearchRange[5][CB_NSTAGES];
-extern const WebRtc_Word16 WebRtcIlbcfix_kFilterRange[];
-
-/* gain quantization tables */
-
-extern const WebRtc_Word16 WebRtcIlbcfix_kGainSq3[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kGainSq4[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kGainSq5[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kGainSq5Sq[];
-extern const WebRtc_Word16* const WebRtcIlbcfix_kGain[];
-
-/* adaptive codebook definitions */
-
-extern const WebRtc_Word16 WebRtcIlbcfix_kCbFiltersRev[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kAlpha[];
-
-/* enhancer definitions */
-
-extern const WebRtc_Word16 WebRtcIlbcfix_kEnhPolyPhaser[ENH_UPS0][ENH_FLO_MULT2_PLUS1];
-extern const WebRtc_Word16 WebRtcIlbcfix_kEnhWt[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kEnhPlocs[];
-
-/* PLC tables */
-
-extern const WebRtc_Word16 WebRtcIlbcfix_kPlcPerSqr[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kPlcPitchFact[];
-extern const WebRtc_Word16 WebRtcIlbcfix_kPlcPfSlope[];
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/create_augmented_vec.c b/trunk/src/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
deleted file mode 100644
index f021c4d..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/create_augmented_vec.c
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CreateAugmentedVec.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * Recreate a specific codebook vector from the augmented part.
- *
- *----------------------------------------------------------------*/
-
-void WebRtcIlbcfix_CreateAugmentedVec(
- WebRtc_Word16 index, /* (i) Index for the augmented vector to be created */
- WebRtc_Word16 *buffer, /* (i) Pointer to the end of the codebook memory that
- is used for creation of the augmented codebook */
- WebRtc_Word16 *cbVec /* (o) The construced codebook vector */
- ) {
- WebRtc_Word16 ilow;
- WebRtc_Word16 *ppo, *ppi;
- WebRtc_Word16 cbVecTmp[4];
-
- ilow = index-4;
-
- /* copy the first noninterpolated part */
- ppo = buffer-index;
- WEBRTC_SPL_MEMCPY_W16(cbVec, ppo, index);
-
- /* interpolation */
- ppo = buffer - 4;
- ppi = buffer - index - 4;
-
- /* perform cbVec[ilow+k] = ((ppi[k]*alphaTbl[k])>>15) + ((ppo[k]*alphaTbl[3-k])>>15);
- for k = 0..3
- */
- WebRtcSpl_ElementwiseVectorMult(&cbVec[ilow], ppi, WebRtcIlbcfix_kAlpha, 4, 15);
- WebRtcSpl_ReverseOrderMultArrayElements(cbVecTmp, ppo, &WebRtcIlbcfix_kAlpha[3], 4, 15);
- WebRtcSpl_AddVectorsAndShift(&cbVec[ilow], &cbVec[ilow], cbVecTmp, 4, 0);
-
- /* copy the second noninterpolated part */
- ppo = buffer - index;
- WEBRTC_SPL_MEMCPY_W16(cbVec+index,ppo,(SUBL-index));
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/create_augmented_vec.h b/trunk/src/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
deleted file mode 100644
index 970a9be..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/create_augmented_vec.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_CreateAugmentedVec.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_CREATE_AUGMENTED_VEC_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Recreate a specific codebook vector from the augmented part.
- *
- *----------------------------------------------------------------*/
-
-void WebRtcIlbcfix_CreateAugmentedVec(
- WebRtc_Word16 index, /* (i) Index for the augmented vector to be created */
- WebRtc_Word16 *buffer, /* (i) Pointer to the end of the codebook memory that
- is used for creation of the augmented codebook */
- WebRtc_Word16 *cbVec /* (o) The construced codebook vector */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/decode.c b/trunk/src/modules/audio_coding/codecs/ilbc/decode.c
deleted file mode 100644
index 827532c..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/decode.c
+++ /dev/null
@@ -1,244 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Decode.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "simple_lsf_dequant.h"
-#include "decoder_interpolate_lsf.h"
-#include "index_conv_dec.h"
-#include "do_plc.h"
-#include "constants.h"
-#include "enhancer_interface.h"
-#include "xcorr_coef.h"
-#include "lsf_check.h"
-#include "decode_residual.h"
-#include "unpack_bits.h"
-#include "hp_output.h"
-#ifndef WEBRTC_BIG_ENDIAN
-#include "swap_bytes.h"
-#endif
-
-/*----------------------------------------------------------------*
- * main decoder function
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_DecodeImpl(
- WebRtc_Word16 *decblock, /* (o) decoded signal block */
- WebRtc_UWord16 *bytes, /* (i) encoded signal bits */
- iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
- structure */
- WebRtc_Word16 mode /* (i) 0: bad packet, PLC,
- 1: normal */
- ) {
- int i;
- WebRtc_Word16 order_plus_one;
-
- WebRtc_Word16 last_bit;
- WebRtc_Word16 *data;
- /* Stack based */
- WebRtc_Word16 decresidual[BLOCKL_MAX];
- WebRtc_Word16 PLCresidual[BLOCKL_MAX + LPC_FILTERORDER];
- WebRtc_Word16 syntdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
- WebRtc_Word16 PLClpc[LPC_FILTERORDER + 1];
- iLBC_bits *iLBCbits_inst = (iLBC_bits*)PLCresidual;
-
- /* Reuse some buffers that are non overlapping in order to save stack memory */
- data = &PLCresidual[LPC_FILTERORDER];
-
- if (mode>0) { /* the data are good */
-
- /* decode data */
-
-#ifndef WEBRTC_BIG_ENDIAN
- WebRtcIlbcfix_SwapBytes((WebRtc_UWord16*)bytes, iLBCdec_inst->no_of_words);
-#endif
-
- /* Unpacketize bits into parameters */
-
- last_bit = WebRtcIlbcfix_UnpackBits(bytes, iLBCbits_inst, iLBCdec_inst->mode);
-
-#ifndef WEBRTC_BIG_ENDIAN
- /* Swap back so that the the input vector "bytes" is unchanged */
- WebRtcIlbcfix_SwapBytes((WebRtc_UWord16*)bytes, iLBCdec_inst->no_of_words);
-#endif
-
- /* Check for bit errors */
- if (iLBCbits_inst->startIdx<1)
- mode = 0;
- if ((iLBCdec_inst->mode==20) && (iLBCbits_inst->startIdx>3))
- mode = 0;
- if ((iLBCdec_inst->mode==30) && (iLBCbits_inst->startIdx>5))
- mode = 0;
- if (last_bit==1)
- mode = 0;
-
- if (mode==1) { /* No bit errors was detected, continue decoding */
- /* Stack based */
- WebRtc_Word16 lsfdeq[LPC_FILTERORDER*LPC_N_MAX];
- WebRtc_Word16 weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
-
- /* adjust index */
- WebRtcIlbcfix_IndexConvDec(iLBCbits_inst->cb_index);
-
- /* decode the lsf */
- WebRtcIlbcfix_SimpleLsfDeQ(lsfdeq, (WebRtc_Word16*)(iLBCbits_inst->lsf), iLBCdec_inst->lpc_n);
- WebRtcIlbcfix_LsfCheck(lsfdeq, LPC_FILTERORDER, iLBCdec_inst->lpc_n);
- WebRtcIlbcfix_DecoderInterpolateLsp(syntdenum, weightdenum,
- lsfdeq, LPC_FILTERORDER, iLBCdec_inst);
-
- /* Decode the residual using the cb and gain indexes */
- WebRtcIlbcfix_DecodeResidual(iLBCdec_inst, iLBCbits_inst, decresidual, syntdenum);
-
- /* preparing the plc for a future loss! */
- WebRtcIlbcfix_DoThePlc( PLCresidual, PLClpc, 0,
- decresidual, syntdenum + (LPC_FILTERORDER + 1)*(iLBCdec_inst->nsub - 1),
- (WebRtc_Word16)(iLBCdec_inst->last_lag), iLBCdec_inst);
-
- /* Use the output from doThePLC */
- WEBRTC_SPL_MEMCPY_W16(decresidual, PLCresidual, iLBCdec_inst->blockl);
- }
-
- }
-
- if (mode == 0) {
- /* the data is bad (either a PLC call
- * was made or a bit error was detected)
- */
-
- /* packet loss conceal */
-
- WebRtcIlbcfix_DoThePlc( PLCresidual, PLClpc, 1,
- decresidual, syntdenum, (WebRtc_Word16)(iLBCdec_inst->last_lag), iLBCdec_inst);
-
- WEBRTC_SPL_MEMCPY_W16(decresidual, PLCresidual, iLBCdec_inst->blockl);
-
- order_plus_one = LPC_FILTERORDER + 1;
-
- for (i = 0; i < iLBCdec_inst->nsub; i++) {
- WEBRTC_SPL_MEMCPY_W16(syntdenum+(i*order_plus_one),
- PLClpc, order_plus_one);
- }
- }
-
- if ((*iLBCdec_inst).use_enhancer == 1) { /* Enhancer activated */
-
- /* Update the filter and filter coefficients if there was a packet loss */
- if (iLBCdec_inst->prev_enh_pl==2) {
- for (i=0;i<iLBCdec_inst->nsub;i++) {
- WEBRTC_SPL_MEMCPY_W16(&(iLBCdec_inst->old_syntdenum[i*(LPC_FILTERORDER+1)]),
- syntdenum, (LPC_FILTERORDER+1));
- }
- }
-
- /* post filtering */
- (*iLBCdec_inst).last_lag =
- WebRtcIlbcfix_EnhancerInterface(data, decresidual, iLBCdec_inst);
-
- /* synthesis filtering */
-
- /* Set up the filter state */
- WEBRTC_SPL_MEMCPY_W16(&data[-LPC_FILTERORDER], iLBCdec_inst->syntMem, LPC_FILTERORDER);
-
- if (iLBCdec_inst->mode==20) {
- /* Enhancer has 40 samples delay */
- i=0;
- WebRtcSpl_FilterARFastQ12(
- data, data,
- iLBCdec_inst->old_syntdenum + (i+iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1),
- LPC_FILTERORDER+1, SUBL);
-
- for (i=1; i < iLBCdec_inst->nsub; i++) {
- WebRtcSpl_FilterARFastQ12(
- data+i*SUBL, data+i*SUBL,
- syntdenum+(i-1)*(LPC_FILTERORDER+1),
- LPC_FILTERORDER+1, SUBL);
- }
-
- } else if (iLBCdec_inst->mode==30) {
- /* Enhancer has 80 samples delay */
- for (i=0; i < 2; i++) {
- WebRtcSpl_FilterARFastQ12(
- data+i*SUBL, data+i*SUBL,
- iLBCdec_inst->old_syntdenum + (i+4)*(LPC_FILTERORDER+1),
- LPC_FILTERORDER+1, SUBL);
- }
- for (i=2; i < iLBCdec_inst->nsub; i++) {
- WebRtcSpl_FilterARFastQ12(
- data+i*SUBL, data+i*SUBL,
- syntdenum+(i-2)*(LPC_FILTERORDER+1),
- LPC_FILTERORDER+1, SUBL);
- }
- }
-
- /* Save the filter state */
- WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &data[iLBCdec_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER);
-
- } else { /* Enhancer not activated */
- WebRtc_Word16 lag;
-
- /* Find last lag (since the enhancer is not called to give this info) */
- lag = 20;
- if (iLBCdec_inst->mode==20) {
- lag = (WebRtc_Word16)WebRtcIlbcfix_XcorrCoef(
- &decresidual[iLBCdec_inst->blockl-60],
- &decresidual[iLBCdec_inst->blockl-60-lag],
- 60,
- 80, lag, -1);
- } else {
- lag = (WebRtc_Word16)WebRtcIlbcfix_XcorrCoef(
- &decresidual[iLBCdec_inst->blockl-ENH_BLOCKL],
- &decresidual[iLBCdec_inst->blockl-ENH_BLOCKL-lag],
- ENH_BLOCKL,
- 100, lag, -1);
- }
-
- /* Store lag (it is needed if next packet is lost) */
- (*iLBCdec_inst).last_lag = (int)lag;
-
- /* copy data and run synthesis filter */
- WEBRTC_SPL_MEMCPY_W16(data, decresidual, iLBCdec_inst->blockl);
-
- /* Set up the filter state */
- WEBRTC_SPL_MEMCPY_W16(&data[-LPC_FILTERORDER], iLBCdec_inst->syntMem, LPC_FILTERORDER);
-
- for (i=0; i < iLBCdec_inst->nsub; i++) {
- WebRtcSpl_FilterARFastQ12(
- data+i*SUBL, data+i*SUBL,
- syntdenum + i*(LPC_FILTERORDER+1),
- LPC_FILTERORDER+1, SUBL);
- }
-
- /* Save the filter state */
- WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &data[iLBCdec_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER);
- }
-
- WEBRTC_SPL_MEMCPY_W16(decblock,data,iLBCdec_inst->blockl);
-
- /* High pass filter the signal (with upscaling a factor 2 and saturation) */
- WebRtcIlbcfix_HpOutput(decblock, (WebRtc_Word16*)WebRtcIlbcfix_kHpOutCoefs,
- iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
- iLBCdec_inst->blockl);
-
- WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->old_syntdenum,
- syntdenum, iLBCdec_inst->nsub*(LPC_FILTERORDER+1));
-
- iLBCdec_inst->prev_enh_pl=0;
-
- if (mode==0) { /* PLC was used */
- iLBCdec_inst->prev_enh_pl=1;
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/decode.h b/trunk/src/modules/audio_coding/codecs/ilbc/decode.h
deleted file mode 100644
index 0252d9c..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/decode.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Decode.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * main decoder function
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_DecodeImpl(
- WebRtc_Word16 *decblock, /* (o) decoded signal block */
- WebRtc_UWord16 *bytes, /* (i) encoded signal bits */
- iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
- structure */
- WebRtc_Word16 mode /* (i) 0: bad packet, PLC,
- 1: normal */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/decode_residual.c b/trunk/src/modules/audio_coding/codecs/ilbc/decode_residual.c
deleted file mode 100644
index 4bc1cd3..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/decode_residual.c
+++ /dev/null
@@ -1,189 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_DecodeResidual.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "state_construct.h"
-#include "cb_construct.h"
-#include "index_conv_dec.h"
-#include "do_plc.h"
-#include "constants.h"
-#include "enhancer_interface.h"
-#include "xcorr_coef.h"
-#include "lsf_check.h"
-
-
-/*----------------------------------------------------------------*
- * frame residual decoder function (subrutine to iLBC_decode)
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_DecodeResidual(
- iLBC_Dec_Inst_t *iLBCdec_inst,
- /* (i/o) the decoder state structure */
- iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits, which are used
- for the decoding */
- WebRtc_Word16 *decresidual, /* (o) decoded residual frame */
- WebRtc_Word16 *syntdenum /* (i) the decoded synthesis filter
- coefficients */
- ) {
- WebRtc_Word16 meml_gotten, Nfor, Nback, diff, start_pos;
- WebRtc_Word16 subcount, subframe;
- WebRtc_Word16 *reverseDecresidual = iLBCdec_inst->enh_buf; /* Reversed decoded data, used for decoding backwards in time (reuse memory in state) */
- WebRtc_Word16 *memVec = iLBCdec_inst->prevResidual; /* Memory for codebook and filter state (reuse memory in state) */
- WebRtc_Word16 *mem = &memVec[CB_HALFFILTERLEN]; /* Memory for codebook */
-
- diff = STATE_LEN - iLBCdec_inst->state_short_len;
-
- if (iLBC_encbits->state_first == 1) {
- start_pos = (iLBC_encbits->startIdx-1)*SUBL;
- } else {
- start_pos = (iLBC_encbits->startIdx-1)*SUBL + diff;
- }
-
- /* decode scalar part of start state */
-
- WebRtcIlbcfix_StateConstruct(iLBC_encbits->idxForMax,
- iLBC_encbits->idxVec, &syntdenum[(iLBC_encbits->startIdx-1)*(LPC_FILTERORDER+1)],
- &decresidual[start_pos], iLBCdec_inst->state_short_len
- );
-
- if (iLBC_encbits->state_first) { /* put adaptive part in the end */
-
- /* setup memory */
-
- WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-iLBCdec_inst->state_short_len));
- WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCdec_inst->state_short_len, decresidual+start_pos,
- iLBCdec_inst->state_short_len);
-
- /* construct decoded vector */
-
- WebRtcIlbcfix_CbConstruct(
- &decresidual[start_pos+iLBCdec_inst->state_short_len],
- iLBC_encbits->cb_index, iLBC_encbits->gain_index,
- mem+CB_MEML-ST_MEM_L_TBL,
- ST_MEM_L_TBL, (WebRtc_Word16)diff
- );
-
- }
- else {/* put adaptive part in the beginning */
-
- /* create reversed vectors for prediction */
-
- WebRtcSpl_MemCpyReversedOrder(reverseDecresidual+diff,
- &decresidual[(iLBC_encbits->startIdx+1)*SUBL-1-STATE_LEN], diff);
-
- /* setup memory */
-
- meml_gotten = iLBCdec_inst->state_short_len;
- WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1,
- decresidual+start_pos, meml_gotten);
- WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-meml_gotten));
-
- /* construct decoded vector */
-
- WebRtcIlbcfix_CbConstruct(
- reverseDecresidual,
- iLBC_encbits->cb_index, iLBC_encbits->gain_index,
- mem+CB_MEML-ST_MEM_L_TBL,
- ST_MEM_L_TBL, diff
- );
-
- /* get decoded residual from reversed vector */
-
- WebRtcSpl_MemCpyReversedOrder(&decresidual[start_pos-1],
- reverseDecresidual, diff);
- }
-
- /* counter for predicted subframes */
-
- subcount=1;
-
- /* forward prediction of subframes */
-
- Nfor = iLBCdec_inst->nsub-iLBC_encbits->startIdx-1;
-
- if( Nfor > 0 ) {
-
- /* setup memory */
- WebRtcSpl_MemSetW16(mem, 0, CB_MEML-STATE_LEN);
- WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-STATE_LEN,
- decresidual+(iLBC_encbits->startIdx-1)*SUBL, STATE_LEN);
-
- /* loop over subframes to encode */
-
- for (subframe=0; subframe<Nfor; subframe++) {
-
- /* construct decoded vector */
- WebRtcIlbcfix_CbConstruct(
- &decresidual[(iLBC_encbits->startIdx+1+subframe)*SUBL],
- iLBC_encbits->cb_index+subcount*CB_NSTAGES,
- iLBC_encbits->gain_index+subcount*CB_NSTAGES,
- mem, MEM_LF_TBL, SUBL
- );
-
- /* update memory */
- WEBRTC_SPL_MEMMOVE_W16(mem, mem+SUBL, CB_MEML-SUBL);
- WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
- &decresidual[(iLBC_encbits->startIdx+1+subframe)*SUBL], SUBL);
-
- subcount++;
- }
-
- }
-
- /* backward prediction of subframes */
-
- Nback = iLBC_encbits->startIdx-1;
-
- if( Nback > 0 ){
-
- /* setup memory */
-
- meml_gotten = SUBL*(iLBCdec_inst->nsub+1-iLBC_encbits->startIdx);
- if( meml_gotten > CB_MEML ) {
- meml_gotten=CB_MEML;
- }
-
- WebRtcSpl_MemCpyReversedOrder(mem+CB_MEML-1,
- decresidual+(iLBC_encbits->startIdx-1)*SUBL, meml_gotten);
- WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-meml_gotten));
-
- /* loop over subframes to decode */
-
- for (subframe=0; subframe<Nback; subframe++) {
-
- /* construct decoded vector */
- WebRtcIlbcfix_CbConstruct(
- &reverseDecresidual[subframe*SUBL],
- iLBC_encbits->cb_index+subcount*CB_NSTAGES,
- iLBC_encbits->gain_index+subcount*CB_NSTAGES,
- mem, MEM_LF_TBL, SUBL
- );
-
- /* update memory */
- WEBRTC_SPL_MEMMOVE_W16(mem, mem+SUBL, CB_MEML-SUBL);
- WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
- &reverseDecresidual[subframe*SUBL], SUBL);
-
- subcount++;
- }
-
- /* get decoded residual from reversed vector */
- WebRtcSpl_MemCpyReversedOrder(decresidual+SUBL*Nback-1,
- reverseDecresidual, SUBL*Nback);
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/decode_residual.h b/trunk/src/modules/audio_coding/codecs/ilbc/decode_residual.h
deleted file mode 100644
index ea7208a..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/decode_residual.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_DecodeResidual.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_RESIDUAL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODE_RESIDUAL_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * frame residual decoder function (subrutine to iLBC_decode)
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_DecodeResidual(
- iLBC_Dec_Inst_t *iLBCdec_inst,
- /* (i/o) the decoder state structure */
- iLBC_bits *iLBC_encbits, /* (i/o) Encoded bits, which are used
- for the decoding */
- WebRtc_Word16 *decresidual, /* (o) decoded residual frame */
- WebRtc_Word16 *syntdenum /* (i) the decoded synthesis filter
- coefficients */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c b/trunk/src/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
deleted file mode 100644
index eee3105..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.c
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_DecoderInterpolateLsp.c
-
-******************************************************************/
-
-#include "lsf_interpolate_to_poly_dec.h"
-#include "bw_expand.h"
-#include "defines.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * obtain synthesis and weighting filters form lsf coefficients
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_DecoderInterpolateLsp(
- WebRtc_Word16 *syntdenum, /* (o) synthesis filter coefficients */
- WebRtc_Word16 *weightdenum, /* (o) weighting denumerator
- coefficients */
- WebRtc_Word16 *lsfdeq, /* (i) dequantized lsf coefficients */
- WebRtc_Word16 length, /* (i) length of lsf coefficient vector */
- iLBC_Dec_Inst_t *iLBCdec_inst
- /* (i) the decoder state structure */
- ){
- int i, pos, lp_length;
- WebRtc_Word16 lp[LPC_FILTERORDER + 1], *lsfdeq2;
-
- lsfdeq2 = lsfdeq + length;
- lp_length = length + 1;
-
- if (iLBCdec_inst->mode==30) {
- /* subframe 1: Interpolation between old and first LSF */
-
- WebRtcIlbcfix_LspInterpolate2PolyDec(lp, (*iLBCdec_inst).lsfdeqold, lsfdeq,
- WebRtcIlbcfix_kLsfWeight30ms[0], length);
- WEBRTC_SPL_MEMCPY_W16(syntdenum,lp,lp_length);
- WebRtcIlbcfix_BwExpand(weightdenum, lp, (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpSyntDenum, (WebRtc_Word16)lp_length);
-
- /* subframes 2 to 6: interpolation between first and last LSF */
-
- pos = lp_length;
- for (i = 1; i < 6; i++) {
- WebRtcIlbcfix_LspInterpolate2PolyDec(lp, lsfdeq, lsfdeq2,
- WebRtcIlbcfix_kLsfWeight30ms[i], length);
- WEBRTC_SPL_MEMCPY_W16(syntdenum + pos,lp,lp_length);
- WebRtcIlbcfix_BwExpand(weightdenum + pos, lp,
- (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpSyntDenum, (WebRtc_Word16)lp_length);
- pos += lp_length;
- }
- } else { /* iLBCdec_inst->mode=20 */
- /* subframes 1 to 4: interpolation between old and new LSF */
- pos = 0;
- for (i = 0; i < iLBCdec_inst->nsub; i++) {
- WebRtcIlbcfix_LspInterpolate2PolyDec(lp, iLBCdec_inst->lsfdeqold, lsfdeq,
- WebRtcIlbcfix_kLsfWeight20ms[i], length);
- WEBRTC_SPL_MEMCPY_W16(syntdenum+pos,lp,lp_length);
- WebRtcIlbcfix_BwExpand(weightdenum+pos, lp,
- (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpSyntDenum, (WebRtc_Word16)lp_length);
- pos += lp_length;
- }
- }
-
- /* update memory */
-
- if (iLBCdec_inst->mode==30) {
- WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, lsfdeq2, length);
- } else {
- WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, lsfdeq, length);
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h b/trunk/src/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
deleted file mode 100644
index 3896ca9..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/decoder_interpolate_lsf.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_DecoderInterpolateLsp.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DECODER_INTERPOLATE_LSF_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * obtain synthesis and weighting filters form lsf coefficients
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_DecoderInterpolateLsp(
- WebRtc_Word16 *syntdenum, /* (o) synthesis filter coefficients */
- WebRtc_Word16 *weightdenum, /* (o) weighting denumerator
- coefficients */
- WebRtc_Word16 *lsfdeq, /* (i) dequantized lsf coefficients */
- WebRtc_Word16 length, /* (i) length of lsf coefficient vector */
- iLBC_Dec_Inst_t *iLBCdec_inst
- /* (i) the decoder state structure */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/defines.h b/trunk/src/modules/audio_coding/codecs/ilbc/defines.h
deleted file mode 100644
index bdeba01..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/defines.h
+++ /dev/null
@@ -1,219 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- define.h
-
-******************************************************************/
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DEFINES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DEFINES_H_
-
-#include "typedefs.h"
-#include "signal_processing_library.h"
-#include <string.h>
-
-/* general codec settings */
-
-#define FS 8000
-#define BLOCKL_20MS 160
-#define BLOCKL_30MS 240
-#define BLOCKL_MAX 240
-#define NSUB_20MS 4
-#define NSUB_30MS 6
-#define NSUB_MAX 6
-#define NASUB_20MS 2
-#define NASUB_30MS 4
-#define NASUB_MAX 4
-#define SUBL 40
-#define STATE_LEN 80
-#define STATE_SHORT_LEN_30MS 58
-#define STATE_SHORT_LEN_20MS 57
-
-/* LPC settings */
-
-#define LPC_FILTERORDER 10
-#define LPC_LOOKBACK 60
-#define LPC_N_20MS 1
-#define LPC_N_30MS 2
-#define LPC_N_MAX 2
-#define LPC_ASYMDIFF 20
-#define LSF_NSPLIT 3
-#define LSF_NUMBER_OF_STEPS 4
-#define LPC_HALFORDER 5
-#define COS_GRID_POINTS 60
-
-/* cb settings */
-
-#define CB_NSTAGES 3
-#define CB_EXPAND 2
-#define CB_MEML 147
-#define CB_FILTERLEN (2*4)
-#define CB_HALFFILTERLEN 4
-#define CB_RESRANGE 34
-#define CB_MAXGAIN_FIXQ6 83 /* error = -0.24% */
-#define CB_MAXGAIN_FIXQ14 21299
-
-/* enhancer */
-
-#define ENH_BLOCKL 80 /* block length */
-#define ENH_BLOCKL_HALF (ENH_BLOCKL/2)
-#define ENH_HL 3 /* 2*ENH_HL+1 is number blocks
- in said second sequence */
-#define ENH_SLOP 2 /* max difference estimated and
- correct pitch period */
-#define ENH_PLOCSL 8 /* pitch-estimates and
- pitch-locations buffer length */
-#define ENH_OVERHANG 2
-#define ENH_UPS0 4 /* upsampling rate */
-#define ENH_FL0 3 /* 2*FLO+1 is the length of each filter */
-#define ENH_FLO_MULT2_PLUS1 7
-#define ENH_VECTL (ENH_BLOCKL+2*ENH_FL0)
-#define ENH_CORRDIM (2*ENH_SLOP+1)
-#define ENH_NBLOCKS (BLOCKL/ENH_BLOCKL)
-#define ENH_NBLOCKS_EXTRA 5
-#define ENH_NBLOCKS_TOT 8 /* ENH_NBLOCKS+ENH_NBLOCKS_EXTRA */
-#define ENH_BUFL (ENH_NBLOCKS_TOT)*ENH_BLOCKL
-#define ENH_BUFL_FILTEROVERHEAD 3
-#define ENH_A0 819 /* Q14 */
-#define ENH_A0_MINUS_A0A0DIV4 848256041 /* Q34 */
-#define ENH_A0DIV2 26843546 /* Q30 */
-
-/* PLC */
-
-/* Down sampling */
-
-#define FILTERORDER_DS_PLUS1 7
-#define DELAY_DS 3
-#define FACTOR_DS 2
-
-/* bit stream defs */
-
-#define NO_OF_BYTES_20MS 38
-#define NO_OF_BYTES_30MS 50
-#define NO_OF_WORDS_20MS 19
-#define NO_OF_WORDS_30MS 25
-#define STATE_BITS 3
-#define BYTE_LEN 8
-#define ULP_CLASSES 3
-
-/* help parameters */
-
-#define TWO_PI_FIX 25736 /* Q12 */
-
-/* Constants for codebook search and creation */
-
-#define ST_MEM_L_TBL 85
-#define MEM_LF_TBL 147
-
-
-/* Struct for the bits */
-typedef struct iLBC_bits_t_ {
- WebRtc_Word16 lsf[LSF_NSPLIT*LPC_N_MAX];
- WebRtc_Word16 cb_index[CB_NSTAGES*(NASUB_MAX+1)]; /* First CB_NSTAGES values contains extra CB index */
- WebRtc_Word16 gain_index[CB_NSTAGES*(NASUB_MAX+1)]; /* First CB_NSTAGES values contains extra CB gain */
- WebRtc_Word16 idxForMax;
- WebRtc_Word16 state_first;
- WebRtc_Word16 idxVec[STATE_SHORT_LEN_30MS];
- WebRtc_Word16 firstbits;
- WebRtc_Word16 startIdx;
-} iLBC_bits;
-
-/* type definition encoder instance */
-typedef struct iLBC_Enc_Inst_t_ {
-
- /* flag for frame size mode */
- WebRtc_Word16 mode;
-
- /* basic parameters for different frame sizes */
- WebRtc_Word16 blockl;
- WebRtc_Word16 nsub;
- WebRtc_Word16 nasub;
- WebRtc_Word16 no_of_bytes, no_of_words;
- WebRtc_Word16 lpc_n;
- WebRtc_Word16 state_short_len;
-
- /* analysis filter state */
- WebRtc_Word16 anaMem[LPC_FILTERORDER];
-
- /* Fix-point old lsf parameters for interpolation */
- WebRtc_Word16 lsfold[LPC_FILTERORDER];
- WebRtc_Word16 lsfdeqold[LPC_FILTERORDER];
-
- /* signal buffer for LP analysis */
- WebRtc_Word16 lpc_buffer[LPC_LOOKBACK + BLOCKL_MAX];
-
- /* state of input HP filter */
- WebRtc_Word16 hpimemx[2];
- WebRtc_Word16 hpimemy[4];
-
-#ifdef SPLIT_10MS
- WebRtc_Word16 weightdenumbuf[66];
- WebRtc_Word16 past_samples[160];
- WebRtc_UWord16 bytes[25];
- WebRtc_Word16 section;
- WebRtc_Word16 Nfor_flag;
- WebRtc_Word16 Nback_flag;
- WebRtc_Word16 start_pos;
- WebRtc_Word16 diff;
-#endif
-
-} iLBC_Enc_Inst_t;
-
-/* type definition decoder instance */
-typedef struct iLBC_Dec_Inst_t_ {
-
- /* flag for frame size mode */
- WebRtc_Word16 mode;
-
- /* basic parameters for different frame sizes */
- WebRtc_Word16 blockl;
- WebRtc_Word16 nsub;
- WebRtc_Word16 nasub;
- WebRtc_Word16 no_of_bytes, no_of_words;
- WebRtc_Word16 lpc_n;
- WebRtc_Word16 state_short_len;
-
- /* synthesis filter state */
- WebRtc_Word16 syntMem[LPC_FILTERORDER];
-
- /* old LSF for interpolation */
- WebRtc_Word16 lsfdeqold[LPC_FILTERORDER];
-
- /* pitch lag estimated in enhancer and used in PLC */
- int last_lag;
-
- /* PLC state information */
- int consPLICount, prev_enh_pl;
- WebRtc_Word16 perSquare;
-
- WebRtc_Word16 prevScale, prevPLI;
- WebRtc_Word16 prevLag, prevLpc[LPC_FILTERORDER+1];
- WebRtc_Word16 prevResidual[NSUB_MAX*SUBL];
- WebRtc_Word16 seed;
-
- /* previous synthesis filter parameters */
-
- WebRtc_Word16 old_syntdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
-
- /* state of output HP filter */
- WebRtc_Word16 hpimemx[2];
- WebRtc_Word16 hpimemy[4];
-
- /* enhancer state information */
- int use_enhancer;
- WebRtc_Word16 enh_buf[ENH_BUFL+ENH_BUFL_FILTEROVERHEAD];
- WebRtc_Word16 enh_period[ENH_NBLOCKS_TOT];
-
-} iLBC_Dec_Inst_t;
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/do_plc.c b/trunk/src/modules/audio_coding/codecs/ilbc/do_plc.c
deleted file mode 100644
index 0dfae2b..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/do_plc.c
+++ /dev/null
@@ -1,308 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_DoThePlc.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-#include "comp_corr.h"
-#include "bw_expand.h"
-
-/*----------------------------------------------------------------*
- * Packet loss concealment routine. Conceals a residual signal
- * and LP parameters. If no packet loss, update state.
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_DoThePlc(
- WebRtc_Word16 *PLCresidual, /* (o) concealed residual */
- WebRtc_Word16 *PLClpc, /* (o) concealed LP parameters */
- WebRtc_Word16 PLI, /* (i) packet loss indicator
- 0 - no PL, 1 = PL */
- WebRtc_Word16 *decresidual, /* (i) decoded residual */
- WebRtc_Word16 *lpc, /* (i) decoded LPC (only used for no PL) */
- WebRtc_Word16 inlag, /* (i) pitch lag */
- iLBC_Dec_Inst_t *iLBCdec_inst
- /* (i/o) decoder instance */
- ){
- WebRtc_Word16 i, pick;
- WebRtc_Word32 cross, ener, cross_comp, ener_comp = 0;
- WebRtc_Word32 measure, maxMeasure, energy;
- WebRtc_Word16 max, crossSquareMax, crossSquare;
- WebRtc_Word16 j, lag, tmp1, tmp2, randlag;
- WebRtc_Word16 shift1, shift2, shift3, shiftMax;
- WebRtc_Word16 scale3;
- WebRtc_Word16 corrLen;
- WebRtc_Word32 tmpW32, tmp2W32;
- WebRtc_Word16 use_gain;
- WebRtc_Word16 tot_gain;
- WebRtc_Word16 max_perSquare;
- WebRtc_Word16 scale1, scale2;
- WebRtc_Word16 totscale;
- WebRtc_Word32 nom;
- WebRtc_Word16 denom;
- WebRtc_Word16 pitchfact;
- WebRtc_Word16 use_lag;
- int ind;
- WebRtc_Word16 randvec[BLOCKL_MAX];
-
- /* Packet Loss */
- if (PLI == 1) {
-
- (*iLBCdec_inst).consPLICount += 1;
-
- /* if previous frame not lost,
- determine pitch pred. gain */
-
- if (iLBCdec_inst->prevPLI != 1) {
-
- /* Maximum 60 samples are correlated, preserve as high accuracy
- as possible without getting overflow */
- max = WebRtcSpl_MaxAbsValueW16((*iLBCdec_inst).prevResidual, (WebRtc_Word16)iLBCdec_inst->blockl);
- scale3 = (WebRtcSpl_GetSizeInBits(max)<<1) - 25;
- if (scale3 < 0) {
- scale3 = 0;
- }
-
- /* Store scale for use when interpolating between the
- * concealment and the received packet */
- iLBCdec_inst->prevScale = scale3;
-
- /* Search around the previous lag +/-3 to find the
- best pitch period */
- lag = inlag - 3;
-
- /* Guard against getting outside the frame */
- corrLen = WEBRTC_SPL_MIN(60, iLBCdec_inst->blockl-(inlag+3));
-
- WebRtcIlbcfix_CompCorr( &cross, &ener,
- iLBCdec_inst->prevResidual, lag, iLBCdec_inst->blockl, corrLen, scale3);
-
- /* Normalize and store cross^2 and the number of shifts */
- shiftMax = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(cross))-15;
- crossSquareMax = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(WEBRTC_SPL_SHIFT_W32(cross, -shiftMax),
- WEBRTC_SPL_SHIFT_W32(cross, -shiftMax), 15);
-
- for (j=inlag-2;j<=inlag+3;j++) {
- WebRtcIlbcfix_CompCorr( &cross_comp, &ener_comp,
- iLBCdec_inst->prevResidual, j, iLBCdec_inst->blockl, corrLen, scale3);
-
- /* Use the criteria (corr*corr)/energy to compare if
- this lag is better or not. To avoid the division,
- do a cross multiplication */
- shift1 = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(cross_comp))-15;
- crossSquare = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(WEBRTC_SPL_SHIFT_W32(cross_comp, -shift1),
- WEBRTC_SPL_SHIFT_W32(cross_comp, -shift1), 15);
-
- shift2 = WebRtcSpl_GetSizeInBits(ener)-15;
- measure = WEBRTC_SPL_MUL_16_16(WEBRTC_SPL_SHIFT_W32(ener, -shift2),
- crossSquare);
-
- shift3 = WebRtcSpl_GetSizeInBits(ener_comp)-15;
- maxMeasure = WEBRTC_SPL_MUL_16_16(WEBRTC_SPL_SHIFT_W32(ener_comp, -shift3),
- crossSquareMax);
-
- /* Calculate shift value, so that the two measures can
- be put in the same Q domain */
- if(((shiftMax<<1)+shift3) > ((shift1<<1)+shift2)) {
- tmp1 = WEBRTC_SPL_MIN(31, (shiftMax<<1)+shift3-(shift1<<1)-shift2);
- tmp2 = 0;
- } else {
- tmp1 = 0;
- tmp2 = WEBRTC_SPL_MIN(31, (shift1<<1)+shift2-(shiftMax<<1)-shift3);
- }
-
- if ((measure>>tmp1) > (maxMeasure>>tmp2)) {
- /* New lag is better => record lag, measure and domain */
- lag = j;
- crossSquareMax = crossSquare;
- cross = cross_comp;
- shiftMax = shift1;
- ener = ener_comp;
- }
- }
-
- /* Calculate the periodicity for the lag with the maximum correlation.
-
- Definition of the periodicity:
- abs(corr(vec1, vec2))/(sqrt(energy(vec1))*sqrt(energy(vec2)))
-
- Work in the Square domain to simplify the calculations
- max_perSquare is less than 1 (in Q15)
- */
- tmp2W32=WebRtcSpl_DotProductWithScale(&iLBCdec_inst->prevResidual[iLBCdec_inst->blockl-corrLen],
- &iLBCdec_inst->prevResidual[iLBCdec_inst->blockl-corrLen],
- corrLen, scale3);
-
- if ((tmp2W32>0)&&(ener_comp>0)) {
- /* norm energies to WebRtc_Word16, compute the product of the energies and
- use the upper WebRtc_Word16 as the denominator */
-
- scale1=(WebRtc_Word16)WebRtcSpl_NormW32(tmp2W32)-16;
- tmp1=(WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(tmp2W32, scale1);
-
- scale2=(WebRtc_Word16)WebRtcSpl_NormW32(ener)-16;
- tmp2=(WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(ener, scale2);
- denom=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp1, tmp2, 16); /* denom in Q(scale1+scale2-16) */
-
- /* Square the cross correlation and norm it such that max_perSquare
- will be in Q15 after the division */
-
- totscale = scale1+scale2-1;
- tmp1 = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(cross, (totscale>>1));
- tmp2 = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(cross, totscale-(totscale>>1));
-
- nom = WEBRTC_SPL_MUL_16_16(tmp1, tmp2);
- max_perSquare = (WebRtc_Word16)WebRtcSpl_DivW32W16(nom, denom);
-
- } else {
- max_perSquare = 0;
- }
- }
-
- /* previous frame lost, use recorded lag and gain */
-
- else {
- lag = iLBCdec_inst->prevLag;
- max_perSquare = iLBCdec_inst->perSquare;
- }
-
- /* Attenuate signal and scale down pitch pred gain if
- several frames lost consecutively */
-
- use_gain = 32767; /* 1.0 in Q15 */
-
- if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>320) {
- use_gain = 29491; /* 0.9 in Q15 */
- } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>640) {
- use_gain = 22938; /* 0.7 in Q15 */
- } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>960) {
- use_gain = 16384; /* 0.5 in Q15 */
- } else if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>1280) {
- use_gain = 0; /* 0.0 in Q15 */
- }
-
- /* Compute mixing factor of picth repeatition and noise:
- for max_per>0.7 set periodicity to 1.0
- 0.4<max_per<0.7 set periodicity to (maxper-0.4)/0.7-0.4)
- max_per<0.4 set periodicity to 0.0
- */
-
- if (max_perSquare>7868) { /* periodicity > 0.7 (0.7^4=0.2401 in Q15) */
- pitchfact = 32767;
- } else if (max_perSquare>839) { /* 0.4 < periodicity < 0.7 (0.4^4=0.0256 in Q15) */
- /* find best index and interpolate from that */
- ind = 5;
- while ((max_perSquare<WebRtcIlbcfix_kPlcPerSqr[ind])&&(ind>0)) {
- ind--;
- }
- /* pitch fact is approximated by first order */
- tmpW32 = (WebRtc_Word32)WebRtcIlbcfix_kPlcPitchFact[ind] +
- WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kPlcPfSlope[ind], (max_perSquare-WebRtcIlbcfix_kPlcPerSqr[ind]), 11);
-
- pitchfact = (WebRtc_Word16)WEBRTC_SPL_MIN(tmpW32, 32767); /* guard against overflow */
-
- } else { /* periodicity < 0.4 */
- pitchfact = 0;
- }
-
- /* avoid repetition of same pitch cycle (buzzyness) */
- use_lag = lag;
- if (lag<80) {
- use_lag = 2*lag;
- }
-
- /* compute concealed residual */
- energy = 0;
-
- for (i=0; i<iLBCdec_inst->blockl; i++) {
-
- /* noise component - 52 < randlagFIX < 117 */
- iLBCdec_inst->seed = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16(iLBCdec_inst->seed, 31821)+(WebRtc_Word32)13849);
- randlag = 53 + (WebRtc_Word16)(iLBCdec_inst->seed & 63);
-
- pick = i - randlag;
-
- if (pick < 0) {
- randvec[i] = iLBCdec_inst->prevResidual[iLBCdec_inst->blockl+pick];
- } else {
- randvec[i] = iLBCdec_inst->prevResidual[pick];
- }
-
- /* pitch repeatition component */
- pick = i - use_lag;
-
- if (pick < 0) {
- PLCresidual[i] = iLBCdec_inst->prevResidual[iLBCdec_inst->blockl+pick];
- } else {
- PLCresidual[i] = PLCresidual[pick];
- }
-
- /* Attinuate total gain for each 10 ms */
- if (i<80) {
- tot_gain=use_gain;
- } else if (i<160) {
- tot_gain=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(31130, use_gain, 15); /* 0.95*use_gain */
- } else {
- tot_gain=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(29491, use_gain, 15); /* 0.9*use_gain */
- }
-
-
- /* mix noise and pitch repeatition */
-
- PLCresidual[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tot_gain,
- (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32( (WEBRTC_SPL_MUL_16_16(pitchfact, PLCresidual[i]) +
- WEBRTC_SPL_MUL_16_16((32767-pitchfact), randvec[i]) + 16384),
- 15),
- 15);
-
- /* Shifting down the result one step extra to ensure that no overflow
- will occur */
- energy += WEBRTC_SPL_MUL_16_16_RSFT(PLCresidual[i],
- PLCresidual[i], (iLBCdec_inst->prevScale+1));
-
- }
-
- /* less than 30 dB, use only noise */
- if (energy < (WEBRTC_SPL_SHIFT_W32(((WebRtc_Word32)iLBCdec_inst->blockl*900),-(iLBCdec_inst->prevScale+1)))) {
- energy = 0;
- for (i=0; i<iLBCdec_inst->blockl; i++) {
- PLCresidual[i] = randvec[i];
- }
- }
-
- /* use the old LPC */
- WEBRTC_SPL_MEMCPY_W16(PLClpc, (*iLBCdec_inst).prevLpc, LPC_FILTERORDER+1);
-
- /* Update state in case there are multiple frame losses */
- iLBCdec_inst->prevLag = lag;
- iLBCdec_inst->perSquare = max_perSquare;
- }
-
- /* no packet loss, copy input */
-
- else {
- WEBRTC_SPL_MEMCPY_W16(PLCresidual, decresidual, iLBCdec_inst->blockl);
- WEBRTC_SPL_MEMCPY_W16(PLClpc, lpc, (LPC_FILTERORDER+1));
- iLBCdec_inst->consPLICount = 0;
- }
-
- /* update state */
- iLBCdec_inst->prevPLI = PLI;
- WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->prevLpc, PLClpc, (LPC_FILTERORDER+1));
- WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->prevResidual, PLCresidual, iLBCdec_inst->blockl);
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/do_plc.h b/trunk/src/modules/audio_coding/codecs/ilbc/do_plc.h
deleted file mode 100644
index c5bcc52..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/do_plc.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_DoThePlc.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_DO_PLC_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Packet loss concealment routine. Conceals a residual signal
- * and LP parameters. If no packet loss, update state.
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_DoThePlc(
- WebRtc_Word16 *PLCresidual, /* (o) concealed residual */
- WebRtc_Word16 *PLClpc, /* (o) concealed LP parameters */
- WebRtc_Word16 PLI, /* (i) packet loss indicator
- 0 - no PL, 1 = PL */
- WebRtc_Word16 *decresidual, /* (i) decoded residual */
- WebRtc_Word16 *lpc, /* (i) decoded LPC (only used for no PL) */
- WebRtc_Word16 inlag, /* (i) pitch lag */
- iLBC_Dec_Inst_t *iLBCdec_inst
- /* (i/o) decoder instance */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/documentation/rfc3951.txt b/trunk/src/modules/audio_coding/codecs/ilbc/documentation/rfc3951.txt
deleted file mode 100644
index d4fba08..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/documentation/rfc3951.txt
+++ /dev/null
@@ -1,10867 +0,0 @@
-
-
-
-
-
-
-Network Working Group S. Andersen
-Request for Comments: 3951 Aalborg University
-Category: Experimental A. Duric
- Telio
- H. Astrom
- R. Hagen
- W. Kleijn
- J. Linden
- Global IP Sound
- December 2004
-
-
- Internet Low Bit Rate Codec (iLBC)
-
-Status of this Memo
-
- This memo defines an Experimental Protocol for the Internet
- community. It does not specify an Internet standard of any kind.
- Discussion and suggestions for improvement are requested.
- Distribution of this memo is unlimited.
-
-Copyright Notice
-
- Copyright (C) The Internet Society (2004).
-
-Abstract
-
- This document specifies a speech codec suitable for robust voice
- communication over IP. The codec is developed by Global IP Sound
- (GIPS). It is designed for narrow band speech and results in a
- payload bit rate of 13.33 kbit/s for 30 ms frames and 15.20 kbit/s
- for 20 ms frames. The codec enables graceful speech quality
- degradation in the case of lost frames, which occurs in connection
- with lost or delayed IP packets.
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-Andersen, et al. Experimental [Page 1]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
-Table of Contents
-
- 1. Introduction . . . . . . . . . . . . . . . . . . . . . . . . . 4
- 2. Outline of the Codec . . . . . . . . . . . . . . . . . . . . . 5
- 2.1. Encoder. . . . . . . . . . . . . . . . . . . . . . . . . 5
- 2.2. Decoder. . . . . . . . . . . . . . . . . . . . . . . . . 7
- 3. Encoder Principles . . . . . . . . . . . . . . . . . . . . . . 7
- 3.1. Pre-processing . . . . . . . . . . . . . . . . . . . . . 9
- 3.2. LPC Analysis and Quantization. . . . . . . . . . . . . . 9
- 3.2.1. Computation of Autocorrelation Coefficients. . . 10
- 3.2.2. Computation of LPC Coefficients. . . . . . . . . 11
- 3.2.3. Computation of LSF Coefficients from LPC
- Coefficients . . . . . . . . . . . . . . . . . . 11
- 3.2.4. Quantization of LSF Coefficients . . . . . . . . 12
- 3.2.5. Stability Check of LSF Coefficients. . . . . . . 13
- 3.2.6. Interpolation of LSF Coefficients. . . . . . . . 13
- 3.2.7. LPC Analysis and Quantization for 20 ms Frames . 14
- 3.3. Calculation of the Residual. . . . . . . . . . . . . . . 15
- 3.4. Perceptual Weighting Filter. . . . . . . . . . . . . . . 15
- 3.5. Start State Encoder. . . . . . . . . . . . . . . . . . . 15
- 3.5.1. Start State Estimation . . . . . . . . . . . . . 16
- 3.5.2. All-Pass Filtering and Scale Quantization. . . . 17
- 3.5.3. Scalar Quantization. . . . . . . . . . . . . . . 18
- 3.6. Encoding the Remaining Samples . . . . . . . . . . . . . 19
- 3.6.1. Codebook Memory. . . . . . . . . . . . . . . . . 20
- 3.6.2. Perceptual Weighting of Codebook Memory
- and Target . . . . . . . . . . . . . . . . . . . 22
- 3.6.3. Codebook Creation. . . . . . . . . . . . . . . . 23
- 3.6.3.1. Creation of a Base Codebook . . . . . . 23
- 3.6.3.2. Codebook Expansion. . . . . . . . . . . 24
- 3.6.3.3. Codebook Augmentation . . . . . . . . . 24
- 3.6.4. Codebook Search. . . . . . . . . . . . . . . . . 26
- 3.6.4.1. Codebook Search at Each Stage . . . . . 26
- 3.6.4.2. Gain Quantization at Each Stage . . . . 27
- 3.6.4.3. Preparation of Target for Next Stage. . 28
- 3.7. Gain Correction Encoding . . . . . . . . . . . . . . . . 28
- 3.8. Bitstream Definition . . . . . . . . . . . . . . . . . . 29
- 4. Decoder Principles . . . . . . . . . . . . . . . . . . . . . . 32
- 4.1. LPC Filter Reconstruction. . . . . . . . . . . . . . . . 33
- 4.2. Start State Reconstruction . . . . . . . . . . . . . . . 33
- 4.3. Excitation Decoding Loop . . . . . . . . . . . . . . . . 34
- 4.4. Multistage Adaptive Codebook Decoding. . . . . . . . . . 35
- 4.4.1. Construction of the Decoded Excitation Signal. . 35
- 4.5. Packet Loss Concealment. . . . . . . . . . . . . . . . . 35
- 4.5.1. Block Received Correctly and Previous Block
- Also Received. . . . . . . . . . . . . . . . . . 35
- 4.5.2. Block Not Received . . . . . . . . . . . . . . . 36
-
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- 4.5.3. Block Received Correctly When Previous Block
- Not Received . . . . . . . . . . . . . . . . . . 36
- 4.6. Enhancement. . . . . . . . . . . . . . . . . . . . . . . 37
- 4.6.1. Estimating the Pitch . . . . . . . . . . . . . . 39
- 4.6.2. Determination of the Pitch-Synchronous
- Sequences. . . . . . . . . . . . . . . . . . . . 39
- 4.6.3. Calculation of the Smoothed Excitation . . . . . 41
- 4.6.4. Enhancer Criterion . . . . . . . . . . . . . . . 41
- 4.6.5. Enhancing the Excitation . . . . . . . . . . . . 42
- 4.7. Synthesis Filtering. . . . . . . . . . . . . . . . . . . 43
- 4.8. Post Filtering . . . . . . . . . . . . . . . . . . . . . 43
- 5. Security Considerations. . . . . . . . . . . . . . . . . . . . 43
- 6. Evaluation of the iLBC Implementations . . . . . . . . . . . . 43
- 7. References . . . . . . . . . . . . . . . . . . . . . . . . . . 43
- 7.1. Normative References . . . . . . . . . . . . . . . . . . 43
- 7.2. Informative References . . . . . . . . . . . . . . . . . 44
- 8. ACKNOWLEDGEMENTS . . . . . . . . . . . . . . . . . . . . . . . 44
- APPENDIX A: Reference Implementation . . . . . . . . . . . . . . . 45
- A.1. iLBC_test.c. . . . . . . . . . . . . . . . . . . . . . . 46
- A.2 iLBC_encode.h. . . . . . . . . . . . . . . . . . . . . . 52
- A.3. iLBC_encode.c. . . . . . . . . . . . . . . . . . . . . . 53
- A.4. iLBC_decode.h. . . . . . . . . . . . . . . . . . . . . . 63
- A.5. iLBC_decode.c. . . . . . . . . . . . . . . . . . . . . . 64
- A.6. iLBC_define.h. . . . . . . . . . . . . . . . . . . . . . 76
- A.7. constants.h. . . . . . . . . . . . . . . . . . . . . . . 80
- A.8. constants.c. . . . . . . . . . . . . . . . . . . . . . . 82
- A.9. anaFilter.h. . . . . . . . . . . . . . . . . . . . . . . 96
- A.10. anaFilter.c. . . . . . . . . . . . . . . . . . . . . . . 97
- A.11. createCB.h . . . . . . . . . . . . . . . . . . . . . . . 98
- A.12. createCB.c . . . . . . . . . . . . . . . . . . . . . . . 99
- A.13. doCPLC.h . . . . . . . . . . . . . . . . . . . . . . . .104
- A.14. doCPLC.c . . . . . . . . . . . . . . . . . . . . . . . .104
- A.15. enhancer.h . . . . . . . . . . . . . . . . . . . . . . .109
- A.16. enhancer.c . . . . . . . . . . . . . . . . . . . . . . .110
- A.17. filter.h . . . . . . . . . . . . . . . . . . . . . . . .123
- A.18. filter.c . . . . . . . . . . . . . . . . . . . . . . . .125
- A.19. FrameClassify.h. . . . . . . . . . . . . . . . . . . . .128
- A.20. FrameClassify.c. . . . . . . . . . . . . . . . . . . . .129
- A.21. gainquant.h. . . . . . . . . . . . . . . . . . . . . . .131
- A.22. gainquant.c. . . . . . . . . . . . . . . . . . . . . . .131
- A.23. getCBvec.h . . . . . . . . . . . . . . . . . . . . . . .134
- A.24. getCBvec.c . . . . . . . . . . . . . . . . . . . . . . .134
- A.25. helpfun.h. . . . . . . . . . . . . . . . . . . . . . . .138
- A.26. helpfun.c. . . . . . . . . . . . . . . . . . . . . . . .140
- A.27. hpInput.h. . . . . . . . . . . . . . . . . . . . . . . .146
- A.28. hpInput.c. . . . . . . . . . . . . . . . . . . . . . . .146
- A.29. hpOutput.h . . . . . . . . . . . . . . . . . . . . . . .148
- A.30. hpOutput.c . . . . . . . . . . . . . . . . . . . . . . .148
-
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- A.31. iCBConstruct.h . . . . . . . . . . . . . . . . . . . . .149
- A.32. iCBConstruct.c . . . . . . . . . . . . . . . . . . . . .150
- A.33. iCBSearch.h. . . . . . . . . . . . . . . . . . . . . . .152
- A.34. iCBSearch.c. . . . . . . . . . . . . . . . . . . . . . .153
- A.35. LPCdecode.h. . . . . . . . . . . . . . . . . . . . . . .163
- A.36. LPCdecode.c. . . . . . . . . . . . . . . . . . . . . . .164
- A.37. LPCencode.h. . . . . . . . . . . . . . . . . . . . . . .167
- A.38. LPCencode.c. . . . . . . . . . . . . . . . . . . . . . .167
- A.39. lsf.h. . . . . . . . . . . . . . . . . . . . . . . . . .172
- A.40. lsf.c. . . . . . . . . . . . . . . . . . . . . . . . . .172
- A.41. packing.h. . . . . . . . . . . . . . . . . . . . . . . .178
- A.42. packing.c. . . . . . . . . . . . . . . . . . . . . . . .179
- A.43. StateConstructW.h. . . . . . . . . . . . . . . . . . . .182
- A.44. StateConstructW.c. . . . . . . . . . . . . . . . . . . .183
- A.45. StateSearchW.h . . . . . . . . . . . . . . . . . . . . .185
- A.46. StateSearchW.c . . . . . . . . . . . . . . . . . . . . .186
- A.47. syntFilter.h . . . . . . . . . . . . . . . . . . . . . .190
- A.48. syntFilter.c . . . . . . . . . . . . . . . . . . . . . .190
- Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . .192
- Full Copyright Statement . . . . . . . . . . . . . . . . . . . . .194
-
-1. Introduction
-
- This document contains the description of an algorithm for the coding
- of speech signals sampled at 8 kHz. The algorithm, called iLBC, uses
- a block-independent linear-predictive coding (LPC) algorithm and has
- support for two basic frame lengths: 20 ms at 15.2 kbit/s and 30 ms
- at 13.33 kbit/s. When the codec operates at block lengths of 20 ms,
- it produces 304 bits per block, which SHOULD be packetized as in [1].
- Similarly, for block lengths of 30 ms it produces 400 bits per block,
- which SHOULD be packetized as in [1]. The two modes for the
- different frame sizes operate in a very similar way. When they
- differ it is explicitly stated in the text, usually with the notation
- x/y, where x refers to the 20 ms mode and y refers to the 30 ms mode.
-
- The described algorithm results in a speech coding system with a
- controlled response to packet losses similar to what is known from
- pulse code modulation (PCM) with packet loss concealment (PLC), such
- as the ITU-T G.711 standard [4], which operates at a fixed bit rate
- of 64 kbit/s. At the same time, the described algorithm enables
- fixed bit rate coding with a quality-versus-bit rate tradeoff close
- to state-of-the-art. A suitable RTP payload format for the iLBC
- codec is specified in [1].
-
- Some of the applications for which this coder is suitable are real
- time communications such as telephony and videoconferencing,
- streaming audio, archival, and messaging.
-
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- Cable Television Laboratories (CableLabs(R)) has adopted iLBC as a
- mandatory PacketCable(TM) audio codec standard for VoIP over Cable
- applications [3].
-
- This document is organized as follows. Section 2 gives a brief
- outline of the codec. The specific encoder and decoder algorithms
- are explained in sections 3 and 4, respectively. Appendix A provides
- a c-code reference implementation.
-
- The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
- "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
- document are to be interpreted as described in BCP 14, RFC 2119 [2].
-
-2. Outline of the Codec
-
- The codec consists of an encoder and a decoder as described in
- sections 2.1 and 2.2, respectively.
-
- The essence of the codec is LPC and block-based coding of the LPC
- residual signal. For each 160/240 (20 ms/30 ms) sample block, the
- following major steps are performed: A set of LPC filters are
- computed, and the speech signal is filtered through them to produce
- the residual signal. The codec uses scalar quantization of the
- dominant part, in terms of energy, of the residual signal for the
- block. The dominant state is of length 57/58 (20 ms/30 ms) samples
- and forms a start state for dynamic codebooks constructed from the
- already coded parts of the residual signal. These dynamic codebooks
- are used to code the remaining parts of the residual signal. By this
- method, coding independence between blocks is achieved, resulting in
- elimination of propagation of perceptual degradations due to packet
- loss. The method facilitates high-quality packet loss concealment
- (PLC).
-
-2.1. Encoder
-
- The input to the encoder SHOULD be 16 bit uniform PCM sampled at 8
- kHz. It SHOULD be partitioned into blocks of BLOCKL=160/240 samples
- for the 20/30 ms frame size. Each block is divided into NSUB=4/6
- consecutive sub-blocks of SUBL=40 samples each. For 30 ms frame
- size, the encoder performs two LPC_FILTERORDER=10 linear-predictive
- coding (LPC) analyses. The first analysis applies a smooth window
- centered over the second sub-block and extending to the middle of the
- fifth sub-block. The second LPC analysis applies a smooth asymmetric
- window centered over the fifth sub-block and extending to the end of
- the sixth sub-block. For 20 ms frame size, one LPC_FILTERORDER=10
- linear-predictive coding (LPC) analysis is performed with a smooth
- window centered over the third sub-frame.
-
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- For each of the LPC analyses, a set of line-spectral frequencies
- (LSFs) are obtained, quantized, and interpolated to obtain LSF
- coefficients for each sub-block. Subsequently, the LPC residual is
- computed by using the quantized and interpolated LPC analysis
- filters.
-
- The two consecutive sub-blocks of the residual exhibiting the maximal
- weighted energy are identified. Within these two sub-blocks, the
- start state (segment) is selected from two choices: the first 57/58
- samples or the last 57/58 samples of the two consecutive sub-blocks.
- The selected segment is the one of higher energy. The start state is
- encoded with scalar quantization.
-
- A dynamic codebook encoding procedure is used to encode 1) the 23/22
- (20 ms/30 ms) remaining samples in the two sub-blocks containing the
- start state; 2) the sub-blocks after the start state in time; and 3)
- the sub-blocks before the start state in time. Thus, the encoding
- target can be either the 23/22 samples remaining of the two sub-
- blocks containing the start state or a 40-sample sub-block. This
- target can consist of samples indexed forward in time or backward in
- time, depending on the location of the start state.
-
- The codebook coding is based on an adaptive codebook built from a
- codebook memory that contains decoded LPC excitation samples from the
- already encoded part of the block. These samples are indexed in the
- same time direction as the target vector, ending at the sample
- instant prior to the first sample instant represented in the target
- vector. The codebook is used in CB_NSTAGES=3 stages in a successive
- refinement approach, and the resulting three code vector gains are
- encoded with 5-, 4-, and 3-bit scalar quantization, respectively.
-
- The codebook search method employs noise shaping derived from the LPC
- filters, and the main decision criterion is to minimize the squared
- error between the target vector and the code vectors. Each code
- vector in this codebook comes from one of CB_EXPAND=2 codebook
- sections. The first section is filled with delayed, already encoded
- residual vectors. The code vectors of the second codebook section
- are constructed by predefined linear combinations of vectors in the
- first section of the codebook.
-
- As codebook encoding with squared-error matching is known to produce
- a coded signal of less power than does the scalar quantized start
- state signal, a gain re-scaling method is implemented by a refined
- search for a better set of codebook gains in terms of power matching
- after encoding. This is done by searching for a higher value of the
- gain factor for the first stage codebook, as the subsequent stage
- codebook gains are scaled by the first stage gain.
-
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-2.2. Decoder
-
- Typically for packet communications, a jitter buffer placed at the
- receiving end decides whether the packet containing an encoded signal
- block has been received or lost. This logic is not part of the codec
- described here. For each encoded signal block received the decoder
- performs a decoding. For each lost signal block, the decoder
- performs a PLC operation.
-
- The decoding for each block starts by decoding and interpolating the
- LPC coefficients. Subsequently the start state is decoded.
-
- For codebook-encoded segments, each segment is decoded by
- constructing the three code vectors given by the received codebook
- indices in the same way that the code vectors were constructed in the
- encoder. The three gain factors are also decoded and the resulting
- decoded signal is given by the sum of the three codebook vectors
- scaled with respective gain.
-
- An enhancement algorithm is applied to the reconstructed excitation
- signal. This enhancement augments the periodicity of voiced speech
- regions. The enhancement is optimized under the constraint that the
- modification signal (defined as the difference between the enhanced
- excitation and the excitation signal prior to enhancement) has a
- short-time energy that does not exceed a preset fraction of the
- short-time energy of the excitation signal prior to enhancement.
-
- A packet loss concealment (PLC) operation is easily embedded in the
- decoder. The PLC operation can, e.g., be based on repeating LPC
- filters and obtaining the LPC residual signal by using a long-term
- prediction estimate from previous residual blocks.
-
-3. Encoder Principles
-
- The following block diagram is an overview of all the components of
- the iLBC encoding procedure. The description of the blocks contains
- references to the section where that particular procedure is further
- described.
-
-
-
-
-
-
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- +-----------+ +---------+ +---------+
- speech -> | 1. Pre P | -> | 2. LPC | -> | 3. Ana | ->
- +-----------+ +---------+ +---------+
-
- +---------------+ +--------------+
- -> | 4. Start Sel | ->| 5. Scalar Qu | ->
- +---------------+ +--------------+
-
- +--------------+ +---------------+
- -> |6. CB Search | -> | 7. Packetize | -> payload
- | +--------------+ | +---------------+
- ----<---------<------
- sub-frame 0..2/4 (20 ms/30 ms)
-
- Figure 3.1. Flow chart of the iLBC encoder
-
- 1. Pre-process speech with a HP filter, if needed (section 3.1).
-
- 2. Compute LPC parameters, quantize, and interpolate (section 3.2).
-
- 3. Use analysis filters on speech to compute residual (section 3.3).
-
- 4. Select position of 57/58-sample start state (section 3.5).
-
- 5. Quantize the 57/58-sample start state with scalar quantization
- (section 3.5).
-
- 6. Search the codebook for each sub-frame. Start with 23/22 sample
- block, then encode sub-blocks forward in time, and then encode
- sub-blocks backward in time. For each block, the steps in Figure
- 3.4 are performed (section 3.6).
-
- 7. Packetize the bits into the payload specified in Table 3.2.
-
- The input to the encoder SHOULD be 16-bit uniform PCM sampled at 8
- kHz. Also it SHOULD be partitioned into blocks of BLOCKL=160/240
- samples. Each block input to the encoder is divided into NSUB=4/6
- consecutive sub-blocks of SUBL=40 samples each.
-
-
-
-
-
-
-
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- 0 39 79 119 159
- +---------------------------------------+
- | 1 | 2 | 3 | 4 |
- +---------------------------------------+
- 20 ms frame
-
- 0 39 79 119 159 199 239
- +-----------------------------------------------------------+
- | 1 | 2 | 3 | 4 | 5 | 6 |
- +-----------------------------------------------------------+
- 30 ms frame
- Figure 3.2. One input block to the encoder for 20 ms (with four sub-
- frames) and 30 ms (with six sub-frames).
-
-3.1. Pre-processing
-
- In some applications, the recorded speech signal contains DC level
- and/or 50/60 Hz noise. If these components have not been removed
- prior to the encoder call, they should be removed by a high-pass
- filter. A reference implementation of this, using a filter with a
- cutoff frequency of 90 Hz, can be found in Appendix A.28.
-
-3.2. LPC Analysis and Quantization
-
- The input to the LPC analysis module is a possibly high-pass filtered
- speech buffer, speech_hp, that contains 240/300 (LPC_LOOKBACK +
- BLOCKL = 80/60 + 160/240 = 240/300) speech samples, where samples 0
- through 79/59 are from the previous block and samples 80/60 through
- 239/299 are from the current block. No look-ahead into the next
- block is used. For the very first block processed, the look-back
- samples are assumed to be zeros.
-
- For each input block, the LPC analysis calculates one/two set(s) of
- LPC_FILTERORDER=10 LPC filter coefficients using the autocorrelation
- method and the Levinson-Durbin recursion. These coefficients are
- converted to the Line Spectrum Frequency representation. In the 20
- ms case, the single lsf set represents the spectral characteristics
- as measured at the center of the third sub-block. For 30 ms frames,
- the first set, lsf1, represents the spectral properties of the input
- signal at the center of the second sub-block, and the other set,
- lsf2, represents the spectral characteristics as measured at the
- center of the fifth sub-block. The details of the computation for 30
- ms frames are described in sections 3.2.1 through 3.2.6. Section
- 3.2.7 explains how the LPC Analysis and Quantization differs for 20
- ms frames.
-
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-3.2.1. Computation of Autocorrelation Coefficients
-
- The first step in the LPC analysis procedure is to calculate
- autocorrelation coefficients by using windowed speech samples. This
- windowing is the only difference in the LPC analysis procedure for
- the two sets of coefficients. For the first set, a 240-sample-long
- standard symmetric Hanning window is applied to samples 0 through 239
- of the input data. The first window, lpc_winTbl, is defined as
-
- lpc_winTbl[i]= 0.5 * (1.0 - cos((2*PI*(i+1))/(BLOCKL+1)));
- i=0,...,119
- lpc_winTbl[i] = winTbl[BLOCKL - i - 1]; i=120,...,239
-
- The windowed speech speech_hp_win1 is then obtained by multiplying
- the first 240 samples of the input speech buffer with the window
- coefficients:
-
- speech_hp_win1[i] = speech_hp[i] * lpc_winTbl[i];
- i=0,...,BLOCKL-1
-
- From these 240 windowed speech samples, 11 (LPC_FILTERORDER + 1)
- autocorrelation coefficients, acf1, are calculated:
-
- acf1[lag] += speech_hp_win1[n] * speech_hp_win1[n + lag];
- lag=0,...,LPC_FILTERORDER; n=0,...,BLOCKL-lag-1
-
- In order to make the analysis more robust against numerical precision
- problems, a spectral smoothing procedure is applied by windowing the
- autocorrelation coefficients before the LPC coefficients are
- computed. Also, a white noise floor is added to the autocorrelation
- function by multiplying coefficient zero by 1.0001 (40dB below the
- energy of the windowed speech signal). These two steps are
- implemented by multiplying the autocorrelation coefficients with the
- following window:
-
- lpc_lagwinTbl[0] = 1.0001;
- lpc_lagwinTbl[i] = exp(-0.5 * ((2 * PI * 60.0 * i) /FS)^2);
- i=1,...,LPC_FILTERORDER
- where FS=8000 is the sampling frequency
-
- Then, the windowed acf function acf1_win is obtained by
-
- acf1_win[i] = acf1[i] * lpc_lagwinTbl[i];
- i=0,...,LPC_FILTERORDER
-
- The second set of autocorrelation coefficients, acf2_win, are
- obtained in a similar manner. The window, lpc_asymwinTbl, is applied
- to samples 60 through 299, i.e., the entire current block. The
-
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- window consists of two segments, the first (samples 0 to 219) being
- half a Hanning window with length 440 and the second a quarter of a
- cycle of a cosine wave. By using this asymmetric window, an LPC
- analysis centered in the fifth sub-block is obtained without the need
- for any look-ahead, which would add delay. The asymmetric window is
- defined as
-
- lpc_asymwinTbl[i] = (sin(PI * (i + 1) / 441))^2; i=0,...,219
-
- lpc_asymwinTbl[i] = cos((i - 220) * PI / 40); i=220,...,239
-
- and the windowed speech is computed by
-
- speech_hp_win2[i] = speech_hp[i + LPC_LOOKBACK] *
- lpc_asymwinTbl[i]; i=0,....BLOCKL-1
-
- The windowed autocorrelation coefficients are then obtained in
- exactly the same way as for the first analysis instance.
-
- The generation of the windows lpc_winTbl, lpc_asymwinTbl, and
- lpc_lagwinTbl are typically done in advance, and the arrays are
- stored in ROM rather than repeating the calculation for every block.
-
-3.2.2. Computation of LPC Coefficients
-
- From the 2 x 11 smoothed autocorrelation coefficients, acf1_win and
- acf2_win, the 2 x 11 LPC coefficients, lp1 and lp2, are calculated
- in the same way for both analysis locations by using the well known
- Levinson-Durbin recursion. The first LPC coefficient is always 1.0,
- resulting in ten unique coefficients.
-
- After determining the LPC coefficients, a bandwidth expansion
- procedure is applied to smooth the spectral peaks in the
- short-term spectrum. The bandwidth addition is obtained by the
- following modification of the LPC coefficients:
-
- lp1_bw[i] = lp1[i] * chirp^i; i=0,...,LPC_FILTERORDER
- lp2_bw[i] = lp2[i] * chirp^i; i=0,...,LPC_FILTERORDER
-
- where "chirp" is a real number between 0 and 1. It is RECOMMENDED to
- use a value of 0.9.
-
-3.2.3. Computation of LSF Coefficients from LPC Coefficients
-
- Thus far, two sets of LPC coefficients that represent the short-term
- spectral characteristics of the speech signal for two different time
- locations within the current block have been determined. These
- coefficients SHOULD be quantized and interpolated. Before this is
-
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- done, it is advantageous to convert the LPC parameters into another
- type of representation called Line Spectral Frequencies (LSF). The
- LSF parameters are used because they are better suited for
- quantization and interpolation than the regular LPC coefficients.
- Many computationally efficient methods for calculating the LSFs from
- the LPC coefficients have been proposed in the literature. The
- detailed implementation of one applicable method can be found in
- Appendix A.26. The two arrays of LSF coefficients obtained, lsf1 and
- lsf2, are of dimension 10 (LPC_FILTERORDER).
-
-3.2.4. Quantization of LSF Coefficients
-
- Because the LPC filters defined by the two sets of LSFs are also
- needed in the decoder, the LSF parameters need to be quantized and
- transmitted as side information. The total number of bits required
- to represent the quantization of the two LSF representations for one
- block of speech is 40, with 20 bits used for each of lsf1 and lsf2.
-
- For computational and storage reasons, the LSF vectors are quantized
- using three-split vector quantization (VQ). That is, the LSF vectors
- are split into three sub-vectors that are each quantized with a
- regular VQ. The quantized versions of lsf1 and lsf2, qlsf1 and
- qlsf2, are obtained by using the same memoryless split VQ. The
- length of each of these two LSF vectors is 10, and they are split
- into three sub-vectors containing 3, 3, and 4 values, respectively.
-
- For each of the sub-vectors, a separate codebook of quantized values
- has been designed with a standard VQ training method for a large
- database containing speech from a large number of speakers recorded
- under various conditions. The size of each of the three codebooks
- associated with the split definitions above is
-
- int size_lsfCbTbl[LSF_NSPLIT] = {64,128,128};
-
- The actual values of the vector quantization codebook that must be
- used can be found in the reference code of Appendix A. Both sets of
- LSF coefficients, lsf1 and lsf2, are quantized with a standard
- memoryless split vector quantization (VQ) structure using the squared
- error criterion in the LSF domain. The split VQ quantization
- consists of the following steps:
-
- 1) Quantize the first three LSF coefficients (1 - 3) with a VQ
- codebook of size 64.
- 2) Quantize the next three LSF coefficients 4 - 6 with VQ a codebook
- of size 128.
- 3) Quantize the last four LSF coefficients (7 - 10) with a VQ
- codebook of size 128.
-
-
-
-
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-
- This procedure, repeated for lsf1 and lsf2, gives six quantization
- indices and the quantized sets of LSF coefficients qlsf1 and qlsf2.
- Each set of three indices is encoded with 6 + 7 + 7 = 20 bits. The
- total number of bits used for LSF quantization in a block is thus 40
- bits.
-
-3.2.5. Stability Check of LSF Coefficients
-
- The LSF representation of the LPC filter has the convenient property
- that the coefficients are ordered by increasing value, i.e., lsf(n-1)
- < lsf(n), 0 < n < 10, if the corresponding synthesis filter is
- stable. As we are employing a split VQ scheme, it is possible that
- at the split boundaries the LSF coefficients are not ordered
- correctly and hence that the corresponding LP filter is unstable. To
- ensure that the filter used is stable, a stability check is performed
- for the quantized LSF vectors. If it turns out that the coefficients
- are not ordered appropriately (with a safety margin of 50 Hz to
- ensure that formant peaks are not too narrow), they will be moved
- apart. The detailed method for this can be found in Appendix A.40.
- The same procedure is performed in the decoder. This ensures that
- exactly the same LSF representations are used in both encoder and
- decoder.
-
-3.2.6. Interpolation of LSF Coefficients
-
- From the two sets of LSF coefficients that are computed for each
- block of speech, different LSFs are obtained for each sub-block by
- means of interpolation. This procedure is performed for the original
- LSFs (lsf1 and lsf2), as well as the quantized versions qlsf1 and
- qlsf2, as both versions are used in the encoder. Here follows a
- brief summary of the interpolation scheme; the details are found in
- the c-code of Appendix A. In the first sub-block, the average of the
- second LSF vector from the previous block and the first LSF vector in
- the current block is used. For sub-blocks two through five, the LSFs
- used are obtained by linear interpolation from lsf1 (and qlsf1) to
- lsf2 (and qlsf2), with lsf1 used in sub-block two and lsf2 in sub-
- block five. In the last sub-block, lsf2 is used. For the very first
- block it is assumed that the last LSF vector of the previous block is
- equal to a predefined vector, lsfmeanTbl, obtained by calculating the
- mean LSF vector of the LSF design database.
-
- lsfmeanTbl[LPC_FILTERORDER] = {0.281738, 0.445801, 0.663330,
- 0.962524, 1.251831, 1.533081, 1.850586, 2.137817,
- 2.481445, 2.777344}
-
-
-
-
-
-
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-
- The interpolation method is standard linear interpolation in the LSF
- domain. The interpolated LSF values are converted to LPC
- coefficients for each sub-block. The unquantized and quantized LPC
- coefficients form two sets of filters respectively. The unquantized
- analysis filter for sub-block k is defined as follows
-
- ___
- \
- Ak(z)= 1 + > ak(i)*z^(-i)
- /__
- i=1...LPC_FILTERORDER
-
- The quantized analysis filter for sub-block k is defined as follows
- ___
- \
- A~k(z)= 1 + > a~k(i)*z^(-i)
- /__
- i=1...LPC_FILTERORDER
-
- A reference implementation of the lsf encoding is given in Appendix
- A.38. A reference implementation of the corresponding decoding can
- be found in Appendix A.36.
-
-3.2.7. LPC Analysis and Quantization for 20 ms Frames
-
- As previously stated, the codec only calculates one set of LPC
- parameters for the 20 ms frame size as opposed to two sets for 30 ms
- frames. A single set of autocorrelation coefficients is calculated
- on the LPC_LOOKBACK + BLOCKL = 80 + 160 = 240 samples. These samples
- are windowed with the asymmetric window lpc_asymwinTbl, centered over
- the third sub-frame, to form speech_hp_win. Autocorrelation
- coefficients, acf, are calculated on the 240 samples in speech_hp_win
- and then windowed exactly as in section 3.2.1 (resulting in
- acf_win).
-
- This single set of windowed autocorrelation coefficients is used to
- calculate LPC coefficients, LSF coefficients, and quantized LSF
- coefficients in exactly the same manner as in sections 3.2.3 through
- 3.2.4. As for the 30 ms frame size, the ten LSF coefficients are
- divided into three sub-vectors of size 3, 3, and 4 and quantized by
- using the same scheme and codebook as in section 3.2.4 to finally get
- 3 quantization indices. The quantized LSF coefficients are
- stabilized with the algorithm described in section 3.2.5.
-
- From the set of LSF coefficients computed for this block and those
- from the previous block, different LSFs are obtained for each sub-
- block by means of interpolation. The interpolation is done linearly
- in the LSF domain over the four sub-blocks, so that the n-th sub-
-
-
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-
- frame uses the weight (4-n)/4 for the LSF from old frame and the
- weight n/4 of the LSF from the current frame. For the very first
- block the mean LSF, lsfmeanTbl, is used as the LSF from the previous
- block. Similarly as seen in section 3.2.6, both unquantized, A(z),
- and quantized, A~(z), analysis filters are calculated for each of the
- four sub-blocks.
-
-3.3. Calculation of the Residual
-
- The block of speech samples is filtered by the quantized and
- interpolated LPC analysis filters to yield the residual signal. In
- particular, the corresponding LPC analysis filter for each 40 sample
- sub-block is used to filter the speech samples for the same sub-
- block. The filter memory at the end of each sub-block is carried
- over to the LPC filter of the next sub-block. The signal at the
- output of each LP analysis filter constitutes the residual signal for
- the corresponding sub-block.
-
- A reference implementation of the LPC analysis filters is given in
- Appendix A.10.
-
-3.4. Perceptual Weighting Filter
-
- In principle any good design of a perceptual weighting filter can be
- applied in the encoder without compromising this codec definition.
- However, it is RECOMMENDED to use the perceptual weighting filter Wk
- for sub-block k specified below:
-
- Wk(z)=1/Ak(z/LPC_CHIRP_WEIGHTDENUM), where
- LPC_CHIRP_WEIGHTDENUM = 0.4222
-
- This is a simple design with low complexity that is applied in the
- LPC residual domain. Here Ak(z) is the filter obtained for sub-block
- k from unquantized but interpolated LSF coefficients.
-
-3.5. Start State Encoder
-
- The start state is quantized by using a common 6-bit scalar quantizer
- for the block and a 3-bit scalar quantizer operating on scaled
- samples in the weighted speech domain. In the following we describe
- the state encoding in greater detail.
-
-
-
-
-
-
-
-
-
-
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-
-3.5.1. Start State Estimation
-
- The two sub-blocks containing the start state are determined by
- finding the two consecutive sub-blocks in the block having the
- highest power. Advantageously, down-weighting is used in the
- beginning and end of the sub-frames, i.e., the following measure is
- computed (NSUB=4/6 for 20/30 ms frame size):
-
- nsub=1,...,NSUB-1
- ssqn[nsub] = 0.0;
- for (i=(nsub-1)*SUBL; i<(nsub-1)*SUBL+5; i++)
- ssqn[nsub] += sampEn_win[i-(nsub-1)*SUBL]*
- residual[i]*residual[i];
- for (i=(nsub-1)*SUBL+5; i<(nsub+1)*SUBL-5; i++)
- ssqn[nsub] += residual[i]*residual[i];
- for (i=(nsub+1)*SUBL-5; i<(nsub+1)*SUBL; i++)
- ssqn[nsub] += sampEn_win[(nsub+1)*SUBL-i-1]*
- residual[i]*residual[i];
-
- where sampEn_win[5]={1/6, 2/6, 3/6, 4/6, 5/6}; MAY be used. The
- sub-frame number corresponding to the maximum value of
- ssqEn_win[nsub-1]*ssqn[nsub] is selected as the start state
- indicator. A weighting of ssqEn_win[]={0.8,0.9,1.0,0.9,0.8} for 30
- ms frames and ssqEn_win[]={0.9,1.0,0.9} for 20 ms frames; MAY
- advantageously be used to bias the start state towards the middle of
- the frame.
-
- For 20 ms frames there are three possible positions for the two-sub-
- block length maximum power segment; the start state position is
- encoded with 2 bits. The start state position, start, MUST be
- encoded as
-
- start=1: start state in sub-frame 0 and 1
- start=2: start state in sub-frame 1 and 2
- start=3: start state in sub-frame 2 and 3
-
- For 30 ms frames there are five possible positions of the two-sub-
- block length maximum power segment, the start state position is
- encoded with 3 bits. The start state position, start, MUST be
- encoded as
-
- start=1: start state in sub-frame 0 and 1
- start=2: start state in sub-frame 1 and 2
- start=3: start state in sub-frame 2 and 3
- start=4: start state in sub-frame 3 and 4
- start=5: start state in sub-frame 4 and 5
-
-
-
-
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-
- Hence, in both cases, index 0 is not used. In order to shorten the
- start state for bit rate efficiency, the start state is brought down
- to STATE_SHORT_LEN=57 samples for 20 ms frames and STATE_SHORT_LEN=58
- samples for 30 ms frames. The power of the first 23/22 and last
- 23/22 samples of the two sub-frame blocks identified above is
- computed as the sum of the squared signal sample values, and the
- 23/22-sample segment with the lowest power is excluded from the start
- state. One bit is transmitted to indicate which of the two possible
- 57/58 sample segments is used. The start state position within the
- two sub-frames determined above, state_first, MUST be encoded as
-
- state_first=1: start state is first STATE_SHORT_LEN samples
- state_first=0: start state is last STATE_SHORT_LEN samples
-
-3.5.2. All-Pass Filtering and Scale Quantization
-
- The block of residual samples in the start state is first filtered by
- an all-pass filter with the quantized LPC coefficients as denominator
- and reversed quantized LPC coefficients as numerator. The purpose of
- this phase-dispersion filter is to get a more even distribution of
- the sample values in the residual signal. The filtering is performed
- by circular convolution, where the initial filter memory is set to
- zero.
-
- res(0..(STATE_SHORT_LEN-1)) = uncoded start state residual
- res((STATE_SHORT_LEN)..(2*STATE_SHORT_LEN-1)) = 0
-
- Pk(z) = A~rk(z)/A~k(z), where
- ___
- \
- A~rk(z)= z^(-LPC_FILTERORDER)+>a~k(i+1)*z^(i-(LPC_FILTERORDER-1))
- /__
- i=0...(LPC_FILTERORDER-1)
-
- and A~k(z) is taken from the block where the start state begins
-
- res -> Pk(z) -> filtered
-
- ccres(k) = filtered(k) + filtered(k+STATE_SHORT_LEN),
- k=0..(STATE_SHORT_LEN-1)
-
- The all-pass filtered block is searched for its largest magnitude
- sample. The 10-logarithm of this magnitude is quantized with a 6-bit
- quantizer, state_frgqTbl, by finding the nearest representation.
-
-
-
-
-
-
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-
- This results in an index, idxForMax, corresponding to a quantized
- value, qmax. The all-pass filtered residual samples in the block are
- then multiplied with a scaling factor scal=4.5/(10^qmax) to yield
- normalized samples.
-
- state_frgqTbl[64] = {1.000085, 1.071695, 1.140395, 1.206868,
- 1.277188, 1.351503, 1.429380, 1.500727, 1.569049,
- 1.639599, 1.707071, 1.781531, 1.840799, 1.901550,
- 1.956695, 2.006750, 2.055474, 2.102787, 2.142819,
- 2.183592, 2.217962, 2.257177, 2.295739, 2.332967,
- 2.369248, 2.402792, 2.435080, 2.468598, 2.503394,
- 2.539284, 2.572944, 2.605036, 2.636331, 2.668939,
- 2.698780, 2.729101, 2.759786, 2.789834, 2.818679,
- 2.848074, 2.877470, 2.906899, 2.936655, 2.967804,
- 3.000115, 3.033367, 3.066355, 3.104231, 3.141499,
- 3.183012, 3.222952, 3.265433, 3.308441, 3.350823,
- 3.395275, 3.442793, 3.490801, 3.542514, 3.604064,
- 3.666050, 3.740994, 3.830749, 3.938770, 4.101764}
-
-3.5.3. Scalar Quantization
-
- The normalized samples are quantized in the perceptually weighted
- speech domain by a sample-by-sample scalar DPCM quantization as
- depicted in Figure 3.3. Each sample in the block is filtered by a
- weighting filter Wk(z), specified in section 3.4, to form a weighted
- speech sample x[n]. The target sample d[n] is formed by subtracting
- a predicted sample y[n], where the prediction filter is given by
-
- Pk(z) = 1 - 1 / Wk(z).
-
- +-------+ x[n] + d[n] +-----------+ u[n]
- residual -->| Wk(z) |-------->(+)---->| Quantizer |------> quantized
- +-------+ - /|\ +-----------+ | residual
- | \|/
- y[n] +--------------------->(+)
- | |
- | +------+ |
- +--------| Pk(z)|<------+
- +------+
-
- Figure 3.3. Quantization of start state samples by DPCM in weighted
- speech domain.
-
- The coded state sample u[n] is obtained by quantizing d[n] with a 3-
- bit quantizer with quantization table state_sq3Tbl.
-
- state_sq3Tbl[8] = {-3.719849, -2.177490, -1.130005, -0.309692,
- 0.444214, 1.329712, 2.436279, 3.983887}
-
-
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-
- The quantized samples are transformed back to the residual domain by
- 1) scaling with 1/scal; 2) time-reversing the scaled samples; 3)
- filtering the time-reversed samples by the same all-pass filter, as
- in section 3.5.2, by using circular convolution; and 4) time-
- reversing the filtered samples. (More detail is in section 4.2.)
-
- A reference implementation of the start-state encoding can be found
- in Appendix A.46.
-
-3.6. Encoding the Remaining Samples
-
- A dynamic codebook is used to encode 1) the 23/22 remaining samples
- in the two sub-blocks containing the start state; 2) the sub-blocks
- after the start state in time; and 3) the sub-blocks before the start
- state in time. Thus, the encoding target can be either the 23/22
- samples remaining of the 2 sub-blocks containing the start state, or
- a 40-sample sub-block. This target can consist of samples that are
- indexed forward in time or backward in time, depending on the
- location of the start state. The length of the target is denoted by
- lTarget.
-
- The coding is based on an adaptive codebook that is built from a
- codebook memory that contains decoded LPC excitation samples from the
- already encoded part of the block. These samples are indexed in the
- same time direction as is the target vector and end at the sample
- instant prior to the first sample instant represented in the target
- vector. The codebook memory has length lMem, which is equal to
- CB_MEML=147 for the two/four 40-sample sub-blocks and 85 for the
- 23/22-sample sub-block.
-
- The following figure shows an overview of the encoding procedure.
-
- +------------+ +---------------+ +-------------+
- -> | 1. Decode | -> | 2. Mem setup | -> | 3. Perc. W. | ->
- +------------+ +---------------+ +-------------+
-
- +------------+ +-----------------+
- -> | 4. Search | -> | 5. Upd. Target | ------------------>
- | +------------+ +------------------ |
- ----<-------------<-----------<----------
- stage=0..2
-
- +----------------+
- -> | 6. Recalc G[0] | ---------------> gains and CB indices
- +----------------+
-
- Figure 3.4. Flow chart of the codebook search in the iLBC encoder.
-
-
-
-
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-
- 1. Decode the part of the residual that has been encoded so far,
- using the codebook without perceptual weighting.
-
- 2. Set up the memory by taking data from the decoded residual. This
- memory is used to construct codebooks. For blocks preceding the
- start state, both the decoded residual and the target are time
- reversed (section 3.6.1).
- 3. Filter the memory + target with the perceptual weighting filter
- (section 3.6.2).
-
- 4. Search for the best match between the target and the codebook
- vector. Compute the optimal gain for this match and quantize that
- gain (section 3.6.4).
-
- 5. Update the perceptually weighted target by subtracting the
- contribution from the selected codebook vector from the
- perceptually weighted memory (quantized gain times selected
- vector). Repeat 4 and 5 for the two additional stages.
-
- 6. Calculate the energy loss due to encoding of the residual. If
- needed, compensate for this loss by an upscaling and
- requantization of the gain for the first stage (section 3.7).
-
- The following sections provide an in-depth description of the
- different blocks of Figure 3.4.
-
-3.6.1. Codebook Memory
-
- The codebook memory is based on the already encoded sub-blocks, so
- the available data for encoding increases for each new sub-block that
- has been encoded. Until enough sub-blocks have been encoded to fill
- the codebook memory with data, it is padded with zeros. The
- following figure shows an example of the order in which the sub-
- blocks are encoded for the 30 ms frame size if the start state is
- located in the last 58 samples of sub-block 2 and 3.
-
- +-----------------------------------------------------+
- | 5 | 1 |///|////////| 2 | 3 | 4 |
- +-----------------------------------------------------+
-
- Figure 3.5. The order from 1 to 5 in which the sub-blocks are
- encoded. The slashed area is the start state.
-
-
-
-
-
-
-
-
-
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-
- The first target sub-block to be encoded is number 1, and the
- corresponding codebook memory is shown in the following figure. As
- the target vector comes before the start state in time, the codebook
- memory and target vector are time reversed; thus, after the block has
- been time reversed the search algorithm can be reused. As only the
- start state has been encoded so far, the last samples of the codebook
- memory are padded with zeros.
-
- +-------------------------
- |zeros|\\\\\\\\|\\\\| 1 |
- +-------------------------
-
- Figure 3.6. The codebook memory, length lMem=85 samples, and the
- target vector 1, length 22 samples.
-
- The next step is to encode sub-block 2 by using the memory that now
- has increased since sub-block 1 has been encoded. The following
- figure shows the codebook memory for encoding of sub-block 2.
-
- +-----------------------------------
- | zeros | 1 |///|////////| 2 |
- +-----------------------------------
-
- Figure 3.7. The codebook memory, length lMem=147 samples, and the
- target vector 2, length 40 samples.
-
- The next step is to encode sub-block 3 by using the memory which has
- been increased yet again since sub-blocks 1 and 2 have been encoded,
- but the sub-block still has to be padded with a few zeros. The
- following figure shows the codebook memory for encoding of sub-block
- 3.
-
- +------------------------------------------
- |zeros| 1 |///|////////| 2 | 3 |
- +------------------------------------------
-
- Figure 3.8. The codebook memory, length lMem=147 samples, and the
- target vector 3, length 40 samples.
-
- The next step is to encode sub-block 4 by using the memory which now
- has increased yet again since sub-blocks 1, 2, and 3 have been
- encoded. This time, the memory does not have to be padded with
- zeros. The following figure shows the codebook memory for encoding
- of sub-block 4.
-
-
-
-
-
-
-
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-
- +------------------------------------------
- |1|///|////////| 2 | 3 | 4 |
- +------------------------------------------
-
- Figure 3.9. The codebook memory, length lMem=147 samples, and the
- target vector 4, length 40 samples.
-
- The final target sub-block to be encoded is number 5, and the
- following figure shows the corresponding codebook memory. As the
- target vector comes before the start state in time, the codebook
- memory and target vector are time reversed.
-
- +-------------------------------------------
- | 3 | 2 |\\\\\\\\|\\\\| 1 | 5 |
- +-------------------------------------------
-
- Figure 3.10. The codebook memory, length lMem=147 samples, and the
- target vector 5, length 40 samples.
-
- For the case of 20 ms frames, the encoding procedure looks almost
- exactly the same. The only difference is that the size of the start
- state is 57 samples and that there are only three sub-blocks to be
- encoded. The encoding order is the same as above, starting with the
- 23-sample target and then encoding the two remaining 40-sample sub-
- blocks, first going forward in time and then going backward in time
- relative to the start state.
-
-3.6.2. Perceptual Weighting of Codebook Memory and Target
-
- To provide a perceptual weighting of the coding error, a
- concatenation of the codebook memory and the target to be coded is
- all-pole filtered with the perceptual weighting filter specified in
- section 3.4. The filter state of the weighting filter is set to
- zero.
-
- in(0..(lMem-1)) = unweighted codebook memory
- in(lMem..(lMem+lTarget-1)) = unweighted target signal
-
-
- in -> Wk(z) -> filtered,
- where Wk(z) is taken from the sub-block of the target
-
- weighted codebook memory = filtered(0..(lMem-1))
- weighted target signal = filtered(lMem..(lMem+lTarget-1))
-
- The codebook search is done with the weighted codebook memory and the
- weighted target, whereas the decoding and the codebook memory update
- uses the unweighted codebook memory.
-
-
-
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-
-3.6.3. Codebook Creation
-
- The codebook for the search is created from the perceptually weighted
- codebook memory. It consists of two sections, where the first is
- referred to as the base codebook and the second as the expanded
- codebook, as it is created by linear combinations of the first. Each
- of these two sections also has a subsection referred to as the
- augmented codebook. The augmented codebook is only created and used
- for the coding of the 40-sample sub-blocks and not for the 23/22-
- sample sub-block case. The codebook size used for the different
- sub-blocks and different stages are summarized in the table below.
-
- Stage
- 1 2 & 3
- --------------------------------------------
- 22 128 (64+0)*2 128 (64+0)*2
- Sub- 1:st 40 256 (108+20)*2 128 (44+20)*2
- Blocks 2:nd 40 256 (108+20)*2 256 (108+20)*2
- 3:rd 40 256 (108+20)*2 256 (108+20)*2
- 4:th 40 256 (108+20)*2 256 (108+20)*2
-
- Table 3.1. Codebook sizes for the 30 ms mode.
-
- Table 3.1 shows the codebook size for the different sub-blocks and
- stages for 30 ms frames. Inside the parentheses it shows how the
- number of codebook vectors is distributed, within the two sections,
- between the base/expanded codebook and the augmented base/expanded
- codebook. It should be interpreted in the following way:
- (base/expanded cb + augmented base/expanded cb). The total number of
- codebook vectors for a specific sub-block and stage is given by the
- following formula:
-
- Tot. cb vectors = base cb + aug. base cb + exp. cb + aug. exp. cb
-
- The corresponding values to Figure 3.1 for 20 ms frames are only
- slightly modified. The short sub-block is 23 instead of 22 samples,
- and the 3:rd and 4:th sub-frame are not present.
-
-3.6.3.1. Creation of a Base Codebook
-
- The base codebook is given by the perceptually weighted codebook
- memory that is mentioned in section 3.5.3. The different codebook
- vectors are given by sliding a window of length 23/22 or 40, given by
- variable lTarget, over the lMem-long perceptually weighted codebook
- memory. The indices are ordered so that the codebook vector
- containing sample (lMem-lTarget-n) to (lMem-n-1) of the codebook
-
-
-
-
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-
- memory vector has index n, where n=0..lMem-lTarget. Thus the total
- number of base codebook vectors is lMem-lTarget+1, and the indices
- are ordered from sample delay lTarget (23/22 or 40) to lMem+1 (86 or
- 148).
-
-3.6.3.2. Codebook Expansion
-
- The base codebook is expanded by a factor of 2, creating an
- additional section in the codebook. This new section is obtained by
- filtering the base codebook, base_cb, with a FIR filter with filter
- length CB_FILTERLEN=8. The construction of the expanded codebook
- compensates for the delay of four samples introduced by the FIR
- filter.
-
- cbfiltersTbl[CB_FILTERLEN]={-0.033691, 0.083740, -0.144043,
- 0.713379, 0.806152, -0.184326,
- 0.108887, -0.034180};
-
- ___
- \
- exp_cb(k)= + > cbfiltersTbl(i)*x(k-i+4)
- /__
- i=0...(LPC_FILTERORDER-1)
-
- where x(j) = base_cb(j) for j=0..lMem-1 and 0 otherwise
-
- The individual codebook vectors of the new filtered codebook, exp_cb,
- and their indices are obtained in the same fashion as described above
- for the base codebook.
-
-3.6.3.3. Codebook Augmentation
-
- For cases where encoding entire sub-blocks, i.e., cbveclen=40, the
- base and expanded codebooks are augmented to increase codebook
- richness. The codebooks are augmented by vectors produced by
- interpolation of segments. The base and expanded codebook,
- constructed above, consists of vectors corresponding to sample delays
- in the range from cbveclen to lMem. The codebook augmentation
- attempts to augment these codebooks with vectors corresponding to
- sample delays from 20 to 39. However, not all of these samples are
- present in the base codebook and expanded codebook, respectively.
- Therefore, the augmentation vectors are constructed as linear
- combinations between samples corresponding to sample delays in the
- range 20 to 39. The general idea of this procedure is presented in
- the following figures and text. The procedure is performed for both
- the base codebook and the expanded codebook.
-
-
-
-
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-
- - - ------------------------|
- codebook memory |
- - - ------------------------|
- |-5-|---15---|-5-|
- pi pp po
-
- | | Codebook vector
- |---15---|-5-|-----20-----| <- corresponding to
- i ii iii sample delay 20
-
- Figure 3.11. Generation of the first augmented codebook.
-
- Figure 3.11 shows the codebook memory with pointers pi, pp, and po,
- where pi points to sample 25, pp to sample 20, and po to sample 5.
- Below the codebook memory, the augmented codebook vector
- corresponding to sample delay 20 is drawn. Segment i consists of
- fifteen samples from pointer pp and forward in time. Segment ii
- consists of five interpolated samples from pi and forward and from po
- and forward. The samples are linearly interpolated with weights
- [0.0, 0.2, 0.4, 0.6, 0.8] for pi and weights [1.0, 0.8, 0.6, 0.4,
- 0.2] for po. Segment iii consists of twenty samples from pp and
- forward. The augmented codebook vector corresponding to sample delay
- 21 is produced by moving pointers pp and pi one sample backward in
- time. This gives us the following figure.
-
- - - ------------------------|
- codebook memory |
- - - ------------------------|
- |-5-|---16---|-5-|
- pi pp po
-
- | | Codebook vector
- |---16---|-5-|-----19-----| <- corresponding to
- i ii iii sample delay 21
-
- Figure 3.12. Generation of the second augmented codebook.
-
- Figure 3.12 shows the codebook memory with pointers pi, pp and po
- where pi points to sample 26, pp to sample 21, and po to sample 5.
- Below the codebook memory, the augmented codebook vector
- corresponding to sample delay 21 is drawn. Segment i now consists of
- sixteen samples from pp and forward. Segment ii consists of five
- interpolated samples from pi and forward and from po and forward, and
- the interpolation weights are the same throughout the procedure.
- Segment iii consists of nineteen samples from pp and forward. The
- same procedure of moving the two pointers is continued until the last
- augmented vector corresponding to sample delay 39 has been created.
- This gives a total of twenty new codebook vectors to each of the two
-
-
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-
- sections. Thus the total number of codebook vectors for each of the
- two sections, when including the augmented codebook, becomes lMem-
- SUBL+1+SUBL/2. This is provided that augmentation is evoked, i.e.,
- that lTarget=SUBL.
-
-3.6.4. Codebook Search
-
- The codebook search uses the codebooks described in the sections
- above to find the best match of the perceptually weighted target, see
- section 3.6.2. The search method is a multi-stage gain-shape
- matching performed as follows. At each stage the best shape vector
- is identified, then the gain is calculated and quantized, and finally
- the target is updated in preparation for the next codebook search
- stage. The number of stages is CB_NSTAGES=3.
-
- If the target is the 23/22-sample vector the codebooks are indexed so
- that the base codebook is followed by the expanded codebook. If the
- target is 40 samples the order is as follows: base codebook,
- augmented base codebook, expanded codebook, and augmented expanded
- codebook. The size of each codebook section and its corresponding
- augmented section is given by Table 3.1 in section 3.6.3.
-
- For example, when the second 40-sample sub-block is coded, indices 0
- - 107 correspond to the base codebook, 108 - 127 correspond to the
- augmented base codebook, 128 - 235 correspond to the expanded
- codebook, and indices 236 - 255 correspond to the augmented expanded
- codebook. The indices are divided in the same fashion for all stages
- in the example. Only in the case of coding the first 40-sample sub-
- block is there a difference between stages (see Table 3.1).
-
-3.6.4.1. Codebook Search at Each Stage
-
- The codebooks are searched to find the best match to the target at
- each stage. When the best match is found, the target is updated and
- the next-stage search is started. The three chosen codebook vectors
- and their corresponding gains constitute the encoded sub-block. The
- best match is decided by the following three criteria:
-
- 1. Compute the measure
-
- (target*cbvec)^2 / ||cbvec||^2
-
- for all codebook vectors, cbvec, and choose the codebook vector
- maximizing the measure. The expression (target*cbvec) is the dot
- product between the target vector to be coded and the codebook vector
- for which we compute the measure. The norm, ||x||, is defined as the
- square root of (x*x).
-
-
-
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-
- 2. The absolute value of the gain, corresponding to the chosen
- codebook vector, cbvec, must be smaller than a fixed limit,
- CB_MAXGAIN=1.3:
-
- |gain| < CB_MAXGAIN
-
- where the gain is computed in the following way:
-
- gain = (target*cbvec) / ||cbvec||^2
-
- 3. For the first stage, the dot product of the chosen codebook vector
- and target must be positive:
-
- target*cbvec > 0
-
- In practice the above criteria are used in a sequential search
- through all codebook vectors. The best match is found by registering
- a new max measure and index whenever the previously registered max
- measure is surpassed and all other criteria are fulfilled. If none
- of the codebook vectors fulfill (2) and (3), the first codebook
- vector is selected.
-
-3.6.4.2. Gain Quantization at Each Stage
-
- The gain follows as a result of the computation
-
- gain = (target*cbvec) / ||cbvec||^2
-
- for the optimal codebook vector found by the procedure in section
- 3.6.4.1.
-
- The three stages quantize the gain, using 5, 4, and 3 bits,
- respectively. In the first stage, the gain is limited to positive
- values. This gain is quantized by finding the nearest value in the
- quantization table gain_sq5Tbl.
-
- gain_sq5Tbl[32]={0.037476, 0.075012, 0.112488, 0.150024, 0.187500,
- 0.224976, 0.262512, 0.299988, 0.337524, 0.375000,
- 0.412476, 0.450012, 0.487488, 0.525024, 0.562500,
- 0.599976, 0.637512, 0.674988, 0.712524, 0.750000,
- 0.787476, 0.825012, 0.862488, 0.900024, 0.937500,
- 0.974976, 1.012512, 1.049988, 1.087524, 1.125000,
- 1.162476, 1.200012}
-
- The gains of the subsequent two stages can be either positive or
- negative. The gains are quantized by using a quantization table
- times a scale factor. The second stage uses the table gain_sq4Tbl,
- and the third stage uses gain_sq3Tbl. The scale factor equates 0.1
-
-
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-
- or the absolute value of the quantized gain representation value
- obtained in the previous stage, whichever is larger. Again, the
- resulting gain index is the index to the nearest value of the
- quantization table times the scale factor.
-
- gainQ = scaleFact * gain_sqXTbl[index]
-
- gain_sq4Tbl[16]={-1.049988, -0.900024, -0.750000, -0.599976,
- -0.450012, -0.299988, -0.150024, 0.000000, 0.150024,
- 0.299988, 0.450012, 0.599976, 0.750000, 0.900024,
- 1.049988, 1.200012}
-
- gain_sq3Tbl[8]={-1.000000, -0.659973, -0.330017,0.000000,
- 0.250000, 0.500000, 0.750000, 1.00000}
-
-3.6.4.3. Preparation of Target for Next Stage
-
- Before performing the search for the next stage, the perceptually
- weighted target vector is updated by subtracting from it the selected
- codebook vector (from the perceptually weighted codebook) times the
- corresponding quantized gain.
-
- target[i] = target[i] - gainQ * selected_vec[i];
-
- A reference implementation of the codebook encoding is found in
- Appendix A.34.
-
-3.7. Gain Correction Encoding
-
- The start state is quantized in a relatively model independent manner
- using 3 bits per sample. In contrast, the remaining parts of the
- block are encoded by using an adaptive codebook. This codebook will
- produce high matching accuracy whenever there is a high correlation
- between the target and the best codebook vector. For unvoiced speech
- segments and background noises, this is not necessarily so, which,
- due to the nature of the squared error criterion, results in a coded
- signal with less power than the target signal. As the coded start
- state has good power matching to the target, the result is a power
- fluctuation within the encoded frame. Perceptually, the main problem
- with this is that the time envelope of the signal energy becomes
- unsteady. To overcome this problem, the gains for the codebooks are
- re-scaled after the codebook encoding by searching for a new gain
- factor for the first stage codebook that provides better power
- matching.
-
- First, the energy for the target signal, tene, is computed along with
- the energy for the coded signal, cene, given by the addition of the
- three gain scaled codebook vectors. Because the gains of the second
-
-
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-
- and third stage scale with the gain of the first stage, when the
- first stage gain is changed from gain[0] to gain_sq5Tbl[i] the energy
- of the coded signal changes from cene to
-
- cene*(gain_sq5Tbl[i]*gain_sq5Tbl[i])/(gain[0]*gain[0])
-
- where gain[0] is the gain for the first stage found in the original
- codebook search. A refined search is performed by testing the gain
- indices i=0 to 31, and as long as the new codebook energy as given
- above is less than tene, the gain index for stage 1 is increased. A
- restriction is applied so that the new gain value for stage 1 cannot
- be more than two times higher than the original value found in the
- codebook search. Note that by using this method we do not change the
- shape of the encoded vector, only the gain or amplitude.
-
-3.8. Bitstream Definition
-
- The total number of bits used to describe one frame of 20 ms speech
- is 304, which fits in 38 bytes and results in a bit rate of 15.20
- kbit/s. For the case of a frame length of 30 ms speech, the total
- number of bits used is 400, which fits in 50 bytes and results in a
- bit rate of 13.33 kbit/s. In the bitstream definition, the bits are
- distributed into three classes according to their bit error or loss
- sensitivity. The most sensitive bits (class 1) are placed first in
- the bitstream for each frame. The less sensitive bits (class 2) are
- placed after the class 1 bits. The least sensitive bits (class 3)
- are placed at the end of the bitstream for each frame.
-
- In the 20/30 ms frame length cases for each class, the following hold
- true: The class 1 bits occupy a total of 6/8 bytes (48/64 bits), the
- class 2 bits occupy 8/12 bytes (64/96 bits), and the class 3 bits
- occupy 24/30 bytes (191/239 bits). This distribution of the bits
- enables the use of uneven level protection (ULP) as is exploited in
- the payload format definition for iLBC [1]. The detailed bit
- allocation is shown in the table below. When a quantization index is
- distributed between more classes, the more significant bits belong to
- the lowest class.
-
-
-
-
-
-
-
-
-
-
-
-
-
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-
- Bitstream structure:
-
- ------------------------------------------------------------------+
- Parameter | Bits Class <1,2,3> |
- | 20 ms frame | 30 ms frame |
- ----------------------------------+---------------+---------------+
- Split 1 | 6 <6,0,0> | 6 <6,0,0> |
- LSF 1 Split 2 | 7 <7,0,0> | 7 <7,0,0> |
- LSF Split 3 | 7 <7,0,0> | 7 <7,0,0> |
- ------------------+---------------+---------------+
- Split 1 | NA (Not Appl.)| 6 <6,0,0> |
- LSF 2 Split 2 | NA | 7 <7,0,0> |
- Split 3 | NA | 7 <7,0,0> |
- ------------------+---------------+---------------+
- Sum | 20 <20,0,0> | 40 <40,0,0> |
- ----------------------------------+---------------+---------------+
- Block Class | 2 <2,0,0> | 3 <3,0,0> |
- ----------------------------------+---------------+---------------+
- Position 22 sample segment | 1 <1,0,0> | 1 <1,0,0> |
- ----------------------------------+---------------+---------------+
- Scale Factor State Coder | 6 <6,0,0> | 6 <6,0,0> |
- ----------------------------------+---------------+---------------+
- Sample 0 | 3 <0,1,2> | 3 <0,1,2> |
- Quantized Sample 1 | 3 <0,1,2> | 3 <0,1,2> |
- Residual : | : : | : : |
- State : | : : | : : |
- Samples : | : : | : : |
- Sample 56 | 3 <0,1,2> | 3 <0,1,2> |
- Sample 57 | NA | 3 <0,1,2> |
- ------------------+---------------+---------------+
- Sum | 171 <0,57,114>| 174 <0,58,116>|
- ----------------------------------+---------------+---------------+
- Stage 1 | 7 <6,0,1> | 7 <4,2,1> |
- CB for 22/23 Stage 2 | 7 <0,0,7> | 7 <0,0,7> |
- sample block Stage 3 | 7 <0,0,7> | 7 <0,0,7> |
- ------------------+---------------+---------------+
- Sum | 21 <6,0,15> | 21 <4,2,15> |
- ----------------------------------+---------------+---------------+
- Stage 1 | 5 <2,0,3> | 5 <1,1,3> |
- Gain for 22/23 Stage 2 | 4 <1,1,2> | 4 <1,1,2> |
- sample block Stage 3 | 3 <0,0,3> | 3 <0,0,3> |
- ------------------+---------------+---------------+
- Sum | 12 <3,1,8> | 12 <2,2,8> |
- ----------------------------------+---------------+---------------+
- Stage 1 | 8 <7,0,1> | 8 <6,1,1> |
- sub-block 1 Stage 2 | 7 <0,0,7> | 7 <0,0,7> |
- Stage 3 | 7 <0,0,7> | 7 <0,0,7> |
- ------------------+---------------+---------------+
-
-
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-
- Stage 1 | 8 <0,0,8> | 8 <0,7,1> |
- sub-block 2 Stage 2 | 8 <0,0,8> | 8 <0,0,8> |
- Indices Stage 3 | 8 <0,0,8> | 8 <0,0,8> |
- for CB ------------------+---------------+---------------+
- sub-blocks Stage 1 | NA | 8 <0,7,1> |
- sub-block 3 Stage 2 | NA | 8 <0,0,8> |
- Stage 3 | NA | 8 <0,0,8> |
- ------------------+---------------+---------------+
- Stage 1 | NA | 8 <0,7,1> |
- sub-block 4 Stage 2 | NA | 8 <0,0,8> |
- Stage 3 | NA | 8 <0,0,8> |
- ------------------+---------------+---------------+
- Sum | 46 <7,0,39> | 94 <6,22,66> |
- ----------------------------------+---------------+---------------+
- Stage 1 | 5 <1,2,2> | 5 <1,2,2> |
- sub-block 1 Stage 2 | 4 <1,1,2> | 4 <1,2,1> |
- Stage 3 | 3 <0,0,3> | 3 <0,0,3> |
- ------------------+---------------+---------------+
- Stage 1 | 5 <1,1,3> | 5 <0,2,3> |
- sub-block 2 Stage 2 | 4 <0,2,2> | 4 <0,2,2> |
- Stage 3 | 3 <0,0,3> | 3 <0,0,3> |
- Gains for ------------------+---------------+---------------+
- sub-blocks Stage 1 | NA | 5 <0,1,4> |
- sub-block 3 Stage 2 | NA | 4 <0,1,3> |
- Stage 3 | NA | 3 <0,0,3> |
- ------------------+---------------+---------------+
- Stage 1 | NA | 5 <0,1,4> |
- sub-block 4 Stage 2 | NA | 4 <0,1,3> |
- Stage 3 | NA | 3 <0,0,3> |
- ------------------+---------------+---------------+
- Sum | 24 <3,6,15> | 48 <2,12,34> |
- ----------------------------------+---------------+---------------+
- Empty frame indicator | 1 <0,0,1> | 1 <0,0,1> |
- -------------------------------------------------------------------
- SUM 304 <48,64,192> 400 <64,96,240>
-
- Table 3.2. The bitstream definition for iLBC for both the 20 ms
- frame size mode and the 30 ms frame size mode.
-
- When packetized into the payload, the bits MUST be sorted as follows:
- All the class 1 bits in the order (from top to bottom) as specified
- in the table, all the class 2 bits (from top to bottom), and all the
- class 3 bits in the same sequential order. The last bit, the empty
- frame indicator, SHOULD be set to zero by the encoder. If this bit
- is set to 1 the decoder SHOULD treat the data as a lost frame. For
- example, this bit can be set to 1 to indicate lost frame for file
- storage format, as in [1].
-
-
-
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-
-4. Decoder Principles
-
- This section describes the principles of each component of the
- decoder algorithm.
-
- +-------------+ +--------+ +---------------+
- payload -> | 1. Get para | -> | 2. LPC | -> | 3. Sc Dequant | ->
- +-------------+ +--------+ +---------------+
-
- +-------------+ +------------------+
- -> | 4. Mem setup| -> | 5. Construct res |------->
- | +-------------+ +------------------- |
- ---------<-----------<-----------<------------
- Sub-frame 0...2/4 (20 ms/30 ms)
-
- +----------------+ +----------+
- -> | 6. Enhance res | -> | 7. Synth | ------------>
- +----------------+ +----------+
-
- +-----------------+
- -> | 8. Post Process | ----------------> decoded speech
- +-----------------+
-
- Figure 4.1. Flow chart of the iLBC decoder. If a frame was lost,
- steps 1 to 5 SHOULD be replaced by a PLC algorithm.
-
- 1. Extract the parameters from the bitstream.
-
- 2. Decode the LPC and interpolate (section 4.1).
-
- 3. Construct the 57/58-sample start state (section 4.2).
-
- 4. Set up the memory by using data from the decoded residual. This
- memory is used for codebook construction. For blocks preceding
- the start state, both the decoded residual and the target are time
- reversed. Sub-frames are decoded in the same order as they were
- encoded.
-
- 5. Construct the residuals of this sub-frame (gain[0]*cbvec[0] +
- gain[1]*cbvec[1] + gain[2]*cbvec[2]). Repeat 4 and 5 until the
- residual of all sub-blocks has been constructed.
-
- 6. Enhance the residual with the post filter (section 4.6).
-
- 7. Synthesis of the residual (section 4.7).
-
- 8. Post process with HP filter, if desired (section 4.8).
-
-
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-
-4.1. LPC Filter Reconstruction
-
- The decoding of the LP filter parameters is very straightforward.
- For a set of three/six indices, the corresponding LSF vector(s) are
- found by simple table lookup. For each of the LSF vectors, the three
- split vectors are concatenated to obtain qlsf1 and qlsf2,
- respectively (in the 20 ms mode only one LSF vector, qlsf, is
- constructed). The next step is the stability check described in
- section 3.2.5 followed by the interpolation scheme described in
- section 3.2.6 (3.2.7 for 20 ms frames). The only difference is that
- only the quantized LSFs are known at the decoder, and hence the
- unquantized LSFs are not processed.
-
- A reference implementation of the LPC filter reconstruction is given
- in Appendix A.36.
-
-4.2. Start State Reconstruction
-
- The scalar encoded STATE_SHORT_LEN=58 (STATE_SHORT_LEN=57 in the 20
- ms mode) state samples are reconstructed by 1) forming a set of
- samples (by table lookup) from the index stream idxVec[n], 2)
- multiplying the set with 1/scal=(10^qmax)/4.5, 3) time reversing the
- 57/58 samples, 4) filtering the time reversed block with the
- dispersion (all-pass) filter used in the encoder (as described in
- section 3.5.2); this compensates for the phase distortion of the
- earlier filter operation, and 5 reversing the 57/58 samples from the
- previous step.
-
- in(0..(STATE_SHORT_LEN-1)) = time reversed samples from table
- look-up,
- idxVecDec((STATE_SHORT_LEN-1)..0)
-
- in(STATE_SHORT_LEN..(2*STATE_SHORT_LEN-1)) = 0
-
- Pk(z) = A~rk(z)/A~k(z), where
- ___
- \
- A~rk(z)= z^(-LPC_FILTERORDER) + > a~ki*z^(i-(LPC_FILTERORDER-1))
- /__
- i=0...(LPC_FILTERORDER-1)
-
- and A~k(z) is taken from the block where the start state begins
-
- in -> Pk(z) -> filtered
-
- out(k) = filtered(STATE_SHORT_LEN-1-k) +
- filtered(2*STATE_SHORT_LEN-1-k),
- k=0..(STATE_SHORT_LEN-1)
-
-
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-
- The remaining 23/22 samples in the state are reconstructed by the
- same adaptive codebook technique described in section 4.3. The
- location bit determines whether these are the first or the last 23/22
- samples of the 80-sample state vector. If the remaining 23/22
- samples are the first samples, then the scalar encoded
- STATE_SHORT_LEN state samples are time-reversed before initialization
- of the adaptive codebook memory vector.
-
- A reference implementation of the start state reconstruction is given
- in Appendix A.44.
-
-4.3. Excitation Decoding Loop
-
- The decoding of the LPC excitation vector proceeds in the same order
- in which the residual was encoded at the encoder. That is, after the
- decoding of the entire 80-sample state vector, the forward sub-blocks
- (corresponding to samples occurring after the state vector samples)
- are decoded, and then the backward sub-blocks (corresponding to
- samples occurring before the state vector) are decoded, resulting in
- a fully decoded block of excitation signal samples.
-
- In particular, each sub-block is decoded by using the multistage
- adaptive codebook decoding module described in section 4.4. This
- module relies upon an adaptive codebook memory constructed before
- each run of the adaptive codebook decoding. The construction of the
- adaptive codebook memory in the decoder is identical to the method
- outlined in section 3.6.3, except that it is done on the codebook
- memory without perceptual weighting.
-
- For the initial forward sub-block, the last STATE_LEN=80 samples of
- the length CB_LMEM=147 adaptive codebook memory are filled with the
- samples of the state vector. For subsequent forward sub-blocks, the
- first SUBL=40 samples of the adaptive codebook memory are discarded,
- the remaining samples are shifted by SUBL samples toward the
- beginning of the vector, and the newly decoded SUBL=40 samples are
- placed at the end of the adaptive codebook memory. For backward
- sub-blocks, the construction is similar, except that every vector of
- samples involved is first time reversed.
-
- A reference implementation of the excitation decoding loop is found
- in Appendix A.5.
-
-
-
-
-
-
-
-
-
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-
-4.4. Multistage Adaptive Codebook Decoding
-
- The Multistage Adaptive Codebook Decoding module is used at both the
- sender (encoder) and the receiver (decoder) ends to produce a
- synthetic signal in the residual domain that is eventually used to
- produce synthetic speech. The module takes the index values used to
- construct vectors that are scaled and summed together to produce a
- synthetic signal that is the output of the module.
-
-4.4.1. Construction of the Decoded Excitation Signal
-
- The unpacked index values provided at the input to the module are
- references to extended codebooks, which are constructed as described
- in section 3.6.3, except that they are based on the codebook memory
- without the perceptual weighting. The unpacked three indices are
- used to look up three codebook vectors. The unpacked three gain
- indices are used to decode the corresponding 3 gains. In this
- decoding, the successive rescaling, as described in section 3.6.4.2,
- is applied.
-
- A reference implementation of the adaptive codebook decoding is
- listed in Appendix A.32.
-
-4.5. Packet Loss Concealment
-
- If packet loss occurs, the decoder receives a signal saying that
- information regarding a block is lost. For such blocks it is
- RECOMMENDED to use a Packet Loss Concealment (PLC) unit to create a
- decoded signal that masks the effect of that packet loss. In the
- following we will describe an example of a PLC unit that can be used
- with the iLBC codec. As the PLC unit is used only at the decoder,
- the PLC unit does not affect interoperability between
- implementations. Other PLC implementations MAY therefore be used.
-
- The PLC described operates on the LP filters and the excitation
- signals and is based on the following principles:
-
-4.5.1. Block Received Correctly and Previous Block Also Received
-
- If the block is received correctly, the PLC only records state
- information of the current block that can be used in case the next
- block is lost. The LP filter coefficients for each sub-block and the
- entire decoded excitation signal are all saved in the decoder state
- structure. All of this information will be needed if the following
- block is lost.
-
-
-
-
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-
-
-4.5.2. Block Not Received
-
- If the block is not received, the block substitution is based on a
- pitch-synchronous repetition of the excitation signal, which is
- filtered by the last LP filter of the previous block. The previous
- block's information is stored in the decoder state structure.
-
- A correlation analysis is performed on the previous block's
- excitation signal in order to detect the amount of pitch periodicity
- and a pitch value. The correlation measure is also used to decide on
- the voicing level (the degree to which the previous block's
- excitation was a voiced or roughly periodic signal). The excitation
- in the previous block is used to create an excitation for the block
- to be substituted, such that the pitch of the previous block is
- maintained. Therefore, the new excitation is constructed in a
- pitch-synchronous manner. In order to avoid a buzzy-sounding
- substituted block, a random excitation is mixed with the new pitch
- periodic excitation, and the relative use of the two components is
- computed from the correlation measure (voicing level).
-
- For the block to be substituted, the newly constructed excitation
- signal is then passed through the LP filter to produce the speech
- that will be substituted for the lost block.
-
- For several consecutive lost blocks, the packet loss concealment
- continues in a similar manner. The correlation measure of the last
- block received is still used along with the same pitch value. The LP
- filters of the last block received are also used again. The energy
- of the substituted excitation for consecutive lost blocks is
- decreased, leading to a dampened excitation, and therefore to
- dampened speech.
-
-4.5.3. Block Received Correctly When Previous Block Not Received
-
- For the case in which a block is received correctly when the previous
- block was not, the correctly received block's directly decoded speech
- (based solely on the received block) is not used as the actual
- output. The reason for this is that the directly decoded speech does
- not necessarily smoothly merge into the synthetic speech generated
- for the previous lost block. If the two signals are not smoothly
- merged, an audible discontinuity is accidentally produced.
- Therefore, a correlation analysis between the two blocks of
- excitation signal (the excitation of the previous concealed block and
- that of the current received block) is performed to find the best
- phase match. Then a simple overlap-add procedure is performed to
- merge the previous excitation smoothly into the current block's
- excitation.
-
-
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-
-
- The exact implementation of the packet loss concealment does not
- influence interoperability of the codec.
-
- A reference implementation of the packet loss concealment is
- suggested in Appendix A.14. Exact compliance with this suggested
- algorithm is not needed for a reference implementation to be fully
- compatible with the overall codec specification.
-
-4.6. Enhancement
-
- The decoder contains an enhancement unit that operates on the
- reconstructed excitation signal. The enhancement unit increases the
- perceptual quality of the reconstructed signal by reducing the
- speech-correlated noise in the voiced speech segments. Compared to
- traditional postfilters, the enhancer has an advantage in that it can
- only modify the excitation signal slightly. This means that there is
- no risk of over enhancement. The enhancer works very similarly for
- both the 20 ms frame size mode and the 30 ms frame size mode.
-
- For the mode with 20 ms frame size, the enhancer uses a memory of six
- 80-sample excitation blocks prior in time plus the two new 80-sample
- excitation blocks. For each block of 160 new unenhanced excitation
- samples, 160 enhanced excitation samples are produced. The enhanced
- excitation is 40-sample delayed compared to the unenhanced
- excitation, as the enhancer algorithm uses lookahead.
-
- For the mode with 30 ms frame size, the enhancer uses a memory of
- five 80-sample excitation blocks prior in time plus the three new
- 80-sample excitation blocks. For each block of 240 new unenhanced
- excitation samples, 240 enhanced excitation samples are produced.
- The enhanced excitation is 80-sample delayed compared to the
- unenhanced excitation, as the enhancer algorithm uses lookahead.
-
- Outline of Enhancer
-
- The speech enhancement unit operates on sub-blocks of 80 samples,
- which means that there are two/three 80 sample sub-blocks per frame.
- Each of these two/three sub-blocks is enhanced separately, but in an
- analogous manner.
-
-
-
-
-
-
-
-
-
-
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-
-
- unenhanced residual
- |
- | +---------------+ +--------------+
- +-> | 1. Pitch Est | -> | 2. Find PSSQ | -------->
- +---------------+ | +--------------+
- +-----<-------<------<--+
- +------------+ enh block 0..1/2 |
- -> | 3. Smooth | |
- +------------+ |
- \ |
- /\ |
- / \ Already |
- / 4. \----------->----------->-----------+ |
- \Crit/ Fulfilled | |
- \? / v |
- \/ | |
- \ +-----------------+ +---------+ | |
- Not +->| 5. Use Constr. | -> | 6. Mix | ----->
- Fulfilled +-----------------+ +---------+
-
- ---------------> enhanced residual
-
- Figure 4.2. Flow chart of the enhancer.
-
- 1. Pitch estimation of each of the two/three new 80-sample blocks.
-
- 2. Find the pitch-period-synchronous sequence n (for block k) by a
- search around the estimated pitch value. Do this for n=1,2,3,
- -1,-2,-3.
-
- 3. Calculate the smoothed residual generated by the six pitch-
- period-synchronous sequences from prior step.
-
- 4. Check if the smoothed residual satisfies the criterion (section
- 4.6.4).
-
- 5. Use constraint to calculate mixing factor (section 4.6.5).
-
- 6. Mix smoothed signal with unenhanced residual (pssq(n) n=0).
-
- The main idea of the enhancer is to find three 80 sample blocks
- before and three 80-sample blocks after the analyzed unenhanced sub-
- block and to use these to improve the quality of the excitation in
- that sub-block. The six blocks are chosen so that they have the
- highest possible correlation with the unenhanced sub-block that is
- being enhanced. In other words, the six blocks are pitch-period-
- synchronous sequences to the unenhanced sub-block.
-
-
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-
- A linear combination of the six pitch-period-synchronous sequences is
- calculated that approximates the sub-block. If the squared error
- between the approximation and the unenhanced sub-block is small
- enough, the enhanced residual is set equal to this approximation.
- For the cases when the squared error criterion is not fulfilled, a
- linear combination of the approximation and the unenhanced residual
- forms the enhanced residual.
-
-4.6.1. Estimating the Pitch
-
- Pitch estimates are needed to determine the locations of the pitch-
- period-synchronous sequences in a complexity-efficient way. For each
- of the new two/three sub-blocks, a pitch estimate is calculated by
- finding the maximum correlation in the range from lag 20 to lag 120.
- These pitch estimates are used to narrow down the search for the best
- possible pitch-period-synchronous sequences.
-
-4.6.2. Determination of the Pitch-Synchronous Sequences
-
- Upon receiving the pitch estimates from the prior step, the enhancer
- analyzes and enhances one 80-sample sub-block at a time. The pitch-
- period-synchronous-sequences pssq(n) can be viewed as vectors of
- length 80 samples each shifted n*lag samples from the current sub-
- block. The six pitch-period-synchronous-sequences, pssq(-3) to
- pssq(-1) and pssq(1) to pssq(3), are found one at a time by the steps
- below:
-
- 1) Calculate the estimate of the position of the pssq(n). For
- pssq(n) in front of pssq(0) (n > 0), the location of the pssq(n)
- is estimated by moving one pitch estimate forward in time from the
- exact location of pssq(n-1). Similarly, pssq(n) behind pssq(0) (n
- < 0) is estimated by moving one pitch estimate backward in time
- from the exact location of pssq(n+1). If the estimated pssq(n)
- vector location is totally within the enhancer memory (Figure
- 4.3), steps 2, 3, and 4 are performed, otherwise the pssq(n) is
- set to zeros.
-
- 2) Compute the correlation between the unenhanced excitation and
- vectors around the estimated location interval of pssq(n). The
- correlation is calculated in the interval estimated location +/- 2
- samples. This results in five correlation values.
-
- 3) The five correlation values are upsampled by a factor of 4, by
- using four simple upsampling filters (MA filters with coefficients
- upsFilter1.. upsFilter4). Within these the maximum value is
- found, which specifies the best pitch-period with a resolution of
- a quarter of a sample.
-
-
-
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-
-
- upsFilter1[7]={0.000000 0.000000 0.000000 1.000000
- 0.000000 0.000000 0.000000}
- upsFilter2[7]={0.015625 -0.076904 0.288330 0.862061
- -0.106445 0.018799 -0.015625}
- upsFilter3[7]={0.023682 -0.124268 0.601563 0.601563
- -0.124268 0.023682 -0.023682}
- upsFilter4[7]={0.018799 -0.106445 0.862061 0.288330
- -0.076904 0.015625 -0.018799}
-
- 4) Generate the pssq(n) vector by upsampling of the excitation memory
- and extracting the sequence that corresponds to the lag delay that
- was calculated in prior step.
-
- With the steps above, all the pssq(n) can be found in an iterative
- manner, first moving backward in time from pssq(0) and then forward
- in time from pssq(0).
-
-
- 0 159 319 479 639
- +---------------------------------------------------------------+
- | -5 | -4 | -3 | -2 | -1 | 0 | 1 | 2 |
- +---------------------------------------------------------------+
- |pssq 0 |
- |pssq -1| |pssq 1 |
- |pssq -2| |pssq 2 |
- |pssq -3| |pssq 3 |
-
- Figure 4.3. Enhancement for 20 ms frame size.
-
- Figure 4.3 depicts pitch-period-synchronous sequences in the
- enhancement of the first 80 sample block in the 20 ms frame size
- mode. The unenhanced signal input is stored in the last two sub-
- blocks (1 - 2), and the six other sub-blocks contain unenhanced
- residual prior-in-time. We perform the enhancement algorithm on two
- blocks of 80 samples, where the first of the two blocks consists of
- the last 40 samples of sub-block 0 and the first 40 samples of sub-
- block 1. The second 80-sample block consists of the last 40 samples
- of sub-block 1 and the first 40 samples of sub-block 2.
-
-
-
-
-
-
-
-
-
-
-
-
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-
-
- 0 159 319 479 639
- +---------------------------------------------------------------+
- | -4 | -3 | -2 | -1 | 0 | 1 | 2 | 3 |
- +---------------------------------------------------------------+
- |pssq 0 |
- |pssq -1| |pssq 1 |
- |pssq -2| |pssq 2 |
- |pssq -3| |pssq 3 |
-
- Figure 4.4. Enhancement for 30 ms frame size.
-
- Figure 4.4 depicts pitch-period-synchronous sequences in the
- enhancement of the first 80-sample block in the 30 ms frame size
- mode. The unenhanced signal input is stored in the last three sub-
- blocks (1 - 3). The five other sub-blocks contain unenhanced
- residual prior-in-time. The enhancement algorithm is performed on
- the three 80 sample sub-blocks 0, 1, and 2.
-
-4.6.3. Calculation of the Smoothed Excitation
-
- A linear combination of the six pssq(n) (n!=0) form a smoothed
- approximation, z, of pssq(0). Most of the weight is put on the
- sequences that are close to pssq(0), as these are likely to be most
- similar to pssq(0). The smoothed vector is also rescaled so that the
- energy of z is the same as the energy of pssq(0).
-
- ___
- \
- y = > pssq(i) * pssq_weight(i)
- /__
- i=-3,-2,-1,1,2,3
-
- pssq_weight(i) = 0.5*(1-cos(2*pi*(i+4)/(2*3+2)))
-
- z = C * y, where C = ||pssq(0)||/||y||
-
-4.6.4. Enhancer Criterion
-
- The criterion of the enhancer is that the enhanced excitation is not
- allowed to differ much from the unenhanced excitation. This
- criterion is checked for each 80-sample sub-block.
-
- e < (b * ||pssq(0)||^2), where b=0.05 and (Constraint 1)
-
- e = (pssq(0)-z)*(pssq(0)-z), and "*" means the dot product
-
-
-
-
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-
-
-4.6.5. Enhancing the excitation
-
- From the criterion in the previous section, it is clear that the
- excitation is not allowed to change much. The purpose of this
- constraint is to prevent the creation of an enhanced signal
- significantly different from the original signal. This also means
- that the constraint limits the numerical size of the errors that the
- enhancement procedure can make. That is especially important in
- unvoiced segments and background noise segments for which increased
- periodicity could lead to lower perceived quality.
-
- When the constraint in the prior section is not met, the enhanced
- residual is instead calculated through a constrained optimization by
- using the Lagrange multiplier technique. The new constraint is that
-
- e = (b * ||pssq(0)||^2) (Constraint 2)
-
- We distinguish two solution regions for the optimization: 1) the
- region where the first constraint is fulfilled and 2) the region
- where the first constraint is not fulfilled and the second constraint
- must be used.
-
- In the first case, where the second constraint is not needed, the
- optimized re-estimated vector is simply z, the energy-scaled version
- of y.
-
- In the second case, where the second constraint is activated and
- becomes an equality constraint, we have
-
- z= A*y + B*pssq(0)
-
- where
-
- A = sqrt((b-b^2/4)*(w00*w00)/ (w11*w00 + w10*w10)) and
-
- w11 = pssq(0)*pssq(0)
- w00 = y*y
- w10 = y*pssq(0) (* symbolizes the dot product)
-
- and
-
- B = 1 - b/2 - A * w10/w00
-
- Appendix A.16 contains a listing of a reference implementation for
- the enhancement method.
-
-
-
-
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-
-
-4.7. Synthesis Filtering
-
- Upon decoding or PLC of the LP excitation block, the decoded speech
- block is obtained by running the decoded LP synthesis filter,
- 1/A~k(z), over the block. The synthesis filters have to be shifted
- to compensate for the delay in the enhancer. For 20 ms frame size
- mode, they SHOULD be shifted one 40-sample sub-block, and for 30 ms
- frame size mode, they SHOULD be shifted two 40-sample sub-blocks.
- The LP coefficients SHOULD be changed at the first sample of every
- sub-block while keeping the filter state. For PLC blocks, one
- solution is to apply the last LP coefficients of the last decoded
- speech block for all sub-blocks.
-
- The reference implementation for the synthesis filtering can be found
- in Appendix A.48.
-
-4.8. Post Filtering
-
- If desired, the decoded block can be filtered by a high-pass filter.
- This removes the low frequencies of the decoded signal. A reference
- implementation of this, with cutoff at 65 Hz, is shown in Appendix
- A.30.
-
-5. Security Considerations
-
- This algorithm for the coding of speech signals is not subject to any
- known security consideration; however, its RTP payload format [1] is
- subject to several considerations, which are addressed there.
- Confidentiality of the media streams is achieved by encryption;
- therefore external mechanisms, such as SRTP [5], MAY be used for that
- purpose.
-
-6. Evaluation of the iLBC Implementations
-
- It is possible and suggested to evaluate certain iLBC implementation
- by utilizing methodology and tools available at
- http://www.ilbcfreeware.org/evaluation.html
-
-7. References
-
-7.1. Normative References
-
- [1] Duric, A. and S. Andersen, "Real-time Transport Protocol (RTP)
- Payload Format for internet Low Bit Rate Codec (iLBC) Speech",
- RFC 3952, December 2004.
-
- [2] Bradner, S., "Key words for use in RFCs to Indicate Requirement
- Levels", BCP 14, RFC 2119, March 1997.
-
-
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-
-
- [3] PacketCable(TM) Audio/Video Codecs Specification, Cable
- Television Laboratories, Inc.
-
-7.2. Informative References
-
- [4] ITU-T Recommendation G.711, available online from the ITU
- bookstore at http://www.itu.int.
-
- [5] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K. Norman,
- "The Secure Real Time Transport Protocol (SRTP)", RFC 3711, March
- 2004.
-
-8. Acknowledgements
-
- This extensive work, besides listed authors, has the following
- authors, who could not have been listed among "official" authors (due
- to IESG restrictions in the number of authors who can be listed):
-
- Manohar N. Murthi (Department of Electrical and Computer
- Engineering, University of Miami), Fredrik Galschiodt, Julian
- Spittka, and Jan Skoglund (Global IP Sound).
-
- The authors are deeply indebted to the following people and thank
- them sincerely:
-
- Henry Sinnreich, Patrik Faltstrom, Alan Johnston, and Jean-
- Francois Mule for great support of the iLBC initiative and for
- valuable feedback and comments.
-
- Peter Vary, Frank Mertz, and Christoph Erdmann (RWTH Aachen);
- Vladimir Cuperman (Niftybox LLC); Thomas Eriksson (Chalmers Univ
- of Tech), and Gernot Kubin (TU Graz), for thorough review of the
- iLBC document and their valuable feedback and remarks.
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
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-
-
-APPENDIX A. Reference Implementation
-
- This appendix contains the complete c-code for a reference
- implementation of encoder and decoder for the specified codec.
-
- The c-code consists of the following files with highest-level
- functions:
-
- iLBC_test.c: main function for evaluation purpose
- iLBC_encode.h: encoder header
- iLBC_encode.c: encoder function
- iLBC_decode.h: decoder header
- iLBC_decode.c: decoder function
-
- The following files contain global defines and constants:
-
- iLBC_define.h: global defines
- constants.h: global constants header
- constants.c: global constants memory allocations
-
- The following files contain subroutines:
-
- anaFilter.h: lpc analysis filter header
- anaFilter.c: lpc analysis filter function
- createCB.h: codebook construction header
- createCB.c: codebook construction function
- doCPLC.h: packet loss concealment header
- doCPLC.c: packet loss concealment function
- enhancer.h: signal enhancement header
- enhancer.c: signal enhancement function
- filter.h: general filter header
- filter.c: general filter functions
- FrameClassify.h: start state classification header
- FrameClassify.c: start state classification function
- gainquant.h: gain quantization header
- gainquant.c: gain quantization function
- getCBvec.h: codebook vector construction header
- getCBvec.c: codebook vector construction function
- helpfun.h: general purpose header
- helpfun.c: general purpose functions
- hpInput.h: input high pass filter header
- hpInput.c: input high pass filter function
- hpOutput.h: output high pass filter header
- hpOutput.c: output high pass filter function
- iCBConstruct.h: excitation decoding header
- iCBConstruct.c: excitation decoding function
- iCBSearch.h: excitation encoding header
- iCBSearch.c: excitation encoding function
-
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-
- LPCdecode.h: lpc decoding header
- LPCdecode.c: lpc decoding function
- LPCencode.h: lpc encoding header
- LPCencode.c: lpc encoding function
- lsf.h: line spectral frequencies header
- lsf.c: line spectral frequencies functions
- packing.h: bitstream packetization header
- packing.c: bitstream packetization functions
- StateConstructW.h: state decoding header
- StateConstructW.c: state decoding functions
- StateSearchW.h: state encoding header
- StateSearchW.c: state encoding function
- syntFilter.h: lpc synthesis filter header
- syntFilter.c: lpc synthesis filter function
-
- The implementation is portable and should work on many different
- platforms. However, it is not difficult to optimize the
- implementation on particular platforms, an exercise left to the
- reader.
-
-A.1. iLBC_test.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- iLBC_test.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include <math.h>
- #include <stdlib.h>
- #include <stdio.h>
- #include <string.h>
- #include "iLBC_define.h"
- #include "iLBC_encode.h"
- #include "iLBC_decode.h"
-
- /* Runtime statistics */
- #include <time.h>
-
- #define ILBCNOOFWORDS_MAX (NO_OF_BYTES_30MS/2)
-
- /*----------------------------------------------------------------*
- * Encoder interface function
-
-
-
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-
-
- *---------------------------------------------------------------*/
-
- short encode( /* (o) Number of bytes encoded */
- iLBC_Enc_Inst_t *iLBCenc_inst,
- /* (i/o) Encoder instance */
- short *encoded_data, /* (o) The encoded bytes */
- short *data /* (i) The signal block to encode*/
- ){
- float block[BLOCKL_MAX];
- int k;
-
- /* convert signal to float */
-
- for (k=0; k<iLBCenc_inst->blockl; k++)
- block[k] = (float)data[k];
-
- /* do the actual encoding */
-
- iLBC_encode((unsigned char *)encoded_data, block, iLBCenc_inst);
-
-
- return (iLBCenc_inst->no_of_bytes);
- }
-
- /*----------------------------------------------------------------*
- * Decoder interface function
- *---------------------------------------------------------------*/
-
- short decode( /* (o) Number of decoded samples */
- iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
- short *decoded_data, /* (o) Decoded signal block*/
- short *encoded_data, /* (i) Encoded bytes */
- short mode /* (i) 0=PL, 1=Normal */
- ){
- int k;
- float decblock[BLOCKL_MAX], dtmp;
-
- /* check if mode is valid */
-
- if (mode<0 || mode>1) {
- printf("\nERROR - Wrong mode - 0, 1 allowed\n"); exit(3);}
-
- /* do actual decoding of block */
-
- iLBC_decode(decblock, (unsigned char *)encoded_data,
- iLBCdec_inst, mode);
-
- /* convert to short */
-
-
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-
-
- for (k=0; k<iLBCdec_inst->blockl; k++){
- dtmp=decblock[k];
-
- if (dtmp<MIN_SAMPLE)
- dtmp=MIN_SAMPLE;
- else if (dtmp>MAX_SAMPLE)
- dtmp=MAX_SAMPLE;
- decoded_data[k] = (short) dtmp;
- }
-
- return (iLBCdec_inst->blockl);
- }
-
- /*---------------------------------------------------------------*
- * Main program to test iLBC encoding and decoding
- *
- * Usage:
- * exefile_name.exe <infile> <bytefile> <outfile> <channel>
- *
- * <infile> : Input file, speech for encoder (16-bit pcm file)
- * <bytefile> : Bit stream output from the encoder
- * <outfile> : Output file, decoded speech (16-bit pcm file)
- * <channel> : Bit error file, optional (16-bit)
- * 1 - Packet received correctly
- * 0 - Packet Lost
- *
- *--------------------------------------------------------------*/
-
- int main(int argc, char* argv[])
- {
-
- /* Runtime statistics */
-
- float starttime;
- float runtime;
- float outtime;
-
- FILE *ifileid,*efileid,*ofileid, *cfileid;
- short data[BLOCKL_MAX];
- short encoded_data[ILBCNOOFWORDS_MAX], decoded_data[BLOCKL_MAX];
- int len;
- short pli, mode;
- int blockcount = 0;
- int packetlosscount = 0;
-
- /* Create structs */
- iLBC_Enc_Inst_t Enc_Inst;
- iLBC_Dec_Inst_t Dec_Inst;
-
-
-
-Andersen, et al. Experimental [Page 48]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- /* get arguments and open files */
-
- if ((argc!=5) && (argc!=6)) {
- fprintf(stderr,
- "\n*-----------------------------------------------*\n");
- fprintf(stderr,
- " %s <20,30> input encoded decoded (channel)\n\n",
- argv[0]);
- fprintf(stderr,
- " mode : Frame size for the encoding/decoding\n");
- fprintf(stderr,
- " 20 - 20 ms\n");
- fprintf(stderr,
- " 30 - 30 ms\n");
- fprintf(stderr,
- " input : Speech for encoder (16-bit pcm file)\n");
- fprintf(stderr,
- " encoded : Encoded bit stream\n");
- fprintf(stderr,
- " decoded : Decoded speech (16-bit pcm file)\n");
- fprintf(stderr,
- " channel : Packet loss pattern, optional (16-bit)\n");
- fprintf(stderr,
- " 1 - Packet received correctly\n");
- fprintf(stderr,
- " 0 - Packet Lost\n");
- fprintf(stderr,
- "*-----------------------------------------------*\n\n");
- exit(1);
- }
- mode=atoi(argv[1]);
- if (mode != 20 && mode != 30) {
- fprintf(stderr,"Wrong mode %s, must be 20, or 30\n",
- argv[1]);
- exit(2);
- }
- if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
- fprintf(stderr,"Cannot open input file %s\n", argv[2]);
- exit(2);}
- if ( (efileid=fopen(argv[3],"wb")) == NULL) {
- fprintf(stderr, "Cannot open encoded file %s\n",
- argv[3]); exit(1);}
- if ( (ofileid=fopen(argv[4],"wb")) == NULL) {
- fprintf(stderr, "Cannot open decoded file %s\n",
- argv[4]); exit(1);}
- if (argc==6) {
- if( (cfileid=fopen(argv[5],"rb")) == NULL) {
- fprintf(stderr, "Cannot open channel file %s\n",
-
-
-
-Andersen, et al. Experimental [Page 49]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- argv[5]);
- exit(1);
- }
- } else {
- cfileid=NULL;
- }
-
- /* print info */
-
- fprintf(stderr, "\n");
- fprintf(stderr,
- "*---------------------------------------------------*\n");
- fprintf(stderr,
- "* *\n");
- fprintf(stderr,
- "* iLBC test program *\n");
- fprintf(stderr,
- "* *\n");
- fprintf(stderr,
- "* *\n");
- fprintf(stderr,
- "*---------------------------------------------------*\n");
- fprintf(stderr,"\nMode : %2d ms\n", mode);
- fprintf(stderr,"Input file : %s\n", argv[2]);
- fprintf(stderr,"Encoded file : %s\n", argv[3]);
- fprintf(stderr,"Output file : %s\n", argv[4]);
- if (argc==6) {
- fprintf(stderr,"Channel file : %s\n", argv[5]);
- }
- fprintf(stderr,"\n");
-
- /* Initialization */
-
- initEncode(&Enc_Inst, mode);
- initDecode(&Dec_Inst, mode, 1);
-
- /* Runtime statistics */
-
- starttime=clock()/(float)CLOCKS_PER_SEC;
-
- /* loop over input blocks */
-
- while (fread(data,sizeof(short),Enc_Inst.blockl,ifileid)==
- Enc_Inst.blockl) {
-
- blockcount++;
-
- /* encoding */
-
-
-
-Andersen, et al. Experimental [Page 50]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- fprintf(stderr, "--- Encoding block %i --- ",blockcount);
- len=encode(&Enc_Inst, encoded_data, data);
- fprintf(stderr, "\r");
-
- /* write byte file */
-
- fwrite(encoded_data, sizeof(unsigned char), len, efileid);
-
- /* get channel data if provided */
- if (argc==6) {
- if (fread(&pli, sizeof(short), 1, cfileid)) {
- if ((pli!=0)&&(pli!=1)) {
- fprintf(stderr, "Error in channel file\n");
- exit(0);
- }
- if (pli==0) {
- /* Packet loss -> remove info from frame */
- memset(encoded_data, 0,
- sizeof(short)*ILBCNOOFWORDS_MAX);
- packetlosscount++;
- }
- } else {
- fprintf(stderr, "Error. Channel file too short\n");
- exit(0);
- }
- } else {
- pli=1;
- }
-
- /* decoding */
-
- fprintf(stderr, "--- Decoding block %i --- ",blockcount);
-
- len=decode(&Dec_Inst, decoded_data, encoded_data, pli);
- fprintf(stderr, "\r");
-
- /* write output file */
-
- fwrite(decoded_data,sizeof(short),len,ofileid);
- }
-
- /* Runtime statistics */
-
- runtime = (float)(clock()/(float)CLOCKS_PER_SEC-starttime);
- outtime = (float)((float)blockcount*(float)mode/1000.0);
- printf("\n\nLength of speech file: %.1f s\n", outtime);
- printf("Packet loss : %.1f%%\n",
- 100.0*(float)packetlosscount/(float)blockcount);
-
-
-
-Andersen, et al. Experimental [Page 51]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- printf("Time to run iLBC :");
- printf(" %.1f s (%.1f %% of realtime)\n\n", runtime,
- (100*runtime/outtime));
-
- /* close files */
-
- fclose(ifileid); fclose(efileid); fclose(ofileid);
- if (argc==6) {
- fclose(cfileid);
- }
- return(0);
- }
-
-A.2. iLBC_encode.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- iLBC_encode.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_ILBCENCODE_H
- #define __iLBC_ILBCENCODE_H
-
- #include "iLBC_define.h"
-
- short initEncode( /* (o) Number of bytes
- encoded */
- iLBC_Enc_Inst_t *iLBCenc_inst, /* (i/o) Encoder instance */
- int mode /* (i) frame size mode */
- );
-
- void iLBC_encode(
-
- unsigned char *bytes, /* (o) encoded data bits iLBC */
- float *block, /* (o) speech vector to
- encode */
- iLBC_Enc_Inst_t *iLBCenc_inst /* (i/o) the general encoder
- state */
- );
-
- #endif
-
-
-
-
-Andersen, et al. Experimental [Page 52]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
-A.3. iLBC_encode.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- iLBC_encode.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include <math.h>
- #include <stdlib.h>
- #include <string.h>
-
- #include "iLBC_define.h"
- #include "LPCencode.h"
- #include "FrameClassify.h"
- #include "StateSearchW.h"
- #include "StateConstructW.h"
- #include "helpfun.h"
- #include "constants.h"
- #include "packing.h"
- #include "iCBSearch.h"
- #include "iCBConstruct.h"
- #include "hpInput.h"
- #include "anaFilter.h"
- #include "syntFilter.h"
-
- /*----------------------------------------------------------------*
- * Initiation of encoder instance.
- *---------------------------------------------------------------*/
-
- short initEncode( /* (o) Number of bytes
- encoded */
- iLBC_Enc_Inst_t *iLBCenc_inst, /* (i/o) Encoder instance */
- int mode /* (i) frame size mode */
- ){
- iLBCenc_inst->mode = mode;
- if (mode==30) {
- iLBCenc_inst->blockl = BLOCKL_30MS;
- iLBCenc_inst->nsub = NSUB_30MS;
- iLBCenc_inst->nasub = NASUB_30MS;
- iLBCenc_inst->lpc_n = LPC_N_30MS;
- iLBCenc_inst->no_of_bytes = NO_OF_BYTES_30MS;
- iLBCenc_inst->no_of_words = NO_OF_WORDS_30MS;
-
-
-
-Andersen, et al. Experimental [Page 53]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- iLBCenc_inst->state_short_len=STATE_SHORT_LEN_30MS;
- /* ULP init */
- iLBCenc_inst->ULP_inst=&ULP_30msTbl;
- }
- else if (mode==20) {
- iLBCenc_inst->blockl = BLOCKL_20MS;
- iLBCenc_inst->nsub = NSUB_20MS;
- iLBCenc_inst->nasub = NASUB_20MS;
- iLBCenc_inst->lpc_n = LPC_N_20MS;
- iLBCenc_inst->no_of_bytes = NO_OF_BYTES_20MS;
- iLBCenc_inst->no_of_words = NO_OF_WORDS_20MS;
- iLBCenc_inst->state_short_len=STATE_SHORT_LEN_20MS;
- /* ULP init */
- iLBCenc_inst->ULP_inst=&ULP_20msTbl;
- }
- else {
- exit(2);
- }
-
- memset((*iLBCenc_inst).anaMem, 0,
- LPC_FILTERORDER*sizeof(float));
- memcpy((*iLBCenc_inst).lsfold, lsfmeanTbl,
- LPC_FILTERORDER*sizeof(float));
- memcpy((*iLBCenc_inst).lsfdeqold, lsfmeanTbl,
- LPC_FILTERORDER*sizeof(float));
- memset((*iLBCenc_inst).lpc_buffer, 0,
- (LPC_LOOKBACK+BLOCKL_MAX)*sizeof(float));
- memset((*iLBCenc_inst).hpimem, 0, 4*sizeof(float));
-
- return (iLBCenc_inst->no_of_bytes);
- }
-
- /*----------------------------------------------------------------*
- * main encoder function
- *---------------------------------------------------------------*/
-
- void iLBC_encode(
- unsigned char *bytes, /* (o) encoded data bits iLBC */
- float *block, /* (o) speech vector to
- encode */
- iLBC_Enc_Inst_t *iLBCenc_inst /* (i/o) the general encoder
- state */
- ){
-
- float data[BLOCKL_MAX];
- float residual[BLOCKL_MAX], reverseResidual[BLOCKL_MAX];
-
- int start, idxForMax, idxVec[STATE_LEN];
-
-
-
-Andersen, et al. Experimental [Page 54]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- float reverseDecresidual[BLOCKL_MAX], mem[CB_MEML];
- int n, k, meml_gotten, Nfor, Nback, i, pos;
- int gain_index[CB_NSTAGES*NASUB_MAX],
- extra_gain_index[CB_NSTAGES];
- int cb_index[CB_NSTAGES*NASUB_MAX],extra_cb_index[CB_NSTAGES];
- int lsf_i[LSF_NSPLIT*LPC_N_MAX];
- unsigned char *pbytes;
- int diff, start_pos, state_first;
- float en1, en2;
- int index, ulp, firstpart;
- int subcount, subframe;
- float weightState[LPC_FILTERORDER];
- float syntdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
- float weightdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
- float decresidual[BLOCKL_MAX];
-
- /* high pass filtering of input signal if such is not done
- prior to calling this function */
-
- hpInput(block, iLBCenc_inst->blockl,
- data, (*iLBCenc_inst).hpimem);
-
- /* otherwise simply copy */
-
- /*memcpy(data,block,iLBCenc_inst->blockl*sizeof(float));*/
-
- /* LPC of hp filtered input data */
-
- LPCencode(syntdenum, weightdenum, lsf_i, data, iLBCenc_inst);
-
-
- /* inverse filter to get residual */
-
- for (n=0; n<iLBCenc_inst->nsub; n++) {
- anaFilter(&data[n*SUBL], &syntdenum[n*(LPC_FILTERORDER+1)],
- SUBL, &residual[n*SUBL], iLBCenc_inst->anaMem);
- }
-
- /* find state location */
-
- start = FrameClassify(iLBCenc_inst, residual);
-
- /* check if state should be in first or last part of the
- two subframes */
-
- diff = STATE_LEN - iLBCenc_inst->state_short_len;
- en1 = 0;
- index = (start-1)*SUBL;
-
-
-
-Andersen, et al. Experimental [Page 55]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- for (i = 0; i < iLBCenc_inst->state_short_len; i++) {
- en1 += residual[index+i]*residual[index+i];
- }
- en2 = 0;
- index = (start-1)*SUBL+diff;
- for (i = 0; i < iLBCenc_inst->state_short_len; i++) {
- en2 += residual[index+i]*residual[index+i];
- }
-
-
- if (en1 > en2) {
- state_first = 1;
- start_pos = (start-1)*SUBL;
- } else {
- state_first = 0;
- start_pos = (start-1)*SUBL + diff;
- }
-
- /* scalar quantization of state */
-
- StateSearchW(iLBCenc_inst, &residual[start_pos],
- &syntdenum[(start-1)*(LPC_FILTERORDER+1)],
- &weightdenum[(start-1)*(LPC_FILTERORDER+1)], &idxForMax,
- idxVec, iLBCenc_inst->state_short_len, state_first);
-
- StateConstructW(idxForMax, idxVec,
- &syntdenum[(start-1)*(LPC_FILTERORDER+1)],
- &decresidual[start_pos], iLBCenc_inst->state_short_len);
-
- /* predictive quantization in state */
-
- if (state_first) { /* put adaptive part in the end */
-
- /* setup memory */
-
- memset(mem, 0,
- (CB_MEML-iLBCenc_inst->state_short_len)*sizeof(float));
- memcpy(mem+CB_MEML-iLBCenc_inst->state_short_len,
- decresidual+start_pos,
- iLBCenc_inst->state_short_len*sizeof(float));
- memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
-
- /* encode sub-frames */
-
- iCBSearch(iLBCenc_inst, extra_cb_index, extra_gain_index,
- &residual[start_pos+iLBCenc_inst->state_short_len],
- mem+CB_MEML-stMemLTbl,
- stMemLTbl, diff, CB_NSTAGES,
-
-
-
-Andersen, et al. Experimental [Page 56]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- &weightdenum[start*(LPC_FILTERORDER+1)],
- weightState, 0);
-
- /* construct decoded vector */
-
- iCBConstruct(
- &decresidual[start_pos+iLBCenc_inst->state_short_len],
- extra_cb_index, extra_gain_index,
- mem+CB_MEML-stMemLTbl,
- stMemLTbl, diff, CB_NSTAGES);
-
- }
- else { /* put adaptive part in the beginning */
-
- /* create reversed vectors for prediction */
-
- for (k=0; k<diff; k++) {
- reverseResidual[k] = residual[(start+1)*SUBL-1
- -(k+iLBCenc_inst->state_short_len)];
- }
-
- /* setup memory */
-
- meml_gotten = iLBCenc_inst->state_short_len;
- for (k=0; k<meml_gotten; k++) {
- mem[CB_MEML-1-k] = decresidual[start_pos + k];
- }
- memset(mem, 0, (CB_MEML-k)*sizeof(float));
- memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
-
- /* encode sub-frames */
-
- iCBSearch(iLBCenc_inst, extra_cb_index, extra_gain_index,
- reverseResidual, mem+CB_MEML-stMemLTbl, stMemLTbl,
- diff, CB_NSTAGES,
- &weightdenum[(start-1)*(LPC_FILTERORDER+1)],
- weightState, 0);
-
- /* construct decoded vector */
-
- iCBConstruct(reverseDecresidual, extra_cb_index,
- extra_gain_index, mem+CB_MEML-stMemLTbl, stMemLTbl,
- diff, CB_NSTAGES);
-
- /* get decoded residual from reversed vector */
-
- for (k=0; k<diff; k++) {
- decresidual[start_pos-1-k] = reverseDecresidual[k];
-
-
-
-Andersen, et al. Experimental [Page 57]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- }
- }
-
- /* counter for predicted sub-frames */
-
- subcount=0;
-
- /* forward prediction of sub-frames */
-
- Nfor = iLBCenc_inst->nsub-start-1;
-
-
- if ( Nfor > 0 ) {
-
- /* setup memory */
-
- memset(mem, 0, (CB_MEML-STATE_LEN)*sizeof(float));
- memcpy(mem+CB_MEML-STATE_LEN, decresidual+(start-1)*SUBL,
- STATE_LEN*sizeof(float));
- memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
-
- /* loop over sub-frames to encode */
-
- for (subframe=0; subframe<Nfor; subframe++) {
-
- /* encode sub-frame */
-
- iCBSearch(iLBCenc_inst, cb_index+subcount*CB_NSTAGES,
- gain_index+subcount*CB_NSTAGES,
- &residual[(start+1+subframe)*SUBL],
- mem+CB_MEML-memLfTbl[subcount],
- memLfTbl[subcount], SUBL, CB_NSTAGES,
- &weightdenum[(start+1+subframe)*
- (LPC_FILTERORDER+1)],
- weightState, subcount+1);
-
- /* construct decoded vector */
-
- iCBConstruct(&decresidual[(start+1+subframe)*SUBL],
- cb_index+subcount*CB_NSTAGES,
- gain_index+subcount*CB_NSTAGES,
- mem+CB_MEML-memLfTbl[subcount],
- memLfTbl[subcount], SUBL, CB_NSTAGES);
-
- /* update memory */
-
- memcpy(mem, mem+SUBL, (CB_MEML-SUBL)*sizeof(float));
- memcpy(mem+CB_MEML-SUBL,
-
-
-
-Andersen, et al. Experimental [Page 58]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- &decresidual[(start+1+subframe)*SUBL],
- SUBL*sizeof(float));
- memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
-
- subcount++;
- }
- }
-
-
- /* backward prediction of sub-frames */
-
- Nback = start-1;
-
-
- if ( Nback > 0 ) {
-
- /* create reverse order vectors */
-
- for (n=0; n<Nback; n++) {
- for (k=0; k<SUBL; k++) {
- reverseResidual[n*SUBL+k] =
- residual[(start-1)*SUBL-1-n*SUBL-k];
- reverseDecresidual[n*SUBL+k] =
- decresidual[(start-1)*SUBL-1-n*SUBL-k];
- }
- }
-
- /* setup memory */
-
- meml_gotten = SUBL*(iLBCenc_inst->nsub+1-start);
-
-
- if ( meml_gotten > CB_MEML ) {
- meml_gotten=CB_MEML;
- }
- for (k=0; k<meml_gotten; k++) {
- mem[CB_MEML-1-k] = decresidual[(start-1)*SUBL + k];
- }
- memset(mem, 0, (CB_MEML-k)*sizeof(float));
- memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
-
- /* loop over sub-frames to encode */
-
- for (subframe=0; subframe<Nback; subframe++) {
-
- /* encode sub-frame */
-
- iCBSearch(iLBCenc_inst, cb_index+subcount*CB_NSTAGES,
-
-
-
-Andersen, et al. Experimental [Page 59]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- gain_index+subcount*CB_NSTAGES,
- &reverseResidual[subframe*SUBL],
- mem+CB_MEML-memLfTbl[subcount],
- memLfTbl[subcount], SUBL, CB_NSTAGES,
- &weightdenum[(start-2-subframe)*
- (LPC_FILTERORDER+1)],
- weightState, subcount+1);
-
- /* construct decoded vector */
-
- iCBConstruct(&reverseDecresidual[subframe*SUBL],
- cb_index+subcount*CB_NSTAGES,
- gain_index+subcount*CB_NSTAGES,
- mem+CB_MEML-memLfTbl[subcount],
- memLfTbl[subcount], SUBL, CB_NSTAGES);
-
- /* update memory */
-
- memcpy(mem, mem+SUBL, (CB_MEML-SUBL)*sizeof(float));
- memcpy(mem+CB_MEML-SUBL,
- &reverseDecresidual[subframe*SUBL],
- SUBL*sizeof(float));
- memset(weightState, 0, LPC_FILTERORDER*sizeof(float));
-
- subcount++;
-
- }
-
- /* get decoded residual from reversed vector */
-
- for (i=0; i<SUBL*Nback; i++) {
- decresidual[SUBL*Nback - i - 1] =
- reverseDecresidual[i];
- }
- }
- /* end encoding part */
-
- /* adjust index */
- index_conv_enc(cb_index);
-
- /* pack bytes */
-
- pbytes=bytes;
- pos=0;
-
- /* loop over the 3 ULP classes */
-
- for (ulp=0; ulp<3; ulp++) {
-
-
-
-Andersen, et al. Experimental [Page 60]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
-
- /* LSF */
- for (k=0; k<LSF_NSPLIT*iLBCenc_inst->lpc_n; k++) {
- packsplit(&lsf_i[k], &firstpart, &lsf_i[k],
- iLBCenc_inst->ULP_inst->lsf_bits[k][ulp],
- iLBCenc_inst->ULP_inst->lsf_bits[k][ulp]+
- iLBCenc_inst->ULP_inst->lsf_bits[k][ulp+1]+
- iLBCenc_inst->ULP_inst->lsf_bits[k][ulp+2]);
- dopack( &pbytes, firstpart,
- iLBCenc_inst->ULP_inst->lsf_bits[k][ulp], &pos);
- }
-
- /* Start block info */
-
- packsplit(&start, &firstpart, &start,
- iLBCenc_inst->ULP_inst->start_bits[ulp],
- iLBCenc_inst->ULP_inst->start_bits[ulp]+
- iLBCenc_inst->ULP_inst->start_bits[ulp+1]+
- iLBCenc_inst->ULP_inst->start_bits[ulp+2]);
- dopack( &pbytes, firstpart,
- iLBCenc_inst->ULP_inst->start_bits[ulp], &pos);
-
- packsplit(&state_first, &firstpart, &state_first,
- iLBCenc_inst->ULP_inst->startfirst_bits[ulp],
- iLBCenc_inst->ULP_inst->startfirst_bits[ulp]+
- iLBCenc_inst->ULP_inst->startfirst_bits[ulp+1]+
- iLBCenc_inst->ULP_inst->startfirst_bits[ulp+2]);
- dopack( &pbytes, firstpart,
- iLBCenc_inst->ULP_inst->startfirst_bits[ulp], &pos);
-
- packsplit(&idxForMax, &firstpart, &idxForMax,
- iLBCenc_inst->ULP_inst->scale_bits[ulp],
- iLBCenc_inst->ULP_inst->scale_bits[ulp]+
- iLBCenc_inst->ULP_inst->scale_bits[ulp+1]+
- iLBCenc_inst->ULP_inst->scale_bits[ulp+2]);
- dopack( &pbytes, firstpart,
- iLBCenc_inst->ULP_inst->scale_bits[ulp], &pos);
-
- for (k=0; k<iLBCenc_inst->state_short_len; k++) {
- packsplit(idxVec+k, &firstpart, idxVec+k,
- iLBCenc_inst->ULP_inst->state_bits[ulp],
- iLBCenc_inst->ULP_inst->state_bits[ulp]+
- iLBCenc_inst->ULP_inst->state_bits[ulp+1]+
- iLBCenc_inst->ULP_inst->state_bits[ulp+2]);
- dopack( &pbytes, firstpart,
- iLBCenc_inst->ULP_inst->state_bits[ulp], &pos);
- }
-
-
-
-
-Andersen, et al. Experimental [Page 61]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- /* 23/22 (20ms/30ms) sample block */
-
- for (k=0;k<CB_NSTAGES;k++) {
- packsplit(extra_cb_index+k, &firstpart,
- extra_cb_index+k,
- iLBCenc_inst->ULP_inst->extra_cb_index[k][ulp],
- iLBCenc_inst->ULP_inst->extra_cb_index[k][ulp]+
- iLBCenc_inst->ULP_inst->extra_cb_index[k][ulp+1]+
- iLBCenc_inst->ULP_inst->extra_cb_index[k][ulp+2]);
- dopack( &pbytes, firstpart,
- iLBCenc_inst->ULP_inst->extra_cb_index[k][ulp],
- &pos);
- }
-
- for (k=0;k<CB_NSTAGES;k++) {
- packsplit(extra_gain_index+k, &firstpart,
- extra_gain_index+k,
- iLBCenc_inst->ULP_inst->extra_cb_gain[k][ulp],
- iLBCenc_inst->ULP_inst->extra_cb_gain[k][ulp]+
- iLBCenc_inst->ULP_inst->extra_cb_gain[k][ulp+1]+
- iLBCenc_inst->ULP_inst->extra_cb_gain[k][ulp+2]);
- dopack( &pbytes, firstpart,
- iLBCenc_inst->ULP_inst->extra_cb_gain[k][ulp],
- &pos);
- }
-
- /* The two/four (20ms/30ms) 40 sample sub-blocks */
-
- for (i=0; i<iLBCenc_inst->nasub; i++) {
- for (k=0; k<CB_NSTAGES; k++) {
- packsplit(cb_index+i*CB_NSTAGES+k, &firstpart,
- cb_index+i*CB_NSTAGES+k,
- iLBCenc_inst->ULP_inst->cb_index[i][k][ulp],
- iLBCenc_inst->ULP_inst->cb_index[i][k][ulp]+
- iLBCenc_inst->ULP_inst->cb_index[i][k][ulp+1]+
- iLBCenc_inst->ULP_inst->cb_index[i][k][ulp+2]);
- dopack( &pbytes, firstpart,
- iLBCenc_inst->ULP_inst->cb_index[i][k][ulp],
- &pos);
- }
- }
-
- for (i=0; i<iLBCenc_inst->nasub; i++) {
- for (k=0; k<CB_NSTAGES; k++) {
- packsplit(gain_index+i*CB_NSTAGES+k, &firstpart,
- gain_index+i*CB_NSTAGES+k,
- iLBCenc_inst->ULP_inst->cb_gain[i][k][ulp],
- iLBCenc_inst->ULP_inst->cb_gain[i][k][ulp]+
-
-
-
-Andersen, et al. Experimental [Page 62]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- iLBCenc_inst->ULP_inst->cb_gain[i][k][ulp+1]+
- iLBCenc_inst->ULP_inst->cb_gain[i][k][ulp+2]);
- dopack( &pbytes, firstpart,
- iLBCenc_inst->ULP_inst->cb_gain[i][k][ulp],
- &pos);
- }
- }
- }
-
- /* set the last bit to zero (otherwise the decoder
- will treat it as a lost frame) */
- dopack( &pbytes, 0, 1, &pos);
- }
-
-A.4. iLBC_decode.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- iLBC_decode.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_ILBCDECODE_H
- #define __iLBC_ILBCDECODE_H
-
- #include "iLBC_define.h"
-
- short initDecode( /* (o) Number of decoded
- samples */
- iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
- int mode, /* (i) frame size mode */
- int use_enhancer /* (i) 1 to use enhancer
- 0 to run without
- enhancer */
- );
-
- void iLBC_decode(
- float *decblock, /* (o) decoded signal block */
- unsigned char *bytes, /* (i) encoded signal bits */
- iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
- structure */
- int mode /* (i) 0: bad packet, PLC,
- 1: normal */
-
-
-
-Andersen, et al. Experimental [Page 63]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- );
-
- #endif
-
-A.5. iLBC_decode.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- iLBC_decode.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include <math.h>
- #include <stdlib.h>
-
- #include "iLBC_define.h"
- #include "StateConstructW.h"
- #include "LPCdecode.h"
- #include "iCBConstruct.h"
- #include "doCPLC.h"
- #include "helpfun.h"
- #include "constants.h"
- #include "packing.h"
- #include "string.h"
- #include "enhancer.h"
- #include "hpOutput.h"
- #include "syntFilter.h"
-
- /*----------------------------------------------------------------*
- * Initiation of decoder instance.
- *---------------------------------------------------------------*/
-
- short initDecode( /* (o) Number of decoded
- samples */
- iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
- int mode, /* (i) frame size mode */
- int use_enhancer /* (i) 1 to use enhancer
- 0 to run without
- enhancer */
- ){
- int i;
-
- iLBCdec_inst->mode = mode;
-
-
-
-Andersen, et al. Experimental [Page 64]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- if (mode==30) {
- iLBCdec_inst->blockl = BLOCKL_30MS;
- iLBCdec_inst->nsub = NSUB_30MS;
- iLBCdec_inst->nasub = NASUB_30MS;
- iLBCdec_inst->lpc_n = LPC_N_30MS;
- iLBCdec_inst->no_of_bytes = NO_OF_BYTES_30MS;
- iLBCdec_inst->no_of_words = NO_OF_WORDS_30MS;
- iLBCdec_inst->state_short_len=STATE_SHORT_LEN_30MS;
- /* ULP init */
- iLBCdec_inst->ULP_inst=&ULP_30msTbl;
- }
- else if (mode==20) {
- iLBCdec_inst->blockl = BLOCKL_20MS;
- iLBCdec_inst->nsub = NSUB_20MS;
- iLBCdec_inst->nasub = NASUB_20MS;
- iLBCdec_inst->lpc_n = LPC_N_20MS;
- iLBCdec_inst->no_of_bytes = NO_OF_BYTES_20MS;
- iLBCdec_inst->no_of_words = NO_OF_WORDS_20MS;
- iLBCdec_inst->state_short_len=STATE_SHORT_LEN_20MS;
- /* ULP init */
- iLBCdec_inst->ULP_inst=&ULP_20msTbl;
- }
- else {
- exit(2);
- }
-
- memset(iLBCdec_inst->syntMem, 0,
- LPC_FILTERORDER*sizeof(float));
- memcpy((*iLBCdec_inst).lsfdeqold, lsfmeanTbl,
- LPC_FILTERORDER*sizeof(float));
-
- memset(iLBCdec_inst->old_syntdenum, 0,
- ((LPC_FILTERORDER + 1)*NSUB_MAX)*sizeof(float));
- for (i=0; i<NSUB_MAX; i++)
- iLBCdec_inst->old_syntdenum[i*(LPC_FILTERORDER+1)]=1.0;
-
- iLBCdec_inst->last_lag = 20;
-
- iLBCdec_inst->prevLag = 120;
- iLBCdec_inst->per = 0.0;
- iLBCdec_inst->consPLICount = 0;
- iLBCdec_inst->prevPLI = 0;
- iLBCdec_inst->prevLpc[0] = 1.0;
- memset(iLBCdec_inst->prevLpc+1,0,
- LPC_FILTERORDER*sizeof(float));
- memset(iLBCdec_inst->prevResidual, 0, BLOCKL_MAX*sizeof(float));
- iLBCdec_inst->seed=777;
-
-
-
-
-Andersen, et al. Experimental [Page 65]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- memset(iLBCdec_inst->hpomem, 0, 4*sizeof(float));
-
- iLBCdec_inst->use_enhancer = use_enhancer;
- memset(iLBCdec_inst->enh_buf, 0, ENH_BUFL*sizeof(float));
- for (i=0;i<ENH_NBLOCKS_TOT;i++)
- iLBCdec_inst->enh_period[i]=(float)40.0;
-
- iLBCdec_inst->prev_enh_pl = 0;
-
- return (iLBCdec_inst->blockl);
- }
-
- /*----------------------------------------------------------------*
- * frame residual decoder function (subrutine to iLBC_decode)
- *---------------------------------------------------------------*/
-
- void Decode(
- iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
- structure */
- float *decresidual, /* (o) decoded residual frame */
- int start, /* (i) location of start
- state */
- int idxForMax, /* (i) codebook index for the
- maximum value */
- int *idxVec, /* (i) codebook indexes for the
- samples in the start
- state */
- float *syntdenum, /* (i) the decoded synthesis
- filter coefficients */
- int *cb_index, /* (i) the indexes for the
- adaptive codebook */
- int *gain_index, /* (i) the indexes for the
- corresponding gains */
- int *extra_cb_index, /* (i) the indexes for the
- adaptive codebook part
- of start state */
- int *extra_gain_index, /* (i) the indexes for the
- corresponding gains */
- int state_first /* (i) 1 if non adaptive part
- of start state comes
- first 0 if that part
- comes last */
- ){
- float reverseDecresidual[BLOCKL_MAX], mem[CB_MEML];
- int k, meml_gotten, Nfor, Nback, i;
- int diff, start_pos;
- int subcount, subframe;
-
-
-
-
-Andersen, et al. Experimental [Page 66]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- diff = STATE_LEN - iLBCdec_inst->state_short_len;
-
- if (state_first == 1) {
- start_pos = (start-1)*SUBL;
- } else {
- start_pos = (start-1)*SUBL + diff;
- }
-
- /* decode scalar part of start state */
-
- StateConstructW(idxForMax, idxVec,
- &syntdenum[(start-1)*(LPC_FILTERORDER+1)],
- &decresidual[start_pos], iLBCdec_inst->state_short_len);
-
-
- if (state_first) { /* put adaptive part in the end */
-
- /* setup memory */
-
- memset(mem, 0,
- (CB_MEML-iLBCdec_inst->state_short_len)*sizeof(float));
- memcpy(mem+CB_MEML-iLBCdec_inst->state_short_len,
- decresidual+start_pos,
- iLBCdec_inst->state_short_len*sizeof(float));
-
- /* construct decoded vector */
-
- iCBConstruct(
- &decresidual[start_pos+iLBCdec_inst->state_short_len],
- extra_cb_index, extra_gain_index, mem+CB_MEML-stMemLTbl,
- stMemLTbl, diff, CB_NSTAGES);
-
- }
- else {/* put adaptive part in the beginning */
-
- /* create reversed vectors for prediction */
-
- for (k=0; k<diff; k++) {
- reverseDecresidual[k] =
- decresidual[(start+1)*SUBL-1-
- (k+iLBCdec_inst->state_short_len)];
- }
-
- /* setup memory */
-
- meml_gotten = iLBCdec_inst->state_short_len;
- for (k=0; k<meml_gotten; k++){
- mem[CB_MEML-1-k] = decresidual[start_pos + k];
-
-
-
-Andersen, et al. Experimental [Page 67]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- }
- memset(mem, 0, (CB_MEML-k)*sizeof(float));
-
- /* construct decoded vector */
-
- iCBConstruct(reverseDecresidual, extra_cb_index,
- extra_gain_index, mem+CB_MEML-stMemLTbl, stMemLTbl,
- diff, CB_NSTAGES);
-
- /* get decoded residual from reversed vector */
-
- for (k=0; k<diff; k++) {
- decresidual[start_pos-1-k] = reverseDecresidual[k];
- }
- }
-
- /* counter for predicted sub-frames */
-
- subcount=0;
-
- /* forward prediction of sub-frames */
-
- Nfor = iLBCdec_inst->nsub-start-1;
-
- if ( Nfor > 0 ){
-
- /* setup memory */
-
- memset(mem, 0, (CB_MEML-STATE_LEN)*sizeof(float));
- memcpy(mem+CB_MEML-STATE_LEN, decresidual+(start-1)*SUBL,
- STATE_LEN*sizeof(float));
-
- /* loop over sub-frames to encode */
-
- for (subframe=0; subframe<Nfor; subframe++) {
-
- /* construct decoded vector */
-
- iCBConstruct(&decresidual[(start+1+subframe)*SUBL],
- cb_index+subcount*CB_NSTAGES,
- gain_index+subcount*CB_NSTAGES,
- mem+CB_MEML-memLfTbl[subcount],
- memLfTbl[subcount], SUBL, CB_NSTAGES);
-
- /* update memory */
-
- memcpy(mem, mem+SUBL, (CB_MEML-SUBL)*sizeof(float));
- memcpy(mem+CB_MEML-SUBL,
-
-
-
-Andersen, et al. Experimental [Page 68]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- &decresidual[(start+1+subframe)*SUBL],
- SUBL*sizeof(float));
-
- subcount++;
-
- }
-
- }
-
- /* backward prediction of sub-frames */
-
- Nback = start-1;
-
- if ( Nback > 0 ) {
-
- /* setup memory */
-
- meml_gotten = SUBL*(iLBCdec_inst->nsub+1-start);
-
- if ( meml_gotten > CB_MEML ) {
- meml_gotten=CB_MEML;
- }
- for (k=0; k<meml_gotten; k++) {
- mem[CB_MEML-1-k] = decresidual[(start-1)*SUBL + k];
- }
- memset(mem, 0, (CB_MEML-k)*sizeof(float));
-
- /* loop over subframes to decode */
-
- for (subframe=0; subframe<Nback; subframe++) {
-
- /* construct decoded vector */
-
- iCBConstruct(&reverseDecresidual[subframe*SUBL],
- cb_index+subcount*CB_NSTAGES,
- gain_index+subcount*CB_NSTAGES,
- mem+CB_MEML-memLfTbl[subcount], memLfTbl[subcount],
- SUBL, CB_NSTAGES);
-
- /* update memory */
-
- memcpy(mem, mem+SUBL, (CB_MEML-SUBL)*sizeof(float));
- memcpy(mem+CB_MEML-SUBL,
- &reverseDecresidual[subframe*SUBL],
- SUBL*sizeof(float));
-
- subcount++;
- }
-
-
-
-Andersen, et al. Experimental [Page 69]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- /* get decoded residual from reversed vector */
-
- for (i=0; i<SUBL*Nback; i++)
- decresidual[SUBL*Nback - i - 1] =
- reverseDecresidual[i];
- }
- }
-
- /*----------------------------------------------------------------*
- * main decoder function
- *---------------------------------------------------------------*/
-
- void iLBC_decode(
- float *decblock, /* (o) decoded signal block */
- unsigned char *bytes, /* (i) encoded signal bits */
- iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) the decoder state
- structure */
- int mode /* (i) 0: bad packet, PLC,
- 1: normal */
- ){
- float data[BLOCKL_MAX];
- float lsfdeq[LPC_FILTERORDER*LPC_N_MAX];
- float PLCresidual[BLOCKL_MAX], PLClpc[LPC_FILTERORDER + 1];
- float zeros[BLOCKL_MAX], one[LPC_FILTERORDER + 1];
- int k, i, start, idxForMax, pos, lastpart, ulp;
- int lag, ilag;
- float cc, maxcc;
- int idxVec[STATE_LEN];
- int check;
- int gain_index[NASUB_MAX*CB_NSTAGES],
- extra_gain_index[CB_NSTAGES];
- int cb_index[CB_NSTAGES*NASUB_MAX], extra_cb_index[CB_NSTAGES];
- int lsf_i[LSF_NSPLIT*LPC_N_MAX];
- int state_first;
- int last_bit;
- unsigned char *pbytes;
- float weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
- int order_plus_one;
- float syntdenum[NSUB_MAX*(LPC_FILTERORDER+1)];
- float decresidual[BLOCKL_MAX];
-
- if (mode>0) { /* the data are good */
-
- /* decode data */
-
- pbytes=bytes;
- pos=0;
-
-
-
-
-Andersen, et al. Experimental [Page 70]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- /* Set everything to zero before decoding */
-
- for (k=0; k<LSF_NSPLIT*LPC_N_MAX; k++) {
- lsf_i[k]=0;
- }
- start=0;
- state_first=0;
- idxForMax=0;
- for (k=0; k<iLBCdec_inst->state_short_len; k++) {
- idxVec[k]=0;
- }
- for (k=0; k<CB_NSTAGES; k++) {
- extra_cb_index[k]=0;
- }
- for (k=0; k<CB_NSTAGES; k++) {
- extra_gain_index[k]=0;
- }
- for (i=0; i<iLBCdec_inst->nasub; i++) {
- for (k=0; k<CB_NSTAGES; k++) {
- cb_index[i*CB_NSTAGES+k]=0;
- }
- }
- for (i=0; i<iLBCdec_inst->nasub; i++) {
- for (k=0; k<CB_NSTAGES; k++) {
- gain_index[i*CB_NSTAGES+k]=0;
- }
- }
-
- /* loop over ULP classes */
-
- for (ulp=0; ulp<3; ulp++) {
-
- /* LSF */
- for (k=0; k<LSF_NSPLIT*iLBCdec_inst->lpc_n; k++){
- unpack( &pbytes, &lastpart,
- iLBCdec_inst->ULP_inst->lsf_bits[k][ulp], &pos);
- packcombine(&lsf_i[k], lastpart,
- iLBCdec_inst->ULP_inst->lsf_bits[k][ulp]);
- }
-
- /* Start block info */
-
- unpack( &pbytes, &lastpart,
- iLBCdec_inst->ULP_inst->start_bits[ulp], &pos);
- packcombine(&start, lastpart,
- iLBCdec_inst->ULP_inst->start_bits[ulp]);
-
- unpack( &pbytes, &lastpart,
-
-
-
-Andersen, et al. Experimental [Page 71]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- iLBCdec_inst->ULP_inst->startfirst_bits[ulp], &pos);
- packcombine(&state_first, lastpart,
- iLBCdec_inst->ULP_inst->startfirst_bits[ulp]);
-
- unpack( &pbytes, &lastpart,
- iLBCdec_inst->ULP_inst->scale_bits[ulp], &pos);
- packcombine(&idxForMax, lastpart,
- iLBCdec_inst->ULP_inst->scale_bits[ulp]);
-
- for (k=0; k<iLBCdec_inst->state_short_len; k++) {
- unpack( &pbytes, &lastpart,
- iLBCdec_inst->ULP_inst->state_bits[ulp], &pos);
- packcombine(idxVec+k, lastpart,
- iLBCdec_inst->ULP_inst->state_bits[ulp]);
- }
-
- /* 23/22 (20ms/30ms) sample block */
-
- for (k=0; k<CB_NSTAGES; k++) {
- unpack( &pbytes, &lastpart,
- iLBCdec_inst->ULP_inst->extra_cb_index[k][ulp],
- &pos);
- packcombine(extra_cb_index+k, lastpart,
- iLBCdec_inst->ULP_inst->extra_cb_index[k][ulp]);
- }
- for (k=0; k<CB_NSTAGES; k++) {
- unpack( &pbytes, &lastpart,
- iLBCdec_inst->ULP_inst->extra_cb_gain[k][ulp],
- &pos);
- packcombine(extra_gain_index+k, lastpart,
- iLBCdec_inst->ULP_inst->extra_cb_gain[k][ulp]);
- }
-
- /* The two/four (20ms/30ms) 40 sample sub-blocks */
-
- for (i=0; i<iLBCdec_inst->nasub; i++) {
- for (k=0; k<CB_NSTAGES; k++) {
- unpack( &pbytes, &lastpart,
- iLBCdec_inst->ULP_inst->cb_index[i][k][ulp],
- &pos);
- packcombine(cb_index+i*CB_NSTAGES+k, lastpart,
- iLBCdec_inst->ULP_inst->cb_index[i][k][ulp]);
- }
- }
-
- for (i=0; i<iLBCdec_inst->nasub; i++) {
- for (k=0; k<CB_NSTAGES; k++) {
- unpack( &pbytes, &lastpart,
-
-
-
-Andersen, et al. Experimental [Page 72]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- iLBCdec_inst->ULP_inst->cb_gain[i][k][ulp],
- &pos);
- packcombine(gain_index+i*CB_NSTAGES+k, lastpart,
- iLBCdec_inst->ULP_inst->cb_gain[i][k][ulp]);
- }
- }
- }
- /* Extract last bit. If it is 1 this indicates an
- empty/lost frame */
- unpack( &pbytes, &last_bit, 1, &pos);
-
- /* Check for bit errors or empty/lost frames */
- if (start<1)
- mode = 0;
- if (iLBCdec_inst->mode==20 && start>3)
- mode = 0;
- if (iLBCdec_inst->mode==30 && start>5)
- mode = 0;
- if (last_bit==1)
- mode = 0;
-
- if (mode==1) { /* No bit errors was detected,
- continue decoding */
-
- /* adjust index */
- index_conv_dec(cb_index);
-
- /* decode the lsf */
-
- SimplelsfDEQ(lsfdeq, lsf_i, iLBCdec_inst->lpc_n);
- check=LSF_check(lsfdeq, LPC_FILTERORDER,
- iLBCdec_inst->lpc_n);
- DecoderInterpolateLSF(syntdenum, weightdenum,
- lsfdeq, LPC_FILTERORDER, iLBCdec_inst);
-
- Decode(iLBCdec_inst, decresidual, start, idxForMax,
- idxVec, syntdenum, cb_index, gain_index,
- extra_cb_index, extra_gain_index,
- state_first);
-
- /* preparing the plc for a future loss! */
-
- doThePLC(PLCresidual, PLClpc, 0, decresidual,
- syntdenum +
- (LPC_FILTERORDER + 1)*(iLBCdec_inst->nsub - 1),
- (*iLBCdec_inst).last_lag, iLBCdec_inst);
-
-
-
-
-
-Andersen, et al. Experimental [Page 73]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- memcpy(decresidual, PLCresidual,
- iLBCdec_inst->blockl*sizeof(float));
- }
-
- }
-
- if (mode == 0) {
- /* the data is bad (either a PLC call
- * was made or a severe bit error was detected)
- */
-
- /* packet loss conceal */
-
- memset(zeros, 0, BLOCKL_MAX*sizeof(float));
-
- one[0] = 1;
- memset(one+1, 0, LPC_FILTERORDER*sizeof(float));
-
- start=0;
-
- doThePLC(PLCresidual, PLClpc, 1, zeros, one,
- (*iLBCdec_inst).last_lag, iLBCdec_inst);
- memcpy(decresidual, PLCresidual,
- iLBCdec_inst->blockl*sizeof(float));
-
- order_plus_one = LPC_FILTERORDER + 1;
- for (i = 0; i < iLBCdec_inst->nsub; i++) {
- memcpy(syntdenum+(i*order_plus_one), PLClpc,
- order_plus_one*sizeof(float));
- }
- }
-
- if (iLBCdec_inst->use_enhancer == 1) {
-
- /* post filtering */
-
- iLBCdec_inst->last_lag =
- enhancerInterface(data, decresidual, iLBCdec_inst);
-
- /* synthesis filtering */
-
- if (iLBCdec_inst->mode==20) {
- /* Enhancer has 40 samples delay */
- i=0;
- syntFilter(data + i*SUBL,
- iLBCdec_inst->old_syntdenum +
- (i+iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1),
- SUBL, iLBCdec_inst->syntMem);
-
-
-
-Andersen, et al. Experimental [Page 74]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- for (i=1; i < iLBCdec_inst->nsub; i++) {
- syntFilter(data + i*SUBL,
- syntdenum + (i-1)*(LPC_FILTERORDER+1),
- SUBL, iLBCdec_inst->syntMem);
- }
- } else if (iLBCdec_inst->mode==30) {
- /* Enhancer has 80 samples delay */
- for (i=0; i < 2; i++) {
- syntFilter(data + i*SUBL,
- iLBCdec_inst->old_syntdenum +
- (i+iLBCdec_inst->nsub-2)*(LPC_FILTERORDER+1),
- SUBL, iLBCdec_inst->syntMem);
- }
- for (i=2; i < iLBCdec_inst->nsub; i++) {
- syntFilter(data + i*SUBL,
- syntdenum + (i-2)*(LPC_FILTERORDER+1), SUBL,
- iLBCdec_inst->syntMem);
- }
- }
-
- } else {
-
- /* Find last lag */
- lag = 20;
- maxcc = xCorrCoef(&decresidual[BLOCKL_MAX-ENH_BLOCKL],
- &decresidual[BLOCKL_MAX-ENH_BLOCKL-lag], ENH_BLOCKL);
-
- for (ilag=21; ilag<120; ilag++) {
- cc = xCorrCoef(&decresidual[BLOCKL_MAX-ENH_BLOCKL],
- &decresidual[BLOCKL_MAX-ENH_BLOCKL-ilag],
- ENH_BLOCKL);
-
- if (cc > maxcc) {
- maxcc = cc;
- lag = ilag;
- }
- }
- iLBCdec_inst->last_lag = lag;
-
- /* copy data and run synthesis filter */
-
- memcpy(data, decresidual,
- iLBCdec_inst->blockl*sizeof(float));
- for (i=0; i < iLBCdec_inst->nsub; i++) {
- syntFilter(data + i*SUBL,
- syntdenum + i*(LPC_FILTERORDER+1), SUBL,
- iLBCdec_inst->syntMem);
- }
-
-
-
-Andersen, et al. Experimental [Page 75]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- }
-
- /* high pass filtering on output if desired, otherwise
- copy to out */
-
- hpOutput(data, iLBCdec_inst->blockl,
- decblock,iLBCdec_inst->hpomem);
-
- /* memcpy(decblock,data,iLBCdec_inst->blockl*sizeof(float));*/
-
- memcpy(iLBCdec_inst->old_syntdenum, syntdenum,
-
- iLBCdec_inst->nsub*(LPC_FILTERORDER+1)*sizeof(float));
-
- iLBCdec_inst->prev_enh_pl=0;
-
- if (mode==0) { /* PLC was used */
- iLBCdec_inst->prev_enh_pl=1;
- }
- }
-
-A.6. iLBC_define.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- iLBC_define.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
- #include <string.h>
-
- #ifndef __iLBC_ILBCDEFINE_H
- #define __iLBC_ILBCDEFINE_H
-
- /* general codec settings */
-
- #define FS (float)8000.0
- #define BLOCKL_20MS 160
- #define BLOCKL_30MS 240
- #define BLOCKL_MAX 240
- #define NSUB_20MS 4
- #define NSUB_30MS 6
- #define NSUB_MAX 6
- #define NASUB_20MS 2
-
-
-
-Andersen, et al. Experimental [Page 76]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- #define NASUB_30MS 4
- #define NASUB_MAX 4
- #define SUBL 40
- #define STATE_LEN 80
- #define STATE_SHORT_LEN_30MS 58
- #define STATE_SHORT_LEN_20MS 57
-
- /* LPC settings */
-
- #define LPC_FILTERORDER 10
- #define LPC_CHIRP_SYNTDENUM (float)0.9025
- #define LPC_CHIRP_WEIGHTDENUM (float)0.4222
- #define LPC_LOOKBACK 60
- #define LPC_N_20MS 1
- #define LPC_N_30MS 2
- #define LPC_N_MAX 2
- #define LPC_ASYMDIFF 20
- #define LPC_BW (float)60.0
- #define LPC_WN (float)1.0001
- #define LSF_NSPLIT 3
- #define LSF_NUMBER_OF_STEPS 4
- #define LPC_HALFORDER (LPC_FILTERORDER/2)
-
- /* cb settings */
-
- #define CB_NSTAGES 3
- #define CB_EXPAND 2
- #define CB_MEML 147
- #define CB_FILTERLEN 2*4
- #define CB_HALFFILTERLEN 4
- #define CB_RESRANGE 34
- #define CB_MAXGAIN (float)1.3
-
- /* enhancer */
-
- #define ENH_BLOCKL 80 /* block length */
- #define ENH_BLOCKL_HALF (ENH_BLOCKL/2)
- #define ENH_HL 3 /* 2*ENH_HL+1 is number blocks
- in said second sequence */
- #define ENH_SLOP 2 /* max difference estimated and
- correct pitch period */
- #define ENH_PLOCSL 20 /* pitch-estimates and pitch-
- locations buffer length */
- #define ENH_OVERHANG 2
- #define ENH_UPS0 4 /* upsampling rate */
- #define ENH_FL0 3 /* 2*FLO+1 is the length of
- each filter */
- #define ENH_VECTL (ENH_BLOCKL+2*ENH_FL0)
-
-
-
-Andersen, et al. Experimental [Page 77]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- #define ENH_CORRDIM (2*ENH_SLOP+1)
- #define ENH_NBLOCKS (BLOCKL_MAX/ENH_BLOCKL)
- #define ENH_NBLOCKS_EXTRA 5
- #define ENH_NBLOCKS_TOT 8 /* ENH_NBLOCKS +
- ENH_NBLOCKS_EXTRA */
- #define ENH_BUFL (ENH_NBLOCKS_TOT)*ENH_BLOCKL
- #define ENH_ALPHA0 (float)0.05
-
- /* Down sampling */
-
- #define FILTERORDER_DS 7
- #define DELAY_DS 3
- #define FACTOR_DS 2
-
- /* bit stream defs */
-
- #define NO_OF_BYTES_20MS 38
- #define NO_OF_BYTES_30MS 50
- #define NO_OF_WORDS_20MS 19
- #define NO_OF_WORDS_30MS 25
- #define STATE_BITS 3
- #define BYTE_LEN 8
- #define ULP_CLASSES 3
-
- /* help parameters */
-
- #define FLOAT_MAX (float)1.0e37
- #define EPS (float)2.220446049250313e-016
- #define PI (float)3.14159265358979323846
- #define MIN_SAMPLE -32768
- #define MAX_SAMPLE 32767
- #define TWO_PI (float)6.283185307
- #define PI2 (float)0.159154943
-
- /* type definition encoder instance */
- typedef struct iLBC_ULP_Inst_t_ {
- int lsf_bits[6][ULP_CLASSES+2];
- int start_bits[ULP_CLASSES+2];
- int startfirst_bits[ULP_CLASSES+2];
- int scale_bits[ULP_CLASSES+2];
- int state_bits[ULP_CLASSES+2];
- int extra_cb_index[CB_NSTAGES][ULP_CLASSES+2];
- int extra_cb_gain[CB_NSTAGES][ULP_CLASSES+2];
- int cb_index[NSUB_MAX][CB_NSTAGES][ULP_CLASSES+2];
- int cb_gain[NSUB_MAX][CB_NSTAGES][ULP_CLASSES+2];
- } iLBC_ULP_Inst_t;
-
- /* type definition encoder instance */
-
-
-
-Andersen, et al. Experimental [Page 78]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- typedef struct iLBC_Enc_Inst_t_ {
-
- /* flag for frame size mode */
- int mode;
-
- /* basic parameters for different frame sizes */
- int blockl;
- int nsub;
- int nasub;
- int no_of_bytes, no_of_words;
- int lpc_n;
- int state_short_len;
- const iLBC_ULP_Inst_t *ULP_inst;
-
- /* analysis filter state */
- float anaMem[LPC_FILTERORDER];
-
- /* old lsf parameters for interpolation */
- float lsfold[LPC_FILTERORDER];
- float lsfdeqold[LPC_FILTERORDER];
-
- /* signal buffer for LP analysis */
- float lpc_buffer[LPC_LOOKBACK + BLOCKL_MAX];
-
- /* state of input HP filter */
- float hpimem[4];
-
- } iLBC_Enc_Inst_t;
-
- /* type definition decoder instance */
- typedef struct iLBC_Dec_Inst_t_ {
-
- /* flag for frame size mode */
- int mode;
-
- /* basic parameters for different frame sizes */
- int blockl;
- int nsub;
- int nasub;
- int no_of_bytes, no_of_words;
- int lpc_n;
- int state_short_len;
- const iLBC_ULP_Inst_t *ULP_inst;
-
- /* synthesis filter state */
- float syntMem[LPC_FILTERORDER];
-
- /* old LSF for interpolation */
-
-
-
-Andersen, et al. Experimental [Page 79]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- float lsfdeqold[LPC_FILTERORDER];
-
- /* pitch lag estimated in enhancer and used in PLC */
- int last_lag;
-
- /* PLC state information */
- int prevLag, consPLICount, prevPLI, prev_enh_pl;
- float prevLpc[LPC_FILTERORDER+1];
- float prevResidual[NSUB_MAX*SUBL];
- float per;
- unsigned long seed;
-
- /* previous synthesis filter parameters */
- float old_syntdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
-
- /* state of output HP filter */
- float hpomem[4];
-
- /* enhancer state information */
- int use_enhancer;
- float enh_buf[ENH_BUFL];
- float enh_period[ENH_NBLOCKS_TOT];
-
- } iLBC_Dec_Inst_t;
-
- #endif
-
-A.7. constants.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- constants.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_CONSTANTS_H
- #define __iLBC_CONSTANTS_H
-
- #include "iLBC_define.h"
-
-
- /* ULP bit allocation */
-
-
-
-
-Andersen, et al. Experimental [Page 80]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- extern const iLBC_ULP_Inst_t ULP_20msTbl;
- extern const iLBC_ULP_Inst_t ULP_30msTbl;
-
- /* high pass filters */
-
- extern float hpi_zero_coefsTbl[];
- extern float hpi_pole_coefsTbl[];
- extern float hpo_zero_coefsTbl[];
- extern float hpo_pole_coefsTbl[];
-
- /* low pass filters */
- extern float lpFilt_coefsTbl[];
-
- /* LPC analysis and quantization */
-
- extern float lpc_winTbl[];
- extern float lpc_asymwinTbl[];
- extern float lpc_lagwinTbl[];
- extern float lsfCbTbl[];
- extern float lsfmeanTbl[];
- extern int dim_lsfCbTbl[];
- extern int size_lsfCbTbl[];
- extern float lsf_weightTbl_30ms[];
- extern float lsf_weightTbl_20ms[];
-
- /* state quantization tables */
-
- extern float state_sq3Tbl[];
- extern float state_frgqTbl[];
-
- /* gain quantization tables */
-
- extern float gain_sq3Tbl[];
- extern float gain_sq4Tbl[];
- extern float gain_sq5Tbl[];
-
- /* adaptive codebook definitions */
-
- extern int search_rangeTbl[5][CB_NSTAGES];
- extern int memLfTbl[];
- extern int stMemLTbl;
- extern float cbfiltersTbl[CB_FILTERLEN];
-
- /* enhancer definitions */
-
- extern float polyphaserTbl[];
- extern float enh_plocsTbl[];
-
-
-
-
-Andersen, et al. Experimental [Page 81]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- #endif
-
-A.8. constants.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- constants.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include "iLBC_define.h"
-
- /* ULP bit allocation */
-
- /* 20 ms frame */
-
- const iLBC_ULP_Inst_t ULP_20msTbl = {
- /* LSF */
- { {6,0,0,0,0}, {7,0,0,0,0}, {7,0,0,0,0},
- {0,0,0,0,0}, {0,0,0,0,0}, {0,0,0,0,0}},
- /* Start state location, gain and samples */
- {2,0,0,0,0},
- {1,0,0,0,0},
- {6,0,0,0,0},
- {0,1,2,0,0},
- /* extra CB index and extra CB gain */
- {{6,0,1,0,0}, {0,0,7,0,0}, {0,0,7,0,0}},
- {{2,0,3,0,0}, {1,1,2,0,0}, {0,0,3,0,0}},
- /* CB index and CB gain */
- { {{7,0,1,0,0}, {0,0,7,0,0}, {0,0,7,0,0}},
- {{0,0,8,0,0}, {0,0,8,0,0}, {0,0,8,0,0}},
- {{0,0,0,0,0}, {0,0,0,0,0}, {0,0,0,0,0}},
- {{0,0,0,0,0}, {0,0,0,0,0}, {0,0,0,0,0}}},
- { {{1,2,2,0,0}, {1,1,2,0,0}, {0,0,3,0,0}},
- {{1,1,3,0,0}, {0,2,2,0,0}, {0,0,3,0,0}},
- {{0,0,0,0,0}, {0,0,0,0,0}, {0,0,0,0,0}},
- {{0,0,0,0,0}, {0,0,0,0,0}, {0,0,0,0,0}}}
- };
-
- /* 30 ms frame */
-
- const iLBC_ULP_Inst_t ULP_30msTbl = {
- /* LSF */
-
-
-
-Andersen, et al. Experimental [Page 82]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- { {6,0,0,0,0}, {7,0,0,0,0}, {7,0,0,0,0},
- {6,0,0,0,0}, {7,0,0,0,0}, {7,0,0,0,0}},
- /* Start state location, gain and samples */
- {3,0,0,0,0},
- {1,0,0,0,0},
- {6,0,0,0,0},
- {0,1,2,0,0},
- /* extra CB index and extra CB gain */
- {{4,2,1,0,0}, {0,0,7,0,0}, {0,0,7,0,0}},
- {{1,1,3,0,0}, {1,1,2,0,0}, {0,0,3,0,0}},
- /* CB index and CB gain */
- { {{6,1,1,0,0}, {0,0,7,0,0}, {0,0,7,0,0}},
- {{0,7,1,0,0}, {0,0,8,0,0}, {0,0,8,0,0}},
- {{0,7,1,0,0}, {0,0,8,0,0}, {0,0,8,0,0}},
- {{0,7,1,0,0}, {0,0,8,0,0}, {0,0,8,0,0}}},
- { {{1,2,2,0,0}, {1,2,1,0,0}, {0,0,3,0,0}},
- {{0,2,3,0,0}, {0,2,2,0,0}, {0,0,3,0,0}},
- {{0,1,4,0,0}, {0,1,3,0,0}, {0,0,3,0,0}},
- {{0,1,4,0,0}, {0,1,3,0,0}, {0,0,3,0,0}}}
- };
-
- /* HP Filters */
-
- float hpi_zero_coefsTbl[3] = {
- (float)0.92727436, (float)-1.8544941, (float)0.92727436
- };
- float hpi_pole_coefsTbl[3] = {
- (float)1.0, (float)-1.9059465, (float)0.9114024
- };
- float hpo_zero_coefsTbl[3] = {
- (float)0.93980581, (float)-1.8795834, (float)0.93980581
- };
- float hpo_pole_coefsTbl[3] = {
- (float)1.0, (float)-1.9330735, (float)0.93589199
- };
-
- /* LP Filter */
-
- float lpFilt_coefsTbl[FILTERORDER_DS]={
- (float)-0.066650, (float)0.125000, (float)0.316650,
- (float)0.414063, (float)0.316650,
- (float)0.125000, (float)-0.066650
- };
-
- /* State quantization tables */
-
- float state_sq3Tbl[8] = {
- (float)-3.719849, (float)-2.177490, (float)-1.130005,
-
-
-
-Andersen, et al. Experimental [Page 83]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- (float)-0.309692, (float)0.444214, (float)1.329712,
- (float)2.436279, (float)3.983887
- };
-
- float state_frgqTbl[64] = {
- (float)1.000085, (float)1.071695, (float)1.140395,
- (float)1.206868, (float)1.277188, (float)1.351503,
- (float)1.429380, (float)1.500727, (float)1.569049,
- (float)1.639599, (float)1.707071, (float)1.781531,
- (float)1.840799, (float)1.901550, (float)1.956695,
- (float)2.006750, (float)2.055474, (float)2.102787,
- (float)2.142819, (float)2.183592, (float)2.217962,
- (float)2.257177, (float)2.295739, (float)2.332967,
- (float)2.369248, (float)2.402792, (float)2.435080,
- (float)2.468598, (float)2.503394, (float)2.539284,
- (float)2.572944, (float)2.605036, (float)2.636331,
- (float)2.668939, (float)2.698780, (float)2.729101,
- (float)2.759786, (float)2.789834, (float)2.818679,
- (float)2.848074, (float)2.877470, (float)2.906899,
- (float)2.936655, (float)2.967804, (float)3.000115,
- (float)3.033367, (float)3.066355, (float)3.104231,
- (float)3.141499, (float)3.183012, (float)3.222952,
- (float)3.265433, (float)3.308441, (float)3.350823,
- (float)3.395275, (float)3.442793, (float)3.490801,
- (float)3.542514, (float)3.604064, (float)3.666050,
- (float)3.740994, (float)3.830749, (float)3.938770,
- (float)4.101764
- };
-
- /* CB tables */
-
- int search_rangeTbl[5][CB_NSTAGES]={{58,58,58}, {108,44,44},
- {108,108,108}, {108,108,108}, {108,108,108}};
- int stMemLTbl=85;
- int memLfTbl[NASUB_MAX]={147,147,147,147};
-
- /* expansion filter(s) */
-
- float cbfiltersTbl[CB_FILTERLEN]={
- (float)-0.034180, (float)0.108887, (float)-0.184326,
- (float)0.806152, (float)0.713379, (float)-0.144043,
- (float)0.083740, (float)-0.033691
- };
-
- /* Gain Quantization */
-
- float gain_sq3Tbl[8]={
- (float)-1.000000, (float)-0.659973, (float)-0.330017,
-
-
-
-Andersen, et al. Experimental [Page 84]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- (float)0.000000, (float)0.250000, (float)0.500000,
- (float)0.750000, (float)1.00000};
-
- float gain_sq4Tbl[16]={
- (float)-1.049988, (float)-0.900024, (float)-0.750000,
- (float)-0.599976, (float)-0.450012, (float)-0.299988,
- (float)-0.150024, (float)0.000000, (float)0.150024,
- (float)0.299988, (float)0.450012, (float)0.599976,
- (float)0.750000, (float)0.900024, (float)1.049988,
- (float)1.200012};
-
- float gain_sq5Tbl[32]={
- (float)0.037476, (float)0.075012, (float)0.112488,
- (float)0.150024, (float)0.187500, (float)0.224976,
- (float)0.262512, (float)0.299988, (float)0.337524,
- (float)0.375000, (float)0.412476, (float)0.450012,
- (float)0.487488, (float)0.525024, (float)0.562500,
- (float)0.599976, (float)0.637512, (float)0.674988,
- (float)0.712524, (float)0.750000, (float)0.787476,
- (float)0.825012, (float)0.862488, (float)0.900024,
- (float)0.937500, (float)0.974976, (float)1.012512,
- (float)1.049988, (float)1.087524, (float)1.125000,
- (float)1.162476, (float)1.200012};
-
- /* Enhancer - Upsamling a factor 4 (ENH_UPS0 = 4) */
- float polyphaserTbl[ENH_UPS0*(2*ENH_FL0+1)]={
- (float)0.000000, (float)0.000000, (float)0.000000,
- (float)1.000000,
- (float)0.000000, (float)0.000000, (float)0.000000,
- (float)0.015625, (float)-0.076904, (float)0.288330,
- (float)0.862061,
- (float)-0.106445, (float)0.018799, (float)-0.015625,
- (float)0.023682, (float)-0.124268, (float)0.601563,
- (float)0.601563,
- (float)-0.124268, (float)0.023682, (float)-0.023682,
- (float)0.018799, (float)-0.106445, (float)0.862061,
- (float)0.288330,
- (float)-0.076904, (float)0.015625, (float)-0.018799};
-
- float enh_plocsTbl[ENH_NBLOCKS_TOT] = {(float)40.0, (float)120.0,
- (float)200.0, (float)280.0, (float)360.0,
- (float)440.0, (float)520.0, (float)600.0};
-
- /* LPC analysis and quantization */
-
- int dim_lsfCbTbl[LSF_NSPLIT] = {3, 3, 4};
- int size_lsfCbTbl[LSF_NSPLIT] = {64,128,128};
-
-
-
-
-Andersen, et al. Experimental [Page 85]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- float lsfmeanTbl[LPC_FILTERORDER] = {
- (float)0.281738, (float)0.445801, (float)0.663330,
- (float)0.962524, (float)1.251831, (float)1.533081,
- (float)1.850586, (float)2.137817, (float)2.481445,
- (float)2.777344};
-
- float lsf_weightTbl_30ms[6] = {(float)(1.0/2.0), (float)1.0,
- (float)(2.0/3.0),
- (float)(1.0/3.0), (float)0.0, (float)0.0};
-
- float lsf_weightTbl_20ms[4] = {(float)(3.0/4.0), (float)(2.0/4.0),
- (float)(1.0/4.0), (float)(0.0)};
-
- /* Hanning LPC window */
- float lpc_winTbl[BLOCKL_MAX]={
- (float)0.000183, (float)0.000671, (float)0.001526,
- (float)0.002716, (float)0.004242, (float)0.006104,
- (float)0.008301, (float)0.010834, (float)0.013702,
- (float)0.016907, (float)0.020416, (float)0.024261,
- (float)0.028442, (float)0.032928, (float)0.037750,
- (float)0.042877, (float)0.048309, (float)0.054047,
- (float)0.060089, (float)0.066437, (float)0.073090,
- (float)0.080017, (float)0.087219, (float)0.094727,
- (float)0.102509, (float)0.110535, (float)0.118835,
- (float)0.127411, (float)0.136230, (float)0.145294,
- (float)0.154602, (float)0.164154, (float)0.173920,
- (float)0.183899, (float)0.194122, (float)0.204529,
- (float)0.215149, (float)0.225952, (float)0.236938,
- (float)0.248108, (float)0.259460, (float)0.270966,
- (float)0.282654, (float)0.294464, (float)0.306396,
- (float)0.318481, (float)0.330688, (float)0.343018,
- (float)0.355438, (float)0.367981, (float)0.380585,
- (float)0.393280, (float)0.406067, (float)0.418884,
- (float)0.431763, (float)0.444702, (float)0.457672,
- (float)0.470673, (float)0.483704, (float)0.496735,
- (float)0.509766, (float)0.522797, (float)0.535828,
- (float)0.548798, (float)0.561768, (float)0.574677,
- (float)0.587524, (float)0.600342, (float)0.613068,
- (float)0.625732, (float)0.638306, (float)0.650787,
- (float)0.663147, (float)0.675415, (float)0.687561,
- (float)0.699585, (float)0.711487, (float)0.723206,
- (float)0.734802, (float)0.746216, (float)0.757477,
- (float)0.768585, (float)0.779480, (float)0.790192,
- (float)0.800720, (float)0.811005, (float)0.821106,
- (float)0.830994, (float)0.840668, (float)0.850067,
- (float)0.859253, (float)0.868225, (float)0.876892,
- (float)0.885345, (float)0.893524, (float)0.901428,
- (float)0.909058, (float)0.916412, (float)0.923492,
-
-
-
-Andersen, et al. Experimental [Page 86]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- (float)0.930267, (float)0.936768, (float)0.942963,
- (float)0.948853, (float)0.954437, (float)0.959717,
- (float)0.964691, (float)0.969360, (float)0.973694,
- (float)0.977692, (float)0.981384, (float)0.984741,
- (float)0.987762, (float)0.990479, (float)0.992828,
- (float)0.994873, (float)0.996552, (float)0.997925,
- (float)0.998932, (float)0.999603, (float)0.999969,
- (float)0.999969, (float)0.999603, (float)0.998932,
- (float)0.997925, (float)0.996552, (float)0.994873,
- (float)0.992828, (float)0.990479, (float)0.987762,
- (float)0.984741, (float)0.981384, (float)0.977692,
- (float)0.973694, (float)0.969360, (float)0.964691,
- (float)0.959717, (float)0.954437, (float)0.948853,
- (float)0.942963, (float)0.936768, (float)0.930267,
- (float)0.923492, (float)0.916412, (float)0.909058,
- (float)0.901428, (float)0.893524, (float)0.885345,
- (float)0.876892, (float)0.868225, (float)0.859253,
- (float)0.850067, (float)0.840668, (float)0.830994,
- (float)0.821106, (float)0.811005, (float)0.800720,
- (float)0.790192, (float)0.779480, (float)0.768585,
- (float)0.757477, (float)0.746216, (float)0.734802,
- (float)0.723206, (float)0.711487, (float)0.699585,
- (float)0.687561, (float)0.675415, (float)0.663147,
- (float)0.650787, (float)0.638306, (float)0.625732,
- (float)0.613068, (float)0.600342, (float)0.587524,
- (float)0.574677, (float)0.561768, (float)0.548798,
- (float)0.535828, (float)0.522797, (float)0.509766,
- (float)0.496735, (float)0.483704, (float)0.470673,
- (float)0.457672, (float)0.444702, (float)0.431763,
- (float)0.418884, (float)0.406067, (float)0.393280,
- (float)0.380585, (float)0.367981, (float)0.355438,
- (float)0.343018, (float)0.330688, (float)0.318481,
- (float)0.306396, (float)0.294464, (float)0.282654,
- (float)0.270966, (float)0.259460, (float)0.248108,
- (float)0.236938, (float)0.225952, (float)0.215149,
- (float)0.204529, (float)0.194122, (float)0.183899,
- (float)0.173920, (float)0.164154, (float)0.154602,
- (float)0.145294, (float)0.136230, (float)0.127411,
- (float)0.118835, (float)0.110535, (float)0.102509,
- (float)0.094727, (float)0.087219, (float)0.080017,
- (float)0.073090, (float)0.066437, (float)0.060089,
- (float)0.054047, (float)0.048309, (float)0.042877,
- (float)0.037750, (float)0.032928, (float)0.028442,
- (float)0.024261, (float)0.020416, (float)0.016907,
- (float)0.013702, (float)0.010834, (float)0.008301,
- (float)0.006104, (float)0.004242, (float)0.002716,
- (float)0.001526, (float)0.000671, (float)0.000183
- };
-
-
-
-Andersen, et al. Experimental [Page 87]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- /* Asymmetric LPC window */
- float lpc_asymwinTbl[BLOCKL_MAX]={
- (float)0.000061, (float)0.000214, (float)0.000458,
- (float)0.000824, (float)0.001282, (float)0.001831,
- (float)0.002472, (float)0.003235, (float)0.004120,
- (float)0.005066, (float)0.006134, (float)0.007294,
- (float)0.008545, (float)0.009918, (float)0.011383,
- (float)0.012939, (float)0.014587, (float)0.016357,
- (float)0.018219, (float)0.020172, (float)0.022217,
- (float)0.024353, (float)0.026611, (float)0.028961,
- (float)0.031372, (float)0.033905, (float)0.036530,
- (float)0.039276, (float)0.042084, (float)0.044983,
- (float)0.047974, (float)0.051086, (float)0.054260,
- (float)0.057526, (float)0.060883, (float)0.064331,
- (float)0.067871, (float)0.071503, (float)0.075226,
- (float)0.079010, (float)0.082916, (float)0.086884,
- (float)0.090942, (float)0.095062, (float)0.099304,
- (float)0.103607, (float)0.107971, (float)0.112427,
- (float)0.116974, (float)0.121582, (float)0.126282,
- (float)0.131073, (float)0.135895, (float)0.140839,
- (float)0.145813, (float)0.150879, (float)0.156006,
- (float)0.161224, (float)0.166504, (float)0.171844,
- (float)0.177246, (float)0.182709, (float)0.188263,
- (float)0.193848, (float)0.199524, (float)0.205231,
- (float)0.211029, (float)0.216858, (float)0.222778,
- (float)0.228729, (float)0.234741, (float)0.240814,
- (float)0.246918, (float)0.253082, (float)0.259308,
- (float)0.265564, (float)0.271881, (float)0.278259,
- (float)0.284668, (float)0.291107, (float)0.297607,
- (float)0.304138, (float)0.310730, (float)0.317322,
- (float)0.323975, (float)0.330658, (float)0.337372,
- (float)0.344147, (float)0.350922, (float)0.357727,
- (float)0.364594, (float)0.371460, (float)0.378357,
- (float)0.385284, (float)0.392212, (float)0.399170,
- (float)0.406158, (float)0.413177, (float)0.420197,
- (float)0.427246, (float)0.434296, (float)0.441376,
- (float)0.448456, (float)0.455536, (float)0.462646,
- (float)0.469757, (float)0.476868, (float)0.483978,
- (float)0.491089, (float)0.498230, (float)0.505341,
- (float)0.512451, (float)0.519592, (float)0.526703,
- (float)0.533813, (float)0.540924, (float)0.548004,
- (float)0.555084, (float)0.562164, (float)0.569244,
- (float)0.576294, (float)0.583313, (float)0.590332,
- (float)0.597321, (float)0.604309, (float)0.611267,
- (float)0.618195, (float)0.625092, (float)0.631989,
- (float)0.638855, (float)0.645660, (float)0.652466,
- (float)0.659241, (float)0.665985, (float)0.672668,
- (float)0.679352, (float)0.685974, (float)0.692566,
-
-
-
-Andersen, et al. Experimental [Page 88]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- (float)0.699127, (float)0.705658, (float)0.712128,
- (float)0.718536, (float)0.724945, (float)0.731262,
- (float)0.737549, (float)0.743805, (float)0.750000,
- (float)0.756134, (float)0.762238, (float)0.768280,
- (float)0.774261, (float)0.780182, (float)0.786072,
- (float)0.791870, (float)0.797638, (float)0.803314,
- (float)0.808960, (float)0.814514, (float)0.820038,
- (float)0.825470, (float)0.830841, (float)0.836151,
- (float)0.841400, (float)0.846558, (float)0.851654,
- (float)0.856689, (float)0.861633, (float)0.866516,
- (float)0.871338, (float)0.876068, (float)0.880737,
- (float)0.885315, (float)0.889801, (float)0.894226,
- (float)0.898560, (float)0.902832, (float)0.907013,
- (float)0.911102, (float)0.915100, (float)0.919037,
- (float)0.922882, (float)0.926636, (float)0.930328,
- (float)0.933899, (float)0.937408, (float)0.940796,
- (float)0.944122, (float)0.947357, (float)0.950470,
- (float)0.953522, (float)0.956482, (float)0.959351,
- (float)0.962097, (float)0.964783, (float)0.967377,
- (float)0.969849, (float)0.972229, (float)0.974518,
- (float)0.976715, (float)0.978821, (float)0.980835,
- (float)0.982727, (float)0.984528, (float)0.986237,
- (float)0.987854, (float)0.989380, (float)0.990784,
- (float)0.992096, (float)0.993317, (float)0.994415,
- (float)0.995422, (float)0.996338, (float)0.997162,
- (float)0.997864, (float)0.998474, (float)0.998962,
- (float)0.999390, (float)0.999695, (float)0.999878,
- (float)0.999969, (float)0.999969, (float)0.996918,
- (float)0.987701, (float)0.972382, (float)0.951050,
- (float)0.923889, (float)0.891022, (float)0.852631,
- (float)0.809021, (float)0.760406, (float)0.707092,
- (float)0.649445, (float)0.587799, (float)0.522491,
- (float)0.453979, (float)0.382690, (float)0.309021,
- (float)0.233459, (float)0.156433, (float)0.078461
- };
-
- /* Lag window for LPC */
- float lpc_lagwinTbl[LPC_FILTERORDER + 1]={
- (float)1.000100, (float)0.998890, (float)0.995569,
- (float)0.990057, (float)0.982392,
- (float)0.972623, (float)0.960816, (float)0.947047,
- (float)0.931405, (float)0.913989, (float)0.894909};
-
- /* LSF quantization*/
- float lsfCbTbl[64 * 3 + 128 * 3 + 128 * 4] = {
- (float)0.155396, (float)0.273193, (float)0.451172,
- (float)0.390503, (float)0.648071, (float)1.002075,
- (float)0.440186, (float)0.692261, (float)0.955688,
-
-
-
-Andersen, et al. Experimental [Page 89]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- (float)0.343628, (float)0.642334, (float)1.071533,
- (float)0.318359, (float)0.491577, (float)0.670532,
- (float)0.193115, (float)0.375488, (float)0.725708,
- (float)0.364136, (float)0.510376, (float)0.658691,
- (float)0.297485, (float)0.527588, (float)0.842529,
- (float)0.227173, (float)0.365967, (float)0.563110,
- (float)0.244995, (float)0.396729, (float)0.636475,
- (float)0.169434, (float)0.300171, (float)0.520264,
- (float)0.312866, (float)0.464478, (float)0.643188,
- (float)0.248535, (float)0.429932, (float)0.626099,
- (float)0.236206, (float)0.491333, (float)0.817139,
- (float)0.334961, (float)0.625122, (float)0.895752,
- (float)0.343018, (float)0.518555, (float)0.698608,
- (float)0.372803, (float)0.659790, (float)0.945435,
- (float)0.176880, (float)0.316528, (float)0.581421,
- (float)0.416382, (float)0.625977, (float)0.805176,
- (float)0.303223, (float)0.568726, (float)0.915039,
- (float)0.203613, (float)0.351440, (float)0.588135,
- (float)0.221191, (float)0.375000, (float)0.614746,
- (float)0.199951, (float)0.323364, (float)0.476074,
- (float)0.300781, (float)0.433350, (float)0.566895,
- (float)0.226196, (float)0.354004, (float)0.507568,
- (float)0.300049, (float)0.508179, (float)0.711670,
- (float)0.312012, (float)0.492676, (float)0.763428,
- (float)0.329956, (float)0.541016, (float)0.795776,
- (float)0.373779, (float)0.604614, (float)0.928833,
- (float)0.210571, (float)0.452026, (float)0.755249,
- (float)0.271118, (float)0.473267, (float)0.662476,
- (float)0.285522, (float)0.436890, (float)0.634399,
- (float)0.246704, (float)0.565552, (float)0.859009,
- (float)0.270508, (float)0.406250, (float)0.553589,
- (float)0.361450, (float)0.578491, (float)0.813843,
- (float)0.342651, (float)0.482788, (float)0.622437,
- (float)0.340332, (float)0.549438, (float)0.743164,
- (float)0.200439, (float)0.336304, (float)0.540894,
- (float)0.407837, (float)0.644775, (float)0.895142,
- (float)0.294678, (float)0.454834, (float)0.699097,
- (float)0.193115, (float)0.344482, (float)0.643188,
- (float)0.275757, (float)0.420776, (float)0.598755,
- (float)0.380493, (float)0.608643, (float)0.861084,
- (float)0.222778, (float)0.426147, (float)0.676514,
- (float)0.407471, (float)0.700195, (float)1.053101,
- (float)0.218384, (float)0.377197, (float)0.669922,
- (float)0.313232, (float)0.454102, (float)0.600952,
- (float)0.347412, (float)0.571533, (float)0.874146,
- (float)0.238037, (float)0.405396, (float)0.729492,
- (float)0.223877, (float)0.412964, (float)0.822021,
- (float)0.395264, (float)0.582153, (float)0.743896,
-
-
-
-Andersen, et al. Experimental [Page 90]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- (float)0.247925, (float)0.485596, (float)0.720581,
- (float)0.229126, (float)0.496582, (float)0.907715,
- (float)0.260132, (float)0.566895, (float)1.012695,
- (float)0.337402, (float)0.611572, (float)0.978149,
- (float)0.267822, (float)0.447632, (float)0.769287,
- (float)0.250610, (float)0.381714, (float)0.530029,
- (float)0.430054, (float)0.805054, (float)1.221924,
- (float)0.382568, (float)0.544067, (float)0.701660,
- (float)0.383545, (float)0.710327, (float)1.149170,
- (float)0.271362, (float)0.529053, (float)0.775513,
- (float)0.246826, (float)0.393555, (float)0.588623,
- (float)0.266846, (float)0.422119, (float)0.676758,
- (float)0.311523, (float)0.580688, (float)0.838623,
- (float)1.331177, (float)1.576782, (float)1.779541,
- (float)1.160034, (float)1.401978, (float)1.768188,
- (float)1.161865, (float)1.525146, (float)1.715332,
- (float)0.759521, (float)0.913940, (float)1.119873,
- (float)0.947144, (float)1.121338, (float)1.282471,
- (float)1.015015, (float)1.557007, (float)1.804932,
- (float)1.172974, (float)1.402100, (float)1.692627,
- (float)1.087524, (float)1.474243, (float)1.665405,
- (float)0.899536, (float)1.105225, (float)1.406250,
- (float)1.148438, (float)1.484741, (float)1.796265,
- (float)0.785645, (float)1.209839, (float)1.567749,
- (float)0.867798, (float)1.166504, (float)1.450684,
- (float)0.922485, (float)1.229858, (float)1.420898,
- (float)0.791260, (float)1.123291, (float)1.409546,
- (float)0.788940, (float)0.966064, (float)1.340332,
- (float)1.051147, (float)1.272827, (float)1.556641,
- (float)0.866821, (float)1.181152, (float)1.538818,
- (float)0.906738, (float)1.373535, (float)1.607910,
- (float)1.244751, (float)1.581421, (float)1.933838,
- (float)0.913940, (float)1.337280, (float)1.539673,
- (float)0.680542, (float)0.959229, (float)1.662720,
- (float)0.887207, (float)1.430542, (float)1.800781,
- (float)0.912598, (float)1.433594, (float)1.683960,
- (float)0.860474, (float)1.060303, (float)1.455322,
- (float)1.005127, (float)1.381104, (float)1.706909,
- (float)0.800781, (float)1.363892, (float)1.829102,
- (float)0.781860, (float)1.124390, (float)1.505981,
- (float)1.003662, (float)1.471436, (float)1.684692,
- (float)0.981323, (float)1.309570, (float)1.618042,
- (float)1.228760, (float)1.554321, (float)1.756470,
- (float)0.734375, (float)0.895752, (float)1.225586,
- (float)0.841797, (float)1.055664, (float)1.249268,
- (float)0.920166, (float)1.119385, (float)1.486206,
- (float)0.894409, (float)1.539063, (float)1.828979,
- (float)1.283691, (float)1.543335, (float)1.858276,
-
-
-
-Andersen, et al. Experimental [Page 91]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- (float)0.676025, (float)0.933105, (float)1.490845,
- (float)0.821289, (float)1.491821, (float)1.739868,
- (float)0.923218, (float)1.144653, (float)1.580566,
- (float)1.057251, (float)1.345581, (float)1.635864,
- (float)0.888672, (float)1.074951, (float)1.353149,
- (float)0.942749, (float)1.195435, (float)1.505493,
- (float)1.492310, (float)1.788086, (float)2.039673,
- (float)1.070313, (float)1.634399, (float)1.860962,
- (float)1.253296, (float)1.488892, (float)1.686035,
- (float)0.647095, (float)0.864014, (float)1.401855,
- (float)0.866699, (float)1.254883, (float)1.453369,
- (float)1.063965, (float)1.532593, (float)1.731323,
- (float)1.167847, (float)1.521484, (float)1.884033,
- (float)0.956055, (float)1.502075, (float)1.745605,
- (float)0.928711, (float)1.288574, (float)1.479614,
- (float)1.088013, (float)1.380737, (float)1.570801,
- (float)0.905029, (float)1.186768, (float)1.371948,
- (float)1.057861, (float)1.421021, (float)1.617432,
- (float)1.108276, (float)1.312500, (float)1.501465,
- (float)0.979492, (float)1.416992, (float)1.624268,
- (float)1.276001, (float)1.661011, (float)2.007935,
- (float)0.993042, (float)1.168579, (float)1.331665,
- (float)0.778198, (float)0.944946, (float)1.235962,
- (float)1.223755, (float)1.491333, (float)1.815674,
- (float)0.852661, (float)1.350464, (float)1.722290,
- (float)1.134766, (float)1.593140, (float)1.787354,
- (float)1.051392, (float)1.339722, (float)1.531006,
- (float)0.803589, (float)1.271240, (float)1.652100,
- (float)0.755737, (float)1.143555, (float)1.639404,
- (float)0.700928, (float)0.837280, (float)1.130371,
- (float)0.942749, (float)1.197876, (float)1.669800,
- (float)0.993286, (float)1.378296, (float)1.566528,
- (float)0.801025, (float)1.095337, (float)1.298950,
- (float)0.739990, (float)1.032959, (float)1.383667,
- (float)0.845703, (float)1.072266, (float)1.543823,
- (float)0.915649, (float)1.072266, (float)1.224487,
- (float)1.021973, (float)1.226196, (float)1.481323,
- (float)0.999878, (float)1.204102, (float)1.555908,
- (float)0.722290, (float)0.913940, (float)1.340210,
- (float)0.673340, (float)0.835938, (float)1.259521,
- (float)0.832397, (float)1.208374, (float)1.394165,
- (float)0.962158, (float)1.576172, (float)1.912842,
- (float)1.166748, (float)1.370850, (float)1.556763,
- (float)0.946289, (float)1.138550, (float)1.400391,
- (float)1.035034, (float)1.218262, (float)1.386475,
- (float)1.393799, (float)1.717773, (float)2.000244,
- (float)0.972656, (float)1.260986, (float)1.760620,
- (float)1.028198, (float)1.288452, (float)1.484619,
-
-
-
-Andersen, et al. Experimental [Page 92]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- (float)0.773560, (float)1.258057, (float)1.756714,
- (float)1.080322, (float)1.328003, (float)1.742676,
- (float)0.823975, (float)1.450806, (float)1.917725,
- (float)0.859009, (float)1.016602, (float)1.191895,
- (float)0.843994, (float)1.131104, (float)1.645020,
- (float)1.189697, (float)1.702759, (float)1.894409,
- (float)1.346680, (float)1.763184, (float)2.066040,
- (float)0.980469, (float)1.253784, (float)1.441650,
- (float)1.338135, (float)1.641968, (float)1.932739,
- (float)1.223267, (float)1.424194, (float)1.626465,
- (float)0.765747, (float)1.004150, (float)1.579102,
- (float)1.042847, (float)1.269165, (float)1.647461,
- (float)0.968750, (float)1.257568, (float)1.555786,
- (float)0.826294, (float)0.993408, (float)1.275146,
- (float)0.742310, (float)0.950439, (float)1.430542,
- (float)1.054321, (float)1.439819, (float)1.828003,
- (float)1.072998, (float)1.261719, (float)1.441895,
- (float)0.859375, (float)1.036377, (float)1.314819,
- (float)0.895752, (float)1.267212, (float)1.605591,
- (float)0.805420, (float)0.962891, (float)1.142334,
- (float)0.795654, (float)1.005493, (float)1.468506,
- (float)1.105347, (float)1.313843, (float)1.584839,
- (float)0.792236, (float)1.221802, (float)1.465698,
- (float)1.170532, (float)1.467651, (float)1.664063,
- (float)0.838257, (float)1.153198, (float)1.342163,
- (float)0.968018, (float)1.198242, (float)1.391235,
- (float)1.250122, (float)1.623535, (float)1.823608,
- (float)0.711670, (float)1.058350, (float)1.512085,
- (float)1.204834, (float)1.454468, (float)1.739136,
- (float)1.137451, (float)1.421753, (float)1.620117,
- (float)0.820435, (float)1.322754, (float)1.578247,
- (float)0.798706, (float)1.005005, (float)1.213867,
- (float)0.980713, (float)1.324951, (float)1.512939,
- (float)1.112305, (float)1.438843, (float)1.735596,
- (float)1.135498, (float)1.356689, (float)1.635742,
- (float)1.101318, (float)1.387451, (float)1.686523,
- (float)0.849854, (float)1.276978, (float)1.523438,
- (float)1.377930, (float)1.627563, (float)1.858154,
- (float)0.884888, (float)1.095459, (float)1.287476,
- (float)1.289795, (float)1.505859, (float)1.756592,
- (float)0.817505, (float)1.384155, (float)1.650513,
- (float)1.446655, (float)1.702148, (float)1.931885,
- (float)0.835815, (float)1.023071, (float)1.385376,
- (float)0.916626, (float)1.139038, (float)1.335327,
- (float)0.980103, (float)1.174072, (float)1.453735,
- (float)1.705688, (float)2.153809, (float)2.398315, (float)2.743408,
- (float)1.797119, (float)2.016846, (float)2.445679, (float)2.701904,
- (float)1.990356, (float)2.219116, (float)2.576416, (float)2.813477,
-
-
-
-Andersen, et al. Experimental [Page 93]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- (float)1.849365, (float)2.190918, (float)2.611572, (float)2.835083,
- (float)1.657959, (float)1.854370, (float)2.159058, (float)2.726196,
- (float)1.437744, (float)1.897705, (float)2.253174, (float)2.655396,
- (float)2.028687, (float)2.247314, (float)2.542358, (float)2.875854,
- (float)1.736938, (float)1.922119, (float)2.185913, (float)2.743408,
- (float)1.521606, (float)1.870972, (float)2.526855, (float)2.786987,
- (float)1.841431, (float)2.050659, (float)2.463623, (float)2.857666,
- (float)1.590088, (float)2.067261, (float)2.427979, (float)2.794434,
- (float)1.746826, (float)2.057373, (float)2.320190, (float)2.800781,
- (float)1.734619, (float)1.940552, (float)2.306030, (float)2.826416,
- (float)1.786255, (float)2.204468, (float)2.457520, (float)2.795288,
- (float)1.861084, (float)2.170532, (float)2.414551, (float)2.763672,
- (float)2.001465, (float)2.307617, (float)2.552734, (float)2.811890,
- (float)1.784424, (float)2.124146, (float)2.381592, (float)2.645508,
- (float)1.888794, (float)2.135864, (float)2.418579, (float)2.861206,
- (float)2.301147, (float)2.531250, (float)2.724976, (float)2.913086,
- (float)1.837769, (float)2.051270, (float)2.261963, (float)2.553223,
- (float)2.012939, (float)2.221191, (float)2.440186, (float)2.678101,
- (float)1.429565, (float)1.858276, (float)2.582275, (float)2.845703,
- (float)1.622803, (float)1.897705, (float)2.367310, (float)2.621094,
- (float)1.581543, (float)1.960449, (float)2.515869, (float)2.736450,
- (float)1.419434, (float)1.933960, (float)2.394653, (float)2.746704,
- (float)1.721924, (float)2.059570, (float)2.421753, (float)2.769653,
- (float)1.911011, (float)2.220703, (float)2.461060, (float)2.740723,
- (float)1.581177, (float)1.860840, (float)2.516968, (float)2.874634,
- (float)1.870361, (float)2.098755, (float)2.432373, (float)2.656494,
- (float)2.059692, (float)2.279785, (float)2.495605, (float)2.729370,
- (float)1.815674, (float)2.181519, (float)2.451538, (float)2.680542,
- (float)1.407959, (float)1.768311, (float)2.343018, (float)2.668091,
- (float)2.168701, (float)2.394653, (float)2.604736, (float)2.829346,
- (float)1.636230, (float)1.865723, (float)2.329102, (float)2.824219,
- (float)1.878906, (float)2.139526, (float)2.376709, (float)2.679810,
- (float)1.765381, (float)1.971802, (float)2.195435, (float)2.586914,
- (float)2.164795, (float)2.410889, (float)2.673706, (float)2.903198,
- (float)2.071899, (float)2.331055, (float)2.645874, (float)2.907104,
- (float)2.026001, (float)2.311523, (float)2.594849, (float)2.863892,
- (float)1.948975, (float)2.180786, (float)2.514893, (float)2.797852,
- (float)1.881836, (float)2.130859, (float)2.478149, (float)2.804199,
- (float)2.238159, (float)2.452759, (float)2.652832, (float)2.868286,
- (float)1.897949, (float)2.101685, (float)2.524292, (float)2.880127,
- (float)1.856445, (float)2.074585, (float)2.541016, (float)2.791748,
- (float)1.695557, (float)2.199097, (float)2.506226, (float)2.742676,
- (float)1.612671, (float)1.877075, (float)2.435425, (float)2.732910,
- (float)1.568848, (float)1.786499, (float)2.194580, (float)2.768555,
- (float)1.953369, (float)2.164551, (float)2.486938, (float)2.874023,
- (float)1.388306, (float)1.725342, (float)2.384521, (float)2.771851,
- (float)2.115356, (float)2.337769, (float)2.592896, (float)2.864014,
- (float)1.905762, (float)2.111328, (float)2.363525, (float)2.789307,
-
-
-
-Andersen, et al. Experimental [Page 94]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- (float)1.882568, (float)2.332031, (float)2.598267, (float)2.827637,
- (float)1.683594, (float)2.088745, (float)2.361938, (float)2.608643,
- (float)1.874023, (float)2.182129, (float)2.536133, (float)2.766968,
- (float)1.861938, (float)2.070435, (float)2.309692, (float)2.700562,
- (float)1.722168, (float)2.107422, (float)2.477295, (float)2.837646,
- (float)1.926880, (float)2.184692, (float)2.442627, (float)2.663818,
- (float)2.123901, (float)2.337280, (float)2.553101, (float)2.777466,
- (float)1.588135, (float)1.911499, (float)2.212769, (float)2.543945,
- (float)2.053955, (float)2.370850, (float)2.712158, (float)2.939941,
- (float)2.210449, (float)2.519653, (float)2.770386, (float)2.958618,
- (float)2.199463, (float)2.474731, (float)2.718262, (float)2.919922,
- (float)1.960083, (float)2.175415, (float)2.608032, (float)2.888794,
- (float)1.953735, (float)2.185181, (float)2.428223, (float)2.809570,
- (float)1.615234, (float)2.036499, (float)2.576538, (float)2.834595,
- (float)1.621094, (float)2.028198, (float)2.431030, (float)2.664673,
- (float)1.824951, (float)2.267456, (float)2.514526, (float)2.747925,
- (float)1.994263, (float)2.229126, (float)2.475220, (float)2.833984,
- (float)1.746338, (float)2.011353, (float)2.588257, (float)2.826904,
- (float)1.562866, (float)2.135986, (float)2.471680, (float)2.687256,
- (float)1.748901, (float)2.083496, (float)2.460938, (float)2.686279,
- (float)1.758057, (float)2.131470, (float)2.636597, (float)2.891602,
- (float)2.071289, (float)2.299072, (float)2.550781, (float)2.814331,
- (float)1.839600, (float)2.094360, (float)2.496460, (float)2.723999,
- (float)1.882202, (float)2.088257, (float)2.636841, (float)2.923096,
- (float)1.957886, (float)2.153198, (float)2.384399, (float)2.615234,
- (float)1.992920, (float)2.351196, (float)2.654419, (float)2.889771,
- (float)2.012817, (float)2.262451, (float)2.643799, (float)2.903076,
- (float)2.025635, (float)2.254761, (float)2.508423, (float)2.784058,
- (float)2.316040, (float)2.589355, (float)2.794189, (float)2.963623,
- (float)1.741211, (float)2.279541, (float)2.578491, (float)2.816284,
- (float)1.845337, (float)2.055786, (float)2.348511, (float)2.822021,
- (float)1.679932, (float)1.926514, (float)2.499756, (float)2.835693,
- (float)1.722534, (float)1.946899, (float)2.448486, (float)2.728760,
- (float)1.829834, (float)2.043213, (float)2.580444, (float)2.867676,
- (float)1.676636, (float)2.071655, (float)2.322510, (float)2.704834,
- (float)1.791504, (float)2.113525, (float)2.469727, (float)2.784058,
- (float)1.977051, (float)2.215088, (float)2.497437, (float)2.726929,
- (float)1.800171, (float)2.106689, (float)2.357788, (float)2.738892,
- (float)1.827759, (float)2.170166, (float)2.525879, (float)2.852417,
- (float)1.918335, (float)2.132813, (float)2.488403, (float)2.728149,
- (float)1.916748, (float)2.225098, (float)2.542603, (float)2.857666,
- (float)1.761230, (float)1.976074, (float)2.507446, (float)2.884521,
- (float)2.053711, (float)2.367432, (float)2.608032, (float)2.837646,
- (float)1.595337, (float)2.000977, (float)2.307129, (float)2.578247,
- (float)1.470581, (float)2.031250, (float)2.375854, (float)2.647583,
- (float)1.801392, (float)2.128052, (float)2.399780, (float)2.822876,
- (float)1.853638, (float)2.066650, (float)2.429199, (float)2.751465,
- (float)1.956299, (float)2.163696, (float)2.394775, (float)2.734253,
-
-
-
-Andersen, et al. Experimental [Page 95]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- (float)1.963623, (float)2.275757, (float)2.585327, (float)2.865234,
- (float)1.887451, (float)2.105469, (float)2.331787, (float)2.587402,
- (float)2.120117, (float)2.443359, (float)2.733887, (float)2.941406,
- (float)1.506348, (float)1.766968, (float)2.400513, (float)2.851807,
- (float)1.664551, (float)1.981079, (float)2.375732, (float)2.774414,
- (float)1.720703, (float)1.978882, (float)2.391479, (float)2.640991,
- (float)1.483398, (float)1.814819, (float)2.434448, (float)2.722290,
- (float)1.769043, (float)2.136597, (float)2.563721, (float)2.774414,
- (float)1.810791, (float)2.049316, (float)2.373901, (float)2.613647,
- (float)1.788330, (float)2.005981, (float)2.359131, (float)2.723145,
- (float)1.785156, (float)1.993164, (float)2.399780, (float)2.832520,
- (float)1.695313, (float)2.022949, (float)2.522583, (float)2.745117,
- (float)1.584106, (float)1.965576, (float)2.299927, (float)2.715576,
- (float)1.894897, (float)2.249878, (float)2.655884, (float)2.897705,
- (float)1.720581, (float)1.995728, (float)2.299438, (float)2.557007,
- (float)1.619385, (float)2.173950, (float)2.574219, (float)2.787964,
- (float)1.883179, (float)2.220459, (float)2.474365, (float)2.825073,
- (float)1.447632, (float)2.045044, (float)2.555542, (float)2.744873,
- (float)1.502686, (float)2.156616, (float)2.653320, (float)2.846558,
- (float)1.711548, (float)1.944092, (float)2.282959, (float)2.685791,
- (float)1.499756, (float)1.867554, (float)2.341064, (float)2.578857,
- (float)1.916870, (float)2.135132, (float)2.568237, (float)2.826050,
- (float)1.498047, (float)1.711182, (float)2.223267, (float)2.755127,
- (float)1.808716, (float)1.997559, (float)2.256470, (float)2.758545,
- (float)2.088501, (float)2.402710, (float)2.667358, (float)2.890259,
- (float)1.545044, (float)1.819214, (float)2.324097, (float)2.692993,
- (float)1.796021, (float)2.012573, (float)2.505737, (float)2.784912,
- (float)1.786499, (float)2.041748, (float)2.290405, (float)2.650757,
- (float)1.938232, (float)2.264404, (float)2.529053, (float)2.796143
- };
-
-A.9. anaFilter.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- anaFilter.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_ANAFILTER_H
- #define __iLBC_ANAFILTER_H
-
- void anaFilter(
-
-
-
-Andersen, et al. Experimental [Page 96]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- float *In, /* (i) Signal to be filtered */
- float *a, /* (i) LP parameters */
- int len,/* (i) Length of signal */
- float *Out, /* (o) Filtered signal */
- float *mem /* (i/o) Filter state */
- );
-
- #endif
-
-A.10. anaFilter.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- anaFilter.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include <string.h>
- #include "iLBC_define.h"
-
- /*----------------------------------------------------------------*
- * LP analysis filter.
- *---------------------------------------------------------------*/
-
- void anaFilter(
- float *In, /* (i) Signal to be filtered */
- float *a, /* (i) LP parameters */
- int len,/* (i) Length of signal */
- float *Out, /* (o) Filtered signal */
- float *mem /* (i/o) Filter state */
- ){
- int i, j;
- float *po, *pi, *pm, *pa;
-
- po = Out;
-
- /* Filter first part using memory from past */
-
- for (i=0; i<LPC_FILTERORDER; i++) {
- pi = &In[i];
- pm = &mem[LPC_FILTERORDER-1];
- pa = a;
- *po=0.0;
-
-
-
-Andersen, et al. Experimental [Page 97]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- for (j=0; j<=i; j++) {
- *po+=(*pa++)*(*pi--);
- }
- for (j=i+1; j<LPC_FILTERORDER+1; j++) {
-
- *po+=(*pa++)*(*pm--);
- }
- po++;
- }
-
- /* Filter last part where the state is entirely
- in the input vector */
-
- for (i=LPC_FILTERORDER; i<len; i++) {
- pi = &In[i];
- pa = a;
- *po=0.0;
- for (j=0; j<LPC_FILTERORDER+1; j++) {
- *po+=(*pa++)*(*pi--);
- }
- po++;
- }
-
- /* Update state vector */
-
- memcpy(mem, &In[len-LPC_FILTERORDER],
- LPC_FILTERORDER*sizeof(float));
- }
-
-A.11. createCB.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- createCB.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_CREATECB_H
- #define __iLBC_CREATECB_H
-
- void filteredCBvecs(
- float *cbvectors, /* (o) Codebook vector for the
- higher section */
-
-
-
-Andersen, et al. Experimental [Page 98]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- float *mem, /* (i) Buffer to create codebook
- vectors from */
- int lMem /* (i) Length of buffer */
- );
-
- void searchAugmentedCB(
- int low, /* (i) Start index for the search */
- int high, /* (i) End index for the search */
- int stage, /* (i) Current stage */
- int startIndex, /* (i) CB index for the first
- augmented vector */
- float *target, /* (i) Target vector for encoding */
- float *buffer, /* (i) Pointer to the end of the
- buffer for augmented codebook
- construction */
- float *max_measure, /* (i/o) Currently maximum measure */
- int *best_index,/* (o) Currently the best index */
- float *gain, /* (o) Currently the best gain */
- float *energy, /* (o) Energy of augmented
- codebook vectors */
- float *invenergy/* (o) Inv energy of aug codebook
- vectors */
- );
-
- void createAugmentedVec(
- int index, /* (i) Index for the aug vector
- to be created */
- float *buffer, /* (i) Pointer to the end of the
- buffer for augmented codebook
- construction */
- float *cbVec /* (o) The construced codebook vector */
- );
-
- #endif
-
-A.12. createCB.c
-
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- createCB.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
-
-
-Andersen, et al. Experimental [Page 99]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- #include "iLBC_define.h"
- #include "constants.h"
- #include <string.h>
- #include <math.h>
-
- /*----------------------------------------------------------------*
- * Construct an additional codebook vector by filtering the
- * initial codebook buffer. This vector is then used to expand
- * the codebook with an additional section.
- *---------------------------------------------------------------*/
-
- void filteredCBvecs(
- float *cbvectors, /* (o) Codebook vectors for the
- higher section */
- float *mem, /* (i) Buffer to create codebook
- vector from */
- int lMem /* (i) Length of buffer */
- ){
- int j, k;
- float *pp, *pp1;
- float tempbuff2[CB_MEML+CB_FILTERLEN];
- float *pos;
-
- memset(tempbuff2, 0, (CB_HALFFILTERLEN-1)*sizeof(float));
- memcpy(&tempbuff2[CB_HALFFILTERLEN-1], mem, lMem*sizeof(float));
- memset(&tempbuff2[lMem+CB_HALFFILTERLEN-1], 0,
- (CB_HALFFILTERLEN+1)*sizeof(float));
-
- /* Create codebook vector for higher section by filtering */
-
- /* do filtering */
- pos=cbvectors;
- memset(pos, 0, lMem*sizeof(float));
- for (k=0; k<lMem; k++) {
- pp=&tempbuff2[k];
- pp1=&cbfiltersTbl[CB_FILTERLEN-1];
- for (j=0;j<CB_FILTERLEN;j++) {
- (*pos)+=(*pp++)*(*pp1--);
- }
- pos++;
- }
- }
-
- /*----------------------------------------------------------------*
- * Search the augmented part of the codebook to find the best
- * measure.
- *----------------------------------------------------------------*/
-
-
-
-
-Andersen, et al. Experimental [Page 100]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- void searchAugmentedCB(
- int low, /* (i) Start index for the search */
- int high, /* (i) End index for the search */
- int stage, /* (i) Current stage */
- int startIndex, /* (i) Codebook index for the first
- aug vector */
- float *target, /* (i) Target vector for encoding */
- float *buffer, /* (i) Pointer to the end of the buffer for
- augmented codebook construction */
- float *max_measure, /* (i/o) Currently maximum measure */
- int *best_index,/* (o) Currently the best index */
- float *gain, /* (o) Currently the best gain */
- float *energy, /* (o) Energy of augmented codebook
- vectors */
- float *invenergy/* (o) Inv energy of augmented codebook
- vectors */
- ) {
- int icount, ilow, j, tmpIndex;
- float *pp, *ppo, *ppi, *ppe, crossDot, alfa;
- float weighted, measure, nrjRecursive;
- float ftmp;
-
- /* Compute the energy for the first (low-5)
- noninterpolated samples */
- nrjRecursive = (float) 0.0;
- pp = buffer - low + 1;
- for (j=0; j<(low-5); j++) {
- nrjRecursive += ( (*pp)*(*pp) );
- pp++;
- }
- ppe = buffer - low;
-
-
- for (icount=low; icount<=high; icount++) {
-
- /* Index of the codebook vector used for retrieving
- energy values */
- tmpIndex = startIndex+icount-20;
-
- ilow = icount-4;
-
- /* Update the energy recursively to save complexity */
- nrjRecursive = nrjRecursive + (*ppe)*(*ppe);
- ppe--;
- energy[tmpIndex] = nrjRecursive;
-
- /* Compute cross dot product for the first (low-5)
- samples */
-
-
-
-Andersen, et al. Experimental [Page 101]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- crossDot = (float) 0.0;
- pp = buffer-icount;
- for (j=0; j<ilow; j++) {
- crossDot += target[j]*(*pp++);
- }
-
- /* interpolation */
- alfa = (float) 0.2;
- ppo = buffer-4;
- ppi = buffer-icount-4;
- for (j=ilow; j<icount; j++) {
- weighted = ((float)1.0-alfa)*(*ppo)+alfa*(*ppi);
- ppo++;
- ppi++;
- energy[tmpIndex] += weighted*weighted;
- crossDot += target[j]*weighted;
- alfa += (float)0.2;
- }
-
- /* Compute energy and cross dot product for the
- remaining samples */
- pp = buffer - icount;
- for (j=icount; j<SUBL; j++) {
- energy[tmpIndex] += (*pp)*(*pp);
- crossDot += target[j]*(*pp++);
- }
-
- if (energy[tmpIndex]>0.0) {
- invenergy[tmpIndex]=(float)1.0/(energy[tmpIndex]+EPS);
- } else {
- invenergy[tmpIndex] = (float) 0.0;
- }
-
- if (stage==0) {
- measure = (float)-10000000.0;
-
- if (crossDot > 0.0) {
- measure = crossDot*crossDot*invenergy[tmpIndex];
- }
- }
- else {
- measure = crossDot*crossDot*invenergy[tmpIndex];
- }
-
- /* check if measure is better */
- ftmp = crossDot*invenergy[tmpIndex];
-
- if ((measure>*max_measure) && (fabs(ftmp)<CB_MAXGAIN)) {
-
-
-
-Andersen, et al. Experimental [Page 102]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- *best_index = tmpIndex;
- *max_measure = measure;
- *gain = ftmp;
- }
- }
- }
-
-
- /*----------------------------------------------------------------*
- * Recreate a specific codebook vector from the augmented part.
- *
- *----------------------------------------------------------------*/
-
- void createAugmentedVec(
- int index, /* (i) Index for the augmented vector
- to be created */
- float *buffer, /* (i) Pointer to the end of the buffer for
- augmented codebook construction */
- float *cbVec/* (o) The construced codebook vector */
- ) {
- int ilow, j;
- float *pp, *ppo, *ppi, alfa, alfa1, weighted;
-
- ilow = index-5;
-
- /* copy the first noninterpolated part */
-
- pp = buffer-index;
- memcpy(cbVec,pp,sizeof(float)*index);
-
- /* interpolation */
-
- alfa1 = (float)0.2;
- alfa = 0.0;
- ppo = buffer-5;
- ppi = buffer-index-5;
- for (j=ilow; j<index; j++) {
- weighted = ((float)1.0-alfa)*(*ppo)+alfa*(*ppi);
- ppo++;
- ppi++;
- cbVec[j] = weighted;
- alfa += alfa1;
- }
-
- /* copy the second noninterpolated part */
-
- pp = buffer - index;
- memcpy(cbVec+index,pp,sizeof(float)*(SUBL-index));
-
-
-
-Andersen, et al. Experimental [Page 103]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- }
-
-A.13. doCPLC.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- doCPLC.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_DOLPC_H
- #define __iLBC_DOLPC_H
-
- void doThePLC(
- float *PLCresidual, /* (o) concealed residual */
- float *PLClpc, /* (o) concealed LP parameters */
- int PLI, /* (i) packet loss indicator
- 0 - no PL, 1 = PL */
- float *decresidual, /* (i) decoded residual */
- float *lpc, /* (i) decoded LPC (only used for no PL) */
- int inlag, /* (i) pitch lag */
- iLBC_Dec_Inst_t *iLBCdec_inst
- /* (i/o) decoder instance */
- );
-
- #endif
-
-A.14. doCPLC.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- doCPLC.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include <math.h>
- #include <string.h>
- #include <stdio.h>
-
-
-
-Andersen, et al. Experimental [Page 104]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- #include "iLBC_define.h"
-
- /*----------------------------------------------------------------*
- * Compute cross correlation and pitch gain for pitch prediction
- * of last subframe at given lag.
- *---------------------------------------------------------------*/
-
- void compCorr(
- float *cc, /* (o) cross correlation coefficient */
- float *gc, /* (o) gain */
- float *pm,
- float *buffer, /* (i) signal buffer */
- int lag, /* (i) pitch lag */
- int bLen, /* (i) length of buffer */
- int sRange /* (i) correlation search length */
- ){
- int i;
- float ftmp1, ftmp2, ftmp3;
-
- /* Guard against getting outside buffer */
- if ((bLen-sRange-lag)<0) {
- sRange=bLen-lag;
- }
-
- ftmp1 = 0.0;
- ftmp2 = 0.0;
- ftmp3 = 0.0;
- for (i=0; i<sRange; i++) {
- ftmp1 += buffer[bLen-sRange+i] *
- buffer[bLen-sRange+i-lag];
- ftmp2 += buffer[bLen-sRange+i-lag] *
- buffer[bLen-sRange+i-lag];
- ftmp3 += buffer[bLen-sRange+i] *
- buffer[bLen-sRange+i];
- }
-
- if (ftmp2 > 0.0) {
- *cc = ftmp1*ftmp1/ftmp2;
- *gc = (float)fabs(ftmp1/ftmp2);
- *pm=(float)fabs(ftmp1)/
- ((float)sqrt(ftmp2)*(float)sqrt(ftmp3));
- }
- else {
- *cc = 0.0;
- *gc = 0.0;
- *pm=0.0;
- }
- }
-
-
-
-Andersen, et al. Experimental [Page 105]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- /*----------------------------------------------------------------*
- * Packet loss concealment routine. Conceals a residual signal
- * and LP parameters. If no packet loss, update state.
- *---------------------------------------------------------------*/
-
- void doThePLC(
- float *PLCresidual, /* (o) concealed residual */
- float *PLClpc, /* (o) concealed LP parameters */
- int PLI, /* (i) packet loss indicator
- 0 - no PL, 1 = PL */
- float *decresidual, /* (i) decoded residual */
- float *lpc, /* (i) decoded LPC (only used for no PL) */
- int inlag, /* (i) pitch lag */
- iLBC_Dec_Inst_t *iLBCdec_inst
- /* (i/o) decoder instance */
- ){
- int lag=20, randlag;
- float gain, maxcc;
- float use_gain;
- float gain_comp, maxcc_comp, per, max_per;
- int i, pick, use_lag;
- float ftmp, randvec[BLOCKL_MAX], pitchfact, energy;
-
- /* Packet Loss */
-
- if (PLI == 1) {
-
- iLBCdec_inst->consPLICount += 1;
-
- /* if previous frame not lost,
- determine pitch pred. gain */
-
- if (iLBCdec_inst->prevPLI != 1) {
-
- /* Search around the previous lag to find the
- best pitch period */
-
- lag=inlag-3;
- compCorr(&maxcc, &gain, &max_per,
- iLBCdec_inst->prevResidual,
- lag, iLBCdec_inst->blockl, 60);
- for (i=inlag-2;i<=inlag+3;i++) {
- compCorr(&maxcc_comp, &gain_comp, &per,
- iLBCdec_inst->prevResidual,
- i, iLBCdec_inst->blockl, 60);
-
- if (maxcc_comp>maxcc) {
- maxcc=maxcc_comp;
-
-
-
-Andersen, et al. Experimental [Page 106]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- gain=gain_comp;
- lag=i;
- max_per=per;
- }
- }
-
- }
-
- /* previous frame lost, use recorded lag and periodicity */
-
- else {
- lag=iLBCdec_inst->prevLag;
- max_per=iLBCdec_inst->per;
- }
-
- /* downscaling */
-
- use_gain=1.0;
- if (iLBCdec_inst->consPLICount*iLBCdec_inst->blockl>320)
- use_gain=(float)0.9;
- else if (iLBCdec_inst->consPLICount*
- iLBCdec_inst->blockl>2*320)
- use_gain=(float)0.7;
- else if (iLBCdec_inst->consPLICount*
- iLBCdec_inst->blockl>3*320)
- use_gain=(float)0.5;
- else if (iLBCdec_inst->consPLICount*
- iLBCdec_inst->blockl>4*320)
- use_gain=(float)0.0;
-
- /* mix noise and pitch repeatition */
- ftmp=(float)sqrt(max_per);
- if (ftmp>(float)0.7)
- pitchfact=(float)1.0;
- else if (ftmp>(float)0.4)
- pitchfact=(ftmp-(float)0.4)/((float)0.7-(float)0.4);
- else
- pitchfact=0.0;
-
-
- /* avoid repetition of same pitch cycle */
- use_lag=lag;
- if (lag<80) {
- use_lag=2*lag;
- }
-
- /* compute concealed residual */
-
-
-
-
-Andersen, et al. Experimental [Page 107]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- energy = 0.0;
- for (i=0; i<iLBCdec_inst->blockl; i++) {
-
- /* noise component */
-
- iLBCdec_inst->seed=(iLBCdec_inst->seed*69069L+1) &
- (0x80000000L-1);
- randlag = 50 + ((signed long) iLBCdec_inst->seed)%70;
- pick = i - randlag;
-
- if (pick < 0) {
- randvec[i] =
- iLBCdec_inst->prevResidual[
- iLBCdec_inst->blockl+pick];
- } else {
- randvec[i] = randvec[pick];
- }
-
- /* pitch repeatition component */
- pick = i - use_lag;
-
- if (pick < 0) {
- PLCresidual[i] =
- iLBCdec_inst->prevResidual[
- iLBCdec_inst->blockl+pick];
- } else {
- PLCresidual[i] = PLCresidual[pick];
- }
-
- /* mix random and periodicity component */
-
- if (i<80)
- PLCresidual[i] = use_gain*(pitchfact *
- PLCresidual[i] +
- ((float)1.0 - pitchfact) * randvec[i]);
- else if (i<160)
- PLCresidual[i] = (float)0.95*use_gain*(pitchfact *
- PLCresidual[i] +
- ((float)1.0 - pitchfact) * randvec[i]);
- else
- PLCresidual[i] = (float)0.9*use_gain*(pitchfact *
- PLCresidual[i] +
- ((float)1.0 - pitchfact) * randvec[i]);
-
- energy += PLCresidual[i] * PLCresidual[i];
- }
-
- /* less than 30 dB, use only noise */
-
-
-
-Andersen, et al. Experimental [Page 108]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
-
- if (sqrt(energy/(float)iLBCdec_inst->blockl) < 30.0) {
- gain=0.0;
- for (i=0; i<iLBCdec_inst->blockl; i++) {
- PLCresidual[i] = randvec[i];
- }
- }
-
- /* use old LPC */
-
- memcpy(PLClpc,iLBCdec_inst->prevLpc,
- (LPC_FILTERORDER+1)*sizeof(float));
-
- }
-
- /* no packet loss, copy input */
-
- else {
- memcpy(PLCresidual, decresidual,
- iLBCdec_inst->blockl*sizeof(float));
- memcpy(PLClpc, lpc, (LPC_FILTERORDER+1)*sizeof(float));
- iLBCdec_inst->consPLICount = 0;
- }
-
- /* update state */
-
- if (PLI) {
- iLBCdec_inst->prevLag = lag;
- iLBCdec_inst->per=max_per;
- }
-
- iLBCdec_inst->prevPLI = PLI;
- memcpy(iLBCdec_inst->prevLpc, PLClpc,
- (LPC_FILTERORDER+1)*sizeof(float));
- memcpy(iLBCdec_inst->prevResidual, PLCresidual,
- iLBCdec_inst->blockl*sizeof(float));
- }
-
-A.15. enhancer.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- enhancer.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
-
-
-Andersen, et al. Experimental [Page 109]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- ******************************************************************/
-
- #ifndef __ENHANCER_H
- #define __ENHANCER_H
-
- #include "iLBC_define.h"
-
- float xCorrCoef(
- float *target, /* (i) first array */
- float *regressor, /* (i) second array */
- int subl /* (i) dimension arrays */
- );
-
- int enhancerInterface(
- float *out, /* (o) the enhanced recidual signal */
- float *in, /* (i) the recidual signal to enhance */
- iLBC_Dec_Inst_t *iLBCdec_inst
- /* (i/o) the decoder state structure */
- );
-
- #endif
-
-A.16. enhancer.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- enhancer.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include <math.h>
- #include <string.h>
- #include "iLBC_define.h"
- #include "constants.h"
- #include "filter.h"
-
- /*----------------------------------------------------------------*
- * Find index in array such that the array element with said
- * index is the element of said array closest to "value"
- * according to the squared-error criterion
- *---------------------------------------------------------------*/
-
- void NearestNeighbor(
-
-
-
-Andersen, et al. Experimental [Page 110]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- int *index, /* (o) index of array element closest
- to value */
- float *array, /* (i) data array */
- float value,/* (i) value */
- int arlength/* (i) dimension of data array */
- ){
- int i;
- float bestcrit,crit;
-
- crit=array[0]-value;
- bestcrit=crit*crit;
- *index=0;
- for (i=1; i<arlength; i++) {
- crit=array[i]-value;
- crit=crit*crit;
-
- if (crit<bestcrit) {
- bestcrit=crit;
- *index=i;
- }
- }
- }
-
- /*----------------------------------------------------------------*
- * compute cross correlation between sequences
- *---------------------------------------------------------------*/
-
- void mycorr1(
- float* corr, /* (o) correlation of seq1 and seq2 */
- float* seq1, /* (i) first sequence */
- int dim1, /* (i) dimension first seq1 */
- const float *seq2, /* (i) second sequence */
- int dim2 /* (i) dimension seq2 */
- ){
- int i,j;
-
- for (i=0; i<=dim1-dim2; i++) {
- corr[i]=0.0;
- for (j=0; j<dim2; j++) {
- corr[i] += seq1[i+j] * seq2[j];
- }
- }
- }
-
- /*----------------------------------------------------------------*
- * upsample finite array assuming zeros outside bounds
- *---------------------------------------------------------------*/
-
-
-
-
-Andersen, et al. Experimental [Page 111]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- void enh_upsample(
- float* useq1, /* (o) upsampled output sequence */
- float* seq1,/* (i) unupsampled sequence */
- int dim1, /* (i) dimension seq1 */
- int hfl /* (i) polyphase filter length=2*hfl+1 */
- ){
- float *pu,*ps;
- int i,j,k,q,filterlength,hfl2;
- const float *polyp[ENH_UPS0]; /* pointers to
- polyphase columns */
- const float *pp;
-
- /* define pointers for filter */
-
- filterlength=2*hfl+1;
-
- if ( filterlength > dim1 ) {
- hfl2=(int) (dim1/2);
- for (j=0; j<ENH_UPS0; j++) {
- polyp[j]=polyphaserTbl+j*filterlength+hfl-hfl2;
- }
- hfl=hfl2;
- filterlength=2*hfl+1;
- }
- else {
- for (j=0; j<ENH_UPS0; j++) {
- polyp[j]=polyphaserTbl+j*filterlength;
- }
- }
-
- /* filtering: filter overhangs left side of sequence */
-
- pu=useq1;
- for (i=hfl; i<filterlength; i++) {
- for (j=0; j<ENH_UPS0; j++) {
- *pu=0.0;
- pp = polyp[j];
- ps = seq1+i;
- for (k=0; k<=i; k++) {
- *pu += *ps-- * *pp++;
- }
- pu++;
- }
- }
-
- /* filtering: simple convolution=inner products */
-
- for (i=filterlength; i<dim1; i++) {
-
-
-
-Andersen, et al. Experimental [Page 112]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- for (j=0;j<ENH_UPS0; j++){
- *pu=0.0;
- pp = polyp[j];
- ps = seq1+i;
- for (k=0; k<filterlength; k++) {
- *pu += *ps-- * *pp++;
- }
- pu++;
- }
- }
-
- /* filtering: filter overhangs right side of sequence */
-
- for (q=1; q<=hfl; q++) {
- for (j=0; j<ENH_UPS0; j++) {
- *pu=0.0;
- pp = polyp[j]+q;
- ps = seq1+dim1-1;
- for (k=0; k<filterlength-q; k++) {
- *pu += *ps-- * *pp++;
- }
- pu++;
- }
- }
- }
-
-
- /*----------------------------------------------------------------*
- * find segment starting near idata+estSegPos that has highest
- * correlation with idata+centerStartPos through
- * idata+centerStartPos+ENH_BLOCKL-1 segment is found at a
- * resolution of ENH_UPSO times the original of the original
- * sampling rate
- *---------------------------------------------------------------*/
-
- void refiner(
- float *seg, /* (o) segment array */
- float *updStartPos, /* (o) updated start point */
- float* idata, /* (i) original data buffer */
- int idatal, /* (i) dimension of idata */
- int centerStartPos, /* (i) beginning center segment */
- float estSegPos,/* (i) estimated beginning other segment */
- float period /* (i) estimated pitch period */
- ){
- int estSegPosRounded,searchSegStartPos,searchSegEndPos,corrdim;
- int tloc,tloc2,i,st,en,fraction;
- float vect[ENH_VECTL],corrVec[ENH_CORRDIM],maxv;
- float corrVecUps[ENH_CORRDIM*ENH_UPS0];
-
-
-
-Andersen, et al. Experimental [Page 113]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- /* defining array bounds */
-
- estSegPosRounded=(int)(estSegPos - 0.5);
-
- searchSegStartPos=estSegPosRounded-ENH_SLOP;
-
- if (searchSegStartPos<0) {
- searchSegStartPos=0;
- }
- searchSegEndPos=estSegPosRounded+ENH_SLOP;
-
- if (searchSegEndPos+ENH_BLOCKL >= idatal) {
- searchSegEndPos=idatal-ENH_BLOCKL-1;
- }
- corrdim=searchSegEndPos-searchSegStartPos+1;
-
- /* compute upsampled correlation (corr33) and find
- location of max */
-
- mycorr1(corrVec,idata+searchSegStartPos,
- corrdim+ENH_BLOCKL-1,idata+centerStartPos,ENH_BLOCKL);
- enh_upsample(corrVecUps,corrVec,corrdim,ENH_FL0);
- tloc=0; maxv=corrVecUps[0];
- for (i=1; i<ENH_UPS0*corrdim; i++) {
-
- if (corrVecUps[i]>maxv) {
- tloc=i;
- maxv=corrVecUps[i];
- }
- }
-
- /* make vector can be upsampled without ever running outside
- bounds */
-
- *updStartPos= (float)searchSegStartPos +
- (float)tloc/(float)ENH_UPS0+(float)1.0;
- tloc2=(int)(tloc/ENH_UPS0);
-
- if (tloc>tloc2*ENH_UPS0) {
- tloc2++;
- }
- st=searchSegStartPos+tloc2-ENH_FL0;
-
- if (st<0) {
- memset(vect,0,-st*sizeof(float));
- memcpy(&vect[-st],idata, (ENH_VECTL+st)*sizeof(float));
- }
- else {
-
-
-
-Andersen, et al. Experimental [Page 114]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- en=st+ENH_VECTL;
-
- if (en>idatal) {
- memcpy(vect, &idata[st],
- (ENH_VECTL-(en-idatal))*sizeof(float));
- memset(&vect[ENH_VECTL-(en-idatal)], 0,
- (en-idatal)*sizeof(float));
- }
- else {
- memcpy(vect, &idata[st], ENH_VECTL*sizeof(float));
- }
- }
- fraction=tloc2*ENH_UPS0-tloc;
-
- /* compute the segment (this is actually a convolution) */
-
- mycorr1(seg,vect,ENH_VECTL,polyphaserTbl+(2*ENH_FL0+1)*fraction,
- 2*ENH_FL0+1);
- }
-
- /*----------------------------------------------------------------*
- * find the smoothed output data
- *---------------------------------------------------------------*/
-
- void smath(
- float *odata, /* (o) smoothed output */
- float *sseq,/* (i) said second sequence of waveforms */
- int hl, /* (i) 2*hl+1 is sseq dimension */
- float alpha0/* (i) max smoothing energy fraction */
- ){
- int i,k;
- float w00,w10,w11,A,B,C,*psseq,err,errs;
- float surround[BLOCKL_MAX]; /* shape contributed by other than
- current */
- float wt[2*ENH_HL+1]; /* waveform weighting to get
- surround shape */
- float denom;
-
- /* create shape of contribution from all waveforms except the
- current one */
-
- for (i=1; i<=2*hl+1; i++) {
- wt[i-1] = (float)0.5*(1 - (float)cos(2*PI*i/(2*hl+2)));
- }
- wt[hl]=0.0; /* for clarity, not used */
- for (i=0; i<ENH_BLOCKL; i++) {
- surround[i]=sseq[i]*wt[0];
- }
-
-
-
-Andersen, et al. Experimental [Page 115]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- for (k=1; k<hl; k++) {
- psseq=sseq+k*ENH_BLOCKL;
- for(i=0;i<ENH_BLOCKL; i++) {
- surround[i]+=psseq[i]*wt[k];
- }
- }
- for (k=hl+1; k<=2*hl; k++) {
- psseq=sseq+k*ENH_BLOCKL;
- for(i=0;i<ENH_BLOCKL; i++) {
- surround[i]+=psseq[i]*wt[k];
- }
- }
-
- /* compute some inner products */
-
- w00 = w10 = w11 = 0.0;
- psseq=sseq+hl*ENH_BLOCKL; /* current block */
- for (i=0; i<ENH_BLOCKL;i++) {
- w00+=psseq[i]*psseq[i];
- w11+=surround[i]*surround[i];
- w10+=surround[i]*psseq[i];
- }
-
- if (fabs(w11) < 1.0) {
- w11=1.0;
- }
- C = (float)sqrt( w00/w11);
-
- /* first try enhancement without power-constraint */
-
- errs=0.0;
- psseq=sseq+hl*ENH_BLOCKL;
- for (i=0; i<ENH_BLOCKL; i++) {
- odata[i]=C*surround[i];
- err=psseq[i]-odata[i];
- errs+=err*err;
- }
-
- /* if constraint violated by first try, add constraint */
-
- if (errs > alpha0 * w00) {
- if ( w00 < 1) {
- w00=1;
- }
- denom = (w11*w00-w10*w10)/(w00*w00);
-
- if (denom > 0.0001) { /* eliminates numerical problems
- for if smooth */
-
-
-
-Andersen, et al. Experimental [Page 116]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- A = (float)sqrt( (alpha0- alpha0*alpha0/4)/denom);
- B = -alpha0/2 - A * w10/w00;
- B = B+1;
- }
- else { /* essentially no difference between cycles;
- smoothing not needed */
- A= 0.0;
- B= 1.0;
- }
-
- /* create smoothed sequence */
-
- psseq=sseq+hl*ENH_BLOCKL;
- for (i=0; i<ENH_BLOCKL; i++) {
- odata[i]=A*surround[i]+B*psseq[i];
- }
- }
- }
-
- /*----------------------------------------------------------------*
- * get the pitch-synchronous sample sequence
- *---------------------------------------------------------------*/
-
- void getsseq(
- float *sseq, /* (o) the pitch-synchronous sequence */
- float *idata, /* (i) original data */
- int idatal, /* (i) dimension of data */
- int centerStartPos, /* (i) where current block starts */
- float *period, /* (i) rough-pitch-period array */
- float *plocs, /* (i) where periods of period array
- are taken */
- int periodl, /* (i) dimension period array */
- int hl /* (i) 2*hl+1 is the number of sequences */
- ){
- int i,centerEndPos,q;
- float blockStartPos[2*ENH_HL+1];
- int lagBlock[2*ENH_HL+1];
- float plocs2[ENH_PLOCSL];
- float *psseq;
-
- centerEndPos=centerStartPos+ENH_BLOCKL-1;
-
- /* present */
-
- NearestNeighbor(lagBlock+hl,plocs,
- (float)0.5*(centerStartPos+centerEndPos),periodl);
-
- blockStartPos[hl]=(float)centerStartPos;
-
-
-
-Andersen, et al. Experimental [Page 117]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- psseq=sseq+ENH_BLOCKL*hl;
- memcpy(psseq, idata+centerStartPos, ENH_BLOCKL*sizeof(float));
-
- /* past */
-
- for (q=hl-1; q>=0; q--) {
- blockStartPos[q]=blockStartPos[q+1]-period[lagBlock[q+1]];
- NearestNeighbor(lagBlock+q,plocs,
- blockStartPos[q]+
- ENH_BLOCKL_HALF-period[lagBlock[q+1]], periodl);
-
-
- if (blockStartPos[q]-ENH_OVERHANG>=0) {
- refiner(sseq+q*ENH_BLOCKL, blockStartPos+q, idata,
- idatal, centerStartPos, blockStartPos[q],
- period[lagBlock[q+1]]);
- } else {
- psseq=sseq+q*ENH_BLOCKL;
- memset(psseq, 0, ENH_BLOCKL*sizeof(float));
- }
- }
-
- /* future */
-
- for (i=0; i<periodl; i++) {
- plocs2[i]=plocs[i]-period[i];
- }
- for (q=hl+1; q<=2*hl; q++) {
- NearestNeighbor(lagBlock+q,plocs2,
- blockStartPos[q-1]+ENH_BLOCKL_HALF,periodl);
-
- blockStartPos[q]=blockStartPos[q-1]+period[lagBlock[q]];
- if (blockStartPos[q]+ENH_BLOCKL+ENH_OVERHANG<idatal) {
- refiner(sseq+ENH_BLOCKL*q, blockStartPos+q, idata,
- idatal, centerStartPos, blockStartPos[q],
- period[lagBlock[q]]);
- }
- else {
- psseq=sseq+q*ENH_BLOCKL;
- memset(psseq, 0, ENH_BLOCKL*sizeof(float));
- }
- }
- }
-
- /*----------------------------------------------------------------*
- * perform enhancement on idata+centerStartPos through
- * idata+centerStartPos+ENH_BLOCKL-1
- *---------------------------------------------------------------*/
-
-
-
-Andersen, et al. Experimental [Page 118]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- void enhancer(
- float *odata, /* (o) smoothed block, dimension blockl */
- float *idata, /* (i) data buffer used for enhancing */
- int idatal, /* (i) dimension idata */
- int centerStartPos, /* (i) first sample current block
- within idata */
- float alpha0, /* (i) max correction-energy-fraction
- (in [0,1]) */
- float *period, /* (i) pitch period array */
- float *plocs, /* (i) locations where period array
- values valid */
- int periodl /* (i) dimension of period and plocs */
- ){
- float sseq[(2*ENH_HL+1)*ENH_BLOCKL];
-
- /* get said second sequence of segments */
-
- getsseq(sseq,idata,idatal,centerStartPos,period,
- plocs,periodl,ENH_HL);
-
- /* compute the smoothed output from said second sequence */
-
- smath(odata,sseq,ENH_HL,alpha0);
-
- }
-
- /*----------------------------------------------------------------*
- * cross correlation
- *---------------------------------------------------------------*/
-
- float xCorrCoef(
- float *target, /* (i) first array */
- float *regressor, /* (i) second array */
- int subl /* (i) dimension arrays */
- ){
- int i;
- float ftmp1, ftmp2;
-
- ftmp1 = 0.0;
- ftmp2 = 0.0;
- for (i=0; i<subl; i++) {
- ftmp1 += target[i]*regressor[i];
- ftmp2 += regressor[i]*regressor[i];
- }
-
- if (ftmp1 > 0.0) {
- return (float)(ftmp1*ftmp1/ftmp2);
- }
-
-
-
-Andersen, et al. Experimental [Page 119]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- else {
- return (float)0.0;
- }
- }
-
- /*----------------------------------------------------------------*
- * interface for enhancer
- *---------------------------------------------------------------*/
-
- int enhancerInterface(
- float *out, /* (o) enhanced signal */
- float *in, /* (i) unenhanced signal */
- iLBC_Dec_Inst_t *iLBCdec_inst /* (i) buffers etc */
- ){
- float *enh_buf, *enh_period;
- int iblock, isample;
- int lag=0, ilag, i, ioffset;
- float cc, maxcc;
- float ftmp1, ftmp2;
- float *inPtr, *enh_bufPtr1, *enh_bufPtr2;
- float plc_pred[ENH_BLOCKL];
-
- float lpState[6], downsampled[(ENH_NBLOCKS*ENH_BLOCKL+120)/2];
- int inLen=ENH_NBLOCKS*ENH_BLOCKL+120;
- int start, plc_blockl, inlag;
-
- enh_buf=iLBCdec_inst->enh_buf;
- enh_period=iLBCdec_inst->enh_period;
-
- memmove(enh_buf, &enh_buf[iLBCdec_inst->blockl],
- (ENH_BUFL-iLBCdec_inst->blockl)*sizeof(float));
-
- memcpy(&enh_buf[ENH_BUFL-iLBCdec_inst->blockl], in,
- iLBCdec_inst->blockl*sizeof(float));
-
- if (iLBCdec_inst->mode==30)
- plc_blockl=ENH_BLOCKL;
- else
- plc_blockl=40;
-
- /* when 20 ms frame, move processing one block */
- ioffset=0;
- if (iLBCdec_inst->mode==20) ioffset=1;
-
- i=3-ioffset;
- memmove(enh_period, &enh_period[i],
- (ENH_NBLOCKS_TOT-i)*sizeof(float));
-
-
-
-
-Andersen, et al. Experimental [Page 120]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- /* Set state information to the 6 samples right before
- the samples to be downsampled. */
-
- memcpy(lpState,
- enh_buf+(ENH_NBLOCKS_EXTRA+ioffset)*ENH_BLOCKL-126,
- 6*sizeof(float));
-
- /* Down sample a factor 2 to save computations */
-
- DownSample(enh_buf+(ENH_NBLOCKS_EXTRA+ioffset)*ENH_BLOCKL-120,
- lpFilt_coefsTbl, inLen-ioffset*ENH_BLOCKL,
- lpState, downsampled);
-
- /* Estimate the pitch in the down sampled domain. */
- for (iblock = 0; iblock<ENH_NBLOCKS-ioffset; iblock++) {
-
- lag = 10;
- maxcc = xCorrCoef(downsampled+60+iblock*
- ENH_BLOCKL_HALF, downsampled+60+iblock*
- ENH_BLOCKL_HALF-lag, ENH_BLOCKL_HALF);
- for (ilag=11; ilag<60; ilag++) {
- cc = xCorrCoef(downsampled+60+iblock*
- ENH_BLOCKL_HALF, downsampled+60+iblock*
- ENH_BLOCKL_HALF-ilag, ENH_BLOCKL_HALF);
-
- if (cc > maxcc) {
- maxcc = cc;
- lag = ilag;
- }
- }
-
- /* Store the estimated lag in the non-downsampled domain */
- enh_period[iblock+ENH_NBLOCKS_EXTRA+ioffset] = (float)lag*2;
-
-
- }
-
-
- /* PLC was performed on the previous packet */
- if (iLBCdec_inst->prev_enh_pl==1) {
-
- inlag=(int)enh_period[ENH_NBLOCKS_EXTRA+ioffset];
-
- lag = inlag-1;
- maxcc = xCorrCoef(in, in+lag, plc_blockl);
- for (ilag=inlag; ilag<=inlag+1; ilag++) {
- cc = xCorrCoef(in, in+ilag, plc_blockl);
-
-
-
-
-Andersen, et al. Experimental [Page 121]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- if (cc > maxcc) {
- maxcc = cc;
- lag = ilag;
- }
- }
-
- enh_period[ENH_NBLOCKS_EXTRA+ioffset-1]=(float)lag;
-
- /* compute new concealed residual for the old lookahead,
- mix the forward PLC with a backward PLC from
- the new frame */
-
- inPtr=&in[lag-1];
-
- enh_bufPtr1=&plc_pred[plc_blockl-1];
-
- if (lag>plc_blockl) {
- start=plc_blockl;
- } else {
- start=lag;
- }
-
- for (isample = start; isample>0; isample--) {
- *enh_bufPtr1-- = *inPtr--;
- }
-
- enh_bufPtr2=&enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl];
- for (isample = (plc_blockl-1-lag); isample>=0; isample--) {
- *enh_bufPtr1-- = *enh_bufPtr2--;
- }
-
- /* limit energy change */
- ftmp2=0.0;
- ftmp1=0.0;
- for (i=0;i<plc_blockl;i++) {
- ftmp2+=enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl-i]*
- enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl-i];
- ftmp1+=plc_pred[i]*plc_pred[i];
- }
- ftmp1=(float)sqrt(ftmp1/(float)plc_blockl);
- ftmp2=(float)sqrt(ftmp2/(float)plc_blockl);
- if (ftmp1>(float)2.0*ftmp2 && ftmp1>0.0) {
- for (i=0;i<plc_blockl-10;i++) {
- plc_pred[i]*=(float)2.0*ftmp2/ftmp1;
- }
- for (i=plc_blockl-10;i<plc_blockl;i++) {
- plc_pred[i]*=(float)(i-plc_blockl+10)*
- ((float)1.0-(float)2.0*ftmp2/ftmp1)/(float)(10)+
-
-
-
-Andersen, et al. Experimental [Page 122]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- (float)2.0*ftmp2/ftmp1;
- }
- }
-
- enh_bufPtr1=&enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl];
- for (i=0; i<plc_blockl; i++) {
- ftmp1 = (float) (i+1) / (float) (plc_blockl+1);
- *enh_bufPtr1 *= ftmp1;
- *enh_bufPtr1 += ((float)1.0-ftmp1)*
- plc_pred[plc_blockl-1-i];
- enh_bufPtr1--;
- }
- }
-
- if (iLBCdec_inst->mode==20) {
- /* Enhancer with 40 samples delay */
- for (iblock = 0; iblock<2; iblock++) {
- enhancer(out+iblock*ENH_BLOCKL, enh_buf,
- ENH_BUFL, (5+iblock)*ENH_BLOCKL+40,
- ENH_ALPHA0, enh_period, enh_plocsTbl,
- ENH_NBLOCKS_TOT);
- }
- } else if (iLBCdec_inst->mode==30) {
- /* Enhancer with 80 samples delay */
- for (iblock = 0; iblock<3; iblock++) {
- enhancer(out+iblock*ENH_BLOCKL, enh_buf,
- ENH_BUFL, (4+iblock)*ENH_BLOCKL,
- ENH_ALPHA0, enh_period, enh_plocsTbl,
- ENH_NBLOCKS_TOT);
- }
- }
-
- return (lag*2);
- }
-
-A.17. filter.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- filter.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
-
-
-
-Andersen, et al. Experimental [Page 123]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- #ifndef __iLBC_FILTER_H
- #define __iLBC_FILTER_H
-
- void AllPoleFilter(
- float *InOut, /* (i/o) on entrance InOut[-orderCoef] to
- InOut[-1] contain the state of the
- filter (delayed samples). InOut[0] to
- InOut[lengthInOut-1] contain the filter
- input, on en exit InOut[-orderCoef] to
- InOut[-1] is unchanged and InOut[0] to
- InOut[lengthInOut-1] contain filtered
- samples */
- float *Coef,/* (i) filter coefficients, Coef[0] is assumed
- to be 1.0 */
- int lengthInOut,/* (i) number of input/output samples */
- int orderCoef /* (i) number of filter coefficients */
- );
-
- void AllZeroFilter(
- float *In, /* (i) In[0] to In[lengthInOut-1] contain
- filter input samples */
- float *Coef,/* (i) filter coefficients (Coef[0] is assumed
- to be 1.0) */
- int lengthInOut,/* (i) number of input/output samples */
- int orderCoef, /* (i) number of filter coefficients */
- float *Out /* (i/o) on entrance Out[-orderCoef] to Out[-1]
- contain the filter state, on exit Out[0]
- to Out[lengthInOut-1] contain filtered
- samples */
- );
-
- void ZeroPoleFilter(
- float *In, /* (i) In[0] to In[lengthInOut-1] contain filter
- input samples In[-orderCoef] to In[-1]
- contain state of all-zero section */
- float *ZeroCoef,/* (i) filter coefficients for all-zero
- section (ZeroCoef[0] is assumed to
- be 1.0) */
- float *PoleCoef,/* (i) filter coefficients for all-pole section
- (ZeroCoef[0] is assumed to be 1.0) */
- int lengthInOut,/* (i) number of input/output samples */
- int orderCoef, /* (i) number of filter coefficients */
- float *Out /* (i/o) on entrance Out[-orderCoef] to Out[-1]
- contain state of all-pole section. On
- exit Out[0] to Out[lengthInOut-1]
- contain filtered samples */
- );
-
-
-
-
-Andersen, et al. Experimental [Page 124]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- void DownSample (
- float *In, /* (i) input samples */
- float *Coef, /* (i) filter coefficients */
- int lengthIn, /* (i) number of input samples */
- float *state, /* (i) filter state */
- float *Out /* (o) downsampled output */
- );
-
- #endif
-
-A.18. filter.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- filter.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include "iLBC_define.h"
-
- /*----------------------------------------------------------------*
- * all-pole filter
- *---------------------------------------------------------------*/
-
- void AllPoleFilter(
- float *InOut, /* (i/o) on entrance InOut[-orderCoef] to
- InOut[-1] contain the state of the
- filter (delayed samples). InOut[0] to
- InOut[lengthInOut-1] contain the filter
- input, on en exit InOut[-orderCoef] to
- InOut[-1] is unchanged and InOut[0] to
- InOut[lengthInOut-1] contain filtered
- samples */
- float *Coef,/* (i) filter coefficients, Coef[0] is assumed
- to be 1.0 */
- int lengthInOut,/* (i) number of input/output samples */
- int orderCoef /* (i) number of filter coefficients */
- ){
- int n,k;
-
- for(n=0;n<lengthInOut;n++){
- for(k=1;k<=orderCoef;k++){
- *InOut -= Coef[k]*InOut[-k];
-
-
-
-Andersen, et al. Experimental [Page 125]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- }
- InOut++;
- }
- }
-
- /*----------------------------------------------------------------*
- * all-zero filter
- *---------------------------------------------------------------*/
-
- void AllZeroFilter(
- float *In, /* (i) In[0] to In[lengthInOut-1] contain
- filter input samples */
- float *Coef,/* (i) filter coefficients (Coef[0] is assumed
- to be 1.0) */
- int lengthInOut,/* (i) number of input/output samples */
- int orderCoef, /* (i) number of filter coefficients */
- float *Out /* (i/o) on entrance Out[-orderCoef] to Out[-1]
- contain the filter state, on exit Out[0]
- to Out[lengthInOut-1] contain filtered
- samples */
- ){
- int n,k;
-
- for(n=0;n<lengthInOut;n++){
- *Out = Coef[0]*In[0];
- for(k=1;k<=orderCoef;k++){
- *Out += Coef[k]*In[-k];
- }
- Out++;
- In++;
- }
- }
-
- /*----------------------------------------------------------------*
- * pole-zero filter
- *---------------------------------------------------------------*/
-
- void ZeroPoleFilter(
- float *In, /* (i) In[0] to In[lengthInOut-1] contain
- filter input samples In[-orderCoef] to
- In[-1] contain state of all-zero
- section */
- float *ZeroCoef,/* (i) filter coefficients for all-zero
- section (ZeroCoef[0] is assumed to
- be 1.0) */
- float *PoleCoef,/* (i) filter coefficients for all-pole section
- (ZeroCoef[0] is assumed to be 1.0) */
- int lengthInOut,/* (i) number of input/output samples */
-
-
-
-Andersen, et al. Experimental [Page 126]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- int orderCoef, /* (i) number of filter coefficients */
- float *Out /* (i/o) on entrance Out[-orderCoef] to Out[-1]
- contain state of all-pole section. On
- exit Out[0] to Out[lengthInOut-1]
- contain filtered samples */
- ){
- AllZeroFilter(In,ZeroCoef,lengthInOut,orderCoef,Out);
- AllPoleFilter(Out,PoleCoef,lengthInOut,orderCoef);
- }
-
- /*----------------------------------------------------------------*
- * downsample (LP filter and decimation)
- *---------------------------------------------------------------*/
-
- void DownSample (
- float *In, /* (i) input samples */
- float *Coef, /* (i) filter coefficients */
- int lengthIn, /* (i) number of input samples */
- float *state, /* (i) filter state */
- float *Out /* (o) downsampled output */
- ){
- float o;
- float *Out_ptr = Out;
- float *Coef_ptr, *In_ptr;
- float *state_ptr;
- int i, j, stop;
-
- /* LP filter and decimate at the same time */
-
- for (i = DELAY_DS; i < lengthIn; i+=FACTOR_DS)
- {
- Coef_ptr = &Coef[0];
- In_ptr = &In[i];
- state_ptr = &state[FILTERORDER_DS-2];
-
- o = (float)0.0;
-
- stop = (i < FILTERORDER_DS) ? i + 1 : FILTERORDER_DS;
-
- for (j = 0; j < stop; j++)
- {
- o += *Coef_ptr++ * (*In_ptr--);
- }
- for (j = i + 1; j < FILTERORDER_DS; j++)
- {
- o += *Coef_ptr++ * (*state_ptr--);
- }
-
-
-
-
-Andersen, et al. Experimental [Page 127]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- *Out_ptr++ = o;
- }
-
- /* Get the last part (use zeros as input for the future) */
-
- for (i=(lengthIn+FACTOR_DS); i<(lengthIn+DELAY_DS);
- i+=FACTOR_DS) {
-
- o=(float)0.0;
-
- if (i<lengthIn) {
- Coef_ptr = &Coef[0];
- In_ptr = &In[i];
- for (j=0; j<FILTERORDER_DS; j++) {
- o += *Coef_ptr++ * (*Out_ptr--);
- }
- } else {
- Coef_ptr = &Coef[i-lengthIn];
- In_ptr = &In[lengthIn-1];
- for (j=0; j<FILTERORDER_DS-(i-lengthIn); j++) {
- o += *Coef_ptr++ * (*In_ptr--);
- }
- }
- *Out_ptr++ = o;
- }
- }
-
-A.19. FrameClassify.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- FrameClassify.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_FRAMECLASSIFY_H
- #define __iLBC_FRAMECLASSIFY_H
-
- int FrameClassify( /* index to the max-energy sub-frame */
- iLBC_Enc_Inst_t *iLBCenc_inst,
- /* (i/o) the encoder state structure */
- float *residual /* (i) lpc residual signal */
- );
-
-
-
-Andersen, et al. Experimental [Page 128]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- #endif
-
-A.20. FrameClassify.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- FrameClassify.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include "iLBC_define.h"
-
- /*---------------------------------------------------------------*
- * Classification of subframes to localize start state
- *--------------------------------------------------------------*/
-
- int FrameClassify( /* index to the max-energy sub-frame */
- iLBC_Enc_Inst_t *iLBCenc_inst,
- /* (i/o) the encoder state structure */
- float *residual /* (i) lpc residual signal */
- ) {
- float max_ssqEn, fssqEn[NSUB_MAX], bssqEn[NSUB_MAX], *pp;
- int n, l, max_ssqEn_n;
- const float ssqEn_win[NSUB_MAX-1]={(float)0.8,(float)0.9,
- (float)1.0,(float)0.9,(float)0.8};
- const float sampEn_win[5]={(float)1.0/(float)6.0,
- (float)2.0/(float)6.0, (float)3.0/(float)6.0,
- (float)4.0/(float)6.0, (float)5.0/(float)6.0};
-
- /* init the front and back energies to zero */
-
- memset(fssqEn, 0, NSUB_MAX*sizeof(float));
- memset(bssqEn, 0, NSUB_MAX*sizeof(float));
-
- /* Calculate front of first seqence */
-
- n=0;
- pp=residual;
- for (l=0; l<5; l++) {
- fssqEn[n] += sampEn_win[l] * (*pp) * (*pp);
- pp++;
- }
- for (l=5; l<SUBL; l++) {
-
-
-
-Andersen, et al. Experimental [Page 129]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- fssqEn[n] += (*pp) * (*pp);
- pp++;
- }
-
- /* Calculate front and back of all middle sequences */
-
- for (n=1; n<iLBCenc_inst->nsub-1; n++) {
- pp=residual+n*SUBL;
- for (l=0; l<5; l++) {
- fssqEn[n] += sampEn_win[l] * (*pp) * (*pp);
- bssqEn[n] += (*pp) * (*pp);
- pp++;
- }
- for (l=5; l<SUBL-5; l++) {
- fssqEn[n] += (*pp) * (*pp);
- bssqEn[n] += (*pp) * (*pp);
- pp++;
- }
- for (l=SUBL-5; l<SUBL; l++) {
- fssqEn[n] += (*pp) * (*pp);
- bssqEn[n] += sampEn_win[SUBL-l-1] * (*pp) * (*pp);
- pp++;
- }
- }
-
- /* Calculate back of last seqence */
-
- n=iLBCenc_inst->nsub-1;
- pp=residual+n*SUBL;
- for (l=0; l<SUBL-5; l++) {
- bssqEn[n] += (*pp) * (*pp);
- pp++;
- }
- for (l=SUBL-5; l<SUBL; l++) {
- bssqEn[n] += sampEn_win[SUBL-l-1] * (*pp) * (*pp);
- pp++;
- }
-
- /* find the index to the weighted 80 sample with
- most energy */
-
- if (iLBCenc_inst->mode==20) l=1;
- else l=0;
-
- max_ssqEn=(fssqEn[0]+bssqEn[1])*ssqEn_win[l];
- max_ssqEn_n=1;
- for (n=2; n<iLBCenc_inst->nsub; n++) {
-
-
-
-
-Andersen, et al. Experimental [Page 130]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- l++;
- if ((fssqEn[n-1]+bssqEn[n])*ssqEn_win[l] > max_ssqEn) {
- max_ssqEn=(fssqEn[n-1]+bssqEn[n]) *
- ssqEn_win[l];
- max_ssqEn_n=n;
- }
- }
-
- return max_ssqEn_n;
- }
-
-A.21. gainquant.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- gainquant.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_GAINQUANT_H
- #define __iLBC_GAINQUANT_H
-
- float gainquant(/* (o) quantized gain value */
- float in, /* (i) gain value */
- float maxIn,/* (i) maximum of gain value */
- int cblen, /* (i) number of quantization indices */
- int *index /* (o) quantization index */
- );
-
- float gaindequant( /* (o) quantized gain value */
- int index, /* (i) quantization index */
- float maxIn,/* (i) maximum of unquantized gain */
- int cblen /* (i) number of quantization indices */
- );
-
- #endif
-
-A.22. gainquant.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
-
-
-
-Andersen, et al. Experimental [Page 131]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- gainquant.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include <string.h>
- #include <math.h>
- #include "constants.h"
- #include "filter.h"
-
- /*----------------------------------------------------------------*
- * quantizer for the gain in the gain-shape coding of residual
- *---------------------------------------------------------------*/
-
- float gainquant(/* (o) quantized gain value */
- float in, /* (i) gain value */
- float maxIn,/* (i) maximum of gain value */
- int cblen, /* (i) number of quantization indices */
- int *index /* (o) quantization index */
- ){
- int i, tindex;
- float minmeasure,measure, *cb, scale;
-
- /* ensure a lower bound on the scaling factor */
-
- scale=maxIn;
-
- if (scale<0.1) {
- scale=(float)0.1;
- }
-
- /* select the quantization table */
-
- if (cblen == 8) {
- cb = gain_sq3Tbl;
- } else if (cblen == 16) {
- cb = gain_sq4Tbl;
- } else {
- cb = gain_sq5Tbl;
- }
-
- /* select the best index in the quantization table */
-
- minmeasure=10000000.0;
- tindex=0;
- for (i=0; i<cblen; i++) {
-
-
-
-Andersen, et al. Experimental [Page 132]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- measure=(in-scale*cb[i])*(in-scale*cb[i]);
-
- if (measure<minmeasure) {
- tindex=i;
- minmeasure=measure;
- }
- }
- *index=tindex;
-
- /* return the quantized value */
-
- return scale*cb[tindex];
- }
-
- /*----------------------------------------------------------------*
- * decoder for quantized gains in the gain-shape coding of
- * residual
- *---------------------------------------------------------------*/
-
- float gaindequant( /* (o) quantized gain value */
- int index, /* (i) quantization index */
- float maxIn,/* (i) maximum of unquantized gain */
- int cblen /* (i) number of quantization indices */
- ){
- float scale;
-
- /* obtain correct scale factor */
-
- scale=(float)fabs(maxIn);
-
- if (scale<0.1) {
- scale=(float)0.1;
- }
-
- /* select the quantization table and return the decoded value */
-
- if (cblen==8) {
- return scale*gain_sq3Tbl[index];
- } else if (cblen==16) {
- return scale*gain_sq4Tbl[index];
- }
- else if (cblen==32) {
- return scale*gain_sq5Tbl[index];
- }
-
- return 0.0;
- }
-
-
-
-
-Andersen, et al. Experimental [Page 133]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
-A.23. getCBvec.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- getCBvec.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_GETCBVEC_H
- #define __iLBC_GETCBVEC_H
-
- void getCBvec(
- float *cbvec, /* (o) Constructed codebook vector */
- float *mem, /* (i) Codebook buffer */
- int index, /* (i) Codebook index */
- int lMem, /* (i) Length of codebook buffer */
- int cbveclen/* (i) Codebook vector length */
- );
-
- #endif
-
-A.24. getCBvec.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- getCBvec.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include "iLBC_define.h"
- #include "constants.h"
- #include <string.h>
-
- /*----------------------------------------------------------------*
- * Construct codebook vector for given index.
- *---------------------------------------------------------------*/
-
- void getCBvec(
-
-
-
-Andersen, et al. Experimental [Page 134]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- float *cbvec, /* (o) Constructed codebook vector */
- float *mem, /* (i) Codebook buffer */
- int index, /* (i) Codebook index */
- int lMem, /* (i) Length of codebook buffer */
- int cbveclen/* (i) Codebook vector length */
- ){
- int j, k, n, memInd, sFilt;
- float tmpbuf[CB_MEML];
- int base_size;
- int ilow, ihigh;
- float alfa, alfa1;
-
- /* Determine size of codebook sections */
-
- base_size=lMem-cbveclen+1;
-
- if (cbveclen==SUBL) {
- base_size+=cbveclen/2;
- }
-
- /* No filter -> First codebook section */
-
- if (index<lMem-cbveclen+1) {
-
- /* first non-interpolated vectors */
-
- k=index+cbveclen;
- /* get vector */
- memcpy(cbvec, mem+lMem-k, cbveclen*sizeof(float));
-
- } else if (index < base_size) {
-
- k=2*(index-(lMem-cbveclen+1))+cbveclen;
-
- ihigh=k/2;
- ilow=ihigh-5;
-
- /* Copy first noninterpolated part */
-
- memcpy(cbvec, mem+lMem-k/2, ilow*sizeof(float));
-
- /* interpolation */
-
- alfa1=(float)0.2;
- alfa=0.0;
- for (j=ilow; j<ihigh; j++) {
- cbvec[j]=((float)1.0-alfa)*mem[lMem-k/2+j]+
- alfa*mem[lMem-k+j];
-
-
-
-Andersen, et al. Experimental [Page 135]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- alfa+=alfa1;
- }
-
- /* Copy second noninterpolated part */
-
- memcpy(cbvec+ihigh, mem+lMem-k+ihigh,
- (cbveclen-ihigh)*sizeof(float));
-
- }
-
- /* Higher codebook section based on filtering */
-
- else {
-
- /* first non-interpolated vectors */
-
- if (index-base_size<lMem-cbveclen+1) {
- float tempbuff2[CB_MEML+CB_FILTERLEN+1];
- float *pos;
- float *pp, *pp1;
-
- memset(tempbuff2, 0,
- CB_HALFFILTERLEN*sizeof(float));
- memcpy(&tempbuff2[CB_HALFFILTERLEN], mem,
- lMem*sizeof(float));
- memset(&tempbuff2[lMem+CB_HALFFILTERLEN], 0,
- (CB_HALFFILTERLEN+1)*sizeof(float));
-
- k=index-base_size+cbveclen;
- sFilt=lMem-k;
- memInd=sFilt+1-CB_HALFFILTERLEN;
-
- /* do filtering */
- pos=cbvec;
- memset(pos, 0, cbveclen*sizeof(float));
- for (n=0; n<cbveclen; n++) {
- pp=&tempbuff2[memInd+n+CB_HALFFILTERLEN];
- pp1=&cbfiltersTbl[CB_FILTERLEN-1];
- for (j=0; j<CB_FILTERLEN; j++) {
- (*pos)+=(*pp++)*(*pp1--);
- }
- pos++;
- }
- }
-
- /* interpolated vectors */
-
- else {
-
-
-
-Andersen, et al. Experimental [Page 136]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- float tempbuff2[CB_MEML+CB_FILTERLEN+1];
-
- float *pos;
- float *pp, *pp1;
- int i;
-
- memset(tempbuff2, 0,
- CB_HALFFILTERLEN*sizeof(float));
- memcpy(&tempbuff2[CB_HALFFILTERLEN], mem,
- lMem*sizeof(float));
- memset(&tempbuff2[lMem+CB_HALFFILTERLEN], 0,
- (CB_HALFFILTERLEN+1)*sizeof(float));
-
- k=2*(index-base_size-
- (lMem-cbveclen+1))+cbveclen;
- sFilt=lMem-k;
- memInd=sFilt+1-CB_HALFFILTERLEN;
-
- /* do filtering */
- pos=&tmpbuf[sFilt];
- memset(pos, 0, k*sizeof(float));
- for (i=0; i<k; i++) {
- pp=&tempbuff2[memInd+i+CB_HALFFILTERLEN];
- pp1=&cbfiltersTbl[CB_FILTERLEN-1];
- for (j=0; j<CB_FILTERLEN; j++) {
- (*pos)+=(*pp++)*(*pp1--);
- }
- pos++;
- }
-
- ihigh=k/2;
- ilow=ihigh-5;
-
- /* Copy first noninterpolated part */
-
- memcpy(cbvec, tmpbuf+lMem-k/2,
- ilow*sizeof(float));
-
- /* interpolation */
-
- alfa1=(float)0.2;
- alfa=0.0;
- for (j=ilow; j<ihigh; j++) {
- cbvec[j]=((float)1.0-alfa)*
- tmpbuf[lMem-k/2+j]+alfa*tmpbuf[lMem-k+j];
- alfa+=alfa1;
- }
-
-
-
-
-Andersen, et al. Experimental [Page 137]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- /* Copy second noninterpolated part */
-
- memcpy(cbvec+ihigh, tmpbuf+lMem-k+ihigh,
- (cbveclen-ihigh)*sizeof(float));
- }
- }
- }
-
-A.25. helpfun.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- helpfun.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_HELPFUN_H
- #define __iLBC_HELPFUN_H
-
- void autocorr(
- float *r, /* (o) autocorrelation vector */
- const float *x, /* (i) data vector */
- int N, /* (i) length of data vector */
- int order /* largest lag for calculated
- autocorrelations */
- );
-
- void window(
- float *z, /* (o) the windowed data */
- const float *x, /* (i) the original data vector */
- const float *y, /* (i) the window */
- int N /* (i) length of all vectors */
- );
-
- void levdurb(
- float *a, /* (o) lpc coefficient vector starting
- with 1.0 */
- float *k, /* (o) reflection coefficients */
- float *r, /* (i) autocorrelation vector */
- int order /* (i) order of lpc filter */
- );
-
- void interpolate(
-
-
-
-Andersen, et al. Experimental [Page 138]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- float *out, /* (o) the interpolated vector */
- float *in1, /* (i) the first vector for the
- interpolation */
- float *in2, /* (i) the second vector for the
- interpolation */
- float coef, /* (i) interpolation weights */
- int length /* (i) length of all vectors */
- );
-
- void bwexpand(
- float *out, /* (o) the bandwidth expanded lpc
- coefficients */
- float *in, /* (i) the lpc coefficients before bandwidth
- expansion */
- float coef, /* (i) the bandwidth expansion factor */
- int length /* (i) the length of lpc coefficient vectors */
- );
-
- void vq(
- float *Xq, /* (o) the quantized vector */
- int *index, /* (o) the quantization index */
- const float *CB,/* (i) the vector quantization codebook */
- float *X, /* (i) the vector to quantize */
- int n_cb, /* (i) the number of vectors in the codebook */
- int dim /* (i) the dimension of all vectors */
- );
-
- void SplitVQ(
- float *qX, /* (o) the quantized vector */
- int *index, /* (o) a vector of indexes for all vector
- codebooks in the split */
- float *X, /* (i) the vector to quantize */
- const float *CB,/* (i) the quantizer codebook */
- int nsplit, /* the number of vector splits */
- const int *dim, /* the dimension of X and qX */
- const int *cbsize /* the number of vectors in the codebook */
- );
-
-
- void sort_sq(
- float *xq, /* (o) the quantized value */
- int *index, /* (o) the quantization index */
- float x, /* (i) the value to quantize */
- const float *cb,/* (i) the quantization codebook */
- int cb_size /* (i) the size of the quantization codebook */
- );
-
- int LSF_check( /* (o) 1 for stable lsf vectors and 0 for
-
-
-
-Andersen, et al. Experimental [Page 139]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- nonstable ones */
- float *lsf, /* (i) a table of lsf vectors */
- int dim, /* (i) the dimension of each lsf vector */
- int NoAn /* (i) the number of lsf vectors in the
- table */
- );
-
- #endif
-
-A.26. helpfun.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- helpfun.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include <math.h>
-
- #include "iLBC_define.h"
- #include "constants.h"
-
- /*----------------------------------------------------------------*
- * calculation of auto correlation
- *---------------------------------------------------------------*/
-
- void autocorr(
- float *r, /* (o) autocorrelation vector */
- const float *x, /* (i) data vector */
- int N, /* (i) length of data vector */
- int order /* largest lag for calculated
- autocorrelations */
- ){
- int lag, n;
- float sum;
-
- for (lag = 0; lag <= order; lag++) {
- sum = 0;
- for (n = 0; n < N - lag; n++) {
- sum += x[n] * x[n+lag];
- }
- r[lag] = sum;
- }
-
-
-
-Andersen, et al. Experimental [Page 140]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- }
-
- /*----------------------------------------------------------------*
- * window multiplication
- *---------------------------------------------------------------*/
-
- void window(
- float *z, /* (o) the windowed data */
- const float *x, /* (i) the original data vector */
- const float *y, /* (i) the window */
- int N /* (i) length of all vectors */
- ){
- int i;
-
- for (i = 0; i < N; i++) {
- z[i] = x[i] * y[i];
- }
- }
-
- /*----------------------------------------------------------------*
- * levinson-durbin solution for lpc coefficients
- *---------------------------------------------------------------*/
-
- void levdurb(
- float *a, /* (o) lpc coefficient vector starting
- with 1.0 */
- float *k, /* (o) reflection coefficients */
- float *r, /* (i) autocorrelation vector */
- int order /* (i) order of lpc filter */
- ){
- float sum, alpha;
- int m, m_h, i;
-
- a[0] = 1.0;
-
- if (r[0] < EPS) { /* if r[0] <= 0, set LPC coeff. to zero */
- for (i = 0; i < order; i++) {
- k[i] = 0;
- a[i+1] = 0;
- }
- } else {
- a[1] = k[0] = -r[1]/r[0];
- alpha = r[0] + r[1] * k[0];
- for (m = 1; m < order; m++){
- sum = r[m + 1];
- for (i = 0; i < m; i++){
- sum += a[i+1] * r[m - i];
- }
-
-
-
-Andersen, et al. Experimental [Page 141]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- k[m] = -sum / alpha;
- alpha += k[m] * sum;
- m_h = (m + 1) >> 1;
- for (i = 0; i < m_h; i++){
- sum = a[i+1] + k[m] * a[m - i];
- a[m - i] += k[m] * a[i+1];
- a[i+1] = sum;
- }
- a[m+1] = k[m];
- }
- }
- }
-
- /*----------------------------------------------------------------*
- * interpolation between vectors
- *---------------------------------------------------------------*/
-
- void interpolate(
- float *out, /* (o) the interpolated vector */
- float *in1, /* (i) the first vector for the
- interpolation */
- float *in2, /* (i) the second vector for the
- interpolation */
- float coef, /* (i) interpolation weights */
- int length /* (i) length of all vectors */
- ){
- int i;
- float invcoef;
-
- invcoef = (float)1.0 - coef;
- for (i = 0; i < length; i++) {
- out[i] = coef * in1[i] + invcoef * in2[i];
- }
- }
-
- /*----------------------------------------------------------------*
- * lpc bandwidth expansion
- *---------------------------------------------------------------*/
-
- void bwexpand(
- float *out, /* (o) the bandwidth expanded lpc
- coefficients */
- float *in, /* (i) the lpc coefficients before bandwidth
- expansion */
- float coef, /* (i) the bandwidth expansion factor */
- int length /* (i) the length of lpc coefficient vectors */
- ){
- int i;
-
-
-
-Andersen, et al. Experimental [Page 142]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- float chirp;
-
- chirp = coef;
-
- out[0] = in[0];
- for (i = 1; i < length; i++) {
- out[i] = chirp * in[i];
- chirp *= coef;
- }
- }
-
- /*----------------------------------------------------------------*
- * vector quantization
- *---------------------------------------------------------------*/
-
- void vq(
- float *Xq, /* (o) the quantized vector */
- int *index, /* (o) the quantization index */
- const float *CB,/* (i) the vector quantization codebook */
- float *X, /* (i) the vector to quantize */
- int n_cb, /* (i) the number of vectors in the codebook */
- int dim /* (i) the dimension of all vectors */
- ){
- int i, j;
- int pos, minindex;
- float dist, tmp, mindist;
-
- pos = 0;
- mindist = FLOAT_MAX;
- minindex = 0;
- for (j = 0; j < n_cb; j++) {
- dist = X[0] - CB[pos];
- dist *= dist;
- for (i = 1; i < dim; i++) {
- tmp = X[i] - CB[pos + i];
- dist += tmp*tmp;
- }
-
- if (dist < mindist) {
- mindist = dist;
- minindex = j;
- }
- pos += dim;
- }
- for (i = 0; i < dim; i++) {
- Xq[i] = CB[minindex*dim + i];
- }
- *index = minindex;
-
-
-
-Andersen, et al. Experimental [Page 143]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- }
-
- /*----------------------------------------------------------------*
- * split vector quantization
- *---------------------------------------------------------------*/
-
- void SplitVQ(
- float *qX, /* (o) the quantized vector */
- int *index, /* (o) a vector of indexes for all vector
- codebooks in the split */
- float *X, /* (i) the vector to quantize */
- const float *CB,/* (i) the quantizer codebook */
- int nsplit, /* the number of vector splits */
- const int *dim, /* the dimension of X and qX */
- const int *cbsize /* the number of vectors in the codebook */
- ){
- int cb_pos, X_pos, i;
-
- cb_pos = 0;
- X_pos= 0;
- for (i = 0; i < nsplit; i++) {
- vq(qX + X_pos, index + i, CB + cb_pos, X + X_pos,
- cbsize[i], dim[i]);
- X_pos += dim[i];
- cb_pos += dim[i] * cbsize[i];
- }
- }
-
- /*----------------------------------------------------------------*
- * scalar quantization
- *---------------------------------------------------------------*/
-
- void sort_sq(
- float *xq, /* (o) the quantized value */
- int *index, /* (o) the quantization index */
- float x, /* (i) the value to quantize */
- const float *cb,/* (i) the quantization codebook */
- int cb_size /* (i) the size of the quantization codebook */
- ){
- int i;
-
- if (x <= cb[0]) {
- *index = 0;
- *xq = cb[0];
- } else {
- i = 0;
- while ((x > cb[i]) && i < cb_size - 1) {
- i++;
-
-
-
-Andersen, et al. Experimental [Page 144]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- }
-
- if (x > ((cb[i] + cb[i - 1])/2)) {
- *index = i;
- *xq = cb[i];
- } else {
- *index = i - 1;
- *xq = cb[i - 1];
- }
- }
- }
-
- /*----------------------------------------------------------------*
- * check for stability of lsf coefficients
- *---------------------------------------------------------------*/
-
- int LSF_check( /* (o) 1 for stable lsf vectors and 0 for
- nonstable ones */
- float *lsf, /* (i) a table of lsf vectors */
- int dim, /* (i) the dimension of each lsf vector */
- int NoAn /* (i) the number of lsf vectors in the
- table */
- ){
- int k,n,m, Nit=2, change=0,pos;
- float tmp;
- static float eps=(float)0.039; /* 50 Hz */
- static float eps2=(float)0.0195;
- static float maxlsf=(float)3.14; /* 4000 Hz */
- static float minlsf=(float)0.01; /* 0 Hz */
-
- /* LSF separation check*/
-
- for (n=0; n<Nit; n++) { /* Run through a couple of times */
- for (m=0; m<NoAn; m++) { /* Number of analyses per frame */
- for (k=0; k<(dim-1); k++) {
- pos=m*dim+k;
-
- if ((lsf[pos+1]-lsf[pos])<eps) {
-
- if (lsf[pos+1]<lsf[pos]) {
- tmp=lsf[pos+1];
- lsf[pos+1]= lsf[pos]+eps2;
- lsf[pos]= lsf[pos+1]-eps2;
- } else {
- lsf[pos]-=eps2;
- lsf[pos+1]+=eps2;
- }
- change=1;
-
-
-
-Andersen, et al. Experimental [Page 145]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- }
-
- if (lsf[pos]<minlsf) {
- lsf[pos]=minlsf;
- change=1;
- }
-
- if (lsf[pos]>maxlsf) {
- lsf[pos]=maxlsf;
- change=1;
- }
- }
- }
- }
-
- return change;
- }
-
-A.27. hpInput.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- hpInput.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_HPINPUT_H
- #define __iLBC_HPINPUT_H
-
- void hpInput(
- float *In, /* (i) vector to filter */
- int len, /* (i) length of vector to filter */
- float *Out, /* (o) the resulting filtered vector */
- float *mem /* (i/o) the filter state */
- );
-
- #endif
-
-A.28. hpInput.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
-
-
-Andersen, et al. Experimental [Page 146]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- hpInput.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include "constants.h"
-
- /*----------------------------------------------------------------*
- * Input high-pass filter
- *---------------------------------------------------------------*/
-
- void hpInput(
- float *In, /* (i) vector to filter */
- int len, /* (i) length of vector to filter */
- float *Out, /* (o) the resulting filtered vector */
- float *mem /* (i/o) the filter state */
- ){
- int i;
- float *pi, *po;
-
- /* all-zero section*/
-
- pi = &In[0];
- po = &Out[0];
- for (i=0; i<len; i++) {
- *po = hpi_zero_coefsTbl[0] * (*pi);
- *po += hpi_zero_coefsTbl[1] * mem[0];
- *po += hpi_zero_coefsTbl[2] * mem[1];
-
- mem[1] = mem[0];
- mem[0] = *pi;
- po++;
- pi++;
-
- }
-
- /* all-pole section*/
-
- po = &Out[0];
- for (i=0; i<len; i++) {
- *po -= hpi_pole_coefsTbl[1] * mem[2];
- *po -= hpi_pole_coefsTbl[2] * mem[3];
-
- mem[3] = mem[2];
- mem[2] = *po;
- po++;
-
-
-
-Andersen, et al. Experimental [Page 147]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- }
- }
-
-A.29. hpOutput.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- hpOutput.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_HPOUTPUT_H
- #define __iLBC_HPOUTPUT_H
-
- void hpOutput(
- float *In, /* (i) vector to filter */
- int len,/* (i) length of vector to filter */
- float *Out, /* (o) the resulting filtered vector */
- float *mem /* (i/o) the filter state */
- );
-
- #endif
-
-A.30. hpOutput.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- hpOutput.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include "constants.h"
-
- /*----------------------------------------------------------------*
- * Output high-pass filter
- *---------------------------------------------------------------*/
-
- void hpOutput(
-
-
-
-Andersen, et al. Experimental [Page 148]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- float *In, /* (i) vector to filter */
- int len,/* (i) length of vector to filter */
- float *Out, /* (o) the resulting filtered vector */
- float *mem /* (i/o) the filter state */
- ){
- int i;
- float *pi, *po;
-
- /* all-zero section*/
-
- pi = &In[0];
- po = &Out[0];
- for (i=0; i<len; i++) {
- *po = hpo_zero_coefsTbl[0] * (*pi);
- *po += hpo_zero_coefsTbl[1] * mem[0];
- *po += hpo_zero_coefsTbl[2] * mem[1];
-
- mem[1] = mem[0];
- mem[0] = *pi;
- po++;
- pi++;
-
- }
-
- /* all-pole section*/
-
- po = &Out[0];
- for (i=0; i<len; i++) {
- *po -= hpo_pole_coefsTbl[1] * mem[2];
- *po -= hpo_pole_coefsTbl[2] * mem[3];
-
- mem[3] = mem[2];
- mem[2] = *po;
- po++;
- }
- }
-
-A.31. iCBConstruct.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- iCBConstruct.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
-
-
-
-Andersen, et al. Experimental [Page 149]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- ******************************************************************/
-
- #ifndef __iLBC_ICBCONSTRUCT_H
- #define __iLBC_ICBCONSTRUCT_H
-
- void index_conv_enc(
- int *index /* (i/o) Codebook indexes */
- );
-
- void index_conv_dec(
- int *index /* (i/o) Codebook indexes */
- );
-
- void iCBConstruct(
- float *decvector, /* (o) Decoded vector */
- int *index, /* (i) Codebook indices */
- int *gain_index,/* (i) Gain quantization indices */
- float *mem, /* (i) Buffer for codevector construction */
- int lMem, /* (i) Length of buffer */
- int veclen, /* (i) Length of vector */
- int nStages /* (i) Number of codebook stages */
- );
-
- #endif
-
-A.32. iCBConstruct.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- iCBConstruct.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include <math.h>
-
- #include "iLBC_define.h"
- #include "gainquant.h"
- #include "getCBvec.h"
-
- /*----------------------------------------------------------------*
- * Convert the codebook indexes to make the search easier
- *---------------------------------------------------------------*/
-
-
-
-
-Andersen, et al. Experimental [Page 150]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- void index_conv_enc(
- int *index /* (i/o) Codebook indexes */
- ){
- int k;
-
- for (k=1; k<CB_NSTAGES; k++) {
-
- if ((index[k]>=108)&&(index[k]<172)) {
- index[k]-=64;
- } else if (index[k]>=236) {
- index[k]-=128;
- } else {
- /* ERROR */
- }
- }
- }
-
- void index_conv_dec(
- int *index /* (i/o) Codebook indexes */
- ){
- int k;
-
- for (k=1; k<CB_NSTAGES; k++) {
-
- if ((index[k]>=44)&&(index[k]<108)) {
- index[k]+=64;
- } else if ((index[k]>=108)&&(index[k]<128)) {
- index[k]+=128;
- } else {
- /* ERROR */
- }
- }
- }
-
- /*----------------------------------------------------------------*
- * Construct decoded vector from codebook and gains.
- *---------------------------------------------------------------*/
-
- void iCBConstruct(
- float *decvector, /* (o) Decoded vector */
- int *index, /* (i) Codebook indices */
- int *gain_index,/* (i) Gain quantization indices */
- float *mem, /* (i) Buffer for codevector construction */
- int lMem, /* (i) Length of buffer */
- int veclen, /* (i) Length of vector */
- int nStages /* (i) Number of codebook stages */
- ){
- int j,k;
-
-
-
-Andersen, et al. Experimental [Page 151]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- float gain[CB_NSTAGES];
- float cbvec[SUBL];
-
- /* gain de-quantization */
-
- gain[0] = gaindequant(gain_index[0], 1.0, 32);
- if (nStages > 1) {
- gain[1] = gaindequant(gain_index[1],
- (float)fabs(gain[0]), 16);
- }
- if (nStages > 2) {
- gain[2] = gaindequant(gain_index[2],
- (float)fabs(gain[1]), 8);
- }
-
- /* codebook vector construction and construction of
- total vector */
-
- getCBvec(cbvec, mem, index[0], lMem, veclen);
- for (j=0;j<veclen;j++){
- decvector[j] = gain[0]*cbvec[j];
- }
- if (nStages > 1) {
- for (k=1; k<nStages; k++) {
- getCBvec(cbvec, mem, index[k], lMem, veclen);
- for (j=0;j<veclen;j++) {
- decvector[j] += gain[k]*cbvec[j];
- }
- }
- }
- }
-
-A.33. iCBSearch.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- iCBSearch.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_ICBSEARCH_H
- #define __iLBC_ICBSEARCH_H
-
-
-
-
-Andersen, et al. Experimental [Page 152]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- void iCBSearch(
- iLBC_Enc_Inst_t *iLBCenc_inst,
- /* (i) the encoder state structure */
- int *index, /* (o) Codebook indices */
- int *gain_index,/* (o) Gain quantization indices */
- float *intarget,/* (i) Target vector for encoding */
- float *mem, /* (i) Buffer for codebook construction */
- int lMem, /* (i) Length of buffer */
- int lTarget, /* (i) Length of vector */
- int nStages, /* (i) Number of codebook stages */
- float *weightDenum, /* (i) weighting filter coefficients */
- float *weightState, /* (i) weighting filter state */
- int block /* (i) the sub-block number */
- );
-
- #endif
-
-A.34. iCBSearch.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- iCBSearch.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include <math.h>
- #include <string.h>
-
- #include "iLBC_define.h"
- #include "gainquant.h"
- #include "createCB.h"
- #include "filter.h"
- #include "constants.h"
-
- /*----------------------------------------------------------------*
- * Search routine for codebook encoding and gain quantization.
- *---------------------------------------------------------------*/
-
- void iCBSearch(
- iLBC_Enc_Inst_t *iLBCenc_inst,
- /* (i) the encoder state structure */
- int *index, /* (o) Codebook indices */
- int *gain_index,/* (o) Gain quantization indices */
-
-
-
-Andersen, et al. Experimental [Page 153]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- float *intarget,/* (i) Target vector for encoding */
- float *mem, /* (i) Buffer for codebook construction */
- int lMem, /* (i) Length of buffer */
- int lTarget, /* (i) Length of vector */
- int nStages, /* (i) Number of codebook stages */
- float *weightDenum, /* (i) weighting filter coefficients */
- float *weightState, /* (i) weighting filter state */
- int block /* (i) the sub-block number */
- ){
- int i, j, icount, stage, best_index, range, counter;
- float max_measure, gain, measure, crossDot, ftmp;
- float gains[CB_NSTAGES];
- float target[SUBL];
- int base_index, sInd, eInd, base_size;
- int sIndAug=0, eIndAug=0;
- float buf[CB_MEML+SUBL+2*LPC_FILTERORDER];
- float invenergy[CB_EXPAND*128], energy[CB_EXPAND*128];
- float *pp, *ppi=0, *ppo=0, *ppe=0;
- float cbvectors[CB_MEML];
- float tene, cene, cvec[SUBL];
- float aug_vec[SUBL];
-
- memset(cvec,0,SUBL*sizeof(float));
-
- /* Determine size of codebook sections */
-
- base_size=lMem-lTarget+1;
-
- if (lTarget==SUBL) {
- base_size=lMem-lTarget+1+lTarget/2;
- }
-
- /* setup buffer for weighting */
-
- memcpy(buf,weightState,sizeof(float)*LPC_FILTERORDER);
- memcpy(buf+LPC_FILTERORDER,mem,lMem*sizeof(float));
- memcpy(buf+LPC_FILTERORDER+lMem,intarget,lTarget*sizeof(float));
-
- /* weighting */
-
- AllPoleFilter(buf+LPC_FILTERORDER, weightDenum,
- lMem+lTarget, LPC_FILTERORDER);
-
- /* Construct the codebook and target needed */
-
- memcpy(target, buf+LPC_FILTERORDER+lMem, lTarget*sizeof(float));
-
- tene=0.0;
-
-
-
-Andersen, et al. Experimental [Page 154]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- for (i=0; i<lTarget; i++) {
- tene+=target[i]*target[i];
- }
-
- /* Prepare search over one more codebook section. This section
- is created by filtering the original buffer with a filter. */
-
- filteredCBvecs(cbvectors, buf+LPC_FILTERORDER, lMem);
-
- /* The Main Loop over stages */
-
- for (stage=0; stage<nStages; stage++) {
-
- range = search_rangeTbl[block][stage];
-
- /* initialize search measure */
-
- max_measure = (float)-10000000.0;
- gain = (float)0.0;
- best_index = 0;
-
- /* Compute cross dot product between the target
- and the CB memory */
-
- crossDot=0.0;
- pp=buf+LPC_FILTERORDER+lMem-lTarget;
- for (j=0; j<lTarget; j++) {
- crossDot += target[j]*(*pp++);
- }
-
- if (stage==0) {
-
- /* Calculate energy in the first block of
- 'lTarget' samples. */
- ppe = energy;
- ppi = buf+LPC_FILTERORDER+lMem-lTarget-1;
- ppo = buf+LPC_FILTERORDER+lMem-1;
-
- *ppe=0.0;
- pp=buf+LPC_FILTERORDER+lMem-lTarget;
- for (j=0; j<lTarget; j++) {
- *ppe+=(*pp)*(*pp++);
- }
-
- if (*ppe>0.0) {
- invenergy[0] = (float) 1.0 / (*ppe + EPS);
- } else {
- invenergy[0] = (float) 0.0;
-
-
-
-Andersen, et al. Experimental [Page 155]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- }
- ppe++;
-
- measure=(float)-10000000.0;
-
- if (crossDot > 0.0) {
- measure = crossDot*crossDot*invenergy[0];
- }
- }
- else {
- measure = crossDot*crossDot*invenergy[0];
- }
-
- /* check if measure is better */
- ftmp = crossDot*invenergy[0];
-
- if ((measure>max_measure) && (fabs(ftmp)<CB_MAXGAIN)) {
- best_index = 0;
- max_measure = measure;
- gain = ftmp;
- }
-
- /* loop over the main first codebook section,
- full search */
-
- for (icount=1; icount<range; icount++) {
-
- /* calculate measure */
-
- crossDot=0.0;
- pp = buf+LPC_FILTERORDER+lMem-lTarget-icount;
-
- for (j=0; j<lTarget; j++) {
- crossDot += target[j]*(*pp++);
- }
-
- if (stage==0) {
- *ppe++ = energy[icount-1] + (*ppi)*(*ppi) -
- (*ppo)*(*ppo);
- ppo--;
- ppi--;
-
- if (energy[icount]>0.0) {
- invenergy[icount] =
- (float)1.0/(energy[icount]+EPS);
- } else {
- invenergy[icount] = (float) 0.0;
- }
-
-
-
-Andersen, et al. Experimental [Page 156]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- measure=(float)-10000000.0;
-
- if (crossDot > 0.0) {
- measure = crossDot*crossDot*invenergy[icount];
- }
- }
- else {
- measure = crossDot*crossDot*invenergy[icount];
- }
-
- /* check if measure is better */
- ftmp = crossDot*invenergy[icount];
-
- if ((measure>max_measure) && (fabs(ftmp)<CB_MAXGAIN)) {
- best_index = icount;
- max_measure = measure;
- gain = ftmp;
- }
- }
-
- /* Loop over augmented part in the first codebook
- * section, full search.
- * The vectors are interpolated.
- */
-
- if (lTarget==SUBL) {
-
- /* Search for best possible cb vector and
- compute the CB-vectors' energy. */
- searchAugmentedCB(20, 39, stage, base_size-lTarget/2,
- target, buf+LPC_FILTERORDER+lMem,
- &max_measure, &best_index, &gain, energy,
- invenergy);
- }
-
- /* set search range for following codebook sections */
-
- base_index=best_index;
-
- /* unrestricted search */
-
- if (CB_RESRANGE == -1) {
- sInd=0;
- eInd=range-1;
- sIndAug=20;
- eIndAug=39;
- }
-
-
-
-
-Andersen, et al. Experimental [Page 157]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- /* restricted search around best index from first
- codebook section */
-
- else {
- /* Initialize search indices */
- sIndAug=0;
- eIndAug=0;
- sInd=base_index-CB_RESRANGE/2;
- eInd=sInd+CB_RESRANGE;
-
- if (lTarget==SUBL) {
-
- if (sInd<0) {
-
- sIndAug = 40 + sInd;
- eIndAug = 39;
- sInd=0;
-
- } else if ( base_index < (base_size-20) ) {
-
- if (eInd > range) {
- sInd -= (eInd-range);
- eInd = range;
- }
- } else { /* base_index >= (base_size-20) */
-
- if (sInd < (base_size-20)) {
- sIndAug = 20;
- sInd = 0;
- eInd = 0;
- eIndAug = 19 + CB_RESRANGE;
-
- if(eIndAug > 39) {
- eInd = eIndAug-39;
- eIndAug = 39;
- }
- } else {
- sIndAug = 20 + sInd - (base_size-20);
- eIndAug = 39;
- sInd = 0;
- eInd = CB_RESRANGE - (eIndAug-sIndAug+1);
- }
- }
-
- } else { /* lTarget = 22 or 23 */
-
- if (sInd < 0) {
- eInd -= sInd;
-
-
-
-Andersen, et al. Experimental [Page 158]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- sInd = 0;
- }
-
- if(eInd > range) {
- sInd -= (eInd - range);
- eInd = range;
- }
- }
- }
-
- /* search of higher codebook section */
-
- /* index search range */
- counter = sInd;
- sInd += base_size;
- eInd += base_size;
-
-
- if (stage==0) {
- ppe = energy+base_size;
- *ppe=0.0;
-
- pp=cbvectors+lMem-lTarget;
- for (j=0; j<lTarget; j++) {
- *ppe+=(*pp)*(*pp++);
- }
-
- ppi = cbvectors + lMem - 1 - lTarget;
- ppo = cbvectors + lMem - 1;
-
- for (j=0; j<(range-1); j++) {
- *(ppe+1) = *ppe + (*ppi)*(*ppi) - (*ppo)*(*ppo);
- ppo--;
- ppi--;
- ppe++;
- }
- }
-
- /* loop over search range */
-
- for (icount=sInd; icount<eInd; icount++) {
-
- /* calculate measure */
-
- crossDot=0.0;
- pp=cbvectors + lMem - (counter++) - lTarget;
-
- for (j=0;j<lTarget;j++) {
-
-
-
-Andersen, et al. Experimental [Page 159]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- crossDot += target[j]*(*pp++);
- }
-
- if (energy[icount]>0.0) {
- invenergy[icount] =(float)1.0/(energy[icount]+EPS);
- } else {
- invenergy[icount] =(float)0.0;
- }
-
- if (stage==0) {
-
- measure=(float)-10000000.0;
-
- if (crossDot > 0.0) {
- measure = crossDot*crossDot*
- invenergy[icount];
- }
- }
- else {
- measure = crossDot*crossDot*invenergy[icount];
- }
-
- /* check if measure is better */
- ftmp = crossDot*invenergy[icount];
-
- if ((measure>max_measure) && (fabs(ftmp)<CB_MAXGAIN)) {
- best_index = icount;
- max_measure = measure;
- gain = ftmp;
- }
- }
-
- /* Search the augmented CB inside the limited range. */
-
- if ((lTarget==SUBL)&&(sIndAug!=0)) {
- searchAugmentedCB(sIndAug, eIndAug, stage,
- 2*base_size-20, target, cbvectors+lMem,
- &max_measure, &best_index, &gain, energy,
- invenergy);
- }
-
- /* record best index */
-
- index[stage] = best_index;
-
- /* gain quantization */
-
- if (stage==0){
-
-
-
-Andersen, et al. Experimental [Page 160]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
-
- if (gain<0.0){
- gain = 0.0;
- }
-
- if (gain>CB_MAXGAIN) {
- gain = (float)CB_MAXGAIN;
- }
- gain = gainquant(gain, 1.0, 32, &gain_index[stage]);
- }
- else {
- if (stage==1) {
- gain = gainquant(gain, (float)fabs(gains[stage-1]),
- 16, &gain_index[stage]);
- } else {
- gain = gainquant(gain, (float)fabs(gains[stage-1]),
- 8, &gain_index[stage]);
- }
- }
-
- /* Extract the best (according to measure)
- codebook vector */
-
- if (lTarget==(STATE_LEN-iLBCenc_inst->state_short_len)) {
-
- if (index[stage]<base_size) {
- pp=buf+LPC_FILTERORDER+lMem-lTarget-index[stage];
- } else {
- pp=cbvectors+lMem-lTarget-
- index[stage]+base_size;
- }
- } else {
-
- if (index[stage]<base_size) {
- if (index[stage]<(base_size-20)) {
- pp=buf+LPC_FILTERORDER+lMem-
- lTarget-index[stage];
- } else {
- createAugmentedVec(index[stage]-base_size+40,
- buf+LPC_FILTERORDER+lMem,aug_vec);
- pp=aug_vec;
- }
- } else {
- int filterno, position;
-
- filterno=index[stage]/base_size;
- position=index[stage]-filterno*base_size;
-
-
-
-
-Andersen, et al. Experimental [Page 161]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
-
- if (position<(base_size-20)) {
- pp=cbvectors+filterno*lMem-lTarget-
- index[stage]+filterno*base_size;
- } else {
- createAugmentedVec(
- index[stage]-(filterno+1)*base_size+40,
- cbvectors+filterno*lMem,aug_vec);
- pp=aug_vec;
- }
- }
- }
-
- /* Subtract the best codebook vector, according
- to measure, from the target vector */
-
- for (j=0;j<lTarget;j++) {
- cvec[j] += gain*(*pp);
- target[j] -= gain*(*pp++);
- }
-
- /* record quantized gain */
-
- gains[stage]=gain;
-
- }/* end of Main Loop. for (stage=0;... */
-
- /* Gain adjustment for energy matching */
- cene=0.0;
- for (i=0; i<lTarget; i++) {
- cene+=cvec[i]*cvec[i];
- }
- j=gain_index[0];
-
- for (i=gain_index[0]; i<32; i++) {
- ftmp=cene*gain_sq5Tbl[i]*gain_sq5Tbl[i];
-
- if ((ftmp<(tene*gains[0]*gains[0])) &&
- (gain_sq5Tbl[j]<(2.0*gains[0]))) {
- j=i;
- }
- }
- gain_index[0]=j;
- }
-
-
-
-
-
-
-
-Andersen, et al. Experimental [Page 162]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
-A.35. LPCdecode.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- LPC_decode.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_LPC_DECODE_H
- #define __iLBC_LPC_DECODE_H
-
- void LSFinterpolate2a_dec(
- float *a, /* (o) lpc coefficients for a sub-frame */
- float *lsf1, /* (i) first lsf coefficient vector */
- float *lsf2, /* (i) second lsf coefficient vector */
- float coef, /* (i) interpolation weight */
- int length /* (i) length of lsf vectors */
- );
-
- void SimplelsfDEQ(
- float *lsfdeq, /* (o) dequantized lsf coefficients */
- int *index, /* (i) quantization index */
- int lpc_n /* (i) number of LPCs */
- );
-
- void DecoderInterpolateLSF(
- float *syntdenum, /* (o) synthesis filter coefficients */
- float *weightdenum, /* (o) weighting denumerator
- coefficients */
- float *lsfdeq, /* (i) dequantized lsf coefficients */
- int length, /* (i) length of lsf coefficient vector */
- iLBC_Dec_Inst_t *iLBCdec_inst
- /* (i) the decoder state structure */
- );
-
- #endif
-
-
-
-
-
-
-
-
-
-
-Andersen, et al. Experimental [Page 163]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
-A.36. LPCdecode.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- LPC_decode.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include <math.h>
- #include <string.h>
-
- #include "helpfun.h"
- #include "lsf.h"
- #include "iLBC_define.h"
- #include "constants.h"
-
- /*---------------------------------------------------------------*
- * interpolation of lsf coefficients for the decoder
- *--------------------------------------------------------------*/
-
- void LSFinterpolate2a_dec(
- float *a, /* (o) lpc coefficients for a sub-frame */
- float *lsf1, /* (i) first lsf coefficient vector */
- float *lsf2, /* (i) second lsf coefficient vector */
- float coef, /* (i) interpolation weight */
- int length /* (i) length of lsf vectors */
- ){
- float lsftmp[LPC_FILTERORDER];
-
- interpolate(lsftmp, lsf1, lsf2, coef, length);
- lsf2a(a, lsftmp);
- }
-
- /*---------------------------------------------------------------*
- * obtain dequantized lsf coefficients from quantization index
- *--------------------------------------------------------------*/
-
- void SimplelsfDEQ(
- float *lsfdeq, /* (o) dequantized lsf coefficients */
- int *index, /* (i) quantization index */
- int lpc_n /* (i) number of LPCs */
- ){
- int i, j, pos, cb_pos;
-
-
-
-Andersen, et al. Experimental [Page 164]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- /* decode first LSF */
-
- pos = 0;
- cb_pos = 0;
- for (i = 0; i < LSF_NSPLIT; i++) {
- for (j = 0; j < dim_lsfCbTbl[i]; j++) {
- lsfdeq[pos + j] = lsfCbTbl[cb_pos +
- (long)(index[i])*dim_lsfCbTbl[i] + j];
- }
- pos += dim_lsfCbTbl[i];
- cb_pos += size_lsfCbTbl[i]*dim_lsfCbTbl[i];
- }
-
- if (lpc_n>1) {
-
- /* decode last LSF */
-
- pos = 0;
- cb_pos = 0;
- for (i = 0; i < LSF_NSPLIT; i++) {
- for (j = 0; j < dim_lsfCbTbl[i]; j++) {
- lsfdeq[LPC_FILTERORDER + pos + j] =
- lsfCbTbl[cb_pos +
- (long)(index[LSF_NSPLIT + i])*
- dim_lsfCbTbl[i] + j];
- }
- pos += dim_lsfCbTbl[i];
- cb_pos += size_lsfCbTbl[i]*dim_lsfCbTbl[i];
- }
- }
- }
-
- /*----------------------------------------------------------------*
- * obtain synthesis and weighting filters form lsf coefficients
- *---------------------------------------------------------------*/
-
- void DecoderInterpolateLSF(
- float *syntdenum, /* (o) synthesis filter coefficients */
- float *weightdenum, /* (o) weighting denumerator
- coefficients */
- float *lsfdeq, /* (i) dequantized lsf coefficients */
- int length, /* (i) length of lsf coefficient vector */
- iLBC_Dec_Inst_t *iLBCdec_inst
- /* (i) the decoder state structure */
- ){
- int i, pos, lp_length;
- float lp[LPC_FILTERORDER + 1], *lsfdeq2;
-
-
-
-
-Andersen, et al. Experimental [Page 165]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- lsfdeq2 = lsfdeq + length;
- lp_length = length + 1;
-
- if (iLBCdec_inst->mode==30) {
- /* sub-frame 1: Interpolation between old and first */
-
- LSFinterpolate2a_dec(lp, iLBCdec_inst->lsfdeqold, lsfdeq,
- lsf_weightTbl_30ms[0], length);
- memcpy(syntdenum,lp,lp_length*sizeof(float));
- bwexpand(weightdenum, lp, LPC_CHIRP_WEIGHTDENUM,
- lp_length);
-
- /* sub-frames 2 to 6: interpolation between first
- and last LSF */
-
- pos = lp_length;
- for (i = 1; i < 6; i++) {
- LSFinterpolate2a_dec(lp, lsfdeq, lsfdeq2,
- lsf_weightTbl_30ms[i], length);
- memcpy(syntdenum + pos,lp,lp_length*sizeof(float));
- bwexpand(weightdenum + pos, lp,
- LPC_CHIRP_WEIGHTDENUM, lp_length);
- pos += lp_length;
- }
- }
- else {
- pos = 0;
- for (i = 0; i < iLBCdec_inst->nsub; i++) {
- LSFinterpolate2a_dec(lp, iLBCdec_inst->lsfdeqold,
- lsfdeq, lsf_weightTbl_20ms[i], length);
- memcpy(syntdenum+pos,lp,lp_length*sizeof(float));
- bwexpand(weightdenum+pos, lp, LPC_CHIRP_WEIGHTDENUM,
- lp_length);
- pos += lp_length;
- }
- }
-
- /* update memory */
-
- if (iLBCdec_inst->mode==30)
- memcpy(iLBCdec_inst->lsfdeqold, lsfdeq2,
- length*sizeof(float));
- else
- memcpy(iLBCdec_inst->lsfdeqold, lsfdeq,
- length*sizeof(float));
-
- }
-
-
-
-
-Andersen, et al. Experimental [Page 166]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
-A.37. LPCencode.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- LPCencode.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_LPCENCOD_H
- #define __iLBC_LPCENCOD_H
-
- void LPCencode(
- float *syntdenum, /* (i/o) synthesis filter coefficients
- before/after encoding */
- float *weightdenum, /* (i/o) weighting denumerator coefficients
- before/after encoding */
- int *lsf_index, /* (o) lsf quantization index */
- float *data, /* (i) lsf coefficients to quantize */
- iLBC_Enc_Inst_t *iLBCenc_inst
- /* (i/o) the encoder state structure */
- );
-
- #endif
-
-A.38. LPCencode.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- LPCencode.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include <string.h>
-
- #include "iLBC_define.h"
- #include "helpfun.h"
- #include "lsf.h"
- #include "constants.h"
-
-
-
-Andersen, et al. Experimental [Page 167]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- /*----------------------------------------------------------------*
- * lpc analysis (subrutine to LPCencode)
- *---------------------------------------------------------------*/
-
- void SimpleAnalysis(
- float *lsf, /* (o) lsf coefficients */
- float *data, /* (i) new data vector */
- iLBC_Enc_Inst_t *iLBCenc_inst
- /* (i/o) the encoder state structure */
- ){
- int k, is;
- float temp[BLOCKL_MAX], lp[LPC_FILTERORDER + 1];
- float lp2[LPC_FILTERORDER + 1];
- float r[LPC_FILTERORDER + 1];
-
- is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl;
- memcpy(iLBCenc_inst->lpc_buffer+is,data,
- iLBCenc_inst->blockl*sizeof(float));
-
- /* No lookahead, last window is asymmetric */
-
- for (k = 0; k < iLBCenc_inst->lpc_n; k++) {
-
- is = LPC_LOOKBACK;
-
- if (k < (iLBCenc_inst->lpc_n - 1)) {
- window(temp, lpc_winTbl,
- iLBCenc_inst->lpc_buffer, BLOCKL_MAX);
- } else {
- window(temp, lpc_asymwinTbl,
- iLBCenc_inst->lpc_buffer + is, BLOCKL_MAX);
- }
-
- autocorr(r, temp, BLOCKL_MAX, LPC_FILTERORDER);
- window(r, r, lpc_lagwinTbl, LPC_FILTERORDER + 1);
-
- levdurb(lp, temp, r, LPC_FILTERORDER);
- bwexpand(lp2, lp, LPC_CHIRP_SYNTDENUM, LPC_FILTERORDER+1);
-
- a2lsf(lsf + k*LPC_FILTERORDER, lp2);
- }
- is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl;
- memmove(iLBCenc_inst->lpc_buffer,
- iLBCenc_inst->lpc_buffer+LPC_LOOKBACK+BLOCKL_MAX-is,
- is*sizeof(float));
- }
-
- /*----------------------------------------------------------------*
-
-
-
-Andersen, et al. Experimental [Page 168]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- * lsf interpolator and conversion from lsf to a coefficients
- * (subrutine to SimpleInterpolateLSF)
- *---------------------------------------------------------------*/
-
- void LSFinterpolate2a_enc(
- float *a, /* (o) lpc coefficients */
- float *lsf1,/* (i) first set of lsf coefficients */
- float *lsf2,/* (i) second set of lsf coefficients */
- float coef, /* (i) weighting coefficient to use between
- lsf1 and lsf2 */
- long length /* (i) length of coefficient vectors */
- ){
- float lsftmp[LPC_FILTERORDER];
-
- interpolate(lsftmp, lsf1, lsf2, coef, length);
- lsf2a(a, lsftmp);
- }
-
- /*----------------------------------------------------------------*
- * lsf interpolator (subrutine to LPCencode)
- *---------------------------------------------------------------*/
-
- void SimpleInterpolateLSF(
- float *syntdenum, /* (o) the synthesis filter denominator
- resulting from the quantized
- interpolated lsf */
- float *weightdenum, /* (o) the weighting filter denominator
- resulting from the unquantized
- interpolated lsf */
- float *lsf, /* (i) the unquantized lsf coefficients */
- float *lsfdeq, /* (i) the dequantized lsf coefficients */
- float *lsfold, /* (i) the unquantized lsf coefficients of
- the previous signal frame */
- float *lsfdeqold, /* (i) the dequantized lsf coefficients of
- the previous signal frame */
- int length, /* (i) should equate LPC_FILTERORDER */
- iLBC_Enc_Inst_t *iLBCenc_inst
- /* (i/o) the encoder state structure */
- ){
- int i, pos, lp_length;
- float lp[LPC_FILTERORDER + 1], *lsf2, *lsfdeq2;
-
- lsf2 = lsf + length;
- lsfdeq2 = lsfdeq + length;
- lp_length = length + 1;
-
- if (iLBCenc_inst->mode==30) {
- /* sub-frame 1: Interpolation between old and first
-
-
-
-Andersen, et al. Experimental [Page 169]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- set of lsf coefficients */
-
- LSFinterpolate2a_enc(lp, lsfdeqold, lsfdeq,
- lsf_weightTbl_30ms[0], length);
- memcpy(syntdenum,lp,lp_length*sizeof(float));
- LSFinterpolate2a_enc(lp, lsfold, lsf,
- lsf_weightTbl_30ms[0], length);
- bwexpand(weightdenum, lp, LPC_CHIRP_WEIGHTDENUM, lp_length);
-
- /* sub-frame 2 to 6: Interpolation between first
- and second set of lsf coefficients */
-
- pos = lp_length;
- for (i = 1; i < iLBCenc_inst->nsub; i++) {
- LSFinterpolate2a_enc(lp, lsfdeq, lsfdeq2,
- lsf_weightTbl_30ms[i], length);
- memcpy(syntdenum + pos,lp,lp_length*sizeof(float));
-
- LSFinterpolate2a_enc(lp, lsf, lsf2,
- lsf_weightTbl_30ms[i], length);
- bwexpand(weightdenum + pos, lp,
- LPC_CHIRP_WEIGHTDENUM, lp_length);
- pos += lp_length;
- }
- }
- else {
- pos = 0;
- for (i = 0; i < iLBCenc_inst->nsub; i++) {
- LSFinterpolate2a_enc(lp, lsfdeqold, lsfdeq,
- lsf_weightTbl_20ms[i], length);
- memcpy(syntdenum+pos,lp,lp_length*sizeof(float));
- LSFinterpolate2a_enc(lp, lsfold, lsf,
- lsf_weightTbl_20ms[i], length);
- bwexpand(weightdenum+pos, lp,
- LPC_CHIRP_WEIGHTDENUM, lp_length);
- pos += lp_length;
- }
- }
-
- /* update memory */
-
- if (iLBCenc_inst->mode==30) {
- memcpy(lsfold, lsf2, length*sizeof(float));
- memcpy(lsfdeqold, lsfdeq2, length*sizeof(float));
- }
- else {
- memcpy(lsfold, lsf, length*sizeof(float));
- memcpy(lsfdeqold, lsfdeq, length*sizeof(float));
-
-
-
-Andersen, et al. Experimental [Page 170]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- }
- }
-
- /*----------------------------------------------------------------*
- * lsf quantizer (subrutine to LPCencode)
- *---------------------------------------------------------------*/
-
- void SimplelsfQ(
- float *lsfdeq, /* (o) dequantized lsf coefficients
- (dimension FILTERORDER) */
- int *index, /* (o) quantization index */
- float *lsf, /* (i) the lsf coefficient vector to be
- quantized (dimension FILTERORDER ) */
- int lpc_n /* (i) number of lsf sets to quantize */
- ){
- /* Quantize first LSF with memoryless split VQ */
- SplitVQ(lsfdeq, index, lsf, lsfCbTbl, LSF_NSPLIT,
- dim_lsfCbTbl, size_lsfCbTbl);
-
- if (lpc_n==2) {
- /* Quantize second LSF with memoryless split VQ */
- SplitVQ(lsfdeq + LPC_FILTERORDER, index + LSF_NSPLIT,
- lsf + LPC_FILTERORDER, lsfCbTbl, LSF_NSPLIT,
- dim_lsfCbTbl, size_lsfCbTbl);
- }
- }
-
- /*----------------------------------------------------------------*
- * lpc encoder
- *---------------------------------------------------------------*/
-
- void LPCencode(
- float *syntdenum, /* (i/o) synthesis filter coefficients
- before/after encoding */
- float *weightdenum, /* (i/o) weighting denumerator
- coefficients before/after
- encoding */
- int *lsf_index, /* (o) lsf quantization index */
- float *data, /* (i) lsf coefficients to quantize */
- iLBC_Enc_Inst_t *iLBCenc_inst
- /* (i/o) the encoder state structure */
- ){
- float lsf[LPC_FILTERORDER * LPC_N_MAX];
- float lsfdeq[LPC_FILTERORDER * LPC_N_MAX];
- int change=0;
-
- SimpleAnalysis(lsf, data, iLBCenc_inst);
- SimplelsfQ(lsfdeq, lsf_index, lsf, iLBCenc_inst->lpc_n);
-
-
-
-Andersen, et al. Experimental [Page 171]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- change=LSF_check(lsfdeq, LPC_FILTERORDER, iLBCenc_inst->lpc_n);
- SimpleInterpolateLSF(syntdenum, weightdenum,
- lsf, lsfdeq, iLBCenc_inst->lsfold,
- iLBCenc_inst->lsfdeqold, LPC_FILTERORDER, iLBCenc_inst);
- }
-
-A.39. lsf.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- lsf.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_LSF_H
- #define __iLBC_LSF_H
-
- void a2lsf(
- float *freq,/* (o) lsf coefficients */
- float *a /* (i) lpc coefficients */
- );
-
- void lsf2a(
- float *a_coef, /* (o) lpc coefficients */
- float *freq /* (i) lsf coefficients */
- );
-
- #endif
-
-A.40. lsf.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- lsf.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include <string.h>
-
-
-
-Andersen, et al. Experimental [Page 172]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- #include <math.h>
-
- #include "iLBC_define.h"
-
- /*----------------------------------------------------------------*
- * conversion from lpc coefficients to lsf coefficients
- *---------------------------------------------------------------*/
-
- void a2lsf(
- float *freq,/* (o) lsf coefficients */
- float *a /* (i) lpc coefficients */
- ){
- float steps[LSF_NUMBER_OF_STEPS] =
- {(float)0.00635, (float)0.003175, (float)0.0015875,
- (float)0.00079375};
- float step;
- int step_idx;
- int lsp_index;
- float p[LPC_HALFORDER];
- float q[LPC_HALFORDER];
- float p_pre[LPC_HALFORDER];
- float q_pre[LPC_HALFORDER];
- float old_p, old_q, *old;
- float *pq_coef;
- float omega, old_omega;
- int i;
- float hlp, hlp1, hlp2, hlp3, hlp4, hlp5;
-
- for (i=0; i<LPC_HALFORDER; i++) {
- p[i] = (float)-1.0 * (a[i + 1] + a[LPC_FILTERORDER - i]);
- q[i] = a[LPC_FILTERORDER - i] - a[i + 1];
- }
-
- p_pre[0] = (float)-1.0 - p[0];
- p_pre[1] = - p_pre[0] - p[1];
- p_pre[2] = - p_pre[1] - p[2];
- p_pre[3] = - p_pre[2] - p[3];
- p_pre[4] = - p_pre[3] - p[4];
- p_pre[4] = p_pre[4] / 2;
-
- q_pre[0] = (float)1.0 - q[0];
- q_pre[1] = q_pre[0] - q[1];
- q_pre[2] = q_pre[1] - q[2];
- q_pre[3] = q_pre[2] - q[3];
- q_pre[4] = q_pre[3] - q[4];
- q_pre[4] = q_pre[4] / 2;
-
- omega = 0.0;
-
-
-
-Andersen, et al. Experimental [Page 173]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- old_omega = 0.0;
-
- old_p = FLOAT_MAX;
- old_q = FLOAT_MAX;
-
- /* Here we loop through lsp_index to find all the
- LPC_FILTERORDER roots for omega. */
-
- for (lsp_index = 0; lsp_index<LPC_FILTERORDER; lsp_index++) {
-
- /* Depending on lsp_index being even or odd, we
- alternatively solve the roots for the two LSP equations. */
-
-
- if ((lsp_index & 0x1) == 0) {
- pq_coef = p_pre;
- old = &old_p;
- } else {
- pq_coef = q_pre;
- old = &old_q;
- }
-
- /* Start with low resolution grid */
-
- for (step_idx = 0, step = steps[step_idx];
- step_idx < LSF_NUMBER_OF_STEPS;){
-
- /* cos(10piw) + pq(0)cos(8piw) + pq(1)cos(6piw) +
- pq(2)cos(4piw) + pq(3)cod(2piw) + pq(4) */
-
- hlp = (float)cos(omega * TWO_PI);
- hlp1 = (float)2.0 * hlp + pq_coef[0];
- hlp2 = (float)2.0 * hlp * hlp1 - (float)1.0 +
- pq_coef[1];
- hlp3 = (float)2.0 * hlp * hlp2 - hlp1 + pq_coef[2];
- hlp4 = (float)2.0 * hlp * hlp3 - hlp2 + pq_coef[3];
- hlp5 = hlp * hlp4 - hlp3 + pq_coef[4];
-
-
- if (((hlp5 * (*old)) <= 0.0) || (omega >= 0.5)){
-
- if (step_idx == (LSF_NUMBER_OF_STEPS - 1)){
-
- if (fabs(hlp5) >= fabs(*old)) {
- freq[lsp_index] = omega - step;
- } else {
- freq[lsp_index] = omega;
- }
-
-
-
-Andersen, et al. Experimental [Page 174]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
-
-
- if ((*old) >= 0.0){
- *old = (float)-1.0 * FLOAT_MAX;
- } else {
- *old = FLOAT_MAX;
- }
-
- omega = old_omega;
- step_idx = 0;
-
- step_idx = LSF_NUMBER_OF_STEPS;
- } else {
-
- if (step_idx == 0) {
- old_omega = omega;
- }
-
- step_idx++;
- omega -= steps[step_idx];
-
- /* Go back one grid step */
-
- step = steps[step_idx];
- }
- } else {
-
- /* increment omega until they are of different sign,
- and we know there is at least one root between omega
- and old_omega */
- *old = hlp5;
- omega += step;
- }
- }
- }
-
- for (i = 0; i<LPC_FILTERORDER; i++) {
- freq[i] = freq[i] * TWO_PI;
- }
- }
-
- /*----------------------------------------------------------------*
- * conversion from lsf coefficients to lpc coefficients
- *---------------------------------------------------------------*/
-
- void lsf2a(
- float *a_coef, /* (o) lpc coefficients */
- float *freq /* (i) lsf coefficients */
-
-
-
-Andersen, et al. Experimental [Page 175]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- ){
- int i, j;
- float hlp;
- float p[LPC_HALFORDER], q[LPC_HALFORDER];
- float a[LPC_HALFORDER + 1], a1[LPC_HALFORDER],
- a2[LPC_HALFORDER];
- float b[LPC_HALFORDER + 1], b1[LPC_HALFORDER],
- b2[LPC_HALFORDER];
-
- for (i=0; i<LPC_FILTERORDER; i++) {
- freq[i] = freq[i] * PI2;
- }
-
- /* Check input for ill-conditioned cases. This part is not
- found in the TIA standard. It involves the following 2 IF
- blocks. If "freq" is judged ill-conditioned, then we first
- modify freq[0] and freq[LPC_HALFORDER-1] (normally
- LPC_HALFORDER = 10 for LPC applications), then we adjust
- the other "freq" values slightly */
-
-
- if ((freq[0] <= 0.0) || (freq[LPC_FILTERORDER - 1] >= 0.5)){
-
-
- if (freq[0] <= 0.0) {
- freq[0] = (float)0.022;
- }
-
-
- if (freq[LPC_FILTERORDER - 1] >= 0.5) {
- freq[LPC_FILTERORDER - 1] = (float)0.499;
- }
-
- hlp = (freq[LPC_FILTERORDER - 1] - freq[0]) /
- (float) (LPC_FILTERORDER - 1);
-
- for (i=1; i<LPC_FILTERORDER; i++) {
- freq[i] = freq[i - 1] + hlp;
- }
- }
-
- memset(a1, 0, LPC_HALFORDER*sizeof(float));
- memset(a2, 0, LPC_HALFORDER*sizeof(float));
- memset(b1, 0, LPC_HALFORDER*sizeof(float));
- memset(b2, 0, LPC_HALFORDER*sizeof(float));
- memset(a, 0, (LPC_HALFORDER+1)*sizeof(float));
- memset(b, 0, (LPC_HALFORDER+1)*sizeof(float));
-
-
-
-
-Andersen, et al. Experimental [Page 176]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- /* p[i] and q[i] compute cos(2*pi*omega_{2j}) and
- cos(2*pi*omega_{2j-1} in eqs. 4.2.2.2-1 and 4.2.2.2-2.
- Note that for this code p[i] specifies the coefficients
- used in .Q_A(z) while q[i] specifies the coefficients used
- in .P_A(z) */
-
- for (i=0; i<LPC_HALFORDER; i++) {
- p[i] = (float)cos(TWO_PI * freq[2 * i]);
- q[i] = (float)cos(TWO_PI * freq[2 * i + 1]);
- }
-
- a[0] = 0.25;
- b[0] = 0.25;
-
- for (i= 0; i<LPC_HALFORDER; i++) {
- a[i + 1] = a[i] - 2 * p[i] * a1[i] + a2[i];
- b[i + 1] = b[i] - 2 * q[i] * b1[i] + b2[i];
- a2[i] = a1[i];
- a1[i] = a[i];
- b2[i] = b1[i];
- b1[i] = b[i];
- }
-
- for (j=0; j<LPC_FILTERORDER; j++) {
-
- if (j == 0) {
- a[0] = 0.25;
- b[0] = -0.25;
- } else {
- a[0] = b[0] = 0.0;
- }
-
- for (i=0; i<LPC_HALFORDER; i++) {
- a[i + 1] = a[i] - 2 * p[i] * a1[i] + a2[i];
- b[i + 1] = b[i] - 2 * q[i] * b1[i] + b2[i];
- a2[i] = a1[i];
- a1[i] = a[i];
- b2[i] = b1[i];
- b1[i] = b[i];
- }
-
- a_coef[j + 1] = 2 * (a[LPC_HALFORDER] + b[LPC_HALFORDER]);
- }
-
- a_coef[0] = 1.0;
- }
-
-
-
-
-
-Andersen, et al. Experimental [Page 177]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
-A.41. packing.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- packing.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __PACKING_H
- #define __PACKING_H
-
- void packsplit(
- int *index, /* (i) the value to split */
- int *firstpart, /* (o) the value specified by most
- significant bits */
- int *rest, /* (o) the value specified by least
- significant bits */
- int bitno_firstpart, /* (i) number of bits in most
- significant part */
- int bitno_total /* (i) number of bits in full range
- of value */
- );
-
- void packcombine(
- int *index, /* (i/o) the msb value in the
- combined value out */
- int rest, /* (i) the lsb value */
- int bitno_rest /* (i) the number of bits in the
- lsb part */
- );
-
- void dopack(
- unsigned char **bitstream, /* (i/o) on entrance pointer to
- place in bitstream to pack
- new data, on exit pointer
- to place in bitstream to
- pack future data */
- int index, /* (i) the value to pack */
- int bitno, /* (i) the number of bits that the
- value will fit within */
- int *pos /* (i/o) write position in the
- current byte */
- );
-
-
-
-Andersen, et al. Experimental [Page 178]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- void unpack(
- unsigned char **bitstream, /* (i/o) on entrance pointer to
- place in bitstream to
- unpack new data from, on
- exit pointer to place in
- bitstream to unpack future
- data from */
- int *index, /* (o) resulting value */
- int bitno, /* (i) number of bits used to
- represent the value */
- int *pos /* (i/o) read position in the
- current byte */
- );
-
- #endif
-
-A.42. packing.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- packing.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include <math.h>
- #include <stdlib.h>
-
- #include "iLBC_define.h"
- #include "constants.h"
- #include "helpfun.h"
- #include "string.h"
-
- /*----------------------------------------------------------------*
- * splitting an integer into first most significant bits and
- * remaining least significant bits
- *---------------------------------------------------------------*/
-
- void packsplit(
- int *index, /* (i) the value to split */
- int *firstpart, /* (o) the value specified by most
- significant bits */
- int *rest, /* (o) the value specified by least
- significant bits */
-
-
-
-Andersen, et al. Experimental [Page 179]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- int bitno_firstpart, /* (i) number of bits in most
- significant part */
- int bitno_total /* (i) number of bits in full range
- of value */
- ){
- int bitno_rest = bitno_total-bitno_firstpart;
-
- *firstpart = *index>>(bitno_rest);
- *rest = *index-(*firstpart<<(bitno_rest));
- }
-
- /*----------------------------------------------------------------*
- * combining a value corresponding to msb's with a value
- * corresponding to lsb's
- *---------------------------------------------------------------*/
-
- void packcombine(
- int *index, /* (i/o) the msb value in the
- combined value out */
- int rest, /* (i) the lsb value */
- int bitno_rest /* (i) the number of bits in the
- lsb part */
- ){
- *index = *index<<bitno_rest;
- *index += rest;
- }
-
- /*----------------------------------------------------------------*
- * packing of bits into bitstream, i.e., vector of bytes
- *---------------------------------------------------------------*/
-
- void dopack(
- unsigned char **bitstream, /* (i/o) on entrance pointer to
- place in bitstream to pack
- new data, on exit pointer
- to place in bitstream to
- pack future data */
- int index, /* (i) the value to pack */
- int bitno, /* (i) the number of bits that the
- value will fit within */
- int *pos /* (i/o) write position in the
- current byte */
- ){
- int posLeft;
-
- /* Clear the bits before starting in a new byte */
-
- if ((*pos)==0) {
-
-
-
-Andersen, et al. Experimental [Page 180]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- **bitstream=0;
- }
-
- while (bitno>0) {
-
- /* Jump to the next byte if end of this byte is reached*/
-
- if (*pos==8) {
- *pos=0;
- (*bitstream)++;
- **bitstream=0;
- }
-
- posLeft=8-(*pos);
-
- /* Insert index into the bitstream */
-
- if (bitno <= posLeft) {
- **bitstream |= (unsigned char)(index<<(posLeft-bitno));
- *pos+=bitno;
- bitno=0;
- } else {
- **bitstream |= (unsigned char)(index>>(bitno-posLeft));
-
- *pos=8;
- index-=((index>>(bitno-posLeft))<<(bitno-posLeft));
-
- bitno-=posLeft;
- }
- }
- }
-
- /*----------------------------------------------------------------*
- * unpacking of bits from bitstream, i.e., vector of bytes
- *---------------------------------------------------------------*/
-
- void unpack(
- unsigned char **bitstream, /* (i/o) on entrance pointer to
- place in bitstream to
- unpack new data from, on
- exit pointer to place in
- bitstream to unpack future
- data from */
- int *index, /* (o) resulting value */
- int bitno, /* (i) number of bits used to
- represent the value */
- int *pos /* (i/o) read position in the
- current byte */
-
-
-
-Andersen, et al. Experimental [Page 181]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- ){
- int BitsLeft;
-
- *index=0;
-
- while (bitno>0) {
-
- /* move forward in bitstream when the end of the
- byte is reached */
-
- if (*pos==8) {
- *pos=0;
- (*bitstream)++;
- }
-
- BitsLeft=8-(*pos);
-
- /* Extract bits to index */
-
- if (BitsLeft>=bitno) {
- *index+=((((**bitstream)<<(*pos)) & 0xFF)>>(8-bitno));
-
- *pos+=bitno;
- bitno=0;
- } else {
-
- if ((8-bitno)>0) {
- *index+=((((**bitstream)<<(*pos)) & 0xFF)>>
- (8-bitno));
- *pos=8;
- } else {
- *index+=(((int)(((**bitstream)<<(*pos)) & 0xFF))<<
- (bitno-8));
- *pos=8;
- }
- bitno-=BitsLeft;
- }
- }
- }
-
-A.43. StateConstructW.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- StateConstructW.h
-
-
-
-
-Andersen, et al. Experimental [Page 182]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_STATECONSTRUCTW_H
- #define __iLBC_STATECONSTRUCTW_H
-
- void StateConstructW(
- int idxForMax, /* (i) 6-bit index for the quantization of
- max amplitude */
- int *idxVec, /* (i) vector of quantization indexes */
- float *syntDenum, /* (i) synthesis filter denumerator */
- float *out, /* (o) the decoded state vector */
- int len /* (i) length of a state vector */
- );
-
- #endif
-
-A.44. StateConstructW.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- StateConstructW.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include <math.h>
- #include <string.h>
-
- #include "iLBC_define.h"
- #include "constants.h"
- #include "filter.h"
-
- /*----------------------------------------------------------------*
- * decoding of the start state
- *---------------------------------------------------------------*/
-
- void StateConstructW(
- int idxForMax, /* (i) 6-bit index for the quantization of
- max amplitude */
- int *idxVec, /* (i) vector of quantization indexes */
- float *syntDenum, /* (i) synthesis filter denumerator */
-
-
-
-Andersen, et al. Experimental [Page 183]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- float *out, /* (o) the decoded state vector */
- int len /* (i) length of a state vector */
- ){
- float maxVal, tmpbuf[LPC_FILTERORDER+2*STATE_LEN], *tmp,
- numerator[LPC_FILTERORDER+1];
- float foutbuf[LPC_FILTERORDER+2*STATE_LEN], *fout;
- int k,tmpi;
-
- /* decoding of the maximum value */
-
- maxVal = state_frgqTbl[idxForMax];
- maxVal = (float)pow(10,maxVal)/(float)4.5;
-
- /* initialization of buffers and coefficients */
-
- memset(tmpbuf, 0, LPC_FILTERORDER*sizeof(float));
- memset(foutbuf, 0, LPC_FILTERORDER*sizeof(float));
- for (k=0; k<LPC_FILTERORDER; k++) {
- numerator[k]=syntDenum[LPC_FILTERORDER-k];
- }
- numerator[LPC_FILTERORDER]=syntDenum[0];
- tmp = &tmpbuf[LPC_FILTERORDER];
- fout = &foutbuf[LPC_FILTERORDER];
-
- /* decoding of the sample values */
-
- for (k=0; k<len; k++) {
- tmpi = len-1-k;
- /* maxVal = 1/scal */
- tmp[k] = maxVal*state_sq3Tbl[idxVec[tmpi]];
- }
-
- /* circular convolution with all-pass filter */
-
- memset(tmp+len, 0, len*sizeof(float));
- ZeroPoleFilter(tmp, numerator, syntDenum, 2*len,
- LPC_FILTERORDER, fout);
- for (k=0;k<len;k++) {
- out[k] = fout[len-1-k]+fout[2*len-1-k];
- }
- }
-
-
-
-
-
-
-
-
-
-
-Andersen, et al. Experimental [Page 184]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
-A.45. StateSearchW.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- StateSearchW.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_STATESEARCHW_H
- #define __iLBC_STATESEARCHW_H
-
- void AbsQuantW(
- iLBC_Enc_Inst_t *iLBCenc_inst,
- /* (i) Encoder instance */
- float *in, /* (i) vector to encode */
- float *syntDenum, /* (i) denominator of synthesis filter */
- float *weightDenum, /* (i) denominator of weighting filter */
- int *out, /* (o) vector of quantizer indexes */
- int len, /* (i) length of vector to encode and
- vector of quantizer indexes */
- int state_first /* (i) position of start state in the
- 80 vec */
- );
-
- void StateSearchW(
- iLBC_Enc_Inst_t *iLBCenc_inst,
- /* (i) Encoder instance */
- float *residual,/* (i) target residual vector */
- float *syntDenum, /* (i) lpc synthesis filter */
- float *weightDenum, /* (i) weighting filter denuminator */
- int *idxForMax, /* (o) quantizer index for maximum
- amplitude */
- int *idxVec, /* (o) vector of quantization indexes */
- int len, /* (i) length of all vectors */
- int state_first /* (i) position of start state in the
- 80 vec */
- );
-
-
- #endif
-
-
-
-
-
-
-Andersen, et al. Experimental [Page 185]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
-A.46. StateSearchW.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- StateSearchW.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include <math.h>
- #include <string.h>
-
- #include "iLBC_define.h"
- #include "constants.h"
- #include "filter.h"
- #include "helpfun.h"
-
- /*----------------------------------------------------------------*
- * predictive noise shaping encoding of scaled start state
- * (subrutine for StateSearchW)
- *---------------------------------------------------------------*/
-
- void AbsQuantW(
- iLBC_Enc_Inst_t *iLBCenc_inst,
- /* (i) Encoder instance */
- float *in, /* (i) vector to encode */
- float *syntDenum, /* (i) denominator of synthesis filter */
- float *weightDenum, /* (i) denominator of weighting filter */
- int *out, /* (o) vector of quantizer indexes */
- int len, /* (i) length of vector to encode and
- vector of quantizer indexes */
- int state_first /* (i) position of start state in the
- 80 vec */
- ){
- float *syntOut;
- float syntOutBuf[LPC_FILTERORDER+STATE_SHORT_LEN_30MS];
- float toQ, xq;
- int n;
- int index;
-
- /* initialization of buffer for filtering */
-
- memset(syntOutBuf, 0, LPC_FILTERORDER*sizeof(float));
-
-
-
-
-Andersen, et al. Experimental [Page 186]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- /* initialization of pointer for filtering */
-
- syntOut = &syntOutBuf[LPC_FILTERORDER];
-
- /* synthesis and weighting filters on input */
-
- if (state_first) {
- AllPoleFilter (in, weightDenum, SUBL, LPC_FILTERORDER);
- } else {
- AllPoleFilter (in, weightDenum,
- iLBCenc_inst->state_short_len-SUBL,
- LPC_FILTERORDER);
- }
-
- /* encoding loop */
-
- for (n=0; n<len; n++) {
-
- /* time update of filter coefficients */
-
- if ((state_first)&&(n==SUBL)){
- syntDenum += (LPC_FILTERORDER+1);
- weightDenum += (LPC_FILTERORDER+1);
-
- /* synthesis and weighting filters on input */
- AllPoleFilter (&in[n], weightDenum, len-n,
- LPC_FILTERORDER);
-
- } else if ((state_first==0)&&
- (n==(iLBCenc_inst->state_short_len-SUBL))) {
- syntDenum += (LPC_FILTERORDER+1);
- weightDenum += (LPC_FILTERORDER+1);
-
- /* synthesis and weighting filters on input */
- AllPoleFilter (&in[n], weightDenum, len-n,
- LPC_FILTERORDER);
-
- }
-
- /* prediction of synthesized and weighted input */
-
- syntOut[n] = 0.0;
- AllPoleFilter (&syntOut[n], weightDenum, 1,
- LPC_FILTERORDER);
-
- /* quantization */
-
- toQ = in[n]-syntOut[n];
-
-
-
-Andersen, et al. Experimental [Page 187]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- sort_sq(&xq, &index, toQ, state_sq3Tbl, 8);
- out[n]=index;
- syntOut[n] = state_sq3Tbl[out[n]];
-
- /* update of the prediction filter */
-
- AllPoleFilter(&syntOut[n], weightDenum, 1,
- LPC_FILTERORDER);
- }
- }
-
- /*----------------------------------------------------------------*
- * encoding of start state
- *---------------------------------------------------------------*/
-
- void StateSearchW(
- iLBC_Enc_Inst_t *iLBCenc_inst,
- /* (i) Encoder instance */
- float *residual,/* (i) target residual vector */
- float *syntDenum, /* (i) lpc synthesis filter */
- float *weightDenum, /* (i) weighting filter denuminator */
- int *idxForMax, /* (o) quantizer index for maximum
- amplitude */
- int *idxVec, /* (o) vector of quantization indexes */
- int len, /* (i) length of all vectors */
- int state_first /* (i) position of start state in the
- 80 vec */
- ){
- float dtmp, maxVal;
- float tmpbuf[LPC_FILTERORDER+2*STATE_SHORT_LEN_30MS];
- float *tmp, numerator[1+LPC_FILTERORDER];
- float foutbuf[LPC_FILTERORDER+2*STATE_SHORT_LEN_30MS], *fout;
- int k;
- float qmax, scal;
-
- /* initialization of buffers and filter coefficients */
-
- memset(tmpbuf, 0, LPC_FILTERORDER*sizeof(float));
- memset(foutbuf, 0, LPC_FILTERORDER*sizeof(float));
- for (k=0; k<LPC_FILTERORDER; k++) {
- numerator[k]=syntDenum[LPC_FILTERORDER-k];
- }
- numerator[LPC_FILTERORDER]=syntDenum[0];
- tmp = &tmpbuf[LPC_FILTERORDER];
- fout = &foutbuf[LPC_FILTERORDER];
-
- /* circular convolution with the all-pass filter */
-
-
-
-
-Andersen, et al. Experimental [Page 188]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- memcpy(tmp, residual, len*sizeof(float));
- memset(tmp+len, 0, len*sizeof(float));
- ZeroPoleFilter(tmp, numerator, syntDenum, 2*len,
- LPC_FILTERORDER, fout);
- for (k=0; k<len; k++) {
- fout[k] += fout[k+len];
- }
-
- /* identification of the maximum amplitude value */
-
- maxVal = fout[0];
- for (k=1; k<len; k++) {
-
- if (fout[k]*fout[k] > maxVal*maxVal){
- maxVal = fout[k];
- }
- }
- maxVal=(float)fabs(maxVal);
-
- /* encoding of the maximum amplitude value */
-
- if (maxVal < 10.0) {
- maxVal = 10.0;
- }
- maxVal = (float)log10(maxVal);
- sort_sq(&dtmp, idxForMax, maxVal, state_frgqTbl, 64);
-
- /* decoding of the maximum amplitude representation value,
- and corresponding scaling of start state */
-
- maxVal=state_frgqTbl[*idxForMax];
- qmax = (float)pow(10,maxVal);
- scal = (float)(4.5)/qmax;
- for (k=0; k<len; k++){
- fout[k] *= scal;
- }
-
- /* predictive noise shaping encoding of scaled start state */
-
- AbsQuantW(iLBCenc_inst, fout,syntDenum,
- weightDenum,idxVec, len, state_first);
- }
-
-
-
-
-
-
-
-
-
-Andersen, et al. Experimental [Page 189]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
-A.47. syntFilter.h
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- syntFilter.h
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #ifndef __iLBC_SYNTFILTER_H
- #define __iLBC_SYNTFILTER_H
-
- void syntFilter(
- float *Out, /* (i/o) Signal to be filtered */
- float *a, /* (i) LP parameters */
- int len, /* (i) Length of signal */
- float *mem /* (i/o) Filter state */
- );
-
- #endif
-
-A.48. syntFilter.c
-
- /******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- syntFilter.c
-
- Copyright (C) The Internet Society (2004).
- All Rights Reserved.
-
- ******************************************************************/
-
- #include "iLBC_define.h"
-
- /*----------------------------------------------------------------*
- * LP synthesis filter.
- *---------------------------------------------------------------*/
-
- void syntFilter(
- float *Out, /* (i/o) Signal to be filtered */
- float *a, /* (i) LP parameters */
- int len, /* (i) Length of signal */
-
-
-
-Andersen, et al. Experimental [Page 190]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- float *mem /* (i/o) Filter state */
- ){
- int i, j;
- float *po, *pi, *pa, *pm;
-
- po=Out;
-
- /* Filter first part using memory from past */
-
- for (i=0; i<LPC_FILTERORDER; i++) {
- pi=&Out[i-1];
- pa=&a[1];
- pm=&mem[LPC_FILTERORDER-1];
- for (j=1; j<=i; j++) {
- *po-=(*pa++)*(*pi--);
- }
- for (j=i+1; j<LPC_FILTERORDER+1; j++) {
- *po-=(*pa++)*(*pm--);
- }
- po++;
- }
-
- /* Filter last part where the state is entirely in
- the output vector */
-
- for (i=LPC_FILTERORDER; i<len; i++) {
- pi=&Out[i-1];
- pa=&a[1];
- for (j=1; j<LPC_FILTERORDER+1; j++) {
- *po-=(*pa++)*(*pi--);
- }
- po++;
- }
-
- /* Update state vector */
-
- memcpy(mem, &Out[len-LPC_FILTERORDER],
- LPC_FILTERORDER*sizeof(float));
- }
-
-
-
-
-
-
-
-
-
-
-
-
-Andersen, et al. Experimental [Page 191]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
-Authors' Addresses
-
- Soren Vang Andersen
- Department of Communication Technology
- Aalborg University
- Fredrik Bajers Vej 7A
- 9200 Aalborg
- Denmark
-
- Phone: ++45 9 6358627
- EMail: sva@kom.auc.dk
-
-
- Alan Duric
- Telio AS
- Stoperigt. 2
- Oslo, N-0250
- Norway
-
- Phone: +47 21673555
- EMail: alan.duric@telio.no
-
-
- Henrik Astrom
- Global IP Sound AB
- Olandsgatan 42
- Stockholm, S-11663
- Sweden
-
- Phone: +46 8 54553040
- EMail: henrik.astrom@globalipsound.com
-
-
- Roar Hagen
- Global IP Sound AB
- Olandsgatan 42
- Stockholm, S-11663
- Sweden
-
- Phone: +46 8 54553040
- EMail: roar.hagen@globalipsound.com
-
-
-
-
-
-
-
-
-
-
-Andersen, et al. Experimental [Page 192]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
- W. Bastiaan Kleijn
- Global IP Sound AB
- Olandsgatan 42
- Stockholm, S-11663
- Sweden
-
- Phone: +46 8 54553040
- EMail: bastiaan.kleijn@globalipsound.com
-
-
- Jan Linden
- Global IP Sound Inc.
- 900 Kearny Street, suite 500
- San Francisco, CA-94133
- USA
-
- Phone: +1 415 397 2555
- EMail: jan.linden@globalipsound.com
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-Andersen, et al. Experimental [Page 193]
-
-RFC 3951 Internet Low Bit Rate Codec December 2004
-
-
-Full Copyright Statement
-
- Copyright (C) The Internet Society (2004).
-
- This document is subject to the rights, licenses and restrictions
- contained in BCP 78, and except as set forth therein, the authors
- retain all their rights.
-
- This document and the information contained herein are provided on an
- "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
- OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET
- ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED,
- INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE
- INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
- WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
-
-Intellectual Property
-
- The IETF takes no position regarding the validity or scope of any
- Intellectual Property Rights or other rights that might be claimed to
- pertain to the implementation or use of the technology described in
- this document or the extent to which any license under such rights
- might or might not be available; nor does it represent that it has
- made any independent effort to identify any such rights. Information
- on the IETF's procedures with respect to rights in IETF Documents can
- be found in BCP 78 and BCP 79.
-
- Copies of IPR disclosures made to the IETF Secretariat and any
- assurances of licenses to be made available, or the result of an
- attempt made to obtain a general license or permission for the use of
- such proprietary rights by implementers or users of this
- specification can be obtained from the IETF on-line IPR repository at
- http://www.ietf.org/ipr.
-
- The IETF invites any interested party to bring to its attention any
- copyrights, patents or patent applications, or other proprietary
- rights that may cover technology that may be required to implement
- this standard. Please address the information to the IETF at ietf-
- ipr@ietf.org.
-
-
-Acknowledgement
-
- Funding for the RFC Editor function is currently provided by the
- Internet Society.
-
-
-
-
-
-
-Andersen, et al. Experimental [Page 194]
-
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/documentation/rfc3952.txt b/trunk/src/modules/audio_coding/codecs/ilbc/documentation/rfc3952.txt
deleted file mode 100644
index eade7cd..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/documentation/rfc3952.txt
+++ /dev/null
@@ -1,731 +0,0 @@
-
-
-
-
-
-
-Network Working Group A. Duric
-Request for Comments: 3952 Telio
-Category: Experimental S. Andersen
- Aalborg University
- December 2004
-
-
- Real-time Transport Protocol (RTP) Payload Format
- for internet Low Bit Rate Codec (iLBC) Speech
-
-Status of this Memo
-
- This memo defines an Experimental Protocol for the Internet
- community. It does not specify an Internet standard of any kind.
- Discussion and suggestions for improvement are requested.
- Distribution of this memo is unlimited.
-
-Copyright Notice
-
- Copyright (C) The Internet Society (2004).
-
-Abstract
-
- This document describes the Real-time Transport Protocol (RTP)
- payload format for the internet Low Bit Rate Codec (iLBC) Speech
- developed by Global IP Sound (GIPS). Also, within the document there
- are included necessary details for the use of iLBC with MIME and
- Session Description Protocol (SDP).
-
-Table of Contents
-
- 1. Introduction. . . . . . . . . . . . . . . . . . . . . . . . . . 2
- 2. Background. . . . . . . . . . . . . . . . . . . . . . . . . . . 2
- 3. RTP Payload Format. . . . . . . . . . . . . . . . . . . . . . . 3
- 3.1. Bitstream definition . . . . . . . . . . . . . . . . . . . 3
- 3.2. Multiple iLBC frames in a RTP packet . . . . . . . . . . . 6
- 4. IANA Considerations . . . . . . . . . . . . . . . . . . . . . . 7
- 4.1. Storage Mode . . . . . . . . . . . . . . . . . . . . . . . 7
- 4.2. MIME registration of iLBC. . . . . . . . . . . . . . . . . 8
- 5. Mapping to SDP Parameters . . . . . . . . . . . . . . . . . . . 9
- 6. Security Considerations . . . . . . . . . . . . . . . . . . . . 11
- 7. References. . . . . . . . . . . . . . . . . . . . . . . . . . . 11
- 7.1. Normative References . . . . . . . . . . . . . . . . . . . 11
- 7.2. Informative References . . . . . . . . . . . . . . . . . . 12
- 8. Acknowledgements. . . . . . . . . . . . . . . . . . . . . . . . 12
- Authors' Addresses . . . . . . . . . . . . . . . . . . . . . . . . 12
- Full Copyright Statement . . . . . . . . . . . . . . . . . . . . . 13
-
-
-
-
-Duric & Andersen Experimental [Page 1]
-
-RFC 3952 RTP Payload Format for iLBC Speech December 2004
-
-
-1. Introduction
-
- This document describes how compressed iLBC speech, as produced by
- the iLBC codec [1], may be formatted for use as an RTP payload type.
- Methods are provided to packetize the codec data frames into RTP
- packets. The sender may send one or more codec data frames per
- packet depending on the application scenario or based on the
- transport network condition, bandwidth restriction, delay
- requirements and packet-loss tolerance.
-
- The key words "MUST", "MUST NOT", "REQUIRED", "SHALL", "SHALL NOT",
- "SHOULD", "SHOULD NOT", "RECOMMENDED", "MAY", and "OPTIONAL" in this
- document are to be interpreted as described in BCP 14, RFC 2119 [2].
-
-2. Background
-
- Global IP Sound (GIPS) has developed a speech compression algorithm
- for use in IP based communications [1]. The iLBC codec enables
- graceful speech quality degradation in the case of lost frames, which
- occurs in connection with lost or delayed IP packets.
-
- This codec is suitable for real time communications such as,
- telephony and videoconferencing, streaming audio, archival and
- messaging.
-
- The iLBC codec [1] is an algorithm that compresses each basic frame
- (20 ms or 30 ms) of 8000 Hz, 16-bit sampled input speech, into output
- frames with rate of 400 bits for 30 ms basic frame size and 304 bits
- for 20 ms basic frame size.
-
- The codec supports two basic frame lengths: 30 ms at 13.33 kbit/s and
- 20 ms at 15.2 kbit/s, using a block independent linear-predictive
- coding (LPC) algorithm. When the codec operates at block lengths of
- 20 ms, it produces 304 bits per block which MUST be packetized in 38
- bytes. Similarly, for block lengths of 30 ms it produces 400 bits
- per block which MUST be packetized in 50 bytes. This algorithm
- results in a speech coding system with a controlled response to
- packet losses similar to what is known from pulse code modulation
- (PCM) with a packet loss concealment (PLC), such as ITU-T G711
- standard [7], which operates at a fixed bit rate of 64 kbit/s. At
- the same time, this algorithm enables fixed bit rate coding with a
- quality-versus-bit rate tradeoff close to what is known from code-
- excited linear prediction (CELP).
-
-
-
-
-
-
-
-
-Duric & Andersen Experimental [Page 2]
-
-RFC 3952 RTP Payload Format for iLBC Speech December 2004
-
-
-3. RTP Payload Format
-
- The iLBC codec uses 20 or 30 ms frames and a sampling rate clock of 8
- kHz, so the RTP timestamp MUST be in units of 1/8000 of a second. The
- RTP payload for iLBC has the format shown in the figure bellow. No
- addition header specific to this payload format is required.
-
- This format is intended for the situations where the sender and the
- receiver send one or more codec data frames per packet. The RTP
- packet looks as follows:
-
- 0 1 2 3
- 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | RTP Header [3] |
- +=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+=+
- | |
- + one or more frames of iLBC [1] |
- | |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
- Figure 1, Packet format diagram
-
- The RTP header of the packetized encoded iLBC speech has the expected
- values as described in [3]. The usage of M bit SHOULD be as
- specified in the applicable RTP profile, for example, RFC 3551 [4]
- specifies that if the sender does not suppress silence (i.e., sends a
- frame on every frame interval), the M bit will always be zero. When
- more then one codec data frame is present in a single RTP packet, the
- timestamp is, as always, the oldest data frame represented in the RTP
- packet.
-
- The assignment of an RTP payload type for this new packet format is
- outside the scope of this document, and will not be specified here.
- It is expected that the RTP profile for a particular class of
- applications will assign a payload type for this encoding, or if that
- is not done, then a payload type in the dynamic range shall be chosen
- by the sender.
-
-3.1. Bitstream definition
-
- The total number of bits used to describe one frame of 20 ms speech
- is 304, which fits in 38 bytes and results in a bit rate of 15.20
- kbit/s. For the case with a frame length of 30 ms speech the total
- number of bits used is 400, which fits in 50 bytes and results in a
- bit rate of 13.33 kbit/s. In the bitstream definition, the bits are
- distributed into three classes according to their bit error or loss
- sensitivity. The most sensitive bits (class 1) are placed first in
-
-
-
-Duric & Andersen Experimental [Page 3]
-
-RFC 3952 RTP Payload Format for iLBC Speech December 2004
-
-
- the bitstream for each frame. The less sensitive bits (class 2) are
- placed after the class 1 bits. The least sensitive bits (class 3)
- are placed at the end of the bitstream for each frame.
-
- Looking at the 20/30 ms frame length cases for each class: The class
- 1 bits occupy a total of 6/8 bytes (48/64 bits), the class 2 bits
- occupy 8/12 bytes (64/96 bits), and the class 3 bits occupy 24/30
- bytes (191/239 bits). This distribution of the bits enables the use
- of uneven level protection (ULP). The detailed bit allocation is
- shown in the table below. When a quantization index is distributed
- between more classes the more significant bits belong to the lowest
- class.
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-
-Duric & Andersen Experimental [Page 4]
-
-RFC 3952 RTP Payload Format for iLBC Speech December 2004
-
-
- Bitstream structure:
-
- ------------------------------------------------------------------+
- Parameter | Bits Class <1,2,3> |
- | 20 ms frame | 30 ms frame |
- ----------------------------------+---------------+---------------+
- Split 1 | 6 <6,0,0> | 6 <6,0,0> |
- LSF 1 Split 2 | 7 <7,0,0> | 7 <7,0,0> |
- LSF Split 3 | 7 <7,0,0> | 7 <7,0,0> |
- ------------------+---------------+---------------+
- Split 1 | NA (Not Appl.)| 6 <6,0,0> |
- LSF 2 Split 2 | NA | 7 <7,0,0> |
- Split 3 | NA | 7 <7,0,0> |
- ------------------+---------------+---------------+
- Sum | 20 <20,0,0> | 40 <40,0,0> |
- ----------------------------------+---------------+---------------+
- Block Class. | 2 <2,0,0> | 3 <3,0,0> |
- ----------------------------------+---------------+---------------+
- Position 22 sample segment | 1 <1,0,0> | 1 <1,0,0> |
- ----------------------------------+---------------+---------------+
- Scale Factor State Coder | 6 <6,0,0> | 6 <6,0,0> |
- ----------------------------------+---------------+---------------+
- Sample 0 | 3 <0,1,2> | 3 <0,1,2> |
- Quantized Sample 1 | 3 <0,1,2> | 3 <0,1,2> |
- Residual : | : : | : : |
- State : | : : | : : |
- Samples : | : : | : : |
- Sample 56 | 3 <0,1,2> | 3 <0,1,2> |
- Sample 57 | NA | 3 <0,1,2> |
- ------------------+---------------+---------------+
- Sum | 171 <0,57,114>| 174 <0,58,116>|
- ----------------------------------+---------------+---------------+
- Stage 1 | 7 <6,0,1> | 7 <4,2,1> |
- CB for 22/23 Stage 2 | 7 <0,0,7> | 7 <0,0,7> |
- sample block Stage 3 | 7 <0,0,7> | 7 <0,0,7> |
- ------------------+---------------+---------------+
- Sum | 21 <6,0,15> | 21 <4,2,15> |
- ----------------------------------+---------------+---------------+
- Stage 1 | 5 <2,0,3> | 5 <1,1,3> |
- Gain for 22/23 Stage 2 | 4 <1,1,2> | 4 <1,1,2> |
- sample block Stage 3 | 3 <0,0,3> | 3 <0,0,3> |
- ------------------+---------------+---------------+
- Sum | 12 <3,1,8> | 12 <2,2,8> |
- ----------------------------------+---------------+---------------+
- Stage 1 | 8 <7,0,1> | 8 <6,1,1> |
- sub-block 1 Stage 2 | 7 <0,0,7> | 7 <0,0,7> |
- Stage 3 | 7 <0,0,7> | 7 <0,0,7> |
- ------------------+---------------+---------------+
-
-
-
-Duric & Andersen Experimental [Page 5]
-
-RFC 3952 RTP Payload Format for iLBC Speech December 2004
-
-
- Stage 1 | 8 <0,0,8> | 8 <0,7,1> |
- sub-block 2 Stage 2 | 8 <0,0,8> | 8 <0,0,8> |
- Indices Stage 3 | 8 <0,0,8> | 8 <0,0,8> |
- for CB ------------------+---------------+---------------+
- sub-blocks Stage 1 | NA | 8 <0,7,1> |
- sub-block 3 Stage 2 | NA | 8 <0,0,8> |
- Stage 3 | NA | 8 <0,0,8> |
- ------------------+---------------+---------------+
- Stage 1 | NA | 8 <0,7,1> |
- sub-block 4 Stage 2 | NA | 8 <0,0,8> |
- Stage 3 | NA | 8 <0,0,8> |
- ------------------+---------------+---------------+
- Sum | 46 <7,0,39> | 94 <6,22,66> |
- ----------------------------------+---------------+---------------+
- Stage 1 | 5 <1,2,2> | 5 <1,2,2> |
- sub-block 1 Stage 2 | 4 <1,1,2> | 4 <1,2,1> |
- Stage 3 | 3 <0,0,3> | 3 <0,0,3> |
- ------------------+---------------+---------------+
- Stage 1 | 5 <1,1,3> | 5 <0,2,3> |
- sub-block 2 Stage 2 | 4 <0,2,2> | 4 <0,2,2> |
- Stage 3 | 3 <0,0,3> | 3 <0,0,3> |
- Gains for ------------------+---------------+---------------+
- sub-blocks Stage 1 | NA | 5 <0,1,4> |
- sub-block 3 Stage 2 | NA | 4 <0,1,3> |
- Stage 3 | NA | 3 <0,0,3> |
- ------------------+---------------+---------------+
- Stage 1 | NA | 5 <0,1,4> |
- sub-block 4 Stage 2 | NA | 4 <0,1,3> |
- Stage 3 | NA | 3 <0,0,3> |
- ------------------+---------------+---------------+
- Sum | 24 <3,6,15> | 48 <2,12,34> |
- -------------------------------------------------------------------
- Empty frame indicator | 1 <0,0,1> | 1 <0,0,1> |
- -------------------------------------------------------------------
- SUM 304 <48,64,192> 400 <64,96,240>
-
- Table 3.1 The bitstream definition for iLBC.
-
- When packetized into the payload, all the class 1 bits MUST be sorted
- in order (from top and down) as they were specified in the table.
- Additionally, all the class 2 bits MUST be sorted (from top and down)
- and all the class 3 bits MUST be sorted in the same sequential order.
-
-3.2. Multiple iLBC frames in a RTP packet
-
- More than one iLBC frame may be included in a single RTP packet by a
- sender.
-
-
-
-
-Duric & Andersen Experimental [Page 6]
-
-RFC 3952 RTP Payload Format for iLBC Speech December 2004
-
-
- It is important to observe that senders have the following additional
- restrictions:
-
- o SHOULD NOT include more iLBC frames in a single RTP packet than
- will fit in the MTU of the RTP transport protocol.
-
- o Frames MUST NOT be split between RTP packets.
-
- o Frames of the different modes (20 ms and 30 ms) MUST NOT be
- included within the same packet.
-
- It is RECOMMENDED that the number of frames contained within an RTP
- packet are consistent with the application. For example, in
- telephony and other real time applications where delay is important,
- the delay is lower depending on the amount of frames per packet
- (i.e., fewer frames per packet, the lower the delay). Whereas for
- bandwidth constrained links or delay insensitive streaming messaging
- application, one or more frames per packet would be acceptable.
-
- Information describing the number of frames contained in an RTP
- packet is not transmitted as part of the RTP payload. The way to
- determine the number of iLBC frames is to count the total number of
- octets within the RTP packet, and divide the octet count by the
- number of expected octets per frame (32/50 per frame).
-
-4. IANA Considerations
-
- One new MIME sub-type as described in this section has been
- registered.
-
-4.1. Storage Mode
-
- The storage mode is used for storing speech frames (e.g., as a file
- or email attachment).
-
- +------------------+
- | Header |
- +------------------+
- | Speech frame 1 |
- +------------------+
- : :
- +------------------+
- | Speech frame n |
- +------------------+
-
- Figure 2, Storage format diagram
-
-
-
-
-
-Duric & Andersen Experimental [Page 7]
-
-RFC 3952 RTP Payload Format for iLBC Speech December 2004
-
-
- The file begins with a header that includes only a magic number to
- identify that it is an iLBC file.
-
- The magic number for iLBC file MUST correspond to the ASCII character
- string:
-
- o for 30 ms frame size mode:"#!iLBC30\n", or "0x23 0x21 0x69
- 0x4C 0x42 0x43 0x33 0x30 0x0A" in hexadecimal form,
-
- o for 20 ms frame size mode:"#!iLBC20\n", or "0x23 0x21 0x69
- 0x4C 0x42 0x43 0x32 0x30 0x0A" in hexadecimal form.
-
- After the header, follow the speech frames in consecutive order.
-
- Speech frames lost in transmission MUST be stored as "empty frames",
- as defined in [1].
-
-4.2. MIME Registration of iLBC
-
- MIME media type name: audio
-
- MIME subtype: iLBC
-
- Optional parameters:
-
- All of the parameters does apply for RTP transfer only.
-
- maxptime:The maximum amount of media which can be encapsulated in
- each packet, expressed as time in milliseconds. The time
- SHALL be calculated as the sum of the time the media present
- in the packet represents. The time SHOULD be a multiple of
- the frame size. This attribute is probably only meaningful
- for audio data, but may be used with other media types if it
- makes sense. It is a media attribute, and is not dependent
- on charset. Note that this attribute was introduced after
- RFC 2327, and non updated implementations will ignore this
- attribute.
-
- mode: The iLBC operating frame mode (20 or 30 ms) that will be
- encapsulated in each packet. Values can be 0, 20 and 30
- (where 0 is reserved, 20 stands for preferred 20 ms frame
- size and 30 stands for preferred 30 ms frame size).
-
- ptime: Defined as usual for RTP audio (see [5]).
-
- Encoding considerations:
- This type is defined for transfer via both RTP (RFC 3550)
- and stored-file methods as described in Section 4.1, of RFC
-
-
-
-Duric & Andersen Experimental [Page 8]
-
-RFC 3952 RTP Payload Format for iLBC Speech December 2004
-
-
- 3952. Audio data is binary data, and must be encoded for
- non-binary transport; the Base64 encoding is suitable for
- email.
-
- Security considerations:
- See Section 6 of RFC 3952.
-
- Public specification:
- Please refer to RFC 3951 [1].
-
- Additional information:
- The following applies to stored-file transfer methods:
-
- Magic number:
- ASCII character string for:
- o 30 ms frame size mode "#!iLBC30\n" (or 0x23 0x21
- 0x69 0x4C 0x42 0x43 0x33 0x30 0x0A in hexadecimal)
- o 20 ms frame size mode "#!iLBC20\n" (or 0x23 0x21
- 0x69 0x4C 0x42 0x43 0x32 0x30 0x0A in hexadecimal)
-
- File extensions: lbc, LBC
- Macintosh file type code: none
- Object identifier or OID: none
-
- Person & email address to contact for further information:
- alan.duric@telio.no
-
- Intended usage: COMMON.
- It is expected that many VoIP applications will use this
- type.
-
- Author/Change controller:
- alan.duric@telio.no
- IETF Audio/Video transport working group
-
-5. Mapping To SDP Parameters
-
- The information carried in the MIME media type specification has a
- specific mapping to fields in the Session Description Protocol (SDP)
- [5], which is commonly used to describe RTP sessions. When SDP is
- used to specify sessions employing the iLBC codec, the mapping is as
- follows:
-
- o The MIME type ("audio") goes in SDP "m=" as the media name.
-
- o The MIME subtype (payload format name) goes in SDP "a=rtpmap" as
- the encoding name.
-
-
-
-
-Duric & Andersen Experimental [Page 9]
-
-RFC 3952 RTP Payload Format for iLBC Speech December 2004
-
-
- o The parameters "ptime" and "maxptime" go in the SDP "a=ptime" and
- "a=maxptime" attributes, respectively.
-
- o The parameter "mode" goes in the SDP "a=fmtp" attribute by copying
- it directly from the MIME media type string as "mode=value".
-
- When conveying information by SDP, the encoding name SHALL be "iLBC"
- (the same as the MIME subtype).
-
- An example of the media representation in SDP for describing iLBC
- might be:
-
- m=audio 49120 RTP/AVP 97
- a=rtpmap:97 iLBC/8000
-
- If 20 ms frame size mode is used, remote iLBC encoder SHALL receive
- "mode" parameter in the SDP "a=fmtp" attribute by copying them
- directly from the MIME media type string as a semicolon separated
- with parameter=value, where parameter is "mode", and values can be 0
- and 20 (where 0 is reserved and 20 stands for preferred 20 ms frame
- size). An example of the media representation in SDP for describing
- iLBC when 20 ms frame size mode is used might be:
-
- m=audio 49120 RTP/AVP 97
- a=rtpmap:97 iLBC/8000
- a=fmtp:97 mode=20
-
- It is important to emphasize the bi-directional character of the
- "mode" parameter - both sides of a bi-directional session MUST use
- the same "mode" value.
-
- The offer contains the preferred mode of the offerer. The answerer
- may agree to that mode by including the same mode in the answer, or
- may include a different mode. The resulting mode used by both
- parties SHALL be the lower of the bandwidth modes in the offer and
- answer.
-
- That is, an offer of "mode=20" receiving an answer of "mode=30" will
- result in "mode=30" being used by both participants. Similarly, an
- offer of "mode=30" and an answer of "mode=20" will result in
- "mode=30" being used by both participants.
-
- This is important when one end point utilizes a bandwidth constrained
- link (e.g., 28.8k modem link or slower), where only the lower frame
- size will work.
-
-
-
-
-
-
-Duric & Andersen Experimental [Page 10]
-
-RFC 3952 RTP Payload Format for iLBC Speech December 2004
-
-
- Parameter ptime can not be used for the purpose of specifying iLBC
- operating mode, due to fact that for the certain values it will be
- impossible to distinguish which mode is about to be used (e.g., when
- ptime=60, it would be impossible to distinguish if packet is carrying
- 2 frames of 30 ms or 3 frames of 20 ms, etc.).
-
- Note that the payload format (encoding) names are commonly shown in
- upper case. MIME subtypes are commonly shown in lower case. These
- names are case-insensitive in both places. Similarly, parameter
- names are case-insensitive both in MIME types and in the default
- mapping to the SDP a=fmtp attribute
-
-6. Security Considerations
-
- RTP packets using the payload format defined in this specification
- are subject to the general security considerations discussed in [3]
- and any appropriate profile (e.g., [4]).
-
- As this format transports encoded speech, the main security issues
- include confidentiality and authentication of the speech itself. The
- payload format itself does not have any built-in security mechanisms.
- Confidentiality of the media streams is achieved by encryption,
- therefore external mechanisms, such as SRTP [6], MAY be used for that
- purpose. The data compression used with this payload format is
- applied end-to-end; hence encryption may be performed after
- compression with no conflict between the two operations.
-
- A potential denial-of-service threat exists for data encoding using
- compression techniques that have non-uniform receiver-end
- computational load. The attacker can inject pathological datagrams
- into the stream which are complex to decode and cause the receiver to
- become overloaded. However, the encodings covered in this document
- do not exhibit any significant non-uniformity.
-
-7. References
-
-7.1. Normative References
-
- [1] Andersen, S., Duric, A., Astrom, H., Hagen, R., Kleijn, W., and
- J. Linden, "Internet Low Bit Rate Codec (iLBC)", RFC 3951,
- December 2004.
-
- [2] Bradner, S., "Key words for use in RFCs to Indicate Requirement
- Levels", BCP 14, RFC 2119, March 1997.
-
- [3] Schulzrinne, H., Casner, S., Frederick, R., and V. Jacobson,
- "RTP: A Transport Protocol for Real-Time Applications", STD 64,
- RFC 3550, July 2003.
-
-
-
-Duric & Andersen Experimental [Page 11]
-
-RFC 3952 RTP Payload Format for iLBC Speech December 2004
-
-
- [4] Schulzrinne, H. and S. Casner, "RTP Profile for Audio and Video
- Conferences with Minimal Control", STD 65, RFC 3551, July 2003.
-
- [5] Handley, M. and V. Jacobson, "SDP: Session Description
- Protocol", RFC 2327, April 1998.
-
- [6] Baugher, M., McGrew, D., Naslund, M., Carrara, E., and K.
- Norrman, "The Secure Real-time Transport Protocol", RFC 3711,
- March 2004.
-
-7.2. Informative References
-
- [7] ITU-T Recommendation G.711, available online from the ITU
- bookstore at http://www.itu.int.
-
-8. Acknowledgements
-
- Henry Sinnreich, Patrik Faltstrom, Alan Johnston and Jean-Francois
- Mule for great support of the iLBC initiative and for valuable
- feedback and comments.
-
-Authors' Addresses
-
- Alan Duric
- Telio AS
- Stoperigt. 2
- Oslo, N-0250
- Norway
-
- Phone: +47 21673505
- EMail: alan.duric@telio.no
-
-
- Soren Vang Andersen
- Department of Communication Technology
- Aalborg University
- Fredrik Bajers Vej 7A
- 9200 Aalborg
- Denmark
-
- Phone: ++45 9 6358627
- EMail: sva@kom.auc.dk
-
-
-
-
-
-
-
-
-
-Duric & Andersen Experimental [Page 12]
-
-RFC 3952 RTP Payload Format for iLBC Speech December 2004
-
-
-Full Copyright Statement
-
- Copyright (C) The Internet Society (2004).
-
- This document is subject to the rights, licenses and restrictions
- contained in BCP 78, and except as set forth therein, the authors
- retain all their rights.
-
- This document and the information contained herein are provided on an
- "AS IS" basis and THE CONTRIBUTOR, THE ORGANIZATION HE/SHE REPRESENTS
- OR IS SPONSORED BY (IF ANY), THE INTERNET SOCIETY AND THE INTERNET
- ENGINEERING TASK FORCE DISCLAIM ALL WARRANTIES, EXPRESS OR IMPLIED,
- INCLUDING BUT NOT LIMITED TO ANY WARRANTY THAT THE USE OF THE
- INFORMATION HEREIN WILL NOT INFRINGE ANY RIGHTS OR ANY IMPLIED
- WARRANTIES OF MERCHANTABILITY OR FITNESS FOR A PARTICULAR PURPOSE.
-
-Intellectual Property
-
- The IETF takes no position regarding the validity or scope of any
- Intellectual Property Rights or other rights that might be claimed to
- pertain to the implementation or use of the technology described in
- this document or the extent to which any license under such rights
- might or might not be available; nor does it represent that it has
- made any independent effort to identify any such rights. Information
- on the IETF's procedures with respect to rights in IETF Documents can
- be found in BCP 78 and BCP 79.
-
- Copies of IPR disclosures made to the IETF Secretariat and any
- assurances of licenses to be made available, or the result of an
- attempt made to obtain a general license or permission for the use of
- such proprietary rights by implementers or users of this
- specification can be obtained from the IETF on-line IPR repository at
- http://www.ietf.org/ipr.
-
- The IETF invites any interested party to bring to its attention any
- copyrights, patents or patent applications, or other proprietary
- rights that may cover technology that may be required to implement
- this standard. Please address the information to the IETF at ietf-
- ipr@ietf.org.
-
-
-Acknowledgement
-
- Funding for the RFC Editor function is currently provided by the
- Internet Society.
-
-
-
-
-
-
-Duric & Andersen Experimental [Page 13]
-
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/encode.c b/trunk/src/modules/audio_coding/codecs/ilbc/encode.c
deleted file mode 100644
index 64e9eab..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/encode.c
+++ /dev/null
@@ -1,518 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Encode.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "lpc_encode.h"
-#include "frame_classify.h"
-#include "state_search.h"
-#include "state_construct.h"
-#include "constants.h"
-#include "cb_search.h"
-#include "cb_construct.h"
-#include "index_conv_enc.h"
-#include "pack_bits.h"
-#include "hp_input.h"
-
-#ifdef SPLIT_10MS
-#include "unpack_bits.h"
-#include "index_conv_dec.h"
-#endif
-#ifndef WEBRTC_BIG_ENDIAN
-#include "swap_bytes.h"
-#endif
-
-/*----------------------------------------------------------------*
- * main encoder function
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_EncodeImpl(
- WebRtc_UWord16 *bytes, /* (o) encoded data bits iLBC */
- WebRtc_Word16 *block, /* (i) speech vector to encode */
- iLBC_Enc_Inst_t *iLBCenc_inst /* (i/o) the general encoder
- state */
- ){
- int n, meml_gotten, Nfor, Nback;
- WebRtc_Word16 diff, start_pos;
- int index;
- int subcount, subframe;
- WebRtc_Word16 start_count, end_count;
- WebRtc_Word16 *residual;
- WebRtc_Word32 en1, en2;
- WebRtc_Word16 scale, max;
- WebRtc_Word16 *syntdenum;
- WebRtc_Word16 *decresidual;
- WebRtc_Word16 *reverseResidual;
- WebRtc_Word16 *reverseDecresidual;
- /* Stack based */
- WebRtc_Word16 weightdenum[(LPC_FILTERORDER + 1)*NSUB_MAX];
- WebRtc_Word16 dataVec[BLOCKL_MAX + LPC_FILTERORDER];
- WebRtc_Word16 memVec[CB_MEML+CB_FILTERLEN];
- WebRtc_Word16 bitsMemory[sizeof(iLBC_bits)/sizeof(WebRtc_Word16)];
- iLBC_bits *iLBCbits_inst = (iLBC_bits*)bitsMemory;
-
-
-#ifdef SPLIT_10MS
- WebRtc_Word16 *weightdenumbuf = iLBCenc_inst->weightdenumbuf;
- WebRtc_Word16 last_bit;
-#endif
-
- WebRtc_Word16 *data = &dataVec[LPC_FILTERORDER];
- WebRtc_Word16 *mem = &memVec[CB_HALFFILTERLEN];
-
- /* Reuse som buffers to save stack memory */
- residual = &iLBCenc_inst->lpc_buffer[LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl];
- syntdenum = mem; /* syntdenum[(LPC_FILTERORDER + 1)*NSUB_MAX] and mem are used non overlapping in the code */
- decresidual = residual; /* Already encoded residual is overwritten by the decoded version */
- reverseResidual = data; /* data and reverseResidual are used non overlapping in the code */
- reverseDecresidual = reverseResidual; /* Already encoded residual is overwritten by the decoded version */
-
-#ifdef SPLIT_10MS
-
- WebRtcSpl_MemSetW16 ( (WebRtc_Word16 *) iLBCbits_inst, 0,
- (WebRtc_Word16) (sizeof(iLBC_bits) / sizeof(WebRtc_Word16)) );
-
- start_pos = iLBCenc_inst->start_pos;
- diff = iLBCenc_inst->diff;
-
- if (iLBCenc_inst->section != 0){
- WEBRTC_SPL_MEMCPY_W16 (weightdenum, weightdenumbuf,
- SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
- /* Un-Packetize the frame into parameters */
- last_bit = WebRtcIlbcfix_UnpackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
- if (last_bit)
- return;
- /* adjust index */
- WebRtcIlbcfix_IndexConvDec (iLBCbits_inst->cb_index);
-
- if (iLBCenc_inst->section == 1){
- /* Save first 80 samples of a 160/240 sample frame for 20/30msec */
- WEBRTC_SPL_MEMCPY_W16 (iLBCenc_inst->past_samples, block, 80);
- }
- else{ // iLBCenc_inst->section == 2 AND mode = 30ms
- /* Save second 80 samples of a 240 sample frame for 30msec */
- WEBRTC_SPL_MEMCPY_W16 (iLBCenc_inst->past_samples + 80, block, 80);
- }
- }
- else{ // iLBCenc_inst->section == 0
- /* form a complete frame of 160/240 for 20msec/30msec mode */
- WEBRTC_SPL_MEMCPY_W16 (data + (iLBCenc_inst->mode * 8) - 80, block, 80);
- WEBRTC_SPL_MEMCPY_W16 (data, iLBCenc_inst->past_samples,
- (iLBCenc_inst->mode * 8) - 80);
- iLBCenc_inst->Nfor_flag = 0;
- iLBCenc_inst->Nback_flag = 0;
-#else
- /* copy input block to data*/
- WEBRTC_SPL_MEMCPY_W16(data,block,iLBCenc_inst->blockl);
-#endif
-
- /* high pass filtering of input signal and scale down the residual (*0.5) */
- WebRtcIlbcfix_HpInput(data, (WebRtc_Word16*)WebRtcIlbcfix_kHpInCoefs,
- iLBCenc_inst->hpimemy, iLBCenc_inst->hpimemx,
- iLBCenc_inst->blockl);
-
- /* LPC of hp filtered input data */
- WebRtcIlbcfix_LpcEncode(syntdenum, weightdenum, iLBCbits_inst->lsf, data,
- iLBCenc_inst);
-
- /* Set up state */
- WEBRTC_SPL_MEMCPY_W16(dataVec, iLBCenc_inst->anaMem, LPC_FILTERORDER);
-
- /* inverse filter to get residual */
- for (n=0; n<iLBCenc_inst->nsub; n++ ) {
- WebRtcSpl_FilterMAFastQ12(
- &data[n*SUBL], &residual[n*SUBL],
- &syntdenum[n*(LPC_FILTERORDER+1)],
- LPC_FILTERORDER+1, SUBL);
- }
-
- /* Copy the state for next frame */
- WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->anaMem, &data[iLBCenc_inst->blockl-LPC_FILTERORDER], LPC_FILTERORDER);
-
- /* find state location */
-
- iLBCbits_inst->startIdx = WebRtcIlbcfix_FrameClassify(iLBCenc_inst,residual);
-
- /* check if state should be in first or last part of the
- two subframes */
-
- index = (iLBCbits_inst->startIdx-1)*SUBL;
- max=WebRtcSpl_MaxAbsValueW16(&residual[index], 2*SUBL);
- scale=WebRtcSpl_GetSizeInBits(WEBRTC_SPL_MUL_16_16(max,max));
-
- /* Scale to maximum 25 bits so that the MAC won't cause overflow */
- scale = scale - 25;
- if(scale < 0) {
- scale = 0;
- }
-
- diff = STATE_LEN - iLBCenc_inst->state_short_len;
- en1=WebRtcSpl_DotProductWithScale(&residual[index], &residual[index],
- iLBCenc_inst->state_short_len, scale);
- index += diff;
- en2=WebRtcSpl_DotProductWithScale(&residual[index], &residual[index],
- iLBCenc_inst->state_short_len, scale);
- if (en1 > en2) {
- iLBCbits_inst->state_first = 1;
- start_pos = (iLBCbits_inst->startIdx-1)*SUBL;
- } else {
- iLBCbits_inst->state_first = 0;
- start_pos = (iLBCbits_inst->startIdx-1)*SUBL + diff;
- }
-
- /* scalar quantization of state */
-
- WebRtcIlbcfix_StateSearch(iLBCenc_inst, iLBCbits_inst, &residual[start_pos],
- &syntdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)],
- &weightdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)]);
-
- WebRtcIlbcfix_StateConstruct(iLBCbits_inst->idxForMax, iLBCbits_inst->idxVec,
- &syntdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)],
- &decresidual[start_pos], iLBCenc_inst->state_short_len
- );
-
- /* predictive quantization in state */
-
- if (iLBCbits_inst->state_first) { /* put adaptive part in the end */
-
- /* setup memory */
-
- WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-iLBCenc_inst->state_short_len));
- WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-iLBCenc_inst->state_short_len,
- decresidual+start_pos, iLBCenc_inst->state_short_len);
-
- /* encode subframes */
-
- WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
- &residual[start_pos+iLBCenc_inst->state_short_len],
- mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL, diff,
- &weightdenum[iLBCbits_inst->startIdx*(LPC_FILTERORDER+1)], 0);
-
- /* construct decoded vector */
-
- WebRtcIlbcfix_CbConstruct(&decresidual[start_pos+iLBCenc_inst->state_short_len],
- iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
- mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL,
- diff
- );
-
- }
- else { /* put adaptive part in the beginning */
-
- /* create reversed vectors for prediction */
-
- WebRtcSpl_MemCpyReversedOrder(&reverseResidual[diff-1],
- &residual[(iLBCbits_inst->startIdx+1)*SUBL-STATE_LEN], diff);
-
- /* setup memory */
-
- meml_gotten = iLBCenc_inst->state_short_len;
- WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[start_pos], meml_gotten);
- WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-iLBCenc_inst->state_short_len));
-
- /* encode subframes */
- WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
- reverseResidual, mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL, diff,
- &weightdenum[(iLBCbits_inst->startIdx-1)*(LPC_FILTERORDER+1)],
- 0);
-
- /* construct decoded vector */
-
- WebRtcIlbcfix_CbConstruct(reverseDecresidual,
- iLBCbits_inst->cb_index, iLBCbits_inst->gain_index,
- mem+CB_MEML-ST_MEM_L_TBL, ST_MEM_L_TBL,
- diff
- );
-
- /* get decoded residual from reversed vector */
-
- WebRtcSpl_MemCpyReversedOrder(&decresidual[start_pos-1], reverseDecresidual, diff);
- }
-
-#ifdef SPLIT_10MS
- iLBCenc_inst->start_pos = start_pos;
- iLBCenc_inst->diff = diff;
- iLBCenc_inst->section++;
- /* adjust index */
- WebRtcIlbcfix_IndexConvEnc (iLBCbits_inst->cb_index);
- /* Packetize the parameters into the frame */
- WebRtcIlbcfix_PackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
- WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum,
- SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
- return;
- }
-#endif
-
- /* forward prediction of subframes */
-
- Nfor = iLBCenc_inst->nsub-iLBCbits_inst->startIdx-1;
-
- /* counter for predicted subframes */
-#ifdef SPLIT_10MS
- if (iLBCenc_inst->mode == 20)
- {
- subcount = 1;
- }
- if (iLBCenc_inst->mode == 30)
- {
- if (iLBCenc_inst->section == 1)
- {
- subcount = 1;
- }
- if (iLBCenc_inst->section == 2)
- {
- subcount = 3;
- }
- }
-#else
- subcount=1;
-#endif
-
- if( Nfor > 0 ){
-
- /* setup memory */
-
- WebRtcSpl_MemSetW16(mem, 0, CB_MEML-STATE_LEN);
- WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-STATE_LEN,
- decresidual+(iLBCbits_inst->startIdx-1)*SUBL, STATE_LEN);
-
-#ifdef SPLIT_10MS
- if (iLBCenc_inst->Nfor_flag > 0)
- {
- for (subframe = 0; subframe < WEBRTC_SPL_MIN (Nfor, 2); subframe++)
- {
- /* update memory */
- WEBRTC_SPL_MEMCPY_W16 (mem, mem + SUBL, (CB_MEML - SUBL));
- WEBRTC_SPL_MEMCPY_W16 (mem + CB_MEML - SUBL,
- &decresidual[(iLBCbits_inst->startIdx + 1 +
- subframe) * SUBL], SUBL);
- }
- }
-
- iLBCenc_inst->Nfor_flag++;
-
- if (iLBCenc_inst->mode == 20)
- {
- start_count = 0;
- end_count = Nfor;
- }
- if (iLBCenc_inst->mode == 30)
- {
- if (iLBCenc_inst->section == 1)
- {
- start_count = 0;
- end_count = WEBRTC_SPL_MIN (Nfor, 2);
- }
- if (iLBCenc_inst->section == 2)
- {
- start_count = WEBRTC_SPL_MIN (Nfor, 2);
- end_count = Nfor;
- }
- }
-#else
- start_count = 0;
- end_count = (WebRtc_Word16)Nfor;
-#endif
-
- /* loop over subframes to encode */
-
- for (subframe = start_count; subframe < end_count; subframe++){
-
- /* encode subframe */
-
- WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index+subcount*CB_NSTAGES,
- iLBCbits_inst->gain_index+subcount*CB_NSTAGES,
- &residual[(iLBCbits_inst->startIdx+1+subframe)*SUBL],
- mem, MEM_LF_TBL, SUBL,
- &weightdenum[(iLBCbits_inst->startIdx+1+subframe)*(LPC_FILTERORDER+1)],
- (WebRtc_Word16)subcount);
-
- /* construct decoded vector */
-
- WebRtcIlbcfix_CbConstruct(&decresidual[(iLBCbits_inst->startIdx+1+subframe)*SUBL],
- iLBCbits_inst->cb_index+subcount*CB_NSTAGES,
- iLBCbits_inst->gain_index+subcount*CB_NSTAGES,
- mem, MEM_LF_TBL,
- SUBL
- );
-
- /* update memory */
-
- WEBRTC_SPL_MEMMOVE_W16(mem, mem+SUBL, (CB_MEML-SUBL));
- WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
- &decresidual[(iLBCbits_inst->startIdx+1+subframe)*SUBL], SUBL);
-
- subcount++;
- }
- }
-
-#ifdef SPLIT_10MS
- if ((iLBCenc_inst->section == 1) &&
- (iLBCenc_inst->mode == 30) && (Nfor > 0) && (end_count == 2))
- {
- iLBCenc_inst->section++;
- /* adjust index */
- WebRtcIlbcfix_IndexConvEnc (iLBCbits_inst->cb_index);
- /* Packetize the parameters into the frame */
- WebRtcIlbcfix_PackBits (iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
- WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum,
- SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
- return;
- }
-#endif
-
- /* backward prediction of subframes */
-
- Nback = iLBCbits_inst->startIdx-1;
-
- if( Nback > 0 ){
-
- /* create reverse order vectors
- (The decresidual does not need to be copied since it is
- contained in the same vector as the residual)
- */
-
- WebRtcSpl_MemCpyReversedOrder(&reverseResidual[Nback*SUBL-1], residual, Nback*SUBL);
-
- /* setup memory */
-
- meml_gotten = SUBL*(iLBCenc_inst->nsub+1-iLBCbits_inst->startIdx);
- if( meml_gotten > CB_MEML ) {
- meml_gotten=CB_MEML;
- }
-
- WebRtcSpl_MemCpyReversedOrder(&mem[CB_MEML-1], &decresidual[Nback*SUBL], meml_gotten);
- WebRtcSpl_MemSetW16(mem, 0, (WebRtc_Word16)(CB_MEML-meml_gotten));
-
-#ifdef SPLIT_10MS
- if (iLBCenc_inst->Nback_flag > 0)
- {
- for (subframe = 0; subframe < WEBRTC_SPL_MAX (2 - Nfor, 0); subframe++)
- {
- /* update memory */
- WEBRTC_SPL_MEMCPY_W16 (mem, mem + SUBL, (CB_MEML - SUBL));
- WEBRTC_SPL_MEMCPY_W16 (mem + CB_MEML - SUBL,
- &reverseDecresidual[subframe * SUBL], SUBL);
- }
- }
-
- iLBCenc_inst->Nback_flag++;
-
-
- if (iLBCenc_inst->mode == 20)
- {
- start_count = 0;
- end_count = Nback;
- }
- if (iLBCenc_inst->mode == 30)
- {
- if (iLBCenc_inst->section == 1)
- {
- start_count = 0;
- end_count = WEBRTC_SPL_MAX (2 - Nfor, 0);
- }
- if (iLBCenc_inst->section == 2)
- {
- start_count = WEBRTC_SPL_MAX (2 - Nfor, 0);
- end_count = Nback;
- }
- }
-#else
- start_count = 0;
- end_count = (WebRtc_Word16)Nback;
-#endif
-
- /* loop over subframes to encode */
-
- for (subframe = start_count; subframe < end_count; subframe++){
-
- /* encode subframe */
-
- WebRtcIlbcfix_CbSearch(iLBCenc_inst, iLBCbits_inst->cb_index+subcount*CB_NSTAGES,
- iLBCbits_inst->gain_index+subcount*CB_NSTAGES, &reverseResidual[subframe*SUBL],
- mem, MEM_LF_TBL, SUBL,
- &weightdenum[(iLBCbits_inst->startIdx-2-subframe)*(LPC_FILTERORDER+1)],
- (WebRtc_Word16)subcount);
-
- /* construct decoded vector */
-
- WebRtcIlbcfix_CbConstruct(&reverseDecresidual[subframe*SUBL],
- iLBCbits_inst->cb_index+subcount*CB_NSTAGES,
- iLBCbits_inst->gain_index+subcount*CB_NSTAGES,
- mem, MEM_LF_TBL, SUBL
- );
-
- /* update memory */
-
- WEBRTC_SPL_MEMMOVE_W16(mem, mem+SUBL, (CB_MEML-SUBL));
- WEBRTC_SPL_MEMCPY_W16(mem+CB_MEML-SUBL,
- &reverseDecresidual[subframe*SUBL], SUBL);
-
- subcount++;
-
- }
-
- /* get decoded residual from reversed vector */
-
- WebRtcSpl_MemCpyReversedOrder(&decresidual[SUBL*Nback-1], reverseDecresidual, SUBL*Nback);
- }
- /* end encoding part */
-
- /* adjust index */
-
- WebRtcIlbcfix_IndexConvEnc(iLBCbits_inst->cb_index);
-
- /* Packetize the parameters into the frame */
-
-#ifdef SPLIT_10MS
- if( (iLBCenc_inst->mode==30) && (iLBCenc_inst->section==1) ){
- WebRtcIlbcfix_PackBits(iLBCenc_inst->bytes, iLBCbits_inst, iLBCenc_inst->mode);
- }
- else{
- WebRtcIlbcfix_PackBits(bytes, iLBCbits_inst, iLBCenc_inst->mode);
- }
-#else
- WebRtcIlbcfix_PackBits(bytes, iLBCbits_inst, iLBCenc_inst->mode);
-#endif
-
-#ifndef WEBRTC_BIG_ENDIAN
- /* Swap bytes for LITTLE ENDIAN since the packbits()
- function assumes BIG_ENDIAN machine */
-#ifdef SPLIT_10MS
- if (( (iLBCenc_inst->section == 1) && (iLBCenc_inst->mode == 20) ) ||
- ( (iLBCenc_inst->section == 2) && (iLBCenc_inst->mode == 30) )){
- WebRtcIlbcfix_SwapBytes(bytes, iLBCenc_inst->no_of_words);
- }
-#else
- WebRtcIlbcfix_SwapBytes(bytes, iLBCenc_inst->no_of_words);
-#endif
-#endif
-
-#ifdef SPLIT_10MS
- if (subcount == (iLBCenc_inst->nsub - 1))
- {
- iLBCenc_inst->section = 0;
- }
- else
- {
- iLBCenc_inst->section++;
- WEBRTC_SPL_MEMCPY_W16 (weightdenumbuf, weightdenum,
- SCRATCH_ENCODE_DATAVEC - SCRATCH_ENCODE_WEIGHTDENUM);
- }
-#endif
-
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/encode.h b/trunk/src/modules/audio_coding/codecs/ilbc/encode.h
deleted file mode 100644
index b553f0c..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/encode.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Encode.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENCODE_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * main encoder function
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_EncodeImpl(
- WebRtc_UWord16 *bytes, /* (o) encoded data bits iLBC */
- WebRtc_Word16 *block, /* (i) speech vector to encode */
- iLBC_Enc_Inst_t *iLBCenc_inst /* (i/o) the general encoder
- state */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/energy_inverse.c b/trunk/src/modules/audio_coding/codecs/ilbc/energy_inverse.c
deleted file mode 100644
index d56069b..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/energy_inverse.c
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_EnergyInverse.c
-
-******************************************************************/
-
-/* Inverses the in vector in into Q29 domain */
-
-#include "energy_inverse.h"
-
-void WebRtcIlbcfix_EnergyInverse(
- WebRtc_Word16 *energy, /* (i/o) Energy and inverse
- energy (in Q29) */
- int noOfEnergies) /* (i) The length of the energy
- vector */
-{
- WebRtc_Word32 Nom=(WebRtc_Word32)0x1FFFFFFF;
- WebRtc_Word16 *energyPtr;
- int i;
-
- /* Set the minimum energy value to 16384 to avoid overflow */
- energyPtr=energy;
- for (i=0; i<noOfEnergies; i++) {
- (*energyPtr)=WEBRTC_SPL_MAX((*energyPtr),16384);
- energyPtr++;
- }
-
- /* Calculate inverse energy in Q29 */
- energyPtr=energy;
- for (i=0; i<noOfEnergies; i++) {
- (*energyPtr) = (WebRtc_Word16)WebRtcSpl_DivW32W16(Nom, (*energyPtr));
- energyPtr++;
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/energy_inverse.h b/trunk/src/modules/audio_coding/codecs/ilbc/energy_inverse.h
deleted file mode 100644
index db13589..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/energy_inverse.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_EnergyInverse.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENERGY_INVERSE_H_
-
-#include "defines.h"
-
-/* Inverses the in vector in into Q29 domain */
-
-void WebRtcIlbcfix_EnergyInverse(
- WebRtc_Word16 *energy, /* (i/o) Energy and inverse
- energy (in Q29) */
- int noOfEnergies); /* (i) The length of the energy
- vector */
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/enh_upsample.c b/trunk/src/modules/audio_coding/codecs/ilbc/enh_upsample.c
deleted file mode 100644
index 3343816..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/enh_upsample.c
+++ /dev/null
@@ -1,110 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_EnhUpsample.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * upsample finite array assuming zeros outside bounds
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_EnhUpsample(
- WebRtc_Word32 *useq1, /* (o) upsampled output sequence */
- WebRtc_Word16 *seq1 /* (i) unupsampled sequence */
- ){
- int j;
- WebRtc_Word32 *pu1, *pu11;
- WebRtc_Word16 *ps, *w16tmp;
- const WebRtc_Word16 *pp;
-
- /* filtering: filter overhangs left side of sequence */
- pu1=useq1;
- for (j=0;j<ENH_UPS0; j++) {
- pu11=pu1;
- /* i = 2 */
- pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1;
- ps=seq1+2;
- (*pu11) = WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
- (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
- (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
- pu11+=ENH_UPS0;
- /* i = 3 */
- pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1;
- ps=seq1+3;
- (*pu11) = WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
- (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
- (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
- (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
- pu11+=ENH_UPS0;
- /* i = 4 */
- pp=WebRtcIlbcfix_kEnhPolyPhaser[j]+1;
- ps=seq1+4;
- (*pu11) = WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
- (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
- (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
- (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
- (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--,*pp++);
- pu1++;
- }
-
- /* filtering: simple convolution=inner products
- (not needed since the sequence is so short)
- */
-
- /* filtering: filter overhangs right side of sequence */
-
- /* Code with loops, which is equivivalent to the expanded version below
-
- filterlength = 5;
- hf1 = 2;
- for(j=0;j<ENH_UPS0; j++){
- pu = useq1 + (filterlength-hfl)*ENH_UPS0 + j;
- for(i=1; i<=hfl; i++){
- *pu=0;
- pp = polyp[j]+i;
- ps = seq1+dim1-1;
- for(k=0;k<filterlength-i;k++) {
- *pu += WEBRTC_SPL_MUL_16_16(*ps--, *pp++);
- }
- pu+=ENH_UPS0;
- }
- }
- */
- pu1 = useq1 + 12;
- w16tmp = seq1+4;
- for (j=0;j<ENH_UPS0; j++) {
- pu11 = pu1;
- /* i = 1 */
- pp = WebRtcIlbcfix_kEnhPolyPhaser[j]+2;
- ps = w16tmp;
- (*pu11) = WEBRTC_SPL_MUL_16_16(*ps--, *pp++);
- (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--, *pp++);
- (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--, *pp++);
- (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--, *pp++);
- pu11+=ENH_UPS0;
- /* i = 2 */
- pp = WebRtcIlbcfix_kEnhPolyPhaser[j]+3;
- ps = w16tmp;
- (*pu11) = WEBRTC_SPL_MUL_16_16(*ps--, *pp++);
- (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--, *pp++);
- (*pu11) += WEBRTC_SPL_MUL_16_16(*ps--, *pp++);
- pu11+=ENH_UPS0;
-
- pu1++;
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/enh_upsample.h b/trunk/src/modules/audio_coding/codecs/ilbc/enh_upsample.h
deleted file mode 100644
index 53534cc..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/enh_upsample.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_EnhUpsample.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENH_UPSAMPLE_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * upsample finite array assuming zeros outside bounds
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_EnhUpsample(
- WebRtc_Word32 *useq1, /* (o) upsampled output sequence */
- WebRtc_Word16 *seq1 /* (i) unupsampled sequence */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/enhancer.c b/trunk/src/modules/audio_coding/codecs/ilbc/enhancer.c
deleted file mode 100644
index b8f3335..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/enhancer.c
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Enhancer.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-#include "get_sync_seq.h"
-#include "smooth.h"
-
-/*----------------------------------------------------------------*
- * perform enhancement on idata+centerStartPos through
- * idata+centerStartPos+ENH_BLOCKL-1
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Enhancer(
- WebRtc_Word16 *odata, /* (o) smoothed block, dimension blockl */
- WebRtc_Word16 *idata, /* (i) data buffer used for enhancing */
- WebRtc_Word16 idatal, /* (i) dimension idata */
- WebRtc_Word16 centerStartPos, /* (i) first sample current block within idata */
- WebRtc_Word16 *period, /* (i) pitch period array (pitch bward-in time) */
- WebRtc_Word16 *plocs, /* (i) locations where period array values valid */
- WebRtc_Word16 periodl /* (i) dimension of period and plocs */
- ){
- /* Stack based */
- WebRtc_Word16 surround[ENH_BLOCKL];
-
- WebRtcSpl_MemSetW16(surround, 0, ENH_BLOCKL);
-
- /* get said second sequence of segments */
-
- WebRtcIlbcfix_GetSyncSeq(idata, idatal, centerStartPos, period, plocs,
- periodl, ENH_HL, surround);
-
- /* compute the smoothed output from said second sequence */
-
- WebRtcIlbcfix_Smooth(odata, idata+centerStartPos, surround);
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/enhancer.h b/trunk/src/modules/audio_coding/codecs/ilbc/enhancer.h
deleted file mode 100644
index e14f559..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/enhancer.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Enhancer.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * perform enhancement on idata+centerStartPos through
- * idata+centerStartPos+ENH_BLOCKL-1
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Enhancer(
- WebRtc_Word16 *odata, /* (o) smoothed block, dimension blockl */
- WebRtc_Word16 *idata, /* (i) data buffer used for enhancing */
- WebRtc_Word16 idatal, /* (i) dimension idata */
- WebRtc_Word16 centerStartPos, /* (i) first sample current block within idata */
- WebRtc_Word16 *period, /* (i) pitch period array (pitch bward-in time) */
- WebRtc_Word16 *plocs, /* (i) locations where period array values valid */
- WebRtc_Word16 periodl /* (i) dimension of period and plocs */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/enhancer_interface.c b/trunk/src/modules/audio_coding/codecs/ilbc/enhancer_interface.c
deleted file mode 100644
index 61b71d1..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/enhancer_interface.c
+++ /dev/null
@@ -1,381 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_EnhancerInterface.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-#include "xcorr_coef.h"
-#include "enhancer.h"
-#include "hp_output.h"
-
-
-
-/*----------------------------------------------------------------*
- * interface for enhancer
- *---------------------------------------------------------------*/
-
-int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
- WebRtc_Word16 *out, /* (o) enhanced signal */
- WebRtc_Word16 *in, /* (i) unenhanced signal */
- iLBC_Dec_Inst_t *iLBCdec_inst /* (i) buffers etc */
- ){
- int iblock;
- int lag=20, tlag=20;
- int inLen=iLBCdec_inst->blockl+120;
- WebRtc_Word16 scale, scale1, plc_blockl;
- WebRtc_Word16 *enh_buf, *enh_period;
- WebRtc_Word32 tmp1, tmp2, max, new_blocks;
- WebRtc_Word16 *enh_bufPtr1;
- int i, k;
- WebRtc_Word16 EnChange;
- WebRtc_Word16 SqrtEnChange;
- WebRtc_Word16 inc;
- WebRtc_Word16 win;
- WebRtc_Word16 *tmpW16ptr;
- WebRtc_Word16 startPos;
- WebRtc_Word16 *plc_pred;
- WebRtc_Word16 *target, *regressor;
- WebRtc_Word16 max16;
- int shifts;
- WebRtc_Word32 ener;
- WebRtc_Word16 enerSh;
- WebRtc_Word16 corrSh;
- WebRtc_Word16 ind, sh;
- WebRtc_Word16 start, stop;
- /* Stack based */
- WebRtc_Word16 totsh[3];
- WebRtc_Word16 downsampled[(BLOCKL_MAX+120)>>1]; /* length 180 */
- WebRtc_Word32 corr32[50];
- WebRtc_Word32 corrmax[3];
- WebRtc_Word16 corr16[3];
- WebRtc_Word16 en16[3];
- WebRtc_Word16 lagmax[3];
-
- plc_pred = downsampled; /* Reuse memory since plc_pred[ENH_BLOCKL] and
- downsampled are non overlapping */
- enh_buf=iLBCdec_inst->enh_buf;
- enh_period=iLBCdec_inst->enh_period;
-
- /* Copy in the new data into the enhancer buffer */
-
- WEBRTC_SPL_MEMMOVE_W16(enh_buf, &enh_buf[iLBCdec_inst->blockl],
- ENH_BUFL-iLBCdec_inst->blockl);
-
- WEBRTC_SPL_MEMCPY_W16(&enh_buf[ENH_BUFL-iLBCdec_inst->blockl], in,
- iLBCdec_inst->blockl);
-
- /* Set variables that are dependent on frame size */
- if (iLBCdec_inst->mode==30) {
- plc_blockl=ENH_BLOCKL;
- new_blocks=3;
- startPos=320; /* Start position for enhancement
- (640-new_blocks*ENH_BLOCKL-80) */
- } else {
- plc_blockl=40;
- new_blocks=2;
- startPos=440; /* Start position for enhancement
- (640-new_blocks*ENH_BLOCKL-40) */
- }
-
- /* Update the pitch prediction for each enhancer block, move the old ones */
- WEBRTC_SPL_MEMMOVE_W16(enh_period, &enh_period[new_blocks],
- (ENH_NBLOCKS_TOT-new_blocks));
-
- k=WebRtcSpl_DownsampleFast(
- enh_buf+ENH_BUFL-inLen, /* Input samples */
- (WebRtc_Word16)(inLen+ENH_BUFL_FILTEROVERHEAD),
- downsampled,
- (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16(inLen, 1),
- (WebRtc_Word16*)WebRtcIlbcfix_kLpFiltCoefs, /* Coefficients in Q12 */
- FILTERORDER_DS_PLUS1, /* Length of filter (order-1) */
- FACTOR_DS,
- DELAY_DS);
-
- /* Estimate the pitch in the down sampled domain. */
- for(iblock = 0; iblock<new_blocks; iblock++){
-
- /* references */
- i=60+WEBRTC_SPL_MUL_16_16(iblock,ENH_BLOCKL_HALF);
- target=downsampled+i;
- regressor=downsampled+i-10;
-
- /* scaling */
- max16=WebRtcSpl_MaxAbsValueW16(®ressor[-50],
- (WebRtc_Word16)(ENH_BLOCKL_HALF+50-1));
- shifts = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_MUL_16_16(max16, max16)) - 25;
- shifts = WEBRTC_SPL_MAX(0, shifts);
-
- /* compute cross correlation */
- WebRtcSpl_CrossCorrelation(corr32, target, regressor,
- ENH_BLOCKL_HALF, 50, (WebRtc_Word16)shifts, -1);
-
- /* Find 3 highest correlations that should be compared for the
- highest (corr*corr)/ener */
-
- for (i=0;i<2;i++) {
- lagmax[i] = WebRtcSpl_MaxIndexW32(corr32, 50);
- corrmax[i] = corr32[lagmax[i]];
- start = lagmax[i] - 2;
- stop = lagmax[i] + 2;
- start = WEBRTC_SPL_MAX(0, start);
- stop = WEBRTC_SPL_MIN(49, stop);
- for (k=start; k<=stop; k++) {
- corr32[k] = 0;
- }
- }
- lagmax[2] = WebRtcSpl_MaxIndexW32(corr32, 50);
- corrmax[2] = corr32[lagmax[2]];
-
- /* Calculate normalized corr^2 and ener */
- for (i=0;i<3;i++) {
- corrSh = 15-WebRtcSpl_GetSizeInBits(corrmax[i]);
- ener = WebRtcSpl_DotProductWithScale(®ressor[-lagmax[i]],
- ®ressor[-lagmax[i]],
- ENH_BLOCKL_HALF, shifts);
- enerSh = 15-WebRtcSpl_GetSizeInBits(ener);
- corr16[i] = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(corrmax[i], corrSh);
- corr16[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(corr16[i],
- corr16[i], 16);
- en16[i] = (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(ener, enerSh);
- totsh[i] = enerSh - WEBRTC_SPL_LSHIFT_W32(corrSh, 1);
- }
-
- /* Compare lagmax[0..3] for the (corr^2)/ener criteria */
- ind = 0;
- for (i=1; i<3; i++) {
- if (totsh[ind] > totsh[i]) {
- sh = WEBRTC_SPL_MIN(31, totsh[ind]-totsh[i]);
- if ( WEBRTC_SPL_MUL_16_16(corr16[ind], en16[i]) <
- WEBRTC_SPL_MUL_16_16_RSFT(corr16[i], en16[ind], sh)) {
- ind = i;
- }
- } else {
- sh = WEBRTC_SPL_MIN(31, totsh[i]-totsh[ind]);
- if (WEBRTC_SPL_MUL_16_16_RSFT(corr16[ind], en16[i], sh) <
- WEBRTC_SPL_MUL_16_16(corr16[i], en16[ind])) {
- ind = i;
- }
- }
- }
-
- lag = lagmax[ind] + 10;
-
- /* Store the estimated lag in the non-downsampled domain */
- enh_period[ENH_NBLOCKS_TOT-new_blocks+iblock] =
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(lag, 8);
-
- /* Store the estimated lag for backward PLC */
- if (iLBCdec_inst->prev_enh_pl==1) {
- if (!iblock) {
- tlag = WEBRTC_SPL_MUL_16_16(lag, 2);
- }
- } else {
- if (iblock==1) {
- tlag = WEBRTC_SPL_MUL_16_16(lag, 2);
- }
- }
-
- lag = WEBRTC_SPL_MUL_16_16(lag, 2);
- }
-
- if ((iLBCdec_inst->prev_enh_pl==1)||(iLBCdec_inst->prev_enh_pl==2)) {
-
- /* Calculate the best lag of the new frame
- This is used to interpolate backwards and mix with the PLC'd data
- */
-
- /* references */
- target=in;
- regressor=in+tlag-1;
-
- /* scaling */
- max16=WebRtcSpl_MaxAbsValueW16(regressor, (WebRtc_Word16)(plc_blockl+3-1));
- if (max16>5000)
- shifts=2;
- else
- shifts=0;
-
- /* compute cross correlation */
- WebRtcSpl_CrossCorrelation(corr32, target, regressor,
- plc_blockl, 3, (WebRtc_Word16)shifts, 1);
-
- /* find lag */
- lag=WebRtcSpl_MaxIndexW32(corr32, 3);
- lag+=tlag-1;
-
- /* Copy the backward PLC to plc_pred */
-
- if (iLBCdec_inst->prev_enh_pl==1) {
- if (lag>plc_blockl) {
- WEBRTC_SPL_MEMCPY_W16(plc_pred, &in[lag-plc_blockl], plc_blockl);
- } else {
- WEBRTC_SPL_MEMCPY_W16(&plc_pred[plc_blockl-lag], in, lag);
- WEBRTC_SPL_MEMCPY_W16(
- plc_pred, &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl+lag],
- (plc_blockl-lag));
- }
- } else {
- int pos;
-
- pos = plc_blockl;
-
- while (lag<pos) {
- WEBRTC_SPL_MEMCPY_W16(&plc_pred[pos-lag], in, lag);
- pos = pos - lag;
- }
- WEBRTC_SPL_MEMCPY_W16(plc_pred, &in[lag-pos], pos);
-
- }
-
- if (iLBCdec_inst->prev_enh_pl==1) {
- /* limit energy change
- if energy in backward PLC is more than 4 times higher than the forward
- PLC, then reduce the energy in the backward PLC vector:
- sample 1...len-16 set energy of the to 4 times forward PLC
- sample len-15..len interpolate between 4 times fw PLC and bw PLC energy
-
- Note: Compared to floating point code there is a slight change,
- the window is 16 samples long instead of 10 samples to simplify the
- calculations
- */
-
- max=WebRtcSpl_MaxAbsValueW16(
- &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl], plc_blockl);
- max16=WebRtcSpl_MaxAbsValueW16(plc_pred, plc_blockl);
- max = WEBRTC_SPL_MAX(max, max16);
- scale=22-(WebRtc_Word16)WebRtcSpl_NormW32(max);
- scale=WEBRTC_SPL_MAX(scale,0);
-
- tmp2 = WebRtcSpl_DotProductWithScale(
- &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl],
- &enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl],
- plc_blockl, scale);
- tmp1 = WebRtcSpl_DotProductWithScale(plc_pred, plc_pred,
- plc_blockl, scale);
-
- /* Check the energy difference */
- if ((tmp1>0)&&((tmp1>>2)>tmp2)) {
- /* EnChange is now guaranteed to be <0.5
- Calculate EnChange=tmp2/tmp1 in Q16
- */
-
- scale1=(WebRtc_Word16)WebRtcSpl_NormW32(tmp1);
- tmp1=WEBRTC_SPL_SHIFT_W32(tmp1, (scale1-16)); /* using 15 bits */
-
- tmp2=WEBRTC_SPL_SHIFT_W32(tmp2, (scale1));
- EnChange = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp2,
- (WebRtc_Word16)tmp1);
-
- /* Calculate the Sqrt of the energy in Q15 ((14+16)/2) */
- SqrtEnChange = (WebRtc_Word16)WebRtcSpl_SqrtFloor(
- WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)EnChange, 14));
-
-
- /* Multiply first part of vector with 2*SqrtEnChange */
- WebRtcSpl_ScaleVector(plc_pred, plc_pred, SqrtEnChange,
- (WebRtc_Word16)(plc_blockl-16), 14);
-
- /* Calculate increase parameter for window part (16 last samples) */
- /* (1-2*SqrtEnChange)/16 in Q15 */
- inc=(2048-WEBRTC_SPL_RSHIFT_W16(SqrtEnChange, 3));
-
- win=0;
- tmpW16ptr=&plc_pred[plc_blockl-16];
-
- for (i=16;i>0;i--) {
- (*tmpW16ptr)=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
- (*tmpW16ptr), (SqrtEnChange+(win>>1)), 14);
- /* multiply by (2.0*SqrtEnChange+win) */
-
- win += inc;
- tmpW16ptr++;
- }
- }
-
- /* Make the linear interpolation between the forward PLC'd data
- and the backward PLC'd data (from the new frame)
- */
-
- if (plc_blockl==40) {
- inc=400; /* 1/41 in Q14 */
- } else { /* plc_blockl==80 */
- inc=202; /* 1/81 in Q14 */
- }
- win=0;
- enh_bufPtr1=&enh_buf[ENH_BUFL-1-iLBCdec_inst->blockl];
- for (i=0; i<plc_blockl; i++) {
- win+=inc;
- *enh_bufPtr1 =
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((*enh_bufPtr1), win, 14);
- *enh_bufPtr1 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
- (16384-win), plc_pred[plc_blockl-1-i], 14);
- enh_bufPtr1--;
- }
- } else {
- WebRtc_Word16 *synt = &downsampled[LPC_FILTERORDER];
-
- enh_bufPtr1=&enh_buf[ENH_BUFL-iLBCdec_inst->blockl-plc_blockl];
- WEBRTC_SPL_MEMCPY_W16(enh_bufPtr1, plc_pred, plc_blockl);
-
- /* Clear fileter memory */
- WebRtcSpl_MemSetW16(iLBCdec_inst->syntMem, 0, LPC_FILTERORDER);
- WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemy, 0, 4);
- WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemx, 0, 2);
-
- /* Initialize filter memory by filtering through 2 lags */
- WEBRTC_SPL_MEMCPY_W16(&synt[-LPC_FILTERORDER], iLBCdec_inst->syntMem,
- LPC_FILTERORDER);
- WebRtcSpl_FilterARFastQ12(
- enh_bufPtr1,
- synt,
- &iLBCdec_inst->old_syntdenum[
- (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
- LPC_FILTERORDER+1, (WebRtc_Word16)lag);
-
- WEBRTC_SPL_MEMCPY_W16(&synt[-LPC_FILTERORDER], &synt[lag-LPC_FILTERORDER],
- LPC_FILTERORDER);
- WebRtcIlbcfix_HpOutput(synt, (WebRtc_Word16*)WebRtcIlbcfix_kHpOutCoefs,
- iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
- (WebRtc_Word16)lag);
- WebRtcSpl_FilterARFastQ12(
- enh_bufPtr1, synt,
- &iLBCdec_inst->old_syntdenum[
- (iLBCdec_inst->nsub-1)*(LPC_FILTERORDER+1)],
- LPC_FILTERORDER+1, (WebRtc_Word16)lag);
-
- WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->syntMem, &synt[lag-LPC_FILTERORDER],
- LPC_FILTERORDER);
- WebRtcIlbcfix_HpOutput(synt, (WebRtc_Word16*)WebRtcIlbcfix_kHpOutCoefs,
- iLBCdec_inst->hpimemy, iLBCdec_inst->hpimemx,
- (WebRtc_Word16)lag);
- }
- }
-
-
- /* Perform enhancement block by block */
-
- for (iblock = 0; iblock<new_blocks; iblock++) {
- WebRtcIlbcfix_Enhancer(out+WEBRTC_SPL_MUL_16_16(iblock, ENH_BLOCKL),
- enh_buf,
- ENH_BUFL,
- (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16(iblock, ENH_BLOCKL)+startPos),
- enh_period,
- (WebRtc_Word16*)WebRtcIlbcfix_kEnhPlocs, ENH_NBLOCKS_TOT);
- }
-
- return (lag);
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/enhancer_interface.h b/trunk/src/modules/audio_coding/codecs/ilbc/enhancer_interface.h
deleted file mode 100644
index 37b27e2..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/enhancer_interface.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_EnhancerInterface.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_ENHANCER_INTERFACE_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * interface for enhancer
- *---------------------------------------------------------------*/
-
-int WebRtcIlbcfix_EnhancerInterface( /* (o) Estimated lag in end of in[] */
- WebRtc_Word16 *out, /* (o) enhanced signal */
- WebRtc_Word16 *in, /* (i) unenhanced signal */
- iLBC_Dec_Inst_t *iLBCdec_inst /* (i) buffers etc */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c b/trunk/src/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
deleted file mode 100644
index 7cece26..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.c
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_FilteredCbVecs.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * Construct an additional codebook vector by filtering the
- * initial codebook buffer. This vector is then used to expand
- * the codebook with an additional section.
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_FilteredCbVecs(
- WebRtc_Word16 *cbvectors, /* (o) Codebook vector for the higher section */
- WebRtc_Word16 *CBmem, /* (i) Codebook memory that is filtered to create a
- second CB section */
- int lMem, /* (i) Length of codebook memory */
- WebRtc_Word16 samples /* (i) Number of samples to filter */
- ) {
-
- /* Set up the memory, start with zero state */
- WebRtcSpl_MemSetW16(CBmem+lMem, 0, CB_HALFFILTERLEN);
- WebRtcSpl_MemSetW16(CBmem-CB_HALFFILTERLEN, 0, CB_HALFFILTERLEN);
- WebRtcSpl_MemSetW16(cbvectors, 0, lMem-samples);
-
- /* Filter to obtain the filtered CB memory */
-
- WebRtcSpl_FilterMAFastQ12(
- CBmem+CB_HALFFILTERLEN+lMem-samples, cbvectors+lMem-samples,
- (WebRtc_Word16*)WebRtcIlbcfix_kCbFiltersRev, CB_FILTERLEN, samples);
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h b/trunk/src/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
deleted file mode 100644
index c502e8f..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/filtered_cb_vecs.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_FilteredCbVecs.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FILTERED_CB_VECS_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Construct an additional codebook vector by filtering the
- * initial codebook buffer. This vector is then used to expand
- * the codebook with an additional section.
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_FilteredCbVecs(
- WebRtc_Word16 *cbvectors, /* (o) Codebook vector for the higher section */
- WebRtc_Word16 *CBmem, /* (i) Codebook memory that is filtered to create a
- second CB section */
- int lMem, /* (i) Length of codebook memory */
- WebRtc_Word16 samples /* (i) Number of samples to filter */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/frame_classify.c b/trunk/src/modules/audio_coding/codecs/ilbc/frame_classify.c
deleted file mode 100644
index ea3675e..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/frame_classify.c
+++ /dev/null
@@ -1,88 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_FrameClassify.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * Classification of subframes to localize start state
- *---------------------------------------------------------------*/
-
-WebRtc_Word16 WebRtcIlbcfix_FrameClassify(
- /* (o) Index to the max-energy sub frame */
- iLBC_Enc_Inst_t *iLBCenc_inst,
- /* (i/o) the encoder state structure */
- WebRtc_Word16 *residualFIX /* (i) lpc residual signal */
- ){
- WebRtc_Word16 max, scale;
- WebRtc_Word32 ssqEn[NSUB_MAX-1];
- WebRtc_Word16 *ssqPtr;
- WebRtc_Word32 *seqEnPtr;
- WebRtc_Word32 maxW32;
- WebRtc_Word16 scale1;
- WebRtc_Word16 pos;
- int n;
-
- /*
- Calculate the energy of each of the 80 sample blocks
- in the draft the 4 first and last samples are windowed with 1/5...4/5
- and 4/5...1/5 respectively. To simplify for the fixpoint we have changed
- this to 0 0 1 1 and 1 1 0 0
- */
-
- max = WebRtcSpl_MaxAbsValueW16(residualFIX, iLBCenc_inst->blockl);
- scale=WebRtcSpl_GetSizeInBits(WEBRTC_SPL_MUL_16_16(max,max));
-
- /* Scale to maximum 24 bits so that it won't overflow for 76 samples */
- scale = scale-24;
- scale1 = WEBRTC_SPL_MAX(0, scale);
-
- /* Calculate energies */
- ssqPtr=residualFIX + 2;
- seqEnPtr=ssqEn;
- for (n=(iLBCenc_inst->nsub-1); n>0; n--) {
- (*seqEnPtr) = WebRtcSpl_DotProductWithScale(ssqPtr, ssqPtr, 76, scale1);
- ssqPtr += 40;
- seqEnPtr++;
- }
-
- /* Scale to maximum 20 bits in order to allow for the 11 bit window */
- maxW32 = WebRtcSpl_MaxValueW32(ssqEn, (WebRtc_Word16)(iLBCenc_inst->nsub-1));
- scale = WebRtcSpl_GetSizeInBits(maxW32) - 20;
- scale1 = WEBRTC_SPL_MAX(0, scale);
-
- /* Window each 80 block with the ssqEn_winTbl window to give higher probability for
- the blocks in the middle
- */
- seqEnPtr=ssqEn;
- if (iLBCenc_inst->mode==20) {
- ssqPtr=(WebRtc_Word16*)WebRtcIlbcfix_kStartSequenceEnrgWin+1;
- } else {
- ssqPtr=(WebRtc_Word16*)WebRtcIlbcfix_kStartSequenceEnrgWin;
- }
- for (n=(iLBCenc_inst->nsub-1); n>0; n--) {
- (*seqEnPtr)=WEBRTC_SPL_MUL(((*seqEnPtr)>>scale1), (*ssqPtr));
- seqEnPtr++;
- ssqPtr++;
- }
-
- /* Extract the best choise of start state */
- pos = WebRtcSpl_MaxIndexW32(ssqEn, (WebRtc_Word16)(iLBCenc_inst->nsub-1)) + 1;
-
- return(pos);
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/frame_classify.h b/trunk/src/modules/audio_coding/codecs/ilbc/frame_classify.h
deleted file mode 100644
index faf4666..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/frame_classify.h
+++ /dev/null
@@ -1,29 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_FrameClassify.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_FRAME_CLASSIFY_H_
-
-WebRtc_Word16 WebRtcIlbcfix_FrameClassify(
- /* (o) Index to the max-energy sub frame */
- iLBC_Enc_Inst_t *iLBCenc_inst,
- /* (i/o) the encoder state structure */
- WebRtc_Word16 *residualFIX /* (i) lpc residual signal */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/gain_dequant.c b/trunk/src/modules/audio_coding/codecs/ilbc/gain_dequant.c
deleted file mode 100644
index 9450a80..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/gain_dequant.c
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_GainDequant.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * decoder for quantized gains in the gain-shape coding of
- * residual
- *---------------------------------------------------------------*/
-
-WebRtc_Word16 WebRtcIlbcfix_GainDequant(
- /* (o) quantized gain value (Q14) */
- WebRtc_Word16 index, /* (i) quantization index */
- WebRtc_Word16 maxIn, /* (i) maximum of unquantized gain (Q14) */
- WebRtc_Word16 stage /* (i) The stage of the search */
- ){
- WebRtc_Word16 scale;
- const WebRtc_Word16 *gain;
-
- /* obtain correct scale factor */
-
- scale=WEBRTC_SPL_ABS_W16(maxIn);
- scale = WEBRTC_SPL_MAX(1638, scale); /* if lower than 0.1, set it to 0.1 */
-
- /* select the quantization table and return the decoded value */
- gain = WebRtcIlbcfix_kGain[stage];
-
- return((WebRtc_Word16)((WEBRTC_SPL_MUL_16_16(scale, gain[index])+8192)>>14));
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/gain_dequant.h b/trunk/src/modules/audio_coding/codecs/ilbc/gain_dequant.h
deleted file mode 100644
index 28f2ceb..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/gain_dequant.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_GainDequant.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_DEQUANT_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * decoder for quantized gains in the gain-shape coding of
- * residual
- *---------------------------------------------------------------*/
-
-WebRtc_Word16 WebRtcIlbcfix_GainDequant(
- /* (o) quantized gain value (Q14) */
- WebRtc_Word16 index, /* (i) quantization index */
- WebRtc_Word16 maxIn, /* (i) maximum of unquantized gain (Q14) */
- WebRtc_Word16 stage /* (i) The stage of the search */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/gain_quant.c b/trunk/src/modules/audio_coding/codecs/ilbc/gain_quant.c
deleted file mode 100644
index bdf88a5..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/gain_quant.c
+++ /dev/null
@@ -1,106 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_GainQuant.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * quantizer for the gain in the gain-shape coding of residual
- *---------------------------------------------------------------*/
-
-WebRtc_Word16 WebRtcIlbcfix_GainQuant( /* (o) quantized gain value */
- WebRtc_Word16 gain, /* (i) gain value Q14 */
- WebRtc_Word16 maxIn, /* (i) maximum of gain value Q14 */
- WebRtc_Word16 stage, /* (i) The stage of the search */
- WebRtc_Word16 *index /* (o) quantization index */
- ) {
-
- WebRtc_Word16 scale, returnVal, cblen;
- WebRtc_Word32 gainW32, measure1, measure2;
- const WebRtc_Word16 *cbPtr, *cb;
- int loc, noMoves, noChecks, i;
-
- /* ensure a lower bound (0.1) on the scaling factor */
-
- scale = WEBRTC_SPL_MAX(1638, maxIn);
-
- /* select the quantization table and calculate
- the length of the table and the number of
- steps in the binary search that are needed */
- cb = WebRtcIlbcfix_kGain[stage];
- cblen = 32>>stage;
- noChecks = 4-stage;
-
- /* Multiply the gain with 2^14 to make the comparison
- easier and with higher precision */
- gainW32 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)gain, 14);
-
- /* Do a binary search, starting in the middle of the CB
- loc - defines the current position in the table
- noMoves - defines the number of steps to move in the CB in order
- to get next CB location
- */
-
- loc = cblen>>1;
- noMoves = loc;
- cbPtr = cb + loc; /* Centre of CB */
-
- for (i=noChecks;i>0;i--) {
- noMoves>>=1;
- measure1=WEBRTC_SPL_MUL_16_16(scale, (*cbPtr));
-
- /* Move up if gain is larger, otherwise move down in table */
- measure1 = measure1 - gainW32;
-
- if (0>measure1) {
- cbPtr+=noMoves;
- loc+=noMoves;
- } else {
- cbPtr-=noMoves;
- loc-=noMoves;
- }
- }
-
- /* Check which value is the closest one: loc-1, loc or loc+1 */
-
- measure1=WEBRTC_SPL_MUL_16_16(scale, (*cbPtr));
- if (gainW32>measure1) {
- /* Check against value above loc */
- measure2=WEBRTC_SPL_MUL_16_16(scale, (*(cbPtr+1)));
- if ((measure2-gainW32)<(gainW32-measure1)) {
- loc+=1;
- }
- } else {
- /* Check against value below loc */
- measure2=WEBRTC_SPL_MUL_16_16(scale, (*(cbPtr-1)));
- if ((gainW32-measure2)<=(measure1-gainW32)) {
- loc-=1;
- }
- }
-
- /* Guard against getting outside the table. The calculation above can give a location
- which is one above the maximum value (in very rare cases) */
- loc=WEBRTC_SPL_MIN(loc, (cblen-1));
- *index=loc;
-
- /* Calculate the quantized gain value (in Q14) */
- returnVal=(WebRtc_Word16)((WEBRTC_SPL_MUL_16_16(scale, cb[loc])+8192)>>14);
-
- /* return the quantized value */
- return(returnVal);
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/gain_quant.h b/trunk/src/modules/audio_coding/codecs/ilbc/gain_quant.h
deleted file mode 100644
index a2f0596..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/gain_quant.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_GainQuant.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GAIN_QUANT_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * quantizer for the gain in the gain-shape coding of residual
- *---------------------------------------------------------------*/
-
-WebRtc_Word16 WebRtcIlbcfix_GainQuant( /* (o) quantized gain value */
- WebRtc_Word16 gain, /* (i) gain value Q14 */
- WebRtc_Word16 maxIn, /* (i) maximum of gain value Q14 */
- WebRtc_Word16 stage, /* (i) The stage of the search */
- WebRtc_Word16 *index /* (o) quantization index */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/get_cd_vec.c b/trunk/src/modules/audio_coding/codecs/ilbc/get_cd_vec.c
deleted file mode 100644
index aba3e31..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/get_cd_vec.c
+++ /dev/null
@@ -1,111 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_GetCbVec.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-#include "create_augmented_vec.h"
-
-/*----------------------------------------------------------------*
- * Construct codebook vector for given index.
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_GetCbVec(
- WebRtc_Word16 *cbvec, /* (o) Constructed codebook vector */
- WebRtc_Word16 *mem, /* (i) Codebook buffer */
- WebRtc_Word16 index, /* (i) Codebook index */
- WebRtc_Word16 lMem, /* (i) Length of codebook buffer */
- WebRtc_Word16 cbveclen /* (i) Codebook vector length */
- ){
- WebRtc_Word16 k, base_size;
- WebRtc_Word16 lag;
- /* Stack based */
- WebRtc_Word16 tempbuff2[SUBL+5];
-
- /* Determine size of codebook sections */
-
- base_size=lMem-cbveclen+1;
-
- if (cbveclen==SUBL) {
- base_size+=WEBRTC_SPL_RSHIFT_W16(cbveclen,1);
- }
-
- /* No filter -> First codebook section */
-
- if (index<lMem-cbveclen+1) {
-
- /* first non-interpolated vectors */
-
- k=index+cbveclen;
- /* get vector */
- WEBRTC_SPL_MEMCPY_W16(cbvec, mem+lMem-k, cbveclen);
-
- } else if (index < base_size) {
-
- /* Calculate lag */
-
- k=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(2, (index-(lMem-cbveclen+1)))+cbveclen;
-
- lag=WEBRTC_SPL_RSHIFT_W16(k, 1);
-
- WebRtcIlbcfix_CreateAugmentedVec(lag, mem+lMem, cbvec);
-
- }
-
- /* Higher codebbok section based on filtering */
-
- else {
-
- WebRtc_Word16 memIndTest;
-
- /* first non-interpolated vectors */
-
- if (index-base_size<lMem-cbveclen+1) {
-
- /* Set up filter memory, stuff zeros outside memory buffer */
-
- memIndTest = lMem-(index-base_size+cbveclen);
-
- WebRtcSpl_MemSetW16(mem-CB_HALFFILTERLEN, 0, CB_HALFFILTERLEN);
- WebRtcSpl_MemSetW16(mem+lMem, 0, CB_HALFFILTERLEN);
-
- /* do filtering to get the codebook vector */
-
- WebRtcSpl_FilterMAFastQ12(
- &mem[memIndTest+4], cbvec, (WebRtc_Word16*)WebRtcIlbcfix_kCbFiltersRev,
- CB_FILTERLEN, cbveclen);
- }
-
- /* interpolated vectors */
-
- else {
- /* Stuff zeros outside memory buffer */
- memIndTest = lMem-cbveclen-CB_FILTERLEN;
- WebRtcSpl_MemSetW16(mem+lMem, 0, CB_HALFFILTERLEN);
-
- /* do filtering */
- WebRtcSpl_FilterMAFastQ12(
- &mem[memIndTest+7], tempbuff2, (WebRtc_Word16*)WebRtcIlbcfix_kCbFiltersRev,
- CB_FILTERLEN, (WebRtc_Word16)(cbveclen+5));
-
- /* Calculate lag index */
- lag = (cbveclen<<1)-20+index-base_size-lMem-1;
-
- WebRtcIlbcfix_CreateAugmentedVec(lag, tempbuff2+SUBL+5, cbvec);
- }
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/get_cd_vec.h b/trunk/src/modules/audio_coding/codecs/ilbc/get_cd_vec.h
deleted file mode 100644
index 99b5d4e..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/get_cd_vec.h
+++ /dev/null
@@ -1,30 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_GetCbVec.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_CD_VEC_H_
-
-void WebRtcIlbcfix_GetCbVec(
- WebRtc_Word16 *cbvec, /* (o) Constructed codebook vector */
- WebRtc_Word16 *mem, /* (i) Codebook buffer */
- WebRtc_Word16 index, /* (i) Codebook index */
- WebRtc_Word16 lMem, /* (i) Length of codebook buffer */
- WebRtc_Word16 cbveclen /* (i) Codebook vector length */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/get_lsp_poly.c b/trunk/src/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
deleted file mode 100644
index c55e918..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/get_lsp_poly.c
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_GetLspPoly.c
-
-******************************************************************/
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Construct the polynomials F1(z) and F2(z) from the LSP
- * (Computations are done in Q24)
- *
- * The expansion is performed using the following recursion:
- *
- * f[0] = 1;
- * tmp = -2.0 * lsp[0];
- * f[1] = tmp;
- * for (i=2; i<=5; i++) {
- * b = -2.0 * lsp[2*i-2];
- * f[i] = tmp*f[i-1] + 2.0*f[i-2];
- * for (j=i; j>=2; j--) {
- * f[j] = f[j] + tmp*f[j-1] + f[j-2];
- * }
- * f[i] = f[i] + tmp;
- * }
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_GetLspPoly(
- WebRtc_Word16 *lsp, /* (i) LSP in Q15 */
- WebRtc_Word32 *f) /* (o) polonymial in Q24 */
-{
- WebRtc_Word32 tmpW32;
- int i, j;
- WebRtc_Word16 high, low;
- WebRtc_Word16 *lspPtr;
- WebRtc_Word32 *fPtr;
-
- lspPtr = lsp;
- fPtr = f;
- /* f[0] = 1.0 (Q24) */
- (*fPtr) = (WebRtc_Word32)16777216;
- fPtr++;
-
- (*fPtr) = WEBRTC_SPL_MUL((*lspPtr), -1024);
- fPtr++;
- lspPtr+=2;
-
- for(i=2; i<=5; i++)
- {
- (*fPtr) = fPtr[-2];
-
- for(j=i; j>1; j--)
- {
- /* Compute f[j] = f[j] + tmp*f[j-1] + f[j-2]; */
- high = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(fPtr[-1], 16);
- low = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(fPtr[-1]-WEBRTC_SPL_LSHIFT_W32(((WebRtc_Word32)high),16), 1);
-
- tmpW32 = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(high, (*lspPtr)), 2) +
- WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16_RSFT(low, (*lspPtr), 15), 2);
-
- (*fPtr) += fPtr[-2];
- (*fPtr) -= tmpW32;
- fPtr--;
- }
- (*fPtr) -= (WebRtc_Word32)WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)(*lspPtr), 10);
-
- fPtr+=i;
- lspPtr+=2;
- }
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/get_lsp_poly.h b/trunk/src/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
deleted file mode 100644
index b0520b4..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/get_lsp_poly.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_GetLspPoly.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_LSP_POLY_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Construct the polynomials F1(z) and F2(z) from the LSP
- * (Computations are done in Q24)
- *
- * The expansion is performed using the following recursion:
- *
- * f[0] = 1;
- * tmp = -2.0 * lsp[0];
- * f[1] = tmp;
- * for (i=2; i<=5; i++) {
- * b = -2.0 * lsp[2*i-2];
- * f[i] = tmp*f[i-1] + 2.0*f[i-2];
- * for (j=i; j>=2; j--) {
- * f[j] = f[j] + tmp*f[j-1] + f[j-2];
- * }
- * f[i] = f[i] + tmp;
- * }
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_GetLspPoly(
- WebRtc_Word16 *lsp, /* (i) LSP in Q15 */
- WebRtc_Word32 *f); /* (o) polonymial in Q24 */
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/get_sync_seq.c b/trunk/src/modules/audio_coding/codecs/ilbc/get_sync_seq.c
deleted file mode 100644
index ce72865..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/get_sync_seq.c
+++ /dev/null
@@ -1,108 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_GetSyncSeq.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-#include "refiner.h"
-#include "nearest_neighbor.h"
-
-/*----------------------------------------------------------------*
- * get the pitch-synchronous sample sequence
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_GetSyncSeq(
- WebRtc_Word16 *idata, /* (i) original data */
- WebRtc_Word16 idatal, /* (i) dimension of data */
- WebRtc_Word16 centerStartPos, /* (i) where current block starts */
- WebRtc_Word16 *period, /* (i) rough-pitch-period array (Q-2) */
- WebRtc_Word16 *plocs, /* (i) where periods of period array are taken (Q-2) */
- WebRtc_Word16 periodl, /* (i) dimension period array */
- WebRtc_Word16 hl, /* (i) 2*hl+1 is the number of sequences */
- WebRtc_Word16 *surround /* (i/o) The contribution from this sequence
- summed with earlier contributions */
- ){
- WebRtc_Word16 i,centerEndPos,q;
- /* Stack based */
- WebRtc_Word16 lagBlock[2*ENH_HL+1];
- WebRtc_Word16 blockStartPos[2*ENH_HL+1]; /* Defines the position to search around (Q2) */
- WebRtc_Word16 plocs2[ENH_PLOCSL];
-
- centerEndPos=centerStartPos+ENH_BLOCKL-1;
-
- /* present (find predicted lag from this position) */
-
- WebRtcIlbcfix_NearestNeighbor(lagBlock+hl,plocs,
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(2, (centerStartPos+centerEndPos)),
- periodl);
-
- blockStartPos[hl]=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, centerStartPos);
-
- /* past (find predicted position and perform a refined
- search to find the best sequence) */
-
- for(q=hl-1;q>=0;q--) {
- blockStartPos[q]=blockStartPos[q+1]-period[lagBlock[q+1]];
-
- WebRtcIlbcfix_NearestNeighbor(lagBlock+q, plocs,
- (WebRtc_Word16)(blockStartPos[q] + (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, ENH_BLOCKL_HALF)-period[lagBlock[q+1]]),
- periodl);
-
- if((blockStartPos[q]-(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, ENH_OVERHANG))>=0) {
-
- /* Find the best possible sequence in the 4 times upsampled
- domain around blockStartPos+q */
- WebRtcIlbcfix_Refiner(blockStartPos+q,idata,idatal,
- centerStartPos,blockStartPos[q],surround,WebRtcIlbcfix_kEnhWt[q]);
-
- } else {
- /* Don't add anything since this sequence would
- be outside the buffer */
- }
- }
-
- /* future (find predicted position and perform a refined
- search to find the best sequence) */
-
- for(i=0;i<periodl;i++) {
- plocs2[i]=(plocs[i]-period[i]);
- }
-
- for(q=hl+1;q<=WEBRTC_SPL_MUL_16_16(2, hl);q++) {
-
- WebRtcIlbcfix_NearestNeighbor(lagBlock+q,plocs2,
- (WebRtc_Word16)(blockStartPos[q-1]+
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, ENH_BLOCKL_HALF)),periodl);
-
- blockStartPos[q]=blockStartPos[q-1]+period[lagBlock[q]];
-
- if( (blockStartPos[q]+(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, (ENH_BLOCKL+ENH_OVERHANG)))
- <
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(4, idatal)) {
-
- /* Find the best possible sequence in the 4 times upsampled
- domain around blockStartPos+q */
- WebRtcIlbcfix_Refiner(blockStartPos+q, idata, idatal,
- centerStartPos,blockStartPos[q],surround,WebRtcIlbcfix_kEnhWt[2*hl-q]);
-
- }
- else {
- /* Don't add anything since this sequence would
- be outside the buffer */
- }
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/get_sync_seq.h b/trunk/src/modules/audio_coding/codecs/ilbc/get_sync_seq.h
deleted file mode 100644
index a0ffd39..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/get_sync_seq.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_GetSyncSeq.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_GET_SYNC_SEQ_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * get the pitch-synchronous sample sequence
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_GetSyncSeq(
- WebRtc_Word16 *idata, /* (i) original data */
- WebRtc_Word16 idatal, /* (i) dimension of data */
- WebRtc_Word16 centerStartPos, /* (i) where current block starts */
- WebRtc_Word16 *period, /* (i) rough-pitch-period array (Q-2) */
- WebRtc_Word16 *plocs, /* (i) where periods of period array are taken (Q-2) */
- WebRtc_Word16 periodl, /* (i) dimension period array */
- WebRtc_Word16 hl, /* (i) 2*hl+1 is the number of sequences */
- WebRtc_Word16 *surround /* (i/o) The contribution from this sequence
- summed with earlier contributions */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/hp_input.c b/trunk/src/modules/audio_coding/codecs/ilbc/hp_input.c
deleted file mode 100644
index f202f62..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/hp_input.c
+++ /dev/null
@@ -1,88 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_HpInput.c
-
-******************************************************************/
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * high-pass filter of input with *0.5 and saturation
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_HpInput(
- WebRtc_Word16 *signal, /* (i/o) signal vector */
- WebRtc_Word16 *ba, /* (i) B- and A-coefficients (2:nd order)
- {b[0] b[1] b[2] -a[1] -a[2]} a[0]
- is assumed to be 1.0 */
- WebRtc_Word16 *y, /* (i/o) Filter state yhi[n-1] ylow[n-1]
- yhi[n-2] ylow[n-2] */
- WebRtc_Word16 *x, /* (i/o) Filter state x[n-1] x[n-2] */
- WebRtc_Word16 len) /* (i) Number of samples to filter */
-{
- int i;
- WebRtc_Word32 tmpW32;
- WebRtc_Word32 tmpW32b;
-
- for (i=0; i<len; i++) {
-
- /*
- y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2]
- + (-a[1])*y[i-1] + (-a[2])*y[i-2];
- */
-
- tmpW32 = WEBRTC_SPL_MUL_16_16(y[1], ba[3]); /* (-a[1])*y[i-1] (low part) */
- tmpW32 += WEBRTC_SPL_MUL_16_16(y[3], ba[4]); /* (-a[2])*y[i-2] (low part) */
- tmpW32 = (tmpW32>>15);
- tmpW32 += WEBRTC_SPL_MUL_16_16(y[0], ba[3]); /* (-a[1])*y[i-1] (high part) */
- tmpW32 += WEBRTC_SPL_MUL_16_16(y[2], ba[4]); /* (-a[2])*y[i-2] (high part) */
- tmpW32 = (tmpW32<<1);
-
- tmpW32 += WEBRTC_SPL_MUL_16_16(signal[i], ba[0]); /* b[0]*x[0] */
- tmpW32 += WEBRTC_SPL_MUL_16_16(x[0], ba[1]); /* b[1]*x[i-1] */
- tmpW32 += WEBRTC_SPL_MUL_16_16(x[1], ba[2]); /* b[2]*x[i-2] */
-
- /* Update state (input part) */
- x[1] = x[0];
- x[0] = signal[i];
-
- /* Rounding in Q(12+1), i.e. add 2^12 */
- tmpW32b = tmpW32 + 4096;
-
- /* Saturate (to 2^28) so that the HP filtered signal does not overflow */
- tmpW32b = WEBRTC_SPL_SAT((WebRtc_Word32)268435455, tmpW32b, (WebRtc_Word32)-268435456);
-
- /* Convert back to Q0 and multiply with 0.5 */
- signal[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32b, 13);
-
- /* Update state (filtered part) */
- y[2] = y[0];
- y[3] = y[1];
-
- /* upshift tmpW32 by 3 with saturation */
- if (tmpW32>268435455) {
- tmpW32 = WEBRTC_SPL_WORD32_MAX;
- } else if (tmpW32<-268435456) {
- tmpW32 = WEBRTC_SPL_WORD32_MIN;
- } else {
- tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);
- }
-
- y[0] = (WebRtc_Word16)(tmpW32 >> 16);
- y[1] = (WebRtc_Word16)((tmpW32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)y[0], 16))>>1);
- }
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/hp_input.h b/trunk/src/modules/audio_coding/codecs/ilbc/hp_input.h
deleted file mode 100644
index f56c4f7..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/hp_input.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_HpInput.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_INPUT_H_
-
-#include "defines.h"
-
-void WebRtcIlbcfix_HpInput(
- WebRtc_Word16 *signal, /* (i/o) signal vector */
- WebRtc_Word16 *ba, /* (i) B- and A-coefficients (2:nd order)
- {b[0] b[1] b[2] -a[1] -a[2]} a[0]
- is assumed to be 1.0 */
- WebRtc_Word16 *y, /* (i/o) Filter state yhi[n-1] ylow[n-1]
- yhi[n-2] ylow[n-2] */
- WebRtc_Word16 *x, /* (i/o) Filter state x[n-1] x[n-2] */
- WebRtc_Word16 len); /* (i) Number of samples to filter */
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/hp_output.c b/trunk/src/modules/audio_coding/codecs/ilbc/hp_output.c
deleted file mode 100644
index 8e1c919..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/hp_output.c
+++ /dev/null
@@ -1,89 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_HpOutput.c
-
-******************************************************************/
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * high-pass filter of output and *2 with saturation
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_HpOutput(
- WebRtc_Word16 *signal, /* (i/o) signal vector */
- WebRtc_Word16 *ba, /* (i) B- and A-coefficients (2:nd order)
- {b[0] b[1] b[2] -a[1] -a[2]} a[0]
- is assumed to be 1.0 */
- WebRtc_Word16 *y, /* (i/o) Filter state yhi[n-1] ylow[n-1]
- yhi[n-2] ylow[n-2] */
- WebRtc_Word16 *x, /* (i/o) Filter state x[n-1] x[n-2] */
- WebRtc_Word16 len) /* (i) Number of samples to filter */
-{
- int i;
- WebRtc_Word32 tmpW32;
- WebRtc_Word32 tmpW32b;
-
- for (i=0; i<len; i++) {
-
- /*
- y[i] = b[0]*x[i] + b[1]*x[i-1] + b[2]*x[i-2]
- + (-a[1])*y[i-1] + (-a[2])*y[i-2];
- */
-
- tmpW32 = WEBRTC_SPL_MUL_16_16(y[1], ba[3]); /* (-a[1])*y[i-1] (low part) */
- tmpW32 += WEBRTC_SPL_MUL_16_16(y[3], ba[4]); /* (-a[2])*y[i-2] (low part) */
- tmpW32 = (tmpW32>>15);
- tmpW32 += WEBRTC_SPL_MUL_16_16(y[0], ba[3]); /* (-a[1])*y[i-1] (high part) */
- tmpW32 += WEBRTC_SPL_MUL_16_16(y[2], ba[4]); /* (-a[2])*y[i-2] (high part) */
- tmpW32 = (tmpW32<<1);
-
- tmpW32 += WEBRTC_SPL_MUL_16_16(signal[i], ba[0]); /* b[0]*x[0] */
- tmpW32 += WEBRTC_SPL_MUL_16_16(x[0], ba[1]); /* b[1]*x[i-1] */
- tmpW32 += WEBRTC_SPL_MUL_16_16(x[1], ba[2]); /* b[2]*x[i-2] */
-
- /* Update state (input part) */
- x[1] = x[0];
- x[0] = signal[i];
-
- /* Rounding in Q(12-1), i.e. add 2^10 */
- tmpW32b = tmpW32 + 1024;
-
- /* Saturate (to 2^26) so that the HP filtered signal does not overflow */
- tmpW32b = WEBRTC_SPL_SAT((WebRtc_Word32)67108863, tmpW32b, (WebRtc_Word32)-67108864);
-
- /* Convert back to Q0 and multiply with 2 */
- signal[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmpW32b, 11);
-
- /* Update state (filtered part) */
- y[2] = y[0];
- y[3] = y[1];
-
- /* upshift tmpW32 by 3 with saturation */
- if (tmpW32>268435455) {
- tmpW32 = WEBRTC_SPL_WORD32_MAX;
- } else if (tmpW32<-268435456) {
- tmpW32 = WEBRTC_SPL_WORD32_MIN;
- } else {
- tmpW32 = WEBRTC_SPL_LSHIFT_W32(tmpW32, 3);
- }
-
- y[0] = (WebRtc_Word16)(tmpW32 >> 16);
- y[1] = (WebRtc_Word16)((tmpW32 - WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)y[0], 16))>>1);
-
- }
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/hp_output.h b/trunk/src/modules/audio_coding/codecs/ilbc/hp_output.h
deleted file mode 100644
index c9a7426..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/hp_output.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_HpOutput.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_HP_OUTPUT_H_
-
-#include "defines.h"
-
-void WebRtcIlbcfix_HpOutput(
- WebRtc_Word16 *signal, /* (i/o) signal vector */
- WebRtc_Word16 *ba, /* (i) B- and A-coefficients (2:nd order)
- {b[0] b[1] b[2] -a[1] -a[2]} a[0]
- is assumed to be 1.0 */
- WebRtc_Word16 *y, /* (i/o) Filter state yhi[n-1] ylow[n-1]
- yhi[n-2] ylow[n-2] */
- WebRtc_Word16 *x, /* (i/o) Filter state x[n-1] x[n-2] */
- WebRtc_Word16 len); /* (i) Number of samples to filter */
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/ilbc.c b/trunk/src/modules/audio_coding/codecs/ilbc/ilbc.c
deleted file mode 100644
index 75b64c4..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/ilbc.c
+++ /dev/null
@@ -1,255 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- iLBCInterface.c
-
-******************************************************************/
-
-#include "ilbc.h"
-#include "defines.h"
-#include "init_encode.h"
-#include "encode.h"
-#include "init_decode.h"
-#include "decode.h"
-#include <stdlib.h>
-
-
-WebRtc_Word16 WebRtcIlbcfix_EncoderAssign(iLBC_encinst_t **iLBC_encinst, WebRtc_Word16 *ILBCENC_inst_Addr, WebRtc_Word16 *size) {
- *iLBC_encinst=(iLBC_encinst_t*)ILBCENC_inst_Addr;
- *size=sizeof(iLBC_Enc_Inst_t)/sizeof(WebRtc_Word16);
- if (*iLBC_encinst!=NULL) {
- return(0);
- } else {
- return(-1);
- }
-}
-
-WebRtc_Word16 WebRtcIlbcfix_DecoderAssign(iLBC_decinst_t **iLBC_decinst, WebRtc_Word16 *ILBCDEC_inst_Addr, WebRtc_Word16 *size) {
- *iLBC_decinst=(iLBC_decinst_t*)ILBCDEC_inst_Addr;
- *size=sizeof(iLBC_Dec_Inst_t)/sizeof(WebRtc_Word16);
- if (*iLBC_decinst!=NULL) {
- return(0);
- } else {
- return(-1);
- }
-}
-
-WebRtc_Word16 WebRtcIlbcfix_EncoderCreate(iLBC_encinst_t **iLBC_encinst) {
- *iLBC_encinst=(iLBC_encinst_t*)malloc(sizeof(iLBC_Enc_Inst_t));
- if (*iLBC_encinst!=NULL) {
- return(0);
- } else {
- return(-1);
- }
-}
-
-WebRtc_Word16 WebRtcIlbcfix_DecoderCreate(iLBC_decinst_t **iLBC_decinst) {
- *iLBC_decinst=(iLBC_decinst_t*)malloc(sizeof(iLBC_Dec_Inst_t));
- if (*iLBC_decinst!=NULL) {
- return(0);
- } else {
- return(-1);
- }
-}
-
-WebRtc_Word16 WebRtcIlbcfix_EncoderFree(iLBC_encinst_t *iLBC_encinst) {
- free(iLBC_encinst);
- return(0);
-}
-
-WebRtc_Word16 WebRtcIlbcfix_DecoderFree(iLBC_decinst_t *iLBC_decinst) {
- free(iLBC_decinst);
- return(0);
-}
-
-
-WebRtc_Word16 WebRtcIlbcfix_EncoderInit(iLBC_encinst_t *iLBCenc_inst, WebRtc_Word16 mode)
-{
- if ((mode==20)||(mode==30)) {
- WebRtcIlbcfix_InitEncode((iLBC_Enc_Inst_t*) iLBCenc_inst, mode);
- return(0);
- } else {
- return(-1);
- }
-}
-
-WebRtc_Word16 WebRtcIlbcfix_Encode(iLBC_encinst_t *iLBCenc_inst, WebRtc_Word16 *speechIn, WebRtc_Word16 len, WebRtc_Word16 *encoded) {
-
- WebRtc_Word16 pos = 0;
- WebRtc_Word16 encpos = 0;
-
- if ((len != ((iLBC_Enc_Inst_t*)iLBCenc_inst)->blockl) &&
-#ifdef SPLIT_10MS
- (len != 80) &&
-#endif
- (len != 2*((iLBC_Enc_Inst_t*)iLBCenc_inst)->blockl) &&
- (len != 3*((iLBC_Enc_Inst_t*)iLBCenc_inst)->blockl))
- {
- /* A maximum of 3 frames/packet is allowed */
- return(-1);
- } else {
-
- /* call encoder */
- while (pos<len) {
- WebRtcIlbcfix_EncodeImpl((WebRtc_UWord16*) &encoded[encpos], &speechIn[pos], (iLBC_Enc_Inst_t*) iLBCenc_inst);
-#ifdef SPLIT_10MS
- pos += 80;
- if(((iLBC_Enc_Inst_t*)iLBCenc_inst)->section == 0)
-#else
- pos += ((iLBC_Enc_Inst_t*)iLBCenc_inst)->blockl;
-#endif
- encpos += ((iLBC_Enc_Inst_t*)iLBCenc_inst)->no_of_words;
- }
- return (encpos*2);
- }
-}
-
-WebRtc_Word16 WebRtcIlbcfix_DecoderInit(iLBC_decinst_t *iLBCdec_inst, WebRtc_Word16 mode) {
- if ((mode==20)||(mode==30)) {
- WebRtcIlbcfix_InitDecode((iLBC_Dec_Inst_t*) iLBCdec_inst, mode, 1);
- return(0);
- } else {
- return(-1);
- }
-}
-WebRtc_Word16 WebRtcIlbcfix_DecoderInit20Ms(iLBC_decinst_t *iLBCdec_inst) {
- WebRtcIlbcfix_InitDecode((iLBC_Dec_Inst_t*) iLBCdec_inst, 20, 1);
- return(0);
-}
-WebRtc_Word16 WebRtcIlbcfix_Decoderinit30Ms(iLBC_decinst_t *iLBCdec_inst) {
- WebRtcIlbcfix_InitDecode((iLBC_Dec_Inst_t*) iLBCdec_inst, 30, 1);
- return(0);
-}
-
-
-WebRtc_Word16 WebRtcIlbcfix_Decode(iLBC_decinst_t *iLBCdec_inst,
- WebRtc_Word16 *encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType)
-{
- int i=0;
- /* Allow for automatic switching between the frame sizes
- (although you do get some discontinuity) */
- if ((len==((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
- (len==2*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
- (len==3*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)) {
- /* ok, do nothing */
- } else {
- /* Test if the mode has changed */
- if (((iLBC_Dec_Inst_t*)iLBCdec_inst)->mode==20) {
- if ((len==NO_OF_BYTES_30MS)||
- (len==2*NO_OF_BYTES_30MS)||
- (len==3*NO_OF_BYTES_30MS)) {
- WebRtcIlbcfix_InitDecode(((iLBC_Dec_Inst_t*)iLBCdec_inst), 30, ((iLBC_Dec_Inst_t*)iLBCdec_inst)->use_enhancer);
- } else {
- /* Unsupported frame length */
- return(-1);
- }
- } else {
- if ((len==NO_OF_BYTES_20MS)||
- (len==2*NO_OF_BYTES_20MS)||
- (len==3*NO_OF_BYTES_20MS)) {
- WebRtcIlbcfix_InitDecode(((iLBC_Dec_Inst_t*)iLBCdec_inst), 20, ((iLBC_Dec_Inst_t*)iLBCdec_inst)->use_enhancer);
- } else {
- /* Unsupported frame length */
- return(-1);
- }
- }
- }
-
- while ((i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)<len) {
- WebRtcIlbcfix_DecodeImpl(&decoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl], (WebRtc_UWord16*) &encoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_words], (iLBC_Dec_Inst_t*) iLBCdec_inst, 1);
- i++;
- }
- /* iLBC does not support VAD/CNG yet */
- *speechType=1;
- return(i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl);
-}
-
-WebRtc_Word16 WebRtcIlbcfix_Decode20Ms(iLBC_decinst_t *iLBCdec_inst,
- WebRtc_Word16 *encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType)
-{
- int i=0;
- if ((len==((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
- (len==2*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
- (len==3*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)) {
- /* ok, do nothing */
- } else {
- return(-1);
- }
-
- while ((i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)<len) {
- WebRtcIlbcfix_DecodeImpl(&decoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl], (WebRtc_UWord16*) &encoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_words], (iLBC_Dec_Inst_t*) iLBCdec_inst, 1);
- i++;
- }
- /* iLBC does not support VAD/CNG yet */
- *speechType=1;
- return(i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl);
-}
-
-WebRtc_Word16 WebRtcIlbcfix_Decode30Ms(iLBC_decinst_t *iLBCdec_inst,
- WebRtc_Word16 *encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType)
-{
- int i=0;
- if ((len==((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
- (len==2*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)||
- (len==3*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)) {
- /* ok, do nothing */
- } else {
- return(-1);
- }
-
- while ((i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_bytes)<len) {
- WebRtcIlbcfix_DecodeImpl(&decoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl], (WebRtc_UWord16*) &encoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->no_of_words], (iLBC_Dec_Inst_t*) iLBCdec_inst, 1);
- i++;
- }
- /* iLBC does not support VAD/CNG yet */
- *speechType=1;
- return(i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl);
-}
-
-WebRtc_Word16 WebRtcIlbcfix_DecodePlc(iLBC_decinst_t *iLBCdec_inst, WebRtc_Word16 *decoded, WebRtc_Word16 noOfLostFrames) {
- int i;
- WebRtc_UWord16 dummy;
-
- for (i=0;i<noOfLostFrames;i++) {
- /* call decoder */
- WebRtcIlbcfix_DecodeImpl(&decoded[i*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl], &dummy, (iLBC_Dec_Inst_t*) iLBCdec_inst, 0);
- }
- return (noOfLostFrames*((iLBC_Dec_Inst_t*)iLBCdec_inst)->blockl);
-}
-
-WebRtc_Word16 WebRtcIlbcfix_NetEqPlc(iLBC_decinst_t *iLBCdec_inst, WebRtc_Word16 *decoded, WebRtc_Word16 noOfLostFrames) {
-
- /* Two input parameters not used, but needed for function pointers in NetEQ */
- (void)(decoded = NULL);
- (void)(noOfLostFrames = 0);
-
- WebRtcSpl_MemSetW16(((iLBC_Dec_Inst_t*)iLBCdec_inst)->enh_buf, 0, ENH_BUFL);
- ((iLBC_Dec_Inst_t*)iLBCdec_inst)->prev_enh_pl = 2;
-
- return (0);
-}
-
-void WebRtcIlbcfix_version(char *version)
-{
- strcpy((char*)version, "1.1.1");
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/ilbc.gypi b/trunk/src/modules/audio_coding/codecs/ilbc/ilbc.gypi
deleted file mode 100644
index 57410c5..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/ilbc.gypi
+++ /dev/null
@@ -1,190 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'iLBC',
- 'type': '<(library)',
- 'dependencies': [
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- ],
- 'include_dirs': [
- 'interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'interface',
- ],
- },
- 'sources': [
- 'interface/ilbc.h',
- 'abs_quant.c',
- 'abs_quant_loop.c',
- 'augmented_cb_corr.c',
- 'bw_expand.c',
- 'cb_construct.c',
- 'cb_mem_energy.c',
- 'cb_mem_energy_augmentation.c',
- 'cb_mem_energy_calc.c',
- 'cb_search.c',
- 'cb_search_core.c',
- 'cb_update_best_index.c',
- 'chebyshev.c',
- 'comp_corr.c',
- 'constants.c',
- 'create_augmented_vec.c',
- 'decode.c',
- 'decode_residual.c',
- 'decoder_interpolate_lsf.c',
- 'do_plc.c',
- 'encode.c',
- 'energy_inverse.c',
- 'enh_upsample.c',
- 'enhancer.c',
- 'enhancer_interface.c',
- 'filtered_cb_vecs.c',
- 'frame_classify.c',
- 'gain_dequant.c',
- 'gain_quant.c',
- 'get_cd_vec.c',
- 'get_lsp_poly.c',
- 'get_sync_seq.c',
- 'hp_input.c',
- 'hp_output.c',
- 'ilbc.c',
- 'index_conv_dec.c',
- 'index_conv_enc.c',
- 'init_decode.c',
- 'init_encode.c',
- 'interpolate.c',
- 'interpolate_samples.c',
- 'lpc_encode.c',
- 'lsf_check.c',
- 'lsf_interpolate_to_poly_dec.c',
- 'lsf_interpolate_to_poly_enc.c',
- 'lsf_to_lsp.c',
- 'lsf_to_poly.c',
- 'lsp_to_lsf.c',
- 'my_corr.c',
- 'nearest_neighbor.c',
- 'pack_bits.c',
- 'poly_to_lsf.c',
- 'poly_to_lsp.c',
- 'refiner.c',
- 'simple_interpolate_lsf.c',
- 'simple_lpc_analysis.c',
- 'simple_lsf_dequant.c',
- 'simple_lsf_quant.c',
- 'smooth.c',
- 'smooth_out_data.c',
- 'sort_sq.c',
- 'split_vq.c',
- 'state_construct.c',
- 'state_search.c',
- 'swap_bytes.c',
- 'unpack_bits.c',
- 'vq3.c',
- 'vq4.c',
- 'window32_w32.c',
- 'xcorr_coef.c',
- 'abs_quant.h',
- 'abs_quant_loop.h',
- 'augmented_cb_corr.h',
- 'bw_expand.h',
- 'cb_construct.h',
- 'cb_mem_energy.h',
- 'cb_mem_energy_augmentation.h',
- 'cb_mem_energy_calc.h',
- 'cb_search.h',
- 'cb_search_core.h',
- 'cb_update_best_index.h',
- 'chebyshev.h',
- 'comp_corr.h',
- 'constants.h',
- 'create_augmented_vec.h',
- 'decode.h',
- 'decode_residual.h',
- 'decoder_interpolate_lsf.h',
- 'do_plc.h',
- 'encode.h',
- 'energy_inverse.h',
- 'enh_upsample.h',
- 'enhancer.h',
- 'enhancer_interface.h',
- 'filtered_cb_vecs.h',
- 'frame_classify.h',
- 'gain_dequant.h',
- 'gain_quant.h',
- 'get_cd_vec.h',
- 'get_lsp_poly.h',
- 'get_sync_seq.h',
- 'hp_input.h',
- 'hp_output.h',
- 'defines.h',
- 'index_conv_dec.h',
- 'index_conv_enc.h',
- 'init_decode.h',
- 'init_encode.h',
- 'interpolate.h',
- 'interpolate_samples.h',
- 'lpc_encode.h',
- 'lsf_check.h',
- 'lsf_interpolate_to_poly_dec.h',
- 'lsf_interpolate_to_poly_enc.h',
- 'lsf_to_lsp.h',
- 'lsf_to_poly.h',
- 'lsp_to_lsf.h',
- 'my_corr.h',
- 'nearest_neighbor.h',
- 'pack_bits.h',
- 'poly_to_lsf.h',
- 'poly_to_lsp.h',
- 'refiner.h',
- 'simple_interpolate_lsf.h',
- 'simple_lpc_analysis.h',
- 'simple_lsf_dequant.h',
- 'simple_lsf_quant.h',
- 'smooth.h',
- 'smooth_out_data.h',
- 'sort_sq.h',
- 'split_vq.h',
- 'state_construct.h',
- 'state_search.h',
- 'swap_bytes.h',
- 'unpack_bits.h',
- 'vq3.h',
- 'vq4.h',
- 'window32_w32.h',
- 'xcorr_coef.h',
- ], # sources
- }, # iLBC
- ], # targets
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'iLBCtest',
- 'type': 'executable',
- 'dependencies': [
- 'iLBC',
- ],
- 'sources': [
- 'test/iLBC_test.c',
- ],
- }, # iLBCtest
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_dec.c b/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_dec.c
deleted file mode 100644
index 0d6346a..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_dec.c
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_IndexConvDec.c
-
-******************************************************************/
-
-#include "defines.h"
-
-void WebRtcIlbcfix_IndexConvDec(
- WebRtc_Word16 *index /* (i/o) Codebook indexes */
- ){
- int k;
-
- for (k=4;k<6;k++) {
- /* Readjust the second and third codebook index for the first 40 sample
- so that they look the same as the first (in terms of lag)
- */
- if ((index[k]>=44)&&(index[k]<108)) {
- index[k]+=64;
- } else if ((index[k]>=108)&&(index[k]<128)) {
- index[k]+=128;
- } else {
- /* ERROR */
- }
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_dec.h b/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_dec.h
deleted file mode 100644
index f29ee23..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_dec.h
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_IndexConvDec.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_DEC_H_
-
-#include "defines.h"
-
-void WebRtcIlbcfix_IndexConvDec(
- WebRtc_Word16 *index /* (i/o) Codebook indexes */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_enc.c b/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_enc.c
deleted file mode 100644
index cbc04b6..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_enc.c
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- IiLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_IndexConvEnc.c
-
-******************************************************************/
-
-#include "defines.h"
-/*----------------------------------------------------------------*
- * Convert the codebook indexes to make the search easier
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_IndexConvEnc(
- WebRtc_Word16 *index /* (i/o) Codebook indexes */
- ){
- int k;
-
- for (k=4;k<6;k++) {
- /* Readjust the second and third codebook index so that it is
- packetized into 7 bits (before it was put in lag-wise the same
- way as for the first codebook which uses 8 bits)
- */
- if ((index[k]>=108)&&(index[k]<172)) {
- index[k]-=64;
- } else if (index[k]>=236) {
- index[k]-=128;
- } else {
- /* ERROR */
- }
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_enc.h b/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_enc.h
deleted file mode 100644
index d28a6e2..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/index_conv_enc.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_IndexConvEnc.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INDEX_CONV_ENC_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Convert the codebook indexes to make the search easier
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_IndexConvEnc(
- WebRtc_Word16 *index /* (i/o) Codebook indexes */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/init_decode.c b/trunk/src/modules/audio_coding/codecs/ilbc/init_decode.c
deleted file mode 100644
index b654f1e..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/init_decode.c
+++ /dev/null
@@ -1,98 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_InitDecode.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * Initiation of decoder instance.
- *---------------------------------------------------------------*/
-
-WebRtc_Word16 WebRtcIlbcfix_InitDecode( /* (o) Number of decoded samples */
- iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
- WebRtc_Word16 mode, /* (i) frame size mode */
- int use_enhancer /* (i) 1 to use enhancer
- 0 to run without enhancer */
- ) {
- int i;
-
- iLBCdec_inst->mode = mode;
-
- /* Set all the variables that are dependent on the frame size mode */
- if (mode==30) {
- iLBCdec_inst->blockl = BLOCKL_30MS;
- iLBCdec_inst->nsub = NSUB_30MS;
- iLBCdec_inst->nasub = NASUB_30MS;
- iLBCdec_inst->lpc_n = LPC_N_30MS;
- iLBCdec_inst->no_of_bytes = NO_OF_BYTES_30MS;
- iLBCdec_inst->no_of_words = NO_OF_WORDS_30MS;
- iLBCdec_inst->state_short_len=STATE_SHORT_LEN_30MS;
- }
- else if (mode==20) {
- iLBCdec_inst->blockl = BLOCKL_20MS;
- iLBCdec_inst->nsub = NSUB_20MS;
- iLBCdec_inst->nasub = NASUB_20MS;
- iLBCdec_inst->lpc_n = LPC_N_20MS;
- iLBCdec_inst->no_of_bytes = NO_OF_BYTES_20MS;
- iLBCdec_inst->no_of_words = NO_OF_WORDS_20MS;
- iLBCdec_inst->state_short_len=STATE_SHORT_LEN_20MS;
- }
- else {
- return(-1);
- }
-
- /* Reset all the previous LSF to mean LSF */
- WEBRTC_SPL_MEMCPY_W16(iLBCdec_inst->lsfdeqold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER);
-
- /* Clear the synthesis filter memory */
- WebRtcSpl_MemSetW16(iLBCdec_inst->syntMem, 0, LPC_FILTERORDER);
-
- /* Set the old synthesis filter to {1.0 0.0 ... 0.0} */
- WebRtcSpl_MemSetW16(iLBCdec_inst->old_syntdenum, 0, ((LPC_FILTERORDER + 1)*NSUB_MAX));
- for (i=0; i<NSUB_MAX; i++) {
- iLBCdec_inst->old_syntdenum[i*(LPC_FILTERORDER+1)] = 4096;
- }
-
- /* Clear the variables that are used for the PLC */
- iLBCdec_inst->last_lag = 20;
- iLBCdec_inst->consPLICount = 0;
- iLBCdec_inst->prevPLI = 0;
- iLBCdec_inst->perSquare = 0;
- iLBCdec_inst->prevLag = 120;
- iLBCdec_inst->prevLpc[0] = 4096;
- WebRtcSpl_MemSetW16(iLBCdec_inst->prevLpc+1, 0, LPC_FILTERORDER);
- WebRtcSpl_MemSetW16(iLBCdec_inst->prevResidual, 0, BLOCKL_MAX);
-
- /* Initialize the seed for the random number generator */
- iLBCdec_inst->seed = 777;
-
- /* Set the filter state of the HP filter to 0 */
- WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemx, 0, 2);
- WebRtcSpl_MemSetW16(iLBCdec_inst->hpimemy, 0, 4);
-
- /* Set the variables that are used in the ehnahcer */
- iLBCdec_inst->use_enhancer = use_enhancer;
- WebRtcSpl_MemSetW16(iLBCdec_inst->enh_buf, 0, (ENH_BUFL+ENH_BUFL_FILTEROVERHEAD));
- for (i=0;i<ENH_NBLOCKS_TOT;i++) {
- iLBCdec_inst->enh_period[i]=160; /* Q(-4) */
- }
-
- iLBCdec_inst->prev_enh_pl = 0;
-
- return (iLBCdec_inst->blockl);
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/init_decode.h b/trunk/src/modules/audio_coding/codecs/ilbc/init_decode.h
deleted file mode 100644
index 3452f34..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/init_decode.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_InitDecode.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_DECODE_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Initiation of decoder instance.
- *---------------------------------------------------------------*/
-
-WebRtc_Word16 WebRtcIlbcfix_InitDecode( /* (o) Number of decoded samples */
- iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
- WebRtc_Word16 mode, /* (i) frame size mode */
- int use_enhancer /* (i) 1 to use enhancer
- 0 to run without enhancer */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/init_encode.c b/trunk/src/modules/audio_coding/codecs/ilbc/init_encode.c
deleted file mode 100644
index e034bb0..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/init_encode.c
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_InitEncode.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * Initiation of encoder instance.
- *---------------------------------------------------------------*/
-
-WebRtc_Word16 WebRtcIlbcfix_InitEncode( /* (o) Number of bytes encoded */
- iLBC_Enc_Inst_t *iLBCenc_inst, /* (i/o) Encoder instance */
- WebRtc_Word16 mode /* (i) frame size mode */
- ){
- iLBCenc_inst->mode = mode;
-
- /* Set all the variables that are dependent on the frame size mode */
- if (mode==30) {
- iLBCenc_inst->blockl = BLOCKL_30MS;
- iLBCenc_inst->nsub = NSUB_30MS;
- iLBCenc_inst->nasub = NASUB_30MS;
- iLBCenc_inst->lpc_n = LPC_N_30MS;
- iLBCenc_inst->no_of_bytes = NO_OF_BYTES_30MS;
- iLBCenc_inst->no_of_words = NO_OF_WORDS_30MS;
- iLBCenc_inst->state_short_len=STATE_SHORT_LEN_30MS;
- }
- else if (mode==20) {
- iLBCenc_inst->blockl = BLOCKL_20MS;
- iLBCenc_inst->nsub = NSUB_20MS;
- iLBCenc_inst->nasub = NASUB_20MS;
- iLBCenc_inst->lpc_n = LPC_N_20MS;
- iLBCenc_inst->no_of_bytes = NO_OF_BYTES_20MS;
- iLBCenc_inst->no_of_words = NO_OF_WORDS_20MS;
- iLBCenc_inst->state_short_len=STATE_SHORT_LEN_20MS;
- }
- else {
- return(-1);
- }
-
- /* Clear the buffers and set the previous LSF and LSP to the mean value */
- WebRtcSpl_MemSetW16(iLBCenc_inst->anaMem, 0, LPC_FILTERORDER);
- WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lsfold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER);
- WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lsfdeqold, WebRtcIlbcfix_kLsfMean, LPC_FILTERORDER);
- WebRtcSpl_MemSetW16(iLBCenc_inst->lpc_buffer, 0, LPC_LOOKBACK + BLOCKL_MAX);
-
- /* Set the filter state of the HP filter to 0 */
- WebRtcSpl_MemSetW16(iLBCenc_inst->hpimemx, 0, 2);
- WebRtcSpl_MemSetW16(iLBCenc_inst->hpimemy, 0, 4);
-
-#ifdef SPLIT_10MS
- /*Zeroing the past samples for 10msec Split*/
- WebRtcSpl_MemSetW16(iLBCenc_inst->past_samples,0,160);
- iLBCenc_inst->section = 0;
-#endif
-
- return (iLBCenc_inst->no_of_bytes);
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/init_encode.h b/trunk/src/modules/audio_coding/codecs/ilbc/init_encode.h
deleted file mode 100644
index f1d1858..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/init_encode.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_InitEncode.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INIT_ENCODE_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Initiation of encoder instance.
- *---------------------------------------------------------------*/
-
-WebRtc_Word16 WebRtcIlbcfix_InitEncode( /* (o) Number of bytes encoded */
- iLBC_Enc_Inst_t *iLBCenc_inst, /* (i/o) Encoder instance */
- WebRtc_Word16 mode /* (i) frame size mode */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/interface/ilbc.h b/trunk/src/modules/audio_coding/codecs/ilbc/interface/ilbc.h
deleted file mode 100644
index ca1d39e..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/interface/ilbc.h
+++ /dev/null
@@ -1,259 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * ilbc.h
- *
- * This header file contains all of the API's for iLBC.
- *
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_INTERFACE_ILBC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_INTERFACE_ILBC_H_
-
-/*
- * Define the fixpoint numeric formats
- */
-
-#include "typedefs.h"
-
-/*
- * Solution to support multiple instances
- * Customer has to cast instance to proper type
- */
-
-typedef struct iLBC_encinst_t_ iLBC_encinst_t;
-
-typedef struct iLBC_decinst_t_ iLBC_decinst_t;
-
-/*
- * Comfort noise constants
- */
-
-#define ILBC_SPEECH 1
-#define ILBC_CNG 2
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
- /****************************************************************************
- * WebRtcIlbcfix_XxxAssign(...)
- *
- * These functions assigns the encoder/decoder instance to the specified
- * memory location
- *
- * Input:
- * - XXX_xxxinst : Pointer to created instance that should be
- * assigned
- * - ILBCXXX_inst_Addr : Pointer to the desired memory space
- * - size : The size that this structure occupies (in Word16)
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIlbcfix_EncoderAssign(iLBC_encinst_t **iLBC_encinst,
- WebRtc_Word16 *ILBCENC_inst_Addr,
- WebRtc_Word16 *size);
- WebRtc_Word16 WebRtcIlbcfix_DecoderAssign(iLBC_decinst_t **iLBC_decinst,
- WebRtc_Word16 *ILBCDEC_inst_Addr,
- WebRtc_Word16 *size);
-
-
- /****************************************************************************
- * WebRtcIlbcfix_XxxAssign(...)
- *
- * These functions create a instance to the specified structure
- *
- * Input:
- * - XXX_inst : Pointer to created instance that should be created
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIlbcfix_EncoderCreate(iLBC_encinst_t **iLBC_encinst);
- WebRtc_Word16 WebRtcIlbcfix_DecoderCreate(iLBC_decinst_t **iLBC_decinst);
-
- /****************************************************************************
- * WebRtcIlbcfix_XxxFree(...)
- *
- * These functions frees the dynamic memory of a specified instance
- *
- * Input:
- * - XXX_inst : Pointer to created instance that should be freed
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIlbcfix_EncoderFree(iLBC_encinst_t *iLBC_encinst);
- WebRtc_Word16 WebRtcIlbcfix_DecoderFree(iLBC_decinst_t *iLBC_decinst);
-
-
- /****************************************************************************
- * WebRtcIlbcfix_EncoderInit(...)
- *
- * This function initializes a iLBC instance
- *
- * Input:
- * - iLBCenc_inst : iLBC instance, i.e. the user that should receive
- * be initialized
- * - frameLen : The frame length of the codec 20/30 (ms)
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIlbcfix_EncoderInit(iLBC_encinst_t *iLBCenc_inst,
- WebRtc_Word16 frameLen);
-
- /****************************************************************************
- * WebRtcIlbcfix_Encode(...)
- *
- * This function encodes one iLBC frame. Input speech length has be a
- * multiple of the frame length.
- *
- * Input:
- * - iLBCenc_inst : iLBC instance, i.e. the user that should encode
- * a package
- * - speechIn : Input speech vector
- * - len : Samples in speechIn (160, 240, 320 or 480)
- *
- * Output:
- * - encoded : The encoded data vector
- *
- * Return value : >0 - Length (in bytes) of coded data
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIlbcfix_Encode(iLBC_encinst_t *iLBCenc_inst,
- WebRtc_Word16 *speechIn,
- WebRtc_Word16 len,
- WebRtc_Word16 *encoded);
-
- /****************************************************************************
- * WebRtcIlbcfix_DecoderInit(...)
- *
- * This function initializes a iLBC instance with either 20 or 30 ms frames
- * Alternatively the WebRtcIlbcfix_DecoderInit_XXms can be used. Then it's
- * not needed to specify the frame length with a variable.
- *
- * Input:
- * - iLBC_decinst_t : iLBC instance, i.e. the user that should receive
- * be initialized
- * - frameLen : The frame length of the codec 20/30 (ms)
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIlbcfix_DecoderInit(iLBC_decinst_t *iLBCdec_inst,
- WebRtc_Word16 frameLen);
- WebRtc_Word16 WebRtcIlbcfix_DecoderInit20Ms(iLBC_decinst_t *iLBCdec_inst);
- WebRtc_Word16 WebRtcIlbcfix_Decoderinit30Ms(iLBC_decinst_t *iLBCdec_inst);
-
- /****************************************************************************
- * WebRtcIlbcfix_Decode(...)
- *
- * This function decodes a packet with iLBC frame(s). Output speech length
- * will be a multiple of 160 or 240 samples ((160 or 240)*frames/packet).
- *
- * Input:
- * - iLBCdec_inst : iLBC instance, i.e. the user that should decode
- * a packet
- * - encoded : Encoded iLBC frame(s)
- * - len : Bytes in encoded vector
- *
- * Output:
- * - decoded : The decoded vector
- * - speechType : 1 normal, 2 CNG
- *
- * Return value : >0 - Samples in decoded vector
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIlbcfix_Decode(iLBC_decinst_t *iLBCdec_inst,
- WebRtc_Word16* encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType);
- WebRtc_Word16 WebRtcIlbcfix_Decode20Ms(iLBC_decinst_t *iLBCdec_inst,
- WebRtc_Word16 *encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType);
- WebRtc_Word16 WebRtcIlbcfix_Decode30Ms(iLBC_decinst_t *iLBCdec_inst,
- WebRtc_Word16 *encoded,
- WebRtc_Word16 len,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 *speechType);
-
- /****************************************************************************
- * WebRtcIlbcfix_DecodePlc(...)
- *
- * This function conducts PLC for iLBC frame(s). Output speech length
- * will be a multiple of 160 or 240 samples.
- *
- * Input:
- * - iLBCdec_inst : iLBC instance, i.e. the user that should perform
- * a PLC
- * - noOfLostFrames : Number of PLC frames to produce
- *
- * Output:
- * - decoded : The "decoded" vector
- *
- * Return value : >0 - Samples in decoded PLC vector
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIlbcfix_DecodePlc(iLBC_decinst_t *iLBCdec_inst,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 noOfLostFrames);
-
- /****************************************************************************
- * WebRtcIlbcfix_NetEqPlc(...)
- *
- * This function updates the decoder when a packet loss has occured, but it
- * does not produce any PLC data. Function can be used if another PLC method
- * is used (i.e NetEq).
- *
- * Input:
- * - iLBCdec_inst : iLBC instance that should be updated
- * - noOfLostFrames : Number of lost frames
- *
- * Output:
- * - decoded : The "decoded" vector (nothing in this case)
- *
- * Return value : >0 - Samples in decoded PLC vector
- * -1 - Error
- */
-
- WebRtc_Word16 WebRtcIlbcfix_NetEqPlc(iLBC_decinst_t *iLBCdec_inst,
- WebRtc_Word16 *decoded,
- WebRtc_Word16 noOfLostFrames);
-
- /****************************************************************************
- * WebRtcIlbcfix_version(...)
- *
- * This function returns the version number of iLBC
- *
- * Output:
- * - version : Version number of iLBC (maximum 20 char)
- */
-
- void WebRtcIlbcfix_version(char *version);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/interpolate.c b/trunk/src/modules/audio_coding/codecs/ilbc/interpolate.c
deleted file mode 100644
index 11cb33c..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/interpolate.c
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Interpolate.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * interpolation between vectors
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Interpolate(
- WebRtc_Word16 *out, /* (o) output vector */
- WebRtc_Word16 *in1, /* (i) first input vector */
- WebRtc_Word16 *in2, /* (i) second input vector */
- WebRtc_Word16 coef, /* (i) weight coefficient in Q14 */
- WebRtc_Word16 length) /* (i) number of sample is vectors */
-{
- int i;
- WebRtc_Word16 invcoef;
-
- /*
- Performs the operation out[i] = in[i]*coef + (1-coef)*in2[i] (with rounding)
- */
-
- invcoef = 16384 - coef; /* 16384 = 1.0 (Q14)*/
- for (i = 0; i < length; i++) {
- out[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
- (WEBRTC_SPL_MUL_16_16(coef, in1[i]) + WEBRTC_SPL_MUL_16_16(invcoef, in2[i]))+8192,
- 14);
- }
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/interpolate.h b/trunk/src/modules/audio_coding/codecs/ilbc/interpolate.h
deleted file mode 100644
index a12021c..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/interpolate.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Interpolate.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * interpolation between vectors
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Interpolate(
- WebRtc_Word16 *out, /* (o) output vector */
- WebRtc_Word16 *in1, /* (i) first input vector */
- WebRtc_Word16 *in2, /* (i) second input vector */
- WebRtc_Word16 coef, /* (i) weight coefficient in Q14 */
- WebRtc_Word16 length); /* (i) number of sample is vectors */
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/interpolate_samples.c b/trunk/src/modules/audio_coding/codecs/ilbc/interpolate_samples.c
deleted file mode 100644
index 31eb52e..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/interpolate_samples.c
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_InterpolateSamples.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-void WebRtcIlbcfix_InterpolateSamples(
- WebRtc_Word16 *interpSamples, /* (o) The interpolated samples */
- WebRtc_Word16 *CBmem, /* (i) The CB memory */
- WebRtc_Word16 lMem /* (i) Length of the CB memory */
- ) {
- WebRtc_Word16 *ppi, *ppo, i, j, temp1, temp2;
- WebRtc_Word16 *tmpPtr;
-
- /* Calculate the 20 vectors of interpolated samples (4 samples each)
- that are used in the codebooks for lag 20 to 39 */
- tmpPtr = interpSamples;
- for (j=0; j<20; j++) {
- temp1 = 0;
- temp2 = 3;
- ppo = CBmem+lMem-4;
- ppi = CBmem+lMem-j-24;
- for (i=0; i<4; i++) {
-
- *tmpPtr++ = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kAlpha[temp2],*ppo, 15) +
- (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kAlpha[temp1], *ppi, 15);
-
- ppo++;
- ppi++;
- temp1++;
- temp2--;
- }
- }
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/interpolate_samples.h b/trunk/src/modules/audio_coding/codecs/ilbc/interpolate_samples.h
deleted file mode 100644
index 5c98aaf..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/interpolate_samples.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_InterpolateSamples.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_INTERPOLATE_SAMPLES_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Construct the interpolated samples for the Augmented CB
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_InterpolateSamples(
- WebRtc_Word16 *interpSamples, /* (o) The interpolated samples */
- WebRtc_Word16 *CBmem, /* (i) The CB memory */
- WebRtc_Word16 lMem /* (i) Length of the CB memory */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lpc_encode.c b/trunk/src/modules/audio_coding/codecs/ilbc/lpc_encode.c
deleted file mode 100644
index 73d67a0..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/lpc_encode.c
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_LpcEncode.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "simple_lpc_analysis.h"
-#include "simple_interpolate_lsf.h"
-#include "simple_lsf_quant.h"
-#include "lsf_check.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * lpc encoder
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_LpcEncode(
- WebRtc_Word16 *syntdenum, /* (i/o) synthesis filter coefficients
- before/after encoding */
- WebRtc_Word16 *weightdenum, /* (i/o) weighting denumerator coefficients
- before/after encoding */
- WebRtc_Word16 *lsf_index, /* (o) lsf quantization index */
- WebRtc_Word16 *data, /* (i) Speech to do LPC analysis on */
- iLBC_Enc_Inst_t *iLBCenc_inst
- /* (i/o) the encoder state structure */
- ) {
- /* Stack based */
- WebRtc_Word16 lsf[LPC_FILTERORDER * LPC_N_MAX];
- WebRtc_Word16 lsfdeq[LPC_FILTERORDER * LPC_N_MAX];
-
- /* Calculate LSF's from the input speech */
- WebRtcIlbcfix_SimpleLpcAnalysis(lsf, data, iLBCenc_inst);
-
- /* Quantize the LSF's */
- WebRtcIlbcfix_SimpleLsfQ(lsfdeq, lsf_index, lsf, iLBCenc_inst->lpc_n);
-
- /* Stableize the LSF's if needed */
- WebRtcIlbcfix_LsfCheck(lsfdeq, LPC_FILTERORDER, iLBCenc_inst->lpc_n);
-
- /* Calculate the synthesis and weighting filter coefficients from
- the optimal LSF and the dequantized LSF */
- WebRtcIlbcfix_SimpleInterpolateLsf(syntdenum, weightdenum,
- lsf, lsfdeq, iLBCenc_inst->lsfold,
- iLBCenc_inst->lsfdeqold, LPC_FILTERORDER, iLBCenc_inst);
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lpc_encode.h b/trunk/src/modules/audio_coding/codecs/ilbc/lpc_encode.h
deleted file mode 100644
index 36967a3..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/lpc_encode.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_LpcEncode.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LPC_ENCODE_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * lpc encoder
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_LpcEncode(
- WebRtc_Word16 *syntdenum, /* (i/o) synthesis filter coefficients
- before/after encoding */
- WebRtc_Word16 *weightdenum, /* (i/o) weighting denumerator coefficients
- before/after encoding */
- WebRtc_Word16 *lsf_index, /* (o) lsf quantization index */
- WebRtc_Word16 *data, /* (i) Speech to do LPC analysis on */
- iLBC_Enc_Inst_t *iLBCenc_inst
- /* (i/o) the encoder state structure */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_check.c b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_check.c
deleted file mode 100644
index 7097d74..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_check.c
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_LsfCheck.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * check for stability of lsf coefficients
- *---------------------------------------------------------------*/
-
-int WebRtcIlbcfix_LsfCheck(
- WebRtc_Word16 *lsf, /* LSF parameters */
- int dim, /* dimension of LSF */
- int NoAn) /* No of analysis per frame */
-{
- int k,n,m, Nit=2, change=0,pos;
- const WebRtc_Word16 eps=319; /* 0.039 in Q13 (50 Hz)*/
- const WebRtc_Word16 eps2=160; /* eps/2.0 in Q13;*/
- const WebRtc_Word16 maxlsf=25723; /* 3.14; (4000 Hz)*/
- const WebRtc_Word16 minlsf=82; /* 0.01; (0 Hz)*/
-
- /* LSF separation check*/
- for (n=0;n<Nit;n++) { /* Run through a 2 times */
- for (m=0;m<NoAn;m++) { /* Number of analyses per frame */
- for (k=0;k<(dim-1);k++) {
- pos=m*dim+k;
-
- /* Seperate coefficients with a safety margin of 50 Hz */
- if ((lsf[pos+1]-lsf[pos])<eps) {
-
- if (lsf[pos+1]<lsf[pos]) {
- lsf[pos+1]= lsf[pos]+eps2;
- lsf[pos]= lsf[pos+1]-eps2;
- } else {
- lsf[pos]-=eps2;
- lsf[pos+1]+=eps2;
- }
- change=1;
- }
-
- /* Limit minimum and maximum LSF */
- if (lsf[pos]<minlsf) {
- lsf[pos]=minlsf;
- change=1;
- }
-
- if (lsf[pos]>maxlsf) {
- lsf[pos]=maxlsf;
- change=1;
- }
- }
- }
- }
-
- return change;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_check.h b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_check.h
deleted file mode 100644
index 830bbed..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_check.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_LsfCheck.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_CHECK_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * check for stability of lsf coefficients
- *---------------------------------------------------------------*/
-
-int WebRtcIlbcfix_LsfCheck(
- WebRtc_Word16 *lsf, /* LSF parameters */
- int dim, /* dimension of LSF */
- int NoAn); /* No of analysis per frame */
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c
deleted file mode 100644
index 3bb23d0..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.c
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_LspInterpolate2PolyDec.c
-
-******************************************************************/
-
-#include "interpolate.h"
-#include "lsf_to_poly.h"
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * interpolation of lsf coefficients for the decoder
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_LspInterpolate2PolyDec(
- WebRtc_Word16 *a, /* (o) lpc coefficients Q12 */
- WebRtc_Word16 *lsf1, /* (i) first set of lsf coefficients Q13 */
- WebRtc_Word16 *lsf2, /* (i) second set of lsf coefficients Q13 */
- WebRtc_Word16 coef, /* (i) weighting coefficient to use between
- lsf1 and lsf2 Q14 */
- WebRtc_Word16 length /* (i) length of coefficient vectors */
- ){
- WebRtc_Word16 lsftmp[LPC_FILTERORDER];
-
- /* interpolate LSF */
- WebRtcIlbcfix_Interpolate(lsftmp, lsf1, lsf2, coef, length);
-
- /* Compute the filter coefficients from the LSF */
- WebRtcIlbcfix_Lsf2Poly(a, lsftmp);
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
deleted file mode 100644
index 23fe3a7..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_dec.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_LspInterpolate2PolyDec.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_DEC_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * interpolation of lsf coefficients for the decoder
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_LspInterpolate2PolyDec(
- WebRtc_Word16 *a, /* (o) lpc coefficients Q12 */
- WebRtc_Word16 *lsf1, /* (i) first set of lsf coefficients Q13 */
- WebRtc_Word16 *lsf2, /* (i) second set of lsf coefficients Q13 */
- WebRtc_Word16 coef, /* (i) weighting coefficient to use between
- lsf1 and lsf2 Q14 */
- WebRtc_Word16 length /* (i) length of coefficient vectors */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c
deleted file mode 100644
index 3b0a34d..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.c
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_LsfInterpolate2PloyEnc.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "interpolate.h"
-#include "lsf_to_poly.h"
-
-/*----------------------------------------------------------------*
- * lsf interpolator and conversion from lsf to a coefficients
- * (subrutine to SimpleInterpolateLSF)
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_LsfInterpolate2PloyEnc(
- WebRtc_Word16 *a, /* (o) lpc coefficients Q12 */
- WebRtc_Word16 *lsf1, /* (i) first set of lsf coefficients Q13 */
- WebRtc_Word16 *lsf2, /* (i) second set of lsf coefficients Q13 */
- WebRtc_Word16 coef, /* (i) weighting coefficient to use between
- lsf1 and lsf2 Q14 */
- WebRtc_Word16 length /* (i) length of coefficient vectors */
- ) {
- /* Stack based */
- WebRtc_Word16 lsftmp[LPC_FILTERORDER];
-
- /* interpolate LSF */
- WebRtcIlbcfix_Interpolate(lsftmp, lsf1, lsf2, coef, length);
-
- /* Compute the filter coefficients from the LSF */
- WebRtcIlbcfix_Lsf2Poly(a, lsftmp);
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
deleted file mode 100644
index 1bbbb80..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_interpolate_to_poly_enc.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_LsfInterpolate2PloyEnc.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_INTERPOLATE_TO_POLY_ENC_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * lsf interpolator and conversion from lsf to a coefficients
- * (subrutine to SimpleInterpolateLSF)
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_LsfInterpolate2PloyEnc(
- WebRtc_Word16 *a, /* (o) lpc coefficients Q12 */
- WebRtc_Word16 *lsf1, /* (i) first set of lsf coefficients Q13 */
- WebRtc_Word16 *lsf2, /* (i) second set of lsf coefficients Q13 */
- WebRtc_Word16 coef, /* (i) weighting coefficient to use between
- lsf1 and lsf2 Q14 */
- WebRtc_Word16 length /* (i) length of coefficient vectors */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c
deleted file mode 100644
index 84278a4..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_lsp.c
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Lsf2Lsp.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * conversion from lsf to lsp coefficients
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Lsf2Lsp(
- WebRtc_Word16 *lsf, /* (i) lsf in Q13 values between 0 and pi */
- WebRtc_Word16 *lsp, /* (o) lsp in Q15 values between -1 and 1 */
- WebRtc_Word16 m /* (i) number of coefficients */
- ) {
- WebRtc_Word16 i, k;
- WebRtc_Word16 diff; /* difference, which is used for the
- linear approximation (Q8) */
- WebRtc_Word16 freq; /* normalized frequency in Q15 (0..1) */
- WebRtc_Word32 tmpW32;
-
- for(i=0; i<m; i++)
- {
- freq = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(lsf[i], 20861, 15);
- /* 20861: 1.0/(2.0*PI) in Q17 */
- /*
- Upper 8 bits give the index k and
- Lower 8 bits give the difference, which needs
- to be approximated linearly
- */
- k = WEBRTC_SPL_RSHIFT_W16(freq, 8);
- diff = (freq&0x00ff);
-
- /* Guard against getting outside table */
-
- if (k>63) {
- k = 63;
- }
-
- /* Calculate linear approximation */
- tmpW32 = WEBRTC_SPL_MUL_16_16(WebRtcIlbcfix_kCosDerivative[k], diff);
- lsp[i] = WebRtcIlbcfix_kCos[k]+(WebRtc_Word16)(WEBRTC_SPL_RSHIFT_W32(tmpW32, 12));
- }
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
deleted file mode 100644
index db6549b..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_lsp.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Lsf2Lsp.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_LSP_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * conversion from lsf to lsp coefficients
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Lsf2Lsp(
- WebRtc_Word16 *lsf, /* (i) lsf in Q13 values between 0 and pi */
- WebRtc_Word16 *lsp, /* (o) lsp in Q15 values between -1 and 1 */
- WebRtc_Word16 m /* (i) number of coefficients */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_poly.c b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_poly.c
deleted file mode 100644
index f1c4a9e..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_poly.c
+++ /dev/null
@@ -1,86 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Lsf2Poly.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "lsf_to_lsp.h"
-#include "get_lsp_poly.h"
-#include "constants.h"
-
-void WebRtcIlbcfix_Lsf2Poly(
- WebRtc_Word16 *a, /* (o) predictor coefficients (order = 10) in Q12 */
- WebRtc_Word16 *lsf /* (i) line spectral frequencies in Q13 */
- ) {
- WebRtc_Word32 f[2][6]; /* f[0][] and f[1][] corresponds to
- F1(z) and F2(z) respectivly */
- WebRtc_Word32 *f1ptr, *f2ptr;
- WebRtc_Word16 *a1ptr, *a2ptr;
- WebRtc_Word32 tmpW32;
- WebRtc_Word16 lsp[10];
- int i;
-
- /* Convert lsf to lsp */
- WebRtcIlbcfix_Lsf2Lsp(lsf, lsp, LPC_FILTERORDER);
-
- /* Get F1(z) and F2(z) from the lsp */
- f1ptr=f[0];
- f2ptr=f[1];
- WebRtcIlbcfix_GetLspPoly(&lsp[0],f1ptr);
- WebRtcIlbcfix_GetLspPoly(&lsp[1],f2ptr);
-
- /* for i = 5 down to 1
- Compute f1[i] += f1[i-1];
- and f2[i] += f2[i-1];
- */
- f1ptr=&f[0][5];
- f2ptr=&f[1][5];
- for (i=5; i>0; i--)
- {
- (*f1ptr) += (*(f1ptr-1));
- (*f2ptr) -= (*(f2ptr-1));
- f1ptr--;
- f2ptr--;
- }
-
- /* Get the A(z) coefficients
- a[0] = 1.0
- for i = 1 to 5
- a[i] = (f1[i] + f2[i] + round)>>13;
- for i = 1 to 5
- a[11-i] = (f1[i] - f2[i] + round)>>13;
- */
- a[0]=4096;
- a1ptr=&a[1];
- a2ptr=&a[10];
- f1ptr=&f[0][1];
- f2ptr=&f[1][1];
- for (i=5; i>0; i--)
- {
- tmpW32 = (*f1ptr) + (*f2ptr);
- (*a1ptr) = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((tmpW32+4096),13);
-
- tmpW32 = (*f1ptr) - (*f2ptr);
- (*a2ptr) = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((tmpW32+4096),13);
-
- a1ptr++;
- a2ptr--;
- f1ptr++;
- f2ptr++;
- }
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_poly.h b/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
deleted file mode 100644
index a00693b..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/lsf_to_poly.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Lsf2Poly.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSF_TO_POLY_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Convert from LSF coefficients to A coefficients
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Lsf2Poly(
- WebRtc_Word16 *a, /* (o) predictor coefficients (order = 10) in Q12 */
- WebRtc_Word16 *lsf /* (i) line spectral frequencies in Q13 */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c b/trunk/src/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c
deleted file mode 100644
index 134afbb..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/lsp_to_lsf.c
+++ /dev/null
@@ -1,84 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Lsp2Lsf.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * conversion from LSP coefficients to LSF coefficients
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Lsp2Lsf(
- WebRtc_Word16 *lsp, /* (i) lsp vector -1...+1 in Q15 */
- WebRtc_Word16 *lsf, /* (o) Lsf vector 0...Pi in Q13
- (ordered, so that lsf[i]<lsf[i+1]) */
- WebRtc_Word16 m /* (i) Number of coefficients */
- )
-{
- WebRtc_Word16 i, k;
- WebRtc_Word16 diff; /* diff between table value and desired value (Q15) */
- WebRtc_Word16 freq; /* lsf/(2*pi) (Q16) */
- WebRtc_Word16 *lspPtr, *lsfPtr, *cosTblPtr;
- WebRtc_Word16 tmp;
-
- /* set the index to maximum index value in WebRtcIlbcfix_kCos */
- k = 63;
-
- /*
- Start with the highest LSP and then work the way down
- For each LSP the lsf is calculated by first order approximation
- of the acos(x) function
- */
- lspPtr = &lsp[9];
- lsfPtr = &lsf[9];
- cosTblPtr=(WebRtc_Word16*)&WebRtcIlbcfix_kCos[k];
- for(i=m-1; i>=0; i--)
- {
- /*
- locate value in the table, which is just above lsp[i],
- basically an approximation to acos(x)
- */
- while( (((WebRtc_Word32)(*cosTblPtr)-(*lspPtr)) < 0)&&(k>0) )
- {
- k-=1;
- cosTblPtr--;
- }
-
- /* Calculate diff, which is used in the linear approximation of acos(x) */
- diff = (*lspPtr)-(*cosTblPtr);
-
- /*
- The linear approximation of acos(lsp[i]) :
- acos(lsp[i])= k*512 + (WebRtcIlbcfix_kAcosDerivative[ind]*offset >> 11)
- */
-
- /* tmp (linear offset) in Q16 */
- tmp = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(WebRtcIlbcfix_kAcosDerivative[k],diff, 11);
-
- /* freq in Q16 */
- freq = (WebRtc_Word16)WEBRTC_SPL_LSHIFT_W16(k,9)+tmp;
-
- /* lsf = freq*2*pi */
- (*lsfPtr) = (WebRtc_Word16)(((WebRtc_Word32)freq*25736)>>15);
-
- lsfPtr--;
- lspPtr--;
- }
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h b/trunk/src/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
deleted file mode 100644
index 97ba7e4..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/lsp_to_lsf.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Lsp2Lsf.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_LSP_TO_LSF_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * conversion from LSP coefficients to LSF coefficients
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Lsp2Lsf(
- WebRtc_Word16 *lsp, /* (i) lsp vector -1...+1 in Q15 */
- WebRtc_Word16 *lsf, /* (o) Lsf vector 0...Pi in Q13
- (ordered, so that lsf[i]<lsf[i+1]) */
- WebRtc_Word16 m /* (i) Number of coefficients */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/my_corr.c b/trunk/src/modules/audio_coding/codecs/ilbc/my_corr.c
deleted file mode 100644
index 2162205..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/my_corr.c
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_MyCorr.c
-
-******************************************************************/
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * compute cross correlation between sequences
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_MyCorr(
- WebRtc_Word32 *corr, /* (o) correlation of seq1 and seq2 */
- WebRtc_Word16 *seq1, /* (i) first sequence */
- WebRtc_Word16 dim1, /* (i) dimension first seq1 */
- const WebRtc_Word16 *seq2, /* (i) second sequence */
- WebRtc_Word16 dim2 /* (i) dimension seq2 */
- ){
- WebRtc_Word16 max, scale, loops;
-
- /* Calculate correlation between the two sequences. Scale the
- result of the multiplcication to maximum 26 bits in order
- to avoid overflow */
- max=WebRtcSpl_MaxAbsValueW16(seq1, dim1);
- scale=WebRtcSpl_GetSizeInBits(max);
-
- scale = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16(2,scale)-26);
- if (scale<0) {
- scale=0;
- }
-
- loops=dim1-dim2+1;
-
- /* Calculate the cross correlations */
- WebRtcSpl_CrossCorrelation(corr, (WebRtc_Word16*)seq2, seq1, dim2, loops, scale, 1);
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/my_corr.h b/trunk/src/modules/audio_coding/codecs/ilbc/my_corr.h
deleted file mode 100644
index f588c53..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/my_corr.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_MyCorr.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_MY_CORR_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * compute cross correlation between sequences
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_MyCorr(
- WebRtc_Word32 *corr, /* (o) correlation of seq1 and seq2 */
- WebRtc_Word16 *seq1, /* (i) first sequence */
- WebRtc_Word16 dim1, /* (i) dimension first seq1 */
- const WebRtc_Word16 *seq2, /* (i) second sequence */
- WebRtc_Word16 dim2 /* (i) dimension seq2 */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/nearest_neighbor.c b/trunk/src/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
deleted file mode 100644
index ea9e1eb..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/nearest_neighbor.c
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_NearestNeighbor.c
-
-******************************************************************/
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Find index in array such that the array element with said
- * index is the element of said array closest to "value"
- * according to the squared-error criterion
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_NearestNeighbor(
- WebRtc_Word16 *index, /* (o) index of array element closest to value */
- WebRtc_Word16 *array, /* (i) data array (Q2) */
- WebRtc_Word16 value, /* (i) value (Q2) */
- WebRtc_Word16 arlength /* (i) dimension of data array (==8) */
- ){
- int i;
- WebRtc_Word16 diff;
- /* Stack based */
- WebRtc_Word32 crit[8];
-
- /* Calculate square distance */
- for(i=0;i<arlength;i++){
- diff=array[i]-value;
- crit[i]=WEBRTC_SPL_MUL_16_16(diff, diff);
- }
-
- /* Find the minimum square distance */
- *index=WebRtcSpl_MinIndexW32(crit, (WebRtc_Word16)arlength);
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/nearest_neighbor.h b/trunk/src/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
deleted file mode 100644
index 705e17a..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/nearest_neighbor.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_NearestNeighbor.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_NEAREST_NEIGHBOR_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Find index in array such that the array element with said
- * index is the element of said array closest to "value"
- * according to the squared-error criterion
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_NearestNeighbor(
- WebRtc_Word16 *index, /* (o) index of array element closest to value */
- WebRtc_Word16 *array, /* (i) data array (Q2) */
- WebRtc_Word16 value, /* (i) value (Q2) */
- WebRtc_Word16 arlength /* (i) dimension of data array (==8) */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/pack_bits.c b/trunk/src/modules/audio_coding/codecs/ilbc/pack_bits.c
deleted file mode 100644
index 3990fbe..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/pack_bits.c
+++ /dev/null
@@ -1,251 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_PackBits.c
-
-******************************************************************/
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * unpacking of bits from bitstream, i.e., vector of bytes
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_PackBits(
- WebRtc_UWord16 *bitstream, /* (o) The packetized bitstream */
- iLBC_bits *enc_bits, /* (i) Encoded bits */
- WebRtc_Word16 mode /* (i) Codec mode (20 or 30) */
- ){
- WebRtc_UWord16 *bitstreamPtr;
- int i, k;
- WebRtc_Word16 *tmpPtr;
-
- bitstreamPtr=bitstream;
-
- /* Class 1 bits of ULP */
- /* First WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)enc_bits->lsf[0])<<10; /* Bit 0..5 */
- (*bitstreamPtr) |= (enc_bits->lsf[1])<<3; /* Bit 6..12 */
- (*bitstreamPtr) |= (enc_bits->lsf[2]&0x70)>>4; /* Bit 13..15 */
- bitstreamPtr++;
- /* Second WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)enc_bits->lsf[2]&0xF)<<12; /* Bit 0..3 */
-
- if (mode==20) {
- (*bitstreamPtr) |= (enc_bits->startIdx)<<10; /* Bit 4..5 */
- (*bitstreamPtr) |= (enc_bits->state_first)<<9; /* Bit 6 */
- (*bitstreamPtr) |= (enc_bits->idxForMax)<<3; /* Bit 7..12 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[0])&0x70)>>4; /* Bit 13..15 */
- bitstreamPtr++;
- /* Third WebRtc_Word16 */
- (*bitstreamPtr) = ((enc_bits->cb_index[0])&0xE)<<12; /* Bit 0..2 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x18)<<8; /* Bit 3..4 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x8)<<7; /* Bit 5 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0xFE)<<2; /* Bit 6..12 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[3])&0x10)>>2; /* Bit 13 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x8)>>2; /* Bit 14 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x10)>>4; /* Bit 15 */
- } else { /* mode==30 */
- (*bitstreamPtr) |= (enc_bits->lsf[3])<<6; /* Bit 4..9 */
- (*bitstreamPtr) |= (enc_bits->lsf[4]&0x7E)>>1; /* Bit 10..15 */
- bitstreamPtr++;
- /* Third WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)enc_bits->lsf[4]&0x1)<<15; /* Bit 0 */
- (*bitstreamPtr) |= (enc_bits->lsf[5])<<8; /* Bit 1..7 */
- (*bitstreamPtr) |= (enc_bits->startIdx)<<5; /* Bit 8..10 */
- (*bitstreamPtr) |= (enc_bits->state_first)<<4; /* Bit 11 */
- (*bitstreamPtr) |= ((enc_bits->idxForMax)&0x3C)>>2; /* Bit 12..15 */
- bitstreamPtr++;
- /* 4:th WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)enc_bits->idxForMax&0x3)<<14; /* Bit 0..1 */
- (*bitstreamPtr) |= (enc_bits->cb_index[0]&0x78)<<7; /* Bit 2..5 */
- (*bitstreamPtr) |= (enc_bits->gain_index[0]&0x10)<<5; /* Bit 6 */
- (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x8)<<5; /* Bit 7 */
- (*bitstreamPtr) |= (enc_bits->cb_index[3]&0xFC); /* Bit 8..13 */
- (*bitstreamPtr) |= (enc_bits->gain_index[3]&0x10)>>3; /* Bit 14 */
- (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x8)>>3; /* Bit 15 */
- }
- /* Class 2 bits of ULP */
- /* 4:th to 6:th WebRtc_Word16 for 20 ms case
- 5:th to 7:th WebRtc_Word16 for 30 ms case */
- bitstreamPtr++;
- tmpPtr=enc_bits->idxVec;
- for (k=0; k<3; k++) {
- (*bitstreamPtr) = 0;
- for (i=15; i>=0; i--) {
- (*bitstreamPtr) |= ((WebRtc_UWord16)((*tmpPtr)&0x4)>>2)<<i;
- /* Bit 15-i */
- tmpPtr++;
- }
- bitstreamPtr++;
- }
-
- if (mode==20) {
- /* 7:th WebRtc_Word16 */
- (*bitstreamPtr) = 0;
- for (i=15; i>6; i--) {
- (*bitstreamPtr) |= ((WebRtc_UWord16)((*tmpPtr)&0x4)>>2)<<i;
- /* Bit 15-i */
- tmpPtr++;
- }
- (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x4)<<4; /* Bit 9 */
- (*bitstreamPtr) |= (enc_bits->gain_index[3]&0xC)<<2; /* Bit 10..11 */
- (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x4)<<1; /* Bit 12 */
- (*bitstreamPtr) |= (enc_bits->gain_index[6]&0x8)>>1; /* Bit 13 */
- (*bitstreamPtr) |= (enc_bits->gain_index[7]&0xC)>>2; /* Bit 14..15 */
-
- } else { /* mode==30 */
- /* 8:th WebRtc_Word16 */
- (*bitstreamPtr) = 0;
- for (i=15; i>5; i--) {
- (*bitstreamPtr) |= ((WebRtc_UWord16)((*tmpPtr)&0x4)>>2)<<i;
- /* Bit 15-i */
- tmpPtr++;
- }
- (*bitstreamPtr) |= (enc_bits->cb_index[0]&0x6)<<3; /* Bit 10..11 */
- (*bitstreamPtr) |= (enc_bits->gain_index[0]&0x8); /* Bit 12 */
- (*bitstreamPtr) |= (enc_bits->gain_index[1]&0x4); /* Bit 13 */
- (*bitstreamPtr) |= (enc_bits->cb_index[3]&0x2); /* Bit 14 */
- (*bitstreamPtr) |= (enc_bits->cb_index[6]&0x80)>>7; /* Bit 15 */
- bitstreamPtr++;
- /* 9:th WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)enc_bits->cb_index[6]&0x7E)<<9;/* Bit 0..5 */
- (*bitstreamPtr) |= (enc_bits->cb_index[9]&0xFE)<<2; /* Bit 6..12 */
- (*bitstreamPtr) |= (enc_bits->cb_index[12]&0xE0)>>5; /* Bit 13..15 */
- bitstreamPtr++;
- /* 10:th WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)enc_bits->cb_index[12]&0x1E)<<11;/* Bit 0..3 */
- (*bitstreamPtr) |= (enc_bits->gain_index[3]&0xC)<<8; /* Bit 4..5 */
- (*bitstreamPtr) |= (enc_bits->gain_index[4]&0x6)<<7; /* Bit 6..7 */
- (*bitstreamPtr) |= (enc_bits->gain_index[6]&0x18)<<3; /* Bit 8..9 */
- (*bitstreamPtr) |= (enc_bits->gain_index[7]&0xC)<<2; /* Bit 10..11 */
- (*bitstreamPtr) |= (enc_bits->gain_index[9]&0x10)>>1; /* Bit 12 */
- (*bitstreamPtr) |= (enc_bits->gain_index[10]&0x8)>>1; /* Bit 13 */
- (*bitstreamPtr) |= (enc_bits->gain_index[12]&0x10)>>3; /* Bit 14 */
- (*bitstreamPtr) |= (enc_bits->gain_index[13]&0x8)>>3; /* Bit 15 */
- }
- bitstreamPtr++;
- /* Class 3 bits of ULP */
- /* 8:th to 14:th WebRtc_Word16 for 20 ms case
- 11:th to 17:th WebRtc_Word16 for 30 ms case */
- tmpPtr=enc_bits->idxVec;
- for (k=0; k<7; k++) {
- (*bitstreamPtr) = 0;
- for (i=14; i>=0; i-=2) {
- (*bitstreamPtr) |= ((WebRtc_UWord16)((*tmpPtr)&0x3))<<i; /* Bit 15-i..14-i*/
- tmpPtr++;
- }
- bitstreamPtr++;
- }
-
- if (mode==20) {
- /* 15:th WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)((enc_bits->idxVec[56])&0x3))<<14;/* Bit 0..1 */
- (*bitstreamPtr) |= (((enc_bits->cb_index[0])&1))<<13; /* Bit 2 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[1]))<<6; /* Bit 3..9 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[2])&0x7E)>>1; /* Bit 10..15 */
- bitstreamPtr++;
- /* 16:th WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)((enc_bits->cb_index[2])&0x1))<<15;
- /* Bit 0 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x7)<<12; /* Bit 1..3 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x3)<<10; /* Bit 4..5 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[2]))<<7; /* Bit 6..8 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0x1)<<6; /* Bit 9 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[4])&0x7E)>>1; /* Bit 10..15 */
- bitstreamPtr++;
- /* 17:th WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)((enc_bits->cb_index[4])&0x1))<<15;
- /* Bit 0 */
- (*bitstreamPtr) |= (enc_bits->cb_index[5])<<8; /* Bit 1..7 */
- (*bitstreamPtr) |= (enc_bits->cb_index[6]); /* Bit 8..15 */
- bitstreamPtr++;
- /* 18:th WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)(enc_bits->cb_index[7]))<<8; /* Bit 0..7 */
- (*bitstreamPtr) |= (enc_bits->cb_index[8]); /* Bit 8..15 */
- bitstreamPtr++;
- /* 19:th WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)((enc_bits->gain_index[3])&0x3))<<14;
- /* Bit 0..1 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x3)<<12; /* Bit 2..3 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[5]))<<9; /* Bit 4..6 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x7)<<6; /* Bit 7..9 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[7])&0x3)<<4; /* Bit 10..11 */
- (*bitstreamPtr) |= (enc_bits->gain_index[8])<<1; /* Bit 12..14 */
- } else { /* mode==30 */
- /* 18:th WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)((enc_bits->idxVec[56])&0x3))<<14;/* Bit 0..1 */
- (*bitstreamPtr) |= (((enc_bits->idxVec[57])&0x3))<<12; /* Bit 2..3 */
- (*bitstreamPtr) |= (((enc_bits->cb_index[0])&1))<<11; /* Bit 4 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[1]))<<4; /* Bit 5..11 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[2])&0x78)>>3; /* Bit 12..15 */
- bitstreamPtr++;
- /* 19:th WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)(enc_bits->cb_index[2])&0x7)<<13;
- /* Bit 0..2 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[0])&0x7)<<10; /* Bit 3..5 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[1])&0x3)<<8; /* Bit 6..7 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[2])&0x7)<<5; /* Bit 8..10 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[3])&0x1)<<4; /* Bit 11 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[4])&0x78)>>3; /* Bit 12..15 */
- bitstreamPtr++;
- /* 20:th WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)(enc_bits->cb_index[4])&0x7)<<13;
- /* Bit 0..2 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[5]))<<6; /* Bit 3..9 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[6])&0x1)<<5; /* Bit 10 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[7])&0xF8)>>3; /* Bit 11..15 */
- bitstreamPtr++;
- /* 21:st WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)(enc_bits->cb_index[7])&0x7)<<13;
- /* Bit 0..2 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[8]))<<5; /* Bit 3..10 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[9])&0x1)<<4; /* Bit 11 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[10])&0xF0)>>4; /* Bit 12..15 */
- bitstreamPtr++;
- /* 22:nd WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)(enc_bits->cb_index[10])&0xF)<<12;
- /* Bit 0..3 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[11]))<<4; /* Bit 4..11 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[12])&0x1)<<3; /* Bit 12 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[13])&0xE0)>>5; /* Bit 13..15 */
- bitstreamPtr++;
- /* 23:rd WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)(enc_bits->cb_index[13])&0x1F)<<11;
- /* Bit 0..4 */
- (*bitstreamPtr) |= ((enc_bits->cb_index[14]))<<3; /* Bit 5..12 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[3])&0x3)<<1; /* Bit 13..14 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[4])&0x1); /* Bit 15 */
- bitstreamPtr++;
- /* 24:rd WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)(enc_bits->gain_index[5]))<<13;
- /* Bit 0..2 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[6])&0x7)<<10; /* Bit 3..5 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[7])&0x3)<<8; /* Bit 6..7 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[8]))<<5; /* Bit 8..10 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[9])&0xF)<<1; /* Bit 11..14 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[10])&0x4)>>2; /* Bit 15 */
- bitstreamPtr++;
- /* 25:rd WebRtc_Word16 */
- (*bitstreamPtr) = ((WebRtc_UWord16)(enc_bits->gain_index[10])&0x3)<<14;
- /* Bit 0..1 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[11]))<<11; /* Bit 2..4 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[12])&0xF)<<7; /* Bit 5..8 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[13])&0x7)<<4; /* Bit 9..11 */
- (*bitstreamPtr) |= ((enc_bits->gain_index[14]))<<1; /* Bit 12..14 */
- }
- /* Last bit is automatically zero */
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/pack_bits.h b/trunk/src/modules/audio_coding/codecs/ilbc/pack_bits.h
deleted file mode 100644
index ed3f224..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/pack_bits.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_PackBits.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_PACK_BITS_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * unpacking of bits from bitstream, i.e., vector of bytes
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_PackBits(
- WebRtc_UWord16 *bitstream, /* (o) The packetized bitstream */
- iLBC_bits *enc_bits, /* (i) Encoded bits */
- WebRtc_Word16 mode /* (i) Codec mode (20 or 30) */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsf.c b/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsf.c
deleted file mode 100644
index fe91851..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsf.c
+++ /dev/null
@@ -1,31 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Poly2Lsf.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-#include "poly_to_lsp.h"
-#include "lsp_to_lsf.h"
-
-void WebRtcIlbcfix_Poly2Lsf(
- WebRtc_Word16 *lsf, /* (o) lsf coefficients (Q13) */
- WebRtc_Word16 *a /* (i) A coefficients (Q12) */
- ) {
- WebRtc_Word16 lsp[10];
- WebRtcIlbcfix_Poly2Lsp(a, lsp, (WebRtc_Word16*)WebRtcIlbcfix_kLspMean);
- WebRtcIlbcfix_Lsp2Lsf(lsp, lsf, 10);
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsf.h b/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
deleted file mode 100644
index 0ea595e..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsf.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Poly2Lsf.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSF_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * conversion from lpc coefficients to lsf coefficients
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Poly2Lsf(
- WebRtc_Word16 *lsf, /* (o) lsf coefficients (Q13) */
- WebRtc_Word16 *a /* (i) A coefficients (Q12) */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsp.c b/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsp.c
deleted file mode 100644
index 29b4213..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsp.c
+++ /dev/null
@@ -1,156 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Poly2Lsp.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-#include "chebyshev.h"
-
-/*----------------------------------------------------------------*
- * conversion from lpc coefficients to lsp coefficients
- * function is only for 10:th order LPC
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Poly2Lsp(
- WebRtc_Word16 *a, /* (o) A coefficients in Q12 */
- WebRtc_Word16 *lsp, /* (i) LSP coefficients in Q15 */
- WebRtc_Word16 *old_lsp /* (i) old LSP coefficients that are used if the new
- coefficients turn out to be unstable */
- ) {
- WebRtc_Word16 f[2][6]; /* f[0][] represents f1 and f[1][] represents f2 */
- WebRtc_Word16 *a_i_ptr, *a_10mi_ptr;
- WebRtc_Word16 *f1ptr, *f2ptr;
- WebRtc_Word32 tmpW32;
- WebRtc_Word16 x, y, xlow, ylow, xmid, ymid, xhigh, yhigh, xint;
- WebRtc_Word16 shifts, sign;
- int i, j;
- int foundFreqs;
- int fi_select;
-
- /*
- Calculate the two polynomials f1(z) and f2(z)
- (the sum and the diff polynomial)
- f1[0] = f2[0] = 1.0;
- f1[i+1] = a[i+1] + a[10-i] - f1[i];
- f2[i+1] = a[i+1] - a[10-i] - f1[i];
- */
-
- a_i_ptr = a + 1;
- a_10mi_ptr = a + 10;
- f1ptr = f[0];
- f2ptr = f[1];
- (*f1ptr) = 1024; /* 1.0 in Q10 */
- (*f2ptr) = 1024; /* 1.0 in Q10 */
- for (i = 0; i < 5; i++) {
- (*(f1ptr+1)) = (WebRtc_Word16)(WEBRTC_SPL_RSHIFT_W32(((WebRtc_Word32)(*a_i_ptr)+(*a_10mi_ptr)), 2) - (*f1ptr));
- (*(f2ptr+1)) = (WebRtc_Word16)(WEBRTC_SPL_RSHIFT_W32(((WebRtc_Word32)(*a_i_ptr)-(*a_10mi_ptr)), 2) + (*f2ptr));
- a_i_ptr++;
- a_10mi_ptr--;
- f1ptr++;
- f2ptr++;
- }
-
- /*
- find the LSPs using the Chebychev pol. evaluation
- */
-
- fi_select = 0; /* selector between f1 and f2, start with f1 */
-
- foundFreqs = 0;
-
- xlow = WebRtcIlbcfix_kCosGrid[0];
- ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]);
-
- /*
- Iterate until all the 10 LSP's have been found or
- all the grid points have been tried. If the 10 LSP's can
- not be found, set the LSP vector to previous LSP
- */
-
- for (j = 1; j < COS_GRID_POINTS && foundFreqs < 10; j++) {
- xhigh = xlow;
- yhigh = ylow;
- xlow = WebRtcIlbcfix_kCosGrid[j];
- ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]);
-
- if (WEBRTC_SPL_MUL_16_16(ylow, yhigh) <= 0) {
- /* Run 4 times to reduce the interval */
- for (i = 0; i < 4; i++) {
- /* xmid =(xlow + xhigh)/2 */
- xmid = WEBRTC_SPL_RSHIFT_W16(xlow, 1) + WEBRTC_SPL_RSHIFT_W16(xhigh, 1);
- ymid = WebRtcIlbcfix_Chebyshev(xmid, f[fi_select]);
-
- if (WEBRTC_SPL_MUL_16_16(ylow, ymid) <= 0) {
- yhigh = ymid;
- xhigh = xmid;
- } else {
- ylow = ymid;
- xlow = xmid;
- }
- }
-
- /*
- Calculater xint by linear interpolation:
- xint = xlow - ylow*(xhigh-xlow)/(yhigh-ylow);
- */
-
- x = xhigh - xlow;
- y = yhigh - ylow;
-
- if (y == 0) {
- xint = xlow;
- } else {
- sign = y;
- y = WEBRTC_SPL_ABS_W16(y);
- shifts = (WebRtc_Word16)WebRtcSpl_NormW32(y)-16;
- y = WEBRTC_SPL_LSHIFT_W16(y, shifts);
- y = (WebRtc_Word16)WebRtcSpl_DivW32W16(536838144, y); /* 1/(yhigh-ylow) */
-
- tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(x, y, (19-shifts));
-
- /* y=(xhigh-xlow)/(yhigh-ylow) */
- y = (WebRtc_Word16)(tmpW32&0xFFFF);
-
- if (sign < 0) {
- y = -y;
- }
- /* tmpW32 = ylow*(xhigh-xlow)/(yhigh-ylow) */
- tmpW32 = WEBRTC_SPL_MUL_16_16_RSFT(ylow, y, 10);
- xint = xlow-(WebRtc_Word16)(tmpW32&0xFFFF);
- }
-
- /* Store the calculated lsp */
- lsp[foundFreqs] = (WebRtc_Word16)xint;
- foundFreqs++;
-
- /* if needed, set xlow and ylow for next recursion */
- if (foundFreqs<10) {
- xlow = xint;
- /* Swap between f1 and f2 (f[0][] and f[1][]) */
- fi_select = ((fi_select+1)&0x1);
-
- ylow = WebRtcIlbcfix_Chebyshev(xlow, f[fi_select]);
- }
- }
- }
-
- /* Check if M roots found, if not then use the old LSP */
- if (foundFreqs < 10) {
- WEBRTC_SPL_MEMCPY_W16(lsp, old_lsp, 10);
- }
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsp.h b/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
deleted file mode 100644
index 7eebb25..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/poly_to_lsp.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Poly2Lsp.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_POLY_TO_LSP_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * conversion from lpc coefficients to lsp coefficients
- * function is only for 10:th order LPC
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Poly2Lsp(
- WebRtc_Word16 *a, /* (o) A coefficients in Q12 */
- WebRtc_Word16 *lsp, /* (i) LSP coefficients in Q15 */
- WebRtc_Word16 *old_lsp /* (i) old LSP coefficients that are used if the new
- coefficients turn out to be unstable */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/refiner.c b/trunk/src/modules/audio_coding/codecs/ilbc/refiner.c
deleted file mode 100644
index 9210092..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/refiner.c
+++ /dev/null
@@ -1,150 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Refiner.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-#include "enh_upsample.h"
-#include "my_corr.h"
-
-/*----------------------------------------------------------------*
- * find segment starting near idata+estSegPos that has highest
- * correlation with idata+centerStartPos through
- * idata+centerStartPos+ENH_BLOCKL-1 segment is found at a
- * resolution of ENH_UPSO times the original of the original
- * sampling rate
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Refiner(
- WebRtc_Word16 *updStartPos, /* (o) updated start point (Q-2) */
- WebRtc_Word16 *idata, /* (i) original data buffer */
- WebRtc_Word16 idatal, /* (i) dimension of idata */
- WebRtc_Word16 centerStartPos, /* (i) beginning center segment */
- WebRtc_Word16 estSegPos, /* (i) estimated beginning other segment (Q-2) */
- WebRtc_Word16 *surround, /* (i/o) The contribution from this sequence
- summed with earlier contributions */
- WebRtc_Word16 gain /* (i) Gain to use for this sequence */
- ){
- WebRtc_Word16 estSegPosRounded,searchSegStartPos,searchSegEndPos,corrdim;
- WebRtc_Word16 tloc,tloc2,i,st,en,fraction;
-
- WebRtc_Word32 maxtemp, scalefact;
- WebRtc_Word16 *filtStatePtr, *polyPtr;
- /* Stack based */
- WebRtc_Word16 filt[7];
- WebRtc_Word32 corrVecUps[ENH_CORRDIM*ENH_UPS0];
- WebRtc_Word32 corrVecTemp[ENH_CORRDIM];
- WebRtc_Word16 vect[ENH_VECTL];
- WebRtc_Word16 corrVec[ENH_CORRDIM];
-
- /* defining array bounds */
-
- estSegPosRounded=WEBRTC_SPL_RSHIFT_W16((estSegPos - 2),2);
-
- searchSegStartPos=estSegPosRounded-ENH_SLOP;
-
- if (searchSegStartPos<0) {
- searchSegStartPos=0;
- }
- searchSegEndPos=estSegPosRounded+ENH_SLOP;
-
- if(searchSegEndPos+ENH_BLOCKL >= idatal) {
- searchSegEndPos=idatal-ENH_BLOCKL-1;
- }
- corrdim=searchSegEndPos-searchSegStartPos+1;
-
- /* compute upsampled correlation and find
- location of max */
-
- WebRtcIlbcfix_MyCorr(corrVecTemp,idata+searchSegStartPos,
- (WebRtc_Word16)(corrdim+ENH_BLOCKL-1),idata+centerStartPos,ENH_BLOCKL);
-
- /* Calculate the rescaling factor for the correlation in order to
- put the correlation in a WebRtc_Word16 vector instead */
- maxtemp=WebRtcSpl_MaxAbsValueW32(corrVecTemp, (WebRtc_Word16)corrdim);
-
- scalefact=WebRtcSpl_GetSizeInBits(maxtemp)-15;
-
- if (scalefact>0) {
- for (i=0;i<corrdim;i++) {
- corrVec[i]=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(corrVecTemp[i], scalefact);
- }
- } else {
- for (i=0;i<corrdim;i++) {
- corrVec[i]=(WebRtc_Word16)corrVecTemp[i];
- }
- }
- /* In order to guarantee that all values are initialized */
- for (i=corrdim;i<ENH_CORRDIM;i++) {
- corrVec[i]=0;
- }
-
- /* Upsample the correlation */
- WebRtcIlbcfix_EnhUpsample(corrVecUps,corrVec);
-
- /* Find maximum */
- tloc=WebRtcSpl_MaxIndexW32(corrVecUps, (WebRtc_Word16) (ENH_UPS0*corrdim));
-
- /* make vector can be upsampled without ever running outside
- bounds */
- *updStartPos = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(searchSegStartPos,4) + tloc + 4;
-
- tloc2 = WEBRTC_SPL_RSHIFT_W16((tloc+3), 2);
-
- st=searchSegStartPos+tloc2-ENH_FL0;
-
- /* initialize the vector to be filtered, stuff with zeros
- when data is outside idata buffer */
- if(st<0){
- WebRtcSpl_MemSetW16(vect, 0, (WebRtc_Word16)(-st));
- WEBRTC_SPL_MEMCPY_W16(&vect[-st], idata, (ENH_VECTL+st));
- }
- else{
- en=st+ENH_VECTL;
-
- if(en>idatal){
- WEBRTC_SPL_MEMCPY_W16(vect, &idata[st],
- (ENH_VECTL-(en-idatal)));
- WebRtcSpl_MemSetW16(&vect[ENH_VECTL-(en-idatal)], 0,
- (WebRtc_Word16)(en-idatal));
- }
- else {
- WEBRTC_SPL_MEMCPY_W16(vect, &idata[st], ENH_VECTL);
- }
- }
- /* Calculate which of the 4 fractions to use */
- fraction=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16(tloc2,ENH_UPS0)-tloc;
-
- /* compute the segment (this is actually a convolution) */
-
- filtStatePtr = filt + 6;
- polyPtr = (WebRtc_Word16*)WebRtcIlbcfix_kEnhPolyPhaser[fraction];
- for (i=0;i<7;i++) {
- *filtStatePtr-- = *polyPtr++;
- }
-
- WebRtcSpl_FilterMAFastQ12(
- &vect[6], vect, filt,
- ENH_FLO_MULT2_PLUS1, ENH_BLOCKL);
-
- /* Add the contribution from this vector (scaled with gain) to the total surround vector */
- WebRtcSpl_AddAffineVectorToVector(
- surround, vect, gain,
- (WebRtc_Word32)32768, 16, ENH_BLOCKL);
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/refiner.h b/trunk/src/modules/audio_coding/codecs/ilbc/refiner.h
deleted file mode 100644
index 559555c..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/refiner.h
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Refiner.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_REFINER_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * find segment starting near idata+estSegPos that has highest
- * correlation with idata+centerStartPos through
- * idata+centerStartPos+ENH_BLOCKL-1 segment is found at a
- * resolution of ENH_UPSO times the original of the original
- * sampling rate
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Refiner(
- WebRtc_Word16 *updStartPos, /* (o) updated start point (Q-2) */
- WebRtc_Word16 *idata, /* (i) original data buffer */
- WebRtc_Word16 idatal, /* (i) dimension of idata */
- WebRtc_Word16 centerStartPos, /* (i) beginning center segment */
- WebRtc_Word16 estSegPos, /* (i) estimated beginning other segment (Q-2) */
- WebRtc_Word16 *surround, /* (i/o) The contribution from this sequence
- summed with earlier contributions */
- WebRtc_Word16 gain /* (i) Gain to use for this sequence */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c b/trunk/src/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
deleted file mode 100644
index ee5e643..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.c
+++ /dev/null
@@ -1,130 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_SimpleInterpolateLsf.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "lsf_interpolate_to_poly_enc.h"
-#include "bw_expand.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * lsf interpolator (subrutine to LPCencode)
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_SimpleInterpolateLsf(
- WebRtc_Word16 *syntdenum, /* (o) the synthesis filter denominator
- resulting from the quantized
- interpolated lsf Q12 */
- WebRtc_Word16 *weightdenum, /* (o) the weighting filter denominator
- resulting from the unquantized
- interpolated lsf Q12 */
- WebRtc_Word16 *lsf, /* (i) the unquantized lsf coefficients Q13 */
- WebRtc_Word16 *lsfdeq, /* (i) the dequantized lsf coefficients Q13 */
- WebRtc_Word16 *lsfold, /* (i) the unquantized lsf coefficients of
- the previous signal frame Q13 */
- WebRtc_Word16 *lsfdeqold, /* (i) the dequantized lsf coefficients of the
- previous signal frame Q13 */
- WebRtc_Word16 length, /* (i) should equate FILTERORDER */
- iLBC_Enc_Inst_t *iLBCenc_inst
- /* (i/o) the encoder state structure */
- ) {
- int i, pos, lp_length;
-
- WebRtc_Word16 *lsf2, *lsfdeq2;
- /* Stack based */
- WebRtc_Word16 lp[LPC_FILTERORDER + 1];
-
- lsf2 = lsf + length;
- lsfdeq2 = lsfdeq + length;
- lp_length = length + 1;
-
- if (iLBCenc_inst->mode==30) {
- /* subframe 1: Interpolation between old and first set of
- lsf coefficients */
-
- /* Calculate Analysis/Syntehsis filter from quantized LSF */
- WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeqold, lsfdeq,
- WebRtcIlbcfix_kLsfWeight30ms[0],
- length);
- WEBRTC_SPL_MEMCPY_W16(syntdenum, lp, lp_length);
-
- /* Calculate Weighting filter from quantized LSF */
- WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfold, lsf,
- WebRtcIlbcfix_kLsfWeight30ms[0],
- length);
- WebRtcIlbcfix_BwExpand(weightdenum, lp,
- (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpWeightDenum,
- (WebRtc_Word16)lp_length);
-
- /* subframe 2 to 6: Interpolation between first and second
- set of lsf coefficients */
-
- pos = lp_length;
- for (i = 1; i < iLBCenc_inst->nsub; i++) {
-
- /* Calculate Analysis/Syntehsis filter from quantized LSF */
- WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeq, lsfdeq2,
- WebRtcIlbcfix_kLsfWeight30ms[i],
- length);
- WEBRTC_SPL_MEMCPY_W16(syntdenum + pos, lp, lp_length);
-
- /* Calculate Weighting filter from quantized LSF */
- WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsf, lsf2,
- WebRtcIlbcfix_kLsfWeight30ms[i],
- length);
- WebRtcIlbcfix_BwExpand(weightdenum + pos, lp,
- (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpWeightDenum,
- (WebRtc_Word16)lp_length);
-
- pos += lp_length;
- }
-
- /* update memory */
-
- WEBRTC_SPL_MEMCPY_W16(lsfold, lsf2, length);
- WEBRTC_SPL_MEMCPY_W16(lsfdeqold, lsfdeq2, length);
-
- } else { /* iLBCenc_inst->mode==20 */
- pos = 0;
- for (i = 0; i < iLBCenc_inst->nsub; i++) {
-
- /* Calculate Analysis/Syntehsis filter from quantized LSF */
- WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfdeqold, lsfdeq,
- WebRtcIlbcfix_kLsfWeight20ms[i],
- length);
- WEBRTC_SPL_MEMCPY_W16(syntdenum + pos, lp, lp_length);
-
- /* Calculate Weighting filter from quantized LSF */
- WebRtcIlbcfix_LsfInterpolate2PloyEnc(lp, lsfold, lsf,
- WebRtcIlbcfix_kLsfWeight20ms[i],
- length);
- WebRtcIlbcfix_BwExpand(weightdenum+pos, lp,
- (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpWeightDenum,
- (WebRtc_Word16)lp_length);
-
- pos += lp_length;
- }
-
- /* update memory */
-
- WEBRTC_SPL_MEMCPY_W16(lsfold, lsf, length);
- WEBRTC_SPL_MEMCPY_W16(lsfdeqold, lsfdeq, length);
-
- }
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h b/trunk/src/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
deleted file mode 100644
index 8cdd7da..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/simple_interpolate_lsf.h
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_SimpleInterpolateLsf.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_INTERPOLATE_LSF_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * lsf interpolator (subrutine to LPCencode)
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_SimpleInterpolateLsf(
- WebRtc_Word16 *syntdenum, /* (o) the synthesis filter denominator
- resulting from the quantized
- interpolated lsf Q12 */
- WebRtc_Word16 *weightdenum, /* (o) the weighting filter denominator
- resulting from the unquantized
- interpolated lsf Q12 */
- WebRtc_Word16 *lsf, /* (i) the unquantized lsf coefficients Q13 */
- WebRtc_Word16 *lsfdeq, /* (i) the dequantized lsf coefficients Q13 */
- WebRtc_Word16 *lsfold, /* (i) the unquantized lsf coefficients of
- the previous signal frame Q13 */
- WebRtc_Word16 *lsfdeqold, /* (i) the dequantized lsf coefficients of the
- previous signal frame Q13 */
- WebRtc_Word16 length, /* (i) should equate FILTERORDER */
- iLBC_Enc_Inst_t *iLBCenc_inst
- /* (i/o) the encoder state structure */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
deleted file mode 100644
index 2d19edd..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.c
+++ /dev/null
@@ -1,94 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_SimpleLpcAnalysis.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "window32_w32.h"
-#include "bw_expand.h"
-#include "poly_to_lsf.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * lpc analysis (subrutine to LPCencode)
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_SimpleLpcAnalysis(
- WebRtc_Word16 *lsf, /* (o) lsf coefficients */
- WebRtc_Word16 *data, /* (i) new block of speech */
- iLBC_Enc_Inst_t *iLBCenc_inst
- /* (i/o) the encoder state structure */
- ) {
- int k;
- int scale;
- WebRtc_Word16 is;
- WebRtc_Word16 stability;
- /* Stack based */
- WebRtc_Word16 A[LPC_FILTERORDER + 1];
- WebRtc_Word32 R[LPC_FILTERORDER + 1];
- WebRtc_Word16 windowedData[BLOCKL_MAX];
- WebRtc_Word16 rc[LPC_FILTERORDER];
-
- is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl;
- WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lpc_buffer+is,data,iLBCenc_inst->blockl);
-
- /* No lookahead, last window is asymmetric */
-
- for (k = 0; k < iLBCenc_inst->lpc_n; k++) {
-
- is = LPC_LOOKBACK;
-
- if (k < (iLBCenc_inst->lpc_n - 1)) {
-
- /* Hanning table WebRtcIlbcfix_kLpcWin[] is in Q15-domain so the output is right-shifted 15 */
- WebRtcSpl_ElementwiseVectorMult(windowedData, iLBCenc_inst->lpc_buffer, WebRtcIlbcfix_kLpcWin, BLOCKL_MAX, 15);
- } else {
-
- /* Hanning table WebRtcIlbcfix_kLpcAsymWin[] is in Q15-domain so the output is right-shifted 15 */
- WebRtcSpl_ElementwiseVectorMult(windowedData, iLBCenc_inst->lpc_buffer+is, WebRtcIlbcfix_kLpcAsymWin, BLOCKL_MAX, 15);
- }
-
- /* Compute autocorrelation */
- WebRtcSpl_AutoCorrelation(windowedData, BLOCKL_MAX, LPC_FILTERORDER, R, &scale);
-
- /* Window autocorrelation vector */
- WebRtcIlbcfix_Window32W32(R, R, WebRtcIlbcfix_kLpcLagWin, LPC_FILTERORDER + 1 );
-
- /* Calculate the A coefficients from the Autocorrelation using Levinson Durbin algorithm */
- stability=WebRtcSpl_LevinsonDurbin(R, A, rc, LPC_FILTERORDER);
-
- /*
- Set the filter to {1.0, 0.0, 0.0,...} if filter from Levinson Durbin algorithm is unstable
- This should basically never happen...
- */
- if (stability!=1) {
- A[0]=4096;
- WebRtcSpl_MemSetW16(&A[1], 0, LPC_FILTERORDER);
- }
-
- /* Bandwidth expand the filter coefficients */
- WebRtcIlbcfix_BwExpand(A, A, (WebRtc_Word16*)WebRtcIlbcfix_kLpcChirpSyntDenum, LPC_FILTERORDER+1);
-
- /* Convert from A to LSF representation */
- WebRtcIlbcfix_Poly2Lsf(lsf + k*LPC_FILTERORDER, A);
- }
-
- is=LPC_LOOKBACK+BLOCKL_MAX-iLBCenc_inst->blockl;
- WEBRTC_SPL_MEMCPY_W16(iLBCenc_inst->lpc_buffer,
- iLBCenc_inst->lpc_buffer+LPC_LOOKBACK+BLOCKL_MAX-is, is);
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
deleted file mode 100644
index 83c1e5b..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lpc_analysis.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_SimpleLpcAnalysis.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LPC_ANALYSIS_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * lpc analysis (subrutine to LPCencode)
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_SimpleLpcAnalysis(
- WebRtc_Word16 *lsf, /* (o) lsf coefficients */
- WebRtc_Word16 *data, /* (i) new block of speech */
- iLBC_Enc_Inst_t *iLBCenc_inst
- /* (i/o) the encoder state structure */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c
deleted file mode 100644
index 7b5efa0..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.c
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_SimpleLsfDeQ.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * obtain dequantized lsf coefficients from quantization index
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_SimpleLsfDeQ(
- WebRtc_Word16 *lsfdeq, /* (o) dequantized lsf coefficients */
- WebRtc_Word16 *index, /* (i) quantization index */
- WebRtc_Word16 lpc_n /* (i) number of LPCs */
- ){
- int i, j, pos, cb_pos;
-
- /* decode first LSF */
-
- pos = 0;
- cb_pos = 0;
- for (i = 0; i < LSF_NSPLIT; i++) {
- for (j = 0; j < WebRtcIlbcfix_kLsfDimCb[i]; j++) {
- lsfdeq[pos + j] = WebRtcIlbcfix_kLsfCb[cb_pos +
- WEBRTC_SPL_MUL_16_16(index[i], WebRtcIlbcfix_kLsfDimCb[i]) + j];
- }
- pos += WebRtcIlbcfix_kLsfDimCb[i];
- cb_pos += WEBRTC_SPL_MUL_16_16(WebRtcIlbcfix_kLsfSizeCb[i], WebRtcIlbcfix_kLsfDimCb[i]);
- }
-
- if (lpc_n>1) {
- /* decode last LSF */
- pos = 0;
- cb_pos = 0;
- for (i = 0; i < LSF_NSPLIT; i++) {
- for (j = 0; j < WebRtcIlbcfix_kLsfDimCb[i]; j++) {
- lsfdeq[LPC_FILTERORDER + pos + j] = WebRtcIlbcfix_kLsfCb[cb_pos +
- WEBRTC_SPL_MUL_16_16(index[LSF_NSPLIT + i], WebRtcIlbcfix_kLsfDimCb[i]) + j];
- }
- pos += WebRtcIlbcfix_kLsfDimCb[i];
- cb_pos += WEBRTC_SPL_MUL_16_16(WebRtcIlbcfix_kLsfSizeCb[i], WebRtcIlbcfix_kLsfDimCb[i]);
- }
- }
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
deleted file mode 100644
index efd3103..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_dequant.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_SimpleLsfDeQ.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_DEQUANT_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * obtain dequantized lsf coefficients from quantization index
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_SimpleLsfDeQ(
- WebRtc_Word16 *lsfdeq, /* (o) dequantized lsf coefficients */
- WebRtc_Word16 *index, /* (i) quantization index */
- WebRtc_Word16 lpc_n /* (i) number of LPCs */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c
deleted file mode 100644
index aa27fb4..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_quant.c
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_SimpleLsfQ.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "split_vq.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * lsf quantizer (subrutine to LPCencode)
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_SimpleLsfQ(
- WebRtc_Word16 *lsfdeq, /* (o) dequantized lsf coefficients
- (dimension FILTERORDER) Q13 */
- WebRtc_Word16 *index, /* (o) quantization index */
- WebRtc_Word16 *lsf, /* (i) the lsf coefficient vector to be
- quantized (dimension FILTERORDER) Q13 */
- WebRtc_Word16 lpc_n /* (i) number of lsf sets to quantize */
- ){
-
- /* Quantize first LSF with memoryless split VQ */
- WebRtcIlbcfix_SplitVq( lsfdeq, index, lsf,
- (WebRtc_Word16*)WebRtcIlbcfix_kLsfCb, (WebRtc_Word16*)WebRtcIlbcfix_kLsfDimCb, (WebRtc_Word16*)WebRtcIlbcfix_kLsfSizeCb);
-
- if (lpc_n==2) {
- /* Quantize second LSF with memoryless split VQ */
- WebRtcIlbcfix_SplitVq( lsfdeq + LPC_FILTERORDER, index + LSF_NSPLIT,
- lsf + LPC_FILTERORDER, (WebRtc_Word16*)WebRtcIlbcfix_kLsfCb,
- (WebRtc_Word16*)WebRtcIlbcfix_kLsfDimCb, (WebRtc_Word16*)WebRtcIlbcfix_kLsfSizeCb);
- }
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h b/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
deleted file mode 100644
index fd17b2e..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/simple_lsf_quant.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_SimpleLsfQ.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SIMPLE_LSF_QUANT_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * lsf quantizer (subrutine to LPCencode)
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_SimpleLsfQ(
- WebRtc_Word16 *lsfdeq, /* (o) dequantized lsf coefficients
- (dimension FILTERORDER) Q13 */
- WebRtc_Word16 *index, /* (o) quantization index */
- WebRtc_Word16 *lsf, /* (i) the lsf coefficient vector to be
- quantized (dimension FILTERORDER) Q13 */
- WebRtc_Word16 lpc_n /* (i) number of lsf sets to quantize */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/smooth.c b/trunk/src/modules/audio_coding/codecs/ilbc/smooth.c
deleted file mode 100644
index b606077..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/smooth.c
+++ /dev/null
@@ -1,211 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Smooth.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-#include "smooth_out_data.h"
-
-/*----------------------------------------------------------------*
- * find the smoothed output data
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Smooth(
- WebRtc_Word16 *odata, /* (o) smoothed output */
- WebRtc_Word16 *current, /* (i) the un enhanced residual for
- this block */
- WebRtc_Word16 *surround /* (i) The approximation from the
- surrounding sequences */
- ) {
- WebRtc_Word16 maxtot, scale, scale1, scale2;
- WebRtc_Word16 A, B, C, denomW16;
- WebRtc_Word32 B_W32, denom, num;
- WebRtc_Word32 errs;
- WebRtc_Word32 w00,w10,w11, endiff, crit;
- WebRtc_Word32 w00prim, w10prim, w11_div_w00;
- WebRtc_Word16 w11prim;
- WebRtc_Word16 bitsw00, bitsw10, bitsw11;
- WebRtc_Word32 w11w00, w10w10, w00w00;
- WebRtc_Word16 max1, max2;
-
- /* compute some inner products (ensure no overflow by first calculating proper scale factor) */
-
- w00 = w10 = w11 = 0;
-
- max1=WebRtcSpl_MaxAbsValueW16(current, ENH_BLOCKL);
- max2=WebRtcSpl_MaxAbsValueW16(surround, ENH_BLOCKL);
- maxtot=WEBRTC_SPL_MAX(max1, max2);
-
- scale=WebRtcSpl_GetSizeInBits(maxtot);
- scale = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(2,scale)-26;
- scale=WEBRTC_SPL_MAX(0, scale);
-
- w00=WebRtcSpl_DotProductWithScale(current,current,ENH_BLOCKL,scale);
- w11=WebRtcSpl_DotProductWithScale(surround,surround,ENH_BLOCKL,scale);
- w10=WebRtcSpl_DotProductWithScale(surround,current,ENH_BLOCKL,scale);
-
- if (w00<0) w00 = WEBRTC_SPL_WORD32_MAX;
- if (w11<0) w11 = WEBRTC_SPL_WORD32_MAX;
-
- /* Rescale w00 and w11 to w00prim and w11prim, so that w00prim/w11prim
- is in Q16 */
-
- bitsw00 = WebRtcSpl_GetSizeInBits(w00);
- bitsw11 = WebRtcSpl_GetSizeInBits(w11);
- bitsw10 = WebRtcSpl_GetSizeInBits(WEBRTC_SPL_ABS_W32(w10));
- scale1 = 31 - bitsw00;
- scale2 = 15 - bitsw11;
-
- if (scale2>(scale1-16)) {
- scale2 = scale1 - 16;
- } else {
- scale1 = scale2 + 16;
- }
-
- w00prim = WEBRTC_SPL_LSHIFT_W32(w00, scale1);
- w11prim = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(w11, scale2);
-
- /* Perform C = sqrt(w11/w00) (C is in Q11 since (16+6)/2=11) */
- if (w11prim>64) {
- endiff = WEBRTC_SPL_LSHIFT_W32(
- (WebRtc_Word32)WebRtcSpl_DivW32W16(w00prim, w11prim), 6);
- C = (WebRtc_Word16)WebRtcSpl_SqrtFloor(endiff); /* C is in Q11 */
- } else {
- C = 1;
- }
-
- /* first try enhancement without power-constraint */
-
- errs = WebRtcIlbcfix_Smooth_odata(odata, current, surround, C);
-
-
-
- /* if constraint violated by first try, add constraint */
-
- if ( (6-scale+scale1) > 31) {
- crit=0;
- } else {
- /* crit = 0.05 * w00 (Result in Q-6) */
- crit = WEBRTC_SPL_SHIFT_W32(
- WEBRTC_SPL_MUL(ENH_A0, WEBRTC_SPL_RSHIFT_W32(w00prim, 14)),
- -(6-scale+scale1));
- }
-
- if (errs > crit) {
-
- if( w00 < 1) {
- w00=1;
- }
-
- /* Calculate w11*w00, w10*w10 and w00*w00 in the same Q domain */
-
- scale1 = bitsw00-15;
- scale2 = bitsw11-15;
-
- if (scale2>scale1) {
- scale = scale2;
- } else {
- scale = scale1;
- }
-
- w11w00 = WEBRTC_SPL_MUL_16_16(
- (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w11, -scale),
- (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w00, -scale));
-
- w10w10 = WEBRTC_SPL_MUL_16_16(
- (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w10, -scale),
- (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w10, -scale));
-
- w00w00 = WEBRTC_SPL_MUL_16_16(
- (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w00, -scale),
- (WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(w00, -scale));
-
- /* Calculate (w11*w00-w10*w10)/(w00*w00) in Q16 */
- if (w00w00>65536) {
- endiff = (w11w00-w10w10);
- endiff = WEBRTC_SPL_MAX(0, endiff);
- /* denom is in Q16 */
- denom = WebRtcSpl_DivW32W16(endiff, (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(w00w00, 16));
- } else {
- denom = 65536;
- }
-
- if( denom > 7){ /* eliminates numerical problems
- for if smooth */
-
- scale=WebRtcSpl_GetSizeInBits(denom)-15;
-
- if (scale>0) {
- /* denomW16 is in Q(16+scale) */
- denomW16=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(denom, scale);
-
- /* num in Q(34-scale) */
- num=WEBRTC_SPL_RSHIFT_W32(ENH_A0_MINUS_A0A0DIV4, scale);
- } else {
- /* denomW16 is in Q16 */
- denomW16=(WebRtc_Word16)denom;
-
- /* num in Q34 */
- num=ENH_A0_MINUS_A0A0DIV4;
- }
-
- /* A sqrt( (ENH_A0-(ENH_A0^2)/4)*(w00*w00)/(w11*w00 + w10*w10) ) in Q9 */
- A = (WebRtc_Word16)WebRtcSpl_SqrtFloor(WebRtcSpl_DivW32W16(num, denomW16));
-
- /* B_W32 is in Q30 ( B = 1 - ENH_A0/2 - A * w10/w00 ) */
- scale1 = 31-bitsw10;
- scale2 = 21-scale1;
- w10prim = WEBRTC_SPL_LSHIFT_W32(w10, scale1);
- w00prim = WEBRTC_SPL_SHIFT_W32(w00, -scale2);
- scale = bitsw00-scale2-15;
-
- if (scale>0) {
- w10prim=WEBRTC_SPL_RSHIFT_W32(w10prim, scale);
- w00prim=WEBRTC_SPL_RSHIFT_W32(w00prim, scale);
- }
-
- if ((w00prim>0)&&(w10prim>0)) {
- w11_div_w00=WebRtcSpl_DivW32W16(w10prim, (WebRtc_Word16)w00prim);
-
- if (WebRtcSpl_GetSizeInBits(w11_div_w00)+WebRtcSpl_GetSizeInBits(A)>31) {
- B_W32 = 0;
- } else {
- B_W32 = (WebRtc_Word32)1073741824 - (WebRtc_Word32)ENH_A0DIV2 -
- WEBRTC_SPL_MUL(A, w11_div_w00);
- }
- B = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(B_W32, 16); /* B in Q14 */
- } else {
- /* No smoothing */
- A = 0;
- B = 16384; /* 1 in Q14 */
- }
- }
- else{ /* essentially no difference between cycles;
- smoothing not needed */
-
- A = 0;
- B = 16384; /* 1 in Q14 */
- }
-
- /* create smoothed sequence */
-
- WebRtcSpl_ScaleAndAddVectors(surround, A, 9,
- current, B, 14,
- odata, ENH_BLOCKL);
- }
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/smooth.h b/trunk/src/modules/audio_coding/codecs/ilbc/smooth.h
deleted file mode 100644
index 88ce805..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/smooth.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Smooth.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * find the smoothed output data
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Smooth(
- WebRtc_Word16 *odata, /* (o) smoothed output */
- WebRtc_Word16 *current, /* (i) the un enhanced residual for
- this block */
- WebRtc_Word16 *surround /* (i) The approximation from the
- surrounding sequences */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/smooth_out_data.c b/trunk/src/modules/audio_coding/codecs/ilbc/smooth_out_data.c
deleted file mode 100644
index 9bacd85..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/smooth_out_data.c
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Smooth_odata.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-WebRtc_Word32 WebRtcIlbcfix_Smooth_odata(
- WebRtc_Word16 *odata,
- WebRtc_Word16 *psseq,
- WebRtc_Word16 *surround,
- WebRtc_Word16 C)
-{
- int i;
-
- WebRtc_Word16 err;
- WebRtc_Word32 errs;
-
- for(i=0;i<80;i++) {
- odata[i]= (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(
- (WEBRTC_SPL_MUL_16_16(C, surround[i])+1024), 11);
- }
-
- errs=0;
- for(i=0;i<80;i++) {
- err=(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W16((psseq[i]-odata[i]), 3);
- errs+=WEBRTC_SPL_MUL_16_16(err, err); /* errs in Q-6 */
- }
-
- return errs;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/smooth_out_data.h b/trunk/src/modules/audio_coding/codecs/ilbc/smooth_out_data.h
deleted file mode 100644
index 6fbe694..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/smooth_out_data.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Smooth_odata.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SMOOTH_OUT_DATA_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * help function to WebRtcIlbcfix_Smooth()
- *---------------------------------------------------------------*/
-
-WebRtc_Word32 WebRtcIlbcfix_Smooth_odata(
- WebRtc_Word16 *odata,
- WebRtc_Word16 *psseq,
- WebRtc_Word16 *surround,
- WebRtc_Word16 C);
-
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/sort_sq.c b/trunk/src/modules/audio_coding/codecs/ilbc/sort_sq.c
deleted file mode 100644
index 9276a7b..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/sort_sq.c
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_SortSq.c
-
-******************************************************************/
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * scalar quantization
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_SortSq(
- WebRtc_Word16 *xq, /* (o) the quantized value */
- WebRtc_Word16 *index, /* (o) the quantization index */
- WebRtc_Word16 x, /* (i) the value to quantize */
- const WebRtc_Word16 *cb, /* (i) the quantization codebook */
- WebRtc_Word16 cb_size /* (i) the size of the quantization codebook */
- ){
- int i;
-
- if (x <= cb[0]) {
- *index = 0;
- *xq = cb[0];
- } else {
- i = 0;
- while ((x > cb[i]) && (i < (cb_size-1))) {
- i++;
- }
-
- if (x > WEBRTC_SPL_RSHIFT_W32(( (WebRtc_Word32)cb[i] + cb[i - 1] + 1),1)) {
- *index = i;
- *xq = cb[i];
- } else {
- *index = i - 1;
- *xq = cb[i - 1];
- }
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/sort_sq.h b/trunk/src/modules/audio_coding/codecs/ilbc/sort_sq.h
deleted file mode 100644
index 2863dc5..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/sort_sq.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_SortSq.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SORT_SQ_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * scalar quantization
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_SortSq(
- WebRtc_Word16 *xq, /* (o) the quantized value */
- WebRtc_Word16 *index, /* (o) the quantization index */
- WebRtc_Word16 x, /* (i) the value to quantize */
- const WebRtc_Word16 *cb, /* (i) the quantization codebook */
- WebRtc_Word16 cb_size /* (i) the size of the quantization codebook */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/split_vq.c b/trunk/src/modules/audio_coding/codecs/ilbc/split_vq.c
deleted file mode 100644
index d908fa2..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/split_vq.c
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_SplitVq.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-#include "vq3.h"
-#include "vq4.h"
-
-/*----------------------------------------------------------------*
- * split vector quantization
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_SplitVq(
- WebRtc_Word16 *qX, /* (o) the quantized vector in Q13 */
- WebRtc_Word16 *index, /* (o) a vector of indexes for all vector
- codebooks in the split */
- WebRtc_Word16 *X, /* (i) the vector to quantize */
- WebRtc_Word16 *CB, /* (i) the quantizer codebook in Q13 */
- WebRtc_Word16 *dim, /* (i) the dimension of X and qX */
- WebRtc_Word16 *cbsize /* (i) the number of vectors in the codebook */
- ) {
-
- WebRtc_Word16 *qXPtr, *indexPtr, *CBPtr, *XPtr;
-
- /* Quantize X with the 3 vectror quantization tables */
-
- qXPtr=qX;
- indexPtr=index;
- CBPtr=CB;
- XPtr=X;
- WebRtcIlbcfix_Vq3(qXPtr, indexPtr, CBPtr, XPtr, cbsize[0]);
-
- qXPtr+=3;
- indexPtr+=1;
- CBPtr+=(dim[0]*cbsize[0]);
- XPtr+=3;
- WebRtcIlbcfix_Vq3(qXPtr, indexPtr, CBPtr, XPtr, cbsize[1]);
-
- qXPtr+=3;
- indexPtr+=1;
- CBPtr+=(dim[1]*cbsize[1]);
- XPtr+=3;
- WebRtcIlbcfix_Vq4(qXPtr, indexPtr, CBPtr, XPtr, cbsize[2]);
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/split_vq.h b/trunk/src/modules/audio_coding/codecs/ilbc/split_vq.h
deleted file mode 100644
index 7264a21..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/split_vq.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_SplitVq.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SPLIT_VQ_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * split vector quantization
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_SplitVq(
- WebRtc_Word16 *qX, /* (o) the quantized vector in Q13 */
- WebRtc_Word16 *index, /* (o) a vector of indexes for all vector
- codebooks in the split */
- WebRtc_Word16 *X, /* (i) the vector to quantize */
- WebRtc_Word16 *CB, /* (i) the quantizer codebook in Q13 */
- WebRtc_Word16 *dim, /* (i) the dimension of X and qX */
- WebRtc_Word16 *cbsize /* (i) the number of vectors in the codebook */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/state_construct.c b/trunk/src/modules/audio_coding/codecs/ilbc/state_construct.c
deleted file mode 100644
index 9d03cc3..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/state_construct.c
+++ /dev/null
@@ -1,111 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_StateConstruct.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * decoding of the start state
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_StateConstruct(
- WebRtc_Word16 idxForMax, /* (i) 6-bit index for the quantization of
- max amplitude */
- WebRtc_Word16 *idxVec, /* (i) vector of quantization indexes */
- WebRtc_Word16 *syntDenum, /* (i) synthesis filter denumerator */
- WebRtc_Word16 *Out_fix, /* (o) the decoded state vector */
- WebRtc_Word16 len /* (i) length of a state vector */
- ) {
- int k;
- WebRtc_Word16 maxVal;
- WebRtc_Word16 *tmp1, *tmp2, *tmp3;
- /* Stack based */
- WebRtc_Word16 numerator[1+LPC_FILTERORDER];
- WebRtc_Word16 sampleValVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
- WebRtc_Word16 sampleMaVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
- WebRtc_Word16 *sampleVal = &sampleValVec[LPC_FILTERORDER];
- WebRtc_Word16 *sampleMa = &sampleMaVec[LPC_FILTERORDER];
- WebRtc_Word16 *sampleAr = &sampleValVec[LPC_FILTERORDER];
-
- /* initialization of coefficients */
-
- for (k=0; k<LPC_FILTERORDER+1; k++){
- numerator[k] = syntDenum[LPC_FILTERORDER-k];
- }
-
- /* decoding of the maximum value */
-
- maxVal = WebRtcIlbcfix_kFrgQuantMod[idxForMax];
-
- /* decoding of the sample values */
- tmp1 = sampleVal;
- tmp2 = &idxVec[len-1];
-
- if (idxForMax<37) {
- for(k=0; k<len; k++){
- /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 2097152 (= 0.5 << 22)
- maxVal is in Q8 and result is in Q(-1) */
- (*tmp1) = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(maxVal,WebRtcIlbcfix_kStateSq3[(*tmp2)])+(WebRtc_Word32)2097152) >> 22);
- tmp1++;
- tmp2--;
- }
- } else if (idxForMax<59) {
- for(k=0; k<len; k++){
- /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 262144 (= 0.5 << 19)
- maxVal is in Q5 and result is in Q(-1) */
- (*tmp1) = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(maxVal,WebRtcIlbcfix_kStateSq3[(*tmp2)])+(WebRtc_Word32)262144) >> 19);
- tmp1++;
- tmp2--;
- }
- } else {
- for(k=0; k<len; k++){
- /*the shifting is due to the Q13 in sq4_fixQ13[i], also the adding of 65536 (= 0.5 << 17)
- maxVal is in Q3 and result is in Q(-1) */
- (*tmp1) = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(maxVal,WebRtcIlbcfix_kStateSq3[(*tmp2)])+(WebRtc_Word32)65536) >> 17);
- tmp1++;
- tmp2--;
- }
- }
-
- /* Set the rest of the data to zero */
- WebRtcSpl_MemSetW16(&sampleVal[len], 0, len);
-
- /* circular convolution with all-pass filter */
-
- /* Set the state to zero */
- WebRtcSpl_MemSetW16(sampleValVec, 0, (LPC_FILTERORDER));
-
- /* Run MA filter + AR filter */
- WebRtcSpl_FilterMAFastQ12(
- sampleVal, sampleMa,
- numerator, LPC_FILTERORDER+1, (WebRtc_Word16)(len + LPC_FILTERORDER));
- WebRtcSpl_MemSetW16(&sampleMa[len + LPC_FILTERORDER], 0, (len - LPC_FILTERORDER));
- WebRtcSpl_FilterARFastQ12(
- sampleMa, sampleAr,
- syntDenum, LPC_FILTERORDER+1, (WebRtc_Word16)(2*len));
-
- tmp1 = &sampleAr[len-1];
- tmp2 = &sampleAr[2*len-1];
- tmp3 = Out_fix;
- for(k=0;k<len;k++){
- (*tmp3) = (*tmp1) + (*tmp2);
- tmp1--;
- tmp2--;
- tmp3++;
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/state_construct.h b/trunk/src/modules/audio_coding/codecs/ilbc/state_construct.h
deleted file mode 100644
index 465699b..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/state_construct.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_StateConstruct.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_CONSTRUCT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_CONSTRUCT_H_
-
-/*----------------------------------------------------------------*
- * Generate the start state from the quantized indexes
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_StateConstruct(
- WebRtc_Word16 idxForMax, /* (i) 6-bit index for the quantization of
- max amplitude */
- WebRtc_Word16 *idxVec, /* (i) vector of quantization indexes */
- WebRtc_Word16 *syntDenum, /* (i) synthesis filter denumerator */
- WebRtc_Word16 *Out_fix, /* (o) the decoded state vector */
- WebRtc_Word16 len /* (i) length of a state vector */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/state_search.c b/trunk/src/modules/audio_coding/codecs/ilbc/state_search.c
deleted file mode 100644
index 824a0ba..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/state_search.c
+++ /dev/null
@@ -1,119 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_StateSearch.c
-
-******************************************************************/
-
-#include "defines.h"
-#include "constants.h"
-#include "abs_quant.h"
-
-/*----------------------------------------------------------------*
- * encoding of start state
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_StateSearch(
- iLBC_Enc_Inst_t *iLBCenc_inst,
- /* (i) Encoder instance */
- iLBC_bits *iLBC_encbits,/* (i/o) Encoded bits (output idxForMax
- and idxVec, input state_first) */
- WebRtc_Word16 *residual, /* (i) target residual vector */
- WebRtc_Word16 *syntDenum, /* (i) lpc synthesis filter */
- WebRtc_Word16 *weightDenum /* (i) weighting filter denuminator */
- ) {
- WebRtc_Word16 k, index;
- WebRtc_Word16 maxVal;
- WebRtc_Word16 scale, shift;
- WebRtc_Word32 maxValsq;
- WebRtc_Word16 scaleRes;
- WebRtc_Word16 max;
- int i;
- /* Stack based */
- WebRtc_Word16 numerator[1+LPC_FILTERORDER];
- WebRtc_Word16 residualLongVec[2*STATE_SHORT_LEN_30MS+LPC_FILTERORDER];
- WebRtc_Word16 sampleMa[2*STATE_SHORT_LEN_30MS];
- WebRtc_Word16 *residualLong = &residualLongVec[LPC_FILTERORDER];
- WebRtc_Word16 *sampleAr = residualLong;
-
- /* Scale to maximum 12 bits to avoid saturation in circular convolution filter */
- max = WebRtcSpl_MaxAbsValueW16(residual, iLBCenc_inst->state_short_len);
- scaleRes = WebRtcSpl_GetSizeInBits(max)-12;
- scaleRes = WEBRTC_SPL_MAX(0, scaleRes);
- /* Set up the filter coefficients for the circular convolution */
- for (i=0; i<LPC_FILTERORDER+1; i++) {
- numerator[i] = (syntDenum[LPC_FILTERORDER-i]>>scaleRes);
- }
-
- /* Copy the residual to a temporary buffer that we can filter
- * and set the remaining samples to zero.
- */
- WEBRTC_SPL_MEMCPY_W16(residualLong, residual, iLBCenc_inst->state_short_len);
- WebRtcSpl_MemSetW16(residualLong + iLBCenc_inst->state_short_len, 0, iLBCenc_inst->state_short_len);
-
- /* Run the Zero-Pole filter (Ciurcular convolution) */
- WebRtcSpl_MemSetW16(residualLongVec, 0, LPC_FILTERORDER);
- WebRtcSpl_FilterMAFastQ12(
- residualLong, sampleMa,
- numerator, LPC_FILTERORDER+1, (WebRtc_Word16)(iLBCenc_inst->state_short_len + LPC_FILTERORDER));
- WebRtcSpl_MemSetW16(&sampleMa[iLBCenc_inst->state_short_len + LPC_FILTERORDER], 0, iLBCenc_inst->state_short_len - LPC_FILTERORDER);
-
- WebRtcSpl_FilterARFastQ12(
- sampleMa, sampleAr,
- syntDenum, LPC_FILTERORDER+1, (WebRtc_Word16)(2*iLBCenc_inst->state_short_len));
-
- for(k=0;k<iLBCenc_inst->state_short_len;k++){
- sampleAr[k] += sampleAr[k+iLBCenc_inst->state_short_len];
- }
-
- /* Find maximum absolute value in the vector */
- maxVal=WebRtcSpl_MaxAbsValueW16(sampleAr, iLBCenc_inst->state_short_len);
-
- /* Find the best index */
-
- if ((((WebRtc_Word32)maxVal)<<scaleRes)<23170) {
- maxValsq=((WebRtc_Word32)maxVal*maxVal)<<(2+2*scaleRes);
- } else {
- maxValsq=(WebRtc_Word32)WEBRTC_SPL_WORD32_MAX;
- }
-
- index=0;
- for (i=0;i<63;i++) {
-
- if (maxValsq>=WebRtcIlbcfix_kChooseFrgQuant[i]) {
- index=i+1;
- } else {
- i=63;
- }
- }
- iLBC_encbits->idxForMax=index;
-
- /* Rescale the vector before quantization */
- scale=WebRtcIlbcfix_kScale[index];
-
- if (index<27) { /* scale table is in Q16, fout[] is in Q(-1) and we want the result to be in Q11 */
- shift=4;
- } else { /* scale table is in Q21, fout[] is in Q(-1) and we want the result to be in Q11 */
- shift=9;
- }
-
- /* Set up vectors for AbsQuant and rescale it with the scale factor */
- WebRtcSpl_ScaleVectorWithSat(sampleAr, sampleAr, scale,
- iLBCenc_inst->state_short_len, (WebRtc_Word16)(shift-scaleRes));
-
- /* Quantize the values in fout[] */
- WebRtcIlbcfix_AbsQuant(iLBCenc_inst, iLBC_encbits, sampleAr, weightDenum);
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/state_search.h b/trunk/src/modules/audio_coding/codecs/ilbc/state_search.h
deleted file mode 100644
index 8b7f298..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/state_search.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_StateSearch.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_STATE_SEARCH_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * encoding of start state
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_StateSearch(
- iLBC_Enc_Inst_t *iLBCenc_inst,
- /* (i) Encoder instance */
- iLBC_bits *iLBC_encbits,/* (i/o) Encoded bits (output idxForMax
- and idxVec, input state_first) */
- WebRtc_Word16 *residual, /* (i) target residual vector */
- WebRtc_Word16 *syntDenum, /* (i) lpc synthesis filter */
- WebRtc_Word16 *weightDenum /* (i) weighting filter denuminator */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/swap_bytes.c b/trunk/src/modules/audio_coding/codecs/ilbc/swap_bytes.c
deleted file mode 100644
index 61b8b7b..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/swap_bytes.c
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_SwapBytes.c
-
-******************************************************************/
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Swap bytes (to simplify operations on Little Endian machines)
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_SwapBytes(
- WebRtc_UWord16 *sequence, /* (i/o) the sequence to swap */
- WebRtc_Word16 wordLength /* (i) number or WebRtc_UWord16 to swap */
- ) {
- int k;
- WebRtc_UWord16 temp=0;
- for( k=wordLength; k>0; k-- ) {
- temp = (*sequence >> 8)|(*sequence << 8);
- *sequence++ = temp;
- //*sequence++ = (*sequence >> 8) | (*sequence << 8);
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/swap_bytes.h b/trunk/src/modules/audio_coding/codecs/ilbc/swap_bytes.h
deleted file mode 100644
index 2f2b3eb..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/swap_bytes.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_SwapBytes.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_SWAP_BYTES_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * Swap bytes (to simplify operations on Little Endian machines)
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_SwapBytes(
- WebRtc_UWord16 *sequence, /* (i/o) the sequence to swap */
- WebRtc_Word16 wordLength /* (i) number or WebRtc_UWord16 to swap */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_test.c b/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
deleted file mode 100644
index d2c9ba5..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_test.c
+++ /dev/null
@@ -1,225 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- iLBC_test.c
-
-******************************************************************/
-
-#include <stdlib.h>
-#include <stdio.h>
-#include <string.h>
-#include "ilbc.h"
-
-/*---------------------------------------------------------------*
- * Main program to test iLBC encoding and decoding
- *
- * Usage:
- * exefile_name.exe <infile> <bytefile> <outfile> <channel>
- *
- * <infile> : Input file, speech for encoder (16-bit pcm file)
- * <bytefile> : Bit stream output from the encoder
- * <outfile> : Output file, decoded speech (16-bit pcm file)
- * <channel> : Bit error file, optional (16-bit)
- * 1 - Packet received correctly
- * 0 - Packet Lost
- *
- *--------------------------------------------------------------*/
-
-#define BLOCKL_MAX 240
-#define ILBCNOOFWORDS_MAX 25
-
-
-int main(int argc, char* argv[])
-{
-
- FILE *ifileid,*efileid,*ofileid, *cfileid;
- WebRtc_Word16 data[BLOCKL_MAX];
- WebRtc_Word16 encoded_data[ILBCNOOFWORDS_MAX], decoded_data[BLOCKL_MAX];
- int len;
- short pli, mode;
- int blockcount = 0;
- int packetlosscount = 0;
- int frameLen;
- WebRtc_Word16 speechType;
- iLBC_encinst_t *Enc_Inst;
- iLBC_decinst_t *Dec_Inst;
-
-#ifdef __ILBC_WITH_40BITACC
- /* Doublecheck that long long exists */
- if (sizeof(long)>=sizeof(long long)) {
- fprintf(stderr, "40-bit simulation is not be supported on this platform\n");
- exit(0);
- }
-#endif
-
- /* get arguments and open files */
-
- if ((argc!=5) && (argc!=6)) {
- fprintf(stderr,
- "\n*-----------------------------------------------*\n");
- fprintf(stderr,
- " %s <20,30> input encoded decoded (channel)\n\n",
- argv[0]);
- fprintf(stderr,
- " mode : Frame size for the encoding/decoding\n");
- fprintf(stderr,
- " 20 - 20 ms\n");
- fprintf(stderr,
- " 30 - 30 ms\n");
- fprintf(stderr,
- " input : Speech for encoder (16-bit pcm file)\n");
- fprintf(stderr,
- " encoded : Encoded bit stream\n");
- fprintf(stderr,
- " decoded : Decoded speech (16-bit pcm file)\n");
- fprintf(stderr,
- " channel : Packet loss pattern, optional (16-bit)\n");
- fprintf(stderr,
- " 1 - Packet received correctly\n");
- fprintf(stderr,
- " 0 - Packet Lost\n");
- fprintf(stderr,
- "*-----------------------------------------------*\n\n");
- exit(1);
- }
- mode=atoi(argv[1]);
- if (mode != 20 && mode != 30) {
- fprintf(stderr,"Wrong mode %s, must be 20, or 30\n",
- argv[1]);
- exit(2);
- }
- if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
- fprintf(stderr,"Cannot open input file %s\n", argv[2]);
- exit(2);}
- if ( (efileid=fopen(argv[3],"wb")) == NULL) {
- fprintf(stderr, "Cannot open encoded file file %s\n",
- argv[3]); exit(1);}
- if ( (ofileid=fopen(argv[4],"wb")) == NULL) {
- fprintf(stderr, "Cannot open decoded file %s\n",
- argv[4]); exit(1);}
- if (argc==6) {
- if( (cfileid=fopen(argv[5],"rb")) == NULL) {
- fprintf(stderr, "Cannot open channel file %s\n",
- argv[5]);
- exit(1);
- }
- } else {
- cfileid=NULL;
- }
-
- /* print info */
-
- fprintf(stderr, "\n");
- fprintf(stderr,
- "*---------------------------------------------------*\n");
- fprintf(stderr,
- "* *\n");
- fprintf(stderr,
- "* iLBC test program *\n");
- fprintf(stderr,
- "* *\n");
- fprintf(stderr,
- "* *\n");
- fprintf(stderr,
- "*---------------------------------------------------*\n");
- fprintf(stderr,"\nMode : %2d ms\n", mode);
- fprintf(stderr,"Input file : %s\n", argv[2]);
- fprintf(stderr,"Encoded file : %s\n", argv[3]);
- fprintf(stderr,"Output file : %s\n", argv[4]);
- if (argc==6) {
- fprintf(stderr,"Channel file : %s\n", argv[5]);
- }
- fprintf(stderr,"\n");
-
- /* Create structs */
- WebRtcIlbcfix_EncoderCreate(&Enc_Inst);
- WebRtcIlbcfix_DecoderCreate(&Dec_Inst);
-
-
- /* Initialization */
-
- WebRtcIlbcfix_EncoderInit(Enc_Inst, mode);
- WebRtcIlbcfix_DecoderInit(Dec_Inst, mode);
- frameLen = mode*8;
-
- /* loop over input blocks */
-
- while (((WebRtc_Word16)fread(data,sizeof(WebRtc_Word16),frameLen,ifileid))==
- frameLen) {
-
- blockcount++;
-
- /* encoding */
-
- fprintf(stderr, "--- Encoding block %i --- ",blockcount);
- len=WebRtcIlbcfix_Encode(Enc_Inst, data, (WebRtc_Word16)frameLen, encoded_data);
- fprintf(stderr, "\r");
-
- /* write byte file */
-
- fwrite(encoded_data, sizeof(WebRtc_Word16), ((len+1)/sizeof(WebRtc_Word16)), efileid);
-
- /* get channel data if provided */
- if (argc==6) {
- if (fread(&pli, sizeof(WebRtc_Word16), 1, cfileid)) {
- if ((pli!=0)&&(pli!=1)) {
- fprintf(stderr, "Error in channel file\n");
- exit(0);
- }
- if (pli==0) {
- /* Packet loss -> remove info from frame */
- memset(encoded_data, 0,
- sizeof(WebRtc_Word16)*ILBCNOOFWORDS_MAX);
- packetlosscount++;
- }
- } else {
- fprintf(stderr, "Error. Channel file too short\n");
- exit(0);
- }
- } else {
- pli=1;
- }
-
- /* decoding */
-
- fprintf(stderr, "--- Decoding block %i --- ",blockcount);
- if (pli==1) {
- len=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data,
- (WebRtc_Word16)len, decoded_data,&speechType);
- } else {
- len=WebRtcIlbcfix_DecodePlc(Dec_Inst, decoded_data, 1);
- }
- fprintf(stderr, "\r");
-
- /* write output file */
-
- fwrite(decoded_data,sizeof(WebRtc_Word16),len,ofileid);
- }
-
- /* close files */
-
- fclose(ifileid); fclose(efileid); fclose(ofileid);
- if (argc==6) {
- fclose(cfileid);
- }
-
- /* Free structs */
- WebRtcIlbcfix_EncoderFree(Enc_Inst);
- WebRtcIlbcfix_DecoderFree(Dec_Inst);
-
-
- printf("\nDone with simulation\n\n");
-
- return(0);
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c b/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
deleted file mode 100644
index ee5e484..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_testLib.c
+++ /dev/null
@@ -1,207 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
-iLBC Speech Coder ANSI-C Source Code
-
-iLBC_test.c
-
-******************************************************************/
-
-#include <math.h>
-#include <stdlib.h>
-#include <stdio.h>
-#include <string.h>
-#include <time.h>
-#include "ilbc.h"
-
-//#define JUNK_DATA
-#ifdef JUNK_DATA
-#define SEED_FILE "randseed.txt"
-#endif
-
-
-/*----------------------------------------------------------------*
-* Main program to test iLBC encoding and decoding
-*
-* Usage:
-* exefile_name.exe <infile> <bytefile> <outfile>
-*
-*---------------------------------------------------------------*/
-
-int main(int argc, char* argv[])
-{
- FILE *ifileid,*efileid,*ofileid, *chfileid;
- short encoded_data[55], data[240], speechType;
- short len, mode, pli;
- int blockcount = 0;
-
- iLBC_encinst_t *Enc_Inst;
- iLBC_decinst_t *Dec_Inst;
-#ifdef JUNK_DATA
- int i;
- FILE *seedfile;
- unsigned int random_seed = (unsigned int) time(NULL);//1196764538
-#endif
-
- /* Create structs */
- WebRtcIlbcfix_EncoderCreate(&Enc_Inst);
- WebRtcIlbcfix_DecoderCreate(&Dec_Inst);
-
- /* get arguments and open files */
-
- if (argc != 6 ) {
- fprintf(stderr, "%s mode inputfile bytefile outputfile channelfile\n",
- argv[0]);
- fprintf(stderr, "Example:\n");
- fprintf(stderr, "%s <30,20> in.pcm byte.dat out.pcm T30.0.dat\n", argv[0]);
- exit(1);
- }
- mode=atoi(argv[1]);
- if (mode != 20 && mode != 30) {
- fprintf(stderr,"Wrong mode %s, must be 20, or 30\n", argv[1]);
- exit(2);
- }
- if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
- fprintf(stderr,"Cannot open input file %s\n", argv[2]);
- exit(2);}
- if ( (efileid=fopen(argv[3],"wb")) == NULL) {
- fprintf(stderr, "Cannot open channelfile file %s\n",
- argv[3]); exit(3);}
- if( (ofileid=fopen(argv[4],"wb")) == NULL) {
- fprintf(stderr, "Cannot open output file %s\n",
- argv[4]); exit(3);}
- if ( (chfileid=fopen(argv[5],"rb")) == NULL) {
- fprintf(stderr,"Cannot open channel file file %s\n", argv[5]);
- exit(2);
- }
- /* print info */
- fprintf(stderr, "\n");
- fprintf(stderr,
- "*---------------------------------------------------*\n");
- fprintf(stderr,
- "* *\n");
- fprintf(stderr,
- "* iLBCtest *\n");
- fprintf(stderr,
- "* *\n");
- fprintf(stderr,
- "* *\n");
- fprintf(stderr,
- "*---------------------------------------------------*\n");
-#ifdef SPLIT_10MS
- fprintf(stderr,"\n10ms split with raw mode: %2d ms\n", mode);
-#else
- fprintf(stderr,"\nMode : %2d ms\n", mode);
-#endif
- fprintf(stderr,"\nInput file : %s\n", argv[2]);
- fprintf(stderr,"Coded file : %s\n", argv[3]);
- fprintf(stderr,"Output file : %s\n\n", argv[4]);
- fprintf(stderr,"Channel file : %s\n\n", argv[5]);
-
-#ifdef JUNK_DATA
- srand(random_seed);
-
- if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
- fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
- }
- else {
- fprintf(seedfile, "%u\n", random_seed);
- fclose(seedfile);
- }
-#endif
-
- /* Initialization */
- WebRtcIlbcfix_EncoderInit(Enc_Inst, mode);
- WebRtcIlbcfix_DecoderInit(Dec_Inst, mode);
-
- /* loop over input blocks */
-#ifdef SPLIT_10MS
- while(fread(data, sizeof(short), 80, ifileid) == 80) {
-#else
- while((short)fread(data,sizeof(short),(mode<<3),ifileid)==(mode<<3)) {
-#endif
- blockcount++;
-
- /* encoding */
- fprintf(stderr, "--- Encoding block %i --- ",blockcount);
-#ifdef SPLIT_10MS
- len=WebRtcIlbcfix_Encode(Enc_Inst, data, 80, encoded_data);
-#else
- len=WebRtcIlbcfix_Encode(Enc_Inst, data, (short)(mode<<3), encoded_data);
-#endif
- fprintf(stderr, "\r");
-
-#ifdef JUNK_DATA
- for ( i = 0; i < len; i++) {
- encoded_data[i] = (short) (encoded_data[i] + (short) rand());
- }
-#endif
- /* write byte file */
- if(len != 0){ //len may be 0 in 10ms split case
- fwrite(encoded_data,1,len,efileid);
- }
-
- if(len != 0){ //len may be 0 in 10ms split case
- /* get channel data if provided */
- if (argc==6) {
- if (fread(&pli, sizeof(WebRtc_Word16), 1, chfileid)) {
- if ((pli!=0)&&(pli!=1)) {
- fprintf(stderr, "Error in channel file\n");
- exit(0);
- }
- if (pli==0) {
- /* Packet loss -> remove info from frame */
- memset(encoded_data, 0, sizeof(WebRtc_Word16)*25);
- }
- } else {
- fprintf(stderr, "Error. Channel file too short\n");
- exit(0);
- }
- } else {
- pli=1;
- }
-
- /* decoding */
- fprintf(stderr, "--- Decoding block %i --- ",blockcount);
- if (pli==1) {
- len=WebRtcIlbcfix_Decode(Dec_Inst, encoded_data, len, data, &speechType);
- } else {
- len=WebRtcIlbcfix_DecodePlc(Dec_Inst, data, 1);
- }
- fprintf(stderr, "\r");
-
- /* write output file */
- fwrite(data,sizeof(short),len,ofileid);
- }
- }
-
-#ifdef JUNK_DATA
- if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
- fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
- }
- else {
- fprintf(seedfile, "ok\n\n");
- fclose(seedfile);
- }
-#endif
-
- /* free structs */
- WebRtcIlbcfix_EncoderFree(Enc_Inst);
- WebRtcIlbcfix_DecoderFree(Dec_Inst);
-
- /* close files */
- fclose(ifileid);
- fclose(efileid);
- fclose(ofileid);
-
- return 0;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c b/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
deleted file mode 100644
index f67945e..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBC_testprogram.c
+++ /dev/null
@@ -1,343 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- iLBC_test.c
-
-******************************************************************/
-
-#include <math.h>
-#include <stdlib.h>
-#include <stdio.h>
-#include <string.h>
-
-#include "defines.h"
-#include "nit_encode.h"
-#include "encode.h"
-#include "init_decode.h"
-#include "decode.h"
-#include "constants.h"
-#include "ilbc.h"
-
-#define ILBCNOOFWORDS_MAX (NO_OF_BYTES_30MS)/2
-
-/* Runtime statistics */
-#include <time.h>
-/* #define CLOCKS_PER_SEC 1000 */
-
-/*----------------------------------------------------------------*
- * Encoder interface function
- *---------------------------------------------------------------*/
-
-short encode( /* (o) Number of bytes encoded */
- iLBC_Enc_Inst_t *iLBCenc_inst, /* (i/o) Encoder instance */
- WebRtc_Word16 *encoded_data, /* (o) The encoded bytes */
- WebRtc_Word16 *data /* (i) The signal block to encode */
- ){
-
- /* do the actual encoding */
- WebRtcIlbcfix_Encode((WebRtc_UWord16 *)encoded_data, data, iLBCenc_inst);
-
- return (iLBCenc_inst->no_of_bytes);
-}
-
-/*----------------------------------------------------------------*
- * Decoder interface function
- *---------------------------------------------------------------*/
-
-short decode( /* (o) Number of decoded samples */
- iLBC_Dec_Inst_t *iLBCdec_inst, /* (i/o) Decoder instance */
- short *decoded_data, /* (o) Decoded signal block */
- short *encoded_data, /* (i) Encoded bytes */
- short mode /* (i) 0=PL, 1=Normal */
- ){
-
- /* check if mode is valid */
-
- if (mode<0 || mode>1) {
- printf("\nERROR - Wrong mode - 0, 1 allowed\n"); exit(3);}
-
- /* do actual decoding of block */
-
- WebRtcIlbcfix_Decode(decoded_data, (WebRtc_UWord16 *)encoded_data,
- iLBCdec_inst, mode);
-
- return (iLBCdec_inst->blockl);
-}
-
-/*----------------------------------------------------------------*
- * Main program to test iLBC encoding and decoding
- *
- * Usage:
- * exefile_name.exe <infile> <bytefile> <outfile> <channelfile>
- *
- *---------------------------------------------------------------*/
-
-#define MAXFRAMES 10000
-#define MAXFILELEN (BLOCKL_MAX*MAXFRAMES)
-
-int main(int argc, char* argv[])
-{
-
- /* Runtime statistics */
-
- float starttime1, starttime2;
- float runtime1, runtime2;
- float outtime;
-
- FILE *ifileid,*efileid,*ofileid, *chfileid;
- short *inputdata, *encodeddata, *decodeddata;
- short *channeldata;
- int blockcount = 0, noOfBlocks=0, i, noOfLostBlocks=0;
- short mode;
- iLBC_Enc_Inst_t Enc_Inst;
- iLBC_Dec_Inst_t Dec_Inst;
-
- short frameLen;
- short count;
-#ifdef SPLIT_10MS
- short size;
-#endif
-
- inputdata=(short*) malloc(MAXFILELEN*sizeof(short));
- if (inputdata==NULL) {
- fprintf(stderr,"Could not allocate memory for vector\n");
- exit(0);
- }
- encodeddata=(short*) malloc(ILBCNOOFWORDS_MAX*MAXFRAMES*sizeof(short));
- if (encodeddata==NULL) {
- fprintf(stderr,"Could not allocate memory for vector\n");
- free(inputdata);
- exit(0);
- }
- decodeddata=(short*) malloc(MAXFILELEN*sizeof(short));
- if (decodeddata==NULL) {
- fprintf(stderr,"Could not allocate memory for vector\n");
- free(inputdata);
- free(encodeddata);
- exit(0);
- }
- channeldata=(short*) malloc(MAXFRAMES*sizeof(short));
- if (channeldata==NULL) {
- fprintf(stderr,"Could not allocate memory for vector\n");
- free(inputdata);
- free(encodeddata);
- free(decodeddata);
- exit(0);
- }
-
- /* get arguments and open files */
-
- if (argc != 6 ) {
- fprintf(stderr, "%s mode inputfile bytefile outputfile channelfile\n",
- argv[0]);
- fprintf(stderr, "Example:\n");
- fprintf(stderr, "%s <30,20> in.pcm byte.dat out.pcm T30.0.dat\n", argv[0]);
- exit(1);
- }
- mode=atoi(argv[1]);
- if (mode != 20 && mode != 30) {
- fprintf(stderr,"Wrong mode %s, must be 20, or 30\n", argv[1]);
- exit(2);
- }
- if ( (ifileid=fopen(argv[2],"rb")) == NULL) {
- fprintf(stderr,"Cannot open input file %s\n", argv[2]);
- exit(2);}
- if ( (efileid=fopen(argv[3],"wb")) == NULL) {
- fprintf(stderr, "Cannot open channelfile file %s\n",
- argv[3]); exit(3);}
- if( (ofileid=fopen(argv[4],"wb")) == NULL) {
- fprintf(stderr, "Cannot open output file %s\n",
- argv[4]); exit(3);}
- if ( (chfileid=fopen(argv[5],"rb")) == NULL) {
- fprintf(stderr,"Cannot open channel file file %s\n", argv[5]);
- exit(2);}
-
-
- /* print info */
-#ifndef PRINT_MIPS
- fprintf(stderr, "\n");
- fprintf(stderr,
- "*---------------------------------------------------*\n");
- fprintf(stderr,
- "* *\n");
- fprintf(stderr,
- "* iLBCtest *\n");
- fprintf(stderr,
- "* *\n");
- fprintf(stderr,
- "* *\n");
- fprintf(stderr,
- "*---------------------------------------------------*\n");
-#ifdef SPLIT_10MS
- fprintf(stderr,"\n10ms split with raw mode: %2d ms\n", mode);
-#else
- fprintf(stderr,"\nMode : %2d ms\n", mode);
-#endif
- fprintf(stderr,"\nInput file : %s\n", argv[2]);
- fprintf(stderr,"Coded file : %s\n", argv[3]);
- fprintf(stderr,"Output file : %s\n\n", argv[4]);
- fprintf(stderr,"Channel file : %s\n\n", argv[5]);
-#endif
-
- /* Initialization */
-
- WebRtcIlbcfix_EncoderInit(&Enc_Inst, mode);
- WebRtcIlbcfix_DecoderInit(&Dec_Inst, mode, 1);
-
- /* extract the input file and channel file */
-
-#ifdef SPLIT_10MS
- frameLen = (mode==20)? 80:160;
- fread(Enc_Inst.past_samples, sizeof(short), frameLen, ifileid);
- Enc_Inst.section = 0;
-
- while( fread(&inputdata[noOfBlocks*80], sizeof(short),
- 80, ifileid) == 80 ) {
- noOfBlocks++;
- }
-
- noOfBlocks += frameLen/80;
- frameLen = 80;
-#else
- frameLen = Enc_Inst.blockl;
-
- while( fread(&inputdata[noOfBlocks*Enc_Inst.blockl],sizeof(short),
- Enc_Inst.blockl,ifileid)==(WebRtc_UWord16)Enc_Inst.blockl){
- noOfBlocks++;
- }
-#endif
-
-
- while ((fread(&channeldata[blockcount],sizeof(short), 1,chfileid)==1)
- && ( blockcount < noOfBlocks/(Enc_Inst.blockl/frameLen) )) {
- blockcount++;
- }
-
- if ( blockcount < noOfBlocks/(Enc_Inst.blockl/frameLen) ) {
- fprintf(stderr,"Channel file %s is too short\n", argv[4]);
- free(inputdata);
- free(encodeddata);
- free(decodeddata);
- free(channeldata);
- exit(0);
- }
-
- count=0;
-
- /* Runtime statistics */
-
- starttime1 = clock()/(float)CLOCKS_PER_SEC;
-
- /* Encoding loop */
-#ifdef PRINT_MIPS
- printf("-1 -1\n");
-#endif
-
-#ifdef SPLIT_10MS
- /* "Enc_Inst.section != 0" is to make sure we run through full
- lengths of all vectors for 10ms split mode.
- */
- // while( (count < noOfBlocks) || (Enc_Inst.section != 0) ) {
- while( count < blockcount * (Enc_Inst.blockl/frameLen) ) {
-
- encode(&Enc_Inst, &encodeddata[Enc_Inst.no_of_words *
- (count/(Enc_Inst.nsub/2))],
- &inputdata[frameLen * count] );
-#else
- while (count < noOfBlocks) {
- encode( &Enc_Inst, &encodeddata[Enc_Inst.no_of_words * count],
- &inputdata[frameLen * count] );
-#endif
-
-#ifdef PRINT_MIPS
- printf("-1 -1\n");
-#endif
-
- count++;
- }
-
- count=0;
-
- /* Runtime statistics */
-
- starttime2=clock()/(float)CLOCKS_PER_SEC;
- runtime1 = (float)(starttime2-starttime1);
-
- /* Decoding loop */
-
- while (count < blockcount) {
- if (channeldata[count]==1) {
- /* Normal decoding */
- decode(&Dec_Inst, &decodeddata[count * Dec_Inst.blockl],
- &encodeddata[Dec_Inst.no_of_words * count], 1);
- } else if (channeldata[count]==0) {
- /* PLC */
- short emptydata[ILBCNOOFWORDS_MAX];
- memset(emptydata, 0, Dec_Inst.no_of_words*sizeof(short));
- decode(&Dec_Inst, &decodeddata[count*Dec_Inst.blockl],
- emptydata, 0);
- noOfLostBlocks++;
- } else {
- printf("Error in channel file (values have to be either 1 or 0)\n");
- exit(0);
- }
-#ifdef PRINT_MIPS
- printf("-1 -1\n");
-#endif
-
- count++;
- }
-
- /* Runtime statistics */
-
- runtime2 = (float)(clock()/(float)CLOCKS_PER_SEC-starttime2);
-
- outtime = (float)((float)blockcount*
- (float)mode/1000.0);
-
-#ifndef PRINT_MIPS
- printf("\nLength of speech file: %.1f s\n", outtime);
- printf("Lost frames : %.1f%%\n\n", 100*(float)noOfLostBlocks/(float)blockcount);
-
- printf("Time to run iLBC_encode+iLBC_decode:");
- printf(" %.1f s (%.1f%% of realtime)\n", runtime1+runtime2,
- (100*(runtime1+runtime2)/outtime));
-
- printf("Time in iLBC_encode :");
- printf(" %.1f s (%.1f%% of total runtime)\n",
- runtime1, 100.0*runtime1/(runtime1+runtime2));
-
- printf("Time in iLBC_decode :");
- printf(" %.1f s (%.1f%% of total runtime)\n\n",
- runtime2, 100.0*runtime2/(runtime1+runtime2));
-#endif
-
- /* Write data to files */
- for (i=0; i<blockcount; i++) {
- fwrite(&encodeddata[i*Enc_Inst.no_of_words], sizeof(short),
- Enc_Inst.no_of_words, efileid);
- }
- for (i=0;i<blockcount;i++) {
- fwrite(&decodeddata[i*Enc_Inst.blockl],sizeof(short),Enc_Inst.blockl,ofileid);
- }
-
- /* return memory and close files */
-
- free(inputdata);
- free(encodeddata);
- free(decodeddata);
- free(channeldata);
- fclose(ifileid); fclose(efileid); fclose(ofileid);
- return(0);
- }
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBCtestscript.txt b/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBCtestscript.txt
deleted file mode 100644
index a52f7c1..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/test/iLBCtestscript.txt
+++ /dev/null
@@ -1,73 +0,0 @@
-#!/bin/bash
-(set -o igncr) 2>/dev/null && set -o igncr; # force bash to ignore \r character
-
-#
-# This script can be used to verify the bit exactness of iLBC fixed-point version 1.0.6
-#
-
-INP=../../../../../../../test/data/audio_coding
-EXEP=../../../../../../../out/Release
-OUTP=./GeneratedFiles
-mkdir ./GeneratedFiles
-
-$EXEP/iLBCtest 20 $INP/F00.INP $OUTP/F00.BIT20 $OUTP/F00.OUT20 $INP/clean.chn
-$EXEP/iLBCtest 20 $INP/F01.INP $OUTP/F01.BIT20 $OUTP/F01.OUT20 $INP/clean.chn
-$EXEP/iLBCtest 20 $INP/F02.INP $OUTP/F02.BIT20 $OUTP/F02.OUT20 $INP/clean.chn
-$EXEP/iLBCtest 20 $INP/F03.INP $OUTP/F03.BIT20 $OUTP/F03.OUT20 $INP/clean.chn
-$EXEP/iLBCtest 20 $INP/F04.INP $OUTP/F04.BIT20 $OUTP/F04.OUT20 $INP/clean.chn
-$EXEP/iLBCtest 20 $INP/F05.INP $OUTP/F05.BIT20 $OUTP/F05.OUT20 $INP/clean.chn
-$EXEP/iLBCtest 20 $INP/F06.INP $OUTP/F06.BIT20 $OUTP/F06.OUT20 $INP/clean.chn
-
-$EXEP/iLBCtest 30 $INP/F00.INP $OUTP/F00.BIT30 $OUTP/F00.OUT30 $INP/clean.chn
-$EXEP/iLBCtest 30 $INP/F01.INP $OUTP/F01.BIT30 $OUTP/F01.OUT30 $INP/clean.chn
-$EXEP/iLBCtest 30 $INP/F02.INP $OUTP/F02.BIT30 $OUTP/F02.OUT30 $INP/clean.chn
-$EXEP/iLBCtest 30 $INP/F03.INP $OUTP/F03.BIT30 $OUTP/F03.OUT30 $INP/clean.chn
-$EXEP/iLBCtest 30 $INP/F04.INP $OUTP/F04.BIT30 $OUTP/F04.OUT30 $INP/clean.chn
-$EXEP/iLBCtest 30 $INP/F05.INP $OUTP/F05.BIT30 $OUTP/F05.OUT30 $INP/clean.chn
-$EXEP/iLBCtest 30 $INP/F06.INP $OUTP/F06.BIT30 $OUTP/F06.OUT30 $INP/clean.chn
-
-$EXEP/iLBCtest 20 $INP/F00.INP $OUTP/F00.BIT20 $OUTP/F00_tlm10.OUT20 $INP/tlm10.chn
-$EXEP/iLBCtest 20 $INP/F01.INP $OUTP/F01.BIT20 $OUTP/F01_tlm10.OUT20 $INP/tlm10.chn
-$EXEP/iLBCtest 20 $INP/F02.INP $OUTP/F02.BIT20 $OUTP/F02_tlm10.OUT20 $INP/tlm10.chn
-$EXEP/iLBCtest 30 $INP/F00.INP $OUTP/F00.BIT30 $OUTP/F00_tlm10.OUT30 $INP/tlm10.chn
-$EXEP/iLBCtest 30 $INP/F01.INP $OUTP/F01.BIT30 $OUTP/F01_tlm10.OUT30 $INP/tlm10.chn
-$EXEP/iLBCtest 30 $INP/F02.INP $OUTP/F02.BIT30 $OUTP/F02_tlm10.OUT30 $INP/tlm10.chn
-
-
-diff $OUTP/F00.BIT20 $INP/F00.BIT20
-diff $OUTP/F01.BIT20 $INP/F01.BIT20
-diff $OUTP/F02.BIT20 $INP/F02.BIT20
-diff $OUTP/F03.BIT20 $INP/F03.BIT20
-diff $OUTP/F04.BIT20 $INP/F04.BIT20
-diff $OUTP/F05.BIT20 $INP/F05.BIT20
-diff $OUTP/F06.BIT20 $INP/F06.BIT20
-diff $OUTP/F00.OUT20 $INP/F00.OUT20
-diff $OUTP/F01.OUT20 $INP/F01.OUT20
-diff $OUTP/F02.OUT20 $INP/F02.OUT20
-diff $OUTP/F03.OUT20 $INP/F03.OUT20
-diff $OUTP/F04.OUT20 $INP/F04.OUT20
-diff $OUTP/F05.OUT20 $INP/F05.OUT20
-diff $OUTP/F06.OUT20 $INP/F06.OUT20
-
-diff $OUTP/F00.BIT30 $INP/F00.BIT30
-diff $OUTP/F01.BIT30 $INP/F01.BIT30
-diff $OUTP/F02.BIT30 $INP/F02.BIT30
-diff $OUTP/F03.BIT30 $INP/F03.BIT30
-diff $OUTP/F04.BIT30 $INP/F04.BIT30
-diff $OUTP/F05.BIT30 $INP/F05.BIT30
-diff $OUTP/F06.BIT30 $INP/F06.BIT30
-diff $OUTP/F00.OUT30 $INP/F00.OUT30
-diff $OUTP/F01.OUT30 $INP/F01.OUT30
-diff $OUTP/F02.OUT30 $INP/F02.OUT30
-diff $OUTP/F03.OUT30 $INP/F03.OUT30
-diff $OUTP/F04.OUT30 $INP/F04.OUT30
-diff $OUTP/F05.OUT30 $INP/F05.OUT30
-diff $OUTP/F06.OUT30 $INP/F06.OUT30
-
-diff $OUTP/F00_tlm10.OUT20 $INP/F00_tlm10.OUT20
-diff $OUTP/F01_tlm10.OUT20 $INP/F01_tlm10.OUT20
-diff $OUTP/F02_tlm10.OUT20 $INP/F02_tlm10.OUT20
-diff $OUTP/F00_tlm10.OUT30 $INP/F00_tlm10.OUT30
-diff $OUTP/F01_tlm10.OUT30 $INP/F01_tlm10.OUT30
-diff $OUTP/F02_tlm10.OUT30 $INP/F02_tlm10.OUT30
-
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/unpack_bits.c b/trunk/src/modules/audio_coding/codecs/ilbc/unpack_bits.c
deleted file mode 100644
index 6c883a7..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/unpack_bits.c
+++ /dev/null
@@ -1,239 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_UnpackBits.c
-
-******************************************************************/
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * unpacking of bits from bitstream, i.e., vector of bytes
- *---------------------------------------------------------------*/
-
-WebRtc_Word16 WebRtcIlbcfix_UnpackBits( /* (o) "Empty" frame indicator */
- WebRtc_UWord16 *bitstream, /* (i) The packatized bitstream */
- iLBC_bits *enc_bits, /* (o) Paramerers from bitstream */
- WebRtc_Word16 mode /* (i) Codec mode (20 or 30) */
- ) {
- WebRtc_UWord16 *bitstreamPtr;
- int i, k;
- WebRtc_Word16 *tmpPtr;
-
- bitstreamPtr=bitstream;
-
- /* First WebRtc_Word16 */
- enc_bits->lsf[0] = (*bitstreamPtr)>>10; /* Bit 0..5 */
- enc_bits->lsf[1] = ((*bitstreamPtr)>>3)&0x7F; /* Bit 6..12 */
- enc_bits->lsf[2] = ((*bitstreamPtr)&0x7)<<4; /* Bit 13..15 */
- bitstreamPtr++;
- /* Second WebRtc_Word16 */
- enc_bits->lsf[2] |= ((*bitstreamPtr)>>12)&0xF; /* Bit 0..3 */
-
- if (mode==20) {
- enc_bits->startIdx = ((*bitstreamPtr)>>10)&0x3; /* Bit 4..5 */
- enc_bits->state_first = ((*bitstreamPtr)>>9)&0x1; /* Bit 6 */
- enc_bits->idxForMax = ((*bitstreamPtr)>>3)&0x3F; /* Bit 7..12 */
- enc_bits->cb_index[0] = ((*bitstreamPtr)&0x7)<<4; /* Bit 13..15 */
- bitstreamPtr++;
- /* Third WebRtc_Word16 */
- enc_bits->cb_index[0] |= ((*bitstreamPtr)>>12)&0xE; /* Bit 0..2 */
- enc_bits->gain_index[0] = ((*bitstreamPtr)>>8)&0x18; /* Bit 3..4 */
- enc_bits->gain_index[1] = ((*bitstreamPtr)>>7)&0x8; /* Bit 5 */
- enc_bits->cb_index[3] = ((*bitstreamPtr)>>2)&0xFE; /* Bit 6..12 */
- enc_bits->gain_index[3] = ((*bitstreamPtr)<<2)&0x10; /* Bit 13 */
- enc_bits->gain_index[4] = ((*bitstreamPtr)<<2)&0x8; /* Bit 14 */
- enc_bits->gain_index[6] = ((*bitstreamPtr)<<4)&0x10; /* Bit 15 */
- } else { /* mode==30 */
- enc_bits->lsf[3] = ((*bitstreamPtr)>>6)&0x3F; /* Bit 4..9 */
- enc_bits->lsf[4] = ((*bitstreamPtr)<<1)&0x7E; /* Bit 10..15 */
- bitstreamPtr++;
- /* Third WebRtc_Word16 */
- enc_bits->lsf[4] |= ((*bitstreamPtr)>>15)&0x1; /* Bit 0 */
- enc_bits->lsf[5] = ((*bitstreamPtr)>>8)&0x7F; /* Bit 1..7 */
- enc_bits->startIdx = ((*bitstreamPtr)>>5)&0x7; /* Bit 8..10 */
- enc_bits->state_first = ((*bitstreamPtr)>>4)&0x1; /* Bit 11 */
- enc_bits->idxForMax = ((*bitstreamPtr)<<2)&0x3C; /* Bit 12..15 */
- bitstreamPtr++;
- /* 4:th WebRtc_Word16 */
- enc_bits->idxForMax |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1 */
- enc_bits->cb_index[0] = ((*bitstreamPtr)>>7)&0x78; /* Bit 2..5 */
- enc_bits->gain_index[0] = ((*bitstreamPtr)>>5)&0x10; /* Bit 6 */
- enc_bits->gain_index[1] = ((*bitstreamPtr)>>5)&0x8; /* Bit 7 */
- enc_bits->cb_index[3] = ((*bitstreamPtr))&0xFC; /* Bit 8..13 */
- enc_bits->gain_index[3] = ((*bitstreamPtr)<<3)&0x10; /* Bit 14 */
- enc_bits->gain_index[4] = ((*bitstreamPtr)<<3)&0x8; /* Bit 15 */
- }
- /* Class 2 bits of ULP */
- /* 4:th to 6:th WebRtc_Word16 for 20 ms case
- 5:th to 7:th WebRtc_Word16 for 30 ms case */
- bitstreamPtr++;
- tmpPtr=enc_bits->idxVec;
- for (k=0; k<3; k++) {
- for (i=15; i>=0; i--) {
- (*tmpPtr) = (((*bitstreamPtr)>>i)<<2)&0x4;
- /* Bit 15-i */
- tmpPtr++;
- }
- bitstreamPtr++;
- }
-
- if (mode==20) {
- /* 7:th WebRtc_Word16 */
- for (i=15; i>6; i--) {
- (*tmpPtr) = (((*bitstreamPtr)>>i)<<2)&0x4;
- /* Bit 15-i */
- tmpPtr++;
- }
- enc_bits->gain_index[1] |= ((*bitstreamPtr)>>4)&0x4; /* Bit 9 */
- enc_bits->gain_index[3] |= ((*bitstreamPtr)>>2)&0xC; /* Bit 10..11 */
- enc_bits->gain_index[4] |= ((*bitstreamPtr)>>1)&0x4; /* Bit 12 */
- enc_bits->gain_index[6] |= ((*bitstreamPtr)<<1)&0x8; /* Bit 13 */
- enc_bits->gain_index[7] = ((*bitstreamPtr)<<2)&0xC; /* Bit 14..15 */
-
- } else { /* mode==30 */
- /* 8:th WebRtc_Word16 */
- for (i=15; i>5; i--) {
- (*tmpPtr) = (((*bitstreamPtr)>>i)<<2)&0x4;
- /* Bit 15-i */
- tmpPtr++;
- }
- enc_bits->cb_index[0] |= ((*bitstreamPtr)>>3)&0x6; /* Bit 10..11 */
- enc_bits->gain_index[0] |= ((*bitstreamPtr))&0x8; /* Bit 12 */
- enc_bits->gain_index[1] |= ((*bitstreamPtr))&0x4; /* Bit 13 */
- enc_bits->cb_index[3] |= ((*bitstreamPtr))&0x2; /* Bit 14 */
- enc_bits->cb_index[6] = ((*bitstreamPtr)<<7)&0x80; /* Bit 15 */
- bitstreamPtr++;
- /* 9:th WebRtc_Word16 */
- enc_bits->cb_index[6] |= ((*bitstreamPtr)>>9)&0x7E; /* Bit 0..5 */
- enc_bits->cb_index[9] = ((*bitstreamPtr)>>2)&0xFE; /* Bit 6..12 */
- enc_bits->cb_index[12] = ((*bitstreamPtr)<<5)&0xE0; /* Bit 13..15 */
- bitstreamPtr++;
- /* 10:th WebRtc_Word16 */
- enc_bits->cb_index[12] |= ((*bitstreamPtr)>>11)&0x1E;/* Bit 0..3 */
- enc_bits->gain_index[3] |= ((*bitstreamPtr)>>8)&0xC; /* Bit 4..5 */
- enc_bits->gain_index[4] |= ((*bitstreamPtr)>>7)&0x6; /* Bit 6..7 */
- enc_bits->gain_index[6] = ((*bitstreamPtr)>>3)&0x18; /* Bit 8..9 */
- enc_bits->gain_index[7] = ((*bitstreamPtr)>>2)&0xC; /* Bit 10..11 */
- enc_bits->gain_index[9] = ((*bitstreamPtr)<<1)&0x10; /* Bit 12 */
- enc_bits->gain_index[10] = ((*bitstreamPtr)<<1)&0x8; /* Bit 13 */
- enc_bits->gain_index[12] = ((*bitstreamPtr)<<3)&0x10; /* Bit 14 */
- enc_bits->gain_index[13] = ((*bitstreamPtr)<<3)&0x8; /* Bit 15 */
- }
- bitstreamPtr++;
- /* Class 3 bits of ULP */
- /* 8:th to 14:th WebRtc_Word16 for 20 ms case
- 11:th to 17:th WebRtc_Word16 for 30 ms case */
- tmpPtr=enc_bits->idxVec;
- for (k=0; k<7; k++) {
- for (i=14; i>=0; i-=2) {
- (*tmpPtr) |= ((*bitstreamPtr)>>i)&0x3; /* Bit 15-i..14-i*/
- tmpPtr++;
- }
- bitstreamPtr++;
- }
-
- if (mode==20) {
- /* 15:th WebRtc_Word16 */
- enc_bits->idxVec[56] |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1 */
- enc_bits->cb_index[0] |= ((*bitstreamPtr)>>13)&0x1; /* Bit 2 */
- enc_bits->cb_index[1] = ((*bitstreamPtr)>>6)&0x7F; /* Bit 3..9 */
- enc_bits->cb_index[2] = ((*bitstreamPtr)<<1)&0x7E; /* Bit 10..15 */
- bitstreamPtr++;
- /* 16:th WebRtc_Word16 */
- enc_bits->cb_index[2] |= ((*bitstreamPtr)>>15)&0x1; /* Bit 0 */
- enc_bits->gain_index[0] |= ((*bitstreamPtr)>>12)&0x7; /* Bit 1..3 */
- enc_bits->gain_index[1] |= ((*bitstreamPtr)>>10)&0x3; /* Bit 4..5 */
- enc_bits->gain_index[2] = ((*bitstreamPtr)>>7)&0x7; /* Bit 6..8 */
- enc_bits->cb_index[3] |= ((*bitstreamPtr)>>6)&0x1; /* Bit 9 */
- enc_bits->cb_index[4] = ((*bitstreamPtr)<<1)&0x7E; /* Bit 10..15 */
- bitstreamPtr++;
- /* 17:th WebRtc_Word16 */
- enc_bits->cb_index[4] |= ((*bitstreamPtr)>>15)&0x1; /* Bit 0 */
- enc_bits->cb_index[5] = ((*bitstreamPtr)>>8)&0x7F; /* Bit 1..7 */
- enc_bits->cb_index[6] = ((*bitstreamPtr))&0xFF; /* Bit 8..15 */
- bitstreamPtr++;
- /* 18:th WebRtc_Word16 */
- enc_bits->cb_index[7] = (*bitstreamPtr)>>8; /* Bit 0..7 */
- enc_bits->cb_index[8] = (*bitstreamPtr)&0xFF; /* Bit 8..15 */
- bitstreamPtr++;
- /* 19:th WebRtc_Word16 */
- enc_bits->gain_index[3] |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1 */
- enc_bits->gain_index[4] |= ((*bitstreamPtr)>>12)&0x3; /* Bit 2..3 */
- enc_bits->gain_index[5] = ((*bitstreamPtr)>>9)&0x7; /* Bit 4..6 */
- enc_bits->gain_index[6] |= ((*bitstreamPtr)>>6)&0x7; /* Bit 7..9 */
- enc_bits->gain_index[7] |= ((*bitstreamPtr)>>4)&0x3; /* Bit 10..11 */
- enc_bits->gain_index[8] = ((*bitstreamPtr)>>1)&0x7; /* Bit 12..14 */
- } else { /* mode==30 */
- /* 18:th WebRtc_Word16 */
- enc_bits->idxVec[56] |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1 */
- enc_bits->idxVec[57] |= ((*bitstreamPtr)>>12)&0x3; /* Bit 2..3 */
- enc_bits->cb_index[0] |= ((*bitstreamPtr)>>11)&1; /* Bit 4 */
- enc_bits->cb_index[1] = ((*bitstreamPtr)>>4)&0x7F; /* Bit 5..11 */
- enc_bits->cb_index[2] = ((*bitstreamPtr)<<3)&0x78; /* Bit 12..15 */
- bitstreamPtr++;
- /* 19:th WebRtc_Word16 */
- enc_bits->cb_index[2] |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2 */
- enc_bits->gain_index[0] |= ((*bitstreamPtr)>>10)&0x7; /* Bit 3..5 */
- enc_bits->gain_index[1] |= ((*bitstreamPtr)>>8)&0x3; /* Bit 6..7 */
- enc_bits->gain_index[2] = ((*bitstreamPtr)>>5)&0x7; /* Bit 8..10 */
- enc_bits->cb_index[3] |= ((*bitstreamPtr)>>4)&0x1; /* Bit 11 */
- enc_bits->cb_index[4] = ((*bitstreamPtr)<<3)&0x78; /* Bit 12..15 */
- bitstreamPtr++;
- /* 20:th WebRtc_Word16 */
- enc_bits->cb_index[4] |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2 */
- enc_bits->cb_index[5] = ((*bitstreamPtr)>>6)&0x7F; /* Bit 3..9 */
- enc_bits->cb_index[6] |= ((*bitstreamPtr)>>5)&0x1; /* Bit 10 */
- enc_bits->cb_index[7] = ((*bitstreamPtr)<<3)&0xF8; /* Bit 11..15 */
- bitstreamPtr++;
- /* 21:st WebRtc_Word16 */
- enc_bits->cb_index[7] |= ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2 */
- enc_bits->cb_index[8] = ((*bitstreamPtr)>>5)&0xFF; /* Bit 3..10 */
- enc_bits->cb_index[9] |= ((*bitstreamPtr)>>4)&0x1; /* Bit 11 */
- enc_bits->cb_index[10] = ((*bitstreamPtr)<<4)&0xF0; /* Bit 12..15 */
- bitstreamPtr++;
- /* 22:nd WebRtc_Word16 */
- enc_bits->cb_index[10] |= ((*bitstreamPtr)>>12)&0xF; /* Bit 0..3 */
- enc_bits->cb_index[11] = ((*bitstreamPtr)>>4)&0xFF; /* Bit 4..11 */
- enc_bits->cb_index[12] |= ((*bitstreamPtr)>>3)&0x1; /* Bit 12 */
- enc_bits->cb_index[13] = ((*bitstreamPtr)<<5)&0xE0; /* Bit 13..15 */
- bitstreamPtr++;
- /* 23:rd WebRtc_Word16 */
- enc_bits->cb_index[13] |= ((*bitstreamPtr)>>11)&0x1F;/* Bit 0..4 */
- enc_bits->cb_index[14] = ((*bitstreamPtr)>>3)&0xFF; /* Bit 5..12 */
- enc_bits->gain_index[3] |= ((*bitstreamPtr)>>1)&0x3; /* Bit 13..14 */
- enc_bits->gain_index[4] |= ((*bitstreamPtr)&0x1); /* Bit 15 */
- bitstreamPtr++;
- /* 24:rd WebRtc_Word16 */
- enc_bits->gain_index[5] = ((*bitstreamPtr)>>13)&0x7; /* Bit 0..2 */
- enc_bits->gain_index[6] |= ((*bitstreamPtr)>>10)&0x7; /* Bit 3..5 */
- enc_bits->gain_index[7] |= ((*bitstreamPtr)>>8)&0x3; /* Bit 6..7 */
- enc_bits->gain_index[8] = ((*bitstreamPtr)>>5)&0x7; /* Bit 8..10 */
- enc_bits->gain_index[9] |= ((*bitstreamPtr)>>1)&0xF; /* Bit 11..14 */
- enc_bits->gain_index[10] |= ((*bitstreamPtr)<<2)&0x4; /* Bit 15 */
- bitstreamPtr++;
- /* 25:rd WebRtc_Word16 */
- enc_bits->gain_index[10] |= ((*bitstreamPtr)>>14)&0x3; /* Bit 0..1 */
- enc_bits->gain_index[11] = ((*bitstreamPtr)>>11)&0x7; /* Bit 2..4 */
- enc_bits->gain_index[12] |= ((*bitstreamPtr)>>7)&0xF; /* Bit 5..8 */
- enc_bits->gain_index[13] |= ((*bitstreamPtr)>>4)&0x7; /* Bit 9..11 */
- enc_bits->gain_index[14] = ((*bitstreamPtr)>>1)&0x7; /* Bit 12..14 */
- }
- /* Last bit should be zero, otherwise it's an "empty" frame */
- if (((*bitstreamPtr)&0x1) == 1) {
- return(1);
- } else {
- return(0);
- }
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/unpack_bits.h b/trunk/src/modules/audio_coding/codecs/ilbc/unpack_bits.h
deleted file mode 100644
index 864865f..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/unpack_bits.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_UnpackBits.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_UNPACK_BITS_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * unpacking of bits from bitstream, i.e., vector of bytes
- *---------------------------------------------------------------*/
-
-WebRtc_Word16 WebRtcIlbcfix_UnpackBits( /* (o) "Empty" frame indicator */
- WebRtc_UWord16 *bitstream, /* (i) The packatized bitstream */
- iLBC_bits *enc_bits, /* (o) Paramerers from bitstream */
- WebRtc_Word16 mode /* (i) Codec mode (20 or 30) */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/vq3.c b/trunk/src/modules/audio_coding/codecs/ilbc/vq3.c
deleted file mode 100644
index 81d1bfa..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/vq3.c
+++ /dev/null
@@ -1,63 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Vq3.c
-
-******************************************************************/
-
-#include "vq3.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * vector quantization
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Vq3(
- WebRtc_Word16 *Xq, /* quantized vector (Q13) */
- WebRtc_Word16 *index,
- WebRtc_Word16 *CB, /* codebook in Q13 */
- WebRtc_Word16 *X, /* vector to quantize (Q13) */
- WebRtc_Word16 n_cb
- ){
- WebRtc_Word16 i, j;
- WebRtc_Word16 pos, minindex=0;
- WebRtc_Word16 tmp;
- WebRtc_Word32 dist, mindist;
-
- pos = 0;
- mindist = WEBRTC_SPL_WORD32_MAX; /* start value */
-
- /* Find the codebook with the lowest square distance */
- for (j = 0; j < n_cb; j++) {
- tmp = X[0] - CB[pos];
- dist = WEBRTC_SPL_MUL_16_16(tmp, tmp);
- for (i = 1; i < 3; i++) {
- tmp = X[i] - CB[pos + i];
- dist += WEBRTC_SPL_MUL_16_16(tmp, tmp);
- }
-
- if (dist < mindist) {
- mindist = dist;
- minindex = j;
- }
- pos += 3;
- }
-
- /* Store the quantized codebook and the index */
- for (i = 0; i < 3; i++) {
- Xq[i] = CB[minindex*3 + i];
- }
- *index = minindex;
-
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/vq3.h b/trunk/src/modules/audio_coding/codecs/ilbc/vq3.h
deleted file mode 100644
index f2628e0..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/vq3.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Vq3.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ3_H_
-
-#include "typedefs.h"
-
-/*----------------------------------------------------------------*
- * Vector quantization of order 3 (based on MSE)
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Vq3(
- WebRtc_Word16 *Xq, /* (o) the quantized vector (Q13) */
- WebRtc_Word16 *index, /* (o) the quantization index */
- WebRtc_Word16 *CB, /* (i) the vector quantization codebook (Q13) */
- WebRtc_Word16 *X, /* (i) the vector to quantize (Q13) */
- WebRtc_Word16 n_cb /* (i) the number of vectors in the codebook */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/vq4.c b/trunk/src/modules/audio_coding/codecs/ilbc/vq4.c
deleted file mode 100644
index 3d4c26d..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/vq4.c
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Vq4.c
-
-******************************************************************/
-
-#include "vq4.h"
-#include "constants.h"
-
-/*----------------------------------------------------------------*
- * vector quantization
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Vq4(
- WebRtc_Word16 *Xq, /* quantized vector (Q13) */
- WebRtc_Word16 *index,
- WebRtc_Word16 *CB, /* codebook in Q13 */
- WebRtc_Word16 *X, /* vector to quantize (Q13) */
- WebRtc_Word16 n_cb
- ){
- WebRtc_Word16 i, j;
- WebRtc_Word16 pos, minindex=0;
- WebRtc_Word16 tmp;
- WebRtc_Word32 dist, mindist;
-
- pos = 0;
- mindist = WEBRTC_SPL_WORD32_MAX; /* start value */
-
- /* Find the codebook with the lowest square distance */
- for (j = 0; j < n_cb; j++) {
- tmp = X[0] - CB[pos];
- dist = WEBRTC_SPL_MUL_16_16(tmp, tmp);
- for (i = 1; i < 4; i++) {
- tmp = X[i] - CB[pos + i];
- dist += WEBRTC_SPL_MUL_16_16(tmp, tmp);
- }
-
- if (dist < mindist) {
- mindist = dist;
- minindex = j;
- }
- pos += 4;
- }
-
- /* Store the quantized codebook and the index */
- for (i = 0; i < 4; i++) {
- Xq[i] = CB[minindex*4 + i];
- }
- *index = minindex;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/vq4.h b/trunk/src/modules/audio_coding/codecs/ilbc/vq4.h
deleted file mode 100644
index 1b8cff2..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/vq4.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Vq4.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_VQ4_H_
-
-#include "typedefs.h"
-
-/*----------------------------------------------------------------*
- * Vector quantization of order 4 (based on MSE)
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Vq4(
- WebRtc_Word16 *Xq, /* (o) the quantized vector (Q13) */
- WebRtc_Word16 *index, /* (o) the quantization index */
- WebRtc_Word16 *CB, /* (i) the vector quantization codebook (Q13) */
- WebRtc_Word16 *X, /* (i) the vector to quantize (Q13) */
- WebRtc_Word16 n_cb /* (i) the number of vectors in the codebook */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/window32_w32.c b/trunk/src/modules/audio_coding/codecs/ilbc/window32_w32.c
deleted file mode 100644
index b0e8406..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/window32_w32.c
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Window32W32.c
-
-******************************************************************/
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * window multiplication
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Window32W32(
- WebRtc_Word32 *z, /* Output */
- WebRtc_Word32 *x, /* Input (same domain as Output)*/
- const WebRtc_Word32 *y, /* Q31 Window */
- WebRtc_Word16 N /* length to process */
- ) {
- WebRtc_Word16 i;
- WebRtc_Word16 x_low, x_hi, y_low, y_hi;
- WebRtc_Word16 left_shifts;
- WebRtc_Word32 temp;
-
- left_shifts = (WebRtc_Word16)WebRtcSpl_NormW32(x[0]);
- WebRtcSpl_VectorBitShiftW32(x, N, x, (WebRtc_Word16)(-left_shifts));
-
-
- /* The double precision numbers use a special representation:
- * w32 = hi<<16 + lo<<1
- */
- for (i = 0; i < N; i++) {
- /* Extract higher bytes */
- x_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(x[i], 16);
- y_hi = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(y[i], 16);
-
- /* Extract lower bytes, defined as (w32 - hi<<16)>>1 */
- temp = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)x_hi, 16);
- x_low = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32((x[i] - temp), 1);
-
- temp = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)y_hi, 16);
- y_low = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32((y[i] - temp), 1);
-
- /* Calculate z by a 32 bit multiplication using both low and high from x and y */
- temp = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_16(x_hi, y_hi), 1);
- temp = (temp + (WEBRTC_SPL_MUL_16_16_RSFT(x_hi, y_low, 14)));
-
- z[i] = (temp + (WEBRTC_SPL_MUL_16_16_RSFT(x_low, y_hi, 14)));
- }
-
- WebRtcSpl_VectorBitShiftW32(z, N, z, left_shifts);
-
- return;
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/window32_w32.h b/trunk/src/modules/audio_coding/codecs/ilbc/window32_w32.h
deleted file mode 100644
index 121188a..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/window32_w32.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_Window32W32.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_WINDOW32_W32_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * window multiplication
- *---------------------------------------------------------------*/
-
-void WebRtcIlbcfix_Window32W32(
- WebRtc_Word32 *z, /* Output */
- WebRtc_Word32 *x, /* Input (same domain as Output)*/
- const WebRtc_Word32 *y, /* Q31 Window */
- WebRtc_Word16 N /* length to process */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/xcorr_coef.c b/trunk/src/modules/audio_coding/codecs/ilbc/xcorr_coef.c
deleted file mode 100644
index 04170ad..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/xcorr_coef.c
+++ /dev/null
@@ -1,142 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_XcorrCoef.c
-
-******************************************************************/
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * cross correlation which finds the optimal lag for the
- * crossCorr*crossCorr/(energy) criteria
- *---------------------------------------------------------------*/
-
-int WebRtcIlbcfix_XcorrCoef(
- WebRtc_Word16 *target, /* (i) first array */
- WebRtc_Word16 *regressor, /* (i) second array */
- WebRtc_Word16 subl, /* (i) dimension arrays */
- WebRtc_Word16 searchLen, /* (i) the search lenght */
- WebRtc_Word16 offset, /* (i) samples offset between arrays */
- WebRtc_Word16 step /* (i) +1 or -1 */
- ){
- int k;
- WebRtc_Word16 maxlag;
- WebRtc_Word16 pos;
- WebRtc_Word16 max;
- WebRtc_Word16 crossCorrScale, Energyscale;
- WebRtc_Word16 crossCorrSqMod, crossCorrSqMod_Max;
- WebRtc_Word32 crossCorr, Energy;
- WebRtc_Word16 crossCorrmod, EnergyMod, EnergyMod_Max;
- WebRtc_Word16 *tp, *rp;
- WebRtc_Word16 *rp_beg, *rp_end;
- WebRtc_Word16 totscale, totscale_max;
- WebRtc_Word16 scalediff;
- WebRtc_Word32 newCrit, maxCrit;
- int shifts;
-
- /* Initializations, to make sure that the first one is selected */
- crossCorrSqMod_Max=0;
- EnergyMod_Max=WEBRTC_SPL_WORD16_MAX;
- totscale_max=-500;
- maxlag=0;
- pos=0;
-
- /* Find scale value and start position */
- if (step==1) {
- max=WebRtcSpl_MaxAbsValueW16(regressor, (WebRtc_Word16)(subl+searchLen-1));
- rp_beg = regressor;
- rp_end = ®ressor[subl];
- } else { /* step==-1 */
- max=WebRtcSpl_MaxAbsValueW16(®ressor[-searchLen], (WebRtc_Word16)(subl+searchLen-1));
- rp_beg = ®ressor[-1];
- rp_end = ®ressor[subl-1];
- }
-
- /* Introduce a scale factor on the Energy in WebRtc_Word32 in
- order to make sure that the calculation does not
- overflow */
-
- if (max>5000) {
- shifts=2;
- } else {
- shifts=0;
- }
-
- /* Calculate the first energy, then do a +/- to get the other energies */
- Energy=WebRtcSpl_DotProductWithScale(regressor, regressor, subl, shifts);
-
- for (k=0;k<searchLen;k++) {
- tp = target;
- rp = ®ressor[pos];
-
- crossCorr=WebRtcSpl_DotProductWithScale(tp, rp, subl, shifts);
-
- if ((Energy>0)&&(crossCorr>0)) {
-
- /* Put cross correlation and energy on 16 bit word */
- crossCorrScale=(WebRtc_Word16)WebRtcSpl_NormW32(crossCorr)-16;
- crossCorrmod=(WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(crossCorr, crossCorrScale);
- Energyscale=(WebRtc_Word16)WebRtcSpl_NormW32(Energy)-16;
- EnergyMod=(WebRtc_Word16)WEBRTC_SPL_SHIFT_W32(Energy, Energyscale);
-
- /* Square cross correlation and store upper WebRtc_Word16 */
- crossCorrSqMod=(WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(crossCorrmod, crossCorrmod, 16);
-
- /* Calculate the total number of (dynamic) right shifts that have
- been performed on (crossCorr*crossCorr)/energy
- */
- totscale=Energyscale-(crossCorrScale<<1);
-
- /* Calculate the shift difference in order to be able to compare the two
- (crossCorr*crossCorr)/energy in the same domain
- */
- scalediff=totscale-totscale_max;
- scalediff=WEBRTC_SPL_MIN(scalediff,31);
- scalediff=WEBRTC_SPL_MAX(scalediff,-31);
-
- /* Compute the cross multiplication between the old best criteria
- and the new one to be able to compare them without using a
- division */
-
- if (scalediff<0) {
- newCrit = ((WebRtc_Word32)crossCorrSqMod*EnergyMod_Max)>>(-scalediff);
- maxCrit = ((WebRtc_Word32)crossCorrSqMod_Max*EnergyMod);
- } else {
- newCrit = ((WebRtc_Word32)crossCorrSqMod*EnergyMod_Max);
- maxCrit = ((WebRtc_Word32)crossCorrSqMod_Max*EnergyMod)>>scalediff;
- }
-
- /* Store the new lag value if the new criteria is larger
- than previous largest criteria */
-
- if (newCrit > maxCrit) {
- crossCorrSqMod_Max = crossCorrSqMod;
- EnergyMod_Max = EnergyMod;
- totscale_max = totscale;
- maxlag = k;
- }
- }
- pos+=step;
-
- /* Do a +/- to get the next energy */
- Energy += step*(WEBRTC_SPL_RSHIFT_W32(
- ((WebRtc_Word32)(*rp_end)*(*rp_end)) - ((WebRtc_Word32)(*rp_beg)*(*rp_beg)),
- shifts));
- rp_beg+=step;
- rp_end+=step;
- }
-
- return(maxlag+offset);
-}
diff --git a/trunk/src/modules/audio_coding/codecs/ilbc/xcorr_coef.h b/trunk/src/modules/audio_coding/codecs/ilbc/xcorr_coef.h
deleted file mode 100644
index ac885c4..0000000
--- a/trunk/src/modules/audio_coding/codecs/ilbc/xcorr_coef.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/******************************************************************
-
- iLBC Speech Coder ANSI-C Source Code
-
- WebRtcIlbcfix_XcorrCoef.h
-
-******************************************************************/
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_MAIN_SOURCE_XCORR_COEF_H_
-
-#include "defines.h"
-
-/*----------------------------------------------------------------*
- * cross correlation which finds the optimal lag for the
- * crossCorr*crossCorr/(energy) criteria
- *---------------------------------------------------------------*/
-
-int WebRtcIlbcfix_XcorrCoef(
- WebRtc_Word16 *target, /* (i) first array */
- WebRtc_Word16 *regressor, /* (i) second array */
- WebRtc_Word16 subl, /* (i) dimension arrays */
- WebRtc_Word16 searchLen, /* (i) the search lenght */
- WebRtc_Word16 offset, /* (i) samples offset between arrays */
- WebRtc_Word16 step /* (i) +1 or -1 */
- );
-
-#endif
diff --git a/trunk/src/modules/audio_coding/codecs/pcm16b/Android.mk b/trunk/src/modules/audio_coding/codecs/pcm16b/Android.mk
deleted file mode 100644
index 2e88fa6..0000000
--- a/trunk/src/modules/audio_coding/codecs/pcm16b/Android.mk
+++ /dev/null
@@ -1,37 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_pcm16b
-LOCAL_MODULE_TAGS := optional
-LOCAL_SRC_FILES := pcm16b.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include \
- $(LOCAL_PATH)/../../../..
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_coding/codecs/pcm16b/include/pcm16b.h b/trunk/src/modules/audio_coding/codecs/pcm16b/include/pcm16b.h
deleted file mode 100644
index e3cac4d..0000000
--- a/trunk/src/modules/audio_coding/codecs/pcm16b/include/pcm16b.h
+++ /dev/null
@@ -1,106 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_MAIN_INTERFACE_PCM16B_H_
-#define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_MAIN_INTERFACE_PCM16B_H_
-/*
- * Define the fixpoint numeric formats
- */
-
-#include "typedefs.h"
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/****************************************************************************
- * WebRtcPcm16b_EncodeW16(...)
- *
- * "Encode" a sample vector to 16 bit linear (Encoded standard is big endian)
- *
- * Input:
- * - speechIn16b : Input speech vector
- * - len : Number of samples in speech vector
- *
- * Output:
- * - speechOut16b : Encoded data vector (big endian 16 bit)
- *
- * Returned value : Size in bytes of speechOut16b
- */
-
-WebRtc_Word16 WebRtcPcm16b_EncodeW16(WebRtc_Word16 *speechIn16b,
- WebRtc_Word16 len,
- WebRtc_Word16 *speechOut16b);
-
-/****************************************************************************
- * WebRtcPcm16b_Encode(...)
- *
- * "Encode" a sample vector to 16 bit linear (Encoded standard is big endian)
- *
- * Input:
- * - speech16b : Input speech vector
- * - len : Number of samples in speech vector
- *
- * Output:
- * - speech8b : Encoded data vector (big endian 16 bit)
- *
- * Returned value : Size in bytes of speech8b
- */
-
-WebRtc_Word16 WebRtcPcm16b_Encode(WebRtc_Word16 *speech16b,
- WebRtc_Word16 len,
- unsigned char *speech8b);
-
-/****************************************************************************
- * WebRtcPcm16b_DecodeW16(...)
- *
- * "Decode" a vector to 16 bit linear (Encoded standard is big endian)
- *
- * Input:
- * - speechIn16b : Encoded data vector (big endian 16 bit)
- * - len : Number of bytes in speechIn16b
- *
- * Output:
- * - speechOut16b : Decoded speech vector
- *
- * Returned value : Samples in speechOut16b
- */
-
-WebRtc_Word16 WebRtcPcm16b_DecodeW16(void *inst,
- WebRtc_Word16 *speechIn16b,
- WebRtc_Word16 len,
- WebRtc_Word16 *speechOut16b,
- WebRtc_Word16* speechType);
-
-/****************************************************************************
- * WebRtcPcm16b_Decode(...)
- *
- * "Decode" a vector to 16 bit linear (Encoded standard is big endian)
- *
- * Input:
- * - speech8b : Encoded data vector (big endian 16 bit)
- * - len : Number of bytes in speech8b
- *
- * Output:
- * - speech16b : Decoded speech vector
- *
- * Returned value : Samples in speech16b
- */
-
-
-WebRtc_Word16 WebRtcPcm16b_Decode(unsigned char *speech8b,
- WebRtc_Word16 len,
- WebRtc_Word16 *speech16b);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif /* PCM16B */
diff --git a/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.c b/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.c
deleted file mode 100644
index 0cff5dd..0000000
--- a/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.c
+++ /dev/null
@@ -1,105 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-#include "pcm16b.h"
-
-#include <stdlib.h>
-
-#include "typedefs.h"
-
-#ifdef WEBRTC_BIG_ENDIAN
-#include "signal_processing_library.h"
-#endif
-
-#define HIGHEND 0xFF00
-#define LOWEND 0xFF
-
-
-
-/* Encoder with WebRtc_Word16 Output */
-WebRtc_Word16 WebRtcPcm16b_EncodeW16(WebRtc_Word16 *speechIn16b,
- WebRtc_Word16 len,
- WebRtc_Word16 *speechOut16b)
-{
-#ifdef WEBRTC_BIG_ENDIAN
- WEBRTC_SPL_MEMCPY_W16(speechOut16b, speechIn16b, len);
-#else
- int i;
- for (i=0;i<len;i++) {
- speechOut16b[i]=(((WebRtc_UWord16)speechIn16b[i])>>8)|((((WebRtc_UWord16)speechIn16b[i])<<8)&0xFF00);
- }
-#endif
- return(len<<1);
-}
-
-
-/* Encoder with char Output (old version) */
-WebRtc_Word16 WebRtcPcm16b_Encode(WebRtc_Word16 *speech16b,
- WebRtc_Word16 len,
- unsigned char *speech8b)
-{
- WebRtc_Word16 samples=len*2;
- WebRtc_Word16 pos;
- WebRtc_Word16 short1;
- WebRtc_Word16 short2;
- for (pos=0;pos<len;pos++) {
- short1=HIGHEND & speech16b[pos];
- short2=LOWEND & speech16b[pos];
- short1=short1>>8;
- speech8b[pos*2]=(unsigned char) short1;
- speech8b[pos*2+1]=(unsigned char) short2;
- }
- return(samples);
-}
-
-
-/* Decoder with WebRtc_Word16 Input instead of char when the WebRtc_Word16 Encoder is used */
-WebRtc_Word16 WebRtcPcm16b_DecodeW16(void *inst,
- WebRtc_Word16 *speechIn16b,
- WebRtc_Word16 len,
- WebRtc_Word16 *speechOut16b,
- WebRtc_Word16* speechType)
-{
-#ifdef WEBRTC_BIG_ENDIAN
- WEBRTC_SPL_MEMCPY_W8(speechOut16b, speechIn16b, ((len*sizeof(WebRtc_Word16)+1)>>1));
-#else
- int i;
- int samples=len>>1;
-
- for (i=0;i<samples;i++) {
- speechOut16b[i]=(((WebRtc_UWord16)speechIn16b[i])>>8)|(((WebRtc_UWord16)(speechIn16b[i]&0xFF))<<8);
- }
-#endif
-
- *speechType=1;
-
- // Avoid warning.
- (void)(inst = NULL);
-
- return(len>>1);
-}
-
-/* "old" version of the decoder that uses char as input (not used in NetEq any more) */
-WebRtc_Word16 WebRtcPcm16b_Decode(unsigned char *speech8b,
- WebRtc_Word16 len,
- WebRtc_Word16 *speech16b)
-{
- WebRtc_Word16 samples=len>>1;
- WebRtc_Word16 pos;
- WebRtc_Word16 shortval;
- for (pos=0;pos<samples;pos++) {
- shortval=((unsigned short) speech8b[pos*2]);
- shortval=(shortval<<8)&HIGHEND;
- shortval=shortval|(((unsigned short) speech8b[pos*2+1])&LOWEND);
- speech16b[pos]=shortval;
- }
- return(samples);
-}
diff --git a/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.gypi b/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.gypi
deleted file mode 100644
index 38754e6..0000000
--- a/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b.gypi
+++ /dev/null
@@ -1,52 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'PCM16B',
- 'type': '<(library)',
- 'include_dirs': [
- 'include',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- ],
- },
- 'sources': [
- 'include/pcm16b.h',
- 'pcm16b.c',
- ],
- },
- ], # targets
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'pcm16b_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'PCM16B',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'sources': [
- 'pcm16b_unittest.cc',
- ],
- }, # PCM16B_unittests
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b_unittest.cc b/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b_unittest.cc
deleted file mode 100644
index eb910b3..0000000
--- a/trunk/src/modules/audio_coding/codecs/pcm16b/pcm16b_unittest.cc
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Empty test just to get code coverage metrics for this dir.
- */
-#include "pcm16b.h"
-#include "gtest/gtest.h"
-
-TEST(Pcm16bTest, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/modules/audio_coding/main/OWNERS b/trunk/src/modules/audio_coding/main/OWNERS
deleted file mode 100644
index e1e6256..0000000
--- a/trunk/src/modules/audio_coding/main/OWNERS
+++ /dev/null
@@ -1,3 +0,0 @@
-tina.legrand@webrtc.org
-turaj@webrtc.org
-jan.skoglund@webrtc.org
diff --git a/trunk/src/modules/audio_coding/main/interface/audio_coding_module.h b/trunk/src/modules/audio_coding/main/interface/audio_coding_module.h
deleted file mode 100644
index ed13689..0000000
--- a/trunk/src/modules/audio_coding/main/interface/audio_coding_module.h
+++ /dev/null
@@ -1,888 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H
-
-#include "audio_coding_module_typedefs.h"
-#include "module.h"
-#include "module_common_types.h"
-
-namespace webrtc {
-
-// forward declarations
-struct CodecInst;
-
-#define WEBRTC_10MS_PCM_AUDIO 960 // 16 bits super wideband 48 Khz
-
-// Callback class used for sending data ready to be packetized
-class AudioPacketizationCallback {
- public:
- virtual ~AudioPacketizationCallback() {}
-
- virtual WebRtc_Word32 SendData(
- FrameType frameType, WebRtc_UWord8 payloadType, WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData, WebRtc_UWord16 payloadSize,
- const RTPFragmentationHeader* fragmentation) = 0;
-};
-
-// Callback class used for inband Dtmf detection
-class AudioCodingFeedback {
- public:
- virtual ~AudioCodingFeedback() {}
-
- virtual WebRtc_Word32 IncomingDtmf(const WebRtc_UWord8 digitDtmf,
- const bool end) = 0;
-};
-
-// Callback class used for reporting VAD decision
-class ACMVADCallback {
- public:
- virtual ~ACMVADCallback() {}
-
- virtual WebRtc_Word32 InFrameType(WebRtc_Word16 frameType) = 0;
-};
-
-// Callback class used for reporting receiver statistics
-class ACMVQMonCallback {
- public:
- virtual ~ACMVQMonCallback() {}
-
- virtual WebRtc_Word32 NetEqStatistics(
- const WebRtc_Word32 id, // current ACM id
- const WebRtc_UWord16 MIUsValid, // valid voice duration in ms
- const WebRtc_UWord16 MIUsReplaced, // concealed voice duration in ms
- const WebRtc_UWord8 eventFlags, // concealed voice flags
- const WebRtc_UWord16 delayMS) = 0; // average delay in ms
-};
-
-class AudioCodingModule: public Module {
- protected:
- AudioCodingModule() {}
- virtual ~AudioCodingModule() {}
-
- public:
- ///////////////////////////////////////////////////////////////////////////
- // Creation and destruction of a ACM
- //
- static AudioCodingModule* Create(const WebRtc_Word32 id);
-
- static void Destroy(AudioCodingModule* module);
-
- ///////////////////////////////////////////////////////////////////////////
- // Utility functions
- //
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_UWord8 NumberOfCodecs()
- // Returns number of supported codecs.
- //
- // Return value:
- // number of supported codecs.
- ///
- static WebRtc_UWord8 NumberOfCodecs();
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 Codec()
- // Get supported codec with list number.
- //
- // Input:
- // -listId : list number.
- //
- // Output:
- // -codec : a structure where the parameters of the codec,
- // given by list number is written to.
- //
- // Return value:
- // -1 if the list number (listId) is invalid.
- // 0 if succeeded.
- //
- static WebRtc_Word32 Codec(const WebRtc_UWord8 listId, CodecInst& codec);
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 Codec()
- // Get supported codec with the given codec name and sampling frequency.
- // If the sampling frequency is -1 then the search will be only based on
- // codec name.
- //
- // Input:
- // -payloadName : name of the codec.
- // -samplingFreqHz : samling frequency of the codec.
- //
- // Output:
- // -codec : a structure where the parameters of the codec,
- // given by name is written to.
- //
- // Return value:
- // -1 if the list number (listId) is invalid.
- // 0 if succeeded.
- //
- static WebRtc_Word32 Codec(const char* payloadName, CodecInst& codec,
- const WebRtc_Word32 samplingFreqHz = -1);
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 Codec()
- //
- // Returns the list number of the given codec name and sampling frequency.
- // If the sampling frequency is -1 then the search will be only based on
- // codec name.
- //
- // Input:
- // -payloadName : name of the codec.
- // -samplingFreqHz : samling frequency of the codec.
- //
- // Return value:
- // if the codec is found, the index of the codec in the list,
- // -1 if the codec is not found.
- //
- static WebRtc_Word32 Codec(const char* payloadName,
- const WebRtc_Word32 samplingFreqHz = -1);
-
- ///////////////////////////////////////////////////////////////////////////
- // bool IsCodecValid()
- // Checks the validity of the parameters of the given codec.
- //
- // Input:
- // -codec : the structur which keeps the parameters of the
- // codec.
- //
- // Reurn value:
- // true if the parameters are valid,
- // false if any parameter is not valid.
- //
- static bool IsCodecValid(const CodecInst& codec);
-
- ///////////////////////////////////////////////////////////////////////////
- // Sender
- //
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 InitializeSender()
- // Any encoder-related state of ACM will be initialized to the
- // same state when ACM is created. This will not interrupt or
- // effect decoding functionality of ACM. ACM will lose all the
- // encoding-related settings by calling this function.
- // For instance, a send codec has to be registered again.
- //
- // Return value:
- // -1 if failed to initialize,
- // 0 if succeeded.
- //
- virtual WebRtc_Word32 InitializeSender() = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 ResetEncoder()
- // This API resets the states of encoder. All the encoder settings, such as
- // send-codec or VAD/DTX, will be preserved.
- //
- // Return value:
- // -1 if failed to initialize,
- // 0 if succeeded.
- //
- virtual WebRtc_Word32 ResetEncoder() = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 RegisterSendCodec()
- // Registers a codec, specified by "sendCodec," as sending codec.
- // This API can be called multiple of times to register Codec. The last codec
- // registered overwrites the previous ones.
- // The API can also be used to change payload type for CNG and RED, which are
- // registered by default to default payload types.
- // Note that registering CNG and RED won't overwrite speech codecs.
- // This API can be called to set/change the send payload-type, frame-size
- // or encoding rate (if applicable for the codec).
- //
- // Input:
- // -sendCodec : Parameters of the codec to be registered, c.f.
- // common_types.h for the definition of
- // CodecInst.
- //
- // Return value:
- // -1 if failed to initialize,
- // 0 if succeeded.
- //
- virtual WebRtc_Word32 RegisterSendCodec(const CodecInst& sendCodec) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 SendCodec()
- // Get parameters for the codec currently registered as send codec.
- //
- // Output:
- // -currentSendCodec : parameters of the send codec.
- //
- // Return value:
- // -1 if failed to get send codec,
- // 0 if succeeded.
- //
- virtual WebRtc_Word32 SendCodec(CodecInst& currentSendCodec) const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 SendFrequency()
- // Get the sampling frequency of the current encoder in Hertz.
- //
- // Return value:
- // positive; sampling frequency [Hz] of the current encoder.
- // -1 if an error has happened.
- //
- virtual WebRtc_Word32 SendFrequency() const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 Bitrate()
- // Get encoding bit-rate in bits per second.
- //
- // Return value:
- // positive; encoding rate in bits/sec,
- // -1 if an error is happened.
- //
- virtual WebRtc_Word32 SendBitrate() const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 SetReceivedEstimatedBandwidth()
- // Set available bandwidth [bits/sec] of the up-link channel.
- // This information is used for traffic shaping, and is currently only
- // supported if iSAC is the send codec.
- //
- // Input:
- // -bw : bandwidth in bits/sec estimated for
- // up-link.
- // Return value
- // -1 if error occurred in setting the bandwidth,
- // 0 bandwidth is set successfully.
- //
- virtual WebRtc_Word32 SetReceivedEstimatedBandwidth(
- const WebRtc_Word32 bw) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 RegisterTransportCallback()
- // Register a transport callback which will be called to deliver
- // the encoded buffers whenever Process() is called and a
- // bit-stream is ready.
- //
- // Input:
- // -transport : pointer to the callback class
- // transport->SendData() is called whenever
- // Process() is called and bit-stream is ready
- // to deliver.
- //
- // Return value:
- // -1 if the transport callback could not be registered
- // 0 if registration is successful.
- //
- virtual WebRtc_Word32 RegisterTransportCallback(
- AudioPacketizationCallback* transport) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 Add10MsData()
- // Add 10MS of raw (PCM) audio data to the encoder. If the sampling
- // frequency of the audio does not match the sampling frequency of the
- // current encoder ACM will resample the audio.
- //
- // Input:
- // -audioFrame : the input audio frame, containing raw audio
- // sampling frequency etc.,
- // c.f. module_common_types.h for definition of
- // AudioFrame.
- //
- // Return value:
- // 0 successfully added the frame.
- // -1 some error occurred and data is not added.
- // < -1 to add the frame to the buffer n samples had to be
- // overwritten, -n is the return value in this case.
- //
- virtual WebRtc_Word32 Add10MsData(const AudioFrame& audioFrame) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // (FEC) Forward Error Correction
- //
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 SetFECStatus(const bool enable)
- // configure FEC status i.e. on/off.
- //
- // RFC 2198 describes a solution which has a single payload type which
- // signifies a packet with redundancy. That packet then becomes a container,
- // encapsulating multiple payloads into a single RTP packet.
- // Such a scheme is flexible, since any amount of redundancy may be
- // encapsulated within a single packet. There is, however, a small overhead
- // since each encapsulated payload must be preceded by a header indicating
- // the type of data enclosed.
- //
- // This means that FEC is actually a RED scheme.
- //
- // Input:
- // -enableFEC : if true FEC is enabled, otherwise FEC is
- // disabled.
- //
- // Return value:
- // -1 if failed to set FEC status,
- // 0 if succeeded.
- //
- virtual WebRtc_Word32 SetFECStatus(const bool enableFEC) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // bool FECStatus()
- // Get FEC status
- //
- // Return value
- // true if FEC is enabled,
- // false if FEC is disabled.
- //
- virtual bool FECStatus() const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // (VAD) Voice Activity Detection
- //
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 SetVAD()
- // If DTX is enabled & the codec does not have internal DTX/VAD
- // WebRtc VAD will be automatically enabled and 'enableVAD' is ignored.
- //
- // If DTX is disabled but VAD is enabled no DTX packets are send,
- // regardless of whether the codec has internal DTX/VAD or not. In this
- // case, WebRtc VAD is running to label frames as active/in-active.
- //
- // Inputs:
- // -enableDTX : if true DTX is enabled,
- // otherwise DTX is disabled.
- // -enableVAD : if true VAD is enabled,
- // otherwise VAD is disabled.
- // -vadMode : determines the aggressiveness of VAD. A more
- // aggressive mode results in more frames labeled
- // as in-active, c.f. definition of
- // ACMVADMode in audio_coding_module_typedefs.h
- // for valid values.
- //
- // Return value:
- // -1 if failed to set up VAD/DTX,
- // 0 if succeeded.
- //
- virtual WebRtc_Word32 SetVAD(const bool enableDTX = true,
- const bool enableVAD = false,
- const ACMVADMode vadMode = VADNormal) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 VAD()
- // Get VAD status.
- //
- // Outputs:
- // -dtxEnabled : is set to true if DTX is enabled, otherwise
- // is set to false.
- // -vadEnabled : is set to true if VAD is enabled, otherwise
- // is set to false.
- // -vadMode : is set to the current aggressiveness of VAD.
- //
- // Return value:
- // -1 if fails to retrieve the setting of DTX/VAD,
- // 0 if succeeeded.
- //
- virtual WebRtc_Word32 VAD(bool& dtxEnabled, bool& vadEnabled,
- ACMVADMode& vadMode) const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 ReplaceInternalDTXWithWebRtc()
- // Used to replace codec internal DTX scheme with WebRtc. This is only
- // supported for G729, where this call replaces AnnexB with WebRtc DTX.
- //
- // Input:
- // -useWebRtcDTX : if false (default) the codec built-in DTX/VAD
- // scheme is used, otherwise the internal DTX is
- // replaced with WebRtc DTX/VAD.
- //
- // Return value:
- // -1 if failed to replace codec internal DTX with WebRtc,
- // 0 if succeeded.
- //
- virtual WebRtc_Word32 ReplaceInternalDTXWithWebRtc(
- const bool useWebRtcDTX = false) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 IsInternalDTXReplacedWithWebRtc()
- // Get status if the codec internal DTX (when such exists) is replaced with
- // WebRtc DTX. This is only supported for G729.
- //
- // Output:
- // -usesWebRtcDTX : is set to true if the codec internal DTX is
- // replaced with WebRtc DTX/VAD, otherwise it is set
- // to false.
- //
- // Return value:
- // -1 if failed to determine if codec internal DTX is replaced with WebRtc,
- // 0 if succeeded.
- //
- virtual WebRtc_Word32 IsInternalDTXReplacedWithWebRtc(
- bool& usesWebRtcDTX) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 RegisterVADCallback()
- // Call this method to register a callback function which is called
- // any time that ACM encounters an empty frame. That is a frame which is
- // recognized inactive. Depending on the codec WebRtc VAD or internal codec
- // VAD is employed to identify a frame as active/inactive.
- //
- // Input:
- // -vadCallback : pointer to a callback function.
- //
- // Return value:
- // -1 if failed to register the callback function.
- // 0 if the callback function is registered successfully.
- //
- virtual WebRtc_Word32 RegisterVADCallback(ACMVADCallback* vadCallback) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // Receiver
- //
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 InitializeReceiver()
- // Any decoder-related state of ACM will be initialized to the
- // same state when ACM is created. This will not interrupt or
- // effect encoding functionality of ACM. ACM would lose all the
- // decoding-related settings by calling this function.
- // For instance, all registered codecs are deleted and have to be
- // registered again.
- //
- // Return value:
- // -1 if failed to initialize,
- // 0 if succeeded.
- //
- virtual WebRtc_Word32 InitializeReceiver() = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 ResetDecoder()
- // This API resets the states of decoders. ACM will not lose any
- // decoder-related settings, such as registered codecs.
- //
- // Return value:
- // -1 if failed to initialize,
- // 0 if succeeded.
- //
- virtual WebRtc_Word32 ResetDecoder() = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 ReceiveFrequency()
- // Get sampling frequency of the last received payload.
- //
- // Return value:
- // non-negative the sampling frequency in Hertz.
- // -1 if an error has occurred.
- //
- virtual WebRtc_Word32 ReceiveFrequency() const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 PlayoutFrequency()
- // Get sampling frequency of audio played out.
- //
- // Return value:
- // the sampling frequency in Hertz.
- //
- virtual WebRtc_Word32 PlayoutFrequency() const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 RegisterReceiveCodec()
- // Register possible decoders, can be called multiple times for
- // codecs, CNG-NB, CNG-WB, CNG-SWB, AVT and RED.
- //
- // Input:
- // -receiveCodec : parameters of the codec to be registered, c.f.
- // common_types.h for the definition of
- // CodecInst.
- //
- // Return value:
- // -1 if failed to register the codec
- // 0 if the codec registered successfully.
- //
- virtual WebRtc_Word32 RegisterReceiveCodec(const CodecInst& receiveCodec) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 UnregisterReceiveCodec()
- // Unregister the codec currently registered with a specific payload type
- // from the list of possible receive codecs.
- //
- // Input:
- // -payloadType : The number representing the payload type to
- // unregister.
- //
- // Output:
- // -1 if the unregistration fails.
- // 0 if the given codec is successfully unregistered.
- //
- virtual WebRtc_Word32 UnregisterReceiveCodec(
- const WebRtc_Word16 receiveCodec) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 ReceiveCodec()
- // Get the codec associated with last received payload.
- //
- // Output:
- // -currRcvCodec : parameters of the codec associated with the last
- // received payload, c.f. common_types.h for
- // the definition of CodecInst.
- //
- // Return value:
- // -1 if failed to retrieve the codec,
- // 0 if the codec is successfully retrieved.
- //
- virtual WebRtc_Word32 ReceiveCodec(CodecInst& currRcvCodec) const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 IncomingPacket()
- // Call this function to insert a parsed RTP packet into ACM.
- //
- // Inputs:
- // -incomingPayload : received payload.
- // -payloadLengthByte : the length of payload in bytes.
- // -rtpInfo : the relevant information retrieved from RTP
- // header.
- //
- // Return value:
- // -1 if failed to push in the payload
- // 0 if payload is successfully pushed in.
- //
- virtual WebRtc_Word32 IncomingPacket(const WebRtc_Word8* incomingPayload,
- const WebRtc_Word32 payloadLengthByte,
- const WebRtcRTPHeader& rtpInfo) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 IncomingPayload()
- // Call this API to push incoming payloads when there is no rtp-info.
- // The rtp-info will be created in ACM. One usage for this API is when
- // pre-encoded files are pushed in ACM
- //
- // Inputs:
- // -incomingPayload : received payload.
- // -payloadLenghtByte : the length, in bytes, of the received payload.
- // -payloadType : the payload-type. This specifies which codec has
- // to be used to decode the payload.
- // -timestamp : send timestamp of the payload. ACM starts with
- // a random value and increment it by the
- // packet-size, which is given when the codec in
- // question is registered by RegisterReceiveCodec().
- // Therefore, it is essential to have the timestamp
- // if the frame-size differ from the registered
- // value or if the incoming payload contains DTX
- // packets.
- //
- // Return value:
- // -1 if failed to push in the payload
- // 0 if payload is successfully pushed in.
- //
- virtual WebRtc_Word32 IncomingPayload(const WebRtc_Word8* incomingPayload,
- const WebRtc_Word32 payloadLengthByte,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timestamp = 0) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 SetMinimumPlayoutDelay()
- // Set Minimum playout delay, used for lip-sync.
- //
- // Input:
- // -timeMs : minimum delay in milliseconds.
- //
- // Return value:
- // -1 if failed to set the delay,
- // 0 if the minimum delay is set.
- //
- virtual WebRtc_Word32 SetMinimumPlayoutDelay(const WebRtc_Word32 timeMs) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 RegisterIncomingMessagesCallback()
- // Used by the module to deliver messages to the codec module/application
- // when a Dtmf tone is detected, as well as when it stopped.
- //
- // Inputs:
- // -inMsgCallback : pointer to callback function which will be called
- // if Dtmf is detected.
- // -cpt : enables CPT (Call Progress Tone) detection for the
- // specified country. c.f. definition of ACMCountries
- // in audio_coding_module_typedefs.h for valid
- // entries. The default value disables CPT
- // detection.
- //
- // Return value:
- // -1 if the message callback could not be registered
- // 0 if registration is successful.
- //
- virtual WebRtc_Word32
- RegisterIncomingMessagesCallback(
- AudioCodingFeedback* inMsgCallback,
- const ACMCountries cpt = ACMDisableCountryDetection) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 SetDtmfPlayoutStatus()
- // Configure Dtmf playout, i.e. whether out-of-band
- // Dtmf tones are played or not.
- //
- // Input:
- // -enable : if true to enable playout out-of-band Dtmf tones,
- // false to disable.
- //
- // Return value:
- // -1 if the method fails, e.g. Dtmf playout is not supported.
- // 0 if the status is set successfully.
- //
- virtual WebRtc_Word32 SetDtmfPlayoutStatus(const bool enable) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // bool DtmfPlayoutStatus()
- // Get Dtmf playout status.
- //
- // Return value:
- // true if out-of-band Dtmf tones are played,
- // false if playout of Dtmf tones is disabled.
- //
- virtual bool DtmfPlayoutStatus() const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 SetBackgroundNoiseMode()
- // Sets the mode of the background noise playout in an event of long
- // packetloss burst. For the valid modes see the declaration of
- // ACMBackgroundNoiseMode in audio_coding_module_typedefs.h.
- //
- // Input:
- // -mode : the mode for the background noise playout.
- //
- // Return value:
- // -1 if failed to set the mode.
- // 0 if succeeded in setting the mode.
- //
- virtual WebRtc_Word32 SetBackgroundNoiseMode(
- const ACMBackgroundNoiseMode mode) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 BackgroundNoiseMode()
- // Call this method to get the mode of the background noise playout.
- // Playout of background noise is a result of a long packetloss burst.
- // See ACMBackgroundNoiseMode in audio_coding_module_typedefs.h for
- // possible modes.
- //
- // Output:
- // -mode : a reference to ACMBackgroundNoiseMode enumerator.
- //
- // Return value:
- // 0 if the output is a valid mode.
- // -1 if ACM failed to output a valid mode.
- //
- virtual WebRtc_Word32 BackgroundNoiseMode(ACMBackgroundNoiseMode& mode) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 PlayoutTimestamp()
- // The send timestamp of an RTP packet is associated with the decoded
- // audio of the packet in question. This function returns the timestamp of
- // the latest audio obtained by calling PlayoutData10ms().
- //
- // Input:
- // -timestamp : a reference to a WebRtc_UWord32 to receive the
- // timestamp.
- // Return value:
- // 0 if the output is a correct timestamp.
- // -1 if failed to output the correct timestamp.
- //
- //
- virtual WebRtc_Word32 PlayoutTimestamp(WebRtc_UWord32& timestamp) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 DecoderEstimatedBandwidth()
- // Get the estimate of the Bandwidth, in bits/second, based on the incoming
- // stream. This API is useful in one-way communication scenarios, where
- // the bandwidth information is sent in an out-of-band fashion.
- // Currently only supported if iSAC is registered as a reciever.
- //
- // Return value:
- // >0 bandwidth in bits/second.
- // -1 if failed to get a bandwidth estimate.
- //
- virtual WebRtc_Word32 DecoderEstimatedBandwidth() const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 SetPlayoutMode()
- // Call this API to set the playout mode. Playout mode could be optimized
- // for i) voice, ii) FAX or iii) streaming. In Voice mode, NetEQ is
- // optimized to deliver highest audio quality while maintaining a minimum
- // delay. In FAX mode, NetEQ is optimized to have few delay changes as
- // possible and maintain a constant delay, perhaps large relative to voice
- // mode, to avoid PLC. In streaming mode, we tolerate a little more delay
- // to acheive better jitter robustness.
- //
- // Input:
- // -mode : playout mode. Possible inputs are:
- // "voice",
- // "fax" and
- // "streaming".
- //
- // Return value:
- // -1 if failed to set the mode,
- // 0 if succeeding.
- //
- virtual WebRtc_Word32 SetPlayoutMode(const AudioPlayoutMode mode) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // AudioPlayoutMode PlayoutMode()
- // Get playout mode, i.e. whether it is speech, FAX or streaming. See
- // audio_coding_module_typedefs.h for definition of AudioPlayoutMode.
- //
- // Return value:
- // voice: is for voice output,
- // fax: a mode that is optimized for receiving FAX signals.
- // In this mode NetEq tries to maintain a constant high
- // delay to avoid PLC if possible.
- // streaming: a mode that is suitable for streaminq. In this mode we
- // accept longer delay to improve jitter robustness.
- //
- virtual AudioPlayoutMode PlayoutMode() const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 PlayoutData10Ms(
- // Get 10 milliseconds of raw audio data for playout, at the given sampling
- // frequency. ACM will perform a resampling if required.
- //
- // Input:
- // -desiredFreqHz : the desired sampling frequency, in Hertz, of the
- // output audio. If set to -1, the function returns the
- // audio at the current sampling frequency.
- //
- // Output:
- // -audioFrame : output audio frame which contains raw audio data
- // and other relevant parameters, c.f.
- // module_common_types.h for the definition of
- // AudioFrame.
- //
- // Return value:
- // -1 if the function fails,
- // 0 if the function succeeds.
- //
- virtual WebRtc_Word32
- PlayoutData10Ms(const WebRtc_Word32 desiredFreqHz,
- AudioFrame &audioFrame) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // (CNG) Comfort Noise Generation
- // Generate comfort noise when receiving DTX packets
- //
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 SetReceiveVADMode()
- // Configure VAD aggressiveness on the incoming stream.
- //
- // Input:
- // -mode : aggressiveness of the VAD on incoming stream.
- // See audio_coding_module_typedefs.h for the
- // definition of ACMVADMode, and possible
- // values for aggressiveness.
- //
- // Return value:
- // -1 if fails to set the mode,
- // 0 if the mode is set successfully.
- //
- virtual WebRtc_Word16 SetReceiveVADMode(const ACMVADMode mode) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // ACMVADMode ReceiveVADMode()
- // Get VAD aggressiveness on the incoming stream.
- //
- // Return value:
- // aggressiveness of VAD, running on the incoming stream. A more
- // aggressive mode means more audio frames will be labeled as in-active.
- // See audio_coding_module_typedefs.h for the definition of
- // ACMVADMode.
- //
- virtual ACMVADMode ReceiveVADMode() const = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // Codec specific
- //
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 SetISACMaxRate()
- // Set the maximum instantaneous rate of iSAC. For a payload of B bits
- // with a frame-size of T sec the instantaneous rate is B/T bist per
- // second. Therefore, (B/T < maxRateBitPerSec) and
- // (B < maxPayloadLenBytes * 8) are always satisfied for iSAC payloads,
- // c.f SetISACMaxPayloadSize().
- //
- // Input:
- // -maxRateBitPerSec : maximum instantaneous bit-rate given in bits/sec.
- //
- // Return value:
- // -1 if failed to set the maximum rate.
- // 0 if the maximum rate is set successfully.
- //
- virtual WebRtc_Word32 SetISACMaxRate(
- const WebRtc_UWord32 maxRateBitPerSec) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 SetISACMaxPayloadSize()
- // Set the maximum payload size of iSAC packets. No iSAC payload,
- // regardless of its frame-size, may exceed the given limit. For
- // an iSAC payload of size B bits and frame-size T sec we have;
- // (B < maxPayloadLenBytes * 8) and (B/T < maxRateBitPerSec), c.f.
- // SetISACMaxRate().
- //
- // Input:
- // -maxPayloadLenBytes : maximum payload size in bytes.
- //
- // Return value:
- // -1 if failed to set the maximm payload-size.
- // 0 if the given linit is seet successfully.
- //
- virtual WebRtc_Word32 SetISACMaxPayloadSize(
- const WebRtc_UWord16 maxPayloadLenBytes) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 ConfigISACBandwidthEstimator()
- // Call this function to configure the bandwidth estimator of ISAC.
- // During the adaptation of bit-rate, iSAC atomatically adjusts the
- // frame-size (either 30 or 60 ms) to save on RTP header. The initial
- // frame-size can be specified by the first argument. The configuration also
- // regards the initial estimate of bandwidths. The estimator starts from
- // this point and converges to the actual bottleneck. This is given by the
- // second parameter. Furthermore, it is also possible to control the
- // adaptation of frame-size. This is specified by the last parameter.
- //
- // Input:
- // -initFrameSizeMsec : initial frame-size in milisecods. For iSAC-wb
- // 30 ms and 60 ms (default) are acceptable values,
- // and for iSAC-swb 30 ms is the only acceptable
- // value. Zero indiates default value.
- // -initRateBitPerSec : initial estimate of the bandwidth. Values
- // between 10000 and 58000 are acceptable.
- // -enforceFrameSize : if true, the frame-size will not be adapted.
- //
- // Return value:
- // -1 if failed to configure the bandwidth estimator,
- // 0 if the configuration was successfully applied.
- //
- virtual WebRtc_Word32 ConfigISACBandwidthEstimator(
- const WebRtc_UWord8 initFrameSizeMsec,
- const WebRtc_UWord16 initRateBitPerSec,
- const bool enforceFrameSize = false) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // statistics
- //
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 NetworkStatistics()
- // Get network statistics.
- //
- // Input:
- // -networkStatistics : a structure that contains network statistics.
- //
- // Return value:
- // -1 if failed to set the network statistics,
- // 0 if statistics are set successfully.
- //
- virtual WebRtc_Word32 NetworkStatistics(
- ACMNetworkStatistics& networkStatistics) const = 0;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_H
diff --git a/trunk/src/modules/audio_coding/main/interface/audio_coding_module_typedefs.h b/trunk/src/modules/audio_coding/main/interface/audio_coding_module_typedefs.h
deleted file mode 100644
index c0e06ef..0000000
--- a/trunk/src/modules/audio_coding/main/interface/audio_coding_module_typedefs.h
+++ /dev/null
@@ -1,197 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_TYPEDEFS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_TYPEDEFS_H_
-
-#include "typedefs.h"
-
-namespace webrtc {
-
-///////////////////////////////////////////////////////////////////////////
-// enum AudioPlayoutMode
-// An enumerator for different playout modes.
-//
-// -voice : This is the standard mode for VoIP calls. The trade-off
-// between low delay and jitter robustness is optimized
-// for high-quality two-way communication.
-// NetEQs packet loss concealment and signal processing
-// capabilities are fully employed.
-// -fax : The fax mode is optimized for decodability of fax signals
-// rather than for perceived audio quality. When this mode
-// is selected, NetEQ will do as few delay changes as possible,
-// trying to maintain a high and constant delay. Meanwhile,
-// the packet loss concealment efforts are reduced.
-//
-// -streaming : In the case of one-way communication such as passive
-// conference participant, a webinar, or a streaming application,
-// this mode can be used to improve the jitter robustness at
-// the cost of increased delay.
-//
-enum AudioPlayoutMode {
- voice = 0,
- fax = 1,
- streaming = 2
-};
-
-///////////////////////////////////////////////////////////////////////////
-// enum ACMSpeechType
-// An enumerator for possible labels of a decoded frame.
-//
-// -normal : a normal speech frame. If VAD is enabled on the
-// incoming stream this label indicate that the
-// frame is active.
-// -PLC : a PLC frame. The corresponding packet was lost
-// and this frame generated by PLC techniques.
-// -CNG : the frame is comfort noise. This happens if VAD
-// is enabled at the sender and we have received
-// SID.
-// -PLCCNG : PLC will fade to comfort noise if the duration
-// of PLC is long. This labels such a case.
-// -VADPassive : the VAD at the receiver recognizes this frame as
-// passive.
-//
-enum ACMSpeechType {
- normal = 0,
- PLC = 1,
- CNG = 2,
- PLCCNG = 3,
- VADPassive = 4
-};
-
-///////////////////////////////////////////////////////////////////////////
-// enum ACMVADMode
-// An enumerator for aggressiveness of VAD
-// -VADNormal : least aggressive mode.
-// -VADLowBitrate : more aggressive than "VADNormal" to save on
-// bit-rate.
-// -VADAggr : an aggressive mode.
-// -VADVeryAggr : the most agressive mode.
-//
-enum ACMVADMode {
- VADNormal = 0,
- VADLowBitrate = 1,
- VADAggr = 2,
- VADVeryAggr = 3
-};
-
-///////////////////////////////////////////////////////////////////////////
-// enum ACMCountries
-// An enumerator for countries, used when enabling CPT for a specific country.
-//
-enum ACMCountries {
- ACMDisableCountryDetection = -1, // disable CPT detection
- ACMUSA = 0,
- ACMJapan,
- ACMCanada,
- ACMFrance,
- ACMGermany,
- ACMAustria,
- ACMBelgium,
- ACMUK,
- ACMCzech,
- ACMDenmark,
- ACMFinland,
- ACMGreece,
- ACMHungary,
- ACMIceland,
- ACMIreland,
- ACMItaly,
- ACMLuxembourg,
- ACMMexico,
- ACMNorway,
- ACMPoland,
- ACMPortugal,
- ACMSpain,
- ACMSweden,
- ACMTurkey,
- ACMChina,
- ACMHongkong,
- ACMTaiwan,
- ACMKorea,
- ACMSingapore,
- ACMNonStandard1
-// non-standard countries
-};
-
-///////////////////////////////////////////////////////////////////////////
-// enum ACMAMRPackingFormat
-// An enumerator for different bit-packing format of AMR codec according to
-// RFC 3267.
-//
-// -AMRUndefined : undefined.
-// -AMRBandwidthEfficient : bandwidth-efficient mode.
-// -AMROctetAlligned : Octet-alligned mode.
-// -AMRFileStorage : file-storage mode.
-//
-enum ACMAMRPackingFormat {
- AMRUndefined = -1,
- AMRBandwidthEfficient = 0,
- AMROctetAlligned = 1,
- AMRFileStorage = 2
-};
-
-
-///////////////////////////////////////////////////////////////////////////
-//
-// Struct containing network statistics
-//
-// -currentBufferSize : current jitter buffer size in ms
-// -preferredBufferSize : preferred (optimal) buffer size in ms
-// -jitterPeaksFound : indicate if peaky-jitter mode is engaged, that is,
-// if severe but sparse network delays have occurred.
-// -currentPacketLossRate : loss rate (network + late) (in Q14)
-// -currentDiscardRate : late loss rate (in Q14)
-// -currentExpandRate : fraction (of original stream) of synthesized
-// speech inserted through expansion (in Q14)
-// -currentPreemptiveRate : fraction of synthesized speech inserted through
-// pre-emptive expansion (in Q14)
-// -currentAccelerateRate : fraction of data removed through acceleration
-// (in Q14)
-// -clockDriftPPM : clock-drift between sender and receiver in parts-
-// per-million. Positive means that receiver sample
-// rate is higher than sender sample rate.
-// -meanWaitingTimeMs : average packet waiting time in the buffer
-// -medianWaitingTimeMs : median packet waiting time in the buffer
-// -minWaitingTimeMs : min packet waiting time in the buffer
-// -maxWaitingTimeMs : max packet waiting time in the buffer
-typedef struct {
- WebRtc_UWord16 currentBufferSize;
- WebRtc_UWord16 preferredBufferSize;
- bool jitterPeaksFound;
- WebRtc_UWord16 currentPacketLossRate;
- WebRtc_UWord16 currentDiscardRate;
- WebRtc_UWord16 currentExpandRate;
- WebRtc_UWord16 currentPreemptiveRate;
- WebRtc_UWord16 currentAccelerateRate;
- int32_t clockDriftPPM;
- int meanWaitingTimeMs;
- int medianWaitingTimeMs;
- int minWaitingTimeMs;
- int maxWaitingTimeMs;
-} ACMNetworkStatistics;
-
-///////////////////////////////////////////////////////////////////////////
-//
-// Enumeration of background noise mode a mapping from NetEQ interface.
-//
-// -On : default "normal" behavior with eternal noise
-// -Fade : noise fades to zero after some time
-// -Off : background noise is always zero
-//
-enum ACMBackgroundNoiseMode {
- On,
- Fade,
- Off
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_INTERFACE_AUDIO_CODING_MODULE_TYPEDEFS_H_
diff --git a/trunk/src/modules/audio_coding/main/source/Android.mk b/trunk/src/modules/audio_coding/main/source/Android.mk
deleted file mode 100644
index 90214a9..0000000
--- a/trunk/src/modules/audio_coding/main/source/Android.mk
+++ /dev/null
@@ -1,67 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_audio_coding
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := \
- acm_cng.cc \
- acm_codec_database.cc \
- acm_dtmf_detection.cc \
- acm_dtmf_playout.cc \
- acm_g722.cc \
- acm_generic_codec.cc \
- acm_ilbc.cc \
- acm_isac.cc \
- acm_neteq.cc \
- acm_pcm16b.cc \
- acm_pcma.cc \
- acm_pcmu.cc \
- acm_red.cc \
- acm_resampler.cc \
- audio_coding_module.cc \
- audio_coding_module_impl.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../codecs/cng/include \
- $(LOCAL_PATH)/../../codecs/g711/include \
- $(LOCAL_PATH)/../../codecs/g722/include \
- $(LOCAL_PATH)/../../codecs/ilbc/interface \
- $(LOCAL_PATH)/../../codecs/iSAC/main/interface \
- $(LOCAL_PATH)/../../codecs/iSAC/fix/interface \
- $(LOCAL_PATH)/../../codecs/pcm16b/include \
- $(LOCAL_PATH)/../../neteq/interface \
- $(LOCAL_PATH)/../../../.. \
- $(LOCAL_PATH)/../../../interface \
- $(LOCAL_PATH)/../../../../common_audio/resampler/include \
- $(LOCAL_PATH)/../../../../common_audio/signal_processing/include \
- $(LOCAL_PATH)/../../../../common_audio/vad/include \
- $(LOCAL_PATH)/../../../../system_wrappers/interface
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_coding/main/source/acm_amr.cc b/trunk/src/modules/audio_coding/main/source/acm_amr.cc
deleted file mode 100644
index 71cf603..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_amr.cc
+++ /dev/null
@@ -1,444 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_amr.h"
-#include "acm_common_defs.h"
-#include "acm_neteq.h"
-#include "audio_coding_module_typedefs.h"
-#include "rw_lock_wrapper.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-
-#ifdef WEBRTC_CODEC_AMR
-// NOTE! GSM AMR is not included in the open-source package. The following
-// interface file is needed:
-//
-// /modules/audio_coding/codecs/amr/main/interface/amr_interface.h
-//
-// The API in the header file should match the one below.
-//
-// int16_t WebRtcAmr_CreateEnc(AMR_encinst_t_** encInst);
-// int16_t WebRtcAmr_CreateDec(AMR_decinst_t_** decInst);
-// int16_t WebRtcAmr_FreeEnc(AMR_encinst_t_* encInst);
-// int16_t WebRtcAmr_FreeDec(AMR_decinst_t_* decInst);
-// int16_t WebRtcAmr_Encode(AMR_encinst_t_* encInst,
-// int16_t* input,
-// int16_t len,
-// int16_t*output,
-// int16_t mode);
-// int16_t WebRtcAmr_EncoderInit(AMR_encinst_t_* encInst,
-// int16_t dtxMode);
-// int16_t WebRtcAmr_EncodeBitmode(AMR_encinst_t_* encInst,
-// int format);
-// int16_t WebRtcAmr_Decode(AMR_decinst_t_* decInst);
-// int16_t WebRtcAmr_DecodePlc(AMR_decinst_t_* decInst);
-// int16_t WebRtcAmr_DecoderInit(AMR_decinst_t_* decInst);
-// int16_t WebRtcAmr_DecodeBitmode(AMR_decinst_t_* decInst,
-// int format);
-// void WebRtcAmr_Version(char *versionStr, short len);
-#include "amr_interface.h"
-#endif
-
-namespace webrtc {
-
-#ifndef WEBRTC_CODEC_AMR
-ACMAMR::ACMAMR(WebRtc_Word16 /* codecID */)
- : _encoderInstPtr(NULL),
- _decoderInstPtr(NULL),
- _encodingMode(-1), // Invalid value.
- _encodingRate(0), // Invalid value.
- _encoderPackingFormat(AMRBandwidthEfficient),
- _decoderPackingFormat(AMRBandwidthEfficient) {
- return;
-}
-
-ACMAMR::~ACMAMR() {
- return;
-}
-
-WebRtc_Word16 ACMAMR::InternalEncode(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */) {
- return -1;
-}
-
-WebRtc_Word16 ACMAMR::DecodeSafe(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */) {
- return -1;
-}
-
-WebRtc_Word16 ACMAMR::EnableDTX() {
- return -1;
-}
-
-WebRtc_Word16 ACMAMR::DisableDTX() {
- return -1;
-}
-
-WebRtc_Word16 ACMAMR::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */) {
- return -1;
-}
-
-WebRtc_Word16 ACMAMR::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */) {
- return -1;
-}
-
-WebRtc_Word32 ACMAMR::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
- const CodecInst& /* codecInst */) {
- return -1;
-}
-
-ACMGenericCodec* ACMAMR::CreateInstance(void) {
- return NULL;
-}
-
-WebRtc_Word16 ACMAMR::InternalCreateEncoder() {
- return -1;
-}
-
-void ACMAMR::DestructEncoderSafe() {
- return;
-}
-
-WebRtc_Word16 ACMAMR::InternalCreateDecoder() {
- return -1;
-}
-
-void ACMAMR::DestructDecoderSafe() {
- return;
-}
-
-WebRtc_Word16 ACMAMR::SetBitRateSafe(const WebRtc_Word32 /* rate */) {
- return -1;
-}
-
-void ACMAMR::InternalDestructEncoderInst(void* /* ptrInst */) {
- return;
-}
-
-WebRtc_Word16 ACMAMR::SetAMREncoderPackingFormat(
- ACMAMRPackingFormat /* packingFormat */) {
- return -1;
-}
-
-ACMAMRPackingFormat ACMAMR::AMREncoderPackingFormat() const {
- return AMRUndefined;
-}
-
-WebRtc_Word16 ACMAMR::SetAMRDecoderPackingFormat(
- ACMAMRPackingFormat /* packingFormat */) {
- return -1;
-}
-
-ACMAMRPackingFormat ACMAMR::AMRDecoderPackingFormat() const {
- return AMRUndefined;
-}
-
-WebRtc_Word16 ACMAMR::UnregisterFromNetEqSafe(ACMNetEQ* /* netEq */,
- WebRtc_Word16 /* payloadType */) {
- return -1;
-}
-
-
-#else //===================== Actual Implementation =======================
-
-#define WEBRTC_AMR_MR475 0
-#define WEBRTC_AMR_MR515 1
-#define WEBRTC_AMR_MR59 2
-#define WEBRTC_AMR_MR67 3
-#define WEBRTC_AMR_MR74 4
-#define WEBRTC_AMR_MR795 5
-#define WEBRTC_AMR_MR102 6
-#define WEBRTC_AMR_MR122 7
-
-ACMAMR::ACMAMR(WebRtc_Word16 codecID)
- : _encoderInstPtr(NULL),
- _decoderInstPtr(NULL),
- _encodingMode(-1), // invalid value
- _encodingRate(0) { // invalid value
- _codecID = codecID;
- _hasInternalDTX = true;
- _encoderPackingFormat = AMRBandwidthEfficient;
- _decoderPackingFormat = AMRBandwidthEfficient;
- return;
-}
-
-ACMAMR::~ACMAMR() {
- if (_encoderInstPtr != NULL) {
- WebRtcAmr_FreeEnc(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
- if (_decoderInstPtr != NULL) {
- WebRtcAmr_FreeDec(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
- return;
-}
-
-WebRtc_Word16 ACMAMR::InternalEncode(WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte) {
- WebRtc_Word16 vadDecision = 1;
- // sanity check, if the rate is set correctly. we might skip this
- // sanity check. if rate is not set correctly, initialization flag
- // should be false and should not be here.
- if ((_encodingMode < WEBRTC_AMR_MR475) ||
- (_encodingMode > WEBRTC_AMR_MR122)) {
- *bitStreamLenByte = 0;
- return -1;
- }
- *bitStreamLenByte = WebRtcAmr_Encode(_encoderInstPtr,
- &_inAudio[_inAudioIxRead],
- _frameLenSmpl,
- (WebRtc_Word16*) bitStream,
- _encodingMode);
-
- // Update VAD, if internal DTX is used
- if (_hasInternalDTX && _dtxEnabled) {
- if (*bitStreamLenByte <= (7 * _frameLenSmpl / 160)) {
- vadDecision = 0;
- }
- for (WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) {
- _vadLabel[n] = vadDecision;
- }
- }
- // increment the read index
- _inAudioIxRead += _frameLenSmpl;
- return *bitStreamLenByte;
-}
-
-WebRtc_Word16 ACMAMR::DecodeSafe(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */) {
- return 0;
-}
-
-WebRtc_Word16 ACMAMR::EnableDTX() {
- if (_dtxEnabled) {
- return 0;
- } else if (_encoderExist) { // check if encoder exist
- // enable DTX
- if (WebRtcAmr_EncoderInit(_encoderInstPtr, 1) < 0) {
- return -1;
- }
- _dtxEnabled = true;
- return 0;
- } else {
- return -1;
- }
-}
-
-WebRtc_Word16 ACMAMR::DisableDTX() {
- if (!_dtxEnabled) {
- return 0;
- } else if (_encoderExist) { // check if encoder exist
- // disable DTX
- if (WebRtcAmr_EncoderInit(_encoderInstPtr, 0) < 0) {
- return -1;
- }
- _dtxEnabled = false;
- return 0;
- } else {
- // encoder doesn't exists, therefore disabling is harmless
- return 0;
- }
-}
-
-WebRtc_Word16 ACMAMR::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
- WebRtc_Word16 status = SetBitRateSafe((codecParams->codecInstant).rate);
- status += (WebRtcAmr_EncoderInit(
- _encoderInstPtr, ((codecParams->enableDTX) ? 1 : 0)) < 0) ? -1 : 0;
- status += (WebRtcAmr_EncodeBitmode(
- _encoderInstPtr, _encoderPackingFormat) < 0) ? -1 : 0;
- return (status < 0) ? -1 : 0;
-}
-
-WebRtc_Word16 ACMAMR::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */) {
- WebRtc_Word16 status =
- ((WebRtcAmr_DecoderInit(_decoderInstPtr) < 0) ? -1 : 0);
- status += WebRtcAmr_DecodeBitmode(_decoderInstPtr, _decoderPackingFormat);
- return (status < 0) ? -1 : 0;
-}
-
-WebRtc_Word32 ACMAMR::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst) {
- if (!_decoderInitialized) {
- // Todo:
- // log error
- return -1;
- }
- // Fill up the structure by calling
- // "SET_CODEC_PAR" & "SET_AMR_FUNCTION."
- // Then call NetEQ to add the codec to it's
- // database.
- SET_CODEC_PAR((codecDef), kDecoderAMR, codecInst.pltype, _decoderInstPtr,
- 8000);
- SET_AMR_FUNCTIONS((codecDef));
- return 0;
-}
-
-ACMGenericCodec* ACMAMR::CreateInstance(void) {
- return NULL;
-}
-
-WebRtc_Word16 ACMAMR::InternalCreateEncoder() {
- return WebRtcAmr_CreateEnc(&_encoderInstPtr);
-}
-
-void ACMAMR::DestructEncoderSafe() {
- if (_encoderInstPtr != NULL) {
- WebRtcAmr_FreeEnc(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
- // there is no encoder set the following
- _encoderExist = false;
- _encoderInitialized = false;
- _encodingMode = -1; // invalid value
- _encodingRate = 0; // invalid value
-}
-
-WebRtc_Word16 ACMAMR::InternalCreateDecoder() {
- return WebRtcAmr_CreateDec(&_decoderInstPtr);
-}
-
-void ACMAMR::DestructDecoderSafe() {
- if (_decoderInstPtr != NULL) {
- WebRtcAmr_FreeDec(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
- // there is no encoder instance set the followings
- _decoderExist = false;
- _decoderInitialized = false;
-}
-
-WebRtc_Word16 ACMAMR::SetBitRateSafe(const WebRtc_Word32 rate) {
- switch (rate) {
- case 4750: {
- _encodingMode = WEBRTC_AMR_MR475;
- _encodingRate = 4750;
- break;
- }
- case 5150: {
- _encodingMode = WEBRTC_AMR_MR515;
- _encodingRate = 5150;
- break;
- }
- case 5900: {
- _encodingMode = WEBRTC_AMR_MR59;
- _encodingRate = 5900;
- break;
- }
- case 6700: {
- _encodingMode = WEBRTC_AMR_MR67;
- _encodingRate = 6700;
- break;
- }
- case 7400: {
- _encodingMode = WEBRTC_AMR_MR74;
- _encodingRate = 7400;
- break;
- }
- case 7950: {
- _encodingMode = WEBRTC_AMR_MR795;
- _encodingRate = 7950;
- break;
- }
- case 10200: {
- _encodingMode = WEBRTC_AMR_MR102;
- _encodingRate = 10200;
- break;
- }
- case 12200: {
- _encodingMode = WEBRTC_AMR_MR122;
- _encodingRate = 12200;
- break;
- }
- default: {
- return -1;
- }
- }
- return 0;
-}
-
-void ACMAMR::InternalDestructEncoderInst(void* ptrInst) {
- // Free the memory where ptrInst is pointing to
- if (ptrInst != NULL) {
- WebRtcAmr_FreeEnc(reinterpret_cast<AMR_encinst_t_*>(ptrInst));
- }
- return;
-}
-
-WebRtc_Word16 ACMAMR::SetAMREncoderPackingFormat(
- ACMAMRPackingFormat packingFormat) {
- if ((packingFormat != AMRBandwidthEfficient) &&
- (packingFormat != AMROctetAlligned) &&
- (packingFormat != AMRFileStorage)) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Invalid AMR Encoder packing-format.");
- return -1;
- } else {
- if (WebRtcAmr_EncodeBitmode(_encoderInstPtr, packingFormat) < 0) {
- return -1;
- } else {
- _encoderPackingFormat = packingFormat;
- return 0;
- }
- }
-}
-
-ACMAMRPackingFormat ACMAMR::AMREncoderPackingFormat() const {
- return _encoderPackingFormat;
-}
-
-WebRtc_Word16 ACMAMR::SetAMRDecoderPackingFormat(
- ACMAMRPackingFormat packingFormat) {
- if ((packingFormat != AMRBandwidthEfficient) &&
- (packingFormat != AMROctetAlligned) &&
- (packingFormat != AMRFileStorage)) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Invalid AMR decoder packing-format.");
- return -1;
- } else {
- if (WebRtcAmr_DecodeBitmode(_decoderInstPtr, packingFormat) < 0) {
- return -1;
- } else {
- _decoderPackingFormat = packingFormat;
- return 0;
- }
- }
-}
-
-ACMAMRPackingFormat ACMAMR::AMRDecoderPackingFormat() const {
- return _decoderPackingFormat;
-}
-
-WebRtc_Word16 ACMAMR::UnregisterFromNetEqSafe(ACMNetEQ* netEq,
- WebRtc_Word16 payloadType) {
- if (payloadType != _decoderParams.codecInstant.pltype) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot unregister codec %s given payload-type %d does not "
- "match the stored payload type",
- _decoderParams.codecInstant.plname, payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
- return netEq->RemoveCodec(kDecoderAMR);
-}
-
-#endif
-
-}
diff --git a/trunk/src/modules/audio_coding/main/source/acm_amr.h b/trunk/src/modules/audio_coding/main/source/acm_amr.h
deleted file mode 100644
index cc342e7..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_amr.h
+++ /dev/null
@@ -1,86 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_
-
-#include "acm_generic_codec.h"
-
-// forward declaration
-struct AMR_encinst_t_;
-struct AMR_decinst_t_;
-
-namespace webrtc {
-
-enum ACMAMRPackingFormat;
-
-class ACMAMR: public ACMGenericCodec {
- public:
- ACMAMR(WebRtc_Word16 codecID);
- ~ACMAMR();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 SetAMREncoderPackingFormat(
- const ACMAMRPackingFormat packingFormat);
-
- ACMAMRPackingFormat AMREncoderPackingFormat() const;
-
- WebRtc_Word16 SetAMRDecoderPackingFormat(
- const ACMAMRPackingFormat packingFormat);
-
- ACMAMRPackingFormat AMRDecoderPackingFormat() const;
-
- protected:
- WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio, WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- void InternalDestructEncoderInst(void* ptrInst);
-
- WebRtc_Word16 SetBitRateSafe(const WebRtc_Word32 rate);
-
- WebRtc_Word16 EnableDTX();
-
- WebRtc_Word16 DisableDTX();
-
- WebRtc_Word16 UnregisterFromNetEqSafe(ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- AMR_encinst_t_* _encoderInstPtr;
- AMR_decinst_t_* _decoderInstPtr;
- WebRtc_Word16 _encodingMode;
- WebRtc_Word16 _encodingRate;
- ACMAMRPackingFormat _encoderPackingFormat;
- ACMAMRPackingFormat _decoderPackingFormat;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMR_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_amrwb.cc b/trunk/src/modules/audio_coding/main/source/acm_amrwb.cc
deleted file mode 100644
index fed2e3f..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_amrwb.cc
+++ /dev/null
@@ -1,450 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_amrwb.h"
-#include "acm_common_defs.h"
-#include "acm_neteq.h"
-#include "audio_coding_module_typedefs.h"
-#include "rw_lock_wrapper.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-
-#ifdef WEBRTC_CODEC_AMRWB
-// NOTE! GSM AMR-wb is not included in the open-source package. The
-// following interface file is needed:
-//
-// /modules/audio_coding/codecs/amrwb/main/interface/amrwb_interface.h
-//
-// The API in the header file should match the one below.
-//
-// int16_t WebRtcAmrWb_CreateEnc(AMRWB_encinst_t_** encInst);
-// int16_t WebRtcAmrWb_CreateDec(AMRWB_decinst_t_** decInst);
-// int16_t WebRtcAmrWb_FreeEnc(AMRWB_encinst_t_* encInst);
-// int16_t WebRtcAmrWb_FreeDec(AMRWB_decinst_t_* decInst);
-// int16_t WebRtcAmrWb_Encode(AMRWB_encinst_t_* encInst, int16_t* input,
-// int16_t len, int16_t* output, int16_t mode);
-// int16_t WebRtcAmrWb_EncoderInit(AMRWB_encinst_t_* encInst,
-// int16_t dtxMode);
-// int16_t WebRtcAmrWb_EncodeBitmode(AMRWB_encinst_t_* encInst,
-// int format);
-// int16_t WebRtcAmrWb_Decode(AMRWB_decinst_t_* decInst);
-// int16_t WebRtcAmrWb_DecodePlc(AMRWB_decinst_t_* decInst);
-// int16_t WebRtcAmrWb_DecoderInit(AMRWB_decinst_t_* decInst);
-// int16_t WebRtcAmrWb_DecodeBitmode(AMRWB_decinst_t_* decInst,
-// int format);
-// void WebRtcAmrWb_Version(char *versionStr, short len);
-#include "amrwb_interface.h"
-#endif
-
-namespace webrtc {
-
-#ifndef WEBRTC_CODEC_AMRWB
-ACMAMRwb::ACMAMRwb(WebRtc_Word16 /* codecID*/)
- : _encoderInstPtr(NULL),
- _decoderInstPtr(NULL),
- _encodingMode(-1), // invalid value
- _encodingRate(0), // invalid value
- _encoderPackingFormat(AMRBandwidthEfficient),
- _decoderPackingFormat(AMRBandwidthEfficient) {
- return;
-}
-
-ACMAMRwb::~ACMAMRwb() {
- return;
-}
-
-WebRtc_Word16 ACMAMRwb::InternalEncode(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */) {
- return -1;
-}
-
-WebRtc_Word16 ACMAMRwb::DecodeSafe(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */) {
- return -1;
-}
-
-WebRtc_Word16 ACMAMRwb::EnableDTX() {
- return -1;
-}
-
-WebRtc_Word16 ACMAMRwb::DisableDTX() {
- return -1;
-}
-
-WebRtc_Word16 ACMAMRwb::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */) {
- return -1;
-}
-
-WebRtc_Word16 ACMAMRwb::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */) {
- return -1;
-}
-
-WebRtc_Word32 ACMAMRwb::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
- const CodecInst& /* codecInst */) {
- return -1;
-}
-
-ACMGenericCodec*
-ACMAMRwb::CreateInstance(void) {
- return NULL;
-}
-
-WebRtc_Word16 ACMAMRwb::InternalCreateEncoder() {
- return -1;
-}
-
-void ACMAMRwb::DestructEncoderSafe() {
- return;
-}
-
-WebRtc_Word16 ACMAMRwb::InternalCreateDecoder() {
- return -1;
-}
-
-void ACMAMRwb::DestructDecoderSafe() {
- return;
-}
-
-WebRtc_Word16 ACMAMRwb::SetBitRateSafe(const WebRtc_Word32 /* rate */) {
- return -1;
-}
-
-void ACMAMRwb::InternalDestructEncoderInst(void* /* ptrInst */) {
- return;
-}
-
-WebRtc_Word16 ACMAMRwb::SetAMRwbEncoderPackingFormat(
- ACMAMRPackingFormat /* packingFormat */) {
- return -1;
-}
-
-ACMAMRPackingFormat ACMAMRwb::AMRwbEncoderPackingFormat() const {
- return AMRUndefined;
-}
-
-WebRtc_Word16 ACMAMRwb::SetAMRwbDecoderPackingFormat(
- ACMAMRPackingFormat /* packingFormat */) {
- return -1;
-}
-
-ACMAMRPackingFormat ACMAMRwb::AMRwbDecoderPackingFormat() const {
- return AMRUndefined;
-}
-
-WebRtc_Word16 ACMAMRwb::UnregisterFromNetEqSafe(
- ACMNetEQ* /* netEq */,
- WebRtc_Word16 /* payloadType */) {
- return -1;
-}
-
-#else //===================== Actual Implementation =======================
-
-#define AMRWB_MODE_7k 0
-#define AMRWB_MODE_9k 1
-#define AMRWB_MODE_12k 2
-#define AMRWB_MODE_14k 3
-#define AMRWB_MODE_16k 4
-#define AMRWB_MODE_18k 5
-#define AMRWB_MODE_20k 6
-#define AMRWB_MODE_23k 7
-#define AMRWB_MODE_24k 8
-
-ACMAMRwb::ACMAMRwb(WebRtc_Word16 codecID)
- : _encoderInstPtr(NULL),
- _decoderInstPtr(NULL),
- _encodingMode(-1), // invalid value
- _encodingRate(0) { // invalid value
- _codecID = codecID;
- _hasInternalDTX = true;
- _encoderPackingFormat = AMRBandwidthEfficient;
- _decoderPackingFormat = AMRBandwidthEfficient;
- return;
-}
-
-ACMAMRwb::~ACMAMRwb() {
- if (_encoderInstPtr != NULL) {
- WebRtcAmrWb_FreeEnc(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
- if (_decoderInstPtr != NULL) {
- WebRtcAmrWb_FreeDec(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
- return;
-}
-
-WebRtc_Word16 ACMAMRwb::InternalEncode(WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte) {
- WebRtc_Word16 vadDecision = 1;
- // sanity check, if the rate is set correctly. we might skip this
- // sanity check. if rate is not set correctly, initialization flag
- // should be false and should not be here.
- if ((_encodingMode < AMRWB_MODE_7k) || (_encodingMode > AMRWB_MODE_24k)) {
- *bitStreamLenByte = 0;
- return -1;
- }
- *bitStreamLenByte = WebRtcAmrWb_Encode(_encoderInstPtr,
- &_inAudio[_inAudioIxRead],
- _frameLenSmpl,
- (WebRtc_Word16*) bitStream,
- _encodingMode);
-
- // Update VAD, if internal DTX is used
- if (_hasInternalDTX && _dtxEnabled) {
- if (*bitStreamLenByte <= (7 * _frameLenSmpl / 160)) {
- vadDecision = 0;
- }
- for (WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++) {
- _vadLabel[n] = vadDecision;
- }
- }
- // increment the read index this tell the caller that how far
- // we have gone forward in reading the audio buffer
- _inAudioIxRead += _frameLenSmpl;
- return *bitStreamLenByte;
-}
-
-WebRtc_Word16 ACMAMRwb::DecodeSafe(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */) {
- return 0;
-}
-
-WebRtc_Word16 ACMAMRwb::EnableDTX() {
- if (_dtxEnabled) {
- return 0;
- } else if (_encoderExist) { // check if encoder exist
- // enable DTX
- if (WebRtcAmrWb_EncoderInit(_encoderInstPtr, 1) < 0) {
- return -1;
- }
- _dtxEnabled = true;
- return 0;
- } else {
- return -1;
- }
-}
-
-WebRtc_Word16 ACMAMRwb::DisableDTX() {
- if (!_dtxEnabled) {
- return 0;
- } else if (_encoderExist) { // check if encoder exist
- // disable DTX
- if (WebRtcAmrWb_EncoderInit(_encoderInstPtr, 0) < 0) {
- return -1;
- }
- _dtxEnabled = false;
- return 0;
- } else {
- // encoder doesn't exists, therefore disabling is harmless
- return 0;
- }
-}
-
-WebRtc_Word16 ACMAMRwb::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
- // sanity check
- if (_encoderInstPtr == NULL) {
- return -1;
- }
-
- WebRtc_Word16 status = SetBitRateSafe((codecParams->codecInstant).rate);
- status += (WebRtcAmrWb_EncoderInit(
- _encoderInstPtr, ((codecParams->enableDTX) ? 1 : 0)) < 0) ? -1 : 0;
- status += (WebRtcAmrWb_EncodeBitmode(
- _encoderInstPtr, _encoderPackingFormat) < 0) ? -1 : 0;
- return (status < 0) ? -1 : 0;
-}
-
-WebRtc_Word16 ACMAMRwb::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */) {
- WebRtc_Word16 status = WebRtcAmrWb_DecodeBitmode(_decoderInstPtr,
- _decoderPackingFormat);
- status += ((WebRtcAmrWb_DecoderInit(_decoderInstPtr) < 0) ? -1 : 0);
- return (status < 0) ? -1 : 0;
-}
-
-WebRtc_Word32 ACMAMRwb::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst) {
- if (!_decoderInitialized) {
- return -1;
- }
-
- // Fill up the structure by calling
- // "SET_CODEC_PAR" & "SET_AMRWB_FUNCTION."
- // Then call NetEQ to add the codec to it's
- // database.
- SET_CODEC_PAR((codecDef), kDecoderAMRWB, codecInst.pltype, _decoderInstPtr,
- 16000);
- SET_AMRWB_FUNCTIONS((codecDef));
- return 0;
-}
-
-ACMGenericCodec* ACMAMRwb::CreateInstance(void) {
- return NULL;
-}
-
-WebRtc_Word16 ACMAMRwb::InternalCreateEncoder() {
- return WebRtcAmrWb_CreateEnc(&_encoderInstPtr);
-}
-
-void ACMAMRwb::DestructEncoderSafe() {
- if (_encoderInstPtr != NULL) {
- WebRtcAmrWb_FreeEnc(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
- // there is no encoder set the following
- _encoderExist = false;
- _encoderInitialized = false;
- _encodingMode = -1; // invalid value
- _encodingRate = 0;
-}
-
-WebRtc_Word16 ACMAMRwb::InternalCreateDecoder() {
- return WebRtcAmrWb_CreateDec(&_decoderInstPtr);
-}
-
-void ACMAMRwb::DestructDecoderSafe() {
- if (_decoderInstPtr != NULL) {
- WebRtcAmrWb_FreeDec(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
- // there is no encoder instance set the followings
- _decoderExist = false;
- _decoderInitialized = false;
-}
-
-WebRtc_Word16 ACMAMRwb::SetBitRateSafe(const WebRtc_Word32 rate) {
- switch (rate) {
- case 7000: {
- _encodingMode = AMRWB_MODE_7k;
- _encodingRate = 7000;
- break;
- }
- case 9000: {
- _encodingMode = AMRWB_MODE_9k;
- _encodingRate = 9000;
- break;
- }
- case 12000: {
- _encodingMode = AMRWB_MODE_12k;
- _encodingRate = 12000;
- break;
- }
- case 14000: {
- _encodingMode = AMRWB_MODE_14k;
- _encodingRate = 14000;
- break;
- }
- case 16000: {
- _encodingMode = AMRWB_MODE_16k;
- _encodingRate = 16000;
- break;
- }
- case 18000: {
- _encodingMode = AMRWB_MODE_18k;
- _encodingRate = 18000;
- break;
- }
- case 20000: {
- _encodingMode = AMRWB_MODE_20k;
- _encodingRate = 20000;
- break;
- }
- case 23000: {
- _encodingMode = AMRWB_MODE_23k;
- _encodingRate = 23000;
- break;
- }
- case 24000: {
- _encodingMode = AMRWB_MODE_24k;
- _encodingRate = 24000;
- break;
- }
- default: {
- return -1;
- }
- }
- return 0;
-}
-
-void ACMAMRwb::InternalDestructEncoderInst(void* ptrInst) {
- if (ptrInst != NULL) {
- WebRtcAmrWb_FreeEnc(static_cast<AMRWB_encinst_t_*>(ptrInst));
- }
- return;
-}
-
-WebRtc_Word16 ACMAMRwb::SetAMRwbEncoderPackingFormat(
- ACMAMRPackingFormat packingFormat) {
- if ((packingFormat != AMRBandwidthEfficient) &&
- (packingFormat != AMROctetAlligned) &&
- (packingFormat != AMRFileStorage)) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Invalid AMRwb encoder packing-format.");
- return -1;
- } else {
- if (WebRtcAmrWb_EncodeBitmode(_encoderInstPtr, packingFormat) < 0) {
- return -1;
- } else {
- _encoderPackingFormat = packingFormat;
- return 0;
- }
- }
-}
-
-ACMAMRPackingFormat ACMAMRwb::AMRwbEncoderPackingFormat() const {
- return _encoderPackingFormat;
-}
-
-WebRtc_Word16 ACMAMRwb::SetAMRwbDecoderPackingFormat(
- ACMAMRPackingFormat packingFormat) {
- if ((packingFormat != AMRBandwidthEfficient) &&
- (packingFormat != AMROctetAlligned) &&
- (packingFormat != AMRFileStorage)) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Invalid AMRwb decoder packing-format.");
- return -1;
- } else {
- if (WebRtcAmrWb_DecodeBitmode(_decoderInstPtr, packingFormat) < 0) {
- return -1;
- } else {
- _decoderPackingFormat = packingFormat;
- return 0;
- }
- }
-}
-
-ACMAMRPackingFormat ACMAMRwb::AMRwbDecoderPackingFormat() const {
- return _decoderPackingFormat;
-}
-
-WebRtc_Word16 ACMAMRwb::UnregisterFromNetEqSafe(ACMNetEQ* netEq,
- WebRtc_Word16 payloadType) {
- if (payloadType != _decoderParams.codecInstant.pltype) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot unregister codec %s given payload-type %d does not"
- "match the stored payload type",
- _decoderParams.codecInstant.plname, payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
- return netEq->RemoveCodec(kDecoderAMRWB);
-}
-#endif
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_amrwb.h b/trunk/src/modules/audio_coding/main/source/acm_amrwb.h
deleted file mode 100644
index 775f07f..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_amrwb.h
+++ /dev/null
@@ -1,87 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_
-
-#include "acm_generic_codec.h"
-
-// forward declaration
-struct AMRWB_encinst_t_;
-struct AMRWB_decinst_t_;
-
-namespace webrtc {
-
-enum ACMAMRPackingFormat;
-
-class ACMAMRwb: public ACMGenericCodec {
- public:
- ACMAMRwb(WebRtc_Word16 codecID);
- ~ACMAMRwb();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams* codecParams);
-
- WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams* codecParams);
-
- WebRtc_Word16 SetAMRwbEncoderPackingFormat(
- const ACMAMRPackingFormat packingFormat);
-
- ACMAMRPackingFormat AMRwbEncoderPackingFormat() const;
-
- WebRtc_Word16 SetAMRwbDecoderPackingFormat(
- const ACMAMRPackingFormat packingFormat);
-
- ACMAMRPackingFormat AMRwbDecoderPackingFormat() const;
-
- protected:
- WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio, WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- void InternalDestructEncoderInst(void* ptrInst);
-
- WebRtc_Word16 SetBitRateSafe(const WebRtc_Word32 rate);
-
- WebRtc_Word16 EnableDTX();
-
- WebRtc_Word16 DisableDTX();
-
- WebRtc_Word16 UnregisterFromNetEqSafe(ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- AMRWB_encinst_t_* _encoderInstPtr;
- AMRWB_decinst_t_* _decoderInstPtr;
-
- WebRtc_Word16 _encodingMode;
- WebRtc_Word16 _encodingRate;
- ACMAMRPackingFormat _encoderPackingFormat;
- ACMAMRPackingFormat _decoderPackingFormat;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_AMRWB_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_celt.cc b/trunk/src/modules/audio_coding/main/source/acm_celt.cc
deleted file mode 100644
index 5b12f32..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_celt.cc
+++ /dev/null
@@ -1,317 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_common_defs.h"
-#include "acm_neteq.h"
-#include "acm_celt.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-// TODO(tlegrand): Add full paths.
-
-#ifdef WEBRTC_CODEC_CELT
-// NOTE! Celt is not included in the open-source package. Modify this file or
-// your codec API to match the function call and name of used Celt API file.
-#include "celt_interface.h"
-#endif
-
-namespace webrtc {
-
-#ifndef WEBRTC_CODEC_CELT
-
-ACMCELT::ACMCELT(int16_t /* codecID */)
- : enc_inst_ptr_(NULL),
- dec_inst_ptr_(NULL),
- sampling_freq_(0),
- bitrate_(0),
- channels_(1),
- dec_channels_(1) {
- return;
-}
-
-ACMCELT::~ACMCELT() {
- return;
-}
-
-int16_t ACMCELT::InternalEncode(uint8_t* /* bitStream */,
- int16_t* /* bitStreamLenByte */) {
- return -1;
-}
-
-int16_t ACMCELT::DecodeSafe(uint8_t* /* bitStream */,
- int16_t /* bitStreamLenByte */,
- int16_t* /* audio */,
- int16_t* /* audioSamples */,
- WebRtc_Word8* /* speechType */) {
- return -1;
-}
-
-int16_t ACMCELT::InternalInitEncoder(WebRtcACMCodecParams* /* codecParams */) {
- return -1;
-}
-
-int16_t ACMCELT::InternalInitDecoder(WebRtcACMCodecParams* /* codecParams */) {
- return -1;
-}
-
-int32_t ACMCELT::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
- const CodecInst& /* codecInst */) {
- return -1;
-}
-
-ACMGenericCodec* ACMCELT::CreateInstance(void) {
- return NULL;
-}
-
-int16_t ACMCELT::InternalCreateEncoder() {
- return -1;
-}
-
-void ACMCELT::DestructEncoderSafe() {
- return;
-}
-
-int16_t ACMCELT::InternalCreateDecoder() {
- return -1;
-}
-
-void ACMCELT::DestructDecoderSafe() {
- return;
-}
-
-void ACMCELT::InternalDestructEncoderInst(void* /* ptrInst */) {
- return;
-}
-
-int16_t ACMCELT::UnregisterFromNetEqSafe(ACMNetEQ* /* netEq */,
- int16_t /* payloadType */) {
- return -1;
-}
-
-bool ACMCELT::IsTrueStereoCodec() {
- return true;
-}
-
-int16_t ACMCELT::SetBitRateSafe(const int32_t /*rate*/) {
- return -1;
-}
-
-#else //===================== Actual Implementation =======================
-
-ACMCELT::ACMCELT(int16_t codecID)
- : enc_inst_ptr_(NULL),
- dec_inst_ptr_(NULL),
- sampling_freq_(32000), // Default sampling frequency.
- bitrate_(64000), // Default rate.
- channels_(1), // Default send mono.
- dec_channels_(1) { // Default receive mono.
- // TODO(tlegrand): remove later when ACMGenericCodec has a new constructor.
- _codecID = codecID;
-
- return;
-}
-
-ACMCELT::~ACMCELT() {
- if (enc_inst_ptr_ != NULL) {
- WebRtcCelt_FreeEnc(enc_inst_ptr_);
- enc_inst_ptr_ = NULL;
- }
- if (dec_inst_ptr_ != NULL) {
- WebRtcCelt_FreeDec(dec_inst_ptr_);
- dec_inst_ptr_ = NULL;
- }
- return;
-}
-
-int16_t ACMCELT::InternalEncode(uint8_t* bitStream, int16_t* bitStreamLenByte) {
- *bitStreamLenByte = 0;
-
- // Call Encoder.
- *bitStreamLenByte = WebRtcCelt_Encode(enc_inst_ptr_,
- &_inAudio[_inAudioIxRead],
- bitStream);
-
- // Increment the read index this tell the caller that how far
- // we have gone forward in reading the audio buffer.
- _inAudioIxRead += _frameLenSmpl * channels_;
-
- if (*bitStreamLenByte < 0) {
- // Error reported from the encoder.
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalEncode: Encode error for Celt");
- *bitStreamLenByte = 0;
- return -1;
- }
-
- return *bitStreamLenByte;
-}
-
-int16_t ACMCELT::DecodeSafe(uint8_t* /* bitStream */,
- int16_t /* bitStreamLenByte */,
- int16_t* /* audio */,
- int16_t* /* audioSamples */,
- WebRtc_Word8* /* speechType */) {
- return 0;
-}
-
-int16_t ACMCELT::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
- // Set bitrate and check that it is within the valid range.
- int16_t status = SetBitRateSafe((codecParams->codecInstant).rate);
- if (status < 0) {
- return -1;
- }
-
- // If number of channels changed we need to re-create memory.
- if (codecParams->codecInstant.channels != channels_) {
- WebRtcCelt_FreeEnc(enc_inst_ptr_);
- enc_inst_ptr_ = NULL;
- // Store new number of channels.
- channels_ = codecParams->codecInstant.channels;
- if (WebRtcCelt_CreateEnc(&enc_inst_ptr_, channels_) < 0) {
- return -1;
- }
- }
-
- // Initiate encoder.
- if (WebRtcCelt_EncoderInit(enc_inst_ptr_, channels_, bitrate_) >= 0) {
- return 0;
- } else {
- return -1;
- }
-}
-
-int16_t ACMCELT::InternalInitDecoder(WebRtcACMCodecParams* codecParams) {
- // If number of channels changed we need to re-create memory.
- if (codecParams->codecInstant.channels != dec_channels_) {
- WebRtcCelt_FreeDec(dec_inst_ptr_);
- dec_inst_ptr_ = NULL;
- // Store new number of channels.
- dec_channels_ = codecParams->codecInstant.channels;
- if (WebRtcCelt_CreateDec(&dec_inst_ptr_, dec_channels_) < 0) {
- return -1;
- }
- }
-
- // Initiate decoder.
- if (WebRtcCelt_DecoderInit(dec_inst_ptr_) < 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalInitDecoder: init decoder failed for Celt.");
- return -1;
- }
- return 0;
-}
-
-int32_t ACMCELT::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst) {
- if (!_decoderInitialized) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "CodecDef: Decoder uninitialized for Celt");
- return -1;
- }
-
- // Fill up the structure by calling
- // "SET_CODEC_PAR" and "SET_CELT_FUNCTIONS" or "SET_CELTSLAVE_FUNCTIONS".
- // Then call NetEQ to add the codec to it's
- // database.
- SET_CODEC_PAR((codecDef), kDecoderCELT_32, codecInst.pltype, dec_inst_ptr_,
- 32000);
- // If this is the master of NetEQ, regular decoder will be added, otherwise
- // the slave decoder will be used.
- if (_isMaster) {
- SET_CELT_FUNCTIONS((codecDef));
- } else {
- SET_CELTSLAVE_FUNCTIONS((codecDef));
- }
- return 0;
-}
-
-ACMGenericCodec* ACMCELT::CreateInstance(void) {
- return NULL;
-}
-
-int16_t ACMCELT::InternalCreateEncoder() {
- if (WebRtcCelt_CreateEnc(&enc_inst_ptr_, _noChannels) < 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalCreateEncoder: create encoder failed for Celt");
- return -1;
- }
- channels_ = _noChannels;
- return 0;
-}
-
-void ACMCELT::DestructEncoderSafe() {
- _encoderExist = false;
- _encoderInitialized = false;
- if (enc_inst_ptr_ != NULL) {
- WebRtcCelt_FreeEnc(enc_inst_ptr_);
- enc_inst_ptr_ = NULL;
- }
-}
-
-int16_t ACMCELT::InternalCreateDecoder() {
- if (WebRtcCelt_CreateDec(&dec_inst_ptr_, dec_channels_) < 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalCreateDecoder: create decoder failed for Celt");
- return -1;
- }
-
- return 0;
-}
-
-void ACMCELT::DestructDecoderSafe() {
- _decoderExist = false;
- _decoderInitialized = false;
- if (dec_inst_ptr_ != NULL) {
- WebRtcCelt_FreeDec(dec_inst_ptr_);
- dec_inst_ptr_ = NULL;
- }
-}
-
-void ACMCELT::InternalDestructEncoderInst(void* ptrInst) {
- if (ptrInst != NULL) {
- WebRtcCelt_FreeEnc(static_cast<CELT_encinst_t*>(ptrInst));
- }
- return;
-}
-
-int16_t ACMCELT::UnregisterFromNetEqSafe(ACMNetEQ* netEq, int16_t payloadType) {
- if (payloadType != _decoderParams.codecInstant.pltype) {
- WEBRTC_TRACE(
- webrtc::kTraceError,
- webrtc::kTraceAudioCoding,
- _uniqueID,
- "Cannot unregister codec: given payload-type does not match the stored "
- "payload type",
- _decoderParams.codecInstant.plname, payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
- return netEq->RemoveCodec(kDecoderCELT_32);
-}
-
-bool ACMCELT::IsTrueStereoCodec() {
- return true;
-}
-
-int16_t ACMCELT::SetBitRateSafe(const int32_t rate) {
- // Check that rate is in the valid range.
- if ((rate >= 48000) && (rate <= 128000)) {
- bitrate_ = rate;
- return 0;
- } else {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "SetBitRateSafe: Invalid rate Celt");
- return -1;
- }
-}
-
-#endif
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_celt.h b/trunk/src/modules/audio_coding/main/source/acm_celt.h
deleted file mode 100644
index 776bd7f..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_celt.h
+++ /dev/null
@@ -1,74 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_
-
-#include "acm_generic_codec.h"
-
-// forward declaration
-struct CELT_encinst_t_;
-struct CELT_decinst_t_;
-
-namespace webrtc {
-
-class ACMCELT : public ACMGenericCodec {
- public:
- ACMCELT(int16_t codecID);
- ~ACMCELT();
-
- ACMGenericCodec* CreateInstance(void);
-
- int16_t InternalEncode(uint8_t* bitstream, int16_t* bitStreamLenByte);
-
- int16_t InternalInitEncoder(WebRtcACMCodecParams *codecParams);
-
- int16_t InternalInitDecoder(WebRtcACMCodecParams *codecParams);
-
- protected:
-
- WebRtc_Word16 DecodeSafe(
- uint8_t* /* bitStream */,
- int16_t /* bitStreamLenByte */,
- int16_t* /* audio */,
- int16_t* /* audioSamples */,
- // TODO(leozwang): use int8_t here when WebRtc_Word8 is properly typed.
- // http://code.google.com/p/webrtc/issues/detail?id=311
- WebRtc_Word8* /* speechType */);
-
- int32_t CodecDef(WebRtcNetEQ_CodecDef& codecDef, const CodecInst& codecInst);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- int16_t InternalCreateEncoder();
-
- int16_t InternalCreateDecoder();
-
- void InternalDestructEncoderInst(void* ptrInst);
-
- int16_t UnregisterFromNetEqSafe(ACMNetEQ* netEq, int16_t payloadType);
-
- bool IsTrueStereoCodec();
-
- int16_t SetBitRateSafe(const int32_t rate);
-
- CELT_encinst_t_* enc_inst_ptr_;
- CELT_decinst_t_* dec_inst_ptr_;
- uint16_t sampling_freq_;
- int32_t bitrate_;
- uint16_t channels_;
- uint16_t dec_channels_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CELT_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_cng.cc b/trunk/src/modules/audio_coding/main/source/acm_cng.cc
deleted file mode 100644
index 1bda752..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_cng.cc
+++ /dev/null
@@ -1,157 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_cng.h"
-#include "acm_codec_database.h"
-#include "acm_common_defs.h"
-#include "acm_neteq.h"
-#include "trace.h"
-#include "webrtc_cng.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-
-namespace webrtc {
-
-ACMCNG::ACMCNG(WebRtc_Word16 codecID) {
- _encoderInstPtr = NULL;
- _decoderInstPtr = NULL;
- _codecID = codecID;
- _sampFreqHz = ACMCodecDB::CodecFreq(_codecID);
- return;
-}
-
-ACMCNG::~ACMCNG() {
- if (_encoderInstPtr != NULL) {
- WebRtcCng_FreeEnc(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
- if (_decoderInstPtr != NULL) {
- WebRtcCng_FreeDec(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
- return;
-}
-
-// CNG is not like a regular encoder, this function
-// should not be called normally
-// instead the following function is called from inside
-// ACMGenericCodec::ProcessFrameVADDTX
-WebRtc_Word16 ACMCNG::InternalEncode(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */) {
- return -1;
-}
-
-WebRtc_Word16 ACMCNG::DecodeSafe(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */) {
- return 0;
-}
-
-// CNG is not like a regular encoder,
-// this function should not be called normally
-// instead the following function is called from inside
-// ACMGenericCodec::ProcessFrameVADDTX
-WebRtc_Word16 ACMCNG::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */) {
- return -1;
-}
-
-WebRtc_Word16 ACMCNG::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */) {
- return WebRtcCng_InitDec(_decoderInstPtr);
-}
-
-WebRtc_Word32 ACMCNG::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst) {
- if (!_decoderInitialized) {
- // TODO (tlegrand): log error
- return -1;
- }
- // Fill up the structure by calling
- // "SET_CODEC_PAR" & "SET_CNG_FUNCTION."
- // Then return the structure back to NetEQ to add the codec to it's
- // database.
-
- if (_sampFreqHz == 8000 || _sampFreqHz == 16000 || _sampFreqHz == 32000) {
- SET_CODEC_PAR((codecDef), kDecoderCNG, codecInst.pltype,
- _decoderInstPtr, _sampFreqHz);
- SET_CNG_FUNCTIONS((codecDef));
- return 0;
- } else {
- return -1;
- }
-}
-
-ACMGenericCodec* ACMCNG::CreateInstance(void) {
- return NULL;
-}
-
-WebRtc_Word16 ACMCNG::InternalCreateEncoder() {
- if (WebRtcCng_CreateEnc(&_encoderInstPtr) < 0) {
- _encoderInstPtr = NULL;
- return -1;
- } else {
- return 0;
- }
-}
-
-void ACMCNG::DestructEncoderSafe() {
- if (_encoderInstPtr != NULL) {
- WebRtcCng_FreeEnc(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
- _encoderExist = false;
- _encoderInitialized = false;
-}
-
-WebRtc_Word16 ACMCNG::InternalCreateDecoder() {
- if (WebRtcCng_CreateDec(&_decoderInstPtr) < 0) {
- _decoderInstPtr = NULL;
- return -1;
- } else {
- return 0;
- }
-}
-
-void ACMCNG::DestructDecoderSafe() {
- if (_decoderInstPtr != NULL) {
- WebRtcCng_FreeDec(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
- _decoderExist = false;
- _decoderInitialized = false;
-}
-
-void ACMCNG::InternalDestructEncoderInst(void* ptrInst) {
- if (ptrInst != NULL) {
- WebRtcCng_FreeEnc(static_cast<CNG_enc_inst*>(ptrInst));
- }
- return;
-}
-
-WebRtc_Word16 ACMCNG::UnregisterFromNetEqSafe(ACMNetEQ* netEq,
- WebRtc_Word16 payloadType) {
- if (payloadType != _decoderParams.codecInstant.pltype) {
- WEBRTC_TRACE(
- webrtc::kTraceError,
- webrtc::kTraceAudioCoding,
- _uniqueID,
- "Cannot unregister codec %s given payload-type %d does not "
- "match the stored payload type",
- _decoderParams.codecInstant.plname, payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
- return netEq->RemoveCodec(kDecoderCNG);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_cng.h b/trunk/src/modules/audio_coding/main/source/acm_cng.h
deleted file mode 100644
index 1781b2b..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_cng.h
+++ /dev/null
@@ -1,73 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_
-
-#include "acm_generic_codec.h"
-
-// forward declaration
-struct WebRtcCngEncInst;
-struct WebRtcCngDecInst;
-
-namespace webrtc {
-
-class ACMCNG: public ACMGenericCodec {
- public:
- ACMCNG(WebRtc_Word16 codecID);
- ~ACMCNG();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
-
-protected:
- WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio, WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- void InternalDestructEncoderInst(void* ptrInst);
-
- WebRtc_Word16 EnableDTX() {
- return -1;
- }
-
- WebRtc_Word16 DisableDTX() {
- return -1;
- }
-
- WebRtc_Word16 UnregisterFromNetEqSafe(ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- WebRtcCngEncInst* _encoderInstPtr;
- WebRtcCngDecInst* _decoderInstPtr;
- WebRtc_Word16 _sampFreqHz;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CNG_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_codec_database.cc b/trunk/src/modules/audio_coding/main/source/acm_codec_database.cc
deleted file mode 100644
index c623d29..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_codec_database.cc
+++ /dev/null
@@ -1,998 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file generates databases with information about all supported audio
- * codecs.
- */
-
-// TODO(tlegrand): Change constant input pointers in all functions to constant
-// references, where appropriate.
-#include "acm_codec_database.h"
-
-#include <stdio.h>
-
-#include "acm_common_defs.h"
-#include "trace.h"
-
-// Includes needed to get version info and to create the codecs.
-// G.711, PCM mu-law and A-law.
-#include "acm_pcma.h"
-#include "acm_pcmu.h"
-#include "g711_interface.h"
-// CNG.
-#include "acm_cng.h"
-#include "webrtc_cng.h"
-// NetEQ.
-#include "webrtc_neteq.h"
-#ifdef WEBRTC_CODEC_ISAC
- #include "acm_isac.h"
- #include "acm_isac_macros.h"
- #include "isac.h"
-#endif
-#ifdef WEBRTC_CODEC_ISACFX
- #include "acm_isac.h"
- #include "acm_isac_macros.h"
- #include "isacfix.h"
-#endif
-#ifdef WEBRTC_CODEC_PCM16
- #include "pcm16b.h"
- #include "acm_pcm16b.h"
-#endif
-#ifdef WEBRTC_CODEC_ILBC
- #include "acm_ilbc.h"
- #include "ilbc.h"
-#endif
-#ifdef WEBRTC_CODEC_AMR
- #include "acm_amr.h"
- #include "amr_interface.h"
-#endif
-#ifdef WEBRTC_CODEC_AMRWB
- #include "acm_amrwb.h"
- #include "amrwb_interface.h"
-#endif
-#ifdef WEBRTC_CODEC_CELT
- #include "acm_celt.h"
- #include "celt_interface.h"
-#endif
-#ifdef WEBRTC_CODEC_G722
- #include "acm_g722.h"
- #include "g722_interface.h"
-#endif
-#ifdef WEBRTC_CODEC_G722_1
- #include "acm_g7221.h"
- #include "g7221_interface.h"
-#endif
-#ifdef WEBRTC_CODEC_G722_1C
- #include "acm_g7221c.h"
- #include "g7221c_interface.h"
-#endif
-#ifdef WEBRTC_CODEC_G729
- #include "acm_g729.h"
- #include "g729_interface.h"
-#endif
-#ifdef WEBRTC_CODEC_G729_1
- #include "acm_g7291.h"
- #include "g7291_interface.h"
-#endif
-#ifdef WEBRTC_CODEC_GSMFR
- #include "acm_gsmfr.h"
- #include "gsmfr_interface.h"
-#endif
-#ifdef WEBRTC_CODEC_SPEEX
- #include "acm_speex.h"
- #include "speex_interface.h"
-#endif
-#ifdef WEBRTC_CODEC_AVT
- #include "acm_dtmf_playout.h"
-#endif
-#ifdef WEBRTC_CODEC_RED
- #include "acm_red.h"
-#endif
-
-namespace webrtc {
-
-// We dynamically allocate some of the dynamic payload types to the defined
-// codecs. Note! There are a limited number of payload types. If more codecs
-// are defined they will receive reserved fixed payload types (values 67-95).
-const int kDynamicPayloadtypes[ACMCodecDB::kMaxNumCodecs] = {
- 105, 107, 108, 109, 110, 111, 112, 113, 114, 115, 116, 117, 118, 119,
- 120, 121, 122, 123, 124, 125, 126, 95, 94, 93, 92, 91, 90, 89,
- 88, 87, 86, 85, 84, 83, 82, 81, 80, 79, 78, 77, 76, 75,
- 74, 73, 72, 71, 70, 69, 68, 67
-};
-
-// Creates database with all supported codec at compile time.
-// Each entry needs the following parameters in the given order:
-// payload type, name, sampling frequency, packet size in samples,
-// default channel support, and default rate.
-#if (defined(WEBRTC_CODEC_PCM16) || \
- defined(WEBRTC_CODEC_AMR) || defined(WEBRTC_CODEC_AMRWB) || \
- defined(WEBRTC_CODEC_CELT) || defined(WEBRTC_CODEC_G729_1) || \
- defined(WEBRTC_CODEC_SPEEX) || defined(WEBRTC_CODEC_G722_1) || \
- defined(WEBRTC_CODEC_G722_1C))
-static int count_database = 0;
-#endif
-
-const CodecInst ACMCodecDB::database_[] = {
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
- {103, "ISAC", 16000, kIsacPacSize480, 1, kIsacWbDefaultRate},
-# if (defined(WEBRTC_CODEC_ISAC))
- {104, "ISAC", 32000, kIsacPacSize960, 1, kIsacSwbDefaultRate},
-# endif
-#endif
-#ifdef WEBRTC_CODEC_PCM16
- {kDynamicPayloadtypes[count_database++], "L16", 8000, 80, 1, 128000},
- {kDynamicPayloadtypes[count_database++], "L16", 16000, 160, 1, 256000},
- {kDynamicPayloadtypes[count_database++], "L16", 32000, 320, 1, 512000},
-#endif
- // G.711, PCM mu-law and A-law.
- {0, "PCMU", 8000, 160, 1, 64000},
- {8, "PCMA", 8000, 160, 1, 64000},
-#ifdef WEBRTC_CODEC_ILBC
- {102, "ILBC", 8000, 240, 1, 13300},
-#endif
-#ifdef WEBRTC_CODEC_AMR
- {kDynamicPayloadtypes[count_database++], "AMR", 8000, 160, 1, 12200},
-#endif
-#ifdef WEBRTC_CODEC_AMRWB
- {kDynamicPayloadtypes[count_database++], "AMR-WB", 16000, 320, 1, 20000},
-#endif
-#ifdef WEBRTC_CODEC_CELT
- {kDynamicPayloadtypes[count_database++], "CELT", 32000, 320, 2, 64000},
-#endif
-#ifdef WEBRTC_CODEC_G722
- {9, "G722", 16000, 320, 1, 64000},
-#endif
-#ifdef WEBRTC_CODEC_G722_1
- {kDynamicPayloadtypes[count_database++], "G7221", 16000, 320, 1, 32000},
- {kDynamicPayloadtypes[count_database++], "G7221", 16000, 320, 1, 24000},
- {kDynamicPayloadtypes[count_database++], "G7221", 16000, 320, 1, 16000},
-#endif
-#ifdef WEBRTC_CODEC_G722_1C
- {kDynamicPayloadtypes[count_database++], "G7221", 32000, 640, 1, 48000},
- {kDynamicPayloadtypes[count_database++], "G7221", 32000, 640, 1, 32000},
- {kDynamicPayloadtypes[count_database++], "G7221", 32000, 640, 1, 24000},
-#endif
-#ifdef WEBRTC_CODEC_G729
- {18, "G729", 8000, 240, 1, 8000},
-#endif
-#ifdef WEBRTC_CODEC_G729_1
- {kDynamicPayloadtypes[count_database++], "G7291", 16000, 320, 1, 32000},
-#endif
-#ifdef WEBRTC_CODEC_GSMFR
- {3, "GSM", 8000, 160, 1, 13200},
-#endif
-#ifdef WEBRTC_CODEC_SPEEX
- {kDynamicPayloadtypes[count_database++], "speex", 8000, 160, 1, 11000},
- {kDynamicPayloadtypes[count_database++], "speex", 16000, 320, 1, 22000},
-#endif
- // Comfort noise for three different sampling frequencies.
- {13, "CN", 8000, 240, 1, 0},
- {98, "CN", 16000, 480, 1, 0},
- {99, "CN", 32000, 960, 1, 0},
-#ifdef WEBRTC_CODEC_AVT
- {106, "telephone-event", 8000, 240, 1, 0},
-#endif
-#ifdef WEBRTC_CODEC_RED
- {127, "red", 8000, 0, 1, 0},
-#endif
- // To prevent compile errors due to trailing commas.
- {-1, "Null", -1, -1, -1, -1}
-};
-
-// Create database with all codec settings at compile time.
-// Each entry needs the following parameters in the given order:
-// Number of allowed packet sizes, a vector with the allowed packet sizes,
-// Basic block samples, max number of channels that are supported.
-const ACMCodecDB::CodecSettings ACMCodecDB::codec_settings_[] = {
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
- {2, {kIsacPacSize480, kIsacPacSize960}, 0, 1},
-# if (defined(WEBRTC_CODEC_ISAC))
- {1, {kIsacPacSize960}, 0, 1},
-# endif
-#endif
-#ifdef WEBRTC_CODEC_PCM16
- {4, {80, 160, 240, 320}, 0, 2},
- {4, {160, 320, 480, 640}, 0, 2},
- {2, {320, 640}, 0, 2},
-#endif
- // G.711, PCM mu-law and A-law.
- {6, {80, 160, 240, 320, 400, 480}, 0, 2},
- {6, {80, 160, 240, 320, 400, 480}, 0, 2},
-#ifdef WEBRTC_CODEC_ILBC
- {4, {160, 240, 320, 480}, 0, 1},
-#endif
-#ifdef WEBRTC_CODEC_AMR
- {3, {160, 320, 480}, 0, 1},
-#endif
-#ifdef WEBRTC_CODEC_AMRWB
- {3, {320, 640, 960}, 0, 1},
-#endif
-#ifdef WEBRTC_CODEC_CELT
- {1, {320}, 0, 2},
-#endif
-#ifdef WEBRTC_CODEC_G722
- {6, {160, 320, 480, 640, 800, 960}, 0, 2},
-#endif
-#ifdef WEBRTC_CODEC_G722_1
- {1, {320}, 320, 2},
- {1, {320}, 320, 2},
- {1, {320}, 320, 2},
-#endif
-#ifdef WEBRTC_CODEC_G722_1C
- {1, {640}, 640, 2},
- {1, {640}, 640, 2},
- {1, {640}, 640, 2},
-#endif
-#ifdef WEBRTC_CODEC_G729
- {6, {80, 160, 240, 320, 400, 480}, 0, 1},
-#endif
-#ifdef WEBRTC_CODEC_G729_1
- {3, {320, 640, 960}, 0, 1},
-#endif
-#ifdef WEBRTC_CODEC_GSMFR
- {3, {160, 320, 480}, 160, 1},
-#endif
-#ifdef WEBRTC_CODEC_SPEEX
- {3, {160, 320, 480}, 0, 1},
- {3, {320, 640, 960}, 0, 1},
-#endif
- // Comfort noise for three different sampling frequencies.
- {1, {240}, 240, 1},
- {1, {480}, 480, 1},
- {1, {960}, 960, 1},
-#ifdef WEBRTC_CODEC_AVT
- {1, {240}, 240, 1},
-#endif
-#ifdef WEBRTC_CODEC_RED
- {1, {0}, 0, 1},
-#endif
- // To prevent compile errors due to trailing commas.
- {-1, {-1}, -1, -1}
-};
-
-// Create a database of all NetEQ decoders at compile time.
-const WebRtcNetEQDecoder ACMCodecDB::neteq_decoders_[] = {
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
- kDecoderISAC,
-# if (defined(WEBRTC_CODEC_ISAC))
- kDecoderISACswb,
-# endif
-#endif
-#ifdef WEBRTC_CODEC_PCM16
- kDecoderPCM16B,
- kDecoderPCM16Bwb,
- kDecoderPCM16Bswb32kHz,
-#endif
- // G.711, PCM mu-las and A-law.
- kDecoderPCMu,
- kDecoderPCMa,
-#ifdef WEBRTC_CODEC_ILBC
- kDecoderILBC,
-#endif
-#ifdef WEBRTC_CODEC_AMR
- kDecoderAMR,
-#endif
-#ifdef WEBRTC_CODEC_AMRWB
- kDecoderAMRWB,
-#endif
-#ifdef WEBRTC_CODEC_CELT
- kDecoderCELT_32,
-#endif
-#ifdef WEBRTC_CODEC_G722
- kDecoderG722,
-#endif
-#ifdef WEBRTC_CODEC_G722_1
- kDecoderG722_1_32,
- kDecoderG722_1_24,
- kDecoderG722_1_16,
-#endif
-#ifdef WEBRTC_CODEC_G722_1C
- kDecoderG722_1C_48,
- kDecoderG722_1C_32,
- kDecoderG722_1C_24,
-#endif
-#ifdef WEBRTC_CODEC_G729
- kDecoderG729,
-#endif
-#ifdef WEBRTC_CODEC_G729_1
- kDecoderG729_1,
-#endif
-#ifdef WEBRTC_CODEC_GSMFR
- kDecoderGSMFR,
-#endif
-#ifdef WEBRTC_CODEC_SPEEX
- kDecoderSPEEX_8,
- kDecoderSPEEX_16,
-#endif
- // Comfort noise for three different sampling frequencies.
- kDecoderCNG,
- kDecoderCNG,
- kDecoderCNG,
-#ifdef WEBRTC_CODEC_AVT
- kDecoderAVT,
-#endif
-#ifdef WEBRTC_CODEC_RED
- kDecoderRED,
-#endif
- kDecoderReservedEnd
-};
-
-// Get codec information from database.
-// TODO(tlegrand): replace memcpy with a pointer to the data base memory.
-int ACMCodecDB::Codec(int codec_id, CodecInst* codec_inst) {
- // Error check to see that codec_id is not out of bounds.
- if ((codec_id < 0) || (codec_id >= kNumCodecs)) {
- return -1;
- }
-
- // Copy database information for the codec to the output.
- memcpy(codec_inst, &database_[codec_id], sizeof(CodecInst));
-
- return 0;
-}
-
-// Enumerator for error codes when asking for codec database id.
-enum {
- kInvalidCodec = -10,
- kInvalidFrequency = -20,
- kInvalidPayloadtype = -30,
- kInvalidPacketSize = -40,
- kInvalidRate = -50
-};
-
-// Gets the codec id number from the database. If there is some mismatch in
-// the codec settings, an error message will be recorded in the error string.
-// NOTE! Only the first mismatch found will be recorded in the error string.
-int ACMCodecDB::CodecNumber(const CodecInst* codec_inst, int* mirror_id,
- char* err_message, int max_message_len_byte) {
- int codec_id = ACMCodecDB::CodecNumber(codec_inst, mirror_id);
-
- // Write error message if ACMCodecDB::CodecNumber() returned error.
- if ((codec_id < 0) && (err_message != NULL)) {
- char my_err_msg[1000];
-
- if (codec_id == kInvalidCodec) {
- sprintf(my_err_msg, "Call to ACMCodecDB::CodecNumber failed, plname=%s "
- "is not a valid codec", codec_inst->plname);
- } else if (codec_id == kInvalidFrequency) {
- sprintf(my_err_msg, "Call to ACMCodecDB::CodecNumber failed, plfreq=%d "
- "is not a valid frequency for the codec %s", codec_inst->plfreq,
- codec_inst->plname);
- } else if (codec_id == kInvalidPayloadtype) {
- sprintf(my_err_msg, "Call to ACMCodecDB::CodecNumber failed, payload "
- "number %d is out of range for %s", codec_inst->pltype,
- codec_inst->plname);
- } else if (codec_id == kInvalidPacketSize) {
- sprintf(my_err_msg, "Call to ACMCodecDB::CodecNumber failed, Packet "
- "size is out of range for %s", codec_inst->plname);
- } else if (codec_id == kInvalidRate) {
- sprintf(my_err_msg, "Call to ACMCodecDB::CodecNumber failed, rate=%d "
- "is not a valid rate for %s", codec_inst->rate,
- codec_inst->plname);
- } else {
- // Other error
- sprintf(my_err_msg, "invalid codec parameters to be registered, "
- "ACMCodecDB::CodecNumber failed");
- }
-
- strncpy(err_message, my_err_msg, max_message_len_byte - 1);
- // make sure that the message is null-terminated.
- err_message[max_message_len_byte - 1] = '\0';
- }
-
- return codec_id;
-}
-
-// Gets the codec id number from the database. If there is some mismatch in
-// the codec settings, the function will return an error code.
-// NOTE! The first mismatch found will generate the return value.
-int ACMCodecDB::CodecNumber(const CodecInst* codec_inst, int* mirror_id) {
- int codec_number = -1;
- bool name_match = false;
-
- // Looks for a matching payload name and frequency in the codec list.
- // Need to check both since some codecs have several codec entries with
- // different frequencies (like iSAC).
- for (int i = 0; i < kNumCodecs; i++) {
- if (STR_CASE_CMP(database_[i].plname, codec_inst->plname) == 0) {
- // We have found a matching codec name in the list.
- name_match = true;
-
- // Checks if frequency match.
- if (codec_inst->plfreq == database_[i].plfreq) {
- codec_number = i;
- break;
- }
- }
- }
-
- // Checks if the error is in the name or in the frequency.
- if (codec_number == -1) {
- if (!name_match) {
- return kInvalidCodec;
- } else {
- return kInvalidFrequency;
- }
- }
-
- // Checks the validity of payload type
- if (!ValidPayloadType(codec_inst->pltype)) {
- return kInvalidPayloadtype;
- }
-
- // Comfort Noise is special case, packet-size & rate is not checked.
- if (STR_CASE_CMP(database_[codec_number].plname, "CN") == 0) {
- *mirror_id = codec_number;
- return codec_number;
- }
-
- // RED is special case, packet-size & rate is not checked.
- if (STR_CASE_CMP(database_[codec_number].plname, "red") == 0) {
- *mirror_id = codec_number;
- return codec_number;
- }
-
- // Checks the validity of packet size.
- if (codec_settings_[codec_number].num_packet_sizes > 0) {
- bool packet_size_ok = false;
- int i;
- int packet_size_samples;
- for (i = 0; i < codec_settings_[codec_number].num_packet_sizes; i++) {
- packet_size_samples =
- codec_settings_[codec_number].packet_sizes_samples[i];
- if (codec_inst->pacsize == packet_size_samples) {
- packet_size_ok = true;
- break;
- }
- }
-
- if (!packet_size_ok) {
- return kInvalidPacketSize;
- }
- }
-
- if (codec_inst->pacsize < 1) {
- return kInvalidPacketSize;
- }
-
-
- // Check the validity of rate. Codecs with multiple rates have their own
- // function for this.
- *mirror_id = codec_number;
- if (STR_CASE_CMP("isac", codec_inst->plname) == 0) {
- if (IsISACRateValid(codec_inst->rate)) {
- // Set mirrorID to iSAC WB which is only created once to be used both for
- // iSAC WB and SWB, because they need to share struct.
- *mirror_id = kISAC;
- return codec_number;
- } else {
- return kInvalidRate;
- }
- } else if (STR_CASE_CMP("ilbc", codec_inst->plname) == 0) {
- return IsILBCRateValid(codec_inst->rate, codec_inst->pacsize)
- ? codec_number : kInvalidRate;
- } else if (STR_CASE_CMP("amr", codec_inst->plname) == 0) {
- return IsAMRRateValid(codec_inst->rate)
- ? codec_number : kInvalidRate;
- } else if (STR_CASE_CMP("amr-wb", codec_inst->plname) == 0) {
- return IsAMRwbRateValid(codec_inst->rate)
- ? codec_number : kInvalidRate;
- } else if (STR_CASE_CMP("g7291", codec_inst->plname) == 0) {
- return IsG7291RateValid(codec_inst->rate)
- ? codec_number : kInvalidRate;
- } else if (STR_CASE_CMP("speex", codec_inst->plname) == 0) {
- return IsSpeexRateValid(codec_inst->rate)
- ? codec_number : kInvalidRate;
- } else if (STR_CASE_CMP("celt", codec_inst->plname) == 0) {
- return IsCeltRateValid(codec_inst->rate)
- ? codec_number : kInvalidRate;
- }
-
- return IsRateValid(codec_number, codec_inst->rate) ?
- codec_number : kInvalidRate;
-}
-
-// Gets codec id number, and mirror id, from database for the receiver.
-int ACMCodecDB::ReceiverCodecNumber(const CodecInst* codec_inst,
- int* mirror_id) {
- int codec_number = -1;
-
- // Looks for a matching payload name and frequency in the codec list.
- // Need to check both since some codecs have several codec entries with
- // different frequencies (like iSAC).
- for (int i = 0; i < kNumCodecs; i++) {
- if (STR_CASE_CMP(database_[i].plname, codec_inst->plname) == 0) {
- // We have found a matching codec name in the list.
-
- // Check if frequency match.
- if (codec_inst->plfreq == database_[i].plfreq) {
- codec_number = i;
- *mirror_id = codec_number;
-
- // Check if codec is iSAC, set mirrorID to iSAC WB which is only
- // created once to be used both for iSAC WB and SWB, because they need
- // to share struct.
- if (STR_CASE_CMP(codec_inst->plname, "ISAC") == 0) {
- *mirror_id = kISAC;
- }
-
- break;
- }
- }
- }
-
- return codec_number;
-}
-
-// Returns the codec sampling frequency for codec with id = "codec_id" in
-// database.
-int ACMCodecDB::CodecFreq(int codec_id) {
- // Error check to see that codec_id is not out of bounds.
- if (codec_id < 0 || codec_id >= kNumCodecs) {
- return -1;
- }
-
- return database_[codec_id].plfreq;
-}
-
-// Returns the codec's basic coding block size in samples.
-int ACMCodecDB::BasicCodingBlock(int codec_id) {
- // Error check to see that codec_id is not out of bounds.
- if (codec_id < 0 || codec_id >= kNumCodecs) {
- return -1;
- }
-
- return codec_settings_[codec_id].basic_block_samples;
-}
-
-// Returns the NetEQ decoder database.
-const WebRtcNetEQDecoder* ACMCodecDB::NetEQDecoders() {
- return neteq_decoders_;
-}
-
-// All version numbers for the codecs in the database are listed in text.
-// TODO(tlegrand): change to use std::string.
-int ACMCodecDB::CodecsVersion(char* version, size_t* remaining_buffer_bytes,
- size_t* position) {
- const size_t kTemporaryBufferSize = 500;
- const size_t kVersionBufferSize = 1000;
- char versions_buffer[kVersionBufferSize];
- char version_num_buf[kTemporaryBufferSize];
- size_t len = *position;
- size_t remaining_size = kVersionBufferSize;
-
- versions_buffer[0] = '\0';
-
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- ACM_ISAC_VERSION(version_num_buf);
- strncat(versions_buffer, "ISAC\t\t", remaining_size);
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, version_num_buf, remaining_size);
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, "\n", remaining_size);
-#endif
-#ifdef WEBRTC_CODEC_PCM16
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, "L16\t\t1.0.0\n", remaining_size);
-#endif
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- WebRtcG711_Version(version_num_buf, kTemporaryBufferSize);
- strncat(versions_buffer, "G.711\t\t", remaining_size);
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, version_num_buf, remaining_size);
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, "\n", remaining_size);
-#ifdef WEBRTC_CODEC_ILBC
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- WebRtcIlbcfix_version(version_num_buf);
- strncat(versions_buffer, "ILBC\t\t", remaining_size);
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, version_num_buf, remaining_size);
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, "\n", remaining_size);
-#endif
-#ifdef WEBRTC_CODEC_AMR
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- WebRtcAmr_Version(version_num_buf, kTemporaryBufferSize);
- strncat(versions_buffer, "AMR\t\t", remaining_size);
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, version_num_buf, remaining_size);
-#endif
-#ifdef WEBRTC_CODEC_AMRWB
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- WebRtcAmrWb_Version(version_num_buf, kTemporaryBufferSize);
- strncat(versions_buffer, "AMR-WB\t\t", remaining_size);
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, version_num_buf, remaining_size);
-#endif
-#ifdef WEBRTC_CODEC_G722
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- WebRtcG722_Version(version_num_buf, kTemporaryBufferSize);
- strncat(versions_buffer, "G.722\t\t", remaining_size);
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, version_num_buf, remaining_size);
-#endif
-#ifdef WEBRTC_CODEC_G722_1
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- WebRtcG7221_Version(version_num_buf, kTemporaryBufferSize);
- strncat(versions_buffer, "G.722.1\t\t", remaining_size);
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, version_num_buf, remaining_size);
-#endif
-#ifdef WEBRTC_CODEC_G722_1C
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- WebRtcG7221c_Version(version_num_buf, kTemporaryBufferSize);
- strncat(versions_buffer, "G.722.1C\t", remaining_size);
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, version_num_buf, remaining_size);
-#endif
-#ifdef WEBRTC_CODEC_G729
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- WebRtcG729_Version(version_num_buf, kTemporaryBufferSize);
- strncat(versions_buffer, "G.729\t\t", remaining_size);
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, version_num_buf, remaining_size);
-#endif
-#ifdef WEBRTC_CODEC_G729_1
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- WebRtcG7291_Version(version_num_buf, kTemporaryBufferSize);
- strncat(versions_buffer, "G.729.1\t\t", remaining_size);
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, version_num_buf, remaining_size);
-#endif
-#ifdef WEBRTC_CODEC_GSMFR
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- WebRtcGSMFR_Version(version_num_buf, kTemporaryBufferSize);
- strncat(versions_buffer, "GSM-FR\t\t", remaining_size);
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, version_num_buf, remaining_size);
-#endif
-#ifdef WEBRTC_CODEC_SPEEX
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- WebRtcSpeex_Version(version_num_buf, kTemporaryBufferSize);
- strncat(versions_buffer, "Speex\t\t", remaining_size);
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, version_num_buf, remaining_size);
-#endif
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- WebRtcCng_Version(version_num_buf);
- strncat(versions_buffer, "CNG\t\t", remaining_size);
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, version_num_buf, remaining_size);
-#ifdef WEBRTC_CODEC_AVT
- remaining_size = kVersionBufferSize - strlen(versions_buffer);
- strncat(versions_buffer, "Tone Generation\t1.0.0\n", remaining_size);
-#endif
- strncpy(&version[len], versions_buffer, *remaining_buffer_bytes);
- *position = strlen(version);
- *remaining_buffer_bytes -= (*position - len);
- if (*remaining_buffer_bytes < strlen(versions_buffer)) {
- return -1;
- }
-
- return 0;
-}
-
-// Gets mirror id. The Id is used for codecs sharing struct for settings that
-// need different payload types.
-int ACMCodecDB::MirrorID(int codec_id) {
- if (STR_CASE_CMP(database_[codec_id].plname, "isac") == 0) {
- return kISAC;
- } else {
- return codec_id;
- }
-}
-
-// Creates memory/instance for storing codec state.
-ACMGenericCodec* ACMCodecDB::CreateCodecInstance(const CodecInst* codec_inst) {
- // All we have support for right now.
- if (!STR_CASE_CMP(codec_inst->plname, "ISAC")) {
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
- return new ACMISAC(kISAC);
-#endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "PCMU")) {
- return new ACMPCMU(kPCMU);
- } else if (!STR_CASE_CMP(codec_inst->plname, "PCMA")) {
- return new ACMPCMA(kPCMA);
- } else if (!STR_CASE_CMP(codec_inst->plname, "ILBC")) {
-#ifdef WEBRTC_CODEC_ILBC
- return new ACMILBC(kILBC);
-#endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "AMR")) {
-#ifdef WEBRTC_CODEC_AMR
- return new ACMAMR(kGSMAMR);
-#endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "AMR-WB")) {
-#ifdef WEBRTC_CODEC_AMRWB
- return new ACMAMRwb(kGSMAMRWB);
-#endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "CELT")) {
-#ifdef WEBRTC_CODEC_CELT
- return new ACMCELT(kCELT32);
-#endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "G722")) {
-#ifdef WEBRTC_CODEC_G722
- return new ACMG722(kG722);
-#endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "G7221")) {
- switch (codec_inst->plfreq) {
- case 16000: {
-#ifdef WEBRTC_CODEC_G722_1
- int codec_id;
- switch (codec_inst->rate) {
- case 16000 : {
- codec_id = kG722_1_16;
- break;
- }
- case 24000 : {
- codec_id = kG722_1_24;
- break;
- }
- case 32000 : {
- codec_id = kG722_1_32;
- break;
- }
- default: {
- return NULL;
- }
- return new ACMG722_1(codec_id);
- }
-#endif
- }
- case 32000: {
-#ifdef WEBRTC_CODEC_G722_1C
- int codec_id;
- switch (codec_inst->rate) {
- case 24000 : {
- codec_id = kG722_1C_24;
- break;
- }
- case 32000 : {
- codec_id = kG722_1C_32;
- break;
- }
- case 48000 : {
- codec_id = kG722_1C_48;
- break;
- }
- default: {
- return NULL;
- }
- return new ACMG722_1C(codec_id);
- }
-#endif
- }
- }
- } else if (!STR_CASE_CMP(codec_inst->plname, "CN")) {
- // For CN we need to check sampling frequency to know what codec to create.
- int codec_id;
- switch (codec_inst->plfreq) {
- case 8000: {
- codec_id = kCNNB;
- break;
- }
- case 16000: {
- codec_id = kCNWB;
- break;
- }
- case 32000: {
- codec_id = kCNSWB;
- break;
- }
- default: {
- return NULL;
- }
- }
- return new ACMCNG(codec_id);
- } else if (!STR_CASE_CMP(codec_inst->plname, "G729")) {
-#ifdef WEBRTC_CODEC_G729
- return new ACMG729(kG729);
-#endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "G7291")) {
-#ifdef WEBRTC_CODEC_G729_1
- return new ACMG729_1(kG729_1);
-#endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "speex")) {
-#ifdef WEBRTC_CODEC_SPEEX
- int codec_id;
- switch (codec_inst->plfreq) {
- case 8000: {
- codec_id = kSPEEX8;
- break;
- }
- case 16000: {
- codec_id = kSPEEX16;
- break;
- }
- default: {
- return NULL;
- }
- }
- return new ACMSPEEX(codec_id);
-#endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "CN")) {
- // For CN we need to check sampling frequency to know what codec to create.
- int codec_id;
- switch (codec_inst->plfreq) {
- case 8000: {
- codec_id = kCNNB;
- break;
- }
- case 16000: {
- codec_id = kCNWB;
- break;
- }
- case 32000: {
- codec_id = kCNSWB;
- break;
- }
- default: {
- return NULL;
- }
- }
- return new ACMCNG(codec_id);
- } else if (!STR_CASE_CMP(codec_inst->plname, "L16")) {
-#ifdef WEBRTC_CODEC_PCM16
- // For L16 we need to check sampling frequency to know what codec to create.
- int codec_id;
- switch (codec_inst->plfreq) {
- case 8000: {
- codec_id = kPCM16B;
- break;
- }
- case 16000: {
- codec_id =kPCM16Bwb;
- break;
- }
- case 32000: {
- codec_id = kPCM16Bswb32kHz;
- break;
- }
- default: {
- return NULL;
- }
- }
- return new ACMPCM16B(codec_id);
-#endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "telephone-event")) {
-#ifdef WEBRTC_CODEC_AVT
- return new ACMDTMFPlayout(kAVT);
-#endif
- } else if (!STR_CASE_CMP(codec_inst->plname, "red")) {
-#ifdef WEBRTC_CODEC_RED
- return new ACMRED(kRED);
-#endif
- }
- return NULL;
-}
-
-// Checks if the bitrate is valid for the codec.
-bool ACMCodecDB::IsRateValid(int codec_id, int rate) {
- if (database_[codec_id].rate == rate) {
- return true;
- } else {
- return false;
- }
-}
-
-// Checks if the bitrate is valid for iSAC.
-bool ACMCodecDB::IsISACRateValid(int rate) {
- if ((rate == -1) || ((rate <= 56000) && (rate >= 10000))) {
- return true;
- } else {
- return false;
- }
-}
-
-// Checks if the bitrate is valid for iLBC.
-bool ACMCodecDB::IsILBCRateValid(int rate, int frame_size_samples) {
- if (((frame_size_samples == 240) || (frame_size_samples == 480)) &&
- (rate == 13300)) {
- return true;
- } else if (((frame_size_samples == 160) || (frame_size_samples == 320)) &&
- (rate == 15200)) {
- return true;
- } else {
- return false;
- }
-}
-
-// Check if the bitrate is valid for the GSM-AMR.
-bool ACMCodecDB::IsAMRRateValid(int rate) {
- switch (rate) {
- case 4750:
- case 5150:
- case 5900:
- case 6700:
- case 7400:
- case 7950:
- case 10200:
- case 12200: {
- return true;
- }
- default: {
- return false;
- }
- }
-}
-
-// Check if the bitrate is valid for GSM-AMR-WB.
-bool ACMCodecDB::IsAMRwbRateValid(int rate) {
- switch (rate) {
- case 7000:
- case 9000:
- case 12000:
- case 14000:
- case 16000:
- case 18000:
- case 20000:
- case 23000:
- case 24000: {
- return true;
- }
- default: {
- return false;
- }
- }
-}
-
-// Check if the bitrate is valid for G.729.1.
-bool ACMCodecDB::IsG7291RateValid(int rate) {
- switch (rate) {
- case 8000:
- case 12000:
- case 14000:
- case 16000:
- case 18000:
- case 20000:
- case 22000:
- case 24000:
- case 26000:
- case 28000:
- case 30000:
- case 32000: {
- return true;
- }
- default: {
- return false;
- }
- }
-}
-
-// Checks if the bitrate is valid for Speex.
-bool ACMCodecDB::IsSpeexRateValid(int rate) {
- if (rate > 2000) {
- return true;
- } else {
- return false;
- }
-}
-
-// Checks if the bitrate is valid for Celt.
-bool ACMCodecDB::IsCeltRateValid(int rate) {
- if ((rate >= 48000) && (rate <= 128000)) {
- return true;
- } else {
- return false;
- }
-}
-
-// Checks if the payload type is in the valid range.
-bool ACMCodecDB::ValidPayloadType(int payload_type) {
- if ((payload_type < 0) || (payload_type > 127)) {
- return false;
- }
- return true;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_codec_database.h b/trunk/src/modules/audio_coding/main/source/acm_codec_database.h
deleted file mode 100644
index 6830e65..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_codec_database.h
+++ /dev/null
@@ -1,308 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file generates databases with information about all supported audio
- * codecs.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_
-
-#include "acm_generic_codec.h"
-#include "common_types.h"
-#include "webrtc_neteq.h"
-
-namespace webrtc {
-
-// TODO(tlegrand): replace class ACMCodecDB with a namespace.
-class ACMCodecDB {
- public:
- // Enum with array indexes for the supported codecs. NOTE! The order MUST
- // be the same as when creating the database in acm_codec_database.cc.
- enum {
- kNone = -1
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
- , kISAC
-# if (defined(WEBRTC_CODEC_ISAC))
- , kISACSWB
-# endif
-#endif
-#ifdef WEBRTC_CODEC_PCM16
- , kPCM16B
- , kPCM16Bwb
- , kPCM16Bswb32kHz
-#endif
- , kPCMU
- , kPCMA
-#ifdef WEBRTC_CODEC_ILBC
- , kILBC
-#endif
-#ifdef WEBRTC_CODEC_AMR
- , kGSMAMR
-#endif
-#ifdef WEBRTC_CODEC_AMRWB
- , kGSMAMRWB
-#endif
-#ifdef WEBRTC_CODEC_CELT
- , kCELT32
-#endif
-#ifdef WEBRTC_CODEC_G722
- , kG722
-#endif
-#ifdef WEBRTC_CODEC_G722_1
- , kG722_1_32
- , kG722_1_24
- , kG722_1_16
-#endif
-#ifdef WEBRTC_CODEC_G722_1C
- , kG722_1C_48
- , kG722_1C_32
- , kG722_1C_24
-#endif
-#ifdef WEBRTC_CODEC_G729
- , kG729
-#endif
-#ifdef WEBRTC_CODEC_G729_1
- , kG729_1
-#endif
-#ifdef WEBRTC_CODEC_GSMFR
- , kGSMFR
-#endif
-#ifdef WEBRTC_CODEC_SPEEX
- , kSPEEX8
- , kSPEEX16
-#endif
- , kCNNB
- , kCNWB
- , kCNSWB
-#ifdef WEBRTC_CODEC_AVT
- , kAVT
-#endif
-#ifdef WEBRTC_CODEC_RED
- , kRED
-#endif
- , kNumCodecs
- };
-
- // Set unsupported codecs to -1
-#ifndef WEBRTC_CODEC_ISAC
- enum {kISACSWB = -1};
-# ifndef WEBRTC_CODEC_ISACFX
- enum {kISAC = -1};
-# endif
-#endif
-#ifndef WEBRTC_CODEC_PCM16
- enum {kPCM16B = -1};
- enum {kPCM16Bwb = -1};
- enum {kPCM16Bswb32kHz = -1};
-#endif
- // 48 kHz not supported, always set to -1.
- enum {kPCM16Bswb48kHz = -1};
-#ifndef WEBRTC_CODEC_ILBC
- enum {kILBC = -1};
-#endif
-#ifndef WEBRTC_CODEC_AMR
- enum {kGSMAMR = -1};
-#endif
-#ifndef WEBRTC_CODEC_AMRWB
- enum {kGSMAMRWB = -1};
-#endif
-#ifndef WEBRTC_CODEC_CELT
- enum {kCELT32 = -1};
-#endif
-#ifndef WEBRTC_CODEC_G722
- enum {kG722 = -1};
-#endif
-#ifndef WEBRTC_CODEC_G722_1
- enum {kG722_1_32 = -1};
- enum {kG722_1_24 = -1};
- enum {kG722_1_16 = -1};
-#endif
-#ifndef WEBRTC_CODEC_G722_1C
- enum {kG722_1C_48 = -1};
- enum {kG722_1C_32 = -1};
- enum {kG722_1C_24 = -1};
-#endif
-#ifndef WEBRTC_CODEC_G729
- enum {kG729 = -1};
-#endif
-#ifndef WEBRTC_CODEC_G729_1
- enum {kG729_1 = -1};
-#endif
-#ifndef WEBRTC_CODEC_GSMFR
- enum {kGSMFR = -1};
-#endif
-#ifndef WEBRTC_CODEC_SPEEX
- enum {kSPEEX8 = -1};
- enum {kSPEEX16 = -1};
-#endif
-#ifndef WEBRTC_CODEC_AVT
- enum {kAVT = -1};
-#endif
-#ifndef WEBRTC_CODEC_RED
- enum {kRED = -1};
-#endif
-
- // kMaxNumCodecs - Maximum number of codecs that can be activated in one
- // build.
- // kMaxNumPacketSize - Maximum number of allowed packet sizes for one codec.
- // These might need to be increased if adding a new codec to the database
- static const int kMaxNumCodecs = 50;
- static const int kMaxNumPacketSize = 6;
-
- // Codec specific settings
- //
- // num_packet_sizes - number of allowed packet sizes.
- // packet_sizes_samples - list of the allowed packet sizes.
- // basic_block_samples - assigned a value different from 0 if the codec
- // requires to be fed with a specific number of samples
- // that can be different from packet size.
- // channel_support - number of channels supported to encode;
- // 1 = mono, 2 = stereo, etc.
- struct CodecSettings {
- int num_packet_sizes;
- int packet_sizes_samples[kMaxNumPacketSize];
- int basic_block_samples;
- int channel_support;
- };
-
- // Gets codec information from database at the position in database given by
- // [codec_id].
- // Input:
- // [codec_id] - number that specifies at what position in the database to
- // get the information.
- // Output:
- // [codec_inst] - filled with information about the codec.
- // Return:
- // 0 if successful, otherwise -1.
- static int Codec(int codec_id, CodecInst* codec_inst);
-
- // Returns codec id and mirror id from database, given the information
- // received in the input [codec_inst]. Mirror id is a number that tells
- // where to find the codec's memory (instance). The number is either the
- // same as codec id (most common), or a number pointing at a different
- // entry in the database, if the codec has several entries with different
- // payload types. This is used for codecs that must share one struct even if
- // the payload type differs.
- // One example is the codec iSAC which has the same struct for both 16 and
- // 32 khz, but they have different entries in the database. Let's say the
- // function is called with iSAC 32kHz. The function will return 1 as that is
- // the entry in the data base, and [mirror_id] = 0, as that is the entry for
- // iSAC 16 kHz, which holds the shared memory.
- // Input:
- // [codec_inst] - Information about the codec for which we require the
- // database id.
- // Output:
- // [mirror_id] - mirror id, which most often is the same as the return
- // value, see above.
- // [err_message] - if present, in the event of a mismatch found between the
- // input and the database, a descriptive error message is
- // written here.
- // [err_message] - if present, the length of error message is returned here.
- // Return:
- // codec id if successful, otherwise < 0.
- static int CodecNumber(const CodecInst* codec_inst, int* mirror_id,
- char* err_message, int max_message_len_byte);
- static int CodecNumber(const CodecInst* codec_inst, int* mirror_id);
- static int ReceiverCodecNumber(const CodecInst* codec_inst, int* mirror_id);
-
- // Returns the codec sampling frequency for codec with id = "codec_id" in
- // database.
- // TODO(tlegrand): Check if function is needed, or if we can change
- // to access database directly.
- // Input:
- // [codec_id] - number that specifies at what position in the database to
- // get the information.
- // Return:
- // codec sampling frequency if successful, otherwise -1.
- static int CodecFreq(int codec_id);
-
- // Return the codec's basic coding block size in samples.
- // TODO(tlegrand): Check if function is needed, or if we can change
- // to access database directly.
- // Input:
- // [codec_id] - number that specifies at what position in the database to
- // get the information.
- // Return:
- // codec basic block size if successful, otherwise -1.
- static int BasicCodingBlock(int codec_id);
-
- // Returns the NetEQ decoder database.
- static const WebRtcNetEQDecoder* NetEQDecoders();
-
- // All version numbers for the codecs in the database are listed in text.
- // Input/Output:
- // [version] - pointer to a char vector with minimum size 1000 bytes.
- // Audio coding module's and all component's versions is
- // written here.
- // [remaining_buffer_bytes] - remaining space in buffer.
- // [position] - current position to write at in buffer.
- // Return:
- // -1 if version information doesn't fit, 0 on success.
- static int CodecsVersion(char* version, size_t* remaining_buffer_bytes,
- size_t* position);
-
- // Returns mirror id, which is a number that tells where to find the codec's
- // memory (instance). It is either the same as codec id (most common), or a
- // number pointing at a different entry in the database, if the codec have
- // several entries with different payload types. This is used for codecs that
- // must share struct even if the payload type differs.
- // TODO(tlegrand): Check if function is needed, or if we can change
- // to access database directly.
- // Input:
- // [codec_id] - number that specifies codec's position in the database.
- // Return:
- // Mirror id on success, otherwise -1.
- static int MirrorID(int codec_id);
-
- // Create memory/instance for storing codec state.
- // Input:
- // [codec_inst] - information about codec. Only name of codec, "plname", is
- // used in this function.
- static ACMGenericCodec* CreateCodecInstance(const CodecInst* codec_inst);
-
- // Checks if the bitrate is valid for the codec.
- // Input:
- // [codec_id] - number that specifies codec's position in the database.
- // [rate] - bitrate to check.
- // [frame_size_samples] - (used for iLBC) specifies which frame size to go
- // with the rate.
- static bool IsRateValid(int codec_id, int rate);
- static bool IsISACRateValid(int rate);
- static bool IsILBCRateValid(int rate, int frame_size_samples);
- static bool IsAMRRateValid(int rate);
- static bool IsAMRwbRateValid(int rate);
- static bool IsG7291RateValid(int rate);
- static bool IsSpeexRateValid(int rate);
- static bool IsCeltRateValid(int rate);
-
- // Check if the payload type is valid, meaning that it is in the valid range
- // of 0 to 127.
- // Input:
- // [payload_type] - payload type.
- static bool ValidPayloadType(int payload_type);
-
- // Databases with information about the supported codecs
- // database_ - stored information about all codecs: payload type, name,
- // sampling frequency, packet size in samples, default channel
- // support, and default rate.
- // codec_settings_ - stored codec settings: number of allowed packet sizes,
- // a vector with the allowed packet sizes, basic block
- // samples, and max number of channels that are supported.
- // neteq_decoders_ - list of supported decoders in NetEQ.
- static const CodecInst database_[kMaxNumCodecs];
- static const CodecSettings codec_settings_[kMaxNumCodecs];
- static const WebRtcNetEQDecoder neteq_decoders_[kMaxNumCodecs];
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_CODEC_DATABASE_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_common_defs.h b/trunk/src/modules/audio_coding/main/source/acm_common_defs.h
deleted file mode 100644
index fd8dbd6..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_common_defs.h
+++ /dev/null
@@ -1,115 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_
-
-#include <string.h>
-
-#include "audio_coding_module_typedefs.h"
-#include "common_types.h"
-#include "engine_configurations.h"
-#include "typedefs.h"
-
-// Checks for enabled codecs, we prevent enabling codecs which are not
-// compatible.
-#if ((defined WEBRTC_CODEC_ISAC) && (defined WEBRTC_CODEC_ISACFX))
-#error iSAC and iSACFX codecs cannot be enabled at the same time
-#endif
-
-#ifdef WIN32
-// OS-dependent case-insensitive string comparison
-#define STR_CASE_CMP(x,y) ::_stricmp(x,y)
-#else
-// OS-dependent case-insensitive string comparison
-#define STR_CASE_CMP(x,y) ::strcasecmp(x,y)
-#endif
-
-namespace webrtc {
-
-// 60 ms is the maximum block size we support. An extra 20 ms is considered
-// for safety if process() method is not called when it should be, i.e. we
-// accept 20 ms of jitter. 80 ms @ 32 kHz (super wide-band) is 2560 samples.
-#define AUDIO_BUFFER_SIZE_W16 2560
-
-// There is one timestamp per each 10 ms of audio
-// the audio buffer, at max, may contain 32 blocks of 10ms
-// audio if the sampling frequency is 8000 Hz (80 samples per block).
-// Therefore, The size of the buffer where we keep timestamps
-// is defined as follows
-#define TIMESTAMP_BUFFER_SIZE_W32 (AUDIO_BUFFER_SIZE_W16/80)
-
-// The maximum size of a payload, that is 60 ms of PCM-16 @ 32 kHz stereo
-#define MAX_PAYLOAD_SIZE_BYTE 7680
-
-// General codec specific defines
-const int kIsacWbDefaultRate = 32000;
-const int kIsacSwbDefaultRate = 56000;
-const int kIsacPacSize480 = 480;
-const int kIsacPacSize960 = 960;
-
-// An encoded bit-stream is labeled by one of the following enumerators.
-//
-// kNoEncoding : There has been no encoding.
-// kActiveNormalEncoded : Active audio frame coded by the codec.
-// kPassiveNormalEncoded : Passive audio frame coded by the codec.
-// kPassiveDTXNB : Passive audio frame coded by narrow-band CN.
-// kPassiveDTXWB : Passive audio frame coded by wide-band CN.
-// kPassiveDTXSWB : Passive audio frame coded by super-wide-band CN.
-//
-enum WebRtcACMEncodingType {
- kNoEncoding,
- kActiveNormalEncoded,
- kPassiveNormalEncoded,
- kPassiveDTXNB,
- kPassiveDTXWB,
- kPassiveDTXSWB
-};
-
-// A structure which contains codec parameters. For instance, used when
-// initializing encoder and decoder.
-//
-// codecInstant : c.f. common_types.h
-// enableDTX : set true to enable DTX. If codec does not have
-// internal DTX, this will enable VAD.
-// enableVAD : set true to enable VAD.
-// vadMode : VAD mode, c.f. audio_coding_module_typedefs.h
-// for possible values.
-struct WebRtcACMCodecParams {
- CodecInst codecInstant;
- bool enableDTX;
- bool enableVAD;
- ACMVADMode vadMode;
-};
-
-// A structure that encapsulates audio buffer and related parameters
-// used for synchronization of audio of two ACMs.
-//
-// inAudio : same as ACMGenericCodec::_inAudio
-// inAudioIxRead : same as ACMGenericCodec::_inAudioIxRead
-// inAudioIxWrite : same as ACMGenericCodec::_inAudioIxWrite
-// inTimestamp : same as ACMGenericCodec::_inTimestamp
-// inTimestampIxWrite : same as ACMGenericCodec::_inTImestampIxWrite
-// lastTimestamp : same as ACMGenericCodec::_lastTimestamp
-// lastInTimestamp : same as AudioCodingModuleImpl::_lastInTimestamp
-//
-struct WebRtcACMAudioBuff {
- WebRtc_Word16 inAudio[AUDIO_BUFFER_SIZE_W16];
- WebRtc_Word16 inAudioIxRead;
- WebRtc_Word16 inAudioIxWrite;
- WebRtc_UWord32 inTimestamp[TIMESTAMP_BUFFER_SIZE_W32];
- WebRtc_Word16 inTimestampIxWrite;
- WebRtc_UWord32 lastTimestamp;
- WebRtc_UWord32 lastInTimestamp;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_COMMON_DEFS_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_dtmf_detection.cc b/trunk/src/modules/audio_coding/main/source/acm_dtmf_detection.cc
deleted file mode 100644
index 7669b07..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_dtmf_detection.cc
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_dtmf_detection.h"
-#include "audio_coding_module_typedefs.h"
-
-namespace webrtc {
-
-ACMDTMFDetection::ACMDTMFDetection()
- : _init(0) {}
-
-ACMDTMFDetection::~ACMDTMFDetection() {}
-
-WebRtc_Word16 ACMDTMFDetection::Enable(ACMCountries /* cpt */) {
- return -1;
-}
-
-WebRtc_Word16 ACMDTMFDetection::Disable() {
- return -1;
-}
-
-WebRtc_Word16 ACMDTMFDetection::Detect(
- const WebRtc_Word16* /* inAudioBuff */,
- const WebRtc_UWord16 /* inBuffLenWord16 */,
- const WebRtc_Word32 /* inFreqHz */,
- bool& /* toneDetected */,
- WebRtc_Word16& /* tone */) {
- return -1;
-}
-
-WebRtc_Word16 ACMDTMFDetection::GetVersion(
- WebRtc_Word8* /* version */,
- WebRtc_UWord32& /* remainingBufferInBytes */,
- WebRtc_UWord32& /* position */) {
- return -1;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_dtmf_detection.h b/trunk/src/modules/audio_coding/main/source/acm_dtmf_detection.h
deleted file mode 100644
index 16aba84..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_dtmf_detection.h
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_DETECTION_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_DETECTION_H_
-
-#include "acm_resampler.h"
-#include "audio_coding_module_typedefs.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-class ACMDTMFDetection {
- public:
- ACMDTMFDetection();
- ~ACMDTMFDetection();
- WebRtc_Word16 Enable(ACMCountries cpt = ACMDisableCountryDetection);
- WebRtc_Word16 Disable();
- WebRtc_Word16 Detect(const WebRtc_Word16* inAudioBuff,
- const WebRtc_UWord16 inBuffLenWord16,
- const WebRtc_Word32 inFreqHz,
- bool& toneDetected,
- WebRtc_Word16& tone);
-
- static WebRtc_Word16 GetVersion(WebRtc_Word8* version,
- WebRtc_UWord32& remainingBufferInBytes,
- WebRtc_UWord32& position);
-
- private:
- ACMResampler _resampler;
- bool _init;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_DETECTION_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_dtmf_playout.cc b/trunk/src/modules/audio_coding/main/source/acm_dtmf_playout.cc
deleted file mode 100644
index e748f8a..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_dtmf_playout.cc
+++ /dev/null
@@ -1,185 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_dtmf_playout.h"
-#include "acm_common_defs.h"
-#include "acm_neteq.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-
-namespace webrtc {
-
-#ifndef WEBRTC_CODEC_AVT
-
-ACMDTMFPlayout::ACMDTMFPlayout(
- WebRtc_Word16 /* codecID */) {
- return;
-}
-
-ACMDTMFPlayout::~ACMDTMFPlayout() {
- return;
-}
-
-WebRtc_Word16 ACMDTMFPlayout::InternalEncode(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */) {
- return -1;
-}
-
-WebRtc_Word16 ACMDTMFPlayout::DecodeSafe(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */) {
- return -1;
-}
-
-WebRtc_Word16 ACMDTMFPlayout::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */) {
- return -1;
-}
-
-WebRtc_Word16 ACMDTMFPlayout::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */) {
- return -1;
-}
-
-WebRtc_Word32 ACMDTMFPlayout::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
- const CodecInst& /* codecInst */) {
- return -1;
-}
-
-ACMGenericCodec* ACMDTMFPlayout::CreateInstance(void) {
- return NULL;
-}
-
-WebRtc_Word16 ACMDTMFPlayout::InternalCreateEncoder() {
- return -1;
-}
-
-WebRtc_Word16 ACMDTMFPlayout::InternalCreateDecoder() {
- return -1;
-}
-
-void ACMDTMFPlayout::InternalDestructEncoderInst(void* /* ptrInst */) {
- return;
-}
-
-void ACMDTMFPlayout::DestructEncoderSafe() {
- return;
-}
-
-void ACMDTMFPlayout::DestructDecoderSafe() {
- return;
-}
-
-WebRtc_Word16 ACMDTMFPlayout::UnregisterFromNetEqSafe(
- ACMNetEQ* /* netEq */,
- WebRtc_Word16 /* payloadType */) {
- return -1;
-}
-
-#else //===================== Actual Implementation =======================
-
-ACMDTMFPlayout::ACMDTMFPlayout(WebRtc_Word16 codecID) {
- _codecID = codecID;
-}
-
-ACMDTMFPlayout::~ACMDTMFPlayout() {
- return;
-}
-
-WebRtc_Word16 ACMDTMFPlayout::InternalEncode(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */) {
- return 0;
-}
-
-WebRtc_Word16 ACMDTMFPlayout::DecodeSafe(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */) {
- return 0;
-}
-
-WebRtc_Word16 ACMDTMFPlayout::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */) {
- // This codec does not need initialization,
- // DTMFPlayout has no instance
- return 0;
-}
-
-WebRtc_Word16 ACMDTMFPlayout::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */) {
- // This codec does not need initialization,
- // DTMFPlayout has no instance
- return 0;
-}
-
-WebRtc_Word32 ACMDTMFPlayout::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst) {
- // Fill up the structure by calling
- // "SET_CODEC_PAR" & "SET_AVT_FUNCTION."
- // Then call NetEQ to add the codec to it's
- // database.
- SET_CODEC_PAR((codecDef), kDecoderAVT, codecInst.pltype, NULL, 8000);
- SET_AVT_FUNCTIONS((codecDef));
- return 0;
-}
-
-ACMGenericCodec* ACMDTMFPlayout::CreateInstance(void) {
- return NULL;
-}
-
-WebRtc_Word16 ACMDTMFPlayout::InternalCreateEncoder() {
- // DTMFPlayout has no instance
- return 0;
-}
-
-WebRtc_Word16 ACMDTMFPlayout::InternalCreateDecoder() {
- // DTMFPlayout has no instance
- return 0;
-}
-
-void ACMDTMFPlayout::InternalDestructEncoderInst(void* /* ptrInst */) {
- // DTMFPlayout has no instance
- return;
-}
-
-void ACMDTMFPlayout::DestructEncoderSafe() {
- // DTMFPlayout has no instance
- return;
-}
-
-void ACMDTMFPlayout::DestructDecoderSafe() {
- // DTMFPlayout has no instance
- return;
-}
-
-WebRtc_Word16 ACMDTMFPlayout::UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType) {
- if (payloadType != _decoderParams.codecInstant.pltype) {
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceAudioCoding,
- _uniqueID,
- "Cannot unregister codec %s given payload-type %d does not "
- "match the stored payload type",
- _decoderParams.codecInstant.plname, payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
- return netEq->RemoveCodec(kDecoderAVT);
-}
-#endif
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_dtmf_playout.h b/trunk/src/modules/audio_coding/main/source/acm_dtmf_playout.h
deleted file mode 100644
index a75d691..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_dtmf_playout.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_
-
-#include "acm_generic_codec.h"
-
-namespace webrtc {
-
-class ACMDTMFPlayout: public ACMGenericCodec {
- public:
- ACMDTMFPlayout(WebRtc_Word16 codecID);
- ~ACMDTMFPlayout();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
-
- protected:
- WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio, WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- WebRtc_Word16 UnregisterFromNetEqSafe(ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- void InternalDestructEncoderInst(void* ptrInst);
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_DTMF_PLAYOUT_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g722.cc b/trunk/src/modules/audio_coding/main/source/acm_g722.cc
deleted file mode 100644
index 40e431c..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_g722.cc
+++ /dev/null
@@ -1,337 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_g722.h"
-#include "acm_codec_database.h"
-#include "acm_common_defs.h"
-#include "acm_neteq.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-#include "g722_interface.h"
-
-namespace webrtc {
-
-#ifndef WEBRTC_CODEC_G722
-
-ACMG722::ACMG722(WebRtc_Word16 /* codecID */)
- : _ptrEncStr(NULL),
- _ptrDecStr(NULL),
- _encoderInstPtr(NULL),
- _encoderInstPtrRight(NULL),
- _decoderInstPtr(NULL) {
- return;
-}
-
-ACMG722::~ACMG722() {
- return;
-}
-
-WebRtc_Word32 ACMG722::Add10MsDataSafe(const WebRtc_UWord32 /* timestamp */,
- const WebRtc_Word16* /* data */,
- const WebRtc_UWord16 /* lengthSmpl */,
- const WebRtc_UWord8 /* audioChannel */) {
- return -1;
-}
-
-WebRtc_Word16 ACMG722::InternalEncode(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */) {
- return -1;
-}
-
-WebRtc_Word16 ACMG722::DecodeSafe(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */) {
- return -1;
-}
-
-WebRtc_Word16 ACMG722::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */) {
- return -1;
-}
-
-WebRtc_Word16 ACMG722::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */) {
- return -1;
-}
-
-WebRtc_Word32 ACMG722::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
- const CodecInst& /* codecInst */) {
- return -1;
-}
-
-ACMGenericCodec* ACMG722::CreateInstance(void) {
- return NULL;
-}
-
-WebRtc_Word16 ACMG722::InternalCreateEncoder() {
- return -1;
-}
-
-void ACMG722::DestructEncoderSafe() {
- return;
-}
-
-WebRtc_Word16 ACMG722::InternalCreateDecoder() {
- return -1;
-}
-
-void ACMG722::DestructDecoderSafe() {
- return;
-}
-
-void ACMG722::InternalDestructEncoderInst(void* /* ptrInst */) {
- return;
-}
-
-WebRtc_Word16 ACMG722::UnregisterFromNetEqSafe(
- ACMNetEQ* /* netEq */,
- WebRtc_Word16 /* payloadType */) {
- return -1;
-}
-
-#else //===================== Actual Implementation =======================
-
-// Encoder and decoder memory
-struct ACMG722EncStr {
- G722EncInst* inst; // instance for left channel in case of stereo
- G722EncInst* instRight; // instance for right channel in case of stereo
-};
-struct ACMG722DecStr {
- G722DecInst* inst; // instance for left channel in case of stereo
- G722DecInst* instRight; // instance for right channel in case of stereo
-};
-
-ACMG722::ACMG722(WebRtc_Word16 codecID)
- : _encoderInstPtr(NULL),
- _encoderInstPtrRight(NULL),
- _decoderInstPtr(NULL) {
- // Encoder
- _ptrEncStr = new ACMG722EncStr;
- if (_ptrEncStr != NULL) {
- _ptrEncStr->inst = NULL;
- _ptrEncStr->instRight = NULL;
- }
- // Decoder
- _ptrDecStr = new ACMG722DecStr;
- if (_ptrDecStr != NULL) {
- _ptrDecStr->inst = NULL;
- _ptrDecStr->instRight = NULL; // Not used
- }
- _codecID = codecID;
- return;
-}
-
-ACMG722::~ACMG722() {
- // Encoder
- if (_ptrEncStr != NULL) {
- if (_ptrEncStr->inst != NULL) {
- WebRtcG722_FreeEncoder(_ptrEncStr->inst);
- _ptrEncStr->inst = NULL;
- }
- if (_ptrEncStr->instRight != NULL) {
- WebRtcG722_FreeEncoder(_ptrEncStr->instRight);
- _ptrEncStr->instRight = NULL;
- }
- delete _ptrEncStr;
- _ptrEncStr = NULL;
- }
- // Decoder
- if (_ptrDecStr != NULL) {
- if (_ptrDecStr->inst != NULL) {
- WebRtcG722_FreeDecoder(_ptrDecStr->inst);
- _ptrDecStr->inst = NULL;
- }
- if (_ptrDecStr->instRight != NULL) {
- WebRtcG722_FreeDecoder(_ptrDecStr->instRight);
- _ptrDecStr->instRight = NULL;
- }
- delete _ptrDecStr;
- _ptrDecStr = NULL;
- }
- return;
-}
-
-WebRtc_Word32 ACMG722::Add10MsDataSafe(const WebRtc_UWord32 timestamp,
- const WebRtc_Word16* data,
- const WebRtc_UWord16 lengthSmpl,
- const WebRtc_UWord8 audioChannel) {
- return ACMGenericCodec::Add10MsDataSafe((timestamp >> 1), data, lengthSmpl,
- audioChannel);
-}
-
-WebRtc_Word16 ACMG722::InternalEncode(WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte) {
- // If stereo, split input signal in left and right channel before encoding
- if (_noChannels == 2) {
- WebRtc_Word16 leftChannel[960];
- WebRtc_Word16 rightChannel[960];
- WebRtc_UWord8 outLeft[480];
- WebRtc_UWord8 outRight[480];
- WebRtc_Word16 lenInBytes;
- for (int i = 0, j = 0; i < _frameLenSmpl * 2; i += 2, j++) {
- leftChannel[j] = _inAudio[_inAudioIxRead + i];
- rightChannel[j] = _inAudio[_inAudioIxRead + i + 1];
- }
- lenInBytes = WebRtcG722_Encode(_encoderInstPtr, leftChannel, _frameLenSmpl,
- (WebRtc_Word16*) outLeft);
- lenInBytes += WebRtcG722_Encode(_encoderInstPtrRight, rightChannel,
- _frameLenSmpl, (WebRtc_Word16*) outRight);
- *bitStreamLenByte = lenInBytes;
-
- // Interleave the 4 bits per sample from left and right channel
- for (int i = 0, j = 0; i < lenInBytes; i += 2, j++) {
- bitStream[i] = (outRight[j] & 0xF0) + (outLeft[j] >> 4);
- bitStream[i + 1] = ((outRight[j] & 0x0F) << 4) + (outLeft[j] & 0x0F);
- }
- } else {
- *bitStreamLenByte = WebRtcG722_Encode(_encoderInstPtr,
- &_inAudio[_inAudioIxRead],
- _frameLenSmpl,
- (WebRtc_Word16*) bitStream);
- }
-
- // increment the read index this tell the caller how far
- // we have gone forward in reading the audio buffer
- _inAudioIxRead += _frameLenSmpl * _noChannels;
- return *bitStreamLenByte;
-}
-
-WebRtc_Word16 ACMG722::DecodeSafe(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */) {
- return 0;
-}
-
-WebRtc_Word16 ACMG722::InternalInitEncoder(WebRtcACMCodecParams* codecParams) {
- if (codecParams->codecInstant.channels == 2) {
- // Create codec struct for right channel
- if (_ptrEncStr->instRight == NULL) {
- WebRtcG722_CreateEncoder(&_ptrEncStr->instRight);
- if (_ptrEncStr->instRight == NULL) {
- return -1;
- }
- }
- _encoderInstPtrRight = _ptrEncStr->instRight;
- if (WebRtcG722_EncoderInit(_encoderInstPtrRight) < 0) {
- return -1;
- }
- }
-
- return WebRtcG722_EncoderInit(_encoderInstPtr);
-}
-
-WebRtc_Word16 ACMG722::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */) {
- return WebRtcG722_DecoderInit(_decoderInstPtr);
-}
-
-WebRtc_Word32 ACMG722::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst) {
- if (!_decoderInitialized) {
- // TODO: log error
- return -1;
- }
- // Fill up the structure by calling
- // "SET_CODEC_PAR" & "SET_G722_FUNCTION."
- // Then call NetEQ to add the codec to it's
- // database.
- SET_CODEC_PAR((codecDef), kDecoderG722, codecInst.pltype, _decoderInstPtr,
- 16000);
- SET_G722_FUNCTIONS((codecDef));
- return 0;
-}
-
-ACMGenericCodec* ACMG722::CreateInstance(void) {
- return NULL;
-}
-
-WebRtc_Word16 ACMG722::InternalCreateEncoder() {
- if (_ptrEncStr == NULL) {
- // this structure must be created at the costructor
- // if it is still NULL then there is a probelm and
- // we dont continue
- return -1;
- }
- WebRtcG722_CreateEncoder(&_ptrEncStr->inst);
- if (_ptrEncStr->inst == NULL) {
- return -1;
- }
- _encoderInstPtr = _ptrEncStr->inst;
- return 0;
-}
-
-void ACMG722::DestructEncoderSafe() {
- if (_ptrEncStr != NULL) {
- if (_ptrEncStr->inst != NULL) {
- WebRtcG722_FreeEncoder(_ptrEncStr->inst);
- _ptrEncStr->inst = NULL;
- }
- }
- _encoderExist = false;
- _encoderInitialized = false;
-}
-
-WebRtc_Word16 ACMG722::InternalCreateDecoder() {
- if (_ptrDecStr == NULL) {
- // this structure must be created at the costructor
- // if it is still NULL then there is a probelm and
- // we dont continue
- return -1;
- }
-
- WebRtcG722_CreateDecoder(&_ptrDecStr->inst);
- if (_ptrDecStr->inst == NULL) {
- return -1;
- }
- _decoderInstPtr = _ptrDecStr->inst;
- return 0;
-}
-
-void ACMG722::DestructDecoderSafe() {
- _decoderExist = false;
- _decoderInitialized = false;
- if (_ptrDecStr != NULL) {
- if (_ptrDecStr->inst != NULL) {
- WebRtcG722_FreeDecoder(_ptrDecStr->inst);
- _ptrDecStr->inst = NULL;
- }
- }
-}
-
-void ACMG722::InternalDestructEncoderInst(void* ptrInst) {
- if (ptrInst != NULL) {
- WebRtcG722_FreeEncoder(static_cast<G722EncInst*>(ptrInst));
- }
- return;
-}
-
-WebRtc_Word16 ACMG722::UnregisterFromNetEqSafe(ACMNetEQ* netEq,
- WebRtc_Word16 payloadType) {
- if (payloadType != _decoderParams.codecInstant.pltype) {
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceAudioCoding,
- _uniqueID,
- "Cannot unregister codec %s given payload-type %d does not "
- "match the stored payload type",
- _decoderParams.codecInstant.plname, payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
- return netEq->RemoveCodec(kDecoderG722);
-}
-
-#endif
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g722.h b/trunk/src/modules/audio_coding/main/source/acm_g722.h
deleted file mode 100644
index 776d188..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_g722.h
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_
-
-#include "acm_generic_codec.h"
-
-typedef struct WebRtcG722EncInst G722EncInst;
-typedef struct WebRtcG722DecInst G722DecInst;
-
-namespace webrtc {
-
-// forward declaration
-struct ACMG722EncStr;
-struct ACMG722DecStr;
-
-class ACMG722: public ACMGenericCodec {
- public:
- ACMG722(WebRtc_Word16 codecID);
- ~ACMG722();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
-
- protected:
- WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio, WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
- WebRtc_Word32 Add10MsDataSafe(const WebRtc_UWord32 timestamp,
- const WebRtc_Word16* data,
- const WebRtc_UWord16 lengthSmpl,
- const WebRtc_UWord8 audioChannel);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- void InternalDestructEncoderInst(void* ptrInst);
-
- WebRtc_Word16 UnregisterFromNetEqSafe(ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- ACMG722EncStr* _ptrEncStr;
- ACMG722DecStr* _ptrDecStr;
-
- G722EncInst* _encoderInstPtr;
- G722EncInst* _encoderInstPtrRight; // Prepared for stereo
- G722DecInst* _decoderInstPtr;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g7221.cc b/trunk/src/modules/audio_coding/main/source/acm_g7221.cc
deleted file mode 100644
index b82d931..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_g7221.cc
+++ /dev/null
@@ -1,528 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_g7221.h"
-#include "acm_codec_database.h"
-#include "acm_common_defs.h"
-#include "acm_neteq.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-
-#ifdef WEBRTC_CODEC_G722_1
-// NOTE! G.722.1 is not included in the open-source package. The following
-// interface file is needed:
-//
-// /modules/audio_coding/codecs/g7221/main/interface/g7221_interface.h
-//
-// The API in the header file should match the one below.
-//
-// int16_t WebRtcG7221_CreateEnc16(G722_1_16_encinst_t_** encInst);
-// int16_t WebRtcG7221_CreateEnc24(G722_1_24_encinst_t_** encInst);
-// int16_t WebRtcG7221_CreateEnc32(G722_1_32_encinst_t_** encInst);
-// int16_t WebRtcG7221_CreateDec16(G722_1_16_decinst_t_** decInst);
-// int16_t WebRtcG7221_CreateDec24(G722_1_24_decinst_t_** decInst);
-// int16_t WebRtcG7221_CreateDec32(G722_1_32_decinst_t_** decInst);
-//
-// int16_t WebRtcG7221_FreeEnc16(G722_1_16_encinst_t_** encInst);
-// int16_t WebRtcG7221_FreeEnc24(G722_1_24_encinst_t_** encInst);
-// int16_t WebRtcG7221_FreeEnc32(G722_1_32_encinst_t_** encInst);
-// int16_t WebRtcG7221_FreeDec16(G722_1_16_decinst_t_** decInst);
-// int16_t WebRtcG7221_FreeDec24(G722_1_24_decinst_t_** decInst);
-// int16_t WebRtcG7221_FreeDec32(G722_1_32_decinst_t_** decInst);
-//
-// int16_t WebRtcG7221_EncoderInit16(G722_1_16_encinst_t_* encInst);
-// int16_t WebRtcG7221_EncoderInit24(G722_1_24_encinst_t_* encInst);
-// int16_t WebRtcG7221_EncoderInit32(G722_1_32_encinst_t_* encInst);
-// int16_t WebRtcG7221_DecoderInit16(G722_1_16_decinst_t_* decInst);
-// int16_t WebRtcG7221_DecoderInit24(G722_1_24_decinst_t_* decInst);
-// int16_t WebRtcG7221_DecoderInit32(G722_1_32_decinst_t_* decInst);
-//
-// int16_t WebRtcG7221_Encode16(G722_1_16_encinst_t_* encInst,
-// int16_t* input,
-// int16_t len,
-// int16_t* output);
-// int16_t WebRtcG7221_Encode24(G722_1_24_encinst_t_* encInst,
-// int16_t* input,
-// int16_t len,
-// int16_t* output);
-// int16_t WebRtcG7221_Encode32(G722_1_32_encinst_t_* encInst,
-// int16_t* input,
-// int16_t len,
-// int16_t* output);
-//
-// int16_t WebRtcG7221_Decode16(G722_1_16_decinst_t_* decInst,
-// int16_t* bitstream,
-// int16_t len,
-// int16_t* output);
-// int16_t WebRtcG7221_Decode24(G722_1_24_decinst_t_* decInst,
-// int16_t* bitstream,
-// int16_t len,
-// int16_t* output);
-// int16_t WebRtcG7221_Decode32(G722_1_32_decinst_t_* decInst,
-// int16_t* bitstream,
-// int16_t len,
-// int16_t* output);
-//
-// int16_t WebRtcG7221_DecodePlc16(G722_1_16_decinst_t_* decInst,
-// int16_t* output,
-// int16_t nrLostFrames);
-// int16_t WebRtcG7221_DecodePlc24(G722_1_24_decinst_t_* decInst,
-// int16_t* output,
-// int16_t nrLostFrames);
-// int16_t WebRtcG7221_DecodePlc32(G722_1_32_decinst_t_* decInst,
-// int16_t* output,
-// int16_t nrLostFrames);
-// void WebRtcG7221_Version(char *versionStr, short len);
-#include "g7221_interface.h"
-#endif
-
-namespace webrtc {
-
-#ifndef WEBRTC_CODEC_G722_1
-
-ACMG722_1::ACMG722_1(WebRtc_Word16 /* codecID */)
- : _operationalRate(-1),
- _encoderInstPtr(NULL),
- _encoderInstPtrRight(NULL),
- _decoderInstPtr(NULL),
- _encoderInst16Ptr(NULL),
- _encoderInst16PtrR(NULL),
- _encoderInst24Ptr(NULL),
- _encoderInst24PtrR(NULL),
- _encoderInst32Ptr(NULL),
- _encoderInst32PtrR(NULL),
- _decoderInst16Ptr(NULL),
- _decoderInst24Ptr(NULL),
- _decoderInst32Ptr(NULL) {
- return;
-}
-
-ACMG722_1::~ACMG722_1() {
- return;
-}
-
-WebRtc_Word16 ACMG722_1::InternalEncode(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */) {
- return -1;
-}
-
-WebRtc_Word16 ACMG722_1::DecodeSafe(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */) {
- return -1;
-}
-
-WebRtc_Word16 ACMG722_1::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */) {
- return -1;
-}
-
-WebRtc_Word16 ACMG722_1::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */) {
- return -1;
-}
-
-WebRtc_Word32 ACMG722_1::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
- const CodecInst& /* codecInst */) {
- return -1;
-}
-
-ACMGenericCodec* ACMG722_1::CreateInstance(void) {
- return NULL;
-}
-
-WebRtc_Word16 ACMG722_1::InternalCreateEncoder() {
- return -1;
-}
-
-void ACMG722_1::DestructEncoderSafe() {
- return;
-}
-
-WebRtc_Word16 ACMG722_1::InternalCreateDecoder() {
- return -1;
-}
-
-void ACMG722_1::DestructDecoderSafe() {
- return;
-}
-
-void ACMG722_1::InternalDestructEncoderInst(void* /* ptrInst */) {
- return;
-}
-
-WebRtc_Word16 ACMG722_1::UnregisterFromNetEqSafe(
- ACMNetEQ* /* netEq */, WebRtc_Word16 /* payloadType */) {
- return -1;
-}
-
-#else //===================== Actual Implementation =======================
-ACMG722_1::ACMG722_1(
- WebRtc_Word16 codecID):
- _encoderInstPtr(NULL),
- _encoderInstPtrRight(NULL),
- _decoderInstPtr(NULL),
- _encoderInst16Ptr(NULL),
- _encoderInst16PtrR(NULL),
- _encoderInst24Ptr(NULL),
- _encoderInst24PtrR(NULL),
- _encoderInst32Ptr(NULL),
- _encoderInst32PtrR(NULL),
- _decoderInst16Ptr(NULL),
- _decoderInst24Ptr(NULL),
- _decoderInst32Ptr(NULL) {
- _codecID = codecID;
- if (_codecID == ACMCodecDB::kG722_1_16) {
- _operationalRate = 16000;
- } else if (_codecID == ACMCodecDB::kG722_1_24) {
- _operationalRate = 24000;
- } else if (_codecID == ACMCodecDB::kG722_1_32) {
- _operationalRate = 32000;
- } else {
- _operationalRate = -1;
- }
- return;
-}
-
-ACMG722_1::~ACMG722_1() {
- if (_encoderInstPtr != NULL) {
- delete _encoderInstPtr;
- _encoderInstPtr = NULL;
- }
- if (_encoderInstPtrRight != NULL) {
- delete _encoderInstPtrRight;
- _encoderInstPtrRight = NULL;
- }
- if (_decoderInstPtr != NULL) {
- delete _decoderInstPtr;
- _decoderInstPtr = NULL;
- }
-
- switch (_operationalRate) {
- case 16000: {
- _encoderInst16Ptr = NULL;
- _encoderInst16PtrR = NULL;
- _decoderInst16Ptr = NULL;
- break;
- }
- case 24000: {
- _encoderInst24Ptr = NULL;
- _encoderInst24PtrR = NULL;
- _decoderInst24Ptr = NULL;
- break;
- }
- case 32000: {
- _encoderInst32Ptr = NULL;
- _encoderInst32PtrR = NULL;
- _decoderInst32Ptr = NULL;
- break;
- }
- default: {
- break;
- }
- }
- return;
-}
-
-WebRtc_Word16 ACMG722_1::InternalEncode(WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte) {
- WebRtc_Word16 leftChannel[320];
- WebRtc_Word16 rightChannel[320];
- WebRtc_Word16 lenInBytes;
- WebRtc_Word16 outB[160];
-
- // If stereo, split input signal in left and right channel before encoding
- if (_noChannels == 2) {
- for (int i = 0, j = 0; i < _frameLenSmpl * 2; i += 2, j++) {
- leftChannel[j] = _inAudio[_inAudioIxRead + i];
- rightChannel[j] = _inAudio[_inAudioIxRead + i + 1];
- }
- } else {
- memcpy(leftChannel, &_inAudio[_inAudioIxRead], 320);
- }
-
- switch (_operationalRate) {
- case 16000: {
- Inst lenInBytes = WebRtcG7221_Encode16(_encoderInst16Ptr, leftChannel,
- 320, &outB[0]);
- if (_noChannels == 2) {
- lenInBytes += WebRtcG7221_Encode16(_encoderInst16PtrR, rightChannel,
- 320, &outB[lenInBytes / 2]);
- }
- break;
- }
- case 24000: {
- lenInBytes = WebRtcG7221_Encode24(_encoderInst24Ptr, leftChannel, 320,
- &outB[0]);
- if (_noChannels == 2) {
- lenInBytes += WebRtcG7221_Encode24(_encoderInst24PtrR, rightChannel,
- 320, &outB[lenInBytes / 2]);
- }
- break;
- }
- case 32000: {
- lenInBytes = WebRtcG7221_Encode32(_encoderInst32Ptr, leftChannel, 320,
- &outB[0]);
- if (_noChannels == 2) {
- lenInBytes += WebRtcG7221_Encode32(_encoderInst32PtrR, rightChannel,
- 320, &outB[lenInBytes / 2]);
- }
- break;
- }
- default: {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalInitEncode: Wrong rate for G722_1.");
- return -1;
- }
- }
- memcpy(bitStream, outB, lenInBytes);
- *bitStreamLenByte = lenInBytes;
-
- // increment the read index this tell the caller that how far
- // we have gone forward in reading the audio buffer
- _inAudioIxRead += 320 * _noChannels;
- return *bitStreamLenByte;
-}
-
-WebRtc_Word16 ACMG722_1::DecodeSafe(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */) {
- return 0;
-}
-
-WebRtc_Word16 ACMG722_1::InternalInitEncoder(
- WebRtcACMCodecParams* codecParams) {
- WebRtc_Word16 ret;
-
- switch (_operationalRate) {
- case 16000: {
- ret = WebRtcG7221_EncoderInit16(_encoderInst16PtrR);
- if (ret < 0) {
- return ret;
- }
- return WebRtcG7221_EncoderInit16(_encoderInst16Ptr);
- }
- case 24000: {
- ret = WebRtcG7221_EncoderInit24(_encoderInst24PtrR);
- if (ret < 0) {
- return ret;
- }
- return WebRtcG7221_EncoderInit24(_encoderInst24Ptr);
- }
- case 32000: {
- ret = WebRtcG7221_EncoderInit32(_encoderInst32PtrR);
- if (ret < 0) {
- return ret;
- }
- return WebRtcG7221_EncoderInit32(_encoderInst32Ptr);
- }
- default: {
- WEBRTC_TRACE(webrtc::kTraceError,Inst webrtc::kTraceAudioCoding,
- _uniqueID, "InternalInitEncoder: Wrong rate for G722_1.");
- return -1;
- }
- }
-}
-
-WebRtc_Word16 ACMG722_1::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */) {
- switch (_operationalRate) {
- case 16000: {
- return WebRtcG7221_DecoderInit16(_decoderInst16Ptr);
- }
- case 24000: {
- return WebRtcG7221_DecoderInit24(_decoderInst24Ptr);
- }
- case 32000: {
- return WebRtcG7221_DecoderInit32(_decoderInst32Ptr);
- }
- default: {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalInitDecoder: Wrong rate for G722_1.");
- return -1;
- }
- }
-}
-
-WebRtc_Word32 ACMG722_1::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst) {
- if (!_decoderInitialized) {
- // Todo:
- // log error
- return -1;
- }
- // NetEq has an array of pointers to WebRtcNetEQ_CodecDef.
- // Get an entry of that array (neteq wrapper will allocate memory)
- // by calling "netEq->CodecDef", where "NETEQ_CODEC_G722_1_XX" would
- // be the index of the entry.
- // Fill up the given structure by calling
- // "SET_CODEC_PAR" & "SET_G722_1_XX_FUNCTION."
- // Then return the structure back to NetEQ to add the codec to it's
- // database.
- switch (_operationalRate) {
- case 16000: {
- SET_CODEC_PAR((codecDef), kDecoderG722_1_16, codecInst.pltype,
- _decoderInst16Ptr, 16000);
- SET_G722_1_16_FUNCTIONS((codecDef));
- break;
- }
- case 24000: {
- SET_CODEC_PAR((codecDef), kDecoderG722_1_24, codecInst.pltype,
- _decoderInst24Ptr, 16000);
- SET_G722_1_24_FUNCTIONS((codecDef));
- break;
- }
- case 32000: {
- SET_CODEC_PAR((codecDef), kDecoderG722_1_32, codecInst.pltype,
- _decoderInst32Ptr, 16000);
- SET_G722_1_32_FUNCTIONS((codecDef));
- break;
- }
- default: {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "CodecDef: Wrong rate for G722_1.");
- return -1;
- }
- }
- return 0;
-}
-
-ACMGenericCodec* ACMG722_1::CreateInstance(void) {
- return NULL;
-}
-
-WebRtc_Word16 ACMG722_1::InternalCreateEncoder() {
- if ((_encoderInstPtr == NULL) || (_encoderInstPtrRight == NULL)) {
- return -1;
- }
- switch (_operationalRate) {
- case 16000: {
- WebRtcG7221_CreateEnc16(&_encoderInst16Ptr);
- WebRtcG7221_CreateEnc16(&_encoderInst16PtrR);
- break;
- }
- case 24000: {
- WebRtcG7221_CreateEnc24(&_encoderInst24Ptr);
- WebRtcG7221_CreateEnc24(&_encoderInst24PtrR);
- break;
- }
- case 32000: {
- WebRtcG7221_CreateEnc32(&_encoderInst32Ptr);
- WebRtcG7221_CreateEnc32(&_encoderInst32PtrR);
- break;
- }
- default: {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalCreateEncoder: Wrong rate for G722_1.");
- return -1;
- }
- }
- return 0;
-}
-
-void ACMG722_1::DestructEncoderSafe() {
- _encoderExist = false;
- _encoderInitialized = false;
- if (_encoderInstPtr != NULL) {
- delete _encoderInstPtr;
- _encoderInstPtr = NULL;
- }
- if (_encoderInstPtrRight != NULL) {
- delete _encoderInstPtrRight;
- _encoderInstPtrRight = NULL;
- }
- _encoderInst16Ptr = NULL;
- _encoderInst24Ptr = NULL;
- _encoderInst32Ptr = NULL;
-}
-
-WebRtc_Word16 ACMG722_1::InternalCreateDecoder() {
- if (_decoderInstPtr == NULL) {
- return -1;
- }
- switch (_operationalRate) {
- case 16000: {
- WebRtcG7221_CreateDec16(&_decoderInst16Ptr);
- break;
- }
- case 24000: {
- WebRtcG7221_CreateDec24(&_decoderInst24Ptr);
- break;
- }
- case 32000: {
- WebRtcG7221_CreateDec32(&_decoderInst32Ptr);
- break;
- }
- default: {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalCreateDecoder: Wrong rate for G722_1.");
- return -1;
- }
- }
- return 0;
-}
-
-void ACMG722_1::DestructDecoderSafe() {
- _decoderExist = false;
- _decoderInitialized = false;
- if (_decoderInstPtr != NULL) {
- delete _decoderInstPtr;
- _decoderInstPtr = NULL;
- }
- _decoderInst16Ptr = NULL;
- _decoderInst24Ptr = NULL;
- _decoderInst32Ptr = NULL;
-}
-
-void ACMG722_1::InternalDestructEncoderInst(void* ptrInst) {
- if (ptrInst != NULL) {
- delete ptrInst;
- }
- return;
-}
-
-WebRtc_Word16 ACMG722_1::UnregisterFromNetEqSafe(ACMNetEQ* netEq,
- WebRtc_Word16 payloadType) {
- if (payloadType != _decoderParams.codecInstant.pltype) {
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceAudioCoding,
- _uniqueID,
- "Cannot unregister codec %s given payload-type %d does not "
- "match the stored payload type",
- _decoderParams.codecInstant.plname, payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
- switch (_operationalRate) {
- case 16000: {
- return netEq->RemoveCodec(kDecoderG722_1_16);
- }
- case 24000: {
- return netEq->RemoveCodec(kDecoderG722_1_24);
- }
- case 32000: {
- return netEq->RemoveCodec(kDecoderG722_1_32);
- }
- default: {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "UnregisterFromNetEqSafe: Wrong rate for G722_1.");
- return -1;
- }
- }
-}
-
-#endif
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g7221.h b/trunk/src/modules/audio_coding/main/source/acm_g7221.h
deleted file mode 100644
index 3130afd..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_g7221.h
+++ /dev/null
@@ -1,85 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_1_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_1_H_
-
-#include "acm_generic_codec.h"
-
-// forward declaration
-struct G722_1_16_encinst_t_;
-struct G722_1_16_decinst_t_;
-struct G722_1_24_encinst_t_;
-struct G722_1_24_decinst_t_;
-struct G722_1_32_encinst_t_;
-struct G722_1_32_decinst_t_;
-struct G722_1_Inst_t_;
-
-namespace webrtc {
-
-class ACMG722_1: public ACMGenericCodec {
- public:
- ACMG722_1(WebRtc_Word16 codecID);
- ~ACMG722_1();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 InternalInitDecoder(WebRtcACMCodecParams *codecParams);
-
- protected:
- WebRtc_Word16 DecodeSafe(WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio, WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- void InternalDestructEncoderInst(void* ptrInst);
-
- WebRtc_Word16 UnregisterFromNetEqSafe(ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- WebRtc_Word32 _operationalRate;
-
- G722_1_Inst_t_* _encoderInstPtr;
- G722_1_Inst_t_* _encoderInstPtrRight; //Used in stereo mode
- G722_1_Inst_t_* _decoderInstPtr;
-
- // Only one set of these pointer is valid at any instance
- G722_1_16_encinst_t_* _encoderInst16Ptr;
- G722_1_16_encinst_t_* _encoderInst16PtrR;
- G722_1_24_encinst_t_* _encoderInst24Ptr;
- G722_1_24_encinst_t_* _encoderInst24PtrR;
- G722_1_32_encinst_t_* _encoderInst32Ptr;
- G722_1_32_encinst_t_* _encoderInst32PtrR;
-
- // Only one of these pointer is valid at any instance
- G722_1_16_decinst_t_* _decoderInst16Ptr;
- G722_1_24_decinst_t_* _decoderInst24Ptr;
- G722_1_32_decinst_t_* _decoderInst32Ptr;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_1_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g7221c.cc b/trunk/src/modules/audio_coding/main/source/acm_g7221c.cc
deleted file mode 100644
index 920965e..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_g7221c.cc
+++ /dev/null
@@ -1,532 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_g7221c.h"
-#include "acm_codec_database.h"
-#include "acm_common_defs.h"
-#include "acm_neteq.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-#include "trace.h"
-
-#ifdef WEBRTC_CODEC_G722_1C
-// NOTE! G.722.1C is not included in the open-source package. The following
-// interface file is needed:
-//
-// /modules/audio_coding/codecs/g7221c/main/interface/g7221c_interface.h
-//
-// The API in the header file should match the one below.
-//
-// int16_t WebRtcG7221C_CreateEnc24(G722_1C_24_encinst_t_** encInst);
-// int16_t WebRtcG7221C_CreateEnc32(G722_1C_32_encinst_t_** encInst);
-// int16_t WebRtcG7221C_CreateEnc48(G722_1C_48_encinst_t_** encInst);
-// int16_t WebRtcG7221C_CreateDec24(G722_1C_24_decinst_t_** decInst);
-// int16_t WebRtcG7221C_CreateDec32(G722_1C_32_decinst_t_** decInst);
-// int16_t WebRtcG7221C_CreateDec48(G722_1C_48_decinst_t_** decInst);
-//
-// int16_t WebRtcG7221C_FreeEnc24(G722_1C_24_encinst_t_** encInst);
-// int16_t WebRtcG7221C_FreeEnc32(G722_1C_32_encinst_t_** encInst);
-// int16_t WebRtcG7221C_FreeEnc48(G722_1C_48_encinst_t_** encInst);
-// int16_t WebRtcG7221C_FreeDec24(G722_1C_24_decinst_t_** decInst);
-// int16_t WebRtcG7221C_FreeDec32(G722_1C_32_decinst_t_** decInst);
-// int16_t WebRtcG7221C_FreeDec48(G722_1C_48_decinst_t_** decInst);
-//
-// int16_t WebRtcG7221C_EncoderInit24(G722_1C_24_encinst_t_* encInst);
-// int16_t WebRtcG7221C_EncoderInit32(G722_1C_32_encinst_t_* encInst);
-// int16_t WebRtcG7221C_EncoderInit48(G722_1C_48_encinst_t_* encInst);
-// int16_t WebRtcG7221C_DecoderInit24(G722_1C_24_decinst_t_* decInst);
-// int16_t WebRtcG7221C_DecoderInit32(G722_1C_32_decinst_t_* decInst);
-// int16_t WebRtcG7221C_DecoderInit48(G722_1C_48_decinst_t_* decInst);
-//
-// int16_t WebRtcG7221C_Encode24(G722_1C_24_encinst_t_* encInst,
-// int16_t* input,
-// int16_t len,
-// int16_t* output);
-// int16_t WebRtcG7221C_Encode32(G722_1C_32_encinst_t_* encInst,
-// int16_t* input,
-// int16_t len,
-// int16_t* output);
-// int16_t WebRtcG7221C_Encode48(G722_1C_48_encinst_t_* encInst,
-// int16_t* input,
-// int16_t len,
-// int16_t* output);
-//
-// int16_t WebRtcG7221C_Decode24(G722_1C_24_decinst_t_* decInst,
-// int16_t* bitstream,
-// int16_t len,
-// int16_t* output);
-// int16_t WebRtcG7221C_Decode32(G722_1C_32_decinst_t_* decInst,
-// int16_t* bitstream,
-// int16_t len,
-// int16_t* output);
-// int16_t WebRtcG7221C_Decode48(G722_1C_48_decinst_t_* decInst,
-// int16_t* bitstream,
-// int16_t len,
-// int16_t* output);
-//
-// int16_t WebRtcG7221C_DecodePlc24(G722_1C_24_decinst_t_* decInst,
-// int16_t* output,
-// int16_t nrLostFrames);
-// int16_t WebRtcG7221C_DecodePlc32(G722_1C_32_decinst_t_* decInst,
-// int16_t* output,
-// int16_t nrLostFrames);
-// int16_t WebRtcG7221C_DecodePlc48(G722_1C_48_decinst_t_* decInst,
-// int16_t* output,
-// int16_t nrLostFrames);
-// void WebRtcG7221C_Version(char *versionStr, short len);
-#include "g7221c_interface.h"
-#endif
-
-namespace webrtc {
-
-#ifndef WEBRTC_CODEC_G722_1C
-
-ACMG722_1C::ACMG722_1C(WebRtc_Word16 /* codecID */)
- : _operationalRate(-1),
- _encoderInstPtr(NULL),
- _encoderInstPtrRight(NULL),
- _decoderInstPtr(NULL),
- _encoderInst24Ptr(NULL),
- _encoderInst24PtrR(NULL),
- _encoderInst32Ptr(NULL),
- _encoderInst32PtrR(NULL),
- _encoderInst48Ptr(NULL),
- _encoderInst48PtrR(NULL),
- _decoderInst24Ptr(NULL),
- _decoderInst32Ptr(NULL),
- _decoderInst48Ptr(NULL) {
- return;
-}
-
-ACMG722_1C::~ACMG722_1C() {
- return;
-}
-
-WebRtc_Word16 ACMG722_1C::InternalEncode(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */) {
- return -1;
-}
-
-WebRtc_Word16 ACMG722_1C::DecodeSafe(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */) {
- return -1;
-}
-
-WebRtc_Word16 ACMG722_1C::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */) {
- return -1;
-}
-
-WebRtc_Word16 ACMG722_1C::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */) {
- return -1;
-}
-
-WebRtc_Word32 ACMG722_1C::CodecDef(WebRtcNetEQ_CodecDef& /* codecDef */,
- const CodecInst& /* codecInst */) {
- return -1;
-}
-
-ACMGenericCodec* ACMG722_1C::CreateInstance(void) {
- return NULL;
-}
-
-WebRtc_Word16 ACMG722_1C::InternalCreateEncoder() {
- return -1;
-}
-
-void ACMG722_1C::DestructEncoderSafe() {
- return;
-}
-
-WebRtc_Word16 ACMG722_1C::InternalCreateDecoder() {
- return -1;
-}
-
-void ACMG722_1C::DestructDecoderSafe() {
- return;
-}
-
-void ACMG722_1C::InternalDestructEncoderInst(void* /* ptrInst */) {
- return;
-}
-
-WebRtc_Word16 ACMG722_1C::UnregisterFromNetEqSafe(
- ACMNetEQ* /* netEq */,
- WebRtc_Word16 /* payloadType */) {
- return -1;
-}
-
-#else //===================== Actual Implementation =======================
-ACMG722_1C::ACMG722_1C(WebRtc_Word16 codecID) :
- _encoderInstPtr(NULL), _encoderInstPtrRight(NULL), _decoderInstPtr(NULL),
- _encoderInst24Ptr(NULL), _encoderInst24PtrR(NULL), _encoderInst32Ptr(NULL),
- _encoderInst32PtrR(NULL), _encoderInst48Ptr(NULL), _encoderInst48PtrR(NULL),
- _decoderInst24Ptr(NULL), _decoderInst32Ptr(NULL), _decoderInst48Ptr(NULL) {
- _codecID = codecID;
- if (_codecID == ACMCodecDB::kG722_1C_24) {
- _operationalRate = 24000;
- } else if (_codecID == ACMCodecDB::kG722_1C_32) {
- _operationalRate = 32000;
- } else if (_codecID == ACMCodecDB::kG722_1C_48) {
- _operationalRate = 48000;
- } else {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Wrong codec id for G722_1c.");
- _operationalRate = -1;
- }
- return;
-}
-
-ACMG722_1C::~ACMG722_1C() {
- if (_encoderInstPtr != NULL) {
- delete _encoderInstPtr;
- _encoderInstPtr = NULL;
- }
- if (_encoderInstPtrRight != NULL) {
- delete _encoderInstPtrRight;
- _encoderInstPtrRight = NULL;
- }
- if (_decoderInstPtr != NULL) {
- delete _decoderInstPtr;
- _decoderInstPtr = NULL;
- }
-
- switch (_operationalRate) {
- case 24000: {
- _encoderInst24Ptr = NULL;
- _encoderInst24PtrR = NULL;
- _decoderInst24Ptr = NULL;
- break;
- }
- case 32000: {
- _encoderInst32Ptr = NULL;
- _encoderInst32PtrR = NULL;
- _decoderInst32Ptr = NULL;
- break;
- }
- case 48000: {
- _encoderInst48Ptr = NULL;
- _encoderInst48PtrR = NULL;
- _decoderInst48Ptr = NULL;
- break;
- }
- default: {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Wrong rate for G722_1c.");
- break;
- }
- }
- return;
-}
-
-WebRtc_Word16 ACMG722_1C::InternalEncode(WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte) {
- WebRtc_Word16 leftChannel[640];
- WebRtc_Word16 rightChannel[640];
- WebRtc_Word16 lenInBytes;
- WebRtc_Word16 outB[240];
-
- // If stereo, split input signal in left and right channel before encoding
- if (_noChannels == 2) {
- for (int i = 0, j = 0; i < _frameLenSmpl * 2; i += 2, j++) {
- leftChannel[j] = _inAudio[_inAudioIxRead + i];
- rightChannel[j] = _inAudio[_inAudioIxRead + i + 1];
- }
- } else {
- memcpy(leftChannel, &_inAudio[_inAudioIxRead], 640);
- }
-
- switch (_operationalRate) {
- case 24000: {
- lenInBytes = WebRtcG7221C_Encode24(_encoderInst24Ptr, leftChannel, 640,
- &outB[0]);
- if (_noChannels == 2) {
- lenInBytes += WebRtcG7221C_Encode24(_encoderInst24PtrR, rightChannel,
- 640, &outB[lenInBytes / 2]);
- }
- break;
- }
- case 32000: {
- lenInBytes = WebRtcG7221C_Encode32(_encoderInst32Ptr, leftChannel, 640,
- &outB[0]);
- if (_noChannels == 2) {
- lenInBytes += WebRtcG7221C_Encode32(_encoderInst32PtrR, rightChannel,
- 640, &outB[lenInBytes / 2]);
- }
- break;
- }
- case 48000: {
- lenInBytes = WebRtcG7221C_Encode48(_encoderInst48Ptr, leftChannel, 640,
- &outB[0]);
- if (_noChannels == 2) {
- lenInBytes += WebRtcG7221C_Encode48(_encoderInst48PtrR, rightChannel,
- 640, &outB[lenInBytes / 2]);
- }
- break;
- }
- default: {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalEncode: Wrong rate for G722_1c.");
- return -1;
- }
- }
-
- memcpy(bitStream, outB, lenInBytes);
- *bitStreamLenByte = lenInBytes;
-
- // increment the read index this tell the caller that how far
- // we have gone forward in reading the audio buffer
- _inAudioIxRead += 640 * _noChannels;
-
- return *bitStreamLenByte;
-}
-
-WebRtc_Word16 ACMG722_1C::DecodeSafe(WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */) {
- return 0;
-}
-
-WebRtc_Word16 ACMG722_1C::InternalInitEncoder(
- WebRtcACMCodecParams* codecParams) {
- WebRtc_Word16 ret;
-
- switch (_operationalRate) {
- case 24000: {
- ret = WebRtcG7221C_EncoderInit24(_encoderInst24PtrR);
- if (ret < 0) {
- return ret;
- }
- return WebRtcG7221C_EncoderInit24(_encoderInst24Ptr);
- }
- case 32000: {
- ret = WebRtcG7221C_EncoderInit32(_encoderInst32PtrR);
- if (ret < 0) {
- return ret;
- }
- return WebRtcG7221C_EncoderInit32(_encoderInst32Ptr);
- }
- case 48000: {
- ret = WebRtcG7221C_EncoderInit48(_encoderInst48PtrR);
- if (ret < 0) {
- return ret;
- }
- return WebRtcG7221C_EncoderInit48(_encoderInst48Ptr);
- }
- default: {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalInitEncode: Wrong rate for G722_1c.");
- return -1;
- }
- }
-}
-
-WebRtc_Word16 ACMG722_1C::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */) {
- switch (_operationalRate) {
- case 24000: {
- return WebRtcG7221C_DecoderInit24(_decoderInst24Ptr);
- }
- case 32000: {
- return WebRtcG7221C_DecoderInit32(_decoderInst32Ptr);
- }
- case 48000: {
- return WebRtcG7221C_DecoderInit48(_decoderInst48Ptr);
- }
- default: {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalInitDecoder: Wrong rate for G722_1c.");
- return -1;
- }
- }
-}
-
-WebRtc_Word32 ACMG722_1C::CodecDef(WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst) {
-
- if (!_decoderInitialized) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "CodeDef: decoder not initialized for G722_1c");
- return -1;
- }
- // NetEq has an array of pointers to WebRtcNetEQ_CodecDef.
- // get an entry of that array (neteq wrapper will allocate memory)
- // by calling "netEq->CodecDef", where "NETEQ_CODEC_G722_1_XX" would
- // be the index of the entry.
- // Fill up the given structure by calling
- // "SET_CODEC_PAR" & "SET_G722_1_XX_FUNCTION."
- // Then return the structure back to NetEQ to add the codec to it's
- // database.
- switch (_operationalRate) {
- case 24000: {
- SET_CODEC_PAR((codecDef), kDecoderG722_1C_24, codecInst.pltype,
- _decoderInst24Ptr, 32000);
- SET_G722_1C_24_FUNCTIONS((codecDef));
- break;
- }
- case 32000: {
- SET_CODEC_PAR((codecDef), kDecoderG722_1C_32, codecInst.pltype,
- _decoderInst32Ptr, 32000);
- SET_G722_1C_32_FUNCTIONS((codecDef));
- break;
- }
- case 48000: {
- SET_CODEC_PAR((codecDef), kDecoderG722_1C_32, codecInst.pltype,
- _decoderInst48Ptr, 32000);
- SET_G722_1C_48_FUNCTIONS((codecDef));
- break;
- }
- default: {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "CodeDef: Wrong rate for G722_1c.");
- return -1;
- }
- }
- return 0;
-}
-
-ACMGenericCodec*
-ACMG722_1C::CreateInstance(void) {
- return NULL;
-}
-
-WebRtc_Word16 ACMG722_1C::InternalCreateEncoder() {
- if ((_encoderInstPtr == NULL) || (_encoderInstPtrRight == NULL)) {
- return -1;
- }
- switch (_operationalRate) {
- case 24000: {
- WebRtcG7221C_CreateEnc24(&_encoderInst24Ptr);
- WebRtcG7221C_CreateEnc24(&_encoderInst24PtrR);
- break;
- }
- case 32000: {
- WebRtcG7221C_CreateEnc32(&_encoderInst32Ptr);
- WebRtcG7221C_CreateEnc32(&_encoderInst32PtrR);
- break;
- }
- case 48000: {
- WebRtcG7221C_CreateEnc48(&_encoderInst48Ptr);
- WebRtcG7221C_CreateEnc48(&_encoderInst48PtrR);
- break;
- }
- default: {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalCreateEncoder: Wrong rate for G722_1c.");
- return -1;
- }
- }
- return 0;
-}
-
-void ACMG722_1C::DestructEncoderSafe() {
- _encoderExist = false;
- _encoderInitialized = false;
- if (_encoderInstPtr != NULL) {
- delete _encoderInstPtr;
- _encoderInstPtr = NULL;
- }
- if (_encoderInstPtrRight != NULL) {
- delete _encoderInstPtrRight;
- _encoderInstPtrRight = NULL;
- }
- _encoderInst24Ptr = NULL;
- _encoderInst32Ptr = NULL;
- _encoderInst48Ptr = NULL;
-}
-
-WebRtc_Word16 ACMG722_1C::InternalCreateDecoder() {
- if (_decoderInstPtr == NULL) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalCreateEncoder: cannot create decoder");
- return -1;
- }
- switch (_operationalRate) {
- case 24000: {
- WebRtcG7221C_CreateDec24(&_decoderInst24Ptr);
- break;
- }
- case 32000: {
- WebRtcG7221C_CreateDec32(&_decoderInst32Ptr);
- break;
- }
- case 48000: {
- WebRtcG7221C_CreateDec48(&_decoderInst48Ptr);
- break;
- }
- default: {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalCreateEncoder: Wrong rate for G722_1c.");
- return -1;
- }
- }
- return 0;
-}
-
-void ACMG722_1C::DestructDecoderSafe() {
- _decoderExist = false;
- _decoderInitialized = false;
- if (_decoderInstPtr != NULL) {
- delete _decoderInstPtr;
- _decoderInstPtr = NULL;
- }
- _decoderInst24Ptr = NULL;
- _decoderInst32Ptr = NULL;
- _decoderInst48Ptr = NULL;
-}
-
-void ACMG722_1C::InternalDestructEncoderInst(void* ptrInst) {
- if (ptrInst != NULL) {
- delete ptrInst;
- }
- return;
-}
-
-WebRtc_Word16 ACMG722_1C::UnregisterFromNetEqSafe(ACMNetEQ* netEq,
- WebRtc_Word16 payloadType) {
- if (payloadType != _decoderParams.codecInstant.pltype) {
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceAudioCoding,
- _uniqueID,
- "Cannot unregister codec %s given payload-type %d does not"
- "match the stored payload type",
- _decoderParams.codecInstant.plname, payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
- switch (_operationalRate) {
- case 24000: {
- return netEq->RemoveCodec(kDecoderG722_1C_24);
- }
- case 32000: {
- return netEq->RemoveCodec(kDecoderG722_1C_32);
- }
- case 48000: {
- return netEq->RemoveCodec(kDecoderG722_1C_48);
- }
- default: {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Could not remove codec from NetEQ for G722_1c."
- "Sampling frequency doesn't match");
- return -1;
- }
- }
-}
-
-#endif
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g7221c.h b/trunk/src/modules/audio_coding/main/source/acm_g7221c.h
deleted file mode 100644
index 5129e4c..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_g7221c.h
+++ /dev/null
@@ -1,94 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_1C_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_1C_H_
-
-#include "acm_generic_codec.h"
-
-// forward declaration
-struct G722_1C_24_encinst_t_;
-struct G722_1C_24_decinst_t_;
-struct G722_1C_32_encinst_t_;
-struct G722_1C_32_decinst_t_;
-struct G722_1C_48_encinst_t_;
-struct G722_1C_48_decinst_t_;
-struct G722_1_Inst_t_;
-
-namespace webrtc {
-
-class ACMG722_1C : public ACMGenericCodec
-{
-public:
- ACMG722_1C(WebRtc_Word16 codecID);
- ~ACMG722_1C();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(
- WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(
- WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 InternalInitDecoder(
- WebRtcACMCodecParams *codecParams);
-
-protected:
- WebRtc_Word16 DecodeSafe(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio,
- WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- void InternalDestructEncoderInst(
- void* ptrInst);
-
- WebRtc_Word16 UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- WebRtc_Word32 _operationalRate;
-
- G722_1_Inst_t_* _encoderInstPtr;
- G722_1_Inst_t_* _encoderInstPtrRight; //Used in stereo mode
- G722_1_Inst_t_* _decoderInstPtr;
-
- // Only one set of these pointer is valid at any instance
- G722_1C_24_encinst_t_* _encoderInst24Ptr;
- G722_1C_24_encinst_t_* _encoderInst24PtrR;
- G722_1C_32_encinst_t_* _encoderInst32Ptr;
- G722_1C_32_encinst_t_* _encoderInst32PtrR;
- G722_1C_48_encinst_t_* _encoderInst48Ptr;
- G722_1C_48_encinst_t_* _encoderInst48PtrR;
-
- // Only one of these pointer is valid at any instance
- G722_1C_24_decinst_t_* _decoderInst24Ptr;
- G722_1C_32_decinst_t_* _decoderInst32Ptr;
- G722_1C_48_decinst_t_* _decoderInst48Ptr;
-};
-
-} // namespace webrtc;
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G722_1C_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g729.cc b/trunk/src/modules/audio_coding/main/source/acm_g729.cc
deleted file mode 100644
index bca2d72..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_g729.cc
+++ /dev/null
@@ -1,546 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_g729.h"
-#include "acm_common_defs.h"
-#include "acm_neteq.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-
-#ifdef WEBRTC_CODEC_G729
- // NOTE! G.729 is not included in the open-source package. The following
- // interface file is needed:
- //
- // /modules/audio_coding/codecs/g729/main/interface/g729_interface.h
- //
- // The API in the header file should match the one below.
- //
- // int16_t WebRtcG729_CreateEnc(G729_encinst_t_** inst);
- // int16_t WebRtcG729_CreateDec(G729_decinst_t_** inst);
- // int16_t WebRtcG729_FreeEnc(G729_encinst_t_* inst);
- // int16_t WebRtcG729_FreeDec(G729_decinst_t_* inst);
- // int16_t WebRtcG729_Encode(G729_encinst_t_* encInst, int16_t* input,
- // int16_t len, int16_t* output);
- // int16_t WebRtcG729_EncoderInit(G729_encinst_t_* encInst, int16_t mode);
- // int16_t WebRtcG729_Decode(G729_decinst_t_* decInst);
- // int16_t WebRtcG729_DecodeBwe(G729_decinst_t_* decInst, int16_t* input);
- // int16_t WebRtcG729_DecodePlc(G729_decinst_t_* decInst);
- // int16_t WebRtcG729_DecoderInit(G729_decinst_t_* decInst);
- // void WebRtcG729_Version(char *versionStr, short len);
- #include "g729_interface.h"
-#endif
-
-namespace webrtc {
-
-#ifndef WEBRTC_CODEC_G729
-
-ACMG729::ACMG729(WebRtc_Word16 /* codecID */)
- : _encoderInstPtr(NULL),
- _decoderInstPtr(NULL) {
- return;
-}
-
-
-ACMG729::~ACMG729()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMG729::InternalEncode(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMG729::EnableDTX()
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMG729::DisableDTX()
-{
- return -1;
-}
-
-WebRtc_Word32
-ACMG729::ReplaceInternalDTXSafe(
- const bool /*replaceInternalDTX*/)
-{
- return -1;
-}
-
-WebRtc_Word32
-ACMG729::IsInternalDTXReplacedSafe(
- bool* /* internalDTXReplaced */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMG729::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMG729::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMG729::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
-}
-
-
-WebRtc_Word32
-ACMG729::CodecDef(
- WebRtcNetEQ_CodecDef& /* codecDef */,
- const CodecInst& /* codecInst */)
-{
- return -1;
-}
-
-
-ACMGenericCodec*
-ACMG729::CreateInstance(void)
-{
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMG729::InternalCreateEncoder()
-{
- return -1;
-}
-
-
-void
-ACMG729::DestructEncoderSafe()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMG729::InternalCreateDecoder()
-{
- return -1;
-}
-
-
-void
-ACMG729::DestructDecoderSafe()
-{
- return;
-}
-
-
-void
-ACMG729::InternalDestructEncoderInst(
- void* /* ptrInst */)
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMG729::UnregisterFromNetEqSafe(
- ACMNetEQ* /* netEq */,
- WebRtc_Word16 /* payloadType */)
-{
- return -1;
-}
-
-
-#else //===================== Actual Implementation =======================
-
-ACMG729::ACMG729(
- WebRtc_Word16 codecID):
-_encoderInstPtr(NULL),
-_decoderInstPtr(NULL)
-{
- _codecID = codecID;
- _hasInternalDTX = true;
- return;
-}
-
-
-ACMG729::~ACMG729()
-{
- if(_encoderInstPtr != NULL)
- {
- // Delete encoder memory
- WebRtcG729_FreeEnc(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
- if(_decoderInstPtr != NULL)
- {
- // Delete decoder memory
- WebRtcG729_FreeDec(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
- return;
-}
-
-
-WebRtc_Word16
-ACMG729::InternalEncode(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte)
-{
- // Initialize before entering the loop
- WebRtc_Word16 noEncodedSamples = 0;
- WebRtc_Word16 tmpLenByte = 0;
- WebRtc_Word16 vadDecision = 0;
- *bitStreamLenByte = 0;
- while(noEncodedSamples < _frameLenSmpl)
- {
- // Call G.729 encoder with pointer to encoder memory, input
- // audio, number of samples and bitsream
- tmpLenByte = WebRtcG729_Encode(_encoderInstPtr,
- &_inAudio[_inAudioIxRead], 80,
- (WebRtc_Word16*)(&(bitStream[*bitStreamLenByte])));
-
- // increment the read index this tell the caller that how far
- // we have gone forward in reading the audio buffer
- _inAudioIxRead += 80;
-
- // sanity check
- if(tmpLenByte < 0)
- {
- // error has happened
- *bitStreamLenByte = 0;
- return -1;
- }
-
- // increment number of written bytes
- *bitStreamLenByte += tmpLenByte;
- switch(tmpLenByte)
- {
- case 0:
- {
- if(0 == noEncodedSamples)
- {
- // this is the first 10 ms in this packet and there is
- // no data generated, perhaps DTX is enabled and the
- // codec is not generating any bit-stream for this 10 ms.
- // we do not continue encoding this frame.
- return 0;
- }
- break;
- }
- case 2:
- {
- // check if G.729 internal DTX is enabled
- if(_hasInternalDTX && _dtxEnabled)
- {
- vadDecision = 0;
- for(WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++)
- {
- _vadLabel[n] = vadDecision;
- }
- }
- // we got a SID and have to send out this packet no matter
- // how much audio we have encoded
- return *bitStreamLenByte;
- }
- case 10:
- {
- vadDecision = 1;
- // this is a valid length just continue encoding
- break;
- }
- default:
- {
- return -1;
- }
- }
-
- // update number of encoded samples
- noEncodedSamples += 80;
- }
-
- // update VAD decision vector
- if(_hasInternalDTX && !vadDecision && _dtxEnabled)
- {
- for(WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++)
- {
- _vadLabel[n] = vadDecision;
- }
- }
-
- // done encoding, return number of encoded bytes
- return *bitStreamLenByte;
-}
-
-
-WebRtc_Word16
-ACMG729::EnableDTX()
-{
- if(_dtxEnabled)
- {
- // DTX already enabled, do nothing
- return 0;
- }
- else if(_encoderExist)
- {
- // Re-init the G.729 encoder to turn on DTX
- if(WebRtcG729_EncoderInit(_encoderInstPtr, 1) < 0)
- {
- return -1;
- }
- _dtxEnabled = true;
- return 0;
- }
- else
- {
- return -1;
- }
-}
-
-
-WebRtc_Word16
-ACMG729::DisableDTX()
-{
- if(!_dtxEnabled)
- {
- // DTX already dissabled, do nothing
- return 0;
- }
- else if(_encoderExist)
- {
- // Re-init the G.729 decoder to turn off DTX
- if(WebRtcG729_EncoderInit(_encoderInstPtr, 0) < 0)
- {
- return -1;
- }
- _dtxEnabled = false;
- return 0;
- }
- else
- {
- // encoder doesn't exists, therefore disabling is harmless
- return 0;
- }
-}
-
-
-WebRtc_Word32
-ACMG729::ReplaceInternalDTXSafe(
- const bool replaceInternalDTX)
-{
- // This function is used to dissable the G.729 built in DTX and use an
- // external instead.
-
- if(replaceInternalDTX == _hasInternalDTX)
- {
- // Make sure we keep the DTX/VAD setting if possible
- bool oldEnableDTX = _dtxEnabled;
- bool oldEnableVAD = _vadEnabled;
- ACMVADMode oldMode = _vadMode;
- if (replaceInternalDTX)
- {
- // Disable internal DTX before enabling external DTX
- DisableDTX();
- }
- else
- {
- // Disable external DTX before enabling internal
- ACMGenericCodec::DisableDTX();
- }
- _hasInternalDTX = !replaceInternalDTX;
- WebRtc_Word16 status = SetVADSafe(oldEnableDTX, oldEnableVAD, oldMode);
- // Check if VAD status has changed from inactive to active, or if error was
- // reported
- if (status == 1) {
- _vadEnabled = true;
- return status;
- } else if (status < 0) {
- _hasInternalDTX = replaceInternalDTX;
- return -1;
- }
- }
- return 0;
-}
-
-
-WebRtc_Word32
-ACMG729::IsInternalDTXReplacedSafe(
- bool* internalDTXReplaced)
-{
- // Get status of wether DTX is replaced or not
- *internalDTXReplaced = !_hasInternalDTX;
- return 0;
-}
-
-
-WebRtc_Word16
-ACMG729::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- // This function is not used. G.729 decoder is called from inside NetEQ
- return 0;
-}
-
-
-WebRtc_Word16
-ACMG729::InternalInitEncoder(
- WebRtcACMCodecParams* codecParams)
-{
- // Init G.729 encoder
- return WebRtcG729_EncoderInit(_encoderInstPtr,
- ((codecParams->enableDTX)? 1:0));
-}
-
-
-WebRtc_Word16
-ACMG729::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- // Init G.729 decoder
- return WebRtcG729_DecoderInit(_decoderInstPtr);
-}
-
-
-WebRtc_Word32
-ACMG729::CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst)
-{
- if (!_decoderInitialized)
- {
- // Todo:
- // log error
- return -1;
- }
-
- // Fill up the structure by calling
- // "SET_CODEC_PAR" & "SET_G729_FUNCTION."
- // Then call NetEQ to add the codec to it's
- // database.
- SET_CODEC_PAR((codecDef), kDecoderG729, codecInst.pltype,
- _decoderInstPtr, 8000);
- SET_G729_FUNCTIONS((codecDef));
- return 0;
-}
-
-
-ACMGenericCodec*
-ACMG729::CreateInstance(void)
-{
- // Function not used
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMG729::InternalCreateEncoder()
-{
- // Create encoder memory
- return WebRtcG729_CreateEnc(&_encoderInstPtr);
-}
-
-
-void
-ACMG729::DestructEncoderSafe()
-{
- // Free encoder memory
- _encoderExist = false;
- _encoderInitialized = false;
- if(_encoderInstPtr != NULL)
- {
- WebRtcG729_FreeEnc(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
-}
-
-
-WebRtc_Word16
-ACMG729::InternalCreateDecoder()
-{
- // Create decoder memory
- return WebRtcG729_CreateDec(&_decoderInstPtr);
-}
-
-
-void
-ACMG729::DestructDecoderSafe()
-{
- // Free decoder memory
- _decoderExist = false;
- _decoderInitialized = false;
- if(_decoderInstPtr != NULL)
- {
- WebRtcG729_FreeDec(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
-}
-
-
-void
-ACMG729::InternalDestructEncoderInst(
- void* ptrInst)
-{
- if(ptrInst != NULL)
- {
- WebRtcG729_FreeEnc((G729_encinst_t_*)ptrInst);
- }
- return;
-}
-
-
-WebRtc_Word16
-ACMG729::UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType)
-{
- // Remove codec from the NetEQ database
- if(payloadType != _decoderParams.codecInstant.pltype)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot unregister codec %s given payload-type %d does not match \
-the stored payload type",
- _decoderParams.codecInstant.plname,
- payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
- return netEq->RemoveCodec(kDecoderG729);
-}
-
-#endif
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g729.h b/trunk/src/modules/audio_coding/main/source/acm_g729.h
deleted file mode 100644
index 50b648a..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_g729.h
+++ /dev/null
@@ -1,84 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_
-
-#include "acm_generic_codec.h"
-
-// forward declaration
-struct G729_encinst_t_;
-struct G729_decinst_t_;
-
-namespace webrtc {
-
-class ACMG729 : public ACMGenericCodec
-{
-public:
- ACMG729(WebRtc_Word16 codecID);
- ~ACMG729();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(
- WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(
- WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 InternalInitDecoder(
- WebRtcACMCodecParams *codecParams);
-
-protected:
- WebRtc_Word16 DecodeSafe(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio,
- WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- void InternalDestructEncoderInst(
- void* ptrInst);
-
- WebRtc_Word16 EnableDTX();
-
- WebRtc_Word16 DisableDTX();
-
- WebRtc_Word32 ReplaceInternalDTXSafe(
- const bool replaceInternalDTX);
-
- WebRtc_Word32 IsInternalDTXReplacedSafe(
- bool* internalDTXReplaced);
-
- WebRtc_Word16 UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- G729_encinst_t_* _encoderInstPtr;
- G729_decinst_t_* _decoderInstPtr;
-
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g7291.cc b/trunk/src/modules/audio_coding/main/source/acm_g7291.cc
deleted file mode 100644
index d832e9c..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_g7291.cc
+++ /dev/null
@@ -1,500 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_g7291.h"
-#include "acm_common_defs.h"
-#include "acm_neteq.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-
-#ifdef WEBRTC_CODEC_G729_1
- // NOTE! G.729.1 is not included in the open-source package. The following
- // interface file is needed:
- //
- // /modules/audio_coding/codecs/g7291/main/interface/g7291_interface.h
- //
- // The API in the header file should match the one below.
- //
- // int16_t WebRtcG7291_Create(G729_1_inst_t_** inst);
- // int16_t WebRtcG7291_Free(G729_1_inst_t_* inst);
- // int16_t WebRtcG7291_Encode(G729_1_inst_t_* encInst, int16_t* input,
- // int16_t* output, int16_t myRate,
- // int16_t nrFrames);
- // int16_t WebRtcG7291_EncoderInit(G729_1_inst_t_* encInst, int16_t myRate,
- // int16_t flag8kHz, int16_t flagG729mode);
- // int16_t WebRtcG7291_Decode(G729_1_inst_t_* decInst);
- // int16_t WebRtcG7291_DecodeBwe(G729_1_inst_t_* decInst, int16_t* input);
- // int16_t WebRtcG7291_DecodePlc(G729_1_inst_t_* decInst);
- // int16_t WebRtcG7291_DecoderInit(G729_1_inst_t_* decInst);
- // void WebRtcG7291_Version(char *versionStr, short len);
- #include "g7291_interface.h"
-#endif
-
-namespace webrtc {
-
-#ifndef WEBRTC_CODEC_G729_1
-
-ACMG729_1::ACMG729_1( WebRtc_Word16 /* codecID */)
- : _encoderInstPtr(NULL),
- _decoderInstPtr(NULL),
- _myRate(32000),
- _flag8kHz(0),
- _flagG729mode(0) {
- return;
-}
-
-
-ACMG729_1::~ACMG729_1()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMG729_1::InternalEncode(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMG729_1::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMG729_1::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMG729_1::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
-}
-
-
-WebRtc_Word32
-ACMG729_1::CodecDef(
- WebRtcNetEQ_CodecDef& /* codecDef */,
- const CodecInst& /* codecInst */)
-{
- return -1;
-}
-
-
-ACMGenericCodec*
-ACMG729_1::CreateInstance(void)
-{
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMG729_1::InternalCreateEncoder()
-{
- return -1;
-}
-
-
-void
-ACMG729_1::DestructEncoderSafe()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMG729_1::InternalCreateDecoder()
-{
- return -1;
-}
-
-
-void
-ACMG729_1::DestructDecoderSafe()
-{
- return;
-}
-
-
-void
-ACMG729_1::InternalDestructEncoderInst(
- void* /* ptrInst */)
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMG729_1::UnregisterFromNetEqSafe(
- ACMNetEQ* /* netEq */,
- WebRtc_Word16 /* payloadType */)
-{
- return -1;
-}
-
-WebRtc_Word16
-ACMG729_1::SetBitRateSafe(
- const WebRtc_Word32 /*rate*/ )
-{
- return -1;
-}
-
-#else //===================== Actual Implementation =======================
-
-struct G729_1_inst_t_;
-
-ACMG729_1::ACMG729_1(WebRtc_Word16 codecID)
- : _encoderInstPtr(NULL),
- _decoderInstPtr(NULL),
- _myRate(32000), // Default rate.
- _flag8kHz(0),
- _flagG729mode(0) {
- // TODO(tlegrand): We should add codecID as a input variable to the
- // constructor of ACMGenericCodec.
- _codecID = codecID;
- return;
-}
-
-ACMG729_1::~ACMG729_1()
-{
- if(_encoderInstPtr != NULL)
- {
- WebRtcG7291_Free(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
- if(_decoderInstPtr != NULL)
- {
- WebRtcG7291_Free(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
- return;
-}
-
-
-WebRtc_Word16
-ACMG729_1::InternalEncode(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte)
-{
-
- // Initialize before entering the loop
- WebRtc_Word16 noEncodedSamples = 0;
- *bitStreamLenByte = 0;
-
- WebRtc_Word16 byteLengthFrame = 0;
-
- // Derive number of 20ms frames per encoded packet.
- // [1,2,3] <=> [20,40,60]ms <=> [320,640,960] samples
- WebRtc_Word16 n20msFrames = (_frameLenSmpl / 320);
- // Byte length for the frame. +1 is for rate information.
- byteLengthFrame = _myRate/(8*50) * n20msFrames + (1 - _flagG729mode);
-
- // The following might be revised if we have G729.1 Annex C (support for DTX);
- do
- {
- *bitStreamLenByte = WebRtcG7291_Encode(_encoderInstPtr, &_inAudio[_inAudioIxRead],
- (WebRtc_Word16*)bitStream, _myRate, n20msFrames);
-
- // increment the read index this tell the caller that how far
- // we have gone forward in reading the audio buffer
- _inAudioIxRead += 160;
-
- // sanity check
- if(*bitStreamLenByte < 0)
- {
- // error has happened
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalEncode: Encode error for G729_1");
- *bitStreamLenByte = 0;
- return -1;
- }
-
- noEncodedSamples += 160;
- } while(*bitStreamLenByte == 0);
-
-
- // This criteria will change if we have Annex C.
- if(*bitStreamLenByte != byteLengthFrame)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalEncode: Encode error for G729_1");
- *bitStreamLenByte = 0;
- return -1;
- }
-
-
- if(noEncodedSamples != _frameLenSmpl)
- {
- *bitStreamLenByte = 0;
- return -1;
- }
-
- return *bitStreamLenByte;
-}
-
-
-WebRtc_Word16
-ACMG729_1::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return 0;
-}
-
-
-WebRtc_Word16
-ACMG729_1::InternalInitEncoder(
- WebRtcACMCodecParams* codecParams)
-{
- //set the bit rate and initialize
- _myRate = codecParams->codecInstant.rate;
- return SetBitRateSafe( (WebRtc_UWord32)_myRate);
-}
-
-
-WebRtc_Word16
-ACMG729_1::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- if (WebRtcG7291_DecoderInit(_decoderInstPtr) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalInitDecoder: init decoder failed for G729_1");
- return -1;
- }
- return 0;
-}
-
-
-WebRtc_Word32
-ACMG729_1::CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst)
-{
- if (!_decoderInitialized)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "CodeDef: Decoder uninitialized for G729_1");
- return -1;
- }
-
- // Fill up the structure by calling
- // "SET_CODEC_PAR" & "SET_G729_FUNCTION."
- // Then call NetEQ to add the codec to it's
- // database.
- SET_CODEC_PAR((codecDef), kDecoderG729_1, codecInst.pltype,
- _decoderInstPtr, 16000);
- SET_G729_1_FUNCTIONS((codecDef));
- return 0;
-}
-
-
-ACMGenericCodec*
-ACMG729_1::CreateInstance(void)
-{
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMG729_1::InternalCreateEncoder()
-{
- if (WebRtcG7291_Create(&_encoderInstPtr) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalCreateEncoder: create encoder failed for G729_1");
- return -1;
- }
- return 0;
-}
-
-
-void
-ACMG729_1::DestructEncoderSafe()
-{
- _encoderExist = false;
- _encoderInitialized = false;
- if(_encoderInstPtr != NULL)
- {
- WebRtcG7291_Free(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
-}
-
-
-WebRtc_Word16
-ACMG729_1::InternalCreateDecoder()
-{
- if (WebRtcG7291_Create(&_decoderInstPtr) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalCreateDecoder: create decoder failed for G729_1");
- return -1;
- }
- return 0;
-}
-
-
-void
-ACMG729_1::DestructDecoderSafe()
-{
- _decoderExist = false;
- _decoderInitialized = false;
- if(_decoderInstPtr != NULL)
- {
- WebRtcG7291_Free(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
-}
-
-
-void
-ACMG729_1::InternalDestructEncoderInst(
- void* ptrInst)
-{
- if(ptrInst != NULL)
- {
- //WebRtcG7291_Free((G729_1_inst_t*)ptrInst);
- }
- return;
-}
-
-
-WebRtc_Word16
-ACMG729_1::UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType)
-{
- if(payloadType != _decoderParams.codecInstant.pltype)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot unregister codec: given payload-type does not match \
-the stored payload type",
- _decoderParams.codecInstant.plname,
- payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
- return netEq->RemoveCodec(kDecoderG729_1);
-}
-
-WebRtc_Word16
-ACMG729_1::SetBitRateSafe(
- const WebRtc_Word32 rate)
-{
- //allowed rates: { 8000, 12000, 14000, 16000, 18000, 20000,
- // 22000, 24000, 26000, 28000, 30000, 32000};
- // TODO(tlegrand): This check exists in one other place two. Should be
- // possible to reuse code.
- switch(rate)
- {
- case 8000:
- {
- _myRate = 8000;
- break;
- }
- case 12000:
- {
- _myRate = 12000;
- break;
- }
- case 14000:
- {
- _myRate = 14000;
- break;
- }
- case 16000:
- {
- _myRate = 16000;
- break;
- }
- case 18000:
- {
- _myRate = 18000;
- break;
- }
- case 20000:
- {
- _myRate = 20000;
- break;
- }
- case 22000:
- {
- _myRate = 22000;
- break;
- }
- case 24000:
- {
- _myRate = 24000;
- break;
- }
- case 26000:
- {
- _myRate = 26000;
- break;
- }
- case 28000:
- {
- _myRate = 28000;
- break;
- }
- case 30000:
- {
- _myRate = 30000;
- break;
- }
- case 32000:
- {
- _myRate = 32000;
- break;
- }
- default:
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "SetBitRateSafe: Invalid rate G729_1");
- return -1;
- }
- }
-
- // Re-init with new rate
- if (WebRtcG7291_EncoderInit(_encoderInstPtr, _myRate, _flag8kHz, _flagG729mode) >= 0)
- {
- _encoderParams.codecInstant.rate = _myRate;
- return 0;
- }
- else
- {
- return -1;
- }
-}
-
-
-#endif
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_g7291.h b/trunk/src/modules/audio_coding/main/source/acm_g7291.h
deleted file mode 100644
index cb27dc9..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_g7291.h
+++ /dev/null
@@ -1,85 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_1_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_1_H_
-
-#include "acm_generic_codec.h"
-
-// forward declaration
-struct G729_1_inst_t_;
-struct G729_1_inst_t_;
-
-namespace webrtc {
-
-class ACMG729_1: public ACMGenericCodec
-{
-public:
- ACMG729_1(WebRtc_Word16 codecID);
- ~ACMG729_1();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(
- WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(
- WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 InternalInitDecoder(
- WebRtcACMCodecParams *codecParams);
-
-protected:
- WebRtc_Word16 DecodeSafe(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio,
- WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- void InternalDestructEncoderInst(
- void* ptrInst);
-
- //WebRtc_Word16 EnableDTX();
- //
- //WebRtc_Word16 DisableDTX();
-
- WebRtc_Word16 UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- WebRtc_Word16 SetBitRateSafe(
- const WebRtc_Word32 rate);
-
- G729_1_inst_t_* _encoderInstPtr;
- G729_1_inst_t_* _decoderInstPtr;
-
- WebRtc_UWord16 _myRate;
- WebRtc_Word16 _flag8kHz;
- WebRtc_Word16 _flagG729mode;
-
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_G729_1_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_generic_codec.cc b/trunk/src/modules/audio_coding/main/source/acm_generic_codec.cc
deleted file mode 100644
index 0e212fb..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_generic_codec.cc
+++ /dev/null
@@ -1,1554 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <assert.h>
-#include <string.h>
-
-#include "acm_codec_database.h"
-#include "acm_common_defs.h"
-#include "acm_generic_codec.h"
-#include "acm_neteq.h"
-#include "trace.h"
-#include "webrtc_vad.h"
-#include "webrtc_cng.h"
-
-namespace webrtc
-{
-
-// Enum for CNG
-enum
-{
- kMaxPLCParamsCNG = WEBRTC_CNG_MAX_LPC_ORDER,
- kNewCNGNumPLCParams = 8
-};
-
-#define ACM_SID_INTERVAL_MSEC 100
-
-// We set some of the variables to invalid values as a check point
-// if a proper initialization has happened. Another approach is
-// to initialize to a default codec that we are sure is always included.
-ACMGenericCodec::ACMGenericCodec()
- : _inAudioIxWrite(0),
- _inAudioIxRead(0),
- _inTimestampIxWrite(0),
- _inAudio(NULL),
- _inTimestamp(NULL),
- _frameLenSmpl(-1), // invalid value
- _noChannels(1),
- _codecID(-1), // invalid value
- _noMissedSamples(0),
- _encoderExist(false),
- _decoderExist(false),
- _encoderInitialized(false),
- _decoderInitialized(false),
- _registeredInNetEq(false),
- _hasInternalDTX(false),
- _ptrVADInst(NULL),
- _vadEnabled(false),
- _vadMode(VADNormal),
- _dtxEnabled(false),
- _ptrDTXInst(NULL),
- _numLPCParams(kNewCNGNumPLCParams),
- _sentCNPrevious(false),
- _isMaster(true),
- _netEqDecodeLock(NULL),
- _codecWrapperLock(*RWLockWrapper::CreateRWLock()),
- _lastEncodedTimestamp(0),
- _lastTimestamp(0xD87F3F9F),
- _isAudioBuffFresh(true),
- _uniqueID(0) {
- // Initialize VAD vector.
- for (int i = 0; i < MAX_FRAME_SIZE_10MSEC; i++) {
- _vadLabel[i] = 0;
- }
-
- // Nullify memory for encoder and decoder, and set payload type to an
- // invalid value.
- memset(&_encoderParams, 0, sizeof(WebRtcACMCodecParams));
- _encoderParams.codecInstant.pltype = -1;
- memset(&_decoderParams, 0, sizeof(WebRtcACMCodecParams));
- _decoderParams.codecInstant.pltype = -1;
-}
-
-ACMGenericCodec::~ACMGenericCodec()
-{
- // Check all the members which are pointers and
- // if they are not NULL delete/free them.
-
- if(_ptrVADInst != NULL)
- {
- WebRtcVad_Free(_ptrVADInst);
- _ptrVADInst = NULL;
- }
-
- if (_inAudio != NULL)
- {
- delete [] _inAudio;
- _inAudio = NULL;
- }
-
- if (_inTimestamp != NULL)
- {
- delete [] _inTimestamp;
- _inTimestamp = NULL;
- }
- if(_ptrDTXInst != NULL)
- {
- WebRtcCng_FreeEnc(_ptrDTXInst);
- _ptrDTXInst = NULL;
- }
- delete &_codecWrapperLock;
-}
-
-WebRtc_Word32
-ACMGenericCodec::Add10MsData(
- const WebRtc_UWord32 timestamp,
- const WebRtc_Word16* data,
- const WebRtc_UWord16 lengthSmpl,
- const WebRtc_UWord8 audioChannel)
-{
- WriteLockScoped wl(_codecWrapperLock);
- return Add10MsDataSafe(timestamp, data, lengthSmpl, audioChannel);
-}
-
-WebRtc_Word32
-ACMGenericCodec::Add10MsDataSafe(
- const WebRtc_UWord32 timestamp,
- const WebRtc_Word16* data,
- const WebRtc_UWord16 lengthSmpl,
- const WebRtc_UWord8 audioChannel)
-{
- // The codec expects to get data in correct sampling rate.
- // get the sampling frequency of the codec
- WebRtc_UWord16 plFreqHz;
-
- if(EncoderSampFreq(plFreqHz) < 0)
- {
- // _codecID is not correct, perhaps the codec is not initialized yet.
- return -1;
- }
-
- // Sanity check, if the length of the input corresponds to 10 ms.
- if((plFreqHz / 100) != lengthSmpl)
- {
- // This is not 10 ms of audio, given the sampling frequency of the
- // codec
- return -1;
- }
- if(_lastTimestamp == timestamp)
- {
- // Same timestamp as the last time, overwrite.
- if((_inAudioIxWrite >= lengthSmpl) && (_inTimestampIxWrite > 0))
- {
- _inAudioIxWrite -= lengthSmpl;
- _inTimestampIxWrite--;
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _uniqueID,
- "Adding 10ms with previous timestamp, \
-overwriting the previous 10ms");
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _uniqueID,
- "Adding 10ms with previous timestamp, this will sound bad");
- }
- }
-
- _lastTimestamp = timestamp;
-
- if ((_inAudioIxWrite + lengthSmpl*audioChannel) > AUDIO_BUFFER_SIZE_W16)
- {
- // Get the number of samples to be overwritten
- WebRtc_Word16 missedSamples = _inAudioIxWrite + lengthSmpl*audioChannel -
- AUDIO_BUFFER_SIZE_W16;
-
- // Move the data (overwite the old data)
- memmove(_inAudio, _inAudio + missedSamples,
- (AUDIO_BUFFER_SIZE_W16 - lengthSmpl*audioChannel)*sizeof(WebRtc_Word16));
- // Copy the new data
- memcpy(_inAudio + (AUDIO_BUFFER_SIZE_W16 - lengthSmpl*audioChannel), data,
- lengthSmpl*audioChannel * sizeof(WebRtc_Word16));
-
- // Get the number of 10 ms blocks which are overwritten
- WebRtc_Word16 missed10MsecBlocks =
- (WebRtc_Word16)((missedSamples/audioChannel * 100) / plFreqHz);
-
- // Move the timestamps
- memmove(_inTimestamp, _inTimestamp + missed10MsecBlocks,
- (_inTimestampIxWrite - missed10MsecBlocks) * sizeof(WebRtc_UWord32));
- _inTimestampIxWrite -= missed10MsecBlocks;
- _inTimestamp[_inTimestampIxWrite] = timestamp;
- _inTimestampIxWrite++;
-
- // Buffer is full
- _inAudioIxWrite = AUDIO_BUFFER_SIZE_W16;
- IncreaseNoMissedSamples(missedSamples);
- _isAudioBuffFresh = false;
- return -missedSamples;
- }
- memcpy(_inAudio + _inAudioIxWrite, data, lengthSmpl*audioChannel * sizeof(WebRtc_Word16));
- _inAudioIxWrite += lengthSmpl*audioChannel;
-
- assert(_inTimestampIxWrite < TIMESTAMP_BUFFER_SIZE_W32);
- assert(_inTimestampIxWrite >= 0);
-
- _inTimestamp[_inTimestampIxWrite] = timestamp;
- _inTimestampIxWrite++;
- _isAudioBuffFresh = false;
- return 0;
-}
-
-WebRtc_Word16
-ACMGenericCodec::Encode(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte,
- WebRtc_UWord32* timeStamp,
- WebRtcACMEncodingType* encodingType)
-{
- WriteLockScoped lockCodec(_codecWrapperLock);
- ReadLockScoped lockNetEq(*_netEqDecodeLock);
- return EncodeSafe(bitStream, bitStreamLenByte,
- timeStamp, encodingType);
-}
-
-
-WebRtc_Word16
-ACMGenericCodec::EncodeSafe(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte,
- WebRtc_UWord32* timeStamp,
- WebRtcACMEncodingType* encodingType)
-{
- // Do we have enough data to encode?
- // we wait until we have a full frame to encode.
- if(_inAudioIxWrite < _frameLenSmpl*_noChannels)
- {
- // There is not enough audio
- *timeStamp = 0;
- *bitStreamLenByte = 0;
- // Doesn't really matter what this parameter set to
- *encodingType = kNoEncoding;
- return 0;
- }
-
- // Not all codecs accept the whole frame to be pushed into
- // encoder at once.
- const WebRtc_Word16 myBasicCodingBlockSmpl =
- ACMCodecDB::BasicCodingBlock(_codecID);
- if((myBasicCodingBlockSmpl < 0) ||
- (!_encoderInitialized) ||
- (!_encoderExist))
- {
- // This should not happen
- *timeStamp = 0;
- *bitStreamLenByte = 0;
- *encodingType = kNoEncoding;
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "EncodeSafe: error, basic coding sample block is negative");
- return -1;
- }
-
- // This makes the internal encoder read from the begining of the buffer
- _inAudioIxRead = 0;
- *timeStamp = _inTimestamp[0];
-
- // Process the audio through VAD the function doesn't set _vadLabels.
- // If VAD is disabled all labels are set to ONE (active)
- WebRtc_Word16 status = 0;
- WebRtc_Word16 dtxProcessedSamples = 0;
-
- status = ProcessFrameVADDTX(bitStream, bitStreamLenByte,
- &dtxProcessedSamples);
-
- if(status < 0)
- {
- *timeStamp = 0;
- *bitStreamLenByte = 0;
- *encodingType = kNoEncoding;
- }
- else
- {
- if(dtxProcessedSamples > 0)
- {
- // Dtx have processed some samples may or may not a bit-stream
- // is generated we should not do any encoding (normally there
- // will be not enough data)
-
- // Setting the following makes that the move of audio data
- // and timestamps happen correctly
- _inAudioIxRead = dtxProcessedSamples;
- // This will let the owner of ACMGenericCodec to know that the
- // generated bit-stream is DTX to use correct payload type
- WebRtc_UWord16 sampFreqHz;
- EncoderSampFreq(sampFreqHz);
- if (sampFreqHz == 8000) {
- *encodingType = kPassiveDTXNB;
- } else if (sampFreqHz == 16000) {
- *encodingType = kPassiveDTXWB;
- } else if (sampFreqHz == 32000) {
- *encodingType = kPassiveDTXSWB;
- } else {
- status = -1;
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "EncodeSafe: Wrong sampling frequency for DTX.");
- }
-
- // Transport empty frame if we have an empty bitstream
- if ((*bitStreamLenByte == 0)
- && (_sentCNPrevious || ((_inAudioIxWrite - _inAudioIxRead) <= 0))
- )
- {
- // Makes sure we transmit an empty frame
- *bitStreamLenByte = 1;
- *encodingType = kNoEncoding;
- }
- _sentCNPrevious = true;
- }
- else
- {
- _sentCNPrevious = false;
- // This will let the caller of the method to know if the frame is
- // Active or non-Active The caller of the method knows that the
- // stream is encoded by codec and can use the info for callbacks,
- // if any registered.
- if(myBasicCodingBlockSmpl == 0)
- {
- // This codec can handle all allowed frame sizes as basic
- // coding block
- status = InternalEncode(bitStream, bitStreamLenByte);
-
- if(status < 0)
- {
- // TODO:
- // Maybe reseting the encoder to be fresh for the next
- // frame
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "EncodeSafe: error in internalEncode");
- *bitStreamLenByte = 0;
- *encodingType = kNoEncoding;
- }
- }
- else
- {
- // A basic-coding-block for this codec is defined so we loop
- // over the audio with the steps of the basic-coding-block.
- // It is not necessary that in each itteration
- WebRtc_Word16 tmpBitStreamLenByte;
-
- // Reset the variables which will be increamented in the loop
- *bitStreamLenByte = 0;
- bool done = false;
- while(!done)
- {
- status = InternalEncode(&bitStream[*bitStreamLenByte],
- &tmpBitStreamLenByte);
- *bitStreamLenByte += tmpBitStreamLenByte;
-
- // Guard Against errors and too large payloads
- if((status < 0) ||
- (*bitStreamLenByte > MAX_PAYLOAD_SIZE_BYTE))
- {
- // Error has happened if we are in the middle of a full
- // frame we have to exit. Before exiting, whatever bits
- // are in the buffer are probably corruptred. Anyways
- // we ignore them.
- *bitStreamLenByte = 0;
- *encodingType = kNoEncoding;
- // We might have come here because of the second
- // condition.
- status = -1;
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding,
- _uniqueID, "EncodeSafe: error in InternalEncode");
- // break from the loop
- break;
- }
-
- done = _inAudioIxRead >= _frameLenSmpl;
- }
- }
- if(status >= 0)
- {
- *encodingType = (_vadLabel[0] == 1)?
- kActiveNormalEncoded:kPassiveNormalEncoded;
- // Transport empty frame if we have an empty bitsteram
- if ((*bitStreamLenByte == 0) && ((_inAudioIxWrite - _inAudioIxRead) <= 0))
- {
- // Makes sure we transmit an empty frame
- *bitStreamLenByte = 1;
- *encodingType = kNoEncoding;
- }
- }
- }
- }
-
- // Move the timestampe buffer according to the number of 10 ms blocks
- // which are read.
- WebRtc_UWord16 sampFreqHz;
- EncoderSampFreq(sampFreqHz);
-
- WebRtc_Word16 num10MsecBlocks =
- (WebRtc_Word16)((_inAudioIxRead/_noChannels * 100) / sampFreqHz);
- if(_inTimestampIxWrite > num10MsecBlocks)
- {
- memmove(_inTimestamp, _inTimestamp + num10MsecBlocks,
- (_inTimestampIxWrite - num10MsecBlocks) * sizeof(WebRtc_Word32));
- }
- _inTimestampIxWrite -= num10MsecBlocks;
-
- // We have to move the audio that is not encoded to the beginning
- // of the buffer and accordingly adjust the read and write indices.
- if(_inAudioIxRead < _inAudioIxWrite)
- {
- memmove(_inAudio, &_inAudio[_inAudioIxRead],
- (_inAudioIxWrite - _inAudioIxRead)*sizeof(WebRtc_Word16));
- }
-
- _inAudioIxWrite -= _inAudioIxRead;
-
- _inAudioIxRead = 0;
- _lastEncodedTimestamp = *timeStamp;
- return (status < 0) ? (-1):(*bitStreamLenByte);
-}
-
-WebRtc_Word16
-ACMGenericCodec::Decode(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio,
- WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType)
-{
- WriteLockScoped wl(_codecWrapperLock);
- return DecodeSafe(bitStream, bitStreamLenByte, audio,
- audioSamples, speechType);
-}
-
-bool
-ACMGenericCodec::EncoderInitialized()
-{
- ReadLockScoped rl(_codecWrapperLock);
- return _encoderInitialized;
-}
-
-bool
-ACMGenericCodec::DecoderInitialized()
-{
- ReadLockScoped rl(_codecWrapperLock);
- return _decoderInitialized;
-}
-
-
-WebRtc_Word32
-ACMGenericCodec::RegisterInNetEq(
- ACMNetEQ* netEq,
- const CodecInst& codecInst)
-{
- WebRtcNetEQ_CodecDef codecDef;
- WriteLockScoped wl(_codecWrapperLock);
-
- if(CodecDef(codecDef, codecInst) < 0)
- {
- // Failed to register
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "RegisterInNetEq: error, failed to register");
- _registeredInNetEq = false;
- return -1;
- }
- else
- {
- if(netEq->AddCodec(&codecDef, _isMaster) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "RegisterInNetEq: error, failed to add codec");
- _registeredInNetEq = false;
- return -1;
- }
- // Registered
- _registeredInNetEq = true;
- return 0;
- }
-}
-
-WebRtc_Word16
-ACMGenericCodec::EncoderParams(
- WebRtcACMCodecParams* encParams)
-{
- ReadLockScoped rl(_codecWrapperLock);
- return EncoderParamsSafe(encParams);
-}
-
-WebRtc_Word16
-ACMGenericCodec::EncoderParamsSafe(
- WebRtcACMCodecParams* encParams)
-{
- // Codec parameters are valid only if the encoder is initialized
- if(_encoderInitialized)
- {
- WebRtc_Word32 currentRate;
- memcpy(encParams, &_encoderParams, sizeof(WebRtcACMCodecParams));
- currentRate = encParams->codecInstant.rate;
- CurrentRate(currentRate);
- encParams->codecInstant.rate = currentRate;
- return 0;
- }
- else
- {
- encParams->codecInstant.plname[0] = '\0';
- encParams->codecInstant.pltype = -1;
- encParams->codecInstant.pacsize = 0;
- encParams->codecInstant.rate = 0;
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "EncoderParamsSafe: error, encoder not initialized");
- return -1;
- }
-}
-
-bool
-ACMGenericCodec::DecoderParams(
- WebRtcACMCodecParams* decParams,
- const WebRtc_UWord8 payloadType)
-{
- ReadLockScoped rl(_codecWrapperLock);
- return DecoderParamsSafe(decParams, payloadType);
-}
-
-bool
-ACMGenericCodec::DecoderParamsSafe(
- WebRtcACMCodecParams* decParams,
- const WebRtc_UWord8 payloadType)
-{
- // Decoder parameters are valid only if decoder is initialized
- if(_decoderInitialized)
- {
- if(payloadType == _decoderParams.codecInstant.pltype)
- {
- memcpy(decParams, &_decoderParams, sizeof(WebRtcACMCodecParams));
- return true;
- }
- }
-
- decParams->codecInstant.plname[0] = '\0';
- decParams->codecInstant.pltype = -1;
- decParams->codecInstant.pacsize = 0;
- decParams->codecInstant.rate = 0;
- return false;
-}
-
-WebRtc_Word16
-ACMGenericCodec::ResetEncoder()
-{
- WriteLockScoped lockCodec(_codecWrapperLock);
- ReadLockScoped lockNetEq(*_netEqDecodeLock);
- return ResetEncoderSafe();
-}
-
-WebRtc_Word16
-ACMGenericCodec::ResetEncoderSafe()
-{
- if(!_encoderExist || !_encoderInitialized)
- {
- // We don't reset if doesn't exists or not initialized yet
- return 0;
- }
-
- _inAudioIxWrite = 0;
- _inAudioIxRead = 0;
- _inTimestampIxWrite = 0;
- _noMissedSamples = 0;
- _isAudioBuffFresh = true;
- memset(_inAudio, 0, AUDIO_BUFFER_SIZE_W16 * sizeof(WebRtc_Word16));
- memset(_inTimestamp, 0, TIMESTAMP_BUFFER_SIZE_W32 * sizeof(WebRtc_Word32));
-
- // Store DTX/VAD params
- bool enableVAD = _vadEnabled;
- bool enableDTX = _dtxEnabled;
- ACMVADMode mode = _vadMode;
-
- // Reset the encoder
- if(InternalResetEncoder() < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "ResetEncoderSafe: error in reset encoder");
- return -1;
- }
-
- // Disable DTX & VAD this deletes the states
- // we like to have fresh start
- DisableDTX();
- DisableVAD();
-
- // Set DTX/VAD
- return SetVADSafe(enableDTX, enableVAD, mode);
-}
-
-WebRtc_Word16
-ACMGenericCodec::InternalResetEncoder()
-{
- // For most of the codecs it is sufficient to
- // call their internal initialization.
- // There are some exceptions.
- // ----
- // For iSAC we don't want to lose BWE history,
- // so for iSAC we have to over-write this function.
- // ----
- return InternalInitEncoder(&_encoderParams);
-}
-
-WebRtc_Word16
-ACMGenericCodec::InitEncoder(
- WebRtcACMCodecParams* codecParams,
- bool forceInitialization)
-{
- WriteLockScoped lockCodec(_codecWrapperLock);
- ReadLockScoped lockNetEq(*_netEqDecodeLock);
- return InitEncoderSafe(codecParams, forceInitialization);
-}
-
-WebRtc_Word16
-ACMGenericCodec::InitEncoderSafe(
- WebRtcACMCodecParams* codecParams,
- bool forceInitialization)
-{
- // Check if we got a valid set of parameters
- int mirrorID;
- int codecNumber =
- ACMCodecDB::CodecNumber(&(codecParams->codecInstant), &mirrorID);
-
- if(codecNumber < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InitEncoderSafe: error, codec number negative");
- return -1;
- }
- // Check if the parameters are for this codec
- if((_codecID >= 0) && (_codecID != codecNumber) && (_codecID != mirrorID))
- {
- // The current codec is not the same as the one given by codecParams
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InitEncoderSafe: current codec is not the same as the one given by codecParams");
- return -1;
- }
-
- if(!CanChangeEncodingParam(codecParams->codecInstant))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InitEncoderSafe: cannot change encoding parameters");
- return -1;
- }
-
- if(_encoderInitialized && !forceInitialization)
- {
- // The encoder is already initialized
- return 0;
- }
- WebRtc_Word16 status;
- if(!_encoderExist)
- {
- _encoderInitialized = false;
- status = CreateEncoder();
- if(status < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InitEncoderSafe: cannot create encoder");
- return -1;
- }
- else
- {
- _encoderExist = true;
- }
- }
- _frameLenSmpl = (codecParams->codecInstant).pacsize;
- _noChannels = codecParams->codecInstant.channels;
- status = InternalInitEncoder(codecParams);
- if(status < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InitEncoderSafe: error in init encoder");
- _encoderInitialized = false;
- return -1;
- }
- else
- {
- memcpy(&_encoderParams, codecParams, sizeof(WebRtcACMCodecParams));
- _encoderInitialized = true;
- if(_inAudio == NULL)
- {
- _inAudio = new WebRtc_Word16[AUDIO_BUFFER_SIZE_W16];
- if(_inAudio == NULL)
- {
- return -1;
- }
- memset(_inAudio, 0, AUDIO_BUFFER_SIZE_W16 * sizeof(WebRtc_Word16));
- }
- if(_inTimestamp == NULL)
- {
- _inTimestamp = new WebRtc_UWord32[TIMESTAMP_BUFFER_SIZE_W32];
- if(_inTimestamp == NULL)
- {
- return -1;
- }
- memset(_inTimestamp, 0, sizeof(WebRtc_UWord32) *
- TIMESTAMP_BUFFER_SIZE_W32);
- }
- _isAudioBuffFresh = true;
- }
- status = SetVADSafe(codecParams->enableDTX, codecParams->enableVAD,
- codecParams->vadMode);
-
- return status;
-}
-
-bool
-ACMGenericCodec::CanChangeEncodingParam(
- CodecInst& /*codecInst*/)
-{
- return true;
-}
-
-WebRtc_Word16
-ACMGenericCodec::InitDecoder(
- WebRtcACMCodecParams* codecParams,
- bool forceInitialization)
-{
- WriteLockScoped lockCodc(_codecWrapperLock);
- WriteLockScoped lockNetEq(*_netEqDecodeLock);
- return InitDecoderSafe(codecParams, forceInitialization);
-}
-
-WebRtc_Word16
-ACMGenericCodec::InitDecoderSafe(
- WebRtcACMCodecParams* codecParams,
- bool forceInitialization)
-{
- int mirrorID;
- // Check if we got a valid set of parameters
- int codecNumber =
- ACMCodecDB::ReceiverCodecNumber(&codecParams->codecInstant, &mirrorID);
-
- if(codecNumber < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InitDecoderSafe: error, invalid codec number");
- return -1;
- }
- // Check if the parameters are for this codec
- if((_codecID >= 0) && (_codecID != codecNumber) && (_codecID != mirrorID))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InitDecoderSafe: current codec is not the same as the one given "
- "by codecParams");
- // The current codec is not the same as the one given by codecParams
- return -1;
- }
-
-
- if(_decoderInitialized && !forceInitialization)
- {
- // The encoder is already initialized
- return 0;
- }
-
- WebRtc_Word16 status;
- if(!_decoderExist)
- {
- _decoderInitialized = false;
- status = CreateDecoder();
- if(status < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InitDecoderSafe: cannot create decoder");
- return -1;
- }
- else
- {
- _decoderExist = true;
- }
- }
-
- status = InternalInitDecoder(codecParams);
- if(status < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InitDecoderSafe: cannot init decoder");
- _decoderInitialized = false;
- return -1;
- }
- else
- {
- // Store the parameters
- SaveDecoderParamSafe(codecParams);
- _decoderInitialized = true;
- }
- return 0;
-}
-
-WebRtc_Word16
-ACMGenericCodec::ResetDecoder(WebRtc_Word16 payloadType)
-{
- WriteLockScoped lockCodec(_codecWrapperLock);
- WriteLockScoped lockNetEq(*_netEqDecodeLock);
- return ResetDecoderSafe(payloadType);
-}
-
-WebRtc_Word16
-ACMGenericCodec::ResetDecoderSafe(WebRtc_Word16 payloadType)
-{
- WebRtcACMCodecParams decoderParams;
- if(!_decoderExist || !_decoderInitialized)
- {
- return 0;
- }
- // Initialization of the decoder should work for all
- // the codec. If there is a codec that has to keep
- // some states then we need to define a virtual and
- // overwrite in that codec
- DecoderParamsSafe(&decoderParams, (WebRtc_UWord8) payloadType);
- return InternalInitDecoder(&decoderParams);
-}
-
-void
-ACMGenericCodec::ResetNoMissedSamples()
-{
- WriteLockScoped cs(_codecWrapperLock);
- _noMissedSamples = 0;
-}
-
-void
-ACMGenericCodec::IncreaseNoMissedSamples(
- const WebRtc_Word16 noSamples)
-{
- _noMissedSamples += noSamples;
-}
-
-// Get the number of missed samples, this can be public
-WebRtc_UWord32
-ACMGenericCodec::NoMissedSamples() const
-{
- ReadLockScoped cs(_codecWrapperLock);
- return _noMissedSamples;
-}
-void
-ACMGenericCodec::DestructEncoder()
-{
- WriteLockScoped wl(_codecWrapperLock);
-
- // Disable VAD and delete the instance
- if(_ptrVADInst != NULL)
- {
- WebRtcVad_Free(_ptrVADInst);
- _ptrVADInst = NULL;
- }
- _vadEnabled = false;
- _vadMode = VADNormal;
-
- //Disable DTX and delete the instance
- _dtxEnabled = false;
- if(_ptrDTXInst != NULL)
- {
- WebRtcCng_FreeEnc(_ptrDTXInst);
- _ptrDTXInst = NULL;
- }
- _numLPCParams = kNewCNGNumPLCParams;
-
- DestructEncoderSafe();
-}
-
-void
-ACMGenericCodec::DestructDecoder()
-{
- WriteLockScoped wl(_codecWrapperLock);
- _decoderParams.codecInstant.pltype = -1;
- DestructDecoderSafe();
-}
-
-WebRtc_Word16
-ACMGenericCodec::SetBitRate(
- const WebRtc_Word32 bitRateBPS)
-{
- WriteLockScoped wl(_codecWrapperLock);
- return SetBitRateSafe(bitRateBPS);
-}
-
-WebRtc_Word16
-ACMGenericCodec::SetBitRateSafe(
- const WebRtc_Word32 bitRateBPS)
-{
- // If the codec can change the bit-rate this function
- // should be overwritten, otherewise the only acceptable
- // value is the one that is in database.
- CodecInst codecParams;
- if(ACMCodecDB::Codec(_codecID, &codecParams) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "SetBitRateSafe: error in ACMCodecDB::Codec");
- return -1;
- }
- if(codecParams.rate != bitRateBPS)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "SetBitRateSafe: rate value is not acceptable");
- return -1;
- }
- else
- {
- return 0;
- }
-}
-
-WebRtc_Word32
-ACMGenericCodec::GetEstimatedBandwidth()
-{
- WriteLockScoped wl(_codecWrapperLock);
- return GetEstimatedBandwidthSafe();
-}
-
-WebRtc_Word32
-ACMGenericCodec::GetEstimatedBandwidthSafe()
-{
- // All codecs but iSAC will return -1
- return -1;
-}
-
-WebRtc_Word32
-ACMGenericCodec::SetEstimatedBandwidth(
- WebRtc_Word32 estimatedBandwidth)
-{
- WriteLockScoped wl(_codecWrapperLock);
- return SetEstimatedBandwidthSafe(estimatedBandwidth);
-}
-
-WebRtc_Word32
-ACMGenericCodec::SetEstimatedBandwidthSafe(
- WebRtc_Word32 /*estimatedBandwidth*/)
-{
- // All codecs but iSAC will return -1
- return -1;
-}
-
-WebRtc_Word32
-ACMGenericCodec::GetRedPayload(
- WebRtc_UWord8* redPayload,
- WebRtc_Word16* payloadBytes)
-{
- WriteLockScoped wl(_codecWrapperLock);
- return GetRedPayloadSafe(redPayload, payloadBytes);
-}
-
-WebRtc_Word32
-ACMGenericCodec::GetRedPayloadSafe(
- WebRtc_UWord8* /* redPayload */,
- WebRtc_Word16* /* payloadBytes */)
-{
- return -1; // Do nothing by default
-}
-
-WebRtc_Word16
-ACMGenericCodec::CreateEncoder()
-{
- WebRtc_Word16 status = 0;
- if(!_encoderExist)
- {
- status = InternalCreateEncoder();
- // We just created the codec and obviously it is not initialized
- _encoderInitialized = false;
- }
-
- if(status < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "CreateEncoder: error in internal create encoder");
- _encoderExist = false;
- }
- else
- {
- _encoderExist = true;
- }
- return status;
-}
-
-WebRtc_Word16
-ACMGenericCodec::CreateDecoder()
-{
- WebRtc_Word16 status = 0;
- if(!_decoderExist)
- {
- status = InternalCreateDecoder();
- // Decoder just created and obviously it is not initialized
- _decoderInitialized = false;
- }
-
- if(status < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "CreateDecoder: error in internal create decoder");
- _decoderExist = false;
- }
- else
- {
- _decoderExist = true;
- }
- return status;
-}
-
-
-void ACMGenericCodec::DestructEncoderInst(void* ptrInst)
-{
- if(ptrInst != NULL)
- {
- WriteLockScoped lockCodec(_codecWrapperLock);
- ReadLockScoped lockNetEq(*_netEqDecodeLock);
- InternalDestructEncoderInst(ptrInst);
- }
-}
-
-
-WebRtc_Word16
-ACMGenericCodec::AudioBuffer(
- WebRtcACMAudioBuff& audioBuff)
-{
- ReadLockScoped cs(_codecWrapperLock);
- memcpy(audioBuff.inAudio, _inAudio,
- AUDIO_BUFFER_SIZE_W16 * sizeof(WebRtc_Word16));
- audioBuff.inAudioIxRead = _inAudioIxRead;
- audioBuff.inAudioIxWrite = _inAudioIxWrite;
- memcpy(audioBuff.inTimestamp, _inTimestamp,
- TIMESTAMP_BUFFER_SIZE_W32*sizeof(WebRtc_UWord32));
- audioBuff.inTimestampIxWrite = _inTimestampIxWrite;
- audioBuff.lastTimestamp = _lastTimestamp;
- return 0;
-}
-
-
-WebRtc_Word16
-ACMGenericCodec::SetAudioBuffer(
- WebRtcACMAudioBuff& audioBuff)
-{
- WriteLockScoped cs(_codecWrapperLock);
- memcpy(_inAudio, audioBuff.inAudio,
- AUDIO_BUFFER_SIZE_W16 * sizeof(WebRtc_Word16));
- _inAudioIxRead = audioBuff.inAudioIxRead;
- _inAudioIxWrite = audioBuff.inAudioIxWrite;
- memcpy(_inTimestamp, audioBuff.inTimestamp,
- TIMESTAMP_BUFFER_SIZE_W32*sizeof(WebRtc_UWord32));
- _inTimestampIxWrite = audioBuff.inTimestampIxWrite;
- _lastTimestamp = audioBuff.lastTimestamp;
- _isAudioBuffFresh = false;
- return 0;
-}
-
-
-WebRtc_UWord32
-ACMGenericCodec::LastEncodedTimestamp() const
-{
- ReadLockScoped cs(_codecWrapperLock);
- return _lastEncodedTimestamp;
-}
-
-
-WebRtc_UWord32
-ACMGenericCodec::EarliestTimestamp() const
-{
- ReadLockScoped cs(_codecWrapperLock);
- return _inTimestamp[0];
-}
-
-
-WebRtc_Word16
-ACMGenericCodec::SetVAD(
- const bool enableDTX,
- const bool enableVAD,
- const ACMVADMode mode)
-{
- WriteLockScoped cs(_codecWrapperLock);
- return SetVADSafe(enableDTX, enableVAD, mode);
-}
-
-
-WebRtc_Word16
-ACMGenericCodec::SetVADSafe(
- const bool enableDTX,
- const bool enableVAD,
- const ACMVADMode mode)
-{
- if(enableDTX)
- {
- // Make G729 AnnexB a special case
- if (!STR_CASE_CMP(_encoderParams.codecInstant.plname, "G729") && !_hasInternalDTX)
- {
- if (ACMGenericCodec::EnableDTX() < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "SetVADSafe: error in enable DTX");
- return -1;
- }
- }
- else
- {
- if(EnableDTX() < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "SetVADSafe: error in enable DTX");
- return -1;
- }
- }
-
- if(_hasInternalDTX)
- {
- // Codec has internal DTX, practically we don't need WebRtc VAD,
- // however, we let the user to turn it on if they need call-backs
- // on silence. Store VAD mode for future even if VAD is off.
- _vadMode = mode;
- return (enableVAD)? EnableVAD(mode):DisableVAD();
- }
- else
- {
- // Codec does not have internal DTX so enabling DTX requires an
- // active VAD. 'enableDTX == true' overwrites VAD status.
- if(EnableVAD(mode) < 0)
- {
- // If we cannot create VAD we have to disable DTX
- if(!_vadEnabled)
- {
- DisableDTX();
- }
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "SetVADSafe: error in enable VAD");
- return -1;
- }
-
- // Return '1', to let the caller know VAD was turned on, even if the
- // function was called with VAD='false'
- if (enableVAD == false) {
- return 1;
- } else {
- return 0;
- }
- }
- }
- else
- {
- // Make G729 AnnexB a special case
- if (!STR_CASE_CMP(_encoderParams.codecInstant.plname, "G729") && !_hasInternalDTX)
- {
- ACMGenericCodec::DisableDTX();
- }
- else
- {
- DisableDTX();
- }
- return (enableVAD)? EnableVAD(mode):DisableVAD();
- }
-}
-
-WebRtc_Word16
-ACMGenericCodec::EnableDTX()
-{
- if(_hasInternalDTX)
- {
- // We should not be here if we have internal DTX
- // this function should be overwritten by the derived
- // class in this case
- return -1;
- }
- if(!_dtxEnabled)
- {
- if(WebRtcCng_CreateEnc(&_ptrDTXInst) < 0)
- {
- _ptrDTXInst = NULL;
- return -1;
- }
- WebRtc_UWord16 freqHz;
- EncoderSampFreq(freqHz);
- if(WebRtcCng_InitEnc(_ptrDTXInst, (WebRtc_Word16)freqHz,
- ACM_SID_INTERVAL_MSEC, _numLPCParams) < 0)
- {
- // Couldn't initialize, has to return -1, and free the memory
- WebRtcCng_FreeEnc(_ptrDTXInst);
- _ptrDTXInst = NULL;
- return -1;
- }
- _dtxEnabled = true;
- }
- return 0;
-}
-
-WebRtc_Word16
-ACMGenericCodec::DisableDTX()
-{
- if(_hasInternalDTX)
- {
- // We should not be here if we have internal DTX
- // this function should be overwritten by the derived
- // class in this case
- return -1;
- }
- if(_ptrDTXInst != NULL)
- {
- WebRtcCng_FreeEnc(_ptrDTXInst);
- _ptrDTXInst = NULL;
- }
- _dtxEnabled = false;
- return 0;
-}
-
-WebRtc_Word16
-ACMGenericCodec::EnableVAD(
- ACMVADMode mode)
-{
- if((mode < VADNormal) || (mode > VADVeryAggr))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "EnableVAD: error in VAD mode range");
- return -1;
- }
-
- if(!_vadEnabled)
- {
- if(WebRtcVad_Create(&_ptrVADInst) < 0)
- {
- _ptrVADInst = NULL;
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "EnableVAD: error in create VAD");
- return -1;
- }
- if(WebRtcVad_Init(_ptrVADInst) < 0)
- {
- WebRtcVad_Free(_ptrVADInst);
- _ptrVADInst = NULL;
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "EnableVAD: error in init VAD");
- return -1;
- }
- }
-
- // Set the vad mode to the given value
- if(WebRtcVad_set_mode(_ptrVADInst, mode) < 0)
- {
- // We failed to set the mode and we have to return -1. If
- // we already have a working VAD (_vadEnabled == true) then
- // we leave it to work. otherwise, the following will be
- // executed.
- if(!_vadEnabled)
- {
- // We just created the instance but cannot set the mode
- // we have to free the memomry.
- WebRtcVad_Free(_ptrVADInst);
- _ptrVADInst = NULL;
- }
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _uniqueID,
- "EnableVAD: failed to set the VAD mode");
- return -1;
- }
- _vadMode = mode;
- _vadEnabled = true;
- return 0;
-}
-
-WebRtc_Word16
-ACMGenericCodec::DisableVAD()
-{
- if(_ptrVADInst != NULL)
- {
- WebRtcVad_Free(_ptrVADInst);
- _ptrVADInst = NULL;
- }
- _vadEnabled = false;
- return 0;
-}
-
-WebRtc_Word32
-ACMGenericCodec::ReplaceInternalDTX(
- const bool replaceInternalDTX)
-{
- WriteLockScoped cs(_codecWrapperLock);
- return ReplaceInternalDTXSafe(replaceInternalDTX);
-}
-
-WebRtc_Word32
-ACMGenericCodec::ReplaceInternalDTXSafe(
- const bool /* replaceInternalDTX */)
-{
- return -1;
-}
-
-WebRtc_Word32
-ACMGenericCodec::IsInternalDTXReplaced(
- bool* internalDTXReplaced)
-{
- WriteLockScoped cs(_codecWrapperLock);
- return IsInternalDTXReplacedSafe(internalDTXReplaced);
-}
-
-WebRtc_Word32
-ACMGenericCodec::IsInternalDTXReplacedSafe(
- bool* internalDTXReplaced)
-{
- *internalDTXReplaced = false;
- return 0;
-}
-
-WebRtc_Word16
-ACMGenericCodec::ProcessFrameVADDTX(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte,
- WebRtc_Word16* samplesProcessed)
-{
- if(!_vadEnabled)
- {
- // VAD not enabled, set all vadLable[] to 1 (speech detected)
- for(WebRtc_Word16 n = 0; n < MAX_FRAME_SIZE_10MSEC; n++)
- {
- _vadLabel[n] = 1;
- }
- *samplesProcessed = 0;
- return 0;
- }
- WebRtc_UWord16 freqHz;
- EncoderSampFreq(freqHz);
-
- // Calculate number of samples in 10 ms blocks, and number ms in one frame
- WebRtc_Word16 samplesIn10Msec = (WebRtc_Word16)(freqHz / 100);
- WebRtc_Word32 frameLenMsec = (((WebRtc_Word32)_frameLenSmpl * 1000) / freqHz);
- WebRtc_Word16 status;
-
- // Vector for storing maximum 30 ms of mono audio at 32 kHz
- WebRtc_Word16 audio[960];
-
- // Calculate number of VAD-blocks to process, and number of samples in each block.
- int noSamplesToProcess[2];
- if (frameLenMsec == 40)
- {
- // 20 ms in each VAD block
- noSamplesToProcess[0] = noSamplesToProcess[1] = 2*samplesIn10Msec;
- }
- else
- {
- // For 10-30 ms framesizes, second VAD block will be size zero ms,
- // for 50 and 60 ms first VAD block will be 30 ms.
- noSamplesToProcess[0] = (frameLenMsec > 30)? 3*samplesIn10Msec : _frameLenSmpl;
- noSamplesToProcess[1] = _frameLenSmpl-noSamplesToProcess[0];
- }
-
- int offSet = 0;
- int loops = (noSamplesToProcess[1]>0) ? 2 : 1;
- for (int i=0; i<loops; i++) {
- // If stereo, calculate mean of the two channels
- if(_noChannels == 2) {
- for (int j=0; j<noSamplesToProcess[i]; j++) {
- audio[j] = (_inAudio[(offSet+j)*2]+_inAudio[(offSet+j)*2+1])/2;
- }
- offSet = noSamplesToProcess[0];
- } else {
- // Mono, copy data from _inAudio to continue work on
- memcpy(audio, _inAudio, sizeof(WebRtc_Word16)*noSamplesToProcess[i]);
- }
-
- // Call VAD
- status = WebRtcVad_Process(_ptrVADInst, (WebRtc_Word16)freqHz,
- audio, noSamplesToProcess[i]);
-
- _vadLabel[i] = status;
-
- if(status < 0)
- {
- // This will force that the data be removed from the buffer
- *samplesProcessed += noSamplesToProcess[i];
- return -1;
- }
-
- // If VAD decision non-active, update DTX. NOTE! We only do this if the first part of
- // a frame gets the VAD decision "inactive". Otherwise DTX might say it is time to
- // transmit SID frame, but we will encode the whole frame, because the first part is
- // active.
- *samplesProcessed = 0;
- if((status == 0) && (i==0) && _dtxEnabled && !_hasInternalDTX)
- {
- WebRtc_Word16 bitStreamLen;
- WebRtc_Word16 num10MsecFrames = noSamplesToProcess[i] / samplesIn10Msec;
- *bitStreamLenByte = 0;
- for(WebRtc_Word16 n = 0; n < num10MsecFrames; n++)
- {
- // This block is (passive) && (vad enabled)
- status = WebRtcCng_Encode(_ptrDTXInst, &audio[n*samplesIn10Msec],
- samplesIn10Msec, bitStream, &bitStreamLen, 0);
- if (status < 0) {
- return -1;
- }
-
- *samplesProcessed += samplesIn10Msec*_noChannels;
-
- // bitStreamLen will only be > 0 once per 100 ms
- *bitStreamLenByte += bitStreamLen;
- }
-
-
- // Check if all samples got processed by the DTX
- if(*samplesProcessed != noSamplesToProcess[i]*_noChannels) {
- // Set to zero since something went wrong. Shouldn't happen.
- *samplesProcessed = 0;
- }
- }
-
- if(*samplesProcessed > 0)
- {
- // The block contains inactive speech, and is processed by DTX.
- // Discontinue running VAD.
- break;
- }
- }
-
- return status;
-}
-
-WebRtc_Word16
-ACMGenericCodec::SamplesLeftToEncode()
-{
- ReadLockScoped rl(_codecWrapperLock);
- return (_frameLenSmpl <= _inAudioIxWrite)?
- 0:(_frameLenSmpl - _inAudioIxWrite);
-}
-
-WebRtc_Word32
-ACMGenericCodec::UnregisterFromNetEq(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType)
-{
- WriteLockScoped wl(_codecWrapperLock);
- if(!_registeredInNetEq)
- {
- return 0;
- }
- if(UnregisterFromNetEqSafe(netEq, payloadType) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "UnregisterFromNetEq: error, cannot unregister from NetEq");
- _registeredInNetEq = true;
- return -1;
- }
- else
- {
- _registeredInNetEq = false;
- return 0;
- }
-}
-
-void
-ACMGenericCodec::SetUniqueID(
- const WebRtc_UWord32 id)
-{
- _uniqueID = id;
-}
-
-bool
-ACMGenericCodec::IsAudioBufferFresh() const
-{
- ReadLockScoped rl(_codecWrapperLock);
- return _isAudioBuffFresh;
-}
-
-// This function is replaced by codec specific functions for some codecs
-WebRtc_Word16
-ACMGenericCodec::EncoderSampFreq(WebRtc_UWord16& sampFreqHz)
-{
- WebRtc_Word32 f;
- f = ACMCodecDB::CodecFreq(_codecID);
- if(f < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "EncoderSampFreq: codec frequency is negative");
- return -1;
- }
- else
- {
- sampFreqHz = (WebRtc_UWord16)f;
- return 0;
- }
-}
-
-
-WebRtc_Word32
-ACMGenericCodec::ConfigISACBandwidthEstimator(
- const WebRtc_UWord8 /* initFrameSizeMsec */,
- const WebRtc_UWord16 /* initRateBitPerSec */,
- const bool /* enforceFrameSize */)
-{
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, _uniqueID,
- "The send-codec is not iSAC, failed to config iSAC bandwidth estimator.");
- return -1;
-}
-
-WebRtc_Word32
-ACMGenericCodec::SetISACMaxRate(
- const WebRtc_UWord32 /* maxRateBitPerSec */)
-{
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, _uniqueID,
- "The send-codec is not iSAC, failed to set iSAC max rate.");
- return -1;
-}
-
-WebRtc_Word32
-ACMGenericCodec::SetISACMaxPayloadSize(
- const WebRtc_UWord16 /* maxPayloadLenBytes */)
-{
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, _uniqueID,
- "The send-codec is not iSAC, failed to set iSAC max payload-size.");
- return -1;
-}
-
-
-void
-ACMGenericCodec::SaveDecoderParam(
- const WebRtcACMCodecParams* codecParams)
-{
- WriteLockScoped wl(_codecWrapperLock);
- SaveDecoderParamSafe(codecParams);
-}
-
-
-void
-ACMGenericCodec::SaveDecoderParamSafe(
- const WebRtcACMCodecParams* codecParams)
-{
- memcpy(&_decoderParams, codecParams, sizeof(WebRtcACMCodecParams));
-}
-
-WebRtc_Word16
-ACMGenericCodec::UpdateEncoderSampFreq(
- WebRtc_UWord16 /* encoderSampFreqHz */)
-{
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "It is asked for a change in smapling frequency while the \
-current send-codec supports only one sampling rate.");
- return -1;
-}
-
-
-void
-ACMGenericCodec::SetIsMaster(
- bool isMaster)
-{
- WriteLockScoped wl(_codecWrapperLock);
- _isMaster = isMaster;
-}
-
-
-
-WebRtc_Word16
-ACMGenericCodec::REDPayloadISAC(
- const WebRtc_Word32 /* isacRate */,
- const WebRtc_Word16 /* isacBwEstimate */,
- WebRtc_UWord8* /* payload */,
- WebRtc_Word16* /* payloadLenBytes */)
-{
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Error: REDPayloadISAC is an iSAC specific function");
- return -1;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_generic_codec.h b/trunk/src/modules/audio_coding/main/source/acm_generic_codec.h
deleted file mode 100644
index 7f8ef8c..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_generic_codec.h
+++ /dev/null
@@ -1,1343 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_
-
-#include "acm_common_defs.h"
-#include "audio_coding_module_typedefs.h"
-#include "rw_lock_wrapper.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-
-#define MAX_FRAME_SIZE_10MSEC 6
-
-// forward declaration
-struct WebRtcVadInst;
-struct WebRtcCngEncInst;
-
-namespace webrtc
-{
-
-// forward declaration
-struct CodecInst;
-class ACMNetEQ;
-
-class ACMGenericCodec
-{
-public:
- ///////////////////////////////////////////////////////////////////////////
- // Constructor of the class
- //
- ACMGenericCodec();
-
-
- ///////////////////////////////////////////////////////////////////////////
- // Destructor of the class.
- //
- virtual ~ACMGenericCodec();
-
-
- ///////////////////////////////////////////////////////////////////////////
- // ACMGenericCodec* CreateInstance();
- // The function will be used for FEC. It is not implemented yet.
- //
- virtual ACMGenericCodec* CreateInstance() = 0;
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 Encode()
- // The function is called to perform an encoding of the audio stored in
- // audio buffer. An encoding is performed only if enough audio, i.e. equal
- // to the frame-size of the codec, exist. The audio frame will be processed
- // by VAD and CN/DTX if required. There are few different cases.
- //
- // A) Neither VAD nor DTX is active; the frame is encoded by the encoder.
- //
- // B) VAD is enabled but not DTX; in this case the audio is processed by VAD
- // and encoded by the encoder. The "*encodingType" will be either
- // "activeNormalEncode" or "passiveNormalEncode" if frame is active or
- // passive, respectively.
- //
- // C) DTX is enabled; if the codec has internal VAD/DTX we just encode the
- // frame by the encoder. Otherwise, the frame is passed through VAD and
- // if identified as passive, then it will be processed by CN/DTX. If the
- // frame is active it will be encoded by the encoder.
- //
- // This function acquires the appropriate locks and calls EncodeSafe() for
- // the actual processing.
- //
- // Outputs:
- // -bitStream : a buffer where bit-stream will be written to.
- // -bitStreamLenByte : contains the length of the bit-stream in
- // bytes.
- // -timeStamp : contains the RTP timestamp, this is the
- // sampling time of the first sample encoded
- // (measured in number of samples).
- // -encodingType : contains the type of encoding applied on the
- // audio samples. The alternatives are
- // (c.f. acm_common_types.h)
- // -kNoEncoding:
- // there was not enough data to encode. or
- // some error has happened that we could
- // not do encoding.
- // -kActiveNormalEncoded:
- // the audio frame is active and encoded by
- // the given codec.
- // -kPassiveNormalEncoded:
- // the audio frame is passive but coded with
- // the given codec (NO DTX).
- // -kPassiveDTXWB:
- // The audio frame is passive and used
- // wide-band CN to encode.
- // -kPassiveDTXNB:
- // The audio frame is passive and used
- // narrow-band CN to encode.
- //
- // Return value:
- // -1 if error is occurred, otherwise the length of the bit-stream in
- // bytes.
- //
- WebRtc_Word16 Encode(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte,
- WebRtc_UWord32* timeStamp,
- WebRtcACMEncodingType* encodingType);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 Decode()
- // This function is used to decode a given bit-stream, without engaging
- // NetEQ.
- //
- // This function acquires the appropriate locks and calls DecodeSafe() for
- // the actual processing. Please note that this is not functional yet.
- //
- // Inputs:
- // -bitStream : a buffer where bit-stream will be read.
- // -bitStreamLenByte : the length of the bit-stream in bytes.
- //
- // Outputs:
- // -audio : pointer to a buffer where the audio will written.
- // -audioSamples : number of audio samples out of decoding the given
- // bit-stream.
- // -speechType : speech type (for future use).
- //
- // Return value:
- // -1 if failed to decode,
- // 0 if succeeded.
- //
- WebRtc_Word16 Decode(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio,
- WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // bool EncoderInitialized();
- //
- // Return value:
- // True if the encoder is successfully initialized,
- // false otherwise.
- //
- bool EncoderInitialized();
-
-
- ///////////////////////////////////////////////////////////////////////////
- // bool DecoderInitialized();
- //
- // Return value:
- // True if the decoder is successfully initialized,
- // false otherwise.
- //
- bool DecoderInitialized();
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 EncoderParams()
- // It is called to get encoder parameters. It will call
- // EncoderParamsSafe() in turn.
- //
- // Output:
- // -encParams : a buffer where the encoder parameters is
- // written to. If the encoder is not
- // initialized this buffer is filled with
- // invalid values
- // Return value:
- // -1 if the encoder is not initialized,
- // 0 otherwise.
- //
- //
- WebRtc_Word16 EncoderParams(
- WebRtcACMCodecParams *encParams);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 DecoderParams(...)
- // It is called to get decoder parameters. It will call DecoderParamsSafe()
- // in turn.
- //
- // Output:
- // -decParams : a buffer where the decoder parameters is
- // written to. If the decoder is not initialized
- // this buffer is filled with invalid values
- //
- // Return value:
- // -1 if the decoder is not initialized,
- // 0 otherwise.
- //
- //
- bool DecoderParams(
- WebRtcACMCodecParams *decParams,
- const WebRtc_UWord8 payloadType);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 InitEncoder(...)
- // This function is called to initialize the encoder with the given
- // parameters.
- //
- // Input:
- // -codecParams : parameters of encoder.
- // -forceInitialization: if false the initialization is invoked only if
- // the encoder is not initialized. If true the
- // encoder is forced to (re)initialize.
- //
- // Return value:
- // 0 if could initialize successfully,
- // -1 if failed to initialize.
- //
- //
- WebRtc_Word16 InitEncoder(
- WebRtcACMCodecParams* codecParams,
- bool forceInitialization);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 InitDecoder()
- // This function is called to initialize the decoder with the given
- // parameters. (c.f. acm_common_defs.h & common_types.h for the
- // definition of the structure)
- //
- // Input:
- // -codecParams : parameters of decoder.
- // -forceInitialization: if false the initialization is invoked only
- // if the decoder is not initialized. If true
- // the encoder is forced to(re)initialize.
- //
- // Return value:
- // 0 if could initialize successfully,
- // -1 if failed to initialize.
- //
- //
- WebRtc_Word16 InitDecoder(
- WebRtcACMCodecParams* codecParams,
- bool forceInitialization);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 RegisterInNetEq(...)
- // This function is called to register the decoder in NetEq, with the given
- // payload-type.
- //
- // Inputs:
- // -netEq : pointer to NetEq Instance
- // -codecInst : instance with of the codec settings of the codec
- //
- // Return values
- // -1 if failed to register,
- // 0 if successfully initialized.
- //
- WebRtc_Word32 RegisterInNetEq(
- ACMNetEQ* netEq,
- const CodecInst& codecInst);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 Add10MsData(...)
- // This function is called to add 10 ms of audio to the audio buffer of
- // the codec.
- //
- // Inputs:
- // -timeStamp : the timestamp of the 10 ms audio. the timestamp
- // is the sampling time of the
- // first sample measured in number of samples.
- // -data : a buffer that contains the audio. The codec
- // expects to get the audio in correct sampling
- // frequency
- // -length : the length of the audio buffer
- // -audioChannel : 0 for mono, 1 for stereo (not supported yet)
- //
- // Return values:
- // -1 if failed
- // 0 otherwise.
- //
- WebRtc_Word32 Add10MsData(
- const WebRtc_UWord32 timeStamp,
- const WebRtc_Word16* data,
- const WebRtc_UWord16 length,
- const WebRtc_UWord8 audioChannel);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_UWord32 NoMissedSamples()
- // This function returns the number of samples which are overwritten in
- // the audio buffer. The audio samples are overwritten if the input audio
- // buffer is full, but Add10MsData() is called. (We might remove this
- // function if it is not used)
- //
- // Return Value:
- // Number of samples which are overwritten.
- //
- WebRtc_UWord32 NoMissedSamples() const;
-
-
- ///////////////////////////////////////////////////////////////////////////
- // void ResetNoMissedSamples()
- // This function resets the number of overwritten samples to zero.
- // (We might remove this function if we remove NoMissedSamples())
- //
- void ResetNoMissedSamples();
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 SetBitRate()
- // The function is called to set the encoding rate.
- //
- // Input:
- // -bitRateBPS : encoding rate in bits per second
- //
- // Return value:
- // -1 if failed to set the rate, due to invalid input or given
- // codec is not rate-adjustable.
- // 0 if the rate is adjusted successfully
- //
- WebRtc_Word16 SetBitRate(const WebRtc_Word32 bitRateBPS);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // DestructEncoderInst()
- // This API is used in conferencing. It will free the memory that is pointed
- // by "ptrInst". "ptrInst" is a pointer to encoder instance, created and
- // filled up by calling EncoderInst(...).
- //
- // Inputs:
- // -ptrInst : pointer to an encoder instance to be deleted.
- //
- //
- void DestructEncoderInst(
- void* ptrInst);
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 AudioBuffer()
- // This is used when synchronization of codecs is required. There are cases
- // that the audio buffers of two codecs have to be synched. By calling this
- // function on can get the audio buffer and other related parameters, such
- // as timestamps...
- //
- // Output:
- // -audioBuff : a pointer to WebRtcACMAudioBuff where the audio
- // buffer of this codec will be written to.
- //
- // Return value:
- // -1 if fails to copy the audio buffer,
- // 0 if succeeded.
- //
- WebRtc_Word16 AudioBuffer(
- WebRtcACMAudioBuff& audioBuff);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_UWord32 EarliestTimestamp()
- // Returns the timestamp of the first 10 ms in audio buffer. This is used
- // to identify if a synchronization of two encoders is required.
- //
- // Return value:
- // timestamp of the first 10 ms audio in the audio buffer.
- //
- WebRtc_UWord32 EarliestTimestamp() const;
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 SetAudioBuffer()
- // This function is called to set the audio buffer and the associated
- // parameters to a given value.
- //
- // Return value:
- // -1 if fails to copy the audio buffer,
- // 0 if succeeded.
- //
- WebRtc_Word16 SetAudioBuffer(WebRtcACMAudioBuff& audioBuff);
-
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 SetVAD()
- // This is called to set VAD & DTX. If the codec has internal DTX that will
- // be used. If DTX is enabled and the codec does not have internal DTX,
- // WebRtc-VAD will be used to decide if the frame is active. If DTX is
- // disabled but VAD is enabled. The audio is passed through VAD to label it
- // as active or passive, but the frame is encoded normally. However the
- // bit-stream is labeled properly so that ACM::Process() can use this
- // information. In case of failure, the previous states of the VAD & DTX
- // are kept.
- //
- // Inputs:
- // -enableDTX : if true DTX will be enabled otherwise the DTX is
- // disabled. If codec has internal DTX that will be
- // used, otherwise WebRtc-CNG is used. In the latter
- // case VAD is automatically activated.
- // -enableVAD : if true WebRtc-VAD is enabled, otherwise VAD is
- // disabled, except for the case that DTX is enabled
- // but codec doesn't have internal DTX. In this case
- // VAD is enabled regardless of the value of
- // "enableVAD."
- // -mode : this specifies the aggressiveness of VAD.
- //
- // Return value
- // -1 if failed to set DTX & VAD as specified,
- // 0 if succeeded.
- //
- WebRtc_Word16 SetVAD(
- const bool enableDTX = true,
- const bool enableVAD = false,
- const ACMVADMode mode = VADNormal);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 ReplaceInternalDTX()
- // This is called to replace the codec internal DTX with WebRtc DTX.
- // This is only valid for G729 where the user has possibility to replace
- // AnnexB with WebRtc DTX. For other codecs this function has no effect.
- //
- // Input:
- // -replaceInternalDTX : if true the internal DTX is replaced with WebRtc.
- //
- // Return value
- // -1 if failed to replace internal DTX,
- // 0 if succeeded.
- //
- WebRtc_Word32 ReplaceInternalDTX(const bool replaceInternalDTX);
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 IsInternalDTXReplaced()
- // This is called to check if the codec internal DTX is replaced by WebRtc DTX.
- // This is only valid for G729 where the user has possibility to replace
- // AnnexB with WebRtc DTX. For other codecs this function has no effect.
- //
- // Output:
- // -internalDTXReplaced : if true the internal DTX is replaced with WebRtc.
- //
- // Return value
- // -1 if failed to check if replace internal DTX or replacement not feasible,
- // 0 if succeeded.
- //
- WebRtc_Word32 IsInternalDTXReplaced(bool* internalDTXReplaced);
-
- ///////////////////////////////////////////////////////////////////////////
- // void SetNetEqDecodeLock()
- // Passes the NetEq lock to the codec.
- //
- // Input:
- // -netEqDecodeLock : pointer to the lock associated with NetEQ of ACM.
- //
- void SetNetEqDecodeLock(
- RWLockWrapper* netEqDecodeLock)
- {
- _netEqDecodeLock = netEqDecodeLock;
- }
-
-
- ///////////////////////////////////////////////////////////////////////////
- // bool HasInternalDTX()
- // Used to check if the codec has internal DTX.
- //
- // Return value:
- // true if the codec has an internal DTX, e.g. G729,
- // false otherwise.
- //
- bool HasInternalDTX() const
- {
- return _hasInternalDTX;
- }
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 GetEstimatedBandwidth()
- // Used to get decoder estimated bandwidth. Only iSAC will provide a value.
- //
- //
- // Return value:
- // -1 if fails to get decoder estimated bandwidth,
- // >0 estimated bandwidth in bits/sec.
- //
- WebRtc_Word32 GetEstimatedBandwidth();
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 SetEstimatedBandwidth()
- // Used to set estiamted bandwidth sent out of band from other side. Only
- // iSAC will have use for the value.
- //
- // Input:
- // -estimatedBandwidth: estimated bandwidth in bits/sec
- //
- // Return value:
- // -1 if fails to set estimated bandwidth,
- // 0 on success.
- //
- WebRtc_Word32 SetEstimatedBandwidth(WebRtc_Word32 estimatedBandwidth);
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 GetRedPayload()
- // Used to get codec specific RED payload (if such is implemented).
- // Currently only done in iSAC.
- //
- // Outputs:
- // -redPayload : a pointer to the data for RED payload.
- // -payloadBytes : number of bytes in RED payload.
- //
- // Return value:
- // -1 if fails to get codec specific RED,
- // 0 if succeeded.
- //
- WebRtc_Word32 GetRedPayload(
- WebRtc_UWord8* redPayload,
- WebRtc_Word16* payloadBytes);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 ResetEncoder()
- // By calling this function you would re-initialize the encoder with the
- // current parameters. All the settings, e.g. VAD/DTX, frame-size... should
- // remain unchanged. (In case of iSAC we don't want to lose BWE history.)
- //
- // Return value
- // -1 if failed,
- // 0 if succeeded.
- //
- WebRtc_Word16 ResetEncoder();
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 ResetEncoder()
- // By calling this function you would re-initialize the decoder with the
- // current parameters.
- //
- // Return value
- // -1 if failed,
- // 0 if succeeded.
- //
- WebRtc_Word16 ResetDecoder(
- WebRtc_Word16 payloadType);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // void DestructEncoder()
- // This function is called to delete the encoder instance, if possible, to
- // have a fresh start. For codecs where encoder and decoder share the same
- // instance we cannot delete the encoder and instead we will initialize the
- // encoder. We also delete VAD and DTX if they have been created.
- //
- void DestructEncoder();
-
-
- ///////////////////////////////////////////////////////////////////////////
- // void DestructDecoder()
- // This function is called to delete the decoder instance, if possible, to
- // have a fresh start. For codecs where encoder and decoder share the same
- // instance we cannot delete the encoder and instead we will initialize the
- // decoder. Before deleting decoder instance it has to be removed from the
- // NetEq list.
- //
- void DestructDecoder();
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 SamplesLeftToEncode()
- // Returns the number of samples required to be able to do encoding.
- //
- // Return value:
- // Number of samples.
- //
- WebRtc_Word16 SamplesLeftToEncode();
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_UWord32 LastEncodedTimestamp()
- // Returns the timestamp of the last frame it encoded.
- //
- // Return value:
- // Timestamp.
- //
- WebRtc_UWord32 LastEncodedTimestamp() const;
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 UnregisterFromNetEq()
- // To remove the codec from NetEQ. If the codec (or the decoder instance)
- // is going to be deleted, first the codec has to be removed from NetEq
- // by calling this function.
- //
- // Input:
- // -netEq : pointer to a NetEq instance that the codec
- // has to be unregistered from.
- //
- // Output:
- // -1 if failed to unregister the codec,
- // 0 if the codec is successfully unregistered.
- //
- WebRtc_Word32 UnregisterFromNetEq(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // SetUniqueID()
- // Set a unique ID for the codec to be used for tracing and debuging
- //
- // Input
- // -id : A number to identify the codec.
- //
- void SetUniqueID(
- const WebRtc_UWord32 id);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // IsAudioBufferFresh()
- // Specifies if ever audio is injected to this codec.
- //
- // Return value
- // -true; no audio is feed into this codec
- // -false; audio has already been fed to the codec.
- //
- bool IsAudioBufferFresh() const;
-
-
- ///////////////////////////////////////////////////////////////////////////
- // UpdateDecoderSampFreq()
- // For most of the codecs this function does nothing. It must be
- // implemented for those codecs that one codec instance serves as the
- // decoder for different flavers of the codec. One example is iSAC. there,
- // iSAC 16 kHz and iSAC 32 kHz are treated as two different codecs with
- // different payload types, however, there is only one iSAC instance to
- // decode. The reason for that is we would like to decode and encode with
- // the same codec instance for bandwidth estimator to work.
- //
- // Each time that we receive a new payload type, we call this funtion to
- // prepare the decoder associated with the new payload. Normally, decoders
- // doesn't have to do anything. For iSAC the decoder has to change it's
- // sampling rate. The input parameter specifies the current flaver of the
- // codec in codec database. For instance, if we just got a SWB payload then
- // the input parameter is ACMCodecDB::isacswb.
- //
- // Input:
- // -codecId : the ID of the codec associated with the
- // payload type that we just received.
- //
- // Return value:
- // 0 if succeeded in updating the decoder.
- // -1 if failed to update.
- //
- virtual WebRtc_Word16 UpdateDecoderSampFreq(
- WebRtc_Word16 /* codecId */)
- {
- return 0;
- }
-
-
- ///////////////////////////////////////////////////////////////////////////
- // UpdateEncoderSampFreq()
- // Call this function to update the encoder sampling frequency. This
- // is for codecs where one payload-name supports several encoder sampling
- // frequencies. Otherwise, to change the sampling frequency we need to
- // register new codec. ACM will consider that as registration of a new
- // codec, not a change in parameter. For iSAC, switching from WB to SWB
- // is treated as a change in parameter. Therefore, we need this function.
- //
- // Input:
- // -encoderSampFreqHz : encoder sampling frequency.
- //
- // Return value:
- // -1 if failed, or if this is meaningless for the given codec.
- // 0 if succeeded.
- //
- virtual WebRtc_Word16 UpdateEncoderSampFreq(
- WebRtc_UWord16 encoderSampFreqHz);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // EncoderSampFreq()
- // Get the sampling frequency that the encoder (WebRtc wrapper) expects.
- //
- // Output:
- // -sampFreqHz : sampling frequency, in Hertz, which the encoder
- // should be fed with.
- //
- // Return value:
- // -1 if failed to output sampling rate.
- // 0 if the sample rate is returned successfully.
- //
- virtual WebRtc_Word16 EncoderSampFreq(
- WebRtc_UWord16& sampFreqHz);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word32 ConfigISACBandwidthEstimator()
- // Call this function to configure the bandwidth estimator of ISAC.
- // During the adaptation of bit-rate, iSAC atomatically adjusts the
- // frame-size (either 30 or 60 ms) to save on RTP header. The initial
- // frame-size can be specified by the first argument. The configuration also
- // regards the initial estimate of bandwidths. The estimator starts from
- // this point and converges to the actual bottleneck. This is given by the
- // second parameter. Furthermore, it is also possible to control the
- // adaptation of frame-size. This is specified by the last parameter.
- //
- // Input:
- // -initFrameSizeMsec : initial frame-size in milisecods. For iSAC-wb
- // 30 ms and 60 ms (default) are acceptable values,
- // and for iSAC-swb 30 ms is the only acceptable
- // value. Zero indiates default value.
- // -initRateBitPerSec : initial estimate of the bandwidth. Values
- // between 10000 and 58000 are acceptable.
- // -enforceFrameSize : if true, the frame-size will not be adapted.
- //
- // Return value:
- // -1 if failed to configure the bandwidth estimator,
- // 0 if the configuration was successfully applied.
- //
- virtual WebRtc_Word32 ConfigISACBandwidthEstimator(
- const WebRtc_UWord8 initFrameSizeMsec,
- const WebRtc_UWord16 initRateBitPerSec,
- const bool enforceFrameSize);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // SetISACMaxPayloadSize()
- // Set the maximum payload size of iSAC packets. No iSAC payload,
- // regardless of its frame-size, may exceed the given limit. For
- // an iSAC payload of size B bits and frame-size T sec we have;
- // (B < maxPayloadLenBytes * 8) and (B/T < maxRateBitPerSec), c.f.
- // SetISACMaxRate().
- //
- // Input:
- // -maxPayloadLenBytes : maximum payload size in bytes.
- //
- // Return value:
- // -1 if failed to set the maximm payload-size.
- // 0 if the given linit is seet successfully.
- //
- virtual WebRtc_Word32 SetISACMaxPayloadSize(
- const WebRtc_UWord16 maxPayloadLenBytes);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // SetISACMaxRate()
- // Set the maximum instantaneous rate of iSAC. For a payload of B bits
- // with a frame-size of T sec the instantaneous rate is B/T bist per
- // second. Therefore, (B/T < maxRateBitPerSec) and
- // (B < maxPayloadLenBytes * 8) are always satisfied for iSAC payloads,
- // c.f SetISACMaxPayloadSize().
- //
- // Input:
- // -maxRateBitPerSec : maximum instantaneous bit-rate given in bits/sec.
- //
- // Return value:
- // -1 if failed to set the maximum rate.
- // 0 if the maximum rate is set successfully.
- //
- virtual WebRtc_Word32 SetISACMaxRate(
- const WebRtc_UWord32 maxRateBitPerSec);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // SaveDecoderParamS()
- // Save the parameters of decoder.
- //
- // Input:
- // -codecParams : pointer to a struct where the parameters of
- // decoder is stored in.
- //
- void SaveDecoderParam(
- const WebRtcACMCodecParams* codecParams);
-
-
- WebRtc_Word32 FrameSize()
- {
- return _frameLenSmpl;
- }
-
- void SetIsMaster(bool isMaster);
-
-
-
-
- ///////////////////////////////////////////////////////////////////////////
- // REDPayloadISAC()
- // This is an iSAC-specific function. The function is called to get RED
- // paylaod from a default-encoder.
- //
- // Inputs:
- // -isacRate : the target rate of the main payload. A RED
- // paylaod is generated according to the rate of
- // main paylaod. Note that we are not specifying the
- // rate of RED payload, but the main payload.
- // -isacBwEstimate : bandwidth information should be inserted in
- // RED payload.
- //
- // Output:
- // -payload : pointer to a buffer where the RED paylaod will
- // written to.
- // -paylaodLenBytes : a place-holder to write the length of the RED
- // payload in Bytes.
- //
- // Return value:
- // -1 if an error occures, otherwise the length of the payload (in Bytes)
- // is returned.
- //
- //
- virtual WebRtc_Word16 REDPayloadISAC(
- const WebRtc_Word32 isacRate,
- const WebRtc_Word16 isacBwEstimate,
- WebRtc_UWord8* payload,
- WebRtc_Word16* payloadLenBytes);
-
- ///////////////////////////////////////////////////////////////////////////
- // IsTrueStereoCodec()
- // Call to see if current encoder is a true stereo codec. This function
- // should be overwritten for codecs which are true stereo codecs
- // Return value:
- // -true if stereo codec
- // -false if not stereo codec.
- //
- virtual bool IsTrueStereoCodec() {
- return false;
- }
-
-protected:
- ///////////////////////////////////////////////////////////////////////////
- // All the functions with FunctionNameSafe(...) contain the actual
- // implementation of FunctionName(...). FunctionName() acquires an
- // appropriate lock and calls FunctionNameSafe() to do the actual work.
- // Therefore, for the description of functionality, input/output arguments
- // and return value we refer to FunctionName()
- //
-
- ///////////////////////////////////////////////////////////////////////////
- // See Encode() for the description of function, input(s)/output(s) and
- // return value.
- //
- WebRtc_Word16 EncodeSafe(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte,
- WebRtc_UWord32* timeStamp,
- WebRtcACMEncodingType* encodingType);
-
- ///////////////////////////////////////////////////////////////////////////
- // See Decode() for the description of function, input(s)/output(s) and
- // return value.
- //
- virtual WebRtc_Word16 DecodeSafe(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio,
- WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // See Add10MsSafe() for the description of function, input(s)/output(s)
- // and return value.
- //
- virtual WebRtc_Word32 Add10MsDataSafe(
- const WebRtc_UWord32 timeStamp,
- const WebRtc_Word16* data,
- const WebRtc_UWord16 length,
- const WebRtc_UWord8 audioChannel);
-
- ///////////////////////////////////////////////////////////////////////////
- // See RegisterInNetEq() for the description of function,
- // input(s)/output(s) and return value.
- //
- virtual WebRtc_Word32 CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst) = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // See EncoderParam() for the description of function, input(s)/output(s)
- // and return value.
- //
- WebRtc_Word16 EncoderParamsSafe(
- WebRtcACMCodecParams *encParams);
-
- ///////////////////////////////////////////////////////////////////////////
- // See DecoderParam for the description of function, input(s)/output(s)
- // and return value.
- //
- // Note:
- // Any Class where a single instance handle several flavers of the
- // same codec, therefore, several payload types are associated with
- // the same instance have to implement this function.
- //
- // Currently only iSAC is implementing it. A single iSAC instance is
- // used for decoding both WB & SWB stream. At one moment both WB & SWB
- // can be registered as receive codec. Hence two payloads are associated
- // with a single codec instance.
- //
- virtual bool DecoderParamsSafe(
- WebRtcACMCodecParams *decParams,
- const WebRtc_UWord8 payloadType);
-
- ///////////////////////////////////////////////////////////////////////////
- // See ResetEncoder() for the description of function, input(s)/output(s)
- // and return value.
- //
- WebRtc_Word16 ResetEncoderSafe();
-
- ///////////////////////////////////////////////////////////////////////////
- // See InitEncoder() for the description of function, input(s)/output(s)
- // and return value.
- //
- WebRtc_Word16 InitEncoderSafe(
- WebRtcACMCodecParams *codecParams,
- bool forceInitialization);
-
- ///////////////////////////////////////////////////////////////////////////
- // See InitDecoder() for the description of function, input(s)/output(s)
- // and return value.
- //
- WebRtc_Word16 InitDecoderSafe(
- WebRtcACMCodecParams *codecParams,
- bool forceInitialization);
-
- ///////////////////////////////////////////////////////////////////////////
- // See ResetDecoder() for the description of function, input(s)/output(s)
- // and return value.
- //
- WebRtc_Word16 ResetDecoderSafe(
- WebRtc_Word16 payloadType);
-
- ///////////////////////////////////////////////////////////////////////////
- // See DestructEncoder() for the description of function,
- // input(s)/output(s) and return value.
- //
- virtual void DestructEncoderSafe() = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // See DestructDecoder() for the description of function,
- // input(s)/output(s) and return value.
- //
- virtual void DestructDecoderSafe() = 0;
-
- ///////////////////////////////////////////////////////////////////////////
- // See SetBitRate() for the description of function, input(s)/output(s)
- // and return value.
- //
- // Any codec that can change the bit-rate has to implement this.
- //
- virtual WebRtc_Word16 SetBitRateSafe(
- const WebRtc_Word32 bitRateBPS);
-
- ///////////////////////////////////////////////////////////////////////////
- // See GetEstimatedBandwidth() for the description of function, input(s)/output(s)
- // and return value.
- //
- virtual WebRtc_Word32 GetEstimatedBandwidthSafe();
-
- ///////////////////////////////////////////////////////////////////////////
- // See SetEstimatedBandwidth() for the description of function, input(s)/output(s)
- // and return value.
- //
- virtual WebRtc_Word32 SetEstimatedBandwidthSafe(WebRtc_Word32 estimatedBandwidth);
-
- ///////////////////////////////////////////////////////////////////////////
- // See GetRedPayload() for the description of function, input(s)/output(s)
- // and return value.
- //
- virtual WebRtc_Word32 GetRedPayloadSafe(
- WebRtc_UWord8* redPayload,
- WebRtc_Word16* payloadBytes);
-
- ///////////////////////////////////////////////////////////////////////////
- // See SetVAD() for the description of function, input(s)/output(s) and
- // return value.
- //
- WebRtc_Word16 SetVADSafe(
- const bool enableDTX = true,
- const bool enableVAD = false,
- const ACMVADMode mode = VADNormal);
-
- ///////////////////////////////////////////////////////////////////////////
- // See ReplaceInternalDTX() for the description of function, input and
- // return value.
- //
- virtual WebRtc_Word32 ReplaceInternalDTXSafe(
- const bool replaceInternalDTX);
-
- ///////////////////////////////////////////////////////////////////////////
- // See IsInternalDTXReplaced() for the description of function, input and
- // return value.
- //
- virtual WebRtc_Word32 IsInternalDTXReplacedSafe(
- bool* internalDTXReplaced);
-
- ///////////////////////////////////////////////////////////////////////////
- // See UnregisterFromNetEq() for the description of function,
- // input(s)/output(s) and return value.
- //
- virtual WebRtc_Word16 UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType) = 0;
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 CreateEncoder()
- // Creates the encoder instance.
- //
- // Return value:
- // -1 if failed,
- // 0 if succeeded.
- //
- WebRtc_Word16 CreateEncoder();
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 CreateDecoder()
- // Creates the decoder instance.
- //
- // Return value:
- // -1 if failed,
- // 0 if succeeded.
- //
- WebRtc_Word16 CreateDecoder();
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 EnableVAD();
- // Enables VAD with the given mode. The VAD instance will be created if
- // it does not exists.
- //
- // Input:
- // -mode : VAD mode c.f. audio_coding_module_typedefs.h for
- // the options.
- //
- // Return value:
- // -1 if failed,
- // 0 if succeeded.
- //
- WebRtc_Word16 EnableVAD(ACMVADMode mode);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 DisableVAD()
- // Disables VAD.
- //
- // Return value:
- // -1 if failed,
- // 0 if succeeded.
- //
- WebRtc_Word16 DisableVAD();
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 EnableDTX()
- // Enables DTX. This method should be overwritten for codecs which have
- // internal DTX.
- //
- // Return value:
- // -1 if failed,
- // 0 if succeeded.
- //
- virtual WebRtc_Word16 EnableDTX();
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 DisableDTX()
- // Disables usage of DTX. This method should be overwritten for codecs which
- // have internal DTX.
- //
- // Return value:
- // -1 if failed,
- // 0 if succeeded.
- //
- virtual WebRtc_Word16 DisableDTX();
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 InternalEncode()
- // This is a codec-specific function called in EncodeSafe() to actually
- // encode a frame of audio.
- //
- // Outputs:
- // -bitStream : pointer to a buffer where the bit-stream is
- // written to.
- // -bitStreamLenByte : the length of the bit-stream in byte, a negative
- // value indicates error.
- //
- // Return value:
- // -1 if failed,
- // otherwise the length of the bit-stream is returned.
- //
- virtual WebRtc_Word16 InternalEncode(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte) = 0;
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 InternalInitEncoder()
- // This is a codec-specific function called in InitEncoderSafe(), it has to
- // do all codec-specific operation to initialize the encoder given the
- // encoder parameters.
- //
- // Input:
- // -codecParams : pointer to a structure that contains parameters to
- // initialize encoder.
- // Set codecParam->CodecInst.rate to -1 for
- // iSAC to operate in adaptive mode.
- // (to do: if frame-length is -1 frame-length will be
- // automatically adjusted, otherwise, given
- // frame-length is forced)
- //
- // Return value:
- // -1 if failed,
- // 0 if succeeded.
- //
- virtual WebRtc_Word16 InternalInitEncoder(
- WebRtcACMCodecParams *codecParams) = 0;
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 InternalInitDecoder()
- // This is a codec-specific function called in InitDecoderSafe(), it has to
- // do all codec-specific operation to initialize the decoder given the
- // decoder parameters.
- //
- // Input:
- // -codecParams : pointer to a structure that contains parameters to
- // initialize encoder.
- //
- // Return value:
- // -1 if failed,
- // 0 if succeeded.
- //
- virtual WebRtc_Word16 InternalInitDecoder(
- WebRtcACMCodecParams *codecParams) = 0;
-
-
- ///////////////////////////////////////////////////////////////////////////
- // void IncreaseNoMissedSamples()
- // This method is called to increase the number of samples that are
- // overwritten in the audio buffer.
- //
- // Input:
- // -noSamples : the number of overwritten samples is incremented
- // by this value.
- //
- void IncreaseNoMissedSamples(
- const WebRtc_Word16 noSamples);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 InternalCreateEncoder()
- // This is a codec-specific method called in CreateEncoderSafe() it is
- // supposed to perform all codec-specific operations to create encoder
- // instance.
- //
- // Return value:
- // -1 if failed,
- // 0 if succeeded.
- //
- virtual WebRtc_Word16 InternalCreateEncoder() = 0;
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 InternalCreateDecoder()
- // This is a codec-specific method called in CreateDecoderSafe() it is
- // supposed to perform all codec-specific operations to create decoder
- // instance.
- //
- // Return value:
- // -1 if failed,
- // 0 if succeeded.
- //
- virtual WebRtc_Word16 InternalCreateDecoder() = 0;
-
-
- ///////////////////////////////////////////////////////////////////////////
- // void InternalDestructEncoderInst()
- // This is a codec-specific method, used in conferencing, called from
- // DestructEncoderInst(). The input argument is pointer to encoder instance
- // (codec instance for codecs that encoder and decoder share the same
- // instance). This method is called to free the memory that "ptrInst" is
- // pointing to.
- //
- // Input:
- // -ptrInst : pointer to encoder instance.
- //
- // Return value:
- // -1 if failed,
- // 0 if succeeded.
- //
- virtual void InternalDestructEncoderInst(
- void* ptrInst) = 0;
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 InternalResetEncoder()
- // This method is called to reset the states of encoder. However, the
- // current parameters, e.g. frame-length, should remain as they are. For
- // most of the codecs a re-initialization of the encoder is what needs to
- // be down. But for iSAC we like to keep the BWE history so we cannot
- // re-initialize. As soon as such an API is implemented in iSAC this method
- // has to be overwritten in ACMISAC class.
- //
- // Return value:
- // -1 if failed,
- // 0 if succeeded.
- //
- virtual WebRtc_Word16 InternalResetEncoder();
-
-
-
- ///////////////////////////////////////////////////////////////////////////
- // WebRtc_Word16 ProcessFrameVADDTX()
- // This function is called when a full frame of audio is available. It will
- // break the audio frame into blocks such that each block could be processed
- // by VAD & CN/DTX. If a frame is divided into two blocks then there are two
- // cases. First, the first block is active, the second block will not be
- // processed by CN/DTX but only by VAD and return to caller with
- // '*samplesProcessed' set to zero. There, the audio frame will be encoded
- // by the encoder. Second, the first block is inactive and is processed by
- // CN/DTX, then we stop processing the next block and return to the caller
- // which is EncodeSafe(), with "*samplesProcessed" equal to the number of
- // samples in first block.
- //
- // Output:
- // -bitStream : pointer to a buffer where DTX frame, if
- // generated, will be written to.
- // -bitStreamLenByte : contains the length of bit-stream in bytes, if
- // generated. Zero if no bit-stream is generated.
- // -noSamplesProcessed : contains no of samples that actually CN has
- // processed. Those samples processed by CN will not
- // be encoded by the encoder, obviously. If
- // contains zero, it means that the frame has been
- // identified as active by VAD. Note that
- // "*noSamplesProcessed" might be non-zero but
- // "*bitStreamLenByte" be zero.
- //
- // Return value:
- // -1 if failed,
- // 0 if succeeded.
- //
- WebRtc_Word16 ProcessFrameVADDTX(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte,
- WebRtc_Word16* samplesProcessed);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // CanChangeEncodingParam()
- // Check if the codec parameters can be changed. In conferencing normally
- // codec parametrs cannot be changed. The exception is bit-rate of isac.
- //
- // return value:
- // -true if codec parameters are allowed to change.
- // -flase otherwise.
- //
- virtual bool CanChangeEncodingParam(CodecInst& codecInst);
-
-
- ///////////////////////////////////////////////////////////////////////////
- // CurrentRate()
- // Call to get the current encoding rate of the encoder. This function
- // should be overwritten for codecs whic automatically change their
- // target rate. One example is iSAC. The output of the function is the
- // current target rate.
- //
- // Output:
- // -rateBitPerSec : the current target rate of the codec.
- //
- virtual void CurrentRate(
- WebRtc_Word32& /* rateBitPerSec */)
- {
- return;
- }
-
- virtual void SaveDecoderParamSafe(
- const WebRtcACMCodecParams* codecParams);
-
- // &_inAudio[_inAudioIxWrite] always point to where new audio can be
- // written to
- WebRtc_Word16 _inAudioIxWrite;
-
- // &_inAudio[_inAudioIxRead] points to where audio has to be read from
- WebRtc_Word16 _inAudioIxRead;
-
- WebRtc_Word16 _inTimestampIxWrite;
-
- // Where the audio is stored before encoding,
- // To save memory the following buffer can be allocated
- // dynamically for 80ms depending on the sampling frequency
- // of the codec.
- WebRtc_Word16* _inAudio;
- WebRtc_UWord32* _inTimestamp;
-
- WebRtc_Word16 _frameLenSmpl;
- WebRtc_UWord16 _noChannels;
-
- // This will point to a static database of the supported codecs
- WebRtc_Word16 _codecID;
-
- // This will account for the No of samples were not encoded
- // the case is rare, either samples are missed due to overwite
- // at input buffer or due to encoding error
- WebRtc_UWord32 _noMissedSamples;
-
- // True if the encoder instance created
- bool _encoderExist;
- bool _decoderExist;
- // True if the ecncoder instance initialized
- bool _encoderInitialized;
- bool _decoderInitialized;
-
- bool _registeredInNetEq;
-
- // VAD/DTX
- bool _hasInternalDTX;
- WebRtcVadInst* _ptrVADInst;
- bool _vadEnabled;
- ACMVADMode _vadMode;
- WebRtc_Word16 _vadLabel[MAX_FRAME_SIZE_10MSEC];
- bool _dtxEnabled;
- WebRtcCngEncInst* _ptrDTXInst;
- WebRtc_UWord8 _numLPCParams;
- bool _sentCNPrevious;
- bool _isMaster;
-
- WebRtcACMCodecParams _encoderParams;
- WebRtcACMCodecParams _decoderParams;
-
- // Used as a global lock for all avaiable decoders
- // so that no decoder is used when NetEQ decodes.
- RWLockWrapper* _netEqDecodeLock;
- // Used to lock wrapper internal data
- // such as buffers and state variables.
- RWLockWrapper& _codecWrapperLock;
-
- WebRtc_UWord32 _lastEncodedTimestamp;
- WebRtc_UWord32 _lastTimestamp;
- bool _isAudioBuffFresh;
- WebRtc_UWord32 _uniqueID;
-};
-
-} // namespace webrt
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GENERIC_CODEC_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_gsmfr.cc b/trunk/src/modules/audio_coding/main/source/acm_gsmfr.cc
deleted file mode 100644
index c1f817c..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_gsmfr.cc
+++ /dev/null
@@ -1,416 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_gsmfr.h"
-#include "acm_common_defs.h"
-#include "acm_neteq.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-
-#ifdef WEBRTC_CODEC_GSMFR
- // NOTE! GSM-FR is not included in the open-source package. The following
- // interface file is needed:
- //
- // /modules/audio_coding/codecs/gsmfr/main/interface/gsmfr_interface.h
- //
- // The API in the header file should match the one below.
- //
- // int16_t WebRtcGSMFR_CreateEnc(GSMFR_encinst_t_** inst);
- // int16_t WebRtcGSMFR_CreateDec(GSMFR_decinst_t_** inst);
- // int16_t WebRtcGSMFR_FreeEnc(GSMFR_encinst_t_* inst);
- // int16_t WebRtcGSMFR_FreeDec(GSMFR_decinst_t_* inst);
- // int16_t WebRtcGSMFR_Encode(GSMFR_encinst_t_* encInst, int16_t* input,
- // int16_t len, int16_t* output);
- // int16_t WebRtcGSMFR_EncoderInit(GSMFR_encinst_t_* encInst, int16_t mode);
- // int16_t WebRtcGSMFR_Decode(GSMFR_decinst_t_* decInst);
- // int16_t WebRtcGSMFR_DecodeBwe(GSMFR_decinst_t_* decInst, int16_t* input);
- // int16_t WebRtcGSMFR_DecodePlc(GSMFR_decinst_t_* decInst);
- // int16_t WebRtcGSMFR_DecoderInit(GSMFR_decinst_t_* decInst);
- // void WebRtcGSMFR_Version(char *versionStr, short len);
- #include "gsmfr_interface.h"
-#endif
-
-namespace webrtc {
-
-#ifndef WEBRTC_CODEC_GSMFR
-
-ACMGSMFR::ACMGSMFR(WebRtc_Word16 /* codecID */)
- : _encoderInstPtr(NULL),
- _decoderInstPtr(NULL) {
- return;
-}
-
-
-ACMGSMFR::~ACMGSMFR()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMGSMFR::InternalEncode(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMGSMFR::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMGSMFR::EnableDTX()
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMGSMFR::DisableDTX()
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMGSMFR::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMGSMFR::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
-}
-
-
-WebRtc_Word32
-ACMGSMFR::CodecDef(
- WebRtcNetEQ_CodecDef& /* codecDef */,
- const CodecInst& /* codecInst */)
-{
- return -1;
-}
-
-
-ACMGenericCodec*
-ACMGSMFR::CreateInstance(void)
-{
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMGSMFR::InternalCreateEncoder()
-{
- return -1;
-}
-
-
-void
-ACMGSMFR::DestructEncoderSafe()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMGSMFR::InternalCreateDecoder()
-{
- return -1;
-}
-
-
-void
-ACMGSMFR::DestructDecoderSafe()
-{
- return;
-}
-
-
-void
-ACMGSMFR::InternalDestructEncoderInst(
- void* /* ptrInst */)
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMGSMFR::UnregisterFromNetEqSafe(
- ACMNetEQ* /* netEq */,
- WebRtc_Word16 /* payloadType */)
-{
- return -1;
-}
-
-
-#else //===================== Actual Implementation =======================
-
-ACMGSMFR::ACMGSMFR(
- WebRtc_Word16 codecID):
-_encoderInstPtr(NULL),
-_decoderInstPtr(NULL)
-{
- _codecID = codecID;
- _hasInternalDTX = true;
- return;
-}
-
-
-ACMGSMFR::~ACMGSMFR()
-{
- if(_encoderInstPtr != NULL)
- {
- WebRtcGSMFR_FreeEnc(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
- if(_decoderInstPtr != NULL)
- {
- WebRtcGSMFR_FreeDec(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
- return;
-}
-
-
-WebRtc_Word16
-ACMGSMFR::InternalEncode(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte)
-{
- *bitStreamLenByte = WebRtcGSMFR_Encode(_encoderInstPtr,
- &_inAudio[_inAudioIxRead], _frameLenSmpl, (WebRtc_Word16*)bitStream);
- // increment the read index this tell the caller that how far
- // we have gone forward in reading the audio buffer
- _inAudioIxRead += _frameLenSmpl;
- return *bitStreamLenByte;
-}
-
-
-WebRtc_Word16
-ACMGSMFR::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return 0;
-}
-
-
-WebRtc_Word16
-ACMGSMFR::EnableDTX()
-{
- if(_dtxEnabled)
- {
- return 0;
- }
- else if(_encoderExist)
- {
- if(WebRtcGSMFR_EncoderInit(_encoderInstPtr, 1) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "EnableDTX: cannot init encoder for GSMFR");
- return -1;
- }
- _dtxEnabled = true;
- return 0;
- }
- else
- {
- return -1;
- }
-}
-
-
-WebRtc_Word16
-ACMGSMFR::DisableDTX()
-{
- if(!_dtxEnabled)
- {
- return 0;
- }
- else if(_encoderExist)
- {
- if(WebRtcGSMFR_EncoderInit(_encoderInstPtr, 0) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "DisableDTX: cannot init encoder for GSMFR");
- return -1;
- }
- _dtxEnabled = false;
- return 0;
- }
- else
- {
- // encoder doesn't exists, therefore disabling is harmless
- return 0;
- }
-}
-
-
-WebRtc_Word16
-ACMGSMFR::InternalInitEncoder(
- WebRtcACMCodecParams* codecParams)
-{
- if (WebRtcGSMFR_EncoderInit(_encoderInstPtr, ((codecParams->enableDTX)? 1:0)) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalInitEncoder: cannot init encoder for GSMFR");
- }
- return 0;
-}
-
-
-WebRtc_Word16
-ACMGSMFR::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- if (WebRtcGSMFR_DecoderInit(_decoderInstPtr) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalInitDecoder: cannot init decoder for GSMFR");
- return -1;
- }
- return 0;
-}
-
-
-WebRtc_Word32
-ACMGSMFR::CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst)
-{
- if (!_decoderInitialized)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "CodecDef: decoder is not initialized for GSMFR");
- return -1;
- }
- // Fill up the structure by calling
- // "SET_CODEC_PAR" & "SET_GSMFR_FUNCTION."
- // Then call NetEQ to add the codec to it's
- // database.
- SET_CODEC_PAR((codecDef), kDecoderGSMFR, codecInst.pltype,
- _decoderInstPtr, 8000);
- SET_GSMFR_FUNCTIONS((codecDef));
- return 0;
-}
-
-
-ACMGenericCodec*
-ACMGSMFR::CreateInstance(void)
-{
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMGSMFR::InternalCreateEncoder()
-{
- if (WebRtcGSMFR_CreateEnc(&_encoderInstPtr) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalCreateEncoder: cannot create instance for GSMFR encoder");
- return -1;
- }
- return 0;
-}
-
-
-void
-ACMGSMFR::DestructEncoderSafe()
-{
- if(_encoderInstPtr != NULL)
- {
- WebRtcGSMFR_FreeEnc(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
- _encoderExist = false;
- _encoderInitialized = false;
-}
-
-
-WebRtc_Word16
-ACMGSMFR::InternalCreateDecoder()
-{
- if (WebRtcGSMFR_CreateDec(&_decoderInstPtr) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalCreateDecoder: cannot create instance for GSMFR decoder");
- return -1;
- }
- return 0;
-}
-
-
-void
-ACMGSMFR::DestructDecoderSafe()
-{
- if(_decoderInstPtr != NULL)
- {
- WebRtcGSMFR_FreeDec(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
- _decoderExist = false;
- _decoderInitialized = false;
-}
-
-
-void
-ACMGSMFR::InternalDestructEncoderInst(
- void* ptrInst)
-{
- if(ptrInst != NULL)
- {
- WebRtcGSMFR_FreeEnc((GSMFR_encinst_t_*)ptrInst);
- }
- return;
-}
-
-
-WebRtc_Word16
-ACMGSMFR::UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType)
-{
- if(payloadType != _decoderParams.codecInstant.pltype)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot unregister codec: payload-type does not match \
-the stored payload type",
- _decoderParams.codecInstant.plname,
- payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
- return netEq->RemoveCodec(kDecoderGSMFR);
-}
-
-#endif
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_gsmfr.h b/trunk/src/modules/audio_coding/main/source/acm_gsmfr.h
deleted file mode 100644
index 43e05fc..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_gsmfr.h
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_
-
-#include "acm_generic_codec.h"
-
-// forward declaration
-struct GSMFR_encinst_t_;
-struct GSMFR_decinst_t_;
-
-namespace webrtc {
-
-class ACMGSMFR : public ACMGenericCodec
-{
-public:
- ACMGSMFR(WebRtc_Word16 codecID);
- ~ACMGSMFR();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(
- WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(
- WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 InternalInitDecoder(
- WebRtcACMCodecParams *codecParams);
-
-protected:
- WebRtc_Word16 DecodeSafe(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio,
- WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- void InternalDestructEncoderInst(
- void* ptrInst);
-
- WebRtc_Word16 EnableDTX();
-
- WebRtc_Word16 DisableDTX();
-
- WebRtc_Word16 UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- GSMFR_encinst_t_* _encoderInstPtr;
- GSMFR_decinst_t_* _decoderInstPtr;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_GSMFR_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_ilbc.cc b/trunk/src/modules/audio_coding/main/source/acm_ilbc.cc
deleted file mode 100644
index f16e2f8..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_ilbc.cc
+++ /dev/null
@@ -1,390 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_common_defs.h"
-#include "acm_ilbc.h"
-#include "acm_neteq.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-
-#ifdef WEBRTC_CODEC_ILBC
- #include "ilbc.h"
-#endif
-
-namespace webrtc
-{
-
-#ifndef WEBRTC_CODEC_ILBC
-
-ACMILBC::ACMILBC(WebRtc_Word16 /* codecID */)
- : _encoderInstPtr(NULL),
- _decoderInstPtr(NULL) {
- return;
-}
-
-
-ACMILBC::~ACMILBC()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMILBC::InternalEncode(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMILBC::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMILBC::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMILBC::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
-}
-
-
-WebRtc_Word32
-ACMILBC::CodecDef(
- WebRtcNetEQ_CodecDef& /* codecDef */,
- const CodecInst& /* codecInst */)
-{
- return -1;
-}
-
-
-ACMGenericCodec*
-ACMILBC::CreateInstance(void)
-{
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMILBC::InternalCreateEncoder()
-{
- return -1;
-}
-
-
-void
-ACMILBC::DestructEncoderSafe()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMILBC::InternalCreateDecoder()
-{
- return -1;
-}
-
-
-void
-ACMILBC::DestructDecoderSafe()
-{
- return;
-}
-
-
-void
-ACMILBC::InternalDestructEncoderInst(
- void* /* ptrInst */)
-{
- return;
-}
-
-WebRtc_Word16
-ACMILBC::SetBitRateSafe(const WebRtc_Word32 /* rate */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMILBC::UnregisterFromNetEqSafe(
- ACMNetEQ* /* netEq */,
- WebRtc_Word16 /* payloadType */)
-{
- return -1;
-}
-
-
-#else //===================== Actual Implementation =======================
-
-
-ACMILBC::ACMILBC(
- WebRtc_Word16 codecID):
-_encoderInstPtr(NULL),
-_decoderInstPtr(NULL)
-{
- _codecID = codecID;
- return;
-}
-
-
-ACMILBC::~ACMILBC()
-{
- if(_encoderInstPtr != NULL)
- {
- WebRtcIlbcfix_EncoderFree(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
- if(_decoderInstPtr != NULL)
- {
- WebRtcIlbcfix_DecoderFree(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
- return;
-}
-
-
-WebRtc_Word16
-ACMILBC::InternalEncode(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte)
-{
- *bitStreamLenByte = WebRtcIlbcfix_Encode(_encoderInstPtr,
- &_inAudio[_inAudioIxRead], _frameLenSmpl, (WebRtc_Word16*)bitStream);
- if (*bitStreamLenByte < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalEncode: error in encode for ILBC");
- return -1;
- }
- // increment the read index this tell the caller that how far
- // we have gone forward in reading the audio buffer
- _inAudioIxRead += _frameLenSmpl;
- return *bitStreamLenByte;
-}
-
-
-WebRtc_Word16
-ACMILBC::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return 0;
-}
-
-
-WebRtc_Word16
-ACMILBC::InternalInitEncoder(
- WebRtcACMCodecParams* codecParams)
-{
- // initialize with a correct processing block length
- if((160 == (codecParams->codecInstant).pacsize) ||
- (320 == (codecParams->codecInstant).pacsize))
- {
- // processing block of 20ms
- return WebRtcIlbcfix_EncoderInit(_encoderInstPtr, 20);
- }
- else if((240 == (codecParams->codecInstant).pacsize) ||
- (480 == (codecParams->codecInstant).pacsize))
- {
- // processing block of 30ms
- return WebRtcIlbcfix_EncoderInit(_encoderInstPtr, 30);
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalInitEncoder: invalid processing block");
- return -1;
- }
-}
-
-
-WebRtc_Word16
-ACMILBC::InternalInitDecoder(
- WebRtcACMCodecParams* codecParams)
-{
- // initialize with a correct processing block length
- if((160 == (codecParams->codecInstant).pacsize) ||
- (320 == (codecParams->codecInstant).pacsize))
- {
- // processing block of 20ms
- return WebRtcIlbcfix_DecoderInit(_decoderInstPtr, 20);
- }
- else if((240 == (codecParams->codecInstant).pacsize) ||
- (480 == (codecParams->codecInstant).pacsize))
- {
- // processing block of 30ms
- return WebRtcIlbcfix_DecoderInit(_decoderInstPtr, 30);
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalInitDecoder: invalid processing block");
- return -1;
- }
-}
-
-
-WebRtc_Word32
-ACMILBC::CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst)
-{
- if (!_decoderInitialized)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "CodeDef: decoder not initialized for ILBC");
- return -1;
- }
- // Fill up the structure by calling
- // "SET_CODEC_PAR" & "SET_ILBC_FUNCTION."
- // Then return the structure back to NetEQ to add the codec to it's
- // database.
- SET_CODEC_PAR((codecDef), kDecoderILBC, codecInst.pltype,
- _decoderInstPtr, 8000);
- SET_ILBC_FUNCTIONS((codecDef));
- return 0;
-}
-
-
-ACMGenericCodec*
-ACMILBC::CreateInstance(void)
-{
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMILBC::InternalCreateEncoder()
-{
- if (WebRtcIlbcfix_EncoderCreate(&_encoderInstPtr) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalCreateEncoder: cannot create instance for ILBC encoder");
- return -1;
- }
- return 0;
-}
-
-
-void
-ACMILBC::DestructEncoderSafe()
-{
- _encoderInitialized = false;
- _encoderExist = false;
- if(_encoderInstPtr != NULL)
- {
- WebRtcIlbcfix_EncoderFree(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
-}
-
-
-WebRtc_Word16
-ACMILBC::InternalCreateDecoder()
-{
- if (WebRtcIlbcfix_DecoderCreate(&_decoderInstPtr) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalCreateDecoder: cannot create instance for ILBC decoder");
- return -1;
- }
- return 0;
-}
-
-
-void
-ACMILBC::DestructDecoderSafe()
-{
- _decoderInitialized = false;
- _decoderExist = false;
- if(_decoderInstPtr != NULL)
- {
- WebRtcIlbcfix_DecoderFree(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
-}
-
-
-void
-ACMILBC::InternalDestructEncoderInst(
- void* ptrInst)
-{
- if(ptrInst != NULL)
- {
- WebRtcIlbcfix_EncoderFree((iLBC_encinst_t_*)ptrInst);
- }
- return;
-}
-
-WebRtc_Word16
-ACMILBC::SetBitRateSafe(const WebRtc_Word32 rate)
-{
- // Check that rate is valid. No need to store the value
- if (rate == 13300)
- {
- WebRtcIlbcfix_EncoderInit(_encoderInstPtr, 30);
- }
- else if (rate == 15200)
- {
- WebRtcIlbcfix_EncoderInit(_encoderInstPtr, 20);
- }
- else
- {
- return -1;
- }
- _encoderParams.codecInstant.rate = rate;
-
- return 0;
-}
-
-WebRtc_Word16
-ACMILBC::UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType)
-{
- if(payloadType != _decoderParams.codecInstant.pltype)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot unregister codec: given payload-type does not match \
-the stored payload type",
- _decoderParams.codecInstant.plname,
- payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
- return netEq->RemoveCodec(kDecoderILBC);
-}
-
-#endif
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_ilbc.h b/trunk/src/modules/audio_coding/main/source/acm_ilbc.h
deleted file mode 100644
index 2d3e420..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_ilbc.h
+++ /dev/null
@@ -1,78 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_
-
-#include "acm_generic_codec.h"
-
-// forward declaration
-struct iLBC_encinst_t_;
-struct iLBC_decinst_t_;
-
-namespace webrtc
-{
-
-class ACMILBC : public ACMGenericCodec
-{
-public:
- ACMILBC(WebRtc_Word16 codecID);
- ~ACMILBC();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(
- WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(
- WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 InternalInitDecoder(
- WebRtcACMCodecParams *codecParams);
-
-protected:
- WebRtc_Word16 DecodeSafe(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio,
- WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
-
- WebRtc_Word16 SetBitRateSafe(
- const WebRtc_Word32 rate);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- void InternalDestructEncoderInst(
- void* ptrInst);
-
- WebRtc_Word16 UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- iLBC_encinst_t_* _encoderInstPtr;
- iLBC_decinst_t_* _decoderInstPtr;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ILBC_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_isac.cc b/trunk/src/modules/audio_coding/main/source/acm_isac.cc
deleted file mode 100644
index adbb9c5..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_isac.cc
+++ /dev/null
@@ -1,1241 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_codec_database.h"
-#include "acm_common_defs.h"
-#include "acm_isac.h"
-#include "acm_neteq.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-
-
-#ifdef WEBRTC_CODEC_ISAC
- #include "acm_isac_macros.h"
- #include "isac.h"
-#endif
-
-#ifdef WEBRTC_CODEC_ISACFX
- #include "acm_isac_macros.h"
- #include "isacfix.h"
-#endif
-
-namespace webrtc
-{
-
-// we need this otherwise we cannot use forward declaration
-// in the header file
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
-struct ACMISACInst
-{
- ACM_ISAC_STRUCT *inst;
-};
-#endif
-
-#define ISAC_MIN_RATE 10000
-#define ISAC_MAX_RATE 56000
-
-
-// How the scaling is computed. iSAC computes a gain based on the
-// bottleneck. It follows the following expression for that
-//
-// G(BN_kbps) = pow(10, (a + b * BN_kbps + c * BN_kbps * BN_kbps) / 20.0)
-// / 3.4641;
-//
-// Where for 30 ms framelength we have,
-//
-// a = -23; b = 0.48; c = 0;
-//
-// As the default encoder is operating at 32kbps we have the scale as
-//
-// S(BN_kbps) = G(BN_kbps) / G(32);
-
-#define ISAC_NUM_SUPPORTED_RATES 9
-const WebRtc_UWord16 isacSuportedRates[ISAC_NUM_SUPPORTED_RATES] = {
- 32000, 30000, 26000, 23000, 21000,
- 19000, 17000, 15000, 12000};
-
-const float isacScale[ISAC_NUM_SUPPORTED_RATES] = {
- 1.0f, 0.8954f, 0.7178f, 0.6081f, 0.5445f,
- 0.4875f, 0.4365f, 0.3908f, 0.3311f};
-
-// Tables for bandwidth estimates
-#define NR_ISAC_BANDWIDTHS 24
-const WebRtc_Word32 isacRatesWB[NR_ISAC_BANDWIDTHS] =
-{
- 10000, 11100, 12300, 13700, 15200, 16900,
- 18800, 20900, 23300, 25900, 28700, 31900,
- 10100, 11200, 12400, 13800, 15300, 17000,
- 18900, 21000, 23400, 26000, 28800, 32000};
-
-
-const WebRtc_Word32 isacRatesSWB[NR_ISAC_BANDWIDTHS] =
-{
- 10000, 11000, 12400, 13800, 15300, 17000,
- 18900, 21000, 23200, 25400, 27600, 29800,
- 32000, 34100, 36300, 38500, 40700, 42900,
- 45100, 47300, 49500, 51700, 53900, 56000,
-};
-
-#if (!defined(WEBRTC_CODEC_ISAC) && !defined(WEBRTC_CODEC_ISACFX))
-
-ACMISAC::ACMISAC(WebRtc_Word16 /* codecID */)
- : _codecInstPtr(NULL),
- _isEncInitialized(false),
- _isacCodingMode(CHANNEL_INDEPENDENT),
- _enforceFrameSize(false),
- _isacCurrentBN(32000),
- _samplesIn10MsAudio(160) { // Initiates to 16 kHz mode.
- // Initiate decoder parameters for the 32 kHz mode.
- memset(&_decoderParams32kHz, 0, sizeof(WebRtcACMCodecParams));
- _decoderParams32kHz.codecInstant.pltype = -1;
-
- return;
-}
-
-
-ACMISAC::~ACMISAC()
-{
- return;
-}
-
-
-ACMGenericCodec*
-ACMISAC::CreateInstance(void)
-{
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMISAC::InternalEncode(
- WebRtc_UWord8* /* bitstream */,
- WebRtc_Word16* /* bitStreamLenByte */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMISAC::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return 0;
-}
-
-
-WebRtc_Word16
-ACMISAC::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMISAC::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMISAC::InternalCreateDecoder()
-{
- return -1;
-}
-
-
-void
-ACMISAC::DestructDecoderSafe()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMISAC::InternalCreateEncoder()
-{
- return -1;
-}
-
-
-void
-ACMISAC::DestructEncoderSafe()
-{
- return;
-}
-
-
-WebRtc_Word32
-ACMISAC::CodecDef(
- WebRtcNetEQ_CodecDef& /* codecDef */,
- const CodecInst& /* codecInst */)
-{
- return -1;
-}
-
-
-void
-ACMISAC::InternalDestructEncoderInst(
- void* /* ptrInst */)
-{
- return;
-}
-
-WebRtc_Word16
-ACMISAC::DeliverCachedIsacData(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */,
- WebRtc_UWord32* /* timestamp */,
- WebRtcACMEncodingType* /* encodingType */,
- const WebRtc_UWord16 /* isacRate */,
- const WebRtc_UWord8 /* isacBWestimate */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMISAC::Transcode(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */,
- WebRtc_Word16 /* qBWE */,
- WebRtc_Word32 /* scale */,
- bool /* isRED */)
-{
- return -1;
-}
-
-WebRtc_Word16
-ACMISAC::SetBitRateSafe(
- WebRtc_Word32 /* bitRate */)
-{
- return -1;
-}
-
-WebRtc_Word32
-ACMISAC::GetEstimatedBandwidthSafe()
-{
- return -1;
-}
-
-WebRtc_Word32
-ACMISAC::SetEstimatedBandwidthSafe(
- WebRtc_Word32 /* estimatedBandwidth */)
-{
- return -1;
-}
-
-WebRtc_Word32
-ACMISAC::GetRedPayloadSafe(
- WebRtc_UWord8* /* redPayload */,
- WebRtc_Word16* /* payloadBytes */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMISAC::UnregisterFromNetEqSafe(
- ACMNetEQ* /* netEq */,
- WebRtc_Word16 /* payloadType */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMISAC::UpdateDecoderSampFreq(
- WebRtc_Word16 /* codecId */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMISAC::UpdateEncoderSampFreq(
- WebRtc_UWord16 /* encoderSampFreqHz */)
-{
- return -1;
-}
-
-WebRtc_Word16
-ACMISAC::EncoderSampFreq(
- WebRtc_UWord16& /* sampFreqHz */)
-{
- return -1;
-}
-
-WebRtc_Word32
-ACMISAC::ConfigISACBandwidthEstimator(
- const WebRtc_UWord8 /* initFrameSizeMsec */,
- const WebRtc_UWord16 /* initRateBitPerSec */,
- const bool /* enforceFrameSize */)
-{
- return -1;
-}
-
-WebRtc_Word32
-ACMISAC::SetISACMaxPayloadSize(
- const WebRtc_UWord16 /* maxPayloadLenBytes */)
-{
- return -1;
-}
-
-WebRtc_Word32
-ACMISAC::SetISACMaxRate(
- const WebRtc_UWord32 /* maxRateBitPerSec */)
-{
- return -1;
-}
-
-
-void
-ACMISAC::UpdateFrameLen()
-{
- return;
-}
-
-void
-ACMISAC::CurrentRate(
- WebRtc_Word32& /*rateBitPerSec */)
-{
- return;
-}
-
-bool
-ACMISAC::DecoderParamsSafe(
- WebRtcACMCodecParams* /* decParams */,
- const WebRtc_UWord8 /* payloadType */)
-{
- return false;
-}
-
-void
-ACMISAC::SaveDecoderParamSafe(
- const WebRtcACMCodecParams* /* codecParams */)
-{
- return;
-}
-
-WebRtc_Word16
-ACMISAC::REDPayloadISAC(
- const WebRtc_Word32 /* isacRate */,
- const WebRtc_Word16 /* isacBwEstimate */,
- WebRtc_UWord8* /* payload */,
- WebRtc_Word16* /* payloadLenBytes */)
-{
- return -1;
-}
-
-
-#else //===================== Actual Implementation =======================
-
-
-
-#ifdef WEBRTC_CODEC_ISACFX
-
-enum IsacSamplingRate
-{
- kIsacWideband = 16,
- kIsacSuperWideband = 32
-};
-
-static float
-ACMISACFixTranscodingScale(
- WebRtc_UWord16 rate)
-{
- // find the scale for transcoding, the scale is rounded
- // downward
- float scale = -1;
- for(WebRtc_Word16 n=0; n < ISAC_NUM_SUPPORTED_RATES; n++)
- {
- if(rate >= isacSuportedRates[n])
- {
- scale = isacScale[n];
- break;
- }
- }
- return scale;
-}
-
-static void
-ACMISACFixGetSendBitrate(
- ACM_ISAC_STRUCT* inst,
- WebRtc_Word32* bottleNeck)
-{
- *bottleNeck = WebRtcIsacfix_GetUplinkBw(inst);
-}
-
-static WebRtc_Word16
-ACMISACFixGetNewBitstream(
- ACM_ISAC_STRUCT* inst,
- WebRtc_Word16 BWEIndex,
- WebRtc_Word16 /* jitterIndex */,
- WebRtc_Word32 rate,
- WebRtc_Word16* bitStream,
- bool isRED)
-{
- if (isRED)
- {
- // RED not supported with iSACFIX
- return -1;
- }
- float scale = ACMISACFixTranscodingScale((WebRtc_UWord16)rate);
- return WebRtcIsacfix_GetNewBitStream(inst, BWEIndex, scale, bitStream);
-}
-
-
-static WebRtc_Word16
-ACMISACFixGetSendBWE(
- ACM_ISAC_STRUCT* inst,
- WebRtc_Word16* rateIndex,
- WebRtc_Word16* /* dummy */)
-{
- WebRtc_Word16 localRateIndex;
- WebRtc_Word16 status = WebRtcIsacfix_GetDownLinkBwIndex(inst, &localRateIndex);
- if(status < 0)
- {
- return -1;
- }
- else
- {
- *rateIndex = localRateIndex;
- return 0;
- }
-}
-
-static WebRtc_Word16
-ACMISACFixControlBWE(
- ACM_ISAC_STRUCT* inst,
- WebRtc_Word32 rateBPS,
- WebRtc_Word16 frameSizeMs,
- WebRtc_Word16 enforceFrameSize)
-{
- return WebRtcIsacfix_ControlBwe(inst, (WebRtc_Word16)rateBPS,
- frameSizeMs, enforceFrameSize);
-}
-
-static WebRtc_Word16
-ACMISACFixControl(
- ACM_ISAC_STRUCT* inst,
- WebRtc_Word32 rateBPS,
- WebRtc_Word16 frameSizeMs)
-{
- return WebRtcIsacfix_Control(inst, (WebRtc_Word16)rateBPS,
- frameSizeMs);
-}
-
-static IsacSamplingRate
-ACMISACFixGetEncSampRate(
- ACM_ISAC_STRUCT* /* inst */)
-{
- return kIsacWideband;
-}
-
-
-static IsacSamplingRate
-ACMISACFixGetDecSampRate(
- ACM_ISAC_STRUCT* /* inst */)
-{
- return kIsacWideband;
-}
-
-#endif
-
-
-
-
-
-
-ACMISAC::ACMISAC(WebRtc_Word16 codecID)
- : _isEncInitialized(false),
- _isacCodingMode(CHANNEL_INDEPENDENT),
- _enforceFrameSize(false),
- _isacCurrentBN(32000),
- _samplesIn10MsAudio(160) { // Initiates to 16 kHz mode.
- _codecID = codecID;
-
- // Create codec instance.
- _codecInstPtr = new ACMISACInst;
- if (_codecInstPtr == NULL) {
- return;
- }
- _codecInstPtr->inst = NULL;
-
- // Initiate decoder parameters for the 32 kHz mode.
- memset(&_decoderParams32kHz, 0, sizeof(WebRtcACMCodecParams));
- _decoderParams32kHz.codecInstant.pltype = -1;
-
- // TODO(tlegrand): Check if the following is really needed, now that
- // ACMGenericCodec has been updated to initialize this value.
- // Initialize values that can be used uninitialized otherwise
- _decoderParams.codecInstant.pltype = -1;
-}
-
-
-ACMISAC::~ACMISAC()
-{
- if (_codecInstPtr != NULL)
- {
- if(_codecInstPtr->inst != NULL)
- {
- ACM_ISAC_FREE(_codecInstPtr->inst);
- _codecInstPtr->inst = NULL;
- }
- delete _codecInstPtr;
- _codecInstPtr = NULL;
- }
- return;
-}
-
-
-ACMGenericCodec*
-ACMISAC::CreateInstance(void)
-{
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMISAC::InternalEncode(
- WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte)
-{
- // ISAC takes 10ms audio everytime we call encoder, therefor,
- // it should be treated like codecs with 'basic coding block'
- // non-zero, and the following 'while-loop' should not be necessary.
- // However, due to a mistake in the codec the frame-size might change
- // at the first 10ms pushed in to iSAC if the bit-rate is low, this is
- // sort of a bug in iSAC. to address this we treat iSAC as the
- // following.
-
- if (_codecInstPtr == NULL)
- {
- return -1;
- }
- *bitStreamLenByte = 0;
- while((*bitStreamLenByte == 0) && (_inAudioIxRead < _frameLenSmpl))
- {
- if(_inAudioIxRead > _inAudioIxWrite)
- {
- // something is wrong.
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "The actual fram-size of iSAC appears to be larger that expected. All audio \
-pushed in but no bit-stream is generated.");
- return -1;
- }
- *bitStreamLenByte = ACM_ISAC_ENCODE(_codecInstPtr->inst,
- &_inAudio[_inAudioIxRead], (WebRtc_Word16*)bitstream);
- // increment the read index this tell the caller that how far
- // we have gone forward in reading the audio buffer
- _inAudioIxRead += _samplesIn10MsAudio;
- }
- if(*bitStreamLenByte == 0)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, _uniqueID,
- "ISAC Has encoded the whole frame but no bit-stream is generated.");
- }
-
- // a packet is generated iSAC, is set in adaptive mode may change
- // the frame length and we like to update the bottleneck value as
- // well, although updating bottleneck is not crucial
- if((*bitStreamLenByte > 0) && (_isacCodingMode == ADAPTIVE))
- {
- //_frameLenSmpl = ACM_ISAC_GETNEWFRAMELEN(_codecInstPtr->inst);
- ACM_ISAC_GETSENDBITRATE(_codecInstPtr->inst, &_isacCurrentBN);
- }
- UpdateFrameLen();
- return *bitStreamLenByte;
-}
-
-
-WebRtc_Word16
-ACMISAC::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return 0;
-}
-
-
-WebRtc_Word16
-ACMISAC::InternalInitEncoder(
- WebRtcACMCodecParams* codecParams)
-{
- // if rate is set to -1 then iSAC has to be in adaptive mode
- if(codecParams->codecInstant.rate == -1)
- {
- _isacCodingMode = ADAPTIVE;
- }
-
- // sanity check that rate is in acceptable range
- else if((codecParams->codecInstant.rate >= ISAC_MIN_RATE) &&
- (codecParams->codecInstant.rate <= ISAC_MAX_RATE))
- {
- _isacCodingMode = CHANNEL_INDEPENDENT;
- _isacCurrentBN = codecParams->codecInstant.rate;
- }
- else
- {
- return -1;
- }
-
- // we need to set the encoder sampling frequency.
- if(UpdateEncoderSampFreq((WebRtc_UWord16)codecParams->codecInstant.plfreq) < 0)
- {
- return -1;
- }
- if(ACM_ISAC_ENCODERINIT(_codecInstPtr->inst, _isacCodingMode) < 0)
- {
- return -1;
- }
-
- // apply the frame-size and rate if operating in
- // channel-independent mode
- if(_isacCodingMode == CHANNEL_INDEPENDENT)
- {
- if(ACM_ISAC_CONTROL(_codecInstPtr->inst,
- codecParams->codecInstant.rate,
- codecParams->codecInstant.pacsize /
- (codecParams->codecInstant.plfreq / 1000)) < 0)
- {
- return -1;
- }
- }
- else
- {
- // We need this for adaptive case and has to be called
- // after initialization
- ACM_ISAC_GETSENDBITRATE(
- _codecInstPtr->inst, &_isacCurrentBN);
- }
- _frameLenSmpl = ACM_ISAC_GETNEWFRAMELEN(_codecInstPtr->inst);
- return 0;
-}
-
-WebRtc_Word16
-ACMISAC::InternalInitDecoder(
- WebRtcACMCodecParams* codecParams)
-{
- if (_codecInstPtr == NULL)
- {
- return -1;
- }
-
- // set decoder sampling frequency.
- if(codecParams->codecInstant.plfreq == 32000)
- {
- UpdateDecoderSampFreq(ACMCodecDB::kISACSWB);
- }
- else
- {
- UpdateDecoderSampFreq(ACMCodecDB::kISAC);
- }
-
- // in a one-way communication we may never register send-codec.
- // However we like that the BWE to work properly so it has to
- // be initialized. The BWE is initialized when iSAC encoder is initialized.
- // Therefore, we need this.
- if(!_encoderInitialized)
- {
- // Since we don't require a valid rate or a valid packet size when initializing
- // the decoder, we set valid values before initializing encoder
- codecParams->codecInstant.rate = kIsacWbDefaultRate;
- codecParams->codecInstant.pacsize = kIsacPacSize960;
- if(InternalInitEncoder(codecParams) < 0)
- {
- return -1;
- }
- _encoderInitialized = true;
- }
-
- return ACM_ISAC_DECODERINIT(_codecInstPtr->inst);
-}
-
-
-WebRtc_Word16
-ACMISAC::InternalCreateDecoder()
-{
- if (_codecInstPtr == NULL)
- {
- return -1;
- }
- WebRtc_Word16 status = ACM_ISAC_CREATE (&(_codecInstPtr->inst));
-
- // specific to codecs with one instance for encoding and decoding
- _encoderInitialized = false;
- if(status < 0)
- {
- _encoderExist = false;
- }
- else
- {
- _encoderExist = true;
- }
- return status;
-}
-
-
-void
-ACMISAC::DestructDecoderSafe()
-{
- // codec with shared instance cannot delete.
- _decoderInitialized = false;
- return;
-}
-
-
-WebRtc_Word16
-ACMISAC::InternalCreateEncoder()
-{
- if (_codecInstPtr == NULL)
- {
- return -1;
- }
- WebRtc_Word16 status = ACM_ISAC_CREATE(&(_codecInstPtr->inst));
-
- // specific to codecs with one instance for encoding and decoding
- _decoderInitialized = false;
- if(status < 0)
- {
- _decoderExist = false;
- }
- else
- {
- _decoderExist = true;
- }
- return status;
-}
-
-
-void
-ACMISAC::DestructEncoderSafe()
-{
- // codec with shared instance cannot delete.
- _encoderInitialized = false;
- return;
-}
-
-
-WebRtc_Word32
-ACMISAC::CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst)
-{
- // Sanity checks
- if (_codecInstPtr == NULL)
- {
- return -1;
- }
- if (!_decoderInitialized || !_decoderExist)
- {
- // Todo:
- // log error
- return -1;
- }
- // Fill up the structure by calling
- // "SET_CODEC_PAR" & "SET_ISAC_FUNCTION."
- // Then call NetEQ to add the codec to it's
- // database.
- if(codecInst.plfreq == 16000)
- {
- SET_CODEC_PAR((codecDef), kDecoderISAC, codecInst.pltype,
- _codecInstPtr->inst, 16000);
-#ifdef WEBRTC_CODEC_ISAC
- SET_ISAC_FUNCTIONS((codecDef));
-#else
- SET_ISACfix_FUNCTIONS((codecDef));
-#endif
- }
- else
- {
-#ifdef WEBRTC_CODEC_ISAC
- SET_CODEC_PAR((codecDef), kDecoderISACswb, codecInst.pltype,
- _codecInstPtr->inst, 32000);
- SET_ISACSWB_FUNCTIONS((codecDef));
-#else
- return -1;
-#endif
- }
-
- return 0;
-}
-
-
-void
-ACMISAC::InternalDestructEncoderInst(
- void* ptrInst)
-{
- if(ptrInst != NULL)
- {
- ACM_ISAC_FREE((ACM_ISAC_STRUCT *)ptrInst);
- }
- return;
-}
-
-WebRtc_Word16
-ACMISAC::Transcode(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte,
- WebRtc_Word16 qBWE,
- WebRtc_Word32 rate,
- bool isRED)
-{
- WebRtc_Word16 jitterInfo = 0;
- // transcode from a higher rate to lower rate
- // sanity check
- if (_codecInstPtr == NULL)
- {
- return -1;
- }
-
- *bitStreamLenByte = ACM_ISAC_GETNEWBITSTREAM(_codecInstPtr->inst,
- qBWE, jitterInfo, rate, (WebRtc_Word16*)bitStream, (isRED)? 1:0);
-
- if(*bitStreamLenByte < 0)
- {
- // error happened
- *bitStreamLenByte = 0;
- return -1;
- }
- else
- {
- return *bitStreamLenByte;
- }
-}
-
-WebRtc_Word16
-ACMISAC::SetBitRateSafe(
- WebRtc_Word32 bitRate)
-{
- if (_codecInstPtr == NULL)
- {
- return -1;
- }
- WebRtc_UWord16 encoderSampFreq;
- EncoderSampFreq(encoderSampFreq);
- bool reinit = false;
- // change the BN of iSAC
- if(bitRate == -1)
- {
- // ADAPTIVE MODE
- // Check if it was already in adaptive mode
- if(_isacCodingMode != ADAPTIVE)
- {
- // was not in adaptive, then set the mode to adaptive
- // and flag for re-initialization
- _isacCodingMode = ADAPTIVE;
- reinit = true;
- }
- }
- // Sanity check if the rate valid
- else if((bitRate >= ISAC_MIN_RATE) &&
- (bitRate <= ISAC_MAX_RATE))
- {
- //check if it was in channel-independent mode before
- if(_isacCodingMode != CHANNEL_INDEPENDENT)
- {
- // was not in channel independent, set the mode to
- // channel-independent and flag for re-initialization
- _isacCodingMode = CHANNEL_INDEPENDENT;
- reinit = true;
- }
- // store the bottleneck
- _isacCurrentBN = (WebRtc_UWord16)bitRate;
- }
- else
- {
- // invlaid rate
- return -1;
- }
-
- WebRtc_Word16 status = 0;
- if(reinit)
- {
- // initialize and check if it is successful
- if(ACM_ISAC_ENCODERINIT(_codecInstPtr->inst, _isacCodingMode) < 0)
- {
- // failed initialization
- return -1;
- }
- }
- if(_isacCodingMode == CHANNEL_INDEPENDENT)
- {
-
- status = ACM_ISAC_CONTROL(_codecInstPtr->inst, _isacCurrentBN,
- (encoderSampFreq == 32000)? 30:(_frameLenSmpl / 16));
- if(status < 0)
- {
- status = -1;
- }
- }
-
- // Update encoder parameters
- _encoderParams.codecInstant.rate = bitRate;
-
- UpdateFrameLen();
- return status;
-}
-
-
-WebRtc_Word32
-ACMISAC::GetEstimatedBandwidthSafe()
-{
- WebRtc_Word16 bandwidthIndex;
- WebRtc_Word16 delayIndex;
- IsacSamplingRate sampRate;
-
- // Get bandwidth information
- ACM_ISAC_GETSENDBWE(_codecInstPtr->inst, &bandwidthIndex, &delayIndex);
-
- // Validy check of index
- if ((bandwidthIndex < 0) || (bandwidthIndex >= NR_ISAC_BANDWIDTHS))
- {
- return -1;
- }
-
- // Check sample frequency
- sampRate = ACM_ISAC_GETDECSAMPRATE(_codecInstPtr->inst);
- if(sampRate == kIsacWideband)
- {
- return isacRatesWB[bandwidthIndex];
- }
- else
- {
- return isacRatesSWB[bandwidthIndex];
- }
-}
-
-WebRtc_Word32
-ACMISAC::SetEstimatedBandwidthSafe(
- WebRtc_Word32 estimatedBandwidth)
-{
- IsacSamplingRate sampRate;
- WebRtc_Word16 bandwidthIndex;
-
- // Check sample frequency and choose appropriate table
- sampRate = ACM_ISAC_GETENCSAMPRATE(_codecInstPtr->inst);
-
- if(sampRate == kIsacWideband)
- {
- // Search through the WB rate table to find the index
-
- bandwidthIndex = NR_ISAC_BANDWIDTHS/2 - 1;
- for (int i=0; i<(NR_ISAC_BANDWIDTHS/2); i++)
- {
- if (estimatedBandwidth == isacRatesWB[i])
- {
- bandwidthIndex = i;
- break;
- } else if (estimatedBandwidth == isacRatesWB[i+NR_ISAC_BANDWIDTHS/2])
- {
- bandwidthIndex = i + NR_ISAC_BANDWIDTHS/2;
- break;
- } else if (estimatedBandwidth < isacRatesWB[i])
- {
- bandwidthIndex = i;
- break;
- }
- }
- }
- else
- {
- // Search through the SWB rate table to find the index
- bandwidthIndex = NR_ISAC_BANDWIDTHS - 1;
- for (int i=0; i<NR_ISAC_BANDWIDTHS; i++)
- {
- if(estimatedBandwidth <= isacRatesSWB[i])
- {
- bandwidthIndex = i;
- break;
- }
- }
- }
-
- // Set iSAC Bandwidth Estimate
- ACM_ISAC_SETBWE(_codecInstPtr->inst, bandwidthIndex);
-
- return 0;
-}
-
-WebRtc_Word32
-ACMISAC::GetRedPayloadSafe(
-#if (!defined(WEBRTC_CODEC_ISAC))
- WebRtc_UWord8* /* redPayload */,
- WebRtc_Word16* /* payloadBytes */)
-{
- return -1;
-#else
- WebRtc_UWord8* redPayload,
- WebRtc_Word16* payloadBytes)
-{
-
- WebRtc_Word16 bytes = WebRtcIsac_GetRedPayload(_codecInstPtr->inst, (WebRtc_Word16*)redPayload);
- if (bytes < 0)
- {
- return -1;
- }
- *payloadBytes = bytes;
- return 0;
-#endif
-}
-
-
-WebRtc_Word16
-ACMISAC::UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType)
-{
- if(payloadType == _decoderParams.codecInstant.pltype)
- {
- return netEq->RemoveCodec(kDecoderISAC);
- }
- else if(payloadType == _decoderParams32kHz.codecInstant.pltype)
- {
- return netEq->RemoveCodec(kDecoderISACswb);
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot unregister codec %s given payload-type %d does not match \
-the stored payload type %d or %d",
- _decoderParams.codecInstant.plname,
- payloadType,
- _decoderParams.codecInstant.pltype,
- _decoderParams32kHz.codecInstant.pltype);
-
- return -1;
- }
-}
-
-
-WebRtc_Word16
-ACMISAC::UpdateDecoderSampFreq(
-#ifdef WEBRTC_CODEC_ISAC
- WebRtc_Word16 codecId)
-{
- if(ACMCodecDB::kISAC == codecId)
- {
- return WebRtcIsac_SetDecSampRate(_codecInstPtr->inst, kIsacWideband);
- }
- else if(ACMCodecDB::kISACSWB == codecId)
- {
- return WebRtcIsac_SetDecSampRate(_codecInstPtr->inst, kIsacSuperWideband);
- }
- else
- {
- return -1;
- }
-
-#else
- WebRtc_Word16 /* codecId */)
-{
- return 0;
-#endif
-}
-
-
-WebRtc_Word16
-ACMISAC::UpdateEncoderSampFreq(
-#ifdef WEBRTC_CODEC_ISAC
- WebRtc_UWord16 encoderSampFreqHz)
-{
- WebRtc_UWord16 currentSampRateHz;
- EncoderSampFreq(currentSampRateHz);
-
- if(currentSampRateHz != encoderSampFreqHz)
- {
- if((encoderSampFreqHz != 16000) && (encoderSampFreqHz != 32000))
- {
- return -1;
- }
- else
- {
- _inAudioIxRead = 0;
- _inAudioIxWrite = 0;
- _inTimestampIxWrite = 0;
- if(encoderSampFreqHz == 16000)
- {
- if(WebRtcIsac_SetEncSampRate(_codecInstPtr->inst, kIsacWideband) < 0)
- {
- return -1;
- }
- _samplesIn10MsAudio = 160;
- }
- else
- {
-
- if(WebRtcIsac_SetEncSampRate(_codecInstPtr->inst, kIsacSuperWideband) < 0)
- {
- return -1;
- }
- _samplesIn10MsAudio = 320;
- }
- _frameLenSmpl = ACM_ISAC_GETNEWFRAMELEN(_codecInstPtr->inst);
- _encoderParams.codecInstant.pacsize = _frameLenSmpl;
- _encoderParams.codecInstant.plfreq = encoderSampFreqHz;
- return 0;
- }
- }
-#else
- WebRtc_UWord16 /* codecId */)
-{
-#endif
- return 0;
-}
-
-WebRtc_Word16
-ACMISAC::EncoderSampFreq(
- WebRtc_UWord16& sampFreqHz)
-{
- IsacSamplingRate sampRate;
- sampRate = ACM_ISAC_GETENCSAMPRATE(_codecInstPtr->inst);
- if(sampRate == kIsacSuperWideband)
- {
- sampFreqHz = 32000;
- }
- else
- {
- sampFreqHz = 16000;
- }
- return 0;
-}
-
-WebRtc_Word32
-ACMISAC::ConfigISACBandwidthEstimator(
- const WebRtc_UWord8 initFrameSizeMsec,
- const WebRtc_UWord16 initRateBitPerSec,
- const bool enforceFrameSize)
-{
- WebRtc_Word16 status;
- {
- WebRtc_UWord16 sampFreqHz;
- EncoderSampFreq(sampFreqHz);
- // @TODO: at 32kHz we hardcode calling with 30ms and enforce
- // the frame-size otherwise we might get error. Revise if
- // control-bwe is changed.
- if(sampFreqHz == 32000)
- {
- status = ACM_ISAC_CONTROL_BWE(_codecInstPtr->inst,
- initRateBitPerSec, 30, 1);
- }
- else
- {
- status = ACM_ISAC_CONTROL_BWE(_codecInstPtr->inst,
- initRateBitPerSec, initFrameSizeMsec, enforceFrameSize? 1:0);
- }
- }
- if(status < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Coutn't config iSAC BWE.");
- return -1;
- }
- UpdateFrameLen();
- ACM_ISAC_GETSENDBITRATE(_codecInstPtr->inst, &_isacCurrentBN);
- return 0;
-}
-
-WebRtc_Word32
-ACMISAC::SetISACMaxPayloadSize(
- const WebRtc_UWord16 maxPayloadLenBytes)
-{
- return ACM_ISAC_SETMAXPAYLOADSIZE(_codecInstPtr->inst, maxPayloadLenBytes);
-}
-
-WebRtc_Word32
-ACMISAC::SetISACMaxRate(
- const WebRtc_UWord32 maxRateBitPerSec)
-{
- return ACM_ISAC_SETMAXRATE(_codecInstPtr->inst, maxRateBitPerSec);
-}
-
-
-void
-ACMISAC::UpdateFrameLen()
-{
- _frameLenSmpl = ACM_ISAC_GETNEWFRAMELEN(_codecInstPtr->inst);
- _encoderParams.codecInstant.pacsize = _frameLenSmpl;
-}
-
-void
-ACMISAC::CurrentRate(WebRtc_Word32& rateBitPerSec)
-{
- if(_isacCodingMode == ADAPTIVE)
- {
- ACM_ISAC_GETSENDBITRATE(_codecInstPtr->inst, &rateBitPerSec);
- }
-}
-
-
-bool
-ACMISAC::DecoderParamsSafe(
- WebRtcACMCodecParams* decParams,
- const WebRtc_UWord8 payloadType)
-{
- if(_decoderInitialized)
- {
- if(payloadType == _decoderParams.codecInstant.pltype)
- {
- memcpy(decParams, &_decoderParams, sizeof(WebRtcACMCodecParams));
- return true;
- }
- if(payloadType == _decoderParams32kHz.codecInstant.pltype)
- {
- memcpy(decParams, &_decoderParams32kHz,
- sizeof(WebRtcACMCodecParams));
- return true;
- }
- }
- return false;
-}
-
-void
-ACMISAC::SaveDecoderParamSafe(
- const WebRtcACMCodecParams* codecParams)
-{
- // set decoder sampling frequency.
- if(codecParams->codecInstant.plfreq == 32000)
- {
- memcpy(&_decoderParams32kHz, codecParams, sizeof(WebRtcACMCodecParams));
- }
- else
- {
- memcpy(&_decoderParams, codecParams, sizeof(WebRtcACMCodecParams));
- }
-}
-
-
-WebRtc_Word16
-ACMISAC::REDPayloadISAC(
- const WebRtc_Word32 isacRate,
- const WebRtc_Word16 isacBwEstimate,
- WebRtc_UWord8* payload,
- WebRtc_Word16* payloadLenBytes)
-{
- WebRtc_Word16 status;
- ReadLockScoped rl(_codecWrapperLock);
- status = Transcode(payload, payloadLenBytes, isacBwEstimate, isacRate, true);
- return status;
-}
-
-#endif
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_isac.h b/trunk/src/modules/audio_coding/main/source/acm_isac.h
deleted file mode 100644
index 6013c61..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_isac.h
+++ /dev/null
@@ -1,153 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_
-
-#include "acm_generic_codec.h"
-
-namespace webrtc
-{
-
-struct ACMISACInst;
-
-enum iSACCodingMode {ADAPTIVE, CHANNEL_INDEPENDENT};
-
-
-class ACMISAC : public ACMGenericCodec
-{
-public:
- ACMISAC(WebRtc_Word16 codecID);
- ~ACMISAC();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(
- WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(
- WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 InternalInitDecoder(
- WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 DeliverCachedIsacData(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte,
- WebRtc_UWord32* timestamp,
- WebRtcACMEncodingType* encodingType,
- const WebRtc_UWord16 isacRate,
- const WebRtc_UWord8 isacBWestimate);
-
- WebRtc_Word16 DeliverCachedData(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */,
- WebRtc_UWord32* /* timestamp */,
- WebRtcACMEncodingType* /* encodingType */)
- {
- return -1;
- }
-
- WebRtc_Word16 UpdateDecoderSampFreq(
- WebRtc_Word16 codecId);
-
- WebRtc_Word16 UpdateEncoderSampFreq(
- WebRtc_UWord16 sampFreqHz);
-
- WebRtc_Word16 EncoderSampFreq(
- WebRtc_UWord16& sampFreqHz);
-
- WebRtc_Word32 ConfigISACBandwidthEstimator(
- const WebRtc_UWord8 initFrameSizeMsec,
- const WebRtc_UWord16 initRateBitPerSec,
- const bool enforceFrameSize);
-
- WebRtc_Word32 SetISACMaxPayloadSize(
- const WebRtc_UWord16 maxPayloadLenBytes);
-
- WebRtc_Word32 SetISACMaxRate(
- const WebRtc_UWord32 maxRateBitPerSec);
-
- WebRtc_Word16 REDPayloadISAC(
- const WebRtc_Word32 isacRate,
- const WebRtc_Word16 isacBwEstimate,
- WebRtc_UWord8* payload,
- WebRtc_Word16* payloadLenBytes);
-
-protected:
- WebRtc_Word16 DecodeSafe(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio,
- WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 SetBitRateSafe(
- const WebRtc_Word32 bitRate);
-
- WebRtc_Word32 GetEstimatedBandwidthSafe();
-
- WebRtc_Word32 SetEstimatedBandwidthSafe(WebRtc_Word32 estimatedBandwidth);
-
- WebRtc_Word32 GetRedPayloadSafe(
- WebRtc_UWord8* redPayload,
- WebRtc_Word16* payloadBytes);
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- void InternalDestructEncoderInst(
- void* ptrInst);
-
- WebRtc_Word16 Transcode(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte,
- WebRtc_Word16 qBWE,
- WebRtc_Word32 rate,
- bool isRED);
-
- WebRtc_Word16 UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- void CurrentRate(WebRtc_Word32& rateBitPerSec);
-
- void UpdateFrameLen();
-
- bool DecoderParamsSafe(
- WebRtcACMCodecParams *decParams,
- const WebRtc_UWord8 payloadType);
-
- void SaveDecoderParamSafe(
- const WebRtcACMCodecParams* codecParams);
-
- ACMISACInst* _codecInstPtr;
-
- bool _isEncInitialized;
- iSACCodingMode _isacCodingMode;
- bool _enforceFrameSize;
- WebRtc_Word32 _isacCurrentBN;
- WebRtc_UWord16 _samplesIn10MsAudio;
- WebRtcACMCodecParams _decoderParams32kHz;
-};
-
-} //namespace
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_isac_macros.h b/trunk/src/modules/audio_coding/main/source/acm_isac_macros.h
deleted file mode 100644
index 9e5f55f..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_isac_macros.h
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_
-
-#include "engine_configurations.h"
-
-namespace webrtc
-{
-
-#ifdef WEBRTC_CODEC_ISAC
-# define ACM_ISAC_CREATE WebRtcIsac_Create
-# define ACM_ISAC_FREE WebRtcIsac_Free
-# define ACM_ISAC_ENCODERINIT WebRtcIsac_EncoderInit
-# define ACM_ISAC_ENCODE WebRtcIsac_Encode
-# define ACM_ISAC_DECODERINIT WebRtcIsac_DecoderInit
-# define ACM_ISAC_DECODE_BWE WebRtcIsac_UpdateBwEstimate
-# define ACM_ISAC_DECODE_B WebRtcIsac_Decode
-# define ACM_ISAC_DECODEPLC WebRtcIsac_DecodePlc
-# define ACM_ISAC_CONTROL WebRtcIsac_Control
-# define ACM_ISAC_CONTROL_BWE WebRtcIsac_ControlBwe
-# define ACM_ISAC_GETFRAMELEN WebRtcIsac_ReadFrameLen
-# define ACM_ISAC_VERSION WebRtcIsac_version
-# define ACM_ISAC_GETERRORCODE WebRtcIsac_GetErrorCode
-# define ACM_ISAC_GETSENDBITRATE WebRtcIsac_GetUplinkBw
-# define ACM_ISAC_SETMAXPAYLOADSIZE WebRtcIsac_SetMaxPayloadSize
-# define ACM_ISAC_SETMAXRATE WebRtcIsac_SetMaxRate
-# define ACM_ISAC_GETNEWBITSTREAM WebRtcIsac_GetNewBitStream
-# define ACM_ISAC_GETSENDBWE WebRtcIsac_GetDownLinkBwIndex
-# define ACM_ISAC_SETBWE WebRtcIsac_UpdateUplinkBw
-# define ACM_ISAC_GETBWE WebRtcIsac_ReadBwIndex
-# define ACM_ISAC_GETNEWFRAMELEN WebRtcIsac_GetNewFrameLen
-# define ACM_ISAC_STRUCT ISACStruct
-# define ACM_ISAC_GETENCSAMPRATE WebRtcIsac_EncSampRate
-# define ACM_ISAC_GETDECSAMPRATE WebRtcIsac_DecSampRate
-#endif
-
-#ifdef WEBRTC_CODEC_ISACFX
-# define ACM_ISAC_CREATE WebRtcIsacfix_Create
-# define ACM_ISAC_FREE WebRtcIsacfix_Free
-# define ACM_ISAC_ENCODERINIT WebRtcIsacfix_EncoderInit
-# define ACM_ISAC_ENCODE WebRtcIsacfix_Encode
-# define ACM_ISAC_DECODERINIT WebRtcIsacfix_DecoderInit
-# define ACM_ISAC_DECODE_BWE WebRtcIsacfix_UpdateBwEstimate
-# define ACM_ISAC_DECODE_B WebRtcIsacfix_Decode
-# define ACM_ISAC_DECODEPLC WebRtcIsacfix_DecodePlc
-# define ACM_ISAC_CONTROL ACMISACFixControl // local Impl
-# define ACM_ISAC_CONTROL_BWE ACMISACFixControlBWE // local Impl
-# define ACM_ISAC_GETFRAMELEN WebRtcIsacfix_ReadFrameLen
-# define ACM_ISAC_VERSION WebRtcIsacfix_version
-# define ACM_ISAC_GETERRORCODE WebRtcIsacfix_GetErrorCode
-# define ACM_ISAC_GETSENDBITRATE ACMISACFixGetSendBitrate // local Impl
-# define ACM_ISAC_SETMAXPAYLOADSIZE WebRtcIsacfix_SetMaxPayloadSize
-# define ACM_ISAC_SETMAXRATE WebRtcIsacfix_SetMaxRate
-# define ACM_ISAC_GETNEWBITSTREAM ACMISACFixGetNewBitstream // local Impl
-# define ACM_ISAC_GETSENDBWE ACMISACFixGetSendBWE // local Impl
-# define ACM_ISAC_SETBWE WebRtcIsacfix_UpdateUplinkBw
-# define ACM_ISAC_GETBWE WebRtcIsacfix_ReadBwIndex
-# define ACM_ISAC_GETNEWFRAMELEN WebRtcIsacfix_GetNewFrameLen
-# define ACM_ISAC_STRUCT ISACFIX_MainStruct
-# define ACM_ISAC_GETENCSAMPRATE ACMISACFixGetEncSampRate // local Impl
-# define ACM_ISAC_GETDECSAMPRATE ACMISACFixGetDecSampRate // local Impl
-#endif
-
-} //namespace
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_ISAC_MACROS_H_
-
diff --git a/trunk/src/modules/audio_coding/main/source/acm_neteq.cc b/trunk/src/modules/audio_coding/main/source/acm_neteq.cc
deleted file mode 100644
index be25918..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_neteq.cc
+++ /dev/null
@@ -1,1270 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-#include <algorithm> // sort
-#include <stdlib.h> // malloc
-#include <vector>
-
-#include "acm_neteq.h"
-#include "common_types.h"
-#include "critical_section_wrapper.h"
-#include "rw_lock_wrapper.h"
-#include "signal_processing_library.h"
-#include "tick_util.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_internal.h"
-
-namespace webrtc
-{
-
-#define RTP_HEADER_SIZE 12
-#define NETEQ_INIT_FREQ 8000
-#define NETEQ_INIT_FREQ_KHZ (NETEQ_INIT_FREQ/1000)
-#define NETEQ_ERR_MSG_LEN_BYTE (WEBRTC_NETEQ_MAX_ERROR_NAME + 1)
-
-
-ACMNetEQ::ACMNetEQ()
-:
-_id(0),
-_currentSampFreqKHz(NETEQ_INIT_FREQ_KHZ),
-_avtPlayout(false),
-_playoutMode(voice),
-_netEqCritSect(CriticalSectionWrapper::CreateCriticalSection()),
-_vadStatus(false),
-_vadMode(VADNormal),
-_decodeLock(RWLockWrapper::CreateRWLock()),
-_numSlaves(0),
-_receivedStereo(false),
-_masterSlaveInfo(NULL),
-_previousAudioActivity(AudioFrame::kVadUnknown),
-_extraDelay(0),
-_callbackCritSect(CriticalSectionWrapper::CreateCriticalSection())
-{
- for(int n = 0; n < MAX_NUM_SLAVE_NETEQ + 1; n++)
- {
- _isInitialized[n] = false;
- _ptrVADInst[n] = NULL;
- _inst[n] = NULL;
- _instMem[n] = NULL;
- _netEqPacketBuffer[n] = NULL;
- }
-}
-
-ACMNetEQ::~ACMNetEQ()
-{
- {
- CriticalSectionScoped lock(*_netEqCritSect);
- for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
- {
- if (_instMem[idx] != NULL)
- {
- free(_instMem[idx]);
- _instMem[idx] = NULL;
- }
- if (_netEqPacketBuffer[idx] != NULL)
- {
- free(_netEqPacketBuffer[idx]);
- _netEqPacketBuffer[idx] = NULL;
- }
- if(_ptrVADInst[idx] != NULL)
- {
- WebRtcVad_Free(_ptrVADInst[idx]);
- _ptrVADInst[idx] = NULL;
- }
- }
- if(_masterSlaveInfo != NULL)
- {
- free(_masterSlaveInfo);
- _masterSlaveInfo = NULL;
- }
- }
- if(_netEqCritSect != NULL)
- {
- delete _netEqCritSect;
- }
-
- if(_decodeLock != NULL)
- {
- delete _decodeLock;
- }
-
- if(_callbackCritSect != NULL)
- {
- delete _callbackCritSect;
- }
-}
-
-WebRtc_Word32
-ACMNetEQ::Init()
-{
- CriticalSectionScoped lock(*_netEqCritSect);
-
- for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
- {
- if(InitByIdxSafe(idx) < 0)
- {
- return -1;
- }
- // delete VAD instance and start fresh if required.
- if(_ptrVADInst[idx] != NULL)
- {
- WebRtcVad_Free(_ptrVADInst[idx]);
- _ptrVADInst[idx] = NULL;
- }
- if(_vadStatus)
- {
- // Has to enable VAD
- if(EnableVADByIdxSafe(idx) < 0)
- {
- // Failed to enable VAD.
- // Delete VAD instance, if it is created
- if(_ptrVADInst[idx] != NULL)
- {
- WebRtcVad_Free(_ptrVADInst[idx]);
- _ptrVADInst[idx] = NULL;
- }
- // We are at initialization of NetEq, if failed to
- // enable VAD, we delete the NetEq instance.
- if (_instMem[idx] != NULL) {
- free(_instMem[idx]);
- _instMem[idx] = NULL;
- _inst[idx] = NULL;
- }
- _isInitialized[idx] = false;
- return -1;
- }
- }
- _isInitialized[idx] = true;
- }
- if (EnableVAD() == -1)
- {
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word16
-ACMNetEQ::InitByIdxSafe(
- const WebRtc_Word16 idx)
-{
- int memorySizeBytes;
- if (WebRtcNetEQ_AssignSize(&memorySizeBytes) != 0)
- {
- LogError("AssignSize", idx);
- return -1;
- }
-
- if(_instMem[idx] != NULL)
- {
- free(_instMem[idx]);
- _instMem[idx] = NULL;
- }
- _instMem[idx] = malloc(memorySizeBytes);
- if (_instMem[idx] == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "InitByIdxSafe: NetEq Initialization error: could not allocate memory for NetEq");
- _isInitialized[idx] = false;
- return -1;
- }
- if (WebRtcNetEQ_Assign(&_inst[idx], _instMem[idx]) != 0)
- {
- if (_instMem[idx] != NULL) {
- free(_instMem[idx]);
- _instMem[idx] = NULL;
- }
- LogError("Assign", idx);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "InitByIdxSafe: NetEq Initialization error: could not Assign");
- _isInitialized[idx] = false;
- return -1;
- }
- if (WebRtcNetEQ_Init(_inst[idx], NETEQ_INIT_FREQ) != 0)
- {
- if (_instMem[idx] != NULL) {
- free(_instMem[idx]);
- _instMem[idx] = NULL;
- }
- LogError("Init", idx);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "InitByIdxSafe: NetEq Initialization error: could not initialize NetEq");
- _isInitialized[idx] = false;
- return -1;
- }
- _isInitialized[idx] = true;
- return 0;
-}
-
-WebRtc_Word16
-ACMNetEQ::EnableVADByIdxSafe(
- const WebRtc_Word16 idx)
-{
- if(_ptrVADInst[idx] == NULL)
- {
- if(WebRtcVad_Create(&_ptrVADInst[idx]) < 0)
- {
- _ptrVADInst[idx] = NULL;
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "EnableVADByIdxSafe: NetEq Initialization error: could not create VAD");
- return -1;
- }
- }
-
- if(WebRtcNetEQ_SetVADInstance(_inst[idx], _ptrVADInst[idx],
- (WebRtcNetEQ_VADInitFunction) WebRtcVad_Init,
- (WebRtcNetEQ_VADSetmodeFunction) WebRtcVad_set_mode,
- (WebRtcNetEQ_VADFunction) WebRtcVad_Process) < 0)
- {
- LogError("setVADinstance", idx);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "EnableVADByIdxSafe: NetEq Initialization error: could not set VAD instance");
- return -1;
- }
-
- if(WebRtcNetEQ_SetVADMode(_inst[idx], _vadMode) < 0)
- {
- LogError("setVADmode", idx);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "EnableVADByIdxSafe: NetEq Initialization error: could not set VAD mode");
- return -1;
- }
- return 0;
-}
-
-
-
-
-WebRtc_Word32
-ACMNetEQ::AllocatePacketBuffer(
- const WebRtcNetEQDecoder* usedCodecs,
- WebRtc_Word16 noOfCodecs)
-{
- // Due to WebRtcNetEQ_GetRecommendedBufferSize
- // the following has to be int otherwise we will have compiler error
- // if not casted
-
- CriticalSectionScoped lock(*_netEqCritSect);
- for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
- {
- if(AllocatePacketBufferByIdxSafe(usedCodecs, noOfCodecs, idx) < 0)
- {
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_Word16
-ACMNetEQ::AllocatePacketBufferByIdxSafe(
- const WebRtcNetEQDecoder* usedCodecs,
- WebRtc_Word16 noOfCodecs,
- const WebRtc_Word16 idx)
-{
- int maxNoPackets;
- int bufferSizeInBytes;
-
- if(!_isInitialized[idx])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "AllocatePacketBufferByIdxSafe: NetEq is not initialized.");
- return -1;
- }
- if (WebRtcNetEQ_GetRecommendedBufferSize(_inst[idx], usedCodecs, noOfCodecs,
- kTCPLargeJitter , &maxNoPackets, &bufferSizeInBytes)
- != 0)
- {
- LogError("GetRecommendedBufferSize", idx);
- return -1;
- }
- if(_netEqPacketBuffer[idx] != NULL)
- {
- free(_netEqPacketBuffer[idx]);
- _netEqPacketBuffer[idx] = NULL;
- }
-
- _netEqPacketBuffer[idx] = (WebRtc_Word16 *)malloc(bufferSizeInBytes);
- if (_netEqPacketBuffer[idx] == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "AllocatePacketBufferByIdxSafe: NetEq Initialization error: could not allocate "
- "memory for NetEq Packet Buffer");
- return -1;
-
- }
- if (WebRtcNetEQ_AssignBuffer(_inst[idx], maxNoPackets, _netEqPacketBuffer[idx],
- bufferSizeInBytes) != 0)
- {
- if (_netEqPacketBuffer[idx] != NULL) {
- free(_netEqPacketBuffer[idx]);
- _netEqPacketBuffer[idx] = NULL;
- }
- LogError("AssignBuffer", idx);
- return -1;
- }
- return 0;
-}
-
-
-
-
-WebRtc_Word32
-ACMNetEQ::SetExtraDelay(
- const WebRtc_Word32 delayInMS)
-{
- CriticalSectionScoped lock(*_netEqCritSect);
-
- for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
- {
- if(!_isInitialized[idx])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "SetExtraDelay: NetEq is not initialized.");
- return -1;
- }
- if(WebRtcNetEQ_SetExtraDelay(_inst[idx], delayInMS) < 0)
- {
- LogError("SetExtraDelay", idx);
- return -1;
- }
- }
- _extraDelay = delayInMS;
- return 0;
-}
-
-
-WebRtc_Word32
-ACMNetEQ::SetAVTPlayout(
- const bool enable)
-{
- CriticalSectionScoped lock(*_netEqCritSect);
- if (_avtPlayout != enable)
- {
- for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
- {
- if(!_isInitialized[idx])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "SetAVTPlayout: NetEq is not initialized.");
- return -1;
- }
- if(WebRtcNetEQ_SetAVTPlayout(_inst[idx], (enable) ? 1 : 0) < 0)
- {
- LogError("SetAVTPlayout", idx);
- return -1;
- }
- }
- }
- _avtPlayout = enable;
- return 0;
-}
-
-
-bool
-ACMNetEQ::AVTPlayout() const
-{
- CriticalSectionScoped lock(*_netEqCritSect);
- return _avtPlayout;
-}
-
-WebRtc_Word32
-ACMNetEQ::CurrentSampFreqHz() const
-{
- CriticalSectionScoped lock(*_netEqCritSect);
- if(!_isInitialized[0])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "CurrentSampFreqHz: NetEq is not initialized.");
- return -1;
- }
- return (WebRtc_Word32)(1000*_currentSampFreqKHz);
-}
-
-
-WebRtc_Word32
-ACMNetEQ::SetPlayoutMode(
- const AudioPlayoutMode mode)
-{
- CriticalSectionScoped lock(*_netEqCritSect);
- if(_playoutMode != mode)
- {
- for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
- {
- if(!_isInitialized[idx])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "SetPlayoutMode: NetEq is not initialized.");
- return -1;
- }
-
- enum WebRtcNetEQPlayoutMode playoutMode = kPlayoutOff;
- switch(mode)
- {
- case voice:
- playoutMode = kPlayoutOn;
- break;
- case fax:
- playoutMode = kPlayoutFax;
- break;
- case streaming:
- playoutMode = kPlayoutStreaming;
- break;
- }
- if(WebRtcNetEQ_SetPlayoutMode(_inst[idx], playoutMode) < 0)
- {
- LogError("SetPlayoutMode", idx);
- return -1;
- }
- }
- _playoutMode = mode;
- }
-
- return 0;
-}
-
-AudioPlayoutMode
-ACMNetEQ::PlayoutMode() const
-{
- CriticalSectionScoped lock(*_netEqCritSect);
- return _playoutMode;
-}
-
-
-WebRtc_Word32
-ACMNetEQ::NetworkStatistics(
- ACMNetworkStatistics* statistics) const
-{
- WebRtcNetEQ_NetworkStatistics stats;
- CriticalSectionScoped lock(*_netEqCritSect);
- if(!_isInitialized[0])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "NetworkStatistics: NetEq is not initialized.");
- return -1;
- }
- if(WebRtcNetEQ_GetNetworkStatistics(_inst[0], &stats) == 0)
- {
- statistics->currentAccelerateRate = stats.currentAccelerateRate;
- statistics->currentBufferSize = stats.currentBufferSize;
- statistics->jitterPeaksFound = (stats.jitterPeaksFound > 0);
- statistics->currentDiscardRate = stats.currentDiscardRate;
- statistics->currentExpandRate = stats.currentExpandRate;
- statistics->currentPacketLossRate = stats.currentPacketLossRate;
- statistics->currentPreemptiveRate = stats.currentPreemptiveRate;
- statistics->preferredBufferSize = stats.preferredBufferSize;
- statistics->clockDriftPPM = stats.clockDriftPPM;
- }
- else
- {
- LogError("getNetworkStatistics", 0);
- return -1;
- }
- const int kArrayLen = 100;
- int waiting_times[kArrayLen];
- int waiting_times_len = WebRtcNetEQ_GetRawFrameWaitingTimes(
- _inst[0], kArrayLen, waiting_times);
- if (waiting_times_len > 0)
- {
- std::vector<int> waiting_times_vec(waiting_times,
- waiting_times + waiting_times_len);
- std::sort(waiting_times_vec.begin(), waiting_times_vec.end());
- size_t size = waiting_times_vec.size();
- assert(size == static_cast<size_t>(waiting_times_len));
- if (size % 2 == 0)
- {
- statistics->medianWaitingTimeMs =
- (waiting_times_vec[size / 2 - 1] +
- waiting_times_vec[size / 2]) / 2;
- }
- else
- {
- statistics->medianWaitingTimeMs = waiting_times_vec[size / 2];
- }
- statistics->minWaitingTimeMs = waiting_times_vec.front();
- statistics->maxWaitingTimeMs = waiting_times_vec.back();
- double sum = 0;
- for (size_t i = 0; i < size; ++i) {
- sum += waiting_times_vec[i];
- }
- statistics->meanWaitingTimeMs = static_cast<int>(sum / size);
- }
- else if (waiting_times_len == 0)
- {
- statistics->meanWaitingTimeMs = -1;
- statistics->medianWaitingTimeMs = -1;
- statistics->minWaitingTimeMs = -1;
- statistics->maxWaitingTimeMs = -1;
- }
- else
- {
- LogError("getRawFrameWaitingTimes", 0);
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32
-ACMNetEQ::RecIn(
- const WebRtc_Word8* incomingPayload,
- const WebRtc_Word32 payloadLength,
- const WebRtcRTPHeader& rtpInfo)
-{
- // translate to NetEq struct
- WebRtcNetEQ_RTPInfo netEqRTPInfo;
- netEqRTPInfo.payloadType = rtpInfo.header.payloadType;
- netEqRTPInfo.sequenceNumber = rtpInfo.header.sequenceNumber;
- netEqRTPInfo.timeStamp = rtpInfo.header.timestamp;
- netEqRTPInfo.SSRC = rtpInfo.header.ssrc;
- netEqRTPInfo.markerBit = rtpInfo.header.markerBit;
-
- CriticalSectionScoped lock(*_netEqCritSect);
- // Down-cast the time to (32-6)-bit since we only care about
- // the least significant bits. (32-6) bits cover 2^(32-6) = 67108864 ms.
- // we masked 6 most significant bits of 32-bit so we don't loose resolution
- // when do the following multiplication.
- const WebRtc_UWord32 nowInMs = static_cast<WebRtc_UWord32>(
- TickTime::MillisecondTimestamp() & 0x03ffffff);
- WebRtc_UWord32 recvTimestamp = static_cast<WebRtc_UWord32>
- (_currentSampFreqKHz * nowInMs);
-
- int status;
-
- if(rtpInfo.type.Audio.channel == 1)
- {
- if(!_isInitialized[0])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "RecIn: NetEq is not initialized.");
- return -1;
- }
- // PUSH into Master
- status = WebRtcNetEQ_RecInRTPStruct(_inst[0], &netEqRTPInfo,
- (WebRtc_UWord8 *)incomingPayload, (WebRtc_Word16)payloadLength,
- recvTimestamp);
- if(status < 0)
- {
- LogError("RecInRTPStruct", 0);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "RecIn: NetEq, error in pushing in Master");
- return -1;
- }
- }
- else if(rtpInfo.type.Audio.channel == 2)
- {
- if(!_isInitialized[1])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "RecIn: NetEq is not initialized.");
- return -1;
- }
- // PUSH into Slave
- status = WebRtcNetEQ_RecInRTPStruct(_inst[1], &netEqRTPInfo,
- (WebRtc_UWord8 *)incomingPayload, (WebRtc_Word16)payloadLength,
- recvTimestamp);
- if(status < 0)
- {
- LogError("RecInRTPStruct", 1);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "RecIn: NetEq, error in pushing in Slave");
- return -1;
- }
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "RecIn: NetEq, error invalid numbe of channels %d \
-(1, for Master stream, and 2, for slave stream, are valid values)",
- rtpInfo.type.Audio.channel);
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32
-ACMNetEQ::RecOut(
- AudioFrame& audioFrame)
-{
- enum WebRtcNetEQOutputType type;
- WebRtc_Word16 payloadLenSample;
- enum WebRtcNetEQOutputType typeMaster;
- enum WebRtcNetEQOutputType typeSlave;
-
- WebRtc_Word16 payloadLenSampleSlave;
-
- CriticalSectionScoped lockNetEq(*_netEqCritSect);
-
- if(!_receivedStereo)
- {
- if(!_isInitialized[0])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "RecOut: NetEq is not initialized.");
- return -1;
- }
- {
- WriteLockScoped lockCodec(*_decodeLock);
- if(WebRtcNetEQ_RecOut(_inst[0], &(audioFrame._payloadData[0]),
- &payloadLenSample) != 0)
- {
- LogError("RecOut", 0);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "RecOut: NetEq, error in pulling out for mono case");
-
- // Check for errors that can be recovered from:
- // RECOUT_ERROR_SAMPLEUNDERRUN = 2003
- int errorCode = WebRtcNetEQ_GetErrorCode(_inst[0]);
- if(errorCode != 2003)
- {
- // Cannot recover; return an error
- return -1;
- }
- }
- }
- WebRtcNetEQ_GetSpeechOutputType(_inst[0], &type);
- audioFrame._audioChannel = 1;
- }
- else
- {
- if(!_isInitialized[0] || !_isInitialized[1])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "RecOut: NetEq is not initialized.");
- return -1;
- }
- WebRtc_Word16 payloadMaster[480];
- WebRtc_Word16 payloadSlave[480];
- {
- WriteLockScoped lockCodec(*_decodeLock);
- if(WebRtcNetEQ_RecOutMasterSlave(_inst[0], payloadMaster,
- &payloadLenSample, _masterSlaveInfo, 1) != 0)
- {
- LogError("RecOutMasterSlave", 0);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "RecOut: NetEq, error in pulling out for master");
-
- // Check for errors that can be recovered from:
- // RECOUT_ERROR_SAMPLEUNDERRUN = 2003
- int errorCode = WebRtcNetEQ_GetErrorCode(_inst[0]);
- if(errorCode != 2003)
- {
- // Cannot recover; return an error
- return -1;
- }
- }
- if(WebRtcNetEQ_RecOutMasterSlave(_inst[1], payloadSlave,
- &payloadLenSampleSlave, _masterSlaveInfo, 0) != 0)
- {
- LogError("RecOutMasterSlave", 1);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "RecOut: NetEq, error in pulling out for slave");
-
- // Check for errors that can be recovered from:
- // RECOUT_ERROR_SAMPLEUNDERRUN = 2003
- int errorCode = WebRtcNetEQ_GetErrorCode(_inst[0]);
- if(errorCode != 2003)
- {
- // Cannot recover; return an error
- return -1;
- }
- }
- }
- if(payloadLenSample != payloadLenSampleSlave)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, _id,
- "RecOut: mismatch between the lenght of the decoded \
-audio by Master (%d samples) and Slave (%d samples).",
- payloadLenSample, payloadLenSampleSlave);
- if(payloadLenSample > payloadLenSampleSlave)
- {
- memset(&payloadSlave[payloadLenSampleSlave], 0,
- (payloadLenSample - payloadLenSampleSlave) * sizeof(WebRtc_Word16));
- }
- }
-
- for(WebRtc_Word16 n = 0; n < payloadLenSample; n++)
- {
- audioFrame._payloadData[n<<1] = payloadMaster[n];
- audioFrame._payloadData[(n<<1)+1] = payloadSlave[n];
- }
- audioFrame._audioChannel = 2;
-
- WebRtcNetEQ_GetSpeechOutputType(_inst[0], &typeMaster);
- WebRtcNetEQ_GetSpeechOutputType(_inst[1], &typeSlave);
- if((typeMaster == kOutputNormal) ||
- (typeSlave == kOutputNormal))
- {
- type = kOutputNormal;
- }
- else
- {
- type = typeMaster;
- }
- }
-
- audioFrame._payloadDataLengthInSamples = static_cast<WebRtc_UWord16>(payloadLenSample);
- // NetEq always returns 10 ms of audio.
- _currentSampFreqKHz = static_cast<float>(audioFrame._payloadDataLengthInSamples) / 10.0f;
- audioFrame._frequencyInHz = audioFrame._payloadDataLengthInSamples * 100;
- if(_vadStatus)
- {
- if(type == kOutputVADPassive)
- {
- audioFrame._vadActivity = AudioFrame::kVadPassive;
- audioFrame._speechType = AudioFrame::kNormalSpeech;
- }
- else if(type == kOutputNormal)
- {
- audioFrame._vadActivity = AudioFrame::kVadActive;
- audioFrame._speechType = AudioFrame::kNormalSpeech;
- }
- else if(type == kOutputPLC)
- {
- audioFrame._vadActivity = _previousAudioActivity;
- audioFrame._speechType = AudioFrame::kPLC;
- }
- else if(type == kOutputCNG)
- {
- audioFrame._vadActivity = AudioFrame::kVadPassive;
- audioFrame._speechType = AudioFrame::kCNG;
- }
- else
- {
- audioFrame._vadActivity = AudioFrame::kVadPassive;
- audioFrame._speechType = AudioFrame::kPLCCNG;
- }
- }
- else
- {
- // Always return kVadUnknown when receive VAD is inactive
- audioFrame._vadActivity = AudioFrame::kVadUnknown;
-
- if(type == kOutputNormal)
- {
- audioFrame._speechType = AudioFrame::kNormalSpeech;
- }
- else if(type == kOutputPLC)
- {
- audioFrame._speechType = AudioFrame::kPLC;
- }
- else if(type == kOutputPLCtoCNG)
- {
- audioFrame._speechType = AudioFrame::kPLCCNG;
- }
- else if(type == kOutputCNG)
- {
- audioFrame._speechType = AudioFrame::kCNG;
- }
- else
- {
- // type is kOutputVADPassive which
- // we don't expect to get if _vadStatus is false
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, _id,
- "RecOut: NetEq returned kVadPassive while _vadStatus is false.");
- audioFrame._vadActivity = AudioFrame::kVadUnknown;
- audioFrame._speechType = AudioFrame::kNormalSpeech;
- }
- }
- _previousAudioActivity = audioFrame._vadActivity;
-
- return 0;
-}
-
-// When ACMGenericCodec has set the codec specific parameters in codecDef
-// it calls AddCodec() to add the new codec to the NetEQ database.
-WebRtc_Word32
-ACMNetEQ::AddCodec(
- WebRtcNetEQ_CodecDef* codecDef,
- bool toMaster)
-{
- if (codecDef == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "ACMNetEQ::AddCodec: error, codecDef is NULL");
- return -1;
- }
- CriticalSectionScoped lock(*_netEqCritSect);
-
- WebRtc_Word16 idx;
- if(toMaster)
- {
- idx = 0;
- }
- else
- {
- idx = 1;
- }
-
- if(!_isInitialized[idx])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "ACMNetEQ::AddCodec: NetEq is not initialized.");
- return -1;
- }
- if(WebRtcNetEQ_CodecDbAdd(_inst[idx], codecDef) < 0)
- {
- LogError("CodecDB_Add", idx);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "ACMNetEQ::AddCodec: NetEq, error in adding codec");
- return -1;
- }
- else
- {
- return 0;
- }
-}
-
-// Creates a Word16 RTP packet out of a Word8 payload and an rtp info struct.
-// Must be byte order safe.
-void
-ACMNetEQ::RTPPack(
- WebRtc_Word16* rtpPacket,
- const WebRtc_Word8* payload,
- const WebRtc_Word32 payloadLengthW8,
- const WebRtcRTPHeader& rtpInfo)
-{
- WebRtc_Word32 idx = 0;
- WEBRTC_SPL_SET_BYTE(rtpPacket, (WebRtc_Word8)0x80, idx);
- idx++;
-
- WEBRTC_SPL_SET_BYTE(rtpPacket, rtpInfo.header.payloadType, idx);
- idx++;
-
- WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
- &(rtpInfo.header.sequenceNumber), 1), idx);
- idx++;
-
- WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
- &(rtpInfo.header.sequenceNumber), 0), idx);
- idx++;
-
- WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
- &(rtpInfo.header.timestamp), 3), idx);
- idx++;
-
- WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
- &(rtpInfo.header.timestamp), 2), idx);
- idx++;
-
- WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
- &(rtpInfo.header.timestamp), 1), idx);
- idx++;
-
- WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
- &(rtpInfo.header.timestamp), 0), idx);
- idx++;
-
- WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
- &(rtpInfo.header.ssrc), 3), idx);
- idx++;
-
- WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
- &(rtpInfo.header.ssrc), 2), idx);
- idx++;
-
- WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
- &(rtpInfo.header.ssrc), 1), idx);
- idx++;
-
- WEBRTC_SPL_SET_BYTE(rtpPacket, WEBRTC_SPL_GET_BYTE(
- &(rtpInfo.header.ssrc), 0), idx);
- idx++;
-
- for (WebRtc_Word16 i=0; i < payloadLengthW8; i++)
- {
- WEBRTC_SPL_SET_BYTE(rtpPacket, payload[i], idx);
- idx++;
- }
- if (payloadLengthW8 & 1)
- {
- // Our 16 bits buffer is one byte too large, set that
- // last byte to zero.
- WEBRTC_SPL_SET_BYTE(rtpPacket, 0x0, idx);
- }
-}
-
-WebRtc_Word16
-ACMNetEQ::EnableVAD()
-{
- CriticalSectionScoped lock(*_netEqCritSect);
- if (_vadStatus)
- {
- return 0;
- }
- for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
- {
- if(!_isInitialized[idx])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "SetVADStatus: NetEq is not initialized.");
- return -1;
- }
- // VAD was off and we have to turn it on
- if(EnableVADByIdxSafe(idx) < 0)
- {
- return -1;
- }
-
- // Set previous VAD status to PASSIVE
- _previousAudioActivity = AudioFrame::kVadPassive;
- }
- _vadStatus = true;
- return 0;
-}
-
-
-ACMVADMode
-ACMNetEQ::VADMode() const
-{
- CriticalSectionScoped lock(*_netEqCritSect);
- return _vadMode;
-}
-
-
-WebRtc_Word16
-ACMNetEQ::SetVADMode(
- const ACMVADMode mode)
-{
- CriticalSectionScoped lock(*_netEqCritSect);
- if((mode < VADNormal) || (mode > VADVeryAggr))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "SetVADMode: NetEq error: could not set VAD mode, mode is not supported");
- return -1;
- }
- else
- {
- for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
- {
- if(!_isInitialized[idx])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "SetVADMode: NetEq is not initialized.");
- return -1;
- }
- if(WebRtcNetEQ_SetVADMode(_inst[idx], mode) < 0)
- {
- LogError("SetVADmode", idx);
- return -1;
- }
- }
- _vadMode = mode;
- return 0;
- }
-}
-
-
-WebRtc_Word32
-ACMNetEQ::FlushBuffers()
-{
- CriticalSectionScoped lock(*_netEqCritSect);
- for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
- {
- if(!_isInitialized[idx])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "FlushBuffers: NetEq is not initialized.");
- return -1;
- }
- if(WebRtcNetEQ_FlushBuffers(_inst[idx]) < 0)
- {
- LogError("FlushBuffers", idx);
- return -1;
- }
- }
- return 0;
-}
-
-
-WebRtc_Word32
-ACMNetEQ::GetVersion(
- char* version,
- WebRtc_UWord32& remainingBufferInBytes,
- WebRtc_UWord32& position)
-{
- WebRtc_UWord32 len = position;
- strncpy(&version[position], "NetEq\t\t", remainingBufferInBytes);
- position = (WebRtc_UWord32)strlen(version);
- remainingBufferInBytes -= (position - len);
- len = position;
-
- char myVersion[100];
- if(WebRtcNetEQ_GetVersion(myVersion) < 0)
- {
- return -1;
- }
-
- strncpy(&version[position], myVersion, remainingBufferInBytes);
- position = (WebRtc_UWord32)strlen(version);
- remainingBufferInBytes -= (position - len);
- len = position;
-
- strncpy(&version[position], "\n", remainingBufferInBytes);
- position = (WebRtc_UWord32)strlen(version);
- remainingBufferInBytes -= (position - len);
- len = position;
-
- return 0;
-}
-
-WebRtc_Word16
-ACMNetEQ::RemoveCodec(
- WebRtcNetEQDecoder codecIdx,
- bool isStereo)
-{
- // sanity check
- if((codecIdx <= kDecoderReservedStart) ||
- (codecIdx >= kDecoderReservedEnd))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "RemoveCodec: NetEq error: could not Remove Codec, codec index out of range");
- return -1;
- }
- CriticalSectionScoped lock(*_netEqCritSect);
- if(!_isInitialized[0])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "RemoveCodec: NetEq is not initialized.");
- return -1;
- }
-
- if(WebRtcNetEQ_CodecDbRemove(_inst[0], codecIdx) < 0)
- {
- LogError("CodecDB_Remove", 0);
- return -1;
- }
-
- if(isStereo)
- {
- if(WebRtcNetEQ_CodecDbRemove(_inst[1], codecIdx) < 0)
- {
- LogError("CodecDB_Remove", 1);
- return -1;
- }
- }
-
- return 0;
-}
-
-WebRtc_Word16
-ACMNetEQ::SetBackgroundNoiseMode(
- const ACMBackgroundNoiseMode mode)
-{
- CriticalSectionScoped lock(*_netEqCritSect);
- for(WebRtc_Word16 idx = 0; idx < _numSlaves + 1; idx++)
- {
- if(!_isInitialized[idx])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "SetBackgroundNoiseMode: NetEq is not initialized.");
- return -1;
- }
- if(WebRtcNetEQ_SetBGNMode(_inst[idx], (WebRtcNetEQBGNMode)mode) < 0)
- {
- LogError("SetBGNMode", idx);
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_Word16
-ACMNetEQ::BackgroundNoiseMode(
- ACMBackgroundNoiseMode& mode)
-{
- WebRtcNetEQBGNMode myMode;
- CriticalSectionScoped lock(*_netEqCritSect);
- if(!_isInitialized[0])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "BackgroundNoiseMode: NetEq is not initialized.");
- return -1;
- }
- if(WebRtcNetEQ_GetBGNMode(_inst[0], &myMode) < 0)
- {
- LogError("WebRtcNetEQ_GetBGNMode", 0);
- return -1;
- }
- else
- {
- mode = (ACMBackgroundNoiseMode)myMode;
- }
- return 0;
-}
-
-void
-ACMNetEQ::SetUniqueId(
- WebRtc_Word32 id)
-{
- CriticalSectionScoped lock(*_netEqCritSect);
- _id = id;
-}
-
-
-void
-ACMNetEQ::LogError(
- const char* neteqFuncName,
- const WebRtc_Word16 idx) const
-{
- char errorName[NETEQ_ERR_MSG_LEN_BYTE];
- char myFuncName[50];
- int neteqErrorCode = WebRtcNetEQ_GetErrorCode(_inst[idx]);
- WebRtcNetEQ_GetErrorName(neteqErrorCode, errorName, NETEQ_ERR_MSG_LEN_BYTE - 1);
- strncpy(myFuncName, neteqFuncName, 49);
- errorName[NETEQ_ERR_MSG_LEN_BYTE - 1] = '\0';
- myFuncName[49] = '\0';
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "NetEq-%d Error in function %s, error-code: %d, error-string: %s",
- idx,
- myFuncName,
- neteqErrorCode,
- errorName);
-}
-
-
-WebRtc_Word32
-ACMNetEQ::PlayoutTimestamp(
- WebRtc_UWord32& timestamp)
-{
- CriticalSectionScoped lock(*_netEqCritSect);
- if(WebRtcNetEQ_GetSpeechTimeStamp(_inst[0], ×tamp) < 0)
- {
- LogError("GetSpeechTimeStamp", 0);
- return -1;
- }
- else
- {
- return 0;
- }
-}
-
-WebRtc_Word16
-ACMNetEQ::AddSlave(
- const WebRtcNetEQDecoder* usedCodecs,
- WebRtc_Word16 noOfCodecs)
-{
- CriticalSectionScoped lock(*_netEqCritSect);
- const WebRtc_Word16 slaveIdx = 1;
- if(_numSlaves < 1)
- {
- // initialize the receiver, this also sets up VAD.
- if(InitByIdxSafe(slaveIdx) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "AddSlave: AddSlave Failed, Could not Initialize");
- return -1;
- }
-
- // Allocate buffer.
- if(AllocatePacketBufferByIdxSafe(usedCodecs, noOfCodecs, slaveIdx) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "AddSlave: AddSlave Failed, Could not Allocate Packet Buffer");
- return -1;
- }
-
- if(_masterSlaveInfo != NULL)
- {
- free(_masterSlaveInfo);
- _masterSlaveInfo = NULL;
- }
- int msInfoSize = WebRtcNetEQ_GetMasterSlaveInfoSize();
- _masterSlaveInfo = malloc(msInfoSize);
-
- if(_masterSlaveInfo == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "AddSlave: AddSlave Failed, Could not Allocate memory for Master-Slave Info");
- return -1;
- }
-
- // We accept this as initialized NetEQ, the rest is to synchronize
- // Slave with Master.
- _numSlaves = 1;
- _isInitialized[slaveIdx] = true;
-
- // Set Slave delay as all other instances.
- if(WebRtcNetEQ_SetExtraDelay(_inst[slaveIdx], _extraDelay) < 0)
- {
- LogError("SetExtraDelay", slaveIdx);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "AddSlave: AddSlave Failed, Could not set delay");
- return -1;
- }
-
- // Set AVT
- if(WebRtcNetEQ_SetAVTPlayout(_inst[slaveIdx], (_avtPlayout) ? 1 : 0) < 0)
- {
- LogError("SetAVTPlayout", slaveIdx);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "AddSlave: AddSlave Failed, Could not set AVT playout.");
- return -1;
- }
-
- // Set Background Noise
- WebRtcNetEQBGNMode currentMode;
- if(WebRtcNetEQ_GetBGNMode(_inst[0], ¤tMode) < 0)
- {
- LogError("GetBGNMode", 0);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "AAddSlave: AddSlave Failed, Could not Get BGN form Master.");
- return -1;
- }
-
- if(WebRtcNetEQ_SetBGNMode(_inst[slaveIdx], (WebRtcNetEQBGNMode)currentMode) < 0)
- {
- LogError("SetBGNMode", slaveIdx);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "AddSlave: AddSlave Failed, Could not set BGN mode.");
- return -1;
- }
-
- enum WebRtcNetEQPlayoutMode playoutMode = kPlayoutOff;
- switch(_playoutMode)
- {
- case voice:
- playoutMode = kPlayoutOn;
- break;
- case fax:
- playoutMode = kPlayoutFax;
- break;
- case streaming:
- playoutMode = kPlayoutStreaming;
- break;
- }
- if(WebRtcNetEQ_SetPlayoutMode(_inst[slaveIdx], playoutMode) < 0)
- {
- LogError("SetPlayoutMode", 1);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "AddSlave: AddSlave Failed, Could not Set Playout Mode.");
- return -1;
- }
- }
-
- return 0;
-}
-
-void
-ACMNetEQ::SetReceivedStereo(
- bool receivedStereo)
-{
- CriticalSectionScoped lock(*_netEqCritSect);
- _receivedStereo = receivedStereo;
-}
-
-WebRtc_UWord8
-ACMNetEQ::NumSlaves()
-{
- CriticalSectionScoped lock(*_netEqCritSect);
- return _numSlaves;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_neteq.h b/trunk/src/modules/audio_coding/main/source/acm_neteq.h
deleted file mode 100644
index 677c622..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_neteq.h
+++ /dev/null
@@ -1,383 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_
-
-#include "audio_coding_module.h"
-#include "audio_coding_module_typedefs.h"
-#include "engine_configurations.h"
-#include "module_common_types.h"
-#include "typedefs.h"
-#include "webrtc_neteq.h"
-#include "webrtc_vad.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class RWLockWrapper;
-struct CodecInst;
-enum AudioPlayoutMode;
-enum ACMSpeechType;
-
-#define MAX_NUM_SLAVE_NETEQ 1
-
-class ACMNetEQ
-{
-public:
- // Constructor of the class
- ACMNetEQ();
-
- // Destructor of the class.
- ~ACMNetEQ();
-
- //
- // GetVersion()
- // Fills the version array with the NetEQ version and updates the
- // remainingBufferInBytes and position variables accordingly.
- //
- // Output:
- // - version : An array to be filled with the version
- // data.
- //
- // Input/Output:
- // - remainingBuffInBytes : The number of free bytes at the end of
- // the version array.
- // - position : Position where the free space starts.
- //
- // Return value : 0 if ok.
- // -1 if NetEQ returned an error.
- //
- static WebRtc_Word32 GetVersion(
- char* version,
- WebRtc_UWord32& remainingBuffInBytes,
- WebRtc_UWord32& position);
-
- //
- // Init()
- // Allocates memory for NetEQ and VAD and initializes them.
- //
- // Return value : 0 if ok.
- // -1 if NetEQ or VAD returned an error or
- // if out of memory.
- //
- WebRtc_Word32 Init();
-
- //
- // RecIn()
- // Gives the payload to NetEQ.
- //
- // Input:
- // - incomingPayload : Incoming audio payload.
- // - payloadLength : Length of incoming audio payload.
- // - rtpInfo : RTP header for the incoming payload containing
- // information about payload type, sequence number,
- // timestamp, ssrc and marker bit.
- //
- // Return value : 0 if ok.
- // <0 if NetEQ returned an error.
- //
- WebRtc_Word32 RecIn(
- const WebRtc_Word8* incomingPayload,
- const WebRtc_Word32 payloadLength,
- const WebRtcRTPHeader& rtpInfo);
-
- //
- // RecOut()
- // Asks NetEQ for 10 ms of decoded audio.
- //
- // Input:
- // -audioFrame : an audio frame were output data and
- // associated parameters are written to.
- //
- // Return value : 0 if ok.
- // -1 if NetEQ returned an error.
- //
- WebRtc_Word32 RecOut(
- AudioFrame& audioFrame);
-
- //
- // AddCodec()
- // Adds a new codec to the NetEQ codec database.
- //
- // Input:
- // - codecDef : The codec to be added.
- // - toMaster : true if the codec has to be added to Master
- // NetEq, otherwise will be added to the Slave
- // NetEQ.
- //
- // Return value : 0 if ok.
- // <0 if NetEQ returned an error.
- //
- WebRtc_Word32 AddCodec(
- WebRtcNetEQ_CodecDef *codecDef,
- bool toMaster = true);
-
- //
- // AllocatePacketBuffer()
- // Allocates the NetEQ packet buffer.
- //
- // Input:
- // - usedCodecs : An array of the codecs to be used by NetEQ.
- // - noOfCodecs : Number of codecs in usedCodecs.
- //
- // Return value : 0 if ok.
- // <0 if NetEQ returned an error.
- //
- WebRtc_Word32 AllocatePacketBuffer(
- const WebRtcNetEQDecoder* usedCodecs,
- WebRtc_Word16 noOfCodecs);
-
- //
- // SetExtraDelay()
- // Sets an delayInMS milliseconds extra delay in NetEQ.
- //
- // Input:
- // - delayInMS : Extra delay in milliseconds.
- //
- // Return value : 0 if ok.
- // <0 if NetEQ returned an error.
- //
- WebRtc_Word32 SetExtraDelay(
- const WebRtc_Word32 delayInMS);
-
- //
- // SetAVTPlayout()
- // Enable/disable playout of AVT payloads.
- //
- // Input:
- // - enable : Enable if true, disable if false.
- //
- // Return value : 0 if ok.
- // <0 if NetEQ returned an error.
- //
- WebRtc_Word32 SetAVTPlayout(
- const bool enable);
-
- //
- // AVTPlayout()
- // Get the current AVT playout state.
- //
- // Return value : True if AVT playout is enabled.
- // False if AVT playout is disabled.
- //
- bool AVTPlayout() const;
-
- //
- // CurrentSampFreqHz()
- // Get the current sampling frequency in Hz.
- //
- // Return value : Sampling frequency in Hz.
- //
- WebRtc_Word32 CurrentSampFreqHz() const;
-
- //
- // SetPlayoutMode()
- // Sets the playout mode to voice or fax.
- //
- // Input:
- // - mode : The playout mode to be used, voice,
- // fax, or streaming.
- //
- // Return value : 0 if ok.
- // <0 if NetEQ returned an error.
- //
- WebRtc_Word32 SetPlayoutMode(
- const AudioPlayoutMode mode);
-
- //
- // PlayoutMode()
- // Get the current playout mode.
- //
- // Return value : The current playout mode.
- //
- AudioPlayoutMode PlayoutMode() const;
-
- //
- // NetworkStatistics()
- // Get the current network statistics from NetEQ.
- //
- // Output:
- // - statistics : The current network statistics.
- //
- // Return value : 0 if ok.
- // <0 if NetEQ returned an error.
- //
- WebRtc_Word32 NetworkStatistics(
- ACMNetworkStatistics* statistics) const;
-
- //
- // VADMode()
- // Get the current VAD Mode.
- //
- // Return value : The current VAD mode.
- //
- ACMVADMode VADMode() const;
-
- //
- // SetVADMode()
- // Set the VAD mode.
- //
- // Input:
- // - mode : The new VAD mode.
- //
- // Return value : 0 if ok.
- // -1 if an error occurred.
- //
- WebRtc_Word16 SetVADMode(
- const ACMVADMode mode);
-
- //
- // DecodeLock()
- // Get the decode lock used to protect decoder instances while decoding.
- //
- // Return value : Pointer to the decode lock.
- //
- RWLockWrapper* DecodeLock() const
- {
- return _decodeLock;
- }
-
- //
- // FlushBuffers()
- // Flushes the NetEQ packet and speech buffers.
- //
- // Return value : 0 if ok.
- // -1 if NetEQ returned an error.
- //
- WebRtc_Word32 FlushBuffers();
-
- //
- // RemoveCodec()
- // Removes a codec from the NetEQ codec database.
- //
- // Input:
- // - codecIdx : Codec to be removed.
- //
- // Return value : 0 if ok.
- // -1 if an error occurred.
- //
- WebRtc_Word16 RemoveCodec(
- WebRtcNetEQDecoder codecIdx,
- bool isStereo = false);
-
-
- //
- // SetBackgroundNoiseMode()
- // Set the mode of the background noise.
- //
- // Input:
- // - mode : an enumerator specifying the mode of the
- // background noise.
- //
- // Return value : 0 if succeeded,
- // -1 if failed to set the mode.
- //
- WebRtc_Word16 SetBackgroundNoiseMode(
- const ACMBackgroundNoiseMode mode);
-
- //
- // BackgroundNoiseMode()
- // return the mode of the background noise.
- //
- // Return value : The mode of background noise.
- //
- WebRtc_Word16 BackgroundNoiseMode(
- ACMBackgroundNoiseMode& mode);
-
- void SetUniqueId(
- WebRtc_Word32 id);
-
- WebRtc_Word32 PlayoutTimestamp(
- WebRtc_UWord32& timestamp);
-
- void SetReceivedStereo(
- bool receivedStereo);
-
- WebRtc_UWord8 NumSlaves();
-
- enum JB {masterJB = 0, slaveJB = 1};
-
- WebRtc_Word16 AddSlave(
- const WebRtcNetEQDecoder* usedCodecs,
- WebRtc_Word16 noOfCodecs);
-
-private:
- //
- // RTPPack()
- // Creates a Word16 RTP packet out of the payload data in Word16 and
- // a WebRtcRTPHeader.
- //
- // Input:
- // - payload : Payload to be packetized.
- // - payloadLengthW8 : Length of the payload in bytes.
- // - rtpInfo : RTP header struct.
- //
- // Output:
- // - rtpPacket : The RTP packet.
- //
- static void RTPPack(
- WebRtc_Word16* rtpPacket,
- const WebRtc_Word8* payload,
- const WebRtc_Word32 payloadLengthW8,
- const WebRtcRTPHeader& rtpInfo);
-
- void LogError(
- const char* neteqFuncName,
- const WebRtc_Word16 idx) const;
-
- WebRtc_Word16 InitByIdxSafe(
- const WebRtc_Word16 idx);
-
- // EnableVAD()
- // Enable VAD.
- //
- // Return value : 0 if ok.
- // -1 if an error occurred.
- //
- WebRtc_Word16 EnableVAD();
-
- WebRtc_Word16 EnableVADByIdxSafe(
- const WebRtc_Word16 idx);
-
- WebRtc_Word16 AllocatePacketBufferByIdxSafe(
- const WebRtcNetEQDecoder* usedCodecs,
- WebRtc_Word16 noOfCodecs,
- const WebRtc_Word16 idx);
-
- void* _inst[MAX_NUM_SLAVE_NETEQ + 1];
- void* _instMem[MAX_NUM_SLAVE_NETEQ + 1];
-
- WebRtc_Word16* _netEqPacketBuffer[MAX_NUM_SLAVE_NETEQ + 1];
-
- WebRtc_Word32 _id;
- float _currentSampFreqKHz;
- bool _avtPlayout;
- AudioPlayoutMode _playoutMode;
- CriticalSectionWrapper* _netEqCritSect;
-
- WebRtcVadInst* _ptrVADInst[MAX_NUM_SLAVE_NETEQ + 1];
-
- bool _vadStatus;
- ACMVADMode _vadMode;
- RWLockWrapper* _decodeLock;
- bool _isInitialized[MAX_NUM_SLAVE_NETEQ + 1];
- WebRtc_UWord8 _numSlaves;
- bool _receivedStereo;
- void* _masterSlaveInfo;
- AudioFrame::VADActivity _previousAudioActivity;
- WebRtc_Word32 _extraDelay;
-
- CriticalSectionWrapper* _callbackCritSect;
-};
-
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_NETEQ_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_neteq_unittest.cc b/trunk/src/modules/audio_coding/main/source/acm_neteq_unittest.cc
deleted file mode 100644
index 7feb2e0..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_neteq_unittest.cc
+++ /dev/null
@@ -1,147 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file contains unit tests for ACM's NetEQ wrapper (class ACMNetEQ).
-
-#include <stdlib.h>
-
-#include "gtest/gtest.h"
-#include "modules/audio_coding/codecs/pcm16b/include/pcm16b.h"
-#include "modules/audio_coding/main/interface/audio_coding_module_typedefs.h"
-#include "modules/audio_coding/main/source/acm_codec_database.h"
-#include "modules/audio_coding/main/source/acm_neteq.h"
-#include "modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
-#include "modules/interface/module_common_types.h"
-#include "typedefs.h" // NOLINT(build/include)
-
-namespace webrtc {
-
-class AcmNetEqTest : public ::testing::Test {
- protected:
- static const size_t kMaxPayloadLen = 5760; // 60 ms, 48 kHz, 16 bit samples.
- static const int kPcm16WbPayloadType = 94;
- AcmNetEqTest() {}
- virtual void SetUp();
- virtual void TearDown() {}
-
- void InsertZeroPacket(uint16_t sequence_number,
- uint32_t timestamp,
- uint8_t payload_type,
- uint32_t ssrc,
- bool marker_bit,
- size_t len_payload_bytes);
- void PullData(int expected_num_samples);
-
- ACMNetEQ neteq_;
-};
-
-void AcmNetEqTest::SetUp() {
- ASSERT_EQ(0, neteq_.Init());
- ASSERT_EQ(0, neteq_.AllocatePacketBuffer(ACMCodecDB::NetEQDecoders(),
- ACMCodecDB::kNumCodecs));
- WebRtcNetEQ_CodecDef codec_def;
- SET_CODEC_PAR(codec_def, kDecoderPCM16Bwb, kPcm16WbPayloadType, NULL, 16000);
- SET_PCM16B_WB_FUNCTIONS(codec_def);
- ASSERT_EQ(0, neteq_.AddCodec(&codec_def, true));
-}
-
-void AcmNetEqTest::InsertZeroPacket(uint16_t sequence_number,
- uint32_t timestamp,
- uint8_t payload_type,
- uint32_t ssrc,
- bool marker_bit,
- size_t len_payload_bytes) {
- ASSERT_TRUE(len_payload_bytes <= kMaxPayloadLen);
- uint16_t payload[kMaxPayloadLen] = {0};
- WebRtcRTPHeader rtp_header;
- rtp_header.header.sequenceNumber = sequence_number;
- rtp_header.header.timestamp = timestamp;
- rtp_header.header.ssrc = ssrc;
- rtp_header.header.payloadType = payload_type;
- rtp_header.header.markerBit = marker_bit;
- rtp_header.type.Audio.channel = 1;
- ASSERT_EQ(0, neteq_.RecIn(reinterpret_cast<WebRtc_Word8*>(payload),
- len_payload_bytes, rtp_header));
-}
-
-void AcmNetEqTest::PullData(int expected_num_samples) {
- AudioFrame out_frame;
- ASSERT_EQ(0, neteq_.RecOut(out_frame));
- ASSERT_EQ(expected_num_samples, out_frame._payloadDataLengthInSamples);
-}
-
-TEST_F(AcmNetEqTest, NetworkStatistics) {
- // Use fax mode to avoid time-scaling. This is to simplify the testing of
- // packet waiting times in the packet buffer.
- neteq_.SetPlayoutMode(fax);
- // Insert 31 dummy packets at once. Each packet contains 10 ms 16 kHz audio.
- int num_frames = 30;
- const int kSamples = 10 * 16;
- const int kPayloadBytes = kSamples * 2;
- int i, j;
- for (i = 0; i < num_frames; ++i) {
- InsertZeroPacket(i, i * kSamples, kPcm16WbPayloadType, 0x1234, false,
- kPayloadBytes);
- }
- // Pull out data once.
- PullData(kSamples);
- // Insert one more packet (to produce different mean and median).
- i = num_frames;
- InsertZeroPacket(i, i * kSamples, kPcm16WbPayloadType, 0x1234, false,
- kPayloadBytes);
- // Pull out all data.
- for (j = 1; j < num_frames + 1; ++j) {
- PullData(kSamples);
- }
-
- ACMNetworkStatistics stats;
- ASSERT_EQ(0, neteq_.NetworkStatistics(&stats));
- EXPECT_EQ(0, stats.currentBufferSize);
- EXPECT_EQ(0, stats.preferredBufferSize);
- EXPECT_FALSE(stats.jitterPeaksFound);
- EXPECT_EQ(0, stats.currentPacketLossRate);
- EXPECT_EQ(0, stats.currentDiscardRate);
- EXPECT_EQ(0, stats.currentExpandRate);
- EXPECT_EQ(0, stats.currentPreemptiveRate);
- EXPECT_EQ(0, stats.currentAccelerateRate);
- EXPECT_EQ(-916, stats.clockDriftPPM); // Initial value is slightly off.
- EXPECT_EQ(300, stats.maxWaitingTimeMs);
- EXPECT_EQ(10, stats.minWaitingTimeMs);
- EXPECT_EQ(159, stats.meanWaitingTimeMs);
- EXPECT_EQ(160, stats.medianWaitingTimeMs);
-}
-
-TEST_F(AcmNetEqTest, TestZeroLengthWaitingTimesVector) {
- // Insert one packet.
- const int kSamples = 10 * 16;
- const int kPayloadBytes = kSamples * 2;
- int i = 0;
- InsertZeroPacket(i, i * kSamples, kPcm16WbPayloadType, 0x1234, false,
- kPayloadBytes);
- // Do not pull out any data.
-
- ACMNetworkStatistics stats;
- ASSERT_EQ(0, neteq_.NetworkStatistics(&stats));
- EXPECT_EQ(0, stats.currentBufferSize);
- EXPECT_EQ(0, stats.preferredBufferSize);
- EXPECT_FALSE(stats.jitterPeaksFound);
- EXPECT_EQ(0, stats.currentPacketLossRate);
- EXPECT_EQ(0, stats.currentDiscardRate);
- EXPECT_EQ(0, stats.currentExpandRate);
- EXPECT_EQ(0, stats.currentPreemptiveRate);
- EXPECT_EQ(0, stats.currentAccelerateRate);
- EXPECT_EQ(-916, stats.clockDriftPPM); // Initial value is slightly off.
- EXPECT_EQ(-1, stats.minWaitingTimeMs);
- EXPECT_EQ(-1, stats.maxWaitingTimeMs);
- EXPECT_EQ(-1, stats.meanWaitingTimeMs);
- EXPECT_EQ(-1, stats.medianWaitingTimeMs);
-}
-
-} // namespace
diff --git a/trunk/src/modules/audio_coding/main/source/acm_opus.cc b/trunk/src/modules/audio_coding/main/source/acm_opus.cc
deleted file mode 100644
index ebdf971..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_opus.cc
+++ /dev/null
@@ -1,477 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_common_defs.h"
-#include "acm_neteq.h"
-#include "acm_opus.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-
-#ifdef WEBRTC_CODEC_OPUS
- // NOTE! Opus is not included in the open-source package. Modify this file or your codec
- // API to match the function call and name of used Opus API file.
- // #include "opus_interface.h"
-#endif
-
-namespace webrtc
-{
-
-#ifndef WEBRTC_CODEC_OPUS
-
-ACMOPUS::ACMOPUS(WebRtc_Word16 /* codecID */)
- : _encoderInstPtr(NULL),
- _decoderInstPtr(NULL),
- _mySampFreq(0),
- _myRate(0),
- _opusMode(0),
- _flagVBR(0) {
- return;
-}
-
-
-ACMOPUS::~ACMOPUS()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMOPUS::InternalEncode(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMOPUS::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMOPUS::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMOPUS::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
-}
-
-
-WebRtc_Word32
-ACMOPUS::CodecDef(
- WebRtcNetEQ_CodecDef& /* codecDef */,
- const CodecInst& /* codecInst */)
-{
- return -1;
-}
-
-
-ACMGenericCodec*
-ACMOPUS::CreateInstance(void)
-{
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMOPUS::InternalCreateEncoder()
-{
- return -1;
-}
-
-
-void
-ACMOPUS::DestructEncoderSafe()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMOPUS::InternalCreateDecoder()
-{
- return -1;
-}
-
-
-void
-ACMOPUS::DestructDecoderSafe()
-{
- return;
-}
-
-
-void
-ACMOPUS::InternalDestructEncoderInst(
- void* /* ptrInst */)
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMOPUS::UnregisterFromNetEqSafe(
- ACMNetEQ* /* netEq */,
- WebRtc_Word16 /* payloadType */)
-{
- return -1;
-}
-
-WebRtc_Word16
-ACMOPUS::SetBitRateSafe(
- const WebRtc_Word32 /*rate*/ )
-{
- return -1;
-}
-
-#else //===================== Actual Implementation =======================
-
-// Remove when integrating a real Opus wrapper
-extern WebRtc_Word16 WebRtcOpus_CreateEnc(OPUS_inst_t_** inst, WebRtc_Word16 samplFreq);
-extern WebRtc_Word16 WebRtcOpus_CreateDec(OPUS_inst_t_** inst, WebRtc_Word16 samplFreq);
-extern WebRtc_Word16 WebRtcOpus_FreeEnc(OPUS_inst_t_* inst);
-extern WebRtc_Word16 WebRtcOpus_FreeDec(OPUS_inst_t_* inst);
-extern WebRtc_Word16 WebRtcOpus_Encode(OPUS_inst_t_* encInst,
- WebRtc_Word16* input,
- WebRtc_Word16* output,
- WebRtc_Word16 len,
- WebRtc_Word16 byteLen);
-extern WebRtc_Word16 WebRtcOpus_EncoderInit(OPUS_inst_t_* encInst,
- WebRtc_Word16 samplFreq,
- WebRtc_Word16 mode,
- WebRtc_Word16 vbrFlag);
-extern WebRtc_Word16 WebRtcOpus_Decode(OPUS_inst_t_* decInst);
-extern WebRtc_Word16 WebRtcOpus_DecodeBwe(OPUS_inst_t_* decInst, WebRtc_Word16* input);
-extern WebRtc_Word16 WebRtcOpus_DecodePlc(OPUS_inst_t_* decInst);
-extern WebRtc_Word16 WebRtcOpus_DecoderInit(OPUS_inst_t_* decInst);
-
-ACMOPUS::ACMOPUS(WebRtc_Word16 codecID)
- : _encoderInstPtr(NULL),
- _decoderInstPtr(NULL),
- _mySampFreq(48000), // Default sampling frequency.
- _myRate(50000), // Default rate.
- _opusMode(1), // Default mode is the hybrid mode.
- _flagVBR(0) { // Default VBR off.
- _codecID = codecID;
-
- // Current implementation doesn't have DTX. That might change.
- _hasInternalDTX = false;
-
- return;
-}
-
-ACMOPUS::~ACMOPUS()
-{
- if(_encoderInstPtr != NULL)
- {
- WebRtcOpus_FreeEnc(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
- if(_decoderInstPtr != NULL)
- {
- WebRtcOpus_FreeDec(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
- return;
-}
-
-
-WebRtc_Word16
-ACMOPUS::InternalEncode(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte)
-{
- WebRtc_Word16 noEncodedSamples = 0;
- WebRtc_Word16 tmpLenByte = 0;
- *bitStreamLenByte = 0;
-
- WebRtc_Word16 byteLengthFrame = 0;
-
- // Derive what byte-length is requested
- byteLengthFrame = _myRate*_frameLenSmpl/(8*_mySampFreq);
-
- // Call Encoder
- *bitStreamLenByte = WebRtcOpus_Encode(_encoderInstPtr, &_inAudio[_inAudioIxRead],
- (WebRtc_Word16*)bitStream, _frameLenSmpl, byteLengthFrame);
-
- // increment the read index this tell the caller that how far
- // we have gone forward in reading the audio buffer
- _inAudioIxRead += _frameLenSmpl;
-
- // sanity check
- if(*bitStreamLenByte < 0)
- {
- // error has happened
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalEncode: Encode error for Opus");
- *bitStreamLenByte = 0;
- return -1;
- }
-
- return *bitStreamLenByte;
-}
-
-
-
-WebRtc_Word16
-ACMOPUS::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return 0;
-}
-
-
-WebRtc_Word16
-ACMOPUS::InternalInitEncoder(
- WebRtcACMCodecParams* codecParams)
-{
- //set the bit rate and initialize
- _myRate = codecParams->codecInstant.rate;
- return SetBitRateSafe( (WebRtc_UWord32)_myRate);
-}
-
-
-WebRtc_Word16
-ACMOPUS::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- if (WebRtcOpus_DecoderInit(_decoderInstPtr) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalInitDecoder: init decoder failed for Opus");
- return -1;
- }
- return 0;
-}
-
-
-WebRtc_Word32
-ACMOPUS::CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst)
-{
- if (!_decoderInitialized)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "CodeDef: Decoder uninitialized for Opus");
- return -1;
- }
-
- // Fill up the structure by calling
- // "SET_CODEC_PAR" & "SET_G729_FUNCTION."
- // Then call NetEQ to add the codec to it's
- // database.
- SET_CODEC_PAR((codecDef), kDecoderOpus, codecInst.pltype,
- _decoderInstPtr, 16000);
- SET_OPUS_FUNCTIONS((codecDef));
- return 0;
-}
-
-
-ACMGenericCodec*
-ACMOPUS::CreateInstance(void)
-{
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMOPUS::InternalCreateEncoder()
-{
- if (WebRtcOpus_CreateEnc(&_encoderInstPtr, _mySampFreq) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalCreateEncoder: create encoder failed for Opus");
- return -1;
- }
- return 0;
-}
-
-
-void
-ACMOPUS::DestructEncoderSafe()
-{
- _encoderExist = false;
- _encoderInitialized = false;
- if(_encoderInstPtr != NULL)
- {
- WebRtcOpus_FreeEnc(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
-}
-
-
-WebRtc_Word16
-ACMOPUS::InternalCreateDecoder()
-{
- if (WebRtcOpus_CreateDec(&_decoderInstPtr, _mySampFreq) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "InternalCreateDecoder: create decoder failed for Opus");
- return -1;
- }
- return 0;
-}
-
-
-void
-ACMOPUS::DestructDecoderSafe()
-{
- _decoderExist = false;
- _decoderInitialized = false;
- if(_decoderInstPtr != NULL)
- {
- WebRtcOpus_FreeDec(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
-}
-
-
-void
-ACMOPUS::InternalDestructEncoderInst(
- void* ptrInst)
-{
- if(ptrInst != NULL)
- {
- WebRtcOpus_FreeEnc((OPUS_inst_t*)ptrInst);
- }
- return;
-}
-
-
-WebRtc_Word16
-ACMOPUS::UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType)
-{
- if(payloadType != _decoderParams.codecInstant.pltype)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot unregister codec: given payload-type does not match \
-the stored payload type",
- _decoderParams.codecInstant.plname,
- payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
- return netEq->RemoveCodec(kDecoderOpus);
-}
-
-WebRtc_Word16
-ACMOPUS::SetBitRateSafe(
- const WebRtc_Word32 rate)
-{
- //allowed rates: {8000, 12000, 14000, 16000, 18000, 20000,
- // 22000, 24000, 26000, 28000, 30000, 32000};
- switch(rate)
- {
- case 8000:
- {
- _myRate = 8000;
- break;
- }
- case 12000:
- {
- _myRate = 12000;
- break;
- }
- case 14000:
- {
- _myRate = 14000;
- break;
- }
- case 16000:
- {
- _myRate = 16000;
- break;
- }
- case 18000:
- {
- _myRate = 18000;
- break;
- }
- case 20000:
- {
- _myRate = 20000;
- break;
- }
- case 22000:
- {
- _myRate = 22000;
- break;
- }
- case 24000:
- {
- _myRate = 24000;
- break;
- }
- case 26000:
- {
- _myRate = 26000;
- break;
- }
- case 28000:
- {
- _myRate = 28000;
- break;
- }
- case 30000:
- {
- _myRate = 30000;
- break;
- }
- case 32000:
- {
- _myRate = 32000;
- break;
- }
- default:
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "SetBitRateSafe: Invalid rate Opus");
- return -1;
- }
- }
-
- // Re-init with new rate
- if (WebRtcOpus_EncoderInit(_encoderInstPtr, _mySampFreq, _opusMode, _flagVBR) >= 0)
- {
- _encoderParams.codecInstant.rate = _myRate;
- return 0;
- }
- else
- {
- return -1;
- }
-}
-
-#endif
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_opus.h b/trunk/src/modules/audio_coding/main/source/acm_opus.h
deleted file mode 100644
index 278c369..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_opus.h
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
-
-#include "acm_generic_codec.h"
-
-// forward declaration
-struct OPUS_inst_t_;
-struct OPUS_inst_t_;
-
-namespace webrtc
-{
-
-class ACMOPUS: public ACMGenericCodec
-{
-public:
- ACMOPUS(WebRtc_Word16 codecID);
- ~ACMOPUS();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(
- WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(
- WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 InternalInitDecoder(
- WebRtcACMCodecParams *codecParams);
-
-protected:
- WebRtc_Word16 DecodeSafe(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio,
- WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- void InternalDestructEncoderInst(
- void* ptrInst);
-
- WebRtc_Word16 UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- WebRtc_Word16 SetBitRateSafe(
- const WebRtc_Word32 rate);
-
- OPUS_inst_t_* _encoderInstPtr;
- OPUS_inst_t_* _decoderInstPtr;
-
- WebRtc_UWord16 _mySampFreq;
- WebRtc_UWord16 _myRate;
- WebRtc_Word16 _opusMode;
- WebRtc_Word16 _flagVBR;
-
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_OPUS_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_pcm16b.cc b/trunk/src/modules/audio_coding/main/source/acm_pcm16b.cc
deleted file mode 100644
index 5c93e90..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_pcm16b.cc
+++ /dev/null
@@ -1,334 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_codec_database.h"
-#include "acm_common_defs.h"
-#include "acm_neteq.h"
-#include "acm_pcm16b.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-
-#ifdef WEBRTC_CODEC_PCM16
- #include "pcm16b.h"
-#endif
-
-namespace webrtc
-{
-
-#ifndef WEBRTC_CODEC_PCM16
-
-ACMPCM16B::ACMPCM16B(
- WebRtc_Word16 /* codecID */)
-{
- return;
-}
-
-
-ACMPCM16B::~ACMPCM16B()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMPCM16B::InternalEncode(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMPCM16B::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMPCM16B::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMPCM16B::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
-}
-
-
-WebRtc_Word32
-ACMPCM16B::CodecDef(
- WebRtcNetEQ_CodecDef& /* codecDef */,
- const CodecInst& /* codecInst */)
-{
- return -1;
-}
-
-
-ACMGenericCodec*
-ACMPCM16B::CreateInstance(void)
-{
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMPCM16B::InternalCreateEncoder()
-{
- return -1;
-}
-
-
-WebRtc_Word16
-ACMPCM16B::InternalCreateDecoder()
-{
- return -1;
-}
-
-
-void
-ACMPCM16B::InternalDestructEncoderInst(
- void* /* ptrInst */)
-{
- return;
-}
-
-
-void
-ACMPCM16B::DestructEncoderSafe()
-{
- return;
-}
-
-void
-ACMPCM16B::DestructDecoderSafe()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMPCM16B::UnregisterFromNetEqSafe(
- ACMNetEQ* /* netEq */,
- WebRtc_Word16 /* payloadType */)
-{
- return -1;
-}
-
-
-
-#else //===================== Actual Implementation =======================
-
-
-ACMPCM16B::ACMPCM16B(
- WebRtc_Word16 codecID)
-{
- _codecID = codecID;
- _samplingFreqHz = ACMCodecDB::CodecFreq(_codecID);
-}
-
-
-ACMPCM16B::~ACMPCM16B()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMPCM16B::InternalEncode(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte)
-{
- *bitStreamLenByte = WebRtcPcm16b_Encode(&_inAudio[_inAudioIxRead],
- _frameLenSmpl*_noChannels,
- bitStream);
- // increment the read index to tell the caller that how far
- // we have gone forward in reading the audio buffer
- _inAudioIxRead += _frameLenSmpl*_noChannels;
- return *bitStreamLenByte;
-}
-
-
-WebRtc_Word16
-ACMPCM16B::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return 0;
-}
-
-
-WebRtc_Word16
-ACMPCM16B::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- // This codec does not need initialization,
- // PCM has no instance
- return 0;
-}
-
-
-WebRtc_Word16
-ACMPCM16B::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- // This codec does not need initialization,
- // PCM has no instance
- return 0;
-}
-
-
-WebRtc_Word32
-ACMPCM16B::CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst)
-{
- // Fill up the structure by calling
- // "SET_CODEC_PAR" & "SET_PCMU_FUNCTION."
- // Then call NetEQ to add the codec to it's
- // database.
- switch(_samplingFreqHz)
- {
- case 8000:
- {
- SET_CODEC_PAR((codecDef), kDecoderPCM16B, codecInst.pltype,
- NULL, 8000);
- SET_PCM16B_FUNCTIONS((codecDef));
- break;
- }
- case 16000:
- {
- SET_CODEC_PAR((codecDef), kDecoderPCM16Bwb, codecInst.pltype,
- NULL, 16000);
- SET_PCM16B_WB_FUNCTIONS((codecDef));
- break;
- }
- case 32000:
- {
- SET_CODEC_PAR((codecDef), kDecoderPCM16Bswb32kHz,
- codecInst.pltype, NULL, 32000);
- SET_PCM16B_SWB32_FUNCTIONS((codecDef));
- break;
- }
- default:
- {
- return -1;
- }
- }
- return 0;
-}
-
-
-ACMGenericCodec*
-ACMPCM16B::CreateInstance(void)
-{
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMPCM16B::InternalCreateEncoder()
-{
- // PCM has no instance
- return 0;
-}
-
-
-WebRtc_Word16
-ACMPCM16B::InternalCreateDecoder()
-{
- // PCM has no instance
- return 0;
-}
-
-
-void
-ACMPCM16B::InternalDestructEncoderInst(
- void* /* ptrInst */)
-{
- // PCM has no instance
- return;
-}
-
-
-void
-ACMPCM16B::DestructEncoderSafe()
-{
- // PCM has no instance
- _encoderExist = false;
- _encoderInitialized = false;
- return;
-}
-
-void
-ACMPCM16B::DestructDecoderSafe()
-{
- // PCM has no instance
- _decoderExist = false;
- _decoderInitialized = false;
- return;
-}
-
-
-WebRtc_Word16
-ACMPCM16B::UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType)
-{
- if(payloadType != _decoderParams.codecInstant.pltype)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot unregister codec %s given payload-type %d does not match \
-the stored payload type",
- _decoderParams.codecInstant.plname,
- payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
-
- switch(_samplingFreqHz)
- {
- case 8000:
- {
- return netEq->RemoveCodec(kDecoderPCM16B);
- }
- case 16000:
- {
- return netEq->RemoveCodec(kDecoderPCM16Bwb);
- }
- case 32000:
- {
- return netEq->RemoveCodec(kDecoderPCM16Bswb32kHz);
- }
- default:
- {
- return -1;
- }
- }
-}
-
-#endif
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_pcm16b.h b/trunk/src/modules/audio_coding/main/source/acm_pcm16b.h
deleted file mode 100644
index a81d8fe..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_pcm16b.h
+++ /dev/null
@@ -1,69 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_
-
-#include "acm_generic_codec.h"
-
-namespace webrtc
-{
-
-class ACMPCM16B : public ACMGenericCodec
-{
-public:
- ACMPCM16B(WebRtc_Word16 codecID);
- ~ACMPCM16B();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(
- WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(
- WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 InternalInitDecoder(
- WebRtcACMCodecParams *codecParams);
-
-protected:
- WebRtc_Word16 DecodeSafe(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio,
- WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- WebRtc_Word16 UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- void InternalDestructEncoderInst(
- void* ptrInst);
-
- WebRtc_Word32 _samplingFreqHz;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCM16B_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_pcma.cc b/trunk/src/modules/audio_coding/main/source/acm_pcma.cc
deleted file mode 100644
index c86bd1c..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_pcma.cc
+++ /dev/null
@@ -1,165 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_common_defs.h"
-#include "acm_neteq.h"
-#include "acm_pcma.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-
-// Codec interface
-#include "g711_interface.h"
-
-namespace webrtc
-{
-
-ACMPCMA::ACMPCMA(WebRtc_Word16 codecID)
-{
- _codecID = codecID;
-}
-
-
-ACMPCMA::~ACMPCMA()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMPCMA::InternalEncode(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte)
-{
- *bitStreamLenByte = WebRtcG711_EncodeA(NULL, &_inAudio[_inAudioIxRead],
- _frameLenSmpl*_noChannels, (WebRtc_Word16*)bitStream);
- // increment the read index this tell the caller that how far
- // we have gone forward in reading the audio buffer
- _inAudioIxRead += _frameLenSmpl*_noChannels;
- return *bitStreamLenByte;
-}
-
-
-WebRtc_Word16
-ACMPCMA::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return 0;
-}
-
-
-WebRtc_Word16
-ACMPCMA::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- // This codec does not need initialization,
- // PCM has no instance
- return 0;
-}
-
-
-WebRtc_Word16
-ACMPCMA::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- // This codec does not need initialization,
- // PCM has no instance
- return 0;
-}
-
-
-WebRtc_Word32 ACMPCMA::CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst)
-{
- // Fill up the structure by calling
- // "SET_CODEC_PAR" & "SET_PCMA_FUNCTION."
- // Then call NetEQ to add the codec to it's
- // database.
- SET_CODEC_PAR((codecDef), kDecoderPCMa, codecInst.pltype, NULL, 8000);
- SET_PCMA_FUNCTIONS((codecDef));
- return 0;
-}
-
-
-ACMGenericCodec*
-ACMPCMA::CreateInstance(void)
-{
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMPCMA::InternalCreateEncoder()
-{
- // PCM has no instance
- return 0;
-}
-
-
-WebRtc_Word16
-ACMPCMA::InternalCreateDecoder()
-{
- // PCM has no instance
- return 0;
-}
-
-
-void
-ACMPCMA::InternalDestructEncoderInst(
- void* /* ptrInst */)
-{
- // PCM has no instance
- return;
-}
-
-
-void
-ACMPCMA::DestructEncoderSafe()
-{
- // PCM has no instance
- return;
-}
-
-
-void
-ACMPCMA::DestructDecoderSafe()
-{
- // PCM has no instance
- _decoderInitialized = false;
- _decoderExist = false;
- return;
-}
-
-
-WebRtc_Word16
-ACMPCMA::UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType)
-{
- if(payloadType != _decoderParams.codecInstant.pltype)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot unregister codec %s given payload-type %d does not match \
-the stored payload type",
- _decoderParams.codecInstant.plname,
- payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
-
- return netEq->RemoveCodec(kDecoderPCMa);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_pcma.h b/trunk/src/modules/audio_coding/main/source/acm_pcma.h
deleted file mode 100644
index db25798..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_pcma.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_
-
-#include "acm_generic_codec.h"
-
-namespace webrtc
-{
-
-class ACMPCMA : public ACMGenericCodec
-{
-public:
- ACMPCMA(WebRtc_Word16 codecID);
- ~ACMPCMA();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(
- WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(
- WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 InternalInitDecoder(
- WebRtcACMCodecParams *codecParams);
-
-protected:
- WebRtc_Word16 DecodeSafe(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio,
- WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- WebRtc_Word16 UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- void InternalDestructEncoderInst(
- void* ptrInst);
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMA_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_pcmu.cc b/trunk/src/modules/audio_coding/main/source/acm_pcmu.cc
deleted file mode 100644
index 320ba5b..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_pcmu.cc
+++ /dev/null
@@ -1,167 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_common_defs.h"
-#include "acm_neteq.h"
-#include "acm_pcmu.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-
-// Codec interface
-#include "g711_interface.h"
-
-namespace webrtc
-{
-
-ACMPCMU::ACMPCMU(WebRtc_Word16 codecID)
-{
- _codecID = codecID;
-}
-
-
-ACMPCMU::~ACMPCMU()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMPCMU::InternalEncode(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte)
-{
- *bitStreamLenByte = WebRtcG711_EncodeU(NULL, &_inAudio[_inAudioIxRead],
- _frameLenSmpl*_noChannels, (WebRtc_Word16*)bitStream);
- // increment the read index this tell the caller that how far
- // we have gone forward in reading the audio buffer
- _inAudioIxRead += _frameLenSmpl*_noChannels;
- return *bitStreamLenByte;
-}
-
-
-WebRtc_Word16
-ACMPCMU::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return 0;
-}
-
-
-WebRtc_Word16
-ACMPCMU::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- // This codec does not need initialization,
- // PCM has no instance
- return 0;
-}
-
-
-WebRtc_Word16
-ACMPCMU::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- // This codec does not need initialization,
- // PCM has no instance
- return 0;
-}
-
-
-WebRtc_Word32
-ACMPCMU::CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst)
-{
- // Fill up the structure by calling
- // "SET_CODEC_PAR" & "SET_PCMU_FUNCTION."
- // Then call NetEQ to add the codec to it's
- // database.
- SET_CODEC_PAR((codecDef), kDecoderPCMu, codecInst.pltype, NULL, 8000);
- SET_PCMU_FUNCTIONS((codecDef));
- return 0;
-}
-
-
-ACMGenericCodec*
-ACMPCMU::CreateInstance(void)
-{
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMPCMU::InternalCreateEncoder()
-{
- // PCM has no instance
- return 0;
-}
-
-
-WebRtc_Word16
-ACMPCMU::InternalCreateDecoder()
-{
- // PCM has no instance
- return 0;
-}
-
-
-void
-ACMPCMU::InternalDestructEncoderInst(
- void* /* ptrInst */)
-{
- // PCM has no instance
- return;
-}
-
-
-void
-ACMPCMU::DestructEncoderSafe()
-{
- // PCM has no instance
- _encoderExist = false;
- _encoderInitialized = false;
- return;
-}
-
-void ACMPCMU::DestructDecoderSafe()
-{
- // PCM has no instance
- _decoderInitialized = false;
- _decoderExist = false;
- return;
-}
-
-
-WebRtc_Word16
-ACMPCMU::UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType)
-{
-
- if(payloadType != _decoderParams.codecInstant.pltype)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot unregister codec %s given payload-type %d does not match \
-the stored payload type",
- _decoderParams.codecInstant.plname,
- payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
-
- return netEq->RemoveCodec(kDecoderPCMu);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_pcmu.h b/trunk/src/modules/audio_coding/main/source/acm_pcmu.h
deleted file mode 100644
index 2fc4223..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_pcmu.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_
-
-#include "acm_generic_codec.h"
-
-namespace webrtc
-{
-
-class ACMPCMU : public ACMGenericCodec
-{
-public:
- ACMPCMU(WebRtc_Word16 codecID);
- ~ACMPCMU();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(
- WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(
- WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 InternalInitDecoder(
- WebRtcACMCodecParams *codecParams);
-
-protected:
- WebRtc_Word16 DecodeSafe(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio,
- WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- WebRtc_Word16 UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- void InternalDestructEncoderInst(
- void* ptrInst);
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_PCMU_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_red.cc b/trunk/src/modules/audio_coding/main/source/acm_red.cc
deleted file mode 100644
index 88dbc24..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_red.cc
+++ /dev/null
@@ -1,163 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_red.h"
-#include "acm_neteq.h"
-#include "acm_common_defs.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-
-namespace webrtc
-{
-
-ACMRED::ACMRED(WebRtc_Word16 codecID)
-{
- _codecID = codecID;
-}
-
-
-ACMRED::~ACMRED()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMRED::InternalEncode(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */)
-{
- // RED is never used as an encoder
- // RED has no instance
- return 0;
-}
-
-
-WebRtc_Word16
-ACMRED::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return 0;
-}
-
-
-WebRtc_Word16
-ACMRED::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- // This codec does not need initialization,
- // RED has no instance
- return 0;
-}
-
-
-WebRtc_Word16
-ACMRED::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- // This codec does not need initialization,
- // RED has no instance
- return 0;
-}
-
-
-WebRtc_Word32
-ACMRED::CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst)
-{
- if (!_decoderInitialized)
- {
- // Todo:
- // log error
- return -1;
- }
-
- // Fill up the structure by calling
- // "SET_CODEC_PAR" & "SET_PCMU_FUNCTION."
- // Then call NetEQ to add the codec to it's
- // database.
- SET_CODEC_PAR((codecDef), kDecoderRED, codecInst.pltype, NULL, 8000);
- SET_RED_FUNCTIONS((codecDef));
- return 0;
-}
-
-
-ACMGenericCodec*
-ACMRED::CreateInstance(void)
-{
- return NULL;
-}
-
-
-WebRtc_Word16
-ACMRED::InternalCreateEncoder()
-{
- // RED has no instance
- return 0;
-}
-
-
-WebRtc_Word16
-ACMRED::InternalCreateDecoder()
-{
- // RED has no instance
- return 0;
-}
-
-
-void
-ACMRED::InternalDestructEncoderInst(
- void* /* ptrInst */)
-{
- // RED has no instance
- return;
-}
-
-
-void
-ACMRED::DestructEncoderSafe()
-{
- // RED has no instance
- return;
-}
-
-void ACMRED::DestructDecoderSafe()
-{
- // RED has no instance
- return;
-}
-
-
-WebRtc_Word16
-ACMRED::UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType)
-{
- if(payloadType != _decoderParams.codecInstant.pltype)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot unregister codec %s given payload-type %d does not match \
-the stored payload type",
- _decoderParams.codecInstant.plname,
- payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
-
- return netEq->RemoveCodec(kDecoderRED);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_red.h b/trunk/src/modules/audio_coding/main/source/acm_red.h
deleted file mode 100644
index d22cb8c..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_red.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_
-
-#include "acm_generic_codec.h"
-
-namespace webrtc
-{
-
-class ACMRED : public ACMGenericCodec
-{
-public:
- ACMRED(WebRtc_Word16 codecID);
- ~ACMRED();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(
- WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(
- WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 InternalInitDecoder(
- WebRtcACMCodecParams *codecParams);
-
-protected:
- WebRtc_Word16 DecodeSafe(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio,
- WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- WebRtc_Word16 UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- void InternalDestructEncoderInst(
- void* ptrInst);
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RED_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_resampler.cc b/trunk/src/modules/audio_coding/main/source/acm_resampler.cc
deleted file mode 100644
index 1e36e73..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_resampler.cc
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string.h>
-
-#include "acm_resampler.h"
-#include "critical_section_wrapper.h"
-#include "resampler.h"
-#include "signal_processing_library.h"
-#include "trace.h"
-
-namespace webrtc
-{
-
-ACMResampler::ACMResampler():
-
-_resamplerCritSect(*CriticalSectionWrapper::CreateCriticalSection())
-{
-}
-
-ACMResampler::~ACMResampler()
-{
-
- delete &_resamplerCritSect;
-}
-
-
-WebRtc_Word16
-ACMResampler::Resample10Msec(
- const WebRtc_Word16* inAudio,
- WebRtc_Word32 inFreqHz,
- WebRtc_Word16* outAudio,
- WebRtc_Word32 outFreqHz,
- WebRtc_UWord8 numAudioChannels)
-{
-
- CriticalSectionScoped cs(_resamplerCritSect);
-
- if(inFreqHz == outFreqHz)
- {
- size_t length = static_cast<size_t>(inFreqHz * numAudioChannels / 100);
- memcpy(outAudio, inAudio, length * sizeof(WebRtc_Word16));
- return static_cast<WebRtc_Word16>(inFreqHz / 100);
- }
-
- int maxLen = 480 * numAudioChannels; //max number of samples for 10ms at 48kHz
- int lengthIn = (WebRtc_Word16)(inFreqHz / 100) * numAudioChannels;
- int outLen;
-
- WebRtc_Word32 ret;
- ResamplerType type;
- type = (numAudioChannels == 1)? kResamplerSynchronous:kResamplerSynchronousStereo;
-
- ret = _resampler.ResetIfNeeded(inFreqHz,outFreqHz,type);
- if (ret < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
- "Error in reset of resampler");
- return -1;
- }
-
- ret = _resampler.Push(inAudio, lengthIn, outAudio, maxLen, outLen);
- if (ret < 0 )
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, 0,
- "Error in resampler: resampler.Push");
- return -1;
- }
-
- WebRtc_Word16 outAudioLenSmpl = (WebRtc_Word16) outLen / numAudioChannels;
-
- return outAudioLenSmpl;
-
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_resampler.h b/trunk/src/modules/audio_coding/main/source/acm_resampler.h
deleted file mode 100644
index 246a4fe..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_resampler.h
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_
-
-#include "resampler.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-
-class ACMResampler
-{
-public:
- ACMResampler();
- ~ACMResampler();
-
- WebRtc_Word16 Resample10Msec(
- const WebRtc_Word16* inAudio,
- const WebRtc_Word32 inFreqHz,
- WebRtc_Word16* outAudio,
- const WebRtc_Word32 outFreqHz,
- WebRtc_UWord8 numAudioChannels);
-
-private:
-
- //Use the Resampler class
- Resampler _resampler;
- CriticalSectionWrapper& _resamplerCritSect;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_RESAMPLER_H_
diff --git a/trunk/src/modules/audio_coding/main/source/acm_speex.cc b/trunk/src/modules/audio_coding/main/source/acm_speex.cc
deleted file mode 100644
index acb2a3b..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_speex.cc
+++ /dev/null
@@ -1,669 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_speex.h"
-#include "acm_codec_database.h"
-#include "acm_common_defs.h"
-#include "acm_neteq.h"
-#include "trace.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_help_macros.h"
-
-#ifdef WEBRTC_CODEC_SPEEX
- // NOTE! Speex is not included in the open-source package. The following
- // interface file is needed:
- //
- // /modules/audio_coding/codecs/speex/main/interface/speex_interface.h
- //
- // The API in the header file should match the one below.
- //
- // int16_t WebRtcSpeex_CreateEnc(SPEEX_encinst_t **SPEEXenc_inst,
- // int32_t fs);
- // int16_t WebRtcSpeex_FreeEnc(SPEEX_encinst_t *SPEEXenc_inst);
- // int16_t WebRtcSpeex_CreateDec(SPEEX_decinst_t **SPEEXdec_inst,
- // int32_t fs,
- // int16_t enh_enabled);
- // int16_t WebRtcSpeex_FreeDec(SPEEX_decinst_t *SPEEXdec_inst);
- // int16_t WebRtcSpeex_Encode(SPEEX_encinst_t *SPEEXenc_inst,
- // int16_t *speechIn,
- // int32_t rate);
- // int16_t WebRtcSpeex_EncoderInit(SPEEX_encinst_t *SPEEXenc_inst,
- // int16_t vbr, int16_t complexity,
- // int16_t vad_enable);
- // int16_t WebRtcSpeex_GetBitstream(SPEEX_encinst_t *SPEEXenc_inst,
- // int16_t *encoded);
- // int16_t WebRtcSpeex_DecodePlc(SPEEX_decinst_t *SPEEXdec_inst,
- // int16_t *decoded, int16_t noOfLostFrames);
- // int16_t WebRtcSpeex_Decode(SPEEX_decinst_t *SPEEXdec_inst,
- // int16_t *encoded, int16_t len,
- // int16_t *decoded, int16_t *speechType);
- // int16_t WebRtcSpeex_DecoderInit(SPEEX_decinst_t *SPEEXdec_inst);
- // void WebRtcSpeex_Version(char *versionStr, short len);
- #include "speex_interface.h"
-#endif
-
-namespace webrtc {
-
-#ifndef WEBRTC_CODEC_SPEEX
-ACMSPEEX::ACMSPEEX(WebRtc_Word16 /* codecID*/)
- : _encoderInstPtr(NULL),
- _decoderInstPtr(NULL),
- _complMode(0),
- _vbrEnabled(false),
- _encodingRate(-1),
- _samplingFrequency(-1),
- _samplesIn20MsAudio(-1) {
- return;
-}
-
-ACMSPEEX::~ACMSPEEX()
-{
- return;
-}
-
-WebRtc_Word16
-ACMSPEEX::InternalEncode(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16* /* bitStreamLenByte */)
-{
- return -1;
-}
-
-WebRtc_Word16
-ACMSPEEX::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return -1;
-}
-
-WebRtc_Word16
-ACMSPEEX::EnableDTX()
-{
- return -1;
-}
-
-WebRtc_Word16
-ACMSPEEX::DisableDTX()
-{
- return -1;
-}
-
-WebRtc_Word16
-ACMSPEEX::InternalInitEncoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
-}
-
-WebRtc_Word16
-ACMSPEEX::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- return -1;
-}
-
-WebRtc_Word32
-ACMSPEEX::CodecDef(
- WebRtcNetEQ_CodecDef& /* codecDef */,
- const CodecInst& /* codecInst */)
-{
- return -1;
-}
-
-ACMGenericCodec*
-ACMSPEEX::CreateInstance(void)
-{
- return NULL;
-}
-
-WebRtc_Word16
-ACMSPEEX::InternalCreateEncoder()
-{
- return -1;
-}
-
-void
-ACMSPEEX::DestructEncoderSafe()
-{
- return;
-}
-
-
-WebRtc_Word16
-ACMSPEEX::InternalCreateDecoder()
-{
- return -1;
-}
-
-void
-ACMSPEEX::DestructDecoderSafe()
-{
- return;
-}
-
-WebRtc_Word16
-ACMSPEEX::SetBitRateSafe(
- const WebRtc_Word32 /* rate */)
-{
- return -1;
-}
-
-void
-ACMSPEEX::InternalDestructEncoderInst(
- void* /* ptrInst */)
-{
- return;
-}
-
-WebRtc_Word16
-ACMSPEEX::UnregisterFromNetEqSafe(
- ACMNetEQ* /* netEq */,
- WebRtc_Word16 /* payloadType */)
-{
- return -1;
-}
-
-#ifdef UNUSEDSPEEX
-WebRtc_Word16
-ACMSPEEX::EnableVBR()
-{
- return -1;
-}
-
-WebRtc_Word16
-ACMSPEEX::DisableVBR()
-{
- return -1;
-}
-
-WebRtc_Word16
-ACMSPEEX::SetComplMode(
- WebRtc_Word16 mode)
-{
- return -1;
-}
-#endif
-
-#else //===================== Actual Implementation =======================
-
-ACMSPEEX::ACMSPEEX(WebRtc_Word16 codecID):
-_encoderInstPtr(NULL),
-_decoderInstPtr(NULL)
-{
- _codecID = codecID;
-
- // Set sampling frequency, frame size and rate Speex
- if(_codecID == ACMCodecDB::kSPEEX8)
- {
- _samplingFrequency = 8000;
- _samplesIn20MsAudio = 160;
- _encodingRate = 11000;
- }
- else if(_codecID == ACMCodecDB::kSPEEX16)
- {
- _samplingFrequency = 16000;
- _samplesIn20MsAudio = 320;
- _encodingRate = 22000;
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Wrong codec id for Speex.");
-
- _samplingFrequency = -1;
- _samplesIn20MsAudio = -1;
- _encodingRate = -1;
- }
-
- _hasInternalDTX = true;
- _dtxEnabled = false;
- _vbrEnabled = false;
- _complMode = 3; // default complexity value
-
- return;
-}
-
-ACMSPEEX::~ACMSPEEX()
-{
- if(_encoderInstPtr != NULL)
- {
- WebRtcSpeex_FreeEnc(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
- if(_decoderInstPtr != NULL)
- {
- WebRtcSpeex_FreeDec(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
- return;
-}
-
-WebRtc_Word16
-ACMSPEEX::InternalEncode(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16* bitStreamLenByte)
-{
- WebRtc_Word16 status;
- WebRtc_Word16 numEncodedSamples = 0;
- WebRtc_Word16 n = 0;
-
- while( numEncodedSamples < _frameLenSmpl)
- {
- status = WebRtcSpeex_Encode(_encoderInstPtr, &_inAudio[_inAudioIxRead],
- _encodingRate);
-
- // increment the read index this tell the caller that how far
- // we have gone forward in reading the audio buffer
- _inAudioIxRead += _samplesIn20MsAudio;
- numEncodedSamples += _samplesIn20MsAudio;
-
- if(status < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Error in Speex encoder");
- return status;
- }
-
- // Update VAD, if internal DTX is used
- if(_hasInternalDTX && _dtxEnabled)
- {
- _vadLabel[n++] = status;
- _vadLabel[n++] = status;
- }
-
- if(status == 0)
- {
- // This frame is detected as inactive. We need send whatever
- // encoded so far.
- *bitStreamLenByte = WebRtcSpeex_GetBitstream(_encoderInstPtr,
- (WebRtc_Word16*)bitStream);
-
- return *bitStreamLenByte;
- }
- }
-
- *bitStreamLenByte = WebRtcSpeex_GetBitstream(_encoderInstPtr,
- (WebRtc_Word16*)bitStream);
- return *bitStreamLenByte;
-}
-
-WebRtc_Word16
-ACMSPEEX::DecodeSafe(
- WebRtc_UWord8* /* bitStream */,
- WebRtc_Word16 /* bitStreamLenByte */,
- WebRtc_Word16* /* audio */,
- WebRtc_Word16* /* audioSamples */,
- WebRtc_Word8* /* speechType */)
-{
- return 0;
-}
-
-WebRtc_Word16
-ACMSPEEX::EnableDTX()
-{
- if(_dtxEnabled)
- {
- return 0;
- }
- else if(_encoderExist) // check if encoder exist
- {
- // enable DTX
- if(WebRtcSpeex_EncoderInit(_encoderInstPtr, (_vbrEnabled ? 1:0), _complMode, 1) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot enable DTX for Speex");
- return -1;
- }
- _dtxEnabled = true;
- return 0;
- }
- else
- {
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word16
-ACMSPEEX::DisableDTX()
-{
- if(!_dtxEnabled)
- {
- return 0;
- }
- else if(_encoderExist) // check if encoder exist
- {
- // disable DTX
- if(WebRtcSpeex_EncoderInit(_encoderInstPtr, (_vbrEnabled ? 1:0), _complMode, 0) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot disable DTX for Speex");
- return -1;
- }
- _dtxEnabled = false;
- return 0;
- }
- else
- {
- // encoder doesn't exists, therefore disabling is harmless
- return 0;
- }
-
- return 0;
-}
-
-WebRtc_Word16
-ACMSPEEX::InternalInitEncoder(
- WebRtcACMCodecParams* codecParams)
-{
- // sanity check
- if (_encoderInstPtr == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot initialize Speex encoder, instance does not exist");
- return -1;
- }
-
- WebRtc_Word16 status = SetBitRateSafe((codecParams->codecInstant).rate);
- status += (WebRtcSpeex_EncoderInit(_encoderInstPtr, _vbrEnabled, _complMode, ((codecParams->enableDTX)? 1:0)) < 0)? -1:0;
-
- if (status >= 0) {
- return 0;
- } else {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Error in initialization of Speex encoder");
- return -1;
- }
-}
-
-WebRtc_Word16
-ACMSPEEX::InternalInitDecoder(
- WebRtcACMCodecParams* /* codecParams */)
-{
- WebRtc_Word16 status;
-
- // sanity check
- if (_decoderInstPtr == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot initialize Speex decoder, instance does not exist");
- return -1;
- }
- status = ((WebRtcSpeex_DecoderInit(_decoderInstPtr) < 0)? -1:0);
-
- if (status >= 0) {
- return 0;
- } else {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Error in initialization of Speex decoder");
- return -1;
- }
-}
-
-WebRtc_Word32
-ACMSPEEX::CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst)
-{
- if (!_decoderInitialized)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Error, Speex decoder is not initialized");
- return -1;
- }
-
- // Fill up the structure by calling
- // "SET_CODEC_PAR" & "SET_SPEEX_FUNCTION."
- // Then call NetEQ to add the codec to it's
- // database.
-
- switch(_samplingFrequency)
- {
- case 8000:
- {
- SET_CODEC_PAR((codecDef), kDecoderSPEEX_8, codecInst.pltype,
- _decoderInstPtr, 8000);
- break;
- }
- case 16000:
- {
- SET_CODEC_PAR((codecDef), kDecoderSPEEX_16, codecInst.pltype,
- _decoderInstPtr, 16000);
- break;
- }
- default:
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Unsupported sampling frequency for Speex");
-
- return -1;
- }
- }
-
- SET_SPEEX_FUNCTIONS((codecDef));
- return 0;
-}
-
-ACMGenericCodec*
-ACMSPEEX::CreateInstance(void)
-{
- return NULL;
-}
-
-WebRtc_Word16
-ACMSPEEX::InternalCreateEncoder()
-{
- return WebRtcSpeex_CreateEnc(&_encoderInstPtr, _samplingFrequency);
-}
-
-void
-ACMSPEEX::DestructEncoderSafe()
-{
- if(_encoderInstPtr != NULL)
- {
- WebRtcSpeex_FreeEnc(_encoderInstPtr);
- _encoderInstPtr = NULL;
- }
- // there is no encoder set the following
- _encoderExist = false;
- _encoderInitialized = false;
- _encodingRate = 0;
-}
-
-
-WebRtc_Word16
-ACMSPEEX::InternalCreateDecoder()
-{
- return WebRtcSpeex_CreateDec(&_decoderInstPtr, _samplingFrequency, 1);
-}
-
-void
-ACMSPEEX::DestructDecoderSafe()
-{
- if(_decoderInstPtr != NULL)
- {
- WebRtcSpeex_FreeDec(_decoderInstPtr);
- _decoderInstPtr = NULL;
- }
- // there is no encoder instance set the followings
- _decoderExist = false;
- _decoderInitialized = false;
-}
-
-WebRtc_Word16
-ACMSPEEX::SetBitRateSafe(
- const WebRtc_Word32 rate)
-{
- // Check if changed rate
- if (rate == _encodingRate) {
- return 0;
- } else if (rate > 2000) {
- _encodingRate = rate;
- _encoderParams.codecInstant.rate = rate;
- } else {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Unsupported encoding rate for Speex");
-
- return -1;
- }
-
- return 0;
-}
-
-
-void
-ACMSPEEX::InternalDestructEncoderInst(
- void* ptrInst)
-{
- if(ptrInst != NULL)
- {
- WebRtcSpeex_FreeEnc((SPEEX_encinst_t_*)ptrInst);
- }
- return;
-}
-
-
-WebRtc_Word16
-ACMSPEEX::UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType)
-{
- if(payloadType != _decoderParams.codecInstant.pltype)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot unregister codec %s given payload-type %d does not match \
-the stored payload type",
- _decoderParams.codecInstant.plname,
- payloadType,
- _decoderParams.codecInstant.pltype);
- return -1;
- }
-
-
- switch(_samplingFrequency)
- {
- case 8000:
- {
- return netEq->RemoveCodec(kDecoderSPEEX_8);
- }
- case 16000:
- {
- return netEq->RemoveCodec(kDecoderSPEEX_16);
- }
- default:
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Could not unregister Speex from NetEQ. Sampling frequency doesn't match");
- return -1;
- }
- }
-}
-
-
-#ifdef UNUSEDSPEEX
-
-// This API is currently not in use. If requested to be able to enable/disable VBR
-// an ACM API need to be added.
-WebRtc_Word16
-ACMSPEEX::EnableVBR()
-{
- if(_vbrEnabled)
- {
- return 0;
- }
- else if(_encoderExist) // check if encoder exist
- {
- // enable Variable Bit Rate (VBR)
- if(WebRtcSpeex_EncoderInit(_encoderInstPtr, 1, _complMode, (_dtxEnabled? 1:0)) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot enable VBR mode for Speex");
-
- return -1;
- }
- _vbrEnabled = true;
- return 0;
- }
- else
- {
- return -1;
- }
-}
-
-
-// This API is currently not in use. If requested to be able to enable/disable VBR
-// an ACM API need to be added.
-WebRtc_Word16
-ACMSPEEX::DisableVBR()
-{
- if(!_vbrEnabled)
- {
- return 0;
- }
- else if(_encoderExist) // check if encoder exist
- {
- // disable DTX
- if(WebRtcSpeex_EncoderInit(_encoderInstPtr, 0, _complMode, (_dtxEnabled? 1:0)) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Cannot disable DTX for Speex");
-
- return -1;
- }
- _vbrEnabled = false;
- return 0;
- }
- else
- {
- // encoder doesn't exists, therefore disabling is harmless
- return 0;
- }
-}
-
-// This API is currently not in use. If requested to be able to set complexity
-// an ACM API need to be added.
-WebRtc_Word16
-ACMSPEEX::SetComplMode(
- WebRtc_Word16 mode)
-{
- // Check if new mode
- if(mode == _complMode)
- {
- return 0;
- }
- else if(_encoderExist) // check if encoder exist
- {
- // Set new mode
- if(WebRtcSpeex_EncoderInit(_encoderInstPtr, 0, mode, (_dtxEnabled? 1:0)) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _uniqueID,
- "Error in complexity mode for Speex");
- return -1;
- }
- _complMode = mode;
- return 0;
- }
- else
- {
- // encoder doesn't exists, therefore disabling is harmless
- return 0;
- }
-}
-
-#endif
-
-#endif
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/acm_speex.h b/trunk/src/modules/audio_coding/main/source/acm_speex.h
deleted file mode 100644
index 7988e45..0000000
--- a/trunk/src/modules/audio_coding/main/source/acm_speex.h
+++ /dev/null
@@ -1,94 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_
-
-#include "acm_generic_codec.h"
-
-// forward declaration
-struct SPEEX_encinst_t_;
-struct SPEEX_decinst_t_;
-
-namespace webrtc {
-
-class ACMSPEEX : public ACMGenericCodec
-{
-public:
- ACMSPEEX(WebRtc_Word16 codecID);
- ~ACMSPEEX();
- // for FEC
- ACMGenericCodec* CreateInstance(void);
-
- WebRtc_Word16 InternalEncode(
- WebRtc_UWord8* bitstream,
- WebRtc_Word16* bitStreamLenByte);
-
- WebRtc_Word16 InternalInitEncoder(
- WebRtcACMCodecParams *codecParams);
-
- WebRtc_Word16 InternalInitDecoder(
- WebRtcACMCodecParams *codecParams);
-
-protected:
- WebRtc_Word16 DecodeSafe(
- WebRtc_UWord8* bitStream,
- WebRtc_Word16 bitStreamLenByte,
- WebRtc_Word16* audio,
- WebRtc_Word16* audioSamples,
- WebRtc_Word8* speechType);
-
- WebRtc_Word32 CodecDef(
- WebRtcNetEQ_CodecDef& codecDef,
- const CodecInst& codecInst);
-
- void DestructEncoderSafe();
-
- void DestructDecoderSafe();
-
- WebRtc_Word16 InternalCreateEncoder();
-
- WebRtc_Word16 InternalCreateDecoder();
-
- void InternalDestructEncoderInst(
- void* ptrInst);
-
- WebRtc_Word16 SetBitRateSafe(
- const WebRtc_Word32 rate);
-
- WebRtc_Word16 EnableDTX();
-
- WebRtc_Word16 DisableDTX();
-
-#ifdef UNUSEDSPEEX
- WebRtc_Word16 EnableVBR();
-
- WebRtc_Word16 DisableVBR();
-
- WebRtc_Word16 SetComplMode(
- WebRtc_Word16 mode);
-#endif
-
- WebRtc_Word16 UnregisterFromNetEqSafe(
- ACMNetEQ* netEq,
- WebRtc_Word16 payloadType);
-
- SPEEX_encinst_t_* _encoderInstPtr;
- SPEEX_decinst_t_* _decoderInstPtr;
- WebRtc_Word16 _complMode;
- bool _vbrEnabled;
- WebRtc_Word32 _encodingRate;
- WebRtc_Word16 _samplingFrequency;
- WebRtc_UWord16 _samplesIn20MsAudio;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_ACM_SPEEX_H_
diff --git a/trunk/src/modules/audio_coding/main/source/audio_coding_module.cc b/trunk/src/modules/audio_coding/main/source/audio_coding_module.cc
deleted file mode 100644
index 2cd959d..0000000
--- a/trunk/src/modules/audio_coding/main/source/audio_coding_module.cc
+++ /dev/null
@@ -1,136 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-#include "acm_dtmf_detection.h"
-#include "audio_coding_module.h"
-#include "audio_coding_module_impl.h"
-#include "trace.h"
-
-namespace webrtc
-{
-
-// Create module
-AudioCodingModule*
-AudioCodingModule::Create(
- const WebRtc_Word32 id)
-{
- return new AudioCodingModuleImpl(id);
-}
-
-// Destroy module
-void
-AudioCodingModule::Destroy(
- AudioCodingModule* module)
-{
- delete static_cast<AudioCodingModuleImpl*> (module);
-}
-
-// Get number of supported codecs
-WebRtc_UWord8 AudioCodingModule::NumberOfCodecs()
-{
- return static_cast<WebRtc_UWord8>(ACMCodecDB::kNumCodecs);
-}
-
-// Get supported codec param with id
-WebRtc_Word32
-AudioCodingModule::Codec(
- const WebRtc_UWord8 listId,
- CodecInst& codec)
-{
- // Get the codec settings for the codec with the given list ID
- return ACMCodecDB::Codec(listId, &codec);
-}
-
-// Get supported codec Param with name
-WebRtc_Word32
-AudioCodingModule::Codec(
- const char* payloadName,
- CodecInst& codec,
- const WebRtc_Word32 samplingFreqHz)
-{
- // Search through codec list for a matching name
- for(int codecCntr = 0; codecCntr < ACMCodecDB::kNumCodecs; codecCntr++)
- {
- // Store codec settings for codec number "codeCntr" in the output struct
- ACMCodecDB::Codec(codecCntr, &codec);
-
- if(!STR_CASE_CMP(codec.plname, payloadName))
- {
- // If samplingFreqHz is set (!= -1), check if frequency matches
- if((samplingFreqHz == codec.plfreq) || (samplingFreqHz == -1))
- {
- // We found a match, return OK
- return 0;
- }
- }
- }
-
- // if we are here we couldn't find anything
- // set the params to unacceptable values
- codec.plname[0] = '\0';
- codec.pltype = -1;
- codec.pacsize = 0;
- codec.rate = 0;
- codec.plfreq = 0;
- return -1;
-}
-
-// Get supported codec Index with name, and frequency if needed
-WebRtc_Word32
-AudioCodingModule::Codec(
- const char* payloadName,
- const WebRtc_Word32 samplingFreqHz)
-{
- CodecInst codec;
-
- // Search through codec list for a matching name
- for(int codecCntr = 0; codecCntr < ACMCodecDB::kNumCodecs; codecCntr++)
- {
- // Temporally store codec settings for codec number "codeCntr" in "codec"
- ACMCodecDB::Codec(codecCntr, &codec);
-
- if(!STR_CASE_CMP(codec.plname, payloadName))
- {
- // If samplingFreqHz is set (!= -1), check if frequency matches
- if((samplingFreqHz == codec.plfreq) || (samplingFreqHz == -1))
- {
- // We found a match, return codec list number (index)
- return codecCntr;
- }
- }
- }
-
- // We did not find a matching codec in the list
- return -1;
-}
-
-// Checks the validity of the parameters of the given codec
-bool
-AudioCodingModule::IsCodecValid(
- const CodecInst& codec)
-{
- int mirrorID;
- char errMsg[500];
-
- int codecNumber = ACMCodecDB::CodecNumber(&codec, &mirrorID, errMsg, 500);
-
- if(codecNumber < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, -1, errMsg);
- return false;
- }
- else
- {
- return true;
- }
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/audio_coding_module.gypi b/trunk/src/modules/audio_coding/main/source/audio_coding_module.gypi
deleted file mode 100644
index 80114e0..0000000
--- a/trunk/src/modules/audio_coding/main/source/audio_coding_module.gypi
+++ /dev/null
@@ -1,152 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'audio_coding_module',
- 'type': '<(library)',
- 'dependencies': [
- 'CNG',
- 'G711',
- 'G722',
- 'iLBC',
- 'iSAC',
- 'iSACFix',
- 'PCM16B',
- 'NetEq',
- '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- '<(webrtc_root)/common_audio/common_audio.gyp:vad',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '../interface',
- '../../../interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- '../../../interface',
- ],
- },
- 'sources': [
- '../interface/audio_coding_module.h',
- '../interface/audio_coding_module_typedefs.h',
- 'acm_amr.cc',
- 'acm_amr.h',
- 'acm_amrwb.cc',
- 'acm_amrwb.h',
- 'acm_celt.cc',
- 'acm_celt.h',
- 'acm_cng.cc',
- 'acm_cng.h',
- 'acm_codec_database.cc',
- 'acm_codec_database.h',
- 'acm_dtmf_detection.cc',
- 'acm_dtmf_detection.h',
- 'acm_dtmf_playout.cc',
- 'acm_dtmf_playout.h',
- 'acm_g722.cc',
- 'acm_g722.h',
- 'acm_g7221.cc',
- 'acm_g7221.h',
- 'acm_g7221c.cc',
- 'acm_g7221c.h',
- 'acm_g729.cc',
- 'acm_g729.h',
- 'acm_g7291.cc',
- 'acm_g7291.h',
- 'acm_generic_codec.cc',
- 'acm_generic_codec.h',
- 'acm_gsmfr.cc',
- 'acm_gsmfr.h',
- 'acm_ilbc.cc',
- 'acm_ilbc.h',
- 'acm_isac.cc',
- 'acm_isac.h',
- 'acm_isac_macros.h',
- 'acm_neteq.cc',
- 'acm_neteq.h',
- 'acm_opus.cc',
- 'acm_opus.h',
- 'acm_speex.cc',
- 'acm_speex.h',
- 'acm_pcm16b.cc',
- 'acm_pcm16b.h',
- 'acm_pcma.cc',
- 'acm_pcma.h',
- 'acm_pcmu.cc',
- 'acm_pcmu.h',
- 'acm_red.cc',
- 'acm_red.h',
- 'acm_resampler.cc',
- 'acm_resampler.h',
- 'audio_coding_module.cc',
- 'audio_coding_module_impl.cc',
- 'audio_coding_module_impl.h',
- ],
- },
- ],
- # Exclude the test targets when building with chromium.
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'audio_coding_module_test',
- 'type': 'executable',
- 'dependencies': [
- 'audio_coding_module',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'sources': [
- '../test/ACMTest.cc',
- '../test/APITest.cc',
- '../test/Channel.cc',
- '../test/EncodeDecodeTest.cc',
- '../test/iSACTest.cc',
- '../test/PCMFile.cc',
- '../test/RTPFile.cc',
- '../test/SpatialAudio.cc',
- '../test/TestAllCodecs.cc',
- '../test/Tester.cc',
- '../test/TestFEC.cc',
- '../test/TestStereo.cc',
- '../test/TestVADDTX.cc',
- '../test/TimedTrace.cc',
- '../test/TwoWayCommunication.cc',
- '../test/utility.cc',
- ],
- },
- {
- 'target_name': 'audio_coding_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'audio_coding_module',
- 'NetEq',
- '<(webrtc_root)/common_audio/common_audio.gyp:vad',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'sources': [
- 'acm_neteq_unittest.cc',
- ],
- }, # audio_coding_unittests
- ],
- }],
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/main/source/audio_coding_module_impl.cc b/trunk/src/modules/audio_coding/main/source/audio_coding_module_impl.cc
deleted file mode 100644
index 18b6f4f..0000000
--- a/trunk/src/modules/audio_coding/main/source/audio_coding_module_impl.cc
+++ /dev/null
@@ -1,2624 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "acm_codec_database.h"
-#include "acm_common_defs.h"
-#include "acm_dtmf_detection.h"
-#include "acm_generic_codec.h"
-#include "acm_resampler.h"
-#include "audio_coding_module_impl.h"
-#include "critical_section_wrapper.h"
-#include "engine_configurations.h"
-#include "rw_lock_wrapper.h"
-#include "trace.h"
-
-#include <assert.h>
-#include <stdlib.h>
-
-#ifdef ACM_QA_TEST
-# include <stdio.h>
-#endif
-
-#ifdef TIMED_LOGGING
- char message[500];
- #include "../test/timedtrace.h"
- #include <string.h>
- #define LOGWITHTIME(logString) \
- sprintf(message, logString, _id); \
- _trace.TimedLogg(message);
-#else
- #define LOGWITHTIME(logString)
-#endif
-
-namespace webrtc
-{
-
-enum {
- kACMToneEnd = 999
-};
-
-AudioCodingModuleImpl::AudioCodingModuleImpl(
- const WebRtc_Word32 id):
- _packetizationCallback(NULL),
- _id(id),
- _lastTimestamp(0),
- _lastInTimestamp(0),
- _vadEnabled(false),
- _dtxEnabled(false),
- _vadMode(VADNormal),
- _stereoSend(false),
- _prev_received_channel(0),
- _expected_channels(1),
- _currentSendCodecIdx(-1), // invalid value
- _sendCodecRegistered(false),
- _acmCritSect(CriticalSectionWrapper::CreateCriticalSection()),
- _vadCallback(NULL),
- _lastRecvAudioCodecPlType(255),
- _isFirstRED(true),
- _fecEnabled(false),
- _fragmentation(NULL),
- _lastFECTimestamp(0),
- _redPayloadType(255),
- _receiveREDPayloadType(255), // invalid value
- _previousPayloadType(255),
- _dummyRTPHeader(NULL),
- _recvPlFrameSizeSmpls(0),
- _receiverInitialized(false),
- _dtmfDetector(NULL),
- _dtmfCallback(NULL),
- _lastDetectedTone(kACMToneEnd),
- _callbackCritSect(CriticalSectionWrapper::CreateCriticalSection())
-{
- _lastTimestamp = 0xD87F3F9F;
- _lastInTimestamp = 0xD87F3F9F;
-
- // Nullify send codec memory, set payload type and set codec name to
- // invalid values.
- memset(&_sendCodecInst, 0, sizeof(CodecInst));
- strncpy(_sendCodecInst.plname, "noCodecRegistered", 31);
- _sendCodecInst.pltype = -1;
-
- // Nullify memory for CNG, DTMF and RED.
- memset(&_cngNB, 0, sizeof(CodecInst));
- memset(&_cngWB, 0, sizeof(CodecInst));
- memset(&_cngSWB, 0, sizeof(CodecInst));
- memset(&_RED, 0, sizeof(CodecInst));
- memset(&_DTMF, 0, sizeof(CodecInst));
-
- for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++)
- {
- _codecs[i] = NULL;
- _registeredPlTypes[i] = -1;
- _stereoReceive[i] = false;
- _slaveCodecs[i] = NULL;
- _mirrorCodecIdx[i] = -1;
- }
-
- _netEq.SetUniqueId(_id);
-
- // Allocate memory for RED
- _redBuffer = new WebRtc_UWord8[MAX_PAYLOAD_SIZE_BYTE];
- _fragmentation = new RTPFragmentationHeader;
- _fragmentation->fragmentationVectorSize = 2;
- _fragmentation->fragmentationOffset = new WebRtc_UWord32[2];
- _fragmentation->fragmentationLength = new WebRtc_UWord32[2];
- _fragmentation->fragmentationTimeDiff = new WebRtc_UWord16[2];
- _fragmentation->fragmentationPlType = new WebRtc_UWord8[2];
-
- // Register the default payload type for RED and for
- // CNG for the three frequencies 8, 16 and 32 kHz
- for (int i = (ACMCodecDB::kNumCodecs - 1); i>=0; i--)
- {
- if((STR_CASE_CMP(ACMCodecDB::database_[i].plname, "red") == 0))
- {
- _redPayloadType = ACMCodecDB::database_[i].pltype;
- }
- else if ((STR_CASE_CMP(ACMCodecDB::database_[i].plname, "CN") == 0))
- {
- if (ACMCodecDB::database_[i].plfreq == 8000)
- {
- memcpy(&_cngNB, &ACMCodecDB::database_[i], sizeof(_cngNB));
- }
- else if (ACMCodecDB::database_[i].plfreq == 16000)
- {
- memcpy(&_cngWB, &ACMCodecDB::database_[i], sizeof(_cngWB));
- } else if (ACMCodecDB::database_[i].plfreq == 32000)
- {
- memcpy(&_cngSWB, &ACMCodecDB::database_[i], sizeof(_cngSWB));
- }
- }
- }
-
- if(InitializeReceiverSafe() < 0 )
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Cannot initialize reciever");
- }
-#ifdef TIMED_LOGGING
- _trace.SetUp("TimedLogg.txt");
-#endif
-
-#ifdef ACM_QA_TEST
- char fileName[500];
- sprintf(fileName, "ACM_QA_incomingPL_%03d_%d%d%d%d%d%d.dat",
- _id,
- rand() % 10,
- rand() % 10,
- rand() % 10,
- rand() % 10,
- rand() % 10,
- rand() % 10);
-
- _incomingPL = fopen(fileName, "wb");
-
- sprintf(fileName, "ACM_QA_outgoingPL_%03d_%d%d%d%d%d%d.dat",
- _id,
- rand() % 10,
- rand() % 10,
- rand() % 10,
- rand() % 10,
- rand() % 10,
- rand() % 10);
- _outgoingPL = fopen(fileName, "wb");
-#endif
-
- WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, id, "Created");
-}
-
-AudioCodingModuleImpl::~AudioCodingModuleImpl()
-{
- {
- CriticalSectionScoped lock(*_acmCritSect);
- _currentSendCodecIdx = -1;
-
- for (int i=0; i < ACMCodecDB::kMaxNumCodecs; i++)
- {
- if (_codecs[i] != NULL)
- {
- assert(_mirrorCodecIdx[i] > -1);
- if(_codecs[_mirrorCodecIdx[i]] != NULL)
- {
- delete _codecs[_mirrorCodecIdx[i]];
- _codecs[_mirrorCodecIdx[i]] = NULL;
- }
- _codecs[i] = NULL;
- }
-
- if(_slaveCodecs[i] != NULL)
- {
- assert(_mirrorCodecIdx[i] > -1);
- if(_slaveCodecs[_mirrorCodecIdx[i]] != NULL)
- {
- delete _slaveCodecs[_mirrorCodecIdx[i]];
- _slaveCodecs[_mirrorCodecIdx[i]] = NULL;
- }
- _slaveCodecs[i] = NULL;
- }
- }
-
- if(_dtmfDetector != NULL)
- {
- delete _dtmfDetector;
- _dtmfDetector = NULL;
- }
- if(_dummyRTPHeader != NULL)
- {
- delete _dummyRTPHeader;
- _dummyRTPHeader = NULL;
- }
- if(_redBuffer != NULL)
- {
- delete [] _redBuffer;
- _redBuffer = NULL;
- }
- if(_fragmentation != NULL)
- {
- // Only need to delete fragmentation header, it will clean
- // up it's own memory
- delete _fragmentation;
- _fragmentation = NULL;
- }
- }
-
-#ifdef ACM_QA_TEST
- if(_incomingPL != NULL)
- {
- fclose(_incomingPL);
- }
-
- if(_outgoingPL != NULL)
- {
- fclose(_outgoingPL);
- }
-#endif
-
- delete _callbackCritSect;
- _callbackCritSect = NULL;
-
- delete _acmCritSect;
- _acmCritSect = NULL;
- WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceAudioCoding, _id, "Destroyed");
-}
-
-WebRtc_Word32
-AudioCodingModuleImpl::ChangeUniqueId(
- const WebRtc_Word32 id)
-{
- {
- CriticalSectionScoped lock(*_acmCritSect);
- _id = id;
-#ifdef ACM_QA_TEST
- if(_incomingPL != NULL)
- {
- fclose(_incomingPL);
- }
-
- if(_outgoingPL != NULL)
- {
- fclose(_outgoingPL);
- }
-
- char fileName[500];
- sprintf(fileName, "ACM_QA_incomingPL_%03d_%d%d%d%d%d%d.dat",
- _id,
- rand() % 10,
- rand() % 10,
- rand() % 10,
- rand() % 10,
- rand() % 10,
- rand() % 10);
-
- _incomingPL = fopen(fileName, "wb");
-
- sprintf(fileName, "ACM_QA_outgoingPL_%03d_%d%d%d%d%d%d.dat",
- _id,
- rand() % 10,
- rand() % 10,
- rand() % 10,
- rand() % 10,
- rand() % 10,
- rand() % 10);
- _outgoingPL = fopen(fileName, "wb");
-#endif
-
- for (int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++)
- {
- if(_codecs[i] != NULL)
- {
- _codecs[i]->SetUniqueID(id);
- }
- }
- }
-
- _netEq.SetUniqueId(_id);
- return 0;
-}
-
-// returns the number of milliseconds until the module want a
-// worker thread to call Process
-WebRtc_Word32
-AudioCodingModuleImpl::TimeUntilNextProcess()
-{
- CriticalSectionScoped lock(*_acmCritSect);
-
- if(!HaveValidEncoder("TimeUntilNextProcess"))
- {
- return -1;
- }
- return _codecs[_currentSendCodecIdx]->SamplesLeftToEncode() /
- (_sendCodecInst.plfreq / 1000);
-}
-
-// Process any pending tasks such as timeouts
-WebRtc_Word32
-AudioCodingModuleImpl::Process()
-{
- WebRtc_UWord8 bitStream[2 * MAX_PAYLOAD_SIZE_BYTE]; // Make room for 1 RED payload
- WebRtc_Word16 lengthBytes = 2 * MAX_PAYLOAD_SIZE_BYTE;
- WebRtc_Word16 redLengthBytes = lengthBytes;
- WebRtc_UWord32 rtpTimestamp;
- WebRtc_Word16 status;
- WebRtcACMEncodingType encodingType;
- FrameType frameType = kAudioFrameSpeech;
- WebRtc_UWord8 currentPayloadType = 0;
- bool hasDataToSend = false;
- bool fecActive = false;
-
- // keep the scope of the ACM critical section limited
- {
- CriticalSectionScoped lock(*_acmCritSect);
- if(!HaveValidEncoder("Process"))
- {
- return -1;
- }
-
- status = _codecs[_currentSendCodecIdx]->Encode(bitStream, &lengthBytes,
- &rtpTimestamp, &encodingType);
- if (status < 0) // Encode failed
- {
- // logging error
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Process(): Encoding Failed");
- lengthBytes = 0;
- return -1;
- }
- else if(status == 0)
- {
- // Not enough data
- return 0;
- }
- else
- {
- switch(encodingType)
- {
- case kNoEncoding:
- {
- currentPayloadType = _previousPayloadType;
- frameType = kFrameEmpty;
- lengthBytes = 0;
- break;
- }
- case kActiveNormalEncoded:
- case kPassiveNormalEncoded:
- {
- currentPayloadType = (WebRtc_UWord8)_sendCodecInst.pltype;
- frameType = kAudioFrameSpeech;
- break;
- }
- case kPassiveDTXNB:
- {
- currentPayloadType = (WebRtc_UWord8)_cngNB.pltype;
- frameType = kAudioFrameCN;
- _isFirstRED = true;
- break;
- }
- case kPassiveDTXWB:
- {
- currentPayloadType = (WebRtc_UWord8)_cngWB.pltype;
- frameType = kAudioFrameCN;
- _isFirstRED = true;
- break;
- }
- case kPassiveDTXSWB:
- {
- currentPayloadType = (WebRtc_UWord8)_cngSWB.pltype;
- frameType = kAudioFrameCN;
- _isFirstRED = true;
- break;
- }
- }
- hasDataToSend = true;
- _previousPayloadType = currentPayloadType;
-
- // Redundancy encode is done here,
- // the two bitstreams packetized into
- // one RTP packet and the fragmentation points
- // are set.
- // Only apply RED on speech data.
- if((_fecEnabled) &&
- ((encodingType == kActiveNormalEncoded) ||
- (encodingType == kPassiveNormalEncoded)))
- {
- // FEC is enabled within this scope.
- //
- // Note that, a special solution exists for iSAC since it is the only codec for
- // which getRedPayload has a non-empty implementation.
- //
- // Summary of the FEC scheme below (use iSAC as example):
- //
- // 1st (_firstRED is true) encoded iSAC frame (primary #1) =>
- // - call getRedPayload() and store redundancy for packet #1 in second
- // fragment of RED buffer (old data)
- // - drop the primary iSAC frame
- // - don't call SendData
- // 2nd (_firstRED is false) encoded iSAC frame (primary #2) =>
- // - store primary #2 in 1st fragment of RED buffer and send the combined
- // packet
- // - the transmitted packet contains primary #2 (new) and reduncancy for
- // packet #1 (old)
- // - call getRedPayload() and store redundancy for packet #2 in second
- // fragment of RED buffer
- //
- // ...
- //
- // Nth encoded iSAC frame (primary #N) =>
- // - store primary #N in 1st fragment of RED buffer and send the combined
- // packet
- // - the transmitted packet contains primary #N (new) and reduncancy for
- // packet #(N-1) (old)
- // - call getRedPayload() and store redundancy for packet #N in second
- // fragment of RED buffer
- //
- // For all other codecs, getRedPayload does nothing and returns -1 =>
- // redundant data is only a copy.
- //
- // First combined packet contains : #2 (new) and #1 (old)
- // Second combined packet contains: #3 (new) and #2 (old)
- // Third combined packet contains : #4 (new) and #3 (old)
- //
- // Hence, even if every second packet is dropped, perfect reconstruction is
- // possible.
- fecActive = true;
-
- hasDataToSend = false;
- if(!_isFirstRED) // skip this part for the first packet in a RED session
- {
- // Rearrange bitStream such that FEC packets are included.
- // Replace bitStream now that we have stored current bitStream.
- memcpy(bitStream + _fragmentation->fragmentationOffset[1], _redBuffer,
- _fragmentation->fragmentationLength[1]);
- // Update the fragmentation time difference vector
- WebRtc_UWord16 timeSinceLastTimestamp =
- WebRtc_UWord16(rtpTimestamp - _lastFECTimestamp);
-
- // Update fragmentation vectors
- _fragmentation->fragmentationPlType[1] =
- _fragmentation->fragmentationPlType[0];
- _fragmentation->fragmentationTimeDiff[1] = timeSinceLastTimestamp;
- hasDataToSend = true;
- }
-
- // Insert new packet length.
- _fragmentation->fragmentationLength[0] = lengthBytes;
-
- // Insert new packet payload type.
- _fragmentation->fragmentationPlType[0] = currentPayloadType;
- _lastFECTimestamp = rtpTimestamp;
-
- // can be modified by the GetRedPayload() call if iSAC is utilized
- redLengthBytes = lengthBytes;
- // A fragmentation header is provided => packetization according to RFC 2198
- // (RTP Payload for Redundant Audio Data) will be used.
- // First fragment is the current data (new).
- // Second fragment is the previous data (old).
- lengthBytes =
- static_cast<WebRtc_Word16> (_fragmentation->fragmentationLength[0] +
- _fragmentation->fragmentationLength[1]);
-
- // Get, and store, redundant data from the encoder based on the recently
- // encoded frame.
- // NOTE - only iSAC contains an implementation; all other codecs does nothing
- // and returns -1.
- if (_codecs[_currentSendCodecIdx]->GetRedPayload(_redBuffer,
- &redLengthBytes) == -1)
- {
- // The codec was not iSAC => use current encoder output as redundant data
- // instead (trivial FEC scheme)
- memcpy(_redBuffer, bitStream, redLengthBytes);
- }
-
- _isFirstRED = false;
- // Update payload type with RED payload type
- currentPayloadType = _redPayloadType;
- }
- }
- }
-
- if(hasDataToSend)
- {
- CriticalSectionScoped lock(*_callbackCritSect);
-#ifdef ACM_QA_TEST
- if(_outgoingPL != NULL)
- {
- fwrite(&rtpTimestamp, sizeof(WebRtc_UWord32), 1, _outgoingPL);
- fwrite(¤tPayloadType, sizeof(WebRtc_UWord8), 1, _outgoingPL);
- fwrite(&lengthBytes, sizeof(WebRtc_Word16), 1, _outgoingPL);
- }
-#endif
-
- if(_packetizationCallback != NULL)
- {
- if (fecActive) {
- _packetizationCallback->SendData(frameType, currentPayloadType,
- rtpTimestamp, bitStream, lengthBytes, _fragmentation);
- } else {
- _packetizationCallback->SendData(frameType, currentPayloadType,
- rtpTimestamp, bitStream, lengthBytes, NULL);
- }
- }
-
- // This is for test
- if(_vadCallback != NULL)
- {
- _vadCallback->InFrameType(((WebRtc_Word16)encodingType));
- }
- }
- if (fecActive) {
- _fragmentation->fragmentationLength[1] = redLengthBytes;
- }
- return lengthBytes;
-}
-
-
-
-
-/////////////////////////////////////////
-// Sender
-//
-
-// Initialize send codec
-WebRtc_Word32
-AudioCodingModuleImpl::InitializeSender()
-{
- CriticalSectionScoped lock(*_acmCritSect);
-
- _sendCodecRegistered = false;
- _currentSendCodecIdx = -1; // invalid value
-
- _sendCodecInst.plname[0] = '\0';
-
- for(int codecCntr = 0; codecCntr < ACMCodecDB::kMaxNumCodecs; codecCntr++)
- {
- if(_codecs[codecCntr] != NULL)
- {
- _codecs[codecCntr]->DestructEncoder();
- }
- }
- // Initialize FEC/RED
- _isFirstRED = true;
- if(_fecEnabled)
- {
- if(_redBuffer != NULL)
- {
- memset(_redBuffer, 0, MAX_PAYLOAD_SIZE_BYTE);
- }
- if(_fragmentation != NULL)
- {
- _fragmentation->fragmentationVectorSize = 2;
- _fragmentation->fragmentationOffset[0] = 0;
- _fragmentation->fragmentationOffset[0] = MAX_PAYLOAD_SIZE_BYTE;
- memset(_fragmentation->fragmentationLength, 0, sizeof(WebRtc_UWord32) * 2);
- memset(_fragmentation->fragmentationTimeDiff, 0, sizeof(WebRtc_UWord16) * 2);
- memset(_fragmentation->fragmentationPlType, 0, sizeof(WebRtc_UWord8) * 2);
- }
- }
-
- return 0;
-}
-
-WebRtc_Word32
-AudioCodingModuleImpl::ResetEncoder()
-{
- CriticalSectionScoped lock(*_acmCritSect);
- if(!HaveValidEncoder("ResetEncoder"))
- {
- return -1;
- }
- return _codecs[_currentSendCodecIdx]->ResetEncoder();
-}
-
-void
-AudioCodingModuleImpl::UnregisterSendCodec()
-{
- CriticalSectionScoped lock(*_acmCritSect);
- _sendCodecRegistered = false;
- _currentSendCodecIdx = -1; // invalid value
-
- return;
-}
-
-ACMGenericCodec*
-AudioCodingModuleImpl::CreateCodec(
- const CodecInst& codec)
-{
- ACMGenericCodec* myCodec = NULL;
-
- myCodec = ACMCodecDB::CreateCodecInstance(&codec);
- if(myCodec == NULL)
- {
- // Error, could not create the codec
-
- // logging error
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "ACMCodecDB::CreateCodecInstance() failed in \
-CreateCodec()");
- return myCodec;
- }
- myCodec->SetUniqueID(_id);
- myCodec->SetNetEqDecodeLock(_netEq.DecodeLock());
-
- return myCodec;
-}
-
-// can be called multiple times for Codec, CNG, RED
-WebRtc_Word32
-AudioCodingModuleImpl::RegisterSendCodec(
- const CodecInst& sendCodec)
-{
- if((sendCodec.channels != 1) && (sendCodec.channels != 2))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Registering Send codec failed due to wrong number of channels, %d. Only\
-mono codecs are supported, i.e. channels=1.", sendCodec.channels);
- return -1;
- }
-
- char errMsg[500];
- int mirrorId;
- int codecID = ACMCodecDB::CodecNumber(&sendCodec, &mirrorId, errMsg,
- sizeof(errMsg));
- CriticalSectionScoped lock(*_acmCritSect);
-
- // Check for reported errors from function CodecNumber()
- if(codecID < 0)
- {
- if(!_sendCodecRegistered)
- {
- // This values has to be NULL if there is no codec registered
- _currentSendCodecIdx = -1; // invalid value
- }
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id, errMsg);
- // Failed to register Send Codec
- return -1;
- }
-
- // telephone-event cannot be a send codec
- if(!STR_CASE_CMP(sendCodec.plname, "telephone-event"))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "telephone-event cannot be registered as send codec");
- return -1;
- }
-
- // RED can be registered with other payload type. If not registered a default
- // payload type is used.
- if(!STR_CASE_CMP(sendCodec.plname, "red"))
- {
- // Check if the payload-type is valid
- if(!ACMCodecDB::ValidPayloadType(sendCodec.pltype))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Invalid payload-type %d for %s.",
- sendCodec.pltype, sendCodec.plname);
- return -1;
- }
- // Set RED payload type
- _redPayloadType = (WebRtc_UWord8)sendCodec.pltype;
- return 0;
- }
-
- // CNG can be registered with other payload type. If not registered the
- // default payload types will be used: CNNB=13 (fixed), CNWB=97, CNSWB=98
- if(!STR_CASE_CMP(sendCodec.plname, "CN"))
- {
- // CNG is registered
- switch(sendCodec.plfreq)
- {
- case 8000:
- {
- memcpy(&_cngNB, &sendCodec, sizeof(_cngNB));
- break;
- }
- case 16000:
- {
- memcpy(&_cngWB, &sendCodec, sizeof(_cngWB));
- break;
- }
- case 32000:
- {
- memcpy(&_cngSWB, &sendCodec, sizeof(_cngSWB));
- break;
- }
- default :
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "RegisterSendCodec() failed, invalid frequency for CNG registeration");
- return -1;
- }
- }
-
- return 0;
- }
-
- // Check if the payload-type is valid
- if(!ACMCodecDB::ValidPayloadType(sendCodec.pltype))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Invalid payload-type %d for %s.",
- sendCodec.pltype, sendCodec.plname);
- return -1;
- }
-
- // Check if codec supports the number of channels
- if(ACMCodecDB::codec_settings_[codecID].channel_support < sendCodec.channels)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "%d number of channels not supportedn for %s.",
- sendCodec.channels, sendCodec.plname);
- return -1;
- }
-
- // Set Stereo
- if (sendCodec.channels == 2)
- {
- _stereoSend = true;
- }
-
- // check if the codec is already registered as send codec
- bool oldCodecFamily;
- if(_sendCodecRegistered)
- {
- int sendCodecMirrorID;
- int sendCodecID =
- ACMCodecDB::CodecNumber(&_sendCodecInst, &sendCodecMirrorID);
- assert(sendCodecID >= 0);
- oldCodecFamily = (sendCodecID == codecID) || (mirrorId == sendCodecMirrorID);
- }
- else
- {
- oldCodecFamily = false;
- }
-
- // If new codec, register
- if (!oldCodecFamily)
- {
- if(_codecs[mirrorId] == NULL)
- {
-
- _codecs[mirrorId] = CreateCodec(sendCodec);
- if(_codecs[mirrorId] == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Cannot Create the codec");
- return -1;
- }
- _mirrorCodecIdx[mirrorId] = mirrorId;
- }
-
- if(mirrorId != codecID)
- {
- _codecs[codecID] = _codecs[mirrorId];
- _mirrorCodecIdx[codecID] = mirrorId;
- }
-
- ACMGenericCodec* tmpCodecPtr = _codecs[codecID];
- WebRtc_Word16 status;
- WebRtcACMCodecParams codecParams;
-
- memcpy(&(codecParams.codecInstant), &sendCodec,
- sizeof(CodecInst));
- codecParams.enableVAD = _vadEnabled;
- codecParams.enableDTX = _dtxEnabled;
- codecParams.vadMode = _vadMode;
- // force initialization
- status = tmpCodecPtr->InitEncoder(&codecParams, true);
-
- // Check if VAD was turned on, or if error is reported
- if (status == 1) {
- _vadEnabled = true;
- } else if (status < 0)
- {
- // could not initialize the encoder
-
- // Check if already have a registered codec
- // Depending on that different messages are logged
- if(!_sendCodecRegistered)
- {
- _currentSendCodecIdx = -1; // invalid value
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Cannot Initialize the encoder No Encoder is registered");
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Cannot Initialize the encoder, continue encoding \
-with the previously registered codec");
- }
- return -1;
- }
-
- // Everything is fine so we can replace the previous codec
- // with this one
- if(_sendCodecRegistered)
- {
- // If we change codec we start fresh with FEC.
- // This is not strictly required by the standard.
- _isFirstRED = true;
-
- if(tmpCodecPtr->SetVAD(_dtxEnabled, _vadEnabled, _vadMode) < 0){
- // SetVAD failed
- _vadEnabled = false;
- _dtxEnabled = false;
- }
-
- }
-
- _currentSendCodecIdx = codecID;
- _sendCodecRegistered = true;
- memcpy(&_sendCodecInst, &sendCodec, sizeof(CodecInst));
- _previousPayloadType = _sendCodecInst.pltype;
- return 0;
- }
- else
- {
- // If codec is the same as already registers check if any parameters
- // has changed compared to the current values.
- // If any parameter is valid then apply it and record.
- bool forceInit = false;
-
- if(mirrorId != codecID)
- {
- _codecs[codecID] = _codecs[mirrorId];
- _mirrorCodecIdx[codecID] = mirrorId;
- }
-
- // check the payload-type
- if(sendCodec.pltype != _sendCodecInst.pltype)
- {
- // At this point check if the given payload type is valid.
- // Record it later when the sampling frequency is changed
- // successfully.
- if(!ACMCodecDB::ValidPayloadType(sendCodec.pltype))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Out of range payload type");
- return -1;
- }
-
- }
-
- // If there is a codec that ONE instance of codec supports multiple
- // sampling frequencies, then we need to take care of it here.
- // one such a codec is iSAC. Both WB and SWB are encoded and decoded
- // with one iSAC instance. Therefore, we need to update the encoder
- // frequency if required.
- if(_sendCodecInst.plfreq != sendCodec.plfreq)
- {
- forceInit = true;
-
- // if sampling frequency is changed we have to start fresh with RED.
- _isFirstRED = true;
- }
-
- // If packet size or number of channels has changed, we need to
- // re-initialize the encoder.
- if(_sendCodecInst.pacsize != sendCodec.pacsize)
- {
- forceInit = true;
- }
- if(_sendCodecInst.channels != sendCodec.channels)
- {
- forceInit = true;
- }
-
- if(forceInit)
- {
- WebRtcACMCodecParams codecParams;
-
- memcpy(&(codecParams.codecInstant), &sendCodec,
- sizeof(CodecInst));
- codecParams.enableVAD = _vadEnabled;
- codecParams.enableDTX = _dtxEnabled;
- codecParams.vadMode = _vadMode;
-
- // force initialization
- if(_codecs[_currentSendCodecIdx]->InitEncoder(&codecParams, true) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Could not change the codec packet-size.");
- return -1;
- }
-
- _sendCodecInst.plfreq = sendCodec.plfreq;
- _sendCodecInst.pacsize = sendCodec.pacsize;
- _sendCodecInst.channels = sendCodec.channels;
- }
-
- // If the change of sampling frequency has been successful then
- // we store the payload-type.
- _sendCodecInst.pltype = sendCodec.pltype;
-
- // check if a change in Rate is required
- if(sendCodec.rate != _sendCodecInst.rate)
- {
- if(_codecs[codecID]->SetBitRate(sendCodec.rate) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Could not change the codec rate.");
- return -1;
- }
- _sendCodecInst.rate = sendCodec.rate;
- }
- _previousPayloadType = _sendCodecInst.pltype;
-
- return 0;
- }
-}
-
-// get current send codec
-WebRtc_Word32
-AudioCodingModuleImpl::SendCodec(
- CodecInst& currentSendCodec) const
-{
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
- "SendCodec()");
- CriticalSectionScoped lock(*_acmCritSect);
-
- if(!_sendCodecRegistered)
- {
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
- "SendCodec Failed, no codec is registered");
-
- return -1;
- }
- WebRtcACMCodecParams encoderParam;
- _codecs[_currentSendCodecIdx]->EncoderParams(&encoderParam);
- encoderParam.codecInstant.pltype = _sendCodecInst.pltype;
- memcpy(¤tSendCodec, &(encoderParam.codecInstant),
- sizeof(CodecInst));
-
- return 0;
-}
-
-// get current send freq
-WebRtc_Word32
-AudioCodingModuleImpl::SendFrequency() const
-{
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
- "SendFrequency()");
- CriticalSectionScoped lock(*_acmCritSect);
-
- if(!_sendCodecRegistered)
- {
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
- "SendFrequency Failed, no codec is registered");
-
- return -1;
- }
-
- return _sendCodecInst.plfreq;
-}
-
-// Get encode bitrate
-// Adaptive rate codecs return their current encode target rate, while other codecs
-// return there longterm avarage or their fixed rate.
-WebRtc_Word32
-AudioCodingModuleImpl::SendBitrate() const
-{
- CriticalSectionScoped lock(*_acmCritSect);
-
- if(!_sendCodecRegistered)
- {
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
- "SendBitrate Failed, no codec is registered");
-
- return -1;
- }
-
- WebRtcACMCodecParams encoderParam;
- _codecs[_currentSendCodecIdx]->EncoderParams(&encoderParam);
-
- return encoderParam.codecInstant.rate;
-}
-
-// set available bandwidth, inform the encoder about the estimated bandwidth
-// received from the remote party
-WebRtc_Word32
-AudioCodingModuleImpl::SetReceivedEstimatedBandwidth(
- const WebRtc_Word32 bw )
-{
- return _codecs[_currentSendCodecIdx]->SetEstimatedBandwidth(bw);
-}
-
-// register a transport callback wich will be called to deliver
-// the encoded buffers
-WebRtc_Word32
-AudioCodingModuleImpl::RegisterTransportCallback(
- AudioPacketizationCallback* transport)
-{
- CriticalSectionScoped lock(*_callbackCritSect);
- _packetizationCallback = transport;
- return 0;
-}
-
-// Used by the module to deliver messages to the codec module/appliation
-// AVT(DTMF)
-WebRtc_Word32
-AudioCodingModuleImpl::RegisterIncomingMessagesCallback(
-#ifndef WEBRTC_DTMF_DETECTION
- AudioCodingFeedback* /* incomingMessagesCallback */,
- const ACMCountries /* cpt */)
-{
- return -1;
-#else
- AudioCodingFeedback* incomingMessagesCallback,
- const ACMCountries cpt)
-{
- WebRtc_Word16 status = 0;
-
- // Enter the critical section for callback
- {
- CriticalSectionScoped lock(*_callbackCritSect);
- _dtmfCallback = incomingMessagesCallback;
- }
- // enter the ACM critical section to set up the DTMF class.
- {
- CriticalSectionScoped lock(*_acmCritSect);
- // Check if the call is to disable or enable the callback
- if(incomingMessagesCallback == NULL)
- {
- // callback is disabled, delete DTMF-detector class
- if(_dtmfDetector != NULL)
- {
- delete _dtmfDetector;
- _dtmfDetector = NULL;
- }
- status = 0;
- }
- else
- {
- status = 0;
- if(_dtmfDetector == NULL)
- {
- _dtmfDetector = new(ACMDTMFDetection);
- if(_dtmfDetector == NULL)
- {
- status = -1;
- }
- }
- if(status >= 0)
- {
- status = _dtmfDetector->Enable(cpt);
- if(status < 0)
- {
- // failed to initialize if DTMF-detection was not enabled before,
- // delete the class, and set the callback to NULL and return -1.
- delete _dtmfDetector;
- _dtmfDetector = NULL;
- }
- }
- }
- }
- // check if we failed in setting up the DTMF-detector class
- if((status < 0))
- {
- // we failed, we cannot have the callback
- CriticalSectionScoped lock(*_callbackCritSect);
- _dtmfCallback = NULL;
- }
-
- return status;
-#endif
-}
-
-
-// Add 10MS of raw (PCM) audio data to the encoder
-WebRtc_Word32
-AudioCodingModuleImpl::Add10MsData(
- const AudioFrame& audioFrame)
-{
- // Do we have a codec registered?
- CriticalSectionScoped lock(*_acmCritSect);
- if(!HaveValidEncoder("Add10MsData"))
- {
- return -1;
- }
-
- if(audioFrame._payloadDataLengthInSamples == 0)
- {
- assert(false);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Cannot Add 10 ms audio, payload length is zero");
- return -1;
- }
- // Allow for 8, 16, 32 and 48kHz input audio
- if((audioFrame._frequencyInHz != 8000) &&
- (audioFrame._frequencyInHz != 16000) &&
- (audioFrame._frequencyInHz != 32000) &&
- (audioFrame._frequencyInHz != 48000))
- {
- assert(false);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Cannot Add 10 ms audio, input frequency not valid");
- return -1;
- }
-
-
- // If the length and frequency matches. We currently just support raw PCM
- if((audioFrame._frequencyInHz/ 100) !=
- audioFrame._payloadDataLengthInSamples)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Cannot Add 10 ms audio, input frequency and length doesn't \
-match");
- return -1;
- }
-
- // Calculate the timestamp that should be pushed to codec.
- // This might be different from the timestamp of the frame
- // due to re-sampling
- bool resamplingRequired =
- ((WebRtc_Word32)audioFrame._frequencyInHz != _sendCodecInst.plfreq);
-
- // If number of channels in audio doesn't match codec mode, we need
- // either mono-to-stereo or stereo-to-mono conversion.
- WebRtc_Word16 audio[WEBRTC_10MS_PCM_AUDIO];
- int audio_channels = _sendCodecInst.channels;
- if (audioFrame._audioChannel != _sendCodecInst.channels) {
- if (_sendCodecInst.channels == 2) {
- // Do mono-to-stereo conversion by copying each sample.
- for (int k = 0; k < audioFrame._payloadDataLengthInSamples; k++) {
- audio[k * 2] = audioFrame._payloadData[k];
- audio[(k * 2) + 1] = audioFrame._payloadData[k];
- }
- } else if (_sendCodecInst.channels == 1) {
- // Do stereo-to-mono conversion by creating the average of the stereo
- // samples.
- for (int k = 0; k < audioFrame._payloadDataLengthInSamples; k++) {
- audio[k] = (audioFrame._payloadData[k * 2] +
- audioFrame._payloadData[(k * 2) + 1]) >> 1;
- }
- }
- } else {
- // Copy payload data for future use.
- size_t length = static_cast<size_t>(
- audioFrame._payloadDataLengthInSamples * audio_channels);
- memcpy(audio, audioFrame._payloadData, length * sizeof(WebRtc_UWord16));
- }
-
- WebRtc_UWord32 currentTimestamp;
- WebRtc_Word32 status;
- // if it is required, we have to do a resampling.
- if(resamplingRequired)
- {
- WebRtc_Word16 resampledAudio[WEBRTC_10MS_PCM_AUDIO];
- WebRtc_Word32 sendPlFreq = _sendCodecInst.plfreq;
- WebRtc_UWord32 diffInputTimestamp;
- WebRtc_Word16 newLengthSmpl;
-
- // calculate the timestamp of this frame
- if(_lastInTimestamp > audioFrame._timeStamp)
- {
- // a wrap around has happened
- diffInputTimestamp = ((WebRtc_UWord32)0xFFFFFFFF - _lastInTimestamp)
- + audioFrame._timeStamp;
- }
- else
- {
- diffInputTimestamp = audioFrame._timeStamp - _lastInTimestamp;
- }
- currentTimestamp = _lastTimestamp + (WebRtc_UWord32)(diffInputTimestamp *
- ((double)_sendCodecInst.plfreq / (double)audioFrame._frequencyInHz));
-
- newLengthSmpl = _inputResampler.Resample10Msec(
- audio, audioFrame._frequencyInHz, resampledAudio, sendPlFreq,
- audio_channels);
-
- if(newLengthSmpl < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Cannot add 10 ms audio, resmapling failed");
- return -1;
- }
- status = _codecs[_currentSendCodecIdx]->Add10MsData(currentTimestamp,
- resampledAudio, newLengthSmpl, audio_channels);
- }
- else
- {
- currentTimestamp = audioFrame._timeStamp;
-
- status = _codecs[_currentSendCodecIdx]->Add10MsData(currentTimestamp,
- audio, audioFrame._payloadDataLengthInSamples,
- audio_channels);
- }
- _lastInTimestamp = audioFrame._timeStamp;
- _lastTimestamp = currentTimestamp;
- return status;
-}
-
-/////////////////////////////////////////
-// (FEC) Forward Error Correction
-//
-
-bool
-AudioCodingModuleImpl::FECStatus() const
-{
- CriticalSectionScoped lock(*_acmCritSect);
- return _fecEnabled;
-}
-
-// configure FEC status i.e on/off
-WebRtc_Word32
-AudioCodingModuleImpl::SetFECStatus(
-#ifdef WEBRTC_CODEC_RED
- const bool enableFEC)
-{
- CriticalSectionScoped lock(*_acmCritSect);
-
- if (_fecEnabled != enableFEC)
- {
- // Reset the RED buffer
- memset(_redBuffer, 0, MAX_PAYLOAD_SIZE_BYTE);
-
- // Reset fragmentation buffers
- _fragmentation->fragmentationVectorSize = 2;
- _fragmentation->fragmentationOffset[0] = 0;
- _fragmentation->fragmentationOffset[1] = MAX_PAYLOAD_SIZE_BYTE;
- memset(_fragmentation->fragmentationLength, 0, sizeof(WebRtc_UWord32) * 2);
- memset(_fragmentation->fragmentationTimeDiff, 0, sizeof(WebRtc_UWord16) * 2);
- memset(_fragmentation->fragmentationPlType, 0, sizeof(WebRtc_UWord8) * 2);
-
- // set _fecEnabled
- _fecEnabled = enableFEC;
- }
- _isFirstRED = true; // Make sure we restart FEC
- return 0;
-#else
- const bool /* enableFEC */)
-{
- _fecEnabled = false;
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, _id,
- " WEBRTC_CODEC_RED is undefined => _fecEnabled = %d", _fecEnabled);
- return -1;
-#endif
-}
-
-
-/////////////////////////////////////////
-// (VAD) Voice Activity Detection
-//
-
-WebRtc_Word32
-AudioCodingModuleImpl::SetVAD(
- const bool enableDTX,
- const bool enableVAD,
- const ACMVADMode vadMode)
-{
- CriticalSectionScoped lock(*_acmCritSect);
-
- // sanity check of the mode
- if((vadMode != VADNormal) &&
- (vadMode != VADLowBitrate) &&
- (vadMode != VADAggr) &&
- (vadMode != VADVeryAggr))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Invalid VAD Mode %d, no change is made to VAD/DTX status",
- (int)vadMode);
- return -1;
- }
-
- // If a send codec is registered, set VAD/DTX for the codec
- if(HaveValidEncoder("SetVAD")) {
- WebRtc_Word16 status =
- _codecs[_currentSendCodecIdx]->SetVAD(enableDTX, enableVAD, vadMode);
- if(status == 1) {
- // Vad was enabled;
- _vadEnabled = true;
- _dtxEnabled = enableDTX;
- _vadMode = vadMode;
-
- return 0;
- } else if (status < 0) {
- // SetVAD failed
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "SetVAD failed");
-
- _vadEnabled = false;
- _dtxEnabled = false;
-
- return -1;
- }
- }
-
- _vadEnabled = enableVAD;
- _dtxEnabled = enableDTX;
- _vadMode = vadMode;
-
- return 0;
-}
-
-WebRtc_Word32
-AudioCodingModuleImpl::VAD(
- bool& dtxEnabled,
- bool& vadEnabled,
- ACMVADMode& vadMode) const
-{
- CriticalSectionScoped lock(*_acmCritSect);
-
- dtxEnabled = _dtxEnabled;
- vadEnabled = _vadEnabled;
- vadMode = _vadMode;
-
- return 0;
-}
-
-/////////////////////////////////////////
-// Receiver
-//
-
-WebRtc_Word32
-AudioCodingModuleImpl::InitializeReceiver()
-{
- CriticalSectionScoped lock(*_acmCritSect);
- return InitializeReceiverSafe();
-}
-
-// Initialize receiver, resets codec database etc
-WebRtc_Word32
-AudioCodingModuleImpl::InitializeReceiverSafe()
-{
- // If the receiver is already initialized then we
- // also like to destruct decoders if any exist. After a call
- // to this function, we should have a clean start-up.
- if(_receiverInitialized)
- {
- for(int codecCntr = 0; codecCntr < ACMCodecDB::kNumCodecs; codecCntr++)
- {
- if(UnregisterReceiveCodecSafe(codecCntr) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "InitializeReceiver() failed, Could not unregister codec");
- return -1;
- }
- }
- }
- if (_netEq.Init() != 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "InitializeReceiver() failed, Could not initialize NetEQ");
- return -1;
- }
- _netEq.SetUniqueId(_id);
- if (_netEq.AllocatePacketBuffer(ACMCodecDB::NetEQDecoders(),
- ACMCodecDB::kNumCodecs) != 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "NetEQ cannot allocatePacket Buffer");
- return -1;
- }
-
- // Register RED and CN
- int regInNeteq = 0;
- for (int i = (ACMCodecDB::kNumCodecs - 1); i>-1; i--) {
- if((STR_CASE_CMP(ACMCodecDB::database_[i].plname, "red") == 0)) {
- regInNeteq = 1;
- } else if ((STR_CASE_CMP(ACMCodecDB::database_[i].plname, "CN") == 0)) {
- regInNeteq = 1;
- }
-
- if (regInNeteq == 1) {
- if(RegisterRecCodecMSSafe(ACMCodecDB::database_[i], i, i,
- ACMNetEQ::masterJB) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Cannot register master codec.");
- return -1;
- }
- _registeredPlTypes[i] = ACMCodecDB::database_[i].pltype;
- regInNeteq = 0;
- }
- }
-
- _receiverInitialized = true;
- return 0;
-}
-
-// Reset the decoder state
-WebRtc_Word32
-AudioCodingModuleImpl::ResetDecoder()
-{
- CriticalSectionScoped lock(*_acmCritSect);
-
- for(int codecCntr = 0; codecCntr < ACMCodecDB::kMaxNumCodecs; codecCntr++)
- {
- if((_codecs[codecCntr] != NULL) && (_registeredPlTypes[codecCntr] != -1))
- {
- if(_codecs[codecCntr]->ResetDecoder(_registeredPlTypes[codecCntr]) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "ResetDecoder failed:");
- return -1;
- }
- }
- }
- return _netEq.FlushBuffers();
-}
-
-// get current receive freq
-WebRtc_Word32
-AudioCodingModuleImpl::ReceiveFrequency() const
-{
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
- "ReceiveFrequency()");
- WebRtcACMCodecParams codecParams;
-
- CriticalSectionScoped lock(*_acmCritSect);
- if(DecoderParamByPlType(_lastRecvAudioCodecPlType, codecParams) < 0)
- {
- return _netEq.CurrentSampFreqHz();
- }
- else
- {
- return codecParams.codecInstant.plfreq;
- }
-}
-
-// get current playout freq
-WebRtc_Word32
-AudioCodingModuleImpl::PlayoutFrequency() const
-{
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
- "PlayoutFrequency()");
-
- CriticalSectionScoped lock(*_acmCritSect);
-
- return _netEq.CurrentSampFreqHz();
-}
-
-
-// register possible reveive codecs, can be called multiple times,
-// for codecs, CNG (NB, WB and SWB), DTMF, RED
-WebRtc_Word32
-AudioCodingModuleImpl::RegisterReceiveCodec(
- const CodecInst& receiveCodec)
-{
- CriticalSectionScoped lock(*_acmCritSect);
-
- if(receiveCodec.channels > 2)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "More than 2 audio channel is not supported.");
- return -1;
- }
-
- int mirrorId;
- int codecId = ACMCodecDB::ReceiverCodecNumber(&receiveCodec, &mirrorId);
-
- if(codecId < 0 || codecId >= ACMCodecDB::kNumCodecs)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Wrong codec params to be registered as receive codec");
- return -1;
- }
- // Check if the payload-type is valid.
- if(!ACMCodecDB::ValidPayloadType(receiveCodec.pltype))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Invalid payload-type %d for %s.",
- receiveCodec.pltype, receiveCodec.plname);
- return -1;
- }
-
- if(!_receiverInitialized)
- {
- if(InitializeReceiverSafe() < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Cannot initialize reciver, so failed registering a codec.");
- return -1;
- }
- }
-
- // If codec already registered, start with unregistering
- if(_registeredPlTypes[codecId] != -1)
- {
- if(UnregisterReceiveCodecSafe(codecId) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Cannot register master codec.");
- return -1;
- }
- }
-
- if(RegisterRecCodecMSSafe(receiveCodec, codecId, mirrorId,
- ACMNetEQ::masterJB) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Cannot register master codec.");
- return -1;
- }
-
- // If receive stereo, make sure we have two instances of NetEQ, one for each channel
- if(receiveCodec.channels == 2)
- {
- if(_netEq.NumSlaves() < 1)
- {
- if(_netEq.AddSlave(ACMCodecDB::NetEQDecoders(),
- ACMCodecDB::kNumCodecs) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Cannot Add Slave jitter buffer to NetEQ.");
- return -1;
- }
-
- // Register RED and CN in slave.
- bool reg_in_neteq = false;
- for (int i = (ACMCodecDB::kNumCodecs - 1); i > -1; i--) {
- if((STR_CASE_CMP(ACMCodecDB::database_[i].plname, "RED") == 0)) {
- reg_in_neteq = true;
- } else if ((STR_CASE_CMP(ACMCodecDB::database_[i].plname, "CN") == 0)) {
- reg_in_neteq = true;
- }
-
- if (reg_in_neteq) {
- if(RegisterRecCodecMSSafe(ACMCodecDB::database_[i], i, i,
- ACMNetEQ::slaveJB) < 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Cannot register slave codec.");
- return -1;
- }
- _registeredPlTypes[i] = ACMCodecDB::database_[i].pltype;
- reg_in_neteq = false;
- }
- }
- }
-
- if(RegisterRecCodecMSSafe(receiveCodec, codecId, mirrorId,
- ACMNetEQ::slaveJB) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Cannot register slave codec.");
- return -1;
- }
-
- if((_stereoReceive[codecId] == false) &&
- (_lastRecvAudioCodecPlType == receiveCodec.pltype))
- {
- _lastRecvAudioCodecPlType = -1;
- }
- _stereoReceive[codecId] = true;
- }
- else
- {
- _stereoReceive[codecId] = false;
- }
-
- _registeredPlTypes[codecId] = receiveCodec.pltype;
-
- if(!STR_CASE_CMP(receiveCodec.plname, "RED"))
- {
- _receiveREDPayloadType = receiveCodec.pltype;
- }
- return 0;
-}
-
-
-
-WebRtc_Word32
-AudioCodingModuleImpl::RegisterRecCodecMSSafe(
- const CodecInst& receiveCodec,
- WebRtc_Word16 codecId,
- WebRtc_Word16 mirrorId,
- ACMNetEQ::JB jitterBuffer)
-{
- ACMGenericCodec** codecArray;
- if(jitterBuffer == ACMNetEQ::masterJB)
- {
- codecArray = &_codecs[0];
- }
- else if(jitterBuffer == ACMNetEQ::slaveJB)
- {
- codecArray = &_slaveCodecs[0];
- if (_codecs[codecId]->IsTrueStereoCodec()) {
- // True stereo codecs need to use the same codec memory
- // for both master and slave.
- _slaveCodecs[mirrorId] = _codecs[mirrorId];
- _mirrorCodecIdx[mirrorId] = mirrorId;
- }
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "RegisterReceiveCodecMSSafe failed, jitterBuffer is neither master or slave ");
- return -1;
- }
-
- if (codecArray[mirrorId] == NULL)
- {
- codecArray[mirrorId] = CreateCodec(receiveCodec);
- if(codecArray[mirrorId] == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Cannot create codec to register as receive codec");
- return -1;
- }
- _mirrorCodecIdx[mirrorId] = mirrorId;
- }
- if(mirrorId != codecId)
- {
- codecArray[codecId] = codecArray[mirrorId];
- _mirrorCodecIdx[codecId] = mirrorId;
- }
-
- codecArray[codecId]->SetIsMaster(jitterBuffer == ACMNetEQ::masterJB);
-
- WebRtc_Word16 status = 0;
- bool registerInNetEq = true;
- WebRtcACMCodecParams codecParams;
- memcpy(&(codecParams.codecInstant), &receiveCodec,
- sizeof(CodecInst));
- codecParams.enableVAD = false;
- codecParams.enableDTX = false;
- codecParams.vadMode = VADNormal;
- if (!codecArray[codecId]->DecoderInitialized())
- {
- // force initialization
- status = codecArray[codecId]->InitDecoder(&codecParams, true);
- if(status < 0)
- {
- // could not initialize the decoder we don't want to
- // continue if we could not initialize properly.
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "could not initialize the receive codec, codec not registered");
-
- return -1;
- }
- }
- else if(mirrorId != codecId)
- {
- // Currently this only happens for iSAC.
- // we have to store the decoder parameters
-
- codecArray[codecId]->SaveDecoderParam(&codecParams);
- }
- if (registerInNetEq)
- {
- if(codecArray[codecId]->RegisterInNetEq(&_netEq, receiveCodec)
- != 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Receive codec could not be registered in NetEQ");
-
- return -1;
- }
- // Guaranty that the same payload-type that is
- // registered in NetEQ is stored in the codec.
- codecArray[codecId]->SaveDecoderParam(&codecParams);
- }
-
- return status;
-}
-
-
-
-// Get current received codec
-WebRtc_Word32
-AudioCodingModuleImpl::ReceiveCodec(
- CodecInst& currentReceiveCodec) const
-{
- WebRtcACMCodecParams decoderParam;
- CriticalSectionScoped lock(*_acmCritSect);
-
- for(int decCntr = 0; decCntr < ACMCodecDB::kMaxNumCodecs; decCntr++)
- {
- if(_codecs[decCntr] != NULL)
- {
- if(_codecs[decCntr]->DecoderInitialized())
- {
- if(_codecs[decCntr]->DecoderParams(&decoderParam,
- _lastRecvAudioCodecPlType))
- {
- memcpy(¤tReceiveCodec, &decoderParam.codecInstant,
- sizeof(CodecInst));
- return 0;
- }
- }
- }
- }
-
- // if we are here then we haven't found any codec
- // set codec pltype to -1 to indicate that the structure
- // is invalid and return -1.
- currentReceiveCodec.pltype = -1;
- return -1;
-}
-
-// Incoming packet from network parsed and ready for decode
-WebRtc_Word32
-AudioCodingModuleImpl::IncomingPacket(
- const WebRtc_Word8* incomingPayload,
- const WebRtc_Word32 payloadLength,
- const WebRtcRTPHeader& rtpInfo)
-{
- if (payloadLength < 0)
- {
- // Log error
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "IncomingPacket() Error, payload-length cannot be negative");
- return -1;
- }
- {
- // store the payload Type. this will be used to retrieve "received codec"
- // and "received frequency."
- CriticalSectionScoped lock(*_acmCritSect);
-#ifdef ACM_QA_TEST
- if(_incomingPL != NULL)
- {
- fwrite(&rtpInfo.header.timestamp, sizeof(WebRtc_UWord32), 1, _incomingPL);
- fwrite(&rtpInfo.header.payloadType, sizeof(WebRtc_UWord8), 1, _incomingPL);
- fwrite(&payloadLength, sizeof(WebRtc_Word16), 1, _incomingPL);
- }
-#endif
-
- WebRtc_UWord8 myPayloadType;
-
- // Check if this is an RED payload
- if(rtpInfo.header.payloadType == _receiveREDPayloadType)
- {
- // get the primary payload-type.
- myPayloadType = (WebRtc_UWord8)(incomingPayload[0] & 0x7F);
- }
- else
- {
- myPayloadType = rtpInfo.header.payloadType;
- }
-
- // If payload is audio, check if received payload is different from previous
- if((!rtpInfo.type.Audio.isCNG) &&
- (myPayloadType != _cngNB.pltype) &&
- (myPayloadType != _cngWB.pltype) &&
- (myPayloadType != _cngSWB.pltype))
- {
- // This is Audio not CNG
-
- if(myPayloadType != _lastRecvAudioCodecPlType)
- {
- // We detect a change in payload type. It is necessary for iSAC
- // we are going to use ONE iSAC instance for decoding both WB and
- // SWB payloads. If payload is changed there might be a need to reset
- // sampling rate of decoder. depending what we have received "now".
- for(int i = 0; i < ACMCodecDB::kMaxNumCodecs; i++)
- {
- if(_registeredPlTypes[i] == myPayloadType)
- {
- if(_codecs[i] == NULL)
- {
- // we found a payload type but the corresponding
- // codec is NULL this should not happen
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "IncomingPacket() Error, payload type found but corresponding "
- "codec is NULL");
- return -1;
- }
- _codecs[i]->UpdateDecoderSampFreq(i);
- _netEq.SetReceivedStereo(_stereoReceive[i]);
-
- // Store number of channels we expect to receive for the
- // current payload type.
- if (_stereoReceive[i]) {
- _expected_channels = 2;
- } else {
- _expected_channels = 1;
- }
-
- // Reset previous received channel
- _prev_received_channel = 0;
-
- break;
- }
- }
- }
- _lastRecvAudioCodecPlType = myPayloadType;
- }
- }
-
- // Check that number of received channels match the setup for the
- // received codec.
- if (_expected_channels == 2) {
- if ((_prev_received_channel == 1) && (rtpInfo.type.Audio.channel == 1)) {
- // We expect every second call to this function to be for channel 2,
- // since we are in stereo-receive mode.
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "IncomingPacket() Error, payload is"
- "mono, but codec registered as stereo.");
- return -1;
- }
- _prev_received_channel = rtpInfo.type.Audio.channel;
- } else if (rtpInfo.type.Audio.channel == 2) {
- // Codec is registered as mono, but we receive a stereo packet.
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "IncomingPacket() Error, payload is"
- "stereo, but codec registered as mono.");
- return -1;
- }
-
- // Insert packet into NetEQ.
- return _netEq.RecIn(incomingPayload, payloadLength, rtpInfo);
-}
-
-// Minimum playout delay (Used for lip-sync)
-WebRtc_Word32
-AudioCodingModuleImpl::SetMinimumPlayoutDelay(
- const WebRtc_Word32 timeMs)
-{
- if((timeMs < 0) || (timeMs > 1000))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Delay must be in the range of 0-1000 milliseconds.");
- return -1;
- }
- return _netEq.SetExtraDelay(timeMs);
-}
-
-// Get Dtmf playout status
-bool
-AudioCodingModuleImpl::DtmfPlayoutStatus() const
-{
-#ifndef WEBRTC_CODEC_AVT
- return false;
-#else
- return _netEq.AVTPlayout();
-#endif
-}
-
-// configure Dtmf playout status i.e on/off
-// playout the incoming outband Dtmf tone
-WebRtc_Word32
-AudioCodingModuleImpl::SetDtmfPlayoutStatus(
-#ifndef WEBRTC_CODEC_AVT
- const bool /* enable */)
-{
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioCoding, _id,
- "SetDtmfPlayoutStatus() failed: AVT is not supported.");
- return -1;
-#else
- const bool enable)
-{
- return _netEq.SetAVTPlayout(enable);
-#endif
-}
-
-// Estimate the Bandwidth based on the incoming stream
-// This is also done in the RTP module
-// need this for one way audio where the RTCP send the BW estimate
-WebRtc_Word32
-AudioCodingModuleImpl::DecoderEstimatedBandwidth() const
-{
- CodecInst codecInst;
- WebRtc_Word16 codecID = -1;
- int plTypWB;
- int plTypSWB;
-
- // Get iSAC settings
- for(int codecCntr = 0; codecCntr < ACMCodecDB::kNumCodecs; codecCntr++)
- {
- // Store codec settings for codec number "codeCntr" in the output struct
- ACMCodecDB::Codec(codecCntr, &codecInst);
-
- if(!STR_CASE_CMP(codecInst.plname, "isac"))
- {
- codecID = 1;
- plTypWB = codecInst.pltype;
-
- ACMCodecDB::Codec(codecCntr+1, &codecInst);
- plTypSWB = codecInst.pltype;
-
- break;
- }
- }
-
- if(codecID < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "DecoderEstimatedBandwidth failed");
- return -1;
- }
-
- if ((_lastRecvAudioCodecPlType == plTypWB) || (_lastRecvAudioCodecPlType == plTypSWB))
- {
- return _codecs[codecID]->GetEstimatedBandwidth();
- } else {
- return -1;
- }
-}
-
-// Set playout mode for: voice, fax, or streaming
-WebRtc_Word32
-AudioCodingModuleImpl::SetPlayoutMode(
- const AudioPlayoutMode mode)
-{
- if((mode != voice) &&
- (mode != fax) &&
- (mode != streaming))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Invalid playout mode.");
- return -1;
- }
- return _netEq.SetPlayoutMode(mode);
-}
-
-// Get playout mode voice, fax
-AudioPlayoutMode
-AudioCodingModuleImpl::PlayoutMode() const
-{
- return _netEq.PlayoutMode();
-}
-
-
-// Get 10 milliseconds of raw audio data to play out
-// automatic resample to the requested frequency
-WebRtc_Word32
-AudioCodingModuleImpl::PlayoutData10Ms(
- const WebRtc_Word32 desiredFreqHz,
- AudioFrame& audioFrame)
-{
- bool stereoMode;
-
- // recOut always returns 10 ms
- if (_netEq.RecOut(_audioFrame) != 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "PlayoutData failed, RecOut Failed");
- return -1;
- }
-
- audioFrame._audioChannel = _audioFrame._audioChannel;
- audioFrame._vadActivity = _audioFrame._vadActivity;
- audioFrame._speechType = _audioFrame._speechType;
-
- stereoMode = (_audioFrame._audioChannel > 1);
- //For stereo playout:
- // Master and Slave samples are interleaved starting with Master
-
- const WebRtc_UWord16 recvFreq = static_cast<WebRtc_UWord16>(_audioFrame._frequencyInHz);
- bool toneDetected = false;
- WebRtc_Word16 lastDetectedTone;
- WebRtc_Word16 tone;
-
- // limit the scope of ACM Critical section
- // perhaps we don't need to have output resampler in
- // critical section, it is supposed to be called in this
- // function and no where else. However, it won't degrade complexity
- {
- CriticalSectionScoped lock(*_acmCritSect);
-
- if ((recvFreq != desiredFreqHz) && (desiredFreqHz != -1))
- {
- // resample payloadData
- WebRtc_Word16 tmpLen = _outputResampler.Resample10Msec(
- _audioFrame._payloadData, recvFreq, audioFrame._payloadData, desiredFreqHz,
- _audioFrame._audioChannel);
-
- if(tmpLen < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "PlayoutData failed, resampler failed");
- return -1;
- }
-
- //Set the payload data length from the resampler
- audioFrame._payloadDataLengthInSamples = (WebRtc_UWord16)tmpLen;
- // set the ssampling frequency
- audioFrame._frequencyInHz = desiredFreqHz;
- }
- else
- {
- memcpy(audioFrame._payloadData, _audioFrame._payloadData,
- _audioFrame._payloadDataLengthInSamples * audioFrame._audioChannel
- * sizeof(WebRtc_Word16));
- // set the payload length
- audioFrame._payloadDataLengthInSamples = _audioFrame._payloadDataLengthInSamples;
- // set the sampling frequency
- audioFrame._frequencyInHz = recvFreq;
- }
-
- //Tone detection done for master channel
- if(_dtmfDetector != NULL)
- {
- // Dtmf Detection
- if(audioFrame._frequencyInHz == 8000)
- {
- // use audioFrame._payloadData then Dtmf detector doesn't
- // need resampling
- if(!stereoMode)
- {
- _dtmfDetector->Detect(audioFrame._payloadData,
- audioFrame._payloadDataLengthInSamples,
- audioFrame._frequencyInHz, toneDetected, tone);
- }
- else
- {
- // we are in 8 kHz so the master channel needs only 80 samples
- WebRtc_Word16 masterChannel[80];
- for(int n = 0; n < 80; n++)
- {
- masterChannel[n] = audioFrame._payloadData[n<<1];
- }
- _dtmfDetector->Detect(masterChannel,
- audioFrame._payloadDataLengthInSamples,
- audioFrame._frequencyInHz, toneDetected, tone);
- }
- }
- else
- {
- // Do the detection on the audio that we got from NetEQ (_audioFrame).
- if(!stereoMode)
- {
- _dtmfDetector->Detect(_audioFrame._payloadData,
- _audioFrame._payloadDataLengthInSamples, recvFreq,
- toneDetected, tone);
- }
- else
- {
- WebRtc_Word16 masterChannel[WEBRTC_10MS_PCM_AUDIO];
- for(int n = 0; n < _audioFrame._payloadDataLengthInSamples; n++)
- {
- masterChannel[n] = _audioFrame._payloadData[n<<1];
- }
- _dtmfDetector->Detect(masterChannel,
- _audioFrame._payloadDataLengthInSamples, recvFreq,
- toneDetected, tone);
- }
- }
- }
-
- // we want to do this while we are in _acmCritSect
- // doesn't really need to initialize the following
- // variable but Linux complains if we don't
- lastDetectedTone = kACMToneEnd;
- if(toneDetected)
- {
- lastDetectedTone = _lastDetectedTone;
- _lastDetectedTone = tone;
- }
- }
-
- if(toneDetected)
- {
- // we will deal with callback here, so enter callback critical
- // section
- CriticalSectionScoped lock(*_callbackCritSect);
-
- if(_dtmfCallback != NULL)
- {
- if(tone != kACMToneEnd)
- {
- // just a tone
- _dtmfCallback->IncomingDtmf((WebRtc_UWord8)tone, false);
- }
- else if((tone == kACMToneEnd) &&
- (lastDetectedTone != kACMToneEnd))
- {
- // The tone is "END" and the previously detected tone is
- // not "END," so call fir an end.
- _dtmfCallback->IncomingDtmf((WebRtc_UWord8)lastDetectedTone,
- true);
- }
- }
- }
-
- audioFrame._id = _id;
- audioFrame._volume = -1;
- audioFrame._energy = -1;
- audioFrame._timeStamp = 0;
-
- return 0;
-}
-
-/////////////////////////////////////////
-// (CNG) Comfort Noise Generation
-// Generate comfort noise when receiving DTX packets
-//
-
-// Get VAD aggressiveness on the incoming stream
-ACMVADMode
-AudioCodingModuleImpl::ReceiveVADMode() const
-{
- return _netEq.VADMode();
-}
-
-// Configure VAD aggressiveness on the incoming stream
-WebRtc_Word16
-AudioCodingModuleImpl::SetReceiveVADMode(
- const ACMVADMode mode)
-{
- return _netEq.SetVADMode(mode);
-}
-
-/////////////////////////////////////////
-// statistics
-//
-
-WebRtc_Word32
-AudioCodingModuleImpl::NetworkStatistics(
- ACMNetworkStatistics& statistics) const
-{
- WebRtc_Word32 status;
- status = _netEq.NetworkStatistics(&statistics);
- return status;
-}
-
-void
-AudioCodingModuleImpl::DestructEncoderInst(
- void* ptrInst)
-{
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _id,
- "DestructEncoderInst()");
- if(!HaveValidEncoder("DestructEncoderInst"))
- {
- return;
- }
-
- _codecs[_currentSendCodecIdx]->DestructEncoderInst(ptrInst);
-}
-
-WebRtc_Word16
-AudioCodingModuleImpl::AudioBuffer(
- WebRtcACMAudioBuff& audioBuff)
-{
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _id,
- "AudioBuffer()");
- if(!HaveValidEncoder("AudioBuffer"))
- {
- return -1;
- }
-
- audioBuff.lastInTimestamp = _lastInTimestamp;
- return _codecs[_currentSendCodecIdx]->AudioBuffer(audioBuff);
-}
-
-WebRtc_Word16
-AudioCodingModuleImpl::SetAudioBuffer(
- WebRtcACMAudioBuff& audioBuff)
-{
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _id,
- "SetAudioBuffer()");
- if(!HaveValidEncoder("SetAudioBuffer"))
- {
- return -1;
- }
-
- return _codecs[_currentSendCodecIdx]->SetAudioBuffer(audioBuff);
-}
-
-
-WebRtc_UWord32
-AudioCodingModuleImpl::EarliestTimestamp() const
-{
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _id,
- "EarliestTimestamp()");
- if(!HaveValidEncoder("EarliestTimestamp"))
- {
- return -1;
- }
-
- return _codecs[_currentSendCodecIdx]->EarliestTimestamp();
-}
-
-WebRtc_Word32
-AudioCodingModuleImpl::RegisterVADCallback(
- ACMVADCallback* vadCallback)
-{
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceAudioCoding, _id,
- "RegisterVADCallback()");
- CriticalSectionScoped lock(*_callbackCritSect);
- _vadCallback = vadCallback;
- return 0;
-}
-
-WebRtc_Word32
-AudioCodingModuleImpl::IncomingPayload(
- const WebRtc_Word8* incomingPayload,
- const WebRtc_Word32 payloadLength,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timestamp)
-{
- if (payloadLength < 0)
- {
- // Log error in trace file.
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "IncomingPacket() Error, payload-length cannot be negative");
- return -1;
- }
-
- if(_dummyRTPHeader == NULL)
- {
- // This is the first time that we are using _dummyRTPHeader
- // so we have to create it.
- WebRtcACMCodecParams codecParams;
- _dummyRTPHeader = new WebRtcRTPHeader;
- if (_dummyRTPHeader == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "IncomingPacket() Error, out of memory");
- return -1;
- }
- _dummyRTPHeader->header.payloadType = payloadType;
- // Don't matter in this case
- _dummyRTPHeader->header.ssrc = 0;
- _dummyRTPHeader->header.markerBit = false;
- // start with random numbers
- _dummyRTPHeader->header.sequenceNumber = rand();
- _dummyRTPHeader->header.timestamp = (((WebRtc_UWord32)rand()) << 16) +
- (WebRtc_UWord32)rand();
- _dummyRTPHeader->type.Audio.channel = 1;
-
- if(DecoderParamByPlType(payloadType, codecParams) < 0)
- {
- // we didn't find a codec with the given payload.
- // something is wrong we exit, but we delete _dummyRTPHeader
- // and set it to NULL to start clean next time
- delete _dummyRTPHeader;
- _dummyRTPHeader = NULL;
- return -1;
- }
- _recvPlFrameSizeSmpls = codecParams.codecInstant.pacsize;
- }
-
- if(payloadType != _dummyRTPHeader->header.payloadType)
- {
- // payload type has changed since the last time we might need to
- // update the frame-size
- WebRtcACMCodecParams codecParams;
- if(DecoderParamByPlType(payloadType, codecParams) < 0)
- {
- // we didn't find a codec with the given payload.
- // something is wrong we exit
- return -1;
- }
- _recvPlFrameSizeSmpls = codecParams.codecInstant.pacsize;
- _dummyRTPHeader->header.payloadType = payloadType;
- }
-
- if(timestamp > 0)
- {
- _dummyRTPHeader->header.timestamp = timestamp;
- }
-
- // store the payload Type. this will be used to retrieve "received codec"
- // and "received frequency."
- _lastRecvAudioCodecPlType = payloadType;
-
- // Insert in NetEQ
- if(_netEq.RecIn(incomingPayload, payloadLength, (*_dummyRTPHeader)) < 0)
- {
- return -1;
- }
-
- // get ready for the next payload
- _dummyRTPHeader->header.sequenceNumber++;
- _dummyRTPHeader->header.timestamp += _recvPlFrameSizeSmpls;
- return 0;
-}
-
-WebRtc_Word16
-AudioCodingModuleImpl::DecoderParamByPlType(
- const WebRtc_UWord8 payloadType,
- WebRtcACMCodecParams& codecParams) const
-{
- CriticalSectionScoped lock(*_acmCritSect);
- for(WebRtc_Word16 codecCntr = 0; codecCntr < ACMCodecDB::kMaxNumCodecs; codecCntr++)
- {
- if(_codecs[codecCntr] != NULL)
- {
- if(_codecs[codecCntr]->DecoderInitialized())
- {
- if(_codecs[codecCntr]->DecoderParams(&codecParams,
- payloadType))
- {
- return 0;
- }
- }
- }
- }
- // if we are here it means that we could not find a
- // codec with that payload type. reset the values to
- // not acceptable values and return -1;
- codecParams.codecInstant.plname[0] = '\0';
- codecParams.codecInstant.pacsize = 0;
- codecParams.codecInstant.rate = 0;
- codecParams.codecInstant.pltype = -1;
- return -1;
-}
-
-
-
-WebRtc_Word16
-AudioCodingModuleImpl::DecoderListIDByPlName(
- const char* payloadName,
- const WebRtc_UWord16 sampFreqHz) const
-{
- WebRtcACMCodecParams codecParams;
- CriticalSectionScoped lock(*_acmCritSect);
- for(WebRtc_Word16 codecCntr = 0; codecCntr < ACMCodecDB::kMaxNumCodecs; codecCntr++)
- {
- if((_codecs[codecCntr] != NULL))
- {
- if(_codecs[codecCntr]->DecoderInitialized())
- {
- assert(_registeredPlTypes[codecCntr] >= 0);
- assert(_registeredPlTypes[codecCntr] <= 255);
- _codecs[codecCntr]->DecoderParams(&codecParams,
- (WebRtc_UWord8)_registeredPlTypes[codecCntr]);
- if(!STR_CASE_CMP(codecParams.codecInstant.plname, payloadName))
- {
- // Check if the given sampling frequency matches.
- // A zero sampling frequency means we matching the names
- // is sufficient and we don't need to check for the
- // frequencies.
- // Currently it is only iSAC which has one name but two
- // sampling frequencies.
- if((sampFreqHz == 0) ||
- (codecParams.codecInstant.plfreq == sampFreqHz))
- {
- return codecCntr;
- }
- }
- }
- }
- }
- // if we are here it means that we could not find a
- // codec with that payload type. return -1;
- return -1;
-}
-
-WebRtc_Word32
-AudioCodingModuleImpl::LastEncodedTimestamp(WebRtc_UWord32& timestamp) const
-{
- CriticalSectionScoped lock(*_acmCritSect);
- if(!HaveValidEncoder("LastEncodedTimestamp"))
- {
- return -1;
- }
- timestamp = _codecs[_currentSendCodecIdx]->LastEncodedTimestamp();
- return 0;
-}
-
-WebRtc_Word32
-AudioCodingModuleImpl::ReplaceInternalDTXWithWebRtc(bool useWebRtcDTX)
-{
- CriticalSectionScoped lock(*_acmCritSect);
-
- if(!HaveValidEncoder("ReplaceInternalDTXWithWebRtc"))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Cannot replace codec internal DTX when no send codec is registered.");
- return -1;
- }
-
- WebRtc_Word32 res = _codecs[_currentSendCodecIdx]->ReplaceInternalDTX(useWebRtcDTX);
- // Check if VAD is turned on, or if there is any error
- if(res == 1)
- {
- _vadEnabled = true;
- } else if(res < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Failed to set ReplaceInternalDTXWithWebRtc(%d)", useWebRtcDTX);
- return res;
- }
-
- return 0;
-}
-
-WebRtc_Word32
-AudioCodingModuleImpl::IsInternalDTXReplacedWithWebRtc(bool& usesWebRtcDTX)
-{
- CriticalSectionScoped lock(*_acmCritSect);
-
- if(!HaveValidEncoder("IsInternalDTXReplacedWithWebRtc"))
- {
- return -1;
- }
- if(_codecs[_currentSendCodecIdx]->IsInternalDTXReplaced(&usesWebRtcDTX) < 0)
- {
- return -1;
- }
- return 0;
-}
-
-
-WebRtc_Word32
-AudioCodingModuleImpl::SetISACMaxRate(
- const WebRtc_UWord32 maxRateBitPerSec)
-{
- CriticalSectionScoped lock(*_acmCritSect);
-
- if(!HaveValidEncoder("SetISACMaxRate"))
- {
- return -1;
- }
-
- return _codecs[_currentSendCodecIdx]->SetISACMaxRate(maxRateBitPerSec);
-}
-
-
-WebRtc_Word32
-AudioCodingModuleImpl::SetISACMaxPayloadSize(
- const WebRtc_UWord16 maxPayloadLenBytes)
-{
- CriticalSectionScoped lock(*_acmCritSect);
-
- if(!HaveValidEncoder("SetISACMaxPayloadSize"))
- {
- return -1;
- }
-
- return _codecs[_currentSendCodecIdx]->SetISACMaxPayloadSize(maxPayloadLenBytes);
-}
-
-WebRtc_Word32
-AudioCodingModuleImpl::ConfigISACBandwidthEstimator(
- const WebRtc_UWord8 initFrameSizeMsec,
- const WebRtc_UWord16 initRateBitPerSec,
- const bool enforceFrameSize)
-{
- CriticalSectionScoped lock(*_acmCritSect);
-
- if(!HaveValidEncoder("ConfigISACBandwidthEstimator"))
- {
- return -1;
- }
-
- return _codecs[_currentSendCodecIdx]->ConfigISACBandwidthEstimator(
- initFrameSizeMsec, initRateBitPerSec, enforceFrameSize);
-}
-
-WebRtc_Word32
-AudioCodingModuleImpl::SetBackgroundNoiseMode(
- const ACMBackgroundNoiseMode mode)
-{
- if((mode < On) ||
- (mode > Off))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "The specified background noise is out of range.\n");
- return -1;
- }
- return _netEq.SetBackgroundNoiseMode(mode);
-}
-
-WebRtc_Word32
-AudioCodingModuleImpl::BackgroundNoiseMode(
- ACMBackgroundNoiseMode& mode)
-{
- return _netEq.BackgroundNoiseMode(mode);
-}
-
-WebRtc_Word32
-AudioCodingModuleImpl::PlayoutTimestamp(
- WebRtc_UWord32& timestamp)
-{
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceAudioCoding, _id,
- "PlayoutTimestamp()");
- return _netEq.PlayoutTimestamp(timestamp);
-}
-
-bool
-AudioCodingModuleImpl::HaveValidEncoder(
- const char* callerName) const
-{
- if((!_sendCodecRegistered) ||
- (_currentSendCodecIdx < 0) ||
- (_currentSendCodecIdx >= ACMCodecDB::kNumCodecs))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "%s failed: No send codec is registered.", callerName);
- return false;
- }
- if((_currentSendCodecIdx < 0) ||
- (_currentSendCodecIdx >= ACMCodecDB::kNumCodecs))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "%s failed: Send codec index out of range.", callerName);
- return false;
- }
- if(_codecs[_currentSendCodecIdx] == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "%s failed: Send codec is NULL pointer.", callerName);
- return false;
- }
- return true;
-}
-
-WebRtc_Word32
-AudioCodingModuleImpl::UnregisterReceiveCodec(
- const WebRtc_Word16 payloadType)
-{
- CriticalSectionScoped lock(*_acmCritSect);
- WebRtc_Word16 codecID;
-
- // Search through the list of registered payload types
- for (codecID = 0; codecID < ACMCodecDB::kMaxNumCodecs; codecID++)
- {
- if (_registeredPlTypes[codecID] == payloadType)
- {
- // we have found the codecID registered with the payload type
- break;
- }
- }
-
- if(codecID >= ACMCodecDB::kNumCodecs)
- {
- // payload type was not registered. No need to unregister
- return 0;
- }
-
- // Unregister the codec with the given payload type
- return UnregisterReceiveCodecSafe(codecID);
-}
-
-WebRtc_Word32
-AudioCodingModuleImpl::UnregisterReceiveCodecSafe(
- const WebRtc_Word16 codecID)
-{
- const WebRtcNetEQDecoder *neteqDecoder = ACMCodecDB::NetEQDecoders();
- WebRtc_Word16 mirrorID = ACMCodecDB::MirrorID(codecID);
- if(_codecs[codecID] != NULL)
- {
- if(_registeredPlTypes[codecID] != -1)
- {
- // before deleting the decoder instance unregister
- // from NetEQ.
- if(_netEq.RemoveCodec(neteqDecoder[codecID], _stereoReceive[codecID]) < 0)
- {
- CodecInst codecInst;
- ACMCodecDB::Codec(codecID, &codecInst);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioCoding, _id,
- "Unregistering %s-%d from NetEQ failed.",
- codecInst.plname, codecInst.plfreq);
- return -1;
- }
-
- // CN is a special case for NetEQ, all three sampling frequencies are
- // deletad if one is deleted
- if(STR_CASE_CMP(ACMCodecDB::database_[codecID].plname, "CN") == 0)
- {
- // Search codecs nearby in the database to unregister all CN.
- for (int i=-2; i<3; i++)
- {
- if (STR_CASE_CMP(ACMCodecDB::database_[codecID+i].plname, "CN") == 0)
- {
- _codecs[codecID+i]->DestructDecoder();
- if(_stereoReceive[codecID+i])
- {
- _slaveCodecs[codecID+i]->DestructDecoder();
- }
- _registeredPlTypes[codecID+i] = -1;
- }
- }
- } else
- {
- if(codecID == mirrorID)
- {
- _codecs[codecID]->DestructDecoder();
- if(_stereoReceive[codecID])
- {
- _slaveCodecs[codecID]->DestructDecoder();
- }
- }
- }
- }
- }
-
- if(_registeredPlTypes[codecID] == _receiveREDPayloadType)
- {
- // RED is going to be unregistered.
- // set the following to an invalid value.
- _receiveREDPayloadType = 255;
- }
- _registeredPlTypes[codecID] = -1;
-
- return 0;
-}
-
-WebRtc_Word32
-AudioCodingModuleImpl::REDPayloadISAC(
- const WebRtc_Word32 isacRate,
- const WebRtc_Word16 isacBwEstimate,
- WebRtc_UWord8* payload,
- WebRtc_Word16* payloadLenByte)
-{
- if(!HaveValidEncoder("EncodeData"))
- {
- return -1;
- }
- WebRtc_Word16 status;
-
- status = _codecs[_currentSendCodecIdx]->REDPayloadISAC(isacRate, isacBwEstimate,
- payload, payloadLenByte);
-
- return status;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/source/audio_coding_module_impl.h b/trunk/src/modules/audio_coding/main/source/audio_coding_module_impl.h
deleted file mode 100644
index 422b22e..0000000
--- a/trunk/src/modules/audio_coding/main/source/audio_coding_module_impl.h
+++ /dev/null
@@ -1,384 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_AUDIO_CODING_MODULE_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_AUDIO_CODING_MODULE_IMPL_H_
-
-#include "acm_codec_database.h"
-#include "acm_neteq.h"
-#include "acm_resampler.h"
-#include "common_types.h"
-#include "engine_configurations.h"
-
-namespace webrtc {
-
-class ACMDTMFDetection;
-class ACMGenericCodec;
-class CriticalSectionWrapper;
-class RWLockWrapper;
-
-//#define TIMED_LOGGING
-
-#ifdef TIMED_LOGGING
- #include "../test/timedtrace.h"
-#endif
-
-#ifdef ACM_QA_TEST
-# include <stdio.h>
-#endif
-
-class AudioCodingModuleImpl : public AudioCodingModule
-{
-public:
- // constructor
- AudioCodingModuleImpl(
- const WebRtc_Word32 id);
-
- // destructor
- ~AudioCodingModuleImpl();
-
- // get version information for ACM and all components
- WebRtc_Word32 Version(
- char* version,
- WebRtc_UWord32& remainingBufferInBytes,
- WebRtc_UWord32& position) const;
-
- // change the unique identifier of this object
- virtual WebRtc_Word32 ChangeUniqueId(
- const WebRtc_Word32 id);
-
- // returns the number of milliseconds until the module want
- // a worker thread to call Process
- WebRtc_Word32 TimeUntilNextProcess();
-
- // Process any pending tasks such as timeouts
- WebRtc_Word32 Process();
-
- // used in conference to go to and from active encoding, hence
- // in and out of mix
- WebRtc_Word32 SetMode(
- const bool passive);
-
-
-
- /////////////////////////////////////////
- // Sender
- //
-
- // initialize send codec
- WebRtc_Word32 InitializeSender();
-
- // reset send codec
- WebRtc_Word32 ResetEncoder();
-
- // can be called multiple times for Codec, CNG, RED
- WebRtc_Word32 RegisterSendCodec(
- const CodecInst& sendCodec);
-
- // get current send codec
- WebRtc_Word32 SendCodec(
- CodecInst& currentSendCodec) const;
-
- // get current send freq
- WebRtc_Word32 SendFrequency() const;
-
- // Get encode bitrate
- // Adaptive rate codecs return their current encode target rate, while other codecs
- // return there longterm avarage or their fixed rate.
- WebRtc_Word32 SendBitrate() const;
-
- // set available bandwidth, inform the encoder about the
- // estimated bandwidth received from the remote party
- virtual WebRtc_Word32 SetReceivedEstimatedBandwidth(
- const WebRtc_Word32 bw);
-
- // register a transport callback which will be
- // called to deliver the encoded buffers
- WebRtc_Word32 RegisterTransportCallback(
- AudioPacketizationCallback* transport);
-
- // Used by the module to deliver messages to the codec module/application
- // AVT(DTMF)
- WebRtc_Word32 RegisterIncomingMessagesCallback(
- AudioCodingFeedback* incomingMessagesCallback,
- const ACMCountries cpt);
-
- // Add 10MS of raw (PCM) audio data to the encoder
- WebRtc_Word32 Add10MsData(
- const AudioFrame& audioFrame);
-
- // set background noise mode for NetEQ, on, off or fade
- WebRtc_Word32 SetBackgroundNoiseMode(
- const ACMBackgroundNoiseMode mode);
-
- // get current background noise mode
- WebRtc_Word32 BackgroundNoiseMode(
- ACMBackgroundNoiseMode& mode);
-
- /////////////////////////////////////////
- // (FEC) Forward Error Correction
- //
-
- // configure FEC status i.e on/off
- WebRtc_Word32 SetFECStatus(
- const bool enable);
-
- // Get FEC status
- bool FECStatus() const;
-
- /////////////////////////////////////////
- // (VAD) Voice Activity Detection
- // and
- // (CNG) Comfort Noise Generation
- //
-
- WebRtc_Word32 SetVAD(
- const bool enableDTX = true,
- const bool enableVAD = false,
- const ACMVADMode vadMode = VADNormal);
-
- WebRtc_Word32 VAD(
- bool& dtxEnabled,
- bool& vadEnabled,
- ACMVADMode& vadMode) const;
-
- WebRtc_Word32 RegisterVADCallback(
- ACMVADCallback* vadCallback);
-
- // Get VAD aggressiveness on the incoming stream
- ACMVADMode ReceiveVADMode() const;
-
- // Configure VAD aggressiveness on the incoming stream
- WebRtc_Word16 SetReceiveVADMode(
- const ACMVADMode mode);
-
-
- /////////////////////////////////////////
- // Receiver
- //
-
- // initialize receiver, resets codec database etc
- WebRtc_Word32 InitializeReceiver();
-
- // reset the decoder state
- WebRtc_Word32 ResetDecoder();
-
- // get current receive freq
- WebRtc_Word32 ReceiveFrequency() const;
-
- // get current playout freq
- WebRtc_Word32 PlayoutFrequency() const;
-
- // register possible reveive codecs, can be called multiple times,
- // for codecs, CNG, DTMF, RED
- WebRtc_Word32 RegisterReceiveCodec(
- const CodecInst& receiveCodec);
-
- // get current received codec
- WebRtc_Word32 ReceiveCodec(
- CodecInst& currentReceiveCodec) const;
-
- // incoming packet from network parsed and ready for decode
- WebRtc_Word32 IncomingPacket(
- const WebRtc_Word8* incomingPayload,
- const WebRtc_Word32 payloadLength,
- const WebRtcRTPHeader& rtpInfo);
-
- // Incoming payloads, without rtp-info, the rtp-info will be created in ACM.
- // One usage for this API is when pre-encoded files are pushed in ACM.
- WebRtc_Word32 IncomingPayload(
- const WebRtc_Word8* incomingPayload,
- const WebRtc_Word32 payloadLength,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timestamp = 0);
-
- // Minimum playout dealy (Used for lip-sync)
- WebRtc_Word32 SetMinimumPlayoutDelay(
- const WebRtc_Word32 timeMs);
-
- // configure Dtmf playout status i.e on/off playout the incoming outband Dtmf tone
- WebRtc_Word32 SetDtmfPlayoutStatus(
- const bool enable);
-
- // Get Dtmf playout status
- bool DtmfPlayoutStatus() const;
-
- // Estimate the Bandwidth based on the incoming stream
- // This is also done in the RTP module
- // need this for one way audio where the RTCP send the BW estimate
- WebRtc_Word32 DecoderEstimatedBandwidth() const;
-
- // Set playout mode voice, fax
- WebRtc_Word32 SetPlayoutMode(
- const AudioPlayoutMode mode);
-
- // Get playout mode voice, fax
- AudioPlayoutMode PlayoutMode() const;
-
- // Get playout timestamp
- WebRtc_Word32 PlayoutTimestamp(
- WebRtc_UWord32& timestamp);
-
- // Get 10 milliseconds of raw audio data to play out
- // automatic resample to the requested frequency if > 0
- WebRtc_Word32 PlayoutData10Ms(
- const WebRtc_Word32 desiredFreqHz,
- AudioFrame &audioFrame);
-
-
- /////////////////////////////////////////
- // Statistics
- //
-
- WebRtc_Word32 NetworkStatistics(
- ACMNetworkStatistics& statistics) const;
-
- void DestructEncoderInst(void* ptrInst);
-
- WebRtc_Word16 AudioBuffer(WebRtcACMAudioBuff& audioBuff);
-
- // GET RED payload for iSAC. The method id called
- // when 'this' ACM is default ACM.
- WebRtc_Word32 REDPayloadISAC(
- const WebRtc_Word32 isacRate,
- const WebRtc_Word16 isacBwEstimate,
- WebRtc_UWord8* payload,
- WebRtc_Word16* payloadLenByte);
-
- WebRtc_Word16 SetAudioBuffer(WebRtcACMAudioBuff& audioBuff);
-
- WebRtc_UWord32 EarliestTimestamp() const;
-
- WebRtc_Word32 LastEncodedTimestamp(WebRtc_UWord32& timestamp) const;
-
- WebRtc_Word32 ReplaceInternalDTXWithWebRtc(
- const bool useWebRtcDTX);
-
- WebRtc_Word32 IsInternalDTXReplacedWithWebRtc(
- bool& usesWebRtcDTX);
-
- WebRtc_Word32 SetISACMaxRate(
- const WebRtc_UWord32 rateBitPerSec);
-
- WebRtc_Word32 SetISACMaxPayloadSize(
- const WebRtc_UWord16 payloadLenBytes);
-
- WebRtc_Word32 ConfigISACBandwidthEstimator(
- const WebRtc_UWord8 initFrameSizeMsec,
- const WebRtc_UWord16 initRateBitPerSec,
- const bool enforceFrameSize = false);
-
- WebRtc_Word32 UnregisterReceiveCodec(
- const WebRtc_Word16 payloadType);
-
-protected:
- void UnregisterSendCodec();
-
- WebRtc_Word32 UnregisterReceiveCodecSafe(
- const WebRtc_Word16 codecID);
-
- ACMGenericCodec* CreateCodec(
- const CodecInst& codec);
-
- WebRtc_Word16 DecoderParamByPlType(
- const WebRtc_UWord8 payloadType,
- WebRtcACMCodecParams& codecParams) const;
-
- WebRtc_Word16 DecoderListIDByPlName(
- const char* payloadName,
- const WebRtc_UWord16 sampFreqHz = 0) const;
-
- WebRtc_Word32 InitializeReceiverSafe();
-
- bool HaveValidEncoder(const char* callerName) const;
-
- WebRtc_Word32 RegisterRecCodecMSSafe(
- const CodecInst& receiveCodec,
- WebRtc_Word16 codecId,
- WebRtc_Word16 mirrorId,
- ACMNetEQ::JB jitterBuffer);
-
-private:
- AudioPacketizationCallback* _packetizationCallback;
- WebRtc_Word32 _id;
- WebRtc_UWord32 _lastTimestamp;
- WebRtc_UWord32 _lastInTimestamp;
- CodecInst _sendCodecInst;
- CodecInst _cngNB;
- CodecInst _cngWB;
- CodecInst _cngSWB;
- CodecInst _RED;
- CodecInst _DTMF;
- bool _vadEnabled;
- bool _dtxEnabled;
- ACMVADMode _vadMode;
- ACMGenericCodec* _codecs[ACMCodecDB::kMaxNumCodecs];
- ACMGenericCodec* _slaveCodecs[ACMCodecDB::kMaxNumCodecs];
- WebRtc_Word16 _mirrorCodecIdx[ACMCodecDB::kMaxNumCodecs];
- bool _stereoReceive[ACMCodecDB::kMaxNumCodecs];
- bool _stereoSend;
- int _prev_received_channel;
- int _expected_channels;
- WebRtc_Word32 _currentSendCodecIdx;
- bool _sendCodecRegistered;
- ACMResampler _inputResampler;
- ACMResampler _outputResampler;
- ACMNetEQ _netEq;
- CriticalSectionWrapper* _acmCritSect;
- ACMVADCallback* _vadCallback;
- WebRtc_UWord8 _lastRecvAudioCodecPlType;
-
- // RED/FEC
- bool _isFirstRED;
- bool _fecEnabled;
- WebRtc_UWord8* _redBuffer;
- RTPFragmentationHeader* _fragmentation;
- WebRtc_UWord32 _lastFECTimestamp;
- WebRtc_UWord8 _redPayloadType;
- // if no RED is registered as receive codec this
- // will have an invalid value.
- WebRtc_UWord8 _receiveREDPayloadType;
-
- // This is to keep track of CN instances where we can send DTMFs
- WebRtc_UWord8 _previousPayloadType;
-
- // This keeps track of payload types associated with _codecs[].
- // We define it as signed variable and initialize with -1 to indicate
- // unused elements.
- WebRtc_Word16 _registeredPlTypes[ACMCodecDB::kMaxNumCodecs];
-
- // Used when payloads are pushed into ACM without any RTP info
- // One example is when pre-encoded bit-stream is pushed from
- // a file.
- WebRtcRTPHeader* _dummyRTPHeader;
- WebRtc_UWord16 _recvPlFrameSizeSmpls;
-
- bool _receiverInitialized;
- ACMDTMFDetection* _dtmfDetector;
-
- AudioCodingFeedback* _dtmfCallback;
- WebRtc_Word16 _lastDetectedTone;
- CriticalSectionWrapper* _callbackCritSect;
-#ifdef TIMED_LOGGING
- TimedTrace _trace;
-#endif
-
- AudioFrame _audioFrame;
-
-#ifdef ACM_QA_TEST
- FILE* _outgoingPL;
- FILE* _incomingPL;
-#endif
-
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CODING_MAIN_SOURCE_AUDIO_CODING_MODULE_IMPL_H_
diff --git a/trunk/src/modules/audio_coding/main/test/ACMTest.cc b/trunk/src/modules/audio_coding/main/test/ACMTest.cc
deleted file mode 100644
index 1bbac0e..0000000
--- a/trunk/src/modules/audio_coding/main/test/ACMTest.cc
+++ /dev/null
@@ -1,16 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "ACMTest.h"
-
-ACMTest::~ACMTest()
-{
-}
-
diff --git a/trunk/src/modules/audio_coding/main/test/ACMTest.h b/trunk/src/modules/audio_coding/main/test/ACMTest.h
deleted file mode 100644
index e965671..0000000
--- a/trunk/src/modules/audio_coding/main/test/ACMTest.h
+++ /dev/null
@@ -1,21 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef ACMTEST_H
-#define ACMTEST_H
-
-class ACMTest
-{
-public:
- virtual ~ACMTest() =0;
- virtual void Perform() =0;
-};
-
-#endif
diff --git a/trunk/src/modules/audio_coding/main/test/APITest.cc b/trunk/src/modules/audio_coding/main/test/APITest.cc
deleted file mode 100644
index ed44178..0000000
--- a/trunk/src/modules/audio_coding/main/test/APITest.cc
+++ /dev/null
@@ -1,1569 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cctype>
-#include <iostream>
-#include <ostream>
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "APITest.h"
-#include "common_types.h"
-#include "engine_configurations.h"
-#include "event_wrapper.h"
-#include "gtest/gtest.h"
-#include "thread_wrapper.h"
-#include "tick_util.h"
-#include "testsupport/fileutils.h"
-#include "trace.h"
-#include "utility.h"
-
-namespace webrtc {
-
-#define TEST_DURATION_SEC 600
-
-#define NUMBER_OF_SENDER_TESTS 6
-
-#define MAX_FILE_NAME_LENGTH_BYTE 500
-#define CHECK_THREAD_NULLITY(myThread, S) \
- if(myThread != NULL) \
- { \
- unsigned int i; \
- (myThread)->Start(i); \
- } \
- else \
- { \
- ADD_FAILURE() << S; \
- }
-
-
-void
-APITest::Wait(WebRtc_UWord32 waitLengthMs)
-{
- if(_randomTest)
- {
- return;
- }
- else
- {
- EventWrapper* myEvent = EventWrapper::Create();
- myEvent->Wait(waitLengthMs);
- delete myEvent;
- return;
- }
-}
-
-
-
-APITest::APITest():
-_acmA(NULL),
-_acmB(NULL),
-_channel_A2B(NULL),
-_channel_B2A(NULL),
-_writeToFile(true),
-_pullEventA(NULL),
-_pushEventA(NULL),
-_processEventA(NULL),
-_apiEventA(NULL),
-_pullEventB(NULL),
-_pushEventB(NULL),
-_processEventB(NULL),
-_apiEventB(NULL),
-_codecCntrA(0),
-_codecCntrB(0),
-_testCntrA(1),
-_testCntrB(1),
-_thereIsEncoderA(false),
-_thereIsEncoderB(false),
-_thereIsDecoderA(false),
-_thereIsDecoderB(false),
-_sendVADA(false),
-_sendDTXA(false),
-_sendVADModeA(VADNormal),
-_sendVADB(false),
-_sendDTXB(false),
-_sendVADModeB(VADNormal),
-_minDelayA(0),
-_minDelayB(0),
-_dotPositionA(0),
-_dotMoveDirectionA(1),
-_dotPositionB(39),
-_dotMoveDirectionB(-1),
-_dtmfCallback(NULL),
-_vadCallbackA(NULL),
-_vadCallbackB(NULL),
-_apiTestRWLock(*RWLockWrapper::CreateRWLock()),
-_randomTest(false),
-_testNumA(0),
-_testNumB(1)
-{
- int n;
- for( n = 0; n < 32; n++)
- {
- _payloadUsed[n] = false;
- }
-
- for(n = 0; n < 3; n++)
- {
- _receiveVADActivityA[n] = 0;
- _receiveVADActivityB[n] = 0;
- }
-
- _movingDot[40] = '\0';
-
- for(int n = 0; n <40; n++)
- {
- _movingDot[n] = ' ';
- }
-}
-
-APITest::~APITest()
-{
- DESTROY_ACM(_acmA);
- DESTROY_ACM(_acmB);
-
- DELETE_POINTER(_channel_A2B);
- DELETE_POINTER(_channel_B2A);
-
- DELETE_POINTER(_pushEventA);
- DELETE_POINTER(_pullEventA);
- DELETE_POINTER(_processEventA);
- DELETE_POINTER(_apiEventA);
-
- DELETE_POINTER(_pushEventB);
- DELETE_POINTER(_pullEventB);
- DELETE_POINTER(_processEventB);
- DELETE_POINTER(_apiEventB);
-
- _inFileA.Close();
- _outFileA.Close();
-
- _inFileB.Close();
- _outFileB.Close();
-
- DELETE_POINTER(_dtmfCallback);
- DELETE_POINTER(_vadCallbackA);
- DELETE_POINTER(_vadCallbackB);
-
- delete &_apiTestRWLock;
-}
-
-
-
-//WebRtc_Word16
-//APITest::SetInFile(char* fileName, WebRtc_UWord16 frequencyHz)
-//{
-// return _inFile.Open(fileName, frequencyHz, "rb");
-//}
-//
-//WebRtc_Word16
-//APITest::SetOutFile(char* fileName, WebRtc_UWord16 frequencyHz)
-//{
-// return _outFile.Open(fileName, frequencyHz, "wb");
-//}
-
-WebRtc_Word16
-APITest::SetUp()
-{
- _acmA = AudioCodingModule::Create(1);
- _acmB = AudioCodingModule::Create(2);
-
- CodecInst dummyCodec;
- int lastPayloadType = 0;
-
- WebRtc_Word16 numCodecs = _acmA->NumberOfCodecs();
- for(WebRtc_UWord8 n = 0; n < numCodecs; n++)
- {
- AudioCodingModule::Codec(n, dummyCodec);
- if((STR_CASE_CMP(dummyCodec.plname, "CN") == 0) &&
- (dummyCodec.plfreq == 32000))
- {
- continue;
- }
-
- printf("Register Receive Codec %s ", dummyCodec.plname);
-
- if((n != 0) && !FixedPayloadTypeCodec(dummyCodec.plname))
- {
- // Check registration with an already occupied payload type
- int currentPayloadType = dummyCodec.pltype;
- dummyCodec.pltype = 97; //lastPayloadType;
- CHECK_ERROR(_acmB->RegisterReceiveCodec(dummyCodec));
- dummyCodec.pltype = currentPayloadType;
- }
-
- if((n < numCodecs - 1) && !FixedPayloadTypeCodec(dummyCodec.plname))
- {
- // test if re-registration works;
- CodecInst nextCodec;
- int currentPayloadType = dummyCodec.pltype;
- AudioCodingModule::Codec(n + 1, nextCodec);
- dummyCodec.pltype = nextCodec.pltype;
- if(!FixedPayloadTypeCodec(nextCodec.plname))
- {
- _acmB->RegisterReceiveCodec(dummyCodec);
- }
- dummyCodec.pltype = currentPayloadType;
- }
-
- if((n < numCodecs - 1) && !FixedPayloadTypeCodec(dummyCodec.plname))
- {
- // test if un-registration works;
- CodecInst nextCodec;
- AudioCodingModule::Codec(n + 1, nextCodec);
- nextCodec.pltype = dummyCodec.pltype;
- if(!FixedPayloadTypeCodec(nextCodec.plname))
- {
- CHECK_ERROR_MT(_acmA->RegisterReceiveCodec(nextCodec));
- CHECK_ERROR_MT(_acmA->UnregisterReceiveCodec(nextCodec.pltype));
- }
- }
-
-
- CHECK_ERROR_MT(_acmA->RegisterReceiveCodec(dummyCodec));
- printf(" side A done!");
- CHECK_ERROR_MT(_acmB->RegisterReceiveCodec(dummyCodec));
- printf(" side B done!\n");
-
- if(!strcmp(dummyCodec.plname, "CN"))
- {
- CHECK_ERROR_MT(_acmA->RegisterSendCodec(dummyCodec));
- CHECK_ERROR_MT(_acmB->RegisterSendCodec(dummyCodec));
- }
- lastPayloadType = dummyCodec.pltype;
- if((lastPayloadType >= 96) && (lastPayloadType <= 127))
- {
- _payloadUsed[lastPayloadType - 96] = true;
- }
- }
- _thereIsDecoderA = true;
- _thereIsDecoderB = true;
-
- // Register Send Codec
- AudioCodingModule::Codec((WebRtc_UWord8)_codecCntrA, dummyCodec);
- CHECK_ERROR_MT(_acmA->RegisterSendCodec(dummyCodec));
- _thereIsEncoderA = true;
- //
- AudioCodingModule::Codec((WebRtc_UWord8)_codecCntrB, dummyCodec);
- CHECK_ERROR_MT(_acmB->RegisterSendCodec(dummyCodec));
- _thereIsEncoderB = true;
-
- char fileName[500];
- WebRtc_UWord16 frequencyHz;
-
- printf("\n\nAPI Test\n");
- printf("========\n");
- printf("Hit enter to accept the default values indicated in []\n\n");
-
- //--- Input A
- strcpy(fileName, "./test/data/audio_coding/testfile32kHz.pcm");
- frequencyHz = 32000;
- printf("Enter input file at side A [%s]: ", fileName);
- PCMFile::ChooseFile(fileName, 499, &frequencyHz);
- _inFileA.Open(fileName, frequencyHz, "rb", true);
-
- //--- Output A
- std::string outputFileA = webrtc::test::OutputPath() + "outA.pcm";
- strcpy(fileName, outputFileA.c_str());
- printf("Enter output file at side A [%s]: ", fileName);
- PCMFile::ChooseFile(fileName, 499, &frequencyHz);
- _outFileA.Open(fileName, frequencyHz, "wb");
-
- //--- Input B
- strcpy(fileName, "./test/data/audio_coding/testfile32kHz.pcm");
- printf("\n\nEnter input file at side B [%s]: ", fileName);
- PCMFile::ChooseFile(fileName, 499, &frequencyHz);
- _inFileB.Open(fileName, frequencyHz, "rb", true);
-
- //--- Output B
- std::string outputFileB = webrtc::test::OutputPath() + "outB.pcm";
- strcpy(fileName, outputFileB.c_str());
- printf("Enter output file at side B [%s]: ", fileName);
- PCMFile::ChooseFile(fileName, 499, &frequencyHz);
- _outFileB.Open(fileName, frequencyHz, "wb");
-
- //--- Set A-to-B channel
- _channel_A2B = new Channel(2);
- CHECK_ERROR_MT(_acmA->RegisterTransportCallback(_channel_A2B));
- _channel_A2B->RegisterReceiverACM(_acmB);
-
- //--- Set B-to-A channel
- _channel_B2A = new Channel(1);
- CHECK_ERROR_MT(_acmB->RegisterTransportCallback(_channel_B2A));
- _channel_B2A->RegisterReceiverACM(_acmA);
-
- //--- EVENT TIMERS
- // A
- _pullEventA = EventWrapper::Create();
- _pushEventA = EventWrapper::Create();
- _processEventA = EventWrapper::Create();
- _apiEventA = EventWrapper::Create();
- // B
- _pullEventB = EventWrapper::Create();
- _pushEventB = EventWrapper::Create();
- _processEventB = EventWrapper::Create();
- _apiEventB = EventWrapper::Create();
-
- //--- I/O params
- // A
- _outFreqHzA = _outFileA.SamplingFrequency();
- // B
- _outFreqHzB = _outFileB.SamplingFrequency();
-
-
- //Trace::SetEncryptedTraceFile("ACMAPITestEncrypted.txt");
-
- char print[11];
-
- printf("\nRandom Test (y/n)?");
- EXPECT_TRUE(fgets(print, 10, stdin) != NULL);
- print[10] = '\0';
- if(strstr(print, "y") != NULL)
- {
- _randomTest = true;
- _verbose = false;
- _writeToFile = false;
- Trace::CreateTrace();
- Trace::SetTraceFile("ACMAPITest.txt");
- //freopen("APITest_log.txt", "w", stdout);
- }
- else
- {
- Trace::CreateTrace();
- Trace::SetTraceFile("ACMAPITest.txt", true);
- _randomTest = false;
- printf("\nPrint Tests (y/n)? ");
- EXPECT_TRUE(fgets(print, 10, stdin) != NULL);
- print[10] = '\0';
- if(strstr(print, "y") == NULL)
- {
- EXPECT_TRUE(freopen("APITest_log.txt", "w", stdout) != 0);
- _verbose = false;
- }
- }
-
-#ifdef WEBRTC_DTMF_DETECTION
- _dtmfCallback = new DTMFDetector;
-#endif
- _vadCallbackA = new VADCallback;
- _vadCallbackB = new VADCallback;
-
- return 0;
-}
-
-bool
-APITest::PushAudioThreadA(void* obj)
-{
- return static_cast<APITest*>(obj)->PushAudioRunA();
-}
-
-bool
-APITest::PushAudioThreadB(void* obj)
-{
- return static_cast<APITest*>(obj)->PushAudioRunB();
-}
-
-bool
-APITest::PullAudioThreadA(void* obj)
-{
- return static_cast<APITest*>(obj)->PullAudioRunA();
-}
-
-bool
-APITest::PullAudioThreadB(void* obj)
-{
- return static_cast<APITest*>(obj)->PullAudioRunB();
-}
-
-bool
-APITest::ProcessThreadA(void* obj)
-{
- return static_cast<APITest*>(obj)->ProcessRunA();
-}
-
-bool
-APITest::ProcessThreadB(void* obj)
-{
- return static_cast<APITest*>(obj)->ProcessRunB();
-}
-
-bool
-APITest::APIThreadA(void* obj)
-{
- return static_cast<APITest*>(obj)->APIRunA();
-}
-
-bool
-APITest::APIThreadB(void* obj)
-{
- return static_cast<APITest*>(obj)->APIRunB();
-}
-
-bool
-APITest::PullAudioRunA()
-{
- _pullEventA->Wait(100);
- AudioFrame audioFrame;
- if(_acmA->PlayoutData10Ms(_outFreqHzA, audioFrame) < 0)
- {
- bool thereIsDecoder;
- {
- ReadLockScoped rl(_apiTestRWLock);
- thereIsDecoder = _thereIsDecoderA;
- }
- if(thereIsDecoder)
- {
- fprintf(stderr, "\n>>>>>> cannot pull audio A <<<<<<<< \n");
- }
- }
- else
- {
- if(_writeToFile)
- {
- _outFileA.Write10MsData(audioFrame);
- }
- _receiveVADActivityA[(int)audioFrame._vadActivity]++;
- }
- return true;
-}
-
-bool
-APITest::PullAudioRunB()
-{
- _pullEventB->Wait(100);
- AudioFrame audioFrame;
- if(_acmB->PlayoutData10Ms(_outFreqHzB, audioFrame) < 0)
- {
- bool thereIsDecoder;
- {
- ReadLockScoped rl(_apiTestRWLock);
- thereIsDecoder = _thereIsDecoderB;
- }
- if(thereIsDecoder)
- {
- fprintf(stderr, "\n>>>>>> cannot pull audio B <<<<<<<< \n");
- fprintf(stderr, "%d %d\n", _testNumA, _testNumB);
- }
- }
- else
- {
- if(_writeToFile)
- {
- _outFileB.Write10MsData(audioFrame);
- }
- _receiveVADActivityB[(int)audioFrame._vadActivity]++;
- }
- return true;
-}
-
-bool
-APITest::PushAudioRunA()
-{
- _pushEventA->Wait(100);
- AudioFrame audioFrame;
- _inFileA.Read10MsData(audioFrame);
- if(_acmA->Add10MsData(audioFrame) < 0)
- {
- bool thereIsEncoder;
- {
- ReadLockScoped rl(_apiTestRWLock);
- thereIsEncoder = _thereIsEncoderA;
- }
- if(thereIsEncoder)
- {
- fprintf(stderr, "\n>>>> add10MsData at A failed <<<<\n");
- }
- }
- return true;
-}
-
-bool
-APITest::PushAudioRunB()
-{
- _pushEventB->Wait(100);
- AudioFrame audioFrame;
- _inFileB.Read10MsData(audioFrame);
- if(_acmB->Add10MsData(audioFrame) < 0)
- {
- bool thereIsEncoder;
- {
- ReadLockScoped rl(_apiTestRWLock);
- thereIsEncoder = _thereIsEncoderB;
- }
-
- if(thereIsEncoder)
- {
- fprintf(stderr, "\n>>>> cannot add audio to B <<<<");
- }
- }
-
- return true;
-}
-
-bool
-APITest::ProcessRunA()
-{
- _processEventA->Wait(100);
- if(_acmA->Process() < 0)
- {
- // do not print error message if there is no encoder
- bool thereIsEncoder;
- {
- ReadLockScoped rl(_apiTestRWLock);
- thereIsEncoder = _thereIsEncoderA;
- }
-
- if(thereIsEncoder)
- {
- fprintf(stderr, "\n>>>>> Process Failed at A <<<<<\n");
- }
- }
- return true;
-}
-
-bool
-APITest::ProcessRunB()
-{
- _processEventB->Wait(100);
- if(_acmB->Process() < 0)
- {
- bool thereIsEncoder;
- {
- ReadLockScoped rl(_apiTestRWLock);
- thereIsEncoder = _thereIsEncoderB;
- }
- if(thereIsEncoder)
- {
- fprintf(stderr, "\n>>>>> Process Failed at B <<<<<\n");
- }
- }
- return true;
-}
-
-/*/
- *
- * In side A we test the APIs which are related to sender Side.
- *
-/*/
-
-
-void
-APITest::RunTest(char thread)
-{
- int testNum;
- {
- WriteLockScoped cs(_apiTestRWLock);
- if(thread == 'A')
- {
- _testNumA = (_testNumB + 1 + (rand() % 6)) % 7;
- testNum = _testNumA;
-
- _movingDot[_dotPositionA] = ' ';
- if(_dotPositionA == 0)
- {
- _dotMoveDirectionA = 1;
- }
- if(_dotPositionA == 19)
- {
- _dotMoveDirectionA = -1;
- }
- _dotPositionA += _dotMoveDirectionA;
- _movingDot[_dotPositionA] = (_dotMoveDirectionA > 0)? '>':'<';
- }
- else
- {
- _testNumB = (_testNumA + 1 + (rand() % 6)) % 7;
- testNum = _testNumB;
-
- _movingDot[_dotPositionB] = ' ';
- if(_dotPositionB == 20)
- {
- _dotMoveDirectionB = 1;
- }
- if(_dotPositionB == 39)
- {
- _dotMoveDirectionB = -1;
- }
- _dotPositionB += _dotMoveDirectionB;
- _movingDot[_dotPositionB] = (_dotMoveDirectionB > 0)? '>':'<';
- }
- //fprintf(stderr, "%c: %d \n", thread, testNum);
- //fflush(stderr);
- }
- switch(testNum)
- {
- case 0:
- CurrentCodec('A');
- ChangeCodec('A');
- break;
- case 1:
- TestPlayout('B');
- break;
- case 2:
- if(!_randomTest)
- {
- fprintf(stdout, "\nTesting Delay ...\n");
- }
- TestDelay('A');
- break;
- case 3:
- TestSendVAD('A');
- break;
- case 4:
- TestRegisteration('A');
- break;
- case 5:
- TestReceiverVAD('A');
- break;
- case 6:
-#ifdef WEBRTC_DTMF_DETECTION
- LookForDTMF('A');
-#endif
- break;
- default:
- fprintf(stderr, "Wrong Test Number\n");
- getchar();
- exit(1);
- }
-}
-
-
-
-bool
-APITest::APIRunA()
-{
- _apiEventA->Wait(50);
-
- bool randomTest;
- {
- ReadLockScoped rl(_apiTestRWLock);
- randomTest = _randomTest;
- }
- if(randomTest)
- {
- RunTest('A');
- }
- else
- {
- CurrentCodec('A');
- ChangeCodec('A');
- TestPlayout('B');
- if(_codecCntrA == 0)
- {
- fprintf(stdout, "\nTesting Delay ...\n");
- TestDelay('A');
- }
- // VAD TEST
- TestSendVAD('A');
- TestRegisteration('A');
- TestReceiverVAD('A');
-#ifdef WEBRTC_DTMF_DETECTION
- LookForDTMF('A');
-#endif
- }
- return true;
-}
-
-bool
-APITest::APIRunB()
-{
- _apiEventB->Wait(50);
- bool randomTest;
- {
- ReadLockScoped rl(_apiTestRWLock);
- randomTest = _randomTest;
- }
- //_apiEventB->Wait(2000);
- if(randomTest)
- {
- RunTest('B');
- }
-
- return true;
-}
-
-void
-APITest::Perform()
-{
- SetUp();
-
- //--- THREADS
- // A
- // PUSH
- ThreadWrapper* myPushAudioThreadA = ThreadWrapper::CreateThread(PushAudioThreadA,
- this, kNormalPriority, "PushAudioThreadA");
- CHECK_THREAD_NULLITY(myPushAudioThreadA, "Unable to start A::PUSH thread");
- // PULL
- ThreadWrapper* myPullAudioThreadA = ThreadWrapper::CreateThread(PullAudioThreadA,
- this, kNormalPriority, "PullAudioThreadA");
- CHECK_THREAD_NULLITY(myPullAudioThreadA, "Unable to start A::PULL thread");
- // Process
- ThreadWrapper* myProcessThreadA = ThreadWrapper::CreateThread(ProcessThreadA,
- this, kNormalPriority, "ProcessThreadA");
- CHECK_THREAD_NULLITY(myProcessThreadA, "Unable to start A::Process thread");
- // API
- ThreadWrapper* myAPIThreadA = ThreadWrapper::CreateThread(APIThreadA,
- this, kNormalPriority, "APIThreadA");
- CHECK_THREAD_NULLITY(myAPIThreadA, "Unable to start A::API thread");
- // B
- // PUSH
- ThreadWrapper* myPushAudioThreadB = ThreadWrapper::CreateThread(PushAudioThreadB,
- this, kNormalPriority, "PushAudioThreadB");
- CHECK_THREAD_NULLITY(myPushAudioThreadB, "Unable to start B::PUSH thread");
- // PULL
- ThreadWrapper* myPullAudioThreadB = ThreadWrapper::CreateThread(PullAudioThreadB,
- this, kNormalPriority, "PullAudioThreadB");
- CHECK_THREAD_NULLITY(myPullAudioThreadB, "Unable to start B::PULL thread");
- // Process
- ThreadWrapper* myProcessThreadB = ThreadWrapper::CreateThread(ProcessThreadB,
- this, kNormalPriority, "ProcessThreadB");
- CHECK_THREAD_NULLITY(myProcessThreadB, "Unable to start B::Process thread");
- // API
- ThreadWrapper* myAPIThreadB = ThreadWrapper::CreateThread(APIThreadB,
- this, kNormalPriority, "APIThreadB");
- CHECK_THREAD_NULLITY(myAPIThreadB, "Unable to start B::API thread");
-
-
- //_apiEventA->StartTimer(true, 5000);
- //_apiEventB->StartTimer(true, 5000);
-
- _processEventA->StartTimer(true, 10);
- _processEventB->StartTimer(true, 10);
-
- _pullEventA->StartTimer(true, 10);
- _pullEventB->StartTimer(true, 10);
-
- _pushEventA->StartTimer(true, 10);
- _pushEventB->StartTimer(true, 10);
-
- // Keep main thread waiting for sender/receiver
- // threads to complete
- EventWrapper* completeEvent = EventWrapper::Create();
- WebRtc_UWord64 startTime = TickTime::MillisecondTimestamp();
- WebRtc_UWord64 currentTime;
- do
- {
- {
- //ReadLockScoped rl(_apiTestRWLock);
- //fprintf(stderr, "\r%s", _movingDot);
- }
- //fflush(stderr);
- completeEvent->Wait(50);
- currentTime = TickTime::MillisecondTimestamp();
- } while((currentTime - startTime) < 120000); // Run test in 2 minutes (120000 ms)
-
- //completeEvent->Wait(0xFFFFFFFF);//(unsigned long)((unsigned long)TEST_DURATION_SEC * (unsigned long)1000));
- delete completeEvent;
-
- myPushAudioThreadA->Stop();
- myPullAudioThreadA->Stop();
- myProcessThreadA->Stop();
- myAPIThreadA->Stop();
-
- delete myPushAudioThreadA;
- delete myPullAudioThreadA;
- delete myProcessThreadA;
- delete myAPIThreadA;
-
-
- myPushAudioThreadB->Stop();
- myPullAudioThreadB->Stop();
- myProcessThreadB->Stop();
- myAPIThreadB->Stop();
-
- delete myPushAudioThreadB;
- delete myPullAudioThreadB;
- delete myProcessThreadB;
- delete myAPIThreadB;
-}
-
-
-void
-APITest::CheckVADStatus(char side)
-{
-
- bool dtxEnabled;
- bool vadEnabled;
- ACMVADMode vadMode;
-
- if(side == 'A')
- {
- _acmA->VAD(dtxEnabled, vadEnabled, vadMode);
- _acmA->RegisterVADCallback(NULL);
- _vadCallbackA->Reset();
- _acmA->RegisterVADCallback(_vadCallbackA);
-
- if(!_randomTest)
- {
- if(_verbose)
- {
- fprintf(stdout, "DTX %3s, VAD %3s, Mode %d",
- dtxEnabled? "ON":"OFF",
- vadEnabled? "ON":"OFF",
- (int)vadMode);
- Wait(5000);
- fprintf(stdout, " => bit-rate %3.0f kbps\n",
- _channel_A2B->BitRate());
- }
- else
- {
- Wait(5000);
- fprintf(stdout, "DTX %3s, VAD %3s, Mode %d => bit-rate %3.0f kbps\n",
- dtxEnabled? "ON":"OFF",
- vadEnabled? "ON":"OFF",
- (int)vadMode,
- _channel_A2B->BitRate());
- }
- _vadCallbackA->PrintFrameTypes();
- }
-
- if(dtxEnabled != _sendDTXA)
- {
- fprintf(stderr, ">>> Error Enabling DTX <<<\n");
- }
- if((vadEnabled != _sendVADA) && (!dtxEnabled))
- {
- fprintf(stderr, ">>> Error Enabling VAD <<<\n");
- }
- if((vadMode != _sendVADModeA) && vadEnabled)
- {
- fprintf(stderr, ">>> Error setting VAD-mode <<<\n");
- }
- }
- else
- {
- _acmB->VAD(dtxEnabled, vadEnabled, vadMode);
-
- _acmB->RegisterVADCallback(NULL);
- _vadCallbackB->Reset();
- _acmB->RegisterVADCallback(_vadCallbackB);
-
- if(!_randomTest)
- {
- if(_verbose)
- {
- fprintf(stdout, "DTX %3s, VAD %3s, Mode %d",
- dtxEnabled? "ON":"OFF",
- vadEnabled? "ON":"OFF",
- (int)vadMode);
- Wait(5000);
- fprintf(stdout, " => bit-rate %3.0f kbps\n",
- _channel_B2A->BitRate());
- }
- else
- {
- Wait(5000);
- fprintf(stdout, "DTX %3s, VAD %3s, Mode %d => bit-rate %3.0f kbps\n",
- dtxEnabled? "ON":"OFF",
- vadEnabled? "ON":"OFF",
- (int)vadMode,
- _channel_B2A->BitRate());
- }
- _vadCallbackB->PrintFrameTypes();
- }
-
- if(dtxEnabled != _sendDTXB)
- {
- fprintf(stderr, ">>> Error Enabling DTX <<<\n");
- }
- if((vadEnabled != _sendVADB) && (!dtxEnabled))
- {
- fprintf(stderr, ">>> Error Enabling VAD <<<\n");
- }
- if((vadMode != _sendVADModeB) && vadEnabled)
- {
- fprintf(stderr, ">>> Error setting VAD-mode <<<\n");
- }
- }
-}
-
-// Set Min delay, get delay, playout timestamp
-void
-APITest::TestDelay(char side)
-{
- AudioCodingModule* myACM;
- Channel* myChannel;
- WebRtc_Word32* myMinDelay;
- EventWrapper* myEvent = EventWrapper::Create();
-
- WebRtc_UWord32 inTimestamp = 0;
- WebRtc_UWord32 outTimestamp = 0;
- double estimDelay = 0;
-
- double averageEstimDelay = 0;
- double averageDelay = 0;
-
- CircularBuffer estimDelayCB(100);
- estimDelayCB.SetArithMean(true);
-
- if(side == 'A')
- {
- myACM = _acmA;
- myChannel = _channel_B2A;
- myMinDelay = &_minDelayA;
- }
- else
- {
- myACM = _acmB;
- myChannel = _channel_A2B;
- myMinDelay = &_minDelayB;
- }
-
-
- CHECK_ERROR_MT(myACM->SetMinimumPlayoutDelay(*myMinDelay));
-
-
- inTimestamp = myChannel->LastInTimestamp();
- CHECK_ERROR_MT(myACM->PlayoutTimestamp(outTimestamp));
-
- if(!_randomTest)
- {
- myEvent->StartTimer(true, 30);
- int n = 0;
- int settlePoint = 5000;
- while(n < settlePoint + 400)
- {
- myEvent->Wait(1000);
-
- inTimestamp = myChannel->LastInTimestamp();
- CHECK_ERROR_MT(myACM->PlayoutTimestamp(outTimestamp));
-
- //std::cout << outTimestamp << std::endl << std::flush;
- estimDelay = (double)((WebRtc_UWord32)(inTimestamp - outTimestamp)) /
- ((double)myACM->ReceiveFrequency() / 1000.0);
-
- estimDelayCB.Update(estimDelay);
-
- estimDelayCB.ArithMean(averageEstimDelay);
- //printf("\n %6.1f \n", estimDelay);
- //std::cout << " " << std::flush;
-
- if(_verbose)
- {
- fprintf(stdout, "\rExpected: %4d, retreived: %6.1f, measured: %6.1f",
- *myMinDelay, averageDelay, averageEstimDelay);
- std::cout << " " << std::flush;
- }
- if((averageDelay > *myMinDelay) && (n < settlePoint))
- {
- settlePoint = n;
- }
- n++;
- }
- myEvent->StopTimer();
- }
-
- if((!_verbose) && (!_randomTest))
- {
- fprintf(stdout, "\nExpected: %4d, retreived: %6.1f, measured: %6.1f",
- *myMinDelay, averageDelay, averageEstimDelay);
- }
-
- *myMinDelay = (rand() % 1000) + 1;
-
- ACMNetworkStatistics networkStat;
- CHECK_ERROR_MT(myACM->NetworkStatistics(networkStat));
-
- if(!_randomTest)
- {
- fprintf(stdout, "\n\nJitter Statistics at Side %c\n", side);
- fprintf(stdout, "--------------------------------------\n");
- fprintf(stdout, "buffer-size............. %d\n", networkStat.currentBufferSize);
- fprintf(stdout, "Preferred buffer-size... %d\n", networkStat.preferredBufferSize);
- fprintf(stdout, "Peaky jitter mode........%d\n", networkStat.jitterPeaksFound);
- fprintf(stdout, "packet-size rate........ %d\n", networkStat.currentPacketLossRate);
- fprintf(stdout, "discard rate............ %d\n", networkStat.currentDiscardRate);
- fprintf(stdout, "expand rate............. %d\n", networkStat.currentExpandRate);
- fprintf(stdout, "Preemptive rate......... %d\n", networkStat.currentPreemptiveRate);
- fprintf(stdout, "Accelerate rate......... %d\n", networkStat.currentAccelerateRate);
- fprintf(stdout, "Clock-drift............. %d\n", networkStat.clockDriftPPM);
- fprintf(stdout, "Mean waiting time....... %d\n", networkStat.meanWaitingTimeMs);
- fprintf(stdout, "Median waiting time..... %d\n", networkStat.medianWaitingTimeMs);
- fprintf(stdout, "Min waiting time........ %d\n", networkStat.minWaitingTimeMs);
- fprintf(stdout, "Max waiting time........ %d\n", networkStat.maxWaitingTimeMs);
- }
-
- CHECK_ERROR_MT(myACM->SetMinimumPlayoutDelay(*myMinDelay));
-
- if(!_randomTest)
- {
- myEvent->Wait(500);
- fprintf(stdout, "\n");
- fprintf(stdout, "\n");
- }
- delete myEvent;
-}
-
-// Unregister a codec & register again.
-void
-APITest::TestRegisteration(char sendSide)
-{
- AudioCodingModule* sendACM;
- AudioCodingModule* receiveACM;
- bool* thereIsDecoder;
- EventWrapper* myEvent = EventWrapper::Create();
-
- if(!_randomTest)
- {
- fprintf(stdout, "\n\n");
- fprintf(stdout, "---------------------------------------------------------\n");
- fprintf(stdout, " Unregister/register Receive Codec\n");
- fprintf(stdout, "---------------------------------------------------------\n");
- }
-
- switch(sendSide)
- {
- case 'A':
- {
- sendACM = _acmA;
- receiveACM = _acmB;
- thereIsDecoder = &_thereIsDecoderB;
- break;
- }
- case 'B':
- {
- sendACM = _acmB;
- receiveACM = _acmA;
- thereIsDecoder = &_thereIsDecoderA;
- break;
- }
- default:
- fprintf(stderr, "Invalid sender-side in TestRegistration(%c)\n", sendSide);
- exit(-1);
- }
-
- CodecInst myCodec;
- if(sendACM->SendCodec(myCodec) < 0)
- {
- AudioCodingModule::Codec(_codecCntrA, myCodec);
- }
-
- if(!_randomTest)
- {
- fprintf(stdout, "Unregistering reveive codec, NO AUDIO.\n");
- fflush(stdout);
- }
- {
- WriteLockScoped wl(_apiTestRWLock);
- *thereIsDecoder = false;
- }
- //myEvent->Wait(20);
- CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec.pltype));
- Wait(1000);
-
- int currentPayload = myCodec.pltype;
-
- if(!FixedPayloadTypeCodec(myCodec.plname))
- {
- WebRtc_Word32 i;
- for(i = 0; i < 32; i++)
- {
- if(!_payloadUsed[i])
- {
- if(!_randomTest)
- {
- fprintf(stdout, "Register receive codec with new Payload, AUDIO BACK.\n");
- }
- //myCodec.pltype = i + 96;
- //CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
- //CHECK_ERROR_MT(sendACM->RegisterSendCodec(myCodec));
- //myEvent->Wait(20);
- //{
- // WriteLockScoped wl(_apiTestRWLock);
- // *thereIsDecoder = true;
- //}
- Wait(1000);
-
- if(!_randomTest)
- {
- fprintf(stdout, "Unregistering reveive codec, NO AUDIO.\n");
- }
- //{
- // WriteLockScoped wl(_apiTestRWLock);
- // *thereIsDecoder = false;
- //}
- //myEvent->Wait(20);
- //CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec.pltype));
- Wait(1000);
-
- myCodec.pltype = currentPayload;
- if(!_randomTest)
- {
- fprintf(stdout, "Register receive codec with default Payload, AUDIO BACK.\n");
- fflush(stdout);
- }
- CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
- //CHECK_ERROR_MT(sendACM->RegisterSendCodec(myCodec));
- myEvent->Wait(20);
- {
- WriteLockScoped wl(_apiTestRWLock);
- *thereIsDecoder = true;
- }
- Wait(1000);
-
- break;
- }
- }
- if(i == 32)
- {
- CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
- {
- WriteLockScoped wl(_apiTestRWLock);
- *thereIsDecoder = true;
- }
- }
- }
- else
- {
- if(!_randomTest)
- {
- fprintf(stdout, "Register receive codec with fixed Payload, AUDIO BACK.\n");
- fflush(stdout);
- }
- CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
- //CHECK_ERROR_MT(receiveACM->UnregisterReceiveCodec(myCodec.pltype));
- //CHECK_ERROR_MT(receiveACM->RegisterReceiveCodec(myCodec));
- myEvent->Wait(20);
- {
- WriteLockScoped wl(_apiTestRWLock);
- *thereIsDecoder = true;
- }
- }
- delete myEvent;
- if(!_randomTest)
- {
- fprintf(stdout, "---------------------------------------------------------\n");
- }
-}
-
-// Playout Mode, background noise mode.
-// Receiver Frequency, playout frequency.
-void
-APITest::TestPlayout(char receiveSide)
-{
- AudioCodingModule* receiveACM;
- AudioPlayoutMode* playoutMode = NULL;
- ACMBackgroundNoiseMode* bgnMode = NULL;
- switch(receiveSide)
- {
- case 'A':
- {
- receiveACM = _acmA;
- playoutMode = &_playoutModeA;
- bgnMode = &_bgnModeA;
- break;
- }
- case 'B':
- {
- receiveACM = _acmB;
- playoutMode = &_playoutModeB;
- bgnMode = &_bgnModeB;
- break;
- }
- default:
- receiveACM = _acmA;
- }
-
- WebRtc_Word32 receiveFreqHz = receiveACM->ReceiveFrequency();
- WebRtc_Word32 playoutFreqHz = receiveACM->PlayoutFrequency();
-
- CHECK_ERROR_MT(receiveFreqHz);
- CHECK_ERROR_MT(playoutFreqHz);
-
- char bgnString[25];
- switch(*bgnMode)
- {
- case On:
- {
- *bgnMode = Fade;
- strncpy(bgnString, "Fade", 25);
- break;
- }
- case Fade:
- {
- *bgnMode = Off;
- strncpy(bgnString, "OFF", 25);
- break;
- }
- case Off:
- {
- *bgnMode = On;
- strncpy(bgnString, "ON", 25);
- break;
- }
- default:
- *bgnMode = On;
- strncpy(bgnString, "ON", 25);
- }
- CHECK_ERROR_MT(receiveACM->SetBackgroundNoiseMode(*bgnMode));
- bgnString[24] = '\0';
-
- char playoutString[25];
- switch(*playoutMode)
- {
- case voice:
- {
- *playoutMode = fax;
- strncpy(playoutString, "FAX", 25);
- break;
- }
- case fax:
- {
- *playoutMode = streaming;
- strncpy(playoutString, "Streaming", 25);
- break;
- }
- case streaming:
- {
- *playoutMode = voice;
- strncpy(playoutString, "Voice", 25);
- break;
- }
- default:
- *playoutMode = voice;
- strncpy(playoutString, "Voice", 25);
- }
- CHECK_ERROR_MT(receiveACM->SetPlayoutMode(*playoutMode));
- playoutString[24] = '\0';
-
- if(!_randomTest)
- {
- fprintf(stdout, "\n");
- fprintf(stdout, "In Side %c\n", receiveSide);
- fprintf(stdout, "---------------------------------\n");
- fprintf(stdout, "Receive Frequency....... %d Hz\n", receiveFreqHz);
- fprintf(stdout, "Playout Frequency....... %d Hz\n", playoutFreqHz);
- fprintf(stdout, "Audio Playout Mode...... %s\n", playoutString);
- fprintf(stdout, "Background Noise Mode... %s\n", bgnString);
- }
-}
-
-// set/get receiver VAD status & mode.
-void
-APITest::TestReceiverVAD(char side)
-{
- AudioCodingModule* myACM;
- int* myReceiveVADActivity;
-
- if(side == 'A')
- {
- myACM = _acmA;
- myReceiveVADActivity = _receiveVADActivityA;
- }
- else
- {
- myACM = _acmB;
- myReceiveVADActivity = _receiveVADActivityB;
- }
-
- ACMVADMode mode = myACM->ReceiveVADMode();
-
- CHECK_ERROR_MT(mode);
-
- if(!_randomTest)
- {
- fprintf(stdout, "\n\nCurrent Receive VAD at side %c\n", side);
- fprintf(stdout, "----------------------------------\n");
- fprintf(stdout, "mode.......... %d\n", (int)mode);
- fprintf(stdout, "VAD Active.... %d\n", myReceiveVADActivity[0]);
- fprintf(stdout, "VAD Passive... %d\n", myReceiveVADActivity[1]);
- fprintf(stdout, "VAD Unknown... %d\n", myReceiveVADActivity[2]);
- }
-
- if(!_randomTest)
- {
- fprintf(stdout, "\nChange Receive VAD at side %c\n\n", side);
- }
-
- switch(mode)
- {
- case VADNormal:
- mode = VADAggr;
- break;
- case VADLowBitrate:
- mode = VADVeryAggr;
- break;
- case VADAggr:
- mode = VADLowBitrate;
- break;
- case VADVeryAggr:
- mode = VADNormal;
- break;
- default:
- mode = VADNormal;
-
- CHECK_ERROR_MT(myACM->SetReceiveVADMode(mode));
- }
- for(int n = 0; n < 3; n++)
- {
- myReceiveVADActivity[n] = 0;
- }
-}
-
-
-void
-APITest::TestSendVAD(char side)
-{
- if(_randomTest)
- {
- return;
- }
-
- bool* vad;
- bool* dtx;
- ACMVADMode* mode;
- Channel* myChannel;
- AudioCodingModule* myACM;
-
- CodecInst myCodec;
- if(!_randomTest)
- {
- fprintf(stdout, "\n\n");
- fprintf(stdout, "-----------------------------------------------\n");
- fprintf(stdout, " Test VAD API\n");
- fprintf(stdout, "-----------------------------------------------\n");
- }
-
- if(side == 'A')
- {
- AudioCodingModule::Codec(_codecCntrA, myCodec);
- vad = &_sendVADA;
- dtx = &_sendDTXA;
- mode = &_sendVADModeA;
- myChannel = _channel_A2B;
- myACM = _acmA;
- }
- else
- {
- AudioCodingModule::Codec(_codecCntrB, myCodec);
- vad = &_sendVADB;
- dtx = &_sendDTXB;
- mode = &_sendVADModeB;
- myChannel = _channel_B2A;
- myACM = _acmB;
- }
-
- CheckVADStatus(side);
- if(!_randomTest)
- {
- fprintf(stdout, "\n\n");
- }
-
- switch(*mode)
- {
- case VADNormal:
- *vad = true;
- *dtx = true;
- *mode = VADAggr;
- break;
- case VADLowBitrate:
- *vad = true;
- *dtx = true;
- *mode = VADVeryAggr;
- break;
- case VADAggr:
- *vad = true;
- *dtx = true;
- *mode = VADLowBitrate;
- break;
- case VADVeryAggr:
- *vad = false;
- *dtx = false;
- *mode = VADNormal;
- break;
- default:
- *mode = VADNormal;
- }
-
- *dtx = (myCodec.plfreq == 32000)? false:*dtx;
-
- CHECK_ERROR_MT(myACM->SetVAD(*dtx, *vad, *mode));
- myChannel->ResetStats();
-
- CheckVADStatus(side);
- if(!_randomTest)
- {
- fprintf(stdout, "\n");
- fprintf(stdout, "-----------------------------------------------\n");
- }
-
- // Fault Test
- CHECK_PROTECTED_MT(myACM->SetVAD(false, true, (ACMVADMode)-1));
- CHECK_PROTECTED_MT(myACM->SetVAD(false, true, (ACMVADMode)4));
-
-
-
-}
-
-
-void
-APITest::CurrentCodec(char side)
-{
- CodecInst myCodec;
- if(side == 'A')
- {
- _acmA->SendCodec(myCodec);
- }
- else
- {
- _acmB->SendCodec(myCodec);
- }
-
- if(!_randomTest)
- {
- fprintf(stdout, "\n\n");
- fprintf(stdout, "Send codec in Side A\n");
- fprintf(stdout, "----------------------------\n");
- fprintf(stdout, "Name................. %s\n", myCodec.plname);
- fprintf(stdout, "Sampling Frequency... %d\n", myCodec.plfreq);
- fprintf(stdout, "Rate................. %d\n", myCodec.rate);
- fprintf(stdout, "Payload-type......... %d\n", myCodec.pltype);
- fprintf(stdout, "Packet-size.......... %d\n", myCodec.pacsize);
- }
-
- Wait(100);
-}
-
-void
-APITest::ChangeCodec(char side)
-{
- CodecInst myCodec;
- AudioCodingModule* myACM;
- WebRtc_UWord8* codecCntr;
- bool* thereIsEncoder;
- bool* vad;
- bool* dtx;
- ACMVADMode* mode;
- Channel* myChannel;
- // Reset and Wait
- if(!_randomTest)
- {
- fprintf(stdout, "Reset Encoder Side A \n");
- }
- if(side == 'A')
- {
- myACM = _acmA;
- codecCntr = &_codecCntrA;
- {
- WriteLockScoped wl(_apiTestRWLock);
- thereIsEncoder = &_thereIsEncoderA;
- }
- vad = &_sendVADA;
- dtx = &_sendDTXA;
- mode = &_sendVADModeA;
- myChannel = _channel_A2B;
- }
- else
- {
- myACM = _acmB;
- codecCntr = &_codecCntrB;
- {
- WriteLockScoped wl(_apiTestRWLock);
- thereIsEncoder = &_thereIsEncoderB;
- }
- vad = &_sendVADB;
- dtx = &_sendDTXB;
- mode = &_sendVADModeB;
- myChannel = _channel_B2A;
- }
-
- myACM->ResetEncoder();
- Wait(100);
-
- // Register the next codec
- do
- {
- *codecCntr = (*codecCntr < AudioCodingModule::NumberOfCodecs() - 1)?
- (*codecCntr + 1):0;
-
- if(*codecCntr == 0)
- {
- //printf("Initialize Sender Side A \n");
- {
- WriteLockScoped wl(_apiTestRWLock);
- *thereIsEncoder = false;
- }
- CHECK_ERROR_MT(myACM->InitializeSender());
- Wait(1000);
-
- // After Initialization CN is lost, re-register them
- if(AudioCodingModule::Codec("CN", myCodec, 8000) >= 0)
- {
- CHECK_ERROR_MT(myACM->RegisterSendCodec(myCodec));
- }
- if(AudioCodingModule::Codec("CN", myCodec, 16000) >= 0)
- {
- CHECK_ERROR_MT(myACM->RegisterSendCodec(myCodec));
- }
- // VAD & DTX are disabled after initialization
- *vad = false;
- *dtx = false;
- _writeToFile = false;
- }
-
- AudioCodingModule::Codec(*codecCntr, myCodec);
- } while(!STR_CASE_CMP(myCodec.plname, "CN") ||
- !STR_CASE_CMP(myCodec.plname, "telephone-event") ||
- !STR_CASE_CMP(myCodec.plname, "RED"));
-
- if(!_randomTest)
- {
- fprintf(stdout, "\n====================================================================\n");
- fprintf(stdout, " Registering New Codec %s, %d kHz, %d kbps\n",
- myCodec.plname, myCodec.plfreq / 1000, myCodec.rate / 1000);
- }
- //std::cout<< std::flush;
-
- // NO DTX for supe-wideband codec at this point
- if(myCodec.plfreq == 32000)
- {
- *dtx = false;
- CHECK_ERROR_MT(myACM->SetVAD(*dtx, *vad, *mode));
-
- }
-
- CHECK_ERROR_MT(myACM->RegisterSendCodec(myCodec));
- myChannel->ResetStats();
- {
- WriteLockScoped wl(_apiTestRWLock);
- *thereIsEncoder = true;
- }
- Wait(500);
-}
-
-
-void
-APITest::LookForDTMF(char side)
-{
- if(!_randomTest)
- {
- fprintf(stdout, "\n\nLooking for DTMF Signal in Side %c\n", side);
- fprintf(stdout, "----------------------------------------\n");
- }
-
- if(side == 'A')
- {
- _acmB->RegisterIncomingMessagesCallback(NULL);
- _acmA->RegisterIncomingMessagesCallback(_dtmfCallback);
- Wait(1000);
- _acmA->RegisterIncomingMessagesCallback(NULL);
- }
- else
- {
- _acmA->RegisterIncomingMessagesCallback(NULL);
- _acmB->RegisterIncomingMessagesCallback(_dtmfCallback);
- Wait(1000);
- _acmB->RegisterIncomingMessagesCallback(NULL);
- }
-}
-
-} // namespace webrtc
-
diff --git a/trunk/src/modules/audio_coding/main/test/APITest.h b/trunk/src/modules/audio_coding/main/test/APITest.h
deleted file mode 100644
index db0a87c..0000000
--- a/trunk/src/modules/audio_coding/main/test/APITest.h
+++ /dev/null
@@ -1,177 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef API_TEST_H
-#define API_TEST_H
-
-#include "ACMTest.h"
-#include "Channel.h"
-#include "PCMFile.h"
-#include "event_wrapper.h"
-#include "utility.h"
-
-namespace webrtc {
-
-enum APITESTAction {TEST_CHANGE_CODEC_ONLY = 0, DTX_TEST = 1};
-
-class APITest : public ACMTest
-{
-public:
- APITest();
- ~APITest();
-
- void Perform();
-private:
- WebRtc_Word16 SetUp();
-
- static bool PushAudioThreadA(void* obj);
- static bool PullAudioThreadA(void* obj);
- static bool ProcessThreadA(void* obj);
- static bool APIThreadA(void* obj);
-
- static bool PushAudioThreadB(void* obj);
- static bool PullAudioThreadB(void* obj);
- static bool ProcessThreadB(void* obj);
- static bool APIThreadB(void* obj);
-
- void CheckVADStatus(char side);
-
- // Set Min delay, get delay, playout timestamp
- void TestDelay(char side);
-
- // Unregister a codec & register again.
- void TestRegisteration(char side);
-
- // Playout Mode, background noise mode.
- // Receiver Frequency, playout frequency.
- void TestPlayout(char receiveSide);
-
- // set/get receiver VAD status & mode.
- void TestReceiverVAD(char side);
-
- //
- void TestSendVAD(char side);
-
- void CurrentCodec(char side);
-
- void ChangeCodec(char side);
-
- void Wait(WebRtc_UWord32 waitLengthMs);
-
- void LookForDTMF(char side);
-
- void RunTest(char thread);
-
- bool PushAudioRunA();
- bool PullAudioRunA();
- bool ProcessRunA();
- bool APIRunA();
-
- bool PullAudioRunB();
- bool PushAudioRunB();
- bool ProcessRunB();
- bool APIRunB();
-
-
-
- //--- ACMs
- AudioCodingModule* _acmA;
- AudioCodingModule* _acmB;
-
- //--- Channels
- Channel* _channel_A2B;
- Channel* _channel_B2A;
-
- //--- I/O files
- // A
- PCMFile _inFileA;
- PCMFile _outFileA;
- // B
- PCMFile _outFileB;
- PCMFile _inFileB;
-
- //--- I/O params
- // A
- WebRtc_Word32 _outFreqHzA;
- // B
- WebRtc_Word32 _outFreqHzB;
-
- // Should we write to file.
- // we might skip writing to file if we
- // run the test for a long time.
- bool _writeToFile;
- //--- Events
- // A
- EventWrapper* _pullEventA; // pulling data from ACM
- EventWrapper* _pushEventA; // pushing data to ACM
- EventWrapper* _processEventA; // process
- EventWrapper* _apiEventA; // API calls
- // B
- EventWrapper* _pullEventB; // pulling data from ACM
- EventWrapper* _pushEventB; // pushing data to ACM
- EventWrapper* _processEventB; // process
- EventWrapper* _apiEventB; // API calls
-
- // keep track of the codec in either side.
- WebRtc_UWord8 _codecCntrA;
- WebRtc_UWord8 _codecCntrB;
-
- // keep track of tests
- WebRtc_UWord8 _testCntrA;
- WebRtc_UWord8 _testCntrB;
-
- // Is set to true if there is no encoder in either side
- bool _thereIsEncoderA;
- bool _thereIsEncoderB;
- bool _thereIsDecoderA;
- bool _thereIsDecoderB;
-
- bool _sendVADA;
- bool _sendDTXA;
- ACMVADMode _sendVADModeA;
-
- bool _sendVADB;
- bool _sendDTXB;
- ACMVADMode _sendVADModeB;
-
- WebRtc_Word32 _minDelayA;
- WebRtc_Word32 _minDelayB;
- bool _payloadUsed[32];
-
- AudioPlayoutMode _playoutModeA;
- AudioPlayoutMode _playoutModeB;
-
- ACMBackgroundNoiseMode _bgnModeA;
- ACMBackgroundNoiseMode _bgnModeB;
-
-
- int _receiveVADActivityA[3];
- int _receiveVADActivityB[3];
- bool _verbose;
-
- int _dotPositionA;
- int _dotMoveDirectionA;
- int _dotPositionB;
- int _dotMoveDirectionB;
-
- char _movingDot[41];
-
- DTMFDetector* _dtmfCallback;
- VADCallback* _vadCallbackA;
- VADCallback* _vadCallbackB;
- RWLockWrapper& _apiTestRWLock;
- bool _randomTest;
- int _testNumA;
- int _testNumB;
-};
-
-} // namespace webrtc
-
-#endif
diff --git a/trunk/src/modules/audio_coding/main/test/Channel.cc b/trunk/src/modules/audio_coding/main/test/Channel.cc
deleted file mode 100644
index 363b106..0000000
--- a/trunk/src/modules/audio_coding/main/test/Channel.cc
+++ /dev/null
@@ -1,483 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <assert.h>
-#include <iostream>
-
-#include "audio_coding_module.h"
-#include "Channel.h"
-#include "tick_util.h"
-#include "typedefs.h"
-#include "common_types.h"
-
-namespace webrtc {
-
-WebRtc_Word32
-Channel::SendData(
- const FrameType frameType,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const RTPFragmentationHeader* fragmentation)
-{
- WebRtcRTPHeader rtpInfo;
- WebRtc_Word32 status;
- WebRtc_UWord16 payloadDataSize = payloadSize;
-
- rtpInfo.header.markerBit = false;
- rtpInfo.header.ssrc = 0;
- rtpInfo.header.sequenceNumber = _seqNo++;
- rtpInfo.header.payloadType = payloadType;
- rtpInfo.header.timestamp = timeStamp;
- if(frameType == kAudioFrameCN)
- {
- rtpInfo.type.Audio.isCNG = true;
- }
- else
- {
- rtpInfo.type.Audio.isCNG = false;
- }
- if(frameType == kFrameEmpty)
- {
- // Skip this frame
- return 0;
- }
-
- rtpInfo.type.Audio.channel = 1;
- // Treat fragmentation separately
- if(fragmentation != NULL)
- {
- if((fragmentation->fragmentationTimeDiff[1] <= 0x3fff) && // silence for too long send only new data
- (fragmentation->fragmentationVectorSize == 2))
- {
- // only 0x80 if we have multiple blocks
- _payloadData[0] = 0x80 + fragmentation->fragmentationPlType[1];
- WebRtc_UWord32 REDheader = (((WebRtc_UWord32)fragmentation->fragmentationTimeDiff[1]) << 10) + fragmentation->fragmentationLength[1];
- _payloadData[1] = WebRtc_UWord8((REDheader >> 16) & 0x000000FF);
- _payloadData[2] = WebRtc_UWord8((REDheader >> 8) & 0x000000FF);
- _payloadData[3] = WebRtc_UWord8(REDheader & 0x000000FF);
-
- _payloadData[4] = fragmentation->fragmentationPlType[0];
- // copy the RED data
- memcpy(_payloadData + 5,
- payloadData + fragmentation->fragmentationOffset[1],
- fragmentation->fragmentationLength[1]);
- // copy the normal data
- memcpy(_payloadData + 5 + fragmentation->fragmentationLength[1],
- payloadData + fragmentation->fragmentationOffset[0],
- fragmentation->fragmentationLength[0]);
- payloadDataSize += 5;
- } else
- {
- // single block (newest one)
- memcpy(_payloadData,
- payloadData + fragmentation->fragmentationOffset[0],
- fragmentation->fragmentationLength[0]);
- payloadDataSize = WebRtc_UWord16(fragmentation->fragmentationLength[0]);
- rtpInfo.header.payloadType = fragmentation->fragmentationPlType[0];
- }
- }
- else
- {
- memcpy(_payloadData, payloadData, payloadDataSize);
- if(_isStereo)
- {
- if(_leftChannel)
- {
- memcpy(&_rtpInfo, &rtpInfo, sizeof(WebRtcRTPHeader));
- _leftChannel = false;
- rtpInfo.type.Audio.channel = 1;
- }
- else
- {
- memcpy(&rtpInfo, &_rtpInfo, sizeof(WebRtcRTPHeader));
- _leftChannel = true;
- rtpInfo.type.Audio.channel = 2;
- }
- }
- }
-
- _channelCritSect->Enter();
- if(_saveBitStream)
- {
- //fwrite(payloadData, sizeof(WebRtc_UWord8), payloadSize, _bitStreamFile);
- }
-
- if(!_isStereo)
- {
- CalcStatistics(rtpInfo, payloadSize);
- }
- _lastInTimestamp = timeStamp;
- _totalBytes += payloadDataSize;
- _channelCritSect->Leave();
-
- if(_useFECTestWithPacketLoss)
- {
- _packetLoss += 1;
- if(_packetLoss == 3)
- {
- _packetLoss = 0;
- return 0;
- }
- }
-
-
- //status = _receiverACM->IncomingPayload((WebRtc_Word8*)_payloadData, payloadSize, payloadType, timeStamp);
- status = _receiverACM->IncomingPacket((WebRtc_Word8*)_payloadData, payloadDataSize, rtpInfo);
-
- //delete [] payloadData;
-
-
-
- return status;
-}
-
-void
-Channel::CalcStatistics(
- WebRtcRTPHeader& rtpInfo,
- WebRtc_UWord16 payloadSize)
-{
- int n;
- if((rtpInfo.header.payloadType != _lastPayloadType) &&
- (_lastPayloadType != -1))
- {
- // payload-type is changed.
- // we have to terminate the calculations on the previous payload type
- // we ignore the last packet in that payload type just to make things
- // easier.
- for(n = 0; n < MAX_NUM_PAYLOADS; n++)
- {
- if(_lastPayloadType == _payloadStats[n].payloadType)
- {
- _payloadStats[n].newPacket = true;
- break;
- }
- }
- }
- _lastPayloadType = rtpInfo.header.payloadType;
-
- bool newPayload = true;
- ACMTestPayloadStats* currentPayloadStr;
- for(n = 0; n < MAX_NUM_PAYLOADS; n++)
- {
- if(rtpInfo.header.payloadType == _payloadStats[n].payloadType)
- {
- newPayload = false;
- currentPayloadStr = &_payloadStats[n];
- break;
- }
- }
-
- if(!newPayload)
- {
- if(!currentPayloadStr->newPacket)
- {
- WebRtc_UWord32 lastFrameSizeSample = (WebRtc_UWord32)((WebRtc_UWord32)rtpInfo.header.timestamp -
- (WebRtc_UWord32)currentPayloadStr->lastTimestamp);
- assert(lastFrameSizeSample > 0);
- int k = 0;
- while((currentPayloadStr->frameSizeStats[k].frameSizeSample !=
- lastFrameSizeSample) &&
- (currentPayloadStr->frameSizeStats[k].frameSizeSample != 0))
- {
- k++;
- }
- ACMTestFrameSizeStats* currentFrameSizeStats =
- &(currentPayloadStr->frameSizeStats[k]);
- currentFrameSizeStats->frameSizeSample = (WebRtc_Word16)lastFrameSizeSample;
-
- // increment the number of encoded samples.
- currentFrameSizeStats->totalEncodedSamples +=
- lastFrameSizeSample;
- // increment the number of recveived packets
- currentFrameSizeStats->numPackets++;
- // increment the total number of bytes (this is based on
- // the previous payload we don't know the frame-size of
- // the current payload.
- currentFrameSizeStats->totalPayloadLenByte +=
- currentPayloadStr->lastPayloadLenByte;
- // store the maximum payload-size (this is based on
- // the previous payload we don't know the frame-size of
- // the current payload.
- if(currentFrameSizeStats->maxPayloadLen <
- currentPayloadStr->lastPayloadLenByte)
- {
- currentFrameSizeStats->maxPayloadLen =
- currentPayloadStr->lastPayloadLenByte;
- }
- // store the current values for the next time
- currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
- currentPayloadStr->lastPayloadLenByte = payloadSize;
- }
- else
- {
- currentPayloadStr->newPacket = false;
- currentPayloadStr->lastPayloadLenByte = payloadSize;
- currentPayloadStr->lastTimestamp = rtpInfo.header.timestamp;
- currentPayloadStr->payloadType = rtpInfo.header.payloadType;
- }
- }
- else
- {
- n = 0;
- while(_payloadStats[n].payloadType != -1)
- {
- n++;
- }
- // first packet
- _payloadStats[n].newPacket = false;
- _payloadStats[n].lastPayloadLenByte = payloadSize;
- _payloadStats[n].lastTimestamp = rtpInfo.header.timestamp;
- _payloadStats[n].payloadType = rtpInfo.header.payloadType;
- }
-}
-
-Channel::Channel(WebRtc_Word16 chID) :
-_receiverACM(NULL),
-_seqNo(0),
-_channelCritSect(CriticalSectionWrapper::CreateCriticalSection()),
-_bitStreamFile(NULL),
-_saveBitStream(false),
-_lastPayloadType(-1),
-_isStereo(false),
-_leftChannel(true),
-_lastInTimestamp(0),
-_packetLoss(0),
-_useFECTestWithPacketLoss(false),
-_chID(chID),
-_beginTime(TickTime::MillisecondTimestamp()),
-_totalBytes(0)
-{
- int n;
- int k;
- for(n = 0; n < MAX_NUM_PAYLOADS; n++)
- {
- _payloadStats[n].payloadType = -1;
- _payloadStats[n].newPacket = true;
- for(k = 0; k < MAX_NUM_FRAMESIZES; k++)
- {
- _payloadStats[n].frameSizeStats[k].frameSizeSample = 0;
- _payloadStats[n].frameSizeStats[k].maxPayloadLen = 0;
- _payloadStats[n].frameSizeStats[k].numPackets = 0;
- _payloadStats[n].frameSizeStats[k].totalPayloadLenByte = 0;
- _payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
- }
- }
- if(chID >= 0)
- {
- _saveBitStream = true;
- char bitStreamFileName[500];
- sprintf(bitStreamFileName, "bitStream_%d.dat", chID);
- _bitStreamFile = fopen(bitStreamFileName, "wb");
- }
- else
- {
- _saveBitStream = false;
- }
-}
-
-Channel::~Channel()
-{
- delete _channelCritSect;
-}
-
-void
-Channel::RegisterReceiverACM(AudioCodingModule* acm)
-{
- _receiverACM = acm;
- return;
-}
-
-void
-Channel::ResetStats()
-{
- int n;
- int k;
- _channelCritSect->Enter();
- _lastPayloadType = -1;
- for(n = 0; n < MAX_NUM_PAYLOADS; n++)
- {
- _payloadStats[n].payloadType = -1;
- _payloadStats[n].newPacket = true;
- for(k = 0; k < MAX_NUM_FRAMESIZES; k++)
- {
- _payloadStats[n].frameSizeStats[k].frameSizeSample = 0;
- _payloadStats[n].frameSizeStats[k].maxPayloadLen = 0;
- _payloadStats[n].frameSizeStats[k].numPackets = 0;
- _payloadStats[n].frameSizeStats[k].totalPayloadLenByte = 0;
- _payloadStats[n].frameSizeStats[k].totalEncodedSamples = 0;
- }
- }
- _beginTime = TickTime::MillisecondTimestamp();
- _totalBytes = 0;
- _channelCritSect->Leave();
-}
-
-WebRtc_Word16
-Channel::Stats(CodecInst& codecInst, ACMTestPayloadStats& payloadStats)
-{
- _channelCritSect->Enter();
- int n;
- payloadStats.payloadType = -1;
- for(n = 0; n < MAX_NUM_PAYLOADS; n++)
- {
- if(_payloadStats[n].payloadType == codecInst.pltype)
- {
- memcpy(&payloadStats, &_payloadStats[n], sizeof(ACMTestPayloadStats));
- break;
- }
- }
- if(payloadStats.payloadType == -1)
- {
- _channelCritSect->Leave();
- return -1;
- }
- for(n = 0; n < MAX_NUM_FRAMESIZES; n++)
- {
- if(payloadStats.frameSizeStats[n].frameSizeSample == 0)
- {
- _channelCritSect->Leave();
- return 0;
- }
- payloadStats.frameSizeStats[n].usageLenSec =
- (double)payloadStats.frameSizeStats[n].totalEncodedSamples
- / (double)codecInst.plfreq;
-
- payloadStats.frameSizeStats[n].rateBitPerSec =
- payloadStats.frameSizeStats[n].totalPayloadLenByte * 8 /
- payloadStats.frameSizeStats[n].usageLenSec;
-
- }
- _channelCritSect->Leave();
- return 0;
-}
-
-void
-Channel::Stats(WebRtc_UWord32* numPackets)
-{
- _channelCritSect->Enter();
- int k;
- int n;
- memset(numPackets, 0, MAX_NUM_PAYLOADS * sizeof(WebRtc_UWord32));
- for(k = 0; k < MAX_NUM_PAYLOADS; k++)
- {
- if(_payloadStats[k].payloadType == -1)
- {
- break;
- }
- numPackets[k] = 0;
- for(n = 0; n < MAX_NUM_FRAMESIZES; n++)
- {
- if(_payloadStats[k].frameSizeStats[n].frameSizeSample == 0)
- {
- break;
- }
- numPackets[k] +=
- _payloadStats[k].frameSizeStats[n].numPackets;
- }
- }
- _channelCritSect->Leave();
-}
-
-void
-Channel::Stats(WebRtc_UWord8* payloadType, WebRtc_UWord32* payloadLenByte)
-{
- _channelCritSect->Enter();
-
- int k;
- int n;
- memset(payloadLenByte, 0, MAX_NUM_PAYLOADS * sizeof(WebRtc_UWord32));
- for(k = 0; k < MAX_NUM_PAYLOADS; k++)
- {
- if(_payloadStats[k].payloadType == -1)
- {
- break;
- }
- payloadType[k] = (WebRtc_UWord8)_payloadStats[k].payloadType;
- payloadLenByte[k] = 0;
- for(n = 0; n < MAX_NUM_FRAMESIZES; n++)
- {
- if(_payloadStats[k].frameSizeStats[n].frameSizeSample == 0)
- {
- break;
- }
- payloadLenByte[k] += (WebRtc_UWord16)
- _payloadStats[k].frameSizeStats[n].totalPayloadLenByte;
- }
- }
-
- _channelCritSect->Leave();
-}
-
-
-void
-Channel::PrintStats(CodecInst& codecInst)
-{
- ACMTestPayloadStats payloadStats;
- Stats(codecInst, payloadStats);
- printf("%s %d kHz\n",
- codecInst.plname,
- codecInst.plfreq / 1000);
- printf("=====================================================\n");
- if(payloadStats.payloadType == -1)
- {
- printf("No Packets are sent with payload-type %d (%s)\n\n",
- codecInst.pltype,
- codecInst.plname);
- return;
- }
- for(int k = 0; k < MAX_NUM_FRAMESIZES; k++)
- {
- if(payloadStats.frameSizeStats[k].frameSizeSample == 0)
- {
- break;
- }
- printf("Frame-size.................... %d samples\n",
- payloadStats.frameSizeStats[k].frameSizeSample);
- printf("Average Rate.................. %.0f bits/sec\n",
- payloadStats.frameSizeStats[k].rateBitPerSec);
- printf("Maximum Payload-Size.......... %d Bytes\n",
- payloadStats.frameSizeStats[k].maxPayloadLen);
- printf("Maximum Instantaneous Rate.... %.0f bits/sec\n",
- ((double)payloadStats.frameSizeStats[k].maxPayloadLen * 8.0 *
- (double)codecInst.plfreq) /
- (double)payloadStats.frameSizeStats[k].frameSizeSample);
- printf("Number of Packets............. %u\n",
- (unsigned int)payloadStats.frameSizeStats[k].numPackets);
- printf("Duration...................... %0.3f sec\n\n",
- payloadStats.frameSizeStats[k].usageLenSec);
-
- }
-
-}
-
-WebRtc_UWord32
-Channel::LastInTimestamp()
-{
- WebRtc_UWord32 timestamp;
- _channelCritSect->Enter();
- timestamp = _lastInTimestamp;
- _channelCritSect->Leave();
- return timestamp;
-}
-
-double
-Channel::BitRate()
-{
- double rate;
- WebRtc_UWord64 currTime = TickTime::MillisecondTimestamp();
- _channelCritSect->Enter();
- rate = ((double)_totalBytes * 8.0)/ (double)(currTime - _beginTime);
- _channelCritSect->Leave();
- return rate;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/Channel.h b/trunk/src/modules/audio_coding/main/test/Channel.h
deleted file mode 100644
index 375bec7..0000000
--- a/trunk/src/modules/audio_coding/main/test/Channel.h
+++ /dev/null
@@ -1,125 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef CHANNEL_H
-#define CHANNEL_H
-
-#include <stdio.h>
-
-#include "audio_coding_module.h"
-#include "critical_section_wrapper.h"
-#include "rw_lock_wrapper.h"
-
-namespace webrtc {
-
-#define MAX_NUM_PAYLOADS 50
-#define MAX_NUM_FRAMESIZES 6
-
-
-struct ACMTestFrameSizeStats
-{
- WebRtc_UWord16 frameSizeSample;
- WebRtc_Word16 maxPayloadLen;
- WebRtc_UWord32 numPackets;
- WebRtc_UWord64 totalPayloadLenByte;
- WebRtc_UWord64 totalEncodedSamples;
- double rateBitPerSec;
- double usageLenSec;
-
-};
-
-struct ACMTestPayloadStats
-{
- bool newPacket;
- WebRtc_Word16 payloadType;
- WebRtc_Word16 lastPayloadLenByte;
- WebRtc_UWord32 lastTimestamp;
- ACMTestFrameSizeStats frameSizeStats[MAX_NUM_FRAMESIZES];
-};
-
-class Channel: public AudioPacketizationCallback
-{
-public:
-
- Channel(
- WebRtc_Word16 chID = -1);
- ~Channel();
-
- WebRtc_Word32 SendData(
- const FrameType frameType,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const RTPFragmentationHeader* fragmentation);
-
- void RegisterReceiverACM(
- AudioCodingModule *acm);
-
- void ResetStats();
-
- WebRtc_Word16 Stats(
- CodecInst& codecInst,
- ACMTestPayloadStats& payloadStats);
-
- void Stats(
- WebRtc_UWord32* numPackets);
-
- void Stats(
- WebRtc_UWord8* payloadLenByte,
- WebRtc_UWord32* payloadType);
-
- void PrintStats(
- CodecInst& codecInst);
-
- void SetIsStereo(bool isStereo)
- {
- _isStereo = isStereo;
- }
-
- WebRtc_UWord32 LastInTimestamp();
-
- void SetFECTestWithPacketLoss(bool usePacketLoss)
- {
- _useFECTestWithPacketLoss = usePacketLoss;
- }
-
- double BitRate();
-
-private:
- void CalcStatistics(
- WebRtcRTPHeader& rtpInfo,
- WebRtc_UWord16 payloadSize);
-
- AudioCodingModule* _receiverACM;
- WebRtc_UWord16 _seqNo;
- // 60msec * 32 sample(max)/msec * 2 description (maybe) * 2 bytes/sample
- WebRtc_UWord8 _payloadData[60 * 32 * 2 * 2];
-
- CriticalSectionWrapper* _channelCritSect;
- FILE* _bitStreamFile;
- bool _saveBitStream;
- WebRtc_Word16 _lastPayloadType;
- ACMTestPayloadStats _payloadStats[MAX_NUM_PAYLOADS];
- bool _isStereo;
- WebRtcRTPHeader _rtpInfo;
- bool _leftChannel;
- WebRtc_UWord32 _lastInTimestamp;
- // FEC Test variables
- WebRtc_Word16 _packetLoss;
- bool _useFECTestWithPacketLoss;
- WebRtc_Word16 _chID;
- WebRtc_UWord64 _beginTime;
- WebRtc_UWord64 _totalBytes;
-};
-
-} // namespace webrtc
-
-#endif
diff --git a/trunk/src/modules/audio_coding/main/test/EncodeDecodeTest.cc b/trunk/src/modules/audio_coding/main/test/EncodeDecodeTest.cc
deleted file mode 100644
index 74bb84c..0000000
--- a/trunk/src/modules/audio_coding/main/test/EncodeDecodeTest.cc
+++ /dev/null
@@ -1,407 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "EncodeDecodeTest.h"
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "audio_coding_module.h"
-#include "common_types.h"
-#include "gtest/gtest.h"
-#include "trace.h"
-#include "testsupport/fileutils.h"
-#include "utility.h"
-
-namespace webrtc {
-
-TestPacketization::TestPacketization(RTPStream *rtpStream,
- WebRtc_UWord16 frequency)
- : _rtpStream(rtpStream),
- _frequency(frequency),
- _seqNo(0) {
-}
-
-TestPacketization::~TestPacketization() { }
-
-WebRtc_Word32 TestPacketization::SendData(
- const FrameType /* frameType */,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const RTPFragmentationHeader* /* fragmentation */) {
- _rtpStream->Write(payloadType, timeStamp, _seqNo++, payloadData, payloadSize,
- _frequency);
- return 1;
-}
-
-Sender::Sender()
- : _acm(NULL),
- _pcmFile(),
- _audioFrame(),
- _payloadSize(0),
- _timeStamp(0),
- _packetization(NULL) {
-}
-
-void Sender::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
- acm->InitializeSender();
- struct CodecInst sendCodec;
- int noOfCodecs = acm->NumberOfCodecs();
- int codecNo;
-
- if (testMode == 1) {
- // Set the codec, input file, and parameters for the current test.
- codecNo = codeId;
- // Use same input file for now.
- char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
- _pcmFile.Open(fileName, 32000, "rb");
- } else if (testMode == 0) {
- // Set the codec, input file, and parameters for the current test.
- codecNo = codeId;
- acm->Codec(codecNo, sendCodec);
- // Use same input file for now.
- char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
- _pcmFile.Open(fileName, 32000, "rb");
- } else {
- printf("List of supported codec.\n");
- for (int n = 0; n < noOfCodecs; n++) {
- acm->Codec(n, sendCodec);
- printf("%d %s\n", n, sendCodec.plname);
- }
- printf("Choose your codec:");
- ASSERT_GT(scanf("%d", &codecNo), 0);
- char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
- _pcmFile.Open(fileName, 32000, "rb");
- }
-
- acm->Codec(codecNo, sendCodec);
- acm->RegisterSendCodec(sendCodec);
- _packetization = new TestPacketization(rtpStream, sendCodec.plfreq);
- if (acm->RegisterTransportCallback(_packetization) < 0) {
- printf("Registering Transport Callback failed, for run: codecId: %d: --\n",
- codeId);
- }
-
- _acm = acm;
- }
-
-void Sender::Teardown() {
- _pcmFile.Close();
- delete _packetization;
-}
-
-bool Sender::Add10MsData() {
- if (!_pcmFile.EndOfFile()) {
- _pcmFile.Read10MsData(_audioFrame);
- WebRtc_Word32 ok = _acm->Add10MsData(_audioFrame);
- if (ok != 0) {
- printf("Error calling Add10MsData: for run: codecId: %d\n", codeId);
- exit(1);
- }
- return true;
- }
- return false;
-}
-
-bool Sender::Process() {
- WebRtc_Word32 ok = _acm->Process();
- if (ok < 0) {
- printf("Error calling Add10MsData: for run: codecId: %d\n", codeId);
- exit(1);
- }
- return true;
-}
-
-void Sender::Run() {
- while (true) {
- if (!Add10MsData()) {
- break;
- }
- if (!Process()) { // This could be done in a processing thread
- break;
- }
- }
-}
-
-Receiver::Receiver()
- : _playoutLengthSmpls(WEBRTC_10MS_PCM_AUDIO),
- _payloadSizeBytes(MAX_INCOMING_PAYLOAD) {
-}
-
-void Receiver::Setup(AudioCodingModule *acm, RTPStream *rtpStream) {
- struct CodecInst recvCodec;
- int noOfCodecs;
- acm->InitializeReceiver();
-
- noOfCodecs = acm->NumberOfCodecs();
- for (int i = 0; i < noOfCodecs; i++) {
- acm->Codec((WebRtc_UWord8) i, recvCodec);
- if (acm->RegisterReceiveCodec(recvCodec) != 0) {
- printf("Unable to register codec: for run: codecId: %d\n", codeId);
- exit(1);
- }
- }
-
- char filename[128];
- _rtpStream = rtpStream;
- int playSampFreq;
-
- if (testMode == 1) {
- playSampFreq=recvCodec.plfreq;
- //output file for current run
- sprintf(filename,"%s/out%dFile.pcm", webrtc::test::OutputPath().c_str(),
- codeId);
- _pcmFile.Open(filename, recvCodec.plfreq, "wb+");
- } else if (testMode == 0) {
- playSampFreq=32000;
- //output file for current run
- sprintf(filename, "%s/encodeDecode_out%d.pcm",
- webrtc::test::OutputPath().c_str(), codeId);
- _pcmFile.Open(filename, 32000/*recvCodec.plfreq*/, "wb+");
- } else {
- printf("\nValid output frequencies:\n");
- printf("8000\n16000\n32000\n-1,");
- printf("which means output freq equal to received signal freq");
- printf("\n\nChoose output sampling frequency: ");
- ASSERT_GT(scanf("%d", &playSampFreq), 0);
- sprintf(filename, "%s/outFile.pcm", webrtc::test::OutputPath().c_str());
- _pcmFile.Open(filename, 32000, "wb+");
- }
-
- _realPayloadSizeBytes = 0;
- _playoutBuffer = new WebRtc_Word16[WEBRTC_10MS_PCM_AUDIO];
- _frequency = playSampFreq;
- _acm = acm;
- _firstTime = true;
-}
-
-void Receiver::Teardown() {
- delete [] _playoutBuffer;
- _pcmFile.Close();
- if (testMode > 1)
- Trace::ReturnTrace();
-}
-
-bool Receiver::IncomingPacket() {
- if (!_rtpStream->EndOfFile()) {
- if (_firstTime) {
- _firstTime = false;
- _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
- _payloadSizeBytes, &_nextTime);
- if (_realPayloadSizeBytes == 0) {
- if (_rtpStream->EndOfFile()) {
- _firstTime = true;
- return true;
- } else {
- printf("Error in reading incoming payload.\n");
- return false;
- }
- }
- }
-
- WebRtc_Word32 ok = _acm->IncomingPacket(_incomingPayload,
- _realPayloadSizeBytes, _rtpInfo);
- if (ok != 0) {
- printf("Error when inserting packet to ACM, for run: codecId: %d\n",
- codeId);
- exit(1);
- }
- _realPayloadSizeBytes = _rtpStream->Read(&_rtpInfo, _incomingPayload,
- _payloadSizeBytes, &_nextTime);
- if (_realPayloadSizeBytes == 0 && _rtpStream->EndOfFile()) {
- _firstTime = true;
- }
- }
- return true;
-}
-
-bool Receiver::PlayoutData() {
- AudioFrame audioFrame;
-
- if (_acm->PlayoutData10Ms(_frequency, audioFrame) != 0) {
- printf("Error when calling PlayoutData10Ms, for run: codecId: %d\n",
- codeId);
- exit(1);
- }
- if (_playoutLengthSmpls == 0) {
- return false;
- }
- _pcmFile.Write10MsData(audioFrame._payloadData,
- audioFrame._payloadDataLengthInSamples);
- return true;
-}
-
-void Receiver::Run() {
- WebRtc_UWord8 counter500Ms = 50;
- WebRtc_UWord32 clock = 0;
-
- while (counter500Ms > 0) {
- if (clock == 0 || clock >= _nextTime) {
- IncomingPacket();
- if (clock == 0) {
- clock = _nextTime;
- }
- }
- if ((clock % 10) == 0) {
- if (!PlayoutData()) {
- clock++;
- continue;
- }
- }
- if (_rtpStream->EndOfFile()) {
- counter500Ms--;
- }
- clock++;
- }
-}
-
-EncodeDecodeTest::EncodeDecodeTest() {
- _testMode = 2;
- Trace::CreateTrace();
- Trace::SetTraceFile("acm_encdec_test.txt");
-}
-
-EncodeDecodeTest::EncodeDecodeTest(int testMode) {
- //testMode == 0 for autotest
- //testMode == 1 for testing all codecs/parameters
- //testMode > 1 for specific user-input test (as it was used before)
- _testMode = testMode;
- if(_testMode != 0) {
- Trace::CreateTrace();
- Trace::SetTraceFile("acm_encdec_test.txt");
- }
-}
-
-void EncodeDecodeTest::Perform() {
- if (_testMode == 0) {
- printf("Running Encode/Decode Test");
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1,
- "---------- EncodeDecodeTest ----------");
- }
-
- int numCodecs = 1;
- int codePars[3]; //freq, pacsize, rate
- int numPars[52]; //number of codec parameters sets (rate,freq,pacsize)to test,
- //for a given codec
-
- codePars[0] = 0;
- codePars[1] = 0;
- codePars[2] = 0;
-
- if (_testMode == 1) {
- AudioCodingModule *acmTmp = AudioCodingModule::Create(0);
- struct CodecInst sendCodecTmp;
- numCodecs = acmTmp->NumberOfCodecs();
- printf("List of supported codec.\n");
- for(int n = 0; n < numCodecs; n++) {
- acmTmp->Codec(n, sendCodecTmp);
- if (STR_CASE_CMP(sendCodecTmp.plname, "telephone-event") == 0) {
- numPars[n] = 0;
- } else if (STR_CASE_CMP(sendCodecTmp.plname, "cn") == 0) {
- numPars[n] = 0;
- } else if (STR_CASE_CMP(sendCodecTmp.plname, "red") == 0) {
- numPars[n] = 0;
- } else {
- numPars[n] = 1;
- printf("%d %s\n", n, sendCodecTmp.plname);
- }
- }
- AudioCodingModule::Destroy(acmTmp);
- } else if (_testMode == 0) {
- AudioCodingModule *acmTmp = AudioCodingModule::Create(0);
- numCodecs = acmTmp->NumberOfCodecs();
- AudioCodingModule::Destroy(acmTmp);
- struct CodecInst dummyCodec;
-
- //chose range of testing for codecs/parameters
- for(int i = 0 ; i < numCodecs ; i++) {
- numPars[i] = 1;
- acmTmp->Codec(i, dummyCodec);
- if (STR_CASE_CMP(dummyCodec.plname, "telephone-event") == 0) {
- numPars[i] = 0;
- } else if (STR_CASE_CMP(dummyCodec.plname, "cn") == 0) {
- numPars[i] = 0;
- } else if (STR_CASE_CMP(dummyCodec.plname, "red") == 0) {
- numPars[i] = 0;
- }
- }
- } else {
- numCodecs = 1;
- numPars[0] = 1;
- }
-
- _receiver.testMode = _testMode;
-
- //loop over all codecs:
- for (int codeId = 0; codeId < numCodecs; codeId++) {
- //only encode using real encoders, not telephone-event anc cn
- for (int loopPars = 1; loopPars <= numPars[codeId]; loopPars++) {
- if (_testMode == 1) {
- printf("\n");
- printf("***FOR RUN: codeId: %d\n", codeId);
- printf("\n");
- } else if (_testMode == 0) {
- printf(".");
- }
-
- EncodeToFile(1, codeId, codePars, _testMode);
-
- AudioCodingModule *acm = AudioCodingModule::Create(10);
- RTPFile rtpFile;
- std::string fileName = webrtc::test::OutputPath() + "outFile.rtp";
- rtpFile.Open(fileName.c_str(), "rb");
-
- _receiver.codeId = codeId;
-
- rtpFile.ReadHeader();
- _receiver.Setup(acm, &rtpFile);
- _receiver.Run();
- _receiver.Teardown();
- rtpFile.Close();
- AudioCodingModule::Destroy(acm);
-
- if (_testMode == 1) {
- printf("***COMPLETED RUN FOR: codecID: %d ***\n", codeId);
- }
- }
- }
- if (_testMode == 0) {
- printf("Done!\n");
- }
- if (_testMode == 1)
- Trace::ReturnTrace();
-}
-
-void EncodeDecodeTest::EncodeToFile(int fileType, int codeId, int* codePars,
- int testMode) {
- AudioCodingModule *acm = AudioCodingModule::Create(0);
- RTPFile rtpFile;
- std::string fileName = webrtc::test::OutputPath() + "outFile.rtp";
- rtpFile.Open(fileName.c_str(), "wb+");
- rtpFile.WriteHeader();
-
- //for auto_test and logging
- _sender.testMode = testMode;
- _sender.codeId = codeId;
-
- _sender.Setup(acm, &rtpFile);
- struct CodecInst sendCodecInst;
- if (acm->SendCodec(sendCodecInst) >= 0) {
- _sender.Run();
- }
- _sender.Teardown();
- rtpFile.Close();
- AudioCodingModule::Destroy(acm);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/EncodeDecodeTest.h b/trunk/src/modules/audio_coding/main/test/EncodeDecodeTest.h
deleted file mode 100644
index a730fea..0000000
--- a/trunk/src/modules/audio_coding/main/test/EncodeDecodeTest.h
+++ /dev/null
@@ -1,117 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
-#define WEBRTC_MODULES_AUDIO_CODING_MAIN_TEST_ENCODEDECODETEST_H_
-
-#include <stdio.h>
-
-#include "ACMTest.h"
-#include "audio_coding_module.h"
-#include "RTPFile.h"
-#include "PCMFile.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-#define MAX_INCOMING_PAYLOAD 8096
-
-// TestPacketization callback which writes the encoded payloads to file
-class TestPacketization: public AudioPacketizationCallback {
- public:
- TestPacketization(RTPStream *rtpStream, WebRtc_UWord16 frequency);
- ~TestPacketization();
- virtual WebRtc_Word32 SendData(const FrameType frameType,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const RTPFragmentationHeader* fragmentation);
-
- private:
- static void MakeRTPheader(WebRtc_UWord8* rtpHeader, WebRtc_UWord8 payloadType,
- WebRtc_Word16 seqNo, WebRtc_UWord32 timeStamp,
- WebRtc_UWord32 ssrc);
- RTPStream* _rtpStream;
- WebRtc_Word32 _frequency;
- WebRtc_Word16 _seqNo;
-};
-
-class Sender {
- public:
- Sender();
- void Setup(AudioCodingModule *acm, RTPStream *rtpStream);
- void Teardown();
- void Run();
- bool Add10MsData();
- bool Process();
-
- //for auto_test and logging
- WebRtc_UWord8 testMode;
- WebRtc_UWord8 codeId;
-
- private:
- AudioCodingModule* _acm;
- PCMFile _pcmFile;
- AudioFrame _audioFrame;
- WebRtc_UWord16 _payloadSize;
- WebRtc_UWord32 _timeStamp;
- TestPacketization* _packetization;
-};
-
-class Receiver {
- public:
- Receiver();
- void Setup(AudioCodingModule *acm, RTPStream *rtpStream);
- void Teardown();
- void Run();
- bool IncomingPacket();
- bool PlayoutData();
-
- //for auto_test and logging
- WebRtc_UWord8 codeId;
- WebRtc_UWord8 testMode;
-
- private:
- AudioCodingModule* _acm;
- bool _rtpEOF;
- RTPStream* _rtpStream;
- PCMFile _pcmFile;
- WebRtc_Word16* _playoutBuffer;
- WebRtc_UWord16 _playoutLengthSmpls;
- WebRtc_Word8 _incomingPayload[MAX_INCOMING_PAYLOAD];
- WebRtc_UWord16 _payloadSizeBytes;
- WebRtc_UWord16 _realPayloadSizeBytes;
- WebRtc_Word32 _frequency;
- bool _firstTime;
- WebRtcRTPHeader _rtpInfo;
- WebRtc_UWord32 _nextTime;
-};
-
-class EncodeDecodeTest: public ACMTest {
- public:
- EncodeDecodeTest();
- EncodeDecodeTest(int testMode);
- virtual void Perform();
-
- WebRtc_UWord16 _playoutFreq;
- WebRtc_UWord8 _testMode;
-
- private:
- void EncodeToFile(int fileType, int codeId, int* codePars, int testMode);
-
- protected:
- Sender _sender;
- Receiver _receiver;
-};
-
-} // namespace webrtc
-
-#endif
diff --git a/trunk/src/modules/audio_coding/main/test/PCMFile.cc b/trunk/src/modules/audio_coding/main/test/PCMFile.cc
deleted file mode 100644
index 520ddbe..0000000
--- a/trunk/src/modules/audio_coding/main/test/PCMFile.cc
+++ /dev/null
@@ -1,302 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "PCMFile.h"
-
-#include <cctype>
-#include <stdio.h>
-#include <string.h>
-
-#include "gtest/gtest.h"
-#include "module_common_types.h"
-
-namespace webrtc {
-
-#define MAX_FILE_NAME_LENGTH_BYTE 500
-
-PCMFile::PCMFile():
-_pcmFile(NULL),
-_nSamples10Ms(160),
-_frequency(16000),
-_endOfFile(false),
-_autoRewind(false),
-_rewinded(false),
-_timestamp(0),
-_readStereo(false),
-_saveStereo(false)
-{
- _timestamp = (((WebRtc_UWord32)rand() & 0x0000FFFF) << 16) |
- ((WebRtc_UWord32)rand() & 0x0000FFFF);
-}
-
-/*
-PCMFile::~PCMFile()
-{
- if(_pcmFile != NULL)
- {
- fclose(_pcmFile);
- _pcmFile = NULL;
- }
-}
-*/
-
-WebRtc_Word16
-PCMFile::ChooseFile(
- char* fileName,
- WebRtc_Word16 maxLen)
-{
- char tmpName[MAX_FILE_NAME_LENGTH_BYTE];
- //strcpy(_fileName, "in.pcm");
- //printf("\n\nPlease enter the input file: ");
- EXPECT_TRUE(fgets(tmpName, MAX_FILE_NAME_LENGTH_BYTE, stdin) != NULL);
- tmpName[MAX_FILE_NAME_LENGTH_BYTE-1] = '\0';
- WebRtc_Word16 n = 0;
-
- // removing leading spaces
- while((isspace(tmpName[n]) || iscntrl(tmpName[n])) &&
- (tmpName[n] != 0) &&
- (n < MAX_FILE_NAME_LENGTH_BYTE))
- {
- n++;
- }
- if(n > 0)
- {
- memmove(tmpName, &tmpName[n], MAX_FILE_NAME_LENGTH_BYTE - n);
- }
-
- //removing trailing spaces
- n = (WebRtc_Word16)(strlen(tmpName) - 1);
- if(n >= 0)
- {
- while((isspace(tmpName[n]) || iscntrl(tmpName[n])) &&
- (n >= 0))
- {
- n--;
- }
- }
- if(n >= 0)
- {
- tmpName[n + 1] = '\0';
- }
-
- WebRtc_Word16 len = (WebRtc_Word16)strlen(tmpName);
- if(len > maxLen)
- {
- return -1;
- }
- if(len > 0)
- {
- strncpy(fileName, tmpName, len+1);
- }
- return 0;
-}
-
-WebRtc_Word16
-PCMFile::ChooseFile(
- char* fileName,
- WebRtc_Word16 maxLen,
- WebRtc_UWord16* frequencyHz)
-{
- char tmpName[MAX_FILE_NAME_LENGTH_BYTE];
- //strcpy(_fileName, "in.pcm");
- //printf("\n\nPlease enter the input file: ");
- EXPECT_TRUE(fgets(tmpName, MAX_FILE_NAME_LENGTH_BYTE, stdin) != NULL);
- tmpName[MAX_FILE_NAME_LENGTH_BYTE-1] = '\0';
- WebRtc_Word16 n = 0;
-
- // removing leading spaces
- while((isspace(tmpName[n]) || iscntrl(tmpName[n])) &&
- (tmpName[n] != 0) &&
- (n < MAX_FILE_NAME_LENGTH_BYTE))
- {
- n++;
- }
- if(n > 0)
- {
- memmove(tmpName, &tmpName[n], MAX_FILE_NAME_LENGTH_BYTE - n);
- }
-
- //removing trailing spaces
- n = (WebRtc_Word16)(strlen(tmpName) - 1);
- if(n >= 0)
- {
- while((isspace(tmpName[n]) || iscntrl(tmpName[n])) &&
- (n >= 0))
- {
- n--;
- }
- }
- if(n >= 0)
- {
- tmpName[n + 1] = '\0';
- }
-
- WebRtc_Word16 len = (WebRtc_Word16)strlen(tmpName);
- if(len > maxLen)
- {
- return -1;
- }
- if(len > 0)
- {
- strncpy(fileName, tmpName, len+1);
- }
- printf("Enter the sampling frequency (in Hz) of the above file [%u]: ", *frequencyHz);
- EXPECT_TRUE(fgets(tmpName, 10, stdin) != NULL);
- WebRtc_UWord16 tmpFreq = (WebRtc_UWord16)atoi(tmpName);
- if(tmpFreq > 0)
- {
- *frequencyHz = tmpFreq;
- }
- return 0;
-}
-
-void
-PCMFile::Open(
- const char* filename,
- WebRtc_UWord16 frequency,
- const char* mode,
- bool autoRewind)
-{
- if ((_pcmFile = fopen(filename, mode)) == NULL)
- {
- printf("Cannot open file %s.\n", filename);
- ADD_FAILURE() << "Unable to read file";
- }
- _frequency = frequency;
- _nSamples10Ms = (WebRtc_UWord16)(_frequency / 100);
- _autoRewind = autoRewind;
- _endOfFile = false;
- _rewinded = false;
-}
-
-WebRtc_Word32
-PCMFile::SamplingFrequency() const
-{
- return _frequency;
-}
-
-WebRtc_UWord16
-PCMFile::PayloadLength10Ms() const
-{
- return _nSamples10Ms;
-}
-
-WebRtc_Word32
-PCMFile::Read10MsData(
- AudioFrame& audioFrame)
-{
- WebRtc_UWord16 noChannels = 1;
- if (_readStereo)
- {
- noChannels = 2;
- }
-
- WebRtc_Word32 payloadSize = (WebRtc_Word32)fread(audioFrame._payloadData, sizeof(WebRtc_UWord16), _nSamples10Ms*noChannels, _pcmFile);
- if (payloadSize < _nSamples10Ms*noChannels) {
- for (int k = payloadSize; k < _nSamples10Ms*noChannels; k++)
- {
- audioFrame._payloadData[k] = 0;
- }
- if(_autoRewind)
- {
- rewind(_pcmFile);
- _rewinded = true;
- }
- else
- {
- _endOfFile = true;
- }
- }
- audioFrame._payloadDataLengthInSamples = _nSamples10Ms;
- audioFrame._frequencyInHz = _frequency;
- audioFrame._audioChannel = noChannels;
- audioFrame._timeStamp = _timestamp;
- _timestamp += _nSamples10Ms;
- return _nSamples10Ms;
-}
-
-void
-PCMFile::Write10MsData(
- AudioFrame& audioFrame)
-{
- if(audioFrame._audioChannel == 1)
- {
- if(!_saveStereo)
- {
- fwrite(audioFrame._payloadData, sizeof(WebRtc_UWord16),
- audioFrame._payloadDataLengthInSamples, _pcmFile);
- }
- else
- {
- WebRtc_Word16* stereoAudio = new WebRtc_Word16[2 *
- audioFrame._payloadDataLengthInSamples];
- int k;
- for(k = 0; k < audioFrame._payloadDataLengthInSamples; k++)
- {
- stereoAudio[k<<1] = audioFrame._payloadData[k];
- stereoAudio[(k<<1) + 1] = audioFrame._payloadData[k];
- }
- fwrite(stereoAudio, sizeof(WebRtc_Word16), 2*audioFrame._payloadDataLengthInSamples,
- _pcmFile);
- delete [] stereoAudio;
- }
- }
- else
- {
- fwrite(audioFrame._payloadData, sizeof(WebRtc_Word16),
- audioFrame._audioChannel * audioFrame._payloadDataLengthInSamples, _pcmFile);
- }
-}
-
-
-void
-PCMFile::Write10MsData(
- WebRtc_Word16* playoutBuffer,
- WebRtc_UWord16 playoutLengthSmpls)
-{
- fwrite(playoutBuffer, sizeof(WebRtc_UWord16), playoutLengthSmpls, _pcmFile);
-}
-
-
-void
-PCMFile::Close()
-{
- fclose(_pcmFile);
- _pcmFile = NULL;
-}
-
-void
-PCMFile::Rewind()
-{
- rewind(_pcmFile);
- _endOfFile = false;
-}
-
-bool
-PCMFile::Rewinded()
-{
- return _rewinded;
-}
-
-void
-PCMFile::SaveStereo(
- bool saveStereo)
-{
- _saveStereo = saveStereo;
-}
-
-void
-PCMFile::ReadStereo(
- bool readStereo)
-{
- _readStereo = readStereo;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/PCMFile.h b/trunk/src/modules/audio_coding/main/test/PCMFile.h
deleted file mode 100644
index a9cb9cf..0000000
--- a/trunk/src/modules/audio_coding/main/test/PCMFile.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef PCMFILE_H
-#define PCMFILE_H
-
-#include "typedefs.h"
-#include "module_common_types.h"
-#include <cstdio>
-#include <cstdlib>
-
-namespace webrtc {
-
-class PCMFile
-{
-public:
- PCMFile();
- ~PCMFile()
- {
- if(_pcmFile != NULL)
- {
- fclose(_pcmFile);
- }
- }
- void Open(const char *filename, WebRtc_UWord16 frequency, const char *mode,
- bool autoRewind = false);
-
- WebRtc_Word32 Read10MsData(AudioFrame& audioFrame);
-
- void Write10MsData(WebRtc_Word16 *playoutBuffer, WebRtc_UWord16 playoutLengthSmpls);
- void Write10MsData(AudioFrame& audioFrame);
-
- WebRtc_UWord16 PayloadLength10Ms() const;
- WebRtc_Word32 SamplingFrequency() const;
- void Close();
- bool EndOfFile() const { return _endOfFile; }
- void Rewind();
- static WebRtc_Word16 ChooseFile(char* fileName, WebRtc_Word16 maxLen,
- WebRtc_UWord16* frequencyHz);
- static WebRtc_Word16 ChooseFile(char* fileName, WebRtc_Word16 maxLen);
- bool Rewinded();
- void SaveStereo(
- bool saveStereo = true);
- void ReadStereo(
- bool readStereo = true);
-private:
- FILE* _pcmFile;
- WebRtc_UWord16 _nSamples10Ms;
- WebRtc_Word32 _frequency;
- bool _endOfFile;
- bool _autoRewind;
- bool _rewinded;
- WebRtc_UWord32 _timestamp;
- bool _readStereo;
- bool _saveStereo;
-};
-
-} // namespace webrtc
-
-#endif
diff --git a/trunk/src/modules/audio_coding/main/test/RTPFile.cc b/trunk/src/modules/audio_coding/main/test/RTPFile.cc
deleted file mode 100644
index 29d6a1d..0000000
--- a/trunk/src/modules/audio_coding/main/test/RTPFile.cc
+++ /dev/null
@@ -1,260 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "RTPFile.h"
-
-#include <stdlib.h>
-
-#ifdef WIN32
-# include <Winsock2.h>
-#else
-# include <arpa/inet.h>
-#endif
-
-#include "audio_coding_module.h"
-#include "engine_configurations.h"
-#include "gtest/gtest.h" // TODO (tlegrand): Consider removing usage of gtest.
-#include "rw_lock_wrapper.h"
-
-namespace webrtc {
-
-void RTPStream::ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const WebRtc_UWord8* rtpHeader)
-{
- rtpInfo->header.payloadType = rtpHeader[1];
- rtpInfo->header.sequenceNumber = (static_cast<WebRtc_UWord16>(rtpHeader[2])<<8) | rtpHeader[3];
- rtpInfo->header.timestamp = (static_cast<WebRtc_UWord32>(rtpHeader[4])<<24) |
- (static_cast<WebRtc_UWord32>(rtpHeader[5])<<16) |
- (static_cast<WebRtc_UWord32>(rtpHeader[6])<<8) |
- rtpHeader[7];
- rtpInfo->header.ssrc = (static_cast<WebRtc_UWord32>(rtpHeader[8])<<24) |
- (static_cast<WebRtc_UWord32>(rtpHeader[9])<<16) |
- (static_cast<WebRtc_UWord32>(rtpHeader[10])<<8) |
- rtpHeader[11];
-}
-
-void RTPStream::MakeRTPheader(WebRtc_UWord8* rtpHeader,
- WebRtc_UWord8 payloadType, WebRtc_Word16 seqNo,
- WebRtc_UWord32 timeStamp, WebRtc_UWord32 ssrc)
-{
- rtpHeader[0]=(unsigned char)0x80;
- rtpHeader[1]=(unsigned char)(payloadType & 0xFF);
- rtpHeader[2]=(unsigned char)((seqNo>>8)&0xFF);
- rtpHeader[3]=(unsigned char)((seqNo)&0xFF);
- rtpHeader[4]=(unsigned char)((timeStamp>>24)&0xFF);
- rtpHeader[5]=(unsigned char)((timeStamp>>16)&0xFF);
-
- rtpHeader[6]=(unsigned char)((timeStamp>>8)&0xFF);
- rtpHeader[7]=(unsigned char)(timeStamp & 0xFF);
-
- rtpHeader[8]=(unsigned char)((ssrc>>24)&0xFF);
- rtpHeader[9]=(unsigned char)((ssrc>>16)&0xFF);
-
- rtpHeader[10]=(unsigned char)((ssrc>>8)&0xFF);
- rtpHeader[11]=(unsigned char)(ssrc & 0xFF);
-}
-
-
-RTPPacket::RTPPacket(WebRtc_UWord8 payloadType, WebRtc_UWord32 timeStamp,
- WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
- WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency)
- :
-payloadType(payloadType),
-timeStamp(timeStamp),
-seqNo(seqNo),
-payloadSize(payloadSize),
-frequency(frequency)
-{
- if (payloadSize > 0)
- {
- this->payloadData = new WebRtc_UWord8[payloadSize];
- memcpy(this->payloadData, payloadData, payloadSize);
- }
-}
-
-RTPPacket::~RTPPacket()
-{
- delete [] payloadData;
-}
-
-RTPBuffer::RTPBuffer()
-{
- _queueRWLock = RWLockWrapper::CreateRWLock();
-}
-
-RTPBuffer::~RTPBuffer()
-{
- delete _queueRWLock;
-}
-
-void
-RTPBuffer::Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
- const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency)
-{
- RTPPacket *packet = new RTPPacket(payloadType, timeStamp, seqNo, payloadData, payloadSize, frequency);
- _queueRWLock->AcquireLockExclusive();
- _rtpQueue.push(packet);
- _queueRWLock->ReleaseLockExclusive();
-}
-
-WebRtc_UWord16
-RTPBuffer::Read(WebRtcRTPHeader* rtpInfo,
- WebRtc_Word8* payloadData,
- WebRtc_UWord16 payloadSize,
- WebRtc_UWord32* offset)
-{
- _queueRWLock->AcquireLockShared();
- RTPPacket *packet = _rtpQueue.front();
- _rtpQueue.pop();
- _queueRWLock->ReleaseLockShared();
- rtpInfo->header.markerBit = 1;
- rtpInfo->header.payloadType = packet->payloadType;
- rtpInfo->header.sequenceNumber = packet->seqNo;
- rtpInfo->header.ssrc = 0;
- rtpInfo->header.timestamp = packet->timeStamp;
- if (packet->payloadSize > 0 && payloadSize >= packet->payloadSize)
- {
- memcpy(payloadData, packet->payloadData, packet->payloadSize);
- }
- else
- {
- return 0;
- }
- *offset = (packet->timeStamp/(packet->frequency/1000));
-
- return packet->payloadSize;
-}
-
-bool
-RTPBuffer::EndOfFile() const
-{
- _queueRWLock->AcquireLockShared();
- bool eof = _rtpQueue.empty();
- _queueRWLock->ReleaseLockShared();
- return eof;
-}
-
-void RTPFile::Open(const char *filename, const char *mode)
-{
- if ((_rtpFile = fopen(filename, mode)) == NULL)
- {
- printf("Cannot write file %s.\n", filename);
- ADD_FAILURE() << "Unable to write file";
- exit(1);
- }
-}
-
-void RTPFile::Close()
-{
- if (_rtpFile != NULL)
- {
- fclose(_rtpFile);
- _rtpFile = NULL;
- }
-}
-
-
-void RTPFile::WriteHeader()
-{
- // Write data in a format that NetEQ and RTP Play can parse
- fprintf(_rtpFile, "#!RTPencode%s\n", "1.0");
- WebRtc_UWord32 dummy_variable = 0; // should be converted to network endian format, but does not matter when 0
- fwrite(&dummy_variable, 4, 1, _rtpFile);
- fwrite(&dummy_variable, 4, 1, _rtpFile);
- fwrite(&dummy_variable, 4, 1, _rtpFile);
- fwrite(&dummy_variable, 2, 1, _rtpFile);
- fwrite(&dummy_variable, 2, 1, _rtpFile);
- fflush(_rtpFile);
-}
-
-void RTPFile::ReadHeader()
-{
- WebRtc_UWord32 start_sec, start_usec, source;
- WebRtc_UWord16 port, padding;
- char fileHeader[40];
- EXPECT_TRUE(fgets(fileHeader, 40, _rtpFile) != 0);
- EXPECT_EQ(1u, fread(&start_sec, 4, 1, _rtpFile));
- start_sec=ntohl(start_sec);
- EXPECT_EQ(1u, fread(&start_usec, 4, 1, _rtpFile));
- start_usec=ntohl(start_usec);
- EXPECT_EQ(1u, fread(&source, 4, 1, _rtpFile));
- source=ntohl(source);
- EXPECT_EQ(1u, fread(&port, 2, 1, _rtpFile));
- port=ntohs(port);
- EXPECT_EQ(1u, fread(&padding, 2, 1, _rtpFile));
- padding=ntohs(padding);
-}
-
-void RTPFile::Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
- const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency)
-{
- /* write RTP packet to file */
- WebRtc_UWord8 rtpHeader[12];
- MakeRTPheader(rtpHeader, payloadType, seqNo, timeStamp, 0);
- WebRtc_UWord16 lengthBytes = htons(12 + payloadSize + 8);
- WebRtc_UWord16 plen = htons(12 + payloadSize);
- WebRtc_UWord32 offsetMs;
-
- offsetMs = (timeStamp/(frequency/1000));
- offsetMs = htonl(offsetMs);
- fwrite(&lengthBytes, 2, 1, _rtpFile);
- fwrite(&plen, 2, 1, _rtpFile);
- fwrite(&offsetMs, 4, 1, _rtpFile);
- fwrite(rtpHeader, 12, 1, _rtpFile);
- fwrite(payloadData, 1, payloadSize, _rtpFile);
-}
-
-WebRtc_UWord16 RTPFile::Read(WebRtcRTPHeader* rtpInfo,
- WebRtc_Word8* payloadData,
- WebRtc_UWord16 payloadSize,
- WebRtc_UWord32* offset)
-{
- WebRtc_UWord16 lengthBytes;
- WebRtc_UWord16 plen;
- WebRtc_UWord8 rtpHeader[12];
- size_t read_len = fread(&lengthBytes, 2, 1, _rtpFile);
- /* Check if we have reached end of file. */
- if ((read_len == 0) && feof(_rtpFile))
- {
- _rtpEOF = true;
- return 0;
- }
- EXPECT_EQ(1u, fread(&plen, 2, 1, _rtpFile));
- EXPECT_EQ(1u, fread(offset, 4, 1, _rtpFile));
- lengthBytes = ntohs(lengthBytes);
- plen = ntohs(plen);
- *offset = ntohl(*offset);
- EXPECT_GT(plen, 11);
-
- EXPECT_EQ(1u, fread(rtpHeader, 12, 1, _rtpFile));
- ParseRTPHeader(rtpInfo, rtpHeader);
- rtpInfo->type.Audio.isCNG = false;
- rtpInfo->type.Audio.channel = 1;
- EXPECT_EQ(lengthBytes, plen + 8);
-
- if (plen == 0)
- {
- return 0;
- }
- if (payloadSize < (lengthBytes - 20))
- {
- return -1;
- }
- if (lengthBytes < 20)
- {
- return -1;
- }
- lengthBytes -= 20;
- EXPECT_EQ(lengthBytes, fread(payloadData, 1, lengthBytes, _rtpFile));
- return lengthBytes;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/RTPFile.h b/trunk/src/modules/audio_coding/main/test/RTPFile.h
deleted file mode 100644
index e11b160..0000000
--- a/trunk/src/modules/audio_coding/main/test/RTPFile.h
+++ /dev/null
@@ -1,103 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef RTPFILE_H
-#define RTPFILE_H
-
-#include "audio_coding_module.h"
-#include "module_common_types.h"
-#include "typedefs.h"
-#include "rw_lock_wrapper.h"
-#include <stdio.h>
-#include <queue>
-
-namespace webrtc {
-
-class RTPStream
-{
-public:
- virtual ~RTPStream(){}
-
- virtual void Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
- const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency) = 0;
-
- // Returns the packet's payload size. Zero should be treated as an
- // end-of-stream (in the case that EndOfFile() is true) or an error.
- virtual WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo,
- WebRtc_Word8* payloadData,
- WebRtc_UWord16 payloadSize,
- WebRtc_UWord32* offset) = 0;
- virtual bool EndOfFile() const = 0;
-
-protected:
- void MakeRTPheader(WebRtc_UWord8* rtpHeader,
- WebRtc_UWord8 payloadType, WebRtc_Word16 seqNo,
- WebRtc_UWord32 timeStamp, WebRtc_UWord32 ssrc);
- void ParseRTPHeader(WebRtcRTPHeader* rtpInfo, const WebRtc_UWord8* rtpHeader);
-};
-
-class RTPPacket
-{
-public:
- RTPPacket(WebRtc_UWord8 payloadType, WebRtc_UWord32 timeStamp,
- WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
- WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency);
- ~RTPPacket();
- WebRtc_UWord8 payloadType;
- WebRtc_UWord32 timeStamp;
- WebRtc_Word16 seqNo;
- WebRtc_UWord8* payloadData;
- WebRtc_UWord16 payloadSize;
- WebRtc_UWord32 frequency;
-};
-
-class RTPBuffer : public RTPStream
-{
-public:
- RTPBuffer();
- ~RTPBuffer();
- void Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
- const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency);
- WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo,
- WebRtc_Word8* payloadData,
- WebRtc_UWord16 payloadSize,
- WebRtc_UWord32* offset);
- virtual bool EndOfFile() const;
-private:
- RWLockWrapper* _queueRWLock;
- std::queue<RTPPacket *> _rtpQueue;
-};
-
-class RTPFile : public RTPStream
-{
-public:
- ~RTPFile(){}
- RTPFile() : _rtpFile(NULL),_rtpEOF(false) {}
- void Open(const char *outFilename, const char *mode);
- void Close();
- void WriteHeader();
- void ReadHeader();
- void Write(const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
- const WebRtc_Word16 seqNo, const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize, WebRtc_UWord32 frequency);
- WebRtc_UWord16 Read(WebRtcRTPHeader* rtpInfo,
- WebRtc_Word8* payloadData,
- WebRtc_UWord16 payloadSize,
- WebRtc_UWord32* offset);
- bool EndOfFile() const { return _rtpEOF; }
-private:
- FILE* _rtpFile;
- bool _rtpEOF;
-};
-
-} // namespace webrtc
-#endif
diff --git a/trunk/src/modules/audio_coding/main/test/SpatialAudio.cc b/trunk/src/modules/audio_coding/main/test/SpatialAudio.cc
deleted file mode 100644
index e19ca44..0000000
--- a/trunk/src/modules/audio_coding/main/test/SpatialAudio.cc
+++ /dev/null
@@ -1,246 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <string.h>
-
-#include <math.h>
-
-#include "common_types.h"
-#include "SpatialAudio.h"
-#include "trace.h"
-#include "testsupport/fileutils.h"
-#include "utility.h"
-
-namespace webrtc {
-
-#define NUM_PANN_COEFFS 10
-
-SpatialAudio::SpatialAudio(int testMode)
-{
- _testMode = testMode;
-}
-
-SpatialAudio::~SpatialAudio()
-{
- AudioCodingModule::Destroy(_acmLeft);
- AudioCodingModule::Destroy(_acmRight);
- AudioCodingModule::Destroy(_acmReceiver);
- delete _channel;
- _inFile.Close();
- _outFile.Close();
-}
-
-WebRtc_Word16
-SpatialAudio::Setup()
-{
- // Create ACMs and the Channel;
- _acmLeft = AudioCodingModule::Create(1);
- _acmRight = AudioCodingModule::Create(2);
- _acmReceiver = AudioCodingModule::Create(3);
- _channel = new Channel;
-
- // Register callback for the sender side.
- CHECK_ERROR(_acmLeft->RegisterTransportCallback(_channel));
- CHECK_ERROR(_acmRight->RegisterTransportCallback(_channel));
- // Register the receiver ACM in channel
- _channel->RegisterReceiverACM(_acmReceiver);
-
- char audioFileName[MAX_FILE_NAME_LENGTH_BYTE];
- WebRtc_UWord16 sampFreqHz = 32000;
-
- strncpy(audioFileName, "./test/data/audio_coding/testfile32kHz.pcm",
- MAX_FILE_NAME_LENGTH_BYTE - 1);
- if(_testMode == 1)
- {
- printf("Enter the input file [%s]: ", audioFileName);
- PCMFile::ChooseFile(audioFileName, MAX_FILE_NAME_LENGTH_BYTE, &sampFreqHz);
- }
- _inFile.Open(audioFileName, sampFreqHz, "rb", false);
-
- if(_testMode == 0)
- {
- std::string outputFile = webrtc::test::OutputPath() +
- "out_spatial_autotest.pcm";
- strncpy(audioFileName, outputFile.c_str(),
- MAX_FILE_NAME_LENGTH_BYTE - 1);
- }
- else if(_testMode == 1)
- {
- printf("\n");
- std::string outputFile = webrtc::test::OutputPath() +
- "testspatial_out.pcm";
- strncpy(audioFileName, outputFile.c_str(),
- MAX_FILE_NAME_LENGTH_BYTE - 1);
- printf("Enter the output file [%s]: ", audioFileName);
- PCMFile::ChooseFile(audioFileName, MAX_FILE_NAME_LENGTH_BYTE, &sampFreqHz);
- }
- else
- {
- std::string outputFile = webrtc::test::OutputPath() +
- "testspatial_out.pcm";
- strncpy(audioFileName, outputFile.c_str(),
- MAX_FILE_NAME_LENGTH_BYTE - 1);
- }
- _outFile.Open(audioFileName, sampFreqHz, "wb", false);
- _outFile.SaveStereo(true);
-
-
- // Register couple of codecs as receive codec
- CodecInst codecInst;
-
- _acmLeft->Codec((WebRtc_UWord8)0, codecInst);
- codecInst.channels = 2;
- CHECK_ERROR(_acmReceiver->RegisterReceiveCodec(codecInst));
-
- _acmLeft->Codec((WebRtc_UWord8)3, codecInst);
- codecInst.channels = 2;
- CHECK_ERROR(_acmReceiver->RegisterReceiveCodec(codecInst));
-
- _acmLeft->Codec((WebRtc_UWord8)1, codecInst);
- CHECK_ERROR(_acmReceiver->RegisterReceiveCodec(codecInst));
-
- _acmLeft->Codec((WebRtc_UWord8)4, codecInst);
- CHECK_ERROR(_acmReceiver->RegisterReceiveCodec(codecInst));
-
- return 0;
-}
-
-void
-SpatialAudio::Perform()
-{
- if(_testMode == 0)
- {
- printf("Running SpatialAudio Test");
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1, "---------- SpatialAudio ----------");
- }
-
- Setup();
-
- CodecInst codecInst;
- _acmLeft->Codec((WebRtc_UWord8)1, codecInst);
- CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
- EncodeDecode();
-
- WebRtc_Word16 pannCntr = 0;
-
- double leftPanning[NUM_PANN_COEFFS] =
- {1.00, 0.95, 0.90, 0.85, 0.80, 0.75, 0.70, 0.60, 0.55, 0.50};
- double rightPanning[NUM_PANN_COEFFS] =
- {0.50, 0.55, 0.60, 0.70, 0.75, 0.80, 0.85, 0.90, 0.95, 1.00};
-
- while((pannCntr + 1) < NUM_PANN_COEFFS)
- {
- _acmLeft->Codec((WebRtc_UWord8)0, codecInst);
- codecInst.pacsize = 480;
- CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
- CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
-
- EncodeDecode(leftPanning[pannCntr], rightPanning[pannCntr]);
- pannCntr++;
-
- // Change codec
- _acmLeft->Codec((WebRtc_UWord8)3, codecInst);
- codecInst.pacsize = 320;
- CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
- CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
-
- EncodeDecode(leftPanning[pannCntr], rightPanning[pannCntr]);
- pannCntr++;
- if(_testMode == 0)
- {
- printf(".");
- }
- }
-
- _acmLeft->Codec((WebRtc_UWord8)4, codecInst);
- CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
- EncodeDecode();
-
- _acmLeft->Codec((WebRtc_UWord8)0, codecInst);
- codecInst.pacsize = 480;
- CHECK_ERROR(_acmLeft->RegisterSendCodec(codecInst));
- CHECK_ERROR(_acmRight->RegisterSendCodec(codecInst));
- pannCntr = NUM_PANN_COEFFS -1;
- while(pannCntr >= 0)
- {
- EncodeDecode(leftPanning[pannCntr], rightPanning[pannCntr]);
- pannCntr--;
- if(_testMode == 0)
- {
- printf(".");
- }
- }
- if(_testMode == 0)
- {
- printf("Done!\n");
- }
-}
-
-void
-SpatialAudio::EncodeDecode(
- const double leftPanning,
- const double rightPanning)
-{
- AudioFrame audioFrame;
- WebRtc_Word32 outFileSampFreq = _outFile.SamplingFrequency();
-
- const double rightToLeftRatio = rightPanning / leftPanning;
-
- _channel->SetIsStereo(true);
-
- while(!_inFile.EndOfFile())
- {
- _inFile.Read10MsData(audioFrame);
- for(int n = 0; n < audioFrame._payloadDataLengthInSamples; n++)
- {
- audioFrame._payloadData[n] = (WebRtc_Word16)floor(
- audioFrame._payloadData[n] * leftPanning + 0.5);
- }
- CHECK_ERROR(_acmLeft->Add10MsData(audioFrame));
-
- for(int n = 0; n < audioFrame._payloadDataLengthInSamples; n++)
- {
- audioFrame._payloadData[n] = (WebRtc_Word16)floor(
- audioFrame._payloadData[n] * rightToLeftRatio + 0.5);
- }
- CHECK_ERROR(_acmRight->Add10MsData(audioFrame));
-
- CHECK_ERROR(_acmLeft->Process());
- CHECK_ERROR(_acmRight->Process());
-
- CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq, audioFrame));
- _outFile.Write10MsData(audioFrame);
- }
- _inFile.Rewind();
-}
-
-void
-SpatialAudio::EncodeDecode()
-{
- AudioFrame audioFrame;
- WebRtc_Word32 outFileSampFreq = _outFile.SamplingFrequency();
-
- _channel->SetIsStereo(false);
-
- while(!_inFile.EndOfFile())
- {
- _inFile.Read10MsData(audioFrame);
- CHECK_ERROR(_acmLeft->Add10MsData(audioFrame));
-
- CHECK_ERROR(_acmLeft->Process());
-
- CHECK_ERROR(_acmReceiver->PlayoutData10Ms(outFileSampFreq, audioFrame));
- _outFile.Write10MsData(audioFrame);
- }
- _inFile.Rewind();
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/SpatialAudio.h b/trunk/src/modules/audio_coding/main/test/SpatialAudio.h
deleted file mode 100644
index 6a88327..0000000
--- a/trunk/src/modules/audio_coding/main/test/SpatialAudio.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef ACM_TEST_SPATIAL_AUDIO_H
-#define ACM_TEST_SPATIAL_AUDIO_H
-
-#include "ACMTest.h"
-#include "Channel.h"
-#include "PCMFile.h"
-#include "audio_coding_module.h"
-#include "utility.h"
-
-#define MAX_FILE_NAME_LENGTH_BYTE 500
-
-namespace webrtc {
-
-class SpatialAudio : public ACMTest
-{
-public:
- SpatialAudio(int testMode);
- ~SpatialAudio();
-
- void Perform();
-private:
- WebRtc_Word16 Setup();
- void EncodeDecode(double leftPanning, double rightPanning);
- void EncodeDecode();
-
- AudioCodingModule* _acmLeft;
- AudioCodingModule* _acmRight;
- AudioCodingModule* _acmReceiver;
- Channel* _channel;
- PCMFile _inFile;
- PCMFile _outFile;
- int _testMode;
-};
-
-} // namespace webrtc
-
-#endif
diff --git a/trunk/src/modules/audio_coding/main/test/TestAllCodecs.cc b/trunk/src/modules/audio_coding/main/test/TestAllCodecs.cc
deleted file mode 100644
index 9d7f1e3..0000000
--- a/trunk/src/modules/audio_coding/main/test/TestAllCodecs.cc
+++ /dev/null
@@ -1,872 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "TestAllCodecs.h"
-
-#include <cassert>
-#include <iostream>
-
-#include "audio_coding_module_typedefs.h"
-#include "common_types.h"
-#include "engine_configurations.h"
-#include "testsupport/fileutils.h"
-#include "trace.h"
-#include "utility.h"
-
-namespace webrtc {
-
-// Class for simulating packet handling
-TestPack::TestPack():
-_receiverACM(NULL),
-_seqNo(0),
-_timeStampDiff(0),
-_lastInTimestamp(0),
-_totalBytes(0),
-_payloadSize(0)
-{
-}
-TestPack::~TestPack()
-{
-}
-
-void
-TestPack::RegisterReceiverACM(AudioCodingModule* acm)
-{
- _receiverACM = acm;
- return;
-}
-WebRtc_Word32
-TestPack::SendData(
- const FrameType frameType,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const RTPFragmentationHeader* fragmentation)
-{
- WebRtcRTPHeader rtpInfo;
- WebRtc_Word32 status;
- WebRtc_UWord16 payloadDataSize = payloadSize;
-
- rtpInfo.header.markerBit = false;
- rtpInfo.header.ssrc = 0;
- rtpInfo.header.sequenceNumber = _seqNo++;
- rtpInfo.header.payloadType = payloadType;
- rtpInfo.header.timestamp = timeStamp;
- if(frameType == kAudioFrameCN)
- {
- rtpInfo.type.Audio.isCNG = true;
- }
- else
- {
- rtpInfo.type.Audio.isCNG = false;
- }
- if(frameType == kFrameEmpty)
- {
- // Skip this frame
- return 0;
- }
-
- rtpInfo.type.Audio.channel = 1;
- memcpy(_payloadData, payloadData, payloadDataSize);
-
- status = _receiverACM->IncomingPacket((WebRtc_Word8*)_payloadData, payloadDataSize, rtpInfo);
-
- _payloadSize = payloadDataSize;
- _timeStampDiff = timeStamp - _lastInTimestamp;
- _lastInTimestamp = timeStamp;
- _totalBytes += payloadDataSize;
- return status;
-}
-
-WebRtc_UWord16
-TestPack::GetPayloadSize()
-{
- return _payloadSize;
-}
-
-
-WebRtc_UWord32
-TestPack::GetTimeStampDiff()
-{
- return _timeStampDiff;
-}
-
-void
-TestPack::ResetPayloadSize()
-{
- _payloadSize = 0;
-}
-
-TestAllCodecs::TestAllCodecs(int testMode):
-_acmA(NULL),
-_acmB(NULL),
-_channelA2B(NULL),
-_testCntr(0),
-_packSizeSamp(0),
-_packSizeBytes(0),
-_counter(0)
-{
- // testMode = 0 for silent test (auto test)
- _testMode = testMode;
-}
-
-TestAllCodecs::~TestAllCodecs()
-{
- if(_acmA != NULL)
- {
- AudioCodingModule::Destroy(_acmA);
- _acmA = NULL;
- }
- if(_acmB != NULL)
- {
- AudioCodingModule::Destroy(_acmB);
- _acmB = NULL;
- }
- if(_channelA2B != NULL)
- {
- delete _channelA2B;
- _channelA2B = NULL;
- }
-}
-
-void TestAllCodecs::Perform()
-{
-
- char file[] = "./test/data/audio_coding/testfile32kHz.pcm";
- _inFileA.Open(file, 32000, "rb");
-
- if(_testMode == 0)
- {
- printf("Running All Codecs Test");
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
- "---------- TestAllCodecs ----------");
- }
-
- _acmA = AudioCodingModule::Create(0);
- _acmB = AudioCodingModule::Create(1);
-
- _acmA->InitializeReceiver();
- _acmB->InitializeReceiver();
-
- WebRtc_UWord8 numEncoders = _acmA->NumberOfCodecs();
- CodecInst myCodecParam;
-
- for(WebRtc_UWord8 n = 0; n < numEncoders; n++)
- {
- _acmB->Codec(n, myCodecParam);
- _acmB->RegisterReceiveCodec(myCodecParam);
- }
-
- // Create and connect the channel
- _channelA2B = new TestPack;
- _acmA->RegisterTransportCallback(_channelA2B);
- _channelA2B->RegisterReceiverACM(_acmB);
-
- // All codecs are tested for all allowed sampling frequencies, rates and packet sizes
-#ifdef WEBRTC_CODEC_GSMAMR
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecAMR[] = "AMR";
- RegisterSendCodec('A', codecAMR, 8000, 4750, 160, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 4750, 320, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 4750, 480, 3);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 5150, 160, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 5150, 320, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 5150, 480, 3);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 5900, 160, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 5900, 320, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 5900, 480, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 6700, 160, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 6700, 320, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 6700, 480, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 7400, 160, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 7400, 320, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 7400, 480, 3);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 7950, 160, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 7950, 320, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 7950, 480, 3);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 10200, 160, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 10200, 320, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 10200, 480, 3);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 12200, 160, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 12200, 320, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMR, 8000, 12200, 480, 3);
- Run(_channelA2B);
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_GSMAMRWB
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- char codecAMRWB[] = "AMR-WB";
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecAMRWB, 16000, 7000, 320, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 7000, 640, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 7000, 960, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 9000, 320, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 9000, 640, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 9000, 960, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 12000, 320, 3);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 12000, 640, 6);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 12000, 960, 8);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 14000, 320, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 14000, 640, 4);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 14000, 960, 5);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 16000, 320, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 16000, 640, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 16000, 960, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 18000, 320, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 18000, 640, 4);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 18000, 960, 5);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 20000, 320, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 20000, 640, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 20000, 960, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 23000, 320, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 23000, 640, 3);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 23000, 960, 3);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 24000, 320, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 24000, 640, 2);
- Run(_channelA2B);
- RegisterSendCodec('A', codecAMRWB, 16000, 24000, 960, 2);
- Run(_channelA2B);
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_G722
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecG722[] = "G722";
- RegisterSendCodec('A', codecG722, 16000, 64000, 160, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG722, 16000, 64000, 320, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG722, 16000, 64000, 480, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG722, 16000, 64000, 640, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG722, 16000, 64000, 800, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG722, 16000, 64000, 960, 0);
- Run(_channelA2B);
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_G722_1
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecG7221_1[] = "G7221";
- RegisterSendCodec('A', codecG7221_1, 16000, 32000, 320, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7221_1, 16000, 24000, 320, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7221_1, 16000, 16000, 320, 0);
- Run(_channelA2B);
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_G722_1C
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecG7221_2[] = "G7221";
- RegisterSendCodec('A', codecG7221_2, 32000, 48000, 640, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7221_2, 32000, 32000, 640, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7221_2, 32000, 24000, 640, 0);
- Run(_channelA2B);
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_G729
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecG729[] = "G729";
- RegisterSendCodec('A', codecG729, 8000, 8000, 80, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG729, 8000, 8000, 160, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG729, 8000, 8000, 240, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG729, 8000, 8000, 320, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG729, 8000, 8000, 400, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG729, 8000, 8000, 480, 0);
- Run(_channelA2B);
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_G729_1
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecG7291[] = "G7291";
- RegisterSendCodec('A', codecG7291, 16000, 8000, 320, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 8000, 640, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 8000, 960, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 12000, 320, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 12000, 640, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 12000, 960, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 14000, 320, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 14000, 640, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 14000, 960, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 16000, 320, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 16000, 640, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 16000, 960, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 18000, 320, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 18000, 640, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 18000, 960, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 20000, 320, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 20000, 640, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 20000, 960, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 22000, 320, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 22000, 640, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 22000, 960, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 24000, 320, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 24000, 640, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 24000, 960, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 26000, 320, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 26000, 640, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 26000, 960, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 28000, 320, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 28000, 640, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 28000, 960, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 30000, 320, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 30000, 640, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 30000, 960, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 32000, 320, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 32000, 640, 1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecG7291, 16000, 32000, 960, 1);
- Run(_channelA2B);
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_GSMFR
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecGSM[] = "GSM";
- RegisterSendCodec('A', codecGSM, 8000, 13200, 160, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecGSM, 8000, 13200, 320, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecGSM, 8000, 13200, 480, 0);
- Run(_channelA2B);
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_ILBC
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecILBC[] = "ILBC";
- RegisterSendCodec('A', codecILBC, 8000, 13300, 240, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecILBC, 8000, 13300, 480, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecILBC, 8000, 15200, 160, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecILBC, 8000, 15200, 320, 0);
- Run(_channelA2B);
- _outFileB.Close();
-#endif
-#if (defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecISAC[] = "ISAC";
- RegisterSendCodec('A', codecISAC, 16000, -1, 480, -1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecISAC, 16000, -1, 960, -1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecISAC, 16000, 15000, 480, -1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecISAC, 16000, 32000, 960, -1);
- Run(_channelA2B);
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_ISAC
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecISAC, 32000, -1, 960, -1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecISAC, 32000, 56000, 960, -1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecISAC, 32000, 37000, 960, -1);
- Run(_channelA2B);
- RegisterSendCodec('A', codecISAC, 32000, 32000, 960, -1);
- Run(_channelA2B);
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_PCM16
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecL16[] = "L16";
- RegisterSendCodec('A', codecL16, 8000, 128000, 80, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecL16, 8000, 128000, 160, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecL16, 8000, 128000, 240, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecL16, 8000, 128000, 320, 0);
- Run(_channelA2B);
- _outFileB.Close();
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecL16, 16000, 256000, 160, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecL16, 16000, 256000, 320, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecL16, 16000, 256000, 480, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecL16, 16000, 256000, 640, 0);
- Run(_channelA2B);
- _outFileB.Close();
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecL16, 32000, 512000, 320, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecL16, 32000, 512000, 640, 0);
- Run(_channelA2B);
- _outFileB.Close();
-#endif
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecPCMA[] = "PCMA";
- RegisterSendCodec('A', codecPCMA, 8000, 64000, 80, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecPCMA, 8000, 64000, 160, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecPCMA, 8000, 64000, 240, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecPCMA, 8000, 64000, 320, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecPCMA, 8000, 64000, 400, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecPCMA, 8000, 64000, 480, 0);
- Run(_channelA2B);
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- char codecPCMU[] = "PCMU";
- RegisterSendCodec('A', codecPCMU, 8000, 64000, 80, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecPCMU, 8000, 64000, 160, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecPCMU, 8000, 64000, 240, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecPCMU, 8000, 64000, 320, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecPCMU, 8000, 64000, 400, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecPCMU, 8000, 64000, 480, 0);
- Run(_channelA2B);
- _outFileB.Close();
-#ifdef WEBRTC_CODEC_SPEEX
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecSPEEX[] = "SPEEX";
- RegisterSendCodec('A', codecSPEEX, 8000, 2400, 160, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecSPEEX, 8000, 8000, 320, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecSPEEX, 8000, 18200, 480, 0);
- Run(_channelA2B);
- _outFileB.Close();
-
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecSPEEX, 16000, 4000, 320, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecSPEEX, 16000, 12800, 640, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecSPEEX, 16000, 34200, 960, 0);
- Run(_channelA2B);
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_CELT
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecCELT_32[] = "CELT";
- RegisterSendCodec('A', codecCELT_32, 32000, 48000, 320, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecCELT_32, 32000, 64000, 320, 0);
- Run(_channelA2B);
- RegisterSendCodec('A', codecCELT_32, 32000, 128000, 320, 0);
- Run(_channelA2B);
- _outFileB.Close();
-#endif
- if(_testMode != 0) {
- printf("=======================================================================\n");
- } else {
- printf("Done!\n");
- }
-
- /* Print out all codecs that were not tested in the run */
- if(_testMode != 0) {
- printf("The following codecs was not included in the test:\n");
-#ifndef WEBRTC_CODEC_GSMAMR
- printf(" GSMAMR\n");
-#endif
-#ifndef WEBRTC_CODEC_GSMAMRWB
- printf(" GSMAMR-wb\n");
-#endif
-#ifndef WEBRTC_CODEC_G722
- printf(" G.722\n");
-#endif
-#ifndef WEBRTC_CODEC_G722_1
- printf(" G.722.1\n");
-#endif
-#ifndef WEBRTC_CODEC_G722_1C
- printf(" G.722.1C\n");
-#endif
-#ifndef WEBRTC_CODEC_G729
- printf(" G.729\n");
-#endif
-#ifndef WEBRTC_CODEC_G729_1
- printf(" G.729.1\n");
-#endif
-#ifndef WEBRTC_CODEC_GSMFR
- printf(" GSMFR\n");
-#endif
-#ifndef WEBRTC_CODEC_ILBC
- printf(" iLBC\n");
-#endif
-#ifndef WEBRTC_CODEC_ISAC
- printf(" ISAC float\n");
-#endif
-#ifndef WEBRTC_CODEC_ISACFX
- printf(" ISAC fix\n");
-#endif
-#ifndef WEBRTC_CODEC_PCM16
- printf(" PCM16\n");
-#endif
-#ifndef WEBRTC_CODEC_SPEEX
- printf(" Speex\n");
-#endif
-
- printf("\nTo complete the test, listen to the %d number of output files.\n", _testCntr);
- }
-}
-
-// Register Codec to use in the test
-//
-// Input: side - which ACM to use, 'A' or 'B'
-// codecName - name to use when register the codec
-// samplingFreqHz - sampling frequency in Herz
-// rate - bitrate in bytes
-// packSize - packet size in samples
-// extraByte - if extra bytes needed compared to the bitrate
-// used when registering, can be an internal header
-// set to -1 if the codec is a variable rate codec
-WebRtc_Word16 TestAllCodecs::RegisterSendCodec(char side,
- char* codecName,
- WebRtc_Word32 samplingFreqHz,
- int rate,
- int packSize,
- int extraByte)
-{
- if(_testMode != 0) {
- // Print out codec and settings
- printf("codec: %s Freq: %d Rate: %d PackSize: %d", codecName, samplingFreqHz, rate, packSize);
- }
-
- // Store packetsize in samples, used to validate the recieved packet
- _packSizeSamp = packSize;
-
- // Store the expected packet size in bytes, used to validate the recieved packet
- // If variable rate codec (extraByte == -1), set to -1 (65535)
- if (extraByte != -1)
- {
- // Add 0.875 to always round up to a whole byte
- _packSizeBytes = (WebRtc_UWord16)((float)(packSize*rate)/(float)(samplingFreqHz*8)+0.875)+extraByte;
- }
- else
- {
- // Packets will have a variable size
- _packSizeBytes = -1;
- }
-
- // Set pointer to the ACM where to register the codec
- AudioCodingModule* myACM;
- switch(side)
- {
- case 'A':
- {
- myACM = _acmA;
- break;
- }
- case 'B':
- {
- myACM = _acmB;
- break;
- }
- default:
- return -1;
- }
-
- if(myACM == NULL)
- {
- assert(false);
- return -1;
- }
- CodecInst myCodecParam;
-
- // Get all codec paramters before registering
- CHECK_ERROR(AudioCodingModule::Codec(codecName, myCodecParam, samplingFreqHz));
- myCodecParam.rate = rate;
- myCodecParam.pacsize = packSize;
- CHECK_ERROR(myACM->RegisterSendCodec(myCodecParam));
-
- // initialization was succesful
- return 0;
-}
-
-void TestAllCodecs::Run(TestPack* channel)
-{
- AudioFrame audioFrame;
-
- WebRtc_Word32 outFreqHzB = _outFileB.SamplingFrequency();
- WebRtc_UWord16 recSize;
- WebRtc_UWord32 timeStampDiff;
- channel->ResetPayloadSize();
- int errorCount = 0;
-
- // Only run 1 second for each test case
- while((_counter<1000)&& (!_inFileA.EndOfFile()))
- {
- // Add 10 msec to ACM
- _inFileA.Read10MsData(audioFrame);
- CHECK_ERROR(_acmA->Add10MsData(audioFrame));
-
- // Run sender side of ACM
- CHECK_ERROR(_acmA->Process());
-
- // Verify that the received packet size matches the settings
- recSize = channel->GetPayloadSize();
- if (recSize) {
- if ((recSize != _packSizeBytes) && (_packSizeBytes < 65535)) {
- errorCount++;
- }
-
- // Verify that the timestamp is updated with expected length
- timeStampDiff = channel->GetTimeStampDiff();
- if ((_counter > 10) && (timeStampDiff != _packSizeSamp))
- errorCount++;
- }
-
-
- // Run received side of ACM
- CHECK_ERROR(_acmB->PlayoutData10Ms(outFreqHzB, audioFrame));
-
- // Write output speech to file
- _outFileB.Write10MsData(audioFrame._payloadData, audioFrame._payloadDataLengthInSamples);
- }
-
- if (errorCount)
- {
- printf(" - test FAILED\n");
- }
- else if(_testMode != 0)
- {
- printf(" - test PASSED\n");
- }
-
- // Reset _counter
- if (_counter == 1000) {
- _counter = 0;
- }
- if (_inFileA.EndOfFile()) {
- _inFileA.Rewind();
- }
-}
-
-void TestAllCodecs::OpenOutFile(WebRtc_Word16 testNumber)
-{
- char fileName[500];
- sprintf(fileName, "%s/testallcodecs_out_%02d.pcm",
- webrtc::test::OutputPath().c_str(), testNumber);
- _outFileB.Open(fileName, 32000, "wb");
-}
-
-void TestAllCodecs::DisplaySendReceiveCodec()
-{
- CodecInst myCodecParam;
- _acmA->SendCodec(myCodecParam);
- printf("%s -> ", myCodecParam.plname);
- _acmB->ReceiveCodec(myCodecParam);
- printf("%s\n", myCodecParam.plname);
-}
-
-} // namespace webrtc
-
diff --git a/trunk/src/modules/audio_coding/main/test/TestAllCodecs.h b/trunk/src/modules/audio_coding/main/test/TestAllCodecs.h
deleted file mode 100644
index e0d621f..0000000
--- a/trunk/src/modules/audio_coding/main/test/TestAllCodecs.h
+++ /dev/null
@@ -1,96 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef TEST_ALL_CODECS_H
-#define TEST_ALL_CODECS_H
-
-#include "ACMTest.h"
-#include "Channel.h"
-#include "PCMFile.h"
-
-namespace webrtc {
-
-class TestPack : public AudioPacketizationCallback
-{
-public:
- TestPack();
- ~TestPack();
-
- void RegisterReceiverACM(AudioCodingModule* acm);
-
- virtual WebRtc_Word32 SendData(const FrameType frameType,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const RTPFragmentationHeader* fragmentation);
-
- WebRtc_UWord16 GetPayloadSize();
- WebRtc_UWord32 GetTimeStampDiff();
- void ResetPayloadSize();
-
-private:
- AudioCodingModule* _receiverACM;
- WebRtc_Word16 _seqNo;
- WebRtc_UWord8 _payloadData[60 * 32 * 2 * 2];
- WebRtc_UWord32 _timeStampDiff;
- WebRtc_UWord32 _lastInTimestamp;
- WebRtc_UWord64 _totalBytes;
- WebRtc_UWord16 _payloadSize;
-};
-
-class TestAllCodecs : public ACMTest
-{
-public:
- TestAllCodecs(int testMode);
- ~TestAllCodecs();
-
- void Perform();
-private:
- // The default value of '-1' indicates that the registration is based only on codec name
- // and a sampling frequncy matching is not required. This is useful for codecs which support
- // several sampling frequency.
- WebRtc_Word16 RegisterSendCodec(char side,
- char* codecName,
- WebRtc_Word32 sampFreqHz,
- int rate,
- int packSize,
- int extraByte);
-
- void Run(TestPack* channel);
- void OpenOutFile(WebRtc_Word16 testNumber);
- void DisplaySendReceiveCodec();
-
- WebRtc_Word32 SendData(
- const FrameType frameType,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const RTPFragmentationHeader* fragmentation);
-
- int _testMode;
-
- AudioCodingModule* _acmA;
- AudioCodingModule* _acmB;
-
- TestPack* _channelA2B;
-
- PCMFile _inFileA;
- PCMFile _outFileB;
- WebRtc_Word16 _testCntr;
- WebRtc_UWord16 _packSizeSamp;
- WebRtc_UWord16 _packSizeBytes;
- int _counter;
-};
-
-#endif // TEST_ALL_CODECS_H
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/TestFEC.cc b/trunk/src/modules/audio_coding/main/test/TestFEC.cc
deleted file mode 100644
index 81ca91a..0000000
--- a/trunk/src/modules/audio_coding/main/test/TestFEC.cc
+++ /dev/null
@@ -1,627 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "TestFEC.h"
-
-#include <cassert>
-#include <iostream>
-
-#include "audio_coding_module_typedefs.h"
-#include "common_types.h"
-#include "engine_configurations.h"
-#include "trace.h"
-#include "testsupport/fileutils.h"
-#include "utility.h"
-
-namespace webrtc {
-
-TestFEC::TestFEC(int testMode):
-_acmA(NULL),
-_acmB(NULL),
-_channelA2B(NULL),
-_testCntr(0)
-{
- _testMode = testMode;
-}
-
-TestFEC::~TestFEC()
-{
- if(_acmA != NULL)
- {
- AudioCodingModule::Destroy(_acmA);
- _acmA = NULL;
- }
- if(_acmB != NULL)
- {
- AudioCodingModule::Destroy(_acmB);
- _acmB = NULL;
- }
- if(_channelA2B != NULL)
- {
- delete _channelA2B;
- _channelA2B = NULL;
- }
-}
-
-void TestFEC::Perform()
-{
-
- if(_testMode == 0)
- {
- printf("Running FEC Test");
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
- "---------- TestFEC ----------");
- }
- char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
- _inFileA.Open(fileName, 32000, "rb");
-
-
- bool fecEnabled;
-
- _acmA = AudioCodingModule::Create(0);
- _acmB = AudioCodingModule::Create(1);
-
- _acmA->InitializeReceiver();
- _acmB->InitializeReceiver();
-
- WebRtc_UWord8 numEncoders = _acmA->NumberOfCodecs();
- CodecInst myCodecParam;
- if(_testMode != 0)
- {
- printf("Registering codecs at receiver... \n");
- }
- for(WebRtc_UWord8 n = 0; n < numEncoders; n++)
- {
- _acmB->Codec(n, myCodecParam);
- if(_testMode != 0)
- {
- printf("%s\n", myCodecParam.plname);
- }
- _acmB->RegisterReceiveCodec(myCodecParam);
- }
-
- // Create and connect the channel
- _channelA2B = new Channel;
- _acmA->RegisterTransportCallback(_channelA2B);
- _channelA2B->RegisterReceiverACM(_acmB);
-
- if(_testMode != 0)
- {
- printf("=======================================================================\n");
- printf("%d ",_testCntr++);
- }
- else
- {
- printf(".");
- }
-#ifndef WEBRTC_CODEC_G722
- printf("G722 needs to be activated to run this test\n");
- exit(-1);
-#endif
- char nameG722[] = "G722";
- RegisterSendCodec('A', nameG722, 16000);
- char nameCN[] = "CN";
- RegisterSendCodec('A', nameCN, 16000);
- char nameRED[] = "RED";
- RegisterSendCodec('A', nameRED);
- OpenOutFile(_testCntr);
- SetVAD(true, true, VADAggr);
- _acmA->SetFECStatus(false);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- Run();
- _outFileB.Close();
-
- if(_testMode != 0)
- {
- printf("=======================================================================\n");
- printf("%d ",_testCntr++);
- }
- else
- {
- printf(".");
- }
- _acmA->SetFECStatus(true);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- OpenOutFile(_testCntr);
- Run();
- _outFileB.Close();
-
-
-
- if(_testMode != 0)
- {
- printf("=======================================================================\n");
- printf("%d ",_testCntr++);
- }
- else
- {
- printf(".");
- }
- char nameISAC[] = "iSAC";
- RegisterSendCodec('A',nameISAC, 16000);
- OpenOutFile(_testCntr);
- SetVAD(true, true, VADVeryAggr);
- _acmA->SetFECStatus(false);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- Run();
- _outFileB.Close();
-
-
-
- if(_testMode != 0)
- {
- printf("=======================================================================\n");
- printf("%d ",_testCntr++);
- }
- else
- {
- printf(".");
- }
- _acmA->SetFECStatus(true);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- OpenOutFile(_testCntr);
- Run();
- _outFileB.Close();
-
-
-
- if(_testMode != 0)
- {
- printf("=======================================================================\n");
- printf("%d ",_testCntr++);
- }
- else
- {
- printf(".");
- }
-
- RegisterSendCodec('A', nameISAC, 32000);
- OpenOutFile(_testCntr);
- SetVAD(true, true, VADVeryAggr);
- _acmA->SetFECStatus(false);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- Run();
- _outFileB.Close();
-
-
-
- if(_testMode != 0)
- {
- printf("=======================================================================\n");
- printf("%d ",_testCntr++);
- }
- else
- {
- printf(".");
- }
- _acmA->SetFECStatus(true);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- OpenOutFile(_testCntr);
- Run();
- _outFileB.Close();
-
-
-
-
- if(_testMode != 0)
- {
- printf("=======================================================================\n");
- printf("%d ",_testCntr++);
- }
- else
- {
- printf(".");
- }
-
- RegisterSendCodec('A', nameISAC, 32000);
- OpenOutFile(_testCntr);
- SetVAD(false, false, VADNormal);
- _acmA->SetFECStatus(true);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- Run();
-
-
- RegisterSendCodec('A', nameISAC, 16000);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- Run();
-
- RegisterSendCodec('A', nameISAC, 32000);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- Run();
-
- RegisterSendCodec('A', nameISAC, 16000);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- Run();
- _outFileB.Close();
-
-
-
-
-
-
-
- _channelA2B->SetFECTestWithPacketLoss(true);
-
-
-
- if(_testMode != 0)
- {
- printf("=======================================================================\n");
- printf("%d ",_testCntr++);
- }
- else
- {
- printf(".");
- }
-
- RegisterSendCodec('A',nameG722);
- RegisterSendCodec('A', nameCN, 16000);
- OpenOutFile(_testCntr);
- SetVAD(true, true, VADAggr);
- _acmA->SetFECStatus(false);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- Run();
- _outFileB.Close();
-
-
-
- if(_testMode != 0)
- {
- printf("=======================================================================\n");
- printf("%d ",_testCntr++);
- }
- else
- {
- printf(".");
- }
- _acmA->SetFECStatus(true);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- OpenOutFile(_testCntr);
- Run();
- _outFileB.Close();
-
-
-
- if(_testMode != 0)
- {
- printf("=======================================================================\n");
- printf("%d ",_testCntr++);
- }
- else
- {
- printf(".");
- }
- RegisterSendCodec('A', nameISAC, 16000);
- OpenOutFile(_testCntr);
- SetVAD(true, true, VADVeryAggr);
- _acmA->SetFECStatus(false);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- Run();
- _outFileB.Close();
-
-
-
- if(_testMode != 0)
- {
- printf("=======================================================================\n");
- printf("%d ",_testCntr++);
- }
- else
- {
- printf(".");
- }
- _acmA->SetFECStatus(true);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- OpenOutFile(_testCntr);
- Run();
- _outFileB.Close();
-
-
-
-
- if(_testMode != 0)
- {
- printf("=======================================================================\n");
- printf("%d ",_testCntr++);
- }
- else
- {
- printf(".");
- }
- RegisterSendCodec('A', nameISAC, 32000);
- OpenOutFile(_testCntr);
- SetVAD(true, true, VADVeryAggr);
- _acmA->SetFECStatus(false);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- Run();
- _outFileB.Close();
-
-
-
- if(_testMode != 0)
- {
- printf("=======================================================================\n");
- printf("%d ",_testCntr++);
- }
- else
- {
- printf(".");
- }
- _acmA->SetFECStatus(true);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- OpenOutFile(_testCntr);
- Run();
- _outFileB.Close();
-
-
-
-
-
-
- if(_testMode != 0)
- {
- printf("=======================================================================\n");
- printf("%d ",_testCntr++);
- }
- else
- {
- printf(".");
- }
- RegisterSendCodec('A', nameISAC, 32000);
- OpenOutFile(_testCntr);
- SetVAD(false, false, VADNormal);
- _acmA->SetFECStatus(true);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- Run();
-
-
- RegisterSendCodec('A', nameISAC, 16000);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- Run();
-
- RegisterSendCodec('A', nameISAC, 32000);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- Run();
-
- RegisterSendCodec('A', nameISAC, 16000);
- fecEnabled = _acmA->FECStatus();
- if(_testMode != 0)
- {
- printf("FEC currently %s\n",(fecEnabled?"ON":"OFF"));
- DisplaySendReceiveCodec();
- }
- Run();
- _outFileB.Close();
-
-
-
- if(_testMode == 0)
- {
- printf("Done!\n");
- }
-}
-
-WebRtc_Word32 TestFEC::SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode)
-{
- if(_testMode != 0)
- {
- printf("DTX %s; VAD %s; VAD-Mode %d\n",
- enableDTX? "ON":"OFF",
- enableVAD? "ON":"OFF",
- (WebRtc_Word16)vadMode);
- }
- return _acmA->SetVAD(enableDTX, enableVAD, vadMode);
-}
-
-WebRtc_Word16 TestFEC::RegisterSendCodec(char side, char* codecName, WebRtc_Word32 samplingFreqHz)
-{
- if(_testMode != 0)
- {
- if(samplingFreqHz > 0)
- {
- printf("Registering %s-%d for side %c\n", codecName, samplingFreqHz, side);
- }
- else
- {
- printf("Registering %s for side %c\n", codecName, side);
- }
- }
- std::cout << std::flush;
- AudioCodingModule* myACM;
- switch(side)
- {
- case 'A':
- {
- myACM = _acmA;
- break;
- }
- case 'B':
- {
- myACM = _acmB;
- break;
- }
- default:
- return -1;
- }
-
- if(myACM == NULL)
- {
- assert(false);
- return -1;
- }
- CodecInst myCodecParam;
-
- CHECK_ERROR(AudioCodingModule::Codec(codecName, myCodecParam, samplingFreqHz));
-
- CHECK_ERROR(myACM->RegisterSendCodec(myCodecParam));
-
- // initialization was succesful
- return 0;
-}
-
-void TestFEC::Run()
-{
- AudioFrame audioFrame;
-
- WebRtc_UWord16 msecPassed = 0;
- WebRtc_UWord32 secPassed = 0;
- WebRtc_Word32 outFreqHzB = _outFileB.SamplingFrequency();
-
- while(!_inFileA.EndOfFile())
- {
- _inFileA.Read10MsData(audioFrame);
- CHECK_ERROR(_acmA->Add10MsData(audioFrame));
- CHECK_ERROR(_acmA->Process());
- CHECK_ERROR(_acmB->PlayoutData10Ms(outFreqHzB, audioFrame));
- _outFileB.Write10MsData(audioFrame._payloadData, audioFrame._payloadDataLengthInSamples);
- msecPassed += 10;
- if(msecPassed >= 1000)
- {
- msecPassed = 0;
- secPassed++;
- }
- if(((secPassed%5) == 4) && (msecPassed == 0) && (_testCntr > 14))
- {
- printf("%3u:%3u ", secPassed, msecPassed);
- _acmA->SetFECStatus(false);
- printf("FEC currently %s\n",(_acmA->FECStatus()?"ON":"OFF"));
- }
- if(((secPassed%5) == 4) && (msecPassed >= 990) && (_testCntr > 14))
- {
- printf("%3u:%3u ", secPassed, msecPassed);
- _acmA->SetFECStatus(true);
- printf("FEC currently %s\n",(_acmA->FECStatus()?"ON":"OFF"));
- }
- }
- _inFileA.Rewind();
-}
-
-void TestFEC::OpenOutFile(WebRtc_Word16 testNumber)
-{
- char fileName[500];
-
- if(_testMode == 0)
- {
- sprintf(fileName, "%s/TestFEC_autoFile_%02d.pcm",
- webrtc::test::OutputPath().c_str(), testNumber);
- }
- else
- {
- sprintf(fileName, "%s/TestFEC_outFile_%02d.pcm",
- webrtc::test::OutputPath().c_str(), testNumber);
- }
- _outFileB.Open(fileName, 32000, "wb");
-}
-
-void TestFEC::DisplaySendReceiveCodec()
-{
- CodecInst myCodecParam;
- _acmA->SendCodec(myCodecParam);
- printf("%s -> ", myCodecParam.plname);
- _acmB->ReceiveCodec(myCodecParam);
- printf("%s\n", myCodecParam.plname);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/TestFEC.h b/trunk/src/modules/audio_coding/main/test/TestFEC.h
deleted file mode 100644
index 00e951f..0000000
--- a/trunk/src/modules/audio_coding/main/test/TestFEC.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef TEST_FEC_H
-#define TEST_FEC_H
-
-#include "ACMTest.h"
-#include "Channel.h"
-#include "PCMFile.h"
-
-namespace webrtc {
-
-class TestFEC : public ACMTest
-{
-public:
- TestFEC(int testMode);
- ~TestFEC();
-
- void Perform();
-private:
- // The default value of '-1' indicates that the registration is based only on codec name
- // and a sampling frequncy matching is not required. This is useful for codecs which support
- // several sampling frequency.
- WebRtc_Word16 RegisterSendCodec(char side, char* codecName, WebRtc_Word32 sampFreqHz = -1);
- void Run();
- void OpenOutFile(WebRtc_Word16 testNumber);
- void DisplaySendReceiveCodec();
- WebRtc_Word32 SetVAD(bool enableDTX, bool enableVAD, ACMVADMode vadMode);
- AudioCodingModule* _acmA;
- AudioCodingModule* _acmB;
-
- Channel* _channelA2B;
-
- PCMFile _inFileA;
- PCMFile _outFileB;
- WebRtc_Word16 _testCntr;
- int _testMode;
-};
-
-} // namespace webrtc
-
-#endif
diff --git a/trunk/src/modules/audio_coding/main/test/TestStereo.cc b/trunk/src/modules/audio_coding/main/test/TestStereo.cc
deleted file mode 100644
index 2105d36..0000000
--- a/trunk/src/modules/audio_coding/main/test/TestStereo.cc
+++ /dev/null
@@ -1,804 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "TestStereo.h"
-
-#include <cassert>
-#include <iostream>
-
-#include "audio_coding_module_typedefs.h"
-#include "common_types.h"
-#include "engine_configurations.h"
-#include "testsupport/fileutils.h"
-#include "trace.h"
-#include "utility.h"
-
-namespace webrtc {
-
-// Class for simulating packet handling
-TestPackStereo::TestPackStereo():
-_receiverACM(NULL),
-_seqNo(0),
-_timeStampDiff(0),
-_lastInTimestamp(0),
-_totalBytes(0),
-_payloadSize(0),
-_noChannels(1),
-_codecType(0)
-{
-}
-TestPackStereo::~TestPackStereo()
-{
-}
-
-void
-TestPackStereo::RegisterReceiverACM(AudioCodingModule* acm)
-{
- _receiverACM = acm;
- return;
-}
-
-
-WebRtc_Word32
-TestPackStereo::SendData(
- const FrameType frameType,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const RTPFragmentationHeader* fragmentation)
-{
- WebRtcRTPHeader rtpInfo;
- WebRtc_Word32 status;
- WebRtc_UWord16 payloadDataSize = payloadSize;
- WebRtc_UWord8 payloadDataMaster[60 * 32 * 2 * 2];
- WebRtc_UWord8 payloadDataSlave[60 * 32 * 2 * 2];
-
- rtpInfo.header.markerBit = false;
- rtpInfo.header.ssrc = 0;
- rtpInfo.header.sequenceNumber = _seqNo++;
- rtpInfo.header.payloadType = payloadType;
- rtpInfo.header.timestamp = timeStamp;
- if(frameType == kFrameEmpty)
- {
- // Skip this frame
- return 0;
- }
- if(frameType != kAudioFrameCN)
- {
- rtpInfo.type.Audio.isCNG = false;
-
- // For stereo we need to call ACM with two incoming packets, one for each channel.
- // Different packet-splitting depending on codec.
- if (_codecType == 0) {
- // one byte per sample
- for (int i=0, j=0; i<payloadDataSize; i+=2, j++)
- {
- payloadDataMaster[j] = payloadData[i];
- payloadDataSlave[j] = payloadData[i+1];
- }
- } else if (_codecType == 1) {
- // two bytes per sample
- for (int i=0, j=0; i<payloadDataSize; i+=4, j+=2)
- {
- payloadDataMaster[j] = payloadData[i];
- payloadDataMaster[j+1] = payloadData[i+1];
- payloadDataSlave[j] = payloadData[i+2];
- payloadDataSlave[j+1] = payloadData[i+3];
- }
- } else if (_codecType == 2) {
- // frameBased
- memcpy(payloadDataMaster, &payloadData[0], payloadDataSize/2);
- memcpy(payloadDataSlave, &payloadData[payloadDataSize/2], payloadDataSize/2);
- } else if (_codecType == 3) {
- // four bits per sample
- for (int i=0, j=0; i<payloadDataSize; i+=2, j++)
- {
- payloadDataMaster[j] = (payloadData[i] & 0xF0) + (payloadData[i+1] >> 4);
- payloadDataSlave[j] = ((payloadData[i] & 0x0F) << 4) + (payloadData[i+1] & 0x0F);
- }
- } else if (_codecType == 4) {
- // True stereo, call both master and slave with whole stream.
- memcpy(payloadDataMaster, payloadData, payloadSize);
- memcpy(payloadDataSlave, payloadData, payloadSize);
- payloadDataSize = payloadSize*2;
- }
- }
- else
- {
- // If CNG packet, send the same packet to both master and slave.
- rtpInfo.type.Audio.isCNG = true;
- memcpy(payloadDataMaster, payloadData, payloadSize);
- memcpy(payloadDataSlave, payloadData, payloadSize);
- payloadDataSize = payloadSize*2;
- }
-
- if (_codecType != 5) {
- // Call ACM with two packets, one for each channel
- rtpInfo.type.Audio.channel = 1;
- status = _receiverACM->IncomingPacket((WebRtc_Word8*)payloadDataMaster, payloadDataSize/2, rtpInfo);
- rtpInfo.type.Audio.channel = 2;
- status = _receiverACM->IncomingPacket((WebRtc_Word8*)payloadDataSlave, payloadDataSize/2, rtpInfo);
- } else {
- // Mono case, call ACM with one packet.
- rtpInfo.type.Audio.channel = 1;
- status = _receiverACM->IncomingPacket((WebRtc_Word8*)payloadData, payloadDataSize, rtpInfo);
- }
-
- if (frameType != kAudioFrameCN) {
- _payloadSize = payloadDataSize;
- } else {
- _payloadSize = -1;
- }
- _timeStampDiff = timeStamp - _lastInTimestamp;
- _lastInTimestamp = timeStamp;
- _totalBytes += payloadDataSize;
- return status;
-}
-
-WebRtc_UWord16
-TestPackStereo::GetPayloadSize()
-{
- return _payloadSize;
-}
-
-
-WebRtc_UWord32
-TestPackStereo::GetTimeStampDiff()
-{
- return _timeStampDiff;
-}
-
-void
-TestPackStereo::ResetPayloadSize()
-{
- _payloadSize = 0;
-}
-
-void
-TestPackStereo::SetCodecType(int codecType)
-{
- _codecType = codecType;
-}
-
-TestStereo::TestStereo(int testMode):
-_acmA(NULL),
-_acmB(NULL),
-_channelA2B(NULL),
-_testCntr(0),
-_packSizeSamp(0),
-_packSizeBytes(0),
-_counter(0)
-{
- // testMode = 0 for silent test (auto test)
- _testMode = testMode;
-}
-
-TestStereo::~TestStereo()
-{
- if(_acmA != NULL)
- {
- AudioCodingModule::Destroy(_acmA);
- _acmA = NULL;
- }
- if(_acmB != NULL)
- {
- AudioCodingModule::Destroy(_acmB);
- _acmB = NULL;
- }
- if(_channelA2B != NULL)
- {
- delete _channelA2B;
- _channelA2B = NULL;
- }
-}
-
-void TestStereo::Perform()
-{
- char file_name_stereo[500];
- char file_name_mono[500];
- WebRtc_UWord16 frequencyHz;
- int audio_channels;
- int codec_channels;
-
- if(_testMode == 0)
- {
- printf("Running Stereo Test");
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
- "---------- TestStereo ----------");
- }
-
- strcpy(file_name_stereo, "./test/data/audio_coding/teststereo32kHz.pcm");
- strcpy(file_name_mono, "./test/data/audio_coding/testfile32kHz.pcm");
- frequencyHz = 32000;
-
- _in_file_stereo.Open(file_name_stereo, frequencyHz, "rb");
- _in_file_stereo.ReadStereo(true);
- _in_file_mono.Open(file_name_mono, frequencyHz, "rb");
- _in_file_mono.ReadStereo(false);
-
- _acmA = AudioCodingModule::Create(0);
- _acmB = AudioCodingModule::Create(1);
-
- _acmA->InitializeReceiver();
- _acmB->InitializeReceiver();
-
- WebRtc_UWord8 numEncoders = _acmA->NumberOfCodecs();
- CodecInst myCodecParam;
-
- // Register receiving codecs as stereo.
- for(WebRtc_UWord8 n = 0; n < numEncoders; n++)
- {
- _acmB->Codec(n, myCodecParam);
- if(!strcmp(myCodecParam.plname, "L16") ||
- !strcmp(myCodecParam.plname, "PCMA")||
- !strcmp(myCodecParam.plname, "PCMU")||
- !strcmp(myCodecParam.plname, "G722")||
- !strcmp(myCodecParam.plname, "CELT"))
- {
- myCodecParam.channels=2;
- _acmB->RegisterReceiveCodec(myCodecParam);
- }
- }
-
- // Create and connect the channel.
- _channelA2B = new TestPackStereo;
- _acmA->RegisterTransportCallback(_channelA2B);
- _channelA2B->RegisterReceiverACM(_acmB);
-
- //
- // Test Stereo-To-Stereo for all codecs.
- //
- audio_channels = 2;
- codec_channels = 2;
-
- // All codecs are tested for all allowed sampling frequencies, rates and packet sizes
-#ifdef WEBRTC_CODEC_G722
- if(_testMode != 0) {
- printf("=======================================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Stereo-to-stereo\n");
- } else {
- printf(".");
- }
- _channelA2B->SetCodecType(3);
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecG722[] = "G722";
- RegisterSendCodec('A', codecG722, 16000, 64000, 160, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecG722, 16000, 64000, 320, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecG722, 16000, 64000, 480, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecG722, 16000, 64000, 640, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecG722, 16000, 64000, 800, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecG722, 16000, 64000, 960, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _acmA->SetVAD(true, true, VADNormal);
- RegisterSendCodec('A', codecG722, 16000, 64000, 320, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _acmA->SetVAD(false, false, VADNormal);
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_PCM16
- if(_testMode != 0) {
- printf("=======================================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Stereo-to-stereo\n");
- } else {
- printf(".");
- }
- _channelA2B->SetCodecType(1);
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecL16[] = "L16";
- RegisterSendCodec('A', codecL16, 8000, 128000, 80, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecL16, 8000, 128000, 160, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecL16, 8000, 128000, 240, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecL16, 8000, 128000, 320, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _acmA->SetVAD(true, true, VADNormal);
- RegisterSendCodec('A', codecL16, 8000, 128000, 80, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _acmA->SetVAD(false, false, VADNormal);
- _outFileB.Close();
-
- if(_testMode != 0) {
- printf("=======================================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Stereo-to-stereo\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecL16, 16000, 256000, 160, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecL16, 16000, 256000, 320, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecL16, 16000, 256000, 480, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecL16, 16000, 256000, 640, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _acmA->SetVAD(true, true, VADNormal);
- RegisterSendCodec('A', codecL16, 16000, 256000, 160, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _acmA->SetVAD(false, false, VADNormal);
- _outFileB.Close();
-
- if(_testMode != 0) {
- printf("=======================================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Stereo-to-stereo\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecL16, 32000, 512000, 320, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecL16, 32000, 512000, 640, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _acmA->SetVAD(true, true, VADNormal);
- RegisterSendCodec('A', codecL16, 32000, 512000, 320, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _acmA->SetVAD(false, false, VADNormal);
- _outFileB.Close();
-#endif
-#define PCMA_AND_PCMU
-#ifdef PCMA_AND_PCMU
- if(_testMode != 0) {
- printf("=======================================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Stereo-to-stereo\n");
- } else {
- printf(".");
- }
- _channelA2B->SetCodecType(0);
- audio_channels = 2;
- codec_channels = 2;
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecPCMA[] = "PCMA";
- RegisterSendCodec('A', codecPCMA, 8000, 64000, 80, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecPCMA, 8000, 64000, 160, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecPCMA, 8000, 64000, 240, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecPCMA, 8000, 64000, 320, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecPCMA, 8000, 64000, 400, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecPCMA, 8000, 64000, 480, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _acmA->SetVAD(true, true, VADNormal);
- RegisterSendCodec('A', codecPCMA, 8000, 64000, 80, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _acmA->SetVAD(false, false, VADNormal);
- _outFileB.Close();
- if(_testMode != 0) {
- printf("=======================================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Stereo-to-stereo\n");
- } else {
- printf(".");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecPCMU[] = "PCMU";
- RegisterSendCodec('A', codecPCMU, 8000, 64000, 80, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecPCMU, 8000, 64000, 160, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecPCMU, 8000, 64000, 240, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecPCMU, 8000, 64000, 320, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecPCMU, 8000, 64000, 400, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecPCMU, 8000, 64000, 480, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _acmA->SetVAD(true, true, VADNormal);
- RegisterSendCodec('A', codecPCMU, 8000, 64000, 80, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _acmA->SetVAD(false, false, VADNormal);
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_CELT
- if(_testMode != 0) {
- printf("=======================================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Stereo-to-stereo\n");
- } else {
- printf(".");
- }
- _channelA2B->SetCodecType(4);
- audio_channels = 2;
- codec_channels = 2;
- _testCntr++;
- OpenOutFile(_testCntr);
- char codecCELT[] = "CELT";
- RegisterSendCodec('A', codecCELT, 32000, 48000, 320, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecCELT, 32000, 64000, 320, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecCELT, 32000, 128000, 320, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _acmA->SetVAD(true, true, VADNormal);
- RegisterSendCodec('A', codecCELT, 32000, 48000, 320, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _acmA->SetVAD(false, false, VADNormal);
- _outFileB.Close();
-#endif
- //
- // Test Mono-To-Stereo for all codecs.
- //
- audio_channels = 1;
- codec_channels = 2;
-
-#ifdef WEBRTC_CODEC_G722
- if(_testMode != 0) {
- printf("===============================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Mono-to-stereo\n");
- }
- _testCntr++;
- _channelA2B->SetCodecType(3);
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecG722, 16000, 64000, 160, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_PCM16
- if(_testMode != 0) {
- printf("===============================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Mono-to-stereo\n");
- }
- _testCntr++;
- _channelA2B->SetCodecType(1);
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecL16, 8000, 128000, 80, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _outFileB.Close();
- if(_testMode != 0) {
- printf("===============================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Mono-to-stereo\n");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecL16, 16000, 256000, 160, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _outFileB.Close();
- if(_testMode != 0) {
- printf("===============================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Mono-to-stereo\n");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecL16, 32000, 512000, 320, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _outFileB.Close();
-#endif
-#ifdef PCMA_AND_PCMU
- if(_testMode != 0) {
- printf("===============================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Mono-to-stereo\n");
- }
- _testCntr++;
- _channelA2B->SetCodecType(0);
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecPCMU, 8000, 64000, 80, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecPCMA, 8000, 64000, 80, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_CELT
- if(_testMode != 0) {
- printf("===============================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Mono-to-stereo\n");
- }
- _testCntr++;
- _channelA2B->SetCodecType(4);
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecCELT, 32000, 64000, 320, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _outFileB.Close();
-#endif
-
- //
- // Test Stereo-To-Mono for all codecs.
- //
- audio_channels = 2;
- codec_channels = 1;
- _channelA2B->SetCodecType(5);
-
- // Register receivers as mono.
- for(WebRtc_UWord8 n = 0; n < numEncoders; n++) {
- _acmB->Codec(n, myCodecParam);
- if(!strcmp(myCodecParam.plname, "L16") ||
- !strcmp(myCodecParam.plname, "PCMA")||
- !strcmp(myCodecParam.plname, "PCMU")||
- !strcmp(myCodecParam.plname, "G722")||
- !strcmp(myCodecParam.plname, "CELT")) {
- myCodecParam.channels = 1;
- _acmB->RegisterReceiveCodec(myCodecParam);
- }
- }
-#ifdef WEBRTC_CODEC_G722
- // Run stereo audio and mono codec.
- if(_testMode != 0) {
- printf("===============================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Stereo-to-mono\n");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecG722, 16000, 64000, 160, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_PCM16
- if(_testMode != 0) {
- printf("===============================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Stereo-to-mono\n");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecL16, 8000, 128000, 80, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _outFileB.Close();
- if(_testMode != 0) {
- printf("===============================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Stereo-to-mono\n");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecL16, 16000, 256000, 160, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _outFileB.Close();
- if(_testMode != 0) {
- printf("===============================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Stereo-to-mono\n");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecL16, 32000, 512000, 320, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _outFileB.Close();
-#endif
-#ifdef PCMA_AND_PCMU
- if(_testMode != 0) {
- printf("===============================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Stereo-to-mono\n");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecPCMU, 8000, 64000, 80, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- RegisterSendCodec('A', codecPCMA, 8000, 64000, 80, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_CELT
- if(_testMode != 0) {
- printf("===============================================================\n");
- printf("Test number: %d\n",_testCntr + 1);
- printf("Test type: Stereo-to-mono\n");
- }
- _testCntr++;
- OpenOutFile(_testCntr);
- RegisterSendCodec('A', codecCELT, 32000, 64000, 320, codec_channels);
- Run(_channelA2B, audio_channels, codec_channels);
- _outFileB.Close();
-#endif
-
- // Print out which codecs were tested, and which were not, in the run.
- if(_testMode != 0) {
- printf("\nThe following codecs was INCLUDED in the test:\n");
-#ifdef WEBRTC_CODEC_G722
- printf(" G.722\n");
-#endif
-#ifdef WEBRTC_CODEC_PCM16
- printf(" PCM16\n");
-#endif
- printf(" G.711\n");
-
- printf("\nTo complete the test, listen to the %d number of output files.\n", _testCntr);
- } else {
- printf("Done!\n");
- }
-}
-
-// Register Codec to use in the test
-//
-// Input: side - which ACM to use, 'A' or 'B'
-// codecName - name to use when register the codec
-// samplingFreqHz - sampling frequency in Herz
-// rate - bitrate in bytes
-// packSize - packet size in samples
-// extraByte - if extra bytes needed compared to the bitrate
-// used when registering, can be an internal header
-// set to -1 if the codec is a variable rate codec
-WebRtc_Word16 TestStereo::RegisterSendCodec(char side,
- char* codecName,
- WebRtc_Word32 samplingFreqHz,
- int rate,
- int packSize,
- int channels)
-{
- if(_testMode != 0) {
- // Print out codec and settings
- printf("Codec: %s Freq: %d Rate: %d PackSize: %d", codecName, samplingFreqHz, rate, packSize);
- }
-
- // Store packetsize in samples, used to validate the recieved packet
- _packSizeSamp = packSize;
-
- // Store the expected packet size in bytes, used to validate the recieved packet
- // Add 0.875 to always round up to a whole byte
- _packSizeBytes = (WebRtc_UWord16)((float)(packSize*rate)/(float)(samplingFreqHz*8)+0.875);
-
- // Set pointer to the ACM where to register the codec
- AudioCodingModule* myACM;
- switch(side)
- {
- case 'A':
- {
- myACM = _acmA;
- break;
- }
- case 'B':
- {
- myACM = _acmB;
- break;
- }
- default:
- return -1;
- }
-
- if(myACM == NULL)
- {
- assert(false);
- return -1;
- }
- CodecInst myCodecParam;
-
- // Get all codec parameters before registering
- CHECK_ERROR(AudioCodingModule::Codec(codecName, myCodecParam, samplingFreqHz));
- myCodecParam.rate = rate;
- myCodecParam.pacsize = packSize;
- // Start with register codec as mono, to test that changing to stereo works.
- myCodecParam.channels = 1;
- CHECK_ERROR(myACM->RegisterSendCodec(myCodecParam));
- // Register codec as stereo.
- if (channels == 2) {
- myCodecParam.channels = 2;
- CHECK_ERROR(myACM->RegisterSendCodec(myCodecParam));
- }
- // Initialization was successful.
- return 0;
-}
-
-void TestStereo::Run(TestPackStereo* channel, int in_channels, int out_channels)
-{
- AudioFrame audioFrame;
-
- WebRtc_Word32 outFreqHzB = _outFileB.SamplingFrequency();
- WebRtc_UWord16 recSize;
- WebRtc_UWord32 timeStampDiff;
- channel->ResetPayloadSize();
- int errorCount = 0;
-
- // Only run 1 second for each test case
- // TODO(tlegrand): either remove |_counter| or start using it as the comment
- // above says. Now |_counter| is always 0.
- while(_counter<1000)
- {
- // Add 10 msec to ACM
- if (in_channels == 1) {
- if (_in_file_mono.EndOfFile()) {
- break;
- }
- _in_file_mono.Read10MsData(audioFrame);
- } else {
- if (_in_file_stereo.EndOfFile()) {
- break;
- }
- _in_file_stereo.Read10MsData(audioFrame);
- }
- CHECK_ERROR(_acmA->Add10MsData(audioFrame));
-
- // Run sender side of ACM
- CHECK_ERROR(_acmA->Process());
-
- // Verify that the received packet size matches the settings
- recSize = channel->GetPayloadSize();
- if ((0 < recSize) & (recSize < 65535)) {
- if ((recSize != _packSizeBytes * out_channels) &&
- (_packSizeBytes < 65535)) {
- errorCount++;
- }
-
- // Verify that the timestamp is updated with expected length
- timeStampDiff = channel->GetTimeStampDiff();
- if ((_counter > 10) && (timeStampDiff != _packSizeSamp)) {
- errorCount++;
- }
- }
-
- // Run received side of ACM
- CHECK_ERROR(_acmB->PlayoutData10Ms(outFreqHzB, audioFrame));
-
- // Write output speech to file
- _outFileB.Write10MsData(
- audioFrame._payloadData,
- audioFrame._payloadDataLengthInSamples * audioFrame._audioChannel);
- }
-
- if (errorCount)
- {
- printf(" - test FAILED\n");
- }
- else if(_testMode != 0)
- {
- printf(" - test PASSED\n");
- }
-
- // Reset _counter
- if (_counter == 1000) {
- _counter = 0;
- }
- if (_in_file_mono.EndOfFile()) {
- _in_file_mono.Rewind();
- }
- if (_in_file_stereo.EndOfFile()) {
- _in_file_stereo.Rewind();
- }
-}
-
-void TestStereo::OpenOutFile(WebRtc_Word16 testNumber)
-{
- char fileName[500];
- sprintf(fileName, "%s/teststereo_out_%02d.pcm",
- webrtc::test::OutputPath().c_str(), testNumber);
- _outFileB.Open(fileName, 32000, "wb");
-}
-
-void TestStereo::DisplaySendReceiveCodec()
-{
- CodecInst myCodecParam;
- _acmA->SendCodec(myCodecParam);
- if(_testMode != 0) {
- printf("%s -> ", myCodecParam.plname);
- }
- _acmB->ReceiveCodec(myCodecParam);
- if(_testMode != 0) {
- printf("%s\n", myCodecParam.plname);
- }
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/TestStereo.h b/trunk/src/modules/audio_coding/main/test/TestStereo.h
deleted file mode 100644
index 94a0c56..0000000
--- a/trunk/src/modules/audio_coding/main/test/TestStereo.h
+++ /dev/null
@@ -1,103 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef TEST_STEREO_H
-#define TEST_STEREO_H
-
-#include "ACMTest.h"
-#include "Channel.h"
-#include "PCMFile.h"
-
-namespace webrtc {
-
-class TestPackStereo : public AudioPacketizationCallback
-{
-public:
- TestPackStereo();
- ~TestPackStereo();
-
- void RegisterReceiverACM(AudioCodingModule* acm);
-
- virtual WebRtc_Word32 SendData(const FrameType frameType,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const RTPFragmentationHeader* fragmentation);
-
- WebRtc_UWord16 GetPayloadSize();
- WebRtc_UWord32 GetTimeStampDiff();
- void ResetPayloadSize();
- void SetCodecType(int codecType);
-
-
-private:
- AudioCodingModule* _receiverACM;
- WebRtc_Word16 _seqNo;
- WebRtc_UWord8 _payloadData[60 * 32 * 2 * 2];
- WebRtc_UWord32 _timeStampDiff;
- WebRtc_UWord32 _lastInTimestamp;
- WebRtc_UWord64 _totalBytes;
- WebRtc_UWord16 _payloadSize;
- WebRtc_UWord16 _noChannels;
- int _codecType;
-};
-
-class TestStereo : public ACMTest
-{
-public:
- TestStereo(int testMode);
- ~TestStereo();
-
- void Perform();
-private:
- // The default value of '-1' indicates that the registration is based only on codec name
- // and a sampling frequncy matching is not required. This is useful for codecs which support
- // several sampling frequency.
- WebRtc_Word16 RegisterSendCodec(char side,
- char* codecName,
- WebRtc_Word32 sampFreqHz,
- int rate,
- int packSize,
- int channels);
-
- void Run(TestPackStereo* channel, int in_channels, int out_channels);
- void OpenOutFile(WebRtc_Word16 testNumber);
- void DisplaySendReceiveCodec();
-
- WebRtc_Word32 SendData(
- const FrameType frameType,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const RTPFragmentationHeader* fragmentation);
-
- int _testMode;
-
- AudioCodingModule* _acmA;
- AudioCodingModule* _acmB;
-
- TestPackStereo* _channelA2B;
-
- PCMFile _in_file_stereo;
- PCMFile _in_file_mono;
- PCMFile _outFileB;
- WebRtc_Word16 _testCntr;
- WebRtc_UWord16 _packSizeSamp;
- WebRtc_UWord16 _packSizeBytes;
- int _counter;
- int _codecType;
-};
-
-} // namespace webrtc
-
-#endif
-
diff --git a/trunk/src/modules/audio_coding/main/test/TestVADDTX.cc b/trunk/src/modules/audio_coding/main/test/TestVADDTX.cc
deleted file mode 100644
index d7329fa..0000000
--- a/trunk/src/modules/audio_coding/main/test/TestVADDTX.cc
+++ /dev/null
@@ -1,511 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "TestVADDTX.h"
-
-#include <iostream>
-
-#include "audio_coding_module_typedefs.h"
-#include "common_types.h"
-#include "engine_configurations.h"
-#include "testsupport/fileutils.h"
-#include "trace.h"
-#include "utility.h"
-
-namespace webrtc {
-
-TestVADDTX::TestVADDTX(int testMode):
-_acmA(NULL),
-_acmB(NULL),
-_channelA2B(NULL),
-_testResults(0)
-{
- //testMode == 1 for more extensive testing
- //testMode == 0 for quick test (autotest)
- _testMode = testMode;
-}
-
-TestVADDTX::~TestVADDTX()
-{
- if(_acmA != NULL)
- {
- AudioCodingModule::Destroy(_acmA);
- _acmA = NULL;
- }
- if(_acmB != NULL)
- {
- AudioCodingModule::Destroy(_acmB);
- _acmB = NULL;
- }
- if(_channelA2B != NULL)
- {
- delete _channelA2B;
- _channelA2B = NULL;
- }
-}
-
-void TestVADDTX::Perform()
-{
- if(_testMode == 0)
- {
- printf("Running VAD/DTX Test");
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1,
- "---------- TestVADDTX ----------");
- }
- char fileName[] = "./test/data/audio_coding/testfile32kHz.pcm";
- _inFileA.Open(fileName, 32000, "rb");
-
- _acmA = AudioCodingModule::Create(0);
- _acmB = AudioCodingModule::Create(1);
-
- _acmA->InitializeReceiver();
- _acmB->InitializeReceiver();
-
- WebRtc_UWord8 numEncoders = _acmA->NumberOfCodecs();
- CodecInst myCodecParam;
- if(_testMode != 0)
- {
- printf("Registering codecs at receiver... \n");
- }
- for(WebRtc_UWord8 n = 0; n < numEncoders; n++)
- {
- _acmB->Codec(n, myCodecParam);
- if(_testMode != 0)
- {
- printf("%s\n", myCodecParam.plname);
- }
- _acmB->RegisterReceiveCodec(myCodecParam);
- }
-
- // Create and connect the channel
- _channelA2B = new Channel;
- _acmA->RegisterTransportCallback(_channelA2B);
- _channelA2B->RegisterReceiverACM(_acmB);
-
- _acmA->RegisterVADCallback(&_monitor);
-
-
- WebRtc_Word16 testCntr = 1;
- WebRtc_Word16 testResults = 0;
-
-#ifdef WEBRTC_CODEC_ISAC
- // Open outputfile
- OpenOutFile(testCntr++);
-
- // Register iSAC WB as send codec
- char nameISAC[] = "ISAC";
- RegisterSendCodec('A', nameISAC, 16000);
-
- // Run the five test cased
- runTestCases();
-
- // Close file
- _outFileB.Close();
-
- // Open outputfile
- OpenOutFile(testCntr++);
-
- // Register iSAC SWB as send codec
- RegisterSendCodec('A', nameISAC, 32000);
-
- // Run the five test cased
- runTestCases();
-
- // Close file
- _outFileB.Close();
-#endif
-#ifdef WEBRTC_CODEC_ILBC
- // Open outputfile
- OpenOutFile(testCntr++);
-
- // Register iLBC as send codec
- char nameILBC[] = "ilbc";
- RegisterSendCodec('A', nameILBC);
-
- // Run the five test cased
- runTestCases();
-
- // Close file
- _outFileB.Close();
-
-#endif
- if(_testMode) {
- printf("Done!\n");
- }
-
- printf("VAD/DTX test completed with %d subtests failed\n", testResults);
- if (testResults > 0)
- {
- printf("Press return\n\n");
- getchar();
- }
-}
-
-void TestVADDTX::runTestCases()
-{
- if(_testMode != 0)
- {
- CodecInst myCodecParam;
- _acmA->SendCodec(myCodecParam);
- printf("%s\n", myCodecParam.plname);
- }
- else
- {
- printf(".");
- }
- // #1 DTX = OFF, VAD = ON, VADNormal
- if(_testMode != 0)
- printf("Test #1 ");
- SetVAD(false, true, VADNormal);
- Run();
- _testResults += VerifyTest();
-
- // #2 DTX = OFF, VAD = ON, VADAggr
- if(_testMode != 0)
- printf("Test #2 ");
- SetVAD(false, true, VADAggr);
- Run();
- _testResults += VerifyTest();
-
- // #3 DTX = ON, VAD = ON, VADLowBitrate
- if(_testMode != 0)
- printf("Test #3 ");
- SetVAD(true, true, VADLowBitrate);
- Run();
- _testResults += VerifyTest();
-
- // #4 DTX = ON, VAD = ON, VADVeryAggr
- if(_testMode != 0)
- printf("Test #4 ");
- SetVAD(true, true, VADVeryAggr);
- Run();
- _testResults += VerifyTest();
-
- // #5 DTX = ON, VAD = OFF, VADNormal
- if(_testMode != 0)
- printf("Test #5 ");
- SetVAD(true, false, VADNormal);
- Run();
- _testResults += VerifyTest();
-
-}
-void TestVADDTX::runTestInternalDTX()
-{
- // #6 DTX = ON, VAD = ON, VADNormal
- if(_testMode != 0)
- printf("Test #6 ");
-
- SetVAD(true, true, VADNormal);
- if(_acmA->ReplaceInternalDTXWithWebRtc(true) < 0) {
- printf("Was not able to replace DTX since CN was not registered\n");
- }
- Run();
- _testResults += VerifyTest();
-}
-
-void TestVADDTX::SetVAD(bool statusDTX, bool statusVAD, WebRtc_Word16 vadMode)
-{
- bool dtxEnabled, vadEnabled;
- ACMVADMode vadModeSet;
-
- if (_acmA->SetVAD(statusDTX, statusVAD, (ACMVADMode) vadMode) < 0) {
- assert(false);
- }
- if (_acmA->VAD(dtxEnabled, vadEnabled, vadModeSet) < 0) {
- assert(false);
- }
-
- if(_testMode != 0)
- {
- if(statusDTX != dtxEnabled)
- {
- printf("DTX: %s not the same as requested: %s\n",
- dtxEnabled? "ON":"OFF", dtxEnabled? "OFF":"ON");
- }
- if(((statusVAD == true) && (vadEnabled == false)) ||
- ((statusVAD == false) && (vadEnabled == false) &&
- (statusDTX == true)))
- {
- printf("VAD: %s not the same as requested: %s\n",
- vadEnabled? "ON":"OFF", vadEnabled? "OFF":"ON");
- }
- if(vadModeSet != vadMode)
- {
- printf("VAD mode: %d not the same as requested: %d\n",
- (WebRtc_Word16)vadModeSet, (WebRtc_Word16)vadMode);
- }
- }
-
- // Requested VAD/DTX settings
- _setStruct.statusDTX = statusDTX;
- _setStruct.statusVAD = statusVAD;
- _setStruct.vadMode = (ACMVADMode) vadMode;
-
- // VAD settings after setting VAD in ACM
- _getStruct.statusDTX = dtxEnabled;
- _getStruct.statusVAD = vadEnabled;
- _getStruct.vadMode = vadModeSet;
-
-}
-
-VADDTXstruct TestVADDTX::GetVAD()
-{
- VADDTXstruct retStruct;
- bool dtxEnabled, vadEnabled;
- ACMVADMode vadModeSet;
-
- if (_acmA->VAD(dtxEnabled, vadEnabled, vadModeSet) < 0) {
- assert(false);
- }
-
- retStruct.statusDTX = dtxEnabled;
- retStruct.statusVAD = vadEnabled;
- retStruct.vadMode = vadModeSet;
- return retStruct;
-}
-
-WebRtc_Word16 TestVADDTX::RegisterSendCodec(char side,
- char* codecName,
- WebRtc_Word32 samplingFreqHz,
- WebRtc_Word32 rateKbps)
-{
- if(_testMode != 0)
- {
- printf("Registering %s for side %c\n", codecName, side);
- }
- std::cout << std::flush;
- AudioCodingModule* myACM;
- switch(side)
- {
- case 'A':
- {
- myACM = _acmA;
- break;
- }
- case 'B':
- {
- myACM = _acmB;
- break;
- }
- default:
- return -1;
- }
-
- if(myACM == NULL)
- {
- return -1;
- }
-
- CodecInst myCodecParam;
- for(WebRtc_Word16 codecCntr = 0; codecCntr < myACM->NumberOfCodecs();
- codecCntr++)
- {
- CHECK_ERROR(myACM->Codec((WebRtc_UWord8)codecCntr, myCodecParam));
- if(!STR_CASE_CMP(myCodecParam.plname, codecName))
- {
- if((samplingFreqHz == -1) || (myCodecParam.plfreq == samplingFreqHz))
- {
- if((rateKbps == -1) || (myCodecParam.rate == rateKbps))
- {
- break;
- }
- }
- }
- }
-
- CHECK_ERROR(myACM->RegisterSendCodec(myCodecParam));
-
- // initialization was succesful
- return 0;
-}
-
-void TestVADDTX::Run()
-{
- AudioFrame audioFrame;
-
- WebRtc_UWord16 SamplesIn10MsecA = _inFileA.PayloadLength10Ms();
- WebRtc_UWord32 timestampA = 1;
- WebRtc_Word32 outFreqHzB = _outFileB.SamplingFrequency();
-
- while(!_inFileA.EndOfFile())
- {
- _inFileA.Read10MsData(audioFrame);
- audioFrame._timeStamp = timestampA;
- timestampA += SamplesIn10MsecA;
- CHECK_ERROR(_acmA->Add10MsData(audioFrame));
-
- CHECK_ERROR(_acmA->Process());
-
- CHECK_ERROR(_acmB->PlayoutData10Ms(outFreqHzB, audioFrame));
- _outFileB.Write10MsData(audioFrame._payloadData, audioFrame._payloadDataLengthInSamples);
- }
-#ifdef PRINT_STAT
- _monitor.PrintStatistics(_testMode);
-#endif
- _inFileA.Rewind();
- _monitor.GetStatistics(_statCounter);
- _monitor.ResetStatistics();
-}
-
-void TestVADDTX::OpenOutFile(WebRtc_Word16 testNumber)
-{
- char fileName[500];
- if(_testMode == 0)
- {
- sprintf(fileName, "%s/testVADDTX_autoFile_%02d.pcm",
- webrtc::test::OutputPath().c_str(), testNumber);
- }
- else
- {
- sprintf(fileName, "%s/testVADDTX_outFile_%02d.pcm",
- webrtc::test::OutputPath().c_str(), testNumber);
- }
- _outFileB.Open(fileName, 16000, "wb");
-}
-
-
-WebRtc_Word16 TestVADDTX::VerifyTest()
-{
- // Verify empty frame result
- WebRtc_UWord8 statusEF = 0;
- WebRtc_UWord8 vadPattern = 0;
- WebRtc_UWord8 emptyFramePattern[6];
- CodecInst myCodecParam;
- _acmA->SendCodec(myCodecParam);
- bool dtxInUse = true;
- bool isReplaced = false;
- if ((STR_CASE_CMP(myCodecParam.plname,"G729") == 0) ||
- (STR_CASE_CMP(myCodecParam.plname,"G723") == 0) ||
- (STR_CASE_CMP(myCodecParam.plname,"AMR") == 0) ||
- (STR_CASE_CMP(myCodecParam.plname,"AMR-wb") == 0) ||
- (STR_CASE_CMP(myCodecParam.plname,"speex") == 0))
- {
- _acmA->IsInternalDTXReplacedWithWebRtc(isReplaced);
- if (!isReplaced)
- {
- dtxInUse = false;
- }
- }
-
- // Check for error in VAD/DTX settings
- if (_getStruct.statusDTX != _setStruct.statusDTX){
- // DTX status doesn't match expected
- vadPattern |= 4;
- }
- if (_getStruct.statusDTX){
- if ((!_getStruct.statusVAD && dtxInUse) || (!dtxInUse && (_getStruct.statusVAD !=_setStruct.statusVAD)))
- {
- // Missmatch in VAD setting
- vadPattern |= 2;
- }
- } else {
- if (_getStruct.statusVAD != _setStruct.statusVAD){
- // VAD status doesn't match expected
- vadPattern |= 2;
- }
- }
- if (_getStruct.vadMode != _setStruct.vadMode){
- // VAD Mode doesn't match expected
- vadPattern |= 1;
- }
-
- // Set expected empty frame pattern
- int ii;
- for (ii = 0; ii < 6; ii++) {
- emptyFramePattern[ii] = 0;
- }
- emptyFramePattern[0] = 1; // "kNoEncoding", not important to check. Codecs with packetsize != 80 samples will get this output.
- emptyFramePattern[1] = 1; // Expect to always receive some frames labeled "kActiveNormalEncoded"
- emptyFramePattern[2] = (((!_getStruct.statusDTX && _getStruct.statusVAD) || (!dtxInUse && _getStruct.statusDTX))); // "kPassiveNormalEncoded"
- emptyFramePattern[3] = ((_getStruct.statusDTX && dtxInUse && (_acmA->SendFrequency() == 8000))); // "kPassiveDTXNB"
- emptyFramePattern[4] = ((_getStruct.statusDTX && dtxInUse && (_acmA->SendFrequency() == 16000))); // "kPassiveDTXWB"
- emptyFramePattern[5] = ((_getStruct.statusDTX && dtxInUse && (_acmA->SendFrequency() == 32000))); // "kPassiveDTXSWB"
-
- // Check pattern 1-5 (skip 0)
- for (int ii = 1; ii < 6; ii++)
- {
- if (emptyFramePattern[ii])
- {
- statusEF |= (_statCounter[ii] == 0);
- }
- else
- {
- statusEF |= (_statCounter[ii] > 0);
- }
- }
- if ((statusEF == 0) && (vadPattern == 0))
- {
- if(_testMode != 0)
- {
- printf(" Test OK!\n");
- }
- return 0;
- }
- else
- {
- if (statusEF)
- {
- printf("\t\t\tUnexpected empty frame result!\n");
- }
- if (vadPattern)
- {
- printf("\t\t\tUnexpected SetVAD() result!\tDTX: %d\tVAD: %d\tMode: %d\n", (vadPattern >> 2) & 1, (vadPattern >> 1) & 1, vadPattern & 1);
- }
- return 1;
- }
-}
-
-ActivityMonitor::ActivityMonitor()
-{
- _counter[0] = _counter[1] = _counter[2] = _counter[3] = _counter[4] = _counter[5] = 0;
-}
-
-ActivityMonitor::~ActivityMonitor()
-{
-}
-
-WebRtc_Word32 ActivityMonitor::InFrameType(WebRtc_Word16 frameType)
-{
- _counter[frameType]++;
- return 0;
-}
-
-void ActivityMonitor::PrintStatistics(int testMode)
-{
- if(testMode != 0)
- {
- printf("\n");
- printf("kActiveNormalEncoded kPassiveNormalEncoded kPassiveDTXWB kPassiveDTXNB kPassiveDTXSWB kFrameEmpty\n");
-
- printf("%19u", _counter[1]);
- printf("%22u", _counter[2]);
- printf("%14u", _counter[3]);
- printf("%14u", _counter[4]);
- printf("%14u", _counter[5]);
- printf("%11u", _counter[0]);
-
- printf("\n\n");
- }
-}
-
-void ActivityMonitor::ResetStatistics()
-{
- _counter[0] = _counter[1] = _counter[2] = _counter[3] = _counter[4] = _counter[5] = 0;
-}
-
-void ActivityMonitor::GetStatistics(WebRtc_UWord32* getCounter)
-{
- for (int ii = 0; ii < 6; ii++)
- {
- getCounter[ii] = _counter[ii];
- }
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/TestVADDTX.h b/trunk/src/modules/audio_coding/main/test/TestVADDTX.h
deleted file mode 100644
index e8f9e1e..0000000
--- a/trunk/src/modules/audio_coding/main/test/TestVADDTX.h
+++ /dev/null
@@ -1,90 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef TEST_VAD_DTX_H
-#define TEST_VAD_DTX_H
-
-#include "ACMTest.h"
-#include "Channel.h"
-#include "PCMFile.h"
-
-namespace webrtc {
-
-typedef struct
-{
- bool statusDTX;
- bool statusVAD;
- ACMVADMode vadMode;
-} VADDTXstruct;
-
-class ActivityMonitor : public ACMVADCallback
-{
-public:
- ActivityMonitor();
- ~ActivityMonitor();
- WebRtc_Word32 InFrameType(WebRtc_Word16 frameType);
- void PrintStatistics(int testMode);
- void ResetStatistics();
- void GetStatistics(WebRtc_UWord32* getCounter);
-private:
- // counting according to
- /*enum WebRtcACMEncodingType
- {
- kNoEncoding,
- kActiveNormalEncoded,
- kPassiveNormalEncoded,
- kPassiveDTXNB,
- kPassiveDTXWB,
- kPassiveDTXSWB
- };*/
- WebRtc_UWord32 _counter[6];
-};
-
-class TestVADDTX : public ACMTest
-{
-public:
- TestVADDTX(int testMode);
- ~TestVADDTX();
-
- void Perform();
-private:
- // Registration can be based on codec name only, codec name and sampling frequency, or
- // codec name, sampling frequency and rate.
- WebRtc_Word16 RegisterSendCodec(char side,
- char* codecName,
- WebRtc_Word32 samplingFreqHz = -1,
- WebRtc_Word32 rateKhz = -1);
- void Run();
- void OpenOutFile(WebRtc_Word16 testNumber);
- void runTestCases();
- void runTestInternalDTX();
- void SetVAD(bool statusDTX, bool statusVAD, WebRtc_Word16 vadMode);
- VADDTXstruct GetVAD();
- WebRtc_Word16 VerifyTest();//VADDTXstruct setDTX, VADDTXstruct getDTX);
- AudioCodingModule* _acmA;
- AudioCodingModule* _acmB;
-
- Channel* _channelA2B;
-
- PCMFile _inFileA;
- PCMFile _outFileB;
-
- ActivityMonitor _monitor;
- WebRtc_UWord32 _statCounter[6];
-
- int _testMode;
- int _testResults;
- VADDTXstruct _setStruct;
- VADDTXstruct _getStruct;
-};
-
-} // namespace webrtc
-
-#endif
diff --git a/trunk/src/modules/audio_coding/main/test/Tester.cc b/trunk/src/modules/audio_coding/main/test/Tester.cc
deleted file mode 100644
index e35a0ec..0000000
--- a/trunk/src/modules/audio_coding/main/test/Tester.cc
+++ /dev/null
@@ -1,120 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <string>
-#include <vector>
-
-#include "audio_coding_module.h"
-#include "trace.h"
-
-#include "APITest.h"
-#include "EncodeDecodeTest.h"
-#include "gtest/gtest.h"
-#include "iSACTest.h"
-#include "SpatialAudio.h"
-#include "TestAllCodecs.h"
-#include "TestFEC.h"
-#include "TestStereo.h"
-#include "TestVADDTX.h"
-#include "TwoWayCommunication.h"
-#include "testsupport/fileutils.h"
-
-using webrtc::AudioCodingModule;
-using webrtc::Trace;
-
-// Be sure to create the following directories before running the tests:
-// ./modules/audio_coding/main/test/res_tests
-// ./modules/audio_coding/main/test/res_autotests
-
-// Choose what tests to run by defining one or more of the following:
-#define ACM_AUTO_TEST // Most common codecs and settings will be tested
-//#define ACM_TEST_ENC_DEC // You decide what to test in run time.
- // Used for debugging and for testing while implementing.
-//#define ACM_TEST_TWO_WAY // Debugging
-//#define ACM_TEST_ALL_ENC_DEC // Loop through all defined codecs and settings
-//#define ACM_TEST_STEREO // Run stereo and spatial audio tests
-//#define ACM_TEST_VAD_DTX // Run all VAD/DTX tests
-//#define ACM_TEST_FEC // Test FEC (also called RED)
-//#define ACM_TEST_CODEC_SPEC_API // Only iSAC has codec specfic APIs in this version
-//#define ACM_TEST_FULL_API // Test all APIs with threads (long test)
-
-
-void PopulateTests(std::vector<ACMTest*>* tests)
-{
-
- Trace::CreateTrace();
- std::string trace_file = webrtc::test::OutputPath() + "acm_trace.txt";
- Trace::SetTraceFile(trace_file.c_str());
-
- printf("The following tests will be executed:\n");
-#ifdef ACM_AUTO_TEST
- printf(" ACM auto test\n");
- tests->push_back(new webrtc::EncodeDecodeTest(0));
- tests->push_back(new webrtc::TwoWayCommunication(0));
- tests->push_back(new webrtc::TestAllCodecs(0));
- tests->push_back(new webrtc::TestStereo(0));
- tests->push_back(new webrtc::SpatialAudio(0));
- tests->push_back(new webrtc::TestVADDTX(0));
- tests->push_back(new webrtc::TestFEC(0));
- tests->push_back(new webrtc::ISACTest(0));
-#endif
-#ifdef ACM_TEST_ENC_DEC
- printf(" ACM encode-decode test\n");
- tests->push_back(new webrtc::EncodeDecodeTest(2));
-#endif
-#ifdef ACM_TEST_TWO_WAY
- printf(" ACM two-way communication test\n");
- tests->push_back(new webrtc::TwoWayCommunication(1));
-#endif
-#ifdef ACM_TEST_ALL_ENC_DEC
- printf(" ACM all codecs test\n");
- tests->push_back(new webrtc::TestAllCodecs(1));
-#endif
-#ifdef ACM_TEST_STEREO
- printf(" ACM stereo test\n");
- tests->push_back(new webrtc::TestStereo(1));
- tests->push_back(new webrtc::SpatialAudio(2));
-#endif
-#ifdef ACM_TEST_VAD_DTX
- printf(" ACM VAD-DTX test\n");
- tests->push_back(new webrtc::TestVADDTX(1));
-#endif
-#ifdef ACM_TEST_FEC
- printf(" ACM FEC test\n");
- tests->push_back(new webrtc::TestFEC(1));
-#endif
-#ifdef ACM_TEST_CODEC_SPEC_API
- printf(" ACM codec API test\n");
- tests->push_back(new webrtc::ISACTest(1));
-#endif
-#ifdef ACM_TEST_FULL_API
- printf(" ACM full API test\n");
- tests->push_back(new webrtc::APITest());
-#endif
- printf("\n");
-}
-
-// TODO(kjellander): Make this a proper gtest instead of using this single test
-// to run all the tests.
-TEST(AudioCodingModuleTest, RunAllTests)
-{
- std::vector<ACMTest*> tests;
- PopulateTests(&tests);
- std::vector<ACMTest*>::iterator it;
- for (it=tests.begin() ; it < tests.end(); it++)
- {
- (*it)->Perform();
- delete (*it);
- }
-
- Trace::ReturnTrace();
- printf("ACM test completed\n");
-}
diff --git a/trunk/src/modules/audio_coding/main/test/TimedTrace.cc b/trunk/src/modules/audio_coding/main/test/TimedTrace.cc
deleted file mode 100644
index 6bf301f..0000000
--- a/trunk/src/modules/audio_coding/main/test/TimedTrace.cc
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "TimedTrace.h"
-#include <math.h>
-
-double TimedTrace::_timeEllapsedSec = 0;
-FILE* TimedTrace::_timedTraceFile = NULL;
-
-TimedTrace::TimedTrace()
-{
-
-}
-
-TimedTrace::~TimedTrace()
-{
- if(_timedTraceFile != NULL)
- {
- fclose(_timedTraceFile);
- }
- _timedTraceFile = NULL;
-}
-
-WebRtc_Word16
-TimedTrace::SetUp(char* fileName)
-{
- if(_timedTraceFile == NULL)
- {
- _timedTraceFile = fopen(fileName, "w");
- }
- if(_timedTraceFile == NULL)
- {
- return -1;
- }
- return 0;
-}
-
-void
-TimedTrace::SetTimeEllapsed(double timeEllapsedSec)
-{
- _timeEllapsedSec = timeEllapsedSec;
-}
-
-double
-TimedTrace::TimeEllapsed()
-{
- return _timeEllapsedSec;
-}
-
-void
-TimedTrace::Tick10Msec()
-{
- _timeEllapsedSec += 0.010;
-}
-
-void
-TimedTrace::TimedLogg(char* message)
-{
- unsigned int minutes = (WebRtc_UWord32)floor(_timeEllapsedSec / 60.0);
- double seconds = _timeEllapsedSec - minutes * 60;
- //char myFormat[100] = "%8.2f, %3u:%05.2f: %s\n";
- if(_timedTraceFile != NULL)
- {
- fprintf(_timedTraceFile, "%8.2f, %3u:%05.2f: %s\n",
- _timeEllapsedSec,
- minutes,
- seconds,
- message);
- }
-}
diff --git a/trunk/src/modules/audio_coding/main/test/TimedTrace.h b/trunk/src/modules/audio_coding/main/test/TimedTrace.h
deleted file mode 100644
index d37d287..0000000
--- a/trunk/src/modules/audio_coding/main/test/TimedTrace.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef TIMED_TRACE_H
-#define TIMED_TRACE_H
-
-#include "typedefs.h"
-
-#include <cstdio>
-#include <cstdlib>
-
-
-class TimedTrace
-{
-public:
- TimedTrace();
- ~TimedTrace();
-
- void SetTimeEllapsed(double myTime);
- double TimeEllapsed();
- void Tick10Msec();
- WebRtc_Word16 SetUp(char* fileName);
- void TimedLogg(char* message);
-
-private:
- static double _timeEllapsedSec;
- static FILE* _timedTraceFile;
-
-};
-
-#endif
diff --git a/trunk/src/modules/audio_coding/main/test/TwoWayCommunication.cc b/trunk/src/modules/audio_coding/main/test/TwoWayCommunication.cc
deleted file mode 100644
index c67733a..0000000
--- a/trunk/src/modules/audio_coding/main/test/TwoWayCommunication.cc
+++ /dev/null
@@ -1,524 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "TwoWayCommunication.h"
-
-#include <cctype>
-#include <stdio.h>
-#include <string.h>
-
-#ifdef WIN32
-#include <Windows.h>
-#endif
-
-#include "common_types.h"
-#include "engine_configurations.h"
-#include "gtest/gtest.h"
-#include "PCMFile.h"
-#include "trace.h"
-#include "testsupport/fileutils.h"
-#include "utility.h"
-
-namespace webrtc {
-
-#define MAX_FILE_NAME_LENGTH_BYTE 500
-
-TwoWayCommunication::TwoWayCommunication(int testMode)
-{
- _testMode = testMode;
-}
-
-TwoWayCommunication::~TwoWayCommunication()
-{
- AudioCodingModule::Destroy(_acmA);
- AudioCodingModule::Destroy(_acmB);
-
- AudioCodingModule::Destroy(_acmRefA);
- AudioCodingModule::Destroy(_acmRefB);
-
- delete _channel_A2B;
- delete _channel_B2A;
-
- delete _channelRef_A2B;
- delete _channelRef_B2A;
-#ifdef WEBRTC_DTMF_DETECTION
- if(_dtmfDetectorA != NULL)
- {
- delete _dtmfDetectorA;
- }
- if(_dtmfDetectorB != NULL)
- {
- delete _dtmfDetectorB;
- }
-#endif
- _inFileA.Close();
- _inFileB.Close();
- _outFileA.Close();
- _outFileB.Close();
- _outFileRefA.Close();
- _outFileRefB.Close();
-}
-
-
-WebRtc_UWord8
-TwoWayCommunication::ChooseCodec(WebRtc_UWord8* codecID_A,
- WebRtc_UWord8* codecID_B)
-{
- AudioCodingModule* tmpACM = AudioCodingModule::Create(0);
- WebRtc_UWord8 noCodec = tmpACM->NumberOfCodecs();
- CodecInst codecInst;
- printf("List of Supported Codecs\n");
- printf("========================\n");
- for(WebRtc_UWord8 codecCntr = 0; codecCntr < noCodec; codecCntr++)
- {
- tmpACM->Codec(codecCntr, codecInst);
- printf("%d- %s\n", codecCntr, codecInst.plname);
- }
- printf("\nChoose a send codec for side A [0]: ");
- char myStr[15] = "";
- EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
- *codecID_A = (WebRtc_UWord8)atoi(myStr);
-
- printf("\nChoose a send codec for side B [0]: ");
- EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
- *codecID_B = (WebRtc_UWord8)atoi(myStr);
-
- AudioCodingModule::Destroy(tmpACM);
- printf("\n");
- return 0;
-}
-
-WebRtc_Word16
-TwoWayCommunication::ChooseFile(char* fileName, WebRtc_Word16 maxLen,
- WebRtc_UWord16* frequencyHz)
-{
- char tmpName[MAX_FILE_NAME_LENGTH_BYTE];
- //strcpy(_fileName, "in.pcm");
- //printf("\n\nPlease enter the input file: ");
- EXPECT_TRUE(fgets(tmpName, MAX_FILE_NAME_LENGTH_BYTE, stdin) != NULL);
- tmpName[MAX_FILE_NAME_LENGTH_BYTE-1] = '\0';
- WebRtc_Word16 n = 0;
-
- // removing leading spaces
- while((isspace(tmpName[n]) || iscntrl(tmpName[n])) &&
- (tmpName[n] != 0) &&
- (n < MAX_FILE_NAME_LENGTH_BYTE))
- {
- n++;
- }
- if(n > 0)
- {
- memmove(tmpName, &tmpName[n], MAX_FILE_NAME_LENGTH_BYTE - n);
- }
-
- //removing trailing spaces
- n = (WebRtc_Word16)(strlen(tmpName) - 1);
- if(n >= 0)
- {
- while((isspace(tmpName[n]) || iscntrl(tmpName[n])) &&
- (n >= 0))
- {
- n--;
- }
- }
- if(n >= 0)
- {
- tmpName[n + 1] = '\0';
- }
-
- WebRtc_Word16 len = (WebRtc_Word16)strlen(tmpName);
- if(len > maxLen)
- {
- return -1;
- }
- if(len > 0)
- {
- strncpy(fileName, tmpName, len+1);
- }
- printf("Enter the sampling frequency (in Hz) of the above file [%u]: ",
- *frequencyHz);
- EXPECT_TRUE(fgets(tmpName, 6, stdin) != NULL);
- WebRtc_UWord16 tmpFreq = (WebRtc_UWord16)atoi(tmpName);
- if(tmpFreq > 0)
- {
- *frequencyHz = tmpFreq;
- }
- return 0;
-}
-
-WebRtc_Word16 TwoWayCommunication::SetUp()
-{
- _acmA = AudioCodingModule::Create(1);
- _acmB = AudioCodingModule::Create(2);
-
- _acmRefA = AudioCodingModule::Create(3);
- _acmRefB = AudioCodingModule::Create(4);
-
- WebRtc_UWord8 codecID_A;
- WebRtc_UWord8 codecID_B;
-
- ChooseCodec(&codecID_A, &codecID_B);
- CodecInst codecInst_A;
- CodecInst codecInst_B;
- CodecInst dummyCodec;
- _acmA->Codec(codecID_A, codecInst_A);
- _acmB->Codec(codecID_B, codecInst_B);
-
- _acmA->Codec(6, dummyCodec);
-
- //--- Set A codecs
- CHECK_ERROR(_acmA->RegisterSendCodec(codecInst_A));
- CHECK_ERROR(_acmA->RegisterReceiveCodec(codecInst_B));
-#ifdef WEBRTC_DTMF_DETECTION
- _dtmfDetectorA = new(DTMFDetector);
- CHECK_ERROR(_acmA->RegisterIncomingMessagesCallback(_dtmfDetectorA,
- ACMUSA));
-#endif
- //--- Set ref-A codecs
- CHECK_ERROR(_acmRefA->RegisterSendCodec(codecInst_A));
- CHECK_ERROR(_acmRefA->RegisterReceiveCodec(codecInst_B));
-
- //--- Set B codecs
- CHECK_ERROR(_acmB->RegisterSendCodec(codecInst_B));
- CHECK_ERROR(_acmB->RegisterReceiveCodec(codecInst_A));
-#ifdef WEBRTC_DTMF_DETECTION
- _dtmfDetectorB = new(DTMFDetector);
- CHECK_ERROR(_acmB->RegisterIncomingMessagesCallback(_dtmfDetectorB,
- ACMUSA));
-#endif
-
- //--- Set ref-B codecs
- CHECK_ERROR(_acmRefB->RegisterSendCodec(codecInst_B));
- CHECK_ERROR(_acmRefB->RegisterReceiveCodec(codecInst_A));
-
- char fileName[500];
- char refFileName[500];
- WebRtc_UWord16 frequencyHz;
-
- //--- Input A
- strcpy(fileName, "./test/data/audio_coding/testfile32kHz.pcm");
- frequencyHz = 32000;
- printf("Enter input file at side A [%s]: ", fileName);
- ChooseFile(fileName, 499, &frequencyHz);
-
-
- _inFileA.Open(fileName, frequencyHz, "rb");
-
- //--- Output A
- std::string outputFileA = webrtc::test::OutputPath() + "outA.pcm";
- strcpy(fileName, outputFileA.c_str());
- frequencyHz = 16000;
- printf("Enter output file at side A [%s]: ", fileName);
- ChooseFile(fileName, 499, &frequencyHz);
- _outFileA.Open(fileName, frequencyHz, "wb");
- strcpy(refFileName, "ref_");
- strcat(refFileName, fileName);
- _outFileRefA.Open(refFileName, frequencyHz, "wb");
-
- //--- Input B
- strcpy(fileName, "./test/data/audio_coding/testfile32kHz.pcm");
- frequencyHz = 32000;
- printf("\n\nEnter input file at side B [%s]: ", fileName);
- ChooseFile(fileName, 499, &frequencyHz);
- _inFileB.Open(fileName, frequencyHz, "rb");
-
- //--- Output B
- std::string outputFileB = webrtc::test::OutputPath() + "outB.pcm";
- strcpy(fileName, outputFileB.c_str());
- frequencyHz = 16000;
- printf("Enter output file at side B [%s]: ", fileName);
- ChooseFile(fileName, 499, &frequencyHz);
- _outFileB.Open(fileName, frequencyHz, "wb");
- strcpy(refFileName, "ref_");
- strcat(refFileName, fileName);
- _outFileRefB.Open(refFileName, frequencyHz, "wb");
-
- //--- Set A-to-B channel
- _channel_A2B = new Channel;
- _acmA->RegisterTransportCallback(_channel_A2B);
- _channel_A2B->RegisterReceiverACM(_acmB);
- //--- Do the same for the reference
- _channelRef_A2B = new Channel;
- _acmRefA->RegisterTransportCallback(_channelRef_A2B);
- _channelRef_A2B->RegisterReceiverACM(_acmRefB);
-
- //--- Set B-to-A channel
- _channel_B2A = new Channel;
- _acmB->RegisterTransportCallback(_channel_B2A);
- _channel_B2A->RegisterReceiverACM(_acmA);
- //--- Do the same for reference
- _channelRef_B2A = new Channel;
- _acmRefB->RegisterTransportCallback(_channelRef_B2A);
- _channelRef_B2A->RegisterReceiverACM(_acmRefA);
-
- // The clicks will be more obvious when we
- // are in FAX mode.
- _acmB->SetPlayoutMode(fax);
- _acmRefB->SetPlayoutMode(fax);
-
- return 0;
-}
-
-WebRtc_Word16 TwoWayCommunication::SetUpAutotest()
-{
- _acmA = AudioCodingModule::Create(1);
- _acmB = AudioCodingModule::Create(2);
-
- _acmRefA = AudioCodingModule::Create(3);
- _acmRefB = AudioCodingModule::Create(4);
-
- CodecInst codecInst_A;
- CodecInst codecInst_B;
- CodecInst dummyCodec;
-
- _acmA->Codec("ISAC", codecInst_A, 16000);
- _acmB->Codec("L16", codecInst_B, 8000);
- _acmA->Codec(6, dummyCodec);
-
- //--- Set A codecs
- CHECK_ERROR(_acmA->RegisterSendCodec(codecInst_A));
- CHECK_ERROR(_acmA->RegisterReceiveCodec(codecInst_B));
-#ifdef WEBRTC_DTMF_DETECTION
- _dtmfDetectorA = new(DTMFDetector);
- CHECK_ERROR(_acmA->RegisterIncomingMessagesCallback(_dtmfDetectorA,
- ACMUSA));
-#endif
-
- //--- Set ref-A codecs
- CHECK_ERROR(_acmRefA->RegisterSendCodec(codecInst_A));
- CHECK_ERROR(_acmRefA->RegisterReceiveCodec(codecInst_B));
-
- //--- Set B codecs
- CHECK_ERROR(_acmB->RegisterSendCodec(codecInst_B));
- CHECK_ERROR(_acmB->RegisterReceiveCodec(codecInst_A));
-#ifdef WEBRTC_DTMF_DETECTION
- _dtmfDetectorB = new(DTMFDetector);
- CHECK_ERROR(_acmB->RegisterIncomingMessagesCallback(_dtmfDetectorB,
- ACMUSA));
-#endif
-
- //--- Set ref-B codecs
- CHECK_ERROR(_acmRefB->RegisterSendCodec(codecInst_B));
- CHECK_ERROR(_acmRefB->RegisterReceiveCodec(codecInst_A));
-
- char fileName[500];
- char refFileName[500];
- WebRtc_UWord16 frequencyHz;
-
-
- //--- Input A
- strcpy(fileName, "./test/data/audio_coding/testfile32kHz.pcm");
- frequencyHz = 16000;
- _inFileA.Open(fileName, frequencyHz, "rb");
-
- //--- Output A
- std::string outputFileA = webrtc::test::OutputPath() + "outAutotestA.pcm";
- strcpy(fileName, outputFileA.c_str());
- frequencyHz = 16000;
- _outFileA.Open(fileName, frequencyHz, "wb");
- std::string outputRefFileA = webrtc::test::OutputPath() + "ref_outAutotestA.pcm";
- strcpy(refFileName, outputRefFileA.c_str());
- _outFileRefA.Open(refFileName, frequencyHz, "wb");
-
- //--- Input B
- strcpy(fileName, "./test/data/audio_coding/testfile32kHz.pcm");
- frequencyHz = 16000;
- _inFileB.Open(fileName, frequencyHz, "rb");
-
- //--- Output B
- std::string outputFileB = webrtc::test::OutputPath() + "outAutotestB.pcm";
- strcpy(fileName, outputFileB.c_str());
- frequencyHz = 16000;
- _outFileB.Open(fileName, frequencyHz, "wb");
- std::string outputRefFileB = webrtc::test::OutputPath() + "ref_outAutotestB.pcm";
- strcpy(refFileName, outputRefFileB.c_str());
- _outFileRefB.Open(refFileName, frequencyHz, "wb");
-
- //--- Set A-to-B channel
- _channel_A2B = new Channel;
- _acmA->RegisterTransportCallback(_channel_A2B);
- _channel_A2B->RegisterReceiverACM(_acmB);
- //--- Do the same for the reference
- _channelRef_A2B = new Channel;
- _acmRefA->RegisterTransportCallback(_channelRef_A2B);
- _channelRef_A2B->RegisterReceiverACM(_acmRefB);
-
- //--- Set B-to-A channel
- _channel_B2A = new Channel;
- _acmB->RegisterTransportCallback(_channel_B2A);
- _channel_B2A->RegisterReceiverACM(_acmA);
- //--- Do the same for reference
- _channelRef_B2A = new Channel;
- _acmRefB->RegisterTransportCallback(_channelRef_B2A);
- _channelRef_B2A->RegisterReceiverACM(_acmRefA);
-
- // The clicks will be more obvious when we
- // are in FAX mode.
- _acmB->SetPlayoutMode(fax);
- _acmRefB->SetPlayoutMode(fax);
-
- return 0;
-}
-
-void
-TwoWayCommunication::Perform()
-{
- if(_testMode == 0)
- {
- printf("Running TwoWayCommunication Test");
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
- "---------- TwoWayCommunication ----------");
- SetUpAutotest();
- }
- else
- {
- SetUp();
- }
- unsigned int msecPassed = 0;
- unsigned int secPassed = 0;
-
- WebRtc_Word32 outFreqHzA = _outFileA.SamplingFrequency();
- WebRtc_Word32 outFreqHzB = _outFileB.SamplingFrequency();
-
- AudioFrame audioFrame;
-
- CodecInst codecInst_B;
- CodecInst dummy;
-
- _acmB->SendCodec(codecInst_B);
-
- if(_testMode != 0)
- {
- printf("\n");
- printf("sec:msec A B\n");
- printf("-------- ----- -----\n");
- }
-
- while(!_inFileA.EndOfFile() && !_inFileB.EndOfFile())
- {
- _inFileA.Read10MsData(audioFrame);
- _acmA->Add10MsData(audioFrame);
- _acmRefA->Add10MsData(audioFrame);
-
- _inFileB.Read10MsData(audioFrame);
- _acmB->Add10MsData(audioFrame);
- _acmRefB->Add10MsData(audioFrame);
-
-
- _acmA->Process();
- _acmB->Process();
- _acmRefA->Process();
- _acmRefB->Process();
-
- _acmA->PlayoutData10Ms(outFreqHzA, audioFrame);
- _outFileA.Write10MsData(audioFrame);
-
- _acmRefA->PlayoutData10Ms(outFreqHzA, audioFrame);
- _outFileRefA.Write10MsData(audioFrame);
-
- _acmB->PlayoutData10Ms(outFreqHzB, audioFrame);
- _outFileB.Write10MsData(audioFrame);
-
- _acmRefB->PlayoutData10Ms(outFreqHzB, audioFrame);
- _outFileRefB.Write10MsData(audioFrame);
-
- msecPassed += 10;
- if(msecPassed >= 1000)
- {
- msecPassed = 0;
- secPassed++;
- }
- if(((secPassed%5) == 4) && (msecPassed == 0))
- {
- if(_testMode != 0)
- {
- printf("%3u:%3u ", secPassed, msecPassed);
- }
- _acmA->ResetEncoder();
- if(_testMode == 0)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
- "---------- Errors epected");
- printf(".");
- }
- else
- {
- printf("Reset Encoder (click in side B) ");
- printf("Initialize Sender (no audio in side A)\n");
- }
- CHECK_ERROR(_acmB->InitializeSender());
- }
- if(((secPassed%5) == 4) && (msecPassed >= 990))
- {
- if(_testMode == 0)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
- "----- END: Errors epected");
- printf(".");
- }
- else
- {
- printf("%3u:%3u ", secPassed, msecPassed);
- printf(" ");
- printf("Register Send Codec (audio back in side A)\n");
- }
- CHECK_ERROR(_acmB->RegisterSendCodec(codecInst_B));
- CHECK_ERROR(_acmB->SendCodec(dummy));
- }
- if(((secPassed%7) == 6) && (msecPassed == 0))
- {
- CHECK_ERROR(_acmB->ResetDecoder());
- if(_testMode == 0)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
- "---------- Errors epected");
- printf(".");
- }
- else
- {
- printf("%3u:%3u ", secPassed, msecPassed);
- printf("Initialize Receiver (no audio in side A) ");
- printf("Reset Decoder\n");
- }
- CHECK_ERROR(_acmA->InitializeReceiver());
- }
- if(((secPassed%7) == 6) && (msecPassed >= 990))
- {
- if(_testMode == 0)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioCoding, -1,
- "----- END: Errors epected");
- printf(".");
- }
- else
- {
- printf("%3u:%3u ", secPassed, msecPassed);
- printf("Register Receive Coded (audio back in side A)\n");
- }
- CHECK_ERROR(_acmA->RegisterReceiveCodec(codecInst_B));
- }
- //Sleep(9);
- }
- if(_testMode == 0)
- {
- printf("Done!\n");
- }
-
-#ifdef WEBRTC_DTMF_DETECTION
- printf("\nDTMF at Side A\n");
- _dtmfDetectorA->PrintDetectedDigits();
-
- printf("\nDTMF at Side B\n");
- _dtmfDetectorB->PrintDetectedDigits();
-#endif
-
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/TwoWayCommunication.h b/trunk/src/modules/audio_coding/main/test/TwoWayCommunication.h
deleted file mode 100644
index 0b33317..0000000
--- a/trunk/src/modules/audio_coding/main/test/TwoWayCommunication.h
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef TWO_WAY_COMMUNICATION_H
-#define TWO_WAY_COMMUNICATION_H
-
-#include "ACMTest.h"
-#include "Channel.h"
-#include "PCMFile.h"
-#include "audio_coding_module.h"
-#include "utility.h"
-
-namespace webrtc {
-
-class TwoWayCommunication : public ACMTest
-{
-public:
- TwoWayCommunication(int testMode = 1);
- ~TwoWayCommunication();
-
- void Perform();
-private:
- WebRtc_UWord8 ChooseCodec(WebRtc_UWord8* codecID_A, WebRtc_UWord8* codecID_B);
- WebRtc_Word16 ChooseFile(char* fileName, WebRtc_Word16 maxLen, WebRtc_UWord16* frequencyHz);
- WebRtc_Word16 SetUp();
- WebRtc_Word16 SetUpAutotest();
-
- AudioCodingModule* _acmA;
- AudioCodingModule* _acmB;
-
- AudioCodingModule* _acmRefA;
- AudioCodingModule* _acmRefB;
-
- Channel* _channel_A2B;
- Channel* _channel_B2A;
-
- Channel* _channelRef_A2B;
- Channel* _channelRef_B2A;
-
- PCMFile _inFileA;
- PCMFile _inFileB;
-
- PCMFile _outFileA;
- PCMFile _outFileB;
-
- PCMFile _outFileRefA;
- PCMFile _outFileRefB;
-
- DTMFDetector* _dtmfDetectorA;
- DTMFDetector* _dtmfDetectorB;
-
- int _testMode;
-};
-
-} // namespace webrtc
-
-#endif
diff --git a/trunk/src/modules/audio_coding/main/test/iSACTest.cc b/trunk/src/modules/audio_coding/main/test/iSACTest.cc
deleted file mode 100644
index 966c4c5..0000000
--- a/trunk/src/modules/audio_coding/main/test/iSACTest.cc
+++ /dev/null
@@ -1,598 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cctype>
-#include <stdio.h>
-#include <string.h>
-
-#if _WIN32
-#include <windows.h>
-#elif WEBRTC_LINUX
-#include <ctime>
-#else
-#include <sys/time.h>
-#include <time.h>
-#endif
-
-#include "event_wrapper.h"
-#include "iSACTest.h"
-#include "utility.h"
-#include "trace.h"
-#include "testsupport/fileutils.h"
-#include "tick_util.h"
-
-namespace webrtc {
-
-void SetISACConfigDefault(
- ACMTestISACConfig& isacConfig)
-{
- isacConfig.currentRateBitPerSec = 0;
- isacConfig.currentFrameSizeMsec = 0;
- isacConfig.maxRateBitPerSec = 0;
- isacConfig.maxPayloadSizeByte = 0;
- isacConfig.encodingMode = -1;
- isacConfig.initRateBitPerSec = 0;
- isacConfig.initFrameSizeInMsec = 0;
- isacConfig.enforceFrameSize = false;
- return;
-}
-
-
-WebRtc_Word16 SetISAConfig(
- ACMTestISACConfig& isacConfig,
- AudioCodingModule* acm,
- int testMode)
-{
-
- if((isacConfig.currentRateBitPerSec != 0) ||
- (isacConfig.currentFrameSizeMsec != 0))
- {
- CodecInst sendCodec;
- acm->SendCodec(sendCodec);
- if(isacConfig.currentRateBitPerSec < 0)
- {
- sendCodec.rate = -1;
- CHECK_ERROR(acm->RegisterSendCodec(sendCodec));
- if(testMode != 0)
- {
- printf("ISAC-%s Registered in adaptive (channel-dependent) mode.\n",
- (sendCodec.plfreq == 32000)? "swb":"wb");
- }
- }
- else
- {
-
- if(isacConfig.currentRateBitPerSec != 0)
- {
- sendCodec.rate = isacConfig.currentRateBitPerSec;
- }
- if(isacConfig.currentFrameSizeMsec != 0)
- {
- sendCodec.pacsize = isacConfig.currentFrameSizeMsec *
- (sendCodec.plfreq / 1000);
- }
- CHECK_ERROR(acm->RegisterSendCodec(sendCodec));
- if(testMode != 0)
- {
- printf("Target rate is set to %d bit/sec with frame-size %d ms \n",
- (int)isacConfig.currentRateBitPerSec,
- (int)sendCodec.pacsize / (sendCodec.plfreq / 1000));
- }
- }
- }
-
- if(isacConfig.maxRateBitPerSec > 0)
- {
- CHECK_ERROR(acm->SetISACMaxRate(isacConfig.maxRateBitPerSec));
- if(testMode != 0)
- {
- printf("Max rate is set to %u bit/sec\n",
- isacConfig.maxRateBitPerSec);
- }
- }
- if(isacConfig.maxPayloadSizeByte > 0)
- {
- CHECK_ERROR(acm->SetISACMaxPayloadSize(isacConfig.maxPayloadSizeByte));
- if(testMode != 0)
- {
- printf("Max payload-size is set to %u bit/sec\n",
- isacConfig.maxPayloadSizeByte);
- }
- }
- if((isacConfig.initFrameSizeInMsec != 0) ||
- (isacConfig.initRateBitPerSec != 0))
- {
- CHECK_ERROR(acm->ConfigISACBandwidthEstimator(
- (WebRtc_UWord8)isacConfig.initFrameSizeInMsec,
- (WebRtc_UWord16)isacConfig.initRateBitPerSec,
- isacConfig.enforceFrameSize));
- if((isacConfig.initFrameSizeInMsec != 0) && (testMode != 0))
- {
- printf("Initialize BWE to %d msec frame-size\n",
- isacConfig.initFrameSizeInMsec);
- }
- if((isacConfig.initRateBitPerSec != 0) && (testMode != 0))
- {
- printf("Initialize BWE to %u bit/sec send-bandwidth\n",
- isacConfig.initRateBitPerSec);
- }
- }
-
- return 0;
-}
-
-
-ISACTest::ISACTest(int testMode)
-{
- _testMode = testMode;
-}
-
-ISACTest::~ISACTest()
-{
- AudioCodingModule::Destroy(_acmA);
- AudioCodingModule::Destroy(_acmB);
-
- delete _channel_A2B;
- delete _channel_B2A;
-}
-
-
-WebRtc_Word16
-ISACTest::Setup()
-{
- int codecCntr;
- CodecInst codecParam;
-
- _acmA = AudioCodingModule::Create(1);
- _acmB = AudioCodingModule::Create(2);
-
- for(codecCntr = 0; codecCntr < AudioCodingModule::NumberOfCodecs(); codecCntr++)
- {
- AudioCodingModule::Codec(codecCntr, codecParam);
- if(!STR_CASE_CMP(codecParam.plname, "ISAC") && codecParam.plfreq == 16000)
- {
- memcpy(&_paramISAC16kHz, &codecParam, sizeof(CodecInst));
- _idISAC16kHz = codecCntr;
- }
- if(!STR_CASE_CMP(codecParam.plname, "ISAC") && codecParam.plfreq == 32000)
- {
- memcpy(&_paramISAC32kHz, &codecParam, sizeof(CodecInst));
- _idISAC32kHz = codecCntr;
- }
- }
-
- // register both iSAC-wb & iSAC-swb in both sides as receiver codecs
- CHECK_ERROR(_acmA->RegisterReceiveCodec(_paramISAC16kHz));
- CHECK_ERROR(_acmA->RegisterReceiveCodec(_paramISAC32kHz));
- CHECK_ERROR(_acmB->RegisterReceiveCodec(_paramISAC16kHz));
- CHECK_ERROR(_acmB->RegisterReceiveCodec(_paramISAC32kHz));
-
- //--- Set A-to-B channel
- _channel_A2B = new Channel;
- CHECK_ERROR(_acmA->RegisterTransportCallback(_channel_A2B));
- _channel_A2B->RegisterReceiverACM(_acmB);
-
- //--- Set B-to-A channel
- _channel_B2A = new Channel;
- CHECK_ERROR(_acmB->RegisterTransportCallback(_channel_B2A));
- _channel_B2A->RegisterReceiverACM(_acmA);
-
- strncpy(_fileNameSWB, "./test/data/audio_coding/testfile32kHz.pcm",
- MAX_FILE_NAME_LENGTH_BYTE);
-
- _acmB->RegisterSendCodec(_paramISAC16kHz);
- _acmA->RegisterSendCodec(_paramISAC32kHz);
-
- if(_testMode != 0)
- {
- printf("Side A Send Codec\n");
- printf("%s %d\n", _paramISAC32kHz.plname, _paramISAC32kHz.plfreq);
-
- printf("Side B Send Codec\n");
- printf("%s %d\n", _paramISAC16kHz.plname, _paramISAC16kHz.plfreq);
- }
-
- _inFileA.Open(_fileNameSWB, 32000, "rb");
- std::string fileNameA = webrtc::test::OutputPath() + "testisac_a.pcm";
- std::string fileNameB = webrtc::test::OutputPath() + "testisac_b.pcm";
- _outFileA.Open(fileNameA.c_str(), 32000, "wb");
- _outFileB.Open(fileNameB.c_str(), 32000, "wb");
-
- while(!_inFileA.EndOfFile())
- {
- Run10ms();
- }
- CodecInst receiveCodec;
- CHECK_ERROR(_acmA->ReceiveCodec(receiveCodec));
- if(_testMode != 0)
- {
- printf("Side A Receive Codec\n");
- printf("%s %d\n", receiveCodec.plname, receiveCodec.plfreq);
- }
-
- CHECK_ERROR(_acmB->ReceiveCodec(receiveCodec));
- if(_testMode != 0)
- {
- printf("Side B Receive Codec\n");
- printf("%s %d\n", receiveCodec.plname, receiveCodec.plfreq);
- }
-
- _inFileA.Close();
- _outFileA.Close();
- _outFileB.Close();
-
- return 0;
-}
-
-
-void
-ISACTest::Perform()
-{
- if(_testMode == 0)
- {
- printf("Running iSAC Test");
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceAudioCoding, -1, "---------- iSACTest ----------");
- }
-
- Setup();
-
- WebRtc_Word16 testNr = 0;
- ACMTestISACConfig wbISACConfig;
- ACMTestISACConfig swbISACConfig;
-
- SetISACConfigDefault(wbISACConfig);
- SetISACConfigDefault(swbISACConfig);
-
- wbISACConfig.currentRateBitPerSec = -1;
- swbISACConfig.currentRateBitPerSec = -1;
- testNr++;
- EncodeDecode(testNr, wbISACConfig, swbISACConfig);
-
- if (_testMode != 0)
- {
- SetISACConfigDefault(wbISACConfig);
- SetISACConfigDefault(swbISACConfig);
-
- wbISACConfig.currentRateBitPerSec = -1;
- swbISACConfig.currentRateBitPerSec = -1;
- wbISACConfig.initRateBitPerSec = 13000;
- wbISACConfig.initFrameSizeInMsec = 60;
- swbISACConfig.initRateBitPerSec = 20000;
- swbISACConfig.initFrameSizeInMsec = 30;
- testNr++;
- EncodeDecode(testNr, wbISACConfig, swbISACConfig);
-
- SetISACConfigDefault(wbISACConfig);
- SetISACConfigDefault(swbISACConfig);
-
- wbISACConfig.currentRateBitPerSec = 20000;
- swbISACConfig.currentRateBitPerSec = 48000;
- testNr++;
- EncodeDecode(testNr, wbISACConfig, swbISACConfig);
-
- wbISACConfig.currentRateBitPerSec = 16000;
- swbISACConfig.currentRateBitPerSec = 30000;
- wbISACConfig.currentFrameSizeMsec = 60;
- testNr++;
- EncodeDecode(testNr, wbISACConfig, swbISACConfig);
- }
-
- SetISACConfigDefault(wbISACConfig);
- SetISACConfigDefault(swbISACConfig);
- testNr++;
- EncodeDecode(testNr, wbISACConfig, swbISACConfig);
-
- int user_input;
- if((_testMode == 0) || (_testMode == 1))
- {
- swbISACConfig.maxPayloadSizeByte = (WebRtc_UWord16)200;
- wbISACConfig.maxPayloadSizeByte = (WebRtc_UWord16)200;
- }
- else
- {
- printf("Enter the max payload-size for side A: ");
- CHECK_ERROR(scanf("%d", &user_input));
- swbISACConfig.maxPayloadSizeByte = (WebRtc_UWord16)user_input;
- printf("Enter the max payload-size for side B: ");
- CHECK_ERROR(scanf("%d", &user_input));
- wbISACConfig.maxPayloadSizeByte = (WebRtc_UWord16)user_input;
- }
- testNr++;
- EncodeDecode(testNr, wbISACConfig, swbISACConfig);
-
- _acmA->ResetEncoder();
- _acmB->ResetEncoder();
- SetISACConfigDefault(wbISACConfig);
- SetISACConfigDefault(swbISACConfig);
-
- if((_testMode == 0) || (_testMode == 1))
- {
- swbISACConfig.maxRateBitPerSec = (WebRtc_UWord32)48000;
- wbISACConfig.maxRateBitPerSec = (WebRtc_UWord32)48000;
- }
- else
- {
- printf("Enter the max rate for side A: ");
- CHECK_ERROR(scanf("%d", &user_input));
- swbISACConfig.maxRateBitPerSec = (WebRtc_UWord32)user_input;
- printf("Enter the max rate for side B: ");
- CHECK_ERROR(scanf("%d", &user_input));
- wbISACConfig.maxRateBitPerSec = (WebRtc_UWord32)user_input;
- }
-
- testNr++;
- EncodeDecode(testNr, wbISACConfig, swbISACConfig);
-
-
- testNr++;
- if(_testMode == 0)
- {
- SwitchingSamplingRate(testNr, 4);
- printf("Done!\n");
- }
- else
- {
- SwitchingSamplingRate(testNr, 80);
- }
-}
-
-
-void
-ISACTest::Run10ms()
-{
- AudioFrame audioFrame;
-
- _inFileA.Read10MsData(audioFrame);
- CHECK_ERROR(_acmA->Add10MsData(audioFrame));
-
- CHECK_ERROR(_acmB->Add10MsData(audioFrame));
-
- CHECK_ERROR(_acmA->Process());
- CHECK_ERROR(_acmB->Process());
-
- CHECK_ERROR(_acmA->PlayoutData10Ms(32000, audioFrame));
- _outFileA.Write10MsData(audioFrame);
-
- CHECK_ERROR(_acmB->PlayoutData10Ms(32000, audioFrame));
- _outFileB.Write10MsData(audioFrame);
-}
-
-void
-ISACTest::EncodeDecode(
- int testNr,
- ACMTestISACConfig& wbISACConfig,
- ACMTestISACConfig& swbISACConfig)
-{
- if(_testMode == 0)
- {
- printf(".");
- }
- else
- {
- printf("\nTest %d:\n\n", testNr);
- }
- char fileNameOut[MAX_FILE_NAME_LENGTH_BYTE];
-
- // Files in Side A
- _inFileA.Open(_fileNameSWB, 32000, "rb", true);
- if(_testMode == 0)
- {
- sprintf(fileNameOut,
- "%s/out_iSACTest_%s_%02d.pcm",
- webrtc::test::OutputPath().c_str(),
- "A",
- testNr);
- }
- else
- {
- sprintf(fileNameOut,
- "%s/out%s_%02d.pcm",
- webrtc::test::OutputPath().c_str(),
- "A",
- testNr);
- }
- _outFileA.Open(fileNameOut, 32000, "wb");
-
- // Files in Side B
- _inFileB.Open(_fileNameSWB, 32000, "rb", true);
- if(_testMode == 0)
- {
- sprintf(fileNameOut,
- "%s/out_iSACTest_%s_%02d.pcm",
- webrtc::test::OutputPath().c_str(),
- "B",
- testNr);
- }
- else
- {
- sprintf(fileNameOut,
- "%s/out%s_%02d.pcm",
- webrtc::test::OutputPath().c_str(),
- "B",
- testNr);
- }
- _outFileB.Open(fileNameOut, 32000, "wb");
-
- CHECK_ERROR(_acmA->RegisterSendCodec(_paramISAC16kHz));
- CHECK_ERROR(_acmA->RegisterSendCodec(_paramISAC32kHz));
-
- CHECK_ERROR(_acmB->RegisterSendCodec(_paramISAC32kHz));
- CHECK_ERROR(_acmB->RegisterSendCodec(_paramISAC16kHz));
- if(_testMode != 0)
- {
- printf("Side A Sending Super-Wideband \n");
- printf("Side B Sending Wideband\n\n");
- }
-
- SetISAConfig(swbISACConfig, _acmA, _testMode);
- SetISAConfig(wbISACConfig, _acmB, _testMode);
-
- bool adaptiveMode = false;
- if((swbISACConfig.currentRateBitPerSec == -1) ||
- (wbISACConfig.currentRateBitPerSec == -1))
- {
- adaptiveMode = true;
- }
- _myTimer.Reset();
- _channel_A2B->ResetStats();
- _channel_B2A->ResetStats();
-
- char currentTime[500];
- if(_testMode == 2) printf("\n");
- CodecInst sendCodec;
- EventWrapper* myEvent = EventWrapper::Create();
- myEvent->StartTimer(true, 10);
- while(!(_inFileA.EndOfFile() || _inFileA.Rewinded()))
- {
- Run10ms();
- _myTimer.Tick10ms();
- _myTimer.CurrentTimeHMS(currentTime);
- if(_testMode == 2) printf("\r%s ", currentTime);
-
- if((adaptiveMode) && (_testMode != 0))
- {
- myEvent->Wait(5000);
-
- _acmA->SendCodec(sendCodec);
- if(_testMode == 2) printf("[%d] ", sendCodec.rate);
- _acmB->SendCodec(sendCodec);
- if(_testMode == 2) printf("[%d] ", sendCodec.rate);
- }
- }
-
- if(_testMode != 0)
- {
- printf("\n\nSide A statistics\n\n");
- _channel_A2B->PrintStats(_paramISAC32kHz);
-
- printf("\n\nSide B statistics\n\n");
- _channel_B2A->PrintStats(_paramISAC16kHz);
- }
-
- _channel_A2B->ResetStats();
- _channel_B2A->ResetStats();
-
- if(_testMode != 0) printf("\n");
- _outFileA.Close();
- _outFileB.Close();
- _inFileA.Close();
- _inFileB.Close();
-}
-
-void
-ISACTest::SwitchingSamplingRate(
- int testNr,
- int maxSampRateChange)
-{
- char fileNameOut[MAX_FILE_NAME_LENGTH_BYTE];
-
- // Files in Side A
- _inFileA.Open(_fileNameSWB, 32000, "rb");
- if(_testMode == 0)
- {
- sprintf(fileNameOut,
- "%s/out_iSACTest_%s_%02d.pcm",
- webrtc::test::OutputPath().c_str(),
- "A",
- testNr);
- }
- else
- {
- printf("\nTest %d", testNr);
- printf(" Alternate between WB and SWB at the sender Side\n\n");
- sprintf(fileNameOut,
- "%s/out%s_%02d.pcm",
- webrtc::test::OutputPath().c_str(),
- "A",
- testNr);
- }
- _outFileA.Open(fileNameOut, 32000, "wb", true);
-
- // Files in Side B
- _inFileB.Open(_fileNameSWB, 32000, "rb");
- if(_testMode == 0)
- {
- sprintf(fileNameOut,
- "%s/out_iSACTest_%s_%02d.pcm",
- webrtc::test::OutputPath().c_str(),
- "B",
- testNr);
- }
- else
- {
- sprintf(fileNameOut, "%s/out%s_%02d.pcm",
- webrtc::test::OutputPath().c_str(),
- "B",
- testNr);
- }
- _outFileB.Open(fileNameOut, 32000, "wb", true);
-
- CHECK_ERROR(_acmA->RegisterSendCodec(_paramISAC32kHz));
- CHECK_ERROR(_acmB->RegisterSendCodec(_paramISAC16kHz));
- if(_testMode != 0)
- {
- printf("Side A Sending Super-Wideband \n");
- printf("Side B Sending Wideband\n");
- }
-
- int numSendCodecChanged = 0;
- _myTimer.Reset();
- char currentTime[50];
- while(numSendCodecChanged < (maxSampRateChange<<1))
- {
- Run10ms();
- _myTimer.Tick10ms();
- _myTimer.CurrentTimeHMS(currentTime);
- if(_testMode == 2) printf("\r%s", currentTime);
- if(_inFileA.EndOfFile())
- {
- if(_inFileA.SamplingFrequency() == 16000)
- {
- if(_testMode != 0) printf("\nSide A switched to Send Super-Wideband\n");
- _inFileA.Close();
- _inFileA.Open(_fileNameSWB, 32000, "rb");
- CHECK_ERROR(_acmA->RegisterSendCodec(_paramISAC32kHz));
- }
- else
- {
- if(_testMode != 0) printf("\nSide A switched to Send Wideband\n");
- _inFileA.Close();
- _inFileA.Open(_fileNameSWB, 32000, "rb");
- CHECK_ERROR(_acmA->RegisterSendCodec(_paramISAC16kHz));
- }
- numSendCodecChanged++;
- }
-
- if(_inFileB.EndOfFile())
- {
- if(_inFileB.SamplingFrequency() == 16000)
- {
- if(_testMode != 0) printf("\nSide B switched to Send Super-Wideband\n");
- _inFileB.Close();
- _inFileB.Open(_fileNameSWB, 32000, "rb");
- CHECK_ERROR(_acmB->RegisterSendCodec(_paramISAC32kHz));
- }
- else
- {
- if(_testMode != 0) printf("\nSide B switched to Send Wideband\n");
- _inFileB.Close();
- _inFileB.Open(_fileNameSWB, 32000, "rb");
- CHECK_ERROR(_acmB->RegisterSendCodec(_paramISAC16kHz));
- }
- numSendCodecChanged++;
- }
- }
- _outFileA.Close();
- _outFileB.Close();
- _inFileA.Close();
- _inFileB.Close();
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/iSACTest.h b/trunk/src/modules/audio_coding/main/test/iSACTest.h
deleted file mode 100644
index 17bacad..0000000
--- a/trunk/src/modules/audio_coding/main/test/iSACTest.h
+++ /dev/null
@@ -1,103 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef ACM_ISAC_TEST_H
-#define ACM_ISAC_TEST_H
-
-#include "ACMTest.h"
-#include "Channel.h"
-#include "PCMFile.h"
-#include "audio_coding_module.h"
-#include "utility.h"
-#include "common_types.h"
-
-#define MAX_FILE_NAME_LENGTH_BYTE 500
-#define NO_OF_CLIENTS 15
-
-namespace webrtc {
-
-struct ACMTestISACConfig
-{
- WebRtc_Word32 currentRateBitPerSec;
- WebRtc_Word16 currentFrameSizeMsec;
- WebRtc_UWord32 maxRateBitPerSec;
- WebRtc_Word16 maxPayloadSizeByte;
- WebRtc_Word16 encodingMode;
- WebRtc_UWord32 initRateBitPerSec;
- WebRtc_Word16 initFrameSizeInMsec;
- bool enforceFrameSize;
-};
-
-
-
-class ISACTest : public ACMTest
-{
-public:
- ISACTest(int testMode);
- ~ISACTest();
-
- void Perform();
-private:
- WebRtc_Word16 Setup();
- WebRtc_Word16 SetupConference();
- WebRtc_Word16 RunConference();
-
-
- void Run10ms();
-
- void EncodeDecode(
- int testNr,
- ACMTestISACConfig& wbISACConfig,
- ACMTestISACConfig& swbISACConfig);
-
- void TestBWE(
- int testNr);
-
- void SwitchingSamplingRate(
- int testNr,
- int maxSampRateChange);
-
- AudioCodingModule* _acmA;
- AudioCodingModule* _acmB;
-
- Channel* _channel_A2B;
- Channel* _channel_B2A;
-
- PCMFile _inFileA;
- PCMFile _inFileB;
-
- PCMFile _outFileA;
- PCMFile _outFileB;
-
- WebRtc_UWord8 _idISAC16kHz;
- WebRtc_UWord8 _idISAC32kHz;
- CodecInst _paramISAC16kHz;
- CodecInst _paramISAC32kHz;
-
- char _fileNameWB[MAX_FILE_NAME_LENGTH_BYTE];
- char _fileNameSWB[MAX_FILE_NAME_LENGTH_BYTE];
-
- ACMTestTimer _myTimer;
- int _testMode;
-
- AudioCodingModule* _defaultACM32;
- AudioCodingModule* _defaultACM16;
-
- AudioCodingModule* _confACM[NO_OF_CLIENTS];
- AudioCodingModule* _clientACM[NO_OF_CLIENTS];
- Channel* _conf2Client[NO_OF_CLIENTS];
- Channel* _client2Conf[NO_OF_CLIENTS];
-
- PCMFile _clientOutFile[NO_OF_CLIENTS];
-};
-
-} // namespace webrtc
-
-#endif
diff --git a/trunk/src/modules/audio_coding/main/test/utility.cc b/trunk/src/modules/audio_coding/main/test/utility.cc
deleted file mode 100644
index 56acbf7..0000000
--- a/trunk/src/modules/audio_coding/main/test/utility.cc
+++ /dev/null
@@ -1,434 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "utility.h"
-
-#include <assert.h>
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "audio_coding_module.h"
-#include "common_types.h"
-#include "gtest/gtest.h"
-
-#define NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE 13
-
-namespace webrtc {
-
-ACMTestTimer::ACMTestTimer() :
-_msec(0),
-_sec(0),
-_min(0),
-_hour(0)
-{
- return;
-}
-
-ACMTestTimer::~ACMTestTimer()
-{
- return;
-}
-
-void ACMTestTimer::Reset()
-{
- _msec = 0;
- _sec = 0;
- _min = 0;
- _hour = 0;
- return;
-}
-void ACMTestTimer::Tick10ms()
-{
- _msec += 10;
- Adjust();
- return;
-}
-
-void ACMTestTimer::Tick1ms()
-{
- _msec++;
- Adjust();
- return;
-}
-
-void ACMTestTimer::Tick100ms()
-{
- _msec += 100;
- Adjust();
- return;
-}
-
-void ACMTestTimer::Tick1sec()
-{
- _sec++;
- Adjust();
- return;
-}
-
-void ACMTestTimer::CurrentTimeHMS(char* currTime)
-{
- sprintf(currTime, "%4lu:%02u:%06.3f", _hour, _min, (double)_sec + (double)_msec / 1000.);
- return;
-}
-
-void ACMTestTimer::CurrentTime(
- unsigned long& h,
- unsigned char& m,
- unsigned char& s,
- unsigned short& ms)
-{
- h = _hour;
- m = _min;
- s = _sec;
- ms = _msec;
- return;
-}
-
-void ACMTestTimer::Adjust()
-{
- unsigned int n;
- if(_msec >= 1000)
- {
- n = _msec / 1000;
- _msec -= (1000 * n);
- _sec += n;
- }
- if(_sec >= 60)
- {
- n = _sec / 60;
- _sec -= (n * 60);
- _min += n;
- }
- if(_min >= 60)
- {
- n = _min / 60;
- _min -= (n * 60);
- _hour += n;
- }
-}
-
-
-WebRtc_Word16
-ChooseCodec(
- CodecInst& codecInst)
-{
-
- PrintCodecs();
- //AudioCodingModule* tmpACM = AudioCodingModule::Create(0);
- WebRtc_UWord8 noCodec = AudioCodingModule::NumberOfCodecs();
- WebRtc_Word8 codecID;
- bool outOfRange = false;
- char myStr[15] = "";
- do
- {
- printf("\nChoose a codec [0]: ");
- EXPECT_TRUE(fgets(myStr, 10, stdin) != NULL);
- codecID = atoi(myStr);
- if((codecID < 0) || (codecID >= noCodec))
- {
- printf("\nOut of range.\n");
- outOfRange = true;
- }
- } while(outOfRange);
-
- CHECK_ERROR(AudioCodingModule::Codec((WebRtc_UWord8)codecID, codecInst));
- return 0;
-}
-
-void
-PrintCodecs()
-{
- WebRtc_UWord8 noCodec = AudioCodingModule::NumberOfCodecs();
-
- CodecInst codecInst;
- printf("No Name [Hz] [bps]\n");
- for(WebRtc_UWord8 codecCntr = 0; codecCntr < noCodec; codecCntr++)
- {
- AudioCodingModule::Codec(codecCntr, codecInst);
- printf("%2d- %-18s %5d %6d\n",
- codecCntr, codecInst.plname, codecInst.plfreq, codecInst.rate);
- }
-
-}
-
-CircularBuffer::CircularBuffer(WebRtc_UWord32 len):
-_buff(NULL),
-_idx(0),
-_buffIsFull(false),
-_calcAvg(false),
-_calcVar(false),
-_sum(0),
-_sumSqr(0)
-{
- _buff = new double[len];
- if(_buff == NULL)
- {
- _buffLen = 0;
- }
- else
- {
- for(WebRtc_UWord32 n = 0; n < len; n++)
- {
- _buff[n] = 0;
- }
- _buffLen = len;
- }
-}
-
-CircularBuffer::~CircularBuffer()
-{
- if(_buff != NULL)
- {
- delete [] _buff;
- _buff = NULL;
- }
-}
-
-void
-CircularBuffer::Update(
- const double newVal)
-{
- assert(_buffLen > 0);
-
- // store the value that is going to be overwritten
- double oldVal = _buff[_idx];
- // record the new value
- _buff[_idx] = newVal;
- // increment the index, to point to where we would
- // write next
- _idx++;
- // it is a circular buffer, if we are at the end
- // we have to cycle to the beginning
- if(_idx >= _buffLen)
- {
- // flag that the buffer is filled up.
- _buffIsFull = true;
- _idx = 0;
- }
-
- // Update
-
- if(_calcAvg)
- {
- // for the average we have to update
- // the sum
- _sum += (newVal - oldVal);
- }
-
- if(_calcVar)
- {
- // to calculate variance we have to update
- // the sum of squares
- _sumSqr += (double)(newVal - oldVal) * (double)(newVal + oldVal);
- }
-}
-
-void
-CircularBuffer::SetArithMean(
- bool enable)
-{
- assert(_buffLen > 0);
-
- if(enable && !_calcAvg)
- {
- WebRtc_UWord32 lim;
- if(_buffIsFull)
- {
- lim = _buffLen;
- }
- else
- {
- lim = _idx;
- }
- _sum = 0;
- for(WebRtc_UWord32 n = 0; n < lim; n++)
- {
- _sum += _buff[n];
- }
- }
- _calcAvg = enable;
-}
-
-void
-CircularBuffer::SetVariance(
- bool enable)
-{
- assert(_buffLen > 0);
-
- if(enable && !_calcVar)
- {
- WebRtc_UWord32 lim;
- if(_buffIsFull)
- {
- lim = _buffLen;
- }
- else
- {
- lim = _idx;
- }
- _sumSqr = 0;
- for(WebRtc_UWord32 n = 0; n < lim; n++)
- {
- _sumSqr += _buff[n] * _buff[n];
- }
- }
- _calcAvg = enable;
-}
-
-WebRtc_Word16
-CircularBuffer::ArithMean(double& mean)
-{
- assert(_buffLen > 0);
-
- if(_buffIsFull)
- {
-
- mean = _sum / (double)_buffLen;
- return 0;
- }
- else
- {
- if(_idx > 0)
- {
- mean = _sum / (double)_idx;
- return 0;
- }
- else
- {
- return -1;
- }
-
- }
-}
-
-WebRtc_Word16
-CircularBuffer::Variance(double& var)
-{
- assert(_buffLen > 0);
-
- if(_buffIsFull)
- {
- var = _sumSqr / (double)_buffLen;
- return 0;
- }
- else
- {
- if(_idx > 0)
- {
- var = _sumSqr / (double)_idx;
- return 0;
- }
- else
- {
- return -1;
- }
- }
-}
-
-
-
-bool
-FixedPayloadTypeCodec(const char* payloadName)
-{
- char fixPayloadTypeCodecs[NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE][32] = {
- "PCMU",
- "PCMA",
- "GSM",
- "G723",
- "DVI4",
- "LPC",
- "PCMA",
- "G722",
- "QCELP",
- "CN",
- "MPA",
- "G728",
- "G729"
- };
-
- for(int n = 0; n < NUM_CODECS_WITH_FIXED_PAYLOAD_TYPE; n++)
- {
- if(!STR_CASE_CMP(payloadName, fixPayloadTypeCodecs[n]))
- {
- return true;
- }
- }
- return false;
-}
-
-DTMFDetector::DTMFDetector()
-{
- for(WebRtc_Word16 n = 0; n < 1000; n++)
- {
- _toneCntr[n] = 0;
- }
-}
-
-DTMFDetector::~DTMFDetector()
-{
-}
-
-WebRtc_Word32 DTMFDetector::IncomingDtmf(const WebRtc_UWord8 digitDtmf, const bool /* toneEnded */)
-{
- fprintf(stdout, "%d-",digitDtmf);
- _toneCntr[digitDtmf]++;
- return 0;
-}
-
-void DTMFDetector::PrintDetectedDigits()
-{
- for(WebRtc_Word16 n = 0; n < 1000; n++)
- {
- if(_toneCntr[n] > 0)
- {
- fprintf(stdout, "%d %u msec, \n", n, _toneCntr[n]*10);
- }
- }
- fprintf(stdout, "\n");
- return;
-}
-
-void
-VADCallback::Reset()
-{
- for(int n = 0; n < 6; n++)
- {
- _numFrameTypes[n] = 0;
- }
-}
-
-VADCallback::VADCallback()
-{
- for(int n = 0; n < 6; n++)
- {
- _numFrameTypes[n] = 0;
- }
-}
-
-void
-VADCallback::PrintFrameTypes()
-{
- fprintf(stdout, "No encoding.................. %d\n", _numFrameTypes[0]);
- fprintf(stdout, "Active normal encoded........ %d\n", _numFrameTypes[1]);
- fprintf(stdout, "Passive normal encoded....... %d\n", _numFrameTypes[2]);
- fprintf(stdout, "Passive DTX wideband......... %d\n", _numFrameTypes[3]);
- fprintf(stdout, "Passive DTX narrowband....... %d\n", _numFrameTypes[4]);
- fprintf(stdout, "Passive DTX super-wideband... %d\n", _numFrameTypes[5]);
-}
-
-WebRtc_Word32
-VADCallback::InFrameType(
- WebRtc_Word16 frameType)
-{
- _numFrameTypes[frameType]++;
- return 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_coding/main/test/utility.h b/trunk/src/modules/audio_coding/main/test/utility.h
deleted file mode 100644
index 887c735..0000000
--- a/trunk/src/modules/audio_coding/main/test/utility.h
+++ /dev/null
@@ -1,195 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef ACM_TEST_UTILITY_H
-#define ACM_TEST_UTILITY_H
-
-#include "audio_coding_module.h"
-#include "gtest/gtest.h"
-
-namespace webrtc {
-
-//-----------------------------
-#define CHECK_ERROR(f) \
- do { \
- EXPECT_GE(f, 0) << "Error Calling API"; \
- }while(0)
-
-//-----------------------------
-#define CHECK_PROTECTED(f) \
- do { \
- if(f >= 0) { \
- ADD_FAILURE() << "Error Calling API"; \
- } \
- else { \
- printf("An expected error is caught.\n"); \
- } \
- }while(0)
-
-//----------------------------
-#define CHECK_ERROR_MT(f) \
- do { \
- if(f < 0) { \
- fprintf(stderr, "Error Calling API in file %s at line %d \n", \
- __FILE__, __LINE__); \
- } \
- }while(0)
-
-//----------------------------
-#define CHECK_PROTECTED_MT(f) \
- do { \
- if(f >= 0) { \
- fprintf(stderr, "Error Calling API in file %s at line %d \n", \
- __FILE__, __LINE__); \
- } \
- else { \
- printf("An expected error is caught.\n"); \
- } \
- }while(0)
-
-
-
-#ifdef WIN32
- /* Exclude rarely-used stuff from Windows headers */
- //#define WIN32_LEAN_AND_MEAN
- /* OS-dependent case-insensitive string comparison */
- #define STR_CASE_CMP(x,y) ::_stricmp(x,y)
-#else
- /* OS-dependent case-insensitive string comparison */
- #define STR_CASE_CMP(x,y) ::strcasecmp(x,y)
-#endif
-
-#define DESTROY_ACM(acm) \
- do { \
- if(acm != NULL) { \
- AudioCodingModule::Destroy(acm); \
- acm = NULL; \
- } \
- } while(0)
-
-
-#define DELETE_POINTER(p) \
- do { \
- if(p != NULL) { \
- delete p; \
- p = NULL; \
- } \
- } while(0)
-
-class ACMTestTimer
-{
-public:
- ACMTestTimer();
- ~ACMTestTimer();
-
- void Reset();
- void Tick10ms();
- void Tick1ms();
- void Tick100ms();
- void Tick1sec();
- void CurrentTimeHMS(
- char* currTime);
- void CurrentTime(
- unsigned long& h,
- unsigned char& m,
- unsigned char& s,
- unsigned short& ms);
-
-private:
- void Adjust();
-
- unsigned short _msec;
- unsigned char _sec;
- unsigned char _min;
- unsigned long _hour;
-};
-
-
-
-class CircularBuffer
-{
-public:
- CircularBuffer(WebRtc_UWord32 len);
- ~CircularBuffer();
-
- void SetArithMean(
- bool enable);
- void SetVariance(
- bool enable);
-
- void Update(
- const double newVal);
- void IsBufferFull();
-
- WebRtc_Word16 Variance(double& var);
- WebRtc_Word16 ArithMean(double& mean);
-
-protected:
- double* _buff;
- WebRtc_UWord32 _idx;
- WebRtc_UWord32 _buffLen;
-
- bool _buffIsFull;
- bool _calcAvg;
- bool _calcVar;
- double _sum;
- double _sumSqr;
-};
-
-
-
-
-
-WebRtc_Word16 ChooseCodec(
- CodecInst& codecInst);
-
-void PrintCodecs();
-
-bool FixedPayloadTypeCodec(const char* payloadName);
-
-
-
-
-class DTMFDetector: public AudioCodingFeedback
-{
-public:
- DTMFDetector();
- ~DTMFDetector();
- // used for inband DTMF detection
- WebRtc_Word32 IncomingDtmf(const WebRtc_UWord8 digitDtmf, const bool toneEnded);
- void PrintDetectedDigits();
-
-private:
- WebRtc_UWord32 _toneCntr[1000];
-
-};
-
-
-
-
-class VADCallback : public ACMVADCallback
-{
-public:
- VADCallback();
- ~VADCallback(){}
-
- WebRtc_Word32 InFrameType(
- WebRtc_Word16 frameType);
-
- void PrintFrameTypes();
- void Reset();
-
-private:
- WebRtc_UWord32 _numFrameTypes[6];
-};
-
-} // namespace webrtc
-
-#endif // ACM_TEST_UTILITY_H
diff --git a/trunk/src/modules/audio_coding/neteq/Android.mk b/trunk/src/modules/audio_coding/neteq/Android.mk
deleted file mode 100644
index 84267be..0000000
--- a/trunk/src/modules/audio_coding/neteq/Android.mk
+++ /dev/null
@@ -1,73 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_neteq
-LOCAL_MODULE_TAGS := optional
-LOCAL_SRC_FILES := \
- accelerate.c \
- automode.c \
- bgn_update.c \
- bufstats_decision.c \
- cng_internal.c \
- codec_db.c \
- correlator.c \
- dsp.c \
- dsp_helpfunctions.c \
- dtmf_buffer.c \
- dtmf_tonegen.c \
- expand.c \
- mcu_address_init.c \
- mcu_dsp_common.c \
- mcu_reset.c \
- merge.c \
- min_distortion.c \
- mix_voice_unvoice.c \
- mute_signal.c \
- normal.c \
- packet_buffer.c \
- peak_detection.c \
- preemptive_expand.c \
- random_vector.c \
- recin.c \
- recout.c \
- rtcp.c \
- rtp.c \
- set_fs.c \
- signal_mcu.c \
- split_and_insert.c \
- unmute_signal.c \
- webrtc_neteq.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS) \
- '-DNETEQ_VOICEENGINE_CODECS'
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/interface \
- $(LOCAL_PATH)/../codecs/cng/include \
- $(LOCAL_PATH)/../../.. \
- $(LOCAL_PATH)/../../../common_audio/signal_processing/include
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_coding/neteq/OWNERS b/trunk/src/modules/audio_coding/neteq/OWNERS
deleted file mode 100644
index 1d25542..0000000
--- a/trunk/src/modules/audio_coding/neteq/OWNERS
+++ /dev/null
@@ -1,2 +0,0 @@
-henrik.lundin@webrtc.org
-tina.legrand@webrtc.org
diff --git a/trunk/src/modules/audio_coding/neteq/accelerate.c b/trunk/src/modules/audio_coding/neteq/accelerate.c
deleted file mode 100644
index 285de4d..0000000
--- a/trunk/src/modules/audio_coding/neteq/accelerate.c
+++ /dev/null
@@ -1,489 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the Accelerate algorithm that is used to reduce
- * the delay by removing a part of the audio stream.
- */
-
-#include "dsp.h"
-
-#include "signal_processing_library.h"
-
-#include "dsp_helpfunctions.h"
-#include "neteq_error_codes.h"
-
-#define ACCELERATE_CORR_LEN 50
-#define ACCELERATE_MIN_LAG 10
-#define ACCELERATE_MAX_LAG 60
-#define ACCELERATE_DOWNSAMPLED_LEN (ACCELERATE_CORR_LEN + ACCELERATE_MAX_LAG)
-
-/* Scratch usage:
-
- Type Name size startpos endpos
- WebRtc_Word16 pw16_downSampSpeech 110 0 109
- WebRtc_Word32 pw32_corr 2*50 110 209
- WebRtc_Word16 pw16_corr 50 0 49
-
- Total: 110+2*50
- */
-
-#define SCRATCH_PW16_DS_SPEECH 0
-#define SCRATCH_PW32_CORR ACCELERATE_DOWNSAMPLED_LEN
-#define SCRATCH_PW16_CORR 0
-
-/****************************************************************************
- * WebRtcNetEQ_Accelerate(...)
- *
- * This function tries to shorten the audio data by removing one or several
- * pitch periods. The operation is only carried out if the correlation is
- * strong or if the signal energy is very low.
- *
- * Input:
- * - inst : NetEQ DSP instance
- * - scratchPtr : Pointer to scratch vector.
- * - decoded : Pointer to newly decoded speech.
- * - len : Length of decoded speech.
- * - BGNonly : If non-zero, Accelerate will only remove the last
- * DEFAULT_TIME_ADJUST seconds of the input.
- * No signal matching is done.
- *
- * Output:
- * - inst : Updated instance
- * - outData : Pointer to a memory space where the output data
- * should be stored
- * - pw16_len : Number of samples written to outData.
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-
-int WebRtcNetEQ_Accelerate(DSPInst_t *inst,
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_scratchPtr,
-#endif
- const WebRtc_Word16 *pw16_decoded, int len,
- WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
- WebRtc_Word16 BGNonly)
-{
-
-#ifdef SCRATCH
- /* Use scratch memory for internal temporary vectors */
- WebRtc_Word16 *pw16_downSampSpeech = pw16_scratchPtr + SCRATCH_PW16_DS_SPEECH;
- WebRtc_Word32 *pw32_corr = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_PW32_CORR);
- WebRtc_Word16 *pw16_corr = pw16_scratchPtr + SCRATCH_PW16_CORR;
-#else
- /* Allocate memory for temporary vectors */
- WebRtc_Word16 pw16_downSampSpeech[ACCELERATE_DOWNSAMPLED_LEN];
- WebRtc_Word32 pw32_corr[ACCELERATE_CORR_LEN];
- WebRtc_Word16 pw16_corr[ACCELERATE_CORR_LEN];
-#endif
- WebRtc_Word16 w16_decodedMax = 0;
- WebRtc_Word16 w16_tmp;
- WebRtc_Word16 w16_tmp2;
- WebRtc_Word32 w32_tmp;
- WebRtc_Word32 w32_tmp2;
-
- const WebRtc_Word16 w16_startLag = ACCELERATE_MIN_LAG;
- const WebRtc_Word16 w16_endLag = ACCELERATE_MAX_LAG;
- const WebRtc_Word16 w16_corrLen = ACCELERATE_CORR_LEN;
- const WebRtc_Word16 *pw16_vec1, *pw16_vec2;
- WebRtc_Word16 *pw16_vectmp;
- WebRtc_Word16 w16_inc, w16_startfact;
- WebRtc_Word16 w16_bestIndex, w16_bestVal;
- WebRtc_Word16 w16_VAD = 1;
- WebRtc_Word16 fsMult;
- WebRtc_Word16 fsMult120;
- WebRtc_Word32 w32_en1, w32_en2, w32_cc;
- WebRtc_Word16 w16_en1, w16_en2;
- WebRtc_Word16 w16_en1Scale, w16_en2Scale;
- WebRtc_Word16 w16_sqrtEn1En2;
- WebRtc_Word16 w16_bestCorr = 0;
- int ok;
-
-#ifdef NETEQ_STEREO
- MasterSlaveInfo *msInfo = inst->msInfo;
-#endif
-
- fsMult = WebRtcNetEQ_CalcFsMult(inst->fs); /* Calculate fs/8000 */
-
- /* Pre-calculate common multiplication with fsMult */
- fsMult120 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fsMult, 120); /* 15 ms */
-
- inst->ExpandInst.w16_consecExp = 0; /* Last was not expand any more */
-
- /* Sanity check for len variable; must be (almost) 30 ms
- (120*fsMult + max(bestIndex)) */
- if (len < (WebRtc_Word16) WEBRTC_SPL_MUL_16_16((120 + 119), fsMult))
- {
- /* Length of decoded data too short */
- inst->w16_mode = MODE_UNSUCCESS_ACCELERATE;
- *pw16_len = len;
-
- /* simply move all data from decoded to outData */
- WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
-
- return NETEQ_OTHER_ERROR;
- }
-
- /***********************************/
- /* Special operations for BGN only */
- /***********************************/
-
- /* Check if "background noise only" flag is set */
- if (BGNonly)
- {
- /* special operation for BGN only; simply remove a chunk of data */
- w16_bestIndex = DEFAULT_TIME_ADJUST * WEBRTC_SPL_LSHIFT_W16(fsMult, 3); /* X*fs/1000 */
-
- /* Sanity check for bestIndex */
- if (w16_bestIndex > len)
- { /* not good, do nothing instead */
- inst->w16_mode = MODE_UNSUCCESS_ACCELERATE;
- *pw16_len = len;
-
- /* simply move all data from decoded to outData */
- WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
-
- return NETEQ_OTHER_ERROR;
- }
-
- /* set length parameter */
- *pw16_len = len - w16_bestIndex; /* we remove bestIndex samples */
-
- /* copy to output */
- WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, *pw16_len);
-
- /* set mode */
- inst->w16_mode = MODE_LOWEN_ACCELERATE;
-
- /* update statistics */
- inst->statInst.accelerateLength += w16_bestIndex;
-
- return 0;
- } /* end of special code for BGN mode */
-
-#ifdef NETEQ_STEREO
-
- /* Sanity for msInfo */
- if (msInfo == NULL)
- {
- /* this should not happen here */
- return MASTER_SLAVE_ERROR;
- }
-
- if (msInfo->msMode != NETEQ_SLAVE)
- {
- /* Find correlation lag only for non-slave instances */
-
-#endif
-
- /****************************************************************/
- /* Find the strongest correlation lag by downsampling to 4 kHz, */
- /* calculating correlation for downsampled signal and finding */
- /* the strongest correlation peak. */
- /****************************************************************/
-
- /* find maximum absolute value */
- w16_decodedMax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (WebRtc_Word16) len);
-
- /* downsample the decoded speech to 4 kHz */
- ok = WebRtcNetEQ_DownSampleTo4kHz(pw16_decoded, len, inst->fs, pw16_downSampSpeech,
- ACCELERATE_DOWNSAMPLED_LEN, 1 /* compensate delay*/);
- if (ok != 0)
- {
- /* error */
- inst->w16_mode = MODE_UNSUCCESS_ACCELERATE;
- *pw16_len = len;
- /* simply move all data from decoded to outData */
- WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
- return NETEQ_OTHER_ERROR;
- }
-
- /*
- * Set scaling factor for cross correlation to protect against overflow
- * (log2(50) => 6)
- */
- w16_tmp = 6 - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_decodedMax, w16_decodedMax));
- w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
-
- /* Perform correlation from lag 10 to lag 60 in 4 kHz domain */
- WebRtcNetEQ_CrossCorr(
- pw32_corr, &pw16_downSampSpeech[w16_endLag],
- &pw16_downSampSpeech[w16_endLag - w16_startLag], w16_corrLen,
- (WebRtc_Word16) (w16_endLag - w16_startLag), w16_tmp, -1);
-
- /* Normalize correlation to 14 bits and put in a WebRtc_Word16 vector */
- w32_tmp = WebRtcSpl_MaxAbsValueW32(pw32_corr, w16_corrLen);
- w16_tmp = 17 - WebRtcSpl_NormW32(w32_tmp);
- w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
-
- WebRtcSpl_VectorBitShiftW32ToW16(pw16_corr, w16_corrLen, pw32_corr, w16_tmp);
-
-#ifdef NETEQ_STEREO
- } /* end if (msInfo->msMode != NETEQ_SLAVE) */
-
- if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO))
- {
- /* Find the strongest correlation peak by using the parabolic fit method */
- WebRtcNetEQ_PeakDetection(pw16_corr, (WebRtc_Word16) w16_corrLen, 1, fsMult,
- &w16_bestIndex, &w16_bestVal);
- /* 0 <= bestIndex <= (2*corrLen - 1)*fsMult = 99*fsMult */
-
- /* Compensate bestIndex for displaced starting position */
- w16_bestIndex = w16_bestIndex + w16_startLag * WEBRTC_SPL_LSHIFT_W16(fsMult, 1);
- /* 20*fsMult <= bestIndex <= 119*fsMult */
-
- msInfo->bestIndex = w16_bestIndex;
- }
- else if (msInfo->msMode == NETEQ_SLAVE)
- {
- if (msInfo->extraInfo == ACC_FAIL)
- {
- /* Master has signaled an unsuccessful accelerate */
- w16_bestIndex = 0;
- }
- else
- {
- /* Get best index from master */
- w16_bestIndex = msInfo->bestIndex;
- }
- }
- else
- {
- /* Invalid mode */
- return MASTER_SLAVE_ERROR;
- }
-
-#else /* NETEQ_STEREO */
-
- /* Find the strongest correlation peak by using the parabolic fit method */
- WebRtcNetEQ_PeakDetection(pw16_corr, (WebRtc_Word16) w16_corrLen, 1, fsMult,
- &w16_bestIndex, &w16_bestVal);
- /* 0 <= bestIndex <= (2*corrLen - 1)*fsMult = 99*fsMult */
-
- /* Compensate bestIndex for displaced starting position */
- w16_bestIndex = w16_bestIndex + w16_startLag * WEBRTC_SPL_LSHIFT_W16(fsMult, 1);
- /* 20*fsMult <= bestIndex <= 119*fsMult */
-
-#endif /* NETEQ_STEREO */
-
-#ifdef NETEQ_STEREO
-
- if (msInfo->msMode != NETEQ_SLAVE)
- {
- /* Calculate correlation only for non-slave instances */
-
-#endif /* NETEQ_STEREO */
-
- /*****************************************************/
- /* Calculate correlation bestCorr for the found lag. */
- /* Also do a simple VAD decision. */
- /*****************************************************/
-
- /*
- * Calculate scaling to ensure that bestIndex samples can be square-summed
- * without overflowing
- */
- w16_tmp = (31
- - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_decodedMax, w16_decodedMax)));
- w16_tmp += (31 - WebRtcSpl_NormW32(w16_bestIndex));
- w16_tmp -= 31;
- w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
-
- /* vec1 starts at 15 ms minus one pitch period */
- pw16_vec1 = &pw16_decoded[fsMult120 - w16_bestIndex];
- /* vec2 start at 15 ms */
- pw16_vec2 = &pw16_decoded[fsMult120];
-
- /* Calculate energies for vec1 and vec2 */
- w32_en1 = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec1,
- (WebRtc_Word16*) pw16_vec1, w16_bestIndex, w16_tmp);
- w32_en2 = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec2,
- (WebRtc_Word16*) pw16_vec2, w16_bestIndex, w16_tmp);
-
- /* Calculate cross-correlation at the found lag */
- w32_cc = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec1, (WebRtc_Word16*) pw16_vec2,
- w16_bestIndex, w16_tmp);
-
- /* Check VAD constraint
- ((en1+en2)/(2*bestIndex)) <= 8*inst->BGNInst.energy */
- w32_tmp = WEBRTC_SPL_RSHIFT_W32(w32_en1 + w32_en2, 4); /* (en1+en2)/(2*8) */
- if (inst->BGNInst.w16_initialized == 1)
- {
- w32_tmp2 = inst->BGNInst.w32_energy;
- }
- else
- {
- /* if BGN parameters have not been estimated, use a fixed threshold */
- w32_tmp2 = 75000;
- }
- w16_tmp2 = 16 - WebRtcSpl_NormW32(w32_tmp2);
- w16_tmp2 = WEBRTC_SPL_MAX(0, w16_tmp2);
- w32_tmp = WEBRTC_SPL_RSHIFT_W32(w32_tmp, w16_tmp2);
- w16_tmp2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_tmp2, w16_tmp2);
- w32_tmp2 = WEBRTC_SPL_MUL_16_16(w16_bestIndex, w16_tmp2);
-
- /* Scale w32_tmp properly before comparing with w32_tmp2 */
- /* (w16_tmp is scaling before energy calculation, thus 2*w16_tmp) */
- if (WebRtcSpl_NormW32(w32_tmp) < WEBRTC_SPL_LSHIFT_W32(w16_tmp,1))
- {
- /* Cannot scale only w32_tmp, must scale w32_temp2 too */
- WebRtc_Word16 tempshift = WebRtcSpl_NormW32(w32_tmp);
- w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, tempshift);
- w32_tmp2 = WEBRTC_SPL_RSHIFT_W32(w32_tmp2,
- WEBRTC_SPL_LSHIFT_W32(w16_tmp,1) - tempshift);
- }
- else
- {
- w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp,
- WEBRTC_SPL_LSHIFT_W32(w16_tmp,1));
- }
-
- if (w32_tmp <= w32_tmp2) /*((en1+en2)/(2*bestIndex)) <= 8*inst->BGNInst.energy */
- {
- /* The signal seems to be passive speech */
- w16_VAD = 0;
- w16_bestCorr = 0; /* Correlation does not matter */
- }
- else
- {
- /* The signal is active speech */
- w16_VAD = 1;
-
- /* Calculate correlation (cc/sqrt(en1*en2)) */
-
- /* Start with calculating scale values */
- w16_en1Scale = 16 - WebRtcSpl_NormW32(w32_en1);
- w16_en1Scale = WEBRTC_SPL_MAX(0, w16_en1Scale);
- w16_en2Scale = 16 - WebRtcSpl_NormW32(w32_en2);
- w16_en2Scale = WEBRTC_SPL_MAX(0, w16_en2Scale);
-
- /* Make sure total scaling is even (to simplify scale factor after sqrt) */
- if ((w16_en1Scale + w16_en2Scale) & 1)
- {
- w16_en1Scale += 1;
- }
-
- /* Convert energies to WebRtc_Word16 */
- w16_en1 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en1, w16_en1Scale);
- w16_en2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale);
-
- /* Calculate energy product */
- w32_tmp = WEBRTC_SPL_MUL_16_16(w16_en1, w16_en2);
-
- /* Calculate square-root of energy product */
- w16_sqrtEn1En2 = (WebRtc_Word16) WebRtcSpl_SqrtFloor(w32_tmp);
-
- /* Calculate cc/sqrt(en1*en2) in Q14 */
- w16_tmp = 14 - WEBRTC_SPL_RSHIFT_W16(w16_en1Scale+w16_en2Scale, 1);
- w32_cc = WEBRTC_SPL_SHIFT_W32(w32_cc, w16_tmp);
- w32_cc = WEBRTC_SPL_MAX(0, w32_cc); /* Don't divide with negative number */
- w16_bestCorr = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_cc, w16_sqrtEn1En2);
- w16_bestCorr = WEBRTC_SPL_MIN(16384, w16_bestCorr); /* set maximum to 1.0 */
- }
-
-#ifdef NETEQ_STEREO
-
- } /* end if (msInfo->msMode != NETEQ_SLAVE) */
-
-#endif /* NETEQ_STEREO */
-
- /************************************************/
- /* Check accelerate criteria and remove samples */
- /************************************************/
-
- /* Check for strong correlation (>0.9) or passive speech */
-#ifdef NETEQ_STEREO
- if ((((w16_bestCorr > 14746) || (w16_VAD == 0)) && (msInfo->msMode != NETEQ_SLAVE))
- || ((msInfo->msMode == NETEQ_SLAVE) && (msInfo->extraInfo != ACC_FAIL)))
-#else
- if ((w16_bestCorr > 14746) || (w16_VAD == 0))
-#endif
- {
- /* Do accelerate operation by overlap add */
-
- /*
- * Calculate cross-fading slope so that the fading factor goes from
- * 1 (16384 in Q14) to 0 in one pitch period (bestIndex).
- */
- w16_inc = (WebRtc_Word16) WebRtcSpl_DivW32W16((WebRtc_Word32) 16384,
- (WebRtc_Word16) (w16_bestIndex + 1)); /* in Q14 */
-
- /* Initiate fading factor */
- w16_startfact = 16384 - w16_inc;
-
- /* vec1 starts at 15 ms minus one pitch period */
- pw16_vec1 = &pw16_decoded[fsMult120 - w16_bestIndex];
- /* vec2 start at 15 ms */
- pw16_vec2 = &pw16_decoded[fsMult120];
-
- /* Copy unmodified part [0 to 15 ms minus 1 pitch period] */
- w16_tmp = (fsMult120 - w16_bestIndex);
- WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, w16_tmp);
-
- /* Generate interpolated part of length bestIndex (1 pitch period) */
- pw16_vectmp = pw16_outData + w16_tmp; /* start of interpolation output */
- /* Reuse mixing function from Expand */
- WebRtcNetEQ_MixVoiceUnvoice(pw16_vectmp, (WebRtc_Word16*) pw16_vec1,
- (WebRtc_Word16*) pw16_vec2, &w16_startfact, w16_inc, w16_bestIndex);
-
- /* Move the last part (also unmodified) */
- /* Take from decoded at 15 ms + 1 pitch period */
- pw16_vec2 = &pw16_decoded[fsMult120 + w16_bestIndex];
- WEBRTC_SPL_MEMMOVE_W16(&pw16_outData[fsMult120], pw16_vec2,
- (WebRtc_Word16) (len - fsMult120 - w16_bestIndex));
-
- /* Set the mode flag */
- if (w16_VAD)
- {
- inst->w16_mode = MODE_SUCCESS_ACCELERATE;
- }
- else
- {
- inst->w16_mode = MODE_LOWEN_ACCELERATE;
- }
-
- /* Calculate resulting length = original length - pitch period */
- *pw16_len = len - w16_bestIndex;
-
- /* Update in-call statistics */
- inst->statInst.accelerateLength += w16_bestIndex;
-
- return 0;
- }
- else
- {
- /* Accelerate not allowed */
-
-#ifdef NETEQ_STEREO
- /* Signal to slave(s) that this was unsuccessful */
- if (msInfo->msMode == NETEQ_MASTER)
- {
- msInfo->extraInfo = ACC_FAIL;
- }
-#endif
-
- /* Set mode flag to unsuccessful accelerate */
- inst->w16_mode = MODE_UNSUCCESS_ACCELERATE;
-
- /* Length is unmodified */
- *pw16_len = len;
-
- /* Simply move all data from decoded to outData */
- WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
-
- return 0;
- }
-}
-
-#undef SCRATCH_PW16_DS_SPEECH
-#undef SCRATCH_PW32_CORR
-#undef SCRATCH_PW16_CORR
diff --git a/trunk/src/modules/audio_coding/neteq/automode.c b/trunk/src/modules/audio_coding/neteq/automode.c
deleted file mode 100644
index da127e0..0000000
--- a/trunk/src/modules/audio_coding/neteq/automode.c
+++ /dev/null
@@ -1,736 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the implementation of automatic buffer level optimization.
- */
-
-#include "automode.h"
-
-#include <assert.h>
-
-#include "signal_processing_library.h"
-
-#include "neteq_defines.h"
-
-#ifdef NETEQ_DELAY_LOGGING
-/* special code for offline delay logging */
-#include <stdio.h>
-#include "delay_logging.h"
-
-extern FILE *delay_fid2; /* file pointer to delay log file */
-#endif /* NETEQ_DELAY_LOGGING */
-
-
-int WebRtcNetEQ_UpdateIatStatistics(AutomodeInst_t *inst, int maxBufLen,
- WebRtc_UWord16 seqNumber, WebRtc_UWord32 timeStamp,
- WebRtc_Word32 fsHz, int mdCodec, int streamingMode)
-{
- WebRtc_UWord32 timeIat; /* inter-arrival time */
- int i;
- WebRtc_Word32 tempsum = 0; /* temp summation */
- WebRtc_Word32 tempvar; /* temporary variable */
- int retval = 0; /* return value */
- WebRtc_Word16 packetLenSamp; /* packet speech length in samples */
-
- /****************/
- /* Sanity check */
- /****************/
-
- if (maxBufLen <= 1 || fsHz <= 0)
- {
- /* maxBufLen must be at least 2 and fsHz must both be strictly positive */
- return -1;
- }
-
- /****************************/
- /* Update packet statistics */
- /****************************/
-
- /* Try calculating packet length from current and previous timestamps */
- if ((timeStamp <= inst->lastTimeStamp) || (seqNumber <= inst->lastSeqNo))
- {
- /* Wrong timestamp or sequence order; revert to backup plan */
- packetLenSamp = inst->packetSpeechLenSamp; /* use stored value */
- }
- else if (timeStamp > inst->lastTimeStamp)
- {
- /* calculate timestamps per packet */
- packetLenSamp = (WebRtc_Word16) WebRtcSpl_DivU32U16(timeStamp - inst->lastTimeStamp,
- seqNumber - inst->lastSeqNo);
- }
-
- /* Check that the packet size is positive; if not, the statistics cannot be updated. */
- if (packetLenSamp > 0)
- { /* packet size ok */
-
- /* calculate inter-arrival time in integer packets (rounding down) */
- timeIat = WebRtcSpl_DivW32W16(inst->packetIatCountSamp, packetLenSamp);
-
- /* Special operations for streaming mode */
- if (streamingMode != 0)
- {
- /*
- * Calculate IAT in Q8, including fractions of a packet (i.e., more accurate
- * than timeIat).
- */
- WebRtc_Word16 timeIatQ8 = (WebRtc_Word16) WebRtcSpl_DivW32W16(
- WEBRTC_SPL_LSHIFT_W32(inst->packetIatCountSamp, 8), packetLenSamp);
-
- /*
- * Calculate cumulative sum iat with sequence number compensation (ideal arrival
- * times makes this sum zero).
- */
- inst->cSumIatQ8 += (timeIatQ8
- - WEBRTC_SPL_LSHIFT_W32(seqNumber - inst->lastSeqNo, 8));
-
- /* subtract drift term */
- inst->cSumIatQ8 -= CSUM_IAT_DRIFT;
-
- /* ensure not negative */
- inst->cSumIatQ8 = WEBRTC_SPL_MAX(inst->cSumIatQ8, 0);
-
- /* remember max */
- if (inst->cSumIatQ8 > inst->maxCSumIatQ8)
- {
- inst->maxCSumIatQ8 = inst->cSumIatQ8;
- inst->maxCSumUpdateTimer = 0;
- }
-
- /* too long since the last maximum was observed; decrease max value */
- if (inst->maxCSumUpdateTimer > (WebRtc_UWord32) WEBRTC_SPL_MUL_32_16(fsHz,
- MAX_STREAMING_PEAK_PERIOD))
- {
- inst->maxCSumIatQ8 -= 4; /* remove 1000*4/256 = 15.6 ms/s */
- }
- } /* end of streaming mode */
-
- /* check for discontinuous packet sequence and re-ordering */
- if (seqNumber > inst->lastSeqNo + 1)
- {
- /* Compensate for gap in the sequence numbers.
- * Reduce IAT with expected extra time due to lost packets, but ensure that
- * the IAT is not negative.
- */
- timeIat -= WEBRTC_SPL_MIN(timeIat,
- (WebRtc_UWord32) (seqNumber - inst->lastSeqNo - 1));
- }
- else if (seqNumber < inst->lastSeqNo)
- {
- /* compensate for re-ordering */
- timeIat += (WebRtc_UWord32) (inst->lastSeqNo + 1 - seqNumber);
- }
-
- /* saturate IAT at maximum value */
- timeIat = WEBRTC_SPL_MIN( timeIat, MAX_IAT );
-
- /* update iatProb = forgetting_factor * iatProb for all elements */
- for (i = 0; i <= MAX_IAT; i++)
- {
- WebRtc_Word32 tempHi, tempLo; /* Temporary variables */
-
- /*
- * Multiply iatProbFact (Q15) with iatProb (Q30) and right-shift 15 steps
- * to come back to Q30. The operation is done in two steps:
- */
-
- /*
- * 1) Multiply the high 16 bits (15 bits + sign) of iatProb. Shift iatProb
- * 16 steps right to get the high 16 bits in a WebRtc_Word16 prior to
- * multiplication, and left-shift with 1 afterwards to come back to
- * Q30 = (Q15 * (Q30>>16)) << 1.
- */
- tempHi = WEBRTC_SPL_MUL_16_16(inst->iatProbFact,
- (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(inst->iatProb[i], 16));
- tempHi = WEBRTC_SPL_LSHIFT_W32(tempHi, 1); /* left-shift 1 step */
-
- /*
- * 2) Isolate and multiply the low 16 bits of iatProb. Right-shift 15 steps
- * afterwards to come back to Q30 = (Q15 * Q30) >> 15.
- */
- tempLo = inst->iatProb[i] & 0x0000FFFF; /* sift out the 16 low bits */
- tempLo = WEBRTC_SPL_MUL_16_U16(inst->iatProbFact,
- (WebRtc_UWord16) tempLo);
- tempLo = WEBRTC_SPL_RSHIFT_W32(tempLo, 15);
-
- /* Finally, add the high and low parts */
- inst->iatProb[i] = tempHi + tempLo;
-
- /* Sum all vector elements while we are at it... */
- tempsum += inst->iatProb[i];
- }
-
- /*
- * Increase the probability for the currently observed inter-arrival time
- * with 1 - iatProbFact. The factor is in Q15, iatProb in Q30;
- * hence, left-shift 15 steps to obtain result in Q30.
- */
- inst->iatProb[timeIat] += (32768 - inst->iatProbFact) << 15;
-
- tempsum += (32768 - inst->iatProbFact) << 15; /* add to vector sum */
-
- /*
- * Update iatProbFact (changes only during the first seconds after reset)
- * The factor converges to IAT_PROB_FACT.
- */
- inst->iatProbFact += (IAT_PROB_FACT - inst->iatProbFact + 3) >> 2;
-
- /* iatProb should sum up to 1 (in Q30). */
- tempsum -= 1 << 30; /* should be zero */
-
- /* Check if it does, correct if it doesn't. */
- if (tempsum > 0)
- {
- /* tempsum too large => decrease a few values in the beginning */
- i = 0;
- while (i <= MAX_IAT && tempsum > 0)
- {
- /* Remove iatProb[i] / 16 from iatProb, but not more than tempsum */
- tempvar = WEBRTC_SPL_MIN(tempsum, inst->iatProb[i] >> 4);
- inst->iatProb[i++] -= tempvar;
- tempsum -= tempvar;
- }
- }
- else if (tempsum < 0)
- {
- /* tempsum too small => increase a few values in the beginning */
- i = 0;
- while (i <= MAX_IAT && tempsum < 0)
- {
- /* Add iatProb[i] / 16 to iatProb, but not more than tempsum */
- tempvar = WEBRTC_SPL_MIN(-tempsum, inst->iatProb[i] >> 4);
- inst->iatProb[i++] += tempvar;
- tempsum += tempvar;
- }
- }
-
- /* Calculate optimal buffer level based on updated statistics */
- tempvar = (WebRtc_Word32) WebRtcNetEQ_CalcOptimalBufLvl(inst, fsHz, mdCodec, timeIat,
- streamingMode);
- if (tempvar > 0)
- {
- inst->optBufLevel = (WebRtc_UWord16) tempvar;
-
- if (streamingMode != 0)
- {
- inst->optBufLevel = WEBRTC_SPL_MAX(inst->optBufLevel,
- inst->maxCSumIatQ8);
- }
-
- /*********/
- /* Limit */
- /*********/
-
- /* Subtract extra delay from maxBufLen */
- if (inst->extraDelayMs > 0 && inst->packetSpeechLenSamp > 0)
- {
- maxBufLen -= inst->extraDelayMs / inst->packetSpeechLenSamp * fsHz / 1000;
- maxBufLen = WEBRTC_SPL_MAX(maxBufLen, 1); // sanity: at least one packet
- }
-
- maxBufLen = WEBRTC_SPL_LSHIFT_W32(maxBufLen, 8); /* shift to Q8 */
-
- /* Enforce upper limit; 75% of maxBufLen */
- inst->optBufLevel = (WebRtc_UWord16) WEBRTC_SPL_MIN( inst->optBufLevel,
- (maxBufLen >> 1) + (maxBufLen >> 2) ); /* 1/2 + 1/4 = 75% */
- }
- else
- {
- retval = (int) tempvar;
- }
-
- } /* end if */
-
- /*******************************/
- /* Update post-call statistics */
- /*******************************/
-
- /* Calculate inter-arrival time in ms = packetIatCountSamp / (fsHz / 1000) */
- timeIat = WEBRTC_SPL_UDIV(
- WEBRTC_SPL_UMUL_32_16(inst->packetIatCountSamp, (WebRtc_Word16) 1000),
- (WebRtc_UWord32) fsHz);
-
- /* Increase counter corresponding to current inter-arrival time */
- if (timeIat > 2000)
- {
- inst->countIAT2000ms++;
- }
- else if (timeIat > 1000)
- {
- inst->countIAT1000ms++;
- }
- else if (timeIat > 500)
- {
- inst->countIAT500ms++;
- }
-
- if (timeIat > inst->longestIATms)
- {
- /* update maximum value */
- inst->longestIATms = timeIat;
- }
-
- /***********************************/
- /* Prepare for next packet arrival */
- /***********************************/
-
- inst->packetIatCountSamp = 0; /* reset inter-arrival time counter */
-
- inst->lastSeqNo = seqNumber; /* remember current sequence number */
-
- inst->lastTimeStamp = timeStamp; /* remember current timestamp */
-
- return retval;
-}
-
-
-WebRtc_Word16 WebRtcNetEQ_CalcOptimalBufLvl(AutomodeInst_t *inst, WebRtc_Word32 fsHz,
- int mdCodec, WebRtc_UWord32 timeIatPkts,
- int streamingMode)
-{
-
- WebRtc_Word32 sum1 = 1 << 30; /* assign to 1 in Q30 */
- WebRtc_Word16 B;
- WebRtc_UWord16 Bopt;
- int i;
- WebRtc_Word32 betaInv; /* optimization parameter */
-
-#ifdef NETEQ_DELAY_LOGGING
- /* special code for offline delay logging */
- int temp_var;
-#endif
-
- /****************/
- /* Sanity check */
- /****************/
-
- if (fsHz <= 0)
- {
- /* fsHz must be strictly positive */
- return -1;
- }
-
- /***********************************************/
- /* Get betaInv parameter based on playout mode */
- /***********************************************/
-
- if (streamingMode)
- {
- /* streaming (listen-only) mode */
- betaInv = AUTOMODE_STREAMING_BETA_INV_Q30;
- }
- else
- {
- /* normal mode */
- betaInv = AUTOMODE_BETA_INV_Q30;
- }
-
- /*******************************************************************/
- /* Calculate optimal buffer level without considering jitter peaks */
- /*******************************************************************/
-
- /*
- * Find the B for which the probability of observing an inter-arrival time larger
- * than or equal to B is less than or equal to betaInv.
- */
- B = 0; /* start from the beginning of iatProb */
- sum1 -= inst->iatProb[B]; /* ensure that optimal level is not less than 1 */
-
- do
- {
- /*
- * Subtract the probabilities one by one until the sum is no longer greater
- * than betaInv.
- */
- sum1 -= inst->iatProb[++B];
- }
- while ((sum1 > betaInv) && (B < MAX_IAT));
-
- Bopt = B; /* This is our primary value for the optimal buffer level Bopt */
-
- if (mdCodec)
- {
- /*
- * Use alternative cost function when multiple description codec is in use.
- * Do not have to re-calculate all points, just back off a few steps from
- * previous value of B.
- */
- WebRtc_Word32 sum2 = sum1; /* copy sum1 */
-
- while ((sum2 <= betaInv + inst->iatProb[Bopt]) && (Bopt > 0))
- {
- /* Go backwards in the sum until the modified cost function solution is found */
- sum2 += inst->iatProb[Bopt--];
- }
-
- Bopt++; /* This is the optimal level when using an MD codec */
-
- /* Now, Bopt and B can have different values. */
- }
-
-#ifdef NETEQ_DELAY_LOGGING
- /* special code for offline delay logging */
- temp_var = NETEQ_DELAY_LOGGING_SIGNAL_OPTBUF;
- fwrite( &temp_var, sizeof(int), 1, delay_fid2 );
- temp_var = (int) (Bopt * inst->packetSpeechLenSamp);
-#endif
-
- /******************************************************************/
- /* Make levelFiltFact adaptive: Larger B <=> larger levelFiltFact */
- /******************************************************************/
-
- switch (B)
- {
- case 0:
- case 1:
- {
- inst->levelFiltFact = 251;
- break;
- }
- case 2:
- case 3:
- {
- inst->levelFiltFact = 252;
- break;
- }
- case 4:
- case 5:
- case 6:
- case 7:
- {
- inst->levelFiltFact = 253;
- break;
- }
- default: /* B > 7 */
- {
- inst->levelFiltFact = 254;
- break;
- }
- }
-
- /************************/
- /* Peak mode operations */
- /************************/
-
- /* Compare current IAT with peak threshold
- *
- * If IAT > optimal level + threshold (+1 for MD codecs)
- * or if IAT > 2 * optimal level (note: optimal level is in Q8):
- */
- if (timeIatPkts > (WebRtc_UWord32) (Bopt + inst->peakThresholdPkt + (mdCodec != 0))
- || timeIatPkts > (WebRtc_UWord32) WEBRTC_SPL_LSHIFT_U16(Bopt, 1))
- {
- /* A peak is observed */
-
- if (inst->peakIndex == -1)
- {
- /* this is the first peak; prepare for next peak */
- inst->peakIndex = 0;
- /* set the mode-disable counter */
- inst->peakModeDisabled = WEBRTC_SPL_LSHIFT_W16(1, NUM_PEAKS_REQUIRED-2);
- }
- else if (inst->peakIatCountSamp
- <=
- (WebRtc_UWord32) WEBRTC_SPL_MUL_32_16(fsHz, MAX_PEAK_PERIOD))
- {
- /* This is not the first peak and the period time is valid */
-
- /* store time elapsed since last peak */
- inst->peakPeriodSamp[inst->peakIndex] = inst->peakIatCountSamp;
-
- /* saturate height to 16 bits */
- inst->peakHeightPkt[inst->peakIndex]
- =
- (WebRtc_Word16) WEBRTC_SPL_MIN(timeIatPkts, WEBRTC_SPL_WORD16_MAX);
-
- /* increment peakIndex and wrap/modulo */
- inst->peakIndex = (inst->peakIndex + 1) & PEAK_INDEX_MASK;
-
- /* process peak vectors */
- inst->curPeakHeight = 0;
- inst->curPeakPeriod = 0;
-
- for (i = 0; i < NUM_PEAKS; i++)
- {
- /* Find maximum of peak heights and peak periods */
- inst->curPeakHeight
- = WEBRTC_SPL_MAX(inst->curPeakHeight, inst->peakHeightPkt[i]);
- inst->curPeakPeriod
- = WEBRTC_SPL_MAX(inst->curPeakPeriod, inst->peakPeriodSamp[i]);
-
- }
-
- inst->peakModeDisabled >>= 1; /* decrease mode-disable "counter" */
-
- }
- else if (inst->peakIatCountSamp > (WebRtc_UWord32) WEBRTC_SPL_MUL_32_16(fsHz,
- WEBRTC_SPL_LSHIFT_W16(MAX_PEAK_PERIOD, 1)))
- {
- /*
- * More than 2 * MAX_PEAK_PERIOD has elapsed since last peak;
- * too long time => reset peak statistics
- */
- inst->curPeakHeight = 0;
- inst->curPeakPeriod = 0;
- for (i = 0; i < NUM_PEAKS; i++)
- {
- inst->peakHeightPkt[i] = 0;
- inst->peakPeriodSamp[i] = 0;
- }
-
- inst->peakIndex = -1; /* Next peak is first peak */
- inst->peakIatCountSamp = 0;
- }
-
- inst->peakIatCountSamp = 0; /* Reset peak interval timer */
- } /* end if peak is observed */
-
- /* Evaluate peak mode conditions */
-
- /*
- * If not disabled (enough peaks have been observed) and
- * time since last peak is less than two peak periods.
- */
- inst->peakFound = 0;
- if ((!inst->peakModeDisabled) && (inst->peakIatCountSamp
- <= WEBRTC_SPL_LSHIFT_W32(inst->curPeakPeriod , 1)))
- {
- /* Engage peak mode */
- inst->peakFound = 1;
- /* Set optimal buffer level to curPeakHeight (if it's not already larger) */
- Bopt = WEBRTC_SPL_MAX(Bopt, inst->curPeakHeight);
-
-#ifdef NETEQ_DELAY_LOGGING
- /* special code for offline delay logging */
- temp_var = (int) -(Bopt * inst->packetSpeechLenSamp);
-#endif
- }
-
- /* Scale Bopt to Q8 */
- Bopt = WEBRTC_SPL_LSHIFT_U16(Bopt,8);
-
-#ifdef NETEQ_DELAY_LOGGING
- /* special code for offline delay logging */
- fwrite( &temp_var, sizeof(int), 1, delay_fid2 );
-#endif
-
- /* Sanity check: Bopt must be strictly positive */
- if (Bopt <= 0)
- {
- Bopt = WEBRTC_SPL_LSHIFT_W16(1, 8); /* 1 in Q8 */
- }
-
- return Bopt; /* return value in Q8 */
-}
-
-
-int WebRtcNetEQ_BufferLevelFilter(WebRtc_Word32 curSizeMs8, AutomodeInst_t *inst,
- int sampPerCall, WebRtc_Word16 fsMult)
-{
-
- WebRtc_Word16 curSizeFrames;
-
- /****************/
- /* Sanity check */
- /****************/
-
- if (sampPerCall <= 0 || fsMult <= 0)
- {
- /* sampPerCall and fsMult must both be strictly positive */
- return -1;
- }
-
- /* Check if packet size has been detected */
- if (inst->packetSpeechLenSamp > 0)
- {
- /*
- * Current buffer level in packet lengths
- * = (curSizeMs8 * fsMult) / packetSpeechLenSamp
- */
- curSizeFrames = (WebRtc_Word16) WebRtcSpl_DivW32W16(
- WEBRTC_SPL_MUL_32_16(curSizeMs8, fsMult), inst->packetSpeechLenSamp);
- }
- else
- {
- curSizeFrames = 0;
- }
-
- /* Filter buffer level */
- if (inst->levelFiltFact > 0) /* check that filter factor is set */
- {
- /* Filter:
- * buffLevelFilt = levelFiltFact * buffLevelFilt
- * + (1-levelFiltFact) * curSizeFrames
- *
- * levelFiltFact is in Q8
- */
- inst->buffLevelFilt = (WebRtc_UWord16) (WEBRTC_SPL_RSHIFT_W32(
- WEBRTC_SPL_MUL_16_U16(inst->levelFiltFact, inst->buffLevelFilt), 8)
- + WEBRTC_SPL_MUL_16_16(256 - inst->levelFiltFact, curSizeFrames));
- }
-
- /* Account for time-scale operations (accelerate and pre-emptive expand) */
- if (inst->prevTimeScale)
- {
- /*
- * Time-scaling has been performed since last filter update.
- * Subtract the sampleMemory from buffLevelFilt after converting sampleMemory
- * from samples to packets in Q8. Make sure that the filtered value is
- * non-negative.
- */
- inst->buffLevelFilt = (WebRtc_UWord16) WEBRTC_SPL_MAX( inst->buffLevelFilt -
- WebRtcSpl_DivW32W16(
- WEBRTC_SPL_LSHIFT_W32(inst->sampleMemory, 8), /* sampleMemory in Q8 */
- inst->packetSpeechLenSamp ), /* divide by packetSpeechLenSamp */
- 0);
-
- /*
- * Reset flag and set timescaleHoldOff timer to prevent further time-scaling
- * for some time.
- */
- inst->prevTimeScale = 0;
- inst->timescaleHoldOff = AUTOMODE_TIMESCALE_LIMIT;
- }
-
- /* Update time counters and HoldOff timer */
- inst->packetIatCountSamp += sampPerCall; /* packet inter-arrival time */
- inst->peakIatCountSamp += sampPerCall; /* peak inter-arrival time */
- inst->timescaleHoldOff >>= 1; /* time-scaling limiter */
- inst->maxCSumUpdateTimer += sampPerCall; /* cumulative-sum timer */
-
- return 0;
-
-}
-
-
-int WebRtcNetEQ_SetPacketSpeechLen(AutomodeInst_t *inst, WebRtc_Word16 newLenSamp,
- WebRtc_Word32 fsHz)
-{
-
- /* Sanity check for newLenSamp and fsHz */
- if (newLenSamp <= 0 || fsHz <= 0)
- {
- return -1;
- }
-
- inst->packetSpeechLenSamp = newLenSamp; /* Store packet size in instance */
-
- /* Make NetEQ wait for first regular packet before starting the timer */
- inst->lastPackCNGorDTMF = 1;
-
- inst->packetIatCountSamp = 0; /* Reset packet time counter */
-
- /*
- * Calculate peak threshold from packet size. The threshold is defined as
- * the (fractional) number of packets that corresponds to PEAK_HEIGHT
- * (in Q8 seconds). That is, threshold = PEAK_HEIGHT/256 * fsHz / packLen.
- */
- inst->peakThresholdPkt = (WebRtc_UWord16) WebRtcSpl_DivW32W16ResW16(
- WEBRTC_SPL_MUL_16_16_RSFT(PEAK_HEIGHT,
- (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(fsHz, 6), 2), inst->packetSpeechLenSamp);
-
- return 0;
-}
-
-
-int WebRtcNetEQ_ResetAutomode(AutomodeInst_t *inst, int maxBufLenPackets)
-{
-
- int i;
- WebRtc_UWord16 tempprob = 0x4002; /* 16384 + 2 = 100000000000010 binary; */
-
- /* Sanity check for maxBufLenPackets */
- if (maxBufLenPackets <= 1)
- {
- /* Invalid value; set to 10 instead (arbitary small number) */
- maxBufLenPackets = 10;
- }
-
- /* Reset filtered buffer level */
- inst->buffLevelFilt = 0;
-
- /* Reset packet size to unknown */
- inst->packetSpeechLenSamp = 0;
-
- /*
- * Flag that last packet was special payload, so that automode will treat the next speech
- * payload as the first payload received.
- */
- inst->lastPackCNGorDTMF = 1;
-
- /* Reset peak detection parameters */
- inst->peakModeDisabled = 1; /* disable peak mode */
- inst->peakIatCountSamp = 0;
- inst->peakIndex = -1; /* indicates that no peak is registered */
- inst->curPeakHeight = 0;
- inst->curPeakPeriod = 0;
- for (i = 0; i < NUM_PEAKS; i++)
- {
- inst->peakHeightPkt[i] = 0;
- inst->peakPeriodSamp[i] = 0;
- }
-
- /*
- * Set the iatProb PDF vector to an exponentially decaying distribution
- * iatProb[i] = 0.5^(i+1), i = 0, 1, 2, ...
- * iatProb is in Q30.
- */
- for (i = 0; i <= MAX_IAT; i++)
- {
- /* iatProb[i] = 0.5^(i+1) = iatProb[i-1] / 2 */
- tempprob = WEBRTC_SPL_RSHIFT_U16(tempprob, 1);
- /* store in PDF vector */
- inst->iatProb[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) tempprob, 16);
- }
-
- /*
- * Calculate the optimal buffer level corresponding to the initial PDF.
- * No need to call WebRtcNetEQ_CalcOptimalBufLvl() since we have just hard-coded
- * all the variables that the buffer level depends on => we know the result
- */
- inst->optBufLevel = WEBRTC_SPL_MIN(4,
- (maxBufLenPackets >> 1) + (maxBufLenPackets >> 1)); /* 75% of maxBufLenPackets */
- inst->levelFiltFact = 253;
-
- /*
- * Reset the iat update forgetting factor to 0 to make the impact of the first
- * incoming packets greater.
- */
- inst->iatProbFact = 0;
-
- /* Reset packet inter-arrival time counter */
- inst->packetIatCountSamp = 0;
-
- /* Clear time-scaling related variables */
- inst->prevTimeScale = 0;
- inst->timescaleHoldOff = AUTOMODE_TIMESCALE_LIMIT; /* don't allow time-scaling immediately */
-
- inst->cSumIatQ8 = 0;
- inst->maxCSumIatQ8 = 0;
-
- return 0;
-}
-
-int32_t WebRtcNetEQ_AverageIAT(const AutomodeInst_t *inst) {
- int i;
- int32_t sum_q24 = 0;
- assert(inst);
- for (i = 0; i <= MAX_IAT; ++i) {
- /* Shift 6 to fit worst case: 2^30 * 64. */
- sum_q24 += (inst->iatProb[i] >> 6) * i;
- }
- /* Subtract the nominal inter-arrival time 1 = 2^24 in Q24. */
- sum_q24 -= (1 << 24);
- /*
- * Multiply with 1000000 / 2^24 = 15625 / 2^18 to get in parts-per-million.
- * Shift 7 to Q17 first, then multiply with 15625 and shift another 11.
- */
- return ((sum_q24 >> 7) * 15625) >> 11;
-}
diff --git a/trunk/src/modules/audio_coding/neteq/automode.h b/trunk/src/modules/audio_coding/neteq/automode.h
deleted file mode 100644
index dbd09cf..0000000
--- a/trunk/src/modules/audio_coding/neteq/automode.h
+++ /dev/null
@@ -1,264 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the functionality for automatic buffer level optimization.
- */
-
-#ifndef AUTOMODE_H
-#define AUTOMODE_H
-
-#include "typedefs.h"
-
-/*************/
-/* Constants */
-/*************/
-
-/* The beta parameter defines the trade-off between delay and underrun probability. */
-/* It is defined through its inverse in Q30 */
-#define AUTOMODE_BETA_INV_Q30 53687091 /* 1/20 in Q30 */
-#define AUTOMODE_STREAMING_BETA_INV_Q30 536871 /* 1/2000 in Q30 */
-
-/* Forgetting factor for the inter-arrival time statistics */
-#define IAT_PROB_FACT 32745 /* 0.9993 in Q15 */
-
-/* Maximum inter-arrival time to register (in "packet-times") */
-#define MAX_IAT 64
-#define PEAK_HEIGHT 20 /* 0.08s in Q8 */
-
-/* The value (1<<5) sets maximum accelerate "speed" to about 100 ms/s */
-#define AUTOMODE_TIMESCALE_LIMIT (1<<5)
-
-/* Peak mode related parameters */
-/* Number of peaks in peak vector; must be a power of 2 */
-#define NUM_PEAKS 8
-
-/* Must be NUM_PEAKS-1 */
-#define PEAK_INDEX_MASK 0x0007
-
-/* Longest accepted peak distance */
-#define MAX_PEAK_PERIOD 10
-#define MAX_STREAMING_PEAK_PERIOD 600 /* 10 minutes */
-
-/* Number of peaks required before peak mode can be engaged */
-#define NUM_PEAKS_REQUIRED 3
-
-/* Drift term for cumulative sum */
-#define CSUM_IAT_DRIFT 2
-
-/*******************/
-/* Automode struct */
-/*******************/
-
-/* The automode struct is a sub-struct of the
- bufstats-struct (BufstatsInst_t). */
-
-typedef struct
-{
-
- /* Filtered current buffer level */
- WebRtc_UWord16 levelFiltFact; /* filter forgetting factor in Q8 */
- WebRtc_UWord16 buffLevelFilt; /* filtered buffer level in Q8 */
-
- /* Inter-arrival time (iat) statistics */
- WebRtc_Word32 iatProb[MAX_IAT + 1]; /* iat probabilities in Q30 */
- WebRtc_Word16 iatProbFact; /* iat forgetting factor in Q15 */
- WebRtc_UWord32 packetIatCountSamp; /* time (in timestamps) elapsed since last
- packet arrival, based on RecOut calls */
- WebRtc_UWord16 optBufLevel; /* current optimal buffer level in Q8 */
-
- /* Packet related information */
- WebRtc_Word16 packetSpeechLenSamp; /* speech samples per incoming packet */
- WebRtc_Word16 lastPackCNGorDTMF; /* indicates that the last received packet
- contained special information */
- WebRtc_UWord16 lastSeqNo; /* sequence number for last packet received */
- WebRtc_UWord32 lastTimeStamp; /* timestamp for the last packet received */
- WebRtc_Word32 sampleMemory; /* memory position for keeping track of how many
- samples we cut during expand */
- WebRtc_Word16 prevTimeScale; /* indicates that the last mode was an accelerate
- or pre-emptive expand operation */
- WebRtc_UWord32 timescaleHoldOff; /* counter that is shifted one step right each
- RecOut call; time-scaling allowed when it has
- reached 0 */
- WebRtc_Word16 extraDelayMs; /* extra delay for sync with video */
-
- /* Peak-detection */
- /* vector with the latest peak periods (peak spacing in samples) */
- WebRtc_UWord32 peakPeriodSamp[NUM_PEAKS];
- /* vector with the latest peak heights (in packets) */
- WebRtc_Word16 peakHeightPkt[NUM_PEAKS];
- WebRtc_Word16 peakIndex; /* index for the vectors peakPeriodSamp and peakHeightPkt;
- -1 if still waiting for first peak */
- WebRtc_UWord16 peakThresholdPkt; /* definition of peak (in packets);
- calculated from PEAK_HEIGHT */
- WebRtc_UWord32 peakIatCountSamp; /* samples elapsed since last peak was observed */
- WebRtc_UWord32 curPeakPeriod; /* current maximum of peakPeriodSamp vector */
- WebRtc_Word16 curPeakHeight; /* derived from peakHeightPkt vector;
- used as optimal buffer level in peak mode */
- WebRtc_Word16 peakModeDisabled; /* ==0 if peak mode can be engaged; >0 if not */
- uint16_t peakFound; /* 1 if peaks are detected and extra delay is applied;
- * 0 otherwise. */
-
- /* Post-call statistics */
- WebRtc_UWord32 countIAT500ms; /* number of times we got small network outage */
- WebRtc_UWord32 countIAT1000ms; /* number of times we got medium network outage */
- WebRtc_UWord32 countIAT2000ms; /* number of times we got large network outage */
- WebRtc_UWord32 longestIATms; /* mSec duration of longest network outage */
-
- WebRtc_Word16 cSumIatQ8; /* cumulative sum of inter-arrival times */
- WebRtc_Word16 maxCSumIatQ8; /* max cumulative sum IAT */
- WebRtc_UWord32 maxCSumUpdateTimer;/* time elapsed since maximum was observed */
-
-} AutomodeInst_t;
-
-/*************/
-/* Functions */
-/*************/
-
-/****************************************************************************
- * WebRtcNetEQ_UpdateIatStatistics(...)
- *
- * Update the packet inter-arrival time statistics when a new packet arrives.
- * This function should be called for every arriving packet, with some
- * exceptions when using DTX/VAD and DTMF. A new optimal buffer level is
- * calculated after the update.
- *
- * Input:
- * - inst : Automode instance
- * - maxBufLen : Maximum number of packets the buffer can hold
- * - seqNumber : RTP sequence number of incoming packet
- * - timeStamp : RTP timestamp of incoming packet
- * - fsHz : Sample rate in Hz
- * - mdCodec : Non-zero if the current codec is a multiple-
- * description codec
- * - streamingMode : A non-zero value will increase jitter robustness (and delay)
- *
- * Output:
- * - inst : Updated automode instance
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-
-int WebRtcNetEQ_UpdateIatStatistics(AutomodeInst_t *inst, int maxBufLen,
- WebRtc_UWord16 seqNumber, WebRtc_UWord32 timeStamp,
- WebRtc_Word32 fsHz, int mdCodec, int streamingMode);
-
-/****************************************************************************
- * WebRtcNetEQ_CalcOptimalBufLvl(...)
- *
- * Calculate the optimal buffer level based on packet inter-arrival time
- * statistics.
- *
- * Input:
- * - inst : Automode instance
- * - fsHz : Sample rate in Hz
- * - mdCodec : Non-zero if the current codec is a multiple-
- * description codec
- * - timeIatPkts : Currently observed inter-arrival time in packets
- * - streamingMode : A non-zero value will increase jitter robustness (and delay)
- *
- * Output:
- * - inst : Updated automode instance
- *
- * Return value : >0 - Optimal buffer level
- * <0 - Error
- */
-
-WebRtc_Word16 WebRtcNetEQ_CalcOptimalBufLvl(AutomodeInst_t *inst, WebRtc_Word32 fsHz,
- int mdCodec, WebRtc_UWord32 timeIatPkts,
- int streamingMode);
-
-/****************************************************************************
- * WebRtcNetEQ_BufferLevelFilter(...)
- *
- * Update filtered buffer level. The function must be called once for each
- * RecOut call, since the timing of automode hinges on counters that are
- * updated by this function.
- *
- * Input:
- * - curSizeMs8 : Total length of unused speech data in packet buffer
- * and sync buffer, in ms * 8
- * - inst : Automode instance
- * - sampPerCall : Number of samples per RecOut call
- * - fsMult : Sample rate in Hz divided by 8000
- *
- * Output:
- * - inst : Updated automode instance
- *
- * Return value : 0 - Ok
- * : <0 - Error
- */
-
-int WebRtcNetEQ_BufferLevelFilter(WebRtc_Word32 curSizeMs8, AutomodeInst_t *inst,
- int sampPerCall, WebRtc_Word16 fsMult);
-
-/****************************************************************************
- * WebRtcNetEQ_SetPacketSpeechLen(...)
- *
- * Provide the number of speech samples extracted from a packet to the
- * automode instance. Several of the calculations within automode depend
- * on knowing the packet size.
- *
- *
- * Input:
- * - inst : Automode instance
- * - newLenSamp : Number of samples per RecOut call
- * - fsHz : Sample rate in Hz
- *
- * Output:
- * - inst : Updated automode instance
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-
-int WebRtcNetEQ_SetPacketSpeechLen(AutomodeInst_t *inst, WebRtc_Word16 newLenSamp,
- WebRtc_Word32 fsHz);
-
-/****************************************************************************
- * WebRtcNetEQ_ResetAutomode(...)
- *
- * Reset the automode instance.
- *
- *
- * Input:
- * - inst : Automode instance
- * - maxBufLenPackets : Maximum number of packets that the packet
- * buffer can hold (>1)
- *
- * Output:
- * - inst : Updated automode instance
- *
- * Return value : 0 - Ok
- */
-
-int WebRtcNetEQ_ResetAutomode(AutomodeInst_t *inst, int maxBufLenPackets);
-
-/****************************************************************************
- * WebRtcNetEQ_AverageIAT(...)
- *
- * Calculate the average inter-arrival time based on current statistics.
- * The average is expressed in parts per million relative the nominal. That is,
- * if the average inter-arrival time is equal to the nominal frame time,
- * the return value is zero. A positive value corresponds to packet spacing
- * being too large, while a negative value means that the packets arrive with
- * less spacing than expected.
- *
- *
- * Input:
- * - inst : Automode instance.
- *
- * Return value : Average relative inter-arrival time in samples.
- */
-
-int32_t WebRtcNetEQ_AverageIAT(const AutomodeInst_t *inst);
-
-#endif /* AUTOMODE_H */
diff --git a/trunk/src/modules/audio_coding/neteq/bgn_update.c b/trunk/src/modules/audio_coding/neteq/bgn_update.c
deleted file mode 100644
index 05956c2..0000000
--- a/trunk/src/modules/audio_coding/neteq/bgn_update.c
+++ /dev/null
@@ -1,247 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the function for updating the background noise estimate.
- */
-
-#include "dsp.h"
-
-#include "signal_processing_library.h"
-
-#include "dsp_helpfunctions.h"
-
-/* Scratch usage:
- Designed for BGN_LPC_ORDER <= 10
-
- Type Name size startpos endpos
- WebRtc_Word32 pw32_autoCorr 22 0 21 (Length (BGN_LPC_ORDER + 1)*2)
- WebRtc_Word16 pw16_tempVec 10 22 31 (Length BGN_LPC_ORDER)
- WebRtc_Word16 pw16_rc 10 32 41 (Length BGN_LPC_ORDER)
- WebRtc_Word16 pw16_outVec 74 0 73 (Length BGN_LPC_ORDER + 64)
-
- Total: 74
- */
-
-#if (BGN_LPC_ORDER > 10) && (defined SCRATCH)
-#error BGN_LPC_ORDER is too large for current scratch memory allocation
-#endif
-
-#define SCRATCH_PW32_AUTO_CORR 0
-#define SCRATCH_PW16_TEMP_VEC 22
-#define SCRATCH_PW16_RC 32
-#define SCRATCH_PW16_OUT_VEC 0
-
-#define NETEQFIX_BGNFRAQINCQ16 229 /* 0.0035 in Q16 */
-
-/****************************************************************************
- * WebRtcNetEQ_BGNUpdate(...)
- *
- * This function updates the background noise parameter estimates.
- *
- * Input:
- * - inst : NetEQ instance, where the speech history is stored.
- * - scratchPtr : Pointer to scratch vector.
- *
- * Output:
- * - inst : Updated information about the BGN characteristics.
- *
- * Return value : No return value
- */
-
-void WebRtcNetEQ_BGNUpdate(
-#ifdef SCRATCH
- DSPInst_t *inst, WebRtc_Word16 *pw16_scratchPtr
-#else
- DSPInst_t *inst
-#endif
-)
-{
- const WebRtc_Word16 w16_vecLen = 256;
- BGNInst_t *BGN_Inst = &(inst->BGNInst);
-#ifdef SCRATCH
- WebRtc_Word32 *pw32_autoCorr = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_PW32_AUTO_CORR);
- WebRtc_Word16 *pw16_tempVec = pw16_scratchPtr + SCRATCH_PW16_TEMP_VEC;
- WebRtc_Word16 *pw16_rc = pw16_scratchPtr + SCRATCH_PW16_RC;
- WebRtc_Word16 *pw16_outVec = pw16_scratchPtr + SCRATCH_PW16_OUT_VEC;
-#else
- WebRtc_Word32 pw32_autoCorr[BGN_LPC_ORDER + 1];
- WebRtc_Word16 pw16_tempVec[BGN_LPC_ORDER];
- WebRtc_Word16 pw16_outVec[BGN_LPC_ORDER + 64];
- WebRtc_Word16 pw16_rc[BGN_LPC_ORDER];
-#endif
- WebRtc_Word16 pw16_A[BGN_LPC_ORDER + 1];
- WebRtc_Word32 w32_tmp;
- WebRtc_Word16 *pw16_vec;
- WebRtc_Word16 w16_maxSample;
- WebRtc_Word16 w16_tmp, w16_tmp2;
- WebRtc_Word16 w16_enSampleShift;
- WebRtc_Word32 w32_en, w32_enBGN;
- WebRtc_Word32 w32_enUpdateThreashold;
- WebRtc_Word16 stability;
-
- pw16_vec = inst->pw16_speechHistory + inst->w16_speechHistoryLen - w16_vecLen;
-
-#ifdef NETEQ_VAD
- if( !inst->VADInst.VADEnabled /* we are not using post-decode VAD */
- || inst->VADInst.VADDecision == 0 )
- { /* ... or, post-decode VAD says passive speaker */
-#endif /* NETEQ_VAD */
-
- /*Insert zeros to guarantee that boundary values do not distort autocorrelation */
- WEBRTC_SPL_MEMCPY_W16(pw16_tempVec, pw16_vec - BGN_LPC_ORDER, BGN_LPC_ORDER);
- WebRtcSpl_MemSetW16(pw16_vec - BGN_LPC_ORDER, 0, BGN_LPC_ORDER);
-
- w16_maxSample = WebRtcSpl_MaxAbsValueW16(pw16_vec, w16_vecLen);
- w16_tmp = 8 /* log2(w16_veclen) = 8 */
- - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_maxSample, w16_maxSample));
- w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
-
- WebRtcNetEQ_CrossCorr(pw32_autoCorr, pw16_vec, pw16_vec, w16_vecLen, BGN_LPC_ORDER + 1,
- w16_tmp, -1);
-
- /* Copy back data */
- WEBRTC_SPL_MEMCPY_W16(pw16_vec - BGN_LPC_ORDER, pw16_tempVec, BGN_LPC_ORDER);
-
- w16_enSampleShift = 8 - w16_tmp; /* Number of shifts to get energy/sample */
- /* pw32_autoCorr[0]>>w16_enSampleShift */
- w32_en = WEBRTC_SPL_RSHIFT_W32(pw32_autoCorr[0], w16_enSampleShift);
- if ((w32_en < BGN_Inst->w32_energyUpdate
-#ifdef NETEQ_VAD
- /* post-decode VAD disabled and w32_en sufficiently low */
- && !inst->VADInst.VADEnabled)
- /* ... or, post-decode VAD says passive speaker */
- || (inst->VADInst.VADEnabled && inst->VADInst.VADDecision == 0)
-#else
- ) /* just close the extra parenthesis */
-#endif /* NETEQ_VAD */
- )
- {
- /* Generate LPC coefficients */
- if (pw32_autoCorr[0] > 0)
- {
- /* regardless of whether the filter is actually updated or not,
- update energy threshold levels, since we have in fact observed
- a low energy signal */
- if (w32_en < BGN_Inst->w32_energyUpdate)
- {
- /* Never get under 1.0 in average sample energy */
- BGN_Inst->w32_energyUpdate = WEBRTC_SPL_MAX(w32_en, 1);
- BGN_Inst->w32_energyUpdateLow = 0;
- }
-
- stability = WebRtcSpl_LevinsonDurbin(pw32_autoCorr, pw16_A, pw16_rc, BGN_LPC_ORDER);
- /* Only update BGN if filter is stable */
- if (stability != 1)
- {
- return;
- }
- }
- else
- {
- /* Do not update */
- return;
- }
- /* Generate the CNG gain factor by looking at the energy of the residual */
- WebRtcSpl_FilterMAFastQ12(pw16_vec + w16_vecLen - 64, pw16_outVec, pw16_A,
- BGN_LPC_ORDER + 1, 64);
- w32_enBGN = WebRtcNetEQ_DotW16W16(pw16_outVec, pw16_outVec, 64, 0);
- /* Dot product should never overflow since it is BGN and residual! */
-
- /*
- * Check spectral flatness
- * Comparing the residual variance with the input signal variance tells
- * if the spectrum is flat or not.
- * (20*w32_enBGN) >= (w32_en<<6)
- * Also ensure that the energy is non-zero.
- */
- if ((WEBRTC_SPL_MUL_32_16(w32_enBGN, 20) >= WEBRTC_SPL_LSHIFT_W32(w32_en, 6))
- && (w32_en > 0))
- {
- /* spectrum is flat enough; save filter parameters */
-
- WEBRTC_SPL_MEMCPY_W16(BGN_Inst->pw16_filter, pw16_A, BGN_LPC_ORDER+1);
- WEBRTC_SPL_MEMCPY_W16(BGN_Inst->pw16_filterState,
- pw16_vec + w16_vecLen - BGN_LPC_ORDER, BGN_LPC_ORDER);
-
- /* Save energy level */
- BGN_Inst->w32_energy = WEBRTC_SPL_MAX(w32_en, 1);
-
- /* Update energy threshold levels */
- /* Never get under 1.0 in average sample energy */
- BGN_Inst->w32_energyUpdate = WEBRTC_SPL_MAX(w32_en, 1);
- BGN_Inst->w32_energyUpdateLow = 0;
-
- /* Normalize w32_enBGN to 29 or 30 bits before sqrt */
- w16_tmp2 = WebRtcSpl_NormW32(w32_enBGN) - 1;
- if (w16_tmp2 & 0x1)
- {
- w16_tmp2 -= 1; /* Even number of shifts required */
- }
- w32_enBGN = WEBRTC_SPL_SHIFT_W32(w32_enBGN, w16_tmp2);
-
- /* Calculate scale and shift factor */
- BGN_Inst->w16_scale = (WebRtc_Word16) WebRtcSpl_SqrtFloor(w32_enBGN);
- BGN_Inst->w16_scaleShift = 13 + ((6 + w16_tmp2) >> 1); /* RANDN table is in Q13, */
- /* 6=log2(64) */
-
- BGN_Inst->w16_initialized = 1;
- }
-
- }
- else
- {
- /*
- * Will only happen if post-decode VAD is disabled and w32_en is not low enough.
- * Increase the threshold for update so that it increases by a factor 4 in four
- * seconds.
- * energy = energy * 1.0035
- */
- w32_tmp = WEBRTC_SPL_MUL_16_16_RSFT(NETEQFIX_BGNFRAQINCQ16,
- BGN_Inst->w32_energyUpdateLow, 16);
- w32_tmp += WEBRTC_SPL_MUL_16_16(NETEQFIX_BGNFRAQINCQ16,
- (WebRtc_Word16)(BGN_Inst->w32_energyUpdate & 0xFF));
- w32_tmp += (WEBRTC_SPL_MUL_16_16(NETEQFIX_BGNFRAQINCQ16,
- (WebRtc_Word16)((BGN_Inst->w32_energyUpdate>>8) & 0xFF)) << 8);
- BGN_Inst->w32_energyUpdateLow += w32_tmp;
-
- BGN_Inst->w32_energyUpdate += WEBRTC_SPL_MUL_16_16(NETEQFIX_BGNFRAQINCQ16,
- (WebRtc_Word16)(BGN_Inst->w32_energyUpdate>>16));
- BGN_Inst->w32_energyUpdate += BGN_Inst->w32_energyUpdateLow >> 16;
- BGN_Inst->w32_energyUpdateLow = (BGN_Inst->w32_energyUpdateLow & 0x0FFFF);
-
- /* Update maximum energy */
- /* Decrease by a factor 1/1024 each time */
- BGN_Inst->w32_energyMax = BGN_Inst->w32_energyMax - (BGN_Inst->w32_energyMax >> 10);
- if (w32_en > BGN_Inst->w32_energyMax)
- {
- BGN_Inst->w32_energyMax = w32_en;
- }
-
- /* Set update level to at the minimum 60.21dB lower then the maximum energy */
- w32_enUpdateThreashold = (BGN_Inst->w32_energyMax + 524288) >> 20;
- if (w32_enUpdateThreashold > BGN_Inst->w32_energyUpdate)
- {
- BGN_Inst->w32_energyUpdate = w32_enUpdateThreashold;
- }
- }
-
-#ifdef NETEQ_VAD
-} /* closing initial if-statement */
-#endif /* NETEQ_VAD */
-
- return;
-}
-
-#undef SCRATCH_PW32_AUTO_CORR
-#undef SCRATCH_PW16_TEMP_VEC
-#undef SCRATCH_PW16_RC
-#undef SCRATCH_PW16_OUT_VEC
-
diff --git a/trunk/src/modules/audio_coding/neteq/buffer_stats.h b/trunk/src/modules/audio_coding/neteq/buffer_stats.h
deleted file mode 100644
index 9820519..0000000
--- a/trunk/src/modules/audio_coding/neteq/buffer_stats.h
+++ /dev/null
@@ -1,95 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Calculates and stores the packet buffer statistics.
- */
-
-#ifndef BUFFER_STATS_H
-#define BUFFER_STATS_H
-
-#include "automode.h"
-#include "webrtc_neteq.h" /* to define enum WebRtcNetEQPlayoutMode */
-
-/* NetEQ related decisions */
-#define BUFSTATS_DO_NORMAL 0
-#define BUFSTATS_DO_ACCELERATE 1
-#define BUFSTATS_DO_MERGE 2
-#define BUFSTATS_DO_EXPAND 3
-#define BUFSTAT_REINIT 4
-#define BUFSTATS_DO_RFC3389CNG_PACKET 5
-#define BUFSTATS_DO_RFC3389CNG_NOPACKET 6
-#define BUFSTATS_DO_INTERNAL_CNG_NOPACKET 7
-#define BUFSTATS_DO_PREEMPTIVE_EXPAND 8
-#define BUFSTAT_REINIT_DECODER 9
-#define BUFSTATS_DO_DTMF_ONLY 10
-/* Decisions related to when NetEQ is switched off (or in FAX mode) */
-#define BUFSTATS_DO_ALTERNATIVE_PLC 11
-#define BUFSTATS_DO_ALTERNATIVE_PLC_INC_TS 12
-#define BUFSTATS_DO_AUDIO_REPETITION 13
-#define BUFSTATS_DO_AUDIO_REPETITION_INC_TS 14
-
-/* Reinit decoder states after this number of expands (upon arrival of new packet) */
-#define REINIT_AFTER_EXPANDS 100
-
-/* Wait no longer than this number of RecOut calls before using an "early" packet */
-#define MAX_WAIT_FOR_PACKET 10
-
-/* CNG modes */
-#define CNG_OFF 0
-#define CNG_RFC3389_ON 1
-#define CNG_INTERNAL_ON 2
-
-typedef struct
-{
-
- /* store statistical data here */
- WebRtc_Word16 w16_cngOn; /* remember if CNG is interrupted by other event (e.g. DTMF) */
- WebRtc_Word16 w16_noExpand;
- WebRtc_Word32 uw32_CNGplayedTS;
-
- /* VQmon data */
- WebRtc_UWord16 avgDelayMsQ8;
- WebRtc_Word16 maxDelayMs;
-
- AutomodeInst_t Automode_inst;
-
-} BufstatsInst_t;
-
-/****************************************************************************
- * WebRtcNetEQ_BufstatsDecision()
- *
- * Gives a decision about what action that is currently desired
- *
- *
- * Input:
- * inst: The bufstat instance
- * cur_size: Current buffer size in ms in Q3 domain
- * targetTS: The desired timestamp to start playout from
- * availableTS: The closest future value available in buffer
- * noPacket 1 if no packet is available, makes availableTS undefined
- * prevPlayMode mode of last NetEq playout
- * timestampsPerCall number of timestamp for 10ms
- *
- * Output:
- * Returns: A decision, as defined above (see top of file)
- *
- */
-
-WebRtc_UWord16 WebRtcNetEQ_BufstatsDecision(BufstatsInst_t *inst, WebRtc_Word16 frameSize,
- WebRtc_Word32 cur_size, WebRtc_UWord32 targetTS,
- WebRtc_UWord32 availableTS, int noPacket,
- int cngPacket, int prevPlayMode,
- enum WebRtcNetEQPlayoutMode playoutMode,
- int timestampsPerCall, int NoOfExpandCalls,
- WebRtc_Word16 fs_mult,
- WebRtc_Word16 lastModeBGNonly, int playDtmf);
-
-#endif
diff --git a/trunk/src/modules/audio_coding/neteq/bufstats_decision.c b/trunk/src/modules/audio_coding/neteq/bufstats_decision.c
deleted file mode 100644
index 3d37e17..0000000
--- a/trunk/src/modules/audio_coding/neteq/bufstats_decision.c
+++ /dev/null
@@ -1,426 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the function where the main decision logic for buffer level
- * adaptation happens.
- */
-
-#include "buffer_stats.h"
-
-#include <assert.h>
-
-#include "signal_processing_library.h"
-
-#include "automode.h"
-#include "neteq_defines.h"
-#include "neteq_error_codes.h"
-#include "webrtc_neteq.h"
-
-#define NETEQ_BUFSTAT_20MS_Q7 2560 /* = 20 ms in Q7 */
-
-WebRtc_UWord16 WebRtcNetEQ_BufstatsDecision(BufstatsInst_t *inst, WebRtc_Word16 frameSize,
- WebRtc_Word32 cur_size, WebRtc_UWord32 targetTS,
- WebRtc_UWord32 availableTS, int noPacket,
- int cngPacket, int prevPlayMode,
- enum WebRtcNetEQPlayoutMode playoutMode,
- int timestampsPerCall, int NoOfExpandCalls,
- WebRtc_Word16 fs_mult,
- WebRtc_Word16 lastModeBGNonly, int playDtmf)
-{
-
- int currentDelayMs;
- WebRtc_Word32 currSizeSamples = cur_size;
- WebRtc_Word16 extraDelayPacketsQ8 = 0;
-
- /* Avoid overflow if the buffer size should be really large (cur_size is limited 256ms) */
- WebRtc_Word32 curr_sizeQ7 = WEBRTC_SPL_LSHIFT_W32(cur_size, 4);
- WebRtc_UWord16 level_limit_hi, level_limit_lo;
-
- inst->Automode_inst.prevTimeScale &= (prevPlayMode == MODE_SUCCESS_ACCELERATE
- || prevPlayMode == MODE_LOWEN_ACCELERATE || prevPlayMode == MODE_SUCCESS_PREEMPTIVE
- || prevPlayMode == MODE_LOWEN_PREEMPTIVE);
-
- if ((prevPlayMode != MODE_RFC3389CNG) && (prevPlayMode != MODE_CODEC_INTERNAL_CNG))
- {
- /*
- * Do not update buffer history if currently playing CNG
- * since it will bias the filtered buffer level.
- */
- WebRtcNetEQ_BufferLevelFilter(cur_size, &(inst->Automode_inst), timestampsPerCall,
- fs_mult);
- }
- else
- {
- /* only update time counters */
- inst->Automode_inst.packetIatCountSamp += timestampsPerCall; /* packet inter-arrival time */
- inst->Automode_inst.peakIatCountSamp += timestampsPerCall; /* peak inter-arrival time */
- inst->Automode_inst.timescaleHoldOff >>= 1; /* time-scaling limiter */
- }
- cur_size = WEBRTC_SPL_MIN(curr_sizeQ7, WEBRTC_SPL_WORD16_MAX);
-
- /* Calculate VQmon related variables */
- /* avgDelay = avgDelay*(511/512) + currentDelay*(1/512) (sample ms delay in Q8) */
- inst->avgDelayMsQ8 = (WebRtc_Word16) (WEBRTC_SPL_MUL_16_16_RSFT(inst->avgDelayMsQ8,511,9)
- + (cur_size >> 9));
-
- /* Update maximum delay if needed */
- currentDelayMs = (curr_sizeQ7 >> 7);
- if (currentDelayMs > inst->maxDelayMs)
- {
- inst->maxDelayMs = currentDelayMs;
- }
-
- /* NetEQ is on with normal or steaming mode */
- if (playoutMode == kPlayoutOn || playoutMode == kPlayoutStreaming)
- {
- /* Guard for errors, so that it should not get stuck in error mode */
- if (prevPlayMode == MODE_ERROR)
- {
- if (noPacket)
- {
- return BUFSTATS_DO_EXPAND;
- }
- else
- {
- return BUFSTAT_REINIT;
- }
- }
-
- if (prevPlayMode != MODE_EXPAND && prevPlayMode != MODE_FADE_TO_BGN)
- {
- inst->w16_noExpand = 1;
- }
- else
- {
- inst->w16_noExpand = 0;
- }
-
- if (cngPacket)
- {
- /* signed difference between wanted and available TS */
- WebRtc_Word32 diffTS = (inst->uw32_CNGplayedTS + targetTS) - availableTS;
- int32_t optimal_level_samp = (inst->Automode_inst.optBufLevel *
- inst->Automode_inst.packetSpeechLenSamp) >> 8;
- int32_t excess_waiting_time_samp = -diffTS - optimal_level_samp;
-
- if (excess_waiting_time_samp > optimal_level_samp / 2)
- {
- /* The waiting time for this packet will be longer than 1.5
- * times the wanted buffer delay. Advance the clock to cut
- * waiting time down to the optimal.
- */
- inst->uw32_CNGplayedTS += excess_waiting_time_samp;
- diffTS += excess_waiting_time_samp;
- }
-
- if ((diffTS) < 0 && (prevPlayMode == MODE_RFC3389CNG))
- {
- /* Not time to play this packet yet. Wait another round before using this
- * packet. Keep on playing CNG from previous CNG parameters. */
- return BUFSTATS_DO_RFC3389CNG_NOPACKET;
- }
-
- /* otherwise, go for the CNG packet now */
- return BUFSTATS_DO_RFC3389CNG_PACKET;
- }
-
- /*Check for expand/cng */
- if (noPacket)
- {
- if (inst->w16_cngOn == CNG_RFC3389_ON)
- {
- /* keep on playing CNG */
- return BUFSTATS_DO_RFC3389CNG_NOPACKET;
- }
- else if (inst->w16_cngOn == CNG_INTERNAL_ON)
- {
- /* keep on playing internal CNG */
- return BUFSTATS_DO_INTERNAL_CNG_NOPACKET;
- }
- else if (playDtmf == 1)
- {
- /* we have not audio data, but can play DTMF */
- return BUFSTATS_DO_DTMF_ONLY;
- }
- else
- {
- /* nothing to play => do Expand */
- return BUFSTATS_DO_EXPAND;
- }
- }
-
- /*
- * If the expand period was very long, reset NetEQ since it is likely that the
- * sender was restarted.
- */
- if (NoOfExpandCalls > REINIT_AFTER_EXPANDS) return BUFSTAT_REINIT_DECODER;
-
- /* Calculate extra delay in Q8 packets */
- if (inst->Automode_inst.extraDelayMs > 0 && inst->Automode_inst.packetSpeechLenSamp
- > 0)
- {
- extraDelayPacketsQ8 = WebRtcSpl_DivW32W16ResW16(
- (WEBRTC_SPL_MUL(inst->Automode_inst.extraDelayMs, 8 * fs_mult) << 8),
- inst->Automode_inst.packetSpeechLenSamp);
- /* (extra delay in samples in Q8) */
- }
-
- /* Check if needed packet is available */
- if (targetTS == availableTS)
- {
-
- /* If last mode was not expand, and there is no DTMF to play */
- if (inst->w16_noExpand == 1 && playDtmf == 0)
- {
- /* If so check for accelerate */
-
- level_limit_lo = ((inst->Automode_inst.optBufLevel) >> 1) /* 50 % */
- + ((inst->Automode_inst.optBufLevel) >> 2); /* ... + 25% = 75% */
-
- /* set upper limit to optBufLevel, but make sure that window is at least 20ms */
- level_limit_hi = WEBRTC_SPL_MAX(inst->Automode_inst.optBufLevel,
- level_limit_lo +
- WebRtcSpl_DivW32W16ResW16((WEBRTC_SPL_MUL(20*8, fs_mult) << 8),
- inst->Automode_inst.packetSpeechLenSamp));
-
- /* if extra delay is non-zero, add it */
- if (extraDelayPacketsQ8 > 0)
- {
- level_limit_hi += extraDelayPacketsQ8;
- level_limit_lo += extraDelayPacketsQ8;
- }
-
- if (((inst->Automode_inst.buffLevelFilt >= level_limit_hi) &&
- (inst->Automode_inst.timescaleHoldOff == 0)) ||
- (inst->Automode_inst.buffLevelFilt >= level_limit_hi << 2))
- {
- /*
- * Buffer level higher than limit and time-scaling allowed,
- * OR buffer level _really_ high.
- */
- return BUFSTATS_DO_ACCELERATE;
- }
- else if ((inst->Automode_inst.buffLevelFilt < level_limit_lo)
- && (inst->Automode_inst.timescaleHoldOff == 0))
- {
- return BUFSTATS_DO_PREEMPTIVE_EXPAND;
- }
- }
- return BUFSTATS_DO_NORMAL;
- }
-
- /* Check for Merge */
- else if (availableTS > targetTS)
- {
-
- /* Check that we do not play a packet "too early" */
- if ((prevPlayMode == MODE_EXPAND)
- && (availableTS - targetTS
- < (WebRtc_UWord32) WEBRTC_SPL_MUL_16_16((WebRtc_Word16)timestampsPerCall,
- (WebRtc_Word16)REINIT_AFTER_EXPANDS))
- && (NoOfExpandCalls < MAX_WAIT_FOR_PACKET)
- && (availableTS
- > targetTS
- + WEBRTC_SPL_MUL_16_16((WebRtc_Word16)timestampsPerCall,
- (WebRtc_Word16)NoOfExpandCalls))
- && (inst->Automode_inst.buffLevelFilt <= inst->Automode_inst.optBufLevel
- + extraDelayPacketsQ8))
- {
- if (playDtmf == 1)
- {
- /* we still have DTMF to play, so do not perform expand */
- return BUFSTATS_DO_DTMF_ONLY;
- }
- else
- {
- /* nothing to play */
- return BUFSTATS_DO_EXPAND;
- }
- }
-
- /* If previous was CNG period or BGNonly then no merge is needed */
- if ((prevPlayMode == MODE_RFC3389CNG) || (prevPlayMode == MODE_CODEC_INTERNAL_CNG)
- || lastModeBGNonly)
- {
- /*
- * Keep the same delay as before the CNG (or maximum 70 ms in buffer as safety
- * precaution), but make sure that the number of samples in buffer is no
- * higher than 4 times the optimal level.
- */
- WebRtc_Word32 diffTS = (inst->uw32_CNGplayedTS + targetTS) - availableTS;
- if (diffTS >= 0
- || (WEBRTC_SPL_MUL_16_16_RSFT( inst->Automode_inst.optBufLevel
- + extraDelayPacketsQ8,
- inst->Automode_inst.packetSpeechLenSamp, 6) < currSizeSamples))
- {
- /* it is time to play this new packet */
- return BUFSTATS_DO_NORMAL;
- }
- else
- {
- /* it is too early to play this new packet => keep on playing CNG */
- if (prevPlayMode == MODE_RFC3389CNG)
- {
- return BUFSTATS_DO_RFC3389CNG_NOPACKET;
- }
- else if (prevPlayMode == MODE_CODEC_INTERNAL_CNG)
- {
- return BUFSTATS_DO_INTERNAL_CNG_NOPACKET;
- }
- else if (playDtmf == 1)
- {
- /* we have not audio data, but can play DTMF */
- return BUFSTATS_DO_DTMF_ONLY;
- }
- else /* lastModeBGNonly */
- {
- /* signal expand, but this will result in BGN again */
- return BUFSTATS_DO_EXPAND;
- }
- }
- }
-
- /* Do not merge unless we have done a Expand before (for complexity reasons) */
- if ((inst->w16_noExpand == 0) || ((frameSize < timestampsPerCall) && (cur_size
- > NETEQ_BUFSTAT_20MS_Q7)))
- {
- return BUFSTATS_DO_MERGE;
- }
- else if (playDtmf == 1)
- {
- /* play DTMF instead of expand */
- return BUFSTATS_DO_DTMF_ONLY;
- }
- else
- {
- return BUFSTATS_DO_EXPAND;
- }
- }
- }
- else
- { /* kPlayoutOff or kPlayoutFax */
- if (cngPacket)
- {
- if (((WebRtc_Word32) ((inst->uw32_CNGplayedTS + targetTS) - availableTS)) >= 0)
- {
- /* time to play this packet now */
- return BUFSTATS_DO_RFC3389CNG_PACKET;
- }
- else
- {
- /* wait before playing this packet */
- return BUFSTATS_DO_RFC3389CNG_NOPACKET;
- }
- }
- if (noPacket)
- {
- /*
- * No packet =>
- * 1. If in CNG mode play as usual
- * 2. Otherwise use other method to generate data and hold TS value
- */
- if (inst->w16_cngOn == CNG_RFC3389_ON)
- {
- /* keep on playing CNG */
- return BUFSTATS_DO_RFC3389CNG_NOPACKET;
- }
- else if (inst->w16_cngOn == CNG_INTERNAL_ON)
- {
- /* keep on playing internal CNG */
- return BUFSTATS_DO_INTERNAL_CNG_NOPACKET;
- }
- else
- {
- /* nothing to play => invent some data to play out */
- if (playoutMode == kPlayoutOff)
- {
- return BUFSTATS_DO_ALTERNATIVE_PLC;
- }
- else if (playoutMode == kPlayoutFax)
- {
- return BUFSTATS_DO_AUDIO_REPETITION;
- }
- else
- {
- /* UNDEFINED, should not get here... */
- assert(0);
- return BUFSTAT_REINIT;
- }
- }
- }
- else if (targetTS == availableTS)
- {
- return BUFSTATS_DO_NORMAL;
- }
- else
- {
- if (((WebRtc_Word32) ((inst->uw32_CNGplayedTS + targetTS) - availableTS)) >= 0)
- {
- return BUFSTATS_DO_NORMAL;
- }
- else if (playoutMode == kPlayoutOff)
- {
- /*
- * If currently playing CNG, continue with that. Don't increase TS
- * since uw32_CNGplayedTS will be increased.
- */
- if (inst->w16_cngOn == CNG_RFC3389_ON)
- {
- return BUFSTATS_DO_RFC3389CNG_NOPACKET;
- }
- else if (inst->w16_cngOn == CNG_INTERNAL_ON)
- {
- return BUFSTATS_DO_INTERNAL_CNG_NOPACKET;
- }
- else
- {
- /*
- * Otherwise, do PLC and increase TS while waiting for the time to
- * play this packet.
- */
- return BUFSTATS_DO_ALTERNATIVE_PLC_INC_TS;
- }
- }
- else if (playoutMode == kPlayoutFax)
- {
- /*
- * If currently playing CNG, continue with that don't increase TS since
- * uw32_CNGplayedTS will be increased.
- */
- if (inst->w16_cngOn == CNG_RFC3389_ON)
- {
- return BUFSTATS_DO_RFC3389CNG_NOPACKET;
- }
- else if (inst->w16_cngOn == CNG_INTERNAL_ON)
- {
- return BUFSTATS_DO_INTERNAL_CNG_NOPACKET;
- }
- else
- {
- /*
- * Otherwise, do audio repetition and increase TS while waiting for the
- * time to play this packet.
- */
- return BUFSTATS_DO_AUDIO_REPETITION_INC_TS;
- }
- }
- else
- {
- /* UNDEFINED, should not get here... */
- assert(0);
- return BUFSTAT_REINIT;
- }
- }
- }
- /* We should not get here (but sometimes we do anyway...) */
- return BUFSTAT_REINIT;
-}
-
diff --git a/trunk/src/modules/audio_coding/neteq/cng_internal.c b/trunk/src/modules/audio_coding/neteq/cng_internal.c
deleted file mode 100644
index f3a10dc..0000000
--- a/trunk/src/modules/audio_coding/neteq/cng_internal.c
+++ /dev/null
@@ -1,155 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the function for obtaining comfort noise from noise parameters
- * according to IETF RFC 3389.
- */
-
-#include "dsp.h"
-
-#include "signal_processing_library.h"
-#include "webrtc_cng.h"
-
-#include "dsp_helpfunctions.h"
-#include "neteq_error_codes.h"
-
-/****************************************************************************
- * WebRtcNetEQ_Cng(...)
- *
- * This function produces CNG according to RFC 3389.
- *
- * Input:
- * - inst : NetEQ DSP instance
- * - len : Number of samples to produce (max 640 or
- * 640 - fsHz*5/8000 for first-time CNG, governed by
- * the definition of WEBRTC_CNG_MAX_OUTSIZE_ORDER in
- * webrtc_cng.h)
- *
- * Output:
- * - pw16_outData : Output CNG
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-
-#ifdef NETEQ_CNG_CODEC
-/* Must compile NetEQ with CNG support to enable this function */
-
-int WebRtcNetEQ_Cng(DSPInst_t *inst, WebRtc_Word16 *pw16_outData, int len)
-{
- WebRtc_Word16 w16_winMute = 0; /* mixing factor for overlap data */
- WebRtc_Word16 w16_winUnMute = 0; /* mixing factor for comfort noise */
- WebRtc_Word16 w16_winMuteInc = 0; /* mixing factor increment (negative) */
- WebRtc_Word16 w16_winUnMuteInc = 0; /* mixing factor increment */
- int i;
-
- /*
- * Check if last RecOut call was other than RFC3389,
- * that is, this call is the first of a CNG period.
- */
- if (inst->w16_mode != MODE_RFC3389CNG)
- {
- /* Reset generation and overlap slightly with old data */
-
- /* Generate len samples + overlap */
- if (WebRtcCng_Generate(inst->CNG_Codec_inst, pw16_outData,
- (WebRtc_Word16) (len + inst->ExpandInst.w16_overlap), 1) < 0)
- {
- /* error returned */
- return -WebRtcCng_GetErrorCodeDec(inst->CNG_Codec_inst);
- }
-
- /* Set windowing parameters depending on sample rate */
- if (inst->fs == 8000)
- {
- /* Windowing in Q15 */
- w16_winMute = NETEQ_OVERLAP_WINMUTE_8KHZ_START;
- w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_8KHZ_INC;
- w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_8KHZ_START;
- w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_8KHZ_INC;
-#ifdef NETEQ_WIDEBAND
- }
- else if (inst->fs == 16000)
- {
- /* Windowing in Q15 */
- w16_winMute = NETEQ_OVERLAP_WINMUTE_16KHZ_START;
- w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_16KHZ_INC;
- w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_16KHZ_START;
- w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_16KHZ_INC;
-#endif
-#ifdef NETEQ_32KHZ_WIDEBAND
- }
- else if (inst->fs == 32000)
- {
- /* Windowing in Q15 */
- w16_winMute = NETEQ_OVERLAP_WINMUTE_32KHZ_START;
- w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_32KHZ_INC;
- w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_32KHZ_START;
- w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_32KHZ_INC;
-#endif
-#ifdef NETEQ_48KHZ_WIDEBAND
- }
- else if (inst->fs == 48000)
- {
- /* Windowing in Q15 */
- w16_winMute = NETEQ_OVERLAP_WINMUTE_48KHZ_START;
- w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_48KHZ_INC;
- w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_48KHZ_START;
- w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_48KHZ_INC;
-#endif
- }
- else
- {
- /* Unsupported sample rate (should not be possible) */
- return NETEQ_OTHER_ERROR;
- }
-
- /* Do overlap add between new vector and overlap */
- for (i = 0; i < inst->ExpandInst.w16_overlap; i++)
- {
- /* overlapVec[i] = WinMute * overlapVec[i] + WinUnMute * outData[i] */
- inst->ExpandInst.pw16_overlapVec[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
- WEBRTC_SPL_MUL_16_16(
- inst->ExpandInst.pw16_overlapVec[i], w16_winMute) +
- WEBRTC_SPL_MUL_16_16(pw16_outData[i], w16_winUnMute)
- + 16384, 15); /* shift with proper rounding */
-
- w16_winMute += w16_winMuteInc; /* decrease mute factor (inc<0) */
- w16_winUnMute += w16_winUnMuteInc; /* increase unmute factor (inc>0) */
-
- }
-
- /*
- * Shift the contents of the outData buffer by overlap samples, since we
- * already used these first samples in the overlapVec above
- */
-
- WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_outData+inst->ExpandInst.w16_overlap, len);
-
- }
- else
- {
- /* This is a subsequent CNG call; no special overlap needed */
-
- /* Generate len samples */
- if (WebRtcCng_Generate(inst->CNG_Codec_inst, pw16_outData, (WebRtc_Word16) len, 0) < 0)
- {
- /* error returned */
- return -WebRtcCng_GetErrorCodeDec(inst->CNG_Codec_inst);
- }
- }
-
- return 0;
-
-}
-
-#endif /* NETEQ_CNG_CODEC */
-
diff --git a/trunk/src/modules/audio_coding/neteq/codec_db.c b/trunk/src/modules/audio_coding/neteq/codec_db.c
deleted file mode 100644
index cb63aea..0000000
--- a/trunk/src/modules/audio_coding/neteq/codec_db.c
+++ /dev/null
@@ -1,743 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Implementation of the codec database.
- */
-
-#include "codec_db.h"
-
-#include <string.h> /* to define NULL */
-
-#include "signal_processing_library.h"
-
-#include "neteq_error_codes.h"
-
-/*
- * Resets the codec database.
- */
-
-int WebRtcNetEQ_DbReset(CodecDbInst_t *inst)
-{
- int i;
-
- WebRtcSpl_MemSetW16((WebRtc_Word16*) inst, 0,
- sizeof(CodecDbInst_t) / sizeof(WebRtc_Word16));
-
- for (i = 0; i < NUM_TOTAL_CODECS; i++)
- {
- inst->position[i] = -1;
- }
-
- for (i = 0; i < NUM_CODECS; i++)
- {
- inst->payloadType[i] = -1;
- }
-
- for (i = 0; i < NUM_CNG_CODECS; i++)
- {
- inst->CNGpayloadType[i] = -1;
- }
-
- return 0;
-}
-
-/*
- * Adds a new codec to the database.
- */
-
-int WebRtcNetEQ_DbAdd(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec,
- WebRtc_Word16 payloadType, FuncDecode funcDecode,
- FuncDecode funcDecodeRCU, FuncDecodePLC funcDecodePLC,
- FuncDecodeInit funcDecodeInit, FuncAddLatePkt funcAddLatePkt,
- FuncGetMDinfo funcGetMDinfo, FuncGetPitchInfo funcGetPitch,
- FuncUpdBWEst funcUpdBWEst, FuncGetErrorCode funcGetErrorCode,
- void* codec_state, WebRtc_UWord16 codec_fs)
-{
-
- int temp;
- int insertCNGcodec = 0, overwriteCNGcodec = 0, CNGpos = -1;
-
-#ifndef NETEQ_RED_CODEC
- if (codec == kDecoderRED)
- {
- return CODEC_DB_UNSUPPORTED_CODEC;
- }
-#endif
- if (((int) codec <= (int) kDecoderReservedStart) || ((int) codec
- >= (int) kDecoderReservedEnd))
- {
- return CODEC_DB_UNSUPPORTED_CODEC;
- }
-
- if ((codec_fs != 8000)
-#ifdef NETEQ_WIDEBAND
- &&(codec_fs!=16000)
-#endif
-#ifdef NETEQ_32KHZ_WIDEBAND
- &&(codec_fs!=32000)
-#endif
-#ifdef NETEQ_48KHZ_WIDEBAND
- &&(codec_fs!=48000)
-#endif
- )
- {
- return CODEC_DB_UNSUPPORTED_FS;
- }
-
- /* Ensure that the codec type is supported */
- switch (codec)
- {
-#ifdef NETEQ_PCM16B_CODEC
- case kDecoderPCM16B :
-#endif
-#ifdef NETEQ_G711_CODEC
- case kDecoderPCMu :
- case kDecoderPCMa :
-#endif
-#ifdef NETEQ_ILBC_CODEC
- case kDecoderILBC :
-#endif
-#ifdef NETEQ_ISAC_CODEC
- case kDecoderISAC :
-#endif
-#ifdef NETEQ_ISAC_SWB_CODEC
- case kDecoderISACswb :
-#endif
-#ifdef NETEQ_G722_CODEC
- case kDecoderG722 :
-#endif
-#ifdef NETEQ_WIDEBAND
- case kDecoderPCM16Bwb :
-#endif
-#ifdef NETEQ_32KHZ_WIDEBAND
- case kDecoderPCM16Bswb32kHz :
-#endif
-#ifdef NETEQ_CNG_CODEC
- case kDecoderCNG :
-#endif
-#ifdef NETEQ_ATEVENT_DECODE
- case kDecoderAVT :
-#endif
-#ifdef NETEQ_RED_CODEC
- case kDecoderRED :
-#endif
-#ifdef NETEQ_48KHZ_WIDEBAND
- case kDecoderPCM16Bswb48kHz :
-#endif
-#ifdef NETEQ_ARBITRARY_CODEC
- case kDecoderArbitrary:
-#endif
-#ifdef NETEQ_G729_CODEC
- case kDecoderG729:
-#endif
-#ifdef NETEQ_G729_1_CODEC
- case kDecoderG729_1 :
-#endif
-#ifdef NETEQ_G726_CODEC
- case kDecoderG726_16 :
- case kDecoderG726_24 :
- case kDecoderG726_32 :
- case kDecoderG726_40 :
-#endif
-#ifdef NETEQ_G722_1_CODEC
- case kDecoderG722_1_16 :
- case kDecoderG722_1_24 :
- case kDecoderG722_1_32 :
-#endif
-#ifdef NETEQ_G722_1C_CODEC
- case kDecoderG722_1C_24 :
- case kDecoderG722_1C_32 :
- case kDecoderG722_1C_48 :
-#endif
-#ifdef NETEQ_SPEEX_CODEC
- case kDecoderSPEEX_8 :
- case kDecoderSPEEX_16 :
-#endif
-#ifdef NETEQ_CELT_CODEC
- case kDecoderCELT_32 :
-#endif
-#ifdef NETEQ_GSMFR_CODEC
- case kDecoderGSMFR :
-#endif
-#ifdef NETEQ_AMR_CODEC
- case kDecoderAMR :
-#endif
-#ifdef NETEQ_AMRWB_CODEC
- case kDecoderAMRWB :
-#endif
- {
- /* If we end up here, the inserted codec is supported => Do nothing */
- break;
- }
- default:
- {
- /* If we get to this point, the inserted codec is not supported */
- return CODEC_DB_UNSUPPORTED_CODEC;
- }
- }
-
- /* Check to see if payload type is taken */
- if (WebRtcNetEQ_DbGetCodec(inst, payloadType) > 0)
- {
- return CODEC_DB_PAYLOAD_TAKEN;
- }
-
- /* Special case for CNG codecs */
- if (codec == kDecoderCNG)
- {
- /* check if this is first CNG codec to be registered */
- if (WebRtcNetEQ_DbGetPayload(inst, codec) == CODEC_DB_NOT_EXIST2)
- {
- /* no other CNG codec found */
- insertCNGcodec = 1;
- }
-
- /* find the appropriate insert position in CNG payload vector */
- switch (codec_fs)
- {
-#ifdef NETEQ_WIDEBAND
- case 16000:
- CNGpos = 1;
- break;
-#endif
-#ifdef NETEQ_32KHZ_WIDEBAND
- case 32000:
- CNGpos = 2;
- break;
-#endif
-#ifdef NETEQ_48KHZ_WIDEBAND
- case 48000:
- CNGpos = 3;
- break;
-#endif
- default: /* 8000 Hz case */
- CNGpos = 0;
- /*
- * The 8 kHz CNG payload type is the one associated with the regular codec DB
- * should override any other setting.
- * Overwrite if this isn't the first CNG
- */
- overwriteCNGcodec = !insertCNGcodec;
- break;
- }
-
- /* insert CNG payload type */
- inst->CNGpayloadType[CNGpos] = payloadType;
-
- }
-
- if ((codec != kDecoderCNG) || (insertCNGcodec == 1) || (overwriteCNGcodec == 1))
- {
- /* Check if we have reached the maximum numbers of simultaneous codecs */
- if (inst->nrOfCodecs == NUM_CODECS) return CODEC_DB_FULL;
-
- /* Check that codec has not already been initialized to DB =>
- remove it and reinitialize according to new spec */
- if ((inst->position[codec] != -1) && (overwriteCNGcodec != 1))
- { /* if registering multiple CNG codecs, don't remove, just overwrite */
- WebRtcNetEQ_DbRemove(inst, codec);
- }
-
- if (overwriteCNGcodec == 1)
- {
- temp = inst->position[codec];
- }
- else
- {
- temp = inst->nrOfCodecs; /* Store this codecs position */
- inst->position[codec] = temp;
- inst->nrOfCodecs++;
- }
-
- inst->payloadType[temp] = payloadType;
-
- /* Copy to database */
- inst->codec_state[temp] = codec_state;
- inst->funcDecode[temp] = funcDecode;
- inst->funcDecodeRCU[temp] = funcDecodeRCU;
- inst->funcAddLatePkt[temp] = funcAddLatePkt;
- inst->funcDecodeInit[temp] = funcDecodeInit;
- inst->funcDecodePLC[temp] = funcDecodePLC;
- inst->funcGetMDinfo[temp] = funcGetMDinfo;
- inst->funcGetPitch[temp] = funcGetPitch;
- inst->funcUpdBWEst[temp] = funcUpdBWEst;
- inst->funcGetErrorCode[temp] = funcGetErrorCode;
- inst->codec_fs[temp] = codec_fs;
-
- }
-
- return 0;
-}
-
-/*
- * Removes a codec from the database.
- */
-
-int WebRtcNetEQ_DbRemove(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec)
-{
- int i;
- int pos = -1;
-
-#ifndef NETEQ_RED_CODEC
- if (codec == kDecoderRED)
- {
- return CODEC_DB_UNSUPPORTED_CODEC;
- }
-#endif
- if (((int) codec <= (int) kDecoderReservedStart) || ((int) codec
- >= (int) kDecoderReservedEnd))
- {
- return CODEC_DB_UNSUPPORTED_CODEC;
- }
-
- pos = inst->position[codec];
- if (pos == -1)
- {
- return CODEC_DB_NOT_EXIST4;
- }
- else
- {
- /* Remove this codec */
- inst->position[codec] = -1;
- for (i = pos; i < (inst->nrOfCodecs - 1); i++)
- {
- inst->payloadType[i] = inst->payloadType[i + 1];
- inst->codec_state[i] = inst->codec_state[i + 1];
- inst->funcDecode[i] = inst->funcDecode[i + 1];
- inst->funcDecodeRCU[i] = inst->funcDecodeRCU[i + 1];
- inst->funcAddLatePkt[i] = inst->funcAddLatePkt[i + 1];
- inst->funcDecodeInit[i] = inst->funcDecodeInit[i + 1];
- inst->funcDecodePLC[i] = inst->funcDecodePLC[i + 1];
- inst->funcGetMDinfo[i] = inst->funcGetMDinfo[i + 1];
- inst->funcGetPitch[i] = inst->funcGetPitch[i + 1];
- inst->funcUpdBWEst[i] = inst->funcUpdBWEst[i + 1];
- inst->funcGetErrorCode[i] = inst->funcGetErrorCode[i + 1];
- inst->codec_fs[i] = inst->codec_fs[i + 1];
- }
- inst->payloadType[i] = -1;
- inst->codec_state[i] = NULL;
- inst->funcDecode[i] = NULL;
- inst->funcDecodeRCU[i] = NULL;
- inst->funcAddLatePkt[i] = NULL;
- inst->funcDecodeInit[i] = NULL;
- inst->funcDecodePLC[i] = NULL;
- inst->funcGetMDinfo[i] = NULL;
- inst->funcGetPitch[i] = NULL;
- inst->funcUpdBWEst[i] = NULL;
- inst->funcGetErrorCode[i] = NULL;
- inst->codec_fs[i] = 0;
- /* Move down all the codecs above this one */
- for (i = 0; i < NUM_TOTAL_CODECS; i++)
- {
- if (inst->position[i] >= pos)
- {
- inst->position[i] = inst->position[i] - 1;
- }
- }
- inst->nrOfCodecs--;
-
- if (codec == kDecoderCNG)
- {
- /* also remove all registered CNG payload types */
- for (i = 0; i < NUM_CNG_CODECS; i++)
- {
- inst->CNGpayloadType[i] = -1;
- }
- }
- }
- return 0;
-}
-
-/*
- * Get the decoder function pointers for a codec.
- */
-
-int WebRtcNetEQ_DbGetPtrs(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec,
- CodecFuncInst_t *ptr_inst)
-{
-
- int pos = inst->position[codec];
- if ((codec <= kDecoderReservedStart) || (codec >= kDecoderReservedEnd) || (codec
- > NUM_TOTAL_CODECS))
- {
- /* ERROR */
- pos = -1;
- }
- if (pos >= 0)
- {
- ptr_inst->codec_state = inst->codec_state[pos];
- ptr_inst->funcAddLatePkt = inst->funcAddLatePkt[pos];
- ptr_inst->funcDecode = inst->funcDecode[pos];
- ptr_inst->funcDecodeRCU = inst->funcDecodeRCU[pos];
- ptr_inst->funcDecodeInit = inst->funcDecodeInit[pos];
- ptr_inst->funcDecodePLC = inst->funcDecodePLC[pos];
- ptr_inst->funcGetMDinfo = inst->funcGetMDinfo[pos];
- ptr_inst->funcUpdBWEst = inst->funcUpdBWEst[pos];
- ptr_inst->funcGetErrorCode = inst->funcGetErrorCode[pos];
- ptr_inst->codec_fs = inst->codec_fs[pos];
- return 0;
- }
- else
- {
- WebRtcSpl_MemSetW16((WebRtc_Word16*) ptr_inst, 0,
- sizeof(CodecFuncInst_t) / sizeof(WebRtc_Word16));
- return CODEC_DB_NOT_EXIST1;
- }
-}
-
-/*
- * Returns payload number given a codec identifier.
- */
-
-int WebRtcNetEQ_DbGetPayload(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codecID)
-{
- if (inst->position[codecID] == -1)
- return CODEC_DB_NOT_EXIST2;
- else
- return (inst->payloadType[inst->position[codecID]]);
-
-}
-
-/*
- * Returns codec identifier given a payload number.
- * Returns -1 if the payload type does not exist.
- */
-
-int WebRtcNetEQ_DbGetCodec(CodecDbInst_t *inst, int payloadType)
-{
- int i, pos;
-
- for (i = 0; i < NUM_TOTAL_CODECS; i++)
- {
- pos = inst->position[i];
- if (pos != -1)
- {
- if (inst->payloadType[pos] == payloadType) return i;
- }
- }
-
- /* did not find payload type */
- /* check if it's a CNG codec */
- if (WebRtcNetEQ_DbIsCNGPayload(inst, payloadType))
- {
- return kDecoderCNG;
- }
-
- /* found no match */
- return CODEC_DB_NOT_EXIST3;
-}
-
-/*
- * Extracts the Payload Split information of the codec with the specified payloadType.
- */
-
-int WebRtcNetEQ_DbGetSplitInfo(SplitInfo_t *inst, enum WebRtcNetEQDecoder codecID,
- int codedsize)
-{
-
- switch (codecID)
- {
-#ifdef NETEQ_ISAC_CODEC
- case kDecoderISAC:
-#endif
-#ifdef NETEQ_ISAC_SWB_CODEC
- case kDecoderISACswb:
-#endif
-#ifdef NETEQ_ARBITRARY_CODEC
- case kDecoderArbitrary:
-#endif
-#ifdef NETEQ_AMR_CODEC
- case kDecoderAMR:
-#endif
-#ifdef NETEQ_AMRWB_CODEC
- case kDecoderAMRWB:
-#endif
-#ifdef NETEQ_G726_CODEC
- /* Treat G726 as non-splittable to simplify the implementation */
- case kDecoderG726_16:
- case kDecoderG726_24:
- case kDecoderG726_32:
- case kDecoderG726_40:
-#endif
-#ifdef NETEQ_SPEEX_CODEC
- case kDecoderSPEEX_8:
- case kDecoderSPEEX_16:
-#endif
-#ifdef NETEQ_CELT_CODEC
- case kDecoderCELT_32 :
-#endif
-#ifdef NETEQ_G729_1_CODEC
- case kDecoderG729_1:
-#endif
- {
- /* These codecs' payloads are not splittable */
- inst->deltaBytes = NO_SPLIT;
- return 0;
- }
-
- /*
- * Sample based coders are a special case.
- * In this case, deltaTime signals the number of bytes per timestamp unit times 2
- * in log2 domain.
- */
-#if (defined NETEQ_G711_CODEC)
- case kDecoderPCMu:
- case kDecoderPCMa:
- {
- inst->deltaBytes = -12;
- inst->deltaTime = 1;
- return 0;
- }
-#endif
-#if (defined NETEQ_G722_CODEC)
- case kDecoderG722:
- {
- inst->deltaBytes = -14;
- inst->deltaTime = 0;
- return 0;
- }
-#endif
-#if (defined NETEQ_PCM16B_CODEC)
- case kDecoderPCM16B:
- {
- inst->deltaBytes = -12;
- inst->deltaTime = 2;
- return 0;
- }
-#endif
-#if ((defined NETEQ_PCM16B_CODEC)&&(defined NETEQ_WIDEBAND))
- case kDecoderPCM16Bwb:
- {
- inst->deltaBytes = -14;
- inst->deltaTime = 2;
- return 0;
- }
-#endif
-#if ((defined NETEQ_PCM16B_CODEC)&&(defined NETEQ_32KHZ_WIDEBAND))
- case kDecoderPCM16Bswb32kHz:
- {
- inst->deltaBytes = -18;
- inst->deltaTime = 2;
- return 0;
- }
-#endif
-#if ((defined NETEQ_PCM16B_CODEC)&&(defined NETEQ_48KHZ_WIDEBAND))
- case kDecoderPCM16Bswb48kHz:
- {
- inst->deltaBytes = -22;
- inst->deltaTime = 2;
- return 0;
- }
-#endif
-
- /* Splittable payloads */
-#ifdef NETEQ_G722_1_CODEC
- case kDecoderG722_1_16:
- {
- inst->deltaBytes = 40;
- inst->deltaTime = 320;
- return 0;
- }
- case kDecoderG722_1_24:
- {
- inst->deltaBytes = 60;
- inst->deltaTime = 320;
- return 0;
- }
- case kDecoderG722_1_32:
- {
- inst->deltaBytes = 80;
- inst->deltaTime = 320;
- return 0;
- }
-#endif
-#ifdef NETEQ_G722_1C_CODEC
- case kDecoderG722_1C_24:
- {
- inst->deltaBytes = 60;
- inst->deltaTime = 640;
- return 0;
- }
- case kDecoderG722_1C_32:
- {
- inst->deltaBytes = 80;
- inst->deltaTime = 640;
- return 0;
- }
- case kDecoderG722_1C_48:
- {
- inst->deltaBytes = 120;
- inst->deltaTime = 640;
- return 0;
- }
-#endif
-#ifdef NETEQ_G729_CODEC
- case kDecoderG729:
- {
- inst->deltaBytes = 10;
- inst->deltaTime = 80;
- return 0;
- }
-#endif
-#ifdef NETEQ_ILBC_CODEC
- case kDecoderILBC:
- {
- /* Check for splitting of iLBC packets.
- * If payload size is a multiple of 50 bytes it should be split into 30ms frames.
- * If payload size is a multiple of 38 bytes it should be split into 20ms frames.
- * Least common multiplier between 38 and 50 is 950, so the payload size must be less than
- * 950 bytes in order to resolve the frames unambiguously.
- * Currently max 12 frames in one bundle.
- */
- switch (codedsize)
- {
- case 50:
- case 100:
- case 150:
- case 200:
- case 250:
- case 300:
- case 350:
- case 400:
- case 450:
- case 500:
- case 550:
- case 600:
- {
- inst->deltaBytes = 50;
- inst->deltaTime = 240;
- break;
- }
- case 38:
- case 76:
- case 114:
- case 152:
- case 190:
- case 228:
- case 266:
- case 304:
- case 342:
- case 380:
- case 418:
- case 456:
- {
- inst->deltaBytes = 38;
- inst->deltaTime = 160;
- break;
- }
- default:
- {
- return AMBIGUOUS_ILBC_FRAME_SIZE; /* Something not supported... */
- }
- }
- return 0;
- }
-#endif
-#ifdef NETEQ_GSMFR_CODEC
- case kDecoderGSMFR:
- {
- inst->deltaBytes = 33;
- inst->deltaTime = 160;
- return 0;
- }
-#endif
- default:
- { /*Unknown codec */
- inst->deltaBytes = NO_SPLIT;
- return CODEC_DB_UNKNOWN_CODEC;
- }
- } /* end of switch */
-}
-
-/*
- * Returns 1 if codec is multiple description, 0 otherwise.
- * NOTE: This function is a stub, since there currently are no MD codecs.
- */
-int WebRtcNetEQ_DbIsMDCodec(enum WebRtcNetEQDecoder codecID)
-{
- if (0) /* Add test for MD codecs here */
- return 1;
- else
- return 0;
-}
-
-/*
- * Returns 1 if payload type is registered as a CNG codec, 0 otherwise
- */
-int WebRtcNetEQ_DbIsCNGPayload(CodecDbInst_t *inst, int payloadType)
-{
-#ifdef NETEQ_CNG_CODEC
- int i;
-
- for(i=0; i<NUM_CNG_CODECS; i++)
- {
- if( (inst->CNGpayloadType[i] != -1) && (inst->CNGpayloadType[i] == payloadType) )
- {
- return 1;
- }
- }
-#endif
-
- return 0;
-
-}
-
-/*
- * Return the sample rate for the codec with the given payload type, 0 if error
- */
-WebRtc_UWord16 WebRtcNetEQ_DbGetSampleRate(CodecDbInst_t *inst, int payloadType)
-{
- int i;
- CodecFuncInst_t codecInst;
-
- /* Sanity */
- if (inst == NULL)
- {
- /* return 0 Hz */
- return 0;
- }
-
- /* Check among CNG payloads */
- for (i = 0; i < NUM_CNG_CODECS; i++)
- {
- if ((inst->CNGpayloadType[i] != -1) && (inst->CNGpayloadType[i] == payloadType))
- {
- switch (i)
- {
- case 1:
- return 16000;
- case 2:
- return 32000;
- case 3:
- return 48000;
- default:
- return 8000;
- }
- }
- }
-
- /* Not a CNG payload, check the other payloads */
- i = WebRtcNetEQ_DbGetCodec(inst, payloadType);
- if (i >= 0)
- {
- if (WebRtcNetEQ_DbGetPtrs(inst, (enum WebRtcNetEQDecoder) i, &codecInst) != 0)
- {
- /* Unexpected error, return 0 Hz */
- return 0;
- }
- return codecInst.codec_fs;
- }
-
- /* If we end up here, we got an error, return 0 Hz */
- return 0;
-
-}
-
diff --git a/trunk/src/modules/audio_coding/neteq/codec_db.h b/trunk/src/modules/audio_coding/neteq/codec_db.h
deleted file mode 100644
index 7f42980..0000000
--- a/trunk/src/modules/audio_coding/neteq/codec_db.h
+++ /dev/null
@@ -1,126 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Interface for the codec database.
- */
-
-#ifndef CODEC_DB_H
-#define CODEC_DB_H
-
-#include "typedefs.h"
-
-#include "webrtc_neteq.h"
-#include "codec_db_defines.h"
-#include "neteq_defines.h"
-
-#if defined(NETEQ_48KHZ_WIDEBAND)
- #define NUM_CNG_CODECS 4
-#elif defined(NETEQ_32KHZ_WIDEBAND)
- #define NUM_CNG_CODECS 3
-#elif defined(NETEQ_WIDEBAND)
- #define NUM_CNG_CODECS 2
-#else
- #define NUM_CNG_CODECS 1
-#endif
-
-typedef struct
-{
-
- WebRtc_Word16 position[NUM_TOTAL_CODECS];
- WebRtc_Word16 nrOfCodecs;
-
- WebRtc_Word16 payloadType[NUM_CODECS];
- FuncDecode funcDecode[NUM_CODECS];
- FuncDecode funcDecodeRCU[NUM_CODECS];
- FuncDecodePLC funcDecodePLC[NUM_CODECS];
- FuncDecodeInit funcDecodeInit[NUM_CODECS];
- FuncAddLatePkt funcAddLatePkt[NUM_CODECS];
- FuncGetMDinfo funcGetMDinfo[NUM_CODECS];
- FuncGetPitchInfo funcGetPitch[NUM_CODECS];
- FuncUpdBWEst funcUpdBWEst[NUM_CODECS];
- FuncGetErrorCode funcGetErrorCode[NUM_CODECS];
- void * codec_state[NUM_CODECS];
- WebRtc_UWord16 codec_fs[NUM_CODECS];
- WebRtc_Word16 CNGpayloadType[NUM_CNG_CODECS];
-
-} CodecDbInst_t;
-
-#define NO_SPLIT -1 /* codec payload cannot be split */
-
-typedef struct
-{
- WebRtc_Word16 deltaBytes;
- WebRtc_Word16 deltaTime;
-} SplitInfo_t;
-
-/*
- * Resets the codec database.
- */
-int WebRtcNetEQ_DbReset(CodecDbInst_t *inst);
-
-/*
- * Adds a new codec to the database.
- */
-int WebRtcNetEQ_DbAdd(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec,
- WebRtc_Word16 payloadType, FuncDecode funcDecode,
- FuncDecode funcDecodeRCU, FuncDecodePLC funcDecodePLC,
- FuncDecodeInit funcDecodeInit, FuncAddLatePkt funcAddLatePkt,
- FuncGetMDinfo funcGetMDinfo, FuncGetPitchInfo funcGetPitch,
- FuncUpdBWEst funcUpdBWEst, FuncGetErrorCode funcGetErrorCode,
- void* codec_state, WebRtc_UWord16 codec_fs);
-
-/*
- * Removes a codec from the database.
- */
-int WebRtcNetEQ_DbRemove(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codec);
-
-/*
- * Get the decoder function pointers for a codec.
- */
-int WebRtcNetEQ_DbGetPtrs(CodecDbInst_t *inst, enum WebRtcNetEQDecoder,
- CodecFuncInst_t *ptr_inst);
-
-/*
- * Returns payload number given a codec identifier.
- */
-
-int WebRtcNetEQ_DbGetPayload(CodecDbInst_t *inst, enum WebRtcNetEQDecoder codecID);
-
-/*
- * Returns codec identifier given a payload number.
- */
-
-int WebRtcNetEQ_DbGetCodec(CodecDbInst_t *inst, int payloadType);
-
-/*
- * Extracts the Payload Split information of the codec with the specified payloadType.
- */
-
-int WebRtcNetEQ_DbGetSplitInfo(SplitInfo_t *inst, enum WebRtcNetEQDecoder codecID,
- int codedsize);
-
-/*
- * Returns 1 if codec is multiple description type, 0 otherwise.
- */
-int WebRtcNetEQ_DbIsMDCodec(enum WebRtcNetEQDecoder codecID);
-
-/*
- * Returns 1 if payload type is registered as a CNG codec, 0 otherwise.
- */
-int WebRtcNetEQ_DbIsCNGPayload(CodecDbInst_t *inst, int payloadType);
-
-/*
- * Return the sample rate for the codec with the given payload type, 0 if error.
- */
-WebRtc_UWord16 WebRtcNetEQ_DbGetSampleRate(CodecDbInst_t *inst, int payloadType);
-
-#endif
-
diff --git a/trunk/src/modules/audio_coding/neteq/codec_db_defines.h b/trunk/src/modules/audio_coding/neteq/codec_db_defines.h
deleted file mode 100644
index 9b78b86..0000000
--- a/trunk/src/modules/audio_coding/neteq/codec_db_defines.h
+++ /dev/null
@@ -1,89 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Some definitions related to the codec database.
- */
-
-#ifndef CODEC_DB_DEFINES_H
-#define CODEC_DB_DEFINES_H
-
-#include "typedefs.h"
-
-#define NUM_CODECS 47 /* probably too large with the limited set of supported codecs*/
-#define NUM_TOTAL_CODECS kDecoderReservedEnd
-
-/*
- * Pointer to decoder function.
- */
-typedef WebRtc_Word16 (*FuncDecode)(void* state, WebRtc_Word16* encoded, WebRtc_Word16 len,
- WebRtc_Word16* decoded, WebRtc_Word16* speechType);
-
-/*
- * Pointer to PLC function.
- */
-typedef WebRtc_Word16 (*FuncDecodePLC)(void* state, WebRtc_Word16* decodec,
- WebRtc_Word16 frames);
-
-/*
- * Pointer to decoder init function.
- */
-typedef WebRtc_Word16 (*FuncDecodeInit)(void* state);
-
-/*
- * Pointer to add late packet function.
- */
-typedef WebRtc_Word16
- (*FuncAddLatePkt)(void* state, WebRtc_Word16* encoded, WebRtc_Word16 len);
-
-/*
- * Pointer to get MD infofunction.
- */
-typedef WebRtc_Word16 (*FuncGetMDinfo)(void* state);
-
-/*
- * Pointer to pitch info function.
- * Return 0 for unvoiced, -1 if pitch not availiable.
- */
-typedef WebRtc_Word16 (*FuncGetPitchInfo)(void* state, WebRtc_Word16* encoded,
- WebRtc_Word16* length);
-
-/*
- * Pointer to the update bandwidth estimate function
- */
-typedef WebRtc_Word16 (*FuncUpdBWEst)(void* state, const WebRtc_UWord16 *encoded,
- WebRtc_Word32 packet_size,
- WebRtc_UWord16 rtp_seq_number, WebRtc_UWord32 send_ts,
- WebRtc_UWord32 arr_ts);
-
-/*
- * Pointer to error code function
- */
-typedef WebRtc_Word16 (*FuncGetErrorCode)(void* state);
-
-typedef struct CodecFuncInst_t_
-{
-
- FuncDecode funcDecode;
- FuncDecode funcDecodeRCU;
- FuncDecodePLC funcDecodePLC;
- FuncDecodeInit funcDecodeInit;
- FuncAddLatePkt funcAddLatePkt;
- FuncGetMDinfo funcGetMDinfo;
- FuncUpdBWEst funcUpdBWEst; /* Currently in use for the ISAC family (without LC) only*/
- FuncGetErrorCode funcGetErrorCode;
- void * codec_state;
- WebRtc_UWord16 codec_fs;
- WebRtc_UWord32 timeStamp;
-
-} CodecFuncInst_t;
-
-#endif
-
diff --git a/trunk/src/modules/audio_coding/neteq/correlator.c b/trunk/src/modules/audio_coding/neteq/correlator.c
deleted file mode 100644
index 97c41da..0000000
--- a/trunk/src/modules/audio_coding/neteq/correlator.c
+++ /dev/null
@@ -1,132 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "dsp.h"
-
-#include "signal_processing_library.h"
-
-#include "dsp_helpfunctions.h"
-
-/* Scratch usage:
-
- Type Name size startpos endpos
- WebRtc_Word16 pw16_corrVec 62 0 61
- WebRtc_Word16 pw16_data_ds 124 0 123
- WebRtc_Word32 pw32_corr 2*54 124 231
-
- Total: 232
- */
-
-#define SCRATCH_pw16_corrVec 0
-#define SCRATCH_pw16_data_ds 0
-#define SCRATCH_pw32_corr 124
-
-#define NETEQ_CORRELATOR_DSVECLEN 124 /* 124 = 60 + 10 + 54 */
-
-WebRtc_Word16 WebRtcNetEQ_Correlator(DSPInst_t *inst,
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_scratchPtr,
-#endif
- WebRtc_Word16 *pw16_data,
- WebRtc_Word16 w16_dataLen,
- WebRtc_Word16 *pw16_corrOut,
- WebRtc_Word16 *pw16_corrScale)
-{
- WebRtc_Word16 w16_corrLen = 60;
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_data_ds = pw16_scratchPtr + SCRATCH_pw16_corrVec;
- WebRtc_Word32 *pw32_corr = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_pw32_corr);
- /* WebRtc_Word16 *pw16_corrVec = pw16_scratchPtr + SCRATCH_pw16_corrVec;*/
-#else
- WebRtc_Word16 pw16_data_ds[NETEQ_CORRELATOR_DSVECLEN];
- WebRtc_Word32 pw32_corr[54];
- /* WebRtc_Word16 pw16_corrVec[4+54+4];*/
-#endif
- /* WebRtc_Word16 *pw16_corr=&pw16_corrVec[4];*/
- WebRtc_Word16 w16_maxVal;
- WebRtc_Word32 w32_maxVal;
- WebRtc_Word16 w16_normVal;
- WebRtc_Word16 w16_normVal2;
- /* WebRtc_Word16 w16_corrUpsLen;*/
- WebRtc_Word16 *pw16_B = NULL;
- WebRtc_Word16 w16_Blen = 0;
- WebRtc_Word16 w16_factor = 0;
-
- /* Set constants depending on frequency used */
- if (inst->fs == 8000)
- {
- w16_Blen = 3;
- w16_factor = 2;
- pw16_B = (WebRtc_Word16*) WebRtcNetEQ_kDownsample8kHzTbl;
-#ifdef NETEQ_WIDEBAND
- }
- else if (inst->fs==16000)
- {
- w16_Blen = 5;
- w16_factor = 4;
- pw16_B = (WebRtc_Word16*)WebRtcNetEQ_kDownsample16kHzTbl;
-#endif
-#ifdef NETEQ_32KHZ_WIDEBAND
- }
- else if (inst->fs==32000)
- {
- w16_Blen = 7;
- w16_factor = 8;
- pw16_B = (WebRtc_Word16*)WebRtcNetEQ_kDownsample32kHzTbl;
-#endif
-#ifdef NETEQ_48KHZ_WIDEBAND
- }
- else /* if inst->fs==48000 */
- {
- w16_Blen = 7;
- w16_factor = 12;
- pw16_B = (WebRtc_Word16*)WebRtcNetEQ_kDownsample48kHzTbl;
-#endif
- }
-
- /* Downsample data in order to work on a 4 kHz sampled signal */
- WebRtcSpl_DownsampleFast(
- pw16_data + w16_dataLen - (NETEQ_CORRELATOR_DSVECLEN * w16_factor),
- (WebRtc_Word16) (NETEQ_CORRELATOR_DSVECLEN * w16_factor), pw16_data_ds,
- NETEQ_CORRELATOR_DSVECLEN, pw16_B, w16_Blen, w16_factor, (WebRtc_Word16) 0);
-
- /* Normalize downsampled vector to using entire 16 bit */
- w16_maxVal = WebRtcSpl_MaxAbsValueW16(pw16_data_ds, 124);
- w16_normVal = 16 - WebRtcSpl_NormW32((WebRtc_Word32) w16_maxVal);
- WebRtcSpl_VectorBitShiftW16(pw16_data_ds, NETEQ_CORRELATOR_DSVECLEN, pw16_data_ds,
- w16_normVal);
-
- /* Correlate from lag 10 to lag 60 (20..120 in NB and 40..240 in WB) */
-
- WebRtcNetEQ_CrossCorr(
- pw32_corr, &pw16_data_ds[NETEQ_CORRELATOR_DSVECLEN - w16_corrLen],
- &pw16_data_ds[NETEQ_CORRELATOR_DSVECLEN - w16_corrLen - 10], 60, 54,
- 6 /*maxValue... shifts*/, -1);
-
- /*
- * Move data from w32 to w16 vector.
- * Normalize downsampled vector to using all 14 bits
- */
- w32_maxVal = WebRtcSpl_MaxAbsValueW32(pw32_corr, 54);
- w16_normVal2 = 18 - WebRtcSpl_NormW32(w32_maxVal);
- w16_normVal2 = WEBRTC_SPL_MAX(w16_normVal2, 0);
-
- WebRtcSpl_VectorBitShiftW32ToW16(pw16_corrOut, 54, pw32_corr, w16_normVal2);
-
- /* Total scale factor (right shifts) of correlation value */
- *pw16_corrScale = 2 * w16_normVal + 6 + w16_normVal2;
-
- return (50 + 1);
-}
-
-#undef SCRATCH_pw16_corrVec
-#undef SCRATCH_pw16_data_ds
-#undef SCRATCH_pw32_corr
-
diff --git a/trunk/src/modules/audio_coding/neteq/delay_logging.h b/trunk/src/modules/audio_coding/neteq/delay_logging.h
deleted file mode 100644
index 04b1c40..0000000
--- a/trunk/src/modules/audio_coding/neteq/delay_logging.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Contains definitions for the delay logging functionality. Only used for debugging and
- * tracing purposes.
- */
-
-#ifndef DELAY_LOGGING_H
-#define DELAY_LOGGING_H
-
-#define NETEQ_DELAY_LOGGING_VERSION_STRING "2.0"
-
-#define NETEQ_DELAY_LOGGING_SIGNAL_RECIN 1
-#define NETEQ_DELAY_LOGGING_SIGNAL_FLUSH 2
-#define NETEQ_DELAY_LOGGING_SIGNAL_CLOCK 3
-#define NETEQ_DELAY_LOGGING_SIGNAL_EOF 4
-#define NETEQ_DELAY_LOGGING_SIGNAL_DECODE 5
-#define NETEQ_DELAY_LOGGING_SIGNAL_CHANGE_FS 6
-#define NETEQ_DELAY_LOGGING_SIGNAL_MERGE_INFO 7
-#define NETEQ_DELAY_LOGGING_SIGNAL_EXPAND_INFO 8
-#define NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO 9
-#define NETEQ_DELAY_LOGGING_SIGNAL_PREEMPTIVE_INFO 10
-#define NETEQ_DELAY_LOGGING_SIGNAL_OPTBUF 11
-#define NETEQ_DELAY_LOGGING_SIGNAL_DECODE_ONE_DESC 12
-
-#endif
diff --git a/trunk/src/modules/audio_coding/neteq/dsp.c b/trunk/src/modules/audio_coding/neteq/dsp.c
deleted file mode 100644
index 96f07d2..0000000
--- a/trunk/src/modules/audio_coding/neteq/dsp.c
+++ /dev/null
@@ -1,522 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains some DSP initialization functions and
- * constant table definitions.
- */
-
-#include "dsp.h"
-
-#include "signal_processing_library.h"
-
-#include "neteq_error_codes.h"
-
-/* Filter coefficients used when downsampling from the indicated
- sample rates (8, 16, 32, 48 kHz) to 4 kHz.
- Coefficients are in Q12. */
-
-/* {0.3, 0.4, 0.3} */
-const WebRtc_Word16 WebRtcNetEQ_kDownsample8kHzTbl[] = { 1229, 1638, 1229 };
-
-#ifdef NETEQ_WIDEBAND
-/* {0.15, 0.2, 0.3, 0.2, 0.15} */
-const WebRtc_Word16 WebRtcNetEQ_kDownsample16kHzTbl[] =
-{ 614, 819, 1229, 819, 614};
-#endif
-
-#ifdef NETEQ_32KHZ_WIDEBAND
-/* {0.1425, 0.1251, 0.1525, 0.1628, 0.1525, 0.1251, 0.1425} */
-const WebRtc_Word16 WebRtcNetEQ_kDownsample32kHzTbl[] =
-{ 584, 512, 625, 667, 625, 512, 584};
-#endif
-
-#ifdef NETEQ_48KHZ_WIDEBAND
-/* {0.2487, 0.0952, 0.1042, 0.1074, 0.1042, 0.0952, 0.2487} */
-const WebRtc_Word16 WebRtcNetEQ_kDownsample48kHzTbl[] =
-{ 1019, 390, 427, 440, 427, 390, 1019};
-#endif
-
-/* Constants used in expand function WebRtcNetEQ_Expand */
-
-/* Q12: -1.264421 + 4.8659148*x - 4.0092827*x^2 + 1.4100529*x^3 */
-const WebRtc_Word16 WebRtcNetEQ_kMixFractionFuncTbl[4] = { -5179, 19931, -16422, 5776 };
-
-/* Tabulated divisions to save complexity */
-/* 1049/{0, .., 6} */
-const WebRtc_Word16 WebRtcNetEQ_k1049div[7] = { 0, 1049, 524, 349, 262, 209, 174 };
-
-/* 2097/{0, .., 6} */
-const WebRtc_Word16 WebRtcNetEQ_k2097div[7] = { 0, 2097, 1048, 699, 524, 419, 349 };
-
-/* 5243/{0, .., 6} */
-const WebRtc_Word16 WebRtcNetEQ_k5243div[7] = { 0, 5243, 2621, 1747, 1310, 1048, 873 };
-
-#ifdef WEBRTC_NETEQ_40BITACC_TEST
-/*
- * Run NetEQ with simulated 40-bit accumulator to run bit-exact to a DSP
- * implementation where the main (spl and NetEQ) functions have been
- * 40-bit optimized. For testing purposes.
- */
-
-/****************************************************************************
- * WebRtcNetEQ_40BitAccCrossCorr(...)
- *
- * Calculates the Cross correlation between two sequences seq1 and seq2. Seq1
- * is fixed and seq2 slides as the pointer is increased with step
- *
- * Input:
- * - seq1 : First sequence (fixed throughout the correlation)
- * - seq2 : Second sequence (slided step_seq2 for each
- * new correlation)
- * - dimSeq : Number of samples to use in the cross correlation.
- * Should be no larger than 1024 to avoid overflow.
- * - dimCrossCorr : Number of CrossCorrelations to calculate (start
- * position for seq2 is updated for each new one)
- * - rShift : Number of right shifts to use
- * - step_seq2 : How many (positive or negative) steps the seq2
- * pointer should be updated for each new cross
- * correlation value
- *
- * Output:
- * - crossCorr : The cross correlation in Q-rShift
- */
-
-void WebRtcNetEQ_40BitAccCrossCorr(WebRtc_Word32 *crossCorr,
- WebRtc_Word16 *seq1,
- WebRtc_Word16 *seq2,
- WebRtc_Word16 dimSeq,
- WebRtc_Word16 dimCrossCorr,
- WebRtc_Word16 rShift,
- WebRtc_Word16 step_seq2)
-{
- int i, j;
- WebRtc_Word16 *seq1Ptr, *seq2Ptr;
- WebRtc_Word64 acc;
-
- for (i = 0; i < dimCrossCorr; i++)
- {
- /* Set the pointer to the static vector, set the pointer to
- the sliding vector and initialize crossCorr */
- seq1Ptr = seq1;
- seq2Ptr = seq2 + (step_seq2 * i);
- acc = 0;
-
- /* Perform the cross correlation */
- for (j = 0; j < dimSeq; j++)
- {
- acc += WEBRTC_SPL_MUL_16_16((*seq1Ptr), (*seq2Ptr));
- seq1Ptr++;
- seq2Ptr++;
- }
-
- (*crossCorr) = (WebRtc_Word32) (acc >> rShift);
- crossCorr++;
- }
-}
-
-/****************************************************************************
- * WebRtcNetEQ_40BitAccDotW16W16(...)
- *
- * Calculates the dot product between two vectors (WebRtc_Word16)
- *
- * Input:
- * - vector1 : Vector 1
- * - vector2 : Vector 2
- * - len : Number of samples in vector
- * Should be no larger than 1024 to avoid overflow.
- * - scaling : The number of left shifts required to avoid overflow
- * in the dot product
- * Return value : The dot product
- */
-
-WebRtc_Word32 WebRtcNetEQ_40BitAccDotW16W16(WebRtc_Word16 *vector1,
- WebRtc_Word16 *vector2,
- int len,
- int scaling)
-{
- WebRtc_Word32 sum;
- int i;
- WebRtc_Word64 acc;
-
- acc = 0;
- for (i = 0; i < len; i++)
- {
- acc += WEBRTC_SPL_MUL_16_16(*vector1++, *vector2++);
- }
-
- sum = (WebRtc_Word32) (acc >> scaling);
-
- return(sum);
-}
-
-#endif /* WEBRTC_NETEQ_40BITACC_TEST */
-
-/****************************************************************************
- * WebRtcNetEQ_DSPInit(...)
- *
- * Initializes DSP side of NetEQ.
- *
- * Input:
- * - inst : NetEq DSP instance
- * - fs : Initial sample rate (may change when decoding data)
- *
- * Output:
- * - inst : Updated instance
- *
- * Return value : 0 - ok
- * : non-zero - error
- */
-
-int WebRtcNetEQ_DSPInit(DSPInst_t *inst, WebRtc_UWord16 fs)
-{
-
- int res = 0;
- WebRtc_Word16 fs_mult;
-
- /* Pointers and values to save before clearing the instance */
-#ifdef NETEQ_CNG_CODEC
- void *savedPtr1 = inst->CNG_Codec_inst;
-#endif
- void *savedPtr2 = inst->pw16_readAddress;
- void *savedPtr3 = inst->pw16_writeAddress;
- void *savedPtr4 = inst->main_inst;
-#ifdef NETEQ_VAD
- void *savedVADptr = inst->VADInst.VADState;
- VADInitFunction savedVADinit = inst->VADInst.initFunction;
- VADSetmodeFunction savedVADsetmode = inst->VADInst.setmodeFunction;
- VADFunction savedVADfunc = inst->VADInst.VADFunction;
- WebRtc_Word16 savedVADEnabled = inst->VADInst.VADEnabled;
- WebRtc_Word16 savedVADMode = inst->VADInst.VADMode;
-#endif /* NETEQ_VAD */
- DSPStats_t saveStats;
- WebRtc_Word16 saveMsPerCall = inst->millisecondsPerCall;
- enum BGNMode saveBgnMode = inst->BGNInst.bgnMode;
-#ifdef NETEQ_STEREO
- MasterSlaveInfo saveMSinfo;
-#endif
-
- /* copy contents of statInst to avoid clearing */WEBRTC_SPL_MEMCPY_W16(&saveStats, &(inst->statInst),
- sizeof(DSPStats_t)/sizeof(WebRtc_Word16));
-
-#ifdef NETEQ_STEREO
- /* copy contents of msInfo to avoid clearing */WEBRTC_SPL_MEMCPY_W16(&saveMSinfo, &(inst->msInfo),
- sizeof(MasterSlaveInfo)/sizeof(WebRtc_Word16));
-#endif
-
- /* check that the sample rate is valid */
- if ((fs != 8000)
-#ifdef NETEQ_WIDEBAND
- &&(fs!=16000)
-#endif
-#ifdef NETEQ_32KHZ_WIDEBAND
- &&(fs!=32000)
-#endif
-#ifdef NETEQ_48KHZ_WIDEBAND
- &&(fs!=48000)
-#endif
- )
- {
- /* invalid rate */
- return (CODEC_DB_UNSUPPORTED_FS);
- }
-
- /* calcualte fs/8000 */
- fs_mult = WebRtcSpl_DivW32W16ResW16(fs, 8000);
-
- /* Set everything to zero since most variables should be zero at start */
- WebRtcSpl_MemSetW16((WebRtc_Word16 *) inst, 0, sizeof(DSPInst_t) / sizeof(WebRtc_Word16));
-
- /* Restore saved pointers */
-#ifdef NETEQ_CNG_CODEC
- inst->CNG_Codec_inst = (CNG_dec_inst *)savedPtr1;
-#endif
- inst->pw16_readAddress = (WebRtc_Word16 *) savedPtr2;
- inst->pw16_writeAddress = (WebRtc_Word16 *) savedPtr3;
- inst->main_inst = savedPtr4;
-#ifdef NETEQ_VAD
- inst->VADInst.VADState = savedVADptr;
- inst->VADInst.initFunction = savedVADinit;
- inst->VADInst.setmodeFunction = savedVADsetmode;
- inst->VADInst.VADFunction = savedVADfunc;
- inst->VADInst.VADEnabled = savedVADEnabled;
- inst->VADInst.VADMode = savedVADMode;
-#endif /* NETEQ_VAD */
-
- /* Initialize main part */
- inst->fs = fs;
- inst->millisecondsPerCall = saveMsPerCall;
- inst->timestampsPerCall = inst->millisecondsPerCall * 8 * fs_mult;
- inst->ExpandInst.w16_overlap = 5 * fs_mult;
- inst->endPosition = 565 * fs_mult;
- inst->curPosition = inst->endPosition - inst->ExpandInst.w16_overlap;
- inst->w16_seedInc = 1;
- inst->uw16_seed = 777;
- inst->w16_muteFactor = 16384; /* 1.0 in Q14 */
- inst->w16_frameLen = 3 * inst->timestampsPerCall; /* Dummy initialize to 30ms */
-
- inst->w16_speechHistoryLen = 256 * fs_mult;
- inst->pw16_speechHistory = &inst->speechBuffer[inst->endPosition
- - inst->w16_speechHistoryLen];
- inst->ExpandInst.pw16_overlapVec = &(inst->pw16_speechHistory[inst->w16_speechHistoryLen
- - inst->ExpandInst.w16_overlap]);
-
- /* Reusage of memory in speechBuffer inside Expand */
- inst->ExpandInst.pw16_expVecs[0] = &inst->speechBuffer[0];
- inst->ExpandInst.pw16_expVecs[1] = &inst->speechBuffer[126 * fs_mult];
- inst->ExpandInst.pw16_arState = &inst->speechBuffer[2 * 126 * fs_mult];
- inst->ExpandInst.pw16_arFilter = &inst->speechBuffer[2 * 126 * fs_mult
- + UNVOICED_LPC_ORDER];
- /* Ends at 2*126*fs_mult+UNVOICED_LPC_ORDER+(UNVOICED_LPC_ORDER+1) */
-
- inst->ExpandInst.w16_expandMuteFactor = 16384; /* 1.0 in Q14 */
-
- /* Initialize BGN part */
- inst->BGNInst.pw16_filter[0] = 4096;
- inst->BGNInst.w16_scale = 20000;
- inst->BGNInst.w16_scaleShift = 24;
- inst->BGNInst.w32_energyUpdate = 500000;
- inst->BGNInst.w32_energyUpdateLow = 0;
- inst->BGNInst.w32_energy = 2500;
- inst->BGNInst.w16_initialized = 0;
- inst->BGNInst.bgnMode = saveBgnMode;
-
- /* Recreate statistics counters */WEBRTC_SPL_MEMCPY_W16(&(inst->statInst), &saveStats,
- sizeof(DSPStats_t)/sizeof(WebRtc_Word16));
-
-#ifdef NETEQ_STEREO
- /* Recreate MSinfo */WEBRTC_SPL_MEMCPY_W16(&(inst->msInfo), &saveMSinfo,
- sizeof(MasterSlaveInfo)/sizeof(WebRtc_Word16));
-#endif
-
-#ifdef NETEQ_CNG_CODEC
- if (inst->CNG_Codec_inst!=NULL)
- {
- /* initialize comfort noise generator */
- res |= WebRtcCng_InitDec(inst->CNG_Codec_inst);
- }
-#endif
-
-#ifdef NETEQ_VAD
- /* initialize PostDecode VAD instance
- (don't bother checking for NULL instance, this is done inside init function) */
- res |= WebRtcNetEQ_InitVAD(&inst->VADInst, fs);
-#endif /* NETEQ_VAD */
-
- return (res);
-}
-
-/****************************************************************************
- * WebRtcNetEQ_AddressInit(...)
- *
- * Initializes the shared-memory communication on the DSP side.
- *
- * Input:
- * - inst : NetEQ DSP instance
- * - data2McuAddress : Pointer to memory where DSP writes / MCU reads
- * - data2DspAddress : Pointer to memory where MCU writes / DSP reads
- * - mainInst : NetEQ main instance
- *
- * Output:
- * - inst : Updated instance
- *
- * Return value : 0 - ok
- */
-
-int WebRtcNetEQ_AddressInit(DSPInst_t *inst, const void *data2McuAddress,
- const void *data2DspAddress, const void *mainInst)
-{
-
- /* set shared-memory addresses in the DSP instance */
- inst->pw16_readAddress = (WebRtc_Word16 *) data2DspAddress;
- inst->pw16_writeAddress = (WebRtc_Word16 *) data2McuAddress;
-
- /* set pointer to main NetEQ instance */
- inst->main_inst = (void *) mainInst;
-
- /* set output frame size to 10 ms = 80 samples in narrowband */
- inst->millisecondsPerCall = 10;
- inst->timestampsPerCall = 80;
-
- return (0);
-
-}
-
-/****************************************************************************
- * NETEQDSP_clearInCallStats(...)
- *
- * Reset in-call statistics variables on DSP side.
- *
- * Input:
- * - inst : NetEQ DSP instance
- *
- * Output:
- * - inst : Updated instance
- *
- * Return value : 0 - ok
- */
-
-int WebRtcNetEQ_ClearInCallStats(DSPInst_t *inst)
-{
-
- /* Reset statistics counters */
- inst->statInst.accelerateLength = 0;
- inst->statInst.expandLength = 0;
- inst->statInst.preemptiveLength = 0;
-
- return (0);
-}
-
-/****************************************************************************
- * WebRtcNetEQ_ClearPostCallStats(...)
- *
- * Reset post-call statistics variables on DSP side.
- *
- * Input:
- * - inst : NetEQ DSP instance
- *
- * Output:
- * - inst : Updated instance
- *
- * Return value : 0 - ok
- */
-
-int WebRtcNetEQ_ClearPostCallStats(DSPInst_t *inst)
-{
-
- /* Reset statistics counters */
- inst->statInst.expandedVoiceSamples = 0;
- inst->statInst.expandedNoiseSamples = 0;
-
- return (0);
-}
-
-#ifdef NETEQ_VAD
-
-/****************************************************************************
- * WebRtcNetEQ_InitVAD(...)
- *
- * Initializes post-decode VAD instance.
- *
- * Input:
- * - VADinst : PostDecodeVAD instance
- * - fs : Initial sample rate
- *
- * Output:
- * - VADinst : Updated instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_InitVAD(PostDecodeVAD_t *VADInst, WebRtc_UWord16 fs)
-{
-
- int res = 0;
-
- /* initially, disable the post-decode VAD */
- VADInst->VADEnabled = 0;
-
- if (VADInst->VADState != NULL /* if VAD state is provided */
- && VADInst->initFunction != NULL /* and all function ... */
- && VADInst->setmodeFunction != NULL /* ... pointers ... */
- && VADInst->VADFunction != NULL) /* ... are defined */
- {
- res = VADInst->initFunction( VADInst->VADState ); /* call VAD init function */
- res |= WebRtcNetEQ_SetVADModeInternal( VADInst, VADInst->VADMode );
-
- if (res!=0)
- {
- /* something is wrong; play it safe and set the VADstate to NULL */
- VADInst->VADState = NULL;
- }
- else if (fs<=16000)
- {
- /* enable VAD if NB or WB (VAD cannot handle SWB) */
- VADInst->VADEnabled = 1;
- }
- }
-
- /* reset SID/CNG interval counter */
- VADInst->SIDintervalCounter = 0;
-
- /* initialize with active-speaker decision */
- VADInst->VADDecision = 1;
-
- return(res);
-
-}
-
-/****************************************************************************
- * WebRtcNetEQ_SetVADModeInternal(...)
- *
- * Set the VAD mode in the VAD struct, and communicate it to the VAD instance
- * if it exists.
- *
- * Input:
- * - VADinst : PostDecodeVAD instance
- * - mode : Mode number passed on to the VAD function
- *
- * Output:
- * - VADinst : Updated instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_SetVADModeInternal(PostDecodeVAD_t *VADInst, int mode)
-{
-
- int res = 0;
-
- VADInst->VADMode = mode;
-
- if (VADInst->VADState != NULL)
- {
- /* call setmode function */
- res = VADInst->setmodeFunction(VADInst->VADState, mode);
- }
-
- return(res);
-
-}
-
-#endif /* NETEQ_VAD */
-
-/****************************************************************************
- * WebRtcNetEQ_FlushSpeechBuffer(...)
- *
- * Flush the speech buffer.
- *
- * Input:
- * - inst : NetEq DSP instance
- *
- * Output:
- * - inst : Updated instance
- *
- * Return value : 0 - ok
- * : non-zero - error
- */
-
-int WebRtcNetEQ_FlushSpeechBuffer(DSPInst_t *inst)
-{
- WebRtc_Word16 fs_mult;
-
- /* calcualte fs/8000 */
- fs_mult = WebRtcSpl_DivW32W16ResW16(inst->fs, 8000);
-
- /* clear buffer */
- WebRtcSpl_MemSetW16(inst->speechBuffer, 0, SPEECH_BUF_SIZE);
- inst->endPosition = 565 * fs_mult;
- inst->curPosition = inst->endPosition - inst->ExpandInst.w16_overlap;
-
- return 0;
-}
-
diff --git a/trunk/src/modules/audio_coding/neteq/dsp.h b/trunk/src/modules/audio_coding/neteq/dsp.h
deleted file mode 100644
index 94213e4..0000000
--- a/trunk/src/modules/audio_coding/neteq/dsp.h
+++ /dev/null
@@ -1,788 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains some DSP initialization functions,
- * constant table definitions and other parameters.
- * Also contains definitions of all DSP-side data structures.
- */
-
-
-#ifndef DSP_H
-#define DSP_H
-
-#include "typedefs.h"
-
-#include "webrtc_cng.h"
-
-#include "codec_db_defines.h"
-#include "neteq_defines.h"
-#include "neteq_statistics.h"
-
-#ifdef NETEQ_ATEVENT_DECODE
-#include "dtmf_tonegen.h"
-#endif
-
-
-
-/*****************************/
-/* Pre-processor definitions */
-/*****************************/
-
-/* FSMULT is the sample rate divided by 8000 */
-#if defined(NETEQ_48KHZ_WIDEBAND)
- #define FSMULT 6
-#elif defined(NETEQ_32KHZ_WIDEBAND)
- #define FSMULT 4
-#elif defined(NETEQ_WIDEBAND)
- #define FSMULT 2
-#else
- #define FSMULT 1
-#endif
-
-/* Size of the speech buffer (or synchronization buffer). */
-/* 60 ms decoding + 10 ms syncbuff + 0.625ms lookahead */
-#define SPEECH_BUF_SIZE (565 * FSMULT)
-
-/* Misc definitions */
-#define BGN_LPC_ORDER (4 + FSMULT) /* 5, 6, 8, or 10 */
-#define UNVOICED_LPC_ORDER 6
-#define RANDVEC_NO_OF_SAMPLES 256
-
-/* Number of milliseconds to remove/add during accelerate/pre-emptive expand
- under BGNonly operation */
-#define DEFAULT_TIME_ADJUST 8
-
-/* Number of RecOut calls without CNG/SID before re-enabling post-decode VAD */
-#define POST_DECODE_VAD_AUTO_ENABLE 3000
-
-/* 8kHz windowing in Q15 (over 5 samples) */
-#define NETEQ_OVERLAP_WINMUTE_8KHZ_START 27307
-#define NETEQ_OVERLAP_WINMUTE_8KHZ_INC -5461
-#define NETEQ_OVERLAP_WINUNMUTE_8KHZ_START 5461
-#define NETEQ_OVERLAP_WINUNMUTE_8KHZ_INC 5461
-/* 16kHz windowing in Q15 (over 10 samples) */
-#define NETEQ_OVERLAP_WINMUTE_16KHZ_START 29789
-#define NETEQ_OVERLAP_WINMUTE_16KHZ_INC -2979
-#define NETEQ_OVERLAP_WINUNMUTE_16KHZ_START 2979
-#define NETEQ_OVERLAP_WINUNMUTE_16KHZ_INC 2979
-/* 32kHz windowing in Q15 (over 20 samples) */
-#define NETEQ_OVERLAP_WINMUTE_32KHZ_START 31208
-#define NETEQ_OVERLAP_WINMUTE_32KHZ_INC -1560
-#define NETEQ_OVERLAP_WINUNMUTE_32KHZ_START 1560
-#define NETEQ_OVERLAP_WINUNMUTE_32KHZ_INC 1560
-/* 48kHz windowing in Q15 (over 30 samples) */
-#define NETEQ_OVERLAP_WINMUTE_48KHZ_START 31711
-#define NETEQ_OVERLAP_WINMUTE_48KHZ_INC -1057
-#define NETEQ_OVERLAP_WINUNMUTE_48KHZ_START 1057
-#define NETEQ_OVERLAP_WINUNMUTE_48KHZ_INC 1057
-
-/* Fade BGN towards zero after this many Expand calls */
-#define FADE_BGN_TIME 200
-
-
-/*******************/
-/* Constant tables */
-/*******************/
-
-extern const WebRtc_Word16 WebRtcNetEQ_kDownsample8kHzTbl[];
-extern const WebRtc_Word16 WebRtcNetEQ_kDownsample16kHzTbl[];
-extern const WebRtc_Word16 WebRtcNetEQ_kDownsample32kHzTbl[];
-extern const WebRtc_Word16 WebRtcNetEQ_kDownsample48kHzTbl[];
-extern const WebRtc_Word16 WebRtcNetEQ_kRandnTbl[];
-extern const WebRtc_Word16 WebRtcNetEQ_kMixFractionFuncTbl[];
-extern const WebRtc_Word16 WebRtcNetEQ_k1049div[];
-extern const WebRtc_Word16 WebRtcNetEQ_k2097div[];
-extern const WebRtc_Word16 WebRtcNetEQ_k5243div[];
-
-
-
-/************/
-/* Typedefs */
-/************/
-
-enum BGNMode
-{
- BGN_ON, /* default "normal" behavior with eternal noise */
- BGN_FADE, /* noise fades to zero after some time */
- BGN_OFF /* background noise is always zero */
-};
-
-#ifdef NETEQ_STEREO
-enum MasterSlaveMode
-{
- NETEQ_MONO, /* stand-alone instance */
- NETEQ_MASTER, /* master instance in a spatial/stereo configuration */
- NETEQ_SLAVE /* slave instance in a spatial/stereo configuration */
-};
-
-enum MasterSlaveExtraInfo
-{
- NO_INFO, /* no info to convey */
- ACC_FAIL, /* signal that accelerate failed */
- PE_EXP_FAIL, /* signal that pre-emptive expand failed */
- DTMF_OVERDUB, /* signal that DTMF overdub is generated */
- DTMF_ONLY /* signal that DTMF only is played */
-};
-#endif
-
-/****************************/
-/* DSP-side data structures */
-/****************************/
-
-/* Background noise (BGN) instance for storing BGN parameters
- (sub-instance of NETEQDSP_inst) */
-typedef struct BGNInst_t_
-{
-
- WebRtc_Word32 w32_energy;
- WebRtc_Word32 w32_energyMax;
- WebRtc_Word32 w32_energyUpdate;
- WebRtc_Word32 w32_energyUpdateLow;
- WebRtc_Word16 pw16_filterState[BGN_LPC_ORDER];
- WebRtc_Word16 pw16_filter[BGN_LPC_ORDER + 1];
- WebRtc_Word16 w16_mutefactor;
- WebRtc_Word16 w16_scale;
- WebRtc_Word16 w16_scaleShift;
- WebRtc_Word16 w16_initialized;
- enum BGNMode bgnMode;
-
-} BGNInst_t;
-
-/* Expansion instance (sub-instance of NETEQDSP_inst) */
-typedef struct ExpandInst_t_
-{
-
- WebRtc_Word16 w16_overlap; /* Constant, 5 for NB and 10 for WB */
- WebRtc_Word16 w16_consecExp; /* Number of consecutive expand calls */
- WebRtc_Word16 *pw16_arFilter; /* length [UNVOICED_LPC_ORDER+1] */
- WebRtc_Word16 *pw16_arState; /* length [UNVOICED_LPC_ORDER] */
- WebRtc_Word16 w16_arGain;
- WebRtc_Word16 w16_arGainScale;
- WebRtc_Word16 w16_vFraction; /* Q14 */
- WebRtc_Word16 w16_currentVFraction; /* Q14 */
- WebRtc_Word16 *pw16_expVecs[2];
- WebRtc_Word16 w16_lags[3];
- WebRtc_Word16 w16_maxLag;
- WebRtc_Word16 *pw16_overlapVec; /* last samples of speech history */
- WebRtc_Word16 w16_lagsDirection;
- WebRtc_Word16 w16_lagsPosition;
- WebRtc_Word16 w16_expandMuteFactor; /* Q14 */
- WebRtc_Word16 w16_stopMuting;
- WebRtc_Word16 w16_onset;
- WebRtc_Word16 w16_muteSlope; /* Q20 */
-
-} ExpandInst_t;
-
-#ifdef NETEQ_VAD
-
-/*
- * VAD function pointer types, replicating the typedefs in webrtc_neteq_internal.h.
- * These function pointers match the definitions of WebRtc VAD functions WebRtcVad_Init,
- * WebRtcVad_set_mode and WebRtcVad_Process, respectively, all found in webrtc_vad.h.
- */
-typedef int (*VADInitFunction)(void *VAD_inst);
-typedef int (*VADSetmodeFunction)(void *VAD_inst, int mode);
-typedef WebRtc_Word16 (*VADFunction)(void *VAD_inst, WebRtc_Word16 fs, WebRtc_Word16 *frame,
- WebRtc_Word16 frameLen);
-
-/* Post-decode VAD instance (sub-instance of NETEQDSP_inst) */
-typedef struct PostDecodeVAD_t_
-{
-
- void *VADState; /* pointer to a VAD instance */
-
- WebRtc_Word16 VADEnabled; /* 1 if enabled, 0 if disabled */
- int VADMode; /* mode parameter to pass to the VAD function */
- WebRtc_Word16 VADDecision; /* 1 for active, 0 for passive */
- WebRtc_Word16 SIDintervalCounter; /* reset when decoding CNG/SID frame,
- increment for each recout call */
-
- /* Function pointers */
- VADInitFunction initFunction; /* VAD init function */
- VADSetmodeFunction setmodeFunction; /* VAD setmode function */
- VADFunction VADFunction; /* VAD function */
-
-} PostDecodeVAD_t;
-
-#endif /* NETEQ_VAD */
-
-#ifdef NETEQ_STEREO
-#define MAX_MS_DECODES 10
-
-typedef struct
-{
- /* Stand-alone, master, or slave */
- enum MasterSlaveMode msMode;
-
- enum MasterSlaveExtraInfo extraInfo;
-
- WebRtc_UWord16 instruction;
- WebRtc_Word16 distLag;
- WebRtc_Word16 corrLag;
- WebRtc_Word16 bestIndex;
-
- WebRtc_UWord32 endTimestamp;
- WebRtc_UWord16 samplesLeftWithOverlap;
-
-} MasterSlaveInfo;
-#endif
-
-
-/* "Main" NetEQ DSP instance */
-typedef struct DSPInst_t_
-{
-
- /* MCU/DSP Communication layer */
- WebRtc_Word16 *pw16_readAddress;
- WebRtc_Word16 *pw16_writeAddress;
- void *main_inst;
-
- /* Output frame size in ms and samples */
- WebRtc_Word16 millisecondsPerCall;
- WebRtc_Word16 timestampsPerCall;
-
- /*
- * Example of speech buffer
- *
- * -----------------------------------------------------------
- * | History T-60 to T | Future |
- * -----------------------------------------------------------
- * ^ ^
- * | |
- * curPosition endPosition
- *
- * History is gradually shifted out to the left when inserting
- * new data at the end.
- */
-
- WebRtc_Word16 speechBuffer[SPEECH_BUF_SIZE]; /* History/future speech buffer */
- int curPosition; /* Next sample to play */
- int endPosition; /* Position that ends future data */
- WebRtc_UWord32 endTimestamp; /* Timestamp value at end of future data */
- WebRtc_UWord32 videoSyncTimestamp; /* (Estimated) timestamp of the last
- played sample (usually same as
- endTimestamp-(endPosition-curPosition)
- except during Expand and CNG) */
- WebRtc_UWord16 fs; /* sample rate in Hz */
- WebRtc_Word16 w16_frameLen; /* decoder frame length in samples */
- WebRtc_Word16 w16_mode; /* operation used during last RecOut call */
- WebRtc_Word16 w16_muteFactor; /* speech mute factor in Q14 */
- WebRtc_Word16 *pw16_speechHistory; /* beginning of speech history during Expand */
- WebRtc_Word16 w16_speechHistoryLen; /* 256 for NB and 512 for WB */
-
- /* random noise seed parameters */
- WebRtc_Word16 w16_seedInc;
- WebRtc_UWord32 uw16_seed;
-
- /* VQmon related variable */
- WebRtc_Word16 w16_concealedTS;
-
- /*****************/
- /* Sub-instances */
- /*****************/
-
- /* Decoder data */
- CodecFuncInst_t codec_ptr_inst;
-
-#ifdef NETEQ_CNG_CODEC
- /* CNG "decoder" instance */
- CNG_dec_inst *CNG_Codec_inst;
-#endif /* NETEQ_CNG_CODEC */
-
-#ifdef NETEQ_ATEVENT_DECODE
- /* DTMF generator instance */
- dtmf_tone_inst_t DTMFInst;
-#endif /* NETEQ_CNG_CODEC */
-
-#ifdef NETEQ_VAD
- /* Post-decode VAD instance */
- PostDecodeVAD_t VADInst;
-#endif /* NETEQ_VAD */
-
- /* Expand instance (defined above) */
- ExpandInst_t ExpandInst;
-
- /* Background noise instance (defined above) */
- BGNInst_t BGNInst;
-
- /* Internal statistics instance */
- DSPStats_t statInst;
-
-#ifdef NETEQ_STEREO
- /* Pointer to Master/Slave info */
- MasterSlaveInfo *msInfo;
-#endif
-
-} DSPInst_t;
-
-
-/*************************/
-/* Function declarations */
-/*************************/
-
-/****************************************************************************
- * WebRtcNetEQ_DSPInit(...)
- *
- * Initializes DSP side of NetEQ.
- *
- * Input:
- * - inst : NetEq DSP instance
- * - fs : Initial sample rate (may change when decoding data)
- *
- * Output:
- * - inst : Updated instance
- *
- * Return value : 0 - ok
- * : non-zero - error
- */
-
-int WebRtcNetEQ_DSPInit(DSPInst_t *inst, WebRtc_UWord16 fs);
-
-/****************************************************************************
- * WebRtcNetEQ_AddressInit(...)
- *
- * Initializes the shared-memory communication on the DSP side.
- *
- * Input:
- * - inst : NetEQ DSP instance
- * - data2McuAddress : Pointer to memory where DSP writes / MCU reads
- * - data2DspAddress : Pointer to memory where MCU writes / DSP reads
- * - mainInst : NetEQ main instance
- *
- * Output:
- * - inst : Updated instance
- *
- * Return value : 0 - ok
- */
-
-int WebRtcNetEQ_AddressInit(DSPInst_t *inst, const void *data2McuAddress,
- const void *data2DspAddress, const void *mainInst);
-
-/****************************************************************************
- * WebRtcNetEQ_ClearInCallStats(...)
- *
- * Reset in-call statistics variables on DSP side.
- *
- * Input:
- * - inst : NetEQ DSP instance
- *
- * Output:
- * - inst : Updated instance
- *
- * Return value : 0 - ok
- */
-
-int WebRtcNetEQ_ClearInCallStats(DSPInst_t *inst);
-
-/****************************************************************************
- * WebRtcNetEQ_ClearPostCallStats(...)
- *
- * Reset post-call statistics variables on DSP side.
- *
- * Input:
- * - inst : NetEQ DSP instance
- *
- * Output:
- * - inst : Updated instance
- *
- * Return value : 0 - ok
- */
-
-int WebRtcNetEQ_ClearPostCallStats(DSPInst_t *inst);
-
-/****************************************************************************
- * WebRtcNetEQ_RecOutInternal(...)
- *
- * This function asks NetEQ for more speech/audio data.
- *
- * Input:
- * - inst : NetEQ instance, i.e. the user that requests more
- * speech/audio data.
- * - outdata : Pointer to a memory space where the output data
- * should be stored.
- * - BGNonly : If non-zero, RecOut will only produce background
- * noise. It will still draw packets from the packet
- * buffer, but they will never be decoded.
- *
- * Output:
- * - inst : Updated user information
- * - len : Number of samples that were outputted from NetEq
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
- WebRtc_Word16 BGNonly);
-
-/****************************************************************************
- * WebRtcNetEQ_Normal(...)
- *
- * This function has the possibility to modify data that is played out in Normal
- * mode, for example adjust the gain of the signal. The length of the signal
- * can not be changed.
- *
- * Input:
- * - inst : NetEQ DSP instance
- * - scratchPtr : Pointer to scratch vector
- * - decoded : Pointer to vector of new data from decoder
- * - len : Number of input samples
- *
- * Output:
- * - inst : Updated user information
- * - pw16_len : Pointer to varibale where the number of samples
- * produced will be written
- *
- * Return value : >=0 - Number of samples written to outData
- * -1 - Error
- */
-
-int WebRtcNetEQ_Normal(DSPInst_t *inst,
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_scratchPtr,
-#endif
- WebRtc_Word16 *pw16_decoded, WebRtc_Word16 len,
- WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len);
-
-/****************************************************************************
- * WebRtcNetEQ_Expand(...)
- *
- * This function produces one "chunk" of expansion data (PLC audio). The
- * lenght of the produced audio depends on the speech history.
- *
- * Input:
- * - inst : NetEQ DSP instance
- * - scratchPtr : Pointer to scratch vector
- * - BGNonly : If non-zero, Expand will only produce background
- * noise.
- * - pw16_len : Desired number of samples (only for BGN mode).
- *
- * Output:
- * - inst : Updated user information
- * - outdata : Pointer to a memory space where the output data
- * should be stored
- * - pw16_len : Number of samples that were outputted from NetEq
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-
-int WebRtcNetEQ_Expand(DSPInst_t *inst,
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_scratchPtr,
-#endif
- WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
- WebRtc_Word16 BGNonly);
-
-/****************************************************************************
- * WebRtcNetEQ_GenerateBGN(...)
- *
- * This function generates and writes len samples of background noise to the
- * output vector. The Expand function will be called repeteadly until the
- * correct number of samples is produced.
- *
- * Input:
- * - inst : NetEQ DSP instance
- * - scratchPtr : Pointer to scratch vector
- * - len : Desired length of produced BGN.
- *
- *
- * Output:
- * - pw16_outData : Pointer to a memory space where the output data
- * should be stored
- *
- * Return value : >=0 - Number of noise samples produced and written
- * to output
- * -1 - Error
- */
-
-int WebRtcNetEQ_GenerateBGN(DSPInst_t *inst,
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_scratchPtr,
-#endif
- WebRtc_Word16 *pw16_outData, WebRtc_Word16 len);
-
-/****************************************************************************
- * WebRtcNetEQ_PreEmptiveExpand(...)
- *
- * This function tries to extend the audio data by repeating one or several
- * pitch periods. The operation is only carried out if the correlation is
- * strong or if the signal energy is very low. The algorithm is the
- * reciprocal of the Accelerate algorithm.
- *
- * Input:
- * - inst : NetEQ DSP instance
- * - scratchPtr : Pointer to scratch vector.
- * - decoded : Pointer to newly decoded speech.
- * - len : Length of decoded speech.
- * - oldDataLen : Length of the part of decoded that has already been played out.
- * - BGNonly : If non-zero, Pre-emptive Expand will only copy
- * the first DEFAULT_TIME_ADJUST seconds of the
- * input and append to the end. No signal matching is
- * done.
- *
- * Output:
- * - inst : Updated instance
- * - outData : Pointer to a memory space where the output data
- * should be stored. The vector must be at least
- * min(len + 120*fs/8000, NETEQ_MAX_OUTPUT_SIZE)
- * elements long.
- * - pw16_len : Number of samples written to outData.
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-
-int WebRtcNetEQ_PreEmptiveExpand(DSPInst_t *inst,
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_scratchPtr,
-#endif
- const WebRtc_Word16 *pw16_decoded, int len, int oldDataLen,
- WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
- WebRtc_Word16 BGNonly);
-
-/****************************************************************************
- * WebRtcNetEQ_Accelerate(...)
- *
- * This function tries to shorten the audio data by removing one or several
- * pitch periods. The operation is only carried out if the correlation is
- * strong or if the signal energy is very low.
- *
- * Input:
- * - inst : NetEQ DSP instance
- * - scratchPtr : Pointer to scratch vector.
- * - decoded : Pointer to newly decoded speech.
- * - len : Length of decoded speech.
- * - BGNonly : If non-zero, Accelerate will only remove the last
- * DEFAULT_TIME_ADJUST seconds of the intput.
- * No signal matching is done.
- *
- *
- * Output:
- * - inst : Updated instance
- * - outData : Pointer to a memory space where the output data
- * should be stored
- * - pw16_len : Number of samples written to outData.
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-
-int WebRtcNetEQ_Accelerate(DSPInst_t *inst,
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_scratchPtr,
-#endif
- const WebRtc_Word16 *pw16_decoded, int len,
- WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
- WebRtc_Word16 BGNonly);
-
-/****************************************************************************
- * WebRtcNetEQ_Merge(...)
- *
- * This function is used to merge new data from the decoder to the exisiting
- * stream in the synchronization buffer. The merge operation is typically
- * done after a packet loss, where the end of the expanded data does not
- * fit naturally with the new decoded data.
- *
- * Input:
- * - inst : NetEQ DSP instance
- * - scratchPtr : Pointer to scratch vector.
- * - decoded : Pointer to new decoded speech.
- * - len : Number of samples in pw16_decoded.
- *
- *
- * Output:
- * - inst : Updated user information
- * - outData : Pointer to a memory space where the output data
- * should be stored
- * - pw16_len : Number of samples written to pw16_outData
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-
-int WebRtcNetEQ_Merge(DSPInst_t *inst,
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_scratchPtr,
-#endif
- WebRtc_Word16 *pw16_decoded, int len, WebRtc_Word16 *pw16_outData,
- WebRtc_Word16 *pw16_len);
-
-/****************************************************************************
- * WebRtcNetEQ_Cng(...)
- *
- * This function produces CNG according to RFC 3389
- *
- * Input:
- * - inst : NetEQ DSP instance
- * - len : Number of samples to produce
- *
- * Output:
- * - pw16_outData : Output CNG
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-
-#ifdef NETEQ_CNG_CODEC
-/* Must compile NetEQ with CNG support to enable this function */
-
-int WebRtcNetEQ_Cng(DSPInst_t *inst, WebRtc_Word16 *pw16_outData, int len);
-
-#endif /* NETEQ_CNG_CODEC */
-
-/****************************************************************************
- * WebRtcNetEQ_BGNUpdate(...)
- *
- * This function updates the background noise parameter estimates.
- *
- * Input:
- * - inst : NetEQ instance, where the speech history is stored.
- * - scratchPtr : Pointer to scratch vector.
- *
- * Output:
- * - inst : Updated information about the BGN characteristics.
- *
- * Return value : No return value
- */
-
-void WebRtcNetEQ_BGNUpdate(
-#ifdef SCRATCH
- DSPInst_t *inst, WebRtc_Word16 *pw16_scratchPtr
-#else
- DSPInst_t *inst
-#endif
- );
-
-#ifdef NETEQ_VAD
-/* Functions used by post-decode VAD */
-
-/****************************************************************************
- * WebRtcNetEQ_InitVAD(...)
- *
- * Initializes post-decode VAD instance.
- *
- * Input:
- * - VADinst : PostDecodeVAD instance
- * - fs : Initial sample rate
- *
- * Output:
- * - VADinst : Updated instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_InitVAD(PostDecodeVAD_t *VADInst, WebRtc_UWord16 fs);
-
-/****************************************************************************
- * WebRtcNetEQ_SetVADModeInternal(...)
- *
- * Set the VAD mode in the VAD struct, and communicate it to the VAD instance
- * if it exists.
- *
- * Input:
- * - VADinst : PostDecodeVAD instance
- * - mode : Mode number passed on to the VAD function
- *
- * Output:
- * - VADinst : Updated instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_SetVADModeInternal(PostDecodeVAD_t *VADInst, int mode);
-
-#endif /* NETEQ_VAD */
-
-/****************************************************************************
- * WebRtcNetEQ_FlushSpeechBuffer(...)
- *
- * Flush the speech buffer.
- *
- * Input:
- * - inst : NetEq DSP instance
- *
- * Output:
- * - inst : Updated instance
- *
- * Return value : 0 - ok
- * : non-zero - error
- */
-
-int WebRtcNetEQ_FlushSpeechBuffer(DSPInst_t *inst);
-
-#ifndef WEBRTC_NETEQ_40BITACC_TEST
-
-#include "signal_processing_library.h"
-/* Map to regular SPL functions */
-#define WebRtcNetEQ_CrossCorr WebRtcSpl_CrossCorrelation
-#define WebRtcNetEQ_DotW16W16 WebRtcSpl_DotProductWithScale
-
-#else /* WEBRTC_NETEQ_40BITACC_TEST defined */
-/* Run NetEQ with simulated 40-bit accumulator to run bit-exact to a DSP
- implementation where the main (splib and NetEQ) functions have been
- 40-bit optimized. */
-
-/* Map to special 40-bit optimized functions, defined below */
-#define WebRtcNetEQ_CrossCorr WebRtcNetEQ_40BitAccCrossCorr
-#define WebRtcNetEQ_DotW16W16 WebRtcNetEQ_40BitAccDotW16W16
-
-/****************************************************************************
- * WebRtcNetEQ_40BitAccCrossCorr(...)
- *
- * Calculates the Cross correlation between two sequences seq1 and seq2. Seq1
- * is fixed and seq2 slides as the pointer is increased with step
- *
- * Input:
- * - seq1 : First sequence (fixed throughout the correlation)
- * - seq2 : Second sequence (slided step_seq2 for each
- * new correlation)
- * - dimSeq : Number of samples to use in the cross correlation.
- * Should be no larger than 1024 to avoid overflow.
- * - dimCrossCorr : Number of CrossCorrelations to calculate (start
- * position for seq2 is updated for each new one)
- * - rShift : Number of right shifts to use
- * - step_seq2 : How many (positive or negative) steps the seq2
- * pointer should be updated for each new cross
- * correlation value
- *
- * Output:
- * - crossCorr : The cross correlation in Q-rShift
- */
-
-void WebRtcNetEQ_40BitAccCrossCorr(WebRtc_Word32 *crossCorr, WebRtc_Word16 *seq1,
- WebRtc_Word16 *seq2, WebRtc_Word16 dimSeq,
- WebRtc_Word16 dimCrossCorr, WebRtc_Word16 rShift,
- WebRtc_Word16 step_seq2);
-
-/****************************************************************************
- * WebRtcNetEQ_40BitAccDotW16W16(...)
- *
- * Calculates the dot product between two vectors (WebRtc_Word16)
- *
- * Input:
- * - vector1 : Vector 1
- * - vector2 : Vector 2
- * - len : Number of samples in vector
- * Should be no larger than 1024 to avoid overflow.
- * - scaling : The number of right shifts (after multiplication)
- * required to avoid overflow in the dot product.
- * Return value : The dot product
- */
-
-WebRtc_Word32 WebRtcNetEQ_40BitAccDotW16W16(WebRtc_Word16 *vector1, WebRtc_Word16 *vector2,
- int len, int scaling);
-
-#endif /* WEBRTC_NETEQ_40BITACC_TEST */
-
-#endif /* DSP_H */
diff --git a/trunk/src/modules/audio_coding/neteq/dsp_helpfunctions.c b/trunk/src/modules/audio_coding/neteq/dsp_helpfunctions.c
deleted file mode 100644
index 6e9a283..0000000
--- a/trunk/src/modules/audio_coding/neteq/dsp_helpfunctions.c
+++ /dev/null
@@ -1,120 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains some help functions that did not fit elsewhere.
- */
-
-#include "dsp_helpfunctions.h"
-
-
-WebRtc_Word16 WebRtcNetEQ_CalcFsMult(WebRtc_UWord16 fsHz)
-{
- switch (fsHz)
- {
- case 8000:
- {
- return 1;
- }
- case 16000:
- {
- return 2;
- }
- case 32000:
- {
- return 4;
- }
- case 48000:
- {
- return 6;
- }
- default:
- {
- return 1;
- }
- }
-}
-
-
-int WebRtcNetEQ_DownSampleTo4kHz(const WebRtc_Word16 *in, int inLen, WebRtc_UWord16 inFsHz,
- WebRtc_Word16 *out, int outLen, int compensateDelay)
-{
- WebRtc_Word16 *B; /* filter coefficients */
- WebRtc_Word16 Blen; /* number of coefficients */
- WebRtc_Word16 filterDelay; /* phase delay in samples */
- WebRtc_Word16 factor; /* conversion rate (inFsHz/8000) */
- int ok;
-
- /* Set constants depending on frequency used */
- /* NOTE: The phase delay values are wrong compared to the true phase delay
- of the filters. However, the error is preserved (through the +1 term)
- for consistency. */
- switch (inFsHz)
- {
- case 8000:
- {
- Blen = 3;
- factor = 2;
- B = (WebRtc_Word16*) WebRtcNetEQ_kDownsample8kHzTbl;
- filterDelay = 1 + 1;
- break;
- }
-#ifdef NETEQ_WIDEBAND
- case 16000:
- {
- Blen = 5;
- factor = 4;
- B = (WebRtc_Word16*) WebRtcNetEQ_kDownsample16kHzTbl;
- filterDelay = 2 + 1;
- break;
- }
-#endif
-#ifdef NETEQ_32KHZ_WIDEBAND
- case 32000:
- {
- Blen = 7;
- factor = 8;
- B = (WebRtc_Word16*) WebRtcNetEQ_kDownsample32kHzTbl;
- filterDelay = 3 + 1;
- break;
- }
-#endif
-#ifdef NETEQ_48KHZ_WIDEBAND
- case 48000:
- {
- Blen = 7;
- factor = 12;
- B = (WebRtc_Word16*) WebRtcNetEQ_kDownsample48kHzTbl;
- filterDelay = 3 + 1;
- break;
- }
-#endif
- default:
- {
- /* unsupported or wrong sample rate */
- return -1;
- }
- }
-
- if (!compensateDelay)
- {
- /* disregard delay compensation */
- filterDelay = 0;
- }
-
- ok = WebRtcSpl_DownsampleFast((WebRtc_Word16*) &in[Blen - 1],
- (WebRtc_Word16) (inLen - (Blen - 1)), /* number of input samples */
- out, (WebRtc_Word16) outLen, /* number of output samples to produce */
- B, Blen, factor, filterDelay); /* filter parameters */
-
- return ok; /* return value is -1 if input signal is too short */
-
-}
-
diff --git a/trunk/src/modules/audio_coding/neteq/dsp_helpfunctions.h b/trunk/src/modules/audio_coding/neteq/dsp_helpfunctions.h
deleted file mode 100644
index f728c09..0000000
--- a/trunk/src/modules/audio_coding/neteq/dsp_helpfunctions.h
+++ /dev/null
@@ -1,220 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Various help functions used by the DSP functions.
- */
-
-#ifndef DSP_HELPFUNCTIONS_H
-#define DSP_HELPFUNCTIONS_H
-
-#include "typedefs.h"
-
-#include "dsp.h"
-
-/****************************************************************************
- * WebRtcNetEQ_Correlator(...)
- *
- * Calculate signal correlation.
- *
- * Input:
- * - inst : DSP instance
- * - data : Speech history to do expand from (older history in data[-4..-1])
- * - dataLen : Length of data
- *
- * Output:
- * - corrOut : CC of downsampled signal
- * - corrScale : Scale factor for correlation (-Qdomain)
- *
- * Return value : Length of correlated data
- */
-
-WebRtc_Word16 WebRtcNetEQ_Correlator(DSPInst_t *inst,
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_scratchPtr,
-#endif
- WebRtc_Word16 *pw16_data, WebRtc_Word16 w16_dataLen,
- WebRtc_Word16 *pw16_corrOut,
- WebRtc_Word16 *pw16_corrScale);
-
-/****************************************************************************
- * WebRtcNetEQ_PeakDetection(...)
- *
- * Peak detection with parabolic fit.
- *
- * Input:
- * - data : Data sequence for peak detection
- * - dataLen : Length of data
- * - nmbPeaks : Number of peaks to detect
- * - fs_mult : Sample rate multiplier
- *
- * Output:
- * - corrIndex : Index of the peak
- * - winner : Value of the peak
- *
- * Return value : 0 for ok
- */
-
-WebRtc_Word16 WebRtcNetEQ_PeakDetection(WebRtc_Word16 *pw16_data, WebRtc_Word16 w16_dataLen,
- WebRtc_Word16 w16_nmbPeaks, WebRtc_Word16 fs_mult,
- WebRtc_Word16 *pw16_corrIndex,
- WebRtc_Word16 *pw16_winners);
-
-/****************************************************************************
- * WebRtcNetEQ_PrblFit(...)
- *
- * Three-point parbola fit.
- *
- * Input:
- * - 3pts : Three input samples
- * - fs_mult : Sample rate multiplier
- *
- * Output:
- * - Ind : Index of the peak
- * - outVal : Value of the peak
- *
- * Return value : 0 for ok
- */
-
-WebRtc_Word16 WebRtcNetEQ_PrblFit(WebRtc_Word16 *pw16_3pts, WebRtc_Word16 *pw16_Ind,
- WebRtc_Word16 *pw16_outVal, WebRtc_Word16 fs_mult);
-
-/****************************************************************************
- * WebRtcNetEQ_MinDistortion(...)
- *
- * Find the lag that results in minimum distortion.
- *
- * Input:
- * - data : Start of speech to perform distortion on, second vector is assumed
- * to be data[-Lag]
- * - minLag : Start lag
- * - maxLag : End lag
- * - len : Length to correlate
- *
- * Output:
- * - dist : Distorion value
- *
- * Return value : Lag for minimum distortion
- */
-
-WebRtc_Word16 WebRtcNetEQ_MinDistortion(const WebRtc_Word16 *pw16_data,
- WebRtc_Word16 w16_minLag, WebRtc_Word16 w16_maxLag,
- WebRtc_Word16 len, WebRtc_Word32 *pw16_dist);
-
-/****************************************************************************
- * WebRtcNetEQ_RandomVec(...)
- *
- * Generate random vector.
- *
- * Input:
- * - seed : Current seed (input/output)
- * - len : Number of samples to generate
- * - incVal : Jump step
- *
- * Output:
- * - randVec : Generated random vector
- */
-
-void WebRtcNetEQ_RandomVec(WebRtc_UWord32 *w32_seed, WebRtc_Word16 *pw16_randVec,
- WebRtc_Word16 w16_len, WebRtc_Word16 w16_incval);
-
-/****************************************************************************
- * WebRtcNetEQ_MixVoiceUnvoice(...)
- *
- * Mix voiced and unvoiced signal.
- *
- * Input:
- * - voicedVec : Voiced input signal
- * - unvoicedVec : Unvoiced input signal
- * - current_vfraction : Current mixing factor
- * - vfraction_change : Mixing factor change per sample
- * - N : Number of samples
- *
- * Output:
- * - outData : Mixed signal
- */
-
-void WebRtcNetEQ_MixVoiceUnvoice(WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_voicedVec,
- WebRtc_Word16 *pw16_unvoicedVec,
- WebRtc_Word16 *w16_current_vfraction,
- WebRtc_Word16 w16_vfraction_change, WebRtc_Word16 N);
-
-/****************************************************************************
- * WebRtcNetEQ_UnmuteSignal(...)
- *
- * Gradually reduce attenuation.
- *
- * Input:
- * - inVec : Input signal
- * - startMuteFact : Starting attenuation
- * - unmuteFact : Factor to "unmute" with (Q20)
- * - N : Number of samples
- *
- * Output:
- * - outVec : Output signal
- */
-
-void WebRtcNetEQ_UnmuteSignal(WebRtc_Word16 *pw16_inVec, WebRtc_Word16 *startMuteFact,
- WebRtc_Word16 *pw16_outVec, WebRtc_Word16 unmuteFact,
- WebRtc_Word16 N);
-
-/****************************************************************************
- * WebRtcNetEQ_MuteSignal(...)
- *
- * Gradually increase attenuation.
- *
- * Input:
- * - inout : Input/output signal
- * - muteSlope : Slope of muting
- * - N : Number of samples
- */
-
-void WebRtcNetEQ_MuteSignal(WebRtc_Word16 *pw16_inout, WebRtc_Word16 muteSlope,
- WebRtc_Word16 N);
-
-/****************************************************************************
- * WebRtcNetEQ_CalcFsMult(...)
- *
- * Calculate the sample rate divided by 8000.
- *
- * Input:
- * - fsHz : Sample rate in Hz in {8000, 16000, 32000, 48000}.
- *
- * Return value : fsHz/8000 for the valid values, 1 for other inputs
- */
-
-WebRtc_Word16 WebRtcNetEQ_CalcFsMult(WebRtc_UWord16 fsHz);
-
-/****************************************************************************
- * WebRtcNetEQ_DownSampleTo4kHz(...)
- *
- * Lowpass filter and downsample a signal to 4 kHz sample rate.
- *
- * Input:
- * - in : Input signal samples.
- * - inLen : Number of input samples.
- * - inFsHz : Input sample rate in Hz.
- * - outLen : Desired number of samples in decimated signal.
- * - compensateDelay : If non-zero, compensate for the phase delay of
- * of the anti-alias filter.
- *
- * Output:
- * - out : Output signal samples.
- *
- * Return value : 0 - Ok
- * -1 - Error
- *
- */
-
-int WebRtcNetEQ_DownSampleTo4kHz(const WebRtc_Word16 *in, int inLen, WebRtc_UWord16 inFsHz,
- WebRtc_Word16 *out, int outLen, int compensateDelay);
-
-#endif
-
diff --git a/trunk/src/modules/audio_coding/neteq/dtmf_buffer.c b/trunk/src/modules/audio_coding/neteq/dtmf_buffer.c
deleted file mode 100644
index f00f9c9..0000000
--- a/trunk/src/modules/audio_coding/neteq/dtmf_buffer.c
+++ /dev/null
@@ -1,232 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Implementation of packet buffer for DTMF messages.
- */
-
-#include "dtmf_buffer.h"
-
-#include "typedefs.h" /* to define endianness */
-#include "signal_processing_library.h"
-
-#include "neteq_error_codes.h"
-
-
-#ifdef NETEQ_ATEVENT_DECODE
-
-WebRtc_Word16 WebRtcNetEQ_DtmfRemoveEvent(dtmf_inst_t *DTMFdec_inst)
-{
-
- int i;
- for (i = 0; i < 3; i++)
- {
- DTMFdec_inst->EventQueue[i] = DTMFdec_inst->EventQueue[i + 1];
- DTMFdec_inst->EventQueueVolume[i] = DTMFdec_inst->EventQueueVolume[i + 1];
- DTMFdec_inst->EventQueueEnded[i] = DTMFdec_inst->EventQueueEnded[i + 1];
- DTMFdec_inst->EventQueueStartTime[i] = DTMFdec_inst->EventQueueStartTime[i + 1];
- DTMFdec_inst->EventQueueEndTime[i] = DTMFdec_inst->EventQueueEndTime[i + 1];
- }
- DTMFdec_inst->EventBufferSize--;
- DTMFdec_inst->EventQueue[3] = -1;
- DTMFdec_inst->EventQueueVolume[3] = 0;
- DTMFdec_inst->EventQueueEnded[3] = 0;
- DTMFdec_inst->EventQueueStartTime[3] = 0;
- DTMFdec_inst->EventQueueEndTime[3] = 0;
-
- return 0;
-}
-
-WebRtc_Word16 WebRtcNetEQ_DtmfDecoderInit(dtmf_inst_t *DTMFdec_inst, WebRtc_UWord16 fs,
- WebRtc_Word16 MaxPLCtime)
-{
- int i;
- if (((fs != 8000) && (fs != 16000) && (fs != 32000) && (fs != 48000)) || (MaxPLCtime < 0))
- {
- return DTMF_DEC_PARAMETER_ERROR;
- }
- if (fs == 8000)
- DTMFdec_inst->framelen = 80;
- else if (fs == 16000)
- DTMFdec_inst->framelen = 160;
- else if (fs == 32000)
- DTMFdec_inst->framelen = 320;
- else
- /* fs == 48000 */
- DTMFdec_inst->framelen = 480;
-
- DTMFdec_inst->MaxPLCtime = MaxPLCtime;
- DTMFdec_inst->CurrentPLCtime = 0;
- DTMFdec_inst->EventBufferSize = 0;
- for (i = 0; i < 4; i++)
- {
- DTMFdec_inst->EventQueue[i] = -1;
- DTMFdec_inst->EventQueueVolume[i] = 0;
- DTMFdec_inst->EventQueueEnded[i] = 0;
- DTMFdec_inst->EventQueueStartTime[i] = 0;
- DTMFdec_inst->EventQueueEndTime[i] = 0;
- }
- return 0;
-}
-
-WebRtc_Word16 WebRtcNetEQ_DtmfInsertEvent(dtmf_inst_t *DTMFdec_inst,
- const WebRtc_Word16 *encoded, WebRtc_Word16 len,
- WebRtc_UWord32 timeStamp)
-{
-
- int i;
- WebRtc_Word16 value;
- const WebRtc_Word16 *EventStart;
- WebRtc_Word16 endEvent;
- WebRtc_Word16 Volume;
- WebRtc_Word16 Duration;
- WebRtc_Word16 position = -1;
-
- /* Extract event */
- if (len == 4)
- {
- EventStart = encoded;
-#ifdef WEBRTC_BIG_ENDIAN
- value=((*EventStart)>>8);
- endEvent=((*EventStart)&0x80)>>7;
- Volume=((*EventStart)&0x3F);
- Duration=EventStart[1];
-#else
- value = ((*EventStart) & 0xFF);
- endEvent = ((*EventStart) & 0x8000) >> 15;
- Volume = ((*EventStart) & 0x3F00) >> 8;
- Duration = (((((WebRtc_UWord16) EventStart[1]) >> 8) & 0xFF)
- | (((WebRtc_UWord16) (EventStart[1] & 0xFF)) << 8));
-#endif
- /* Only events between 0-15 are supported (DTMF tones) */
- if ((value < 0) || (value > 15))
- {
- return 0;
- }
-
- /* Discard all DTMF tones with really low volume (<-36dbm0) */
- if (Volume > 36)
- {
- return 0;
- }
-
- /*Are there any unended events of the same type? */
- for (i = 0; i < DTMFdec_inst->EventBufferSize; i++)
- {
- /* Going through the whole queue even when we have found a match will
- ensure that we add to the latest applicable event */
- if ((DTMFdec_inst->EventQueue[i] == value) && (!DTMFdec_inst->EventQueueEnded[i]
- || endEvent)) position = i;
- }
- if (position > -1)
- {
- DTMFdec_inst->EventQueueVolume[position] = Volume;
- if ((timeStamp + Duration) > DTMFdec_inst->EventQueueEndTime[position]) DTMFdec_inst->EventQueueEndTime[position]
- = DTMFdec_inst->EventQueueStartTime[position] + Duration;
- if (endEvent) DTMFdec_inst->EventQueueEnded[position] = 1;
- }
- else
- {
- if (DTMFdec_inst->EventBufferSize == MAX_DTMF_QUEUE_SIZE)
- { /* Buffer full */
- /* Remove one event */
- DTMFdec_inst->EventBufferSize--;
- }
- /* Store data in the instance on a new position*/
- DTMFdec_inst->EventQueue[DTMFdec_inst->EventBufferSize] = value;
- DTMFdec_inst->EventQueueVolume[DTMFdec_inst->EventBufferSize] = Volume;
- DTMFdec_inst->EventQueueEnded[DTMFdec_inst->EventBufferSize] = endEvent;
- DTMFdec_inst->EventQueueStartTime[DTMFdec_inst->EventBufferSize] = timeStamp;
- DTMFdec_inst->EventQueueEndTime[DTMFdec_inst->EventBufferSize] = timeStamp
- + Duration;
- DTMFdec_inst->EventBufferSize++;
- }
- return 0;
- }
- return DTMF_INSERT_ERROR;
-}
-
-WebRtc_Word16 WebRtcNetEQ_DtmfDecode(dtmf_inst_t *DTMFdec_inst, WebRtc_Word16 *event,
- WebRtc_Word16 *volume, WebRtc_UWord32 currTimeStamp)
-{
-
- if (DTMFdec_inst->EventBufferSize < 1) return 0; /* No events to play */
-
- /* We have events, is it time to play them? */
- if (currTimeStamp < DTMFdec_inst->EventQueueStartTime[0])
- {
- /*No, just return zero */
- return 0;
- }
-
- /* Continue on the event that is currently ongoing */
- *event = DTMFdec_inst->EventQueue[0];
- *volume = DTMFdec_inst->EventQueueVolume[0];
-
- if (DTMFdec_inst->EventQueueEndTime[0] >= (currTimeStamp + DTMFdec_inst->framelen))
- {
-
- /* Still at least framLen to play */
-
- DTMFdec_inst->CurrentPLCtime = 0;
- if ((DTMFdec_inst->EventQueueEndTime[0] == (currTimeStamp + DTMFdec_inst->framelen))
- && (DTMFdec_inst->EventQueueEnded[0]))
- { /* We are done */
- /*Remove the event from Queue*/
- WebRtcNetEQ_DtmfRemoveEvent(DTMFdec_inst);
- }
- return DTMFdec_inst->framelen;
-
- }
- else
- {
- if ((DTMFdec_inst->EventQueueEnded[0]) || (DTMFdec_inst->EventQueue[1] > -1))
- {
- /*
- * Less than frameLen to play and end of event or already received next event.
- * Give our a whole frame size of audio to simplify things.
- */
-
- /*Remove the event from Queue*/
- WebRtcNetEQ_DtmfRemoveEvent(DTMFdec_inst);
- DTMFdec_inst->CurrentPLCtime = 0;
-
- return DTMFdec_inst->framelen;
-
- }
- else
- {
- /* Less than frameLen to play and not end of event. */
- DTMFdec_inst->CurrentPLCtime = (WebRtc_Word16) (currTimeStamp
- - DTMFdec_inst->EventQueueEndTime[0]);
-
- if ((DTMFdec_inst->CurrentPLCtime > DTMFdec_inst->MaxPLCtime)
- || (DTMFdec_inst->CurrentPLCtime < -DTMFdec_inst->MaxPLCtime))
- {
- /*Remove the event from queue*/
- WebRtcNetEQ_DtmfRemoveEvent(DTMFdec_inst);
- DTMFdec_inst->CurrentPLCtime = 0;
- }
-
- /* If we have a new event that it's time to play */
- if ((DTMFdec_inst->EventQueue[1] > -1) && (DTMFdec_inst->EventQueueStartTime[1]
- >= (currTimeStamp + DTMFdec_inst->framelen)))
- {
- /*Remove the event from queue*/
- WebRtcNetEQ_DtmfRemoveEvent(DTMFdec_inst);
- DTMFdec_inst->CurrentPLCtime = 0;
- }
-
- return DTMFdec_inst->framelen;
- }
- }
-}
-
-#endif
diff --git a/trunk/src/modules/audio_coding/neteq/dtmf_buffer.h b/trunk/src/modules/audio_coding/neteq/dtmf_buffer.h
deleted file mode 100644
index e185411..0000000
--- a/trunk/src/modules/audio_coding/neteq/dtmf_buffer.h
+++ /dev/null
@@ -1,101 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Packet buffer for DTMF messages.
- */
-
-#ifndef DTMF_BUFFER_H
-#define DTMF_BUFFER_H
-
-#include "typedefs.h"
-
-#include "neteq_defines.h"
-
-/* Include this code only if ATEVENT (DTMF) is defined in */
-#ifdef NETEQ_ATEVENT_DECODE
-
-#define MAX_DTMF_QUEUE_SIZE 4
-
-typedef struct dtmf_inst_t_
-{
- WebRtc_Word16 MaxPLCtime;
- WebRtc_Word16 CurrentPLCtime;
- WebRtc_Word16 EventQueue[MAX_DTMF_QUEUE_SIZE];
- WebRtc_Word16 EventQueueVolume[MAX_DTMF_QUEUE_SIZE];
- WebRtc_Word16 EventQueueEnded[MAX_DTMF_QUEUE_SIZE];
- WebRtc_UWord32 EventQueueStartTime[MAX_DTMF_QUEUE_SIZE];
- WebRtc_UWord32 EventQueueEndTime[MAX_DTMF_QUEUE_SIZE];
- WebRtc_Word16 EventBufferSize;
- WebRtc_Word16 framelen;
-} dtmf_inst_t;
-
-/****************************************************************************
- * WebRtcNetEQ_DtmfDecoderInit(...)
- *
- * This function initializes a DTMF instance.
- *
- * Input:
- * - DTMF_decinst_t : DTMF instance
- * - fs : The sample rate used for the DTMF
- * - MaxPLCtime : Maximum length for a PLC before zeros should be inserted
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcNetEQ_DtmfDecoderInit(dtmf_inst_t *DTMFdec_inst, WebRtc_UWord16 fs,
- WebRtc_Word16 MaxPLCtime);
-
-/****************************************************************************
- * WebRtcNetEQ_DtmfInsertEvent(...)
- *
- * This function decodes a packet with DTMF frames.
- *
- * Input:
- * - DTMFdec_inst : DTMF instance
- * - encoded : Encoded DTMF frame(s)
- * - len : Bytes in encoded vector
- *
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcNetEQ_DtmfInsertEvent(dtmf_inst_t *DTMFdec_inst,
- const WebRtc_Word16 *encoded, WebRtc_Word16 len,
- WebRtc_UWord32 timeStamp);
-
-/****************************************************************************
- * WebRtcNetEQ_DtmfDecode(...)
- *
- * This function decodes a packet with DTMF frame(s). Output will be the
- * event that should be played for next 10 ms.
- *
- * Input:
- * - DTMFdec_inst : DTMF instance
- * - currTimeStamp : The current playout timestamp
- *
- * Output:
- * - event : Event number to be played
- * - volume : Event volume to be played
- *
- * Return value : >0 - There is a event to be played
- * 0 - No event to be played
- * -1 - Error
- */
-
-WebRtc_Word16 WebRtcNetEQ_DtmfDecode(dtmf_inst_t *DTMFdec_inst, WebRtc_Word16 *event,
- WebRtc_Word16 *volume, WebRtc_UWord32 currTimeStamp);
-
-#endif /* NETEQ_ATEVENT_DECODE */
-
-#endif /* DTMF_BUFFER_H */
-
diff --git a/trunk/src/modules/audio_coding/neteq/dtmf_tonegen.c b/trunk/src/modules/audio_coding/neteq/dtmf_tonegen.c
deleted file mode 100644
index a52f9bc..0000000
--- a/trunk/src/modules/audio_coding/neteq/dtmf_tonegen.c
+++ /dev/null
@@ -1,371 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the DTMF tone generator and its parameters.
- *
- * A sinusoid is generated using the recursive oscillator model
- *
- * y[n] = sin(w*n + phi) = 2*cos(w) * y[n-1] - y[n-2]
- * = a * y[n-1] - y[n-2]
- *
- * initialized with
- * y[-2] = 0
- * y[-1] = sin(w)
- *
- * A DTMF signal is a combination of two sinusoids, depending
- * on which event is sent (i.e, which key is pressed). The following
- * table maps each key (event codes in parentheses) into two tones:
- *
- * 1209 Hz 1336 Hz 1477 Hz 1633 Hz
- * 697 Hz 1 (ev. 1) 2 (ev. 2) 3 (ev. 3) A (ev. 12)
- * 770 Hz 4 (ev. 4) 5 (ev. 5) 6 (ev. 6) B (ev. 13)
- * 852 Hz 7 (ev. 7) 8 (ev. 8) 9 (ev. 9) C (ev. 14)
- * 941 Hz * (ev. 10) 0 (ev. 0) # (ev. 11) D (ev. 15)
- *
- * The two tones are added to form the DTMF signal.
- *
- */
-
-#include "dtmf_tonegen.h"
-
-#include "signal_processing_library.h"
-
-#include "neteq_error_codes.h"
-
-#ifdef NETEQ_ATEVENT_DECODE
-/* Must compile NetEQ with DTMF support to enable the functionality */
-
-/*******************/
-/* Constant tables */
-/*******************/
-
-/*
- * All tables corresponding to the oscillator model are organized so that
- * the coefficients for a specific frequency is found in the same position
- * in every table. The positions for the tones follow this layout:
- *
- * dummyVector[8] =
- * {
- * 697 Hz, 770 Hz, 852 Hz, 941 Hz,
- * 1209 Hz, 1336 Hz, 1477 Hz, 1633 Hz
- * };
- */
-
-/*
- * Tables for the constant a = 2*cos(w) = 2*cos(2*pi*f/fs)
- * in the oscillator model, for 8, 16, 32 and 48 kHz sample rate.
- * Table values in Q14.
- */
-
-const WebRtc_Word16 WebRtcNetEQ_dtfm_aTbl8Khz[8] =
-{
- 27980, 26956, 25701, 24219,
- 19073, 16325, 13085, 9315
-};
-
-#ifdef NETEQ_WIDEBAND
-const WebRtc_Word16 WebRtcNetEQ_dtfm_aTbl16Khz[8]=
-{
- 31548, 31281, 30951, 30556,
- 29144, 28361, 27409, 26258
-};
-#endif
-
-#ifdef NETEQ_32KHZ_WIDEBAND
-const WebRtc_Word16 WebRtcNetEQ_dtfm_aTbl32Khz[8]=
-{
- 32462, 32394, 32311, 32210,
- 31849, 31647, 31400, 31098
-};
-#endif
-
-#ifdef NETEQ_48KHZ_WIDEBAND
-const WebRtc_Word16 WebRtcNetEQ_dtfm_aTbl48Khz[8]=
-{
- 32632, 32602, 32564, 32520,
- 32359, 32268, 32157, 32022
-};
-#endif
-
-/*
- * Initialization values y[-1] = sin(w) = sin(2*pi*f/fs), for 8, 16, 32 and 48 kHz sample rate.
- * Table values in Q14.
- */
-
-const WebRtc_Word16 WebRtcNetEQ_dtfm_yInitTab8Khz[8] =
-{
- 8528, 9315, 10163, 11036,
- 13323, 14206,15021, 15708
-};
-
-#ifdef NETEQ_WIDEBAND
-const WebRtc_Word16 WebRtcNetEQ_dtfm_yInitTab16Khz[8]=
-{
- 4429, 4879, 5380, 5918,
- 7490, 8207, 8979, 9801
-};
-#endif
-
-#ifdef NETEQ_32KHZ_WIDEBAND
-const WebRtc_Word16 WebRtcNetEQ_dtfm_yInitTab32Khz[8]=
-{
- 2235, 2468, 2728, 3010,
- 3853, 4249, 4685, 5164
-};
-#endif
-
-#ifdef NETEQ_48KHZ_WIDEBAND
-const WebRtc_Word16 WebRtcNetEQ_dtfm_yInitTab48Khz[8]=
-{
- 1493, 1649, 1823, 2013,
- 2582, 2851, 3148, 3476
-};
-#endif
-
-/* Volume in dBm0 from 0 to -63, where 0 is the first table entry.
- Everything below -36 is discarded, wherefore the table stops at -36.
- Table entries are in Q14.
- */
-
-const WebRtc_Word16 WebRtcNetEQ_dtfm_dBm0[37] = { 16141, 14386, 12821, 11427, 10184, 9077, 8090,
- 7210, 6426, 5727, 5104, 4549, 4054, 3614,
- 3221, 2870, 2558, 2280, 2032, 1811, 1614,
- 1439, 1282, 1143, 1018, 908, 809, 721, 643,
- 573, 510, 455, 405, 361, 322, 287, 256 };
-
-/****************************************************************************
- * WebRtcNetEQ_DTMFGenerate(...)
- *
- * Generate 10 ms DTMF signal according to input parameters.
- *
- * Input:
- * - DTMFdecInst : DTMF instance
- * - value : DTMF event number (0-15)
- * - volume : Volume of generated signal (0-36)
- * Volume is given in negative dBm0, i.e., volume == 0
- * means 0 dBm0 while volume == 36 mean -36 dBm0.
- * - sampFreq : Sample rate in Hz
- *
- * Output:
- * - signal : Pointer to vector where DTMF signal is stored;
- * Vector must be at least sampFreq/100 samples long.
- * - DTMFdecInst : Updated DTMF instance
- *
- * Return value : >0 - Number of samples written to signal
- * : <0 - error
- */
-
-WebRtc_Word16 WebRtcNetEQ_DTMFGenerate(dtmf_tone_inst_t *DTMFdecInst, WebRtc_Word16 value,
- WebRtc_Word16 volume, WebRtc_Word16 *signal,
- WebRtc_UWord16 sampFreq, WebRtc_Word16 extFrameLen)
-{
- const WebRtc_Word16 *aTbl; /* pointer to a-coefficient table */
- const WebRtc_Word16 *yInitTable; /* pointer to initialization value table */
- WebRtc_Word16 a1 = 0; /* a-coefficient for first tone (low tone) */
- WebRtc_Word16 a2 = 0; /* a-coefficient for second tone (high tone) */
- int i;
- int frameLen; /* number of samples to generate */
- int lowIndex;
- int highIndex;
- WebRtc_Word32 tempVal;
- WebRtc_Word16 tempValLow;
- WebRtc_Word16 tempValHigh;
-
- /* Sanity check for volume */
- if ((volume < 0) || (volume > 36))
- {
- return DTMF_DEC_PARAMETER_ERROR;
- }
-
- /* Sanity check for extFrameLen */
- if (extFrameLen < -1)
- {
- return DTMF_DEC_PARAMETER_ERROR;
- }
-
- /* Select oscillator coefficient tables based on sample rate */
- if (sampFreq == 8000)
- {
- aTbl = WebRtcNetEQ_dtfm_aTbl8Khz;
- yInitTable = WebRtcNetEQ_dtfm_yInitTab8Khz;
- frameLen = 80;
-#ifdef NETEQ_WIDEBAND
- }
- else if (sampFreq == 16000)
- {
- aTbl = WebRtcNetEQ_dtfm_aTbl16Khz;
- yInitTable = WebRtcNetEQ_dtfm_yInitTab16Khz;
- frameLen = 160;
-#endif
-#ifdef NETEQ_32KHZ_WIDEBAND
- }
- else if (sampFreq == 32000)
- {
- aTbl = WebRtcNetEQ_dtfm_aTbl32Khz;
- yInitTable = WebRtcNetEQ_dtfm_yInitTab32Khz;
- frameLen = 320;
-#endif
-#ifdef NETEQ_48KHZ_WIDEBAND
- }
- else if (sampFreq == 48000)
- {
- aTbl = WebRtcNetEQ_dtfm_aTbl48Khz;
- yInitTable = WebRtcNetEQ_dtfm_yInitTab48Khz;
- frameLen = 480;
-#endif
- }
- else
- {
- /* unsupported sample rate */
- return DTMF_GEN_UNKNOWN_SAMP_FREQ;
- }
-
- if (extFrameLen >= 0)
- {
- frameLen = extFrameLen;
- }
-
- /* select low frequency based on event value */
- switch (value)
- {
- case 1:
- case 2:
- case 3:
- case 12: /* first row on keypad */
- {
- lowIndex = 0; /* low frequency: 697 Hz */
- break;
- }
- case 4:
- case 5:
- case 6:
- case 13: /* second row on keypad */
- {
- lowIndex = 1; /* low frequency: 770 Hz */
- break;
- }
- case 7:
- case 8:
- case 9:
- case 14: /* third row on keypad */
- {
- lowIndex = 2; /* low frequency: 852 Hz */
- break;
- }
- case 0:
- case 10:
- case 11:
- case 15: /* fourth row on keypad */
- {
- lowIndex = 3; /* low frequency: 941 Hz */
- break;
- }
- default:
- {
- return DTMF_DEC_PARAMETER_ERROR;
- }
- } /* end switch */
-
- /* select high frequency based on event value */
- switch (value)
- {
- case 1:
- case 4:
- case 7:
- case 10: /* first column on keypad */
- {
- highIndex = 4; /* high frequency: 1209 Hz */
- break;
- }
- case 2:
- case 5:
- case 8:
- case 0: /* second column on keypad */
- {
- highIndex = 5;/* high frequency: 1336 Hz */
- break;
- }
- case 3:
- case 6:
- case 9:
- case 11: /* third column on keypad */
- {
- highIndex = 6;/* high frequency: 1477 Hz */
- break;
- }
- case 12:
- case 13:
- case 14:
- case 15: /* fourth column on keypad (special) */
- {
- highIndex = 7;/* high frequency: 1633 Hz */
- break;
- }
- default:
- {
- return DTMF_DEC_PARAMETER_ERROR;
- }
- } /* end switch */
-
- /* select coefficients based on results from switches above */
- a1 = aTbl[lowIndex]; /* coefficient for first (low) tone */
- a2 = aTbl[highIndex]; /* coefficient for second (high) tone */
-
- if (DTMFdecInst->reinit)
- {
- /* set initial values for the recursive model */
- DTMFdecInst->oldOutputLow[0] = yInitTable[lowIndex];
- DTMFdecInst->oldOutputLow[1] = 0;
- DTMFdecInst->oldOutputHigh[0] = yInitTable[highIndex];
- DTMFdecInst->oldOutputHigh[1] = 0;
-
- /* reset reinit flag */
- DTMFdecInst->reinit = 0;
- }
-
- /* generate signal sample by sample */
- for (i = 0; i < frameLen; i++)
- {
-
- /* Use rescursion formula y[n] = a*y[n-1] - y[n-2] */
- tempValLow
- = (WebRtc_Word16) (((WEBRTC_SPL_MUL_16_16(a1, DTMFdecInst->oldOutputLow[1])
- + 8192) >> 14) - DTMFdecInst->oldOutputLow[0]);
- tempValHigh
- = (WebRtc_Word16) (((WEBRTC_SPL_MUL_16_16(a2, DTMFdecInst->oldOutputHigh[1])
- + 8192) >> 14) - DTMFdecInst->oldOutputHigh[0]);
-
- /* Update recursion memory */
- DTMFdecInst->oldOutputLow[0] = DTMFdecInst->oldOutputLow[1];
- DTMFdecInst->oldOutputLow[1] = tempValLow;
- DTMFdecInst->oldOutputHigh[0] = DTMFdecInst->oldOutputHigh[1];
- DTMFdecInst->oldOutputHigh[1] = tempValHigh;
-
- /* scale high tone with 32768 (15 left shifts)
- and low tone with 23171 (3dB lower than high tone) */
- tempVal = WEBRTC_SPL_MUL_16_16(DTMF_AMP_LOW, tempValLow)
- + WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)tempValHigh, 15);
-
- /* Norm the signal to Q14 (with proper rounding) */
- tempVal = (tempVal + 16384) >> 15;
-
- /* Scale the signal to correct dbM0 value */
- signal[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
- (WEBRTC_SPL_MUL_16_16(tempVal, WebRtcNetEQ_dtfm_dBm0[volume])
- + 8192), 14); /* volume value is in Q14; use proper rounding */
- }
-
- return frameLen;
-
-}
-
-#endif /* NETEQ_ATEVENT_DECODE */
-
diff --git a/trunk/src/modules/audio_coding/neteq/dtmf_tonegen.h b/trunk/src/modules/audio_coding/neteq/dtmf_tonegen.h
deleted file mode 100644
index add6eb1..0000000
--- a/trunk/src/modules/audio_coding/neteq/dtmf_tonegen.h
+++ /dev/null
@@ -1,73 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the DTMF tone generator function.
- */
-
-#ifndef DTMF_TONEGEN_H
-#define DTMF_TONEGEN_H
-
-#include "typedefs.h"
-
-#include "neteq_defines.h"
-
-#ifdef NETEQ_ATEVENT_DECODE
-/* Must compile NetEQ with DTMF support to enable the functionality */
-
-#define DTMF_AMP_LOW 23171 /* 3 dB lower than the high frequency */
-
-/* The DTMF generator struct (part of DSP main struct DSPInst_t) */
-typedef struct dtmf_tone_inst_t_
-{
-
- WebRtc_Word16 reinit; /* non-zero if the oscillator model should
- be reinitialized for next event */
- WebRtc_Word16 oldOutputLow[2]; /* oscillator recursion history (low tone) */
- WebRtc_Word16 oldOutputHigh[2]; /* oscillator recursion history (high tone) */
-
- int lastDtmfSample; /* index to the first non-DTMF sample in the
- speech history, if non-negative */
-}dtmf_tone_inst_t;
-
-/****************************************************************************
- * WebRtcNetEQ_DTMFGenerate(...)
- *
- * Generate 10 ms DTMF signal according to input parameters.
- *
- * Input:
- * - DTMFdecInst : DTMF instance
- * - value : DTMF event number (0-15)
- * - volume : Volume of generated signal (0-36)
- * Volume is given in negative dBm0, i.e., volume == 0
- * means 0 dBm0 while volume == 36 mean -36 dBm0.
- * - sampFreq : Sample rate in Hz
- *
- * Output:
- * - signal : Pointer to vector where DTMF signal is stored;
- * Vector must be at least sampFreq/100 samples long.
- * - DTMFdecInst : Updated DTMF instance
- *
- * Return value : >0 - Number of samples written to signal
- * : <0 - Error
- */
-
-WebRtc_Word16 WebRtcNetEQ_DTMFGenerate(dtmf_tone_inst_t *DTMFdecInst,
- WebRtc_Word16 value,
- WebRtc_Word16 volume,
- WebRtc_Word16 *signal,
- WebRtc_UWord16 sampFreq,
- WebRtc_Word16 frameLen
-);
-
-#endif /* NETEQ_ATEVENT_DECODE */
-
-#endif /* DTMF_TONEGEN_H */
-
diff --git a/trunk/src/modules/audio_coding/neteq/expand.c b/trunk/src/modules/audio_coding/neteq/expand.c
deleted file mode 100644
index 3db7a2a..0000000
--- a/trunk/src/modules/audio_coding/neteq/expand.c
+++ /dev/null
@@ -1,1216 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This is the function to expand from the speech history, to produce concealment data or
- * increasing delay.
- */
-
-#include "dsp.h"
-
-#include <assert.h>
-
-#include "signal_processing_library.h"
-
-#include "dsp_helpfunctions.h"
-#include "neteq_error_codes.h"
-
-#define CHECK_NO_OF_CORRMAX 3
-#define DISTLEN 20
-#define LPCANALASYSLEN 160
-
-/* Scratch usage:
-
- Type Name size startpos endpos
- (First part of first expand)
- WebRtc_Word16 pw16_bestCorrIndex 3 0 2
- WebRtc_Word16 pw16_bestCorr 3 3 5
- WebRtc_Word16 pw16_bestDistIndex 3 6 8
- WebRtc_Word16 pw16_bestDist 3 9 11
- WebRtc_Word16 pw16_corrVec 102*fs/8000 12 11+102*fs/8000
- func WebRtcNetEQ_Correlator 232 12+102*fs/8000 243+102*fs/8000
-
- (Second part of first expand)
- WebRtc_Word32 pw32_corr2 99*fs/8000+1 0 99*fs/8000
- WebRtc_Word32 pw32_autoCorr 2*7 0 13
- WebRtc_Word16 pw16_rc 6 14 19
-
- Signal combination:
- WebRtc_Word16 pw16_randVec 30+120*fs/8000 0 29+120*fs/8000
- WebRtc_Word16 pw16_scaledRandVec 125*fs/8000 30+120*fs/8000 29+245*fs/8000
- WebRtc_Word16 pw16_unvoicedVecSpace 10+125*fs/8000 30+245*fs/8000 39+370*fs/8000
-
- Total: 40+370*fs/8000 (size depends on UNVOICED_LPC_ORDER and BGN_LPC_ORDER)
- */
-
-#if ((BGN_LPC_ORDER > 10) || (UNVOICED_LPC_ORDER > 10)) && (defined SCRATCH)
-#error BGN_LPC_ORDER and/or BGN_LPC_ORDER are too large for current scratch memory allocation
-#endif
-
-#define SCRATCH_PW16_BEST_CORR_INDEX 0
-#define SCRATCH_PW16_BEST_CORR 3
-#define SCRATCH_PW16_BEST_DIST_INDEX 6
-#define SCRATCH_PW16_BEST_DIST 9
-#define SCRATCH_PW16_CORR_VEC 12
-#define SCRATCH_PW16_CORR2 0
-#define SCRATCH_PW32_AUTO_CORR 0
-#define SCRATCH_PW16_RC 14
-#define SCRATCH_PW16_RAND_VEC 0
-
-#if (defined(NETEQ_48KHZ_WIDEBAND))
-#define SCRATCH_NETEQDSP_CORRELATOR 624
-#define SCRATCH_PW16_SCALED_RAND_VEC 750
-#define SCRATCH_PW16_UNVOICED_VEC_SPACE 1500
-#elif (defined(NETEQ_32KHZ_WIDEBAND))
-#define SCRATCH_NETEQDSP_CORRELATOR 420
-#define SCRATCH_PW16_SCALED_RAND_VEC 510
-#define SCRATCH_PW16_UNVOICED_VEC_SPACE 1010
-#elif (defined(NETEQ_WIDEBAND))
-#define SCRATCH_NETEQDSP_CORRELATOR 216
-#define SCRATCH_PW16_SCALED_RAND_VEC 270
-#define SCRATCH_PW16_UNVOICED_VEC_SPACE 520
-#else /* NB */
-#define SCRATCH_NETEQDSP_CORRELATOR 114
-#define SCRATCH_PW16_SCALED_RAND_VEC 150
-#define SCRATCH_PW16_UNVOICED_VEC_SPACE 275
-#endif
-
-/****************************************************************************
- * WebRtcNetEQ_Expand(...)
- *
- * This function produces one "chunk" of expansion data (PLC audio). The
- * length of the produced audio depends on the speech history.
- *
- * Input:
- * - inst : DSP instance
- * - scratchPtr : Pointer to scratch vector
- * - outdata : Pointer to a memory space where the output data
- * should be stored
- * - BGNonly : If non-zero, "expand" will only produce background noise.
- * - pw16_len : Desired number of samples (only for BGN mode).
- *
- * Output:
- * - inst : Updated instance
- * - pw16_len : Number of samples that were output from NetEq
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-
-int WebRtcNetEQ_Expand(DSPInst_t *inst,
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_scratchPtr,
-#endif
- WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
- WebRtc_Word16 BGNonly)
-{
-
- WebRtc_Word16 fs_mult;
- ExpandInst_t *ExpandState = &(inst->ExpandInst);
- BGNInst_t *BGNState = &(inst->BGNInst);
- int i;
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_randVec = pw16_scratchPtr + SCRATCH_PW16_RAND_VEC;
- WebRtc_Word16 *pw16_scaledRandVec = pw16_scratchPtr + SCRATCH_PW16_SCALED_RAND_VEC;
- WebRtc_Word16 *pw16_unvoicedVecSpace = pw16_scratchPtr + SCRATCH_PW16_UNVOICED_VEC_SPACE;
-#else
- WebRtc_Word16 pw16_randVec[FSMULT * 120 + 30]; /* 150 for NB and 270 for WB */
- WebRtc_Word16 pw16_scaledRandVec[FSMULT * 125]; /* 125 for NB and 250 for WB */
- WebRtc_Word16 pw16_unvoicedVecSpace[BGN_LPC_ORDER + FSMULT * 125];
-#endif
- /* 125 for NB and 250 for WB etc. Reuse pw16_outData[] for this vector */
- WebRtc_Word16 *pw16_voicedVecStorage = pw16_outData;
- WebRtc_Word16 *pw16_voicedVec = &pw16_voicedVecStorage[ExpandState->w16_overlap];
- WebRtc_Word16 *pw16_unvoicedVec = pw16_unvoicedVecSpace + UNVOICED_LPC_ORDER;
- WebRtc_Word16 *pw16_cngVec = pw16_unvoicedVecSpace + BGN_LPC_ORDER;
- WebRtc_Word16 w16_expVecsLen, w16_lag = 0, w16_expVecPos;
- WebRtc_Word16 w16_randLen;
- WebRtc_Word16 w16_vfractionChange; /* in Q14 */
- WebRtc_Word16 w16_winMute = 0, w16_winMuteInc = 0, w16_winUnMute = 0, w16_winUnMuteInc = 0;
- WebRtc_Word32 w32_tmp;
- WebRtc_Word16 w16_tmp, w16_tmp2;
- WebRtc_Word16 stability;
- enum BGNMode bgnMode = inst->BGNInst.bgnMode;
-
- /* Pre-calculate common multiplications with fs_mult */
- WebRtc_Word16 fsMult4;
- WebRtc_Word16 fsMult20;
- WebRtc_Word16 fsMult120;
- WebRtc_Word16 fsMultDistLen;
- WebRtc_Word16 fsMultLPCAnalasysLen;
-
-#ifdef NETEQ_STEREO
- MasterSlaveInfo *msInfo = inst->msInfo;
-#endif
-
- /* fs is WebRtc_UWord16 (to hold fs=48000) */
- fs_mult = WebRtcNetEQ_CalcFsMult(inst->fs); /* calculate fs/8000 */
-
- /* Pre-calculate common multiplications with fs_mult */
- fsMult4 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fs_mult, 4);
- fsMult20 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fs_mult, 20);
- fsMult120 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fs_mult, 120);
- fsMultDistLen = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fs_mult, DISTLEN);
- fsMultLPCAnalasysLen = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fs_mult, LPCANALASYSLEN);
-
- /*
- * Perform all the initial setup if it's the first expansion.
- * If background noise (BGN) only, this setup is not needed.
- */
- if (ExpandState->w16_consecExp == 0 && !BGNonly)
- {
- /* Setup more variables */
-#ifdef SCRATCH
- WebRtc_Word32 *pw32_autoCorr = (WebRtc_Word32*) (pw16_scratchPtr
- + SCRATCH_PW32_AUTO_CORR);
- WebRtc_Word16 *pw16_rc = pw16_scratchPtr + SCRATCH_PW16_RC;
- WebRtc_Word16 *pw16_bestCorrIndex = pw16_scratchPtr + SCRATCH_PW16_BEST_CORR_INDEX;
- WebRtc_Word16 *pw16_bestCorr = pw16_scratchPtr + SCRATCH_PW16_BEST_CORR;
- WebRtc_Word16 *pw16_bestDistIndex = pw16_scratchPtr + SCRATCH_PW16_BEST_DIST_INDEX;
- WebRtc_Word16 *pw16_bestDist = pw16_scratchPtr + SCRATCH_PW16_BEST_DIST;
- WebRtc_Word16 *pw16_corrVec = pw16_scratchPtr + SCRATCH_PW16_CORR_VEC;
- WebRtc_Word32 *pw32_corr2 = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_PW16_CORR2);
-#else
- WebRtc_Word32 pw32_autoCorr[UNVOICED_LPC_ORDER+1];
- WebRtc_Word16 pw16_rc[UNVOICED_LPC_ORDER];
- WebRtc_Word16 pw16_corrVec[FSMULT*102]; /* 102 for NB */
- WebRtc_Word16 pw16_bestCorrIndex[CHECK_NO_OF_CORRMAX];
- WebRtc_Word16 pw16_bestCorr[CHECK_NO_OF_CORRMAX];
- WebRtc_Word16 pw16_bestDistIndex[CHECK_NO_OF_CORRMAX];
- WebRtc_Word16 pw16_bestDist[CHECK_NO_OF_CORRMAX];
- WebRtc_Word32 pw32_corr2[(99*FSMULT)+1];
-#endif
- WebRtc_Word32 pw32_bestDist[CHECK_NO_OF_CORRMAX];
- WebRtc_Word16 w16_ind = 0;
- WebRtc_Word16 w16_corrVecLen;
- WebRtc_Word16 w16_corrScale;
- WebRtc_Word16 w16_distScale;
- WebRtc_Word16 w16_indMin, w16_indMax;
- WebRtc_Word16 w16_len;
- WebRtc_Word32 w32_en1, w32_en2, w32_cc;
- WebRtc_Word16 w16_en1Scale, w16_en2Scale;
- WebRtc_Word16 w16_en1, w16_en2;
- WebRtc_Word32 w32_en1_mul_en2;
- WebRtc_Word16 w16_sqrt_en1en2;
- WebRtc_Word16 w16_ccShiftL;
- WebRtc_Word16 w16_bestcorr; /* Correlation in Q14 */
- WebRtc_Word16 *pw16_vec1, *pw16_vec2;
- WebRtc_Word16 w16_factor;
- WebRtc_Word16 w16_DistLag, w16_CorrLag, w16_diffLag;
- WebRtc_Word16 w16_energyLen;
- WebRtc_Word16 w16_slope;
- WebRtc_Word16 w16_startInd;
- WebRtc_Word16 w16_noOfcorr2;
- WebRtc_Word16 w16_scale;
-
- /* Initialize some variables */
- ExpandState->w16_lagsDirection = 1;
- ExpandState->w16_lagsPosition = -1;
- ExpandState->w16_expandMuteFactor = 16384; /* Start from 1.0 (Q14) */
- BGNState->w16_mutefactor = 0; /* Start with 0 gain for BGN (value in Q14) */
- inst->w16_seedInc = 1;
-
-#ifdef NETEQ_STEREO
- /* Sanity for msInfo */
- if (msInfo == NULL)
- {
- /* this should not happen here */
- return MASTER_SLAVE_ERROR;
- }
-
- /*
- * Do not calculate correlations for slave instance(s)
- * unless lag info from master is corrupt
- */
- if ((msInfo->msMode != NETEQ_SLAVE)
- || ((msInfo->distLag <= 0) || (msInfo->corrLag <= 0)))
- {
-#endif
- /* Calculate correlation vector in downsampled domain (4 kHz sample rate) */
- w16_corrVecLen = WebRtcNetEQ_Correlator(inst,
-#ifdef SCRATCH
- pw16_scratchPtr + SCRATCH_NETEQDSP_CORRELATOR,
-#endif
- inst->pw16_speechHistory, inst->w16_speechHistoryLen, pw16_corrVec,
- &w16_corrScale);
-
- /* Find peaks in correlation vector using parabolic fit method */
- WebRtcNetEQ_PeakDetection(pw16_corrVec, w16_corrVecLen, CHECK_NO_OF_CORRMAX, fs_mult,
- pw16_bestCorrIndex, pw16_bestCorr);
-
- /*
- * Adjust peak locations; cross-correlation lags start at 2.5 ms
- * (20*fs_mult samples)
- */
- pw16_bestCorrIndex[0] += fsMult20;
- pw16_bestCorrIndex[1] += fsMult20;
- pw16_bestCorrIndex[2] += fsMult20;
-
- /* Calculate distortion around the 3 (CHECK_NO_OF_CORRMAX) best lags */
- w16_distScale = 0;
- for (i = 0; i < CHECK_NO_OF_CORRMAX; i++)
- {
- w16_tmp = fsMult20;
- w16_tmp2 = pw16_bestCorrIndex[i] - fsMult4;
- w16_indMin = WEBRTC_SPL_MAX(w16_tmp, w16_tmp2);
- w16_tmp = fsMult120 - 1;
- w16_tmp2 = pw16_bestCorrIndex[i] + fsMult4;
- w16_indMax = WEBRTC_SPL_MIN(w16_tmp, w16_tmp2);
-
- pw16_bestDistIndex[i] = WebRtcNetEQ_MinDistortion(
- &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - fsMultDistLen]),
- w16_indMin, w16_indMax, fsMultDistLen, &pw32_bestDist[i]);
-
- w16_distScale
- = WEBRTC_SPL_MAX(16 - WebRtcSpl_NormW32(pw32_bestDist[i]), w16_distScale);
-
- }
-
- /* Shift the distortion values to fit in WebRtc_Word16 */
- WebRtcSpl_VectorBitShiftW32ToW16(pw16_bestDist, CHECK_NO_OF_CORRMAX, pw32_bestDist,
- w16_distScale);
-
- /*
- * Find index of maximum criteria, where crit[i] = bestCorr[i])/(bestDist[i])
- * Do this by a cross multiplication.
- */
-
- w32_en1 = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[0],pw16_bestDist[1]);
- w32_en2 = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[1],pw16_bestDist[0]);
- if (w32_en1 >= w32_en2)
- {
- /* 0 wins over 1 */
- w32_en1
- = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[0], pw16_bestDist[2]);
- w32_en2
- = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[2], pw16_bestDist[0]);
- if (w32_en1 >= w32_en2)
- {
- /* 0 wins over 2 */
- w16_ind = 0;
- }
- else
- {
- /* 2 wins over 0 */
- w16_ind = 2;
- }
- }
- else
- {
- /* 1 wins over 0 */
- w32_en1
- = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[1],pw16_bestDist[2]);
- w32_en2
- = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) pw16_bestCorr[2],pw16_bestDist[1]);
- if ((WebRtc_Word32) w32_en1 >= (WebRtc_Word32) w32_en2)
- {
- /* 1 wins over 2 */
- w16_ind = 1;
- }
- else
- {
- /* 2 wins over 1 */
- w16_ind = 2;
- }
- }
-
-#ifdef NETEQ_STEREO
- }
-
- /* Store DistLag and CorrLag of the position with highest criteria */
- if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO)
- || ((msInfo->msMode == NETEQ_SLAVE) && (msInfo->distLag <= 0 || msInfo->corrLag
- <= 0)))
- {
- /* lags not provided externally */
- w16_DistLag = pw16_bestDistIndex[w16_ind];
- w16_CorrLag = pw16_bestCorrIndex[w16_ind];
- if (msInfo->msMode == NETEQ_MASTER)
- {
- msInfo->distLag = w16_DistLag;
- msInfo->corrLag = w16_CorrLag;
- }
- }
- else if (msInfo->msMode == NETEQ_SLAVE)
- {
- /* lags provided externally (from master) */
- w16_DistLag = msInfo->distLag;
- w16_CorrLag = msInfo->corrLag;
-
- /* sanity for lag values */
- if ((w16_DistLag <= 0) || (w16_CorrLag <= 0))
- {
- return MASTER_SLAVE_ERROR;
- }
- }
- else
- {
- /* Invalid mode */
- return MASTER_SLAVE_ERROR;
- }
-#else /* not NETEQ_STEREO */
- w16_DistLag = pw16_bestDistIndex[w16_ind];
- w16_CorrLag = pw16_bestCorrIndex[w16_ind];
-#endif
-
- ExpandState->w16_maxLag = WEBRTC_SPL_MAX(w16_DistLag, w16_CorrLag);
-
- /* Calculate the exact best correlation (in the range within CorrLag-DistLag) */
- w16_len = w16_DistLag + 10;
- w16_len = WEBRTC_SPL_MIN(w16_len, fsMult120);
- w16_len = WEBRTC_SPL_MAX(w16_len, 60 * fs_mult);
-
- w16_startInd = WEBRTC_SPL_MIN(w16_DistLag, w16_CorrLag);
- w16_noOfcorr2 = WEBRTC_SPL_ABS_W16((w16_DistLag-w16_CorrLag)) + 1;
- /* w16_noOfcorr2 maximum value is 99*fs_mult + 1 */
-
- /* Calculate suitable scaling */
- w16_tmp
- = WebRtcSpl_MaxAbsValueW16(
- &inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len - w16_startInd
- - w16_noOfcorr2],
- (WebRtc_Word16) (w16_len + w16_startInd + w16_noOfcorr2 - 1));
- w16_corrScale = ((31 - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_tmp, w16_tmp)))
- + (31 - WebRtcSpl_NormW32(w16_len))) - 31;
- w16_corrScale = WEBRTC_SPL_MAX(0, w16_corrScale);
-
- /*
- * Perform the correlation, store in pw32_corr2
- */
-
- WebRtcNetEQ_CrossCorr(pw32_corr2,
- &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len]),
- &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len - w16_startInd]),
- w16_len, w16_noOfcorr2, w16_corrScale, -1);
-
- /* Find maximizing index */
- w16_ind = WebRtcSpl_MaxIndexW32(pw32_corr2, w16_noOfcorr2);
- w32_cc = pw32_corr2[w16_ind]; /* this is maximum correlation */
- w16_ind = w16_ind + w16_startInd; /* correct index for start offset */
-
- /* Calculate energies */
- w32_en1 = WebRtcNetEQ_DotW16W16(
- &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len]),
- &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len]), w16_len,
- w16_corrScale);
- w32_en2 = WebRtcNetEQ_DotW16W16(
- &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len - w16_ind]),
- &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_len - w16_ind]),
- w16_len, w16_corrScale);
-
- /* Calculate the correlation value w16_bestcorr */
- if ((w32_en1 > 0) && (w32_en2 > 0))
- {
- w16_en1Scale = 16 - WebRtcSpl_NormW32(w32_en1);
- w16_en1Scale = WEBRTC_SPL_MAX(0, w16_en1Scale);
- w16_en2Scale = 16 - WebRtcSpl_NormW32(w32_en2);
- w16_en2Scale = WEBRTC_SPL_MAX(0, w16_en2Scale);
- /* Make sure total scaling is even (to simplify scale factor after sqrt) */
- if ((w16_en1Scale + w16_en2Scale) & 1)
- {
- /* if sum is odd */
- w16_en1Scale += 1;
- }
- w16_en1 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en1, w16_en1Scale);
- w16_en2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale);
- w32_en1_mul_en2 = WEBRTC_SPL_MUL_16_16(w16_en1, w16_en2);
- w16_sqrt_en1en2 = (WebRtc_Word16) WebRtcSpl_SqrtFloor(w32_en1_mul_en2);
-
- /* Calculate cc/sqrt(en1*en2) in Q14 */
- w16_ccShiftL = 14 - ((w16_en1Scale + w16_en2Scale) >> 1);
- w32_cc = WEBRTC_SPL_SHIFT_W32(w32_cc, w16_ccShiftL);
- w16_bestcorr = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_cc, w16_sqrt_en1en2);
- w16_bestcorr = WEBRTC_SPL_MIN(16384, w16_bestcorr); /* set maximum to 1.0 */
-
- }
- else
- {
- /* if either en1 or en2 is zero */
- w16_bestcorr = 0;
- }
-
- /*
- * Extract the two vectors, pw16_expVecs[0][] and pw16_expVecs[1][],
- * from the SpeechHistory[]
- */
- w16_expVecsLen = ExpandState->w16_maxLag + ExpandState->w16_overlap;
- pw16_vec1 = &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - w16_expVecsLen]);
- pw16_vec2 = pw16_vec1 - w16_DistLag;
- /* Normalize the second vector to the same energy as the first */
- w32_en1 = WebRtcNetEQ_DotW16W16(pw16_vec1, pw16_vec1, w16_expVecsLen, w16_corrScale);
- w32_en2 = WebRtcNetEQ_DotW16W16(pw16_vec2, pw16_vec2, w16_expVecsLen, w16_corrScale);
-
- /*
- * Confirm that energy factor sqrt(w32_en1/w32_en2) is within difference 0.5 - 2.0
- * w32_en1/w32_en2 within 0.25 - 4
- */
- if (((w32_en1 >> 2) < w32_en2) && ((w32_en1) > (w32_en2 >> 2)))
- {
-
- /* Energy constraint fulfilled => use both vectors and scale them accordingly */
- w16_en2Scale = 16 - WebRtcSpl_NormW32(w32_en2);
- w16_en2Scale = WEBRTC_SPL_MAX(0, w16_en2Scale);
- w16_en1Scale = w16_en2Scale - 13;
-
- /* calculate w32_en1/w32_en2 in Q13 */
- w32_en1_mul_en2 = WebRtcSpl_DivW32W16(
- WEBRTC_SPL_SHIFT_W32(w32_en1, -w16_en1Scale),
- (WebRtc_Word16) (WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale)));
-
- /* calculate factor in Q13 (sqrt of en1/en2 in Q26) */
- w16_factor = (WebRtc_Word16) WebRtcSpl_SqrtFloor(
- WEBRTC_SPL_LSHIFT_W32(w32_en1_mul_en2, 13));
-
- /* Copy the two vectors and give them the same energy */
-
- WEBRTC_SPL_MEMCPY_W16(ExpandState->pw16_expVecs[0], pw16_vec1, w16_expVecsLen);
- WebRtcSpl_AffineTransformVector(ExpandState->pw16_expVecs[1], pw16_vec2,
- w16_factor, 4096, 13, w16_expVecsLen);
-
- }
- else
- {
- /* Energy change constraint not fulfilled => only use last vector */
-
- WEBRTC_SPL_MEMCPY_W16(ExpandState->pw16_expVecs[0], pw16_vec1, w16_expVecsLen);
- WEBRTC_SPL_MEMCPY_W16(ExpandState->pw16_expVecs[1], ExpandState->pw16_expVecs[0],
- w16_expVecsLen);
-
- /* Set the w16_factor since it is used by muting slope */
- if (((w32_en1 >> 2) < w32_en2) || (w32_en2 == 0))
- {
- w16_factor = 4096; /* 0.5 in Q13*/
- }
- else
- {
- w16_factor = 16384; /* 2.0 in Q13*/
- }
- }
-
- /* Set the 3 lag values */
- w16_diffLag = w16_DistLag - w16_CorrLag;
- if (w16_diffLag == 0)
- {
- /* DistLag and CorrLag are equal */
- ExpandState->w16_lags[0] = w16_DistLag;
- ExpandState->w16_lags[1] = w16_DistLag;
- ExpandState->w16_lags[2] = w16_DistLag;
- }
- else
- {
- /* DistLag and CorrLag are not equal; use different combinations of the two */
- ExpandState->w16_lags[0] = w16_DistLag; /* DistLag only */
- ExpandState->w16_lags[1] = ((w16_DistLag + w16_CorrLag) >> 1); /* 50/50 */
- /* Third lag, move one half-step towards CorrLag (in both cases) */
- if (w16_diffLag > 0)
- {
- ExpandState->w16_lags[2] = (w16_DistLag + w16_CorrLag - 1) >> 1;
- }
- else
- {
- ExpandState->w16_lags[2] = (w16_DistLag + w16_CorrLag + 1) >> 1;
- }
- }
-
- /*************************************************
- * Calculate the LPC and the gain of the filters *
- *************************************************/
-
- /* Calculate scale value needed for autocorrelation */
- w16_tmp = WebRtcSpl_MaxAbsValueW16(
- &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - fsMultLPCAnalasysLen]),
- fsMultLPCAnalasysLen);
-
- w16_tmp = 16 - WebRtcSpl_NormW32(w16_tmp);
- w16_tmp = WEBRTC_SPL_MIN(w16_tmp,0);
- w16_tmp = (w16_tmp << 1) + 7;
- w16_tmp = WEBRTC_SPL_MAX(w16_tmp,0);
-
- /* set w16_ind to simplify the following expressions */
- w16_ind = inst->w16_speechHistoryLen - fsMultLPCAnalasysLen - UNVOICED_LPC_ORDER;
-
- /* store first UNVOICED_LPC_ORDER samples in pw16_rc */
-
- WEBRTC_SPL_MEMCPY_W16(pw16_rc, &inst->pw16_speechHistory[w16_ind], UNVOICED_LPC_ORDER);
-
- /* set first samples to zero */
- WebRtcSpl_MemSetW16(&inst->pw16_speechHistory[w16_ind], 0, UNVOICED_LPC_ORDER);
-
- /* Calculate UNVOICED_LPC_ORDER+1 lags of the ACF */
-
- WebRtcNetEQ_CrossCorr(
- pw32_autoCorr, &(inst->pw16_speechHistory[w16_ind + UNVOICED_LPC_ORDER]),
- &(inst->pw16_speechHistory[w16_ind + UNVOICED_LPC_ORDER]), fsMultLPCAnalasysLen,
- UNVOICED_LPC_ORDER + 1, w16_tmp, -1);
-
- /* Recover the stored samples from pw16_rc */
-
- WEBRTC_SPL_MEMCPY_W16(&inst->pw16_speechHistory[w16_ind], pw16_rc, UNVOICED_LPC_ORDER);
-
- if (pw32_autoCorr[0] > 0)
- { /* check that variance is positive */
-
- /* estimate AR filter parameters using Levinson-Durbin algorithm
- (UNVOICED_LPC_ORDER+1 filter coefficients) */
- stability = WebRtcSpl_LevinsonDurbin(pw32_autoCorr, ExpandState->pw16_arFilter,
- pw16_rc, UNVOICED_LPC_ORDER);
-
- /* Only update BGN if filter is stable */
- if (stability != 1)
- {
- /* Set first coefficient to 4096 (1.0 in Q12)*/
- ExpandState->pw16_arFilter[0] = 4096;
- /* Set remaining UNVOICED_LPC_ORDER coefficients to zero */
- WebRtcSpl_MemSetW16(ExpandState->pw16_arFilter + 1, 0, UNVOICED_LPC_ORDER);
- }
-
- }
-
- if (w16_DistLag < 40)
- {
- w16_energyLen = 2 * w16_DistLag;
- }
- else
- {
- w16_energyLen = w16_DistLag;
- }
- w16_randLen = w16_energyLen + 30; /* Startup part */
-
- /* Extract a noise segment */
- if (w16_randLen <= RANDVEC_NO_OF_SAMPLES)
- {
- WEBRTC_SPL_MEMCPY_W16(pw16_randVec,
- (WebRtc_Word16*) WebRtcNetEQ_kRandnTbl, w16_randLen);
- }
- else
- { /* only applies to SWB where length could be larger than 256 */
-#if FSMULT >= 2 /* Makes pw16_randVec longer than RANDVEC_NO_OF_SAMPLES. */
- WEBRTC_SPL_MEMCPY_W16(pw16_randVec, (WebRtc_Word16*) WebRtcNetEQ_kRandnTbl,
- RANDVEC_NO_OF_SAMPLES);
- inst->w16_seedInc = (inst->w16_seedInc + 2) & (RANDVEC_NO_OF_SAMPLES - 1);
- assert(w16_randLen <= FSMULT * 120 + 30);
- WebRtcNetEQ_RandomVec(&inst->uw16_seed, &pw16_randVec[RANDVEC_NO_OF_SAMPLES],
- (WebRtc_Word16) (w16_randLen - RANDVEC_NO_OF_SAMPLES), inst->w16_seedInc);
-#else
- assert(0);
-#endif
- }
-
- /* Set up state vector and calculate scale factor for unvoiced filtering */
-
- WEBRTC_SPL_MEMCPY_W16(ExpandState->pw16_arState,
- &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - UNVOICED_LPC_ORDER]),
- UNVOICED_LPC_ORDER);
- WEBRTC_SPL_MEMCPY_W16(pw16_unvoicedVec - UNVOICED_LPC_ORDER,
- &(inst->pw16_speechHistory[inst->w16_speechHistoryLen - 128 - UNVOICED_LPC_ORDER]),
- UNVOICED_LPC_ORDER);
- WebRtcSpl_FilterMAFastQ12(&inst->pw16_speechHistory[inst->w16_speechHistoryLen - 128],
- pw16_unvoicedVec, ExpandState->pw16_arFilter, UNVOICED_LPC_ORDER + 1, 128);
- if (WebRtcSpl_MaxAbsValueW16(pw16_unvoicedVec, 128) > 4000)
- {
- w16_scale = 4;
- }
- else
- {
- w16_scale = 0;
- }
- w32_tmp = WebRtcNetEQ_DotW16W16(pw16_unvoicedVec, pw16_unvoicedVec, 128, w16_scale);
-
- /* Normalize w32_tmp to 28 or 29 bits to preserve sqrt() accuracy */
- w16_tmp = WebRtcSpl_NormW32(w32_tmp) - 3;
- w16_tmp += ((w16_tmp & 0x1) ^ 0x1); /* Make sure we do an odd number of shifts since we
- from earlier have 7 shifts from dividing with 128.*/
- w32_tmp = WEBRTC_SPL_SHIFT_W32(w32_tmp, w16_tmp);
- w32_tmp = WebRtcSpl_SqrtFloor(w32_tmp);
- ExpandState->w16_arGainScale = 13 + ((w16_tmp + 7 - w16_scale) >> 1);
- ExpandState->w16_arGain = (WebRtc_Word16) w32_tmp;
-
- /********************************************************************
- * Calculate vfraction from bestcorr *
- * if (bestcorr>0.480665) *
- * vfraction = ((bestcorr-0.4)/(1-0.4)).^2 *
- * else vfraction = 0 *
- * *
- * approximation (coefficients in Q12): *
- * if (x>0.480665) (y(x)<0.3) *
- * y(x) = -1.264421 + 4.8659148*x - 4.0092827*x^2 + 1.4100529*x^3 *
- * else y(x) = 0; *
- ********************************************************************/
-
- if (w16_bestcorr > 7875)
- {
- /* if x>0.480665 */
- WebRtc_Word16 w16_x1, w16_x2, w16_x3;
- w16_x1 = w16_bestcorr;
- w32_tmp = WEBRTC_SPL_MUL_16_16((WebRtc_Word32) w16_x1, w16_x1);
- w16_x2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 14);
- w32_tmp = WEBRTC_SPL_MUL_16_16(w16_x1, w16_x2);
- w16_x3 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 14);
- w32_tmp
- = (WebRtc_Word32) WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) WebRtcNetEQ_kMixFractionFuncTbl[0], 14);
- w32_tmp
- += (WebRtc_Word32) WEBRTC_SPL_MUL_16_16(WebRtcNetEQ_kMixFractionFuncTbl[1], w16_x1);
- w32_tmp
- += (WebRtc_Word32) WEBRTC_SPL_MUL_16_16(WebRtcNetEQ_kMixFractionFuncTbl[2], w16_x2);
- w32_tmp
- += (WebRtc_Word32) WEBRTC_SPL_MUL_16_16(WebRtcNetEQ_kMixFractionFuncTbl[3], w16_x3);
- ExpandState->w16_vFraction = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 12);
- ExpandState->w16_vFraction = WEBRTC_SPL_MIN(ExpandState->w16_vFraction, 16384);
- ExpandState->w16_vFraction = WEBRTC_SPL_MAX(ExpandState->w16_vFraction, 0);
- }
- else
- {
- ExpandState->w16_vFraction = 0;
- }
-
- /***********************************************************************
- * Calculate muting slope, reuse value from earlier scaling of ExpVecs *
- ***********************************************************************/
- w16_slope = w16_factor;
-
- if (w16_slope > 12288)
- {
- /* w16_slope > 1.5 ? */
- /* Calculate (1-(1/slope))/w16_DistLag = (slope-1)/(w16_DistLag*slope) */
- w32_tmp = w16_slope - 8192;
- w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, 12); /* Value in Q25 (13+12=25) */
- w16_tmp = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(w16_DistLag,
- w16_slope, 8); /* Value in Q5 (13-8=5) */
- w16_tmp = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_tmp,
- w16_tmp); /* Res in Q20 (25-5=20) */
-
- if (w16_slope > 14746)
- { /* w16_slope > 1.8 ? */
- ExpandState->w16_muteSlope = (w16_tmp + 1) >> 1;
- }
- else
- {
- ExpandState->w16_muteSlope = (w16_tmp + 4) >> 3;
- }
- ExpandState->w16_onset = 1;
- }
- else if (ExpandState->w16_vFraction > 13107)
- {
- /* w16_vFraction > 0.8 ? */
- if (w16_slope > 8028)
- {
- /* w16_vFraction > 0.98 ? */
- ExpandState->w16_muteSlope = 0;
- }
- else
- {
- /* Calculate (1-slope)/w16_DistLag */
- w32_tmp = 8192 - w16_slope;
- w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, 7); /* Value in Q20 (13+7=20) */
- ExpandState->w16_muteSlope = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_tmp,
- w16_DistLag); /* Res in Q20 (20-0=20) */
- }
- ExpandState->w16_onset = 0;
- }
- else
- {
- /*
- * Use the minimum of 0.005 (0.9 on 50 samples in NB and the slope)
- * and ((1-slope)/w16_DistLag)
- */
- w32_tmp = 8192 - w16_slope;
- w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, 7); /* Value in Q20 (13+7=20) */
- w32_tmp = WEBRTC_SPL_MAX(w32_tmp, 0);
- ExpandState->w16_muteSlope = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_tmp,
- w16_DistLag); /* Res in Q20 (20-0=20) */
- w16_tmp = WebRtcNetEQ_k5243div[fs_mult]; /* 0.005/fs_mult = 5243/fs_mult */
- ExpandState->w16_muteSlope = WEBRTC_SPL_MAX(w16_tmp, ExpandState->w16_muteSlope);
- ExpandState->w16_onset = 0;
- }
- }
- else
- {
- /* This is not the first Expansion, parameters are already estimated. */
-
- /* Extract a noise segment */
- if (BGNonly) /* If we should produce nothing but background noise */
- {
- if (*pw16_len > 0)
- {
- /*
- * Set length to input parameter length, but not more than length
- * of pw16_randVec
- */
- w16_lag = WEBRTC_SPL_MIN(*pw16_len, FSMULT * 120 + 30);
- }
- else
- {
- /* set length to 15 ms */
- w16_lag = fsMult120;
- }
- w16_randLen = w16_lag;
- }
- else
- {
- w16_randLen = ExpandState->w16_maxLag;
- }
-
- if (w16_randLen <= RANDVEC_NO_OF_SAMPLES)
- {
- inst->w16_seedInc = (inst->w16_seedInc + 2) & (RANDVEC_NO_OF_SAMPLES - 1);
- WebRtcNetEQ_RandomVec(&inst->uw16_seed, pw16_randVec, w16_randLen,
- inst->w16_seedInc);
- }
- else
- { /* only applies to SWB where length could be larger than 256 */
-#if FSMULT >= 2 /* Makes pw16_randVec longer than RANDVEC_NO_OF_SAMPLES. */
- inst->w16_seedInc = (inst->w16_seedInc + 2) & (RANDVEC_NO_OF_SAMPLES - 1);
- WebRtcNetEQ_RandomVec(&inst->uw16_seed, pw16_randVec, RANDVEC_NO_OF_SAMPLES,
- inst->w16_seedInc);
- inst->w16_seedInc = (inst->w16_seedInc + 2) & (RANDVEC_NO_OF_SAMPLES - 1);
- assert(w16_randLen <= FSMULT * 120 + 30);
- WebRtcNetEQ_RandomVec(&inst->uw16_seed, &pw16_randVec[RANDVEC_NO_OF_SAMPLES],
- (WebRtc_Word16) (w16_randLen - RANDVEC_NO_OF_SAMPLES), inst->w16_seedInc);
-#else
- assert(0);
-#endif
- }
- } /* end if(first expand or BGNonly) ... else ... */
-
- if (!BGNonly) /* Voiced and unvoiced parts not used if generating BGN only */
- {
-
- /*************************************************
- * Generate signal *
- *************************************************/
-
- /*
- * Voiced part
- */
-
- /* Linearly mute the use_vfraction value from 1 to vfraction */
- if (ExpandState->w16_consecExp == 0)
- {
- ExpandState->w16_currentVFraction = 16384; /* 1.0 in Q14 */
- }
-
- ExpandState->w16_lagsPosition = ExpandState->w16_lagsPosition
- + ExpandState->w16_lagsDirection;
-
- /* Change direction if needed */
- if (ExpandState->w16_lagsPosition == 0)
- {
- ExpandState->w16_lagsDirection = 1;
- }
- if (ExpandState->w16_lagsPosition == 2)
- {
- ExpandState->w16_lagsDirection = -1;
- }
-
- /* Generate a weighted vector with the selected lag */
- w16_expVecsLen = ExpandState->w16_maxLag + ExpandState->w16_overlap;
- w16_lag = ExpandState->w16_lags[ExpandState->w16_lagsPosition];
- /* Copy lag+overlap data */
- w16_expVecPos = w16_expVecsLen - w16_lag - ExpandState->w16_overlap;
- w16_tmp = w16_lag + ExpandState->w16_overlap;
- if (ExpandState->w16_lagsPosition == 0)
- {
- WEBRTC_SPL_MEMCPY_W16(pw16_voicedVecStorage,
- &(ExpandState->pw16_expVecs[0][w16_expVecPos]), w16_tmp);
- }
- else if (ExpandState->w16_lagsPosition == 1)
- {
- WebRtcSpl_ScaleAndAddVectorsWithRound(&ExpandState->pw16_expVecs[0][w16_expVecPos], 3,
- &ExpandState->pw16_expVecs[1][w16_expVecPos], 1, 2, pw16_voicedVecStorage,
- w16_tmp);
-
- }
- else if (ExpandState->w16_lagsPosition == 2)
- {
- WebRtcSpl_ScaleAndAddVectorsWithRound(&ExpandState->pw16_expVecs[0][w16_expVecPos], 1,
- &ExpandState->pw16_expVecs[1][w16_expVecPos], 1, 1, pw16_voicedVecStorage,
- w16_tmp);
- }
-
- if (inst->fs == 8000)
- {
- /* Windowing in Q15 */
- w16_winMute = NETEQ_OVERLAP_WINMUTE_8KHZ_START;
- w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_8KHZ_INC;
- w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_8KHZ_START;
- w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_8KHZ_INC;
-#ifdef NETEQ_WIDEBAND
- }
- else if (inst->fs == 16000)
- {
- /* Windowing in Q15 */
- w16_winMute = NETEQ_OVERLAP_WINMUTE_16KHZ_START;
- w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_16KHZ_INC;
- w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_16KHZ_START;
- w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_16KHZ_INC;
-#endif
-#ifdef NETEQ_32KHZ_WIDEBAND
- }
- else if (inst->fs == 32000)
- {
- /* Windowing in Q15 */
- w16_winMute = NETEQ_OVERLAP_WINMUTE_32KHZ_START;
- w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_32KHZ_INC;
- w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_32KHZ_START;
- w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_32KHZ_INC;
-#endif
-#ifdef NETEQ_48KHZ_WIDEBAND
- }
- else /* if (inst->fs==48000) */
- {
- /* Windowing in Q15 */
- w16_winMute = NETEQ_OVERLAP_WINMUTE_48KHZ_START;
- w16_winMuteInc = NETEQ_OVERLAP_WINMUTE_48KHZ_INC;
- w16_winUnMute = NETEQ_OVERLAP_WINUNMUTE_48KHZ_START;
- w16_winUnMuteInc = NETEQ_OVERLAP_WINUNMUTE_48KHZ_INC;
-#endif
- }
-
- /* Smooth the expanded if it has not been muted to or vfraction is larger than 0.5 */
- if ((ExpandState->w16_expandMuteFactor > 819) && (ExpandState->w16_currentVFraction
- > 8192))
- {
- for (i = 0; i < ExpandState->w16_overlap; i++)
- {
- /* Do overlap add between new vector and overlap */
- ExpandState->pw16_overlapVec[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
- WEBRTC_SPL_MUL_16_16(ExpandState->pw16_overlapVec[i], w16_winMute) +
- WEBRTC_SPL_MUL_16_16(
- WEBRTC_SPL_MUL_16_16_RSFT(ExpandState->w16_expandMuteFactor,
- pw16_voicedVecStorage[i], 14), w16_winUnMute) + 16384, 15);
- w16_winMute += w16_winMuteInc;
- w16_winUnMute += w16_winUnMuteInc;
- }
- }
- else if (ExpandState->w16_expandMuteFactor == 0
-#ifdef NETEQ_STEREO
- && msInfo->msMode == NETEQ_MONO /* only if mono mode is selected */
-#endif
- )
- {
- /* if ExpandState->w16_expandMuteFactor = 0 => all is CNG component
- set the output length to 15ms (for best CNG production) */
- w16_tmp = fsMult120;
- ExpandState->w16_maxLag = w16_tmp;
- ExpandState->w16_lags[0] = w16_tmp;
- ExpandState->w16_lags[1] = w16_tmp;
- ExpandState->w16_lags[2] = w16_tmp;
- }
-
- /*
- * Unvoiced part
- */
-
- WEBRTC_SPL_MEMCPY_W16(pw16_unvoicedVec - UNVOICED_LPC_ORDER,
- ExpandState->pw16_arState,
- UNVOICED_LPC_ORDER);
- if (ExpandState->w16_arGainScale > 0)
- {
- w32_tmp = ((WebRtc_Word32) 1) << (ExpandState->w16_arGainScale - 1);
- }
- else
- {
- w32_tmp = 0;
- }
-
- /* Note that shift value can be >16 which complicates things for some DSPs */
- WebRtcSpl_AffineTransformVector(pw16_scaledRandVec, pw16_randVec,
- ExpandState->w16_arGain, w32_tmp, ExpandState->w16_arGainScale, w16_lag);
-
- WebRtcSpl_FilterARFastQ12(pw16_scaledRandVec, pw16_unvoicedVec,
- ExpandState->pw16_arFilter, UNVOICED_LPC_ORDER + 1, w16_lag);
-
- WEBRTC_SPL_MEMCPY_W16(ExpandState->pw16_arState,
- &(pw16_unvoicedVec[w16_lag - UNVOICED_LPC_ORDER]),
- UNVOICED_LPC_ORDER);
-
- /*
- * Voiced + Unvoiced
- */
-
- /* For lag =
- <=31*fs_mult => go from 1 to 0 in about 8 ms
- (>=31..<=63)*fs_mult => go from 1 to 0 in about 16 ms
- >=64*fs_mult => go from 1 to 0 in about 32 ms
- */
- w16_tmp = (31 - WebRtcSpl_NormW32(ExpandState->w16_maxLag)) - 5; /* getbits(w16_maxLag) -5 */
- w16_vfractionChange = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(256, w16_tmp);
- if (ExpandState->w16_stopMuting == 1)
- {
- w16_vfractionChange = 0;
- }
-
- /* Create combined signal (unmuted) by shifting in more and more of unvoiced part */
- w16_tmp = 8 - w16_tmp; /* getbits(w16_vfractionChange) */
- w16_tmp = (ExpandState->w16_currentVFraction - ExpandState->w16_vFraction) >> w16_tmp;
- w16_tmp = WEBRTC_SPL_MIN(w16_tmp, w16_lag);
- WebRtcNetEQ_MixVoiceUnvoice(pw16_outData, pw16_voicedVec, pw16_unvoicedVec,
- &ExpandState->w16_currentVFraction, w16_vfractionChange, w16_tmp);
-
- if (w16_tmp < w16_lag)
- {
- if (w16_vfractionChange != 0)
- {
- ExpandState->w16_currentVFraction = ExpandState->w16_vFraction;
- }
- w16_tmp2 = 16384 - ExpandState->w16_currentVFraction;
- WebRtcSpl_ScaleAndAddVectorsWithRound(pw16_voicedVec + w16_tmp,
- ExpandState->w16_currentVFraction, pw16_unvoicedVec + w16_tmp, w16_tmp2, 14,
- pw16_outData + w16_tmp, (WebRtc_Word16) (w16_lag - w16_tmp));
- }
-
- /* Select muting factor */
- if (ExpandState->w16_consecExp == 3)
- {
- /* 0.95 on 50 samples in NB (0.0010/fs_mult in Q20) */
- ExpandState->w16_muteSlope = WEBRTC_SPL_MAX(ExpandState->w16_muteSlope,
- WebRtcNetEQ_k1049div[fs_mult]);
- }
- if (ExpandState->w16_consecExp == 7)
- {
- /* 0.90 on 50 samples in NB (0.0020/fs_mult in Q20) */
- ExpandState->w16_muteSlope = WEBRTC_SPL_MAX(ExpandState->w16_muteSlope,
- WebRtcNetEQ_k2097div[fs_mult]);
- }
-
- /* Mute segment according to slope value */
- if ((ExpandState->w16_consecExp != 0) || (ExpandState->w16_onset != 1))
- {
- /* Mute to the previous level, then continue with the muting */
- WebRtcSpl_AffineTransformVector(pw16_outData, pw16_outData,
- ExpandState->w16_expandMuteFactor, 8192, 14, w16_lag);
-
- if ((ExpandState->w16_stopMuting != 1))
- {
- WebRtcNetEQ_MuteSignal(pw16_outData, ExpandState->w16_muteSlope, w16_lag);
-
- w16_tmp = 16384 - (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(w16_lag,
- ExpandState->w16_muteSlope) + 8192) >> 6); /* 20-14 = 6 */
- w16_tmp = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(w16_tmp,
- ExpandState->w16_expandMuteFactor) + 8192) >> 14);
-
- /* Guard against getting stuck with very small (but sometimes audible) gain */
- if ((ExpandState->w16_consecExp > 3) && (w16_tmp
- >= ExpandState->w16_expandMuteFactor))
- {
- ExpandState->w16_expandMuteFactor = 0;
- }
- else
- {
- ExpandState->w16_expandMuteFactor = w16_tmp;
- }
- }
- }
-
- } /* end if(!BGNonly) */
-
- /*
- * BGN
- */
-
- if (BGNState->w16_initialized == 1)
- {
- /* BGN parameters are initialized; use them */
-
- WEBRTC_SPL_MEMCPY_W16(pw16_cngVec - BGN_LPC_ORDER,
- BGNState->pw16_filterState,
- BGN_LPC_ORDER);
-
- if (BGNState->w16_scaleShift > 1)
- {
- w32_tmp = ((WebRtc_Word32) 1) << (BGNState->w16_scaleShift - 1);
- }
- else
- {
- w32_tmp = 0;
- }
-
- /* Scale random vector to correct energy level */
- /* Note that shift value can be >16 which complicates things for some DSPs */
- WebRtcSpl_AffineTransformVector(pw16_scaledRandVec, pw16_randVec,
- BGNState->w16_scale, w32_tmp, BGNState->w16_scaleShift, w16_lag);
-
- WebRtcSpl_FilterARFastQ12(pw16_scaledRandVec, pw16_cngVec, BGNState->pw16_filter,
- BGN_LPC_ORDER + 1, w16_lag);
-
- WEBRTC_SPL_MEMCPY_W16(BGNState->pw16_filterState,
- &(pw16_cngVec[w16_lag-BGN_LPC_ORDER]),
- BGN_LPC_ORDER);
-
- /* Unmute the insertion of background noise */
-
- if (bgnMode == BGN_FADE && ExpandState->w16_consecExp >= FADE_BGN_TIME
- && BGNState->w16_mutefactor > 0)
- {
- /* fade BGN to zero */
- /* calculate muting slope, approx 2^18/fsHz */
- WebRtc_Word16 muteFactor;
- if (fs_mult == 1)
- {
- muteFactor = -32;
- }
- else if (fs_mult == 2)
- {
- muteFactor = -16;
- }
- else if (fs_mult == 4)
- {
- muteFactor = -8;
- }
- else
- {
- muteFactor = -5;
- }
- /* use UnmuteSignal function with negative slope */
- WebRtcNetEQ_UnmuteSignal(pw16_cngVec, &BGNState->w16_mutefactor, /* In Q14 */
- pw16_cngVec, muteFactor, /* In Q20 */
- w16_lag);
- }
- else if (BGNState->w16_mutefactor < 16384 && !BGNonly)
- {
- /* if (w16_mutefactor < 1) and not BGN only (since then we use no muting) */
-
- /*
- * If BGN_OFF, or if BNG_FADE has started fading,
- * mutefactor should not be increased.
- */
- if (ExpandState->w16_stopMuting != 1 && bgnMode != BGN_OFF && !(bgnMode
- == BGN_FADE && ExpandState->w16_consecExp >= FADE_BGN_TIME))
- {
- WebRtcNetEQ_UnmuteSignal(pw16_cngVec, &BGNState->w16_mutefactor, /* In Q14 */
- pw16_cngVec, ExpandState->w16_muteSlope, /* In Q20 */
- w16_lag);
- }
- else
- {
- /* BGN_ON and stop muting, or
- * BGN_OFF (mute factor is always 0), or
- * BGN_FADE has reached 0 */
- WebRtcSpl_AffineTransformVector(pw16_cngVec, pw16_cngVec,
- BGNState->w16_mutefactor, 8192, 14, w16_lag);
- }
- }
- }
- else
- {
- /* BGN parameters have not been initialized; use zero noise */
- WebRtcSpl_MemSetW16(pw16_cngVec, 0, w16_lag);
- }
-
- if (BGNonly)
- {
- /* Copy BGN to outdata */
- for (i = 0; i < w16_lag; i++)
- {
- pw16_outData[i] = pw16_cngVec[i];
- }
- }
- else
- {
- /* Add CNG vector to the Voiced + Unvoiced vectors */
- for (i = 0; i < w16_lag; i++)
- {
- pw16_outData[i] = pw16_outData[i] + pw16_cngVec[i];
- }
-
- /* increase call number */
- ExpandState->w16_consecExp = ExpandState->w16_consecExp + 1;
- if (ExpandState->w16_consecExp < 0) /* Guard against overflow */
- ExpandState->w16_consecExp = FADE_BGN_TIME; /* "Arbitrary" large num of expands */
- }
-
- inst->w16_mode = MODE_EXPAND;
- *pw16_len = w16_lag;
-
- /* Update in-call and post-call statistics */
- if (ExpandState->w16_stopMuting != 1 || BGNonly)
- {
- /*
- * Only do this if StopMuting != 1 or if explicitly BGNonly, otherwise Expand is
- * called from Merge or Normal and special measures must be taken.
- */
- inst->statInst.expandLength += (WebRtc_UWord32) *pw16_len;
- if (ExpandState->w16_expandMuteFactor == 0 || BGNonly)
- {
- /* Only noise expansion */
- inst->statInst.expandedNoiseSamples += *pw16_len;
- }
- else
- {
- /* Voice expand (note: not necessarily _voiced_) */
- inst->statInst.expandedVoiceSamples += *pw16_len;
- }
- }
-
- return 0;
-}
-
-/****************************************************************************
- * WebRtcNetEQ_GenerateBGN(...)
- *
- * This function generates and writes len samples of background noise to the
- * output vector. The Expand function will be called repeatedly until the
- * correct number of samples is produced.
- *
- * Input:
- * - inst : NetEq instance, i.e. the user that requests more
- * speech/audio data
- * - scratchPtr : Pointer to scratch vector
- * - len : Desired length of produced BGN.
- *
- *
- * Output:
- * - pw16_outData : Pointer to a memory space where the output data
- * should be stored
- *
- * Return value : >=0 - Number of noise samples produced and written
- * to output
- * -1 - Error
- */
-
-int WebRtcNetEQ_GenerateBGN(DSPInst_t *inst,
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_scratchPtr,
-#endif
- WebRtc_Word16 *pw16_outData, WebRtc_Word16 len)
-{
-
- WebRtc_Word16 pos = 0;
- WebRtc_Word16 tempLen = len;
-
- while (tempLen > 0)
- {
- /* while we still need more noise samples, call Expand to obtain background noise */
- WebRtcNetEQ_Expand(inst,
-#ifdef SCRATCH
- pw16_scratchPtr,
-#endif
- &pw16_outData[pos], &tempLen, 1 /*BGNonly*/);
-
- pos += tempLen; /* we got this many samples */
- tempLen = len - pos; /* this is the number of samples we still need */
- }
-
- return pos;
-}
-
-#undef SCRATCH_PW16_BEST_CORR_INDEX
-#undef SCRATCH_PW16_BEST_CORR
-#undef SCRATCH_PW16_BEST_DIST_INDEX
-#undef SCRATCH_PW16_BEST_DIST
-#undef SCRATCH_PW16_CORR_VEC
-#undef SCRATCH_PW16_CORR2
-#undef SCRATCH_PW32_AUTO_CORR
-#undef SCRATCH_PW16_RC
-#undef SCRATCH_PW16_RAND_VEC
-#undef SCRATCH_NETEQDSP_CORRELATOR
-#undef SCRATCH_PW16_SCALED_RAND_VEC
-#undef SCRATCH_PW16_UNVOICED_VEC_SPACE
-
diff --git a/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq.h b/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq.h
deleted file mode 100644
index 909131b..0000000
--- a/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq.h
+++ /dev/null
@@ -1,215 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This is the main API for NetEQ. Helper macros are located in webrtc_neteq_help_macros.h,
- * while some internal API functions are found in webrtc_neteq_internal.h.
- */
-
-#include "typedefs.h"
-
-#ifndef WEBRTC_NETEQ_H
-#define WEBRTC_NETEQ_H
-
-#ifdef __cplusplus
-extern "C"
-{
-#endif
-
-/**********************************************************
- * Definitions
- */
-
-enum WebRtcNetEQDecoder
-{
- kDecoderReservedStart,
- kDecoderPCMu,
- kDecoderPCMa,
- kDecoderILBC,
- kDecoderISAC,
- kDecoderISACswb,
- kDecoderPCM16B,
- kDecoderPCM16Bwb,
- kDecoderPCM16Bswb32kHz,
- kDecoderPCM16Bswb48kHz,
- kDecoderG722,
- kDecoderRED,
- kDecoderAVT,
- kDecoderCNG,
- kDecoderArbitrary,
- kDecoderG729,
- kDecoderG729_1,
- kDecoderG726_16,
- kDecoderG726_24,
- kDecoderG726_32,
- kDecoderG726_40,
- kDecoderG722_1_16,
- kDecoderG722_1_24,
- kDecoderG722_1_32,
- kDecoderG722_1C_24,
- kDecoderG722_1C_32,
- kDecoderG722_1C_48,
- kDecoderSPEEX_8,
- kDecoderSPEEX_16,
- kDecoderCELT_32,
- kDecoderGSMFR,
- kDecoderAMR,
- kDecoderAMRWB,
- kDecoderReservedEnd
-};
-
-enum WebRtcNetEQNetworkType
-{
- kUDPNormal,
- kUDPVideoSync,
- kTCPNormal,
- kTCPLargeJitter,
- kTCPXLargeJitter
-};
-
-enum WebRtcNetEQOutputType
-{
- kOutputNormal,
- kOutputPLC,
- kOutputCNG,
- kOutputPLCtoCNG,
- kOutputVADPassive
-};
-
-enum WebRtcNetEQPlayoutMode
-{
- kPlayoutOn, kPlayoutOff, kPlayoutFax, kPlayoutStreaming
-};
-
-/* Available modes for background noise (inserted after long expands) */
-enum WebRtcNetEQBGNMode
-{
- kBGNOn, /* default "normal" behavior with eternal noise */
- kBGNFade, /* noise fades to zero after some time */
- kBGNOff
-/* background noise is always zero */
-};
-
-/*************************************************
- * Definitions of decoder calls and the default
- * API function calls for each codec
- */
-
-typedef WebRtc_Word16 (*WebRtcNetEQ_FuncDecode)(void* state, WebRtc_Word16* encoded,
- WebRtc_Word16 len, WebRtc_Word16* decoded,
- WebRtc_Word16* speechType);
-typedef WebRtc_Word16 (*WebRtcNetEQ_FuncDecodePLC)(void* state, WebRtc_Word16* decoded,
- WebRtc_Word16 frames);
-typedef WebRtc_Word16 (*WebRtcNetEQ_FuncDecodeInit)(void* state);
-typedef WebRtc_Word16 (*WebRtcNetEQ_FuncAddLatePkt)(void* state, WebRtc_Word16* encoded,
- WebRtc_Word16 len);
-typedef WebRtc_Word16 (*WebRtcNetEQ_FuncGetMDinfo)(void* state);
-typedef WebRtc_Word16 (*WebRtcNetEQ_FuncGetPitchInfo)(void* state, WebRtc_Word16* encoded,
- WebRtc_Word16* length);
-typedef WebRtc_Word16 (*WebRtcNetEQ_FuncUpdBWEst)(void* state, const WebRtc_UWord16 *encoded,
- WebRtc_Word32 packet_size,
- WebRtc_UWord16 rtp_seq_number,
- WebRtc_UWord32 send_ts,
- WebRtc_UWord32 arr_ts);
-typedef WebRtc_Word16 (*WebRtcNetEQ_FuncGetErrorCode)(void* state);
-
-/**********************************************************
- * Structures
- */
-
-typedef struct
-{
- enum WebRtcNetEQDecoder codec;
- WebRtc_Word16 payloadType;
- WebRtcNetEQ_FuncDecode funcDecode;
- WebRtcNetEQ_FuncDecode funcDecodeRCU;
- WebRtcNetEQ_FuncDecodePLC funcDecodePLC;
- WebRtcNetEQ_FuncDecodeInit funcDecodeInit;
- WebRtcNetEQ_FuncAddLatePkt funcAddLatePkt;
- WebRtcNetEQ_FuncGetMDinfo funcGetMDinfo;
- WebRtcNetEQ_FuncGetPitchInfo funcGetPitch;
- WebRtcNetEQ_FuncUpdBWEst funcUpdBWEst;
- WebRtcNetEQ_FuncGetErrorCode funcGetErrorCode;
- void* codec_state;
- WebRtc_UWord16 codec_fs;
-} WebRtcNetEQ_CodecDef;
-
-typedef struct
-{
- WebRtc_UWord16 fraction_lost;
- WebRtc_UWord32 cum_lost;
- WebRtc_UWord32 ext_max;
- WebRtc_UWord32 jitter;
-} WebRtcNetEQ_RTCPStat;
-
-/**********************************************************
- * NETEQ Functions
- */
-
-/* Info functions */
-
-#define WEBRTC_NETEQ_MAX_ERROR_NAME 40
-int WebRtcNetEQ_GetVersion(char *version);
-int WebRtcNetEQ_GetErrorCode(void *inst);
-int WebRtcNetEQ_GetErrorName(int errorCode, char *errorName, int maxStrLen);
-
-/* Instance memory assign functions */
-
-int WebRtcNetEQ_AssignSize(int *sizeinbytes);
-int WebRtcNetEQ_Assign(void **inst, void *NETEQ_inst_Addr);
-int WebRtcNetEQ_GetRecommendedBufferSize(void *inst, const enum WebRtcNetEQDecoder *codec,
- int noOfCodecs, enum WebRtcNetEQNetworkType nwType,
- int *MaxNoOfPackets, int *sizeinbytes);
-int WebRtcNetEQ_AssignBuffer(void *inst, int MaxNoOfPackets, void *NETEQ_Buffer_Addr,
- int sizeinbytes);
-
-/* Init functions */
-
-int WebRtcNetEQ_Init(void *inst, WebRtc_UWord16 fs);
-int WebRtcNetEQ_SetAVTPlayout(void *inst, int PlayoutAVTon);
-int WebRtcNetEQ_SetExtraDelay(void *inst, int DelayInMs);
-int WebRtcNetEQ_SetPlayoutMode(void *inst, enum WebRtcNetEQPlayoutMode playoutMode);
-int WebRtcNetEQ_SetBGNMode(void *inst, enum WebRtcNetEQBGNMode bgnMode);
-int WebRtcNetEQ_GetBGNMode(const void *inst, enum WebRtcNetEQBGNMode *bgnMode);
-
-/* Codec Database functions */
-
-int WebRtcNetEQ_CodecDbReset(void *inst);
-int WebRtcNetEQ_CodecDbAdd(void *inst, WebRtcNetEQ_CodecDef *codecInst);
-int WebRtcNetEQ_CodecDbRemove(void *inst, enum WebRtcNetEQDecoder codec);
-int WebRtcNetEQ_CodecDbGetSizeInfo(void *inst, WebRtc_Word16 *UsedEntries,
- WebRtc_Word16 *MaxEntries);
-int WebRtcNetEQ_CodecDbGetCodecInfo(void *inst, WebRtc_Word16 Entry,
- enum WebRtcNetEQDecoder *codec);
-
-/* Real-time functions */
-
-int WebRtcNetEQ_RecIn(void *inst, WebRtc_Word16 *p_w16datagramstart, WebRtc_Word16 w16_RTPlen,
- WebRtc_UWord32 uw32_timeRec);
-int WebRtcNetEQ_RecOut(void *inst, WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len);
-int WebRtcNetEQ_GetRTCPStats(void *inst, WebRtcNetEQ_RTCPStat *RTCP_inst);
-int WebRtcNetEQ_GetRTCPStatsNoReset(void *inst, WebRtcNetEQ_RTCPStat *RTCP_inst);
-int WebRtcNetEQ_GetSpeechTimeStamp(void *inst, WebRtc_UWord32 *timestamp);
-int WebRtcNetEQ_GetSpeechOutputType(void *inst, enum WebRtcNetEQOutputType *outputType);
-
-/* VQmon related functions */
-int WebRtcNetEQ_VQmonRecOutStatistics(void *inst, WebRtc_UWord16 *validVoiceDurationMs,
- WebRtc_UWord16 *concealedVoiceDurationMs,
- WebRtc_UWord8 *concealedVoiceFlags);
-int WebRtcNetEQ_VQmonGetConfiguration(void *inst, WebRtc_UWord16 *absMaxDelayMs,
- WebRtc_UWord8 *adaptationRate);
-int WebRtcNetEQ_VQmonGetRxStatistics(void *inst, WebRtc_UWord16 *avgDelayMs,
- WebRtc_UWord16 *maxDelayMs);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif
diff --git a/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h b/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h
deleted file mode 100644
index 214bd10..0000000
--- a/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h
+++ /dev/null
@@ -1,387 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains some helper macros that can be used when loading the
- * NetEQ codec database.
- */
-
-#ifndef WEBRTC_NETEQ_HELP_MACROS_H
-#define WEBRTC_NETEQ_HELP_MACROS_H
-
-#ifndef NULL
-#define NULL 0
-#endif
-
-/**********************************************************
- * Help macros for NetEQ initialization
- */
-
-#define SET_CODEC_PAR(inst,decoder,pt,state,fs) \
- inst.codec=decoder; \
- inst.payloadType=pt; \
- inst.codec_state=state; \
- inst.codec_fs=fs;
-
-#define SET_PCMU_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG711_DecodeU; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=NULL; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_PCMA_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG711_DecodeA; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=NULL; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_ILBC_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcIlbcfix_Decode; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcIlbcfix_NetEqPlc; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcIlbcfix_Decoderinit30Ms; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_ISAC_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcIsac_Decode; \
- inst.funcDecodeRCU=(WebRtcNetEQ_FuncDecode)WebRtcIsac_DecodeRcu; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcIsac_DecoderInit; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=(WebRtcNetEQ_FuncUpdBWEst)WebRtcIsac_UpdateBwEstimate; \
- inst.funcGetErrorCode=(WebRtcNetEQ_FuncGetErrorCode)WebRtcIsac_GetErrorCode;
-
-#define SET_ISACfix_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcIsacfix_Decode; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcIsacfix_DecoderInit; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=(WebRtcNetEQ_FuncUpdBWEst)WebRtcIsacfix_UpdateBwEstimate; \
- inst.funcGetErrorCode=(WebRtcNetEQ_FuncGetErrorCode)WebRtcIsacfix_GetErrorCode;
-
-#define SET_ISACSWB_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcIsac_Decode; \
- inst.funcDecodeRCU=(WebRtcNetEQ_FuncDecode)WebRtcIsac_DecodeRcu; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcIsac_DecoderInit; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=(WebRtcNetEQ_FuncUpdBWEst)WebRtcIsac_UpdateBwEstimate; \
- inst.funcGetErrorCode=(WebRtcNetEQ_FuncGetErrorCode)WebRtcIsac_GetErrorCode;
-
-#define SET_G729_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG729_Decode; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG729_DecodePlc; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG729_DecoderInit; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_G729_1_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7291_Decode; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7291_DecoderInit; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=(WebRtcNetEQ_FuncUpdBWEst)WebRtcG7291_DecodeBwe; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_PCM16B_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcPcm16b_DecodeW16; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=NULL; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_PCM16B_WB_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcPcm16b_DecodeW16; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=NULL; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_PCM16B_SWB32_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcPcm16b_DecodeW16; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=NULL; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-
-#define SET_PCM16B_SWB48_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcPcm16b_DecodeW16; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=NULL; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_G722_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG722_Decode; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG722_DecoderInit;\
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_G722_1_16_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221_Decode16; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221_DecodePlc16; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221_DecoderInit16; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_G722_1_24_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221_Decode24; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221_DecodePlc24; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221_DecoderInit24; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_G722_1_32_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221_Decode32; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221_DecodePlc32; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221_DecoderInit32; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_G722_1C_24_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221C_Decode24; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221C_DecodePlc24; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221C_DecoderInit24; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_G722_1C_32_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221C_Decode32; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221C_DecodePlc32; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221C_DecoderInit32; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_G722_1C_48_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG7221C_Decode48; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcG7221C_DecodePlc48; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG7221C_DecoderInit48; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_AMR_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcAmr_Decode; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcAmr_DecodePlc; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcAmr_DecoderInit; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_AMRWB_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcAmrWb_Decode; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcAmrWb_DecodePlc; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcAmrWb_DecoderInit; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_GSMFR_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcGSMFR_Decode; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcGSMFR_DecodePlc; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcGSMFR_DecoderInit; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_G726_16_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG726_decode16; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG726_decoderinit16; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_G726_24_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG726_decode24; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG726_decoderinit24; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_G726_32_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG726_decode32; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG726_decoderinit32; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_G726_40_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcG726_decode40; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcG726_decoderinit40; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_SPEEX_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcSpeex_Decode; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcSpeex_DecodePlc; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcSpeex_DecoderInit; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_CELT_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcCelt_Decode; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcCelt_DecodePlc; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcCelt_DecoderInit; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_CELTSLAVE_FUNCTIONS(inst) \
- inst.funcDecode=(WebRtcNetEQ_FuncDecode)WebRtcCelt_DecodeSlave; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=(WebRtcNetEQ_FuncDecodePLC)WebRtcCelt_DecodePlc; \
- inst.funcDecodeInit=(WebRtcNetEQ_FuncDecodeInit)WebRtcCelt_DecoderInit; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_RED_FUNCTIONS(inst) \
- inst.funcDecode=NULL; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=NULL; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_AVT_FUNCTIONS(inst) \
- inst.funcDecode=NULL; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=NULL; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#define SET_CNG_FUNCTIONS(inst) \
- inst.funcDecode=NULL; \
- inst.funcDecodeRCU=NULL; \
- inst.funcDecodePLC=NULL; \
- inst.funcDecodeInit=NULL; \
- inst.funcAddLatePkt=NULL; \
- inst.funcGetMDinfo=NULL; \
- inst.funcGetPitch=NULL; \
- inst.funcUpdBWEst=NULL; \
- inst.funcGetErrorCode=NULL;
-
-#endif /* WEBRTC_NETEQ_HELP_MACROS_H */
-
diff --git a/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h b/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h
deleted file mode 100644
index 70ed044..0000000
--- a/trunk/src/modules/audio_coding/neteq/interface/webrtc_neteq_internal.h
+++ /dev/null
@@ -1,236 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the internal API functions.
- */
-
-#include "typedefs.h"
-
-#ifndef WEBRTC_NETEQ_INTERNAL_H
-#define WEBRTC_NETEQ_INTERNAL_H
-
-#ifdef __cplusplus
-extern "C"
-{
-#endif
-
-typedef struct
-{
- WebRtc_UWord8 payloadType;
- WebRtc_UWord16 sequenceNumber;
- WebRtc_UWord32 timeStamp;
- WebRtc_UWord32 SSRC;
- WebRtc_UWord8 markerBit;
-} WebRtcNetEQ_RTPInfo;
-
-/****************************************************************************
- * WebRtcNetEQ_RecInRTPStruct(...)
- *
- * Alternative RecIn function, used when the RTP data has already been
- * parsed into an RTP info struct (WebRtcNetEQ_RTPInfo).
- *
- * Input:
- * - inst : NetEQ instance
- * - rtpInfo : Pointer to RTP info
- * - payloadPtr : Pointer to the RTP payload (first byte after header)
- * - payloadLenBytes : Length (in bytes) of the payload in payloadPtr
- * - timeRec : Receive time (in timestamps of the used codec)
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcNetEQ_RecInRTPStruct(void *inst, WebRtcNetEQ_RTPInfo *rtpInfo,
- const WebRtc_UWord8 *payloadPtr, WebRtc_Word16 payloadLenBytes,
- WebRtc_UWord32 timeRec);
-
-/****************************************************************************
- * WebRtcNetEQ_GetMasterSlaveInfoSize(...)
- *
- * Get size in bytes for master/slave struct msInfo used in
- * WebRtcNetEQ_RecOutMasterSlave.
- *
- * Return value : Struct size in bytes
- *
- */
-
-int WebRtcNetEQ_GetMasterSlaveInfoSize();
-
-/****************************************************************************
- * WebRtcNetEQ_RecOutMasterSlave(...)
- *
- * RecOut function for running several NetEQ instances in master/slave mode.
- * One master can be used to control several slaves.
- * The MasterSlaveInfo struct must be allocated outside NetEQ.
- * Use function WebRtcNetEQ_GetMasterSlaveInfoSize to get the size needed.
- *
- * Input:
- * - inst : NetEQ instance
- * - isMaster : Non-zero indicates that this is the master channel
- * - msInfo : (slave only) Information from master
- *
- * Output:
- * - inst : Updated NetEQ instance
- * - pw16_outData : Pointer to vector where output should be written
- * - pw16_len : Pointer to variable where output length is returned
- * - msInfo : (master only) Information to slave(s)
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_RecOutMasterSlave(void *inst, WebRtc_Word16 *pw16_outData,
- WebRtc_Word16 *pw16_len, void *msInfo,
- WebRtc_Word16 isMaster);
-
-typedef struct
-{
- uint16_t currentBufferSize; /* Current jitter buffer size in ms. */
- uint16_t preferredBufferSize; /* Preferred buffer size in ms. */
- uint16_t jitterPeaksFound; /* 1 if adding extra delay due to peaky
- * jitter; 0 otherwise. */
- uint16_t currentPacketLossRate; /* Loss rate (network + late) (Q14). */
- uint16_t currentDiscardRate; /* Late loss rate (Q14). */
- uint16_t currentExpandRate; /* Fraction (of original stream) of
- * synthesized speech inserted through
- * expansion (in Q14). */
- uint16_t currentPreemptiveRate; /* Fraction of data inserted through
- * pre-emptive expansion (in Q14). */
- uint16_t currentAccelerateRate; /* Fraction of data removed through
- * acceleration (in Q14). */
- int32_t clockDriftPPM; /* Average clock-drift in parts-per-
- * million (positive or negative). */
-} WebRtcNetEQ_NetworkStatistics;
-
-/*
- * Get the "in-call" statistics from NetEQ.
- * The statistics are reset after the query.
- */
-int WebRtcNetEQ_GetNetworkStatistics(void *inst, WebRtcNetEQ_NetworkStatistics *stats);
-
-/*
- * Get the raw waiting times for decoded frames. The function writes the last
- * recorded waiting times (from frame arrival to frame decoding) to the memory
- * pointed to by waitingTimeMs. The number of elements written is in the return
- * value. No more than maxLength elements are written. Statistics are reset on
- * each query.
- */
-int WebRtcNetEQ_GetRawFrameWaitingTimes(void *inst,
- int max_length,
- int* waiting_times_ms);
-
-/***********************************************/
-/* Functions for post-decode VAD functionality */
-/***********************************************/
-
-/* NetEQ must be compiled with the flag NETEQ_VAD enabled for these functions to work. */
-
-/*
- * VAD function pointer types
- *
- * These function pointers match the definitions of webrtc VAD functions WebRtcVad_Init,
- * WebRtcVad_set_mode and WebRtcVad_Process, respectively, all found in webrtc_vad.h.
- */
-typedef int (*WebRtcNetEQ_VADInitFunction)(void *VAD_inst);
-typedef int (*WebRtcNetEQ_VADSetmodeFunction)(void *VAD_inst, int mode);
-typedef WebRtc_Word16 (*WebRtcNetEQ_VADFunction)(void *VAD_inst, WebRtc_Word16 fs,
- WebRtc_Word16 *frame, WebRtc_Word16 frameLen);
-
-/****************************************************************************
- * WebRtcNetEQ_SetVADInstance(...)
- *
- * Provide a pointer to an allocated VAD instance. If function is never
- * called or it is called with NULL pointer as VAD_inst, the post-decode
- * VAD functionality is disabled. Also provide pointers to init, setmode
- * and VAD functions. These are typically pointers to WebRtcVad_Init,
- * WebRtcVad_set_mode and WebRtcVad_Process, respectively, all found in the
- * interface file webrtc_vad.h.
- *
- * Input:
- * - NetEQ_inst : NetEQ instance
- * - VADinst : VAD instance
- * - initFunction : Pointer to VAD init function
- * - setmodeFunction : Pointer to VAD setmode function
- * - VADfunction : Pointer to VAD function
- *
- * Output:
- * - NetEQ_inst : Updated NetEQ instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_SetVADInstance(void *NetEQ_inst, void *VAD_inst,
- WebRtcNetEQ_VADInitFunction initFunction,
- WebRtcNetEQ_VADSetmodeFunction setmodeFunction,
- WebRtcNetEQ_VADFunction VADFunction);
-
-/****************************************************************************
- * WebRtcNetEQ_SetVADMode(...)
- *
- * Pass an aggressiveness mode parameter to the post-decode VAD instance.
- * If this function is never called, mode 0 (quality mode) is used as default.
- *
- * Input:
- * - inst : NetEQ instance
- * - mode : mode parameter (same range as WebRtc VAD mode)
- *
- * Output:
- * - inst : Updated NetEQ instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_SetVADMode(void *NetEQ_inst, int mode);
-
-/****************************************************************************
- * WebRtcNetEQ_RecOutNoDecode(...)
- *
- * Special RecOut that does not do any decoding.
- *
- * Input:
- * - inst : NetEQ instance
- *
- * Output:
- * - inst : Updated NetEQ instance
- * - pw16_outData : Pointer to vector where output should be written
- * - pw16_len : Pointer to variable where output length is returned
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_RecOutNoDecode(void *inst, WebRtc_Word16 *pw16_outData,
- WebRtc_Word16 *pw16_len);
-
-/****************************************************************************
- * WebRtcNetEQ_FlushBuffers(...)
- *
- * Flush packet and speech buffers. Does not reset codec database or
- * jitter statistics.
- *
- * Input:
- * - inst : NetEQ instance
- *
- * Output:
- * - inst : Updated NetEQ instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_FlushBuffers(void *inst);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif
diff --git a/trunk/src/modules/audio_coding/neteq/mcu.h b/trunk/src/modules/audio_coding/neteq/mcu.h
deleted file mode 100644
index 499684a..0000000
--- a/trunk/src/modules/audio_coding/neteq/mcu.h
+++ /dev/null
@@ -1,284 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * MCU struct and functions related to the MCU side operations.
- */
-
-#ifndef MCU_H
-#define MCU_H
-
-#include "typedefs.h"
-
-#include "codec_db.h"
-#include "rtcp.h"
-#include "packet_buffer.h"
-#include "buffer_stats.h"
-#include "neteq_statistics.h"
-
-#ifdef NETEQ_ATEVENT_DECODE
-#include "dtmf_buffer.h"
-#endif
-
-#define MAX_ONE_DESC 5 /* cannot do more than this many consecutive one-descriptor decodings */
-#define MAX_LOSS_REPORT_PERIOD 60 /* number of seconds between auto-reset */
-
-enum TsScaling
-{
- kTSnoScaling = 0,
- kTSscalingTwo,
- kTSscalingTwoThirds,
- kTSscalingFourThirds
-};
-
-enum { kLenWaitingTimes = 100 };
-
-typedef struct
-{
-
- WebRtc_Word16 current_Codec;
- WebRtc_Word16 current_Payload;
- WebRtc_UWord32 timeStamp; /* Next timestamp that should be played */
- WebRtc_Word16 millisecondsPerCall;
- WebRtc_UWord16 timestampsPerCall; /* Output chunk size */
- WebRtc_UWord16 fs;
- WebRtc_UWord32 ssrc; /* Current ssrc */
- WebRtc_Word16 new_codec;
- WebRtc_Word16 first_packet;
-
- /* MCU/DSP Communication layer */
- WebRtc_Word16 *pw16_readAddress;
- WebRtc_Word16 *pw16_writeAddress;
- void *main_inst;
-
- CodecDbInst_t codec_DB_inst; /* Information about all the codecs, i.e. which
- functions to use and which codpoints that
- have been assigned */
- SplitInfo_t PayloadSplit_inst; /* Information about how the current codec
- payload should be splitted */
- WebRtcNetEQ_RTCP_t RTCP_inst; /* RTCP statistics */
- PacketBuf_t PacketBuffer_inst; /* The packet buffer */
- BufstatsInst_t BufferStat_inst; /* Statistics that are used to make decision
- for what the DSP should perform */
-#ifdef NETEQ_ATEVENT_DECODE
- dtmf_inst_t DTMF_inst;
-#endif
- int NoOfExpandCalls;
- WebRtc_Word16 AVT_PlayoutOn;
- enum WebRtcNetEQPlayoutMode NetEqPlayoutMode;
-
- WebRtc_Word16 one_desc; /* Number of times running on one desc */
-
- WebRtc_UWord32 lostTS; /* Number of timestamps lost */
- WebRtc_UWord32 lastReportTS; /* Timestamp elapsed since last report was given */
-
- int waiting_times[kLenWaitingTimes]; /* Waiting time statistics storage. */
- int len_waiting_times;
- int next_waiting_time_index;
-
- WebRtc_UWord32 externalTS;
- WebRtc_UWord32 internalTS;
- WebRtc_Word16 TSscalingInitialized;
- enum TsScaling scalingFactor;
-
-#ifdef NETEQ_STEREO
- int usingStereo;
-#endif
-
-} MCUInst_t;
-
-/****************************************************************************
- * WebRtcNetEQ_McuReset(...)
- *
- * Reset the MCU instance.
- *
- * Input:
- * - inst : MCU instance
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-int WebRtcNetEQ_McuReset(MCUInst_t *inst);
-
-/****************************************************************************
- * WebRtcNetEQ_ResetMcuInCallStats(...)
- *
- * Reset MCU-side statistics variables for the in-call statistics.
- *
- * Input:
- * - inst : MCU instance
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-int WebRtcNetEQ_ResetMcuInCallStats(MCUInst_t *inst);
-
-/****************************************************************************
- * WebRtcNetEQ_ResetWaitingTimeStats(...)
- *
- * Reset waiting-time statistics.
- *
- * Input:
- * - inst : MCU instance.
- *
- * Return value : n/a
- */
-void WebRtcNetEQ_ResetWaitingTimeStats(MCUInst_t *inst);
-
-/****************************************************************************
- * WebRtcNetEQ_LogWaitingTime(...)
- *
- * Log waiting-time to the statistics.
- *
- * Input:
- * - inst : MCU instance.
- * - waiting_time : Waiting time in "RecOut calls" (i.e., 1 call = 10 ms).
- *
- * Return value : n/a
- */
-void WebRtcNetEQ_StoreWaitingTime(MCUInst_t *inst, int waiting_time);
-
-/****************************************************************************
- * WebRtcNetEQ_ResetMcuJitterStat(...)
- *
- * Reset MCU-side statistics variables for the post-call statistics.
- *
- * Input:
- * - inst : MCU instance
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-int WebRtcNetEQ_ResetMcuJitterStat(MCUInst_t *inst);
-
-/****************************************************************************
- * WebRtcNetEQ_McuAddressInit(...)
- *
- * Initializes MCU with read address and write address.
- *
- * Input:
- * - inst : MCU instance
- * - Data2McuAddress : Pointer to MCU address
- * - Data2DspAddress : Pointer to DSP address
- * - main_inst : Pointer to NetEQ main instance
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-int WebRtcNetEQ_McuAddressInit(MCUInst_t *inst, void * Data2McuAddress,
- void * Data2DspAddress, void *main_inst);
-
-/****************************************************************************
- * WebRtcNetEQ_McuSetFs(...)
- *
- * Initializes MCU with read address and write address.
- *
- * Input:
- * - inst : MCU instance
- * - fs_hz : Sample rate in Hz -- 8000, 16000, 32000, (48000)
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-int WebRtcNetEQ_McuSetFs(MCUInst_t *inst, WebRtc_UWord16 fs_hz);
-
-/****************************************************************************
- * WebRtcNetEQ_SignalMcu(...)
- *
- * Signal the MCU that data is available and ask for a RecOut decision.
- *
- * Input:
- * - inst : MCU instance
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-int WebRtcNetEQ_SignalMcu(MCUInst_t *inst);
-
-/****************************************************************************
- * WebRtcNetEQ_RecInInternal(...)
- *
- * This function inserts a packet into the jitter buffer.
- *
- * Input:
- * - MCU_inst : MCU instance
- * - RTPpacket : The RTP packet, parsed into NetEQ's internal RTP struct
- * - uw32_timeRec : Time stamp for the arrival of the packet (not RTP timestamp)
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacket,
- WebRtc_UWord32 uw32_timeRec);
-
-/****************************************************************************
- * WebRtcNetEQ_RecInInternal(...)
- *
- * Split the packet according to split_inst and inserts the parts into
- * Buffer_inst.
- *
- * Input:
- * - MCU_inst : MCU instance
- * - RTPpacket : The RTP packet, parsed into NetEQ's internal RTP struct
- * - uw32_timeRec : Time stamp for the arrival of the packet (not RTP timestamp)
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcNetEQ_SplitAndInsertPayload(RTPPacket_t *packet, PacketBuf_t *Buffer_inst,
- SplitInfo_t *split_inst, WebRtc_Word16 *flushed);
-
-/****************************************************************************
- * WebRtcNetEQ_GetTimestampScaling(...)
- *
- * Update information about timestamp scaling for a payload type
- * in MCU_inst->scalingFactor.
- *
- * Input:
- * - MCU_inst : MCU instance
- * - rtpPayloadType : RTP payload number
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_GetTimestampScaling(MCUInst_t *MCU_inst, int rtpPayloadType);
-
-/****************************************************************************
- * WebRtcNetEQ_ScaleTimestampExternalToInternal(...)
- *
- * Convert from external to internal timestamp using current scaling info.
- *
- * Input:
- * - MCU_inst : MCU instance
- * - externalTS : External timestamp
- *
- * Return value : Internal timestamp
- */
-
-WebRtc_UWord32 WebRtcNetEQ_ScaleTimestampExternalToInternal(const MCUInst_t *MCU_inst,
- WebRtc_UWord32 externalTS);
-
-/****************************************************************************
- * WebRtcNetEQ_ScaleTimestampInternalToExternal(...)
- *
- * Convert from external to internal timestamp using current scaling info.
- *
- * Input:
- * - MCU_inst : MCU instance
- * - externalTS : Internal timestamp
- *
- * Return value : External timestamp
- */
-
-WebRtc_UWord32 WebRtcNetEQ_ScaleTimestampInternalToExternal(const MCUInst_t *MCU_inst,
- WebRtc_UWord32 internalTS);
-#endif
diff --git a/trunk/src/modules/audio_coding/neteq/mcu_address_init.c b/trunk/src/modules/audio_coding/neteq/mcu_address_init.c
deleted file mode 100644
index 0306a85..0000000
--- a/trunk/src/modules/audio_coding/neteq/mcu_address_init.c
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "mcu.h"
-
-#include <string.h> /* to define NULL */
-
-/*
- * Initializes MCU with read address and write address
- */
-int WebRtcNetEQ_McuAddressInit(MCUInst_t *inst, void * Data2McuAddress,
- void * Data2DspAddress, void *main_inst)
-{
-
- inst->pw16_readAddress = (WebRtc_Word16*) Data2McuAddress;
- inst->pw16_writeAddress = (WebRtc_Word16*) Data2DspAddress;
- inst->main_inst = main_inst;
-
- inst->millisecondsPerCall = 10;
-
- /* Do expansions in the beginning */
- if (inst->pw16_writeAddress != NULL) inst->pw16_writeAddress[0] = DSP_INSTR_EXPAND;
-
- return (0);
-}
-
diff --git a/trunk/src/modules/audio_coding/neteq/mcu_dsp_common.c b/trunk/src/modules/audio_coding/neteq/mcu_dsp_common.c
deleted file mode 100644
index 13025d4..0000000
--- a/trunk/src/modules/audio_coding/neteq/mcu_dsp_common.c
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Communication between MCU and DSP sides.
- */
-
-#include "mcu_dsp_common.h"
-
-#include <string.h>
-
-/* Initialize instances with read and write address */
-int WebRtcNetEQ_DSPinit(MainInst_t *inst)
-{
- int res = 0;
-
- res |= WebRtcNetEQ_AddressInit(&inst->DSPinst, NULL, NULL, inst);
- res |= WebRtcNetEQ_McuAddressInit(&inst->MCUinst, NULL, NULL, inst);
-
- return res;
-
-}
-
-/* The DSP side will call this function to interrupt the MCU side */
-int WebRtcNetEQ_DSP2MCUinterrupt(MainInst_t *inst, WebRtc_Word16 *pw16_shared_mem)
-{
- inst->MCUinst.pw16_readAddress = pw16_shared_mem;
- inst->MCUinst.pw16_writeAddress = pw16_shared_mem;
- return WebRtcNetEQ_SignalMcu(&inst->MCUinst);
-}
diff --git a/trunk/src/modules/audio_coding/neteq/mcu_dsp_common.h b/trunk/src/modules/audio_coding/neteq/mcu_dsp_common.h
deleted file mode 100644
index e3f4213..0000000
--- a/trunk/src/modules/audio_coding/neteq/mcu_dsp_common.h
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * The main NetEQ instance, which is where the DSP and MCU sides join.
- */
-
-#ifndef MCU_DSP_COMMON_H
-#define MCU_DSP_COMMON_H
-
-#include "typedefs.h"
-
-#include "dsp.h"
-#include "mcu.h"
-
-/* Define size of shared memory area. */
-#if defined(NETEQ_48KHZ_WIDEBAND)
- #define SHARED_MEM_SIZE (6*640)
-#elif defined(NETEQ_32KHZ_WIDEBAND)
- #define SHARED_MEM_SIZE (4*640)
-#elif defined(NETEQ_WIDEBAND)
- #define SHARED_MEM_SIZE (2*640)
-#else
- #define SHARED_MEM_SIZE 640
-#endif
-
-/* Struct to hold the NetEQ instance */
-typedef struct
-{
- DSPInst_t DSPinst; /* DSP part of the NetEQ instance */
- MCUInst_t MCUinst; /* MCU part of the NetEQ instance */
- WebRtc_Word16 ErrorCode; /* Store last error code */
-#ifdef NETEQ_STEREO
- WebRtc_Word16 masterSlave; /* 0 = not set, 1 = master, 2 = slave */
-#endif /* NETEQ_STEREO */
-} MainInst_t;
-
-/* Struct used for communication between DSP and MCU sides of NetEQ */
-typedef struct
-{
- WebRtc_UWord32 playedOutTS; /* Timestamp position at end of DSP data */
- WebRtc_UWord16 samplesLeft; /* Number of samples stored */
- WebRtc_Word16 MD; /* Multiple description codec information */
- WebRtc_Word16 lastMode; /* Latest mode of NetEQ playout */
- WebRtc_Word16 frameLen; /* Frame length of previously decoded packet */
-} DSP2MCU_info_t;
-
-/* Initialize instances with read and write address */
-int WebRtcNetEQ_DSPinit(MainInst_t *inst);
-
-/* The DSP side will call this function to interrupt the MCU side */
-int WebRtcNetEQ_DSP2MCUinterrupt(MainInst_t *inst, WebRtc_Word16 *pw16_shared_mem);
-
-#endif
diff --git a/trunk/src/modules/audio_coding/neteq/mcu_reset.c b/trunk/src/modules/audio_coding/neteq/mcu_reset.c
deleted file mode 100644
index 3aae4ce..0000000
--- a/trunk/src/modules/audio_coding/neteq/mcu_reset.c
+++ /dev/null
@@ -1,128 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Reset MCU side data.
- */
-
-#include "mcu.h"
-
-#include <assert.h>
-#include <string.h>
-
-#include "automode.h"
-
-int WebRtcNetEQ_McuReset(MCUInst_t *inst)
-{
-
-#ifdef NETEQ_ATEVENT_DECODE
- int ok;
-#endif
-
- /* MCU/DSP Communication layer */
- inst->pw16_readAddress = NULL;
- inst->pw16_writeAddress = NULL;
- inst->main_inst = NULL;
- inst->one_desc = 0;
- inst->BufferStat_inst.Automode_inst.extraDelayMs = 0;
- inst->NetEqPlayoutMode = kPlayoutOn;
-
- WebRtcNetEQ_DbReset(&inst->codec_DB_inst);
- memset(&inst->PayloadSplit_inst, 0, sizeof(SplitInfo_t));
-
- /* Clear the Packet buffer and the pointer to memory storage */
- WebRtcNetEQ_PacketBufferFlush(&inst->PacketBuffer_inst);
- inst->PacketBuffer_inst.memorySizeW16 = 0;
- inst->PacketBuffer_inst.maxInsertPositions = 0;
-
- /* Clear the decision and delay history */
- memset(&inst->BufferStat_inst, 0, sizeof(BufstatsInst_t));
-#ifdef NETEQ_ATEVENT_DECODE
- ok = WebRtcNetEQ_DtmfDecoderInit(&inst->DTMF_inst, 8000, 560);
- if (ok != 0)
- {
- return ok;
- }
-#endif
- inst->NoOfExpandCalls = 0;
- inst->current_Codec = -1;
- inst->current_Payload = -1;
-
- inst->millisecondsPerCall = 10;
- inst->timestampsPerCall = inst->millisecondsPerCall * 8;
- inst->fs = 8000;
- inst->first_packet = 1;
-
- WebRtcNetEQ_ResetMcuInCallStats(inst);
-
- WebRtcNetEQ_ResetWaitingTimeStats(inst);
-
- WebRtcNetEQ_ResetMcuJitterStat(inst);
-
- WebRtcNetEQ_ResetAutomode(&(inst->BufferStat_inst.Automode_inst),
- inst->PacketBuffer_inst.maxInsertPositions);
-
- return 0;
-}
-
-/*
- * Reset MCU-side statistics variables for the in-call statistics.
- */
-
-int WebRtcNetEQ_ResetMcuInCallStats(MCUInst_t *inst)
-{
- inst->lostTS = 0;
- inst->lastReportTS = 0;
- inst->PacketBuffer_inst.discardedPackets = 0;
-
- return 0;
-}
-
-/*
- * Reset waiting-time statistics.
- */
-
-void WebRtcNetEQ_ResetWaitingTimeStats(MCUInst_t *inst) {
- memset(inst->waiting_times, 0,
- kLenWaitingTimes * sizeof(inst->waiting_times[0]));
- inst->len_waiting_times = 0;
- inst->next_waiting_time_index = 0;
-}
-
-/*
- * Store waiting-time in the statistics.
- */
-
-void WebRtcNetEQ_StoreWaitingTime(MCUInst_t *inst, int waiting_time) {
- assert(inst->next_waiting_time_index < kLenWaitingTimes);
- inst->waiting_times[inst->next_waiting_time_index] = waiting_time;
- inst->next_waiting_time_index++;
- if (inst->next_waiting_time_index >= kLenWaitingTimes) {
- inst->next_waiting_time_index = 0;
- }
- if (inst->len_waiting_times < kLenWaitingTimes) {
- inst->len_waiting_times++;
- }
-}
-
-/*
- * Reset all MCU-side statistics variables for the post-call statistics.
- */
-
-int WebRtcNetEQ_ResetMcuJitterStat(MCUInst_t *inst)
-{
- inst->BufferStat_inst.Automode_inst.countIAT500ms = 0;
- inst->BufferStat_inst.Automode_inst.countIAT1000ms = 0;
- inst->BufferStat_inst.Automode_inst.countIAT2000ms = 0;
- inst->BufferStat_inst.Automode_inst.longestIATms = 0;
-
- return 0;
-}
-
diff --git a/trunk/src/modules/audio_coding/neteq/merge.c b/trunk/src/modules/audio_coding/neteq/merge.c
deleted file mode 100644
index 5f020a9..0000000
--- a/trunk/src/modules/audio_coding/neteq/merge.c
+++ /dev/null
@@ -1,564 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This is the function to merge a new packet with expanded data after a packet loss.
- */
-
-#include "dsp.h"
-
-#include "signal_processing_library.h"
-
-#include "dsp_helpfunctions.h"
-#include "neteq_error_codes.h"
-
-/****************************************************************************
- * WebRtcNetEQ_Merge(...)
- *
- * This function...
- *
- * Input:
- * - inst : NetEQ DSP instance
- * - scratchPtr : Pointer to scratch vector.
- * - decoded : Pointer to new decoded speech.
- * - len : Number of samples in pw16_decoded.
- *
- *
- * Output:
- * - inst : Updated user information
- * - outData : Pointer to a memory space where the output data
- * should be stored
- * - pw16_len : Number of samples written to pw16_outData
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-
-/* Scratch usage:
-
- Type Name size startpos endpos
- WebRtc_Word16 pw16_expanded 210*fs/8000 0 209*fs/8000
- WebRtc_Word16 pw16_expandedLB 100 210*fs/8000 99+210*fs/8000
- WebRtc_Word16 pw16_decodedLB 40 100+210*fs/8000 139+210*fs/8000
- WebRtc_Word32 pw32_corr 2*60 140+210*fs/8000 260+210*fs/8000
- WebRtc_Word16 pw16_corrVec 68 210*fs/8000 67+210*fs/8000
-
- [gap in scratch vector]
-
- func WebRtcNetEQ_Expand 40+370*fs/8000 126*fs/8000 39+496*fs/8000
-
- Total: 40+496*fs/8000
- */
-
-#define SCRATCH_pw16_expanded 0
-#if (defined(NETEQ_48KHZ_WIDEBAND))
-#define SCRATCH_pw16_expandedLB 1260
-#define SCRATCH_pw16_decodedLB 1360
-#define SCRATCH_pw32_corr 1400
-#define SCRATCH_pw16_corrVec 1260
-#define SCRATCH_NETEQ_EXPAND 756
-#elif (defined(NETEQ_32KHZ_WIDEBAND))
-#define SCRATCH_pw16_expandedLB 840
-#define SCRATCH_pw16_decodedLB 940
-#define SCRATCH_pw32_corr 980
-#define SCRATCH_pw16_corrVec 840
-#define SCRATCH_NETEQ_EXPAND 504
-#elif (defined(NETEQ_WIDEBAND))
-#define SCRATCH_pw16_expandedLB 420
-#define SCRATCH_pw16_decodedLB 520
-#define SCRATCH_pw32_corr 560
-#define SCRATCH_pw16_corrVec 420
-#define SCRATCH_NETEQ_EXPAND 252
-#else /* NB */
-#define SCRATCH_pw16_expandedLB 210
-#define SCRATCH_pw16_decodedLB 310
-#define SCRATCH_pw32_corr 350
-#define SCRATCH_pw16_corrVec 210
-#define SCRATCH_NETEQ_EXPAND 126
-#endif
-
-int WebRtcNetEQ_Merge(DSPInst_t *inst,
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_scratchPtr,
-#endif
- WebRtc_Word16 *pw16_decoded, int len, WebRtc_Word16 *pw16_outData,
- WebRtc_Word16 *pw16_len)
-{
-
- WebRtc_Word16 fs_mult;
- WebRtc_Word16 fs_shift;
- WebRtc_Word32 w32_En_new_frame, w32_En_old_frame;
- WebRtc_Word16 w16_expmax, w16_newmax;
- WebRtc_Word16 w16_tmp, w16_tmp2;
- WebRtc_Word32 w32_tmp;
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_expanded = pw16_scratchPtr + SCRATCH_pw16_expanded;
- WebRtc_Word16 *pw16_expandedLB = pw16_scratchPtr + SCRATCH_pw16_expandedLB;
- WebRtc_Word16 *pw16_decodedLB = pw16_scratchPtr + SCRATCH_pw16_decodedLB;
- WebRtc_Word32 *pw32_corr = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_pw32_corr);
- WebRtc_Word16 *pw16_corrVec = pw16_scratchPtr + SCRATCH_pw16_corrVec;
-#else
- WebRtc_Word16 pw16_expanded[(125+80+5)*FSMULT];
- WebRtc_Word16 pw16_expandedLB[100];
- WebRtc_Word16 pw16_decodedLB[40];
- WebRtc_Word32 pw32_corr[60];
- WebRtc_Word16 pw16_corrVec[4+60+4];
-#endif
- WebRtc_Word16 *pw16_corr = &pw16_corrVec[4];
- WebRtc_Word16 w16_stopPos = 0, w16_bestIndex, w16_interpLen;
- WebRtc_Word16 w16_bestVal; /* bestVal is dummy */
- WebRtc_Word16 w16_startfact, w16_inc;
- WebRtc_Word16 w16_expandedLen;
- WebRtc_Word16 w16_startPos;
- WebRtc_Word16 w16_expLen, w16_newLen = 0;
- WebRtc_Word16 *pw16_decodedOut;
- WebRtc_Word16 w16_muted;
-
- int w16_decodedLen = len;
-
-#ifdef NETEQ_STEREO
- MasterSlaveInfo *msInfo = inst->msInfo;
-#endif
-
- fs_mult = WebRtcSpl_DivW32W16ResW16(inst->fs, 8000);
- fs_shift = 30 - WebRtcSpl_NormW32(fs_mult); /* Note that this is not "exact" for 48kHz */
-
- /*************************************
- * Generate data to merge with
- *************************************/
- /*
- * Check how much data that is left since earlier
- * (at least there should be the overlap)...
- */
- w16_startPos = inst->endPosition - inst->curPosition;
- /* Get one extra expansion to merge and overlap with */
- inst->ExpandInst.w16_stopMuting = 1;
- inst->ExpandInst.w16_lagsDirection = 1; /* make sure we get the "optimal" lag */
- inst->ExpandInst.w16_lagsPosition = -1; /* out of the 3 possible ones */
- w16_expandedLen = 0; /* Does not fill any function currently */
-
- if (w16_startPos >= 210 * FSMULT)
- {
- /*
- * The number of samples available in the sync buffer is more than what fits in
- * pw16_expanded.Keep the first 210*FSMULT samples, but shift them towards the end of
- * the buffer. This is ok, since all of the buffer will be expand data anyway, so as
- * long as the beginning is left untouched, we're fine.
- */
-
- w16_tmp = w16_startPos - 210 * FSMULT; /* length difference */
-
- WEBRTC_SPL_MEMMOVE_W16(&inst->speechBuffer[inst->curPosition+w16_tmp] ,
- &inst->speechBuffer[inst->curPosition], 210*FSMULT);
-
- inst->curPosition += w16_tmp; /* move start position of sync buffer accordingly */
- w16_startPos = 210 * FSMULT; /* this is the truncated length */
- }
-
- WebRtcNetEQ_Expand(inst,
-#ifdef SCRATCH
- pw16_scratchPtr + SCRATCH_NETEQ_EXPAND,
-#endif
- pw16_expanded, /* let Expand write to beginning of pw16_expanded to avoid overflow */
- &w16_newLen, 0);
-
- /*
- * Now shift the data in pw16_expanded to where it belongs.
- * Truncate all that ends up outside the vector.
- */
-
- WEBRTC_SPL_MEMMOVE_W16(&pw16_expanded[w16_startPos], pw16_expanded,
- WEBRTC_SPL_MIN(w16_newLen,
- WEBRTC_SPL_MAX(210*FSMULT - w16_startPos, 0) ) );
-
- inst->ExpandInst.w16_stopMuting = 0;
-
- /* Copy what is left since earlier into the expanded vector */
-
- WEBRTC_SPL_MEMCPY_W16(pw16_expanded, &inst->speechBuffer[inst->curPosition], w16_startPos);
-
- /*
- * Do "ugly" copy and paste from the expanded in order to generate more data
- * to correlate (but not interpolate) with.
- */
- w16_expandedLen = (120 + 80 + 2) * fs_mult;
- w16_expLen = w16_startPos + w16_newLen;
-
- if (w16_expLen < w16_expandedLen)
- {
- while ((w16_expLen + w16_newLen) < w16_expandedLen)
- {
- WEBRTC_SPL_MEMCPY_W16(&pw16_expanded[w16_expLen], &pw16_expanded[w16_startPos],
- w16_newLen);
- w16_expLen += w16_newLen;
- }
-
- /* Copy last part (fraction of a whole expansion) */
-
- WEBRTC_SPL_MEMCPY_W16(&pw16_expanded[w16_expLen], &pw16_expanded[w16_startPos],
- (w16_expandedLen-w16_expLen));
- }
- w16_expLen = w16_expandedLen;
-
- /* Adjust muting factor (main muting factor times expand muting factor) */
- inst->w16_muteFactor
- = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(inst->w16_muteFactor,
- inst->ExpandInst.w16_expandMuteFactor, 14);
-
- /* Adjust muting factor if new vector is more or less of the BGN energy */
- len = WEBRTC_SPL_MIN(64*fs_mult, w16_decodedLen);
- w16_expmax = WebRtcSpl_MaxAbsValueW16(pw16_expanded, (WebRtc_Word16) len);
- w16_newmax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (WebRtc_Word16) len);
-
- /* Calculate energy of old data */
- w16_tmp = 6 + fs_shift - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_expmax, w16_expmax));
- w16_tmp = WEBRTC_SPL_MAX(w16_tmp,0);
- w32_En_old_frame = WebRtcNetEQ_DotW16W16(pw16_expanded, pw16_expanded, len, w16_tmp);
-
- /* Calculate energy of new data */
- w16_tmp2 = 6 + fs_shift - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_newmax, w16_newmax));
- w16_tmp2 = WEBRTC_SPL_MAX(w16_tmp2,0);
- w32_En_new_frame = WebRtcNetEQ_DotW16W16(pw16_decoded, pw16_decoded, len, w16_tmp2);
-
- /* Align to same Q-domain */
- if (w16_tmp2 > w16_tmp)
- {
- w32_En_old_frame = WEBRTC_SPL_RSHIFT_W32(w32_En_old_frame, (w16_tmp2-w16_tmp));
- }
- else
- {
- w32_En_new_frame = WEBRTC_SPL_RSHIFT_W32(w32_En_new_frame, (w16_tmp-w16_tmp2));
- }
-
- /* Calculate muting factor to use for new frame */
- if (w32_En_new_frame > w32_En_old_frame)
- {
- /* Normalize w32_En_new_frame to 14 bits */
- w16_tmp = WebRtcSpl_NormW32(w32_En_new_frame) - 17;
- w32_En_new_frame = WEBRTC_SPL_SHIFT_W32(w32_En_new_frame, w16_tmp);
-
- /*
- * Put w32_En_old_frame in a domain 14 higher, so that
- * w32_En_old_frame/w32_En_new_frame is in Q14
- */
- w16_tmp = w16_tmp + 14;
- w32_En_old_frame = WEBRTC_SPL_SHIFT_W32(w32_En_old_frame, w16_tmp);
- w16_tmp
- = WebRtcSpl_DivW32W16ResW16(w32_En_old_frame, (WebRtc_Word16) w32_En_new_frame);
- /* Calculate sqrt(w32_En_old_frame/w32_En_new_frame) in Q14 */
- w16_muted = (WebRtc_Word16) WebRtcSpl_SqrtFloor(
- WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)w16_tmp,14));
- }
- else
- {
- w16_muted = 16384; /* Set = 1.0 when old frame has higher energy than new */
- }
-
- /* Set the raise the continued muting factor w16_muted if w16_muteFactor is lower */
- if (w16_muted > inst->w16_muteFactor)
- {
- inst->w16_muteFactor = WEBRTC_SPL_MIN(w16_muted, 16384);
- }
-
-#ifdef NETEQ_STEREO
-
- /* Sanity for msInfo */
- if (msInfo == NULL)
- {
- /* this should not happen here */
- return MASTER_SLAVE_ERROR;
- }
-
- /* do not downsample and calculate correlations for slave instance(s) */
- if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO))
- {
-#endif
-
- /*********************************************
- * Downsample to 4kHz and find best overlap
- *********************************************/
-
- /* Downsample to 4 kHz */
- if (inst->fs == 8000)
- {
- WebRtcSpl_DownsampleFast(&pw16_expanded[2], (WebRtc_Word16) (w16_expandedLen - 2),
- pw16_expandedLB, (WebRtc_Word16) (100),
- (WebRtc_Word16*) WebRtcNetEQ_kDownsample8kHzTbl, (WebRtc_Word16) 3,
- (WebRtc_Word16) 2, (WebRtc_Word16) 0);
- if (w16_decodedLen <= 80)
- {
- /* Not quite long enough, so we have to cheat a bit... */
- WebRtc_Word16 temp_len = w16_decodedLen - 2;
- w16_tmp = temp_len / 2;
- WebRtcSpl_DownsampleFast(&pw16_decoded[2], temp_len,
- pw16_decodedLB, w16_tmp,
- (WebRtc_Word16*) WebRtcNetEQ_kDownsample8kHzTbl,
- (WebRtc_Word16) 3, (WebRtc_Word16) 2, (WebRtc_Word16) 0);
- WebRtcSpl_MemSetW16(&pw16_decodedLB[w16_tmp], 0, (40 - w16_tmp));
- }
- else
- {
- WebRtcSpl_DownsampleFast(&pw16_decoded[2],
- (WebRtc_Word16) (w16_decodedLen - 2), pw16_decodedLB,
- (WebRtc_Word16) (40), (WebRtc_Word16*) WebRtcNetEQ_kDownsample8kHzTbl,
- (WebRtc_Word16) 3, (WebRtc_Word16) 2, (WebRtc_Word16) 0);
- }
-#ifdef NETEQ_WIDEBAND
- }
- else if (inst->fs==16000)
- {
- WebRtcSpl_DownsampleFast(
- &pw16_expanded[4], (WebRtc_Word16)(w16_expandedLen-4),
- pw16_expandedLB, (WebRtc_Word16)(100),
- (WebRtc_Word16*)WebRtcNetEQ_kDownsample16kHzTbl, (WebRtc_Word16)5,
- (WebRtc_Word16)4, (WebRtc_Word16)0);
- if (w16_decodedLen<=160)
- {
- /* Not quite long enough, so we have to cheat a bit... */
- WebRtc_Word16 temp_len = w16_decodedLen - 4;
- w16_tmp = temp_len / 4;
- WebRtcSpl_DownsampleFast(
- &pw16_decoded[4], temp_len,
- pw16_decodedLB, w16_tmp,
- (WebRtc_Word16*)WebRtcNetEQ_kDownsample16kHzTbl, (WebRtc_Word16)5,
- (WebRtc_Word16)4, (WebRtc_Word16)0);
- WebRtcSpl_MemSetW16(&pw16_decodedLB[w16_tmp], 0, (40-w16_tmp));
- }
- else
- {
- WebRtcSpl_DownsampleFast(
- &pw16_decoded[4], (WebRtc_Word16)(w16_decodedLen-4),
- pw16_decodedLB, (WebRtc_Word16)(40),
- (WebRtc_Word16*)WebRtcNetEQ_kDownsample16kHzTbl, (WebRtc_Word16)5,
- (WebRtc_Word16)4, (WebRtc_Word16)0);
- }
-#endif
-#ifdef NETEQ_32KHZ_WIDEBAND
- }
- else if (inst->fs==32000)
- {
- /*
- * TODO(hlundin) Why is the offset into pw16_expanded 6?
- */
- WebRtcSpl_DownsampleFast(
- &pw16_expanded[6], (WebRtc_Word16)(w16_expandedLen-6),
- pw16_expandedLB, (WebRtc_Word16)(100),
- (WebRtc_Word16*)WebRtcNetEQ_kDownsample32kHzTbl, (WebRtc_Word16)7,
- (WebRtc_Word16)8, (WebRtc_Word16)0);
- if (w16_decodedLen<=320)
- {
- /* Not quite long enough, so we have to cheat a bit... */
- WebRtc_Word16 temp_len = w16_decodedLen - 6;
- w16_tmp = temp_len / 8;
- WebRtcSpl_DownsampleFast(
- &pw16_decoded[6], temp_len,
- pw16_decodedLB, w16_tmp,
- (WebRtc_Word16*)WebRtcNetEQ_kDownsample32kHzTbl, (WebRtc_Word16)7,
- (WebRtc_Word16)8, (WebRtc_Word16)0);
- WebRtcSpl_MemSetW16(&pw16_decodedLB[w16_tmp], 0, (40-w16_tmp));
- }
- else
- {
- WebRtcSpl_DownsampleFast(
- &pw16_decoded[6], (WebRtc_Word16)(w16_decodedLen-6),
- pw16_decodedLB, (WebRtc_Word16)(40),
- (WebRtc_Word16*)WebRtcNetEQ_kDownsample32kHzTbl, (WebRtc_Word16)7,
- (WebRtc_Word16)8, (WebRtc_Word16)0);
- }
-#endif
-#ifdef NETEQ_48KHZ_WIDEBAND
- }
- else /* if (inst->fs==48000) */
- {
- /*
- * TODO(hlundin) Why is the offset into pw16_expanded 6?
- */
- WebRtcSpl_DownsampleFast(
- &pw16_expanded[6], (WebRtc_Word16)(w16_expandedLen-6),
- pw16_expandedLB, (WebRtc_Word16)(100),
- (WebRtc_Word16*)WebRtcNetEQ_kDownsample48kHzTbl, (WebRtc_Word16)7,
- (WebRtc_Word16)12, (WebRtc_Word16)0);
- if (w16_decodedLen<=320)
- {
- /* Not quite long enough, so we have to cheat a bit... */
- /*
- * TODO(hlundin): Is this correct? Downsampling is a factor 12
- * but w16_tmp = temp_len / 8.
- * (Was w16_tmp = ((w16_decodedLen-6)>>3) before re-write.)
- */
- WebRtc_Word16 temp_len = w16_decodedLen - 6;
- w16_tmp = temp_len / 8;
- WebRtcSpl_DownsampleFast(
- &pw16_decoded[6], temp_len,
- pw16_decodedLB, w16_tmp,
- (WebRtc_Word16*)WebRtcNetEQ_kDownsample48kHzTbl, (WebRtc_Word16)7,
- (WebRtc_Word16)12, (WebRtc_Word16)0);
- WebRtcSpl_MemSetW16(&pw16_decodedLB[w16_tmp], 0, (40-w16_tmp));
- }
- else
- {
- WebRtcSpl_DownsampleFast(
- &pw16_decoded[6], (WebRtc_Word16)(w16_decodedLen-6),
- pw16_decodedLB, (WebRtc_Word16)(40),
- (WebRtc_Word16*)WebRtcNetEQ_kDownsample48kHzTbl, (WebRtc_Word16)7,
- (WebRtc_Word16)12, (WebRtc_Word16)0);
- }
-#endif
- }
-
- /* Calculate correlation without any normalization (40 samples) */
- w16_tmp = WebRtcSpl_DivW32W16ResW16((WebRtc_Word32) inst->ExpandInst.w16_maxLag,
- (WebRtc_Word16) (fs_mult * 2)) + 1;
- w16_stopPos = WEBRTC_SPL_MIN(60, w16_tmp);
- w32_tmp = WEBRTC_SPL_MUL_16_16(w16_expmax, w16_newmax);
- if (w32_tmp > 26843546)
- {
- w16_tmp = 3;
- }
- else
- {
- w16_tmp = 0;
- }
-
- WebRtcNetEQ_CrossCorr(pw32_corr, pw16_decodedLB, pw16_expandedLB, 40,
- (WebRtc_Word16) w16_stopPos, w16_tmp, 1);
-
- /* Normalize correlation to 14 bits and put in a WebRtc_Word16 vector */
- WebRtcSpl_MemSetW16(pw16_corrVec, 0, (4 + 60 + 4));
- w32_tmp = WebRtcSpl_MaxAbsValueW32(pw32_corr, w16_stopPos);
- w16_tmp = 17 - WebRtcSpl_NormW32(w32_tmp);
- w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
-
- WebRtcSpl_VectorBitShiftW32ToW16(pw16_corr, w16_stopPos, pw32_corr, w16_tmp);
-
- /* Calculate allowed starting point for peak finding.
- The peak location bestIndex must fulfill two criteria:
- (1) w16_bestIndex+w16_decodedLen < inst->timestampsPerCall+inst->ExpandInst.w16_overlap
- (2) w16_bestIndex+w16_decodedLen < w16_startPos */
- w16_tmp = WEBRTC_SPL_MAX(0, WEBRTC_SPL_MAX(w16_startPos,
- inst->timestampsPerCall+inst->ExpandInst.w16_overlap) - w16_decodedLen);
- /* Downscale starting index to 4kHz domain */
- w16_tmp2 = WebRtcSpl_DivW32W16ResW16((WebRtc_Word32) w16_tmp,
- (WebRtc_Word16) (fs_mult << 1));
-
-#ifdef NETEQ_STEREO
- } /* end if (msInfo->msMode != NETEQ_SLAVE) */
-
- if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO))
- {
- /* This is master or mono instance; find peak */
- WebRtcNetEQ_PeakDetection(&pw16_corr[w16_tmp2], w16_stopPos, 1, fs_mult, &w16_bestIndex,
- &w16_bestVal);
- w16_bestIndex += w16_tmp; /* compensate for modified starting index */
- msInfo->bestIndex = w16_bestIndex;
- }
- else if (msInfo->msMode == NETEQ_SLAVE)
- {
- /* Get peak location from master instance */
- w16_bestIndex = msInfo->bestIndex;
- }
- else
- {
- /* Invalid mode */
- return MASTER_SLAVE_ERROR;
- }
-
-#else /* NETEQ_STEREO */
-
- /* Find peak */
- WebRtcNetEQ_PeakDetection(&pw16_corr[w16_tmp2], w16_stopPos, 1, fs_mult, &w16_bestIndex,
- &w16_bestVal);
- w16_bestIndex += w16_tmp; /* compensate for modified starting index */
-
-#endif /* NETEQ_STEREO */
-
- /*
- * Ensure that underrun does not occur for 10ms case => we have to get at least
- * 10ms + overlap . (This should never happen thanks to the above modification of
- * peak-finding starting point.)
- * */
- while ((w16_bestIndex + w16_decodedLen) < (inst->timestampsPerCall
- + inst->ExpandInst.w16_overlap) || w16_bestIndex + w16_decodedLen < w16_startPos)
- {
- w16_bestIndex += w16_newLen; /* Jump one lag ahead */
- }
- pw16_decodedOut = pw16_outData + w16_bestIndex;
-
- /* Mute the new decoded data if needed (and unmute it linearly) */
- w16_interpLen = WEBRTC_SPL_MIN(60*fs_mult,
- w16_expandedLen-w16_bestIndex); /* this is the overlapping part of pw16_expanded */
- w16_interpLen = WEBRTC_SPL_MIN(w16_interpLen, w16_decodedLen);
- w16_inc = WebRtcSpl_DivW32W16ResW16(4194,
- fs_mult); /* in Q20, 0.004 for NB and 0.002 for WB */
- if (inst->w16_muteFactor < 16384)
- {
- WebRtcNetEQ_UnmuteSignal(pw16_decoded, &inst->w16_muteFactor, pw16_decoded, w16_inc,
- (WebRtc_Word16) w16_interpLen);
- WebRtcNetEQ_UnmuteSignal(&pw16_decoded[w16_interpLen], &inst->w16_muteFactor,
- &pw16_decodedOut[w16_interpLen], w16_inc,
- (WebRtc_Word16) (w16_decodedLen - w16_interpLen));
- }
- else
- {
- /* No muting needed */
-
- WEBRTC_SPL_MEMMOVE_W16(&pw16_decodedOut[w16_interpLen], &pw16_decoded[w16_interpLen],
- (w16_decodedLen-w16_interpLen));
- }
-
- /* Do overlap and interpolate linearly */
- w16_inc = WebRtcSpl_DivW32W16ResW16(16384, (WebRtc_Word16) (w16_interpLen + 1)); /* Q14 */
- w16_startfact = (16384 - w16_inc);
- WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_expanded, w16_bestIndex);
- WebRtcNetEQ_MixVoiceUnvoice(pw16_decodedOut, &pw16_expanded[w16_bestIndex], pw16_decoded,
- &w16_startfact, w16_inc, w16_interpLen);
-
- inst->w16_mode = MODE_MERGE;
- inst->ExpandInst.w16_consecExp = 0; /* Last was not expand any more */
-
- /* New added length (w16_startPos samples were borrowed) */
- *pw16_len = w16_bestIndex + w16_decodedLen - w16_startPos;
-
- /* Update VQmon parameter */
- inst->w16_concealedTS += (*pw16_len - w16_decodedLen);
- inst->w16_concealedTS = WEBRTC_SPL_MAX(0, inst->w16_concealedTS);
-
- /* Update in-call and post-call statistics */
- if (inst->ExpandInst.w16_expandMuteFactor == 0)
- {
- /* expansion generates noise only */
- inst->statInst.expandedNoiseSamples += (*pw16_len - w16_decodedLen);
- }
- else
- {
- /* expansion generates more than only noise */
- inst->statInst.expandedVoiceSamples += (*pw16_len - w16_decodedLen);
- }
- inst->statInst.expandLength += (*pw16_len - w16_decodedLen);
-
-
- /* Copy back the first part of the data to the speechHistory */
-
- WEBRTC_SPL_MEMCPY_W16(&inst->speechBuffer[inst->curPosition], pw16_outData, w16_startPos);
-
-
- /* Move data to within outData */
-
- WEBRTC_SPL_MEMMOVE_W16(pw16_outData, &pw16_outData[w16_startPos], (*pw16_len));
-
- return 0;
-}
-
-#undef SCRATCH_pw16_expanded
-#undef SCRATCH_pw16_expandedLB
-#undef SCRATCH_pw16_decodedLB
-#undef SCRATCH_pw32_corr
-#undef SCRATCH_pw16_corrVec
-#undef SCRATCH_NETEQ_EXPAND
diff --git a/trunk/src/modules/audio_coding/neteq/min_distortion.c b/trunk/src/modules/audio_coding/neteq/min_distortion.c
deleted file mode 100644
index 4c9ee1c..0000000
--- a/trunk/src/modules/audio_coding/neteq/min_distortion.c
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Calculate best overlap fit according to distortion measure.
- */
-
-#include "dsp_helpfunctions.h"
-
-#include "signal_processing_library.h"
-
-WebRtc_Word16 WebRtcNetEQ_MinDistortion(const WebRtc_Word16 *pw16_data,
- WebRtc_Word16 w16_minLag, WebRtc_Word16 w16_maxLag,
- WebRtc_Word16 len, WebRtc_Word32 *pw16_dist)
-{
- int i, j;
- const WebRtc_Word16 *pw16_data1;
- const WebRtc_Word16 *pw16_data2;
- WebRtc_Word32 w32_diff;
- WebRtc_Word32 w32_sumdiff;
- WebRtc_Word16 bestIndex = -1;
- WebRtc_Word32 minDist = WEBRTC_SPL_WORD32_MAX;
-
- for (i = w16_minLag; i <= w16_maxLag; i++)
- {
- w32_sumdiff = 0;
- pw16_data1 = pw16_data;
- pw16_data2 = pw16_data - i;
-
- for (j = 0; j < len; j++)
- {
- w32_diff = pw16_data1[j] - pw16_data2[j];
- w32_sumdiff += WEBRTC_SPL_ABS_W32(w32_diff);
- }
-
- /* Compare with previous minimum */
- if (w32_sumdiff < minDist)
- {
- minDist = w32_sumdiff;
- bestIndex = i;
- }
- }
-
- *pw16_dist = minDist;
-
- return bestIndex;
-}
-
diff --git a/trunk/src/modules/audio_coding/neteq/mix_voice_unvoice.c b/trunk/src/modules/audio_coding/neteq/mix_voice_unvoice.c
deleted file mode 100644
index 9895630..0000000
--- a/trunk/src/modules/audio_coding/neteq/mix_voice_unvoice.c
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This function mixes a voiced signal with an unvoiced signal and
- * updates the weight on a sample by sample basis.
- */
-
-#include "dsp_helpfunctions.h"
-
-#include "signal_processing_library.h"
-
-void WebRtcNetEQ_MixVoiceUnvoice(WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_voicedVec,
- WebRtc_Word16 *pw16_unvoicedVec,
- WebRtc_Word16 *w16_current_vfraction,
- WebRtc_Word16 w16_vfraction_change, WebRtc_Word16 N)
-{
- int i;
- WebRtc_Word16 w16_tmp2;
- WebRtc_Word16 vfraction = *w16_current_vfraction;
-
- w16_tmp2 = 16384 - vfraction;
- for (i = 0; i < N; i++)
- {
- pw16_outData[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
- WEBRTC_SPL_MUL_16_16(vfraction, pw16_voicedVec[i]) +
- WEBRTC_SPL_MUL_16_16(w16_tmp2, pw16_unvoicedVec[i]) + 8192,
- 14);
- vfraction -= w16_vfraction_change;
- w16_tmp2 += w16_vfraction_change;
- }
- *w16_current_vfraction = vfraction;
-}
-
diff --git a/trunk/src/modules/audio_coding/neteq/mute_signal.c b/trunk/src/modules/audio_coding/neteq/mute_signal.c
deleted file mode 100644
index ee899cf..0000000
--- a/trunk/src/modules/audio_coding/neteq/mute_signal.c
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This function mutes a signal linearly on a sample by sample basis.
- */
-
-#include "dsp_helpfunctions.h"
-
-#include "signal_processing_library.h"
-
-void WebRtcNetEQ_MuteSignal(WebRtc_Word16 *pw16_inout, WebRtc_Word16 muteSlope,
- WebRtc_Word16 N)
-{
- int i;
- WebRtc_Word32 w32_tmp = 1048608; /* (16384<<6 + 32) */
-
- for (i = 0; i < N; i++)
- {
- pw16_inout[i]
- = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16((WebRtc_Word16)(w32_tmp>>6), pw16_inout[i])
- + 8192) >> 14);
- w32_tmp -= muteSlope;
- }
-}
-
diff --git a/trunk/src/modules/audio_coding/neteq/neteq.gypi b/trunk/src/modules/audio_coding/neteq/neteq.gypi
deleted file mode 100644
index c72efdd..0000000
--- a/trunk/src/modules/audio_coding/neteq/neteq.gypi
+++ /dev/null
@@ -1,289 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'NetEq',
- 'type': '<(library)',
- 'dependencies': [
- 'CNG',
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- ],
- 'defines': [
- 'NETEQ_VOICEENGINE_CODECS', # TODO: Should create a Chrome define which
- 'SCRATCH', # specifies a subset of codecs to support.
- ],
- 'include_dirs': [
- 'interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'interface',
- ],
- },
- 'sources': [
- 'interface/webrtc_neteq.h',
- 'interface/webrtc_neteq_help_macros.h',
- 'interface/webrtc_neteq_internal.h',
- 'accelerate.c',
- 'automode.c',
- 'automode.h',
- 'bgn_update.c',
- 'buffer_stats.h',
- 'bufstats_decision.c',
- 'cng_internal.c',
- 'codec_db.c',
- 'codec_db.h',
- 'codec_db_defines.h',
- 'correlator.c',
- 'delay_logging.h',
- 'dsp.c',
- 'dsp.h',
- 'dsp_helpfunctions.c',
- 'dsp_helpfunctions.h',
- 'dtmf_buffer.c',
- 'dtmf_buffer.h',
- 'dtmf_tonegen.c',
- 'dtmf_tonegen.h',
- 'expand.c',
- 'mcu.h',
- 'mcu_address_init.c',
- 'mcu_dsp_common.c',
- 'mcu_dsp_common.h',
- 'mcu_reset.c',
- 'merge.c',
- 'min_distortion.c',
- 'mix_voice_unvoice.c',
- 'mute_signal.c',
- 'neteq_defines.h',
- 'neteq_error_codes.h',
- 'neteq_statistics.h',
- 'normal.c',
- 'packet_buffer.c',
- 'packet_buffer.h',
- 'peak_detection.c',
- 'preemptive_expand.c',
- 'random_vector.c',
- 'recin.c',
- 'recout.c',
- 'rtcp.c',
- 'rtcp.h',
- 'rtp.c',
- 'rtp.h',
- 'set_fs.c',
- 'signal_mcu.c',
- 'split_and_insert.c',
- 'unmute_signal.c',
- 'webrtc_neteq.c',
- ],
- },
- ], # targets
- # Exclude the test target when building with chromium.
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'neteq_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'NetEq',
- 'NetEqTestTools',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- ],
- 'sources': [
- 'webrtc_neteq_unittest.cc',
- ],
- }, # neteq_unittests
- {
- 'target_name': 'NetEqRTPplay',
- 'type': 'executable',
- 'dependencies': [
- 'NetEq', # NetEQ library defined above
- 'NetEqTestTools',# Test helpers
- 'G711',
- 'G722',
- 'PCM16B',
- 'iLBC',
- 'iSAC',
- 'CNG',
- ],
- 'defines': [
- # TODO: Make codec selection conditional on definitions in target NetEq
- 'CODEC_ILBC',
- 'CODEC_PCM16B',
- 'CODEC_G711',
- 'CODEC_G722',
- 'CODEC_ISAC',
- 'CODEC_PCM16B_WB',
- 'CODEC_ISAC_SWB',
- 'CODEC_PCM16B_32KHZ',
- 'CODEC_CNGCODEC8',
- 'CODEC_CNGCODEC16',
- 'CODEC_CNGCODEC32',
- 'CODEC_ATEVENT_DECODE',
- 'CODEC_RED',
- ],
- 'include_dirs': [
- '.',
- 'test',
- ],
- 'sources': [
- 'test/NetEqRTPplay.cc',
- ],
- },
- {
- 'target_name': 'RTPencode',
- 'type': 'executable',
- 'dependencies': [
- 'NetEqTestTools',# Test helpers
- 'G711',
- 'G722',
- 'PCM16B',
- 'iLBC',
- 'iSAC',
- 'CNG',
- '<(webrtc_root)/common_audio/common_audio.gyp:vad',
- ],
- 'defines': [
- # TODO: Make codec selection conditional on definitions in target NetEq
- 'CODEC_ILBC',
- 'CODEC_PCM16B',
- 'CODEC_G711',
- 'CODEC_G722',
- 'CODEC_ISAC',
- 'CODEC_PCM16B_WB',
- 'CODEC_ISAC_SWB',
- 'CODEC_PCM16B_32KHZ',
- 'CODEC_CNGCODEC8',
- 'CODEC_CNGCODEC16',
- 'CODEC_CNGCODEC32',
- 'CODEC_ATEVENT_DECODE',
- 'CODEC_RED',
- ],
- 'include_dirs': [
- 'interface',
- 'test',
- ],
- 'sources': [
- 'test/RTPencode.cc',
- ],
- },
- {
- 'target_name': 'RTPjitter',
- 'type': 'executable',
- 'dependencies': [
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'sources': [
- 'test/RTPjitter.cc',
- ],
- },
- {
- 'target_name': 'RTPanalyze',
- 'type': 'executable',
- 'dependencies': [
- 'NetEqTestTools',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'sources': [
- 'test/RTPanalyze.cc',
- ],
- },
- {
- 'target_name': 'RTPchange',
- 'type': 'executable',
- 'dependencies': [
- 'NetEqTestTools',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'sources': [
- 'test/RTPchange.cc',
- ],
- },
- {
- 'target_name': 'RTPtimeshift',
- 'type': 'executable',
- 'dependencies': [
- 'NetEqTestTools',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'sources': [
- 'test/RTPtimeshift.cc',
- ],
- },
- {
- 'target_name': 'RTPcat',
- 'type': 'executable',
- 'dependencies': [
- 'NetEqTestTools',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'sources': [
- 'test/RTPcat.cc',
- ],
- },
- {
- 'target_name': 'NetEqTestTools',
- # Collection of useful functions used in other tests
- 'type': '<(library)',
- 'dependencies': [
- 'G711',
- 'G722',
- 'PCM16B',
- 'iLBC',
- 'iSAC',
- 'CNG',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'interface',
- 'test',
- ],
- },
- 'defines': [
- # TODO: Make codec selection conditional on definitions in target NetEq
- 'CODEC_ILBC',
- 'CODEC_PCM16B',
- 'CODEC_G711',
- 'CODEC_G722',
- 'CODEC_ISAC',
- 'CODEC_PCM16B_WB',
- 'CODEC_ISAC_SWB',
- 'CODEC_PCM16B_32KHZ',
- 'CODEC_CNGCODEC8',
- 'CODEC_CNGCODEC16',
- 'CODEC_CNGCODEC32',
- 'CODEC_ATEVENT_DECODE',
- 'CODEC_RED',
- ],
- 'include_dirs': [
- 'interface',
- 'test',
- ],
- 'sources': [
- 'test/NETEQTEST_NetEQClass.cc',
- 'test/NETEQTEST_RTPpacket.cc',
- 'test/NETEQTEST_CodecClass.cc',
- 'test/NETEQTEST_NetEQClass.h',
- 'test/NETEQTEST_RTPpacket.h',
- 'test/NETEQTEST_CodecClass.h',
- ],
- },
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_coding/neteq/neteq_defines.h b/trunk/src/modules/audio_coding/neteq/neteq_defines.h
deleted file mode 100644
index 318e6bb..0000000
--- a/trunk/src/modules/audio_coding/neteq/neteq_defines.h
+++ /dev/null
@@ -1,356 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*****************************************************************************************
- *
- * Compilation flags in NetEQ:
- *
- *****************************************************************************************
- *
- ***** Platform flags ******
- *
- * SCRATCH Run NetEQ with "Scratch memory" to save some stack memory.
- * Definition can be used on all platforms
- *
- ***** Summary flags ******
- *
- * NETEQ_ALL_SPECIAL_CODECS Add support for special codecs (CN/RED/DTMF)
- *
- * NETEQ_ALL_NB_CODECS Add support for all NB codecs (except CN/RED/DTMF)
- *
- * NETEQ_ALL_WB_CODECS Add support for all WB codecs (except CN/RED/DTMF)
- *
- * NETEQ_VOICEENGINE_CODECS Support for all NB, WB and SWB32 codecs and CN, RED and DTMF
- *
- * NETEQ_ALL_CODECS Support for all NB, WB, SWB 32kHz and SWB 48kHz as well as
- * CN, RED and DTMF
- *
- ***** Sampling frequency ******
- * (Note: usually not needed when Summary flags are used)
- *
- * NETEQ_WIDEBAND Wideband enabled
- *
- * NETEQ_32KHZ_WIDEBAND Super wideband @ 32kHz enabled
- *
- * NETEQ_48KHZ_WIDEBAND Super wideband @ 48kHz enabled
- *
- ***** Special Codec ******
- * (Note: not needed if NETEQ_ALL_CODECS is used)
- *
- * NETEQ_RED_CODEC With this flag you enable NetEQ to understand redundancy in
- * the RTP. NetEQ will use the redundancy if it's the same
- * codec
- *
- * NETEQ_CNG_CODEC Enable DTX with the CN payload
- *
- * NETEQ_ATEVENT_DECODE Enable AVT event and play out the corresponding DTMF tone
- *
- ***** Speech Codecs *****
- * (Note: Not needed if Summary flags are used)
- *
- * NETEQ_G711_CODEC Enable G.711 u- and A-law
- *
- * NETEQ_PCM16B_CODEC Enable uncompressed 16-bit
- *
- * NETEQ_ILBC_CODEC Enable iLBC
- *
- * NETEQ_ISAC_CODEC Enable iSAC
- *
- * NETEQ_ISAC_SWB_CODEC Enable iSAC-SWB
- *
- * NETEQ_G722_CODEC Enable G.722
- *
- * NETEQ_G729_CODEC Enable G.729
- *
- * NETEQ_G729_1_CODEC Enable G.729.1
- *
- * NETEQ_G726_CODEC Enable G.726
- *
- * NETEQ_G722_1_CODEC Enable G722.1
- *
- * NETEQ_G722_1C_CODEC Enable G722.1 Annex C
- *
- * NETEQ_SPEEX_CODEC Enable Speex (at 8 and 16 kHz sample rate)
- *
- * NETEQ_CELT_CODEC Enable Celt (at 32 kHz sample rate)
- *
- * NETEQ_GSMFR_CODEC Enable GSM-FR
- *
- * NETEQ_AMR_CODEC Enable AMR (narrowband)
- *
- * NETEQ_AMRWB_CODEC Enable AMR-WB
- *
- * NETEQ_CNG_CODEC Enable DTX with the CNG payload
- *
- * NETEQ_ATEVENT_DECODE Enable AVT event and play out the corresponding DTMF tone
- *
- ***** Test flags ******
- *
- * WEBRTC_NETEQ_40BITACC_TEST Run NetEQ with simulated 40-bit accumulator to run
- * bit-exact to a DSP implementation where the main (splib
- * and NetEQ) functions have been 40-bit optimized
- *
- *****************************************************************************************
- */
-
-#if !defined NETEQ_DEFINES_H
-#define NETEQ_DEFINES_H
-
-/* Data block structure for MCU to DSP communication:
- *
- *
- * First 3 16-bit words are pre-header that contains instructions and timestamp update
- * Fourth 16-bit word is length of data block 1
- * Rest is payload data
- *
- * 0 48 64 80
- * -------------...----------------------------------------------------------------------
- * | PreHeader ... | Length 1 | Payload data 1 ...... | Lenght 2| Data block 2.... | ...
- * -------------...----------------------------------------------------------------------
- *
- *
- * Preheader:
- * 4 MSB can be either of:
- */
-
-#define DSP_INSTR_NORMAL 0x1000
-/* Payload data will contain the encoded frames */
-
-#define DSP_INSTR_MERGE 0x2000
-/* Payload data block 1 will contain the encoded frame */
-/* Info block will contain the number of missing samples */
-
-#define DSP_INSTR_EXPAND 0x3000
-/* Payload data will be empty */
-
-#define DSP_INSTR_ACCELERATE 0x4000
-/* Payload data will contain the encoded frame */
-
-#define DSP_INSTR_DO_RFC3389CNG 0x5000
-/* Payload data will contain the SID frame if there is one*/
-
-#define DSP_INSTR_DTMF_GENERATE 0x6000
-/* Payload data will be one WebRtc_Word16 with the current DTMF value and one
- * WebRtc_Word16 with the current volume value
- */
-#define DSP_INSTR_NORMAL_ONE_DESC 0x7000
-/* No encoded frames */
-
-#define DSP_INSTR_DO_CODEC_INTERNAL_CNG 0x8000
-/* Codec has a built-in VAD/DTX scheme (use the above for "no transmission") */
-
-#define DSP_INSTR_PREEMPTIVE_EXPAND 0x9000
-/* Payload data will contain the encoded frames, if any */
-
-#define DSP_INSTR_DO_ALTERNATIVE_PLC 0xB000
-/* NetEQ switched off and packet missing... */
-
-#define DSP_INSTR_DO_ALTERNATIVE_PLC_INC_TS 0xC000
-/* NetEQ switched off and packet missing... */
-
-#define DSP_INSTR_DO_AUDIO_REPETITION 0xD000
-/* NetEQ switched off and packet missing... */
-
-#define DSP_INSTR_DO_AUDIO_REPETITION_INC_TS 0xE000
-/* NetEQ switched off and packet missing... */
-
-#define DSP_INSTR_FADE_TO_BGN 0xF000
-/* Exception handling: fade out to BGN (expand) */
-
-/*
- * Next 4 bits signal additional data that needs to be transmitted
- */
-
-#define DSP_CODEC_NO_CHANGE 0x0100
-#define DSP_CODEC_NEW_CODEC 0x0200
-#define DSP_CODEC_ADD_LATE_PKT 0x0300
-#define DSP_CODEC_RESET 0x0400
-#define DSP_DTMF_PAYLOAD 0x0010
-
-/*
- * The most significant bit of the payload-length
- * is used to flag whether the associated payload
- * is redundant payload. This currently useful only for
- * iSAC, where redundant payloads have to be treated
- * differently. Every time the length is read it must be
- * masked by DSP_CODEC_MASK_RED_FLAG to ignore the flag.
- * Use DSP_CODEC_RED_FLAG to set or retrieve the flag.
- */
-#define DSP_CODEC_MASK_RED_FLAG 0x7FFF
-#define DSP_CODEC_RED_FLAG 0x8000
-
-/*
- * The first block of payload data consist of decode function pointers,
- * and then the speech blocks.
- *
- */
-
-
-/*
- * The playout modes that NetEq produced (i.e. gives more info about if the
- * Accelerate was successful or not)
- */
-
-#define MODE_NORMAL 0x0000
-#define MODE_EXPAND 0x0001
-#define MODE_MERGE 0x0002
-#define MODE_SUCCESS_ACCELERATE 0x0003
-#define MODE_UNSUCCESS_ACCELERATE 0x0004
-#define MODE_RFC3389CNG 0x0005
-#define MODE_LOWEN_ACCELERATE 0x0006
-#define MODE_DTMF 0x0007
-#define MODE_ONE_DESCRIPTOR 0x0008
-#define MODE_CODEC_INTERNAL_CNG 0x0009
-#define MODE_SUCCESS_PREEMPTIVE 0x000A
-#define MODE_UNSUCCESS_PREEMPTIVE 0x000B
-#define MODE_LOWEN_PREEMPTIVE 0x000C
-#define MODE_FADE_TO_BGN 0x000D
-
-#define MODE_ERROR 0x0010
-
-#define MODE_AWAITING_CODEC_PTR 0x0100
-
-#define MODE_BGN_ONLY 0x0200
-
-#define MODE_MASTER_DTMF_SIGNAL 0x0400
-
-#define MODE_USING_STEREO 0x0800
-
-
-
-/***********************/
-/* Group codec defines */
-/***********************/
-
-#if (defined(NETEQ_ALL_SPECIAL_CODECS))
- #define NETEQ_CNG_CODEC
- #define NETEQ_ATEVENT_DECODE
- #define NETEQ_RED_CODEC
- #define NETEQ_VAD
- #define NETEQ_ARBITRARY_CODEC
-#endif
-
-#if (defined(NETEQ_ALL_NB_CODECS)) /* Except RED, DTMF and CNG */
- #define NETEQ_PCM16B_CODEC
- #define NETEQ_G711_CODEC
- #define NETEQ_ILBC_CODEC
- #define NETEQ_G729_CODEC
- #define NETEQ_G726_CODEC
- #define NETEQ_GSMFR_CODEC
- #define NETEQ_AMR_CODEC
-#endif
-
-#if (defined(NETEQ_ALL_WB_CODECS)) /* Except RED, DTMF and CNG */
- #define NETEQ_ISAC_CODEC
- #define NETEQ_G722_CODEC
- #define NETEQ_G722_1_CODEC
- #define NETEQ_G729_1_CODEC
- #define NETEQ_SPEEX_CODEC
- #define NETEQ_AMRWB_CODEC
- #define NETEQ_WIDEBAND
-#endif
-
-#if (defined(NETEQ_ALL_WB32_CODECS)) /* AAC, RED, DTMF and CNG */
- #define NETEQ_ISAC_SWB_CODEC
- #define NETEQ_32KHZ_WIDEBAND
- #define NETEQ_G722_1C_CODEC
- #define NETEQ_CELT_CODEC
-#endif
-
-#if (defined(NETEQ_VOICEENGINE_CODECS))
- /* Special codecs */
- #define NETEQ_CNG_CODEC
- #define NETEQ_ATEVENT_DECODE
- #define NETEQ_RED_CODEC
- #define NETEQ_VAD
- #define NETEQ_ARBITRARY_CODEC
-
- /* Narrowband codecs */
- #define NETEQ_PCM16B_CODEC
- #define NETEQ_G711_CODEC
- #define NETEQ_ILBC_CODEC
- #define NETEQ_AMR_CODEC
- #define NETEQ_G729_CODEC
- #define NETEQ_GSMFR_CODEC
-
- /* Wideband codecs */
- #define NETEQ_WIDEBAND
- #define NETEQ_ISAC_CODEC
- #define NETEQ_G722_CODEC
- #define NETEQ_G722_1_CODEC
- #define NETEQ_G729_1_CODEC
- #define NETEQ_AMRWB_CODEC
- #define NETEQ_SPEEX_CODEC
-
- /* Super wideband 32kHz codecs */
- #define NETEQ_ISAC_SWB_CODEC
- #define NETEQ_32KHZ_WIDEBAND
- #define NETEQ_G722_1C_CODEC
- #define NETEQ_CELT_CODEC
-
-#endif
-
-#if (defined(NETEQ_ALL_CODECS))
- /* Special codecs */
- #define NETEQ_CNG_CODEC
- #define NETEQ_ATEVENT_DECODE
- #define NETEQ_RED_CODEC
- #define NETEQ_VAD
- #define NETEQ_ARBITRARY_CODEC
-
- /* Narrowband codecs */
- #define NETEQ_PCM16B_CODEC
- #define NETEQ_G711_CODEC
- #define NETEQ_ILBC_CODEC
- #define NETEQ_G729_CODEC
- #define NETEQ_G726_CODEC
- #define NETEQ_GSMFR_CODEC
- #define NETEQ_AMR_CODEC
-
- /* Wideband codecs */
- #define NETEQ_WIDEBAND
- #define NETEQ_ISAC_CODEC
- #define NETEQ_G722_CODEC
- #define NETEQ_G722_1_CODEC
- #define NETEQ_G729_1_CODEC
- #define NETEQ_SPEEX_CODEC
- #define NETEQ_AMRWB_CODEC
-
- /* Super wideband 32kHz codecs */
- #define NETEQ_ISAC_SWB_CODEC
- #define NETEQ_32KHZ_WIDEBAND
- #define NETEQ_G722_1C_CODEC
- #define NETEQ_CELT_CODEC
-
- /* Super wideband 48kHz codecs */
- #define NETEQ_48KHZ_WIDEBAND
-#endif
-
-/* Max output size from decoding one frame */
-#if defined(NETEQ_48KHZ_WIDEBAND)
- #define NETEQ_MAX_FRAME_SIZE 2880 /* 60 ms super wideband */
- #define NETEQ_MAX_OUTPUT_SIZE 3600 /* 60+15 ms super wideband (60 ms decoded + 15 ms for merge overlap) */
-#elif defined(NETEQ_32KHZ_WIDEBAND)
- #define NETEQ_MAX_FRAME_SIZE 1920 /* 60 ms super wideband */
- #define NETEQ_MAX_OUTPUT_SIZE 2400 /* 60+15 ms super wideband (60 ms decoded + 15 ms for merge overlap) */
-#elif defined(NETEQ_WIDEBAND)
- #define NETEQ_MAX_FRAME_SIZE 960 /* 60 ms wideband */
- #define NETEQ_MAX_OUTPUT_SIZE 1200 /* 60+15 ms wideband (60 ms decoded + 10 ms for merge overlap) */
-#else
- #define NETEQ_MAX_FRAME_SIZE 480 /* 60 ms narrowband */
- #define NETEQ_MAX_OUTPUT_SIZE 600 /* 60+15 ms narrowband (60 ms decoded + 10 ms for merge overlap) */
-#endif
-
-
-/* Enable stereo */
-#define NETEQ_STEREO
-
-#endif /* #if !defined NETEQ_DEFINES_H */
-
diff --git a/trunk/src/modules/audio_coding/neteq/neteq_error_codes.h b/trunk/src/modules/audio_coding/neteq/neteq_error_codes.h
deleted file mode 100644
index 1ce4680..0000000
--- a/trunk/src/modules/audio_coding/neteq/neteq_error_codes.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Definition of error codes.
- *
- * NOTE: When modifying the error codes,
- * also modify the function WebRtcNetEQ_GetErrorCode!
- */
-
-#if !defined NETEQ_ERROR_CODES_H
-#define NETEQ_ERROR_CODES_H
-
-/* Misc Error */
-#define NETEQ_OTHER_ERROR -1000
-
-/* Misc Recout Errors */
-#define FAULTY_INSTRUCTION -1001
-#define FAULTY_NETWORK_TYPE -1002
-#define FAULTY_DELAYVALUE -1003
-#define FAULTY_PLAYOUTMODE -1004
-#define CORRUPT_INSTANCE -1005
-#define ILLEGAL_MASTER_SLAVE_SWITCH -1006
-#define MASTER_SLAVE_ERROR -1007
-
-/* Misc Recout problems */
-#define UNKNOWN_BUFSTAT_DECISION -2001
-#define RECOUT_ERROR_DECODING -2002
-#define RECOUT_ERROR_SAMPLEUNDERRUN -2003
-#define RECOUT_ERROR_DECODED_TOO_MUCH -2004
-
-/* Misc RecIn problems */
-#define RECIN_CNG_ERROR -3001
-#define RECIN_UNKNOWNPAYLOAD -3002
-#define RECIN_BUFFERINSERT_ERROR -3003
-
-/* PBUFFER/BUFSTAT ERRORS */
-#define PBUFFER_INIT_ERROR -4001
-#define PBUFFER_INSERT_ERROR1 -4002
-#define PBUFFER_INSERT_ERROR2 -4003
-#define PBUFFER_INSERT_ERROR3 -4004
-#define PBUFFER_INSERT_ERROR4 -4005
-#define PBUFFER_INSERT_ERROR5 -4006
-#define UNKNOWN_G723_HEADER -4007
-#define PBUFFER_NONEXISTING_PACKET -4008
-#define PBUFFER_NOT_INITIALIZED -4009
-#define AMBIGUOUS_ILBC_FRAME_SIZE -4010
-
-/* CODEC DATABASE ERRORS */
-#define CODEC_DB_FULL -5001
-#define CODEC_DB_NOT_EXIST1 -5002
-#define CODEC_DB_NOT_EXIST2 -5003
-#define CODEC_DB_NOT_EXIST3 -5004
-#define CODEC_DB_NOT_EXIST4 -5005
-#define CODEC_DB_UNKNOWN_CODEC -5006
-#define CODEC_DB_PAYLOAD_TAKEN -5007
-#define CODEC_DB_UNSUPPORTED_CODEC -5008
-#define CODEC_DB_UNSUPPORTED_FS -5009
-
-/* DTMF ERRORS */
-#define DTMF_DEC_PARAMETER_ERROR -6001
-#define DTMF_INSERT_ERROR -6002
-#define DTMF_GEN_UNKNOWN_SAMP_FREQ -6003
-#define DTMF_NOT_SUPPORTED -6004
-
-/* RTP/PACKET ERRORS */
-#define RED_SPLIT_ERROR1 -7001
-#define RED_SPLIT_ERROR2 -7002
-#define RTP_TOO_SHORT_PACKET -7003
-#define RTP_CORRUPT_PACKET -7004
-
-#endif
diff --git a/trunk/src/modules/audio_coding/neteq/neteq_statistics.h b/trunk/src/modules/audio_coding/neteq/neteq_statistics.h
deleted file mode 100644
index d07f330..0000000
--- a/trunk/src/modules/audio_coding/neteq/neteq_statistics.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Definitions of statistics data structures for MCU and DSP sides.
- */
-
-#include "typedefs.h"
-
-#ifndef NETEQ_STATISTICS_H
-#define NETEQ_STATISTICS_H
-
-/*
- * Statistics struct on DSP side
- */
-typedef struct
-{
-
- /* variables for in-call statistics; queried through WebRtcNetEQ_GetNetworkStatistics */
- WebRtc_UWord32 expandLength; /* number of samples produced through expand */
- WebRtc_UWord32 preemptiveLength; /* number of samples produced through pre-emptive
- expand */
- WebRtc_UWord32 accelerateLength; /* number of samples removed through accelerate */
-
- /* variables for post-call statistics; queried through WebRtcNetEQ_GetJitterStatistics */
- WebRtc_UWord32 expandedVoiceSamples; /* number of voice samples produced through expand */
- WebRtc_UWord32 expandedNoiseSamples; /* number of noise (background) samples produced
- through expand */
-
-} DSPStats_t;
-
-#endif
-
diff --git a/trunk/src/modules/audio_coding/neteq/normal.c b/trunk/src/modules/audio_coding/neteq/normal.c
deleted file mode 100644
index b33940a..0000000
--- a/trunk/src/modules/audio_coding/neteq/normal.c
+++ /dev/null
@@ -1,279 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the function for handling "normal" speech operation.
- */
-#include "dsp.h"
-
-#include "signal_processing_library.h"
-
-#include "dsp_helpfunctions.h"
-
-/* Scratch usage:
-
- Type Name size startpos endpos
- WebRtc_Word16 pw16_expanded 125*fs/8000 0 125*fs/8000-1
-
- func WebRtcNetEQ_Expand 40+370*fs/8000 125*fs/8000 39+495*fs/8000
-
- Total: 40+495*fs/8000
- */
-
-#define SCRATCH_PW16_EXPANDED 0
-#if (defined(NETEQ_48KHZ_WIDEBAND))
-#define SCRATCH_NETEQ_EXPAND 756
-#elif (defined(NETEQ_32KHZ_WIDEBAND))
-#define SCRATCH_NETEQ_EXPAND 504
-#elif (defined(NETEQ_WIDEBAND))
-#define SCRATCH_NETEQ_EXPAND 252
-#else /* NB */
-#define SCRATCH_NETEQ_EXPAND 126
-#endif
-
-/****************************************************************************
- * WebRtcNetEQ_Normal(...)
- *
- * This function has the possibility to modify data that is played out in Normal
- * mode, for example adjust the gain of the signal. The length of the signal
- * can not be changed.
- *
- * Input:
- * - inst : NetEq instance, i.e. the user that requests more
- * speech/audio data
- * - scratchPtr : Pointer to scratch vector
- * - decoded : Pointer to vector of new data from decoder
- * (Vector contents may be altered by the function)
- * - len : Number of input samples
- *
- * Output:
- * - inst : Updated user information
- * - outData : Pointer to a memory space where the output data
- * should be stored
- * - pw16_len : Pointer to variable where the number of samples
- * produced will be written
- *
- * Return value : >=0 - Number of samples written to outData
- * -1 - Error
- */
-
-int WebRtcNetEQ_Normal(DSPInst_t *inst,
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_scratchPtr,
-#endif
- WebRtc_Word16 *pw16_decoded, WebRtc_Word16 len,
- WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len)
-{
-
- int i;
- WebRtc_Word16 fs_mult;
- WebRtc_Word16 fs_shift;
- WebRtc_Word32 w32_En_speech;
- WebRtc_Word16 enLen;
- WebRtc_Word16 w16_muted;
- WebRtc_Word16 w16_inc, w16_frac;
- WebRtc_Word16 w16_tmp;
- WebRtc_Word32 w32_tmp;
-
- /* Sanity check */
- if (len < 0)
- {
- /* Cannot have negative length of input vector */
- return (-1);
- }
-
- if (len == 0)
- {
- /* Still got some data to play => continue with the same mode */
- *pw16_len = len;
- return (len);
- }
-
- fs_mult = WebRtcSpl_DivW32W16ResW16(inst->fs, 8000);
- fs_shift = 30 - WebRtcSpl_NormW32(fs_mult); /* Note that this is not "exact" for 48kHz */
-
- /*
- * Check if last RecOut call resulted in an Expand or a FadeToBGN. If so, we have to take
- * care of some cross-fading and unmuting.
- */
- if (inst->w16_mode == MODE_EXPAND || inst->w16_mode == MODE_FADE_TO_BGN)
- {
-
- /* Define memory where temporary result from Expand algorithm can be stored. */
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_expanded = pw16_scratchPtr + SCRATCH_PW16_EXPANDED;
-#else
- WebRtc_Word16 pw16_expanded[FSMULT * 125];
-#endif
- WebRtc_Word16 expandedLen = 0;
- WebRtc_Word16 w16_decodedMax;
-
- /* Find largest value in new data */
- w16_decodedMax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (WebRtc_Word16) len);
-
- /* Generate interpolation data using Expand */
- /* First, set Expand parameters to appropriate values. */
- inst->ExpandInst.w16_lagsPosition = 0;
- inst->ExpandInst.w16_lagsDirection = 0;
- inst->ExpandInst.w16_stopMuting = 1; /* Do not mute signal any more */
-
- /* Call Expand */
- WebRtcNetEQ_Expand(inst,
-#ifdef SCRATCH
- pw16_scratchPtr + SCRATCH_NETEQ_EXPAND,
-#endif
- pw16_expanded, &expandedLen, (WebRtc_Word16) (inst->w16_mode == MODE_FADE_TO_BGN));
-
- inst->ExpandInst.w16_stopMuting = 0; /* Restore value */
- inst->ExpandInst.w16_consecExp = 0; /* Last was not Expand any more */
-
- /* Adjust muting factor (main muting factor times expand muting factor) */
- if (inst->w16_mode == MODE_FADE_TO_BGN)
- {
- /* If last mode was FadeToBGN, the mute factor should be zero. */
- inst->w16_muteFactor = 0;
- }
- else
- {
- /* w16_muteFactor * w16_expandMuteFactor */
- inst->w16_muteFactor
- = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16_RSFT(inst->w16_muteFactor,
- inst->ExpandInst.w16_expandMuteFactor, 14);
- }
-
- /* Adjust muting factor if needed (to BGN level) */
- enLen = WEBRTC_SPL_MIN(fs_mult<<6, len); /* min( fs_mult * 64, len ) */
- w16_tmp = 6 + fs_shift - WebRtcSpl_NormW32(
- WEBRTC_SPL_MUL_16_16(w16_decodedMax, w16_decodedMax));
- w16_tmp = WEBRTC_SPL_MAX(w16_tmp, 0);
- w32_En_speech = WebRtcNetEQ_DotW16W16(pw16_decoded, pw16_decoded, enLen, w16_tmp);
- w32_En_speech = WebRtcSpl_DivW32W16(w32_En_speech, (WebRtc_Word16) (enLen >> w16_tmp));
-
- if ((w32_En_speech != 0) && (w32_En_speech > inst->BGNInst.w32_energy))
- {
- /* Normalize new frame energy to 15 bits */
- w16_tmp = WebRtcSpl_NormW32(w32_En_speech) - 16;
- /* we want inst->BGNInst.energy/En_speech in Q14 */
- w32_tmp = WEBRTC_SPL_SHIFT_W32(inst->BGNInst.w32_energy, (w16_tmp+14));
- w16_tmp = (WebRtc_Word16) WEBRTC_SPL_SHIFT_W32(w32_En_speech, w16_tmp);
- w16_tmp = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_tmp, w16_tmp);
- w16_muted = (WebRtc_Word16) WebRtcSpl_SqrtFloor(
- WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32) w16_tmp,
- 14)); /* w16_muted in Q14 (sqrt(Q28)) */
- }
- else
- {
- w16_muted = 16384; /* 1.0 in Q14 */
- }
- if (w16_muted > inst->w16_muteFactor)
- {
- inst->w16_muteFactor = WEBRTC_SPL_MIN(w16_muted, 16384);
- }
-
- /* If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14) */
- w16_inc = WebRtcSpl_DivW32W16ResW16(64, fs_mult);
- for (i = 0; i < len; i++)
- {
- /* scale with mute factor */
- w32_tmp = WEBRTC_SPL_MUL_16_16(pw16_decoded[i], inst->w16_muteFactor);
- /* shift 14 with proper rounding */
- pw16_decoded[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32((w32_tmp + 8192), 14);
- /* increase mute_factor towards 16384 */
- inst->w16_muteFactor = WEBRTC_SPL_MIN(16384, (inst->w16_muteFactor+w16_inc));
- }
-
- /*
- * Interpolate the expanded data into the new vector
- * (NB/WB/SWB32/SWB40 8/16/32/32 samples)
- */
- fs_shift = WEBRTC_SPL_MIN(3, fs_shift); /* Set to 3 for >32kHz */
- w16_inc = 4 >> fs_shift;
- w16_frac = w16_inc;
- for (i = 0; i < 8 * fs_mult; i++)
- {
- pw16_decoded[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
- (WEBRTC_SPL_MUL_16_16(w16_frac, pw16_decoded[i]) +
- WEBRTC_SPL_MUL_16_16((32 - w16_frac), pw16_expanded[i]) + 8),
- 5);
- w16_frac += w16_inc;
- }
-
-#ifdef NETEQ_CNG_CODEC
- }
- else if (inst->w16_mode==MODE_RFC3389CNG)
- { /* previous was RFC 3389 CNG...*/
- WebRtc_Word16 pw16_CngInterp[32];
- /* Reset mute factor and start up fresh */
- inst->w16_muteFactor = 16384;
- if (inst->CNG_Codec_inst != NULL)
- {
- /* Generate long enough for 32kHz */
- if(WebRtcCng_Generate(inst->CNG_Codec_inst,pw16_CngInterp, 32, 0)<0)
- {
- /* error returned; set return vector to all zeros */
- WebRtcSpl_MemSetW16(pw16_CngInterp, 0, 32);
- }
- }
- else
- {
- /*
- * If no CNG instance is defined, just copy from the decoded data.
- * (This will result in interpolating the decoded with itself.)
- */
- WEBRTC_SPL_MEMCPY_W16(pw16_CngInterp, pw16_decoded, fs_mult * 8);
- }
- /*
- * Interpolate the CNG into the new vector
- * (NB/WB/SWB32kHz/SWB48kHz 8/16/32/32 samples)
- */
- fs_shift = WEBRTC_SPL_MIN(3, fs_shift); /* Set to 3 for >32kHz */
- w16_inc = 4>>fs_shift;
- w16_frac = w16_inc;
- for (i = 0; i < 8 * fs_mult; i++)
- {
- pw16_decoded[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
- (WEBRTC_SPL_MUL_16_16(w16_frac, pw16_decoded[i]) +
- WEBRTC_SPL_MUL_16_16((32-w16_frac), pw16_CngInterp[i]) + 8),
- 5);
- w16_frac += w16_inc;
- }
-#endif
-
- }
- else if (inst->w16_muteFactor < 16384)
- {
- /*
- * Previous was neither of Expand, FadeToBGN or RFC3389_CNG, but we are still
- * ramping up from previous muting.
- * If muted increase by 0.64 for every 20 ms (NB/WB 0.0040/0.0020 in Q14)
- */
- w16_inc = WebRtcSpl_DivW32W16ResW16(64, fs_mult);
- for (i = 0; i < len; i++)
- {
- /* scale with mute factor */
- w32_tmp = WEBRTC_SPL_MUL_16_16(pw16_decoded[i], inst->w16_muteFactor);
- /* shift 14 with proper rounding */
- pw16_decoded[i] = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32((w32_tmp + 8192), 14);
- /* increase mute_factor towards 16384 */
- inst->w16_muteFactor = WEBRTC_SPL_MIN(16384, (inst->w16_muteFactor+w16_inc));
- }
- }
-
- /* Copy data to other buffer */WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, len);
-
- inst->w16_mode = MODE_NORMAL;
- *pw16_len = len;
- return (len);
-
-}
-
-#undef SCRATCH_PW16_EXPANDED
-#undef SCRATCH_NETEQ_EXPAND
-
diff --git a/trunk/src/modules/audio_coding/neteq/packet_buffer.c b/trunk/src/modules/audio_coding/neteq/packet_buffer.c
deleted file mode 100644
index 8b9073c..0000000
--- a/trunk/src/modules/audio_coding/neteq/packet_buffer.c
+++ /dev/null
@@ -1,735 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Implementation of the actual packet buffer data structure.
- */
-
-#include "packet_buffer.h"
-
-#include <string.h> /* to define NULL */
-
-#include "signal_processing_library.h"
-
-#include "neteq_error_codes.h"
-
-#ifdef NETEQ_DELAY_LOGGING
-/* special code for offline delay logging */
-#include "delay_logging.h"
-#include <stdio.h>
-
-extern FILE *delay_fid2; /* file pointer to delay log file */
-extern WebRtc_UWord32 tot_received_packets;
-#endif /* NETEQ_DELAY_LOGGING */
-
-
-int WebRtcNetEQ_PacketBufferInit(PacketBuf_t *bufferInst, int maxNoOfPackets,
- WebRtc_Word16 *pw16_memory, int memorySize)
-{
- int i;
- int pos = 0;
-
- /* Sanity check */
- if ((memorySize < PBUFFER_MIN_MEMORY_SIZE) || (pw16_memory == NULL)
- || (maxNoOfPackets < 2) || (maxNoOfPackets > 600))
- {
- /* Invalid parameters */
- return (PBUFFER_INIT_ERROR);
- }
-
- /* Clear the buffer instance */
- WebRtcSpl_MemSetW16((WebRtc_Word16*) bufferInst, 0,
- sizeof(PacketBuf_t) / sizeof(WebRtc_Word16));
-
- /* Clear the buffer memory */
- WebRtcSpl_MemSetW16((WebRtc_Word16*) pw16_memory, 0, memorySize);
-
- /* Set maximum number of packets */
- bufferInst->maxInsertPositions = maxNoOfPackets;
-
- /* Initialize array pointers */
- /* After each pointer has been set, the index pos is advanced to point immediately
- * after the the recently allocated vector. Note that one step for the pos index
- * corresponds to a WebRtc_Word16.
- */
-
- bufferInst->timeStamp = (WebRtc_UWord32*) &pw16_memory[pos];
- pos += maxNoOfPackets << 1; /* advance maxNoOfPackets * WebRtc_UWord32 */
-
- bufferInst->payloadLocation = (WebRtc_Word16**) &pw16_memory[pos];
- pos += maxNoOfPackets * (sizeof(WebRtc_Word16*) / sizeof(WebRtc_Word16)); /* advance */
-
- bufferInst->seqNumber = (WebRtc_UWord16*) &pw16_memory[pos];
- pos += maxNoOfPackets; /* advance maxNoOfPackets * WebRtc_UWord16 */
-
- bufferInst->payloadType = &pw16_memory[pos];
- pos += maxNoOfPackets; /* advance maxNoOfPackets * WebRtc_Word16 */
-
- bufferInst->payloadLengthBytes = &pw16_memory[pos];
- pos += maxNoOfPackets; /* advance maxNoOfPackets * WebRtc_Word16 */
-
- bufferInst->rcuPlCntr = &pw16_memory[pos];
- pos += maxNoOfPackets; /* advance maxNoOfPackets * WebRtc_Word16 */
-
- bufferInst->waitingTime = (int*) (&pw16_memory[pos]);
- /* Advance maxNoOfPackets * sizeof(waitingTime element). */
- pos += maxNoOfPackets *
- sizeof(*bufferInst->waitingTime) / sizeof(*pw16_memory);
-
- /* The payload memory starts after the slot arrays */
- bufferInst->startPayloadMemory = &pw16_memory[pos];
- bufferInst->currentMemoryPos = bufferInst->startPayloadMemory;
- bufferInst->memorySizeW16 = (memorySize - pos); /* Remaining memory */
-
- /* Initialize each payload slot as empty with infinite delay */
- for (i = 0; i < bufferInst->maxInsertPositions; i++)
- {
- bufferInst->payloadType[i] = -1;
- }
-
- /* Reset buffer parameters */
- bufferInst->numPacketsInBuffer = 0;
- bufferInst->packSizeSamples = 0;
- bufferInst->insertPosition = 0;
-
- /* Reset buffer statistics */
- bufferInst->discardedPackets = 0;
-
- return (0);
-}
-
-
-int WebRtcNetEQ_PacketBufferFlush(PacketBuf_t *bufferInst)
-{
- int i;
-
- /* Sanity check */
- if (bufferInst->startPayloadMemory == NULL)
- {
- /* Packet buffer has not been initialized */
- /* Don't do the flushing operation, since we do not
- know the state of the struct variables */
- return (0);
- }
-
- /* Set all payload lengths to zero */
- WebRtcSpl_MemSetW16(bufferInst->payloadLengthBytes, 0, bufferInst->maxInsertPositions);
-
- /* Reset buffer variables */
- bufferInst->numPacketsInBuffer = 0;
- bufferInst->currentMemoryPos = bufferInst->startPayloadMemory;
- bufferInst->insertPosition = 0;
-
- /* Clear all slots, starting with the last one */
- for (i = (bufferInst->maxInsertPositions - 1); i >= 0; i--)
- {
- bufferInst->payloadType[i] = -1;
- bufferInst->timeStamp[i] = 0;
- bufferInst->seqNumber[i] = 0;
- }
-
- return (0);
-}
-
-
-int WebRtcNetEQ_PacketBufferInsert(PacketBuf_t *bufferInst, const RTPPacket_t *RTPpacket,
- WebRtc_Word16 *flushed)
-{
- int nextPos;
- int i;
-
-#ifdef NETEQ_DELAY_LOGGING
- /* special code for offline delay logging */
- int temp_var;
-#endif /* NETEQ_DELAY_LOGGING */
-
- /* Initialize to "no flush" */
- *flushed = 0;
-
- /* Sanity check */
- if (bufferInst->startPayloadMemory == NULL)
- {
- /* packet buffer has not been initialized */
- return (-1);
- }
-
- /* Sanity check for payload length
- (payloadLen in bytes and memory size in WebRtc_Word16) */
- if ((RTPpacket->payloadLen > (bufferInst->memorySizeW16 << 1)) || (RTPpacket->payloadLen
- <= 0))
- {
- /* faulty or too long payload length */
- return (-1);
- }
-
- /* Find a position in the buffer for this packet */
- if (bufferInst->numPacketsInBuffer != 0)
- {
- /* Get the next slot */
- bufferInst->insertPosition++;
- if (bufferInst->insertPosition >= bufferInst->maxInsertPositions)
- {
- /* "Wrap around" and start from the beginning */
- bufferInst->insertPosition = 0;
- }
-
- /* Check if there is enough space for the new packet */
- if (bufferInst->currentMemoryPos + ((RTPpacket->payloadLen + 1) >> 1)
- >= &bufferInst->startPayloadMemory[bufferInst->memorySizeW16])
- {
- WebRtc_Word16 *tempMemAddress;
-
- /*
- * Payload does not fit at the end of the memory, put it in the beginning
- * instead
- */
- bufferInst->currentMemoryPos = bufferInst->startPayloadMemory;
-
- /*
- * Now, we must search for the next non-empty payload,
- * finding the one with the lowest start address for the payload
- */
- tempMemAddress = &bufferInst->startPayloadMemory[bufferInst->memorySizeW16];
- nextPos = -1;
-
- /* Loop through all slots again */
- for (i = 0; i < bufferInst->maxInsertPositions; i++)
- {
- /* Look for the non-empty slot with the lowest
- payload location address */
- if (bufferInst->payloadLengthBytes[i] != 0 && bufferInst->payloadLocation[i]
- < tempMemAddress)
- {
- tempMemAddress = bufferInst->payloadLocation[i];
- nextPos = i;
- }
- }
-
- /* Check that we did find a previous payload */
- if (nextPos == -1)
- {
- /* The buffer is corrupt => flush and return error */
- WebRtcNetEQ_PacketBufferFlush(bufferInst);
- *flushed = 1;
- return (-1);
- }
- }
- else
- {
- /* Payload fits at the end of memory. */
-
- /* Find the next non-empty slot. */
- nextPos = bufferInst->insertPosition + 1;
-
- /* Increase nextPos until a non-empty slot is found or end of array is encountered*/
- while ((bufferInst->payloadLengthBytes[nextPos] == 0) && (nextPos
- < bufferInst->maxInsertPositions))
- {
- nextPos++;
- }
-
- if (nextPos == bufferInst->maxInsertPositions)
- {
- /*
- * Reached the end of the array, so there must be a packet in the first
- * position instead
- */
- nextPos = 0;
-
- /* Increase nextPos until a non-empty slot is found */
- while (bufferInst->payloadLengthBytes[nextPos] == 0)
- {
- nextPos++;
- }
- }
- } /* end if-else */
-
- /*
- * Check if the new payload will extend into a payload later in memory.
- * If so, the buffer is full.
- */
- if ((bufferInst->currentMemoryPos <= bufferInst->payloadLocation[nextPos])
- && ((&bufferInst->currentMemoryPos[(RTPpacket->payloadLen + 1) >> 1])
- > bufferInst->payloadLocation[nextPos]))
- {
- /* Buffer is full, so the buffer must be flushed */
- WebRtcNetEQ_PacketBufferFlush(bufferInst);
- *flushed = 1;
- }
-
- if (bufferInst->payloadLengthBytes[bufferInst->insertPosition] != 0)
- {
- /* All positions are already taken and entire buffer should be flushed */
- WebRtcNetEQ_PacketBufferFlush(bufferInst);
- *flushed = 1;
- }
-
- }
- else
- {
- /* Buffer is empty, just insert the packet at the beginning */
- bufferInst->currentMemoryPos = bufferInst->startPayloadMemory;
- bufferInst->insertPosition = 0;
- }
-
- /* Insert packet in the found position */
- if (RTPpacket->starts_byte1 == 0)
- {
- /* Payload is 16-bit aligned => just copy it */
-
- WEBRTC_SPL_MEMCPY_W16(bufferInst->currentMemoryPos,
- RTPpacket->payload, (RTPpacket->payloadLen + 1) >> 1);
- }
- else
- {
- /* Payload is not 16-bit aligned => align it during copy operation */
- for (i = 0; i < RTPpacket->payloadLen; i++)
- {
- /* copy the (i+1)-th byte to the i-th byte */
-
- WEBRTC_SPL_SET_BYTE(bufferInst->currentMemoryPos,
- (WEBRTC_SPL_GET_BYTE(RTPpacket->payload, (i + 1))), i);
- }
- }
-
- /* Copy the packet information */
- bufferInst->payloadLocation[bufferInst->insertPosition] = bufferInst->currentMemoryPos;
- bufferInst->payloadLengthBytes[bufferInst->insertPosition] = RTPpacket->payloadLen;
- bufferInst->payloadType[bufferInst->insertPosition] = RTPpacket->payloadType;
- bufferInst->seqNumber[bufferInst->insertPosition] = RTPpacket->seqNumber;
- bufferInst->timeStamp[bufferInst->insertPosition] = RTPpacket->timeStamp;
- bufferInst->rcuPlCntr[bufferInst->insertPosition] = RTPpacket->rcuPlCntr;
- bufferInst->rcuPlCntr[bufferInst->insertPosition] = 0;
- bufferInst->waitingTime[bufferInst->insertPosition] = 0;
- /* Update buffer parameters */
- bufferInst->numPacketsInBuffer++;
- bufferInst->currentMemoryPos += (RTPpacket->payloadLen + 1) >> 1;
-
-#ifdef NETEQ_DELAY_LOGGING
- /* special code for offline delay logging */
- if (*flushed)
- {
- temp_var = NETEQ_DELAY_LOGGING_SIGNAL_FLUSH;
- fwrite( &temp_var, sizeof(int), 1, delay_fid2 );
- }
- temp_var = NETEQ_DELAY_LOGGING_SIGNAL_RECIN;
- fwrite( &temp_var, sizeof(int), 1, delay_fid2 );
- fwrite( &RTPpacket->timeStamp, sizeof(WebRtc_UWord32), 1, delay_fid2 );
- fwrite( &RTPpacket->seqNumber, sizeof(WebRtc_UWord16), 1, delay_fid2 );
- fwrite( &RTPpacket->payloadType, sizeof(int), 1, delay_fid2 );
- fwrite( &RTPpacket->payloadLen, sizeof(WebRtc_Word16), 1, delay_fid2 );
- tot_received_packets++;
-#endif /* NETEQ_DELAY_LOGGING */
-
- return (0);
-}
-
-
-int WebRtcNetEQ_PacketBufferExtract(PacketBuf_t *bufferInst, RTPPacket_t *RTPpacket,
- int bufferPosition, int *waitingTime)
-{
-
- /* Sanity check */
- if (bufferInst->startPayloadMemory == NULL)
- {
- /* packet buffer has not been initialized */
- return (PBUFFER_NOT_INITIALIZED);
- }
-
- if (bufferPosition < 0 || bufferPosition >= bufferInst->maxInsertPositions)
- {
- /* buffer position is outside valid range */
- return (NETEQ_OTHER_ERROR);
- }
-
- /* Check that there is a valid payload in the specified position */
- if (bufferInst->payloadLengthBytes[bufferPosition] <= 0)
- {
- /* The position does not contain a valid payload */
- RTPpacket->payloadLen = 0; /* Set zero length */
- return (PBUFFER_NONEXISTING_PACKET); /* Return error */
- }
-
- /* Payload exists => extract payload data */
-
- /* Copy the actual data payload to RTP packet struct */
-
- WEBRTC_SPL_MEMCPY_W16((WebRtc_Word16*) RTPpacket->payload,
- bufferInst->payloadLocation[bufferPosition],
- (bufferInst->payloadLengthBytes[bufferPosition] + 1) >> 1); /*length in WebRtc_Word16*/
-
- /* Copy payload parameters */
- RTPpacket->payloadLen = bufferInst->payloadLengthBytes[bufferPosition];
- RTPpacket->payloadType = bufferInst->payloadType[bufferPosition];
- RTPpacket->seqNumber = bufferInst->seqNumber[bufferPosition];
- RTPpacket->timeStamp = bufferInst->timeStamp[bufferPosition];
- RTPpacket->rcuPlCntr = bufferInst->rcuPlCntr[bufferPosition];
- *waitingTime = bufferInst->waitingTime[bufferPosition];
- RTPpacket->starts_byte1 = 0; /* payload is 16-bit aligned */
-
- /* Clear the position in the packet buffer */
- bufferInst->payloadType[bufferPosition] = -1;
- bufferInst->payloadLengthBytes[bufferPosition] = 0;
- bufferInst->seqNumber[bufferPosition] = 0;
- bufferInst->timeStamp[bufferPosition] = 0;
- bufferInst->waitingTime[bufferPosition] = 0;
- bufferInst->payloadLocation[bufferPosition] = bufferInst->startPayloadMemory;
-
- /* Reduce packet counter with one */
- bufferInst->numPacketsInBuffer--;
-
- return (0);
-}
-
-int WebRtcNetEQ_PacketBufferFindLowestTimestamp(PacketBuf_t* buffer_inst,
- uint32_t current_time_stamp,
- uint32_t* time_stamp,
- int* buffer_position,
- int erase_old_packets,
- int16_t* payload_type) {
- int32_t time_stamp_diff = WEBRTC_SPL_WORD32_MAX; /* Smallest diff found. */
- int32_t new_diff;
- int i;
- int16_t rcu_payload_cntr;
-
- if (buffer_inst->startPayloadMemory == NULL) {
- /* Packet buffer has not been initialized. */
- return PBUFFER_NOT_INITIALIZED;
- }
-
- /* Initialize all return values. */
- *time_stamp = 0;
- *payload_type = -1; /* Indicates that no packet was found. */
- *buffer_position = -1; /* Indicates that no packet was found. */
- rcu_payload_cntr = WEBRTC_SPL_WORD16_MAX; /* Indicates no packet found. */
-
- /* Check if buffer is empty. */
- if (buffer_inst->numPacketsInBuffer <= 0) {
- return 0;
- }
-
- /* Loop through all slots in buffer. */
- if (erase_old_packets) { /* If old payloads should be discarded. */
- for (i = 0; i < buffer_inst->maxInsertPositions; ++i) {
- /* Calculate difference between this slot and current_time_stamp. */
- new_diff = (int32_t)(buffer_inst->timeStamp[i] - current_time_stamp);
-
- /* Check if payload should be discarded. */
- if ((new_diff < 0) /* Payload is too old */
- && (new_diff > -30000) /* Account for TS wrap-around. */
- && (buffer_inst->payloadLengthBytes[i] > 0)) { /* Payload exists. */
- /* Throw away old packet. */
-
- /* Clear the position in the buffer. */
- buffer_inst->payloadType[i] = -1;
- buffer_inst->payloadLengthBytes[i] = 0;
-
- /* Reduce packet counter by one. */
- buffer_inst->numPacketsInBuffer--;
- /* Increase discard counter for in-call statistics. */
- buffer_inst->discardedPackets++;
- } else if (((new_diff < time_stamp_diff)
- || ((new_diff == time_stamp_diff)
- && (buffer_inst->rcuPlCntr[i] < rcu_payload_cntr)))
- && (buffer_inst->payloadLengthBytes[i] > 0)) {
- /* New diff is smaller than previous diffs or we have a candidate with a
- * time stamp as previous candidate but better RCU-counter;
- * and the payload exists.
- */
- /* Save this position as the best candidate. */
- *buffer_position = i;
- time_stamp_diff = new_diff;
- *payload_type = buffer_inst->payloadType[i];
- rcu_payload_cntr = buffer_inst->rcuPlCntr[i];
- }
- }
- } else {
- for (i = 0; i < buffer_inst->maxInsertPositions; ++i) {
- /* Calculate difference between this slot and current_time_stamp. */
- new_diff = (int32_t)(buffer_inst->timeStamp[i] - current_time_stamp);
-
- /* Check if this is the oldest packet. */
- if (((new_diff < time_stamp_diff)
- || ((new_diff == time_stamp_diff)
- && (buffer_inst->rcuPlCntr[i] < rcu_payload_cntr)))
- && (buffer_inst->payloadLengthBytes[i] > 0)) {
- /* New diff is smaller than previous diffs or we have a candidate with a
- * time_stamp as previous candidate but better RCU-counter;
- * and the payload exists.
- */
- /* Save this position as the best candidate. */
- *buffer_position = i;
- time_stamp_diff = new_diff;
- *payload_type = buffer_inst->payloadType[i];
- rcu_payload_cntr = buffer_inst->rcuPlCntr[i];
- }
- }
- }
-
- /* Check that we did find a real position. */
- if (*buffer_position >= 0) {
- /* Get the time_stamp for the best position. */
- *time_stamp = buffer_inst->timeStamp[*buffer_position];
- }
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcNetEQ_PacketBufferGetSize(const PacketBuf_t *bufferInst)
-{
- int i, count;
- WebRtc_Word32 sizeSamples;
-
- count = 0;
-
- /* Loop through all slots in the buffer */
- for (i = 0; i < bufferInst->maxInsertPositions; i++)
- {
- /* Only count the packets with non-zero size */
- if (bufferInst->payloadLengthBytes[i] != 0)
- {
- count++;
- }
- }
-
- /*
- * Calculate buffer size as number of packets times packet size
- * (packet size is that of the latest decoded packet)
- */
- sizeSamples = WEBRTC_SPL_MUL_16_16(bufferInst->packSizeSamples, count);
-
- /* Sanity check; size cannot be negative */
- if (sizeSamples < 0)
- {
- sizeSamples = 0;
- }
-
- return sizeSamples;
-}
-
-void WebRtcNetEQ_IncrementWaitingTimes(PacketBuf_t *buffer_inst) {
- int i;
- /* Loop through all slots in the buffer. */
- for (i = 0; i < buffer_inst->maxInsertPositions; ++i) {
- /* Only increment waiting time for the packets with non-zero size. */
- if (buffer_inst->payloadLengthBytes[i] != 0) {
- buffer_inst->waitingTime[i]++;
- }
- }
-}
-
-int WebRtcNetEQ_GetDefaultCodecSettings(const enum WebRtcNetEQDecoder *codecID,
- int noOfCodecs, int *maxBytes, int *maxSlots)
-{
- int i;
- int ok = 0;
- WebRtc_Word16 w16_tmp;
- WebRtc_Word16 codecBytes;
- WebRtc_Word16 codecBuffers;
-
- /* Initialize return variables to zero */
- *maxBytes = 0;
- *maxSlots = 0;
-
- /* Loop through all codecs supplied to function */
- for (i = 0; i < noOfCodecs; i++)
- {
- /* Find current codec and set parameters accordingly */
-
- if (codecID[i] == kDecoderPCMu)
- {
- codecBytes = 1680; /* Up to 210ms @ 64kbps */
- codecBuffers = 30; /* Down to 5ms frames */
- }
- else if (codecID[i] == kDecoderPCMa)
- {
- codecBytes = 1680; /* Up to 210ms @ 64kbps */
- codecBuffers = 30; /* Down to 5ms frames */
- }
- else if (codecID[i] == kDecoderILBC)
- {
- codecBytes = 380; /* 200ms @ 15.2kbps (20ms frames) */
- codecBuffers = 10;
- }
- else if (codecID[i] == kDecoderISAC)
- {
- codecBytes = 960; /* 240ms @ 32kbps (60ms frames) */
- codecBuffers = 8;
- }
- else if (codecID[i] == kDecoderISACswb)
- {
- codecBytes = 1560; /* 240ms @ 52kbps (30ms frames) */
- codecBuffers = 8;
- }
- else if (codecID[i] == kDecoderPCM16B)
- {
- codecBytes = 3360; /* 210ms */
- codecBuffers = 15;
- }
- else if (codecID[i] == kDecoderPCM16Bwb)
- {
- codecBytes = 6720; /* 210ms */
- codecBuffers = 15;
- }
- else if (codecID[i] == kDecoderPCM16Bswb32kHz)
- {
- codecBytes = 13440; /* 210ms */
- codecBuffers = 15;
- }
- else if (codecID[i] == kDecoderPCM16Bswb48kHz)
- {
- codecBytes = 20160; /* 210ms */
- codecBuffers = 15;
- }
- else if (codecID[i] == kDecoderG722)
- {
- codecBytes = 1680; /* 210ms @ 64kbps */
- codecBuffers = 15;
- }
- else if (codecID[i] == kDecoderRED)
- {
- codecBytes = 0; /* Should not be max... */
- codecBuffers = 0;
- }
- else if (codecID[i] == kDecoderAVT)
- {
- codecBytes = 0; /* Should not be max... */
- codecBuffers = 0;
- }
- else if (codecID[i] == kDecoderCNG)
- {
- codecBytes = 0; /* Should not be max... */
- codecBuffers = 0;
- }
- else if (codecID[i] == kDecoderG729)
- {
- codecBytes = 210; /* 210ms @ 8kbps */
- codecBuffers = 20; /* max 200ms supported for 10ms frames */
- }
- else if (codecID[i] == kDecoderG729_1)
- {
- codecBytes = 840; /* 210ms @ 32kbps */
- codecBuffers = 10; /* max 200ms supported for 20ms frames */
- }
- else if (codecID[i] == kDecoderG726_16)
- {
- codecBytes = 400; /* 200ms @ 16kbps */
- codecBuffers = 10;
- }
- else if (codecID[i] == kDecoderG726_24)
- {
- codecBytes = 600; /* 200ms @ 24kbps */
- codecBuffers = 10;
- }
- else if (codecID[i] == kDecoderG726_32)
- {
- codecBytes = 800; /* 200ms @ 32kbps */
- codecBuffers = 10;
- }
- else if (codecID[i] == kDecoderG726_40)
- {
- codecBytes = 1000; /* 200ms @ 40kbps */
- codecBuffers = 10;
- }
- else if (codecID[i] == kDecoderG722_1_16)
- {
- codecBytes = 420; /* 210ms @ 16kbps */
- codecBuffers = 10;
- }
- else if (codecID[i] == kDecoderG722_1_24)
- {
- codecBytes = 630; /* 210ms @ 24kbps */
- codecBuffers = 10;
- }
- else if (codecID[i] == kDecoderG722_1_32)
- {
- codecBytes = 840; /* 210ms @ 32kbps */
- codecBuffers = 10;
- }
- else if (codecID[i] == kDecoderG722_1C_24)
- {
- codecBytes = 630; /* 210ms @ 24kbps */
- codecBuffers = 10;
- }
- else if (codecID[i] == kDecoderG722_1C_32)
- {
- codecBytes = 840; /* 210ms @ 32kbps */
- codecBuffers = 10;
- }
- else if (codecID[i] == kDecoderG722_1C_48)
- {
- codecBytes = 1260; /* 210ms @ 48kbps */
- codecBuffers = 10;
- }
- else if (codecID[i] == kDecoderSPEEX_8)
- {
- codecBytes = 1250; /* 210ms @ 50kbps */
- codecBuffers = 10;
- }
- else if (codecID[i] == kDecoderSPEEX_16)
- {
- codecBytes = 1250; /* 210ms @ 50kbps */
- codecBuffers = 10;
- }
- else if (codecID[i] == kDecoderCELT_32)
- {
- codecBytes = 1250; /* 210ms @ 50kbps */
- codecBuffers = 10;
- }
- else if (codecID[i] == kDecoderGSMFR)
- {
- codecBytes = 340; /* 200ms */
- codecBuffers = 10;
- }
- else if (codecID[i] == kDecoderAMR)
- {
- codecBytes = 384; /* 240ms @ 12.2kbps+headers (60ms frames) */
- codecBuffers = 10;
- }
- else if (codecID[i] == kDecoderAMRWB)
- {
- codecBytes = 744;
- codecBuffers = 10;
- }
- else if (codecID[i] == kDecoderArbitrary)
- {
- codecBytes = 6720; /* Assume worst case uncompressed WB 210ms */
- codecBuffers = 15;
- }
- else
- {
- /*Unknow codec */
- codecBytes = 0;
- codecBuffers = 0;
- ok = CODEC_DB_UNKNOWN_CODEC;
- }
-
- /* Update max variables */
- *maxBytes = WEBRTC_SPL_MAX((*maxBytes), codecBytes);
- *maxSlots = WEBRTC_SPL_MAX((*maxSlots), codecBuffers);
-
- } /* end of for loop */
-
- /*
- * Add size needed by the additional pointers for each slot inside struct,
- * as indicated on each line below.
- */
- w16_tmp = (sizeof(WebRtc_UWord32) /* timeStamp */
- + sizeof(WebRtc_Word16*) /* payloadLocation */
- + sizeof(WebRtc_UWord16) /* seqNumber */
- + sizeof(WebRtc_Word16) /* payloadType */
- + sizeof(WebRtc_Word16) /* payloadLengthBytes */
- + sizeof(WebRtc_Word16) /* rcuPlCntr */
- + sizeof(int)); /* waitingTime */
- /* Add the extra size per slot to the memory count */
- *maxBytes += w16_tmp * (*maxSlots);
-
- return ok;
-}
diff --git a/trunk/src/modules/audio_coding/neteq/packet_buffer.h b/trunk/src/modules/audio_coding/neteq/packet_buffer.h
deleted file mode 100644
index 662f8af..0000000
--- a/trunk/src/modules/audio_coding/neteq/packet_buffer.h
+++ /dev/null
@@ -1,220 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Interface for the actual packet buffer data structure.
- */
-
-#ifndef PACKET_BUFFER_H
-#define PACKET_BUFFER_H
-
-#include "typedefs.h"
-
-#include "webrtc_neteq.h"
-#include "rtp.h"
-
-/* Define minimum allowed buffer memory, in 16-bit words */
-#define PBUFFER_MIN_MEMORY_SIZE 150
-
-/****************************/
-/* The packet buffer struct */
-/****************************/
-
-typedef struct
-{
-
- /* Variables common to the entire buffer */
- WebRtc_UWord16 packSizeSamples; /* packet size in samples of last decoded packet */
- WebRtc_Word16 *startPayloadMemory; /* pointer to the payload memory */
- int memorySizeW16; /* the size (in WebRtc_Word16) of the payload memory */
- WebRtc_Word16 *currentMemoryPos; /* The memory position to insert next payload */
- int numPacketsInBuffer; /* The number of packets in the buffer */
- int insertPosition; /* The position to insert next packet */
- int maxInsertPositions; /* Maximum number of packets allowed */
-
- /* Arrays with one entry per packet slot */
- /* NOTE: If these are changed, the changes must be accounted for at the end of
- the function WebRtcNetEQ_GetDefaultCodecSettings(). */
- WebRtc_UWord32 *timeStamp; /* Timestamp in slot n */
- WebRtc_Word16 **payloadLocation; /* Memory location of payload in slot n */
- WebRtc_UWord16 *seqNumber; /* Sequence number in slot n */
- WebRtc_Word16 *payloadType; /* Payload type of packet in slot n */
- WebRtc_Word16 *payloadLengthBytes; /* Payload length of packet in slot n */
- WebRtc_Word16 *rcuPlCntr; /* zero for non-RCU payload, 1 for main payload
- 2 for redundant payload */
- int *waitingTime;
-
-
- /* Statistics counter */
- WebRtc_UWord16 discardedPackets; /* Number of discarded packets */
-
-} PacketBuf_t;
-
-/*************************/
-/* Function declarations */
-/*************************/
-
-/****************************************************************************
- * WebRtcNetEQ_PacketBufferInit(...)
- *
- * This function initializes the packet buffer.
- *
- * Input:
- * - bufferInst : Buffer instance to be initialized
- * - noOfPackets : Maximum number of packets that buffer should hold
- * - memory : Pointer to the storage memory for the payloads
- * - memorySize : The size of the payload memory (in WebRtc_Word16)
- *
- * Output:
- * - bufferInst : Updated buffer instance
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-
-int WebRtcNetEQ_PacketBufferInit(PacketBuf_t *bufferInst, int maxNoOfPackets,
- WebRtc_Word16 *pw16_memory, int memorySize);
-
-/****************************************************************************
- * WebRtcNetEQ_PacketBufferFlush(...)
- *
- * This function flushes all the packets in the buffer.
- *
- * Input:
- * - bufferInst : Buffer instance to be flushed
- *
- * Output:
- * - bufferInst : Flushed buffer instance
- *
- * Return value : 0 - Ok
- */
-
-int WebRtcNetEQ_PacketBufferFlush(PacketBuf_t *bufferInst);
-
-/****************************************************************************
- * WebRtcNetEQ_PacketBufferInsert(...)
- *
- * This function inserts an RTP packet into the packet buffer.
- *
- * Input:
- * - bufferInst : Buffer instance
- * - RTPpacket : An RTP packet struct (with payload, sequence
- * number, etc.)
- *
- * Output:
- * - bufferInst : Updated buffer instance
- * - flushed : 1 if buffer was flushed, 0 otherwise
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_PacketBufferInsert(PacketBuf_t *bufferInst, const RTPPacket_t *RTPpacket,
- WebRtc_Word16 *flushed);
-
-/****************************************************************************
- * WebRtcNetEQ_PacketBufferExtract(...)
- *
- * This function extracts a payload from the buffer.
- *
- * Input:
- * - bufferInst : Buffer instance
- * - bufferPosition: Position of the packet that should be extracted
- *
- * Output:
- * - RTPpacket : An RTP packet struct (with payload, sequence
- * number, etc)
- * - bufferInst : Updated buffer instance
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-
-int WebRtcNetEQ_PacketBufferExtract(PacketBuf_t *bufferInst, RTPPacket_t *RTPpacket,
- int bufferPosition, int *waitingTime);
-
-/****************************************************************************
- * WebRtcNetEQ_PacketBufferFindLowestTimestamp(...)
- *
- * This function finds the next packet with the lowest timestamp.
- *
- * Input:
- * - buffer_inst : Buffer instance.
- * - current_time_stamp : The timestamp to compare packet timestamps with.
- * - erase_old_packets : If non-zero, erase packets older than currentTS.
- *
- * Output:
- * - time_stamp : Lowest timestamp that was found.
- * - buffer_position : Position of this packet (-1 if there are no
- * packets in the buffer).
- * - payload_type : Payload type of the found payload.
- *
- * Return value : 0 - Ok;
- * < 0 - Error.
- */
-
-int WebRtcNetEQ_PacketBufferFindLowestTimestamp(PacketBuf_t* buffer_inst,
- uint32_t current_time_stamp,
- uint32_t* time_stamp,
- int* buffer_position,
- int erase_old_packets,
- int16_t* payload_type);
-
-/****************************************************************************
- * WebRtcNetEQ_PacketBufferGetSize(...)
- *
- * Calculate and return an estimate of the total data length (in samples)
- * currently in the buffer. The estimate is calculated as the number of
- * packets currently in the buffer (which does not have any remaining waiting
- * time), multiplied with the number of samples obtained from the last
- * decoded packet.
- *
- * Input:
- * - bufferInst : Buffer instance
- *
- * Return value : The buffer size in samples
- */
-
-WebRtc_Word32 WebRtcNetEQ_PacketBufferGetSize(const PacketBuf_t *bufferInst);
-
-/****************************************************************************
- * WebRtcNetEQ_IncrementWaitingTimes(...)
- *
- * Increment the waiting time for all packets in the buffer by one.
- *
- * Input:
- * - bufferInst : Buffer instance
- *
- * Return value : n/a
- */
-
-void WebRtcNetEQ_IncrementWaitingTimes(PacketBuf_t *buffer_inst);
-
-/****************************************************************************
- * WebRtcNetEQ_GetDefaultCodecSettings(...)
- *
- * Calculates a recommended buffer size for a specific set of codecs.
- *
- * Input:
- * - codecID : An array of codec types that will be used
- * - noOfCodecs : Number of codecs in array codecID
- *
- * Output:
- * - maxBytes : Recommended buffer memory size in bytes
- * - maxSlots : Recommended number of slots in buffer
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-
-int WebRtcNetEQ_GetDefaultCodecSettings(const enum WebRtcNetEQDecoder *codecID,
- int noOfCodecs, int *maxBytes, int *maxSlots);
-
-#endif /* PACKET_BUFFER_H */
diff --git a/trunk/src/modules/audio_coding/neteq/peak_detection.c b/trunk/src/modules/audio_coding/neteq/peak_detection.c
deleted file mode 100644
index 678c7f9..0000000
--- a/trunk/src/modules/audio_coding/neteq/peak_detection.c
+++ /dev/null
@@ -1,232 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Implementation of the peak detection used for finding correlation peaks.
- */
-
-#include "dsp_helpfunctions.h"
-
-#include "signal_processing_library.h"
-
-/* Table of constants used in parabolic fit function WebRtcNetEQ_PrblFit */
-const WebRtc_Word16 WebRtcNetEQ_kPrblCf[17][3] = { { 120, 32, 64 }, { 140, 44, 75 },
- { 150, 50, 80 }, { 160, 57, 85 },
- { 180, 72, 96 }, { 200, 89, 107 },
- { 210, 98, 112 }, { 220, 108, 117 },
- { 240, 128, 128 }, { 260, 150, 139 },
- { 270, 162, 144 }, { 280, 174, 149 },
- { 300, 200, 160 }, { 320, 228, 171 },
- { 330, 242, 176 }, { 340, 257, 181 },
- { 360, 288, 192 } };
-
-WebRtc_Word16 WebRtcNetEQ_PeakDetection(WebRtc_Word16 *pw16_data, WebRtc_Word16 w16_dataLen,
- WebRtc_Word16 w16_nmbPeaks, WebRtc_Word16 fs_mult,
- WebRtc_Word16 *pw16_winIndex,
- WebRtc_Word16 *pw16_winValue)
-{
- /* Local variables */
- int i;
- WebRtc_Word16 w16_tmp;
- WebRtc_Word16 w16_tmp2;
- WebRtc_Word16 indMin = 0;
- WebRtc_Word16 indMax = 0;
-
- /* Peak detection */
-
- for (i = 0; i <= (w16_nmbPeaks - 1); i++)
- {
- if (w16_nmbPeaks == 1)
- {
- /*
- * Single peak
- * The parabola fit assumes that an extra point is available; worst case it gets
- * a zero on the high end of the signal.
- */
- w16_dataLen++;
- }
-
- pw16_winIndex[i] = WebRtcSpl_MaxIndexW16(pw16_data, (WebRtc_Word16) (w16_dataLen - 1));
-
- if (i != w16_nmbPeaks - 1)
- {
- w16_tmp = pw16_winIndex[i] - 2; /* *fs_mult; */
- indMin = WEBRTC_SPL_MAX(0, w16_tmp);
- w16_tmp = pw16_winIndex[i] + 2; /* *fs_mult; */
- w16_tmp2 = w16_dataLen - 1;
- indMax = WEBRTC_SPL_MIN(w16_tmp2, w16_tmp);
- }
-
- if ((pw16_winIndex[i] != 0) && (pw16_winIndex[i] != (w16_dataLen - 2)))
- {
- /* Parabola fit*/
- WebRtcNetEQ_PrblFit(&(pw16_data[pw16_winIndex[i] - 1]), &(pw16_winIndex[i]),
- &(pw16_winValue[i]), fs_mult);
- }
- else
- {
- if (pw16_winIndex[i] == (w16_dataLen - 2))
- {
- if (pw16_data[pw16_winIndex[i]] > pw16_data[pw16_winIndex[i] + 1])
- {
- WebRtcNetEQ_PrblFit(&(pw16_data[pw16_winIndex[i] - 1]),
- &(pw16_winIndex[i]), &(pw16_winValue[i]), fs_mult);
- }
- else if (pw16_data[pw16_winIndex[i]] <= pw16_data[pw16_winIndex[i] + 1])
- {
- pw16_winValue[i] = (pw16_data[pw16_winIndex[i]]
- + pw16_data[pw16_winIndex[i] + 1]) >> 1; /* lin approx */
- pw16_winIndex[i] = (pw16_winIndex[i] * 2 + 1) * fs_mult;
- }
- }
- else
- {
- pw16_winValue[i] = pw16_data[pw16_winIndex[i]];
- pw16_winIndex[i] = pw16_winIndex[i] * 2 * fs_mult;
- }
- }
-
- if (i != w16_nmbPeaks - 1)
- {
- WebRtcSpl_MemSetW16(&(pw16_data[indMin]), 0, (indMax - indMin + 1));
- /* for (j=indMin; j<=indMax; j++) pw16_data[j] = 0; */
- }
- }
-
- return 0;
-}
-
-WebRtc_Word16 WebRtcNetEQ_PrblFit(WebRtc_Word16 *pw16_3pts, WebRtc_Word16 *pw16_Ind,
- WebRtc_Word16 *pw16_outVal, WebRtc_Word16 fs_mult)
-{
- /* Variables */
- WebRtc_Word32 Num, Den;
- WebRtc_Word32 temp;
- WebRtc_Word16 flag, stp, strt, lmt;
- WebRtc_UWord16 PFind[13];
-
- if (fs_mult == 1)
- {
- PFind[0] = 0;
- PFind[1] = 8;
- PFind[2] = 16;
- }
- else if (fs_mult == 2)
- {
- PFind[0] = 0;
- PFind[1] = 4;
- PFind[2] = 8;
- PFind[3] = 12;
- PFind[4] = 16;
- }
- else if (fs_mult == 4)
- {
- PFind[0] = 0;
- PFind[1] = 2;
- PFind[2] = 4;
- PFind[3] = 6;
- PFind[4] = 8;
- PFind[5] = 10;
- PFind[6] = 12;
- PFind[7] = 14;
- PFind[8] = 16;
- }
- else
- {
- PFind[0] = 0;
- PFind[1] = 1;
- PFind[2] = 3;
- PFind[3] = 4;
- PFind[4] = 5;
- PFind[5] = 7;
- PFind[6] = 8;
- PFind[7] = 9;
- PFind[8] = 11;
- PFind[9] = 12;
- PFind[10] = 13;
- PFind[11] = 15;
- PFind[12] = 16;
- }
-
- /* Num = -3*pw16_3pts[0] + 4*pw16_3pts[1] - pw16_3pts[2]; */
- /* Den = pw16_3pts[0] - 2*pw16_3pts[1] + pw16_3pts[2]; */
- Num = WEBRTC_SPL_MUL_16_16(pw16_3pts[0],-3) + WEBRTC_SPL_MUL_16_16(pw16_3pts[1],4)
- - pw16_3pts[2];
-
- Den = pw16_3pts[0] + WEBRTC_SPL_MUL_16_16(pw16_3pts[1],-2) + pw16_3pts[2];
-
- temp = (WebRtc_Word32) WEBRTC_SPL_MUL(Num, (WebRtc_Word32)120); /* need 32_16 really */
- flag = 1;
- stp = WebRtcNetEQ_kPrblCf[PFind[fs_mult]][0] - WebRtcNetEQ_kPrblCf[PFind[fs_mult - 1]][0];
- strt = (WebRtcNetEQ_kPrblCf[PFind[fs_mult]][0]
- + WebRtcNetEQ_kPrblCf[PFind[fs_mult - 1]][0]) >> 1;
-
- if (temp < (WebRtc_Word32) WEBRTC_SPL_MUL(-Den,(WebRtc_Word32)strt))
- {
- lmt = strt - stp;
- while (flag)
- {
- if ((flag == fs_mult) || (temp
- > (WebRtc_Word32) WEBRTC_SPL_MUL(-Den,(WebRtc_Word32)lmt)))
- {
- *pw16_outVal
- = (WebRtc_Word16)
- (((WebRtc_Word32) ((WebRtc_Word32) WEBRTC_SPL_MUL(Den,(WebRtc_Word32)WebRtcNetEQ_kPrblCf[PFind[fs_mult-flag]][1])
- + (WebRtc_Word32) WEBRTC_SPL_MUL(Num,(WebRtc_Word32)WebRtcNetEQ_kPrblCf[PFind[fs_mult-flag]][2])
- + WEBRTC_SPL_MUL_16_16(pw16_3pts[0],256))) >> 8);
- *pw16_Ind = (*pw16_Ind) * (fs_mult << 1) - flag;
- flag = 0;
- }
- else
- {
- flag++;
- lmt -= stp;
- }
- }
- }
- else if (temp > (WebRtc_Word32) WEBRTC_SPL_MUL(-Den,(WebRtc_Word32)(strt+stp)))
- {
- lmt = strt + (stp << 1);
- while (flag)
- {
- if ((flag == fs_mult) || (temp
- < (WebRtc_Word32) WEBRTC_SPL_MUL(-Den,(WebRtc_Word32)lmt)))
- {
- WebRtc_Word32 temp_term_1, temp_term_2, temp_term_3;
-
- temp_term_1 = WEBRTC_SPL_MUL(Den,
- (WebRtc_Word32) WebRtcNetEQ_kPrblCf[PFind[fs_mult+flag]][1]);
- temp_term_2 = WEBRTC_SPL_MUL(Num,
- (WebRtc_Word32) WebRtcNetEQ_kPrblCf[PFind[fs_mult+flag]][2]);
- temp_term_3 = WEBRTC_SPL_MUL_16_16(pw16_3pts[0],256);
-
- *pw16_outVal
- = (WebRtc_Word16) ((temp_term_1 + temp_term_2 + temp_term_3) >> 8);
-
- *pw16_Ind = (*pw16_Ind) * (fs_mult << 1) + flag;
- flag = 0;
- }
- else
- {
- flag++;
- lmt += stp;
- }
- }
-
- }
- else
- {
- *pw16_outVal = pw16_3pts[1];
- *pw16_Ind = (*pw16_Ind) * 2 * fs_mult;
- }
-
- return 0;
-}
-
diff --git a/trunk/src/modules/audio_coding/neteq/preemptive_expand.c b/trunk/src/modules/audio_coding/neteq/preemptive_expand.c
deleted file mode 100644
index 167bc3a..0000000
--- a/trunk/src/modules/audio_coding/neteq/preemptive_expand.c
+++ /dev/null
@@ -1,524 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the Pre-emptive Expand algorithm that is used to increase
- * the delay by repeating a part of the audio stream.
- */
-
-#include "dsp.h"
-
-#include "signal_processing_library.h"
-
-#include "dsp_helpfunctions.h"
-#include "neteq_error_codes.h"
-
-#define PREEMPTIVE_CORR_LEN 50
-#define PREEMPTIVE_MIN_LAG 10
-#define PREEMPTIVE_MAX_LAG 60
-#define PREEMPTIVE_DOWNSAMPLED_LEN (PREEMPTIVE_CORR_LEN + PREEMPTIVE_MAX_LAG)
-
-/* Scratch usage:
-
- Type Name size startpos endpos
- WebRtc_Word16 pw16_downSampSpeech 110 0 109
- WebRtc_Word32 pw32_corr 2*50 110 209
- WebRtc_Word16 pw16_corr 50 0 49
-
- Total: 110+2*50
- */
-
-#define SCRATCH_PW16_DS_SPEECH 0
-#define SCRATCH_PW32_CORR PREEMPTIVE_DOWNSAMPLED_LEN
-#define SCRATCH_PW16_CORR 0
-
-/****************************************************************************
- * WebRtcNetEQ_PreEmptiveExpand(...)
- *
- * This function tries to extend the audio data by repeating one or several
- * pitch periods. The operation is only carried out if the correlation is
- * strong or if the signal energy is very low. The algorithm is the
- * reciprocal of the Accelerate algorithm.
- *
- * Input:
- * - inst : NetEQ DSP instance
- * - scratchPtr : Pointer to scratch vector.
- * - decoded : Pointer to newly decoded speech.
- * - len : Length of decoded speech.
- * - oldDataLen : Length of the part of decoded that has already been played out.
- * - BGNonly : If non-zero, Pre-emptive Expand will only copy
- * the first DEFAULT_TIME_ADJUST seconds of the
- * input and append to the end. No signal matching is
- * done.
- *
- * Output:
- * - inst : Updated instance
- * - outData : Pointer to a memory space where the output data
- * should be stored. The vector must be at least
- * min(len + 120*fs/8000, NETEQ_MAX_OUTPUT_SIZE)
- * elements long.
- * - pw16_len : Number of samples written to outData.
- *
- * Return value : 0 - Ok
- * <0 - Error
- */
-
-int WebRtcNetEQ_PreEmptiveExpand(DSPInst_t *inst,
-#ifdef SCRATCH
- WebRtc_Word16 *pw16_scratchPtr,
-#endif
- const WebRtc_Word16 *pw16_decoded, int len, int oldDataLen,
- WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len,
- WebRtc_Word16 BGNonly)
-{
-
-#ifdef SCRATCH
- /* Use scratch memory for internal temporary vectors */
- WebRtc_Word16 *pw16_downSampSpeech = pw16_scratchPtr + SCRATCH_PW16_DS_SPEECH;
- WebRtc_Word32 *pw32_corr = (WebRtc_Word32*) (pw16_scratchPtr + SCRATCH_PW32_CORR);
- WebRtc_Word16 *pw16_corr = pw16_scratchPtr + SCRATCH_PW16_CORR;
-#else
- /* Allocate memory for temporary vectors */
- WebRtc_Word16 pw16_downSampSpeech[PREEMPTIVE_DOWNSAMPLED_LEN];
- WebRtc_Word32 pw32_corr[PREEMPTIVE_CORR_LEN];
- WebRtc_Word16 pw16_corr[PREEMPTIVE_CORR_LEN];
-#endif
- WebRtc_Word16 w16_decodedMax = 0;
- WebRtc_Word16 w16_tmp = 0;
- WebRtc_Word16 w16_tmp2;
- WebRtc_Word32 w32_tmp;
- WebRtc_Word32 w32_tmp2;
-
- const WebRtc_Word16 w16_startLag = PREEMPTIVE_MIN_LAG;
- const WebRtc_Word16 w16_endLag = PREEMPTIVE_MAX_LAG;
- const WebRtc_Word16 w16_corrLen = PREEMPTIVE_CORR_LEN;
- const WebRtc_Word16 *pw16_vec1, *pw16_vec2;
- WebRtc_Word16 *pw16_vectmp;
- WebRtc_Word16 w16_inc, w16_startfact;
- WebRtc_Word16 w16_bestIndex, w16_bestVal;
- WebRtc_Word16 w16_VAD = 1;
- WebRtc_Word16 fsMult;
- WebRtc_Word16 fsMult120;
- WebRtc_Word32 w32_en1, w32_en2, w32_cc;
- WebRtc_Word16 w16_en1, w16_en2;
- WebRtc_Word16 w16_en1Scale, w16_en2Scale;
- WebRtc_Word16 w16_sqrtEn1En2;
- WebRtc_Word16 w16_bestCorr = 0;
- int ok;
-
-#ifdef NETEQ_STEREO
- MasterSlaveInfo *msInfo = inst->msInfo;
-#endif
-
- fsMult = WebRtcNetEQ_CalcFsMult(inst->fs); /* Calculate fs/8000 */
-
- /* Pre-calculate common multiplication with fsMult */
- fsMult120 = (WebRtc_Word16) WEBRTC_SPL_MUL_16_16(fsMult, 120); /* 15 ms */
-
- inst->ExpandInst.w16_consecExp = 0; /* Last was not expand any more */
-
- /*
- * Sanity check for len variable; must be (almost) 30 ms (120*fsMult + max(bestIndex)).
- * Also, the new part must be at least .625 ms (w16_overlap).
- */
- if (len < (WebRtc_Word16) WEBRTC_SPL_MUL_16_16((120 + 119), fsMult) || oldDataLen >= len
- - inst->ExpandInst.w16_overlap)
- {
- /* Length of decoded data too short */
- inst->w16_mode = MODE_UNSUCCESS_PREEMPTIVE;
- *pw16_len = len;
-
-
- /* simply move all data from decoded to outData */
-
- WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
-
- return NETEQ_OTHER_ERROR;
- }
-
- /***********************************/
- /* Special operations for BGN only */
- /***********************************/
-
- /* Check if "background noise only" flag is set */
- if (BGNonly)
- {
- /* special operation for BGN only; simply insert a chunk of data */
- w16_bestIndex = DEFAULT_TIME_ADJUST * (fsMult << 3); /* X*fs/1000 */
-
- /* Sanity check for bestIndex */
- if (w16_bestIndex > len)
- { /* not good, do nothing instead */
- inst->w16_mode = MODE_UNSUCCESS_PREEMPTIVE;
- *pw16_len = len;
-
-
- /* simply move all data from decoded to outData */
-
- WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
-
- return NETEQ_OTHER_ERROR;
- }
-
- /* set length parameter */
- *pw16_len = len + w16_bestIndex;
-
-
- /* copy to output */
-
- WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, len);
- WEBRTC_SPL_MEMCPY_W16(&pw16_outData[len], pw16_decoded, w16_bestIndex);
-
- /* set mode */
- inst->w16_mode = MODE_LOWEN_PREEMPTIVE;
-
- /* update statistics */
- inst->statInst.preemptiveLength += w16_bestIndex;
-
- return 0;
- } /* end of special code for BGN mode */
-
-#ifdef NETEQ_STEREO
-
- /* Sanity for msInfo */
- if (msInfo == NULL)
- {
- /* this should not happen here */
- return MASTER_SLAVE_ERROR;
- }
-
- if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO))
- {
- /* Find correlation lag only for non-slave instances */
-
-#endif
-
- /****************************************************************/
- /* Find the strongest correlation lag by downsampling to 4 kHz, */
- /* calculating correlation for downsampled signal and finding */
- /* the strongest correlation peak. */
- /****************************************************************/
-
- /* find maximum absolute value */
- w16_decodedMax = WebRtcSpl_MaxAbsValueW16(pw16_decoded, (WebRtc_Word16) len);
-
- /* downsample the decoded speech to 4 kHz */
- ok = WebRtcNetEQ_DownSampleTo4kHz(pw16_decoded, len, inst->fs, pw16_downSampSpeech,
- PREEMPTIVE_DOWNSAMPLED_LEN, 1 /* compensate delay*/);
- if (ok != 0)
- {
- /* error */
- inst->w16_mode = MODE_UNSUCCESS_PREEMPTIVE;
- *pw16_len = len;
-
-
- /* simply move all data from decoded to outData */
-
- WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
-
- return NETEQ_OTHER_ERROR;
- }
-
- /*
- * Set scaling factor for cross correlation to protect against
- * overflow (log2(50) => 6)
- */
- w16_tmp = 6 - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_decodedMax, w16_decodedMax));
- w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
-
- /* Perform correlation from lag 10 to lag 60 in 4 kHz domain */WebRtcNetEQ_CrossCorr(
- pw32_corr, &pw16_downSampSpeech[w16_endLag],
- &pw16_downSampSpeech[w16_endLag - w16_startLag], w16_corrLen,
- (WebRtc_Word16) (w16_endLag - w16_startLag), w16_tmp, -1);
-
- /* Normalize correlation to 14 bits and put in a WebRtc_Word16 vector */
- w32_tmp = WebRtcSpl_MaxAbsValueW32(pw32_corr, w16_corrLen);
- w16_tmp = 17 - WebRtcSpl_NormW32(w32_tmp);
- w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
-
- WebRtcSpl_VectorBitShiftW32ToW16(pw16_corr, w16_corrLen, pw32_corr, w16_tmp);
-
- /* Find limits for peak finding, in order to avoid overful NetEQ algorithm buffer. */
- /* Calculate difference between MAX_OUTPUT_SIZE and len in 4 kHz domain. */
- w16_tmp = WebRtcSpl_DivW32W16ResW16((WebRtc_Word32) (NETEQ_MAX_OUTPUT_SIZE - len),
- (WebRtc_Word16) (fsMult << 1)) - w16_startLag;
- w16_tmp = WEBRTC_SPL_MIN(w16_corrLen, w16_tmp); /* no more than corrLen = 50 */
-
-#ifdef NETEQ_STEREO
- } /* end if (msInfo->msMode != NETEQ_SLAVE) */
-
- if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO))
- {
- /* Find the strongest correlation peak by using the parabolic fit method */
- WebRtcNetEQ_PeakDetection(pw16_corr, w16_tmp, 1, fsMult, &w16_bestIndex, &w16_bestVal);
- /* 0 <= bestIndex <= (2*w16_tmp - 1)*fsMult <= (2*corrLen - 1)*fsMult = 99*fsMult */
-
- /* Compensate bestIndex for displaced starting position */
- w16_bestIndex = w16_bestIndex + w16_startLag * WEBRTC_SPL_LSHIFT_W16(fsMult, 1);
- /* 20*fsMult <= bestIndex <= 119*fsMult */
-
- msInfo->bestIndex = w16_bestIndex;
- }
- else if (msInfo->msMode == NETEQ_SLAVE)
- {
- if (msInfo->extraInfo == PE_EXP_FAIL)
- {
- /* Master has signaled an unsuccessful preemptive expand */
- w16_bestIndex = 0;
- }
- else
- {
- /* Get best index from master */
- w16_bestIndex = msInfo->bestIndex;
- }
- }
- else
- {
- /* Invalid mode */
- return (MASTER_SLAVE_ERROR);
- }
-
-#else /* NETEQ_STEREO */
-
- /* Find the strongest correlation peak by using the parabolic fit method */
- WebRtcNetEQ_PeakDetection(pw16_corr, w16_tmp, 1, fsMult, &w16_bestIndex, &w16_bestVal);
- /* 0 <= bestIndex <= (2*w16_tmp - 1)*fsMult <= (2*corrLen - 1)*fsMult = 99*fsMult */
-
- /* Compensate bestIndex for displaced starting position */
- w16_bestIndex = w16_bestIndex + w16_startLag * WEBRTC_SPL_LSHIFT_W16(fsMult, 1);
- /* 20*fsMult <= bestIndex <= 119*fsMult */
-
-#endif /* NETEQ_STEREO */
-
-#ifdef NETEQ_STEREO
-
- if ((msInfo->msMode == NETEQ_MASTER) || (msInfo->msMode == NETEQ_MONO))
- {
- /* Calculate correlation only for non-slave instances */
-
-#endif /* NETEQ_STEREO */
-
- /*****************************************************/
- /* Calculate correlation bestCorr for the found lag. */
- /* Also do a simple VAD decision. */
- /*****************************************************/
-
- /*
- * Calculate scaling to ensure that bestIndex samples can be square-summed
- * without overflowing
- */
- w16_tmp = (31
- - WebRtcSpl_NormW32(WEBRTC_SPL_MUL_16_16(w16_decodedMax, w16_decodedMax)));
- w16_tmp += (31 - WebRtcSpl_NormW32(w16_bestIndex));
- w16_tmp -= 31;
- w16_tmp = WEBRTC_SPL_MAX(0, w16_tmp);
-
- /* vec1 starts at 15 ms minus one pitch period */
- pw16_vec1 = &pw16_decoded[fsMult120 - w16_bestIndex];
- /* vec2 start at 15 ms */
- pw16_vec2 = &pw16_decoded[fsMult120];
-
- /* Calculate energies for vec1 and vec2 */
- w32_en1 = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec1,
- (WebRtc_Word16*) pw16_vec1, w16_bestIndex, w16_tmp);
- w32_en2 = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec2,
- (WebRtc_Word16*) pw16_vec2, w16_bestIndex, w16_tmp);
-
- /* Calculate cross-correlation at the found lag */
- w32_cc = WebRtcNetEQ_DotW16W16((WebRtc_Word16*) pw16_vec1, (WebRtc_Word16*) pw16_vec2,
- w16_bestIndex, w16_tmp);
-
- /* Check VAD constraint
- ((en1+en2)/(2*bestIndex)) <= 8*inst->BGNInst.energy */
- w32_tmp = WEBRTC_SPL_RSHIFT_W32(w32_en1 + w32_en2, 4); /* (en1+en2)/(2*8) */
- if (inst->BGNInst.w16_initialized == 1)
- {
- w32_tmp2 = inst->BGNInst.w32_energy;
- }
- else
- {
- /* if BGN parameters have not been estimated, use a fixed threshold */
- w32_tmp2 = 75000;
- }
- w16_tmp2 = 16 - WebRtcSpl_NormW32(w32_tmp2);
- w16_tmp2 = WEBRTC_SPL_MAX(0, w16_tmp2);
- w32_tmp = WEBRTC_SPL_RSHIFT_W32(w32_tmp, w16_tmp2);
- w16_tmp2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_tmp2, w16_tmp2);
- w32_tmp2 = WEBRTC_SPL_MUL_16_16(w16_bestIndex, w16_tmp2);
-
- /* Scale w32_tmp properly before comparing with w32_tmp2 */
- /* (w16_tmp is scaling before energy calculation, thus 2*w16_tmp) */
- if (WebRtcSpl_NormW32(w32_tmp) < WEBRTC_SPL_LSHIFT_W32(w16_tmp,1))
- {
- /* Cannot scale only w32_tmp, must scale w32_temp2 too */
- WebRtc_Word16 tempshift = WebRtcSpl_NormW32(w32_tmp);
- w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp, tempshift);
- w32_tmp2 = WEBRTC_SPL_RSHIFT_W32(w32_tmp2,
- WEBRTC_SPL_LSHIFT_W32(w16_tmp,1) - tempshift);
- }
- else
- {
- w32_tmp = WEBRTC_SPL_LSHIFT_W32(w32_tmp,
- WEBRTC_SPL_LSHIFT_W32(w16_tmp,1));
- }
-
- if (w32_tmp <= w32_tmp2) /*((en1+en2)/(2*bestIndex)) <= 8*inst->BGNInst.energy */
- {
- /* The signal seems to be passive speech */
- w16_VAD = 0;
- w16_bestCorr = 0; /* Correlation does not matter */
-
- /* For low energy expansion, the new data can be less than 15 ms,
- but we must ensure that bestIndex is not larger than the new data. */
- w16_bestIndex = WEBRTC_SPL_MIN( w16_bestIndex, len - oldDataLen );
- }
- else
- {
- /* The signal is active speech */
- w16_VAD = 1;
-
- /* Calculate correlation (cc/sqrt(en1*en2)) */
-
- /* Start with calculating scale values */
- w16_en1Scale = 16 - WebRtcSpl_NormW32(w32_en1);
- w16_en1Scale = WEBRTC_SPL_MAX(0, w16_en1Scale);
- w16_en2Scale = 16 - WebRtcSpl_NormW32(w32_en2);
- w16_en2Scale = WEBRTC_SPL_MAX(0, w16_en2Scale);
-
- /* Make sure total scaling is even (to simplify scale factor after sqrt) */
- if ((w16_en1Scale + w16_en2Scale) & 1)
- {
- w16_en1Scale += 1;
- }
-
- /* Convert energies to WebRtc_Word16 */
- w16_en1 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en1, w16_en1Scale);
- w16_en2 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(w32_en2, w16_en2Scale);
-
- /* Calculate energy product */
- w32_tmp = WEBRTC_SPL_MUL_16_16(w16_en1, w16_en2);
-
- /* Calculate square-root of energy product */
- w16_sqrtEn1En2 = (WebRtc_Word16) WebRtcSpl_SqrtFloor(w32_tmp);
-
- /* Calculate cc/sqrt(en1*en2) in Q14 */
- w16_tmp = 14 - ((w16_en1Scale + w16_en2Scale) >> 1);
- w32_cc = WEBRTC_SPL_SHIFT_W32(w32_cc, w16_tmp);
- w32_cc = WEBRTC_SPL_MAX(0, w32_cc); /* Don't divide with negative number */
- w16_bestCorr = (WebRtc_Word16) WebRtcSpl_DivW32W16(w32_cc, w16_sqrtEn1En2);
- w16_bestCorr = WEBRTC_SPL_MIN(16384, w16_bestCorr); /* set maximum to 1.0 */
- }
-
-#ifdef NETEQ_STEREO
-
- } /* end if (msInfo->msMode != NETEQ_SLAVE) */
-
-#endif /* NETEQ_STEREO */
-
- /*******************************************************/
- /* Check preemptive expand criteria and insert samples */
- /*******************************************************/
-
- /* Check for strong correlation (>0.9) and at least 15 ms new data,
- or passive speech */
-#ifdef NETEQ_STEREO
- if (((((w16_bestCorr > 14746) && (oldDataLen <= fsMult120)) || (w16_VAD == 0))
- && (msInfo->msMode != NETEQ_SLAVE)) || ((msInfo->msMode == NETEQ_SLAVE)
- && (msInfo->extraInfo != PE_EXP_FAIL)))
-#else
- if (((w16_bestCorr > 14746) && (oldDataLen <= fsMult120))
- || (w16_VAD == 0))
-#endif
- {
- /* Do expand operation by overlap add */
-
- /* Set length of the first part, not to be modified */
- WebRtc_Word16 w16_startIndex = WEBRTC_SPL_MAX(oldDataLen, fsMult120);
-
- /*
- * Calculate cross-fading slope so that the fading factor goes from
- * 1 (16384 in Q14) to 0 in one pitch period (bestIndex).
- */
- w16_inc = (WebRtc_Word16) WebRtcSpl_DivW32W16((WebRtc_Word32) 16384,
- (WebRtc_Word16) (w16_bestIndex + 1)); /* in Q14 */
-
- /* Initiate fading factor */
- w16_startfact = 16384 - w16_inc;
-
- /* vec1 starts at 15 ms minus one pitch period */
- pw16_vec1 = &pw16_decoded[w16_startIndex - w16_bestIndex];
- /* vec2 start at 15 ms */
- pw16_vec2 = &pw16_decoded[w16_startIndex];
-
-
- /* Copy unmodified part [0 to 15 ms] */
-
- WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, w16_startIndex);
-
- /* Generate interpolated part of length bestIndex (1 pitch period) */
- pw16_vectmp = pw16_outData + w16_startIndex;
- /* Reuse mixing function from Expand */
- WebRtcNetEQ_MixVoiceUnvoice(pw16_vectmp, (WebRtc_Word16*) pw16_vec2,
- (WebRtc_Word16*) pw16_vec1, &w16_startfact, w16_inc, w16_bestIndex);
-
- /* Move the last part (also unmodified) */
- /* Take from decoded at 15 ms */
- pw16_vec2 = &pw16_decoded[w16_startIndex];
- WEBRTC_SPL_MEMMOVE_W16(&pw16_outData[w16_startIndex + w16_bestIndex], pw16_vec2,
- (WebRtc_Word16) (len - w16_startIndex));
-
- /* Set the mode flag */
- if (w16_VAD)
- {
- inst->w16_mode = MODE_SUCCESS_PREEMPTIVE;
- }
- else
- {
- inst->w16_mode = MODE_LOWEN_PREEMPTIVE;
- }
-
- /* Calculate resulting length = original length + pitch period */
- *pw16_len = len + w16_bestIndex;
-
- /* Update in-call statistics */
- inst->statInst.preemptiveLength += w16_bestIndex;
-
- return 0;
- }
- else
- {
- /* Preemptive Expand not allowed */
-
-#ifdef NETEQ_STEREO
- /* Signal to slave(s) that this was unsuccessful */
- if (msInfo->msMode == NETEQ_MASTER)
- {
- msInfo->extraInfo = PE_EXP_FAIL;
- }
-#endif
-
- /* Set mode flag to unsuccessful preemptive expand */
- inst->w16_mode = MODE_UNSUCCESS_PREEMPTIVE;
-
- /* Length is unmodified */
- *pw16_len = len;
-
-
- /* Simply move all data from decoded to outData */
-
- WEBRTC_SPL_MEMMOVE_W16(pw16_outData, pw16_decoded, (WebRtc_Word16) len);
-
- return 0;
- }
-}
-
-#undef SCRATCH_PW16_DS_SPEECH
-#undef SCRATCH_PW32_CORR
-#undef SCRATCH_PW16_CORR
diff --git a/trunk/src/modules/audio_coding/neteq/random_vector.c b/trunk/src/modules/audio_coding/neteq/random_vector.c
deleted file mode 100644
index 217bacd..0000000
--- a/trunk/src/modules/audio_coding/neteq/random_vector.c
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This function generates a pseudo-random vector.
- */
-
-#include "dsp_helpfunctions.h"
-
-/*
- * Values are normalized so that
- * sqrt(dot(pw16_NETEQFIX_RANDN_TBL,pw16_NETEQFIX_RANDN_TBL)/256)=2^13
- */
-const WebRtc_Word16 WebRtcNetEQ_kRandnTbl[RANDVEC_NO_OF_SAMPLES] =
-{
- 2680, 5532, 441, 5520, 16170, -5146, -1024, -8733, 3115, 9598, -10380, -4959, -1280, -21716, 7133, -1522,
- 13458, -3902, 2789, -675, 3441, 5016, -13599, -4003, -2739, 3922, -7209, 13352, -11617, -7241, 12905, -2314,
- 5426, 10121, -9702, 11207, -13542, 1373, 816, -5934, -12504, 4798, 1811, 4112, -613, 201, -10367, -2960,
- -2419, 3442, 4299, -6116, -6092, 1552, -1650, -480, -1237, 18720, -11858, -8303, -8212, 865, -2890, -16968,
- 12052, -5845, -5912, 9777, -5665, -6294, 5426, -4737, -6335, 1652, 761, 3832, 641, -8552, -9084, -5753,
- 8146, 12156, -4915, 15086, -1231, -1869, 11749, -9319, -6403, 11407, 6232, -1683, 24340, -11166, 4017, -10448,
- 3153, -2936, 6212, 2891, -866, -404, -4807, -2324, -1917, -2388, -6470, -3895, -10300, 5323, -5403, 2205,
- 4640, 7022, -21186, -6244, -882, -10031, -3395, -12885, 7155, -5339, 5079, -2645, -9515, 6622, 14651, 15852,
- 359, 122, 8246, -3502, -6696, -3679, -13535, -1409, -704, -7403, -4007, 1798, 279, -420, -12796, -14219,
- 1141, 3359, 11434, 7049, -6684, -7473, 14283, -4115, -9123, -8969, 4152, 4117, 13792, 5742, 16168, 8661,
- -1609, -6095, 1881, 14380, -5588, 6758, -6425, -22969, -7269, 7031, 1119, -1611, -5850, -11281, 3559, -8952,
- -10146, -4667, -16251, -1538, 2062, -1012, -13073, 227, -3142, -5265, 20, 5770, -7559, 4740, -4819, 992,
- -8208, -7130, -4652, 6725, 7369, -1036, 13144, -1588, -5304, -2344, -449, -5705, -8894, 5205, -17904, -11188,
- -1022, 4852, 10101, -5255, -4200, -752, 7941, -1543, 5959, 14719, 13346, 17045, -15605, -1678, -1600, -9230,
- 68, 23348, 1172, 7750, 11212, -18227, 9956, 4161, 883, 3947, 4341, 1014, -4889, -2603, 1246, -5630,
- -3596, -870, -1298, 2784, -3317, -6612, -20541, 4166, 4181, -8625, 3562, 12890, 4761, 3205, -12259, -8579
-};
-
-
-void WebRtcNetEQ_RandomVec(WebRtc_UWord32 *w32_seed, WebRtc_Word16 *pw16_randVec,
- WebRtc_Word16 w16_len, WebRtc_Word16 w16_incval)
-{
- int i;
- WebRtc_Word16 w16_pos;
- for (i = 0; i < w16_len; i++)
- {
- *w32_seed = (*w32_seed) + w16_incval;
- w16_pos = (WebRtc_Word16) ((*w32_seed) & (RANDVEC_NO_OF_SAMPLES - 1));
- pw16_randVec[i] = WebRtcNetEQ_kRandnTbl[w16_pos];
- }
-}
-
diff --git a/trunk/src/modules/audio_coding/neteq/recin.c b/trunk/src/modules/audio_coding/neteq/recin.c
deleted file mode 100644
index 608eb6e..0000000
--- a/trunk/src/modules/audio_coding/neteq/recin.c
+++ /dev/null
@@ -1,472 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Implementation of the RecIn function, which is the main function for inserting RTP
- * packets into NetEQ.
- */
-
-#include "mcu.h"
-
-#include <string.h>
-
-#include "signal_processing_library.h"
-
-#include "automode.h"
-#include "dtmf_buffer.h"
-#include "neteq_defines.h"
-#include "neteq_error_codes.h"
-
-
-int WebRtcNetEQ_RecInInternal(MCUInst_t *MCU_inst, RTPPacket_t *RTPpacketInput,
- WebRtc_UWord32 uw32_timeRec)
-{
- RTPPacket_t RTPpacket[2];
- int i_k;
- int i_ok = 0, i_No_Of_Payloads = 1;
- WebRtc_Word16 flushed = 0;
- WebRtc_Word16 codecPos;
- int curr_Codec;
- WebRtc_Word16 isREDPayload = 0;
- WebRtc_Word32 temp_bufsize = MCU_inst->PacketBuffer_inst.numPacketsInBuffer;
-#ifdef NETEQ_RED_CODEC
- RTPPacket_t* RTPpacketPtr[2]; /* Support for redundancy up to 2 payloads */
- RTPpacketPtr[0] = &RTPpacket[0];
- RTPpacketPtr[1] = &RTPpacket[1];
-#endif
-
- /*
- * Copy from input RTP packet to local copy
- * (mainly to enable multiple payloads using RED)
- */
-
- WEBRTC_SPL_MEMCPY_W8(&RTPpacket[0], RTPpacketInput, sizeof(RTPPacket_t));
-
- /* Reinitialize NetEq if it's needed (changed SSRC or first call) */
-
- if ((RTPpacket[0].ssrc != MCU_inst->ssrc) || (MCU_inst->first_packet == 1))
- {
- WebRtcNetEQ_RTCPInit(&MCU_inst->RTCP_inst, RTPpacket[0].seqNumber);
- MCU_inst->first_packet = 0;
-
- /* Flush the buffer */
- WebRtcNetEQ_PacketBufferFlush(&MCU_inst->PacketBuffer_inst);
-
- /* Store new SSRC */
- MCU_inst->ssrc = RTPpacket[0].ssrc;
-
- /* Update codecs */
- MCU_inst->timeStamp = RTPpacket[0].timeStamp;
- MCU_inst->current_Payload = RTPpacket[0].payloadType;
-
- /*Set MCU to update codec on next SignalMCU call */
- MCU_inst->new_codec = 1;
-
- /* Reset timestamp scaling */
- MCU_inst->TSscalingInitialized = 0;
-
- }
-
- /* Call RTCP statistics */
- i_ok |= WebRtcNetEQ_RTCPUpdate(&(MCU_inst->RTCP_inst), RTPpacket[0].seqNumber,
- RTPpacket[0].timeStamp, uw32_timeRec);
-
- /* If Redundancy is supported and this is the redundancy payload, separate the payloads */
-#ifdef NETEQ_RED_CODEC
- if (RTPpacket[0].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst,
- kDecoderRED))
- {
-
- /* Split the payload into a main and a redundancy payloads */
- i_ok = WebRtcNetEQ_RedundancySplit(RTPpacketPtr, 2, &i_No_Of_Payloads);
- if (i_ok < 0)
- {
- /* error returned */
- return i_ok;
- }
-
- /*
- * Only accept a few redundancies of the same type as the main data,
- * AVT events and CNG.
- */
- if ((i_No_Of_Payloads > 1) && (RTPpacket[0].payloadType != RTPpacket[1].payloadType)
- && (RTPpacket[0].payloadType != WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst,
- kDecoderAVT)) && (RTPpacket[1].payloadType != WebRtcNetEQ_DbGetPayload(
- &MCU_inst->codec_DB_inst, kDecoderAVT)) && (!WebRtcNetEQ_DbIsCNGPayload(
- &MCU_inst->codec_DB_inst, RTPpacket[0].payloadType))
- && (!WebRtcNetEQ_DbIsCNGPayload(&MCU_inst->codec_DB_inst, RTPpacket[1].payloadType)))
- {
- i_No_Of_Payloads = 1;
- }
- isREDPayload = 1;
- }
-#endif
-
- /* loop over the number of payloads */
- for (i_k = 0; i_k < i_No_Of_Payloads; i_k++)
- {
-
- if (isREDPayload == 1)
- {
- RTPpacket[i_k].rcuPlCntr = i_k;
- }
- else
- {
- RTPpacket[i_k].rcuPlCntr = 0;
- }
-
- /* Force update of SplitInfo if it's iLBC because of potential change between 20/30ms */
- if (RTPpacket[i_k].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst,
- kDecoderILBC))
- {
- i_ok = WebRtcNetEQ_DbGetSplitInfo(
- &MCU_inst->PayloadSplit_inst,
- (enum WebRtcNetEQDecoder) WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst,
- RTPpacket[i_k].payloadType), RTPpacket[i_k].payloadLen);
- if (i_ok < 0)
- {
- /* error returned */
- return i_ok;
- }
- }
-
- /* Get information about timestamp scaling for this payload type */
- i_ok = WebRtcNetEQ_GetTimestampScaling(MCU_inst, RTPpacket[i_k].payloadType);
- if (i_ok < 0)
- {
- /* error returned */
- return i_ok;
- }
-
- if (MCU_inst->TSscalingInitialized == 0 && MCU_inst->scalingFactor != kTSnoScaling)
- {
- /* Must initialize scaling with current timestamps */
- MCU_inst->externalTS = RTPpacket[i_k].timeStamp;
- MCU_inst->internalTS = RTPpacket[i_k].timeStamp;
- MCU_inst->TSscalingInitialized = 1;
- }
-
- /* Adjust timestamp if timestamp scaling is needed (e.g. SILK or G.722) */
- if (MCU_inst->TSscalingInitialized == 1)
- {
- WebRtc_UWord32 newTS = WebRtcNetEQ_ScaleTimestampExternalToInternal(MCU_inst,
- RTPpacket[i_k].timeStamp);
-
- /* save the incoming timestamp for next time */
- MCU_inst->externalTS = RTPpacket[i_k].timeStamp;
-
- /* add the scaled difference to last scaled timestamp and save ... */
- MCU_inst->internalTS = newTS;
-
- RTPpacket[i_k].timeStamp = newTS;
- }
-
- /* Is this a DTMF packet?*/
- if (RTPpacket[i_k].payloadType == WebRtcNetEQ_DbGetPayload(&MCU_inst->codec_DB_inst,
- kDecoderAVT))
- {
-#ifdef NETEQ_ATEVENT_DECODE
- if (MCU_inst->AVT_PlayoutOn)
- {
- i_ok = WebRtcNetEQ_DtmfInsertEvent(&MCU_inst->DTMF_inst,
- RTPpacket[i_k].payload, RTPpacket[i_k].payloadLen,
- RTPpacket[i_k].timeStamp);
- if (i_ok != 0)
- {
- return i_ok;
- }
- }
-#endif
-#ifdef NETEQ_STEREO
- if (MCU_inst->usingStereo == 0)
- {
- /* do not set this for DTMF packets when using stereo mode */
- MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 1;
- }
-#else
- MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 1;
-#endif
- }
- else if (WebRtcNetEQ_DbIsCNGPayload(&MCU_inst->codec_DB_inst,
- RTPpacket[i_k].payloadType))
- {
- /* Is this a CNG packet? how should we handle this?*/
-#ifdef NETEQ_CNG_CODEC
- /* Get CNG sample rate */
- WebRtc_UWord16 fsCng = WebRtcNetEQ_DbGetSampleRate(&MCU_inst->codec_DB_inst,
- RTPpacket[i_k].payloadType);
- if ((fsCng != MCU_inst->fs) && (fsCng > 8000))
- {
- /*
- * We have received CNG with a different sample rate from what we are using
- * now (must be > 8000, since we may use only one CNG type (default) for all
- * frequencies). Flush buffer and signal new codec.
- */
- WebRtcNetEQ_PacketBufferFlush(&MCU_inst->PacketBuffer_inst);
- MCU_inst->new_codec = 1;
- MCU_inst->current_Codec = -1;
- }
- i_ok = WebRtcNetEQ_PacketBufferInsert(&MCU_inst->PacketBuffer_inst,
- &RTPpacket[i_k], &flushed);
- if (i_ok < 0)
- {
- return RECIN_CNG_ERROR;
- }
- MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 1;
-#else /* NETEQ_CNG_CODEC not defined */
- return RECIN_UNKNOWNPAYLOAD;
-#endif /* NETEQ_CNG_CODEC */
- }
- else
- {
- /* Reinitialize the splitting if the payload and/or the payload length has changed */
- curr_Codec = WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst,
- RTPpacket[i_k].payloadType);
- if (curr_Codec != MCU_inst->current_Codec)
- {
- if (curr_Codec < 0)
- {
- return RECIN_UNKNOWNPAYLOAD;
- }
- MCU_inst->current_Codec = curr_Codec;
- MCU_inst->current_Payload = RTPpacket[i_k].payloadType;
- i_ok = WebRtcNetEQ_DbGetSplitInfo(&MCU_inst->PayloadSplit_inst,
- (enum WebRtcNetEQDecoder) MCU_inst->current_Codec,
- RTPpacket[i_k].payloadLen);
- if (i_ok < 0)
- { /* error returned */
- return i_ok;
- }
- WebRtcNetEQ_PacketBufferFlush(&MCU_inst->PacketBuffer_inst);
- MCU_inst->new_codec = 1;
- }
-
- /* Parse the payload and insert it into the buffer */
- i_ok = WebRtcNetEQ_SplitAndInsertPayload(&RTPpacket[i_k],
- &MCU_inst->PacketBuffer_inst, &MCU_inst->PayloadSplit_inst, &flushed);
- if (i_ok < 0)
- {
- return i_ok;
- }
- if (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF != 0)
- {
- /* first normal packet after CNG or DTMF */
- MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = -1;
- }
- }
- /* Reset DSP timestamp etc. if packet buffer flushed */
- if (flushed)
- {
- MCU_inst->new_codec = 1;
- }
- }
-
- /*
- * Update Bandwidth Estimate
- * Only send the main payload to BWE
- */
- if ((curr_Codec = WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst,
- RTPpacket[0].payloadType)) >= 0)
- {
- codecPos = MCU_inst->codec_DB_inst.position[curr_Codec];
- if (MCU_inst->codec_DB_inst.funcUpdBWEst[codecPos] != NULL) /* codec has BWE function */
- {
- if (RTPpacket[0].starts_byte1) /* check for shifted byte alignment */
- {
- /* re-align to 16-bit alignment */
- for (i_k = 0; i_k < RTPpacket[0].payloadLen; i_k++)
- {
- WEBRTC_SPL_SET_BYTE(RTPpacket[0].payload,
- WEBRTC_SPL_GET_BYTE(RTPpacket[0].payload, i_k+1),
- i_k);
- }
- RTPpacket[0].starts_byte1 = 0;
- }
-
- MCU_inst->codec_DB_inst.funcUpdBWEst[codecPos](
- MCU_inst->codec_DB_inst.codec_state[codecPos],
- (G_CONST WebRtc_UWord16 *) RTPpacket[0].payload,
- (WebRtc_Word32) RTPpacket[0].payloadLen, RTPpacket[0].seqNumber,
- (WebRtc_UWord32) RTPpacket[0].timeStamp, (WebRtc_UWord32) uw32_timeRec);
- }
- }
-
- if (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF == 0)
- {
- /* Calculate the total speech length carried in each packet */
- temp_bufsize = MCU_inst->PacketBuffer_inst.numPacketsInBuffer - temp_bufsize;
- temp_bufsize *= MCU_inst->PacketBuffer_inst.packSizeSamples;
-
- if ((temp_bufsize > 0) && (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF
- == 0) && (temp_bufsize
- != MCU_inst->BufferStat_inst.Automode_inst.packetSpeechLenSamp))
- {
- /* Change the auto-mode parameters if packet length has changed */
- WebRtcNetEQ_SetPacketSpeechLen(&(MCU_inst->BufferStat_inst.Automode_inst),
- (WebRtc_Word16) temp_bufsize, MCU_inst->fs);
- }
-
- /* update statistics */
- if ((WebRtc_Word32) (RTPpacket[0].timeStamp - MCU_inst->timeStamp) >= 0
- && !MCU_inst->new_codec)
- {
- /*
- * Only update statistics if incoming packet is not older than last played out
- * packet, and if new codec flag is not set.
- */
- WebRtcNetEQ_UpdateIatStatistics(&MCU_inst->BufferStat_inst.Automode_inst,
- MCU_inst->PacketBuffer_inst.maxInsertPositions, RTPpacket[0].seqNumber,
- RTPpacket[0].timeStamp, MCU_inst->fs,
- WebRtcNetEQ_DbIsMDCodec((enum WebRtcNetEQDecoder) MCU_inst->current_Codec),
- (MCU_inst->NetEqPlayoutMode == kPlayoutStreaming));
- }
- }
- else if (MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF == -1)
- {
- /*
- * This is first "normal" packet after CNG or DTMF.
- * Reset packet time counter and measure time until next packet,
- * but don't update statistics.
- */
- MCU_inst->BufferStat_inst.Automode_inst.lastPackCNGorDTMF = 0;
- MCU_inst->BufferStat_inst.Automode_inst.packetIatCountSamp = 0;
- }
- return 0;
-
-}
-
-int WebRtcNetEQ_GetTimestampScaling(MCUInst_t *MCU_inst, int rtpPayloadType)
-{
- enum WebRtcNetEQDecoder codec;
- int codecNumber;
-
- codecNumber = WebRtcNetEQ_DbGetCodec(&MCU_inst->codec_DB_inst, rtpPayloadType);
- if (codecNumber < 0)
- {
- /* error */
- return codecNumber;
- }
-
- /* cast to enumerator */
- codec = (enum WebRtcNetEQDecoder) codecNumber;
-
- /*
- * The factor obtained below is the number with which the RTP timestamp must be
- * multiplied to get the true sample count.
- */
- switch (codec)
- {
- case kDecoderG722:
- {
- /* Use timestamp scaling with factor 2 (two output samples per RTP timestamp) */
- MCU_inst->scalingFactor = kTSscalingTwo;
- break;
- }
- case kDecoderAVT:
- case kDecoderCNG:
- {
- /* do not change the timestamp scaling settings */
- break;
- }
- default:
- {
- /* do not use timestamp scaling */
- MCU_inst->scalingFactor = kTSnoScaling;
- break;
- }
- }
- return 0;
-}
-
-WebRtc_UWord32 WebRtcNetEQ_ScaleTimestampExternalToInternal(const MCUInst_t *MCU_inst,
- WebRtc_UWord32 externalTS)
-{
- WebRtc_Word32 timestampDiff;
- WebRtc_UWord32 internalTS;
-
- /* difference between this and last incoming timestamp */
- timestampDiff = externalTS - MCU_inst->externalTS;
-
- switch (MCU_inst->scalingFactor)
- {
- case kTSscalingTwo:
- {
- /* multiply with 2 */
- timestampDiff = WEBRTC_SPL_LSHIFT_W32(timestampDiff, 1);
- break;
- }
- case kTSscalingTwoThirds:
- {
- /* multiply with 2/3 */
- timestampDiff = WEBRTC_SPL_LSHIFT_W32(timestampDiff, 1);
- timestampDiff = WebRtcSpl_DivW32W16(timestampDiff, 3);
- break;
- }
- case kTSscalingFourThirds:
- {
- /* multiply with 4/3 */
- timestampDiff = WEBRTC_SPL_LSHIFT_W32(timestampDiff, 2);
- timestampDiff = WebRtcSpl_DivW32W16(timestampDiff, 3);
- break;
- }
- default:
- {
- /* no scaling */
- }
- }
-
- /* add the scaled difference to last scaled timestamp and save ... */
- internalTS = MCU_inst->internalTS + timestampDiff;
-
- return internalTS;
-}
-
-WebRtc_UWord32 WebRtcNetEQ_ScaleTimestampInternalToExternal(const MCUInst_t *MCU_inst,
- WebRtc_UWord32 internalTS)
-{
- WebRtc_Word32 timestampDiff;
- WebRtc_UWord32 externalTS;
-
- /* difference between this and last incoming timestamp */
- timestampDiff = (WebRtc_Word32) internalTS - MCU_inst->internalTS;
-
- switch (MCU_inst->scalingFactor)
- {
- case kTSscalingTwo:
- {
- /* divide by 2 */
- timestampDiff = WEBRTC_SPL_RSHIFT_W32(timestampDiff, 1);
- break;
- }
- case kTSscalingTwoThirds:
- {
- /* multiply with 3/2 */
- timestampDiff = WEBRTC_SPL_MUL_32_16(timestampDiff, 3);
- timestampDiff = WEBRTC_SPL_RSHIFT_W32(timestampDiff, 1);
- break;
- }
- case kTSscalingFourThirds:
- {
- /* multiply with 3/4 */
- timestampDiff = WEBRTC_SPL_MUL_32_16(timestampDiff, 3);
- timestampDiff = WEBRTC_SPL_RSHIFT_W32(timestampDiff, 2);
- break;
- }
- default:
- {
- /* no scaling */
- }
- }
-
- /* add the scaled difference to last scaled timestamp and save ... */
- externalTS = MCU_inst->externalTS + timestampDiff;
-
- return externalTS;
-}
diff --git a/trunk/src/modules/audio_coding/neteq/recout.c b/trunk/src/modules/audio_coding/neteq/recout.c
deleted file mode 100644
index dab5540..0000000
--- a/trunk/src/modules/audio_coding/neteq/recout.c
+++ /dev/null
@@ -1,1429 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Implementation of RecOut function, which is the main function for the audio output
- * process. This function must be called (through the NetEQ API) once every 10 ms.
- */
-
-#include "dsp.h"
-
-#include <assert.h>
-#include <string.h> /* to define NULL */
-
-#include "signal_processing_library.h"
-
-#include "dsp_helpfunctions.h"
-#include "neteq_error_codes.h"
-#include "neteq_defines.h"
-#include "mcu_dsp_common.h"
-
-/* Audio types */
-#define TYPE_SPEECH 1
-#define TYPE_CNG 2
-
-#ifdef NETEQ_DELAY_LOGGING
-#include "delay_logging.h"
-#include <stdio.h>
-#pragma message("*******************************************************************")
-#pragma message("You have specified to use NETEQ_DELAY_LOGGING in the NetEQ library.")
-#pragma message("Make sure that your test application supports this.")
-#pragma message("*******************************************************************")
-#endif
-
-/* Scratch usage:
-
- Type Name size startpos endpos
- WebRtc_Word16 pw16_NetEqAlgorithm_buffer 600*fs/8000 0 600*fs/8000-1
- struct dspInfo 6 600*fs/8000 605*fs/8000
-
- func WebRtcNetEQ_Normal 40+495*fs/8000 0 39+495*fs/8000
- func WebRtcNetEQ_Merge 40+496*fs/8000 0 39+496*fs/8000
- func WebRtcNetEQ_Expand 40+370*fs/8000 126*fs/800 39+496*fs/8000
- func WebRtcNetEQ_Accelerate 210 240*fs/8000 209+240*fs/8000
- func WebRtcNetEQ_BGNUpdate 69 480*fs/8000 68+480*fs/8000
-
- Total: 605*fs/8000
- */
-
-#define SCRATCH_ALGORITHM_BUFFER 0
-#define SCRATCH_NETEQ_NORMAL 0
-#define SCRATCH_NETEQ_MERGE 0
-
-#if (defined(NETEQ_48KHZ_WIDEBAND))
-#define SCRATCH_DSP_INFO 3600
-#define SCRATCH_NETEQ_ACCELERATE 1440
-#define SCRATCH_NETEQ_BGN_UPDATE 2880
-#define SCRATCH_NETEQ_EXPAND 756
-#elif (defined(NETEQ_32KHZ_WIDEBAND))
-#define SCRATCH_DSP_INFO 2400
-#define SCRATCH_NETEQ_ACCELERATE 960
-#define SCRATCH_NETEQ_BGN_UPDATE 1920
-#define SCRATCH_NETEQ_EXPAND 504
-#elif (defined(NETEQ_WIDEBAND))
-#define SCRATCH_DSP_INFO 1200
-#define SCRATCH_NETEQ_ACCELERATE 480
-#define SCRATCH_NETEQ_BGN_UPDATE 960
-#define SCRATCH_NETEQ_EXPAND 252
-#else /* NB */
-#define SCRATCH_DSP_INFO 600
-#define SCRATCH_NETEQ_ACCELERATE 240
-#define SCRATCH_NETEQ_BGN_UPDATE 480
-#define SCRATCH_NETEQ_EXPAND 126
-#endif
-
-#if (defined(NETEQ_48KHZ_WIDEBAND))
-#define SIZE_SCRATCH_BUFFER 3636
-#elif (defined(NETEQ_32KHZ_WIDEBAND))
-#define SIZE_SCRATCH_BUFFER 2424
-#elif (defined(NETEQ_WIDEBAND))
-#define SIZE_SCRATCH_BUFFER 1212
-#else /* NB */
-#define SIZE_SCRATCH_BUFFER 606
-#endif
-
-#ifdef NETEQ_DELAY_LOGGING
-extern FILE *delay_fid2; /* file pointer to delay log file */
-extern WebRtc_UWord32 tot_received_packets;
-#endif
-
-
-int WebRtcNetEQ_RecOutInternal(DSPInst_t *inst, WebRtc_Word16 *pw16_outData,
- WebRtc_Word16 *pw16_len, WebRtc_Word16 BGNonly)
-{
-
- WebRtc_Word16 blockLen, payloadLen, len = 0, pos;
- WebRtc_Word16 w16_tmp1, w16_tmp2, w16_tmp3, DataEnough;
- WebRtc_Word16 *blockPtr;
- WebRtc_Word16 MD = 0;
-
- WebRtc_Word16 speechType = TYPE_SPEECH;
- WebRtc_UWord16 instr;
- WebRtc_UWord16 uw16_tmp;
-#ifdef SCRATCH
- char pw8_ScratchBuffer[((SIZE_SCRATCH_BUFFER + 1) * 2)];
- WebRtc_Word16 *pw16_scratchPtr = (WebRtc_Word16*) pw8_ScratchBuffer;
- WebRtc_Word16 pw16_decoded_buffer[NETEQ_MAX_FRAME_SIZE];
- WebRtc_Word16 *pw16_NetEqAlgorithm_buffer = pw16_scratchPtr
- + SCRATCH_ALGORITHM_BUFFER;
- DSP2MCU_info_t *dspInfo = (DSP2MCU_info_t*) (pw16_scratchPtr + SCRATCH_DSP_INFO);
-#else
- WebRtc_Word16 pw16_decoded_buffer[NETEQ_MAX_FRAME_SIZE];
- WebRtc_Word16 pw16_NetEqAlgorithm_buffer[NETEQ_MAX_OUTPUT_SIZE];
- DSP2MCU_info_t dspInfoStruct;
- DSP2MCU_info_t *dspInfo = &dspInfoStruct;
-#endif
- WebRtc_Word16 fs_mult;
- int borrowedSamples;
- int oldBorrowedSamples;
- int return_value = 0;
- WebRtc_Word16 lastModeBGNonly = (inst->w16_mode & MODE_BGN_ONLY) != 0; /* check BGN flag */
- void *mainInstBackup = inst->main_inst;
-
-#ifdef NETEQ_DELAY_LOGGING
- int temp_var;
-#endif
- WebRtc_Word16 dtmfValue = -1;
- WebRtc_Word16 dtmfVolume = -1;
- int playDtmf = 0;
-#ifdef NETEQ_ATEVENT_DECODE
- int dtmfSwitch = 0;
-#endif
-#ifdef NETEQ_STEREO
- MasterSlaveInfo *msInfo = inst->msInfo;
-#endif
- WebRtc_Word16 *sharedMem = pw16_NetEqAlgorithm_buffer; /* Reuse memory SHARED_MEM_SIZE size */
- inst->pw16_readAddress = sharedMem;
- inst->pw16_writeAddress = sharedMem;
-
- /* Get information about if there is one descriptor left */
- if (inst->codec_ptr_inst.funcGetMDinfo != NULL)
- {
- MD = inst->codec_ptr_inst.funcGetMDinfo(inst->codec_ptr_inst.codec_state);
- if (MD > 0)
- MD = 1;
- else
- MD = 0;
- }
-
-#ifdef NETEQ_STEREO
- if ((msInfo->msMode == NETEQ_SLAVE) && (inst->codec_ptr_inst.funcDecode != NULL))
- {
- /*
- * Valid function pointers indicate that we have decoded something,
- * and that the timestamp information is correct.
- */
-
- /* Get the information from master to correct synchronization */
- WebRtc_UWord32 currentMasterTimestamp;
- WebRtc_UWord32 currentSlaveTimestamp;
-
- currentMasterTimestamp = msInfo->endTimestamp - msInfo->samplesLeftWithOverlap;
- currentSlaveTimestamp = inst->endTimestamp - (inst->endPosition - inst->curPosition);
-
- if (currentSlaveTimestamp < currentMasterTimestamp)
- {
- /* brute-force discard a number of samples to catch up */
- inst->curPosition += currentMasterTimestamp - currentSlaveTimestamp;
-
- /* make sure we have at least "overlap" samples left */
- inst->curPosition = WEBRTC_SPL_MIN(inst->curPosition,
- inst->endPosition - inst->ExpandInst.w16_overlap);
- }
- else if (currentSlaveTimestamp > currentMasterTimestamp)
- {
- /* back off current position to slow down */
- inst->curPosition -= currentSlaveTimestamp - currentMasterTimestamp;
-
- /* make sure we do not end up outside the speech history */
- inst->curPosition = WEBRTC_SPL_MAX(inst->curPosition, 0);
- }
- }
-#endif
-
- /* Write status data to shared memory */
- dspInfo->playedOutTS = inst->endTimestamp;
- dspInfo->samplesLeft = inst->endPosition - inst->curPosition
- - inst->ExpandInst.w16_overlap;
- dspInfo->MD = MD;
- dspInfo->lastMode = inst->w16_mode;
- dspInfo->frameLen = inst->w16_frameLen;
-
- /* Force update of codec if codec function is NULL */
- if (inst->codec_ptr_inst.funcDecode == NULL)
- {
- dspInfo->lastMode |= MODE_AWAITING_CODEC_PTR;
- }
-
-#ifdef NETEQ_STEREO
- if (msInfo->msMode == NETEQ_SLAVE && (msInfo->extraInfo == DTMF_OVERDUB
- || msInfo->extraInfo == DTMF_ONLY))
- {
- /* Signal that the master instance generated DTMF tones */
- dspInfo->lastMode |= MODE_MASTER_DTMF_SIGNAL;
- }
-
- if (msInfo->msMode != NETEQ_MONO)
- {
- /* We are using stereo mode; signal this to MCU side */
- dspInfo->lastMode |= MODE_USING_STEREO;
- }
-#endif
-
- WEBRTC_SPL_MEMCPY_W8(inst->pw16_writeAddress,dspInfo,sizeof(DSP2MCU_info_t));
-
- /* Signal MCU with "interrupt" call to main inst*/
-#ifdef NETEQ_STEREO
- assert(msInfo != NULL);
- if (msInfo->msMode == NETEQ_MASTER)
- {
- /* clear info to slave */
- WebRtcSpl_MemSetW16((WebRtc_Word16 *) msInfo, 0,
- sizeof(MasterSlaveInfo) / sizeof(WebRtc_Word16));
- /* re-set mode */
- msInfo->msMode = NETEQ_MASTER;
-
- /* Store some information to slave */
- msInfo->endTimestamp = inst->endTimestamp;
- msInfo->samplesLeftWithOverlap = inst->endPosition - inst->curPosition;
- }
-#endif
-
- /*
- * This call will trigger the MCU side to make a decision based on buffer contents and
- * decision history. Instructions, encoded data and function pointers will be written
- * to the shared memory.
- */
- return_value = WebRtcNetEQ_DSP2MCUinterrupt((MainInst_t *) inst->main_inst, sharedMem);
-
- /* Read MCU data and instructions */
- instr = (WebRtc_UWord16) (inst->pw16_readAddress[0] & 0xf000);
-
-#ifdef NETEQ_STEREO
- if (msInfo->msMode == NETEQ_MASTER)
- {
- msInfo->instruction = instr;
- }
- else if (msInfo->msMode == NETEQ_SLAVE)
- {
- /* Nothing to do */
- }
-#endif
-
- /* check for error returned from MCU side, if so, return error */
- if (return_value < 0)
- {
- inst->w16_mode = MODE_ERROR;
- dspInfo->lastMode = MODE_ERROR;
- return return_value;
- }
-
- blockPtr = &((inst->pw16_readAddress)[3]);
-
- /* Check for DTMF payload flag */
- if ((inst->pw16_readAddress[0] & DSP_DTMF_PAYLOAD) != 0)
- {
- playDtmf = 1;
- dtmfValue = blockPtr[1];
- dtmfVolume = blockPtr[2];
- blockPtr += 3;
-
-#ifdef NETEQ_STEREO
- if (msInfo->msMode == NETEQ_MASTER)
- {
- /* signal to slave that master is using DTMF */
- msInfo->extraInfo = DTMF_OVERDUB;
- }
-#endif
- }
-
- blockLen = (((*blockPtr) & DSP_CODEC_MASK_RED_FLAG) + 1) >> 1; /* In # of WebRtc_Word16 */
- payloadLen = ((*blockPtr) & DSP_CODEC_MASK_RED_FLAG);
- blockPtr++;
-
- /* Do we have to change our decoder? */
- if ((inst->pw16_readAddress[0] & 0x0f00) == DSP_CODEC_NEW_CODEC)
- {
- WEBRTC_SPL_MEMCPY_W16(&inst->codec_ptr_inst,blockPtr,(payloadLen+1)>>1);
- if (inst->codec_ptr_inst.codec_fs != 0)
- {
- return_value = WebRtcNetEQ_DSPInit(inst, inst->codec_ptr_inst.codec_fs);
- if (return_value != 0)
- { /* error returned */
- instr = DSP_INSTR_FADE_TO_BGN; /* emergency instruction */
- }
-#ifdef NETEQ_DELAY_LOGGING
- temp_var = NETEQ_DELAY_LOGGING_SIGNAL_CHANGE_FS;
- fwrite(&temp_var, sizeof(int), 1, delay_fid2);
- fwrite(&inst->fs, sizeof(WebRtc_UWord16), 1, delay_fid2);
-#endif
- }
-
- /* Copy it again since the init destroys this part */
-
- WEBRTC_SPL_MEMCPY_W16(&inst->codec_ptr_inst,blockPtr,(payloadLen+1)>>1);
- inst->endTimestamp = inst->codec_ptr_inst.timeStamp;
- inst->videoSyncTimestamp = inst->codec_ptr_inst.timeStamp;
- blockPtr += blockLen;
- blockLen = (((*blockPtr) & DSP_CODEC_MASK_RED_FLAG) + 1) >> 1;
- payloadLen = ((*blockPtr) & DSP_CODEC_MASK_RED_FLAG);
- blockPtr++;
- if (inst->codec_ptr_inst.funcDecodeInit != NULL)
- {
- inst->codec_ptr_inst.funcDecodeInit(inst->codec_ptr_inst.codec_state);
- }
-
-#ifdef NETEQ_CNG_CODEC
-
- /* Also update the CNG state as this might be uninitialized */
-
- WEBRTC_SPL_MEMCPY_W16(&inst->CNG_Codec_inst,blockPtr,(payloadLen+1)>>1);
- blockPtr += blockLen;
- blockLen = (((*blockPtr) & DSP_CODEC_MASK_RED_FLAG) + 1) >> 1;
- payloadLen = ((*blockPtr) & DSP_CODEC_MASK_RED_FLAG);
- blockPtr++;
- if (inst->CNG_Codec_inst != NULL)
- {
- WebRtcCng_InitDec(inst->CNG_Codec_inst);
- }
-#endif
- }
- else if ((inst->pw16_readAddress[0] & 0x0f00) == DSP_CODEC_RESET)
- {
- /* Reset the current codec (but not DSP struct) */
- if (inst->codec_ptr_inst.funcDecodeInit != NULL)
- {
- inst->codec_ptr_inst.funcDecodeInit(inst->codec_ptr_inst.codec_state);
- }
-
-#ifdef NETEQ_CNG_CODEC
- /* And reset CNG */
- if (inst->CNG_Codec_inst != NULL)
- {
- WebRtcCng_InitDec(inst->CNG_Codec_inst);
- }
-#endif /*NETEQ_CNG_CODEC*/
- }
-
- fs_mult = WebRtcNetEQ_CalcFsMult(inst->fs);
-
- /* Add late packet? */
- if ((inst->pw16_readAddress[0] & 0x0f00) == DSP_CODEC_ADD_LATE_PKT)
- {
- if (inst->codec_ptr_inst.funcAddLatePkt != NULL)
- {
- /* Only do this if the codec has support for Add Late Pkt */
- inst->codec_ptr_inst.funcAddLatePkt(inst->codec_ptr_inst.codec_state, blockPtr,
- payloadLen);
- }
- blockPtr += blockLen;
- blockLen = (((*blockPtr) & DSP_CODEC_MASK_RED_FLAG) + 1) >> 1; /* In # of Word16 */
- payloadLen = ((*blockPtr) & DSP_CODEC_MASK_RED_FLAG);
- blockPtr++;
- }
-
- /* Do we have to decode data? */
- if ((instr == DSP_INSTR_NORMAL) || (instr == DSP_INSTR_ACCELERATE) || (instr
- == DSP_INSTR_MERGE) || (instr == DSP_INSTR_PREEMPTIVE_EXPAND))
- {
- /* Do we need to update codec-internal PLC state? */
- if ((instr == DSP_INSTR_MERGE) && (inst->codec_ptr_inst.funcDecodePLC != NULL))
- {
- len = 0;
- len = inst->codec_ptr_inst.funcDecodePLC(inst->codec_ptr_inst.codec_state,
- &pw16_decoded_buffer[len], 1);
- }
- len = 0;
-
- /* Do decoding */
- while ((blockLen > 0) && (len < (240 * fs_mult))) /* Guard somewhat against overflow */
- {
- if (inst->codec_ptr_inst.funcDecode != NULL)
- {
- WebRtc_Word16 dec_Len;
- if (!BGNonly)
- {
- /* Do decoding as normal
- *
- * blockPtr is pointing to payload, at this point,
- * the most significant bit of *(blockPtr - 1) is a flag if set to 1
- * indicates that the following payload is the redundant payload.
- */
- if (((*(blockPtr - 1) & DSP_CODEC_RED_FLAG) != 0)
- && (inst->codec_ptr_inst.funcDecodeRCU != NULL))
- {
- dec_Len = inst->codec_ptr_inst.funcDecodeRCU(
- inst->codec_ptr_inst.codec_state, blockPtr, payloadLen,
- &pw16_decoded_buffer[len], &speechType);
- }
- else
- {
- dec_Len = inst->codec_ptr_inst.funcDecode(
- inst->codec_ptr_inst.codec_state, blockPtr, payloadLen,
- &pw16_decoded_buffer[len], &speechType);
- }
- }
- else
- {
- /*
- * Background noise mode: don't decode, just produce the same length BGN.
- * Don't call Expand for BGN here, since Expand uses the memory where the
- * bitstreams are stored (sharemem).
- */
- dec_Len = inst->w16_frameLen;
- }
-
- if (dec_Len > 0)
- {
- len += dec_Len;
- /* Update frameLen */
- inst->w16_frameLen = dec_Len;
- }
- else if (dec_Len < 0)
- {
- /* Error */
- len = -1;
- break;
- }
- /*
- * Sanity check (although we might still write outside memory when this
- * happens...)
- */
- if (len > NETEQ_MAX_FRAME_SIZE)
- {
- WebRtcSpl_MemSetW16(pw16_outData, 0, inst->timestampsPerCall);
- *pw16_len = inst->timestampsPerCall;
- inst->w16_mode = MODE_ERROR;
- dspInfo->lastMode = MODE_ERROR;
- return RECOUT_ERROR_DECODED_TOO_MUCH;
- }
-
- /* Verify that instance was not corrupted by decoder */
- if (mainInstBackup != inst->main_inst)
- {
- /* Instance is corrupt */
- return CORRUPT_INSTANCE;
- }
-
- }
- blockPtr += blockLen;
- blockLen = (((*blockPtr) & DSP_CODEC_MASK_RED_FLAG) + 1) >> 1; /* In # of Word16 */
- payloadLen = ((*blockPtr) & DSP_CODEC_MASK_RED_FLAG);
- blockPtr++;
- }
-
- if (len < 0)
- {
- len = 0;
- inst->endTimestamp += inst->w16_frameLen; /* advance one frame */
- if (inst->codec_ptr_inst.funcGetErrorCode != NULL)
- {
- return_value = -inst->codec_ptr_inst.funcGetErrorCode(
- inst->codec_ptr_inst.codec_state);
- }
- else
- {
- return_value = RECOUT_ERROR_DECODING;
- }
- instr = DSP_INSTR_FADE_TO_BGN;
- }
- if (speechType != TYPE_CNG)
- {
- /*
- * Don't increment timestamp if codec returned CNG speech type
- * since in this case, the MCU side will increment the CNGplayedTS counter.
- */
- inst->endTimestamp += len;
- }
- }
- else if (instr == DSP_INSTR_NORMAL_ONE_DESC)
- {
- if (inst->codec_ptr_inst.funcDecode != NULL)
- {
- len = inst->codec_ptr_inst.funcDecode(inst->codec_ptr_inst.codec_state, NULL, 0,
- pw16_decoded_buffer, &speechType);
-#ifdef NETEQ_DELAY_LOGGING
- temp_var = NETEQ_DELAY_LOGGING_SIGNAL_DECODE_ONE_DESC;
- fwrite(&temp_var, sizeof(int), 1, delay_fid2);
- fwrite(&inst->endTimestamp, sizeof(WebRtc_UWord32), 1, delay_fid2);
- fwrite(&dspInfo->samplesLeft, sizeof(WebRtc_UWord16), 1, delay_fid2);
- tot_received_packets++;
-#endif
- }
- if (speechType != TYPE_CNG)
- {
- /*
- * Don't increment timestamp if codec returned CNG speech type
- * since in this case, the MCU side will increment the CNGplayedTS counter.
- */
- inst->endTimestamp += len;
- }
-
- /* Verify that instance was not corrupted by decoder */
- if (mainInstBackup != inst->main_inst)
- {
- /* Instance is corrupt */
- return CORRUPT_INSTANCE;
- }
-
- if (len <= 0)
- {
- len = 0;
- if (inst->codec_ptr_inst.funcGetErrorCode != NULL)
- {
- return_value = -inst->codec_ptr_inst.funcGetErrorCode(
- inst->codec_ptr_inst.codec_state);
- }
- else
- {
- return_value = RECOUT_ERROR_DECODING;
- }
- if ((inst->codec_ptr_inst.funcDecodeInit != NULL)
- && (inst->codec_ptr_inst.codec_state != NULL))
- {
- /* Reinitialize codec state as something is obviously wrong */
- inst->codec_ptr_inst.funcDecodeInit(inst->codec_ptr_inst.codec_state);
- }
- inst->endTimestamp += inst->w16_frameLen; /* advance one frame */
- instr = DSP_INSTR_FADE_TO_BGN;
- }
- }
-
- if (len == 0 && lastModeBGNonly) /* no new data */
- {
- BGNonly = 1; /* force BGN this time too */
- }
-
-#ifdef NETEQ_VAD
- if ((speechType == TYPE_CNG) /* decoder responded with codec-internal CNG */
- || ((instr == DSP_INSTR_DO_RFC3389CNG) && (blockLen > 0)) /* ... or, SID frame */
- || (inst->fs > 16000)) /* ... or, if not NB or WB */
- {
- /* disable post-decode VAD upon first sign of send-side DTX/VAD active, or if SWB */
- inst->VADInst.VADEnabled = 0;
- inst->VADInst.VADDecision = 1; /* set to always active, just to be on the safe side */
- inst->VADInst.SIDintervalCounter = 0; /* reset SID interval counter */
- }
- else if (!inst->VADInst.VADEnabled) /* VAD disabled and no SID/CNG data observed this time */
- {
- inst->VADInst.SIDintervalCounter++; /* increase counter */
- }
-
- /* check for re-enabling the VAD */
- if (inst->VADInst.SIDintervalCounter >= POST_DECODE_VAD_AUTO_ENABLE)
- {
- /*
- * It's been a while since the last CNG/SID frame was observed => re-enable VAD.
- * (Do not care to look for a VAD instance, since this is done inside the init
- * function)
- */
- WebRtcNetEQ_InitVAD(&inst->VADInst, inst->fs);
- }
-
- if (len > 0 /* if we decoded any data */
- && inst->VADInst.VADEnabled /* and VAD enabled */
- && inst->fs <= 16000) /* can only do VAD for NB and WB */
- {
- int VADframeSize; /* VAD frame size in ms */
- int VADSamplePtr = 0;
-
- inst->VADInst.VADDecision = 0;
-
- if (inst->VADInst.VADFunction != NULL) /* make sure that VAD function is provided */
- {
- /* divide the data into groups, as large as possible */
- for (VADframeSize = 30; VADframeSize >= 10; VADframeSize -= 10)
- {
- /* loop through 30, 20, 10 */
-
- while (inst->VADInst.VADDecision == 0
- && len - VADSamplePtr >= VADframeSize * fs_mult * 8)
- {
- /*
- * Only continue until first active speech found, and as long as there is
- * one VADframeSize left.
- */
-
- /* call VAD with new decoded data */
- inst->VADInst.VADDecision |= inst->VADInst.VADFunction(
- inst->VADInst.VADState, (WebRtc_Word16) inst->fs,
- (WebRtc_Word16 *) &pw16_decoded_buffer[VADSamplePtr],
- (WebRtc_Word16) (VADframeSize * fs_mult * 8));
-
- VADSamplePtr += VADframeSize * fs_mult * 8; /* increment sample counter */
- }
- }
- }
- else
- { /* VAD function is NULL */
- inst->VADInst.VADDecision = 1; /* set decision to active */
- inst->VADInst.VADEnabled = 0; /* disable VAD since we have no VAD function */
- }
-
- }
-#endif /* NETEQ_VAD */
-
- /* Adjust timestamp if needed */
- uw16_tmp = (WebRtc_UWord16) inst->pw16_readAddress[1];
- inst->endTimestamp += (((WebRtc_UWord32) uw16_tmp) << 16);
- uw16_tmp = (WebRtc_UWord16) inst->pw16_readAddress[2];
- inst->endTimestamp += uw16_tmp;
-
- if (BGNonly && len > 0)
- {
- /*
- * If BGN mode, we did not produce any data at decoding.
- * Do it now instead.
- */
-
- WebRtcNetEQ_GenerateBGN(inst,
-#ifdef SCRATCH
- pw16_scratchPtr + SCRATCH_NETEQ_EXPAND,
-#endif
- pw16_decoded_buffer, len);
- }
-
- /* Switch on the instruction received from the MCU side. */
- switch (instr)
- {
- case DSP_INSTR_NORMAL:
-
- /* Allow for signal processing to apply gain-back etc */
- WebRtcNetEQ_Normal(inst,
-#ifdef SCRATCH
- pw16_scratchPtr + SCRATCH_NETEQ_NORMAL,
-#endif
- pw16_decoded_buffer, len, pw16_NetEqAlgorithm_buffer, &len);
-
- /* If last packet was decoded as a inband CNG set mode to CNG instead */
- if ((speechType == TYPE_CNG) || ((inst->w16_mode == MODE_CODEC_INTERNAL_CNG)
- && (len == 0)))
- {
- inst->w16_mode = MODE_CODEC_INTERNAL_CNG;
- }
-
-#ifdef NETEQ_ATEVENT_DECODE
- if (playDtmf == 0)
- {
- inst->DTMFInst.reinit = 1;
- }
-#endif
- break;
- case DSP_INSTR_NORMAL_ONE_DESC:
-
- /* Allow for signal processing to apply gain-back etc */
- WebRtcNetEQ_Normal(inst,
-#ifdef SCRATCH
- pw16_scratchPtr + SCRATCH_NETEQ_NORMAL,
-#endif
- pw16_decoded_buffer, len, pw16_NetEqAlgorithm_buffer, &len);
-#ifdef NETEQ_ATEVENT_DECODE
- if (playDtmf == 0)
- {
- inst->DTMFInst.reinit = 1;
- }
-#endif
- inst->w16_mode = MODE_ONE_DESCRIPTOR;
- break;
- case DSP_INSTR_MERGE:
-#ifdef NETEQ_DELAY_LOGGING
- temp_var = NETEQ_DELAY_LOGGING_SIGNAL_MERGE_INFO;
- fwrite(&temp_var, sizeof(int), 1, delay_fid2);
- temp_var = -len;
-#endif
- /* Call Merge with history*/
- return_value = WebRtcNetEQ_Merge(inst,
-#ifdef SCRATCH
- pw16_scratchPtr + SCRATCH_NETEQ_MERGE,
-#endif
- pw16_decoded_buffer, len, pw16_NetEqAlgorithm_buffer, &len);
-
- if (return_value < 0)
- {
- /* error */
- return return_value;
- }
-
-#ifdef NETEQ_DELAY_LOGGING
- temp_var += len;
- fwrite(&temp_var, sizeof(int), 1, delay_fid2);
-#endif
- /* If last packet was decoded as a inband CNG set mode to CNG instead */
- if (speechType == TYPE_CNG) inst->w16_mode = MODE_CODEC_INTERNAL_CNG;
-#ifdef NETEQ_ATEVENT_DECODE
- if (playDtmf == 0)
- {
- inst->DTMFInst.reinit = 1;
- }
-#endif
- break;
-
- case DSP_INSTR_EXPAND:
- len = 0;
- pos = 0;
- while ((inst->endPosition - inst->curPosition - inst->ExpandInst.w16_overlap + pos)
- < (inst->timestampsPerCall))
- {
- return_value = WebRtcNetEQ_Expand(inst,
-#ifdef SCRATCH
- pw16_scratchPtr + SCRATCH_NETEQ_EXPAND,
-#endif
- pw16_NetEqAlgorithm_buffer, &len, BGNonly);
- if (return_value < 0)
- {
- /* error */
- return return_value;
- }
-
- /*
- * Update buffer, but only end part (otherwise expand state is destroyed
- * since it reuses speechBuffer[] memory
- */
-
- WEBRTC_SPL_MEMMOVE_W16(inst->pw16_speechHistory,
- inst->pw16_speechHistory + len,
- (inst->w16_speechHistoryLen-len));
- WEBRTC_SPL_MEMCPY_W16(&inst->pw16_speechHistory[inst->w16_speechHistoryLen-len],
- pw16_NetEqAlgorithm_buffer, len);
-
- inst->curPosition -= len;
-
- /* Update variables for VQmon */
- inst->w16_concealedTS += len;
-#ifdef NETEQ_DELAY_LOGGING
- temp_var = NETEQ_DELAY_LOGGING_SIGNAL_EXPAND_INFO;
- fwrite(&temp_var, sizeof(int), 1, delay_fid2);
- temp_var = len;
- fwrite(&temp_var, sizeof(int), 1, delay_fid2);
-#endif
- len = 0; /* already written the data, so do not write it again further down. */
- }
-#ifdef NETEQ_ATEVENT_DECODE
- if (playDtmf == 0)
- {
- inst->DTMFInst.reinit = 1;
- }
-#endif
- break;
-
- case DSP_INSTR_ACCELERATE:
- if (len < 3 * 80 * fs_mult)
- {
- /* We need to move data from the speechBuffer[] in order to get 30 ms */
- borrowedSamples = 3 * 80 * fs_mult - len;
-
- WEBRTC_SPL_MEMMOVE_W16(&pw16_decoded_buffer[borrowedSamples],
- pw16_decoded_buffer, len);
- WEBRTC_SPL_MEMCPY_W16(pw16_decoded_buffer,
- &(inst->speechBuffer[inst->endPosition-borrowedSamples]),
- borrowedSamples);
-
- return_value = WebRtcNetEQ_Accelerate(inst,
-#ifdef SCRATCH
- pw16_scratchPtr + SCRATCH_NETEQ_ACCELERATE,
-#endif
- pw16_decoded_buffer, 3 * inst->timestampsPerCall,
- pw16_NetEqAlgorithm_buffer, &len, BGNonly);
-
- if (return_value < 0)
- {
- /* error */
- return return_value;
- }
-
- /* Copy back samples to the buffer */
- if (len < borrowedSamples)
- {
- /*
- * This destroys the beginning of the buffer, but will not cause any
- * problems
- */
-
- WEBRTC_SPL_MEMCPY_W16(&inst->speechBuffer[inst->endPosition-borrowedSamples],
- pw16_NetEqAlgorithm_buffer, len);
- WEBRTC_SPL_MEMMOVE_W16(&inst->speechBuffer[borrowedSamples-len],
- inst->speechBuffer,
- (inst->endPosition-(borrowedSamples-len)));
-
- inst->curPosition += (borrowedSamples - len);
-#ifdef NETEQ_DELAY_LOGGING
- temp_var = NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO;
- fwrite(&temp_var, sizeof(int), 1, delay_fid2);
- temp_var = 3 * inst->timestampsPerCall - len;
- fwrite(&temp_var, sizeof(int), 1, delay_fid2);
-#endif
- len = 0;
- }
- else
- {
- WEBRTC_SPL_MEMCPY_W16(&inst->speechBuffer[inst->endPosition-borrowedSamples],
- pw16_NetEqAlgorithm_buffer, borrowedSamples);
- WEBRTC_SPL_MEMMOVE_W16(pw16_NetEqAlgorithm_buffer,
- &pw16_NetEqAlgorithm_buffer[borrowedSamples],
- (len-borrowedSamples));
-#ifdef NETEQ_DELAY_LOGGING
- temp_var = NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO;
- fwrite(&temp_var, sizeof(int), 1, delay_fid2);
- temp_var = 3 * inst->timestampsPerCall - len;
- fwrite(&temp_var, sizeof(int), 1, delay_fid2);
-#endif
- len = len - borrowedSamples;
- }
-
- }
- else
- {
-#ifdef NETEQ_DELAY_LOGGING
- temp_var = NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO;
- fwrite(&temp_var, sizeof(int), 1, delay_fid2);
- temp_var = len;
-#endif
- return_value = WebRtcNetEQ_Accelerate(inst,
-#ifdef SCRATCH
- pw16_scratchPtr + SCRATCH_NETEQ_ACCELERATE,
-#endif
- pw16_decoded_buffer, len, pw16_NetEqAlgorithm_buffer, &len, BGNonly);
-
- if (return_value < 0)
- {
- /* error */
- return return_value;
- }
-
-#ifdef NETEQ_DELAY_LOGGING
- temp_var -= len;
- fwrite(&temp_var, sizeof(int), 1, delay_fid2);
-#endif
- }
- /* If last packet was decoded as a inband CNG set mode to CNG instead */
- if (speechType == TYPE_CNG) inst->w16_mode = MODE_CODEC_INTERNAL_CNG;
-#ifdef NETEQ_ATEVENT_DECODE
- if (playDtmf == 0)
- {
- inst->DTMFInst.reinit = 1;
- }
-#endif
- break;
-
- case DSP_INSTR_DO_RFC3389CNG:
-#ifdef NETEQ_CNG_CODEC
- if (blockLen > 0)
- {
- if (WebRtcCng_UpdateSid(inst->CNG_Codec_inst, (WebRtc_UWord8*) blockPtr,
- payloadLen) < 0)
- {
- /* error returned from CNG function */
- return_value = -WebRtcCng_GetErrorCodeDec(inst->CNG_Codec_inst);
- len = inst->timestampsPerCall;
- WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0, len);
- break;
- }
- }
-
- if (BGNonly)
- {
- /* Get data from BGN function instead of CNG */
- len = WebRtcNetEQ_GenerateBGN(inst,
-#ifdef SCRATCH
- pw16_scratchPtr + SCRATCH_NETEQ_EXPAND,
-#endif
- pw16_NetEqAlgorithm_buffer, inst->timestampsPerCall);
- if (len != inst->timestampsPerCall)
- {
- /* this is not good, treat this as an error */
- return_value = -1;
- }
- }
- else
- {
- return_value = WebRtcNetEQ_Cng(inst, pw16_NetEqAlgorithm_buffer,
- inst->timestampsPerCall);
- }
- len = inst->timestampsPerCall;
- inst->ExpandInst.w16_consecExp = 0;
- inst->w16_mode = MODE_RFC3389CNG;
-#ifdef NETEQ_ATEVENT_DECODE
- if (playDtmf == 0)
- {
- inst->DTMFInst.reinit = 1;
- }
-#endif
-
- if (return_value < 0)
- {
- /* error returned */
- WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0, len);
- }
-
- break;
-#else
- return FAULTY_INSTRUCTION;
-#endif
- case DSP_INSTR_DO_CODEC_INTERNAL_CNG:
- /*
- * This represents the case when there is no transmission and the decoder should
- * do internal CNG.
- */
- len = 0;
- if (inst->codec_ptr_inst.funcDecode != NULL && !BGNonly)
- {
- len = inst->codec_ptr_inst.funcDecode(inst->codec_ptr_inst.codec_state,
- blockPtr, 0, pw16_decoded_buffer, &speechType);
- }
- else
- {
- /* get BGN data */
- len = WebRtcNetEQ_GenerateBGN(inst,
-#ifdef SCRATCH
- pw16_scratchPtr + SCRATCH_NETEQ_EXPAND,
-#endif
- pw16_decoded_buffer, inst->timestampsPerCall);
- }
- WebRtcNetEQ_Normal(inst,
-#ifdef SCRATCH
- pw16_scratchPtr + SCRATCH_NETEQ_NORMAL,
-#endif
- pw16_decoded_buffer, len, pw16_NetEqAlgorithm_buffer, &len);
- inst->w16_mode = MODE_CODEC_INTERNAL_CNG;
- inst->ExpandInst.w16_consecExp = 0;
- break;
-
- case DSP_INSTR_DTMF_GENERATE:
-#ifdef NETEQ_ATEVENT_DECODE
- dtmfSwitch = 0;
- if ((inst->w16_mode != MODE_DTMF) && (inst->DTMFInst.reinit == 0))
- {
- /* Special case; see below.
- * We must catch this before calling DTMFGenerate,
- * since reinit is set to 0 in that call.
- */
- dtmfSwitch = 1;
- }
-
- len = WebRtcNetEQ_DTMFGenerate(&inst->DTMFInst, dtmfValue, dtmfVolume,
- pw16_NetEqAlgorithm_buffer, inst->fs, -1);
- if (len < 0)
- {
- /* error occurred */
- return_value = len;
- len = inst->timestampsPerCall;
- WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0, len);
- }
-
- if (dtmfSwitch == 1)
- {
- /*
- * This is the special case where the previous operation was DTMF overdub.
- * but the current instruction is "regular" DTMF. We must make sure that the
- * DTMF does not have any discontinuities. The first DTMF sample that we
- * generate now must be played out immediately, wherefore it must be copied to
- * the speech buffer.
- */
-
- /*
- * Generate extra DTMF data to fill the space between
- * curPosition and endPosition
- */
- WebRtc_Word16 tempLen;
-
- tempLen = WebRtcNetEQ_DTMFGenerate(&inst->DTMFInst, dtmfValue, dtmfVolume,
- &pw16_NetEqAlgorithm_buffer[len], inst->fs,
- inst->endPosition - inst->curPosition);
- if (tempLen < 0)
- {
- /* error occurred */
- return_value = tempLen;
- len = inst->endPosition - inst->curPosition;
- WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0,
- inst->endPosition - inst->curPosition);
- }
-
- /* Add to total length */
- len += tempLen;
-
- /* Overwrite the "future" part of the speech buffer with the new DTMF data */
-
- WEBRTC_SPL_MEMCPY_W16(&inst->speechBuffer[inst->curPosition],
- pw16_NetEqAlgorithm_buffer,
- inst->endPosition - inst->curPosition);
-
- /* Shuffle the remaining data to the beginning of algorithm buffer */
- len -= (inst->endPosition - inst->curPosition);
- WEBRTC_SPL_MEMMOVE_W16(pw16_NetEqAlgorithm_buffer,
- &pw16_NetEqAlgorithm_buffer[inst->endPosition - inst->curPosition],
- len);
- }
-
- inst->endTimestamp += inst->timestampsPerCall;
- inst->DTMFInst.reinit = 0;
- inst->ExpandInst.w16_consecExp = 0;
- inst->w16_mode = MODE_DTMF;
- BGNonly = 0; /* override BGN only and let DTMF through */
-
- playDtmf = 0; /* set to zero because the DTMF is already in the Algorithm buffer */
- /*
- * If playDtmf is 1, an extra DTMF vector will be generated and overdubbed
- * on the output.
- */
-
-#ifdef NETEQ_STEREO
- if (msInfo->msMode == NETEQ_MASTER)
- {
- /* signal to slave that master is using DTMF only */
- msInfo->extraInfo = DTMF_ONLY;
- }
-#endif
-
- break;
-#else
- inst->w16_mode = MODE_ERROR;
- dspInfo->lastMode = MODE_ERROR;
- return FAULTY_INSTRUCTION;
-#endif
-
- case DSP_INSTR_DO_ALTERNATIVE_PLC:
- if (inst->codec_ptr_inst.funcDecodePLC != 0)
- {
- len = inst->codec_ptr_inst.funcDecodePLC(inst->codec_ptr_inst.codec_state,
- pw16_NetEqAlgorithm_buffer, 1);
- }
- else
- {
- len = inst->timestampsPerCall;
- /* ZeroStuffing... */
- WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0, len);
- }
- inst->ExpandInst.w16_consecExp = 0;
- break;
- case DSP_INSTR_DO_ALTERNATIVE_PLC_INC_TS:
- if (inst->codec_ptr_inst.funcDecodePLC != 0)
- {
- len = inst->codec_ptr_inst.funcDecodePLC(inst->codec_ptr_inst.codec_state,
- pw16_NetEqAlgorithm_buffer, 1);
- }
- else
- {
- len = inst->timestampsPerCall;
- /* ZeroStuffing... */
- WebRtcSpl_MemSetW16(pw16_NetEqAlgorithm_buffer, 0, len);
- }
- inst->ExpandInst.w16_consecExp = 0;
- inst->endTimestamp += len;
- break;
- case DSP_INSTR_DO_AUDIO_REPETITION:
- len = inst->timestampsPerCall;
- /* copy->paste... */
- WEBRTC_SPL_MEMCPY_W16(pw16_NetEqAlgorithm_buffer,
- &inst->speechBuffer[inst->endPosition-len], len);
- inst->ExpandInst.w16_consecExp = 0;
- break;
- case DSP_INSTR_DO_AUDIO_REPETITION_INC_TS:
- len = inst->timestampsPerCall;
- /* copy->paste... */
- WEBRTC_SPL_MEMCPY_W16(pw16_NetEqAlgorithm_buffer,
- &inst->speechBuffer[inst->endPosition-len], len);
- inst->ExpandInst.w16_consecExp = 0;
- inst->endTimestamp += len;
- break;
-
- case DSP_INSTR_PREEMPTIVE_EXPAND:
- if (len < 3 * inst->timestampsPerCall)
- {
- /* borrow samples from sync buffer if necessary */
- borrowedSamples = 3 * inst->timestampsPerCall - len; /* borrow this many samples */
- /* calculate how many of these are already played out */
- oldBorrowedSamples = WEBRTC_SPL_MAX(0,
- borrowedSamples - (inst->endPosition - inst->curPosition));
- WEBRTC_SPL_MEMMOVE_W16(&pw16_decoded_buffer[borrowedSamples],
- pw16_decoded_buffer, len);
- WEBRTC_SPL_MEMCPY_W16(pw16_decoded_buffer,
- &(inst->speechBuffer[inst->endPosition-borrowedSamples]),
- borrowedSamples);
- }
- else
- {
- borrowedSamples = 0;
- oldBorrowedSamples = 0;
- }
-
-#ifdef NETEQ_DELAY_LOGGING
- w16_tmp1 = len;
-#endif
- /* do the expand */
- return_value = WebRtcNetEQ_PreEmptiveExpand(inst,
-#ifdef SCRATCH
- /* use same scratch memory as Accelerate */
- pw16_scratchPtr + SCRATCH_NETEQ_ACCELERATE,
-#endif
- pw16_decoded_buffer, len + borrowedSamples, oldBorrowedSamples,
- pw16_NetEqAlgorithm_buffer, &len, BGNonly);
-
- if (return_value < 0)
- {
- /* error */
- return return_value;
- }
-
- if (borrowedSamples > 0)
- {
- /* return borrowed samples */
-
- /* Copy back to last part of speechBuffer from beginning of output buffer */
- WEBRTC_SPL_MEMCPY_W16( &(inst->speechBuffer[inst->endPosition-borrowedSamples]),
- pw16_NetEqAlgorithm_buffer,
- borrowedSamples);
-
- len -= borrowedSamples; /* remove the borrowed samples from new total length */
-
- /* Move to beginning of output buffer from end of output buffer */
- WEBRTC_SPL_MEMMOVE_W16( pw16_NetEqAlgorithm_buffer,
- &pw16_NetEqAlgorithm_buffer[borrowedSamples],
- len);
- }
-
-#ifdef NETEQ_DELAY_LOGGING
- temp_var = NETEQ_DELAY_LOGGING_SIGNAL_PREEMPTIVE_INFO;
- fwrite(&temp_var, sizeof(int), 1, delay_fid2);
- temp_var = len - w16_tmp1; /* number of samples added */
- fwrite(&temp_var, sizeof(int), 1, delay_fid2);
-#endif
- /* If last packet was decoded as inband CNG, set mode to CNG instead */
- if (speechType == TYPE_CNG) inst->w16_mode = MODE_CODEC_INTERNAL_CNG;
-#ifdef NETEQ_ATEVENT_DECODE
- if (playDtmf == 0)
- {
- inst->DTMFInst.reinit = 1;
- }
-#endif
- break;
-
- case DSP_INSTR_FADE_TO_BGN:
- {
- int tempReturnValue;
- /* do not overwrite return_value, since it likely contains an error code */
-
- /* calculate interpolation length */
- w16_tmp3 = WEBRTC_SPL_MIN(inst->endPosition - inst->curPosition,
- inst->timestampsPerCall);
- /* check that it will fit in pw16_NetEqAlgorithm_buffer */
- if (w16_tmp3 + inst->w16_frameLen > NETEQ_MAX_OUTPUT_SIZE)
- {
- w16_tmp3 = NETEQ_MAX_OUTPUT_SIZE - inst->w16_frameLen;
- }
-
- /* call Expand */
- len = inst->timestampsPerCall + inst->ExpandInst.w16_overlap;
- pos = 0;
-
- tempReturnValue = WebRtcNetEQ_Expand(inst,
-#ifdef SCRATCH
- pw16_scratchPtr + SCRATCH_NETEQ_EXPAND,
-#endif
- pw16_NetEqAlgorithm_buffer, &len, 1);
-
- if (tempReturnValue < 0)
- {
- /* error */
- /* this error value will override return_value */
- return tempReturnValue;
- }
-
- pos += len; /* got len samples from expand */
-
- /* copy to fill the demand */
- while (pos + len <= inst->w16_frameLen + w16_tmp3)
- {
- WEBRTC_SPL_MEMCPY_W16(&pw16_NetEqAlgorithm_buffer[pos],
- pw16_NetEqAlgorithm_buffer, len);
- pos += len;
- }
-
- /* fill with fraction of the expand vector if needed */
- if (pos < inst->w16_frameLen + w16_tmp3)
- {
- WEBRTC_SPL_MEMCPY_W16(&pw16_NetEqAlgorithm_buffer[pos], pw16_NetEqAlgorithm_buffer,
- inst->w16_frameLen + w16_tmp3 - pos);
- }
-
- len = inst->w16_frameLen + w16_tmp3; /* truncate any surplus samples since we don't want these */
-
- /*
- * Mix with contents in sync buffer. Find largest power of two that is less than
- * interpolate length divide 16384 with this number; result is in w16_tmp2.
- */
- w16_tmp1 = 2;
- w16_tmp2 = 16384;
- while (w16_tmp1 <= w16_tmp3)
- {
- w16_tmp2 >>= 1; /* divide with 2 */
- w16_tmp1 <<= 1; /* increase with a factor of 2 */
- }
-
- w16_tmp1 = 0;
- pos = 0;
- while (w16_tmp1 < 16384)
- {
- inst->speechBuffer[inst->curPosition + pos]
- =
- (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(
- WEBRTC_SPL_MUL_16_16( inst->speechBuffer[inst->endPosition - w16_tmp3 + pos],
- 16384-w16_tmp1 ) +
- WEBRTC_SPL_MUL_16_16( pw16_NetEqAlgorithm_buffer[pos], w16_tmp1 ),
- 14 );
- w16_tmp1 += w16_tmp2;
- pos++;
- }
-
- /* overwrite remainder of speech buffer */
-
- WEBRTC_SPL_MEMCPY_W16( &inst->speechBuffer[inst->endPosition - w16_tmp3 + pos],
- &pw16_NetEqAlgorithm_buffer[pos], w16_tmp3 - pos);
-
- len -= w16_tmp3;
- /* shift algorithm buffer */
-
- WEBRTC_SPL_MEMMOVE_W16( pw16_NetEqAlgorithm_buffer,
- &pw16_NetEqAlgorithm_buffer[w16_tmp3],
- len );
-
- /* Update variables for VQmon */
- inst->w16_concealedTS += len;
-
- inst->w16_mode = MODE_FADE_TO_BGN;
-#ifdef NETEQ_DELAY_LOGGING
- temp_var = NETEQ_DELAY_LOGGING_SIGNAL_EXPAND_INFO;
- fwrite(&temp_var, sizeof(int), 1, delay_fid2);
- temp_var = len;
- fwrite(&temp_var, sizeof(int), 1, delay_fid2);
-#endif
-
- break;
- }
-
- default:
- inst->w16_mode = MODE_ERROR;
- dspInfo->lastMode = MODE_ERROR;
- return FAULTY_INSTRUCTION;
- } /* end of grand switch */
-
- /* Copy data directly to output buffer */
-
- w16_tmp2 = 0;
- if ((inst->endPosition + len - inst->curPosition - inst->ExpandInst.w16_overlap)
- >= inst->timestampsPerCall)
- {
- w16_tmp2 = inst->endPosition - inst->curPosition;
- w16_tmp2 = WEBRTC_SPL_MAX(w16_tmp2, 0); /* Additional error protection, just in case */
- w16_tmp1 = WEBRTC_SPL_MIN(w16_tmp2, inst->timestampsPerCall);
- w16_tmp2 = inst->timestampsPerCall - w16_tmp1;
- WEBRTC_SPL_MEMCPY_W16(pw16_outData, &inst->speechBuffer[inst->curPosition], w16_tmp1);
- WEBRTC_SPL_MEMCPY_W16(&pw16_outData[w16_tmp1], pw16_NetEqAlgorithm_buffer, w16_tmp2);
- DataEnough = 1;
- }
- else
- {
- DataEnough = 0;
- }
-
- if (playDtmf != 0)
- {
-#ifdef NETEQ_ATEVENT_DECODE
- WebRtc_Word16 outDataIndex = 0;
- WebRtc_Word16 overdubLen = -1; /* default len */
- WebRtc_Word16 dtmfLen;
-
- /*
- * Overdub the output with DTMF. Note that this is not executed if the
- * DSP_INSTR_DTMF_GENERATE operation is performed above.
- */
- if (inst->DTMFInst.lastDtmfSample - inst->curPosition > 0)
- {
- /* special operation for transition from "DTMF only" to "DTMF overdub" */
- outDataIndex
- = WEBRTC_SPL_MIN(inst->DTMFInst.lastDtmfSample - inst->curPosition,
- inst->timestampsPerCall);
- overdubLen = inst->timestampsPerCall - outDataIndex;
- }
-
- dtmfLen = WebRtcNetEQ_DTMFGenerate(&inst->DTMFInst, dtmfValue, dtmfVolume,
- &pw16_outData[outDataIndex], inst->fs, overdubLen);
- if (dtmfLen < 0)
- {
- /* error occurred */
- return_value = dtmfLen;
- }
- inst->DTMFInst.reinit = 0;
-#else
- inst->w16_mode = MODE_ERROR;
- dspInfo->lastMode = MODE_ERROR;
- return FAULTY_INSTRUCTION;
-#endif
- }
-
- /*
- * Shuffle speech buffer to allow more data. Move data from pw16_NetEqAlgorithm_buffer
- * to speechBuffer.
- */
- if (instr != DSP_INSTR_EXPAND)
- {
- w16_tmp1 = WEBRTC_SPL_MIN(inst->endPosition, len);
- WEBRTC_SPL_MEMMOVE_W16(inst->speechBuffer, inst->speechBuffer + w16_tmp1,
- (inst->endPosition-w16_tmp1));
- WEBRTC_SPL_MEMCPY_W16(&inst->speechBuffer[inst->endPosition-w16_tmp1],
- &pw16_NetEqAlgorithm_buffer[len-w16_tmp1], w16_tmp1);
-#ifdef NETEQ_ATEVENT_DECODE
- /* Update index to end of DTMF data in speech buffer */
- if (instr == DSP_INSTR_DTMF_GENERATE)
- {
- /* We have written DTMF data to the end of speech buffer */
- inst->DTMFInst.lastDtmfSample = inst->endPosition;
- }
- else if (inst->DTMFInst.lastDtmfSample > 0)
- {
- /* The end of DTMF data in speech buffer has been shuffled */
- inst->DTMFInst.lastDtmfSample -= w16_tmp1;
- }
-#endif
- /*
- * Update the BGN history if last operation was not expand (nor Merge, Accelerate
- * or Pre-emptive expand, to save complexity).
- */
- if ((inst->w16_mode != MODE_EXPAND) && (inst->w16_mode != MODE_MERGE)
- && (inst->w16_mode != MODE_SUCCESS_ACCELERATE) && (inst->w16_mode
- != MODE_LOWEN_ACCELERATE) && (inst->w16_mode != MODE_SUCCESS_PREEMPTIVE)
- && (inst->w16_mode != MODE_LOWEN_PREEMPTIVE) && (inst->w16_mode
- != MODE_FADE_TO_BGN) && (inst->w16_mode != MODE_DTMF) && (!BGNonly))
- {
- WebRtcNetEQ_BGNUpdate(inst
-#ifdef SCRATCH
- , pw16_scratchPtr + SCRATCH_NETEQ_BGN_UPDATE
-#endif
- );
- }
- }
- else /* instr == DSP_INSTR_EXPAND */
- {
- /* Nothing should be done since data is already copied to output. */
- }
-
- inst->curPosition -= len;
-
- /*
- * Extra protection in case something should go totally wrong in terms of sizes...
- * If everything is ok this should NEVER happen.
- */
- if (inst->curPosition < -inst->timestampsPerCall)
- {
- inst->curPosition = -inst->timestampsPerCall;
- }
-
- if ((instr != DSP_INSTR_EXPAND) && (instr != DSP_INSTR_MERGE) && (instr
- != DSP_INSTR_FADE_TO_BGN))
- {
- /* Reset concealed TS parameter if it does not seem to have been flushed */
- if (inst->w16_concealedTS > inst->timestampsPerCall)
- {
- inst->w16_concealedTS = 0;
- }
- }
-
- /*
- * Double-check that we actually have 10 ms to play. If we haven't, there has been a
- * serious error.The decoder might have returned way too few samples
- */
- if (!DataEnough)
- {
- /* This should not happen. Set outdata to zeros, and return error. */
- WebRtcSpl_MemSetW16(pw16_outData, 0, inst->timestampsPerCall);
- *pw16_len = inst->timestampsPerCall;
- inst->w16_mode = MODE_ERROR;
- dspInfo->lastMode = MODE_ERROR;
- return RECOUT_ERROR_SAMPLEUNDERRUN;
- }
-
- /*
- * Update Videosync timestamp (this special timestamp is needed since the endTimestamp
- * stops during CNG and Expand periods.
- */
- if ((inst->w16_mode != MODE_EXPAND) && (inst->w16_mode != MODE_RFC3389CNG))
- {
- WebRtc_UWord32 uw32_tmpTS;
- uw32_tmpTS = inst->endTimestamp - (inst->endPosition - inst->curPosition);
- if ((WebRtc_Word32) (uw32_tmpTS - inst->videoSyncTimestamp) > 0)
- {
- inst->videoSyncTimestamp = uw32_tmpTS;
- }
- }
- else
- {
- inst->videoSyncTimestamp += inst->timestampsPerCall;
- }
-
- /* After this, regardless of what has happened, deliver 10 ms of future data */
- inst->curPosition += inst->timestampsPerCall;
- *pw16_len = inst->timestampsPerCall;
-
- /* Remember if BGNonly was used */
- if (BGNonly)
- {
- inst->w16_mode |= MODE_BGN_ONLY;
- }
-
- return return_value;
-}
-
-#undef SCRATCH_ALGORITHM_BUFFER
-#undef SCRATCH_NETEQ_NORMAL
-#undef SCRATCH_NETEQ_MERGE
-#undef SCRATCH_NETEQ_BGN_UPDATE
-#undef SCRATCH_NETEQ_EXPAND
-#undef SCRATCH_DSP_INFO
-#undef SCRATCH_NETEQ_ACCELERATE
-#undef SIZE_SCRATCH_BUFFER
diff --git a/trunk/src/modules/audio_coding/neteq/rtcp.c b/trunk/src/modules/audio_coding/neteq/rtcp.c
deleted file mode 100644
index 35f73da..0000000
--- a/trunk/src/modules/audio_coding/neteq/rtcp.c
+++ /dev/null
@@ -1,134 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Implementation of RTCP statistics reporting.
- */
-
-#include "rtcp.h"
-
-#include <string.h>
-
-#include "signal_processing_library.h"
-
-int WebRtcNetEQ_RTCPInit(WebRtcNetEQ_RTCP_t *RTCP_inst, WebRtc_UWord16 uw16_seqNo)
-{
- /*
- * Initialize everything to zero and then set the start values for the RTP packet stream.
- */
- WebRtcSpl_MemSetW16((WebRtc_Word16*) RTCP_inst, 0,
- sizeof(WebRtcNetEQ_RTCP_t) / sizeof(WebRtc_Word16));
- RTCP_inst->base_seq = uw16_seqNo;
- RTCP_inst->max_seq = uw16_seqNo;
- return 0;
-}
-
-int WebRtcNetEQ_RTCPUpdate(WebRtcNetEQ_RTCP_t *RTCP_inst, WebRtc_UWord16 uw16_seqNo,
- WebRtc_UWord32 uw32_timeStamp, WebRtc_UWord32 uw32_recTime)
-{
- WebRtc_Word16 w16_SeqDiff;
- WebRtc_Word32 w32_TimeDiff;
- WebRtc_Word32 w32_JitterDiff;
-
- /*
- * Update number of received packets, and largest packet number received.
- */
- RTCP_inst->received++;
- w16_SeqDiff = uw16_seqNo - RTCP_inst->max_seq;
- if (w16_SeqDiff >= 0)
- {
- if (uw16_seqNo < RTCP_inst->max_seq)
- {
- /* Wrap around detected */
- RTCP_inst->cycles++;
- }
- RTCP_inst->max_seq = uw16_seqNo;
- }
-
- /* Calculate Jitter, and update previous timestamps */
- /* Note that the value in RTCP_inst->jitter is in Q4. */
- if (RTCP_inst->received > 1)
- {
- w32_TimeDiff = (uw32_recTime - (uw32_timeStamp - RTCP_inst->transit));
- w32_TimeDiff = WEBRTC_SPL_ABS_W32(w32_TimeDiff);
- w32_JitterDiff = WEBRTC_SPL_LSHIFT_W16(w32_TimeDiff, 4) - RTCP_inst->jitter;
- RTCP_inst->jitter = RTCP_inst->jitter + WEBRTC_SPL_RSHIFT_W32((w32_JitterDiff + 8), 4);
- }
- RTCP_inst->transit = (uw32_timeStamp - uw32_recTime);
- return 0;
-}
-
-int WebRtcNetEQ_RTCPGetStats(WebRtcNetEQ_RTCP_t *RTCP_inst,
- WebRtc_UWord16 *puw16_fraction_lost,
- WebRtc_UWord32 *puw32_cum_lost, WebRtc_UWord32 *puw32_ext_max,
- WebRtc_UWord32 *puw32_jitter, WebRtc_Word16 doNotReset)
-{
- WebRtc_UWord32 uw32_exp_nr, uw32_exp_interval, uw32_rec_interval;
- WebRtc_Word32 w32_lost;
-
- /* Extended highest sequence number received */
- *puw32_ext_max
- = (WebRtc_UWord32) WEBRTC_SPL_LSHIFT_W32((WebRtc_UWord32)RTCP_inst->cycles, 16)
- + RTCP_inst->max_seq;
-
- /*
- * Calculate expected number of packets and compare it to the number of packets that
- * were actually received => the cumulative number of packets lost can be extracted.
- */
- uw32_exp_nr = *puw32_ext_max - RTCP_inst->base_seq + 1;
- if (RTCP_inst->received == 0)
- {
- /* no packets received, assume none lost */
- *puw32_cum_lost = 0;
- }
- else if (uw32_exp_nr > RTCP_inst->received)
- {
- *puw32_cum_lost = uw32_exp_nr - RTCP_inst->received;
- if (*puw32_cum_lost > (WebRtc_UWord32) 0xFFFFFF)
- {
- *puw32_cum_lost = 0xFFFFFF;
- }
- }
- else
- {
- *puw32_cum_lost = 0;
- }
-
- /* Fraction lost (Since last report) */
- uw32_exp_interval = uw32_exp_nr - RTCP_inst->exp_prior;
- if (!doNotReset)
- {
- RTCP_inst->exp_prior = uw32_exp_nr;
- }
- uw32_rec_interval = RTCP_inst->received - RTCP_inst->rec_prior;
- if (!doNotReset)
- {
- RTCP_inst->rec_prior = RTCP_inst->received;
- }
- w32_lost = (WebRtc_Word32) (uw32_exp_interval - uw32_rec_interval);
- if (uw32_exp_interval == 0 || w32_lost <= 0 || RTCP_inst->received == 0)
- {
- *puw16_fraction_lost = 0;
- }
- else
- {
- *puw16_fraction_lost = (WebRtc_UWord16) (WEBRTC_SPL_LSHIFT_W32(w32_lost, 8)
- / uw32_exp_interval);
- }
- if (*puw16_fraction_lost > 0xFF)
- {
- *puw16_fraction_lost = 0xFF;
- }
-
- /* Inter-arrival jitter */
- *puw32_jitter = (RTCP_inst->jitter) >> 4; /* scaling from Q4 */
- return 0;
-}
-
diff --git a/trunk/src/modules/audio_coding/neteq/rtcp.h b/trunk/src/modules/audio_coding/neteq/rtcp.h
deleted file mode 100644
index 009e019..0000000
--- a/trunk/src/modules/audio_coding/neteq/rtcp.h
+++ /dev/null
@@ -1,102 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * RTCP statistics reporting.
- */
-
-#ifndef RTCP_H
-#define RTCP_H
-
-#include "typedefs.h"
-
-typedef struct
-{
- WebRtc_UWord16 cycles; /* The number of wrap-arounds for the sequence number */
- WebRtc_UWord16 max_seq; /* The maximum sequence number received
- (starts from 0 again after wrap around) */
- WebRtc_UWord16 base_seq; /* The sequence number of the first packet that arrived */
- WebRtc_UWord32 received; /* The number of packets that has been received */
- WebRtc_UWord32 rec_prior; /* Number of packets received when last report was generated */
- WebRtc_UWord32 exp_prior; /* Number of packets that should have been received if no
- packets were lost. Stored value from last report. */
- WebRtc_UWord32 jitter; /* Jitter statistics at this instance (calculated according to RFC) */
- WebRtc_Word32 transit; /* Clock difference for previous packet (RTPtimestamp - LOCALtime_rec) */
-} WebRtcNetEQ_RTCP_t;
-
-/****************************************************************************
- * WebRtcNetEQ_RTCPInit(...)
- *
- * This function calculates the parameters that are needed for the RTCP
- * report.
- *
- * Input:
- * - RTCP_inst : RTCP instance, that contains information about the
- * packets that have been received etc.
- * - seqNo : Packet number of the first received frame.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_RTCPInit(WebRtcNetEQ_RTCP_t *RTCP_inst, WebRtc_UWord16 uw16_seqNo);
-
-/****************************************************************************
- * WebRtcNetEQ_RTCPUpdate(...)
- *
- * This function calculates the parameters that are needed for the RTCP
- * report.
- *
- * Input:
- * - RTCP_inst : RTCP instance, that contains information about the
- * packets that have been received etc.
- * - seqNo : Packet number of the first received frame.
- * - timeStamp : Time stamp from the RTP header.
- * - recTime : Time (in RTP timestamps) when this packet was received.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_RTCPUpdate(WebRtcNetEQ_RTCP_t *RTCP_inst, WebRtc_UWord16 uw16_seqNo,
- WebRtc_UWord32 uw32_timeStamp, WebRtc_UWord32 uw32_recTime);
-
-/****************************************************************************
- * WebRtcNetEQ_RTCPGetStats(...)
- *
- * This function calculates the parameters that are needed for the RTCP
- * report.
- *
- * Input:
- * - RTCP_inst : RTCP instance, that contains information about the
- * packets that have been received etc.
- * - doNotReset : If non-zero, the fraction lost statistics will not
- * be reset.
- *
- * Output:
- * - RTCP_inst : Updated RTCP information (some statistics are
- * reset when generating this report)
- * - fraction_lost : Number of lost RTP packets divided by the number of
- * expected packets, since the last RTCP Report.
- * - cum_lost : Cumulative number of lost packets during this
- * session.
- * - ext_max : Extended highest sequence number received.
- * - jitter : Inter-arrival jitter.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_RTCPGetStats(WebRtcNetEQ_RTCP_t *RTCP_inst,
- WebRtc_UWord16 *puw16_fraction_lost,
- WebRtc_UWord32 *puw32_cum_lost, WebRtc_UWord32 *puw32_ext_max,
- WebRtc_UWord32 *puw32_jitter, WebRtc_Word16 doNotReset);
-
-#endif
diff --git a/trunk/src/modules/audio_coding/neteq/rtp.c b/trunk/src/modules/audio_coding/neteq/rtp.c
deleted file mode 100644
index bd4f9a2..0000000
--- a/trunk/src/modules/audio_coding/neteq/rtp.c
+++ /dev/null
@@ -1,240 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * RTP related functions.
- */
-
-#include "rtp.h"
-
-#include "typedefs.h" /* to define endianness */
-
-#include "neteq_error_codes.h"
-
-int WebRtcNetEQ_RTPPayloadInfo(WebRtc_Word16* pw16_Datagram, int i_DatagramLen,
- RTPPacket_t* RTPheader)
-{
- int i_P, i_X, i_CC, i_startPosition;
- int i_IPver;
- int i_extlength = -1; /* Default value is there is no extension */
- int i_padlength = 0; /* Default value if there is no padding */
-
- if (i_DatagramLen < 12)
- {
- return RTP_TOO_SHORT_PACKET;
- }
-
-#ifdef WEBRTC_BIG_ENDIAN
- i_IPver = (((WebRtc_UWord16) (pw16_Datagram[0] & 0xC000)) >> 14); /* Extract the version */
- i_P = (((WebRtc_UWord16) (pw16_Datagram[0] & 0x2000)) >> 13); /* Extract the P bit */
- i_X = (((WebRtc_UWord16) (pw16_Datagram[0] & 0x1000)) >> 12); /* Extract the X bit */
- i_CC = ((WebRtc_UWord16) (pw16_Datagram[0] >> 8) & 0xF); /* Get the CC number */
- RTPheader->payloadType = pw16_Datagram[0] & 0x7F; /* Get the coder type */
- RTPheader->seqNumber = pw16_Datagram[1]; /* Get the sequence number */
- RTPheader->timeStamp = ((((WebRtc_UWord32) ((WebRtc_UWord16) pw16_Datagram[2])) << 16)
- | (WebRtc_UWord16) (pw16_Datagram[3])); /* Get timestamp */
- RTPheader->ssrc = (((WebRtc_UWord32) pw16_Datagram[4]) << 16)
- + (((WebRtc_UWord32) pw16_Datagram[5])); /* Get the SSRC */
-
- if (i_X == 1)
- {
- /* Extension header exists. Find out how many WebRtc_Word32 it consists of. */
- i_extlength = pw16_Datagram[7 + 2 * i_CC];
- }
- if (i_P == 1)
- {
- /* Padding exists. Find out how many bytes the padding consists of. */
- if (i_DatagramLen & 0x1)
- {
- /* odd number of bytes => last byte in higher byte */
- i_padlength = (((WebRtc_UWord16) pw16_Datagram[i_DatagramLen >> 1]) >> 8);
- }
- else
- {
- /* even number of bytes => last byte in lower byte */
- i_padlength = ((pw16_Datagram[(i_DatagramLen >> 1) - 1]) & 0xFF);
- }
- }
-#else /* WEBRTC_LITTLE_ENDIAN */
- i_IPver = (((WebRtc_UWord16) (pw16_Datagram[0] & 0xC0)) >> 6); /* Extract the IP version */
- i_P = (((WebRtc_UWord16) (pw16_Datagram[0] & 0x20)) >> 5); /* Extract the P bit */
- i_X = (((WebRtc_UWord16) (pw16_Datagram[0] & 0x10)) >> 4); /* Extract the X bit */
- i_CC = (WebRtc_UWord16) (pw16_Datagram[0] & 0xF); /* Get the CC number */
- RTPheader->payloadType = (pw16_Datagram[0] >> 8) & 0x7F; /* Get the coder type */
- RTPheader->seqNumber = (((((WebRtc_UWord16) pw16_Datagram[1]) >> 8) & 0xFF)
- | (((WebRtc_UWord16) (pw16_Datagram[1] & 0xFF)) << 8)); /* Get the packet number */
- RTPheader->timeStamp = ((((WebRtc_UWord16) pw16_Datagram[2]) & 0xFF) << 24)
- | ((((WebRtc_UWord16) pw16_Datagram[2]) & 0xFF00) << 8)
- | ((((WebRtc_UWord16) pw16_Datagram[3]) >> 8) & 0xFF)
- | ((((WebRtc_UWord16) pw16_Datagram[3]) & 0xFF) << 8); /* Get timestamp */
- RTPheader->ssrc = ((((WebRtc_UWord16) pw16_Datagram[4]) & 0xFF) << 24)
- | ((((WebRtc_UWord16) pw16_Datagram[4]) & 0xFF00) << 8)
- | ((((WebRtc_UWord16) pw16_Datagram[5]) >> 8) & 0xFF)
- | ((((WebRtc_UWord16) pw16_Datagram[5]) & 0xFF) << 8); /* Get the SSRC */
-
- if (i_X == 1)
- {
- /* Extension header exists. Find out how many WebRtc_Word32 it consists of. */
- i_extlength = (((((WebRtc_UWord16) pw16_Datagram[7 + 2 * i_CC]) >> 8) & 0xFF)
- | (((WebRtc_UWord16) (pw16_Datagram[7 + 2 * i_CC] & 0xFF)) << 8));
- }
- if (i_P == 1)
- {
- /* Padding exists. Find out how many bytes the padding consists of. */
- if (i_DatagramLen & 0x1)
- {
- /* odd number of bytes => last byte in higher byte */
- i_padlength = (pw16_Datagram[i_DatagramLen >> 1] & 0xFF);
- }
- else
- {
- /* even number of bytes => last byte in lower byte */
- i_padlength = (((WebRtc_UWord16) pw16_Datagram[(i_DatagramLen >> 1) - 1]) >> 8);
- }
- }
-#endif
-
- i_startPosition = 12 + 4 * (i_extlength + 1) + 4 * i_CC;
- RTPheader->payload = &pw16_Datagram[i_startPosition >> 1];
- RTPheader->payloadLen = i_DatagramLen - i_startPosition - i_padlength;
- RTPheader->starts_byte1 = 0;
-
- if ((i_IPver != 2) || (RTPheader->payloadLen <= 0) || (RTPheader->payloadLen >= 16000)
- || (i_startPosition < 12) || (i_startPosition > i_DatagramLen))
- {
- return RTP_CORRUPT_PACKET;
- }
-
- return 0;
-}
-
-#ifdef NETEQ_RED_CODEC
-
-int WebRtcNetEQ_RedundancySplit(RTPPacket_t* RTPheader[], int i_MaximumPayloads,
- int *i_No_Of_Payloads)
-{
- const WebRtc_Word16 *pw16_data = RTPheader[0]->payload; /* Pointer to the data */
- WebRtc_UWord16 uw16_offsetTimeStamp = 65535, uw16_secondPayload = 65535;
- int i_blockLength, i_k;
- int i_discardedBlockLength = 0;
- int singlePayload = 0;
-
-#ifdef WEBRTC_BIG_ENDIAN
- if ((pw16_data[0] & 0x8000) == 0)
- {
- /* Only one payload in this packet*/
- singlePayload = 1;
- /* set the blocklength to -4 to deduce the non-existent 4-byte RED header */
- i_blockLength = -4;
- RTPheader[0]->payloadType = ((((WebRtc_UWord16)pw16_data[0]) & 0x7F00) >> 8);
- }
- else
- {
- /* Discard all but the two last payloads. */
- while (((pw16_data[2] & 0x8000) == 1)&&
- (pw16_data<((RTPheader[0]->payload)+((RTPheader[0]->payloadLen+1)>>1))))
- {
- i_discardedBlockLength += (4+(((WebRtc_UWord16)pw16_data[1]) & 0x3FF));
- pw16_data+=2;
- }
- if (pw16_data>=(RTPheader[0]->payload+((RTPheader[0]->payloadLen+1)>>1)))
- {
- return RED_SPLIT_ERROR2; /* Error, we are outside the packet */
- }
- singlePayload = 0; /* the packet contains more than one payload */
- uw16_secondPayload = ((((WebRtc_UWord16)pw16_data[0]) & 0x7F00) >> 8);
- RTPheader[0]->payloadType = ((((WebRtc_UWord16)pw16_data[2]) & 0x7F00) >> 8);
- uw16_offsetTimeStamp = ((((WebRtc_UWord16)pw16_data[0]) & 0xFF) << 6) +
- ((((WebRtc_UWord16)pw16_data[1]) & 0xFC00) >> 10);
- i_blockLength = (((WebRtc_UWord16)pw16_data[1]) & 0x3FF);
- }
-#else /* WEBRTC_LITTLE_ENDIAN */
- if ((pw16_data[0] & 0x80) == 0)
- {
- /* Only one payload in this packet */
- singlePayload = 1;
- /* set the blocklength to -4 to deduce the non-existent 4-byte RED header */
- i_blockLength = -4;
- RTPheader[0]->payloadType = (((WebRtc_UWord16) pw16_data[0]) & 0x7F);
- }
- else
- {
- /* Discard all but the two last payloads. */
- while (((pw16_data[2] & 0x80) == 1) && (pw16_data < ((RTPheader[0]->payload)
- + ((RTPheader[0]->payloadLen + 1) >> 1))))
- {
- i_discardedBlockLength += (4 + ((((WebRtc_UWord16) pw16_data[1]) & 0x3) << 8)
- + ((((WebRtc_UWord16) pw16_data[1]) & 0xFF00) >> 8));
- pw16_data += 2;
- }
- if (pw16_data >= (RTPheader[0]->payload + ((RTPheader[0]->payloadLen + 1) >> 1)))
- {
- return RED_SPLIT_ERROR2; /* Error, we are outside the packet */;
- }
- singlePayload = 0; /* the packet contains more than one payload */
- uw16_secondPayload = (((WebRtc_UWord16) pw16_data[0]) & 0x7F);
- RTPheader[0]->payloadType = (((WebRtc_UWord16) pw16_data[2]) & 0x7F);
- uw16_offsetTimeStamp = ((((WebRtc_UWord16) pw16_data[0]) & 0xFF00) >> 2)
- + ((((WebRtc_UWord16) pw16_data[1]) & 0xFC) >> 2);
- i_blockLength = ((((WebRtc_UWord16) pw16_data[1]) & 0x3) << 8)
- + ((((WebRtc_UWord16) pw16_data[1]) & 0xFF00) >> 8);
- }
-#endif
-
- if (i_MaximumPayloads < 2 || singlePayload == 1)
- {
- /* Reject the redundancy; or no redundant payload present. */
- for (i_k = 1; i_k < i_MaximumPayloads; i_k++)
- {
- RTPheader[i_k]->payloadType = -1;
- RTPheader[i_k]->payloadLen = 0;
- }
-
- /* update the pointer for the main data */
- pw16_data = &pw16_data[(5 + i_blockLength) >> 1];
- RTPheader[0]->starts_byte1 = (5 + i_blockLength) & 0x1;
- RTPheader[0]->payloadLen = RTPheader[0]->payloadLen - (i_blockLength + 5)
- - i_discardedBlockLength;
- RTPheader[0]->payload = pw16_data;
-
- *i_No_Of_Payloads = 1;
-
- }
- else
- {
- /* Redundancy accepted, put the redundancy in second RTPheader. */
- RTPheader[1]->payloadType = uw16_secondPayload;
- RTPheader[1]->payload = &pw16_data[5 >> 1];
- RTPheader[1]->starts_byte1 = 5 & 0x1;
- RTPheader[1]->seqNumber = RTPheader[0]->seqNumber;
- RTPheader[1]->timeStamp = RTPheader[0]->timeStamp - uw16_offsetTimeStamp;
- RTPheader[1]->ssrc = RTPheader[0]->ssrc;
- RTPheader[1]->payloadLen = i_blockLength;
-
- /* Modify first RTP packet, so that it contains the main data. */
- RTPheader[0]->payload = &pw16_data[(5 + i_blockLength) >> 1];
- RTPheader[0]->starts_byte1 = (5 + i_blockLength) & 0x1;
- RTPheader[0]->payloadLen = RTPheader[0]->payloadLen - (i_blockLength + 5)
- - i_discardedBlockLength;
-
- /* Clear the following payloads. */
- for (i_k = 2; i_k < i_MaximumPayloads; i_k++)
- {
- RTPheader[i_k]->payloadType = -1;
- RTPheader[i_k]->payloadLen = 0;
- }
-
- *i_No_Of_Payloads = 2;
- }
- return 0;
-}
-
-#endif
-
diff --git a/trunk/src/modules/audio_coding/neteq/rtp.h b/trunk/src/modules/audio_coding/neteq/rtp.h
deleted file mode 100644
index 8490d62..0000000
--- a/trunk/src/modules/audio_coding/neteq/rtp.h
+++ /dev/null
@@ -1,78 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * RTP data struct and related functions.
- */
-
-#ifndef RTP_H
-#define RTP_H
-
-#include "typedefs.h"
-
-#include "codec_db.h"
-
-typedef struct
-{
- WebRtc_UWord16 seqNumber;
- WebRtc_UWord32 timeStamp;
- WebRtc_UWord32 ssrc;
- int payloadType;
- const WebRtc_Word16 *payload;
- WebRtc_Word16 payloadLen;
- WebRtc_Word16 starts_byte1;
- WebRtc_Word16 rcuPlCntr;
-} RTPPacket_t;
-
-/****************************************************************************
- * WebRtcNetEQ_RTPPayloadInfo(...)
- *
- * Converts a datagram into an RTP header struct.
- *
- * Input:
- * - Datagram : UDP datagram from the network
- * - DatagramLen : Length in bytes of the datagram
- *
- * Output:
- * - RTPheader : Structure with the datagram info
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_RTPPayloadInfo(WebRtc_Word16* pw16_Datagram, int i_DatagramLen,
- RTPPacket_t* RTPheader);
-
-/****************************************************************************
- * WebRtcNetEQ_RedundancySplit(...)
- *
- * Splits a Redundancy RTP struct into two RTP structs. User has to check
- * that it's really the redundancy payload. No such check is done inside this
- * function.
- *
- * Input:
- * - RTPheader : First header holds the whole RTP packet (with the redundancy payload)
- * - MaximumPayloads:
- * The maximum number of RTP payloads that should be
- * extracted (1+maximum_no_of_Redundancies).
- *
- * Output:
- * - RTPheader : First header holds the main RTP data, while 2..N
- * holds the redundancy data.
- * - No_Of
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_RedundancySplit(RTPPacket_t* RTPheader[], int i_MaximumPayloads,
- int *i_No_Of_Payloads);
-
-#endif
diff --git a/trunk/src/modules/audio_coding/neteq/set_fs.c b/trunk/src/modules/audio_coding/neteq/set_fs.c
deleted file mode 100644
index b2ad5ca..0000000
--- a/trunk/src/modules/audio_coding/neteq/set_fs.c
+++ /dev/null
@@ -1,78 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Function were the sample rate is set.
- */
-
-#include "mcu.h"
-
-#include "dtmf_buffer.h"
-#include "neteq_error_codes.h"
-
-int WebRtcNetEQ_McuSetFs(MCUInst_t *inst, WebRtc_UWord16 fs)
-{
- WebRtc_Word16 ok = 0;
-
- switch (fs)
- {
- case 8000:
- {
-#ifdef NETEQ_ATEVENT_DECODE
- ok = WebRtcNetEQ_DtmfDecoderInit(&inst->DTMF_inst, 8000, 560);
-#endif
- inst->timestampsPerCall = inst->millisecondsPerCall * 8;
- break;
- }
-
-#ifdef NETEQ_WIDEBAND
- case 16000:
- {
-#ifdef NETEQ_ATEVENT_DECODE
- ok = WebRtcNetEQ_DtmfDecoderInit(&inst->DTMF_inst, 16000, 1120);
-#endif
- inst->timestampsPerCall = inst->millisecondsPerCall * 16;
- break;
- }
-#endif
-
-#ifdef NETEQ_32KHZ_WIDEBAND
- case 32000:
- {
-#ifdef NETEQ_ATEVENT_DECODE
- ok = WebRtcNetEQ_DtmfDecoderInit(&inst->DTMF_inst, 32000, 2240);
-#endif
- inst->timestampsPerCall = inst->millisecondsPerCall * 32;
- break;
- }
-#endif
-
-#ifdef NETEQ_48KHZ_WIDEBAND
- case 48000:
- {
-#ifdef NETEQ_ATEVENT_DECODE
- ok = WebRtcNetEQ_DtmfDecoderInit(&inst->DTMF_inst, 48000, 3360);
-#endif
- inst->timestampsPerCall = inst->millisecondsPerCall * 48;
- break;
- }
-#endif
-
- default:
- {
- /* Not supported yet */
- return CODEC_DB_UNSUPPORTED_FS;
- }
- } /* end switch */
-
- inst->fs = fs;
-
- return ok;
-}
diff --git a/trunk/src/modules/audio_coding/neteq/signal_mcu.c b/trunk/src/modules/audio_coding/neteq/signal_mcu.c
deleted file mode 100644
index 0180f38..0000000
--- a/trunk/src/modules/audio_coding/neteq/signal_mcu.c
+++ /dev/null
@@ -1,766 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Signal the MCU that data is available and ask for a RecOut decision.
- */
-
-#include "mcu.h"
-
-#include <string.h>
-
-#include "signal_processing_library.h"
-
-#include "automode.h"
-#include "dtmf_buffer.h"
-#include "mcu_dsp_common.h"
-#include "neteq_error_codes.h"
-
-#ifdef NETEQ_DELAY_LOGGING
-#include "delay_logging.h"
-#include <stdio.h>
-
-extern FILE *delay_fid2; /* file pointer to delay log file */
-#endif
-
-
-/*
- * Signals the MCU that DSP status data is available.
- */
-int WebRtcNetEQ_SignalMcu(MCUInst_t *inst)
-{
-
- int i_bufferpos, i_res;
- WebRtc_UWord16 uw16_instr;
- DSP2MCU_info_t dspInfo;
- WebRtc_Word16 *blockPtr, blockLen;
- WebRtc_UWord32 uw32_availableTS;
- RTPPacket_t temp_pkt;
- WebRtc_Word32 w32_bufsize, w32_tmp;
- WebRtc_Word16 payloadType = -1;
- WebRtc_Word16 wantedNoOfTimeStamps;
- WebRtc_Word32 totalTS;
- WebRtc_Word16 oldPT, latePacketExist = 0;
- WebRtc_UWord32 oldTS, prevTS, uw32_tmp;
- WebRtc_UWord16 prevSeqNo;
- WebRtc_Word16 nextSeqNoAvail;
- WebRtc_Word16 fs_mult, w16_tmp;
- WebRtc_Word16 lastModeBGNonly = 0;
-#ifdef NETEQ_DELAY_LOGGING
- int temp_var;
-#endif
- int playDtmf = 0;
-
- fs_mult = WebRtcSpl_DivW32W16ResW16(inst->fs, 8000);
-
- /* Increment counter since last statistics report */
- inst->lastReportTS += inst->timestampsPerCall;
-
- /* Increment waiting time for all packets. */
- WebRtcNetEQ_IncrementWaitingTimes(&inst->PacketBuffer_inst);
-
- /* Read info from DSP so we now current status */
-
- WEBRTC_SPL_MEMCPY_W8(&dspInfo,inst->pw16_readAddress,sizeof(DSP2MCU_info_t));
-
- /* Set blockPtr to first payload block */
- blockPtr = &inst->pw16_writeAddress[3];
-
- /* Clear instruction word and number of lost samples (2*WebRtc_Word16) */
- inst->pw16_writeAddress[0] = 0;
- inst->pw16_writeAddress[1] = 0;
- inst->pw16_writeAddress[2] = 0;
-
- if ((dspInfo.lastMode & MODE_AWAITING_CODEC_PTR) != 0)
- {
- /*
- * Make sure state is adjusted so that a codec update is
- * performed when first packet arrives.
- */
- if (inst->new_codec != 1)
- {
- inst->current_Codec = -1;
- }
- dspInfo.lastMode = (dspInfo.lastMode ^ MODE_AWAITING_CODEC_PTR);
- }
-
-#ifdef NETEQ_STEREO
- if ((dspInfo.lastMode & MODE_MASTER_DTMF_SIGNAL) != 0)
- {
- playDtmf = 1; /* force DTMF decision */
- dspInfo.lastMode = (dspInfo.lastMode ^ MODE_MASTER_DTMF_SIGNAL);
- }
-
- if ((dspInfo.lastMode & MODE_USING_STEREO) != 0)
- {
- if (inst->usingStereo == 0)
- {
- /* stereo mode changed; reset automode instance to re-synchronize statistics */
- WebRtcNetEQ_ResetAutomode(&(inst->BufferStat_inst.Automode_inst),
- inst->PacketBuffer_inst.maxInsertPositions);
- }
- inst->usingStereo = 1;
- dspInfo.lastMode = (dspInfo.lastMode ^ MODE_USING_STEREO);
- }
- else
- {
- inst->usingStereo = 0;
- }
-#endif
-
- /* detect if BGN_ONLY flag is set in lastMode */
- if ((dspInfo.lastMode & MODE_BGN_ONLY) != 0)
- {
- lastModeBGNonly = 1; /* remember flag */
- dspInfo.lastMode ^= MODE_BGN_ONLY; /* clear the flag */
- }
-
- if ((dspInfo.lastMode == MODE_RFC3389CNG) || (dspInfo.lastMode == MODE_CODEC_INTERNAL_CNG)
- || (dspInfo.lastMode == MODE_EXPAND))
- {
- /*
- * If last mode was CNG (or Expand, since this could be covering up for a lost CNG
- * packet), increase the CNGplayedTS counter.
- */
- inst->BufferStat_inst.uw32_CNGplayedTS += inst->timestampsPerCall;
-
- if (dspInfo.lastMode == MODE_RFC3389CNG)
- {
- /* remember that RFC3389CNG is on (needed if CNG is interrupted by DTMF) */
- inst->BufferStat_inst.w16_cngOn = CNG_RFC3389_ON;
- }
- else if (dspInfo.lastMode == MODE_CODEC_INTERNAL_CNG)
- {
- /* remember that internal CNG is on (needed if CNG is interrupted by DTMF) */
- inst->BufferStat_inst.w16_cngOn = CNG_INTERNAL_ON;
- }
-
- }
-
- /* Update packet size from previously decoded packet */
- if (dspInfo.frameLen > 0)
- {
- inst->PacketBuffer_inst.packSizeSamples = dspInfo.frameLen;
- }
-
- /* Look for late packet (unless codec has changed) */
- if (inst->new_codec != 1)
- {
- if (WebRtcNetEQ_DbIsMDCodec((enum WebRtcNetEQDecoder) inst->current_Codec))
- {
- WebRtcNetEQ_PacketBufferFindLowestTimestamp(&inst->PacketBuffer_inst,
- inst->timeStamp, &uw32_availableTS, &i_bufferpos, 1, &payloadType);
- if ((inst->new_codec != 1) && (inst->timeStamp == uw32_availableTS)
- && (inst->timeStamp < dspInfo.playedOutTS) && (i_bufferpos != -1)
- && (WebRtcNetEQ_DbGetPayload(&(inst->codec_DB_inst),
- (enum WebRtcNetEQDecoder) inst->current_Codec) == payloadType))
- {
- int waitingTime;
- temp_pkt.payload = blockPtr + 1;
- i_res = WebRtcNetEQ_PacketBufferExtract(&inst->PacketBuffer_inst, &temp_pkt,
- i_bufferpos, &waitingTime);
- if (i_res < 0)
- { /* error returned */
- return i_res;
- }
- WebRtcNetEQ_StoreWaitingTime(inst, waitingTime);
- *blockPtr = temp_pkt.payloadLen;
- /* set the flag if this is a redundant payload */
- if (temp_pkt.rcuPlCntr > 0)
- {
- *blockPtr = (*blockPtr) | (DSP_CODEC_RED_FLAG);
- }
- blockPtr += ((temp_pkt.payloadLen + 1) >> 1) + 1;
-
- /*
- * Close the data with a zero size block, in case we will not write any
- * more data.
- */
- *blockPtr = 0;
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0xf0ff)
- | DSP_CODEC_ADD_LATE_PKT;
- latePacketExist = 1;
- }
- }
- }
-
- i_res = WebRtcNetEQ_PacketBufferFindLowestTimestamp(&inst->PacketBuffer_inst,
- dspInfo.playedOutTS, &uw32_availableTS, &i_bufferpos, (inst->new_codec == 0),
- &payloadType);
- if (i_res < 0)
- { /* error returned */
- return i_res;
- }
-
- if (inst->BufferStat_inst.w16_cngOn == CNG_RFC3389_ON)
- {
- /*
- * Because of timestamp peculiarities, we have to "manually" disallow using a CNG
- * packet with the same timestamp as the one that was last played. This can happen
- * when using redundancy and will cause the timing to shift.
- */
- while (i_bufferpos != -1 && WebRtcNetEQ_DbIsCNGPayload(&inst->codec_DB_inst,
- payloadType) && dspInfo.playedOutTS >= uw32_availableTS)
- {
-
- /* Don't use this packet, discard it */
- inst->PacketBuffer_inst.payloadType[i_bufferpos] = -1;
- inst->PacketBuffer_inst.payloadLengthBytes[i_bufferpos] = 0;
- inst->PacketBuffer_inst.numPacketsInBuffer--;
-
- /* Check buffer again */
- WebRtcNetEQ_PacketBufferFindLowestTimestamp(&inst->PacketBuffer_inst,
- dspInfo.playedOutTS, &uw32_availableTS, &i_bufferpos, (inst->new_codec == 0),
- &payloadType);
- }
- }
-
- /* Check packet buffer */
- w32_bufsize = WebRtcNetEQ_PacketBufferGetSize(&inst->PacketBuffer_inst);
-
- if (dspInfo.lastMode == MODE_SUCCESS_ACCELERATE || dspInfo.lastMode
- == MODE_LOWEN_ACCELERATE || dspInfo.lastMode == MODE_SUCCESS_PREEMPTIVE
- || dspInfo.lastMode == MODE_LOWEN_PREEMPTIVE)
- {
- /* Subtract (dspInfo.samplesLeft + inst->timestampsPerCall) from sampleMemory */
- inst->BufferStat_inst.Automode_inst.sampleMemory -= dspInfo.samplesLeft
- + inst->timestampsPerCall;
- }
-
- /* calculate total current buffer size (in ms*8), including sync buffer */
- w32_bufsize = WebRtcSpl_DivW32W16((w32_bufsize + dspInfo.samplesLeft), fs_mult);
-
-#ifdef NETEQ_ATEVENT_DECODE
- /* DTMF data will affect the decision */
- if (WebRtcNetEQ_DtmfDecode(&inst->DTMF_inst, blockPtr + 1, blockPtr + 2,
- dspInfo.playedOutTS + inst->BufferStat_inst.uw32_CNGplayedTS) > 0)
- {
- playDtmf = 1;
-
- /* Flag DTMF payload */
- inst->pw16_writeAddress[0] = inst->pw16_writeAddress[0] | DSP_DTMF_PAYLOAD;
-
- /* Block Length in bytes */
- blockPtr[0] = 4;
- /* Advance to next payload position */
- blockPtr += 3;
- }
-#endif
-
- /* Update statistics and make decision */
- uw16_instr = WebRtcNetEQ_BufstatsDecision(&inst->BufferStat_inst,
- inst->PacketBuffer_inst.packSizeSamples, w32_bufsize, dspInfo.playedOutTS,
- uw32_availableTS, i_bufferpos == -1,
- WebRtcNetEQ_DbIsCNGPayload(&inst->codec_DB_inst, payloadType), dspInfo.lastMode,
- inst->NetEqPlayoutMode, inst->timestampsPerCall, inst->NoOfExpandCalls, fs_mult,
- lastModeBGNonly, playDtmf);
-
- /* Check if time to reset loss counter */
- if (inst->lastReportTS > WEBRTC_SPL_UMUL(inst->fs, MAX_LOSS_REPORT_PERIOD))
- {
- /* reset loss counter */
- WebRtcNetEQ_ResetMcuInCallStats(inst);
- }
-
- /* Check sync buffer size */
- if ((dspInfo.samplesLeft >= inst->timestampsPerCall) && (uw16_instr
- != BUFSTATS_DO_ACCELERATE) && (uw16_instr != BUFSTATS_DO_MERGE) && (uw16_instr
- != BUFSTATS_DO_PREEMPTIVE_EXPAND))
- {
- *blockPtr = 0;
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) | DSP_INSTR_NORMAL;
- return 0;
- }
-
- if (uw16_instr == BUFSTATS_DO_EXPAND)
- {
- inst->NoOfExpandCalls++;
- }
- else
- {
- /* reset counter */
- inst->NoOfExpandCalls = 0;
- }
-
- /* New codec or big change in packet number? */
- if (((inst->new_codec) || (uw16_instr == BUFSTAT_REINIT)) && (uw16_instr
- != BUFSTATS_DO_EXPAND))
- {
- CodecFuncInst_t cinst;
-
- /* Clear other instructions */
- blockPtr = &inst->pw16_writeAddress[3];
- /* Clear instruction word */
- inst->pw16_writeAddress[0] = 0;
-
- inst->timeStamp = uw32_availableTS;
- dspInfo.playedOutTS = uw32_availableTS;
- if (inst->current_Codec != -1)
- {
- i_res = WebRtcNetEQ_DbGetPtrs(&inst->codec_DB_inst,
- (enum WebRtcNetEQDecoder) inst->current_Codec, &cinst);
- if (i_res < 0)
- { /* error returned */
- return i_res;
- }
- }
- else
- {
- /* The main codec has not been initialized yet (first packets are DTMF or CNG). */
- if (WebRtcNetEQ_DbIsCNGPayload(&inst->codec_DB_inst, payloadType))
- {
- /* The currently extracted packet is CNG; get CNG fs */
- WebRtc_UWord16 tempFs;
-
- tempFs = WebRtcNetEQ_DbGetSampleRate(&inst->codec_DB_inst, payloadType);
- if (tempFs > 0)
- {
- inst->fs = tempFs;
- }
- }
- WebRtcSpl_MemSetW16((WebRtc_Word16*) &cinst, 0,
- sizeof(CodecFuncInst_t) / sizeof(WebRtc_Word16));
- cinst.codec_fs = inst->fs;
- }
- cinst.timeStamp = inst->timeStamp;
- blockLen = (sizeof(CodecFuncInst_t)) >> (sizeof(WebRtc_Word16) - 1); /* in Word16 */
- *blockPtr = blockLen * 2;
- blockPtr++;
- WEBRTC_SPL_MEMCPY_W8(blockPtr,&cinst,sizeof(CodecFuncInst_t));
- blockPtr += blockLen;
- inst->new_codec = 0;
-
- /* Reinitialize the MCU fs */
- i_res = WebRtcNetEQ_McuSetFs(inst, cinst.codec_fs);
- if (i_res < 0)
- { /* error returned */
- return i_res;
- }
-
- /* Set the packet size by guessing */
- inst->PacketBuffer_inst.packSizeSamples = inst->timestampsPerCall * 3;
-
- WebRtcNetEQ_ResetAutomode(&(inst->BufferStat_inst.Automode_inst),
- inst->PacketBuffer_inst.maxInsertPositions);
-
-#ifdef NETEQ_CNG_CODEC
- /* Also insert CNG state as this might be needed by DSP */
- i_res = WebRtcNetEQ_DbGetPtrs(&inst->codec_DB_inst, kDecoderCNG, &cinst);
- if ((i_res < 0) && (i_res != CODEC_DB_NOT_EXIST1))
- {
- /* other error returned */
- /* (CODEC_DB_NOT_EXIST1 simply indicates that CNG is not used */
- return i_res;
- }
- else
- {
- /* CNG exists */
- blockLen = (sizeof(cinst.codec_state)) >> (sizeof(WebRtc_Word16) - 1);
- *blockPtr = blockLen * 2;
- blockPtr++;
- WEBRTC_SPL_MEMCPY_W8(blockPtr,&cinst.codec_state,sizeof(cinst.codec_state));
- blockPtr += blockLen;
- }
-#endif
-
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0xf0ff)
- | DSP_CODEC_NEW_CODEC;
-
- if (uw16_instr == BUFSTATS_DO_RFC3389CNG_NOPACKET)
- {
- /*
- * Change decision to CNG packet, since we do have a CNG packet, but it was
- * considered too early to use. Now, use it anyway.
- */
- uw16_instr = BUFSTATS_DO_RFC3389CNG_PACKET;
- }
- else if (uw16_instr != BUFSTATS_DO_RFC3389CNG_PACKET)
- {
- uw16_instr = BUFSTATS_DO_NORMAL;
- }
-
- /* reset loss counter */
- WebRtcNetEQ_ResetMcuInCallStats(inst);
- }
-
- /* Should we just reset the decoder? */
- if (uw16_instr == BUFSTAT_REINIT_DECODER)
- {
- /* Change decision to normal and flag decoder reset */
- uw16_instr = BUFSTATS_DO_NORMAL;
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0xf0ff) | DSP_CODEC_RESET;
- }
-
- /* Expand requires no new packet */
- if (uw16_instr == BUFSTATS_DO_EXPAND)
- {
-
- inst->timeStamp = dspInfo.playedOutTS;
-
- /* Have we got one descriptor left? */
- if (WebRtcNetEQ_DbIsMDCodec((enum WebRtcNetEQDecoder) inst->current_Codec)
- && (dspInfo.MD || latePacketExist))
- {
-
- if (dspInfo.lastMode != MODE_ONE_DESCRIPTOR)
- {
- /* this is the first "consecutive" one-descriptor decoding; reset counter */
- inst->one_desc = 0;
- }
- if (inst->one_desc < MAX_ONE_DESC)
- {
- /* use that one descriptor */
- inst->one_desc++; /* increase counter */
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_NORMAL_ONE_DESC;
-
- /* decrease counter since we did no Expand */
- inst->NoOfExpandCalls = WEBRTC_SPL_MAX(inst->NoOfExpandCalls - 1, 0);
- return 0;
- }
- else
- {
- /* too many consecutive one-descriptor decodings; do expand instead */
- inst->one_desc = 0; /* reset counter */
- }
-
- }
-
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff) | DSP_INSTR_EXPAND;
- return 0;
- }
-
- /* Merge is not needed if we still have a descriptor */
- if ((uw16_instr == BUFSTATS_DO_MERGE) && (dspInfo.MD != 0))
- {
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_NORMAL_ONE_DESC;
- *blockPtr = 0;
- return 0;
- }
-
- /* Do CNG without trying to extract any packets from buffer */
- if (uw16_instr == BUFSTATS_DO_RFC3389CNG_NOPACKET)
- {
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_DO_RFC3389CNG;
- *blockPtr = 0;
- return 0;
- }
-
- /* Do built-in CNG without extracting any new packets from buffer */
- if (uw16_instr == BUFSTATS_DO_INTERNAL_CNG_NOPACKET)
- {
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_DO_CODEC_INTERNAL_CNG;
- *blockPtr = 0;
- return 0;
- }
-
- /* Do DTMF without extracting any new packets from buffer */
- if (uw16_instr == BUFSTATS_DO_DTMF_ONLY)
- {
- WebRtc_UWord32 timeStampJump = 0;
-
- /* Update timestamp */
- if ((inst->BufferStat_inst.uw32_CNGplayedTS > 0) && (dspInfo.lastMode != MODE_DTMF))
- {
- /* Jump in timestamps if needed */
- timeStampJump = inst->BufferStat_inst.uw32_CNGplayedTS;
- inst->pw16_writeAddress[1] = (WebRtc_UWord16) (timeStampJump >> 16);
- inst->pw16_writeAddress[2] = (WebRtc_UWord16) (timeStampJump & 0xFFFF);
- }
-
- inst->timeStamp = dspInfo.playedOutTS + timeStampJump;
-
- inst->BufferStat_inst.uw32_CNGplayedTS = 0;
- inst->NoOfExpandCalls = 0;
-
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_DTMF_GENERATE;
- *blockPtr = 0;
- return 0;
- }
-
- if (uw16_instr == BUFSTATS_DO_ACCELERATE)
- {
- /* In order to do a Accelerate we need at least 30 ms of data */
- if (dspInfo.samplesLeft >= (3 * 80 * fs_mult))
- {
- /* Already have enough data, so we do not need to extract any more */
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_ACCELERATE;
- *blockPtr = 0;
- inst->BufferStat_inst.Automode_inst.sampleMemory
- = (WebRtc_Word32) dspInfo.samplesLeft;
- inst->BufferStat_inst.Automode_inst.prevTimeScale = 1;
- return 0;
- }
- else if ((dspInfo.samplesLeft >= (1 * 80 * fs_mult))
- && (inst->PacketBuffer_inst.packSizeSamples >= (240 * fs_mult)))
- {
- /* Avoid decoding more data as it might overflow playout buffer */
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_NORMAL;
- *blockPtr = 0;
- return 0;
- }
- else if ((dspInfo.samplesLeft < (1 * 80 * fs_mult))
- && (inst->PacketBuffer_inst.packSizeSamples >= (240 * fs_mult)))
- {
- /* For >= 30ms allow Accelerate with a decoding to avoid overflow in playout buffer */
- wantedNoOfTimeStamps = inst->timestampsPerCall;
- }
- else if (dspInfo.samplesLeft >= (2 * 80 * fs_mult))
- {
- /* We need to decode another 10 ms in order to do an Accelerate */
- wantedNoOfTimeStamps = inst->timestampsPerCall;
- }
- else
- {
- /*
- * Build up decoded data by decoding at least 20 ms of data.
- * Do not perform Accelerate yet, but wait until we only need to do one decoding.
- */
- wantedNoOfTimeStamps = 2 * inst->timestampsPerCall;
- uw16_instr = BUFSTATS_DO_NORMAL;
- }
- }
- else if (uw16_instr == BUFSTATS_DO_PREEMPTIVE_EXPAND)
- {
- /* In order to do a Preemptive Expand we need at least 30 ms of data */
- if (dspInfo.samplesLeft >= (3 * 80 * fs_mult))
- {
- /* Already have enough data, so we do not need to extract any more */
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_PREEMPTIVE_EXPAND;
- *blockPtr = 0;
- inst->BufferStat_inst.Automode_inst.sampleMemory
- = (WebRtc_Word32) dspInfo.samplesLeft;
- inst->BufferStat_inst.Automode_inst.prevTimeScale = 1;
- return 0;
- }
- else if ((dspInfo.samplesLeft >= (1 * 80 * fs_mult))
- && (inst->PacketBuffer_inst.packSizeSamples >= (240 * fs_mult)))
- {
- /*
- * Avoid decoding more data as it might overflow playout buffer;
- * still try Preemptive Expand though.
- */
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_PREEMPTIVE_EXPAND;
- *blockPtr = 0;
- inst->BufferStat_inst.Automode_inst.sampleMemory
- = (WebRtc_Word32) dspInfo.samplesLeft;
- inst->BufferStat_inst.Automode_inst.prevTimeScale = 1;
- return 0;
- }
- else if ((dspInfo.samplesLeft < (1 * 80 * fs_mult))
- && (inst->PacketBuffer_inst.packSizeSamples >= (240 * fs_mult)))
- {
- /*
- * For >= 30ms allow Preemptive Expand with a decoding to avoid overflow in
- * playout buffer
- */
- wantedNoOfTimeStamps = inst->timestampsPerCall;
- }
- else if (dspInfo.samplesLeft >= (2 * 80 * fs_mult))
- {
- /* We need to decode another 10 ms in order to do an Preemptive Expand */
- wantedNoOfTimeStamps = inst->timestampsPerCall;
- }
- else
- {
- /*
- * Build up decoded data by decoding at least 20 ms of data,
- * Still try to perform Preemptive Expand.
- */
- wantedNoOfTimeStamps = 2 * inst->timestampsPerCall;
- }
- }
- else
- {
- wantedNoOfTimeStamps = inst->timestampsPerCall;
- }
-
- /* Otherwise get data from buffer, try to get at least 10ms */
- totalTS = 0;
- oldTS = uw32_availableTS;
- if ((i_bufferpos > -1) && (uw16_instr != BUFSTATS_DO_ALTERNATIVE_PLC) && (uw16_instr
- != BUFSTATS_DO_ALTERNATIVE_PLC_INC_TS) && (uw16_instr != BUFSTATS_DO_AUDIO_REPETITION)
- && (uw16_instr != BUFSTATS_DO_AUDIO_REPETITION_INC_TS))
- {
- uw32_tmp = (uw32_availableTS - dspInfo.playedOutTS);
- inst->pw16_writeAddress[1] = (WebRtc_UWord16) (uw32_tmp >> 16);
- inst->pw16_writeAddress[2] = (WebRtc_UWord16) (uw32_tmp & 0xFFFF);
- if (inst->BufferStat_inst.w16_cngOn == CNG_OFF)
- {
- /*
- * Adjustment of TS only corresponds to an actual packet loss
- * if comfort noise is not played. If comfort noise was just played,
- * this adjustment of TS is only done to get back in sync with the
- * stream TS; no loss to report.
- */
- inst->lostTS += uw32_tmp;
- }
-
- if (uw16_instr != BUFSTATS_DO_RFC3389CNG_PACKET)
- {
- /* We are about to decode and use a non-CNG packet => CNG period is ended */
- inst->BufferStat_inst.w16_cngOn = CNG_OFF;
- }
-
- /*
- * Reset CNG timestamp as a new packet will be delivered.
- * (Also if CNG packet, since playedOutTS is updated.)
- */
- inst->BufferStat_inst.uw32_CNGplayedTS = 0;
-
- prevSeqNo = inst->PacketBuffer_inst.seqNumber[i_bufferpos];
- prevTS = inst->PacketBuffer_inst.timeStamp[i_bufferpos];
- oldPT = inst->PacketBuffer_inst.payloadType[i_bufferpos];
-
- /* clear flag bits */
- inst->pw16_writeAddress[0] = inst->pw16_writeAddress[0] & 0xFF3F;
- do
- {
- int waitingTime;
- inst->timeStamp = uw32_availableTS;
- /* Write directly to shared memory */
- temp_pkt.payload = blockPtr + 1;
- i_res = WebRtcNetEQ_PacketBufferExtract(&inst->PacketBuffer_inst, &temp_pkt,
- i_bufferpos, &waitingTime);
-
- if (i_res < 0)
- {
- /* error returned */
- return i_res;
- }
- WebRtcNetEQ_StoreWaitingTime(inst, waitingTime);
-
-#ifdef NETEQ_DELAY_LOGGING
- temp_var = NETEQ_DELAY_LOGGING_SIGNAL_DECODE;
- fwrite(&temp_var,sizeof(int),1,delay_fid2);
- fwrite(&temp_pkt.timeStamp,sizeof(WebRtc_UWord32),1,delay_fid2);
- fwrite(&dspInfo.samplesLeft, sizeof(WebRtc_UWord16), 1, delay_fid2);
-#endif
-
- *blockPtr = temp_pkt.payloadLen;
- /* set the flag if this is a redundant payload */
- if (temp_pkt.rcuPlCntr > 0)
- {
- *blockPtr = (*blockPtr) | (DSP_CODEC_RED_FLAG);
- }
- blockPtr += ((temp_pkt.payloadLen + 1) >> 1) + 1;
-
- if (i_bufferpos > -1)
- {
- /*
- * Store number of TS extracted (last extracted is assumed to be of
- * packSizeSamples).
- */
- totalTS = uw32_availableTS - oldTS + inst->PacketBuffer_inst.packSizeSamples;
- }
- /* Check what next packet is available */
- WebRtcNetEQ_PacketBufferFindLowestTimestamp(&inst->PacketBuffer_inst,
- inst->timeStamp, &uw32_availableTS, &i_bufferpos, 0, &payloadType);
-
- nextSeqNoAvail = 0;
- if ((i_bufferpos > -1) && (oldPT
- == inst->PacketBuffer_inst.payloadType[i_bufferpos]))
- {
- w16_tmp = inst->PacketBuffer_inst.seqNumber[i_bufferpos] - prevSeqNo;
- w32_tmp = inst->PacketBuffer_inst.timeStamp[i_bufferpos] - prevTS;
- if ((w16_tmp == 1) || /* Next packet */
- ((w16_tmp == 0) && (w32_tmp == inst->PacketBuffer_inst.packSizeSamples)))
- { /* or packet split into frames */
- nextSeqNoAvail = 1;
- }
- prevSeqNo = inst->PacketBuffer_inst.seqNumber[i_bufferpos];
- }
-
- }
- while ((totalTS < wantedNoOfTimeStamps) && (nextSeqNoAvail == 1));
- }
-
- if ((uw16_instr == BUFSTATS_DO_ACCELERATE)
- || (uw16_instr == BUFSTATS_DO_PREEMPTIVE_EXPAND))
- {
- /* Check that we have enough data (30ms) to do the Accelearate */
- if ((totalTS + dspInfo.samplesLeft) < WEBRTC_SPL_MUL(3,inst->timestampsPerCall)
- && (uw16_instr == BUFSTATS_DO_ACCELERATE))
- {
- /* Not enough, do normal operation instead */
- uw16_instr = BUFSTATS_DO_NORMAL;
- }
- else
- {
- inst->BufferStat_inst.Automode_inst.sampleMemory
- = (WebRtc_Word32) dspInfo.samplesLeft + totalTS;
- inst->BufferStat_inst.Automode_inst.prevTimeScale = 1;
- }
- }
-
- /* Close the data with a zero size block */
- *blockPtr = 0;
-
- /* Write data to DSP */
- switch (uw16_instr)
- {
- case BUFSTATS_DO_NORMAL:
- /* Normal with decoding included */
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_NORMAL;
- break;
- case BUFSTATS_DO_ACCELERATE:
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_ACCELERATE;
- break;
- case BUFSTATS_DO_MERGE:
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_MERGE;
- break;
- case BUFSTATS_DO_RFC3389CNG_PACKET:
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_DO_RFC3389CNG;
- break;
- case BUFSTATS_DO_ALTERNATIVE_PLC:
- inst->pw16_writeAddress[1] = 0;
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_DO_ALTERNATIVE_PLC;
- break;
- case BUFSTATS_DO_ALTERNATIVE_PLC_INC_TS:
- inst->pw16_writeAddress[1] = 0;
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_DO_ALTERNATIVE_PLC_INC_TS;
- break;
- case BUFSTATS_DO_AUDIO_REPETITION:
- inst->pw16_writeAddress[1] = 0;
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_DO_AUDIO_REPETITION;
- break;
- case BUFSTATS_DO_AUDIO_REPETITION_INC_TS:
- inst->pw16_writeAddress[1] = 0;
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_DO_AUDIO_REPETITION_INC_TS;
- break;
- case BUFSTATS_DO_PREEMPTIVE_EXPAND:
- inst->pw16_writeAddress[0] = (inst->pw16_writeAddress[0] & 0x0fff)
- | DSP_INSTR_PREEMPTIVE_EXPAND;
- break;
- default:
- return UNKNOWN_BUFSTAT_DECISION;
- }
-
- inst->timeStamp = dspInfo.playedOutTS;
- return 0;
-
-}
-
diff --git a/trunk/src/modules/audio_coding/neteq/split_and_insert.c b/trunk/src/modules/audio_coding/neteq/split_and_insert.c
deleted file mode 100644
index 03c1569..0000000
--- a/trunk/src/modules/audio_coding/neteq/split_and_insert.c
+++ /dev/null
@@ -1,141 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Split an RTP payload (if possible and suitable) and insert into packet buffer.
- */
-
-#include "mcu.h"
-
-#include <string.h>
-
-#include "signal_processing_library.h"
-
-#include "neteq_error_codes.h"
-
-int WebRtcNetEQ_SplitAndInsertPayload(RTPPacket_t *packet, PacketBuf_t *Buffer_inst,
- SplitInfo_t *split_inst, WebRtc_Word16 *flushed)
-{
-
- int i_ok;
- int len;
- int i;
- RTPPacket_t temp_packet;
- WebRtc_Word16 localFlushed = 0;
- const WebRtc_Word16 *pw16_startPayload;
- *flushed = 0;
-
- len = packet->payloadLen;
-
- /* Copy to temp packet that can be modified. */
-
- WEBRTC_SPL_MEMCPY_W8(&temp_packet,packet,sizeof(RTPPacket_t));
-
- if (split_inst->deltaBytes == NO_SPLIT)
- {
- /* Not splittable codec */
- i_ok = WebRtcNetEQ_PacketBufferInsert(Buffer_inst, packet, &localFlushed);
- *flushed |= localFlushed;
- if (i_ok < 0)
- {
- return PBUFFER_INSERT_ERROR5;
- }
- }
- else if (split_inst->deltaBytes < -10)
- {
- /* G711, PCM16B or G722, use "soft splitting" */
- int split_size = packet->payloadLen;
- int mult = WEBRTC_SPL_ABS_W32(split_inst->deltaBytes) - 10;
-
- /* Find "chunk size" >= 20 ms and < 40 ms
- * split_inst->deltaTime in this case contains the number of bytes per
- * timestamp unit times 2
- */
- while (split_size >= ((80 << split_inst->deltaTime) * mult))
- {
- split_size >>= 1;
- }
-
- /* Make the size an even value. */
- if (split_size > 1)
- {
- split_size >>= 1;
- split_size *= 2;
- }
-
- temp_packet.payloadLen = split_size;
- pw16_startPayload = temp_packet.payload;
- i = 0;
- while (len >= (2 * split_size))
- {
- /* insert every chunk */
- i_ok = WebRtcNetEQ_PacketBufferInsert(Buffer_inst, &temp_packet, &localFlushed);
- *flushed |= localFlushed;
- temp_packet.timeStamp += ((2 * split_size) >> split_inst->deltaTime);
- i++;
- temp_packet.payload = &(pw16_startPayload[(i * split_size) >> 1]);
- temp_packet.starts_byte1 = temp_packet.starts_byte1 ^ (split_size & 0x1);
-
- len -= split_size;
- if (i_ok < 0)
- {
- return PBUFFER_INSERT_ERROR1;
- }
- }
-
- /* Insert the rest */
- temp_packet.payloadLen = len;
- i_ok = WebRtcNetEQ_PacketBufferInsert(Buffer_inst, &temp_packet, &localFlushed);
- *flushed |= localFlushed;
- if (i_ok < 0)
- {
- return PBUFFER_INSERT_ERROR2;
- }
- }
- else
- {
- /* Frame based codec, use hard splitting. */
- i = 0;
- pw16_startPayload = temp_packet.payload;
- while (len >= split_inst->deltaBytes)
- {
-
- temp_packet.payloadLen = split_inst->deltaBytes;
- i_ok = WebRtcNetEQ_PacketBufferInsert(Buffer_inst, &temp_packet, &localFlushed);
- *flushed |= localFlushed;
- i++;
- temp_packet.payload = &(pw16_startPayload[(i * split_inst->deltaBytes) >> 1]);
- temp_packet.timeStamp += split_inst->deltaTime;
- temp_packet.starts_byte1 = temp_packet.starts_byte1 ^ (split_inst->deltaBytes
- & 0x1);
-
- if (i_ok < 0)
- {
- return PBUFFER_INSERT_ERROR3;
- }
- len -= split_inst->deltaBytes;
-
- }
- if (len > 0)
- {
- /* Must be a either an error or a SID frame at the end of the packet. */
- temp_packet.payloadLen = len;
- i_ok = WebRtcNetEQ_PacketBufferInsert(Buffer_inst, &temp_packet, &localFlushed);
- *flushed |= localFlushed;
- if (i_ok < 0)
- {
- return PBUFFER_INSERT_ERROR4;
- }
- }
- }
-
- return 0;
-}
-
diff --git a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.cc b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.cc
deleted file mode 100644
index 64bf508..0000000
--- a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.cc
+++ /dev/null
@@ -1,657 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "NETEQTEST_CodecClass.h"
-
-#include <stdlib.h> // exit
-
-#include "webrtc_neteq_help_macros.h"
-
-NETEQTEST_Decoder::NETEQTEST_Decoder(enum WebRtcNetEQDecoder type, WebRtc_UWord16 fs, const char * name, WebRtc_UWord8 pt)
-:
-_decoder(NULL),
-_decoderType(type),
-_pt(pt),
-_fs(fs),
-_name(name)
-{
-}
-
-int NETEQTEST_Decoder::loadToNetEQ(NETEQTEST_NetEQClass & neteq, WebRtcNetEQ_CodecDef & codecInst)
-{
- SET_CODEC_PAR(codecInst, _decoderType, _pt, _decoder, _fs);
- int err = neteq.loadCodec(codecInst);
-
- if (err)
- {
- printf("Error loading codec %s into NetEQ database\n", _name.c_str());
- }
-
- return(err);
-}
-
-
-// iSAC
-#ifdef CODEC_ISAC
-#include "isac.h"
-
-decoder_iSAC::decoder_iSAC(WebRtc_UWord8 pt)
-:
-NETEQTEST_Decoder(kDecoderISAC, 16000, "iSAC", pt)
-{
- WebRtc_Word16 err = WebRtcIsac_Create((ISACStruct **) &_decoder);
- if (err)
- {
- exit(EXIT_FAILURE);
- }
-
- WebRtcIsac_EncoderInit((ISACStruct *) _decoder, 0);
- WebRtcIsac_SetDecSampRate((ISACStruct *) _decoder, kIsacWideband);
-}
-
-
-decoder_iSAC::~decoder_iSAC()
-{
- if (_decoder)
- {
- WebRtcIsac_Free((ISACStruct *) _decoder);
- _decoder = NULL;
- }
-}
-
-
-int decoder_iSAC::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_ISAC_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-
-}
-#endif
-
-#ifdef CODEC_ISAC_SWB
-decoder_iSACSWB::decoder_iSACSWB(WebRtc_UWord8 pt)
-:
-NETEQTEST_Decoder(kDecoderISACswb, 32000, "iSAC swb", pt)
-{
- WebRtc_Word16 err = WebRtcIsac_Create((ISACStruct **) &_decoder);
- if (err)
- {
- exit(EXIT_FAILURE);
- }
-
- WebRtcIsac_EncoderInit((ISACStruct *) _decoder, 0);
- WebRtcIsac_SetDecSampRate((ISACStruct *) _decoder, kIsacSuperWideband);
-}
-
-decoder_iSACSWB::~decoder_iSACSWB()
-{
- if (_decoder)
- {
- WebRtcIsac_Free((ISACStruct *) _decoder);
- _decoder = NULL;
- }
-}
-
-int decoder_iSACSWB::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_ISACSWB_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-
-}
-#endif
-
-// PCM u/A
-#ifdef CODEC_G711
-#include "g711_interface.h"
-
-decoder_PCMU::decoder_PCMU(WebRtc_UWord8 pt)
-:
-NETEQTEST_Decoder(kDecoderPCMu, 8000, "G.711-u", pt)
-{
- // no state to crate or init
-}
-
-int decoder_PCMU::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_PCMU_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-
-}
-
-decoder_PCMA::decoder_PCMA(WebRtc_UWord8 pt)
-:
-NETEQTEST_Decoder(kDecoderPCMa, 8000, "G.711-A", pt)
-{
- // no state to crate or init
-}
-
-int decoder_PCMA::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_PCMA_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-// Linear PCM16b
-#if (defined(CODEC_PCM16B) || defined(CODEC_PCM16B_WB) || \
- defined(CODEC_PCM16B_32KHZ) || defined(CODEC_PCM16B_48KHZ))
-#include "pcm16b.h"
-#endif
-
-#ifdef CODEC_PCM16B
-int decoder_PCM16B_NB::loadToNetEQ(NETEQTEST_NetEQClass &neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_PCM16B_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_PCM16B_WB
-int decoder_PCM16B_WB::loadToNetEQ(NETEQTEST_NetEQClass &neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_PCM16B_WB_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_PCM16B_32KHZ
-int decoder_PCM16B_SWB32::loadToNetEQ(NETEQTEST_NetEQClass &neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_PCM16B_SWB32_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_PCM16B_48KHZ
-int decoder_PCM16B_SWB48::loadToNetEQ(NETEQTEST_NetEQClass &neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_PCM16B_SWB48_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_ILBC
-#include "ilbc.h"
-decoder_ILBC::decoder_ILBC(WebRtc_UWord8 pt)
-:
-NETEQTEST_Decoder(kDecoderILBC, 8000, "iLBC", pt)
-{
- WebRtc_Word16 err = WebRtcIlbcfix_DecoderCreate((iLBC_decinst_t **) &_decoder);
- if (err)
- {
- exit(EXIT_FAILURE);
- }
-}
-
-decoder_ILBC::~decoder_ILBC()
-{
- WebRtcIlbcfix_DecoderFree((iLBC_decinst_t *) _decoder);
-}
-
-int decoder_ILBC::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_ILBC_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_G729
-#include "G729Interface.h"
-decoder_G729::decoder_G729(WebRtc_UWord8 pt)
-:
-NETEQTEST_Decoder(kDecoderG729, 8000, "G.729", pt)
-{
- WebRtc_Word16 err = WebRtcG729_CreateDec((G729_decinst_t **) &_decoder);
- if (err)
- {
- exit(EXIT_FAILURE);
- }
-}
-
-decoder_G729::~decoder_G729()
-{
- WebRtcG729_FreeDec((G729_decinst_t *) _decoder);
-}
-
-int decoder_G729::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_G729_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_G729_1
-#include "G729_1Interface.h"
-decoder_G729_1::decoder_G729_1(WebRtc_UWord8 pt)
-:
-NETEQTEST_Decoder(kDecoderG729_1, 16000, "G.729.1", pt)
-{
- WebRtc_Word16 err = WebRtcG7291_Create((G729_1_inst_t **) &_decoder);
- if (err)
- {
- exit(EXIT_FAILURE);
- }
-}
-
-decoder_G729_1::~decoder_G729_1()
-{
- WebRtcG7291_Free((G729_1_inst_t *) _decoder);
-}
-
-int decoder_G729_1::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_G729_1_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_G722
-#include "g722_interface.h"
-decoder_G722::decoder_G722(WebRtc_UWord8 pt)
-:
-NETEQTEST_Decoder(kDecoderG722, 16000, "G.722", pt)
-{
- WebRtc_Word16 err = WebRtcG722_CreateDecoder((G722DecInst **) &_decoder);
- if (err)
- {
- exit(EXIT_FAILURE);
- }
-}
-
-decoder_G722::~decoder_G722()
-{
- WebRtcG722_FreeDecoder((G722DecInst *) _decoder);
-}
-
-int decoder_G722::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_G722_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#if (defined(CODEC_G722_1_16) || defined(CODEC_G722_1_24) || \
- defined(CODEC_G722_1_32) || defined(CODEC_G722_1C_24) || \
- defined(CODEC_G722_1C_32) || defined(CODEC_G722_1C_48))
-#include "G722_1Interface.h"
-#endif
-
-#ifdef CODEC_G722_1_16
-decoder_G722_1_16::decoder_G722_1_16(WebRtc_UWord8 pt)
-:
-NETEQTEST_Decoder(kDecoderG722_1_16, 16000, "G.722.1 (16 kbps)", pt)
-{
- if (WebRtcG7221_CreateDec16((G722_1_16_decinst_t **) &_decoder))
- {
- exit(EXIT_FAILURE);
- }
-}
-
-decoder_G722_1_16::~decoder_G722_1_16()
-{
- WebRtcG7221_FreeDec16((G722_1_16_decinst_t *) _decoder);
-}
-
-int decoder_G722_1_16::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_G722_1_16_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_G722_1_24
-decoder_G722_1_24::decoder_G722_1_24(WebRtc_UWord8 pt)
-:
-NETEQTEST_Decoder(kDecoderG722_1_24, 16000, "G.722.1 (24 kbps)", pt)
-{
- if (WebRtcG7221_CreateDec24((G722_1_24_decinst_t **) &_decoder))
- {
- exit(EXIT_FAILURE);
- }
-}
-
-decoder_G722_1_24::~decoder_G722_1_24()
-{
- WebRtcG7221_FreeDec24((G722_1_24_decinst_t *) _decoder);
-}
-
-int decoder_G722_1_24::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_G722_1_24_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_G722_1_32
-decoder_G722_1_32::decoder_G722_1_32(WebRtc_UWord8 pt)
-:
-NETEQTEST_Decoder(kDecoderG722_1_32, 16000, "G.722.1 (32 kbps)", pt)
-{
- if (WebRtcG7221_CreateDec32((G722_1_32_decinst_t **) &_decoder))
- {
- exit(EXIT_FAILURE);
- }
-}
-
-decoder_G722_1_32::~decoder_G722_1_32()
-{
- WebRtcG7221_FreeDec32((G722_1_32_decinst_t *) _decoder);
-}
-
-int decoder_G722_1_32::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_G722_1_32_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_G722_1C_24
-decoder_G722_1C_24::decoder_G722_1C_24(WebRtc_UWord8 pt)
-:
-NETEQTEST_Decoder(kDecoderG722_1C_24, 32000, "G.722.1C (24 kbps)", pt)
-{
- if (WebRtcG7221C_CreateDec24((G722_1C_24_decinst_t **) &_decoder))
- exit(EXIT_FAILURE);
-}
-
-decoder_G722_1C_24::~decoder_G722_1C_24()
-{
- WebRtcG7221C_FreeDec24((G722_1C_24_decinst_t *) _decoder);
-}
-
-int decoder_G722_1C_24::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_G722_1C_24_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_G722_1C_32
-decoder_G722_1C_32::decoder_G722_1C_32(WebRtc_UWord8 pt)
-:
-NETEQTEST_Decoder(kDecoderG722_1C_32, 32000, "G.722.1C (32 kbps)", pt)
-{
- if (WebRtcG7221C_CreateDec32((G722_1C_32_decinst_t **) &_decoder))
- exit(EXIT_FAILURE);
-}
-
-decoder_G722_1C_32::~decoder_G722_1C_32()
-{
- WebRtcG7221C_FreeDec32((G722_1C_32_decinst_t *) _decoder);
-}
-
-int decoder_G722_1C_32::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_G722_1C_32_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_G722_1C_48
-decoder_G722_1C_48::decoder_G722_1C_48(WebRtc_UWord8 pt)
-:
-NETEQTEST_Decoder(kDecoderG722_1C_48, 32000, "G.722.1C (48 kbps)", pt)
-{
- if (WebRtcG7221C_CreateDec48((G722_1C_48_decinst_t **) &_decoder))
- exit(EXIT_FAILURE);
-}
-
-decoder_G722_1C_48::~decoder_G722_1C_48()
-{
- WebRtcG7221C_FreeDec48((G722_1C_48_decinst_t *) _decoder);
-}
-
-int decoder_G722_1C_48::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_G722_1C_48_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_AMR
-#include "AMRInterface.h"
-#include "AMRCreation.h"
-decoder_AMR::decoder_AMR(WebRtc_UWord8 pt)
-:
-NETEQTEST_Decoder(kDecoderAMR, 8000, "AMR", pt)
-{
- if (WebRtcAmr_CreateDec((AMR_decinst_t **) &_decoder))
- exit(EXIT_FAILURE);
-
- WebRtcAmr_DecodeBitmode((AMR_decinst_t *) _decoder, AMRBandwidthEfficient);
-}
-
-decoder_AMR::~decoder_AMR()
-{
- WebRtcAmr_FreeDec((AMR_decinst_t *) _decoder);
-}
-
-int decoder_AMR::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_AMR_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_AMRWB
-#include "AMRWBInterface.h"
-#include "AMRWBCreation.h"
-decoder_AMRWB::decoder_AMRWB(WebRtc_UWord8 pt)
-:
-NETEQTEST_Decoder(kDecoderAMRWB, 16000, "AMR wb", pt)
-{
- if (WebRtcAmrWb_CreateDec((AMRWB_decinst_t **) &_decoder))
- exit(EXIT_FAILURE);
-
- WebRtcAmrWb_DecodeBitmode((AMRWB_decinst_t *) _decoder, AMRBandwidthEfficient);
-}
-
-decoder_AMRWB::~decoder_AMRWB()
-{
- WebRtcAmrWb_FreeDec((AMRWB_decinst_t *) _decoder);
-}
-
-int decoder_AMRWB::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_AMRWB_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_GSMFR
-#include "GSMFRInterface.h"
-#include "GSMFRCreation.h"
-decoder_GSMFR::decoder_GSMFR(WebRtc_UWord8 pt)
-:
-NETEQTEST_Decoder(kDecoderGSMFR, 8000, "GSM-FR", pt)
-{
- if (WebRtcGSMFR_CreateDec((GSMFR_decinst_t **) &_decoder))
- exit(EXIT_FAILURE);
-}
-
-decoder_GSMFR::~decoder_GSMFR()
-{
- WebRtcGSMFR_FreeDec((GSMFR_decinst_t *) _decoder);
-}
-
-int decoder_GSMFR::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_GSMFR_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#if (defined(CODEC_SPEEX_8) || defined (CODEC_SPEEX_16))
-#include "SpeexInterface.h"
-decoder_SPEEX::decoder_SPEEX(WebRtc_UWord8 pt, WebRtc_UWord16 fs)
-:
-NETEQTEST_Decoder(fs == 8000 ? kDecoderSPEEX_8 : kDecoderSPEEX_16,
- fs, "SPEEX " + fs/1000, pt)
-{
- if (fs != 8000 && fs != 16000)
- throw std::exception("Wrong sample rate for SPEEX");
-
- if (WebRtcSpeex_CreateDec((SPEEX_decinst_t **) &_decoder, fs, 1))
- exit(EXIT_FAILURE);
-}
-
-decoder_SPEEX::~decoder_SPEEX()
-{
- WebRtcSpeex_FreeDec((SPEEX_decinst_t *) _decoder);
-}
-
-int decoder_SPEEX::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_SPEEX_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_CELT_32
-#include "celt_interface.h"
-decoder_CELT::decoder_CELT(WebRtc_UWord8 pt, WebRtc_UWord16 fs)
-:
-NETEQTEST_Decoder(kDecoderCELT_32, fs, "CELT", pt)
-{
- if (WebRtcCelt_CreateDec((CELT_decinst_t **) &_decoder, 1))
- exit(EXIT_FAILURE);
-}
-
-decoder_CELT::~decoder_CELT()
-{
- WebRtcCelt_FreeDec((CELT_decinst_t *) _decoder);
-}
-
-int decoder_CELT::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_CELT_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_RED
-int decoder_RED::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_RED_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#ifdef CODEC_ATEVENT_DECODE
-int decoder_AVT::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_AVT_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
-
-#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
- defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
-#include "webrtc_cng.h"
-decoder_CNG::decoder_CNG(WebRtc_UWord8 pt, WebRtc_UWord16 fs)
-:
-NETEQTEST_Decoder(kDecoderCNG, fs, "CNG " + fs/1000, pt)
-{
- if (fs != 8000 && fs != 16000 && fs != 32000 && fs != 48000)
- exit(EXIT_FAILURE);
-
- if (WebRtcCng_CreateDec((CNG_dec_inst **) &_decoder))
- exit(EXIT_FAILURE);
-}
-
-decoder_CNG::~decoder_CNG()
-{
- WebRtcCng_FreeDec((CNG_dec_inst *) _decoder);
-}
-
-int decoder_CNG::loadToNetEQ(NETEQTEST_NetEQClass & neteq)
-{
- WebRtcNetEQ_CodecDef codecInst;
-
- SET_CNG_FUNCTIONS(codecInst);
-
- return(NETEQTEST_Decoder::loadToNetEQ(neteq, codecInst));
-}
-#endif
diff --git a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h
deleted file mode 100644
index ff49049..0000000
--- a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h
+++ /dev/null
@@ -1,301 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef NETEQTEST_CODECCLASS_H
-#define NETEQTEST_CODECCLASS_H
-
-#include <string>
-#include <string.h>
-
-#include "typedefs.h"
-#include "webrtc_neteq.h"
-#include "NETEQTEST_NetEQClass.h"
-
-class NETEQTEST_Decoder
-{
-public:
- NETEQTEST_Decoder(enum WebRtcNetEQDecoder type, WebRtc_UWord16 fs, const char * name, WebRtc_UWord8 pt = 0);
- virtual ~NETEQTEST_Decoder() {};
-
- virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq) = 0;
-
- int getName(char * name, int maxLen) const { strncpy( name, _name.c_str(), maxLen ); return 0;};
-
- void setPT(WebRtc_UWord8 pt) { _pt = pt; };
- WebRtc_UWord16 getFs() const { return (_fs); };
- enum WebRtcNetEQDecoder getType() const { return (_decoderType); };
- WebRtc_UWord8 getPT() const { return (_pt); };
-
-protected:
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq, WebRtcNetEQ_CodecDef & codecInst);
-
- void * _decoder;
- enum WebRtcNetEQDecoder _decoderType;
- WebRtc_UWord8 _pt;
- WebRtc_UWord16 _fs;
- std::string _name;
-
-private:
-};
-
-
-class decoder_iSAC : public NETEQTEST_Decoder
-{
-public:
- decoder_iSAC(WebRtc_UWord8 pt = 0);
- virtual ~decoder_iSAC();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-
-class decoder_iSACSWB : public NETEQTEST_Decoder
-{
-public:
- decoder_iSACSWB(WebRtc_UWord8 pt = 0);
- virtual ~decoder_iSACSWB();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-
-class decoder_PCMU : public NETEQTEST_Decoder
-{
-public:
- decoder_PCMU(WebRtc_UWord8 pt = 0);
- virtual ~decoder_PCMU() {};
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-
-class decoder_PCMA : public NETEQTEST_Decoder
-{
-public:
- decoder_PCMA(WebRtc_UWord8 pt = 0);
- virtual ~decoder_PCMA() {};
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-class decoder_PCM16B_NB : public NETEQTEST_Decoder
-{
-public:
- decoder_PCM16B_NB(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderPCM16B, 8000, "PCM16 nb", pt) {};
- virtual ~decoder_PCM16B_NB() {};
- virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-class decoder_PCM16B_WB : public NETEQTEST_Decoder
-{
-public:
- decoder_PCM16B_WB(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderPCM16Bwb, 16000, "PCM16 wb", pt) {};
- virtual ~decoder_PCM16B_WB() {};
- virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-class decoder_PCM16B_SWB32 : public NETEQTEST_Decoder
-{
-public:
- decoder_PCM16B_SWB32(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderPCM16Bswb32kHz, 32000, "PCM16 swb32", pt) {};
- virtual ~decoder_PCM16B_SWB32() {};
- virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-class decoder_PCM16B_SWB48 : public NETEQTEST_Decoder
-{
-public:
- decoder_PCM16B_SWB48(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderPCM16Bswb48kHz, 48000, "PCM16 swb48", pt) {};
- virtual ~decoder_PCM16B_SWB48() {};
- virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-
-class decoder_ILBC : public NETEQTEST_Decoder
-{
-public:
- decoder_ILBC(WebRtc_UWord8 pt = 0);
- virtual ~decoder_ILBC();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-
-class decoder_G729 : public NETEQTEST_Decoder
-{
-public:
- decoder_G729(WebRtc_UWord8 pt = 0);
- virtual ~decoder_G729();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-class decoder_G729_1 : public NETEQTEST_Decoder
-{
-public:
- decoder_G729_1(WebRtc_UWord8 pt = 0);
- virtual ~decoder_G729_1();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-
-class decoder_G722 : public NETEQTEST_Decoder
-{
-public:
- decoder_G722(WebRtc_UWord8 pt = 0);
- virtual ~decoder_G722();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-
-class decoder_G722_1_16 : public NETEQTEST_Decoder
-{
-public:
- decoder_G722_1_16(WebRtc_UWord8 pt = 0);
- virtual ~decoder_G722_1_16();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-class decoder_G722_1_24 : public NETEQTEST_Decoder
-{
-public:
- decoder_G722_1_24(WebRtc_UWord8 pt = 0);
- virtual ~decoder_G722_1_24();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-class decoder_G722_1_32 : public NETEQTEST_Decoder
-{
-public:
- decoder_G722_1_32(WebRtc_UWord8 pt = 0);
- virtual ~decoder_G722_1_32();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-
-class decoder_G722_1C_24 : public NETEQTEST_Decoder
-{
-public:
- decoder_G722_1C_24(WebRtc_UWord8 pt = 0);
- virtual ~decoder_G722_1C_24();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-class decoder_G722_1C_32 : public NETEQTEST_Decoder
-{
-public:
- decoder_G722_1C_32(WebRtc_UWord8 pt = 0);
- virtual ~decoder_G722_1C_32();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-class decoder_G722_1C_48 : public NETEQTEST_Decoder
-{
-public:
- decoder_G722_1C_48(WebRtc_UWord8 pt = 0);
- virtual ~decoder_G722_1C_48();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-
-class decoder_AMR : public NETEQTEST_Decoder
-{
-public:
- decoder_AMR(WebRtc_UWord8 pt = 0);
- virtual ~decoder_AMR();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-class decoder_AMRWB : public NETEQTEST_Decoder
-{
-public:
- decoder_AMRWB(WebRtc_UWord8 pt = 0);
- virtual ~decoder_AMRWB();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-class decoder_GSMFR : public NETEQTEST_Decoder
-{
-public:
- decoder_GSMFR(WebRtc_UWord8 pt = 0);
- virtual ~decoder_GSMFR();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-class decoder_G726 : public NETEQTEST_Decoder
-{
-public:
- //virtual decoder_G726(WebRtc_UWord8 pt = 0) = 0;
- decoder_G726(enum WebRtcNetEQDecoder type, const char * name, WebRtc_UWord8 pt = 0);
- virtual ~decoder_G726();
- virtual int loadToNetEQ(NETEQTEST_NetEQClass & neteq) = 0;
-};
-
-class decoder_G726_16 : public decoder_G726
-{
-public:
- decoder_G726_16(WebRtc_UWord8 pt = 0) : decoder_G726(kDecoderG726_16, "G.726 (16 kbps)", pt) {};
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-class decoder_G726_24 : public decoder_G726
-{
-public:
- decoder_G726_24(WebRtc_UWord8 pt = 0) : decoder_G726(kDecoderG726_24, "G.726 (24 kbps)", pt) {};
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-class decoder_G726_32 : public decoder_G726
-{
-public:
- decoder_G726_32(WebRtc_UWord8 pt = 0) : decoder_G726(kDecoderG726_32, "G.726 (32 kbps)", pt) {};
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-class decoder_G726_40 : public decoder_G726
-{
-public:
- decoder_G726_40(WebRtc_UWord8 pt = 0) : decoder_G726(kDecoderG726_40, "G.726 (40 kbps)", pt) {};
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-class decoder_SPEEX : public NETEQTEST_Decoder
-{
-public:
- decoder_SPEEX(WebRtc_UWord8 pt = 0, WebRtc_UWord16 fs = 8000);
- virtual ~decoder_SPEEX();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-class decoder_CELT : public NETEQTEST_Decoder
-{
-public:
- decoder_CELT(WebRtc_UWord8 pt = 0, WebRtc_UWord16 fs = 32000);
- virtual ~decoder_CELT();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-class decoder_RED : public NETEQTEST_Decoder
-{
-public:
- decoder_RED(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderRED, 8000, "RED", pt) {};
- virtual ~decoder_RED() {};
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-class decoder_AVT : public NETEQTEST_Decoder
-{
-public:
- decoder_AVT(WebRtc_UWord8 pt = 0) : NETEQTEST_Decoder(kDecoderAVT, 8000, "AVT", pt) {};
- virtual ~decoder_AVT() {};
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-
-class decoder_CNG : public NETEQTEST_Decoder
-{
-public:
- decoder_CNG(WebRtc_UWord8 pt = 0, WebRtc_UWord16 fs = 8000);
- virtual ~decoder_CNG();
- int loadToNetEQ(NETEQTEST_NetEQClass & neteq);
-};
-
-#endif //NETEQTEST_CODECCLASS_H
diff --git a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.cc b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.cc
deleted file mode 100644
index 2e60658..0000000
--- a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.cc
+++ /dev/null
@@ -1,384 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <memory.h>
-
-#include "NETEQTEST_NetEQClass.h"
-
-
-NETEQTEST_NetEQClass::NETEQTEST_NetEQClass()
- :
- _inst(NULL),
- _instMem(NULL),
- _bufferMem(NULL),
- _preparseRTP(false),
- _fsmult(1),
- _isMaster(true)
-{
-#ifdef WINDOWS_TIMING
- _totTimeRecIn.QuadPart = 0;
- _totTimeRecOut.QuadPart = 0;
-#endif
-}
-
-NETEQTEST_NetEQClass::NETEQTEST_NetEQClass(enum WebRtcNetEQDecoder *usedCodec, int noOfCodecs,
- WebRtc_UWord16 fs, WebRtcNetEQNetworkType nwType)
- :
- _inst(NULL),
- _instMem(NULL),
- _bufferMem(NULL),
- _preparseRTP(false),
- _fsmult(1),
- _isMaster(true)
-{
-#ifdef WINDOWS_TIMING
- _totTimeRecIn.QuadPart = 0;
- _totTimeRecOut.QuadPart = 0;
-#endif
-
- if (assign() == 0)
- {
- if (init(fs) == 0)
- {
- assignBuffer(usedCodec, noOfCodecs, nwType);
- }
- }
-}
-
-
-NETEQTEST_NetEQClass::~NETEQTEST_NetEQClass()
-{
- if (_instMem)
- {
- delete [] _instMem;
- _instMem = NULL;
- }
-
- if (_bufferMem)
- {
- delete [] _bufferMem;
- _bufferMem = NULL;
- }
-
- _inst = NULL;
-}
-
-int NETEQTEST_NetEQClass::assign()
-{
- int memSize;
-
- WebRtcNetEQ_AssignSize(&memSize);
-
- if (_instMem)
- {
- delete [] _instMem;
- _instMem = NULL;
- }
-
- _instMem = new WebRtc_Word8[memSize];
-
- int ret = WebRtcNetEQ_Assign(&_inst, _instMem);
-
- if (ret)
- {
- printError();
- }
-
- return (ret);
-}
-
-
-int NETEQTEST_NetEQClass::init(WebRtc_UWord16 fs)
-{
- int ret;
-
- if (!_inst)
- {
- // not assigned
- ret = assign();
-
- if (ret != 0)
- {
- printError();
- return (ret);
- }
- }
-
- ret = WebRtcNetEQ_Init(_inst, fs);
-
- if (ret != 0)
- {
- printError();
- }
-
- return (ret);
-
-}
-
-
-int NETEQTEST_NetEQClass::assignBuffer(enum WebRtcNetEQDecoder *usedCodec, int noOfCodecs, WebRtcNetEQNetworkType nwType)
-{
- int numPackets, memSize, ret;
-
- if (!_inst)
- {
- // not assigned
- ret = assign();
-
- if (ret != 0)
- {
- printError();
- return (ret);
- }
-
- ret = init();
-
- if (ret != 0)
- {
- printError();
- return (ret);
- }
- }
-
- ret = WebRtcNetEQ_GetRecommendedBufferSize(_inst, usedCodec, noOfCodecs, nwType, &numPackets, &memSize);
-
- if (ret != 0)
- {
- printError();
- return (ret);
- }
-
- if (_bufferMem)
- {
- delete [] _bufferMem;
- _bufferMem = NULL;
- }
-
- _bufferMem = new WebRtc_Word8[memSize];
-
- memset(_bufferMem, -1, memSize);
-
- ret = WebRtcNetEQ_AssignBuffer(_inst, numPackets, _bufferMem, memSize);
-
- if (ret != 0)
- {
- printError();
- }
-
- return (ret);
-}
-
-int NETEQTEST_NetEQClass::loadCodec(WebRtcNetEQ_CodecDef &codecInst)
-{
- int err = WebRtcNetEQ_CodecDbAdd(_inst, &codecInst);
-
- if (err)
- {
- printError();
- }
-
- return (err);
-}
-
-void NETEQTEST_NetEQClass::printError()
-{
- if (_inst)
- {
- int errorCode = WebRtcNetEQ_GetErrorCode(_inst);
-
- if (errorCode)
- {
- char errorName[WEBRTC_NETEQ_MAX_ERROR_NAME];
-
- WebRtcNetEQ_GetErrorName(errorCode, errorName, WEBRTC_NETEQ_MAX_ERROR_NAME);
-
- printf("Error %i: %s\n", errorCode, errorName);
- }
- }
-}
-
-void NETEQTEST_NetEQClass::printError(NETEQTEST_RTPpacket &rtp)
-{
- // print regular error info
- printError();
-
- // print extra info from packet
- printf("\tRTP: TS=%u, SN=%u, PT=%u, M=%i, len=%i\n",
- rtp.timeStamp(), rtp.sequenceNumber(), rtp.payloadType(),
- rtp.markerBit(), rtp.payloadLen());
-
-}
-
-int NETEQTEST_NetEQClass::recIn(NETEQTEST_RTPpacket &rtp)
-{
-
- int err;
-#ifdef WINDOWS_TIMING
- LARGE_INTEGER countA, countB;
-#endif
-
- if (_preparseRTP)
- {
- WebRtcNetEQ_RTPInfo rtpInfo;
- // parse RTP header
- rtp.parseHeader(rtpInfo);
-
-#ifdef WINDOWS_TIMING
- QueryPerformanceCounter(&countA); // get start count for processor
-#endif
-
- err = WebRtcNetEQ_RecInRTPStruct(_inst, &rtpInfo, rtp.payload(), rtp.payloadLen(), rtp.time() * _fsmult * 8);
-
-#ifdef WINDOWS_TIMING
- QueryPerformanceCounter(&countB); // get stop count for processor
- _totTimeRecIn.QuadPart += (countB.QuadPart - countA.QuadPart);
-#endif
-
- }
- else
- {
-
-#ifdef WINDOWS_TIMING
- QueryPerformanceCounter(&countA); // get start count for processor
-#endif
-
- err = WebRtcNetEQ_RecIn(_inst, (WebRtc_Word16 *) rtp.datagram(), rtp.dataLen(), rtp.time() * _fsmult * 8);
-
-#ifdef WINDOWS_TIMING
- QueryPerformanceCounter(&countB); // get stop count for processor
- _totTimeRecIn.QuadPart += (countB.QuadPart - countA.QuadPart);
-#endif
-
- }
-
- if (err)
- {
- printError(rtp);
- }
-
- return (err);
-
-}
-
-
-WebRtc_Word16 NETEQTEST_NetEQClass::recOut(WebRtc_Word16 *outData, void *msInfo, enum WebRtcNetEQOutputType *outputType)
-{
- int err;
- WebRtc_Word16 outLen = 0;
-#ifdef WINDOWS_TIMING
- LARGE_INTEGER countA, countB;
-#endif
-
-#ifdef WINDOWS_TIMING
- QueryPerformanceCounter(&countA); // get start count for processor
-#endif
-
- if (!msInfo)
- {
- // no msInfo given, do mono mode
- err = WebRtcNetEQ_RecOut(_inst, outData, &outLen);
- }
- else
- {
- // master/slave mode
- err = WebRtcNetEQ_RecOutMasterSlave(_inst, outData, &outLen, msInfo, static_cast<WebRtc_Word16>(_isMaster));
- }
-
-#ifdef WINDOWS_TIMING
- QueryPerformanceCounter(&countB); // get stop count for processor
- _totTimeRecOut.QuadPart += (countB.QuadPart - countA.QuadPart);
-#endif
-
- if (err)
- {
- printError();
- }
- else
- {
- int newfsmult = static_cast<int>(outLen / 80);
-
- if (newfsmult != _fsmult)
- {
-#ifdef NETEQTEST_PRINT_WARNINGS
- printf("Warning: output sample rate changed\n");
-#endif // NETEQTEST_PRINT_WARNINGS
- _fsmult = newfsmult;
- }
- }
-
- if (outputType != NULL)
- {
- err = WebRtcNetEQ_GetSpeechOutputType(_inst, outputType);
-
- if (err)
- {
- printError();
- }
- }
-
- return (outLen);
-}
-
-
-WebRtc_UWord32 NETEQTEST_NetEQClass::getSpeechTimeStamp()
-{
-
- WebRtc_UWord32 ts = 0;
- int err;
-
- err = WebRtcNetEQ_GetSpeechTimeStamp(_inst, &ts);
-
- if (err)
- {
- printError();
- ts = 0;
- }
-
- return (ts);
-
-}
-
-WebRtcNetEQOutputType NETEQTEST_NetEQClass::getOutputType() {
- WebRtcNetEQOutputType type;
-
- int err = WebRtcNetEQ_GetSpeechOutputType(_inst, &type);
- if (err)
- {
- printError();
- type = kOutputNormal;
- }
- return (type);
-}
-
-//NETEQTEST_NetEQVector::NETEQTEST_NetEQVector(int numChannels)
-//:
-//channels(numChannels, new NETEQTEST_NetEQClass())
-//{
-// //for (int i = 0; i < numChannels; i++)
-// //{
-// // channels.push_back(new NETEQTEST_NetEQClass());
-// //}
-//}
-//
-//NETEQTEST_NetEQVector::NETEQTEST_NetEQVector(int numChannels, enum WebRtcNetEQDecoder *usedCodec, int noOfCodecs,
-// WebRtc_UWord16 fs, WebRtcNetEQNetworkType nwType)
-// :
-//channels(numChannels, new NETEQTEST_NetEQClass(usedCodec, noOfCodecs, fs, nwType))
-//{
-// //for (int i = 0; i < numChannels; i++)
-// //{
-// // channels.push_back(new NETEQTEST_NetEQClass(usedCodec, noOfCodecs, fs, nwType));
-// //}
-//}
-//
-//NETEQTEST_NetEQVector::~NETEQTEST_NetEQVector()
-//{
-//}
-
diff --git a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.h b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.h
deleted file mode 100644
index c425b58..0000000
--- a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.h
+++ /dev/null
@@ -1,92 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef NETEQTEST_NETEQCLASS_H
-#define NETEQTEST_NETEQCLASS_H
-
-#include <stdio.h>
-#include <vector>
-
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_internal.h"
-
-#include "NETEQTEST_RTPpacket.h"
-
-#ifdef WIN32
-#define WINDOWS_TIMING // complexity measurement only implemented for windows
-//TODO(hlundin):Add complexity testing for Linux.
-#include <windows.h>
-#endif
-
-class NETEQTEST_NetEQClass
-{
-public:
- NETEQTEST_NetEQClass();
- NETEQTEST_NetEQClass(enum WebRtcNetEQDecoder *usedCodec, int noOfCodecs,
- WebRtc_UWord16 fs = 8000, WebRtcNetEQNetworkType nwType = kTCPLargeJitter);
- ~NETEQTEST_NetEQClass();
-
- int assign();
- int init(WebRtc_UWord16 fs = 8000);
- int assignBuffer(enum WebRtcNetEQDecoder *usedCodec, int noOfCodecs, WebRtcNetEQNetworkType nwType = kTCPLargeJitter);
- int loadCodec(WebRtcNetEQ_CodecDef & codecInst);
- int recIn(NETEQTEST_RTPpacket & rtp);
- WebRtc_Word16 recOut(WebRtc_Word16 *outData, void *msInfo = NULL, enum WebRtcNetEQOutputType *outputType = NULL);
- WebRtc_UWord32 getSpeechTimeStamp();
- WebRtcNetEQOutputType getOutputType();
-
- void * instance() { return (_inst); };
- void usePreparseRTP( bool useIt = true ) { _preparseRTP = useIt; };
- bool usingPreparseRTP() { return (_preparseRTP); };
- void setMaster( bool isMaster = true ) { _isMaster = isMaster; };
- void setSlave() { _isMaster = false; };
- bool isMaster() { return (_isMaster); };
- bool isSlave() { return (!_isMaster); };
-
-#ifdef WINDOWS_TIMING
- double getRecInTime() { return (static_cast<double>( _totTimeRecIn.QuadPart )); };
- double getRecOutTime() { return (static_cast<double>( _totTimeRecOut.QuadPart )); };
-#else
- double getRecInTime() { return (0.0); };
- double getRecOutTime() { return (0.0); };
-
-#endif
-
- void printError();
- void printError(NETEQTEST_RTPpacket & rtp);
-
-private:
- void * _inst;
- WebRtc_Word8 * _instMem;
- WebRtc_Word8 * _bufferMem;
- bool _preparseRTP;
- int _fsmult;
- bool _isMaster;
-#ifdef WINDOWS_TIMING
- LARGE_INTEGER _totTimeRecIn;
- LARGE_INTEGER _totTimeRecOut;
-#endif
-};
-
-
-
-//class NETEQTEST_NetEQVector
-//{
-//public:
-// NETEQTEST_NetEQVector(int numChannels);
-// NETEQTEST_NetEQVector(int numChannels, enum WebRtcNetEQDecoder *usedCodec, int noOfCodecs,
-// WebRtc_UWord16 fs = 8000, WebRtcNetEQNetworkType nwType = kTCPLargeJitter);
-// ~NETEQTEST_NetEQVector();
-//
-//private:
-// std::vector<NETEQTEST_NetEQClass *> channels;
-//};
-
-#endif //NETEQTEST_NETEQCLASS_H
diff --git a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc
deleted file mode 100644
index 0412f06..0000000
--- a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.cc
+++ /dev/null
@@ -1,875 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "NETEQTEST_RTPpacket.h"
-
-#include <assert.h>
-#include <string.h>
-
-#ifdef WIN32
-#include <winsock2.h>
-#else
-#include <netinet/in.h> // for htons, htonl, etc
-#endif
-
-#include <cstdlib>
-
-#include "gtest/gtest.h"
-
-#define HDR_SIZE 8 // rtpplay packet header size in bytes
-
-
-NETEQTEST_RTPpacket::NETEQTEST_RTPpacket()
-:
-_datagram(NULL),
-_payloadPtr(NULL),
-_memSize(0),
-_datagramLen(-1),
-_payloadLen(0),
-_rtpParsed(false),
-_receiveTime(0),
-_lost(false)
-{
- memset(&_rtpInfo, 0, sizeof(_rtpInfo));
- _blockList.clear();
-}
-
-NETEQTEST_RTPpacket::NETEQTEST_RTPpacket(const NETEQTEST_RTPpacket& copyFromMe)
-{
-
- memcpy(this, ©FromMe, sizeof(NETEQTEST_RTPpacket));
-
- _datagram = NULL;
- _payloadPtr = NULL;
-
- if(copyFromMe._datagram)
- {
- _datagram = new WebRtc_UWord8[_memSize];
-
- if(_datagram)
- {
- memcpy(_datagram, copyFromMe._datagram, _memSize);
- }
- }
-
- if(copyFromMe._payloadPtr)
- {
- _payloadPtr = _datagram + (copyFromMe._payloadPtr - copyFromMe._datagram);
- }
-
- _blockList = copyFromMe._blockList;
-
-}
-
-
-NETEQTEST_RTPpacket & NETEQTEST_RTPpacket::operator = (const NETEQTEST_RTPpacket & other)
-{
- if (this != &other) // protect against invalid self-assignment
- {
-
- // deallocate datagram memory if allocated
- if(_datagram)
- {
- delete [] _datagram;
- }
-
- // do shallow copy
- memcpy(this, &other, sizeof(NETEQTEST_RTPpacket));
-
- // reset pointers
- _datagram = NULL;
- _payloadPtr = NULL;
-
- if(other._datagram)
- {
- _datagram = new WebRtc_UWord8[other._memSize];
- _memSize = other._memSize;
-
- if(_datagram)
- {
- memcpy(_datagram, other._datagram, _memSize);
- }
- }
-
- if(other._payloadPtr)
- {
- _payloadPtr = _datagram + (other._payloadPtr - other._datagram);
- }
-
- // copy the blocking list (map)
- _blockList = other._blockList;
-
- }
-
- // by convention, always return *this
- return *this;
-}
-
-
-
-NETEQTEST_RTPpacket::~NETEQTEST_RTPpacket()
-{
- if(_datagram)
- {
- delete [] _datagram;
- }
-}
-
-
-void NETEQTEST_RTPpacket::reset()
-{
- if(_datagram) {
- delete [] _datagram;
- }
- _datagram = NULL;
- _memSize = 0;
- _datagramLen = -1;
- _payloadLen = 0;
- _payloadPtr = NULL;
- _receiveTime = 0;
- memset(&_rtpInfo, 0, sizeof(_rtpInfo));
- _rtpParsed = false;
-
-}
-
-int NETEQTEST_RTPpacket::skipFileHeader(FILE *fp)
-{
- if (!fp) {
- return -1;
- }
-
- const int kFirstLineLength = 40;
- char firstline[kFirstLineLength];
- if (fgets(firstline, kFirstLineLength, fp) == NULL) {
- return -1;
- }
- if (strncmp(firstline, "#!rtpplay", 9) == 0) {
- if (strncmp(firstline, "#!rtpplay1.0", 12) != 0) {
- return -1;
- }
- }
- else if (strncmp(firstline, "#!RTPencode", 11) == 0) {
- if (strncmp(firstline, "#!RTPencode1.0", 14) != 0) {
- return -1;
- }
- }
- else
- {
- return -1;
- }
-
- const int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
- if (fseek(fp, kRtpDumpHeaderSize, SEEK_CUR) != 0)
- {
- return -1;
- }
- return 0;
-}
-
-int NETEQTEST_RTPpacket::readFromFile(FILE *fp)
-{
- if(!fp)
- {
- return(-1);
- }
-
- WebRtc_UWord16 length, plen;
- WebRtc_UWord32 offset;
-
- if (fread(&length,2,1,fp)==0)
- {
- reset();
- return(-2);
- }
- length = ntohs(length);
-
- if (fread(&plen,2,1,fp)==0)
- {
- reset();
- return(-1);
- }
- int packetLen = ntohs(plen);
-
- if (fread(&offset,4,1,fp)==0)
- {
- reset();
- return(-1);
- }
- WebRtc_UWord32 receiveTime = ntohl(offset); // store in local variable until we have passed the reset below
-
- // Use length here because a plen of 0 specifies rtcp
- length = (WebRtc_UWord16) (length - HDR_SIZE);
-
- // check buffer size
- if (_datagram && _memSize < length)
- {
- reset();
- }
-
- if (!_datagram)
- {
- _datagram = new WebRtc_UWord8[length];
- _memSize = length;
- }
-
- if (fread((unsigned short *) _datagram,1,length,fp) != length)
- {
- reset();
- return(-1);
- }
-
- _datagramLen = length;
- _receiveTime = receiveTime;
-
- if (!_blockList.empty() && _blockList.count(payloadType()) > 0)
- {
- // discard this payload
- return(readFromFile(fp));
- }
-
- return(packetLen);
-
-}
-
-
-int NETEQTEST_RTPpacket::readFixedFromFile(FILE *fp, size_t length)
-{
- if (!fp)
- {
- return -1;
- }
-
- // check buffer size
- if (_datagram && _memSize < static_cast<int>(length))
- {
- reset();
- }
-
- if (!_datagram)
- {
- _datagram = new WebRtc_UWord8[length];
- _memSize = length;
- }
-
- if (fread(_datagram, 1, length, fp) != length)
- {
- reset();
- return -1;
- }
-
- _datagramLen = length;
- _receiveTime = 0;
-
- if (!_blockList.empty() && _blockList.count(payloadType()) > 0)
- {
- // discard this payload
- return readFromFile(fp);
- }
-
- return length;
-
-}
-
-
-int NETEQTEST_RTPpacket::writeToFile(FILE *fp)
-{
- if (!fp)
- {
- return -1;
- }
-
- WebRtc_UWord16 length, plen;
- WebRtc_UWord32 offset;
-
- // length including RTPplay header
- length = htons(_datagramLen + HDR_SIZE);
- if (fwrite(&length, 2, 1, fp) != 1)
- {
- return -1;
- }
-
- // payload length
- plen = htons(_datagramLen);
- if (fwrite(&plen, 2, 1, fp) != 1)
- {
- return -1;
- }
-
- // offset (=receive time)
- offset = htonl(_receiveTime);
- if (fwrite(&offset, 4, 1, fp) != 1)
- {
- return -1;
- }
-
-
- // write packet data
- if (fwrite(_datagram, 1, _datagramLen, fp) !=
- static_cast<size_t>(_datagramLen))
- {
- return -1;
- }
-
- return _datagramLen + HDR_SIZE; // total number of bytes written
-
-}
-
-
-void NETEQTEST_RTPpacket::blockPT(WebRtc_UWord8 pt)
-{
- _blockList[pt] = true;
-}
-
-
-void NETEQTEST_RTPpacket::parseHeader()
-{
- if (_rtpParsed)
- {
- // nothing to do
- return;
- }
-
- if (_datagramLen < 12)
- {
- // corrupt packet?
- return;
- }
-
- _payloadLen = parseRTPheader(_datagram, _datagramLen, &_rtpInfo, &_payloadPtr);
-
- _rtpParsed = true;
-
- return;
-
-}
-
-void NETEQTEST_RTPpacket::parseHeader(WebRtcNetEQ_RTPInfo & rtpInfo)
-{
- if (!_rtpParsed)
- {
- // parse the header
- parseHeader();
- }
-
- memcpy(&rtpInfo, &_rtpInfo, sizeof(WebRtcNetEQ_RTPInfo));
-}
-
-WebRtcNetEQ_RTPInfo const * NETEQTEST_RTPpacket::RTPinfo() const
-{
- if (_rtpParsed)
- {
- return &_rtpInfo;
- }
- else
- {
- return NULL;
- }
-}
-
-WebRtc_UWord8 * NETEQTEST_RTPpacket::datagram() const
-{
- if (_datagramLen > 0)
- {
- return _datagram;
- }
- else
- {
- return NULL;
- }
-}
-
-WebRtc_UWord8 * NETEQTEST_RTPpacket::payload() const
-{
- if (_payloadLen > 0)
- {
- return _payloadPtr;
- }
- else
- {
- return NULL;
- }
-}
-
-WebRtc_Word16 NETEQTEST_RTPpacket::payloadLen() const
-{
- return _payloadLen;
-}
-
-WebRtc_Word16 NETEQTEST_RTPpacket::dataLen() const
-{
- return _datagramLen;
-}
-
-bool NETEQTEST_RTPpacket::isParsed() const
-{
- return _rtpParsed;
-}
-
-bool NETEQTEST_RTPpacket::isLost() const
-{
- return _lost;
-}
-
-WebRtc_UWord8 NETEQTEST_RTPpacket::payloadType() const
-{
- WebRtcNetEQ_RTPInfo tempRTPinfo;
-
- if(_datagram)
- {
- parseRTPheader(_datagram, _datagramLen, &tempRTPinfo);
- }
- else
- {
- return 0;
- }
-
- return tempRTPinfo.payloadType;
-}
-
-WebRtc_UWord16 NETEQTEST_RTPpacket::sequenceNumber() const
-{
- WebRtcNetEQ_RTPInfo tempRTPinfo;
-
- if(_datagram)
- {
- parseRTPheader(_datagram, _datagramLen, &tempRTPinfo);
- }
- else
- {
- return 0;
- }
-
- return tempRTPinfo.sequenceNumber;
-}
-
-WebRtc_UWord32 NETEQTEST_RTPpacket::timeStamp() const
-{
- WebRtcNetEQ_RTPInfo tempRTPinfo;
-
- if(_datagram)
- {
- parseRTPheader(_datagram, _datagramLen, &tempRTPinfo);
- }
- else
- {
- return 0;
- }
-
- return tempRTPinfo.timeStamp;
-}
-
-WebRtc_UWord32 NETEQTEST_RTPpacket::SSRC() const
-{
- WebRtcNetEQ_RTPInfo tempRTPinfo;
-
- if(_datagram)
- {
- parseRTPheader(_datagram, _datagramLen, &tempRTPinfo);
- }
- else
- {
- return 0;
- }
-
- return tempRTPinfo.SSRC;
-}
-
-WebRtc_UWord8 NETEQTEST_RTPpacket::markerBit() const
-{
- WebRtcNetEQ_RTPInfo tempRTPinfo;
-
- if(_datagram)
- {
- parseRTPheader(_datagram, _datagramLen, &tempRTPinfo);
- }
- else
- {
- return 0;
- }
-
- return tempRTPinfo.markerBit;
-}
-
-
-
-int NETEQTEST_RTPpacket::setPayloadType(WebRtc_UWord8 pt)
-{
-
- if (_datagramLen < 12)
- {
- return -1;
- }
-
- if (!_rtpParsed)
- {
- _rtpInfo.payloadType = pt;
- }
-
- _datagram[1]=(unsigned char)(pt & 0xFF);
-
- return 0;
-
-}
-
-int NETEQTEST_RTPpacket::setSequenceNumber(WebRtc_UWord16 sn)
-{
-
- if (_datagramLen < 12)
- {
- return -1;
- }
-
- if (!_rtpParsed)
- {
- _rtpInfo.sequenceNumber = sn;
- }
-
- _datagram[2]=(unsigned char)((sn>>8)&0xFF);
- _datagram[3]=(unsigned char)((sn)&0xFF);
-
- return 0;
-
-}
-
-int NETEQTEST_RTPpacket::setTimeStamp(WebRtc_UWord32 ts)
-{
-
- if (_datagramLen < 12)
- {
- return -1;
- }
-
- if (!_rtpParsed)
- {
- _rtpInfo.timeStamp = ts;
- }
-
- _datagram[4]=(unsigned char)((ts>>24)&0xFF);
- _datagram[5]=(unsigned char)((ts>>16)&0xFF);
- _datagram[6]=(unsigned char)((ts>>8)&0xFF);
- _datagram[7]=(unsigned char)(ts & 0xFF);
-
- return 0;
-
-}
-
-int NETEQTEST_RTPpacket::setSSRC(WebRtc_UWord32 ssrc)
-{
-
- if (_datagramLen < 12)
- {
- return -1;
- }
-
- if (!_rtpParsed)
- {
- _rtpInfo.SSRC = ssrc;
- }
-
- _datagram[8]=(unsigned char)((ssrc>>24)&0xFF);
- _datagram[9]=(unsigned char)((ssrc>>16)&0xFF);
- _datagram[10]=(unsigned char)((ssrc>>8)&0xFF);
- _datagram[11]=(unsigned char)(ssrc & 0xFF);
-
- return 0;
-
-}
-
-int NETEQTEST_RTPpacket::setMarkerBit(WebRtc_UWord8 mb)
-{
-
- if (_datagramLen < 12)
- {
- return -1;
- }
-
- if (_rtpParsed)
- {
- _rtpInfo.markerBit = mb;
- }
-
- if (mb)
- {
- _datagram[0] |= 0x01;
- }
- else
- {
- _datagram[0] &= 0xFE;
- }
-
- return 0;
-
-}
-
-int NETEQTEST_RTPpacket::setRTPheader(const WebRtcNetEQ_RTPInfo *RTPinfo)
-{
- if (_datagramLen < 12)
- {
- // this packet is not ok
- return -1;
- }
-
- makeRTPheader(_datagram,
- RTPinfo->payloadType,
- RTPinfo->sequenceNumber,
- RTPinfo->timeStamp,
- RTPinfo->SSRC,
- RTPinfo->markerBit);
-
- return 0;
-}
-
-
-int NETEQTEST_RTPpacket::splitStereo(NETEQTEST_RTPpacket& slaveRtp, enum stereoModes mode)
-{
- // if mono, do nothing
- if (mode == stereoModeMono)
- {
- return 0;
- }
-
- // check that the RTP header info is parsed
- parseHeader();
-
- // start by copying the main rtp packet
- slaveRtp = *this;
-
- if(_payloadLen == 0)
- {
- // do no more
- return 0;
- }
-
- if(_payloadLen%2 != 0)
- {
- // length must be a factor of 2
- return -1;
- }
-
- switch(mode)
- {
- case stereoModeSample1:
- {
- // sample based codec with 1-byte samples
- splitStereoSample(slaveRtp, 1 /* 1 byte/sample */);
- break;
- }
- case stereoModeSample2:
- {
- // sample based codec with 2-byte samples
- splitStereoSample(slaveRtp, 2 /* 2 bytes/sample */);
- break;
- }
- case stereoModeFrame:
- {
- // frame based codec
- splitStereoFrame(slaveRtp);
- break;
- }
- case stereoModeMono:
- {
- assert(false);
- return -1;
- }
- }
-
- return 0;
-}
-
-
-void NETEQTEST_RTPpacket::makeRTPheader(unsigned char* rtp_data, WebRtc_UWord8 payloadType, WebRtc_UWord16 seqNo, WebRtc_UWord32 timestamp, WebRtc_UWord32 ssrc, WebRtc_UWord8 markerBit) const
-{
- rtp_data[0]=(unsigned char)0x80;
- if (markerBit)
- {
- rtp_data[0] |= 0x01;
- }
- else
- {
- rtp_data[0] &= 0xFE;
- }
- rtp_data[1]=(unsigned char)(payloadType & 0xFF);
- rtp_data[2]=(unsigned char)((seqNo>>8)&0xFF);
- rtp_data[3]=(unsigned char)((seqNo)&0xFF);
- rtp_data[4]=(unsigned char)((timestamp>>24)&0xFF);
- rtp_data[5]=(unsigned char)((timestamp>>16)&0xFF);
-
- rtp_data[6]=(unsigned char)((timestamp>>8)&0xFF);
- rtp_data[7]=(unsigned char)(timestamp & 0xFF);
-
- rtp_data[8]=(unsigned char)((ssrc>>24)&0xFF);
- rtp_data[9]=(unsigned char)((ssrc>>16)&0xFF);
-
- rtp_data[10]=(unsigned char)((ssrc>>8)&0xFF);
- rtp_data[11]=(unsigned char)(ssrc & 0xFF);
-}
-
-
-WebRtc_UWord16 NETEQTEST_RTPpacket::parseRTPheader(const WebRtc_UWord8 *datagram, int datagramLen, WebRtcNetEQ_RTPInfo *RTPinfo, WebRtc_UWord8 **payloadPtr) const
-{
- WebRtc_Word16 *rtp_data = (WebRtc_Word16 *) datagram;
- int i_P, i_X, i_CC, i_extlength=-1, i_padlength=0, i_startPosition;
-
- i_P=(((WebRtc_UWord16)(rtp_data[0] & 0x20))>>5); /* Extract the P bit */
- i_X=(((WebRtc_UWord16)(rtp_data[0] & 0x10))>>4); /* Extract the X bit */
- i_CC=(WebRtc_UWord16)(rtp_data[0] & 0xF); /* Get the CC number */
- RTPinfo->markerBit = (WebRtc_UWord8) ((rtp_data[0] >> 15) & 0x01); /* Get the marker bit */
- RTPinfo->payloadType = (WebRtc_UWord8) ((rtp_data[0] >> 8) & 0x7F); /* Get the coder type */
- RTPinfo->sequenceNumber = ((( ((WebRtc_UWord16)rtp_data[1]) >> 8) & 0xFF) |
- ( ((WebRtc_UWord16)(rtp_data[1] & 0xFF)) << 8)); /* Get the packet number */
- RTPinfo->timeStamp = ((((WebRtc_UWord16)rtp_data[2]) & 0xFF) << 24) |
- ((((WebRtc_UWord16)rtp_data[2]) & 0xFF00) << 8) |
- ((((WebRtc_UWord16)rtp_data[3]) >> 8) & 0xFF) |
- ((((WebRtc_UWord16)rtp_data[3]) & 0xFF) << 8); /* Get timestamp */
- RTPinfo->SSRC=((((WebRtc_UWord16)rtp_data[4]) & 0xFF) << 24) |
- ((((WebRtc_UWord16)rtp_data[4]) & 0xFF00) << 8) |
- ((((WebRtc_UWord16)rtp_data[5]) >> 8) & 0xFF) |
- ((((WebRtc_UWord16)rtp_data[5]) & 0xFF) << 8); /* Get the SSRC */
-
- if (i_X==1) {
- /* Extention header exists. Find out how many WebRtc_Word32 it consists of */
- i_extlength=((( ((WebRtc_UWord16)rtp_data[7+2*i_CC]) >> 8) & 0xFF) |
- ( ((WebRtc_UWord16)(rtp_data[7+2*i_CC]&0xFF)) << 8));
- }
- if (i_P==1) {
- /* Padding exists. Find out how many bytes the padding consists of */
- if (datagramLen & 0x1) {
- /* odd number of bytes => last byte in higher byte */
- i_padlength=(rtp_data[datagramLen>>1] & 0xFF);
- } else {
- /* even number of bytes => last byte in lower byte */
- i_padlength=(((WebRtc_UWord16)rtp_data[(datagramLen>>1)-1]) >> 8);
- }
- }
-
- i_startPosition=12+4*(i_extlength+1)+4*i_CC;
-
- if (payloadPtr) {
- *payloadPtr = (WebRtc_UWord8*) &rtp_data[i_startPosition>>1];
- }
-
- return (WebRtc_UWord16) (datagramLen-i_startPosition-i_padlength);
-}
-
-//void NETEQTEST_RTPpacket::splitStereoSample(WebRtc_UWord8 *data, WebRtc_UWord16 *lenBytes, WebRtc_UWord8 *slaveData, WebRtc_UWord16 *slaveLenBytes, int stride)
-void NETEQTEST_RTPpacket::splitStereoSample(NETEQTEST_RTPpacket& slaveRtp, int stride)
-{
- if(!_payloadPtr || !slaveRtp._payloadPtr
- || _payloadLen <= 0 || slaveRtp._memSize < _memSize)
- {
- return;
- }
-
- WebRtc_UWord8 *readDataPtr = _payloadPtr;
- WebRtc_UWord8 *writeDataPtr = _payloadPtr;
- WebRtc_UWord8 *slaveData = slaveRtp._payloadPtr;
-
- while (readDataPtr - _payloadPtr < _payloadLen)
- {
- // master data
- for (int ix = 0; ix < stride; ix++) {
- *writeDataPtr = *readDataPtr;
- writeDataPtr++;
- readDataPtr++;
- }
-
- // slave data
- for (int ix = 0; ix < stride; ix++) {
- *slaveData = *readDataPtr;
- slaveData++;
- readDataPtr++;
- }
- }
-
- _payloadLen /= 2;
- slaveRtp._payloadLen = _payloadLen;
-}
-
-
-//void NETEQTEST_RTPpacket::splitStereoFrame(WebRtc_UWord8 *data, WebRtc_UWord16 *lenBytes, WebRtc_UWord8 *slaveData, WebRtc_UWord16 *slaveLenBytes)
-void NETEQTEST_RTPpacket::splitStereoFrame(NETEQTEST_RTPpacket& slaveRtp)
-{
- if(!_payloadPtr || !slaveRtp._payloadPtr
- || _payloadLen <= 0 || slaveRtp._memSize < _memSize)
- {
- return;
- }
-
- memmove(slaveRtp._payloadPtr, _payloadPtr + _payloadLen/2, _payloadLen/2);
-
- _payloadLen /= 2;
- slaveRtp._payloadLen = _payloadLen;
-}
-
-// Get the RTP header for the RED payload indicated by argument index.
-// The first RED payload is index = 0.
-int NETEQTEST_RTPpacket::extractRED(int index, WebRtcNetEQ_RTPInfo& red)
-{
-//
-// 0 1 2 3
-// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// |1| block PT | timestamp offset | block length |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// |1| ... |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// |0| block PT |
-// +-+-+-+-+-+-+-+-+
-//
-
- parseHeader();
-
- WebRtc_UWord8* ptr = payload();
- WebRtc_UWord8* payloadEndPtr = ptr + payloadLen();
- int num_encodings = 0;
- int total_len = 0;
-
- while ((ptr < payloadEndPtr) && (*ptr & 0x80))
- {
- int len = ((ptr[2] & 0x03) << 8) + ptr[3];
- if (num_encodings == index)
- {
- // Header found.
- red.payloadType = ptr[0] & 0x7F;
- WebRtc_UWord32 offset = (ptr[1] << 6) + ((ptr[2] & 0xFC) >> 2);
- red.sequenceNumber = sequenceNumber();
- red.timeStamp = timeStamp() - offset;
- red.markerBit = markerBit();
- red.SSRC = SSRC();
- return len;
- }
- ++num_encodings;
- total_len += len;
- ptr += 4;
- }
- if ((ptr < payloadEndPtr) && (num_encodings == index))
- {
- // Last header.
- red.payloadType = ptr[0] & 0x7F;
- red.sequenceNumber = sequenceNumber();
- red.timeStamp = timeStamp();
- red.markerBit = markerBit();
- red.SSRC = SSRC();
- ++ptr;
- return payloadLen() - (ptr - payload()) - total_len;
- }
- return -1;
-}
-
-// Randomize the payload, not the RTP header.
-void NETEQTEST_RTPpacket::scramblePayload(void)
-{
- parseHeader();
-
- for (int i = 0; i < _payloadLen; ++i)
- {
- _payloadPtr[i] = static_cast<WebRtc_UWord8>(std::rand());
- }
-}
diff --git a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h b/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
deleted file mode 100644
index 0478568..0000000
--- a/trunk/src/modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h
+++ /dev/null
@@ -1,91 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef NETEQTEST_RTPPACKET_H
-#define NETEQTEST_RTPPACKET_H
-
-#include <map>
-#include <stdio.h>
-#include "typedefs.h"
-#include "webrtc_neteq_internal.h"
-
-enum stereoModes {
- stereoModeMono,
- stereoModeSample1,
- stereoModeSample2,
- stereoModeFrame
-};
-
-class NETEQTEST_RTPpacket
-{
-public:
- NETEQTEST_RTPpacket();
- NETEQTEST_RTPpacket(const NETEQTEST_RTPpacket& copyFromMe);
- NETEQTEST_RTPpacket & operator = (const NETEQTEST_RTPpacket & other);
- bool operator !() const { return (dataLen() < 0); };
- ~NETEQTEST_RTPpacket();
- void reset();
- static int skipFileHeader(FILE *fp);
- int readFromFile(FILE *fp);
- int readFixedFromFile(FILE *fp, size_t len);
- int writeToFile(FILE *fp);
- void blockPT(WebRtc_UWord8 pt);
- //WebRtc_Word16 payloadType();
- void parseHeader();
- void parseHeader(WebRtcNetEQ_RTPInfo & rtpInfo);
- WebRtcNetEQ_RTPInfo const * RTPinfo() const;
- WebRtc_UWord8 * datagram() const;
- WebRtc_UWord8 * payload() const;
- WebRtc_Word16 payloadLen() const;
- WebRtc_Word16 dataLen() const;
- bool isParsed() const;
- bool isLost() const;
- WebRtc_UWord32 time() const { return _receiveTime; };
-
- WebRtc_UWord8 payloadType() const;
- WebRtc_UWord16 sequenceNumber() const;
- WebRtc_UWord32 timeStamp() const;
- WebRtc_UWord32 SSRC() const;
- WebRtc_UWord8 markerBit() const;
-
- int setPayloadType(WebRtc_UWord8 pt);
- int setSequenceNumber(WebRtc_UWord16 sn);
- int setTimeStamp(WebRtc_UWord32 ts);
- int setSSRC(WebRtc_UWord32 ssrc);
- int setMarkerBit(WebRtc_UWord8 mb);
- void setTime(WebRtc_UWord32 receiveTime) { _receiveTime = receiveTime; };
-
- int setRTPheader(const WebRtcNetEQ_RTPInfo *RTPinfo);
-
- int splitStereo(NETEQTEST_RTPpacket& slaveRtp, enum stereoModes mode);
-
- int extractRED(int index, WebRtcNetEQ_RTPInfo& red);
-
- void scramblePayload(void);
-
- WebRtc_UWord8 * _datagram;
- WebRtc_UWord8 * _payloadPtr;
- int _memSize;
- WebRtc_Word16 _datagramLen;
- WebRtc_Word16 _payloadLen;
- WebRtcNetEQ_RTPInfo _rtpInfo;
- bool _rtpParsed;
- WebRtc_UWord32 _receiveTime;
- bool _lost;
- std::map<WebRtc_UWord8, bool> _blockList;
-
-private:
- void makeRTPheader(unsigned char* rtp_data, WebRtc_UWord8 payloadType, WebRtc_UWord16 seqNo, WebRtc_UWord32 timestamp, WebRtc_UWord32 ssrc, WebRtc_UWord8 markerBit) const;
- WebRtc_UWord16 parseRTPheader(const WebRtc_UWord8 *datagram, int datagramLen, WebRtcNetEQ_RTPInfo *RTPinfo, WebRtc_UWord8 **payloadPtr = NULL) const;
- void splitStereoSample(NETEQTEST_RTPpacket& slaveRtp, int stride);
- void splitStereoFrame(NETEQTEST_RTPpacket& slaveRtp);
-};
-
-#endif //NETEQTEST_RTPPACKET_H
diff --git a/trunk/src/modules/audio_coding/neteq/test/NetEqRTPplay.cc b/trunk/src/modules/audio_coding/neteq/test/NetEqRTPplay.cc
deleted file mode 100644
index d3f7eb5..0000000
--- a/trunk/src/modules/audio_coding/neteq/test/NetEqRTPplay.cc
+++ /dev/null
@@ -1,1736 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//TODO(hlundin): Reformat file to meet style guide.
-
-/* header includes */
-#include "typedefs.h"
-#include "stdio.h"
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_internal.h"
-#include "webrtc_neteq_help_macros.h"
-#include "neteq_error_codes.h" // for the API test
-
-#include "NETEQTEST_RTPpacket.h"
-#include "NETEQTEST_NetEQClass.h"
-#include "NETEQTEST_CodecClass.h"
-
-#include <string.h>
-#include <stdlib.h>
-#include <time.h>
-#include <map>
-#include <vector>
-
-#ifdef WIN32
-#include <cassert>
-#include <windows.h>
-#endif
-
-#ifdef WEBRTC_LINUX
-#include <netinet/in.h>
-#include <libgen.h>
-#include <cassert>
-#endif
-
-//#include "vld.h"
-
-//#define NETEQ_DELAY_LOGGING
-//#define DUMMY_SLAVE_CHANNEL
-
-#ifdef NETEQ_DELAY_LOGGING
-#include "delay_logging.h"
-#define DUMMY_SLAVE_CHANNEL // do not use a slave channel, only generate zeros instead
-#endif
-
-
-/************************/
-/* Define payload types */
-/************************/
-
-// Payload types are defined in the textfile ptypes.txt, and can be changed after compilation.
-
-
-
-/*********************/
-/* Misc. definitions */
-/*********************/
-
-#define TIME_STEP 1
-#define FIRSTLINELEN 40
-#define MAX_NETEQ_BUFFERSIZE 170000 //100000
-#define CHECK_ZERO(a) {int errCode = a; char tempErrName[WEBRTC_NETEQ_MAX_ERROR_NAME]; if((errCode)!=0){errCode = WebRtcNetEQ_GetErrorCode(inst); WebRtcNetEQ_GetErrorName(errCode, tempErrName, WEBRTC_NETEQ_MAX_ERROR_NAME); printf("\n %s \n line: %d \n error at %s\n Error Code = %d\n",__FILE__,__LINE__,#a, errCode); exit(0);}}
-#define CHECK_NOT_NULL(a) if((a)==NULL){printf("\n %s \n line: %d \nerror at %s\n",__FILE__,__LINE__,#a );return(-1);}
-//#define PLAY_CLEAN // ignore arrival times and let the packets arrive according to RTP timestamps
-#define HDR_SIZE 8 // rtpplay packet header size in bytes
-//#define JUNK_DATA // scramble the payloads to test error resilience
-//#define ZERO_TS_START
-
-#ifdef JUNK_DATA
- #define SEED_FILE "randseed.txt"
-#endif
-
-#ifdef WIN32
-#define MY_MAX_DRIVE _MAX_DRIVE
-#define MY_MAX_PATH _MAX_PATH
-#define MY_MAX_FNAME _MAX_FNAME
-#define MY_MAX_EXT _MAX_EXT
-
-#elif defined(WEBRTC_LINUX)
-#include <linux/limits.h>
-#define MY_MAX_PATH PATH_MAX
-
-#elif defined(WEBRTC_MAC)
-#include <sys/syslimits.h>
-#define MY_MAX_PATH PATH_MAX
-#endif // WEBRTC_MAC
-
-/************/
-/* Typedefs */
-/************/
-
-typedef struct {
- enum WebRtcNetEQDecoder codec;
- enum stereoModes stereo;
- NETEQTEST_Decoder * decoder[2];
- int fs;
-} decoderStruct;
-
-
-/*************************/
-/* Function declarations */
-/*************************/
-
-void stereoInterleave(WebRtc_Word16 *data, WebRtc_Word16 totalLen);
-int getNextRecoutTime(FILE *fp, WebRtc_UWord32 *nextTime);
-void getNextExtraDelay(FILE *fp, WebRtc_UWord32 *t, int *d);
-bool splitStereo(NETEQTEST_RTPpacket& rtp, NETEQTEST_RTPpacket& rtpSlave,
- const WebRtc_Word16 *stereoPtype, const enum stereoModes *stereoMode, int noOfStereoCodecs,
- const WebRtc_Word16 *cngPtype, int noOfCngCodecs,
- bool *isStereo);
-void parsePtypeFile(FILE *ptypeFile, std::map<WebRtc_UWord8, decoderStruct>* decoders);
-int populateUsedCodec(std::map<WebRtc_UWord8, decoderStruct>* decoders, enum WebRtcNetEQDecoder *usedCodec);
-void createAndInsertDecoders (NETEQTEST_NetEQClass *neteq, std::map<WebRtc_UWord8, decoderStruct>* decoders, int channelNumber);
-void free_coders(std::map<WebRtc_UWord8, decoderStruct> & decoders);
-int doAPItest();
-bool changeStereoMode(NETEQTEST_RTPpacket & rtp, std::map<WebRtc_UWord8, decoderStruct> & decoders, enum stereoModes *stereoMode);
-
-
-
-/********************/
-/* Global variables */
-/********************/
-
-WebRtc_Word16 NetEqPacketBuffer[MAX_NETEQ_BUFFERSIZE>>1];
-WebRtc_Word16 NetEqPacketBufferSlave[MAX_NETEQ_BUFFERSIZE>>1];
-
-#ifdef NETEQ_DELAY_LOGGING
-extern "C" {
- FILE *delay_fid2; /* file pointer */
- WebRtc_UWord32 tot_received_packets=0;
-}
-#endif
-
-#ifdef DEF_BUILD_DATE
-extern char BUILD_DATE;
-#endif
-
-WebRtc_UWord32 writtenSamples = 0;
-WebRtc_UWord32 simClock=0;
-
-int main(int argc, char* argv[])
-{
- std::vector<NETEQTEST_NetEQClass *> NetEQvector;
- NETEQTEST_RTPpacket rtp;
- char version[20];
-
- NETEQTEST_RTPpacket slaveRtp;
- //bool switchMS = false;
- //bool duplicatePayload = false;
- enum WebRtcNetEQDecoder usedCodec[kDecoderReservedEnd-1];
- int noOfCodecs;
- int ok;
- WebRtc_Word16 out_data[640*2];
- WebRtc_Word16 outLen, writeLen;
- int fs = 8000;
- WebRtcNetEQ_RTCPStat RTCPstat;
-#ifdef WIN32
- char outdrive[MY_MAX_DRIVE];
- char outpath[MY_MAX_PATH];
- char outfile[MY_MAX_FNAME];
- char outext[MY_MAX_EXT];
-#endif
- char outfilename[MY_MAX_PATH];
-#ifdef NETEQ_DELAY_LOGGING
- float clock_float;
- int temp_var;
-#endif
-#ifdef JUNK_DATA
- FILE *seedfile;
-#endif
- FILE *recoutTimes = NULL;
- FILE *extraDelays = NULL;
- WebRtcNetEQPlayoutMode streamingMode = kPlayoutOn;
- bool preParseRTP = false;
- bool rtpOnly = false;
- int packetLen = 0;
- int packetCount = 0;
- std::map<WebRtc_UWord8, decoderStruct> decoders;
-
- /* get the version string */
- WebRtcNetEQ_GetVersion(version);
- printf("\n\nNetEq version: %s\n", version);
-#ifdef DEF_BUILD_DATE
- printf("Build time: %s\n", __BUILD_DATE);
-#endif
-
- /* check number of parameters */
- if ((argc < 3)
-#ifdef WIN32 // implicit output file name possible for windows
- && (argc < 2)
-#endif
- ) {
- /* print help text and exit */
- printf("Test program for NetEQ.\n");
- printf("The program reads an RTP stream from file and inserts it into NetEQ.\n");
- printf("The format of the RTP stream file should be the same as for rtpplay,\n");
- printf("and can be obtained e.g., from Ethereal by using\n");
- printf("Statistics -> RTP -> Show All Streams -> [select a stream] -> Save As\n\n");
- printf("Usage:\n\n");
-#ifdef WIN32
- printf("%s RTPfile [outfile] [-options]\n", argv[0]);
-#else
- printf("%s RTPfile outfile [-options]\n", argv[0]);
-#endif
- printf("where:\n");
-
- printf("RTPfile : RTP stream input file\n\n");
-
- printf("outfile : PCM speech output file\n");
- printf(" Output file name is derived from RTP file name if omitted\n\n");
-
- printf("-options are optional switches:\n");
- printf("\t-recout datfile : supply recout times\n");
- printf("\t-extradelay datfile : supply extra delay settings and timing\n");
- printf("\t-streaming : engage streaming mode\n");
- printf("\t-fax : engage fax mode\n");
- printf("\t-preparsertp : use RecIn with pre-parsed RTP\n");
- printf("\t-rtponly packLenBytes : input file consists of constant size RTP packets without RTPplay headers\n");
- //printf("\t-switchms : switch from mono to stereo (copy channel) after 10 seconds\n");
- //printf("\t-duplicate : use two instances with identical input (2-channel mono)\n");
-
- return(0);
- }
-
- if (strcmp(argv[1], "-apitest")==0) {
- // do API test and then return
- ok=doAPItest();
-
- if (ok==0)
- printf("API test successful!\n");
- else
- printf("API test failed!\n");
-
- return(ok);
- }
-
- FILE* in_file=fopen(argv[1],"rb");
- CHECK_NOT_NULL(in_file);
- printf("Input file: %s\n",argv[1]);
-
- int argIx = 2; // index of next argument from command line
-
- if ( argc >= 3 && argv[2][0] != '-' ) { // output name given on command line
- strcpy(outfilename, argv[2]);
- argIx++;
- } else { // derive output name from input name
-#ifdef WIN32
- _splitpath(argv[1],outdrive,outpath,outfile,outext);
- _makepath(outfilename,outdrive,outpath,outfile,"pcm");
-#else
- fprintf(stderr,"Output file name must be specified.\n");
- return(-1);
-#endif
- }
- FILE* out_file=fopen(outfilename,"wb");
- if (out_file==NULL) {
- fprintf(stderr,"Could not open file %s for writing\n", outfilename);
- return(-1);
- }
- printf("Output file: %s\n",outfilename);
-
- // Parse for more arguments, all beginning with '-'
-
- while( argIx < argc ) {
- if (argv[argIx][0] != '-') {
- fprintf(stderr,"Unknown input argument %s\n", argv[argIx]);
- return(-1);
- }
-
- if( strcmp(argv[argIx], "-recout") == 0 ) {
- argIx++;
- recoutTimes = fopen(argv[argIx], "rb");
- CHECK_NOT_NULL(recoutTimes);
- argIx++;
- }
- else if( strcmp(argv[argIx], "-extradelay") == 0 ) {
- argIx++;
- extraDelays = fopen(argv[argIx], "rb");
- CHECK_NOT_NULL(extraDelays);
- argIx++;
- }
- else if( strcmp(argv[argIx], "-streaming") == 0 ) {
- argIx++;
- streamingMode = kPlayoutStreaming;
- }
- else if( strcmp(argv[argIx], "-fax") == 0 ) {
- argIx++;
- streamingMode = kPlayoutFax;
- }
- else if( strcmp(argv[argIx], "-preparsertp") == 0 ) {
- argIx++;
- preParseRTP = true;
- }
- else if( strcmp(argv[argIx], "-rtponly") == 0 ) {
- argIx++;
- rtpOnly = true;
- packetLen = atoi(argv[argIx]);
- argIx++;
- if (packetLen <= 0)
- {
- printf("Wrong packet size used with argument -rtponly.\n");
- exit(1);
- }
- }
- //else if( strcmp(argv[argIx], "-switchms") == 0 ) {
- // argIx++;
- // switchMS = true;
- //}
- //else if( strcmp(argv[argIx], "-duplicate") == 0 ) {
- // argIx++;
- // duplicatePayload = true;
- //}
- else {
- fprintf(stderr,"Unknown input argument %s\n", argv[argIx]);
- return(-1);
- }
- }
-
-
-
-#ifdef NETEQ_DELAY_LOGGING
- char delayfile[MY_MAX_PATH];
-#ifdef WIN32
- _splitpath(outfilename,outdrive,outpath,outfile,outext);
- _makepath(delayfile,outdrive,outpath,outfile,"d");
-#else
- sprintf(delayfile, "%s.d", outfilename);
-#endif
- delay_fid2 = fopen(delayfile,"wb");
- fprintf(delay_fid2, "#!NetEQ_Delay_Logging%s\n", NETEQ_DELAY_LOGGING_VERSION_STRING);
-#endif
-
- char ptypesfile[MY_MAX_PATH];
-#ifdef WIN32
- _splitpath(argv[0],outdrive,outpath,outfile,outext);
- _makepath(ptypesfile,outdrive,outpath,"ptypes","txt");
-#else
- // TODO(hlundin): Include path to ptypes, as for WIN32 above.
- strcpy(ptypesfile, "ptypes.txt");
-#endif
- FILE *ptypeFile = fopen(ptypesfile,"rt");
- if (!ptypeFile) {
- // Check if we can find the file at the usual place in the trunk.
- if (strstr(argv[0], "out/Debug/")) {
- int path_len = strstr(argv[0], "out/Debug/") - argv[0];
- strncpy(ptypesfile, argv[0], path_len);
- ptypesfile[path_len] = '\0';
- strcat(ptypesfile,
- "src/modules/audio_coding/NetEQ/main/test/ptypes.txt");
- ptypeFile = fopen(ptypesfile,"rt");
- }
- }
- CHECK_NOT_NULL(ptypeFile);
- printf("Ptypes file: %s\n\n", ptypesfile);
-
- parsePtypeFile(ptypeFile, &decoders);
- fclose(ptypeFile);
-
- noOfCodecs = populateUsedCodec(&decoders, usedCodec);
-
-
- /* read RTP file header */
- if (!rtpOnly)
- {
- if (NETEQTEST_RTPpacket::skipFileHeader(in_file) != 0)
- {
- fprintf(stderr, "Wrong format in RTP file.\n");
- return -1;
- }
- }
-
- /* check payload type for first speech packet */
- long tempFilePos = ftell(in_file);
- enum stereoModes stereoMode = stereoModeMono;
-
- if (!rtpOnly)
- {
- while (rtp.readFromFile(in_file) >= 0)
- {
- if (decoders.count(rtp.payloadType()) > 0
- && decoders[rtp.payloadType()].codec != kDecoderRED
- && decoders[rtp.payloadType()].codec != kDecoderAVT
- && decoders[rtp.payloadType()].codec != kDecoderCNG )
- {
- stereoMode = decoders[rtp.payloadType()].stereo;
- fs = decoders[rtp.payloadType()].fs;
- break;
- }
- }
- }
- else
- {
- while (rtp.readFixedFromFile(in_file, packetLen) >= 0)
- {
- if (decoders.count(rtp.payloadType()) > 0
- && decoders[rtp.payloadType()].codec != kDecoderRED
- && decoders[rtp.payloadType()].codec != kDecoderAVT
- && decoders[rtp.payloadType()].codec != kDecoderCNG )
- {
- stereoMode = decoders[rtp.payloadType()].stereo;
- fs = decoders[rtp.payloadType()].fs;
- break;
- }
- }
- }
-
- fseek(in_file, tempFilePos, SEEK_SET /* from beginning */);
-
-
- /* block some payload types */
- //rtp.blockPT(72);
- //rtp.blockPT(23);
-
- /* read first packet */
- if (!rtpOnly)
- {
- rtp.readFromFile(in_file);
- }
- else
- {
- rtp.readFixedFromFile(in_file, packetLen);
- rtp.setTime((1000 * rtp.timeStamp()) / fs);
- }
- if (!rtp)
- {
- printf("\nWarning: RTP file is empty\n\n");
- }
-
-
- /* Initialize NetEQ instances */
- int numInst = 1;
- if (stereoMode > stereoModeMono)
- {
- numInst = 2;
- }
-
- for (int i = 0; i < numInst; i++)
- {
- // create memory, allocate, initialize, and allocate packet buffer memory
- NetEQvector.push_back (new NETEQTEST_NetEQClass(usedCodec, noOfCodecs, static_cast<WebRtc_UWord16>(fs), kTCPLargeJitter));
-
- createAndInsertDecoders (NetEQvector[i], &decoders, i /* channel */);
-
- WebRtcNetEQ_SetAVTPlayout(NetEQvector[i]->instance(),1); // enable DTMF playout
-
- WebRtcNetEQ_SetPlayoutMode(NetEQvector[i]->instance(), streamingMode);
-
- NetEQvector[i]->usePreparseRTP(preParseRTP);
-
- if (numInst > 1)
- {
- // we are using master/slave mode
- if (i == 0)
- {
- // first instance is master
- NetEQvector[i]->isMaster();
- }
- else
- {
- // all other are slaves
- NetEQvector[i]->isSlave();
- }
- }
- }
-
-
-#ifdef ZERO_TS_START
- WebRtc_UWord32 firstTS = rtp.timeStamp();
- rtp.setTimeStamp(0);
-#else
- WebRtc_UWord32 firstTS = 0;
-#endif
-
- // check stereo mode
- if (stereoMode > stereoModeMono)
- {
- if(rtp.splitStereo(slaveRtp, stereoMode))
- {
- printf("Error in splitStereo\n");
- }
- }
-
-#ifdef PLAY_CLEAN
- WebRtc_UWord32 prevTS = rtp.timeStamp();
- WebRtc_UWord32 currTS, prev_time;
-#endif
-
-#ifdef JUNK_DATA
- unsigned int random_seed = (unsigned int) /*1196764538; */time(NULL);
- srand(random_seed);
-
- if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
- fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
- }
- else {
- fprintf(seedfile, "%u\n", random_seed);
- fclose(seedfile);
- }
-#endif
-
- WebRtc_UWord32 nextRecoutTime;
- int lastRecout = getNextRecoutTime(recoutTimes, &nextRecoutTime); // does nothing if recoutTimes == NULL
-
- if (recoutTimes)
- simClock = (rtp.time() < nextRecoutTime ? rtp.time(): nextRecoutTime);
- else
- simClock = rtp.time(); // start immediately with first packet
-
- WebRtc_UWord32 start_clock = simClock;
-
- WebRtc_UWord32 nextExtraDelayTime;
- int extraDelay = -1;
- getNextExtraDelay(extraDelays, &nextExtraDelayTime, &extraDelay);
-
- void *msInfo;
- msInfo = malloc(WebRtcNetEQ_GetMasterSlaveInfoSize());
- if(msInfo == NULL)
- return(-1);
-
- while(rtp.dataLen() >= 0 || (recoutTimes && !lastRecout)) {
-// printf("simClock = %Lu\n", simClock);
-
-#ifdef NETEQ_DELAY_LOGGING
- temp_var = NETEQ_DELAY_LOGGING_SIGNAL_CLOCK;
- clock_float = (float) simClock;
- fwrite(&temp_var,sizeof(int),1,delay_fid2);
- fwrite(&clock_float, sizeof(float),1,delay_fid2);
-#endif
- /* time to set extra delay */
- if (extraDelay > -1 && simClock >= nextExtraDelayTime) {
- // set extra delay for all instances
- for (int i = 0; i < numInst; i++)
- {
- WebRtcNetEQ_SetExtraDelay(NetEQvector[i]->instance(), extraDelay);
- }
- getNextExtraDelay(extraDelays, &nextExtraDelayTime, &extraDelay);
- }
-
- /* check if time to receive */
- while (simClock >= rtp.time() && rtp.dataLen() >= 0)
- {
- if (rtp.dataLen() > 0)
- {
-
- // insert main packet
- NetEQvector[0]->recIn(rtp);
-
- if (stereoMode > stereoModeMono
- && slaveRtp.dataLen() > 0)
- {
- // insert slave packet
- NetEQvector[1]->recIn(slaveRtp);
- }
-
- }
-
- /* get next packet */
-#ifdef PLAY_CLEAN
- prev_time = rtp.time();
-#endif
- if (!rtpOnly)
- {
- rtp.readFromFile(in_file);
- }
- else
- {
- rtp.readFixedFromFile(in_file, packetLen);
- rtp.setTime((1000 * rtp.timeStamp()) / fs);
- }
-
- if (rtp.dataLen() >= 0)
- {
- rtp.setTimeStamp(rtp.timeStamp() - firstTS);
- }
-
- packetCount++;
-
- if (changeStereoMode(rtp, decoders, &stereoMode))
- {
- printf("Warning: stereo mode changed\n");
- }
-
- if (stereoMode > stereoModeMono)
- {
- if(rtp.splitStereo(slaveRtp, stereoMode))
- {
- printf("Error in splitStereo\n");
- }
- }
-
-#ifdef PLAY_CLEAN
- currTS = rtp.timeStamp();
- rtp.setTime(prev_time + (currTS-prevTS)/(fs/1000));
- prevTS = currTS;
-#endif
- }
-
- /* check if time to RecOut */
- if ( (!recoutTimes && (simClock%10)==0) // recout times not given from file
- || ( recoutTimes && (simClock >= nextRecoutTime) ) ) // recout times given from file
- {
- if (stereoMode > stereoModeMono)
- {
- // stereo
- WebRtc_Word16 tempLen;
- tempLen = NetEQvector[0]->recOut( out_data, msInfo ); // master
- outLen = NetEQvector[1]->recOut( &out_data[tempLen], msInfo ); // slave
-
- assert(tempLen == outLen);
-
- writeLen = outLen * 2;
- stereoInterleave(out_data, writeLen);
- }
- else
- {
- // mono
- outLen = NetEQvector[0]->recOut( out_data );
- writeLen = outLen;
- }
-
- // write to file
- fwrite(out_data,writeLen,2,out_file);
- writtenSamples += writeLen;
-
-
- lastRecout = getNextRecoutTime(recoutTimes, &nextRecoutTime); // does nothing if recoutTimes == NULL
-
- /* ask for statistics */
- WebRtcNetEQ_NetworkStatistics inCallStats;
- WebRtcNetEQ_GetNetworkStatistics(NetEQvector[0]->instance(), &inCallStats);
-
- }
-
- /* increase time */
- simClock+=TIME_STEP;
- }
-
- fclose(in_file);
- fclose(out_file);
-
-#ifdef NETEQ_DELAY_LOGGING
- temp_var = NETEQ_DELAY_LOGGING_SIGNAL_EOF;
- fwrite(&temp_var,sizeof(int),1,delay_fid2);
- fwrite(&tot_received_packets,sizeof(WebRtc_UWord32),1,delay_fid2);
- fprintf(delay_fid2,"End of file\n");
- fclose(delay_fid2);
-#endif
-
- WebRtcNetEQ_GetRTCPStats(NetEQvector[0]->instance(), &RTCPstat);
- printf("RTCP statistics:\n");
- printf(" cum_lost : %d\n", (int) RTCPstat.cum_lost);
- printf(" ext_max : %d\n", (int) RTCPstat.ext_max);
- printf(" fraction_lost : %d (%f%%)\n", RTCPstat.fraction_lost, (float)(100.0*RTCPstat.fraction_lost/256.0));
- printf(" jitter : %d\n", (int) RTCPstat.jitter);
-
- printf("\n Call duration ms : %u\n", simClock-start_clock);
-
- printf("\nComplexity estimates (including sub-components):\n");
- printf(" RecIn complexity : %.2f MCPS\n", NetEQvector[0]->getRecInTime() / ((float) 1000*(simClock-start_clock)));
- printf(" RecOut complexity : %.2f MCPS\n", NetEQvector[0]->getRecOutTime() / ((float) 1000*(simClock-start_clock)));
-
- free_coders(decoders);
- //free_coders(0 /* first channel */);
- // if (stereoMode > stereoModeMono) {
- // free_coders(1 /* second channel */);
- // }
- free(msInfo);
-
- for (std::vector<NETEQTEST_NetEQClass *>::iterator it = NetEQvector.begin();
- it < NetEQvector.end(); delete *it++);
-
- printf("\nSimulation done!\n");
-
-#ifdef JUNK_DATA
- if ( (seedfile = fopen(SEED_FILE, "a+t") ) == NULL ) {
- fprintf(stderr, "Error: Could not open file %s\n", SEED_FILE);
- }
- else {
- fprintf(seedfile, "ok\n\n");
- fclose(seedfile);
- }
-#endif
-
-
- // Log complexity to file
-/* FILE *statfile;
- statfile = fopen("complexity.txt","at");
- fprintf(statfile,"%.4f, %.4f\n", (float) totTime_RecIn.QuadPart / ((float) 1000*(simClock-start_clock)), (float) totTime_RecOut.QuadPart / ((float) 1000*(simClock-start_clock)));
- fclose(statfile);*/
-
- return(0);
-
-}
-
-
-
-
-
-/****************/
-/* Subfunctions */
-/****************/
-
-bool splitStereo(NETEQTEST_RTPpacket& rtp, NETEQTEST_RTPpacket& rtpSlave,
- const WebRtc_Word16 *stereoPtype, const enum stereoModes *stereoMode, int noOfStereoCodecs,
- const WebRtc_Word16 *cngPtype, int noOfCngCodecs,
- bool *isStereo)
-{
-
- // init
- //bool isStereo = false;
- enum stereoModes tempStereoMode = stereoModeMono;
- bool isCng = false;
-
- // check payload length
- if (rtp.dataLen() <= 0) {
- //*isStereo = false; // don't change
- return(*isStereo);
- }
-
- // check payload type
- WebRtc_Word16 ptype = rtp.payloadType();
-
- // is this a cng payload?
- for (int k = 0; k < noOfCngCodecs; k++) {
- if (ptype == cngPtype[k]) {
- // do not change stereo state
- isCng = true;
- tempStereoMode = stereoModeFrame;
- }
- }
-
- if (!isCng)
- {
- *isStereo = false;
-
- // is this payload type a stereo codec? which type?
- for (int k = 0; k < noOfStereoCodecs; k++) {
- if (ptype == stereoPtype[k]) {
- tempStereoMode = stereoMode[k];
- *isStereo = true;
- break; // exit for loop
- }
- }
- }
-
- if (*isStereo)
- {
- // split the payload if stereo
-
- if(rtp.splitStereo(rtpSlave, tempStereoMode))
- {
- printf("Error in splitStereo\n");
- }
-
- }
-
- return(*isStereo);
-
-}
-
-void stereoInterleave(WebRtc_Word16 *data, WebRtc_Word16 totalLen)
-{
- int k;
-
- for(k = totalLen/2; k < totalLen; k++) {
- WebRtc_Word16 temp = data[k];
- memmove(&data[2*k - totalLen + 2], &data[2*k - totalLen + 1], (totalLen - k -1) * sizeof(WebRtc_Word16));
- data[2*k - totalLen + 1] = temp;
- }
-}
-
-
-int getNextRecoutTime(FILE *fp, WebRtc_UWord32 *nextTime) {
-
- float tempTime;
-
- if (!fp) {
- return -1;
- }
-
- if (fread(&tempTime, sizeof(float), 1, fp) != 0) {
- // not end of file
- *nextTime = (WebRtc_UWord32) tempTime;
- return 0;
- }
-
- *nextTime = 0;
- fclose(fp);
-
- return 1;
-}
-
-void getNextExtraDelay(FILE *fp, WebRtc_UWord32 *t, int *d) {
-
- float temp[2];
-
- if(!fp) {
- *d = -1;
- return;
- }
-
- if (fread(&temp, sizeof(float), 2, fp) != 0) {
- // not end of file
- *t = (WebRtc_UWord32) temp[0];
- *d = (int) temp[1];
- return;
- }
-
- *d = -1;
- fclose(fp);
-
- return;
-}
-
-
-void parsePtypeFile(FILE *ptypeFile, std::map<WebRtc_UWord8, decoderStruct>* decoders)
-{
- int n, pt;
- char codec[100];
- decoderStruct tempDecoder;
-
- // read first line
- n = fscanf(ptypeFile, "%s %i\n", codec, &pt);
-
- while (n==2)
- {
- memset(&tempDecoder, 0, sizeof(decoderStruct));
- tempDecoder.stereo = stereoModeMono;
-
- if( pt >= 0 // < 0 disables this codec
- && isalpha(codec[0]) ) // and is a letter
- {
-
- /* check for stereo */
- int L = strlen(codec);
- bool isStereo = false;
-
- if (codec[L-1] == '*') {
- // stereo codec
- isStereo = true;
-
- // remove '*'
- codec[L-1] = '\0';
- }
-
-#ifdef CODEC_G711
- if(strcmp(codec, "pcmu") == 0) {
- tempDecoder.codec = kDecoderPCMu;
- tempDecoder.fs = 8000;
- }
- else if(strcmp(codec, "pcma") == 0) {
- tempDecoder.codec = kDecoderPCMa;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_IPCMU
- else if(strcmp(codec, "eg711u") == 0) {
- tempDecoder.codec = kDecoderEG711u;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_IPCMA
- else if(strcmp(codec, "eg711a") == 0) {
- tempDecoder.codec = kDecoderEG711a;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_ILBC
- else if(strcmp(codec, "ilbc") == 0) {
- tempDecoder.codec = kDecoderILBC;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_ISAC
- else if(strcmp(codec, "isac") == 0) {
- tempDecoder.codec = kDecoderISAC;
- tempDecoder.fs = 16000;
- }
-#endif
-#ifdef CODEC_ISACLC
- else if(strcmp(codec, "isaclc") == 0) {
- tempDecoder.codec = NETEQ_CODEC_ISACLC;
- tempDecoder.fs = 16000;
- }
-#endif
-#ifdef CODEC_ISAC_SWB
- else if(strcmp(codec, "isacswb") == 0) {
- tempDecoder.codec = kDecoderISACswb;
- tempDecoder.fs = 32000;
- }
-#endif
-#ifdef CODEC_IPCMWB
- else if(strcmp(codec, "ipcmwb") == 0) {
- tempDecoder.codec = kDecoderIPCMwb;
- tempDecoder.fs = 16000;
- }
-#endif
-#ifdef CODEC_G722
- else if(strcmp(codec, "g722") == 0) {
- tempDecoder.codec = kDecoderG722;
- tempDecoder.fs = 16000;
- }
-#endif
-#ifdef CODEC_G722_1_16
- else if(strcmp(codec, "g722_1_16") == 0) {
- tempDecoder.codec = kDecoderG722_1_16;
- tempDecoder.fs = 16000;
- }
-#endif
-#ifdef CODEC_G722_1_24
- else if(strcmp(codec, "g722_1_24") == 0) {
- tempDecoder.codec = kDecoderG722_1_24;
- tempDecoder.fs = 16000;
- }
-#endif
-#ifdef CODEC_G722_1_32
- else if(strcmp(codec, "g722_1_32") == 0) {
- tempDecoder.codec = kDecoderG722_1_32;
- tempDecoder.fs = 16000;
- }
-#endif
-#ifdef CODEC_G722_1C_24
- else if(strcmp(codec, "g722_1c_24") == 0) {
- tempDecoder.codec = kDecoderG722_1C_24;
- tempDecoder.fs = 32000;
- }
-#endif
-#ifdef CODEC_G722_1C_32
- else if(strcmp(codec, "g722_1c_32") == 0) {
- tempDecoder.codec = kDecoderG722_1C_32;
- tempDecoder.fs = 32000;
- }
-#endif
-#ifdef CODEC_G722_1C_48
- else if(strcmp(codec, "g722_1c_48") == 0) {
- tempDecoder.codec = kDecoderG722_1C_48;
- tempDecoder.fs = 32000;
- }
-#endif
-#ifdef CODEC_G723
- else if(strcmp(codec, "g723") == 0) {
- tempDecoder.codec = NETEQ_CODEC_G723;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_G726
- else if(strcmp(codec, "g726_16") == 0) {
- tempDecoder.codec = kDecoderG726_16;
- tempDecoder.fs = 8000;
- }
- else if(strcmp(codec, "g726_24") == 0) {
- tempDecoder.codec = kDecoderG726_24;
- tempDecoder.fs = 8000;
- }
- else if(strcmp(codec, "g726_32") == 0) {
- tempDecoder.codec = kDecoderG726_32;
- tempDecoder.fs = 8000;
- }
- else if(strcmp(codec, "g726_40") == 0) {
- tempDecoder.codec = kDecoderG726_40;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_G729
- else if(strcmp(codec, "g729") == 0) {
- tempDecoder.codec = kDecoderG729;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_G729D
- else if(strcmp(codec, "g729d") == 0) {
- tempDecoder.codec = NETEQ_CODEC_G729D;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_G729_1
- else if(strcmp(codec, "g729_1") == 0) {
- tempDecoder.codec = kDecoderG729_1;
- tempDecoder.fs = 16000;
- }
-#endif
-#ifdef CODEC_GSMFR
- else if(strcmp(codec, "gsmfr") == 0) {
- tempDecoder.codec = kDecoderGSMFR;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_GSMEFR
- else if(strcmp(codec, "gsmefr") == 0) {
- tempDecoder.codec = NETEQ_CODEC_GSMEFR;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_AMR
- else if(strcmp(codec, "amr") == 0) {
- tempDecoder.codec = kDecoderAMR;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_AMRWB
- else if(strcmp(codec, "amrwb") == 0) {
- tempDecoder.codec = kDecoderAMRWB;
- tempDecoder.fs = 16000;
- }
-#endif
-#ifdef CODEC_DVI4
- else if(strcmp(codec, "dvi4") == 0) {
- tempDecoder.codec = NETEQ_CODEC_DVI4;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_SPEEX_8
- else if(strcmp(codec, "speex8") == 0) {
- tempDecoder.codec = kDecoderSPEEX_8;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_SPEEX_16
- else if(strcmp(codec, "speex16") == 0) {
- tempDecoder.codec = kDecoderSPEEX_16;
- tempDecoder.fs = 16000;
- }
-#endif
-#ifdef CODEC_CELT_32
- else if(strcmp(codec, "celt32") == 0) {
- tempDecoder.codec = kDecoderCELT_32;
- tempDecoder.fs = 32000;
- }
-#endif
-#ifdef CODEC_SILK_NB
- else if(strcmp(codec, "silk8") == 0) {
- tempDecoder.codec = NETEQ_CODEC_SILK_8;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_SILK_WB
- else if(strcmp(codec, "silk12") == 0) {
- tempDecoder.codec = NETEQ_CODEC_SILK_12;
- tempDecoder.fs = 16000;
- }
- else if(strcmp(codec, "silk16") == 0) {
- tempDecoder.codec = NETEQ_CODEC_SILK_16;
- tempDecoder.fs = 16000;
- }
-#endif
-#ifdef CODEC_SILK_SWB
- else if(strcmp(codec, "silk24") == 0) {
- tempDecoder.codec = NETEQ_CODEC_SILK_24;
- tempDecoder.fs = 32000;
- }
-#endif
-#ifdef CODEC_MELPE
- else if(strcmp(codec, "melpe") == 0) {
- tempDecoder.codec = NETEQ_CODEC_MELPE;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_PCM16B
- else if(strcmp(codec, "pcm16b") == 0) {
- tempDecoder.codec = kDecoderPCM16B;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_PCM16B_WB
- else if(strcmp(codec, "pcm16b_wb") == 0) {
- tempDecoder.codec = kDecoderPCM16Bwb;
- tempDecoder.fs = 16000;
- }
-#endif
-#ifdef CODEC_PCM16B_32KHZ
- else if(strcmp(codec, "pcm16b_swb32khz") == 0) {
- tempDecoder.codec = kDecoderPCM16Bswb32kHz;
- tempDecoder.fs = 32000;
- }
-#endif
-#ifdef CODEC_PCM16B_48KHZ
- else if(strcmp(codec, "pcm16b_swb48khz") == 0) {
- tempDecoder.codec = kDecoderPCM16Bswb48kHz;
- tempDecoder.fs = 48000;
- }
-#endif
-#ifdef CODEC_CNGCODEC8
- else if(strcmp(codec, "cn") == 0) {
- tempDecoder.codec = kDecoderCNG;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_CNGCODEC16
- else if(strcmp(codec, "cn_wb") == 0) {
- tempDecoder.codec = kDecoderCNG;
- tempDecoder.fs = 16000;
- }
-#endif
-#ifdef CODEC_CNGCODEC32
- else if(strcmp(codec, "cn_swb32") == 0) {
- tempDecoder.codec = kDecoderCNG;
- tempDecoder.fs = 32000;
- }
-#endif
-#ifdef CODEC_CNGCODEC48
- else if(strcmp(codec, "cn_swb48") == 0) {
- tempDecoder.codec = kDecoderCNG;
- tempDecoder.fs = 48000;
- }
-#endif
-#ifdef CODEC_ATEVENT_DECODE
- else if(strcmp(codec, "avt") == 0) {
- tempDecoder.codec = kDecoderAVT;
- tempDecoder.fs = 8000;
- }
-#endif
-#ifdef CODEC_RED
- else if(strcmp(codec, "red") == 0) {
- tempDecoder.codec = kDecoderRED;
- tempDecoder.fs = 8000;
- }
-#endif
- else if(isalpha(codec[0])) {
- printf("Unsupported codec %s\n", codec);
- // read next line and continue while loop
- n = fscanf(ptypeFile, "%s %i\n", codec, &pt);
- continue;
- }
- else {
- // name is not recognized, and does not start with a letter
- // hence, it is commented out
- // read next line and continue while loop
- n = fscanf(ptypeFile, "%s %i\n", codec, &pt);
- continue;
- }
-
- // handle stereo
- if (tempDecoder.codec == kDecoderCNG)
- {
- // always set stereo mode for CNG, even if it is not marked at stereo
- tempDecoder.stereo = stereoModeFrame;
- }
- else if(isStereo)
- {
- switch(tempDecoder.codec) {
- // sample based codecs
- case kDecoderPCMu:
- case kDecoderPCMa:
- case kDecoderG722:
- {
- // 1 octet per sample
- tempDecoder.stereo = stereoModeSample1;
- break;
- }
- case kDecoderPCM16B:
- case kDecoderPCM16Bwb:
- case kDecoderPCM16Bswb32kHz:
- case kDecoderPCM16Bswb48kHz:
- {
- // 2 octets per sample
- tempDecoder.stereo = stereoModeSample2;
- break;
- }
-
- // fixed-rate frame codecs
-// case kDecoderG729:
-// case NETEQ_CODEC_G729D:
-// case NETEQ_CODEC_G729E:
-// case kDecoderG722_1_16:
-// case kDecoderG722_1_24:
-// case kDecoderG722_1_32:
-// case kDecoderG722_1C_24:
-// case kDecoderG722_1C_32:
-// case kDecoderG722_1C_48:
-// case NETEQ_CODEC_MELPE:
-// {
-// tempDecoder.stereo = stereoModeFrame;
-// break;
-// }
- default:
- {
- printf("Cannot use codec %s as stereo codec\n", codec);
- exit(0);
- }
- }
- }
-
- if (pt > 127)
- {
- printf("Payload type must be less than 128\n");
- exit(0);
- }
-
- // insert into codecs map
- (*decoders)[static_cast<WebRtc_UWord8>(pt)] = tempDecoder;
-
- }
-
- n = fscanf(ptypeFile, "%s %i\n", codec, &pt);
- } // end while
-
-}
-
-
-bool changeStereoMode(NETEQTEST_RTPpacket & rtp, std::map<WebRtc_UWord8, decoderStruct> & decoders, enum stereoModes *stereoMode)
-{
- if (decoders.count(rtp.payloadType()) > 0
- && decoders[rtp.payloadType()].codec != kDecoderRED
- && decoders[rtp.payloadType()].codec != kDecoderAVT
- && decoders[rtp.payloadType()].codec != kDecoderCNG )
- {
- if (decoders[rtp.payloadType()].stereo != *stereoMode)
- {
- *stereoMode = decoders[rtp.payloadType()].stereo;
- return true; // stereo mode did change
- }
- }
-
- return false; // stereo mode did not change
-}
-
-
-int populateUsedCodec(std::map<WebRtc_UWord8, decoderStruct>* decoders, enum WebRtcNetEQDecoder *usedCodec)
-{
- int numCodecs = 0;
-
- std::map<WebRtc_UWord8, decoderStruct>::iterator it;
-
- it = decoders->begin();
-
- for (int i = 0; i < static_cast<int>(decoders->size()); i++, it++)
- {
- usedCodec[numCodecs] = (*it).second.codec;
- numCodecs++;
- }
-
- return numCodecs;
-}
-
-
-void createAndInsertDecoders (NETEQTEST_NetEQClass *neteq, std::map<WebRtc_UWord8, decoderStruct>* decoders, int channelNumber)
-{
- std::map<WebRtc_UWord8, decoderStruct>::iterator it;
-
- for (it = decoders->begin(); it != decoders->end(); it++)
- {
- if (channelNumber == 0 ||
- ((*it).second.stereo > stereoModeMono ))
- {
- // create decoder instance
- WebRtc_UWord8 pt = static_cast<WebRtc_UWord8>( (*it).first );
- NETEQTEST_Decoder **dec = &((*it).second.decoder[channelNumber]);
- enum WebRtcNetEQDecoder type = (*it).second.codec;
-
- switch (type)
- {
-#ifdef CODEC_G711
- case kDecoderPCMu:
- *dec = new decoder_PCMU( pt );
- break;
- case kDecoderPCMa:
- *dec = new decoder_PCMA( pt );
- break;
-#endif
-#ifdef CODEC_IPCMU
- case kDecoderEG711u:
- *dec = new decoder_IPCMU( pt );
- break;
-#endif
-#ifdef CODEC_IPCMA
- case kDecoderEG711a:
- *dec = new decoder_IPCMA( pt );
- break;
-#endif
-#ifdef CODEC_IPCMWB
- case kDecoderIPCMwb:
- *dec = new decoder_IPCMWB( pt );
- break;
-#endif
-#ifdef CODEC_ILBC
- case kDecoderILBC:
- *dec = new decoder_ILBC( pt );
- break;
-#endif
-#ifdef CODEC_ISAC
- case kDecoderISAC:
- *dec = new decoder_iSAC( pt );
- break;
-#endif
-#ifdef CODEC_ISAC_SWB
- case kDecoderISACswb:
- *dec = new decoder_iSACSWB( pt );
- break;
-#endif
-#ifdef CODEC_G729
- case kDecoderG729:
- *dec = new decoder_G729( pt );
- break;
- case NETEQ_CODEC_G729D:
- printf("Error: G729D not supported\n");
- break;
-#endif
-#ifdef CODEC_G729E
- case NETEQ_CODEC_G729E:
- *dec = new decoder_G729E( pt );
- break;
-#endif
-#ifdef CODEC_G729_1
- case kDecoderG729_1:
- *dec = new decoder_G729_1( pt );
- break;
-#endif
-#ifdef CODEC_G723
- case NETEQ_CODEC_G723:
- *dec = new decoder_G723( pt );
- break;
-#endif
-#ifdef CODEC_PCM16B
- case kDecoderPCM16B:
- *dec = new decoder_PCM16B_NB( pt );
- break;
-#endif
-#ifdef CODEC_PCM16B_WB
- case kDecoderPCM16Bwb:
- *dec = new decoder_PCM16B_WB( pt );
- break;
-#endif
-#ifdef CODEC_PCM16B_32KHZ
- case kDecoderPCM16Bswb32kHz:
- *dec = new decoder_PCM16B_SWB32( pt );
- break;
-#endif
-#ifdef CODEC_PCM16B_48KHZ
- case kDecoderPCM16Bswb48kHz:
- *dec = new decoder_PCM16B_SWB48( pt );
- break;
-#endif
-#ifdef CODEC_DVI4
- case NETEQ_CODEC_DVI4:
- *dec = new decoder_DVI4( pt );
- break;
-#endif
-#ifdef CODEC_G722
- case kDecoderG722:
- *dec = new decoder_G722( pt );
- break;
-#endif
-#ifdef CODEC_G722_1_16
- case kDecoderG722_1_16:
- *dec = new decoder_G722_1_16( pt );
- break;
-#endif
-#ifdef CODEC_G722_1_24
- case kDecoderG722_1_24:
- *dec = new decoder_G722_1_24( pt );
- break;
-#endif
-#ifdef CODEC_G722_1_32
- case kDecoderG722_1_32:
- *dec = new decoder_G722_1_32( pt );
- break;
-#endif
-#ifdef CODEC_G722_1C_24
- case kDecoderG722_1C_24:
- *dec = new decoder_G722_1C_24( pt );
- break;
-#endif
-#ifdef CODEC_G722_1C_32
- case kDecoderG722_1C_32:
- *dec = new decoder_G722_1C_32( pt );
- break;
-#endif
-#ifdef CODEC_G722_1C_48
- case kDecoderG722_1C_48:
- *dec = new decoder_G722_1C_48( pt );
- break;
-#endif
-#ifdef CODEC_AMR
- case kDecoderAMR:
- *dec = new decoder_AMR( pt );
- break;
-#endif
-#ifdef CODEC_AMRWB
- case kDecoderAMRWB:
- *dec = new decoder_AMRWB( pt );
- break;
-#endif
-#ifdef CODEC_GSMFR
- case kDecoderGSMFR:
- *dec = new decoder_GSMFR( pt );
- break;
-#endif
-#ifdef CODEC_GSMEFR
- case NETEQ_CODEC_GSMEFR:
- *dec = new decoder_GSMEFR( pt );
- break;
-#endif
-#ifdef CODEC_G726
- case kDecoderG726_16:
- *dec = new decoder_G726_16( pt );
- break;
- case kDecoderG726_24:
- *dec = new decoder_G726_24( pt );
- break;
- case kDecoderG726_32:
- *dec = new decoder_G726_32( pt );
- break;
- case kDecoderG726_40:
- *dec = new decoder_G726_40( pt );
- break;
-#endif
-#ifdef CODEC_MELPE
- case NETEQ_CODEC_MELPE:
-#if (_MSC_VER >= 1400) && !defined(_WIN64) // only for Visual 2005 or later, and not for x64
- *dec = new decoder_MELPE( pt );
-#endif
- break;
-#endif
-#ifdef CODEC_SPEEX_8
- case kDecoderSPEEX_8:
- *dec = new decoder_SPEEX( pt, 8000 );
- break;
-#endif
-#ifdef CODEC_SPEEX_16
- case kDecoderSPEEX_16:
- *dec = new decoder_SPEEX( pt, 16000 );
- break;
-#endif
-#ifdef CODEC_CELT_32
- case kDecoderCELT_32:
- *dec = new decoder_CELT( pt, 32000 );
- break;
-#endif
-#ifdef CODEC_RED
- case kDecoderRED:
- *dec = new decoder_RED( pt );
- break;
-#endif
-#ifdef CODEC_ATEVENT_DECODE
- case kDecoderAVT:
- *dec = new decoder_AVT( pt );
- break;
-#endif
-#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
- defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
- case kDecoderCNG:
- *dec = new decoder_CNG( pt, static_cast<WebRtc_UWord16>((*it).second.fs) );
- break;
-#endif
-#ifdef CODEC_ISACLC
- case NETEQ_CODEC_ISACLC:
- *dec = new decoder_iSACLC( pt );
- break;
-#endif
-#ifdef CODEC_SILK_NB
- case NETEQ_CODEC_SILK_8:
-#if (_MSC_VER >= 1400) && !defined(_WIN64) // only for Visual 2005 or later, and not for x64
- *dec = new decoder_SILK8( pt );
-#endif
- break;
-#endif
-#ifdef CODEC_SILK_WB
- case NETEQ_CODEC_SILK_12:
-#if (_MSC_VER >= 1400) && !defined(_WIN64) // only for Visual 2005 or later, and not for x64
- *dec = new decoder_SILK12( pt );
-#endif
- break;
-#endif
-#ifdef CODEC_SILK_WB
- case NETEQ_CODEC_SILK_16:
-#if (_MSC_VER >= 1400) && !defined(_WIN64) // only for Visual 2005 or later, and not for x64
- *dec = new decoder_SILK16( pt );
-#endif
- break;
-#endif
-#ifdef CODEC_SILK_SWB
- case NETEQ_CODEC_SILK_24:
-#if (_MSC_VER >= 1400) && !defined(_WIN64) // only for Visual 2005 or later, and not for x64
- *dec = new decoder_SILK24( pt );
-#endif
- break;
-#endif
-
- default:
- printf("Unknown codec type encountered in createAndInsertDecoders\n");
- exit(0);
- }
-
- // insert into codec DB
- if (*dec)
- {
- (*dec)->loadToNetEQ(*neteq);
- }
- }
- }
-
-}
-
-
-void free_coders(std::map<WebRtc_UWord8, decoderStruct> & decoders)
-{
- std::map<WebRtc_UWord8, decoderStruct>::iterator it;
-
- for (it = decoders.begin(); it != decoders.end(); it++)
- {
- if ((*it).second.decoder[0])
- {
- delete (*it).second.decoder[0];
- }
-
- if ((*it).second.decoder[1])
- {
- delete (*it).second.decoder[1];
- }
- }
-}
-
-
-
-#include "pcm16b.h"
-#include "g711_interface.h"
-#include "isac.h"
-
-int doAPItest() {
-
- char version[20];
- void *inst;
- enum WebRtcNetEQDecoder usedCodec;
- int NetEqBufferMaxPackets, BufferSizeInBytes;
- WebRtcNetEQ_CodecDef codecInst;
- WebRtcNetEQ_RTCPStat RTCPstat;
- WebRtc_UWord32 timestamp;
- int memorySize;
- int ok;
-
- printf("API-test:\n");
-
- /* get the version string */
- WebRtcNetEQ_GetVersion(version);
- printf("NetEq version: %s\n\n", version);
-
- /* test that API functions return -1 if instance is NULL */
-#define CHECK_MINUS_ONE(x) {int errCode = x; if((errCode)!=-1){printf("\n API test failed at line %d: %s. Function did not return -1 as expected\n",__LINE__,#x); return(-1);}}
-//#define RESET_ERROR(x) ((MainInst_t*) x)->ErrorCode = 0;
- inst = NULL;
-
- CHECK_MINUS_ONE(WebRtcNetEQ_GetErrorCode(inst))
- CHECK_MINUS_ONE(WebRtcNetEQ_Assign(&inst, NULL))
-// printf("WARNING: Test of WebRtcNetEQ_Assign() is disabled due to a bug.\n");
- usedCodec=kDecoderPCMu;
- CHECK_MINUS_ONE(WebRtcNetEQ_GetRecommendedBufferSize(inst, &usedCodec, 1, kTCPLargeJitter, &NetEqBufferMaxPackets, &BufferSizeInBytes))
- CHECK_MINUS_ONE(WebRtcNetEQ_AssignBuffer(inst, NetEqBufferMaxPackets, NetEqPacketBuffer, BufferSizeInBytes))
-
- CHECK_MINUS_ONE(WebRtcNetEQ_Init(inst, 8000))
- CHECK_MINUS_ONE(WebRtcNetEQ_SetAVTPlayout(inst, 0))
- CHECK_MINUS_ONE(WebRtcNetEQ_SetExtraDelay(inst, 17))
- CHECK_MINUS_ONE(WebRtcNetEQ_SetPlayoutMode(inst, kPlayoutOn))
-
- CHECK_MINUS_ONE(WebRtcNetEQ_CodecDbReset(inst))
- CHECK_MINUS_ONE(WebRtcNetEQ_CodecDbAdd(inst, &codecInst))
- CHECK_MINUS_ONE(WebRtcNetEQ_CodecDbRemove(inst, usedCodec))
- WebRtc_Word16 temp1, temp2;
- CHECK_MINUS_ONE(WebRtcNetEQ_CodecDbGetSizeInfo(inst, &temp1, &temp2))
- CHECK_MINUS_ONE(WebRtcNetEQ_CodecDbGetCodecInfo(inst, 0, &usedCodec))
-
- CHECK_MINUS_ONE(WebRtcNetEQ_RecIn(inst, &temp1, 17, 4711))
- CHECK_MINUS_ONE(WebRtcNetEQ_RecOut(inst, &temp1, &temp2))
- CHECK_MINUS_ONE(WebRtcNetEQ_GetRTCPStats(inst, &RTCPstat)); // error here!!!
- CHECK_MINUS_ONE(WebRtcNetEQ_GetSpeechTimeStamp(inst, ×tamp))
- WebRtcNetEQOutputType temptype;
- CHECK_MINUS_ONE(WebRtcNetEQ_GetSpeechOutputType(inst, &temptype))
-
- WebRtc_UWord8 tempFlags;
- WebRtc_UWord16 utemp1, utemp2;
- CHECK_MINUS_ONE(WebRtcNetEQ_VQmonRecOutStatistics(inst, &utemp1, &utemp2, &tempFlags))
- CHECK_MINUS_ONE(WebRtcNetEQ_VQmonGetRxStatistics(inst, &utemp1, &utemp2))
-
- WebRtcNetEQ_AssignSize(&memorySize);
- CHECK_ZERO(WebRtcNetEQ_Assign(&inst, malloc(memorySize)))
-
- /* init with wrong sample frequency */
- CHECK_MINUS_ONE(WebRtcNetEQ_Init(inst, 17))
-
- /* init with correct fs */
- CHECK_ZERO(WebRtcNetEQ_Init(inst, 8000))
-
- /* GetRecommendedBufferSize with wrong codec */
- usedCodec=kDecoderReservedStart;
- ok = WebRtcNetEQ_GetRecommendedBufferSize(inst, &usedCodec, 1, kTCPLargeJitter , &NetEqBufferMaxPackets, &BufferSizeInBytes);
- if((ok!=-1) || ((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_UNKNOWN_CODEC))){
- printf("WebRtcNetEQ_GetRecommendedBufferSize() did not return proper error code for wrong codec.\n");
- printf("return value = %d; error code = %d\n", ok, WebRtcNetEQ_GetErrorCode(inst));
- }
- //RESET_ERROR(inst)
-
- /* GetRecommendedBufferSize with wrong network type */
- usedCodec = kDecoderPCMu;
- ok=WebRtcNetEQ_GetRecommendedBufferSize(inst, &usedCodec, 1, (enum WebRtcNetEQNetworkType) 4711 , &NetEqBufferMaxPackets, &BufferSizeInBytes);
- if ((ok!=-1) || ((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-FAULTY_NETWORK_TYPE))) {
- printf("WebRtcNetEQ_GetRecommendedBufferSize() did not return proper error code for wrong network type.\n");
- printf("return value = %d; error code = %d\n", ok, WebRtcNetEQ_GetErrorCode(inst));
- //RESET_ERROR(inst)
- }
- CHECK_ZERO(WebRtcNetEQ_GetRecommendedBufferSize(inst, &usedCodec, 1, kTCPLargeJitter , &NetEqBufferMaxPackets, &BufferSizeInBytes))
-
- /* try to do RecIn before assigning the packet buffer */
-/* makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, 17,4711, 1235412312);
- makeDTMFpayload(&rtp_data[12], 1, 1, 10, 100);
- ok = WebRtcNetEQ_RecIn(inst, (short *) rtp_data, 12+4, 4711);
- printf("return value = %d; error code = %d\n", ok, WebRtcNetEQ_GetErrorCode(inst));*/
-
- /* check all limits of WebRtcNetEQ_AssignBuffer */
- ok=WebRtcNetEQ_AssignBuffer(inst, NetEqBufferMaxPackets, NetEqPacketBuffer, 149<<1);
- if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-PBUFFER_INIT_ERROR))) {
- printf("WebRtcNetEQ_AssignBuffer() did not return proper error code for wrong sizeinbytes\n");
- }
- ok=WebRtcNetEQ_AssignBuffer(inst, NetEqBufferMaxPackets, NULL, BufferSizeInBytes);
- if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-PBUFFER_INIT_ERROR))) {
- printf("WebRtcNetEQ_AssignBuffer() did not return proper error code for NULL memory pointer\n");
- }
- ok=WebRtcNetEQ_AssignBuffer(inst, 1, NetEqPacketBuffer, BufferSizeInBytes);
- if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-PBUFFER_INIT_ERROR))) {
- printf("WebRtcNetEQ_AssignBuffer() did not return proper error code for wrong MaxNoOfPackets\n");
- }
- ok=WebRtcNetEQ_AssignBuffer(inst, 601, NetEqPacketBuffer, BufferSizeInBytes);
- if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-PBUFFER_INIT_ERROR))) {
- printf("WebRtcNetEQ_AssignBuffer() did not return proper error code for wrong MaxNoOfPackets\n");
- }
-
- /* do correct assignbuffer */
- CHECK_ZERO(WebRtcNetEQ_AssignBuffer(inst, NetEqBufferMaxPackets, NetEqPacketBuffer, BufferSizeInBytes))
-
- ok=WebRtcNetEQ_SetExtraDelay(inst, -1);
- if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-FAULTY_DELAYVALUE))) {
- printf("WebRtcNetEQ_SetExtraDelay() did not return proper error code for too small delay\n");
- }
- ok=WebRtcNetEQ_SetExtraDelay(inst, 1001);
- if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-FAULTY_DELAYVALUE))) {
- printf("WebRtcNetEQ_SetExtraDelay() did not return proper error code for too large delay\n");
- }
-
- ok=WebRtcNetEQ_SetPlayoutMode(inst,(enum WebRtcNetEQPlayoutMode) 4711);
- if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-FAULTY_PLAYOUTMODE))) {
- printf("WebRtcNetEQ_SetPlayoutMode() did not return proper error code for wrong mode\n");
- }
-
- /* number of codecs should return zero before adding any codecs */
- WebRtcNetEQ_CodecDbGetSizeInfo(inst, &temp1, &temp2);
- if(temp1!=0)
- printf("WebRtcNetEQ_CodecDbGetSizeInfo() return non-zero number of codecs in DB before adding any codecs\n");
-
- /* get info from empty database */
- ok=WebRtcNetEQ_CodecDbGetCodecInfo(inst, 17, &usedCodec);
- if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_NOT_EXIST1))) {
- printf("WebRtcNetEQ_CodecDbGetCodecInfo() did not return proper error code for out-of-range entry number\n");
- }
-
- /* remove codec from empty database */
- ok=WebRtcNetEQ_CodecDbRemove(inst,kDecoderPCMa);
- if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_NOT_EXIST4))) {
- printf("WebRtcNetEQ_CodecDbRemove() did not return proper error code when removing codec that has not been added\n");
- }
-
- /* add codec with unsupported fs */
-#ifdef CODEC_PCM16B
-#ifndef NETEQ_48KHZ_WIDEBAND
- SET_CODEC_PAR(codecInst,kDecoderPCM16Bswb48kHz,77,NULL,48000);
- SET_PCM16B_SWB48_FUNCTIONS(codecInst);
- ok=WebRtcNetEQ_CodecDbAdd(inst, &codecInst);
- if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_UNSUPPORTED_FS))) {
- printf("WebRtcNetEQ_CodecDbAdd() did not return proper error code when adding codec with unsupported sample freq\n");
- }
-#else
- printf("Could not test adding codec with unsupported sample frequency since NetEQ is compiled with 48kHz support.\n");
-#endif
-#else
- printf("Could not test adding codec with unsupported sample frequency since NetEQ is compiled without PCM16B support.\n");
-#endif
-
- /* add two codecs with identical payload types */
- SET_CODEC_PAR(codecInst,kDecoderPCMa,17,NULL,8000);
- SET_PCMA_FUNCTIONS(codecInst);
- CHECK_ZERO(WebRtcNetEQ_CodecDbAdd(inst, &codecInst))
-
- SET_CODEC_PAR(codecInst,kDecoderPCMu,17,NULL,8000);
- SET_PCMU_FUNCTIONS(codecInst);
- ok=WebRtcNetEQ_CodecDbAdd(inst, &codecInst);
- if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_PAYLOAD_TAKEN))) {
- printf("WebRtcNetEQ_CodecDbAdd() did not return proper error code when adding two codecs with identical payload types\n");
- }
-
- /* try adding several payload types for CNG codecs */
- SET_CODEC_PAR(codecInst,kDecoderCNG,105,NULL,16000);
- SET_CNG_FUNCTIONS(codecInst);
- CHECK_ZERO(WebRtcNetEQ_CodecDbAdd(inst, &codecInst))
- SET_CODEC_PAR(codecInst,kDecoderCNG,13,NULL,8000);
- SET_CNG_FUNCTIONS(codecInst);
- CHECK_ZERO(WebRtcNetEQ_CodecDbAdd(inst, &codecInst))
-
- /* try adding a speech codec over a CNG codec */
- SET_CODEC_PAR(codecInst,kDecoderISAC,105,NULL,16000); /* same as WB CNG above */
- SET_ISAC_FUNCTIONS(codecInst);
- ok=WebRtcNetEQ_CodecDbAdd(inst, &codecInst);
- if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_PAYLOAD_TAKEN))) {
- printf("WebRtcNetEQ_CodecDbAdd() did not return proper error code when adding a speech codec over a CNG codec\n");
- }
-
- /* try adding a CNG codec over a speech codec */
- SET_CODEC_PAR(codecInst,kDecoderCNG,17,NULL,32000); /* same as PCMU above */
- SET_CNG_FUNCTIONS(codecInst);
- ok=WebRtcNetEQ_CodecDbAdd(inst, &codecInst);
- if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_PAYLOAD_TAKEN))) {
- printf("WebRtcNetEQ_CodecDbAdd() did not return proper error code when adding a speech codec over a CNG codec\n");
- }
-
-
- /* remove codec out of range */
- ok=WebRtcNetEQ_CodecDbRemove(inst,kDecoderReservedStart);
- if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_UNSUPPORTED_CODEC))) {
- printf("WebRtcNetEQ_CodecDbRemove() did not return proper error code when removing codec that is out of range\n");
- }
- ok=WebRtcNetEQ_CodecDbRemove(inst,kDecoderReservedEnd);
- if((ok!=-1)||((ok==-1)&&(WebRtcNetEQ_GetErrorCode(inst)!=-CODEC_DB_UNSUPPORTED_CODEC))) {
- printf("WebRtcNetEQ_CodecDbRemove() did not return proper error code when removing codec that is out of range\n");
- }
-
- /*SET_CODEC_PAR(codecInst,kDecoderEG711a,NETEQ_CODEC_EG711A_PT,NetEqiPCMAState,8000);
- SET_IPCMA_FUNCTIONS(codecInst);
- CHECK_ZERO(WebRtcNetEQ_CodecDbAdd(inst, &codecInst))
-*/
- free(inst);
-
- return(0);
-
-}
diff --git a/trunk/src/modules/audio_coding/neteq/test/PayloadTypes.h b/trunk/src/modules/audio_coding/neteq/test/PayloadTypes.h
deleted file mode 100644
index f6cc3da..0000000
--- a/trunk/src/modules/audio_coding/neteq/test/PayloadTypes.h
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/* PayloadTypes.h */
-/* Used by NetEqRTPplay application */
-
-/* RTP defined codepoints */
-#define NETEQ_CODEC_PCMU_PT 0
-#define NETEQ_CODEC_GSMFR_PT 3
-#define NETEQ_CODEC_G723_PT 4
-#define NETEQ_CODEC_DVI4_PT 125 // 8 kHz version
-//#define NETEQ_CODEC_DVI4_16_PT 6 // 16 kHz version
-#define NETEQ_CODEC_PCMA_PT 8
-#define NETEQ_CODEC_G722_PT 9
-#define NETEQ_CODEC_CN_PT 13
-//#define NETEQ_CODEC_G728_PT 15
-//#define NETEQ_CODEC_DVI4_11_PT 16 // 11.025 kHz version
-//#define NETEQ_CODEC_DVI4_22_PT 17 // 22.050 kHz version
-#define NETEQ_CODEC_G729_PT 18
-
-/* Dynamic RTP codepoints as defined in VoiceEngine (file VEAPI.cpp) */
-#define NETEQ_CODEC_IPCMWB_PT 97
-#define NETEQ_CODEC_SPEEX8_PT 98
-#define NETEQ_CODEC_SPEEX16_PT 99
-#define NETEQ_CODEC_EG711U_PT 100
-#define NETEQ_CODEC_EG711A_PT 101
-#define NETEQ_CODEC_ILBC_PT 102
-#define NETEQ_CODEC_ISAC_PT 103
-#define NETEQ_CODEC_ISACLC_PT 119
-#define NETEQ_CODEC_ISACSWB_PT 104
-#define NETEQ_CODEC_AVT_PT 106
-#define NETEQ_CODEC_G722_1_16_PT 108
-#define NETEQ_CODEC_G722_1_24_PT 109
-#define NETEQ_CODEC_G722_1_32_PT 110
-#define NETEQ_CODEC_SC3_PT 111
-#define NETEQ_CODEC_AMR_PT 112
-#define NETEQ_CODEC_GSMEFR_PT 113
-//#define NETEQ_CODEC_ILBCRCU_PT 114
-#define NETEQ_CODEC_G726_16_PT 115
-#define NETEQ_CODEC_G726_24_PT 116
-#define NETEQ_CODEC_G726_32_PT 121
-#define NETEQ_CODEC_RED_PT 117
-#define NETEQ_CODEC_G726_40_PT 118
-//#define NETEQ_CODEC_ENERGY_PT 120
-#define NETEQ_CODEC_CN_WB_PT 105
-#define NETEQ_CODEC_CN_SWB_PT 126
-#define NETEQ_CODEC_G729_1_PT 107
-#define NETEQ_CODEC_G729D_PT 123
-#define NETEQ_CODEC_MELPE_PT 124
-#define NETEQ_CODEC_CELT32_PT 114
-
-/* Extra dynamic codepoints */
-#define NETEQ_CODEC_AMRWB_PT 120
-#define NETEQ_CODEC_PCM16B_PT 93
-#define NETEQ_CODEC_PCM16B_WB_PT 94
-#define NETEQ_CODEC_PCM16B_SWB32KHZ_PT 95
-#define NETEQ_CODEC_PCM16B_SWB48KHZ_PT 96
-#define NETEQ_CODEC_MPEG4AAC_PT 122
-
-
-/* Not default in VoiceEngine */
-#define NETEQ_CODEC_G722_1C_24_PT 84
-#define NETEQ_CODEC_G722_1C_32_PT 85
-#define NETEQ_CODEC_G722_1C_48_PT 86
-
-#define NETEQ_CODEC_SILK_8_PT 80
-#define NETEQ_CODEC_SILK_12_PT 81
-#define NETEQ_CODEC_SILK_16_PT 82
-#define NETEQ_CODEC_SILK_24_PT 83
-
diff --git a/trunk/src/modules/audio_coding/neteq/test/RTPanalyze.cc b/trunk/src/modules/audio_coding/neteq/test/RTPanalyze.cc
deleted file mode 100644
index 12617dd..0000000
--- a/trunk/src/modules/audio_coding/neteq/test/RTPanalyze.cc
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <assert.h>
-#include <stdio.h>
-#include <vector>
-
-#include "modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
-
-enum {
- kRedPayloadType = 127
-};
-
-int main(int argc, char* argv[]) {
- FILE* in_file = fopen(argv[1], "rb");
- if (!in_file) {
- printf("Cannot open input file %s\n", argv[1]);
- return -1;
- }
- printf("Input file: %s\n", argv[1]);
-
- FILE* out_file = fopen(argv[2], "wt");
- if (!out_file) {
- printf("Cannot open output file %s\n", argv[2]);
- return -1;
- }
- printf("Output file: %s\n\n", argv[2]);
-
- // Print file header.
- fprintf(out_file, "SeqNo TimeStamp SendTime Size PT M\n");
-
- // Read file header.
- NETEQTEST_RTPpacket::skipFileHeader(in_file);
- NETEQTEST_RTPpacket packet;
-
- while (packet.readFromFile(in_file) >= 0) {
- // Write packet data to file.
- fprintf(out_file, "%5u %10u %10u %5i %5i %2i\n",
- packet.sequenceNumber(), packet.timeStamp(), packet.time(),
- packet.dataLen(), packet.payloadType(), packet.markerBit());
- if (packet.payloadType() == kRedPayloadType) {
- WebRtcNetEQ_RTPInfo red_header;
- int len;
- int red_index = 0;
- while ((len = packet.extractRED(red_index++, red_header)) >= 0) {
- fprintf(out_file, "* %5u %10u %10u %5i %5i\n",
- red_header.sequenceNumber, red_header.timeStamp,
- packet.time(), len, red_header.payloadType);
- }
- assert(red_index > 1); // We must get at least one payload.
- }
- }
-
- fclose(in_file);
- fclose(out_file);
-
- return 0;
-}
diff --git a/trunk/src/modules/audio_coding/neteq/test/RTPcat.cc b/trunk/src/modules/audio_coding/neteq/test/RTPcat.cc
deleted file mode 100644
index 001b00b..0000000
--- a/trunk/src/modules/audio_coding/neteq/test/RTPcat.cc
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-
-#include <algorithm>
-#include <vector>
-
-#include "gtest/gtest.h"
-#include "modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
-
-#define FIRSTLINELEN 40
-
-int main(int argc, char* argv[]) {
- if (argc < 3) {
- printf("Usage: RTPcat in1.rtp int2.rtp [...] out.rtp\n");
- exit(1);
- }
-
- FILE* in_file = fopen(argv[1], "rb");
- if (!in_file) {
- printf("Cannot open input file %s\n", argv[1]);
- return -1;
- }
-
- FILE* out_file = fopen(argv[argc - 1], "wb"); // Last parameter is out file.
- if (!out_file) {
- printf("Cannot open output file %s\n", argv[argc - 1]);
- return -1;
- }
- printf("Output RTP file: %s\n\n", argv[argc - 1]);
-
- // Read file header and write directly to output file.
- char firstline[FIRSTLINELEN];
- const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
- EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, in_file) != NULL);
- EXPECT_GT(fputs(firstline, out_file), 0);
- EXPECT_EQ(kRtpDumpHeaderSize, fread(firstline, 1, kRtpDumpHeaderSize,
- in_file));
- EXPECT_EQ(kRtpDumpHeaderSize, fwrite(firstline, 1, kRtpDumpHeaderSize,
- out_file));
-
- // Close input file and re-open it later (easier to write the loop below).
- fclose(in_file);
-
- for (int i = 1; i < argc - 1; i++) {
- in_file = fopen(argv[i], "rb");
- if (!in_file) {
- printf("Cannot open input file %s\n", argv[i]);
- return -1;
- }
- printf("Input RTP file: %s\n", argv[i]);
-
- NETEQTEST_RTPpacket::skipFileHeader(in_file);
- NETEQTEST_RTPpacket packet;
- int pack_len = packet.readFromFile(in_file);
- if (pack_len < 0) {
- exit(1);
- }
- while (pack_len >= 0) {
- packet.writeToFile(out_file);
- pack_len = packet.readFromFile(in_file);
- }
- fclose(in_file);
- }
- fclose(out_file);
- return 0;
-}
diff --git a/trunk/src/modules/audio_coding/neteq/test/RTPchange.cc b/trunk/src/modules/audio_coding/neteq/test/RTPchange.cc
deleted file mode 100644
index ecbd81c..0000000
--- a/trunk/src/modules/audio_coding/neteq/test/RTPchange.cc
+++ /dev/null
@@ -1,128 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-
-#include <algorithm>
-#include <vector>
-
-#include "gtest/gtest.h"
-#include "modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
-
-#define FIRSTLINELEN 40
-
-static bool pktCmp(NETEQTEST_RTPpacket *a, NETEQTEST_RTPpacket *b) {
- return (a->time() < b->time());
-}
-
-int main(int argc, char* argv[]) {
- FILE* in_file = fopen(argv[1], "rb");
- if (!in_file) {
- printf("Cannot open input file %s\n", argv[1]);
- return -1;
- }
- printf("Input RTP file: %s\n", argv[1]);
-
- FILE* stat_file = fopen(argv[2], "rt");
- if (!stat_file) {
- printf("Cannot open timing file %s\n", argv[2]);
- return -1;
- }
- printf("Timing file: %s\n", argv[2]);
-
- FILE* out_file = fopen(argv[3], "wb");
- if (!out_file) {
- printf("Cannot open output file %s\n", argv[3]);
- return -1;
- }
- printf("Output RTP file: %s\n\n", argv[3]);
-
- // Read all statistics and insert into map.
- // Read first line.
- char temp_str[100];
- if (fgets(temp_str, 100, stat_file) == NULL) {
- printf("Failed to read timing file %s\n", argv[2]);
- return -1;
- }
- // Define map.
- std::map<std::pair<uint16_t, uint32_t>, uint32_t> packet_stats;
- uint16_t seq_no;
- uint32_t ts;
- uint32_t send_time;
-
- while (fscanf(stat_file,
- "%hu %u %u %*i %*i\n", &seq_no, &ts, &send_time) == 3) {
- std::pair<uint16_t, uint32_t>
- temp_pair = std::pair<uint16_t, uint32_t>(seq_no, ts);
-
- packet_stats[temp_pair] = send_time;
- }
-
- fclose(stat_file);
-
- // Read file header and write directly to output file.
- char first_line[FIRSTLINELEN];
- if (fgets(first_line, FIRSTLINELEN, in_file) == NULL) {
- printf("Failed to read first line of input file %s\n", argv[1]);
- return -1;
- }
- fputs(first_line, out_file);
- // start_sec + start_usec + source + port + padding
- const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
- if (fread(first_line, 1, kRtpDumpHeaderSize, in_file)
- != kRtpDumpHeaderSize) {
- printf("Failed to read RTP dump header from input file %s\n", argv[1]);
- return -1;
- }
- if (fwrite(first_line, 1, kRtpDumpHeaderSize, out_file)
- != kRtpDumpHeaderSize) {
- printf("Failed to write RTP dump header to output file %s\n", argv[3]);
- return -1;
- }
-
- std::vector<NETEQTEST_RTPpacket *> packet_vec;
-
- while (1) {
- // Insert in vector.
- NETEQTEST_RTPpacket *new_packet = new NETEQTEST_RTPpacket();
- if (new_packet->readFromFile(in_file) < 0) {
- // End of file.
- break;
- }
-
- // Look for new send time in statistics vector.
- std::pair<uint16_t, uint32_t> temp_pair =
- std::pair<uint16_t, uint32_t>(new_packet->sequenceNumber(),
- new_packet->timeStamp());
-
- uint32_t new_send_time = packet_stats[temp_pair];
- new_packet->setTime(new_send_time); // Set new send time.
- packet_vec.push_back(new_packet); // Insert in vector.
- }
-
- // Sort the vector according to send times.
- std::sort(packet_vec.begin(), packet_vec.end(), pktCmp);
-
- std::vector<NETEQTEST_RTPpacket *>::iterator it;
- for (it = packet_vec.begin(); it != packet_vec.end(); it++) {
- // Write to out file.
- if ((*it)->writeToFile(out_file) < 0) {
- printf("Error writing to file\n");
- return -1;
- }
- // Delete packet.
- delete *it;
- }
-
- fclose(in_file);
- fclose(out_file);
-
- return 0;
-}
diff --git a/trunk/src/modules/audio_coding/neteq/test/RTPencode.cc b/trunk/src/modules/audio_coding/neteq/test/RTPencode.cc
deleted file mode 100644
index f328794..0000000
--- a/trunk/src/modules/audio_coding/neteq/test/RTPencode.cc
+++ /dev/null
@@ -1,2031 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//TODO(hlundin): Reformat file to meet style guide.
-
-/* header includes */
-#include "typedefs.h"
-#include "stdio.h"
-#include "webrtc_neteq.h" // needed for enum WebRtcNetEQDecoder
-#include <string.h>
-#include <stdlib.h>
-#include <cassert>
-
-#ifdef WIN32
-#include <winsock2.h>
-#endif
-#ifdef WEBRTC_LINUX
-#include <netinet/in.h>
-#endif
-
-
-/************************/
-/* Define payload types */
-/************************/
-
-#include "PayloadTypes.h"
-
-
-
-/*********************/
-/* Misc. definitions */
-/*********************/
-
-#define STOPSENDTIME 3000
-#define RESTARTSENDTIME 0 //162500
-#define FIRSTLINELEN 40
-#define CHECK_NOT_NULL(a) if((a)==0){printf("\n %s \n line: %d \nerror at %s\n",__FILE__,__LINE__,#a );return(-1);}
-
-//#define MULTIPLE_SAME_TIMESTAMP
-#define REPEAT_PACKET_DISTANCE 17
-#define REPEAT_PACKET_COUNT 1 // number of extra packets to send
-
-//#define INSERT_OLD_PACKETS
-#define OLD_PACKET 5 // how many seconds too old should the packet be?
-
-//#define TIMESTAMP_WRAPAROUND
-
-//#define RANDOM_DATA
-//#define RANDOM_PAYLOAD_DATA
-#define RANDOM_SEED 10
-
-//#define INSERT_DTMF_PACKETS
-//#define NO_DTMF_OVERDUB
-#define DTMF_PACKET_INTERVAL 2000
-#define DTMF_DURATION 500
-
-#define STEREO_MODE_FRAME 0
-#define STEREO_MODE_SAMPLE_1 1 //1 octet per sample
-#define STEREO_MODE_SAMPLE_2 2 //2 octets per sample
-
-/*************************/
-/* Function declarations */
-/*************************/
-
-void NetEQTest_GetCodec_and_PT(char * name, enum WebRtcNetEQDecoder *codec, int *PT, int frameLen, int *fs, int *bitrate, int *useRed);
-int NetEQTest_init_coders(enum WebRtcNetEQDecoder coder, int enc_frameSize, int bitrate, int sampfreq , int vad, int numChannels);
-void defineCodecs(enum WebRtcNetEQDecoder *usedCodec, int *noOfCodecs );
-int NetEQTest_free_coders(enum WebRtcNetEQDecoder coder, int numChannels);
-int NetEQTest_encode(int coder, WebRtc_Word16 *indata, int frameLen, unsigned char * encoded,int sampleRate , int * vad, int useVAD, int bitrate, int numChannels);
-void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, WebRtc_UWord32 timestamp, WebRtc_UWord32 ssrc);
-int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, WebRtc_UWord32 *timestamp, WebRtc_UWord16 *blockLen,
- int seqNo, WebRtc_UWord32 ssrc);
-int makeDTMFpayload(unsigned char* payload_data, int Event, int End, int Volume, int Duration);
-void stereoDeInterleave(WebRtc_Word16* audioSamples, int numSamples);
-void stereoInterleave(unsigned char* data, int dataLen, int stride);
-
-/*********************/
-/* Codec definitions */
-/*********************/
-
-#include "webrtc_vad.h"
-
-#if ((defined CODEC_PCM16B)||(defined NETEQ_ARBITRARY_CODEC))
- #include "pcm16b.h"
-#endif
-#ifdef CODEC_G711
- #include "g711_interface.h"
-#endif
-#ifdef CODEC_G729
- #include "G729Interface.h"
-#endif
-#ifdef CODEC_G729_1
- #include "G729_1Interface.h"
-#endif
-#ifdef CODEC_AMR
- #include "AMRInterface.h"
- #include "AMRCreation.h"
-#endif
-#ifdef CODEC_AMRWB
- #include "AMRWBInterface.h"
- #include "AMRWBCreation.h"
-#endif
-#ifdef CODEC_ILBC
- #include "ilbc.h"
-#endif
-#if (defined CODEC_ISAC || defined CODEC_ISAC_SWB)
- #include "isac.h"
-#endif
-#ifdef NETEQ_ISACFIX_CODEC
- #include "isacfix.h"
- #ifdef CODEC_ISAC
- #error Cannot have both ISAC and ISACfix defined. Please de-select one in the beginning of RTPencode.cpp
- #endif
-#endif
-#ifdef CODEC_G722
- #include "g722_interface.h"
-#endif
-#ifdef CODEC_G722_1_24
- #include "G722_1Interface.h"
-#endif
-#ifdef CODEC_G722_1_32
- #include "G722_1Interface.h"
-#endif
-#ifdef CODEC_G722_1_16
- #include "G722_1Interface.h"
-#endif
-#ifdef CODEC_G722_1C_24
- #include "G722_1Interface.h"
-#endif
-#ifdef CODEC_G722_1C_32
- #include "G722_1Interface.h"
-#endif
-#ifdef CODEC_G722_1C_48
- #include "G722_1Interface.h"
-#endif
-#ifdef CODEC_G726
- #include "G726Creation.h"
- #include "G726Interface.h"
-#endif
-#ifdef CODEC_GSMFR
- #include "GSMFRInterface.h"
- #include "GSMFRCreation.h"
-#endif
-#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
- defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
- #include "webrtc_cng.h"
-#endif
-#if ((defined CODEC_SPEEX_8)||(defined CODEC_SPEEX_16))
- #include "SpeexInterface.h"
-#endif
-#ifdef CODEC_CELT_32
-#include "celt_interface.h"
-#endif
-
-
-/***********************************/
-/* Global codec instance variables */
-/***********************************/
-
-WebRtcVadInst *VAD_inst[2];
-
-#ifdef CODEC_G722
- G722EncInst *g722EncState[2];
-#endif
-
-#ifdef CODEC_G722_1_24
- G722_1_24_encinst_t *G722_1_24enc_inst[2];
-#endif
-#ifdef CODEC_G722_1_32
- G722_1_32_encinst_t *G722_1_32enc_inst[2];
-#endif
-#ifdef CODEC_G722_1_16
- G722_1_16_encinst_t *G722_1_16enc_inst[2];
-#endif
-#ifdef CODEC_G722_1C_24
- G722_1C_24_encinst_t *G722_1C_24enc_inst[2];
-#endif
-#ifdef CODEC_G722_1C_32
- G722_1C_32_encinst_t *G722_1C_32enc_inst[2];
-#endif
-#ifdef CODEC_G722_1C_48
- G722_1C_48_encinst_t *G722_1C_48enc_inst[2];
-#endif
-#ifdef CODEC_G726
- G726_encinst_t *G726enc_inst[2];
-#endif
-#ifdef CODEC_G729
- G729_encinst_t *G729enc_inst[2];
-#endif
-#ifdef CODEC_G729_1
- G729_1_inst_t *G729_1_inst[2];
-#endif
-#ifdef CODEC_AMR
- AMR_encinst_t *AMRenc_inst[2];
- WebRtc_Word16 AMR_bitrate;
-#endif
-#ifdef CODEC_AMRWB
- AMRWB_encinst_t *AMRWBenc_inst[2];
- WebRtc_Word16 AMRWB_bitrate;
-#endif
-#ifdef CODEC_ILBC
- iLBC_encinst_t *iLBCenc_inst[2];
-#endif
-#ifdef CODEC_ISAC
- ISACStruct *ISAC_inst[2];
-#endif
-#ifdef NETEQ_ISACFIX_CODEC
- ISACFIX_MainStruct *ISAC_inst[2];
-#endif
-#ifdef CODEC_ISAC_SWB
- ISACStruct *ISACSWB_inst[2];
-#endif
-#ifdef CODEC_GSMFR
- GSMFR_encinst_t *GSMFRenc_inst[2];
-#endif
-#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
- defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
- CNG_enc_inst *CNGenc_inst[2];
-#endif
-#ifdef CODEC_SPEEX_8
- SPEEX_encinst_t *SPEEX8enc_inst[2];
-#endif
-#ifdef CODEC_SPEEX_16
- SPEEX_encinst_t *SPEEX16enc_inst[2];
-#endif
-#ifdef CODEC_CELT_32
- CELT_encinst_t *CELT32enc_inst[2];
-#endif
-#ifdef CODEC_G711
- void *G711state[2]={NULL, NULL};
-#endif
-
-
-int main(int argc, char* argv[])
-{
- int packet_size, fs;
- enum WebRtcNetEQDecoder usedCodec;
- int payloadType;
- int bitrate = 0;
- int useVAD, vad;
- int useRed=0;
- int len, enc_len;
- WebRtc_Word16 org_data[4000];
- unsigned char rtp_data[8000];
- WebRtc_Word16 seqNo=0xFFF;
- WebRtc_UWord32 ssrc=1235412312;
- WebRtc_UWord32 timestamp=0xAC1245;
- WebRtc_UWord16 length, plen;
- WebRtc_UWord32 offset;
- double sendtime = 0;
- int red_PT[2] = {0};
- WebRtc_UWord32 red_TS[2] = {0};
- WebRtc_UWord16 red_len[2] = {0};
- int RTPheaderLen=12;
- unsigned char red_data[8000];
-#ifdef INSERT_OLD_PACKETS
- WebRtc_UWord16 old_length, old_plen;
- int old_enc_len;
- int first_old_packet=1;
- unsigned char old_rtp_data[8000];
- int packet_age=0;
-#endif
-#ifdef INSERT_DTMF_PACKETS
- int NTone = 1;
- int DTMFfirst = 1;
- WebRtc_UWord32 DTMFtimestamp;
- bool dtmfSent = false;
-#endif
- bool usingStereo = false;
- int stereoMode = 0;
- int numChannels = 1;
-
- /* check number of parameters */
- if ((argc != 6) && (argc != 7)) {
- /* print help text and exit */
- printf("Application to encode speech into an RTP stream.\n");
- printf("The program reads a PCM file and encodes is using the specified codec.\n");
- printf("The coded speech is packetized in RTP packest and written to the output file.\n");
- printf("The format of the RTP stream file is simlilar to that of rtpplay,\n");
- printf("but with the receive time euqal to 0 for all packets.\n");
- printf("Usage:\n\n");
- printf("%s PCMfile RTPfile frameLen codec useVAD bitrate\n", argv[0]);
- printf("where:\n");
-
- printf("PCMfile : PCM speech input file\n\n");
-
- printf("RTPfile : RTP stream output file\n\n");
-
- printf("frameLen : 80...960... Number of samples per packet (limit depends on codec)\n\n");
-
- printf("codecName\n");
-#ifdef CODEC_PCM16B
- printf(" : pcm16b 16 bit PCM (8kHz)\n");
-#endif
-#ifdef CODEC_PCM16B_WB
- printf(" : pcm16b_wb 16 bit PCM (16kHz)\n");
-#endif
-#ifdef CODEC_PCM16B_32KHZ
- printf(" : pcm16b_swb32 16 bit PCM (32kHz)\n");
-#endif
-#ifdef CODEC_PCM16B_48KHZ
- printf(" : pcm16b_swb48 16 bit PCM (48kHz)\n");
-#endif
-#ifdef CODEC_G711
- printf(" : pcma g711 A-law (8kHz)\n");
-#endif
-#ifdef CODEC_G711
- printf(" : pcmu g711 u-law (8kHz)\n");
-#endif
-#ifdef CODEC_G729
- printf(" : g729 G729 (8kHz and 8kbps) CELP (One-Three frame(s)/packet)\n");
-#endif
-#ifdef CODEC_G729_1
- printf(" : g729.1 G729.1 (16kHz) variable rate (8--32 kbps)\n");
-#endif
-#ifdef CODEC_G722_1_16
- printf(" : g722.1_16 G722.1 coder (16kHz) (g722.1 with 16kbps)\n");
-#endif
-#ifdef CODEC_G722_1_24
- printf(" : g722.1_24 G722.1 coder (16kHz) (the 24kbps version)\n");
-#endif
-#ifdef CODEC_G722_1_32
- printf(" : g722.1_32 G722.1 coder (16kHz) (the 32kbps version)\n");
-#endif
-#ifdef CODEC_G722_1C_24
- printf(" : g722.1C_24 G722.1 C coder (32kHz) (the 24kbps version)\n");
-#endif
-#ifdef CODEC_G722_1C_32
- printf(" : g722.1C_32 G722.1 C coder (32kHz) (the 32kbps version)\n");
-#endif
-#ifdef CODEC_G722_1C_48
- printf(" : g722.1C_48 G722.1 C coder (32kHz) (the 48kbps)\n");
-#endif
-
-#ifdef CODEC_G726
- printf(" : g726_16 G726 coder (8kHz) 16kbps\n");
- printf(" : g726_24 G726 coder (8kHz) 24kbps\n");
- printf(" : g726_32 G726 coder (8kHz) 32kbps\n");
- printf(" : g726_40 G726 coder (8kHz) 40kbps\n");
-#endif
-#ifdef CODEC_AMR
- printf(" : AMRXk Adaptive Multi Rate CELP codec (8kHz)\n");
- printf(" X = 4.75, 5.15, 5.9, 6.7, 7.4, 7.95, 10.2 or 12.2\n");
-#endif
-#ifdef CODEC_AMRWB
- printf(" : AMRwbXk Adaptive Multi Rate Wideband CELP codec (16kHz)\n");
- printf(" X = 7, 9, 12, 14, 16, 18, 20, 23 or 24\n");
-#endif
-#ifdef CODEC_ILBC
- printf(" : ilbc iLBC codec (8kHz and 13.8kbps)\n");
-#endif
-#ifdef CODEC_ISAC
- printf(" : isac iSAC (16kHz and 32.0 kbps). To set rate specify a rate parameter as last parameter\n");
-#endif
-#ifdef CODEC_ISAC_SWB
- printf(" : isacswb iSAC SWB (32kHz and 32.0-52.0 kbps). To set rate specify a rate parameter as last parameter\n");
-#endif
-#ifdef CODEC_GSMFR
- printf(" : gsmfr GSM FR codec (8kHz and 13kbps)\n");
-#endif
-#ifdef CODEC_G722
- printf(" : g722 g722 coder (16kHz) (the 64kbps version)\n");
-#endif
-#ifdef CODEC_SPEEX_8
- printf(" : speex8 speex coder (8 kHz)\n");
-#endif
-#ifdef CODEC_SPEEX_16
- printf(" : speex16 speex coder (16 kHz)\n");
-#endif
-#ifdef CODEC_CELT_32
- printf(" : celt32 celt coder (32 kHz)\n");
-#endif
-#ifdef CODEC_RED
-#ifdef CODEC_G711
- printf(" : red_pcm Redundancy RTP packet with 2*G711A frames\n");
-#endif
-#ifdef CODEC_ISAC
- printf(" : red_isac Redundancy RTP packet with 2*iSAC frames\n");
-#endif
-#endif
- printf("\n");
-
-#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
- defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
- printf("useVAD : 0 Voice Activity Detection is switched off\n");
- printf(" : 1 Voice Activity Detection is switched on\n\n");
-#else
- printf("useVAD : 0 Voice Activity Detection switched off (on not supported)\n\n");
-#endif
- printf("bitrate : Codec bitrate in bps (only applies to vbr codecs)\n\n");
-
- return(0);
- }
-
- FILE* in_file=fopen(argv[1],"rb");
- CHECK_NOT_NULL(in_file);
- printf("Input file: %s\n",argv[1]);
- FILE* out_file=fopen(argv[2],"wb");
- CHECK_NOT_NULL(out_file);
- printf("Output file: %s\n\n",argv[2]);
- packet_size=atoi(argv[3]);
- CHECK_NOT_NULL(packet_size);
- printf("Packet size: %i\n",packet_size);
-
- // check for stereo
- if(argv[4][strlen(argv[4])-1] == '*') {
- // use stereo
- usingStereo = true;
- numChannels = 2;
- argv[4][strlen(argv[4])-1] = '\0';
- }
-
- NetEQTest_GetCodec_and_PT(argv[4], &usedCodec, &payloadType, packet_size, &fs, &bitrate, &useRed);
-
- if(useRed) {
- RTPheaderLen = 12 + 4 + 1; /* standard RTP = 12; 4 bytes per redundant payload, except last one which is 1 byte */
- }
-
- useVAD=atoi(argv[5]);
-#if !(defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
- defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
- if (useVAD!=0) {
- printf("Error: this simulation does not support VAD/DTX/CNG\n");
- }
-#endif
-
- // check stereo type
- if(usingStereo)
- {
- switch(usedCodec)
- {
- // sample based codecs
- case kDecoderPCMu:
- case kDecoderPCMa:
- case kDecoderG722:
- {
- // 1 octet per sample
- stereoMode = STEREO_MODE_SAMPLE_1;
- break;
- }
- case kDecoderPCM16B:
- case kDecoderPCM16Bwb:
- case kDecoderPCM16Bswb32kHz:
- case kDecoderPCM16Bswb48kHz:
- {
- // 2 octets per sample
- stereoMode = STEREO_MODE_SAMPLE_2;
- break;
- }
-
- // fixed-rate frame codecs (with internal VAD)
- case kDecoderG729:
- {
- if(useVAD) {
- printf("Cannot use codec-internal VAD and stereo\n");
- exit(0);
- }
- // break intentionally omitted
- }
- case kDecoderG722_1_16:
- case kDecoderG722_1_24:
- case kDecoderG722_1_32:
- case kDecoderG722_1C_24:
- case kDecoderG722_1C_32:
- case kDecoderG722_1C_48:
- {
- stereoMode = STEREO_MODE_FRAME;
- break;
- }
- default:
- {
- printf("Cannot use codec %s as stereo codec\n", argv[4]);
- exit(0);
- }
- }
- }
-
- if ((usedCodec == kDecoderISAC) || (usedCodec == kDecoderISACswb))
- {
- if (argc != 7)
- {
- if (usedCodec == kDecoderISAC)
- {
- bitrate = 32000;
- printf(
- "Running iSAC at default bitrate of 32000 bps (to specify explicitly add the bps as last parameter)\n");
- }
- else // (usedCodec==kDecoderISACswb)
- {
- bitrate = 56000;
- printf(
- "Running iSAC at default bitrate of 56000 bps (to specify explicitly add the bps as last parameter)\n");
- }
- }
- else
- {
- bitrate = atoi(argv[6]);
- if (usedCodec == kDecoderISAC)
- {
- if ((bitrate < 10000) || (bitrate > 32000))
- {
- printf(
- "Error: iSAC bitrate must be between 10000 and 32000 bps (%i is invalid)\n",
- bitrate);
- exit(0);
- }
- printf("Running iSAC at bitrate of %i bps\n", bitrate);
- }
- else // (usedCodec==kDecoderISACswb)
- {
- if ((bitrate < 32000) || (bitrate > 56000))
- {
- printf(
- "Error: iSAC SWB bitrate must be between 32000 and 56000 bps (%i is invalid)\n",
- bitrate);
- exit(0);
- }
- }
- }
- }
- else
- {
- if (argc == 7)
- {
- printf(
- "Error: Bitrate parameter can only be specified for iSAC, G.723, and G.729.1\n");
- exit(0);
- }
- }
-
- if(useRed) {
- printf("Redundancy engaged. ");
- }
- printf("Used codec: %i\n",usedCodec);
- printf("Payload type: %i\n",payloadType);
-
- NetEQTest_init_coders(usedCodec, packet_size, bitrate, fs, useVAD, numChannels);
-
- /* write file header */
- //fprintf(out_file, "#!RTPencode%s\n", "1.0");
- fprintf(out_file, "#!rtpplay%s \n", "1.0"); // this is the string that rtpplay needs
- WebRtc_UWord32 dummy_variable = 0; // should be converted to network endian format, but does not matter when 0
- fwrite(&dummy_variable, 4, 1, out_file);
- fwrite(&dummy_variable, 4, 1, out_file);
- fwrite(&dummy_variable, 4, 1, out_file);
- fwrite(&dummy_variable, 2, 1, out_file);
- fwrite(&dummy_variable, 2, 1, out_file);
-
-#ifdef TIMESTAMP_WRAPAROUND
- timestamp = 0xFFFFFFFF - fs*10; /* should give wrap-around in 10 seconds */
-#endif
-#if defined(RANDOM_DATA) | defined(RANDOM_PAYLOAD_DATA)
- srand(RANDOM_SEED);
-#endif
-
- /* if redundancy is used, the first redundant payload is zero length */
- red_len[0] = 0;
-
- /* read first frame */
- len=fread(org_data,2,packet_size * numChannels,in_file) / numChannels;
-
- /* de-interleave if stereo */
- if ( usingStereo )
- {
- stereoDeInterleave(org_data, len * numChannels);
- }
-
- while (len==packet_size) {
-
-#ifdef INSERT_DTMF_PACKETS
- dtmfSent = false;
-
- if ( sendtime >= NTone * DTMF_PACKET_INTERVAL ) {
- if ( sendtime < NTone * DTMF_PACKET_INTERVAL + DTMF_DURATION ) {
- // tone has not ended
- if (DTMFfirst==1) {
- DTMFtimestamp = timestamp; // save this timestamp
- DTMFfirst=0;
- }
- makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo,DTMFtimestamp, ssrc);
- enc_len = makeDTMFpayload(&rtp_data[12], NTone % 12, 0, 4, (int) (sendtime - NTone * DTMF_PACKET_INTERVAL)*(fs/1000) + len);
- }
- else {
- // tone has ended
- makeRTPheader(rtp_data, NETEQ_CODEC_AVT_PT, seqNo,DTMFtimestamp, ssrc);
- enc_len = makeDTMFpayload(&rtp_data[12], NTone % 12, 1, 4, DTMF_DURATION*(fs/1000));
- NTone++;
- DTMFfirst=1;
- }
-
- /* write RTP packet to file */
- length = htons(12 + enc_len + 8);
- plen = htons(12 + enc_len);
- offset = (WebRtc_UWord32) sendtime; //(timestamp/(fs/1000));
- offset = htonl(offset);
- fwrite(&length, 2, 1, out_file);
- fwrite(&plen, 2, 1, out_file);
- fwrite(&offset, 4, 1, out_file);
- fwrite(rtp_data, 12 + enc_len, 1, out_file);
-
- dtmfSent = true;
- }
-#endif
-
-#ifdef NO_DTMF_OVERDUB
- /* If DTMF is sent, we should not send any speech packets during the same time */
- if (dtmfSent) {
- enc_len = 0;
- }
- else {
-#endif
- /* encode frame */
- enc_len=NetEQTest_encode(usedCodec, org_data, packet_size, &rtp_data[12] ,fs,&vad, useVAD, bitrate, numChannels);
- if (enc_len==-1) {
- printf("Error encoding frame\n");
- exit(0);
- }
-
- if ( usingStereo &&
- stereoMode != STEREO_MODE_FRAME &&
- vad == 1 )
- {
- // interleave the encoded payload for sample-based codecs (not for CNG)
- stereoInterleave(&rtp_data[12], enc_len, stereoMode);
- }
-#ifdef NO_DTMF_OVERDUB
- }
-#endif
-
- if (enc_len > 0 && (sendtime <= STOPSENDTIME || sendtime > RESTARTSENDTIME)) {
- if(useRed) {
- if(red_len[0] > 0) {
- memmove(&rtp_data[RTPheaderLen+red_len[0]], &rtp_data[12], enc_len);
- memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
-
- red_len[1] = enc_len;
- red_TS[1] = timestamp;
- if(vad)
- red_PT[1] = payloadType;
- else
- red_PT[1] = NETEQ_CODEC_CN_PT;
-
- makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++, ssrc);
-
-
- enc_len += red_len[0] + RTPheaderLen - 12;
- }
- else { // do not use redundancy payload for this packet, i.e., only last payload
- memmove(&rtp_data[RTPheaderLen-4], &rtp_data[12], enc_len);
- //memcpy(&rtp_data[RTPheaderLen], red_data, red_len[0]);
-
- red_len[1] = enc_len;
- red_TS[1] = timestamp;
- if(vad)
- red_PT[1] = payloadType;
- else
- red_PT[1] = NETEQ_CODEC_CN_PT;
-
- makeRedundantHeader(rtp_data, red_PT, 2, red_TS, red_len, seqNo++, ssrc);
-
-
- enc_len += red_len[0] + RTPheaderLen - 4 - 12; // 4 is length of redundancy header (not used)
- }
- }
- else {
-
- /* make RTP header */
- if (vad) // regular speech data
- makeRTPheader(rtp_data, payloadType, seqNo++,timestamp, ssrc);
- else // CNG data
- makeRTPheader(rtp_data, NETEQ_CODEC_CN_PT, seqNo++,timestamp, ssrc);
-
- }
-#ifdef MULTIPLE_SAME_TIMESTAMP
- int mult_pack=0;
- do {
-#endif //MULTIPLE_SAME_TIMESTAMP
- /* write RTP packet to file */
- length = htons(12 + enc_len + 8);
- plen = htons(12 + enc_len);
- offset = (WebRtc_UWord32) sendtime; //(timestamp/(fs/1000));
- offset = htonl(offset);
- fwrite(&length, 2, 1, out_file);
- fwrite(&plen, 2, 1, out_file);
- fwrite(&offset, 4, 1, out_file);
-#ifdef RANDOM_DATA
- for (int k=0; k<12+enc_len; k++) {
- rtp_data[k] = rand() + rand();
- }
-#endif
-#ifdef RANDOM_PAYLOAD_DATA
- for (int k=12; k<12+enc_len; k++) {
- rtp_data[k] = rand() + rand();
- }
-#endif
- fwrite(rtp_data, 12 + enc_len, 1, out_file);
-#ifdef MULTIPLE_SAME_TIMESTAMP
- } while ( (seqNo%REPEAT_PACKET_DISTANCE == 0) && (mult_pack++ < REPEAT_PACKET_COUNT) );
-#endif //MULTIPLE_SAME_TIMESTAMP
-
-#ifdef INSERT_OLD_PACKETS
- if (packet_age >= OLD_PACKET*fs) {
- if (!first_old_packet) {
- // send the old packet
- fwrite(&old_length, 2, 1, out_file);
- fwrite(&old_plen, 2, 1, out_file);
- fwrite(&offset, 4, 1, out_file);
- fwrite(old_rtp_data, 12 + old_enc_len, 1, out_file);
- }
- // store current packet as old
- old_length=length;
- old_plen=plen;
- memcpy(old_rtp_data,rtp_data,12+enc_len);
- old_enc_len=enc_len;
- first_old_packet=0;
- packet_age=0;
-
- }
- packet_age += packet_size;
-#endif
-
- if(useRed) {
- /* move data to redundancy store */
-#ifdef CODEC_ISAC
- if(usedCodec==kDecoderISAC)
- {
- assert(!usingStereo); // Cannot handle stereo yet
- red_len[0] = WebRtcIsac_GetRedPayload(ISAC_inst[0], (WebRtc_Word16*)red_data);
- }
- else
- {
-#endif
- memcpy(red_data, &rtp_data[RTPheaderLen+red_len[0]], enc_len);
- red_len[0]=red_len[1];
-#ifdef CODEC_ISAC
- }
-#endif
- red_TS[0]=red_TS[1];
- red_PT[0]=red_PT[1];
- }
-
- }
-
- /* read next frame */
- len=fread(org_data,2,packet_size * numChannels,in_file) / numChannels;
- /* de-interleave if stereo */
- if ( usingStereo )
- {
- stereoDeInterleave(org_data, len * numChannels);
- }
-
- if (payloadType==NETEQ_CODEC_G722_PT)
- timestamp+=len>>1;
- else
- timestamp+=len;
-
- sendtime += (double) len/(fs/1000);
- }
-
- NetEQTest_free_coders(usedCodec, numChannels);
- fclose(in_file);
- fclose(out_file);
- printf("Done!\n");
-
- return(0);
-}
-
-
-
-
-/****************/
-/* Subfunctions */
-/****************/
-
-void NetEQTest_GetCodec_and_PT(char * name, enum WebRtcNetEQDecoder *codec, int *PT, int frameLen, int *fs, int *bitrate, int *useRed) {
-
- *bitrate = 0; /* Default bitrate setting */
- *useRed = 0; /* Default no redundancy */
-
- if(!strcmp(name,"pcmu")){
- *codec=kDecoderPCMu;
- *PT=NETEQ_CODEC_PCMU_PT;
- *fs=8000;
- }
- else if(!strcmp(name,"pcma")){
- *codec=kDecoderPCMa;
- *PT=NETEQ_CODEC_PCMA_PT;
- *fs=8000;
- }
- else if(!strcmp(name,"pcm16b")){
- *codec=kDecoderPCM16B;
- *PT=NETEQ_CODEC_PCM16B_PT;
- *fs=8000;
- }
- else if(!strcmp(name,"pcm16b_wb")){
- *codec=kDecoderPCM16Bwb;
- *PT=NETEQ_CODEC_PCM16B_WB_PT;
- *fs=16000;
- }
- else if(!strcmp(name,"pcm16b_swb32")){
- *codec=kDecoderPCM16Bswb32kHz;
- *PT=NETEQ_CODEC_PCM16B_SWB32KHZ_PT;
- *fs=32000;
- }
- else if(!strcmp(name,"pcm16b_swb48")){
- *codec=kDecoderPCM16Bswb48kHz;
- *PT=NETEQ_CODEC_PCM16B_SWB48KHZ_PT;
- *fs=48000;
- }
- else if(!strcmp(name,"g722")){
- *codec=kDecoderG722;
- *PT=NETEQ_CODEC_G722_PT;
- *fs=16000;
- }
- else if(!strcmp(name,"g722.1_16")){
- *codec=kDecoderG722_1_16;
- *PT=NETEQ_CODEC_G722_1_16_PT;
- *fs=16000;
- }
- else if(!strcmp(name,"g722.1_24")){
- *codec=kDecoderG722_1_24;
- *PT=NETEQ_CODEC_G722_1_24_PT;
- *fs=16000;
- }
- else if(!strcmp(name,"g722.1_32")){
- *codec=kDecoderG722_1_32;
- *PT=NETEQ_CODEC_G722_1_32_PT;
- *fs=16000;
- }
- else if(!strcmp(name,"g722.1C_24")){
- *codec=kDecoderG722_1C_24;
- *PT=NETEQ_CODEC_G722_1C_24_PT;
- *fs=32000;
- }
- else if(!strcmp(name,"g722.1C_32")){
- *codec=kDecoderG722_1C_32;
- *PT=NETEQ_CODEC_G722_1C_32_PT;
- *fs=32000;
- }
- else if(!strcmp(name,"g722.1C_48")){
- *codec=kDecoderG722_1C_48;
- *PT=NETEQ_CODEC_G722_1C_48_PT;
- *fs=32000;
- }
- else if(!strcmp(name,"g726_16")){
- *fs=8000;
- *codec=kDecoderG726_16;
- *PT=NETEQ_CODEC_G726_16_PT;
- *bitrate=16;
- }
- else if(!strcmp(name,"g726_24")){
- *fs=8000;
- *codec=kDecoderG726_24;
- *PT=NETEQ_CODEC_G726_24_PT;
- *bitrate=24;
- }
- else if(!strcmp(name,"g726_32")){
- *fs=8000;
- *codec=kDecoderG726_32;
- *PT=NETEQ_CODEC_G726_32_PT;
- *bitrate=32;
- }
- else if(!strcmp(name,"g726_40")){
- *fs=8000;
- *codec=kDecoderG726_40;
- *PT=NETEQ_CODEC_G726_40_PT;
- *bitrate=40;
- }
- else if((!strcmp(name,"amr4.75k"))||(!strcmp(name,"amr5.15k"))||(!strcmp(name,"amr5.9k"))||
- (!strcmp(name,"amr6.7k"))||(!strcmp(name,"amr7.4k"))||(!strcmp(name,"amr7.95k"))||
- (!strcmp(name,"amr10.2k"))||(!strcmp(name,"amr12.2k"))) {
- *fs=8000;
- if (!strcmp(name,"amr4.75k"))
- *bitrate = 0;
- if (!strcmp(name,"amr5.15k"))
- *bitrate = 1;
- if (!strcmp(name,"amr5.9k"))
- *bitrate = 2;
- if (!strcmp(name,"amr6.7k"))
- *bitrate = 3;
- if (!strcmp(name,"amr7.4k"))
- *bitrate = 4;
- if (!strcmp(name,"amr7.95k"))
- *bitrate = 5;
- if (!strcmp(name,"amr10.2k"))
- *bitrate = 6;
- if (!strcmp(name,"amr12.2k"))
- *bitrate = 7;
- *codec=kDecoderAMR;
- *PT=NETEQ_CODEC_AMR_PT;
- }
- else if((!strcmp(name,"amrwb7k"))||(!strcmp(name,"amrwb9k"))||(!strcmp(name,"amrwb12k"))||
- (!strcmp(name,"amrwb14k"))||(!strcmp(name,"amrwb16k"))||(!strcmp(name,"amrwb18k"))||
- (!strcmp(name,"amrwb20k"))||(!strcmp(name,"amrwb23k"))||(!strcmp(name,"amrwb24k"))) {
- *fs=16000;
- if (!strcmp(name,"amrwb7k"))
- *bitrate = 7000;
- if (!strcmp(name,"amrwb9k"))
- *bitrate = 9000;
- if (!strcmp(name,"amrwb12k"))
- *bitrate = 12000;
- if (!strcmp(name,"amrwb14k"))
- *bitrate = 14000;
- if (!strcmp(name,"amrwb16k"))
- *bitrate = 16000;
- if (!strcmp(name,"amrwb18k"))
- *bitrate = 18000;
- if (!strcmp(name,"amrwb20k"))
- *bitrate = 20000;
- if (!strcmp(name,"amrwb23k"))
- *bitrate = 23000;
- if (!strcmp(name,"amrwb24k"))
- *bitrate = 24000;
- *codec=kDecoderAMRWB;
- *PT=NETEQ_CODEC_AMRWB_PT;
- }
- else if((!strcmp(name,"ilbc"))&&((frameLen%240==0)||(frameLen%160==0))){
- *fs=8000;
- *codec=kDecoderILBC;
- *PT=NETEQ_CODEC_ILBC_PT;
- }
- else if(!strcmp(name,"isac")){
- *fs=16000;
- *codec=kDecoderISAC;
- *PT=NETEQ_CODEC_ISAC_PT;
- }
- else if(!strcmp(name,"isacswb")){
- *fs=32000;
- *codec=kDecoderISACswb;
- *PT=NETEQ_CODEC_ISACSWB_PT;
- }
- else if(!strcmp(name,"g729")){
- *fs=8000;
- *codec=kDecoderG729;
- *PT=NETEQ_CODEC_G729_PT;
- }
- else if(!strcmp(name,"g729.1")){
- *fs=16000;
- *codec=kDecoderG729_1;
- *PT=NETEQ_CODEC_G729_1_PT;
- }
- else if(!strcmp(name,"gsmfr")){
- *fs=8000;
- *codec=kDecoderGSMFR;
- *PT=NETEQ_CODEC_GSMFR_PT;
- }
- else if(!strcmp(name,"speex8")){
- *fs=8000;
- *codec=kDecoderSPEEX_8;
- *PT=NETEQ_CODEC_SPEEX8_PT;
- }
- else if(!strcmp(name,"speex16")){
- *fs=16000;
- *codec=kDecoderSPEEX_16;
- *PT=NETEQ_CODEC_SPEEX16_PT;
- }
- else if(!strcmp(name,"celt32")){
- *fs=32000;
- *codec=kDecoderCELT_32;
- *PT=NETEQ_CODEC_CELT32_PT;
- }
- else if(!strcmp(name,"red_pcm")){
- *codec=kDecoderPCMa;
- *PT=NETEQ_CODEC_PCMA_PT; /* this will be the PT for the sub-headers */
- *fs=8000;
- *useRed = 1;
- } else if(!strcmp(name,"red_isac")){
- *codec=kDecoderISAC;
- *PT=NETEQ_CODEC_ISAC_PT; /* this will be the PT for the sub-headers */
- *fs=16000;
- *useRed = 1;
- } else {
- printf("Error: Not a supported codec (%s)\n", name);
- exit(0);
- }
-
-}
-
-
-
-
-int NetEQTest_init_coders(enum WebRtcNetEQDecoder coder, int enc_frameSize, int bitrate, int sampfreq , int vad, int numChannels){
-
- int ok=0;
-
- for (int k = 0; k < numChannels; k++)
- {
- ok=WebRtcVad_Create(&VAD_inst[k]);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for VAD instance\n");
- exit(0);
- }
- ok=WebRtcVad_Init(VAD_inst[k]);
- if (ok==-1) {
- printf("Error: Initialization of VAD struct failed\n");
- exit(0);
- }
-
-
-#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
- defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
- ok=WebRtcCng_CreateEnc(&CNGenc_inst[k]);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for CNG encoding instance\n");
- exit(0);
- }
- if(sampfreq <= 16000) {
- ok=WebRtcCng_InitEnc(CNGenc_inst[k],sampfreq, 200, 5);
- if (ok==-1) {
- printf("Error: Initialization of CNG struct failed. Error code %d\n",
- WebRtcCng_GetErrorCodeEnc(CNGenc_inst[k]));
- exit(0);
- }
- }
-#endif
-
- switch (coder) {
- case kDecoderReservedStart : // dummy codec
-#ifdef CODEC_PCM16B
- case kDecoderPCM16B :
-#endif
-#ifdef CODEC_PCM16B_WB
- case kDecoderPCM16Bwb :
-#endif
-#ifdef CODEC_PCM16B_32KHZ
- case kDecoderPCM16Bswb32kHz :
-#endif
-#ifdef CODEC_PCM16B_48KHZ
- case kDecoderPCM16Bswb48kHz :
-#endif
-#ifdef CODEC_G711
- case kDecoderPCMu :
- case kDecoderPCMa :
-#endif
- // do nothing
- break;
-#ifdef CODEC_G729
- case kDecoderG729:
- if (sampfreq==8000) {
- if ((enc_frameSize==80)||(enc_frameSize==160)||(enc_frameSize==240)||(enc_frameSize==320)||(enc_frameSize==400)||(enc_frameSize==480)) {
- ok=WebRtcG729_CreateEnc(&G729enc_inst[k]);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for G729 encoding instance\n");
- exit(0);
- }
- } else {
- printf("\nError: g729 only supports 10, 20, 30, 40, 50 or 60 ms!!\n\n");
- exit(0);
- }
- WebRtcG729_EncoderInit(G729enc_inst[k], vad);
- if ((vad==1)&&(enc_frameSize!=80)) {
- printf("\nError - This simulation only supports VAD for G729 at 10ms packets (not %dms)\n", (enc_frameSize>>3));
- }
- } else {
- printf("\nError - g729 is only developed for 8kHz \n");
- exit(0);
- }
- break;
-#endif
-#ifdef CODEC_G729_1
- case kDecoderG729_1:
- if (sampfreq==16000) {
- if ((enc_frameSize==320)||(enc_frameSize==640)||(enc_frameSize==960)
- ) {
- ok=WebRtcG7291_Create(&G729_1_inst[k]);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for G.729.1 codec instance\n");
- exit(0);
- }
- } else {
- printf("\nError: G.729.1 only supports 20, 40 or 60 ms!!\n\n");
- exit(0);
- }
- if (!(((bitrate >= 12000) && (bitrate <= 32000) && (bitrate%2000 == 0)) || (bitrate == 8000))) {
- /* must be 8, 12, 14, 16, 18, 20, 22, 24, 26, 28, 30, or 32 kbps */
- printf("\nError: G.729.1 bitrate must be 8000 or 12000--32000 in steps of 2000 bps\n");
- exit(0);
- }
- WebRtcG7291_EncoderInit(G729_1_inst[k], bitrate, 0 /* flag8kHz*/, 0 /*flagG729mode*/);
- } else {
- printf("\nError - G.729.1 input is always 16 kHz \n");
- exit(0);
- }
- break;
-#endif
-#ifdef CODEC_SPEEX_8
- case kDecoderSPEEX_8 :
- if (sampfreq==8000) {
- if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) {
- ok=WebRtcSpeex_CreateEnc(&SPEEX8enc_inst[k], sampfreq);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for Speex encoding instance\n");
- exit(0);
- }
- } else {
- printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
- exit(0);
- }
- if ((vad==1)&&(enc_frameSize!=160)) {
- printf("\nError - This simulation only supports VAD for Speex at 20ms packets (not %dms)\n", (enc_frameSize>>3));
- vad=0;
- }
- ok=WebRtcSpeex_EncoderInit(SPEEX8enc_inst[k], 0/*vbr*/, 3 /*complexity*/, vad);
- if (ok!=0) exit(0);
- } else {
- printf("\nError - Speex8 called with sample frequency other than 8 kHz.\n\n");
- }
- break;
-#endif
-#ifdef CODEC_SPEEX_16
- case kDecoderSPEEX_16 :
- if (sampfreq==16000) {
- if ((enc_frameSize==320)||(enc_frameSize==640)||(enc_frameSize==960)) {
- ok=WebRtcSpeex_CreateEnc(&SPEEX16enc_inst[k], sampfreq);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for Speex encoding instance\n");
- exit(0);
- }
- } else {
- printf("\nError: Speex only supports 20, 40, and 60 ms!!\n\n");
- exit(0);
- }
- if ((vad==1)&&(enc_frameSize!=320)) {
- printf("\nError - This simulation only supports VAD for Speex at 20ms packets (not %dms)\n", (enc_frameSize>>4));
- vad=0;
- }
- ok=WebRtcSpeex_EncoderInit(SPEEX16enc_inst[k], 0/*vbr*/, 3 /*complexity*/, vad);
- if (ok!=0) exit(0);
- } else {
- printf("\nError - Speex16 called with sample frequency other than 16 kHz.\n\n");
- }
- break;
-#endif
-#ifdef CODEC_CELT_32
- case kDecoderCELT_32 :
- if (sampfreq==32000) {
- if (enc_frameSize==320) {
- ok=WebRtcCelt_CreateEnc(&CELT32enc_inst[k], 1 /*mono*/);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for Celt encoding instance\n");
- exit(0);
- }
- } else {
- printf("\nError: Celt only supports 10 ms!!\n\n");
- exit(0);
- }
- ok=WebRtcCelt_EncoderInit(CELT32enc_inst[k], 1 /*mono*/, 48000 /*bitrate*/);
- if (ok!=0) exit(0);
- } else {
- printf("\nError - Celt32 called with sample frequency other than 32 kHz.\n\n");
- }
- break;
-#endif
-
-#ifdef CODEC_G722_1_16
- case kDecoderG722_1_16 :
- if (sampfreq==16000) {
- ok=WebRtcG7221_CreateEnc16(&G722_1_16enc_inst[k]);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for G.722.1 instance\n");
- exit(0);
- }
- if (enc_frameSize==320) {
- } else {
- printf("\nError: G722.1 only supports 20 ms!!\n\n");
- exit(0);
- }
- WebRtcG7221_EncoderInit16((G722_1_16_encinst_t*)G722_1_16enc_inst[k]);
- } else {
- printf("\nError - G722.1 is only developed for 16kHz \n");
- exit(0);
- }
- break;
-#endif
-#ifdef CODEC_G722_1_24
- case kDecoderG722_1_24 :
- if (sampfreq==16000) {
- ok=WebRtcG7221_CreateEnc24(&G722_1_24enc_inst[k]);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for G.722.1 instance\n");
- exit(0);
- }
- if (enc_frameSize==320) {
- } else {
- printf("\nError: G722.1 only supports 20 ms!!\n\n");
- exit(0);
- }
- WebRtcG7221_EncoderInit24((G722_1_24_encinst_t*)G722_1_24enc_inst[k]);
- } else {
- printf("\nError - G722.1 is only developed for 16kHz \n");
- exit(0);
- }
- break;
-#endif
-#ifdef CODEC_G722_1_32
- case kDecoderG722_1_32 :
- if (sampfreq==16000) {
- ok=WebRtcG7221_CreateEnc32(&G722_1_32enc_inst[k]);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for G.722.1 instance\n");
- exit(0);
- }
- if (enc_frameSize==320) {
- } else {
- printf("\nError: G722.1 only supports 20 ms!!\n\n");
- exit(0);
- }
- WebRtcG7221_EncoderInit32((G722_1_32_encinst_t*)G722_1_32enc_inst[k]);
- } else {
- printf("\nError - G722.1 is only developed for 16kHz \n");
- exit(0);
- }
- break;
-#endif
-#ifdef CODEC_G722_1C_24
- case kDecoderG722_1C_24 :
- if (sampfreq==32000) {
- ok=WebRtcG7221C_CreateEnc24(&G722_1C_24enc_inst[k]);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for G.722.1C instance\n");
- exit(0);
- }
- if (enc_frameSize==640) {
- } else {
- printf("\nError: G722.1 C only supports 20 ms!!\n\n");
- exit(0);
- }
- WebRtcG7221C_EncoderInit24((G722_1C_24_encinst_t*)G722_1C_24enc_inst[k]);
- } else {
- printf("\nError - G722.1 C is only developed for 32kHz \n");
- exit(0);
- }
- break;
-#endif
-#ifdef CODEC_G722_1C_32
- case kDecoderG722_1C_32 :
- if (sampfreq==32000) {
- ok=WebRtcG7221C_CreateEnc32(&G722_1C_32enc_inst[k]);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for G.722.1C instance\n");
- exit(0);
- }
- if (enc_frameSize==640) {
- } else {
- printf("\nError: G722.1 C only supports 20 ms!!\n\n");
- exit(0);
- }
- WebRtcG7221C_EncoderInit32((G722_1C_32_encinst_t*)G722_1C_32enc_inst[k]);
- } else {
- printf("\nError - G722.1 C is only developed for 32kHz \n");
- exit(0);
- }
- break;
-#endif
-#ifdef CODEC_G722_1C_48
- case kDecoderG722_1C_48 :
- if (sampfreq==32000) {
- ok=WebRtcG7221C_CreateEnc48(&G722_1C_48enc_inst[k]);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for G.722.1C instance\n");
- exit(0);
- }
- if (enc_frameSize==640) {
- } else {
- printf("\nError: G722.1 C only supports 20 ms!!\n\n");
- exit(0);
- }
- WebRtcG7221C_EncoderInit48((G722_1C_48_encinst_t*)G722_1C_48enc_inst[k]);
- } else {
- printf("\nError - G722.1 C is only developed for 32kHz \n");
- exit(0);
- }
- break;
-#endif
-#ifdef CODEC_G722
- case kDecoderG722 :
- if (sampfreq==16000) {
- if (enc_frameSize%2==0) {
- } else {
- printf("\nError - g722 frames must have an even number of enc_frameSize\n");
- exit(0);
- }
- WebRtcG722_CreateEncoder(&g722EncState[k]);
- WebRtcG722_EncoderInit(g722EncState[k]);
- } else {
- printf("\nError - g722 is only developed for 16kHz \n");
- exit(0);
- }
- break;
-#endif
-#ifdef CODEC_AMR
- case kDecoderAMR :
- if (sampfreq==8000) {
- ok=WebRtcAmr_CreateEnc(&AMRenc_inst[k]);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for AMR encoding instance\n");
- exit(0);
- }if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) {
- } else {
- printf("\nError - AMR must have a multiple of 160 enc_frameSize\n");
- exit(0);
- }
- WebRtcAmr_EncoderInit(AMRenc_inst[k], vad);
- WebRtcAmr_EncodeBitmode(AMRenc_inst[k], AMRBandwidthEfficient);
- AMR_bitrate = bitrate;
- } else {
- printf("\nError - AMR is only developed for 8kHz \n");
- exit(0);
- }
- break;
-#endif
-#ifdef CODEC_AMRWB
- case kDecoderAMRWB :
- if (sampfreq==16000) {
- ok=WebRtcAmrWb_CreateEnc(&AMRWBenc_inst[k]);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for AMRWB encoding instance\n");
- exit(0);
- }
- if (((enc_frameSize/320)<0)||((enc_frameSize/320)>3)||((enc_frameSize%320)!=0)) {
- printf("\nError - AMRwb must have frameSize of 20, 40 or 60ms\n");
- exit(0);
- }
- WebRtcAmrWb_EncoderInit(AMRWBenc_inst[k], vad);
- if (bitrate==7000) {
- AMRWB_bitrate = AMRWB_MODE_7k;
- } else if (bitrate==9000) {
- AMRWB_bitrate = AMRWB_MODE_9k;
- } else if (bitrate==12000) {
- AMRWB_bitrate = AMRWB_MODE_12k;
- } else if (bitrate==14000) {
- AMRWB_bitrate = AMRWB_MODE_14k;
- } else if (bitrate==16000) {
- AMRWB_bitrate = AMRWB_MODE_16k;
- } else if (bitrate==18000) {
- AMRWB_bitrate = AMRWB_MODE_18k;
- } else if (bitrate==20000) {
- AMRWB_bitrate = AMRWB_MODE_20k;
- } else if (bitrate==23000) {
- AMRWB_bitrate = AMRWB_MODE_23k;
- } else if (bitrate==24000) {
- AMRWB_bitrate = AMRWB_MODE_24k;
- }
- WebRtcAmrWb_EncodeBitmode(AMRWBenc_inst[k], AMRBandwidthEfficient);
-
- } else {
- printf("\nError - AMRwb is only developed for 16kHz \n");
- exit(0);
- }
- break;
-#endif
-#ifdef CODEC_ILBC
- case kDecoderILBC :
- if (sampfreq==8000) {
- ok=WebRtcIlbcfix_EncoderCreate(&iLBCenc_inst[k]);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for iLBC encoding instance\n");
- exit(0);
- }
- if ((enc_frameSize==160)||(enc_frameSize==240)||(enc_frameSize==320)||(enc_frameSize==480)) {
- } else {
- printf("\nError - iLBC only supports 160, 240, 320 and 480 enc_frameSize (20, 30, 40 and 60 ms)\n");
- exit(0);
- }
- if ((enc_frameSize==160)||(enc_frameSize==320)) {
- /* 20 ms version */
- WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 20);
- } else {
- /* 30 ms version */
- WebRtcIlbcfix_EncoderInit(iLBCenc_inst[k], 30);
- }
- } else {
- printf("\nError - iLBC is only developed for 8kHz \n");
- exit(0);
- }
- break;
-#endif
-#ifdef CODEC_ISAC
- case kDecoderISAC:
- if (sampfreq==16000) {
- ok=WebRtcIsac_Create(&ISAC_inst[k]);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for iSAC instance\n");
- exit(0);
- }if ((enc_frameSize==480)||(enc_frameSize==960)) {
- } else {
- printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n");
- exit(0);
- }
- WebRtcIsac_EncoderInit(ISAC_inst[k],1);
- if ((bitrate<10000)||(bitrate>32000)) {
- printf("\nError - iSAC bitrate has to be between 10000 and 32000 bps (not %i)\n", bitrate);
- exit(0);
- }
- WebRtcIsac_Control(ISAC_inst[k], bitrate, enc_frameSize>>4);
- } else {
- printf("\nError - iSAC only supports 480 or 960 enc_frameSize (30 or 60 ms)\n");
- exit(0);
- }
- break;
-#endif
-#ifdef NETEQ_ISACFIX_CODEC
- case kDecoderISAC:
- if (sampfreq==16000) {
- ok=WebRtcIsacfix_Create(&ISAC_inst[k]);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for iSAC instance\n");
- exit(0);
- }if ((enc_frameSize==480)||(enc_frameSize==960)) {
- } else {
- printf("\nError - iSAC only supports frameSize (30 and 60 ms)\n");
- exit(0);
- }
- WebRtcIsacfix_EncoderInit(ISAC_inst[k],1);
- if ((bitrate<10000)||(bitrate>32000)) {
- printf("\nError - iSAC bitrate has to be between 10000 and 32000 bps (not %i)\n", bitrate);
- exit(0);
- }
- WebRtcIsacfix_Control(ISAC_inst[k], bitrate, enc_frameSize>>4);
- } else {
- printf("\nError - iSAC only supports 480 or 960 enc_frameSize (30 or 60 ms)\n");
- exit(0);
- }
- break;
-#endif
-#ifdef CODEC_ISAC_SWB
- case kDecoderISACswb:
- if (sampfreq==32000) {
- ok=WebRtcIsac_Create(&ISACSWB_inst[k]);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for iSAC SWB instance\n");
- exit(0);
- }if (enc_frameSize==960) {
- } else {
- printf("\nError - iSAC SWB only supports frameSize 30 ms\n");
- exit(0);
- }
- ok = WebRtcIsac_SetEncSampRate(ISACSWB_inst[k], kIsacSuperWideband);
- if (ok!=0) {
- printf("Error: Couldn't set sample rate for iSAC SWB instance\n");
- exit(0);
- }
- WebRtcIsac_EncoderInit(ISACSWB_inst[k],1);
- if ((bitrate<32000)||(bitrate>56000)) {
- printf("\nError - iSAC SWB bitrate has to be between 32000 and 56000 bps (not %i)\n", bitrate);
- exit(0);
- }
- WebRtcIsac_Control(ISACSWB_inst[k], bitrate, enc_frameSize>>5);
- } else {
- printf("\nError - iSAC SWB only supports 960 enc_frameSize (30 ms)\n");
- exit(0);
- }
- break;
-#endif
-#ifdef CODEC_GSMFR
- case kDecoderGSMFR:
- if (sampfreq==8000) {
- ok=WebRtcGSMFR_CreateEnc(&GSMFRenc_inst[k]);
- if (ok!=0) {
- printf("Error: Couldn't allocate memory for GSM FR encoding instance\n");
- exit(0);
- }
- if ((enc_frameSize==160)||(enc_frameSize==320)||(enc_frameSize==480)) {
- } else {
- printf("\nError - GSM FR must have a multiple of 160 enc_frameSize\n");
- exit(0);
- }
- WebRtcGSMFR_EncoderInit(GSMFRenc_inst[k], 0);
- } else {
- printf("\nError - GSM FR is only developed for 8kHz \n");
- exit(0);
- }
- break;
-#endif
- default :
- printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
- exit(0);
- break;
- }
-
- if (ok != 0) {
- return(ok);
- }
- } // end for
-
- return(0);
-}
-
-
-
-
-int NetEQTest_free_coders(enum WebRtcNetEQDecoder coder, int numChannels) {
-
- for (int k = 0; k < numChannels; k++)
- {
- WebRtcVad_Free(VAD_inst[k]);
-#if (defined(CODEC_CNGCODEC8) || defined(CODEC_CNGCODEC16) || \
- defined(CODEC_CNGCODEC32) || defined(CODEC_CNGCODEC48))
- WebRtcCng_FreeEnc(CNGenc_inst[k]);
-#endif
-
- switch (coder)
- {
- case kDecoderReservedStart : // dummy codec
-#ifdef CODEC_PCM16B
- case kDecoderPCM16B :
-#endif
-#ifdef CODEC_PCM16B_WB
- case kDecoderPCM16Bwb :
-#endif
-#ifdef CODEC_PCM16B_32KHZ
- case kDecoderPCM16Bswb32kHz :
-#endif
-#ifdef CODEC_PCM16B_48KHZ
- case kDecoderPCM16Bswb48kHz :
-#endif
-#ifdef CODEC_G711
- case kDecoderPCMu :
- case kDecoderPCMa :
-#endif
- // do nothing
- break;
-#ifdef CODEC_G729
- case kDecoderG729:
- WebRtcG729_FreeEnc(G729enc_inst[k]);
- break;
-#endif
-#ifdef CODEC_G729_1
- case kDecoderG729_1:
- WebRtcG7291_Free(G729_1_inst[k]);
- break;
-#endif
-#ifdef CODEC_SPEEX_8
- case kDecoderSPEEX_8 :
- WebRtcSpeex_FreeEnc(SPEEX8enc_inst[k]);
- break;
-#endif
-#ifdef CODEC_SPEEX_16
- case kDecoderSPEEX_16 :
- WebRtcSpeex_FreeEnc(SPEEX16enc_inst[k]);
- break;
-#endif
-#ifdef CODEC_CELT_32
- case kDecoderCELT_32 :
- WebRtcCelt_FreeEnc(CELT32enc_inst[k]);
- break;
-#endif
-
-#ifdef CODEC_G722_1_16
- case kDecoderG722_1_16 :
- WebRtcG7221_FreeEnc16(G722_1_16enc_inst[k]);
- break;
-#endif
-#ifdef CODEC_G722_1_24
- case kDecoderG722_1_24 :
- WebRtcG7221_FreeEnc24(G722_1_24enc_inst[k]);
- break;
-#endif
-#ifdef CODEC_G722_1_32
- case kDecoderG722_1_32 :
- WebRtcG7221_FreeEnc32(G722_1_32enc_inst[k]);
- break;
-#endif
-#ifdef CODEC_G722_1C_24
- case kDecoderG722_1C_24 :
- WebRtcG7221C_FreeEnc24(G722_1C_24enc_inst[k]);
- break;
-#endif
-#ifdef CODEC_G722_1C_32
- case kDecoderG722_1C_32 :
- WebRtcG7221C_FreeEnc32(G722_1C_32enc_inst[k]);
- break;
-#endif
-#ifdef CODEC_G722_1C_48
- case kDecoderG722_1C_48 :
- WebRtcG7221C_FreeEnc48(G722_1C_48enc_inst[k]);
- break;
-#endif
-#ifdef CODEC_G722
- case kDecoderG722 :
- WebRtcG722_FreeEncoder(g722EncState[k]);
- break;
-#endif
-#ifdef CODEC_AMR
- case kDecoderAMR :
- WebRtcAmr_FreeEnc(AMRenc_inst[k]);
- break;
-#endif
-#ifdef CODEC_AMRWB
- case kDecoderAMRWB :
- WebRtcAmrWb_FreeEnc(AMRWBenc_inst[k]);
- break;
-#endif
-#ifdef CODEC_ILBC
- case kDecoderILBC :
- WebRtcIlbcfix_EncoderFree(iLBCenc_inst[k]);
- break;
-#endif
-#ifdef CODEC_ISAC
- case kDecoderISAC:
- WebRtcIsac_Free(ISAC_inst[k]);
- break;
-#endif
-#ifdef NETEQ_ISACFIX_CODEC
- case kDecoderISAC:
- WebRtcIsacfix_Free(ISAC_inst[k]);
- break;
-#endif
-#ifdef CODEC_ISAC_SWB
- case kDecoderISACswb:
- WebRtcIsac_Free(ISACSWB_inst[k]);
- break;
-#endif
-#ifdef CODEC_GSMFR
- case kDecoderGSMFR:
- WebRtcGSMFR_FreeEnc(GSMFRenc_inst[k]);
- break;
-#endif
- default :
- printf("Error: unknown codec in call to NetEQTest_init_coders.\n");
- exit(0);
- break;
- }
- }
-
- return(0);
-}
-
-
-
-
-
-
-int NetEQTest_encode(int coder, WebRtc_Word16 *indata, int frameLen, unsigned char * encoded,int sampleRate ,
- int * vad, int useVAD, int bitrate, int numChannels){
-
- short cdlen = 0;
- WebRtc_Word16 *tempdata;
- static int first_cng=1;
- WebRtc_Word16 tempLen;
-
- *vad =1;
-
- // check VAD first
- if(useVAD&&
- (coder!=kDecoderG729)&&(coder!=kDecoderAMR)&&
- (coder!=kDecoderSPEEX_8)&&(coder!=kDecoderSPEEX_16))
- {
- *vad = 0;
-
- for (int k = 0; k < numChannels; k++)
- {
- tempLen = frameLen;
- tempdata = &indata[k*frameLen];
- int localVad=0;
- /* Partition the signal and test each chunk for VAD.
- All chunks must be VAD=0 to produce a total VAD=0. */
- while (tempLen >= 10*sampleRate/1000) {
- if ((tempLen % 30*sampleRate/1000) == 0) { // tempLen is multiple of 30ms
- localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 30*sampleRate/1000);
- tempdata += 30*sampleRate/1000;
- tempLen -= 30*sampleRate/1000;
- }
- else if (tempLen >= 20*sampleRate/1000) { // tempLen >= 20ms
- localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 20*sampleRate/1000);
- tempdata += 20*sampleRate/1000;
- tempLen -= 20*sampleRate/1000;
- }
- else { // use 10ms
- localVad |= WebRtcVad_Process(VAD_inst[k] ,sampleRate, tempdata, 10*sampleRate/1000);
- tempdata += 10*sampleRate/1000;
- tempLen -= 10*sampleRate/1000;
- }
- }
-
- // aggregate all VAD decisions over all channels
- *vad |= localVad;
- }
-
- if(!*vad){
- // all channels are silent
- cdlen = 0;
- for (int k = 0; k < numChannels; k++)
- {
- WebRtcCng_Encode(CNGenc_inst[k],&indata[k*frameLen], (frameLen <= 640 ? frameLen : 640) /* max 640 */,
- encoded,&tempLen,first_cng);
- encoded += tempLen;
- cdlen += tempLen;
- }
- *vad=0;
- first_cng=0;
- return(cdlen);
- }
- }
-
-
- // loop over all channels
- int totalLen = 0;
-
- for (int k = 0; k < numChannels; k++)
- {
- /* Encode with the selected coder type */
- if (coder==kDecoderPCMu) { /*g711 u-law */
-#ifdef CODEC_G711
- cdlen = WebRtcG711_EncodeU(G711state[k], indata, frameLen, (WebRtc_Word16*) encoded);
-#endif
- }
- else if (coder==kDecoderPCMa) { /*g711 A-law */
-#ifdef CODEC_G711
- cdlen = WebRtcG711_EncodeA(G711state[k], indata, frameLen, (WebRtc_Word16*) encoded);
- }
-#endif
-#ifdef CODEC_PCM16B
- else if ((coder==kDecoderPCM16B)||(coder==kDecoderPCM16Bwb)||
- (coder==kDecoderPCM16Bswb32kHz)||(coder==kDecoderPCM16Bswb48kHz)) { /*pcm16b (8kHz, 16kHz, 32kHz or 48kHz) */
- cdlen = WebRtcPcm16b_EncodeW16(indata, frameLen, (WebRtc_Word16*) encoded);
- }
-#endif
-#ifdef CODEC_G722
- else if (coder==kDecoderG722) { /*g722 */
- cdlen=WebRtcG722_Encode(g722EncState[k], indata, frameLen, (WebRtc_Word16*)encoded);
- cdlen=frameLen>>1;
- }
-#endif
-#ifdef CODEC_G722_1_16
- else if (coder==kDecoderG722_1_16) { /* g722.1 16kbit/s mode */
- cdlen=WebRtcG7221_Encode16((G722_1_16_encinst_t*)G722_1_16enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
- }
-#endif
-#ifdef CODEC_G722_1_24
- else if (coder==kDecoderG722_1_24) { /* g722.1 24kbit/s mode*/
- cdlen=WebRtcG7221_Encode24((G722_1_24_encinst_t*)G722_1_24enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
- }
-#endif
-#ifdef CODEC_G722_1_32
- else if (coder==kDecoderG722_1_32) { /* g722.1 32kbit/s mode */
- cdlen=WebRtcG7221_Encode32((G722_1_32_encinst_t*)G722_1_32enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
- }
-#endif
-#ifdef CODEC_G722_1C_24
- else if (coder==kDecoderG722_1C_24) { /* g722.1 32 kHz 24kbit/s mode*/
- cdlen=WebRtcG7221C_Encode24((G722_1C_24_encinst_t*)G722_1C_24enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
- }
-#endif
-#ifdef CODEC_G722_1C_32
- else if (coder==kDecoderG722_1C_32) { /* g722.1 32 kHz 32kbit/s mode */
- cdlen=WebRtcG7221C_Encode32((G722_1C_32_encinst_t*)G722_1C_32enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
- }
-#endif
-#ifdef CODEC_G722_1C_48
- else if (coder==kDecoderG722_1C_48) { /* g722.1 32 kHz 48kbit/s mode */
- cdlen=WebRtcG7221C_Encode48((G722_1C_48_encinst_t*)G722_1C_48enc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
- }
-#endif
-#ifdef CODEC_G729
- else if (coder==kDecoderG729) { /*g729 */
- WebRtc_Word16 dataPos=0;
- WebRtc_Word16 len=0;
- cdlen = 0;
- for (dataPos=0;dataPos<frameLen;dataPos+=80) {
- len=WebRtcG729_Encode(G729enc_inst[k], &indata[dataPos], 80, (WebRtc_Word16*)(&encoded[cdlen]));
- cdlen += len;
- }
- }
-#endif
-#ifdef CODEC_G729_1
- else if (coder==kDecoderG729_1) { /*g729.1 */
- WebRtc_Word16 dataPos=0;
- WebRtc_Word16 len=0;
- cdlen = 0;
- for (dataPos=0;dataPos<frameLen;dataPos+=160) {
- len=WebRtcG7291_Encode(G729_1_inst[k], &indata[dataPos], (WebRtc_Word16*)(&encoded[cdlen]), bitrate, frameLen/320 /* num 20ms frames*/);
- cdlen += len;
- }
- }
-#endif
-#ifdef CODEC_AMR
- else if (coder==kDecoderAMR) { /*AMR */
- cdlen=WebRtcAmr_Encode(AMRenc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded, AMR_bitrate);
- }
-#endif
-#ifdef CODEC_AMRWB
- else if (coder==kDecoderAMRWB) { /*AMR-wb */
- cdlen=WebRtcAmrWb_Encode(AMRWBenc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded, AMRWB_bitrate);
- }
-#endif
-#ifdef CODEC_ILBC
- else if (coder==kDecoderILBC) { /*iLBC */
- cdlen=WebRtcIlbcfix_Encode(iLBCenc_inst[k], indata,frameLen,(WebRtc_Word16*)encoded);
- }
-#endif
-#if (defined(CODEC_ISAC) || defined(NETEQ_ISACFIX_CODEC)) // TODO(hlundin): remove all NETEQ_ISACFIX_CODEC
- else if (coder==kDecoderISAC) { /*iSAC */
- int noOfCalls=0;
- cdlen=0;
- while (cdlen<=0) {
-#ifdef CODEC_ISAC /* floating point */
- cdlen=WebRtcIsac_Encode(ISAC_inst[k],&indata[noOfCalls*160],(WebRtc_Word16*)encoded);
-#else /* fixed point */
- cdlen=WebRtcIsacfix_Encode(ISAC_inst[k],&indata[noOfCalls*160],(WebRtc_Word16*)encoded);
-#endif
- noOfCalls++;
- }
- }
-#endif
-#ifdef CODEC_ISAC_SWB
- else if (coder==kDecoderISACswb) { /* iSAC SWB */
- int noOfCalls=0;
- cdlen=0;
- while (cdlen<=0) {
- cdlen=WebRtcIsac_Encode(ISACSWB_inst[k],&indata[noOfCalls*320],(WebRtc_Word16*)encoded);
- noOfCalls++;
- }
- }
-#endif
-#ifdef CODEC_GSMFR
- else if (coder==kDecoderGSMFR) { /* GSM FR */
- cdlen=WebRtcGSMFR_Encode(GSMFRenc_inst[k], indata, frameLen, (WebRtc_Word16*)encoded);
- }
-#endif
-#ifdef CODEC_SPEEX_8
- else if (coder==kDecoderSPEEX_8) { /* Speex */
- int encodedLen = 0;
- int retVal = 1;
- while (retVal == 1 && encodedLen < frameLen) {
- retVal = WebRtcSpeex_Encode(SPEEX8enc_inst[k], &indata[encodedLen], 15000);
- encodedLen += 20*8; /* 20 ms */
- }
- if( (retVal == 0 && encodedLen != frameLen) || retVal < 0) {
- printf("Error encoding speex frame!\n");
- exit(0);
- }
- cdlen=WebRtcSpeex_GetBitstream(SPEEX8enc_inst[k], (WebRtc_Word16*)encoded);
- }
-#endif
-#ifdef CODEC_SPEEX_16
- else if (coder==kDecoderSPEEX_16) { /* Speex */
- int encodedLen = 0;
- int retVal = 1;
- while (retVal == 1 && encodedLen < frameLen) {
- retVal = WebRtcSpeex_Encode(SPEEX16enc_inst[k], &indata[encodedLen], 15000);
- encodedLen += 20*16; /* 20 ms */
- }
- if( (retVal == 0 && encodedLen != frameLen) || retVal < 0) {
- printf("Error encoding speex frame!\n");
- exit(0);
- }
- cdlen=WebRtcSpeex_GetBitstream(SPEEX16enc_inst[k], (WebRtc_Word16*)encoded);
- }
-#endif
-#ifdef CODEC_CELT_32
- else if (coder==kDecoderCELT_32) { /* Celt */
- int encodedLen = 0;
- cdlen = 0;
- while (cdlen <= 0) {
- cdlen = WebRtcCelt_Encode(CELT32enc_inst[k], &indata[encodedLen], encoded);
- encodedLen += 10*32; /* 10 ms */
- }
- if( (encodedLen != frameLen) || cdlen < 0) {
- printf("Error encoding Celt frame!\n");
- exit(0);
- }
- }
-#endif
-
- indata += frameLen;
- encoded += cdlen;
- totalLen += cdlen;
-
- } // end for
-
- first_cng=1;
- return(totalLen);
-}
-
-
-
-void makeRTPheader(unsigned char* rtp_data, int payloadType, int seqNo, WebRtc_UWord32 timestamp, WebRtc_UWord32 ssrc){
-
- rtp_data[0]=(unsigned char)0x80;
- rtp_data[1]=(unsigned char)(payloadType & 0xFF);
- rtp_data[2]=(unsigned char)((seqNo>>8)&0xFF);
- rtp_data[3]=(unsigned char)((seqNo)&0xFF);
- rtp_data[4]=(unsigned char)((timestamp>>24)&0xFF);
- rtp_data[5]=(unsigned char)((timestamp>>16)&0xFF);
-
- rtp_data[6]=(unsigned char)((timestamp>>8)&0xFF);
- rtp_data[7]=(unsigned char)(timestamp & 0xFF);
-
- rtp_data[8]=(unsigned char)((ssrc>>24)&0xFF);
- rtp_data[9]=(unsigned char)((ssrc>>16)&0xFF);
-
- rtp_data[10]=(unsigned char)((ssrc>>8)&0xFF);
- rtp_data[11]=(unsigned char)(ssrc & 0xFF);
-}
-
-
-int makeRedundantHeader(unsigned char* rtp_data, int *payloadType, int numPayloads, WebRtc_UWord32 *timestamp, WebRtc_UWord16 *blockLen,
- int seqNo, WebRtc_UWord32 ssrc)
-{
-
- int i;
- unsigned char *rtpPointer;
- WebRtc_UWord16 offset;
-
- /* first create "standard" RTP header */
- makeRTPheader(rtp_data, NETEQ_CODEC_RED_PT, seqNo, timestamp[numPayloads-1], ssrc);
-
- rtpPointer = &rtp_data[12];
-
- /* add one sub-header for each redundant payload (not the primary) */
- for(i=0; i<numPayloads-1; i++) { /* |0 1 2 3 4 5 6 7| */
- if(blockLen[i] > 0) {
- offset = (WebRtc_UWord16) (timestamp[numPayloads-1] - timestamp[i]);
-
- rtpPointer[0] = (unsigned char) ( 0x80 | (0x7F & payloadType[i]) ); /* |F| block PT | */
- rtpPointer[1] = (unsigned char) ((offset >> 6) & 0xFF); /* | timestamp- | */
- rtpPointer[2] = (unsigned char) ( ((offset & 0x3F)<<2) |
- ( (blockLen[i]>>8) & 0x03 ) ); /* | -offset |bl-| */
- rtpPointer[3] = (unsigned char) ( blockLen[i] & 0xFF ); /* | -ock length | */
-
- rtpPointer += 4;
- }
- }
-
- /* last sub-header */
- rtpPointer[0]= (unsigned char) (0x00 | (0x7F&payloadType[numPayloads-1]));/* |F| block PT | */
- rtpPointer += 1;
-
- return(rtpPointer - rtp_data); /* length of header in bytes */
-}
-
-
-
-int makeDTMFpayload(unsigned char* payload_data, int Event, int End, int Volume, int Duration) {
- unsigned char E,R,V;
- R=0;
- V=(unsigned char)Volume;
- if (End==0) {
- E = 0x00;
- } else {
- E = 0x80;
- }
- payload_data[0]=(unsigned char)Event;
- payload_data[1]=(unsigned char)(E|R|V);
- //Duration equals 8 times time_ms, default is 8000 Hz.
- payload_data[2]=(unsigned char)((Duration>>8)&0xFF);
- payload_data[3]=(unsigned char)(Duration&0xFF);
- return(4);
-}
-
-void stereoDeInterleave(WebRtc_Word16* audioSamples, int numSamples)
-{
-
- WebRtc_Word16 *tempVec;
- WebRtc_Word16 *readPtr, *writeL, *writeR;
-
- if (numSamples <= 0)
- return;
-
- tempVec = (WebRtc_Word16 *) malloc(sizeof(WebRtc_Word16) * numSamples);
- if (tempVec == NULL) {
- printf("Error allocating memory\n");
- exit(0);
- }
-
- memcpy(tempVec, audioSamples, numSamples*sizeof(WebRtc_Word16));
-
- writeL = audioSamples;
- writeR = &audioSamples[numSamples/2];
- readPtr = tempVec;
-
- for (int k = 0; k < numSamples; k += 2)
- {
- *writeL = *readPtr;
- readPtr++;
- *writeR = *readPtr;
- readPtr++;
- writeL++;
- writeR++;
- }
-
- free(tempVec);
-
-}
-
-
-void stereoInterleave(unsigned char* data, int dataLen, int stride)
-{
-
- unsigned char *ptrL, *ptrR;
- unsigned char temp[10];
-
- if (stride > 10)
- {
- exit(0);
- }
-
- if (dataLen%1 != 0)
- {
- // must be even number of samples
- printf("Error: cannot interleave odd sample number\n");
- exit(0);
- }
-
- ptrL = data + stride;
- ptrR = &data[dataLen/2];
-
- while (ptrL < ptrR) {
- // copy from right pointer to temp
- memcpy(temp, ptrR, stride);
-
- // shift data between pointers
- memmove(ptrL + stride, ptrL, ptrR - ptrL);
-
- // copy from temp to left pointer
- memcpy(ptrL, temp, stride);
-
- // advance pointers
- ptrL += stride*2;
- ptrR += stride;
- }
-
-}
diff --git a/trunk/src/modules/audio_coding/neteq/test/RTPjitter.cc b/trunk/src/modules/audio_coding/neteq/test/RTPjitter.cc
deleted file mode 100644
index 795111f..0000000
--- a/trunk/src/modules/audio_coding/neteq/test/RTPjitter.cc
+++ /dev/null
@@ -1,196 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//TODO(hlundin): Reformat file to meet style guide.
-
-/* header includes */
-#include "typedefs.h"
-#include <stdio.h>
-#include <stdlib.h>
-#ifdef WIN32
-#include <winsock2.h>
-#include <io.h>
-#endif
-#ifdef WEBRTC_LINUX
-#include <netinet/in.h>
-#endif
-#include <search.h>
-#include <float.h>
-
-#include "gtest/gtest.h"
-
-/*********************/
-/* Misc. definitions */
-/*********************/
-
-#define FIRSTLINELEN 40
-#define CHECK_ZERO(a) {int errCode = a; if((errCode)!=0){fprintf(stderr,"\n %s \n line: %d \n error at %s\n Error Code = %d\n",__FILE__,__LINE__,#a, WebRtcNetEQ_GetErrorCode(inst)); exit(0);}}
-#define CHECK_NOT_NULL(a) if((a)==NULL){fprintf(stderr,"\n %s \n line: %d \nerror at %s\n",__FILE__,__LINE__,#a );return(-1);}
-
-struct arr_time {
- float time;
- WebRtc_UWord32 ix;
-};
-
-int filelen(FILE *fid)
-{
- fpos_t cur_pos;
- int len;
-
- if (!fid || fgetpos(fid, &cur_pos)) {
- return(-1);
- }
-
- fseek(fid, 0, SEEK_END);
- len = ftell(fid);
-
- fsetpos(fid, &cur_pos);
-
- return (len);
-}
-
-int compare_arr_time(const void *x, const void *y);
-
-int main(int argc, char* argv[])
-{
- unsigned int dat_len, rtp_len, Npack, k;
- arr_time *time_vec;
- char firstline[FIRSTLINELEN];
- unsigned char *rtp_vec = NULL, **packet_ptr, *temp_packet;
- const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
- WebRtc_UWord16 len;
- WebRtc_UWord32 *offset;
-
-/* check number of parameters */
- if (argc != 4) {
- /* print help text and exit */
- printf("Apply jitter on RTP stream.\n");
- printf("The program reads an RTP stream and packet timing from two files.\n");
- printf("The RTP stream is modified to have the same jitter as described in the timing files.\n");
- printf("The format of the RTP stream file should be the same as for rtpplay,\n");
- printf("and can be obtained e.g., from Ethereal by using\n");
- printf("Statistics -> RTP -> Show All Streams -> [select a stream] -> Save As\n\n");
- printf("Usage:\n\n");
- printf("%s RTP_infile dat_file RTP_outfile\n", argv[0]);
- printf("where:\n");
-
- printf("RTP_infile : RTP stream input file\n\n");
-
- printf("dat_file : file with packet arrival times in ms\n\n");
-
- printf("RTP_outfile : RTP stream output file\n\n");
-
- return(0);
- }
-
- FILE* in_file=fopen(argv[1],"rb");
- CHECK_NOT_NULL(in_file);
- printf("Input file: %s\n",argv[1]);
- FILE* dat_file=fopen(argv[2],"rb");
- CHECK_NOT_NULL(dat_file);
- printf("Dat-file: %s\n",argv[2]);
- FILE* out_file=fopen(argv[3],"wb");
- CHECK_NOT_NULL(out_file);
- printf("Output file: %s\n\n",argv[3]);
-
- time_vec = (arr_time *) malloc(sizeof(arr_time)*(filelen(dat_file)/sizeof(float)) + 1000); // add 1000 bytes to avoid (rare) strange error
- if (time_vec==NULL) {
- fprintf(stderr, "Error: could not allocate memory for reading dat file\n");
- goto closing;
- }
-
- dat_len=0;
- while(fread(&(time_vec[dat_len].time),sizeof(float),1,dat_file)>0) {
- time_vec[dat_len].ix=dat_len;
- dat_len++;
- }
-
- qsort(time_vec,dat_len,sizeof(arr_time),compare_arr_time);
-
-
- rtp_vec = (unsigned char *) malloc(sizeof(unsigned char)*filelen(in_file));
- if (rtp_vec==NULL) {
- fprintf(stderr,"Error: could not allocate memory for reading rtp file\n");
- goto closing;
- }
-
- // read file header and write directly to output file
- EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, in_file) != NULL);
- EXPECT_GT(fputs(firstline, out_file), 0);
- EXPECT_EQ(kRtpDumpHeaderSize, fread(firstline, 1, kRtpDumpHeaderSize,
- in_file));
- EXPECT_EQ(kRtpDumpHeaderSize, fwrite(firstline, 1, kRtpDumpHeaderSize,
- out_file));
-
- // read all RTP packets into vector
- rtp_len=0;
- Npack=0;
- len=(WebRtc_UWord16) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file); // read length of first packet
- while(len==2) {
- len = ntohs(*((WebRtc_UWord16 *)(rtp_vec + rtp_len)));
- rtp_len += 2;
- if(fread(&rtp_vec[rtp_len], sizeof(unsigned char), len-2, in_file)!=(unsigned) (len-2)) {
- fprintf(stderr,"Error: currupt packet length\n");
- goto closing;
- }
- rtp_len += len-2;
- Npack++;
- len=(WebRtc_UWord16) fread(&rtp_vec[rtp_len], sizeof(unsigned char), 2, in_file); // read length of next packet
- }
-
- packet_ptr = (unsigned char **) malloc(Npack*sizeof(unsigned char*));
-
- packet_ptr[0]=rtp_vec;
- k=1;
- while(k<Npack) {
- len = ntohs(*((WebRtc_UWord16 *) packet_ptr[k-1]));
- packet_ptr[k]=packet_ptr[k-1]+len;
- k++;
- }
-
- for(k=0; k<dat_len && k<Npack; k++) {
- if(time_vec[k].time < FLT_MAX && time_vec[k].ix < Npack){
- temp_packet = packet_ptr[time_vec[k].ix];
- offset = (WebRtc_UWord32 *) (temp_packet+4);
- if ( time_vec[k].time >= 0 ) {
- *offset = htonl((WebRtc_UWord32) time_vec[k].time);
- }
- else {
- *offset = htonl((WebRtc_UWord32) 0);
- fprintf(stderr, "Warning: negative receive time in dat file transformed to 0.\n");
- }
-
- // write packet to file
- fwrite(temp_packet, sizeof(unsigned char), ntohs(*((WebRtc_UWord16*) temp_packet)), out_file);
- }
- }
-
-
-closing:
- free(time_vec);
- free(rtp_vec);
- fclose(in_file);
- fclose(dat_file);
- fclose(out_file);
-
- return(0);
-}
-
-
-
-int compare_arr_time(const void *xp, const void *yp) {
-
- if(((arr_time *)xp)->time == ((arr_time *)yp)->time)
- return(0);
- else if(((arr_time *)xp)->time > ((arr_time *)yp)->time)
- return(1);
-
- return(-1);
-}
diff --git a/trunk/src/modules/audio_coding/neteq/test/RTPtimeshift.cc b/trunk/src/modules/audio_coding/neteq/test/RTPtimeshift.cc
deleted file mode 100644
index dc7ff9f..0000000
--- a/trunk/src/modules/audio_coding/neteq/test/RTPtimeshift.cc
+++ /dev/null
@@ -1,97 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <algorithm>
-#include <stdio.h>
-#include <vector>
-
-#include "NETEQTEST_RTPpacket.h"
-#include "gtest/gtest.h"
-
-/*********************/
-/* Misc. definitions */
-/*********************/
-
-#define FIRSTLINELEN 40
-
-
-int main(int argc, char* argv[])
-{
- if(argc < 4 || argc > 6)
- {
- printf("Usage: RTPtimeshift in.rtp out.rtp newStartTS [newStartSN [newStartArrTime]]\n");
- exit(1);
- }
-
- FILE *inFile=fopen(argv[1],"rb");
- if (!inFile)
- {
- printf("Cannot open input file %s\n", argv[1]);
- return(-1);
- }
- printf("Input RTP file: %s\n",argv[1]);
-
- FILE *outFile=fopen(argv[2],"wb");
- if (!outFile)
- {
- printf("Cannot open output file %s\n", argv[2]);
- return(-1);
- }
- printf("Output RTP file: %s\n\n",argv[2]);
-
- // read file header and write directly to output file
- const unsigned int kRtpDumpHeaderSize = 4 + 4 + 4 + 2 + 2;
- char firstline[FIRSTLINELEN];
- EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, inFile) != NULL);
- EXPECT_GT(fputs(firstline, outFile), 0);
- EXPECT_EQ(kRtpDumpHeaderSize,
- fread(firstline, 1, kRtpDumpHeaderSize, inFile));
- EXPECT_EQ(kRtpDumpHeaderSize,
- fwrite(firstline, 1, kRtpDumpHeaderSize, outFile));
- NETEQTEST_RTPpacket packet;
- int packLen = packet.readFromFile(inFile);
- if (packLen < 0)
- {
- exit(1);
- }
-
- // get new start TS and start SeqNo from arguments
- WebRtc_UWord32 TSdiff = atoi(argv[3]) - packet.timeStamp();
- WebRtc_UWord16 SNdiff = 0;
- WebRtc_UWord32 ATdiff = 0;
- if (argc > 4)
- {
- if (argv[4] >= 0)
- SNdiff = atoi(argv[4]) - packet.sequenceNumber();
- if (argc > 5)
- {
- if (argv[5] >= 0)
- ATdiff = atoi(argv[5]) - packet.time();
- }
- }
-
- while (packLen >= 0)
- {
-
- packet.setTimeStamp(packet.timeStamp() + TSdiff);
- packet.setSequenceNumber(packet.sequenceNumber() + SNdiff);
- packet.setTime(packet.time() + ATdiff);
-
- packet.writeToFile(outFile);
-
- packLen = packet.readFromFile(inFile);
-
- }
-
- fclose(inFile);
- fclose(outFile);
-
- return 0;
-}
diff --git a/trunk/src/modules/audio_coding/neteq/test/delay_tool/parse_delay_file.m b/trunk/src/modules/audio_coding/neteq/test/delay_tool/parse_delay_file.m
deleted file mode 100644
index 77b394f..0000000
--- a/trunk/src/modules/audio_coding/neteq/test/delay_tool/parse_delay_file.m
+++ /dev/null
@@ -1,191 +0,0 @@
-function outStruct = parse_delay_file(file)
-
-fid = fopen(file, 'rb');
-if fid == -1
- error('Cannot open file %s', file);
-end
-
-textline = fgetl(fid);
-if ~strncmp(textline, '#!NetEQ_Delay_Logging', 21)
- error('Wrong file format');
-end
-
-ver = sscanf(textline, '#!NetEQ_Delay_Logging%d.%d');
-if ~all(ver == [2; 0])
- error('Wrong version of delay logging function')
-end
-
-
-start_pos = ftell(fid);
-fseek(fid, -12, 'eof');
-textline = fgetl(fid);
-if ~strncmp(textline, 'End of file', 21)
- error('File ending is not correct. Seems like the simulation ended abnormally.');
-end
-
-fseek(fid,-12-4, 'eof');
-Npackets = fread(fid, 1, 'int32');
-fseek(fid, start_pos, 'bof');
-
-rtpts = zeros(Npackets, 1);
-seqno = zeros(Npackets, 1);
-pt = zeros(Npackets, 1);
-plen = zeros(Npackets, 1);
-recin_t = nan*ones(Npackets, 1);
-decode_t = nan*ones(Npackets, 1);
-playout_delay = zeros(Npackets, 1);
-optbuf = zeros(Npackets, 1);
-
-fs_ix = 1;
-clock = 0;
-ts_ix = 1;
-ended = 0;
-late_packets = 0;
-fs_now = 8000;
-last_decode_k = 0;
-tot_expand = 0;
-tot_accelerate = 0;
-tot_preemptive = 0;
-
-while not(ended)
- signal = fread(fid, 1, '*int32');
-
- switch signal
- case 3 % NETEQ_DELAY_LOGGING_SIGNAL_CLOCK
- clock = fread(fid, 1, '*float32');
-
- % keep on reading batches of M until the signal is no longer "3"
- % read int32 + float32 in one go
- % this is to save execution time
- temp = [3; 0];
- M = 120;
- while all(temp(1,:) == 3)
- fp = ftell(fid);
- temp = fread(fid, [2 M], '*int32');
- end
-
- % back up to last clock event
- fseek(fid, fp - ftell(fid) + ...
- (find(temp(1,:) ~= 3, 1 ) - 2) * 2 * 4 + 4, 'cof');
- % read the last clock value
- clock = fread(fid, 1, '*float32');
-
- case 1 % NETEQ_DELAY_LOGGING_SIGNAL_RECIN
- temp_ts = fread(fid, 1, 'uint32');
-
- if late_packets > 0
- temp_ix = ts_ix - 1;
- while (temp_ix >= 1) && (rtpts(temp_ix) ~= temp_ts)
- % TODO(hlundin): use matlab vector search instead?
- temp_ix = temp_ix - 1;
- end
-
- if temp_ix >= 1
- % the ts was found in the vector
- late_packets = late_packets - 1;
- else
- temp_ix = ts_ix;
- ts_ix = ts_ix + 1;
- end
- else
- temp_ix = ts_ix;
- ts_ix = ts_ix + 1;
- end
-
- rtpts(temp_ix) = temp_ts;
- seqno(temp_ix) = fread(fid, 1, 'uint16');
- pt(temp_ix) = fread(fid, 1, 'int32');
- plen(temp_ix) = fread(fid, 1, 'int16');
- recin_t(temp_ix) = clock;
-
- case 2 % NETEQ_DELAY_LOGGING_SIGNAL_FLUSH
- % do nothing
-
- case 4 % NETEQ_DELAY_LOGGING_SIGNAL_EOF
- ended = 1;
-
- case 5 % NETEQ_DELAY_LOGGING_SIGNAL_DECODE
- last_decode_ts = fread(fid, 1, 'uint32');
- temp_delay = fread(fid, 1, 'uint16');
-
- k = find(rtpts(1:(ts_ix - 1))==last_decode_ts,1,'last');
- if ~isempty(k)
- decode_t(k) = clock;
- playout_delay(k) = temp_delay + ...
- 5 * fs_now / 8000; % add overlap length
- last_decode_k = k;
- end
-
- case 6 % NETEQ_DELAY_LOGGING_SIGNAL_CHANGE_FS
- fsvec(fs_ix) = fread(fid, 1, 'uint16');
- fschange_ts(fs_ix) = last_decode_ts;
- fs_now = fsvec(fs_ix);
- fs_ix = fs_ix + 1;
-
- case 7 % NETEQ_DELAY_LOGGING_SIGNAL_MERGE_INFO
- playout_delay(last_decode_k) = playout_delay(last_decode_k) ...
- + fread(fid, 1, 'int32');
-
- case 8 % NETEQ_DELAY_LOGGING_SIGNAL_EXPAND_INFO
- temp = fread(fid, 1, 'int32');
- if last_decode_k ~= 0
- tot_expand = tot_expand + temp / (fs_now / 1000);
- end
-
- case 9 % NETEQ_DELAY_LOGGING_SIGNAL_ACCELERATE_INFO
- temp = fread(fid, 1, 'int32');
- if last_decode_k ~= 0
- tot_accelerate = tot_accelerate + temp / (fs_now / 1000);
- end
-
- case 10 % NETEQ_DELAY_LOGGING_SIGNAL_PREEMPTIVE_INFO
- temp = fread(fid, 1, 'int32');
- if last_decode_k ~= 0
- tot_preemptive = tot_preemptive + temp / (fs_now / 1000);
- end
-
- case 11 % NETEQ_DELAY_LOGGING_SIGNAL_OPTBUF
- optbuf(last_decode_k) = fread(fid, 1, 'int32');
-
- case 12 % NETEQ_DELAY_LOGGING_SIGNAL_DECODE_ONE_DESC
- last_decode_ts = fread(fid, 1, 'uint32');
- k = ts_ix - 1;
-
- while (k >= 1) && (rtpts(k) ~= last_decode_ts)
- % TODO(hlundin): use matlab vector search instead?
- k = k - 1;
- end
-
- if k < 1
- % packet not received yet
- k = ts_ix;
- rtpts(ts_ix) = last_decode_ts;
- late_packets = late_packets + 1;
- end
-
- decode_t(k) = clock;
- playout_delay(k) = fread(fid, 1, 'uint16') + ...
- 5 * fs_now / 8000; % add overlap length
- last_decode_k = k;
-
- end
-
-end
-
-
-fclose(fid);
-
-outStruct = struct(...
- 'ts', rtpts, ...
- 'sn', seqno, ...
- 'pt', pt,...
- 'plen', plen,...
- 'arrival', recin_t,...
- 'decode', decode_t,...
- 'fs', fsvec(:),...
- 'fschange_ts', fschange_ts(:),...
- 'playout_delay', playout_delay,...
- 'tot_expand', tot_expand,...
- 'tot_accelerate', tot_accelerate,...
- 'tot_preemptive', tot_preemptive,...
- 'optbuf', optbuf);
diff --git a/trunk/src/modules/audio_coding/neteq/test/delay_tool/plot_neteq_delay.m b/trunk/src/modules/audio_coding/neteq/test/delay_tool/plot_neteq_delay.m
deleted file mode 100644
index bc1c85a..0000000
--- a/trunk/src/modules/audio_coding/neteq/test/delay_tool/plot_neteq_delay.m
+++ /dev/null
@@ -1,187 +0,0 @@
-function [delay_struct, delayvalues] = plot_neteq_delay(delayfile, varargin)
-
-% InfoStruct = plot_neteq_delay(delayfile)
-% InfoStruct = plot_neteq_delay(delayfile, 'skipdelay', skip_seconds)
-%
-% Henrik Lundin, 2006-11-17
-% Henrik Lundin, 2011-05-17
-%
-
-try
- s = parse_delay_file(delayfile);
-catch
- error(lasterr);
-end
-
-delayskip=0;
-noplot=0;
-arg_ptr=1;
-delaypoints=[];
-
-s.sn=unwrap_seqno(s.sn);
-
-while arg_ptr+1 <= nargin
- switch lower(varargin{arg_ptr})
- case {'skipdelay', 'delayskip'}
- % skip a number of seconds in the beginning when calculating delays
- delayskip = varargin{arg_ptr+1};
- arg_ptr = arg_ptr + 2;
- case 'noplot'
- noplot=1;
- arg_ptr = arg_ptr + 1;
- case {'get_delay', 'getdelay'}
- % return a vector of delay values for the points in the given vector
- delaypoints = varargin{arg_ptr+1};
- arg_ptr = arg_ptr + 2;
- otherwise
- warning('Unknown switch %s\n', varargin{arg_ptr});
- arg_ptr = arg_ptr + 1;
- end
-end
-
-% find lost frames that were covered by one-descriptor decoding
-one_desc_ix=find(isnan(s.arrival));
-for k=1:length(one_desc_ix)
- ix=find(s.ts==max(s.ts(s.ts(one_desc_ix(k))>s.ts)));
- s.sn(one_desc_ix(k))=s.sn(ix)+1;
- s.pt(one_desc_ix(k))=s.pt(ix);
- s.arrival(one_desc_ix(k))=s.arrival(ix)+s.decode(one_desc_ix(k))-s.decode(ix);
-end
-
-% remove duplicate received frames that were never decoded (RED codec)
-if length(unique(s.ts(isfinite(s.ts)))) < length(s.ts(isfinite(s.ts)))
- ix=find(isfinite(s.decode));
- s.sn=s.sn(ix);
- s.ts=s.ts(ix);
- s.arrival=s.arrival(ix);
- s.playout_delay=s.playout_delay(ix);
- s.pt=s.pt(ix);
- s.optbuf=s.optbuf(ix);
- plen=plen(ix);
- s.decode=s.decode(ix);
-end
-
-% find non-unique sequence numbers
-[~,un_ix]=unique(s.sn);
-nonun_ix=setdiff(1:length(s.sn),un_ix);
-if ~isempty(nonun_ix)
- warning('RTP sequence numbers are in error');
-end
-
-% sort vectors
-[s.sn,sort_ix]=sort(s.sn);
-s.ts=s.ts(sort_ix);
-s.arrival=s.arrival(sort_ix);
-s.decode=s.decode(sort_ix);
-s.playout_delay=s.playout_delay(sort_ix);
-s.pt=s.pt(sort_ix);
-
-send_t=s.ts-s.ts(1);
-if length(s.fs)<1
- warning('No info about sample rate found in file. Using default 8000.');
- s.fs(1)=8000;
- s.fschange_ts(1)=min(s.ts);
-elseif s.fschange_ts(1)>min(s.ts)
- s.fschange_ts(1)=min(s.ts);
-end
-
-end_ix=length(send_t);
-for k=length(s.fs):-1:1
- start_ix=find(s.ts==s.fschange_ts(k));
- send_t(start_ix:end_ix)=send_t(start_ix:end_ix)/s.fs(k)*1000;
- s.playout_delay(start_ix:end_ix)=s.playout_delay(start_ix:end_ix)/s.fs(k)*1000;
- s.optbuf(start_ix:end_ix)=s.optbuf(start_ix:end_ix)/s.fs(k)*1000;
- end_ix=start_ix-1;
-end
-
-tot_time=max(send_t)-min(send_t);
-
-seq_ix=s.sn-min(s.sn)+1;
-send_t=send_t+max(min(s.arrival-send_t),0);
-
-plot_send_t=nan*ones(max(seq_ix),1);
-plot_send_t(seq_ix)=send_t;
-plot_nw_delay=nan*ones(max(seq_ix),1);
-plot_nw_delay(seq_ix)=s.arrival-send_t;
-
-cng_ix=find(s.pt~=13); % find those packets that are not CNG/SID
-
-if noplot==0
- h=plot(plot_send_t/1000,plot_nw_delay);
- set(h,'color',0.75*[1 1 1]);
- hold on
- if any(s.optbuf~=0)
- peak_ix=find(s.optbuf(cng_ix)<0); % peak mode is labeled with negative values
- no_peak_ix=find(s.optbuf(cng_ix)>0); %setdiff(1:length(cng_ix),peak_ix);
- h1=plot(send_t(cng_ix(peak_ix))/1000,...
- s.arrival(cng_ix(peak_ix))+abs(s.optbuf(cng_ix(peak_ix)))-send_t(cng_ix(peak_ix)),...
- 'r.');
- h2=plot(send_t(cng_ix(no_peak_ix))/1000,...
- s.arrival(cng_ix(no_peak_ix))+abs(s.optbuf(cng_ix(no_peak_ix)))-send_t(cng_ix(no_peak_ix)),...
- 'g.');
- set([h1, h2],'markersize',1)
- end
- %h=plot(send_t(seq_ix)/1000,s.decode+s.playout_delay-send_t(seq_ix));
- h=plot(send_t(cng_ix)/1000,s.decode(cng_ix)+s.playout_delay(cng_ix)-send_t(cng_ix));
- set(h,'linew',1.5);
- hold off
- ax1=axis;
- axis tight
- ax2=axis;
- axis([ax2(1:3) ax1(4)])
-end
-
-
-% calculate delays and other parameters
-
-delayskip_ix = find(send_t-send_t(1)>=delayskip*1000, 1 );
-
-use_ix = intersect(cng_ix,... % use those that are not CNG/SID frames...
- intersect(find(isfinite(s.decode)),... % ... that did arrive ...
- (delayskip_ix:length(s.decode))')); % ... and are sent after delayskip seconds
-
-mean_delay = mean(s.decode(use_ix)+s.playout_delay(use_ix)-send_t(use_ix));
-neteq_delay = mean(s.decode(use_ix)+s.playout_delay(use_ix)-s.arrival(use_ix));
-
-Npack=max(s.sn(delayskip_ix:end))-min(s.sn(delayskip_ix:end))+1;
-nw_lossrate=(Npack-length(s.sn(delayskip_ix:end)))/Npack;
-neteq_lossrate=(length(s.sn(delayskip_ix:end))-length(use_ix))/Npack;
-
-delay_struct=struct('mean_delay',mean_delay,'neteq_delay',neteq_delay,...
- 'nw_lossrate',nw_lossrate,'neteq_lossrate',neteq_lossrate,...
- 'tot_expand',round(s.tot_expand),'tot_accelerate',round(s.tot_accelerate),...
- 'tot_preemptive',round(s.tot_preemptive),'tot_time',tot_time,...
- 'filename',delayfile,'units','ms','fs',unique(s.fs));
-
-if not(isempty(delaypoints))
- delayvalues=interp1(send_t(cng_ix),...
- s.decode(cng_ix)+s.playout_delay(cng_ix)-send_t(cng_ix),...
- delaypoints,'nearest',NaN);
-else
- delayvalues=[];
-end
-
-
-
-% SUBFUNCTIONS %
-
-function y=unwrap_seqno(x)
-
-jumps=find(abs((diff(x)-1))>65000);
-
-while ~isempty(jumps)
- n=jumps(1);
- if x(n+1)-x(n) < 0
- % negative jump
- x(n+1:end)=x(n+1:end)+65536;
- else
- % positive jump
- x(n+1:end)=x(n+1:end)-65536;
- end
-
- jumps=find(abs((diff(x(n+1:end))-1))>65000);
-end
-
-y=x;
-
-return;
diff --git a/trunk/src/modules/audio_coding/neteq/test/ptypes.txt b/trunk/src/modules/audio_coding/neteq/test/ptypes.txt
deleted file mode 100644
index c3d4e25..0000000
--- a/trunk/src/modules/audio_coding/neteq/test/ptypes.txt
+++ /dev/null
@@ -1,20 +0,0 @@
-pcmu 0
-pcma 8
-cn 13
-//ipcmwb 97
-//eg711u 100
-//eg711a 101
-ilbc 102
-isac 103
-isacswb 104
-avt 106
-red 117
-cn_wb 98
-cn_swb32 99
-pcm16b 93
-pcm16b_wb 94
-pcm16b_swb32khz 95
-//pcm16b_swb48khz 96
-//mpeg4aac 122
-g722 9
-celt32 114
diff --git a/trunk/src/modules/audio_coding/neteq/unmute_signal.c b/trunk/src/modules/audio_coding/neteq/unmute_signal.c
deleted file mode 100644
index ee9daa8..0000000
--- a/trunk/src/modules/audio_coding/neteq/unmute_signal.c
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This function "unmutes" a vector on a sample by sample basis.
- */
-
-#include "dsp_helpfunctions.h"
-
-#include "signal_processing_library.h"
-
-
-void WebRtcNetEQ_UnmuteSignal(WebRtc_Word16 *pw16_inVec, WebRtc_Word16 *startMuteFact,
- WebRtc_Word16 *pw16_outVec, WebRtc_Word16 unmuteFact,
- WebRtc_Word16 N)
-{
- int i;
- WebRtc_UWord16 w16_tmp;
- WebRtc_Word32 w32_tmp;
-
- w16_tmp = (WebRtc_UWord16) *startMuteFact;
- w32_tmp = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)w16_tmp,6) + 32;
- for (i = 0; i < N; i++)
- {
- pw16_outVec[i]
- = (WebRtc_Word16) ((WEBRTC_SPL_MUL_16_16(w16_tmp, pw16_inVec[i]) + 8192) >> 14);
- w32_tmp += unmuteFact;
- w32_tmp = WEBRTC_SPL_MAX(0, w32_tmp);
- w16_tmp = (WebRtc_UWord16) WEBRTC_SPL_RSHIFT_W32(w32_tmp, 6); /* 20 - 14 = 6 */
- w16_tmp = WEBRTC_SPL_MIN(16384, w16_tmp);
- }
- *startMuteFact = (WebRtc_Word16) w16_tmp;
-}
-
diff --git a/trunk/src/modules/audio_coding/neteq/webrtc_neteq.c b/trunk/src/modules/audio_coding/neteq/webrtc_neteq.c
deleted file mode 100644
index 5e99fd8..0000000
--- a/trunk/src/modules/audio_coding/neteq/webrtc_neteq.c
+++ /dev/null
@@ -1,1654 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Implementation of main NetEQ API.
- */
-
-#include "webrtc_neteq.h"
-#include "webrtc_neteq_internal.h"
-
-#include <assert.h>
-#include <string.h>
-
-#include "typedefs.h"
-#include "signal_processing_library.h"
-
-#include "neteq_error_codes.h"
-#include "mcu_dsp_common.h"
-#include "rtcp.h"
-
-#define RETURN_ON_ERROR( macroExpr, macroInstPtr ) { \
- if ((macroExpr) != 0) { \
- if ((macroExpr) == -1) { \
- (macroInstPtr)->ErrorCode = - (NETEQ_OTHER_ERROR); \
- } else { \
- (macroInstPtr)->ErrorCode = -((WebRtc_Word16) (macroExpr)); \
- } \
- return(-1); \
- } }
-
-int WebRtcNetEQ_strncpy(char *strDest, int numberOfElements,
- const char *strSource, int count)
-{
- /* check vector lengths */
- if (count > numberOfElements)
- {
- strDest[0] = '\0';
- return (-1);
- }
- else
- {
- strncpy(strDest, strSource, count);
- return (0);
- }
-}
-
-/**********************************************************
- * NETEQ Functions
- */
-
-/*****************************************
- * Info functions
- */
-
-int WebRtcNetEQ_GetVersion(char *version)
-{
- char versionString[] = "3.3.0\0 ";
- char endChar[] = " ";
- int i = 0;
- while ((versionString[i] != endChar[0]) && (i <= 20))
- {
- version[i] = versionString[i]; /* To avoid using strcpy */
- i++;
- }
- return (0);
-}
-
-int WebRtcNetEQ_GetErrorCode(void *inst)
-{
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL) return (-1);
- return (NetEqMainInst->ErrorCode);
-}
-
-int WebRtcNetEQ_GetErrorName(int errorCode, char *errorName, int maxStrLen)
-{
- if ((errorName == NULL) || (maxStrLen <= 0))
- {
- return (-1);
- }
-
- if (errorCode < 0)
- {
- errorCode = -errorCode; // absolute value
- }
-
- switch (errorCode)
- {
- case 1: // could be -1
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "OTHER_ERROR", maxStrLen);
- break;
- }
- case 1001:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "FAULTY_INSTRUCTION", maxStrLen);
- break;
- }
- case 1002:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "FAULTY_NETWORK_TYPE", maxStrLen);
- break;
- }
- case 1003:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "FAULTY_DELAYVALUE", maxStrLen);
- break;
- }
- case 1004:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "FAULTY_PLAYOUTMODE", maxStrLen);
- break;
- }
- case 1005:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "CORRUPT_INSTANCE", maxStrLen);
- break;
- }
- case 1006:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "ILLEGAL_MASTER_SLAVE_SWITCH", maxStrLen);
- break;
- }
- case 1007:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "MASTER_SLAVE_ERROR", maxStrLen);
- break;
- }
- case 2001:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "UNKNOWN_BUFSTAT_DECISION", maxStrLen);
- break;
- }
- case 2002:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECOUT_ERROR_DECODING", maxStrLen);
- break;
- }
- case 2003:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECOUT_ERROR_SAMPLEUNDERRUN", maxStrLen);
- break;
- }
- case 2004:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECOUT_ERROR_DECODED_TOO_MUCH",
- maxStrLen);
- break;
- }
- case 3001:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECIN_CNG_ERROR", maxStrLen);
- break;
- }
- case 3002:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECIN_UNKNOWNPAYLOAD", maxStrLen);
- break;
- }
- case 3003:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "RECIN_BUFFERINSERT_ERROR", maxStrLen);
- break;
- }
- case 4001:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "PBUFFER_INIT_ERROR", maxStrLen);
- break;
- }
- case 4002:
- case 4003:
- case 4004:
- case 4005:
- case 4006:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "PBUFFER_INSERT_ERROR1", maxStrLen);
- break;
- }
- case 4007:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "UNKNOWN_G723_HEADER", maxStrLen);
- break;
- }
- case 4008:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "PBUFFER_NONEXISTING_PACKET", maxStrLen);
- break;
- }
- case 4009:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "PBUFFER_NOT_INITIALIZED", maxStrLen);
- break;
- }
- case 4010:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "AMBIGUOUS_ILBC_FRAME_SIZE", maxStrLen);
- break;
- }
- case 5001:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_FULL", maxStrLen);
- break;
- }
- case 5002:
- case 5003:
- case 5004:
- case 5005:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_NOT_EXIST", maxStrLen);
- break;
- }
- case 5006:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_UNKNOWN_CODEC", maxStrLen);
- break;
- }
- case 5007:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_PAYLOAD_TAKEN", maxStrLen);
- break;
- }
- case 5008:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_UNSUPPORTED_CODEC", maxStrLen);
- break;
- }
- case 5009:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "CODEC_DB_UNSUPPORTED_FS", maxStrLen);
- break;
- }
- case 6001:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "DTMF_DEC_PARAMETER_ERROR", maxStrLen);
- break;
- }
- case 6002:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "DTMF_INSERT_ERROR", maxStrLen);
- break;
- }
- case 6003:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "DTMF_GEN_UNKNOWN_SAMP_FREQ", maxStrLen);
- break;
- }
- case 6004:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "DTMF_NOT_SUPPORTED", maxStrLen);
- break;
- }
- case 7001:
- case 7002:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "RED_SPLIT_ERROR", maxStrLen);
- break;
- }
- case 7003:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "RTP_TOO_SHORT_PACKET", maxStrLen);
- break;
- }
- case 7004:
- {
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "RTP_CORRUPT_PACKET", maxStrLen);
- break;
- }
- default:
- {
- /* check for decoder error ranges */
- if (errorCode >= 6010 && errorCode <= 6810)
- {
- /* iSAC error code */
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "iSAC ERROR", maxStrLen);
- break;
- }
-
- WebRtcNetEQ_strncpy(errorName, maxStrLen, "UNKNOWN_ERROR", maxStrLen);
- return (-1);
- }
- }
-
- return (0);
-}
-
-/* Assign functions (create not allowed in order to avoid malloc in lib) */
-int WebRtcNetEQ_AssignSize(int *sizeinbytes)
-{
- *sizeinbytes = (sizeof(MainInst_t) * 2) / sizeof(WebRtc_Word16);
- return (0);
-}
-
-int WebRtcNetEQ_Assign(void **inst, void *NETEQ_inst_Addr)
-{
- int ok = 0;
- MainInst_t *NetEqMainInst = (MainInst_t*) NETEQ_inst_Addr;
- *inst = NETEQ_inst_Addr;
- if (*inst == NULL) return (-1);
- /* Clear memory */
- WebRtcSpl_MemSetW16((WebRtc_Word16*) NetEqMainInst, 0,
- (sizeof(MainInst_t) / sizeof(WebRtc_Word16)));
- ok = WebRtcNetEQ_McuReset(&NetEqMainInst->MCUinst);
- if (ok != 0)
- {
- NetEqMainInst->ErrorCode = -ok;
- return (-1);
- }
- return (0);
-}
-
-int WebRtcNetEQ_GetRecommendedBufferSize(void *inst, const enum WebRtcNetEQDecoder *codec,
- int noOfCodecs, enum WebRtcNetEQNetworkType nwType,
- int *MaxNoOfPackets, int *sizeinbytes)
-{
- int ok = 0;
- int multiplier;
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL) return (-1);
- *MaxNoOfPackets = 0;
- *sizeinbytes = 0;
- ok = WebRtcNetEQ_GetDefaultCodecSettings(codec, noOfCodecs, sizeinbytes, MaxNoOfPackets);
- if (ok != 0)
- {
- NetEqMainInst->ErrorCode = -ok;
- return (-1);
- }
- if (nwType == kUDPNormal)
- {
- multiplier = 1;
- }
- else if (nwType == kUDPVideoSync)
- {
- multiplier = 4;
- }
- else if (nwType == kTCPNormal)
- {
- multiplier = 4;
- }
- else if (nwType == kTCPLargeJitter)
- {
- multiplier = 8;
- }
- else if (nwType == kTCPXLargeJitter)
- {
- multiplier = 20;
- }
- else
- {
- NetEqMainInst->ErrorCode = -FAULTY_NETWORK_TYPE;
- return (-1);
- }
- *MaxNoOfPackets = (*MaxNoOfPackets) * multiplier;
- *sizeinbytes = (*sizeinbytes) * multiplier;
- if (ok != 0)
- {
- NetEqMainInst->ErrorCode = -ok;
- return (-1);
- }
- return (ok);
-}
-
-int WebRtcNetEQ_AssignBuffer(void *inst, int MaxNoOfPackets, void *NETEQ_Buffer_Addr,
- int sizeinbytes)
-{
- int ok;
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL) return (-1);
- ok = WebRtcNetEQ_PacketBufferInit(&NetEqMainInst->MCUinst.PacketBuffer_inst,
- MaxNoOfPackets, (WebRtc_Word16*) NETEQ_Buffer_Addr, (sizeinbytes >> 1));
- if (ok != 0)
- {
- NetEqMainInst->ErrorCode = -ok;
- return (-1);
- }
- return (ok);
-}
-
-/************************************************
- * Init functions
- */
-
-/****************************************************************************
- * WebRtcNetEQ_Init(...)
- *
- * Initialize NetEQ.
- *
- * Input:
- * - inst : NetEQ instance
- * - fs : Initial sample rate in Hz (may change with payload)
- *
- * Output:
- * - inst : Initialized NetEQ instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_Init(void *inst, WebRtc_UWord16 fs)
-{
- int ok = 0;
-
- /* Typecast inst to internal instance format */
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
-
- if (NetEqMainInst == NULL)
- {
- return (-1);
- }
-
-#ifdef NETEQ_VAD
- /* Start out with no PostDecode VAD instance */
- NetEqMainInst->DSPinst.VADInst.VADState = NULL;
- /* Also set all VAD function pointers to NULL */
- NetEqMainInst->DSPinst.VADInst.initFunction = NULL;
- NetEqMainInst->DSPinst.VADInst.setmodeFunction = NULL;
- NetEqMainInst->DSPinst.VADInst.VADFunction = NULL;
-#endif /* NETEQ_VAD */
-
- ok = WebRtcNetEQ_DSPinit(NetEqMainInst); /* Init addresses between MCU and DSP */
- RETURN_ON_ERROR(ok, NetEqMainInst);
-
- ok = WebRtcNetEQ_DSPInit(&NetEqMainInst->DSPinst, fs); /* Init dsp side */
- RETURN_ON_ERROR(ok, NetEqMainInst);
- /* set BGN mode to default, since it is not cleared by DSP init function */
- NetEqMainInst->DSPinst.BGNInst.bgnMode = BGN_ON;
-
- /* init statistics functions and counters */
- ok = WebRtcNetEQ_ClearInCallStats(&NetEqMainInst->DSPinst);
- RETURN_ON_ERROR(ok, NetEqMainInst);
- ok = WebRtcNetEQ_ClearPostCallStats(&NetEqMainInst->DSPinst);
- RETURN_ON_ERROR(ok, NetEqMainInst);
- ok = WebRtcNetEQ_ResetMcuJitterStat(&NetEqMainInst->MCUinst);
- RETURN_ON_ERROR(ok, NetEqMainInst);
-
- /* flush packet buffer */
- ok = WebRtcNetEQ_PacketBufferFlush(&NetEqMainInst->MCUinst.PacketBuffer_inst);
- RETURN_ON_ERROR(ok, NetEqMainInst);
-
- /* set some variables to initial values */
- NetEqMainInst->MCUinst.current_Codec = -1;
- NetEqMainInst->MCUinst.current_Payload = -1;
- NetEqMainInst->MCUinst.first_packet = 1;
- NetEqMainInst->MCUinst.one_desc = 0;
- NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.extraDelayMs = 0;
- NetEqMainInst->MCUinst.NoOfExpandCalls = 0;
- NetEqMainInst->MCUinst.fs = fs;
-
-#ifdef NETEQ_ATEVENT_DECODE
- /* init DTMF decoder */
- ok = WebRtcNetEQ_DtmfDecoderInit(&(NetEqMainInst->MCUinst.DTMF_inst),fs,560);
- RETURN_ON_ERROR(ok, NetEqMainInst);
-#endif
-
- /* init RTCP statistics */
- WebRtcNetEQ_RTCPInit(&(NetEqMainInst->MCUinst.RTCP_inst), 0);
-
- /* set BufferStat struct to zero */
- WebRtcSpl_MemSetW16((WebRtc_Word16*) &(NetEqMainInst->MCUinst.BufferStat_inst), 0,
- sizeof(BufstatsInst_t) / sizeof(WebRtc_Word16));
-
- /* reset automode */
- WebRtcNetEQ_ResetAutomode(&(NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst),
- NetEqMainInst->MCUinst.PacketBuffer_inst.maxInsertPositions);
-
- NetEqMainInst->ErrorCode = 0;
-
-#ifdef NETEQ_STEREO
- /* set master/slave info to undecided */
- NetEqMainInst->masterSlave = 0;
-#endif
-
- return (ok);
-}
-
-int WebRtcNetEQ_FlushBuffers(void *inst)
-{
- int ok = 0;
-
- /* Typecast inst to internal instance format */
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
-
- if (NetEqMainInst == NULL)
- {
- return (-1);
- }
-
- /* Flush packet buffer */
- ok = WebRtcNetEQ_PacketBufferFlush(&NetEqMainInst->MCUinst.PacketBuffer_inst);
- RETURN_ON_ERROR(ok, NetEqMainInst);
-
- /* Set MCU to wait for new codec */
- NetEqMainInst->MCUinst.first_packet = 1;
-
- /* Flush speech buffer */
- ok = WebRtcNetEQ_FlushSpeechBuffer(&NetEqMainInst->DSPinst);
- RETURN_ON_ERROR(ok, NetEqMainInst);
-
- return 0;
-}
-
-int WebRtcNetEQ_SetAVTPlayout(void *inst, int PlayoutAVTon)
-{
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL) return (-1);
-#ifdef NETEQ_ATEVENT_DECODE
- NetEqMainInst->MCUinst.AVT_PlayoutOn = PlayoutAVTon;
- return(0);
-#else
- if (PlayoutAVTon != 0)
- {
- NetEqMainInst->ErrorCode = -DTMF_NOT_SUPPORTED;
- return (-1);
- }
- else
- {
- return (0);
- }
-#endif
-}
-
-int WebRtcNetEQ_SetExtraDelay(void *inst, int DelayInMs)
-{
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL) return (-1);
- if ((DelayInMs < 0) || (DelayInMs > 1000))
- {
- NetEqMainInst->ErrorCode = -FAULTY_DELAYVALUE;
- return (-1);
- }
- NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.extraDelayMs = DelayInMs;
- return (0);
-}
-
-int WebRtcNetEQ_SetPlayoutMode(void *inst, enum WebRtcNetEQPlayoutMode playoutMode)
-{
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL) return (-1);
- if ((playoutMode != kPlayoutOn) && (playoutMode != kPlayoutOff) && (playoutMode
- != kPlayoutFax) && (playoutMode != kPlayoutStreaming))
- {
- NetEqMainInst->ErrorCode = -FAULTY_PLAYOUTMODE;
- return (-1);
- }
- else
- {
- NetEqMainInst->MCUinst.NetEqPlayoutMode = playoutMode;
- return (0);
- }
-}
-
-int WebRtcNetEQ_SetBGNMode(void *inst, enum WebRtcNetEQBGNMode bgnMode)
-{
-
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
-
- /* Instance sanity */
- if (NetEqMainInst == NULL) return (-1);
-
- /* Check for corrupt/cleared instance */
- if (NetEqMainInst->MCUinst.main_inst != NetEqMainInst)
- {
- /* Instance is corrupt */
- NetEqMainInst->ErrorCode = CORRUPT_INSTANCE;
- return (-1);
- }
-
- NetEqMainInst->DSPinst.BGNInst.bgnMode = (enum BGNMode) bgnMode;
-
- return (0);
-}
-
-int WebRtcNetEQ_GetBGNMode(const void *inst, enum WebRtcNetEQBGNMode *bgnMode)
-{
-
- const MainInst_t *NetEqMainInst = (const MainInst_t*) inst;
-
- /* Instance sanity */
- if (NetEqMainInst == NULL) return (-1);
-
- *bgnMode = (enum WebRtcNetEQBGNMode) NetEqMainInst->DSPinst.BGNInst.bgnMode;
-
- return (0);
-}
-
-/************************************************
- * CodecDB functions
- */
-
-int WebRtcNetEQ_CodecDbReset(void *inst)
-{
- int ok = 0;
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL) return (-1);
- ok = WebRtcNetEQ_DbReset(&NetEqMainInst->MCUinst.codec_DB_inst);
- if (ok != 0)
- {
- NetEqMainInst->ErrorCode = -ok;
- return (-1);
- }
-
- /* set function pointers to NULL to prevent RecOut from using the codec */
- NetEqMainInst->DSPinst.codec_ptr_inst.funcDecode = NULL;
- NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodeRCU = NULL;
- NetEqMainInst->DSPinst.codec_ptr_inst.funcAddLatePkt = NULL;
- NetEqMainInst->DSPinst.codec_ptr_inst.funcDecode = NULL;
- NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodeInit = NULL;
- NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodePLC = NULL;
- NetEqMainInst->DSPinst.codec_ptr_inst.funcGetMDinfo = NULL;
- NetEqMainInst->DSPinst.codec_ptr_inst.funcUpdBWEst = NULL;
- NetEqMainInst->DSPinst.codec_ptr_inst.funcGetErrorCode = NULL;
-
- return (0);
-}
-
-int WebRtcNetEQ_CodecDbGetSizeInfo(void *inst, WebRtc_Word16 *UsedEntries,
- WebRtc_Word16 *MaxEntries)
-{
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL) return (-1);
- *MaxEntries = NUM_CODECS;
- *UsedEntries = NetEqMainInst->MCUinst.codec_DB_inst.nrOfCodecs;
- return (0);
-}
-
-int WebRtcNetEQ_CodecDbGetCodecInfo(void *inst, WebRtc_Word16 Entry,
- enum WebRtcNetEQDecoder *codec)
-{
- int i;
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL) return (-1);
- *codec = (enum WebRtcNetEQDecoder) 0;
- if ((Entry >= 0) && (Entry < NetEqMainInst->MCUinst.codec_DB_inst.nrOfCodecs))
- {
- for (i = 0; i < NUM_TOTAL_CODECS; i++)
- {
- if (NetEqMainInst->MCUinst.codec_DB_inst.position[i] == Entry)
- {
- *codec = (enum WebRtcNetEQDecoder) i;
- }
- }
- }
- else
- {
- NetEqMainInst->ErrorCode = -(CODEC_DB_NOT_EXIST1);
- return (-1);
- }
- return (0);
-}
-
-int WebRtcNetEQ_CodecDbAdd(void *inst, WebRtcNetEQ_CodecDef *codecInst)
-{
- int ok = 0;
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL) return (-1);
- ok = WebRtcNetEQ_DbAdd(&NetEqMainInst->MCUinst.codec_DB_inst, codecInst->codec,
- codecInst->payloadType, codecInst->funcDecode, codecInst->funcDecodeRCU,
- codecInst->funcDecodePLC, codecInst->funcDecodeInit, codecInst->funcAddLatePkt,
- codecInst->funcGetMDinfo, codecInst->funcGetPitch, codecInst->funcUpdBWEst,
- codecInst->funcGetErrorCode, codecInst->codec_state, codecInst->codec_fs);
- if (ok != 0)
- {
- NetEqMainInst->ErrorCode = -ok;
- return (-1);
- }
- return (ok);
-}
-
-int WebRtcNetEQ_CodecDbRemove(void *inst, enum WebRtcNetEQDecoder codec)
-{
- int ok = 0;
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL) return (-1);
-
- /* check if currently used codec is being removed */
- if (NetEqMainInst->MCUinst.current_Codec == (WebRtc_Word16) codec)
- {
- /* set function pointers to NULL to prevent RecOut from using the codec */
- NetEqMainInst->DSPinst.codec_ptr_inst.funcDecode = NULL;
- NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodeRCU = NULL;
- NetEqMainInst->DSPinst.codec_ptr_inst.funcAddLatePkt = NULL;
- NetEqMainInst->DSPinst.codec_ptr_inst.funcDecode = NULL;
- NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodeInit = NULL;
- NetEqMainInst->DSPinst.codec_ptr_inst.funcDecodePLC = NULL;
- NetEqMainInst->DSPinst.codec_ptr_inst.funcGetMDinfo = NULL;
- NetEqMainInst->DSPinst.codec_ptr_inst.funcUpdBWEst = NULL;
- NetEqMainInst->DSPinst.codec_ptr_inst.funcGetErrorCode = NULL;
- }
-
- ok = WebRtcNetEQ_DbRemove(&NetEqMainInst->MCUinst.codec_DB_inst, codec);
- if (ok != 0)
- {
- NetEqMainInst->ErrorCode = -ok;
- return (-1);
- }
- return (ok);
-}
-
-/*********************************
- * Real-time functions
- */
-
-int WebRtcNetEQ_RecIn(void *inst, WebRtc_Word16 *p_w16datagramstart, WebRtc_Word16 w16_RTPlen,
- WebRtc_UWord32 uw32_timeRec)
-{
- int ok = 0;
- RTPPacket_t RTPpacket;
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL) return (-1);
-
- /* Check for corrupt/cleared instance */
- if (NetEqMainInst->MCUinst.main_inst != NetEqMainInst)
- {
- /* Instance is corrupt */
- NetEqMainInst->ErrorCode = CORRUPT_INSTANCE;
- return (-1);
- }
-
- /* Parse RTP header */
- ok = WebRtcNetEQ_RTPPayloadInfo(p_w16datagramstart, w16_RTPlen, &RTPpacket);
- if (ok != 0)
- {
- NetEqMainInst->ErrorCode = -ok;
- return (-1);
- }
-
- ok = WebRtcNetEQ_RecInInternal(&NetEqMainInst->MCUinst, &RTPpacket, uw32_timeRec);
- if (ok != 0)
- {
- NetEqMainInst->ErrorCode = -ok;
- return (-1);
- }
- return (ok);
-}
-
-/****************************************************************************
- * WebRtcNetEQ_RecInRTPStruct(...)
- *
- * Alternative RecIn function, used when the RTP data has already been
- * parsed into an RTP info struct (WebRtcNetEQ_RTPInfo).
- *
- * Input:
- * - inst : NetEQ instance
- * - rtpInfo : Pointer to RTP info
- * - payloadPtr : Pointer to the RTP payload (first byte after header)
- * - payloadLenBytes : Length (in bytes) of the payload in payloadPtr
- * - timeRec : Receive time (in timestamps of the used codec)
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcNetEQ_RecInRTPStruct(void *inst, WebRtcNetEQ_RTPInfo *rtpInfo,
- const WebRtc_UWord8 *payloadPtr, WebRtc_Word16 payloadLenBytes,
- WebRtc_UWord32 uw32_timeRec)
-{
- int ok = 0;
- RTPPacket_t RTPpacket;
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL)
- {
- return (-1);
- }
-
- /* Check for corrupt/cleared instance */
- if (NetEqMainInst->MCUinst.main_inst != NetEqMainInst)
- {
- /* Instance is corrupt */
- NetEqMainInst->ErrorCode = CORRUPT_INSTANCE;
- return (-1);
- }
-
- /* Load NetEQ's RTP struct from Module RTP struct */
- RTPpacket.payloadType = rtpInfo->payloadType;
- RTPpacket.seqNumber = rtpInfo->sequenceNumber;
- RTPpacket.timeStamp = rtpInfo->timeStamp;
- RTPpacket.ssrc = rtpInfo->SSRC;
- RTPpacket.payload = (const WebRtc_Word16*) payloadPtr;
- RTPpacket.payloadLen = payloadLenBytes;
- RTPpacket.starts_byte1 = 0;
-
- ok = WebRtcNetEQ_RecInInternal(&NetEqMainInst->MCUinst, &RTPpacket, uw32_timeRec);
- if (ok != 0)
- {
- NetEqMainInst->ErrorCode = -ok;
- return (-1);
- }
- return (ok);
-}
-
-int WebRtcNetEQ_RecOut(void *inst, WebRtc_Word16 *pw16_outData, WebRtc_Word16 *pw16_len)
-{
- int ok = 0;
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
-#ifdef NETEQ_STEREO
- MasterSlaveInfo msInfo;
- msInfo.msMode = NETEQ_MONO;
-#endif
-
- if (NetEqMainInst == NULL) return (-1);
-
- /* Check for corrupt/cleared instance */
- if (NetEqMainInst->DSPinst.main_inst != NetEqMainInst)
- {
- /* Instance is corrupt */
- NetEqMainInst->ErrorCode = CORRUPT_INSTANCE;
- return (-1);
- }
-
-#ifdef NETEQ_STEREO
- NetEqMainInst->DSPinst.msInfo = &msInfo;
-#endif
-
- ok = WebRtcNetEQ_RecOutInternal(&NetEqMainInst->DSPinst, pw16_outData,
- pw16_len, 0 /* not BGN only */);
- if (ok != 0)
- {
- NetEqMainInst->ErrorCode = -ok;
- return (-1);
- }
- return (ok);
-}
-
-/****************************************************************************
- * WebRtcNetEQ_RecOutMasterSlave(...)
- *
- * RecOut function for running several NetEQ instances in master/slave mode.
- * One master can be used to control several slaves.
- *
- * Input:
- * - inst : NetEQ instance
- * - isMaster : Non-zero indicates that this is the master channel
- * - msInfo : (slave only) Information from master
- *
- * Output:
- * - inst : Updated NetEQ instance
- * - pw16_outData : Pointer to vector where output should be written
- * - pw16_len : Pointer to variable where output length is returned
- * - msInfo : (master only) Information to slave(s)
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_RecOutMasterSlave(void *inst, WebRtc_Word16 *pw16_outData,
- WebRtc_Word16 *pw16_len, void *msInfo,
- WebRtc_Word16 isMaster)
-{
-#ifndef NETEQ_STEREO
- /* Stereo not supported */
- return(-1);
-#else
- int ok = 0;
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
-
- if (NetEqMainInst == NULL) return (-1);
-
- /* Check for corrupt/cleared instance */
- if (NetEqMainInst->DSPinst.main_inst != NetEqMainInst)
- {
- /* Instance is corrupt */
- NetEqMainInst->ErrorCode = CORRUPT_INSTANCE;
- return (-1);
- }
-
- if (msInfo == NULL)
- {
- /* msInfo not provided */
- NetEqMainInst->ErrorCode = NETEQ_OTHER_ERROR;
- return (-1);
- }
-
- /* translate from external to internal Master/Slave information */
- NetEqMainInst->DSPinst.msInfo = (MasterSlaveInfo *) msInfo;
-
- /* check that we have not done a master/slave switch without first re-initializing */
- if ((NetEqMainInst->masterSlave == 1 && !isMaster) || /* switch from master to slave */
- (NetEqMainInst->masterSlave == 2 && isMaster)) /* switch from slave to master */
- {
- NetEqMainInst->ErrorCode = ILLEGAL_MASTER_SLAVE_SWITCH;
- return (-1);
- }
-
- if (!isMaster)
- {
- /* this is the slave */
- NetEqMainInst->masterSlave = 2;
- NetEqMainInst->DSPinst.msInfo->msMode = NETEQ_SLAVE;
- }
- else
- {
- NetEqMainInst->DSPinst.msInfo->msMode = NETEQ_MASTER;
- }
-
- ok = WebRtcNetEQ_RecOutInternal(&NetEqMainInst->DSPinst, pw16_outData,
- pw16_len, 0 /* not BGN only */);
- if (ok != 0)
- {
- NetEqMainInst->ErrorCode = -ok;
- return (-1);
- }
-
- if (isMaster)
- {
- /* this is the master */
- NetEqMainInst->masterSlave = 1;
- }
-
- return (ok);
-#endif
-}
-
-int WebRtcNetEQ_GetMasterSlaveInfoSize()
-{
-#ifdef NETEQ_STEREO
- return (sizeof(MasterSlaveInfo));
-#else
- return(-1);
-#endif
-}
-
-/* Special RecOut that does not do any decoding. */
-int WebRtcNetEQ_RecOutNoDecode(void *inst, WebRtc_Word16 *pw16_outData,
- WebRtc_Word16 *pw16_len)
-{
- int ok = 0;
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
-#ifdef NETEQ_STEREO
- MasterSlaveInfo msInfo;
-#endif
-
- if (NetEqMainInst == NULL) return (-1);
-
- /* Check for corrupt/cleared instance */
- if (NetEqMainInst->DSPinst.main_inst != NetEqMainInst)
- {
- /* Instance is corrupt */
- NetEqMainInst->ErrorCode = CORRUPT_INSTANCE;
- return (-1);
- }
-
-#ifdef NETEQ_STEREO
- /* keep same mode as before */
- switch (NetEqMainInst->masterSlave)
- {
- case 1:
- {
- msInfo.msMode = NETEQ_MASTER;
- break;
- }
- case 2:
- {
- msInfo.msMode = NETEQ_SLAVE;
- break;
- }
- default:
- {
- msInfo.msMode = NETEQ_MONO;
- break;
- }
- }
-
- NetEqMainInst->DSPinst.msInfo = &msInfo;
-#endif
-
- ok = WebRtcNetEQ_RecOutInternal(&NetEqMainInst->DSPinst, pw16_outData,
- pw16_len, 1 /* BGN only */);
- if (ok != 0)
- {
- NetEqMainInst->ErrorCode = -ok;
- return (-1);
- }
- return (ok);
-}
-
-int WebRtcNetEQ_GetRTCPStats(void *inst, WebRtcNetEQ_RTCPStat *RTCP_inst)
-{
- int ok = 0;
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL) return (-1);
- ok = WebRtcNetEQ_RTCPGetStats(&NetEqMainInst->MCUinst.RTCP_inst,
- &RTCP_inst->fraction_lost, &RTCP_inst->cum_lost, &RTCP_inst->ext_max,
- &RTCP_inst->jitter, 0);
- if (ok != 0)
- {
- NetEqMainInst->ErrorCode = -ok;
- return (-1);
- }
- return (ok);
-}
-
-int WebRtcNetEQ_GetRTCPStatsNoReset(void *inst, WebRtcNetEQ_RTCPStat *RTCP_inst)
-{
- int ok = 0;
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL) return (-1);
- ok = WebRtcNetEQ_RTCPGetStats(&NetEqMainInst->MCUinst.RTCP_inst,
- &RTCP_inst->fraction_lost, &RTCP_inst->cum_lost, &RTCP_inst->ext_max,
- &RTCP_inst->jitter, 1);
- if (ok != 0)
- {
- NetEqMainInst->ErrorCode = -ok;
- return (-1);
- }
- return (ok);
-}
-
-int WebRtcNetEQ_GetSpeechTimeStamp(void *inst, WebRtc_UWord32 *timestamp)
-{
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL) return (-1);
-
- if (NetEqMainInst->MCUinst.TSscalingInitialized)
- {
- *timestamp = WebRtcNetEQ_ScaleTimestampInternalToExternal(&NetEqMainInst->MCUinst,
- NetEqMainInst->DSPinst.videoSyncTimestamp);
- }
- else
- {
- *timestamp = NetEqMainInst->DSPinst.videoSyncTimestamp;
- }
-
- return (0);
-}
-
-/****************************************************************************
- * WebRtcNetEQ_GetSpeechOutputType(...)
- *
- * Get the output type for the audio provided by the latest call to
- * WebRtcNetEQ_RecOut().
- *
- * kOutputNormal = normal audio (possibly processed)
- * kOutputPLC = loss concealment through stretching audio
- * kOutputCNG = comfort noise (codec-internal or RFC3389)
- * kOutputPLCtoCNG = background noise only due to long expand or error
- * kOutputVADPassive = PostDecode VAD signalling passive speaker
- *
- * Input:
- * - inst : NetEQ instance
- *
- * Output:
- * - outputType : Output type from enum list WebRtcNetEQOutputType
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_GetSpeechOutputType(void *inst, enum WebRtcNetEQOutputType *outputType)
-{
- /* Typecast to internal instance type */
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
-
- if (NetEqMainInst == NULL)
- {
- return (-1);
- }
-
- if ((NetEqMainInst->DSPinst.w16_mode & MODE_BGN_ONLY) != 0)
- {
- /* If last mode was background noise only */
- *outputType = kOutputPLCtoCNG;
-
- }
- else if ((NetEqMainInst->DSPinst.w16_mode == MODE_CODEC_INTERNAL_CNG)
- || (NetEqMainInst->DSPinst.w16_mode == MODE_RFC3389CNG))
- {
- /* If CN or internal CNG */
- *outputType = kOutputCNG;
-
-#ifdef NETEQ_VAD
- }
- else if ( NetEqMainInst->DSPinst.VADInst.VADDecision == 0 )
- {
- /* post-decode VAD says passive speaker */
- *outputType = kOutputVADPassive;
-#endif /* NETEQ_VAD */
-
- }
- else if ((NetEqMainInst->DSPinst.w16_mode == MODE_EXPAND)
- && (NetEqMainInst->DSPinst.ExpandInst.w16_expandMuteFactor == 0))
- {
- /* Expand mode has faded down to background noise only (very long expand) */
- *outputType = kOutputPLCtoCNG;
-
- }
- else if (NetEqMainInst->DSPinst.w16_mode == MODE_EXPAND)
- {
- /* PLC mode */
- *outputType = kOutputPLC;
-
- }
- else
- {
- /* Normal speech output type (can still be manipulated, e.g., accelerated) */
- *outputType = kOutputNormal;
- }
-
- return (0);
-}
-
-/**********************************
- * Functions related to VQmon
- */
-
-#define WEBRTC_NETEQ_CONCEALMENTFLAG_LOST 0x01
-#define WEBRTC_NETEQ_CONCEALMENTFLAG_DISCARDED 0x02
-#define WEBRTC_NETEQ_CONCEALMENTFLAG_SUPRESS 0x04
-#define WEBRTC_NETEQ_CONCEALMENTFLAG_CNGACTIVE 0x80
-
-int WebRtcNetEQ_VQmonRecOutStatistics(void *inst, WebRtc_UWord16 *validVoiceDurationMs,
- WebRtc_UWord16 *concealedVoiceDurationMs,
- WebRtc_UWord8 *concealedVoiceFlags)
-{
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- WebRtc_Word16 fs_mult;
- WebRtc_Word16 ms_lost;
- if (NetEqMainInst == NULL) return (-1);
- fs_mult = WebRtcSpl_DivW32W16ResW16(NetEqMainInst->MCUinst.fs, 8000);
-
- ms_lost = WebRtcSpl_DivW32W16ResW16(
- (WebRtc_Word32) NetEqMainInst->DSPinst.w16_concealedTS, (WebRtc_Word16) (8 * fs_mult));
- if (ms_lost > NetEqMainInst->DSPinst.millisecondsPerCall) ms_lost
- = NetEqMainInst->DSPinst.millisecondsPerCall;
-
- *validVoiceDurationMs = NetEqMainInst->DSPinst.millisecondsPerCall - ms_lost;
- *concealedVoiceDurationMs = ms_lost;
- if (ms_lost > 0)
- {
- *concealedVoiceFlags = WEBRTC_NETEQ_CONCEALMENTFLAG_LOST;
- }
- else
- {
- *concealedVoiceFlags = 0;
- }
- NetEqMainInst->DSPinst.w16_concealedTS -= ms_lost * (8 * fs_mult);
-
- return (0);
-}
-
-int WebRtcNetEQ_VQmonGetConfiguration(void *inst, WebRtc_UWord16 *absMaxDelayMs,
- WebRtc_UWord8 *adaptationRate)
-{
- /* Dummy check the inst, just to avoid compiler warnings. */
- if (inst == NULL)
- {
- /* Do nothing. */
- }
-
- /* Hardcoded variables that are used for VQmon as jitter buffer parameters */
- *absMaxDelayMs = 240;
- *adaptationRate = 1;
- return (0);
-}
-
-int WebRtcNetEQ_VQmonGetRxStatistics(void *inst, WebRtc_UWord16 *avgDelayMs,
- WebRtc_UWord16 *maxDelayMs)
-{
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL) return (-1);
- *avgDelayMs = (WebRtc_UWord16) (NetEqMainInst->MCUinst.BufferStat_inst.avgDelayMsQ8 >> 8);
- *maxDelayMs = (WebRtc_UWord16) NetEqMainInst->MCUinst.BufferStat_inst.maxDelayMs;
- return (0);
-}
-
-/*************************************
- * Statistics functions
- */
-
-/* Get the "in-call" statistics from NetEQ.
- * The statistics are reset after the query. */
-int WebRtcNetEQ_GetNetworkStatistics(void *inst, WebRtcNetEQ_NetworkStatistics *stats)
-
-{
-
- WebRtc_UWord16 tempU16;
- WebRtc_UWord32 tempU32, tempU32_2;
- int numShift;
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
-
- /* Instance sanity */
- if (NetEqMainInst == NULL) return (-1);
-
- /*******************/
- /* Get buffer size */
- /*******************/
-
- if (NetEqMainInst->MCUinst.fs != 0)
- {
- WebRtc_Word32 temp32;
- /* Query packet buffer for number of samples. */
- temp32 = WebRtcNetEQ_PacketBufferGetSize(
- &NetEqMainInst->MCUinst.PacketBuffer_inst);
-
- /* Divide by sample rate.
- * Calculate temp32 * 1000 / fs to get result in ms. */
- stats->currentBufferSize = (WebRtc_UWord16)
- WebRtcSpl_DivU32U16(temp32 * 1000, NetEqMainInst->MCUinst.fs);
-
- /* Add number of samples yet to play in sync buffer. */
- temp32 = (WebRtc_Word32) (NetEqMainInst->DSPinst.endPosition -
- NetEqMainInst->DSPinst.curPosition);
- stats->currentBufferSize += (WebRtc_UWord16)
- WebRtcSpl_DivU32U16(temp32 * 1000, NetEqMainInst->MCUinst.fs);
- }
- else
- {
- /* Sample rate not initialized. */
- stats->currentBufferSize = 0;
- }
-
- /***************************/
- /* Get optimal buffer size */
- /***************************/
-
- if (NetEqMainInst->MCUinst.fs != 0 && NetEqMainInst->MCUinst.fs <= WEBRTC_SPL_WORD16_MAX)
- {
- /* preferredBufferSize = Bopt * packSizeSamples / (fs/1000) */
- stats->preferredBufferSize
- = (WebRtc_UWord16) WEBRTC_SPL_MUL_16_16(
- (WebRtc_Word16) ((NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.optBufLevel) >> 8), /* optimal buffer level in packets shifted to Q0 */
- WebRtcSpl_DivW32W16ResW16(
- (WebRtc_Word32) NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.packetSpeechLenSamp, /* samples per packet */
- WebRtcSpl_DivW32W16ResW16( (WebRtc_Word32) NetEqMainInst->MCUinst.fs, (WebRtc_Word16) 1000 ) /* samples per ms */
- ) );
-
- /* add extra delay */
- if (NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.extraDelayMs > 0)
- {
- stats->preferredBufferSize
- += NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.extraDelayMs;
- }
- }
- else
- {
- /* sample rate not initialized */
- stats->preferredBufferSize = 0;
- }
-
- /***********************************/
- /* Check if jitter peaks are found */
- /***********************************/
-
- stats->jitterPeaksFound =
- NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst.peakFound;
-
- /***********************/
- /* Calculate loss rate */
- /***********************/
-
- /* timestamps elapsed since last report */
- tempU32 = NetEqMainInst->MCUinst.lastReportTS;
-
- if (NetEqMainInst->MCUinst.lostTS == 0)
- {
- /* no losses */
- stats->currentPacketLossRate = 0;
- }
- else if (NetEqMainInst->MCUinst.lostTS < tempU32)
- {
- /* calculate shifts; we want the result in Q14 */
- numShift = WebRtcSpl_NormU32(NetEqMainInst->MCUinst.lostTS); /* numerator shift for normalize */
-
- if (numShift < 14)
- {
- /* cannot shift numerator 14 steps; shift denominator too */
- tempU32 = WEBRTC_SPL_RSHIFT_U32(tempU32, 14-numShift); /* right-shift */
- }
- else
- {
- /* shift no more than 14 steps */
- numShift = 14;
- }
-
- if (tempU32 == 0)
- {
- /* check for zero denominator; result should be zero in this case */
- stats->currentPacketLossRate = 0;
- }
- else
- {
- /* check that denominator fits in signed 16-bit */
- while (tempU32 > WEBRTC_SPL_WORD16_MAX)
- {
- tempU32 >>= 1; /* right-shift 1 step */
- numShift--; /* compensate in numerator */
- }
- tempU16 = (WebRtc_UWord16) tempU32;
-
- /* do the shift of numerator */
- tempU32
- = WEBRTC_SPL_SHIFT_W32( (WebRtc_UWord32) NetEqMainInst->MCUinst.lostTS, numShift);
-
- stats->currentPacketLossRate = (WebRtc_UWord16) WebRtcSpl_DivU32U16(tempU32,
- tempU16);
- }
- }
- else
- {
- /* lost count is larger than elapsed time count; probably timestamp wrap-around or something else wrong */
- /* set loss rate = 1 */
- stats->currentPacketLossRate = 1 << 14; /* 1 in Q14 */
- }
-
- /**************************/
- /* Calculate discard rate */
- /**************************/
-
- /* timestamps elapsed since last report */
- tempU32 = NetEqMainInst->MCUinst.lastReportTS;
-
- /* number of discarded samples */
- tempU32_2
- = WEBRTC_SPL_MUL_16_U16( (WebRtc_Word16) NetEqMainInst->MCUinst.PacketBuffer_inst.packSizeSamples,
- NetEqMainInst->MCUinst.PacketBuffer_inst.discardedPackets);
-
- if (tempU32_2 == 0)
- {
- /* no discarded samples */
- stats->currentDiscardRate = 0;
- }
- else if (tempU32_2 < tempU32)
- {
- /* calculate shifts; we want the result in Q14 */
- numShift = WebRtcSpl_NormU32(tempU32_2); /* numerator shift for normalize */
-
- if (numShift < 14)
- {
- /* cannot shift numerator 14 steps; shift denominator too */
- tempU32 = WEBRTC_SPL_RSHIFT_U32(tempU32, 14-numShift); /* right-shift */
- }
- else
- {
- /* shift no more than 14 steps */
- numShift = 14;
- }
-
- if (tempU32 == 0)
- {
- /* check for zero denominator; result should be zero in this case */
- stats->currentDiscardRate = 0;
- }
- else
- {
- /* check that denominator fits in signed 16-bit */
- while (tempU32 > WEBRTC_SPL_WORD16_MAX)
- {
- tempU32 >>= 1; /* right-shift 1 step */
- numShift--; /* compensate in numerator */
- }
- tempU16 = (WebRtc_UWord16) tempU32;
-
- /* do the shift of numerator */
- tempU32 = WEBRTC_SPL_SHIFT_W32( tempU32_2, numShift);
-
- stats->currentDiscardRate = (WebRtc_UWord16) WebRtcSpl_DivU32U16(tempU32, tempU16);
- }
- }
- else
- {
- /* lost count is larger than elapsed time count; probably timestamp wrap-around or something else wrong */
- /* set loss rate = 1 */
- stats->currentDiscardRate = 1 << 14; /* 1 in Q14 */
- }
-
- /*************************************************************/
- /* Calculate Accelerate, Expand and Pre-emptive Expand rates */
- /*************************************************************/
-
- /* timestamps elapsed since last report */
- tempU32 = NetEqMainInst->MCUinst.lastReportTS;
-
- if (NetEqMainInst->DSPinst.statInst.accelerateLength == 0)
- {
- /* no accelerate */
- stats->currentAccelerateRate = 0;
- }
- else if (NetEqMainInst->DSPinst.statInst.accelerateLength < tempU32)
- {
- /* calculate shifts; we want the result in Q14 */
- numShift = WebRtcSpl_NormU32(NetEqMainInst->DSPinst.statInst.accelerateLength); /* numerator shift for normalize */
-
- if (numShift < 14)
- {
- /* cannot shift numerator 14 steps; shift denominator too */
- tempU32 = WEBRTC_SPL_RSHIFT_U32(tempU32, 14-numShift); /* right-shift */
- }
- else
- {
- /* shift no more than 14 steps */
- numShift = 14;
- }
-
- if (tempU32 == 0)
- {
- /* check for zero denominator; result should be zero in this case */
- stats->currentAccelerateRate = 0;
- }
- else
- {
- /* check that denominator fits in signed 16-bit */
- while (tempU32 > WEBRTC_SPL_WORD16_MAX)
- {
- tempU32 >>= 1; /* right-shift 1 step */
- numShift--; /* compensate in numerator */
- }
- tempU16 = (WebRtc_UWord16) tempU32;
-
- /* do the shift of numerator */
- tempU32
- = WEBRTC_SPL_SHIFT_W32( NetEqMainInst->DSPinst.statInst.accelerateLength, numShift);
-
- stats->currentAccelerateRate = (WebRtc_UWord16) WebRtcSpl_DivU32U16(tempU32,
- tempU16);
- }
- }
- else
- {
- /* lost count is larger than elapsed time count; probably timestamp wrap-around or something else wrong */
- /* set loss rate = 1 */
- stats->currentAccelerateRate = 1 << 14; /* 1 in Q14 */
- }
-
- /* timestamps elapsed since last report */
- tempU32 = NetEqMainInst->MCUinst.lastReportTS;
-
- if (NetEqMainInst->DSPinst.statInst.expandLength == 0)
- {
- /* no expand */
- stats->currentExpandRate = 0;
- }
- else if (NetEqMainInst->DSPinst.statInst.expandLength < tempU32)
- {
- /* calculate shifts; we want the result in Q14 */
- numShift = WebRtcSpl_NormU32(NetEqMainInst->DSPinst.statInst.expandLength); /* numerator shift for normalize */
-
- if (numShift < 14)
- {
- /* cannot shift numerator 14 steps; shift denominator too */
- tempU32 = WEBRTC_SPL_RSHIFT_U32(tempU32, 14-numShift); /* right-shift */
- }
- else
- {
- /* shift no more than 14 steps */
- numShift = 14;
- }
-
- if (tempU32 == 0)
- {
- /* check for zero denominator; result should be zero in this case */
- stats->currentExpandRate = 0;
- }
- else
- {
- /* check that denominator fits in signed 16-bit */
- while (tempU32 > WEBRTC_SPL_WORD16_MAX)
- {
- tempU32 >>= 1; /* right-shift 1 step */
- numShift--; /* compensate in numerator */
- }
- tempU16 = (WebRtc_UWord16) tempU32;
-
- /* do the shift of numerator */
- tempU32
- = WEBRTC_SPL_SHIFT_W32( NetEqMainInst->DSPinst.statInst.expandLength, numShift);
-
- stats->currentExpandRate = (WebRtc_UWord16) WebRtcSpl_DivU32U16(tempU32, tempU16);
- }
- }
- else
- {
- /* lost count is larger than elapsed time count; probably timestamp wrap-around or something else wrong */
- /* set loss rate = 1 */
- stats->currentExpandRate = 1 << 14; /* 1 in Q14 */
- }
-
- /* timestamps elapsed since last report */
- tempU32 = NetEqMainInst->MCUinst.lastReportTS;
-
- if (NetEqMainInst->DSPinst.statInst.preemptiveLength == 0)
- {
- /* no pre-emptive expand */
- stats->currentPreemptiveRate = 0;
- }
- else if (NetEqMainInst->DSPinst.statInst.preemptiveLength < tempU32)
- {
- /* calculate shifts; we want the result in Q14 */
- numShift = WebRtcSpl_NormU32(NetEqMainInst->DSPinst.statInst.preemptiveLength); /* numerator shift for normalize */
-
- if (numShift < 14)
- {
- /* cannot shift numerator 14 steps; shift denominator too */
- tempU32 = WEBRTC_SPL_RSHIFT_U32(tempU32, 14-numShift); /* right-shift */
- }
- else
- {
- /* shift no more than 14 steps */
- numShift = 14;
- }
-
- if (tempU32 == 0)
- {
- /* check for zero denominator; result should be zero in this case */
- stats->currentPreemptiveRate = 0;
- }
- else
- {
- /* check that denominator fits in signed 16-bit */
- while (tempU32 > WEBRTC_SPL_WORD16_MAX)
- {
- tempU32 >>= 1; /* right-shift 1 step */
- numShift--; /* compensate in numerator */
- }
- tempU16 = (WebRtc_UWord16) tempU32;
-
- /* do the shift of numerator */
- tempU32
- = WEBRTC_SPL_SHIFT_W32( NetEqMainInst->DSPinst.statInst.preemptiveLength, numShift);
-
- stats->currentPreemptiveRate = (WebRtc_UWord16) WebRtcSpl_DivU32U16(tempU32,
- tempU16);
- }
- }
- else
- {
- /* lost count is larger than elapsed time count; probably timestamp wrap-around or something else wrong */
- /* set loss rate = 1 */
- stats->currentPreemptiveRate = 1 << 14; /* 1 in Q14 */
- }
-
- stats->clockDriftPPM = WebRtcNetEQ_AverageIAT(
- &NetEqMainInst->MCUinst.BufferStat_inst.Automode_inst);
-
- /* reset counters */
- WebRtcNetEQ_ResetMcuInCallStats(&(NetEqMainInst->MCUinst));
- WebRtcNetEQ_ClearInCallStats(&(NetEqMainInst->DSPinst));
-
- return (0);
-}
-
-int WebRtcNetEQ_GetRawFrameWaitingTimes(void *inst,
- int max_length,
- int* waiting_times_ms) {
- int i = 0;
- MainInst_t *main_inst = (MainInst_t*) inst;
- if (main_inst == NULL) return -1;
-
- while ((i < max_length) && (i < main_inst->MCUinst.len_waiting_times)) {
- waiting_times_ms[i] = main_inst->MCUinst.waiting_times[i] *
- main_inst->DSPinst.millisecondsPerCall;
- ++i;
- }
- assert(i <= kLenWaitingTimes);
- WebRtcNetEQ_ResetWaitingTimeStats(&main_inst->MCUinst);
- return i;
-}
-
-/****************************************************************************
- * WebRtcNetEQ_SetVADInstance(...)
- *
- * Provide a pointer to an allocated VAD instance. If function is never
- * called or it is called with NULL pointer as VAD_inst, the post-decode
- * VAD functionality is disabled. Also provide pointers to init, setmode
- * and VAD functions. These are typically pointers to WebRtcVad_Init,
- * WebRtcVad_set_mode and WebRtcVad_Process, respectively, all found in the
- * interface file webrtc_vad.h.
- *
- * Input:
- * - NetEQ_inst : NetEQ instance
- * - VADinst : VAD instance
- * - initFunction : Pointer to VAD init function
- * - setmodeFunction : Pointer to VAD setmode function
- * - VADfunction : Pointer to VAD function
- *
- * Output:
- * - NetEQ_inst : Updated NetEQ instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_SetVADInstance(void *NetEQ_inst, void *VAD_inst,
- WebRtcNetEQ_VADInitFunction initFunction,
- WebRtcNetEQ_VADSetmodeFunction setmodeFunction,
- WebRtcNetEQ_VADFunction VADFunction)
-{
-
- /* Typecast to internal instance type */
- MainInst_t *NetEqMainInst = (MainInst_t*) NetEQ_inst;
- if (NetEqMainInst == NULL)
- {
- return (-1);
- }
-
-#ifdef NETEQ_VAD
-
- /* Store pointer in PostDecode VAD struct */
- NetEqMainInst->DSPinst.VADInst.VADState = VAD_inst;
-
- /* Store function pointers */
- NetEqMainInst->DSPinst.VADInst.initFunction = initFunction;
- NetEqMainInst->DSPinst.VADInst.setmodeFunction = setmodeFunction;
- NetEqMainInst->DSPinst.VADInst.VADFunction = VADFunction;
-
- /* Call init function and return the result (ok or fail) */
- return(WebRtcNetEQ_InitVAD(&NetEqMainInst->DSPinst.VADInst, NetEqMainInst->DSPinst.fs));
-
-#else /* NETEQ_VAD not defined */
- return (-1);
-#endif /* NETEQ_VAD */
-
-}
-
-/****************************************************************************
- * WebRtcNetEQ_SetVADMode(...)
- *
- * Pass an aggressiveness mode parameter to the post-decode VAD instance.
- * If this function is never called, mode 0 (quality mode) is used as default.
- *
- * Input:
- * - inst : NetEQ instance
- * - mode : mode parameter (same range as WebRtc VAD mode)
- *
- * Output:
- * - inst : Updated NetEQ instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-
-int WebRtcNetEQ_SetVADMode(void *inst, int mode)
-{
-
- /* Typecast to internal instance type */
- MainInst_t *NetEqMainInst = (MainInst_t*) inst;
- if (NetEqMainInst == NULL)
- {
- return (-1);
- }
-
-#ifdef NETEQ_VAD
-
- /* Set mode and return result */
- return(WebRtcNetEQ_SetVADModeInternal(&NetEqMainInst->DSPinst.VADInst, mode));
-
-#else /* NETEQ_VAD not defined */
- return (-1);
-#endif /* NETEQ_VAD */
-
-}
diff --git a/trunk/src/modules/audio_coding/neteq/webrtc_neteq_unittest.cc b/trunk/src/modules/audio_coding/neteq/webrtc_neteq_unittest.cc
deleted file mode 100644
index fbb9cc2..0000000
--- a/trunk/src/modules/audio_coding/neteq/webrtc_neteq_unittest.cc
+++ /dev/null
@@ -1,605 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file includes unit tests for NetEQ.
- */
-
-#include <stdlib.h>
-#include <string.h> // memset
-
-#include <string>
-#include <vector>
-
-#include "gtest/gtest.h"
-
-#include "modules/audio_coding/neteq/interface/webrtc_neteq.h"
-#include "modules/audio_coding/neteq/interface/webrtc_neteq_help_macros.h"
-#include "modules/audio_coding/neteq/interface/webrtc_neteq_internal.h"
-#include "modules/audio_coding/neteq/test/NETEQTEST_CodecClass.h"
-#include "modules/audio_coding/neteq/test/NETEQTEST_NetEQClass.h"
-#include "modules/audio_coding/neteq/test/NETEQTEST_RTPpacket.h"
-#include "testsupport/fileutils.h"
-#include "typedefs.h" // NOLINT(build/include)
-
-namespace webrtc {
-
-class RefFiles {
- public:
- RefFiles(const std::string& input_file, const std::string& output_file);
- ~RefFiles();
- template<class T> void ProcessReference(const T& test_results);
- template<typename T, size_t n> void ProcessReference(
- const T (&test_results)[n],
- size_t length);
- template<typename T, size_t n> void WriteToFile(
- const T (&test_results)[n],
- size_t length);
- template<typename T, size_t n> void ReadFromFileAndCompare(
- const T (&test_results)[n],
- size_t length);
- void WriteToFile(const WebRtcNetEQ_NetworkStatistics& stats);
- void ReadFromFileAndCompare(const WebRtcNetEQ_NetworkStatistics& stats);
- void WriteToFile(const WebRtcNetEQ_RTCPStat& stats);
- void ReadFromFileAndCompare(const WebRtcNetEQ_RTCPStat& stats);
-
- FILE* input_fp_;
- FILE* output_fp_;
-};
-
-RefFiles::RefFiles(const std::string &input_file,
- const std::string &output_file)
- : input_fp_(NULL),
- output_fp_(NULL) {
- if (!input_file.empty()) {
- input_fp_ = fopen(input_file.c_str(), "rb");
- EXPECT_TRUE(input_fp_ != NULL);
- }
- if (!output_file.empty()) {
- output_fp_ = fopen(output_file.c_str(), "wb");
- EXPECT_TRUE(output_fp_ != NULL);
- }
-}
-
-RefFiles::~RefFiles() {
- if (input_fp_) {
- EXPECT_EQ(EOF, fgetc(input_fp_)); // Make sure that we reached the end.
- fclose(input_fp_);
- }
- if (output_fp_) fclose(output_fp_);
-}
-
-template<class T>
-void RefFiles::ProcessReference(const T& test_results) {
- WriteToFile(test_results);
- ReadFromFileAndCompare(test_results);
-}
-
-template<typename T, size_t n>
-void RefFiles::ProcessReference(const T (&test_results)[n], size_t length) {
- WriteToFile(test_results, length);
- ReadFromFileAndCompare(test_results, length);
-}
-
-template<typename T, size_t n>
-void RefFiles::WriteToFile(const T (&test_results)[n], size_t length) {
- if (output_fp_) {
- ASSERT_EQ(length, fwrite(&test_results, sizeof(T), length, output_fp_));
- }
-}
-
-template<typename T, size_t n>
-void RefFiles::ReadFromFileAndCompare(const T (&test_results)[n],
- size_t length) {
- if (input_fp_) {
- // Read from ref file.
- T* ref = new T[length];
- ASSERT_EQ(length, fread(ref, sizeof(T), length, input_fp_));
- // Compare
- EXPECT_EQ(0, memcmp(&test_results, ref, sizeof(T) * length));
- delete [] ref;
- }
-}
-
-void RefFiles::WriteToFile(const WebRtcNetEQ_NetworkStatistics& stats) {
- if (output_fp_) {
- ASSERT_EQ(1u, fwrite(&stats, sizeof(WebRtcNetEQ_NetworkStatistics), 1,
- output_fp_));
- }
-}
-
-void RefFiles::ReadFromFileAndCompare(
- const WebRtcNetEQ_NetworkStatistics& stats) {
- if (input_fp_) {
- // Read from ref file.
- size_t stat_size = sizeof(WebRtcNetEQ_NetworkStatistics);
- WebRtcNetEQ_NetworkStatistics ref_stats;
- ASSERT_EQ(1u, fread(&ref_stats, stat_size, 1, input_fp_));
- // Compare
- EXPECT_EQ(0, memcmp(&stats, &ref_stats, stat_size));
- }
-}
-
-void RefFiles::WriteToFile(const WebRtcNetEQ_RTCPStat& stats) {
- if (output_fp_) {
- ASSERT_EQ(1u, fwrite(&(stats.fraction_lost), sizeof(stats.fraction_lost), 1,
- output_fp_));
- ASSERT_EQ(1u, fwrite(&(stats.cum_lost), sizeof(stats.cum_lost), 1,
- output_fp_));
- ASSERT_EQ(1u, fwrite(&(stats.ext_max), sizeof(stats.ext_max), 1,
- output_fp_));
- ASSERT_EQ(1u, fwrite(&(stats.jitter), sizeof(stats.jitter), 1,
- output_fp_));
- }
-}
-
-void RefFiles::ReadFromFileAndCompare(
- const WebRtcNetEQ_RTCPStat& stats) {
- if (input_fp_) {
- // Read from ref file.
- WebRtcNetEQ_RTCPStat ref_stats;
- ASSERT_EQ(1u, fread(&(ref_stats.fraction_lost),
- sizeof(ref_stats.fraction_lost), 1, input_fp_));
- ASSERT_EQ(1u, fread(&(ref_stats.cum_lost), sizeof(ref_stats.cum_lost), 1,
- input_fp_));
- ASSERT_EQ(1u, fread(&(ref_stats.ext_max), sizeof(ref_stats.ext_max), 1,
- input_fp_));
- ASSERT_EQ(1u, fread(&(ref_stats.jitter), sizeof(ref_stats.jitter), 1,
- input_fp_));
- // Compare
- EXPECT_EQ(ref_stats.fraction_lost, stats.fraction_lost);
- EXPECT_EQ(ref_stats.cum_lost, stats.cum_lost);
- EXPECT_EQ(ref_stats.ext_max, stats.ext_max);
- EXPECT_EQ(ref_stats.jitter, stats.jitter);
- }
-}
-
-class NetEqDecodingTest : public ::testing::Test {
- protected:
- // NetEQ must be polled for data once every 10 ms. Thus, neither of the
- // constants below can be changed.
- static const int kTimeStepMs = 10;
- static const int kBlockSize8kHz = kTimeStepMs * 8;
- static const int kBlockSize16kHz = kTimeStepMs * 16;
- static const int kBlockSize32kHz = kTimeStepMs * 32;
- static const int kMaxBlockSize = kBlockSize32kHz;
-
- NetEqDecodingTest();
- virtual void SetUp();
- virtual void TearDown();
- void SelectDecoders(WebRtcNetEQDecoder* used_codec);
- void LoadDecoders();
- void OpenInputFile(const std::string &rtp_file);
- void Process(NETEQTEST_RTPpacket* rtp_ptr, int16_t* out_len);
- void DecodeAndCompare(const std::string &rtp_file,
- const std::string &ref_file);
- void DecodeAndCheckStats(const std::string &rtp_file,
- const std::string &stat_ref_file,
- const std::string &rtcp_ref_file);
- static void PopulateRtpInfo(int frame_index,
- int timestamp,
- WebRtcNetEQ_RTPInfo* rtp_info);
- static void PopulateCng(int frame_index,
- int timestamp,
- WebRtcNetEQ_RTPInfo* rtp_info,
- uint8_t* payload,
- int* payload_len);
-
- NETEQTEST_NetEQClass* neteq_inst_;
- std::vector<NETEQTEST_Decoder*> dec_;
- FILE* rtp_fp_;
- unsigned int sim_clock_;
- int16_t out_data_[kMaxBlockSize];
-};
-
-NetEqDecodingTest::NetEqDecodingTest()
- : neteq_inst_(NULL),
- rtp_fp_(NULL),
- sim_clock_(0) {
- memset(out_data_, 0, sizeof(out_data_));
-}
-
-void NetEqDecodingTest::SetUp() {
- WebRtcNetEQDecoder usedCodec[kDecoderReservedEnd - 1];
-
- SelectDecoders(usedCodec);
- neteq_inst_ = new NETEQTEST_NetEQClass(usedCodec, dec_.size(), 8000,
- kTCPLargeJitter);
- ASSERT_TRUE(neteq_inst_);
- LoadDecoders();
-}
-
-void NetEqDecodingTest::TearDown() {
- if (neteq_inst_)
- delete neteq_inst_;
- for (size_t i = 0; i < dec_.size(); ++i) {
- if (dec_[i])
- delete dec_[i];
- }
- if (rtp_fp_)
- fclose(rtp_fp_);
-}
-
-void NetEqDecodingTest::SelectDecoders(WebRtcNetEQDecoder* used_codec) {
- *used_codec++ = kDecoderPCMu;
- dec_.push_back(new decoder_PCMU(0));
- *used_codec++ = kDecoderPCMa;
- dec_.push_back(new decoder_PCMA(8));
- *used_codec++ = kDecoderILBC;
- dec_.push_back(new decoder_ILBC(102));
- *used_codec++ = kDecoderISAC;
- dec_.push_back(new decoder_iSAC(103));
- *used_codec++ = kDecoderISACswb;
- dec_.push_back(new decoder_iSACSWB(104));
- *used_codec++ = kDecoderPCM16B;
- dec_.push_back(new decoder_PCM16B_NB(93));
- *used_codec++ = kDecoderPCM16Bwb;
- dec_.push_back(new decoder_PCM16B_WB(94));
- *used_codec++ = kDecoderPCM16Bswb32kHz;
- dec_.push_back(new decoder_PCM16B_SWB32(95));
- *used_codec++ = kDecoderCNG;
- dec_.push_back(new decoder_CNG(13, 8000));
- *used_codec++ = kDecoderCNG;
- dec_.push_back(new decoder_CNG(98, 16000));
-}
-
-void NetEqDecodingTest::LoadDecoders() {
- for (size_t i = 0; i < dec_.size(); ++i) {
- ASSERT_EQ(0, dec_[i]->loadToNetEQ(*neteq_inst_));
- }
-}
-
-void NetEqDecodingTest::OpenInputFile(const std::string &rtp_file) {
- rtp_fp_ = fopen(rtp_file.c_str(), "rb");
- ASSERT_TRUE(rtp_fp_ != NULL);
- ASSERT_EQ(0, NETEQTEST_RTPpacket::skipFileHeader(rtp_fp_));
-}
-
-void NetEqDecodingTest::Process(NETEQTEST_RTPpacket* rtp, int16_t* out_len) {
- // Check if time to receive.
- while ((sim_clock_ >= rtp->time()) &&
- (rtp->dataLen() >= 0)) {
- if (rtp->dataLen() > 0) {
- ASSERT_EQ(0, neteq_inst_->recIn(*rtp));
- }
- // Get next packet.
- ASSERT_NE(-1, rtp->readFromFile(rtp_fp_));
- }
-
- // RecOut
- *out_len = neteq_inst_->recOut(out_data_);
- ASSERT_TRUE((*out_len == kBlockSize8kHz) ||
- (*out_len == kBlockSize16kHz) ||
- (*out_len == kBlockSize32kHz));
-
- // Increase time.
- sim_clock_ += kTimeStepMs;
-}
-
-void NetEqDecodingTest::DecodeAndCompare(const std::string &rtp_file,
- const std::string &ref_file) {
- OpenInputFile(rtp_file);
-
- std::string ref_out_file = "";
- if (ref_file.empty()) {
- ref_out_file = webrtc::test::OutputPath() + "neteq_out.pcm";
- }
- RefFiles ref_files(ref_file, ref_out_file);
-
- NETEQTEST_RTPpacket rtp;
- ASSERT_GT(rtp.readFromFile(rtp_fp_), 0);
- while (rtp.dataLen() >= 0) {
- int16_t out_len;
- Process(&rtp, &out_len);
- ref_files.ProcessReference(out_data_, out_len);
- }
-}
-
-void NetEqDecodingTest::DecodeAndCheckStats(const std::string &rtp_file,
- const std::string &stat_ref_file,
- const std::string &rtcp_ref_file) {
- OpenInputFile(rtp_file);
- std::string stat_out_file = "";
- if (stat_ref_file.empty()) {
- stat_out_file = webrtc::test::OutputPath() +
- "neteq_network_stats.dat";
- }
- RefFiles network_stat_files(stat_ref_file, stat_out_file);
-
- std::string rtcp_out_file = "";
- if (rtcp_ref_file.empty()) {
- rtcp_out_file = webrtc::test::OutputPath() +
- "neteq_rtcp_stats.dat";
- }
- RefFiles rtcp_stat_files(rtcp_ref_file, rtcp_out_file);
-
- NETEQTEST_RTPpacket rtp;
- ASSERT_GT(rtp.readFromFile(rtp_fp_), 0);
- while (rtp.dataLen() >= 0) {
- int16_t out_len;
- Process(&rtp, &out_len);
-
- // Query the network statistics API once per second
- if (sim_clock_ % 1000 == 0) {
- // Process NetworkStatistics.
- WebRtcNetEQ_NetworkStatistics network_stats;
- ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(),
- &network_stats));
- network_stat_files.ProcessReference(network_stats);
-
- // Process RTCPstat.
- WebRtcNetEQ_RTCPStat rtcp_stats;
- ASSERT_EQ(0, WebRtcNetEQ_GetRTCPStats(neteq_inst_->instance(),
- &rtcp_stats));
- rtcp_stat_files.ProcessReference(rtcp_stats);
- }
- }
-}
-
-void NetEqDecodingTest::PopulateRtpInfo(int frame_index,
- int timestamp,
- WebRtcNetEQ_RTPInfo* rtp_info) {
- rtp_info->sequenceNumber = frame_index;
- rtp_info->timeStamp = timestamp;
- rtp_info->SSRC = 0x1234; // Just an arbitrary SSRC.
- rtp_info->payloadType = 94; // PCM16b WB codec.
- rtp_info->markerBit = 0;
-}
-
-void NetEqDecodingTest::PopulateCng(int frame_index,
- int timestamp,
- WebRtcNetEQ_RTPInfo* rtp_info,
- uint8_t* payload,
- int* payload_len) {
- rtp_info->sequenceNumber = frame_index;
- rtp_info->timeStamp = timestamp;
- rtp_info->SSRC = 0x1234; // Just an arbitrary SSRC.
- rtp_info->payloadType = 98; // WB CNG.
- rtp_info->markerBit = 0;
- payload[0] = 64; // Noise level -64 dBov, quite arbitrarily chosen.
- *payload_len = 1; // Only noise level, no spectral parameters.
-}
-
-TEST_F(NetEqDecodingTest, TestBitExactness) {
- const std::string kInputRtpFile = webrtc::test::ProjectRootPath() +
- "resources/neteq_universal.rtp";
- const std::string kInputRefFile =
- webrtc::test::ResourcePath("neteq_universal_ref", "pcm");
- DecodeAndCompare(kInputRtpFile, kInputRefFile);
-}
-
-TEST_F(NetEqDecodingTest, TestNetworkStatistics) {
- const std::string kInputRtpFile = webrtc::test::ProjectRootPath() +
- "resources/neteq_universal.rtp";
- const std::string kNetworkStatRefFile =
- webrtc::test::ResourcePath("neteq_network_stats", "dat");
- const std::string kRtcpStatRefFile =
- webrtc::test::ResourcePath("neteq_rtcp_stats", "dat");
- DecodeAndCheckStats(kInputRtpFile, kNetworkStatRefFile, kRtcpStatRefFile);
-}
-
-TEST_F(NetEqDecodingTest, TestFrameWaitingTimeStatistics) {
- // Use fax mode to avoid time-scaling. This is to simplify the testing of
- // packet waiting times in the packet buffer.
- ASSERT_EQ(0,
- WebRtcNetEQ_SetPlayoutMode(neteq_inst_->instance(), kPlayoutFax));
- // Insert 30 dummy packets at once. Each packet contains 10 ms 16 kHz audio.
- int num_frames = 30;
- const int kSamples = 10 * 16;
- const int kPayloadBytes = kSamples * 2;
- for (int i = 0; i < num_frames; ++i) {
- uint16_t payload[kSamples] = {0};
- WebRtcNetEQ_RTPInfo rtp_info;
- rtp_info.sequenceNumber = i;
- rtp_info.timeStamp = i * kSamples;
- rtp_info.SSRC = 0x1234; // Just an arbitrary SSRC.
- rtp_info.payloadType = 94; // PCM16b WB codec.
- rtp_info.markerBit = 0;
- ASSERT_EQ(0, WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), &rtp_info,
- reinterpret_cast<uint8_t*>(payload),
- kPayloadBytes, 0));
- }
- // Pull out all data.
- for (int i = 0; i < num_frames; ++i) {
- ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
- }
- const int kVecLen = 110; // More than kLenWaitingTimes in mcu.h.
- int waiting_times[kVecLen];
- int len = WebRtcNetEQ_GetRawFrameWaitingTimes(neteq_inst_->instance(),
- kVecLen, waiting_times);
- EXPECT_EQ(num_frames, len);
- // Since all frames are dumped into NetEQ at once, but pulled out with 10 ms
- // spacing (per definition), we expect the delay to increase with 10 ms for
- // each packet.
- for (int i = 0; i < len; ++i) {
- EXPECT_EQ((i + 1) * 10, waiting_times[i]);
- }
-
- // Check statistics again and make sure it's been reset.
- EXPECT_EQ(0, WebRtcNetEQ_GetRawFrameWaitingTimes(neteq_inst_->instance(),
- kVecLen, waiting_times));
-
- // Process > 100 frames, and make sure that that we get statistics
- // only for 100 frames. Note the new SSRC, causing NetEQ to reset.
- num_frames = 110;
- for (int i = 0; i < num_frames; ++i) {
- uint16_t payload[kSamples] = {0};
- WebRtcNetEQ_RTPInfo rtp_info;
- rtp_info.sequenceNumber = i;
- rtp_info.timeStamp = i * kSamples;
- rtp_info.SSRC = 0x1235; // Just an arbitrary SSRC.
- rtp_info.payloadType = 94; // PCM16b WB codec.
- rtp_info.markerBit = 0;
- ASSERT_EQ(0, WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(), &rtp_info,
- reinterpret_cast<uint8_t*>(payload),
- kPayloadBytes, 0));
- ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
- }
-
- len = WebRtcNetEQ_GetRawFrameWaitingTimes(neteq_inst_->instance(),
- kVecLen, waiting_times);
- EXPECT_EQ(100, len);
-}
-
-TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimeNegative) {
- const int kNumFrames = 3000; // Needed for convergence.
- int frame_index = 0;
- const int kSamples = 10 * 16;
- const int kPayloadBytes = kSamples * 2;
- while (frame_index < kNumFrames) {
- // Insert one packet each time, except every 10th time where we insert two
- // packets at once. This will create a negative clock-drift of approx. 10%.
- int num_packets = (frame_index % 10 == 0 ? 2 : 1);
- for (int n = 0; n < num_packets; ++n) {
- uint8_t payload[kPayloadBytes] = {0};
- WebRtcNetEQ_RTPInfo rtp_info;
- PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
- ASSERT_EQ(0,
- WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(),
- &rtp_info,
- payload,
- kPayloadBytes, 0));
- ++frame_index;
- }
-
- // Pull out data once.
- ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
- }
-
- WebRtcNetEQ_NetworkStatistics network_stats;
- ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(),
- &network_stats));
- EXPECT_EQ(-106911, network_stats.clockDriftPPM);
-}
-
-TEST_F(NetEqDecodingTest, TestAverageInterArrivalTimePositive) {
- const int kNumFrames = 5000; // Needed for convergence.
- int frame_index = 0;
- const int kSamples = 10 * 16;
- const int kPayloadBytes = kSamples * 2;
- for (int i = 0; i < kNumFrames; ++i) {
- // Insert one packet each time, except every 10th time where we don't insert
- // any packet. This will create a positive clock-drift of approx. 11%.
- int num_packets = (i % 10 == 9 ? 0 : 1);
- for (int n = 0; n < num_packets; ++n) {
- uint8_t payload[kPayloadBytes] = {0};
- WebRtcNetEQ_RTPInfo rtp_info;
- PopulateRtpInfo(frame_index, frame_index * kSamples, &rtp_info);
- ASSERT_EQ(0,
- WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(),
- &rtp_info,
- payload,
- kPayloadBytes, 0));
- ++frame_index;
- }
-
- // Pull out data once.
- ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
- }
-
- WebRtcNetEQ_NetworkStatistics network_stats;
- ASSERT_EQ(0, WebRtcNetEQ_GetNetworkStatistics(neteq_inst_->instance(),
- &network_stats));
- EXPECT_EQ(108352, network_stats.clockDriftPPM);
-}
-
-TEST_F(NetEqDecodingTest, LongCngWithClockDrift) {
- uint16_t seq_no = 0;
- uint32_t timestamp = 0;
- const int kFrameSizeMs = 30;
- const int kSamples = kFrameSizeMs * 16;
- const int kPayloadBytes = kSamples * 2;
- // Apply a clock drift of -25 ms / s (sender faster than receiver).
- const double kDriftFactor = 1000.0 / (1000.0 + 25.0);
- double next_input_time_ms = 0.0;
- double t_ms;
-
- // Insert speech for 5 seconds.
- const int kSpeechDurationMs = 5000;
- for (t_ms = 0; t_ms < kSpeechDurationMs; t_ms += 10) {
- // Each turn in this for loop is 10 ms.
- while (next_input_time_ms <= t_ms) {
- // Insert one 30 ms speech frame.
- uint8_t payload[kPayloadBytes] = {0};
- WebRtcNetEQ_RTPInfo rtp_info;
- PopulateRtpInfo(seq_no, timestamp, &rtp_info);
- ASSERT_EQ(0,
- WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(),
- &rtp_info,
- payload,
- kPayloadBytes, 0));
- ++seq_no;
- timestamp += kSamples;
- next_input_time_ms += static_cast<double>(kFrameSizeMs) * kDriftFactor;
- }
- // Pull out data once.
- ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
- }
-
- EXPECT_EQ(kOutputNormal, neteq_inst_->getOutputType());
- int32_t delay_before = timestamp - neteq_inst_->getSpeechTimeStamp();
-
- // Insert CNG for 1 minute (= 60000 ms).
- const int kCngPeriodMs = 100;
- const int kCngPeriodSamples = kCngPeriodMs * 16; // Period in 16 kHz samples.
- const int kCngDurationMs = 60000;
- for (; t_ms < kSpeechDurationMs + kCngDurationMs; t_ms += 10) {
- // Each turn in this for loop is 10 ms.
- while (next_input_time_ms <= t_ms) {
- // Insert one CNG frame each 100 ms.
- uint8_t payload[kPayloadBytes];
- int payload_len;
- WebRtcNetEQ_RTPInfo rtp_info;
- PopulateCng(seq_no, timestamp, &rtp_info, payload, &payload_len);
- ASSERT_EQ(0,
- WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(),
- &rtp_info,
- payload,
- payload_len, 0));
- ++seq_no;
- timestamp += kCngPeriodSamples;
- next_input_time_ms += static_cast<double>(kCngPeriodMs) * kDriftFactor;
- }
- // Pull out data once.
- ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
- }
-
- EXPECT_EQ(kOutputCNG, neteq_inst_->getOutputType());
-
- // Insert speech again until output type is speech.
- while (neteq_inst_->getOutputType() != kOutputNormal) {
- // Each turn in this for loop is 10 ms.
- while (next_input_time_ms <= t_ms) {
- // Insert one 30 ms speech frame.
- uint8_t payload[kPayloadBytes] = {0};
- WebRtcNetEQ_RTPInfo rtp_info;
- PopulateRtpInfo(seq_no, timestamp, &rtp_info);
- ASSERT_EQ(0,
- WebRtcNetEQ_RecInRTPStruct(neteq_inst_->instance(),
- &rtp_info,
- payload,
- kPayloadBytes, 0));
- ++seq_no;
- timestamp += kSamples;
- next_input_time_ms += static_cast<double>(kFrameSizeMs) * kDriftFactor;
- }
- // Pull out data once.
- ASSERT_TRUE(kBlockSize16kHz == neteq_inst_->recOut(out_data_));
- // Increase clock.
- t_ms += 10;
- }
-
- int32_t delay_after = timestamp - neteq_inst_->getSpeechTimeStamp();
- // Compare delay before and after, and make sure it differs less than 20 ms.
- EXPECT_LE(delay_after, delay_before + 20 * 16);
- EXPECT_GE(delay_after, delay_before - 20 * 16);
-}
-
-} // namespace
diff --git a/trunk/src/modules/audio_conference_mixer/OWNERS b/trunk/src/modules/audio_conference_mixer/OWNERS
deleted file mode 100644
index 7dc791e..0000000
--- a/trunk/src/modules/audio_conference_mixer/OWNERS
+++ /dev/null
@@ -1,3 +0,0 @@
-henrike@webrtc.org
-pwestin@webrtc.org
-andrew@webrtc.org
\ No newline at end of file
diff --git a/trunk/src/modules/audio_conference_mixer/interface/audio_conference_mixer.h b/trunk/src/modules/audio_conference_mixer/interface/audio_conference_mixer.h
deleted file mode 100644
index 4ece1bf..0000000
--- a/trunk/src/modules/audio_conference_mixer/interface/audio_conference_mixer.h
+++ /dev/null
@@ -1,86 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
-
-#include "audio_conference_mixer_defines.h"
-#include "module.h"
-#include "module_common_types.h"
-
-namespace webrtc {
-class AudioMixerOutputReceiver;
-class AudioMixerStatusReceiver;
-class MixerParticipant;
-class Trace;
-
-class AudioConferenceMixer : public Module
-{
-public:
- enum {kMaximumAmountOfMixedParticipants = 3};
- enum Frequency
- {
- kNbInHz = 8000,
- kWbInHz = 16000,
- kSwbInHz = 32000,
- kLowestPossible = -1,
- kDefaultFrequency = kWbInHz
- };
-
- // Factory method. Constructor disabled.
- static AudioConferenceMixer* Create(int id);
- virtual ~AudioConferenceMixer() {}
-
- // Module functions
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0;
- virtual WebRtc_Word32 TimeUntilNextProcess() = 0 ;
- virtual WebRtc_Word32 Process() = 0;
-
- // Register/unregister a callback class for receiving the mixed audio.
- virtual WebRtc_Word32 RegisterMixedStreamCallback(
- AudioMixerOutputReceiver& receiver) = 0;
- virtual WebRtc_Word32 UnRegisterMixedStreamCallback() = 0;
-
- // Register/unregister a callback class for receiving status information.
- virtual WebRtc_Word32 RegisterMixerStatusCallback(
- AudioMixerStatusReceiver& mixerStatusCallback,
- const WebRtc_UWord32 amountOf10MsBetweenCallbacks) = 0;
- virtual WebRtc_Word32 UnRegisterMixerStatusCallback() = 0;
-
- // Add/remove participants as candidates for mixing.
- virtual WebRtc_Word32 SetMixabilityStatus(
- MixerParticipant& participant,
- const bool mixable) = 0;
- // mixable is set to true if a participant is a candidate for mixing.
- virtual WebRtc_Word32 MixabilityStatus(
- MixerParticipant& participant,
- bool& mixable) = 0;
-
- // Inform the mixer that the participant should always be mixed and not
- // count toward the number of mixed participants. Note that a participant
- // must have been added to the mixer (by calling SetMixabilityStatus())
- // before this function can be successfully called.
- virtual WebRtc_Word32 SetAnonymousMixabilityStatus(
- MixerParticipant& participant, const bool mixable) = 0;
- // mixable is set to true if the participant is mixed anonymously.
- virtual WebRtc_Word32 AnonymousMixabilityStatus(
- MixerParticipant& participant, bool& mixable) = 0;
-
- // Set the minimum sampling frequency at which to mix. The mixing algorithm
- // may still choose to mix at a higher samling frequency to avoid
- // downsampling of audio contributing to the mixed audio.
- virtual WebRtc_Word32 SetMinimumMixingFrequency(Frequency freq) = 0;
-
-protected:
- AudioConferenceMixer() {}
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_H_
diff --git a/trunk/src/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h b/trunk/src/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h
deleted file mode 100644
index 718470d..0000000
--- a/trunk/src/modules/audio_conference_mixer/interface/audio_conference_mixer_defines.h
+++ /dev/null
@@ -1,106 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
-
-#include "map_wrapper.h"
-#include "module_common_types.h"
-#include "typedefs.h"
-
-namespace webrtc {
-class MixHistory;
-
-// A callback class that all mixer participants must inherit from/implement.
-class MixerParticipant
-{
-public:
- // The implementation of this function should update audioFrame with new
- // audio every time it's called.
- //
- // If it returns -1, the frame will not be added to the mix.
- virtual WebRtc_Word32 GetAudioFrame(const WebRtc_Word32 id,
- AudioFrame& audioFrame) = 0;
-
- // mixed will be set to true if the participant was mixed this mix iteration
- WebRtc_Word32 IsMixed(bool& mixed) const;
-
- // This function specifies the sampling frequency needed for the AudioFrame
- // for future GetAudioFrame(..) calls.
- virtual WebRtc_Word32 NeededFrequency(const WebRtc_Word32 id) = 0;
-
- MixHistory* _mixHistory;
-protected:
- MixerParticipant();
- virtual ~MixerParticipant();
-};
-
-// Container struct for participant statistics.
-struct ParticipantStatistics
-{
- WebRtc_Word32 participant;
- WebRtc_Word32 level;
-};
-
-class AudioMixerStatusReceiver
-{
-public:
- // Callback function that provides an array of ParticipantStatistics for the
- // participants that were mixed last mix iteration.
- virtual void MixedParticipants(
- const WebRtc_Word32 id,
- const ParticipantStatistics* participantStatistics,
- const WebRtc_UWord32 size) = 0;
- // Callback function that provides an array of the ParticipantStatistics for
- // the participants that had a positiv VAD last mix iteration.
- virtual void VADPositiveParticipants(
- const WebRtc_Word32 id,
- const ParticipantStatistics* participantStatistics,
- const WebRtc_UWord32 size) = 0;
- // Callback function that provides the audio level of the mixed audio frame
- // from the last mix iteration.
- virtual void MixedAudioLevel(
- const WebRtc_Word32 id,
- const WebRtc_UWord32 level) = 0;
-protected:
- AudioMixerStatusReceiver() {}
- virtual ~AudioMixerStatusReceiver() {}
-};
-
-class AudioMixerOutputReceiver
-{
-public:
- // This callback function provides the mixed audio for this mix iteration.
- // Note that uniqueAudioFrames is an array of AudioFrame pointers with the
- // size according to the size parameter.
- virtual void NewMixedAudio(const WebRtc_Word32 id,
- const AudioFrame& generalAudioFrame,
- const AudioFrame** uniqueAudioFrames,
- const WebRtc_UWord32 size) = 0;
-protected:
- AudioMixerOutputReceiver() {}
- virtual ~AudioMixerOutputReceiver() {}
-};
-
-class AudioRelayReceiver
-{
-public:
- // This callback function provides the mix decision for this mix iteration.
- // mixerList is a list of elements of the type
- // [int,MixerParticipant*]
- virtual void NewAudioToRelay(const WebRtc_Word32 id,
- const MapWrapper& mixerList) = 0;
-protected:
- AudioRelayReceiver() {}
- virtual ~AudioRelayReceiver() {}
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_INTERFACE_AUDIO_CONFERENCE_MIXER_DEFINES_H_
diff --git a/trunk/src/modules/audio_conference_mixer/source/Android.mk b/trunk/src/modules/audio_conference_mixer/source/Android.mk
deleted file mode 100644
index a90ff1d..0000000
--- a/trunk/src/modules/audio_conference_mixer/source/Android.mk
+++ /dev/null
@@ -1,44 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../android-webrtc.mk
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_audio_conference_mixer
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := \
- audio_frame_manipulator.cc \
- level_indicator.cc \
- audio_conference_mixer_impl.cc \
- time_scheduler.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../interface \
- $(LOCAL_PATH)/../../audio_processing/include \
- $(LOCAL_PATH)/../../.. \
- $(LOCAL_PATH)/../../../system_wrappers/interface
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer.gypi b/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer.gypi
deleted file mode 100644
index fd116c6..0000000
--- a/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer.gypi
+++ /dev/null
@@ -1,70 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'audio_conference_mixer',
- 'type': '<(library)',
- 'dependencies': [
- 'audio_processing',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '../interface',
- '../../interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- '../../interface',
- ],
- },
- 'sources': [
- '../interface/audio_conference_mixer.h',
- '../interface/audio_conference_mixer_defines.h',
- 'audio_frame_manipulator.cc',
- 'audio_frame_manipulator.h',
- 'level_indicator.cc',
- 'level_indicator.h',
- 'memory_pool.h',
- 'memory_pool_posix.h',
- 'memory_pool_win.h',
- 'audio_conference_mixer_impl.cc',
- 'audio_conference_mixer_impl.h',
- 'time_scheduler.cc',
- 'time_scheduler.h',
- ],
- },
- ], # targets
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'audio_conference_mixer_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'audio_conference_mixer',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'sources': [
- 'audio_conference_mixer_unittest.cc',
- ],
- }, # audio_conference_mixer_unittests
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc b/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
deleted file mode 100644
index 8da0ecd..0000000
--- a/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.cc
+++ /dev/null
@@ -1,1187 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_conference_mixer_defines.h"
-#include "audio_conference_mixer_impl.h"
-#include "audio_frame_manipulator.h"
-#include "audio_processing.h"
-#include "critical_section_wrapper.h"
-#include "map_wrapper.h"
-#include "trace.h"
-
-namespace webrtc {
-namespace {
-void SetParticipantStatistics(ParticipantStatistics* stats,
- const AudioFrame& frame)
-{
- stats->participant = frame._id;
- stats->level = frame._volume;
-}
-} // namespace
-
-MixerParticipant::MixerParticipant()
- : _mixHistory(new MixHistory())
-{
-}
-
-MixerParticipant::~MixerParticipant()
-{
- delete _mixHistory;
-}
-
-WebRtc_Word32 MixerParticipant::IsMixed(bool& mixed) const
-{
- return _mixHistory->IsMixed(mixed);
-}
-
-MixHistory::MixHistory()
- : _isMixed(0)
-{
-}
-
-MixHistory::~MixHistory()
-{
-}
-
-WebRtc_Word32 MixHistory::IsMixed(bool& mixed) const
-{
- mixed = (_isMixed.Value() == 1);
- return 0;
-}
-
-WebRtc_Word32 MixHistory::WasMixed(bool& wasMixed) const
-{
- // Was mixed is the same as is mixed depending on perspective. This function
- // is for the perspective of AudioConferenceMixerImpl.
- return IsMixed(wasMixed);
-}
-
-WebRtc_Word32 MixHistory::SetIsMixed(const bool mixed)
-{
- _isMixed = mixed ? 1 : 0;
- return 0;
-}
-
-void MixHistory::ResetMixedStatus()
-{
- _isMixed = 0;
-}
-
-AudioConferenceMixer* AudioConferenceMixer::Create(int id)
-{
- AudioConferenceMixerImpl* mixer = new AudioConferenceMixerImpl(id);
- if(!mixer->Init())
- {
- delete mixer;
- return NULL;
- }
- return mixer;
-}
-
-AudioConferenceMixerImpl::AudioConferenceMixerImpl(int id)
- : _scratchParticipantsToMixAmount(0),
- _scratchMixedParticipants(),
- _scratchVadPositiveParticipantsAmount(0),
- _scratchVadPositiveParticipants(),
- _crit(NULL),
- _cbCrit(NULL),
- _id(id),
- _minimumMixingFreq(kLowestPossible),
- _mixReceiver(NULL),
- _mixerStatusCallback(NULL),
- _amountOf10MsBetweenCallbacks(1),
- _amountOf10MsUntilNextCallback(0),
- _mixerStatusCb(false),
- _outputFrequency(kDefaultFrequency),
- _sampleSize(0),
- _audioFramePool(NULL),
- _participantList(),
- _additionalParticipantList(),
- _numMixedParticipants(0),
- _timeStamp(0),
- _timeScheduler(kProcessPeriodicityInMs),
- _mixedAudioLevel(),
- _processCalls(0),
- _limiter(NULL)
-{}
-
-bool AudioConferenceMixerImpl::Init()
-{
- _crit.reset(CriticalSectionWrapper::CreateCriticalSection());
- if (_crit.get() == NULL)
- return false;
-
- _cbCrit.reset(CriticalSectionWrapper::CreateCriticalSection());
- if(_cbCrit.get() == NULL)
- return false;
-
- _limiter.reset(AudioProcessing::Create(_id));
- if(_limiter.get() == NULL)
- return false;
-
- MemoryPool<AudioFrame>::CreateMemoryPool(_audioFramePool,
- DEFAULT_AUDIO_FRAME_POOLSIZE);
- if(_audioFramePool == NULL)
- return false;
-
- if(SetOutputFrequency(kDefaultFrequency) == -1)
- return false;
-
- // Assume mono.
- if (!SetNumLimiterChannels(1))
- return false;
-
- if(_limiter->gain_control()->set_mode(GainControl::kFixedDigital) !=
- _limiter->kNoError)
- return false;
-
- // We smoothly limit the mixed frame to -7 dbFS. -6 would correspond to the
- // divide-by-2 but -7 is used instead to give a bit of headroom since the
- // AGC is not a hard limiter.
- if(_limiter->gain_control()->set_target_level_dbfs(7) != _limiter->kNoError)
- return false;
-
- if(_limiter->gain_control()->set_compression_gain_db(0)
- != _limiter->kNoError)
- return false;
-
- if(_limiter->gain_control()->enable_limiter(true) != _limiter->kNoError)
- return false;
-
- if(_limiter->gain_control()->Enable(true) != _limiter->kNoError)
- return false;
-
- return true;
-}
-
-AudioConferenceMixerImpl::~AudioConferenceMixerImpl()
-{
- MemoryPool<AudioFrame>::DeleteMemoryPool(_audioFramePool);
- assert(_audioFramePool == NULL);
-}
-
-WebRtc_Word32 AudioConferenceMixerImpl::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
- return 0;
-}
-
-// Process should be called every kProcessPeriodicityInMs ms
-WebRtc_Word32 AudioConferenceMixerImpl::TimeUntilNextProcess()
-{
- WebRtc_Word32 timeUntilNextProcess = 0;
- CriticalSectionScoped cs(_crit.get());
- if(_timeScheduler.TimeToNextUpdate(timeUntilNextProcess) != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
- "failed in TimeToNextUpdate() call");
- // Sanity check
- assert(false);
- return -1;
- }
- return timeUntilNextProcess;
-}
-
-WebRtc_Word32 AudioConferenceMixerImpl::Process()
-{
- WebRtc_UWord32 remainingParticipantsAllowedToMix =
- kMaximumAmountOfMixedParticipants;
- {
- CriticalSectionScoped cs(_crit.get());
- assert(_processCalls == 0);
- _processCalls++;
-
- // Let the scheduler know that we are running one iteration.
- _timeScheduler.UpdateScheduler();
- }
-
- ListWrapper mixList;
- ListWrapper rampOutList;
- ListWrapper additionalFramesList;
- MapWrapper mixedParticipantsMap;
- {
- CriticalSectionScoped cs(_cbCrit.get());
-
- WebRtc_Word32 lowFreq = GetLowestMixingFrequency();
- // SILK can run in 12 kHz and 24 kHz. These frequencies are not
- // supported so use the closest higher frequency to not lose any
- // information.
- // TODO(henrike): this is probably more appropriate to do in
- // GetLowestMixingFrequency().
- if (lowFreq == 12000)
- {
- lowFreq = 16000;
- } else if (lowFreq == 24000) {
- lowFreq = 32000;
- }
- if(lowFreq <= 0)
- {
- CriticalSectionScoped cs(_crit.get());
- _processCalls--;
- return 0;
- } else {
- switch(lowFreq)
- {
- case 8000:
- if(OutputFrequency() != kNbInHz)
- {
- SetOutputFrequency(kNbInHz);
- }
- break;
- case 16000:
- if(OutputFrequency() != kWbInHz)
- {
- SetOutputFrequency(kWbInHz);
- }
- break;
- case 32000:
- if(OutputFrequency() != kSwbInHz)
- {
- SetOutputFrequency(kSwbInHz);
- }
- break;
- default:
- assert(false);
-
- CriticalSectionScoped cs(_crit.get());
- _processCalls--;
- return -1;
- }
- }
-
- UpdateToMix(mixList, rampOutList, mixedParticipantsMap,
- remainingParticipantsAllowedToMix);
-
- GetAdditionalAudio(additionalFramesList);
- UpdateMixedStatus(mixedParticipantsMap);
- _scratchParticipantsToMixAmount = mixedParticipantsMap.Size();
- }
-
- // Clear mixedParticipantsMap to avoid memory leak warning.
- // Please note that the mixedParticipantsMap doesn't own any dynamically
- // allocated memory.
- while(mixedParticipantsMap.Erase(mixedParticipantsMap.First()) == 0) {}
-
- // Get an AudioFrame for mixing from the memory pool.
- AudioFrame* mixedAudio = NULL;
- if(_audioFramePool->PopMemory(mixedAudio) == -1)
- {
- WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id,
- "failed PopMemory() call");
- assert(false);
- return -1;
- }
-
- bool timeForMixerCallback = false;
- int retval = 0;
- WebRtc_Word32 audioLevel = 0;
- {
- const ListItem* firstItem = mixList.First();
- // Assume mono.
- WebRtc_UWord8 numberOfChannels = 1;
- if(firstItem != NULL)
- {
- // Use the same number of channels as the first frame to be mixed.
- numberOfChannels = static_cast<const AudioFrame*>(
- firstItem->GetItem())->_audioChannel;
- }
- // TODO(henrike): it might be better to decide the number of channels
- // with an API instead of dynamically.
-
- CriticalSectionScoped cs(_crit.get());
- if (!SetNumLimiterChannels(numberOfChannels))
- retval = -1;
-
- mixedAudio->UpdateFrame(-1, _timeStamp, NULL, 0, _outputFrequency,
- AudioFrame::kNormalSpeech,
- AudioFrame::kVadPassive, numberOfChannels);
-
- _timeStamp += _sampleSize;
-
- MixFromList(*mixedAudio, mixList);
- MixAnonomouslyFromList(*mixedAudio, additionalFramesList);
- MixAnonomouslyFromList(*mixedAudio, rampOutList);
-
- if(mixedAudio->_payloadDataLengthInSamples == 0)
- {
- // Nothing was mixed, set the audio samples to silence.
- memset(mixedAudio->_payloadData, 0, _sampleSize);
- mixedAudio->_payloadDataLengthInSamples = _sampleSize;
- }
- else
- {
- // Only call the limiter if we have something to mix.
- if(!LimitMixedAudio(*mixedAudio))
- retval = -1;
- }
-
- _mixedAudioLevel.ComputeLevel(mixedAudio->_payloadData,_sampleSize);
- audioLevel = _mixedAudioLevel.GetLevel();
-
- if(_mixerStatusCb)
- {
- _scratchVadPositiveParticipantsAmount = 0;
- UpdateVADPositiveParticipants(mixList);
- if(_amountOf10MsUntilNextCallback-- == 0)
- {
- _amountOf10MsUntilNextCallback = _amountOf10MsBetweenCallbacks;
- timeForMixerCallback = true;
- }
- }
- }
-
- {
- CriticalSectionScoped cs(_cbCrit.get());
- if(_mixReceiver != NULL)
- {
- const AudioFrame** dummy = NULL;
- _mixReceiver->NewMixedAudio(
- _id,
- *mixedAudio,
- dummy,
- 0);
- }
-
- if((_mixerStatusCallback != NULL) &&
- timeForMixerCallback)
- {
- _mixerStatusCallback->MixedParticipants(
- _id,
- _scratchMixedParticipants,
- _scratchParticipantsToMixAmount);
-
- _mixerStatusCallback->VADPositiveParticipants(
- _id,
- _scratchVadPositiveParticipants,
- _scratchVadPositiveParticipantsAmount);
- _mixerStatusCallback->MixedAudioLevel(_id,audioLevel);
- }
- }
-
- // Reclaim all outstanding memory.
- _audioFramePool->PushMemory(mixedAudio);
- ClearAudioFrameList(mixList);
- ClearAudioFrameList(rampOutList);
- ClearAudioFrameList(additionalFramesList);
- {
- CriticalSectionScoped cs(_crit.get());
- _processCalls--;
- }
- return retval;
-}
-
-WebRtc_Word32 AudioConferenceMixerImpl::RegisterMixedStreamCallback(
- AudioMixerOutputReceiver& mixReceiver)
-{
- CriticalSectionScoped cs(_cbCrit.get());
- if(_mixReceiver != NULL)
- {
- return -1;
- }
- _mixReceiver = &mixReceiver;
- return 0;
-}
-
-WebRtc_Word32 AudioConferenceMixerImpl::UnRegisterMixedStreamCallback()
-{
- CriticalSectionScoped cs(_cbCrit.get());
- if(_mixReceiver == NULL)
- {
- return -1;
- }
- _mixReceiver = NULL;
- return 0;
-}
-
-WebRtc_Word32 AudioConferenceMixerImpl::SetOutputFrequency(
- const Frequency frequency)
-{
- CriticalSectionScoped cs(_crit.get());
- const int error = _limiter->set_sample_rate_hz(frequency);
- if(error != _limiter->kNoError)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
- "Error from AudioProcessing: %d", error);
- return -1;
- }
-
- _outputFrequency = frequency;
- _sampleSize = (_outputFrequency*kProcessPeriodicityInMs) / 1000;
-
- return 0;
-}
-
-AudioConferenceMixer::Frequency
-AudioConferenceMixerImpl::OutputFrequency() const
-{
- CriticalSectionScoped cs(_crit.get());
- return _outputFrequency;
-}
-
-bool AudioConferenceMixerImpl::SetNumLimiterChannels(int numChannels)
-{
- if(_limiter->num_input_channels() != numChannels)
- {
- const int error = _limiter->set_num_channels(numChannels,
- numChannels);
- if(error != _limiter->kNoError)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
- "Error from AudioProcessing: %d", error);
- assert(false);
- return false;
- }
- }
-
- return true;
-}
-
-WebRtc_Word32 AudioConferenceMixerImpl::RegisterMixerStatusCallback(
- AudioMixerStatusReceiver& mixerStatusCallback,
- const WebRtc_UWord32 amountOf10MsBetweenCallbacks)
-{
- if(amountOf10MsBetweenCallbacks == 0)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceAudioMixerServer,
- _id,
- "amountOf10MsBetweenCallbacks(%d) needs to be larger than 0");
- return -1;
- }
- {
- CriticalSectionScoped cs(_cbCrit.get());
- if(_mixerStatusCallback != NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
- "Mixer status callback already registered");
- return -1;
- }
- _mixerStatusCallback = &mixerStatusCallback;
- }
- {
- CriticalSectionScoped cs(_crit.get());
- _amountOf10MsBetweenCallbacks = amountOf10MsBetweenCallbacks;
- _amountOf10MsUntilNextCallback = 0;
- _mixerStatusCb = true;
- }
- return 0;
-}
-
-WebRtc_Word32 AudioConferenceMixerImpl::UnRegisterMixerStatusCallback()
-{
- {
- CriticalSectionScoped cs(_crit.get());
- if(!_mixerStatusCb)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
- "Mixer status callback not registered");
- return -1;
- }
- _mixerStatusCb = false;
- }
- {
- CriticalSectionScoped cs(_cbCrit.get());
- _mixerStatusCallback = NULL;
- }
- return 0;
-}
-
-WebRtc_Word32 AudioConferenceMixerImpl::SetMixabilityStatus(
- MixerParticipant& participant,
- const bool mixable)
-{
- if (!mixable)
- {
- // Anonymous participants are in a separate list. Make sure that the
- // participant is in the _participantList if it is being mixed.
- SetAnonymousMixabilityStatus(participant, false);
- }
- WebRtc_UWord32 numMixedParticipants;
- {
- CriticalSectionScoped cs(_cbCrit.get());
- const bool isMixed =
- IsParticipantInList(participant,_participantList);
- // API must be called with a new state.
- if(!(mixable ^ isMixed))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
- "Mixable is aready %s",
- isMixed ? "ON" : "off");
- return -1;
- }
- bool success = false;
- if(mixable)
- {
- success = AddParticipantToList(participant,_participantList);
- }
- else
- {
- success = RemoveParticipantFromList(participant,_participantList);
- }
- if(!success)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
- "failed to %s participant",
- mixable ? "add" : "remove");
- assert(false);
- return -1;
- }
-
- int numMixedNonAnonymous = _participantList.GetSize();
- if (numMixedNonAnonymous > kMaximumAmountOfMixedParticipants)
- {
- numMixedNonAnonymous = kMaximumAmountOfMixedParticipants;
- }
- numMixedParticipants = numMixedNonAnonymous +
- _additionalParticipantList.GetSize();
- }
- // A MixerParticipant was added or removed. Make sure the scratch
- // buffer is updated if necessary.
- // Note: The scratch buffer may only be updated in Process().
- CriticalSectionScoped cs(_crit.get());
- _numMixedParticipants = numMixedParticipants;
- return 0;
-}
-
-WebRtc_Word32 AudioConferenceMixerImpl::MixabilityStatus(
- MixerParticipant& participant,
- bool& mixable)
-{
- CriticalSectionScoped cs(_cbCrit.get());
- mixable = IsParticipantInList(participant, _participantList);
- return 0;
-}
-
-WebRtc_Word32 AudioConferenceMixerImpl::SetAnonymousMixabilityStatus(
- MixerParticipant& participant, const bool anonymous)
-{
- CriticalSectionScoped cs(_cbCrit.get());
- if(IsParticipantInList(participant, _additionalParticipantList))
- {
- if(anonymous)
- {
- return 0;
- }
- if(!RemoveParticipantFromList(participant, _additionalParticipantList))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
- "unable to remove participant from anonymous list");
- assert(false);
- return -1;
- }
- return AddParticipantToList(participant, _participantList) ? 0 : -1;
- }
- if(!anonymous)
- {
- return 0;
- }
- const bool mixable = RemoveParticipantFromList(participant,
- _participantList);
- if(!mixable)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceAudioMixerServer,
- _id,
- "participant must be registered before turning it into anonymous");
- // Setting anonymous status is only possible if MixerParticipant is
- // already registered.
- return -1;
- }
- return AddParticipantToList(participant, _additionalParticipantList) ?
- 0 : -1;
-}
-
-WebRtc_Word32 AudioConferenceMixerImpl::AnonymousMixabilityStatus(
- MixerParticipant& participant, bool& mixable)
-{
- CriticalSectionScoped cs(_cbCrit.get());
- mixable = IsParticipantInList(participant,
- _additionalParticipantList);
- return 0;
-}
-
-WebRtc_Word32 AudioConferenceMixerImpl::SetMinimumMixingFrequency(
- Frequency freq)
-{
- // Make sure that only allowed sampling frequencies are used. Use closest
- // higher sampling frequency to avoid losing information.
- if (static_cast<int>(freq) == 12000)
- {
- freq = kWbInHz;
- } else if (static_cast<int>(freq) == 24000) {
- freq = kSwbInHz;
- }
-
- if((freq == kNbInHz) || (freq == kWbInHz) || (freq == kSwbInHz) ||
- (freq == kLowestPossible))
- {
- _minimumMixingFreq=freq;
- return 0;
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
- "SetMinimumMixingFrequency incorrect frequency: %i",freq);
- assert(false);
- return -1;
- }
-}
-
-// Check all AudioFrames that are to be mixed. The highest sampling frequency
-// found is the lowest that can be used without losing information.
-WebRtc_Word32 AudioConferenceMixerImpl::GetLowestMixingFrequency()
-{
- const int participantListFrequency =
- GetLowestMixingFrequencyFromList(_participantList);
- const int anonymousListFrequency =
- GetLowestMixingFrequencyFromList(_additionalParticipantList);
- const int highestFreq =
- (participantListFrequency > anonymousListFrequency) ?
- participantListFrequency : anonymousListFrequency;
- // Check if the user specified a lowest mixing frequency.
- if(_minimumMixingFreq != kLowestPossible)
- {
- if(_minimumMixingFreq > highestFreq)
- {
- return _minimumMixingFreq;
- }
- }
- return highestFreq;
-}
-
-WebRtc_Word32 AudioConferenceMixerImpl::GetLowestMixingFrequencyFromList(
- ListWrapper& mixList)
-{
- WebRtc_Word32 highestFreq = 8000;
- ListItem* item = mixList.First();
- while(item)
- {
- MixerParticipant* participant =
- static_cast<MixerParticipant*>(item->GetItem());
- const WebRtc_Word32 neededFrequency = participant->NeededFrequency(_id);
- if(neededFrequency > highestFreq)
- {
- highestFreq = neededFrequency;
- }
- item = mixList.Next(item);
- }
- return highestFreq;
-}
-
-void AudioConferenceMixerImpl::UpdateToMix(
- ListWrapper& mixList,
- ListWrapper& rampOutList,
- MapWrapper& mixParticipantList,
- WebRtc_UWord32& maxAudioFrameCounter)
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
- "UpdateToMix(mixList,rampOutList,mixParticipantList,%d)",
- maxAudioFrameCounter);
- const WebRtc_UWord32 mixListStartSize = mixList.GetSize();
- ListWrapper activeList; // Elements are AudioFrames
- // Struct needed by the passive lists to keep track of which AudioFrame
- // belongs to which MixerParticipant.
- struct ParticipantFramePair
- {
- MixerParticipant* participant;
- AudioFrame* audioFrame;
- };
- ListWrapper passiveWasNotMixedList; // Elements are MixerParticipant
- ListWrapper passiveWasMixedList; // Elements are MixerParticipant
- ListItem* item = _participantList.First();
- while(item)
- {
- // Stop keeping track of passive participants if there are already
- // enough participants available (they wont be mixed anyway).
- bool mustAddToPassiveList = (maxAudioFrameCounter >
- (activeList.GetSize() +
- passiveWasMixedList.GetSize() +
- passiveWasNotMixedList.GetSize()));
-
- MixerParticipant* participant = static_cast<MixerParticipant*>(
- item->GetItem());
- bool wasMixed = false;
- participant->_mixHistory->WasMixed(wasMixed);
- AudioFrame* audioFrame = NULL;
- if(_audioFramePool->PopMemory(audioFrame) == -1)
- {
- WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id,
- "failed PopMemory() call");
- assert(false);
- return;
- }
- audioFrame->_frequencyInHz = _outputFrequency;
-
- if(participant->GetAudioFrame(_id,*audioFrame) != 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
- "failed to GetAudioFrame() from participant");
- _audioFramePool->PushMemory(audioFrame);
- item = _participantList.Next(item);
- continue;
- }
- // TODO(henrike): this assert triggers in some test cases where SRTP is
- // used which prevents NetEQ from making a VAD. Temporarily disable this
- // assert until the problem is fixed on a higher level.
- // assert(audioFrame->_vadActivity != AudioFrame::kVadUnknown);
- if (audioFrame->_vadActivity == AudioFrame::kVadUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
- "invalid VAD state from participant");
- }
-
- if(audioFrame->_vadActivity == AudioFrame::kVadActive)
- {
- if(!wasMixed)
- {
- RampIn(*audioFrame);
- }
-
- if(activeList.GetSize() >= maxAudioFrameCounter)
- {
- // There are already more active participants than should be
- // mixed. Only keep the ones with the highest energy.
- ListItem* replaceItem = NULL;
- CalculateEnergy(*audioFrame);
- WebRtc_UWord32 lowestEnergy = audioFrame->_energy;
-
- ListItem* activeItem = activeList.First();
- while(activeItem)
- {
- AudioFrame* replaceFrame = static_cast<AudioFrame*>(
- activeItem->GetItem());
- CalculateEnergy(*replaceFrame);
- if(replaceFrame->_energy < lowestEnergy)
- {
- replaceItem = activeItem;
- lowestEnergy = replaceFrame->_energy;
- }
- activeItem = activeList.Next(activeItem);
- }
- if(replaceItem != NULL)
- {
- AudioFrame* replaceFrame = static_cast<AudioFrame*>(
- replaceItem->GetItem());
-
- bool replaceWasMixed = false;
- MapItem* replaceParticipant = mixParticipantList.Find(
- replaceFrame->_id);
- // When a frame is pushed to |activeList| it is also pushed
- // to mixParticipantList with the frame's id. This means
- // that the Find call above should never fail.
- if(replaceParticipant == NULL)
- {
- assert(false);
- } else {
- static_cast<MixerParticipant*>(
- replaceParticipant->GetItem())->_mixHistory->
- WasMixed(replaceWasMixed);
-
- mixParticipantList.Erase(replaceFrame->_id);
- activeList.Erase(replaceItem);
-
- activeList.PushFront(static_cast<void*>(audioFrame));
- mixParticipantList.Insert(
- audioFrame->_id,
- static_cast<void*>(participant));
- assert(mixParticipantList.Size() <=
- kMaximumAmountOfMixedParticipants);
-
- if(replaceWasMixed)
- {
- RampOut(*replaceFrame);
- rampOutList.PushBack(
- static_cast<void*>(replaceFrame));
- assert(rampOutList.GetSize() <=
- kMaximumAmountOfMixedParticipants);
- } else {
- _audioFramePool->PushMemory(replaceFrame);
- }
- }
- } else {
- if(wasMixed)
- {
- RampOut(*audioFrame);
- rampOutList.PushBack(static_cast<void*>(audioFrame));
- assert(rampOutList.GetSize() <=
- kMaximumAmountOfMixedParticipants);
- } else {
- _audioFramePool->PushMemory(audioFrame);
- }
- }
- } else {
- activeList.PushFront(static_cast<void*>(audioFrame));
- mixParticipantList.Insert(audioFrame->_id,
- static_cast<void*>(participant));
- assert(mixParticipantList.Size() <=
- kMaximumAmountOfMixedParticipants);
- }
- } else {
- if(wasMixed)
- {
- ParticipantFramePair* pair = new ParticipantFramePair;
- pair->audioFrame = audioFrame;
- pair->participant = participant;
- passiveWasMixedList.PushBack(static_cast<void*>(pair));
- } else if(mustAddToPassiveList) {
- RampIn(*audioFrame);
- ParticipantFramePair* pair = new ParticipantFramePair;
- pair->audioFrame = audioFrame;
- pair->participant = participant;
- passiveWasNotMixedList.PushBack(static_cast<void*>(pair));
- } else {
- _audioFramePool->PushMemory(audioFrame);
- }
- }
- item = _participantList.Next(item);
- }
- assert(activeList.GetSize() <= maxAudioFrameCounter);
- // At this point it is known which participants should be mixed. Transfer
- // this information to this functions output parameters.
- while(!activeList.Empty())
- {
- ListItem* mixItem = activeList.First();
- mixList.PushBack(mixItem->GetItem());
- activeList.Erase(mixItem);
- }
- // Always mix a constant number of AudioFrames. If there aren't enough
- // active participants mix passive ones. Starting with those that was mixed
- // last iteration.
- while(!passiveWasMixedList.Empty())
- {
- ListItem* mixItem = passiveWasMixedList.First();
- ParticipantFramePair* pair = static_cast<ParticipantFramePair*>(
- mixItem->GetItem());
- if(mixList.GetSize() < maxAudioFrameCounter + mixListStartSize)
- {
- mixList.PushBack(pair->audioFrame);
- mixParticipantList.Insert(pair->audioFrame->_id,
- static_cast<void*>(pair->participant));
- assert(mixParticipantList.Size() <=
- kMaximumAmountOfMixedParticipants);
- }
- else
- {
- _audioFramePool->PushMemory(pair->audioFrame);
- }
- delete pair;
- passiveWasMixedList.Erase(mixItem);
- }
- // And finally the ones that have not been mixed for a while.
- while(!passiveWasNotMixedList.Empty())
- {
- ListItem* mixItem = passiveWasNotMixedList.First();
- ParticipantFramePair* pair = static_cast<ParticipantFramePair*>(
- mixItem->GetItem());
- if(mixList.GetSize() < maxAudioFrameCounter + mixListStartSize)
- {
- mixList.PushBack(pair->audioFrame);
- mixParticipantList.Insert(pair->audioFrame->_id,
- static_cast<void*>(pair->participant));
- assert(mixParticipantList.Size() <=
- kMaximumAmountOfMixedParticipants);
- }
- else
- {
- _audioFramePool->PushMemory(pair->audioFrame);
- }
- delete pair;
- passiveWasNotMixedList.Erase(mixItem);
- }
- assert(maxAudioFrameCounter + mixListStartSize >= mixList.GetSize());
- maxAudioFrameCounter += mixListStartSize - mixList.GetSize();
-}
-
-void AudioConferenceMixerImpl::GetAdditionalAudio(
- ListWrapper& additionalFramesList)
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
- "GetAdditionalAudio(additionalFramesList)");
- ListItem* item = _additionalParticipantList.First();
- while(item)
- {
- // The GetAudioFrame() callback may remove the current item. Store the
- // next item just in case that happens.
- ListItem* nextItem = _additionalParticipantList.Next(item);
-
- MixerParticipant* participant = static_cast<MixerParticipant*>(
- item->GetItem());
- AudioFrame* audioFrame = NULL;
- if(_audioFramePool->PopMemory(audioFrame) == -1)
- {
- WEBRTC_TRACE(kTraceMemory, kTraceAudioMixerServer, _id,
- "failed PopMemory() call");
- assert(false);
- return;
- }
- audioFrame->_frequencyInHz = _outputFrequency;
- if(participant->GetAudioFrame(_id, *audioFrame) != 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioMixerServer, _id,
- "failed to GetAudioFrame() from participant");
- _audioFramePool->PushMemory(audioFrame);
- item = nextItem;
- continue;
- }
- if(audioFrame->_payloadDataLengthInSamples == 0)
- {
- // Empty frame. Don't use it.
- _audioFramePool->PushMemory(audioFrame);
- item = nextItem;
- continue;
- }
- additionalFramesList.PushBack(static_cast<void*>(audioFrame));
- item = nextItem;
- }
-}
-
-void AudioConferenceMixerImpl::UpdateMixedStatus(
- MapWrapper& mixedParticipantsMap)
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
- "UpdateMixedStatus(mixedParticipantsMap)");
- assert(mixedParticipantsMap.Size() <= kMaximumAmountOfMixedParticipants);
-
- // Loop through all participants. If they are in the mix map they
- // were mixed.
- ListItem* participantItem = _participantList.First();
- while(participantItem != NULL)
- {
- bool isMixed = false;
- MixerParticipant* participant =
- static_cast<MixerParticipant*>(participantItem->GetItem());
-
- MapItem* mixedItem = mixedParticipantsMap.First();
- while(mixedItem)
- {
- if(participant == mixedItem->GetItem())
- {
- isMixed = true;
- break;
- }
- mixedItem = mixedParticipantsMap.Next(mixedItem);
- }
- participant->_mixHistory->SetIsMixed(isMixed);
- participantItem = _participantList.Next(participantItem);
- }
-}
-
-void AudioConferenceMixerImpl::ClearAudioFrameList(ListWrapper& audioFrameList)
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
- "ClearAudioFrameList(audioFrameList)");
- ListItem* item = audioFrameList.First();
- while(item)
- {
- AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem());
- _audioFramePool->PushMemory(audioFrame);
- audioFrameList.Erase(item);
- item = audioFrameList.First();
- }
-}
-
-void AudioConferenceMixerImpl::UpdateVADPositiveParticipants(
- ListWrapper& mixList)
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
- "UpdateVADPositiveParticipants(mixList)");
-
- ListItem* item = mixList.First();
- while(item != NULL)
- {
- AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem());
- CalculateEnergy(*audioFrame);
- if(audioFrame->_vadActivity == AudioFrame::kVadActive)
- {
- _scratchVadPositiveParticipants[
- _scratchVadPositiveParticipantsAmount].participant =
- audioFrame->_id;
- _scratchVadPositiveParticipants[
- _scratchVadPositiveParticipantsAmount].level =
- audioFrame->_volume;
- _scratchVadPositiveParticipantsAmount++;
- }
- item = mixList.Next(item);
- }
-}
-
-bool AudioConferenceMixerImpl::IsParticipantInList(
- MixerParticipant& participant,
- ListWrapper& participantList)
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
- "IsParticipantInList(participant,participantList)");
- ListItem* item = participantList.First();
- while(item != NULL)
- {
- MixerParticipant* rhsParticipant =
- static_cast<MixerParticipant*>(item->GetItem());
- if(&participant == rhsParticipant)
- {
- return true;
- }
- item = participantList.Next(item);
- }
- return false;
-}
-
-bool AudioConferenceMixerImpl::AddParticipantToList(
- MixerParticipant& participant,
- ListWrapper& participantList)
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
- "AddParticipantToList(participant, participantList)");
- if(participantList.PushBack(static_cast<void*>(&participant)) == -1)
- {
- return false;
- }
- // Make sure that the mixed status is correct for new MixerParticipant.
- participant._mixHistory->ResetMixedStatus();
- return true;
-}
-
-bool AudioConferenceMixerImpl::RemoveParticipantFromList(
- MixerParticipant& participant,
- ListWrapper& participantList)
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
- "RemoveParticipantFromList(participant, participantList)");
- ListItem* item = participantList.First();
- while(item)
- {
- if(item->GetItem() == &participant)
- {
- participantList.Erase(item);
- // Participant is no longer mixed, reset to default.
- participant._mixHistory->ResetMixedStatus();
- return true;
- }
- item = participantList.Next(item);
- }
- return false;
-}
-
-WebRtc_Word32 AudioConferenceMixerImpl::MixFromList(
- AudioFrame& mixedAudio,
- const ListWrapper& audioFrameList)
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
- "MixFromList(mixedAudio, audioFrameList)");
- WebRtc_UWord32 position = 0;
- ListItem* item = audioFrameList.First();
- if(item == NULL)
- {
- return 0;
- }
-
- if(_numMixedParticipants == 1)
- {
- // No mixing required here; skip the saturation protection.
- AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem());
- mixedAudio = *audioFrame;
- SetParticipantStatistics(&_scratchMixedParticipants[position],
- *audioFrame);
- return 0;
- }
-
- while(item != NULL)
- {
- if(position >= kMaximumAmountOfMixedParticipants)
- {
- WEBRTC_TRACE(
- kTraceMemory,
- kTraceAudioMixerServer,
- _id,
- "Trying to mix more than max amount of mixed participants:%d!",
- kMaximumAmountOfMixedParticipants);
- // Assert and avoid crash
- assert(false);
- position = 0;
- }
- AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem());
-
- // Divide by two to avoid saturation in the mixing.
- *audioFrame >>= 1;
- mixedAudio += *audioFrame;
-
- SetParticipantStatistics(&_scratchMixedParticipants[position],
- *audioFrame);
-
- position++;
- item = audioFrameList.Next(item);
- }
-
- return 0;
-}
-
-// TODO(andrew): consolidate this function with MixFromList.
-WebRtc_Word32 AudioConferenceMixerImpl::MixAnonomouslyFromList(
- AudioFrame& mixedAudio,
- const ListWrapper& audioFrameList)
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioMixerServer, _id,
- "MixAnonomouslyFromList(mixedAudio, audioFrameList)");
- ListItem* item = audioFrameList.First();
- if(item == NULL)
- return 0;
-
- if(_numMixedParticipants == 1)
- {
- // No mixing required here; skip the saturation protection.
- AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem());
- mixedAudio = *audioFrame;
- return 0;
- }
-
- while(item != NULL)
- {
- AudioFrame* audioFrame = static_cast<AudioFrame*>(item->GetItem());
- // Divide by two to avoid saturation in the mixing.
- *audioFrame >>= 1;
- mixedAudio += *audioFrame;
- item = audioFrameList.Next(item);
- }
- return 0;
-}
-
-bool AudioConferenceMixerImpl::LimitMixedAudio(AudioFrame& mixedAudio)
-{
- if(_numMixedParticipants == 1)
- {
- return true;
- }
-
- // Smoothly limit the mixed frame.
- const int error = _limiter->ProcessStream(&mixedAudio);
-
- // And now we can safely restore the level. This procedure results in
- // some loss of resolution, deemed acceptable.
- //
- // It's possible to apply the gain in the AGC (with a target level of 0 dbFS
- // and compression gain of 6 dB). However, in the transition frame when this
- // is enabled (moving from one to two participants) it has the potential to
- // create discontinuities in the mixed frame.
- //
- // Instead we double the frame (with addition since left-shifting a
- // negative value is undefined).
- mixedAudio += mixedAudio;
-
- if(error != _limiter->kNoError)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioMixerServer, _id,
- "Error from AudioProcessing: %d", error);
- assert(false);
- return false;
- }
- return true;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h b/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
deleted file mode 100644
index efa5d68..0000000
--- a/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_impl.h
+++ /dev/null
@@ -1,209 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
-
-#include "atomic32_wrapper.h"
-#include "audio_conference_mixer.h"
-#include "engine_configurations.h"
-#include "level_indicator.h"
-#include "list_wrapper.h"
-#include "memory_pool.h"
-#include "module_common_types.h"
-#include "scoped_ptr.h"
-#include "time_scheduler.h"
-
-namespace webrtc {
-class AudioProcessing;
-class CriticalSectionWrapper;
-
-// Cheshire cat implementation of MixerParticipant's non virtual functions.
-class MixHistory
-{
-public:
- MixHistory();
- ~MixHistory();
-
- // MixerParticipant function
- WebRtc_Word32 IsMixed(bool& mixed) const;
-
- // Sets wasMixed to true if the participant was mixed previous mix
- // iteration.
- WebRtc_Word32 WasMixed(bool& wasMixed) const;
-
- // Updates the mixed status.
- WebRtc_Word32 SetIsMixed(const bool mixed);
-
- void ResetMixedStatus();
-private:
- Atomic32Wrapper _isMixed; // 0 = false, 1 = true
-};
-
-class AudioConferenceMixerImpl : public AudioConferenceMixer
-{
-public:
- // AudioProcessing only accepts 10 ms frames.
- enum {kProcessPeriodicityInMs = 10};
-
- AudioConferenceMixerImpl(int id);
- ~AudioConferenceMixerImpl();
-
- // Must be called after ctor.
- bool Init();
-
- // Module functions
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
- virtual WebRtc_Word32 TimeUntilNextProcess();
- virtual WebRtc_Word32 Process();
-
- // AudioConferenceMixer functions
- virtual WebRtc_Word32 RegisterMixedStreamCallback(
- AudioMixerOutputReceiver& mixReceiver);
- virtual WebRtc_Word32 UnRegisterMixedStreamCallback();
- virtual WebRtc_Word32 RegisterMixerStatusCallback(
- AudioMixerStatusReceiver& mixerStatusCallback,
- const WebRtc_UWord32 amountOf10MsBetweenCallbacks);
- virtual WebRtc_Word32 UnRegisterMixerStatusCallback();
- virtual WebRtc_Word32 SetMixabilityStatus(MixerParticipant& participant,
- const bool mixable);
- virtual WebRtc_Word32 MixabilityStatus(MixerParticipant& participant,
- bool& mixable);
- virtual WebRtc_Word32 SetMinimumMixingFrequency(Frequency freq);
- virtual WebRtc_Word32 SetAnonymousMixabilityStatus(
- MixerParticipant& participant, const bool mixable);
- virtual WebRtc_Word32 AnonymousMixabilityStatus(
- MixerParticipant& participant, bool& mixable);
-private:
- enum{DEFAULT_AUDIO_FRAME_POOLSIZE = 50};
-
- // Set/get mix frequency
- WebRtc_Word32 SetOutputFrequency(const Frequency frequency);
- Frequency OutputFrequency() const;
-
- // Must be called whenever an audio frame indicates the number of channels
- // has changed.
- bool SetNumLimiterChannels(int numChannels);
-
- // Fills mixList with the AudioFrames pointers that should be used when
- // mixing. Fills mixParticipantList with ParticipantStatistics for the
- // participants who's AudioFrames are inside mixList.
- // maxAudioFrameCounter both input and output specifies how many more
- // AudioFrames that are allowed to be mixed.
- // rampOutList contain AudioFrames corresponding to an audio stream that
- // used to be mixed but shouldn't be mixed any longer. These AudioFrames
- // should be ramped out over this AudioFrame to avoid audio discontinuities.
- void UpdateToMix(ListWrapper& mixList, ListWrapper& rampOutList,
- MapWrapper& mixParticipantList,
- WebRtc_UWord32& maxAudioFrameCounter);
-
- // Return the lowest mixing frequency that can be used without having to
- // downsample any audio.
- WebRtc_Word32 GetLowestMixingFrequency();
- WebRtc_Word32 GetLowestMixingFrequencyFromList(ListWrapper& mixList);
-
- // Return the AudioFrames that should be mixed anonymously.
- void GetAdditionalAudio(ListWrapper& additionalFramesList);
-
- // Update the MixHistory of all MixerParticipants. mixedParticipantsList
- // should contain a map of MixerParticipants that have been mixed.
- void UpdateMixedStatus(MapWrapper& mixedParticipantsList);
-
- // Clears audioFrameList and reclaims all memory associated with it.
- void ClearAudioFrameList(ListWrapper& audioFrameList);
-
- // Update the list of MixerParticipants who have a positive VAD. mixList
- // should be a list of AudioFrames
- void UpdateVADPositiveParticipants(
- ListWrapper& mixList);
-
- // This function returns true if it finds the MixerParticipant in the
- // specified list of MixerParticipants.
- bool IsParticipantInList(
- MixerParticipant& participant,
- ListWrapper& participantList);
-
- // Add/remove the MixerParticipant to the specified
- // MixerParticipant list.
- bool AddParticipantToList(
- MixerParticipant& participant,
- ListWrapper& participantList);
- bool RemoveParticipantFromList(
- MixerParticipant& removeParticipant,
- ListWrapper& participantList);
-
- // Mix the AudioFrames stored in audioFrameList into mixedAudio.
- WebRtc_Word32 MixFromList(
- AudioFrame& mixedAudio,
- const ListWrapper& audioFrameList);
- // Mix the AudioFrames stored in audioFrameList into mixedAudio. No
- // record will be kept of this mix (e.g. the corresponding MixerParticipants
- // will not be marked as IsMixed()
- WebRtc_Word32 MixAnonomouslyFromList(AudioFrame& mixedAudio,
- const ListWrapper& audioFrameList);
-
- bool LimitMixedAudio(AudioFrame& mixedAudio);
-
- // Scratch memory
- // Note that the scratch memory may only be touched in the scope of
- // Process().
- WebRtc_UWord32 _scratchParticipantsToMixAmount;
- ParticipantStatistics _scratchMixedParticipants[
- kMaximumAmountOfMixedParticipants];
- WebRtc_UWord32 _scratchVadPositiveParticipantsAmount;
- ParticipantStatistics _scratchVadPositiveParticipants[
- kMaximumAmountOfMixedParticipants];
-
- scoped_ptr<CriticalSectionWrapper> _crit;
- scoped_ptr<CriticalSectionWrapper> _cbCrit;
-
- WebRtc_Word32 _id;
-
- Frequency _minimumMixingFreq;
-
- // Mix result callback
- AudioMixerOutputReceiver* _mixReceiver;
-
- AudioMixerStatusReceiver* _mixerStatusCallback;
- WebRtc_UWord32 _amountOf10MsBetweenCallbacks;
- WebRtc_UWord32 _amountOf10MsUntilNextCallback;
- bool _mixerStatusCb;
-
- // The current sample frequency and sample size when mixing.
- Frequency _outputFrequency;
- WebRtc_UWord16 _sampleSize;
-
- // Memory pool to avoid allocating/deallocating AudioFrames
- MemoryPool<AudioFrame>* _audioFramePool;
-
- // List of all participants. Note all lists are disjunct
- ListWrapper _participantList; // May be mixed.
- ListWrapper _additionalParticipantList; // Always mixed, anonomously.
-
- WebRtc_UWord32 _numMixedParticipants;
-
- WebRtc_UWord32 _timeStamp;
-
- // Metronome class.
- TimeScheduler _timeScheduler;
-
- // Smooth level indicator.
- LevelIndicator _mixedAudioLevel;
-
- // Counter keeping track of concurrent calls to process.
- // Note: should never be higher than 1 or lower than 0.
- WebRtc_Word16 _processCalls;
-
- // Used for inhibiting saturation in mixing.
- scoped_ptr<AudioProcessing> _limiter;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_CONFERENCE_MIXER_IMPL_H_
diff --git a/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_unittest.cc b/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_unittest.cc
deleted file mode 100644
index f895fbd..0000000
--- a/trunk/src/modules/audio_conference_mixer/source/audio_conference_mixer_unittest.cc
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Empty test just to get code coverage metrics for this dir.
- */
-#include "audio_conference_mixer.h"
-#include "gtest/gtest.h"
-
-TEST(AudioConferenceMixerTest, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/modules/audio_conference_mixer/source/audio_frame_manipulator.cc b/trunk/src/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
deleted file mode 100644
index 8716454..0000000
--- a/trunk/src/modules/audio_conference_mixer/source/audio_frame_manipulator.cc
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_frame_manipulator.h"
-#include "module_common_types.h"
-#include "typedefs.h"
-
-namespace {
-// Linear ramping over 80 samples.
-// TODO(hellner): ramp using fix point?
-const float rampArray[] = {0.0000f, 0.0127f, 0.0253f, 0.0380f,
- 0.0506f, 0.0633f, 0.0759f, 0.0886f,
- 0.1013f, 0.1139f, 0.1266f, 0.1392f,
- 0.1519f, 0.1646f, 0.1772f, 0.1899f,
- 0.2025f, 0.2152f, 0.2278f, 0.2405f,
- 0.2532f, 0.2658f, 0.2785f, 0.2911f,
- 0.3038f, 0.3165f, 0.3291f, 0.3418f,
- 0.3544f, 0.3671f, 0.3797f, 0.3924f,
- 0.4051f, 0.4177f, 0.4304f, 0.4430f,
- 0.4557f, 0.4684f, 0.4810f, 0.4937f,
- 0.5063f, 0.5190f, 0.5316f, 0.5443f,
- 0.5570f, 0.5696f, 0.5823f, 0.5949f,
- 0.6076f, 0.6203f, 0.6329f, 0.6456f,
- 0.6582f, 0.6709f, 0.6835f, 0.6962f,
- 0.7089f, 0.7215f, 0.7342f, 0.7468f,
- 0.7595f, 0.7722f, 0.7848f, 0.7975f,
- 0.8101f, 0.8228f, 0.8354f, 0.8481f,
- 0.8608f, 0.8734f, 0.8861f, 0.8987f,
- 0.9114f, 0.9241f, 0.9367f, 0.9494f,
- 0.9620f, 0.9747f, 0.9873f, 1.0000f};
-const int rampSize = sizeof(rampArray)/sizeof(rampArray[0]);
-} // namespace
-
-namespace webrtc {
-void CalculateEnergy(AudioFrame& audioFrame)
-{
- if(audioFrame._energy != 0xffffffff)
- {
- return;
- }
- audioFrame._energy = 0;
- for(int position = 0; position < audioFrame._payloadDataLengthInSamples;
- position++)
- {
- // TODO(andrew): this can easily overflow.
- audioFrame._energy += audioFrame._payloadData[position] *
- audioFrame._payloadData[position];
- }
-}
-
-void RampIn(AudioFrame& audioFrame)
-{
- assert(rampSize <= audioFrame._payloadDataLengthInSamples);
- for(int i = 0; i < rampSize; i++)
- {
- audioFrame._payloadData[i] = static_cast<WebRtc_Word16>
- (rampArray[i] * audioFrame._payloadData[i]);
- }
-}
-
-void RampOut(AudioFrame& audioFrame)
-{
- assert(rampSize <= audioFrame._payloadDataLengthInSamples);
- for(int i = 0; i < rampSize; i++)
- {
- const int rampPos = rampSize - 1 - i;
- audioFrame._payloadData[i] = static_cast<WebRtc_Word16>
- (rampArray[rampPos] * audioFrame._payloadData[i]);
- }
- memset(&audioFrame._payloadData[rampSize], 0,
- (audioFrame._payloadDataLengthInSamples - rampSize) *
- sizeof(audioFrame._payloadData[0]));
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_conference_mixer/source/audio_frame_manipulator.h b/trunk/src/modules/audio_conference_mixer/source/audio_frame_manipulator.h
deleted file mode 100644
index fdf5d33..0000000
--- a/trunk/src/modules/audio_conference_mixer/source/audio_frame_manipulator.h
+++ /dev/null
@@ -1,26 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_FRAME_MANIPULATOR_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_FRAME_MANIPULATOR_H_
-
-namespace webrtc {
-class AudioFrame;
-
-// Updates the audioFrame's energy (based on its samples).
-void CalculateEnergy(AudioFrame& audioFrame);
-
-// Apply linear step function that ramps in/out the audio samples in audioFrame
-void RampIn(AudioFrame& audioFrame);
-void RampOut(AudioFrame& audioFrame);
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_AUDIO_FRAME_MANIPULATOR_H_
diff --git a/trunk/src/modules/audio_conference_mixer/source/level_indicator.cc b/trunk/src/modules/audio_conference_mixer/source/level_indicator.cc
deleted file mode 100644
index 799a47d..0000000
--- a/trunk/src/modules/audio_conference_mixer/source/level_indicator.cc
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "level_indicator.h"
-
-namespace webrtc {
-// Array for adding smothing to level changes (ad-hoc).
-const WebRtc_UWord32 perm[] =
- {0,1,2,3,4,4,5,5,5,5,6,6,6,6,6,7,7,7,7,8,8,8,9,9,9,9,9,9,9,9,9,9,9};
-
-LevelIndicator::LevelIndicator()
- : _max(0),
- _count(0),
- _currentLevel(0)
-{
-}
-
-LevelIndicator::~LevelIndicator()
-{
-}
-
-// Level is based on the highest absolute value for all samples.
-void LevelIndicator::ComputeLevel(const WebRtc_Word16* speech,
- const WebRtc_UWord16 nrOfSamples)
-{
- WebRtc_Word32 min = 0;
- for(WebRtc_UWord32 i = 0; i < nrOfSamples; i++)
- {
- if(_max < speech[i])
- {
- _max = speech[i];
- }
- if(min > speech[i])
- {
- min = speech[i];
- }
- }
-
- // Absolute max value.
- if(-min > _max)
- {
- _max = -min;
- }
-
- if(_count == TICKS_BEFORE_CALCULATION)
- {
- // Highest sample value maps directly to a level.
- WebRtc_Word32 position = _max / 1000;
- if ((position == 0) &&
- (_max > 250))
- {
- position = 1;
- }
- _currentLevel = perm[position];
- // The max value is decayed and stored so that it can be reused to slow
- // down decreases in level.
- _max = _max >> 1;
- _count = 0;
- } else {
- _count++;
- }
-}
-
-WebRtc_Word32 LevelIndicator::GetLevel()
-{
- return _currentLevel;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_conference_mixer/source/level_indicator.h b/trunk/src/modules/audio_conference_mixer/source/level_indicator.h
deleted file mode 100644
index bdcdf8e..0000000
--- a/trunk/src/modules/audio_conference_mixer/source/level_indicator.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_
-
-#include "typedefs.h"
-
-namespace webrtc {
-class LevelIndicator
-{
-public:
- enum{TICKS_BEFORE_CALCULATION = 10};
-
- LevelIndicator();
- ~LevelIndicator();
-
- // Updates the level.
- void ComputeLevel(const WebRtc_Word16* speech,
- const WebRtc_UWord16 nrOfSamples);
-
- WebRtc_Word32 GetLevel();
-private:
- WebRtc_Word32 _max;
- WebRtc_UWord32 _count;
- WebRtc_UWord32 _currentLevel;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_LEVEL_INDICATOR_H_
diff --git a/trunk/src/modules/audio_conference_mixer/source/memory_pool.h b/trunk/src/modules/audio_conference_mixer/source/memory_pool.h
deleted file mode 100644
index caf5d93..0000000
--- a/trunk/src/modules/audio_conference_mixer/source/memory_pool.h
+++ /dev/null
@@ -1,122 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_H_
-
-#include <assert.h>
-
-#include "typedefs.h"
-
-#if _WIN32
-#include "memory_pool_win.h"
-#else
-#include "memory_pool_posix.h"
-#endif
-
-namespace webrtc {
-
-template<class MemoryType>
-class MemoryPool
-{
-public:
- // Factory method, constructor disabled.
- static WebRtc_Word32 CreateMemoryPool(MemoryPool*& memoryPool,
- WebRtc_UWord32 initialPoolSize);
-
- // Try to delete the memory pool. Fail with return value -1 if there is
- // outstanding memory.
- static WebRtc_Word32 DeleteMemoryPool(
- MemoryPool*& memoryPool);
-
- // Get/return unused memory.
- WebRtc_Word32 PopMemory(MemoryType*& memory);
- WebRtc_Word32 PushMemory(MemoryType*& memory);
-private:
- MemoryPool(WebRtc_Word32 initialPoolSize);
- ~MemoryPool();
-
- MemoryPoolImpl<MemoryType>* _ptrImpl;
-};
-
-template<class MemoryType>
-MemoryPool<MemoryType>::MemoryPool(WebRtc_Word32 initialPoolSize)
-{
- _ptrImpl = new MemoryPoolImpl<MemoryType>(initialPoolSize);
-}
-
-template<class MemoryType>
-MemoryPool<MemoryType>::~MemoryPool()
-{
- delete _ptrImpl;
-}
-
-template<class MemoryType> WebRtc_Word32
-MemoryPool<MemoryType>::CreateMemoryPool(MemoryPool*& memoryPool,
- WebRtc_UWord32 initialPoolSize)
-{
- memoryPool = new MemoryPool(initialPoolSize);
- if(memoryPool == NULL)
- {
- return -1;
- }
- if(memoryPool->_ptrImpl == NULL)
- {
- delete memoryPool;
- memoryPool = NULL;
- return -1;
- }
- if(!memoryPool->_ptrImpl->Initialize())
- {
- delete memoryPool;
- memoryPool = NULL;
- return -1;
- }
- return 0;
-}
-
-template<class MemoryType>
-WebRtc_Word32 MemoryPool<MemoryType>::DeleteMemoryPool(MemoryPool*& memoryPool)
-{
- if(memoryPool == NULL)
- {
- return -1;
- }
- if(memoryPool->_ptrImpl == NULL)
- {
- return -1;
- }
- if(memoryPool->_ptrImpl->Terminate() == -1)
- {
- return -1;
- }
- delete memoryPool;
- memoryPool = NULL;
- return 0;
-}
-
-template<class MemoryType>
-WebRtc_Word32 MemoryPool<MemoryType>::PopMemory(MemoryType*& memory)
-{
- return _ptrImpl->PopMemory(memory);
-}
-
-template<class MemoryType>
-WebRtc_Word32 MemoryPool<MemoryType>::PushMemory(MemoryType*& memory)
-{
- if(memory == NULL)
- {
- return -1;
- }
- return _ptrImpl->PushMemory(memory);
-}
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_H_
diff --git a/trunk/src/modules/audio_conference_mixer/source/memory_pool_posix.h b/trunk/src/modules/audio_conference_mixer/source/memory_pool_posix.h
deleted file mode 100644
index 45f800b..0000000
--- a/trunk/src/modules/audio_conference_mixer/source/memory_pool_posix.h
+++ /dev/null
@@ -1,168 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_GENERIC_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_GENERIC_H_
-
-#include <assert.h>
-
-#include "critical_section_wrapper.h"
-#include "list_wrapper.h"
-#include "typedefs.h"
-
-namespace webrtc {
-template<class MemoryType>
-class MemoryPoolImpl
-{
-public:
- // MemoryPool functions.
- WebRtc_Word32 PopMemory(MemoryType*& memory);
- WebRtc_Word32 PushMemory(MemoryType*& memory);
-
- MemoryPoolImpl(WebRtc_Word32 initialPoolSize);
- ~MemoryPoolImpl();
-
- // Atomic functions
- WebRtc_Word32 Terminate();
- bool Initialize();
-private:
- // Non-atomic function.
- WebRtc_Word32 CreateMemory(WebRtc_UWord32 amountToCreate);
-
- CriticalSectionWrapper* _crit;
-
- bool _terminate;
-
- ListWrapper _memoryPool;
-
- WebRtc_UWord32 _initialPoolSize;
- WebRtc_UWord32 _createdMemory;
- WebRtc_UWord32 _outstandingMemory;
-};
-
-template<class MemoryType>
-MemoryPoolImpl<MemoryType>::MemoryPoolImpl(WebRtc_Word32 initialPoolSize)
- : _crit(CriticalSectionWrapper::CreateCriticalSection()),
- _terminate(false),
- _memoryPool(),
- _initialPoolSize(initialPoolSize),
- _createdMemory(0),
- _outstandingMemory(0)
-{
-}
-
-template<class MemoryType>
-MemoryPoolImpl<MemoryType>::~MemoryPoolImpl()
-{
- // Trigger assert if there is outstanding memory.
- assert(_createdMemory == 0);
- assert(_outstandingMemory == 0);
- delete _crit;
-}
-
-template<class MemoryType>
-WebRtc_Word32 MemoryPoolImpl<MemoryType>::PopMemory(MemoryType*& memory)
-{
- CriticalSectionScoped cs(_crit);
- if(_terminate)
- {
- memory = NULL;
- return -1;
- }
- ListItem* item = _memoryPool.First();
- if(item == NULL)
- {
- // _memoryPool empty create new memory.
- CreateMemory(_initialPoolSize);
- item = _memoryPool.First();
- if(item == NULL)
- {
- memory = NULL;
- return -1;
- }
- }
- memory = static_cast<MemoryType*>(item->GetItem());
- _memoryPool.Erase(item);
- _outstandingMemory++;
- return 0;
-}
-
-template<class MemoryType>
-WebRtc_Word32 MemoryPoolImpl<MemoryType>::PushMemory(MemoryType*& memory)
-{
- if(memory == NULL)
- {
- return -1;
- }
- CriticalSectionScoped cs(_crit);
- _outstandingMemory--;
- if(_memoryPool.GetSize() > (_initialPoolSize << 1))
- {
- // Reclaim memory if less than half of the pool is unused.
- _createdMemory--;
- delete memory;
- memory = NULL;
- return 0;
- }
- _memoryPool.PushBack(static_cast<void*>(memory));
- memory = NULL;
- return 0;
-}
-
-template<class MemoryType>
-bool MemoryPoolImpl<MemoryType>::Initialize()
-{
- CriticalSectionScoped cs(_crit);
- return CreateMemory(_initialPoolSize) == 0;
-}
-
-template<class MemoryType>
-WebRtc_Word32 MemoryPoolImpl<MemoryType>::Terminate()
-{
- CriticalSectionScoped cs(_crit);
- assert(_createdMemory == _outstandingMemory + _memoryPool.GetSize());
-
- _terminate = true;
- // Reclaim all memory.
- while(_createdMemory > 0)
- {
- ListItem* item = _memoryPool.First();
- if(item == NULL)
- {
- // There is memory that hasn't been returned yet.
- return -1;
- }
- MemoryType* memory = static_cast<MemoryType*>(item->GetItem());
- delete memory;
- _memoryPool.Erase(item);
- _createdMemory--;
- }
- return 0;
-}
-
-template<class MemoryType>
-WebRtc_Word32 MemoryPoolImpl<MemoryType>::CreateMemory(
- WebRtc_UWord32 amountToCreate)
-{
- for(WebRtc_UWord32 i = 0; i < amountToCreate; i++)
- {
- MemoryType* memory = new MemoryType();
- if(memory == NULL)
- {
- return -1;
- }
- _memoryPool.PushBack(static_cast<void*>(memory));
- _createdMemory++;
- }
- return 0;
-}
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_GENERIC_H_
diff --git a/trunk/src/modules/audio_conference_mixer/source/memory_pool_win.h b/trunk/src/modules/audio_conference_mixer/source/memory_pool_win.h
deleted file mode 100644
index 1275ca1..0000000
--- a/trunk/src/modules/audio_conference_mixer/source/memory_pool_win.h
+++ /dev/null
@@ -1,199 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_WINDOWS_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_WINDOWS_H_
-
-#include <assert.h>
-#include <windows.h>
-
-#include "aligned_malloc.h"
-#include "atomic32_wrapper.h"
-#include "typedefs.h"
-
-namespace webrtc {
-template<class MemoryType> struct MemoryPoolItem;
-
-template<class MemoryType>
-struct MemoryPoolItemPayload
-{
- MemoryPoolItemPayload()
- : memoryType(),
- base(NULL)
- {
- }
- MemoryType memoryType;
- MemoryPoolItem<MemoryType>* base;
-};
-
-template<class MemoryType>
-struct MemoryPoolItem
-{
- // Atomic single linked list entry header.
- SLIST_ENTRY itemEntry;
- // Atomic single linked list payload.
- MemoryPoolItemPayload<MemoryType>* payload;
-};
-
-template<class MemoryType>
-class MemoryPoolImpl
-{
-public:
- // MemoryPool functions.
- WebRtc_Word32 PopMemory(MemoryType*& memory);
- WebRtc_Word32 PushMemory(MemoryType*& memory);
-
- MemoryPoolImpl(WebRtc_Word32 /*initialPoolSize*/);
- ~MemoryPoolImpl();
-
- // Atomic functions.
- WebRtc_Word32 Terminate();
- bool Initialize();
-private:
- // Non-atomic function.
- MemoryPoolItem<MemoryType>* CreateMemory();
-
- // Windows implementation of single linked atomic list, documented here:
- // http://msdn.microsoft.com/en-us/library/ms686962(VS.85).aspx
-
- // Atomic single linked list head.
- PSLIST_HEADER _pListHead;
-
- Atomic32Wrapper _createdMemory;
- Atomic32Wrapper _outstandingMemory;
-};
-
-template<class MemoryType>
-MemoryPoolImpl<MemoryType>::MemoryPoolImpl(
- WebRtc_Word32 /*initialPoolSize*/)
- : _pListHead(NULL),
- _createdMemory(0),
- _outstandingMemory(0)
-{
-}
-
-template<class MemoryType>
-MemoryPoolImpl<MemoryType>::~MemoryPoolImpl()
-{
- Terminate();
- if(_pListHead != NULL)
- {
- AlignedFree(reinterpret_cast<void*>(_pListHead));
- _pListHead = NULL;
- }
- // Trigger assert if there is outstanding memory.
- assert(_createdMemory.Value() == 0);
- assert(_outstandingMemory.Value() == 0);
-}
-
-template<class MemoryType>
-WebRtc_Word32 MemoryPoolImpl<MemoryType>::PopMemory(MemoryType*& memory)
-{
- PSLIST_ENTRY pListEntry = InterlockedPopEntrySList(_pListHead);
- if(pListEntry == NULL)
- {
- MemoryPoolItem<MemoryType>* item = CreateMemory();
- if(item == NULL)
- {
- return -1;
- }
- pListEntry = &(item->itemEntry);
- }
- ++_outstandingMemory;
- memory = &((MemoryPoolItem<MemoryType>*)pListEntry)->payload->memoryType;
- return 0;
-}
-
-template<class MemoryType>
-WebRtc_Word32 MemoryPoolImpl<MemoryType>::PushMemory(MemoryType*& memory)
-{
- if(memory == NULL)
- {
- return -1;
- }
-
- MemoryPoolItem<MemoryType>* item =
- ((MemoryPoolItemPayload<MemoryType>*)memory)->base;
-
- const WebRtc_Word32 usedItems = --_outstandingMemory;
- const WebRtc_Word32 totalItems = _createdMemory.Value();
- const WebRtc_Word32 freeItems = totalItems - usedItems;
- if(freeItems < 0)
- {
- assert(false);
- delete item->payload;
- AlignedFree(item);
- return -1;
- }
- if(freeItems >= totalItems>>1)
- {
- delete item->payload;
- AlignedFree(item);
- --_createdMemory;
- return 0;
- }
- InterlockedPushEntrySList(_pListHead,&(item->itemEntry));
- return 0;
-}
-
-template<class MemoryType>
-bool MemoryPoolImpl<MemoryType>::Initialize()
-{
- _pListHead = (PSLIST_HEADER)AlignedMalloc(sizeof(SLIST_HEADER),
- MEMORY_ALLOCATION_ALIGNMENT);
- if(_pListHead == NULL)
- {
- return false;
- }
- InitializeSListHead(_pListHead);
- return true;
-}
-
-template<class MemoryType>
-WebRtc_Word32 MemoryPoolImpl<MemoryType>::Terminate()
-{
- WebRtc_Word32 itemsFreed = 0;
- PSLIST_ENTRY pListEntry = InterlockedPopEntrySList(_pListHead);
- while(pListEntry != NULL)
- {
- MemoryPoolItem<MemoryType>* item = ((MemoryPoolItem<MemoryType>*)pListEntry);
- delete item->payload;
- AlignedFree(item);
- --_createdMemory;
- itemsFreed++;
- pListEntry = InterlockedPopEntrySList(_pListHead);
- }
- return itemsFreed;
-}
-
-template<class MemoryType>
-MemoryPoolItem<MemoryType>* MemoryPoolImpl<MemoryType>::CreateMemory()
-{
- MemoryPoolItem<MemoryType>* returnValue = (MemoryPoolItem<MemoryType>*)
- AlignedMalloc(sizeof(MemoryPoolItem<MemoryType>),
- MEMORY_ALLOCATION_ALIGNMENT);
- if(returnValue == NULL)
- {
- return NULL;
- }
-
- returnValue->payload = new MemoryPoolItemPayload<MemoryType>();
- if(returnValue->payload == NULL)
- {
- delete returnValue;
- return NULL;
- }
- returnValue->payload->base = returnValue;
- ++_createdMemory;
- return returnValue;
-}
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_MEMORY_POOL_WINDOWS_H_
diff --git a/trunk/src/modules/audio_conference_mixer/source/time_scheduler.cc b/trunk/src/modules/audio_conference_mixer/source/time_scheduler.cc
deleted file mode 100644
index 183005e..0000000
--- a/trunk/src/modules/audio_conference_mixer/source/time_scheduler.cc
+++ /dev/null
@@ -1,102 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "critical_section_wrapper.h"
-#include "time_scheduler.h"
-
-namespace webrtc {
-TimeScheduler::TimeScheduler(const WebRtc_UWord32 periodicityInMs)
- : _crit(CriticalSectionWrapper::CreateCriticalSection()),
- _isStarted(false),
- _lastPeriodMark(),
- _periodicityInMs(periodicityInMs),
- _periodicityInTicks(TickTime::MillisecondsToTicks(periodicityInMs)),
- _missedPeriods(0)
- {
- }
-
-TimeScheduler::~TimeScheduler()
-{
- delete _crit;
-}
-
-WebRtc_Word32 TimeScheduler::UpdateScheduler()
-{
- CriticalSectionScoped cs(_crit);
- if(!_isStarted)
- {
- _isStarted = true;
- _lastPeriodMark = TickTime::Now();
- return 0;
- }
- // Don't perform any calculations until the debt of pending periods have
- // been worked off.
- if(_missedPeriods > 0)
- {
- _missedPeriods--;
- return 0;
- }
-
- // Calculate the time that has past since previous call to this function.
- TickTime tickNow = TickTime::Now();
- TickInterval amassedTicks = tickNow - _lastPeriodMark;
- WebRtc_Word64 amassedMs = amassedTicks.Milliseconds();
-
- // Calculate the number of periods the time that has passed correspond to.
- WebRtc_Word32 periodsToClaim = (WebRtc_Word32)amassedMs /
- ((WebRtc_Word32)_periodicityInMs);
-
- // One period will be worked off by this call. Make sure that the number of
- // pending periods don't end up being negative (e.g. if this function is
- // called to often).
- if(periodsToClaim < 1)
- {
- periodsToClaim = 1;
- }
-
- // Update the last period mark without introducing any drifting.
- // Note that if this fuunction is called to often _lastPeriodMark can
- // refer to a time in the future which in turn will yield TimeToNextUpdate
- // that is greater than the periodicity
- for(WebRtc_Word32 i = 0; i < periodsToClaim; i++)
- {
- _lastPeriodMark += _periodicityInTicks;
- }
-
- // Update the total amount of missed periods note that we have processed
- // one period hence the - 1
- _missedPeriods += periodsToClaim - 1;
- return 0;
-}
-
-WebRtc_Word32 TimeScheduler::TimeToNextUpdate(
- WebRtc_Word32& updateTimeInMS) const
-{
- CriticalSectionScoped cs(_crit);
- // Missed periods means that the next UpdateScheduler() should happen
- // immediately.
- if(_missedPeriods > 0)
- {
- updateTimeInMS = 0;
- return 0;
- }
-
- // Calculate the time (in ms) that has past since last call to
- // UpdateScheduler()
- TickTime tickNow = TickTime::Now();
- TickInterval ticksSinceLastUpdate = tickNow - _lastPeriodMark;
- const WebRtc_Word32 millisecondsSinceLastUpdate =
- (WebRtc_Word32) ticksSinceLastUpdate.Milliseconds();
-
- updateTimeInMS = _periodicityInMs - millisecondsSinceLastUpdate;
- updateTimeInMS = (updateTimeInMS < 0) ? 0 : updateTimeInMS;
- return 0;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_conference_mixer/source/time_scheduler.h b/trunk/src/modules/audio_conference_mixer/source/time_scheduler.h
deleted file mode 100644
index e2674d9..0000000
--- a/trunk/src/modules/audio_conference_mixer/source/time_scheduler.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// The TimeScheduler class keeps track of periodic events. It is non-drifting
-// and keeps track of any missed periods so that it is possible to catch up.
-// (compare to a metronome)
-
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
-
-#include "tick_util.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class TimeScheduler
-{
-public:
- TimeScheduler(const WebRtc_UWord32 periodicityInMs);
- ~TimeScheduler();
-
- // Signal that a periodic event has been triggered.
- WebRtc_Word32 UpdateScheduler();
-
- // Set updateTimeInMs to the amount of time until UpdateScheduler() should
- // be called. This time will never be negative.
- WebRtc_Word32 TimeToNextUpdate(WebRtc_Word32& updateTimeInMS) const;
-
-private:
- CriticalSectionWrapper* _crit;
-
- bool _isStarted;
- TickTime _lastPeriodMark;
-
- WebRtc_UWord32 _periodicityInMs;
- WebRtc_Word64 _periodicityInTicks;
- WebRtc_UWord32 _missedPeriods;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_SOURCE_TIME_SCHEDULER_H_
diff --git a/trunk/src/modules/audio_conference_mixer/test/FunctionTest/functionTest.cc b/trunk/src/modules/audio_conference_mixer/test/FunctionTest/functionTest.cc
deleted file mode 100644
index e34a3cb..0000000
--- a/trunk/src/modules/audio_conference_mixer/test/FunctionTest/functionTest.cc
+++ /dev/null
@@ -1,1098 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <assert.h>
-#include <iostream>
-#include <time.h>
-
-#include "functionTest.h"
-#include "event_wrapper.h"
-#include "trace.h"
-#include "thread_wrapper.h"
-#include "webrtc_vad.h"
-
-#if (defined(WEBRTC_LINUX) || defined(WEBRTC_MAC))
- #include <sys/stat.h>
- #define MY_PERMISSION_MASK S_IRWXU | S_IRWXG | S_IRWXO
- #define MKDIR(directory) mkdir(directory,MY_PERMISSION_MASK)
-#else // defined(WINDOWS)
- #include <direct.h>
- #define MKDIR(directory) mkdir(directory)
-#endif
-
-int main(int /*argc*/, char* /*argv[]*/)
-{
- // Initialize random number generator
- //unsigned int seed = 1220716312; // just a seed that can be used
- unsigned int seed = (unsigned)time( NULL );
- srand(seed);
- std::cout << "Starting function test. Seed = " << seed << std::endl;
- std::cout << "Press enter to continue" << std::endl;
- getchar();
- MixerWrapper* testInstance1 = MixerWrapper::CreateMixerWrapper();
- MixerWrapper* testInstance2 = MixerWrapper::CreateMixerWrapper();
- if((testInstance1 == NULL) ||
- (testInstance2 == NULL))
- {
- assert(false);
- return 0;
- }
-
- char versionString[256] = "";
- WebRtc_UWord32 remainingBufferInBytes = 256;
- WebRtc_UWord32 position = 0;
- AudioConferenceMixer::GetVersion(versionString,remainingBufferInBytes,position);
-
- int read = 1;
- while(read != 0)
- {
- std::cout << versionString << std::endl;
- std::cout << "--------Menu-----------" << std::endl;
- std::cout << std::endl;
- std::cout << "0. Quit" << std::endl;
- std::cout << "2. StartMixing" << std::endl;
- std::cout << "3. StopMixing" << std::endl;
- std::cout << "4. Create participant(s)" << std::endl;
- std::cout << "5. Delete participant(s)" << std::endl;
- std::cout << "6. List participants " << std::endl;
- std::cout << "7. Print mix status " << std::endl;
- std::cout << "8. Run identical scenario:" << std::endl;
- std::cout << " a. 1 VIP, 3 regular, amount of mixed = 3" << std::endl;
- std::cout << " b. 1 anonymous, 3 regular, amount of mixed = 2" << std::endl;
- scanf("%i",&read);
- getchar();
- MixerParticipant::ParticipantType participantType;
- int option = 0;
- WebRtc_UWord32 id = 0;
- ListItem* item = NULL;
- ListWrapper participants;
- if(read == 0)
- {
- // donothing
- }
- else if(read == 1)
- {
- }
- else if(read == 2)
- {
- testInstance1->StartMixing();
- }
- else if(read == 3)
- {
- testInstance1->StopMixing();
- }
- else if(read == 4)
- {
- while(true)
- {
- std::cout << "VIP(music) = " << MixerParticipant::VIP << std::endl;
- std::cout << "Regular(speech) = " << MixerParticipant::REGULAR << std::endl;
- std::cout << "Anonymous(music) = " << MixerParticipant::MIXED_ANONYMOUS << std::endl;
- std::cout << "Select type of participant: ";
- scanf("%i",&option);
- if(option == MixerParticipant::VIP ||
- option == MixerParticipant::REGULAR ||
- option == MixerParticipant::MIXED_ANONYMOUS)
- {
- break;
- }
- }
- participantType = (MixerParticipant::ParticipantType)option;
- testInstance1->CreateParticipant(participantType);
- }
- else if(read == 5)
- {
- std::cout << "Select participant to delete: ";
- scanf("%i",&option);
- id = option;
- testInstance1->DeleteParticipant(id);
- break;
- }
- else if(read == 6)
- {
- testInstance1->GetParticipantList(participants);
- item = participants.First();
- std::cout << "The following participants have been created: " << std::endl;
- while(item)
- {
- WebRtc_UWord32 id = item->GetUnsignedItem();
- std::cout << id;
- item = participants.Next(item);
- if(item != NULL)
- {
- std::cout << ", ";
- }
- else
- {
- std::cout << std::endl;
- }
- }
- }
- else if(read == 7)
- {
- std::cout << "-------------Mixer Status-------------" << std::endl;
- testInstance1->PrintStatus();
- testInstance2->PrintStatus();
- std::cout << "Press enter to continue";
- getchar();
- std::cout << std::endl;
- std::cout << std::endl;
- }
- else if(read == 8)
- {
- const WebRtc_Word32 amountOfParticipants = 4;
- MixerParticipant::ParticipantType instance1Participants[] =
- {MixerParticipant::VIP,
- MixerParticipant::REGULAR,
- MixerParticipant::REGULAR,
- MixerParticipant::REGULAR};
- MixerParticipant::ParticipantType instance2Participants[] =
- {MixerParticipant::MIXED_ANONYMOUS,
- MixerParticipant::REGULAR,
- MixerParticipant::REGULAR,
- MixerParticipant::REGULAR};
- for(WebRtc_Word32 i = 0; i < amountOfParticipants; i++)
- {
- WebRtc_Word32 startPosition = 0;
- GenerateRandomPosition(startPosition);
- testInstance1->CreateParticipant(instance1Participants[i],startPosition);
- testInstance2->CreateParticipant(instance2Participants[i],startPosition);
- }
- bool success = true;
- success = testInstance1->StartMixing();
- assert(success);
- success = testInstance2->StartMixing(2);
- assert(success);
- }
- }
-
- std::cout << "Press enter to stop" << std::endl;
- getchar();
- delete testInstance1;
- delete testInstance2;
- return 0;
-}
-
-FileWriter::FileWriter()
- :
- _file(NULL)
-{
-}
-
-FileWriter::~FileWriter()
-{
- if(_file)
- {
- fclose(_file);
- }
-}
-
-bool
-FileWriter::SetFileName(
- const char* fileName)
-{
- if(_file)
- {
- fclose(_file);
- }
- _file = fopen(fileName,"wb");
- return _file != NULL;
-}
-
-bool
-FileWriter::WriteToFile(
- const AudioFrame& audioFrame)
-{
- WebRtc_Word32 written = (WebRtc_Word32)fwrite(audioFrame._payloadData,sizeof(WebRtc_Word16),audioFrame._payloadDataLengthInSamples,_file);
- // Do not flush buffers since that will add (a lot of) delay
- return written == audioFrame._payloadDataLengthInSamples;
-}
-
-FileReader::FileReader()
- :
- _frequency(kDefaultFrequency),
- _sampleSize((_frequency*kProcessPeriodicityInMs)/1000),
- _timeStamp(0),
- _file(NULL),
- _vadInstr(NULL),
- _automaticVad(false),
- _vad(false)
-{
- if(WebRtcVad_Create(&_vadInstr) == 0)
- {
- if(WebRtcVad_Init(_vadInstr) != 0)
- {
- assert(false);
- WebRtcVad_Free(_vadInstr);
- _vadInstr = NULL;
- }
- }
- else
- {
- assert(false);
- }
-}
-
-FileReader::~FileReader()
-{
- if(_file)
- {
- fclose(_file);
- }
- if(_vadInstr)
- {
- WebRtcVad_Free(_vadInstr);
- }
-}
-
-bool
-FileReader::SetFileName(
- const char* fileName)
-{
- if(_file)
- {
- fclose(_file);
- }
- _file = fopen(fileName,"rb");
- return _file != NULL;
-}
-
-bool
-FileReader::ReadFromFile(
- AudioFrame& audioFrame)
-{
-
- WebRtc_Word16 buffer[AudioFrame::kMaxAudioFrameSizeSamples];
- LoopedFileRead(buffer,AudioFrame::kMaxAudioFrameSizeSamples,_sampleSize,_file);
-
- bool vad = false;
- GetVAD(buffer,_sampleSize,vad);
- AudioFrame::VADActivity activity = vad ? AudioFrame::kVadActive :
- AudioFrame::kVadPassive;
-
- _volumeCalculator.ComputeLevel(buffer,_sampleSize);
- const WebRtc_Word32 level = _volumeCalculator.GetLevel();
- return audioFrame.UpdateFrame( -1,
- _timeStamp,
- buffer,
- _sampleSize,
- _frequency,
- AudioFrame::kNormalSpeech,
- activity,
- 0,
- level) == 0;
-
-}
-
-bool
-FileReader::FastForwardFile(
- const WebRtc_Word32 samples)
-{
- WebRtc_Word16* tempBuffer = new WebRtc_Word16[samples];
- bool success = LoopedFileRead(tempBuffer,samples,samples,_file);
- delete[] tempBuffer;
- return success;
-}
-
-bool
-FileReader::EnableAutomaticVAD(
- bool enable,
- int mode)
-{
- if(!_automaticVad &&
- enable)
- {
- if(WebRtcVad_Init(_vadInstr) == -1)
- {
- return false;
- }
- }
- WebRtcVad_set_mode(_vadInstr,mode);
- _automaticVad = enable;
- return true;
-}
-
-bool
-FileReader::SetVAD(
- bool vad)
-{
- if(_automaticVad)
- {
- return false;
- }
- _vad = vad;
- return true;
-}
-
-bool
-FileReader::GetVAD(
- WebRtc_Word16* buffer,
- WebRtc_UWord8 bufferLengthInSamples,
- bool& vad)
-{
- if(_automaticVad)
- {
- WebRtc_Word16 result = WebRtcVad_Process(_vadInstr,_frequency,buffer,bufferLengthInSamples);
- if(result == -1)
- {
- assert(false);
- return false;
- }
- _vad = vad = (result == 1);
- }
- vad = _vad;
- return true;
-}
-
-MixerParticipant*
-MixerParticipant::CreateParticipant(
- const WebRtc_UWord32 id,
- ParticipantType participantType,
- const WebRtc_Word32 startPosition,
- char* outputPath)
-{
- if(participantType == RANDOM)
- {
- participantType = (ParticipantType)(rand() % 3);
- }
- MixerParticipant* participant = new MixerParticipant(id,participantType);
- // Randomize the start position so we only need one input file
- // assume file is smaller than 1 minute wideband = 60 * 16000
- // Always start at a multiple of 10ms wideband
- if(!participant->InitializeFileReader(startPosition) ||
- !participant->InitializeFileWriter(outputPath))
- {
- delete participant;
- return NULL;
- }
- return participant;
-}
-
-MixerParticipant::MixerParticipant(
- const WebRtc_UWord32 id,
- ParticipantType participantType)
- :
- _id(id),
- _participantType(participantType),
- _fileReader(),
- _fileWriter()
-{
-}
-
-MixerParticipant::~MixerParticipant()
-{
-}
-
-WebRtc_Word32
-MixerParticipant::GetAudioFrame(
- const WebRtc_Word32 /*id*/,
- AudioFrame& audioFrame)
-{
- if(!_fileReader.ReadFromFile(audioFrame))
- {
- return -1;
- }
- audioFrame._id = _id;
- return 0;
-}
-
-WebRtc_Word32
-MixerParticipant::MixedAudioFrame(
- const AudioFrame& audioFrame)
-{
- return _fileWriter.WriteToFile(audioFrame);
-}
-
-WebRtc_Word32
-MixerParticipant::GetParticipantType(
- ParticipantType& participantType)
-{
- participantType = _participantType;
- return 0;
-}
-
-bool
-MixerParticipant::InitializeFileReader(
- const WebRtc_Word32 startPositionInSamples)
-{
- char fileName[128] = "";
- if(_participantType == REGULAR)
- {
- sprintf(fileName,"convFile.pcm");
- }
- else
- {
- sprintf(fileName,"musicFile.pcm");
- }
- if(!_fileReader.SetFileName(fileName))
- {
- return false;
- }
- if(!_fileReader.EnableAutomaticVAD(true,2))
- {
- assert(false);
- }
- return _fileReader.FastForwardFile(startPositionInSamples);
-}
-
-bool
-MixerParticipant::InitializeFileWriter(
- char* outputPath)
-{
- const WebRtc_Word32 stringsize = 128;
- char fileName[stringsize] = "";
- strncpy(fileName,outputPath,stringsize);
- fileName[stringsize-1] = '\0';
-
- char tempName[stringsize];
- tempName[0] = '\0';
- sprintf(tempName,"outputFile%d.pcm",(int)_id);
- strncat(fileName,tempName,(stringsize - strlen(fileName)));
- fileName[stringsize-1] = '\0';
-
- return _fileWriter.SetFileName(fileName);
-}
-
-StatusReceiver::StatusReceiver(
- const WebRtc_Word32 id)
- :
- _id(id),
- _mixedParticipants(NULL),
- _mixedParticipantsAmount(0),
- _mixedParticipantsSize(0),
- _vadPositiveParticipants(NULL),
- _vadPositiveParticipantsAmount(0),
- _vadPositiveParticipantsSize(0),
- _mixedAudioLevel(0)
-{
-}
-
-StatusReceiver::~StatusReceiver()
-{
- delete[] _mixedParticipants;
- delete[] _vadPositiveParticipants;
-}
-
-void
-StatusReceiver::MixedParticipants(
- const WebRtc_Word32 id,
- const ParticipantStatistics* participantStatistics,
- const WebRtc_UWord32 size)
-{
- if(id != _id)
- {
- assert(false);
- }
- if(_mixedParticipantsSize < size)
- {
- delete[] _mixedParticipants;
- _mixedParticipantsSize = size;
- _mixedParticipants = new ParticipantStatistics[size];
- }
- _mixedParticipantsAmount = size;
- memcpy(_mixedParticipants,participantStatistics,sizeof(ParticipantStatistics)*size);
-}
-
-void
-StatusReceiver::VADPositiveParticipants(
- const WebRtc_Word32 id,
- const ParticipantStatistics* participantStatistics,
- const WebRtc_UWord32 size)
-{
- if(id != _id)
- {
- assert(false);
- }
-
- if(_vadPositiveParticipantsSize < size)
- {
- delete[] _vadPositiveParticipants;
- _vadPositiveParticipantsSize = size;
- _vadPositiveParticipants = new ParticipantStatistics[size];
- }
- _vadPositiveParticipantsAmount = size;
- memcpy(_vadPositiveParticipants,participantStatistics,sizeof(ParticipantStatistics)*size);
-}
-
-void
-StatusReceiver::MixedAudioLevel(
- const WebRtc_Word32 id,
- const WebRtc_UWord32 level)
-{
- if(id != _id)
- {
- assert(false);
- }
- _mixedAudioLevel = level;
-}
-
-void
-StatusReceiver::PrintMixedParticipants()
-{
- std::cout << "Mixed participants" << std::endl;
- if(_mixedParticipantsAmount == 0)
- {
- std::cout << "N/A" << std::endl;
- }
- for(WebRtc_UWord16 i = 0; i < _mixedParticipantsAmount; i++)
- {
- std::cout << i + 1 << ". Participant " << _mixedParticipants[i].participant << ": level = " << _mixedParticipants[i].level << std::endl;
- }
-}
-
-void
-StatusReceiver::PrintVadPositiveParticipants()
-{
- std::cout << "VAD positive participants" << std::endl;
- if(_mixedParticipantsAmount == 0)
- {
- std::cout << "N/A" << std::endl;
- }
- for(WebRtc_UWord16 i = 0; i < _mixedParticipantsAmount; i++)
- {
- std::cout << i + 1 << ". Participant " << _mixedParticipants[i].participant << ": level = " << _mixedParticipants[i].level << std::endl;
- }
-}
-
-void
-StatusReceiver::PrintMixedAudioLevel()
-{
- std::cout << "Mixed audio level = " << _mixedAudioLevel << std::endl;
-}
-
-WebRtc_Word32 MixerWrapper::_mixerWrapperIdCounter = 0;
-
-MixerWrapper::MixerWrapper()
- :
- _processThread(NULL),
- _threadId(0),
- _firstProcessCall(true),
- _previousTime(),
- _periodicityInTicks(TickTime::MillisecondsToTicks(FileReader::kProcessPeriodicityInMs)),
- _synchronizationEvent(EventWrapper::Create()),
- _freeItemIds(),
- _itemIdCounter(0),
- _mixerParticipants(),
- _mixerWrappererId(_mixerWrapperIdCounter++),
- _instanceOutputPath(),
- _trace(NULL),
- _statusReceiver(_mixerWrappererId),
- _generalAudioWriter()
-{
- sprintf(_instanceOutputPath,"instance%d/",(int)_mixerWrappererId);
- MKDIR(_instanceOutputPath);
- _mixer = AudioConferenceMixer::CreateAudioConferenceMixer(
- _mixerWrappererId);
- if(_mixer != NULL)
- {
- bool success = true;
-
- success = _mixer->RegisterMixedStreamCallback(*this) == 0;
- assert(success);
- success = _mixer->RegisterMixedStreamCallback(*this) == -1;
- assert(success);
- success = _mixer->UnRegisterMixedStreamCallback() == 0;
- assert(success);
- success = _mixer->UnRegisterMixedStreamCallback() == -1;
- assert(success);
- success = _mixer->RegisterMixedStreamCallback(*this) == 0;
- assert(success);
-
- success = _mixer->RegisterMixerStatusCallback(_statusReceiver,2) == 0;
- assert(success);
- success = _mixer->RegisterMixerStatusCallback(_statusReceiver,1) == -1;
- assert(success);
- success = _mixer->UnRegisterMixerStatusCallback() == 0;
- assert(success);
- success = _mixer->UnRegisterMixerStatusCallback() == -1;
- assert(success);
- success = _mixer->RegisterMixerStatusCallback(_statusReceiver,1) == 0;
- assert(success);
- }
- else
- {
- assert(false);
- std::cout << "Failed to create mixer instance";
- }
-}
-
-MixerWrapper*
-MixerWrapper::CreateMixerWrapper()
-{
- MixerWrapper* mixerWrapper = new MixerWrapper();
- if(!mixerWrapper->InitializeFileWriter())
- {
- delete mixerWrapper;
- return NULL;
- }
- return mixerWrapper;
-}
-
-MixerWrapper::~MixerWrapper()
-{
- StopMixing();
- ClearAllItemIds();
- _synchronizationEvent->StopTimer();
- delete _synchronizationEvent;
- delete _mixer;
-}
-
-bool
-MixerWrapper::CreateParticipant(
- MixerParticipant::ParticipantType participantType)
-{
- WebRtc_Word32 startPosition = 0;
- GenerateRandomPosition(startPosition);
- return CreateParticipant(participantType,startPosition);
-}
-
-bool
-MixerWrapper::CreateParticipant(
- MixerParticipant::ParticipantType participantType,
- const WebRtc_Word32 startPosition)
-{
- WebRtc_UWord32 id;
- if(!GetFreeItemIds(id))
- {
- return false;
- }
-
- MixerParticipant* participant = MixerParticipant::CreateParticipant(id,participantType,startPosition,_instanceOutputPath);
- if(!participant)
- {
- return false;
- }
- if(_mixerParticipants.Insert(id,static_cast<void*>(participant)) != 0)
- {
- delete participant;
- return false;
- }
- if(!StartMixingParticipant(id))
- {
- DeleteParticipant(id);
- return false;
- }
- return true;
-}
-
-bool
-MixerWrapper::DeleteParticipant(
- const WebRtc_UWord32 id)
-{
- bool success = StopMixingParticipant(id);
- if(!success)
- {
- assert(false);
- return false;
- }
- MapItem* item = _mixerParticipants.Find(id);
- if(item == NULL)
- {
- return false;
- }
- MixerParticipant* participant = static_cast<MixerParticipant*>(item->GetItem());
- delete participant;
- _mixerParticipants.Erase(item);
- AddFreeItemIds(id);
- return true;
-}
-
-bool
-MixerWrapper::StartMixing(
- const WebRtc_UWord32 mixedParticipants)
-{
- if(_processThread)
- {
- return false;
- }
- if(_mixer->SetAmountOfMixedParticipants(mixedParticipants) != 0)
- {
- assert(false);
- }
- WebRtc_UWord32 mixedParticipantsTest = 0;
- _mixer->AmountOfMixedParticipants(mixedParticipantsTest);
- assert(mixedParticipantsTest == mixedParticipants);
-
- if(!_synchronizationEvent->StartTimer(true,10))
- {
- assert(false);
- return false;
- }
- _processThread = ThreadWrapper::CreateThread(Process, this, kLowPriority);
- if(!_processThread->Start(_threadId))
- {
- delete _processThread;
- _processThread = NULL;
- assert(false);
- return false;
- }
-
- return true;
-}
-
-bool
-MixerWrapper::StopMixing()
-{
- while(_processThread &&
- !_processThread->Stop())
- {}
- _synchronizationEvent->StopTimer();
-
- delete _processThread;
- _processThread = NULL;
- return true;
-}
-
-void
-MixerWrapper::NewMixedAudio(
- const WebRtc_Word32 id,
- const AudioFrame& generalAudioFrame,
- const AudioFrame** uniqueAudioFrames,
- const WebRtc_UWord32 size)
-{
- if(id < 0)
- {
- assert(false);
- }
- // Store the general audio
- _generalAudioWriter.WriteToFile(generalAudioFrame);
-
- // Send the unique audio frames to its corresponding participants
- ListWrapper uniqueAudioFrameList;
- for(WebRtc_UWord32 i = 0; i < size; i++)
- {
- WebRtc_UWord32 id = (uniqueAudioFrames[i])->_id;
- MapItem* resultItem = _mixerParticipants.Find(id);
- if(resultItem == NULL)
- {
- assert(false);
- continue;
- }
- MixerParticipant* participant = static_cast<MixerParticipant*>(resultItem->GetItem());
- participant->MixedAudioFrame(*(uniqueAudioFrames[i]));
- uniqueAudioFrameList.PushBack(resultItem->GetItem());
- }
-
- // Send the general audio frames to the remaining participants
- MapItem* item = _mixerParticipants.First();
- while(item)
- {
- bool isUnique = false;
- ListItem* compareItem = uniqueAudioFrameList.First();
- while(compareItem)
- {
- if(compareItem->GetItem() == item->GetItem())
- {
- isUnique = true;
- break;
- }
- compareItem = uniqueAudioFrameList.Next(compareItem);
- }
- if(!isUnique)
- {
- MixerParticipant* participant = static_cast<MixerParticipant*>(item->GetItem());
- participant->MixedAudioFrame(generalAudioFrame);
- }
- item = _mixerParticipants.Next(item);
- }
-}
-
-bool
-MixerWrapper::GetParticipantList(
- ListWrapper& participants)
-{
- MapItem* item = _mixerParticipants.First();
- while(item)
- {
- participants.PushBack(item->GetId());
- item = _mixerParticipants.Next(item);
- }
- return true;
-}
-
-void
-MixerWrapper::PrintStatus()
-{
- std::cout << "instance " << _mixerWrappererId << std::endl;
- std::cout << std::endl;
- _statusReceiver.PrintMixedParticipants();
- std::cout << std::endl;
- _statusReceiver.PrintVadPositiveParticipants();
- std::cout << std::endl;
- _statusReceiver.PrintMixedAudioLevel();
- std::cout << "---------------------------------------" << std::endl;
-}
-
-bool
-MixerWrapper::InitializeFileWriter()
-{
- const WebRtc_Word32 stringsize = 128;
- char fileName[stringsize] = "";
- strncpy(fileName,_instanceOutputPath,stringsize);
- fileName[stringsize-1] = '\0';
-
- strncat(fileName,"generalOutputFile.pcm",(stringsize - strlen(fileName)));
- fileName[stringsize-1] = '\0';
- return _generalAudioWriter.SetFileName(fileName);
-}
-
-bool
-MixerWrapper::Process(
- void* instance)
-{
- MixerWrapper* mixerWrapper = static_cast<MixerWrapper*>(instance);
- return mixerWrapper->Process();
-}
-
-bool
-MixerWrapper::Process()
-{
- switch(_synchronizationEvent->Wait(1000))
- {
- case kEventSignaled:
- // Normal operation, ~10 ms has passed
- break;
- case kEventError:
- // Error occured end the thread and throw an assertion
- assert(false);
- return false;
- case kEventTimeout:
- // One second has passed without a timeout something is wrong
- // end the thread and throw an assertion
- assert(false);
- return false;
- }
- WebRtc_Word32 processOfset = 0;
- const TickTime currentTime = TickTime::Now();
- if(_firstProcessCall)
- {
- _previousTime = TickTime::Now();
- _firstProcessCall = false;
- }
- else
- {
- TickInterval deltaTime = (currentTime - _previousTime);
- _previousTime += _periodicityInTicks;
- processOfset = (WebRtc_Word32) deltaTime.Milliseconds();
- processOfset -= FileReader::kProcessPeriodicityInMs;
- }
-
- _mixer->Process();
- WebRtc_Word32 timeUntilNextProcess = _mixer->TimeUntilNextProcess();
- if(processOfset > FileReader::kProcessPeriodicityInMs)
- {
- std::cout << "Performance Warning: Process running " << processOfset << " too slow" << std::endl;
- _previousTime = currentTime;
- if(timeUntilNextProcess > 0)
- {
- std::cout << "Performance Warning: test performance and module performance missmatch" << std::endl;
- }
- }
- else if(processOfset < -FileReader::kProcessPeriodicityInMs)
- {
- std::cout << "Performance Warning: Process running " << -processOfset << " too fast" << std::endl;
- _previousTime = currentTime;
- if(timeUntilNextProcess < FileReader::kProcessPeriodicityInMs)
- {
- std::cout << "Performance Warning: test performance and module performance missmatch" << std::endl;
- }
- }
- return true;
-}
-
-
-bool
-MixerWrapper::StartMixingParticipant(
- const WebRtc_UWord32 id)
-{
- MapItem* item = _mixerParticipants.Find(id);
- if(item == NULL)
- {
- return false;
- }
- MixerParticipant* participant = static_cast<MixerParticipant*>(item->GetItem());
- MixerParticipant::ParticipantType participantType = MixerParticipant::REGULAR;
- participant->GetParticipantType(participantType);
- if(participantType == MixerParticipant::MIXED_ANONYMOUS)
- {
- bool anonymouslyMixed = false;
- bool success = _mixer->SetAnonymousMixabilityStatus(*participant,true) == 0;
- assert(success);
- success = _mixer->AnonymousMixabilityStatus(*participant,anonymouslyMixed) == 0;
- assert(success);
- assert(anonymouslyMixed);
- success = _mixer->SetAnonymousMixabilityStatus(*participant,true) == -1;
- assert(success);
- success = _mixer->SetAnonymousMixabilityStatus(*participant,false) == 0;
- assert(success);
- success = _mixer->AnonymousMixabilityStatus(*participant,anonymouslyMixed) == 0;
- assert(success);
- assert(!anonymouslyMixed);
- success = _mixer->SetAnonymousMixabilityStatus(*participant,false) == -1;
- assert(success);
- success = _mixer->SetAnonymousMixabilityStatus(*participant,true) == 0;
- assert(success);
- success = _mixer->AnonymousMixabilityStatus(*participant,anonymouslyMixed) == 0;
- assert(success);
- assert(anonymouslyMixed);
- return success;
- }
- WebRtc_UWord32 previousAmountOfMixableParticipants = 0;
- bool success = _mixer->AmountOfMixables(previousAmountOfMixableParticipants) == 0;
- assert(success);
-
- success = _mixer->SetMixabilityStatus(*participant,true) == 0;
- assert(success);
- success = _mixer->SetMixabilityStatus(*participant,true) == -1;
- assert(success);
- success = _mixer->SetMixabilityStatus(*participant,false) == 0;
- assert(success);
- success = _mixer->SetMixabilityStatus(*participant,false) == -1;
- assert(success);
- success = _mixer->SetMixabilityStatus(*participant,true) == 0;
- assert(success);
- if(!success)
- {
- return false;
- }
-
- WebRtc_UWord32 currentAmountOfMixableParticipants = 0;
- success = _mixer->AmountOfMixables(currentAmountOfMixableParticipants) == 0;
- assert(currentAmountOfMixableParticipants == previousAmountOfMixableParticipants + 1);
-
- bool mixable = true;
- success = _mixer->MixabilityStatus(*participant,mixable) == 0;
- assert(success);
- assert(mixable);
- if(participantType == MixerParticipant::REGULAR)
- {
- return true;
- }
- bool IsVIP = false;
- success = _mixer->SetVIPStatus(*participant,true) == 0;
- assert(success);
- success = _mixer->VIPStatus(*participant,IsVIP) == 0;
- assert(success);
- assert(IsVIP);
- success = _mixer->SetVIPStatus(*participant,true) == -1;
- assert(success);
- success = _mixer->SetVIPStatus(*participant,false) == 0;
- assert(success);
- success = _mixer->VIPStatus(*participant,IsVIP) == 0;
- assert(success);
- assert(!IsVIP);
- success = _mixer->SetVIPStatus(*participant,false) == -1;
- assert(success);
- success = _mixer->SetVIPStatus(*participant,true) == 0;
- assert(success);
- success = _mixer->VIPStatus(*participant,IsVIP) == 0;
- assert(success);
- assert(IsVIP);
- assert(success);
- return success;
-}
-
-bool
-MixerWrapper::StopMixingParticipant(
- const WebRtc_UWord32 id)
-{
- MapItem* item = _mixerParticipants.Find(id);
- if(item == NULL)
- {
- return false;
- }
- MixerParticipant* participant = static_cast<MixerParticipant*>(item->GetItem());
- bool success = false;
- WebRtc_UWord32 previousAmountOfMixableParticipants = 0;
- success = _mixer->AmountOfMixables(previousAmountOfMixableParticipants) == 0;
- assert(success);
- success = _mixer->SetMixabilityStatus(*participant,false) == 0;
- assert(success);
- WebRtc_UWord32 currentAmountOfMixableParticipants = 0;
- success = _mixer->AmountOfMixables(currentAmountOfMixableParticipants) == 0;
- assert(success);
- assert(success ? currentAmountOfMixableParticipants == previousAmountOfMixableParticipants -1 :
- currentAmountOfMixableParticipants == previousAmountOfMixableParticipants);
- return success;
-}
-
-bool
-MixerWrapper::GetFreeItemIds(
- WebRtc_UWord32& itemId)
-{
- if(!_freeItemIds.Empty())
- {
- ListItem* item = _freeItemIds.First();
- WebRtc_UWord32* id = static_cast<WebRtc_UWord32*>(item->GetItem());
- itemId = *id;
- delete id;
- return true;
- }
- if(_itemIdCounter == (WebRtc_UWord32) -1)
- {
- return false;
- }
- itemId = _itemIdCounter++;
- return true;
-}
-
-void
-MixerWrapper::AddFreeItemIds(
- const WebRtc_UWord32 itemId)
-{
- WebRtc_UWord32* id = new WebRtc_UWord32;
- *id = itemId;
- _freeItemIds.PushBack(static_cast<void*>(id));
-}
-
-void
-MixerWrapper::ClearAllItemIds()
-{
- ListItem* item = _freeItemIds.First();
- while(item != NULL)
- {
- WebRtc_UWord32* id = static_cast<WebRtc_UWord32*>(item->GetItem());
- delete id;
- _freeItemIds.Erase(item);
- item = _freeItemIds.First();
- }
-}
-
-bool
-LoopedFileRead(
- WebRtc_Word16* buffer,
- WebRtc_UWord32 bufferSizeInSamples,
- WebRtc_UWord32 samplesToRead,
- FILE* file)
-{
- if(bufferSizeInSamples < samplesToRead)
- {
- return false;
- }
- WebRtc_UWord32 gottenSamples = (WebRtc_UWord32)fread(buffer,sizeof(WebRtc_Word16),samplesToRead,file);
- if(gottenSamples != samplesToRead)
- {
- WebRtc_UWord32 missingSamples = samplesToRead - gottenSamples;
- fseek(file,0,0);
- gottenSamples += (WebRtc_UWord32)fread(&buffer[gottenSamples],sizeof(WebRtc_Word16),missingSamples,file);
- }
- if(gottenSamples != samplesToRead)
- {
- return false;
- }
- return true;
-}
-
-void
-GenerateRandomPosition(
- WebRtc_Word32& startPosition)
-{
- startPosition = (rand() % (60*16000/160)) * 160;
-}
diff --git a/trunk/src/modules/audio_conference_mixer/test/FunctionTest/functionTest.h b/trunk/src/modules/audio_conference_mixer/test/FunctionTest/functionTest.h
deleted file mode 100644
index f25e5f1..0000000
--- a/trunk/src/modules/audio_conference_mixer/test/FunctionTest/functionTest.h
+++ /dev/null
@@ -1,276 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_TEST_FUNCTIONTEST_FUNCTIONTEST_H_
-#define WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_TEST_FUNCTIONTEST_FUNCTIONTEST_H_
-
-#include "module_common_types.h"
-#include "level_indicator.h"
-#include "list_wrapper.h"
-#include "map_wrapper.h"
-#include "audio_conference_mixer.h"
-#include "audio_conference_mixer_defines.h"
-#include "tick_util.h"
-
-namespace webrtc {
-class EventWrapper;
-class ThreadWrapper;
-class Trace;
-}
-struct WebRtcVadInst;
-
-class FileWriter
-{
-public:
- FileWriter();
- ~FileWriter();
-
- bool SetFileName(
- const char* fileName);
-
- bool WriteToFile(
- const AudioFrame& audioFrame);
-private:
- FILE* _file;
-};
-
-class FileReader
-{
-public:
- enum {kProcessPeriodicityInMs = 10};
- enum Frequency
- {
- kNbInHz = 8000,
- kWbInHz = 16000,
- kDefaultFrequency = kWbInHz
- };
-
- FileReader();
- ~FileReader();
-
- bool SetFileName(
- const char* fileName);
-
- bool ReadFromFile(
- AudioFrame& audioFrame);
-
- bool FastForwardFile(
- const WebRtc_Word32 samples);
-
- bool EnableAutomaticVAD(
- bool enable,
- int mode);
-
- bool SetVAD(
- bool vad);
-private:
- bool GetVAD(
- WebRtc_Word16* buffer,
- WebRtc_UWord8 bufferLengthInSamples,
- bool& vad);
-
- Frequency _frequency;
- WebRtc_UWord8 _sampleSize;
-
- WebRtc_UWord32 _timeStamp;
-
- FILE* _file;
-
- WebRtcVadInst* _vadInstr;
- bool _automaticVad;
- bool _vad;
-
- LevelIndicator _volumeCalculator;
-};
-
-class MixerParticipant : public MixerParticipant
-{
-public:
- enum ParticipantType
- {
- VIP = 0,
- REGULAR = 1,
- MIXED_ANONYMOUS = 2,
- RANDOM = 3
- };
-
- static MixerParticipant* CreateParticipant(
- const WebRtc_UWord32 id,
- ParticipantType participantType,
- const WebRtc_Word32 startPosition,
- char* outputPath);
- ~MixerParticipant();
-
- WebRtc_Word32 GetAudioFrame(
- const WebRtc_Word32 id,
- AudioFrame& audioFrame);
-
- WebRtc_Word32 MixedAudioFrame(
- const AudioFrame& audioFrame);
-
- WebRtc_Word32 GetParticipantType(
- ParticipantType& participantType);
-private:
- MixerParticipant(
- const WebRtc_UWord32 id,
- ParticipantType participantType);
-
- bool InitializeFileReader(
- const WebRtc_Word32 startPositionInSamples);
-
- bool InitializeFileWriter(
- char* outputPath);
-
- WebRtc_UWord32 _id;
- ParticipantType _participantType;
-
- FileReader _fileReader;
- FileWriter _fileWriter;
-};
-
-class StatusReceiver : public AudioMixerStatusReceiver
-{
-public:
- StatusReceiver(
- const WebRtc_Word32 id);
- ~StatusReceiver();
-
- void MixedParticipants(
- const WebRtc_Word32 id,
- const ParticipantStatistics* participantStatistics,
- const WebRtc_UWord32 size);
-
- void VADPositiveParticipants(
- const WebRtc_Word32 id,
- const ParticipantStatistics* participantStatistics,
- const WebRtc_UWord32 size);
-
- void MixedAudioLevel(
- const WebRtc_Word32 id,
- const WebRtc_UWord32 level);
-
- void PrintMixedParticipants();
-
- void PrintVadPositiveParticipants();
-
- void PrintMixedAudioLevel();
-private:
- WebRtc_Word32 _id;
-
- ParticipantStatistics* _mixedParticipants;
- WebRtc_UWord32 _mixedParticipantsAmount;
- WebRtc_UWord32 _mixedParticipantsSize;
-
- ParticipantStatistics* _vadPositiveParticipants;
- WebRtc_UWord32 _vadPositiveParticipantsAmount;
- WebRtc_UWord32 _vadPositiveParticipantsSize;
-
- WebRtc_UWord32 _mixedAudioLevel;
-};
-
-class MixerWrapper : public AudioMixerOutputReceiver
-{
-public:
- static MixerWrapper* CreateMixerWrapper();
- ~MixerWrapper();
-
- bool SetMixFrequency(
- const AudioConferenceMixer::Frequency frequency);
-
- bool CreateParticipant(
- MixerParticipant::ParticipantType participantType);
-
- bool CreateParticipant(
- MixerParticipant::ParticipantType participantType,
- const WebRtc_Word32 startPosition);
-
- bool DeleteParticipant(
- const WebRtc_UWord32 id);
-
- bool StartMixing(
- const WebRtc_UWord32 mixedParticipants = AudioConferenceMixer::kDefaultAmountOfMixedParticipants);
-
- bool StopMixing();
-
- void NewMixedAudio(
- const WebRtc_Word32 id,
- const AudioFrame& generalAudioFrame,
- const AudioFrame** uniqueAudioFrames,
- const WebRtc_UWord32 size);
-
- bool GetParticipantList(
- ListWrapper& participants);
-
- void PrintStatus();
-private:
- MixerWrapper();
-
- bool InitializeFileWriter();
-
- static bool Process(
- void* instance);
-
- bool Process();
-
- bool StartMixingParticipant(
- const WebRtc_UWord32 id);
-
- bool StopMixingParticipant(
- const WebRtc_UWord32 id);
-
- bool GetFreeItemIds(
- WebRtc_UWord32& itemId);
-
- void AddFreeItemIds(
- const WebRtc_UWord32 itemId);
-
- void ClearAllItemIds();
-
- webrtc::ThreadWrapper* _processThread;
- unsigned int _threadId;
-
- // Performance hooks
- enum{WARNING_COUNTER = 100};
-
- bool _firstProcessCall;
- TickTime _previousTime; // Tick time of previous process
- const WebRtc_Word64 _periodicityInTicks; // Periodicity
-
- webrtc::EventWrapper* _synchronizationEvent;
-
- ListWrapper _freeItemIds;
- WebRtc_UWord32 _itemIdCounter;
-
- MapWrapper _mixerParticipants;
-
- static WebRtc_Word32 _mixerWrapperIdCounter;
- WebRtc_Word32 _mixerWrappererId;
- char _instanceOutputPath[128];
-
- webrtc::Trace* _trace;
- AudioConferenceMixer* _mixer;
-
- StatusReceiver _statusReceiver;
-
- FileWriter _generalAudioWriter;
-};
-
-bool
-LoopedFileRead(
- WebRtc_Word16* buffer,
- WebRtc_UWord32 bufferSizeInSamples,
- WebRtc_UWord32 samplesToRead,
- FILE* file);
-
-void
-GenerateRandomPosition(
- WebRtc_Word32& startPosition);
-
-#endif // WEBRTC_MODULES_AUDIO_CONFERENCE_MIXER_TEST_FUNCTIONTEST_FUNCTIONTEST_H_
diff --git a/trunk/src/modules/audio_device/OWNERS b/trunk/src/modules/audio_device/OWNERS
deleted file mode 100644
index a07ced3..0000000
--- a/trunk/src/modules/audio_device/OWNERS
+++ /dev/null
@@ -1,4 +0,0 @@
-henrikg@webrtc.org
-henrika@webrtc.org
-niklas.enbom@webrtc.org
-xians@webrtc.org
diff --git a/trunk/src/modules/audio_device/main/interface/audio_device.h b/trunk/src/modules/audio_device/main/interface/audio_device.h
deleted file mode 100644
index 1149055..0000000
--- a/trunk/src/modules/audio_device/main/interface/audio_device.h
+++ /dev/null
@@ -1,207 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
-#define MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
-
-#include "module.h"
-#include "audio_device_defines.h"
-
-namespace webrtc {
-
-class AudioDeviceModule : public RefCountedModule {
- public:
- enum ErrorCode {
- kAdmErrNone = 0,
- kAdmErrArgument = 1
- };
-
- enum AudioLayer {
- kPlatformDefaultAudio = 0,
- kWindowsWaveAudio = 1,
- kWindowsCoreAudio = 2,
- kLinuxAlsaAudio = 3,
- kLinuxPulseAudio = 4,
- kDummyAudio = 5
- };
-
- enum WindowsDeviceType {
- kDefaultCommunicationDevice = -1,
- kDefaultDevice = -2
- };
-
- enum BufferType {
- kFixedBufferSize = 0,
- kAdaptiveBufferSize = 1
- };
-
- enum ChannelType {
- kChannelLeft = 0,
- kChannelRight = 1,
- kChannelBoth = 2
- };
-
- public:
- // Retrieve the currently utilized audio layer
- virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const = 0;
-
- // Error handling
- virtual ErrorCode LastError() const = 0;
- virtual int32_t RegisterEventObserver(AudioDeviceObserver* eventCallback) = 0;
-
- // Full-duplex transportation of PCM audio
- virtual int32_t RegisterAudioCallback(AudioTransport* audioCallback) = 0;
-
- // Main initialization and termination
- virtual int32_t Init() = 0;
- virtual int32_t Terminate() = 0;
- virtual bool Initialized() const = 0;
-
- // Device enumeration
- virtual int16_t PlayoutDevices() = 0;
- virtual int16_t RecordingDevices() = 0;
- virtual int32_t PlayoutDeviceName(uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) = 0;
- virtual int32_t RecordingDeviceName(uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) = 0;
-
- // Device selection
- virtual int32_t SetPlayoutDevice(uint16_t index) = 0;
- virtual int32_t SetPlayoutDevice(WindowsDeviceType device) = 0;
- virtual int32_t SetRecordingDevice(uint16_t index) = 0;
- virtual int32_t SetRecordingDevice(WindowsDeviceType device) = 0;
-
- // Audio transport initialization
- virtual int32_t PlayoutIsAvailable(bool* available) = 0;
- virtual int32_t InitPlayout() = 0;
- virtual bool PlayoutIsInitialized() const = 0;
- virtual int32_t RecordingIsAvailable(bool* available) = 0;
- virtual int32_t InitRecording() = 0;
- virtual bool RecordingIsInitialized() const = 0;
-
- // Audio transport control
- virtual int32_t StartPlayout() = 0;
- virtual int32_t StopPlayout() = 0;
- virtual bool Playing() const = 0;
- virtual int32_t StartRecording() = 0;
- virtual int32_t StopRecording() = 0;
- virtual bool Recording() const = 0;
-
- // Microphone Automatic Gain Control (AGC)
- virtual int32_t SetAGC(bool enable) = 0;
- virtual bool AGC() const = 0;
-
- // Volume control based on the Windows Wave API (Windows only)
- virtual int32_t SetWaveOutVolume(uint16_t volumeLeft,
- uint16_t volumeRight) = 0;
- virtual int32_t WaveOutVolume(uint16_t* volumeLeft,
- uint16_t* volumeRight) const = 0;
-
- // Audio mixer initialization
- virtual int32_t SpeakerIsAvailable(bool* available) = 0;
- virtual int32_t InitSpeaker() = 0;
- virtual bool SpeakerIsInitialized() const = 0;
- virtual int32_t MicrophoneIsAvailable(bool* available) = 0;
- virtual int32_t InitMicrophone() = 0;
- virtual bool MicrophoneIsInitialized() const = 0;
-
- // Speaker volume controls
- virtual int32_t SpeakerVolumeIsAvailable(bool* available) = 0;
- virtual int32_t SetSpeakerVolume(uint32_t volume) = 0;
- virtual int32_t SpeakerVolume(uint32_t* volume) const = 0;
- virtual int32_t MaxSpeakerVolume(uint32_t* maxVolume) const = 0;
- virtual int32_t MinSpeakerVolume(uint32_t* minVolume) const = 0;
- virtual int32_t SpeakerVolumeStepSize(uint16_t* stepSize) const = 0;
-
- // Microphone volume controls
- virtual int32_t MicrophoneVolumeIsAvailable(bool* available) = 0;
- virtual int32_t SetMicrophoneVolume(uint32_t volume) = 0;
- virtual int32_t MicrophoneVolume(uint32_t* volume) const = 0;
- virtual int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const = 0;
- virtual int32_t MinMicrophoneVolume(uint32_t* minVolume) const = 0;
- virtual int32_t MicrophoneVolumeStepSize(uint16_t* stepSize) const = 0;
-
- // Speaker mute control
- virtual int32_t SpeakerMuteIsAvailable(bool* available) = 0;
- virtual int32_t SetSpeakerMute(bool enable) = 0;
- virtual int32_t SpeakerMute(bool* enabled) const = 0;
-
- // Microphone mute control
- virtual int32_t MicrophoneMuteIsAvailable(bool* available) = 0;
- virtual int32_t SetMicrophoneMute(bool enable) = 0;
- virtual int32_t MicrophoneMute(bool* enabled) const = 0;
-
- // Microphone boost control
- virtual int32_t MicrophoneBoostIsAvailable(bool* available) = 0;
- virtual int32_t SetMicrophoneBoost(bool enable) = 0;
- virtual int32_t MicrophoneBoost(bool* enabled) const = 0;
-
- // Stereo support
- virtual int32_t StereoPlayoutIsAvailable(bool* available) const = 0;
- virtual int32_t SetStereoPlayout(bool enable) = 0;
- virtual int32_t StereoPlayout(bool* enabled) const = 0;
- virtual int32_t StereoRecordingIsAvailable(bool* available) const = 0;
- virtual int32_t SetStereoRecording(bool enable) = 0;
- virtual int32_t StereoRecording(bool* enabled) const = 0;
- virtual int32_t SetRecordingChannel(const ChannelType channel) = 0;
- virtual int32_t RecordingChannel(ChannelType* channel) const = 0;
-
- // Delay information and control
- virtual int32_t SetPlayoutBuffer(const BufferType type,
- uint16_t sizeMS = 0) = 0;
- virtual int32_t PlayoutBuffer(BufferType* type, uint16_t* sizeMS) const = 0;
- virtual int32_t PlayoutDelay(uint16_t* delayMS) const = 0;
- virtual int32_t RecordingDelay(uint16_t* delayMS) const = 0;
-
- // CPU load
- virtual int32_t CPULoad(uint16_t* load) const = 0;
-
- // Recording of raw PCM data
- virtual int32_t StartRawOutputFileRecording(
- const char pcmFileNameUTF8[kAdmMaxFileNameSize]) = 0;
- virtual int32_t StopRawOutputFileRecording() = 0;
- virtual int32_t StartRawInputFileRecording(
- const char pcmFileNameUTF8[kAdmMaxFileNameSize]) = 0;
- virtual int32_t StopRawInputFileRecording() = 0;
-
- // Native sample rate controls (samples/sec)
- virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) = 0;
- virtual int32_t RecordingSampleRate(uint32_t* samplesPerSec) const = 0;
- virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) = 0;
- virtual int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const = 0;
-
- // Mobile device specific functions
- virtual int32_t ResetAudioDevice() = 0;
- virtual int32_t SetLoudspeakerStatus(bool enable) = 0;
- virtual int32_t GetLoudspeakerStatus(bool* enabled) const = 0;
-
- // *Experimental - not recommended for use.*
- // Enables the Windows Core Audio built-in AEC. Fails on other platforms.
- //
- // Must be called before InitRecording(). When enabled:
- // 1. StartPlayout() must be called before StartRecording().
- // 2. StopRecording() should be called before StopPlayout().
- // The reverse order may cause garbage audio to be rendered or the
- // capture side to halt until StopRecording() is called.
- virtual int32_t EnableBuiltInAEC(bool enable) { return -1; }
- virtual bool BuiltInAECIsEnabled() const { return false; }
-
- protected:
- virtual ~AudioDeviceModule() {};
-};
-
-AudioDeviceModule* CreateAudioDeviceModule(
- WebRtc_Word32 id, AudioDeviceModule::AudioLayer audioLayer);
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_MAIN_INTERFACE_AUDIO_DEVICE_H_
diff --git a/trunk/src/modules/audio_device/main/interface/audio_device_defines.h b/trunk/src/modules/audio_device/main/interface/audio_device_defines.h
deleted file mode 100644
index 67db65d..0000000
--- a/trunk/src/modules/audio_device/main/interface/audio_device_defines.h
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
-
-#include "typedefs.h"
-
-namespace webrtc {
-
-static const int kAdmMaxDeviceNameSize = 128;
-static const int kAdmMaxFileNameSize = 512;
-static const int kAdmMaxGuidSize = 128;
-
-static const int kAdmMinPlayoutBufferSizeMs = 10;
-static const int kAdmMaxPlayoutBufferSizeMs = 250;
-
-// ----------------------------------------------------------------------------
-// AudioDeviceObserver
-// ----------------------------------------------------------------------------
-
-class AudioDeviceObserver
-{
-public:
- enum ErrorCode
- {
- kRecordingError = 0,
- kPlayoutError = 1
- };
- enum WarningCode
- {
- kRecordingWarning = 0,
- kPlayoutWarning = 1
- };
-
- virtual void OnErrorIsReported(const ErrorCode error) = 0;
- virtual void OnWarningIsReported(const WarningCode warning) = 0;
-
-protected:
- virtual ~AudioDeviceObserver() {}
-};
-
-// ----------------------------------------------------------------------------
-// AudioTransport
-// ----------------------------------------------------------------------------
-
-class AudioTransport
-{
-public:
- virtual int32_t RecordedDataIsAvailable(const char* audioSamples,
- const uint32_t nSamples,
- const uint8_t nBytesPerSample,
- const uint8_t nChannels,
- const uint32_t samplesPerSec,
- const uint32_t totalDelayMS,
- const int32_t clockDrift,
- const uint32_t currentMicLevel,
- uint32_t& newMicLevel) = 0;
-
- virtual int32_t NeedMorePlayData(const uint32_t nSamples,
- const uint8_t nBytesPerSample,
- const uint8_t nChannels,
- const uint32_t samplesPerSec,
- char* audioSamples,
- uint32_t& nSamplesOut) = 0;
-
-protected:
- virtual ~AudioTransport() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DEFINES_H
diff --git a/trunk/src/modules/audio_device/main/source/Android.mk b/trunk/src/modules/audio_device/main/source/Android.mk
deleted file mode 100644
index affa5e1..0000000
--- a/trunk/src/modules/audio_device/main/source/Android.mk
+++ /dev/null
@@ -1,57 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_audio_device
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := \
- audio_device_buffer.cc \
- audio_device_generic.cc \
- audio_device_utility.cc \
- audio_device_impl.cc \
- android/audio_device_android_opensles.cc \
- android/audio_device_utility_android.cc \
- dummy/audio_device_utility_dummy.cc \
- dummy/audio_device_dummy.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS) \
- '-DWEBRTC_ANDROID_OPENSLES'
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/android \
- $(LOCAL_PATH)/dummy \
- $(LOCAL_PATH)/linux \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../../.. \
- $(LOCAL_PATH)/../../../interface \
- $(LOCAL_PATH)/../../../../common_audio/resampler/include \
- $(LOCAL_PATH)/../../../../common_audio/signal_processing/include \
- $(LOCAL_PATH)/../../../../system_wrappers/interface \
- system/media/wilhelm/include
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport \
- libOpenSLES
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_device/main/source/android/audio_device_android_jni.cc b/trunk/src/modules/audio_device/main/source/android/audio_device_android_jni.cc
deleted file mode 100644
index c275171..0000000
--- a/trunk/src/modules/audio_device/main/source/android/audio_device_android_jni.cc
+++ /dev/null
@@ -1,2920 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Android audio device implementation (JNI/AudioTrack/AudioRecord usage)
- */
-
-// TODO(xians): Break out attach and detach current thread to JVM to
-// separate functions.
-
-#include <stdlib.h>
-#include "audio_device_utility.h"
-#include "audio_device_android_jni.h"
-#include "audio_device_config.h"
-
-#include "trace.h"
-#include "thread_wrapper.h"
-#include "event_wrapper.h"
-
-// Android logging, uncomment to print trace to logcat instead of
-// trace file/callback
-//#include <android/log.h>
-//#define WEBRTC_TRACE(a,b,c,...) __android_log_print(ANDROID_LOG_DEBUG, \
-// "WebRTC AD jni", __VA_ARGS__)
-
-namespace webrtc
-{
-
-JavaVM* globalJvm = NULL;
-JNIEnv* globalJNIEnv = NULL;
-jobject globalSndContext = NULL;
-jclass globalScClass = NULL;
-
-// ----------------------------------------------------------------------------
-// SetAndroidAudioDeviceObjects
-//
-// Global function for setting Java pointers and creating Java
-// objects that are global to all instances of VoiceEngine used
-// by the same Java application.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 SetAndroidAudioDeviceObjects(void* javaVM, void* env,
- void* context)
-{
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, -1, "%s",
- __FUNCTION__);
-
- globalJvm = (JavaVM*) javaVM;
- globalSndContext = (jobject) context;
-
- if (env)
- {
- globalJNIEnv = (JNIEnv *) env;
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, -1,
- "%s: will find class", __FUNCTION__);
-
- // get java class type (note path to class packet)
- jclass
- javaScClassLocal =
- globalJNIEnv->FindClass(
- "org/webrtc/voiceengine/AudioDeviceAndroid");
- if (!javaScClassLocal)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
- "%s: could not find java class", __FUNCTION__);
- return -1; /* exception thrown */
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, -1,
- "%s: will create global reference", __FUNCTION__);
-
- // create a global reference to the class (to tell JNI that we are
- // referencing it after this function has returned)
- globalScClass
- = reinterpret_cast<jclass> (globalJNIEnv->NewGlobalRef(
- javaScClassLocal));
- if (!globalScClass)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
- "%s: could not create reference", __FUNCTION__);
- return -1;
- }
-
- // Delete local class ref, we only use the global ref
- globalJNIEnv->DeleteLocalRef(javaScClassLocal);
- }
- else // User is resetting the env variable
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1,
- "%s: env is NULL, assuming deinit", __FUNCTION__);
-
- if (!globalJNIEnv)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, -1,
- "%s: saved env already NULL", __FUNCTION__);
- return 0;
- }
-
- globalJNIEnv->DeleteGlobalRef(globalScClass);
- globalJNIEnv = (JNIEnv *) NULL;
- }
-
- return 0;
-}
-
-// ============================================================================
-// Construction & Destruction
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// AudioDeviceAndroidJni - ctor
-// ----------------------------------------------------------------------------
-
-AudioDeviceAndroidJni::AudioDeviceAndroidJni(const WebRtc_Word32 id) :
- _ptrAudioBuffer(NULL),
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _id(id),
- _timeEventRec(*EventWrapper::Create()),
- _timeEventPlay(*EventWrapper::Create()),
- _recStartStopEvent(*EventWrapper::Create()),
- _playStartStopEvent(*EventWrapper::Create()),
- _ptrThreadPlay(NULL),
- _ptrThreadRec(NULL),
- _recThreadID(0),
- _playThreadID(0),
- _playThreadIsInitialized(false),
- _recThreadIsInitialized(false),
- _shutdownPlayThread(false),
- _shutdownRecThread(false),
- // _recBuffer[2*REC_BUF_SIZE_IN_SAMPLES]
- _recordingDeviceIsSpecified(false),
- _playoutDeviceIsSpecified(false), _initialized(false),
- _recording(false), _playing(false), _recIsInitialized(false),
- _playIsInitialized(false), _micIsInitialized(false),
- _speakerIsInitialized(false), _startRec(false), _stopRec(false),
- _startPlay(false), _stopPlay(false), _playWarning(0),
- _playError(0), _recWarning(0), _recError(0), _delayPlayout(0),
- _delayRecording(0),
- _AGC(false),
- _samplingFreqIn(0),
- _samplingFreqOut(0),
- _maxSpeakerVolume(0),
- _loudSpeakerOn(false),
- _recAudioSource(1), // 1 is AudioSource.MIC which is our default
- _javaVM(NULL), _javaContext(NULL), _jniEnvPlay(NULL),
- _jniEnvRec(NULL), _javaScClass(0), _javaScObj(0),
- _javaPlayBuffer(0), _javaRecBuffer(0), _javaDirectPlayBuffer(NULL),
- _javaDirectRecBuffer(NULL), _javaMidPlayAudio(0),
- _javaMidRecAudio(0)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
- "%s created", __FUNCTION__);
-
- memset(_recBuffer, 0, sizeof(_recBuffer));
-}
-
-// ----------------------------------------------------------------------------
-// AudioDeviceAndroidJni - dtor
-// ----------------------------------------------------------------------------
-
-AudioDeviceAndroidJni::~AudioDeviceAndroidJni()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
- "%s destroyed", __FUNCTION__);
-
- Terminate();
-
- delete &_recStartStopEvent;
- delete &_playStartStopEvent;
- delete &_timeEventRec;
- delete &_timeEventPlay;
- delete &_critSect;
-}
-
-// ============================================================================
-// API
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// AttachAudioBuffer
-// ----------------------------------------------------------------------------
-
-void AudioDeviceAndroidJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- _ptrAudioBuffer = audioBuffer;
-
- // inform the AudioBuffer about default settings for this implementation
- _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
- _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
- _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
- _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
-}
-
-// ----------------------------------------------------------------------------
-// ActiveAudioLayer
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::ActiveAudioLayer(
- AudioDeviceModule::AudioLayer& audioLayer) const
-{
-
- audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Init
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::Init()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_initialized)
- {
- return 0;
- }
-
- _playWarning = 0;
- _playError = 0;
- _recWarning = 0;
- _recError = 0;
-
- // Init Java member variables
- // and set up JNI interface to
- // AudioDeviceAndroid java class
- if (InitJavaResources() != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: Failed to init Java resources", __FUNCTION__);
- return -1;
- }
-
- // Check the sample rate to be used for playback and recording
- // and the max playout volume
- if (InitSampleRate() != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: Failed to init samplerate", __FUNCTION__);
- return -1;
- }
-
- // RECORDING
- const char* threadName = "webrtc_jni_audio_capture_thread";
- _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc, this,
- kRealtimePriority, threadName);
- if (_ptrThreadRec == NULL)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " failed to create the rec audio thread");
- return -1;
- }
-
- unsigned int threadID(0);
- if (!_ptrThreadRec->Start(threadID))
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " failed to start the rec audio thread");
- delete _ptrThreadRec;
- _ptrThreadRec = NULL;
- return -1;
- }
- _recThreadID = threadID;
-
- // PLAYOUT
- threadName = "webrtc_jni_audio_render_thread";
- _ptrThreadPlay = ThreadWrapper::CreateThread(PlayThreadFunc, this,
- kRealtimePriority, threadName);
- if (_ptrThreadPlay == NULL)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " failed to create the play audio thread");
- return -1;
- }
-
- threadID = 0;
- if (!_ptrThreadPlay->Start(threadID))
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " failed to start the play audio thread");
- delete _ptrThreadPlay;
- _ptrThreadPlay = NULL;
- return -1;
- }
- _playThreadID = threadID;
-
- _initialized = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Terminate
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::Terminate()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_initialized)
- {
- return 0;
- }
-
- // RECORDING
- StopRecording();
- _shutdownRecThread = true;
- _timeEventRec.Set(); // Release rec thread from waiting state
- if (_ptrThreadRec)
- {
- // First, the thread must detach itself from Java VM
- _critSect.Leave();
- if (kEventSignaled != _recStartStopEvent.Wait(5000))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceAudioDevice,
- _id,
- "%s: Recording thread shutdown timed out, cannot "
- "terminate thread",
- __FUNCTION__);
- // If we close thread anyway, the app will crash
- return -1;
- }
- _recStartStopEvent.Reset();
- _critSect.Enter();
-
- // Close down rec thread
- ThreadWrapper* tmpThread = _ptrThreadRec;
- _ptrThreadRec = NULL;
- _critSect.Leave();
- tmpThread->SetNotAlive();
- // Release again, we might have returned to waiting state
- _timeEventRec.Set();
- if (tmpThread->Stop())
- {
- delete tmpThread;
- _jniEnvRec = NULL;
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " failed to close down the rec audio thread");
- }
- _critSect.Enter();
-
- _recThreadIsInitialized = false;
- }
- _micIsInitialized = false;
- _recordingDeviceIsSpecified = false;
-
- // PLAYOUT
- StopPlayout();
- _shutdownPlayThread = true;
- _timeEventPlay.Set(); // Release rec thread from waiting state
- if (_ptrThreadPlay)
- {
- // First, the thread must detach itself from Java VM
- _critSect.Leave();
- if (kEventSignaled != _playStartStopEvent.Wait(5000))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceAudioDevice,
- _id,
- "%s: Playout thread shutdown timed out, cannot "
- "terminate thread",
- __FUNCTION__);
- // If we close thread anyway, the app will crash
- return -1;
- }
- _playStartStopEvent.Reset();
- _critSect.Enter();
-
- // Close down play thread
- ThreadWrapper* tmpThread = _ptrThreadPlay;
- _ptrThreadPlay = NULL;
- _critSect.Leave();
- tmpThread->SetNotAlive();
- _timeEventPlay.Set();
- if (tmpThread->Stop())
- {
- delete tmpThread;
- _jniEnvPlay = NULL;
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " failed to close down the play audio thread");
- }
- _critSect.Enter();
-
- _playThreadIsInitialized = false;
- }
- _speakerIsInitialized = false;
- _playoutDeviceIsSpecified = false;
-
- // get the JNI env for this thread
- JNIEnv *env;
- bool isAttached = false;
-
- // get the JNI env for this thread
- if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = _javaVM->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: Could not attach thread to JVM (%d, %p)",
- __FUNCTION__, res, env);
- return -1;
- }
- isAttached = true;
- }
-
- // Make method IDs and buffer pointers unusable
- _javaMidPlayAudio = 0;
- _javaMidRecAudio = 0;
- _javaDirectPlayBuffer = NULL;
- _javaDirectRecBuffer = NULL;
-
- // Delete the references to the java buffers, this allows the
- // garbage collector to delete them
- env->DeleteGlobalRef(_javaPlayBuffer);
- _javaPlayBuffer = 0;
- env->DeleteGlobalRef(_javaRecBuffer);
- _javaRecBuffer = 0;
-
- // Delete the references to the java object and class, this allows the
- // garbage collector to delete them
- env->DeleteGlobalRef(_javaScObj);
- _javaScObj = 0;
- _javaScClass = 0;
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (_javaVM->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "%s: Could not detach thread from JVM", __FUNCTION__);
- }
- }
-
- _initialized = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Initialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceAndroidJni::Initialized() const
-{
-
- return (_initialized);
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SpeakerIsAvailable(bool& available)
-{
-
- // We always assume it's available
- available = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitSpeaker
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::InitSpeaker()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_playing)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Playout already started");
- return -1;
- }
-
- if (!_playoutDeviceIsSpecified)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Playout device is not specified");
- return -1;
- }
-
- // Nothing needs to be done here, we use a flag to have consistent
- // behavior with other platforms
- _speakerIsInitialized = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::MicrophoneIsAvailable(bool& available)
-{
-
- // We always assume it's available
- available = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitMicrophone
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::InitMicrophone()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_recording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Recording already started");
- return -1;
- }
-
- if (!_recordingDeviceIsSpecified)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Recording device is not specified");
- return -1;
- }
-
- // Nothing needs to be done here, we use a flag to have consistent
- // behavior with other platforms
- _micIsInitialized = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceAndroidJni::SpeakerIsInitialized() const
-{
-
- return _speakerIsInitialized;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceAndroidJni::MicrophoneIsInitialized() const
-{
-
- return _micIsInitialized;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolumeIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SpeakerVolumeIsAvailable(bool& available)
-{
-
- available = true; // We assume we are always be able to set/get volume
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetSpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SetSpeakerVolume(WebRtc_UWord32 volume)
-{
-
- if (!_speakerIsInitialized)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Speaker not initialized");
- return -1;
- }
- if (!_javaContext)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Context is not set");
- return -1;
- }
-
- // get the JNI env for this thread
- JNIEnv *env;
- bool isAttached = false;
-
- if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = _javaVM->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not attach thread to JVM (%d, %p)", res, env);
- return -1;
- }
- isAttached = true;
- }
-
- // get the method ID
- jmethodID setPlayoutVolumeID = env->GetMethodID(_javaScClass,
- "SetPlayoutVolume", "(I)I");
-
- // call java sc object method
- jint res = env->CallIntMethod(_javaScObj, setPlayoutVolumeID,
- static_cast<int> (volume));
- if (res < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "SetPlayoutVolume failed (%d)", res);
- return -1;
- }
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (_javaVM->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Could not detach thread from JVM");
- }
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SpeakerVolume(WebRtc_UWord32& volume) const
-{
-
- if (!_speakerIsInitialized)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Speaker not initialized");
- return -1;
- }
- if (!_javaContext)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Context is not set");
- return -1;
- }
-
- // get the JNI env for this thread
- JNIEnv *env;
- bool isAttached = false;
-
- if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = _javaVM->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not attach thread to JVM (%d, %p)", res, env);
- return -1;
- }
- isAttached = true;
- }
-
- // get the method ID
- jmethodID getPlayoutVolumeID = env->GetMethodID(_javaScClass,
- "GetPlayoutVolume", "()I");
-
- // call java sc object method
- jint level = env->CallIntMethod(_javaScObj, getPlayoutVolumeID);
- if (level < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "GetPlayoutVolume failed (%d)", level);
- return -1;
- }
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (_javaVM->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Could not detach thread from JVM");
- }
- }
-
- volume = static_cast<WebRtc_UWord32> (level);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetWaveOutVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SetWaveOutVolume(
- WebRtc_UWord16 /*volumeLeft*/,
- WebRtc_UWord16 /*volumeRight*/)
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// WaveOutVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::WaveOutVolume(
- WebRtc_UWord16& /*volumeLeft*/,
- WebRtc_UWord16& /*volumeRight*/) const
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MaxSpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::MaxSpeakerVolume(
- WebRtc_UWord32& maxVolume) const
-{
-
- if (!_speakerIsInitialized)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Speaker not initialized");
- return -1;
- }
-
- maxVolume = _maxSpeakerVolume;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MinSpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::MinSpeakerVolume(
- WebRtc_UWord32& minVolume) const
-{
-
- if (!_speakerIsInitialized)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Speaker not initialized");
- return -1;
- }
-
- minVolume = 0;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolumeStepSize
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SpeakerVolumeStepSize(
- WebRtc_UWord16& stepSize) const
-{
-
- if (!_speakerIsInitialized)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Speaker not initialized");
- return -1;
- }
-
- stepSize = 1;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerMuteIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SpeakerMuteIsAvailable(bool& available)
-{
-
- available = false; // Speaker mute not supported on Android
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetSpeakerMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SetSpeakerMute(bool /*enable*/)
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SpeakerMute(bool& /*enabled*/) const
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneMuteIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::MicrophoneMuteIsAvailable(bool& available)
-{
-
- available = false; // Mic mute not supported on Android
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SetMicrophoneMute(bool /*enable*/)
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::MicrophoneMute(bool& /*enabled*/) const
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneBoostIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::MicrophoneBoostIsAvailable(bool& available)
-{
-
- available = false; // Mic boost not supported on Android
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneBoost
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SetMicrophoneBoost(bool enable)
-{
-
- if (!_micIsInitialized)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Microphone not initialized");
- return -1;
- }
-
- if (enable)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Enabling not available");
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneBoost
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::MicrophoneBoost(bool& enabled) const
-{
-
- if (!_micIsInitialized)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Microphone not initialized");
- return -1;
- }
-
- enabled = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoRecordingIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::StereoRecordingIsAvailable(bool& available)
-{
-
- available = false; // Stereo recording not supported on Android
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetStereoRecording
-//
-// Specifies the number of input channels.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SetStereoRecording(bool enable)
-{
-
- if (enable)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Enabling not available");
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::StereoRecording(bool& enabled) const
-{
-
- enabled = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoPlayoutIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::StereoPlayoutIsAvailable(bool& available)
-{
-
- available = false; // Stereo playout not supported on Android
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetStereoPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SetStereoPlayout(bool enable)
-{
-
- if (enable)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Enabling not available");
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::StereoPlayout(bool& enabled) const
-{
-
- enabled = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetAGC
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SetAGC(bool enable)
-{
-
- _AGC = enable;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// AGC
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceAndroidJni::AGC() const
-{
-
- return _AGC;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolumeIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::MicrophoneVolumeIsAvailable(
- bool& available)
-{
-
- available = false; // Mic volume not supported on Android
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SetMicrophoneVolume(
- WebRtc_UWord32 /*volume*/)
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::MicrophoneVolume(
- WebRtc_UWord32& /*volume*/) const
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MaxMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::MaxMicrophoneVolume(
- WebRtc_UWord32& /*maxVolume*/) const
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MinMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::MinMicrophoneVolume(
- WebRtc_UWord32& /*minVolume*/) const
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolumeStepSize
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::MicrophoneVolumeStepSize(
- WebRtc_UWord16& /*stepSize*/) const
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDevices
-// ----------------------------------------------------------------------------
-
-WebRtc_Word16 AudioDeviceAndroidJni::PlayoutDevices()
-{
-
- // There is one device only
- return 1;
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutDevice I (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SetPlayoutDevice(WebRtc_UWord16 index)
-{
-
- if (_playIsInitialized)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Playout already initialized");
- return -1;
- }
-
- if (0 != index)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Device index is out of range [0,0]");
- return -1;
- }
-
- // Do nothing but set a flag, this is to have consistent behavior
- // with other platforms
- _playoutDeviceIsSpecified = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutDevice II (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SetPlayoutDevice(
- AudioDeviceModule::WindowsDeviceType /*device*/)
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDeviceName
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::PlayoutDeviceName(
- WebRtc_UWord16 index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize])
-{
-
- if (0 != index)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Device index is out of range [0,0]");
- return -1;
- }
-
- // Return empty string
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDeviceName
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::RecordingDeviceName(
- WebRtc_UWord16 index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize])
-{
-
- if (0 != index)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Device index is out of range [0,0]");
- return -1;
- }
-
- // Return empty string
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDevices
-// ----------------------------------------------------------------------------
-
-WebRtc_Word16 AudioDeviceAndroidJni::RecordingDevices()
-{
-
- // There is one device only
- return 1;
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingDevice I (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SetRecordingDevice(WebRtc_UWord16 index)
-{
-
- if (_recIsInitialized)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Recording already initialized");
- return -1;
- }
-
- // Recording device index will be used for specifying recording
- // audio source, allow any value
- _recAudioSource = index;
- _recordingDeviceIsSpecified = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingDevice II (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SetRecordingDevice(
- AudioDeviceModule::WindowsDeviceType /*device*/)
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::PlayoutIsAvailable(bool& available)
-{
-
- available = false;
-
- // Try to initialize the playout side
- WebRtc_Word32 res = InitPlayout();
-
- // Cancel effect of initialization
- StopPlayout();
-
- if (res != -1)
- {
- available = true;
- }
-
- return res;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::RecordingIsAvailable(bool& available)
-{
-
- available = false;
-
- // Try to initialize the playout side
- WebRtc_Word32 res = InitRecording();
-
- // Cancel effect of initialization
- StopRecording();
-
- if (res != -1)
- {
- available = true;
- }
-
- return res;
-}
-
-// ----------------------------------------------------------------------------
-// InitPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::InitPlayout()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_initialized)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Not initialized");
- return -1;
- }
-
- if (_playing)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Playout already started");
- return -1;
- }
-
- if (!_playoutDeviceIsSpecified)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Playout device is not specified");
- return -1;
- }
-
- if (_playIsInitialized)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Playout already initialized");
- return 0;
- }
-
- // Initialize the speaker
- if (InitSpeaker() == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " InitSpeaker() failed");
- }
-
- // get the JNI env for this thread
- JNIEnv *env;
- bool isAttached = false;
-
- // get the JNI env for this thread
- if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- "attaching");
-
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = _javaVM->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not attach thread to JVM (%d, %p)", res, env);
- return -1;
- }
- isAttached = true;
- }
-
- // get the method ID
- jmethodID initPlaybackID = env->GetMethodID(_javaScClass, "InitPlayback",
- "(I)I");
-
- int samplingFreq = 44100;
- if (_samplingFreqOut != 44)
- {
- samplingFreq = _samplingFreqOut * 1000;
- }
-
- int retVal = -1;
-
- // Call java sc object method
- jint res = env->CallIntMethod(_javaScObj, initPlaybackID, samplingFreq);
- if (res < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "InitPlayback failed (%d)", res);
- }
- else
- {
- // Set the audio device buffer sampling rate
- _ptrAudioBuffer->SetPlayoutSampleRate(_samplingFreqOut * 1000);
- _playIsInitialized = true;
- retVal = 0;
- }
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- "detaching");
- if (_javaVM->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Could not detach thread from JVM");
- }
- }
-
- return retVal;
-}
-
-// ----------------------------------------------------------------------------
-// InitRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::InitRecording()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_initialized)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Not initialized");
- return -1;
- }
-
- if (_recording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Recording already started");
- return -1;
- }
-
- if (!_recordingDeviceIsSpecified)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Recording device is not specified");
- return -1;
- }
-
- if (_recIsInitialized)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Recording already initialized");
- return 0;
- }
-
- // Initialize the microphone
- if (InitMicrophone() == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " InitMicrophone() failed");
- }
-
- // get the JNI env for this thread
- JNIEnv *env;
- bool isAttached = false;
-
- // get the JNI env for this thread
- if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = _javaVM->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not attach thread to JVM (%d, %p)", res, env);
- return -1;
- }
- isAttached = true;
- }
-
- // get the method ID
- jmethodID initRecordingID = env->GetMethodID(_javaScClass, "InitRecording",
- "(II)I");
-
- int samplingFreq = 44100;
- if (_samplingFreqIn != 44)
- {
- samplingFreq = _samplingFreqIn * 1000;
- }
-
- int retVal = -1;
-
- // call java sc object method
- jint res = env->CallIntMethod(_javaScObj, initRecordingID, _recAudioSource,
- samplingFreq);
- if (res < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "InitRecording failed (%d)", res);
- }
- else
- {
- // Set the audio device buffer sampling rate
- _ptrAudioBuffer->SetRecordingSampleRate(_samplingFreqIn * 1000);
-
- // the init rec function returns a fixed delay
- _delayRecording = res / _samplingFreqIn;
-
- _recIsInitialized = true;
- retVal = 0;
- }
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (_javaVM->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Could not detach thread from JVM");
- }
- }
-
- return retVal;
-}
-
-// ----------------------------------------------------------------------------
-// StartRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::StartRecording()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_recIsInitialized)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Recording not initialized");
- return -1;
- }
-
- if (_recording)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Recording already started");
- return 0;
- }
-
- // get the JNI env for this thread
- JNIEnv *env;
- bool isAttached = false;
-
- // get the JNI env for this thread
- if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = _javaVM->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not attach thread to JVM (%d, %p)", res, env);
- return -1;
- }
- isAttached = true;
- }
-
- // get the method ID
- jmethodID startRecordingID = env->GetMethodID(_javaScClass,
- "StartRecording", "()I");
-
- // Call java sc object method
- jint res = env->CallIntMethod(_javaScObj, startRecordingID);
- if (res < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "StartRecording failed (%d)", res);
- return -1;
- }
-
- _recWarning = 0;
- _recError = 0;
-
- // Signal to recording thread that we want to start
- _startRec = true;
- _timeEventRec.Set(); // Release thread from waiting state
- _critSect.Leave();
- // Wait for thread to init
- if (kEventSignaled != _recStartStopEvent.Wait(5000))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Timeout or error starting");
- }
- _recStartStopEvent.Reset();
- _critSect.Enter();
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (_javaVM->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Could not detach thread from JVM");
- }
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StopRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::StopRecording()
-
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_recIsInitialized)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Recording is not initialized");
- return 0;
- }
-
- // make sure we don't start recording (it's asynchronous),
- // assuming that we are under lock
- _startRec = false;
-
- // get the JNI env for this thread
- JNIEnv *env;
- bool isAttached = false;
-
- // get the JNI env for this thread
- if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = _javaVM->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not attach thread to JVM (%d, %p)", res, env);
- return -1;
- }
- isAttached = true;
- }
-
- // get the method ID
- jmethodID stopRecordingID = env->GetMethodID(_javaScClass, "StopRecording",
- "()I");
-
- // Call java sc object method
- jint res = env->CallIntMethod(_javaScObj, stopRecordingID);
- if (res < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "StopRecording failed (%d)", res);
- }
-
- _recIsInitialized = false;
- _recording = false;
- _recWarning = 0;
- _recError = 0;
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (_javaVM->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Could not detach thread from JVM");
- }
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceAndroidJni::RecordingIsInitialized() const
-{
-
- return _recIsInitialized;
-}
-
-// ----------------------------------------------------------------------------
-// Recording
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceAndroidJni::Recording() const
-{
-
- return _recording;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceAndroidJni::PlayoutIsInitialized() const
-{
-
- return _playIsInitialized;
-}
-
-// ----------------------------------------------------------------------------
-// StartPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::StartPlayout()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_playIsInitialized)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Playout not initialized");
- return -1;
- }
-
- if (_playing)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Playout already started");
- return 0;
- }
-
- // get the JNI env for this thread
- JNIEnv *env;
- bool isAttached = false;
-
- // get the JNI env for this thread
- if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = _javaVM->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not attach thread to JVM (%d, %p)", res, env);
- return -1;
- }
- isAttached = true;
- }
-
- // get the method ID
- jmethodID startPlaybackID = env->GetMethodID(_javaScClass, "StartPlayback",
- "()I");
-
- // Call java sc object method
- jint res = env->CallIntMethod(_javaScObj, startPlaybackID);
- if (res < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "StartPlayback failed (%d)", res);
- return -1;
- }
-
- _playWarning = 0;
- _playError = 0;
-
- // Signal to playout thread that we want to start
- _startPlay = true;
- _timeEventPlay.Set(); // Release thread from waiting state
- _critSect.Leave();
- // Wait for thread to init
- if (kEventSignaled != _playStartStopEvent.Wait(5000))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Timeout or error starting");
- }
- _playStartStopEvent.Reset();
- _critSect.Enter();
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (_javaVM->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Could not detach thread from JVM");
- }
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StopPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::StopPlayout()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_playIsInitialized)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Playout is not initialized");
- return 0;
- }
-
- // get the JNI env for this thread
- JNIEnv *env;
- bool isAttached = false;
-
- // get the JNI env for this thread
- if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = _javaVM->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not attach thread to JVM (%d, %p)", res, env);
- return -1;
- }
- isAttached = true;
- }
-
- // get the method ID
- jmethodID stopPlaybackID = env->GetMethodID(_javaScClass, "StopPlayback",
- "()I");
-
- // Call java sc object method
- jint res = env->CallIntMethod(_javaScObj, stopPlaybackID);
- if (res < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "StopPlayback failed (%d)", res);
- }
-
- _playIsInitialized = false;
- _playing = false;
- _playWarning = 0;
- _playError = 0;
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (_javaVM->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Could not detach thread from JVM");
- }
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDelay
-//
-// Remaining amount of data still in the playout buffer.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::PlayoutDelay(WebRtc_UWord16& delayMS) const
-{
- delayMS = _delayPlayout;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDelay
-//
-// Remaining amount of data still in the recording buffer.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::RecordingDelay(
- WebRtc_UWord16& delayMS) const
-{
- delayMS = _delayRecording;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Playing
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceAndroidJni::Playing() const
-{
-
- return _playing;
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutBuffer
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SetPlayoutBuffer(
- const AudioDeviceModule::BufferType /*type*/,
- WebRtc_UWord16 /*sizeMS*/)
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutBuffer
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::PlayoutBuffer(
- AudioDeviceModule::BufferType& type,
- WebRtc_UWord16& sizeMS) const
-{
-
- type = AudioDeviceModule::kAdaptiveBufferSize;
- sizeMS = _delayPlayout; // Set to current playout delay
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// CPULoad
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::CPULoad(WebRtc_UWord16& /*load*/) const
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutWarning
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceAndroidJni::PlayoutWarning() const
-{
- return (_playWarning > 0);
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutError
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceAndroidJni::PlayoutError() const
-{
- return (_playError > 0);
-}
-
-// ----------------------------------------------------------------------------
-// RecordingWarning
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceAndroidJni::RecordingWarning() const
-{
- return (_recWarning > 0);
-}
-
-// ----------------------------------------------------------------------------
-// RecordingError
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceAndroidJni::RecordingError() const
-{
- return (_recError > 0);
-}
-
-// ----------------------------------------------------------------------------
-// ClearPlayoutWarning
-// ----------------------------------------------------------------------------
-
-void AudioDeviceAndroidJni::ClearPlayoutWarning()
-{
- _playWarning = 0;
-}
-
-// ----------------------------------------------------------------------------
-// ClearPlayoutError
-// ----------------------------------------------------------------------------
-
-void AudioDeviceAndroidJni::ClearPlayoutError()
-{
- _playError = 0;
-}
-
-// ----------------------------------------------------------------------------
-// ClearRecordingWarning
-// ----------------------------------------------------------------------------
-
-void AudioDeviceAndroidJni::ClearRecordingWarning()
-{
- _recWarning = 0;
-}
-
-// ----------------------------------------------------------------------------
-// ClearRecordingError
-// ----------------------------------------------------------------------------
-
-void AudioDeviceAndroidJni::ClearRecordingError()
-{
- _recError = 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingSampleRate
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SetRecordingSampleRate(
- const WebRtc_UWord32 samplesPerSec)
-{
-
- if (samplesPerSec > 48000 || samplesPerSec < 8000)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Invalid sample rate");
- return -1;
- }
-
- // set the recording sample rate to use
- if (samplesPerSec == 44100)
- {
- _samplingFreqIn = 44;
- }
- else
- {
- _samplingFreqIn = samplesPerSec / 1000;
- }
-
- // Update the AudioDeviceBuffer
- _ptrAudioBuffer->SetRecordingSampleRate(samplesPerSec);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutSampleRate
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SetPlayoutSampleRate(
- const WebRtc_UWord32 samplesPerSec)
-{
-
- if (samplesPerSec > 48000 || samplesPerSec < 8000)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Invalid sample rate");
- return -1;
- }
-
- // set the playout sample rate to use
- if (samplesPerSec == 44100)
- {
- _samplingFreqOut = 44;
- }
- else
- {
- _samplingFreqOut = samplesPerSec / 1000;
- }
-
- // Update the AudioDeviceBuffer
- _ptrAudioBuffer->SetPlayoutSampleRate(samplesPerSec);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetLoudspeakerStatus
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::SetLoudspeakerStatus(bool enable)
-{
-
- if (!_javaContext)
- {
- WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
- " Context is not set");
- return -1;
- }
-
- // get the JNI env for this thread
- JNIEnv *env;
- bool isAttached = false;
-
- if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = _javaVM->AttachCurrentThread(&env, NULL);
-
- // Get the JNI env for this thread
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
- " Could not attach thread to JVM (%d, %p)", res, env);
- return -1;
- }
- isAttached = true;
- }
-
- // get the method ID
- jmethodID setPlayoutSpeakerID = env->GetMethodID(_javaScClass,
- "SetPlayoutSpeaker",
- "(Z)I");
-
- // call java sc object method
- jint res = env->CallIntMethod(_javaScObj, setPlayoutSpeakerID, enable);
- if (res < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
- " SetPlayoutSpeaker failed (%d)", res);
- return -1;
- }
-
- _loudSpeakerOn = enable;
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (_javaVM->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, -1,
- " Could not detach thread from JVM");
- }
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// GetLoudspeakerStatus
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::GetLoudspeakerStatus(bool& enabled) const
-{
-
- enabled = _loudSpeakerOn;
-
- return 0;
-}
-
-// ============================================================================
-// Private Methods
-// ============================================================================
-
-
-// ----------------------------------------------------------------------------
-// InitJavaResources
-//
-// Initializes needed Java resources like the JNI interface to
-// AudioDeviceAndroid.java
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::InitJavaResources()
-{
- // todo: Check if we already have created the java object
- _javaVM = globalJvm;
- _javaContext = globalSndContext;
- _javaScClass = globalScClass;
-
- // use the jvm that has been set
- if (!_javaVM)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: Not a valid Java VM pointer", __FUNCTION__);
- return -1;
- }
-
- // get the JNI env for this thread
- JNIEnv *env;
- bool isAttached = false;
-
- // get the JNI env for this thread
- if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = _javaVM->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: Could not attach thread to JVM (%d, %p)",
- __FUNCTION__, res, env);
- return -1;
- }
- isAttached = true;
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- "get method id");
-
- // get the method ID for the void(void) constructor
- jmethodID cid = env->GetMethodID(_javaScClass, "<init>", "()V");
- if (cid == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: could not get constructor ID", __FUNCTION__);
- return -1; /* exception thrown */
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- "construct object", __FUNCTION__);
-
- // construct the object
- jobject javaScObjLocal = env->NewObject(_javaScClass, cid);
- if (!javaScObjLocal)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "%s: could not create Java sc object", __FUNCTION__);
- return -1;
- }
-
- // create a reference to the object (to tell JNI that we are referencing it
- // after this function has returned)
- _javaScObj = env->NewGlobalRef(javaScObjLocal);
- if (!_javaScObj)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: could not create Java sc object reference",
- __FUNCTION__);
- return -1;
- }
-
- // Delete local object ref, we only use the global ref
- env->DeleteLocalRef(javaScObjLocal);
-
- //////////////////////
- // AUDIO MANAGEMENT
-
- // This is not mandatory functionality
- if (_javaContext)
- {
- // Get Context field ID
- jfieldID fidContext = env->GetFieldID(_javaScClass, "_context",
- "Landroid/content/Context;");
- if (!fidContext)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: could not get Context fid", __FUNCTION__);
- return -1;
- }
-
- // Set the Java application Context so we can use AudioManager
- // Get Context object and check it
- jobject javaContext = (jobject) _javaContext;
- env->SetObjectField(_javaScObj, fidContext, javaContext);
- javaContext = env->GetObjectField(_javaScObj, fidContext);
- if (!javaContext)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: could not set Context", __FUNCTION__);
- return -1;
- }
-
- // Delete local object ref
- env->DeleteLocalRef(javaContext);
- }
- else
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceAudioDevice,
- _id,
- "%s: did not set Context - some functionality is not "
- "supported",
- __FUNCTION__);
- }
-
- /////////////
- // PLAYOUT
-
- // Get play buffer field ID
- jfieldID fidPlayBuffer = env->GetFieldID(_javaScClass, "_playBuffer",
- "Ljava/nio/ByteBuffer;");
- if (!fidPlayBuffer)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: could not get play buffer fid", __FUNCTION__);
- return -1;
- }
-
- // Get play buffer object
- jobject javaPlayBufferLocal =
- env->GetObjectField(_javaScObj, fidPlayBuffer);
- if (!javaPlayBufferLocal)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: could not get play buffer", __FUNCTION__);
- return -1;
- }
-
- // Create a global reference to the object (to tell JNI that we are
- // referencing it after this function has returned)
- // NOTE: we are referencing it only through the direct buffer (see below)
- _javaPlayBuffer = env->NewGlobalRef(javaPlayBufferLocal);
- if (!_javaPlayBuffer)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: could not get play buffer reference", __FUNCTION__);
- return -1;
- }
-
- // Delete local object ref, we only use the global ref
- env->DeleteLocalRef(javaPlayBufferLocal);
-
- // Get direct buffer
- _javaDirectPlayBuffer = env->GetDirectBufferAddress(_javaPlayBuffer);
- if (!_javaDirectPlayBuffer)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: could not get direct play buffer", __FUNCTION__);
- return -1;
- }
-
- // Get the play audio method ID
- _javaMidPlayAudio = env->GetMethodID(_javaScClass, "PlayAudio", "(I)I");
- if (!_javaMidPlayAudio)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: could not get play audio mid", __FUNCTION__);
- return -1;
- }
-
- //////////////
- // RECORDING
-
- // Get rec buffer field ID
- jfieldID fidRecBuffer = env->GetFieldID(_javaScClass, "_recBuffer",
- "Ljava/nio/ByteBuffer;");
- if (!fidRecBuffer)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: could not get rec buffer fid", __FUNCTION__);
- return -1;
- }
-
- // Get rec buffer object
- jobject javaRecBufferLocal = env->GetObjectField(_javaScObj, fidRecBuffer);
- if (!javaRecBufferLocal)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: could not get rec buffer", __FUNCTION__);
- return -1;
- }
-
- // Create a global reference to the object (to tell JNI that we are
- // referencing it after this function has returned)
- // NOTE: we are referencing it only through the direct buffer (see below)
- _javaRecBuffer = env->NewGlobalRef(javaRecBufferLocal);
- if (!_javaRecBuffer)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: could not get rec buffer reference", __FUNCTION__);
- return -1;
- }
-
- // Delete local object ref, we only use the global ref
- env->DeleteLocalRef(javaRecBufferLocal);
-
- // Get direct buffer
- _javaDirectRecBuffer = env->GetDirectBufferAddress(_javaRecBuffer);
- if (!_javaDirectRecBuffer)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: could not get direct rec buffer", __FUNCTION__);
- return -1;
- }
-
- // Get the rec audio method ID
- _javaMidRecAudio = env->GetMethodID(_javaScClass, "RecordAudio", "(I)I");
- if (!_javaMidRecAudio)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: could not get rec audio mid", __FUNCTION__);
- return -1;
- }
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (_javaVM->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "%s: Could not detach thread from JVM", __FUNCTION__);
- }
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitSampleRate
-//
-// checks supported sample rates for playback
-// and recording and initializes the rates to be used
-// Also stores the max playout volume returned from InitPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidJni::InitSampleRate()
-{
- int samplingFreq = 44100;
- jint res = 0;
-
- // get the JNI env for this thread
- JNIEnv *env;
- bool isAttached = false;
-
- // get the JNI env for this thread
- if (_javaVM->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = _javaVM->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: Could not attach thread to JVM (%d, %p)",
- __FUNCTION__, res, env);
- return -1;
- }
- isAttached = true;
- }
-
- if (_samplingFreqIn > 0)
- {
- // read the configured sampling rate
- samplingFreq = 44100;
- if (_samplingFreqIn != 44)
- {
- samplingFreq = _samplingFreqIn * 1000;
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
- " Trying configured recording sampling rate %d",
- samplingFreq);
- }
-
- // get the method ID
- jmethodID initRecordingID = env->GetMethodID(_javaScClass, "InitRecording",
- "(II)I");
-
- bool keepTrying = true;
- while (keepTrying)
- {
- // call java sc object method
- res = env->CallIntMethod(_javaScObj, initRecordingID, _recAudioSource,
- samplingFreq);
- if (res < 0)
- {
- switch (samplingFreq)
- {
- case 44100:
- samplingFreq = 16000;
- break;
- case 16000:
- samplingFreq = 8000;
- break;
- default: // error
- WEBRTC_TRACE(kTraceError,
- kTraceAudioDevice, _id,
- "%s: InitRecording failed (%d)", __FUNCTION__,
- res);
- return -1;
- }
- }
- else
- {
- keepTrying = false;
- }
- }
-
- // set the recording sample rate to use
- if (samplingFreq == 44100)
- {
- _samplingFreqIn = 44;
- }
- else
- {
- _samplingFreqIn = samplingFreq / 1000;
- }
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
- "Recording sample rate set to (%d)", _samplingFreqIn);
-
- // get the method ID
- jmethodID stopRecordingID = env->GetMethodID(_javaScClass, "StopRecording",
- "()I");
-
- // Call java sc object method
- res = env->CallIntMethod(_javaScObj, stopRecordingID);
- if (res < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "StopRecording failed (%d)", res);
- }
-
- // get the method ID
- jmethodID initPlaybackID = env->GetMethodID(_javaScClass, "InitPlayback",
- "(I)I");
-
- if (_samplingFreqOut > 0)
- {
- // read the configured sampling rate
- samplingFreq = 44100;
- if (_samplingFreqOut != 44)
- {
- samplingFreq = _samplingFreqOut * 1000;
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
- " Trying configured playback sampling rate %d",
- samplingFreq);
- }
- else
- {
- // set the preferred sampling frequency
- if (samplingFreq == 8000)
- {
- // try 16000
- samplingFreq = 16000;
- }
- // else use same as recording
- }
-
- keepTrying = true;
- while (keepTrying)
- {
- // call java sc object method
- res = env->CallIntMethod(_javaScObj, initPlaybackID, samplingFreq);
- if (res < 0)
- {
- switch (samplingFreq)
- {
- case 44100:
- samplingFreq = 16000;
- break;
- case 16000:
- samplingFreq = 8000;
- break;
- default: // error
- WEBRTC_TRACE(kTraceError,
- kTraceAudioDevice, _id,
- "InitPlayback failed (%d)", res);
- return -1;
- }
- }
- else
- {
- keepTrying = false;
- }
- }
-
- // Store max playout volume
- _maxSpeakerVolume = static_cast<WebRtc_UWord32> (res);
- if (_maxSpeakerVolume < 1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Did not get valid max speaker volume value (%d)",
- _maxSpeakerVolume);
- }
-
- // set the playback sample rate to use
- if (samplingFreq == 44100)
- {
- _samplingFreqOut = 44;
- }
- else
- {
- _samplingFreqOut = samplingFreq / 1000;
- }
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
- "Playback sample rate set to (%d)", _samplingFreqOut);
-
- // get the method ID
- jmethodID stopPlaybackID = env->GetMethodID(_javaScClass, "StopPlayback",
- "()I");
-
- // Call java sc object method
- res = env->CallIntMethod(_javaScObj, stopPlaybackID);
- if (res < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "StopPlayback failed (%d)", res);
- }
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (_javaVM->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "%s: Could not detach thread from JVM", __FUNCTION__);
- }
- }
-
- return 0;
-}
-
-// ============================================================================
-// Thread Methods
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// PlayThreadFunc
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceAndroidJni::PlayThreadFunc(void* pThis)
-{
- return (static_cast<AudioDeviceAndroidJni*> (pThis)->PlayThreadProcess());
-}
-
-// ----------------------------------------------------------------------------
-// RecThreadFunc
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceAndroidJni::RecThreadFunc(void* pThis)
-{
- return (static_cast<AudioDeviceAndroidJni*> (pThis)->RecThreadProcess());
-}
-
-// ----------------------------------------------------------------------------
-// PlayThreadProcess
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceAndroidJni::PlayThreadProcess()
-{
- if (!_playThreadIsInitialized)
- {
- // Do once when thread is started
-
- // Attach this thread to JVM and get the JNI env for this thread
- jint res = _javaVM->AttachCurrentThread(&_jniEnvPlay, NULL);
- if ((res < 0) || !_jniEnvPlay)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
- _id,
- "Could not attach playout thread to JVM (%d, %p)",
- res, _jniEnvPlay);
- return false; // Close down thread
- }
-
- _playThreadIsInitialized = true;
- }
-
- if (!_playing)
- {
- switch (_timeEventPlay.Wait(1000))
- {
- case kEventSignaled:
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
- _id, "Playout thread event signal");
- _timeEventPlay.Reset();
- break;
- case kEventError:
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
- _id, "Playout thread event error");
- return true;
- case kEventTimeout:
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
- _id, "Playout thread event timeout");
- return true;
- }
- }
-
- Lock();
-
- if (_startPlay)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "_startPlay true, performing initial actions");
- _startPlay = false;
- _playing = true;
- _playWarning = 0;
- _playError = 0;
- _playStartStopEvent.Set();
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- "Sent signal");
- }
-
- if (_playing)
- {
- WebRtc_Word8 playBuffer[2 * 480]; // Max 10 ms @ 48 kHz / 16 bit
- WebRtc_UWord32 samplesToPlay = _samplingFreqOut * 10;
-
- // ask for new PCM data to be played out using the AudioDeviceBuffer
- // ensure that this callback is executed without taking the
- // audio-thread lock
- UnLock();
- WebRtc_UWord32 nSamples =
- _ptrAudioBuffer->RequestPlayoutData(samplesToPlay);
- Lock();
-
- // Check again since play may have stopped during unlocked period
- if (!_playing)
- {
- UnLock();
- return true;
- }
-
- nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
- if (nSamples != samplesToPlay)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " invalid number of output samples(%d)", nSamples);
- _playWarning = 1;
- }
-
- // Copy data to our direct buffer (held by java sc object)
- // todo: Give _javaDirectPlayBuffer directly to VoE?
- memcpy(_javaDirectPlayBuffer, playBuffer, nSamples * 2);
-
- UnLock();
-
- // Call java sc object method to process data in direct buffer
- // Will block until data has been put in OS playout buffer
- // (see java sc class)
- jint res = _jniEnvPlay->CallIntMethod(_javaScObj, _javaMidPlayAudio,
- 2 * nSamples);
- if (res < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "PlayAudio failed (%d)", res);
- _playWarning = 1;
- }
- else if (res > 0)
- {
- // we are not recording and have got a delay value from playback
- _delayPlayout = res / _samplingFreqOut;
- }
- // If 0 is returned we are recording and then play delay is updated
- // in RecordProcess
-
- Lock();
-
- } // _playing
-
- if (_shutdownPlayThread)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- "Detaching thread from Java VM");
-
- // Detach thread from Java VM
- if (_javaVM->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
- _id, "Could not detach playout thread from JVM");
- _shutdownPlayThread = false;
- // If we say OK (i.e. set event) and close thread anyway,
- // app will crash
- }
- else
- {
- _jniEnvPlay = NULL;
- _shutdownPlayThread = false;
- _playStartStopEvent.Set(); // Signal to Terminate() that we are done
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- "Sent signal");
- }
- }
-
- UnLock();
- return true;
-}
-
-// ----------------------------------------------------------------------------
-// RecThreadProcess
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceAndroidJni::RecThreadProcess()
-{
- if (!_recThreadIsInitialized)
- {
- // Do once when thread is started
-
- // Attach this thread to JVM
- jint res = _javaVM->AttachCurrentThread(&_jniEnvRec, NULL);
-
- // Get the JNI env for this thread
- if ((res < 0) || !_jniEnvRec)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
- _id, "Could not attach rec thread to JVM (%d, %p)",
- res, _jniEnvRec);
- return false; // Close down thread
- }
-
- _recThreadIsInitialized = true;
- }
-
- // just sleep if rec has not started
- if (!_recording)
- {
- switch (_timeEventRec.Wait(1000))
- {
- case kEventSignaled:
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
- _id, "Recording thread event signal");
- _timeEventRec.Reset();
- break;
- case kEventError:
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
- _id, "Recording thread event error");
- return true;
- case kEventTimeout:
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
- _id, "Recording thread event timeout");
- return true;
- }
- }
-
- Lock();
-
- if (_startRec)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "_startRec true, performing initial actions");
- _startRec = false;
- _recording = true;
- _recWarning = 0;
- _recError = 0;
- _recStartStopEvent.Set();
- }
-
- if (_recording)
- {
- WebRtc_UWord32 samplesToRec = _samplingFreqIn * 10;
-
- // Call java sc object method to record data to direct buffer
- // Will block until data has been recorded (see java sc class),
- // therefore we must release the lock
- UnLock();
- jint playDelayInSamples = _jniEnvRec->CallIntMethod(_javaScObj,
- _javaMidRecAudio,
- 2 * samplesToRec);
- if (playDelayInSamples < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "RecordAudio failed");
- _recWarning = 1;
- }
- else
- {
- _delayPlayout = playDelayInSamples / _samplingFreqOut;
- }
- Lock();
-
- // Check again since recording may have stopped during Java call
- if (_recording)
- {
-// WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
-// "total delay is %d", msPlayDelay + _delayRecording);
-
- // Copy data to our direct buffer (held by java sc object)
- // todo: Give _javaDirectRecBuffer directly to VoE?
- // todo: Check count <= 480 ?
- memcpy(_recBuffer, _javaDirectRecBuffer, 2 * samplesToRec);
-
- // store the recorded buffer (no action will be taken if the
- // #recorded samples is not a full buffer)
- _ptrAudioBuffer->SetRecordedBuffer(_recBuffer, samplesToRec);
-
- // store vqe delay values
- _ptrAudioBuffer->SetVQEData(_delayPlayout, _delayRecording, 0);
-
- // deliver recorded samples at specified sample rate, mic level
- // etc. to the observer using callback
- UnLock();
- _ptrAudioBuffer->DeliverRecordedData();
- Lock();
- }
-
- } // _recording
-
- if (_shutdownRecThread)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- "Detaching rec thread from Java VM");
-
- // Detach thread from Java VM
- if (_javaVM->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
- _id, "Could not detach recording thread from JVM");
- _shutdownRecThread = false;
- // If we say OK (i.e. set event) and close thread anyway,
- // app will crash
- }
- else
- {
- _jniEnvRec = NULL;
- _shutdownRecThread = false;
- _recStartStopEvent.Set(); // Signal to Terminate() that we are done
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- "Sent signal rec");
- }
- }
-
- UnLock();
- return true;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/android/audio_device_android_jni.h b/trunk/src/modules/audio_device/main/source/android/audio_device_android_jni.h
deleted file mode 100644
index e127e26..0000000
--- a/trunk/src/modules/audio_device/main/source/android/audio_device_android_jni.h
+++ /dev/null
@@ -1,268 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Android audio device interface (JNI/AudioTrack/AudioRecord usage)
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ANDROID_JNI_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ANDROID_JNI_H
-
-#include "audio_device_generic.h"
-#include "critical_section_wrapper.h"
-
-#include <jni.h> // For accessing AudioDeviceAndroid java class
-
-namespace webrtc
-{
-class EventWrapper;
-
-const WebRtc_UWord32 N_REC_SAMPLES_PER_SEC = 44000; // Default is 44.1 kHz
-const WebRtc_UWord32 N_PLAY_SAMPLES_PER_SEC = 44000; // Default is 44.1 kHz
-
-const WebRtc_UWord32 N_REC_CHANNELS = 1; // default is mono recording
-const WebRtc_UWord32 N_PLAY_CHANNELS = 1; // default is mono playout
-
-const WebRtc_UWord32 REC_BUF_SIZE_IN_SAMPLES = 480; // Handle max 10 ms @ 48 kHz
-
-
-WebRtc_Word32 SetAndroidAudioDeviceObjects(void* javaVM, void* env,
- void* context);
-
-class ThreadWrapper;
-
-class AudioDeviceAndroidJni: public AudioDeviceGeneric
-{
-public:
- AudioDeviceAndroidJni(const WebRtc_Word32 id);
- ~AudioDeviceAndroidJni();
-
- virtual WebRtc_Word32 ActiveAudioLayer(
- AudioDeviceModule::AudioLayer& audioLayer) const;
-
- virtual WebRtc_Word32 Init();
- virtual WebRtc_Word32 Terminate();
- virtual bool Initialized() const;
-
- virtual WebRtc_Word16 PlayoutDevices();
- virtual WebRtc_Word16 RecordingDevices();
- virtual WebRtc_Word32 PlayoutDeviceName(
- WebRtc_UWord16 index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]);
- virtual WebRtc_Word32 RecordingDeviceName(
- WebRtc_UWord16 index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]);
-
- virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32 SetPlayoutDevice(
- AudioDeviceModule::WindowsDeviceType device);
- virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32 SetRecordingDevice(
- AudioDeviceModule::WindowsDeviceType device);
-
- virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
- virtual WebRtc_Word32 InitPlayout();
- virtual bool PlayoutIsInitialized() const;
- virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
- virtual WebRtc_Word32 InitRecording();
- virtual bool RecordingIsInitialized() const;
-
- virtual WebRtc_Word32 StartPlayout();
- virtual WebRtc_Word32 StopPlayout();
- virtual bool Playing() const;
- virtual WebRtc_Word32 StartRecording();
- virtual WebRtc_Word32 StopRecording();
- virtual bool Recording() const;
-
- virtual WebRtc_Word32 SetAGC(bool enable);
- virtual bool AGC() const;
-
- virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
- WebRtc_UWord16 volumeRight);
- virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft,
- WebRtc_UWord16& volumeRight) const;
-
- virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
- virtual WebRtc_Word32 InitSpeaker();
- virtual bool SpeakerIsInitialized() const;
- virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
- virtual WebRtc_Word32 InitMicrophone();
- virtual bool MicrophoneIsInitialized() const;
-
- virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
- virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
- virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
- virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
- virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
-
- virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
- virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
- virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
- virtual WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize)
- const;
-
- virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
- virtual WebRtc_Word32 SetSpeakerMute(bool enable);
- virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
-
- virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
- virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
-
- virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
- virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
-
- virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
- virtual WebRtc_Word32 SetStereoPlayout(bool enable);
- virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
- virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
- virtual WebRtc_Word32 SetStereoRecording(bool enable);
- virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
-
- virtual WebRtc_Word32 SetPlayoutBuffer(
- const AudioDeviceModule::BufferType type, WebRtc_UWord16 sizeMS);
- virtual WebRtc_Word32 PlayoutBuffer(
- AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const;
- virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
- virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
-
- virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
-
- virtual bool PlayoutWarning() const;
- virtual bool PlayoutError() const;
- virtual bool RecordingWarning() const;
- virtual bool RecordingError() const;
- virtual void ClearPlayoutWarning();
- virtual void ClearPlayoutError();
- virtual void ClearRecordingWarning();
- virtual void ClearRecordingError();
-
- virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
- virtual WebRtc_Word32 SetRecordingSampleRate(
- const WebRtc_UWord32 samplesPerSec);
- virtual WebRtc_Word32 SetPlayoutSampleRate(
- const WebRtc_UWord32 samplesPerSec);
-
- virtual WebRtc_Word32 SetLoudspeakerStatus(bool enable);
- virtual WebRtc_Word32 GetLoudspeakerStatus(bool& enable) const;
-
-private:
- // Lock
- void Lock()
- {
- _critSect.Enter();
- };
- void UnLock()
- {
- _critSect.Leave();
- };
-
- // Init
- WebRtc_Word32 InitJavaResources();
- WebRtc_Word32 InitSampleRate();
-
- // Threads
- static bool RecThreadFunc(void*);
- static bool PlayThreadFunc(void*);
- bool RecThreadProcess();
- bool PlayThreadProcess();
-
- // Misc
- AudioDeviceBuffer* _ptrAudioBuffer;
- CriticalSectionWrapper& _critSect;
- WebRtc_Word32 _id;
-
- // Events
- EventWrapper& _timeEventRec;
- EventWrapper& _timeEventPlay;
- EventWrapper& _recStartStopEvent;
- EventWrapper& _playStartStopEvent;
-
- // Threads
- ThreadWrapper* _ptrThreadPlay;
- ThreadWrapper* _ptrThreadRec;
- WebRtc_UWord32 _recThreadID;
- WebRtc_UWord32 _playThreadID;
- bool _playThreadIsInitialized;
- bool _recThreadIsInitialized;
- bool _shutdownPlayThread;
- bool _shutdownRecThread;
-
- // Rec buffer
- WebRtc_Word8 _recBuffer[2 * REC_BUF_SIZE_IN_SAMPLES];
-
- // States
- bool _recordingDeviceIsSpecified;
- bool _playoutDeviceIsSpecified;
- bool _initialized;
- bool _recording;
- bool _playing;
- bool _recIsInitialized;
- bool _playIsInitialized;
- bool _micIsInitialized;
- bool _speakerIsInitialized;
-
- // Signal flags to threads
- bool _startRec;
- bool _stopRec;
- bool _startPlay;
- bool _stopPlay;
-
- // Warnings and errors
- WebRtc_UWord16 _playWarning;
- WebRtc_UWord16 _playError;
- WebRtc_UWord16 _recWarning;
- WebRtc_UWord16 _recError;
-
- // Delay
- WebRtc_UWord16 _delayPlayout;
- WebRtc_UWord16 _delayRecording;
-
- // AGC state
- bool _AGC;
-
- // Stored device properties
- WebRtc_UWord16 _samplingFreqIn; // Sampling frequency for Mic
- WebRtc_UWord16 _samplingFreqOut; // Sampling frequency for Speaker
- WebRtc_UWord32 _maxSpeakerVolume; // The maximum speaker volume value
- bool _loudSpeakerOn;
- // Stores the desired audio source to use, set in SetRecordingDevice
- int _recAudioSource;
-
- // JNI and Java
- JavaVM* _javaVM; // denotes a Java VM
- jobject _javaContext; // the application context
-
- JNIEnv* _jniEnvPlay; // The JNI env for playout thread
- JNIEnv* _jniEnvRec; // The JNI env for recording thread
-
- jclass _javaScClass; // AudioDeviceAndroid class
- jobject _javaScObj; // AudioDeviceAndroid object
-
- // The play buffer field in AudioDeviceAndroid object (global ref)
- jobject _javaPlayBuffer;
- // The rec buffer field in AudioDeviceAndroid object (global ref)
- jobject _javaRecBuffer;
- void* _javaDirectPlayBuffer; // Direct buffer pointer to play buffer
- void* _javaDirectRecBuffer; // Direct buffer pointer to rec buffer
- jmethodID _javaMidPlayAudio; // Method ID of play in AudioDeviceAndroid
- jmethodID _javaMidRecAudio; // Method ID of rec in AudioDeviceAndroid
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ANDROID_JNI_H
diff --git a/trunk/src/modules/audio_device/main/source/android/audio_device_android_opensles.cc b/trunk/src/modules/audio_device/main/source/android/audio_device_android_opensles.cc
deleted file mode 100644
index da88a0e..0000000
--- a/trunk/src/modules/audio_device/main/source/android/audio_device_android_opensles.cc
+++ /dev/null
@@ -1,2164 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <time.h>
-#include <sys/time.h>
-
-#include "audio_device_utility.h"
-#include "audio_device_android_opensles.h"
-#include "audio_device_config.h"
-
-#include "trace.h"
-#include "thread_wrapper.h"
-#include "event_wrapper.h"
-
-#ifdef WEBRTC_ANDROID_DEBUG
-#include <android/log.h>
-#define WEBRTC_TRACE(a,b,c,...) __android_log_print( \
- ANDROID_LOG_DEBUG, "WebRTC ADM OpenSLES", __VA_ARGS__)
-#endif
-
-namespace webrtc {
-
-// ============================================================================
-// Construction & Destruction
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// AudioDeviceAndroidOpenSLES - ctor
-// ----------------------------------------------------------------------------
-
-AudioDeviceAndroidOpenSLES::AudioDeviceAndroidOpenSLES(const WebRtc_Word32 id) :
- _ptrAudioBuffer(NULL),
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _id(id),
- _slEngineObject(NULL),
- _slPlayer(NULL),
- _slEngine(NULL),
- _slPlayerPlay(NULL),
- _slOutputMixObject(NULL),
- _slSpeakerVolume(NULL),
- _slRecorder(NULL),
- _slRecorderRecord(NULL),
- _slAudioIODeviceCapabilities(NULL),
- _slRecorderSimpleBufferQueue(NULL),
- _slMicVolume(NULL),
- _micDeviceId(0),
- _recQueueSeq(0),
- _timeEventRec(*EventWrapper::Create()),
- _ptrThreadRec(NULL),
- _recThreadID(0),
- _playQueueSeq(0),
- _recCurrentSeq(0),
- _recBufferTotalSize(0),
- _recordingDeviceIsSpecified(false),
- _playoutDeviceIsSpecified(false),
- _initialized(false),
- _recording(false),
- _playing(false),
- _recIsInitialized(false),
- _playIsInitialized(false),
- _micIsInitialized(false),
- _speakerIsInitialized(false),
- _playWarning(0),
- _playError(0),
- _recWarning(0),
- _recError(0),
- _playoutDelay(0),
- _recordingDelay(0),
- _AGC(false),
- _adbSampleRate(0),
- _samplingRateIn(SL_SAMPLINGRATE_16),
- _samplingRateOut(SL_SAMPLINGRATE_16),
- _maxSpeakerVolume(0),
- _minSpeakerVolume(0),
- _loudSpeakerOn(false) {
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created",
- __FUNCTION__);
- memset(_playQueueBuffer, 0, sizeof(_playQueueBuffer));
- memset(_recQueueBuffer, 0, sizeof(_recQueueBuffer));
- memset(_recBuffer, 0, sizeof(_recBuffer));
- memset(_recLength, 0, sizeof(_recLength));
- memset(_recSeqNumber, 0, sizeof(_recSeqNumber));
-}
-
-AudioDeviceAndroidOpenSLES::~AudioDeviceAndroidOpenSLES() {
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed",
- __FUNCTION__);
-
- Terminate();
-
- delete &_timeEventRec;
- delete &_critSect;
-}
-
-// ============================================================================
-// API
-// ============================================================================
-
-void AudioDeviceAndroidOpenSLES::AttachAudioBuffer(
- AudioDeviceBuffer* audioBuffer) {
-
- CriticalSectionScoped lock(_critSect);
-
- _ptrAudioBuffer = audioBuffer;
-
- // inform the AudioBuffer about default settings for this implementation
- _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
- _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
- _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
- _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::ActiveAudioLayer(
- AudioDeviceModule::AudioLayer& audioLayer) const {
-
- audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::Init() {
-
- CriticalSectionScoped lock(_critSect);
-
- if (_initialized) {
- return 0;
- }
-
- _playWarning = 0;
- _playError = 0;
- _recWarning = 0;
- _recError = 0;
-
- SLEngineOption EngineOption[] = { (SLuint32) SL_ENGINEOPTION_THREADSAFE,
- (SLuint32) SL_BOOLEAN_TRUE };
- WebRtc_Word32 res = slCreateEngine(&_slEngineObject, 1, EngineOption, 0,
- NULL, NULL);
- //WebRtc_Word32 res = slCreateEngine( &_slEngineObject, 0, NULL, 0, NULL,
- // NULL);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to create SL Engine Object");
- return -1;
- }
- /* Realizing the SL Engine in synchronous mode. */
- if ((*_slEngineObject)->Realize(_slEngineObject, SL_BOOLEAN_FALSE)
- != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to Realize SL Engine");
- return -1;
- }
-
- if ((*_slEngineObject)->GetInterface(_slEngineObject, SL_IID_ENGINE,
- (void*) &_slEngine)
- != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get SL Engine interface");
- return -1;
- }
-
- // Check the sample rate to be used for playback and recording
- if (InitSampleRate() != 0) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "%s: Failed to init samplerate", __FUNCTION__);
- return -1;
- }
-
- // Set the audio device buffer sampling rate, we assume we get the same
- // for play and record
- if (_ptrAudioBuffer->SetRecordingSampleRate(_adbSampleRate) < 0) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not set audio device buffer recording "
- "sampling rate (%d)", _adbSampleRate);
- }
- if (_ptrAudioBuffer->SetPlayoutSampleRate(_adbSampleRate) < 0) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Could not set audio device buffer playout sampling "
- "rate (%d)", _adbSampleRate);
- }
-
- _initialized = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::Terminate() {
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_initialized) {
- return 0;
- }
-
- // RECORDING
- StopRecording();
-
- _micIsInitialized = false;
- _recordingDeviceIsSpecified = false;
-
- // PLAYOUT
- StopPlayout();
-
- if (_slEngineObject != NULL) {
- (*_slEngineObject)->Destroy(_slEngineObject);
- _slEngineObject = NULL;
- _slEngine = NULL;
- }
-
- _initialized = false;
-
- return 0;
-}
-
-bool AudioDeviceAndroidOpenSLES::Initialized() const {
-
- return (_initialized);
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SpeakerIsAvailable(bool& available) {
-
- // We always assume it's available
- available = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::InitSpeaker() {
-
- CriticalSectionScoped lock(_critSect);
-
- if (_playing) {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Playout already started");
- return -1;
- }
-
- if (!_playoutDeviceIsSpecified) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Playout device is not specified");
- return -1;
- }
-
- // Nothing needs to be done here, we use a flag to have consistent
- // behavior with other platforms
- _speakerIsInitialized = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::MicrophoneIsAvailable(bool& available) {
-
- // We always assume it's available
- available = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::InitMicrophone() {
-
- CriticalSectionScoped lock(_critSect);
-
- if (_recording) {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Recording already started");
- return -1;
- }
-
- if (!_recordingDeviceIsSpecified) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Recording device is not specified");
- return -1;
- }
-
- // Nothing needs to be done here, we use a flag to have consistent
- // behavior with other platforms
- _micIsInitialized = true;
-
- return 0;
-}
-
-bool AudioDeviceAndroidOpenSLES::SpeakerIsInitialized() const {
-
- return _speakerIsInitialized;
-}
-
-bool AudioDeviceAndroidOpenSLES::MicrophoneIsInitialized() const {
-
- return _micIsInitialized;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SpeakerVolumeIsAvailable(
- bool& available) {
-
- available = true; // We assume we are always be able to set/get volume
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetSpeakerVolume(
- WebRtc_UWord32 volume) {
-
- if (!_speakerIsInitialized) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Speaker not initialized");
- return -1;
- }
-
- if (_slEngineObject == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "SetSpeakerVolume, SL Engine object doesnt exist");
- return -1;
- }
-
- if (_slEngine == NULL) {
- // Get the SL Engine Interface which is implicit
- if ((*_slEngineObject)->GetInterface(_slEngineObject, SL_IID_ENGINE,
- (void*) &_slEngine)
- != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to GetInterface SL Engine Interface");
- return -1;
- }
- }
- /* if (_slOutputMixObject == NULL && _slEngine != NULL)
- {
- // Set arrays required[] and iidArray[] for VOLUME interface
- const SLInterfaceID ids[1] = {SL_IID_VOLUME};
- const SLboolean req[1] = {SL_BOOLEAN_TRUE};
- // Create Output Mix object to be used by player
- if ((*_slEngine)->CreateOutputMix(_slEngine, &_slOutputMixObject, 1, ids,
- req) != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to create Output Mix object");
- return -1;
- }
-
- // Realizing the Output Mix object in synchronous mode.
- if ((*_slOutputMixObject)->Realize(_slOutputMixObject,
- SL_BOOLEAN_FALSE) != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to realize the output mix object");
- return -1;
- }
- }
- if (_slSpeakerVolume == NULL && _slOutputMixObject != NULL)
- {
- if ((*_slOutputMixObject)->GetInterface(_slOutputMixObject,
- SL_IID_VOLUME, (void*)&_slSpeakerVolume) != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get interface for Speaker Volume");
- return -1;
- }
- }
-
- WebRtc_Word32 vol(0);
- vol = ((volume*(_maxSpeakerVolume-_minSpeakerVolume) + (int)(255/2))
- / (255)) + _minSpeakerVolume;
- if (_slSpeakerVolume != NULL)
- {
- if ((*_slSpeakerVolume)->SetVolumeLevel(_slSpeakerVolume, vol)
- != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to set speaker volume");
- return -1;
- }
- }
- */
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SpeakerVolume(
- WebRtc_UWord32& volume) const {
-
- /* if (!_speakerIsInitialized)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Speaker not initialized");
- return -1;
- }
-
- if (_slEngineObject == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "SetSpeakerVolume, SL Engine object doesnt exist");
- return -1;
- }
-
- if (_slEngine == NULL && _slEngineObject != NULL)
- {
- // Get the SL Engine Interface which is implicit
- if ((*_slEngineObject)->GetInterface(_slEngineObject,
- SL_IID_ENGINE, (void*)&_slEngine) != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get SL Engine interface");
- return -1;
- }
- }
- if (_slOutputMixObject == NULL && _slEngine != NULL)
- {
- // Set arrays required[] and iidArray[] for VOLUME interface
- const SLInterfaceID ids[1] = {SL_IID_VOLUME};
- const SLboolean req[1] = {SL_BOOLEAN_TRUE};
- // Create Output Mix object to be used by player
- if ((*_slEngine)->CreateOutputMix(_slEngine,
- &_slOutputMixObject, 1, ids, req) != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to create Output Mixer object");
- return -1;
- }
- // Realizing the Output Mix object in synchronous mode.
- if ((*_slOutputMixObject)->Realize(_slOutputMixObject,
- SL_BOOLEAN_FALSE) != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to realize Output Mix object");
- return -1;
- }
- }
- if (_slSpeakerVolume == NULL && _slOutputMixObject != NULL)
- {
- if ((*_slOutputMixObject)->GetInterface(_slOutputMixObject,
- SL_IID_VOLUME, (void*)&_slSpeakerVolume) != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get speaker volume interface");
- return -1;
- }
- }
-
- SLmillibel vol(0);
- if (_slSpeakerVolume != NULL)
- {
- if ((*_slSpeakerVolume)->GetVolumeLevel(_slSpeakerVolume, &vol)
- != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get speaker volume");
- return -1;
- }
- }
- // volume has to be mapped from millibel to [0, 255]
- // volume = (WebRtc_UWord32) (((vol - _minSpeakerVolume) * 255 +
- * (int)((_maxSpeakerVolume - _minSpeakerVolume)/2)) /
- * (_maxSpeakerVolume - _minSpeakerVolume));
- */
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetWaveOutVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetWaveOutVolume(
- WebRtc_UWord16 /*volumeLeft*/,
- WebRtc_UWord16 /*volumeRight*/) {
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// WaveOutVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::WaveOutVolume(
- WebRtc_UWord16& /*volumeLeft*/,
- WebRtc_UWord16& /*volumeRight*/) const {
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::MaxSpeakerVolume(
- WebRtc_UWord32& maxVolume) const {
-
- if (!_speakerIsInitialized) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Speaker not initialized");
- return -1;
- }
-
- maxVolume = _maxSpeakerVolume;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::MinSpeakerVolume(
- WebRtc_UWord32& minVolume) const {
-
- if (!_speakerIsInitialized) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Speaker not initialized");
- return -1;//
- }
-
- minVolume = _minSpeakerVolume;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SpeakerVolumeStepSize(
- WebRtc_UWord16& stepSize) const {
-
- if (!_speakerIsInitialized) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Speaker not initialized");
- return -1;
- }
- stepSize = 1;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerMuteIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SpeakerMuteIsAvailable(
- bool& available) {
-
- available = false; // Speaker mute not supported on Android
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetSpeakerMute(bool /*enable*/) {
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SpeakerMute(bool& /*enabled*/) const {
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::MicrophoneMuteIsAvailable(
- bool& available) {
-
- available = false; // Mic mute not supported on Android
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetMicrophoneMute(bool /*enable*/) {
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::MicrophoneMute(
- bool& /*enabled*/) const {
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::MicrophoneBoostIsAvailable(
- bool& available) {
-
- available = false; // Mic boost not supported on Android
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetMicrophoneBoost(bool enable) {
-
- if (!_micIsInitialized) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Microphone not initialized");
- return -1;
- }
-
- if (enable) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Enabling not available");
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::MicrophoneBoost(bool& enabled) const {
-
- if (!_micIsInitialized) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Microphone not initialized");
- return -1;
- }
-
- enabled = false;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::StereoRecordingIsAvailable(
- bool& available) {
-
- available = false; // Stereo recording not supported on Android
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetStereoRecording(bool enable) {
-
- if (enable) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Enabling not available");
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::StereoRecording(bool& enabled) const {
-
- enabled = false;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::StereoPlayoutIsAvailable(
- bool& available) {
-
- available = false; // Stereo playout not supported on Android
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetStereoPlayout(bool enable) {
-
- if (enable) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Enabling not available");
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::StereoPlayout(bool& enabled) const {
-
- enabled = false;
-
- return 0;
-}
-
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetAGC(bool enable) {
-
- _AGC = enable;
-
- return 0;
-}
-
-bool AudioDeviceAndroidOpenSLES::AGC() const {
-
- return _AGC;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::MicrophoneVolumeIsAvailable(
- bool& available) {
-
- available = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetMicrophoneVolume(
- WebRtc_UWord32 volume) {
-
- if (_slEngineObject == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "SetMicrophoneVolume, SL Engine Object doesnt exist");
- return -1;
- }
-
- /* Get the optional DEVICE VOLUME interface from the engine */
- if (_slMicVolume == NULL) {
- // Get the optional DEVICE VOLUME interface from the engine
- if ((*_slEngineObject)->GetInterface(_slEngineObject,
- SL_IID_DEVICEVOLUME,
- (void*) &_slMicVolume)
- != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to create Output Mix object");
- }
- }
-
- if (_slMicVolume != NULL) {
- WebRtc_Word32 vol(0);
- vol = ((volume * (_maxSpeakerVolume - _minSpeakerVolume) + (int) (255
- / 2)) / (255)) + _minSpeakerVolume;
- if ((*_slMicVolume)->SetVolume(_slMicVolume, _micDeviceId, vol)
- != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to create Output Mix object");
- }
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::MicrophoneVolume(
- WebRtc_UWord32& /*volume*/) const {
-
- /* if (_slEngineObject == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "MicrophoneVolume, SL Engine Object doesnt exist");
- return -1;
- }
-
- // Get the optional DEVICE VOLUME interface from the engine
- if (_slMicVolume == NULL)
- {
- // Get the optional DEVICE VOLUME interface from the engine
- if ((*_slEngineObject)->GetInterface(_slEngineObject,
- SL_IID_DEVICEVOLUME, (void*)&_slMicVolume) != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get Microphone Volume interface");
- }
- }
-
- SLint32 vol(0);
- if (_slMicVolume != NULL)
- {
- if ((*_slMicVolume)->GetVolume(_slMicVolume, _micDeviceId, &vol)
- != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get Microphone Volume");
- }
- }
- // volume has to be mapped from millibel to [0, 255]
- // volume = (WebRtc_UWord32) (((vol - _minSpeakerVolume) * 255 +
- * (int)((_maxSpeakerVolume - _minSpeakerVolume)/2)) /
- * (_maxSpeakerVolume - _minSpeakerVolume));
- */
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::MaxMicrophoneVolume(
- WebRtc_UWord32& /*maxVolume*/) const {
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::MinMicrophoneVolume(
- WebRtc_UWord32& minVolume) const {
-
- minVolume = 0;
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::MicrophoneVolumeStepSize(
- WebRtc_UWord16& stepSize) const {
-
- stepSize = 1;
- return 0;
-}
-
-WebRtc_Word16 AudioDeviceAndroidOpenSLES::PlayoutDevices() {
-
- return 1;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetPlayoutDevice(WebRtc_UWord16 index) {
-
- if (_playIsInitialized) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Playout already initialized");
- return -1;
- }
-
- if (0 != index) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Device index is out of range [0,0]");
- return -1;
- }
-
- // Do nothing but set a flag, this is to have consistent behaviour
- // with other platforms
- _playoutDeviceIsSpecified = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetPlayoutDevice(
- AudioDeviceModule::WindowsDeviceType /*device*/) {
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::PlayoutDeviceName(
- WebRtc_UWord16 index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) {
-
- if (0 != index) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Device index is out of range [0,0]");
- return -1;
- }
-
- // Return empty string
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid) {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::RecordingDeviceName(
- WebRtc_UWord16 index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) {
-
- if (0 != index) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Device index is out of range [0,0]");
- return -1;
- }
-
- // Return empty string
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid) {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- return 0;
-}
-
-WebRtc_Word16 AudioDeviceAndroidOpenSLES::RecordingDevices() {
-
- return 1;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetRecordingDevice(
- WebRtc_UWord16 index) {
-
- if (_recIsInitialized) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Recording already initialized");
- return -1;
- }
-
- if (0 != index) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Device index is out of range [0,0]");
- return -1;
- }
-
- // Do nothing but set a flag, this is to have consistent behaviour with
- // other platforms
- _recordingDeviceIsSpecified = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetRecordingDevice(
- AudioDeviceModule::WindowsDeviceType /*device*/) {
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::PlayoutIsAvailable(bool& available) {
-
- available = false;
-
- // Try to initialize the playout side
- WebRtc_Word32 res = InitPlayout();
-
- // Cancel effect of initialization
- StopPlayout();
-
- if (res != -1) {
- available = true;
- }
-
- return res;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::RecordingIsAvailable(bool& available) {
-
- available = false;
-
- // Try to initialize the playout side
- WebRtc_Word32 res = InitRecording();
-
- // Cancel effect of initialization
- StopRecording();
-
- if (res != -1) {
- available = true;
- }
-
- return res;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::InitPlayout() {
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_initialized) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, " Not initialized");
- return -1;
- }
-
- if (_playing) {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Playout already started");
- return -1;
- }
-
- if (!_playoutDeviceIsSpecified) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Playout device is not specified");
- return -1;
- }
-
- if (_playIsInitialized) {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Playout already initialized");
- return 0;
- }
-
- // Initialize the speaker
- if (InitSpeaker() == -1) {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " InitSpeaker() failed");
- }
-
- if (_slEngineObject == NULL || _slEngine == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " SLObject or Engiine is NULL");
- return -1;
- }
-
- WebRtc_Word32 res(-1);
- SLDataFormat_PCM pcm;
- SLDataSource audioSource;
- SLDataLocator_AndroidSimpleBufferQueue simpleBufferQueue;
- SLDataSink audioSink;
- SLDataLocator_OutputMix locator_outputmix;
-
- // Create Output Mix object to be used by player
- SLInterfaceID ids[N_MAX_INTERFACES];
- SLboolean req[N_MAX_INTERFACES];
- for (unsigned int i = 0; i < N_MAX_INTERFACES; i++) {
- ids[i] = SL_IID_NULL;
- req[i] = SL_BOOLEAN_FALSE;
- }
- ids[0] = SL_IID_ENVIRONMENTALREVERB;
- res = (*_slEngine)->CreateOutputMix(_slEngine, &_slOutputMixObject, 1, ids,
- req);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get SL Output Mix object");
- return -1;
- }
- // Realizing the Output Mix object in synchronous mode.
- res = (*_slOutputMixObject)->Realize(_slOutputMixObject, SL_BOOLEAN_FALSE);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to realize SL Output Mix object");
- return -1;
- }
- // Get the speaker mixer
- /* res = (*_slOutputMixObject)->GetInterface(_slOutputMixObject,
- * SL_IID_VOLUME, (void*)&_slSpeakerVolume);
- if ( res != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get Speaker Mixer");
- return -1;
- }
- */
- // The code below can be moved to startplayout instead
- /* Setup the data source structure for the buffer queue */
- simpleBufferQueue.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
- /* Two buffers in our buffer queue, to have low latency*/
- simpleBufferQueue.numBuffers = N_PLAY_QUEUE_BUFFERS;
- // TODO(xians), figure out if we should support stereo playout for android
- /* Setup the format of the content in the buffer queue */
- pcm.formatType = SL_DATAFORMAT_PCM;
- pcm.numChannels = 1;
- // _samplingRateOut is initilized in InitSampleRate()
- pcm.samplesPerSec = SL_SAMPLINGRATE_16;
- pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
- pcm.containerSize = SL_PCMSAMPLEFORMAT_FIXED_16;
- pcm.channelMask = SL_SPEAKER_FRONT_CENTER;
- pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
- audioSource.pFormat = (void *) &pcm;
- audioSource.pLocator = (void *) &simpleBufferQueue;
- /* Setup the data sink structure */
- locator_outputmix.locatorType = SL_DATALOCATOR_OUTPUTMIX;
- locator_outputmix.outputMix = _slOutputMixObject;
- audioSink.pLocator = (void *) &locator_outputmix;
- audioSink.pFormat = NULL;
-
- // Set arrays required[] and iidArray[] for SEEK interface
- // (PlayItf is implicit)
- ids[0] = SL_IID_BUFFERQUEUE;
- ids[1] = SL_IID_EFFECTSEND;
- req[0] = SL_BOOLEAN_TRUE;
- req[1] = SL_BOOLEAN_TRUE;
- // Create the music player
- res = (*_slEngine)->CreateAudioPlayer(_slEngine, &_slPlayer, &audioSource,
- &audioSink, 2, ids, req);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to create Audio Player");
- return -1;
- }
-
- // Realizing the player in synchronous mode.
- res = (*_slPlayer)->Realize(_slPlayer, SL_BOOLEAN_FALSE);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to realize the player");
- return -1;
- }
- // Get seek and play interfaces
- res = (*_slPlayer)->GetInterface(_slPlayer, SL_IID_PLAY,
- (void*) &_slPlayerPlay);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get Player interface");
- return -1;
- }
- res = (*_slPlayer)->GetInterface(_slPlayer, SL_IID_BUFFERQUEUE,
- (void*) &_slPlayerSimpleBufferQueue);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get Player Simple Buffer Queue interface");
- return -1;
- }
-
- // Setup to receive buffer queue event callbacks
- res = (*_slPlayerSimpleBufferQueue)->RegisterCallback(
- _slPlayerSimpleBufferQueue,
- PlayerSimpleBufferQueueCallback,
- this);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to register Player Callback");
- return -1;
- }
- _playIsInitialized = true;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::InitRecording() {
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_initialized) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, " Not initialized");
- return -1;
- }
-
- if (_recording) {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Recording already started");
- return -1;
- }
-
- if (!_recordingDeviceIsSpecified) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Recording device is not specified");
- return -1;
- }
-
- if (_recIsInitialized) {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Recording already initialized");
- return 0;
- }
-
- // Initialize the microphone
- if (InitMicrophone() == -1) {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " InitMicrophone() failed");
- }
-
- if (_slEngineObject == NULL || _slEngine == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Recording object is NULL");
- return -1;
- }
- WebRtc_Word32 res(-1);
- WebRtc_Word32 numInputs(0);
- WebRtc_UWord32 inputDeviceIDs[N_MAX_INPUT_DEVICES];
- SLAudioInputDescriptor audioInputDescriptor;
- SLDataSource audioSource;
- SLDataLocator_IODevice micLocator;
- SLDataSink audioSink;
- SLDataFormat_PCM pcm;
- SLDataLocator_AndroidSimpleBufferQueue simpleBufferQueue;
- bool micAvailable(false);
- WebRtc_UWord32 micDeviceID(0);
-
- /* // Get the Audio IO DEVICE CAPABILITIES interface, which is also implicit
- res = (*_slEngineObject)->GetInterface(_slEngineObject,
- SL_IID_AUDIOIODEVICECAPABILITIES, (void*)&_slAudioIODeviceCapabilities);
- if ( res != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get Audio IO device Capacilities interface");
- return -1;
- }
- numInputs = N_MAX_OUTPUT_DEVICES;
-
- res = (*_slAudioIODeviceCapabilities)->GetAvailableAudioInputs(
- _slAudioIODeviceCapabilities, &numInputs, inputDeviceIDs);
- if ( res != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get the number of Input Devices");
- return -1;
- }
-
- // Search for either earpiece microphone or headset microphone input
- // device - with a preference for the latter
- for (int i=0;i<numInputs; i++)
- {
- res = (*_slAudioIODeviceCapabilities)->QueryAudioInputCapabilities(
- _slAudioIODeviceCapabilities, inputDeviceIDs[i], &audioInputDescriptor);
- if ( res != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to query info for the Input Devices");
- return -1;
- }
- if((audioInputDescriptor.deviceConnection ==
- SL_DEVCONNECTION_ATTACHED_WIRED)
- && (audioInputDescriptor.deviceScope == SL_DEVSCOPE_USER)
- && (audioInputDescriptor.deviceLocation == SL_DEVLOCATION_HEADSET))
- {
- micDeviceID = inputDeviceIDs[i];
- micAvailable = true;
- break;
- }
- else if((audioInputDescriptor.deviceConnection ==
- SL_DEVCONNECTION_INTEGRATED)
- && (audioInputDescriptor.deviceScope == SL_DEVSCOPE_USER)
- && (audioInputDescriptor.deviceLocation == SL_DEVLOCATION_HANDSET))
- {
- micDeviceID = inputDeviceIDs[i];
- micAvailable = true;
- break;
- }
- }
- // If neither of the preferred input audio devices is available,
- // no point in continuing
- if (!micAvailable)
- {
- return -1;
- }
- */
-
- // Get the optional DEVICE VOLUME interface from the engine,
- // should this be done somewhere else
- /* res = (*_slEngineObject)->GetInterface(_slEngineObject,
- * SL_IID_DEVICEVOLUME, (void*)&_slMicVolume);
- if ( res != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get Microphone Volume interface");
- return -1;
- }
- */
- // Setup the data source structure
- micLocator.locatorType = SL_DATALOCATOR_IODEVICE;
- micLocator.deviceType = SL_IODEVICE_AUDIOINPUT;
- micLocator.deviceID = SL_DEFAULTDEVICEID_AUDIOINPUT; //micDeviceID;
- micLocator.device = NULL;
- audioSource.pLocator = (void *) &micLocator;
- audioSource.pFormat = NULL;
-
- /* Setup the data source structure for the buffer queue */
- simpleBufferQueue.locatorType = SL_DATALOCATOR_ANDROIDSIMPLEBUFFERQUEUE;
- simpleBufferQueue.numBuffers = N_REC_QUEUE_BUFFERS;
- /* Setup the format of the content in the buffer queue */
- pcm.formatType = SL_DATAFORMAT_PCM;
- pcm.numChannels = 1;
- // _samplingRateIn is initialized in initSampleRate()
- pcm.samplesPerSec = SL_SAMPLINGRATE_16;
- pcm.bitsPerSample = SL_PCMSAMPLEFORMAT_FIXED_16;
- pcm.containerSize = 16;
- pcm.channelMask = SL_SPEAKER_FRONT_CENTER;
- pcm.endianness = SL_BYTEORDER_LITTLEENDIAN;
- audioSink.pFormat = (void *) &pcm;
- audioSink.pLocator = (void *) &simpleBufferQueue;
-
- // Create audio recorder
- const SLInterfaceID id[1] = { SL_IID_ANDROIDSIMPLEBUFFERQUEUE };
- const SLboolean req[1] = { SL_BOOLEAN_TRUE };
- res = (*_slEngine)->CreateAudioRecorder(_slEngine, &_slRecorder,
- &audioSource, &audioSink, 1, id,
- req);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to create Recorder");
- return -1;
- }
-
- // Realizing the recorder in synchronous mode.
- res = (*_slRecorder)->Realize(_slRecorder, SL_BOOLEAN_FALSE);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to realize Recorder");
- return -1;
- }
-
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " get Recorder interface");
- // Get the RECORD interface - it is an implicit interface
- res = (*_slRecorder)->GetInterface(_slRecorder, SL_IID_RECORD,
- (void*) &_slRecorderRecord);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get Recorder interface");
- return -1;
- }
-
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " get Recorder Simple Buffer Queue 2");
- // Get the simpleBufferQueue interface
- res = (*_slRecorder)->GetInterface(_slRecorder,
- SL_IID_ANDROIDSIMPLEBUFFERQUEUE,
- (void*) &_slRecorderSimpleBufferQueue);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get Recorder Simple Buffer Queue");
- return -1;
- }
-
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " register Recorder Callback 2");
- // Setup to receive buffer queue event callbacks
- res = (*_slRecorderSimpleBufferQueue)->RegisterCallback(
- _slRecorderSimpleBufferQueue,
- RecorderSimpleBufferQueueCallback,
- this);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to register Recorder Callback");
- return -1;
- }
-
- _recIsInitialized = true;
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::StartRecording() {
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_recIsInitialized) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Recording not initialized");
- return -1;
- }
-
- if (_recording) {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Recording already started");
- return 0;
- }
-
- if (_slRecorderRecord == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, " RecordITF is NULL");
- return -1;
- }
-
- if (_slRecorderSimpleBufferQueue == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Recorder Simple Buffer Queue is NULL");
- return -1;
- }
-
- // Reset recording buffer
- memset(_recQueueBuffer, 0, sizeof(_recQueueBuffer)); // empty the queue
- _recQueueSeq = 0;
-
- const char* threadName = "webrtc_opensles_audio_capture_thread";
- _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc, this,
- kRealtimePriority, threadName);
- if (_ptrThreadRec == NULL)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " failed to create the rec audio thread");
- return -1;
- }
-
- unsigned int threadID(0);
- if (!_ptrThreadRec->Start(threadID))
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " failed to start the rec audio thread");
- delete _ptrThreadRec;
- _ptrThreadRec = NULL;
- return -1;
- }
- _recThreadID = threadID;
- _recThreadIsInitialized = true;
- memset(_recBuffer, 0, sizeof(_recBuffer));
- memset(_recLength, 0, sizeof(_recLength));
- memset(_recSeqNumber, 0, sizeof(_recSeqNumber));
- _recCurrentSeq = 0;
- _recBufferTotalSize = 0;
- _recWarning = 0;
- _recError = 0;
-
- // Enqueue N_REC_QUEUE_BUFFERS -1 zero buffers to get the ball rolling
- // find out how it behaves when the sample rate is 44100
- WebRtc_Word32 res(-1);
- WebRtc_UWord32 nSample10ms = _adbSampleRate / 100;
- for (int i = 0; i < (N_REC_QUEUE_BUFFERS - 1); i++) {
- // We assign 10ms buffer to each queue, size given in bytes.
- res = (*_slRecorderSimpleBufferQueue)->Enqueue(
- _slRecorderSimpleBufferQueue,
- (void*) _recQueueBuffer[_recQueueSeq],
- 2 * nSample10ms);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to Enqueue Empty Buffer to recorder");
- return -1;
- }
- _recQueueSeq++;
- }
- // Record the audio
- res = (*_slRecorderRecord)->SetRecordState(_slRecorderRecord,
- SL_RECORDSTATE_RECORDING);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to start recording");
- return -1;
- }
- _recording = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::StopRecording() {
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_recIsInitialized) {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Recording is not initialized");
- return 0;
- }
-
- // Stop the recording thread
- if (_ptrThreadRec != NULL)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- "Stopping capture thread");
- bool res = _ptrThreadRec->Stop();
- if (!res) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Failed to stop Capture thread ");
- } else {
- delete _ptrThreadRec;
- _ptrThreadRec = NULL;
- _recThreadIsInitialized = false;
- }
- }
-
- if ((_slRecorderRecord != NULL) && (_slRecorder != NULL)) {
- // Record the audio
- WebRtc_Word32 res = (*_slRecorderRecord)->SetRecordState(
- _slRecorderRecord,
- SL_RECORDSTATE_STOPPED);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to stop recording");
- return -1;
- }
- res = (*_slRecorderSimpleBufferQueue)->Clear(
- _slRecorderSimpleBufferQueue);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to clear recorder buffer queue");
- return -1;
- }
-
- // Destroy the recorder object
- (*_slRecorder)->Destroy(_slRecorder);
- _slRecorder = NULL;
- _slRecorderRecord = NULL;
- _slRecorderRecord = NULL;
- }
-
- _recIsInitialized = false;
- _recording = false;
- _recWarning = 0;
- _recError = 0;
- _recQueueSeq = 0;
- return 0;
-}
-
-bool AudioDeviceAndroidOpenSLES::RecordingIsInitialized() const {
-
- return _recIsInitialized;
-}
-
-
-bool AudioDeviceAndroidOpenSLES::Recording() const {
-
- return _recording;
-}
-
-bool AudioDeviceAndroidOpenSLES::PlayoutIsInitialized() const {
-
- return _playIsInitialized;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::StartPlayout() {
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_playIsInitialized) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Playout not initialized");
- return -1;
- }
-
- if (_playing) {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Playout already started");
- return 0;
- }
-
- if (_slPlayerPlay == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, " PlayItf is NULL");
- return -1;
- }
- if (_slPlayerSimpleBufferQueue == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " PlayerSimpleBufferQueue is NULL");
- return -1;
- }
-
- _recQueueSeq = 0;
-
- WebRtc_Word32 res(-1);
- /* Enqueue a set of zero buffers to get the ball rolling */
- WebRtc_UWord32 nSample10ms = _adbSampleRate / 100;
- WebRtc_Word8 playBuffer[2 * nSample10ms];
- WebRtc_UWord32 noSamplesOut(0);
- /* res = (*_slPlayerSimpleBufferQueue)->Clear(_slPlayerSimpleBufferQueue);
- if (res != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " player simpler buffer queue clean failed");
- //return ; dong return
- }
- */// for (int i = 0; i<(N_PLAY_QUEUE_BUFFERS -1); i++)
- {
- noSamplesOut = _ptrAudioBuffer->RequestPlayoutData(nSample10ms);
- //Lock();
- // Get data from Audio Device Buffer
- noSamplesOut = _ptrAudioBuffer->GetPlayoutData(playBuffer);
- // Insert what we have in data buffer
- memcpy(_playQueueBuffer[_playQueueSeq], playBuffer, 2 * noSamplesOut);
- //UnLock();
-
- //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- // "_playQueueSeq (%u) noSamplesOut (%d)", _playQueueSeq,
- //noSamplesOut);
- // write the buffer data we got from VoE into the device
- res = (*_slPlayerSimpleBufferQueue)->Enqueue(
- _slPlayerSimpleBufferQueue,
- (void*) _playQueueBuffer[_playQueueSeq],
- 2 * noSamplesOut);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " player simpler buffer queue Enqueue failed, %d",
- noSamplesOut);
- //return ; dong return
- }
- _playQueueSeq = (_playQueueSeq + 1) % N_PLAY_QUEUE_BUFFERS;
- }
-
- // Play the PCM samples using a buffer queue
- res = (*_slPlayerPlay)->SetPlayState(_slPlayerPlay, SL_PLAYSTATE_PLAYING);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to start playout");
- return -1;
- }
-
- _playWarning = 0;
- _playError = 0;
- _playing = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::StopPlayout() {
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_playIsInitialized) {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Playout is not initialized");
- return 0;
- }
-
- if ((_slPlayerPlay != NULL) && (_slOutputMixObject == NULL) && (_slPlayer
- == NULL)) {
- // Make sure player is stopped
- WebRtc_Word32 res =
- (*_slPlayerPlay)->SetPlayState(_slPlayerPlay,
- SL_PLAYSTATE_STOPPED);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to stop playout");
- return -1;
- }
- res = (*_slPlayerSimpleBufferQueue)->Clear(_slPlayerSimpleBufferQueue);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to clear recorder buffer queue");
- return -1;
- }
-
- // Destroy the player
- (*_slPlayer)->Destroy(_slPlayer);
- // Destroy Output Mix object
- (*_slOutputMixObject)->Destroy(_slOutputMixObject);
- _slPlayer = NULL;
- _slPlayerPlay = NULL;
- _slPlayerSimpleBufferQueue = NULL;
- _slOutputMixObject = NULL;
- }
-
- _playIsInitialized = false;
- _playing = false;
- _playWarning = 0;
- _playError = 0;
- _playQueueSeq = 0;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::PlayoutDelay(WebRtc_UWord16& delayMS) const {
- delayMS = _playoutDelay;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::RecordingDelay(WebRtc_UWord16& delayMS) const {
- delayMS = _recordingDelay;
-
- return 0;
-}
-
-bool AudioDeviceAndroidOpenSLES::Playing() const {
-
- return _playing;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetPlayoutBuffer(
- const AudioDeviceModule::BufferType /*type*/,
- WebRtc_UWord16 /*sizeMS*/) {
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::PlayoutBuffer(
- AudioDeviceModule::BufferType& type,
- WebRtc_UWord16& sizeMS) const {
-
- type = AudioDeviceModule::kAdaptiveBufferSize;
- sizeMS = _playoutDelay; // Set to current playout delay
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::CPULoad(WebRtc_UWord16& /*load*/) const {
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-bool AudioDeviceAndroidOpenSLES::PlayoutWarning() const {
- return (_playWarning > 0);
-}
-
-bool AudioDeviceAndroidOpenSLES::PlayoutError() const {
- return (_playError > 0);
-}
-
-bool AudioDeviceAndroidOpenSLES::RecordingWarning() const {
- return (_recWarning > 0);
-}
-
-bool AudioDeviceAndroidOpenSLES::RecordingError() const {
- return (_recError > 0);
-}
-
-void AudioDeviceAndroidOpenSLES::ClearPlayoutWarning() {
- _playWarning = 0;
-}
-
-void AudioDeviceAndroidOpenSLES::ClearPlayoutError() {
- _playError = 0;
-}
-
-void AudioDeviceAndroidOpenSLES::ClearRecordingWarning() {
- _recWarning = 0;
-}
-
-void AudioDeviceAndroidOpenSLES::ClearRecordingError() {
- _recError = 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::SetLoudspeakerStatus(bool enable) {
-
- // if (!_javaContext)
- // {
- // WEBRTC_TRACE(kTraceError, kTraceUtility, -1, " Context is not set");
- // return -1;
- // }
-
- // get the JNI env for this thread
- // JNIEnv *env;
- // bool isAttached = false;
-
- // if (_javaVM->GetEnv((void**)&env, JNI_VERSION_1_4) != JNI_OK)
- // {
- // try to attach the thread and get the env
- // Attach this thread to JVMslPlayoutCallback
- // jint res = _javaVM->AttachCurrentThread(&env, NULL);
-
- // Get the JNI env for this thread
- // if ((res < 0) || !env)
- // {
- // WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
- // " Could not attach thread to JVM (%d, %p)", res, env);
- // return -1;
- // }
- // isAttached = true;
- // }
-
- // get the method ID
- // jmethodID setPlayoutSpeakerID = env->GetMethodID(_javaScClass,
- // "SetPlayoutSpeaker", "(Z)I");
-
- // call java sc object method
- // jint res = env->CallIntMethod(_javaScObj, setPlayoutSpeakerID, enable);
- // if (res < 0)
- // {
- // WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
- // " SetPlayoutSpeaker failed (%d)", res);
- // return -1;
- // }
-
- _loudSpeakerOn = enable;
-
- // Detach this thread if it was attached
- // if (isAttached)
- // {
- // if (_javaVM->DetachCurrentThread() < 0)
- // {
- // WEBRTC_TRACE(kTraceWarning, kTraceUtility, -1,
- // " Could not detach thread from JVM");
- // }
- // }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::GetLoudspeakerStatus(
- bool& enabled) const {
-
- enabled = _loudSpeakerOn;
- return 0;
-}
-
-// ============================================================================
-// Private Methods
-// ============================================================================
-
-void AudioDeviceAndroidOpenSLES::PlayerSimpleBufferQueueCallback(
- SLAndroidSimpleBufferQueueItf queueItf,
- void *pContext) {
- AudioDeviceAndroidOpenSLES* ptrThis =
- static_cast<AudioDeviceAndroidOpenSLES*> (pContext);
- ptrThis->PlayerSimpleBufferQueueCallbackHandler(queueItf);
-}
-
-void AudioDeviceAndroidOpenSLES::PlayerSimpleBufferQueueCallbackHandler(
- SLAndroidSimpleBufferQueueItf queueItf) {
- WebRtc_Word32 res;
- //Lock();
- //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- //"_playQueueSeq (%u)", _playQueueSeq);
- if (_playing && (_playQueueSeq < N_PLAY_QUEUE_BUFFERS)) {
- //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
- //_id, "playout callback ");
- unsigned int noSamp10ms = _adbSampleRate / 100;
- // Max 10 ms @ samplerate kHz / 16 bit
- WebRtc_Word8 playBuffer[2 * noSamp10ms];
- int noSamplesOut = 0;
-
- // Assumption for implementation
- // assert(PLAYBUFSIZESAMPLES == noSamp10ms);
-
- // TODO(xians), update the playout delay
- //UnLock();
-
- noSamplesOut = _ptrAudioBuffer->RequestPlayoutData(noSamp10ms);
- //Lock();
- // Get data from Audio Device Buffer
- noSamplesOut = _ptrAudioBuffer->GetPlayoutData(playBuffer);
- // Cast OK since only equality comparison
- if (noSamp10ms != (unsigned int) noSamplesOut) {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "noSamp10ms (%u) != noSamplesOut (%d)", noSamp10ms,
- noSamplesOut);
-
- if (_playWarning > 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Pending play warning exists");
- }
- _playWarning = 1;
- }
- // Insert what we have in data buffer
- memcpy(_playQueueBuffer[_playQueueSeq], playBuffer, 2 * noSamplesOut);
- //UnLock();
-
- //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- //"_playQueueSeq (%u) noSamplesOut (%d)", _playQueueSeq, noSamplesOut);
- // write the buffer data we got from VoE into the device
- res
- = (*_slPlayerSimpleBufferQueue)->Enqueue(
- _slPlayerSimpleBufferQueue,
- _playQueueBuffer[_playQueueSeq],
- 2 * noSamplesOut);
- if (res != SL_RESULT_SUCCESS) {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " player simpler buffer queue Enqueue failed, %d",
- noSamplesOut);
- return;
- }
- // update the playout delay
- UpdatePlayoutDelay(noSamplesOut);
- // update the play buffer sequency
- _playQueueSeq = (_playQueueSeq + 1) % N_PLAY_QUEUE_BUFFERS;
- }
-}
-
-void AudioDeviceAndroidOpenSLES::RecorderSimpleBufferQueueCallback(
- SLAndroidSimpleBufferQueueItf queueItf,
- void *pContext) {
- AudioDeviceAndroidOpenSLES* ptrThis =
- static_cast<AudioDeviceAndroidOpenSLES*> (pContext);
- ptrThis->RecorderSimpleBufferQueueCallbackHandler(queueItf);
-}
-
-void AudioDeviceAndroidOpenSLES::RecorderSimpleBufferQueueCallbackHandler(
- SLAndroidSimpleBufferQueueItf queueItf) {
- WebRtc_Word32 res;
- //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- //" RecorderSimpleBufferQueueCallbackHandler");
- if (_recording) {
- // Insert all data in temp buffer into recording buffers
- // There is zero or one buffer partially full at any given time,
- // all others are full or empty
- // Full means filled with noSamp10ms samples.
-
- const unsigned int noSamp10ms = _adbSampleRate / 100;
- // WebRtc_UWord16 queuePos = 0;
- // WebRtc_UWord16 checkQueuePos = 0;
- unsigned int dataPos = 0;
- WebRtc_UWord16 bufPos = 0;
- WebRtc_Word16 insertPos = -1;
- unsigned int nCopy = 0; // Number of samples to copy
- // WebRtc_Word32 isData = 0;
-
- while (dataPos < noSamp10ms)//REC_BUF_SIZE_IN_SAMPLES) //noSamp10ms)
-
- {
- // Loop over all recording buffers or until we find the partially
- // full buffer
- // First choice is to insert into partially full buffer,
- // second choice is to insert into empty buffer
- bufPos = 0;
- insertPos = -1;
- nCopy = 0;
- while (bufPos < N_REC_BUFFERS)
- {
- if ((_recLength[bufPos] > 0) && (_recLength[bufPos]
- < noSamp10ms))
- {
- // Found the partially full buffer
- insertPos = static_cast<WebRtc_Word16> (bufPos);
- bufPos = N_REC_BUFFERS; // Don't need to search more
- }
- else if ((-1 == insertPos) && (0 == _recLength[bufPos]))
- {
- // Found an empty buffer
- insertPos = static_cast<WebRtc_Word16> (bufPos);
- }
- ++bufPos;
- }
-
- if (insertPos > -1)
- {
- // We found a non-full buffer, copy data from the buffer queue
- // o recBuffer
- unsigned int dataToCopy = noSamp10ms - dataPos;
- unsigned int currentRecLen = _recLength[insertPos];
- unsigned int roomInBuffer = noSamp10ms - currentRecLen;
- nCopy = (dataToCopy < roomInBuffer ? dataToCopy : roomInBuffer);
- memcpy(&_recBuffer[insertPos][currentRecLen],
- &_recQueueBuffer[_recQueueSeq][dataPos],
- nCopy * sizeof(short));
- if (0 == currentRecLen)
- {
- _recSeqNumber[insertPos] = _recCurrentSeq;
- ++_recCurrentSeq;
- }
- _recBufferTotalSize += nCopy;
- // Has to be done last to avoid interrupt problems
- // between threads
- _recLength[insertPos] += nCopy;
- dataPos += nCopy;
- }
- else
- {
- // Didn't find a non-full buffer
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
- _id, " Could not insert into recording buffer");
- if (_recWarning > 0)
- {
- WEBRTC_TRACE(kTraceWarning,
- kTraceAudioDevice, _id,
- " Pending rec warning exists");
- }
- _recWarning = 1;
- dataPos = noSamp10ms; // Don't try to insert more
- }
- }
-
- // clean the queue buffer
- // Start with empty buffer
- memset(_recQueueBuffer[_recQueueSeq], 0, 2 * REC_BUF_SIZE_IN_SAMPLES);
- // write the empty buffer to the queue
- res = (*_slRecorderSimpleBufferQueue)->Enqueue(
- _slRecorderSimpleBufferQueue,
- (void*) _recQueueBuffer[_recQueueSeq],
- 2 * noSamp10ms);
- if (res != SL_RESULT_SUCCESS) {
- return;
- }
- // update the rec queue seq
- _recQueueSeq = (_recQueueSeq + 1) % N_REC_QUEUE_BUFFERS;
- // wake up the recording thread
- _timeEventRec.Set();
- }
-}
-
-void AudioDeviceAndroidOpenSLES::CheckErr(SLresult res) {
- if (res != SL_RESULT_SUCCESS) {
- // Debug printing to be placed here
- exit(-1);
- }
-}
-
-void AudioDeviceAndroidOpenSLES::UpdatePlayoutDelay(WebRtc_UWord32 nSamplePlayed) {
- // currently just do some simple calculation, should we setup a timer for
- // the callback to have a more accurate delay
- // Android CCD asks for 10ms as the maximum warm output latency, so we
- // simply add (nPlayQueueBuffer -1 + 0.5)*10ms
- // This playout delay should be seldom changed
- _playoutDelay = (N_PLAY_QUEUE_BUFFERS - 0.5) * 10 + N_PLAY_QUEUE_BUFFERS
- * nSamplePlayed / (_adbSampleRate / 1000);
-}
-
-void AudioDeviceAndroidOpenSLES::UpdateRecordingDelay() {
- // // Android CCD asks for 10ms as the maximum warm input latency,
- // so we simply add 10ms
- _recordingDelay = 10;
- const WebRtc_UWord32 noSamp10ms = _adbSampleRate / 100;
- // if (_recBufferTotalSize > noSamp10ms)
- // {
- _recordingDelay += (N_REC_QUEUE_BUFFERS * noSamp10ms) / (_adbSampleRate
- / 1000);
- // }
-}
-
-WebRtc_Word32 AudioDeviceAndroidOpenSLES::InitSampleRate() {
-
- WebRtc_Word32 res(-1);
- WebRtc_Word32 numOutputs(0);
- WebRtc_UWord32 headsetDeviceID(0);
- WebRtc_UWord32 earpieceDeviceID(0);
- bool headsetAvailable(false);
- bool earpieceAvailable(false);
- bool foundSampleRate(false);
- WebRtc_UWord32 outputDeviceIDs[N_MAX_OUTPUT_DEVICES];
- SLAudioOutputDescriptor audioOutputDescriptor;
-
- if (_slEngineObject == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, " SL Object is NULL");
- return -1;
- }
-
- /* // Get the Audio IO DEVICE CAPABILITIES interface
- res = (*_slEngineObject)->GetInterface(_slEngineObject,
- SL_IID_AUDIOIODEVICECAPABILITIES, (void*)&_slAudioIODeviceCapabilities);
- if ( res != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get Device Capabilities interface");
- return -1;
- }
- numOutputs = N_MAX_OUTPUT_DEVICES;
- res = (*_slAudioIODeviceCapabilities)->GetAvailableAudioOutputs(
- _slAudioIODeviceCapabilities, &numOutputs, outputDeviceIDs);
- if ( res != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to get number of Output Devices");
- return -1;
- }
-
- // Search for headset output and phone handsfree speaker device,
- // we prefer headset to earpiece
- for (int i=0;i<numOutputs; i++)
- {
- res = (*_slAudioIODeviceCapabilities)->QueryAudioOutputCapabilities(
- _slAudioIODeviceCapabilities, outputDeviceIDs[i], &audioOutputDescriptor);
- if ( res != SL_RESULT_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to query Output Devices info");
- return -1;
- }
- if((audioOutputDescriptor.deviceConnection ==
- SL_DEVCONNECTION_ATTACHED_WIRED)&&
- (audioOutputDescriptor.deviceScope == SL_DEVSCOPE_USER)&&
- (audioOutputDescriptor.deviceLocation == SL_DEVLOCATION_HEADSET))
- {
- headsetDeviceID = outputDeviceIDs[i];
- headsetAvailable = true;
- break;
- }
- else if((audioOutputDescriptor.deviceConnection ==
- SL_DEVCONNECTION_INTEGRATED)&&
- (audioOutputDescriptor.deviceScope == SL_DEVSCOPE_ENVIRONMENT)&&
- (audioOutputDescriptor.deviceLocation == SL_DEVLOCATION_HANDSET))
- {
- earpieceDeviceID = outputDeviceIDs[i];
- earpieceAvailable = true;
- break;
- }
- }
- // Neither headset output nor phone handsfree speaker is available
- if (headsetAvailable == false && earpieceAvailable == false)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, " No playout device");
- return -1;
- }
-
- _sampleRateInMilliHz = SL_SAMPLINGRATE_16;
- if (audioOutputDescriptor.isFreqRangeContinuous == SL_BOOLEAN_FALSE)
- {
- while (!foundSampleRate)
- {
- for (int i=0; i<audioOutputDescriptor.numOfSamplingRatesSupported; i++)
- {
- if (audioOutputDescriptor.samplingRatesSupported[i]
- == _sampleRateInMilliHz) // supported sampling rate in milliHertz
- {
- switch (_sampleRateInMilliHz)
- {
- case SL_SAMPLINGRATE_44_1:
- _adbSampleRate = 44000;
- break;
- case SL_SAMPLINGRATE_16:
- _adbSampleRate = 16000;
- break;
- case SL_SAMPLINGRATE_8:
- _adbSampleRate = 8000;
- break;
- default: // error
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " SampleRate(%d) is not supported", _sampleRateInMilliHz);
- return -1;
- } // switch
- foundSampleRate = true;
- break;
- } //if(audioOutputDescriptor.samplingRatesSupported[i] == _sampleRate
- } //for (int i=0; i<audioOutputDescriptor.numOfSamplingRatesSupported; i++)
- switch (_sampleRateInMilliHz)
- {
- case SL_SAMPLINGRATE_16:
- _sampleRateInMilliHz = SL_SAMPLINGRATE_44_1;
- break;
- case SL_SAMPLINGRATE_44_1:
- _sampleRateInMilliHz = SL_SAMPLINGRATE_8;
- break;
- default:
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, " SampleRate is not supported");
- return -1;
- } // switch
- } // while (!foundSampleRate)
- }
- else //audioOutputDescriptor.isFreqRangeContinuous == SL_BOOLEAN_TRUE
- {
- // minSampleRate < SL_SAMPLINGRATE_16 < maxSampleRate
- if ((audioOutputDescriptor.minSampleRate < SL_SAMPLINGRATE_16) &&
- (SL_SAMPLINGRATE_16 < audioOutputDescriptor.maxSampleRate))
- {
- _adbSampleRate = 16000;
- } // minSampleRate < SL_SAMPLINGRATE_44_1 < maxSampleRate
- else if((audioOutputDescriptor.minSampleRate < SL_SAMPLINGRATE_44_1) &&
- (SL_SAMPLINGRATE_44_1 < audioOutputDescriptor.maxSampleRate))
- {
- _adbSampleRate = 44000;
- } // minSampleRate < SL_SAMPLINGRATE_8 < maxSampleRate
- else if ((audioOutputDescriptor.minSampleRate < SL_SAMPLINGRATE_8) &&
- (SL_SAMPLINGRATE_8 < audioOutputDescriptor.maxSampleRate))
- {
- _adbSampleRate = 8000;
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " SampleRate is not supported");
- return -1;
- }
- } // else
- */
- _samplingRateIn = SL_SAMPLINGRATE_16;
- _samplingRateOut = SL_SAMPLINGRATE_16;
- _adbSampleRate = 16000;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
- " sample rate set to (%d)", _adbSampleRate);
- return 0;
-
-}
-
-// ============================================================================
-// Thread Methods
-// ============================================================================
-
-bool AudioDeviceAndroidOpenSLES::RecThreadFunc(void* pThis) {
- return (static_cast<AudioDeviceAndroidOpenSLES*> (pThis)->RecThreadProcess());
-}
-
-bool AudioDeviceAndroidOpenSLES::RecThreadProcess() {
-
- // Lock();
- // Wait for 100ms for the signal from device callback
- // In case no callback comes in 100ms, we check the buffer anyway
- _timeEventRec.Wait(100);
-
- int bufPos = 0;
- unsigned int lowestSeq = 0;
- int lowestSeqBufPos = 0;
- bool foundBuf = true;
- const unsigned int noSamp10ms = _adbSampleRate / 100;
-
- while (foundBuf)
- {
- // Check if we have any buffer with data to insert into the
- // Audio Device Buffer,
- // and find the one with the lowest seq number
- foundBuf = false;
-
- for (bufPos = 0; bufPos < N_REC_BUFFERS; ++bufPos)
- {
- if (noSamp10ms == _recLength[bufPos])
- {
- if (!foundBuf) {
- lowestSeq = _recSeqNumber[bufPos];
- lowestSeqBufPos = bufPos;
- foundBuf = true;
- } else if (_recSeqNumber[bufPos] < lowestSeq)
- {
- lowestSeq = _recSeqNumber[bufPos];
- lowestSeqBufPos = bufPos;
- }
- }
- } // for
-
- // Insert data into the Audio Device Buffer if found any
- if (foundBuf)
- {
- UpdateRecordingDelay();
- // Set the recorded buffer
- _ptrAudioBuffer->SetRecordedBuffer(_recBuffer[lowestSeqBufPos],
- noSamp10ms);
-
- // Don't need to set the current mic level in ADB since we only
- // support digital AGC,
- // and besides we cannot get or set the iPhone mic level anyway.
-
- // Set VQE info, use clockdrift == 0
- _ptrAudioBuffer->SetVQEData(_playoutDelay, _recordingDelay, 0);
-
- // Deliver recorded samples at specified sample rate, mic level
- // etc. to the observer using callback
- //UnLock();
- _ptrAudioBuffer->DeliverRecordedData();
- //Lock();
-
- // Make buffer available
- _recSeqNumber[lowestSeqBufPos] = 0;
- _recBufferTotalSize -= _recLength[lowestSeqBufPos];
- // Must be done last to avoid interrupt problems between threads
- _recLength[lowestSeqBufPos] = 0;
- }
-
- } // while (foundBuf)
- //UnLock();
- return true;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/android/audio_device_android_opensles.h b/trunk/src/modules/audio_device/main/source/android/audio_device_android_opensles.h
deleted file mode 100644
index 974ae90..0000000
--- a/trunk/src/modules/audio_device/main/source/android/audio_device_android_opensles.h
+++ /dev/null
@@ -1,310 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ANDROID_OPENSLES_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ANDROID_OPENSLES_H
-
-#include "audio_device_generic.h"
-#include "critical_section_wrapper.h"
-
-#include <jni.h> // For accessing AudioDeviceAndroid.java
-#include <stdio.h>
-#include <stdlib.h>
-
-#include <SLES/OpenSLES.h>
-#include <SLES/OpenSLES_Android.h>
-#include <SLES/OpenSLES_AndroidConfiguration.h>
-
-namespace webrtc
-{
-class EventWrapper;
-
-const WebRtc_UWord32 N_MAX_INTERFACES = 3;
-const WebRtc_UWord32 N_MAX_OUTPUT_DEVICES = 6;
-const WebRtc_UWord32 N_MAX_INPUT_DEVICES = 3;
-
-const WebRtc_UWord32 N_REC_SAMPLES_PER_SEC = 16000;//44000; // Default fs
-const WebRtc_UWord32 N_PLAY_SAMPLES_PER_SEC = 16000;//44000; // Default fs
-
-const WebRtc_UWord32 N_REC_CHANNELS = 1; // default is mono recording
-const WebRtc_UWord32 N_PLAY_CHANNELS = 1; // default is mono playout
-
-const WebRtc_UWord32 REC_BUF_SIZE_IN_SAMPLES = 480; // Handle max 10 ms @ 48 kHz
-const WebRtc_UWord32 PLAY_BUF_SIZE_IN_SAMPLES = 480;
-
-// Number of the buffers in playout queue
-const WebRtc_UWord16 N_PLAY_QUEUE_BUFFERS = 2;
-// Number of buffers in recording queue
-const WebRtc_UWord16 N_REC_QUEUE_BUFFERS = 2;
-// Number of 10 ms recording blocks in rec buffer
-const WebRtc_UWord16 N_REC_BUFFERS = 20;
-
-class ThreadWrapper;
-
-class AudioDeviceAndroidOpenSLES: public AudioDeviceGeneric
-{
-public:
- AudioDeviceAndroidOpenSLES(const WebRtc_Word32 id);
- ~AudioDeviceAndroidOpenSLES();
-
- // Retrieve the currently utilized audio layer
- virtual WebRtc_Word32
- ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
-
- // Main initializaton and termination
- virtual WebRtc_Word32 Init();
- virtual WebRtc_Word32 Terminate();
- virtual bool Initialized() const;
-
- // Device enumeration
- virtual WebRtc_Word16 PlayoutDevices();
- virtual WebRtc_Word16 RecordingDevices();
- virtual WebRtc_Word32
- PlayoutDeviceName(WebRtc_UWord16 index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]);
- virtual WebRtc_Word32
- RecordingDeviceName(WebRtc_UWord16 index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]);
-
- // Device selection
- virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32
- SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
- virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32
- SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device);
-
- // Audio transport initialization
- virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
- virtual WebRtc_Word32 InitPlayout();
- virtual bool PlayoutIsInitialized() const;
- virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
- virtual WebRtc_Word32 InitRecording();
- virtual bool RecordingIsInitialized() const;
-
- // Audio transport control
- virtual WebRtc_Word32 StartPlayout();
- virtual WebRtc_Word32 StopPlayout();
- virtual bool Playing() const;
- virtual WebRtc_Word32 StartRecording();
- virtual WebRtc_Word32 StopRecording();
- virtual bool Recording() const;
-
- // Microphone Automatic Gain Control (AGC)
- virtual WebRtc_Word32 SetAGC(bool enable);
- virtual bool AGC() const;
-
- // Volume control based on the Windows Wave API (Windows only)
- virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
- WebRtc_UWord16 volumeRight);
- virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft,
- WebRtc_UWord16& volumeRight) const;
-
- // Audio mixer initialization
- virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
- virtual WebRtc_Word32 InitSpeaker();
- virtual bool SpeakerIsInitialized() const;
- SLPlayItf playItf;
- virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
- virtual WebRtc_Word32 InitMicrophone();
- virtual bool MicrophoneIsInitialized() const;
-
- // Speaker volume controls
- virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
- virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
- virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
- virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
- virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
-
- // Microphone volume controls
- virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
- virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
- virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
- virtual WebRtc_Word32
- MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
-
- // Speaker mute control
- virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
- virtual WebRtc_Word32 SetSpeakerMute(bool enable);
- virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
-
- // Microphone mute control
- virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
- virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
-
- // Microphone boost control
- virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
- virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
-
- // Stereo support
- virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
- virtual WebRtc_Word32 SetStereoPlayout(bool enable);
- virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
- virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
- virtual WebRtc_Word32 SetStereoRecording(bool enable);
- virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
-
- // Delay information and control
- virtual WebRtc_Word32
- SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
- WebRtc_UWord16 sizeMS);
- virtual WebRtc_Word32 PlayoutBuffer(AudioDeviceModule::BufferType& type,
- WebRtc_UWord16& sizeMS) const;
- virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
- virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
-
- // CPU load
- virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
-
- // Error and warning information
- virtual bool PlayoutWarning() const;
- virtual bool PlayoutError() const;
- virtual bool RecordingWarning() const;
- virtual bool RecordingError() const;
- virtual void ClearPlayoutWarning();
- virtual void ClearPlayoutError();
- virtual void ClearRecordingWarning();
- virtual void ClearRecordingError();
-
- // Attach audio buffer
- virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
- // Speaker audio routing
- virtual WebRtc_Word32 SetLoudspeakerStatus(bool enable);
- virtual WebRtc_Word32 GetLoudspeakerStatus(bool& enable) const;
-
-private:
- // Lock
- void Lock()
- {
- _critSect.Enter();
- };
- void UnLock()
- {
- _critSect.Leave();
- };
-
- static void PlayerSimpleBufferQueueCallback(
- SLAndroidSimpleBufferQueueItf queueItf,
- void *pContext);
- void PlayerSimpleBufferQueueCallbackHandler(
- SLAndroidSimpleBufferQueueItf queueItf);
- static void RecorderSimpleBufferQueueCallback(
- SLAndroidSimpleBufferQueueItf queueItf,
- void *pContext);
- void RecorderSimpleBufferQueueCallbackHandler(
- SLAndroidSimpleBufferQueueItf queueItf);
- void CheckErr(SLresult res);
-
- // Delay updates
- void UpdateRecordingDelay();
- void UpdatePlayoutDelay(WebRtc_UWord32 nSamplePlayed);
-
- // Init
- WebRtc_Word32 InitSampleRate();
-
- // Threads
- static bool RecThreadFunc(void*);
- static bool PlayThreadFunc(void*);
- bool RecThreadProcess();
- bool PlayThreadProcess();
-
- // Misc
- AudioDeviceBuffer* _ptrAudioBuffer;
- CriticalSectionWrapper& _critSect;
- WebRtc_Word32 _id;
-
- // audio unit
- SLObjectItf _slEngineObject;
-
- // playout device
- SLObjectItf _slPlayer;
- SLEngineItf _slEngine;
- SLPlayItf _slPlayerPlay;
- SLAndroidSimpleBufferQueueItf _slPlayerSimpleBufferQueue;
- SLObjectItf _slOutputMixObject;
- SLVolumeItf _slSpeakerVolume;
-
- // recording device
- SLObjectItf _slRecorder;
- SLRecordItf _slRecorderRecord;
- SLAudioIODeviceCapabilitiesItf _slAudioIODeviceCapabilities;
- SLAndroidSimpleBufferQueueItf _slRecorderSimpleBufferQueue;
- SLDeviceVolumeItf _slMicVolume;
-
- WebRtc_UWord32 _micDeviceId;
-
- // Events
- EventWrapper& _timeEventRec;
- // Threads
- ThreadWrapper* _ptrThreadRec;
- WebRtc_UWord32 _recThreadID;
- // TODO(xians), remove the following flag
- bool _recThreadIsInitialized;
-
- // Playout buffer
- WebRtc_Word8 _playQueueBuffer[N_PLAY_QUEUE_BUFFERS][2
- * PLAY_BUF_SIZE_IN_SAMPLES];
- WebRtc_UWord32 _playQueueSeq;
- // Recording buffer
- WebRtc_Word8 _recQueueBuffer[N_REC_QUEUE_BUFFERS][2
- * REC_BUF_SIZE_IN_SAMPLES];
- WebRtc_UWord32 _recQueueSeq;
- WebRtc_Word8 _recBuffer[N_REC_BUFFERS][2*REC_BUF_SIZE_IN_SAMPLES];
- WebRtc_UWord32 _recLength[N_REC_BUFFERS];
- WebRtc_UWord32 _recSeqNumber[N_REC_BUFFERS];
- WebRtc_UWord32 _recCurrentSeq;
- // Current total size all data in buffers, used for delay estimate
- WebRtc_UWord32 _recBufferTotalSize;
-
- // States
- bool _recordingDeviceIsSpecified;
- bool _playoutDeviceIsSpecified;
- bool _initialized;
- bool _recording;
- bool _playing;
- bool _recIsInitialized;
- bool _playIsInitialized;
- bool _micIsInitialized;
- bool _speakerIsInitialized;
-
- // Warnings and errors
- WebRtc_UWord16 _playWarning;
- WebRtc_UWord16 _playError;
- WebRtc_UWord16 _recWarning;
- WebRtc_UWord16 _recError;
-
- // Delay
- WebRtc_UWord16 _playoutDelay;
- WebRtc_UWord16 _recordingDelay;
-
- // AGC state
- bool _AGC;
-
- // The sampling rate to use with Audio Device Buffer
- WebRtc_UWord32 _adbSampleRate;
- // Stored device properties
- WebRtc_UWord32 _samplingRateIn; // Sampling frequency for Mic
- WebRtc_UWord32 _samplingRateOut; // Sampling frequency for Speaker
- WebRtc_UWord32 _maxSpeakerVolume; // The maximum speaker volume value
- WebRtc_UWord32 _minSpeakerVolume; // The minimum speaker volume value
- bool _loudSpeakerOn;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ANDROID_OPENSLES_H
diff --git a/trunk/src/modules/audio_device/main/source/android/audio_device_utility_android.cc b/trunk/src/modules/audio_device/main/source/android/audio_device_utility_android.cc
deleted file mode 100644
index 84b4c60..0000000
--- a/trunk/src/modules/audio_device/main/source/android/audio_device_utility_android.cc
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Android audio device utility implementation
- */
-
-#include "audio_device_utility_android.h"
-
-#include "critical_section_wrapper.h"
-#include "trace.h"
-
-namespace webrtc
-{
-
-AudioDeviceUtilityAndroid::AudioDeviceUtilityAndroid(const WebRtc_Word32 id) :
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()), _id(id),
- _lastError(AudioDeviceModule::kAdmErrNone)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
- "%s created", __FUNCTION__);
-}
-
-AudioDeviceUtilityAndroid::~AudioDeviceUtilityAndroid()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
- "%s destroyed", __FUNCTION__);
- {
- CriticalSectionScoped lock(_critSect);
- }
-
- delete &_critSect;
-}
-
-WebRtc_Word32 AudioDeviceUtilityAndroid::Init()
-{
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
- " OS info: %s", "Android");
-
- return 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/android/audio_device_utility_android.h b/trunk/src/modules/audio_device/main/source/android/audio_device_utility_android.h
deleted file mode 100644
index 81f685a..0000000
--- a/trunk/src/modules/audio_device/main/source/android/audio_device_utility_android.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Android audio device utility interface
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_ANDROID_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_ANDROID_H
-
-#include "audio_device_utility.h"
-#include "audio_device.h"
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-
-class AudioDeviceUtilityAndroid: public AudioDeviceUtility
-{
-public:
- AudioDeviceUtilityAndroid(const WebRtc_Word32 id);
- ~AudioDeviceUtilityAndroid();
-
- virtual WebRtc_Word32 Init();
-
-private:
- CriticalSectionWrapper& _critSect;
- WebRtc_Word32 _id;
- AudioDeviceModule::ErrorCode _lastError;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_ANDROID_H
diff --git a/trunk/src/modules/audio_device/main/source/android/org/webrtc/voiceengine/AudioDeviceAndroid.java b/trunk/src/modules/audio_device/main/source/android/org/webrtc/voiceengine/AudioDeviceAndroid.java
deleted file mode 100644
index b56085b..0000000
--- a/trunk/src/modules/audio_device/main/source/android/org/webrtc/voiceengine/AudioDeviceAndroid.java
+++ /dev/null
@@ -1,509 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Android audio device test app
- */
-
-package org.webrtc.voiceengine;
-
-import java.nio.ByteBuffer;
-import java.util.concurrent.locks.ReentrantLock;
-
-import android.content.Context;
-import android.media.AudioFormat;
-import android.media.AudioManager;
-import android.media.AudioRecord;
-import android.media.AudioTrack;
-import android.util.Log;
-
-
-class AudioDeviceAndroid {
- private AudioTrack _audioTrack = null;
- private AudioRecord _audioRecord = null;
-
- private Context _context;
- private AudioManager _audioManager;
-
- private ByteBuffer _playBuffer;
- private ByteBuffer _recBuffer;
- private byte[] _tempBufPlay;
- private byte[] _tempBufRec;
-
- private final ReentrantLock _playLock = new ReentrantLock();
- private final ReentrantLock _recLock = new ReentrantLock();
-
- private boolean _doPlayInit = true;
- private boolean _doRecInit = true;
- private boolean _isRecording = false;
- private boolean _isPlaying = false;
-
- private int _bufferedRecSamples = 0;
- private int _bufferedPlaySamples = 0;
- private int _playPosition = 0;
-
- AudioDeviceAndroid() {
- try {
- _playBuffer = ByteBuffer.allocateDirect(2 * 480); // Max 10 ms @ 48
- // kHz
- _recBuffer = ByteBuffer.allocateDirect(2 * 480); // Max 10 ms @ 48
- // kHz
- } catch (Exception e) {
- DoLog(e.getMessage());
- }
-
- _tempBufPlay = new byte[2 * 480];
- _tempBufRec = new byte[2 * 480];
- }
-
- @SuppressWarnings("unused")
- private int InitRecording(int audioSource, int sampleRate) {
- // get the minimum buffer size that can be used
- int minRecBufSize =
- AudioRecord.getMinBufferSize(sampleRate,
- AudioFormat.CHANNEL_CONFIGURATION_MONO,
- AudioFormat.ENCODING_PCM_16BIT);
-
- // DoLog("min rec buf size is " + minRecBufSize);
-
- // double size to be more safe
- int recBufSize = minRecBufSize * 2;
- _bufferedRecSamples = (5 * sampleRate) / 200;
- // DoLog("rough rec delay set to " + _bufferedRecSamples);
-
- // release the object
- if (_audioRecord != null) {
- _audioRecord.release();
- _audioRecord = null;
- }
-
- try {
- _audioRecord = new AudioRecord(
- audioSource,
- sampleRate,
- AudioFormat.CHANNEL_CONFIGURATION_MONO,
- AudioFormat.ENCODING_PCM_16BIT,
- recBufSize);
-
- } catch (Exception e) {
- DoLog(e.getMessage());
- return -1;
- }
-
- // check that the audioRecord is ready to be used
- if (_audioRecord.getState() != AudioRecord.STATE_INITIALIZED) {
- // DoLog("rec not initialized " + sampleRate);
- return -1;
- }
-
- // DoLog("rec sample rate set to " + sampleRate);
-
- return _bufferedRecSamples;
- }
-
- @SuppressWarnings("unused")
- private int StartRecording() {
- if (_isPlaying == false) {
- SetAudioMode(true);
- }
-
- // start recording
- try {
- _audioRecord.startRecording();
-
- } catch (IllegalStateException e) {
- e.printStackTrace();
- return -1;
- }
-
- _isRecording = true;
- return 0;
- }
-
- @SuppressWarnings("unused")
- private int InitPlayback(int sampleRate) {
- // get the minimum buffer size that can be used
- int minPlayBufSize =
- AudioTrack.getMinBufferSize(sampleRate,
- AudioFormat.CHANNEL_CONFIGURATION_MONO,
- AudioFormat.ENCODING_PCM_16BIT);
-
- // DoLog("min play buf size is " + minPlayBufSize);
-
- int playBufSize = minPlayBufSize;
- if (playBufSize < 6000) {
- playBufSize *= 2;
- }
- _bufferedPlaySamples = 0;
- // DoLog("play buf size is " + playBufSize);
-
- // release the object
- if (_audioTrack != null) {
- _audioTrack.release();
- _audioTrack = null;
- }
-
- try {
- _audioTrack = new AudioTrack(
- AudioManager.STREAM_VOICE_CALL,
- sampleRate,
- AudioFormat.CHANNEL_CONFIGURATION_MONO,
- AudioFormat.ENCODING_PCM_16BIT,
- playBufSize, AudioTrack.MODE_STREAM);
- } catch (Exception e) {
- DoLog(e.getMessage());
- return -1;
- }
-
- // check that the audioRecord is ready to be used
- if (_audioTrack.getState() != AudioTrack.STATE_INITIALIZED) {
- // DoLog("play not initialized " + sampleRate);
- return -1;
- }
-
- // DoLog("play sample rate set to " + sampleRate);
-
- if (_audioManager == null && _context != null) {
- _audioManager = (AudioManager)
- _context.getSystemService(Context.AUDIO_SERVICE);
- }
-
- // Return max playout volume
- if (_audioManager == null) {
- // Don't know the max volume but still init is OK for playout,
- // so we should not return error.
- return 0;
- }
- return _audioManager.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL);
- }
-
- @SuppressWarnings("unused")
- private int StartPlayback() {
- if (_isRecording == false) {
- SetAudioMode(true);
- }
-
- // start playout
- try {
- _audioTrack.play();
-
- } catch (IllegalStateException e) {
- e.printStackTrace();
- return -1;
- }
-
- _isPlaying = true;
- return 0;
- }
-
- @SuppressWarnings("unused")
- private int StopRecording() {
- _recLock.lock();
- try {
- // only stop if we are recording
- if (_audioRecord.getRecordingState() ==
- AudioRecord.RECORDSTATE_RECORDING) {
- // stop recording
- try {
- _audioRecord.stop();
- } catch (IllegalStateException e) {
- e.printStackTrace();
- return -1;
- }
- }
-
- // release the object
- _audioRecord.release();
- _audioRecord = null;
-
- } finally {
- // Ensure we always unlock, both for success, exception or error
- // return.
- _doRecInit = true;
- _recLock.unlock();
- }
-
- if (_isPlaying == false) {
- SetAudioMode(false);
- }
-
- _isRecording = false;
- return 0;
- }
-
- @SuppressWarnings("unused")
- private int StopPlayback() {
- _playLock.lock();
- try {
- // only stop if we are playing
- if (_audioTrack.getPlayState() == AudioTrack.PLAYSTATE_PLAYING) {
- // stop playout
- try {
- _audioTrack.stop();
- } catch (IllegalStateException e) {
- e.printStackTrace();
- return -1;
- }
-
- // flush the buffers
- _audioTrack.flush();
- }
-
- // release the object
- _audioTrack.release();
- _audioTrack = null;
-
- } finally {
- // Ensure we always unlock, both for success, exception or error
- // return.
- _doPlayInit = true;
- _playLock.unlock();
- }
-
- if (_isRecording == false) {
- SetAudioMode(false);
- }
-
- _isPlaying = false;
- return 0;
- }
-
- @SuppressWarnings("unused")
- private int PlayAudio(int lengthInBytes) {
-
- int bufferedSamples = 0;
-
- _playLock.lock();
- try {
- if (_audioTrack == null) {
- return -2; // We have probably closed down while waiting for
- // play lock
- }
-
- // Set priority, only do once
- if (_doPlayInit == true) {
- try {
- android.os.Process.setThreadPriority(
- android.os.Process.THREAD_PRIORITY_URGENT_AUDIO);
- } catch (Exception e) {
- DoLog("Set play thread priority failed: " + e.getMessage());
- }
- _doPlayInit = false;
- }
-
- int written = 0;
- _playBuffer.get(_tempBufPlay);
- written = _audioTrack.write(_tempBufPlay, 0, lengthInBytes);
- _playBuffer.rewind(); // Reset the position to start of buffer
-
- // DoLog("Wrote data to sndCard");
-
- // increase by number of written samples
- _bufferedPlaySamples += (written >> 1);
-
- // decrease by number of played samples
- int pos = _audioTrack.getPlaybackHeadPosition();
- if (pos < _playPosition) { // wrap or reset by driver
- _playPosition = 0; // reset
- }
- _bufferedPlaySamples -= (pos - _playPosition);
- _playPosition = pos;
-
- if (!_isRecording) {
- bufferedSamples = _bufferedPlaySamples;
- }
-
- if (written != lengthInBytes) {
- // DoLog("Could not write all data to sc (written = " + written
- // + ", length = " + lengthInBytes + ")");
- return -1;
- }
-
- } finally {
- // Ensure we always unlock, both for success, exception or error
- // return.
- _playLock.unlock();
- }
-
- return bufferedSamples;
- }
-
- @SuppressWarnings("unused")
- private int RecordAudio(int lengthInBytes) {
- _recLock.lock();
-
- try {
- if (_audioRecord == null) {
- return -2; // We have probably closed down while waiting for rec
- // lock
- }
-
- // Set priority, only do once
- if (_doRecInit == true) {
- try {
- android.os.Process.setThreadPriority(
- android.os.Process.THREAD_PRIORITY_URGENT_AUDIO);
- } catch (Exception e) {
- DoLog("Set rec thread priority failed: " + e.getMessage());
- }
- _doRecInit = false;
- }
-
- int readBytes = 0;
- _recBuffer.rewind(); // Reset the position to start of buffer
- readBytes = _audioRecord.read(_tempBufRec, 0, lengthInBytes);
- // DoLog("read " + readBytes + "from SC");
- _recBuffer.put(_tempBufRec);
-
- if (readBytes != lengthInBytes) {
- // DoLog("Could not read all data from sc (read = " + readBytes
- // + ", length = " + lengthInBytes + ")");
- return -1;
- }
-
- } catch (Exception e) {
- DoLogErr("RecordAudio try failed: " + e.getMessage());
-
- } finally {
- // Ensure we always unlock, both for success, exception or error
- // return.
- _recLock.unlock();
- }
-
- return (_bufferedPlaySamples);
- }
-
- @SuppressWarnings("unused")
- private int SetPlayoutSpeaker(boolean loudspeakerOn) {
- // create audio manager if needed
- if (_audioManager == null && _context != null) {
- _audioManager = (AudioManager)
- _context.getSystemService(Context.AUDIO_SERVICE);
- }
-
- if (_audioManager == null) {
- DoLogErr("Could not change audio routing - no audio manager");
- return -1;
- }
-
- int apiLevel = Integer.parseInt(android.os.Build.VERSION.SDK);
-
- if ((3 == apiLevel) || (4 == apiLevel)) {
- // 1.5 and 1.6 devices
- if (loudspeakerOn) {
- // route audio to back speaker
- _audioManager.setMode(AudioManager.MODE_NORMAL);
- } else {
- // route audio to earpiece
- _audioManager.setMode(AudioManager.MODE_IN_CALL);
- }
- } else {
- // 2.x devices
- if ((android.os.Build.BRAND.equals("Samsung") ||
- android.os.Build.BRAND.equals("samsung")) &&
- ((5 == apiLevel) || (6 == apiLevel) ||
- (7 == apiLevel))) {
- // Samsung 2.0, 2.0.1 and 2.1 devices
- if (loudspeakerOn) {
- // route audio to back speaker
- _audioManager.setMode(AudioManager.MODE_IN_CALL);
- _audioManager.setSpeakerphoneOn(loudspeakerOn);
- } else {
- // route audio to earpiece
- _audioManager.setSpeakerphoneOn(loudspeakerOn);
- _audioManager.setMode(AudioManager.MODE_NORMAL);
- }
- } else {
- // Non-Samsung and Samsung 2.2 and up devices
- _audioManager.setSpeakerphoneOn(loudspeakerOn);
- }
- }
-
- return 0;
- }
-
- @SuppressWarnings("unused")
- private int SetPlayoutVolume(int level) {
-
- // create audio manager if needed
- if (_audioManager == null && _context != null) {
- _audioManager = (AudioManager)
- _context.getSystemService(Context.AUDIO_SERVICE);
- }
-
- int retVal = -1;
-
- if (_audioManager != null) {
- _audioManager.setStreamVolume(AudioManager.STREAM_VOICE_CALL,
- level, 0);
- retVal = 0;
- }
-
- return retVal;
- }
-
- @SuppressWarnings("unused")
- private int GetPlayoutVolume() {
-
- // create audio manager if needed
- if (_audioManager == null && _context != null) {
- _audioManager = (AudioManager)
- _context.getSystemService(Context.AUDIO_SERVICE);
- }
-
- int level = -1;
-
- if (_audioManager != null) {
- level = _audioManager.getStreamVolume(
- AudioManager.STREAM_VOICE_CALL);
- }
-
- return level;
- }
-
- private void SetAudioMode(boolean startCall) {
- int apiLevel = Integer.parseInt(android.os.Build.VERSION.SDK);
-
- if (_audioManager == null && _context != null) {
- _audioManager = (AudioManager)
- _context.getSystemService(Context.AUDIO_SERVICE);
- }
-
- if (_audioManager == null) {
- DoLogErr("Could not set audio mode - no audio manager");
- return;
- }
-
- // ***IMPORTANT*** When the API level for honeycomb (H) has been
- // decided,
- // the condition should be changed to include API level 8 to H-1.
- if ((android.os.Build.BRAND.equals("Samsung") || android.os.Build.BRAND
- .equals("samsung")) && (8 == apiLevel)) {
- // Set Samsung specific VoIP mode for 2.2 devices
- int mode =
- (startCall ? 4 /* VoIP mode */
- : AudioManager.MODE_NORMAL);
- _audioManager.setMode(mode);
- if (_audioManager.getMode() != mode) {
- DoLogErr("Could not set audio mode for Samsung device");
- }
- }
- }
-
- final String logTag = "WebRTC AD java";
-
- private void DoLog(String msg) {
- Log.d(logTag, msg);
- }
-
- private void DoLogErr(String msg) {
- Log.e(logTag, msg);
- }
-}
diff --git a/trunk/src/modules/audio_device/main/source/audio_device.gypi b/trunk/src/modules/audio_device/main/source/audio_device.gypi
deleted file mode 100644
index 2f3f9b1..0000000
--- a/trunk/src/modules/audio_device/main/source/audio_device.gypi
+++ /dev/null
@@ -1,210 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'audio_device',
- 'type': '<(library)',
- 'dependencies': [
- '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '.',
- '../../../interface',
- '../interface',
- 'dummy', # dummy audio device
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../../../../',
- '../../../interface',
- '../interface',
- ],
- },
- # TODO(xians): Rename files to e.g. *_linux.{ext}, remove sources in conditions section
- 'sources': [
- '../interface/audio_device.h',
- '../interface/audio_device_defines.h',
- 'audio_device_buffer.cc',
- 'audio_device_buffer.h',
- 'audio_device_generic.cc',
- 'audio_device_generic.h',
- 'audio_device_utility.cc',
- 'audio_device_utility.h',
- 'audio_device_impl.cc',
- 'audio_device_impl.h',
- 'audio_device_config.h',
- 'dummy/audio_device_dummy.cc',
- 'dummy/audio_device_dummy.h',
- 'dummy/audio_device_utility_dummy.cc',
- 'dummy/audio_device_utility_dummy.h',
- ],
- 'conditions': [
- ['OS=="linux"', {
- 'include_dirs': [
- 'linux',
- ],
- }], # OS==linux
- ['OS=="mac"', {
- 'include_dirs': [
- 'mac',
- ],
- }], # OS==mac
- ['OS=="win"', {
- 'include_dirs': [
- 'win',
- '../../../../../..',
- ],
- }],
- ['OS=="android"', {
- 'include_dirs': [
- 'android',
- ],
- }], # OS==android
- ['include_internal_audio_device==0', {
- 'defines': [
- 'WEBRTC_DUMMY_AUDIO_BUILD',
- ],
- }],
- ['include_internal_audio_device==1', {
- 'sources': [
- 'linux/alsasymboltable_linux.cc',
- 'linux/alsasymboltable_linux.h',
- 'linux/audio_device_alsa_linux.cc',
- 'linux/audio_device_alsa_linux.h',
- 'linux/audio_device_utility_linux.cc',
- 'linux/audio_device_utility_linux.h',
- 'linux/audio_mixer_manager_alsa_linux.cc',
- 'linux/audio_mixer_manager_alsa_linux.h',
- 'linux/latebindingsymboltable_linux.cc',
- 'linux/latebindingsymboltable_linux.h',
- 'mac/audio_device_mac.cc',
- 'mac/audio_device_mac.h',
- 'mac/audio_device_utility_mac.cc',
- 'mac/audio_device_utility_mac.h',
- 'mac/audio_mixer_manager_mac.cc',
- 'mac/audio_mixer_manager_mac.h',
- 'mac/portaudio/pa_memorybarrier.h',
- 'mac/portaudio/pa_ringbuffer.c',
- 'mac/portaudio/pa_ringbuffer.h',
- 'win/audio_device_core_win.cc',
- 'win/audio_device_core_win.h',
- 'win/audio_device_wave_win.cc',
- 'win/audio_device_wave_win.h',
- 'win/audio_device_utility_win.cc',
- 'win/audio_device_utility_win.h',
- 'win/audio_mixer_manager_win.cc',
- 'win/audio_mixer_manager_win.h',
- 'android/audio_device_android_opensles.cc',
- 'android/audio_device_android_opensles.h',
- 'android/audio_device_utility_android.cc',
- 'android/audio_device_utility_android.h',
- ],
- 'conditions': [
- ['OS=="linux"', {
- 'defines': [
- 'LINUX_ALSA',
- ],
- 'link_settings': {
- 'libraries': [
- '-ldl',
- '-lasound',
- ],
- },
- 'conditions': [
- ['include_pulse_audio==1', {
- 'defines': [
- 'LINUX_PULSE',
- ],
- 'sources': [
- 'linux/audio_device_pulse_linux.cc',
- 'linux/audio_device_pulse_linux.h',
- 'linux/audio_mixer_manager_pulse_linux.cc',
- 'linux/audio_mixer_manager_pulse_linux.h',
- 'linux/pulseaudiosymboltable_linux.cc',
- 'linux/pulseaudiosymboltable_linux.h',
- ],
- 'link_settings': {
- 'libraries': [
- '-lpulse',
- ],
- },
- }],
- ],
- }],
- ['OS=="mac"', {
- 'link_settings': {
- 'libraries': [
- '$(SDKROOT)/System/Library/Frameworks/AudioToolbox.framework',
- '$(SDKROOT)/System/Library/Frameworks/CoreAudio.framework',
- ],
- },
- }],
- ['OS=="win"', {
- 'link_settings': {
- 'libraries': [
- # Required for the built-in WASAPI AEC.
- '-ldmoguids.lib',
- '-lwmcodecdspuuid.lib',
- '-lamstrmid.lib',
- '-lmsdmo.lib',
- ],
- },
- }],
- ], # conditions
- }], # include_internal_audio_device==1
- ], # conditions
- },
- ],
- # Exclude the test targets when building with chromium.
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'audio_device_test_api',
- 'type': 'executable',
- 'dependencies': [
- 'audio_device',
- 'webrtc_utility',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'sources': [
- '../test/audio_device_test_api.cc',
- '../test/audio_device_test_defines.h',
- ],
- },
- {
- 'target_name': 'audio_device_test_func',
- 'type': 'executable',
- 'dependencies': [
- 'audio_device',
- 'webrtc_utility',
- '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/../test/test.gyp:test_support',
- ],
- 'sources': [
- '../test/audio_device_test_func.cc',
- '../test/audio_device_test_defines.h',
- '../test/func_test_manager.cc',
- '../test/func_test_manager.h',
- ],
- },
- ],
- }],
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_buffer.cc b/trunk/src/modules/audio_device/main/source/audio_device_buffer.cc
deleted file mode 100644
index 6edec6a..0000000
--- a/trunk/src/modules/audio_device/main/source/audio_device_buffer.cc
+++ /dev/null
@@ -1,652 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "trace.h"
-#include "critical_section_wrapper.h"
-#include "audio_device_buffer.h"
-#include "audio_device_utility.h"
-#include "audio_device_config.h"
-
-#include <stdlib.h>
-#include <string.h>
-#include <cassert>
-
-#include "signal_processing_library.h"
-
-namespace webrtc {
-
-// ----------------------------------------------------------------------------
-// ctor
-// ----------------------------------------------------------------------------
-
-AudioDeviceBuffer::AudioDeviceBuffer() :
- _id(-1),
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _critSectCb(*CriticalSectionWrapper::CreateCriticalSection()),
- _ptrCbAudioTransport(NULL),
- _recSampleRate(0),
- _playSampleRate(0),
- _recChannels(0),
- _playChannels(0),
- _recChannel(AudioDeviceModule::kChannelBoth),
- _recBytesPerSample(0),
- _playBytesPerSample(0),
- _recSamples(0),
- _recSize(0),
- _playSamples(0),
- _playSize(0),
- _recFile(*FileWrapper::Create()),
- _playFile(*FileWrapper::Create()),
- _currentMicLevel(0),
- _newMicLevel(0),
- _playDelayMS(0),
- _recDelayMS(0),
- _clockDrift(0),
- _measureDelay(false), // should always be 'false' (EXPERIMENTAL)
- _pulseList(),
- _lastPulseTime(AudioDeviceUtility::GetTimeInMS())
-{
- // valid ID will be set later by SetId, use -1 for now
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s created", __FUNCTION__);
- memset(_recBuffer, 0, kMaxBufferSizeBytes);
- memset(_playBuffer, 0, kMaxBufferSizeBytes);
-}
-
-// ----------------------------------------------------------------------------
-// dtor
-// ----------------------------------------------------------------------------
-
-AudioDeviceBuffer::~AudioDeviceBuffer()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
- {
- CriticalSectionScoped lock(_critSect);
-
- _recFile.Flush();
- _recFile.CloseFile();
- delete &_recFile;
-
- _playFile.Flush();
- _playFile.CloseFile();
- delete &_playFile;
-
- _EmptyList();
- }
-
- delete &_critSect;
- delete &_critSectCb;
-}
-
-// ----------------------------------------------------------------------------
-// SetId
-// ----------------------------------------------------------------------------
-
-void AudioDeviceBuffer::SetId(WebRtc_UWord32 id)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "AudioDeviceBuffer::SetId(id=%d)", id);
- _id = id;
-}
-
-// ----------------------------------------------------------------------------
-// RegisterAudioCallback
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::RegisterAudioCallback(AudioTransport* audioCallback)
-{
-
- CriticalSectionScoped lock(_critSectCb);
- _ptrCbAudioTransport = audioCallback;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::InitPlayout()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_measureDelay)
- {
- _EmptyList();
- _lastPulseTime = AudioDeviceUtility::GetTimeInMS();
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::InitRecording()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_measureDelay)
- {
- _EmptyList();
- _lastPulseTime = AudioDeviceUtility::GetTimeInMS();
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingSampleRate
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::SetRecordingSampleRate(WebRtc_UWord32 fsHz)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetRecordingSampleRate(fsHz=%u)", fsHz);
-
- CriticalSectionScoped lock(_critSect);
- _recSampleRate = fsHz;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutSampleRate
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::SetPlayoutSampleRate(WebRtc_UWord32 fsHz)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetPlayoutSampleRate(fsHz=%u)", fsHz);
-
- CriticalSectionScoped lock(_critSect);
- _playSampleRate = fsHz;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingSampleRate
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::RecordingSampleRate() const
-{
- return _recSampleRate;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutSampleRate
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::PlayoutSampleRate() const
-{
- return _playSampleRate;
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingChannels
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::SetRecordingChannels(WebRtc_UWord8 channels)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetRecordingChannels(channels=%u)", channels);
-
- CriticalSectionScoped lock(_critSect);
- _recChannels = channels;
- _recBytesPerSample = 2*channels; // 16 bits per sample in mono, 32 bits in stereo
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutChannels
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::SetPlayoutChannels(WebRtc_UWord8 channels)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "AudioDeviceBuffer::SetPlayoutChannels(channels=%u)", channels);
-
- CriticalSectionScoped lock(_critSect);
- _playChannels = channels;
- // 16 bits per sample in mono, 32 bits in stereo
- _playBytesPerSample = 2*channels;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingChannel
-//
-// Select which channel to use while recording.
-// This API requires that stereo is enabled.
-//
-// Note that, the nChannel parameter in RecordedDataIsAvailable will be
-// set to 2 even for kChannelLeft and kChannelRight. However, nBytesPerSample
-// will be 2 instead of 4 four these cases.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::SetRecordingChannel(const AudioDeviceModule::ChannelType channel)
-{
- CriticalSectionScoped lock(_critSect);
-
- if (_recChannels == 1)
- {
- return -1;
- }
-
- if (channel == AudioDeviceModule::kChannelBoth)
- {
- // two bytes per channel
- _recBytesPerSample = 4;
- }
- else
- {
- // only utilize one out of two possible channels (left or right)
- _recBytesPerSample = 2;
- }
- _recChannel = channel;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingChannel
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::RecordingChannel(AudioDeviceModule::ChannelType& channel) const
-{
- channel = _recChannel;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingChannels
-// ----------------------------------------------------------------------------
-
-WebRtc_UWord8 AudioDeviceBuffer::RecordingChannels() const
-{
- return _recChannels;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutChannels
-// ----------------------------------------------------------------------------
-
-WebRtc_UWord8 AudioDeviceBuffer::PlayoutChannels() const
-{
- return _playChannels;
-}
-
-// ----------------------------------------------------------------------------
-// SetCurrentMicLevel
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::SetCurrentMicLevel(WebRtc_UWord32 level)
-{
- _currentMicLevel = level;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// NewMicLevel
-// ----------------------------------------------------------------------------
-
-WebRtc_UWord32 AudioDeviceBuffer::NewMicLevel() const
-{
- return _newMicLevel;
-}
-
-// ----------------------------------------------------------------------------
-// SetVQEData
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::SetVQEData(WebRtc_UWord32 playDelayMS, WebRtc_UWord32 recDelayMS, WebRtc_Word32 clockDrift)
-{
- if ((playDelayMS + recDelayMS) > 300)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "too long delay (play:%i rec:%i)", playDelayMS, recDelayMS, clockDrift);
- }
-
- _playDelayMS = playDelayMS;
- _recDelayMS = recDelayMS;
- _clockDrift = clockDrift;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StartInputFileRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::StartInputFileRecording(const WebRtc_Word8 fileName[kAdmMaxFileNameSize])
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- _recFile.Flush();
- _recFile.CloseFile();
-
- return (_recFile.OpenFile(fileName, false, false, false));
-}
-
-// ----------------------------------------------------------------------------
-// StopInputFileRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::StopInputFileRecording()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- _recFile.Flush();
- _recFile.CloseFile();
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StartOutputFileRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::StartOutputFileRecording(const WebRtc_Word8 fileName[kAdmMaxFileNameSize])
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- _playFile.Flush();
- _playFile.CloseFile();
-
- return (_playFile.OpenFile(fileName, false, false, false));
-}
-
-// ----------------------------------------------------------------------------
-// StopOutputFileRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::StopOutputFileRecording()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- _playFile.Flush();
- _playFile.CloseFile();
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordedBuffer
-//
-// Store recorded audio buffer in local memory ready for the actual
-// "delivery" using a callback.
-//
-// This method can also parse out left or right channel from a stereo
-// input signal, i.e., emulate mono.
-//
-// Examples:
-//
-// 16-bit,48kHz mono, 10ms => nSamples=480 => _recSize=2*480=960 bytes
-// 16-bit,48kHz stereo,10ms => nSamples=480 => _recSize=4*480=1920 bytes
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::SetRecordedBuffer(const WebRtc_Word8* audioBuffer, WebRtc_UWord32 nSamples)
-{
- CriticalSectionScoped lock(_critSect);
-
- if (_recBytesPerSample == 0)
- {
- assert(false);
- return -1;
- }
-
- _recSamples = nSamples;
- _recSize = _recBytesPerSample*nSamples; // {2,4}*nSamples
- if (_recSize > kMaxBufferSizeBytes)
- {
- assert(false);
- return -1;
- }
-
- if (nSamples != _recSamples)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "invalid number of recorded samples (%d)", nSamples);
- return -1;
- }
-
- if (_recChannel == AudioDeviceModule::kChannelBoth)
- {
- // (default) copy the complete input buffer to the local buffer
- memcpy(&_recBuffer[0], audioBuffer, _recSize);
- }
- else
- {
- WebRtc_Word16* ptr16In = (WebRtc_Word16*)audioBuffer;
- WebRtc_Word16* ptr16Out = (WebRtc_Word16*)&_recBuffer[0];
-
- if (AudioDeviceModule::kChannelRight == _recChannel)
- {
- ptr16In++;
- }
-
- // exctract left or right channel from input buffer to the local buffer
- for (WebRtc_UWord32 i = 0; i < _recSamples; i++)
- {
- *ptr16Out = *ptr16In;
- ptr16Out++;
- ptr16In++;
- ptr16In++;
- }
- }
-
- if (_recFile.Open())
- {
- // write to binary file in mono or stereo (interleaved)
- _recFile.Write(&_recBuffer[0], _recSize);
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// DeliverRecordedData
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::DeliverRecordedData()
-{
- CriticalSectionScoped lock(_critSectCb);
-
- // Ensure that user has initialized all essential members
- if ((_recSampleRate == 0) ||
- (_recSamples == 0) ||
- (_recBytesPerSample == 0) ||
- (_recChannels == 0))
- {
- assert(false);
- return -1;
- }
-
- if (_ptrCbAudioTransport == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to deliver recorded data (AudioTransport does not exist)");
- return 0;
- }
-
- WebRtc_Word32 res(0);
- WebRtc_UWord32 newMicLevel(0);
- WebRtc_UWord32 totalDelayMS = _playDelayMS +_recDelayMS;
-
- if (_measureDelay)
- {
- CriticalSectionScoped lock(_critSect);
-
- memset(&_recBuffer[0], 0, _recSize);
- WebRtc_UWord32 time = AudioDeviceUtility::GetTimeInMS();
- if (time - _lastPulseTime > 500)
- {
- _pulseList.PushBack(time);
- _lastPulseTime = time;
-
- WebRtc_Word16* ptr16 = (WebRtc_Word16*)&_recBuffer[0];
- *ptr16 = 30000;
- }
- }
-
- res = _ptrCbAudioTransport->RecordedDataIsAvailable(&_recBuffer[0],
- _recSamples,
- _recBytesPerSample,
- _recChannels,
- _recSampleRate,
- totalDelayMS,
- _clockDrift,
- _currentMicLevel,
- newMicLevel);
- if (res != -1)
- {
- _newMicLevel = newMicLevel;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RequestPlayoutData
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::RequestPlayoutData(WebRtc_UWord32 nSamples)
-{
- {
- CriticalSectionScoped lock(_critSect);
-
- // Ensure that user has initialized all essential members
- if ((_playBytesPerSample == 0) ||
- (_playChannels == 0) ||
- (_playSampleRate == 0))
- {
- assert(false);
- return -1;
- }
-
- _playSamples = nSamples;
- _playSize = _playBytesPerSample * nSamples; // {2,4}*nSamples
- if (_playSize > kMaxBufferSizeBytes)
- {
- assert(false);
- return -1;
- }
-
- if (nSamples != _playSamples)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "invalid number of samples to be played out (%d)", nSamples);
- return -1;
- }
- }
-
- WebRtc_UWord32 nSamplesOut(0);
-
- CriticalSectionScoped lock(_critSectCb);
-
- if (_ptrCbAudioTransport == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to feed data to playout (AudioTransport does not exist)");
- return 0;
- }
-
- if (_ptrCbAudioTransport)
- {
- WebRtc_UWord32 res(0);
-
- res = _ptrCbAudioTransport->NeedMorePlayData(_playSamples,
- _playBytesPerSample,
- _playChannels,
- _playSampleRate,
- &_playBuffer[0],
- nSamplesOut);
- if (res != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "NeedMorePlayData() failed");
- }
-
- // --- Experimental delay-measurement implementation
- // *** not be used in released code ***
-
- if (_measureDelay)
- {
- CriticalSectionScoped lock(_critSect);
-
- WebRtc_Word16 maxAbs = WebRtcSpl_MaxAbsValueW16((const WebRtc_Word16*)&_playBuffer[0], (WebRtc_Word16)nSamplesOut*_playChannels);
- if (maxAbs > 1000)
- {
- WebRtc_UWord32 nowTime = AudioDeviceUtility::GetTimeInMS();
-
- if (!_pulseList.Empty())
- {
- ListItem* item = _pulseList.First();
- if (item)
- {
- WebRtc_Word16 maxIndex = WebRtcSpl_MaxAbsIndexW16((const WebRtc_Word16*)&_playBuffer[0], (WebRtc_Word16)nSamplesOut*_playChannels);
- WebRtc_UWord32 pulseTime = item->GetUnsignedItem();
- WebRtc_UWord32 diff = nowTime - pulseTime + (10*maxIndex)/(nSamplesOut*_playChannels);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "diff time in playout delay (%d)", diff);
- }
- _pulseList.PopFront();
- }
- }
- }
- }
-
- return nSamplesOut;
-}
-
-// ----------------------------------------------------------------------------
-// GetPlayoutData
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceBuffer::GetPlayoutData(WebRtc_Word8* audioBuffer)
-{
- CriticalSectionScoped lock(_critSect);
-
- if (_playSize > kMaxBufferSizeBytes)
- {
- WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "_playSize %i exceeds "
- "kMaxBufferSizeBytes in AudioDeviceBuffer::GetPlayoutData", _playSize);
- assert(false);
- return -1;
- }
-
- memcpy(audioBuffer, &_playBuffer[0], _playSize);
-
- if (_playFile.Open())
- {
- // write to binary file in mono or stereo (interleaved)
- _playFile.Write(&_playBuffer[0], _playSize);
- }
-
- return _playSamples;
-}
-
-// ----------------------------------------------------------------------------
-// _EmptyList
-// ----------------------------------------------------------------------------
-
-void AudioDeviceBuffer::_EmptyList()
-{
- while (!_pulseList.Empty())
- {
- ListItem* item = _pulseList.First();
- if (item)
- {
- // WebRtc_UWord32 ts = item->GetUnsignedItem();
- }
- _pulseList.PopFront();
- }
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_buffer.h b/trunk/src/modules/audio_device/main/source/audio_device_buffer.h
deleted file mode 100644
index e7db275..0000000
--- a/trunk/src/modules/audio_device/main/source/audio_device_buffer.h
+++ /dev/null
@@ -1,122 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
-
-#include "typedefs.h"
-#include "../../../../common_audio/resampler/include/resampler.h"
-#include "file_wrapper.h"
-#include "audio_device.h"
-#include "list_wrapper.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-const WebRtc_UWord32 kPulsePeriodMs = 1000;
-const WebRtc_UWord32 kMaxBufferSizeBytes = 3840; // 10ms in stereo @ 96kHz
-
-class AudioDeviceObserver;
-class MediaFile;
-
-class AudioDeviceBuffer
-{
-public:
- void SetId(WebRtc_UWord32 id);
- WebRtc_Word32 RegisterAudioCallback(AudioTransport* audioCallback);
-
- WebRtc_Word32 InitPlayout();
- WebRtc_Word32 InitRecording();
-
- WebRtc_Word32 SetRecordingSampleRate(WebRtc_UWord32 fsHz);
- WebRtc_Word32 SetPlayoutSampleRate(WebRtc_UWord32 fsHz);
- WebRtc_Word32 RecordingSampleRate() const;
- WebRtc_Word32 PlayoutSampleRate() const;
-
- WebRtc_Word32 SetRecordingChannels(WebRtc_UWord8 channels);
- WebRtc_Word32 SetPlayoutChannels(WebRtc_UWord8 channels);
- WebRtc_UWord8 RecordingChannels() const;
- WebRtc_UWord8 PlayoutChannels() const;
- WebRtc_Word32 SetRecordingChannel(const AudioDeviceModule::ChannelType channel);
- WebRtc_Word32 RecordingChannel(AudioDeviceModule::ChannelType& channel) const;
-
- WebRtc_Word32 SetRecordedBuffer(const WebRtc_Word8* audioBuffer, WebRtc_UWord32 nSamples);
- WebRtc_Word32 SetCurrentMicLevel(WebRtc_UWord32 level);
- WebRtc_Word32 SetVQEData(WebRtc_UWord32 playDelayMS, WebRtc_UWord32 recDelayMS, WebRtc_Word32 clockDrift);
- WebRtc_Word32 DeliverRecordedData();
- WebRtc_UWord32 NewMicLevel() const;
-
- WebRtc_Word32 RequestPlayoutData(WebRtc_UWord32 nSamples);
- WebRtc_Word32 GetPlayoutData(WebRtc_Word8* audioBuffer);
-
- WebRtc_Word32 StartInputFileRecording(const WebRtc_Word8 fileName[kAdmMaxFileNameSize]);
- WebRtc_Word32 StopInputFileRecording();
- WebRtc_Word32 StartOutputFileRecording(const WebRtc_Word8 fileName[kAdmMaxFileNameSize]);
- WebRtc_Word32 StopOutputFileRecording();
-
- AudioDeviceBuffer();
- ~AudioDeviceBuffer();
-
-private:
- void _EmptyList();
-
-private:
- WebRtc_Word32 _id;
- CriticalSectionWrapper& _critSect;
- CriticalSectionWrapper& _critSectCb;
-
- AudioTransport* _ptrCbAudioTransport;
-
- WebRtc_UWord32 _recSampleRate;
- WebRtc_UWord32 _playSampleRate;
-
- WebRtc_UWord8 _recChannels;
- WebRtc_UWord8 _playChannels;
-
- // selected recording channel (left/right/both)
- AudioDeviceModule::ChannelType _recChannel;
-
- // 2 or 4 depending on mono or stereo
- WebRtc_UWord8 _recBytesPerSample;
- WebRtc_UWord8 _playBytesPerSample;
-
- // 10ms in stereo @ 96kHz
- WebRtc_Word8 _recBuffer[kMaxBufferSizeBytes];
-
- // one sample <=> 2 or 4 bytes
- WebRtc_UWord32 _recSamples;
- WebRtc_UWord32 _recSize; // in bytes
-
- // 10ms in stereo @ 96kHz
- WebRtc_Word8 _playBuffer[kMaxBufferSizeBytes];
-
- // one sample <=> 2 or 4 bytes
- WebRtc_UWord32 _playSamples;
- WebRtc_UWord32 _playSize; // in bytes
-
- FileWrapper& _recFile;
- FileWrapper& _playFile;
-
- WebRtc_UWord32 _currentMicLevel;
- WebRtc_UWord32 _newMicLevel;
-
- WebRtc_UWord32 _playDelayMS;
- WebRtc_UWord32 _recDelayMS;
-
- WebRtc_Word32 _clockDrift;
-
- bool _measureDelay;
- ListWrapper _pulseList;
- WebRtc_UWord32 _lastPulseTime;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_BUFFER_H
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_config.h b/trunk/src/modules/audio_device/main/source/audio_device_config.h
deleted file mode 100644
index 23b9d55..0000000
--- a/trunk/src/modules/audio_device/main/source/audio_device_config.h
+++ /dev/null
@@ -1,44 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H
-
-// Enumerators
-//
-enum { kAdmMaxIdleTimeProcess = 1000 };
-enum { GET_MIC_VOLUME_INTERVAL_MS = 1000 };
-
-// Platform specifics
-//
-#if defined(_WIN32)
-#if (_MSC_VER >= 1400)
-// Windows Core Audio is the default audio layer in Windows.
-// Only supported for VS 2005 and higher.
-#define WEBRTC_WINDOWS_CORE_AUDIO_BUILD
-#endif
-#endif
-
-#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
-#include <windows.h>
-#include <tchar.h>
-#include <strsafe.h>
-#define DEBUG_PRINT(...) \
-{ \
- TCHAR msg[256]; \
- StringCchPrintf(msg, 256, __VA_ARGS__); \
- OutputDebugString(msg); \
-}
-#else
-#define DEBUG_PRINT(exp) ((void)0)
-#endif
-
-#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CONFIG_H
-
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_generic.cc b/trunk/src/modules/audio_device/main/source/audio_device_generic.cc
deleted file mode 100644
index 7093d80..0000000
--- a/trunk/src/modules/audio_device/main/source/audio_device_generic.cc
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_device_generic.h"
-#include "trace.h"
-
-namespace webrtc {
-
-WebRtc_Word32 AudioDeviceGeneric::SetRecordingSampleRate(
- const WebRtc_UWord32 samplesPerSec)
-{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
- "Set recording sample rate not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceGeneric::SetPlayoutSampleRate(
- const WebRtc_UWord32 samplesPerSec)
-{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
- "Set playout sample rate not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceGeneric::SetLoudspeakerStatus(bool enable)
-{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
- "Set loudspeaker status not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceGeneric::GetLoudspeakerStatus(bool& enable) const
-{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
- "Get loudspeaker status not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceGeneric::ResetAudioDevice()
-{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
- "Reset audio device not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceGeneric::SoundDeviceControl(unsigned int par1,
- unsigned int par2, unsigned int par3, unsigned int par4)
-{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
- "Sound device control not supported on this platform");
- return -1;
-}
-
-int32_t AudioDeviceGeneric::EnableBuiltInAEC(bool enable)
-{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
- "Windows AEC not supported on this platform");
- return -1;
-}
-
-bool AudioDeviceGeneric::BuiltInAECIsEnabled() const
-{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
- "Windows AEC not supported on this platform");
- return false;
-}
-
-} // namespace webrtc
-
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_generic.h b/trunk/src/modules/audio_device/main/source/audio_device_generic.h
deleted file mode 100644
index 13d5f83..0000000
--- a/trunk/src/modules/audio_device/main/source/audio_device_generic.h
+++ /dev/null
@@ -1,183 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_GENERIC_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_GENERIC_H
-
-#include "audio_device.h"
-#include "audio_device_buffer.h"
-
-namespace webrtc {
-
-class AudioDeviceGeneric
-{
- public:
-
- // Retrieve the currently utilized audio layer
- virtual WebRtc_Word32 ActiveAudioLayer(
- AudioDeviceModule::AudioLayer& audioLayer) const = 0;
-
- // Main initializaton and termination
- virtual WebRtc_Word32 Init() = 0;
- virtual WebRtc_Word32 Terminate() = 0;
- virtual bool Initialized() const = 0;
-
- // Device enumeration
- virtual WebRtc_Word16 PlayoutDevices() = 0;
- virtual WebRtc_Word16 RecordingDevices() = 0;
- virtual WebRtc_Word32 PlayoutDeviceName(
- WebRtc_UWord16 index,
- WebRtc_Word8 name[kAdmMaxDeviceNameSize],
- WebRtc_Word8 guid[kAdmMaxGuidSize]) = 0;
- virtual WebRtc_Word32 RecordingDeviceName(
- WebRtc_UWord16 index,
- WebRtc_Word8 name[kAdmMaxDeviceNameSize],
- WebRtc_Word8 guid[kAdmMaxGuidSize]) = 0;
-
- // Device selection
- virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index) = 0;
- virtual WebRtc_Word32 SetPlayoutDevice(
- AudioDeviceModule::WindowsDeviceType device) = 0;
- virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index) = 0;
- virtual WebRtc_Word32 SetRecordingDevice(
- AudioDeviceModule::WindowsDeviceType device) = 0;
-
- // Audio transport initialization
- virtual WebRtc_Word32 PlayoutIsAvailable(bool& available) = 0;
- virtual WebRtc_Word32 InitPlayout() = 0;
- virtual bool PlayoutIsInitialized() const = 0;
- virtual WebRtc_Word32 RecordingIsAvailable(bool& available) = 0;
- virtual WebRtc_Word32 InitRecording() = 0;
- virtual bool RecordingIsInitialized() const = 0;
-
- // Audio transport control
- virtual WebRtc_Word32 StartPlayout() = 0;
- virtual WebRtc_Word32 StopPlayout() = 0;
- virtual bool Playing() const = 0;
- virtual WebRtc_Word32 StartRecording() = 0;
- virtual WebRtc_Word32 StopRecording() = 0;
- virtual bool Recording() const = 0;
-
- // Microphone Automatic Gain Control (AGC)
- virtual WebRtc_Word32 SetAGC(bool enable) = 0;
- virtual bool AGC() const = 0;
-
- // Volume control based on the Windows Wave API (Windows only)
- virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
- WebRtc_UWord16 volumeRight) = 0;
- virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft,
- WebRtc_UWord16& volumeRight) const = 0;
-
- // Audio mixer initialization
- virtual WebRtc_Word32 SpeakerIsAvailable(bool& available) = 0;
- virtual WebRtc_Word32 InitSpeaker() = 0;
- virtual bool SpeakerIsInitialized() const = 0;
- virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available) = 0;
- virtual WebRtc_Word32 InitMicrophone() = 0;
- virtual bool MicrophoneIsInitialized() const = 0;
-
- // Speaker volume controls
- virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available) = 0;
- virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume) = 0;
- virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const = 0;
- virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const = 0;
- virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const = 0;
- virtual WebRtc_Word32 SpeakerVolumeStepSize(
- WebRtc_UWord16& stepSize) const = 0;
-
- // Microphone volume controls
- virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available) = 0;
- virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume) = 0;
- virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const = 0;
- virtual WebRtc_Word32 MaxMicrophoneVolume(
- WebRtc_UWord32& maxVolume) const = 0;
- virtual WebRtc_Word32 MinMicrophoneVolume(
- WebRtc_UWord32& minVolume) const = 0;
- virtual WebRtc_Word32 MicrophoneVolumeStepSize(
- WebRtc_UWord16& stepSize) const = 0;
-
- // Speaker mute control
- virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available) = 0;
- virtual WebRtc_Word32 SetSpeakerMute(bool enable) = 0;
- virtual WebRtc_Word32 SpeakerMute(bool& enabled) const = 0;
-
- // Microphone mute control
- virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available) = 0;
- virtual WebRtc_Word32 SetMicrophoneMute(bool enable) = 0;
- virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const = 0;
-
- // Microphone boost control
- virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available) = 0;
- virtual WebRtc_Word32 SetMicrophoneBoost(bool enable) = 0;
- virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const = 0;
-
- // Stereo support
- virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available) = 0;
- virtual WebRtc_Word32 SetStereoPlayout(bool enable) = 0;
- virtual WebRtc_Word32 StereoPlayout(bool& enabled) const = 0;
- virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available) = 0;
- virtual WebRtc_Word32 SetStereoRecording(bool enable) = 0;
- virtual WebRtc_Word32 StereoRecording(bool& enabled) const = 0;
-
- // Delay information and control
- virtual WebRtc_Word32 SetPlayoutBuffer(
- const AudioDeviceModule::BufferType type,
- WebRtc_UWord16 sizeMS = 0) = 0;
- virtual WebRtc_Word32 PlayoutBuffer(
- AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const = 0;
- virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const = 0;
- virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const = 0;
-
- // CPU load
- virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const = 0;
-
- // Native sample rate controls (samples/sec)
- virtual WebRtc_Word32 SetRecordingSampleRate(
- const WebRtc_UWord32 samplesPerSec);
- virtual WebRtc_Word32 SetPlayoutSampleRate(
- const WebRtc_UWord32 samplesPerSec);
-
- // Speaker audio routing (for mobile devices)
- virtual WebRtc_Word32 SetLoudspeakerStatus(bool enable);
- virtual WebRtc_Word32 GetLoudspeakerStatus(bool& enable) const;
-
- // Reset Audio Device (for mobile devices)
- virtual WebRtc_Word32 ResetAudioDevice();
-
- // Sound Audio Device control (for WinCE only)
- virtual WebRtc_Word32 SoundDeviceControl(unsigned int par1 = 0,
- unsigned int par2 = 0,
- unsigned int par3 = 0,
- unsigned int par4 = 0);
-
- // Windows Core Audio only.
- virtual int32_t EnableBuiltInAEC(bool enable);
- virtual bool BuiltInAECIsEnabled() const;
-
-public:
- virtual bool PlayoutWarning() const = 0;
- virtual bool PlayoutError() const = 0;
- virtual bool RecordingWarning() const = 0;
- virtual bool RecordingError() const = 0;
- virtual void ClearPlayoutWarning() = 0;
- virtual void ClearPlayoutError() = 0;
- virtual void ClearRecordingWarning() = 0;
- virtual void ClearRecordingError() = 0;
-
-public:
- virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) = 0;
-
- virtual ~AudioDeviceGeneric() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_GENERIC_H
-
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_impl.cc b/trunk/src/modules/audio_device/main/source/audio_device_impl.cc
deleted file mode 100644
index 1c4c445..0000000
--- a/trunk/src/modules/audio_device/main/source/audio_device_impl.cc
+++ /dev/null
@@ -1,2062 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_device_impl.h"
-#include "audio_device_config.h"
-#include "system_wrappers/interface/ref_count.h"
-
-#include <assert.h>
-#include <string.h>
-
-#if defined(_WIN32)
- #include "audio_device_utility_win.h"
- #include "audio_device_wave_win.h"
- #if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
- #include "audio_device_core_win.h"
- #endif
-#elif defined(WEBRTC_ANDROID_OPENSLES)
- #include <stdlib.h>
- #include "audio_device_utility_android.h"
- #include "audio_device_android_opensles.h"
-#elif defined(WEBRTC_ANDROID)
- #include <stdlib.h>
- #include "audio_device_utility_android.h"
- #include "audio_device_android_jni.h"
-#elif defined(WEBRTC_LINUX)
- #include "audio_device_utility_linux.h"
- #if defined(LINUX_ALSA)
- #include "audio_device_alsa_linux.h"
- #endif
- #if defined(LINUX_PULSE)
- #include "audio_device_pulse_linux.h"
- #endif
-#elif defined(MAC_IPHONE)
- #include "audio_device_utility_iphone.h"
- #include "audio_device_iphone.h"
-#elif (defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_MAC))
- #include "audio_device_utility_mac.h"
- #include "audio_device_mac.h"
-#endif
-#include "audio_device_dummy.h"
-#include "audio_device_utility_dummy.h"
-#include "critical_section_wrapper.h"
-#include "trace.h"
-
-#define CHECK_INITIALIZED() \
-{ \
- if (!_initialized) { \
- return -1; \
- }; \
-}
-
-#define CHECK_INITIALIZED_BOOL() \
-{ \
- if (!_initialized) { \
- return false; \
- }; \
-}
-
-namespace webrtc
-{
-
-AudioDeviceModule* CreateAudioDeviceModule(
- WebRtc_Word32 id, AudioDeviceModule::AudioLayer audioLayer) {
- return AudioDeviceModuleImpl::Create(id, audioLayer);
-}
-
-
-// ============================================================================
-// Static methods
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// AudioDeviceModule::Create()
-// ----------------------------------------------------------------------------
-
-AudioDeviceModule* AudioDeviceModuleImpl::Create(const WebRtc_Word32 id,
- const AudioLayer audioLayer)
-{
-
- // Create the generic ref counted (platform independent) implementation.
- RefCountImpl<AudioDeviceModuleImpl>* audioDevice =
- new RefCountImpl<AudioDeviceModuleImpl>(id, audioLayer);
-
- // Ensure that the current platform is supported.
- if (audioDevice->CheckPlatform() == -1)
- {
- delete audioDevice;
- return NULL;
- }
-
- // Create the platform-dependent implementation.
- if (audioDevice->CreatePlatformSpecificObjects() == -1)
- {
- delete audioDevice;
- return NULL;
- }
-
- // Ensure that the generic audio buffer can communicate with the
- // platform-specific parts.
- if (audioDevice->AttachAudioBuffer() == -1)
- {
- delete audioDevice;
- return NULL;
- }
-
- return audioDevice;
-}
-
-// ============================================================================
-// Construction & Destruction
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// AudioDeviceModuleImpl - ctor
-// ----------------------------------------------------------------------------
-
-AudioDeviceModuleImpl::AudioDeviceModuleImpl(const WebRtc_Word32 id, const AudioLayer audioLayer) :
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _critSectEventCb(*CriticalSectionWrapper::CreateCriticalSection()),
- _critSectAudioCb(*CriticalSectionWrapper::CreateCriticalSection()),
- _ptrCbAudioDeviceObserver(NULL),
- _ptrAudioDeviceUtility(NULL),
- _ptrAudioDevice(NULL),
- _id(id),
- _platformAudioLayer(audioLayer),
- _lastProcessTime(AudioDeviceUtility::GetTimeInMS()),
- _platformType(kPlatformNotSupported),
- _initialized(false),
- _lastError(kAdmErrNone)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
-}
-
-// ----------------------------------------------------------------------------
-// CheckPlatform
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::CheckPlatform()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- // Ensure that the current platform is supported
- //
- PlatformType platform(kPlatformNotSupported);
-
-#if defined(_WIN32)
- platform = kPlatformWin32;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "current platform is WIN32");
-#elif defined(WEBRTC_ANDROID)
- platform = kPlatformAndroid;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "current platform is ANDROID");
-#elif defined(WEBRTC_LINUX)
- platform = kPlatformLinux;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "current platform is LINUX");
-#elif (defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_MAC))
- platform = kPlatformMac;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "current platform is MAC");
-#endif
-
- if (platform == kPlatformNotSupported)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "current platform is not supported => this module will self destruct!");
- return -1;
- }
-
- // Store valid output results
- //
- _platformType = platform;
-
- return 0;
-}
-
-
-// ----------------------------------------------------------------------------
-// CreatePlatformSpecificObjects
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::CreatePlatformSpecificObjects()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- AudioDeviceGeneric* ptrAudioDevice(NULL);
- AudioDeviceUtility* ptrAudioDeviceUtility(NULL);
-
-#if defined(WEBRTC_DUMMY_AUDIO_BUILD)
- ptrAudioDevice = new AudioDeviceDummy(Id());
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Dummy Audio APIs will be utilized");
-
- if (ptrAudioDevice != NULL)
- {
- ptrAudioDeviceUtility = new AudioDeviceUtilityDummy(Id());
- }
-#else
- const AudioLayer audioLayer(PlatformAudioLayer());
-
- // Create the *Windows* implementation of the Audio Device
- //
-#if defined(_WIN32)
- if ((audioLayer == kWindowsWaveAudio)
-#if !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
- // Wave audio is default if Core audio is not supported in this build
- || (audioLayer == kPlatformDefaultAudio)
-#endif
- )
- {
- // create *Windows Wave Audio* implementation
- ptrAudioDevice = new AudioDeviceWindowsWave(Id());
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Windows Wave APIs will be utilized");
- }
-#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
- if ((audioLayer == kWindowsCoreAudio) ||
- (audioLayer == kPlatformDefaultAudio)
- )
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "attempting to use the Windows Core Audio APIs...");
-
- if (AudioDeviceWindowsCore::CoreAudioIsSupported())
- {
- // create *Windows Core Audio* implementation
- ptrAudioDevice = new AudioDeviceWindowsCore(Id());
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Windows Core Audio APIs will be utilized");
- }
- else
- {
- // create *Windows Wave Audio* implementation
- ptrAudioDevice = new AudioDeviceWindowsWave(Id());
- if (ptrAudioDevice != NULL)
- {
- // Core Audio was not supported => revert to Windows Wave instead
- _platformAudioLayer = kWindowsWaveAudio; // modify the state set at construction
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "Windows Core Audio is *not* supported => Wave APIs will be utilized instead");
- }
- }
- }
-#endif // defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
- if (ptrAudioDevice != NULL)
- {
- // Create the Windows implementation of the Device Utility.
- // This class is independent of the selected audio layer
- // for Windows.
- //
- ptrAudioDeviceUtility = new AudioDeviceUtilityWindows(Id());
- }
-#endif // #if defined(_WIN32)
-
- // Create the *Android OpenSLES* implementation of the Audio Device
- //
-#if defined(WEBRTC_ANDROID_OPENSLES)
- if (audioLayer == kPlatformDefaultAudio)
- {
- // Create *Android OpenELSE Audio* implementation
- ptrAudioDevice = new AudioDeviceAndroidOpenSLES(Id());
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Android OpenSLES Audio APIs will be utilized");
- }
-
- if (ptrAudioDevice != NULL)
- {
- // Create the Android implementation of the Device Utility.
- ptrAudioDeviceUtility = new AudioDeviceUtilityAndroid(Id());
- }
- // END #if defined(WEBRTC_ANDROID_OPENSLES)
-
- // Create the *Android Java* implementation of the Audio Device
- //
-#elif defined(WEBRTC_ANDROID)
- if (audioLayer == kPlatformDefaultAudio)
- {
- // Create *Android JNI Audio* implementation
- ptrAudioDevice = new AudioDeviceAndroidJni(Id());
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Android JNI Audio APIs will be utilized");
- }
-
- if (ptrAudioDevice != NULL)
- {
- // Create the Android implementation of the Device Utility.
- ptrAudioDeviceUtility = new AudioDeviceUtilityAndroid(Id());
- }
- // END #if defined(WEBRTC_ANDROID)
-
- // Create the *Linux* implementation of the Audio Device
- //
-#elif defined(WEBRTC_LINUX)
- if ((audioLayer == kLinuxPulseAudio) || (audioLayer == kPlatformDefaultAudio))
- {
-#if defined(LINUX_PULSE)
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "attempting to use the Linux PulseAudio APIs...");
-
- if (AudioDeviceLinuxPulse::PulseAudioIsSupported())
- {
- // create *Linux PulseAudio* implementation
- ptrAudioDevice = new AudioDeviceLinuxPulse(Id());
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Linux PulseAudio APIs will be utilized");
- }
- else
- {
-#endif
-#if defined(LINUX_ALSA)
- // create *Linux ALSA Audio* implementation
- ptrAudioDevice = new AudioDeviceLinuxALSA(Id());
- if (ptrAudioDevice != NULL)
- {
- // Pulse Audio was not supported => revert to ALSA instead
- _platformAudioLayer = kLinuxAlsaAudio; // modify the state set at construction
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "Linux PulseAudio is *not* supported => ALSA APIs will be utilized instead");
- }
-#endif
-#if defined(LINUX_PULSE)
- }
-#endif
- }
- else if (audioLayer == kLinuxAlsaAudio)
- {
-#if defined(LINUX_ALSA)
- // create *Linux ALSA Audio* implementation
- ptrAudioDevice = new AudioDeviceLinuxALSA(Id());
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Linux ALSA APIs will be utilized");
-#endif
- }
-
- if (ptrAudioDevice != NULL)
- {
- // Create the Linux implementation of the Device Utility.
- // This class is independent of the selected audio layer
- // for Linux.
- //
- ptrAudioDeviceUtility = new AudioDeviceUtilityLinux(Id());
- }
-#endif // #if defined(WEBRTC_LINUX)
-
- // Create the *iPhone* implementation of the Audio Device
- //
-#if defined(MAC_IPHONE)
- if (audioLayer == kPlatformDefaultAudio)
- {
- // Create *iPhone Audio* implementation
- ptrAudioDevice = new AudioDeviceIPhone(Id());
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "iPhone Audio APIs will be utilized");
- }
-
- if (ptrAudioDevice != NULL)
- {
- // Create the Mac implementation of the Device Utility.
- ptrAudioDeviceUtility = new AudioDeviceUtilityIPhone(Id());
- }
- // END #if defined(MAC_IPHONE)
-
- // Create the *Mac* implementation of the Audio Device
- //
-#elif defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_MAC)
- if (audioLayer == kPlatformDefaultAudio)
- {
- // Create *Mac Audio* implementation
- ptrAudioDevice = new AudioDeviceMac(Id());
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Mac OS X Audio APIs will be utilized");
- }
-
- if (ptrAudioDevice != NULL)
- {
- // Create the Mac implementation of the Device Utility.
- ptrAudioDeviceUtility = new AudioDeviceUtilityMac(Id());
- }
-#endif // #if defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_MAC)
-
- // Create the *Dummy* implementation of the Audio Device
- // Available for all platforms
- //
- if (audioLayer == kDummyAudio)
- {
- // Create *Dummy Audio* implementation
- assert(!ptrAudioDevice);
- ptrAudioDevice = new AudioDeviceDummy(Id());
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Dummy Audio APIs will be utilized");
-
- if (ptrAudioDevice != NULL)
- {
- ptrAudioDeviceUtility = new AudioDeviceUtilityDummy(Id());
- }
- }
-#endif // if defined(WEBRTC_DUMMY_AUDIO_BUILD)
-
- if (ptrAudioDevice == NULL)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "unable to create the platform specific audio device implementation");
- return -1;
- }
-
- if (ptrAudioDeviceUtility == NULL)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "unable to create the platform specific audio device utility");
- return -1;
- }
-
- // Store valid output pointers
- //
- _ptrAudioDevice = ptrAudioDevice;
- _ptrAudioDeviceUtility = ptrAudioDeviceUtility;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// AttachAudioBuffer
-//
-// Install "bridge" between the platform implemetation and the generic
-// implementation. The "child" shall set the native sampling rate and the
-// number of channels in this function call.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::AttachAudioBuffer()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- _audioDeviceBuffer.SetId(_id);
- _ptrAudioDevice->AttachAudioBuffer(&_audioDeviceBuffer);
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// ~AudioDeviceModuleImpl - dtor
-// ----------------------------------------------------------------------------
-
-AudioDeviceModuleImpl::~AudioDeviceModuleImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
-
- if (_ptrAudioDevice)
- {
- delete _ptrAudioDevice;
- _ptrAudioDevice = NULL;
- }
-
- if (_ptrAudioDeviceUtility)
- {
- delete _ptrAudioDeviceUtility;
- _ptrAudioDeviceUtility = NULL;
- }
-
- delete &_critSect;
- delete &_critSectEventCb;
- delete &_critSectAudioCb;
-}
-
-// ============================================================================
-// Module
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// Module::ChangeUniqueId
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Module::TimeUntilNextProcess
-//
-// Returns the number of milliseconds until the module want a worker thread
-// to call Process().
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::TimeUntilNextProcess()
-{
- WebRtc_UWord32 now = AudioDeviceUtility::GetTimeInMS();
- WebRtc_Word32 deltaProcess = kAdmMaxIdleTimeProcess - (now - _lastProcessTime);
- return (deltaProcess);
-}
-
-// ----------------------------------------------------------------------------
-// Module::Process
-//
-// Check for posted error and warning reports. Generate callbacks if
-// new reports exists.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::Process()
-{
-
- _lastProcessTime = AudioDeviceUtility::GetTimeInMS();
-
- // kPlayoutWarning
- if (_ptrAudioDevice->PlayoutWarning())
- {
- CriticalSectionScoped lock(_critSectEventCb);
- if (_ptrCbAudioDeviceObserver)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "=> OnWarningIsReported(kPlayoutWarning)");
- _ptrCbAudioDeviceObserver->OnWarningIsReported(AudioDeviceObserver::kPlayoutWarning);
- }
- _ptrAudioDevice->ClearPlayoutWarning();
- }
-
- // kPlayoutError
- if (_ptrAudioDevice->PlayoutError())
- {
- CriticalSectionScoped lock(_critSectEventCb);
- if (_ptrCbAudioDeviceObserver)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "=> OnErrorIsReported(kPlayoutError)");
- _ptrCbAudioDeviceObserver->OnErrorIsReported(AudioDeviceObserver::kPlayoutError);
- }
- _ptrAudioDevice->ClearPlayoutError();
- }
-
- // kRecordingWarning
- if (_ptrAudioDevice->RecordingWarning())
- {
- CriticalSectionScoped lock(_critSectEventCb);
- if (_ptrCbAudioDeviceObserver)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "=> OnWarningIsReported(kRecordingWarning)");
- _ptrCbAudioDeviceObserver->OnWarningIsReported(AudioDeviceObserver::kRecordingWarning);
- }
- _ptrAudioDevice->ClearRecordingWarning();
- }
-
- // kRecordingError
- if (_ptrAudioDevice->RecordingError())
- {
- CriticalSectionScoped lock(_critSectEventCb);
- if (_ptrCbAudioDeviceObserver)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "=> OnErrorIsReported(kRecordingError)");
- _ptrCbAudioDeviceObserver->OnErrorIsReported(AudioDeviceObserver::kRecordingError);
- }
- _ptrAudioDevice->ClearRecordingError();
- }
-
- return 0;
-}
-
-// ============================================================================
-// Public API
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// ActiveAudioLayer
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::ActiveAudioLayer(AudioLayer* audioLayer) const
-{
-
- AudioLayer activeAudio;
-
- if (_ptrAudioDevice->ActiveAudioLayer(activeAudio) == -1)
- {
- return -1;
- }
-
- *audioLayer = activeAudio;
-
- if (*audioLayer == AudioDeviceModule::kWindowsWaveAudio)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kWindowsWaveAudio");
- }
- else if (*audioLayer == AudioDeviceModule::kWindowsCoreAudio)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kWindowsCoreAudio");
- }
- else if (*audioLayer == AudioDeviceModule::kLinuxAlsaAudio)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kLinuxAlsaAudio");
- }
- else
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: NOT_SUPPORTED");
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// LastError
-// ----------------------------------------------------------------------------
-
-AudioDeviceModule::ErrorCode AudioDeviceModuleImpl::LastError() const
-{
- return _lastError;
-}
-
-// ----------------------------------------------------------------------------
-// Init
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::Init()
-{
-
- if (_initialized)
- return 0;
-
- if (!_ptrAudioDeviceUtility)
- return -1;
-
- if (!_ptrAudioDevice)
- return -1;
-
- _ptrAudioDeviceUtility->Init();
-
- if (_ptrAudioDevice->Init() == -1)
- {
- return -1;
- }
-
- _initialized = true;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Terminate
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::Terminate()
-{
-
- if (!_initialized)
- return 0;
-
- if (_ptrAudioDevice->Terminate() == -1)
- {
- return -1;
- }
-
- _initialized = false;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Initialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceModuleImpl::Initialized() const
-{
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: %d", _initialized);
- return (_initialized);
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SpeakerIsAvailable(bool* available)
-{
- CHECK_INITIALIZED();
-
- bool isAvailable(0);
-
- if (_ptrAudioDevice->SpeakerIsAvailable(isAvailable) == -1)
- {
- return -1;
- }
-
- *available = isAvailable;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", available);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// InitSpeaker
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::InitSpeaker()
-{
- CHECK_INITIALIZED();
- return (_ptrAudioDevice->InitSpeaker());
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::MicrophoneIsAvailable(bool* available)
-{
- CHECK_INITIALIZED();
-
- bool isAvailable(0);
-
- if (_ptrAudioDevice->MicrophoneIsAvailable(isAvailable) == -1)
- {
- return -1;
- }
-
- *available = isAvailable;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// InitMicrophone
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::InitMicrophone()
-{
- CHECK_INITIALIZED();
- return (_ptrAudioDevice->InitMicrophone());
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolumeIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SpeakerVolumeIsAvailable(bool* available)
-{
- CHECK_INITIALIZED();
-
- bool isAvailable(0);
-
- if (_ptrAudioDevice->SpeakerVolumeIsAvailable(isAvailable) == -1)
- {
- return -1;
- }
-
- *available = isAvailable;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SetSpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetSpeakerVolume(WebRtc_UWord32 volume)
-{
- CHECK_INITIALIZED();
- return (_ptrAudioDevice->SetSpeakerVolume(volume));
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SpeakerVolume(WebRtc_UWord32* volume) const
-{
- CHECK_INITIALIZED();
-
- WebRtc_UWord32 level(0);
-
- if (_ptrAudioDevice->SpeakerVolume(level) == -1)
- {
- return -1;
- }
-
- *volume = level;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: volume=%u", *volume);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SetWaveOutVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetWaveOutVolume(WebRtc_UWord16 volumeLeft, WebRtc_UWord16 volumeRight)
-{
- CHECK_INITIALIZED();
- return (_ptrAudioDevice->SetWaveOutVolume(volumeLeft, volumeRight));
-}
-
-// ----------------------------------------------------------------------------
-// WaveOutVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::WaveOutVolume(WebRtc_UWord16* volumeLeft, WebRtc_UWord16* volumeRight) const
-{
- CHECK_INITIALIZED();
-
- WebRtc_UWord16 volLeft(0);
- WebRtc_UWord16 volRight(0);
-
- if (_ptrAudioDevice->WaveOutVolume(volLeft, volRight) == -1)
- {
- return -1;
- }
-
- *volumeLeft = volLeft;
- *volumeRight = volRight;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "outputs: volumeLeft=%u, volumeRight=%u",
- *volumeLeft, *volumeRight);
-
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceModuleImpl::SpeakerIsInitialized() const
-{
- CHECK_INITIALIZED_BOOL();
-
- bool isInitialized = _ptrAudioDevice->SpeakerIsInitialized();
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: %d", isInitialized);
- return (isInitialized);
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceModuleImpl::MicrophoneIsInitialized() const
-{
- CHECK_INITIALIZED_BOOL();
-
- bool isInitialized = _ptrAudioDevice->MicrophoneIsInitialized();
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: %d", isInitialized);
- return (isInitialized);
-}
-
-// ----------------------------------------------------------------------------
-// MaxSpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::MaxSpeakerVolume(WebRtc_UWord32* maxVolume) const
-{
- CHECK_INITIALIZED();
-
- WebRtc_UWord32 maxVol(0);
-
- if (_ptrAudioDevice->MaxSpeakerVolume(maxVol) == -1)
- {
- return -1;
- }
-
- *maxVolume = maxVol;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: maxVolume=%d", *maxVolume);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// MinSpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::MinSpeakerVolume(WebRtc_UWord32* minVolume) const
-{
- CHECK_INITIALIZED();
-
- WebRtc_UWord32 minVol(0);
-
- if (_ptrAudioDevice->MinSpeakerVolume(minVol) == -1)
- {
- return -1;
- }
-
- *minVolume = minVol;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: minVolume=%u", *minVolume);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolumeStepSize
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SpeakerVolumeStepSize(WebRtc_UWord16* stepSize) const
-{
- CHECK_INITIALIZED();
-
- WebRtc_UWord16 delta(0);
-
- if (_ptrAudioDevice->SpeakerVolumeStepSize(delta) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to retrieve the speaker-volume step size");
- return -1;
- }
-
- *stepSize = delta;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: stepSize=%u", *stepSize);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerMuteIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SpeakerMuteIsAvailable(bool* available)
-{
- CHECK_INITIALIZED();
-
- bool isAvailable(0);
-
- if (_ptrAudioDevice->SpeakerMuteIsAvailable(isAvailable) == -1)
- {
- return -1;
- }
-
- *available = isAvailable;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SetSpeakerMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetSpeakerMute(bool enable)
-{
- CHECK_INITIALIZED();
- return (_ptrAudioDevice->SetSpeakerMute(enable));
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SpeakerMute(bool* enabled) const
-{
- CHECK_INITIALIZED();
-
- bool muted(false);
-
- if (_ptrAudioDevice->SpeakerMute(muted) == -1)
- {
- return -1;
- }
-
- *enabled = muted;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: enabled=%u", *enabled);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneMuteIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::MicrophoneMuteIsAvailable(bool* available)
-{
- CHECK_INITIALIZED();
-
- bool isAvailable(0);
-
- if (_ptrAudioDevice->MicrophoneMuteIsAvailable(isAvailable) == -1)
- {
- return -1;
- }
-
- *available = isAvailable;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetMicrophoneMute(bool enable)
-{
- CHECK_INITIALIZED();
- return (_ptrAudioDevice->SetMicrophoneMute(enable));
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::MicrophoneMute(bool* enabled) const
-{
- CHECK_INITIALIZED();
-
- bool muted(false);
-
- if (_ptrAudioDevice->MicrophoneMute(muted) == -1)
- {
- return -1;
- }
-
- *enabled = muted;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: enabled=%u", *enabled);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneBoostIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::MicrophoneBoostIsAvailable(bool* available)
-{
- CHECK_INITIALIZED();
-
- bool isAvailable(0);
-
- if (_ptrAudioDevice->MicrophoneBoostIsAvailable(isAvailable) == -1)
- {
- return -1;
- }
-
- *available = isAvailable;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneBoost
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetMicrophoneBoost(bool enable)
-{
- CHECK_INITIALIZED();
- return (_ptrAudioDevice->SetMicrophoneBoost(enable));
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneBoost
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::MicrophoneBoost(bool* enabled) const
-{
- CHECK_INITIALIZED();
-
- bool onOff(false);
-
- if (_ptrAudioDevice->MicrophoneBoost(onOff) == -1)
- {
- return -1;
- }
-
- *enabled = onOff;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: enabled=%u", *enabled);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolumeIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::MicrophoneVolumeIsAvailable(bool* available)
-{
- CHECK_INITIALIZED();
-
- bool isAvailable(0);
-
- if (_ptrAudioDevice->MicrophoneVolumeIsAvailable(isAvailable) == -1)
- {
- return -1;
- }
-
- *available = isAvailable;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetMicrophoneVolume(WebRtc_UWord32 volume)
-{
- CHECK_INITIALIZED();
- return (_ptrAudioDevice->SetMicrophoneVolume(volume));
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::MicrophoneVolume(WebRtc_UWord32* volume) const
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
- CHECK_INITIALIZED();
-
- WebRtc_UWord32 level(0);
-
- if (_ptrAudioDevice->MicrophoneVolume(level) == -1)
- {
- return -1;
- }
-
- *volume = level;
-
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "output: volume=%u", *volume);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// StereoRecordingIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::StereoRecordingIsAvailable(bool* available) const
-{
- CHECK_INITIALIZED();
-
- bool isAvailable(0);
-
- if (_ptrAudioDevice->StereoRecordingIsAvailable(isAvailable) == -1)
- {
- return -1;
- }
-
- *available = isAvailable;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SetStereoRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetStereoRecording(bool enable)
-{
- CHECK_INITIALIZED();
-
- if (_ptrAudioDevice->RecordingIsInitialized())
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "recording in stereo is not supported");
- return -1;
- }
-
- if (_ptrAudioDevice->SetStereoRecording(enable) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to enable stereo recording");
- return -1;
- }
-
- WebRtc_Word8 nChannels(1);
- if (enable)
- {
- nChannels = 2;
- }
- _audioDeviceBuffer.SetRecordingChannels(nChannels);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::StereoRecording(bool* enabled) const
-{
- CHECK_INITIALIZED();
-
- bool stereo(false);
-
- if (_ptrAudioDevice->StereoRecording(stereo) == -1)
- {
- return -1;
- }
-
- *enabled = stereo;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: enabled=%u", *enabled);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingChannel
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetRecordingChannel(const ChannelType channel)
-{
- if (channel == kChannelBoth)
- {
- }
- else if (channel == kChannelLeft)
- {
- }
- else
- {
- }
- CHECK_INITIALIZED();
-
- bool stereo(false);
-
- if (_ptrAudioDevice->StereoRecording(stereo) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "recording in stereo is not supported");
- return -1;
- }
-
- return (_audioDeviceBuffer.SetRecordingChannel(channel));
-}
-
-// ----------------------------------------------------------------------------
-// RecordingChannel
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::RecordingChannel(ChannelType* channel) const
-{
- CHECK_INITIALIZED();
-
- ChannelType chType;
-
- if (_audioDeviceBuffer.RecordingChannel(chType) == -1)
- {
- return -1;
- }
-
- *channel = chType;
-
- if (*channel == kChannelBoth)
- {
- }
- else if (*channel == kChannelLeft)
- {
- }
- else
- {
- }
-
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// StereoPlayoutIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::StereoPlayoutIsAvailable(bool* available) const
-{
- CHECK_INITIALIZED();
-
- bool isAvailable(0);
-
- if (_ptrAudioDevice->StereoPlayoutIsAvailable(isAvailable) == -1)
- {
- return -1;
- }
-
- *available = isAvailable;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SetStereoPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetStereoPlayout(bool enable)
-{
- CHECK_INITIALIZED();
-
- if (_ptrAudioDevice->PlayoutIsInitialized())
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "unable to set stereo mode while playing side is initialized");
- return -1;
- }
-
- if (_ptrAudioDevice->SetStereoPlayout(enable))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "stereo playout is not supported");
- return -1;
- }
-
- WebRtc_Word8 nChannels(1);
- if (enable)
- {
- nChannels = 2;
- }
- _audioDeviceBuffer.SetPlayoutChannels(nChannels);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::StereoPlayout(bool* enabled) const
-{
- CHECK_INITIALIZED();
-
- bool stereo(false);
-
- if (_ptrAudioDevice->StereoPlayout(stereo) == -1)
- {
- return -1;
- }
-
- *enabled = stereo;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: enabled=%u", *enabled);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SetAGC
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetAGC(bool enable)
-{
- CHECK_INITIALIZED();
- return (_ptrAudioDevice->SetAGC(enable));
-}
-
-// ----------------------------------------------------------------------------
-// AGC
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceModuleImpl::AGC() const
-{
- CHECK_INITIALIZED_BOOL();
- return (_ptrAudioDevice->AGC());
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::PlayoutIsAvailable(bool* available)
-{
- CHECK_INITIALIZED();
-
- bool isAvailable(0);
-
- if (_ptrAudioDevice->PlayoutIsAvailable(isAvailable) == -1)
- {
- return -1;
- }
-
- *available = isAvailable;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// RecordingIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::RecordingIsAvailable(bool* available)
-{
- CHECK_INITIALIZED();
-
- bool isAvailable(0);
-
- if (_ptrAudioDevice->RecordingIsAvailable(isAvailable) == -1)
- {
- return -1;
- }
-
- *available = isAvailable;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: available=%d", *available);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// MaxMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::MaxMicrophoneVolume(WebRtc_UWord32* maxVolume) const
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
- CHECK_INITIALIZED();
-
- WebRtc_UWord32 maxVol(0);
-
- if (_ptrAudioDevice->MaxMicrophoneVolume(maxVol) == -1)
- {
- return -1;
- }
-
- *maxVolume = maxVol;
-
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "output: maxVolume=%d", *maxVolume);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// MinMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::MinMicrophoneVolume(WebRtc_UWord32* minVolume) const
-{
- CHECK_INITIALIZED();
-
- WebRtc_UWord32 minVol(0);
-
- if (_ptrAudioDevice->MinMicrophoneVolume(minVol) == -1)
- {
- return -1;
- }
-
- *minVolume = minVol;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: minVolume=%u", *minVolume);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolumeStepSize
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::MicrophoneVolumeStepSize(WebRtc_UWord16* stepSize) const
-{
- CHECK_INITIALIZED();
-
- WebRtc_UWord16 delta(0);
-
- if (_ptrAudioDevice->MicrophoneVolumeStepSize(delta) == -1)
- {
- return -1;
- }
-
- *stepSize = delta;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: stepSize=%u", *stepSize);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDevices
-// ----------------------------------------------------------------------------
-
-WebRtc_Word16 AudioDeviceModuleImpl::PlayoutDevices()
-{
- CHECK_INITIALIZED();
-
- WebRtc_UWord16 nPlayoutDevices = _ptrAudioDevice->PlayoutDevices();
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: #playout devices=%d", nPlayoutDevices);
- return ((WebRtc_Word16)(nPlayoutDevices));
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutDevice I (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetPlayoutDevice(WebRtc_UWord16 index)
-{
- CHECK_INITIALIZED();
- return (_ptrAudioDevice->SetPlayoutDevice(index));
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutDevice II (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetPlayoutDevice(WindowsDeviceType device)
-{
- if (device == kDefaultDevice)
- {
- }
- else
- {
- }
- CHECK_INITIALIZED();
-
- return (_ptrAudioDevice->SetPlayoutDevice(device));
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDeviceName
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::PlayoutDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
-{
- CHECK_INITIALIZED();
-
- if (name == NULL)
- {
- _lastError = kAdmErrArgument;
- return -1;
- }
-
- if (_ptrAudioDevice->PlayoutDeviceName(index, name, guid) == -1)
- {
- return -1;
- }
-
- if (name != NULL)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: name=%s", name);
- }
- if (guid != NULL)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: guid=%s", guid);
- }
-
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDeviceName
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::RecordingDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
-{
- CHECK_INITIALIZED();
-
- if (name == NULL)
- {
- _lastError = kAdmErrArgument;
- return -1;
- }
-
- if (_ptrAudioDevice->RecordingDeviceName(index, name, guid) == -1)
- {
- return -1;
- }
-
- if (name != NULL)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: name=%s", name);
- }
- if (guid != NULL)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: guid=%s", guid);
- }
-
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDevices
-// ----------------------------------------------------------------------------
-
-WebRtc_Word16 AudioDeviceModuleImpl::RecordingDevices()
-{
- CHECK_INITIALIZED();
-
- WebRtc_UWord16 nRecordingDevices = _ptrAudioDevice->RecordingDevices();
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: #recording devices=%d", nRecordingDevices);
- return ((WebRtc_Word16)nRecordingDevices);
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingDevice I (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetRecordingDevice(WebRtc_UWord16 index)
-{
- CHECK_INITIALIZED();
- return (_ptrAudioDevice->SetRecordingDevice(index));
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingDevice II (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetRecordingDevice(WindowsDeviceType device)
-{
- if (device == kDefaultDevice)
- {
- }
- else
- {
- }
- CHECK_INITIALIZED();
-
- return (_ptrAudioDevice->SetRecordingDevice(device));
-}
-
-// ----------------------------------------------------------------------------
-// InitPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::InitPlayout()
-{
- CHECK_INITIALIZED();
- _audioDeviceBuffer.InitPlayout();
- return (_ptrAudioDevice->InitPlayout());
-}
-
-// ----------------------------------------------------------------------------
-// InitRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::InitRecording()
-{
- CHECK_INITIALIZED();
- _audioDeviceBuffer.InitRecording();
- return (_ptrAudioDevice->InitRecording());
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceModuleImpl::PlayoutIsInitialized() const
-{
- CHECK_INITIALIZED_BOOL();
- return (_ptrAudioDevice->PlayoutIsInitialized());
-}
-
-// ----------------------------------------------------------------------------
-// RecordingIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceModuleImpl::RecordingIsInitialized() const
-{
- CHECK_INITIALIZED_BOOL();
- return (_ptrAudioDevice->RecordingIsInitialized());
-}
-
-// ----------------------------------------------------------------------------
-// StartPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::StartPlayout()
-{
- CHECK_INITIALIZED();
- return (_ptrAudioDevice->StartPlayout());
-}
-
-// ----------------------------------------------------------------------------
-// StopPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::StopPlayout()
-{
- CHECK_INITIALIZED();
- return (_ptrAudioDevice->StopPlayout());
-}
-
-// ----------------------------------------------------------------------------
-// Playing
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceModuleImpl::Playing() const
-{
- CHECK_INITIALIZED_BOOL();
- return (_ptrAudioDevice->Playing());
-}
-
-// ----------------------------------------------------------------------------
-// StartRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::StartRecording()
-{
- CHECK_INITIALIZED();
- return (_ptrAudioDevice->StartRecording());
-}
-// ----------------------------------------------------------------------------
-// StopRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::StopRecording()
-{
- CHECK_INITIALIZED();
- return (_ptrAudioDevice->StopRecording());
-}
-
-// ----------------------------------------------------------------------------
-// Recording
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceModuleImpl::Recording() const
-{
- CHECK_INITIALIZED_BOOL();
- return (_ptrAudioDevice->Recording());
-}
-
-// ----------------------------------------------------------------------------
-// RegisterEventObserver
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::RegisterEventObserver(AudioDeviceObserver* eventCallback)
-{
-
- CriticalSectionScoped lock(_critSectEventCb);
- _ptrCbAudioDeviceObserver = eventCallback;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RegisterAudioCallback
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::RegisterAudioCallback(AudioTransport* audioCallback)
-{
-
- CriticalSectionScoped lock(_critSectAudioCb);
- _audioDeviceBuffer.RegisterAudioCallback(audioCallback);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StartRawInputFileRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::StartRawInputFileRecording(const WebRtc_Word8 pcmFileNameUTF8[kAdmMaxFileNameSize])
-{
- CHECK_INITIALIZED();
-
- if (NULL == pcmFileNameUTF8)
- {
- return -1;
- }
-
- return (_audioDeviceBuffer.StartInputFileRecording(pcmFileNameUTF8));
-}
-
-// ----------------------------------------------------------------------------
-// StopRawInputFileRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::StopRawInputFileRecording()
-{
- CHECK_INITIALIZED();
-
- return (_audioDeviceBuffer.StopInputFileRecording());
-}
-
-// ----------------------------------------------------------------------------
-// StartRawOutputFileRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::StartRawOutputFileRecording(const WebRtc_Word8 pcmFileNameUTF8[kAdmMaxFileNameSize])
-{
- CHECK_INITIALIZED();
-
- if (NULL == pcmFileNameUTF8)
- {
- return -1;
- }
-
- return (_audioDeviceBuffer.StartOutputFileRecording(pcmFileNameUTF8));
-}
-
-// ----------------------------------------------------------------------------
-// StopRawOutputFileRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::StopRawOutputFileRecording()
-{
- CHECK_INITIALIZED();
-
- return (_audioDeviceBuffer.StopOutputFileRecording());
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutBuffer
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetPlayoutBuffer(const BufferType type, WebRtc_UWord16 sizeMS)
-{
- CHECK_INITIALIZED();
-
- if (_ptrAudioDevice->PlayoutIsInitialized())
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "unable to modify the playout buffer while playing side is initialized");
- return -1;
- }
-
- WebRtc_Word32 ret(0);
-
- if (kFixedBufferSize == type)
- {
- if (sizeMS < kAdmMinPlayoutBufferSizeMs || sizeMS > kAdmMaxPlayoutBufferSizeMs)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "size parameter is out of range");
- return -1;
- }
- }
-
- if ((ret = _ptrAudioDevice->SetPlayoutBuffer(type, sizeMS)) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to set the playout buffer (error: %d)", LastError());
- }
-
- return ret;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutBuffer
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::PlayoutBuffer(BufferType* type, WebRtc_UWord16* sizeMS) const
-{
- CHECK_INITIALIZED();
-
- BufferType bufType;
- WebRtc_UWord16 size(0);
-
- if (_ptrAudioDevice->PlayoutBuffer(bufType, size) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to retrieve the buffer type and size");
- return -1;
- }
-
- *type = bufType;
- *sizeMS = size;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: type=%u, sizeMS=%u", *type, *sizeMS);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDelay
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::PlayoutDelay(WebRtc_UWord16* delayMS) const
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
- CHECK_INITIALIZED();
-
- WebRtc_UWord16 delay(0);
-
- if (_ptrAudioDevice->PlayoutDelay(delay) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to retrieve the playout delay");
- return -1;
- }
-
- *delayMS = delay;
-
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "output: delayMS=%u", *delayMS);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDelay
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::RecordingDelay(WebRtc_UWord16* delayMS) const
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
- CHECK_INITIALIZED();
-
- WebRtc_UWord16 delay(0);
-
- if (_ptrAudioDevice->RecordingDelay(delay) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to retrieve the recording delay");
- return -1;
- }
-
- *delayMS = delay;
-
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "output: delayMS=%u", *delayMS);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// CPULoad
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::CPULoad(WebRtc_UWord16* load) const
-{
- CHECK_INITIALIZED();
-
- WebRtc_UWord16 cpuLoad(0);
-
- if (_ptrAudioDevice->CPULoad(cpuLoad) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to retrieve the CPU load");
- return -1;
- }
-
- *load = cpuLoad;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: load=%u", *load);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingSampleRate
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetRecordingSampleRate(const WebRtc_UWord32 samplesPerSec)
-{
- CHECK_INITIALIZED();
-
- if (_ptrAudioDevice->SetRecordingSampleRate(samplesPerSec) != 0)
- {
- return -1;
- }
-
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// RecordingSampleRate
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::RecordingSampleRate(WebRtc_UWord32* samplesPerSec) const
-{
- CHECK_INITIALIZED();
-
- WebRtc_Word32 sampleRate = _audioDeviceBuffer.RecordingSampleRate();
-
- if (sampleRate == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to retrieve the sample rate");
- return -1;
- }
-
- *samplesPerSec = sampleRate;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: samplesPerSec=%u", *samplesPerSec);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutSampleRate
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetPlayoutSampleRate(const WebRtc_UWord32 samplesPerSec)
-{
- CHECK_INITIALIZED();
-
- if (_ptrAudioDevice->SetPlayoutSampleRate(samplesPerSec) != 0)
- {
- return -1;
- }
-
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutSampleRate
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::PlayoutSampleRate(WebRtc_UWord32* samplesPerSec) const
-{
- CHECK_INITIALIZED();
-
- WebRtc_Word32 sampleRate = _audioDeviceBuffer.PlayoutSampleRate();
-
- if (sampleRate == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to retrieve the sample rate");
- return -1;
- }
-
- *samplesPerSec = sampleRate;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: samplesPerSec=%u", *samplesPerSec);
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// ResetAudioDevice
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::ResetAudioDevice()
-{
- CHECK_INITIALIZED();
-
-
- if (_ptrAudioDevice->ResetAudioDevice() == -1)
- {
- return -1;
- }
-
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SetLoudspeakerStatus
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::SetLoudspeakerStatus(bool enable)
-{
- CHECK_INITIALIZED();
-
- if (_ptrAudioDevice->SetLoudspeakerStatus(enable) != 0)
- {
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// GetLoudspeakerStatus
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceModuleImpl::GetLoudspeakerStatus(bool* enabled) const
-{
- CHECK_INITIALIZED();
-
- if (_ptrAudioDevice->GetLoudspeakerStatus(*enabled) != 0)
- {
- return -1;
- }
-
- return 0;
-}
-
-int32_t AudioDeviceModuleImpl::EnableBuiltInAEC(bool enable)
-{
- CHECK_INITIALIZED();
-
- return _ptrAudioDevice->EnableBuiltInAEC(enable);
-}
-
-bool AudioDeviceModuleImpl::BuiltInAECIsEnabled() const
-{
- CHECK_INITIALIZED_BOOL();
-
- return _ptrAudioDevice->BuiltInAECIsEnabled();
-}
-
-// ============================================================================
-// Private Methods
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// Platform
-// ----------------------------------------------------------------------------
-
-AudioDeviceModuleImpl::PlatformType AudioDeviceModuleImpl::Platform() const
-{
- return _platformType;
-}
-
-// ----------------------------------------------------------------------------
-// PlatformAudioLayer
-// ----------------------------------------------------------------------------
-
-AudioDeviceModule::AudioLayer AudioDeviceModuleImpl::PlatformAudioLayer() const
-{
-
- switch (_platformAudioLayer)
- {
- case kPlatformDefaultAudio:
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kPlatformDefaultAudio");
- break;
- case kWindowsWaveAudio:
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kWindowsWaveAudio");
- break;
- case kWindowsCoreAudio:
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kWindowsCoreAudio");
- break;
- case kLinuxAlsaAudio:
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kLinuxAlsaAudio");
- break;
- case kDummyAudio:
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, "output: kDummyAudio");
- break;
- default:
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "output: INVALID");
- break;
- }
-
- return _platformAudioLayer;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_impl.h b/trunk/src/modules/audio_device/main/source/audio_device_impl.h
deleted file mode 100644
index 5edec0d..0000000
--- a/trunk/src/modules/audio_device/main/source/audio_device_impl.h
+++ /dev/null
@@ -1,235 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IMPL_H
-
-#include "audio_device.h"
-#include "audio_device_buffer.h"
-
-namespace webrtc
-{
-
-class AudioDeviceGeneric;
-class AudioDeviceUtility;
-class CriticalSectionWrapper;
-
-class AudioDeviceModuleImpl : public AudioDeviceModule
-{
-public:
- enum PlatformType
- {
- kPlatformNotSupported = 0,
- kPlatformWin32 = 1,
- kPlatformWinCe = 2,
- kPlatformLinux = 3,
- kPlatformMac = 4,
- kPlatformAndroid = 5
- };
-
- WebRtc_Word32 CheckPlatform();
- WebRtc_Word32 CreatePlatformSpecificObjects();
- WebRtc_Word32 AttachAudioBuffer();
-
- AudioDeviceModuleImpl(const WebRtc_Word32 id, const AudioLayer audioLayer);
- virtual ~AudioDeviceModuleImpl();
-
-public: // RefCountedModule
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
- virtual WebRtc_Word32 TimeUntilNextProcess();
- virtual WebRtc_Word32 Process();
-
-public:
- // Factory methods (resource allocation/deallocation)
- static AudioDeviceModule* Create(
- const WebRtc_Word32 id,
- const AudioLayer audioLayer = kPlatformDefaultAudio);
-
- // Retrieve the currently utilized audio layer
- virtual WebRtc_Word32 ActiveAudioLayer(AudioLayer* audioLayer) const;
-
- // Error handling
- virtual ErrorCode LastError() const;
- virtual WebRtc_Word32 RegisterEventObserver(
- AudioDeviceObserver* eventCallback);
-
- // Full-duplex transportation of PCM audio
- virtual WebRtc_Word32 RegisterAudioCallback(
- AudioTransport* audioCallback);
-
- // Main initializaton and termination
- virtual WebRtc_Word32 Init();
- virtual WebRtc_Word32 Terminate();
- virtual bool Initialized() const;
-
- // Device enumeration
- virtual WebRtc_Word16 PlayoutDevices();
- virtual WebRtc_Word16 RecordingDevices();
- virtual WebRtc_Word32 PlayoutDeviceName(
- WebRtc_UWord16 index,
- WebRtc_Word8 name[kAdmMaxDeviceNameSize],
- WebRtc_Word8 guid[kAdmMaxGuidSize]);
- virtual WebRtc_Word32 RecordingDeviceName(
- WebRtc_UWord16 index,
- WebRtc_Word8 name[kAdmMaxDeviceNameSize],
- WebRtc_Word8 guid[kAdmMaxGuidSize]);
-
- // Device selection
- virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32 SetPlayoutDevice(WindowsDeviceType device);
- virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32 SetRecordingDevice(WindowsDeviceType device);
-
- // Audio transport initialization
- virtual WebRtc_Word32 PlayoutIsAvailable(bool* available);
- virtual WebRtc_Word32 InitPlayout();
- virtual bool PlayoutIsInitialized() const;
- virtual WebRtc_Word32 RecordingIsAvailable(bool* available);
- virtual WebRtc_Word32 InitRecording();
- virtual bool RecordingIsInitialized() const;
-
- // Audio transport control
- virtual WebRtc_Word32 StartPlayout();
- virtual WebRtc_Word32 StopPlayout();
- virtual bool Playing() const;
- virtual WebRtc_Word32 StartRecording();
- virtual WebRtc_Word32 StopRecording();
- virtual bool Recording() const;
-
- // Microphone Automatic Gain Control (AGC)
- virtual WebRtc_Word32 SetAGC(bool enable);
- virtual bool AGC() const;
-
- // Volume control based on the Windows Wave API (Windows only)
- virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
- WebRtc_UWord16 volumeRight);
- virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16* volumeLeft,
- WebRtc_UWord16* volumeRight) const;
-
- // Audio mixer initialization
- virtual WebRtc_Word32 SpeakerIsAvailable(bool* available);
- virtual WebRtc_Word32 InitSpeaker();
- virtual bool SpeakerIsInitialized() const;
- virtual WebRtc_Word32 MicrophoneIsAvailable(bool* available);
- virtual WebRtc_Word32 InitMicrophone();
- virtual bool MicrophoneIsInitialized() const;
-
- // Speaker volume controls
- virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool* available);
- virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32* volume) const;
- virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32* maxVolume) const;
- virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32* minVolume) const;
- virtual WebRtc_Word32 SpeakerVolumeStepSize(
- WebRtc_UWord16* stepSize) const;
-
- // Microphone volume controls
- virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool* available);
- virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32* volume) const;
- virtual WebRtc_Word32 MaxMicrophoneVolume(
- WebRtc_UWord32* maxVolume) const;
- virtual WebRtc_Word32 MinMicrophoneVolume(
- WebRtc_UWord32* minVolume) const;
- virtual WebRtc_Word32 MicrophoneVolumeStepSize(
- WebRtc_UWord16* stepSize) const;
-
- // Speaker mute control
- virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool* available);
- virtual WebRtc_Word32 SetSpeakerMute(bool enable);
- virtual WebRtc_Word32 SpeakerMute(bool* enabled) const;
-
- // Microphone mute control
- virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool* available);
- virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
- virtual WebRtc_Word32 MicrophoneMute(bool* enabled) const;
-
- // Microphone boost control
- virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool* available);
- virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
- virtual WebRtc_Word32 MicrophoneBoost(bool* enabled) const;
-
- // Stereo support
- virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool* available) const;
- virtual WebRtc_Word32 SetStereoPlayout(bool enable);
- virtual WebRtc_Word32 StereoPlayout(bool* enabled) const;
- virtual WebRtc_Word32 StereoRecordingIsAvailable(bool* available) const;
- virtual WebRtc_Word32 SetStereoRecording(bool enable);
- virtual WebRtc_Word32 StereoRecording(bool* enabled) const;
- virtual WebRtc_Word32 SetRecordingChannel(const ChannelType channel);
- virtual WebRtc_Word32 RecordingChannel(ChannelType* channel) const;
-
- // Delay information and control
- virtual WebRtc_Word32 SetPlayoutBuffer(const BufferType type,
- WebRtc_UWord16 sizeMS = 0);
- virtual WebRtc_Word32 PlayoutBuffer(BufferType* type,
- WebRtc_UWord16* sizeMS) const;
- virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16* delayMS) const;
- virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16* delayMS) const;
-
- // CPU load
- virtual WebRtc_Word32 CPULoad(WebRtc_UWord16* load) const;
-
- // Recording of raw PCM data
- virtual WebRtc_Word32 StartRawOutputFileRecording(
- const WebRtc_Word8 pcmFileNameUTF8[kAdmMaxFileNameSize]);
- virtual WebRtc_Word32 StopRawOutputFileRecording();
- virtual WebRtc_Word32 StartRawInputFileRecording(
- const WebRtc_Word8 pcmFileNameUTF8[kAdmMaxFileNameSize]);
- virtual WebRtc_Word32 StopRawInputFileRecording();
-
- // Native sample rate controls (samples/sec)
- virtual WebRtc_Word32 SetRecordingSampleRate(
- const WebRtc_UWord32 samplesPerSec);
- virtual WebRtc_Word32 RecordingSampleRate(
- WebRtc_UWord32* samplesPerSec) const;
- virtual WebRtc_Word32 SetPlayoutSampleRate(
- const WebRtc_UWord32 samplesPerSec);
- virtual WebRtc_Word32 PlayoutSampleRate(
- WebRtc_UWord32* samplesPerSec) const;
-
- // Mobile device specific functions
- virtual WebRtc_Word32 ResetAudioDevice();
- virtual WebRtc_Word32 SetLoudspeakerStatus(bool enable);
- virtual WebRtc_Word32 GetLoudspeakerStatus(bool* enabled) const;
-
- virtual int32_t EnableBuiltInAEC(bool enable);
- virtual bool BuiltInAECIsEnabled() const;
-
-public:
- WebRtc_Word32 Id() {return _id;}
-
-private:
- PlatformType Platform() const;
- AudioLayer PlatformAudioLayer() const;
-
-private:
- CriticalSectionWrapper& _critSect;
- CriticalSectionWrapper& _critSectEventCb;
- CriticalSectionWrapper& _critSectAudioCb;
-
- AudioDeviceObserver* _ptrCbAudioDeviceObserver;
-
- AudioDeviceUtility* _ptrAudioDeviceUtility;
- AudioDeviceGeneric* _ptrAudioDevice;
-
- AudioDeviceBuffer _audioDeviceBuffer;
-
- WebRtc_Word32 _id;
- AudioLayer _platformAudioLayer;
- WebRtc_UWord32 _lastProcessTime;
- PlatformType _platformType;
- bool _initialized;
- mutable ErrorCode _lastError;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_INTERFACE_AUDIO_DEVICE_IMPL_H_
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_utility.cc b/trunk/src/modules/audio_device/main/source/audio_device_utility.cc
deleted file mode 100644
index c256f9e..0000000
--- a/trunk/src/modules/audio_device/main/source/audio_device_utility.cc
+++ /dev/null
@@ -1,124 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cassert>
-
-#include "audio_device_utility.h"
-
-#if defined(_WIN32)
-
-// ============================================================================
-// Windows
-// ============================================================================
-
-#include <windows.h>
-#include <conio.h>
-#include <ctype.h>
-#include <stdio.h>
-#include <mmsystem.h>
-
-namespace webrtc
-{
-
-void AudioDeviceUtility::Sleep(WebRtc_UWord32 milliseconds)
-{
- return ::Sleep(milliseconds);
-}
-
-void AudioDeviceUtility::WaitForKey()
-{
- _getch();
-}
-
-WebRtc_UWord32 AudioDeviceUtility::GetTimeInMS()
-{
- return timeGetTime();
-}
-
-bool AudioDeviceUtility::StringCompare(const WebRtc_Word8* str1 , const WebRtc_Word8* str2, const WebRtc_UWord32 length)
-{
- return ((_strnicmp(str1, str2, length) == 0) ? true : false);
-}
-
-} // namespace webrtc
-
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
-
-// ============================================================================
-// Linux & Mac
-// ============================================================================
-
-#include <sys/time.h> // gettimeofday
-#include <time.h> // nanosleep, gettimeofday
-#include <string.h> // strncasecmp
-#include <stdio.h> // getchar
-#include <termios.h> // tcgetattr
-
-#include <unistd.h>
-
-namespace webrtc
-{
-
-void AudioDeviceUtility::WaitForKey()
-{
-
- struct termios oldt, newt;
-
- tcgetattr( STDIN_FILENO, &oldt );
-
- // we don't want getchar to echo!
-
- newt = oldt;
- newt.c_lflag &= ~( ICANON | ECHO );
- tcsetattr( STDIN_FILENO, TCSANOW, &newt );
-
- // catch any newline that's hanging around...
-
- // you'll have to hit enter twice if you
-
- // choose enter out of all available keys
-
- if (getchar() == '\n')
- {
- getchar();
- }
-
- tcsetattr( STDIN_FILENO, TCSANOW, &oldt );
-}
-
-WebRtc_UWord32 AudioDeviceUtility::GetTimeInMS()
-{
- struct timeval tv;
- struct timezone tz;
- WebRtc_UWord32 val;
-
- gettimeofday(&tv, &tz);
- val = (WebRtc_UWord32)(tv.tv_sec*1000 + tv.tv_usec/1000);
- return val;
-}
-
-void AudioDeviceUtility::Sleep(WebRtc_UWord32 milliseconds)
-{
- timespec t;
- t.tv_sec = milliseconds/1000;
- t.tv_nsec = (milliseconds-(milliseconds/1000)*1000)*1000000;
- nanosleep(&t,NULL);
-}
-
-bool AudioDeviceUtility::StringCompare(const WebRtc_Word8* str1 , const WebRtc_Word8* str2, const WebRtc_UWord32 length)
-{
- return (strncasecmp(str1, str2, length) == 0)?true: false;
-}
-
-} // namespace webrtc
-
-#endif // defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
-
-
diff --git a/trunk/src/modules/audio_device/main/source/audio_device_utility.h b/trunk/src/modules/audio_device/main/source/audio_device_utility.h
deleted file mode 100644
index b967950..0000000
--- a/trunk/src/modules/audio_device/main/source/audio_device_utility.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_H
-
-#include "typedefs.h"
-
-namespace webrtc
-{
-
-class AudioDeviceUtility
-{
-public:
- static WebRtc_UWord32 GetTimeInMS();
- static void Sleep(WebRtc_UWord32 milliseconds);
- static void WaitForKey();
- static bool StringCompare(const WebRtc_Word8* str1,
- const WebRtc_Word8* str2,
- const WebRtc_UWord32 length);
- virtual WebRtc_Word32 Init() = 0;
-
- virtual ~AudioDeviceUtility() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_H
-
diff --git a/trunk/src/modules/audio_device/main/source/dummy/audio_device_dummy.cc b/trunk/src/modules/audio_device/main/source/dummy/audio_device_dummy.cc
deleted file mode 100644
index 2c8d5c7..0000000
--- a/trunk/src/modules/audio_device/main/source/dummy/audio_device_dummy.cc
+++ /dev/null
@@ -1,1336 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_device_dummy.h"
-
-#include <string.h>
-
-#include "trace.h"
-#include "thread_wrapper.h"
-#include "event_wrapper.h"
-
-// Enable to record playout data
-//#define RECORD_PLAYOUT 1
-
-namespace webrtc {
-
-const WebRtc_UWord32 REC_TIMER_PERIOD_MS = 10;
-const WebRtc_UWord32 PLAY_TIMER_PERIOD_MS = 10;
-
-// ============================================================================
-// Construction & Destruction
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// AudioDeviceDummy() - ctor
-// ----------------------------------------------------------------------------
-
-AudioDeviceDummy::AudioDeviceDummy(const WebRtc_Word32 id) :
- _ptrAudioBuffer(NULL),
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _id(id),
- _timeEventRec(*EventWrapper::Create()),
- _timeEventPlay(*EventWrapper::Create()),
- _recStartEvent(*EventWrapper::Create()),
- _playStartEvent(*EventWrapper::Create()),
- _ptrThreadRec(NULL),
- _ptrThreadPlay(NULL),
- _recThreadID(0),
- _playThreadID(0),
- _initialized(false),
- _recording(false),
- _playing(false),
- _recIsInitialized(false),
- _playIsInitialized(false),
- _speakerIsInitialized(false),
- _microphoneIsInitialized(false),
- _playDataFile(NULL)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
-
- memset(_recBuffer, 0, sizeof(_recBuffer));
- WebRtc_Word16* tmp = (WebRtc_Word16*)_recBuffer;
-
- // Saw tooth -16000 to 16000, 100 Hz @ fs = 16 kHz
-// for(int i=0; i<160; ++i)
-// {
-// tmp[i] = i*200-16000;
-// }
-
- // Rough sinus 2 kHz @ fs = 16 kHz
- for(int i=0; i<20; ++i)
- {
- tmp[i*8] = 0;
- tmp[i*8+1] = -5000;
- tmp[i*8+2] = -16000;
- tmp[i*8+3] = -5000;
- tmp[i*8+4] = 0;
- tmp[i*8+5] = 5000;
- tmp[i*8+6] = 16000;
- tmp[i*8+7] = 5000;
- }
-
-#ifdef RECORD_PLAYOUT
- _playDataFile = fopen("webrtc_VoiceEngine_playout.pcm", "wb");
- if (!_playDataFile)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Could not open file for writing playout data");
- }
-#endif
-}
-
-// ----------------------------------------------------------------------------
-// AudioDeviceDummy() - dtor
-// ----------------------------------------------------------------------------
-
-AudioDeviceDummy::~AudioDeviceDummy()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
-
- Terminate();
-
- _ptrAudioBuffer = NULL;
-
- delete &_recStartEvent;
- delete &_playStartEvent;
- delete &_timeEventRec;
- delete &_timeEventPlay;
- delete &_critSect;
-
- if (_playDataFile)
- {
- fclose(_playDataFile);
- }
-}
-
-// ============================================================================
-// API
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// AttachAudioBuffer
-// ----------------------------------------------------------------------------
-
-void AudioDeviceDummy::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
-{
-
- _ptrAudioBuffer = audioBuffer;
-
- // Inform the AudioBuffer about default settings for this implementation.
- _ptrAudioBuffer->SetRecordingSampleRate(16000);
- _ptrAudioBuffer->SetPlayoutSampleRate(16000);
- _ptrAudioBuffer->SetRecordingChannels(1);
- _ptrAudioBuffer->SetPlayoutChannels(1);
-}
-
-// ----------------------------------------------------------------------------
-// ActiveAudioLayer
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const
-{
- audioLayer = AudioDeviceModule::kDummyAudio;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Init
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::Init()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_initialized)
- {
- return 0;
- }
-
- const bool periodic(true);
- unsigned int threadID(0);
- char threadName[64] = {0};
-
- // RECORDING
- strncpy(threadName, "webrtc_audio_module_rec_thread", 63);
- _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc, this, kRealtimePriority, threadName);
- if (_ptrThreadRec == NULL)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, " failed to create the rec audio thread");
- return -1;
- }
-
- if (!_ptrThreadRec->Start(threadID))
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, " failed to start the rec audio thread");
- delete _ptrThreadRec;
- _ptrThreadRec = NULL;
- return -1;
- }
- _recThreadID = threadID;
-
- if (!_timeEventRec.StartTimer(periodic, REC_TIMER_PERIOD_MS))
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, " failed to start the rec timer event");
- if (_ptrThreadRec->Stop())
- {
- delete _ptrThreadRec;
- _ptrThreadRec = NULL;
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " unable to stop the activated rec thread");
- }
- return -1;
- }
-
- // PLAYOUT
- strncpy(threadName, "webrtc_audio_module_play_thread", 63);
- _ptrThreadPlay = ThreadWrapper::CreateThread(PlayThreadFunc, this, kRealtimePriority, threadName);
- if (_ptrThreadPlay == NULL)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, " failed to create the play audio thread");
- return -1;
- }
-
- threadID = 0;
- if (!_ptrThreadPlay->Start(threadID))
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, " failed to start the play audio thread");
- delete _ptrThreadPlay;
- _ptrThreadPlay = NULL;
- return -1;
- }
- _playThreadID = threadID;
-
- if (!_timeEventPlay.StartTimer(periodic, PLAY_TIMER_PERIOD_MS))
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, " failed to start the play timer event");
- if (_ptrThreadPlay->Stop())
- {
- delete _ptrThreadPlay;
- _ptrThreadPlay = NULL;
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " unable to stop the activated play thread");
- }
- return -1;
- }
-
- _initialized = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Terminate
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::Terminate()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_initialized)
- {
- return 0;
- }
-
- // RECORDING
- if (_ptrThreadRec)
- {
- ThreadWrapper* tmpThread = _ptrThreadRec;
- _ptrThreadRec = NULL;
- _critSect.Leave();
-
- tmpThread->SetNotAlive();
- _timeEventRec.Set();
-
- if (tmpThread->Stop())
- {
- delete tmpThread;
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " failed to close down the rec audio thread");
- }
-
- _critSect.Enter();
- }
-
- _timeEventRec.StopTimer();
-
- // PLAYOUT
- if (_ptrThreadPlay)
- {
- ThreadWrapper* tmpThread = _ptrThreadPlay;
- _ptrThreadPlay = NULL;
- _critSect.Leave();
-
- tmpThread->SetNotAlive();
- _timeEventPlay.Set();
-
- if (tmpThread->Stop())
- {
- delete tmpThread;
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " failed to close down the play audio thread");
- }
-
- _critSect.Enter();
- }
-
- _timeEventPlay.StopTimer();
-
- _initialized = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Initialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceDummy::Initialized() const
-{
- return (_initialized);
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SpeakerIsAvailable(bool& available)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- available = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitSpeaker
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::InitSpeaker()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_playing)
- {
- return -1;
- }
-
- _speakerIsInitialized = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::MicrophoneIsAvailable(bool& available)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- available = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitMicrophone
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::InitMicrophone()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_recording)
- {
- return -1;
- }
-
- _microphoneIsInitialized = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceDummy::SpeakerIsInitialized() const
-{
-
- return (_speakerIsInitialized);
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceDummy::MicrophoneIsInitialized() const
-{
-
- return (_microphoneIsInitialized);
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolumeIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SpeakerVolumeIsAvailable(bool& available)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- available = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetSpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SetSpeakerVolume(WebRtc_UWord32 volume)
-{
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SpeakerVolume(WebRtc_UWord32& volume) const
-{
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// SetWaveOutVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SetWaveOutVolume(WebRtc_UWord16 volumeLeft, WebRtc_UWord16 volumeRight)
-{
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// WaveOutVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::WaveOutVolume(WebRtc_UWord16& volumeLeft, WebRtc_UWord16& volumeRight) const
-{
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MaxSpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const
-{
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MinSpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::MinSpeakerVolume(WebRtc_UWord32& minVolume) const
-{
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolumeStepSize
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const
-{
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerMuteIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SpeakerMuteIsAvailable(bool& available)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- available = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetSpeakerMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SetSpeakerMute(bool enable)
-{
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SpeakerMute(bool& enabled) const
-{
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneMuteIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::MicrophoneMuteIsAvailable(bool& available)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- available = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SetMicrophoneMute(bool enable)
-{
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::MicrophoneMute(bool& enabled) const
-{
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneBoostIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::MicrophoneBoostIsAvailable(bool& available)
-{
-
- available = false;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneBoost
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SetMicrophoneBoost(bool enable)
-{
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneBoost
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::MicrophoneBoost(bool& enabled) const
-{
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// StereoRecordingIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::StereoRecordingIsAvailable(bool& available)
-{
-
- available = false;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetStereoRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SetStereoRecording(bool enable)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (enable)
- {
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::StereoRecording(bool& enabled) const
-{
-
- enabled = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoPlayoutIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::StereoPlayoutIsAvailable(bool& available)
-{
-
- available = false;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetStereoPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SetStereoPlayout(bool enable)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (enable)
- {
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::StereoPlayout(bool& enabled) const
-{
-
- enabled = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetAGC
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SetAGC(bool enable)
-{
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// AGC
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceDummy::AGC() const
-{
- // WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
- return false;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolumeIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::MicrophoneVolumeIsAvailable(bool& available)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- available = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SetMicrophoneVolume(WebRtc_UWord32 volume)
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "AudioDeviceDummy::SetMicrophoneVolume(volume=%u)", volume);
-
- CriticalSectionScoped lock(_critSect);
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::MicrophoneVolume(WebRtc_UWord32& volume) const
-{
- // WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MaxMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MinMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::MinMicrophoneVolume(WebRtc_UWord32& minVolume) const
-{
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolumeStepSize
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const
-{
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDevices
-// ----------------------------------------------------------------------------
-
-WebRtc_Word16 AudioDeviceDummy::PlayoutDevices()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- return 1;
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutDevice I (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SetPlayoutDevice(WebRtc_UWord16 index)
-{
-
- if (_playIsInitialized)
- {
- return -1;
- }
-
- if (index != 0)
- {
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutDevice II (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device)
-{
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDeviceName
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::PlayoutDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
-{
-
- if (index != 0)
- {
- return -1;
- }
-
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDeviceName
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::RecordingDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
-{
-
- if (index != 0)
- {
- return -1;
- }
-
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDevices
-// ----------------------------------------------------------------------------
-
-WebRtc_Word16 AudioDeviceDummy::RecordingDevices()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- return 1;
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingDevice I (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SetRecordingDevice(WebRtc_UWord16 index)
-{
-
- if (_recIsInitialized)
- {
- return -1;
- }
-
- if (index != 0 )
- {
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingDevice II (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device)
-{
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::PlayoutIsAvailable(bool& available)
-{
-
- available = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::RecordingIsAvailable(bool& available)
-{
-
- available = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::InitPlayout()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_playing)
- {
- return -1;
- }
-
- if (_playIsInitialized)
- {
- return 0;
- }
-
- // Initialize the speaker (devices might have been added or removed)
- if (InitSpeaker() == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " InitSpeaker() failed");
- }
-
- _playIsInitialized = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::InitRecording()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_recording)
- {
- return -1;
- }
-
- if (_recIsInitialized)
- {
- return 0;
- }
-
- // Initialize the microphone (devices might have been added or removed)
- if (InitMicrophone() == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " InitMicrophone() failed");
- }
-
- _recIsInitialized = true;
-
- return 0;
-
-}
-
-// ----------------------------------------------------------------------------
-// StartRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::StartRecording()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_recIsInitialized)
- {
- return -1;
- }
-
- if (_recording)
- {
- return 0;
- }
-
- _recording = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StopRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::StopRecording()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_recIsInitialized)
- {
- return 0;
- }
-
- _recIsInitialized = false;
- _recording = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceDummy::RecordingIsInitialized() const
-{
- return (_recIsInitialized);
-}
-
-// ----------------------------------------------------------------------------
-// Recording
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceDummy::Recording() const
-{
- return (_recording);
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceDummy::PlayoutIsInitialized() const
-{
-
- return (_playIsInitialized);
-}
-
-// ----------------------------------------------------------------------------
-// StartPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::StartPlayout()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_playIsInitialized)
- {
- return -1;
- }
-
- if (_playing)
- {
- return 0;
- }
-
- _playing = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StopPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::StopPlayout()
-{
-
- if (!_playIsInitialized)
- {
- return 0;
- }
-
- _playIsInitialized = false;
- _playing = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDelay
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::PlayoutDelay(WebRtc_UWord16& delayMS) const
-{
- CriticalSectionScoped lock(_critSect);
- delayMS = 0;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDelay
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::RecordingDelay(WebRtc_UWord16& delayMS) const
-{
- CriticalSectionScoped lock(_critSect);
- delayMS = 0;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Playing
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceDummy::Playing() const
-{
- return (_playing);
-}
-// ----------------------------------------------------------------------------
-// SetPlayoutBuffer
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::SetPlayoutBuffer(const AudioDeviceModule::BufferType type, WebRtc_UWord16 sizeMS)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- // Just ignore
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutBuffer
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::PlayoutBuffer(AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const
-{
- CriticalSectionScoped lock(_critSect);
-
- type = AudioDeviceModule::kAdaptiveBufferSize;
- sizeMS = 0;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// CPULoad
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceDummy::CPULoad(WebRtc_UWord16& load) const
-{
-
- load = 0;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutWarning
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceDummy::PlayoutWarning() const
-{
- return false;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutError
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceDummy::PlayoutError() const
-{
- return false;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingWarning
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceDummy::RecordingWarning() const
-{
- return false;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingError
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceDummy::RecordingError() const
-{
- return false;
-}
-
-// ----------------------------------------------------------------------------
-// ClearPlayoutWarning
-// ----------------------------------------------------------------------------
-
-void AudioDeviceDummy::ClearPlayoutWarning()
-{
-}
-
-// ----------------------------------------------------------------------------
-// ClearPlayoutError
-// ----------------------------------------------------------------------------
-
-void AudioDeviceDummy::ClearPlayoutError()
-{
-}
-
-// ----------------------------------------------------------------------------
-// ClearRecordingWarning
-// ----------------------------------------------------------------------------
-
-void AudioDeviceDummy::ClearRecordingWarning()
-{
-}
-
-// ----------------------------------------------------------------------------
-// ClearRecordingError
-// ----------------------------------------------------------------------------
-
-void AudioDeviceDummy::ClearRecordingError()
-{
-}
-
-// ============================================================================
-// Thread Methods
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// PlayThreadFunc
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceDummy::PlayThreadFunc(void* pThis)
-{
- return (static_cast<AudioDeviceDummy*>(pThis)->PlayThreadProcess());
-}
-
-// ----------------------------------------------------------------------------
-// RecThreadFunc
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceDummy::RecThreadFunc(void* pThis)
-{
- return (static_cast<AudioDeviceDummy*>(pThis)->RecThreadProcess());
-}
-
-// ----------------------------------------------------------------------------
-// PlayThreadProcess
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceDummy::PlayThreadProcess()
-{
- switch (_timeEventPlay.Wait(1000))
- {
- case kEventSignaled:
- break;
- case kEventError:
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "EventWrapper::Wait() failed => restarting timer");
- _timeEventPlay.StopTimer();
- _timeEventPlay.StartTimer(true, PLAY_TIMER_PERIOD_MS);
- return true;
- case kEventTimeout:
- return true;
- }
-
- Lock();
-
- if(_playing)
- {
- WebRtc_Word8 playBuffer[2*160];
-
- UnLock();
- WebRtc_Word32 nSamples = (WebRtc_Word32)_ptrAudioBuffer->RequestPlayoutData(160);
- Lock();
-
- if (!_playing)
- {
- UnLock();
- return true;
- }
-
- nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
- if (nSamples != 160)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " invalid number of output samples(%d)", nSamples);
- }
-
- if (_playDataFile)
- {
- int wr = fwrite(playBuffer, 2, 160, _playDataFile);
- if (wr != 160)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Could not write playout data to file (%d) ferror = %d",
- wr, ferror(_playDataFile));
- }
- }
- }
-
- UnLock();
- return true;
-}
-
-// ----------------------------------------------------------------------------
-// RecThreadProcess
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceDummy::RecThreadProcess()
-{
- switch (_timeEventRec.Wait(1000))
- {
- case kEventSignaled:
- break;
- case kEventError:
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "EventWrapper::Wait() failed => restarting timer");
- _timeEventRec.StopTimer();
- _timeEventRec.StartTimer(true, REC_TIMER_PERIOD_MS);
- return true;
- case kEventTimeout:
- return true;
- }
-
- Lock();
-
- if (_recording)
- {
- // store the recorded buffer
- _ptrAudioBuffer->SetRecordedBuffer(_recBuffer, 160);
-
- // store vqe delay values
- _ptrAudioBuffer->SetVQEData(0, 0, 0);
-
- // deliver recorded samples at specified sample rate, mic level etc. to the observer using callback
- UnLock();
- _ptrAudioBuffer->DeliverRecordedData();
- }
- else
- {
- UnLock();
- }
-
- return true;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/dummy/audio_device_dummy.h b/trunk/src/modules/audio_device/main/source/dummy/audio_device_dummy.h
deleted file mode 100644
index 5a979d2..0000000
--- a/trunk/src/modules/audio_device/main/source/dummy/audio_device_dummy.h
+++ /dev/null
@@ -1,178 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DUMMY_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DUMMY_H
-
-#include <stdio.h>
-
-#include "audio_device_generic.h"
-#include "critical_section_wrapper.h"
-
-namespace webrtc {
-class EventWrapper;
-class ThreadWrapper;
-
-class AudioDeviceDummy : public AudioDeviceGeneric
-{
-public:
- AudioDeviceDummy(const WebRtc_Word32 id);
- ~AudioDeviceDummy();
-
- // Retrieve the currently utilized audio layer
- virtual WebRtc_Word32 ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
-
- // Main initializaton and termination
- virtual WebRtc_Word32 Init();
- virtual WebRtc_Word32 Terminate();
- virtual bool Initialized() const;
-
- // Device enumeration
- virtual WebRtc_Word16 PlayoutDevices();
- virtual WebRtc_Word16 RecordingDevices();
- virtual WebRtc_Word32 PlayoutDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize]);
- virtual WebRtc_Word32 RecordingDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize]);
-
- // Device selection
- virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32 SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
- virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32 SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device);
-
- // Audio transport initialization
- virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
- virtual WebRtc_Word32 InitPlayout();
- virtual bool PlayoutIsInitialized() const;
- virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
- virtual WebRtc_Word32 InitRecording();
- virtual bool RecordingIsInitialized() const;
-
- // Audio transport control
- virtual WebRtc_Word32 StartPlayout();
- virtual WebRtc_Word32 StopPlayout();
- virtual bool Playing() const;
- virtual WebRtc_Word32 StartRecording();
- virtual WebRtc_Word32 StopRecording();
- virtual bool Recording() const;
-
- // Microphone Automatic Gain Control (AGC)
- virtual WebRtc_Word32 SetAGC(bool enable);
- virtual bool AGC() const;
-
- // Volume control based on the Windows Wave API (Windows only)
- virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft, WebRtc_UWord16 volumeRight);
- virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft, WebRtc_UWord16& volumeRight) const;
-
- // Audio mixer initialization
- virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
- virtual WebRtc_Word32 InitSpeaker();
- virtual bool SpeakerIsInitialized() const;
- virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
- virtual WebRtc_Word32 InitMicrophone();
- virtual bool MicrophoneIsInitialized() const;
-
- // Speaker volume controls
- virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
- virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
- virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
- virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
- virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
-
- // Microphone volume controls
- virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
- virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
- virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
- virtual WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
-
- // Speaker mute control
- virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
- virtual WebRtc_Word32 SetSpeakerMute(bool enable);
- virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
-
- // Microphone mute control
- virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
- virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
-
- // Microphone boost control
- virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
- virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
-
- // Stereo support
- virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
- virtual WebRtc_Word32 SetStereoPlayout(bool enable);
- virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
- virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
- virtual WebRtc_Word32 SetStereoRecording(bool enable);
- virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
-
- // Delay information and control
- virtual WebRtc_Word32 SetPlayoutBuffer(const AudioDeviceModule::BufferType type, WebRtc_UWord16 sizeMS);
- virtual WebRtc_Word32 PlayoutBuffer(AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const;
- virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
- virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
-
- // CPU load
- virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
-
- virtual bool PlayoutWarning() const;
- virtual bool PlayoutError() const;
- virtual bool RecordingWarning() const;
- virtual bool RecordingError() const;
- virtual void ClearPlayoutWarning();
- virtual void ClearPlayoutError();
- virtual void ClearRecordingWarning();
- virtual void ClearRecordingError();
-
- virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
-private:
- void Lock() { _critSect.Enter(); };
- void UnLock() { _critSect.Leave(); };
-
- static bool RecThreadFunc(void*);
- static bool PlayThreadFunc(void*);
- bool RecThreadProcess();
- bool PlayThreadProcess();
-
- AudioDeviceBuffer* _ptrAudioBuffer;
- CriticalSectionWrapper& _critSect;
- WebRtc_Word32 _id;
-
- EventWrapper& _timeEventRec;
- EventWrapper& _timeEventPlay;
- EventWrapper& _recStartEvent;
- EventWrapper& _playStartEvent;
-
- ThreadWrapper* _ptrThreadRec;
- ThreadWrapper* _ptrThreadPlay;
- WebRtc_UWord32 _recThreadID;
- WebRtc_UWord32 _playThreadID;
-
- bool _initialized;
- bool _recording;
- bool _playing;
- bool _recIsInitialized;
- bool _playIsInitialized;
- bool _speakerIsInitialized;
- bool _microphoneIsInitialized;
-
- WebRtc_Word8 _recBuffer[2*160];
-
- FILE* _playDataFile;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_DUMMY_H
diff --git a/trunk/src/modules/audio_device/main/source/dummy/audio_device_utility_dummy.cc b/trunk/src/modules/audio_device/main/source/dummy/audio_device_utility_dummy.cc
deleted file mode 100644
index 44f2c25..0000000
--- a/trunk/src/modules/audio_device/main/source/dummy/audio_device_utility_dummy.cc
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_device_utility_dummy.h"
-#include "audio_device_config.h" // DEBUG_PRINT()
-#include "critical_section_wrapper.h"
-#include "trace.h"
-
-namespace webrtc
-{
-
-AudioDeviceUtilityDummy::AudioDeviceUtilityDummy(const WebRtc_Word32 id) :
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _id(id),
- _lastError(AudioDeviceModule::kAdmErrNone)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
- "%s created", __FUNCTION__);
-}
-
-AudioDeviceUtilityDummy::~AudioDeviceUtilityDummy()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
- "%s destroyed", __FUNCTION__);
- {
- CriticalSectionScoped lock(_critSect);
-
- // free stuff here...
- }
-
- delete &_critSect;
-}
-
-// ============================================================================
-// API
-// ============================================================================
-
-
-WebRtc_Word32 AudioDeviceUtilityDummy::Init()
-{
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
- " OS info: %s", "Dummy");
-
- return 0;
-}
-
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/dummy/audio_device_utility_dummy.h b/trunk/src/modules/audio_device/main/source/dummy/audio_device_utility_dummy.h
deleted file mode 100644
index 601c448..0000000
--- a/trunk/src/modules/audio_device/main/source/dummy/audio_device_utility_dummy.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_DUMMY_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_DUMMY_H
-
-#include "audio_device_utility.h"
-#include "audio_device.h"
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-
-class AudioDeviceUtilityDummy: public AudioDeviceUtility
-{
-public:
- AudioDeviceUtilityDummy(const WebRtc_Word32 id);
- ~AudioDeviceUtilityDummy();
-
- virtual WebRtc_Word32 Init();
-
-private:
- CriticalSectionWrapper& _critSect;
- WebRtc_Word32 _id;
- AudioDeviceModule::ErrorCode _lastError;
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_UTILITY_DUMMY_H_
diff --git a/trunk/src/modules/audio_device/main/source/linux/alsasymboltable_linux.cc b/trunk/src/modules/audio_device/main/source/linux/alsasymboltable_linux.cc
deleted file mode 100644
index 1b1707c..0000000
--- a/trunk/src/modules/audio_device/main/source/linux/alsasymboltable_linux.cc
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "alsasymboltable_linux.h"
-
-namespace webrtc_adm_linux_alsa {
-
-LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(AlsaSymbolTable, "libasound.so.2")
-#define X(sym) \
- LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(AlsaSymbolTable, sym)
-ALSA_SYMBOLS_LIST
-#undef X
-LATE_BINDING_SYMBOL_TABLE_DEFINE_END(AlsaSymbolTable)
-
-} // namespace webrtc_adm_linux_alsa
diff --git a/trunk/src/modules/audio_device/main/source/linux/alsasymboltable_linux.h b/trunk/src/modules/audio_device/main/source/linux/alsasymboltable_linux.h
deleted file mode 100644
index d25bbd7..0000000
--- a/trunk/src/modules/audio_device/main/source/linux/alsasymboltable_linux.h
+++ /dev/null
@@ -1,147 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_ALSASYMBOLTABLE_LINUX_H
-#define WEBRTC_AUDIO_DEVICE_ALSASYMBOLTABLE_LINUX_H
-
-#include "latebindingsymboltable_linux.h"
-
-namespace webrtc_adm_linux_alsa {
-
-// The ALSA symbols we need, as an X-Macro list.
-// This list must contain precisely every libasound function that is used in
-// alsasoundsystem.cc.
-#define ALSA_SYMBOLS_LIST \
- X(snd_device_name_free_hint) \
- X(snd_device_name_get_hint) \
- X(snd_device_name_hint) \
- X(snd_pcm_avail_update) \
- X(snd_pcm_close) \
- X(snd_pcm_delay) \
- X(snd_pcm_drop) \
- X(snd_pcm_open) \
- X(snd_pcm_prepare) \
- X(snd_pcm_readi) \
- X(snd_pcm_recover) \
- X(snd_pcm_resume) \
- X(snd_pcm_reset) \
- X(snd_pcm_state) \
- X(snd_pcm_set_params) \
- X(snd_pcm_get_params) \
- X(snd_pcm_start) \
- X(snd_pcm_stream) \
- X(snd_pcm_frames_to_bytes) \
- X(snd_pcm_bytes_to_frames) \
- X(snd_pcm_wait) \
- X(snd_pcm_writei) \
- X(snd_pcm_info_get_class) \
- X(snd_pcm_info_get_subdevices_avail) \
- X(snd_pcm_info_get_subdevice_name) \
- X(snd_pcm_info_set_subdevice) \
- X(snd_pcm_info_get_id) \
- X(snd_pcm_info_set_device) \
- X(snd_pcm_info_set_stream) \
- X(snd_pcm_info_get_name) \
- X(snd_pcm_info_get_subdevices_count) \
- X(snd_pcm_info_sizeof) \
- X(snd_pcm_hw_params) \
- X(snd_pcm_hw_params_malloc) \
- X(snd_pcm_hw_params_free) \
- X(snd_pcm_hw_params_any) \
- X(snd_pcm_hw_params_set_access) \
- X(snd_pcm_hw_params_set_format) \
- X(snd_pcm_hw_params_set_channels) \
- X(snd_pcm_hw_params_set_rate_near) \
- X(snd_pcm_hw_params_set_buffer_size_near) \
- X(snd_card_next) \
- X(snd_card_get_name) \
- X(snd_config_update) \
- X(snd_config_copy) \
- X(snd_config_get_id) \
- X(snd_ctl_open) \
- X(snd_ctl_close) \
- X(snd_ctl_card_info) \
- X(snd_ctl_card_info_sizeof) \
- X(snd_ctl_card_info_get_id) \
- X(snd_ctl_card_info_get_name) \
- X(snd_ctl_pcm_next_device) \
- X(snd_ctl_pcm_info) \
- X(snd_mixer_load) \
- X(snd_mixer_free) \
- X(snd_mixer_detach) \
- X(snd_mixer_close) \
- X(snd_mixer_open) \
- X(snd_mixer_attach) \
- X(snd_mixer_first_elem) \
- X(snd_mixer_elem_next) \
- X(snd_mixer_selem_get_name) \
- X(snd_mixer_selem_is_active) \
- X(snd_mixer_selem_register) \
- X(snd_mixer_selem_set_playback_volume_all) \
- X(snd_mixer_selem_get_playback_volume) \
- X(snd_mixer_selem_has_playback_volume) \
- X(snd_mixer_selem_get_playback_volume_range) \
- X(snd_mixer_selem_has_playback_switch) \
- X(snd_mixer_selem_get_playback_switch) \
- X(snd_mixer_selem_set_playback_switch_all) \
- X(snd_mixer_selem_has_capture_switch) \
- X(snd_mixer_selem_get_capture_switch) \
- X(snd_mixer_selem_set_capture_switch_all) \
- X(snd_mixer_selem_has_capture_volume) \
- X(snd_mixer_selem_set_capture_volume_all) \
- X(snd_mixer_selem_get_capture_volume) \
- X(snd_mixer_selem_get_capture_volume_range) \
- X(snd_dlopen) \
- X(snd_dlclose) \
- X(snd_config) \
- X(snd_config_search) \
- X(snd_config_get_string) \
- X(snd_config_search_definition) \
- X(snd_config_get_type) \
- X(snd_config_delete) \
- X(snd_config_iterator_entry) \
- X(snd_config_iterator_first) \
- X(snd_config_iterator_next) \
- X(snd_config_iterator_end) \
- X(snd_config_delete_compound_members) \
- X(snd_config_get_integer) \
- X(snd_config_get_bool) \
- X(snd_dlsym) \
- X(snd_strerror) \
- X(snd_lib_error) \
- X(snd_lib_error_set_handler)
-
-LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(AlsaSymbolTable)
-#define X(sym) \
- LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(AlsaSymbolTable, sym)
-ALSA_SYMBOLS_LIST
-#undef X
-LATE_BINDING_SYMBOL_TABLE_DECLARE_END(AlsaSymbolTable)
-
-} // namespace webrtc_adm_linux_alsa
-
-#endif // WEBRTC_AUDIO_DEVICE_ALSASYMBOLTABLE_LINUX_H
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_device_alsa_linux.cc b/trunk/src/modules/audio_device/main/source/linux/audio_device_alsa_linux.cc
deleted file mode 100644
index f82d502..0000000
--- a/trunk/src/modules/audio_device/main/source/linux/audio_device_alsa_linux.cc
+++ /dev/null
@@ -1,2320 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cassert>
-
-#include "audio_device_utility.h"
-#include "audio_device_alsa_linux.h"
-#include "audio_device_config.h"
-
-#include "event_wrapper.h"
-#include "trace.h"
-#include "thread_wrapper.h"
-
-
-webrtc_adm_linux_alsa::AlsaSymbolTable AlsaSymbolTable;
-
-// Accesses ALSA functions through our late-binding symbol table instead of
-// directly. This way we don't have to link to libasound, which means our binary
-// will work on systems that don't have it.
-#define LATE(sym) \
- LATESYM_GET(webrtc_adm_linux_alsa::AlsaSymbolTable, &AlsaSymbolTable, sym)
-
-// Redefine these here to be able to do late-binding
-#undef snd_ctl_card_info_alloca
-#define snd_ctl_card_info_alloca(ptr) \
- do { *ptr = (snd_ctl_card_info_t *) \
- __builtin_alloca (LATE(snd_ctl_card_info_sizeof)()); \
- memset(*ptr, 0, LATE(snd_ctl_card_info_sizeof)()); } while (0)
-
-#undef snd_pcm_info_alloca
-#define snd_pcm_info_alloca(pInfo) \
- do { *pInfo = (snd_pcm_info_t *) \
- __builtin_alloca (LATE(snd_pcm_info_sizeof)()); \
- memset(*pInfo, 0, LATE(snd_pcm_info_sizeof)()); } while (0)
-
-// snd_lib_error_handler_t
-void WebrtcAlsaErrorHandler(const char *file,
- int line,
- const char *function,
- int err,
- const char *fmt,...){};
-
-namespace webrtc
-{
-static const unsigned int ALSA_PLAYOUT_FREQ = 48000;
-static const unsigned int ALSA_PLAYOUT_CH = 2;
-static const unsigned int ALSA_PLAYOUT_LATENCY = 40*1000; // in us
-static const unsigned int ALSA_CAPTURE_FREQ = 48000;
-static const unsigned int ALSA_CAPTURE_CH = 2;
-static const unsigned int ALSA_CAPTURE_LATENCY = 40*1000; // in us
-static const unsigned int ALSA_PLAYOUT_WAIT_TIMEOUT = 5; // in ms
-static const unsigned int ALSA_CAPTURE_WAIT_TIMEOUT = 5; // in ms
-
-#define FUNC_GET_NUM_OF_DEVICE 0
-#define FUNC_GET_DEVICE_NAME 1
-#define FUNC_GET_DEVICE_NAME_FOR_AN_ENUM 2
-
-AudioDeviceLinuxALSA::AudioDeviceLinuxALSA(const WebRtc_Word32 id) :
- _ptrAudioBuffer(NULL),
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _ptrThreadRec(NULL),
- _ptrThreadPlay(NULL),
- _recThreadID(0),
- _playThreadID(0),
- _id(id),
- _mixerManager(id),
- _inputDeviceIndex(0),
- _outputDeviceIndex(0),
- _inputDeviceIsSpecified(false),
- _outputDeviceIsSpecified(false),
- _handleRecord(NULL),
- _handlePlayout(NULL),
- _recordingBuffersizeInFrame(0),
- _recordingPeriodSizeInFrame(0),
- _playoutBufferSizeInFrame(0),
- _playoutPeriodSizeInFrame(0),
- _recordingBufferSizeIn10MS(0),
- _playoutBufferSizeIn10MS(0),
- _recordingFramesIn10MS(0),
- _playoutFramesIn10MS(0),
- _recordingFreq(ALSA_CAPTURE_FREQ),
- _playoutFreq(ALSA_PLAYOUT_FREQ),
- _recChannels(ALSA_CAPTURE_CH),
- _playChannels(ALSA_PLAYOUT_CH),
- _recordingBuffer(NULL),
- _playoutBuffer(NULL),
- _recordingFramesLeft(0),
- _playoutFramesLeft(0),
- _playbackBufferSize(0),
- _playBufType(AudioDeviceModule::kFixedBufferSize),
- _initialized(false),
- _recording(false),
- _playing(false),
- _recIsInitialized(false),
- _playIsInitialized(false),
- _AGC(false),
- _recordingDelay(0),
- _playoutDelay(0),
- _writeErrors(0),
- _playWarning(0),
- _playError(0),
- _recWarning(0),
- _recError(0),
- _playBufDelay(80),
- _playBufDelayFixed(80)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
- "%s created", __FUNCTION__);
-}
-
-// ----------------------------------------------------------------------------
-// AudioDeviceLinuxALSA - dtor
-// ----------------------------------------------------------------------------
-
-AudioDeviceLinuxALSA::~AudioDeviceLinuxALSA()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
- "%s destroyed", __FUNCTION__);
-
- Terminate();
-
- // Clean up the recording buffer and playout buffer.
- if (_recordingBuffer)
- {
- delete [] _recordingBuffer;
- _recordingBuffer = NULL;
- }
- if (_playoutBuffer)
- {
- delete [] _playoutBuffer;
- _playoutBuffer = NULL;
- }
- delete &_critSect;
-}
-
-void AudioDeviceLinuxALSA::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- _ptrAudioBuffer = audioBuffer;
-
- // Inform the AudioBuffer about default settings for this implementation.
- // Set all values to zero here since the actual settings will be done by
- // InitPlayout and InitRecording later.
- _ptrAudioBuffer->SetRecordingSampleRate(0);
- _ptrAudioBuffer->SetPlayoutSampleRate(0);
- _ptrAudioBuffer->SetRecordingChannels(0);
- _ptrAudioBuffer->SetPlayoutChannels(0);
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::ActiveAudioLayer(
- AudioDeviceModule::AudioLayer& audioLayer) const
-{
- audioLayer = AudioDeviceModule::kLinuxAlsaAudio;
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::Init()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- // Load libasound
- if (!AlsaSymbolTable.Load())
- {
- // Alsa is not installed on
- // this system
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to load symbol table");
- return -1;
- }
-
- if (_initialized)
- {
- return 0;
- }
-
- _playWarning = 0;
- _playError = 0;
- _recWarning = 0;
- _recError = 0;
-
- _initialized = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::Terminate()
-{
-
- if (!_initialized)
- {
- return 0;
- }
-
- CriticalSectionScoped lock(_critSect);
-
- _mixerManager.Close();
-
- // RECORDING
- if (_ptrThreadRec)
- {
- ThreadWrapper* tmpThread = _ptrThreadRec;
- _ptrThreadRec = NULL;
- _critSect.Leave();
-
- tmpThread->SetNotAlive();
-
- if (tmpThread->Stop())
- {
- delete tmpThread;
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " failed to close down the rec audio thread");
- }
-
- _critSect.Enter();
- }
-
- // PLAYOUT
- if (_ptrThreadPlay)
- {
- ThreadWrapper* tmpThread = _ptrThreadPlay;
- _ptrThreadPlay = NULL;
- _critSect.Leave();
-
- tmpThread->SetNotAlive();
-
- if (tmpThread->Stop())
- {
- delete tmpThread;
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " failed to close down the play audio thread");
- }
-
- _critSect.Enter();
- }
-
- _initialized = false;
- _outputDeviceIsSpecified = false;
- _inputDeviceIsSpecified = false;
-
- return 0;
-}
-
-bool AudioDeviceLinuxALSA::Initialized() const
-{
- return (_initialized);
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerIsAvailable(bool& available)
-{
-
- bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
- // Make an attempt to open up the
- // output mixer corresponding to the currently selected output device.
- //
- if (!wasInitialized && InitSpeaker() == -1)
- {
- available = false;
- return 0;
- }
-
- // Given that InitSpeaker was successful, we know that a valid speaker
- // exists
- available = true;
-
- // Close the initialized output mixer
- //
- if (!wasInitialized)
- {
- _mixerManager.CloseSpeaker();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::InitSpeaker()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_playing)
- {
- return -1;
- }
-
- char devName[kAdmMaxDeviceNameSize] = {0};
- GetDevicesInfo(2, true, _outputDeviceIndex, devName, kAdmMaxDeviceNameSize);
- return _mixerManager.OpenSpeaker(devName);
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneIsAvailable(bool& available)
-{
-
- bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
- // Make an attempt to open up the
- // input mixer corresponding to the currently selected output device.
- //
- if (!wasInitialized && InitMicrophone() == -1)
- {
- available = false;
- return 0;
- }
-
- // Given that InitMicrophone was successful, we know that a valid
- // microphone exists
- available = true;
-
- // Close the initialized input mixer
- //
- if (!wasInitialized)
- {
- _mixerManager.CloseMicrophone();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::InitMicrophone()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_recording)
- {
- return -1;
- }
-
- char devName[kAdmMaxDeviceNameSize] = {0};
- GetDevicesInfo(2, false, _inputDeviceIndex, devName, kAdmMaxDeviceNameSize);
- return _mixerManager.OpenMicrophone(devName);
-}
-
-bool AudioDeviceLinuxALSA::SpeakerIsInitialized() const
-{
- return (_mixerManager.SpeakerIsInitialized());
-}
-
-bool AudioDeviceLinuxALSA::MicrophoneIsInitialized() const
-{
- return (_mixerManager.MicrophoneIsInitialized());
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerVolumeIsAvailable(bool& available)
-{
-
- bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
- // Make an attempt to open up the
- // output mixer corresponding to the currently selected output device.
- if (!wasInitialized && InitSpeaker() == -1)
- {
- // If we end up here it means that the selected speaker has no volume
- // control.
- available = false;
- return 0;
- }
-
- // Given that InitSpeaker was successful, we know that a volume control
- // exists
- available = true;
-
- // Close the initialized output mixer
- if (!wasInitialized)
- {
- _mixerManager.CloseSpeaker();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SetSpeakerVolume(WebRtc_UWord32 volume)
-{
-
- return (_mixerManager.SetSpeakerVolume(volume));
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerVolume(WebRtc_UWord32& volume) const
-{
-
- WebRtc_UWord32 level(0);
-
- if (_mixerManager.SpeakerVolume(level) == -1)
- {
- return -1;
- }
-
- volume = level;
-
- return 0;
-}
-
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
- WebRtc_UWord16 volumeRight)
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::WaveOutVolume(
- WebRtc_UWord16& /*volumeLeft*/,
- WebRtc_UWord16& /*volumeRight*/) const
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::MaxSpeakerVolume(
- WebRtc_UWord32& maxVolume) const
-{
-
- WebRtc_UWord32 maxVol(0);
-
- if (_mixerManager.MaxSpeakerVolume(maxVol) == -1)
- {
- return -1;
- }
-
- maxVolume = maxVol;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::MinSpeakerVolume(
- WebRtc_UWord32& minVolume) const
-{
-
- WebRtc_UWord32 minVol(0);
-
- if (_mixerManager.MinSpeakerVolume(minVol) == -1)
- {
- return -1;
- }
-
- minVolume = minVol;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerVolumeStepSize(
- WebRtc_UWord16& stepSize) const
-{
-
- WebRtc_UWord16 delta(0);
-
- if (_mixerManager.SpeakerVolumeStepSize(delta) == -1)
- {
- return -1;
- }
-
- stepSize = delta;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerMuteIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
- bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
- // Make an attempt to open up the
- // output mixer corresponding to the currently selected output device.
- //
- if (!wasInitialized && InitSpeaker() == -1)
- {
- // If we end up here it means that the selected speaker has no volume
- // control, hence it is safe to state that there is no mute control
- // already at this stage.
- available = false;
- return 0;
- }
-
- // Check if the selected speaker has a mute control
- _mixerManager.SpeakerMuteIsAvailable(isAvailable);
-
- available = isAvailable;
-
- // Close the initialized output mixer
- if (!wasInitialized)
- {
- _mixerManager.CloseSpeaker();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SetSpeakerMute(bool enable)
-{
- return (_mixerManager.SetSpeakerMute(enable));
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SpeakerMute(bool& enabled) const
-{
-
- bool muted(0);
-
- if (_mixerManager.SpeakerMute(muted) == -1)
- {
- return -1;
- }
-
- enabled = muted;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneMuteIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
- bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
- // Make an attempt to open up the
- // input mixer corresponding to the currently selected input device.
- //
- if (!wasInitialized && InitMicrophone() == -1)
- {
- // If we end up here it means that the selected microphone has no volume
- // control, hence it is safe to state that there is no mute control
- // already at this stage.
- available = false;
- return 0;
- }
-
- // Check if the selected microphone has a mute control
- //
- _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized input mixer
- //
- if (!wasInitialized)
- {
- _mixerManager.CloseMicrophone();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SetMicrophoneMute(bool enable)
-{
- return (_mixerManager.SetMicrophoneMute(enable));
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneMute(bool& enabled) const
-{
-
- bool muted(0);
-
- if (_mixerManager.MicrophoneMute(muted) == -1)
- {
- return -1;
- }
-
- enabled = muted;
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneBoostIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
- bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
- // Enumerate all avaliable microphone and make an attempt to open up the
- // input mixer corresponding to the currently selected input device.
- //
- if (!wasInitialized && InitMicrophone() == -1)
- {
- // If we end up here it means that the selected microphone has no volume
- // control, hence it is safe to state that there is no boost control
- // already at this stage.
- available = false;
- return 0;
- }
-
- // Check if the selected microphone has a boost control
- _mixerManager.MicrophoneBoostIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized input mixer
- if (!wasInitialized)
- {
- _mixerManager.CloseMicrophone();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SetMicrophoneBoost(bool enable)
-{
-
- return (_mixerManager.SetMicrophoneBoost(enable));
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneBoost(bool& enabled) const
-{
-
- bool onOff(0);
-
- if (_mixerManager.MicrophoneBoost(onOff) == -1)
- {
- return -1;
- }
-
- enabled = onOff;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::StereoRecordingIsAvailable(bool& available)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- // If we already have initialized in stereo it's obviously available
- if (_recIsInitialized && (2 == _recChannels))
- {
- available = true;
- return 0;
- }
-
- // Save rec states and the number of rec channels
- bool recIsInitialized = _recIsInitialized;
- bool recording = _recording;
- int recChannels = _recChannels;
-
- available = false;
-
- // Stop/uninitialize recording if initialized (and possibly started)
- if (_recIsInitialized)
- {
- StopRecording();
- }
-
- // Try init in stereo;
- _recChannels = 2;
- if (InitRecording() == 0)
- {
- available = true;
- }
-
- // Stop/uninitialize recording
- StopRecording();
-
- // Recover previous states
- _recChannels = recChannels;
- if (recIsInitialized)
- {
- InitRecording();
- }
- if (recording)
- {
- StartRecording();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SetStereoRecording(bool enable)
-{
-
- if (enable)
- _recChannels = 2;
- else
- _recChannels = 1;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::StereoRecording(bool& enabled) const
-{
-
- if (_recChannels == 2)
- enabled = true;
- else
- enabled = false;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::StereoPlayoutIsAvailable(bool& available)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- // If we already have initialized in stereo it's obviously available
- if (_playIsInitialized && (2 == _playChannels))
- {
- available = true;
- return 0;
- }
-
- // Save rec states and the number of rec channels
- bool playIsInitialized = _playIsInitialized;
- bool playing = _playing;
- int playChannels = _playChannels;
-
- available = false;
-
- // Stop/uninitialize recording if initialized (and possibly started)
- if (_playIsInitialized)
- {
- StopPlayout();
- }
-
- // Try init in stereo;
- _playChannels = 2;
- if (InitPlayout() == 0)
- {
- available = true;
- }
-
- // Stop/uninitialize recording
- StopPlayout();
-
- // Recover previous states
- _playChannels = playChannels;
- if (playIsInitialized)
- {
- InitPlayout();
- }
- if (playing)
- {
- StartPlayout();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SetStereoPlayout(bool enable)
-{
-
- if (enable)
- _playChannels = 2;
- else
- _playChannels = 1;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::StereoPlayout(bool& enabled) const
-{
-
- if (_playChannels == 2)
- enabled = true;
- else
- enabled = false;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SetAGC(bool enable)
-{
-
- _AGC = enable;
-
- return 0;
-}
-
-bool AudioDeviceLinuxALSA::AGC() const
-{
-
- return _AGC;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneVolumeIsAvailable(bool& available)
-{
-
- bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
- // Make an attempt to open up the
- // input mixer corresponding to the currently selected output device.
- if (!wasInitialized && InitMicrophone() == -1)
- {
- // If we end up here it means that the selected microphone has no volume
- // control.
- available = false;
- return 0;
- }
-
- // Given that InitMicrophone was successful, we know that a volume control
- // exists
- available = true;
-
- // Close the initialized input mixer
- if (!wasInitialized)
- {
- _mixerManager.CloseMicrophone();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SetMicrophoneVolume(WebRtc_UWord32 volume)
-{
-
- return (_mixerManager.SetMicrophoneVolume(volume));
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneVolume(WebRtc_UWord32& volume) const
-{
-
- WebRtc_UWord32 level(0);
-
- if (_mixerManager.MicrophoneVolume(level) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " failed to retrive current microphone level");
- return -1;
- }
-
- volume = level;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::MaxMicrophoneVolume(
- WebRtc_UWord32& maxVolume) const
-{
-
- WebRtc_UWord32 maxVol(0);
-
- if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1)
- {
- return -1;
- }
-
- maxVolume = maxVol;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::MinMicrophoneVolume(
- WebRtc_UWord32& minVolume) const
-{
-
- WebRtc_UWord32 minVol(0);
-
- if (_mixerManager.MinMicrophoneVolume(minVol) == -1)
- {
- return -1;
- }
-
- minVolume = minVol;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::MicrophoneVolumeStepSize(
- WebRtc_UWord16& stepSize) const
-{
-
- WebRtc_UWord16 delta(0);
-
- if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1)
- {
- return -1;
- }
-
- stepSize = delta;
-
- return 0;
-}
-
-WebRtc_Word16 AudioDeviceLinuxALSA::PlayoutDevices()
-{
-
- return (WebRtc_Word16)GetDevicesInfo(0, true);
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SetPlayoutDevice(WebRtc_UWord16 index)
-{
-
- if (_playIsInitialized)
- {
- return -1;
- }
-
- WebRtc_UWord32 nDevices = GetDevicesInfo(0, true);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " number of availiable audio output devices is %u", nDevices);
-
- if (index > (nDevices-1))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " device index is out of range [0,%u]", (nDevices-1));
- return -1;
- }
-
- _outputDeviceIndex = index;
- _outputDeviceIsSpecified = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SetPlayoutDevice(
- AudioDeviceModule::WindowsDeviceType /*device*/)
-{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "WindowsDeviceType not supported");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::PlayoutDeviceName(
- WebRtc_UWord16 index,
- WebRtc_Word8 name[kAdmMaxDeviceNameSize],
- WebRtc_Word8 guid[kAdmMaxGuidSize])
-{
-
- const WebRtc_UWord16 nDevices(PlayoutDevices());
-
- if ((index > (nDevices-1)) || (name == NULL))
- {
- return -1;
- }
-
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- return GetDevicesInfo(1, true, index, name, kAdmMaxDeviceNameSize);
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::RecordingDeviceName(
- WebRtc_UWord16 index,
- WebRtc_Word8 name[kAdmMaxDeviceNameSize],
- WebRtc_Word8 guid[kAdmMaxGuidSize])
-{
-
- const WebRtc_UWord16 nDevices(RecordingDevices());
-
- if ((index > (nDevices-1)) || (name == NULL))
- {
- return -1;
- }
-
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- return GetDevicesInfo(1, false, index, name, kAdmMaxDeviceNameSize);
-}
-
-WebRtc_Word16 AudioDeviceLinuxALSA::RecordingDevices()
-{
-
- return (WebRtc_Word16)GetDevicesInfo(0, false);
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SetRecordingDevice(WebRtc_UWord16 index)
-{
-
- if (_recIsInitialized)
- {
- return -1;
- }
-
- WebRtc_UWord32 nDevices = GetDevicesInfo(0, false);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " number of availiable audio input devices is %u", nDevices);
-
- if (index > (nDevices-1))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " device index is out of range [0,%u]", (nDevices-1));
- return -1;
- }
-
- _inputDeviceIndex = index;
- _inputDeviceIsSpecified = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingDevice II (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SetRecordingDevice(
- AudioDeviceModule::WindowsDeviceType /*device*/)
-{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "WindowsDeviceType not supported");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::PlayoutIsAvailable(bool& available)
-{
-
- available = false;
-
- // Try to initialize the playout side with mono
- // Assumes that user set num channels after calling this function
- _playChannels = 1;
- WebRtc_Word32 res = InitPlayout();
-
- // Cancel effect of initialization
- StopPlayout();
-
- if (res != -1)
- {
- available = true;
- }
- else
- {
- // It may be possible to play out in stereo
- res = StereoPlayoutIsAvailable(available);
- if (available)
- {
- // Then set channels to 2 so InitPlayout doesn't fail
- _playChannels = 2;
- }
- }
-
- return res;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::RecordingIsAvailable(bool& available)
-{
-
- available = false;
-
- // Try to initialize the recording side with mono
- // Assumes that user set num channels after calling this function
- _recChannels = 1;
- WebRtc_Word32 res = InitRecording();
-
- // Cancel effect of initialization
- StopRecording();
-
- if (res != -1)
- {
- available = true;
- }
- else
- {
- // It may be possible to record in stereo
- res = StereoRecordingIsAvailable(available);
- if (available)
- {
- // Then set channels to 2 so InitPlayout doesn't fail
- _recChannels = 2;
- }
- }
-
- return res;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::InitPlayout()
-{
-
- int errVal = 0;
-
- CriticalSectionScoped lock(_critSect);
- if (_playing)
- {
- return -1;
- }
-
- if (!_outputDeviceIsSpecified)
- {
- return -1;
- }
-
- if (_playIsInitialized)
- {
- return 0;
- }
- // Initialize the speaker (devices might have been added or removed)
- if (InitSpeaker() == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " InitSpeaker() failed");
- }
-
- // Start by closing any existing wave-output devices
- //
- if (_handlePlayout != NULL)
- {
- LATE(snd_pcm_close)(_handlePlayout);
- _handlePlayout = NULL;
- _playIsInitialized = false;
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error closing current playout sound device, error:"
- " %s", LATE(snd_strerror)(errVal));
- }
- }
-
- // Open PCM device for playout
- char deviceName[kAdmMaxDeviceNameSize] = {0};
- GetDevicesInfo(2, true, _outputDeviceIndex, deviceName,
- kAdmMaxDeviceNameSize);
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " InitPlayout open (%s)", deviceName);
-
- errVal = LATE(snd_pcm_open)
- (&_handlePlayout,
- deviceName,
- SND_PCM_STREAM_PLAYBACK,
- SND_PCM_NONBLOCK);
-
- if (errVal == -EBUSY) // Device busy - try some more!
- {
- for (int i=0; i < 5; i++)
- {
- sleep(1);
- errVal = LATE(snd_pcm_open)
- (&_handlePlayout,
- deviceName,
- SND_PCM_STREAM_PLAYBACK,
- SND_PCM_NONBLOCK);
- if (errVal == 0)
- {
- break;
- }
- }
- }
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " unable to open playback device: %s (%d)",
- LATE(snd_strerror)(errVal),
- errVal);
- _handlePlayout = NULL;
- return -1;
- }
-
- _playoutFramesIn10MS = _playoutFreq/100;
- if ((errVal = LATE(snd_pcm_set_params)( _handlePlayout,
-#if defined(WEBRTC_BIG_ENDIAN)
- SND_PCM_FORMAT_S16_BE,
-#else
- SND_PCM_FORMAT_S16_LE, //format
-#endif
- SND_PCM_ACCESS_RW_INTERLEAVED, //access
- _playChannels, //channels
- _playoutFreq, //rate
- 1, //soft_resample
- ALSA_PLAYOUT_LATENCY //40*1000 //latency required overall latency in us
- )) < 0)
- { /* 0.5sec */
- _playoutFramesIn10MS = 0;
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " unable to set playback device: %s (%d)",
- LATE(snd_strerror)(errVal),
- errVal);
- ErrorRecovery(errVal, _handlePlayout);
- errVal = LATE(snd_pcm_close)(_handlePlayout);
- _handlePlayout = NULL;
- return -1;
- }
-
- errVal = LATE(snd_pcm_get_params)(_handlePlayout,
- &_playoutBufferSizeInFrame, &_playoutPeriodSizeInFrame);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " snd_pcm_get_params %s",
- LATE(snd_strerror)(errVal),
- errVal);
- _playoutBufferSizeInFrame = 0;
- _playoutPeriodSizeInFrame = 0;
- }
- else {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " playout snd_pcm_get_params "
- "buffer_size:%d period_size :%d",
- _playoutBufferSizeInFrame, _playoutPeriodSizeInFrame);
- }
-
- if (_ptrAudioBuffer)
- {
- // Update webrtc audio buffer with the selected parameters
- _ptrAudioBuffer->SetPlayoutSampleRate(_playoutFreq);
- _ptrAudioBuffer->SetPlayoutChannels(_playChannels);
- }
-
- // Set play buffer size
- _playoutBufferSizeIn10MS = LATE(snd_pcm_frames_to_bytes)(
- _handlePlayout, _playoutFramesIn10MS);
-
- // Init varaibles used for play
- _playWarning = 0;
- _playError = 0;
-
- if (_handlePlayout != NULL)
- {
- _playIsInitialized = true;
- return 0;
- }
- else
- {
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::InitRecording()
-{
-
- int errVal = 0;
-
- CriticalSectionScoped lock(_critSect);
-
- if (_recording)
- {
- return -1;
- }
-
- if (!_inputDeviceIsSpecified)
- {
- return -1;
- }
-
- if (_recIsInitialized)
- {
- return 0;
- }
-
- // Initialize the microphone (devices might have been added or removed)
- if (InitMicrophone() == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " InitMicrophone() failed");
- }
-
- // Start by closing any existing pcm-input devices
- //
- if (_handleRecord != NULL)
- {
- int errVal = LATE(snd_pcm_close)(_handleRecord);
- _handleRecord = NULL;
- _recIsInitialized = false;
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error closing current recording sound device,"
- " error: %s",
- LATE(snd_strerror)(errVal));
- }
- }
-
- // Open PCM device for recording
- // The corresponding settings for playout are made after the record settings
- char deviceName[kAdmMaxDeviceNameSize] = {0};
- GetDevicesInfo(2, false, _inputDeviceIndex, deviceName,
- kAdmMaxDeviceNameSize);
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "InitRecording open (%s)", deviceName);
- errVal = LATE(snd_pcm_open)
- (&_handleRecord,
- deviceName,
- SND_PCM_STREAM_CAPTURE,
- SND_PCM_NONBLOCK);
-
- // Available modes: 0 = blocking, SND_PCM_NONBLOCK, SND_PCM_ASYNC
- if (errVal == -EBUSY) // Device busy - try some more!
- {
- for (int i=0; i < 5; i++)
- {
- sleep(1);
- errVal = LATE(snd_pcm_open)
- (&_handleRecord,
- deviceName,
- SND_PCM_STREAM_CAPTURE,
- SND_PCM_NONBLOCK);
- if (errVal == 0)
- {
- break;
- }
- }
- }
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " unable to open record device: %s",
- LATE(snd_strerror)(errVal));
- _handleRecord = NULL;
- return -1;
- }
-
- _recordingFramesIn10MS = _recordingFreq/100;
- if ((errVal = LATE(snd_pcm_set_params)(_handleRecord,
-#if defined(WEBRTC_BIG_ENDIAN)
- SND_PCM_FORMAT_S16_BE, //format
-#else
- SND_PCM_FORMAT_S16_LE, //format
-#endif
- SND_PCM_ACCESS_RW_INTERLEAVED, //access
- _recChannels, //channels
- _recordingFreq, //rate
- 1, //soft_resample
- ALSA_CAPTURE_LATENCY //latency in us
- )) < 0)
- {
- // Fall back to another mode then.
- if (_recChannels == 1)
- _recChannels = 2;
- else
- _recChannels = 1;
-
- if ((errVal = LATE(snd_pcm_set_params)(_handleRecord,
-#if defined(WEBRTC_BIG_ENDIAN)
- SND_PCM_FORMAT_S16_BE, //format
-#else
- SND_PCM_FORMAT_S16_LE, //format
-#endif
- SND_PCM_ACCESS_RW_INTERLEAVED, //access
- _recChannels, //channels
- _recordingFreq, //rate
- 1, //soft_resample
- ALSA_CAPTURE_LATENCY //latency in us
- )) < 0)
- {
- _recordingFramesIn10MS = 0;
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " unable to set record settings: %s (%d)",
- LATE(snd_strerror)(errVal), errVal);
- ErrorRecovery(errVal, _handleRecord);
- errVal = LATE(snd_pcm_close)(_handleRecord);
- _handleRecord = NULL;
- return -1;
- }
- }
-
- errVal = LATE(snd_pcm_get_params)(_handleRecord,
- &_recordingBuffersizeInFrame, &_recordingPeriodSizeInFrame);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " snd_pcm_get_params %s",
- LATE(snd_strerror)(errVal), errVal);
- _recordingBuffersizeInFrame = 0;
- _recordingPeriodSizeInFrame = 0;
- }
- else {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " capture snd_pcm_get_params "
- "buffer_size:%d period_size:%d",
- _recordingBuffersizeInFrame, _recordingPeriodSizeInFrame);
- }
-
- if (_ptrAudioBuffer)
- {
- // Update webrtc audio buffer with the selected parameters
- _ptrAudioBuffer->SetRecordingSampleRate(_recordingFreq);
- _ptrAudioBuffer->SetRecordingChannels(_recChannels);
- }
-
- // Set rec buffer size and create buffer
- _recordingBufferSizeIn10MS = LATE(snd_pcm_frames_to_bytes)(
- _handleRecord, _recordingFramesIn10MS);
-
- if (_handleRecord != NULL)
- {
- // Mark recording side as initialized
- _recIsInitialized = true;
- return 0;
- }
- else
- {
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::StartRecording()
-{
-
- if (!_recIsInitialized)
- {
- return -1;
- }
-
- if (_recording)
- {
- return 0;
- }
-
- _recording = true;
-
- int errVal = 0;
- _recordingFramesLeft = _recordingFramesIn10MS;
-
- // Make sure we only create the buffer once.
- if (!_recordingBuffer)
- _recordingBuffer = new WebRtc_Word8[_recordingBufferSizeIn10MS];
- if (!_recordingBuffer)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " failed to alloc recording buffer");
- _recording = false;
- return -1;
- }
- // RECORDING
- const char* threadName = "webrtc_audio_module_capture_thread";
- _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc,
- this,
- kRealtimePriority,
- threadName);
- if (_ptrThreadRec == NULL)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " failed to create the rec audio thread");
- _recording = false;
- delete [] _recordingBuffer;
- _recordingBuffer = NULL;
- return -1;
- }
-
- unsigned int threadID(0);
- if (!_ptrThreadRec->Start(threadID))
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " failed to start the rec audio thread");
- _recording = false;
- delete _ptrThreadRec;
- _ptrThreadRec = NULL;
- delete [] _recordingBuffer;
- _recordingBuffer = NULL;
- return -1;
- }
- _recThreadID = threadID;
-
- errVal = LATE(snd_pcm_prepare)(_handleRecord);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " capture snd_pcm_prepare failed (%s)\n",
- LATE(snd_strerror)(errVal));
- // just log error
- // if snd_pcm_open fails will return -1
- }
-
- errVal = LATE(snd_pcm_start)(_handleRecord);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " capture snd_pcm_start err: %s",
- LATE(snd_strerror)(errVal));
- errVal = LATE(snd_pcm_start)(_handleRecord);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " capture snd_pcm_start 2nd try err: %s",
- LATE(snd_strerror)(errVal));
- StopRecording();
- return -1;
- }
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::StopRecording()
-{
-
- {
- CriticalSectionScoped lock(_critSect);
-
- if (!_recIsInitialized)
- {
- return 0;
- }
-
- if (_handleRecord == NULL)
- {
- return -1;
- }
-
- // Make sure we don't start recording (it's asynchronous).
- _recIsInitialized = false;
- _recording = false;
- }
-
- if (_ptrThreadRec && !_ptrThreadRec->Stop())
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to stop the rec audio thread");
- return -1;
- }
- else {
- delete _ptrThreadRec;
- _ptrThreadRec = NULL;
- }
-
- CriticalSectionScoped lock(_critSect);
- _recordingFramesLeft = 0;
- if (_recordingBuffer)
- {
- delete [] _recordingBuffer;
- _recordingBuffer = NULL;
- }
-
- // Stop and close pcm recording device.
- int errVal = LATE(snd_pcm_drop)(_handleRecord);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error stop recording: %s",
- LATE(snd_strerror)(errVal));
- return -1;
- }
-
- errVal = LATE(snd_pcm_close)(_handleRecord);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error closing record sound device, error: %s",
- LATE(snd_strerror)(errVal));
- return -1;
- }
-
- // Check if we have muted and unmute if so.
- bool muteEnabled = false;
- MicrophoneMute(muteEnabled);
- if (muteEnabled)
- {
- SetMicrophoneMute(false);
- }
-
- // set the pcm input handle to NULL
- _handleRecord = NULL;
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " handle_capture is now set to NULL");
-
- return 0;
-}
-
-bool AudioDeviceLinuxALSA::RecordingIsInitialized() const
-{
- return (_recIsInitialized);
-}
-
-bool AudioDeviceLinuxALSA::Recording() const
-{
- return (_recording);
-}
-
-bool AudioDeviceLinuxALSA::PlayoutIsInitialized() const
-{
- return (_playIsInitialized);
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::StartPlayout()
-{
- if (!_playIsInitialized)
- {
- return -1;
- }
-
- if (_playing)
- {
- return 0;
- }
-
- _playing = true;
-
- _playoutFramesLeft = 0;
- if (!_playoutBuffer)
- _playoutBuffer = new WebRtc_Word8[_playoutBufferSizeIn10MS];
- if (!_playoutBuffer)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to alloc playout buf");
- _playing = false;
- return -1;
- }
-
- // PLAYOUT
- const char* threadName = "webrtc_audio_module_play_thread";
- _ptrThreadPlay = ThreadWrapper::CreateThread(PlayThreadFunc,
- this,
- kRealtimePriority,
- threadName);
- if (_ptrThreadPlay == NULL)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " failed to create the play audio thread");
- _playing = false;
- delete [] _playoutBuffer;
- _playoutBuffer = NULL;
- return -1;
- }
-
- unsigned int threadID(0);
- if (!_ptrThreadPlay->Start(threadID))
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " failed to start the play audio thread");
- _playing = false;
- delete _ptrThreadPlay;
- _ptrThreadPlay = NULL;
- delete [] _playoutBuffer;
- _playoutBuffer = NULL;
- return -1;
- }
- _playThreadID = threadID;
-
- int errVal = LATE(snd_pcm_prepare)(_handlePlayout);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " playout snd_pcm_prepare failed (%s)\n",
- LATE(snd_strerror)(errVal));
- // just log error
- // if snd_pcm_open fails will return -1
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::StopPlayout()
-{
-
- {
- CriticalSectionScoped lock(_critSect);
-
- if (!_playIsInitialized)
- {
- return 0;
- }
-
- if (_handlePlayout == NULL)
- {
- return -1;
- }
-
- _playing = false;
- }
-
- // stop playout thread first
- if (_ptrThreadPlay && !_ptrThreadPlay->Stop())
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to stop the play audio thread");
- return -1;
- }
- else {
- delete _ptrThreadPlay;
- _ptrThreadPlay = NULL;
- }
-
- CriticalSectionScoped lock(_critSect);
-
- _playoutFramesLeft = 0;
- delete [] _playoutBuffer;
- _playoutBuffer = NULL;
-
- // stop and close pcm playout device
- int errVal = LATE(snd_pcm_drop)(_handlePlayout);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error stop playing: %s",
- LATE(snd_strerror)(errVal));
- }
-
- errVal = LATE(snd_pcm_close)(_handlePlayout);
- if (errVal < 0)
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error closing playout sound device, error: %s",
- LATE(snd_strerror)(errVal));
-
- // set the pcm input handle to NULL
- _playIsInitialized = false;
- _handlePlayout = NULL;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " handle_playout is now set to NULL");
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::PlayoutDelay(WebRtc_UWord16& delayMS) const
-{
- delayMS = (WebRtc_UWord16)_playoutDelay * 1000 / _playoutFreq;
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::RecordingDelay(WebRtc_UWord16& delayMS) const
-{
- // Adding 10ms adjusted value to the record delay due to 10ms buffering.
- delayMS = (WebRtc_UWord16)(10 + _recordingDelay * 1000 / _recordingFreq);
- return 0;
-}
-
-bool AudioDeviceLinuxALSA::Playing() const
-{
- return (_playing);
-}
-// ----------------------------------------------------------------------------
-// SetPlayoutBuffer
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceLinuxALSA::SetPlayoutBuffer(
- const AudioDeviceModule::BufferType type,
- WebRtc_UWord16 sizeMS)
-{
- _playBufType = type;
- if (type == AudioDeviceModule::kFixedBufferSize)
- {
- _playBufDelayFixed = sizeMS;
- }
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::PlayoutBuffer(
- AudioDeviceModule::BufferType& type,
- WebRtc_UWord16& sizeMS) const
-{
- type = _playBufType;
- if (type == AudioDeviceModule::kFixedBufferSize)
- {
- sizeMS = _playBufDelayFixed;
- }
- else
- {
- sizeMS = _playBufDelay;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::CPULoad(WebRtc_UWord16& load) const
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-bool AudioDeviceLinuxALSA::PlayoutWarning() const
-{
- return (_playWarning > 0);
-}
-
-bool AudioDeviceLinuxALSA::PlayoutError() const
-{
- return (_playError > 0);
-}
-
-bool AudioDeviceLinuxALSA::RecordingWarning() const
-{
- return (_recWarning > 0);
-}
-
-bool AudioDeviceLinuxALSA::RecordingError() const
-{
- return (_recError > 0);
-}
-
-void AudioDeviceLinuxALSA::ClearPlayoutWarning()
-{
- _playWarning = 0;
-}
-
-void AudioDeviceLinuxALSA::ClearPlayoutError()
-{
- _playError = 0;
-}
-
-void AudioDeviceLinuxALSA::ClearRecordingWarning()
-{
- _recWarning = 0;
-}
-
-void AudioDeviceLinuxALSA::ClearRecordingError()
-{
- _recError = 0;
-}
-
-// ============================================================================
-// Private Methods
-// ============================================================================
-
-WebRtc_Word32 AudioDeviceLinuxALSA::GetDevicesInfo(
- const WebRtc_Word32 function,
- const bool playback,
- const WebRtc_Word32 enumDeviceNo,
- char* enumDeviceName,
- const WebRtc_Word32 ednLen) const
-{
-
- // Device enumeration based on libjingle implementation
- // by Tristan Schmelcher at Google Inc.
-
- const char *type = playback ? "Output" : "Input";
- // dmix and dsnoop are only for playback and capture, respectively, but ALSA
- // stupidly includes them in both lists.
- const char *ignorePrefix = playback ? "dsnoop:" : "dmix:" ;
- // (ALSA lists many more "devices" of questionable interest, but we show them
- // just in case the weird devices may actually be desirable for some
- // users/systems.)
-
- int err;
- int enumCount(0);
- bool keepSearching(true);
-
- void **hints;
- err = LATE(snd_device_name_hint)(-1, // All cards
- "pcm", // Only PCM devices
- &hints);
- if (err != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "GetDevicesInfo - device name hint error: %s",
- LATE(snd_strerror)(err));
- return -1;
- }
-
- enumCount++; // default is 0
- if (function == FUNC_GET_DEVICE_NAME && enumDeviceNo == 0)
- {
- strcpy(enumDeviceName, "default");
- return 0;
- }
- if (function == FUNC_GET_DEVICE_NAME_FOR_AN_ENUM && enumDeviceNo == 0)
- {
- strcpy(enumDeviceName, "default");
- return 0;
- }
-
- for (void **list = hints; *list != NULL; ++list)
- {
- char *actualType = LATE(snd_device_name_get_hint)(*list, "IOID");
- if (actualType)
- { // NULL means it's both.
- bool wrongType = (strcmp(actualType, type) != 0);
- free(actualType);
- if (wrongType)
- {
- // Wrong type of device (i.e., input vs. output).
- continue;
- }
- }
-
- char *name = LATE(snd_device_name_get_hint)(*list, "NAME");
- if (!name)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Device has no name");
- // Skip it.
- continue;
- }
-
- // Now check if we actually want to show this device.
- if (strcmp(name, "default") != 0 &&
- strcmp(name, "null") != 0 &&
- strcmp(name, "pulse") != 0 &&
- strncmp(name, ignorePrefix, strlen(ignorePrefix)) != 0)
- {
- // Yes, we do.
- char *desc = LATE(snd_device_name_get_hint)(*list, "DESC");
- if (!desc)
- {
- // Virtual devices don't necessarily have descriptions.
- // Use their names instead
- desc = name;
- }
-
- if (FUNC_GET_NUM_OF_DEVICE == function)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Enum device %d - %s", enumCount, name);
-
- }
- if ((FUNC_GET_DEVICE_NAME == function) &&
- (enumDeviceNo == enumCount))
- {
-
- // We have found the enum device, copy the name to buffer
- strncpy(enumDeviceName, desc, ednLen);
- enumDeviceName[ednLen-1] = '\0';
- keepSearching = false;
- // replace '\n' with '-'
- char * pret = strchr(enumDeviceName, '\n'/*0xa*/); //LF
- if (pret)
- *pret = '-';
- }
- if ((FUNC_GET_DEVICE_NAME_FOR_AN_ENUM == function) &&
- (enumDeviceNo == enumCount))
- {
- // We have found the enum device, copy the name to buffer
- strncpy(enumDeviceName, name, ednLen);
- enumDeviceName[ednLen-1] = '\0';
- keepSearching = false;
- }
- if (keepSearching)
- {
- ++enumCount;
- }
-
- if (desc != name)
- {
- free(desc);
- }
- }
-
- free(name);
-
- if (!keepSearching)
- {
- break;
- }
- }
-
- err = LATE(snd_device_name_free_hint)(hints);
- if (err != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "GetDevicesInfo - device name free hint error: %s",
- LATE(snd_strerror)(err));
- // Continue and return true anyways, since we did get the whole list.
- }
-
- if (FUNC_GET_NUM_OF_DEVICE == function)
- {
- if (enumCount == 1) // only default?
- enumCount = 0;
- return enumCount; // Normal return point for function 0
- }
-
- if (keepSearching)
- {
- // If we get here for function 1 and 2, we didn't find the specified
- // enum device
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "GetDevicesInfo - Could not find device name or numbers");
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::InputSanityCheckAfterUnlockedPeriod() const
-{
- if (_handleRecord == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " input state has been modified during unlocked period");
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::OutputSanityCheckAfterUnlockedPeriod() const
-{
- if (_handlePlayout == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " output state has been modified during unlocked period");
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxALSA::ErrorRecovery(WebRtc_Word32 error,
- snd_pcm_t* deviceHandle)
-{
- int st = LATE(snd_pcm_state)(deviceHandle);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Trying to recover from error: %s (%d) (state %d)",
- (LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_CAPTURE) ?
- "capture" : "playout", LATE(snd_strerror)(error), error, st);
-
- // It is recommended to use snd_pcm_recover for all errors. If that function
- // cannot handle the error, the input error code will be returned, otherwise
- // 0 is returned. From snd_pcm_recover API doc: "This functions handles
- // -EINTR (4) (interrupted system call), -EPIPE (32) (playout overrun or
- // capture underrun) and -ESTRPIPE (86) (stream is suspended) error codes
- // trying to prepare given stream for next I/O."
-
- /** Open */
- // SND_PCM_STATE_OPEN = 0,
- /** Setup installed */
- // SND_PCM_STATE_SETUP,
- /** Ready to start */
- // SND_PCM_STATE_PREPARED,
- /** Running */
- // SND_PCM_STATE_RUNNING,
- /** Stopped: underrun (playback) or overrun (capture) detected */
- // SND_PCM_STATE_XRUN,= 4
- /** Draining: running (playback) or stopped (capture) */
- // SND_PCM_STATE_DRAINING,
- /** Paused */
- // SND_PCM_STATE_PAUSED,
- /** Hardware is suspended */
- // SND_PCM_STATE_SUSPENDED,
- // ** Hardware is disconnected */
- // SND_PCM_STATE_DISCONNECTED,
- // SND_PCM_STATE_LAST = SND_PCM_STATE_DISCONNECTED
-
- // snd_pcm_recover isn't available in older alsa, e.g. on the FC4 machine
- // in Sthlm lab.
-
- int res = LATE(snd_pcm_recover)(deviceHandle, error, 1);
- if (0 == res)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Recovery - snd_pcm_recover OK");
-
- if ((error == -EPIPE || error == -ESTRPIPE) && // Buf underrun/overrun.
- _recording &&
- LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_CAPTURE)
- {
- // For capture streams we also have to repeat the explicit start()
- // to get data flowing again.
- int err = LATE(snd_pcm_start)(deviceHandle);
- if (err != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Recovery - snd_pcm_start error: %u", err);
- return -1;
- }
- }
-
- if ((error == -EPIPE || error == -ESTRPIPE) && // Buf underrun/overrun.
- _playing &&
- LATE(snd_pcm_stream)(deviceHandle) == SND_PCM_STREAM_PLAYBACK)
- {
- // For capture streams we also have to repeat the explicit start() to get
- // data flowing again.
- int err = LATE(snd_pcm_start)(deviceHandle);
- if (err != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Recovery - snd_pcm_start error: %s",
- LATE(snd_strerror)(err));
- return -1;
- }
- }
-
- return -EPIPE == error ? 1 : 0;
- }
- else {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Terriable, it shouldn't happen");
- }
-
- return res;
-}
-
-// ============================================================================
-// Thread Methods
-// ============================================================================
-
-bool AudioDeviceLinuxALSA::PlayThreadFunc(void* pThis)
-{
- return (static_cast<AudioDeviceLinuxALSA*>(pThis)->PlayThreadProcess());
-}
-
-bool AudioDeviceLinuxALSA::RecThreadFunc(void* pThis)
-{
- return (static_cast<AudioDeviceLinuxALSA*>(pThis)->RecThreadProcess());
-}
-
-bool AudioDeviceLinuxALSA::PlayThreadProcess()
-{
- if(!_playing)
- return false;
-
- int err;
- snd_pcm_sframes_t frames;
- snd_pcm_sframes_t avail_frames;
-
- Lock();
- //return a positive number of frames ready otherwise a negative error code
- avail_frames = LATE(snd_pcm_avail_update)(_handlePlayout);
- if (avail_frames < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "playout snd_pcm_avail_update error: %s",
- LATE(snd_strerror)(avail_frames));
- ErrorRecovery(avail_frames, _handlePlayout);
- UnLock();
- return true;
- }
- else if (avail_frames == 0)
- {
- UnLock();
-
- //maximum tixe in milliseconds to wait, a negative value means infinity
- err = LATE(snd_pcm_wait)(_handlePlayout, 2);
- if (err == 0)
- { //timeout occured
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id,
- "playout snd_pcm_wait timeout");
- }
-
- return true;
- }
-
- if (_playoutFramesLeft <= 0)
- {
- UnLock();
- _ptrAudioBuffer->RequestPlayoutData(_playoutFramesIn10MS);
- Lock();
-
- _playoutFramesLeft = _ptrAudioBuffer->GetPlayoutData(_playoutBuffer);
- assert(_playoutFramesLeft == _playoutFramesIn10MS);
- }
-
- if (static_cast<WebRtc_UWord32>(avail_frames) > _playoutFramesLeft)
- avail_frames = _playoutFramesLeft;
-
- int size = LATE(snd_pcm_frames_to_bytes)(_handlePlayout,
- _playoutFramesLeft);
- frames = LATE(snd_pcm_writei)(
- _handlePlayout,
- &_playoutBuffer[_playoutBufferSizeIn10MS - size],
- avail_frames);
-
- if (frames < 0)
- {
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id,
- "playout snd_pcm_avail_update error: %s",
- LATE(snd_strerror)(frames));
- _playoutFramesLeft = 0;
- ErrorRecovery(frames, _handlePlayout);
- UnLock();
- return true;
- }
- else {
- assert(frames==avail_frames);
- _playoutFramesLeft -= frames;
- }
-
- UnLock();
- return true;
-}
-
-bool AudioDeviceLinuxALSA::RecThreadProcess()
-{
- if (!_recording)
- return false;
-
- int err;
- snd_pcm_sframes_t frames;
- snd_pcm_sframes_t avail_frames;
- WebRtc_Word8 buffer[_recordingBufferSizeIn10MS];
-
- Lock();
-
- //return a positive number of frames ready otherwise a negative error code
- avail_frames = LATE(snd_pcm_avail_update)(_handleRecord);
- if (avail_frames < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "capture snd_pcm_avail_update error: %s",
- LATE(snd_strerror)(avail_frames));
- ErrorRecovery(avail_frames, _handleRecord);
- UnLock();
- return true;
- }
- else if (avail_frames == 0)
- { // no frame is available now
- UnLock();
-
- //maximum time in milliseconds to wait, a negative value means infinity
- err = LATE(snd_pcm_wait)(_handleRecord,
- ALSA_CAPTURE_WAIT_TIMEOUT);
- if (err == 0) //timeout occured
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id,
- "caputre snd_pcm_wait timeout");
-
- return true;
- }
-
- if (static_cast<WebRtc_UWord32>(avail_frames) > _recordingFramesLeft)
- avail_frames = _recordingFramesLeft;
-
- frames = LATE(snd_pcm_readi)(_handleRecord,
- buffer, avail_frames); // frames to be written
- if (frames < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "caputre snd_pcm_readi error: %s",
- LATE(snd_strerror)(frames));
- ErrorRecovery(frames, _handleRecord);
- UnLock();
- return true;
- }
- else if (frames > 0)
- {
- assert(frames == avail_frames);
-
- int left_size = LATE(snd_pcm_frames_to_bytes)(_handleRecord,
- _recordingFramesLeft);
- int size = LATE(snd_pcm_frames_to_bytes)(_handleRecord, frames);
-
- memcpy(&_recordingBuffer[_recordingBufferSizeIn10MS - left_size],
- buffer, size);
- _recordingFramesLeft -= frames;
-
- if (!_recordingFramesLeft)
- { // buf is full
- _recordingFramesLeft = _recordingFramesIn10MS;
-
- // store the recorded buffer (no action will be taken if the
- // #recorded samples is not a full buffer)
- _ptrAudioBuffer->SetRecordedBuffer(_recordingBuffer,
- _recordingFramesIn10MS);
-
- WebRtc_UWord32 currentMicLevel = 0;
- WebRtc_UWord32 newMicLevel = 0;
-
- if (AGC())
- {
- // store current mic level in the audio buffer if AGC is enabled
- if (MicrophoneVolume(currentMicLevel) == 0)
- {
- if (currentMicLevel == 0xffffffff)
- currentMicLevel = 100;
- // this call does not affect the actual microphone volume
- _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
- }
- }
-
- // calculate delay
- _playoutDelay = 0;
- _recordingDelay = 0;
- if (_handlePlayout)
- {
- err = LATE(snd_pcm_delay)(_handlePlayout,
- &_playoutDelay); // returned delay in frames
- if (err < 0)
- {
- // TODO(xians): Shall we call ErrorRecovery() here?
- _playoutDelay = 0;
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "playout snd_pcm_delay: %s",
- LATE(snd_strerror)(err));
- }
- }
-
- err = LATE(snd_pcm_delay)(_handleRecord,
- &_recordingDelay); // returned delay in frames
- if (err < 0)
- {
- // TODO(xians): Shall we call ErrorRecovery() here?
- _recordingDelay = 0;
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "caputre snd_pcm_delay: %s",
- LATE(snd_strerror)(err));
- }
-
- // TODO(xians): Shall we add 10ms buffer delay to the record delay?
- _ptrAudioBuffer->SetVQEData(
- _playoutDelay * 1000 / _playoutFreq,
- _recordingDelay * 1000 / _recordingFreq, 0);
-
- // Deliver recorded samples at specified sample rate, mic level etc.
- // to the observer using callback.
- UnLock();
- _ptrAudioBuffer->DeliverRecordedData();
- Lock();
-
- if (AGC())
- {
- newMicLevel = _ptrAudioBuffer->NewMicLevel();
- if (newMicLevel != 0)
- {
- // The VQE will only deliver non-zero microphone levels when a
- // change is needed. Set this new mic level (received from the
- // observer as return value in the callback).
- if (SetMicrophoneVolume(newMicLevel) == -1)
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " the required modification of the "
- "microphone volume failed");
- }
- }
- }
- }
-
- UnLock();
- return true;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_device_alsa_linux.h b/trunk/src/modules/audio_device/main/source/linux/audio_device_alsa_linux.h
deleted file mode 100644
index 9a5a032..0000000
--- a/trunk/src/modules/audio_device/main/source/linux/audio_device_alsa_linux.h
+++ /dev/null
@@ -1,253 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H
-
-#include "audio_device_generic.h"
-#include "critical_section_wrapper.h"
-#include "audio_mixer_manager_alsa_linux.h"
-
-#include <sys/soundcard.h>
-#include <sys/ioctl.h>
-
-#include <alsa/asoundlib.h>
-
-namespace webrtc
-{
-class EventWrapper;
-class ThreadWrapper;
-
-class AudioDeviceLinuxALSA : public AudioDeviceGeneric
-{
-public:
- AudioDeviceLinuxALSA(const WebRtc_Word32 id);
- ~AudioDeviceLinuxALSA();
-
- // Retrieve the currently utilized audio layer
- virtual WebRtc_Word32 ActiveAudioLayer(
- AudioDeviceModule::AudioLayer& audioLayer) const;
-
- // Main initializaton and termination
- virtual WebRtc_Word32 Init();
- virtual WebRtc_Word32 Terminate();
- virtual bool Initialized() const;
-
- // Device enumeration
- virtual WebRtc_Word16 PlayoutDevices();
- virtual WebRtc_Word16 RecordingDevices();
- virtual WebRtc_Word32 PlayoutDeviceName(
- WebRtc_UWord16 index,
- WebRtc_Word8 name[kAdmMaxDeviceNameSize],
- WebRtc_Word8 guid[kAdmMaxGuidSize]);
- virtual WebRtc_Word32 RecordingDeviceName(
- WebRtc_UWord16 index,
- WebRtc_Word8 name[kAdmMaxDeviceNameSize],
- WebRtc_Word8 guid[kAdmMaxGuidSize]);
-
- // Device selection
- virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32 SetPlayoutDevice(
- AudioDeviceModule::WindowsDeviceType device);
- virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32 SetRecordingDevice(
- AudioDeviceModule::WindowsDeviceType device);
-
- // Audio transport initialization
- virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
- virtual WebRtc_Word32 InitPlayout();
- virtual bool PlayoutIsInitialized() const;
- virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
- virtual WebRtc_Word32 InitRecording();
- virtual bool RecordingIsInitialized() const;
-
- // Audio transport control
- virtual WebRtc_Word32 StartPlayout();
- virtual WebRtc_Word32 StopPlayout();
- virtual bool Playing() const;
- virtual WebRtc_Word32 StartRecording();
- virtual WebRtc_Word32 StopRecording();
- virtual bool Recording() const;
-
- // Microphone Automatic Gain Control (AGC)
- virtual WebRtc_Word32 SetAGC(bool enable);
- virtual bool AGC() const;
-
- // Volume control based on the Windows Wave API (Windows only)
- virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
- WebRtc_UWord16 volumeRight);
- virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft,
- WebRtc_UWord16& volumeRight) const;
-
- // Audio mixer initialization
- virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
- virtual WebRtc_Word32 InitSpeaker();
- virtual bool SpeakerIsInitialized() const;
- virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
- virtual WebRtc_Word32 InitMicrophone();
- virtual bool MicrophoneIsInitialized() const;
-
- // Speaker volume controls
- virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
- virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
- virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
- virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
- virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
-
- // Microphone volume controls
- virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
- virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
- virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
- virtual WebRtc_Word32 MicrophoneVolumeStepSize(
- WebRtc_UWord16& stepSize) const;
-
- // Speaker mute control
- virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
- virtual WebRtc_Word32 SetSpeakerMute(bool enable);
- virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
-
- // Microphone mute control
- virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
- virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
-
- // Microphone boost control
- virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
- virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
-
- // Stereo support
- virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
- virtual WebRtc_Word32 SetStereoPlayout(bool enable);
- virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
- virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
- virtual WebRtc_Word32 SetStereoRecording(bool enable);
- virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
-
- // Delay information and control
- virtual WebRtc_Word32 SetPlayoutBuffer(
- const AudioDeviceModule::BufferType type,
- WebRtc_UWord16 sizeMS);
- virtual WebRtc_Word32 PlayoutBuffer(
- AudioDeviceModule::BufferType& type,
- WebRtc_UWord16& sizeMS) const;
- virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
- virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
-
- // CPU load
- virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
-
-public:
- virtual bool PlayoutWarning() const;
- virtual bool PlayoutError() const;
- virtual bool RecordingWarning() const;
- virtual bool RecordingError() const;
- virtual void ClearPlayoutWarning();
- virtual void ClearPlayoutError();
- virtual void ClearRecordingWarning();
- virtual void ClearRecordingError();
-
-public:
- virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
-private:
- WebRtc_Word32 GetDevicesInfo(const WebRtc_Word32 function,
- const bool playback,
- const WebRtc_Word32 enumDeviceNo = 0,
- char* enumDeviceName = NULL,
- const WebRtc_Word32 ednLen = 0) const;
- WebRtc_Word32 ErrorRecovery(WebRtc_Word32 error, snd_pcm_t* deviceHandle);
-
-private:
- void Lock() { _critSect.Enter(); };
- void UnLock() { _critSect.Leave(); };
-private:
- inline WebRtc_Word32 InputSanityCheckAfterUnlockedPeriod() const;
- inline WebRtc_Word32 OutputSanityCheckAfterUnlockedPeriod() const;
-
-private:
- static bool RecThreadFunc(void*);
- static bool PlayThreadFunc(void*);
- bool RecThreadProcess();
- bool PlayThreadProcess();
-
-private:
- AudioDeviceBuffer* _ptrAudioBuffer;
-
- CriticalSectionWrapper& _critSect;
-
- ThreadWrapper* _ptrThreadRec;
- ThreadWrapper* _ptrThreadPlay;
- WebRtc_UWord32 _recThreadID;
- WebRtc_UWord32 _playThreadID;
-
- WebRtc_Word32 _id;
-
- AudioMixerManagerLinuxALSA _mixerManager;
-
- WebRtc_UWord16 _inputDeviceIndex;
- WebRtc_UWord16 _outputDeviceIndex;
- bool _inputDeviceIsSpecified;
- bool _outputDeviceIsSpecified;
-
- snd_pcm_t* _handleRecord;
- snd_pcm_t* _handlePlayout;
-
- snd_pcm_uframes_t _recordingBuffersizeInFrame;
- snd_pcm_uframes_t _recordingPeriodSizeInFrame;
- snd_pcm_uframes_t _playoutBufferSizeInFrame;
- snd_pcm_uframes_t _playoutPeriodSizeInFrame;
-
- ssize_t _recordingBufferSizeIn10MS;
- ssize_t _playoutBufferSizeIn10MS;
- WebRtc_UWord32 _recordingFramesIn10MS;
- WebRtc_UWord32 _playoutFramesIn10MS;
-
- WebRtc_UWord32 _recordingFreq;
- WebRtc_UWord32 _playoutFreq;
- WebRtc_UWord8 _recChannels;
- WebRtc_UWord8 _playChannels;
-
- WebRtc_Word8* _recordingBuffer; // in byte
- WebRtc_Word8* _playoutBuffer; // in byte
- WebRtc_UWord32 _recordingFramesLeft;
- WebRtc_UWord32 _playoutFramesLeft;
-
- WebRtc_UWord32 _playbackBufferSize;
-
- AudioDeviceModule::BufferType _playBufType;
-
-private:
- bool _initialized;
- bool _recording;
- bool _playing;
- bool _recIsInitialized;
- bool _playIsInitialized;
- bool _AGC;
-
- snd_pcm_sframes_t _recordingDelay;
- snd_pcm_sframes_t _playoutDelay;
-
- WebRtc_Word32 _writeErrors;
- WebRtc_UWord16 _playWarning;
- WebRtc_UWord16 _playError;
- WebRtc_UWord16 _recWarning;
- WebRtc_UWord16 _recError;
-
- WebRtc_UWord16 _playBufDelay; // playback delay
- WebRtc_UWord16 _playBufDelayFixed; // fixed playback delay
-};
-
-}
-
-#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_ALSA_LINUX_H_
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_device_pulse_linux.cc b/trunk/src/modules/audio_device/main/source/linux/audio_device_pulse_linux.cc
deleted file mode 100644
index 16e1d33..0000000
--- a/trunk/src/modules/audio_device/main/source/linux/audio_device_pulse_linux.cc
+++ /dev/null
@@ -1,3150 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cassert>
-
-#include "audio_device_utility.h"
-#include "audio_device_pulse_linux.h"
-#include "audio_device_config.h"
-
-#include "event_wrapper.h"
-#include "trace.h"
-#include "thread_wrapper.h"
-
-webrtc_adm_linux_pulse::PulseAudioSymbolTable PaSymbolTable;
-
-// Accesses Pulse functions through our late-binding symbol table instead of
-// directly. This way we don't have to link to libpulse, which means our binary
-// will work on systems that don't have it.
-#define LATE(sym) \
- LATESYM_GET(webrtc_adm_linux_pulse::PulseAudioSymbolTable, &PaSymbolTable, sym)
-
-namespace webrtc
-{
-
-// ============================================================================
-// Static Methods
-// ============================================================================
-
-bool AudioDeviceLinuxPulse::PulseAudioIsSupported()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, -1, "%s",
- __FUNCTION__);
-
- bool pulseAudioIsSupported(true);
-
- // Check that we can initialize
- AudioDeviceLinuxPulse* admPulse = new AudioDeviceLinuxPulse(-1);
- if (admPulse->InitPulseAudio() == -1)
- {
- pulseAudioIsSupported = false;
- }
- admPulse->TerminatePulseAudio();
- delete admPulse;
-
- if (pulseAudioIsSupported)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1,
- "*** Linux Pulse Audio is supported ***");
- } else
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1,
- "*** Linux Pulse Audio is NOT supported => will revert to the ALSA API ***");
- }
-
- return (pulseAudioIsSupported);
-}
-
-AudioDeviceLinuxPulse::AudioDeviceLinuxPulse(const WebRtc_Word32 id) :
- _ptrAudioBuffer(NULL),
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _timeEventRec(*EventWrapper::Create()),
- _timeEventPlay(*EventWrapper::Create()),
- _recStartEvent(*EventWrapper::Create()),
- _playStartEvent(*EventWrapper::Create()),
- _ptrThreadPlay(NULL),
- _ptrThreadRec(NULL),
- _recThreadID(0),
- _playThreadID(0),
- _id(id),
- _mixerManager(id),
- _inputDeviceIndex(0),
- _outputDeviceIndex(0),
- _inputDeviceIsSpecified(false),
- _outputDeviceIsSpecified(false),
- _samplingFreq(0),
- _recChannels(1),
- _playChannels(1),
- _playBufType(AudioDeviceModule::kFixedBufferSize),
- _initialized(false),
- _recording(false),
- _playing(false),
- _recIsInitialized(false),
- _playIsInitialized(false),
- _startRec(false),
- _stopRec(false),
- _startPlay(false),
- _stopPlay(false),
- _AGC(false),
- _playBufDelayFixed(20),
- _sndCardPlayDelay(0),
- _sndCardRecDelay(0),
- _writeErrors(0),
- _playWarning(0),
- _playError(0),
- _recWarning(0),
- _recError(0),
- _deviceIndex(-1),
- _numPlayDevices(0),
- _numRecDevices(0),
- _playDeviceName(NULL),
- _recDeviceName(NULL),
- _playDisplayDeviceName(NULL),
- _recDisplayDeviceName(NULL),
- _playBuffer(NULL),
- _playbackBufferSize(0),
- _playbackBufferUnused(0),
- _tempBufferSpace(0),
- _recBuffer(NULL),
- _recordBufferSize(0),
- _recordBufferUsed(0),
- _tempSampleData(NULL),
- _tempSampleDataSize(0),
- _configuredLatencyPlay(0),
- _configuredLatencyRec(0),
- _paDeviceIndex(-1),
- _paStateChanged(false),
- _paMainloop(NULL),
- _paMainloopApi(NULL),
- _paContext(NULL),
- _recStream(NULL),
- _playStream(NULL),
- _recStreamFlags(0),
- _playStreamFlags(0)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
- "%s created", __FUNCTION__);
-
- memset(_paServerVersion, 0, sizeof(_paServerVersion));
- memset(&_playBufferAttr, 0, sizeof(_playBufferAttr));
- memset(&_recBufferAttr, 0, sizeof(_recBufferAttr));
-}
-
-AudioDeviceLinuxPulse::~AudioDeviceLinuxPulse()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
- "%s destroyed", __FUNCTION__);
-
- Terminate();
-
- if (_recBuffer)
- {
- delete [] _recBuffer;
- _recBuffer = NULL;
- }
- if (_playBuffer)
- {
- delete [] _playBuffer;
- _playBuffer = NULL;
- }
- if (_playDeviceName)
- {
- delete [] _playDeviceName;
- _playDeviceName = NULL;
- }
- if (_recDeviceName)
- {
- delete [] _recDeviceName;
- _recDeviceName = NULL;
- }
-
- delete &_recStartEvent;
- delete &_playStartEvent;
- delete &_timeEventRec;
- delete &_timeEventPlay;
- delete &_critSect;
-}
-
-void AudioDeviceLinuxPulse::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- _ptrAudioBuffer = audioBuffer;
-
- // Inform the AudioBuffer about default settings for this implementation.
- // Set all values to zero here since the actual settings will be done by
- // InitPlayout and InitRecording later.
- _ptrAudioBuffer->SetRecordingSampleRate(0);
- _ptrAudioBuffer->SetPlayoutSampleRate(0);
- _ptrAudioBuffer->SetRecordingChannels(0);
- _ptrAudioBuffer->SetPlayoutChannels(0);
-}
-
-// ----------------------------------------------------------------------------
-// ActiveAudioLayer
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceLinuxPulse::ActiveAudioLayer(
- AudioDeviceModule::AudioLayer& audioLayer) const
-{
- audioLayer = AudioDeviceModule::kLinuxPulseAudio;
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::Init()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_initialized)
- {
- return 0;
- }
-
- // Initialize PulseAudio
- if (InitPulseAudio() < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to initialize PulseAudio");
-
- if (TerminatePulseAudio() < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to terminate PulseAudio");
- }
-
- return -1;
- }
-
- _playWarning = 0;
- _playError = 0;
- _recWarning = 0;
- _recError = 0;
-
- // RECORDING
- const char* threadName = "webrtc_audio_module_rec_thread";
- _ptrThreadRec = ThreadWrapper::CreateThread(RecThreadFunc, this,
- kRealtimePriority, threadName);
- if (_ptrThreadRec == NULL)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " failed to create the rec audio thread");
- return -1;
- }
-
- unsigned int threadID(0);
- if (!_ptrThreadRec->Start(threadID))
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " failed to start the rec audio thread");
-
- delete _ptrThreadRec;
- _ptrThreadRec = NULL;
- return -1;
- }
- _recThreadID = threadID;
-
- // PLAYOUT
- threadName = "webrtc_audio_module_play_thread";
- _ptrThreadPlay = ThreadWrapper::CreateThread(PlayThreadFunc, this,
- kRealtimePriority, threadName);
- if (_ptrThreadPlay == NULL)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " failed to create the play audio thread");
- return -1;
- }
-
- threadID = 0;
- if (!_ptrThreadPlay->Start(threadID))
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " failed to start the play audio thread");
-
- delete _ptrThreadPlay;
- _ptrThreadPlay = NULL;
- return -1;
- }
- _playThreadID = threadID;
-
- _initialized = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::Terminate()
-{
-
- if (!_initialized)
- {
- return 0;
- }
-
- Lock();
-
- _mixerManager.Close();
-
- // RECORDING
- if (_ptrThreadRec)
- {
- ThreadWrapper* tmpThread = _ptrThreadRec;
- _ptrThreadRec = NULL;
- UnLock();
-
- tmpThread->SetNotAlive();
- _timeEventRec.Set();
- if (tmpThread->Stop())
- {
- delete tmpThread;
- } else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " failed to close down the rec audio thread");
- }
- // Lock again since we need to protect _ptrThreadPlay.
- Lock();
- }
-
- // PLAYOUT
- if (_ptrThreadPlay)
- {
- ThreadWrapper* tmpThread = _ptrThreadPlay;
- _ptrThreadPlay = NULL;
- _critSect.Leave();
-
- tmpThread->SetNotAlive();
- _timeEventPlay.Set();
- if (tmpThread->Stop())
- {
- delete tmpThread;
- } else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " failed to close down the play audio thread");
- }
- } else {
- UnLock();
- }
-
- // Terminate PulseAudio
- if (TerminatePulseAudio() < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to terminate PulseAudio");
- return -1;
- }
-
- _initialized = false;
- _outputDeviceIsSpecified = false;
- _inputDeviceIsSpecified = false;
-
- return 0;
-}
-
-bool AudioDeviceLinuxPulse::Initialized() const
-{
- return (_initialized);
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SpeakerIsAvailable(bool& available)
-{
-
- bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
- // Make an attempt to open up the
- // output mixer corresponding to the currently selected output device.
- //
- if (!wasInitialized && InitSpeaker() == -1)
- {
- available = false;
- return 0;
- }
-
- // Given that InitSpeaker was successful, we know that a valid speaker exists
- //
- available = true;
-
- // Close the initialized output mixer
- //
- if (!wasInitialized)
- {
- _mixerManager.CloseSpeaker();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::InitSpeaker()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_playing)
- {
- return -1;
- }
-
- if (!_outputDeviceIsSpecified)
- {
- return -1;
- }
-
- // check if default device
- if (_outputDeviceIndex == 0)
- {
- WebRtc_UWord16 deviceIndex = 0;
- GetDefaultDeviceInfo(false, NULL, deviceIndex);
- _paDeviceIndex = deviceIndex;
- } else
- {
- // get the PA device index from
- // the callback
- _deviceIndex = _outputDeviceIndex;
-
- // get playout devices
- PlayoutDevices();
- }
-
- // the callback has now set the _paDeviceIndex to
- // the PulseAudio index of the device
- if (_mixerManager.OpenSpeaker(_paDeviceIndex) == -1)
- {
- return -1;
- }
-
- // clear _deviceIndex
- _deviceIndex = -1;
- _paDeviceIndex = -1;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::MicrophoneIsAvailable(bool& available)
-{
-
- bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
- // Make an attempt to open up the
- // input mixer corresponding to the currently selected output device.
- //
- if (!wasInitialized && InitMicrophone() == -1)
- {
- available = false;
- return 0;
- }
-
- // Given that InitMicrophone was successful, we know that a valid microphone
- // exists
- available = true;
-
- // Close the initialized input mixer
- //
- if (!wasInitialized)
- {
- _mixerManager.CloseMicrophone();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::InitMicrophone()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_recording)
- {
- return -1;
- }
-
- if (!_inputDeviceIsSpecified)
- {
- return -1;
- }
-
- // Check if default device
- if (_inputDeviceIndex == 0)
- {
- WebRtc_UWord16 deviceIndex = 0;
- GetDefaultDeviceInfo(true, NULL, deviceIndex);
- _paDeviceIndex = deviceIndex;
- } else
- {
- // Get the PA device index from
- // the callback
- _deviceIndex = _inputDeviceIndex;
-
- // get recording devices
- RecordingDevices();
- }
-
- // The callback has now set the _paDeviceIndex to
- // the PulseAudio index of the device
- if (_mixerManager.OpenMicrophone(_paDeviceIndex) == -1)
- {
- return -1;
- }
-
- // Clear _deviceIndex
- _deviceIndex = -1;
- _paDeviceIndex = -1;
-
- return 0;
-}
-
-bool AudioDeviceLinuxPulse::SpeakerIsInitialized() const
-{
- return (_mixerManager.SpeakerIsInitialized());
-}
-
-bool AudioDeviceLinuxPulse::MicrophoneIsInitialized() const
-{
- return (_mixerManager.MicrophoneIsInitialized());
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SpeakerVolumeIsAvailable(bool& available)
-{
-
- bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
- // Make an attempt to open up the
- // output mixer corresponding to the currently selected output device.
- if (!wasInitialized && InitSpeaker() == -1)
- {
- // If we end up here it means that the selected speaker has no volume
- // control.
- available = false;
- return 0;
- }
-
- // Given that InitSpeaker was successful, we know that a volume control exists
- available = true;
-
- // Close the initialized output mixer
- if (!wasInitialized)
- {
- _mixerManager.CloseSpeaker();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SetSpeakerVolume(WebRtc_UWord32 volume)
-{
-
- return (_mixerManager.SetSpeakerVolume(volume));
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SpeakerVolume(WebRtc_UWord32& volume) const
-{
-
- WebRtc_UWord32 level(0);
-
- if (_mixerManager.SpeakerVolume(level) == -1)
- {
- return -1;
- }
-
- volume = level;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SetWaveOutVolume(
- WebRtc_UWord16 volumeLeft,
- WebRtc_UWord16 volumeRight)
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::WaveOutVolume(
- WebRtc_UWord16& /*volumeLeft*/,
- WebRtc_UWord16& /*volumeRight*/) const
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::MaxSpeakerVolume(
- WebRtc_UWord32& maxVolume) const
-{
-
- WebRtc_UWord32 maxVol(0);
-
- if (_mixerManager.MaxSpeakerVolume(maxVol) == -1)
- {
- return -1;
- }
-
- maxVolume = maxVol;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::MinSpeakerVolume(
- WebRtc_UWord32& minVolume) const
-{
-
- WebRtc_UWord32 minVol(0);
-
- if (_mixerManager.MinSpeakerVolume(minVol) == -1)
- {
- return -1;
- }
-
- minVolume = minVol;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SpeakerVolumeStepSize(
- WebRtc_UWord16& stepSize) const
-{
-
- WebRtc_UWord16 delta(0);
-
- if (_mixerManager.SpeakerVolumeStepSize(delta) == -1)
- {
- return -1;
- }
-
- stepSize = delta;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SpeakerMuteIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
- bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
- // Make an attempt to open up the
- // output mixer corresponding to the currently selected output device.
- //
- if (!wasInitialized && InitSpeaker() == -1)
- {
- // If we end up here it means that the selected speaker has no volume
- // control, hence it is safe to state that there is no mute control
- // already at this stage.
- available = false;
- return 0;
- }
-
- // Check if the selected speaker has a mute control
- _mixerManager.SpeakerMuteIsAvailable(isAvailable);
-
- available = isAvailable;
-
- // Close the initialized output mixer
- if (!wasInitialized)
- {
- _mixerManager.CloseSpeaker();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SetSpeakerMute(bool enable)
-{
-
- return (_mixerManager.SetSpeakerMute(enable));
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SpeakerMute(bool& enabled) const
-{
-
- bool muted(0);
- if (_mixerManager.SpeakerMute(muted) == -1)
- {
- return -1;
- }
-
- enabled = muted;
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::MicrophoneMuteIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
- bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
- // Make an attempt to open up the
- // input mixer corresponding to the currently selected input device.
- //
- if (!wasInitialized && InitMicrophone() == -1)
- {
- // If we end up here it means that the selected microphone has no volume
- // control, hence it is safe to state that there is no boost control
- // already at this stage.
- available = false;
- return 0;
- }
-
- // Check if the selected microphone has a mute control
- //
- _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized input mixer
- //
- if (!wasInitialized)
- {
- _mixerManager.CloseMicrophone();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SetMicrophoneMute(bool enable)
-{
-
- return (_mixerManager.SetMicrophoneMute(enable));
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::MicrophoneMute(bool& enabled) const
-{
-
- bool muted(0);
- if (_mixerManager.MicrophoneMute(muted) == -1)
- {
- return -1;
- }
-
- enabled = muted;
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::MicrophoneBoostIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
- bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
- // Enumerate all avaliable microphone and make an attempt to open up the
- // input mixer corresponding to the currently selected input device.
- //
- if (!wasInitialized && InitMicrophone() == -1)
- {
- // If we end up here it means that the selected microphone has no volume
- // control, hence it is safe to state that there is no boost control
- // already at this stage.
- available = false;
- return 0;
- }
-
- // Check if the selected microphone has a boost control
- _mixerManager.MicrophoneBoostIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized input mixer
- if (!wasInitialized)
- {
- _mixerManager.CloseMicrophone();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SetMicrophoneBoost(bool enable)
-{
-
- return (_mixerManager.SetMicrophoneBoost(enable));
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::MicrophoneBoost(bool& enabled) const
-{
-
- bool onOff(0);
-
- if (_mixerManager.MicrophoneBoost(onOff) == -1)
- {
- return -1;
- }
-
- enabled = onOff;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::StereoRecordingIsAvailable(bool& available)
-{
-
- if (_recChannels == 2 && _recording) {
- available = true;
- return 0;
- }
-
- available = false;
- bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
- int error = 0;
-
- if (!wasInitialized && InitMicrophone() == -1)
- {
- // Cannot open the specified device
- available = false;
- return 0;
- }
-
-#ifndef WEBRTC_PA_GTALK
- // Check if the selected microphone can record stereo.
- bool isAvailable(false);
- error = _mixerManager.StereoRecordingIsAvailable(isAvailable);
- if (!error)
- available = isAvailable;
-#endif
-
- // Close the initialized input mixer
- if (!wasInitialized)
- {
- _mixerManager.CloseMicrophone();
- }
-
- return error;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SetStereoRecording(bool enable)
-{
-
-#ifndef WEBRTC_PA_GTALK
- if (enable)
- _recChannels = 2;
- else
- _recChannels = 1;
-#endif
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::StereoRecording(bool& enabled) const
-{
-
- if (_recChannels == 2)
- enabled = true;
- else
- enabled = false;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::StereoPlayoutIsAvailable(bool& available)
-{
-
- if (_playChannels == 2 && _playing) {
- available = true;
- return 0;
- }
-
- available = false;
- bool wasInitialized = _mixerManager.SpeakerIsInitialized();
- int error = 0;
-
- if (!wasInitialized && InitSpeaker() == -1)
- {
- // Cannot open the specified device.
- return -1;
- }
-
-#ifndef WEBRTC_PA_GTALK
- // Check if the selected speaker can play stereo.
- bool isAvailable(false);
- error = _mixerManager.StereoPlayoutIsAvailable(isAvailable);
- if (!error)
- available = isAvailable;
-#endif
-
- // Close the initialized input mixer
- if (!wasInitialized)
- {
- _mixerManager.CloseSpeaker();
- }
-
- return error;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SetStereoPlayout(bool enable)
-{
-
-#ifndef WEBRTC_PA_GTALK
- if (enable)
- _playChannels = 2;
- else
- _playChannels = 1;
-#endif
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::StereoPlayout(bool& enabled) const
-{
-
- if (_playChannels == 2)
- enabled = true;
- else
- enabled = false;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SetAGC(bool enable)
-{
-
- _AGC = enable;
-
- return 0;
-}
-
-bool AudioDeviceLinuxPulse::AGC() const
-{
-
- return _AGC;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::MicrophoneVolumeIsAvailable(
- bool& available)
-{
-
- bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
- // Make an attempt to open up the
- // input mixer corresponding to the currently selected output device.
- if (!wasInitialized && InitMicrophone() == -1)
- {
- // If we end up here it means that the selected microphone has no volume
- // control.
- available = false;
- return 0;
- }
-
- // Given that InitMicrophone was successful, we know that a volume control
- // exists
- available = true;
-
- // Close the initialized input mixer
- if (!wasInitialized)
- {
- _mixerManager.CloseMicrophone();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SetMicrophoneVolume(WebRtc_UWord32 volume)
-{
-
- return (_mixerManager.SetMicrophoneVolume(volume));
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::MicrophoneVolume(
- WebRtc_UWord32& volume) const
-{
-
- WebRtc_UWord32 level(0);
-
- if (_mixerManager.MicrophoneVolume(level) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " failed to retrive current microphone level");
- return -1;
- }
-
- volume = level;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::MaxMicrophoneVolume(
- WebRtc_UWord32& maxVolume) const
-{
-
- WebRtc_UWord32 maxVol(0);
-
- if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1)
- {
- return -1;
- }
-
- maxVolume = maxVol;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::MinMicrophoneVolume(
- WebRtc_UWord32& minVolume) const
-{
-
- WebRtc_UWord32 minVol(0);
-
- if (_mixerManager.MinMicrophoneVolume(minVol) == -1)
- {
- return -1;
- }
-
- minVolume = minVol;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::MicrophoneVolumeStepSize(
- WebRtc_UWord16& stepSize) const
-{
-
- WebRtc_UWord16 delta(0);
-
- if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1)
- {
- return -1;
- }
-
- stepSize = delta;
-
- return 0;
-}
-
-WebRtc_Word16 AudioDeviceLinuxPulse::PlayoutDevices()
-{
-
- PaLock();
-
- pa_operation* paOperation = NULL;
- _numPlayDevices = 1; // init to 1 to account for "default"
-
- // get the whole list of devices and update _numPlayDevices
- paOperation = LATE(pa_context_get_sink_info_list)(_paContext,
- PaSinkInfoCallback,
- this);
-
- WaitForOperationCompletion(paOperation);
-
- PaUnLock();
-
- return _numPlayDevices;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SetPlayoutDevice(WebRtc_UWord16 index)
-{
-
- if (_playIsInitialized)
- {
- return -1;
- }
-
- const WebRtc_UWord16 nDevices = PlayoutDevices();
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " number of availiable output devices is %u", nDevices);
-
- if (index > (nDevices - 1))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " device index is out of range [0,%u]", (nDevices - 1));
- return -1;
- }
-
- _outputDeviceIndex = index;
- _outputDeviceIsSpecified = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SetPlayoutDevice(
- AudioDeviceModule::WindowsDeviceType /*device*/)
-{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "WindowsDeviceType not supported");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::PlayoutDeviceName(
- WebRtc_UWord16 index,
- WebRtc_Word8 name[kAdmMaxDeviceNameSize],
- WebRtc_Word8 guid[kAdmMaxGuidSize])
-{
-
- const WebRtc_UWord16 nDevices = PlayoutDevices();
-
- if ((index > (nDevices - 1)) || (name == NULL))
- {
- return -1;
- }
-
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- // Check if default device
- if (index == 0)
- {
- WebRtc_UWord16 deviceIndex = 0;
- return GetDefaultDeviceInfo(false, name, deviceIndex);
- }
-
- // Tell the callback that we want
- // The name for this device
- _playDisplayDeviceName = name;
- _deviceIndex = index;
-
- // get playout devices
- PlayoutDevices();
-
- // clear device name and index
- _playDisplayDeviceName = NULL;
- _deviceIndex = -1;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::RecordingDeviceName(
- WebRtc_UWord16 index,
- WebRtc_Word8 name[kAdmMaxDeviceNameSize],
- WebRtc_Word8 guid[kAdmMaxGuidSize])
-{
-
- const WebRtc_UWord16 nDevices(RecordingDevices());
-
- if ((index > (nDevices - 1)) || (name == NULL))
- {
- return -1;
- }
-
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- // Check if default device
- if (index == 0)
- {
- WebRtc_UWord16 deviceIndex = 0;
- return GetDefaultDeviceInfo(true, name, deviceIndex);
- }
-
- // Tell the callback that we want
- // the name for this device
- _recDisplayDeviceName = name;
- _deviceIndex = index;
-
- // Get recording devices
- RecordingDevices();
-
- // Clear device name and index
- _recDisplayDeviceName = NULL;
- _deviceIndex = -1;
-
- return 0;
-}
-
-WebRtc_Word16 AudioDeviceLinuxPulse::RecordingDevices()
-{
-
- PaLock();
-
- pa_operation* paOperation = NULL;
- _numRecDevices = 1; // Init to 1 to account for "default"
-
- // Get the whole list of devices and update _numRecDevices
- paOperation = LATE(pa_context_get_source_info_list)(_paContext,
- PaSourceInfoCallback,
- this);
-
- WaitForOperationCompletion(paOperation);
-
- PaUnLock();
-
- return _numRecDevices;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SetRecordingDevice(WebRtc_UWord16 index)
-{
-
- if (_recIsInitialized)
- {
- return -1;
- }
-
- const WebRtc_UWord16 nDevices(RecordingDevices());
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " number of availiable input devices is %u", nDevices);
-
- if (index > (nDevices - 1))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " device index is out of range [0,%u]", (nDevices - 1));
- return -1;
- }
-
- _inputDeviceIndex = index;
- _inputDeviceIsSpecified = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SetRecordingDevice(
- AudioDeviceModule::WindowsDeviceType /*device*/)
-{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "WindowsDeviceType not supported");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::PlayoutIsAvailable(bool& available)
-{
-
- available = false;
-
- // Try to initialize the playout side
- WebRtc_Word32 res = InitPlayout();
-
- // Cancel effect of initialization
- StopPlayout();
-
- if (res != -1)
- {
- available = true;
- }
-
- return res;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::RecordingIsAvailable(bool& available)
-{
-
- available = false;
-
- // Try to initialize the playout side
- WebRtc_Word32 res = InitRecording();
-
- // Cancel effect of initialization
- StopRecording();
-
- if (res != -1)
- {
- available = true;
- }
-
- return res;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::InitPlayout()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_playing)
- {
- return -1;
- }
-
- if (!_outputDeviceIsSpecified)
- {
- return -1;
- }
-
- if (_playIsInitialized)
- {
- return 0;
- }
-
- // Initialize the speaker (devices might have been added or removed)
- if (InitSpeaker() == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " InitSpeaker() failed");
- }
-
- // Set sampling rate to use
- WebRtc_UWord32 samplingRate = _samplingFreq * 1000;
- if (samplingRate == 44000)
- {
- samplingRate = 44100;
- }
-
- // Set the play sample specification
- pa_sample_spec playSampleSpec;
- playSampleSpec.channels = _playChannels;
- playSampleSpec.format = PA_SAMPLE_S16LE;
- playSampleSpec.rate = samplingRate;
-
- // Create a new play stream
- _playStream = LATE(pa_stream_new)(_paContext, "playStream",
- &playSampleSpec, NULL);
-
- if (!_playStream)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to create play stream, err=%d",
- LATE(pa_context_errno)(_paContext));
- return -1;
- }
-
- // Provide the playStream to the mixer
- _mixerManager.SetPlayStream(_playStream);
-
- if (_ptrAudioBuffer)
- {
- // Update audio buffer with the selected parameters
- _ptrAudioBuffer->SetPlayoutSampleRate(_samplingFreq * 1000);
- _ptrAudioBuffer->SetPlayoutChannels((WebRtc_UWord8) _playChannels);
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " stream state %d\n", LATE(pa_stream_get_state)(_playStream));
-
- // Set stream flags
- _playStreamFlags = (pa_stream_flags_t) (PA_STREAM_AUTO_TIMING_UPDATE
- | PA_STREAM_INTERPOLATE_TIMING);
-
- if (_configuredLatencyPlay != WEBRTC_PA_NO_LATENCY_REQUIREMENTS)
- {
- // If configuring a specific latency then we want to specify
- // PA_STREAM_ADJUST_LATENCY to make the server adjust parameters
- // automatically to reach that target latency. However, that flag doesn't
- // exist in Ubuntu 8.04 and many people still use that, so we have to check
- // the protocol version of libpulse.
- if (LATE(pa_context_get_protocol_version)(_paContext)
- >= WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION)
- {
- _playStreamFlags |= PA_STREAM_ADJUST_LATENCY;
- }
-
- const pa_sample_spec *spec =
- LATE(pa_stream_get_sample_spec)(_playStream);
- if (!spec)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " pa_stream_get_sample_spec()");
- return -1;
- }
-
- size_t bytesPerSec = LATE(pa_bytes_per_second)(spec);
- WebRtc_UWord32 latency = bytesPerSec
- * WEBRTC_PA_PLAYBACK_LATENCY_MINIMUM_MSECS / WEBRTC_PA_MSECS_PER_SEC;
-
- // Set the play buffer attributes
- _playBufferAttr.maxlength = latency; // num bytes stored in the buffer
- _playBufferAttr.tlength = latency; // target fill level of play buffer
- // minimum free num bytes before server request more data
- _playBufferAttr.minreq = latency / WEBRTC_PA_PLAYBACK_REQUEST_FACTOR;
- _playBufferAttr.prebuf = _playBufferAttr.tlength
- - _playBufferAttr.minreq; // prebuffer tlength before starting playout
-
- _configuredLatencyPlay = latency;
- }
-
- // num samples in bytes * num channels
- _playbackBufferSize = _samplingFreq * 10 * 2 * _playChannels;
- _playbackBufferUnused = _playbackBufferSize;
- _playBuffer = new WebRtc_Word8[_playbackBufferSize];
-
- // Enable underflow callback
- LATE(pa_stream_set_underflow_callback)(_playStream,
- PaStreamUnderflowCallback, this);
-
- // Set the state callback function for the stream
- LATE(pa_stream_set_state_callback)(_playStream, PaStreamStateCallback, this);
-
- // Mark playout side as initialized
- _playIsInitialized = true;
- _sndCardPlayDelay = 0;
- _sndCardRecDelay = 0;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::InitRecording()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_recording)
- {
- return -1;
- }
-
- if (!_inputDeviceIsSpecified)
- {
- return -1;
- }
-
- if (_recIsInitialized)
- {
- return 0;
- }
-
- // Initialize the microphone (devices might have been added or removed)
- if (InitMicrophone() == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " InitMicrophone() failed");
- }
-
- // Set sampling rate to use
- WebRtc_UWord32 samplingRate = _samplingFreq * 1000;
- if (samplingRate == 44000)
- {
- samplingRate = 44100;
- }
-
- // Set the rec sample specification
- pa_sample_spec recSampleSpec;
- recSampleSpec.channels = _recChannels;
- recSampleSpec.format = PA_SAMPLE_S16LE;
- recSampleSpec.rate = samplingRate;
-
- // Create a new rec stream
- _recStream = LATE(pa_stream_new)(_paContext, "recStream", &recSampleSpec,
- NULL);
- if (!_recStream)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to create rec stream, err=%d",
- LATE(pa_context_errno)(_paContext));
- return -1;
- }
-
- // Provide the recStream to the mixer
- _mixerManager.SetRecStream(_recStream);
-
- if (_ptrAudioBuffer)
- {
- // Update audio buffer with the selected parameters
- _ptrAudioBuffer->SetRecordingSampleRate(_samplingFreq * 1000);
- _ptrAudioBuffer->SetRecordingChannels((WebRtc_UWord8) _recChannels);
- }
-
- if (_configuredLatencyRec != WEBRTC_PA_NO_LATENCY_REQUIREMENTS)
- {
- _recStreamFlags = (pa_stream_flags_t) (PA_STREAM_AUTO_TIMING_UPDATE
- | PA_STREAM_INTERPOLATE_TIMING);
-
- // If configuring a specific latency then we want to specify
- // PA_STREAM_ADJUST_LATENCY to make the server adjust parameters
- // automatically to reach that target latency. However, that flag doesn't
- // exist in Ubuntu 8.04 and many people still use that, so we have to check
- // the protocol version of libpulse.
- if (LATE(pa_context_get_protocol_version)(_paContext)
- >= WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION)
- {
- _recStreamFlags |= PA_STREAM_ADJUST_LATENCY;
- }
-
- const pa_sample_spec *spec =
- LATE(pa_stream_get_sample_spec)(_recStream);
- if (!spec)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " pa_stream_get_sample_spec(rec)");
- return -1;
- }
-
- size_t bytesPerSec = LATE(pa_bytes_per_second)(spec);
- WebRtc_UWord32 latency = bytesPerSec
- * WEBRTC_PA_LOW_CAPTURE_LATENCY_MSECS / WEBRTC_PA_MSECS_PER_SEC;
-
- // Set the rec buffer attributes
- // Note: fragsize specifies a maximum transfer size, not a minimum, so
- // it is not possible to force a high latency setting, only a low one.
- _recBufferAttr.fragsize = latency; // size of fragment
- _recBufferAttr.maxlength = latency + bytesPerSec
- * WEBRTC_PA_CAPTURE_BUFFER_EXTRA_MSECS / WEBRTC_PA_MSECS_PER_SEC;
-
- _configuredLatencyRec = latency;
- }
-
- _recordBufferSize = _samplingFreq * 10 * 2 * _recChannels;
- _recordBufferUsed = 0;
- _recBuffer = new WebRtc_Word8[_recordBufferSize];
-
- // Enable overflow callback
- LATE(pa_stream_set_overflow_callback)(_recStream, PaStreamOverflowCallback,
- this);
-
- // Set the state callback function for the stream
- LATE(pa_stream_set_state_callback)(_recStream, PaStreamStateCallback, this);
-
- // Mark recording side as initialized
- _recIsInitialized = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::StartRecording()
-{
-
- if (!_recIsInitialized)
- {
- return -1;
- }
-
- if (_recording)
- {
- return 0;
- }
-
- // set state to ensure that the recording starts from the audio thread
- _startRec = true;
-
- // the audio thread will signal when recording has started
- _timeEventRec.Set();
- if (kEventTimeout == _recStartEvent.Wait(10000))
- {
- _startRec = false;
- StopRecording();
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to activate recording");
- return -1;
- }
-
- if (_recording)
- {
- // the recording state is set by the audio thread after recording has started
- } else
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to activate recording");
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::StopRecording()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_recIsInitialized)
- {
- return 0;
- }
-
- if (_recStream == NULL)
- {
- return -1;
- }
-
- _recIsInitialized = false;
- _recording = false;
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " stopping recording");
-
- // Stop Recording
- PaLock();
-
- DisableReadCallback();
- LATE(pa_stream_set_overflow_callback)(_recStream, NULL, NULL);
-
- // Unset this here so that we don't get a TERMINATED callback
- LATE(pa_stream_set_state_callback)(_recStream, NULL, NULL);
-
- if (LATE(pa_stream_get_state)(_recStream) != PA_STREAM_UNCONNECTED)
- {
- // Disconnect the stream
- if (LATE(pa_stream_disconnect)(_recStream) != PA_OK)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to disconnect rec stream, err=%d\n",
- LATE(pa_context_errno)(_paContext));
- PaUnLock();
- return -1;
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " disconnected recording");
- }
-
- LATE(pa_stream_unref)(_recStream);
- _recStream = NULL;
-
- PaUnLock();
-
- // Provide the recStream to the mixer
- _mixerManager.SetRecStream(_recStream);
-
- if (_recBuffer)
- {
- delete [] _recBuffer;
- _recBuffer = NULL;
- }
-
- return 0;
-}
-
-bool AudioDeviceLinuxPulse::RecordingIsInitialized() const
-{
- return (_recIsInitialized);
-}
-
-bool AudioDeviceLinuxPulse::Recording() const
-{
- return (_recording);
-}
-
-bool AudioDeviceLinuxPulse::PlayoutIsInitialized() const
-{
- return (_playIsInitialized);
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::StartPlayout()
-{
-
- if (!_playIsInitialized)
- {
- return -1;
- }
-
- if (_playing)
- {
- return 0;
- }
-
- // set state to ensure that playout starts from the audio thread
- _startPlay = true;
-
- // the audio thread will signal when playout has started
- _timeEventPlay.Set();
- if (kEventTimeout == _playStartEvent.Wait(10000))
- {
- _startPlay = false;
- StopPlayout();
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to activate playout");
- return -1;
- }
-
- if (_playing)
- {
- // the playing state is set by the audio thread after playout has started
- } else
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to activate playing");
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::StopPlayout()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_playIsInitialized)
- {
- return 0;
- }
-
- if (_playStream == NULL)
- {
- return -1;
- }
-
- _playIsInitialized = false;
- _playing = false;
- _sndCardPlayDelay = 0;
- _sndCardRecDelay = 0;
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " stopping playback");
-
- // Stop Playout
- PaLock();
-
- DisableWriteCallback();
- LATE(pa_stream_set_underflow_callback)(_playStream, NULL, NULL);
-
- // Unset this here so that we don't get a TERMINATED callback
- LATE(pa_stream_set_state_callback)(_playStream, NULL, NULL);
-
- if (LATE(pa_stream_get_state)(_playStream) != PA_STREAM_UNCONNECTED)
- {
- // Disconnect the stream
- if (LATE(pa_stream_disconnect)(_playStream) != PA_OK)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to disconnect play stream, err=%d",
- LATE(pa_context_errno)(_paContext));
- PaUnLock();
- return -1;
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " disconnected playback");
- }
-
- LATE(pa_stream_unref)(_playStream);
- _playStream = NULL;
-
- PaUnLock();
-
- // Provide the playStream to the mixer
- _mixerManager.SetPlayStream(_playStream);
-
- if (_playBuffer)
- {
- delete [] _playBuffer;
- _playBuffer = NULL;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::PlayoutDelay(WebRtc_UWord16& delayMS) const
-{
- delayMS = (WebRtc_UWord16) _sndCardPlayDelay;
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::RecordingDelay(WebRtc_UWord16& delayMS) const
-{
- delayMS = (WebRtc_UWord16) _sndCardRecDelay;
- return 0;
-}
-
-bool AudioDeviceLinuxPulse::Playing() const
-{
- return (_playing);
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::SetPlayoutBuffer(
- const AudioDeviceModule::BufferType type,
- WebRtc_UWord16 sizeMS)
-{
-
- if (type != AudioDeviceModule::kFixedBufferSize)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Adaptive buffer size not supported on this platform");
- return -1;
- }
-
- _playBufType = type;
- _playBufDelayFixed = sizeMS;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::PlayoutBuffer(
- AudioDeviceModule::BufferType& type,
- WebRtc_UWord16& sizeMS) const
-{
-
- type = _playBufType;
- sizeMS = _playBufDelayFixed;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::CPULoad(WebRtc_UWord16& /*load*/) const
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-bool AudioDeviceLinuxPulse::PlayoutWarning() const
-{
- CriticalSectionScoped lock(_critSect);
- return (_playWarning > 0);
-}
-
-bool AudioDeviceLinuxPulse::PlayoutError() const
-{
- CriticalSectionScoped lock(_critSect);
- return (_playError > 0);
-}
-
-bool AudioDeviceLinuxPulse::RecordingWarning() const
-{
- CriticalSectionScoped lock(_critSect);
- return (_recWarning > 0);
-}
-
-bool AudioDeviceLinuxPulse::RecordingError() const
-{
- CriticalSectionScoped lock(_critSect);
- return (_recError > 0);
-}
-
-void AudioDeviceLinuxPulse::ClearPlayoutWarning()
-{
- CriticalSectionScoped lock(_critSect);
- _playWarning = 0;
-}
-
-void AudioDeviceLinuxPulse::ClearPlayoutError()
-{
- CriticalSectionScoped lock(_critSect);
- _playError = 0;
-}
-
-void AudioDeviceLinuxPulse::ClearRecordingWarning()
-{
- CriticalSectionScoped lock(_critSect);
- _recWarning = 0;
-}
-
-void AudioDeviceLinuxPulse::ClearRecordingError()
-{
- CriticalSectionScoped lock(_critSect);
- _recError = 0;
-}
-
-// ============================================================================
-// Private Methods
-// ============================================================================
-
-void AudioDeviceLinuxPulse::PaContextStateCallback(pa_context *c, void *pThis)
-{
- static_cast<AudioDeviceLinuxPulse*> (pThis)->PaContextStateCallbackHandler(
- c);
-}
-
-// ----------------------------------------------------------------------------
-// PaSinkInfoCallback
-// ----------------------------------------------------------------------------
-
-void AudioDeviceLinuxPulse::PaSinkInfoCallback(pa_context */*c*/,
- const pa_sink_info *i, int eol,
- void *pThis)
-{
- static_cast<AudioDeviceLinuxPulse*> (pThis)->PaSinkInfoCallbackHandler(
- i, eol);
-}
-
-void AudioDeviceLinuxPulse::PaSourceInfoCallback(pa_context */*c*/,
- const pa_source_info *i,
- int eol, void *pThis)
-{
- static_cast<AudioDeviceLinuxPulse*> (pThis)->PaSourceInfoCallbackHandler(
- i, eol);
-}
-
-void AudioDeviceLinuxPulse::PaServerInfoCallback(pa_context */*c*/,
- const pa_server_info *i,
- void *pThis)
-{
- static_cast<AudioDeviceLinuxPulse*> (pThis)->PaServerInfoCallbackHandler(i);
-}
-
-void AudioDeviceLinuxPulse::PaStreamStateCallback(pa_stream *p, void *pThis)
-{
- static_cast<AudioDeviceLinuxPulse*> (pThis)->PaStreamStateCallbackHandler(p);
-}
-
-void AudioDeviceLinuxPulse::PaContextStateCallbackHandler(pa_context *c)
-{
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " context state cb");
-
- pa_context_state_t state = LATE(pa_context_get_state)(c);
- switch (state)
- {
- case PA_CONTEXT_UNCONNECTED:
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " unconnected");
- break;
- case PA_CONTEXT_CONNECTING:
- case PA_CONTEXT_AUTHORIZING:
- case PA_CONTEXT_SETTING_NAME:
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " no state");
- break;
- case PA_CONTEXT_FAILED:
- case PA_CONTEXT_TERMINATED:
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " failed");
- _paStateChanged = true;
- LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
- break;
- case PA_CONTEXT_READY:
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " ready");
- _paStateChanged = true;
- LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
- break;
- }
-}
-
-void AudioDeviceLinuxPulse::PaSinkInfoCallbackHandler(const pa_sink_info *i,
- int eol)
-{
- if (eol)
- {
- // Signal that we are done
- LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
- return;
- }
-
- if (_numPlayDevices == _deviceIndex)
- {
- // Convert the device index to the one of the sink
- _paDeviceIndex = i->index;
-
- if (_playDeviceName)
- {
- // Copy the sink name
- strncpy(_playDeviceName, i->name, kAdmMaxDeviceNameSize);
- _playDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
- }
- if (_playDisplayDeviceName)
- {
- // Copy the sink display name
- strncpy(_playDisplayDeviceName, i->description,
- kAdmMaxDeviceNameSize);
- _playDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
- }
- }
-
- _numPlayDevices++;
-}
-
-void AudioDeviceLinuxPulse::PaSourceInfoCallbackHandler(
- const pa_source_info *i,
- int eol)
-{
- if (eol)
- {
- // Signal that we are done
- LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
- return;
- }
-
- // We don't want to list output devices
- if (i->monitor_of_sink == PA_INVALID_INDEX)
- {
- if (_numRecDevices == _deviceIndex)
- {
- // Convert the device index to the one of the source
- _paDeviceIndex = i->index;
-
- if (_recDeviceName)
- {
- // copy the source name
- strncpy(_recDeviceName, i->name, kAdmMaxDeviceNameSize);
- _recDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
- }
- if (_recDisplayDeviceName)
- {
- // Copy the source display name
- strncpy(_recDisplayDeviceName, i->description,
- kAdmMaxDeviceNameSize);
- _recDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
- }
- }
-
- _numRecDevices++;
- }
-}
-
-void AudioDeviceLinuxPulse::PaServerInfoCallbackHandler(const pa_server_info *i)
-{
- // Use PA native sampling rate
- WebRtc_UWord32 paSampleRate = i->sample_spec.rate;
- if (paSampleRate == 44100)
- {
-#ifdef WEBRTC_PA_GTALK
- paSampleRate = 48000;
-#else
- paSampleRate = 44000;
-#endif
- }
-
- _samplingFreq = paSampleRate / 1000;
-
- // Copy the PA server version
- if (_paServerVersion)
- {
- strncpy(_paServerVersion, i->server_version, 31);
- _paServerVersion[31] = '\0';
- }
-
- if (_recDisplayDeviceName)
- {
- // Copy the source name
- strncpy(_recDisplayDeviceName, i->default_source_name,
- kAdmMaxDeviceNameSize);
- _recDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
- }
-
- if (_playDisplayDeviceName)
- {
- // Copy the sink name
- strncpy(_playDisplayDeviceName, i->default_sink_name,
- kAdmMaxDeviceNameSize);
- _playDisplayDeviceName[kAdmMaxDeviceNameSize - 1] = '\0';
- }
-
- LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
-}
-
-void AudioDeviceLinuxPulse::PaStreamStateCallbackHandler(pa_stream *p)
-{
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " stream state cb");
-
- pa_stream_state_t state = LATE(pa_stream_get_state)(p);
- switch (state)
- {
- case PA_STREAM_UNCONNECTED:
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " unconnected");
- break;
- case PA_STREAM_CREATING:
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " creating");
- break;
- case PA_STREAM_FAILED:
- case PA_STREAM_TERMINATED:
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " failed");
- break;
- case PA_STREAM_READY:
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " ready");
- break;
- }
-
- LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::CheckPulseAudioVersion()
-{
- /*WebRtc_Word32 index = 0;
- WebRtc_Word32 partIndex = 0;
- WebRtc_Word32 partNum = 1;
- WebRtc_Word32 minVersion[3] = {0, 9, 15};
- bool versionOk = false;
- char str[8] = {0};*/
-
- PaLock();
-
- pa_operation* paOperation = NULL;
-
- // get the server info and update deviceName
- paOperation = LATE(pa_context_get_server_info)(_paContext,
- PaServerInfoCallback, this);
-
- WaitForOperationCompletion(paOperation);
-
- PaUnLock();
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1,
- " checking PulseAudio version: %s", _paServerVersion);
-
- /* Saved because it may turn out that we need to check the version in the future
- while (true)
- {
- if (_paServerVersion[index] == '.')
- {
- index++;
- str[partIndex] = '\0';
- partIndex = 0;
-
- if(partNum == 2)
- {
- if (atoi(str) < minVersion[1])
- {
- break;
- }
- partNum = 3;
- }
- else
- {
- if (atoi(str) > minVersion[0])
- {
- versionOk = true;
- break;
- }
- partNum = 2;
- }
- }
- else if (_paServerVersion[index] == '\0' || _paServerVersion[index] == '-')
- {
- str[partIndex] = '\0';
- if (atoi(str) >= minVersion[2])
- {
- versionOk = true;
- }
- break;
- }
-
- str[partIndex] = _paServerVersion[index];
- index++;
- partIndex++;
- }
-
- if (!versionOk)
- {
- return -1;
- }
- */
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::InitSamplingFrequency()
-{
- PaLock();
-
- pa_operation* paOperation = NULL;
-
- // Get the server info and update _samplingFreq
- paOperation = LATE(pa_context_get_server_info)(_paContext,
- PaServerInfoCallback, this);
-
- WaitForOperationCompletion(paOperation);
-
- PaUnLock();
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::GetDefaultDeviceInfo(bool recDevice,
- WebRtc_Word8* name,
- WebRtc_UWord16& index)
-{
- WebRtc_Word8 tmpName[kAdmMaxDeviceNameSize] = {0};
- // subtract length of "default: "
- WebRtc_UWord16 nameLen = kAdmMaxDeviceNameSize - 9;
- WebRtc_Word8* pName = NULL;
-
- if (name)
- {
- // Add "default: "
- strcpy(name, "default: ");
- pName = &name[9];
- }
-
- // Tell the callback that we want
- // the name for this device
- if (recDevice)
- {
- _recDisplayDeviceName = tmpName;
- } else
- {
- _playDisplayDeviceName = tmpName;
- }
-
- // Set members
- _paDeviceIndex = -1;
- _deviceIndex = 0;
- _numPlayDevices = 0;
- _numRecDevices = 0;
-
- PaLock();
-
- pa_operation* paOperation = NULL;
-
- // Get the server info and update deviceName
- paOperation = LATE(pa_context_get_server_info)(_paContext,
- PaServerInfoCallback, this);
-
- WaitForOperationCompletion(paOperation);
-
- // Get the device index
- if (recDevice)
- {
- paOperation
- = LATE(pa_context_get_source_info_by_name)(_paContext,
- (char *) tmpName,
- PaSourceInfoCallback,
- this);
- } else
- {
- paOperation
- = LATE(pa_context_get_sink_info_by_name)(_paContext,
- (char *) tmpName,
- PaSinkInfoCallback, this);
- }
-
- WaitForOperationCompletion(paOperation);
-
- PaUnLock();
-
- // Set the index
- index = _paDeviceIndex;
-
- if (name)
- {
- // Copy to name string
- strncpy(pName, tmpName, nameLen);
- }
-
- // Clear members
- _playDisplayDeviceName = NULL;
- _recDisplayDeviceName = NULL;
- _paDeviceIndex = -1;
- _deviceIndex = -1;
- _numPlayDevices = 0;
- _numRecDevices = 0;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::InitPulseAudio()
-{
- int retVal = 0;
-
- // Load libpulse
- if (!PaSymbolTable.Load())
- {
- // Most likely the Pulse library and sound server are not installed on
- // this system
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to load symbol table");
- return -1;
- }
-
- // Create a mainloop API and connection to the default server
- // the mainloop is the internal asynchronous API event loop
- if (_paMainloop) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " PA mainloop has already existed");
- return -1;
- }
- _paMainloop = LATE(pa_threaded_mainloop_new)();
- if (!_paMainloop)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " could not create mainloop");
- return -1;
- }
-
- // Start the threaded main loop
- retVal = LATE(pa_threaded_mainloop_start)(_paMainloop);
- if (retVal != PA_OK)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to start main loop, error=%d", retVal);
- return -1;
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " mainloop running!");
-
- PaLock();
-
- _paMainloopApi = LATE(pa_threaded_mainloop_get_api)(_paMainloop);
- if (!_paMainloopApi)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " could not create mainloop API");
- PaUnLock();
- return -1;
- }
-
- // Create a new PulseAudio context
- if (_paContext){
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " PA context has already existed");
- PaUnLock();
- return -1;
- }
- _paContext = LATE(pa_context_new)(_paMainloopApi, "WEBRTC VoiceEngine");
-
- if (!_paContext)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " could not create context");
- PaUnLock();
- return -1;
- }
-
- // Set state callback function
- LATE(pa_context_set_state_callback)(_paContext, PaContextStateCallback,
- this);
-
- // Connect the context to a server (default)
- _paStateChanged = false;
- retVal = LATE(pa_context_connect)(_paContext, NULL, PA_CONTEXT_NOAUTOSPAWN,
- NULL);
-
- if (retVal != PA_OK)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to connect context, error=%d", retVal);
- PaUnLock();
- return -1;
- }
-
- // Wait for state change
- while (!_paStateChanged)
- {
- LATE(pa_threaded_mainloop_wait)(_paMainloop);
- }
-
- // Now check to see what final state we reached.
- pa_context_state_t state = LATE(pa_context_get_state)(_paContext);
-
- if (state != PA_CONTEXT_READY)
- {
- if (state == PA_CONTEXT_FAILED)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to connect to PulseAudio sound server");
- } else if (state == PA_CONTEXT_TERMINATED)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " PulseAudio connection terminated early");
- } else
- {
- // Shouldn't happen, because we only signal on one of those three
- // states
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " unknown problem connecting to PulseAudio");
- }
- PaUnLock();
- return -1;
- }
-
- PaUnLock();
-
- // Give the objects to the mixer manager
- _mixerManager.SetPulseAudioObjects(_paMainloop, _paContext);
-
- // Check the version
- if (CheckPulseAudioVersion() < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " PulseAudio version %s not supported", _paServerVersion);
- return -1;
- }
-
- // Initialize sampling frequency
- if (InitSamplingFrequency() < 0 || _samplingFreq == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to initialize sampling frequency, set to %d",
- _samplingFreq);
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::TerminatePulseAudio()
-{
- // Do nothing if the instance doesn't exist
- // likely PaSymbolTable.Load() fails
- if (!_paMainloop) {
- return 0;
- }
-
- PaLock();
-
- // Disconnect the context
- if (_paContext)
- {
- LATE(pa_context_disconnect)(_paContext);
- }
-
- // Unreference the context
- if (_paContext)
- {
- LATE(pa_context_unref)(_paContext);
- }
-
- PaUnLock();
- _paContext = NULL;
-
- // Stop the threaded main loop
- if (_paMainloop)
- {
- LATE(pa_threaded_mainloop_stop)(_paMainloop);
- }
-
- // Free the mainloop
- if (_paMainloop)
- {
- LATE(pa_threaded_mainloop_free)(_paMainloop);
- }
-
- _paMainloop = NULL;
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " PulseAudio terminated");
-
- return 0;
-}
-
-void AudioDeviceLinuxPulse::PaLock()
-{
- LATE(pa_threaded_mainloop_lock)(_paMainloop);
-}
-
-void AudioDeviceLinuxPulse::PaUnLock()
-{
- LATE(pa_threaded_mainloop_unlock)(_paMainloop);
-}
-
-void AudioDeviceLinuxPulse::WaitForOperationCompletion(
- pa_operation* paOperation) const
-{
- if (!paOperation)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "paOperation NULL in WaitForOperationCompletion");
- return;
- }
-
- while (LATE(pa_operation_get_state)(paOperation) == PA_OPERATION_RUNNING)
- {
- LATE(pa_threaded_mainloop_wait)(_paMainloop);
- }
-
- LATE(pa_operation_unref)(paOperation);
-}
-
-// ============================================================================
-// Thread Methods
-// ============================================================================
-
-void AudioDeviceLinuxPulse::EnableWriteCallback()
-{
- if (LATE(pa_stream_get_state)(_playStream) == PA_STREAM_READY)
- {
- // May already have available space. Must check.
- _tempBufferSpace = LATE(pa_stream_writable_size)(_playStream);
- if (_tempBufferSpace > 0)
- {
- // Yup, there is already space available, so if we register a write
- // callback then it will not receive any event. So dispatch one ourself
- // instead
- _timeEventPlay.Set();
- return;
- }
- }
-
- LATE(pa_stream_set_write_callback)(_playStream, &PaStreamWriteCallback,
- this);
-}
-
-void AudioDeviceLinuxPulse::DisableWriteCallback()
-{
- LATE(pa_stream_set_write_callback)(_playStream, NULL, NULL);
-}
-
-void AudioDeviceLinuxPulse::PaStreamWriteCallback(pa_stream */*unused*/,
- size_t buffer_space,
- void *pThis)
-{
- static_cast<AudioDeviceLinuxPulse*> (pThis)->PaStreamWriteCallbackHandler(
- buffer_space);
-}
-
-void AudioDeviceLinuxPulse::PaStreamWriteCallbackHandler(size_t bufferSpace)
-{
- _tempBufferSpace = bufferSpace;
-
- // Since we write the data asynchronously on a different thread, we have
- // to temporarily disable the write callback or else Pulse will call it
- // continuously until we write the data. We re-enable it below.
- DisableWriteCallback();
- _timeEventPlay.Set();
-}
-
-void AudioDeviceLinuxPulse::PaStreamUnderflowCallback(pa_stream */*unused*/,
- void *pThis)
-{
- static_cast<AudioDeviceLinuxPulse*> (pThis)->PaStreamUnderflowCallbackHandler();
-}
-
-void AudioDeviceLinuxPulse::PaStreamUnderflowCallbackHandler()
-{
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Playout underflow");
-
- if (_configuredLatencyPlay == WEBRTC_PA_NO_LATENCY_REQUIREMENTS)
- {
- // We didn't configure a pa_buffer_attr before, so switching to one now
- // would be questionable.
- return;
- }
-
- // Otherwise reconfigure the stream with a higher target latency.
-
- const pa_sample_spec *spec = LATE(pa_stream_get_sample_spec)(_playStream);
- if (!spec)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " pa_stream_get_sample_spec()");
- return;
- }
-
- size_t bytesPerSec = LATE(pa_bytes_per_second)(spec);
- WebRtc_UWord32 newLatency = _configuredLatencyPlay + bytesPerSec
- * WEBRTC_PA_PLAYBACK_LATENCY_INCREMENT_MSECS / WEBRTC_PA_MSECS_PER_SEC;
-
- // Set the play buffer attributes
- _playBufferAttr.maxlength = newLatency;
- _playBufferAttr.tlength = newLatency;
- _playBufferAttr.minreq = newLatency / WEBRTC_PA_PLAYBACK_REQUEST_FACTOR;
- _playBufferAttr.prebuf = _playBufferAttr.tlength - _playBufferAttr.minreq;
-
- pa_operation *op = LATE(pa_stream_set_buffer_attr)(_playStream,
- &_playBufferAttr, NULL,
- NULL);
- if (!op)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " pa_stream_set_buffer_attr()");
- return;
- }
-
- // Don't need to wait for this to complete.
- LATE(pa_operation_unref)(op);
-
- // Save the new latency in case we underflow again.
- _configuredLatencyPlay = newLatency;
-}
-
-void AudioDeviceLinuxPulse::EnableReadCallback()
-{
- LATE(pa_stream_set_read_callback)(_recStream, &PaStreamReadCallback, this);
-}
-
-void AudioDeviceLinuxPulse::DisableReadCallback()
-{
- LATE(pa_stream_set_read_callback)(_recStream, NULL, NULL);
-}
-
-void AudioDeviceLinuxPulse::PaStreamReadCallback(pa_stream */*unused1*/,
- size_t /*unused2*/,
- void *pThis)
-{
- static_cast<AudioDeviceLinuxPulse*> (pThis)->PaStreamReadCallbackHandler();
-}
-
-void AudioDeviceLinuxPulse::PaStreamReadCallbackHandler()
-{
- // We get the data pointer and size now in order to save one Lock/Unlock
- // in the worker thread
- if (LATE(pa_stream_peek)(_recStream, &_tempSampleData, &_tempSampleDataSize)
- != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Can't read data!");
- return;
- }
-
- // Since we consume the data asynchronously on a different thread, we have
- // to temporarily disable the read callback or else Pulse will call it
- // continuously until we consume the data. We re-enable it below
- DisableReadCallback();
- _timeEventRec.Set();
-}
-
-void AudioDeviceLinuxPulse::PaStreamOverflowCallback(pa_stream */*unused*/,
- void *pThis)
-{
- static_cast<AudioDeviceLinuxPulse*> (pThis)->PaStreamOverflowCallbackHandler();
-}
-
-void AudioDeviceLinuxPulse::PaStreamOverflowCallbackHandler()
-{
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Recording overflow");
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::LatencyUsecs(pa_stream *stream)
-{
- if (!WEBRTC_PA_REPORT_LATENCY)
- {
- return 0;
- }
-
- if (!stream)
- {
- return 0;
- }
-
- pa_usec_t latency;
- int negative;
- if (LATE(pa_stream_get_latency)(stream, &latency, &negative) != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Can't query latency");
- // We'd rather continue playout/capture with an incorrect delay than stop
- // it altogether, so return a valid value.
- return 0;
- }
-
- if (negative)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " warning: pa_stream_get_latency reported negative delay");
-
- // The delay can be negative for monitoring streams if the captured
- // samples haven't been played yet. In such a case, "latency" contains the
- // magnitude, so we must negate it to get the real value.
- WebRtc_Word32 tmpLatency = (WebRtc_Word32) -latency;
- if (tmpLatency < 0)
- {
- // Make sure that we don't use a negative delay
- tmpLatency = 0;
- }
-
- return tmpLatency;
- } else
- {
- return (WebRtc_Word32) latency;
- }
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::ReadRecordedData(const void* bufferData,
- size_t bufferSize)
-{
- size_t size = bufferSize;
- WebRtc_UWord32 numRecSamples = _recordBufferSize / (2 * _recChannels);
-
- // Account for the peeked data and the used data
- WebRtc_UWord32 recDelay = (WebRtc_UWord32) ((LatencyUsecs(_recStream)
- / 1000) + 10 * ((size + _recordBufferUsed) / _recordBufferSize));
-
- _sndCardRecDelay = recDelay;
-
- if (_playStream)
- {
- // Get the playout delay
- _sndCardPlayDelay = (WebRtc_UWord32) (LatencyUsecs(_playStream) / 1000);
- }
-
- if (_recordBufferUsed > 0)
- {
- // Have to copy to the buffer until it is full
- size_t copy = _recordBufferSize - _recordBufferUsed;
- if (size < copy)
- {
- copy = size;
- }
-
- memcpy(&_recBuffer[_recordBufferUsed], bufferData, copy);
- _recordBufferUsed += copy;
- bufferData = static_cast<const char *> (bufferData) + copy;
- size -= copy;
-
- if (_recordBufferUsed != _recordBufferSize)
- {
- // Not enough data yet to pass to VoE
- return 0;
- }
-
- // Provide data to VoiceEngine
- if (ProcessRecordedData(_recBuffer, numRecSamples, recDelay) == -1)
- {
- // We have stopped recording
- return -1;
- }
-
- _recordBufferUsed = 0;
- }
-
- // Now process full 10ms sample sets directly from the input
- while (size >= _recordBufferSize)
- {
- // Provide data to VoiceEngine
- if (ProcessRecordedData(
- static_cast<WebRtc_Word8 *> (const_cast<void *> (bufferData)),
- numRecSamples, recDelay) == -1)
- {
- // We have stopped recording
- return -1;
- }
-
- bufferData = static_cast<const char *> (bufferData) + _recordBufferSize;
- size -= _recordBufferSize;
-
- // We have consumed 10ms of data
- recDelay -= 10;
- }
-
- // Now save any leftovers for later.
- if (size > 0)
- {
- memcpy(_recBuffer, bufferData, size);
- _recordBufferUsed = size;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceLinuxPulse::ProcessRecordedData(
- WebRtc_Word8 *bufferData,
- WebRtc_UWord32 bufferSizeInSamples,
- WebRtc_UWord32 recDelay)
-{
- WebRtc_UWord32 currentMicLevel(0);
- WebRtc_UWord32 newMicLevel(0);
-
- _ptrAudioBuffer->SetRecordedBuffer(bufferData, bufferSizeInSamples);
-
- if (AGC())
- {
- // Store current mic level in the audio buffer if AGC is enabled
- if (MicrophoneVolume(currentMicLevel) == 0)
- {
- // This call does not affect the actual microphone volume
- _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
- }
- }
-
- // Set vqe data
- const WebRtc_UWord32 clockDrift(0);
- _ptrAudioBuffer->SetVQEData(_sndCardPlayDelay, recDelay, clockDrift);
-
- // Deliver recorded samples at specified sample rate,
- // mic level etc. to the observer using callback
- UnLock();
- _ptrAudioBuffer->DeliverRecordedData();
- Lock();
-
- // We have been unlocked - check the flag again
- if (!_recording)
- {
- return -1;
- }
-
- if (AGC())
- {
- newMicLevel = _ptrAudioBuffer->NewMicLevel();
- if (newMicLevel != 0)
- {
- // The VQE will only deliver non-zero microphone levels when a
- // change is needed.
- // Set this new mic level (received from the observer as return
- // value in the callback).
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id,
- " AGC change of volume: old=%u => new=%u",
- currentMicLevel, newMicLevel);
- if (SetMicrophoneVolume(newMicLevel) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
- _id,
- " the required modification of the microphone "
- "volume failed");
- }
- }
- }
-
- return 0;
-}
-
-bool AudioDeviceLinuxPulse::PlayThreadFunc(void* pThis)
-{
- return (static_cast<AudioDeviceLinuxPulse*> (pThis)->PlayThreadProcess());
-}
-
-bool AudioDeviceLinuxPulse::RecThreadFunc(void* pThis)
-{
- return (static_cast<AudioDeviceLinuxPulse*> (pThis)->RecThreadProcess());
-}
-
-bool AudioDeviceLinuxPulse::PlayThreadProcess()
-{
- switch (_timeEventPlay.Wait(1000))
- {
- case kEventSignaled:
- _timeEventPlay.Reset();
- break;
- case kEventError:
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "EventWrapper::Wait() failed");
- return true;
- case kEventTimeout:
- return true;
- }
-
- Lock();
-
- if (_startPlay)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "_startPlay true, performing initial actions");
-
- _startPlay = false;
- _playDeviceName = NULL;
-
- // Set if not default device
- if (_outputDeviceIndex > 0)
- {
- // Get the playout device name
- _playDeviceName = new WebRtc_Word8[kAdmMaxDeviceNameSize];
- _deviceIndex = _outputDeviceIndex;
- PlayoutDevices();
- }
-
- // Start muted only supported on 0.9.11 and up
- if (LATE(pa_context_get_protocol_version)(_paContext)
- >= WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION)
- {
- // Get the currently saved speaker mute status
- // and set the initial mute status accordingly
- bool enabled(false);
- _mixerManager.SpeakerMute(enabled);
- if (enabled)
- {
- _playStreamFlags |= PA_STREAM_START_MUTED;
- }
- }
-
- // Get the currently saved speaker volume
- WebRtc_UWord32 volume = 0;
- _mixerManager.SpeakerVolume(volume);
-
- PaLock();
-
- // Set the same volume for all channels
- pa_cvolume cVolumes;
- const pa_sample_spec *spec =
- LATE(pa_stream_get_sample_spec)(_playStream);
- LATE(pa_cvolume_set)(&cVolumes, spec->channels, volume);
-
- // Connect the stream to a sink
- if (LATE(pa_stream_connect_playback)(
- _playStream,
- _playDeviceName,
- &_playBufferAttr,
- (pa_stream_flags_t) _playStreamFlags,
- &cVolumes, NULL) != PA_OK)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to connect play stream, err=%d",
- LATE(pa_context_errno)(_paContext));
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " play stream connected");
-
- // Wait for state change
- while (LATE(pa_stream_get_state)(_playStream) != PA_STREAM_READY)
- {
- LATE(pa_threaded_mainloop_wait)(_paMainloop);
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " play stream ready");
-
- // We can now handle write callbacks
- EnableWriteCallback();
-
- PaUnLock();
-
- // Clear device name
- if (_playDeviceName)
- {
- delete [] _playDeviceName;
- _playDeviceName = NULL;
- }
-
- _playing = true;
- _playStartEvent.Set();
-
- UnLock();
- return true;
- }
-
- if (_playing)
- {
- if (!_recording)
- {
- // Update the playout delay
- _sndCardPlayDelay = (WebRtc_UWord32) (LatencyUsecs(_playStream)
- / 1000);
- }
-
- if (_playbackBufferUnused < _playbackBufferSize)
- {
-
- size_t write = _playbackBufferSize - _playbackBufferUnused;
- if (_tempBufferSpace < write)
- {
- write = _tempBufferSpace;
- }
-
- PaLock();
- if (LATE(pa_stream_write)(
- _playStream,
- (void *) &_playBuffer[_playbackBufferUnused],
- write, NULL, (int64_t) 0,
- PA_SEEK_RELATIVE) != PA_OK)
- {
- _writeErrors++;
- if (_writeErrors > 10)
- {
- if (_playError == 1)
- {
- WEBRTC_TRACE(kTraceWarning,
- kTraceUtility, _id,
- " pending playout error exists");
- }
- _playError = 1; // Triggers callback from module process thread
- WEBRTC_TRACE(
- kTraceError,
- kTraceUtility,
- _id,
- " kPlayoutError message posted: "
- "_writeErrors=%u, error=%d",
- _writeErrors,
- LATE(pa_context_errno)(_paContext));
- _writeErrors = 0;
- }
- }
- PaUnLock();
-
- _playbackBufferUnused += write;
- _tempBufferSpace -= write;
- }
-
- WebRtc_UWord32 numPlaySamples = _playbackBufferSize / (2
- * _playChannels);
- if (_tempBufferSpace > 0) // Might have been reduced to zero by the above
- {
- // Ask for new PCM data to be played out using the AudioDeviceBuffer
- // ensure that this callback is executed without taking the
- // audio-thread lock
- UnLock();
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " requesting data");
- WebRtc_UWord32 nSamples =
- _ptrAudioBuffer->RequestPlayoutData(numPlaySamples);
- Lock();
-
- // We have been unlocked - check the flag again
- if (!_playing)
- {
- UnLock();
- return true;
- }
-
- nSamples = _ptrAudioBuffer->GetPlayoutData(_playBuffer);
- if (nSamples != numPlaySamples)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice,
- _id, " invalid number of output samples(%d)",
- nSamples);
- }
-
- size_t write = _playbackBufferSize;
- if (_tempBufferSpace < write)
- {
- write = _tempBufferSpace;
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " will write");
- PaLock();
- if (LATE(pa_stream_write)(_playStream, (void *) &_playBuffer[0],
- write, NULL, (int64_t) 0,
- PA_SEEK_RELATIVE) != PA_OK)
- {
- _writeErrors++;
- if (_writeErrors > 10)
- {
- if (_playError == 1)
- {
- WEBRTC_TRACE(kTraceWarning,
- kTraceUtility, _id,
- " pending playout error exists");
- }
- _playError = 1; // triggers callback from module process thread
- WEBRTC_TRACE(
- kTraceError,
- kTraceUtility,
- _id,
- " kPlayoutError message posted: "
- "_writeErrors=%u, error=%d",
- _writeErrors,
- LATE(pa_context_errno)(_paContext));
- _writeErrors = 0;
- }
- }
- PaUnLock();
-
- _playbackBufferUnused = write;
- }
-
- _tempBufferSpace = 0;
- PaLock();
- EnableWriteCallback();
- PaUnLock();
-
- } // _playing
-
- UnLock();
- return true;
-}
-
-bool AudioDeviceLinuxPulse::RecThreadProcess()
-{
- switch (_timeEventRec.Wait(1000))
- {
- case kEventSignaled:
- _timeEventRec.Reset();
- break;
- case kEventError:
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "EventWrapper::Wait() failed");
- return true;
- case kEventTimeout:
- return true;
- }
-
- Lock();
-
- if (_startRec)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "_startRec true, performing initial actions");
-
- _recDeviceName = NULL;
-
- // Set if not default device
- if (_inputDeviceIndex > 0)
- {
- // Get the recording device name
- _recDeviceName = new WebRtc_Word8[kAdmMaxDeviceNameSize];
- _deviceIndex = _inputDeviceIndex;
- RecordingDevices();
- }
-
- PaLock();
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " connecting stream");
-
- // Connect the stream to a source
- if (LATE(pa_stream_connect_record)(_recStream, _recDeviceName,
- &_recBufferAttr,
- (pa_stream_flags_t) _recStreamFlags)
- != PA_OK)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to connect rec stream, err=%d",
- LATE(pa_context_errno)(_paContext));
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " connected");
-
- // Wait for state change
- while (LATE(pa_stream_get_state)(_recStream) != PA_STREAM_READY)
- {
- LATE(pa_threaded_mainloop_wait)(_paMainloop);
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " done");
-
- // We can now handle read callbacks
- EnableReadCallback();
-
- PaUnLock();
-
- // Clear device name
- if (_recDeviceName)
- {
- delete [] _recDeviceName;
- _recDeviceName = NULL;
- }
-
- _startRec = false;
- _recording = true;
- _recStartEvent.Set();
-
- UnLock();
- return true;
- }
-
- if (_recording)
- {
- // Read data and provide it to VoiceEngine
- if (ReadRecordedData(_tempSampleData, _tempSampleDataSize) == -1)
- {
- UnLock();
- return true;
- }
-
- _tempSampleData = NULL;
- _tempSampleDataSize = 0;
-
- PaLock();
- while (true)
- {
- // Ack the last thing we read
- if (LATE(pa_stream_drop)(_recStream) != 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
- _id, " failed to drop, err=%d\n",
- LATE(pa_context_errno)(_paContext));
- }
-
- if (LATE(pa_stream_readable_size)(_recStream) <= 0)
- {
- // Then that was all the data
- break;
- }
-
- // Else more data.
- const void *sampleData;
- size_t sampleDataSize;
-
- if (LATE(pa_stream_peek)(_recStream, &sampleData, &sampleDataSize)
- != 0)
- {
- _recError = 1; // triggers callback from module process thread
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice,
- _id, " RECORD_ERROR message posted, error = %d",
- LATE(pa_context_errno)(_paContext));
- break;
- }
-
- _sndCardRecDelay = (WebRtc_UWord32) (LatencyUsecs(_recStream)
- / 1000);
-
- // Drop lock for sigslot dispatch, which could take a while.
- PaUnLock();
- // Read data and provide it to VoiceEngine
- if (ReadRecordedData(sampleData, sampleDataSize) == -1)
- {
- UnLock();
- return true;
- }
- PaLock();
-
- // Return to top of loop for the ack and the check for more data.
- }
-
- EnableReadCallback();
- PaUnLock();
-
- } // _recording
-
- UnLock();
- return true;
-}
-
-}
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_device_pulse_linux.h b/trunk/src/modules/audio_device/main/source/linux/audio_device_pulse_linux.h
deleted file mode 100644
index 693a32b..0000000
--- a/trunk/src/modules/audio_device/main/source/linux/audio_device_pulse_linux.h
+++ /dev/null
@@ -1,385 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
-
-#include "audio_device_generic.h"
-#include "audio_mixer_manager_pulse_linux.h"
-#include "critical_section_wrapper.h"
-
-#include <pulse/pulseaudio.h>
-
-// Set this define to make the code behave like in GTalk/libjingle
-//#define WEBRTC_PA_GTALK
-
-// We define this flag if it's missing from our headers, because we want to be
-// able to compile against old headers but still use PA_STREAM_ADJUST_LATENCY
-// if run against a recent version of the library.
-#ifndef PA_STREAM_ADJUST_LATENCY
-#define PA_STREAM_ADJUST_LATENCY 0x2000U
-#endif
-#ifndef PA_STREAM_START_MUTED
-#define PA_STREAM_START_MUTED 0x1000U
-#endif
-
-// Set this constant to 0 to disable latency reading
-const WebRtc_UWord32 WEBRTC_PA_REPORT_LATENCY = 1;
-
-// Constants from implementation by Tristan Schmelcher [tschmelcher@google.com]
-
-// First PulseAudio protocol version that supports PA_STREAM_ADJUST_LATENCY.
-const WebRtc_UWord32 WEBRTC_PA_ADJUST_LATENCY_PROTOCOL_VERSION = 13;
-
-// Some timing constants for optimal operation. See
-// https://tango.0pointer.de/pipermail/pulseaudio-discuss/2008-January/001170.html
-// for a good explanation of some of the factors that go into this.
-
-// Playback.
-
-// For playback, there is a round-trip delay to fill the server-side playback
-// buffer, so setting too low of a latency is a buffer underflow risk. We will
-// automatically increase the latency if a buffer underflow does occur, but we
-// also enforce a sane minimum at start-up time. Anything lower would be
-// virtually guaranteed to underflow at least once, so there's no point in
-// allowing lower latencies.
-const WebRtc_UWord32 WEBRTC_PA_PLAYBACK_LATENCY_MINIMUM_MSECS = 20;
-
-// Every time a playback stream underflows, we will reconfigure it with target
-// latency that is greater by this amount.
-const WebRtc_UWord32 WEBRTC_PA_PLAYBACK_LATENCY_INCREMENT_MSECS = 20;
-
-// We also need to configure a suitable request size. Too small and we'd burn
-// CPU from the overhead of transfering small amounts of data at once. Too large
-// and the amount of data remaining in the buffer right before refilling it
-// would be a buffer underflow risk. We set it to half of the buffer size.
-const WebRtc_UWord32 WEBRTC_PA_PLAYBACK_REQUEST_FACTOR = 2;
-
-// Capture.
-
-// For capture, low latency is not a buffer overflow risk, but it makes us burn
-// CPU from the overhead of transfering small amounts of data at once, so we set
-// a recommended value that we use for the kLowLatency constant (but if the user
-// explicitly requests something lower then we will honour it).
-// 1ms takes about 6-7% CPU. 5ms takes about 5%. 10ms takes about 4.x%.
-const WebRtc_UWord32 WEBRTC_PA_LOW_CAPTURE_LATENCY_MSECS = 10;
-
-// There is a round-trip delay to ack the data to the server, so the
-// server-side buffer needs extra space to prevent buffer overflow. 20ms is
-// sufficient, but there is no penalty to making it bigger, so we make it huge.
-// (750ms is libpulse's default value for the _total_ buffer size in the
-// kNoLatencyRequirements case.)
-const WebRtc_UWord32 WEBRTC_PA_CAPTURE_BUFFER_EXTRA_MSECS = 750;
-
-const WebRtc_UWord32 WEBRTC_PA_MSECS_PER_SEC = 1000;
-
-// Init _configuredLatencyRec/Play to this value to disable latency requirements
-const WebRtc_Word32 WEBRTC_PA_NO_LATENCY_REQUIREMENTS = -1;
-
-// Set this const to 1 to account for peeked and used data in latency calculation
-const WebRtc_UWord32 WEBRTC_PA_CAPTURE_BUFFER_LATENCY_ADJUSTMENT = 0;
-
-namespace webrtc
-{
-class EventWrapper;
-class ThreadWrapper;
-
-class AudioDeviceLinuxPulse: public AudioDeviceGeneric
-{
-public:
- AudioDeviceLinuxPulse(const WebRtc_Word32 id);
- ~AudioDeviceLinuxPulse();
-
- static bool PulseAudioIsSupported();
-
- // Retrieve the currently utilized audio layer
- virtual WebRtc_Word32
- ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
-
- // Main initializaton and termination
- virtual WebRtc_Word32 Init();
- virtual WebRtc_Word32 Terminate();
- virtual bool Initialized() const;
-
- // Device enumeration
- virtual WebRtc_Word16 PlayoutDevices();
- virtual WebRtc_Word16 RecordingDevices();
- virtual WebRtc_Word32 PlayoutDeviceName(
- WebRtc_UWord16 index,
- WebRtc_Word8 name[kAdmMaxDeviceNameSize],
- WebRtc_Word8 guid[kAdmMaxGuidSize]);
- virtual WebRtc_Word32 RecordingDeviceName(
- WebRtc_UWord16 index,
- WebRtc_Word8 name[kAdmMaxDeviceNameSize],
- WebRtc_Word8 guid[kAdmMaxGuidSize]);
-
- // Device selection
- virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32 SetPlayoutDevice(
- AudioDeviceModule::WindowsDeviceType device);
- virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32 SetRecordingDevice(
- AudioDeviceModule::WindowsDeviceType device);
-
- // Audio transport initialization
- virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
- virtual WebRtc_Word32 InitPlayout();
- virtual bool PlayoutIsInitialized() const;
- virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
- virtual WebRtc_Word32 InitRecording();
- virtual bool RecordingIsInitialized() const;
-
- // Audio transport control
- virtual WebRtc_Word32 StartPlayout();
- virtual WebRtc_Word32 StopPlayout();
- virtual bool Playing() const;
- virtual WebRtc_Word32 StartRecording();
- virtual WebRtc_Word32 StopRecording();
- virtual bool Recording() const;
-
- // Microphone Automatic Gain Control (AGC)
- virtual WebRtc_Word32 SetAGC(bool enable);
- virtual bool AGC() const;
-
- // Volume control based on the Windows Wave API (Windows only)
- virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
- WebRtc_UWord16 volumeRight);
- virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft,
- WebRtc_UWord16& volumeRight) const;
-
- // Audio mixer initialization
- virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
- virtual WebRtc_Word32 InitSpeaker();
- virtual bool SpeakerIsInitialized() const;
- virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
- virtual WebRtc_Word32 InitMicrophone();
- virtual bool MicrophoneIsInitialized() const;
-
- // Speaker volume controls
- virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
- virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
- virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
- virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
- virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
-
- // Microphone volume controls
- virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
- virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
- virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
- virtual WebRtc_Word32 MicrophoneVolumeStepSize(
- WebRtc_UWord16& stepSize) const;
-
- // Speaker mute control
- virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
- virtual WebRtc_Word32 SetSpeakerMute(bool enable);
- virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
-
- // Microphone mute control
- virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
- virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
-
- // Microphone boost control
- virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
- virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
-
- // Stereo support
- virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
- virtual WebRtc_Word32 SetStereoPlayout(bool enable);
- virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
- virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
- virtual WebRtc_Word32 SetStereoRecording(bool enable);
- virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
-
- // Delay information and control
- virtual WebRtc_Word32
- SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
- WebRtc_UWord16 sizeMS);
- virtual WebRtc_Word32 PlayoutBuffer(AudioDeviceModule::BufferType& type,
- WebRtc_UWord16& sizeMS) const;
- virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
- virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
-
- // CPU load
- virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
-
-public:
- virtual bool PlayoutWarning() const;
- virtual bool PlayoutError() const;
- virtual bool RecordingWarning() const;
- virtual bool RecordingError() const;
- virtual void ClearPlayoutWarning();
- virtual void ClearPlayoutError();
- virtual void ClearRecordingWarning();
- virtual void ClearRecordingError();
-
-public:
- virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
-private:
- void Lock()
- {
- _critSect.Enter();
- }
- ;
- void UnLock()
- {
- _critSect.Leave();
- }
- ;
- void WaitForOperationCompletion(pa_operation* paOperation) const;
- void WaitForSuccess(pa_operation* paOperation) const;
-
-private:
- static void PaContextStateCallback(pa_context *c, void *pThis);
- static void PaSinkInfoCallback(pa_context *c, const pa_sink_info *i,
- int eol, void *pThis);
- static void PaSourceInfoCallback(pa_context *c, const pa_source_info *i,
- int eol, void *pThis);
- static void PaServerInfoCallback(pa_context *c, const pa_server_info *i,
- void *pThis);
- static void PaStreamStateCallback(pa_stream *p, void *pThis);
- void PaContextStateCallbackHandler(pa_context *c);
- void PaSinkInfoCallbackHandler(const pa_sink_info *i, int eol);
- void PaSourceInfoCallbackHandler(const pa_source_info *i, int eol);
- void PaServerInfoCallbackHandler(const pa_server_info *i);
- void PaStreamStateCallbackHandler(pa_stream *p);
-
- void EnableWriteCallback();
- void DisableWriteCallback();
- static void PaStreamWriteCallback(pa_stream *unused, size_t buffer_space,
- void *pThis);
- void PaStreamWriteCallbackHandler(size_t buffer_space);
- static void PaStreamUnderflowCallback(pa_stream *unused, void *pThis);
- void PaStreamUnderflowCallbackHandler();
- void EnableReadCallback();
- void DisableReadCallback();
- static void PaStreamReadCallback(pa_stream *unused1, size_t unused2,
- void *pThis);
- void PaStreamReadCallbackHandler();
- static void PaStreamOverflowCallback(pa_stream *unused, void *pThis);
- void PaStreamOverflowCallbackHandler();
- WebRtc_Word32 LatencyUsecs(pa_stream *stream);
- WebRtc_Word32 ReadRecordedData(const void* bufferData, size_t bufferSize);
- WebRtc_Word32 ProcessRecordedData(WebRtc_Word8 *bufferData,
- WebRtc_UWord32 bufferSizeInSamples,
- WebRtc_UWord32 recDelay);
-
- WebRtc_Word32 CheckPulseAudioVersion();
- WebRtc_Word32 InitSamplingFrequency();
- WebRtc_Word32 GetDefaultDeviceInfo(bool recDevice, WebRtc_Word8* name,
- WebRtc_UWord16& index);
- WebRtc_Word32 InitPulseAudio();
- WebRtc_Word32 TerminatePulseAudio();
-
- void PaLock();
- void PaUnLock();
-
- static bool RecThreadFunc(void*);
- static bool PlayThreadFunc(void*);
- bool RecThreadProcess();
- bool PlayThreadProcess();
-
-private:
- AudioDeviceBuffer* _ptrAudioBuffer;
-
- CriticalSectionWrapper& _critSect;
- EventWrapper& _timeEventRec;
- EventWrapper& _timeEventPlay;
- EventWrapper& _recStartEvent;
- EventWrapper& _playStartEvent;
-
- ThreadWrapper* _ptrThreadPlay;
- ThreadWrapper* _ptrThreadRec;
- WebRtc_UWord32 _recThreadID;
- WebRtc_UWord32 _playThreadID;
- WebRtc_Word32 _id;
-
- AudioMixerManagerLinuxPulse _mixerManager;
-
- WebRtc_UWord16 _inputDeviceIndex;
- WebRtc_UWord16 _outputDeviceIndex;
- bool _inputDeviceIsSpecified;
- bool _outputDeviceIsSpecified;
-
- WebRtc_UWord32 _samplingFreq;
- WebRtc_UWord8 _recChannels;
- WebRtc_UWord8 _playChannels;
-
- AudioDeviceModule::BufferType _playBufType;
-
-private:
- bool _initialized;
- bool _recording;
- bool _playing;
- bool _recIsInitialized;
- bool _playIsInitialized;
- bool _startRec;
- bool _stopRec;
- bool _startPlay;
- bool _stopPlay;
- bool _AGC;
-
-private:
- WebRtc_UWord16 _playBufDelayFixed; // fixed playback delay
-
- WebRtc_UWord32 _sndCardPlayDelay;
- WebRtc_UWord32 _sndCardRecDelay;
-
- WebRtc_Word32 _writeErrors;
- WebRtc_UWord16 _playWarning;
- WebRtc_UWord16 _playError;
- WebRtc_UWord16 _recWarning;
- WebRtc_UWord16 _recError;
-
- WebRtc_UWord16 _deviceIndex;
- WebRtc_Word16 _numPlayDevices;
- WebRtc_Word16 _numRecDevices;
- WebRtc_Word8* _playDeviceName;
- WebRtc_Word8* _recDeviceName;
- WebRtc_Word8* _playDisplayDeviceName;
- WebRtc_Word8* _recDisplayDeviceName;
- WebRtc_Word8 _paServerVersion[32];
-
- WebRtc_Word8* _playBuffer;
- size_t _playbackBufferSize;
- size_t _playbackBufferUnused;
- size_t _tempBufferSpace;
- WebRtc_Word8* _recBuffer;
- size_t _recordBufferSize;
- size_t _recordBufferUsed;
- const void* _tempSampleData;
- size_t _tempSampleDataSize;
- WebRtc_Word32 _configuredLatencyPlay;
- WebRtc_Word32 _configuredLatencyRec;
-
- // PulseAudio
- WebRtc_UWord16 _paDeviceIndex;
- bool _paStateChanged;
-
- pa_threaded_mainloop* _paMainloop;
- pa_mainloop_api* _paMainloopApi;
- pa_context* _paContext;
-
- pa_stream* _recStream;
- pa_stream* _playStream;
- WebRtc_UWord32 _recStreamFlags;
- WebRtc_UWord32 _playStreamFlags;
- pa_buffer_attr _playBufferAttr;
- pa_buffer_attr _recBufferAttr;
-};
-
-}
-
-#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_PULSE_LINUX_H_
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_device_utility_linux.cc b/trunk/src/modules/audio_device/main/source/linux/audio_device_utility_linux.cc
deleted file mode 100644
index 0b9dd54..0000000
--- a/trunk/src/modules/audio_device/main/source/linux/audio_device_utility_linux.cc
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_device_utility_linux.h"
-#include "audio_device_config.h" // DEBUG_PRINT()
-#include "critical_section_wrapper.h"
-#include "trace.h"
-
-namespace webrtc
-{
-
-AudioDeviceUtilityLinux::AudioDeviceUtilityLinux(const WebRtc_Word32 id) :
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()), _id(id),
- _lastError(AudioDeviceModule::kAdmErrNone)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
- "%s created", __FUNCTION__);
-}
-
-AudioDeviceUtilityLinux::~AudioDeviceUtilityLinux()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
- "%s destroyed", __FUNCTION__);
- {
- CriticalSectionScoped lock(_critSect);
-
- // free stuff here...
- }
-
- delete &_critSect;
-}
-
-// ============================================================================
-// API
-// ============================================================================
-
-
-WebRtc_Word32 AudioDeviceUtilityLinux::Init()
-{
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
- " OS info: %s", "Linux");
-
- return 0;
-}
-
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_device_utility_linux.h b/trunk/src/modules/audio_device/main/source/linux/audio_device_utility_linux.h
deleted file mode 100644
index 8df7acc..0000000
--- a/trunk/src/modules/audio_device/main/source/linux/audio_device_utility_linux.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_LINUX_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_LINUX_H
-
-#include "audio_device_utility.h"
-#include "audio_device.h"
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-
-class AudioDeviceUtilityLinux: public AudioDeviceUtility
-{
-public:
- AudioDeviceUtilityLinux(const WebRtc_Word32 id);
- ~AudioDeviceUtilityLinux();
-
- virtual WebRtc_Word32 Init();
-
-private:
- CriticalSectionWrapper& _critSect;
- WebRtc_Word32 _id;
- AudioDeviceModule::ErrorCode _lastError;
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_DEVICE_UTILITY_LINUX_H_
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_alsa_linux.cc b/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_alsa_linux.cc
deleted file mode 100644
index 306ad5d..0000000
--- a/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_alsa_linux.cc
+++ /dev/null
@@ -1,1317 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cassert>
-
-#include "audio_mixer_manager_alsa_linux.h"
-#include "trace.h"
-
-extern webrtc_adm_linux_alsa::AlsaSymbolTable AlsaSymbolTable;
-
-// Accesses ALSA functions through our late-binding symbol table instead of
-// directly. This way we don't have to link to libalsa, which means our binary
-// will work on systems that don't have it.
-#define LATE(sym) \
- LATESYM_GET(webrtc_adm_linux_alsa::AlsaSymbolTable, &AlsaSymbolTable, sym)
-
-namespace webrtc
-{
-
-AudioMixerManagerLinuxALSA::AudioMixerManagerLinuxALSA(const WebRtc_Word32 id) :
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _id(id),
- _outputMixerHandle(NULL),
- _inputMixerHandle(NULL),
- _outputMixerElement(NULL),
- _inputMixerElement(NULL)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
- "%s constructed", __FUNCTION__);
-
- memset(_outputMixerStr, 0, kAdmMaxDeviceNameSize);
- memset(_inputMixerStr, 0, kAdmMaxDeviceNameSize);
-}
-
-AudioMixerManagerLinuxALSA::~AudioMixerManagerLinuxALSA()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
- "%s destructed", __FUNCTION__);
-
- Close();
-
- delete &_critSect;
-}
-
-// ============================================================================
-// PUBLIC METHODS
-// ============================================================================
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::Close()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
- __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- CloseSpeaker();
- CloseMicrophone();
-
- return 0;
-
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::CloseSpeaker()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
- __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- int errVal = 0;
-
- if (_outputMixerHandle != NULL)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Closing playout mixer");
- LATE(snd_mixer_free)(_outputMixerHandle);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error freeing playout mixer: %s",
- LATE(snd_strerror)(errVal));
- }
- errVal = LATE(snd_mixer_detach)(_outputMixerHandle, _outputMixerStr);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error detachinging playout mixer: %s",
- LATE(snd_strerror)(errVal));
- }
- errVal = LATE(snd_mixer_close)(_outputMixerHandle);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error snd_mixer_close(handleMixer) errVal=%d",
- errVal);
- }
- _outputMixerHandle = NULL;
- _outputMixerElement = NULL;
- }
- memset(_outputMixerStr, 0, kAdmMaxDeviceNameSize);
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::CloseMicrophone()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- int errVal = 0;
-
- if (_inputMixerHandle != NULL)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Closing record mixer");
-
- LATE(snd_mixer_free)(_inputMixerHandle);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error freeing record mixer: %s",
- LATE(snd_strerror)(errVal));
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Closing record mixer 2");
-
- errVal = LATE(snd_mixer_detach)(_inputMixerHandle, _inputMixerStr);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error detachinging record mixer: %s",
- LATE(snd_strerror)(errVal));
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Closing record mixer 3");
-
- errVal = LATE(snd_mixer_close)(_inputMixerHandle);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error snd_mixer_close(handleMixer) errVal=%d",
- errVal);
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Closing record mixer 4");
- _inputMixerHandle = NULL;
- _inputMixerElement = NULL;
- }
- memset(_inputMixerStr, 0, kAdmMaxDeviceNameSize);
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::OpenSpeaker(char* deviceName)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerLinuxALSA::OpenSpeaker(name=%s)", deviceName);
-
- CriticalSectionScoped lock(_critSect);
-
- int errVal = 0;
-
- // Close any existing output mixer handle
- //
- if (_outputMixerHandle != NULL)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Closing playout mixer");
-
- LATE(snd_mixer_free)(_outputMixerHandle);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error freeing playout mixer: %s",
- LATE(snd_strerror)(errVal));
- }
- errVal = LATE(snd_mixer_detach)(_outputMixerHandle, _outputMixerStr);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error detachinging playout mixer: %s",
- LATE(snd_strerror)(errVal));
- }
- errVal = LATE(snd_mixer_close)(_outputMixerHandle);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error snd_mixer_close(handleMixer) errVal=%d",
- errVal);
- }
- }
- _outputMixerHandle = NULL;
- _outputMixerElement = NULL;
-
- errVal = LATE(snd_mixer_open)(&_outputMixerHandle, 0);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "snd_mixer_open(&_outputMixerHandle, 0) - error");
- return -1;
- }
-
- char controlName[kAdmMaxDeviceNameSize] = { 0 };
- GetControlName(controlName, deviceName);
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " snd_mixer_attach(_outputMixerHandle, %s)", controlName);
-
- errVal = LATE(snd_mixer_attach)(_outputMixerHandle, controlName);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " snd_mixer_attach(_outputMixerHandle, %s) error: %s",
- controlName, LATE(snd_strerror)(errVal));
- _outputMixerHandle = NULL;
- return -1;
- }
- strcpy(_outputMixerStr, controlName);
-
- errVal = LATE(snd_mixer_selem_register)(_outputMixerHandle, NULL, NULL);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " snd_mixer_selem_register(_outputMixerHandle,"
- " NULL, NULL), error: %s",
- LATE(snd_strerror)(errVal));
- _outputMixerHandle = NULL;
- return -1;
- }
-
- // Load and find the proper mixer element
- if (LoadSpeakerMixerElement() < 0)
- {
- return -1;
- }
-
- if (_outputMixerHandle != NULL)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " the output mixer device is now open (0x%x)",
- _outputMixerHandle);
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::OpenMicrophone(char *deviceName)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerLinuxALSA::OpenMicrophone(name=%s)",
- deviceName);
-
- CriticalSectionScoped lock(_critSect);
-
- int errVal = 0;
-
- // Close any existing input mixer handle
- //
- if (_inputMixerHandle != NULL)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Closing record mixer");
-
- LATE(snd_mixer_free)(_inputMixerHandle);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error freeing record mixer: %s",
- LATE(snd_strerror)(errVal));
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Closing record mixer");
-
- errVal = LATE(snd_mixer_detach)(_inputMixerHandle, _inputMixerStr);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error detachinging record mixer: %s",
- LATE(snd_strerror)(errVal));
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Closing record mixer");
-
- errVal = LATE(snd_mixer_close)(_inputMixerHandle);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error snd_mixer_close(handleMixer) errVal=%d",
- errVal);
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Closing record mixer");
- }
- _inputMixerHandle = NULL;
- _inputMixerElement = NULL;
-
- errVal = LATE(snd_mixer_open)(&_inputMixerHandle, 0);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " snd_mixer_open(&_inputMixerHandle, 0) - error");
- return -1;
- }
-
- char controlName[kAdmMaxDeviceNameSize] = { 0 };
- GetControlName(controlName, deviceName);
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " snd_mixer_attach(_inputMixerHandle, %s)", controlName);
-
- errVal = LATE(snd_mixer_attach)(_inputMixerHandle, controlName);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " snd_mixer_attach(_inputMixerHandle, %s) error: %s",
- controlName, LATE(snd_strerror)(errVal));
-
- _inputMixerHandle = NULL;
- return -1;
- }
- strcpy(_inputMixerStr, controlName);
-
- errVal = LATE(snd_mixer_selem_register)(_inputMixerHandle, NULL, NULL);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " snd_mixer_selem_register(_inputMixerHandle,"
- " NULL, NULL), error: %s",
- LATE(snd_strerror)(errVal));
-
- _inputMixerHandle = NULL;
- return -1;
- }
- // Load and find the proper mixer element
- if (LoadMicMixerElement() < 0)
- {
- return -1;
- }
-
- if (_inputMixerHandle != NULL)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " the input mixer device is now open (0x%x)",
- _inputMixerHandle);
- }
-
- return 0;
-}
-
-bool AudioMixerManagerLinuxALSA::SpeakerIsInitialized() const
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- return (_outputMixerHandle != NULL);
-}
-
-bool AudioMixerManagerLinuxALSA::MicrophoneIsInitialized() const
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s",
- __FUNCTION__);
-
- return (_inputMixerHandle != NULL);
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::SetSpeakerVolume(
- WebRtc_UWord32 volume)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerLinuxALSA::SetSpeakerVolume(volume=%u)",
- volume);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_outputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable output mixer element exists");
- return -1;
- }
-
- int errVal =
- LATE(snd_mixer_selem_set_playback_volume_all)(_outputMixerElement,
- volume);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error changing master volume: %s",
- LATE(snd_strerror)(errVal));
- return -1;
- }
-
- return (0);
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::SpeakerVolume(
- WebRtc_UWord32& volume) const
-{
-
- if (_outputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable output mixer element exists");
- return -1;
- }
-
- long int vol(0);
-
- int
- errVal = LATE(snd_mixer_selem_get_playback_volume)(
- _outputMixerElement,
- (snd_mixer_selem_channel_id_t) 0,
- &vol);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Error getting outputvolume: %s",
- LATE(snd_strerror)(errVal));
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " AudioMixerManagerLinuxALSA::SpeakerVolume() => vol=%i",
- vol);
-
- volume = static_cast<WebRtc_UWord32> (vol);
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::MaxSpeakerVolume(
- WebRtc_UWord32& maxVolume) const
-{
-
- if (_outputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avilable output mixer element exists");
- return -1;
- }
-
- long int minVol(0);
- long int maxVol(0);
-
- int errVal =
- LATE(snd_mixer_selem_get_playback_volume_range)(_outputMixerElement,
- &minVol, &maxVol);
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Playout hardware volume range, min: %d, max: %d",
- minVol, maxVol);
-
- if (maxVol <= minVol)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error getting get_playback_volume_range: %s",
- LATE(snd_strerror)(errVal));
- }
-
- maxVolume = static_cast<WebRtc_UWord32> (maxVol);
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::MinSpeakerVolume(
- WebRtc_UWord32& minVolume) const
-{
-
- if (_outputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable output mixer element exists");
- return -1;
- }
-
- long int minVol(0);
- long int maxVol(0);
-
- int errVal =
- LATE(snd_mixer_selem_get_playback_volume_range)(_outputMixerElement,
- &minVol, &maxVol);
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Playout hardware volume range, min: %d, max: %d",
- minVol, maxVol);
-
- if (maxVol <= minVol)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error getting get_playback_volume_range: %s",
- LATE(snd_strerror)(errVal));
- }
-
- minVolume = static_cast<WebRtc_UWord32> (minVol);
-
- return 0;
-}
-
-// TL: Have done testnig with these but they don't seem reliable and
-// they were therefore not added
-/*
- // ----------------------------------------------------------------------------
- // SetMaxSpeakerVolume
- // ----------------------------------------------------------------------------
-
- WebRtc_Word32 AudioMixerManagerLinuxALSA::SetMaxSpeakerVolume(
- WebRtc_UWord32 maxVolume)
- {
-
- if (_outputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable output mixer element exists");
- return -1;
- }
-
- long int minVol(0);
- long int maxVol(0);
-
- int errVal = snd_mixer_selem_get_playback_volume_range(
- _outputMixerElement, &minVol, &maxVol);
- if ((maxVol <= minVol) || (errVal != 0))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Error getting playback volume range: %s", snd_strerror(errVal));
- }
-
- maxVol = maxVolume;
- errVal = snd_mixer_selem_set_playback_volume_range(
- _outputMixerElement, minVol, maxVol);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Playout hardware volume range, min: %d, max: %d", minVol, maxVol);
- if (errVal != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error setting playback volume range: %s", snd_strerror(errVal));
- return -1;
- }
-
- return 0;
- }
-
- // ----------------------------------------------------------------------------
- // SetMinSpeakerVolume
- // ----------------------------------------------------------------------------
-
- WebRtc_Word32 AudioMixerManagerLinuxALSA::SetMinSpeakerVolume(
- WebRtc_UWord32 minVolume)
- {
-
- if (_outputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable output mixer element exists");
- return -1;
- }
-
- long int minVol(0);
- long int maxVol(0);
-
- int errVal = snd_mixer_selem_get_playback_volume_range(
- _outputMixerElement, &minVol, &maxVol);
- if ((maxVol <= minVol) || (errVal != 0))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Error getting playback volume range: %s", snd_strerror(errVal));
- }
-
- minVol = minVolume;
- errVal = snd_mixer_selem_set_playback_volume_range(
- _outputMixerElement, minVol, maxVol);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Playout hardware volume range, min: %d, max: %d", minVol, maxVol);
- if (errVal != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error setting playback volume range: %s", snd_strerror(errVal));
- return -1;
- }
-
- return 0;
- }
- */
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::SpeakerVolumeStepSize(
- WebRtc_UWord16& stepSize) const
-{
-
- if (_outputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable output mixer exists");
- return -1;
- }
-
- // The step size is always 1 for ALSA
- stepSize = 1;
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::SpeakerVolumeIsAvailable(
- bool& available)
-{
- if (_outputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable output mixer element exists");
- return -1;
- }
-
- available = LATE(snd_mixer_selem_has_playback_volume)(_outputMixerElement);
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::SpeakerMuteIsAvailable(
- bool& available)
-{
- if (_outputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable output mixer element exists");
- return -1;
- }
-
- available = LATE(snd_mixer_selem_has_playback_switch)(_outputMixerElement);
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::SetSpeakerMute(bool enable)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerLinuxALSA::SetSpeakerMute(enable=%u)",
- enable);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_outputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable output mixer element exists");
- return -1;
- }
-
- // Ensure that the selected speaker destination has a valid mute control.
- bool available(false);
- SpeakerMuteIsAvailable(available);
- if (!available)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " it is not possible to mute the speaker");
- return -1;
- }
-
- // Note value = 0 (off) means muted
- int errVal =
- LATE(snd_mixer_selem_set_playback_switch_all)(_outputMixerElement,
- !enable);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error setting playback switch: %s",
- LATE(snd_strerror)(errVal));
- return -1;
- }
-
- return (0);
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::SpeakerMute(bool& enabled) const
-{
-
- if (_outputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable output mixer exists");
- return -1;
- }
-
- // Ensure that the selected speaker destination has a valid mute control.
- bool available =
- LATE(snd_mixer_selem_has_playback_switch)(_outputMixerElement);
- if (!available)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " it is not possible to mute the speaker");
- return -1;
- }
-
- int value(false);
-
- // Retrieve one boolean control value for a specified mute-control
- //
- int
- errVal = LATE(snd_mixer_selem_get_playback_switch)(
- _outputMixerElement,
- (snd_mixer_selem_channel_id_t) 0,
- &value);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error getting playback switch: %s",
- LATE(snd_strerror)(errVal));
- return -1;
- }
-
- // Note value = 0 (off) means muted
- enabled = (bool) !value;
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::MicrophoneMuteIsAvailable(
- bool& available)
-{
- if (_inputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable input mixer element exists");
- return -1;
- }
-
- available = LATE(snd_mixer_selem_has_capture_switch)(_inputMixerElement);
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::SetMicrophoneMute(bool enable)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerLinuxALSA::SetMicrophoneMute(enable=%u)",
- enable);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_inputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable input mixer element exists");
- return -1;
- }
-
- // Ensure that the selected microphone destination has a valid mute control.
- bool available(false);
- MicrophoneMuteIsAvailable(available);
- if (!available)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " it is not possible to mute the microphone");
- return -1;
- }
-
- // Note value = 0 (off) means muted
- int errVal =
- LATE(snd_mixer_selem_set_capture_switch_all)(_inputMixerElement,
- !enable);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error setting capture switch: %s",
- LATE(snd_strerror)(errVal));
- return -1;
- }
-
- return (0);
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::MicrophoneMute(bool& enabled) const
-{
-
- if (_inputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable input mixer exists");
- return -1;
- }
-
- // Ensure that the selected microphone destination has a valid mute control.
- bool available =
- LATE(snd_mixer_selem_has_capture_switch)(_inputMixerElement);
- if (!available)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " it is not possible to mute the microphone");
- return -1;
- }
-
- int value(false);
-
- // Retrieve one boolean control value for a specified mute-control
- //
- int
- errVal = LATE(snd_mixer_selem_get_capture_switch)(
- _inputMixerElement,
- (snd_mixer_selem_channel_id_t) 0,
- &value);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error getting capture switch: %s",
- LATE(snd_strerror)(errVal));
- return -1;
- }
-
- // Note value = 0 (off) means muted
- enabled = (bool) !value;
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::MicrophoneBoostIsAvailable(
- bool& available)
-{
- if (_inputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable input mixer exists");
- return -1;
- }
-
- // Microphone boost cannot be enabled through ALSA Simple Mixer Interface
- available = false;
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::SetMicrophoneBoost(bool enable)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerLinuxALSA::SetMicrophoneBoost(enable=%u)",
- enable);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_inputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable input mixer exists");
- return -1;
- }
-
- // Ensure that the selected microphone destination has a valid mute control.
- bool available(false);
- MicrophoneMuteIsAvailable(available);
- if (!available)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " it is not possible to enable microphone boost");
- return -1;
- }
-
- // It is assumed that the call above fails!
-
- return (0);
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::MicrophoneBoost(bool& enabled) const
-{
-
- if (_inputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable input mixer exists");
- return -1;
- }
-
- // Microphone boost cannot be enabled on this platform!
- enabled = false;
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::MicrophoneVolumeIsAvailable(
- bool& available)
-{
- if (_inputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable input mixer element exists");
- return -1;
- }
-
- available = LATE(snd_mixer_selem_has_capture_volume)(_inputMixerElement);
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::SetMicrophoneVolume(
- WebRtc_UWord32 volume)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerLinuxALSA::SetMicrophoneVolume(volume=%u)",
- volume);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_inputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable input mixer element exists");
- return -1;
- }
-
- int
- errVal =
- LATE(snd_mixer_selem_set_capture_volume_all)(_inputMixerElement,
- volume);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error changing microphone volume: %s",
- LATE(snd_strerror)(errVal));
- return -1;
- }
-
- return (0);
-}
-
-// TL: Have done testnig with these but they don't seem reliable and
-// they were therefore not added
-/*
- // ----------------------------------------------------------------------------
- // SetMaxMicrophoneVolume
- // ----------------------------------------------------------------------------
-
- WebRtc_Word32 AudioMixerManagerLinuxALSA::SetMaxMicrophoneVolume(
- WebRtc_UWord32 maxVolume)
- {
-
- if (_inputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable output mixer element exists");
- return -1;
- }
-
- long int minVol(0);
- long int maxVol(0);
-
- int errVal = snd_mixer_selem_get_capture_volume_range(_inputMixerElement,
- &minVol, &maxVol);
- if ((maxVol <= minVol) || (errVal != 0))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Error getting capture volume range: %s", snd_strerror(errVal));
- }
-
- maxVol = (long int)maxVolume;
- printf("min %d max %d", minVol, maxVol);
- errVal = snd_mixer_selem_set_capture_volume_range(_inputMixerElement, minVol, maxVol);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Capture hardware volume range, min: %d, max: %d", minVol, maxVol);
- if (errVal != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error setting capture volume range: %s", snd_strerror(errVal));
- return -1;
- }
-
- return 0;
- }
-
- // ----------------------------------------------------------------------------
- // SetMinMicrophoneVolume
- // ----------------------------------------------------------------------------
-
- WebRtc_Word32 AudioMixerManagerLinuxALSA::SetMinMicrophoneVolume(
- WebRtc_UWord32 minVolume)
- {
-
- if (_inputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable output mixer element exists");
- return -1;
- }
-
- long int minVol(0);
- long int maxVol(0);
-
- int errVal = snd_mixer_selem_get_capture_volume_range(
- _inputMixerElement, &minVol, &maxVol);
- if (maxVol <= minVol)
- {
- //maxVol = 255;
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Error getting capture volume range: %s", snd_strerror(errVal));
- }
-
- printf("min %d max %d", minVol, maxVol);
- minVol = (long int)minVolume;
- errVal = snd_mixer_selem_set_capture_volume_range(
- _inputMixerElement, minVol, maxVol);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Capture hardware volume range, min: %d, max: %d", minVol, maxVol);
- if (errVal != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error setting capture volume range: %s", snd_strerror(errVal));
- return -1;
- }
-
- return 0;
- }
- */
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::MicrophoneVolume(
- WebRtc_UWord32& volume) const
-{
-
- if (_inputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable input mixer element exists");
- return -1;
- }
-
- long int vol(0);
-
- int
- errVal =
- LATE(snd_mixer_selem_get_capture_volume)(
- _inputMixerElement,
- (snd_mixer_selem_channel_id_t) 0,
- &vol);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Error getting inputvolume: %s",
- LATE(snd_strerror)(errVal));
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " AudioMixerManagerLinuxALSA::MicrophoneVolume() => vol=%i",
- vol);
-
- volume = static_cast<WebRtc_UWord32> (vol);
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::MaxMicrophoneVolume(
- WebRtc_UWord32& maxVolume) const
-{
-
- if (_inputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable input mixer element exists");
- return -1;
- }
-
- long int minVol(0);
- long int maxVol(0);
-
- // check if we have mic volume at all
- if (!LATE(snd_mixer_selem_has_capture_volume)(_inputMixerElement))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " No microphone volume available");
- return -1;
- }
-
- int errVal =
- LATE(snd_mixer_selem_get_capture_volume_range)(_inputMixerElement,
- &minVol, &maxVol);
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Microphone hardware volume range, min: %d, max: %d",
- minVol, maxVol);
- if (maxVol <= minVol)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error getting microphone volume range: %s",
- LATE(snd_strerror)(errVal));
- }
-
- maxVolume = static_cast<WebRtc_UWord32> (maxVol);
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::MinMicrophoneVolume(
- WebRtc_UWord32& minVolume) const
-{
-
- if (_inputMixerElement == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable input mixer element exists");
- return -1;
- }
-
- long int minVol(0);
- long int maxVol(0);
-
- int errVal =
- LATE(snd_mixer_selem_get_capture_volume_range)(_inputMixerElement,
- &minVol, &maxVol);
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Microphone hardware volume range, min: %d, max: %d",
- minVol, maxVol);
- if (maxVol <= minVol)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error getting microphone volume range: %s",
- LATE(snd_strerror)(errVal));
- }
-
- minVolume = static_cast<WebRtc_UWord32> (minVol);
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::MicrophoneVolumeStepSize(
- WebRtc_UWord16& stepSize) const
-{
-
- if (_inputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " no avaliable input mixer exists");
- return -1;
- }
-
- // The step size is always 1 for ALSA
- stepSize = 1;
-
- return 0;
-}
-
-// ============================================================================
-// Private Methods
-// ============================================================================
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::LoadMicMixerElement() const
-{
- int errVal = LATE(snd_mixer_load)(_inputMixerHandle);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "snd_mixer_load(_inputMixerHandle), error: %s",
- LATE(snd_strerror)(errVal));
- _inputMixerHandle = NULL;
- return -1;
- }
-
- snd_mixer_elem_t *elem = NULL;
- snd_mixer_elem_t *micElem = NULL;
- unsigned mixerIdx = 0;
- const char *selemName = NULL;
-
- // Find and store handles to the right mixer elements
- for (elem = LATE(snd_mixer_first_elem)(_inputMixerHandle); elem; elem
- = LATE(snd_mixer_elem_next)(elem), mixerIdx++)
- {
- if (LATE(snd_mixer_selem_is_active)(elem))
- {
- selemName = LATE(snd_mixer_selem_get_name)(elem);
- if (strcmp(selemName, "Capture") == 0) // "Capture", "Mic"
- {
- _inputMixerElement = elem;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
- _id, " Capture element set");
- } else if (strcmp(selemName, "Mic") == 0)
- {
- micElem = elem;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
- _id, " Mic element found");
- }
- }
-
- if (_inputMixerElement)
- {
- // Use the first Capture element that is found
- // The second one may not work
- break;
- }
- }
-
- if (_inputMixerElement == NULL)
- {
- // We didn't find a Capture handle, use Mic.
- if (micElem != NULL)
- {
- _inputMixerElement = micElem;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Using Mic as capture volume.");
- } else
- {
- _inputMixerElement = NULL;
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Could not find capture volume on the mixer.");
-
- return -1;
- }
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxALSA::LoadSpeakerMixerElement() const
-{
- int errVal = LATE(snd_mixer_load)(_outputMixerHandle);
- if (errVal < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " snd_mixer_load(_outputMixerHandle), error: %s",
- LATE(snd_strerror)(errVal));
- _outputMixerHandle = NULL;
- return -1;
- }
-
- snd_mixer_elem_t *elem = NULL;
- snd_mixer_elem_t *masterElem = NULL;
- snd_mixer_elem_t *speakerElem = NULL;
- unsigned mixerIdx = 0;
- const char *selemName = NULL;
-
- // Find and store handles to the right mixer elements
- for (elem = LATE(snd_mixer_first_elem)(_outputMixerHandle); elem; elem
- = LATE(snd_mixer_elem_next)(elem), mixerIdx++)
- {
- if (LATE(snd_mixer_selem_is_active)(elem))
- {
- selemName = LATE(snd_mixer_selem_get_name)(elem);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "snd_mixer_selem_get_name %d: %s =%x", mixerIdx,
- selemName, elem);
-
- // "Master", "PCM", "Wave", "Master Mono", "PC Speaker", "PCM", "Wave"
- if (strcmp(selemName, "PCM") == 0)
- {
- _outputMixerElement = elem;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
- _id, " PCM element set");
- } else if (strcmp(selemName, "Master") == 0)
- {
- masterElem = elem;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
- _id, " Master element found");
- } else if (strcmp(selemName, "Speaker") == 0)
- {
- speakerElem = elem;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
- _id, " Speaker element found");
- }
- }
-
- if (_outputMixerElement)
- {
- // We have found the element we want
- break;
- }
- }
-
- // If we didn't find a PCM Handle, use Master or Speaker
- if (_outputMixerElement == NULL)
- {
- if (masterElem != NULL)
- {
- _outputMixerElement = masterElem;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Using Master as output volume.");
- } else if (speakerElem != NULL)
- {
- _outputMixerElement = speakerElem;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Using Speaker as output volume.");
- } else
- {
- _outputMixerElement = NULL;
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Could not find output volume in the mixer.");
- return -1;
- }
- }
-
- return 0;
-}
-
-void AudioMixerManagerLinuxALSA::GetControlName(char* controlName,
- char* deviceName) const
-{
- // Example
- // deviceName: "front:CARD=Intel,DEV=0"
- // controlName: "hw:CARD=Intel"
- char* pos1 = strchr(deviceName, ':');
- char* pos2 = strchr(deviceName, ',');
- if (!pos2)
- {
- // Can also be default:CARD=Intel
- pos2 = &deviceName[strlen(deviceName)];
- }
- if (pos1 && pos2)
- {
- strcpy(controlName, "hw");
- int nChar = (int) (pos2 - pos1);
- strncpy(&controlName[2], pos1, nChar);
- controlName[2 + nChar] = '\0';
- } else
- {
- strcpy(controlName, deviceName);
- }
-
-}
-
-}
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_alsa_linux.h b/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_alsa_linux.h
deleted file mode 100644
index 94ea982..0000000
--- a/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_alsa_linux.h
+++ /dev/null
@@ -1,78 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_ALSA_LINUX_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_ALSA_LINUX_H
-
-#include "typedefs.h"
-#include "audio_device.h"
-#include "critical_section_wrapper.h"
-#include "alsasymboltable_linux.h"
-
-#include <alsa/asoundlib.h>
-
-namespace webrtc
-{
-
-class AudioMixerManagerLinuxALSA
-{
-public:
- WebRtc_Word32 OpenSpeaker(char* deviceName);
- WebRtc_Word32 OpenMicrophone(char* deviceName);
- WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
- WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
- WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
- WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
- WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
- WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
- WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
- WebRtc_Word32 SetSpeakerMute(bool enable);
- WebRtc_Word32 SpeakerMute(bool& enabled) const;
- WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
- WebRtc_Word32 SetMicrophoneMute(bool enable);
- WebRtc_Word32 MicrophoneMute(bool& enabled) const;
- WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
- WebRtc_Word32 SetMicrophoneBoost(bool enable);
- WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
- WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
- WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
- WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
- WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
- WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
- WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
- WebRtc_Word32 Close();
- WebRtc_Word32 CloseSpeaker();
- WebRtc_Word32 CloseMicrophone();
- bool SpeakerIsInitialized() const;
- bool MicrophoneIsInitialized() const;
-
-public:
- AudioMixerManagerLinuxALSA(const WebRtc_Word32 id);
- ~AudioMixerManagerLinuxALSA();
-
-private:
- WebRtc_Word32 LoadMicMixerElement() const;
- WebRtc_Word32 LoadSpeakerMixerElement() const;
- void GetControlName(char *controlName, char* deviceName) const;
-
-private:
- CriticalSectionWrapper& _critSect;
- WebRtc_Word32 _id;
- mutable snd_mixer_t* _outputMixerHandle;
- char _outputMixerStr[kAdmMaxDeviceNameSize];
- mutable snd_mixer_t* _inputMixerHandle;
- char _inputMixerStr[kAdmMaxDeviceNameSize];
- mutable snd_mixer_elem_t* _outputMixerElement;
- mutable snd_mixer_elem_t* _inputMixerElement;
-};
-
-}
-
-#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_MIXER_MANAGER_ALSA_LINUX_H_
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_pulse_linux.cc b/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_pulse_linux.cc
deleted file mode 100644
index 01f5172..0000000
--- a/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_pulse_linux.cc
+++ /dev/null
@@ -1,1271 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cassert>
-
-#include "audio_mixer_manager_pulse_linux.h"
-#include "trace.h"
-
-extern webrtc_adm_linux_pulse::PulseAudioSymbolTable PaSymbolTable;
-
-// Accesses Pulse functions through our late-binding symbol table instead of
-// directly. This way we don't have to link to libpulse, which means our binary
-// will work on systems that don't have it.
-#define LATE(sym) \
- LATESYM_GET(webrtc_adm_linux_pulse::PulseAudioSymbolTable, &PaSymbolTable, sym)
-
-namespace webrtc
-{
-
-enum { kMaxRetryOnFailure = 2 };
-
-AudioMixerManagerLinuxPulse::AudioMixerManagerLinuxPulse(const WebRtc_Word32 id) :
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _id(id),
- _paOutputDeviceIndex(-1),
- _paInputDeviceIndex(-1),
- _paPlayStream(NULL),
- _paRecStream(NULL),
- _paMainloop(NULL),
- _paContext(NULL),
- _paVolume(0),
- _paMute(0),
- _paVolSteps(0),
- _paSpeakerMute(false),
- _paSpeakerVolume(0),
- _paChannels(0),
- _paObjectsSet(false),
- _callbackValues(false)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
- "%s constructed", __FUNCTION__);
-}
-
-AudioMixerManagerLinuxPulse::~AudioMixerManagerLinuxPulse()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
- "%s destructed", __FUNCTION__);
-
- Close();
-
- delete &_critSect;
-}
-
-// ============================================================================
-// PUBLIC METHODS
-// ============================================================================
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::SetPulseAudioObjects(
- pa_threaded_mainloop* mainloop,
- pa_context* context)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
- __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- if (!mainloop || !context)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " could not set PulseAudio objects for mixer");
- return -1;
- }
-
- _paMainloop = mainloop;
- _paContext = context;
- _paObjectsSet = true;
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " the PulseAudio objects for the mixer has been set");
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::Close()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
- __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- CloseSpeaker();
- CloseMicrophone();
-
- _paMainloop = NULL;
- _paContext = NULL;
- _paObjectsSet = false;
-
- return 0;
-
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::CloseSpeaker()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
- __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- // Reset the index to -1
- _paOutputDeviceIndex = -1;
- _paPlayStream = NULL;
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::CloseMicrophone()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
- __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- // Reset the index to -1
- _paInputDeviceIndex = -1;
- _paRecStream = NULL;
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::SetPlayStream(pa_stream* playStream)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerLinuxPulse::SetPlayStream(playStream)");
-
- CriticalSectionScoped lock(_critSect);
- _paPlayStream = playStream;
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::SetRecStream(pa_stream* recStream)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerLinuxPulse::SetRecStream(recStream)");
-
- CriticalSectionScoped lock(_critSect);
- _paRecStream = recStream;
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::OpenSpeaker(
- WebRtc_UWord16 deviceIndex)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerLinuxPulse::OpenSpeaker(deviceIndex=%d)",
- deviceIndex);
-
- CriticalSectionScoped lock(_critSect);
-
- // No point in opening the speaker
- // if PA objects have not been set
- if (!_paObjectsSet)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " PulseAudio objects has not been set");
- return -1;
- }
-
- // Set the index for the PulseAudio
- // output device to control
- _paOutputDeviceIndex = deviceIndex;
-
- // Init the speaker volume to the normal volume
- _paSpeakerVolume = PA_VOLUME_NORM;
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " the output mixer device is now open");
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::OpenMicrophone(
- WebRtc_UWord16 deviceIndex)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerLinuxPulse::OpenMicrophone(deviceIndex=%d)",
- deviceIndex);
-
- CriticalSectionScoped lock(_critSect);
-
- // No point in opening the microphone
- // if PA objects have not been set
- if (!_paObjectsSet)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " PulseAudio objects have not been set");
- return -1;
- }
-
- // Set the index for the PulseAudio
- // input device to control
- _paInputDeviceIndex = deviceIndex;
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " the input mixer device is now open");
-
- return 0;
-}
-
-bool AudioMixerManagerLinuxPulse::SpeakerIsInitialized() const
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s",
- __FUNCTION__);
-
- return (_paOutputDeviceIndex != -1);
-}
-
-bool AudioMixerManagerLinuxPulse::MicrophoneIsInitialized() const
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s",
- __FUNCTION__);
-
- return (_paInputDeviceIndex != -1);
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::SetSpeakerVolume(
- WebRtc_UWord32 volume)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerLinuxPulse::SetSpeakerVolume(volume=%u)",
- volume);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_paOutputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " output device index has not been set");
- return -1;
- }
-
- bool setFailed(false);
-
- if (_paPlayStream && (LATE(pa_stream_get_state)(_paPlayStream)
- != PA_STREAM_UNCONNECTED))
- {
- // We can only really set the volume if we have a connected stream
- PaLock();
-
- // Get the number of channels from the sample specification
- const pa_sample_spec *spec =
- LATE(pa_stream_get_sample_spec)(_paPlayStream);
- if (!spec)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " could not get sample specification");
- PaUnLock();
- return -1;
- }
-
- // Set the same volume for all channels
- pa_cvolume cVolumes;
- LATE(pa_cvolume_set)(&cVolumes, spec->channels, volume);
-
- pa_operation* paOperation = NULL;
- paOperation = LATE(pa_context_set_sink_input_volume)(
- _paContext,
- LATE(pa_stream_get_index)(_paPlayStream),
- &cVolumes,
- PaSetVolumeCallback, NULL);
- if (!paOperation)
- {
- setFailed = true;
- }
-
- // Don't need to wait for the completion
- LATE(pa_operation_unref)(paOperation);
-
- PaUnLock();
- } else
- {
- // We have not created a stream or it's not connected to the sink
- // Save the volume to be set at connection
- _paSpeakerVolume = volume;
- }
-
- if (setFailed)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " could not set speaker volume, error%d",
- LATE(pa_context_errno)(_paContext));
-
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerLinuxPulse::SpeakerVolume(WebRtc_UWord32& volume) const
-{
-
- if (_paOutputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " output device index has not been set");
- return -1;
- }
-
- if (_paPlayStream && (LATE(pa_stream_get_state)(_paPlayStream)
- != PA_STREAM_UNCONNECTED))
- {
- // We can only get the volume if we have a connected stream
- if (!GetSinkInputInfo())
- return -1;
-
- volume = static_cast<WebRtc_UWord32> (_paVolume);
- ResetCallbackVariables();
- } else
- {
- volume = _paSpeakerVolume;
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " AudioMixerManagerLinuxPulse::SpeakerVolume() => vol=%i",
- volume);
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerLinuxPulse::MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const
-{
-
- if (_paOutputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " output device index has not been set");
- return -1;
- }
-
- // PA_VOLUME_NORM corresponds to 100% (0db)
- // but PA allows up to 150 db amplification
- maxVolume = static_cast<WebRtc_UWord32> (PA_VOLUME_NORM);
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerLinuxPulse::MinSpeakerVolume(WebRtc_UWord32& minVolume) const
-{
-
- if (_paOutputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " output device index has not been set");
- return -1;
- }
-
- minVolume = static_cast<WebRtc_UWord32> (PA_VOLUME_MUTED);
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerLinuxPulse::SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const
-{
-
- if (_paOutputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " output device index has not been set");
- return -1;
- }
-
- // The sink input (stream) will always have step size = 1
- // There are PA_VOLUME_NORM+1 steps
- stepSize = 1;
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " AudioMixerManagerLinuxPulse::SpeakerVolumeStepSize() => "
- "size=%i, stepSize");
-
- // Reset members modified by callback
- ResetCallbackVariables();
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerLinuxPulse::SpeakerVolumeIsAvailable(bool& available)
-{
- if (_paOutputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " output device index has not been set");
- return -1;
- }
-
- // Always available in Pulse Audio
- available = true;
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerLinuxPulse::SpeakerMuteIsAvailable(bool& available)
-{
- if (_paOutputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " output device index has not been set");
- return -1;
- }
-
- // Always available in Pulse Audio
- available = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::SetSpeakerMute(bool enable)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerLinuxPulse::SetSpeakerMute(enable=%u)",
- enable);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_paOutputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " output device index has not been set");
- return -1;
- }
-
- bool setFailed(false);
-
- if (_paPlayStream && (LATE(pa_stream_get_state)(_paPlayStream)
- != PA_STREAM_UNCONNECTED))
- {
- // We can only really mute if we have a connected stream
- PaLock();
-
- pa_operation* paOperation = NULL;
- paOperation = LATE(pa_context_set_sink_input_mute)(
- _paContext,
- LATE(pa_stream_get_index)(_paPlayStream),
- (int) enable,
- PaSetVolumeCallback,
- NULL);
- if (!paOperation)
- {
- setFailed = true;
- }
-
- // Don't need to wait for the completion
- LATE(pa_operation_unref)(paOperation);
-
- PaUnLock();
- } else
- {
- // We have not created a stream or it's not connected to the sink
- // Save the mute status to be set at connection
- _paSpeakerMute = enable;
- }
-
- if (setFailed)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " could not mute speaker, error%d",
- LATE(pa_context_errno)(_paContext));
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::SpeakerMute(bool& enabled) const
-{
-
- if (_paOutputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " output device index has not been set");
- return -1;
- }
-
- if (_paPlayStream && (LATE(pa_stream_get_state)(_paPlayStream)
- != PA_STREAM_UNCONNECTED))
- {
- // We can only get the mute status if we have a connected stream
- if (!GetSinkInputInfo())
- return -1;
-
- enabled = static_cast<bool> (_paMute);
- ResetCallbackVariables();
- } else
- {
- enabled = _paSpeakerMute;
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " AudioMixerManagerLinuxPulse::SpeakerMute() => "
- "enabled=%i, enabled");
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerLinuxPulse::StereoPlayoutIsAvailable(bool& available)
-{
- if (_paOutputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " output device index has not been set");
- return -1;
- }
-
- uint32_t deviceIndex = (uint32_t) _paOutputDeviceIndex;
-
- PaLock();
-
- // Get the actual stream device index if we have a connected stream
- // The device used by the stream can be changed
- // during the call
- if (_paPlayStream && (LATE(pa_stream_get_state)(_paPlayStream)
- != PA_STREAM_UNCONNECTED))
- {
- deviceIndex = LATE(pa_stream_get_device_index)(_paPlayStream);
- }
-
- PaUnLock();
-
- if (!GetSinkInfoByIndex(deviceIndex))
- return -1;
-
- available = static_cast<bool> (_paChannels == 2);
-
- // Reset members modified by callback
- ResetCallbackVariables();
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable(bool& available)
-{
- if (_paInputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " input device index has not been set");
- return -1;
- }
-
- uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
-
- PaLock();
-
- // Get the actual stream device index if we have a connected stream
- // The device used by the stream can be changed
- // during the call
- if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
- != PA_STREAM_UNCONNECTED))
- {
- deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
- }
-
- pa_operation* paOperation = NULL;
- ResetCallbackVariables();
-
- // Get info for this source
- // We want to know if the actual device can record in stereo
- paOperation = LATE(pa_context_get_source_info_by_index)(
- _paContext, deviceIndex,
- PaSourceInfoCallback,
- (void*) this);
-
- WaitForOperationCompletion(paOperation);
- PaUnLock();
-
- if (!_callbackValues)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Error getting number of input channels: %d",
- LATE(pa_context_errno)(_paContext));
- return -1;
- }
-
- available = static_cast<bool> (_paChannels == 2);
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " AudioMixerManagerLinuxPulse::StereoRecordingIsAvailable()"
- " => available=%i, available");
-
- // Reset members modified by callback
- ResetCallbackVariables();
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::MicrophoneMuteIsAvailable(
- bool& available)
-{
- if (_paInputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " input device index has not been set");
- return -1;
- }
-
- // Always available in Pulse Audio
- available = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::SetMicrophoneMute(bool enable)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerLinuxPulse::SetMicrophoneMute(enable=%u)",
- enable);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_paInputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " input device index has not been set");
- return -1;
- }
-
- bool setFailed(false);
- pa_operation* paOperation = NULL;
- ResetCallbackVariables();
-
- uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
-
- PaLock();
-
- // Get the actual stream device index if we have a connected stream
- // The device used by the stream can be changed
- // during the call
- if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
- != PA_STREAM_UNCONNECTED))
- {
- deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
- }
-
- // Set mute switch for the source
- paOperation = LATE(pa_context_set_source_mute_by_index)(
- _paContext, deviceIndex,
- enable,
- PaSetVolumeCallback, NULL);
-
- if (!paOperation)
- {
- setFailed = true;
- }
-
- // Don't need to wait for this to complete.
- LATE(pa_operation_unref)(paOperation);
-
- PaUnLock();
-
- // Reset variables altered by callback
- ResetCallbackVariables();
-
- if (setFailed)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " could not mute microphone, error%d",
- LATE(pa_context_errno)(_paContext));
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::MicrophoneMute(bool& enabled) const
-{
-
- if (_paInputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " input device index has not been set");
- return -1;
- }
-
- uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
-
- PaLock();
-
- // Get the actual stream device index if we have a connected stream
- // The device used by the stream can be changed
- // during the call
- if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
- != PA_STREAM_UNCONNECTED))
- {
- deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
- }
-
- PaUnLock();
-
- if (!GetSourceInfoByIndex(deviceIndex))
- return -1;
-
- enabled = static_cast<bool> (_paMute);
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " AudioMixerManagerLinuxPulse::MicrophoneMute() =>"
- " enabled=%i, enabled");
-
- // Reset members modified by callback
- ResetCallbackVariables();
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerLinuxPulse::MicrophoneBoostIsAvailable(bool& available)
-{
- if (_paInputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " input device index has not been set");
- return -1;
- }
-
- // Always unavailable in Pulse Audio
- // Could make it possible to use PA_VOLUME_MAX
- // but that gives bad audio with some sound cards
- available = false;
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::SetMicrophoneBoost(bool enable)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerLinuxPulse::SetMicrophoneBoost(enable=%u)",
- enable);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_paInputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " input device index has not been set");
- return -1;
- }
-
- // Ensure that the selected microphone destination has a valid boost control
- bool available(false);
- MicrophoneBoostIsAvailable(available);
- if (!available)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " it is not possible to enable microphone boost");
- return -1;
- }
-
- // It is assumed that the call above fails!
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::MicrophoneBoost(bool& enabled) const
-{
-
- if (_paInputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " input device index has not been set");
- return -1;
- }
-
- // Microphone boost cannot be enabled on this platform!
- enabled = false;
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::MicrophoneVolumeIsAvailable(
- bool& available)
-{
- if (_paInputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " input device index has not been set");
- return -1;
- }
-
- // Always available in Pulse Audio
- available = true;
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerLinuxPulse::SetMicrophoneVolume(WebRtc_UWord32 volume)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerLinuxPulse::SetMicrophoneVolume(volume=%u)",
- volume);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_paInputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " input device index has not been set");
- return -1;
- }
-
- // Unlike output streams, input streams have no concept of a stream volume,
- // only a device volume. So we have to change the volume of the device
- // itself.
-
- // The device may have a different number of channels than the stream and
- // their mapping may be different, so we don't want to use the channel count
- // from our sample spec. We could use PA_CHANNELS_MAX to cover our bases,
- // and the server allows that even if the device's channel count is lower,
- // but some buggy PA clients don't like that (the pavucontrol on Hardy dies
- // in an assert if the channel count is different). So instead we look up
- // the actual number of channels that the device has.
-
- uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
-
- PaLock();
-
- // Get the actual stream device index if we have a connected stream
- // The device used by the stream can be changed
- // during the call
- if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
- != PA_STREAM_UNCONNECTED))
- {
- deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
- }
-
- bool setFailed(false);
- pa_operation* paOperation = NULL;
- ResetCallbackVariables();
-
- // Get the number of channels for this source
- paOperation
- = LATE(pa_context_get_source_info_by_index)(_paContext, deviceIndex,
- PaSourceInfoCallback,
- (void*) this);
-
- WaitForOperationCompletion(paOperation);
-
- if (!_callbackValues)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Error getting input channels: %d",
- LATE(pa_context_errno)(_paContext));
- PaUnLock();
- return -1;
- }
-
- WebRtc_UWord8 channels = _paChannels;
- ResetCallbackVariables();
-
- pa_cvolume cVolumes;
- LATE(pa_cvolume_set)(&cVolumes, channels, volume);
-
- // Set the volume for the source
- paOperation
- = LATE(pa_context_set_source_volume_by_index)(_paContext, deviceIndex,
- &cVolumes,
- PaSetVolumeCallback, NULL);
-
- if (!paOperation)
- {
- setFailed = true;
- }
-
- // Don't need to wait for this to complete.
- LATE(pa_operation_unref)(paOperation);
-
- PaUnLock();
-
- // Reset variables altered by callback
- ResetCallbackVariables();
-
- if (setFailed)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " could not set microphone volume, error%d",
- LATE(pa_context_errno)(_paContext));
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerLinuxPulse::MicrophoneVolume(WebRtc_UWord32& volume) const
-{
-
- if (_paInputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " input device index has not been set");
- return -1;
- }
-
- uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
-
- PaLock();
-
- // Get the actual stream device index if we have a connected stream
- // The device used by the stream can be changed
- // during the call
- if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
- != PA_STREAM_UNCONNECTED))
- {
- deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
- }
-
- PaUnLock();
-
- if (!GetSourceInfoByIndex(deviceIndex))
- return -1;
-
- volume = static_cast<WebRtc_UWord32> (_paVolume);
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " AudioMixerManagerLinuxPulse::MicrophoneVolume() => vol=%i, volume");
-
- // Reset members modified by callback
- ResetCallbackVariables();
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerLinuxPulse::MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const
-{
-
- if (_paInputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " input device index has not been set");
- return -1;
- }
-
- // PA_VOLUME_NORM corresponds to 100% (0db)
- // PA allows up to 150 db amplification (PA_VOLUME_MAX)
- // but that doesn't work well for all sound cards
- maxVolume = static_cast<WebRtc_UWord32> (PA_VOLUME_NORM);
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerLinuxPulse::MinMicrophoneVolume(WebRtc_UWord32& minVolume) const
-{
-
- if (_paInputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " input device index has not been set");
- return -1;
- }
-
- minVolume = static_cast<WebRtc_UWord32> (PA_VOLUME_MUTED);
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerLinuxPulse::MicrophoneVolumeStepSize(
- WebRtc_UWord16& stepSize) const
-{
-
- if (_paInputDeviceIndex == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " input device index has not been set");
- return -1;
- }
-
- uint32_t deviceIndex = (uint32_t) _paInputDeviceIndex;
-
- PaLock();
-
- // Get the actual stream device index if we have a connected stream
- // The device used by the stream can be changed
- // during the call
- if (_paRecStream && (LATE(pa_stream_get_state)(_paRecStream)
- != PA_STREAM_UNCONNECTED))
- {
- deviceIndex = LATE(pa_stream_get_device_index)(_paRecStream);
- }
-
- pa_operation* paOperation = NULL;
- ResetCallbackVariables();
-
- // Get info for this source
- paOperation
- = LATE(pa_context_get_source_info_by_index)(_paContext, deviceIndex,
- PaSourceInfoCallback,
- (void*) this);
-
- WaitForOperationCompletion(paOperation);
-
- PaUnLock();
-
- if (!_callbackValues)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Error getting step size: %d",
- LATE(pa_context_errno)(_paContext));
- return -1;
- }
-
- stepSize = static_cast<WebRtc_UWord16> ((PA_VOLUME_NORM + 1) / _paVolSteps);
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " AudioMixerManagerLinuxPulse::MicrophoneVolumeStepSize()"
- " => size=%i, stepSize");
-
- // Reset members modified by callback
- ResetCallbackVariables();
-
- return 0;
-}
-
-// ============================================================================
-// Private Methods
-// ============================================================================
-
-void AudioMixerManagerLinuxPulse::PaSinkInfoCallback(pa_context */*c*/,
- const pa_sink_info *i,
- int eol, void *pThis)
-{
- static_cast<AudioMixerManagerLinuxPulse*> (pThis)-> PaSinkInfoCallbackHandler(
- i, eol);
-}
-
-void AudioMixerManagerLinuxPulse::PaSinkInputInfoCallback(
- pa_context */*c*/,
- const pa_sink_input_info *i,
- int eol, void *pThis)
-{
- static_cast<AudioMixerManagerLinuxPulse*> (pThis)->
- PaSinkInputInfoCallbackHandler(i, eol);
-}
-
-
-void AudioMixerManagerLinuxPulse::PaSourceInfoCallback(pa_context */*c*/,
- const pa_source_info *i,
- int eol, void *pThis)
-{
- static_cast<AudioMixerManagerLinuxPulse*> (pThis)->
- PaSourceInfoCallbackHandler(i, eol);
-}
-
-void AudioMixerManagerLinuxPulse::PaSetVolumeCallback(pa_context * c,
- int success, void */*pThis*/)
-{
- if (!success)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
- " failed to set volume");
- }
-}
-
-void AudioMixerManagerLinuxPulse::PaSinkInfoCallbackHandler(
- const pa_sink_info *i,
- int eol)
-{
- if (eol)
- {
- // Signal that we are done
- LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
- return;
- }
-
- _callbackValues = true;
- _paChannels = i->channel_map.channels; // Get number of channels
- pa_volume_t paVolume = PA_VOLUME_MUTED; // Minimum possible value.
- for (int j = 0; j < _paChannels; ++j)
- {
- if (paVolume < i->volume.values[j])
- {
- paVolume = i->volume.values[j];
- }
- }
- _paVolume = paVolume; // get the max volume for any channel
- _paMute = i->mute; // get mute status
-
- // supported since PA 0.9.15
- //_paVolSteps = i->n_volume_steps; // get the number of volume steps
- // default value is PA_VOLUME_NORM+1
- _paVolSteps = PA_VOLUME_NORM + 1;
-}
-
-void AudioMixerManagerLinuxPulse::PaSinkInputInfoCallbackHandler(
- const pa_sink_input_info *i,
- int eol)
-{
- if (eol)
- {
- // Signal that we are done.
- LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
- return;
- }
-
- _callbackValues = true;
- _paChannels = i->channel_map.channels; // Get number of channels
- pa_volume_t paVolume = PA_VOLUME_MUTED; // Minimum possible value.
- for (int j = 0; j < _paChannels; ++j)
- {
- if (paVolume < i->volume.values[j])
- {
- paVolume = i->volume.values[j];
- }
- }
- _paVolume = paVolume; // Get the max volume for any channel
- _paMute = i->mute; // Get mute status
-}
-
-void AudioMixerManagerLinuxPulse::PaSourceInfoCallbackHandler(
- const pa_source_info *i,
- int eol)
-{
- if (eol)
- {
- // Signal that we are done
- LATE(pa_threaded_mainloop_signal)(_paMainloop, 0);
- return;
- }
-
- _callbackValues = true;
- _paChannels = i->channel_map.channels; // Get number of channels
- pa_volume_t paVolume = PA_VOLUME_MUTED; // Minimum possible value.
- for (int j = 0; j < _paChannels; ++j)
- {
- if (paVolume < i->volume.values[j])
- {
- paVolume = i->volume.values[j];
- }
- }
- _paVolume = paVolume; // Get the max volume for any channel
- _paMute = i->mute; // Get mute status
-
- // supported since PA 0.9.15
- //_paVolSteps = i->n_volume_steps; // Get the number of volume steps
- // default value is PA_VOLUME_NORM+1
- _paVolSteps = PA_VOLUME_NORM + 1;
-}
-
-void AudioMixerManagerLinuxPulse::ResetCallbackVariables() const
-{
- _paVolume = 0;
- _paMute = 0;
- _paVolSteps = 0;
- _paChannels = 0;
- _callbackValues = false;
-}
-
-void AudioMixerManagerLinuxPulse::WaitForOperationCompletion(
- pa_operation* paOperation) const
-{
- while (LATE(pa_operation_get_state)(paOperation) == PA_OPERATION_RUNNING)
- {
- LATE(pa_threaded_mainloop_wait)(_paMainloop);
- }
-
- LATE(pa_operation_unref)(paOperation);
-}
-
-void AudioMixerManagerLinuxPulse::PaLock() const
-{
- LATE(pa_threaded_mainloop_lock)(_paMainloop);
-}
-
-void AudioMixerManagerLinuxPulse::PaUnLock() const
-{
- LATE(pa_threaded_mainloop_unlock)(_paMainloop);
-}
-
-bool AudioMixerManagerLinuxPulse::GetSinkInputInfo() const {
- pa_operation* paOperation = NULL;
- ResetCallbackVariables();
-
- PaLock();
- for (int retries = 0; retries < kMaxRetryOnFailure && !_callbackValues;
- retries ++) {
- // Get info for this stream (sink input).
- paOperation = LATE(pa_context_get_sink_input_info)(
- _paContext,
- LATE(pa_stream_get_index)(_paPlayStream),
- PaSinkInputInfoCallback,
- (void*) this);
-
- WaitForOperationCompletion(paOperation);
- }
- PaUnLock();
-
- if (!_callbackValues) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "GetSinkInputInfo failed to get volume info : %d",
- LATE(pa_context_errno)(_paContext));
- return false;
- }
-
- return true;
-}
-
-bool AudioMixerManagerLinuxPulse::GetSinkInfoByIndex(
- int device_index) const {
- pa_operation* paOperation = NULL;
- ResetCallbackVariables();
-
- PaLock();
- for (int retries = 0; retries < kMaxRetryOnFailure && !_callbackValues;
- retries ++) {
- paOperation = LATE(pa_context_get_sink_info_by_index)(_paContext,
- device_index, PaSinkInfoCallback, (void*) this);
-
- WaitForOperationCompletion(paOperation);
- }
- PaUnLock();
-
- if (!_callbackValues) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "GetSinkInfoByIndex failed to get volume info: %d",
- LATE(pa_context_errno)(_paContext));
- return false;
- }
-
- return true;
-}
-
-bool AudioMixerManagerLinuxPulse::GetSourceInfoByIndex(
- int device_index) const {
- pa_operation* paOperation = NULL;
- ResetCallbackVariables();
-
- PaLock();
- for (int retries = 0; retries < kMaxRetryOnFailure && !_callbackValues;
- retries ++) {
- paOperation = LATE(pa_context_get_source_info_by_index)(
- _paContext, device_index, PaSourceInfoCallback, (void*) this);
-
- WaitForOperationCompletion(paOperation);
- }
-
- PaUnLock();
-
- if (!_callbackValues) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "GetSourceInfoByIndex error: %d",
- LATE(pa_context_errno)(_paContext));
- return false;
- }
-
- return true;
-}
-
-}
-
diff --git a/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_pulse_linux.h b/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_pulse_linux.h
deleted file mode 100644
index 22d6da5..0000000
--- a/trunk/src/modules/audio_device/main/source/linux/audio_mixer_manager_pulse_linux.h
+++ /dev/null
@@ -1,121 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_PULSE_LINUX_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_PULSE_LINUX_H
-
-#include "typedefs.h"
-#include "audio_device.h"
-#include "critical_section_wrapper.h"
-#include "pulseaudiosymboltable_linux.h"
-
-#include <stdint.h>
-#include <pulse/pulseaudio.h>
-
-#ifndef UINT32_MAX
-#define UINT32_MAX ((uint32_t)-1)
-#endif
-
-namespace webrtc
-{
-
-class AudioMixerManagerLinuxPulse
-{
-public:
- WebRtc_Word32 SetPlayStream(pa_stream* playStream);
- WebRtc_Word32 SetRecStream(pa_stream* recStream);
- WebRtc_Word32 OpenSpeaker(WebRtc_UWord16 deviceIndex);
- WebRtc_Word32 OpenMicrophone(WebRtc_UWord16 deviceIndex);
- WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
- WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
- WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
- WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
- WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
- WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
- WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
- WebRtc_Word32 SetSpeakerMute(bool enable);
- WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
- WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
- WebRtc_Word32 SpeakerMute(bool& enabled) const;
- WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
- WebRtc_Word32 SetMicrophoneMute(bool enable);
- WebRtc_Word32 MicrophoneMute(bool& enabled) const;
- WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
- WebRtc_Word32 SetMicrophoneBoost(bool enable);
- WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
- WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
- WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
- WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
- WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
- WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
- WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
- WebRtc_Word32 SetPulseAudioObjects(pa_threaded_mainloop* mainloop,
- pa_context* context);
- WebRtc_Word32 Close();
- WebRtc_Word32 CloseSpeaker();
- WebRtc_Word32 CloseMicrophone();
- bool SpeakerIsInitialized() const;
- bool MicrophoneIsInitialized() const;
-
-public:
- AudioMixerManagerLinuxPulse(const WebRtc_Word32 id);
- ~AudioMixerManagerLinuxPulse();
-
-private:
- static void PaSinkInfoCallback(pa_context *c, const pa_sink_info *i,
- int eol, void *pThis);
- static void PaSinkInputInfoCallback(pa_context *c,
- const pa_sink_input_info *i, int eol,
- void *pThis);
- static void PaSourceInfoCallback(pa_context *c, const pa_source_info *i,
- int eol, void *pThis);
- static void
- PaSetVolumeCallback(pa_context * /*c*/, int success, void */*pThis*/);
- void PaSinkInfoCallbackHandler(const pa_sink_info *i, int eol);
- void PaSinkInputInfoCallbackHandler(const pa_sink_input_info *i, int eol);
- void PaSourceInfoCallbackHandler(const pa_source_info *i, int eol);
-
- void ResetCallbackVariables() const;
- void WaitForOperationCompletion(pa_operation* paOperation) const;
- void PaLock() const;
- void PaUnLock() const;
-
- bool GetSinkInputInfo() const;
- bool GetSinkInfoByIndex(int device_index)const ;
- bool GetSourceInfoByIndex(int device_index) const;
-
-private:
- CriticalSectionWrapper& _critSect;
- WebRtc_Word32 _id;
- WebRtc_Word16 _paOutputDeviceIndex;
- WebRtc_Word16 _paInputDeviceIndex;
-
- pa_stream* _paPlayStream;
- pa_stream* _paRecStream;
-
- pa_threaded_mainloop* _paMainloop;
- pa_context* _paContext;
-
- mutable WebRtc_UWord32 _paVolume;
- mutable WebRtc_UWord32 _paMute;
- mutable WebRtc_UWord32 _paVolSteps;
- bool _paSpeakerMute;
- mutable WebRtc_UWord32 _paSpeakerVolume;
- mutable WebRtc_UWord8 _paChannels;
- bool _paObjectsSet;
- mutable bool _callbackValues;
-
- WebRtc_UWord8 _micVolChannels;
- WebRtc_UWord8 _spkVolChannels;
-};
-
-}
-
-#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_LINUX_AUDIO_MIXER_MANAGER_PULSE_LINUX_H_
diff --git a/trunk/src/modules/audio_device/main/source/linux/latebindingsymboltable_linux.cc b/trunk/src/modules/audio_device/main/source/linux/latebindingsymboltable_linux.cc
deleted file mode 100644
index 8f3c7c8..0000000
--- a/trunk/src/modules/audio_device/main/source/linux/latebindingsymboltable_linux.cc
+++ /dev/null
@@ -1,116 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "latebindingsymboltable_linux.h"
-
-#ifdef WEBRTC_LINUX
-#include <dlfcn.h>
-#endif
-
-// TODO(grunell): Either put inside webrtc namespace or use webrtc:: instead.
-using namespace webrtc;
-
-namespace webrtc_adm_linux {
-
-inline static const char *GetDllError() {
-#ifdef WEBRTC_LINUX
- char *err = dlerror();
- if (err) {
- return err;
- } else {
- return "No error";
- }
-#else
-#error Not implemented
-#endif
-}
-
-DllHandle InternalLoadDll(const char dll_name[]) {
-#ifdef WEBRTC_LINUX
- DllHandle handle = dlopen(dll_name, RTLD_NOW);
-#else
-#error Not implemented
-#endif
- if (handle == kInvalidDllHandle) {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, -1,
- "Can't load %s : %d", dll_name, GetDllError());
- }
- return handle;
-}
-
-void InternalUnloadDll(DllHandle handle) {
-#ifdef WEBRTC_LINUX
- if (dlclose(handle) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
- "%d", GetDllError());
- }
-#else
-#error Not implemented
-#endif
-}
-
-static bool LoadSymbol(DllHandle handle,
- const char *symbol_name,
- void **symbol) {
-#ifdef WEBRTC_LINUX
- *symbol = dlsym(handle, symbol_name);
- char *err = dlerror();
- if (err) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
- "Error loading symbol %s : %d", symbol_name, err);
- return false;
- } else if (!*symbol) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
- "Symbol %s is NULL", symbol_name);
- return false;
- }
- return true;
-#else
-#error Not implemented
-#endif
-}
-
-// This routine MUST assign SOME value for every symbol, even if that value is
-// NULL, or else some symbols may be left with uninitialized data that the
-// caller may later interpret as a valid address.
-bool InternalLoadSymbols(DllHandle handle,
- int num_symbols,
- const char *const symbol_names[],
- void *symbols[]) {
-#ifdef WEBRTC_LINUX
- // Clear any old errors.
- dlerror();
-#endif
- for (int i = 0; i < num_symbols; ++i) {
- if (!LoadSymbol(handle, symbol_names[i], &symbols[i])) {
- return false;
- }
- }
- return true;
-}
-
-} // namespace webrtc_adm_linux
diff --git a/trunk/src/modules/audio_device/main/source/linux/latebindingsymboltable_linux.h b/trunk/src/modules/audio_device/main/source/linux/latebindingsymboltable_linux.h
deleted file mode 100644
index 91d25aa..0000000
--- a/trunk/src/modules/audio_device/main/source/linux/latebindingsymboltable_linux.h
+++ /dev/null
@@ -1,195 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_LATEBINDINGSYMBOLTABLE_LINUX_H
-#define WEBRTC_AUDIO_DEVICE_LATEBINDINGSYMBOLTABLE_LINUX_H
-
-#include <assert.h>
-#include <stddef.h> // for NULL
-#include <string.h>
-
-#include "constructor_magic.h"
-#include "trace.h"
-
-// This file provides macros for creating "symbol table" classes to simplify the
-// dynamic loading of symbols from DLLs. Currently the implementation only
-// supports Linux and pure C symbols.
-// See talk/sound/pulseaudiosymboltable.(h|cc) for an example.
-
-namespace webrtc_adm_linux {
-
-#ifdef WEBRTC_LINUX
-typedef void *DllHandle;
-
-const DllHandle kInvalidDllHandle = NULL;
-#else
-#error Not implemented
-#endif
-
-// These are helpers for use only by the class below.
-DllHandle InternalLoadDll(const char dll_name[]);
-
-void InternalUnloadDll(DllHandle handle);
-
-bool InternalLoadSymbols(DllHandle handle,
- int num_symbols,
- const char *const symbol_names[],
- void *symbols[]);
-
-template <int SYMBOL_TABLE_SIZE,
- const char kDllName[],
- const char *const kSymbolNames[]>
-class LateBindingSymbolTable {
- public:
- LateBindingSymbolTable()
- : handle_(kInvalidDllHandle),
- undefined_symbols_(false) {
- memset(symbols_, 0, sizeof(symbols_));
- }
-
- ~LateBindingSymbolTable() {
- Unload();
- }
-
- static int NumSymbols() {
- return SYMBOL_TABLE_SIZE;
- }
-
- // We do not use this, but we offer it for theoretical convenience.
- static const char *GetSymbolName(int index) {
- assert(index < NumSymbols());
- return kSymbolNames[index];
- }
-
- bool IsLoaded() const {
- return handle_ != kInvalidDllHandle;
- }
-
- // Loads the DLL and the symbol table. Returns true iff the DLL and symbol
- // table loaded successfully.
- bool Load() {
- if (IsLoaded()) {
- return true;
- }
- if (undefined_symbols_) {
- // We do not attempt to load again because repeated attempts are not
- // likely to succeed and DLL loading is costly.
- //WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
- // "We know there are undefined symbols");
- return false;
- }
- handle_ = InternalLoadDll(kDllName);
- if (!IsLoaded()) {
- return false;
- }
- if (!InternalLoadSymbols(handle_, NumSymbols(), kSymbolNames, symbols_)) {
- undefined_symbols_ = true;
- Unload();
- return false;
- }
- return true;
- }
-
- void Unload() {
- if (!IsLoaded()) {
- return;
- }
- InternalUnloadDll(handle_);
- handle_ = kInvalidDllHandle;
- memset(symbols_, 0, sizeof(symbols_));
- }
-
- // Retrieves the given symbol. NOTE: Recommended to use LATESYM_GET below
- // instead of this.
- void *GetSymbol(int index) const {
- assert(IsLoaded());
- assert(index < NumSymbols());
- return symbols_[index];
- }
-
- private:
- DllHandle handle_;
- bool undefined_symbols_;
- void *symbols_[SYMBOL_TABLE_SIZE];
-
- DISALLOW_COPY_AND_ASSIGN(LateBindingSymbolTable);
-};
-
-// This macro must be invoked in a header to declare a symbol table class.
-#define LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(ClassName) \
-enum {
-
-// This macro must be invoked in the header declaration once for each symbol
-// (recommended to use an X-Macro to avoid duplication).
-// This macro defines an enum with names built from the symbols, which
-// essentially creates a hash table in the compiler from symbol names to their
-// indices in the symbol table class.
-#define LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(ClassName, sym) \
- ClassName##_SYMBOL_TABLE_INDEX_##sym,
-
-// This macro completes the header declaration.
-#define LATE_BINDING_SYMBOL_TABLE_DECLARE_END(ClassName) \
- ClassName##_SYMBOL_TABLE_SIZE \
-}; \
-\
-extern const char ClassName##_kDllName[]; \
-extern const char *const \
- ClassName##_kSymbolNames[ClassName##_SYMBOL_TABLE_SIZE]; \
-\
-typedef ::webrtc_adm_linux::LateBindingSymbolTable<ClassName##_SYMBOL_TABLE_SIZE, \
- ClassName##_kDllName, \
- ClassName##_kSymbolNames> \
- ClassName;
-
-// This macro must be invoked in a .cc file to define a previously-declared
-// symbol table class.
-#define LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(ClassName, dllName) \
-const char ClassName##_kDllName[] = dllName; \
-const char *const ClassName##_kSymbolNames[ClassName##_SYMBOL_TABLE_SIZE] = {
-
-// This macro must be invoked in the .cc definition once for each symbol
-// (recommended to use an X-Macro to avoid duplication).
-// This would have to use the mangled name if we were to ever support C++
-// symbols.
-#define LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(ClassName, sym) \
- #sym,
-
-#define LATE_BINDING_SYMBOL_TABLE_DEFINE_END(ClassName) \
-};
-
-// Index of a given symbol in the given symbol table class.
-#define LATESYM_INDEXOF(ClassName, sym) \
- (ClassName##_SYMBOL_TABLE_INDEX_##sym)
-
-// Returns a reference to the given late-binded symbol, with the correct type.
-#define LATESYM_GET(ClassName, inst, sym) \
- (*reinterpret_cast<typeof(&sym)>( \
- (inst)->GetSymbol(LATESYM_INDEXOF(ClassName, sym))))
-
-} // namespace webrtc_adm_linux
-
-#endif // WEBRTC_ADM_LATEBINDINGSYMBOLTABLE_LINUX_H
diff --git a/trunk/src/modules/audio_device/main/source/linux/pulseaudiosymboltable_linux.cc b/trunk/src/modules/audio_device/main/source/linux/pulseaudiosymboltable_linux.cc
deleted file mode 100644
index ae663f7..0000000
--- a/trunk/src/modules/audio_device/main/source/linux/pulseaudiosymboltable_linux.cc
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#include "pulseaudiosymboltable_linux.h"
-
-namespace webrtc_adm_linux_pulse {
-
-LATE_BINDING_SYMBOL_TABLE_DEFINE_BEGIN(PulseAudioSymbolTable, "libpulse.so.0")
-#define X(sym) \
- LATE_BINDING_SYMBOL_TABLE_DEFINE_ENTRY(PulseAudioSymbolTable, sym)
-PULSE_AUDIO_SYMBOLS_LIST
-#undef X
-LATE_BINDING_SYMBOL_TABLE_DEFINE_END(PulseAudioSymbolTable)
-
-} // namespace webrtc_adm_linux_pulse
diff --git a/trunk/src/modules/audio_device/main/source/linux/pulseaudiosymboltable_linux.h b/trunk/src/modules/audio_device/main/source/linux/pulseaudiosymboltable_linux.h
deleted file mode 100644
index 049509b..0000000
--- a/trunk/src/modules/audio_device/main/source/linux/pulseaudiosymboltable_linux.h
+++ /dev/null
@@ -1,104 +0,0 @@
-/*
- * libjingle
- * Copyright 2004--2010, Google Inc.
- *
- * Redistribution and use in source and binary forms, with or without
- * modification, are permitted provided that the following conditions are met:
- *
- * 1. Redistributions of source code must retain the above copyright notice,
- * this list of conditions and the following disclaimer.
- * 2. Redistributions in binary form must reproduce the above copyright notice,
- * this list of conditions and the following disclaimer in the documentation
- * and/or other materials provided with the distribution.
- * 3. The name of the author may not be used to endorse or promote products
- * derived from this software without specific prior written permission.
- *
- * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
- * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
- * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
- * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
- * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
- * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
- * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
- * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
- * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
- * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_LINUX_H
-#define WEBRTC_AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_LINUX_H
-
-#include "latebindingsymboltable_linux.h"
-
-namespace webrtc_adm_linux_pulse {
-
-// The PulseAudio symbols we need, as an X-Macro list.
-// This list must contain precisely every libpulse function that is used in
-// the ADM LINUX PULSE Device and Mixer classes
-#define PULSE_AUDIO_SYMBOLS_LIST \
- X(pa_bytes_per_second) \
- X(pa_context_connect) \
- X(pa_context_disconnect) \
- X(pa_context_errno) \
- X(pa_context_get_protocol_version) \
- X(pa_context_get_server_info) \
- X(pa_context_get_sink_info_list) \
- X(pa_context_get_sink_info_by_index) \
- X(pa_context_get_sink_info_by_name) \
- X(pa_context_get_sink_input_info) \
- X(pa_context_get_source_info_by_index) \
- X(pa_context_get_source_info_by_name) \
- X(pa_context_get_source_info_list) \
- X(pa_context_get_state) \
- X(pa_context_new) \
- X(pa_context_set_sink_input_volume) \
- X(pa_context_set_sink_input_mute) \
- X(pa_context_set_source_volume_by_index) \
- X(pa_context_set_source_mute_by_index) \
- X(pa_context_set_state_callback) \
- X(pa_context_unref) \
- X(pa_cvolume_set) \
- X(pa_operation_get_state) \
- X(pa_operation_unref) \
- X(pa_stream_connect_playback) \
- X(pa_stream_connect_record) \
- X(pa_stream_disconnect) \
- X(pa_stream_drop) \
- X(pa_stream_get_device_index) \
- X(pa_stream_get_index) \
- X(pa_stream_get_latency) \
- X(pa_stream_get_sample_spec) \
- X(pa_stream_get_state) \
- X(pa_stream_new) \
- X(pa_stream_peek) \
- X(pa_stream_readable_size) \
- X(pa_stream_set_buffer_attr) \
- X(pa_stream_set_overflow_callback) \
- X(pa_stream_set_read_callback) \
- X(pa_stream_set_state_callback) \
- X(pa_stream_set_underflow_callback) \
- X(pa_stream_set_write_callback) \
- X(pa_stream_unref) \
- X(pa_stream_writable_size) \
- X(pa_stream_write) \
- X(pa_strerror) \
- X(pa_threaded_mainloop_free) \
- X(pa_threaded_mainloop_get_api) \
- X(pa_threaded_mainloop_lock) \
- X(pa_threaded_mainloop_new) \
- X(pa_threaded_mainloop_signal) \
- X(pa_threaded_mainloop_start) \
- X(pa_threaded_mainloop_stop) \
- X(pa_threaded_mainloop_unlock) \
- X(pa_threaded_mainloop_wait)
-
-LATE_BINDING_SYMBOL_TABLE_DECLARE_BEGIN(PulseAudioSymbolTable)
-#define X(sym) \
- LATE_BINDING_SYMBOL_TABLE_DECLARE_ENTRY(PulseAudioSymbolTable, sym)
-PULSE_AUDIO_SYMBOLS_LIST
-#undef X
-LATE_BINDING_SYMBOL_TABLE_DECLARE_END(PulseAudioSymbolTable)
-
-} // namespace webrtc_adm_linux_pulse
-
-#endif // WEBRTC_AUDIO_DEVICE_PULSEAUDIOSYMBOLTABLE_LINUX_H
diff --git a/trunk/src/modules/audio_device/main/source/mac/audio_device_mac.cc b/trunk/src/modules/audio_device/main/source/mac/audio_device_mac.cc
deleted file mode 100644
index e927ae7..0000000
--- a/trunk/src/modules/audio_device/main/source/mac/audio_device_mac.cc
+++ /dev/null
@@ -1,3241 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_device_utility.h"
-#include "audio_device_mac.h"
-#include "audio_device_config.h"
-
-#include "event_wrapper.h"
-#include "trace.h"
-#include "thread_wrapper.h"
-
-#include <cassert>
-
-#include <sys/sysctl.h> // sysctlbyname()
-#include <mach/mach.h> // mach_task_self()
-#include <libkern/OSAtomic.h> // OSAtomicCompareAndSwap()
-#include "portaudio/pa_ringbuffer.h"
-
-namespace webrtc
-{
-
-#define WEBRTC_CA_RETURN_ON_ERR(expr) \
- do { \
- err = expr; \
- if (err != noErr) { \
- logCAMsg(kTraceError, kTraceAudioDevice, _id, \
- "Error in " #expr, (const char *)&err); \
- return -1; \
- } \
- } while(0)
-
-#define WEBRTC_CA_LOG_ERR(expr) \
- do { \
- err = expr; \
- if (err != noErr) { \
- logCAMsg(kTraceError, kTraceAudioDevice, _id, \
- "Error in " #expr, (const char *)&err); \
- } \
- } while(0)
-
-#define WEBRTC_CA_LOG_WARN(expr) \
- do { \
- err = expr; \
- if (err != noErr) { \
- logCAMsg(kTraceWarning, kTraceAudioDevice, _id, \
- "Error in " #expr, (const char *)&err); \
- } \
- } while(0)
-
-enum
-{
- MaxNumberDevices = 64
-};
-
-void AudioDeviceMac::AtomicSet32(int32_t* theValue, int32_t newValue)
-{
- while (1)
- {
- int32_t oldValue = *theValue;
- if (OSAtomicCompareAndSwap32Barrier(oldValue, newValue, theValue)
- == true)
- {
- return;
- }
- }
-}
-
-int32_t AudioDeviceMac::AtomicGet32(int32_t* theValue)
-{
- while (1)
- {
- WebRtc_Word32 value = *theValue;
- if (OSAtomicCompareAndSwap32Barrier(value, value, theValue) == true)
- {
- return value;
- }
- }
-}
-
-// CoreAudio errors are best interpreted as four character strings.
-void AudioDeviceMac::logCAMsg(const TraceLevel level,
- const TraceModule module,
- const WebRtc_Word32 id, const char *msg,
- const char *err)
-{
- assert(msg != NULL);
- assert(err != NULL);
-
-#ifdef WEBRTC_BIG_ENDIAN
- WEBRTC_TRACE(level, module, id, "%s: %.4s", msg, err);
-#else
- // We need to flip the characters in this case.
- WEBRTC_TRACE(level, module, id, "%s: %.1s%.1s%.1s%.1s", msg, err + 3, err
- + 2, err + 1, err);
-#endif
-}
-
-AudioDeviceMac::AudioDeviceMac(const WebRtc_Word32 id) :
- _ptrAudioBuffer(NULL),
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _stopEventRec(*EventWrapper::Create()),
- _stopEvent(*EventWrapper::Create()),
- _captureWorkerThread(NULL),
- _renderWorkerThread(NULL),
- _captureWorkerThreadId(0),
- _renderWorkerThreadId(0),
- _id(id),
- _mixerManager(id),
- _inputDeviceIndex(0),
- _outputDeviceIndex(0),
- _inputDeviceID(kAudioObjectUnknown),
- _outputDeviceID(kAudioObjectUnknown),
- _inputDeviceIsSpecified(false),
- _outputDeviceIsSpecified(false),
- _recChannels(N_REC_CHANNELS),
- _playChannels(N_PLAY_CHANNELS),
- _captureBufData(NULL),
- _renderBufData(NULL),
- _playBufType(AudioDeviceModule::kFixedBufferSize),
- _initialized(false),
- _isShutDown(false),
- _recording(false),
- _playing(false),
- _recIsInitialized(false),
- _playIsInitialized(false),
- _startRec(false),
- _stopRec(false),
- _stopPlay(false),
- _AGC(false),
- _renderDeviceIsAlive(1),
- _captureDeviceIsAlive(1),
- _twoDevices(true),
- _doStop(false),
- _doStopRec(false),
- _macBookPro(false),
- _macBookProPanRight(false),
- _captureLatencyUs(0),
- _renderLatencyUs(0),
- _captureDelayUs(0),
- _renderDelayUs(0),
- _renderDelayOffsetSamples(0),
- _playBufDelayFixed(20),
- _playWarning(0),
- _playError(0),
- _recWarning(0),
- _recError(0),
- _paCaptureBuffer(NULL),
- _paRenderBuffer(NULL),
- _captureBufSizeSamples(0),
- _renderBufSizeSamples(0)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
- "%s created", __FUNCTION__);
-
- assert(&_stopEvent != NULL);
- assert(&_stopEventRec != NULL);
-
- memset(_renderConvertData, 0, sizeof(_renderConvertData));
- memset(&_outStreamFormat, 0, sizeof(AudioStreamBasicDescription));
- memset(&_outDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
- memset(&_inStreamFormat, 0, sizeof(AudioStreamBasicDescription));
- memset(&_inDesiredFormat, 0, sizeof(AudioStreamBasicDescription));
-}
-
-
-AudioDeviceMac::~AudioDeviceMac()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
- "%s destroyed", __FUNCTION__);
-
- if (!_isShutDown)
- {
- Terminate();
- }
-
- if (_captureWorkerThread)
- {
- delete _captureWorkerThread;
- _captureWorkerThread = NULL;
- }
-
- if (_renderWorkerThread)
- {
- delete _renderWorkerThread;
- _renderWorkerThread = NULL;
- }
-
- if (_paRenderBuffer)
- {
- delete _paRenderBuffer;
- _paRenderBuffer = NULL;
- }
-
- if (_paCaptureBuffer)
- {
- delete _paCaptureBuffer;
- _paCaptureBuffer = NULL;
- }
-
- if (_renderBufData)
- {
- delete[] _renderBufData;
- _renderBufData = NULL;
- }
-
- if (_captureBufData)
- {
- delete[] _captureBufData;
- _captureBufData = NULL;
- }
-
- kern_return_t kernErr = KERN_SUCCESS;
- kernErr = semaphore_destroy(mach_task_self(), _renderSemaphore);
- if (kernErr != KERN_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " semaphore_destroy() error: %d", kernErr);
- }
-
- kernErr = semaphore_destroy(mach_task_self(), _captureSemaphore);
- if (kernErr != KERN_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " semaphore_destroy() error: %d", kernErr);
- }
-
- delete &_stopEvent;
- delete &_stopEventRec;
- delete &_critSect;
-}
-
-// ============================================================================
-// API
-// ============================================================================
-
-void AudioDeviceMac::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- _ptrAudioBuffer = audioBuffer;
-
- // inform the AudioBuffer about default settings for this implementation
- _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
- _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
- _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
- _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
-}
-
-WebRtc_Word32 AudioDeviceMac::ActiveAudioLayer(
- AudioDeviceModule::AudioLayer& audioLayer) const
-{
- audioLayer = AudioDeviceModule::kPlatformDefaultAudio;
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::Init()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_initialized)
- {
- return 0;
- }
-
- OSStatus err = noErr;
-
- _isShutDown = false;
-
- // PortAudio ring buffers require an elementCount which is a power of two.
- if (_renderBufData == NULL)
- {
- UInt32 powerOfTwo = 1;
- while (powerOfTwo < PLAY_BUF_SIZE_IN_SAMPLES)
- {
- powerOfTwo <<= 1;
- }
- _renderBufSizeSamples = powerOfTwo;
- _renderBufData = new SInt16[_renderBufSizeSamples];
- }
-
- if (_paRenderBuffer == NULL)
- {
- _paRenderBuffer = new PaUtilRingBuffer;
- ring_buffer_size_t bufSize = -1;
- bufSize = PaUtil_InitializeRingBuffer(_paRenderBuffer, sizeof(SInt16),
- _renderBufSizeSamples,
- _renderBufData);
- if (bufSize == -1)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
- _id, " PaUtil_InitializeRingBuffer() error");
- return -1;
- }
- }
-
- if (_captureBufData == NULL)
- {
- UInt32 powerOfTwo = 1;
- while (powerOfTwo < REC_BUF_SIZE_IN_SAMPLES)
- {
- powerOfTwo <<= 1;
- }
- _captureBufSizeSamples = powerOfTwo;
- _captureBufData = new Float32[_captureBufSizeSamples];
- }
-
- if (_paCaptureBuffer == NULL)
- {
- _paCaptureBuffer = new PaUtilRingBuffer;
- ring_buffer_size_t bufSize = -1;
- bufSize = PaUtil_InitializeRingBuffer(_paCaptureBuffer,
- sizeof(Float32),
- _captureBufSizeSamples,
- _captureBufData);
- if (bufSize == -1)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
- _id, " PaUtil_InitializeRingBuffer() error");
- return -1;
- }
- }
-
- if (_renderWorkerThread == NULL)
- {
- _renderWorkerThread
- = ThreadWrapper::CreateThread(RunRender, this, kRealtimePriority,
- "RenderWorkerThread");
- if (_renderWorkerThread == NULL)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
- _id, " Render CreateThread() error");
- return -1;
- }
- }
-
- if (_captureWorkerThread == NULL)
- {
- _captureWorkerThread
- = ThreadWrapper::CreateThread(RunCapture, this, kRealtimePriority,
- "CaptureWorkerThread");
- if (_captureWorkerThread == NULL)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice,
- _id, " Capture CreateThread() error");
- return -1;
- }
- }
-
- kern_return_t kernErr = KERN_SUCCESS;
- kernErr = semaphore_create(mach_task_self(), &_renderSemaphore,
- SYNC_POLICY_FIFO, 0);
- if (kernErr != KERN_SUCCESS)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " semaphore_create() error: %d", kernErr);
- return -1;
- }
-
- kernErr = semaphore_create(mach_task_self(), &_captureSemaphore,
- SYNC_POLICY_FIFO, 0);
- if (kernErr != KERN_SUCCESS)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- " semaphore_create() error: %d", kernErr);
- return -1;
- }
-
- // Setting RunLoop to NULL here instructs HAL to manage its own thread for
- // notifications. This was the default behaviour on OS X 10.5 and earlier, but now
- // must be explicitly specified. HAL would otherwise try to use the main thread to
- // issue notifications.
- AudioObjectPropertyAddress propertyAddress = {
- kAudioHardwarePropertyRunLoop,
- kAudioObjectPropertyScopeGlobal,
- kAudioObjectPropertyElementMaster };
- CFRunLoopRef runLoop = NULL;
- UInt32 size = sizeof(CFRunLoopRef);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(kAudioObjectSystemObject,
- &propertyAddress, 0, NULL, size, &runLoop));
-
- // Listen for any device changes.
- propertyAddress.mSelector = kAudioHardwarePropertyDevices;
- WEBRTC_CA_LOG_ERR(AudioObjectAddPropertyListener(kAudioObjectSystemObject,
- &propertyAddress, &objectListenerProc, this));
-
- // Determine if this is a MacBook Pro
- _macBookPro = false;
- _macBookProPanRight = false;
- char buf[128];
- size_t length = sizeof(buf);
- memset(buf, 0, length);
-
- int intErr = sysctlbyname("hw.model", buf, &length, NULL, 0);
- if (intErr != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error in sysctlbyname(): %d", err);
- } else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Hardware model: %s", buf);
- if (strncmp(buf, "MacBookPro", 10) == 0)
- {
- _macBookPro = true;
- }
- }
-
- _playWarning = 0;
- _playError = 0;
- _recWarning = 0;
- _recError = 0;
-
- _initialized = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::Terminate()
-{
-
- if (!_initialized)
- {
- return 0;
- }
-
- if (_recording)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Recording must be stopped");
- return -1;
- }
-
- if (_playing)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Playback must be stopped");
- return -1;
- }
-
- _critSect.Enter();
-
- _mixerManager.Close();
-
- OSStatus err = noErr;
- int retVal = 0;
-
- AudioObjectPropertyAddress propertyAddress = {
- kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
- kAudioObjectPropertyElementMaster };
- WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(kAudioObjectSystemObject,
- &propertyAddress, &objectListenerProc, this));
-
- err = AudioHardwareUnload();
- if (err != noErr)
- {
- logCAMsg(kTraceError, kTraceAudioDevice, _id,
- "Error in AudioHardwareUnload()", (const char*) &err);
- retVal = -1;
- }
-
- _critSect.Leave();
-
- _isShutDown = true;
- _initialized = false;
- _outputDeviceIsSpecified = false;
- _inputDeviceIsSpecified = false;
-
- return retVal;
-}
-
-bool AudioDeviceMac::Initialized() const
-{
- return (_initialized);
-}
-
-WebRtc_Word32 AudioDeviceMac::SpeakerIsAvailable(bool& available)
-{
-
- bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
- // Make an attempt to open up the
- // output mixer corresponding to the currently selected output device.
- //
- if (!wasInitialized && InitSpeaker() == -1)
- {
- available = false;
- return 0;
- }
-
- // Given that InitSpeaker was successful, we know that a valid speaker exists
- //
- available = true;
-
- // Close the initialized output mixer
- //
- if (!wasInitialized)
- {
- _mixerManager.CloseSpeaker();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::InitSpeaker()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_playing)
- {
- return -1;
- }
-
- if (InitDevice(_outputDeviceIndex, _outputDeviceID, false) == -1)
- {
- return -1;
- }
-
- if (_inputDeviceID == _outputDeviceID)
- {
- _twoDevices = false;
- } else
- {
- _twoDevices = true;
- }
-
- if (_mixerManager.OpenSpeaker(_outputDeviceID) == -1)
- {
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::MicrophoneIsAvailable(bool& available)
-{
-
- bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
- // Make an attempt to open up the
- // input mixer corresponding to the currently selected output device.
- //
- if (!wasInitialized && InitMicrophone() == -1)
- {
- available = false;
- return 0;
- }
-
- // Given that InitMicrophone was successful, we know that a valid microphone exists
- //
- available = true;
-
- // Close the initialized input mixer
- //
- if (!wasInitialized)
- {
- _mixerManager.CloseMicrophone();
- }
-
- return 0;
-}
-
-
-WebRtc_Word32 AudioDeviceMac::InitMicrophone()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_recording)
- {
- return -1;
- }
-
- if (InitDevice(_inputDeviceIndex, _inputDeviceID, true) == -1)
- {
- return -1;
- }
-
- if (_inputDeviceID == _outputDeviceID)
- {
- _twoDevices = false;
- } else
- {
- _twoDevices = true;
- }
-
- if (_mixerManager.OpenMicrophone(_inputDeviceID) == -1)
- {
- return -1;
- }
-
- return 0;
-}
-
-bool AudioDeviceMac::SpeakerIsInitialized() const
-{
- return (_mixerManager.SpeakerIsInitialized());
-}
-
-bool AudioDeviceMac::MicrophoneIsInitialized() const
-{
- return (_mixerManager.MicrophoneIsInitialized());
-}
-
-WebRtc_Word32 AudioDeviceMac::SpeakerVolumeIsAvailable(bool& available)
-{
-
- bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
- // Make an attempt to open up the
- // output mixer corresponding to the currently selected output device.
- //
- if (!wasInitialized && InitSpeaker() == -1)
- {
- // If we end up here it means that the selected speaker has no volume
- // control.
- available = false;
- return 0;
- }
-
- // Given that InitSpeaker was successful, we know that a volume control exists
- //
- available = true;
-
- // Close the initialized output mixer
- //
- if (!wasInitialized)
- {
- _mixerManager.CloseSpeaker();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::SetSpeakerVolume(WebRtc_UWord32 volume)
-{
-
- return (_mixerManager.SetSpeakerVolume(volume));
-}
-
-WebRtc_Word32 AudioDeviceMac::SpeakerVolume(WebRtc_UWord32& volume) const
-{
-
- WebRtc_UWord32 level(0);
-
- if (_mixerManager.SpeakerVolume(level) == -1)
- {
- return -1;
- }
-
- volume = level;
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
- WebRtc_UWord16 volumeRight)
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32
-AudioDeviceMac::WaveOutVolume(WebRtc_UWord16& /*volumeLeft*/,
- WebRtc_UWord16& /*volumeRight*/) const
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceMac::MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const
-{
-
- WebRtc_UWord32 maxVol(0);
-
- if (_mixerManager.MaxSpeakerVolume(maxVol) == -1)
- {
- return -1;
- }
-
- maxVolume = maxVol;
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::MinSpeakerVolume(WebRtc_UWord32& minVolume) const
-{
-
- WebRtc_UWord32 minVol(0);
-
- if (_mixerManager.MinSpeakerVolume(minVol) == -1)
- {
- return -1;
- }
-
- minVolume = minVol;
- return 0;
-}
-
-WebRtc_Word32
-AudioDeviceMac::SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const
-{
-
- WebRtc_UWord16 delta(0);
-
- if (_mixerManager.SpeakerVolumeStepSize(delta) == -1)
- {
- return -1;
- }
-
- stepSize = delta;
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::SpeakerMuteIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
- bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
- // Make an attempt to open up the
- // output mixer corresponding to the currently selected output device.
- //
- if (!wasInitialized && InitSpeaker() == -1)
- {
- // If we end up here it means that the selected speaker has no volume
- // control, hence it is safe to state that there is no mute control
- // already at this stage.
- available = false;
- return 0;
- }
-
- // Check if the selected speaker has a mute control
- //
- _mixerManager.SpeakerMuteIsAvailable(isAvailable);
-
- available = isAvailable;
-
- // Close the initialized output mixer
- //
- if (!wasInitialized)
- {
- _mixerManager.CloseSpeaker();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::SetSpeakerMute(bool enable)
-{
- return (_mixerManager.SetSpeakerMute(enable));
-}
-
-WebRtc_Word32 AudioDeviceMac::SpeakerMute(bool& enabled) const
-{
-
- bool muted(0);
-
- if (_mixerManager.SpeakerMute(muted) == -1)
- {
- return -1;
- }
-
- enabled = muted;
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::MicrophoneMuteIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
- bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
- // Make an attempt to open up the
- // input mixer corresponding to the currently selected input device.
- //
- if (!wasInitialized && InitMicrophone() == -1)
- {
- // If we end up here it means that the selected microphone has no volume
- // control, hence it is safe to state that there is no boost control
- // already at this stage.
- available = false;
- return 0;
- }
-
- // Check if the selected microphone has a mute control
- //
- _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized input mixer
- //
- if (!wasInitialized)
- {
- _mixerManager.CloseMicrophone();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::SetMicrophoneMute(bool enable)
-{
- return (_mixerManager.SetMicrophoneMute(enable));
-}
-
-WebRtc_Word32 AudioDeviceMac::MicrophoneMute(bool& enabled) const
-{
-
- bool muted(0);
-
- if (_mixerManager.MicrophoneMute(muted) == -1)
- {
- return -1;
- }
-
- enabled = muted;
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::MicrophoneBoostIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
- bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
- // Enumerate all avaliable microphone and make an attempt to open up the
- // input mixer corresponding to the currently selected input device.
- //
- if (!wasInitialized && InitMicrophone() == -1)
- {
- // If we end up here it means that the selected microphone has no volume
- // control, hence it is safe to state that there is no boost control
- // already at this stage.
- available = false;
- return 0;
- }
-
- // Check if the selected microphone has a boost control
- //
- _mixerManager.MicrophoneBoostIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized input mixer
- //
- if (!wasInitialized)
- {
- _mixerManager.CloseMicrophone();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::SetMicrophoneBoost(bool enable)
-{
-
- return (_mixerManager.SetMicrophoneBoost(enable));
-}
-
-WebRtc_Word32 AudioDeviceMac::MicrophoneBoost(bool& enabled) const
-{
-
- bool onOff(0);
-
- if (_mixerManager.MicrophoneBoost(onOff) == -1)
- {
- return -1;
- }
-
- enabled = onOff;
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::StereoRecordingIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
- bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
- if (!wasInitialized && InitMicrophone() == -1)
- {
- // Cannot open the specified device
- available = false;
- return 0;
- }
-
- // Check if the selected microphone can record stereo
- //
- _mixerManager.StereoRecordingIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized input mixer
- //
- if (!wasInitialized)
- {
- _mixerManager.CloseMicrophone();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::SetStereoRecording(bool enable)
-{
-
- if (enable)
- _recChannels = 2;
- else
- _recChannels = 1;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::StereoRecording(bool& enabled) const
-{
-
- if (_recChannels == 2)
- enabled = true;
- else
- enabled = false;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::StereoPlayoutIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
- bool wasInitialized = _mixerManager.SpeakerIsInitialized();
-
- if (!wasInitialized && InitSpeaker() == -1)
- {
- // Cannot open the specified device
- available = false;
- return 0;
- }
-
- // Check if the selected microphone can record stereo
- //
- _mixerManager.StereoPlayoutIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized input mixer
- //
- if (!wasInitialized)
- {
- _mixerManager.CloseSpeaker();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::SetStereoPlayout(bool enable)
-{
-
- if (enable)
- _playChannels = 2;
- else
- _playChannels = 1;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::StereoPlayout(bool& enabled) const
-{
-
- if (_playChannels == 2)
- enabled = true;
- else
- enabled = false;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::SetAGC(bool enable)
-{
-
- _AGC = enable;
-
- return 0;
-}
-
-bool AudioDeviceMac::AGC() const
-{
-
- return _AGC;
-}
-
-WebRtc_Word32 AudioDeviceMac::MicrophoneVolumeIsAvailable(bool& available)
-{
-
- bool wasInitialized = _mixerManager.MicrophoneIsInitialized();
-
- // Make an attempt to open up the
- // input mixer corresponding to the currently selected output device.
- //
- if (!wasInitialized && InitMicrophone() == -1)
- {
- // If we end up here it means that the selected microphone has no volume
- // control.
- available = false;
- return 0;
- }
-
- // Given that InitMicrophone was successful, we know that a volume control
- // exists
- //
- available = true;
-
- // Close the initialized input mixer
- //
- if (!wasInitialized)
- {
- _mixerManager.CloseMicrophone();
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::SetMicrophoneVolume(WebRtc_UWord32 volume)
-{
-
- return (_mixerManager.SetMicrophoneVolume(volume));
-}
-
-WebRtc_Word32 AudioDeviceMac::MicrophoneVolume(WebRtc_UWord32& volume) const
-{
-
- WebRtc_UWord32 level(0);
-
- if (_mixerManager.MicrophoneVolume(level) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " failed to retrive current microphone level");
- return -1;
- }
-
- volume = level;
- return 0;
-}
-
-WebRtc_Word32
-AudioDeviceMac::MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const
-{
-
- WebRtc_UWord32 maxVol(0);
-
- if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1)
- {
- return -1;
- }
-
- maxVolume = maxVol;
- return 0;
-}
-
-WebRtc_Word32
-AudioDeviceMac::MinMicrophoneVolume(WebRtc_UWord32& minVolume) const
-{
-
- WebRtc_UWord32 minVol(0);
-
- if (_mixerManager.MinMicrophoneVolume(minVol) == -1)
- {
- return -1;
- }
-
- minVolume = minVol;
- return 0;
-}
-
-WebRtc_Word32
-AudioDeviceMac::MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const
-{
-
- WebRtc_UWord16 delta(0);
-
- if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1)
- {
- return -1;
- }
-
- stepSize = delta;
- return 0;
-}
-
-WebRtc_Word16 AudioDeviceMac::PlayoutDevices()
-{
-
- AudioDeviceID playDevices[MaxNumberDevices];
- return GetNumberDevices(kAudioDevicePropertyScopeOutput, playDevices,
- MaxNumberDevices);
-}
-
-WebRtc_Word32 AudioDeviceMac::SetPlayoutDevice(WebRtc_UWord16 index)
-{
-
- if (_playIsInitialized)
- {
- return -1;
- }
-
- AudioDeviceID playDevices[MaxNumberDevices];
- WebRtc_UWord32 nDevices = GetNumberDevices(kAudioDevicePropertyScopeOutput,
- playDevices, MaxNumberDevices);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " number of availiable waveform-audio output devices is %u",
- nDevices);
-
- if (index > (nDevices - 1))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " device index is out of range [0,%u]", (nDevices - 1));
- return -1;
- }
-
- _outputDeviceIndex = index;
- _outputDeviceIsSpecified = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::SetPlayoutDevice(
- AudioDeviceModule::WindowsDeviceType /*device*/)
-{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "WindowsDeviceType not supported");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceMac::PlayoutDeviceName(
- WebRtc_UWord16 index,
- WebRtc_Word8 name[kAdmMaxDeviceNameSize],
- WebRtc_Word8 guid[kAdmMaxGuidSize])
-{
-
- const WebRtc_UWord16 nDevices(PlayoutDevices());
-
- if ((index > (nDevices - 1)) || (name == NULL))
- {
- return -1;
- }
-
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- return GetDeviceName(kAudioDevicePropertyScopeOutput, index, name);
-}
-
-WebRtc_Word32 AudioDeviceMac::RecordingDeviceName(
- WebRtc_UWord16 index,
- WebRtc_Word8 name[kAdmMaxDeviceNameSize],
- WebRtc_Word8 guid[kAdmMaxGuidSize])
-{
-
- const WebRtc_UWord16 nDevices(RecordingDevices());
-
- if ((index > (nDevices - 1)) || (name == NULL))
- {
- return -1;
- }
-
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- return GetDeviceName(kAudioDevicePropertyScopeInput, index, name);
-}
-
-WebRtc_Word16 AudioDeviceMac::RecordingDevices()
-{
-
- AudioDeviceID recDevices[MaxNumberDevices];
- return GetNumberDevices(kAudioDevicePropertyScopeInput, recDevices,
- MaxNumberDevices);
-}
-
-WebRtc_Word32 AudioDeviceMac::SetRecordingDevice(WebRtc_UWord16 index)
-{
-
- if (_recIsInitialized)
- {
- return -1;
- }
-
- AudioDeviceID recDevices[MaxNumberDevices];
- WebRtc_UWord32 nDevices = GetNumberDevices(kAudioDevicePropertyScopeInput,
- recDevices, MaxNumberDevices);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " number of availiable waveform-audio input devices is %u",
- nDevices);
-
- if (index > (nDevices - 1))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " device index is out of range [0,%u]", (nDevices - 1));
- return -1;
- }
-
- _inputDeviceIndex = index;
- _inputDeviceIsSpecified = true;
-
- return 0;
-}
-
-
-WebRtc_Word32
-AudioDeviceMac::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType /*device*/)
-{
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "WindowsDeviceType not supported");
- return -1;
-}
-
-WebRtc_Word32 AudioDeviceMac::PlayoutIsAvailable(bool& available)
-{
-
- available = true;
-
- // Try to initialize the playout side
- if (InitPlayout() == -1)
- {
- available = false;
- }
-
- // We destroy the IOProc created by InitPlayout() in implDeviceIOProc().
- // We must actually start playout here in order to have the IOProc
- // deleted by calling StopPlayout().
- if (StartPlayout() == -1)
- {
- available = false;
- }
-
- // Cancel effect of initialization
- if (StopPlayout() == -1)
- {
- available = false;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::RecordingIsAvailable(bool& available)
-{
-
- available = true;
-
- // Try to initialize the recording side
- if (InitRecording() == -1)
- {
- available = false;
- }
-
- // We destroy the IOProc created by InitRecording() in implInDeviceIOProc().
- // We must actually start recording here in order to have the IOProc
- // deleted by calling StopRecording().
- if (StartRecording() == -1)
- {
- available = false;
- }
-
- // Cancel effect of initialization
- if (StopRecording() == -1)
- {
- available = false;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::InitPlayout()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_playing)
- {
- return -1;
- }
-
- if (!_outputDeviceIsSpecified)
- {
- return -1;
- }
-
- if (_playIsInitialized)
- {
- return 0;
- }
-
- // Initialize the speaker (devices might have been added or removed)
- if (InitSpeaker() == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " InitSpeaker() failed");
- }
-
- if (!MicrophoneIsInitialized())
- {
- // Make this call to check if we are using
- // one or two devices (_twoDevices)
- bool available = false;
- if (MicrophoneIsAvailable(available) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " MicrophoneIsAvailable() failed");
- }
- }
-
- PaUtil_FlushRingBuffer(_paRenderBuffer);
-
- OSStatus err = noErr;
- UInt32 size = 0;
- _renderDelayOffsetSamples = 0;
- _renderDelayUs = 0;
- _renderLatencyUs = 0;
- _renderDeviceIsAlive = 1;
- _doStop = false;
-
- // The internal microphone of a MacBook Pro is located under the left speaker
- // grille. When the internal speakers are in use, we want to fully stereo
- // pan to the right.
- AudioObjectPropertyAddress
- propertyAddress = { kAudioDevicePropertyDataSource,
- kAudioDevicePropertyScopeOutput, 0 };
- if (_macBookPro)
- {
- _macBookProPanRight = false;
- Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
- &propertyAddress);
- if (hasProperty)
- {
- UInt32 dataSource = 0;
- size = sizeof(dataSource);
- WEBRTC_CA_LOG_WARN(AudioObjectGetPropertyData(_outputDeviceID,
- &propertyAddress, 0, NULL, &size, &dataSource));
-
- if (dataSource == 'ispk')
- {
- _macBookProPanRight = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
- _id,
- "MacBook Pro using internal speakers; stereo"
- " panning right");
- } else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice,
- _id, "MacBook Pro not using internal speakers");
- }
-
- // Add a listener to determine if the status changes.
- WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_outputDeviceID,
- &propertyAddress, &objectListenerProc, this));
- }
- }
-
- // Get current stream description
- propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
- memset(&_outStreamFormat, 0, sizeof(_outStreamFormat));
- size = sizeof(_outStreamFormat);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
- &propertyAddress, 0, NULL, &size, &_outStreamFormat));
-
- if (_outStreamFormat.mFormatID != kAudioFormatLinearPCM)
- {
- logCAMsg(kTraceError, kTraceAudioDevice, _id,
- "Unacceptable output stream format -> mFormatID",
- (const char *) &_outStreamFormat.mFormatID);
- return -1;
- }
-
- if (_outStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Too many channels on device -> mChannelsPerFrame = %d",
- _outStreamFormat.mChannelsPerFrame);
- return -1;
- }
-
- if (_outStreamFormat.mFormatFlags & kAudioFormatFlagIsNonInterleaved)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Non-interleaved audio data is not supported.",
- "AudioHardware streams should not have this format.");
- return -1;
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Ouput stream format:");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "mSampleRate = %f, mChannelsPerFrame = %u",
- _outStreamFormat.mSampleRate,
- _outStreamFormat.mChannelsPerFrame);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "mBytesPerPacket = %u, mFramesPerPacket = %u",
- _outStreamFormat.mBytesPerPacket,
- _outStreamFormat.mFramesPerPacket);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "mBytesPerFrame = %u, mBitsPerChannel = %u",
- _outStreamFormat.mBytesPerFrame,
- _outStreamFormat.mBitsPerChannel);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "mFormatFlags = %u, mChannelsPerFrame = %u",
- _outStreamFormat.mFormatFlags,
- _outStreamFormat.mChannelsPerFrame);
- logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
- (const char *) &_outStreamFormat.mFormatID);
-
- // Our preferred format to work with
- _outDesiredFormat.mSampleRate = N_PLAY_SAMPLES_PER_SEC;
- if (_outStreamFormat.mChannelsPerFrame >= 2 && (_playChannels == 2))
- {
- _outDesiredFormat.mChannelsPerFrame = 2;
- } else
- {
- // Disable stereo playout when we only have one channel on the device.
- _outDesiredFormat.mChannelsPerFrame = 1;
- _playChannels = 1;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Stereo playout unavailable on this device");
- }
-
- if (_ptrAudioBuffer)
- {
- // Update audio buffer with the selected parameters
- _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
- _ptrAudioBuffer->SetPlayoutChannels((WebRtc_UWord8) _playChannels);
- }
-
- _renderDelayOffsetSamples = _renderBufSizeSamples - N_BUFFERS_OUT
- * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * _outDesiredFormat.mChannelsPerFrame;
-
- _outDesiredFormat.mBytesPerPacket = _outDesiredFormat.mChannelsPerFrame
- * sizeof(SInt16);
- _outDesiredFormat.mFramesPerPacket = 1; // In uncompressed audio,
- // a packet is one frame.
- _outDesiredFormat.mBytesPerFrame = _outDesiredFormat.mChannelsPerFrame
- * sizeof(SInt16);
- _outDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
-
- _outDesiredFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger
- | kLinearPCMFormatFlagIsPacked;
-#ifdef WEBRTC_BIG_ENDIAN
- _outDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
-#endif
- _outDesiredFormat.mFormatID = kAudioFormatLinearPCM;
-
- WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_outDesiredFormat, &_outStreamFormat,
- &_renderConverter));
-
- // First try to set buffer size to desired value (_playBufDelayFixed)
- UInt32 bufByteCount = (UInt32)((_outStreamFormat.mSampleRate / 1000.0)
- * _playBufDelayFixed * _outStreamFormat.mChannelsPerFrame
- * sizeof(Float32));
- if (_outStreamFormat.mFramesPerPacket != 0)
- {
- if (bufByteCount % _outStreamFormat.mFramesPerPacket != 0)
- {
- bufByteCount = ((UInt32)(bufByteCount
- / _outStreamFormat.mFramesPerPacket) + 1)
- * _outStreamFormat.mFramesPerPacket;
- }
- }
-
- // Ensure the buffer size is within the acceptable range provided by the device.
- propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
- AudioValueRange range;
- size = sizeof(range);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
- &propertyAddress, 0, NULL, &size, &range));
- if (range.mMinimum > bufByteCount)
- {
- bufByteCount = range.mMinimum;
- } else if (range.mMaximum < bufByteCount)
- {
- bufByteCount = range.mMaximum;
- }
-
- propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
- size = sizeof(bufByteCount);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
- &propertyAddress, 0, NULL, size, &bufByteCount));
-
- // Get render device latency
- propertyAddress.mSelector = kAudioDevicePropertyLatency;
- UInt32 latency = 0;
- size = sizeof(UInt32);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
- &propertyAddress, 0, NULL, &size, &latency));
- _renderLatencyUs = (WebRtc_UWord32) ((1.0e6 * latency)
- / _outStreamFormat.mSampleRate);
-
- // Get render stream latency
- propertyAddress.mSelector = kAudioDevicePropertyStreams;
- AudioStreamID stream = 0;
- size = sizeof(AudioStreamID);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
- &propertyAddress, 0, NULL, &size, &stream));
- propertyAddress.mSelector = kAudioStreamPropertyLatency;
- size = sizeof(UInt32);
- latency = 0;
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
- &propertyAddress, 0, NULL, &size, &latency));
- _renderLatencyUs += (WebRtc_UWord32) ((1.0e6 * latency)
- / _outStreamFormat.mSampleRate);
-
- // Listen for format changes
- propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(_outputDeviceID,
- &propertyAddress, &objectListenerProc, this));
-
- // Listen for processor overloads
- propertyAddress.mSelector = kAudioDeviceProcessorOverload;
- WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_outputDeviceID,
- &propertyAddress, &objectListenerProc, this));
-
- if (_twoDevices || !_recIsInitialized)
- {
- WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_outputDeviceID,
- deviceIOProc, this, &_deviceIOProcID));
- }
-
- // Mark playout side as initialized
- _playIsInitialized = true;
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " initial playout status: _renderDelayOffsetSamples=%d,"
- " _renderDelayUs=%d, _renderLatencyUs=%d",
- _renderDelayOffsetSamples, _renderDelayUs, _renderLatencyUs);
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::InitRecording()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_recording)
- {
- return -1;
- }
-
- if (!_inputDeviceIsSpecified)
- {
- return -1;
- }
-
- if (_recIsInitialized)
- {
- return 0;
- }
-
- // Initialize the microphone (devices might have been added or removed)
- if (InitMicrophone() == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " InitMicrophone() failed");
- }
-
- if (!SpeakerIsInitialized())
- {
- // Make this call to check if we are using
- // one or two devices (_twoDevices)
- bool available = false;
- if (SpeakerIsAvailable(available) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " SpeakerIsAvailable() failed");
- }
- }
-
- OSStatus err = noErr;
- UInt32 size = 0;
-
- PaUtil_FlushRingBuffer(_paCaptureBuffer);
-
- _captureDelayUs = 0;
- _captureLatencyUs = 0;
- _captureDeviceIsAlive = 1;
- _doStopRec = false;
-
- // Get current stream description
- AudioObjectPropertyAddress
- propertyAddress = { kAudioDevicePropertyStreamFormat,
- kAudioDevicePropertyScopeInput, 0 };
- memset(&_inStreamFormat, 0, sizeof(_inStreamFormat));
- size = sizeof(_inStreamFormat);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
- &propertyAddress, 0, NULL, &size, &_inStreamFormat));
-
- if (_inStreamFormat.mFormatID != kAudioFormatLinearPCM)
- {
- logCAMsg(kTraceError, kTraceAudioDevice, _id,
- "Unacceptable input stream format -> mFormatID",
- (const char *) &_inStreamFormat.mFormatID);
- return -1;
- }
-
- if (_inStreamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- ", Too many channels on device (mChannelsPerFrame = %d)",
- _inStreamFormat.mChannelsPerFrame);
- return -1;
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Input stream format:");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " mSampleRate = %f, mChannelsPerFrame = %u",
- _inStreamFormat.mSampleRate, _inStreamFormat.mChannelsPerFrame);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " mBytesPerPacket = %u, mFramesPerPacket = %u",
- _inStreamFormat.mBytesPerPacket,
- _inStreamFormat.mFramesPerPacket);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " mBytesPerFrame = %u, mBitsPerChannel = %u",
- _inStreamFormat.mBytesPerFrame,
- _inStreamFormat.mBitsPerChannel);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " mFormatFlags = %u, mChannelsPerFrame = %u",
- _inStreamFormat.mFormatFlags,
- _inStreamFormat.mChannelsPerFrame);
- logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
- (const char *) &_inStreamFormat.mFormatID);
-
- // Our preferred format to work with
- if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2))
- {
- _inDesiredFormat.mChannelsPerFrame = 2;
- } else
- {
- // Disable stereo recording when we only have one channel on the device.
- _inDesiredFormat.mChannelsPerFrame = 1;
- _recChannels = 1;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Stereo recording unavailable on this device");
- }
-
- if (_ptrAudioBuffer)
- {
- // Update audio buffer with the selected parameters
- _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
- _ptrAudioBuffer->SetRecordingChannels((WebRtc_UWord8) _recChannels);
- }
-
- _inDesiredFormat.mSampleRate = N_REC_SAMPLES_PER_SEC;
- _inDesiredFormat.mBytesPerPacket = _inDesiredFormat.mChannelsPerFrame
- * sizeof(SInt16);
- _inDesiredFormat.mFramesPerPacket = 1;
- _inDesiredFormat.mBytesPerFrame = _inDesiredFormat.mChannelsPerFrame
- * sizeof(SInt16);
- _inDesiredFormat.mBitsPerChannel = sizeof(SInt16) * 8;
-
- _inDesiredFormat.mFormatFlags = kLinearPCMFormatFlagIsSignedInteger
- | kLinearPCMFormatFlagIsPacked;
-#ifdef WEBRTC_BIG_ENDIAN
- _inDesiredFormat.mFormatFlags |= kLinearPCMFormatFlagIsBigEndian;
-#endif
- _inDesiredFormat.mFormatID = kAudioFormatLinearPCM;
-
- WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_inStreamFormat, &_inDesiredFormat,
- &_captureConverter));
-
- // First try to set buffer size to desired value (10 ms * N_BLOCKS_IO)
- // TODO(xians): investigate this block.
- UInt32 bufByteCount = (UInt32)((_inStreamFormat.mSampleRate / 1000.0)
- * 10.0 * N_BLOCKS_IO * _inStreamFormat.mChannelsPerFrame
- * sizeof(Float32));
- if (_inStreamFormat.mFramesPerPacket != 0)
- {
- if (bufByteCount % _inStreamFormat.mFramesPerPacket != 0)
- {
- bufByteCount = ((UInt32)(bufByteCount
- / _inStreamFormat.mFramesPerPacket) + 1)
- * _inStreamFormat.mFramesPerPacket;
- }
- }
-
- // Ensure the buffer size is within the acceptable range provided by the device.
- propertyAddress.mSelector = kAudioDevicePropertyBufferSizeRange;
- AudioValueRange range;
- size = sizeof(range);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
- &propertyAddress, 0, NULL, &size, &range));
- if (range.mMinimum > bufByteCount)
- {
- bufByteCount = range.mMinimum;
- } else if (range.mMaximum < bufByteCount)
- {
- bufByteCount = range.mMaximum;
- }
-
- propertyAddress.mSelector = kAudioDevicePropertyBufferSize;
- size = sizeof(bufByteCount);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
- &propertyAddress, 0, NULL, size, &bufByteCount));
-
- // Get capture device latency
- propertyAddress.mSelector = kAudioDevicePropertyLatency;
- UInt32 latency = 0;
- size = sizeof(UInt32);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
- &propertyAddress, 0, NULL, &size, &latency));
- _captureLatencyUs = (UInt32)((1.0e6 * latency)
- / _inStreamFormat.mSampleRate);
-
- // Get capture stream latency
- propertyAddress.mSelector = kAudioDevicePropertyStreams;
- AudioStreamID stream = 0;
- size = sizeof(AudioStreamID);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
- &propertyAddress, 0, NULL, &size, &stream));
- propertyAddress.mSelector = kAudioStreamPropertyLatency;
- size = sizeof(UInt32);
- latency = 0;
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
- &propertyAddress, 0, NULL, &size, &latency));
- _captureLatencyUs += (UInt32)((1.0e6 * latency)
- / _inStreamFormat.mSampleRate);
-
- // Listen for format changes
- // TODO(xians): should we be using kAudioDevicePropertyDeviceHasChanged?
- propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectAddPropertyListener(_inputDeviceID,
- &propertyAddress, &objectListenerProc, this));
-
- // Listen for processor overloads
- propertyAddress.mSelector = kAudioDeviceProcessorOverload;
- WEBRTC_CA_LOG_WARN(AudioObjectAddPropertyListener(_inputDeviceID,
- &propertyAddress, &objectListenerProc, this));
-
- if (_twoDevices)
- {
- WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_inputDeviceID,
- inDeviceIOProc, this, &_inDeviceIOProcID));
- } else if (!_playIsInitialized)
- {
- WEBRTC_CA_RETURN_ON_ERR(AudioDeviceCreateIOProcID(_inputDeviceID,
- deviceIOProc, this, &_deviceIOProcID));
- }
-
- // Mark recording side as initialized
- _recIsInitialized = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::StartRecording()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_recIsInitialized)
- {
- return -1;
- }
-
- if (_recording)
- {
- return 0;
- }
-
- if (!_initialized)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Recording worker thread has not been started");
- return -1;
- }
-
- OSStatus err = noErr;
-
- unsigned int threadID(0);
- if (_captureWorkerThread != NULL)
- {
- _captureWorkerThread->Start(threadID);
- }
- _captureWorkerThreadId = threadID;
-
- if (_twoDevices)
- {
- WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_inputDeviceID, _inDeviceIOProcID));
- } else if (!_playing)
- {
- WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_inputDeviceID, _deviceIOProcID));
- }
-
- _recording = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::StopRecording()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_recIsInitialized)
- {
- return 0;
- }
-
- OSStatus err = noErr;
-
- // Stop device
- int32_t captureDeviceIsAlive = AtomicGet32(&_captureDeviceIsAlive);
- if (_twoDevices)
- {
- if (_recording && captureDeviceIsAlive == 1)
- {
- _recording = false;
- _doStopRec = true; // Signal to io proc to stop audio device
- _critSect.Leave(); // Cannot be under lock, risk of deadlock
- if (kEventTimeout == _stopEventRec.Wait(2000))
- {
- CriticalSectionScoped critScoped(_critSect);
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Timed out stopping the capture IOProc. "
- "We may have failed to detect a device removal.");
-
- WEBRTC_CA_LOG_WARN(AudioDeviceStop(_inputDeviceID,
- _inDeviceIOProcID));
- WEBRTC_CA_LOG_WARN(
- AudioDeviceDestroyIOProcID(_inputDeviceID,
- _inDeviceIOProcID));
- }
- _critSect.Enter();
- _doStopRec = false;
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " Recording stopped");
- }
- }
- else
- {
- // We signal a stop for a shared device even when rendering has
- // not yet ended. This is to ensure the IOProc will return early as
- // intended (by checking |_recording|) before accessing
- // resources we free below (e.g. the capture converter).
- //
- // In the case of a shared devcie, the IOProc will verify
- // rendering has ended before stopping itself.
- if (_recording && captureDeviceIsAlive == 1)
- {
- _recording = false;
- _doStop = true; // Signal to io proc to stop audio device
- _critSect.Leave(); // Cannot be under lock, risk of deadlock
- if (kEventTimeout == _stopEvent.Wait(2000))
- {
- CriticalSectionScoped critScoped(_critSect);
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Timed out stopping the shared IOProc. "
- "We may have failed to detect a device removal.");
-
- // We assume rendering on a shared device has stopped as well if
- // the IOProc times out.
- WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID,
- _deviceIOProcID));
- WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
- _deviceIOProcID));
- }
- _critSect.Enter();
- _doStop = false;
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- " Recording stopped (shared)");
- }
- }
-
- // Setting this signal will allow the worker thread to be stopped.
- AtomicSet32(&_captureDeviceIsAlive, 0);
- _critSect.Leave();
- if (_captureWorkerThread != NULL)
- {
- if (!_captureWorkerThread->Stop())
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Timed out waiting for the render worker thread to "
- "stop.");
- }
- }
- _critSect.Enter();
-
- WEBRTC_CA_LOG_WARN(AudioConverterDispose(_captureConverter));
-
- // Remove listeners.
- AudioObjectPropertyAddress
- propertyAddress = { kAudioDevicePropertyStreamFormat,
- kAudioDevicePropertyScopeInput, 0 };
- WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_inputDeviceID,
- &propertyAddress, &objectListenerProc, this));
-
- propertyAddress.mSelector = kAudioDeviceProcessorOverload;
- WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_inputDeviceID,
- &propertyAddress, &objectListenerProc, this));
-
- _recIsInitialized = false;
- _recording = false;
-
- return 0;
-}
-
-bool AudioDeviceMac::RecordingIsInitialized() const
-{
- return (_recIsInitialized);
-}
-
-bool AudioDeviceMac::Recording() const
-{
- return (_recording);
-}
-
-bool AudioDeviceMac::PlayoutIsInitialized() const
-{
- return (_playIsInitialized);
-}
-
-WebRtc_Word32 AudioDeviceMac::StartPlayout()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_playIsInitialized)
- {
- return -1;
- }
-
- if (_playing)
- {
- return 0;
- }
-
- OSStatus err = noErr;
-
- unsigned int threadID(0);
- if (_renderWorkerThread != NULL)
- {
- _renderWorkerThread->Start(threadID);
- }
- _renderWorkerThreadId = threadID;
-
- if (_twoDevices || !_recording)
- {
- WEBRTC_CA_RETURN_ON_ERR(AudioDeviceStart(_outputDeviceID, _deviceIOProcID));
- }
- _playing = true;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::StopPlayout()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_playIsInitialized)
- {
- return 0;
- }
-
- OSStatus err = noErr;
-
- int32_t renderDeviceIsAlive = AtomicGet32(&_renderDeviceIsAlive);
- if (_playing && renderDeviceIsAlive == 1)
- {
- // We signal a stop for a shared device even when capturing has not
- // yet ended. This is to ensure the IOProc will return early as
- // intended (by checking |_playing|) before accessing resources we
- // free below (e.g. the render converter).
- //
- // In the case of a shared device, the IOProc will verify capturing
- // has ended before stopping itself.
- _playing = false;
- _doStop = true; // Signal to io proc to stop audio device
- _critSect.Leave(); // Cannot be under lock, risk of deadlock
- if (kEventTimeout == _stopEvent.Wait(2000))
- {
- CriticalSectionScoped critScoped(_critSect);
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Timed out stopping the render IOProc. "
- "We may have failed to detect a device removal.");
-
- // We assume capturing on a shared device has stopped as well if the
- // IOProc times out.
- WEBRTC_CA_LOG_WARN(AudioDeviceStop(_outputDeviceID,
- _deviceIOProcID));
- WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
- _deviceIOProcID));
- }
- _critSect.Enter();
- _doStop = false;
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- "Playout stopped");
- }
-
- // Setting this signal will allow the worker thread to be stopped.
- AtomicSet32(&_renderDeviceIsAlive, 0);
- _critSect.Leave();
- if (_renderWorkerThread != NULL)
- {
- if (!_renderWorkerThread->Stop())
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Timed out waiting for the render worker thread to "
- "stop.");
- }
- }
- _critSect.Enter();
-
- WEBRTC_CA_LOG_WARN(AudioConverterDispose(_renderConverter));
-
- // Remove listeners.
- AudioObjectPropertyAddress propertyAddress = {
- kAudioDevicePropertyStreamFormat, kAudioDevicePropertyScopeOutput,
- 0 };
- WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
- &propertyAddress, &objectListenerProc, this));
-
- propertyAddress.mSelector = kAudioDeviceProcessorOverload;
- WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
- &propertyAddress, &objectListenerProc, this));
-
- if (_macBookPro)
- {
- Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
- &propertyAddress);
- if (hasProperty)
- {
- propertyAddress.mSelector = kAudioDevicePropertyDataSource;
- WEBRTC_CA_LOG_WARN(AudioObjectRemovePropertyListener(_outputDeviceID,
- &propertyAddress, &objectListenerProc, this));
- }
- }
-
- _playIsInitialized = false;
- _playing = false;
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::PlayoutDelay(WebRtc_UWord16& delayMS) const
-{
- int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
- delayMS = static_cast<WebRtc_UWord16> (1e-3 * (renderDelayUs
- + _renderLatencyUs) + 0.5);
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::RecordingDelay(WebRtc_UWord16& delayMS) const
-{
- int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
- delayMS = static_cast<WebRtc_UWord16> (1e-3 * (captureDelayUs
- + _captureLatencyUs) + 0.5);
- return 0;
-}
-
-bool AudioDeviceMac::Playing() const
-{
- return (_playing);
-}
-
-WebRtc_Word32 AudioDeviceMac::SetPlayoutBuffer(
- const AudioDeviceModule::BufferType type,
- WebRtc_UWord16 sizeMS)
-{
-
- if (type != AudioDeviceModule::kFixedBufferSize)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Adaptive buffer size not supported on this platform");
- return -1;
- }
-
- _playBufType = type;
- _playBufDelayFixed = sizeMS;
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::PlayoutBuffer(
- AudioDeviceModule::BufferType& type,
- WebRtc_UWord16& sizeMS) const
-{
-
- type = _playBufType;
- sizeMS = _playBufDelayFixed;
-
- return 0;
-}
-
-// Not implemented for Mac.
-WebRtc_Word32 AudioDeviceMac::CPULoad(WebRtc_UWord16& /*load*/) const
-{
-
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " API call not supported on this platform");
-
- return -1;
-}
-
-bool AudioDeviceMac::PlayoutWarning() const
-{
- return (_playWarning > 0);
-}
-
-bool AudioDeviceMac::PlayoutError() const
-{
- return (_playError > 0);
-}
-
-bool AudioDeviceMac::RecordingWarning() const
-{
- return (_recWarning > 0);
-}
-
-bool AudioDeviceMac::RecordingError() const
-{
- return (_recError > 0);
-}
-
-void AudioDeviceMac::ClearPlayoutWarning()
-{
- _playWarning = 0;
-}
-
-void AudioDeviceMac::ClearPlayoutError()
-{
- _playError = 0;
-}
-
-void AudioDeviceMac::ClearRecordingWarning()
-{
- _recWarning = 0;
-}
-
-void AudioDeviceMac::ClearRecordingError()
-{
- _recError = 0;
-}
-
-// ============================================================================
-// Private Methods
-// ============================================================================
-
-WebRtc_Word32
-AudioDeviceMac::GetNumberDevices(const AudioObjectPropertyScope scope,
- AudioDeviceID scopedDeviceIds[],
- const WebRtc_UWord32 deviceListLength)
-{
- OSStatus err = noErr;
-
- AudioObjectPropertyAddress propertyAddress = {
- kAudioHardwarePropertyDevices, kAudioObjectPropertyScopeGlobal,
- kAudioObjectPropertyElementMaster };
- UInt32 size = 0;
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyDataSize(kAudioObjectSystemObject,
- &propertyAddress, 0, NULL, &size));
- if (size == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "No devices");
- return 0;
- }
-
- AudioDeviceID* deviceIds = (AudioDeviceID*) malloc(size);
- UInt32 numberDevices = size / sizeof(AudioDeviceID);
- AudioBufferList* bufferList = NULL;
- UInt32 numberScopedDevices = 0;
-
- // First check if there is a default device and list it
- UInt32 hardwareProperty = 0;
- if (scope == kAudioDevicePropertyScopeOutput)
- {
- hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
- } else
- {
- hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
- }
-
- AudioObjectPropertyAddress
- propertyAddressDefault = { hardwareProperty,
- kAudioObjectPropertyScopeGlobal,
- kAudioObjectPropertyElementMaster };
-
- AudioDeviceID usedID;
- UInt32 uintSize = sizeof(UInt32);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
- &propertyAddressDefault, 0, NULL, &uintSize, &usedID));
- if (usedID != kAudioDeviceUnknown)
- {
- scopedDeviceIds[numberScopedDevices] = usedID;
- numberScopedDevices++;
- } else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "GetNumberDevices(): Default device unknown");
- }
-
- // Then list the rest of the devices
- bool listOK = true;
-
- WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
- &propertyAddress, 0, NULL, &size, deviceIds));
- if (err != noErr)
- {
- listOK = false;
- } else
- {
- propertyAddress.mSelector = kAudioDevicePropertyStreamConfiguration;
- propertyAddress.mScope = scope;
- propertyAddress.mElement = 0;
- for (UInt32 i = 0; i < numberDevices; i++)
- {
- // Check for input channels
- WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyDataSize(deviceIds[i],
- &propertyAddress, 0, NULL, &size));
- if (err == kAudioHardwareBadDeviceError)
- {
- // This device doesn't actually exist; continue iterating.
- continue;
- } else if (err != noErr)
- {
- listOK = false;
- break;
- }
-
- bufferList = (AudioBufferList*) malloc(size);
- WEBRTC_CA_LOG_ERR(AudioObjectGetPropertyData(deviceIds[i],
- &propertyAddress, 0, NULL, &size, bufferList));
- if (err != noErr)
- {
- listOK = false;
- break;
- }
-
- if (bufferList->mNumberBuffers > 0)
- {
- if (numberScopedDevices >= deviceListLength)
- {
- WEBRTC_TRACE(kTraceError,
- kTraceAudioDevice, _id,
- "Device list is not long enough");
- listOK = false;
- break;
- }
-
- scopedDeviceIds[numberScopedDevices] = deviceIds[i];
- numberScopedDevices++;
- }
-
- free(bufferList);
- bufferList = NULL;
- } // for
- }
-
- if (!listOK)
- {
- if (deviceIds)
- {
- free(deviceIds);
- deviceIds = NULL;
- }
-
- if (bufferList)
- {
- free(bufferList);
- bufferList = NULL;
- }
-
- return -1;
- }
-
- // Happy ending
- if (deviceIds)
- {
- free(deviceIds);
- deviceIds = NULL;
- }
-
- return numberScopedDevices;
-}
-
-WebRtc_Word32
-AudioDeviceMac::GetDeviceName(const AudioObjectPropertyScope scope,
- const WebRtc_UWord16 index,
- char* name)
-{
- OSStatus err = noErr;
- UInt32 len = kAdmMaxDeviceNameSize;
- AudioDeviceID deviceIds[MaxNumberDevices];
-
- int numberDevices = GetNumberDevices(scope, deviceIds, MaxNumberDevices);
- if (numberDevices < 0)
- {
- return -1;
- } else if (numberDevices == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "No devices");
- return -1;
- }
-
- // If the number is below the number of devices, assume it's "WEBRTC ID"
- // otherwise assume it's a CoreAudio ID
- AudioDeviceID usedID;
-
- // Check if there is a default device
- bool isDefaultDevice = false;
- if (index == 0)
- {
- UInt32 hardwareProperty = 0;
- if (scope == kAudioDevicePropertyScopeOutput)
- {
- hardwareProperty = kAudioHardwarePropertyDefaultOutputDevice;
- } else
- {
- hardwareProperty = kAudioHardwarePropertyDefaultInputDevice;
- }
- AudioObjectPropertyAddress propertyAddress = { hardwareProperty,
- kAudioObjectPropertyScopeGlobal,
- kAudioObjectPropertyElementMaster };
- UInt32 size = sizeof(UInt32);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
- &propertyAddress, 0, NULL, &size, &usedID));
- if (usedID == kAudioDeviceUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "GetDeviceName(): Default device unknown");
- } else
- {
- isDefaultDevice = true;
- }
- }
-
- AudioObjectPropertyAddress propertyAddress = {
- kAudioDevicePropertyDeviceName, scope, 0 };
-
- if (isDefaultDevice)
- {
- char devName[len];
-
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID,
- &propertyAddress, 0, NULL, &len, devName));
-
- sprintf(name, "default (%s)", devName);
- } else
- {
- if (index < numberDevices)
- {
- usedID = deviceIds[index];
- } else
- {
- usedID = index;
- }
-
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(usedID,
- &propertyAddress, 0, NULL, &len, name));
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::InitDevice(const WebRtc_UWord16 userDeviceIndex,
- AudioDeviceID& deviceId,
- const bool isInput)
-{
- OSStatus err = noErr;
- UInt32 size = 0;
- AudioObjectPropertyScope deviceScope;
- AudioObjectPropertySelector defaultDeviceSelector;
- AudioDeviceID deviceIds[MaxNumberDevices];
-
- if (isInput)
- {
- deviceScope = kAudioDevicePropertyScopeInput;
- defaultDeviceSelector = kAudioHardwarePropertyDefaultInputDevice;
- } else
- {
- deviceScope = kAudioDevicePropertyScopeOutput;
- defaultDeviceSelector = kAudioHardwarePropertyDefaultOutputDevice;
- }
-
- AudioObjectPropertyAddress
- propertyAddress = { defaultDeviceSelector,
- kAudioObjectPropertyScopeGlobal,
- kAudioObjectPropertyElementMaster };
-
- // Get the actual device IDs
- int numberDevices = GetNumberDevices(deviceScope, deviceIds,
- MaxNumberDevices);
- if (numberDevices < 0)
- {
- return -1;
- } else if (numberDevices == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "InitDevice(): No devices");
- return -1;
- }
-
- bool isDefaultDevice = false;
- deviceId = kAudioDeviceUnknown;
- if (userDeviceIndex == 0)
- {
- // Try to use default system device
- size = sizeof(AudioDeviceID);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(kAudioObjectSystemObject,
- &propertyAddress, 0, NULL, &size, &deviceId));
- if (deviceId == kAudioDeviceUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " No default device exists");
- } else
- {
- isDefaultDevice = true;
- }
- }
-
- if (!isDefaultDevice)
- {
- deviceId = deviceIds[userDeviceIndex];
- }
-
- // Obtain device name and manufacturer for logging.
- // Also use this as a test to ensure a user-set device ID is valid.
- char devName[128];
- char devManf[128];
- memset(devName, 0, sizeof(devName));
- memset(devManf, 0, sizeof(devManf));
-
- propertyAddress.mSelector = kAudioDevicePropertyDeviceName;
- propertyAddress.mScope = deviceScope;
- propertyAddress.mElement = 0;
- size = sizeof(devName);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId,
- &propertyAddress, 0, NULL, &size, devName));
-
- propertyAddress.mSelector = kAudioDevicePropertyDeviceManufacturer;
- size = sizeof(devManf);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(deviceId,
- &propertyAddress, 0, NULL, &size, devManf));
-
- if (isInput)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Input device: %s %s", devManf, devName);
- } else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Output device: %s %s", devManf, devName);
- }
-
- return 0;
-}
-
-OSStatus AudioDeviceMac::objectListenerProc(
- AudioObjectID objectId,
- UInt32 numberAddresses,
- const AudioObjectPropertyAddress addresses[],
- void* clientData)
-{
- AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
- assert(ptrThis != NULL);
-
- ptrThis->implObjectListenerProc(objectId, numberAddresses, addresses);
-
- // AudioObjectPropertyListenerProc functions are supposed to return 0
- return 0;
-}
-
-OSStatus AudioDeviceMac::implObjectListenerProc(
- const AudioObjectID objectId,
- const UInt32 numberAddresses,
- const AudioObjectPropertyAddress addresses[])
-{
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- "AudioDeviceMac::implObjectListenerProc()");
-
- for (UInt32 i = 0; i < numberAddresses; i++)
- {
- if (addresses[i].mSelector == kAudioHardwarePropertyDevices)
- {
- HandleDeviceChange();
- } else if (addresses[i].mSelector == kAudioDevicePropertyStreamFormat)
- {
- HandleStreamFormatChange(objectId, addresses[i]);
- } else if (addresses[i].mSelector == kAudioDevicePropertyDataSource)
- {
- HandleDataSourceChange(objectId, addresses[i]);
- } else if (addresses[i].mSelector == kAudioDeviceProcessorOverload)
- {
- HandleProcessorOverload(addresses[i]);
- }
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::HandleDeviceChange()
-{
- OSStatus err = noErr;
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- "kAudioHardwarePropertyDevices");
-
- // A device has changed. Check if our registered devices have been removed.
- // Ensure the devices have been initialized, meaning the IDs are valid.
- if (MicrophoneIsInitialized())
- {
- AudioObjectPropertyAddress propertyAddress = {
- kAudioDevicePropertyDeviceIsAlive,
- kAudioDevicePropertyScopeInput, 0 };
- UInt32 deviceIsAlive = 1;
- UInt32 size = sizeof(UInt32);
- err = AudioObjectGetPropertyData(_inputDeviceID, &propertyAddress, 0,
- NULL, &size, &deviceIsAlive);
-
- if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "Capture device is not alive (probably removed)");
- AtomicSet32(&_captureDeviceIsAlive, 0);
- _mixerManager.CloseMicrophone();
- if (_recError == 1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
- _id, " pending recording error exists");
- }
- _recError = 1; // triggers callback from module process thread
- } else if (err != noErr)
- {
- logCAMsg(kTraceError, kTraceAudioDevice, _id,
- "Error in AudioDeviceGetProperty()", (const char*) &err);
- return -1;
- }
- }
-
- if (SpeakerIsInitialized())
- {
- AudioObjectPropertyAddress propertyAddress = {
- kAudioDevicePropertyDeviceIsAlive,
- kAudioDevicePropertyScopeOutput, 0 };
- UInt32 deviceIsAlive = 1;
- UInt32 size = sizeof(UInt32);
- err = AudioObjectGetPropertyData(_outputDeviceID, &propertyAddress, 0,
- NULL, &size, &deviceIsAlive);
-
- if (err == kAudioHardwareBadDeviceError || deviceIsAlive == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "Render device is not alive (probably removed)");
- AtomicSet32(&_renderDeviceIsAlive, 0);
- _mixerManager.CloseSpeaker();
- if (_playError == 1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice,
- _id, " pending playout error exists");
- }
- _playError = 1; // triggers callback from module process thread
- } else if (err != noErr)
- {
- logCAMsg(kTraceError, kTraceAudioDevice, _id,
- "Error in AudioDeviceGetProperty()", (const char*) &err);
- return -1;
- }
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::HandleStreamFormatChange(
- const AudioObjectID objectId,
- const AudioObjectPropertyAddress propertyAddress)
-{
- OSStatus err = noErr;
-
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- "Stream format changed");
-
- if (objectId != _inputDeviceID && objectId != _outputDeviceID)
- {
- return 0;
- }
-
- // Get the new device format
- AudioStreamBasicDescription streamFormat;
- UInt32 size = sizeof(streamFormat);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(objectId,
- &propertyAddress, 0, NULL, &size, &streamFormat));
-
- if (streamFormat.mFormatID != kAudioFormatLinearPCM)
- {
- logCAMsg(kTraceError, kTraceAudioDevice, _id,
- "Unacceptable input stream format -> mFormatID",
- (const char *) &streamFormat.mFormatID);
- return -1;
- }
-
- if (streamFormat.mChannelsPerFrame > N_DEVICE_CHANNELS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Too many channels on device (mChannelsPerFrame = %d)",
- streamFormat.mChannelsPerFrame);
- return -1;
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Stream format:");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "mSampleRate = %f, mChannelsPerFrame = %u",
- streamFormat.mSampleRate, streamFormat.mChannelsPerFrame);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "mBytesPerPacket = %u, mFramesPerPacket = %u",
- streamFormat.mBytesPerPacket, streamFormat.mFramesPerPacket);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "mBytesPerFrame = %u, mBitsPerChannel = %u",
- streamFormat.mBytesPerFrame, streamFormat.mBitsPerChannel);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "mFormatFlags = %u, mChannelsPerFrame = %u",
- streamFormat.mFormatFlags, streamFormat.mChannelsPerFrame);
- logCAMsg(kTraceInfo, kTraceAudioDevice, _id, "mFormatID",
- (const char *) &streamFormat.mFormatID);
-
- if (propertyAddress.mScope == kAudioDevicePropertyScopeInput)
- {
- memcpy(&_inStreamFormat, &streamFormat, sizeof(streamFormat));
-
- if (_inStreamFormat.mChannelsPerFrame >= 2 && (_recChannels == 2))
- {
- _inDesiredFormat.mChannelsPerFrame = 2;
- } else
- {
- // Disable stereo recording when we only have one channel on the device.
- _inDesiredFormat.mChannelsPerFrame = 1;
- _recChannels = 1;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Stereo recording unavailable on this device");
- }
-
- if (_ptrAudioBuffer)
- {
- // Update audio buffer with the selected parameters
- _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
- _ptrAudioBuffer->SetRecordingChannels((WebRtc_UWord8) _recChannels);
- }
-
- // Recreate the converter with the new format
- // TODO(xians): make this thread safe
- WEBRTC_CA_RETURN_ON_ERR(AudioConverterDispose(_captureConverter));
-
- WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&streamFormat, &_inDesiredFormat,
- &_captureConverter));
- } else
- {
- memcpy(&_outStreamFormat, &streamFormat, sizeof(streamFormat));
-
- if (_outStreamFormat.mChannelsPerFrame >= 2 && (_playChannels == 2))
- {
- _outDesiredFormat.mChannelsPerFrame = 2;
- } else
- {
- // Disable stereo playout when we only have one channel on the device.
- _outDesiredFormat.mChannelsPerFrame = 1;
- _playChannels = 1;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Stereo playout unavailable on this device");
- }
-
- if (_ptrAudioBuffer)
- {
- // Update audio buffer with the selected parameters
- _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
- _ptrAudioBuffer->SetPlayoutChannels((WebRtc_UWord8) _playChannels);
- }
-
- _renderDelayOffsetSamples = _renderBufSizeSamples - N_BUFFERS_OUT
- * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES
- * _outDesiredFormat.mChannelsPerFrame;
-
- // Recreate the converter with the new format
- // TODO(xians): make this thread safe
- WEBRTC_CA_RETURN_ON_ERR(AudioConverterDispose(_renderConverter));
-
- WEBRTC_CA_RETURN_ON_ERR(AudioConverterNew(&_outDesiredFormat, &streamFormat,
- &_renderConverter));
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioDeviceMac::HandleDataSourceChange(
- const AudioObjectID objectId,
- const AudioObjectPropertyAddress propertyAddress)
-{
- OSStatus err = noErr;
-
- if (_macBookPro && propertyAddress.mScope
- == kAudioDevicePropertyScopeOutput)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice, _id,
- "Data source changed");
-
- _macBookProPanRight = false;
- UInt32 dataSource = 0;
- UInt32 size = sizeof(UInt32);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(objectId,
- &propertyAddress, 0, NULL, &size, &dataSource));
- if (dataSource == 'ispk')
- {
- _macBookProPanRight = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "MacBook Pro using internal speakers; stereo panning right");
- } else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "MacBook Pro not using internal speakers");
- }
- }
-
- return 0;
-}
-WebRtc_Word32 AudioDeviceMac::HandleProcessorOverload(
- const AudioObjectPropertyAddress propertyAddress)
-{
- // TODO(xians): we probably want to notify the user in some way of the
- // overload. However, the Windows interpretations of these errors seem to
- // be more severe than what ProcessorOverload is thrown for.
- //
- // We don't log the notification, as it's sent from the HAL's IO thread. We
- // don't want to slow it down even further.
- if (propertyAddress.mScope == kAudioDevicePropertyScopeInput)
- {
- //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "Capture processor
- // overload");
- //_callback->ProblemIsReported(
- // SndCardStreamObserver::ERecordingProblem);
- } else
- {
- //WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- // "Render processor overload");
- //_callback->ProblemIsReported(
- // SndCardStreamObserver::EPlaybackProblem);
- }
-
- return 0;
-}
-
-// ============================================================================
-// Thread Methods
-// ============================================================================
-
-OSStatus AudioDeviceMac::deviceIOProc(AudioDeviceID, const AudioTimeStamp*,
- const AudioBufferList* inputData,
- const AudioTimeStamp* inputTime,
- AudioBufferList* outputData,
- const AudioTimeStamp* outputTime,
- void *clientData)
-{
- AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
- assert(ptrThis != NULL);
-
- ptrThis->implDeviceIOProc(inputData, inputTime, outputData, outputTime);
-
- // AudioDeviceIOProc functions are supposed to return 0
- return 0;
-}
-
-OSStatus AudioDeviceMac::outConverterProc(AudioConverterRef,
- UInt32 *numberDataPackets,
- AudioBufferList *data,
- AudioStreamPacketDescription **,
- void *userData)
-{
- AudioDeviceMac *ptrThis = (AudioDeviceMac *) userData;
- assert(ptrThis != NULL);
-
- return ptrThis->implOutConverterProc(numberDataPackets, data);
-}
-
-OSStatus AudioDeviceMac::inDeviceIOProc(AudioDeviceID, const AudioTimeStamp*,
- const AudioBufferList* inputData,
- const AudioTimeStamp* inputTime,
- AudioBufferList*,
- const AudioTimeStamp*, void* clientData)
-{
- AudioDeviceMac *ptrThis = (AudioDeviceMac *) clientData;
- assert(ptrThis != NULL);
-
- ptrThis->implInDeviceIOProc(inputData, inputTime);
-
- // AudioDeviceIOProc functions are supposed to return 0
- return 0;
-}
-
-OSStatus AudioDeviceMac::inConverterProc(
- AudioConverterRef,
- UInt32 *numberDataPackets,
- AudioBufferList *data,
- AudioStreamPacketDescription ** /*dataPacketDescription*/,
- void *userData)
-{
- AudioDeviceMac *ptrThis = static_cast<AudioDeviceMac*> (userData);
- assert(ptrThis != NULL);
-
- return ptrThis->implInConverterProc(numberDataPackets, data);
-}
-
-OSStatus AudioDeviceMac::implDeviceIOProc(const AudioBufferList *inputData,
- const AudioTimeStamp *inputTime,
- AudioBufferList *outputData,
- const AudioTimeStamp *outputTime)
-{
- OSStatus err = noErr;
- UInt64 outputTimeNs = AudioConvertHostTimeToNanos(outputTime->mHostTime);
- UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
-
- if (!_twoDevices && _recording)
- {
- implInDeviceIOProc(inputData, inputTime);
- }
-
- // Check if we should close down audio device
- // Double-checked locking optimization to remove locking overhead
- if (_doStop)
- {
- _critSect.Enter();
- if (_doStop)
- {
- if (_twoDevices || (!_recording && !_playing))
- {
- // In the case of a shared device, the single driving ioProc
- // is stopped here
- WEBRTC_CA_LOG_ERR(AudioDeviceStop(_outputDeviceID,
- _deviceIOProcID));
- WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_outputDeviceID,
- _deviceIOProcID));
- if (err == noErr)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
- _id, " Playout or shared device stopped");
- }
- }
-
- _doStop = false;
- _stopEvent.Set();
- _critSect.Leave();
- return 0;
- }
- _critSect.Leave();
- }
-
- if (!_playing)
- {
- // This can be the case when a shared device is capturing but not
- // rendering. We allow the checks above before returning to avoid a
- // timeout when capturing is stopped.
- return 0;
- }
-
- assert(_outStreamFormat.mBytesPerFrame != 0);
- UInt32 size = outputData->mBuffers->mDataByteSize
- / _outStreamFormat.mBytesPerFrame;
-
- // TODO(xians): signal an error somehow?
- err = AudioConverterFillComplexBuffer(_renderConverter, outConverterProc,
- this, &size, outputData, NULL);
- if (err != noErr)
- {
- if (err == 1)
- {
- // This is our own error.
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " Error in AudioConverterFillComplexBuffer()");
- return 1;
- } else
- {
- logCAMsg(kTraceError, kTraceAudioDevice, _id,
- "Error in AudioConverterFillComplexBuffer()",
- (const char *) &err);
- return 1;
- }
- }
-
- ring_buffer_size_t bufSizeSamples =
- PaUtil_GetRingBufferReadAvailable(_paRenderBuffer);
-
- int32_t renderDelayUs = static_cast<int32_t> (1e-3 * (outputTimeNs - nowNs)
- + 0.5);
- renderDelayUs += static_cast<int32_t> ((1.0e6 * bufSizeSamples)
- / _outDesiredFormat.mChannelsPerFrame / _outDesiredFormat.mSampleRate
- + 0.5);
-
- AtomicSet32(&_renderDelayUs, renderDelayUs);
-
- return 0;
-}
-
-OSStatus AudioDeviceMac::implOutConverterProc(UInt32 *numberDataPackets,
- AudioBufferList *data)
-{
- assert(data->mNumberBuffers == 1);
- ring_buffer_size_t numSamples = *numberDataPackets
- * _outDesiredFormat.mChannelsPerFrame;
-
- data->mBuffers->mNumberChannels = _outDesiredFormat.mChannelsPerFrame;
- // Always give the converter as much as it wants, zero padding as required.
- data->mBuffers->mDataByteSize = *numberDataPackets
- * _outDesiredFormat.mBytesPerPacket;
- data->mBuffers->mData = _renderConvertData;
- memset(_renderConvertData, 0, sizeof(_renderConvertData));
-
- PaUtil_ReadRingBuffer(_paRenderBuffer, _renderConvertData, numSamples);
-
- kern_return_t kernErr = semaphore_signal_all(_renderSemaphore);
- if (kernErr != KERN_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " semaphore_signal_all() error: %d", kernErr);
- return 1;
- }
-
- return 0;
-}
-
-OSStatus AudioDeviceMac::implInDeviceIOProc(const AudioBufferList *inputData,
- const AudioTimeStamp *inputTime)
-{
- OSStatus err = noErr;
- UInt64 inputTimeNs = AudioConvertHostTimeToNanos(inputTime->mHostTime);
- UInt64 nowNs = AudioConvertHostTimeToNanos(AudioGetCurrentHostTime());
-
- // Check if we should close down audio device
- // Double-checked locking optimization to remove locking overhead
- if (_doStopRec)
- {
- _critSect.Enter();
- if (_doStopRec)
- {
- // This will be signalled only when a shared device is not in use.
- WEBRTC_CA_LOG_ERR(AudioDeviceStop(_inputDeviceID, _inDeviceIOProcID));
- WEBRTC_CA_LOG_WARN(AudioDeviceDestroyIOProcID(_inputDeviceID,
- _inDeviceIOProcID));
- if (err == noErr)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceAudioDevice,
- _id, " Recording device stopped");
- }
-
- _doStopRec = false;
- _stopEventRec.Set();
- _critSect.Leave();
- return 0;
- }
- _critSect.Leave();
- }
-
- if (!_recording)
- {
- // Allow above checks to avoid a timeout on stopping capture.
- return 0;
- }
-
- ring_buffer_size_t bufSizeSamples =
- PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer);
-
- int32_t captureDelayUs = static_cast<int32_t> (1e-3 * (nowNs - inputTimeNs)
- + 0.5);
- captureDelayUs
- += static_cast<int32_t> ((1.0e6 * bufSizeSamples)
- / _inStreamFormat.mChannelsPerFrame / _inStreamFormat.mSampleRate
- + 0.5);
-
- AtomicSet32(&_captureDelayUs, captureDelayUs);
-
- assert(inputData->mNumberBuffers == 1);
- ring_buffer_size_t numSamples = inputData->mBuffers->mDataByteSize
- * _inStreamFormat.mChannelsPerFrame / _inStreamFormat.mBytesPerPacket;
- PaUtil_WriteRingBuffer(_paCaptureBuffer, inputData->mBuffers->mData,
- numSamples);
-
- kern_return_t kernErr = semaphore_signal_all(_captureSemaphore);
- if (kernErr != KERN_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " semaphore_signal_all() error: %d", kernErr);
- }
-
- return err;
-}
-
-OSStatus AudioDeviceMac::implInConverterProc(UInt32 *numberDataPackets,
- AudioBufferList *data)
-{
- assert(data->mNumberBuffers == 1);
- ring_buffer_size_t numSamples = *numberDataPackets
- * _inStreamFormat.mChannelsPerFrame;
-
- while (PaUtil_GetRingBufferReadAvailable(_paCaptureBuffer) < numSamples)
- {
- mach_timespec_t timeout;
- timeout.tv_sec = 0;
- timeout.tv_nsec = TIMER_PERIOD_MS;
-
- kern_return_t kernErr = semaphore_timedwait(_captureSemaphore, timeout);
- if (kernErr == KERN_OPERATION_TIMED_OUT)
- {
- int32_t signal = AtomicGet32(&_captureDeviceIsAlive);
- if (signal == 0)
- {
- // The capture device is no longer alive; stop the worker thread.
- *numberDataPackets = 0;
- return 1;
- }
- } else if (kernErr != KERN_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " semaphore_wait() error: %d", kernErr);
- }
- }
-
- // Pass the read pointer directly to the converter to avoid a memcpy.
- void* dummyPtr;
- ring_buffer_size_t dummySize;
- PaUtil_GetRingBufferReadRegions(_paCaptureBuffer, numSamples,
- &data->mBuffers->mData, &numSamples,
- &dummyPtr, &dummySize);
- PaUtil_AdvanceRingBufferReadIndex(_paCaptureBuffer, numSamples);
-
- data->mBuffers->mNumberChannels = _inStreamFormat.mChannelsPerFrame;
- *numberDataPackets = numSamples / _inStreamFormat.mChannelsPerFrame;
- data->mBuffers->mDataByteSize = *numberDataPackets
- * _inStreamFormat.mBytesPerPacket;
-
- return 0;
-}
-
-bool AudioDeviceMac::RunRender(void* ptrThis)
-{
- return static_cast<AudioDeviceMac*> (ptrThis)->RenderWorkerThread();
-}
-
-bool AudioDeviceMac::RenderWorkerThread()
-{
- ring_buffer_size_t numSamples = ENGINE_PLAY_BUF_SIZE_IN_SAMPLES
- * _outDesiredFormat.mChannelsPerFrame;
- while (PaUtil_GetRingBufferWriteAvailable(_paRenderBuffer)
- - _renderDelayOffsetSamples < numSamples)
- {
- mach_timespec_t timeout;
- timeout.tv_sec = 0;
- timeout.tv_nsec = TIMER_PERIOD_MS;
-
- kern_return_t kernErr = semaphore_timedwait(_renderSemaphore, timeout);
- if (kernErr == KERN_OPERATION_TIMED_OUT)
- {
- int32_t signal = AtomicGet32(&_renderDeviceIsAlive);
- if (signal == 0)
- {
- // The render device is no longer alive; stop the worker thread.
- return false;
- }
- } else if (kernErr != KERN_SUCCESS)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " semaphore_timedwait() error: %d", kernErr);
- }
- }
-
- WebRtc_Word8 playBuffer[4 * ENGINE_PLAY_BUF_SIZE_IN_SAMPLES];
-
- if (!_ptrAudioBuffer)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " capture AudioBuffer is invalid");
- return false;
- }
-
- // Ask for new PCM data to be played out using the AudioDeviceBuffer.
- WebRtc_UWord32 nSamples =
- _ptrAudioBuffer->RequestPlayoutData(ENGINE_PLAY_BUF_SIZE_IN_SAMPLES);
-
- nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
- if (nSamples != ENGINE_PLAY_BUF_SIZE_IN_SAMPLES)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " invalid number of output samples(%d)", nSamples);
- }
-
- WebRtc_UWord32 nOutSamples = nSamples * _outDesiredFormat.mChannelsPerFrame;
-
- SInt16 *pPlayBuffer = (SInt16 *) &playBuffer;
- if (_macBookProPanRight && (_playChannels == 2))
- {
- // Mix entirely into the right channel and zero the left channel.
- SInt32 sampleInt32 = 0;
- for (WebRtc_UWord32 sampleIdx = 0; sampleIdx < nOutSamples; sampleIdx
- += 2)
- {
- sampleInt32 = pPlayBuffer[sampleIdx];
- sampleInt32 += pPlayBuffer[sampleIdx + 1];
- sampleInt32 /= 2;
-
- if (sampleInt32 > 32767)
- {
- sampleInt32 = 32767;
- } else if (sampleInt32 < -32768)
- {
- sampleInt32 = -32768;
- }
-
- pPlayBuffer[sampleIdx] = 0;
- pPlayBuffer[sampleIdx + 1] = static_cast<SInt16> (sampleInt32);
- }
- }
-
- PaUtil_WriteRingBuffer(_paRenderBuffer, pPlayBuffer, nOutSamples);
-
- return true;
-}
-
-bool AudioDeviceMac::RunCapture(void* ptrThis)
-{
- return static_cast<AudioDeviceMac*> (ptrThis)->CaptureWorkerThread();
-}
-
-bool AudioDeviceMac::CaptureWorkerThread()
-{
- OSStatus err = noErr;
- UInt32 noRecSamples = ENGINE_REC_BUF_SIZE_IN_SAMPLES
- * _inDesiredFormat.mChannelsPerFrame;
- SInt16 recordBuffer[noRecSamples];
- UInt32 size = ENGINE_REC_BUF_SIZE_IN_SAMPLES;
-
- AudioBufferList engineBuffer;
- engineBuffer.mNumberBuffers = 1; // Interleaved channels.
- engineBuffer.mBuffers->mNumberChannels = _inDesiredFormat.mChannelsPerFrame;
- engineBuffer.mBuffers->mDataByteSize = _inDesiredFormat.mBytesPerPacket
- * noRecSamples;
- engineBuffer.mBuffers->mData = recordBuffer;
-
- err = AudioConverterFillComplexBuffer(_captureConverter, inConverterProc,
- this, &size, &engineBuffer, NULL);
- if (err != noErr)
- {
- if (err == 1)
- {
- // This is our own error.
- return false;
- } else
- {
- logCAMsg(kTraceError, kTraceAudioDevice, _id,
- "Error in AudioConverterFillComplexBuffer()",
- (const char *) &err);
- return false;
- }
- }
-
- // TODO(xians): what if the returned size is incorrect?
- if (size == ENGINE_REC_BUF_SIZE_IN_SAMPLES)
- {
- WebRtc_UWord32 currentMicLevel(0);
- WebRtc_UWord32 newMicLevel(0);
- WebRtc_Word32 msecOnPlaySide;
- WebRtc_Word32 msecOnRecordSide;
-
- int32_t captureDelayUs = AtomicGet32(&_captureDelayUs);
- int32_t renderDelayUs = AtomicGet32(&_renderDelayUs);
-
- msecOnPlaySide = static_cast<WebRtc_Word32> (1e-3 * (renderDelayUs
- + _renderLatencyUs) + 0.5);
- msecOnRecordSide = static_cast<WebRtc_Word32> (1e-3 * (captureDelayUs
- + _captureLatencyUs) + 0.5);
-
- if (!_ptrAudioBuffer)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " capture AudioBuffer is invalid");
- return false;
- }
-
- // store the recorded buffer (no action will be taken if the
- // #recorded samples is not a full buffer)
- _ptrAudioBuffer->SetRecordedBuffer((WebRtc_Word8*) &recordBuffer,
- (WebRtc_UWord32) size);
-
- if (AGC())
- {
- // store current mic level in the audio buffer if AGC is enabled
- if (MicrophoneVolume(currentMicLevel) == 0)
- {
- // this call does not affect the actual microphone volume
- _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
- }
- }
-
- _ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide, 0);
-
- // deliver recorded samples at specified sample rate, mic level etc.
- // to the observer using callback
- _ptrAudioBuffer->DeliverRecordedData();
-
- if (AGC())
- {
- newMicLevel = _ptrAudioBuffer->NewMicLevel();
- if (newMicLevel != 0)
- {
- // The VQE will only deliver non-zero microphone levels when
- // a change is needed.
- // Set this new mic level (received from the observer as return
- // value in the callback).
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice,
- _id, " AGC change of volume: old=%u => new=%u",
- currentMicLevel, newMicLevel);
- if (SetMicrophoneVolume(newMicLevel) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " the required modification of the microphone "
- "volume failed");
- }
- }
- }
- }
-
- return true;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/mac/audio_device_mac.h b/trunk/src/modules/audio_device/main/source/mac/audio_device_mac.h
deleted file mode 100644
index 9dc8917..0000000
--- a/trunk/src/modules/audio_device/main/source/mac/audio_device_mac.h
+++ /dev/null
@@ -1,399 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_MAC_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_MAC_H
-
-#include "audio_device_generic.h"
-#include "critical_section_wrapper.h"
-#include "audio_mixer_manager_mac.h"
-
-#include <CoreAudio/CoreAudio.h>
-#include <AudioToolbox/AudioConverter.h>
-#include <mach/semaphore.h>
-
-struct PaUtilRingBuffer;
-
-namespace webrtc
-{
-class EventWrapper;
-class ThreadWrapper;
-
-const WebRtc_UWord32 N_REC_SAMPLES_PER_SEC = 48000;
-const WebRtc_UWord32 N_PLAY_SAMPLES_PER_SEC = 48000;
-
-const WebRtc_UWord32 N_REC_CHANNELS = 1; // default is mono recording
-const WebRtc_UWord32 N_PLAY_CHANNELS = 2; // default is stereo playout
-const WebRtc_UWord32 N_DEVICE_CHANNELS = 8;
-
-const WebRtc_UWord32 ENGINE_REC_BUF_SIZE_IN_SAMPLES = (N_REC_SAMPLES_PER_SEC
- / 100);
-const WebRtc_UWord32 ENGINE_PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC
- / 100);
-
-enum
-{
- N_BLOCKS_IO = 2
-};
-enum
-{
- N_BUFFERS_IN = 10
-};
-enum
-{
- N_BUFFERS_OUT = 3
-}; // Must be at least N_BLOCKS_IO
-
-const WebRtc_UWord32 TIMER_PERIOD_MS = (2 * 10 * N_BLOCKS_IO * 1000000);
-
-const WebRtc_UWord32 REC_BUF_SIZE_IN_SAMPLES = (ENGINE_REC_BUF_SIZE_IN_SAMPLES
- * N_DEVICE_CHANNELS * N_BUFFERS_IN);
-const WebRtc_UWord32 PLAY_BUF_SIZE_IN_SAMPLES =
- (ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * N_PLAY_CHANNELS * N_BUFFERS_OUT);
-
-class AudioDeviceMac: public AudioDeviceGeneric
-{
-public:
- AudioDeviceMac(const WebRtc_Word32 id);
- ~AudioDeviceMac();
-
- // Retrieve the currently utilized audio layer
- virtual WebRtc_Word32
- ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
-
- // Main initializaton and termination
- virtual WebRtc_Word32 Init();
- virtual WebRtc_Word32 Terminate();
- virtual bool Initialized() const;
-
- // Device enumeration
- virtual WebRtc_Word16 PlayoutDevices();
- virtual WebRtc_Word16 RecordingDevices();
- virtual WebRtc_Word32 PlayoutDeviceName(
- WebRtc_UWord16 index,
- WebRtc_Word8 name[kAdmMaxDeviceNameSize],
- WebRtc_Word8 guid[kAdmMaxGuidSize]);
- virtual WebRtc_Word32 RecordingDeviceName(
- WebRtc_UWord16 index,
- WebRtc_Word8 name[kAdmMaxDeviceNameSize],
- WebRtc_Word8 guid[kAdmMaxGuidSize]);
-
- // Device selection
- virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32 SetPlayoutDevice(
- AudioDeviceModule::WindowsDeviceType device);
- virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32 SetRecordingDevice(
- AudioDeviceModule::WindowsDeviceType device);
-
- // Audio transport initialization
- virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
- virtual WebRtc_Word32 InitPlayout();
- virtual bool PlayoutIsInitialized() const;
- virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
- virtual WebRtc_Word32 InitRecording();
- virtual bool RecordingIsInitialized() const;
-
- // Audio transport control
- virtual WebRtc_Word32 StartPlayout();
- virtual WebRtc_Word32 StopPlayout();
- virtual bool Playing() const;
- virtual WebRtc_Word32 StartRecording();
- virtual WebRtc_Word32 StopRecording();
- virtual bool Recording() const;
-
- // Microphone Automatic Gain Control (AGC)
- virtual WebRtc_Word32 SetAGC(bool enable);
- virtual bool AGC() const;
-
- // Volume control based on the Windows Wave API (Windows only)
- virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
- WebRtc_UWord16 volumeRight);
- virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft,
- WebRtc_UWord16& volumeRight) const;
-
- // Audio mixer initialization
- virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
- virtual WebRtc_Word32 InitSpeaker();
- virtual bool SpeakerIsInitialized() const;
- virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
- virtual WebRtc_Word32 InitMicrophone();
- virtual bool MicrophoneIsInitialized() const;
-
- // Speaker volume controls
- virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
- virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
- virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
- virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
- virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
-
- // Microphone volume controls
- virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
- virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
- virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
- virtual WebRtc_Word32
- MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
-
- // Microphone mute control
- virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
- virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
-
- // Speaker mute control
- virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
- virtual WebRtc_Word32 SetSpeakerMute(bool enable);
- virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
-
- // Microphone boost control
- virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
- virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
-
- // Stereo support
- virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
- virtual WebRtc_Word32 SetStereoPlayout(bool enable);
- virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
- virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
- virtual WebRtc_Word32 SetStereoRecording(bool enable);
- virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
-
- // Delay information and control
- virtual WebRtc_Word32
- SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
- WebRtc_UWord16 sizeMS);
- virtual WebRtc_Word32 PlayoutBuffer(AudioDeviceModule::BufferType& type,
- WebRtc_UWord16& sizeMS) const;
- virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
- virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
-
- // CPU load
- virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
-
-public:
- virtual bool PlayoutWarning() const;
- virtual bool PlayoutError() const;
- virtual bool RecordingWarning() const;
- virtual bool RecordingError() const;
- virtual void ClearPlayoutWarning();
- virtual void ClearPlayoutError();
- virtual void ClearRecordingWarning();
- virtual void ClearRecordingError();
-
-public:
- virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
-private:
- void Lock()
- {
- _critSect.Enter();
- }
- ;
- void UnLock()
- {
- _critSect.Leave();
- }
- ;
- WebRtc_Word32 Id()
- {
- return _id;
- }
-
- static void AtomicSet32(int32_t* theValue, int32_t newValue);
- static int32_t AtomicGet32(int32_t* theValue);
-
- static void logCAMsg(const TraceLevel level,
- const TraceModule module,
- const WebRtc_Word32 id, const char *msg,
- const char *err);
-
- WebRtc_Word32 GetNumberDevices(const AudioObjectPropertyScope scope,
- AudioDeviceID scopedDeviceIds[],
- const WebRtc_UWord32 deviceListLength);
-
- WebRtc_Word32 GetDeviceName(const AudioObjectPropertyScope scope,
- const WebRtc_UWord16 index, char* name);
-
- WebRtc_Word32 InitDevice(WebRtc_UWord16 userDeviceIndex,
- AudioDeviceID& deviceId, bool isInput);
-
- static OSStatus
- objectListenerProc(AudioObjectID objectId, UInt32 numberAddresses,
- const AudioObjectPropertyAddress addresses[],
- void* clientData);
-
- OSStatus
- implObjectListenerProc(AudioObjectID objectId, UInt32 numberAddresses,
- const AudioObjectPropertyAddress addresses[]);
-
- WebRtc_Word32 HandleDeviceChange();
-
- WebRtc_Word32
- HandleStreamFormatChange(AudioObjectID objectId,
- AudioObjectPropertyAddress propertyAddress);
-
- WebRtc_Word32
- HandleDataSourceChange(AudioObjectID objectId,
- AudioObjectPropertyAddress propertyAddress);
-
- WebRtc_Word32
- HandleProcessorOverload(AudioObjectPropertyAddress propertyAddress);
-
-private:
- static OSStatus deviceIOProc(AudioDeviceID device,
- const AudioTimeStamp *now,
- const AudioBufferList *inputData,
- const AudioTimeStamp *inputTime,
- AudioBufferList *outputData,
- const AudioTimeStamp* outputTime,
- void *clientData);
-
- static OSStatus
- outConverterProc(AudioConverterRef audioConverter,
- UInt32 *numberDataPackets, AudioBufferList *data,
- AudioStreamPacketDescription **dataPacketDescription,
- void *userData);
-
- static OSStatus inDeviceIOProc(AudioDeviceID device,
- const AudioTimeStamp *now,
- const AudioBufferList *inputData,
- const AudioTimeStamp *inputTime,
- AudioBufferList *outputData,
- const AudioTimeStamp *outputTime,
- void *clientData);
-
- static OSStatus
- inConverterProc(AudioConverterRef audioConverter,
- UInt32 *numberDataPackets, AudioBufferList *data,
- AudioStreamPacketDescription **dataPacketDescription,
- void *inUserData);
-
- OSStatus implDeviceIOProc(const AudioBufferList *inputData,
- const AudioTimeStamp *inputTime,
- AudioBufferList *outputData,
- const AudioTimeStamp *outputTime);
-
- OSStatus implOutConverterProc(UInt32 *numberDataPackets,
- AudioBufferList *data);
-
- OSStatus implInDeviceIOProc(const AudioBufferList *inputData,
- const AudioTimeStamp *inputTime);
-
- OSStatus implInConverterProc(UInt32 *numberDataPackets,
- AudioBufferList *data);
-
- static bool RunCapture(void*);
- static bool RunRender(void*);
- bool CaptureWorkerThread();
- bool RenderWorkerThread();
-
-private:
- AudioDeviceBuffer* _ptrAudioBuffer;
-
- CriticalSectionWrapper& _critSect;
-
- EventWrapper& _stopEventRec;
- EventWrapper& _stopEvent;
-
- ThreadWrapper* _captureWorkerThread;
- ThreadWrapper* _renderWorkerThread;
- WebRtc_UWord32 _captureWorkerThreadId;
- WebRtc_UWord32 _renderWorkerThreadId;
-
- WebRtc_Word32 _id;
-
- AudioMixerManagerMac _mixerManager;
-
- WebRtc_UWord16 _inputDeviceIndex;
- WebRtc_UWord16 _outputDeviceIndex;
- AudioDeviceID _inputDeviceID;
- AudioDeviceID _outputDeviceID;
-#if __MAC_OS_X_VERSION_MAX_ALLOWED >= 1050
- AudioDeviceIOProcID _inDeviceIOProcID;
- AudioDeviceIOProcID _deviceIOProcID;
-#endif
- bool _inputDeviceIsSpecified;
- bool _outputDeviceIsSpecified;
-
- WebRtc_UWord8 _recChannels;
- WebRtc_UWord8 _playChannels;
-
- Float32* _captureBufData;
- SInt16* _renderBufData;
-
- SInt16 _renderConvertData[PLAY_BUF_SIZE_IN_SAMPLES];
-
- AudioDeviceModule::BufferType _playBufType;
-
-private:
- bool _initialized;
- bool _isShutDown;
- bool _recording;
- bool _playing;
- bool _recIsInitialized;
- bool _playIsInitialized;
- bool _startRec;
- bool _stopRec;
- bool _stopPlay;
- bool _AGC;
-
- // Atomically set varaibles
- int32_t _renderDeviceIsAlive;
- int32_t _captureDeviceIsAlive;
-
- bool _twoDevices;
- bool _doStop; // For play if not shared device or play+rec if shared device
- bool _doStopRec; // For rec if not shared device
- bool _macBookPro;
- bool _macBookProPanRight;
- bool _stereoRender;
- bool _stereoRenderRequested;
-
- AudioConverterRef _captureConverter;
- AudioConverterRef _renderConverter;
-
- AudioStreamBasicDescription _outStreamFormat;
- AudioStreamBasicDescription _outDesiredFormat;
- AudioStreamBasicDescription _inStreamFormat;
- AudioStreamBasicDescription _inDesiredFormat;
-
- WebRtc_UWord32 _captureLatencyUs;
- WebRtc_UWord32 _renderLatencyUs;
-
- // Atomically set variables
- mutable int32_t _captureDelayUs;
- mutable int32_t _renderDelayUs;
-
- WebRtc_Word32 _renderDelayOffsetSamples;
-
-private:
- WebRtc_UWord16 _playBufDelay; // playback delay
- WebRtc_UWord16 _playBufDelayFixed; // fixed playback delay
-
- WebRtc_UWord16 _playWarning;
- WebRtc_UWord16 _playError;
- WebRtc_UWord16 _recWarning;
- WebRtc_UWord16 _recError;
-
- PaUtilRingBuffer* _paCaptureBuffer;
- PaUtilRingBuffer* _paRenderBuffer;
-
- semaphore_t _renderSemaphore;
- semaphore_t _captureSemaphore;
-
- WebRtc_UWord32 _captureBufSizeSamples;
- WebRtc_UWord32 _renderBufSizeSamples;
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_MAC_H_
diff --git a/trunk/src/modules/audio_device/main/source/mac/audio_device_utility_mac.cc b/trunk/src/modules/audio_device/main/source/mac/audio_device_utility_mac.cc
deleted file mode 100644
index 94bdf26..0000000
--- a/trunk/src/modules/audio_device/main/source/mac/audio_device_utility_mac.cc
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_device_utility_mac.h"
-#include "audio_device_config.h" // DEBUG_PRINT()
-#include "critical_section_wrapper.h"
-#include "trace.h"
-
-namespace webrtc
-{
-
-AudioDeviceUtilityMac::AudioDeviceUtilityMac(const WebRtc_Word32 id) :
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _id(id),
- _lastError(AudioDeviceModule::kAdmErrNone)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id,
- "%s created", __FUNCTION__);
-}
-
-// ----------------------------------------------------------------------------
-// AudioDeviceUtilityMac() - dtor
-// ----------------------------------------------------------------------------
-
-AudioDeviceUtilityMac::~AudioDeviceUtilityMac()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
- "%s destroyed", __FUNCTION__);
- {
- CriticalSectionScoped lock(_critSect);
-
- // free stuff here...
- }
-
- delete &_critSect;
-}
-
-WebRtc_Word32 AudioDeviceUtilityMac::Init()
-{
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id,
- " OS info: %s", "OS X");
-
- return 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/mac/audio_device_utility_mac.h b/trunk/src/modules/audio_device/main/source/mac/audio_device_utility_mac.h
deleted file mode 100644
index ccb3d99..0000000
--- a/trunk/src/modules/audio_device/main/source/mac/audio_device_utility_mac.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_MAC_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_MAC_H
-
-#include "audio_device_utility.h"
-#include "audio_device.h"
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-
-class AudioDeviceUtilityMac: public AudioDeviceUtility
-{
-public:
- AudioDeviceUtilityMac(const WebRtc_Word32 id);
- ~AudioDeviceUtilityMac();
-
- virtual WebRtc_Word32 Init();
-
-private:
- CriticalSectionWrapper& _critSect;
- WebRtc_Word32 _id;
- AudioDeviceModule::ErrorCode _lastError;
-};
-
-} // namespace webrtc
-
-#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_UTILITY_MAC_H_
diff --git a/trunk/src/modules/audio_device/main/source/mac/audio_mixer_manager_mac.cc b/trunk/src/modules/audio_device/main/source/mac/audio_mixer_manager_mac.cc
deleted file mode 100644
index 40850d4..0000000
--- a/trunk/src/modules/audio_device/main/source/mac/audio_mixer_manager_mac.cc
+++ /dev/null
@@ -1,1167 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_mixer_manager_mac.h"
-#include "trace.h"
-
-#include <unistd.h> // getpid()
-
-namespace webrtc {
-
-#define WEBRTC_CA_RETURN_ON_ERR(expr) \
- do { \
- err = expr; \
- if (err != noErr) { \
- logCAMsg(kTraceError, kTraceAudioDevice, _id, \
- "Error in " #expr, (const char *)&err); \
- return -1; \
- } \
- } while(0)
-
-#define WEBRTC_CA_LOG_ERR(expr) \
- do { \
- err = expr; \
- if (err != noErr) { \
- logCAMsg(kTraceError, kTraceAudioDevice, _id, \
- "Error in " #expr, (const char *)&err); \
- } \
- } while(0)
-
-#define WEBRTC_CA_LOG_WARN(expr) \
- do { \
- err = expr; \
- if (err != noErr) { \
- logCAMsg(kTraceWarning, kTraceAudioDevice, _id, \
- "Error in " #expr, (const char *)&err); \
- } \
- } while(0)
-
-AudioMixerManagerMac::AudioMixerManagerMac(const WebRtc_Word32 id) :
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _id(id),
- _inputDeviceID(kAudioObjectUnknown),
- _outputDeviceID(kAudioObjectUnknown),
- _noInputChannels(0),
- _noOutputChannels(0)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
- "%s constructed", __FUNCTION__);
-}
-
-AudioMixerManagerMac::~AudioMixerManagerMac()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id,
- "%s destructed", __FUNCTION__);
-
- Close();
-
- delete &_critSect;
-}
-
-// ============================================================================
-// PUBLIC METHODS
-// ============================================================================
-
-WebRtc_Word32 AudioMixerManagerMac::Close()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
- __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- CloseSpeaker();
- CloseMicrophone();
-
- return 0;
-
-}
-
-WebRtc_Word32 AudioMixerManagerMac::CloseSpeaker()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
- __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- _outputDeviceID = kAudioObjectUnknown;
- _noOutputChannels = 0;
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::CloseMicrophone()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s",
- __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- _inputDeviceID = kAudioObjectUnknown;
- _noInputChannels = 0;
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::OpenSpeaker(AudioDeviceID deviceID)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerMac::OpenSpeaker(id=%d)", deviceID);
-
- CriticalSectionScoped lock(_critSect);
-
- OSStatus err = noErr;
- UInt32 size = 0;
- pid_t hogPid = -1;
-
- _outputDeviceID = deviceID;
-
- // Check which process, if any, has hogged the device.
- AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyHogMode,
- kAudioDevicePropertyScopeOutput, 0 };
-
- size = sizeof(hogPid);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
- &propertyAddress, 0, NULL, &size, &hogPid));
-
- if (hogPid == -1)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " No process has hogged the input device");
- }
- // getpid() is apparently "always successful"
- else if (hogPid == getpid())
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Our process has hogged the input device");
- } else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Another process (pid = %d) has hogged the input device",
- static_cast<int> (hogPid));
-
- return -1;
- }
-
- // get number of channels from stream format
- propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
-
- // Get the stream format, to be able to read the number of channels.
- AudioStreamBasicDescription streamFormat;
- size = sizeof(AudioStreamBasicDescription);
- memset(&streamFormat, 0, size);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
- &propertyAddress, 0, NULL, &size, &streamFormat));
-
- _noOutputChannels = streamFormat.mChannelsPerFrame;
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::OpenMicrophone(AudioDeviceID deviceID)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerMac::OpenMicrophone(id=%d)", deviceID);
-
- CriticalSectionScoped lock(_critSect);
-
- OSStatus err = noErr;
- UInt32 size = 0;
- pid_t hogPid = -1;
-
- _inputDeviceID = deviceID;
-
- // Check which process, if any, has hogged the device.
- AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyHogMode,
- kAudioDevicePropertyScopeInput, 0 };
- size = sizeof(hogPid);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
- &propertyAddress, 0, NULL, &size, &hogPid));
- if (hogPid == -1)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " No process has hogged the input device");
- }
- // getpid() is apparently "always successful"
- else if (hogPid == getpid())
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " Our process has hogged the input device");
- } else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Another process (pid = %d) has hogged the input device",
- static_cast<int> (hogPid));
-
- return -1;
- }
-
- // get number of channels from stream format
- propertyAddress.mSelector = kAudioDevicePropertyStreamFormat;
-
- // Get the stream format, to be able to read the number of channels.
- AudioStreamBasicDescription streamFormat;
- size = sizeof(AudioStreamBasicDescription);
- memset(&streamFormat, 0, size);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
- &propertyAddress, 0, NULL, &size, &streamFormat));
-
- _noInputChannels = streamFormat.mChannelsPerFrame;
-
- return 0;
-}
-
-bool AudioMixerManagerMac::SpeakerIsInitialized() const
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s",
- __FUNCTION__);
-
- return (_outputDeviceID != kAudioObjectUnknown);
-}
-
-bool AudioMixerManagerMac::MicrophoneIsInitialized() const
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s",
- __FUNCTION__);
-
- return (_inputDeviceID != kAudioObjectUnknown);
-}
-
-WebRtc_Word32 AudioMixerManagerMac::SetSpeakerVolume(WebRtc_UWord32 volume)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerMac::SetSpeakerVolume(volume=%u)", volume);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_outputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- OSStatus err = noErr;
- UInt32 size = 0;
- bool success = false;
-
- // volume range is 0.0 - 1.0, convert from 0 -255
- const Float32 vol = (Float32)(volume / 255.0);
-
- assert(vol <= 1.0 && vol >= 0.0);
-
- // Does the capture device have a master volume control?
- // If so, use it exclusively.
- AudioObjectPropertyAddress propertyAddress = {
- kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput,
- 0 };
- Boolean isSettable = false;
- err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
- &isSettable);
- if (err == noErr && isSettable)
- {
- size = sizeof(vol);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
- &propertyAddress, 0, NULL, size, &vol));
-
- return 0;
- }
-
- // Otherwise try to set each channel.
- for (UInt32 i = 1; i <= _noOutputChannels; i++)
- {
- propertyAddress.mElement = i;
- isSettable = false;
- err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
- &isSettable);
- if (err == noErr && isSettable)
- {
- size = sizeof(vol);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
- &propertyAddress, 0, NULL, size, &vol));
- }
- success = true;
- }
-
- if (!success)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Unable to set a volume on any output channel");
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::SpeakerVolume(WebRtc_UWord32& volume) const
-{
-
- if (_outputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- OSStatus err = noErr;
- UInt32 size = 0;
- unsigned int channels = 0;
- Float32 channelVol = 0;
- Float32 vol = 0;
-
- // Does the device have a master volume control?
- // If so, use it exclusively.
- AudioObjectPropertyAddress propertyAddress = {
- kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput,
- 0 };
- Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
- &propertyAddress);
- if (hasProperty)
- {
- size = sizeof(vol);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
- &propertyAddress, 0, NULL, &size, &vol));
-
- // vol 0.0 to 1.0 -> convert to 0 - 255
- volume = static_cast<WebRtc_UWord32> (vol * 255 + 0.5);
- } else
- {
- // Otherwise get the average volume across channels.
- vol = 0;
- for (UInt32 i = 1; i <= _noOutputChannels; i++)
- {
- channelVol = 0;
- propertyAddress.mElement = i;
- hasProperty = AudioObjectHasProperty(_outputDeviceID,
- &propertyAddress);
- if (hasProperty)
- {
- size = sizeof(channelVol);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
- &propertyAddress, 0, NULL, &size, &channelVol));
-
- vol += channelVol;
- channels++;
- }
- }
-
- if (channels == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Unable to get a volume on any channel");
- return -1;
- }
-
- assert(channels > 0);
- // vol 0.0 to 1.0 -> convert to 0 - 255
- volume = static_cast<WebRtc_UWord32> (255 * vol / channels + 0.5);
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " AudioMixerManagerMac::SpeakerVolume() => vol=%i", vol);
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerMac::MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const
-{
-
- if (_outputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- // volume range is 0.0 to 1.0
- // we convert that to 0 - 255
- maxVolume = 255;
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerMac::MinSpeakerVolume(WebRtc_UWord32& minVolume) const
-{
-
- if (_outputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- // volume range is 0.0 to 1.0
- // we convert that to 0 - 255
- minVolume = 0;
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerMac::SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const
-{
-
- if (_outputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- // volume range is 0.0 to 1.0
- // we convert that to 0 - 255
- stepSize = 1;
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::SpeakerVolumeIsAvailable(bool& available)
-{
- if (_outputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- OSStatus err = noErr;
-
- // Does the capture device have a master volume control?
- // If so, use it exclusively.
- AudioObjectPropertyAddress propertyAddress = {
- kAudioDevicePropertyVolumeScalar, kAudioDevicePropertyScopeOutput,
- 0 };
- Boolean isSettable = false;
- err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
- &isSettable);
- if (err == noErr && isSettable)
- {
- available = true;
- return 0;
- }
-
- // Otherwise try to set each channel.
- for (UInt32 i = 1; i <= _noOutputChannels; i++)
- {
- propertyAddress.mElement = i;
- isSettable = false;
- err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
- &isSettable);
- if (err != noErr || !isSettable)
- {
- available = false;
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Volume cannot be set for output channel %d, err=%d",
- i, err);
- return -1;
- }
- }
-
- available = true;
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::SpeakerMuteIsAvailable(bool& available)
-{
- if (_outputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- OSStatus err = noErr;
-
- // Does the capture device have a master mute control?
- // If so, use it exclusively.
- AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
- kAudioDevicePropertyScopeOutput, 0 };
- Boolean isSettable = false;
- err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
- &isSettable);
- if (err == noErr && isSettable)
- {
- available = true;
- return 0;
- }
-
- // Otherwise try to set each channel.
- for (UInt32 i = 1; i <= _noOutputChannels; i++)
- {
- propertyAddress.mElement = i;
- isSettable = false;
- err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
- &isSettable);
- if (err != noErr || !isSettable)
- {
- available = false;
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Mute cannot be set for output channel %d, err=%d",
- i, err);
- return -1;
- }
- }
-
- available = true;
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::SetSpeakerMute(bool enable)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerMac::SetSpeakerMute(enable=%u)", enable);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_outputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- OSStatus err = noErr;
- UInt32 size = 0;
- UInt32 mute = enable ? 1 : 0;
- bool success = false;
-
- // Does the render device have a master mute control?
- // If so, use it exclusively.
- AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
- kAudioDevicePropertyScopeOutput, 0 };
- Boolean isSettable = false;
- err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
- &isSettable);
- if (err == noErr && isSettable)
- {
- size = sizeof(mute);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
- &propertyAddress, 0, NULL, size, &mute));
-
- return 0;
- }
-
- // Otherwise try to set each channel.
- for (UInt32 i = 1; i <= _noOutputChannels; i++)
- {
- propertyAddress.mElement = i;
- isSettable = false;
- err = AudioObjectIsPropertySettable(_outputDeviceID, &propertyAddress,
- &isSettable);
- if (err == noErr && isSettable)
- {
- size = sizeof(mute);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_outputDeviceID,
- &propertyAddress, 0, NULL, size, &mute));
- }
- success = true;
- }
-
- if (!success)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Unable to set mute on any input channel");
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::SpeakerMute(bool& enabled) const
-{
-
- if (_outputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- OSStatus err = noErr;
- UInt32 size = 0;
- unsigned int channels = 0;
- UInt32 channelMuted = 0;
- UInt32 muted = 0;
-
- // Does the device have a master volume control?
- // If so, use it exclusively.
- AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
- kAudioDevicePropertyScopeOutput, 0 };
- Boolean hasProperty = AudioObjectHasProperty(_outputDeviceID,
- &propertyAddress);
- if (hasProperty)
- {
- size = sizeof(muted);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
- &propertyAddress, 0, NULL, &size, &muted));
-
- // 1 means muted
- enabled = static_cast<bool> (muted);
- } else
- {
- // Otherwise check if all channels are muted.
- for (UInt32 i = 1; i <= _noOutputChannels; i++)
- {
- muted = 0;
- propertyAddress.mElement = i;
- hasProperty = AudioObjectHasProperty(_outputDeviceID,
- &propertyAddress);
- if (hasProperty)
- {
- size = sizeof(channelMuted);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_outputDeviceID,
- &propertyAddress, 0, NULL, &size, &channelMuted));
-
- muted = (muted && channelMuted);
- channels++;
- }
- }
-
- if (channels == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Unable to get mute for any channel");
- return -1;
- }
-
- assert(channels > 0);
- // 1 means muted
- enabled = static_cast<bool> (muted);
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " AudioMixerManagerMac::SpeakerMute() => enabled=%d, enabled");
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::StereoPlayoutIsAvailable(bool& available)
-{
- if (_outputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- available = (_noOutputChannels == 2);
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::StereoRecordingIsAvailable(bool& available)
-{
- if (_inputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- available = (_noInputChannels == 2);
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::MicrophoneMuteIsAvailable(bool& available)
-{
- if (_inputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- OSStatus err = noErr;
-
- // Does the capture device have a master mute control?
- // If so, use it exclusively.
- AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
- kAudioDevicePropertyScopeInput, 0 };
- Boolean isSettable = false;
- err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
- &isSettable);
- if (err == noErr && isSettable)
- {
- available = true;
- return 0;
- }
-
- // Otherwise try to set each channel.
- for (UInt32 i = 1; i <= _noInputChannels; i++)
- {
- propertyAddress.mElement = i;
- isSettable = false;
- err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
- &isSettable);
- if (err != noErr || !isSettable)
- {
- available = false;
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Mute cannot be set for output channel %d, err=%d",
- i, err);
- return -1;
- }
- }
-
- available = true;
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::SetMicrophoneMute(bool enable)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerMac::SetMicrophoneMute(enable=%u)", enable);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_inputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- OSStatus err = noErr;
- UInt32 size = 0;
- UInt32 mute = enable ? 1 : 0;
- bool success = false;
-
- // Does the capture device have a master mute control?
- // If so, use it exclusively.
- AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
- kAudioDevicePropertyScopeInput, 0 };
- Boolean isSettable = false;
- err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
- &isSettable);
- if (err == noErr && isSettable)
- {
- size = sizeof(mute);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
- &propertyAddress, 0, NULL, size, &mute));
-
- return 0;
- }
-
- // Otherwise try to set each channel.
- for (UInt32 i = 1; i <= _noInputChannels; i++)
- {
- propertyAddress.mElement = i;
- isSettable = false;
- err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
- &isSettable);
- if (err == noErr && isSettable)
- {
- size = sizeof(mute);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
- &propertyAddress, 0, NULL, size, &mute));
- }
- success = true;
- }
-
- if (!success)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Unable to set mute on any input channel");
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::MicrophoneMute(bool& enabled) const
-{
-
- if (_inputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- OSStatus err = noErr;
- UInt32 size = 0;
- unsigned int channels = 0;
- UInt32 channelMuted = 0;
- UInt32 muted = 0;
-
- // Does the device have a master volume control?
- // If so, use it exclusively.
- AudioObjectPropertyAddress propertyAddress = { kAudioDevicePropertyMute,
- kAudioDevicePropertyScopeInput, 0 };
- Boolean hasProperty = AudioObjectHasProperty(_inputDeviceID,
- &propertyAddress);
- if (hasProperty)
- {
- size = sizeof(muted);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
- &propertyAddress, 0, NULL, &size, &muted));
-
- // 1 means muted
- enabled = static_cast<bool> (muted);
- } else
- {
- // Otherwise check if all channels are muted.
- for (UInt32 i = 1; i <= _noInputChannels; i++)
- {
- muted = 0;
- propertyAddress.mElement = i;
- hasProperty = AudioObjectHasProperty(_inputDeviceID,
- &propertyAddress);
- if (hasProperty)
- {
- size = sizeof(channelMuted);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
- &propertyAddress, 0, NULL, &size, &channelMuted));
-
- muted = (muted && channelMuted);
- channels++;
- }
- }
-
- if (channels == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Unable to get mute for any channel");
- return -1;
- }
-
- assert(channels > 0);
- // 1 means muted
- enabled = static_cast<bool> (muted);
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " AudioMixerManagerMac::MicrophoneMute() => enabled=%d",
- enabled);
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::MicrophoneBoostIsAvailable(bool& available)
-{
- if (_inputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- available = false; // No AudioObjectPropertySelector value for Mic Boost
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::SetMicrophoneBoost(bool enable)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerMac::SetMicrophoneBoost(enable=%u)", enable);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_inputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- // Ensure that the selected microphone has a valid boost control.
- bool available(false);
- MicrophoneBoostIsAvailable(available);
- if (!available)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " it is not possible to enable microphone boost");
- return -1;
- }
-
- // It is assumed that the call above fails!
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::MicrophoneBoost(bool& enabled) const
-{
-
- if (_inputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- // Microphone boost cannot be enabled on this platform!
- enabled = false;
-
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::MicrophoneVolumeIsAvailable(bool& available)
-{
- if (_inputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- OSStatus err = noErr;
-
- // Does the capture device have a master volume control?
- // If so, use it exclusively.
- AudioObjectPropertyAddress
- propertyAddress = { kAudioDevicePropertyVolumeScalar,
- kAudioDevicePropertyScopeInput, 0 };
- Boolean isSettable = false;
- err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
- &isSettable);
- if (err == noErr && isSettable)
- {
- available = true;
- return 0;
- }
-
- // Otherwise try to set each channel.
- for (UInt32 i = 1; i <= _noInputChannels; i++)
- {
- propertyAddress.mElement = i;
- isSettable = false;
- err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
- &isSettable);
- if (err != noErr || !isSettable)
- {
- available = false;
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Volume cannot be set for input channel %d, err=%d",
- i, err);
- return -1;
- }
- }
-
- available = true;
- return 0;
-}
-
-WebRtc_Word32 AudioMixerManagerMac::SetMicrophoneVolume(WebRtc_UWord32 volume)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioMixerManagerMac::SetMicrophoneVolume(volume=%u)", volume);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_inputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- OSStatus err = noErr;
- UInt32 size = 0;
- bool success = false;
-
- // volume range is 0.0 - 1.0, convert from 0 - 255
- const Float32 vol = (Float32)(volume / 255.0);
-
- assert(vol <= 1.0 && vol >= 0.0);
-
- // Does the capture device have a master volume control?
- // If so, use it exclusively.
- AudioObjectPropertyAddress
- propertyAddress = { kAudioDevicePropertyVolumeScalar,
- kAudioDevicePropertyScopeInput, 0 };
- Boolean isSettable = false;
- err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
- &isSettable);
- if (err == noErr && isSettable)
- {
- size = sizeof(vol);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
- &propertyAddress, 0, NULL, size, &vol));
-
- return 0;
- }
-
- // Otherwise try to set each channel.
- for (UInt32 i = 1; i <= _noInputChannels; i++)
- {
- propertyAddress.mElement = i;
- isSettable = false;
- err = AudioObjectIsPropertySettable(_inputDeviceID, &propertyAddress,
- &isSettable);
- if (err == noErr && isSettable)
- {
- size = sizeof(vol);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectSetPropertyData(_inputDeviceID,
- &propertyAddress, 0, NULL, size, &vol));
- }
- success = true;
- }
-
- if (!success)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Unable to set a level on any input channel");
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerMac::MicrophoneVolume(WebRtc_UWord32& volume) const
-{
-
- if (_inputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- OSStatus err = noErr;
- UInt32 size = 0;
- unsigned int channels = 0;
- Float32 channelVol = 0;
- Float32 volFloat32 = 0;
-
- // Does the device have a master volume control?
- // If so, use it exclusively.
- AudioObjectPropertyAddress
- propertyAddress = { kAudioDevicePropertyVolumeScalar,
- kAudioDevicePropertyScopeInput, 0 };
- Boolean hasProperty = AudioObjectHasProperty(_inputDeviceID,
- &propertyAddress);
- if (hasProperty)
- {
- size = sizeof(volFloat32);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
- &propertyAddress, 0, NULL, &size, &volFloat32));
-
- // vol 0.0 to 1.0 -> convert to 0 - 255
- volume = static_cast<WebRtc_UWord32> (volFloat32 * 255 + 0.5);
- } else
- {
- // Otherwise get the average volume across channels.
- volFloat32 = 0;
- for (UInt32 i = 1; i <= _noInputChannels; i++)
- {
- channelVol = 0;
- propertyAddress.mElement = i;
- hasProperty = AudioObjectHasProperty(_inputDeviceID,
- &propertyAddress);
- if (hasProperty)
- {
- size = sizeof(channelVol);
- WEBRTC_CA_RETURN_ON_ERR(AudioObjectGetPropertyData(_inputDeviceID,
- &propertyAddress, 0, NULL, &size, &channelVol));
-
- volFloat32 += channelVol;
- channels++;
- }
- }
-
- if (channels == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " Unable to get a level on any channel");
- return -1;
- }
-
- assert(channels > 0);
- // vol 0.0 to 1.0 -> convert to 0 - 255
- volume = static_cast<WebRtc_UWord32>
- (255 * volFloat32 / channels + 0.5);
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " AudioMixerManagerMac::MicrophoneVolume() => vol=%u",
- volume);
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerMac::MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const
-{
-
- if (_inputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- // volume range is 0.0 to 1.0
- // we convert that to 0 - 255
- maxVolume = 255;
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerMac::MinMicrophoneVolume(WebRtc_UWord32& minVolume) const
-{
-
- if (_inputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- // volume range is 0.0 to 1.0
- // we convert that to 0 - 10
- minVolume = 0;
-
- return 0;
-}
-
-WebRtc_Word32
-AudioMixerManagerMac::MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const
-{
-
- if (_inputDeviceID == kAudioObjectUnknown)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " device ID has not been set");
- return -1;
- }
-
- // volume range is 0.0 to 1.0
- // we convert that to 0 - 10
- stepSize = 1;
-
- return 0;
-}
-
-// ============================================================================
-// Private Methods
-// ============================================================================
-
-// CoreAudio errors are best interpreted as four character strings.
-void AudioMixerManagerMac::logCAMsg(const TraceLevel level,
- const TraceModule module,
- const WebRtc_Word32 id, const char *msg,
- const char *err)
-{
- assert(msg != NULL);
- assert(err != NULL);
-
-#ifdef WEBRTC_BIG_ENDIAN
- WEBRTC_TRACE(level, module, id, "%s: %.4s", msg, err);
-#else
- // We need to flip the characters in this case.
- WEBRTC_TRACE(level, module, id, "%s: %.1s%.1s%.1s%.1s", msg, err + 3, err
- + 2, err + 1, err);
-#endif
-}
-
-} // namespace webrtc
-// EOF
diff --git a/trunk/src/modules/audio_device/main/source/mac/audio_mixer_manager_mac.h b/trunk/src/modules/audio_device/main/source/mac/audio_mixer_manager_mac.h
deleted file mode 100644
index 7209f91..0000000
--- a/trunk/src/modules/audio_device/main/source/mac/audio_mixer_manager_mac.h
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_MAC_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_MAC_H
-
-#include "typedefs.h"
-#include "audio_device.h"
-#include "critical_section_wrapper.h"
-
-#include <CoreAudio/CoreAudio.h>
-
-namespace webrtc {
-
-class AudioMixerManagerMac
-{
-public:
- WebRtc_Word32 OpenSpeaker(AudioDeviceID deviceID);
- WebRtc_Word32 OpenMicrophone(AudioDeviceID deviceID);
- WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
- WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
- WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
- WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
- WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
- WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
- WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
- WebRtc_Word32 SetSpeakerMute(bool enable);
- WebRtc_Word32 SpeakerMute(bool& enabled) const;
- WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
- WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
- WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
- WebRtc_Word32 SetMicrophoneMute(bool enable);
- WebRtc_Word32 MicrophoneMute(bool& enabled) const;
- WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
- WebRtc_Word32 SetMicrophoneBoost(bool enable);
- WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
- WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
- WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
- WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
- WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
- WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
- WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
- WebRtc_Word32 Close();
- WebRtc_Word32 CloseSpeaker();
- WebRtc_Word32 CloseMicrophone();
- bool SpeakerIsInitialized() const;
- bool MicrophoneIsInitialized() const;
-
-public:
- AudioMixerManagerMac(const WebRtc_Word32 id);
- ~AudioMixerManagerMac();
-
-private:
- static void logCAMsg(const TraceLevel level,
- const TraceModule module,
- const WebRtc_Word32 id, const char *msg,
- const char *err);
-
-private:
- CriticalSectionWrapper& _critSect;
- WebRtc_Word32 _id;
-
- AudioDeviceID _inputDeviceID;
- AudioDeviceID _outputDeviceID;
-
- WebRtc_UWord16 _noInputChannels;
- WebRtc_UWord16 _noOutputChannels;
-
-};
-
-} //namespace webrtc
-
-#endif // AUDIO_MIXER_MAC_H
diff --git a/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_memorybarrier.h b/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_memorybarrier.h
deleted file mode 100644
index f689622..0000000
--- a/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_memorybarrier.h
+++ /dev/null
@@ -1,127 +0,0 @@
-/*
- * $Id: pa_memorybarrier.h 1240 2007-07-17 13:05:07Z bjornroche $
- * Portable Audio I/O Library
- * Memory barrier utilities
- *
- * Author: Bjorn Roche, XO Audio, LLC
- *
- * This program uses the PortAudio Portable Audio Library.
- * For more information see: http://www.portaudio.com
- * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-/*
- * The text above constitutes the entire PortAudio license; however,
- * the PortAudio community also makes the following non-binding requests:
- *
- * Any person wishing to distribute modifications to the Software is
- * requested to send the modifications to the original developer so that
- * they can be incorporated into the canonical version. It is also
- * requested that these non-binding requests be included along with the
- * license above.
- */
-
-/**
- @file pa_memorybarrier.h
- @ingroup common_src
-*/
-
-/****************
- * Some memory barrier primitives based on the system.
- * right now only OS X, FreeBSD, and Linux are supported. In addition to providing
- * memory barriers, these functions should ensure that data cached in registers
- * is written out to cache where it can be snooped by other CPUs. (ie, the volatile
- * keyword should not be required)
- *
- * the primitives that must be defined are:
- *
- * PaUtil_FullMemoryBarrier()
- * PaUtil_ReadMemoryBarrier()
- * PaUtil_WriteMemoryBarrier()
- *
- ****************/
-
-#if defined(__APPLE__)
-# include <libkern/OSAtomic.h>
- /* Here are the memory barrier functions. Mac OS X only provides
- full memory barriers, so the three types of barriers are the same,
- however, these barriers are superior to compiler-based ones. */
-# define PaUtil_FullMemoryBarrier() OSMemoryBarrier()
-# define PaUtil_ReadMemoryBarrier() OSMemoryBarrier()
-# define PaUtil_WriteMemoryBarrier() OSMemoryBarrier()
-#elif defined(__GNUC__)
- /* GCC >= 4.1 has built-in intrinsics. We'll use those */
-# if (__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 1)
-# define PaUtil_FullMemoryBarrier() __sync_synchronize()
-# define PaUtil_ReadMemoryBarrier() __sync_synchronize()
-# define PaUtil_WriteMemoryBarrier() __sync_synchronize()
- /* as a fallback, GCC understands volatile asm and "memory" to mean it
- * should not reorder memory read/writes */
- /* Note that it is not clear that any compiler actually defines __PPC__,
- * it can probably removed safely. */
-# elif defined( __ppc__ ) || defined( __powerpc__) || defined( __PPC__ )
-# define PaUtil_FullMemoryBarrier() asm volatile("sync":::"memory")
-# define PaUtil_ReadMemoryBarrier() asm volatile("sync":::"memory")
-# define PaUtil_WriteMemoryBarrier() asm volatile("sync":::"memory")
-# elif defined( __i386__ ) || defined( __i486__ ) || defined( __i586__ ) || \
- defined( __i686__ ) || defined( __x86_64__ )
-# define PaUtil_FullMemoryBarrier() asm volatile("mfence":::"memory")
-# define PaUtil_ReadMemoryBarrier() asm volatile("lfence":::"memory")
-# define PaUtil_WriteMemoryBarrier() asm volatile("sfence":::"memory")
-# else
-# ifdef ALLOW_SMP_DANGERS
-# warning Memory barriers not defined on this system or system unknown
-# warning For SMP safety, you should fix this.
-# define PaUtil_FullMemoryBarrier()
-# define PaUtil_ReadMemoryBarrier()
-# define PaUtil_WriteMemoryBarrier()
-# else
-# error Memory barriers are not defined on this system. You can still compile by defining ALLOW_SMP_DANGERS, but SMP safety will not be guaranteed.
-# endif
-# endif
-#elif (_MSC_VER >= 1400) && !defined(_WIN32_WCE)
-# include <intrin.h>
-# pragma intrinsic(_ReadWriteBarrier)
-# pragma intrinsic(_ReadBarrier)
-# pragma intrinsic(_WriteBarrier)
-# define PaUtil_FullMemoryBarrier() _ReadWriteBarrier()
-# define PaUtil_ReadMemoryBarrier() _ReadBarrier()
-# define PaUtil_WriteMemoryBarrier() _WriteBarrier()
-#elif defined(_WIN32_WCE)
-# define PaUtil_FullMemoryBarrier()
-# define PaUtil_ReadMemoryBarrier()
-# define PaUtil_WriteMemoryBarrier()
-#elif defined(_MSC_VER) || defined(__BORLANDC__)
-# define PaUtil_FullMemoryBarrier() _asm { lock add [esp], 0 }
-# define PaUtil_ReadMemoryBarrier() _asm { lock add [esp], 0 }
-# define PaUtil_WriteMemoryBarrier() _asm { lock add [esp], 0 }
-#else
-# ifdef ALLOW_SMP_DANGERS
-# warning Memory barriers not defined on this system or system unknown
-# warning For SMP safety, you should fix this.
-# define PaUtil_FullMemoryBarrier()
-# define PaUtil_ReadMemoryBarrier()
-# define PaUtil_WriteMemoryBarrier()
-# else
-# error Memory barriers are not defined on this system. You can still compile by defining ALLOW_SMP_DANGERS, but SMP safety will not be guaranteed.
-# endif
-#endif
diff --git a/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_ringbuffer.c b/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_ringbuffer.c
deleted file mode 100644
index 310d719..0000000
--- a/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_ringbuffer.c
+++ /dev/null
@@ -1,227 +0,0 @@
-/*
- * $Id: pa_ringbuffer.c 1421 2009-11-18 16:09:05Z bjornroche $
- * Portable Audio I/O Library
- * Ring Buffer utility.
- *
- * Author: Phil Burk, http://www.softsynth.com
- * modified for SMP safety on Mac OS X by Bjorn Roche
- * modified for SMP safety on Linux by Leland Lucius
- * also, allowed for const where possible
- * modified for multiple-byte-sized data elements by Sven Fischer
- *
- * Note that this is safe only for a single-thread reader and a
- * single-thread writer.
- *
- * This program uses the PortAudio Portable Audio Library.
- * For more information see: http://www.portaudio.com
- * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-/*
- * The text above constitutes the entire PortAudio license; however,
- * the PortAudio community also makes the following non-binding requests:
- *
- * Any person wishing to distribute modifications to the Software is
- * requested to send the modifications to the original developer so that
- * they can be incorporated into the canonical version. It is also
- * requested that these non-binding requests be included along with the
- * license above.
- */
-
-/**
- @file
- @ingroup common_src
-*/
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <math.h>
-#include "pa_ringbuffer.h"
-#include <string.h>
-#include "pa_memorybarrier.h"
-
-/***************************************************************************
- * Initialize FIFO.
- * elementCount must be power of 2, returns -1 if not.
- */
-ring_buffer_size_t PaUtil_InitializeRingBuffer( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementSizeBytes, ring_buffer_size_t elementCount, void *dataPtr )
-{
- if( ((elementCount-1) & elementCount) != 0) return -1; /* Not Power of two. */
- rbuf->bufferSize = elementCount;
- rbuf->buffer = (char *)dataPtr;
- PaUtil_FlushRingBuffer( rbuf );
- rbuf->bigMask = (elementCount*2)-1;
- rbuf->smallMask = (elementCount)-1;
- rbuf->elementSizeBytes = elementSizeBytes;
- return 0;
-}
-
-/***************************************************************************
-** Return number of elements available for reading. */
-ring_buffer_size_t PaUtil_GetRingBufferReadAvailable( PaUtilRingBuffer *rbuf )
-{
- PaUtil_ReadMemoryBarrier();
- return ( (rbuf->writeIndex - rbuf->readIndex) & rbuf->bigMask );
-}
-/***************************************************************************
-** Return number of elements available for writing. */
-ring_buffer_size_t PaUtil_GetRingBufferWriteAvailable( PaUtilRingBuffer *rbuf )
-{
- /* Since we are calling PaUtil_GetRingBufferReadAvailable, we don't need an aditional MB */
- return ( rbuf->bufferSize - PaUtil_GetRingBufferReadAvailable(rbuf));
-}
-
-/***************************************************************************
-** Clear buffer. Should only be called when buffer is NOT being read. */
-void PaUtil_FlushRingBuffer( PaUtilRingBuffer *rbuf )
-{
- rbuf->writeIndex = rbuf->readIndex = 0;
-}
-
-/***************************************************************************
-** Get address of region(s) to which we can write data.
-** If the region is contiguous, size2 will be zero.
-** If non-contiguous, size2 will be the size of second region.
-** Returns room available to be written or elementCount, whichever is smaller.
-*/
-ring_buffer_size_t PaUtil_GetRingBufferWriteRegions( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount,
- void **dataPtr1, ring_buffer_size_t *sizePtr1,
- void **dataPtr2, ring_buffer_size_t *sizePtr2 )
-{
- ring_buffer_size_t index;
- ring_buffer_size_t available = PaUtil_GetRingBufferWriteAvailable( rbuf );
- if( elementCount > available ) elementCount = available;
- /* Check to see if write is not contiguous. */
- index = rbuf->writeIndex & rbuf->smallMask;
- if( (index + elementCount) > rbuf->bufferSize )
- {
- /* Write data in two blocks that wrap the buffer. */
- ring_buffer_size_t firstHalf = rbuf->bufferSize - index;
- *dataPtr1 = &rbuf->buffer[index*rbuf->elementSizeBytes];
- *sizePtr1 = firstHalf;
- *dataPtr2 = &rbuf->buffer[0];
- *sizePtr2 = elementCount - firstHalf;
- }
- else
- {
- *dataPtr1 = &rbuf->buffer[index*rbuf->elementSizeBytes];
- *sizePtr1 = elementCount;
- *dataPtr2 = NULL;
- *sizePtr2 = 0;
- }
- return elementCount;
-}
-
-
-/***************************************************************************
-*/
-ring_buffer_size_t PaUtil_AdvanceRingBufferWriteIndex( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount )
-{
- /* we need to ensure that previous writes are seen before we update the write index */
- PaUtil_WriteMemoryBarrier();
- return rbuf->writeIndex = (rbuf->writeIndex + elementCount) & rbuf->bigMask;
-}
-
-/***************************************************************************
-** Get address of region(s) from which we can read data.
-** If the region is contiguous, size2 will be zero.
-** If non-contiguous, size2 will be the size of second region.
-** Returns room available to be written or elementCount, whichever is smaller.
-*/
-ring_buffer_size_t PaUtil_GetRingBufferReadRegions( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount,
- void **dataPtr1, ring_buffer_size_t *sizePtr1,
- void **dataPtr2, ring_buffer_size_t *sizePtr2 )
-{
- ring_buffer_size_t index;
- ring_buffer_size_t available = PaUtil_GetRingBufferReadAvailable( rbuf );
- if( elementCount > available ) elementCount = available;
- /* Check to see if read is not contiguous. */
- index = rbuf->readIndex & rbuf->smallMask;
- if( (index + elementCount) > rbuf->bufferSize )
- {
- /* Write data in two blocks that wrap the buffer. */
- ring_buffer_size_t firstHalf = rbuf->bufferSize - index;
- *dataPtr1 = &rbuf->buffer[index*rbuf->elementSizeBytes];
- *sizePtr1 = firstHalf;
- *dataPtr2 = &rbuf->buffer[0];
- *sizePtr2 = elementCount - firstHalf;
- }
- else
- {
- *dataPtr1 = &rbuf->buffer[index*rbuf->elementSizeBytes];
- *sizePtr1 = elementCount;
- *dataPtr2 = NULL;
- *sizePtr2 = 0;
- }
- return elementCount;
-}
-/***************************************************************************
-*/
-ring_buffer_size_t PaUtil_AdvanceRingBufferReadIndex( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount )
-{
- /* we need to ensure that previous writes are always seen before updating the index. */
- PaUtil_WriteMemoryBarrier();
- return rbuf->readIndex = (rbuf->readIndex + elementCount) & rbuf->bigMask;
-}
-
-/***************************************************************************
-** Return elements written. */
-ring_buffer_size_t PaUtil_WriteRingBuffer( PaUtilRingBuffer *rbuf, const void *data, ring_buffer_size_t elementCount )
-{
- ring_buffer_size_t size1, size2, numWritten;
- void *data1, *data2;
- numWritten = PaUtil_GetRingBufferWriteRegions( rbuf, elementCount, &data1, &size1, &data2, &size2 );
- if( size2 > 0 )
- {
-
- memcpy( data1, data, size1*rbuf->elementSizeBytes );
- data = ((char *)data) + size1*rbuf->elementSizeBytes;
- memcpy( data2, data, size2*rbuf->elementSizeBytes );
- }
- else
- {
- memcpy( data1, data, size1*rbuf->elementSizeBytes );
- }
- PaUtil_AdvanceRingBufferWriteIndex( rbuf, numWritten );
- return numWritten;
-}
-
-/***************************************************************************
-** Return elements read. */
-ring_buffer_size_t PaUtil_ReadRingBuffer( PaUtilRingBuffer *rbuf, void *data, ring_buffer_size_t elementCount )
-{
- ring_buffer_size_t size1, size2, numRead;
- void *data1, *data2;
- numRead = PaUtil_GetRingBufferReadRegions( rbuf, elementCount, &data1, &size1, &data2, &size2 );
- if( size2 > 0 )
- {
- memcpy( data, data1, size1*rbuf->elementSizeBytes );
- data = ((char *)data) + size1*rbuf->elementSizeBytes;
- memcpy( data, data2, size2*rbuf->elementSizeBytes );
- }
- else
- {
- memcpy( data, data1, size1*rbuf->elementSizeBytes );
- }
- PaUtil_AdvanceRingBufferReadIndex( rbuf, numRead );
- return numRead;
-}
diff --git a/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_ringbuffer.h b/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_ringbuffer.h
deleted file mode 100644
index 393f6f8..0000000
--- a/trunk/src/modules/audio_device/main/source/mac/portaudio/pa_ringbuffer.h
+++ /dev/null
@@ -1,233 +0,0 @@
-#ifndef WEBRTC_AUDIO_DEVICE_PA_RINGBUFFER_H
-#define WEBRTC_AUDIO_DEVICE_PA_RINGBUFFER_H
-/*
- * $Id: pa_ringbuffer.h 1421 2009-11-18 16:09:05Z bjornroche $
- * Portable Audio I/O Library
- * Ring Buffer utility.
- *
- * Author: Phil Burk, http://www.softsynth.com
- * modified for SMP safety on OS X by Bjorn Roche.
- * also allowed for const where possible.
- * modified for multiple-byte-sized data elements by Sven Fischer
- *
- * Note that this is safe only for a single-thread reader
- * and a single-thread writer.
- *
- * This program is distributed with the PortAudio Portable Audio Library.
- * For more information see: http://www.portaudio.com
- * Copyright (c) 1999-2000 Ross Bencina and Phil Burk
- *
- * Permission is hereby granted, free of charge, to any person obtaining
- * a copy of this software and associated documentation files
- * (the "Software"), to deal in the Software without restriction,
- * including without limitation the rights to use, copy, modify, merge,
- * publish, distribute, sublicense, and/or sell copies of the Software,
- * and to permit persons to whom the Software is furnished to do so,
- * subject to the following conditions:
- *
- * The above copyright notice and this permission notice shall be
- * included in all copies or substantial portions of the Software.
- *
- * THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND,
- * EXPRESS OR IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF
- * MERCHANTABILITY, FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
- * IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR
- * ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF
- * CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION
- * WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE.
- */
-
-/*
- * The text above constitutes the entire PortAudio license; however,
- * the PortAudio community also makes the following non-binding requests:
- *
- * Any person wishing to distribute modifications to the Software is
- * requested to send the modifications to the original developer so that
- * they can be incorporated into the canonical version. It is also
- * requested that these non-binding requests be included along with the
- * license above.
- */
-
-/** @file
- @ingroup common_src
- @brief Single-reader single-writer lock-free ring buffer
-
- PaUtilRingBuffer is a ring buffer used to transport samples between
- different execution contexts (threads, OS callbacks, interrupt handlers)
- without requiring the use of any locks. This only works when there is
- a single reader and a single writer (ie. one thread or callback writes
- to the ring buffer, another thread or callback reads from it).
-
- The PaUtilRingBuffer structure manages a ring buffer containing N
- elements, where N must be a power of two. An element may be any size
- (specified in bytes).
-
- The memory area used to store the buffer elements must be allocated by
- the client prior to calling PaUtil_InitializeRingBuffer() and must outlive
- the use of the ring buffer.
-*/
-
-#if defined(__APPLE__)
-#include <sys/types.h>
-typedef int32_t ring_buffer_size_t;
-#elif defined( __GNUC__ )
-typedef long ring_buffer_size_t;
-#elif (_MSC_VER >= 1400)
-typedef long ring_buffer_size_t;
-#elif defined(_MSC_VER) || defined(__BORLANDC__)
-typedef long ring_buffer_size_t;
-#else
-typedef long ring_buffer_size_t;
-#endif
-
-
-
-#ifdef __cplusplus
-extern "C"
-{
-#endif /* __cplusplus */
-
-typedef struct PaUtilRingBuffer
-{
- ring_buffer_size_t bufferSize; /**< Number of elements in FIFO. Power of 2. Set by PaUtil_InitRingBuffer. */
- ring_buffer_size_t writeIndex; /**< Index of next writable element. Set by PaUtil_AdvanceRingBufferWriteIndex. */
- ring_buffer_size_t readIndex; /**< Index of next readable element. Set by PaUtil_AdvanceRingBufferReadIndex. */
- ring_buffer_size_t bigMask; /**< Used for wrapping indices with extra bit to distinguish full/empty. */
- ring_buffer_size_t smallMask; /**< Used for fitting indices to buffer. */
- ring_buffer_size_t elementSizeBytes; /**< Number of bytes per element. */
- char *buffer; /**< Pointer to the buffer containing the actual data. */
-}PaUtilRingBuffer;
-
-/** Initialize Ring Buffer.
-
- @param rbuf The ring buffer.
-
- @param elementSizeBytes The size of a single data element in bytes.
-
- @param elementCount The number of elements in the buffer (must be power of 2).
-
- @param dataPtr A pointer to a previously allocated area where the data
- will be maintained. It must be elementCount*elementSizeBytes long.
-
- @return -1 if elementCount is not a power of 2, otherwise 0.
-*/
-ring_buffer_size_t PaUtil_InitializeRingBuffer( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementSizeBytes, ring_buffer_size_t elementCount, void *dataPtr );
-
-/** Clear buffer. Should only be called when buffer is NOT being read.
-
- @param rbuf The ring buffer.
-*/
-void PaUtil_FlushRingBuffer( PaUtilRingBuffer *rbuf );
-
-/** Retrieve the number of elements available in the ring buffer for writing.
-
- @param rbuf The ring buffer.
-
- @return The number of elements available for writing.
-*/
-ring_buffer_size_t PaUtil_GetRingBufferWriteAvailable( PaUtilRingBuffer *rbuf );
-
-/** Retrieve the number of elements available in the ring buffer for reading.
-
- @param rbuf The ring buffer.
-
- @return The number of elements available for reading.
-*/
-ring_buffer_size_t PaUtil_GetRingBufferReadAvailable( PaUtilRingBuffer *rbuf );
-
-/** Write data to the ring buffer.
-
- @param rbuf The ring buffer.
-
- @param data The address of new data to write to the buffer.
-
- @param elementCount The number of elements to be written.
-
- @return The number of elements written.
-*/
-ring_buffer_size_t PaUtil_WriteRingBuffer( PaUtilRingBuffer *rbuf, const void *data, ring_buffer_size_t elementCount );
-
-/** Read data from the ring buffer.
-
- @param rbuf The ring buffer.
-
- @param data The address where the data should be stored.
-
- @param elementCount The number of elements to be read.
-
- @return The number of elements read.
-*/
-ring_buffer_size_t PaUtil_ReadRingBuffer( PaUtilRingBuffer *rbuf, void *data, ring_buffer_size_t elementCount );
-
-/** Get address of region(s) to which we can write data.
-
- @param rbuf The ring buffer.
-
- @param elementCount The number of elements desired.
-
- @param dataPtr1 The address where the first (or only) region pointer will be
- stored.
-
- @param sizePtr1 The address where the first (or only) region length will be
- stored.
-
- @param dataPtr2 The address where the second region pointer will be stored if
- the first region is too small to satisfy elementCount.
-
- @param sizePtr2 The address where the second region length will be stored if
- the first region is too small to satisfy elementCount.
-
- @return The room available to be written or elementCount, whichever is smaller.
-*/
-ring_buffer_size_t PaUtil_GetRingBufferWriteRegions( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount,
- void **dataPtr1, ring_buffer_size_t *sizePtr1,
- void **dataPtr2, ring_buffer_size_t *sizePtr2 );
-
-/** Advance the write index to the next location to be written.
-
- @param rbuf The ring buffer.
-
- @param elementCount The number of elements to advance.
-
- @return The new position.
-*/
-ring_buffer_size_t PaUtil_AdvanceRingBufferWriteIndex( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount );
-
-/** Get address of region(s) from which we can write data.
-
- @param rbuf The ring buffer.
-
- @param elementCount The number of elements desired.
-
- @param dataPtr1 The address where the first (or only) region pointer will be
- stored.
-
- @param sizePtr1 The address where the first (or only) region length will be
- stored.
-
- @param dataPtr2 The address where the second region pointer will be stored if
- the first region is too small to satisfy elementCount.
-
- @param sizePtr2 The address where the second region length will be stored if
- the first region is too small to satisfy elementCount.
-
- @return The number of elements available for reading.
-*/
-ring_buffer_size_t PaUtil_GetRingBufferReadRegions( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount,
- void **dataPtr1, ring_buffer_size_t *sizePtr1,
- void **dataPtr2, ring_buffer_size_t *sizePtr2 );
-
-/** Advance the read index to the next location to be read.
-
- @param rbuf The ring buffer.
-
- @param elementCount The number of elements to advance.
-
- @return The new position.
-*/
-ring_buffer_size_t PaUtil_AdvanceRingBufferReadIndex( PaUtilRingBuffer *rbuf, ring_buffer_size_t elementCount );
-
-#ifdef __cplusplus
-}
-#endif /* __cplusplus */
-#endif /* MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_PORTAUDIO_PA_RINGBUFFER_H_ */
diff --git a/trunk/src/modules/audio_device/main/source/win/audio_device_core_win.cc b/trunk/src/modules/audio_device/main/source/win/audio_device_core_win.cc
deleted file mode 100644
index d0418d2..0000000
--- a/trunk/src/modules/audio_device/main/source/win/audio_device_core_win.cc
+++ /dev/null
@@ -1,5173 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#pragma warning(disable: 4995) // name was marked as #pragma deprecated
-
-#if (_MSC_VER >= 1310) && (_MSC_VER < 1400)
-// Reports the major and minor versions of the compiler.
-// For example, 1310 for Microsoft Visual C++ .NET 2003. 1310 represents version 13 and a 1.0 point release.
-// The Visual C++ 2005 compiler version is 1400.
-// Type cl /? at the command line to see the major and minor versions of your compiler along with the build number.
-#pragma message(">> INFO: Windows Core Audio is not supported in VS 2003")
-#endif
-
-#include "audio_device_config.h"
-
-#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
-#pragma message(">> INFO: WEBRTC_WINDOWS_CORE_AUDIO_BUILD is defined")
-#else
-#pragma message(">> INFO: WEBRTC_WINDOWS_CORE_AUDIO_BUILD is *not* defined")
-#endif
-
-#ifdef WEBRTC_WINDOWS_CORE_AUDIO_BUILD
-
-#include "audio_device_core_win.h"
-
-#include <assert.h>
-#include <string.h>
-
-#include <windows.h>
-#include <comdef.h>
-#include <dmo.h>
-#include "Functiondiscoverykeys_devpkey.h"
-#include <mmsystem.h>
-#include <strsafe.h>
-#include <uuids.h>
-
-#include "audio_device_utility.h"
-#include "trace.h"
-
-// Macro that calls a COM method returning HRESULT value.
-#define EXIT_ON_ERROR(hres) do { if (FAILED(hres)) goto Exit; } while(0)
-
-// Macro that releases a COM object if not NULL.
-#define SAFE_RELEASE(p) do { if ((p)) { (p)->Release(); (p) = NULL; } } while(0)
-
-#define ROUND(x) ((x) >=0 ? (int)((x) + 0.5) : (int)((x) - 0.5))
-
-// REFERENCE_TIME time units per millisecond
-#define REFTIMES_PER_MILLISEC 10000
-
-typedef struct tagTHREADNAME_INFO
-{
- DWORD dwType; // must be 0x1000
- LPCSTR szName; // pointer to name (in user addr space)
- DWORD dwThreadID; // thread ID (-1=caller thread)
- DWORD dwFlags; // reserved for future use, must be zero
-} THREADNAME_INFO;
-
-namespace webrtc {
-namespace {
-
-enum { COM_THREADING_MODEL = COINIT_MULTITHREADED };
-
-enum
-{
- kAecCaptureStreamIndex = 0,
- kAecRenderStreamIndex = 1
-};
-
-// An implementation of IMediaBuffer, as required for
-// IMediaObject::ProcessOutput(). After consuming data provided by
-// ProcessOutput(), call SetLength() to update the buffer availability.
-//
-// Example implementation:
-// http://msdn.microsoft.com/en-us/library/dd376684(v=vs.85).aspx
-class MediaBufferImpl : public IMediaBuffer
-{
-public:
- explicit MediaBufferImpl(DWORD maxLength)
- : _data(new BYTE[maxLength]),
- _length(0),
- _maxLength(maxLength),
- _refCount(0)
- {}
-
- // IMediaBuffer methods.
- STDMETHOD(GetBufferAndLength(BYTE** ppBuffer, DWORD* pcbLength))
- {
- if (!ppBuffer || !pcbLength)
- {
- return E_POINTER;
- }
-
- *ppBuffer = _data;
- *pcbLength = _length;
-
- return S_OK;
- }
-
- STDMETHOD(GetMaxLength(DWORD* pcbMaxLength))
- {
- if (!pcbMaxLength)
- {
- return E_POINTER;
- }
-
- *pcbMaxLength = _maxLength;
- return S_OK;
- }
-
- STDMETHOD(SetLength(DWORD cbLength))
- {
- if (cbLength > _maxLength)
- {
- return E_INVALIDARG;
- }
-
- _length = cbLength;
- return S_OK;
- }
-
- // IUnknown methods.
- STDMETHOD_(ULONG, AddRef())
- {
- return InterlockedIncrement(&_refCount);
- }
-
- STDMETHOD(QueryInterface(REFIID riid, void** ppv))
- {
- if (!ppv)
- {
- return E_POINTER;
- }
- else if (riid != IID_IMediaBuffer && riid != IID_IUnknown)
- {
- return E_NOINTERFACE;
- }
-
- *ppv = static_cast<IMediaBuffer*>(this);
- AddRef();
- return S_OK;
- }
-
- STDMETHOD_(ULONG, Release())
- {
- LONG refCount = InterlockedDecrement(&_refCount);
- if (refCount == 0)
- {
- delete this;
- }
-
- return refCount;
- }
-
-private:
- ~MediaBufferImpl()
- {
- delete [] _data;
- }
-
- BYTE* _data;
- DWORD _length;
- const DWORD _maxLength;
- LONG _refCount;
-};
-} // namespace
-
-// ============================================================================
-// Static Methods
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// CoreAudioIsSupported
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsCore::CoreAudioIsSupported()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, -1, "%s", __FUNCTION__);
-
- bool MMDeviceIsAvailable(false);
- bool coreAudioIsSupported(false);
-
- HRESULT hr(S_OK);
- TCHAR buf[MAXERRORLENGTH];
- TCHAR errorText[MAXERRORLENGTH];
-
- // 1) Check if Windows version is Vista SP1 or later.
- //
- // CoreAudio is only available on Vista SP1 and later.
- //
- OSVERSIONINFOEX osvi;
- DWORDLONG dwlConditionMask = 0;
- int op = VER_LESS_EQUAL;
-
- // Initialize the OSVERSIONINFOEX structure.
- ZeroMemory(&osvi, sizeof(OSVERSIONINFOEX));
- osvi.dwOSVersionInfoSize = sizeof(OSVERSIONINFOEX);
- osvi.dwMajorVersion = 6;
- osvi.dwMinorVersion = 0;
- osvi.wServicePackMajor = 0;
- osvi.wServicePackMinor = 0;
- osvi.wProductType = VER_NT_WORKSTATION;
-
- // Initialize the condition mask.
- VER_SET_CONDITION(dwlConditionMask, VER_MAJORVERSION, op);
- VER_SET_CONDITION(dwlConditionMask, VER_MINORVERSION, op);
- VER_SET_CONDITION(dwlConditionMask, VER_SERVICEPACKMAJOR, op);
- VER_SET_CONDITION(dwlConditionMask, VER_SERVICEPACKMINOR, op);
- VER_SET_CONDITION(dwlConditionMask, VER_PRODUCT_TYPE, VER_EQUAL);
-
- DWORD dwTypeMask = VER_MAJORVERSION | VER_MINORVERSION |
- VER_SERVICEPACKMAJOR | VER_SERVICEPACKMINOR |
- VER_PRODUCT_TYPE;
-
- // Perform the test.
- BOOL isVistaRTMorXP = VerifyVersionInfo(&osvi, dwTypeMask,
- dwlConditionMask);
- if (isVistaRTMorXP != 0)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1,
- "*** Windows Core Audio is only supported on Vista SP1 or later "
- "=> will revert to the Wave API ***");
- return false;
- }
-
- // 2) Initializes the COM library for use by the calling thread.
-
- // The COM init wrapper sets the thread's concurrency model to MTA,
- // and creates a new apartment for the thread if one is required. The
- // wrapper also ensures that each call to CoInitializeEx is balanced
- // by a corresponding call to CoUninitialize.
- //
- ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
- if (!comInit.succeeded()) {
- // Things will work even if an STA thread is calling this method but we
- // want to ensure that MTA is used and therefore return false here.
- return false;
- }
-
- // 3) Check if the MMDevice API is available.
- //
- // The Windows Multimedia Device (MMDevice) API enables audio clients to
- // discover audio endpoint devices, determine their capabilities, and create
- // driver instances for those devices.
- // Header file Mmdeviceapi.h defines the interfaces in the MMDevice API.
- // The MMDevice API consists of several interfaces. The first of these is the
- // IMMDeviceEnumerator interface. To access the interfaces in the MMDevice API,
- // a client obtains a reference to the IMMDeviceEnumerator interface of a
- // device-enumerator object by calling the CoCreateInstance function.
- //
- // Through the IMMDeviceEnumerator interface, the client can obtain references
- // to the other interfaces in the MMDevice API. The MMDevice API implements
- // the following interfaces:
- //
- // IMMDevice Represents an audio device.
- // IMMDeviceCollection Represents a collection of audio devices.
- // IMMDeviceEnumerator Provides methods for enumerating audio devices.
- // IMMEndpoint Represents an audio endpoint device.
- //
- IMMDeviceEnumerator* pIMMD(NULL);
- const CLSID CLSID_MMDeviceEnumerator = __uuidof(MMDeviceEnumerator);
- const IID IID_IMMDeviceEnumerator = __uuidof(IMMDeviceEnumerator);
-
- hr = CoCreateInstance(
- CLSID_MMDeviceEnumerator, // GUID value of MMDeviceEnumerator coclass
- NULL,
- CLSCTX_ALL,
- IID_IMMDeviceEnumerator, // GUID value of the IMMDeviceEnumerator interface
- (void**)&pIMMD );
-
- if (FAILED(hr))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, -1,
- "AudioDeviceWindowsCore::CoreAudioIsSupported() Failed to create the required COM object", hr);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, -1,
- "AudioDeviceWindowsCore::CoreAudioIsSupported() CoCreateInstance(MMDeviceEnumerator) failed (hr=0x%x)", hr);
-
- const DWORD dwFlags = FORMAT_MESSAGE_FROM_SYSTEM |
- FORMAT_MESSAGE_IGNORE_INSERTS;
- const DWORD dwLangID = MAKELANGID(LANG_ENGLISH, SUBLANG_ENGLISH_US);
-
- // Gets the system's human readable message string for this HRESULT.
- // All error message in English by default.
- DWORD messageLength = ::FormatMessageW(dwFlags,
- 0,
- hr,
- dwLangID,
- errorText,
- MAXERRORLENGTH,
- NULL);
-
- assert(messageLength <= MAXERRORLENGTH);
-
- // Trims tailing white space (FormatMessage() leaves a trailing cr-lf.).
- for (; messageLength && ::isspace(errorText[messageLength - 1]);
- --messageLength)
- {
- errorText[messageLength - 1] = '\0';
- }
-
- StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
- StringCchCat(buf, MAXERRORLENGTH, errorText);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, -1, "%S", buf);
- }
- else
- {
- MMDeviceIsAvailable = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, -1,
- "AudioDeviceWindowsCore::CoreAudioIsSupported() CoCreateInstance(MMDeviceEnumerator) succeeded", hr);
- SAFE_RELEASE(pIMMD);
- }
-
- // 4) Verify that we can create and initialize our Core Audio class.
- //
- // Also, perform a limited "API test" to ensure that Core Audio is supported for all devices.
- //
- if (MMDeviceIsAvailable)
- {
- coreAudioIsSupported = false;
-
- AudioDeviceWindowsCore* p = new AudioDeviceWindowsCore(-1);
- if (p == NULL)
- {
- return false;
- }
-
- int ok(0);
- int temp_ok(0);
- bool available(false);
-
- ok |= p->Init();
-
- WebRtc_Word16 numDevsRec = p->RecordingDevices();
- for (WebRtc_UWord16 i = 0; i < numDevsRec; i++)
- {
- ok |= p->SetRecordingDevice(i);
- temp_ok = p->RecordingIsAvailable(available);
- ok |= temp_ok;
- ok |= (available == false);
- if (available)
- {
- ok |= p->InitMicrophone();
- }
- if (ok)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, -1,
- "AudioDeviceWindowsCore::CoreAudioIsSupported() Failed to use Core Audio Recording for device id=%i", i);
- }
- }
-
- WebRtc_Word16 numDevsPlay = p->PlayoutDevices();
- for (WebRtc_UWord16 i = 0; i < numDevsPlay; i++)
- {
- ok |= p->SetPlayoutDevice(i);
- temp_ok = p->PlayoutIsAvailable(available);
- ok |= temp_ok;
- ok |= (available == false);
- if (available)
- {
- ok |= p->InitSpeaker();
- }
- if (ok)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, -1 ,
- "AudioDeviceWindowsCore::CoreAudioIsSupported() Failed to use Core Audio Playout for device id=%i", i);
- }
- }
-
- ok |= p->Terminate();
-
- if (ok == 0)
- {
- coreAudioIsSupported = true;
- }
-
- delete p;
- }
-
- if (coreAudioIsSupported)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1, "*** Windows Core Audio is supported ***");
- }
- else
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, -1, "*** Windows Core Audio is NOT supported => will revert to the Wave API ***");
- }
-
- return (coreAudioIsSupported);
-}
-
-// ============================================================================
-// Construction & Destruction
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// AudioDeviceWindowsCore() - ctor
-// ----------------------------------------------------------------------------
-
-AudioDeviceWindowsCore::AudioDeviceWindowsCore(const WebRtc_Word32 id) :
- _comInit(ScopedCOMInitializer::kMTA),
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _volumeMutex(*CriticalSectionWrapper::CreateCriticalSection()),
- _id(id),
- _ptrAudioBuffer(NULL),
- _ptrEnumerator(NULL),
- _ptrRenderCollection(NULL),
- _ptrCaptureCollection(NULL),
- _ptrDeviceOut(NULL),
- _ptrDeviceIn(NULL),
- _ptrClientOut(NULL),
- _ptrClientIn(NULL),
- _ptrRenderClient(NULL),
- _ptrCaptureClient(NULL),
- _ptrCaptureVolume(NULL),
- _ptrRenderSimpleVolume(NULL),
- _dmo(NULL),
- _mediaBuffer(NULL),
- _builtInAecEnabled(false),
- _playAudioFrameSize(0),
- _playSampleRate(0),
- _playBlockSize(0),
- _playChannels(2),
- _sndCardPlayDelay(0),
- _sndCardRecDelay(0),
- _sampleDriftAt48kHz(0),
- _driftAccumulator(0),
- _writtenSamples(0),
- _readSamples(0),
- _playAcc(0),
- _recAudioFrameSize(0),
- _recSampleRate(0),
- _recBlockSize(0),
- _recChannels(2),
- _avrtLibrary(NULL),
- _winSupportAvrt(false),
- _hRenderSamplesReadyEvent(NULL),
- _hPlayThread(NULL),
- _hCaptureSamplesReadyEvent(NULL),
- _hRecThread(NULL),
- _hShutdownRenderEvent(NULL),
- _hShutdownCaptureEvent(NULL),
- _hRenderStartedEvent(NULL),
- _hCaptureStartedEvent(NULL),
- _hGetCaptureVolumeThread(NULL),
- _hSetCaptureVolumeThread(NULL),
- _hSetCaptureVolumeEvent(NULL),
- _hMmTask(NULL),
- _initialized(false),
- _recording(false),
- _playing(false),
- _recIsInitialized(false),
- _playIsInitialized(false),
- _speakerIsInitialized(false),
- _microphoneIsInitialized(false),
- _AGC(false),
- _playWarning(0),
- _playError(0),
- _recWarning(0),
- _recError(0),
- _playBufType(AudioDeviceModule::kAdaptiveBufferSize),
- _playBufDelay(80),
- _playBufDelayFixed(80),
- _usingInputDeviceIndex(false),
- _usingOutputDeviceIndex(false),
- _inputDevice(AudioDeviceModule::kDefaultCommunicationDevice),
- _outputDevice(AudioDeviceModule::kDefaultCommunicationDevice),
- _inputDeviceIndex(0),
- _outputDeviceIndex(0),
- _newMicLevel(0)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
- assert(_comInit.succeeded());
-
- // Try to load the Avrt DLL
- if (!_avrtLibrary)
- {
- // Get handle to the Avrt DLL module.
- _avrtLibrary = LoadLibrary(TEXT("Avrt.dll"));
- if (_avrtLibrary)
- {
- // Handle is valid (should only happen if OS larger than vista & win7).
- // Try to get the function addresses.
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::AudioDeviceWindowsCore() The Avrt DLL module is now loaded");
-
- _PAvRevertMmThreadCharacteristics = (PAvRevertMmThreadCharacteristics)GetProcAddress(_avrtLibrary, "AvRevertMmThreadCharacteristics");
- _PAvSetMmThreadCharacteristicsA = (PAvSetMmThreadCharacteristicsA)GetProcAddress(_avrtLibrary, "AvSetMmThreadCharacteristicsA");
- _PAvSetMmThreadPriority = (PAvSetMmThreadPriority)GetProcAddress(_avrtLibrary, "AvSetMmThreadPriority");
-
- if ( _PAvRevertMmThreadCharacteristics &&
- _PAvSetMmThreadCharacteristicsA &&
- _PAvSetMmThreadPriority)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::AudioDeviceWindowsCore() AvRevertMmThreadCharacteristics() is OK");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::AudioDeviceWindowsCore() AvSetMmThreadCharacteristicsA() is OK");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::AudioDeviceWindowsCore() AvSetMmThreadPriority() is OK");
- _winSupportAvrt = true;
- }
- }
- }
-
- // Create our samples ready events - we want auto reset events that start in the not-signaled state.
- // The state of an auto-reset event object remains signaled until a single waiting thread is released,
- // at which time the system automatically sets the state to nonsignaled. If no threads are waiting,
- // the event object's state remains signaled.
- // (Except for _hShutdownCaptureEvent, which is used to shutdown multiple threads).
- _hRenderSamplesReadyEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
- _hCaptureSamplesReadyEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
- _hShutdownRenderEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
- _hShutdownCaptureEvent = CreateEvent(NULL, TRUE, FALSE, NULL);
- _hRenderStartedEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
- _hCaptureStartedEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
- _hSetCaptureVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
-
- _perfCounterFreq.QuadPart = 1;
- _perfCounterFactor = 0.0;
- _avgCPULoad = 0.0;
-
- // list of number of channels to use on recording side
- _recChannelsPrioList[0] = 2; // stereo is prio 1
- _recChannelsPrioList[1] = 1; // mono is prio 2
-
- // list of number of channels to use on playout side
- _playChannelsPrioList[0] = 2; // stereo is prio 1
- _playChannelsPrioList[1] = 1; // mono is prio 2
-
- HRESULT hr;
-
- // We know that this API will work since it has already been verified in
- // CoreAudioIsSupported, hence no need to check for errors here as well.
-
- // Retrive the IMMDeviceEnumerator API (should load the MMDevAPI.dll)
- // TODO(henrika): we should probably move this allocation to Init() instead
- // and deallocate in Terminate() to make the implementation more symmetric.
- CoCreateInstance(
- __uuidof(MMDeviceEnumerator),
- NULL,
- CLSCTX_ALL,
- __uuidof(IMMDeviceEnumerator),
- reinterpret_cast<void**>(&_ptrEnumerator));
- assert(NULL != _ptrEnumerator);
-
- // DMO initialization for built-in WASAPI AEC.
- {
- IMediaObject* ptrDMO = NULL;
- hr = CoCreateInstance(CLSID_CWMAudioAEC,
- NULL,
- CLSCTX_INPROC_SERVER,
- IID_IMediaObject,
- reinterpret_cast<void**>(&ptrDMO));
- if (FAILED(hr) || ptrDMO == NULL)
- {
- // Since we check that _dmo is non-NULL in EnableBuiltInAEC(), the
- // feature is prevented from being enabled.
- _builtInAecEnabled = false;
- _TraceCOMError(hr);
- }
- _dmo = ptrDMO;
- SAFE_RELEASE(ptrDMO);
- }
-}
-
-// ----------------------------------------------------------------------------
-// AudioDeviceWindowsCore() - dtor
-// ----------------------------------------------------------------------------
-
-AudioDeviceWindowsCore::~AudioDeviceWindowsCore()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
-
- Terminate();
-
- // The IMMDeviceEnumerator is created during construction. Must release
- // it here and not in Terminate() since we don't recreate it in Init().
- SAFE_RELEASE(_ptrEnumerator);
-
- _ptrAudioBuffer = NULL;
-
- if (NULL != _hRenderSamplesReadyEvent)
- {
- CloseHandle(_hRenderSamplesReadyEvent);
- _hRenderSamplesReadyEvent = NULL;
- }
-
- if (NULL != _hCaptureSamplesReadyEvent)
- {
- CloseHandle(_hCaptureSamplesReadyEvent);
- _hCaptureSamplesReadyEvent = NULL;
- }
-
- if (NULL != _hRenderStartedEvent)
- {
- CloseHandle(_hRenderStartedEvent);
- _hRenderStartedEvent = NULL;
- }
-
- if (NULL != _hCaptureStartedEvent)
- {
- CloseHandle(_hCaptureStartedEvent);
- _hCaptureStartedEvent = NULL;
- }
-
- if (NULL != _hShutdownRenderEvent)
- {
- CloseHandle(_hShutdownRenderEvent);
- _hShutdownRenderEvent = NULL;
- }
-
- if (NULL != _hShutdownCaptureEvent)
- {
- CloseHandle(_hShutdownCaptureEvent);
- _hShutdownCaptureEvent = NULL;
- }
-
- if (NULL != _hSetCaptureVolumeEvent)
- {
- CloseHandle(_hSetCaptureVolumeEvent);
- _hSetCaptureVolumeEvent = NULL;
- }
-
- if (_avrtLibrary)
- {
- BOOL freeOK = FreeLibrary(_avrtLibrary);
- if (!freeOK)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "AudioDeviceWindowsCore::~AudioDeviceWindowsCore() failed to free the loaded Avrt DLL module correctly");
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "AudioDeviceWindowsCore::~AudioDeviceWindowsCore() the Avrt DLL module is now unloaded");
- }
- }
-
- delete &_critSect;
- delete &_volumeMutex;
-}
-
-// ============================================================================
-// API
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// AttachAudioBuffer
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsCore::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
-{
-
- _ptrAudioBuffer = audioBuffer;
-
- // Inform the AudioBuffer about default settings for this implementation.
- // Set all values to zero here since the actual settings will be done by
- // InitPlayout and InitRecording later.
- _ptrAudioBuffer->SetRecordingSampleRate(0);
- _ptrAudioBuffer->SetPlayoutSampleRate(0);
- _ptrAudioBuffer->SetRecordingChannels(0);
- _ptrAudioBuffer->SetPlayoutChannels(0);
-}
-
-// ----------------------------------------------------------------------------
-// ActiveAudioLayer
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const
-{
- audioLayer = AudioDeviceModule::kWindowsCoreAudio;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Init
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::Init()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_initialized)
- {
- return 0;
- }
-
- _playWarning = 0;
- _playError = 0;
- _recWarning = 0;
- _recError = 0;
-
- // Enumerate all audio rendering and capturing endpoint devices.
- // Note that, some of these will not be able to select by the user.
- // The complete collection is for internal use only.
- //
- _EnumerateEndpointDevicesAll(eRender);
- _EnumerateEndpointDevicesAll(eCapture);
-
- _initialized = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Terminate
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::Terminate()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_initialized) {
- return 0;
- }
-
- _initialized = false;
- _speakerIsInitialized = false;
- _microphoneIsInitialized = false;
- _playing = false;
- _recording = false;
-
- SAFE_RELEASE(_ptrRenderCollection);
- SAFE_RELEASE(_ptrCaptureCollection);
- SAFE_RELEASE(_ptrDeviceOut);
- SAFE_RELEASE(_ptrDeviceIn);
- SAFE_RELEASE(_ptrClientOut);
- SAFE_RELEASE(_ptrClientIn);
- SAFE_RELEASE(_ptrRenderClient);
- SAFE_RELEASE(_ptrCaptureClient);
- SAFE_RELEASE(_ptrCaptureVolume);
- SAFE_RELEASE(_ptrRenderSimpleVolume);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Initialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsCore::Initialized() const
-{
- return (_initialized);
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SpeakerIsAvailable(bool& available)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
-
- available = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitSpeaker
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::InitSpeaker()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_playing)
- {
- return -1;
- }
-
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
-
- if (_usingOutputDeviceIndex)
- {
- WebRtc_Word16 nDevices = PlayoutDevices();
- if (_outputDeviceIndex > (nDevices - 1))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "current device selection is invalid => unable to initialize");
- return -1;
- }
- }
-
- WebRtc_Word32 ret(0);
-
- SAFE_RELEASE(_ptrDeviceOut);
- if (_usingOutputDeviceIndex)
- {
- // Refresh the selected rendering endpoint device using current index
- ret = _GetListDevice(eRender, _outputDeviceIndex, &_ptrDeviceOut);
- }
- else
- {
- ERole role;
- (_outputDevice == AudioDeviceModule::kDefaultDevice) ? role = eConsole : role = eCommunications;
- // Refresh the selected rendering endpoint device using role
- ret = _GetDefaultDevice(eRender, role, &_ptrDeviceOut);
- }
-
- if (ret != 0 || (_ptrDeviceOut == NULL))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to initialize the rendering enpoint device");
- SAFE_RELEASE(_ptrDeviceOut);
- return -1;
- }
-
- IAudioSessionManager* pManager = NULL;
- ret = _ptrDeviceOut->Activate(__uuidof(IAudioSessionManager),
- CLSCTX_ALL,
- NULL,
- (void**)&pManager);
- if (ret != 0 || pManager == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to initialize the render manager");
- SAFE_RELEASE(pManager);
- return -1;
- }
-
- SAFE_RELEASE(_ptrRenderSimpleVolume);
- ret = pManager->GetSimpleAudioVolume(NULL, FALSE, &_ptrRenderSimpleVolume);
- if (ret != 0 || _ptrRenderSimpleVolume == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to initialize the render simple volume");
- SAFE_RELEASE(pManager);
- SAFE_RELEASE(_ptrRenderSimpleVolume);
- return -1;
- }
- SAFE_RELEASE(pManager);
-
- _speakerIsInitialized = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::MicrophoneIsAvailable(bool& available)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_ptrDeviceIn == NULL)
- {
- return -1;
- }
-
- available = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitMicrophone
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::InitMicrophone()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_recording)
- {
- return -1;
- }
-
- if (_ptrDeviceIn == NULL)
- {
- return -1;
- }
-
- if (_usingInputDeviceIndex)
- {
- WebRtc_Word16 nDevices = RecordingDevices();
- if (_inputDeviceIndex > (nDevices - 1))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "current device selection is invalid => unable to initialize");
- return -1;
- }
- }
-
- WebRtc_Word32 ret(0);
-
- SAFE_RELEASE(_ptrDeviceIn);
- if (_usingInputDeviceIndex)
- {
- // Refresh the selected capture endpoint device using current index
- ret = _GetListDevice(eCapture, _inputDeviceIndex, &_ptrDeviceIn);
- }
- else
- {
- ERole role;
- (_inputDevice == AudioDeviceModule::kDefaultDevice) ? role = eConsole : role = eCommunications;
- // Refresh the selected capture endpoint device using role
- ret = _GetDefaultDevice(eCapture, role, &_ptrDeviceIn);
- }
-
- if (ret != 0 || (_ptrDeviceIn == NULL))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to initialize the capturing enpoint device");
- SAFE_RELEASE(_ptrDeviceIn);
- return -1;
- }
-
- ret = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume),
- CLSCTX_ALL,
- NULL,
- reinterpret_cast<void **>(&_ptrCaptureVolume));
- if (ret != 0 || _ptrCaptureVolume == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to initialize the capture volume");
- SAFE_RELEASE(_ptrCaptureVolume);
- return -1;
- }
-
- _microphoneIsInitialized = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsCore::SpeakerIsInitialized() const
-{
-
- return (_speakerIsInitialized);
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsCore::MicrophoneIsInitialized() const
-{
-
- return (_microphoneIsInitialized);
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolumeIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SpeakerVolumeIsAvailable(bool& available)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
-
- HRESULT hr = S_OK;
- IAudioSessionManager* pManager = NULL;
- ISimpleAudioVolume* pVolume = NULL;
-
- hr = _ptrDeviceOut->Activate(__uuidof(IAudioSessionManager), CLSCTX_ALL, NULL, (void**)&pManager);
- EXIT_ON_ERROR(hr);
-
- hr = pManager->GetSimpleAudioVolume(NULL, FALSE, &pVolume);
- EXIT_ON_ERROR(hr);
-
- float volume(0.0f);
- hr = pVolume->GetMasterVolume(&volume);
- if (FAILED(hr))
- {
- available = false;
- }
- available = true;
-
- SAFE_RELEASE(pManager);
- SAFE_RELEASE(pVolume);
-
- return 0;
-
-Exit:
- _TraceCOMError(hr);
- SAFE_RELEASE(pManager);
- SAFE_RELEASE(pVolume);
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// SetSpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SetSpeakerVolume(WebRtc_UWord32 volume)
-{
-
- {
- CriticalSectionScoped lock(_critSect);
-
- if (!_speakerIsInitialized)
- {
- return -1;
- }
-
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
- }
-
- if (volume < (WebRtc_UWord32)MIN_CORE_SPEAKER_VOLUME ||
- volume > (WebRtc_UWord32)MAX_CORE_SPEAKER_VOLUME)
- {
- return -1;
- }
-
- HRESULT hr = S_OK;
-
- // scale input volume to valid range (0.0 to 1.0)
- const float fLevel = (float)volume/MAX_CORE_SPEAKER_VOLUME;
- _volumeMutex.Enter();
- hr = _ptrRenderSimpleVolume->SetMasterVolume(fLevel,NULL);
- _volumeMutex.Leave();
- EXIT_ON_ERROR(hr);
-
- return 0;
-
-Exit:
- _TraceCOMError(hr);
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SpeakerVolume(WebRtc_UWord32& volume) const
-{
-
- {
- CriticalSectionScoped lock(_critSect);
-
- if (!_speakerIsInitialized)
- {
- return -1;
- }
-
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
- }
-
- HRESULT hr = S_OK;
- float fLevel(0.0f);
-
- _volumeMutex.Enter();
- hr = _ptrRenderSimpleVolume->GetMasterVolume(&fLevel);
- _volumeMutex.Leave();
- EXIT_ON_ERROR(hr);
-
- // scale input volume range [0.0,1.0] to valid output range
- volume = static_cast<WebRtc_UWord32> (fLevel*MAX_CORE_SPEAKER_VOLUME);
-
- return 0;
-
-Exit:
- _TraceCOMError(hr);
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// SetWaveOutVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SetWaveOutVolume(WebRtc_UWord16 volumeLeft, WebRtc_UWord16 volumeRight)
-{
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// WaveOutVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::WaveOutVolume(WebRtc_UWord16& volumeLeft, WebRtc_UWord16& volumeRight) const
-{
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MaxSpeakerVolume
-//
-// The internal range for Core Audio is 0.0 to 1.0, where 0.0 indicates
-// silence and 1.0 indicates full volume (no attenuation).
-// We add our (webrtc-internal) own max level to match the Wave API and
-// how it is used today in VoE.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const
-{
-
- if (!_speakerIsInitialized)
- {
- return -1;
- }
-
- maxVolume = static_cast<WebRtc_UWord32> (MAX_CORE_SPEAKER_VOLUME);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MinSpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::MinSpeakerVolume(WebRtc_UWord32& minVolume) const
-{
-
- if (!_speakerIsInitialized)
- {
- return -1;
- }
-
- minVolume = static_cast<WebRtc_UWord32> (MIN_CORE_SPEAKER_VOLUME);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolumeStepSize
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const
-{
-
- if (!_speakerIsInitialized)
- {
- return -1;
- }
-
- stepSize = CORE_SPEAKER_VOLUME_STEP_SIZE;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerMuteIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SpeakerMuteIsAvailable(bool& available)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
-
- HRESULT hr = S_OK;
- IAudioEndpointVolume* pVolume = NULL;
-
- // Query the speaker system mute state.
- hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume),
- CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
- EXIT_ON_ERROR(hr);
-
- BOOL mute;
- hr = pVolume->GetMute(&mute);
- if (FAILED(hr))
- available = false;
- else
- available = true;
-
- SAFE_RELEASE(pVolume);
-
- return 0;
-
-Exit:
- _TraceCOMError(hr);
- SAFE_RELEASE(pVolume);
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// SetSpeakerMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SetSpeakerMute(bool enable)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_speakerIsInitialized)
- {
- return -1;
- }
-
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
-
- HRESULT hr = S_OK;
- IAudioEndpointVolume* pVolume = NULL;
-
- // Set the speaker system mute state.
- hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
- EXIT_ON_ERROR(hr);
-
- const BOOL mute(enable);
- hr = pVolume->SetMute(mute, NULL);
- EXIT_ON_ERROR(hr);
-
- SAFE_RELEASE(pVolume);
-
- return 0;
-
-Exit:
- _TraceCOMError(hr);
- SAFE_RELEASE(pVolume);
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SpeakerMute(bool& enabled) const
-{
-
- if (!_speakerIsInitialized)
- {
- return -1;
- }
-
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
-
- HRESULT hr = S_OK;
- IAudioEndpointVolume* pVolume = NULL;
-
- // Query the speaker system mute state.
- hr = _ptrDeviceOut->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
- EXIT_ON_ERROR(hr);
-
- BOOL mute;
- hr = pVolume->GetMute(&mute);
- EXIT_ON_ERROR(hr);
-
- enabled = (mute == TRUE) ? true : false;
-
- SAFE_RELEASE(pVolume);
-
- return 0;
-
-Exit:
- _TraceCOMError(hr);
- SAFE_RELEASE(pVolume);
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneMuteIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::MicrophoneMuteIsAvailable(bool& available)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_ptrDeviceIn == NULL)
- {
- return -1;
- }
-
- HRESULT hr = S_OK;
- IAudioEndpointVolume* pVolume = NULL;
-
- // Query the microphone system mute state.
- hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
- EXIT_ON_ERROR(hr);
-
- BOOL mute;
- hr = pVolume->GetMute(&mute);
- if (FAILED(hr))
- available = false;
- else
- available = true;
-
- SAFE_RELEASE(pVolume);
- return 0;
-
-Exit:
- _TraceCOMError(hr);
- SAFE_RELEASE(pVolume);
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SetMicrophoneMute(bool enable)
-{
-
- if (!_microphoneIsInitialized)
- {
- return -1;
- }
-
- if (_ptrDeviceIn == NULL)
- {
- return -1;
- }
-
- HRESULT hr = S_OK;
- IAudioEndpointVolume* pVolume = NULL;
-
- // Set the microphone system mute state.
- hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
- EXIT_ON_ERROR(hr);
-
- const BOOL mute(enable);
- hr = pVolume->SetMute(mute, NULL);
- EXIT_ON_ERROR(hr);
-
- SAFE_RELEASE(pVolume);
- return 0;
-
-Exit:
- _TraceCOMError(hr);
- SAFE_RELEASE(pVolume);
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::MicrophoneMute(bool& enabled) const
-{
-
- if (!_microphoneIsInitialized)
- {
- return -1;
- }
-
- HRESULT hr = S_OK;
- IAudioEndpointVolume* pVolume = NULL;
-
- // Query the microphone system mute state.
- hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
- EXIT_ON_ERROR(hr);
-
- BOOL mute;
- hr = pVolume->GetMute(&mute);
- EXIT_ON_ERROR(hr);
-
- enabled = (mute == TRUE) ? true : false;
-
- SAFE_RELEASE(pVolume);
- return 0;
-
-Exit:
- _TraceCOMError(hr);
- SAFE_RELEASE(pVolume);
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneBoostIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::MicrophoneBoostIsAvailable(bool& available)
-{
-
- available = false;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneBoost
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SetMicrophoneBoost(bool enable)
-{
-
- if (!_microphoneIsInitialized)
- {
- return -1;
- }
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneBoost
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::MicrophoneBoost(bool& enabled) const
-{
-
- if (!_microphoneIsInitialized)
- {
- return -1;
- }
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// StereoRecordingIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::StereoRecordingIsAvailable(bool& available)
-{
-
- available = true;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetStereoRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SetStereoRecording(bool enable)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (enable)
- {
- _recChannelsPrioList[0] = 2; // try stereo first
- _recChannelsPrioList[1] = 1;
- _recChannels = 2;
- }
- else
- {
- _recChannelsPrioList[0] = 1; // try mono first
- _recChannelsPrioList[1] = 2;
- _recChannels = 1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::StereoRecording(bool& enabled) const
-{
-
- if (_recChannels == 2)
- enabled = true;
- else
- enabled = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoPlayoutIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::StereoPlayoutIsAvailable(bool& available)
-{
-
- available = true;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetStereoPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SetStereoPlayout(bool enable)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (enable)
- {
- _playChannelsPrioList[0] = 2; // try stereo first
- _playChannelsPrioList[1] = 1;
- _playChannels = 2;
- }
- else
- {
- _playChannelsPrioList[0] = 1; // try mono first
- _playChannelsPrioList[1] = 2;
- _playChannels = 1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::StereoPlayout(bool& enabled) const
-{
-
- if (_playChannels == 2)
- enabled = true;
- else
- enabled = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetAGC
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SetAGC(bool enable)
-{
- CriticalSectionScoped lock(_critSect);
- _AGC = enable;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// AGC
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsCore::AGC() const
-{
- CriticalSectionScoped lock(_critSect);
- return _AGC;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolumeIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::MicrophoneVolumeIsAvailable(bool& available)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_ptrDeviceIn == NULL)
- {
- return -1;
- }
-
- HRESULT hr = S_OK;
- IAudioEndpointVolume* pVolume = NULL;
-
- hr = _ptrDeviceIn->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL, NULL, reinterpret_cast<void**>(&pVolume));
- EXIT_ON_ERROR(hr);
-
- float volume(0.0f);
- hr = pVolume->GetMasterVolumeLevelScalar(&volume);
- if (FAILED(hr))
- {
- available = false;
- }
- available = true;
-
- SAFE_RELEASE(pVolume);
- return 0;
-
-Exit:
- _TraceCOMError(hr);
- SAFE_RELEASE(pVolume);
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SetMicrophoneVolume(WebRtc_UWord32 volume)
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "AudioDeviceWindowsCore::SetMicrophoneVolume(volume=%u)", volume);
-
- {
- CriticalSectionScoped lock(_critSect);
-
- if (!_microphoneIsInitialized)
- {
- return -1;
- }
-
- if (_ptrDeviceIn == NULL)
- {
- return -1;
- }
- }
-
- if (volume < static_cast<WebRtc_UWord32>(MIN_CORE_MICROPHONE_VOLUME) ||
- volume > static_cast<WebRtc_UWord32>(MAX_CORE_MICROPHONE_VOLUME))
- {
- return -1;
- }
-
- HRESULT hr = S_OK;
- // scale input volume to valid range (0.0 to 1.0)
- const float fLevel = static_cast<float>(volume)/MAX_CORE_MICROPHONE_VOLUME;
- _volumeMutex.Enter();
- _ptrCaptureVolume->SetMasterVolumeLevelScalar(fLevel, NULL);
- _volumeMutex.Leave();
- EXIT_ON_ERROR(hr);
-
- return 0;
-
-Exit:
- _TraceCOMError(hr);
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::MicrophoneVolume(WebRtc_UWord32& volume) const
-{
- {
- CriticalSectionScoped lock(_critSect);
-
- if (!_microphoneIsInitialized)
- {
- return -1;
- }
-
- if (_ptrDeviceIn == NULL)
- {
- return -1;
- }
- }
-
- HRESULT hr = S_OK;
- float fLevel(0.0f);
- volume = 0;
- _volumeMutex.Enter();
- hr = _ptrCaptureVolume->GetMasterVolumeLevelScalar(&fLevel);
- _volumeMutex.Leave();
- EXIT_ON_ERROR(hr);
-
- // scale input volume range [0.0,1.0] to valid output range
- volume = static_cast<WebRtc_UWord32> (fLevel*MAX_CORE_MICROPHONE_VOLUME);
-
- return 0;
-
-Exit:
- _TraceCOMError(hr);
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// MaxMicrophoneVolume
-//
-// The internal range for Core Audio is 0.0 to 1.0, where 0.0 indicates
-// silence and 1.0 indicates full volume (no attenuation).
-// We add our (webrtc-internal) own max level to match the Wave API and
-// how it is used today in VoE.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- if (!_microphoneIsInitialized)
- {
- return -1;
- }
-
- maxVolume = static_cast<WebRtc_UWord32> (MAX_CORE_MICROPHONE_VOLUME);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MinMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::MinMicrophoneVolume(WebRtc_UWord32& minVolume) const
-{
-
- if (!_microphoneIsInitialized)
- {
- return -1;
- }
-
- minVolume = static_cast<WebRtc_UWord32> (MIN_CORE_MICROPHONE_VOLUME);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolumeStepSize
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const
-{
-
- if (!_microphoneIsInitialized)
- {
- return -1;
- }
-
- stepSize = CORE_MICROPHONE_VOLUME_STEP_SIZE;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDevices
-// ----------------------------------------------------------------------------
-
-WebRtc_Word16 AudioDeviceWindowsCore::PlayoutDevices()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_RefreshDeviceList(eRender) != -1)
- {
- return (_DeviceListCount(eRender));
- }
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutDevice I (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SetPlayoutDevice(WebRtc_UWord16 index)
-{
-
- if (_playIsInitialized)
- {
- return -1;
- }
-
- // Get current number of available rendering endpoint devices and refresh the rendering collection.
- UINT nDevices = PlayoutDevices();
-
- if (index < 0 || index > (nDevices-1))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device index is out of range [0,%u]", (nDevices-1));
- return -1;
- }
-
- CriticalSectionScoped lock(_critSect);
-
- HRESULT hr(S_OK);
-
- assert(_ptrRenderCollection != NULL);
-
- // Select an endpoint rendering device given the specified index
- SAFE_RELEASE(_ptrDeviceOut);
- hr = _ptrRenderCollection->Item(
- index,
- &_ptrDeviceOut);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(_ptrDeviceOut);
- return -1;
- }
-
- WCHAR szDeviceName[MAX_PATH];
- const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
-
- // Get the endpoint device's friendly-name
- if (_GetDeviceName(_ptrDeviceOut, szDeviceName, bufferLen) == 0)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", szDeviceName);
- }
-
- _usingOutputDeviceIndex = true;
- _outputDeviceIndex = index;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutDevice II (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device)
-{
- if (_playIsInitialized)
- {
- return -1;
- }
-
- ERole role(eCommunications);
-
- if (device == AudioDeviceModule::kDefaultDevice)
- {
- role = eConsole;
- }
- else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
- {
- role = eCommunications;
- }
-
- CriticalSectionScoped lock(_critSect);
-
- // Refresh the list of rendering endpoint devices
- _RefreshDeviceList(eRender);
-
- HRESULT hr(S_OK);
-
- assert(_ptrEnumerator != NULL);
-
- // Select an endpoint rendering device given the specified role
- SAFE_RELEASE(_ptrDeviceOut);
- hr = _ptrEnumerator->GetDefaultAudioEndpoint(
- eRender,
- role,
- &_ptrDeviceOut);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(_ptrDeviceOut);
- return -1;
- }
-
- WCHAR szDeviceName[MAX_PATH];
- const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
-
- // Get the endpoint device's friendly-name
- if (_GetDeviceName(_ptrDeviceOut, szDeviceName, bufferLen) == 0)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", szDeviceName);
- }
-
- _usingOutputDeviceIndex = false;
- _outputDevice = device;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDeviceName
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::PlayoutDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
-{
-
- bool defaultCommunicationDevice(false);
- const WebRtc_Word16 nDevices(PlayoutDevices()); // also updates the list of devices
-
- // Special fix for the case when the user selects '-1' as index (<=> Default Communication Device)
- if (index == (WebRtc_UWord16)(-1))
- {
- defaultCommunicationDevice = true;
- index = 0;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Default Communication endpoint device will be used");
- }
-
- if ((index > (nDevices-1)) || (name == NULL))
- {
- return -1;
- }
-
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- CriticalSectionScoped lock(_critSect);
-
- HRESULT hr(S_OK);
- WebRtc_Word32 ret(-1);
- WCHAR szDeviceName[MAX_PATH];
- const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
-
- // Get the endpoint device's friendly-name
- if (defaultCommunicationDevice)
- {
- ret = _GetDefaultDeviceName(eRender, eCommunications, szDeviceName, bufferLen);
- }
- else
- {
- ret = _GetListDeviceName(eRender, index, szDeviceName, bufferLen);
- }
-
- if (ret == 0)
- {
- // Convert the endpoint device's friendly-name to UTF-8
- if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, name, kAdmMaxDeviceNameSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d", GetLastError());
- }
- }
-
- // Get the endpoint ID string (uniquely identifies the device among all audio endpoint devices)
- if (defaultCommunicationDevice)
- {
- ret = _GetDefaultDeviceID(eRender, eCommunications, szDeviceName, bufferLen);
- }
- else
- {
- ret = _GetListDeviceID(eRender, index, szDeviceName, bufferLen);
- }
-
- if (guid != NULL && ret == 0)
- {
- // Convert the endpoint device's ID string to UTF-8
- if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d", GetLastError());
- }
- }
-
- return ret;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDeviceName
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::RecordingDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
-{
-
- bool defaultCommunicationDevice(false);
- const WebRtc_Word16 nDevices(RecordingDevices()); // also updates the list of devices
-
- // Special fix for the case when the user selects '-1' as index (<=> Default Communication Device)
- if (index == (WebRtc_UWord16)(-1))
- {
- defaultCommunicationDevice = true;
- index = 0;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Default Communication endpoint device will be used");
- }
-
- if ((index > (nDevices-1)) || (name == NULL))
- {
- return -1;
- }
-
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- CriticalSectionScoped lock(_critSect);
-
- HRESULT hr(S_OK);
- WebRtc_Word32 ret(-1);
- WCHAR szDeviceName[MAX_PATH];
- const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
-
- // Get the endpoint device's friendly-name
- if (defaultCommunicationDevice)
- {
- ret = _GetDefaultDeviceName(eCapture, eCommunications, szDeviceName, bufferLen);
- }
- else
- {
- ret = _GetListDeviceName(eCapture, index, szDeviceName, bufferLen);
- }
-
- if (ret == 0)
- {
- // Convert the endpoint device's friendly-name to UTF-8
- if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, name, kAdmMaxDeviceNameSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d", GetLastError());
- }
- }
-
- // Get the endpoint ID string (uniquely identifies the device among all audio endpoint devices)
- if (defaultCommunicationDevice)
- {
- ret = _GetDefaultDeviceID(eCapture, eCommunications, szDeviceName, bufferLen);
- }
- else
- {
- ret = _GetListDeviceID(eCapture, index, szDeviceName, bufferLen);
- }
-
- if (guid != NULL && ret == 0)
- {
- // Convert the endpoint device's ID string to UTF-8
- if (WideCharToMultiByte(CP_UTF8, 0, szDeviceName, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d", GetLastError());
- }
- }
-
- return ret;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDevices
-// ----------------------------------------------------------------------------
-
-WebRtc_Word16 AudioDeviceWindowsCore::RecordingDevices()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_RefreshDeviceList(eCapture) != -1)
- {
- return (_DeviceListCount(eCapture));
- }
-
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingDevice I (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SetRecordingDevice(WebRtc_UWord16 index)
-{
-
- if (_recIsInitialized)
- {
- return -1;
- }
-
- // Get current number of available capture endpoint devices and refresh the capture collection.
- UINT nDevices = RecordingDevices();
-
- if (index < 0 || index > (nDevices-1))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device index is out of range [0,%u]", (nDevices-1));
- return -1;
- }
-
- CriticalSectionScoped lock(_critSect);
-
- HRESULT hr(S_OK);
-
- assert(_ptrCaptureCollection != NULL);
-
- // Select an endpoint capture device given the specified index
- SAFE_RELEASE(_ptrDeviceIn);
- hr = _ptrCaptureCollection->Item(
- index,
- &_ptrDeviceIn);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(_ptrDeviceIn);
- return -1;
- }
-
- WCHAR szDeviceName[MAX_PATH];
- const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
-
- // Get the endpoint device's friendly-name
- if (_GetDeviceName(_ptrDeviceIn, szDeviceName, bufferLen) == 0)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", szDeviceName);
- }
-
- _usingInputDeviceIndex = true;
- _inputDeviceIndex = index;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingDevice II (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device)
-{
- if (_recIsInitialized)
- {
- return -1;
- }
-
- ERole role(eCommunications);
-
- if (device == AudioDeviceModule::kDefaultDevice)
- {
- role = eConsole;
- }
- else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
- {
- role = eCommunications;
- }
-
- CriticalSectionScoped lock(_critSect);
-
- // Refresh the list of capture endpoint devices
- _RefreshDeviceList(eCapture);
-
- HRESULT hr(S_OK);
-
- assert(_ptrEnumerator != NULL);
-
- // Select an endpoint capture device given the specified role
- SAFE_RELEASE(_ptrDeviceIn);
- hr = _ptrEnumerator->GetDefaultAudioEndpoint(
- eCapture,
- role,
- &_ptrDeviceIn);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(_ptrDeviceIn);
- return -1;
- }
-
- WCHAR szDeviceName[MAX_PATH];
- const int bufferLen = sizeof(szDeviceName)/sizeof(szDeviceName)[0];
-
- // Get the endpoint device's friendly-name
- if (_GetDeviceName(_ptrDeviceIn, szDeviceName, bufferLen) == 0)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", szDeviceName);
- }
-
- _usingInputDeviceIndex = false;
- _inputDevice = device;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::PlayoutIsAvailable(bool& available)
-{
-
- available = false;
-
- // Try to initialize the playout side
- WebRtc_Word32 res = InitPlayout();
-
- // Cancel effect of initialization
- StopPlayout();
-
- if (res != -1)
- {
- available = true;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::RecordingIsAvailable(bool& available)
-{
-
- available = false;
-
- // Try to initialize the recording side
- WebRtc_Word32 res = InitRecording();
-
- // Cancel effect of initialization
- StopRecording();
-
- if (res != -1)
- {
- available = true;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::InitPlayout()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_playing)
- {
- return -1;
- }
-
- if (_playIsInitialized)
- {
- return 0;
- }
-
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
-
- // Initialize the speaker (devices might have been added or removed)
- if (InitSpeaker() == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "InitSpeaker() failed");
- }
-
- // Ensure that the updated rendering endpoint device is valid
- if (_ptrDeviceOut == NULL)
- {
- return -1;
- }
-
- if (_builtInAecEnabled && _recIsInitialized)
- {
- // Ensure the correct render device is configured in case
- // InitRecording() was called before InitPlayout().
- if (SetDMOProperties() == -1)
- {
- return -1;
- }
- }
-
- HRESULT hr = S_OK;
- WAVEFORMATEX* pWfxOut = NULL;
- WAVEFORMATEX Wfx;
- WAVEFORMATEX* pWfxClosestMatch = NULL;
-
- // Create COM object with IAudioClient interface.
- SAFE_RELEASE(_ptrClientOut);
- hr = _ptrDeviceOut->Activate(
- __uuidof(IAudioClient),
- CLSCTX_ALL,
- NULL,
- (void**)&_ptrClientOut);
- EXIT_ON_ERROR(hr);
-
- // Retrieve the stream format that the audio engine uses for its internal
- // processing (mixing) of shared-mode streams.
- hr = _ptrClientOut->GetMixFormat(&pWfxOut);
- if (SUCCEEDED(hr))
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Audio Engine's current rendering mix format:");
- // format type
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wFormatTag : 0x%X (%u)", pWfxOut->wFormatTag, pWfxOut->wFormatTag);
- // number of channels (i.e. mono, stereo...)
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels : %d", pWfxOut->nChannels);
- // sample rate
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nSamplesPerSec : %d", pWfxOut->nSamplesPerSec);
- // for buffer estimation
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nAvgBytesPerSec: %d", pWfxOut->nAvgBytesPerSec);
- // block size of data
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nBlockAlign : %d", pWfxOut->nBlockAlign);
- // number of bits per sample of mono data
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wBitsPerSample : %d", pWfxOut->wBitsPerSample);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "cbSize : %d", pWfxOut->cbSize);
- }
-
- // Set wave format
- Wfx.wFormatTag = WAVE_FORMAT_PCM;
- Wfx.wBitsPerSample = 16;
- Wfx.cbSize = 0;
-
- const int freqs[] = {48000, 44100, 16000, 96000, 32000, 8000};
- hr = S_FALSE;
-
- // Iterate over frequencies and channels, in order of priority
- for (int freq = 0; freq < sizeof(freqs)/sizeof(freqs[0]); freq++)
- {
- for (int chan = 0; chan < sizeof(_playChannelsPrioList)/sizeof(_playChannelsPrioList[0]); chan++)
- {
- Wfx.nChannels = _playChannelsPrioList[chan];
- Wfx.nSamplesPerSec = freqs[freq];
- Wfx.nBlockAlign = Wfx.nChannels * Wfx.wBitsPerSample / 8;
- Wfx.nAvgBytesPerSec = Wfx.nSamplesPerSec * Wfx.nBlockAlign;
- // If the method succeeds and the audio endpoint device supports the specified stream format,
- // it returns S_OK. If the method succeeds and provides a closest match to the specified format,
- // it returns S_FALSE.
- hr = _ptrClientOut->IsFormatSupported(
- AUDCLNT_SHAREMODE_SHARED,
- &Wfx,
- &pWfxClosestMatch);
- if (hr == S_OK)
- {
- break;
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels=%d, nSamplesPerSec=%d is not supported",
- Wfx.nChannels, Wfx.nSamplesPerSec);
- }
- }
- if (hr == S_OK)
- break;
- }
-
- // TODO(andrew): what happens in the event of failure in the above loop?
- // Is _ptrClientOut->Initialize expected to fail?
- // Same in InitRecording().
- if (hr == S_OK)
- {
- _playAudioFrameSize = Wfx.nBlockAlign;
- _playBlockSize = Wfx.nSamplesPerSec/100;
- _playSampleRate = Wfx.nSamplesPerSec;
- _devicePlaySampleRate = Wfx.nSamplesPerSec; // The device itself continues to run at 44.1 kHz.
- _devicePlayBlockSize = Wfx.nSamplesPerSec/100;
- if (_playBlockSize == 441)
- {
- _playSampleRate = 44000; // we are actually running at 44000 Hz and *not* 44100 Hz
- _playBlockSize = 440; // adjust to size we can handle
- }
- _playChannels = Wfx.nChannels;
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "VoE selected this rendering format:");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wFormatTag : 0x%X (%u)", Wfx.wFormatTag, Wfx.wFormatTag);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels : %d", Wfx.nChannels);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nSamplesPerSec : %d", Wfx.nSamplesPerSec);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nAvgBytesPerSec : %d", Wfx.nAvgBytesPerSec);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nBlockAlign : %d", Wfx.nBlockAlign);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wBitsPerSample : %d", Wfx.wBitsPerSample);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "cbSize : %d", Wfx.cbSize);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Additional settings:");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_playAudioFrameSize: %d", _playAudioFrameSize);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_playBlockSize : %d", _playBlockSize);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_playChannels : %d", _playChannels);
- }
-
- _Get44kHzDrift();
-
- // Create a rendering stream.
- //
- // ****************************************************************************
- // For a shared-mode stream that uses event-driven buffering, the caller must
- // set both hnsPeriodicity and hnsBufferDuration to 0. The Initialize method
- // determines how large a buffer to allocate based on the scheduling period
- // of the audio engine. Although the client's buffer processing thread is
- // event driven, the basic buffer management process, as described previously,
- // is unaltered.
- // Each time the thread awakens, it should call IAudioClient::GetCurrentPadding
- // to determine how much data to write to a rendering buffer or read from a capture
- // buffer. In contrast to the two buffers that the Initialize method allocates
- // for an exclusive-mode stream that uses event-driven buffering, a shared-mode
- // stream requires a single buffer.
- // ****************************************************************************
- //
- REFERENCE_TIME hnsBufferDuration = 0; // ask for minimum buffer size (default)
- if (_devicePlaySampleRate == 44100)
- {
- // Ask for a larger buffer size (30ms) when using 44.1kHz as render rate.
- // There seems to be a larger risk of underruns for 44.1 compared
- // with the default rate (48kHz). When using default, we set the requested
- // buffer duration to 0, which sets the buffer to the minimum size
- // required by the engine thread. The actual buffer size can then be
- // read by GetBufferSize() and it is 20ms on most machines.
- hnsBufferDuration = 30*10000;
- }
- hr = _ptrClientOut->Initialize(
- AUDCLNT_SHAREMODE_SHARED, // share Audio Engine with other applications
- AUDCLNT_STREAMFLAGS_EVENTCALLBACK, // processing of the audio buffer by the client will be event driven
- hnsBufferDuration, // requested buffer capacity as a time value (in 100-nanosecond units)
- 0, // periodicity
- &Wfx, // selected wave format
- NULL); // session GUID
-
- if (FAILED(hr))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "IAudioClient::Initialize() failed:");
- if (pWfxClosestMatch != NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "closest mix format: #channels=%d, samples/sec=%d, bits/sample=%d",
- pWfxClosestMatch->nChannels, pWfxClosestMatch->nSamplesPerSec, pWfxClosestMatch->wBitsPerSample);
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "no format suggested");
- }
- }
- EXIT_ON_ERROR(hr);
-
- if (_ptrAudioBuffer)
- {
- // Update the audio buffer with the selected parameters
- _ptrAudioBuffer->SetPlayoutSampleRate(_playSampleRate);
- _ptrAudioBuffer->SetPlayoutChannels((WebRtc_UWord8)_playChannels);
- }
- else
- {
- // We can enter this state during CoreAudioIsSupported() when no AudioDeviceImplementation
- // has been created, hence the AudioDeviceBuffer does not exist.
- // It is OK to end up here since we don't initiate any media in CoreAudioIsSupported().
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceBuffer must be attached before streaming can start");
- }
-
- // Get the actual size of the shared (endpoint buffer).
- // Typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
- UINT bufferFrameCount(0);
- hr = _ptrClientOut->GetBufferSize(
- &bufferFrameCount);
- if (SUCCEEDED(hr))
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "IAudioClient::GetBufferSize() => %u (<=> %u bytes)",
- bufferFrameCount, bufferFrameCount*_playAudioFrameSize);
- }
-
- // Set the event handle that the system signals when an audio buffer is ready
- // to be processed by the client.
- hr = _ptrClientOut->SetEventHandle(
- _hRenderSamplesReadyEvent);
- EXIT_ON_ERROR(hr);
-
- // Get an IAudioRenderClient interface.
- SAFE_RELEASE(_ptrRenderClient);
- hr = _ptrClientOut->GetService(
- __uuidof(IAudioRenderClient),
- (void**)&_ptrRenderClient);
- EXIT_ON_ERROR(hr);
-
- // Mark playout side as initialized
- _playIsInitialized = true;
-
- CoTaskMemFree(pWfxOut);
- CoTaskMemFree(pWfxClosestMatch);
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "render side is now initialized");
- return 0;
-
-Exit:
- _TraceCOMError(hr);
- CoTaskMemFree(pWfxOut);
- CoTaskMemFree(pWfxClosestMatch);
- SAFE_RELEASE(_ptrClientOut);
- SAFE_RELEASE(_ptrRenderClient);
- return -1;
-}
-
-// Capture initialization when the built-in AEC DirectX Media Object (DMO) is
-// used. Called from InitRecording(), most of which is skipped over. The DMO
-// handles device initialization itself.
-// Reference: http://msdn.microsoft.com/en-us/library/ff819492(v=vs.85).aspx
-WebRtc_Word32 AudioDeviceWindowsCore::InitRecordingDMO()
-{
- assert(_builtInAecEnabled);
- assert(_dmo != NULL);
-
- if (SetDMOProperties() == -1)
- {
- return -1;
- }
-
- DMO_MEDIA_TYPE mt = {0};
- HRESULT hr = MoInitMediaType(&mt, sizeof(WAVEFORMATEX));
- if (FAILED(hr))
- {
- MoFreeMediaType(&mt);
- _TraceCOMError(hr);
- return -1;
- }
- mt.majortype = MEDIATYPE_Audio;
- mt.subtype = MEDIASUBTYPE_PCM;
- mt.formattype = FORMAT_WaveFormatEx;
-
- // Supported formats
- // nChannels: 1 (in AEC-only mode)
- // nSamplesPerSec: 8000, 11025, 16000, 22050
- // wBitsPerSample: 16
- WAVEFORMATEX* ptrWav = reinterpret_cast<WAVEFORMATEX*>(mt.pbFormat);
- ptrWav->wFormatTag = WAVE_FORMAT_PCM;
- ptrWav->nChannels = 1;
- // 16000 is the highest we can support with our resampler.
- ptrWav->nSamplesPerSec = 16000;
- ptrWav->nAvgBytesPerSec = 32000;
- ptrWav->nBlockAlign = 2;
- ptrWav->wBitsPerSample = 16;
- ptrWav->cbSize = 0;
-
- // Set the VoE format equal to the AEC output format.
- _recAudioFrameSize = ptrWav->nBlockAlign;
- _recSampleRate = ptrWav->nSamplesPerSec;
- _recBlockSize = ptrWav->nSamplesPerSec / 100;
- _recChannels = ptrWav->nChannels;
-
- // Set the DMO output format parameters.
- hr = _dmo->SetOutputType(kAecCaptureStreamIndex, &mt, 0);
- MoFreeMediaType(&mt);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- return -1;
- }
-
- if (_ptrAudioBuffer)
- {
- _ptrAudioBuffer->SetRecordingSampleRate(_recSampleRate);
- _ptrAudioBuffer->SetRecordingChannels(_recChannels);
- }
- else
- {
- // Refer to InitRecording() for comments.
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "AudioDeviceBuffer must be attached before streaming can start");
- }
-
- _mediaBuffer = new MediaBufferImpl(_recBlockSize * _recAudioFrameSize);
-
- // Optional, but if called, must be after media types are set.
- hr = _dmo->AllocateStreamingResources();
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- return -1;
- }
-
- _recIsInitialized = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Capture side is now initialized");
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::InitRecording()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_recording)
- {
- return -1;
- }
-
- if (_recIsInitialized)
- {
- return 0;
- }
-
- if (QueryPerformanceFrequency(&_perfCounterFreq) == 0)
- {
- return -1;
- }
- _perfCounterFactor = 10000000.0 / (double)_perfCounterFreq.QuadPart;
-
- if (_ptrDeviceIn == NULL)
- {
- return -1;
- }
-
- // Initialize the microphone (devices might have been added or removed)
- if (InitMicrophone() == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "InitMicrophone() failed");
- }
-
- // Ensure that the updated capturing endpoint device is valid
- if (_ptrDeviceIn == NULL)
- {
- return -1;
- }
-
- if (_builtInAecEnabled)
- {
- // The DMO will configure the capture device.
- return InitRecordingDMO();
- }
-
- HRESULT hr = S_OK;
- WAVEFORMATEX* pWfxIn = NULL;
- WAVEFORMATEX Wfx;
- WAVEFORMATEX* pWfxClosestMatch = NULL;
-
- // Create COM object with IAudioClient interface.
- SAFE_RELEASE(_ptrClientIn);
- hr = _ptrDeviceIn->Activate(
- __uuidof(IAudioClient),
- CLSCTX_ALL,
- NULL,
- (void**)&_ptrClientIn);
- EXIT_ON_ERROR(hr);
-
- // Retrieve the stream format that the audio engine uses for its internal
- // processing (mixing) of shared-mode streams.
- hr = _ptrClientIn->GetMixFormat(&pWfxIn);
- if (SUCCEEDED(hr))
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Audio Engine's current capturing mix format:");
- // format type
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wFormatTag : 0x%X (%u)", pWfxIn->wFormatTag, pWfxIn->wFormatTag);
- // number of channels (i.e. mono, stereo...)
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels : %d", pWfxIn->nChannels);
- // sample rate
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nSamplesPerSec : %d", pWfxIn->nSamplesPerSec);
- // for buffer estimation
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nAvgBytesPerSec: %d", pWfxIn->nAvgBytesPerSec);
- // block size of data
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nBlockAlign : %d", pWfxIn->nBlockAlign);
- // number of bits per sample of mono data
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wBitsPerSample : %d", pWfxIn->wBitsPerSample);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "cbSize : %d", pWfxIn->cbSize);
- }
-
- // Set wave format
- Wfx.wFormatTag = WAVE_FORMAT_PCM;
- Wfx.wBitsPerSample = 16;
- Wfx.cbSize = 0;
-
- const int freqs[6] = {48000, 44100, 16000, 96000, 32000, 8000};
- hr = S_FALSE;
-
- // Iterate over frequencies and channels, in order of priority
- for (int freq = 0; freq < sizeof(freqs)/sizeof(freqs[0]); freq++)
- {
- for (int chan = 0; chan < sizeof(_recChannelsPrioList)/sizeof(_recChannelsPrioList[0]); chan++)
- {
- Wfx.nChannels = _recChannelsPrioList[chan];
- Wfx.nSamplesPerSec = freqs[freq];
- Wfx.nBlockAlign = Wfx.nChannels * Wfx.wBitsPerSample / 8;
- Wfx.nAvgBytesPerSec = Wfx.nSamplesPerSec * Wfx.nBlockAlign;
- // If the method succeeds and the audio endpoint device supports the specified stream format,
- // it returns S_OK. If the method succeeds and provides a closest match to the specified format,
- // it returns S_FALSE.
- hr = _ptrClientIn->IsFormatSupported(
- AUDCLNT_SHAREMODE_SHARED,
- &Wfx,
- &pWfxClosestMatch);
- if (hr == S_OK)
- {
- break;
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels=%d, nSamplesPerSec=%d is not supported",
- Wfx.nChannels, Wfx.nSamplesPerSec);
- }
- }
- if (hr == S_OK)
- break;
- }
-
- if (hr == S_OK)
- {
- _recAudioFrameSize = Wfx.nBlockAlign;
- _recSampleRate = Wfx.nSamplesPerSec;
- _recBlockSize = Wfx.nSamplesPerSec/100;
- _recChannels = Wfx.nChannels;
- if (_recBlockSize == 441)
- {
- _recSampleRate = 44000; // we are actually using 44000 Hz and *not* 44100 Hz
- _recBlockSize = 440; // adjust to size we can handle
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "VoE selected this capturing format:");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wFormatTag : 0x%X (%u)", Wfx.wFormatTag, Wfx.wFormatTag);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nChannels : %d", Wfx.nChannels);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nSamplesPerSec : %d", Wfx.nSamplesPerSec);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nAvgBytesPerSec : %d", Wfx.nAvgBytesPerSec);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "nBlockAlign : %d", Wfx.nBlockAlign);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wBitsPerSample : %d", Wfx.wBitsPerSample);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "cbSize : %d", Wfx.cbSize);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Additional settings:");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_recAudioFrameSize: %d", _recAudioFrameSize);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_recBlockSize : %d", _recBlockSize);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_recChannels : %d", _recChannels);
- }
-
- _Get44kHzDrift();
-
- // Create a capturing stream.
- hr = _ptrClientIn->Initialize(
- AUDCLNT_SHAREMODE_SHARED, // share Audio Engine with other applications
- AUDCLNT_STREAMFLAGS_EVENTCALLBACK | // processing of the audio buffer by the client will be event driven
- AUDCLNT_STREAMFLAGS_NOPERSIST, // volume and mute settings for an audio session will not persist across system restarts
- 0, // required for event-driven shared mode
- 0, // periodicity
- &Wfx, // selected wave format
- NULL); // session GUID
-
-
- if (hr != S_OK)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "IAudioClient::Initialize() failed:");
- if (pWfxClosestMatch != NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "closest mix format: #channels=%d, samples/sec=%d, bits/sample=%d",
- pWfxClosestMatch->nChannels, pWfxClosestMatch->nSamplesPerSec, pWfxClosestMatch->wBitsPerSample);
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "no format suggested");
- }
- }
- EXIT_ON_ERROR(hr);
-
- if (_ptrAudioBuffer)
- {
- // Update the audio buffer with the selected parameters
- _ptrAudioBuffer->SetRecordingSampleRate(_recSampleRate);
- _ptrAudioBuffer->SetRecordingChannels((WebRtc_UWord8)_recChannels);
- }
- else
- {
- // We can enter this state during CoreAudioIsSupported() when no AudioDeviceImplementation
- // has been created, hence the AudioDeviceBuffer does not exist.
- // It is OK to end up here since we don't initiate any media in CoreAudioIsSupported().
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioDeviceBuffer must be attached before streaming can start");
- }
-
- // Get the actual size of the shared (endpoint buffer).
- // Typical value is 960 audio frames <=> 20ms @ 48kHz sample rate.
- UINT bufferFrameCount(0);
- hr = _ptrClientIn->GetBufferSize(
- &bufferFrameCount);
- if (SUCCEEDED(hr))
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "IAudioClient::GetBufferSize() => %u (<=> %u bytes)",
- bufferFrameCount, bufferFrameCount*_recAudioFrameSize);
- }
-
- // Set the event handle that the system signals when an audio buffer is ready
- // to be processed by the client.
- hr = _ptrClientIn->SetEventHandle(
- _hCaptureSamplesReadyEvent);
- EXIT_ON_ERROR(hr);
-
- // Get an IAudioCaptureClient interface.
- SAFE_RELEASE(_ptrCaptureClient);
- hr = _ptrClientIn->GetService(
- __uuidof(IAudioCaptureClient),
- (void**)&_ptrCaptureClient);
- EXIT_ON_ERROR(hr);
-
- // Mark capture side as initialized
- _recIsInitialized = true;
-
- CoTaskMemFree(pWfxIn);
- CoTaskMemFree(pWfxClosestMatch);
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "capture side is now initialized");
- return 0;
-
-Exit:
- _TraceCOMError(hr);
- CoTaskMemFree(pWfxIn);
- CoTaskMemFree(pWfxClosestMatch);
- SAFE_RELEASE(_ptrClientIn);
- SAFE_RELEASE(_ptrCaptureClient);
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// StartRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::StartRecording()
-{
-
- if (!_recIsInitialized)
- {
- return -1;
- }
-
- if (_hRecThread != NULL)
- {
- return 0;
- }
-
- if (_recording)
- {
- return 0;
- }
-
- HRESULT hr = S_OK;
- {
- CriticalSectionScoped critScoped(_critSect);
-
- // Create thread which will drive the capturing
- LPTHREAD_START_ROUTINE lpStartAddress = WSAPICaptureThread;
- if (_builtInAecEnabled)
- {
- // Redirect to the DMO polling method.
- lpStartAddress = WSAPICaptureThreadPollDMO;
-
- if (!_playing)
- {
- // The DMO won't provide us captured output data unless we
- // give it render data to process.
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Playout must be started before recording when using the "
- "built-in AEC");
- return -1;
- }
- }
-
- assert(_hRecThread == NULL);
- _hRecThread = CreateThread(NULL,
- 0,
- lpStartAddress,
- this,
- 0,
- NULL);
- if (_hRecThread == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "failed to create the recording thread");
- return -1;
- }
-
- // Set thread priority to highest possible
- SetThreadPriority(_hRecThread, THREAD_PRIORITY_TIME_CRITICAL);
-
- assert(_hGetCaptureVolumeThread == NULL);
- _hGetCaptureVolumeThread = CreateThread(NULL,
- 0,
- GetCaptureVolumeThread,
- this,
- 0,
- NULL);
- if (_hGetCaptureVolumeThread == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to create the volume getter thread");
- return -1;
- }
-
- assert(_hSetCaptureVolumeThread == NULL);
- _hSetCaptureVolumeThread = CreateThread(NULL,
- 0,
- SetCaptureVolumeThread,
- this,
- 0,
- NULL);
- if (_hSetCaptureVolumeThread == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to create the volume setter thread");
- return -1;
- }
- } // critScoped
-
- DWORD ret = WaitForSingleObject(_hCaptureStartedEvent, 1000);
- if (ret != WAIT_OBJECT_0)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "capturing did not start up properly");
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "capture audio stream has now started...");
-
- _avgCPULoad = 0.0f;
- _playAcc = 0;
- _recording = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StopRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::StopRecording()
-{
- WebRtc_Word32 err = 0;
-
- if (!_recIsInitialized)
- {
- return 0;
- }
-
- _Lock();
-
- if (_hRecThread == NULL)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "no capturing stream is active => close down WASAPI only");
- SAFE_RELEASE(_ptrClientIn);
- SAFE_RELEASE(_ptrCaptureClient);
- _recIsInitialized = false;
- _recording = false;
- _UnLock();
- return 0;
- }
-
- // Stop the driving thread...
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "closing down the webrtc_core_audio_capture_thread...");
- // Manual-reset event; it will remain signalled to stop all capture threads.
- SetEvent(_hShutdownCaptureEvent);
-
- _UnLock();
- DWORD ret = WaitForSingleObject(_hRecThread, 2000);
- if (ret != WAIT_OBJECT_0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "failed to close down webrtc_core_audio_capture_thread");
- err = -1;
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "webrtc_core_audio_capture_thread is now closed");
- }
-
- ret = WaitForSingleObject(_hGetCaptureVolumeThread, 2000);
- if (ret != WAIT_OBJECT_0)
- {
- // the thread did not stop as it should
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to close down volume getter thread");
- err = -1;
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " volume getter thread is now closed");
- }
-
- ret = WaitForSingleObject(_hSetCaptureVolumeThread, 2000);
- if (ret != WAIT_OBJECT_0)
- {
- // the thread did not stop as it should
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to close down volume setter thread");
- err = -1;
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " volume setter thread is now closed");
- }
- _Lock();
-
- ResetEvent(_hShutdownCaptureEvent); // Must be manually reset.
- // Ensure that the thread has released these interfaces properly.
- assert(err == -1 || _ptrClientIn == NULL);
- assert(err == -1 || _ptrCaptureClient == NULL);
-
- _recIsInitialized = false;
- _recording = false;
-
- // These will create thread leaks in the result of an error,
- // but we can at least resume the call.
- CloseHandle(_hRecThread);
- _hRecThread = NULL;
-
- CloseHandle(_hGetCaptureVolumeThread);
- _hGetCaptureVolumeThread = NULL;
-
- CloseHandle(_hSetCaptureVolumeThread);
- _hSetCaptureVolumeThread = NULL;
-
- if (_builtInAecEnabled)
- {
- assert(_dmo != NULL);
- // This is necessary. Otherwise the DMO can generate garbage render
- // audio even after rendering has stopped.
- HRESULT hr = _dmo->FreeStreamingResources();
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- err = -1;
- }
- }
-
- _UnLock();
-
- // Reset the recording delay value.
- _sndCardRecDelay = 0;
-
- return err;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsCore::RecordingIsInitialized() const
-{
- return (_recIsInitialized);
-}
-
-// ----------------------------------------------------------------------------
-// Recording
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsCore::Recording() const
-{
- return (_recording);
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsCore::PlayoutIsInitialized() const
-{
-
- return (_playIsInitialized);
-}
-
-// ----------------------------------------------------------------------------
-// StartPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::StartPlayout()
-{
-
- if (!_playIsInitialized)
- {
- return -1;
- }
-
- if (_hPlayThread != NULL)
- {
- return 0;
- }
-
- if (_playing)
- {
- return 0;
- }
-
- HRESULT hr = S_OK;
- {
- CriticalSectionScoped critScoped(_critSect);
-
- // Create thread which will drive the rendering.
- assert(_hPlayThread == NULL);
- _hPlayThread = CreateThread(
- NULL,
- 0,
- WSAPIRenderThread,
- this,
- 0,
- NULL);
- if (_hPlayThread == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "failed to create the playout thread");
- return -1;
- }
-
- // Set thread priority to highest possible.
- SetThreadPriority(_hPlayThread, THREAD_PRIORITY_TIME_CRITICAL);
- } // critScoped
-
- DWORD ret = WaitForSingleObject(_hRenderStartedEvent, 1000);
- if (ret != WAIT_OBJECT_0)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "rendering did not start up properly");
- return -1;
- }
-
- _playing = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "rendering audio stream has now started...");
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StopPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::StopPlayout()
-{
-
- if (!_playIsInitialized)
- {
- return 0;
- }
-
- {
- CriticalSectionScoped critScoped(_critSect) ;
-
- if (_hPlayThread == NULL)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "no rendering stream is active => close down WASAPI only");
- SAFE_RELEASE(_ptrClientOut);
- SAFE_RELEASE(_ptrRenderClient);
- _playIsInitialized = false;
- _playing = false;
- return 0;
- }
-
- // stop the driving thread...
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "closing down the webrtc_core_audio_render_thread...");
- SetEvent(_hShutdownRenderEvent);
- } // critScoped
-
- DWORD ret = WaitForSingleObject(_hPlayThread, 2000);
- if (ret != WAIT_OBJECT_0)
- {
- // the thread did not stop as it should
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "failed to close down webrtc_core_audio_render_thread");
- CloseHandle(_hPlayThread);
- _hPlayThread = NULL;
- _playIsInitialized = false;
- _playing = false;
- return -1;
- }
-
- {
- CriticalSectionScoped critScoped(_critSect);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "webrtc_core_audio_render_thread is now closed");
-
- // to reset this event manually at each time we finish with it,
- // in case that the render thread has exited before StopPlayout(),
- // this event might be caught by the new render thread within same VoE instance.
- ResetEvent(_hShutdownRenderEvent);
-
- SAFE_RELEASE(_ptrClientOut);
- SAFE_RELEASE(_ptrRenderClient);
-
- _playIsInitialized = false;
- _playing = false;
-
- CloseHandle(_hPlayThread);
- _hPlayThread = NULL;
-
- if (_builtInAecEnabled && _recording)
- {
- // The DMO won't provide us captured output data unless we
- // give it render data to process.
- //
- // We still permit the playout to shutdown, and trace a warning.
- // Otherwise, VoE can get into a state which will never permit
- // playout to stop properly.
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "Recording should be stopped before playout when using the "
- "built-in AEC");
- }
- } // critScoped
-
- // Reset the playout delay value.
- _sndCardPlayDelay = 0;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDelay
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::PlayoutDelay(WebRtc_UWord16& delayMS) const
-{
- delayMS = static_cast<WebRtc_UWord16>(_sndCardPlayDelay.Value());
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDelay
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::RecordingDelay(WebRtc_UWord16& delayMS) const
-{
- delayMS = static_cast<WebRtc_UWord16>(_sndCardRecDelay.Value());
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Playing
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsCore::Playing() const
-{
- return (_playing);
-}
-// ----------------------------------------------------------------------------
-// SetPlayoutBuffer
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::SetPlayoutBuffer(const AudioDeviceModule::BufferType type, WebRtc_UWord16 sizeMS)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- _playBufType = type;
-
- if (type == AudioDeviceModule::kFixedBufferSize)
- {
- _playBufDelayFixed = sizeMS;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutBuffer
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::PlayoutBuffer(AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const
-{
- {
- CriticalSectionScoped lock(_critSect);
- type = _playBufType;
- }
-
- if (type == AudioDeviceModule::kFixedBufferSize)
- {
- CriticalSectionScoped lock(_critSect);
- sizeMS = _playBufDelayFixed;
- }
- else
- {
- // Use same value as for PlayoutDelay
- sizeMS = static_cast<WebRtc_UWord16>(_sndCardPlayDelay.Value());
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// CPULoad
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::CPULoad(WebRtc_UWord16& load) const
-{
-
- load = static_cast<WebRtc_UWord16> (100*_avgCPULoad);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutWarning
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsCore::PlayoutWarning() const
-{
- return ( _playWarning > 0);
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutError
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsCore::PlayoutError() const
-{
- return ( _playError > 0);
-}
-
-// ----------------------------------------------------------------------------
-// RecordingWarning
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsCore::RecordingWarning() const
-{
- return ( _recWarning > 0);
-}
-
-// ----------------------------------------------------------------------------
-// RecordingError
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsCore::RecordingError() const
-{
- return ( _recError > 0);
-}
-
-// ----------------------------------------------------------------------------
-// ClearPlayoutWarning
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsCore::ClearPlayoutWarning()
-{
- _playWarning = 0;
-}
-
-// ----------------------------------------------------------------------------
-// ClearPlayoutError
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsCore::ClearPlayoutError()
-{
- _playError = 0;
-}
-
-// ----------------------------------------------------------------------------
-// ClearRecordingWarning
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsCore::ClearRecordingWarning()
-{
- _recWarning = 0;
-}
-
-// ----------------------------------------------------------------------------
-// ClearRecordingError
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsCore::ClearRecordingError()
-{
- _recError = 0;
-}
-
-// ============================================================================
-// Private Methods
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// [static] WSAPIRenderThread
-// ----------------------------------------------------------------------------
-
-DWORD WINAPI AudioDeviceWindowsCore::WSAPIRenderThread(LPVOID context)
-{
- return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
- DoRenderThread();
-}
-
-// ----------------------------------------------------------------------------
-// [static] WSAPICaptureThread
-// ----------------------------------------------------------------------------
-
-DWORD WINAPI AudioDeviceWindowsCore::WSAPICaptureThread(LPVOID context)
-{
- return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
- DoCaptureThread();
-}
-
-DWORD WINAPI AudioDeviceWindowsCore::WSAPICaptureThreadPollDMO(LPVOID context)
-{
- return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
- DoCaptureThreadPollDMO();
-}
-
-DWORD WINAPI AudioDeviceWindowsCore::GetCaptureVolumeThread(LPVOID context)
-{
- return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
- DoGetCaptureVolumeThread();
-}
-
-DWORD WINAPI AudioDeviceWindowsCore::SetCaptureVolumeThread(LPVOID context)
-{
- return reinterpret_cast<AudioDeviceWindowsCore*>(context)->
- DoSetCaptureVolumeThread();
-}
-
-DWORD AudioDeviceWindowsCore::DoGetCaptureVolumeThread()
-{
- HANDLE waitObject = _hShutdownCaptureEvent;
-
- while (1)
- {
- if (AGC())
- {
- WebRtc_UWord32 currentMicLevel = 0;
- if (MicrophoneVolume(currentMicLevel) == 0)
- {
- // This doesn't set the system volume, just stores it.
- _Lock();
- if (_ptrAudioBuffer)
- {
- _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
- }
- _UnLock();
- }
- }
-
- DWORD waitResult = WaitForSingleObject(waitObject,
- GET_MIC_VOLUME_INTERVAL_MS);
- switch (waitResult)
- {
- case WAIT_OBJECT_0: // _hShutdownCaptureEvent
- return 0;
- case WAIT_TIMEOUT: // timeout notification
- break;
- default: // unexpected error
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " unknown wait termination on get volume thread");
- return -1;
- }
- }
-}
-
-DWORD AudioDeviceWindowsCore::DoSetCaptureVolumeThread()
-{
- HANDLE waitArray[2] = {_hShutdownCaptureEvent, _hSetCaptureVolumeEvent};
-
- while (1)
- {
- DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, INFINITE);
- switch (waitResult)
- {
- case WAIT_OBJECT_0: // _hShutdownCaptureEvent
- return 0;
- case WAIT_OBJECT_0 + 1: // _hSetCaptureVolumeEvent
- break;
- default: // unexpected error
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " unknown wait termination on set volume thread");
- return -1;
- }
-
- _Lock();
- WebRtc_UWord32 newMicLevel = _newMicLevel;
- _UnLock();
-
- if (SetMicrophoneVolume(newMicLevel) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " the required modification of the microphone volume failed");
- }
- }
-}
-
-// ----------------------------------------------------------------------------
-// DoRenderThread
-// ----------------------------------------------------------------------------
-
-DWORD AudioDeviceWindowsCore::DoRenderThread()
-{
-
- bool keepPlaying = true;
- HANDLE waitArray[2] = {_hShutdownRenderEvent, _hRenderSamplesReadyEvent};
- HRESULT hr = S_OK;
- HANDLE hMmTask = NULL;
-
- LARGE_INTEGER t1;
- LARGE_INTEGER t2;
- WebRtc_Word32 time(0);
-
- // Initialize COM as MTA in this thread.
- ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
- if (!comInit.succeeded()) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "failed to initialize COM in render thread");
- return -1;
- }
-
- _SetThreadName(-1, "webrtc_core_audio_render_thread");
-
- // Use Multimedia Class Scheduler Service (MMCSS) to boost the thread priority.
- //
- if (_winSupportAvrt)
- {
- DWORD taskIndex(0);
- hMmTask = _PAvSetMmThreadCharacteristicsA("Pro Audio", &taskIndex);
- if (hMmTask)
- {
- if (FALSE == _PAvSetMmThreadPriority(hMmTask, AVRT_PRIORITY_CRITICAL))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to boost play-thread using MMCSS");
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "render thread is now registered with MMCSS (taskIndex=%d)", taskIndex);
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to enable MMCSS on render thread (err=%d)", GetLastError());
- _TraceCOMError(GetLastError());
- }
- }
-
- _Lock();
-
- // Get size of rendering buffer (length is expressed as the number of audio frames the buffer can hold).
- // This value is fixed during the rendering session.
- //
- UINT32 bufferLength = 0;
- hr = _ptrClientOut->GetBufferSize(&bufferLength);
- EXIT_ON_ERROR(hr);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[REND] size of buffer : %u", bufferLength);
-
- // Get maximum latency for the current stream (will not change for the lifetime of the IAudioClient object).
- //
- REFERENCE_TIME latency;
- _ptrClientOut->GetStreamLatency(&latency);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[REND] max stream latency : %u (%3.2f ms)",
- (DWORD)latency, (double)(latency/10000.0));
-
- // Get the length of the periodic interval separating successive processing passes by
- // the audio engine on the data in the endpoint buffer.
- //
- // The period between processing passes by the audio engine is fixed for a particular
- // audio endpoint device and represents the smallest processing quantum for the audio engine.
- // This period plus the stream latency between the buffer and endpoint device represents
- // the minimum possible latency that an audio application can achieve.
- // Typical value: 100000 <=> 0.01 sec = 10ms.
- //
- REFERENCE_TIME devPeriod = 0;
- REFERENCE_TIME devPeriodMin = 0;
- _ptrClientOut->GetDevicePeriod(&devPeriod, &devPeriodMin);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[REND] device period : %u (%3.2f ms)",
- (DWORD)devPeriod, (double)(devPeriod/10000.0));
-
- // The Event Driven renderer will be woken up every defaultDevicePeriod hundred-nano-seconds.
- // Convert that time into a number of frames.
- //
- double devicePeriodInSeconds = devPeriod / (10000.0*1000.0);
- UINT32 devicePeriodInFrames = static_cast<UINT32>(_playSampleRate * devicePeriodInSeconds + 0.5);
-
- // Derive inital rendering delay.
- // Example: 10*(960/480) + 15 = 20 + 15 = 35ms
- //
- int playout_delay = 10 * (bufferLength / _playBlockSize) +
- (int)((latency + devPeriod) / 10000);
- _sndCardPlayDelay = playout_delay;
- _writtenSamples = 0;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "[REND] initial delay : %u", playout_delay);
-
- double endpointBufferSizeMS = 10.0 * ((double)bufferLength / (double)_devicePlayBlockSize);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[REND] endpointBufferSizeMS : %3.2f", endpointBufferSizeMS);
-
- // Before starting the stream, fill the rendering buffer with silence.
- //
- BYTE *pData = NULL;
- hr = _ptrRenderClient->GetBuffer(bufferLength, &pData);
- EXIT_ON_ERROR(hr);
-
- hr = _ptrRenderClient->ReleaseBuffer(bufferLength, AUDCLNT_BUFFERFLAGS_SILENT);
- EXIT_ON_ERROR(hr);
-
- _writtenSamples += bufferLength;
-
- IAudioClock* clock = NULL;
- hr = _ptrClientOut->GetService(__uuidof(IAudioClock), (void**)&clock);
- if (FAILED(hr)) {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "failed to get IAudioClock interface from the IAudioClient");
- }
-
- // Start up the rendering audio stream.
- hr = _ptrClientOut->Start();
- EXIT_ON_ERROR(hr);
-
- _UnLock();
-
- // Set event which will ensure that the calling thread modifies the playing state to true.
- //
- SetEvent(_hRenderStartedEvent);
-
- // >> ------------------ THREAD LOOP ------------------
-
- while (keepPlaying)
- {
- // Wait for a render notification event or a shutdown event
- DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, 500);
- switch (waitResult)
- {
- case WAIT_OBJECT_0 + 0: // _hShutdownRenderEvent
- keepPlaying = false;
- break;
- case WAIT_OBJECT_0 + 1: // _hRenderSamplesReadyEvent
- break;
- case WAIT_TIMEOUT: // timeout notification
- _ptrClientOut->Stop();
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "render event timed out after 0.5 seconds");
- goto Exit;
- default: // unexpected error
- _ptrClientOut->Stop();
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "unknown wait termination on render side");
- goto Exit;
- }
-
- while (keepPlaying)
- {
- _Lock();
-
- // Get the number of frames of padding (queued up to play) in the endpoint buffer.
- UINT32 padding = 0;
- hr = _ptrClientOut->GetCurrentPadding(&padding);
- EXIT_ON_ERROR(hr);
-
- // Derive the amount of available space in the output buffer
- WebRtc_UWord32 framesAvailable = bufferLength - padding;
- // WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "#avaliable audio frames = %u", framesAvailable);
-
- // Do we have 10 ms available in the render buffer?
- if (framesAvailable < _playBlockSize)
- {
- // Not enough space in render buffer to store next render packet.
- _UnLock();
- break;
- }
-
- // Write n*10ms buffers to the render buffer
- const WebRtc_UWord32 n10msBuffers = (framesAvailable / _playBlockSize);
- for (WebRtc_UWord32 n = 0; n < n10msBuffers; n++)
- {
- // Get pointer (i.e., grab the buffer) to next space in the shared render buffer.
- hr = _ptrRenderClient->GetBuffer(_playBlockSize, &pData);
- EXIT_ON_ERROR(hr);
-
- QueryPerformanceCounter(&t1); // measure time: START
-
- if (_ptrAudioBuffer)
- {
- // Request data to be played out (#bytes = _playBlockSize*_audioFrameSize)
- _UnLock();
- WebRtc_Word32 nSamples =
- _ptrAudioBuffer->RequestPlayoutData(_playBlockSize);
- _Lock();
-
- if (nSamples == -1)
- {
- _UnLock();
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- "failed to read data from render client");
- goto Exit;
- }
-
- // Sanity check to ensure that essential states are not modified during the unlocked period
- if (_ptrRenderClient == NULL || _ptrClientOut == NULL)
- {
- _UnLock();
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "output state has been modified during unlocked period");
- goto Exit;
- }
- if (nSamples != _playBlockSize)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "nSamples(%d) != _playBlockSize(%d)", nSamples, _playBlockSize);
- }
-
- // Get the actual (stored) data
- nSamples = _ptrAudioBuffer->GetPlayoutData((WebRtc_Word8*)pData);
- }
-
- QueryPerformanceCounter(&t2); // measure time: STOP
- time = (int)(t2.QuadPart-t1.QuadPart);
- _playAcc += time;
-
- DWORD dwFlags(0);
- hr = _ptrRenderClient->ReleaseBuffer(_playBlockSize, dwFlags);
- // See http://msdn.microsoft.com/en-us/library/dd316605(VS.85).aspx
- // for more details regarding AUDCLNT_E_DEVICE_INVALIDATED.
- EXIT_ON_ERROR(hr);
-
- _writtenSamples += _playBlockSize;
- }
-
- // Check the current delay on the playout side.
- if (clock) {
- UINT64 pos = 0;
- UINT64 freq = 1;
- clock->GetPosition(&pos, NULL);
- clock->GetFrequency(&freq);
- playout_delay = ROUND((double(_writtenSamples) /
- _devicePlaySampleRate - double(pos) / freq) * 1000.0);
- _sndCardPlayDelay = playout_delay;
- }
-
- _UnLock();
- }
- }
-
- // ------------------ THREAD LOOP ------------------ <<
-
- Sleep(static_cast<DWORD>(endpointBufferSizeMS+0.5));
- hr = _ptrClientOut->Stop();
-
-Exit:
- SAFE_RELEASE(clock);
-
- if (FAILED(hr))
- {
- _UnLock();
- _ptrClientOut->Stop();
- _TraceCOMError(hr);
- }
-
- if (_winSupportAvrt)
- {
- if (NULL != hMmTask)
- {
- _PAvRevertMmThreadCharacteristics(hMmTask);
- }
- }
-
- if (keepPlaying)
- {
- hr = _ptrClientOut->Stop();
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- }
- hr = _ptrClientOut->Reset();
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- }
-
- // Trigger callback from module process thread
- _playError = 1;
- WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kPlayoutError message posted: rendering thread has ended pre-maturely");
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_Rendering thread is now terminated properly");
- }
-
- return (DWORD)hr;
-}
-
-DWORD AudioDeviceWindowsCore::InitCaptureThreadPriority()
-{
- HRESULT hr = S_OK;
- _hMmTask = NULL;
-
- _SetThreadName(-1, "webrtc_core_audio_capture_thread");
-
- // Use Multimedia Class Scheduler Service (MMCSS) to boost the thread
- // priority.
- if (_winSupportAvrt)
- {
- DWORD taskIndex(0);
- _hMmTask = _PAvSetMmThreadCharacteristicsA("Pro Audio", &taskIndex);
- if (_hMmTask)
- {
- if (!_PAvSetMmThreadPriority(_hMmTask, AVRT_PRIORITY_CRITICAL))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "failed to boost rec-thread using MMCSS");
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "capture thread is now registered with MMCSS (taskIndex=%d)",
- taskIndex);
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "failed to enable MMCSS on capture thread (err=%d)",
- GetLastError());
- _TraceCOMError(GetLastError());
- }
- }
-
- return S_OK;
-}
-
-void AudioDeviceWindowsCore::RevertCaptureThreadPriority()
-{
- if (_winSupportAvrt)
- {
- if (NULL != _hMmTask)
- {
- _PAvRevertMmThreadCharacteristics(_hMmTask);
- }
- }
-
- _hMmTask = NULL;
-}
-
-DWORD AudioDeviceWindowsCore::DoCaptureThreadPollDMO()
-{
- assert(_mediaBuffer != NULL);
- bool keepRecording = true;
-
- // Initialize COM as MTA in this thread.
- ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
- if (!comInit.succeeded()) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "failed to initialize COM in polling DMO thread");
- return -1;
- }
-
- HRESULT hr = InitCaptureThreadPriority();
- if (FAILED(hr))
- {
- return hr;
- }
-
- // Set event which will ensure that the calling thread modifies the
- // recording state to true.
- SetEvent(_hCaptureStartedEvent);
-
- // >> ---------------------------- THREAD LOOP ----------------------------
- while (keepRecording)
- {
- // Poll the DMO every 5 ms.
- // (The same interval used in the Wave implementation.)
- DWORD waitResult = WaitForSingleObject(_hShutdownCaptureEvent, 5);
- switch (waitResult)
- {
- case WAIT_OBJECT_0: // _hShutdownCaptureEvent
- keepRecording = false;
- break;
- case WAIT_TIMEOUT: // timeout notification
- break;
- default: // unexpected error
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "Unknown wait termination on capture side");
- hr = -1; // To signal an error callback.
- keepRecording = false;
- break;
- }
-
- while (keepRecording)
- {
- CriticalSectionScoped critScoped(_critSect);
-
- DWORD dwStatus = 0;
- {
- DMO_OUTPUT_DATA_BUFFER dmoBuffer = {0};
- dmoBuffer.pBuffer = _mediaBuffer;
- dmoBuffer.pBuffer->AddRef();
-
- // Poll the DMO for AEC processed capture data. The DMO will
- // copy available data to |dmoBuffer|, and should only return
- // 10 ms frames. The value of |dwStatus| should be ignored.
- hr = _dmo->ProcessOutput(0, 1, &dmoBuffer, &dwStatus);
- SAFE_RELEASE(dmoBuffer.pBuffer);
- dwStatus = dmoBuffer.dwStatus;
- }
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- keepRecording = false;
- assert(false);
- break;
- }
-
- ULONG bytesProduced = 0;
- BYTE* data;
- // Get a pointer to the data buffer. This should be valid until
- // the next call to ProcessOutput.
- hr = _mediaBuffer->GetBufferAndLength(&data, &bytesProduced);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- keepRecording = false;
- assert(false);
- break;
- }
-
- // TODO(andrew): handle AGC.
-
- if (bytesProduced > 0)
- {
- const int kSamplesProduced = bytesProduced / _recAudioFrameSize;
- // TODO(andrew): verify that this is always satisfied. It might
- // be that ProcessOutput will try to return more than 10 ms if
- // we fail to call it frequently enough.
- assert(kSamplesProduced == _recBlockSize);
- assert(sizeof(BYTE) == sizeof(WebRtc_Word8));
- _ptrAudioBuffer->SetRecordedBuffer(
- reinterpret_cast<WebRtc_Word8*>(data),
- kSamplesProduced);
- _ptrAudioBuffer->SetVQEData(0, 0, 0);
-
- _UnLock(); // Release lock while making the callback.
- _ptrAudioBuffer->DeliverRecordedData();
- _Lock();
- }
-
- // Reset length to indicate buffer availability.
- hr = _mediaBuffer->SetLength(0);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- keepRecording = false;
- assert(false);
- break;
- }
-
- if (!(dwStatus & DMO_OUTPUT_DATA_BUFFERF_INCOMPLETE))
- {
- // The DMO cannot currently produce more data. This is the
- // normal case; otherwise it means the DMO had more than 10 ms
- // of data available and ProcessOutput should be called again.
- break;
- }
- }
- }
- // ---------------------------- THREAD LOOP ---------------------------- <<
-
- RevertCaptureThreadPriority();
-
- if (FAILED(hr))
- {
- // Trigger callback from module process thread
- _recError = 1;
- WEBRTC_TRACE(kTraceError, kTraceUtility, _id,
- "kRecordingError message posted: capturing thread has ended "
- "prematurely");
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Capturing thread is now terminated properly");
- }
-
- return hr;
-}
-
-
-// ----------------------------------------------------------------------------
-// DoCaptureThread
-// ----------------------------------------------------------------------------
-
-DWORD AudioDeviceWindowsCore::DoCaptureThread()
-{
-
- bool keepRecording = true;
- HANDLE waitArray[2] = {_hShutdownCaptureEvent, _hCaptureSamplesReadyEvent};
- HRESULT hr = S_OK;
- HANDLE hMmTask = NULL;
-
- LARGE_INTEGER t1;
- LARGE_INTEGER t2;
- WebRtc_Word32 time(0);
-
- BYTE* syncBuffer = NULL;
- UINT32 syncBufIndex = 0;
-
- WebRtc_UWord32 newMicLevel(0);
- WebRtc_UWord32 currentMicLevel(0);
-
- _readSamples = 0;
-
- // Initialize COM as MTA in this thread.
- ScopedCOMInitializer comInit(ScopedCOMInitializer::kMTA);
- if (!comInit.succeeded()) {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "failed to initialize COM in capture thread");
- return -1;
- }
-
- hr = InitCaptureThreadPriority();
- if (FAILED(hr))
- {
- return hr;
- }
-
- _Lock();
-
- // Get size of capturing buffer (length is expressed as the number of audio frames the buffer can hold).
- // This value is fixed during the capturing session.
- //
- UINT32 bufferLength = 0;
- hr = _ptrClientIn->GetBufferSize(&bufferLength);
- EXIT_ON_ERROR(hr);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] size of buffer : %u", bufferLength);
-
- // Allocate memory for sync buffer.
- // It is used for compensation between native 44.1 and internal 44.0 and
- // for cases when the capture buffer is larger than 10ms.
- //
- const UINT32 syncBufferSize = 2*(bufferLength * _recAudioFrameSize);
- syncBuffer = new BYTE[syncBufferSize];
- if (syncBuffer == NULL)
- {
- return E_POINTER;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] size of sync buffer : %u [bytes]", syncBufferSize);
-
- // Get maximum latency for the current stream (will not change for the lifetime of the IAudioClient object).
- //
- REFERENCE_TIME latency;
- _ptrClientIn->GetStreamLatency(&latency);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] max stream latency : %u (%3.2f ms)",
- (DWORD)latency, (double)(latency / 10000.0));
-
- // Get the length of the periodic interval separating successive processing passes by
- // the audio engine on the data in the endpoint buffer.
- //
- REFERENCE_TIME devPeriod = 0;
- REFERENCE_TIME devPeriodMin = 0;
- _ptrClientIn->GetDevicePeriod(&devPeriod, &devPeriodMin);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] device period : %u (%3.2f ms)",
- (DWORD)devPeriod, (double)(devPeriod / 10000.0));
-
- double extraDelayMS = (double)((latency + devPeriod) / 10000.0);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] extraDelayMS : %3.2f", extraDelayMS);
-
- double endpointBufferSizeMS = 10.0 * ((double)bufferLength / (double)_recBlockSize);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[CAPT] endpointBufferSizeMS : %3.2f", endpointBufferSizeMS);
-
- // Start up the capturing stream.
- //
- hr = _ptrClientIn->Start();
- EXIT_ON_ERROR(hr);
-
- _UnLock();
-
- // Set event which will ensure that the calling thread modifies the recording state to true.
- //
- SetEvent(_hCaptureStartedEvent);
-
- // >> ---------------------------- THREAD LOOP ----------------------------
-
- while (keepRecording)
- {
- // Wait for a capture notification event or a shutdown event
- DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, 500);
- switch (waitResult)
- {
- case WAIT_OBJECT_0 + 0: // _hShutdownCaptureEvent
- keepRecording = false;
- break;
- case WAIT_OBJECT_0 + 1: // _hCaptureSamplesReadyEvent
- break;
- case WAIT_TIMEOUT: // timeout notification
- _ptrClientIn->Stop();
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "capture event timed out after 0.5 seconds");
- goto Exit;
- default: // unexpected error
- _ptrClientIn->Stop();
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "unknown wait termination on capture side");
- goto Exit;
- }
-
- while (keepRecording)
- {
- BYTE *pData = 0;
- UINT32 framesAvailable = 0;
- DWORD flags = 0;
- UINT64 recTime = 0;
- UINT64 recPos = 0;
-
- _Lock();
-
- // Find out how much capture data is available
- //
- hr = _ptrCaptureClient->GetBuffer(&pData, // packet which is ready to be read by used
- &framesAvailable, // #frames in the captured packet (can be zero)
- &flags, // support flags (check)
- &recPos, // device position of first audio frame in data packet
- &recTime); // value of performance counter at the time of recording the first audio frame
-
- if (SUCCEEDED(hr))
- {
- if (AUDCLNT_S_BUFFER_EMPTY == hr)
- {
- // Buffer was empty => start waiting for a new capture notification event
- _UnLock();
- break;
- }
-
- if (flags & AUDCLNT_BUFFERFLAGS_SILENT)
- {
- // Treat all of the data in the packet as silence and ignore the actual data values.
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "AUDCLNT_BUFFERFLAGS_SILENT");
- pData = NULL;
- }
-
- assert(framesAvailable != 0);
-
- if (pData)
- {
- CopyMemory(&syncBuffer[syncBufIndex*_recAudioFrameSize], pData, framesAvailable*_recAudioFrameSize);
- }
- else
- {
- ZeroMemory(&syncBuffer[syncBufIndex*_recAudioFrameSize], framesAvailable*_recAudioFrameSize);
- }
- assert(syncBufferSize >= (syncBufIndex*_recAudioFrameSize)+framesAvailable*_recAudioFrameSize);
-
- // Release the capture buffer
- //
- hr = _ptrCaptureClient->ReleaseBuffer(framesAvailable);
- EXIT_ON_ERROR(hr);
-
- _readSamples += framesAvailable;
- syncBufIndex += framesAvailable;
-
- QueryPerformanceCounter(&t1);
-
- // Get the current recording and playout delay.
- WebRtc_UWord32 sndCardRecDelay = (WebRtc_UWord32)
- (((((UINT64)t1.QuadPart * _perfCounterFactor) - recTime)
- / 10000) + (10*syncBufIndex) / _recBlockSize - 10);
- WebRtc_UWord32 sndCardPlayDelay =
- static_cast<WebRtc_UWord32>(_sndCardPlayDelay.Value());
-
- _sndCardRecDelay = sndCardRecDelay;
-
- while (syncBufIndex >= _recBlockSize)
- {
- if (_ptrAudioBuffer)
- {
- _ptrAudioBuffer->SetRecordedBuffer((const WebRtc_Word8*)syncBuffer, _recBlockSize);
-
- _driftAccumulator += _sampleDriftAt48kHz;
- const WebRtc_Word32 clockDrift =
- static_cast<WebRtc_Word32>(_driftAccumulator);
- _driftAccumulator -= clockDrift;
-
- _ptrAudioBuffer->SetVQEData(sndCardPlayDelay,
- sndCardRecDelay,
- clockDrift);
-
- QueryPerformanceCounter(&t1); // measure time: START
-
- _UnLock(); // release lock while making the callback
- _ptrAudioBuffer->DeliverRecordedData();
- _Lock(); // restore the lock
-
- QueryPerformanceCounter(&t2); // measure time: STOP
-
- // Measure "average CPU load".
- // Basically what we do here is to measure how many percent of our 10ms period
- // is used for encoding and decoding. This value shuld be used as a warning indicator
- // only and not seen as an absolute value. Running at ~100% will lead to bad QoS.
- time = (int)(t2.QuadPart - t1.QuadPart);
- _avgCPULoad = (float)(_avgCPULoad*.99 + (time + _playAcc) / (double)(_perfCounterFreq.QuadPart));
- _playAcc = 0;
-
- // Sanity check to ensure that essential states are not modified during the unlocked period
- if (_ptrCaptureClient == NULL || _ptrClientIn == NULL)
- {
- _UnLock();
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id, "input state has been modified during unlocked period");
- goto Exit;
- }
- }
-
- // store remaining data which was not able to deliver as 10ms segment
- MoveMemory(&syncBuffer[0], &syncBuffer[_recBlockSize*_recAudioFrameSize], (syncBufIndex-_recBlockSize)*_recAudioFrameSize);
- syncBufIndex -= _recBlockSize;
- sndCardRecDelay -= 10;
- }
-
- if (_AGC)
- {
- WebRtc_UWord32 newMicLevel = _ptrAudioBuffer->NewMicLevel();
- if (newMicLevel != 0)
- {
- // The VQE will only deliver non-zero microphone levels when a change is needed.
- // Set this new mic level (received from the observer as return value in the callback).
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "AGC change of volume: new=%u", newMicLevel);
- // We store this outside of the audio buffer to avoid
- // having it overwritten by the getter thread.
- _newMicLevel = newMicLevel;
- SetEvent(_hSetCaptureVolumeEvent);
- }
- }
- }
- else
- {
- // If GetBuffer returns AUDCLNT_E_BUFFER_ERROR, the thread consuming the audio samples
- // must wait for the next processing pass. The client might benefit from keeping a count
- // of the failed GetBuffer calls. If GetBuffer returns this error repeatedly, the client
- // can start a new processing loop after shutting down the current client by calling
- // IAudioClient::Stop, IAudioClient::Reset, and releasing the audio client.
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "IAudioCaptureClient::GetBuffer returned AUDCLNT_E_BUFFER_ERROR, hr = 0x%08X", hr);
- goto Exit;
- }
-
- _UnLock();
- }
- }
-
- // ---------------------------- THREAD LOOP ---------------------------- <<
-
- hr = _ptrClientIn->Stop();
-
-Exit:
- if (FAILED(hr))
- {
- _UnLock();
- _ptrClientIn->Stop();
- _TraceCOMError(hr);
- }
-
- RevertCaptureThreadPriority();
-
- if (keepRecording)
- {
- if (_ptrClientIn != NULL)
- {
- hr = _ptrClientIn->Stop();
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- }
- hr = _ptrClientIn->Reset();
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- }
- }
-
- // Trigger callback from module process thread
- _recError = 1;
- WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kRecordingError message posted: capturing thread has ended pre-maturely");
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_Capturing thread is now terminated properly");
- }
-
- SAFE_RELEASE(_ptrClientIn);
- SAFE_RELEASE(_ptrCaptureClient);
-
- if (syncBuffer)
- {
- delete [] syncBuffer;
- }
-
- return (DWORD)hr;
-}
-
-int32_t AudioDeviceWindowsCore::EnableBuiltInAEC(bool enable)
-{
-
- if (_recIsInitialized)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Attempt to set Windows AEC with recording already initialized");
- return -1;
- }
-
- if (_dmo == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Built-in AEC DMO was not initialized properly at create time");
- return -1;
- }
-
- _builtInAecEnabled = enable;
- return 0;
-}
-
-bool AudioDeviceWindowsCore::BuiltInAECIsEnabled() const
-{
- return _builtInAecEnabled;
-}
-
-int AudioDeviceWindowsCore::SetDMOProperties()
-{
- HRESULT hr = S_OK;
- assert(_dmo != NULL);
-
- scoped_refptr<IPropertyStore> ps;
- {
- IPropertyStore* ptrPS = NULL;
- hr = _dmo->QueryInterface(IID_IPropertyStore,
- reinterpret_cast<void**>(&ptrPS));
- if (FAILED(hr) || ptrPS == NULL)
- {
- _TraceCOMError(hr);
- return -1;
- }
- ps = ptrPS;
- SAFE_RELEASE(ptrPS);
- }
-
- // Set the AEC system mode.
- // SINGLE_CHANNEL_AEC - AEC processing only.
- if (SetVtI4Property(ps,
- MFPKEY_WMAAECMA_SYSTEM_MODE,
- SINGLE_CHANNEL_AEC))
- {
- return -1;
- }
-
- // Set the AEC source mode.
- // VARIANT_TRUE - Source mode (we poll the AEC for captured data).
- if (SetBoolProperty(ps,
- MFPKEY_WMAAECMA_DMO_SOURCE_MODE,
- VARIANT_TRUE) == -1)
- {
- return -1;
- }
-
- // Enable the feature mode.
- // This lets us override all the default processing settings below.
- if (SetBoolProperty(ps,
- MFPKEY_WMAAECMA_FEATURE_MODE,
- VARIANT_TRUE) == -1)
- {
- return -1;
- }
-
- // Disable analog AGC (default enabled).
- if (SetBoolProperty(ps,
- MFPKEY_WMAAECMA_MIC_GAIN_BOUNDER,
- VARIANT_FALSE) == -1)
- {
- return -1;
- }
-
- // Disable noise suppression (default enabled).
- // 0 - Disabled, 1 - Enabled
- if (SetVtI4Property(ps,
- MFPKEY_WMAAECMA_FEATR_NS,
- 0) == -1)
- {
- return -1;
- }
-
- // Relevant parameters to leave at default settings:
- // MFPKEY_WMAAECMA_FEATR_AGC - Digital AGC (disabled).
- // MFPKEY_WMAAECMA_FEATR_CENTER_CLIP - AEC center clipping (enabled).
- // MFPKEY_WMAAECMA_FEATR_ECHO_LENGTH - Filter length (256 ms).
- // TODO(andrew): investigate decresing the length to 128 ms.
- // MFPKEY_WMAAECMA_FEATR_FRAME_SIZE - Frame size (0).
- // 0 is automatic; defaults to 160 samples (or 10 ms frames at the
- // selected 16 kHz) as long as mic array processing is disabled.
- // MFPKEY_WMAAECMA_FEATR_NOISE_FILL - Comfort noise (enabled).
- // MFPKEY_WMAAECMA_FEATR_VAD - VAD (disabled).
-
- // Set the devices selected by VoE. If using a default device, we need to
- // search for the device index.
- int inDevIndex = _inputDeviceIndex;
- int outDevIndex = _outputDeviceIndex;
- if (!_usingInputDeviceIndex)
- {
- ERole role = eCommunications;
- if (_inputDevice == AudioDeviceModule::kDefaultDevice)
- {
- role = eConsole;
- }
-
- if (_GetDefaultDeviceIndex(eCapture, role, &inDevIndex) == -1)
- {
- return -1;
- }
- }
-
- if (!_usingOutputDeviceIndex)
- {
- ERole role = eCommunications;
- if (_outputDevice == AudioDeviceModule::kDefaultDevice)
- {
- role = eConsole;
- }
-
- if (_GetDefaultDeviceIndex(eRender, role, &outDevIndex) == -1)
- {
- return -1;
- }
- }
-
- DWORD devIndex = static_cast<uint32_t>(outDevIndex << 16) +
- static_cast<uint32_t>(0x0000ffff & inDevIndex);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "Capture device index: %d, render device index: %d",
- inDevIndex, outDevIndex);
- if (SetVtI4Property(ps,
- MFPKEY_WMAAECMA_DEVICE_INDEXES,
- devIndex) == -1)
- {
- return -1;
- }
-
- return 0;
-}
-
-int AudioDeviceWindowsCore::SetBoolProperty(IPropertyStore* ptrPS,
- REFPROPERTYKEY key,
- VARIANT_BOOL value)
-{
- PROPVARIANT pv;
- PropVariantInit(&pv);
- pv.vt = VT_BOOL;
- pv.boolVal = value;
- HRESULT hr = ptrPS->SetValue(key, pv);
- PropVariantClear(&pv);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- return -1;
- }
- return 0;
-}
-
-int AudioDeviceWindowsCore::SetVtI4Property(IPropertyStore* ptrPS,
- REFPROPERTYKEY key,
- LONG value)
-{
- PROPVARIANT pv;
- PropVariantInit(&pv);
- pv.vt = VT_I4;
- pv.lVal = value;
- HRESULT hr = ptrPS->SetValue(key, pv);
- PropVariantClear(&pv);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- return -1;
- }
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// _RefreshDeviceList
-//
-// Creates a new list of endpoint rendering or capture devices after
-// deleting any previously created (and possibly out-of-date) list of
-// such devices.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::_RefreshDeviceList(EDataFlow dir)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- HRESULT hr = S_OK;
- IMMDeviceCollection *pCollection = NULL;
-
- assert(dir == eRender || dir == eCapture);
- assert(_ptrEnumerator != NULL);
-
- // Create a fresh list of devices using the specified direction
- hr = _ptrEnumerator->EnumAudioEndpoints(
- dir,
- DEVICE_STATE_ACTIVE,
- &pCollection);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(pCollection);
- return -1;
- }
-
- if (dir == eRender)
- {
- SAFE_RELEASE(_ptrRenderCollection);
- _ptrRenderCollection = pCollection;
- }
- else
- {
- SAFE_RELEASE(_ptrCaptureCollection);
- _ptrCaptureCollection = pCollection;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// _DeviceListCount
-//
-// Gets a count of the endpoint rendering or capture devices in the
-// current list of such devices.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word16 AudioDeviceWindowsCore::_DeviceListCount(EDataFlow dir)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- HRESULT hr = S_OK;
- UINT count = 0;
-
- assert(eRender == dir || eCapture == dir);
-
- if (eRender == dir && NULL != _ptrRenderCollection)
- {
- hr = _ptrRenderCollection->GetCount(&count);
- }
- else if (NULL != _ptrCaptureCollection)
- {
- hr = _ptrCaptureCollection->GetCount(&count);
- }
-
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- return -1;
- }
-
- return static_cast<WebRtc_Word16> (count);
-}
-
-// ----------------------------------------------------------------------------
-// _GetListDeviceName
-//
-// Gets the friendly name of an endpoint rendering or capture device
-// from the current list of such devices. The caller uses an index
-// into the list to identify the device.
-//
-// Uses: _ptrRenderCollection or _ptrCaptureCollection which is updated
-// in _RefreshDeviceList().
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::_GetListDeviceName(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- HRESULT hr = S_OK;
- IMMDevice *pDevice = NULL;
-
- assert(dir == eRender || dir == eCapture);
-
- if (eRender == dir && NULL != _ptrRenderCollection)
- {
- hr = _ptrRenderCollection->Item(index, &pDevice);
- }
- else if (NULL != _ptrCaptureCollection)
- {
- hr = _ptrCaptureCollection->Item(index, &pDevice);
- }
-
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(pDevice);
- return -1;
- }
-
- WebRtc_Word32 res = _GetDeviceName(pDevice, szBuffer, bufferLen);
- SAFE_RELEASE(pDevice);
- return res;
-}
-
-// ----------------------------------------------------------------------------
-// _GetDefaultDeviceName
-//
-// Gets the friendly name of an endpoint rendering or capture device
-// given a specified device role.
-//
-// Uses: _ptrEnumerator
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::_GetDefaultDeviceName(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- HRESULT hr = S_OK;
- IMMDevice *pDevice = NULL;
-
- assert(dir == eRender || dir == eCapture);
- assert(role == eConsole || role == eCommunications);
- assert(_ptrEnumerator != NULL);
-
- hr = _ptrEnumerator->GetDefaultAudioEndpoint(
- dir,
- role,
- &pDevice);
-
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(pDevice);
- return -1;
- }
-
- WebRtc_Word32 res = _GetDeviceName(pDevice, szBuffer, bufferLen);
- SAFE_RELEASE(pDevice);
- return res;
-}
-
-// ----------------------------------------------------------------------------
-// _GetListDeviceID
-//
-// Gets the unique ID string of an endpoint rendering or capture device
-// from the current list of such devices. The caller uses an index
-// into the list to identify the device.
-//
-// Uses: _ptrRenderCollection or _ptrCaptureCollection which is updated
-// in _RefreshDeviceList().
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::_GetListDeviceID(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- HRESULT hr = S_OK;
- IMMDevice *pDevice = NULL;
-
- assert(dir == eRender || dir == eCapture);
-
- if (eRender == dir && NULL != _ptrRenderCollection)
- {
- hr = _ptrRenderCollection->Item(index, &pDevice);
- }
- else if (NULL != _ptrCaptureCollection)
- {
- hr = _ptrCaptureCollection->Item(index, &pDevice);
- }
-
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(pDevice);
- return -1;
- }
-
- WebRtc_Word32 res = _GetDeviceID(pDevice, szBuffer, bufferLen);
- SAFE_RELEASE(pDevice);
- return res;
-}
-
-// ----------------------------------------------------------------------------
-// _GetDefaultDeviceID
-//
-// Gets the uniqe device ID of an endpoint rendering or capture device
-// given a specified device role.
-//
-// Uses: _ptrEnumerator
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::_GetDefaultDeviceID(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- HRESULT hr = S_OK;
- IMMDevice *pDevice = NULL;
-
- assert(dir == eRender || dir == eCapture);
- assert(role == eConsole || role == eCommunications);
- assert(_ptrEnumerator != NULL);
-
- hr = _ptrEnumerator->GetDefaultAudioEndpoint(
- dir,
- role,
- &pDevice);
-
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(pDevice);
- return -1;
- }
-
- WebRtc_Word32 res = _GetDeviceID(pDevice, szBuffer, bufferLen);
- SAFE_RELEASE(pDevice);
- return res;
-}
-
-WebRtc_Word32 AudioDeviceWindowsCore::_GetDefaultDeviceIndex(EDataFlow dir,
- ERole role,
- int* index)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- HRESULT hr = S_OK;
- WCHAR szDefaultDeviceID[MAX_PATH] = {0};
- WCHAR szDeviceID[MAX_PATH] = {0};
-
- const size_t kDeviceIDLength = sizeof(szDeviceID)/sizeof(szDeviceID[0]);
- assert(kDeviceIDLength ==
- sizeof(szDefaultDeviceID) / sizeof(szDefaultDeviceID[0]));
-
- if (_GetDefaultDeviceID(dir,
- role,
- szDefaultDeviceID,
- kDeviceIDLength) == -1)
- {
- return -1;
- }
-
- IMMDeviceCollection* collection = _ptrCaptureCollection;
- if (dir == eRender)
- {
- collection = _ptrRenderCollection;
- }
-
- if (!collection)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Device collection not valid");
- return -1;
- }
-
- UINT count = 0;
- hr = collection->GetCount(&count);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- return -1;
- }
-
- *index = -1;
- for (UINT i = 0; i < count; i++)
- {
- memset(szDeviceID, 0, sizeof(szDeviceID));
- scoped_refptr<IMMDevice> device;
- {
- IMMDevice* ptrDevice = NULL;
- hr = collection->Item(i, &ptrDevice);
- if (FAILED(hr) || ptrDevice == NULL)
- {
- _TraceCOMError(hr);
- return -1;
- }
- device = ptrDevice;
- SAFE_RELEASE(ptrDevice);
- }
-
- if (_GetDeviceID(device, szDeviceID, kDeviceIDLength) == -1)
- {
- return -1;
- }
-
- if (wcsncmp(szDefaultDeviceID, szDeviceID, kDeviceIDLength) == 0)
- {
- // Found a match.
- *index = i;
- break;
- }
-
- }
-
- if (*index == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Unable to find collection index for default device");
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// _GetDeviceName
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::_GetDeviceName(IMMDevice* pDevice,
- LPWSTR pszBuffer,
- int bufferLen)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- static const WCHAR szDefault[] = L"<Device not available>";
-
- HRESULT hr = E_FAIL;
- IPropertyStore *pProps = NULL;
- PROPVARIANT varName;
-
- assert(pszBuffer != NULL);
- assert(bufferLen > 0);
-
- if (pDevice != NULL)
- {
- hr = pDevice->OpenPropertyStore(STGM_READ, &pProps);
- if (FAILED(hr))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "IMMDevice::OpenPropertyStore failed, hr = 0x%08X", hr);
- }
- }
-
- // Initialize container for property value.
- PropVariantInit(&varName);
-
- if (SUCCEEDED(hr))
- {
- // Get the endpoint device's friendly-name property.
- hr = pProps->GetValue(PKEY_Device_FriendlyName, &varName);
- if (FAILED(hr))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "IPropertyStore::GetValue failed, hr = 0x%08X", hr);
- }
- }
-
- if ((SUCCEEDED(hr)) && (VT_EMPTY == varName.vt))
- {
- hr = E_FAIL;
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "IPropertyStore::GetValue returned no value, hr = 0x%08X", hr);
- }
-
- if ((SUCCEEDED(hr)) && (VT_LPWSTR != varName.vt))
- {
- // The returned value is not a wide null terminated string.
- hr = E_UNEXPECTED;
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "IPropertyStore::GetValue returned unexpected type, hr = 0x%08X", hr);
- }
-
- if (SUCCEEDED(hr) && (varName.pwszVal != NULL))
- {
- // Copy the valid device name to the provided ouput buffer.
- wcsncpy_s(pszBuffer, bufferLen, varName.pwszVal, _TRUNCATE);
- }
- else
- {
- // Failed to find the device name.
- wcsncpy_s(pszBuffer, bufferLen, szDefault, _TRUNCATE);
- }
-
- PropVariantClear(&varName);
- SAFE_RELEASE(pProps);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// _GetDeviceID
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::_GetDeviceID(IMMDevice* pDevice, LPWSTR pszBuffer, int bufferLen)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- static const WCHAR szDefault[] = L"<Device not available>";
-
- HRESULT hr = E_FAIL;
- LPWSTR pwszID = NULL;
-
- assert(pszBuffer != NULL);
- assert(bufferLen > 0);
-
- if (pDevice != NULL)
- {
- hr = pDevice->GetId(&pwszID);
- }
-
- if (hr == S_OK)
- {
- // Found the device ID.
- wcsncpy_s(pszBuffer, bufferLen, pwszID, _TRUNCATE);
- }
- else
- {
- // Failed to find the device ID.
- wcsncpy_s(pszBuffer, bufferLen, szDefault, _TRUNCATE);
- }
-
- CoTaskMemFree(pwszID);
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// _GetDefaultDevice
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::_GetDefaultDevice(EDataFlow dir, ERole role, IMMDevice** ppDevice)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- HRESULT hr(S_OK);
-
- assert(_ptrEnumerator != NULL);
-
- hr = _ptrEnumerator->GetDefaultAudioEndpoint(
- dir,
- role,
- ppDevice);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// _GetListDevice
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::_GetListDevice(EDataFlow dir, int index, IMMDevice** ppDevice)
-{
- HRESULT hr(S_OK);
-
- assert(_ptrEnumerator != NULL);
-
- IMMDeviceCollection *pCollection = NULL;
-
- hr = _ptrEnumerator->EnumAudioEndpoints(
- dir,
- DEVICE_STATE_ACTIVE, // only active endpoints are OK
- &pCollection);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(pCollection);
- return -1;
- }
-
- hr = pCollection->Item(
- index,
- ppDevice);
- if (FAILED(hr))
- {
- _TraceCOMError(hr);
- SAFE_RELEASE(pCollection);
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// _EnumerateEndpointDevicesAll
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsCore::_EnumerateEndpointDevicesAll(EDataFlow dataFlow) const
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- assert(_ptrEnumerator != NULL);
-
- HRESULT hr = S_OK;
- IMMDeviceCollection *pCollection = NULL;
-
- // Generate a collection of audio endpoint devices in the system.
- // Get states for *all* endpoint devices.
- // Output: IMMDeviceCollection interface.
- hr = _ptrEnumerator->EnumAudioEndpoints(
- dataFlow, // data-flow direction (input parameter)
- DEVICE_STATE_ACTIVE | DEVICE_STATE_DISABLED | DEVICE_STATE_NOTPRESENT | DEVICE_STATE_UNPLUGGED,
- &pCollection); // release interface when done
-
- EXIT_ON_ERROR(hr);
-
- // use the IMMDeviceCollection interface...
-
- UINT count;
- IMMDevice *pEndpoint = NULL;
- IPropertyStore *pProps = NULL;
- IAudioEndpointVolume* pEndpointVolume = NULL;
- LPWSTR pwszID = NULL;
-
- // Retrieve a count of the devices in the device collection.
- hr = pCollection->GetCount(&count);
- EXIT_ON_ERROR(hr);
- if (dataFlow == eRender)
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#rendering endpoint devices (counting all): %u", count);
- else if (dataFlow == eCapture)
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#capturing endpoint devices (counting all): %u", count);
-
- if (count == 0)
- {
- return 0;
- }
-
- // Each loop prints the name of an endpoint device.
- for (ULONG i = 0; i < count; i++)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Endpoint %d:", i);
-
- // Get pointer to endpoint number i.
- // Output: IMMDevice interface.
- hr = pCollection->Item(
- i,
- &pEndpoint);
- EXIT_ON_ERROR(hr);
-
- // use the IMMDevice interface of the specified endpoint device...
-
- // Get the endpoint ID string (uniquely identifies the device among all audio endpoint devices)
- hr = pEndpoint->GetId(&pwszID);
- EXIT_ON_ERROR(hr);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "ID string : %S", pwszID);
-
- // Retrieve an interface to the device's property store.
- // Output: IPropertyStore interface.
- hr = pEndpoint->OpenPropertyStore(
- STGM_READ,
- &pProps);
- EXIT_ON_ERROR(hr);
-
- // use the IPropertyStore interface...
-
- PROPVARIANT varName;
- // Initialize container for property value.
- PropVariantInit(&varName);
-
- // Get the endpoint's friendly-name property.
- // Example: "Speakers (Realtek High Definition Audio)"
- hr = pProps->GetValue(
- PKEY_Device_FriendlyName,
- &varName);
- EXIT_ON_ERROR(hr);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "friendly name: \"%S\"", varName.pwszVal);
-
- // Get the endpoint's current device state
- DWORD dwState;
- hr = pEndpoint->GetState(&dwState);
- EXIT_ON_ERROR(hr);
- if (dwState & DEVICE_STATE_ACTIVE)
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "state (0x%x) : *ACTIVE*", dwState);
- if (dwState & DEVICE_STATE_DISABLED)
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "state (0x%x) : DISABLED", dwState);
- if (dwState & DEVICE_STATE_NOTPRESENT)
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "state (0x%x) : NOTPRESENT", dwState);
- if (dwState & DEVICE_STATE_UNPLUGGED)
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "state (0x%x) : UNPLUGGED", dwState);
-
- // Check the hardware volume capabilities.
- DWORD dwHwSupportMask = 0;
- hr = pEndpoint->Activate(__uuidof(IAudioEndpointVolume), CLSCTX_ALL,
- NULL, (void**)&pEndpointVolume);
- EXIT_ON_ERROR(hr);
- hr = pEndpointVolume->QueryHardwareSupport(&dwHwSupportMask);
- EXIT_ON_ERROR(hr);
- if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_VOLUME)
- // The audio endpoint device supports a hardware volume control
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "hwmask (0x%x) : HARDWARE_SUPPORT_VOLUME", dwHwSupportMask);
- if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_MUTE)
- // The audio endpoint device supports a hardware mute control
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "hwmask (0x%x) : HARDWARE_SUPPORT_MUTE", dwHwSupportMask);
- if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_METER)
- // The audio endpoint device supports a hardware peak meter
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "hwmask (0x%x) : HARDWARE_SUPPORT_METER", dwHwSupportMask);
-
- // Check the channel count (#channels in the audio stream that enters or leaves the audio endpoint device)
- UINT nChannelCount(0);
- hr = pEndpointVolume->GetChannelCount(
- &nChannelCount);
- EXIT_ON_ERROR(hr);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#channels : %u", nChannelCount);
-
- if (dwHwSupportMask & ENDPOINT_HARDWARE_SUPPORT_VOLUME)
- {
- // Get the volume range.
- float fLevelMinDB(0.0);
- float fLevelMaxDB(0.0);
- float fVolumeIncrementDB(0.0);
- hr = pEndpointVolume->GetVolumeRange(
- &fLevelMinDB,
- &fLevelMaxDB,
- &fVolumeIncrementDB);
- EXIT_ON_ERROR(hr);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "volume range : %4.2f (min), %4.2f (max), %4.2f (inc) [dB]",
- fLevelMinDB, fLevelMaxDB, fVolumeIncrementDB);
-
- // The volume range from vmin = fLevelMinDB to vmax = fLevelMaxDB is divided
- // into n uniform intervals of size vinc = fVolumeIncrementDB, where
- // n = (vmax ?vmin) / vinc.
- // The values vmin, vmax, and vinc are measured in decibels. The client can set
- // the volume level to one of n + 1 discrete values in the range from vmin to vmax.
- int n = (int)((fLevelMaxDB-fLevelMinDB)/fVolumeIncrementDB);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#intervals : %d", n);
-
- // Get information about the current step in the volume range.
- // This method represents the volume level of the audio stream that enters or leaves
- // the audio endpoint device as an index or "step" in a range of discrete volume levels.
- // Output value nStepCount is the number of steps in the range. Output value nStep
- // is the step index of the current volume level. If the number of steps is n = nStepCount,
- // then step index nStep can assume values from 0 (minimum volume) to n ?1 (maximum volume).
- UINT nStep(0);
- UINT nStepCount(0);
- hr = pEndpointVolume->GetVolumeStepInfo(
- &nStep,
- &nStepCount);
- EXIT_ON_ERROR(hr);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "volume steps : %d (nStep), %d (nStepCount)", nStep, nStepCount);
- }
-
- CoTaskMemFree(pwszID);
- pwszID = NULL;
- PropVariantClear(&varName);
- SAFE_RELEASE(pProps);
- SAFE_RELEASE(pEndpoint);
- SAFE_RELEASE(pEndpointVolume);
- }
- SAFE_RELEASE(pCollection);
- return 0;
-
-Exit:
- _TraceCOMError(hr);
- CoTaskMemFree(pwszID);
- pwszID = NULL;
- SAFE_RELEASE(pCollection);
- SAFE_RELEASE(pEndpoint);
- SAFE_RELEASE(pEndpointVolume);
- SAFE_RELEASE(pProps);
- return -1;
-}
-
-// ----------------------------------------------------------------------------
-// _TraceCOMError
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsCore::_TraceCOMError(HRESULT hr) const
-{
- TCHAR buf[MAXERRORLENGTH];
- TCHAR errorText[MAXERRORLENGTH];
-
- const DWORD dwFlags = FORMAT_MESSAGE_FROM_SYSTEM |
- FORMAT_MESSAGE_IGNORE_INSERTS;
- const DWORD dwLangID = MAKELANGID(LANG_ENGLISH, SUBLANG_ENGLISH_US);
-
- // Gets the system's human readable message string for this HRESULT.
- // All error message in English by default.
- DWORD messageLength = ::FormatMessageW(dwFlags,
- 0,
- hr,
- dwLangID,
- errorText,
- MAXERRORLENGTH,
- NULL);
-
- assert(messageLength <= MAXERRORLENGTH);
-
- // Trims tailing white space (FormatMessage() leaves a trailing cr-lf.).
- for (; messageLength && ::isspace(errorText[messageLength - 1]);
- --messageLength)
- {
- errorText[messageLength - 1] = '\0';
- }
-
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- "Core Audio method failed (hr=0x%x)", hr);
- StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
- StringCchCat(buf, MAXERRORLENGTH, errorText);
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
-}
-
-// ----------------------------------------------------------------------------
-// _SetThreadName
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsCore::_SetThreadName(DWORD dwThreadID, LPCSTR szThreadName)
-{
- // See http://msdn.microsoft.com/en-us/library/xcb2z8hs(VS.71).aspx for details on the code
- // in this function. Name of article is "Setting a Thread Name (Unmanaged)".
-
- THREADNAME_INFO info;
- info.dwType = 0x1000;
- info.szName = szThreadName;
- info.dwThreadID = dwThreadID;
- info.dwFlags = 0;
-
- __try
- {
- RaiseException( 0x406D1388, 0, sizeof(info)/sizeof(DWORD), (ULONG_PTR *)&info );
- }
- __except (EXCEPTION_CONTINUE_EXECUTION)
- {
- }
-}
-
-// ----------------------------------------------------------------------------
-// _Get44kHzDrift
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsCore::_Get44kHzDrift()
-{
- // We aren't able to resample at 44.1 kHz. Instead we run at 44 kHz and push/pull
- // from the engine faster to compensate. If only one direction is set to 44.1 kHz
- // the result is indistinguishable from clock drift to the AEC. We can compensate
- // internally if we inform the AEC about the drift.
- _sampleDriftAt48kHz = 0;
- _driftAccumulator = 0;
-
- if (_playSampleRate == 44000 && _recSampleRate != 44000)
- {
- _sampleDriftAt48kHz = 480.0f/440;
- }
- else if(_playSampleRate != 44000 && _recSampleRate == 44000)
- {
- _sampleDriftAt48kHz = -480.0f/441;
- }
-}
-
-// ----------------------------------------------------------------------------
-// WideToUTF8
-// ----------------------------------------------------------------------------
-
-char* AudioDeviceWindowsCore::WideToUTF8(const TCHAR* src) const {
-#ifdef UNICODE
- const size_t kStrLen = sizeof(_str);
- memset(_str, 0, kStrLen);
- // Get required size (in bytes) to be able to complete the conversion.
- int required_size = WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, 0, 0, 0);
- if (required_size <= kStrLen)
- {
- // Process the entire input string, including the terminating null char.
- if (WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, kStrLen, 0, 0) == 0)
- memset(_str, 0, kStrLen);
- }
- return _str;
-#else
- return const_cast<char*>(src);
-#endif
-}
-
-} // namespace webrtc
-
-#endif // WEBRTC_WINDOWS_CORE_AUDIO_BUILD
diff --git a/trunk/src/modules/audio_device/main/source/win/audio_device_core_win.h b/trunk/src/modules/audio_device/main/source/win/audio_device_core_win.h
deleted file mode 100644
index da37a8c..0000000
--- a/trunk/src/modules/audio_device/main/source/win/audio_device_core_win.h
+++ /dev/null
@@ -1,381 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CORE_WIN_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CORE_WIN_H
-
-#if (_MSC_VER >= 1400) // only include for VS 2005 and higher
-
-#include "audio_device_generic.h"
-
-#pragma once
-#include <wmcodecdsp.h> // CLSID_CWMAudioAEC
- // (must be before audioclient.h)
-#include <Audioclient.h> // WASAPI
-#include <Audiopolicy.h>
-#include <avrt.h> // Avrt
-#include <endpointvolume.h>
-#include <mediaobj.h> // IMediaObject
-#include <Mmdeviceapi.h> // MMDevice
-
-#include "atomic32_wrapper.h"
-#include "critical_section_wrapper.h"
-#include "scoped_refptr.h"
-
-// Use Multimedia Class Scheduler Service (MMCSS) to boost the thread priority
-#pragma comment( lib, "avrt.lib" )
-// AVRT function pointers
-typedef BOOL (WINAPI *PAvRevertMmThreadCharacteristics)(HANDLE);
-typedef HANDLE (WINAPI *PAvSetMmThreadCharacteristicsA)(LPCSTR, LPDWORD);
-typedef BOOL (WINAPI *PAvSetMmThreadPriority)(HANDLE, AVRT_PRIORITY);
-
-namespace webrtc {
-
-const float MAX_CORE_SPEAKER_VOLUME = 255.0f;
-const float MIN_CORE_SPEAKER_VOLUME = 0.0f;
-const float MAX_CORE_MICROPHONE_VOLUME = 255.0f;
-const float MIN_CORE_MICROPHONE_VOLUME = 0.0f;
-const WebRtc_UWord16 CORE_SPEAKER_VOLUME_STEP_SIZE = 1;
-const WebRtc_UWord16 CORE_MICROPHONE_VOLUME_STEP_SIZE = 1;
-
-// Utility class which initializes COM in the constructor (STA or MTA),
-// and uninitializes COM in the destructor.
-class ScopedCOMInitializer {
- public:
- // Enum value provided to initialize the thread as an MTA instead of STA.
- enum SelectMTA { kMTA };
-
- // Constructor for STA initialization.
- ScopedCOMInitializer() {
- Initialize(COINIT_APARTMENTTHREADED);
- }
-
- // Constructor for MTA initialization.
- explicit ScopedCOMInitializer(SelectMTA mta) {
- Initialize(COINIT_MULTITHREADED);
- }
-
- ScopedCOMInitializer::~ScopedCOMInitializer() {
- if (SUCCEEDED(hr_))
- CoUninitialize();
- }
-
- bool succeeded() const { return SUCCEEDED(hr_); }
-
- private:
- void Initialize(COINIT init) {
- hr_ = CoInitializeEx(NULL, init);
- }
-
- HRESULT hr_;
-
- ScopedCOMInitializer(const ScopedCOMInitializer&);
- void operator=(const ScopedCOMInitializer&);
-};
-
-
-class AudioDeviceWindowsCore : public AudioDeviceGeneric
-{
-public:
- AudioDeviceWindowsCore(const WebRtc_Word32 id);
- ~AudioDeviceWindowsCore();
-
- static bool CoreAudioIsSupported();
-
- // Retrieve the currently utilized audio layer
- virtual WebRtc_Word32 ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
-
- // Main initializaton and termination
- virtual WebRtc_Word32 Init();
- virtual WebRtc_Word32 Terminate();
- virtual bool Initialized() const;
-
- // Device enumeration
- virtual WebRtc_Word16 PlayoutDevices();
- virtual WebRtc_Word16 RecordingDevices();
- virtual WebRtc_Word32 PlayoutDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize]);
- virtual WebRtc_Word32 RecordingDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize]);
-
- // Device selection
- virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32 SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
- virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32 SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device);
-
- // Audio transport initialization
- virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
- virtual WebRtc_Word32 InitPlayout();
- virtual bool PlayoutIsInitialized() const;
- virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
- virtual WebRtc_Word32 InitRecording();
- virtual bool RecordingIsInitialized() const;
-
- // Audio transport control
- virtual WebRtc_Word32 StartPlayout();
- virtual WebRtc_Word32 StopPlayout();
- virtual bool Playing() const;
- virtual WebRtc_Word32 StartRecording();
- virtual WebRtc_Word32 StopRecording();
- virtual bool Recording() const;
-
- // Microphone Automatic Gain Control (AGC)
- virtual WebRtc_Word32 SetAGC(bool enable);
- virtual bool AGC() const;
-
- // Volume control based on the Windows Wave API (Windows only)
- virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft, WebRtc_UWord16 volumeRight);
- virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft, WebRtc_UWord16& volumeRight) const;
-
- // Audio mixer initialization
- virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
- virtual WebRtc_Word32 InitSpeaker();
- virtual bool SpeakerIsInitialized() const;
- virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
- virtual WebRtc_Word32 InitMicrophone();
- virtual bool MicrophoneIsInitialized() const;
-
- // Speaker volume controls
- virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
- virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
- virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
- virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
- virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
-
- // Microphone volume controls
- virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
- virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
- virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
- virtual WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
-
- // Speaker mute control
- virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
- virtual WebRtc_Word32 SetSpeakerMute(bool enable);
- virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
-
- // Microphone mute control
- virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
- virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
-
- // Microphone boost control
- virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
- virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
-
- // Stereo support
- virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
- virtual WebRtc_Word32 SetStereoPlayout(bool enable);
- virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
- virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
- virtual WebRtc_Word32 SetStereoRecording(bool enable);
- virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
-
- // Delay information and control
- virtual WebRtc_Word32 SetPlayoutBuffer(const AudioDeviceModule::BufferType type, WebRtc_UWord16 sizeMS);
- virtual WebRtc_Word32 PlayoutBuffer(AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const;
- virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
- virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
-
- // CPU load
- virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
-
- virtual int32_t EnableBuiltInAEC(bool enable);
- virtual bool BuiltInAECIsEnabled() const;
-
-public:
- virtual bool PlayoutWarning() const;
- virtual bool PlayoutError() const;
- virtual bool RecordingWarning() const;
- virtual bool RecordingError() const;
- virtual void ClearPlayoutWarning();
- virtual void ClearPlayoutError();
- virtual void ClearRecordingWarning();
- virtual void ClearRecordingError();
-
-public:
- virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
-private: // avrt function pointers
- PAvRevertMmThreadCharacteristics _PAvRevertMmThreadCharacteristics;
- PAvSetMmThreadCharacteristicsA _PAvSetMmThreadCharacteristicsA;
- PAvSetMmThreadPriority _PAvSetMmThreadPriority;
- HMODULE _avrtLibrary;
- bool _winSupportAvrt;
-
-private: // thread functions
- DWORD InitCaptureThreadPriority();
- void RevertCaptureThreadPriority();
- static DWORD WINAPI WSAPICaptureThread(LPVOID context);
- DWORD DoCaptureThread();
-
- static DWORD WINAPI WSAPICaptureThreadPollDMO(LPVOID context);
- DWORD DoCaptureThreadPollDMO();
-
- static DWORD WINAPI WSAPIRenderThread(LPVOID context);
- DWORD DoRenderThread();
-
- static DWORD WINAPI GetCaptureVolumeThread(LPVOID context);
- DWORD DoGetCaptureVolumeThread();
-
- static DWORD WINAPI SetCaptureVolumeThread(LPVOID context);
- DWORD DoSetCaptureVolumeThread();
-
- void _SetThreadName(DWORD dwThreadID, LPCSTR szThreadName);
- void _Lock() { _critSect.Enter(); };
- void _UnLock() { _critSect.Leave(); };
-
-private:
- WebRtc_Word32 Id() {return _id;}
-
-private:
- int SetDMOProperties();
-
- int SetBoolProperty(IPropertyStore* ptrPS,
- REFPROPERTYKEY key,
- VARIANT_BOOL value);
-
- int SetVtI4Property(IPropertyStore* ptrPS,
- REFPROPERTYKEY key,
- LONG value);
-
- WebRtc_Word32 _EnumerateEndpointDevicesAll(EDataFlow dataFlow) const;
- void _TraceCOMError(HRESULT hr) const;
-
- WebRtc_Word32 _RefreshDeviceList(EDataFlow dir);
- WebRtc_Word16 _DeviceListCount(EDataFlow dir);
- WebRtc_Word32 _GetDefaultDeviceName(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen);
- WebRtc_Word32 _GetListDeviceName(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen);
- WebRtc_Word32 _GetDeviceName(IMMDevice* pDevice, LPWSTR pszBuffer, int bufferLen);
- WebRtc_Word32 _GetListDeviceID(EDataFlow dir, int index, LPWSTR szBuffer, int bufferLen);
- WebRtc_Word32 _GetDefaultDeviceID(EDataFlow dir, ERole role, LPWSTR szBuffer, int bufferLen);
- WebRtc_Word32 _GetDefaultDeviceIndex(EDataFlow dir, ERole role, int* index);
- WebRtc_Word32 _GetDeviceID(IMMDevice* pDevice, LPWSTR pszBuffer, int bufferLen);
- WebRtc_Word32 _GetDefaultDevice(EDataFlow dir, ERole role, IMMDevice** ppDevice);
- WebRtc_Word32 _GetListDevice(EDataFlow dir, int index, IMMDevice** ppDevice);
-
- void _Get44kHzDrift();
-
- // Converts from wide-char to UTF-8 if UNICODE is defined.
- // Does nothing if UNICODE is undefined.
- char* WideToUTF8(const TCHAR* src) const;
-
- WebRtc_Word32 InitRecordingDMO();
-
-private:
- ScopedCOMInitializer _comInit;
- AudioDeviceBuffer* _ptrAudioBuffer;
- CriticalSectionWrapper& _critSect;
- CriticalSectionWrapper& _volumeMutex;
- WebRtc_Word32 _id;
-
-private: // MMDevice
- IMMDeviceEnumerator* _ptrEnumerator;
- IMMDeviceCollection* _ptrRenderCollection;
- IMMDeviceCollection* _ptrCaptureCollection;
- IMMDevice* _ptrDeviceOut;
- IMMDevice* _ptrDeviceIn;
-
-private: // WASAPI
- IAudioClient* _ptrClientOut;
- IAudioClient* _ptrClientIn;
- IAudioRenderClient* _ptrRenderClient;
- IAudioCaptureClient* _ptrCaptureClient;
- IAudioEndpointVolume* _ptrCaptureVolume;
- ISimpleAudioVolume* _ptrRenderSimpleVolume;
-
- // DirectX Media Object (DMO) for the built-in AEC.
- scoped_refptr<IMediaObject> _dmo;
- scoped_refptr<IMediaBuffer> _mediaBuffer;
- bool _builtInAecEnabled;
-
- HANDLE _hRenderSamplesReadyEvent;
- HANDLE _hPlayThread;
- HANDLE _hRenderStartedEvent;
- HANDLE _hShutdownRenderEvent;
-
- HANDLE _hCaptureSamplesReadyEvent;
- HANDLE _hRecThread;
- HANDLE _hCaptureStartedEvent;
- HANDLE _hShutdownCaptureEvent;
-
- HANDLE _hGetCaptureVolumeThread;
- HANDLE _hSetCaptureVolumeThread;
- HANDLE _hSetCaptureVolumeEvent;
-
- HANDLE _hMmTask;
-
- UINT _playAudioFrameSize;
- WebRtc_UWord32 _playSampleRate;
- WebRtc_UWord32 _devicePlaySampleRate;
- WebRtc_UWord32 _playBlockSize;
- WebRtc_UWord32 _devicePlayBlockSize;
- WebRtc_UWord32 _playChannels;
- Atomic32Wrapper _sndCardPlayDelay;
- UINT64 _writtenSamples;
- LONGLONG _playAcc;
-
- UINT _recAudioFrameSize;
- WebRtc_UWord32 _recSampleRate;
- WebRtc_UWord32 _recBlockSize;
- WebRtc_UWord32 _recChannels;
- UINT64 _readSamples;
- Atomic32Wrapper _sndCardRecDelay;
-
- float _sampleDriftAt48kHz;
- float _driftAccumulator;
-
- WebRtc_UWord16 _recChannelsPrioList[2];
- WebRtc_UWord16 _playChannelsPrioList[2];
-
- LARGE_INTEGER _perfCounterFreq;
- double _perfCounterFactor;
- float _avgCPULoad;
-
-private:
- bool _initialized;
- bool _recording;
- bool _playing;
- bool _recIsInitialized;
- bool _playIsInitialized;
- bool _speakerIsInitialized;
- bool _microphoneIsInitialized;
-
- bool _usingInputDeviceIndex;
- bool _usingOutputDeviceIndex;
- AudioDeviceModule::WindowsDeviceType _inputDevice;
- AudioDeviceModule::WindowsDeviceType _outputDevice;
- WebRtc_UWord16 _inputDeviceIndex;
- WebRtc_UWord16 _outputDeviceIndex;
-
- bool _AGC;
-
- WebRtc_UWord16 _playWarning;
- WebRtc_UWord16 _playError;
- WebRtc_UWord16 _recWarning;
- WebRtc_UWord16 _recError;
-
- AudioDeviceModule::BufferType _playBufType;
- WebRtc_UWord16 _playBufDelay;
- WebRtc_UWord16 _playBufDelayFixed;
-
- WebRtc_UWord16 _newMicLevel;
-
- mutable char _str[512];
-};
-
-#endif // #if (_MSC_VER >= 1400)
-
-} // namespace webrtc
-
-#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_CORE_WIN_H
-
diff --git a/trunk/src/modules/audio_device/main/source/win/audio_device_utility_win.cc b/trunk/src/modules/audio_device/main/source/win/audio_device_utility_win.cc
deleted file mode 100644
index 5a29972..0000000
--- a/trunk/src/modules/audio_device/main/source/win/audio_device_utility_win.cc
+++ /dev/null
@@ -1,232 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_device_utility_win.h"
-#include "audio_device_config.h"
-
-#include "critical_section_wrapper.h"
-#include "trace.h"
-
-#include <windows.h>
-#include <tchar.h>
-#include <strsafe.h>
-
-#define STRING_MAX_SIZE 256
-
-typedef void (WINAPI *PGNSI)(LPSYSTEM_INFO);
-typedef BOOL (WINAPI *PGPI)(DWORD, DWORD, DWORD, DWORD, PDWORD);
-
-namespace webrtc
-{
-
-// ============================================================================
-// Construction & Destruction
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// AudioDeviceUtilityWindows() - ctor
-// ----------------------------------------------------------------------------
-
-AudioDeviceUtilityWindows::AudioDeviceUtilityWindows(const WebRtc_Word32 id) :
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _id(id),
- _lastError(AudioDeviceModule::kAdmErrNone)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
-}
-
-// ----------------------------------------------------------------------------
-// AudioDeviceUtilityWindows() - dtor
-// ----------------------------------------------------------------------------
-
-AudioDeviceUtilityWindows::~AudioDeviceUtilityWindows()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
- {
- CriticalSectionScoped lock(_critSect);
-
- // free stuff here...
- }
-
- delete &_critSect;
-}
-
-// ============================================================================
-// API
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// Init()
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceUtilityWindows::Init()
-{
-
- TCHAR szOS[STRING_MAX_SIZE];
-
- if (GetOSDisplayString(szOS))
- {
-#ifdef _UNICODE
- char os[STRING_MAX_SIZE];
- if (WideCharToMultiByte(CP_UTF8, 0, szOS, -1, os, STRING_MAX_SIZE, NULL, NULL) == 0)
- {
- DWORD err = GetLastError();
- strncpy(os, "Could not get OS info", STRING_MAX_SIZE);
- }
- // DEBUG_PRINTP("OS info: %s\n", os);
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, " OS info: %s", os);
-#else
- // DEBUG_PRINTP("OS info: %s\n", szOS);
- WEBRTC_TRACE(kTraceStateInfo, kTraceAudioDevice, _id, " OS info: %s", szOS);
-#endif
- }
-
- return 0;
-}
-
-// ============================================================================
-// Private Methods
-// ============================================================================
-
-BOOL AudioDeviceUtilityWindows::GetOSDisplayString(LPTSTR pszOS)
-{
- OSVERSIONINFOEX osvi;
- SYSTEM_INFO si;
- PGNSI pGNSI;
- BOOL bOsVersionInfoEx;
-
- ZeroMemory(&si, sizeof(SYSTEM_INFO));
- ZeroMemory(&osvi, sizeof(OSVERSIONINFOEX));
-
- osvi.dwOSVersionInfoSize = sizeof(OSVERSIONINFOEX);
-
- // Retrieve information about the current operating system
- //
- if (!(bOsVersionInfoEx = GetVersionEx((OSVERSIONINFO *) &osvi)))
- return FALSE;
-
- // Parse our OS version string
- //
- if (VER_PLATFORM_WIN32_NT == osvi.dwPlatformId && osvi.dwMajorVersion > 4)
- {
- StringCchCopy(pszOS, STRING_MAX_SIZE, TEXT("Microsoft "));
-
- // Test for the specific product
- //
- // Operating system Version number
- // --------------------------------------
- // Windows 7 6.1
- // Windows Server 2008 R2 6.1
- // Windows Server 2008 6.0
- // Windows Vista 6.0
- // - - - - - - - - - - - - - - - - - - -
- // Windows Server 2003 R2 5.2
- // Windows Server 2003 5.2
- // Windows XP 5.1
- // Windows 2000 5.0
- //
- // see http://msdn.microsoft.com/en-us/library/ms724832(VS.85).aspx for details
- //
- if (osvi.dwMajorVersion == 6)
- {
- if (osvi.dwMinorVersion == 0)
- {
- // Windows Vista or Server 2008
- if (osvi.wProductType == VER_NT_WORKSTATION)
- StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows Vista "));
- else
- StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows Server 2008 " ));
- }
-
- if (osvi.dwMinorVersion == 1)
- {
- // Windows 7 or Server 2008 R2
- if (osvi.wProductType == VER_NT_WORKSTATION)
- StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows 7 "));
- else
- StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows Server 2008 R2 " ));
- }
- }
-
- if (osvi.dwMajorVersion == 5 && osvi.dwMinorVersion == 2)
- {
- StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows Server 2003"));
- }
-
- if (osvi.dwMajorVersion == 5 && osvi.dwMinorVersion == 1)
- {
- StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows XP "));
- if (osvi.wSuiteMask & VER_SUITE_PERSONAL)
- StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Home Edition" ));
- else
- StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Professional" ));
- }
-
- if (osvi.dwMajorVersion == 5 && osvi.dwMinorVersion == 0)
- {
- StringCchCat(pszOS, STRING_MAX_SIZE, TEXT("Windows 2000 "));
-
- if (osvi.wProductType == VER_NT_WORKSTATION )
- {
- StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Professional" ));
- }
- else
- {
- if (osvi.wSuiteMask & VER_SUITE_DATACENTER)
- StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Datacenter Server" ));
- else if (osvi.wSuiteMask & VER_SUITE_ENTERPRISE)
- StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Advanced Server" ));
- else StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( "Server" ));
- }
- }
-
- // Include service pack (if any)
- //
- if (_tcslen(osvi.szCSDVersion) > 0)
- {
- StringCchCat(pszOS, STRING_MAX_SIZE, TEXT(" "));
- StringCchCat(pszOS, STRING_MAX_SIZE, osvi.szCSDVersion);
- }
-
- TCHAR buf[80];
-
- // Include build number
- //
- StringCchPrintf( buf, 80, TEXT(" (build %d)"), osvi.dwBuildNumber);
- StringCchCat(pszOS, STRING_MAX_SIZE, buf);
-
- // Call GetNativeSystemInfo if supported or GetSystemInfo otherwise
- //
- pGNSI = (PGNSI) GetProcAddress(GetModuleHandle(TEXT("kernel32.dll")), "GetNativeSystemInfo");
- if (NULL != pGNSI)
- pGNSI(&si);
- else
- GetSystemInfo(&si);
-
- // Add 64-bit or 32-bit for OS versions "later than" Vista
- //
- if (osvi.dwMajorVersion >= 6)
- {
- if ((si.wProcessorArchitecture == PROCESSOR_ARCHITECTURE_AMD64) ||
- (si.wProcessorArchitecture == PROCESSOR_ARCHITECTURE_IA64))
- StringCchCat(pszOS, STRING_MAX_SIZE, TEXT( ", 64-bit" ));
- else if (si.wProcessorArchitecture == PROCESSOR_ARCHITECTURE_INTEL )
- StringCchCat(pszOS, STRING_MAX_SIZE, TEXT(", 32-bit"));
- }
-
- return TRUE;
- }
- else
- {
- return FALSE;
- }
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/win/audio_device_utility_win.h b/trunk/src/modules/audio_device/main/source/win/audio_device_utility_win.h
deleted file mode 100644
index 77b4c22..0000000
--- a/trunk/src/modules/audio_device/main/source/win/audio_device_utility_win.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_WIN_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_WIN_H
-
-#include "audio_device_utility.h"
-#include "audio_device.h"
-#include <windows.h>
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-
-class AudioDeviceUtilityWindows : public AudioDeviceUtility
-{
-public:
- AudioDeviceUtilityWindows(const WebRtc_Word32 id);
- ~AudioDeviceUtilityWindows();
-
- virtual WebRtc_Word32 Init();
-
-private:
- BOOL GetOSDisplayString(LPTSTR pszOS);
-
-private:
- CriticalSectionWrapper& _critSect;
- WebRtc_Word32 _id;
- AudioDeviceModule::ErrorCode _lastError;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_UTILITY_WIN_H
diff --git a/trunk/src/modules/audio_device/main/source/win/audio_device_wave_win.cc b/trunk/src/modules/audio_device/main/source/win/audio_device_wave_win.cc
deleted file mode 100644
index 509732c..0000000
--- a/trunk/src/modules/audio_device/main/source/win/audio_device_wave_win.cc
+++ /dev/null
@@ -1,3818 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_device_utility.h"
-#include "audio_device_wave_win.h"
-#include "audio_device_config.h"
-
-#include "trace.h"
-#include "thread_wrapper.h"
-#include "event_wrapper.h"
-
-#include <windows.h>
-#include <objbase.h> // CoTaskMemAlloc, CoTaskMemFree
-#include <strsafe.h> // StringCchCopy(), StringCchCat(), StringCchPrintf()
-#include <cassert>
-
-// Avoids the need of Windows 7 SDK
-#ifndef WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE
-#define WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE 0x0010
-#endif
-
-// Supported in Windows Vista and Windows 7.
-// http://msdn.microsoft.com/en-us/library/dd370819(v=VS.85).aspx
-// Taken from Mmddk.h.
-#define DRV_RESERVED 0x0800
-#define DRV_QUERYFUNCTIONINSTANCEID (DRV_RESERVED + 17)
-#define DRV_QUERYFUNCTIONINSTANCEIDSIZE (DRV_RESERVED + 18)
-
-#define POW2(A) (2 << ((A) - 1))
-
-namespace webrtc {
-
-// ============================================================================
-// Construction & Destruction
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// AudioDeviceWindowsWave - ctor
-// ----------------------------------------------------------------------------
-
-AudioDeviceWindowsWave::AudioDeviceWindowsWave(const WebRtc_Word32 id) :
- _ptrAudioBuffer(NULL),
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _timeEvent(*EventWrapper::Create()),
- _recStartEvent(*EventWrapper::Create()),
- _playStartEvent(*EventWrapper::Create()),
- _hGetCaptureVolumeThread(NULL),
- _hShutdownGetVolumeEvent(NULL),
- _hSetCaptureVolumeThread(NULL),
- _hShutdownSetVolumeEvent(NULL),
- _hSetCaptureVolumeEvent(NULL),
- _ptrThread(NULL),
- _threadID(0),
- _critSectCb(*CriticalSectionWrapper::CreateCriticalSection()),
- _id(id),
- _mixerManager(id),
- _usingInputDeviceIndex(false),
- _usingOutputDeviceIndex(false),
- _inputDevice(AudioDeviceModule::kDefaultDevice),
- _outputDevice(AudioDeviceModule::kDefaultDevice),
- _inputDeviceIndex(0),
- _outputDeviceIndex(0),
- _inputDeviceIsSpecified(false),
- _outputDeviceIsSpecified(false),
- _initialized(false),
- _recIsInitialized(false),
- _playIsInitialized(false),
- _recording(false),
- _playing(false),
- _startRec(false),
- _stopRec(false),
- _startPlay(false),
- _stopPlay(false),
- _AGC(false),
- _hWaveIn(NULL),
- _hWaveOut(NULL),
- _recChannels(N_REC_CHANNELS),
- _playChannels(N_PLAY_CHANNELS),
- _recBufCount(0),
- _recPutBackDelay(0),
- _recDelayCount(0),
- _playBufCount(0),
- _prevPlayTime(0),
- _prevRecTime(0),
- _prevTimerCheckTime(0),
- _timesdwBytes(0),
- _timerFaults(0),
- _timerRestartAttempts(0),
- _no_of_msecleft_warnings(0),
- _MAX_minBuffer(65),
- _useHeader(0),
- _dTcheckPlayBufDelay(10),
- _playBufDelay(80),
- _playBufDelayFixed(80),
- _minPlayBufDelay(20),
- _avgCPULoad(0),
- _sndCardPlayDelay(0),
- _sndCardRecDelay(0),
- _plSampOld(0),
- _rcSampOld(0),
- _playBufType(AudioDeviceModule::kAdaptiveBufferSize),
- _recordedBytes(0),
- _playWarning(0),
- _playError(0),
- _recWarning(0),
- _recError(0),
- _newMicLevel(0),
- _minMicVolume(0),
- _maxMicVolume(0)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, id, "%s created", __FUNCTION__);
-
- // Initialize value, set to 0 if it fails
- if (!QueryPerformanceFrequency(&_perfFreq))
- {
- _perfFreq.QuadPart = 0;
- }
-
- _hShutdownGetVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
- _hShutdownSetVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
- _hSetCaptureVolumeEvent = CreateEvent(NULL, FALSE, FALSE, NULL);
-}
-
-// ----------------------------------------------------------------------------
-// AudioDeviceWindowsWave - dtor
-// ----------------------------------------------------------------------------
-
-AudioDeviceWindowsWave::~AudioDeviceWindowsWave()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destroyed", __FUNCTION__);
-
- Terminate();
-
- delete &_recStartEvent;
- delete &_playStartEvent;
- delete &_timeEvent;
- delete &_critSect;
- delete &_critSectCb;
-
- if (NULL != _hShutdownGetVolumeEvent)
- {
- CloseHandle(_hShutdownGetVolumeEvent);
- _hShutdownGetVolumeEvent = NULL;
- }
-
- if (NULL != _hShutdownSetVolumeEvent)
- {
- CloseHandle(_hShutdownSetVolumeEvent);
- _hShutdownSetVolumeEvent = NULL;
- }
-
- if (NULL != _hSetCaptureVolumeEvent)
- {
- CloseHandle(_hSetCaptureVolumeEvent);
- _hSetCaptureVolumeEvent = NULL;
- }
-}
-
-// ============================================================================
-// API
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// AttachAudioBuffer
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsWave::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
-{
-
- CriticalSectionScoped lock(_critSect);
-
- _ptrAudioBuffer = audioBuffer;
-
- // inform the AudioBuffer about default settings for this implementation
- _ptrAudioBuffer->SetRecordingSampleRate(N_REC_SAMPLES_PER_SEC);
- _ptrAudioBuffer->SetPlayoutSampleRate(N_PLAY_SAMPLES_PER_SEC);
- _ptrAudioBuffer->SetRecordingChannels(N_REC_CHANNELS);
- _ptrAudioBuffer->SetPlayoutChannels(N_PLAY_CHANNELS);
-}
-
-// ----------------------------------------------------------------------------
-// ActiveAudioLayer
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const
-{
- audioLayer = AudioDeviceModule::kWindowsWaveAudio;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Init
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::Init()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_initialized)
- {
- return 0;
- }
-
- const WebRtc_UWord32 nowTime(AudioDeviceUtility::GetTimeInMS());
-
- _recordedBytes = 0;
- _prevRecByteCheckTime = nowTime;
- _prevRecTime = nowTime;
- _prevPlayTime = nowTime;
- _prevTimerCheckTime = nowTime;
-
- _playWarning = 0;
- _playError = 0;
- _recWarning = 0;
- _recError = 0;
-
- _mixerManager.EnumerateAll();
-
- if (_ptrThread)
- {
- // thread is already created and active
- return 0;
- }
-
- const char* threadName = "webrtc_audio_module_thread";
- _ptrThread = ThreadWrapper::CreateThread(ThreadFunc,
- this,
- kRealtimePriority,
- threadName);
- if (_ptrThread == NULL)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- "failed to create the audio thread");
- return -1;
- }
-
- unsigned int threadID(0);
- if (!_ptrThread->Start(threadID))
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- "failed to start the audio thread");
- delete _ptrThread;
- _ptrThread = NULL;
- return -1;
- }
- _threadID = threadID;
-
- const bool periodic(true);
- if (!_timeEvent.StartTimer(periodic, TIMER_PERIOD_MS))
- {
- WEBRTC_TRACE(kTraceCritical, kTraceAudioDevice, _id,
- "failed to start the timer event");
- if (_ptrThread->Stop())
- {
- delete _ptrThread;
- _ptrThread = NULL;
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "unable to stop the activated thread");
- }
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "periodic timer (dT=%d) is now active", TIMER_PERIOD_MS);
-
- _hGetCaptureVolumeThread = CreateThread(NULL,
- 0,
- GetCaptureVolumeThread,
- this,
- 0,
- NULL);
- if (_hGetCaptureVolumeThread == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to create the volume getter thread");
- return -1;
- }
-
- SetThreadPriority(_hGetCaptureVolumeThread, THREAD_PRIORITY_NORMAL);
-
- _hSetCaptureVolumeThread = CreateThread(NULL,
- 0,
- SetCaptureVolumeThread,
- this,
- 0,
- NULL);
- if (_hSetCaptureVolumeThread == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to create the volume setter thread");
- return -1;
- }
-
- SetThreadPriority(_hSetCaptureVolumeThread, THREAD_PRIORITY_NORMAL);
-
- _initialized = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Terminate
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::Terminate()
-{
-
- if (!_initialized)
- {
- return 0;
- }
-
- _critSect.Enter();
-
- _mixerManager.Close();
-
- if (_ptrThread)
- {
- ThreadWrapper* tmpThread = _ptrThread;
- _ptrThread = NULL;
- _critSect.Leave();
-
- tmpThread->SetNotAlive();
- _timeEvent.Set();
-
- if (tmpThread->Stop())
- {
- delete tmpThread;
- }
- else
- {
- _critSect.Leave();
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- "failed to close down the audio thread");
- return -1;
- }
- }
- else
- {
- _critSect.Leave();
- }
-
- _critSect.Enter();
- SetEvent(_hShutdownGetVolumeEvent);
- _critSect.Leave();
- WebRtc_Word32 ret = WaitForSingleObject(_hGetCaptureVolumeThread, 2000);
- if (ret != WAIT_OBJECT_0)
- {
- // the thread did not stop as it should
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to close down volume getter thread");
- CloseHandle(_hGetCaptureVolumeThread);
- _hGetCaptureVolumeThread = NULL;
- return -1;
- }
- _critSect.Enter();
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " volume getter thread is now closed");
-
- SetEvent(_hShutdownSetVolumeEvent);
- _critSect.Leave();
- ret = WaitForSingleObject(_hSetCaptureVolumeThread, 2000);
- if (ret != WAIT_OBJECT_0)
- {
- // the thread did not stop as it should
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id,
- " failed to close down volume setter thread");
- CloseHandle(_hSetCaptureVolumeThread);
- _hSetCaptureVolumeThread = NULL;
- return -1;
- }
- _critSect.Enter();
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- " volume setter thread is now closed");
-
- CloseHandle(_hGetCaptureVolumeThread);
- _hGetCaptureVolumeThread = NULL;
-
- CloseHandle(_hSetCaptureVolumeThread);
- _hSetCaptureVolumeThread = NULL;
-
- _critSect.Leave();
-
- _timeEvent.StopTimer();
-
- _initialized = false;
- _outputDeviceIsSpecified = false;
- _inputDeviceIsSpecified = false;
-
- return 0;
-}
-
-
-DWORD WINAPI AudioDeviceWindowsWave::GetCaptureVolumeThread(LPVOID context)
-{
- return(((AudioDeviceWindowsWave*)context)->DoGetCaptureVolumeThread());
-}
-
-DWORD WINAPI AudioDeviceWindowsWave::SetCaptureVolumeThread(LPVOID context)
-{
- return(((AudioDeviceWindowsWave*)context)->DoSetCaptureVolumeThread());
-}
-
-DWORD AudioDeviceWindowsWave::DoGetCaptureVolumeThread()
-{
- HANDLE waitObject = _hShutdownGetVolumeEvent;
-
- while (1)
- {
- DWORD waitResult = WaitForSingleObject(waitObject,
- GET_MIC_VOLUME_INTERVAL_MS);
- switch (waitResult)
- {
- case WAIT_OBJECT_0: // _hShutdownGetVolumeEvent
- return 0;
- case WAIT_TIMEOUT: // timeout notification
- break;
- default: // unexpected error
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " unknown wait termination on get volume thread");
- return -1;
- }
-
- if (AGC())
- {
- WebRtc_UWord32 currentMicLevel = 0;
- if (MicrophoneVolume(currentMicLevel) == 0)
- {
- // This doesn't set the system volume, just stores it.
- _critSect.Enter();
- if (_ptrAudioBuffer)
- {
- _ptrAudioBuffer->SetCurrentMicLevel(currentMicLevel);
- }
- _critSect.Leave();
- }
- }
- }
-}
-
-DWORD AudioDeviceWindowsWave::DoSetCaptureVolumeThread()
-{
- HANDLE waitArray[2] = {_hShutdownSetVolumeEvent, _hSetCaptureVolumeEvent};
-
- while (1)
- {
- DWORD waitResult = WaitForMultipleObjects(2, waitArray, FALSE, INFINITE);
- switch (waitResult)
- {
- case WAIT_OBJECT_0: // _hShutdownSetVolumeEvent
- return 0;
- case WAIT_OBJECT_0 + 1: // _hSetCaptureVolumeEvent
- break;
- default: // unexpected error
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " unknown wait termination on set volume thread");
- return -1;
- }
-
- _critSect.Enter();
- WebRtc_UWord32 newMicLevel = _newMicLevel;
- _critSect.Leave();
-
- if (SetMicrophoneVolume(newMicLevel) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " the required modification of the microphone volume failed");
- }
- }
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Initialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::Initialized() const
-{
- return (_initialized);
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SpeakerIsAvailable(bool& available)
-{
-
- // Enumerate all avaliable speakers and make an attempt to open up the
- // output mixer corresponding to the currently selected output device.
- //
- if (InitSpeaker() == -1)
- {
- available = false;
- return 0;
- }
-
- // Given that InitSpeaker was successful, we know that a valid speaker exists
- //
- available = true;
-
- // Close the initialized output mixer
- //
- _mixerManager.CloseSpeaker();
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitSpeaker
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::InitSpeaker()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_playing)
- {
- return -1;
- }
-
- if (_mixerManager.EnumerateSpeakers() == -1)
- {
- // failed to locate any valid/controllable speaker
- return -1;
- }
-
- if (IsUsingOutputDeviceIndex())
- {
- if (_mixerManager.OpenSpeaker(OutputDeviceIndex()) == -1)
- {
- return -1;
- }
- }
- else
- {
- if (_mixerManager.OpenSpeaker(OutputDevice()) == -1)
- {
- return -1;
- }
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneIsAvailable(bool& available)
-{
-
- // Enumerate all avaliable microphones and make an attempt to open up the
- // input mixer corresponding to the currently selected output device.
- //
- if (InitMicrophone() == -1)
- {
- available = false;
- return 0;
- }
-
- // Given that InitMicrophone was successful, we know that a valid microphone exists
- //
- available = true;
-
- // Close the initialized input mixer
- //
- _mixerManager.CloseMicrophone();
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitMicrophone
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::InitMicrophone()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_recording)
- {
- return -1;
- }
-
- if (_mixerManager.EnumerateMicrophones() == -1)
- {
- // failed to locate any valid/controllable microphone
- return -1;
- }
-
- if (IsUsingInputDeviceIndex())
- {
- if (_mixerManager.OpenMicrophone(InputDeviceIndex()) == -1)
- {
- return -1;
- }
- }
- else
- {
- if (_mixerManager.OpenMicrophone(InputDevice()) == -1)
- {
- return -1;
- }
- }
-
- WebRtc_UWord32 maxVol = 0;
- if (_mixerManager.MaxMicrophoneVolume(maxVol) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " unable to retrieve max microphone volume");
- }
- _maxMicVolume = maxVol;
-
- WebRtc_UWord32 minVol = 0;
- if (_mixerManager.MinMicrophoneVolume(minVol) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id,
- " unable to retrieve min microphone volume");
- }
- _minMicVolume = minVol;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::SpeakerIsInitialized() const
-{
- return (_mixerManager.SpeakerIsInitialized());
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::MicrophoneIsInitialized() const
-{
- return (_mixerManager.MicrophoneIsInitialized());
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolumeIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SpeakerVolumeIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
-
- // Enumerate all avaliable speakers and make an attempt to open up the
- // output mixer corresponding to the currently selected output device.
- //
- if (InitSpeaker() == -1)
- {
- // failed to find a valid speaker
- available = false;
- return 0;
- }
-
- // Check if the selected speaker has a volume control
- //
- _mixerManager.SpeakerVolumeIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized output mixer
- //
- _mixerManager.CloseSpeaker();
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetSpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SetSpeakerVolume(WebRtc_UWord32 volume)
-{
-
- return (_mixerManager.SetSpeakerVolume(volume));
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SpeakerVolume(WebRtc_UWord32& volume) const
-{
-
- WebRtc_UWord32 level(0);
-
- if (_mixerManager.SpeakerVolume(level) == -1)
- {
- return -1;
- }
-
- volume = level;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetWaveOutVolume
-//
-// The low-order word contains the left-channel volume setting, and the
-// high-order word contains the right-channel setting.
-// A value of 0xFFFF represents full volume, and a value of 0x0000 is silence.
-//
-// If a device does not support both left and right volume control,
-// the low-order word of dwVolume specifies the volume level,
-// and the high-order word is ignored.
-//
-// Most devices do not support the full 16 bits of volume-level control
-// and will not use the least-significant bits of the requested volume setting.
-// For example, if a device supports 4 bits of volume control, the values
-// 0x4000, 0x4FFF, and 0x43BE will all be truncated to 0x4000.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SetWaveOutVolume(WebRtc_UWord16 volumeLeft, WebRtc_UWord16 volumeRight)
-{
-
- MMRESULT res(0);
- WAVEOUTCAPS caps;
-
- CriticalSectionScoped lock(_critSect);
-
- if (_hWaveOut == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no open playout device exists => using default");
- }
-
- // To determine whether the device supports volume control on both
- // the left and right channels, use the WAVECAPS_LRVOLUME flag.
- //
- res = waveOutGetDevCaps((UINT_PTR)_hWaveOut, &caps, sizeof(WAVEOUTCAPS));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCaps() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
- if (!(caps.dwSupport & WAVECAPS_VOLUME))
- {
- // this device does not support volume control using the waveOutSetVolume API
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device does not support volume control using the Wave API");
- return -1;
- }
- if (!(caps.dwSupport & WAVECAPS_LRVOLUME))
- {
- // high-order word (right channel) is ignored
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "device does not support volume control on both channels");
- }
-
- DWORD dwVolume(0x00000000);
- dwVolume = (DWORD)(((volumeRight & 0xFFFF) << 16) | (volumeLeft & 0xFFFF));
-
- res = waveOutSetVolume(_hWaveOut, dwVolume);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveOutSetVolume() failed (err=%d)", res);
- TraceWaveOutError(res);
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// WaveOutVolume
-//
-// The low-order word of this location contains the left-channel volume setting,
-// and the high-order word contains the right-channel setting.
-// A value of 0xFFFF (65535) represents full volume, and a value of 0x0000
-// is silence.
-//
-// If a device does not support both left and right volume control,
-// the low-order word of the specified location contains the mono volume level.
-//
-// The full 16-bit setting(s) set with the waveOutSetVolume function is returned,
-// regardless of whether the device supports the full 16 bits of volume-level
-// control.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::WaveOutVolume(WebRtc_UWord16& volumeLeft, WebRtc_UWord16& volumeRight) const
-{
-
- MMRESULT res(0);
- WAVEOUTCAPS caps;
-
- CriticalSectionScoped lock(_critSect);
-
- if (_hWaveOut == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no open playout device exists => using default");
- }
-
- // To determine whether the device supports volume control on both
- // the left and right channels, use the WAVECAPS_LRVOLUME flag.
- //
- res = waveOutGetDevCaps((UINT_PTR)_hWaveOut, &caps, sizeof(WAVEOUTCAPS));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCaps() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
- if (!(caps.dwSupport & WAVECAPS_VOLUME))
- {
- // this device does not support volume control using the waveOutSetVolume API
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device does not support volume control using the Wave API");
- return -1;
- }
- if (!(caps.dwSupport & WAVECAPS_LRVOLUME))
- {
- // high-order word (right channel) is ignored
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "device does not support volume control on both channels");
- }
-
- DWORD dwVolume(0x00000000);
-
- res = waveOutGetVolume(_hWaveOut, &dwVolume);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveOutGetVolume() failed (err=%d)", res);
- TraceWaveOutError(res);
- return -1;
- }
-
- WORD wVolumeLeft = LOWORD(dwVolume);
- WORD wVolumeRight = HIWORD(dwVolume);
-
- volumeLeft = static_cast<WebRtc_UWord16> (wVolumeLeft);
- volumeRight = static_cast<WebRtc_UWord16> (wVolumeRight);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MaxSpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const
-{
-
- WebRtc_UWord32 maxVol(0);
-
- if (_mixerManager.MaxSpeakerVolume(maxVol) == -1)
- {
- return -1;
- }
-
- maxVolume = maxVol;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MinSpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::MinSpeakerVolume(WebRtc_UWord32& minVolume) const
-{
-
- WebRtc_UWord32 minVol(0);
-
- if (_mixerManager.MinSpeakerVolume(minVol) == -1)
- {
- return -1;
- }
-
- minVolume = minVol;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolumeStepSize
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const
-{
-
- WebRtc_UWord16 delta(0);
-
- if (_mixerManager.SpeakerVolumeStepSize(delta) == -1)
- {
- return -1;
- }
-
- stepSize = delta;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerMuteIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SpeakerMuteIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
-
- // Enumerate all avaliable speakers and make an attempt to open up the
- // output mixer corresponding to the currently selected output device.
- //
- if (InitSpeaker() == -1)
- {
- // If we end up here it means that the selected speaker has no volume
- // control, hence it is safe to state that there is no mute control
- // already at this stage.
- available = false;
- return 0;
- }
-
- // Check if the selected speaker has a mute control
- //
- _mixerManager.SpeakerMuteIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized output mixer
- //
- _mixerManager.CloseSpeaker();
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetSpeakerMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SetSpeakerMute(bool enable)
-{
- return (_mixerManager.SetSpeakerMute(enable));
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SpeakerMute(bool& enabled) const
-{
-
- bool muted(0);
-
- if (_mixerManager.SpeakerMute(muted) == -1)
- {
- return -1;
- }
-
- enabled = muted;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneMuteIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneMuteIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
-
- // Enumerate all avaliable microphones and make an attempt to open up the
- // input mixer corresponding to the currently selected input device.
- //
- if (InitMicrophone() == -1)
- {
- // If we end up here it means that the selected microphone has no volume
- // control, hence it is safe to state that there is no boost control
- // already at this stage.
- available = false;
- return 0;
- }
-
- // Check if the selected microphone has a mute control
- //
- _mixerManager.MicrophoneMuteIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized input mixer
- //
- _mixerManager.CloseMicrophone();
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SetMicrophoneMute(bool enable)
-{
- return (_mixerManager.SetMicrophoneMute(enable));
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneMute(bool& enabled) const
-{
-
- bool muted(0);
-
- if (_mixerManager.MicrophoneMute(muted) == -1)
- {
- return -1;
- }
-
- enabled = muted;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneBoostIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneBoostIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
-
- // Enumerate all avaliable microphones and make an attempt to open up the
- // input mixer corresponding to the currently selected input device.
- //
- if (InitMicrophone() == -1)
- {
- // If we end up here it means that the selected microphone has no volume
- // control, hence it is safe to state that there is no boost control
- // already at this stage.
- available = false;
- return 0;
- }
-
- // Check if the selected microphone has a boost control
- //
- _mixerManager.MicrophoneBoostIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized input mixer
- //
- _mixerManager.CloseMicrophone();
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneBoost
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SetMicrophoneBoost(bool enable)
-{
-
- return (_mixerManager.SetMicrophoneBoost(enable));
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneBoost
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneBoost(bool& enabled) const
-{
-
- bool onOff(0);
-
- if (_mixerManager.MicrophoneBoost(onOff) == -1)
- {
- return -1;
- }
-
- enabled = onOff;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoRecordingIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::StereoRecordingIsAvailable(bool& available)
-{
- available = true;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetStereoRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SetStereoRecording(bool enable)
-{
-
- if (enable)
- _recChannels = 2;
- else
- _recChannels = 1;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::StereoRecording(bool& enabled) const
-{
-
- if (_recChannels == 2)
- enabled = true;
- else
- enabled = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoPlayoutIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::StereoPlayoutIsAvailable(bool& available)
-{
- available = true;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetStereoPlayout
-//
-// Specifies the number of output channels.
-//
-// NOTE - the setting will only have an effect after InitPlayout has
-// been called.
-//
-// 16-bit mono:
-//
-// Each sample is 2 bytes. Sample 1 is followed by samples 2, 3, 4, and so on.
-// For each sample, the first byte is the low-order byte of channel 0 and the
-// second byte is the high-order byte of channel 0.
-//
-// 16-bit stereo:
-//
-// Each sample is 4 bytes. Sample 1 is followed by samples 2, 3, 4, and so on.
-// For each sample, the first byte is the low-order byte of channel 0 (left channel);
-// the second byte is the high-order byte of channel 0; the third byte is the
-// low-order byte of channel 1 (right channel); and the fourth byte is the
-// high-order byte of channel 1.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SetStereoPlayout(bool enable)
-{
-
- if (enable)
- _playChannels = 2;
- else
- _playChannels = 1;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StereoPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::StereoPlayout(bool& enabled) const
-{
-
- if (_playChannels == 2)
- enabled = true;
- else
- enabled = false;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetAGC
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SetAGC(bool enable)
-{
-
- _AGC = enable;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// AGC
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::AGC() const
-{
- return _AGC;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolumeIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneVolumeIsAvailable(bool& available)
-{
-
- bool isAvailable(false);
-
- // Enumerate all avaliable microphones and make an attempt to open up the
- // input mixer corresponding to the currently selected output device.
- //
- if (InitMicrophone() == -1)
- {
- // Failed to find valid microphone
- available = false;
- return 0;
- }
-
- // Check if the selected microphone has a volume control
- //
- _mixerManager.MicrophoneVolumeIsAvailable(isAvailable);
- available = isAvailable;
-
- // Close the initialized input mixer
- //
- _mixerManager.CloseMicrophone();
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SetMicrophoneVolume(WebRtc_UWord32 volume)
-{
- return (_mixerManager.SetMicrophoneVolume(volume));
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneVolume(WebRtc_UWord32& volume) const
-{
- WebRtc_UWord32 level(0);
-
- if (_mixerManager.MicrophoneVolume(level) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to retrive current microphone level");
- return -1;
- }
-
- volume = level;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MaxMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const
-{
- // _maxMicVolume can be zero in AudioMixerManager::MaxMicrophoneVolume():
- // (1) API GetLineControl() returns failure at querying the max Mic level.
- // (2) API GetLineControl() returns maxVolume as zero in rare cases.
- // Both cases show we don't have access to the mixer controls.
- // We return -1 here to indicate that.
- if (_maxMicVolume == 0)
- {
- return -1;
- }
-
- maxVolume = _maxMicVolume;;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MinMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::MinMicrophoneVolume(WebRtc_UWord32& minVolume) const
-{
- minVolume = _minMicVolume;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolumeStepSize
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const
-{
-
- WebRtc_UWord16 delta(0);
-
- if (_mixerManager.MicrophoneVolumeStepSize(delta) == -1)
- {
- return -1;
- }
-
- stepSize = delta;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDevices
-// ----------------------------------------------------------------------------
-
-WebRtc_Word16 AudioDeviceWindowsWave::PlayoutDevices()
-{
-
- return (waveOutGetNumDevs());
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutDevice I (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SetPlayoutDevice(WebRtc_UWord16 index)
-{
-
- if (_playIsInitialized)
- {
- return -1;
- }
-
- UINT nDevices = waveOutGetNumDevs();
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "number of availiable waveform-audio output devices is %u", nDevices);
-
- if (index < 0 || index > (nDevices-1))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device index is out of range [0,%u]", (nDevices-1));
- return -1;
- }
-
- _usingOutputDeviceIndex = true;
- _outputDeviceIndex = index;
- _outputDeviceIsSpecified = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetPlayoutDevice II (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device)
-{
- if (_playIsInitialized)
- {
- return -1;
- }
-
- if (device == AudioDeviceModule::kDefaultDevice)
- {
- }
- else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
- {
- }
-
- _usingOutputDeviceIndex = false;
- _outputDevice = device;
- _outputDeviceIsSpecified = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDeviceName
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::PlayoutDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
-{
-
- WebRtc_UWord16 nDevices(PlayoutDevices());
-
- // Special fix for the case when the user asks for the name of the default device.
- //
- if (index == (WebRtc_UWord16)(-1))
- {
- index = 0;
- }
-
- if ((index > (nDevices-1)) || (name == NULL))
- {
- return -1;
- }
-
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- WAVEOUTCAPSW caps; // szPname member (product name (NULL terminated) is a WCHAR
- MMRESULT res;
-
- res = waveOutGetDevCapsW(index, &caps, sizeof(WAVEOUTCAPSW));
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCapsW() failed (err=%d)", res);
- return -1;
- }
- if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, name, kAdmMaxDeviceNameSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 1", GetLastError());
- }
-
- if (guid == NULL)
- {
- return 0;
- }
-
- // It is possible to get the unique endpoint ID string using the Wave API.
- // However, it is only supported on Windows Vista and Windows 7.
-
- size_t cbEndpointId(0);
-
- // Get the size (including the terminating null) of the endpoint ID string of the waveOut device.
- // Windows Vista supports the DRV_QUERYFUNCTIONINSTANCEIDSIZE and DRV_QUERYFUNCTIONINSTANCEID messages.
- res = waveOutMessage((HWAVEOUT)IntToPtr(index),
- DRV_QUERYFUNCTIONINSTANCEIDSIZE,
- (DWORD_PTR)&cbEndpointId, NULL);
- if (res != MMSYSERR_NOERROR)
- {
- // DRV_QUERYFUNCTIONINSTANCEIDSIZE is not supported <=> earlier version of Windows than Vista
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveOutMessage(DRV_QUERYFUNCTIONINSTANCEIDSIZE) failed (err=%d)", res);
- TraceWaveOutError(res);
- // Best we can do is to copy the friendly name and use it as guid
- if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 2", GetLastError());
- }
- return 0;
- }
-
- // waveOutMessage(DRV_QUERYFUNCTIONINSTANCEIDSIZE) worked => we are on a Vista or Windows 7 device
-
- WCHAR *pstrEndpointId = NULL;
- pstrEndpointId = (WCHAR*)CoTaskMemAlloc(cbEndpointId);
-
- // Get the endpoint ID string for this waveOut device.
- res = waveOutMessage((HWAVEOUT)IntToPtr(index),
- DRV_QUERYFUNCTIONINSTANCEID,
- (DWORD_PTR)pstrEndpointId,
- cbEndpointId);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveOutMessage(DRV_QUERYFUNCTIONINSTANCEID) failed (err=%d)", res);
- TraceWaveOutError(res);
- // Best we can do is to copy the friendly name and use it as guid
- if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 3", GetLastError());
- }
- CoTaskMemFree(pstrEndpointId);
- return 0;
- }
-
- if (WideCharToMultiByte(CP_UTF8, 0, pstrEndpointId, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 4", GetLastError());
- }
- CoTaskMemFree(pstrEndpointId);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDeviceName
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::RecordingDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize])
-{
-
- WebRtc_UWord16 nDevices(RecordingDevices());
-
- // Special fix for the case when the user asks for the name of the default device.
- //
- if (index == (WebRtc_UWord16)(-1))
- {
- index = 0;
- }
-
- if ((index > (nDevices-1)) || (name == NULL))
- {
- return -1;
- }
-
- memset(name, 0, kAdmMaxDeviceNameSize);
-
- if (guid != NULL)
- {
- memset(guid, 0, kAdmMaxGuidSize);
- }
-
- WAVEINCAPSW caps; // szPname member (product name (NULL terminated) is a WCHAR
- MMRESULT res;
-
- res = waveInGetDevCapsW(index, &caps, sizeof(WAVEINCAPSW));
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetDevCapsW() failed (err=%d)", res);
- return -1;
- }
- if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, name, kAdmMaxDeviceNameSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 1", GetLastError());
- }
-
- if (guid == NULL)
- {
- return 0;
- }
-
- // It is possible to get the unique endpoint ID string using the Wave API.
- // However, it is only supported on Windows Vista and Windows 7.
-
- size_t cbEndpointId(0);
-
- // Get the size (including the terminating null) of the endpoint ID string of the waveOut device.
- // Windows Vista supports the DRV_QUERYFUNCTIONINSTANCEIDSIZE and DRV_QUERYFUNCTIONINSTANCEID messages.
- res = waveInMessage((HWAVEIN)IntToPtr(index),
- DRV_QUERYFUNCTIONINSTANCEIDSIZE,
- (DWORD_PTR)&cbEndpointId, NULL);
- if (res != MMSYSERR_NOERROR)
- {
- // DRV_QUERYFUNCTIONINSTANCEIDSIZE is not supported <=> earlier version of Windows than Vista
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveInMessage(DRV_QUERYFUNCTIONINSTANCEIDSIZE) failed (err=%d)", res);
- TraceWaveInError(res);
- // Best we can do is to copy the friendly name and use it as guid
- if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 2", GetLastError());
- }
- return 0;
- }
-
- // waveOutMessage(DRV_QUERYFUNCTIONINSTANCEIDSIZE) worked => we are on a Vista or Windows 7 device
-
- WCHAR *pstrEndpointId = NULL;
- pstrEndpointId = (WCHAR*)CoTaskMemAlloc(cbEndpointId);
-
- // Get the endpoint ID string for this waveOut device.
- res = waveInMessage((HWAVEIN)IntToPtr(index),
- DRV_QUERYFUNCTIONINSTANCEID,
- (DWORD_PTR)pstrEndpointId,
- cbEndpointId);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveInMessage(DRV_QUERYFUNCTIONINSTANCEID) failed (err=%d)", res);
- TraceWaveInError(res);
- // Best we can do is to copy the friendly name and use it as guid
- if (WideCharToMultiByte(CP_UTF8, 0, caps.szPname, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 3", GetLastError());
- }
- CoTaskMemFree(pstrEndpointId);
- return 0;
- }
-
- if (WideCharToMultiByte(CP_UTF8, 0, pstrEndpointId, -1, guid, kAdmMaxGuidSize, NULL, NULL) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "WideCharToMultiByte(CP_UTF8) failed with error code %d - 4", GetLastError());
- }
- CoTaskMemFree(pstrEndpointId);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDevices
-// ----------------------------------------------------------------------------
-
-WebRtc_Word16 AudioDeviceWindowsWave::RecordingDevices()
-{
-
- return (waveInGetNumDevs());
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingDevice I (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SetRecordingDevice(WebRtc_UWord16 index)
-{
-
- if (_recIsInitialized)
- {
- return -1;
- }
-
- UINT nDevices = waveInGetNumDevs();
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "number of availiable waveform-audio input devices is %u", nDevices);
-
- if (index < 0 || index > (nDevices-1))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "device index is out of range [0,%u]", (nDevices-1));
- return -1;
- }
-
- _usingInputDeviceIndex = true;
- _inputDeviceIndex = index;
- _inputDeviceIsSpecified = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetRecordingDevice II (II)
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device)
-{
- if (device == AudioDeviceModule::kDefaultDevice)
- {
- }
- else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
- {
- }
-
- if (_recIsInitialized)
- {
- return -1;
- }
-
- _usingInputDeviceIndex = false;
- _inputDevice = device;
- _inputDeviceIsSpecified = true;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::PlayoutIsAvailable(bool& available)
-{
-
- available = false;
-
- // Try to initialize the playout side
- WebRtc_Word32 res = InitPlayout();
-
- // Cancel effect of initialization
- StopPlayout();
-
- if (res != -1)
- {
- available = true;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::RecordingIsAvailable(bool& available)
-{
-
- available = false;
-
- // Try to initialize the recording side
- WebRtc_Word32 res = InitRecording();
-
- // Cancel effect of initialization
- StopRecording();
-
- if (res != -1)
- {
- available = true;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::InitPlayout()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_playing)
- {
- return -1;
- }
-
- if (!_outputDeviceIsSpecified)
- {
- return -1;
- }
-
- if (_playIsInitialized)
- {
- return 0;
- }
-
- // Initialize the speaker (devices might have been added or removed)
- if (InitSpeaker() == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "InitSpeaker() failed");
- }
-
- // Enumerate all availiable output devices
- EnumeratePlayoutDevices();
-
- // Start by closing any existing wave-output devices
- //
- MMRESULT res(MMSYSERR_ERROR);
-
- if (_hWaveOut != NULL)
- {
- res = waveOutClose(_hWaveOut);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutClose() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
- }
-
- // Set the output wave format
- //
- WAVEFORMATEX waveFormat;
-
- waveFormat.wFormatTag = WAVE_FORMAT_PCM;
- waveFormat.nChannels = _playChannels; // mono <=> 1, stereo <=> 2
- waveFormat.nSamplesPerSec = N_PLAY_SAMPLES_PER_SEC;
- waveFormat.wBitsPerSample = 16;
- waveFormat.nBlockAlign = waveFormat.nChannels * (waveFormat.wBitsPerSample/8);
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
- waveFormat.cbSize = 0;
-
- // Open the given waveform-audio output device for playout
- //
- HWAVEOUT hWaveOut(NULL);
-
- if (IsUsingOutputDeviceIndex())
- {
- // verify settings first
- res = waveOutOpen(NULL, _outputDeviceIndex, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_FORMAT_QUERY);
- if (MMSYSERR_NOERROR == res)
- {
- // open the given waveform-audio output device for recording
- res = waveOutOpen(&hWaveOut, _outputDeviceIndex, &waveFormat, 0, 0, CALLBACK_NULL);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening output device corresponding to device ID %u", _outputDeviceIndex);
- }
- }
- else
- {
- if (_outputDevice == AudioDeviceModule::kDefaultCommunicationDevice)
- {
- // check if it is possible to open the default communication device (supported on Windows 7)
- res = waveOutOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE | WAVE_FORMAT_QUERY);
- if (MMSYSERR_NOERROR == res)
- {
- // if so, open the default communication device for real
- res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default communication device");
- }
- else
- {
- // use default device since default communication device was not avaliable
- res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "unable to open default communication device => using default instead");
- }
- }
- else if (_outputDevice == AudioDeviceModule::kDefaultDevice)
- {
- // open default device since it has been requested
- res = waveOutOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_FORMAT_QUERY);
- if (MMSYSERR_NOERROR == res)
- {
- res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default output device");
- }
- }
- }
-
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveOutOpen() failed (err=%d)", res);
- TraceWaveOutError(res);
- return -1;
- }
-
- // Log information about the aquired output device
- //
- WAVEOUTCAPS caps;
-
- res = waveOutGetDevCaps((UINT_PTR)hWaveOut, &caps, sizeof(WAVEOUTCAPS));
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCaps() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
-
- UINT deviceID(0);
- res = waveOutGetID(hWaveOut, &deviceID);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetID() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "utilized device ID : %u", deviceID);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", caps.szPname);
-
- // Store valid handle for the open waveform-audio output device
- _hWaveOut = hWaveOut;
-
- // Store the input wave header as well
- _waveFormatOut = waveFormat;
-
- // Prepare wave-out headers
- //
- const WebRtc_UWord8 bytesPerSample = 2*_playChannels;
-
- for (int n = 0; n < N_BUFFERS_OUT; n++)
- {
- // set up the output wave header
- _waveHeaderOut[n].lpData = _playBuffer[n];
- _waveHeaderOut[n].dwBufferLength = bytesPerSample*PLAY_BUF_SIZE_IN_SAMPLES;
- _waveHeaderOut[n].dwFlags = 0;
- _waveHeaderOut[n].dwLoops = 0;
-
- memset(_playBuffer[n], 0, bytesPerSample*PLAY_BUF_SIZE_IN_SAMPLES);
-
- // The waveOutPrepareHeader function prepares a waveform-audio data block for playback.
- // The lpData, dwBufferLength, and dwFlags members of the WAVEHDR structure must be set
- // before calling this function.
- //
- res = waveOutPrepareHeader(_hWaveOut, &_waveHeaderOut[n], sizeof(WAVEHDR));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutPrepareHeader(%d) failed (err=%d)", n, res);
- TraceWaveOutError(res);
- }
-
- // perform extra check to ensure that the header is prepared
- if (_waveHeaderOut[n].dwFlags != WHDR_PREPARED)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutPrepareHeader(%d) failed (dwFlags != WHDR_PREPARED)", n);
- }
- }
-
- // Mark playout side as initialized
- _playIsInitialized = true;
-
- _dTcheckPlayBufDelay = 10; // check playback buffer delay every 10 ms
- _playBufCount = 0; // index of active output wave header (<=> output buffer index)
- _playBufDelay = 80; // buffer delay/size is initialized to 80 ms and slowly decreased until er < 25
- _minPlayBufDelay = 25; // minimum playout buffer delay
- _MAX_minBuffer = 65; // adaptive minimum playout buffer delay cannot be larger than this value
- _intro = 1; // Used to make sure that adaption starts after (2000-1700)/100 seconds
- _waitCounter = 1700; // Counter for start of adaption of playback buffer
- _erZeroCounter = 0; // Log how many times er = 0 in consequtive calls to RecTimeProc
- _useHeader = 0; // Counts number of "useHeader" detections. Stops at 2.
-
- _writtenSamples = 0;
- _writtenSamplesOld = 0;
- _playedSamplesOld = 0;
- _sndCardPlayDelay = 0;
- _sndCardRecDelay = 0;
-
- WEBRTC_TRACE(kTraceInfo, kTraceUtility, _id,"initial playout status: _playBufDelay=%d, _minPlayBufDelay=%d",
- _playBufDelay, _minPlayBufDelay);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// InitRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::InitRecording()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_recording)
- {
- return -1;
- }
-
- if (!_inputDeviceIsSpecified)
- {
- return -1;
- }
-
- if (_recIsInitialized)
- {
- return 0;
- }
-
- _avgCPULoad = 0;
- _playAcc = 0;
-
- // Initialize the microphone (devices might have been added or removed)
- if (InitMicrophone() == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "InitMicrophone() failed");
- }
-
- // Enumerate all availiable input devices
- EnumerateRecordingDevices();
-
- // Start by closing any existing wave-input devices
- //
- MMRESULT res(MMSYSERR_ERROR);
-
- if (_hWaveIn != NULL)
- {
- res = waveInClose(_hWaveIn);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInClose() failed (err=%d)", res);
- TraceWaveInError(res);
- }
- }
-
- // Set the input wave format
- //
- WAVEFORMATEX waveFormat;
-
- waveFormat.wFormatTag = WAVE_FORMAT_PCM;
- waveFormat.nChannels = _recChannels; // mono <=> 1, stereo <=> 2
- waveFormat.nSamplesPerSec = N_REC_SAMPLES_PER_SEC;
- waveFormat.wBitsPerSample = 16;
- waveFormat.nBlockAlign = waveFormat.nChannels * (waveFormat.wBitsPerSample/8);
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
- waveFormat.cbSize = 0;
-
- // Open the given waveform-audio input device for recording
- //
- HWAVEIN hWaveIn(NULL);
-
- if (IsUsingInputDeviceIndex())
- {
- // verify settings first
- res = waveInOpen(NULL, _inputDeviceIndex, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_FORMAT_QUERY);
- if (MMSYSERR_NOERROR == res)
- {
- // open the given waveform-audio input device for recording
- res = waveInOpen(&hWaveIn, _inputDeviceIndex, &waveFormat, 0, 0, CALLBACK_NULL);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening input device corresponding to device ID %u", _inputDeviceIndex);
- }
- }
- else
- {
- if (_inputDevice == AudioDeviceModule::kDefaultCommunicationDevice)
- {
- // check if it is possible to open the default communication device (supported on Windows 7)
- res = waveInOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE | WAVE_FORMAT_QUERY);
- if (MMSYSERR_NOERROR == res)
- {
- // if so, open the default communication device for real
- res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_DEFAULT_COMMUNICATION_DEVICE);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default communication device");
- }
- else
- {
- // use default device since default communication device was not avaliable
- res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "unable to open default communication device => using default instead");
- }
- }
- else if (_inputDevice == AudioDeviceModule::kDefaultDevice)
- {
- // open default device since it has been requested
- res = waveInOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_FORMAT_QUERY);
- if (MMSYSERR_NOERROR == res)
- {
- res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default input device");
- }
- }
- }
-
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveInOpen() failed (err=%d)", res);
- TraceWaveInError(res);
- return -1;
- }
-
- // Log information about the aquired input device
- //
- WAVEINCAPS caps;
-
- res = waveInGetDevCaps((UINT_PTR)hWaveIn, &caps, sizeof(WAVEINCAPS));
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetDevCaps() failed (err=%d)", res);
- TraceWaveInError(res);
- }
-
- UINT deviceID(0);
- res = waveInGetID(hWaveIn, &deviceID);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetID() failed (err=%d)", res);
- TraceWaveInError(res);
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "utilized device ID : %u", deviceID);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", caps.szPname);
-
- // Store valid handle for the open waveform-audio input device
- _hWaveIn = hWaveIn;
-
- // Store the input wave header as well
- _waveFormatIn = waveFormat;
-
- // Mark recording side as initialized
- _recIsInitialized = true;
-
- _recBufCount = 0; // index of active input wave header (<=> input buffer index)
- _recDelayCount = 0; // ensures that input buffers are returned with certain delay
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StartRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::StartRecording()
-{
-
- if (!_recIsInitialized)
- {
- return -1;
- }
-
- if (_recording)
- {
- return 0;
- }
-
- // set state to ensure that the recording starts from the audio thread
- _startRec = true;
-
- // the audio thread will signal when recording has stopped
- if (kEventTimeout == _recStartEvent.Wait(10000))
- {
- _startRec = false;
- StopRecording();
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to activate recording");
- return -1;
- }
-
- if (_recording)
- {
- // the recording state is set by the audio thread after recording has started
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to activate recording");
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StopRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::StopRecording()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_recIsInitialized)
- {
- return 0;
- }
-
- if (_hWaveIn == NULL)
- {
- return -1;
- }
-
- bool wasRecording = _recording;
- _recIsInitialized = false;
- _recording = false;
-
- MMRESULT res;
-
- // Stop waveform-adio input. If there are any buffers in the queue, the
- // current buffer will be marked as done (the dwBytesRecorded member in
- // the header will contain the length of data), but any empty buffers in
- // the queue will remain there.
- //
- res = waveInStop(_hWaveIn);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInStop() failed (err=%d)", res);
- TraceWaveInError(res);
- }
-
- // Stop input on the given waveform-audio input device and resets the current
- // position to zero. All pending buffers are marked as done and returned to
- // the application.
- //
- res = waveInReset(_hWaveIn);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInReset() failed (err=%d)", res);
- TraceWaveInError(res);
- }
-
- // Clean up the preparation performed by the waveInPrepareHeader function.
- // Only unprepare header if recording was ever started (and headers are prepared).
- //
- if (wasRecording)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "waveInUnprepareHeader() will be performed");
- for (int n = 0; n < N_BUFFERS_IN; n++)
- {
- res = waveInUnprepareHeader(_hWaveIn, &_waveHeaderIn[n], sizeof(WAVEHDR));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInUnprepareHeader() failed (err=%d)", res);
- TraceWaveInError(res);
- }
- }
- }
-
- // Close the given waveform-audio input device.
- //
- res = waveInClose(_hWaveIn);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInClose() failed (err=%d)", res);
- TraceWaveInError(res);
- }
-
- // Set the wave input handle to NULL
- //
- _hWaveIn = NULL;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_hWaveIn is now set to NULL");
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::RecordingIsInitialized() const
-{
- return (_recIsInitialized);
-}
-
-// ----------------------------------------------------------------------------
-// Recording
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::Recording() const
-{
- return (_recording);
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::PlayoutIsInitialized() const
-{
- return (_playIsInitialized);
-}
-
-// ----------------------------------------------------------------------------
-// StartPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::StartPlayout()
-{
-
- if (!_playIsInitialized)
- {
- return -1;
- }
-
- if (_playing)
- {
- return 0;
- }
-
- // set state to ensure that playout starts from the audio thread
- _startPlay = true;
-
- // the audio thread will signal when recording has started
- if (kEventTimeout == _playStartEvent.Wait(10000))
- {
- _startPlay = false;
- StopPlayout();
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to activate playout");
- return -1;
- }
-
- if (_playing)
- {
- // the playing state is set by the audio thread after playout has started
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to activate playing");
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StopPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::StopPlayout()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (!_playIsInitialized)
- {
- return 0;
- }
-
- if (_hWaveOut == NULL)
- {
- return -1;
- }
-
- _playIsInitialized = false;
- _playing = false;
- _sndCardPlayDelay = 0;
- _sndCardRecDelay = 0;
-
- MMRESULT res;
-
- // The waveOutReset function stops playback on the given waveform-audio
- // output device and resets the current position to zero. All pending
- // playback buffers are marked as done (WHDR_DONE) and returned to the application.
- // After this function returns, the application can send new playback buffers
- // to the device by calling waveOutWrite, or close the device by calling waveOutClose.
- //
- res = waveOutReset(_hWaveOut);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutReset() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
-
- // The waveOutUnprepareHeader function cleans up the preparation performed
- // by the waveOutPrepareHeader function. This function must be called after
- // the device driver is finished with a data block.
- // You must call this function before freeing the buffer.
- //
- for (int n = 0; n < N_BUFFERS_OUT; n++)
- {
- res = waveOutUnprepareHeader(_hWaveOut, &_waveHeaderOut[n], sizeof(WAVEHDR));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutUnprepareHeader() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
- }
-
- // The waveOutClose function closes the given waveform-audio output device.
- // The close operation fails if the device is still playing a waveform-audio
- // buffer that was previously sent by calling waveOutWrite. Before calling
- // waveOutClose, the application must wait for all buffers to finish playing
- // or call the waveOutReset function to terminate playback.
- //
- res = waveOutClose(_hWaveOut);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutClose() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
-
- _hWaveOut = NULL;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "_hWaveOut is now set to NULL");
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutDelay
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::PlayoutDelay(WebRtc_UWord16& delayMS) const
-{
- CriticalSectionScoped lock(_critSect);
- delayMS = (WebRtc_UWord16)_sndCardPlayDelay;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// RecordingDelay
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::RecordingDelay(WebRtc_UWord16& delayMS) const
-{
- CriticalSectionScoped lock(_critSect);
- delayMS = (WebRtc_UWord16)_sndCardRecDelay;
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Playing
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::Playing() const
-{
- return (_playing);
-}
-// ----------------------------------------------------------------------------
-// SetPlayoutBuffer
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::SetPlayoutBuffer(const AudioDeviceModule::BufferType type, WebRtc_UWord16 sizeMS)
-{
- CriticalSectionScoped lock(_critSect);
- _playBufType = type;
- if (type == AudioDeviceModule::kFixedBufferSize)
- {
- _playBufDelayFixed = sizeMS;
- }
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutBuffer
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::PlayoutBuffer(AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const
-{
- CriticalSectionScoped lock(_critSect);
- type = _playBufType;
- if (type == AudioDeviceModule::kFixedBufferSize)
- {
- sizeMS = _playBufDelayFixed;
- }
- else
- {
- sizeMS = _playBufDelay;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// CPULoad
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::CPULoad(WebRtc_UWord16& load) const
-{
-
- load = static_cast<WebRtc_UWord16>(100*_avgCPULoad);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutWarning
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::PlayoutWarning() const
-{
- return ( _playWarning > 0);
-}
-
-// ----------------------------------------------------------------------------
-// PlayoutError
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::PlayoutError() const
-{
- return ( _playError > 0);
-}
-
-// ----------------------------------------------------------------------------
-// RecordingWarning
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::RecordingWarning() const
-{
- return ( _recWarning > 0);
-}
-
-// ----------------------------------------------------------------------------
-// RecordingError
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::RecordingError() const
-{
- return ( _recError > 0);
-}
-
-// ----------------------------------------------------------------------------
-// ClearPlayoutWarning
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsWave::ClearPlayoutWarning()
-{
- _playWarning = 0;
-}
-
-// ----------------------------------------------------------------------------
-// ClearPlayoutError
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsWave::ClearPlayoutError()
-{
- _playError = 0;
-}
-
-// ----------------------------------------------------------------------------
-// ClearRecordingWarning
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsWave::ClearRecordingWarning()
-{
- _recWarning = 0;
-}
-
-// ----------------------------------------------------------------------------
-// ClearRecordingError
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsWave::ClearRecordingError()
-{
- _recError = 0;
-}
-
-// ============================================================================
-// Private Methods
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// InputSanityCheckAfterUnlockedPeriod
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::InputSanityCheckAfterUnlockedPeriod() const
-{
- if (_hWaveIn == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "input state has been modified during unlocked period");
- return -1;
- }
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// OutputSanityCheckAfterUnlockedPeriod
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::OutputSanityCheckAfterUnlockedPeriod() const
-{
- if (_hWaveOut == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "output state has been modified during unlocked period");
- return -1;
- }
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// EnumeratePlayoutDevices
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::EnumeratePlayoutDevices()
-{
-
- WebRtc_UWord16 nDevices(PlayoutDevices());
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#output devices: %u", nDevices);
-
- WAVEOUTCAPS caps;
- MMRESULT res;
-
- for (UINT deviceID = 0; deviceID < nDevices; deviceID++)
- {
- res = waveOutGetDevCaps(deviceID, &caps, sizeof(WAVEOUTCAPS));
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetDevCaps() failed (err=%d)", res);
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Device ID %u:", deviceID);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "manufacturer ID : %u", caps.wMid);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product ID : %u",caps.wPid);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "version of driver : %u.%u", HIBYTE(caps.vDriverVersion), LOBYTE(caps.vDriverVersion));
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", caps.szPname);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "dwFormats : 0x%x", caps.dwFormats);
- if (caps.dwFormats & WAVE_FORMAT_48S16)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " 48kHz,stereo,16bit : SUPPORTED");
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " 48kHz,stereo,16bit : *NOT* SUPPORTED");
- }
- if (caps.dwFormats & WAVE_FORMAT_48M16)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " 48kHz,mono,16bit : SUPPORTED");
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " 48kHz,mono,16bit : *NOT* SUPPORTED");
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wChannels : %u", caps.wChannels);
- TraceSupportFlags(caps.dwSupport);
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// EnumerateRecordingDevices
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::EnumerateRecordingDevices()
-{
-
- WebRtc_UWord16 nDevices(RecordingDevices());
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#input devices: %u", nDevices);
-
- WAVEINCAPS caps;
- MMRESULT res;
-
- for (UINT deviceID = 0; deviceID < nDevices; deviceID++)
- {
- res = waveInGetDevCaps(deviceID, &caps, sizeof(WAVEINCAPS));
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetDevCaps() failed (err=%d)", res);
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Device ID %u:", deviceID);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "manufacturer ID : %u", caps.wMid);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product ID : %u",caps.wPid);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "version of driver : %u.%u", HIBYTE(caps.vDriverVersion), LOBYTE(caps.vDriverVersion));
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", caps.szPname);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "dwFormats : 0x%x", caps.dwFormats);
- if (caps.dwFormats & WAVE_FORMAT_48S16)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " 48kHz,stereo,16bit : SUPPORTED");
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " 48kHz,stereo,16bit : *NOT* SUPPORTED");
- }
- if (caps.dwFormats & WAVE_FORMAT_48M16)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " 48kHz,mono,16bit : SUPPORTED");
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, " 48kHz,mono,16bit : *NOT* SUPPORTED");
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "wChannels : %u", caps.wChannels);
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// TraceSupportFlags
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsWave::TraceSupportFlags(DWORD dwSupport) const
-{
- TCHAR buf[256];
-
- StringCchPrintf(buf, 128, TEXT("support flags : 0x%x "), dwSupport);
-
- if (dwSupport & WAVECAPS_PITCH)
- {
- // supports pitch control
- StringCchCat(buf, 256, TEXT("(PITCH)"));
- }
- if (dwSupport & WAVECAPS_PLAYBACKRATE)
- {
- // supports playback rate control
- StringCchCat(buf, 256, TEXT("(PLAYBACKRATE)"));
- }
- if (dwSupport & WAVECAPS_VOLUME)
- {
- // supports volume control
- StringCchCat(buf, 256, TEXT("(VOLUME)"));
- }
- if (dwSupport & WAVECAPS_LRVOLUME)
- {
- // supports separate left and right volume control
- StringCchCat(buf, 256, TEXT("(LRVOLUME)"));
- }
- if (dwSupport & WAVECAPS_SYNC)
- {
- // the driver is synchronous and will block while playing a buffer
- StringCchCat(buf, 256, TEXT("(SYNC)"));
- }
- if (dwSupport & WAVECAPS_SAMPLEACCURATE)
- {
- // returns sample-accurate position information
- StringCchCat(buf, 256, TEXT("(SAMPLEACCURATE)"));
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", buf);
-}
-
-// ----------------------------------------------------------------------------
-// TraceWaveInError
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsWave::TraceWaveInError(MMRESULT error) const
-{
- TCHAR buf[MAXERRORLENGTH];
- TCHAR msg[MAXERRORLENGTH];
-
- StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
- waveInGetErrorText(error, msg, MAXERRORLENGTH);
- StringCchCat(buf, MAXERRORLENGTH, msg);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", buf);
-}
-
-// ----------------------------------------------------------------------------
-// TraceWaveOutError
-// ----------------------------------------------------------------------------
-
-void AudioDeviceWindowsWave::TraceWaveOutError(MMRESULT error) const
-{
- TCHAR buf[MAXERRORLENGTH];
- TCHAR msg[MAXERRORLENGTH];
-
- StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
- waveOutGetErrorText(error, msg, MAXERRORLENGTH);
- StringCchCat(buf, MAXERRORLENGTH, msg);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", buf);
-}
-
-// ----------------------------------------------------------------------------
-// PrepareStartPlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::PrepareStartPlayout()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_hWaveOut == NULL)
- {
- return -1;
- }
-
- // A total of 30ms of data is immediately placed in the SC buffer
- //
- WebRtc_Word8 zeroVec[4*PLAY_BUF_SIZE_IN_SAMPLES]; // max allocation
- memset(zeroVec, 0, 4*PLAY_BUF_SIZE_IN_SAMPLES);
-
- {
- Write(zeroVec, PLAY_BUF_SIZE_IN_SAMPLES);
- Write(zeroVec, PLAY_BUF_SIZE_IN_SAMPLES);
- Write(zeroVec, PLAY_BUF_SIZE_IN_SAMPLES);
- }
-
- _playAcc = 0;
- _playWarning = 0;
- _playError = 0;
- _dc_diff_mean = 0;
- _dc_y_prev = 0;
- _dc_penalty_counter = 20;
- _dc_prevtime = 0;
- _dc_prevplay = 0;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// PrepareStartRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::PrepareStartRecording()
-{
-
- CriticalSectionScoped lock(_critSect);
-
- if (_hWaveIn == NULL)
- {
- return -1;
- }
-
- _playAcc = 0;
- _recordedBytes = 0;
- _recPutBackDelay = REC_PUT_BACK_DELAY;
-
- MMRESULT res;
- MMTIME mmtime;
- mmtime.wType = TIME_SAMPLES;
-
- res = waveInGetPosition(_hWaveIn, &mmtime, sizeof(mmtime));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetPosition(TIME_SAMPLES) failed (err=%d)", res);
- TraceWaveInError(res);
- }
-
- _read_samples = mmtime.u.sample;
- _read_samples_old = _read_samples;
- _rec_samples_old = mmtime.u.sample;
- _wrapCounter = 0;
-
- for (int n = 0; n < N_BUFFERS_IN; n++)
- {
- const WebRtc_UWord8 nBytesPerSample = 2*_recChannels;
-
- // set up the input wave header
- _waveHeaderIn[n].lpData = _recBuffer[n];
- _waveHeaderIn[n].dwBufferLength = nBytesPerSample * REC_BUF_SIZE_IN_SAMPLES;
- _waveHeaderIn[n].dwFlags = 0;
- _waveHeaderIn[n].dwBytesRecorded = 0;
- _waveHeaderIn[n].dwUser = 0;
-
- memset(_recBuffer[n], 0, nBytesPerSample * REC_BUF_SIZE_IN_SAMPLES);
-
- // prepare a buffer for waveform-audio input
- res = waveInPrepareHeader(_hWaveIn, &_waveHeaderIn[n], sizeof(WAVEHDR));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInPrepareHeader(%d) failed (err=%d)", n, res);
- TraceWaveInError(res);
- }
-
- // send an input buffer to the given waveform-audio input device
- res = waveInAddBuffer(_hWaveIn, &_waveHeaderIn[n], sizeof(WAVEHDR));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInAddBuffer(%d) failed (err=%d)", n, res);
- TraceWaveInError(res);
- }
- }
-
- // start input on the given waveform-audio input device
- res = waveInStart(_hWaveIn);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInStart() failed (err=%d)", res);
- TraceWaveInError(res);
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// GetPlayoutBufferDelay
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::GetPlayoutBufferDelay(WebRtc_UWord32& writtenSamples, WebRtc_UWord32& playedSamples)
-{
- int i;
- int ms_Header;
- long playedDifference;
- int msecInPlayoutBuffer(0); // #milliseconds of audio in the playout buffer
-
- const WebRtc_UWord16 nSamplesPerMs = (WebRtc_UWord16)(N_PLAY_SAMPLES_PER_SEC/1000); // default is 48000/1000 = 48
-
- MMRESULT res;
- MMTIME mmtime;
-
- if (!_playing)
- {
- playedSamples = 0;
- return (0);
- }
-
- // Retrieve the current playback position.
- //
- mmtime.wType = TIME_SAMPLES; // number of waveform-audio samples
- res = waveOutGetPosition(_hWaveOut, &mmtime, sizeof(mmtime));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutGetPosition() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
-
- writtenSamples = _writtenSamples; // #samples written to the playout buffer
- playedSamples = mmtime.u.sample; // current playout position in the playout buffer
-
- // derive remaining amount (in ms) of data in the playout buffer
- msecInPlayoutBuffer = ((writtenSamples - playedSamples)/nSamplesPerMs);
- // DEBUG_PRINTP("msecInPlayoutBuffer=%u\n", msecInPlayoutBuffer);
-
- playedDifference = (long) (_playedSamplesOld - playedSamples);
-
- if (playedDifference > 64000)
- {
- // If the sound cards number-of-played-out-samples variable wraps around before
- // written_sampels wraps around this needs to be adjusted. This can happen on
- // sound cards that uses less than 32 bits to keep track of number of played out
- // sampels. To avoid being fooled by sound cards that sometimes produces false
- // output we compare old value minus the new value with a large value. This is
- // neccessary because some SC:s produce an output like 153, 198, 175, 230 which
- // would trigger the wrap-around function if we didn't compare with a large value.
- // The value 64000 is chosen because 2^16=65536 so we allow wrap around at 16 bits.
-
- i = 31;
- while((_playedSamplesOld <= (unsigned long)POW2(i)) && (i > 14)) {
- i--;
- }
-
- if((i < 31) && (i > 14)) {
- // Avoid adjusting when there is 32-bit wrap-around since that is
- // something neccessary.
- //
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "msecleft() => wrap around occured: %d bits used by sound card)", (i+1));
-
- _writtenSamples = _writtenSamples - POW2(i + 1);
- writtenSamples = _writtenSamples;
- msecInPlayoutBuffer = ((writtenSamples - playedSamples)/nSamplesPerMs);
- }
- }
- else if ((_writtenSamplesOld > POW2(31)) && (writtenSamples < 96000))
- {
- // Wrap around as expected after having used all 32 bits. (But we still
- // test if the wrap around happened earlier which it should not)
-
- i = 31;
- while (_writtenSamplesOld <= (unsigned long)POW2(i)) {
- i--;
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, " msecleft() (wrap around occured after having used all 32 bits)");
-
- _writtenSamplesOld = writtenSamples;
- _playedSamplesOld = playedSamples;
- msecInPlayoutBuffer = (int)((writtenSamples + POW2(i + 1) - playedSamples)/nSamplesPerMs);
-
- }
- else if ((writtenSamples < 96000) && (playedSamples > POW2(31)))
- {
- // Wrap around has, as expected, happened for written_sampels before
- // playedSampels so we have to adjust for this until also playedSampels
- // has had wrap around.
-
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, " msecleft() (wrap around occured: correction of output is done)");
-
- _writtenSamplesOld = writtenSamples;
- _playedSamplesOld = playedSamples;
- msecInPlayoutBuffer = (int)((writtenSamples + POW2(32) - playedSamples)/nSamplesPerMs);
- }
-
- _writtenSamplesOld = writtenSamples;
- _playedSamplesOld = playedSamples;
-
-
- // We use the following formaula to track that playout works as it should
- // y=playedSamples/48 - timeGetTime();
- // y represent the clock drift between system clock and sound card clock - should be fairly stable
- // When the exponential mean value of diff(y) goes away from zero something is wrong
- // The exponential formula will accept 1% clock drift but not more
- // The driver error means that we will play to little audio and have a high negative clock drift
- // We kick in our alternative method when the clock drift reaches 20%
-
- int diff,y;
- int unsigned time =0;
-
- // If we have other problems that causes playout glitches
- // we don't want to switch playout method.
- // Check if playout buffer is extremely low, or if we haven't been able to
- // exectue our code in more than 40 ms
-
- time = timeGetTime();
-
- if ((msecInPlayoutBuffer < 20) || (time - _dc_prevtime > 40))
- {
- _dc_penalty_counter = 100;
- }
-
- if ((playedSamples != 0))
- {
- y = playedSamples/48 - time;
- if ((_dc_y_prev != 0) && (_dc_penalty_counter == 0))
- {
- diff = y - _dc_y_prev;
- _dc_diff_mean = (990*_dc_diff_mean)/1000 + 10*diff;
- }
- _dc_y_prev = y;
- }
-
- if (_dc_penalty_counter)
- {
- _dc_penalty_counter--;
- }
-
- if (_dc_diff_mean < -200)
- {
- // Always reset the filter
- _dc_diff_mean = 0;
-
- // Problem is detected. Switch delay method and set min buffer to 80.
- // Reset the filter and keep monitoring the filter output.
- // If issue is detected a second time, increase min buffer to 100.
- // If that does not help, we must modify this scheme further.
-
- _useHeader++;
- if (_useHeader == 1)
- {
- _minPlayBufDelay = 80;
- _playWarning = 1; // only warn first time
- WEBRTC_TRACE(kTraceInfo, kTraceUtility, -1, "Modification #1: _useHeader = %d, _minPlayBufDelay = %d", _useHeader, _minPlayBufDelay);
- }
- else if (_useHeader == 2)
- {
- _minPlayBufDelay = 100; // add some more safety
- WEBRTC_TRACE(kTraceInfo, kTraceUtility, -1, "Modification #2: _useHeader = %d, _minPlayBufDelay = %d", _useHeader, _minPlayBufDelay);
- }
- else
- {
- // This state should not be entered... (HA)
- WEBRTC_TRACE (kTraceWarning, kTraceUtility, -1, "further actions are required!");
- }
- if (_playWarning == 1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending playout warning exists");
- }
- _playWarning = 1; // triggers callback from module process thread
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "kPlayoutWarning message posted: switching to alternative playout delay method");
- }
- _dc_prevtime = time;
- _dc_prevplay = playedSamples;
-
- // Try a very rough method of looking at how many buffers are still playing
- ms_Header = 0;
- for (i = 0; i < N_BUFFERS_OUT; i++) {
- if ((_waveHeaderOut[i].dwFlags & WHDR_INQUEUE)!=0) {
- ms_Header += 10;
- }
- }
-
- if ((ms_Header-50) > msecInPlayoutBuffer) {
- // Test for cases when GetPosition appears to be screwed up (currently just log....)
- TCHAR infoStr[300];
- if (_no_of_msecleft_warnings%20==0)
- {
- StringCchPrintf(infoStr, 300, TEXT("writtenSamples=%i, playedSamples=%i, msecInPlayoutBuffer=%i, ms_Header=%i"), writtenSamples, playedSamples, msecInPlayoutBuffer, ms_Header);
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, (const char*)infoStr);
- }
- _no_of_msecleft_warnings++;
- }
-
- // If this is true we have had a problem with the playout
- if (_useHeader > 0)
- {
- return (ms_Header);
- }
-
-
- if (ms_Header < msecInPlayoutBuffer)
- {
- if (_no_of_msecleft_warnings % 100 == 0)
- {
- TCHAR str[300];
- StringCchPrintf(str, 300, TEXT("_no_of_msecleft_warnings=%i, msecInPlayoutBuffer=%i ms_Header=%i (minBuffer=%i buffersize=%i writtenSamples=%i playedSamples=%i)"),
- _no_of_msecleft_warnings, msecInPlayoutBuffer, ms_Header, _minPlayBufDelay, _playBufDelay, writtenSamples, playedSamples);
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, (const char*)str);
- }
- _no_of_msecleft_warnings++;
- ms_Header -= 6; // Round off as we only have 10ms resolution + Header info is usually slightly delayed compared to GetPosition
-
- if (ms_Header < 0)
- ms_Header = 0;
-
- return (ms_Header);
- }
- else
- {
- return (msecInPlayoutBuffer);
- }
-}
-
-// ----------------------------------------------------------------------------
-// GetRecordingBufferDelay
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::GetRecordingBufferDelay(WebRtc_UWord32& readSamples, WebRtc_UWord32& recSamples)
-{
- long recDifference;
- MMTIME mmtime;
- MMRESULT mmr;
-
- const WebRtc_UWord16 nSamplesPerMs = (WebRtc_UWord16)(N_REC_SAMPLES_PER_SEC/1000); // default is 48000/1000 = 48
-
- // Retrieve the current input position of the given waveform-audio input device
- //
- mmtime.wType = TIME_SAMPLES;
- mmr = waveInGetPosition(_hWaveIn, &mmtime, sizeof(mmtime));
- if (MMSYSERR_NOERROR != mmr)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInGetPosition() failed (err=%d)", mmr);
- TraceWaveInError(mmr);
- }
-
- readSamples = _read_samples; // updated for each full fram in RecProc()
- recSamples = mmtime.u.sample; // remaining time in input queue (recorded but not read yet)
-
- recDifference = (long) (_rec_samples_old - recSamples);
-
- if( recDifference > 64000) {
- WEBRTC_TRACE (kTraceDebug, kTraceUtility, -1,"WRAP 1 (recDifference =%d)", recDifference);
- // If the sound cards number-of-recorded-samples variable wraps around before
- // read_sampels wraps around this needs to be adjusted. This can happen on
- // sound cards that uses less than 32 bits to keep track of number of played out
- // sampels. To avoid being fooled by sound cards that sometimes produces false
- // output we compare old value minus the new value with a large value. This is
- // neccessary because some SC:s produce an output like 153, 198, 175, 230 which
- // would trigger the wrap-around function if we didn't compare with a large value.
- // The value 64000 is chosen because 2^16=65536 so we allow wrap around at 16 bits.
- //
- int i = 31;
- while((_rec_samples_old <= (unsigned long)POW2(i)) && (i > 14))
- i--;
-
- if((i < 31) && (i > 14)) {
- // Avoid adjusting when there is 32-bit wrap-around since that is
- // somethying neccessary.
- //
- _read_samples = _read_samples - POW2(i + 1);
- readSamples = _read_samples;
- _wrapCounter++;
- } else {
- WEBRTC_TRACE (kTraceWarning, kTraceUtility, -1,"AEC (_rec_samples_old %d recSamples %d)",_rec_samples_old, recSamples);
- }
- }
-
- if((_wrapCounter>200)){
- // Do nothing, handled later
- }
- else if((_rec_samples_old > POW2(31)) && (recSamples < 96000)) {
- WEBRTC_TRACE (kTraceDebug, kTraceUtility, -1,"WRAP 2 (_rec_samples_old %d recSamples %d)",_rec_samples_old, recSamples);
- // Wrap around as expected after having used all 32 bits.
- _read_samples_old = readSamples;
- _rec_samples_old = recSamples;
- _wrapCounter++;
- return (int)((recSamples + POW2(32) - readSamples)/nSamplesPerMs);
-
-
- } else if((recSamples < 96000) && (readSamples > POW2(31))) {
- WEBRTC_TRACE (kTraceDebug, kTraceUtility, -1,"WRAP 3 (readSamples %d recSamples %d)",readSamples, recSamples);
- // Wrap around has, as expected, happened for rec_sampels before
- // readSampels so we have to adjust for this until also readSampels
- // has had wrap around.
- _read_samples_old = readSamples;
- _rec_samples_old = recSamples;
- _wrapCounter++;
- return (int)((recSamples + POW2(32) - readSamples)/nSamplesPerMs);
- }
-
- _read_samples_old = _read_samples;
- _rec_samples_old = recSamples;
- int res=(((int)_rec_samples_old - (int)_read_samples_old)/nSamplesPerMs);
-
- if((res > 2000)||(res < 0)||(_wrapCounter>200)){
- // Reset everything
- WEBRTC_TRACE (kTraceWarning, kTraceUtility, -1,"msec_read error (res %d wrapCounter %d)",res, _wrapCounter);
- MMTIME mmtime;
- mmtime.wType = TIME_SAMPLES;
-
- mmr=waveInGetPosition(_hWaveIn, &mmtime, sizeof(mmtime));
- if (mmr != MMSYSERR_NOERROR) {
- WEBRTC_TRACE (kTraceWarning, kTraceUtility, -1, "waveInGetPosition failed (mmr=%d)", mmr);
- }
- _read_samples=mmtime.u.sample;
- _read_samples_old=_read_samples;
- _rec_samples_old=mmtime.u.sample;
-
- // Guess a decent value
- res = 20;
- }
-
- _wrapCounter = 0;
- return res;
-}
-
-// ============================================================================
-// Thread Methods
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// ThreadFunc
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::ThreadFunc(void* pThis)
-{
- return (static_cast<AudioDeviceWindowsWave*>(pThis)->ThreadProcess());
-}
-
-// ----------------------------------------------------------------------------
-// ThreadProcess
-// ----------------------------------------------------------------------------
-
-bool AudioDeviceWindowsWave::ThreadProcess()
-{
- WebRtc_UWord32 time(0);
- WebRtc_UWord32 playDiff(0);
- WebRtc_UWord32 recDiff(0);
-
- LONGLONG playTime(0);
- LONGLONG recTime(0);
-
- switch (_timeEvent.Wait(1000))
- {
- case kEventSignaled:
- break;
- case kEventError:
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "EventWrapper::Wait() failed => restarting timer");
- _timeEvent.StopTimer();
- _timeEvent.StartTimer(true, TIMER_PERIOD_MS);
- return true;
- case kEventTimeout:
- return true;
- }
-
- time = AudioDeviceUtility::GetTimeInMS();
-
- if (_startPlay)
- {
- if (PrepareStartPlayout() == 0)
- {
- _prevTimerCheckTime = time;
- _prevPlayTime = time;
- _startPlay = false;
- _playing = true;
- _playStartEvent.Set();
- }
- }
-
- if (_startRec)
- {
- if (PrepareStartRecording() == 0)
- {
- _prevTimerCheckTime = time;
- _prevRecTime = time;
- _prevRecByteCheckTime = time;
- _startRec = false;
- _recording = true;
- _recStartEvent.Set();
- }
- }
-
- if (_playing)
- {
- playDiff = time - _prevPlayTime;
- }
-
- if (_recording)
- {
- recDiff = time - _prevRecTime;
- }
-
- if (_playing || _recording)
- {
- RestartTimerIfNeeded(time);
- }
-
- if (_playing &&
- (playDiff > (WebRtc_UWord32)(_dTcheckPlayBufDelay - 1)) ||
- (playDiff < 0))
- {
- Lock();
- if (_playing)
- {
- if (PlayProc(playTime) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "PlayProc() failed");
- }
- _prevPlayTime = time;
- if (playTime != 0)
- _playAcc += playTime;
- }
- UnLock();
- }
-
- if (_playing && (playDiff > 12))
- {
- // It has been a long time since we were able to play out, try to
- // compensate by calling PlayProc again.
- //
- Lock();
- if (_playing)
- {
- if (PlayProc(playTime))
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "PlayProc() failed");
- }
- _prevPlayTime = time;
- if (playTime != 0)
- _playAcc += playTime;
- }
- UnLock();
- }
-
- if (_recording &&
- (recDiff > REC_CHECK_TIME_PERIOD_MS) ||
- (recDiff < 0))
- {
- Lock();
- if (_recording)
- {
- WebRtc_Word32 nRecordedBytes(0);
- WebRtc_UWord16 maxIter(10);
-
- // Deliver all availiable recorded buffers and update the CPU load measurement.
- // We use a while loop here to compensate for the fact that the multi-media timer
- // can sometimed enter a "bad state" after hibernation where the resolution is
- // reduced from ~1ms to ~10-15 ms.
- //
- while ((nRecordedBytes = RecProc(recTime)) > 0)
- {
- maxIter--;
- _recordedBytes += nRecordedBytes;
- if (recTime && _perfFreq.QuadPart)
- {
- // Measure the average CPU load:
- // This is a simplified expression where an exponential filter is used:
- // _avgCPULoad = 0.99 * _avgCPULoad + 0.01 * newCPU,
- // newCPU = (recTime+playAcc)/f is time in seconds
- // newCPU / 0.01 is the fraction of a 10 ms period
- // The two 0.01 cancels each other.
- // NOTE - assumes 10ms audio buffers.
- //
- _avgCPULoad = (float)(_avgCPULoad*.99 + (recTime+_playAcc)/(double)(_perfFreq.QuadPart));
- _playAcc = 0;
- }
- if (maxIter == 0)
- {
- // If we get this message ofte, our compensation scheme is not sufficient.
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "failed to compensate for reduced MM-timer resolution");
- }
- }
-
- if (nRecordedBytes == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "RecProc() failed");
- }
-
- _prevRecTime = time;
-
- // Monitor the recording process and generate error/warning callbacks if needed
- MonitorRecording(time);
- }
- UnLock();
- }
-
- if (!_recording)
- {
- _prevRecByteCheckTime = time;
- _avgCPULoad = 0;
- }
-
- return true;
-}
-
-// ----------------------------------------------------------------------------
-// RecProc
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::RecProc(LONGLONG& consumedTime)
-{
- MMRESULT res;
- WebRtc_UWord32 bufCount(0);
- WebRtc_UWord32 nBytesRecorded(0);
-
- consumedTime = 0;
-
- // count modulo N_BUFFERS_IN (0,1,2,...,(N_BUFFERS_IN-1),0,1,2,..)
- if (_recBufCount == N_BUFFERS_IN)
- {
- _recBufCount = 0;
- }
-
- bufCount = _recBufCount;
-
- // take mono/stereo mode into account when deriving size of a full buffer
- const WebRtc_UWord16 bytesPerSample = 2*_recChannels;
- const WebRtc_UWord32 fullBufferSizeInBytes = bytesPerSample * REC_BUF_SIZE_IN_SAMPLES;
-
- // read number of recorded bytes for the given input-buffer
- nBytesRecorded = _waveHeaderIn[bufCount].dwBytesRecorded;
-
- if (nBytesRecorded == fullBufferSizeInBytes ||
- (nBytesRecorded > 0))
- {
- WebRtc_Word32 msecOnPlaySide;
- WebRtc_Word32 msecOnRecordSide;
- WebRtc_UWord32 writtenSamples;
- WebRtc_UWord32 playedSamples;
- WebRtc_UWord32 readSamples, recSamples;
- bool send = true;
-
- WebRtc_UWord32 nSamplesRecorded = (nBytesRecorded/bytesPerSample); // divide by 2 or 4 depending on mono or stereo
-
- if (nBytesRecorded == fullBufferSizeInBytes)
- {
- _timesdwBytes = 0;
- }
- else
- {
- // Test if it is stuck on this buffer
- _timesdwBytes++;
- if (_timesdwBytes < 5)
- {
- // keep trying
- return (0);
- }
- else
- {
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id,"nBytesRecorded=%d => don't use", nBytesRecorded);
- _timesdwBytes = 0;
- send = false;
- }
- }
-
- // store the recorded buffer (no action will be taken if the #recorded samples is not a full buffer)
- _ptrAudioBuffer->SetRecordedBuffer(_waveHeaderIn[bufCount].lpData, nSamplesRecorded);
-
- // update #samples read
- _read_samples += nSamplesRecorded;
-
- // Check how large the playout and recording buffers are on the sound card.
- // This info is needed by the AEC.
- //
- msecOnPlaySide = GetPlayoutBufferDelay(writtenSamples, playedSamples);
- msecOnRecordSide = GetRecordingBufferDelay(readSamples, recSamples);
-
- // If we use the alternative playout delay method, skip the clock drift compensation
- // since it will be an unreliable estimate and might degrade AEC performance.
- WebRtc_Word32 drift = (_useHeader > 0) ? 0 : GetClockDrift(playedSamples, recSamples);
-
- _ptrAudioBuffer->SetVQEData(msecOnPlaySide, msecOnRecordSide, drift);
-
- // Store the play and rec delay values for video synchronization
- _sndCardPlayDelay = msecOnPlaySide;
- _sndCardRecDelay = msecOnRecordSide;
-
- LARGE_INTEGER t1,t2;
-
- if (send)
- {
- QueryPerformanceCounter(&t1);
-
- // deliver recorded samples at specified sample rate, mic level etc. to the observer using callback
- UnLock();
- _ptrAudioBuffer->DeliverRecordedData();
- Lock();
-
- QueryPerformanceCounter(&t2);
-
- if (InputSanityCheckAfterUnlockedPeriod() == -1)
- {
- // assert(false);
- return -1;
- }
- }
-
- if (_AGC)
- {
- WebRtc_UWord32 newMicLevel = _ptrAudioBuffer->NewMicLevel();
- if (newMicLevel != 0)
- {
- // The VQE will only deliver non-zero microphone levels when a change is needed.
- WEBRTC_TRACE(kTraceStream, kTraceUtility, _id,"AGC change of volume: => new=%u", newMicLevel);
-
- // We store this outside of the audio buffer to avoid
- // having it overwritten by the getter thread.
- _newMicLevel = newMicLevel;
- SetEvent(_hSetCaptureVolumeEvent);
- }
- }
-
- // return utilized buffer to queue after specified delay (default is 4)
- if (_recDelayCount > (_recPutBackDelay-1))
- {
- // deley buffer counter to compensate for "put-back-delay"
- bufCount = (bufCount + N_BUFFERS_IN - _recPutBackDelay) % N_BUFFERS_IN;
-
- // reset counter so we can make new detection
- _waveHeaderIn[bufCount].dwBytesRecorded = 0;
-
- // return the utilized wave-header after certain delay (given by _recPutBackDelay)
- res = waveInUnprepareHeader(_hWaveIn, &(_waveHeaderIn[bufCount]), sizeof(WAVEHDR));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInUnprepareHeader(%d) failed (err=%d)", bufCount, res);
- TraceWaveInError(res);
- }
-
- // ensure that the utilized header can be used again
- res = waveInPrepareHeader(_hWaveIn, &(_waveHeaderIn[bufCount]), sizeof(WAVEHDR));
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveInPrepareHeader(%d) failed (err=%d)", bufCount, res);
- TraceWaveInError(res);
- return -1;
- }
-
- // add the utilized buffer to the queue again
- res = waveInAddBuffer(_hWaveIn, &(_waveHeaderIn[bufCount]), sizeof(WAVEHDR));
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveInAddBuffer(%d) failed (err=%d)", bufCount, res);
- TraceWaveInError(res);
- if (_recPutBackDelay < 50)
- {
- _recPutBackDelay++;
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "_recPutBackDelay increased to %d", _recPutBackDelay);
- }
- else
- {
- if (_recError == 1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending recording error exists");
- }
- _recError = 1; // triggers callback from module process thread
- WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kRecordingError message posted: _recPutBackDelay=%u", _recPutBackDelay);
- }
- }
- } // if (_recDelayCount > (_recPutBackDelay-1))
-
- if (_recDelayCount < (_recPutBackDelay+1))
- {
- _recDelayCount++;
- }
-
- // increase main buffer count since one complete buffer has now been delivered
- _recBufCount++;
-
- if (send) {
- // Calculate processing time
- consumedTime = (int)(t2.QuadPart-t1.QuadPart);
- // handle wraps, time should not be higher than a second
- if ((consumedTime > _perfFreq.QuadPart) || (consumedTime < 0))
- consumedTime = 0;
- }
-
- } // if ((nBytesRecorded == fullBufferSizeInBytes))
-
- return nBytesRecorded;
-}
-
-// ----------------------------------------------------------------------------
-// PlayProc
-// ----------------------------------------------------------------------------
-
-int AudioDeviceWindowsWave::PlayProc(LONGLONG& consumedTime)
-{
- WebRtc_Word32 remTimeMS(0);
- WebRtc_Word8 playBuffer[4*PLAY_BUF_SIZE_IN_SAMPLES];
- WebRtc_UWord32 writtenSamples(0);
- WebRtc_UWord32 playedSamples(0);
-
- LARGE_INTEGER t1;
- LARGE_INTEGER t2;
-
- consumedTime = 0;
- _waitCounter++;
-
- // Get number of ms of sound that remains in the sound card buffer for playback.
- //
- remTimeMS = GetPlayoutBufferDelay(writtenSamples, playedSamples);
-
- // The threshold can be adaptive or fixed. The adaptive scheme is updated
- // also for fixed mode but the updated threshold is not utilized.
- //
- const WebRtc_UWord16 thresholdMS =
- (_playBufType == AudioDeviceModule::kAdaptiveBufferSize) ? _playBufDelay : _playBufDelayFixed;
-
- if (remTimeMS < thresholdMS + 9)
- {
- _dTcheckPlayBufDelay = 5;
-
- if (remTimeMS == 0)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceUtility, _id, "playout buffer is empty => we must adapt...");
- if (_waitCounter > 30)
- {
- _erZeroCounter++;
- if (_erZeroCounter == 2)
- {
- _playBufDelay += 15;
- _minPlayBufDelay += 20;
- _waitCounter = 50;
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "New playout states (er=0,erZero=2): minPlayBufDelay=%u, playBufDelay=%u", _minPlayBufDelay, _playBufDelay);
- }
- else if (_erZeroCounter == 3)
- {
- _erZeroCounter = 0;
- _playBufDelay += 30;
- _minPlayBufDelay += 25;
- _waitCounter = 0;
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "New playout states (er=0, erZero=3): minPlayBufDelay=%u, playBufDelay=%u", _minPlayBufDelay, _playBufDelay);
- }
- else
- {
- _minPlayBufDelay += 10;
- _playBufDelay += 15;
- _waitCounter = 50;
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "New playout states (er=0, erZero=1): minPlayBufDelay=%u, playBufDelay=%u", _minPlayBufDelay, _playBufDelay);
- }
- }
- }
- else if (remTimeMS < _minPlayBufDelay)
- {
- // If there is less than 25 ms of audio in the play out buffer
- // increase the buffersize limit value. _waitCounter prevents
- // _playBufDelay to be increased every time this function is called.
-
- if (_waitCounter > 30)
- {
- _playBufDelay += 10;
- if (_intro == 0)
- _waitCounter = 0;
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Playout threshold is increased: playBufDelay=%u", _playBufDelay);
- }
- }
- else if (remTimeMS < thresholdMS - 9)
- {
- _erZeroCounter = 0;
- }
- else
- {
- _erZeroCounter = 0;
- _dTcheckPlayBufDelay = 10;
- }
-
- QueryPerformanceCounter(&t1); // measure time: START
-
- // Ask for new PCM data to be played out using the AudioDeviceBuffer.
- // Ensure that this callback is executed without taking the audio-thread lock.
- //
- UnLock();
- WebRtc_UWord32 nSamples = _ptrAudioBuffer->RequestPlayoutData(PLAY_BUF_SIZE_IN_SAMPLES);
- Lock();
-
- if (OutputSanityCheckAfterUnlockedPeriod() == -1)
- {
- // assert(false);
- return -1;
- }
-
- nSamples = _ptrAudioBuffer->GetPlayoutData(playBuffer);
- if (nSamples != PLAY_BUF_SIZE_IN_SAMPLES)
- {
- WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "invalid number of output samples(%d)", nSamples);
- }
-
- QueryPerformanceCounter(&t2); // measure time: STOP
- consumedTime = (int)(t2.QuadPart - t1.QuadPart);
-
- Write(playBuffer, PLAY_BUF_SIZE_IN_SAMPLES);
-
- } // if (er < thresholdMS + 9)
- else if (thresholdMS + 9 < remTimeMS )
- {
- _erZeroCounter = 0;
- _dTcheckPlayBufDelay = 2; // check buffer more often
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Need to check playout buffer more often (dT=%u, remTimeMS=%u)", _dTcheckPlayBufDelay, remTimeMS);
- }
-
- // If the buffersize has been stable for 20 seconds try to decrease the buffer size
- if (_waitCounter > 2000)
- {
- _intro = 0;
- _playBufDelay--;
- _waitCounter = 1990;
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Playout threshold is decreased: playBufDelay=%u", _playBufDelay);
- }
-
- // Limit the minimum sound card (playback) delay to adaptive minimum delay
- if (_playBufDelay < _minPlayBufDelay)
- {
- _playBufDelay = _minPlayBufDelay;
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Playout threshold is limited to %u", _minPlayBufDelay);
- }
-
- // Limit the maximum sound card (playback) delay to 150 ms
- if (_playBufDelay > 150)
- {
- _playBufDelay = 150;
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Playout threshold is limited to %d", _playBufDelay);
- }
-
- // Upper limit of the minimum sound card (playback) delay to 65 ms.
- // Deactivated during "useHeader mode" (_useHeader > 0).
- if (_minPlayBufDelay > _MAX_minBuffer &&
- (_useHeader == 0))
- {
- _minPlayBufDelay = _MAX_minBuffer;
- WEBRTC_TRACE(kTraceDebug, kTraceUtility, _id, "Minimum playout threshold is limited to %d", _MAX_minBuffer);
- }
-
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// Write
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::Write(WebRtc_Word8* data, WebRtc_UWord16 nSamples)
-{
- if (_hWaveOut == NULL)
- {
- return -1;
- }
-
- if (_playIsInitialized)
- {
- MMRESULT res;
-
- const WebRtc_UWord16 bufCount(_playBufCount);
-
- // Place data in the memory associated with _waveHeaderOut[bufCount]
- //
- const WebRtc_Word16 nBytes = (2*_playChannels)*nSamples;
- memcpy(&_playBuffer[bufCount][0], &data[0], nBytes);
-
- // Send a data block to the given waveform-audio output device.
- //
- // When the buffer is finished, the WHDR_DONE bit is set in the dwFlags
- // member of the WAVEHDR structure. The buffer must be prepared with the
- // waveOutPrepareHeader function before it is passed to waveOutWrite.
- // Unless the device is paused by calling the waveOutPause function,
- // playback begins when the first data block is sent to the device.
- //
- res = waveOutWrite(_hWaveOut, &_waveHeaderOut[bufCount], sizeof(_waveHeaderOut[bufCount]));
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "waveOutWrite(%d) failed (err=%d)", bufCount, res);
- TraceWaveOutError(res);
-
- _writeErrors++;
- if (_writeErrors > 10)
- {
- if (_playError == 1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending playout error exists");
- }
- _playError = 1; // triggers callback from module process thread
- WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kPlayoutError message posted: _writeErrors=%u", _writeErrors);
- }
-
- return -1;
- }
-
- _playBufCount = (_playBufCount+1) % N_BUFFERS_OUT; // increase buffer counter modulo size of total buffer
- _writtenSamples += nSamples; // each sample is 2 or 4 bytes
- _writeErrors = 0;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// GetClockDrift
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::GetClockDrift(const WebRtc_UWord32 plSamp, const WebRtc_UWord32 rcSamp)
-{
- int drift = 0;
- unsigned int plSampDiff = 0, rcSampDiff = 0;
-
- if (plSamp >= _plSampOld)
- {
- plSampDiff = plSamp - _plSampOld;
- }
- else
- {
- // Wrap
- int i = 31;
- while(_plSampOld <= (unsigned int)POW2(i))
- {
- i--;
- }
-
- // Add the amount remaining prior to wrapping
- plSampDiff = plSamp + POW2(i + 1) - _plSampOld;
- }
-
- if (rcSamp >= _rcSampOld)
- {
- rcSampDiff = rcSamp - _rcSampOld;
- }
- else
- { // Wrap
- int i = 31;
- while(_rcSampOld <= (unsigned int)POW2(i))
- {
- i--;
- }
-
- rcSampDiff = rcSamp + POW2(i + 1) - _rcSampOld;
- }
-
- drift = plSampDiff - rcSampDiff;
-
- _plSampOld = plSamp;
- _rcSampOld = rcSamp;
-
- return drift;
-}
-
-// ----------------------------------------------------------------------------
-// MonitorRecording
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::MonitorRecording(const WebRtc_UWord32 time)
-{
- const WebRtc_UWord16 bytesPerSample = 2*_recChannels;
- const WebRtc_UWord32 nRecordedSamples = _recordedBytes/bytesPerSample;
-
- if (nRecordedSamples > 5*N_REC_SAMPLES_PER_SEC)
- {
- // 5 seconds of audio has been recorded...
- if ((time - _prevRecByteCheckTime) > 5700)
- {
- // ...and it was more than 5.7 seconds since we last did this check <=>
- // we have not been able to record 5 seconds of audio in 5.7 seconds,
- // hence a problem should be reported.
- // This problem can be related to USB overload.
- //
- if (_recWarning == 1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending recording warning exists");
- }
- _recWarning = 1; // triggers callback from module process thread
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "kRecordingWarning message posted: time-_prevRecByteCheckTime=%d", time - _prevRecByteCheckTime);
- }
-
- _recordedBytes = 0; // restart "check again when 5 seconds are recorded"
- _prevRecByteCheckTime = time; // reset timer to measure time for recording of 5 seconds
- }
-
- if ((time - _prevRecByteCheckTime) > 8000)
- {
- // It has been more than 8 seconds since we able to confirm that 5 seconds of
- // audio was recorded, hence we have not been able to record 5 seconds in
- // 8 seconds => the complete recording process is most likely dead.
- //
- if (_recError == 1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, "pending recording error exists");
- }
- _recError = 1; // triggers callback from module process thread
- WEBRTC_TRACE(kTraceError, kTraceUtility, _id, "kRecordingError message posted: time-_prevRecByteCheckTime=%d", time - _prevRecByteCheckTime);
-
- _prevRecByteCheckTime = time;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MonitorRecording
-//
-// Restart timer if needed (they seem to be messed up after a hibernate).
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioDeviceWindowsWave::RestartTimerIfNeeded(const WebRtc_UWord32 time)
-{
- const WebRtc_UWord32 diffMS = time - _prevTimerCheckTime;
- _prevTimerCheckTime = time;
-
- if (diffMS > 7)
- {
- // one timer-issue detected...
- _timerFaults++;
- if (_timerFaults > 5 && _timerRestartAttempts < 2)
- {
- // Reinitialize timer event if event fails to execute at least every 5ms.
- // On some machines it helps and the timer starts working as it should again;
- // however, not all machines (we have seen issues on e.g. IBM T60).
- // Therefore, the scheme below ensures that we do max 2 attempts to restart the timer.
- // For the cases where restart does not do the trick, we compensate for the reduced
- // resolution on both the recording and playout sides.
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, _id, " timer issue detected => timer is restarted");
- _timeEvent.StopTimer();
- _timeEvent.StartTimer(true, TIMER_PERIOD_MS);
- // make sure timer gets time to start up and we don't kill/start timer serveral times over and over again
- _timerFaults = -20;
- _timerRestartAttempts++;
- }
- }
- else
- {
- // restart timer-check scheme since we are OK
- _timerFaults = 0;
- _timerRestartAttempts = 0;
- }
-
- return 0;
-}
-
-} // namespace webrtc
-
diff --git a/trunk/src/modules/audio_device/main/source/win/audio_device_wave_win.h b/trunk/src/modules/audio_device/main/source/win/audio_device_wave_win.h
deleted file mode 100644
index 76607f6..0000000
--- a/trunk/src/modules/audio_device/main/source/win/audio_device_wave_win.h
+++ /dev/null
@@ -1,333 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_WAVE_WIN_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_WAVE_WIN_H
-
-#include "audio_device_generic.h"
-#include "audio_mixer_manager_win.h"
-
-#pragma comment( lib, "winmm.lib" )
-
-namespace webrtc {
-class EventWrapper;
-class ThreadWrapper;
-
-const WebRtc_UWord32 TIMER_PERIOD_MS = 2;
-const WebRtc_UWord32 REC_CHECK_TIME_PERIOD_MS = 4;
-const WebRtc_UWord16 REC_PUT_BACK_DELAY = 4;
-
-const WebRtc_UWord32 N_REC_SAMPLES_PER_SEC = 48000;
-const WebRtc_UWord32 N_PLAY_SAMPLES_PER_SEC = 48000;
-
-const WebRtc_UWord32 N_REC_CHANNELS = 1; // default is mono recording
-const WebRtc_UWord32 N_PLAY_CHANNELS = 2; // default is stereo playout
-
-// NOTE - CPU load will not be correct for other sizes than 10ms
-const WebRtc_UWord32 REC_BUF_SIZE_IN_SAMPLES = (N_REC_SAMPLES_PER_SEC/100);
-const WebRtc_UWord32 PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC/100);
-
-enum { N_BUFFERS_IN = 200 };
-enum { N_BUFFERS_OUT = 200 };
-
-class AudioDeviceWindowsWave : public AudioDeviceGeneric
-{
-public:
- AudioDeviceWindowsWave(const WebRtc_Word32 id);
- ~AudioDeviceWindowsWave();
-
- // Retrieve the currently utilized audio layer
- virtual WebRtc_Word32 ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
-
- // Main initializaton and termination
- virtual WebRtc_Word32 Init();
- virtual WebRtc_Word32 Terminate();
- virtual bool Initialized() const;
-
- // Device enumeration
- virtual WebRtc_Word16 PlayoutDevices();
- virtual WebRtc_Word16 RecordingDevices();
- virtual WebRtc_Word32 PlayoutDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize]);
- virtual WebRtc_Word32 RecordingDeviceName(WebRtc_UWord16 index, WebRtc_Word8 name[kAdmMaxDeviceNameSize], WebRtc_Word8 guid[kAdmMaxGuidSize]);
-
- // Device selection
- virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32 SetPlayoutDevice(AudioDeviceModule::WindowsDeviceType device);
- virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
- virtual WebRtc_Word32 SetRecordingDevice(AudioDeviceModule::WindowsDeviceType device);
-
- // Audio transport initialization
- virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
- virtual WebRtc_Word32 InitPlayout();
- virtual bool PlayoutIsInitialized() const;
- virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
- virtual WebRtc_Word32 InitRecording();
- virtual bool RecordingIsInitialized() const;
-
- // Audio transport control
- virtual WebRtc_Word32 StartPlayout();
- virtual WebRtc_Word32 StopPlayout();
- virtual bool Playing() const;
- virtual WebRtc_Word32 StartRecording();
- virtual WebRtc_Word32 StopRecording();
- virtual bool Recording() const;
-
- // Microphone Automatic Gain Control (AGC)
- virtual WebRtc_Word32 SetAGC(bool enable);
- virtual bool AGC() const;
-
- // Volume control based on the Windows Wave API (Windows only)
- virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft, WebRtc_UWord16 volumeRight);
- virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft, WebRtc_UWord16& volumeRight) const;
-
- // Audio mixer initialization
- virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
- virtual WebRtc_Word32 InitSpeaker();
- virtual bool SpeakerIsInitialized() const;
- virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
- virtual WebRtc_Word32 InitMicrophone();
- virtual bool MicrophoneIsInitialized() const;
-
- // Speaker volume controls
- virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
- virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
- virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
- virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
- virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
-
- // Microphone volume controls
- virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
- virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
- virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
- virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
- virtual WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
-
- // Speaker mute control
- virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
- virtual WebRtc_Word32 SetSpeakerMute(bool enable);
- virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
-
- // Microphone mute control
- virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
- virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
-
- // Microphone boost control
- virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
- virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
- virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
-
- // Stereo support
- virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
- virtual WebRtc_Word32 SetStereoPlayout(bool enable);
- virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
- virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
- virtual WebRtc_Word32 SetStereoRecording(bool enable);
- virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
-
- // Delay information and control
- virtual WebRtc_Word32 SetPlayoutBuffer(const AudioDeviceModule::BufferType type, WebRtc_UWord16 sizeMS);
- virtual WebRtc_Word32 PlayoutBuffer(AudioDeviceModule::BufferType& type, WebRtc_UWord16& sizeMS) const;
- virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
- virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
-
- // CPU load
- virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
-
-public:
- virtual bool PlayoutWarning() const;
- virtual bool PlayoutError() const;
- virtual bool RecordingWarning() const;
- virtual bool RecordingError() const;
- virtual void ClearPlayoutWarning();
- virtual void ClearPlayoutError();
- virtual void ClearRecordingWarning();
- virtual void ClearRecordingError();
-
-public:
- virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
-
-private:
- void Lock() { _critSect.Enter(); };
- void UnLock() { _critSect.Leave(); };
- WebRtc_Word32 Id() {return _id;}
- bool IsUsingOutputDeviceIndex() const {return _usingOutputDeviceIndex;}
- AudioDeviceModule::WindowsDeviceType OutputDevice() const {return _outputDevice;}
- WebRtc_UWord16 OutputDeviceIndex() const {return _outputDeviceIndex;}
- bool IsUsingInputDeviceIndex() const {return _usingInputDeviceIndex;}
- AudioDeviceModule::WindowsDeviceType InputDevice() const {return _inputDevice;}
- WebRtc_UWord16 InputDeviceIndex() const {return _inputDeviceIndex;}
-
-private:
- inline WebRtc_Word32 InputSanityCheckAfterUnlockedPeriod() const;
- inline WebRtc_Word32 OutputSanityCheckAfterUnlockedPeriod() const;
-
-private:
- WebRtc_Word32 EnumeratePlayoutDevices();
- WebRtc_Word32 EnumerateRecordingDevices();
- void TraceSupportFlags(DWORD dwSupport) const;
- void TraceWaveInError(MMRESULT error) const;
- void TraceWaveOutError(MMRESULT error) const;
- WebRtc_Word32 PrepareStartRecording();
- WebRtc_Word32 PrepareStartPlayout();
-
- WebRtc_Word32 RecProc(LONGLONG& consumedTime);
- int PlayProc(LONGLONG& consumedTime);
-
- WebRtc_Word32 GetPlayoutBufferDelay(WebRtc_UWord32& writtenSamples, WebRtc_UWord32& playedSamples);
- WebRtc_Word32 GetRecordingBufferDelay(WebRtc_UWord32& readSamples, WebRtc_UWord32& recSamples);
- WebRtc_Word32 Write(WebRtc_Word8* data, WebRtc_UWord16 nSamples);
- WebRtc_Word32 GetClockDrift(const WebRtc_UWord32 plSamp, const WebRtc_UWord32 rcSamp);
- WebRtc_Word32 MonitorRecording(const WebRtc_UWord32 time);
- WebRtc_Word32 RestartTimerIfNeeded(const WebRtc_UWord32 time);
-
-private:
- static bool ThreadFunc(void*);
- bool ThreadProcess();
-
- static DWORD WINAPI GetCaptureVolumeThread(LPVOID context);
- DWORD DoGetCaptureVolumeThread();
-
- static DWORD WINAPI SetCaptureVolumeThread(LPVOID context);
- DWORD DoSetCaptureVolumeThread();
-
-private:
- AudioDeviceBuffer* _ptrAudioBuffer;
-
- CriticalSectionWrapper& _critSect;
- EventWrapper& _timeEvent;
- EventWrapper& _recStartEvent;
- EventWrapper& _playStartEvent;
-
- HANDLE _hGetCaptureVolumeThread;
- HANDLE _hShutdownGetVolumeEvent;
- HANDLE _hSetCaptureVolumeThread;
- HANDLE _hShutdownSetVolumeEvent;
- HANDLE _hSetCaptureVolumeEvent;
-
- ThreadWrapper* _ptrThread;
- WebRtc_UWord32 _threadID;
-
- CriticalSectionWrapper& _critSectCb;
-
- WebRtc_Word32 _id;
-
- AudioMixerManager _mixerManager;
-
- bool _usingInputDeviceIndex;
- bool _usingOutputDeviceIndex;
- AudioDeviceModule::WindowsDeviceType _inputDevice;
- AudioDeviceModule::WindowsDeviceType _outputDevice;
- WebRtc_UWord16 _inputDeviceIndex;
- WebRtc_UWord16 _outputDeviceIndex;
- bool _inputDeviceIsSpecified;
- bool _outputDeviceIsSpecified;
-
- WAVEFORMATEX _waveFormatIn;
- WAVEFORMATEX _waveFormatOut;
-
- HWAVEIN _hWaveIn;
- HWAVEOUT _hWaveOut;
-
- WAVEHDR _waveHeaderIn[N_BUFFERS_IN];
- WAVEHDR _waveHeaderOut[N_BUFFERS_OUT];
-
- WebRtc_UWord8 _recChannels;
- WebRtc_UWord8 _playChannels;
- WebRtc_UWord16 _recBufCount;
- WebRtc_UWord16 _recDelayCount;
- WebRtc_UWord16 _recPutBackDelay;
-
- WebRtc_Word8 _recBuffer[N_BUFFERS_IN][4*REC_BUF_SIZE_IN_SAMPLES];
- WebRtc_Word8 _playBuffer[N_BUFFERS_OUT][4*PLAY_BUF_SIZE_IN_SAMPLES];
-
- AudioDeviceModule::BufferType _playBufType;
-
-private:
- bool _initialized;
- bool _recording;
- bool _playing;
- bool _recIsInitialized;
- bool _playIsInitialized;
- bool _startRec;
- bool _stopRec;
- bool _startPlay;
- bool _stopPlay;
- bool _AGC;
-
-private:
- WebRtc_UWord32 _prevPlayTime;
- WebRtc_UWord32 _prevRecTime;
- WebRtc_UWord32 _prevTimerCheckTime;
-
- WebRtc_UWord16 _playBufCount; // playout buffer index
- WebRtc_UWord16 _dTcheckPlayBufDelay; // dT for check of play buffer, {2,5,10} [ms]
- WebRtc_UWord16 _playBufDelay; // playback delay
- WebRtc_UWord16 _playBufDelayFixed; // fixed playback delay
- WebRtc_UWord16 _minPlayBufDelay; // minimum playback delay
- WebRtc_UWord16 _MAX_minBuffer; // level of (adaptive) min threshold must be < _MAX_minBuffer
-
- WebRtc_Word32 _erZeroCounter; // counts "buffer-is-empty" events
- WebRtc_Word32 _intro;
- WebRtc_Word32 _waitCounter;
-
- WebRtc_UWord32 _writtenSamples;
- WebRtc_UWord32 _writtenSamplesOld;
- WebRtc_UWord32 _playedSamplesOld;
-
- WebRtc_UWord32 _sndCardPlayDelay;
- WebRtc_UWord32 _sndCardRecDelay;
-
- WebRtc_UWord32 _plSampOld;
- WebRtc_UWord32 _rcSampOld;
-
- WebRtc_UWord32 _read_samples;
- WebRtc_UWord32 _read_samples_old;
- WebRtc_UWord32 _rec_samples_old;
-
- // State that detects driver problems:
- WebRtc_Word32 _dc_diff_mean;
- WebRtc_Word32 _dc_y_prev;
- WebRtc_Word32 _dc_penalty_counter;
- WebRtc_Word32 _dc_prevtime;
- WebRtc_UWord32 _dc_prevplay;
-
- WebRtc_UWord32 _recordedBytes; // accumulated #recorded bytes (reset periodically)
- WebRtc_UWord32 _prevRecByteCheckTime; // time when we last checked the recording process
-
- // CPU load measurements
- LARGE_INTEGER _perfFreq;
- LONGLONG _playAcc; // accumulated time for playout callback
- float _avgCPULoad; // average total (rec+play) CPU load
-
- WebRtc_Word32 _wrapCounter;
-
- WebRtc_Word32 _useHeader;
- WebRtc_Word16 _timesdwBytes;
- WebRtc_Word32 _no_of_msecleft_warnings;
- WebRtc_Word32 _writeErrors;
- WebRtc_Word32 _timerFaults;
- WebRtc_Word32 _timerRestartAttempts;
-
- WebRtc_UWord16 _playWarning;
- WebRtc_UWord16 _playError;
- WebRtc_UWord16 _recWarning;
- WebRtc_UWord16 _recError;
-
- WebRtc_UWord32 _newMicLevel;
- WebRtc_UWord32 _minMicVolume;
- WebRtc_UWord32 _maxMicVolume;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_WAVE_WIN_H
diff --git a/trunk/src/modules/audio_device/main/source/win/audio_mixer_manager_win.cc b/trunk/src/modules/audio_device/main/source/win/audio_mixer_manager_win.cc
deleted file mode 100644
index c3cf72e..0000000
--- a/trunk/src/modules/audio_device/main/source/win/audio_mixer_manager_win.cc
+++ /dev/null
@@ -1,2722 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_mixer_manager_win.h"
-#include "trace.h"
-
-#include <strsafe.h> // StringCchCopy(), StringCchCat(), StringCchPrintf()
-#include <cassert> // assert()
-
-#ifdef _WIN32
-// removes warning: "reinterpret_cast: conversion from 'UINT' to 'HMIXEROBJ'
-// of greater size"
-#pragma warning(disable:4312)
-#endif
-
-// Avoids the need of Windows 7 SDK
-#ifndef WAVE_MAPPED_kDefaultCommunicationDevice
-#define WAVE_MAPPED_kDefaultCommunicationDevice 0x0010
-#endif
-
-namespace webrtc {
-
-// ============================================================================
-// CONSTRUCTION/DESTRUCTION
-// ============================================================================
-
-AudioMixerManager::AudioMixerManager(const WebRtc_Word32 id) :
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _id(id),
- _inputMixerHandle(NULL),
- _outputMixerHandle(NULL)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s constructed", __FUNCTION__);
- ClearSpeakerState();
- ClearMicrophoneState();
-}
-
-AudioMixerManager::~AudioMixerManager()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s destructed", __FUNCTION__);
-
- Close();
-
- delete &_critSect;
-}
-
-// ============================================================================
-// PUBLIC METHODS
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// Close
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::Close()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_outputMixerHandle != NULL)
- {
- mixerClose(_outputMixerHandle);
- _outputMixerHandle = NULL;
- }
- if (_inputMixerHandle != NULL)
- {
- mixerClose(_inputMixerHandle);
- _inputMixerHandle = NULL;
- }
- return 0;
-
-}
-
-// ----------------------------------------------------------------------------
-// CloseSpeaker
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::CloseSpeaker()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_outputMixerHandle == NULL)
- {
- return -1;
- }
-
- ClearSpeakerState(_outputMixerID);
-
- mixerClose(_outputMixerHandle);
- _outputMixerHandle = NULL;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// CloseMicrophone
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::CloseMicrophone()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_inputMixerHandle == NULL)
- {
- return -1;
- }
-
- ClearMicrophoneState(_inputMixerID);
-
- mixerClose(_inputMixerHandle);
- _inputMixerHandle = NULL;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// EnumerateAll
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::EnumerateAll()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- UINT nDevices = mixerGetNumDevs();
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#mixer devices: %u", nDevices);
-
- MIXERCAPS caps;
- MIXERLINE destLine;
- MIXERLINE sourceLine;
- MIXERCONTROL controlArray[MAX_NUMBER_OF_LINE_CONTROLS];
-
- UINT mixId(0);
- UINT destId(0);
- UINT sourceId(0);
-
- for (mixId = 0; mixId < nDevices; mixId++)
- {
- if (!GetCapabilities(mixId, caps, true))
- continue;
-
- for (destId = 0; destId < caps.cDestinations; destId++)
- {
- GetDestinationLineInfo(mixId, destId, destLine, true);
- GetAllLineControls(mixId, destLine, controlArray, true);
-
- for (sourceId = 0; sourceId < destLine.cConnections; sourceId++)
- {
- GetSourceLineInfo(mixId, destId, sourceId, sourceLine, true);
- GetAllLineControls(mixId, sourceLine, controlArray, true);
- }
- }
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// EnumerateSpeakers
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::EnumerateSpeakers()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- UINT nDevices = mixerGetNumDevs();
- if (nDevices > MAX_NUMBER_MIXER_DEVICES)
- {
- assert(false);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#mixer devices: %u", nDevices);
-
- MIXERCAPS caps;
- MIXERLINE destLine;
- MIXERCONTROL controlArray[MAX_NUMBER_OF_LINE_CONTROLS];
-
- UINT mixId(0);
- UINT destId(0);
-
- ClearSpeakerState();
-
- // scan all avaliable mixer devices
- for (mixId = 0; mixId < nDevices; mixId++)
- {
- // get capabilities for the specified mixer ID
- GetCapabilities(mixId, caps);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[mixerID=%d] %s: ", mixId, WideToUTF8(caps.szPname));
- // scan all avaliable destinations for this mixer
- for (destId = 0; destId < caps.cDestinations; destId++)
- {
- GetDestinationLineInfo(mixId, destId, destLine);
- if ((destLine.cControls == 0) || // no controls or
- (destLine.cConnections == 0) || // no source lines or
- (destLine.fdwLine & MIXERLINE_LINEF_DISCONNECTED) || // disconnected or
- !(destLine.fdwLine & MIXERLINE_LINEF_ACTIVE)) // inactive
- {
- // don't store this line ID since it will not be possible to control
- continue;
- }
- if ((destLine.dwComponentType == MIXERLINE_COMPONENTTYPE_DST_SPEAKERS) ||
- (destLine.dwComponentType == MIXERLINE_COMPONENTTYPE_DST_HEADPHONES))
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found valid speaker/headphone (name: %s, ID: %u)", WideToUTF8(destLine.szName), destLine.dwLineID);
- _speakerState[mixId].dwLineID = destLine.dwLineID;
- _speakerState[mixId].speakerIsValid = true;
- // retrieve all controls for the speaker component
- GetAllLineControls(mixId, destLine, controlArray);
- for (UINT c = 0; c < destLine.cControls; c++)
- {
- if (controlArray[c].dwControlType == MIXERCONTROL_CONTROLTYPE_VOLUME)
- {
- _speakerState[mixId].dwVolumeControlID = controlArray[c].dwControlID;
- _speakerState[mixId].volumeControlIsValid = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found volume control (name: %s, ID: %u)", WideToUTF8(controlArray[c].szName), controlArray[c].dwControlID);
- }
- else if (controlArray[c].dwControlType == MIXERCONTROL_CONTROLTYPE_MUTE)
- {
- _speakerState[mixId].dwMuteControlID = controlArray[c].dwControlID;
- _speakerState[mixId].muteControlIsValid = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found mute control (name: %s, ID: %u)", WideToUTF8(controlArray[c].szName), controlArray[c].dwControlID);
- }
- }
- break;
- }
- }
- if (!SpeakerIsValid(mixId))
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "unable to find a valid speaker destination line", mixId);
- }
- }
-
- if (ValidSpeakers() == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to locate any valid speaker line");
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// EnumerateMicrophones
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::EnumerateMicrophones()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- UINT nDevices = mixerGetNumDevs();
- if (nDevices > MAX_NUMBER_MIXER_DEVICES)
- {
- assert(false);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "#mixer devices: %u", nDevices);
-
- MIXERCAPS caps;
- MIXERLINE destLine;
- MIXERLINE sourceLine;
- MIXERCONTROL controlArray[MAX_NUMBER_OF_LINE_CONTROLS];
-
- UINT mixId(0);
- UINT destId(0);
-
- ClearMicrophoneState();
-
- // scan all avaliable mixer devices
- for (mixId = 0; mixId < nDevices; mixId++)
- {
- // get capabilities for the specified mixer ID
- GetCapabilities(mixId, caps);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "[mixerID=%d] %s: ", mixId, WideToUTF8(caps.szPname));
- // scan all avaliable destinations for this mixer
- for (destId = 0; destId < caps.cDestinations; destId++)
- {
- GetDestinationLineInfo(mixId, destId, destLine);
-
- if ((destLine.cConnections == 0) || // no source lines or
- (destLine.fdwLine & MIXERLINE_LINEF_DISCONNECTED) || // disconnected or
- !(destLine.fdwLine & MIXERLINE_LINEF_ACTIVE)) // inactive
- {
- // Don't store this line ID since there are no sources connected to this destination.
- // Compare with the speaker side where we also exclude lines with no controls.
- continue;
- }
-
- if (destLine.dwComponentType == MIXERLINE_COMPONENTTYPE_DST_WAVEIN)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found valid Wave In destination (name: %s, ID: %u)", WideToUTF8(destLine.szName), destLine.dwLineID);
- _microphoneState[mixId].dwLineID = destLine.dwLineID;
- _microphoneState[mixId].microphoneIsValid = true;
-
- // retrieve all controls for the identified wave-in destination
- if (!GetAllLineControls(mixId, destLine, controlArray))
- {
- // This destination has no controls. We must try to control
- // one of its sources instead.
- // This is a rare state but has been found for some
- // Logitech USB headsets.
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "this destination has no controls => must control source");
- for (DWORD sourceId = 0; sourceId < destLine.cConnections; sourceId++)
- {
- GetSourceLineInfo(mixId, destId, sourceId, sourceLine, false);
- if (sourceLine.dwComponentType ==
- MIXERLINE_COMPONENTTYPE_SRC_MICROPHONE)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "found microphone source ( name: %s, ID: %u)",
- WideToUTF8(sourceLine.szName), sourceId);
- GetAllLineControls(mixId, sourceLine, controlArray, false);
- // scan the controls for this source and search for volume,
- // mute and on/off (<=> boost) controls
- for (UINT sc = 0; sc < sourceLine.cControls; sc++)
- {
- if (controlArray[sc].dwControlType ==
- MIXERCONTROL_CONTROLTYPE_VOLUME)
- {
- // store this volume control
- _microphoneState[mixId].dwVolumeControlID =
- controlArray[sc].dwControlID;
- _microphoneState[mixId].volumeControlIsValid = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "found volume control (name: %s, ID: %u)",
- WideToUTF8(controlArray[sc].szName),
- controlArray[sc].dwControlID);
- }
- else if (controlArray[sc].dwControlType ==
- MIXERCONTROL_CONTROLTYPE_MUTE)
- {
- // store this mute control
- _microphoneState[mixId].dwMuteControlID =
- controlArray[sc].dwControlID;
- _microphoneState[mixId].muteControlIsValid = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "found mute control (name: %s, ID: %u)",
- WideToUTF8(controlArray[sc].szName),
- controlArray[sc].dwControlID);
- }
- else if (controlArray[sc].dwControlType ==
- MIXERCONTROL_CONTROLTYPE_ONOFF ||
- controlArray[sc].dwControlType ==
- MIXERCONTROL_CONTROLTYPE_LOUDNESS)
- {
- // store this on/off control (most likely a Boost control)
- _microphoneState[mixId].dwOnOffControlID =
- controlArray[sc].dwControlID;
- _microphoneState[mixId].onOffControlIsValid = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "found on/off control (name: %s, ID: %u)",
- WideToUTF8(controlArray[sc].szName),
- controlArray[sc].dwControlID);
- }
- }
- }
- }
-
- break;
- }
-
- // It seems like there are three different configurations we can find in this state:
- //
- // (1) The Wave-in destination contains one MUX control only
- // (2) The Wave-in destination contains one or more controls where one is a volume control
- // (3) On Vista and Win 7, it seems like case 2 above is extended.
- // It is common that a Wave-in destination has two master controls (volume and mute),
- // AND a microphone source as well with its own volume and mute controls with unique
- // identifiers. Initial tests have shown that it is sufficient to modify the master
- // controls only. The source controls will "follow" the master settings, hence the
- // source controls seem to be redundant.
- //
- // For case 1, we should locate the selected source and its controls. The MUX setting will
- // give us the selected source. NOTE - the selecion might not be a microphone.
- //
- // For case 2, the volume control works as a master level control and we should use that one.
- //
- // For case 3, we use the master controls only and assume that the source control will "follow".
- //
- // Examples of case 1: - SigmaTel Audio (built-in)
- // - add more..
- //
- // Examples of case 2: - Plantronics USB Headset
- // - Eutectics IPP 200 USB phone
- // - add more...
- //
- // Examples of case 3: - Realtek High Definition on Vista (TL)
- // - add more...
-
- if ((destLine.cControls == 1) &&
- (controlArray[0].dwControlType == MIXERCONTROL_CONTROLTYPE_MUX))
- {
- // Case 1: MUX control detected => locate the selected source and its volume control
- // Note that, the selecion might not be a microphone. A warning is given for
- // this case only, i.e., it is OK to control a selected Line In source as long
- // as it is connected to the wave-in destination.
-
- UINT selection(0);
- const DWORD nItemsInMux(controlArray[0].cMultipleItems);
-
- // decide which source line that is selected in the mux
- if (GetSelectedMuxSource(mixId, controlArray[0].dwControlID, nItemsInMux, selection))
- {
- // selection now contains the index of the selected source =>
- // read the line information for this source
- // if conditions listed below
- // condition 1: invalid source
- // condition 2: no controls
- // condition 3: disconnected
- // condition 4: inactive
- if (!GetSourceLineInfo(mixId, destId, selection, sourceLine) ||
- (sourceLine.cControls == 0) ||
- (sourceLine.fdwLine & MIXERLINE_LINEF_DISCONNECTED) ||
- !(sourceLine.fdwLine & MIXERLINE_LINEF_ACTIVE))
- {
- continue;
- }
-
- if (sourceLine.dwComponentType != MIXERLINE_COMPONENTTYPE_SRC_MICROPHONE)
- {
- // add more details about the selected source (not a microphone)
- TraceComponentType(sourceLine.dwComponentType);
- // send a warning just to inform about the fact that a non-microphone source will be controlled
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "the selected (to be controlled) source is not a microphone type");
- }
-
- // retrieve all controls for the selected source
- GetAllLineControls(mixId, sourceLine, controlArray);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "MUX selection is %u [0,%u]", selection, nItemsInMux-1);
-
- // scan the controls for this source and search for volume, mute and on/off (<=> boost) controls
- for (UINT sc = 0; sc < sourceLine.cControls; sc++)
- {
- if (controlArray[sc].dwControlType == MIXERCONTROL_CONTROLTYPE_VOLUME)
- {
- // store this volume control
- _microphoneState[mixId].dwVolumeControlID = controlArray[sc].dwControlID;
- _microphoneState[mixId].volumeControlIsValid = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found volume control (name: %s, ID: %u)", WideToUTF8(controlArray[sc].szName), controlArray[sc].dwControlID);
- }
- else if (controlArray[sc].dwControlType == MIXERCONTROL_CONTROLTYPE_MUTE)
- {
- // store this mute control
- _microphoneState[mixId].dwMuteControlID = controlArray[sc].dwControlID;
- _microphoneState[mixId].muteControlIsValid = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found mute control (name: %s, ID: %u)", WideToUTF8(controlArray[sc].szName), controlArray[sc].dwControlID);
- }
- else if (controlArray[sc].dwControlType == MIXERCONTROL_CONTROLTYPE_ONOFF ||
- controlArray[sc].dwControlType == MIXERCONTROL_CONTROLTYPE_LOUDNESS)
- {
- // store this on/off control (most likely a Boost control)
- _microphoneState[mixId].dwOnOffControlID = controlArray[sc].dwControlID;
- _microphoneState[mixId].onOffControlIsValid = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found on/off control (name: %s, ID: %u)", WideToUTF8(controlArray[sc].szName), controlArray[sc].dwControlID);
- }
- }
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to detect which source to control");
- }
-
- }
- else if (destLine.cConnections == 1)
- {
- // Case 2 or Case 3:
-
- GetSourceLineInfo(mixId, destId, 0, sourceLine);
- if ((sourceLine.dwComponentType == MIXERLINE_COMPONENTTYPE_SRC_MICROPHONE) &&
- (sourceLine.cControls > 0))
- {
- // Case 3: same as Case 2 below but we have also detected a Microphone source
- // with its own controls. So far, I have not been able to find any device
- // where it is required to modify these controls. Until I have found such
- // a device, this case will be handled as a Case 2 (see below).
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "microphone source controls will not be controlled");
- }
- else if ((sourceLine.dwComponentType == MIXERLINE_COMPONENTTYPE_SRC_MICROPHONE) &&
- (sourceLine.cControls == 0))
- {
- // default state on non Vista/Win 7 machines
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "microphone source has no controls => use master controls instead");
- }
- else
- {
- // add more details about the selected source (not a microphone)
- TraceComponentType(sourceLine.dwComponentType);
- // send a warning just to inform about the fact that a non-microphone source will be controlled
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "the connected (to be controlled) source is not a microphone type");
- }
-
- // Case 2 : one source only and no MUX control detected =>
- // locate the master volume control (and mute + boost controls if possible)
-
- // scan the controls for this wave-in destination and search for volume, mute and on/off (<=> boost) controls
- for (UINT dc = 0; dc < destLine.cControls; dc++)
- {
- if (controlArray[dc].dwControlType == MIXERCONTROL_CONTROLTYPE_VOLUME)
- {
- // store this volume control
- _microphoneState[mixId].dwVolumeControlID = controlArray[dc].dwControlID;
- _microphoneState[mixId].volumeControlIsValid = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found volume control (name: %s, ID: %u)", WideToUTF8(controlArray[dc].szName), controlArray[dc].dwControlID);
- }
- else if (controlArray[dc].dwControlType == MIXERCONTROL_CONTROLTYPE_MUTE)
- {
- // store this mute control
- _microphoneState[mixId].dwMuteControlID = controlArray[dc].dwControlID;
- _microphoneState[mixId].muteControlIsValid = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found mute control (name: %s, ID: %u)", WideToUTF8(controlArray[dc].szName), controlArray[dc].dwControlID);
- }
- else if (controlArray[dc].dwControlType == MIXERCONTROL_CONTROLTYPE_ONOFF ||
- controlArray[dc].dwControlType == MIXERCONTROL_CONTROLTYPE_LOUDNESS ||
- controlArray[dc].dwControlType == MIXERCONTROL_CONTROLTYPE_BOOLEAN)
- {
- // store this on/off control
- _microphoneState[mixId].dwOnOffControlID = controlArray[dc].dwControlID;
- _microphoneState[mixId].onOffControlIsValid = true;
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "found on/off control (name: %s, ID: %u)", WideToUTF8(controlArray[dc].szName), controlArray[dc].dwControlID);
- }
- }
- }
- else
- {
- // We are in a state where more than one source is connected to the wave-in destination.
- // I am bailing out here for now until I understand this case better.
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "failed to locate valid microphone controls for this mixer");
- }
- break;
- }
- } // for (destId = 0; destId < caps.cDestinations; destId++)
-
- if (!MicrophoneIsValid(mixId))
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "unable to find a valid microphone destination line", mixId);
- }
- } // for (mixId = 0; mixId < nDevices; mixId++)
-
- if (ValidMicrophones() == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "failed to locate any valid microphone line");
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// OpenSpeaker I(II)
-//
-// Verifies that the mixer contains a valid speaker destination line.
-// Avoids opening the mixer if valid control has not been found.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::OpenSpeaker(AudioDeviceModule::WindowsDeviceType device)
-{
- if (device == AudioDeviceModule::kDefaultDevice)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::OpenSpeaker(kDefaultDevice)");
- }
- else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::OpenSpeaker(kDefaultCommunicationDevice)");
- }
-
- CriticalSectionScoped lock(_critSect);
-
- // Close any existing output mixer handle
- //
- if (_outputMixerHandle != NULL)
- {
- mixerClose(_outputMixerHandle);
- _outputMixerHandle = NULL;
- }
-
- MMRESULT res;
- WAVEFORMATEX waveFormat;
- HWAVEOUT hWaveOut(NULL);
-
- waveFormat.wFormatTag = WAVE_FORMAT_PCM ;
- waveFormat.nChannels = 2;
- waveFormat.nSamplesPerSec = 48000;
- waveFormat.wBitsPerSample = 16;
- waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
- waveFormat.cbSize = 0;
-
- // We need a waveform-audio output handle for the currently selected output device.
- // This handle will then give us the corresponding mixer identifier. Once the mixer
- // ID is known, it is possible to open the output mixer.
- //
- if (device == AudioDeviceModule::kDefaultCommunicationDevice)
- {
- // check if it is possible to open the default communication device (supported on Windows 7)
- res = waveOutOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL |
- WAVE_MAPPED_kDefaultCommunicationDevice | WAVE_FORMAT_QUERY);
- if (MMSYSERR_NOERROR == res)
- {
- // if so, open the default communication device for real
- res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_kDefaultCommunicationDevice);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default communication device");
- }
- else
- {
- // use default device since default communication device was not avaliable
- res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "unable to open default communication device => using default instead");
- }
- }
- else if (device == AudioDeviceModule::kDefaultDevice)
- {
- // open default device since it has been requested
- res = waveOutOpen(&hWaveOut, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default output device");
- }
-
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutOpen() failed (err=%d)", res);
- TraceWaveOutError(res);
- }
-
- UINT mixerId(0);
- HMIXER hMixer(NULL);
-
- // Retrieve the device identifier for a mixer device associated with the
- // aquired waveform-audio output handle.
- //
- res = mixerGetID((HMIXEROBJ)hWaveOut, &mixerId, MIXER_OBJECTF_HWAVEOUT);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetID(MIXER_OBJECTF_HWAVEOUT) failed (err=%d)", res);
- // identification failed => use default mixer identifier (=0)
- mixerId = 0;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "specified output device <=> mixer ID %u", mixerId);
-
- // The waveform-audio output handle is no longer needed.
- //
- waveOutClose(hWaveOut);
-
- // Verify that the mixer contains a valid speaker destination line.
- // Avoid opening the mixer if valid control has not been found.
- //
- if (!SpeakerIsValid(mixerId))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "it is not possible to control the speaker volume for this mixer device");
- return -1;
- }
-
- // Open the specified mixer device and ensure that the device will not
- // be removed until the application closes the handle.
- //
- res = mixerOpen(&hMixer, mixerId, 0, 0, MIXER_OBJECTF_MIXER);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerOpen() failed (err=%d)", res);
- }
-
- // Store the output mixer handle and active mixer identifier
- //
- _outputMixerHandle = hMixer;
- _outputMixerID = mixerId;
-
- if (_outputMixerHandle != NULL)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "the output mixer device is now open (0x%x)", _outputMixerHandle);
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// OpenSpeaker II(II)
-//
-// Verifies that the mixer contains a valid speaker destination line.
-// Avoids opening the mixer if valid control has not been found.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::OpenSpeaker(WebRtc_UWord16 index)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::OpenSpeaker(index=%d)", index);
-
- CriticalSectionScoped lock(_critSect);
-
- // Close any existing output mixer handle
- //
- if (_outputMixerHandle != NULL)
- {
- mixerClose(_outputMixerHandle);
- _outputMixerHandle = NULL;
- }
-
- MMRESULT res;
- WAVEFORMATEX waveFormat;
- HWAVEOUT hWaveOut(NULL);
-
- const UINT deviceID(index); // use index parameter as device identifier
-
- waveFormat.wFormatTag = WAVE_FORMAT_PCM ;
- waveFormat.nChannels = 2;
- waveFormat.nSamplesPerSec = 48000;
- waveFormat.wBitsPerSample = 16;
- waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
- waveFormat.cbSize = 0;
-
- // We need a waveform-audio output handle for the currently selected output device.
- // This handle will then give us the corresponding mixer identifier. Once the mixer
- // ID is known, it is possible to open the output mixer.
- //
- res = waveOutOpen(&hWaveOut, deviceID, &waveFormat, 0, 0, CALLBACK_NULL);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveOutOpen(deviceID=%u) failed (err=%d)", index, res);
- TraceWaveOutError(res);
- }
-
- UINT mixerId(0);
- HMIXER hMixer(NULL);
-
- // Retrieve the device identifier for a mixer device associated with the
- // aquired waveform-audio output handle.
- //
- res = mixerGetID((HMIXEROBJ)hWaveOut, &mixerId, MIXER_OBJECTF_HWAVEOUT);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetID(MIXER_OBJECTF_HWAVEOUT) failed (err=%d)", res);
- // identification failed => use default mixer identifier (=0)
- mixerId = 0;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "specified output device <=> mixer ID %u", mixerId);
-
- // The waveform-audio output handle is no longer needed.
- //
- waveOutClose(hWaveOut);
-
- // Verify that the mixer contains a valid speaker destination line.
- // Avoid opening the mixer if valid control has not been found.
- //
- if (!SpeakerIsValid(mixerId))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "it is not possible to control the speaker volume for this mixer device");
- return -1;
- }
-
- // Open the specified mixer device and ensure that the device will not
- // be removed until the application closes the handle.
- //
- res = mixerOpen(&hMixer, mixerId, 0, 0, MIXER_OBJECTF_MIXER);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerOpen() failed (err=%d)", res);
- }
-
- // Store the output mixer handle and active mixer identifier
- //
- _outputMixerHandle = hMixer;
- _outputMixerID = mixerId;
-
- if (_outputMixerHandle != NULL)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "the output mixer device is now open (0x%x)", _outputMixerHandle);
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// OpenMicrophone I(II)
-//
-// Verifies that the mixer contains a valid wave-in destination line.
-// Avoids opening the mixer if valid control has not been found.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::OpenMicrophone(AudioDeviceModule::WindowsDeviceType device)
-{
- if (device == AudioDeviceModule::kDefaultDevice)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::OpenMicrophone(kDefaultDevice)");
- }
- else if (device == AudioDeviceModule::kDefaultCommunicationDevice)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::OpenMicrophone(kDefaultCommunicationDevice)");
- }
-
- CriticalSectionScoped lock(_critSect);
-
- // Close any existing output mixer handle
- //
- if (_inputMixerHandle != NULL)
- {
- mixerClose(_inputMixerHandle);
- _inputMixerHandle = NULL;
- }
-
- MMRESULT res;
- WAVEFORMATEX waveFormat;
- HWAVEIN hWaveIn(NULL);
-
- waveFormat.wFormatTag = WAVE_FORMAT_PCM ;
- waveFormat.nChannels = 1;
- waveFormat.nSamplesPerSec = 48000;
- waveFormat.wBitsPerSample = 16;
- waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
- waveFormat.cbSize = 0 ;
-
- // We need a waveform-audio input handle for the currently selected input device.
- // This handle will then give us the corresponding mixer identifier. Once the mixer
- // ID is known, it is possible to open the input mixer.
- //
- if (device == AudioDeviceModule::kDefaultCommunicationDevice)
- {
- // check if it is possible to open the default communication device (supported on Windows 7)
- res = waveInOpen(NULL, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL |
- WAVE_MAPPED_kDefaultCommunicationDevice | WAVE_FORMAT_QUERY);
- if (MMSYSERR_NOERROR == res)
- {
- // if so, open the default communication device for real
- res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL | WAVE_MAPPED_kDefaultCommunicationDevice);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default communication device");
- }
- else
- {
- // use default device since default communication device was not avaliable
- res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id,
- "unable to open default communication device => using default instead");
- }
- }
- else if (device == AudioDeviceModule::kDefaultDevice)
- {
- // open default device since it has been requested
- res = waveInOpen(&hWaveIn, WAVE_MAPPER, &waveFormat, 0, 0, CALLBACK_NULL);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "opening default input device");
- }
-
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInOpen() failed (err=%d)", res);
- TraceWaveInError(res);
- }
-
- UINT mixerId(0);
- HMIXER hMixer(NULL);
-
- // Retrieve the device identifier for a mixer device associated with the
- // aquired waveform-audio input handle.
- //
- res = mixerGetID((HMIXEROBJ)hWaveIn, &mixerId, MIXER_OBJECTF_HWAVEIN);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetID(MIXER_OBJECTF_HWAVEIN) failed (err=%d)", res);
- // identification failed => use default mixer identifier (=0)
- mixerId = 0;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "specified input device <=> mixer ID %u", mixerId);
-
- // The waveform-audio input handle is no longer needed.
- //
- waveInClose(hWaveIn);
-
- // Verify that the mixer contains a valid wave-in destination line and a volume control.
- // Avoid opening the mixer if valid control has not been found.
- //
- if (!MicrophoneIsValid(mixerId))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "it is not possible to control the microphone volume for this mixer device");
- return -1;
- }
-
- // Open the specified mixer device and ensure that the device will not
- // be removed until the application closes the handle.
- //
- res = mixerOpen(&hMixer, mixerId, 0, 0, MIXER_OBJECTF_MIXER);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerOpen() failed (err=%d)", res);
- }
-
- // Store the input mixer handle and active mixer identifier
- //
- _inputMixerHandle = hMixer;
- _inputMixerID = mixerId;
-
- if (_inputMixerHandle != NULL)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "the input mixer device is now open (0x%x)", _inputMixerHandle);
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// OpenMicrophone II(II)
-//
-// Verifies that the mixer contains a valid wave-in destination line.
-// Avoids opening the mixer if valid control has not been found.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::OpenMicrophone(WebRtc_UWord16 index)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::OpenMicrophone(index=%d)", index);
-
- CriticalSectionScoped lock(_critSect);
-
- // Close any existing input mixer handle
- //
- if (_inputMixerHandle != NULL)
- {
- mixerClose(_inputMixerHandle);
- _inputMixerHandle = NULL;
- }
-
- MMRESULT res;
- WAVEFORMATEX waveFormat;
- HWAVEIN hWaveIn(NULL);
-
- const UINT deviceID(index); // use index parameter as device identifier
-
- waveFormat.wFormatTag = WAVE_FORMAT_PCM ;
- waveFormat.nChannels = 1;
- waveFormat.nSamplesPerSec = 48000;
- waveFormat.wBitsPerSample = 16;
- waveFormat.nBlockAlign = waveFormat.nChannels * waveFormat.wBitsPerSample / 8;
- waveFormat.nAvgBytesPerSec = waveFormat.nSamplesPerSec * waveFormat.nBlockAlign;
- waveFormat.cbSize = 0;
-
- // We need a waveform-audio input handle for the currently selected input device.
- // This handle will then give us the corresponding mixer identifier. Once the mixer
- // ID is known, it is possible to open the input mixer.
- //
- res = waveInOpen(&hWaveIn, deviceID, &waveFormat, 0, 0, CALLBACK_NULL);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "waveInOpen(deviceID=%u) failed (err=%d)", index, res);
- TraceWaveInError(res);
- }
-
- UINT mixerId(0);
- HMIXER hMixer(NULL);
-
- // Retrieve the device identifier for a mixer device associated with the
- // aquired waveform-audio input handle.
- //
- res = mixerGetID((HMIXEROBJ)hWaveIn, &mixerId, MIXER_OBJECTF_HWAVEIN);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetID(MIXER_OBJECTF_HWAVEIN) failed (err=%d)", res);
- // identification failed => use default mixer identifier (=0)
- mixerId = 0;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "specified input device <=> mixer ID %u", mixerId);
-
- // The waveform-audio input handle is no longer needed.
- //
- waveInClose(hWaveIn);
-
- // Verify that the mixer contains a valid wave-in destination line.
- // Avoid opening the mixer if valid control has not been found.
- //
- if (!MicrophoneIsValid(mixerId))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "it is not possible to control the microphone volume for this mixer device");
- return -1;
- }
-
- // Open the specified mixer device and ensure that the device will not
- // be removed until the application closes the handle.
- //
- res = mixerOpen(&hMixer, mixerId, 0, 0, MIXER_OBJECTF_MIXER);
- if (MMSYSERR_NOERROR != res)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerOpen() failed (err=%d)", res);
- }
-
- // Store the input mixer handle and active mixer identifier
- //
- _inputMixerHandle = hMixer;
- _inputMixerID = mixerId;
-
- if (_inputMixerHandle != NULL)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "the input mixer device is now open (0x%x)", _inputMixerHandle);
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioMixerManager::SpeakerIsInitialized() const
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- return (_outputMixerHandle != NULL);
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneIsInitialized
-// ----------------------------------------------------------------------------
-
-bool AudioMixerManager::MicrophoneIsInitialized() const
-{
- WEBRTC_TRACE(kTraceMemory, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- return (_inputMixerHandle != NULL);
-}
-
-// ----------------------------------------------------------------------------
-// SetSpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::SetSpeakerVolume(WebRtc_UWord32 volume)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::SetSpeakerVolume(volume=%u)", volume);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_outputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
- return -1;
- }
-
- const UINT mixerID(_outputMixerID);
- const DWORD dwControlID(_speakerState[_outputMixerID].dwVolumeControlID);
- DWORD dwValue(volume);
-
- // Set one unsigned control value for a specified volume-control identifier
- //
- if (!SetUnsignedControlValue(mixerID, dwControlID, dwValue))
- {
- return -1;
- }
-
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolume
-//
-// Note that (MIXERCONTROL_CONTROLTYPE_VOLUME & MIXERCONTROL_CT_UNITS_MASK)
-// always equals MIXERCONTROL_CT_UNITS_UNSIGNED;
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::SpeakerVolume(WebRtc_UWord32& volume) const
-{
-
- if (_outputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
- return -1;
- }
-
- const UINT mixerID(_outputMixerID);
- const DWORD dwControlID(_speakerState[_outputMixerID].dwVolumeControlID);
- DWORD dwValue(0);
-
- // Retrieve one unsigned control value for a specified volume-control identifier
- //
- if (!GetUnsignedControlValue(mixerID, dwControlID, dwValue))
- {
- return -1;
- }
-
- volume = dwValue;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MaxSpeakerVolume
-//
-// Note that (MIXERCONTROL_CONTROLTYPE_VOLUME & MIXERCONTROL_CT_UNITS_MASK)
-// always equals MIXERCONTROL_CT_UNITS_UNSIGNED
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const
-{
-
- if (_outputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
- return -1;
- }
-
- const UINT mixerID(_outputMixerID);
- const DWORD dwControlID(_speakerState[_outputMixerID].dwVolumeControlID);
- MIXERCONTROL mixerControl;
-
- // Retrieve one control line for a specified volume-control identifier
- //
- if (!GetLineControl(mixerID, dwControlID, mixerControl))
- {
- return -1;
- }
-
- maxVolume = mixerControl.Bounds.dwMaximum;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MinSpeakerVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::MinSpeakerVolume(WebRtc_UWord32& minVolume) const
-{
-
- if (_outputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
- return -1;
- }
-
- const UINT mixerID(_outputMixerID);
- const DWORD dwControlID(_speakerState[_outputMixerID].dwVolumeControlID);
- MIXERCONTROL mixerControl;
-
- // Retrieve one control line for a specified volume-control identifier
- //
- if (!GetLineControl(mixerID, dwControlID, mixerControl))
- {
- return -1;
- }
-
- minVolume = mixerControl.Bounds.dwMinimum;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolumeStepSize
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const
-{
-
- if (_outputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
- return -1;
- }
-
- const UINT mixerID(_outputMixerID);
- const DWORD dwControlID(_speakerState[_outputMixerID].dwVolumeControlID);
- MIXERCONTROL mixerControl;
-
- // Retrieve one control line for a specified volume-control identifier
- //
- if (!GetLineControl(mixerID, _speakerState[mixerID].dwVolumeControlID, mixerControl))
- {
- return -1;
- }
-
- stepSize = static_cast<WebRtc_UWord16> (mixerControl.Metrics.cSteps);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerVolumeIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::SpeakerVolumeIsAvailable(bool& available)
-{
- if (_outputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
- return -1;
- }
-
- available = _speakerState[_outputMixerID].volumeControlIsValid;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerMuteIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::SpeakerMuteIsAvailable(bool& available)
-{
- if (_outputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
- return -1;
- }
-
- available = _speakerState[_outputMixerID].muteControlIsValid;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetSpeakerMute
-//
-// This mute function works a master mute for the output speaker.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::SetSpeakerMute(bool enable)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::SetSpeakerMute(enable=%u)", enable);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_outputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
- return -1;
- }
-
- // Ensure that the selected speaker destination has a valid mute control.
- // If so, its identifier was stored during the enumeration phase which must
- // have taken place since the output mixer handle exists.
- //
- if (!_speakerState[_outputMixerID].muteControlIsValid)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "it is not possible to mute this speaker line");
- return -1;
- }
-
- const DWORD dwControlID(_speakerState[_outputMixerID].dwMuteControlID);
-
- // Set one boolean control value for the specified mute-control
- //
- if (!SetBooleanControlValue(_outputMixerID, dwControlID, enable))
- {
- return -1;
- }
-
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::SpeakerMute(bool& enabled) const
-{
-
- if (_outputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable output mixer exists");
- return -1;
- }
-
- // Ensure that the selected speaker destination has a valid mute control.
- // If so, its identifier was stored during the enumeration phase which must
- // have taken place since the output mixer handle exists.
- //
- if (!_speakerState[_outputMixerID].muteControlIsValid)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "it is not possible to mute this speaker line");
- return -1;
- }
-
- const DWORD dwControlID(_speakerState[_outputMixerID].dwMuteControlID);
- bool value(false);
-
- // Retrieve one boolean control value for a specified mute-control identifier
- //
- if (!GetBooleanControlValue(_outputMixerID, dwControlID, value))
- {
- return -1;
- }
-
- enabled = value;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneMuteIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::MicrophoneMuteIsAvailable(bool& available)
-{
- if (_inputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
- return -1;
- }
-
- available = _microphoneState[_inputMixerID].muteControlIsValid;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneMute
-//
-// This mute function works a master mute for the input microphone.
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::SetMicrophoneMute(bool enable)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::SetMicrophoneMute(enable=%u)", enable);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_inputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
- return -1;
- }
-
- // Ensure that the selected wave-in destinationhas a valid mute control.
- // If so, its identifier was stored during the enumeration phase which must
- // have taken place since the input mixer handle exists.
- //
- if (!_microphoneState[_inputMixerID].muteControlIsValid)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "it is not possible to mute this microphone line");
- return -1;
- }
-
- const DWORD dwControlID(_microphoneState[_inputMixerID].dwMuteControlID);
-
- // Set one boolean control value for the specified mute-control
- //
- if (!SetBooleanControlValue(_inputMixerID, dwControlID, enable))
- {
- return -1;
- }
-
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneMute
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::MicrophoneMute(bool& enabled) const
-{
-
- if (_inputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
- return -1;
- }
-
- // Ensure that the selected wave-in destinationhas a valid mute control.
- // If so, its identifier was stored during the enumeration phase which must
- // have taken place since the input mixer handle exists.
- //
- if (!_microphoneState[_inputMixerID].muteControlIsValid)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "it is not possible to mute this microphone line");
- return -1;
- }
-
- const DWORD dwControlID(_microphoneState[_inputMixerID].dwMuteControlID);
- bool value(false);
-
- // Retrieve one boolean control value for a specified mute-control identifier
- //
- if (!GetBooleanControlValue(_inputMixerID, dwControlID, value))
- {
- return -1;
- }
-
- enabled = value;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneBoostIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::MicrophoneBoostIsAvailable(bool& available)
-{
- if (_inputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
- return -1;
- }
-
- available = _microphoneState[_inputMixerID].onOffControlIsValid;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneBoost
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::SetMicrophoneBoost(bool enable)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::SetMicrophoneBoost(enable=%u)", enable);
-
- CriticalSectionScoped lock(_critSect);
-
- if (_inputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
- return -1;
- }
-
- // Ensure that the selected wave-in destination has a valid boost (on/off) control.
- // If so, its identifier was stored during the enumeration phase which must
- // have taken place since the input mixer handle exists.
- //
- if (!_microphoneState[_inputMixerID].onOffControlIsValid)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no boost control exists for this wave-in line");
- return -1;
- }
-
- const DWORD dwControlID(_microphoneState[_inputMixerID].dwOnOffControlID);
-
- // Set one boolean control value for the specified boost (on/off) control
- //
- if (!SetBooleanControlValue(_inputMixerID, dwControlID, enable))
- {
- return -1;
- }
-
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneBoost
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::MicrophoneBoost(bool& enabled) const
-{
-
- if (_inputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
- return -1;
- }
-
- // Ensure that the selected wave-in destination has a valid boost (on/off) control.
- // If so, its identifier was stored during the enumeration phase which must
- // have taken place since the input mixer handle exists.
- //
- if (!_microphoneState[_inputMixerID].onOffControlIsValid)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no boost control exists for this wave-in line");
- return -1;
- }
-
- const DWORD dwControlID(_microphoneState[_inputMixerID].dwOnOffControlID);
- bool value(false);
-
- // Retrieve one boolean control value for a specified boost-control identifier
- //
- if (!GetBooleanControlValue(_inputMixerID, dwControlID, value))
- {
- return -1;
- }
-
- enabled = value;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolumeIsAvailable
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::MicrophoneVolumeIsAvailable(bool& available)
-{
- if (_inputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
- return -1;
- }
-
- available = _microphoneState[_inputMixerID].volumeControlIsValid;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// SetMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::SetMicrophoneVolume(WebRtc_UWord32 volume)
-{
- CriticalSectionScoped lock(_critSect);
-
- if (_inputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
- return -1;
- }
-
- const UINT mixerID(_inputMixerID);
- const DWORD dwControlID(_microphoneState[_inputMixerID].dwVolumeControlID);
- DWORD dwValue(volume);
-
- // Set one unsigned control value for a specified volume-control identifier
- //
- if (!SetUnsignedControlValue(mixerID, dwControlID, dwValue))
- {
- return -1;
- }
-
- return (0);
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::MicrophoneVolume(WebRtc_UWord32& volume) const
-{
- CriticalSectionScoped lock(_critSect);
-
- if (_inputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
- return -1;
- }
-
- const UINT mixerID(_inputMixerID);
- const DWORD dwControlID(_microphoneState[_inputMixerID].dwVolumeControlID);
- DWORD dwValue(0);
-
- // Retrieve one unsigned control value for a specified volume-control identifier
- //
- if (!GetUnsignedControlValue(mixerID, dwControlID, dwValue))
- {
- return -1;
- }
-
- volume = dwValue;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MaxMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "%s", __FUNCTION__);
-
- if (_inputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
- return -1;
- }
-
- const UINT mixerID(_inputMixerID);
- const DWORD dwControlID(_microphoneState[_inputMixerID].dwVolumeControlID);
- MIXERCONTROL mixerControl;
-
- // Retrieve one control line for a specified volume-control identifier
- //
- if (!GetLineControl(mixerID, dwControlID, mixerControl))
- {
- return -1;
- }
-
- maxVolume = mixerControl.Bounds.dwMaximum;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MinMicrophoneVolume
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::MinMicrophoneVolume(WebRtc_UWord32& minVolume) const
-{
-
- if (_inputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
- return -1;
- }
-
- const UINT mixerID(_inputMixerID);
- const DWORD dwControlID(_microphoneState[_inputMixerID].dwVolumeControlID);
- MIXERCONTROL mixerControl;
-
- // Retrieve one control line for a specified volume-control identifier
- //
- if (!GetLineControl(mixerID, dwControlID, mixerControl))
- {
- return -1;
- }
-
- minVolume = mixerControl.Bounds.dwMinimum;
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneVolumeStepSize
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioMixerManager::MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const
-{
-
- if (_inputMixerHandle == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "no avaliable input mixer exists");
- return -1;
- }
-
- const UINT mixerID(_inputMixerID);
- const DWORD dwControlID(_microphoneState[_inputMixerID].dwVolumeControlID);
- MIXERCONTROL mixerControl;
-
- // Retrieve one control line for a specified volume-control identifier
- //
- if (!GetLineControl(mixerID, dwControlID, mixerControl))
- {
- return -1;
- }
-
- stepSize = static_cast<WebRtc_UWord16> (mixerControl.Metrics.cSteps);
-
- return 0;
-}
-
-// ============================================================================
-// PRIVATE METHODS
-// ============================================================================
-
-// ----------------------------------------------------------------------------
-// Devices
-//
-// A given audio card has one Mixer device associated with it. All of the
-// various components on that card are controlled through that card's one
-// Mixer device.
-// ----------------------------------------------------------------------------
-
-UINT AudioMixerManager::Devices() const
-{
- UINT nDevs = mixerGetNumDevs();
- return nDevs;
-}
-
-// ----------------------------------------------------------------------------
-// DestinationLines
-//
-// # destination lines given mixer ID.
-// ----------------------------------------------------------------------------
-
-UINT AudioMixerManager::DestinationLines(UINT mixId) const
-{
- MIXERCAPS caps;
- if (!GetCapabilities(mixId, caps))
- {
- return 0;
- }
- return (caps.cDestinations);
-}
-// ----------------------------------------------------------------------------
-// DestinationLines
-//
-// # source lines given mixer ID and destination ID.
-// ----------------------------------------------------------------------------
-
-UINT AudioMixerManager::SourceLines(UINT mixId, DWORD destId) const
-{
- MIXERLINE dline;
- if (!GetDestinationLineInfo(mixId, destId, dline))
- {
- return 0;
- }
- return (dline.cConnections);
-}
-
-// ----------------------------------------------------------------------------
-// GetCapabilities
-//
-// Queries a specified mixer device to determine its capabilities.
-// ----------------------------------------------------------------------------
-
-bool AudioMixerManager::GetCapabilities(UINT mixId, MIXERCAPS& caps, bool trace) const
-{
- MMRESULT res;
- MIXERCAPS mcaps;
-
- res = mixerGetDevCaps(mixId, &mcaps, sizeof(MIXERCAPS));
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetDevCaps() failed (err=%d)", res);
- return false;
- }
-
- memcpy(&caps, &mcaps, sizeof(MIXERCAPS));
-
- if (trace)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "Mixer ID %u:", mixId);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "manufacturer ID : %u", caps.wMid);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product ID : %u", caps.wPid);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "version of driver : %u", caps.vDriverVersion);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", WideToUTF8(caps.szPname));
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "misc. support bits : %u", caps.fdwSupport);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "count of destinations: %u (+)", caps.cDestinations);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "===============================================================");
- }
-
- if (caps.cDestinations == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceAudioDevice, _id, "invalid number of mixer destinations");
- return false;
- }
-
- return true;
-}
-
-// ----------------------------------------------------------------------------
-// GetDestinationLineInfo
-// ----------------------------------------------------------------------------
-
-bool AudioMixerManager::GetDestinationLineInfo(UINT mixId, DWORD destId, MIXERLINE& line, bool trace) const
-{
- MMRESULT res;
- MIXERLINE mline;
-
- mline.cbStruct = sizeof(MIXERLINE);
- mline.dwDestination = destId; // max destination index is cDestinations-1
- mline.dwSource = 0; // not set for MIXER_GETLINEINFOF_DESTINATION
-
- // Retrieve information about the specified destination line of a mixer device.
- // Note that we use the mixer ID here and not a handle to an opened mixer.
- // It is not required to open the mixer for enumeration purposes only.
- //
- res = mixerGetLineInfo(reinterpret_cast<HMIXEROBJ>(mixId), &mline, MIXER_OBJECTF_MIXER | MIXER_GETLINEINFOF_DESTINATION);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetLineInfo(MIXER_GETLINEINFOF_DESTINATION) failed (err=%d)", res);
- return false;
- }
-
- memcpy(&line, &mline, sizeof(MIXERLINE));
-
- if (trace)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "> Destination Line ID %u:", destId);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -");
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "destination line index : %u", mline.dwDestination);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "dwLineID : %lu (unique)", mline.dwLineID);
- TraceStatusAndSupportFlags(mline.fdwLine);
- TraceComponentType(mline.dwComponentType);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "count of channels : %u", mline.cChannels);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "# audio source lines : %u (+)", mline.cConnections); // valid only for destinations
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "# controls : %u (*)", mline.cControls); // can be zero
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "short name : %s", WideToUTF8(mline.szShortName));
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "full name : %s", WideToUTF8(mline.szName));
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -");
- TraceTargetType(mline.Target.dwType);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "target device ID : %lu", mline.Target.dwDeviceID);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "manufacturer ID : %u", mline.Target.wMid);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product ID : %u", mline.Target.wPid);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "driver version : %u", mline.Target.vDriverVersion);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "product name : %s", WideToUTF8(mline.Target.szPname));
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "---------------------------------------------------------------");
- }
-
- return true;
-}
-
-// ----------------------------------------------------------------------------
-// GetSourceLineInfo
-// ----------------------------------------------------------------------------
-
-bool AudioMixerManager::GetSourceLineInfo(UINT mixId, DWORD destId, DWORD srcId, MIXERLINE& line, bool trace) const
-{
- MMRESULT res;
- MIXERLINE mline;
-
- mline.cbStruct = sizeof(MIXERLINE);
- mline.dwDestination = destId; // we want the source info for this destination
- mline.dwSource = srcId; // source index (enumerate over these)
-
- // Retrieve information about the specified source line of a mixer device.
- // Note that we use the mixer ID here and not a handle to an opened mixer.
- // It is not required to open the mixer for enumeration purposes only.
- //
- res = mixerGetLineInfo(reinterpret_cast<HMIXEROBJ>(mixId), &mline, MIXER_OBJECTF_MIXER | MIXER_GETLINEINFOF_SOURCE);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetLineInfo(MIXER_GETLINEINFOF_SOURCE) failed (err=%d)", res);
- return false;
- }
-
- memcpy(&line, &mline, sizeof(MIXERLINE));
-
- if (trace)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " >> Source Line ID %u:", srcId);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "destination line index : %u", mline.dwDestination);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "dwSource : %u", mline.dwSource);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "dwLineID : %lu (unique)", mline.dwLineID);
- TraceStatusAndSupportFlags(mline.fdwLine);
- TraceComponentType(mline.dwComponentType);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "# controls : %u (*)", mline.cControls);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "full name : %s", WideToUTF8(mline.szName));
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "- - - - - - - - - - - - - - - - - - - - - - - - - - - - - - - -");
- TraceTargetType(mline.Target.dwType);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "---------------------------------------------------------------");
- }
-
- return true;
-}
-
-// ----------------------------------------------------------------------------
-// GetAllLineControls
-// ----------------------------------------------------------------------------
-
-bool AudioMixerManager::GetAllLineControls(UINT mixId, const MIXERLINE& line, MIXERCONTROL* controlArray, bool trace) const
-{
- // Ensure that we don't try to aquire information if there are no controls for this line
- //
- if (line.cControls == 0)
- return false;
-
- MMRESULT res;
- MIXERLINECONTROLS mlineControls; // contains information about the controls of an audio line
-
- mlineControls.dwLineID = line.dwLineID; // unique audio line identifier
- mlineControls.cControls = line.cControls; // number of controls associated with the line
- mlineControls.pamxctrl = controlArray; // points to the first MIXERCONTROL structure to be filled
- mlineControls.cbStruct = sizeof(MIXERLINECONTROLS);
- mlineControls.cbmxctrl = sizeof(MIXERCONTROL);
-
- // Get information on ALL controls associated with the specified audio line
- //
- res = mixerGetLineControls(reinterpret_cast<HMIXEROBJ>(mixId), &mlineControls, MIXER_OBJECTF_MIXER | MIXER_GETLINECONTROLSF_ALL);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetLineControls(MIXER_GETLINECONTROLSF_ALL) failed (err=%d)", res);
- return false;
- }
-
- if (trace)
- {
- for (UINT c = 0; c < line.cControls; c++)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, " >> Control ID %u:", c);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "dwControlID : %u (unique)", controlArray[c].dwControlID);
- TraceControlType(controlArray[c].dwControlType);
- TraceControlStatusAndSupportFlags(controlArray[c].fdwControl);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "cMultipleItems : %u", controlArray[c].cMultipleItems);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "short name : %s", WideToUTF8(controlArray[c].szShortName));
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "full name : %s", WideToUTF8(controlArray[c].szName));
- if ((controlArray[c].dwControlType & MIXERCONTROL_CT_UNITS_MASK) == MIXERCONTROL_CT_UNITS_SIGNED)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "min signed value : %d", controlArray[c].Bounds.lMinimum);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "max signed value : %d", controlArray[c].Bounds.lMaximum);
- }
- else if ((controlArray[c].dwControlType & MIXERCONTROL_CT_UNITS_MASK) == MIXERCONTROL_CT_UNITS_UNSIGNED ||
- (controlArray[c].dwControlType & MIXERCONTROL_CT_UNITS_MASK) == MIXERCONTROL_CT_UNITS_BOOLEAN)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "min unsigned value : %u", controlArray[c].Bounds.dwMinimum);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "max unsigned value : %u", controlArray[c].Bounds.dwMaximum);
- }
- if (controlArray[c].dwControlType != MIXERCONTROL_CONTROLTYPE_CUSTOM)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "cSteps : %u", controlArray[c].Metrics.cSteps);
- }
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "...............................................................");
- GetControlDetails(mixId, controlArray[c], true);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "...............................................................");
-
- }
- }
-
- return true;
-}
-
-// ----------------------------------------------------------------------------
-// GetLineControls
-// ----------------------------------------------------------------------------
-
-bool AudioMixerManager::GetLineControl(UINT mixId, DWORD dwControlID, MIXERCONTROL& control) const
-{
- MMRESULT res;
- MIXERLINECONTROLS mlineControl;
-
- mlineControl.dwControlID = dwControlID;
- mlineControl.cControls = 1;
- mlineControl.pamxctrl = &control;
- mlineControl.cbStruct = sizeof(MIXERLINECONTROLS);
- mlineControl.cbmxctrl = sizeof(MIXERCONTROL);
-
- // Get information on one controls associated with the specified conrol identifier
- //
- res = mixerGetLineControls(reinterpret_cast<HMIXEROBJ>(mixId), &mlineControl, MIXER_OBJECTF_MIXER | MIXER_GETLINECONTROLSF_ONEBYID);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetLineControls(MIXER_GETLINECONTROLSF_ONEBYID) failed (err=%d)", res);
- return false;
- }
-
- return true;
-}
-
-// ----------------------------------------------------------------------------
-// GetControlDetails
-// ----------------------------------------------------------------------------
-
-bool AudioMixerManager::GetControlDetails(UINT mixId, MIXERCONTROL& controlArray, bool trace) const
-{
- assert(controlArray.cMultipleItems <= MAX_NUMBER_OF_MULTIPLE_ITEMS);
-
- MMRESULT res;
- MIXERCONTROLDETAILS controlDetails;
-
- MIXERCONTROLDETAILS_UNSIGNED valueUnsigned[MAX_NUMBER_OF_MULTIPLE_ITEMS];
- MIXERCONTROLDETAILS_SIGNED valueSigned[MAX_NUMBER_OF_MULTIPLE_ITEMS];
- MIXERCONTROLDETAILS_BOOLEAN valueBoolean[MAX_NUMBER_OF_MULTIPLE_ITEMS];
-
- enum ControlType
- {
- CT_UNITS_UNSIGNED,
- CT_UNITS_SIGNED,
- CT_UNITS_BOOLEAN
- };
-
- ControlType ctype(CT_UNITS_UNSIGNED);
-
- controlDetails.cbStruct = sizeof(MIXERCONTROLDETAILS);
- controlDetails.dwControlID = controlArray.dwControlID; // control identifier
- controlDetails.cChannels = 1; // we need to set values as if they were uniform
- controlDetails.cMultipleItems = controlArray.cMultipleItems; // only nonzero for CONTROLF_MULTIPLE controls
- // can e.g. happen for CONTROLTYPE_MUX
- if (controlDetails.cMultipleItems > MAX_NUMBER_OF_MULTIPLE_ITEMS)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "cMultipleItems > %d", MAX_NUMBER_OF_MULTIPLE_ITEMS);
- controlDetails.cMultipleItems = MAX_NUMBER_OF_MULTIPLE_ITEMS;
- }
-
- if ((controlArray.dwControlType & MIXERCONTROL_CT_UNITS_MASK) == MIXERCONTROL_CT_UNITS_SIGNED)
- {
- ctype = CT_UNITS_SIGNED;
- controlDetails.cbDetails = sizeof(MIXERCONTROLDETAILS_SIGNED);
- controlDetails.paDetails = &valueSigned[0];
- }
- else if ((controlArray.dwControlType & MIXERCONTROL_CT_UNITS_MASK) == MIXERCONTROL_CT_UNITS_UNSIGNED)
- {
- ctype = CT_UNITS_UNSIGNED;
- controlDetails.cbDetails = sizeof(MIXERCONTROLDETAILS_UNSIGNED);
- controlDetails.paDetails = &valueUnsigned[0];
- }
- else if ((controlArray.dwControlType & MIXERCONTROL_CT_UNITS_MASK) == MIXERCONTROL_CT_UNITS_BOOLEAN)
- {
- ctype = CT_UNITS_BOOLEAN;
- controlDetails.cbDetails = sizeof(MIXERCONTROLDETAILS_BOOLEAN);
- controlDetails.paDetails = &valueBoolean[0];
- }
-
- // Retrieve a control's value
- //
- res = mixerGetControlDetails(reinterpret_cast<HMIXEROBJ>(mixId), &controlDetails, MIXER_OBJECTF_MIXER | MIXER_GETCONTROLDETAILSF_VALUE);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetControlDetails(MIXER_GETCONTROLDETAILSF_VALUE) failed (err=%d)", res);
- return false;
- }
-
- if (trace)
- {
- UINT nItems(1);
- nItems = (controlDetails.cMultipleItems > 0 ? controlDetails.cMultipleItems : 1);
- for (UINT i = 0; i < nItems; i++)
- {
- if (ctype == CT_UNITS_SIGNED)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "signed value : %d", valueSigned[i].lValue);
- }
- else if (ctype == CT_UNITS_UNSIGNED)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "unsigned value : %u", valueUnsigned[i].dwValue);
- }
- else if (ctype == CT_UNITS_BOOLEAN)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "boolean value : %u", valueBoolean[i].fValue);
- }
- }
- }
-
- return true;
-}
-
-// ----------------------------------------------------------------------------
-// GetUnsignedControlValue
-// ----------------------------------------------------------------------------
-
-bool AudioMixerManager::GetUnsignedControlValue(UINT mixId, DWORD dwControlID, DWORD& dwValue) const
-{
- MMRESULT res;
- MIXERCONTROLDETAILS controlDetails;
- MIXERCONTROLDETAILS_UNSIGNED valueUnsigned;
-
- controlDetails.dwControlID = dwControlID;
- controlDetails.cbStruct = sizeof(MIXERCONTROLDETAILS);
- controlDetails.cChannels = 1;
- controlDetails.cMultipleItems = 0;
- controlDetails.cbDetails = sizeof(MIXERCONTROLDETAILS_UNSIGNED);
- controlDetails.paDetails = &valueUnsigned;
-
- // Retrieve the unsigned value
- //
- res = mixerGetControlDetails(reinterpret_cast<HMIXEROBJ>(mixId), &controlDetails, MIXER_OBJECTF_MIXER | MIXER_GETCONTROLDETAILSF_VALUE);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetControlDetails(MIXER_GETCONTROLDETAILSF_VALUE) failed (err=%d)", res);
- return false;
- }
-
- // Deliver the retrieved value
- //
- dwValue = valueUnsigned.dwValue;
-
- return true;
-}
-
-// ----------------------------------------------------------------------------
-// SetUnsignedControlValue
-// ----------------------------------------------------------------------------
-
-bool AudioMixerManager::SetUnsignedControlValue(UINT mixId, DWORD dwControlID, DWORD dwValue) const
-{
- WEBRTC_TRACE(kTraceStream, kTraceAudioDevice, _id, "AudioMixerManager::SetUnsignedControlValue(mixId=%u, dwControlID=%d, dwValue=%d)", mixId, dwControlID, dwValue);
-
- MMRESULT res;
- MIXERCONTROLDETAILS controlDetails;
- MIXERCONTROLDETAILS_UNSIGNED valueUnsigned;
-
- controlDetails.dwControlID = dwControlID;
- controlDetails.cbStruct = sizeof(MIXERCONTROLDETAILS);
- controlDetails.cChannels = 1;
- controlDetails.cMultipleItems = 0;
- controlDetails.cbDetails = sizeof(MIXERCONTROLDETAILS_UNSIGNED);
- controlDetails.paDetails = &valueUnsigned;
-
- valueUnsigned.dwValue = dwValue;
-
- // Set the unsigned value
- //
- res = mixerSetControlDetails(reinterpret_cast<HMIXEROBJ>(mixId), &controlDetails, MIXER_OBJECTF_MIXER | MIXER_GETCONTROLDETAILSF_VALUE);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerSetControlDetails(MIXER_GETCONTROLDETAILSF_VALUE) failed (err=%d)", res);
- return false;
- }
-
- return true;
-}
-
-// ----------------------------------------------------------------------------
-// SetBooleanControlValue
-// ----------------------------------------------------------------------------
-
-bool AudioMixerManager::SetBooleanControlValue(UINT mixId, DWORD dwControlID, bool value) const
-{
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "AudioMixerManager::SetBooleanControlValue(mixId=%u, dwControlID=%d, value=%d)", mixId, dwControlID, value);
-
- MMRESULT res;
- MIXERCONTROLDETAILS controlDetails;
- MIXERCONTROLDETAILS_BOOLEAN valueBoolean;
-
- controlDetails.dwControlID = dwControlID;
- controlDetails.cbStruct = sizeof(MIXERCONTROLDETAILS);
- controlDetails.cChannels = 1;
- controlDetails.cMultipleItems = 0;
- controlDetails.cbDetails = sizeof(MIXERCONTROLDETAILS_BOOLEAN);
- controlDetails.paDetails = &valueBoolean;
-
- if (value == true)
- valueBoolean.fValue = TRUE;
- else
- valueBoolean.fValue = FALSE;
-
- // Set the boolean value
- //
- res = mixerSetControlDetails(reinterpret_cast<HMIXEROBJ>(mixId), &controlDetails, MIXER_OBJECTF_MIXER | MIXER_GETCONTROLDETAILSF_VALUE);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerSetControlDetails(MIXER_GETCONTROLDETAILSF_VALUE) failed (err=%d)", res);
- return false;
- }
-
- return true;
-}
-
-// ----------------------------------------------------------------------------
-// GetBooleanControlValue
-// ----------------------------------------------------------------------------
-
-bool AudioMixerManager::GetBooleanControlValue(UINT mixId, DWORD dwControlID, bool& value) const
-{
- MMRESULT res;
- MIXERCONTROLDETAILS controlDetails;
- MIXERCONTROLDETAILS_BOOLEAN valueBoolean;
-
- controlDetails.dwControlID = dwControlID;
- controlDetails.cbStruct = sizeof(MIXERCONTROLDETAILS);
- controlDetails.cChannels = 1;
- controlDetails.cMultipleItems = 0;
- controlDetails.cbDetails = sizeof(MIXERCONTROLDETAILS_BOOLEAN);
- controlDetails.paDetails = &valueBoolean;
-
- // Retrieve the boolean value
- //
- res = mixerGetControlDetails(reinterpret_cast<HMIXEROBJ>(mixId), &controlDetails, MIXER_OBJECTF_MIXER | MIXER_GETCONTROLDETAILSF_VALUE);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetControlDetails(MIXER_GETCONTROLDETAILSF_VALUE) failed (err=%d)", res);
- return false;
- }
-
- // Deliver the retrieved value
- //
- if (valueBoolean.fValue == 0)
- value = false;
- else
- value = true;
-
- return true;
-}
-
-// ----------------------------------------------------------------------------
-// GetSelectedMuxSource
-// ----------------------------------------------------------------------------
-
-bool AudioMixerManager::GetSelectedMuxSource(UINT mixId, DWORD dwControlID, DWORD cMultipleItems, UINT& index) const
-{
- assert(cMultipleItems <= MAX_NUMBER_OF_MULTIPLE_ITEMS);
-
- MMRESULT res;
- MIXERCONTROLDETAILS controlDetails;
- MIXERCONTROLDETAILS_BOOLEAN valueBoolean[MAX_NUMBER_OF_MULTIPLE_ITEMS];
- memset(&valueBoolean, 0, sizeof(valueBoolean));
-
- controlDetails.dwControlID = dwControlID;
- controlDetails.cbStruct = sizeof(MIXERCONTROLDETAILS);
- controlDetails.cChannels = 1;
- controlDetails.cMultipleItems = cMultipleItems;
- controlDetails.cbDetails = sizeof(MIXERCONTROLDETAILS_BOOLEAN);
- controlDetails.paDetails = &valueBoolean;
-
- // Retrieve the boolean values
- //
- res = mixerGetControlDetails(reinterpret_cast<HMIXEROBJ>(mixId), &controlDetails, MIXER_OBJECTF_MIXER | MIXER_GETCONTROLDETAILSF_VALUE);
- if (res != MMSYSERR_NOERROR)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceAudioDevice, _id, "mixerGetControlDetails(MIXER_GETCONTROLDETAILSF_VALUE) failed (err=%d)", res);
- return false;
- }
-
- // Map the current MUX setting to an index corresponding to a source index.
- // e.g. with cMultipleItems = 3,
- // valueBoolean[] = {1,0,0} => index = 2
- // valueBoolean[] = {0,1,0} => index = 1
- // valueBoolean[] = {0,0,1} => index = 0
- //
- // If there is no "1" in the array, we assume index should be 0.
- index = 0;
- for (DWORD i = 0; i < cMultipleItems; i++)
- {
- if (valueBoolean[i].fValue > 0)
- {
- index = (cMultipleItems - 1) - i;
- break;
- }
- }
-
- return true;
-}
-
-// ----------------------------------------------------------------------------
-// TraceStatusAndSupportFlags
-// ----------------------------------------------------------------------------
-
-void AudioMixerManager::TraceStatusAndSupportFlags(DWORD fdwLine) const
-{
- TCHAR buf[128];
-
- StringCchPrintf(buf, 128, TEXT("status & support flags : 0x%x "), fdwLine);
-
- switch (fdwLine)
- {
- case MIXERLINE_LINEF_ACTIVE:
- StringCchCat(buf, 128, TEXT("(ACTIVE DESTINATION)"));
- break;
- case MIXERLINE_LINEF_DISCONNECTED:
- StringCchCat(buf, 128, TEXT("(DISCONNECTED)"));
- break;
- case MIXERLINE_LINEF_SOURCE:
- StringCchCat(buf, 128, TEXT("(INACTIVE SOURCE)"));
- break;
- case MIXERLINE_LINEF_SOURCE | MIXERLINE_LINEF_ACTIVE:
- StringCchCat(buf, 128, TEXT("(ACTIVE SOURCE)"));
- break;
- default:
- StringCchCat(buf, 128, TEXT("(INVALID)"));
- break;
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
-}
-
-// ----------------------------------------------------------------------------
-// TraceComponentType
-// ----------------------------------------------------------------------------
-
-void AudioMixerManager::TraceComponentType(DWORD dwComponentType) const
-{
- TCHAR buf[128];
-
- StringCchPrintf(buf, 128, TEXT("component type : 0x%x "), dwComponentType);
-
- switch (dwComponentType)
- {
- // Destination
- case MIXERLINE_COMPONENTTYPE_DST_UNDEFINED:
- StringCchCat(buf, 128, TEXT("(DST_UNDEFINED)"));
- break;
- case MIXERLINE_COMPONENTTYPE_DST_DIGITAL:
- StringCchCat(buf, 128, TEXT("(DST_DIGITAL)"));
- break;
- case MIXERLINE_COMPONENTTYPE_DST_LINE:
- StringCchCat(buf, 128, TEXT("(DST_LINE)"));
- break;
- case MIXERLINE_COMPONENTTYPE_DST_MONITOR:
- StringCchCat(buf, 128, TEXT("(DST_MONITOR)"));
- break;
- case MIXERLINE_COMPONENTTYPE_DST_SPEAKERS:
- StringCchCat(buf, 128, TEXT("(DST_SPEAKERS)"));
- break;
- case MIXERLINE_COMPONENTTYPE_DST_HEADPHONES:
- StringCchCat(buf, 128, TEXT("(DST_HEADPHONES)"));
- break;
- case MIXERLINE_COMPONENTTYPE_DST_TELEPHONE:
- StringCchCat(buf, 128, TEXT("(DST_TELEPHONE)"));
- break;
- case MIXERLINE_COMPONENTTYPE_DST_WAVEIN:
- StringCchCat(buf, 128, TEXT("(DST_WAVEIN)"));
- break;
- case MIXERLINE_COMPONENTTYPE_DST_VOICEIN:
- StringCchCat(buf, 128, TEXT("(DST_VOICEIN)"));
- break;
- // Source
- case MIXERLINE_COMPONENTTYPE_SRC_UNDEFINED:
- StringCchCat(buf, 128, TEXT("(SRC_UNDEFINED)"));
- break;
- case MIXERLINE_COMPONENTTYPE_SRC_DIGITAL:
- StringCchCat(buf, 128, TEXT("(SRC_DIGITAL)"));
- break;
- case MIXERLINE_COMPONENTTYPE_SRC_LINE:
- StringCchCat(buf, 128, TEXT("(SRC_LINE)"));
- break;
- case MIXERLINE_COMPONENTTYPE_SRC_MICROPHONE:
- StringCchCat(buf, 128, TEXT("(SRC_MICROPHONE)"));
- break;
- case MIXERLINE_COMPONENTTYPE_SRC_SYNTHESIZER:
- StringCchCat(buf, 128, TEXT("(SRC_SYNTHESIZER)"));
- break;
- case MIXERLINE_COMPONENTTYPE_SRC_COMPACTDISC:
- StringCchCat(buf, 128, TEXT("(SRC_COMPACTDISC)"));
- break;
- case MIXERLINE_COMPONENTTYPE_SRC_TELEPHONE:
- StringCchCat(buf, 128, TEXT("(SRC_TELEPHONE)"));
- break;
- case MIXERLINE_COMPONENTTYPE_SRC_PCSPEAKER:
- StringCchCat(buf, 128, TEXT("(SRC_PCSPEAKER)"));
- break;
- case MIXERLINE_COMPONENTTYPE_SRC_WAVEOUT:
- StringCchCat(buf, 128, TEXT("(SRC_WAVEOUT)"));
- break;
- case MIXERLINE_COMPONENTTYPE_SRC_AUXILIARY:
- StringCchCat(buf, 128, TEXT("(SRC_AUXILIARY)"));
- break;
- case MIXERLINE_COMPONENTTYPE_SRC_ANALOG:
- StringCchCat(buf, 128, TEXT("(SRC_ANALOG)"));
- break;
- default:
- StringCchCat(buf, 128, TEXT("(INVALID)"));
- break;
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
-}
-
-// ----------------------------------------------------------------------------
-// TraceTargetType
-// ----------------------------------------------------------------------------
-
-void AudioMixerManager::TraceTargetType(DWORD dwType) const
-{
- TCHAR buf[128];
-
- StringCchPrintf(buf, 128, TEXT("media device type : 0x%x "), dwType);
-
- switch (dwType)
- {
- case MIXERLINE_TARGETTYPE_UNDEFINED:
- StringCchCat(buf, 128, TEXT("(UNDEFINED)"));
- break;
- case MIXERLINE_TARGETTYPE_WAVEOUT:
- StringCchCat(buf, 128, TEXT("(WAVEOUT)"));
- break;
- case MIXERLINE_TARGETTYPE_WAVEIN:
- StringCchCat(buf, 128, TEXT("(WAVEIN)"));
- break;
- case MIXERLINE_TARGETTYPE_MIDIOUT:
- StringCchCat(buf, 128, TEXT("(MIDIOUT)"));
- break;
- case MIXERLINE_TARGETTYPE_MIDIIN:
- StringCchCat(buf, 128, TEXT("(MIDIIN)"));
- break;
- default:
- StringCchCat(buf, 128, TEXT("(INVALID)"));
- break;
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
-}
-
-// ----------------------------------------------------------------------------
-// TraceControlType
-// ----------------------------------------------------------------------------
-
-void AudioMixerManager::TraceControlType(DWORD dwControlType) const
-{
- TCHAR buf[128];
-
- // Class type classification
- //
- StringCchPrintf(buf, 128, TEXT("class type : 0x%x "), dwControlType);
-
- switch (dwControlType & MIXERCONTROL_CT_CLASS_MASK)
- {
- case MIXERCONTROL_CT_CLASS_CUSTOM:
- StringCchCat(buf, 128, TEXT("(CT_CLASS_CUSTOM)"));
- break;
- case MIXERCONTROL_CT_CLASS_METER:
- StringCchCat(buf, 128, TEXT("(CT_CLASS_METER)"));
- break;
- case MIXERCONTROL_CT_CLASS_SWITCH:
- StringCchCat(buf, 128, TEXT("(CT_CLASS_SWITCH)"));
- break;
- case MIXERCONTROL_CT_CLASS_NUMBER:
- StringCchCat(buf, 128, TEXT("(CT_CLASS_NUMBER)"));
- break;
- case MIXERCONTROL_CT_CLASS_SLIDER:
- StringCchCat(buf, 128, TEXT("(CT_CLASS_SLIDER)"));
- break;
- case MIXERCONTROL_CT_CLASS_FADER:
- StringCchCat(buf, 128, TEXT("(CT_CLASS_FADER)"));
- break;
- case MIXERCONTROL_CT_CLASS_TIME:
- StringCchCat(buf, 128, TEXT("(CT_CLASS_TIME)"));
- break;
- case MIXERCONTROL_CT_CLASS_LIST:
- StringCchCat(buf, 128, TEXT("(CT_CLASS_LIST)"));
- break;
- default:
- StringCchCat(buf, 128, TEXT("(INVALID)"));
- break;
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
-
- // Control type (for each class)
- //
- StringCchPrintf(buf, 128, TEXT("control type : 0x%x "), dwControlType);
-
- switch (dwControlType)
- {
- case MIXERCONTROL_CONTROLTYPE_CUSTOM:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_CUSTOM)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_BOOLEANMETER:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_BOOLEANMETER)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_SIGNEDMETER:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_SIGNEDMETER)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_PEAKMETER:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_PEAKMETER)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_UNSIGNEDMETER:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_UNSIGNEDMETER)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_BOOLEAN:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_BOOLEAN)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_ONOFF:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_ONOFF)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_MUTE:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_MUTE)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_MONO:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_MONO)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_LOUDNESS:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_LOUDNESS)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_STEREOENH:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_STEREOENH)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_BASS_BOOST:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_BASS_BOOST)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_BUTTON:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_BUTTON)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_DECIBELS:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_DECIBELS)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_SIGNED:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_SIGNED)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_UNSIGNED:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_UNSIGNED)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_PERCENT:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_PERCENT)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_SLIDER:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_SLIDER)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_PAN:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_PAN)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_QSOUNDPAN:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_QSOUNDPAN)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_FADER:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_FADER)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_VOLUME:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_VOLUME)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_BASS:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_BASS)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_TREBLE:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_TREBLE)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_EQUALIZER:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_EQUALIZER)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_SINGLESELECT:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_SINGLESELECT)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_MUX:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_MUX)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_MULTIPLESELECT:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_MULTIPLESELECT)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_MIXER:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_MIXER)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_MICROTIME:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_MICROTIME)"));
- break;
- case MIXERCONTROL_CONTROLTYPE_MILLITIME:
- StringCchCat(buf, 128, TEXT("(CONTROLTYPE_MILLITIME)"));
- break;
- default:
- StringCchCat(buf, 128, TEXT("(INVALID)"));
- break;
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
-}
-
-// ----------------------------------------------------------------------------
-// TraceControlStatusAndSupportFlags
-//
-// fdwControl
-//
-// Status and support flags for the audio line control. The following values
-// are defined:
-//
-// MIXERCONTROL_CONTROLF_DISABLED
-//
-// The control is disabled, perhaps due to other settings for the mixer hardware,
-// and cannot be used. An application can read current settings from a
-// disabled control, but it cannot apply settings.
-//
-// MIXERCONTROL_CONTROLF_MULTIPLE
-//
-// The control has two or more settings per channel. An equalizer, for example,
-// requires this flag because each frequency band can be set to a different value.
-// An equalizer that affects both channels of a stereo line in a uniform fashion
-// will also specify the MIXERCONTROL_CONTROLF_UNIFORM flag.
-//
-// MIXERCONTROL_CONTROLF_UNIFORM
-//
-// The control acts on all channels of a multichannel line in a uniform fashion.
-// For example, a control that mutes both channels of a stereo line would set
-// this flag. Most MIXERCONTROL_CONTROLTYPE_MUX and
-// MIXERCONTROL_CONTROLTYPE_MIXER controls also specify the
-// MIXERCONTROL_CONTROLF_UNIFORM flag.
-// ----------------------------------------------------------------------------
-
-void AudioMixerManager::TraceControlStatusAndSupportFlags(DWORD fdwControl) const
-{
- TCHAR buf[128];
-
- StringCchPrintf(buf, 128, TEXT("control support flags : 0x%x "), fdwControl);
-
- if (fdwControl & MIXERCONTROL_CONTROLF_DISABLED)
- {
- // The control is disabled, perhaps due to other settings for the mixer hardware,
- // and cannot be used. An application can read current settings from a disabled
- // control, but it cannot apply settings.
- StringCchCat(buf, 128, TEXT("(CONTROLF_DISABLED)"));
- }
-
- if (fdwControl & MIXERCONTROL_CONTROLF_MULTIPLE)
- {
- // The control has two or more settings per channel. An equalizer, for example,
- // requires this flag because each frequency band can be set to a different
- // value. An equalizer that affects both channels of a stereo line in a
- // uniform fashion will also specify the MIXERCONTROL_CONTROLF_UNIFORM flag.
- StringCchCat(buf, 128, TEXT("(CONTROLF_MULTIPLE)"));
- }
-
- if (fdwControl & MIXERCONTROL_CONTROLF_UNIFORM)
- {
- // The control acts on all channels of a multichannel line in a uniform
- // fashion. For example, a control that mutes both channels of a stereo
- // line would set this flag. Most MIXERCONTROL_CONTROLTYPE_MUX and
- // MIXERCONTROL_CONTROLTYPE_MIXER controls also specify the
- // MIXERCONTROL_CONTROLF_UNIFORM flag.
- StringCchCat(buf, 128, TEXT("(CONTROLF_UNIFORM)"));
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
-}
-
-// ----------------------------------------------------------------------------
-// ClearSpeakerState I (II)
-// ----------------------------------------------------------------------------
-
-void AudioMixerManager::ClearSpeakerState(UINT idx)
-{
- _speakerState[idx].dwLineID = 0L;
- _speakerState[idx].dwVolumeControlID = 0L;
- _speakerState[idx].dwMuteControlID = 0L;
- _speakerState[idx].speakerIsValid = false;
- _speakerState[idx].muteControlIsValid = false;
- _speakerState[idx].volumeControlIsValid = false;
-}
-
-// ----------------------------------------------------------------------------
-// ClearSpeakerState II (II)
-// ----------------------------------------------------------------------------
-
-void AudioMixerManager::ClearSpeakerState()
-{
- for (int i = 0; i < MAX_NUMBER_MIXER_DEVICES; i++)
- {
- ClearSpeakerState(i);
- }
-}
-
-// ----------------------------------------------------------------------------
-// SpeakerIsValid
-// ----------------------------------------------------------------------------
-
-bool AudioMixerManager::SpeakerIsValid(UINT idx) const
-{
- return (_speakerState[idx].speakerIsValid);
-}
-
-// ----------------------------------------------------------------------------
-// ValidSpeakers
-//
-// Counts number of valid speaker destinations for all mixer devices.
-// ----------------------------------------------------------------------------
-
-UINT AudioMixerManager::ValidSpeakers() const
-{
- UINT nSpeakers(0);
- for (int i = 0; i < MAX_NUMBER_MIXER_DEVICES; i++)
- {
- if (SpeakerIsValid(i))
- nSpeakers++;
- }
- return nSpeakers;
-}
-
-// ----------------------------------------------------------------------------
-// ClearMicrophoneState I (II)
-// ----------------------------------------------------------------------------
-
-void AudioMixerManager::ClearMicrophoneState(UINT idx)
-{
- _microphoneState[idx].dwLineID = 0L;
- _microphoneState[idx].dwVolumeControlID = 0L;
- _microphoneState[idx].dwMuteControlID = 0L;
- _microphoneState[idx].dwOnOffControlID = 0L;
- _microphoneState[idx].microphoneIsValid = false;
- _microphoneState[idx].muteControlIsValid = false;
- _microphoneState[idx].volumeControlIsValid = false;
- _microphoneState[idx].onOffControlIsValid = false;
-}
-
-// ----------------------------------------------------------------------------
-// ClearMicrophoneState II (II)
-// ----------------------------------------------------------------------------
-
-void AudioMixerManager::ClearMicrophoneState()
-{
- for (int i = 0; i < MAX_NUMBER_MIXER_DEVICES; i++)
- {
- ClearMicrophoneState(i);
- }
-}
-
-// ----------------------------------------------------------------------------
-// MicrophoneIsValid
-// ----------------------------------------------------------------------------
-
-bool AudioMixerManager::MicrophoneIsValid(UINT idx) const
-{
- return (_microphoneState[idx].microphoneIsValid);
-
-}
-
-// ----------------------------------------------------------------------------
-// ValidMicrophones
-//
-// Counts number of valid speaker destinations for all mixer devices.
-// To be valid, a speaker destination line must exist.
-// ----------------------------------------------------------------------------
-
-UINT AudioMixerManager::ValidMicrophones() const
-{
- UINT nMicrophones(0);
- for (int i = 0; i < MAX_NUMBER_MIXER_DEVICES; i++)
- {
- if (MicrophoneIsValid(i))
- nMicrophones++;
- }
- return nMicrophones;
-}
-
-// ----------------------------------------------------------------------------
-// TraceWaveInError
-// ----------------------------------------------------------------------------
-
-void AudioMixerManager::TraceWaveInError(MMRESULT error) const
-{
- TCHAR buf[MAXERRORLENGTH];
- TCHAR msg[MAXERRORLENGTH];
-
- StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
- waveInGetErrorText(error, msg, MAXERRORLENGTH);
- StringCchCat(buf, MAXERRORLENGTH, msg);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
-}
-
-// ----------------------------------------------------------------------------
-// TraceWaveOutError
-// ----------------------------------------------------------------------------
-
-void AudioMixerManager::TraceWaveOutError(MMRESULT error) const
-{
- TCHAR buf[MAXERRORLENGTH];
- TCHAR msg[MAXERRORLENGTH];
-
- StringCchPrintf(buf, MAXERRORLENGTH, TEXT("Error details: "));
- waveOutGetErrorText(error, msg, MAXERRORLENGTH);
- StringCchCat(buf, MAXERRORLENGTH, msg);
- WEBRTC_TRACE(kTraceInfo, kTraceAudioDevice, _id, "%s", WideToUTF8(buf));
-}
-
-// ----------------------------------------------------------------------------
-// WideToUTF8
-// ----------------------------------------------------------------------------
-
-char* AudioMixerManager::WideToUTF8(const TCHAR* src) const {
-#ifdef UNICODE
- const size_t kStrLen = sizeof(_str);
- memset(_str, 0, kStrLen);
- // Get required size (in bytes) to be able to complete the conversion.
- int required_size = WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, 0, 0, 0);
- if (required_size <= kStrLen)
- {
- // Process the entire input string, including the terminating null char.
- if (WideCharToMultiByte(CP_UTF8, 0, src, -1, _str, kStrLen, 0, 0) == 0)
- memset(_str, 0, kStrLen);
- }
- return _str;
-#else
- return const_cast<char*>(src);
-#endif
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_device/main/source/win/audio_mixer_manager_win.h b/trunk/src/modules/audio_device/main/source/win/audio_mixer_manager_win.h
deleted file mode 100644
index da9de47..0000000
--- a/trunk/src/modules/audio_device/main/source/win/audio_mixer_manager_win.h
+++ /dev/null
@@ -1,139 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_WIN_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_WIN_H
-
-#include "typedefs.h"
-#include "audio_device.h"
-#include "critical_section_wrapper.h"
-#include <Windows.h>
-#include <mmsystem.h>
-
-namespace webrtc {
-
-class AudioMixerManager
-{
-public:
- enum { MAX_NUMBER_MIXER_DEVICES = 40 };
- enum { MAX_NUMBER_OF_LINE_CONTROLS = 20 };
- enum { MAX_NUMBER_OF_MULTIPLE_ITEMS = 20 };
- struct SpeakerLineInfo
- {
- DWORD dwLineID;
- bool speakerIsValid;
- DWORD dwVolumeControlID;
- bool volumeControlIsValid;
- DWORD dwMuteControlID;
- bool muteControlIsValid;
- };
- struct MicrophoneLineInfo
- {
- DWORD dwLineID;
- bool microphoneIsValid;
- DWORD dwVolumeControlID;
- bool volumeControlIsValid;
- DWORD dwMuteControlID;
- bool muteControlIsValid;
- DWORD dwOnOffControlID;
- bool onOffControlIsValid;
- };
-public:
- WebRtc_Word32 EnumerateAll();
- WebRtc_Word32 EnumerateSpeakers();
- WebRtc_Word32 EnumerateMicrophones();
- WebRtc_Word32 OpenSpeaker(AudioDeviceModule::WindowsDeviceType device);
- WebRtc_Word32 OpenSpeaker(WebRtc_UWord16 index);
- WebRtc_Word32 OpenMicrophone(AudioDeviceModule::WindowsDeviceType device);
- WebRtc_Word32 OpenMicrophone(WebRtc_UWord16 index);
- WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
- WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
- WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
- WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
- WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
- WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
- WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
- WebRtc_Word32 SetSpeakerMute(bool enable);
- WebRtc_Word32 SpeakerMute(bool& enabled) const;
- WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
- WebRtc_Word32 SetMicrophoneMute(bool enable);
- WebRtc_Word32 MicrophoneMute(bool& enabled) const;
- WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
- WebRtc_Word32 SetMicrophoneBoost(bool enable);
- WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
- WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
- WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
- WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
- WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
- WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
- WebRtc_Word32 MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
- WebRtc_Word32 Close();
- WebRtc_Word32 CloseSpeaker();
- WebRtc_Word32 CloseMicrophone();
- bool SpeakerIsInitialized() const;
- bool MicrophoneIsInitialized() const;
- UINT Devices() const;
-
-private:
- UINT DestinationLines(UINT mixId) const;
- UINT SourceLines(UINT mixId, DWORD destId) const;
- bool GetCapabilities(UINT mixId, MIXERCAPS& caps, bool trace = false) const;
- bool GetDestinationLineInfo(UINT mixId, DWORD destId, MIXERLINE& line, bool trace = false) const;
- bool GetSourceLineInfo(UINT mixId, DWORD destId, DWORD srcId, MIXERLINE& line, bool trace = false) const;
-
- bool GetAllLineControls(UINT mixId, const MIXERLINE& line, MIXERCONTROL* controlArray, bool trace = false) const;
- bool GetLineControl(UINT mixId, DWORD dwControlID, MIXERCONTROL& control) const;
- bool GetControlDetails(UINT mixId, MIXERCONTROL& controlArray, bool trace = false) const;
- bool GetUnsignedControlValue(UINT mixId, DWORD dwControlID, DWORD& dwValue) const;
- bool SetUnsignedControlValue(UINT mixId, DWORD dwControlID, DWORD dwValue) const;
- bool SetBooleanControlValue(UINT mixId, DWORD dwControlID, bool value) const;
- bool GetBooleanControlValue(UINT mixId, DWORD dwControlID, bool& value) const;
- bool GetSelectedMuxSource(UINT mixId, DWORD dwControlID, DWORD cMultipleItems, UINT& index) const;
-
-private:
- void ClearSpeakerState();
- void ClearSpeakerState(UINT idx);
- void ClearMicrophoneState();
- void ClearMicrophoneState(UINT idx);
- bool SpeakerIsValid(UINT idx) const;
- UINT ValidSpeakers() const;
- bool MicrophoneIsValid(UINT idx) const;
- UINT ValidMicrophones() const;
-
- void TraceStatusAndSupportFlags(DWORD fdwLine) const;
- void TraceTargetType(DWORD dwType) const;
- void TraceComponentType(DWORD dwComponentType) const;
- void TraceControlType(DWORD dwControlType) const;
- void TraceControlStatusAndSupportFlags(DWORD fdwControl) const;
- void TraceWaveInError(MMRESULT error) const;
- void TraceWaveOutError(MMRESULT error) const;
- // Converts from wide-char to UTF-8 if UNICODE is defined.
- // Does nothing if UNICODE is undefined.
- char* WideToUTF8(const TCHAR* src) const;
-
-public:
- AudioMixerManager(const WebRtc_Word32 id);
- ~AudioMixerManager();
-
-private:
- CriticalSectionWrapper& _critSect;
- WebRtc_Word32 _id;
- HMIXER _outputMixerHandle;
- UINT _outputMixerID;
- HMIXER _inputMixerHandle;
- UINT _inputMixerID;
- SpeakerLineInfo _speakerState[MAX_NUMBER_MIXER_DEVICES];
- MicrophoneLineInfo _microphoneState[MAX_NUMBER_MIXER_DEVICES];
- mutable char _str[MAXERRORLENGTH];
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_AUDIO_DEVICE_AUDIO_MIXER_MANAGER_H
diff --git a/trunk/src/modules/audio_device/main/test/README.txt b/trunk/src/modules/audio_device/main/test/README.txt
deleted file mode 100644
index 7435ac5..0000000
--- a/trunk/src/modules/audio_device/main/test/README.txt
+++ /dev/null
@@ -1,23 +0,0 @@
-INSTRUCTIONS:
-
-- Start with test #3 (Device enumeration) to get an overview of the available
- audio devices.
-- Next, proceed with test #4 (Device selection) to get more details about
- the supported functions for each audio device.
-- Verify two-way audio in test #5.
- Repeat this test for different selections of playout and recording devices.
-- More detailed tests (volume, mute etc.) can also be performed using #6-#11.
-
-NOTE:
-
-- Some tests requires that the user opens up the audio mixer dialog and
- verifies that a certain action (e.g. Mute ON/OFF) is executed correctly.
-- Files can be recorded during some tests to enable off-line analysis.
-- Full support of 'Default Communication' devices requires Windows 7.
-- If a test consists of several sub tests, press any key to start a new sub test.
-
-KNOWN ISSUES:
-
-- Microphone Boost control is not supported on Windows Vista or Windows 7.
-- Speaker and microphone volume controls will not work as intended on Windows
- Vista if a 'Default Communication' device is selected in any direction.
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/.classpath b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/.classpath
deleted file mode 100644
index 6e9239f..0000000
--- a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/.classpath
+++ /dev/null
@@ -1,7 +0,0 @@
-<?xml version="1.0" encoding="UTF-8"?>
-<classpath>
- <classpathentry kind="src" path="src"/>
- <classpathentry kind="src" path="gen"/>
- <classpathentry kind="con" path="com.android.ide.eclipse.adt.ANDROID_FRAMEWORK"/>
- <classpathentry kind="output" path="bin"/>
-</classpath>
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/AndroidManifest.xml b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/AndroidManifest.xml
deleted file mode 100644
index d8117f5..0000000
--- a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/AndroidManifest.xml
+++ /dev/null
@@ -1,22 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<manifest xmlns:android="http://schemas.android.com/apk/res/android"
- android:versionCode="1"
- android:versionName="1.0" package="org.webrtc.voiceengine.test">
- <application android:icon="@drawable/icon"
- android:label="@string/app_name"
- android:debuggable="true">
- <activity android:label="@string/app_name"
- android:name="AudioDeviceAndroidTest">
- <intent-filter>
- <action android:name="android.intent.action.MAIN" />
- <category android:name="android.intent.category.LAUNCHER" />
- </intent-filter>
- </activity>
- </application>
-
- <uses-sdk android:minSdkVersion="3"></uses-sdk>
- <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS">
- </uses-permission>
- <uses-permission android:name="android.permission.RECORD_AUDIO">
- </uses-permission>
-</manifest>
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/default.properties b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/default.properties
deleted file mode 100644
index 19ddebd..0000000
--- a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/default.properties
+++ /dev/null
@@ -1,11 +0,0 @@
-# This file is automatically generated by Android Tools.
-# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
-#
-# This file must be checked in Version Control Systems.
-#
-# To customize properties used by the Ant build system use,
-# "build.properties", and override values to adapt the script to your
-# project structure.
-
-# Project target.
-target=android-3
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/gen/org/webrtc/voiceengine/test/R.java b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/gen/org/webrtc/voiceengine/test/R.java
deleted file mode 100644
index a295780..0000000
--- a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/gen/org/webrtc/voiceengine/test/R.java
+++ /dev/null
@@ -1,26 +0,0 @@
-/* AUTO-GENERATED FILE. DO NOT MODIFY.
- *
- * This class was automatically generated by the
- * aapt tool from the resource data it found. It
- * should not be modified by hand.
- */
-
-package org.webrtc.voiceengine.test;
-
-public final class R {
- public static final class attr {
- }
- public static final class drawable {
- public static final int icon=0x7f020000;
- }
- public static final class id {
- public static final int Button01=0x7f050000;
- }
- public static final class layout {
- public static final int main=0x7f030000;
- }
- public static final class string {
- public static final int app_name=0x7f040000;
- public static final int run_button=0x7f040001;
- }
-}
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/jni/audio_device_android_test.cc b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/jni/audio_device_android_test.cc
deleted file mode 100644
index f46c6b2..0000000
--- a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/jni/audio_device_android_test.cc
+++ /dev/null
@@ -1,108 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string.h> // memset
-#include <android/log.h>
-
-#include "org_webrtc_voiceengine_test_AudioDeviceAndroidTest.h"
-
-#include "../../../../interface/audio_device.h"
-
-#define LOG_TAG "WebRTC ADM Native"
-
-void api_test();
-void func_test(int);
-
-typedef struct
-{
- // Other
- JavaVM* jvm;
-} AdmData;
-
-static AdmData admData;
-
-jint JNI_OnLoad(JavaVM* vm, void* /*reserved*/)
-{
- if (!vm)
- {
- __android_log_write(ANDROID_LOG_ERROR, LOG_TAG,
- "JNI_OnLoad did not receive a valid VM pointer");
- return -1;
- }
-
- // Get JNI
- JNIEnv* env;
- if (JNI_OK != vm->GetEnv(reinterpret_cast<void**> (&env),
- JNI_VERSION_1_4))
- {
- __android_log_write(ANDROID_LOG_ERROR, LOG_TAG,
- "JNI_OnLoad could not get JNI env");
- return -1;
- }
-
- // Get class to register the native functions with
- // jclass regClass =
- // env->FindClass("org/webrtc/voiceengine/test/AudioDeviceAndroidTest");
- // if (!regClass) {
- // return -1; // Exception thrown
- // }
-
- // Register native functions
- // JNINativeMethod methods[1];
- // methods[0].name = NULL;
- // methods[0].signature = NULL;
- // methods[0].fnPtr = NULL;
- // if (JNI_OK != env->RegisterNatives(regClass, methods, 1))
- // {
- // return -1;
- // }
-
- // Init VoiceEngine data
- memset(&admData, 0, sizeof(admData));
-
- // Store the JVM
- admData.jvm = vm;
-
- return JNI_VERSION_1_4;
-}
-
-JNIEXPORT jboolean JNICALL
-Java_org_webrtc_voiceengine_test_AudioDeviceAndroidTest_NativeInit(JNIEnv * env,
- jclass)
-{
- // Look up and cache any interesting class, field and method IDs for
- // any used java class here
-
- return true;
-}
-
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AudioDeviceAndroidTest_RunTest(JNIEnv *env,
- jobject context,
- jint test)
-{
- // Set instance independent Java objects
- webrtc::AudioDeviceModule::SetAndroidObjects(admData.jvm, env, context);
-
- // Start test
- if (0 == test)
- {
- api_test();
- }
- else
- {
- func_test(test);
- }
-
- // Clear instance independent Java objects
- webrtc::AudioDeviceModule::SetAndroidObjects(NULL, NULL, NULL);
-
- return 0;
-}
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/jni/org_webrtc_voiceengine_test_AudioDeviceAndroidTest.h b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/jni/org_webrtc_voiceengine_test_AudioDeviceAndroidTest.h
deleted file mode 100644
index 5cbc56f..0000000
--- a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/jni/org_webrtc_voiceengine_test_AudioDeviceAndroidTest.h
+++ /dev/null
@@ -1,29 +0,0 @@
-/* DO NOT EDIT THIS FILE - it is machine generated */
-#include <jni.h>
-/* Header for class org_webrtc_voiceengine_test_AudioDeviceAndroidTest */
-
-#ifndef _Included_org_webrtc_voiceengine_test_AudioDeviceAndroidTest
-#define _Included_org_webrtc_voiceengine_test_AudioDeviceAndroidTest
-#ifdef __cplusplus
-extern "C" {
-#endif
-/*
- * Class: org_webrtc_voiceengine_test_AudioDeviceAndroidTest
- * Method: NativeInit
- * Signature: ()Z
- */
-JNIEXPORT jboolean JNICALL Java_org_webrtc_voiceengine_test_AudioDeviceAndroidTest_NativeInit
- (JNIEnv *, jclass);
-
-/*
- * Class: org_webrtc_voiceengine_test_AudioDeviceAndroidTest
- * Method: RunTest
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AudioDeviceAndroidTest_RunTest
- (JNIEnv *, jobject, jint);
-
-#ifdef __cplusplus
-}
-#endif
-#endif
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/drawable/icon.png b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/drawable/icon.png
deleted file mode 100644
index a07c69f..0000000
--- a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/drawable/icon.png
+++ /dev/null
Binary files differ
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/layout/main.xml b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/layout/main.xml
deleted file mode 100644
index 6161f1d..0000000
--- a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/layout/main.xml
+++ /dev/null
@@ -1,12 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android"
- android:orientation="vertical"
- android:layout_width="fill_parent"
- android:layout_height="fill_parent">
-
- <Button android:text="@string/run_button"
- android:id="@+id/Button01"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </Button>
-</LinearLayout>
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/values/strings.xml b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/values/strings.xml
deleted file mode 100644
index bbb6f51..0000000
--- a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/res/values/strings.xml
+++ /dev/null
@@ -1,6 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<resources>
-
- <string name="app_name">WebRTC Audio Device Android Test</string>
-<string name="run_button">Run Test</string>
-</resources>
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/src/org/webrtc/voiceengine/AudioDeviceAndroid.java b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/src/org/webrtc/voiceengine/AudioDeviceAndroid.java
deleted file mode 100644
index 4863168..0000000
--- a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/src/org/webrtc/voiceengine/AudioDeviceAndroid.java
+++ /dev/null
@@ -1 +0,0 @@
-../../../../../../../source/android/org/webrtc/voiceengine/AudioDeviceAndroid.java
diff --git a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/src/org/webrtc/voiceengine/test/AudioDeviceAndroidTest.java b/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/src/org/webrtc/voiceengine/test/AudioDeviceAndroidTest.java
deleted file mode 100644
index b87af46..0000000
--- a/trunk/src/modules/audio_device/main/test/android/audio_device_android_test/src/org/webrtc/voiceengine/test/AudioDeviceAndroidTest.java
+++ /dev/null
@@ -1,69 +0,0 @@
-package org.webrtc.voiceengine.test;
-
-import android.app.Activity;
-import android.media.AudioManager;
-import android.os.Bundle;
-import android.util.Log;
-import android.view.View;
-import android.widget.Button;
-
-public class AudioDeviceAndroidTest extends Activity {
- private Thread _testThread;
-
- /** Called when the activity is first created. */
- @Override
- public void onCreate(Bundle savedInstanceState) {
- super.onCreate(savedInstanceState);
- setContentView(R.layout.main);
-
- final Button buttonStart = (Button) findViewById(R.id.Button01);
- // buttonStart.setWidth(200);
- // button.layout(50, 50, 100, 40);
- buttonStart.setOnClickListener(new View.OnClickListener() {
- public void onClick(View v) {
- _testThread = new Thread(_testProc);
- _testThread.start();
- }
- });
-
- // Suggest to use the voice call audio stream for hardware volume
- // controls
- setVolumeControlStream(AudioManager.STREAM_VOICE_CALL);
-
- DoLog("Started WebRTC Android ADM Test");
- }
-
- private Runnable _testProc = new Runnable() {
- public void run() {
- // TODO(xians), choose test from GUI
- // Select test here, 0 for API test, 1-> for Func tests
- RunTest(5);
- }
- };
-
- private void DoLog(String msg) {
- Log.d("*WebRTC ADM*", msg);
- }
-
- // //////////////// Native function prototypes ////////////////////
-
- // Init wrapper
- private native static boolean NativeInit();
-
- // Function used to call test
- private native int RunTest(int testType);
-
- // Load native library
- static {
- Log.d("*WebRTC ADM*", "Loading audio_device_android_test...");
- System.loadLibrary("audio_device_android_test");
-
- Log.d("*WebRTC ADM*", "Calling native init...");
- if (!NativeInit()) {
- Log.e("*WebRTC ADM*", "Native init failed");
- throw new RuntimeException("Native init failed");
- } else {
- Log.d("*WebRTC ADM*", "Native init successful");
- }
- }
-}
diff --git a/trunk/src/modules/audio_device/main/test/audio_device_test_api.cc b/trunk/src/modules/audio_device/main/test/audio_device_test_api.cc
deleted file mode 100644
index 22ac91a..0000000
--- a/trunk/src/modules/audio_device/main/test/audio_device_test_api.cc
+++ /dev/null
@@ -1,2243 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <string.h>
-
-#include "audio_device_test_defines.h"
-
-#include "../source/audio_device_config.h"
-#include "../source/audio_device_impl.h"
-#include "../source/audio_device_utility.h"
-
-// Helper functions
-#if defined(ANDROID)
-char filenameStr[2][256] =
-{ 0}; // Allow two buffers for those API calls taking two filenames
-int currentStr = 0;
-
-char* GetFilename(char* filename)
-{
- currentStr = !currentStr;
- sprintf(filenameStr[currentStr], "/sdcard/admtest/%s", filename);
- return filenameStr[currentStr];
-}
-const char* GetFilename(const char* filename)
-{
- currentStr = !currentStr;
- sprintf(filenameStr[currentStr], "/sdcard/admtest/%s", filename);
- return filenameStr[currentStr];
-}
-int GetResource(char* resource, char* dest, int destLen)
-{
- currentStr = !currentStr;
- sprintf(filenameStr[currentStr], "/sdcard/admtest/%s", resource);
- strncpy(dest, filenameStr[currentStr], destLen-1);
- return 0;
-}
-char* GetResource(char* resource)
-{
- currentStr = !currentStr;
- sprintf(filenameStr[currentStr], "/sdcard/admtest/%s", resource);
- return filenameStr[currentStr];
-}
-const char* GetResource(const char* resource)
-{
- currentStr = !currentStr;
- sprintf(filenameStr[currentStr], "/sdcard/admtest/%s", resource);
- return filenameStr[currentStr];
-}
-#elif !defined(MAC_IPHONE)
-char* GetFilename(char* filename)
-{
- return filename;
-}
-const char* GetFilename(const char* filename)
-{
- return filename;
-}
-char* GetResource(char* resource)
-{
- return resource;
-}
-const char* GetResource(const char* resource)
-{
- return resource;
-}
-#endif
-
-using namespace webrtc;
-
-// ----------------------------------------------------------------------------
-// AudioEventObserverAPI
-// ----------------------------------------------------------------------------
-
-class AudioEventObserverAPI: public AudioDeviceObserver
-{
-public:
- AudioEventObserverAPI(AudioDeviceModule* audioDevice) :
- _error(kRecordingError),
- _warning(kRecordingWarning),
- _audioDevice(audioDevice)
- {
- }
- ;
- ~AudioEventObserverAPI()
- {
- }
- ;
- virtual void OnErrorIsReported(const ErrorCode error)
- {
- TEST_LOG("\n[*** ERROR ***] => OnErrorIsReported(%d)\n\n", error);
- _error = error;
- // TEST(_audioDevice->StopRecording() == 0);
- // TEST(_audioDevice->StopPlayout() == 0);
- }
- ;
- virtual void OnWarningIsReported(const WarningCode warning)
- {
- TEST_LOG("\n[*** WARNING ***] => OnWarningIsReported(%d)\n\n", warning);
- _warning = warning;
- TEST(_audioDevice->StopRecording() == 0);
- TEST(_audioDevice->StopPlayout() == 0);
- }
- ;
-public:
- ErrorCode _error;
- WarningCode _warning;
-private:
- AudioDeviceModule* _audioDevice;
-};
-
-class AudioTransportAPI: public AudioTransport
-{
-public:
- AudioTransportAPI(AudioDeviceModule* audioDevice) :
- _audioDevice(audioDevice), _recCount(0), _playCount(0)
- {
- }
- ;
-
- ~AudioTransportAPI()
- {
- }
- ;
-
- virtual WebRtc_Word32 RecordedDataIsAvailable(
- const WebRtc_Word8* audioSamples,
- const WebRtc_UWord32 nSamples,
- const WebRtc_UWord8 nBytesPerSample,
- const WebRtc_UWord8 nChannels,
- const WebRtc_UWord32 sampleRate,
- const WebRtc_UWord32 totalDelay,
- const WebRtc_Word32 clockSkew,
- const WebRtc_UWord32 currentMicLevel,
- WebRtc_UWord32& newMicLevel)
- {
- _recCount++;
- if (_recCount % 100 == 0)
- {
- if (nChannels == 1)
- {
- // mono
- TEST_LOG("-");
- } else if ((nChannels == 2) && (nBytesPerSample == 2))
- {
- // stereo but only using one channel
- TEST_LOG("-|");
- } else
- {
- // stereo
- TEST_LOG("--");
- }
- }
-
- return 0;
- }
-
- virtual WebRtc_Word32 NeedMorePlayData(const WebRtc_UWord32 nSamples,
- const WebRtc_UWord8 nBytesPerSample,
- const WebRtc_UWord8 nChannels,
- const WebRtc_UWord32 sampleRate,
- WebRtc_Word8* audioSamples,
- WebRtc_UWord32& nSamplesOut)
- {
- _playCount++;
- if (_playCount % 100 == 0)
- {
- if (nChannels == 1)
- {
- TEST_LOG("+");
- } else
- {
- TEST_LOG("++");
- }
- }
-
- nSamplesOut = 480;
-
- return 0;
- }
- ;
-private:
- AudioDeviceModule* _audioDevice;
- WebRtc_UWord32 _recCount;
- WebRtc_UWord32 _playCount;
-};
-
-int api_test();
-
-
-#if !defined(MAC_IPHONE) && !defined(ANDROID)
-int api_test();
-
-int main(int /*argc*/, char* /*argv*/[])
-{
- api_test();
-}
-#endif
-
-int api_test()
-{
- int i(0);
-
- TEST_LOG("========================================\n");
- TEST_LOG("API Test of the WebRtcAudioDevice Module\n");
- TEST_LOG("========================================\n\n");
-
- ProcessThread* processThread = ProcessThread::CreateProcessThread();
- processThread->Start();
-
- // =======================================================
- // AudioDeviceModuleImpl::Create
- //
- // Windows:
- // if (WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
- // user can select between default (Core) or Wave
- // else
- // user can select between default (Wave) or Wave
- // =======================================================
-
- const WebRtc_Word32 myId = 444;
- AudioDeviceModule* audioDevice(NULL);
-
-#if defined(_WIN32)
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kLinuxAlsaAudio)) == NULL);
-#if defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
- TEST_LOG("WEBRTC_WINDOWS_CORE_AUDIO_BUILD is defined!\n\n");
- // create default implementation (=Core Audio) instance
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
- audioDevice->AddRef();
- TEST(audioDevice->Release() == 0);
- // create non-default (=Wave Audio) instance
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kWindowsWaveAudio)) != NULL);
- audioDevice->AddRef();
- TEST(audioDevice->Release() == 0);
- // explicitly specify usage of Core Audio (same as default)
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kWindowsCoreAudio)) != NULL);
-#else
- TEST_LOG("WEBRTC_WINDOWS_CORE_AUDIO_BUILD is *not* defined!\n");
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kWindowsCoreAudio)) == NULL);
- // create default implementation (=Wave Audio) instance
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
- audioDevice->AddRef();
- TEST(audioDevice->Release() == 0);
- // explicitly specify usage of Wave Audio (same as default)
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kWindowsWaveAudio)) != NULL);
-#endif
-#endif
-
-#if defined(ANDROID)
- // Fails tests
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kWindowsWaveAudio)) == NULL);
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kWindowsCoreAudio)) == NULL);
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kLinuxAlsaAudio)) == NULL);
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kLinuxPulseAudio)) == NULL);
- // Create default implementation instance
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
-#elif defined(WEBRTC_LINUX)
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kWindowsWaveAudio)) == NULL);
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kWindowsCoreAudio)) == NULL);
- // create default implementation instance
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
- audioDevice->AddRef();
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Release() == 0);
- // explicitly specify usage of Pulse Audio (same as default)
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kLinuxPulseAudio)) != NULL);
-#endif
-
-#if defined(WEBRTC_MAC)
- // Fails tests
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kWindowsWaveAudio)) == NULL);
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kWindowsCoreAudio)) == NULL);
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kLinuxAlsaAudio)) == NULL);
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kLinuxPulseAudio)) == NULL);
- // Create default implementation instance
- TEST((audioDevice = AudioDeviceModuleImpl::Create(
- myId, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
-#endif
-
- if (audioDevice == NULL)
- {
-#ifdef _WIN32
- goto Exit;
-#else
- TEST_LOG("Failed creating audio device object! \n");
- return 0;
-#endif
- }
-
- // The ADM is reference counted.
- audioDevice->AddRef();
-
- processThread->RegisterModule(audioDevice);
-
- AudioDeviceModule::AudioLayer audioLayer =
- AudioDeviceModule::kPlatformDefaultAudio;
- TEST(audioDevice->ActiveAudioLayer(&audioLayer) == 0);
- if (audioLayer == AudioDeviceModule::kLinuxAlsaAudio) {
- TEST_LOG("API Test is not available on ALSA. \n");
- processThread->DeRegisterModule(audioDevice);
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Release() == 0);
- return 0;
- }
-
- TEST_LOG("Testing...\n\n");
-
- // =====================
- // RegisterEventObserver
- // =====================
-
- AudioEventObserverAPI* eventObserver =
- new AudioEventObserverAPI(audioDevice);
-
- TEST(audioDevice->RegisterEventObserver(NULL) == 0);
- TEST(audioDevice->RegisterEventObserver(eventObserver) == 0);
- TEST(audioDevice->RegisterEventObserver(NULL) == 0);
-
- // =====================
- // RegisterAudioCallback
- // =====================
-
- AudioTransportAPI* audioTransport = new AudioTransportAPI(audioDevice);
-
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
- TEST(audioDevice->RegisterAudioCallback(audioTransport) == 0);
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
-
- // ====
- // Init
- // ====
-
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Initialized() == false);
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Initialized() == false);
-
- // =========
- // Terminate
- // =========
-
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Initialized() == false);
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Initialized() == false);
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Initialized() == false);
-
- // ------------------------------------------------------------------------
- // Ensure that we keep audio device initialized for all API tests:
- //
- TEST(audioDevice->Init() == 0);
- // ------------------------------------------------------------------------
-
- // goto SHORTCUT;
-
- WebRtc_Word16 nDevices(0);
-
- // ==============
- // PlayoutDevices
- // ==============
-
- TEST((nDevices = audioDevice->PlayoutDevices()) > 0);
- TEST((nDevices = audioDevice->PlayoutDevices()) > 0);
-
- // ================
- // RecordingDevices
- // ================
-
- TEST((nDevices = audioDevice->RecordingDevices()) > 0);
- TEST((nDevices = audioDevice->RecordingDevices()) > 0);
-
- // =================
- // PlayoutDeviceName
- // =================
-
- WebRtc_Word8 name[kAdmMaxDeviceNameSize];
- WebRtc_Word8 guid[kAdmMaxGuidSize];
-
- nDevices = audioDevice->PlayoutDevices();
-
- // fail tests
- TEST(audioDevice->PlayoutDeviceName(-2, name, guid) == -1);
- TEST(audioDevice->PlayoutDeviceName(nDevices, name, guid) == -1);
- TEST(audioDevice->PlayoutDeviceName(0, NULL, guid) == -1);
-
- // bulk tests
- TEST(audioDevice->PlayoutDeviceName(0, name, NULL) == 0);
-#ifdef _WIN32
- TEST(audioDevice->PlayoutDeviceName(-1, name, NULL) == 0); // shall be mapped to 0
-#else
- TEST(audioDevice->PlayoutDeviceName(-1, name, NULL) == -1);
-#endif
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->PlayoutDeviceName(i, name, guid) == 0);
- TEST(audioDevice->PlayoutDeviceName(i, name, NULL) == 0);
- }
-
- // ===================
- // RecordingDeviceName
- // ===================
-
- nDevices = audioDevice->RecordingDevices();
-
- // fail tests
- TEST(audioDevice->RecordingDeviceName(-2, name, guid) == -1);
- TEST(audioDevice->RecordingDeviceName(nDevices, name, guid) == -1);
- TEST(audioDevice->RecordingDeviceName(0, NULL, guid) == -1);
-
- // bulk tests
- TEST(audioDevice->RecordingDeviceName(0, name, NULL) == 0);
-#ifdef _WIN32
- TEST(audioDevice->RecordingDeviceName(-1, name, NULL) == 0); // shall me mapped to 0
-#else
- TEST(audioDevice->RecordingDeviceName(-1, name, NULL) == -1);
-#endif
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->RecordingDeviceName(i, name, guid) == 0);
- TEST(audioDevice->RecordingDeviceName(i, name, NULL) == 0);
- }
-
- // ================
- // SetPlayoutDevice
- // ================
-
- nDevices = audioDevice->PlayoutDevices();
-
- // fail tests
- TEST(audioDevice->SetPlayoutDevice(-1) == -1);
- TEST(audioDevice->SetPlayoutDevice(nDevices) == -1);
-
- // bulk tests
-#ifdef _WIN32
- TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice) == 0);
-#else
- TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultCommunicationDevice) == -1);
- TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice) == -1);
-#endif
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetPlayoutDevice(i) == 0);
- }
-
- // ==================
- // SetRecordingDevice
- // ==================
-
- nDevices = audioDevice->RecordingDevices();
-
- // fail tests
- TEST(audioDevice->SetRecordingDevice(-1) == -1);
- TEST(audioDevice->SetRecordingDevice(nDevices) == -1);
-
- // bulk tests
-#ifdef _WIN32
- TEST(audioDevice->SetRecordingDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->SetRecordingDevice(AudioDeviceModule::kDefaultDevice) == 0);
-#else
- TEST(audioDevice->SetRecordingDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == -1);
- TEST(audioDevice->SetRecordingDevice(
- AudioDeviceModule::kDefaultDevice) == -1);
-#endif
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetRecordingDevice(i) == 0);
- }
-
- // ==================
- // PlayoutIsAvailable
- // ==================
-
- bool available(false);
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
-#ifdef _WIN32
- TEST(audioDevice->SetPlayoutDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- TEST(audioDevice->PlayoutIsInitialized() == false); // availability check should not initialize
-
- TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice) == 0);
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- TEST(audioDevice->PlayoutIsInitialized() == false);
-#endif
-
- nDevices = audioDevice->PlayoutDevices();
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetPlayoutDevice(i) == 0);
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- TEST(audioDevice->PlayoutIsInitialized() == false);
- }
-
- // ====================
- // RecordingIsAvailable
- // ====================
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
-#ifdef _WIN32
- TEST(audioDevice->SetRecordingDevice(AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- TEST(audioDevice->RecordingIsInitialized() == false);
-
- TEST(audioDevice->SetRecordingDevice(AudioDeviceModule::kDefaultDevice) == 0);
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- TEST(audioDevice->RecordingIsInitialized() == false);
-#endif
-
- nDevices = audioDevice->RecordingDevices();
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetRecordingDevice(i) == 0);
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- TEST(audioDevice->RecordingIsInitialized() == false);
- }
-
- // ===========
- // InitPlayout
- // ===========
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial state
- TEST(audioDevice->PlayoutIsInitialized() == false);
-
- // ensure that device must be set before we can initialize
- TEST(audioDevice->InitPlayout() == -1);
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->PlayoutIsInitialized() == true);
-
- // bulk tests
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->PlayoutIsInitialized() == true);
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == -1);
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->PlayoutIsInitialized() == false);
- }
-
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitPlayout() == 0);
- // Sleep is needed for e.g. iPhone since we after stopping then starting may
- // have a hangover time of a couple of ms before initialized.
- AudioDeviceUtility::Sleep(50);
- TEST(audioDevice->PlayoutIsInitialized() == true);
- }
-
- nDevices = audioDevice->PlayoutDevices();
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->SetPlayoutDevice(i) == 0);
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->PlayoutIsInitialized() == true);
- }
- }
- }
-
- TEST(audioDevice->StopPlayout() == 0);
-
- // =============
- // InitRecording
- // =============
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial state
- TEST(audioDevice->RecordingIsInitialized() == false);
-
- // ensure that device must be set before we can initialize
- TEST(audioDevice->InitRecording() == -1);
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->RecordingIsInitialized() == true);
-
- // bulk tests
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->RecordingIsInitialized() == true);
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == -1);
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->RecordingIsInitialized() == false);
- }
-
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitRecording() == 0);
- AudioDeviceUtility::Sleep(50);
- TEST(audioDevice->RecordingIsInitialized() == true);
- }
-
- nDevices = audioDevice->RecordingDevices();
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->SetRecordingDevice(i) == 0);
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->RecordingIsInitialized() == true);
- }
- }
- }
-
- TEST(audioDevice->StopRecording() == 0);
-
- // ============
- // StartPlayout
- // StopPlayout
- // ============
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->Playing() == false);
-
- TEST(audioDevice->StartPlayout() == -1);
- TEST(audioDevice->StopPlayout() == 0);
-
-#ifdef _WIN32
- // kDefaultCommunicationDevice
- TEST(audioDevice->SetPlayoutDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->StartPlayout() == 0);
- TEST(audioDevice->Playing() == true);
- TEST(audioDevice->RegisterAudioCallback(audioTransport) == 0);
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->Playing() == false);
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
- }
-#endif
-
- // repeat test but for kDefaultDevice
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->StartPlayout() == 0);
- TEST(audioDevice->Playing() == true);
- TEST(audioDevice->RegisterAudioCallback(audioTransport) == 0);
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->Playing() == false);
- }
-
- // repeat test for all devices
- nDevices = audioDevice->PlayoutDevices();
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetPlayoutDevice(i) == 0);
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->StartPlayout() == 0);
- TEST(audioDevice->Playing() == true);
- TEST(audioDevice->RegisterAudioCallback(audioTransport) == 0);
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->Playing() == false);
- }
- }
-
- // ==============
- // StartRecording
- // StopRecording
- // ==============
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->Recording() == false);
-
- TEST(audioDevice->StartRecording() == -1);
- TEST(audioDevice->StopRecording() == 0);
-
-#ifdef _WIN32
- // kDefaultCommunicationDevice
- TEST(audioDevice->SetRecordingDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->StartRecording() == 0);
- TEST(audioDevice->Recording() == true);
- TEST(audioDevice->RegisterAudioCallback(audioTransport) == 0);
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->Recording() == false);
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
- }
-#endif
-
- // repeat test but for kDefaultDevice
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->StartRecording() == 0);
- TEST(audioDevice->Recording() == true);
- TEST(audioDevice->RegisterAudioCallback(audioTransport) == 0);
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->Recording() == false);
- }
-
- // repeat test for all devices
- nDevices = audioDevice->RecordingDevices();
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetRecordingDevice(i) == 0);
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->StartRecording() == 0);
- TEST(audioDevice->Recording() == true);
- TEST(audioDevice->RegisterAudioCallback(audioTransport) == 0);
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->Recording() == false);
- }
- }
-
- WebRtc_UWord32 vol(0);
-
-#if defined(_WIN32) && !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
-
- // ================
- // SetWaveOutVolume
- // GetWaveOutVolume
- // ================
-
- // NOTE 1: Windows Wave only!
- // NOTE 2: It seems like the waveOutSetVolume API returns
- // MMSYSERR_NOTSUPPORTED on some Vista machines!
-
- const WebRtc_UWord16 maxVol(0xFFFF);
- WebRtc_UWord16 volL, volR;
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->Playing() == false);
-
- // make dummy test to see if this API is supported
- WebRtc_Word32 works = audioDevice->SetWaveOutVolume(vol, vol);
- WARNING(works == 0);
-
- if (works == 0)
- {
- // set volume without open playout device
- for (vol = 0; vol <= maxVol; vol += (maxVol/5))
- {
- TEST(audioDevice->SetWaveOutVolume(vol, vol) == 0);
- TEST(audioDevice->WaveOutVolume(volL, volR) == 0);
- TEST((volL==vol) && (volR==vol));
- }
-
- // repeat test but this time with an open (default) output device
- TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice) == 0);
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->PlayoutIsInitialized() == true);
- for (vol = 0; vol <= maxVol; vol += (maxVol/5))
- {
- TEST(audioDevice->SetWaveOutVolume(vol, vol) == 0);
- TEST(audioDevice->WaveOutVolume(volL, volR) == 0);
- TEST((volL==vol) && (volR==vol));
- }
-
- // as above but while playout is active
- TEST(audioDevice->StartPlayout() == 0);
- TEST(audioDevice->Playing() == true);
- for (vol = 0; vol <= maxVol; vol += (maxVol/5))
- {
- TEST(audioDevice->SetWaveOutVolume(vol, vol) == 0);
- TEST(audioDevice->WaveOutVolume(volL, volR) == 0);
- TEST((volL==vol) && (volR==vol));
- }
- }
-
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->Playing() == false);
-
-#endif // defined(_WIN32) && !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
- // ==================
- // SpeakerIsAvailable
- // ==================
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->Playing() == false);
- TEST(audioDevice->SpeakerIsInitialized() == false);
-
-#ifdef _WIN32
- // check the kDefaultCommunicationDevice
- TEST(audioDevice->SetPlayoutDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
- // check for availability should not lead to initialization
- TEST(audioDevice->SpeakerIsInitialized() == false);
-#endif
-
- // check the kDefaultDevice
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
- TEST(audioDevice->SpeakerIsInitialized() == false);
-
- // check all availiable devices
- nDevices = audioDevice->PlayoutDevices();
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetPlayoutDevice(i) == 0);
- TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
- TEST(audioDevice->SpeakerIsInitialized() == false);
- }
-
- // ===========
- // InitSpeaker
- // ===========
-
- // NOTE: we call Terminate followed by Init to ensure that any existing output mixer
- // handle is set to NULL. The mixer handle is closed and reopend again for each call to
- // SetPlayoutDevice.
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->Playing() == false);
- TEST(audioDevice->SpeakerIsInitialized() == false);
-
- // kDefaultCommunicationDevice
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
- TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitSpeaker() == 0);
- }
-
- // fail tests
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->StartPlayout() == 0);
- TEST(audioDevice->InitSpeaker() == -1);
- TEST(audioDevice->StopPlayout() == 0);
- }
-
- // kDefaultDevice
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitSpeaker() == 0);
- }
-
- // repeat test for all devices
- nDevices = audioDevice->PlayoutDevices();
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetPlayoutDevice(i) == 0);
- TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitSpeaker() == 0);
- }
- }
-
- // =====================
- // MicrophoneIsAvailable
- // =====================
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->Recording() == false);
- TEST(audioDevice->MicrophoneIsInitialized() == false);
-
-#ifdef _WIN32
- // check the kDefaultCommunicationDevice
- TEST(audioDevice->SetRecordingDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
- // check for availability should not lead to initialization
- TEST(audioDevice->MicrophoneIsInitialized() == false);
-#endif
-
- // check the kDefaultDevice
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
- TEST(audioDevice->MicrophoneIsInitialized() == false);
-
- // check all availiable devices
- nDevices = audioDevice->RecordingDevices();
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetRecordingDevice(i) == 0);
- TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
- TEST(audioDevice->MicrophoneIsInitialized() == false);
- }
-
- // ==============
- // InitMicrophone
- // ==============
-
- // NOTE: we call Terminate followed by Init to ensure that any existing input mixer
- // handle is set to NULL. The mixer handle is closed and reopend again for each call to
- // SetRecordingDevice.
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->Recording() == false);
- TEST(audioDevice->MicrophoneIsInitialized() == false);
-
- // kDefaultCommunicationDevice
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
- TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitMicrophone() == 0);
- }
-
- // fail tests
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->StartRecording() == 0);
- TEST(audioDevice->InitMicrophone() == -1);
- TEST(audioDevice->StopRecording() == 0);
- }
-
- // kDefaultDevice
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitMicrophone() == 0);
- }
-
- // repeat test for all devices
- nDevices = audioDevice->RecordingDevices();
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetRecordingDevice(i) == 0);
- TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitMicrophone() == 0);
- }
- }
-
- // ========================
- // SpeakerVolumeIsAvailable
- // ========================
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->Playing() == false);
- TEST(audioDevice->SpeakerIsInitialized() == false);
-
-#ifdef _WIN32
- // check the kDefaultCommunicationDevice
- TEST(audioDevice->SetPlayoutDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
- // check for availability should not lead to initialization
- TEST(audioDevice->SpeakerIsInitialized() == false);
-#endif
-
- // check the kDefaultDevice
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
- TEST(audioDevice->SpeakerIsInitialized() == false);
-
- // check all availiable devices
- nDevices = audioDevice->PlayoutDevices();
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetPlayoutDevice(i) == 0);
- TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
- TEST(audioDevice->SpeakerIsInitialized() == false);
- }
-
- // ================
- // SetSpeakerVolume
- // SpeakerVolume
- // MaxSpeakerVolume
- // MinSpeakerVolume
- // ================
-
- WebRtc_UWord32 volume(0);
- WebRtc_UWord32 maxVolume(0);
- WebRtc_UWord32 minVolume(0);
- WebRtc_UWord16 stepSize(0);
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->Playing() == false);
- TEST(audioDevice->SpeakerIsInitialized() == false);
-
- // fail tests
- TEST(audioDevice->SetSpeakerVolume(0) == -1); // speaker must be initialized first
- TEST(audioDevice->SpeakerVolume(&volume) == -1);
- TEST(audioDevice->MaxSpeakerVolume(&maxVolume) == -1);
- TEST(audioDevice->MinSpeakerVolume(&minVolume) == -1);
- TEST(audioDevice->SpeakerVolumeStepSize(&stepSize) == -1);
-
-#if defined(_WIN32) && !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
- // test for warning (can e.g. happen on Vista with Wave API)
- TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice) == 0);
- TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitSpeaker() == 0);
- TEST(audioDevice->SetSpeakerVolume(19001) == 0);
- TEST(audioDevice->SpeakerVolume(&volume) == 0);
- WARNING(volume == 19001);
- }
-#endif
-
-#ifdef _WIN32
- // use kDefaultCommunicationDevice and modify/retrieve the volume
- TEST(audioDevice->SetPlayoutDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitSpeaker() == 0);
- TEST(audioDevice->MaxSpeakerVolume(&maxVolume) == 0);
- TEST(audioDevice->MinSpeakerVolume(&minVolume) == 0);
- TEST(audioDevice->SpeakerVolumeStepSize(&stepSize) == 0);
- for (vol = minVolume; vol < (int)maxVolume; vol += 20*stepSize)
- {
- TEST(audioDevice->SetSpeakerVolume(vol) == 0);
- TEST(audioDevice->SpeakerVolume(&volume) == 0);
- TEST((volume == vol) || (volume == vol-1));
- }
- }
-#endif
-
- // use kDefaultDevice and modify/retrieve the volume
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitSpeaker() == 0);
- TEST(audioDevice->MaxSpeakerVolume(&maxVolume) == 0);
- TEST(audioDevice->MinSpeakerVolume(&minVolume) == 0);
- TEST(audioDevice->SpeakerVolumeStepSize(&stepSize) == 0);
- WebRtc_UWord32 step = (maxVolume - minVolume) / 10;
- step = (step < stepSize ? stepSize : step);
- for (vol = minVolume; vol <= maxVolume; vol += step)
- {
- TEST(audioDevice->SetSpeakerVolume(vol) == 0);
- TEST(audioDevice->SpeakerVolume(&volume) == 0);
- TEST((volume == vol) || (volume == vol-1));
- }
- }
-
- // use all (indexed) devices and modify/retrieve the volume
- nDevices = audioDevice->PlayoutDevices();
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetPlayoutDevice(i) == 0);
- TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitSpeaker() == 0);
- TEST(audioDevice->MaxSpeakerVolume(&maxVolume) == 0);
- TEST(audioDevice->MinSpeakerVolume(&minVolume) == 0);
- TEST(audioDevice->SpeakerVolumeStepSize(&stepSize) == 0);
- WebRtc_UWord32 step = (maxVolume - minVolume) / 10;
- step = (step < stepSize ? stepSize : step);
- for (vol = minVolume; vol <= maxVolume; vol += step)
- {
- TEST(audioDevice->SetSpeakerVolume(vol) == 0);
- TEST(audioDevice->SpeakerVolume(&volume) == 0);
- TEST((volume == vol) || (volume == vol-1));
- }
- }
- }
-
- // restore reasonable level
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitSpeaker() == 0);
- TEST(audioDevice->MaxSpeakerVolume(&maxVolume) == 0);
- TEST(audioDevice->SetSpeakerVolume(maxVolume < 10 ?
- maxVolume/3 : maxVolume/10) == 0);
- }
-
- // ======
- // SetAGC
- // AGC
- // ======
-
- // NOTE: The AGC API only enables/disables the AGC. To ensure that it will
- // have an effect, use it in combination with MicrophoneVolumeIsAvailable.
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->Recording() == false);
- TEST(audioDevice->MicrophoneIsInitialized() == false);
- TEST(audioDevice->AGC() == false);
-
- // set/get tests
- TEST(audioDevice->SetAGC(true) == 0);
- TEST(audioDevice->AGC() == true);
- TEST(audioDevice->SetAGC(false) == 0);
- TEST(audioDevice->AGC() == false);
-
- // ===========================
- // MicrophoneVolumeIsAvailable
- // ===========================
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->Recording() == false);
- TEST(audioDevice->MicrophoneIsInitialized() == false);
-
-#ifdef _WIN32
- // check the kDefaultCommunicationDevice
- TEST(audioDevice->SetRecordingDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
- // check for availability should not lead to initialization
- TEST(audioDevice->MicrophoneIsInitialized() == false);
-#endif
-
- // check the kDefaultDevice
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
- TEST(audioDevice->MicrophoneIsInitialized() == false);
-
- // check all availiable devices
- nDevices = audioDevice->RecordingDevices();
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetRecordingDevice(i) == 0);
- TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
- TEST(audioDevice->MicrophoneIsInitialized() == false);
- }
-
- // ===================
- // SetMicrophoneVolume
- // MicrophoneVolume
- // MaxMicrophoneVolume
- // MinMicrophoneVolume
- // ===================
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->Recording() == false);
- TEST(audioDevice->MicrophoneIsInitialized() == false);
-
- // fail tests
- TEST(audioDevice->SetMicrophoneVolume(0) == -1); // must be initialized first
- TEST(audioDevice->MicrophoneVolume(&volume) == -1);
- TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == -1);
- TEST(audioDevice->MinMicrophoneVolume(&minVolume) == -1);
- TEST(audioDevice->MicrophoneVolumeStepSize(&stepSize) == -1);
-
-#if defined(_WIN32) && !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
- // test for warning (can e.g. happen on Vista with Wave API)
- TEST(audioDevice->SetRecordingDevice(AudioDeviceModule::kDefaultDevice) == 0);
- TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitMicrophone() == 0);
- TEST(audioDevice->SetMicrophoneVolume(19001) == 0);
- TEST(audioDevice->MicrophoneVolume(&volume) == 0);
- WARNING(volume == 19001);
- }
-#endif
-
-#ifdef _WIN32
- // initialize kDefaultCommunicationDevice and modify/retrieve the volume
- TEST(audioDevice->SetRecordingDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitMicrophone() == 0);
- TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
- TEST(audioDevice->MinMicrophoneVolume(&minVolume) == 0);
- TEST(audioDevice->MicrophoneVolumeStepSize(&stepSize) == 0);
- for (vol = minVolume; vol < (int)maxVolume; vol += 10*stepSize)
- {
- TEST(audioDevice->SetMicrophoneVolume(vol) == 0);
- TEST(audioDevice->MicrophoneVolume(&volume) == 0);
- TEST((volume == vol) || (volume == vol-1));
- }
- }
-#endif
-
- // reinitialize kDefaultDevice and modify/retrieve the volume
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitMicrophone() == 0);
- TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
- TEST(audioDevice->MinMicrophoneVolume(&minVolume) == 0);
- TEST(audioDevice->MicrophoneVolumeStepSize(&stepSize) == 0);
- for (vol = minVolume; vol < maxVolume; vol += 10 * stepSize)
- {
- TEST(audioDevice->SetMicrophoneVolume(vol) == 0);
- TEST(audioDevice->MicrophoneVolume(&volume) == 0);
- TEST((volume == vol) || (volume == vol-1));
- }
- }
-
- // use all (indexed) devices and modify/retrieve the volume
- nDevices = audioDevice->RecordingDevices();
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetRecordingDevice(i) == 0);
- TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitMicrophone() == 0);
- TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
- TEST(audioDevice->MinMicrophoneVolume(&minVolume) == 0);
- TEST(audioDevice->MicrophoneVolumeStepSize(&stepSize) == 0);
- for (vol = minVolume; vol < maxVolume; vol += 20 * stepSize)
- {
- TEST(audioDevice->SetMicrophoneVolume(vol) == 0);
- TEST(audioDevice->MicrophoneVolume(&volume) == 0);
- TEST((volume == vol) || (volume == vol-1));
- }
- }
- }
-
- // restore reasonable level
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitMicrophone() == 0);
- TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
- TEST(audioDevice->SetMicrophoneVolume(maxVolume/10) == 0);
- }
-
- // ======================
- // SpeakerMuteIsAvailable
- // ======================
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->Playing() == false);
- TEST(audioDevice->SpeakerIsInitialized() == false);
-
-#ifdef _WIN32
- // check the kDefaultCommunicationDevice
- TEST(audioDevice->SetPlayoutDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
- // check for availability should not lead to initialization
- TEST(audioDevice->SpeakerIsInitialized() == false);
-#endif
-
- // check the kDefaultDevice
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
- TEST(audioDevice->SpeakerIsInitialized() == false);
-
- // check all availiable devices
- nDevices = audioDevice->PlayoutDevices();
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetPlayoutDevice(i) == 0);
- TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
- TEST(audioDevice->SpeakerIsInitialized() == false);
- }
-
- // =========================
- // MicrophoneMuteIsAvailable
- // =========================
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->Recording() == false);
- TEST(audioDevice->MicrophoneIsInitialized() == false);
-
-#ifdef _WIN32
- // check the kDefaultCommunicationDevice
- TEST(audioDevice->SetRecordingDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
- // check for availability should not lead to initialization
- #endif
- TEST(audioDevice->MicrophoneIsInitialized() == false);
-
- // check the kDefaultDevice
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
- TEST(audioDevice->MicrophoneIsInitialized() == false);
-
- // check all availiable devices
- nDevices = audioDevice->RecordingDevices();
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetRecordingDevice(i) == 0);
- TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
- TEST(audioDevice->MicrophoneIsInitialized() == false);
- }
-
- // ==========================
- // MicrophoneBoostIsAvailable
- // ==========================
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->Recording() == false);
- TEST(audioDevice->MicrophoneIsInitialized() == false);
-
-#ifdef _WIN32
- // check the kDefaultCommunicationDevice
- TEST(audioDevice->SetRecordingDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
- // check for availability should not lead to initialization
- TEST(audioDevice->MicrophoneIsInitialized() == false);
-#endif
-
- // check the kDefaultDevice
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
- TEST(audioDevice->MicrophoneIsInitialized() == false);
-
- // check all availiable devices
- nDevices = audioDevice->RecordingDevices();
- for (i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetRecordingDevice(i) == 0);
- TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
- TEST(audioDevice->MicrophoneIsInitialized() == false);
- }
-
- // ==============
- // SetSpeakerMute
- // SpeakerMute
- // ==============
-
- bool enabled(false);
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->Playing() == false);
- TEST(audioDevice->SpeakerIsInitialized() == false);
-
- // fail tests
- TEST(audioDevice->SetSpeakerMute(true) == -1); // requires initialization
- TEST(audioDevice->SpeakerMute(&enabled) == -1);
-
-#ifdef _WIN32
- // initialize kDefaultCommunicationDevice and modify/retrieve the mute state
- TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitSpeaker() == 0);
- TEST(audioDevice->SetSpeakerMute(true) == 0);
- TEST(audioDevice->SpeakerMute(&enabled) == 0);
- TEST(enabled == true);
- TEST(audioDevice->SetSpeakerMute(false) == 0);
- TEST(audioDevice->SpeakerMute(&enabled) == 0);
- TEST(enabled == false);
- }
-#endif
-
- // reinitialize kDefaultDevice and modify/retrieve the mute state
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitSpeaker() == 0);
- TEST(audioDevice->SetSpeakerMute(true) == 0);
- TEST(audioDevice->SpeakerMute(&enabled) == 0);
- TEST(enabled == true);
- TEST(audioDevice->SetSpeakerMute(false) == 0);
- TEST(audioDevice->SpeakerMute(&enabled) == 0);
- TEST(enabled == false);
- }
-
- // reinitialize the default device (0) and modify/retrieve the mute state
- TEST(audioDevice->SetPlayoutDevice(0) == 0);
- TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitSpeaker() == 0);
- TEST(audioDevice->SetSpeakerMute(true) == 0);
- TEST(audioDevice->SpeakerMute(&enabled) == 0);
- TEST(enabled == true);
- TEST(audioDevice->SetSpeakerMute(false) == 0);
- TEST(audioDevice->SpeakerMute(&enabled) == 0);
- TEST(enabled == false);
- }
-
- // ==================
- // SetMicrophoneMute
- // MicrophoneMute
- // ==================
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->Recording() == false);
- TEST(audioDevice->MicrophoneIsInitialized() == false);
-
- // fail tests
- TEST(audioDevice->SetMicrophoneMute(true) == -1); // requires initialization
- TEST(audioDevice->MicrophoneMute(&enabled) == -1);
-
-#ifdef _WIN32
- // initialize kDefaultCommunicationDevice and modify/retrieve the mute
- TEST(audioDevice->SetRecordingDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitMicrophone() == 0);
- TEST(audioDevice->SetMicrophoneMute(true) == 0);
- TEST(audioDevice->MicrophoneMute(&enabled) == 0);
- TEST(enabled == true);
- TEST(audioDevice->SetMicrophoneMute(false) == 0);
- TEST(audioDevice->MicrophoneMute(&enabled) == 0);
- TEST(enabled == false);
- }
-#endif
-
- // reinitialize kDefaultDevice and modify/retrieve the mute
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitMicrophone() == 0);
- TEST(audioDevice->SetMicrophoneMute(true) == 0);
- TEST(audioDevice->MicrophoneMute(&enabled) == 0);
- TEST(enabled == true);
- TEST(audioDevice->SetMicrophoneMute(false) == 0);
- TEST(audioDevice->MicrophoneMute(&enabled) == 0);
- TEST(enabled == false);
- }
-
- // reinitialize the default device (0) and modify/retrieve the Mute
- TEST(audioDevice->SetRecordingDevice(0) == 0);
- TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitMicrophone() == 0);
- TEST(audioDevice->SetMicrophoneMute(true) == 0);
- TEST(audioDevice->MicrophoneMute(&enabled) == 0);
- TEST(enabled == true);
- TEST(audioDevice->SetMicrophoneMute(false) == 0);
- TEST(audioDevice->MicrophoneMute(&enabled) == 0);
- TEST(enabled == false);
- }
-
- // ==================
- // SetMicrophoneBoost
- // MicrophoneBoost
- // ==================
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->Recording() == false);
- TEST(audioDevice->MicrophoneIsInitialized() == false);
-
- // fail tests
- TEST(audioDevice->SetMicrophoneBoost(true) == -1); // requires initialization
- TEST(audioDevice->MicrophoneBoost(&enabled) == -1);
-
-#ifdef _WIN32
- // initialize kDefaultCommunicationDevice and modify/retrieve the boost
- TEST(audioDevice->SetRecordingDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitMicrophone() == 0);
- TEST(audioDevice->SetMicrophoneBoost(true) == 0);
- TEST(audioDevice->MicrophoneBoost(&enabled) == 0);
- TEST(enabled == true);
- TEST(audioDevice->SetMicrophoneBoost(false) == 0);
- TEST(audioDevice->MicrophoneBoost(&enabled) == 0);
- TEST(enabled == false);
- }
-#endif
-
- // reinitialize kDefaultDevice and modify/retrieve the boost
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitMicrophone() == 0);
- TEST(audioDevice->SetMicrophoneBoost(true) == 0);
- TEST(audioDevice->MicrophoneBoost(&enabled) == 0);
- TEST(enabled == true);
- TEST(audioDevice->SetMicrophoneBoost(false) == 0);
- TEST(audioDevice->MicrophoneBoost(&enabled) == 0);
- TEST(enabled == false);
- }
-
- // reinitialize the default device (0) and modify/retrieve the boost
- TEST(audioDevice->SetRecordingDevice(0) == 0);
- TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitMicrophone() == 0);
- TEST(audioDevice->SetMicrophoneBoost(true) == 0);
- TEST(audioDevice->MicrophoneBoost(&enabled) == 0);
- TEST(enabled == true);
- TEST(audioDevice->SetMicrophoneBoost(false) == 0);
- TEST(audioDevice->MicrophoneBoost(&enabled) == 0);
- TEST(enabled == false);
- }
-
- // ================
- // SetStereoPlayout
- // StereoPlayout
- // ================
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->Playing() == false);
-
- // fail tests
- TEST(audioDevice->InitPlayout() == -1);
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->PlayoutIsInitialized() == true);
- // must be performed before initialization
- TEST(audioDevice->SetStereoPlayout(true) == -1);
-
- // ensure that we can set the stereo mode for playout
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->PlayoutIsInitialized() == false);
-
- // initialize kDefaultCommunicationDevice and modify/retrieve stereo support
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
- TEST(audioDevice->StereoPlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->SetStereoPlayout(true) == 0);
- TEST(audioDevice->StereoPlayout(&enabled) == 0);
- TEST(enabled == true);
- TEST(audioDevice->SetStereoPlayout(false) == 0);
- TEST(audioDevice->StereoPlayout(&enabled) == 0);
- TEST(enabled == false);
- TEST(audioDevice->SetStereoPlayout(true) == 0);
- TEST(audioDevice->StereoPlayout(&enabled) == 0);
- TEST(enabled == true);
- }
-
- // initialize kDefaultDevice and modify/retrieve stereo support
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->StereoPlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->SetStereoPlayout(true) == 0);
- TEST(audioDevice->StereoPlayout(&enabled) == 0);
- TEST(enabled == true);
- TEST(audioDevice->SetStereoPlayout(false) == 0);
- TEST(audioDevice->StereoPlayout(&enabled) == 0);
- TEST(enabled == false);
- TEST(audioDevice->SetStereoPlayout(true) == 0);
- TEST(audioDevice->StereoPlayout(&enabled) == 0);
- TEST(enabled == true);
- }
-
- // initialize default device (0) and modify/retrieve stereo support
- TEST(audioDevice->SetPlayoutDevice(0) == 0);
- TEST(audioDevice->StereoPlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->SetStereoPlayout(true) == 0);
- TEST(audioDevice->StereoPlayout(&enabled) == 0);
- TEST(enabled == true);
- TEST(audioDevice->SetStereoPlayout(false) == 0);
- TEST(audioDevice->StereoPlayout(&enabled) == 0);
- TEST(enabled == false);
- TEST(audioDevice->SetStereoPlayout(true) == 0);
- TEST(audioDevice->StereoPlayout(&enabled) == 0);
- TEST(enabled == true);
- }
-
- // ==================
- // SetStereoRecording
- // StereoRecording
- // ==================
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->Playing() == false);
-
- // fail tests
- TEST(audioDevice->InitRecording() == -1);
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->RecordingIsInitialized() == true);
- // must be performed before initialization
- TEST(audioDevice->SetStereoRecording(true) == -1);
-
- // ensures that we can set the stereo mode for recording
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->RecordingIsInitialized() == false);
-
- // initialize kDefaultCommunicationDevice and modify/retrieve stereo support
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
- TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->SetStereoRecording(true) == 0);
- TEST(audioDevice->StereoRecording(&enabled) == 0);
- TEST(enabled == true);
- TEST(audioDevice->SetStereoRecording(false) == 0);
- TEST(audioDevice->StereoRecording(&enabled) == 0);
- TEST(enabled == false);
- }
-
- // initialize kDefaultDevice and modify/retrieve stereo support
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->SetStereoRecording(true) == 0);
- TEST(audioDevice->StereoRecording(&enabled) == 0);
- TEST(enabled == true);
- TEST(audioDevice->SetStereoRecording(false) == 0);
- TEST(audioDevice->StereoRecording(&enabled) == 0);
- TEST(enabled == false);
- }
-
- // initialize default device (0) and modify/retrieve stereo support
- TEST(audioDevice->SetRecordingDevice(0) == 0);
- TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->SetStereoRecording(true) == 0);
- TEST(audioDevice->StereoRecording(&enabled) == 0);
- TEST(enabled == true);
- TEST(audioDevice->SetStereoRecording(false) == 0);
- TEST(audioDevice->StereoRecording(&enabled) == 0);
- TEST(enabled == false);
- }
-
- // ===================
- // SetRecordingChannel
- // RecordingChannel
- // ==================
-
- // the user in Win Core Audio
-
- AudioDeviceModule::ChannelType channelType(AudioDeviceModule::kChannelBoth);
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->Playing() == false);
-
- // fail tests
- TEST(audioDevice->SetStereoRecording(false) == 0);
- TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelBoth) == -1);
-
- // initialize kDefaultCommunicationDevice and modify/retrieve stereo support
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
- TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->SetStereoRecording(true) == 0);
- TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelBoth) == 0);
- TEST(audioDevice->RecordingChannel(&channelType) == 0);
- TEST(channelType == AudioDeviceModule::kChannelBoth);
- TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelLeft) == 0);
- TEST(audioDevice->RecordingChannel(&channelType) == 0);
- TEST(channelType == AudioDeviceModule::kChannelLeft);
- TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelRight) == 0);
- TEST(audioDevice->RecordingChannel(&channelType) == 0);
- TEST(channelType == AudioDeviceModule::kChannelRight);
- }
-
- // ================
- // SetPlayoutBuffer
- // PlayoutBuffer
- // ================
-
- AudioDeviceModule::BufferType bufferType;
- WebRtc_UWord16 sizeMS(0);
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->Playing() == false);
- TEST(audioDevice->PlayoutBuffer(&bufferType, &sizeMS) == 0);
-#if defined(_WIN32) || defined(ANDROID) || defined(MAC_IPHONE)
- TEST(bufferType == AudioDeviceModule::kAdaptiveBufferSize);
-#else
- TEST(bufferType == AudioDeviceModule::kFixedBufferSize);
-#endif
-
- // fail tests
- TEST(audioDevice->InitPlayout() == -1); // must set device first
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->PlayoutIsInitialized() == true);
- TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kAdaptiveBufferSize,
- 100) == -1);
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kFixedBufferSize,
- kAdmMinPlayoutBufferSizeMs-1) == -1);
- TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kFixedBufferSize,
- kAdmMaxPlayoutBufferSizeMs+1) == -1);
-
- // bulk tests (all should be successful)
- TEST(audioDevice->PlayoutIsInitialized() == false);
-#ifdef _WIN32
- TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kAdaptiveBufferSize,
- 0) == 0);
- TEST(audioDevice->PlayoutBuffer(&bufferType, &sizeMS) == 0);
- TEST(bufferType == AudioDeviceModule::kAdaptiveBufferSize);
- TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kAdaptiveBufferSize,
- 10000) == 0);
- TEST(audioDevice->PlayoutBuffer(&bufferType, &sizeMS) == 0);
- TEST(bufferType == AudioDeviceModule::kAdaptiveBufferSize);
-#endif
-#if defined(ANDROID) || defined(MAC_IPHONE)
- TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kFixedBufferSize,
- kAdmMinPlayoutBufferSizeMs) == -1);
-#else
- TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kFixedBufferSize,
- kAdmMinPlayoutBufferSizeMs) == 0);
- TEST(audioDevice->PlayoutBuffer(&bufferType, &sizeMS) == 0);
- TEST(bufferType == AudioDeviceModule::kFixedBufferSize);
- TEST(sizeMS == kAdmMinPlayoutBufferSizeMs);
- TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kFixedBufferSize,
- kAdmMaxPlayoutBufferSizeMs) == 0);
- TEST(audioDevice->PlayoutBuffer(&bufferType, &sizeMS) == 0);
- TEST(bufferType == AudioDeviceModule::kFixedBufferSize);
- TEST(sizeMS == kAdmMaxPlayoutBufferSizeMs);
- TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kFixedBufferSize,
- 100) == 0);
- TEST(audioDevice->PlayoutBuffer(&bufferType, &sizeMS) == 0);
- TEST(bufferType == AudioDeviceModule::kFixedBufferSize);
- TEST(sizeMS == 100);
-#endif
-
-#ifdef _WIN32
- // restore default
- TEST(audioDevice->SetPlayoutBuffer(AudioDeviceModule::kAdaptiveBufferSize,
- 0) == 0);
- TEST(audioDevice->PlayoutBuffer(&bufferType, &sizeMS) == 0);
-#endif
-
- // ============
- // PlayoutDelay
- // ============
-
- // NOTE: this API is better tested in a functional test
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->Playing() == false);
-
- // bulk tests
- TEST(audioDevice->PlayoutDelay(&sizeMS) == 0);
- TEST(audioDevice->PlayoutDelay(&sizeMS) == 0);
-
- // ==============
- // RecordingDelay
- // ==============
-
- // NOTE: this API is better tested in a functional test
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->Recording() == false);
-
- // bulk tests
- TEST(audioDevice->RecordingDelay(&sizeMS) == 0);
- TEST(audioDevice->RecordingDelay(&sizeMS) == 0);
-
- // =======
- // CPULoad
- // =======
-
- // NOTE: this API is better tested in a functional test
-
- WebRtc_UWord16 load(0);
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
-
- // bulk tests
-#ifdef _WIN32
- TEST(audioDevice->CPULoad(&load) == 0);
- TEST(load == 0);
-#else
- TEST(audioDevice->CPULoad(&load) == -1);
-#endif
-
- // ===========================
- // StartRawOutputFileRecording
- // StopRawOutputFileRecording
- // ===========================
-
- // NOTE: this API is better tested in a functional test
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->Playing() == false);
-
- // fail tests
- TEST(audioDevice->StartRawOutputFileRecording(NULL) == -1);
-
- // bulk tests
- TEST(audioDevice->StartRawOutputFileRecording(
- GetFilename("raw_output_not_playing.pcm")) == 0);
- TEST(audioDevice->StopRawOutputFileRecording() == 0);
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_COMMUNICATION_DEVICE) == 0);
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->StartPlayout() == 0);
- TEST(audioDevice->StartRawOutputFileRecording(
- GetFilename("raw_output_playing.pcm")) == 0);
- AudioDeviceUtility::Sleep(100);
- TEST(audioDevice->StopRawOutputFileRecording() == 0);
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->StartRawOutputFileRecording(
- GetFilename("raw_output_not_playing.pcm")) == 0);
- TEST(audioDevice->StopRawOutputFileRecording() == 0);
-
- // results after this test:
- //
- // - size of raw_output_not_playing.pcm shall be 0
- // - size of raw_output_playing.pcm shall be > 0
-
- // ==========================
- // StartRawInputFileRecording
- // StopRawInputFileRecording
- // ==========================
-
- // NOTE: this API is better tested in a functional test
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->Playing() == false);
-
- // fail tests
- TEST(audioDevice->StartRawInputFileRecording(NULL) == -1);
-
- // bulk tests
- TEST(audioDevice->StartRawInputFileRecording(
- GetFilename("raw_input_not_recording.pcm")) == 0);
- TEST(audioDevice->StopRawInputFileRecording() == 0);
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->StartRecording() == 0);
- TEST(audioDevice->StartRawInputFileRecording(
- GetFilename("raw_input_recording.pcm")) == 0);
- AudioDeviceUtility::Sleep(100);
- TEST(audioDevice->StopRawInputFileRecording() == 0);
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->StartRawInputFileRecording(
- GetFilename("raw_input_not_recording.pcm")) == 0);
- TEST(audioDevice->StopRawInputFileRecording() == 0);
-
- // results after this test:
- //
- // - size of raw_input_not_recording.pcm shall be 0
- // - size of raw_input_not_recording.pcm shall be > 0
-
- // ===================
- // RecordingSampleRate
- // ===================
-
- WebRtc_UWord32 sampleRate(0);
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
-
- // bulk tests
- TEST(audioDevice->RecordingSampleRate(&sampleRate) == 0);
-#if defined(_WIN32) && !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
- TEST(sampleRate == 48000);
-#elif defined(ANDROID)
- TEST_LOG("Recording sample rate is %u\n\n", sampleRate);
- TEST((sampleRate == 44000) || (sampleRate == 16000));
-#elif defined(MAC_IPHONE)
- TEST_LOG("Recording sample rate is %u\n\n", sampleRate);
- TEST((sampleRate == 44000) || (sampleRate == 16000) || (sampleRate == 8000));
-#endif
-
- // @TODO(xians) - add tests for all platforms here...
-
- // =================
- // PlayoutSampleRate
- // =================
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
-
- // bulk tests
- TEST(audioDevice->PlayoutSampleRate(&sampleRate) == 0);
-#if defined(_WIN32) && !defined(WEBRTC_WINDOWS_CORE_AUDIO_BUILD)
- TEST(sampleRate == 48000);
-#elif defined(ANDROID)
- TEST_LOG("Playout sample rate is %u\n\n", sampleRate);
- TEST((sampleRate == 44000) || (sampleRate == 16000));
-#elif defined(MAC_IPHONE)
- TEST_LOG("Playout sample rate is %u\n\n", sampleRate);
- TEST((sampleRate == 44000) || (sampleRate == 16000) || (sampleRate == 8000));
-#endif
-
- // ==========================
- // ResetAudioDevice
- // ==========================
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->Playing() == false);
- TEST(audioDevice->RecordingIsInitialized() == false);
- TEST(audioDevice->Recording() == false);
-
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
- TEST(audioDevice->SetRecordingDevice(MACRO_DEFAULT_DEVICE) == 0);
-
-#if defined(MAC_IPHONE)
- // Not playing or recording, should just return 0
- TEST(audioDevice->ResetAudioDevice() == 0);
-
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->StartRecording() == 0);
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->StartPlayout() == 0);
- for (int l=0; l<20; ++l)
- {
- TEST_LOG("Resetting sound device several time with pause %d ms\n", l);
- TEST(audioDevice->ResetAudioDevice() == 0);
- AudioDeviceUtility::Sleep(l);
- }
-#else
- // Fail tests
- TEST(audioDevice->ResetAudioDevice() == -1);
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->StartRecording() == 0);
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->StartPlayout() == 0);
- TEST(audioDevice->ResetAudioDevice() == -1);
-#endif
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->StopPlayout() == 0);
-
- // ==========================
- // SetPlayoutSpeaker
- // ==========================
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Init() == 0);
-
- // check initial states
- TEST(audioDevice->Initialized() == true);
- TEST(audioDevice->PlayoutIsInitialized() == false);
- TEST(audioDevice->Playing() == false);
-
- TEST(audioDevice->SetPlayoutDevice(MACRO_DEFAULT_DEVICE) == 0);
-
- bool loudspeakerOn(false);
-#if defined(MAC_IPHONE)
- // Not playing or recording, should just return a success
- TEST(audioDevice->SetLoudspeakerStatus(true) == 0);
- TEST(audioDevice->GetLoudspeakerStatus(loudspeakerOn) == 0);
- TEST(loudspeakerOn == true);
- TEST(audioDevice->SetLoudspeakerStatus(false) == 0);
- TEST(audioDevice->GetLoudspeakerStatus(loudspeakerOn) == 0);
- TEST(loudspeakerOn == false);
-
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->StartPlayout() == 0);
- TEST(audioDevice->SetLoudspeakerStatus(true) == 0);
- TEST(audioDevice->GetLoudspeakerStatus(loudspeakerOn) == 0);
- TEST(loudspeakerOn == true);
- TEST(audioDevice->SetLoudspeakerStatus(false) == 0);
- TEST(audioDevice->GetLoudspeakerStatus(loudspeakerOn) == 0);
- TEST(loudspeakerOn == false);
-
-#else
- // Fail tests
- TEST(audioDevice->SetLoudspeakerStatus(true) == -1);
- TEST(audioDevice->SetLoudspeakerStatus(false) == -1);
- TEST(audioDevice->SetLoudspeakerStatus(true) == -1);
- TEST(audioDevice->SetLoudspeakerStatus(false) == -1);
-
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->StartPlayout() == 0);
- TEST(audioDevice->GetLoudspeakerStatus(&loudspeakerOn) == -1);
-#endif
- TEST(audioDevice->StopPlayout() == 0);
-
-#ifdef _WIN32
- Exit:
-#endif
-
- // ------------------------------------------------------------------------
- // Terminate the module when all tests are done:
- //
- TEST(audioDevice->Terminate() == 0);
- // ------------------------------------------------------------------------
-
- // ===================================================
- // AudioDeviceModuleImpl::Destroy
- // ===================================================
-
-
- // release the ProcessThread object
- if (processThread)
- {
- processThread->DeRegisterModule(audioDevice);
- processThread->Stop();
- ProcessThread::DestroyProcessThread(processThread);
- }
-
- // delete the event observer
- if (eventObserver)
- {
- delete eventObserver;
- eventObserver = NULL;
- }
-
- // delete the audio transport
- if (audioTransport)
- {
- delete audioTransport;
- audioTransport = NULL;
- }
-
- // release the AudioDeviceModule object
- if (audioDevice) {
- TEST(audioDevice->Release() == 0);
- }
-
- TEST_LOG("\n");
- PRINT_TEST_RESULTS;
-
- return 0;
-}
diff --git a/trunk/src/modules/audio_device/main/test/audio_device_test_defines.h b/trunk/src/modules/audio_device/main/test/audio_device_test_defines.h
deleted file mode 100644
index 5c5f4e2..0000000
--- a/trunk/src/modules/audio_device/main/test/audio_device_test_defines.h
+++ /dev/null
@@ -1,116 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_TEST_DEFINES_H
-#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_TEST_DEFINES_H
-
-#include "common_types.h"
-#include "audio_device.h"
-#include "process_thread.h"
-#include "trace.h"
-
-#ifdef _WIN32
-#define MACRO_DEFAULT_DEVICE AudioDeviceModule::kDefaultDevice
-#define MACRO_DEFAULT_COMMUNICATION_DEVICE AudioDeviceModule::kDefaultCommunicationDevice
-#else
-#define MACRO_DEFAULT_DEVICE 0
-#define MACRO_DEFAULT_COMMUNICATION_DEVICE 0
-#endif
-
-#ifdef ANDROID
-#include <android/log.h>
-#define LOG_TAG "WebRtc ADM TEST"
-#define TEST_LOG(...) __android_log_print(ANDROID_LOG_DEBUG, LOG_TAG, __VA_ARGS__)
-#define TEST_LOG_ERROR(...) __android_log_print(ANDROID_LOG_ERROR, LOG_TAG, __VA_ARGS__)
-#else
-#define TEST_LOG printf
-#define TEST_LOG_ERROR(...) fprintf(stderr, __VA_ARGS__)
-#endif
-
-static int testCount = 0;
-static int errorCount = 0;
-static int warningCount = 0;
-
-#define RESET_TEST \
- do { \
- testCount = 0; \
- errorCount = 0; \
- warningCount = 0; \
- } while(0) \
-
-#define PRINT_ERR_MSG(msg) \
- do { \
- TEST_LOG_ERROR("Error at line %i of %s\n%s", \
- __LINE__, __FILE__, msg); \
- } while(0)
-
-#define WARNING(expr) \
- do { \
- testCount++; \
- if (!(expr)) { \
- TEST_LOG_ERROR("WARNING #%d: at line %i\n\n", \
- warningCount+1, __LINE__); \
- warningCount++; \
- } \
- } while(0)
-
-
-#define TEST(expr) \
- do { \
- testCount++; \
- if (!(expr)) { \
- PRINT_ERR_MSG("Assertion failed: " #expr "\n\n"); \
- errorCount++; \
- } \
- } while(0)
-
-#define TEST_ERR(expr, err) \
- do { \
- testCount++; \
- if (!(expr)) { \
- PRINT_ERR_MSG("Assertion failed: " #expr "\n\n"); \
- errorCount++; \
- } \
- if (audioDevice->LastError() != err) { \
- PRINT_ERR_MSG("Assertion failed: " #err "\n\n"); \
- errorCount++; \
- } \
- } while(0)
-
-
-#define PRINT_TEST_RESULTS \
- do { \
- TEST_LOG("\n>> %i tests completed <<\n", testCount); \
- if (errorCount > 0) { \
- TEST_LOG(">> %i FAILED! <<\n\n", errorCount); \
- } \
- else if (warningCount > 0) \
- { \
- TEST_LOG(">> ALL PASSED (with %d warnings) <<\n\n", \
- warningCount); \
- } \
- else \
- { \
- TEST_LOG(">> ALL PASSED <<\n\n"); \
- } \
- } while(0)
-
-// Helper functions
-// For iPhone, they are defined in iPhone specific test code.
-// For Android, they are defined in API test only (since both
-// API and Func tests are built into the same lib).
-// For other, they are defined in both API test and Func test.
-char* GetFilename(char* filename);
-const char* GetFilename(const char* filename);
-char* GetResource(char* resource);
-const char* GetResource(const char* resource);
-
-#endif // WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_TEST_DEFINES_H
-
diff --git a/trunk/src/modules/audio_device/main/test/audio_device_test_func.cc b/trunk/src/modules/audio_device/main/test/audio_device_test_func.cc
deleted file mode 100644
index 87a70aa..0000000
--- a/trunk/src/modules/audio_device/main/test/audio_device_test_func.cc
+++ /dev/null
@@ -1,164 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include "audio_device_test_defines.h"
-#include "func_test_manager.h"
-
-#ifndef __GNUC__
-// Disable warning message 4996 ('scanf': This function or variable may be unsafe)
-#pragma warning( disable : 4996 )
-#endif
-
-using namespace webrtc;
-
-int func_test(int);
-
-// ----------------------------------------------------------------------------
-// main()
-// ----------------------------------------------------------------------------
-
-#if !defined(MAC_IPHONE) && !defined(ANDROID)
-int main(int /*argc*/, char* /*argv*/[])
-{
- func_test(0);
-}
-#endif
-
-// ----------------------------------------------------------------------------
-// func_test()
-// ----------------------------------------------------------------------------
-
-int func_test(int sel)
-{
- TEST_LOG("=========================================\n");
- TEST_LOG("Func Test of the WebRtcAudioDevice Module\n");
- TEST_LOG("=========================================\n\n");
-
- // Initialize the counters here to get rid of "unused variables" warnings.
- testCount = 0;
- errorCount = 0;
- warningCount = 0;
-
- FuncTestManager funcMgr;
-
- funcMgr.Init();
-
- bool quit(false);
-
- while (!quit)
- {
- TEST_LOG("---------------------------------------\n");
- TEST_LOG("Select type of test\n\n");
- TEST_LOG(" (0) Quit\n");
- TEST_LOG(" (1) All\n");
- TEST_LOG("- - - - - - - - - - - - - - - - - - - -\n");
- TEST_LOG(" (2) Audio-layer selection\n");
- TEST_LOG(" (3) Device enumeration\n");
- TEST_LOG(" (4) Device selection\n");
- TEST_LOG(" (5) Audio transport\n");
- TEST_LOG(" (6) Speaker volume\n");
- TEST_LOG(" (7) Microphone volume\n");
- TEST_LOG(" (8) Speaker mute\n");
- TEST_LOG(" (9) Microphone mute\n");
- TEST_LOG(" (10) Microphone boost\n");
- TEST_LOG(" (11) Microphone AGC\n");
- TEST_LOG(" (12) Loopback measurements\n");
- TEST_LOG(" (13) Device removal\n");
- TEST_LOG(" (14) Advanced mobile device API\n");
- TEST_LOG(" (66) XTEST\n");
- TEST_LOG("- - - - - - - - - - - - - - - - - - - -\n");
- TEST_LOG("\n: ");
-
- int selection(0);
- enum TestType testType(TTInvalid);
-
-SHOW_MENU:
-
- if (sel > 0)
- {
- selection = sel;
- }
- else
- {
- if (scanf("%d", &selection) < 0) {
- perror("Failed to get selection.");
- }
- }
-
- switch (selection)
- {
- case 0:
- quit = true;
- break;
- case 1:
- testType = TTAll;
- break;
- case 2:
- testType = TTAudioLayerSelection;
- break;
- case 3:
- testType = TTDeviceEnumeration;
- break;
- case 4:
- testType = TTDeviceSelection;
- break;
- case 5:
- testType = TTAudioTransport;
- break;
- case 6:
- testType = TTSpeakerVolume;
- break;
- case 7:
- testType = TTMicrophoneVolume;
- break;
- case 8:
- testType = TTSpeakerMute;
- break;
- case 9:
- testType = TTMicrophoneMute;
- break;
- case 10:
- testType = TTMicrophoneBoost;
- break;
- case 11:
- testType = TTMicrophoneAGC;
- break;
- case 12:
- testType = TTLoopback;
- break;
- case 13:
- testType = TTDeviceRemoval;
- break;
- case 14:
- testType = TTMobileAPI;
- break;
- case 66:
- testType = TTTest;
- break;
- default:
- testType = TTInvalid;
- TEST_LOG(": ");
- goto SHOW_MENU;
- break;
- }
-
- funcMgr.DoTest(testType);
-
- if (sel > 0)
- {
- quit = true;
- }
- }
-
- funcMgr.Close();
-
- return 0;
-}
diff --git a/trunk/src/modules/audio_device/main/test/func_test_manager.cc b/trunk/src/modules/audio_device/main/test/func_test_manager.cc
deleted file mode 100644
index 8bc7646..0000000
--- a/trunk/src/modules/audio_device/main/test/func_test_manager.cc
+++ /dev/null
@@ -1,2745 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <ctype.h>
-#include <cassert>
-#include <string.h>
-
-#include "func_test_manager.h"
-#include "testsupport/fileutils.h"
-
-#include "../source/audio_device_config.h"
-#include "../source/audio_device_impl.h"
-
-#ifndef __GNUC__
-// Disable warning message ('sprintf': name was marked as #pragma deprecated)
-#pragma warning( disable : 4995 )
-// Disable warning message 4996 ('scanf': This function or variable may be unsafe)
-#pragma warning( disable : 4996 )
-#endif
-
-const char* RecordedMicrophoneFile = "recorded_microphone_mono_48.pcm";
-const char* RecordedMicrophoneVolumeFile =
-"recorded_microphone_volume_mono_48.pcm";
-const char* RecordedMicrophoneMuteFile = "recorded_microphone_mute_mono_48.pcm";
-const char* RecordedMicrophoneBoostFile =
-"recorded_microphone_boost_mono_48.pcm";
-const char* RecordedMicrophoneAGCFile = "recorded_microphone_AGC_mono_48.pcm";
-const char* RecordedSpeakerFile = "recorded_speaker_48.pcm";
-
-struct AudioPacket
-{
- WebRtc_UWord8 dataBuffer[4 * 960];
- WebRtc_UWord16 nSamples;
- WebRtc_UWord16 nBytesPerSample;
- WebRtc_UWord8 nChannels;
- WebRtc_UWord32 samplesPerSec;
-};
-
-// Helper functions
-#if !defined(MAC_IPHONE) && !defined(ANDROID)
-char* GetFilename(char* filename)
-{
- return filename;
-}
-const char* GetFilename(const char* filename)
-{
- return filename;
-}
-char* GetResource(char* resource)
-{
- return resource;
-}
-const char* GetResource(const char* resource)
-{
- return resource;
-}
-#endif
-
-namespace webrtc
-{
-
-AudioEventObserver::AudioEventObserver(AudioDeviceModule* audioDevice) :
- _audioDevice(audioDevice)
-{
-}
-;
-
-AudioEventObserver::~AudioEventObserver()
-{
-}
-;
-
-void AudioEventObserver::OnErrorIsReported(const ErrorCode error)
-{
- TEST_LOG("\n[*** ERROR ***] => OnErrorIsReported(%d)\n \n", error);
- _error = error;
- // TEST(_audioDevice->StopRecording() == 0);
- // TEST(_audioDevice->StopPlayout() == 0);
-}
-;
-
-
-void AudioEventObserver::OnWarningIsReported(const WarningCode warning)
-{
- TEST_LOG("\n[*** WARNING ***] => OnWarningIsReported(%d)\n \n", warning);
- _warning = warning;
- //TEST(_audioDevice->StopRecording() == 0);
- //TEST(_audioDevice->StopPlayout() == 0);
-}
-;
-
-AudioTransportImpl::AudioTransportImpl(AudioDeviceModule* audioDevice) :
- _audioDevice(audioDevice),
- _playFromFile(false),
- _fullDuplex(false),
- _speakerVolume(false),
- _speakerMute(false),
- _microphoneVolume(false),
- _microphoneMute(false),
- _microphoneBoost(false),
- _microphoneAGC(false),
- _loopBackMeasurements(false),
- _playFile(*FileWrapper::Create()),
- _recCount(0),
- _playCount(0),
- _audioList()
-{
- _resampler.Reset(48000, 48000, kResamplerSynchronousStereo);
-}
-;
-
-AudioTransportImpl::~AudioTransportImpl()
-{
- _playFile.Flush();
- _playFile.CloseFile();
- delete &_playFile;
-
- while (!_audioList.Empty())
- {
- ListItem* item = _audioList.First();
- if (item)
- {
- AudioPacket* packet = static_cast<AudioPacket*> (item->GetItem());
- if (packet)
- {
- delete packet;
- }
- }
- _audioList.PopFront();
- }
-}
-;
-
-// ----------------------------------------------------------------------------
-// AudioTransportImpl::SetFilePlayout
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32 AudioTransportImpl::SetFilePlayout(bool enable,
- const WebRtc_Word8* fileName)
-{
- _playFromFile = enable;
- if (enable)
- {
- return (_playFile.OpenFile(fileName, true, true, false));
- } else
- {
- _playFile.Flush();
- return (_playFile.CloseFile());
- }
-}
-;
-
-void AudioTransportImpl::SetFullDuplex(bool enable)
-{
- _fullDuplex = enable;
-
- while (!_audioList.Empty())
- {
- ListItem* item = _audioList.First();
- if (item)
- {
- AudioPacket* packet = static_cast<AudioPacket*> (item->GetItem());
- if (packet)
- {
- delete packet;
- }
- }
- _audioList.PopFront();
- }
-}
-;
-
-WebRtc_Word32 AudioTransportImpl::RecordedDataIsAvailable(
- const WebRtc_Word8* audioSamples,
- const WebRtc_UWord32 nSamples,
- const WebRtc_UWord8 nBytesPerSample,
- const WebRtc_UWord8 nChannels,
- const WebRtc_UWord32 samplesPerSec,
- const WebRtc_UWord32 totalDelayMS,
- const WebRtc_Word32 clockDrift,
- const WebRtc_UWord32 currentMicLevel,
- WebRtc_UWord32& newMicLevel)
-{
- if (_fullDuplex && _audioList.GetSize() < 15)
- {
- AudioPacket* packet = new AudioPacket();
- memcpy(packet->dataBuffer, audioSamples, nSamples * nBytesPerSample);
- packet->nSamples = (WebRtc_UWord16) nSamples;
- packet->nBytesPerSample = nBytesPerSample;
- packet->nChannels = nChannels;
- packet->samplesPerSec = samplesPerSec;
- _audioList.PushBack(packet);
- }
-
- _recCount++;
- if (_recCount % 100 == 0)
- {
- bool addMarker(true);
-
- if (_loopBackMeasurements)
- {
- addMarker = false;
- }
-
- if (_microphoneVolume)
- {
- WebRtc_UWord32 maxVolume(0);
- WebRtc_UWord32 minVolume(0);
- WebRtc_UWord32 volume(0);
- WebRtc_UWord16 stepSize(0);
- TEST(_audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
- TEST(_audioDevice->MinMicrophoneVolume(&minVolume) == 0);
- TEST(_audioDevice->MicrophoneVolumeStepSize(&stepSize) == 0);
- TEST(_audioDevice->MicrophoneVolume(&volume) == 0);
- if (volume == 0)
- {
- TEST_LOG("[0]");
- addMarker = false;
- }
- int stepScale = (int) ((maxVolume - minVolume) / (stepSize * 10));
- volume += (stepScale * stepSize);
- if (volume > maxVolume)
- {
- TEST_LOG("[MAX]");
- volume = 0;
- addMarker = false;
- }
- TEST(_audioDevice->SetMicrophoneVolume(volume) == 0);
- }
-
- if (_microphoneAGC)
- {
- WebRtc_UWord32 maxVolume(0);
- WebRtc_UWord32 minVolume(0);
- WebRtc_UWord16 stepSize(0);
- TEST(_audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
- TEST(_audioDevice->MinMicrophoneVolume(&minVolume) == 0);
- TEST(_audioDevice->MicrophoneVolumeStepSize(&stepSize) == 0);
- // emulate real AGC (min->max->min->max etc.)
- if (currentMicLevel <= 1)
- {
- TEST_LOG("[MIN]");
- addMarker = false;
- }
- int stepScale = (int) ((maxVolume - minVolume) / (stepSize * 10));
- newMicLevel = currentMicLevel + (stepScale * stepSize);
- if (newMicLevel > maxVolume)
- {
- TEST_LOG("[MAX]");
- newMicLevel = 1; // set lowest (non-zero) AGC level
- addMarker = false;
- }
- }
-
- if (_microphoneMute && (_recCount % 500 == 0))
- {
- bool muted(false);
- TEST(_audioDevice->MicrophoneMute(&muted) == 0);
- muted = !muted;
- TEST(_audioDevice->SetMicrophoneMute(muted) == 0);
- if (muted)
- {
- TEST_LOG("[MUTE ON]");
- addMarker = false;
- } else
- {
- TEST_LOG("[MUTE OFF]");
- addMarker = false;
- }
- }
-
- if (_microphoneBoost && (_recCount % 500 == 0))
- {
- bool boosted(false);
- TEST(_audioDevice->MicrophoneBoost(&boosted) == 0);
- boosted = !boosted;
- TEST(_audioDevice->SetMicrophoneBoost(boosted) == 0);
- if (boosted)
- {
- TEST_LOG("[BOOST ON]");
- addMarker = false;
- } else
- {
- TEST_LOG("[BOOST OFF]");
- addMarker = false;
- }
- }
-
- if ((nChannels == 1) && addMarker)
- {
- // mono
- TEST_LOG("-");
- } else if ((nChannels == 2) && (nBytesPerSample == 2) && addMarker)
- {
- AudioDeviceModule::ChannelType
- chType(AudioDeviceModule::kChannelLeft);
- TEST(_audioDevice->RecordingChannel(&chType) == 0);
- if (chType == AudioDeviceModule::kChannelLeft)
- TEST_LOG("-|");
- else
- TEST_LOG("|-");
- } else if (addMarker)
- {
- // stereo
- TEST_LOG("--");
- }
-
- if (nChannels == 2 && nBytesPerSample == 2)
- {
- // TEST_LOG("=> emulated mono (one channel exctracted from stereo input)\n");
- }
- }
-
- return 0;
-}
-
-
-WebRtc_Word32 AudioTransportImpl::NeedMorePlayData(
- const WebRtc_UWord32 nSamples,
- const WebRtc_UWord8 nBytesPerSample,
- const WebRtc_UWord8 nChannels,
- const WebRtc_UWord32 samplesPerSec,
- WebRtc_Word8* audioSamples,
- WebRtc_UWord32& nSamplesOut)
-{
- if (_fullDuplex)
- {
- if (_audioList.Empty())
- {
- // use zero stuffing when not enough data
- memset(audioSamples, 0, nBytesPerSample * nSamples);
- } else
- {
- ListItem* item = _audioList.First();
- AudioPacket* packet = static_cast<AudioPacket*> (item->GetItem());
- if (packet)
- {
- int ret(0);
- int lenOut(0);
- WebRtc_Word16 tmpBuf_96kHz[80 * 12];
- WebRtc_Word16* ptr16In = NULL;
- WebRtc_Word16* ptr16Out = NULL;
-
- const WebRtc_UWord16 nSamplesIn = packet->nSamples;
- const WebRtc_UWord8 nChannelsIn = packet->nChannels;
- const WebRtc_UWord32 samplesPerSecIn = packet->samplesPerSec;
- const WebRtc_UWord16 nBytesPerSampleIn =
- packet->nBytesPerSample;
-
- WebRtc_Word32 fsInHz(samplesPerSecIn);
- WebRtc_Word32 fsOutHz(samplesPerSec);
-
- if (fsInHz == 44100)
- fsInHz = 44000;
-
- if (fsOutHz == 44100)
- fsOutHz = 44000;
-
- if (nChannelsIn == 2 && nBytesPerSampleIn == 4)
- {
- // input is stereo => we will resample in stereo
- ret = _resampler.ResetIfNeeded(fsInHz, fsOutHz,
- kResamplerSynchronousStereo);
- if (ret == 0)
- {
- if (nChannels == 2)
- {
- _resampler.Push(
- (const WebRtc_Word16*) packet->dataBuffer,
- 2 * nSamplesIn,
- (WebRtc_Word16*) audioSamples, 2
- * nSamples, lenOut);
- } else
- {
- _resampler.Push(
- (const WebRtc_Word16*) packet->dataBuffer,
- 2 * nSamplesIn, tmpBuf_96kHz, 2
- * nSamples, lenOut);
-
- ptr16In = &tmpBuf_96kHz[0];
- ptr16Out = (WebRtc_Word16*) audioSamples;
-
- // do stereo -> mono
- for (unsigned int i = 0; i < nSamples; i++)
- {
- *ptr16Out = *ptr16In; // use left channel
- ptr16Out++;
- ptr16In++;
- ptr16In++;
- }
- }
- assert(2*nSamples == (WebRtc_UWord32)lenOut);
- } else
- {
- if (_playCount % 100 == 0)
- TEST_LOG(
- "ERROR: unable to resample from %d to %d\n",
- samplesPerSecIn, samplesPerSec);
- }
- } else
- {
- // input is mono (can be "reduced from stereo" as well) =>
- // we will resample in mono
- ret = _resampler.ResetIfNeeded(fsInHz, fsOutHz,
- kResamplerSynchronous);
- if (ret == 0)
- {
- if (nChannels == 1)
- {
- _resampler.Push(
- (const WebRtc_Word16*) packet->dataBuffer,
- nSamplesIn,
- (WebRtc_Word16*) audioSamples,
- nSamples, lenOut);
- } else
- {
- _resampler.Push(
- (const WebRtc_Word16*) packet->dataBuffer,
- nSamplesIn, tmpBuf_96kHz, nSamples,
- lenOut);
-
- ptr16In = &tmpBuf_96kHz[0];
- ptr16Out = (WebRtc_Word16*) audioSamples;
-
- // do mono -> stereo
- for (unsigned int i = 0; i < nSamples; i++)
- {
- *ptr16Out = *ptr16In; // left
- ptr16Out++;
- *ptr16Out = *ptr16In; // right (same as left sample)
- ptr16Out++;
- ptr16In++;
- }
- }
- assert(nSamples == (WebRtc_UWord32)lenOut);
- } else
- {
- if (_playCount % 100 == 0)
- TEST_LOG("ERROR: unable to resample from %d to %d\n",
- samplesPerSecIn, samplesPerSec);
- }
- }
- nSamplesOut = nSamples;
- delete packet;
- }
- _audioList.PopFront();
- }
- } // if (_fullDuplex)
-
- if (_playFromFile && _playFile.Open())
- {
- WebRtc_Word16 fileBuf[480];
-
- // read mono-file
- WebRtc_Word32 len = _playFile.Read((WebRtc_Word8*) fileBuf, 2
- * nSamples);
- if (len != 2 * (WebRtc_Word32) nSamples)
- {
- _playFile.Rewind();
- _playFile.Read((WebRtc_Word8*) fileBuf, 2 * nSamples);
- }
-
- // convert to stero if required
- if (nChannels == 1)
- {
- memcpy(audioSamples, fileBuf, 2 * nSamples);
- } else
- {
- // mono sample from file is duplicated and sent to left and right
- // channels
- WebRtc_Word16* audio16 = (WebRtc_Word16*) audioSamples;
- for (unsigned int i = 0; i < nSamples; i++)
- {
- (*audio16) = fileBuf[i]; // left
- audio16++;
- (*audio16) = fileBuf[i]; // right
- audio16++;
- }
- }
- } // if (_playFromFile && _playFile.Open())
-
- _playCount++;
-
- if (_playCount % 100 == 0)
- {
- bool addMarker(true);
-
- if (_speakerVolume)
- {
- WebRtc_UWord32 maxVolume(0);
- WebRtc_UWord32 minVolume(0);
- WebRtc_UWord32 volume(0);
- WebRtc_UWord16 stepSize(0);
- TEST(_audioDevice->MaxSpeakerVolume(&maxVolume) == 0);
- TEST(_audioDevice->MinSpeakerVolume(&minVolume) == 0);
- TEST(_audioDevice->SpeakerVolumeStepSize(&stepSize) == 0);
- TEST(_audioDevice->SpeakerVolume(&volume) == 0);
- if (volume == 0)
- {
- TEST_LOG("[0]");
- addMarker = false;
- }
- WebRtc_UWord32 step = (maxVolume - minVolume) / 10;
- step = (step < stepSize ? stepSize : step);
- volume += step;
- if (volume > maxVolume)
- {
- TEST_LOG("[MAX]");
- volume = 0;
- addMarker = false;
- }
- TEST(_audioDevice->SetSpeakerVolume(volume) == 0);
- }
-
- if (_speakerMute && (_playCount % 500 == 0))
- {
- bool muted(false);
- TEST(_audioDevice->SpeakerMute(&muted) == 0);
- muted = !muted;
- TEST(_audioDevice->SetSpeakerMute(muted) == 0);
- if (muted)
- {
- TEST_LOG("[MUTE ON]");
- addMarker = false;
- } else
- {
- TEST_LOG("[MUTE OFF]");
- addMarker = false;
- }
- }
-
- if (_loopBackMeasurements)
- {
- WebRtc_UWord16 recDelayMS(0);
- WebRtc_UWord16 playDelayMS(0);
- WebRtc_UWord32 nItemsInList(0);
-
- nItemsInList = _audioList.GetSize();
- TEST(_audioDevice->RecordingDelay(&recDelayMS) == 0);
- TEST(_audioDevice->PlayoutDelay(&playDelayMS) == 0);
- TEST_LOG("Delay (rec+play)+buf: %3u (%3u+%3u)+%3u [ms]\n",
- recDelayMS + playDelayMS + 10 * (nItemsInList + 1),
- recDelayMS, playDelayMS, 10 * (nItemsInList + 1));
-
- addMarker = false;
- }
-
- if ((nChannels == 1) && addMarker)
- {
- TEST_LOG("+");
- } else if ((nChannels == 2) && addMarker)
- {
- TEST_LOG("++");
- }
- } // if (_playCount % 100 == 0)
-
- nSamplesOut = nSamples;
-
- return 0;
-}
-;
-
-FuncTestManager::FuncTestManager() :
- _resourcePath(webrtc::test::ProjectRootPath() +
- "test/data/audio_device/"),
- _processThread(NULL),
- _audioDevice(NULL),
- _audioEventObserver(NULL),
- _audioTransport(NULL)
-{
- assert(!_resourcePath.empty());
- _playoutFile48 = _resourcePath + "audio_short48.pcm";
- _playoutFile44 = _resourcePath + "audio_short44.pcm";
- _playoutFile16 = _resourcePath + "audio_short16.pcm";
- _playoutFile8 = _resourcePath + "audio_short8.pcm";
-}
-
-FuncTestManager::~FuncTestManager()
-{
-}
-
-WebRtc_Word32 FuncTestManager::Init()
-{
- TEST((_processThread = ProcessThread::CreateProcessThread()) != NULL);
- if (_processThread == NULL)
- {
- return -1;
- }
- _processThread->Start();
-
- // create the Audio Device module
- TEST((_audioDevice = AudioDeviceModuleImpl::Create(
- 555, ADM_AUDIO_LAYER)) != NULL);
- if (_audioDevice == NULL)
- {
- return -1;
- }
- TEST(_audioDevice->AddRef() == 1);
-
- // register the Audio Device module
- _processThread->RegisterModule(_audioDevice);
-
- // register event observer
- _audioEventObserver = new AudioEventObserver(_audioDevice);
- TEST(_audioDevice->RegisterEventObserver(_audioEventObserver) == 0);
-
- // register audio transport
- _audioTransport = new AudioTransportImpl(_audioDevice);
- TEST(_audioDevice->RegisterAudioCallback(_audioTransport) == 0);
-
- return 0;
-}
-
-WebRtc_Word32 FuncTestManager::Close()
-{
- TEST(_audioDevice->RegisterEventObserver(NULL) == 0);
- TEST(_audioDevice->RegisterAudioCallback(NULL) == 0);
- TEST(_audioDevice->Terminate() == 0);
-
- // release the ProcessThread object
- if (_processThread)
- {
- _processThread->DeRegisterModule(_audioDevice);
- _processThread->Stop();
- ProcessThread::DestroyProcessThread(_processThread);
- }
-
- // delete the audio observer
- if (_audioEventObserver)
- {
- delete _audioEventObserver;
- _audioEventObserver = NULL;
- }
-
- // delete the audio transport
- if (_audioTransport)
- {
- delete _audioTransport;
- _audioTransport = NULL;
- }
-
- // release the AudioDeviceModule object
- if (_audioDevice)
- {
- TEST(_audioDevice->Release() == 0);
- _audioDevice = NULL;
- }
-
- // return the ThreadWrapper (singleton)
- Trace::ReturnTrace();
-
- // PRINT_TEST_RESULTS;
-
- return 0;
-}
-
-WebRtc_Word32 FuncTestManager::DoTest(const TestType testType)
-{
- switch (testType)
- {
- case TTAll:
- TestAudioLayerSelection();
- TestDeviceEnumeration();
- TestDeviceSelection();
- TestAudioTransport();
- TestSpeakerVolume();
- TestMicrophoneVolume();
- TestLoopback();
- case TTAudioLayerSelection:
- TestAudioLayerSelection();
- break;
- case TTDeviceEnumeration:
- TestDeviceEnumeration();
- break;
- case TTDeviceSelection:
- TestDeviceSelection();
- break;
- case TTAudioTransport:
- TestAudioTransport();
- break;
- case TTSpeakerVolume:
- TestSpeakerVolume();
- break;
- case TTMicrophoneVolume:
- TestMicrophoneVolume();
- break;
- case TTSpeakerMute:
- TestSpeakerMute();
- break;
- case TTMicrophoneMute:
- TestMicrophoneMute();
- break;
- case TTMicrophoneBoost:
- TestMicrophoneBoost();
- break;
- case TTMicrophoneAGC:
- TestMicrophoneAGC();
- break;
- case TTLoopback:
- TestLoopback();
- break;
- case TTDeviceRemoval:
- TestDeviceRemoval();
- break;
- case TTMobileAPI:
- TestAdvancedMBAPI();
- case TTTest:
- TestExtra();
- break;
- default:
- break;
- }
-
- return 0;
-}
-;
-
-WebRtc_Word32 FuncTestManager::TestAudioLayerSelection()
-{
- TEST_LOG("\n=======================================\n");
- TEST_LOG(" Audio Layer test:\n");
- TEST_LOG("=======================================\n");
-
- if (_audioDevice == NULL)
- {
- return -1;
- }
-
- RESET_TEST;
-
- AudioDeviceModule* audioDevice = _audioDevice;
-
- AudioDeviceModule::AudioLayer audioLayer;
- TEST(audioDevice->ActiveAudioLayer(&audioLayer) == 0);
-
- if (audioLayer == AudioDeviceModule::kWindowsWaveAudio)
- {
- TEST_LOG("\nActiveAudioLayer: kWindowsWaveAudio\n \n");
- } else if (audioLayer == AudioDeviceModule::kWindowsCoreAudio)
- {
- TEST_LOG("\nActiveAudioLayer: kWindowsCoreAudio\n \n");
- } else if (audioLayer == AudioDeviceModule::kLinuxAlsaAudio)
- {
- TEST_LOG("\nActiveAudioLayer: kLinuxAlsaAudio\n \n");
- } else if (audioLayer == AudioDeviceModule::kLinuxPulseAudio)
- {
- TEST_LOG("\nActiveAudioLayer: kLinuxPulseAudio\n \n");
- } else
- {
- TEST_LOG("\nActiveAudioLayer: INVALID\n \n");
- }
-
- char ch;
- bool tryWinWave(false);
- bool tryWinCore(false);
-
- if (audioLayer == AudioDeviceModule::kWindowsWaveAudio)
- {
- TEST_LOG("Would you like to try kWindowsCoreAudio instead "
- "[requires Win Vista or Win 7] (Y/N)?\n: ");
- TEST(scanf(" %c", &ch) > 0);
- ch = toupper(ch);
- if (ch == 'Y')
- {
- tryWinCore = true;
- }
- } else if (audioLayer == AudioDeviceModule::kWindowsCoreAudio)
- {
- TEST_LOG("Would you like to try kWindowsWaveAudio instead (Y/N)?\n: ");
- TEST(scanf(" %c", &ch) > 0);
- ch = toupper(ch);
- if (ch == 'Y')
- {
- tryWinWave = true;
- }
- }
-
- if (tryWinWave || tryWinCore)
- {
- // =======================================
- // First, close down what we have started
-
- // terminate
- TEST(_audioDevice->RegisterEventObserver(NULL) == 0);
- TEST(_audioDevice->RegisterAudioCallback(NULL) == 0);
- TEST(_audioDevice->Terminate() == 0);
-
- // release the ProcessThread object
- if (_processThread)
- {
- _processThread->DeRegisterModule(_audioDevice);
- _processThread->Stop();
- ProcessThread::DestroyProcessThread(_processThread);
- }
-
- // delete the audio observer
- if (_audioEventObserver)
- {
- delete _audioEventObserver;
- _audioEventObserver = NULL;
- }
-
- // delete the audio transport
- if (_audioTransport)
- {
- delete _audioTransport;
- _audioTransport = NULL;
- }
-
- // release the AudioDeviceModule object
- if (_audioDevice)
- {
- TEST(_audioDevice->Release() == 0);
- _audioDevice = NULL;
- }
-
- // ==================================================
- // Next, try to make fresh start with new audio layer
-
- TEST((_processThread = ProcessThread::CreateProcessThread()) != NULL);
- if (_processThread == NULL)
- {
- return -1;
- }
- _processThread->Start();
-
- // create the Audio Device module based on selected audio layer
- if (tryWinWave)
- {
- _audioDevice = AudioDeviceModuleImpl::Create(
- 555,
- AudioDeviceModule::kWindowsWaveAudio);
- } else if (tryWinCore)
- {
- _audioDevice = AudioDeviceModuleImpl::Create(
- 555,
- AudioDeviceModule::kWindowsCoreAudio);
- }
-
- if (_audioDevice == NULL)
- {
- TEST_LOG("\nERROR: Switch of audio layer failed!\n");
- // restore default audio layer instead
- TEST((_audioDevice = AudioDeviceModuleImpl::Create(
- 555, AudioDeviceModule::kPlatformDefaultAudio)) != NULL);
- }
-
- if (_audioDevice == NULL)
- {
- TEST_LOG("\nERROR: Failed to revert back to default audio layer!\n");
- return -1;
- }
-
- TEST(_audioDevice->AddRef() == 1);
-
- // register the Audio Device module
- _processThread->RegisterModule(_audioDevice);
-
- // register event observer
- _audioEventObserver = new AudioEventObserver(_audioDevice);
- TEST(_audioDevice->RegisterEventObserver(_audioEventObserver) == 0);
-
- // register audio transport
- _audioTransport = new AudioTransportImpl(_audioDevice);
- TEST(_audioDevice->RegisterAudioCallback(_audioTransport) == 0);
-
- TEST(_audioDevice->ActiveAudioLayer(&audioLayer) == 0);
-
- if (audioLayer == AudioDeviceModule::kWindowsWaveAudio)
- {
- if (tryWinCore)
- TEST_LOG("\nActiveAudioLayer: kWindowsWaveAudio <=> "
- "switch was *not* possible\n \n");
- else
- TEST_LOG("\nActiveAudioLayer: kWindowsWaveAudio <=> "
- "switch was possible\n \n");
- } else if (audioLayer == AudioDeviceModule::kWindowsCoreAudio)
- {
- if (tryWinWave)
- TEST_LOG("\nActiveAudioLayer: kWindowsCoreAudio <=> "
- "switch was *not* possible\n \n");
- else
- TEST_LOG("\nActiveAudioLayer: kWindowsCoreAudio <=> "
- "switch was possible\n \n");
- }
- } // if (tryWinWave || tryWinCore)
-
- PRINT_TEST_RESULTS;
-
- return 0;
-}
-
-WebRtc_Word32 FuncTestManager::TestDeviceEnumeration()
-{
- TEST_LOG("\n=======================================\n");
- TEST_LOG(" Device Enumeration test:\n");
- TEST_LOG("=======================================\n");
-
- if (_audioDevice == NULL)
- {
- return -1;
- }
-
- RESET_TEST;
-
- AudioDeviceModule* audioDevice = _audioDevice;
-
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
-
- WebRtc_Word8 name[kAdmMaxDeviceNameSize];
- WebRtc_Word8 guid[kAdmMaxGuidSize];
-
- const WebRtc_Word16 nPlayoutDevices(audioDevice->PlayoutDevices());
- TEST(nPlayoutDevices >= 0);
- TEST_LOG("\nPlayoutDevices: %u\n \n", nPlayoutDevices);
- for (int n = 0; n < nPlayoutDevices; n++)
- {
- TEST(audioDevice->PlayoutDeviceName(n, name, guid) == 0);
- TEST_LOG(
- "PlayoutDeviceName(%d) : name=%s \n \
- guid=%s\n",
- n, name, guid);
- }
-
-#ifdef _WIN32
- // default (-1)
- TEST(audioDevice->PlayoutDeviceName(-1, name, guid) == 0);
- TEST_LOG("PlayoutDeviceName(%d): default name=%s \n \
- default guid=%s\n", -1, name, guid);
-#else
- // should fail
- TEST(audioDevice->PlayoutDeviceName(-1, name, guid) == -1);
-#endif
-
- const WebRtc_Word16 nRecordingDevices(audioDevice->RecordingDevices());
- TEST(nRecordingDevices >= 0);
- TEST_LOG("\nRecordingDevices: %u\n \n", nRecordingDevices);
- for (int n = 0; n < nRecordingDevices; n++)
- {
- TEST(audioDevice->RecordingDeviceName(n, name, guid) == 0);
- TEST_LOG(
- "RecordingDeviceName(%d) : name=%s \n \
- guid=%s\n",
- n, name, guid);
- }
-
-#ifdef _WIN32
- // default (-1)
- TEST(audioDevice->RecordingDeviceName(-1, name, guid) == 0);
- TEST_LOG("RecordingDeviceName(%d): default name=%s \n \
- default guid=%s\n", -1, name, guid);
-#else
- // should fail
- TEST(audioDevice->PlayoutDeviceName(-1, name, guid) == -1);
-#endif
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Initialized() == false);
-
- PRINT_TEST_RESULTS;
-
- return 0;
-}
-
-WebRtc_Word32 FuncTestManager::TestDeviceSelection()
-{
- TEST_LOG("\n=======================================\n");
- TEST_LOG(" Device Selection test:\n");
- TEST_LOG("=======================================\n");
-
- if (_audioDevice == NULL)
- {
- return -1;
- }
-
- RESET_TEST;
-
-#define PRINT_HEADING(a, b) \
- { \
- TEST_LOG("Set" #a "Device(" #b ") => \n"); \
- } \
-
-#define PRINT_HEADING_IDX(a, b,c ) \
- { \
- TEST_LOG("Set" #a "Device(%d) (%s) => \n", b, c); \
- } \
-
-#define PRINT_STR(a, b) \
- { \
- char str[128]; \
- (b == true) ? (sprintf(str, " %-17s: available\n", #a)) : (sprintf(str, " %-17s: NA\n", #a)); \
- TEST_LOG("%s", str); \
- } \
-
- AudioDeviceModule* audioDevice = _audioDevice;
-
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
-
- bool available(false);
- WebRtc_Word16 nDevices(-1);
- WebRtc_Word8 name[kAdmMaxDeviceNameSize];
- WebRtc_Word8 guid[kAdmMaxGuidSize];
-
- // =======
- // Playout
-
- nDevices = audioDevice->PlayoutDevices();
- TEST(nDevices >= 0);
-
- TEST_LOG("\n");
-#ifdef _WIN32
- TEST(audioDevice->SetPlayoutDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- PRINT_HEADING(Playout, kDefaultCommunicationDevice);
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- PRINT_STR(Playout, available);
- if (available)
- {
- TEST(audioDevice->StereoPlayoutIsAvailable(&available) == 0);
- PRINT_STR(Stereo Playout, available);
- }
- else
- {
- PRINT_STR(Stereo Playout, false);
- }
- TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
- PRINT_STR(Speaker, available);
- TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
- PRINT_STR(Speaker Volume, available);
- TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
- PRINT_STR(Speaker Mute, available);
-
- TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice) == 0);
- PRINT_HEADING(Playout, kDefaultDevice);
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- PRINT_STR(Playout, available);
- if (available)
- {
- TEST(audioDevice->StereoPlayoutIsAvailable(&available) == 0);
- PRINT_STR(Stereo Playout, available);
- }
- else
- {
- PRINT_STR(Stereo Playout, false);
- }
- TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
- PRINT_STR(Speaker, available);
- TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
- PRINT_STR(Speaker Volume, available);
- TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
- PRINT_STR(Speaker Mute, available);
-#else
- TEST(audioDevice->SetPlayoutDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == -1);
- TEST(audioDevice->SetPlayoutDevice(AudioDeviceModule::kDefaultDevice) == -1);
-#endif
-
- for (int i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetPlayoutDevice(i) == 0);
- TEST(audioDevice->PlayoutDeviceName(i, name, guid) == 0);
- PRINT_HEADING_IDX(Playout, i, name);
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- PRINT_STR(Playout, available);
- if (available)
- {
- TEST(audioDevice->StereoPlayoutIsAvailable(&available) == 0);
- PRINT_STR(Stereo Playout, available);
- } else
- {
- PRINT_STR(Stereo Playout, false);
- }
- TEST(audioDevice->SpeakerIsAvailable(&available) == 0);
- PRINT_STR(Speaker, available);
- TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
- PRINT_STR(Speaker Volume, available);
- TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
- PRINT_STR(Speaker Mute, available);
- }
-
- // =========
- // Recording
-
- nDevices = audioDevice->RecordingDevices();
- TEST(nDevices >= 0);
-
- TEST_LOG("\n");
-#ifdef _WIN32
- TEST(audioDevice->SetRecordingDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == 0);
- PRINT_HEADING(Recording, kDefaultCommunicationDevice);
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- PRINT_STR(Recording, available);
- if (available)
- {
- TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
- PRINT_STR(Stereo Recording, available);
- }
- else
- {
- // special fix to ensure that we don't log 'available' when recording is not OK
- PRINT_STR(Stereo Recording, false);
- }
- TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
- PRINT_STR(Microphone, available);
- TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
- PRINT_STR(Microphone Volume, available);
- TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
- PRINT_STR(Microphone Mute, available);
- TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
- PRINT_STR(Microphone Boost, available);
-
- TEST(audioDevice->SetRecordingDevice(AudioDeviceModule::kDefaultDevice) == 0);
- PRINT_HEADING(Recording, kDefaultDevice);
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- PRINT_STR(Recording, available);
- if (available)
- {
- TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
- PRINT_STR(Stereo Recording, available);
- }
- else
- {
- // special fix to ensure that we don't log 'available' when recording is not OK
- PRINT_STR(Stereo Recording, false);
- }
- TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
- PRINT_STR(Microphone, available);
- TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
- PRINT_STR(Microphone Volume, available);
- TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
- PRINT_STR(Microphone Mute, available);
- TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
- PRINT_STR(Microphone Boost, available);
-#else
- TEST(audioDevice->SetRecordingDevice(
- AudioDeviceModule::kDefaultCommunicationDevice) == -1);
- TEST(audioDevice->SetRecordingDevice(AudioDeviceModule::kDefaultDevice) == -1);
-#endif
-
- for (int i = 0; i < nDevices; i++)
- {
- TEST(audioDevice->SetRecordingDevice(i) == 0);
- TEST(audioDevice->RecordingDeviceName(i, name, guid) == 0);
- PRINT_HEADING_IDX(Recording, i, name);
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- PRINT_STR(Recording, available);
- if (available)
- {
- TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
- PRINT_STR(Stereo Recording, available);
- } else
- {
- // special fix to ensure that we don't log 'available' when recording
- // is not OK
- PRINT_STR(Stereo Recording, false);
- }
- TEST(audioDevice->MicrophoneIsAvailable(&available) == 0);
- PRINT_STR(Microphone, available);
- TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
- PRINT_STR(Microphone Volume, available);
- TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
- PRINT_STR(Microphone Mute, available);
- TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
- PRINT_STR(Microphone Boost, available);
- }
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Initialized() == false);
-
- PRINT_TEST_RESULTS;
-
- return 0;
-}
-
-WebRtc_Word32 FuncTestManager::TestAudioTransport()
-{
- TEST_LOG("\n=======================================\n");
- TEST_LOG(" Audio Transport test:\n");
- TEST_LOG("=======================================\n");
-
- if (_audioDevice == NULL)
- {
- return -1;
- }
-
- RESET_TEST;
-
- AudioDeviceModule* audioDevice = _audioDevice;
-
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
-
- bool recIsAvailable(false);
- bool playIsAvailable(false);
-
- if (SelectRecordingDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
-
- TEST(audioDevice->RecordingIsAvailable(&recIsAvailable) == 0);
- if (!recIsAvailable)
- {
- TEST_LOG(
- "\nWARNING: Recording is not available for the selected device!\n \n");
- }
-
- if (SelectPlayoutDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
-
- TEST(audioDevice->PlayoutIsAvailable(&playIsAvailable) == 0);
- if (recIsAvailable && playIsAvailable)
- {
- _audioTransport->SetFullDuplex(true);
- } else if (!playIsAvailable)
- {
- TEST_LOG(
- "\nWARNING: Playout is not available for the selected device!\n \n");
- }
-
- bool available(false);
- WebRtc_UWord32 samplesPerSec(0);
-
- if (playIsAvailable)
- {
- // =========================================
- // Start by playing out an existing PCM file
-
- TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
- if (available)
- {
- WebRtc_UWord32 maxVolume(0);
- TEST(audioDevice->MaxSpeakerVolume(&maxVolume) == 0);
- TEST(audioDevice->SetSpeakerVolume(maxVolume/2) == 0);
- }
-
- TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
-
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->PlayoutSampleRate(&samplesPerSec) == 0);
- if (samplesPerSec == 48000) {
- _audioTransport->SetFilePlayout(
- true, GetResource(_playoutFile48.c_str()));
- } else if (samplesPerSec == 44100 || samplesPerSec == 44000) {
- _audioTransport->SetFilePlayout(
- true, GetResource(_playoutFile44.c_str()));
- } else if (samplesPerSec == 16000) {
- _audioTransport->SetFilePlayout(
- true, GetResource(_playoutFile16.c_str()));
- } else if (samplesPerSec == 8000) {
- _audioTransport->SetFilePlayout(
- true, GetResource(_playoutFile8.c_str()));
- } else {
- TEST_LOG("\nERROR: Sample rate (%u) is not supported!\n \n",
- samplesPerSec);
- return -1;
- }
- TEST(audioDevice->StartPlayout() == 0);
-
- if (audioDevice->Playing())
- {
- TEST_LOG("\n> Listen to the file being played (fs=%d) out "
- "and verify that the audio quality is OK.\n"
- "> Press any key to stop playing...\n \n",
- samplesPerSec);
- PAUSE(DEFAULT_PAUSE_TIME);
- }
-
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
-
- _audioTransport->SetFilePlayout(false);
- }
-
- bool enabled(false);
- if (recIsAvailable)
- {
- // ====================================
- // Next, record from microphone to file
-
- TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
- if (available)
- {
- WebRtc_UWord32 maxVolume(0);
- TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
- TEST(audioDevice->SetMicrophoneVolume(maxVolume) == 0);
- }
-
- TEST(audioDevice->StartRawInputFileRecording(
- GetFilename(RecordedMicrophoneFile)) == 0);
- TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
-
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->StereoRecording(&enabled) == 0);
- if (enabled)
- {
- // ensure file recording in mono
- TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelLeft) == 0);
- }
- TEST(audioDevice->StartRecording() == 0);
- AudioDeviceUtility::Sleep(100);
-
- TEST(audioDevice->Recording() == true);
- if (audioDevice->Recording())
- {
- TEST_LOG("\n \n> The microphone input signal is now being recorded "
- "to a PCM file.\n"
- "> Speak into the microphone to ensure that your voice is"
- " recorded.\n> Press any key to stop recording...\n \n");
- PAUSE(DEFAULT_PAUSE_TIME);
- }
-
- TEST(audioDevice->StereoRecording(&enabled) == 0);
- if (enabled)
- {
- TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelBoth) == 0);
- }
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
- TEST(audioDevice->StopRawInputFileRecording() == 0);
- }
-
- if (recIsAvailable && playIsAvailable)
- {
- // ==========================
- // Play out the recorded file
-
- _audioTransport->SetFilePlayout(true,
- GetFilename(RecordedMicrophoneFile));
-
- TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->StartPlayout() == 0);
- AudioDeviceUtility::Sleep(100);
- }
-
- TEST(audioDevice->Playing() == true);
- if (audioDevice->Playing())
- {
- TEST_LOG("\n \n> Listen to the recorded file and verify that the "
- "audio quality is OK.\n"
- "> Press any key to stop listening...\n \n");
- PAUSE(DEFAULT_PAUSE_TIME);
- }
-
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
-
- _audioTransport->SetFilePlayout(false);
- }
-
- if (recIsAvailable && playIsAvailable)
- {
- // ==============================
- // Finally, make full duplex test
-
- WebRtc_UWord32 playSamplesPerSec(0);
- WebRtc_UWord32 recSamplesPerSecRec(0);
-
- TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
-
- _audioTransport->SetFullDuplex(true);
-
- TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
- if (available)
- {
- WebRtc_UWord32 maxVolume(0);
- TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
- TEST(audioDevice->SetMicrophoneVolume(maxVolume) == 0);
- }
-
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->PlayoutSampleRate(&playSamplesPerSec) == 0);
- TEST(audioDevice->RecordingSampleRate(&recSamplesPerSecRec) == 0);
- if (playSamplesPerSec != recSamplesPerSecRec)
- {
- TEST_LOG("\nERROR: sample rates does not match (fs_play=%u, fs_rec=%u)",
- playSamplesPerSec, recSamplesPerSecRec);
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
- _audioTransport->SetFullDuplex(false);
- return -1;
- }
-
- TEST(audioDevice->StartRecording() == 0);
- TEST(audioDevice->StartPlayout() == 0);
- AudioDeviceUtility::Sleep(100);
-
- if (audioDevice->Playing() && audioDevice->Recording())
- {
- TEST_LOG("\n \n> Full duplex audio (fs=%u) is now active.\n"
- "> Speak into the microphone and verify that your voice is "
- "played out in loopback.\n> Press any key to stop...\n \n",
- playSamplesPerSec);
- PAUSE(DEFAULT_PAUSE_TIME);
- }
-
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
-
- _audioTransport->SetFullDuplex(false);
- }
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Initialized() == false);
-
- TEST_LOG("\n");
- PRINT_TEST_RESULTS;
-
- return 0;
-}
-
-WebRtc_Word32 FuncTestManager::TestSpeakerVolume()
-{
- TEST_LOG("\n=======================================\n");
- TEST_LOG(" Speaker Volume test:\n");
- TEST_LOG("=======================================\n");
-
- if (_audioDevice == NULL)
- {
- return -1;
- }
-
- RESET_TEST;
-
- AudioDeviceModule* audioDevice = _audioDevice;
-
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
-
- if (SelectPlayoutDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
-
- bool available(false);
- WebRtc_UWord32 startVolume(0);
- WebRtc_UWord32 samplesPerSec(0);
-
- TEST(audioDevice->SpeakerVolumeIsAvailable(&available) == 0);
- if (available)
- {
- _audioTransport->SetSpeakerVolume(true);
- } else
- {
- TEST_LOG("\nERROR: Volume control is not available for the selected "
- "device!\n \n");
- return -1;
- }
-
- // store initial volume setting
- TEST(audioDevice->InitSpeaker() == 0);
- TEST(audioDevice->SpeakerVolume(&startVolume) == 0);
-
- // start at volume 0
- TEST(audioDevice->SetSpeakerVolume(0) == 0);
-
- // ======================================
- // Start playing out an existing PCM file
-
- TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->PlayoutSampleRate(&samplesPerSec) == 0);
- if (48000 == samplesPerSec) {
- _audioTransport->SetFilePlayout(
- true, GetResource(_playoutFile48.c_str()));
- } else if (44100 == samplesPerSec || samplesPerSec == 44000) {
- _audioTransport->SetFilePlayout(
- true, GetResource(_playoutFile44.c_str()));
- } else if (samplesPerSec == 16000) {
- _audioTransport->SetFilePlayout(
- true, GetResource(_playoutFile16.c_str()));
- } else if (samplesPerSec == 8000) {
- _audioTransport->SetFilePlayout(
- true, GetResource(_playoutFile8.c_str()));
- } else {
- TEST_LOG("\nERROR: Sample rate (%d) is not supported!\n \n",
- samplesPerSec);
- return -1;
- }
- TEST(audioDevice->StartPlayout() == 0);
- }
-
- TEST(audioDevice->Playing() == true);
- if (audioDevice->Playing())
- {
- TEST_LOG("\n> Listen to the file being played out and verify that the "
- "selected speaker volume is varied between [~0] and [~MAX].\n"
- "> The file shall be played out with an increasing volume level "
- "correlated to the speaker volume.\n"
- "> Press any key to stop playing...\n \n");
- PAUSE(10000);
- }
-
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
-
- _audioTransport->SetSpeakerVolume(false);
- _audioTransport->SetFilePlayout(false);
-
- // restore volume setting
- TEST(audioDevice->SetSpeakerVolume(startVolume) == 0);
-
- TEST_LOG("\n");
- PRINT_TEST_RESULTS;
-
- return 0;
-}
-
-WebRtc_Word32 FuncTestManager::TestSpeakerMute()
-{
- TEST_LOG("\n=======================================\n");
- TEST_LOG(" Speaker Mute test:\n");
- TEST_LOG("=======================================\n");
-
- if (_audioDevice == NULL)
- {
- return -1;
- }
-
- RESET_TEST;
-
- AudioDeviceModule* audioDevice = _audioDevice;
-
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
-
- if (SelectPlayoutDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
-
- bool available(false);
- bool startMute(false);
- WebRtc_UWord32 samplesPerSec(0);
-
- TEST(audioDevice->SpeakerMuteIsAvailable(&available) == 0);
- if (available)
- {
- _audioTransport->SetSpeakerMute(true);
- } else
- {
- TEST_LOG(
- "\nERROR: Mute control is not available for the selected"
- " device!\n \n");
- return -1;
- }
-
- // store initial mute setting
- TEST(audioDevice->InitSpeaker() == 0);
- TEST(audioDevice->SpeakerMute(&startMute) == 0);
-
- // start with no mute
- TEST(audioDevice->SetSpeakerMute(false) == 0);
-
- // ======================================
- // Start playing out an existing PCM file
-
- TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->PlayoutSampleRate(&samplesPerSec) == 0);
- if (48000 == samplesPerSec)
- _audioTransport->SetFilePlayout(true, _playoutFile48.c_str());
- else if (44100 == samplesPerSec || 44000 == samplesPerSec)
- _audioTransport->SetFilePlayout(true, _playoutFile44.c_str());
- else
- {
- TEST_LOG("\nERROR: Sample rate (%d) is not supported!\n \n",
- samplesPerSec);
- return -1;
- }
- TEST(audioDevice->StartPlayout() == 0);
- }
-
- TEST(audioDevice->Playing() == true);
- if (audioDevice->Playing())
- {
- TEST_LOG("\n> Listen to the file being played out and verify that the"
- " selected speaker mute control is toggled between [MUTE ON] and"
- " [MUTE OFF].\n> You should only hear the file during the"
- " 'MUTE OFF' periods.\n"
- "> Press any key to stop playing...\n \n");
- PAUSE(DEFAULT_PAUSE_TIME);
- }
-
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
-
- _audioTransport->SetSpeakerMute(false);
- _audioTransport->SetFilePlayout(false);
-
- // restore mute setting
- TEST(audioDevice->SetSpeakerMute(startMute) == 0);
-
- TEST_LOG("\n");
- PRINT_TEST_RESULTS;
-
- return 0;
-}
-
-WebRtc_Word32 FuncTestManager::TestMicrophoneVolume()
-{
- TEST_LOG("\n=======================================\n");
- TEST_LOG(" Microphone Volume test:\n");
- TEST_LOG("=======================================\n");
-
- if (_audioDevice == NULL)
- {
- return -1;
- }
-
- RESET_TEST;
-
- AudioDeviceModule* audioDevice = _audioDevice;
-
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
-
- if (SelectRecordingDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
-
- bool available(false);
- TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
- if (available)
- {
- _audioTransport->SetMicrophoneVolume(true);
- } else
- {
- TEST_LOG("\nERROR: Volume control is not available for the selected "
- "device!\n \n");
- return -1;
- }
-
- if (SelectPlayoutDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
-
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- _audioTransport->SetFullDuplex(true);
- } else
- {
- TEST_LOG("\nERROR: Playout is not available for the selected "
- "device!\n \n");
- return -1;
- }
-
- TEST_LOG("\nEnable recording of microphone input to file (%s) during this"
- " test (Y/N)?\n: ",
- RecordedMicrophoneVolumeFile);
- char ch;
- bool fileRecording(false);
- TEST(scanf(" %c", &ch) > 0);
- ch = toupper(ch);
- if (ch == 'Y')
- {
- fileRecording = true;
- }
-
- WebRtc_UWord32 startVolume(0);
- bool enabled(false);
-
- // store initial volume setting
- TEST(audioDevice->InitMicrophone() == 0);
- TEST(audioDevice->MicrophoneVolume(&startVolume) == 0);
-
- // start at volume 0
- TEST(audioDevice->SetMicrophoneVolume(0) == 0);
-
- // ======================================================================
- // Start recording from the microphone while the mic volume is changed
- // continuously.
- // Also, start playing out the input to enable real-time verification.
-
- if (fileRecording)
- {
- TEST(audioDevice->StartRawInputFileRecording(RecordedMicrophoneVolumeFile) == 0);
- }
- TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->StereoRecording(&enabled) == 0);
- if (enabled)
- {
- // ensures a mono file
- TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelRight) == 0);
- }
- TEST(audioDevice->StartRecording() == 0);
- }
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->StartPlayout() == 0);
- }
-
- TEST(audioDevice->Recording() == true);
- TEST(audioDevice->Playing() == true);
- if (audioDevice->Recording() && audioDevice->Playing())
- {
- TEST_LOG("\n> Speak into the microphone and verify that the selected "
- "microphone volume is varied between [~0] and [~MAX].\n"
- "> You should hear your own voice with an increasing volume level"
- " correlated to the microphone volume.\n"
- "> After a finalized test (and if file recording was enabled) "
- "verify the recorded result off line.\n"
- "> Press any key to stop...\n \n");
- PAUSE(DEFAULT_PAUSE_TIME);
- }
-
- if (fileRecording)
- {
- TEST(audioDevice->StopRawInputFileRecording() == 0);
- }
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
- TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
-
- _audioTransport->SetMicrophoneVolume(false);
- _audioTransport->SetFullDuplex(false);
-
- // restore volume setting
- TEST(audioDevice->SetMicrophoneVolume(startVolume) == 0);
-
- TEST_LOG("\n");
- PRINT_TEST_RESULTS;
-
- return 0;
-}
-
-WebRtc_Word32 FuncTestManager::TestMicrophoneMute()
-{
- TEST_LOG("\n=======================================\n");
- TEST_LOG(" Microphone Mute test:\n");
- TEST_LOG("=======================================\n");
-
- if (_audioDevice == NULL)
- {
- return -1;
- }
-
- RESET_TEST;
-
- AudioDeviceModule* audioDevice = _audioDevice;
-
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
-
- if (SelectRecordingDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
-
- bool available(false);
- TEST(audioDevice->MicrophoneMuteIsAvailable(&available) == 0);
- if (available)
- {
- _audioTransport->SetMicrophoneMute(true);
- } else
- {
- TEST_LOG("\nERROR: Mute control is not available for the selected"
- " device!\n \n");
- return -1;
- }
-
- if (SelectPlayoutDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
-
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- _audioTransport->SetFullDuplex(true);
- } else
- {
- TEST_LOG("\nERROR: Playout is not available for the selected "
- "device!\n \n");
- return -1;
- }
-
- TEST_LOG("\nEnable recording of microphone input to file (%s) during this "
- "test (Y/N)?\n: ",
- RecordedMicrophoneMuteFile);
- char ch;
- bool fileRecording(false);
- TEST(scanf(" %c", &ch) > 0);
- ch = toupper(ch);
- if (ch == 'Y')
- {
- fileRecording = true;
- }
-
- bool startMute(false);
- bool enabled(false);
-
- // store initial volume setting
- TEST(audioDevice->InitMicrophone() == 0);
- TEST(audioDevice->MicrophoneMute(&startMute) == 0);
-
- // start at no mute
- TEST(audioDevice->SetMicrophoneMute(false) == 0);
-
- // ==================================================================
- // Start recording from the microphone while the mic mute is toggled
- // continuously.
- // Also, start playing out the input to enable real-time verification.
-
- if (fileRecording)
- {
- TEST(audioDevice->StartRawInputFileRecording(RecordedMicrophoneMuteFile) == 0);
- }
- TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->StereoRecording(&enabled) == 0);
- if (enabled)
- {
- // ensure file recording in mono
- TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelLeft) == 0);
- }
- TEST(audioDevice->StartRecording() == 0);
- }
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->StartPlayout() == 0);
- }
-
- TEST(audioDevice->Recording() == true);
- TEST(audioDevice->Playing() == true);
- if (audioDevice->Recording() && audioDevice->Playing())
- {
- TEST_LOG("\n> Speak into the microphone and verify that the selected "
- "microphone mute control is toggled between [MUTE ON] and [MUTE OFF]."
- "\n> You should only hear your own voice in loopback during the"
- " 'MUTE OFF' periods.\n> After a finalized test (and if file "
- "recording was enabled) verify the recorded result off line.\n"
- "> Press any key to stop...\n \n");
- PAUSE(DEFAULT_PAUSE_TIME);
- }
-
- if (fileRecording)
- {
- TEST(audioDevice->StopRawInputFileRecording() == 0);
- }
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
-
- _audioTransport->SetMicrophoneMute(false);
- _audioTransport->SetFullDuplex(false);
-
- // restore volume setting
- TEST(audioDevice->SetMicrophoneMute(startMute) == 0);
-
- TEST_LOG("\n");
- PRINT_TEST_RESULTS;
-
- return 0;
-}
-
-WebRtc_Word32 FuncTestManager::TestMicrophoneBoost()
-{
- TEST_LOG("\n=======================================\n");
- TEST_LOG(" Microphone Boost test:\n");
- TEST_LOG("=======================================\n");
-
- if (_audioDevice == NULL)
- {
- return -1;
- }
-
- RESET_TEST;
-
- AudioDeviceModule* audioDevice = _audioDevice;
-
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
-
- if (SelectRecordingDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
-
- bool available(false);
- TEST(audioDevice->MicrophoneBoostIsAvailable(&available) == 0);
- if (available)
- {
- _audioTransport->SetMicrophoneBoost(true);
- } else
- {
- TEST_LOG(
- "\nERROR: Boost control is not available for the selected device!\n \n");
- return -1;
- }
-
- if (SelectPlayoutDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
-
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- _audioTransport->SetFullDuplex(true);
- } else
- {
- TEST_LOG("\nERROR: Playout is not available for the selected device!\n \n");
- return -1;
- }
-
- TEST_LOG("\nEnable recording of microphone input to file (%s) during this "
- "test (Y/N)?\n: ",
- RecordedMicrophoneBoostFile);
- char ch;
- bool fileRecording(false);
- TEST(scanf(" %c", &ch) > 0);
- ch = toupper(ch);
- if (ch == 'Y')
- {
- fileRecording = true;
- }
-
- bool startBoost(false);
- bool enabled(false);
-
- // store initial volume setting
- TEST(audioDevice->InitMicrophone() == 0);
- TEST(audioDevice->MicrophoneBoost(&startBoost) == 0);
-
- // start at no boost
- TEST(audioDevice->SetMicrophoneBoost(false) == 0);
-
- // ==================================================================
- // Start recording from the microphone while the mic boost is toggled
- // continuously.
- // Also, start playing out the input to enable real-time verification.
-
- if (fileRecording)
- {
- TEST(audioDevice->StartRawInputFileRecording(RecordedMicrophoneBoostFile) == 0);
- }
- TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->StereoRecording(&enabled) == 0);
- if (enabled)
- {
- // ensure file recording in mono
- TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelLeft) == 0);
- }
- TEST(audioDevice->StartRecording() == 0);
- }
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->StartPlayout() == 0);
- }
-
- TEST(audioDevice->Recording() == true);
- TEST(audioDevice->Playing() == true);
- if (audioDevice->Recording() && audioDevice->Playing())
- {
- TEST_LOG("\n> Speak into the microphone and verify that the selected "
- "microphone boost control is toggled between [BOOST ON] and [BOOST OFF].\n"
- "> You should hear your own voice with an increased volume level "
- "during the 'BOOST ON' periods.\n \n"
- "> After a finalized test (and if file recording was enabled) verify"
- " the recorded result off line.\n"
- "> Press any key to stop...\n \n");
- PAUSE(DEFAULT_PAUSE_TIME);
- }
-
- if (fileRecording)
- {
- TEST(audioDevice->StopRawInputFileRecording() == 0);
- }
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
-
- _audioTransport->SetMicrophoneBoost(false);
- _audioTransport->SetFullDuplex(false);
-
- // restore boost setting
- TEST(audioDevice->SetMicrophoneBoost(startBoost) == 0);
-
- TEST_LOG("\n");
- PRINT_TEST_RESULTS;
-
- return 0;
-}
-
-WebRtc_Word32 FuncTestManager::TestMicrophoneAGC()
-{
- TEST_LOG("\n=======================================\n");
- TEST_LOG(" Microphone AGC test:\n");
- TEST_LOG("=======================================\n");
-
- if (_audioDevice == NULL)
- {
- return -1;
- }
-
- RESET_TEST;
-
- AudioDeviceModule* audioDevice = _audioDevice;
-
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
-
- if (SelectRecordingDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
-
- bool available(false);
- TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
- if (available)
- {
- _audioTransport->SetMicrophoneAGC(true);
- } else
- {
- TEST_LOG("\nERROR: It is not possible to control the microphone volume"
- " for the selected device!\n \n");
- return -1;
- }
-
- if (SelectPlayoutDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
-
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- _audioTransport->SetFullDuplex(true);
- } else
- {
- TEST_LOG("\nERROR: Playout is not available for the selected device!\n \n");
- return -1;
- }
-
- TEST_LOG("\nEnable recording of microphone input to file (%s) during "
- "this test (Y/N)?\n: ",
- RecordedMicrophoneAGCFile);
- char ch;
- bool fileRecording(false);
- TEST(scanf(" %c", &ch) > 0);
- ch = toupper(ch);
- if (ch == 'Y')
- {
- fileRecording = true;
- }
-
- WebRtc_UWord32 startVolume(0);
- bool enabled(false);
-
- // store initial volume setting
- TEST(audioDevice->InitMicrophone() == 0);
- TEST(audioDevice->MicrophoneVolume(&startVolume) == 0);
-
- // ====================================================================
- // Start recording from the microphone while the mic volume is changed
- // continuously
- // by the emulated AGC (implemented by our audio transport).
- // Also, start playing out the input to enable real-time verification.
-
- if (fileRecording)
- {
- TEST(audioDevice->StartRawInputFileRecording(RecordedMicrophoneAGCFile) == 0);
- }
- TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
- TEST(audioDevice->RecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->SetAGC(true) == 0);
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->StereoRecording(&enabled) == 0);
- if (enabled)
- {
- // ensures a mono file
- TEST(audioDevice->SetRecordingChannel(AudioDeviceModule::kChannelRight) == 0);
- }
- TEST(audioDevice->StartRecording() == 0);
- }
- TEST(audioDevice->PlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->StartPlayout() == 0);
- }
-
- TEST(audioDevice->AGC() == true);
- TEST(audioDevice->Recording() == true);
- TEST(audioDevice->Playing() == true);
- if (audioDevice->Recording() && audioDevice->Playing())
- {
- TEST_LOG("\n> Speak into the microphone and verify that the volume of"
- " the selected microphone is varied between [~0] and [~MAX].\n"
- "> You should hear your own voice with an increasing volume level"
- " correlated to an emulated AGC setting.\n"
- "> After a finalized test (and if file recording was enabled) verify"
- " the recorded result off line.\n"
- "> Press any key to stop...\n \n");
- PAUSE(DEFAULT_PAUSE_TIME);
- }
-
- if (fileRecording)
- {
- TEST(audioDevice->StopRawInputFileRecording() == 0);
- }
- TEST(audioDevice->SetAGC(false) == 0);
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
- TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
-
- _audioTransport->SetMicrophoneAGC(false);
- _audioTransport->SetFullDuplex(false);
-
- // restore volume setting
- TEST(audioDevice->SetMicrophoneVolume(startVolume) == 0);
-
- TEST_LOG("\n");
- PRINT_TEST_RESULTS;
-
- return 0;
-}
-
-WebRtc_Word32 FuncTestManager::TestLoopback()
-{
- TEST_LOG("\n=======================================\n");
- TEST_LOG(" Loopback measurement test:\n");
- TEST_LOG("=======================================\n");
-
- if (_audioDevice == NULL)
- {
- return -1;
- }
-
- RESET_TEST;
-
- AudioDeviceModule* audioDevice = _audioDevice;
-
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
-
- bool recIsAvailable(false);
- bool playIsAvailable(false);
- WebRtc_UWord8 nPlayChannels(0);
- WebRtc_UWord8 nRecChannels(0);
-
- if (SelectRecordingDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
-
- TEST(audioDevice->RecordingIsAvailable(&recIsAvailable) == 0);
- if (!recIsAvailable)
- {
- TEST_LOG("\nERROR: Recording is not available for the selected device!\n \n");
- return -1;
- }
-
- if (SelectPlayoutDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
-
- TEST(audioDevice->PlayoutIsAvailable(&playIsAvailable) == 0);
- if (recIsAvailable && playIsAvailable)
- {
- _audioTransport->SetFullDuplex(true);
- _audioTransport->SetLoopbackMeasurements(true);
- } else if (!playIsAvailable)
- {
- TEST_LOG("\nERROR: Playout is not available for the selected device!\n \n");
- return -1;
- }
-
- bool enabled(false);
- bool available(false);
-
- if (recIsAvailable && playIsAvailable)
- {
- WebRtc_UWord32 playSamplesPerSec(0);
- WebRtc_UWord32 recSamplesPerSecRec(0);
-
- TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
-
- _audioTransport->SetFullDuplex(true);
-
- TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->SetStereoRecording(true) == 0);
- }
-
- TEST(audioDevice->StereoPlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->SetStereoPlayout(true) == 0);
- }
-
- TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
- if (available)
- {
- WebRtc_UWord32 maxVolume(0);
- TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
- TEST(audioDevice->SetMicrophoneVolume(maxVolume) == 0);
- }
-
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->PlayoutSampleRate(&playSamplesPerSec) == 0);
- TEST(audioDevice->RecordingSampleRate(&recSamplesPerSecRec) == 0);
- TEST(audioDevice->StereoPlayout(&enabled) == 0);
- enabled ? nPlayChannels = 2 : nPlayChannels = 1;
- TEST(audioDevice->StereoRecording(&enabled) == 0);
- enabled ? nRecChannels = 2 : nRecChannels = 1;
- TEST(audioDevice->StartRecording() == 0);
- TEST(audioDevice->StartPlayout() == 0);
-
- if (audioDevice->Playing() && audioDevice->Recording())
- {
- TEST_LOG("\n \n> Loopback audio is now active.\n"
- "> Rec : fs=%u, #channels=%u.\n"
- "> Play: fs=%u, #channels=%u.\n"
- "> Speak into the microphone and verify that your voice is"
- " played out in loopback.\n"
- "> Press any key to stop...\n \n",
- recSamplesPerSecRec, nRecChannels, playSamplesPerSec,
- nPlayChannels);
- PAUSE(30000);
- }
-
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
-
- _audioTransport->SetFullDuplex(false);
- _audioTransport->SetLoopbackMeasurements(false);
- }
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Initialized() == false);
-
- TEST_LOG("\n");
- PRINT_TEST_RESULTS;
-
- return 0;
-}
-
-WebRtc_Word32 FuncTestManager::TestDeviceRemoval()
-{
- TEST_LOG("\n=======================================\n");
- TEST_LOG(" Device removal test:\n");
- TEST_LOG("=======================================\n");
-
- if (_audioDevice == NULL)
- {
- return -1;
- }
-
- RESET_TEST;
-
- AudioDeviceModule* audioDevice = _audioDevice;
-
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
-
- bool recIsAvailable(false);
- bool playIsAvailable(false);
- WebRtc_UWord8 nPlayChannels(0);
- WebRtc_UWord8 nRecChannels(0);
- WebRtc_UWord8 loopCount(0);
-
- while (loopCount < 2)
- {
- if (SelectRecordingDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
-
- TEST(audioDevice->RecordingIsAvailable(&recIsAvailable) == 0);
- if (!recIsAvailable)
- {
- TEST_LOG("\nERROR: Recording is not available for the selected device!\n \n");
- return -1;
- }
-
- if (SelectPlayoutDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
-
- TEST(audioDevice->PlayoutIsAvailable(&playIsAvailable) == 0);
- if (recIsAvailable && playIsAvailable)
- {
- _audioTransport->SetFullDuplex(true);
- } else if (!playIsAvailable)
- {
- TEST_LOG("\nERROR: Playout is not available for the selected device!\n \n");
- return -1;
- }
-
- bool available(false);
- bool enabled(false);
-
- if (recIsAvailable && playIsAvailable)
- {
- WebRtc_UWord32 playSamplesPerSec(0);
- WebRtc_UWord32 recSamplesPerSecRec(0);
-
- TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
-
- _audioTransport->SetFullDuplex(true);
-
- TEST(audioDevice->StereoRecordingIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->SetStereoRecording(true) == 0);
- }
-
- TEST(audioDevice->StereoPlayoutIsAvailable(&available) == 0);
- if (available)
- {
- TEST(audioDevice->SetStereoPlayout(true) == 0);
- }
-
- TEST(audioDevice->MicrophoneVolumeIsAvailable(&available) == 0);
- if (available)
- {
- WebRtc_UWord32 maxVolume(0);
- TEST(audioDevice->MaxMicrophoneVolume(&maxVolume) == 0);
- TEST(audioDevice->SetMicrophoneVolume(maxVolume) == 0);
- }
-
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->PlayoutSampleRate(&playSamplesPerSec) == 0);
- TEST(audioDevice->RecordingSampleRate(&recSamplesPerSecRec) == 0);
- TEST(audioDevice->StereoPlayout(&enabled) == 0);
- enabled ? nPlayChannels = 2 : nPlayChannels = 1;
- TEST(audioDevice->StereoRecording(&enabled) == 0);
- enabled ? nRecChannels = 2 : nRecChannels = 1;
- TEST(audioDevice->StartRecording() == 0);
- TEST(audioDevice->StartPlayout() == 0);
-
- AudioDeviceModule::AudioLayer audioLayer;
- TEST(audioDevice->ActiveAudioLayer(&audioLayer) == 0);
-
- if (audioLayer == AudioDeviceModule::kLinuxPulseAudio)
- {
- TEST_LOG("\n \n> PulseAudio loopback audio is now active.\n"
- "> Rec : fs=%u, #channels=%u.\n"
- "> Play: fs=%u, #channels=%u.\n"
- "> Speak into the microphone and verify that your voice is"
- " played out in loopback.\n"
- "> Unplug the device and make sure that your voice is played"
- " out in loop back on the built-in soundcard.\n"
- "> Then press any key...\n",
- recSamplesPerSecRec, nRecChannels, playSamplesPerSec,
- nPlayChannels);
-
- PAUSE(DEFAULT_PAUSE_TIME);
- } else if (audioDevice->Playing() && audioDevice->Recording())
- {
- if (loopCount < 1)
- {
- TEST_LOG("\n \n> Loopback audio is now active.\n"
- "> Rec : fs=%u, #channels=%u.\n"
- "> Play: fs=%u, #channels=%u.\n"
- "> Speak into the microphone and verify that your voice"
- " is played out in loopback.\n"
- "> Unplug the device and wait for the error message...\n",
- recSamplesPerSecRec, nRecChannels,
- playSamplesPerSec, nPlayChannels);
-
- _audioEventObserver->_error
- = (AudioDeviceObserver::ErrorCode) (-1);
- while (_audioEventObserver->_error
- == (AudioDeviceObserver::ErrorCode) (-1))
- {
- SLEEP(500);
- }
- } else
- {
- TEST_LOG("\n \n> Loopback audio is now active.\n"
- "> Rec : fs=%u, #channels=%u.\n"
- "> Play: fs=%u, #channels=%u.\n"
- "> Speak into the microphone and verify that your voice"
- " is played out in loopback.\n"
- "> Press any key to stop...\n",
- recSamplesPerSecRec, nRecChannels,
- playSamplesPerSec, nPlayChannels);
-
- PAUSE(DEFAULT_PAUSE_TIME);
- }
- }
-
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
-
- _audioTransport->SetFullDuplex(false);
-
- if (loopCount < 1)
- {
- TEST_LOG("\n \n> Stopped!\n");
- TEST_LOG("> Now reinsert device if you want to enumerate it.\n");
- TEST_LOG("> Press any key when done.\n");
- PAUSE(DEFAULT_PAUSE_TIME);
- }
-
- loopCount++;
- }
- } // loopCount
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Initialized() == false);
-
- TEST_LOG("\n");
- PRINT_TEST_RESULTS;
-
- return 0;
-}
-
-WebRtc_Word32 FuncTestManager::TestExtra()
-{
- TEST_LOG("\n=======================================\n");
- TEST_LOG(" Extra test:\n");
- TEST_LOG("=======================================\n");
-
- if (_audioDevice == NULL)
- {
- return -1;
- }
-
- RESET_TEST;
-
- AudioDeviceModule* audioDevice = _audioDevice;
-
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
-
- TEST(audioDevice->Terminate() == 0);
- TEST(audioDevice->Initialized() == false);
-
- TEST_LOG("\n");
- PRINT_TEST_RESULTS;
-
- return 0;
-}
-
-WebRtc_Word32 FuncTestManager::SelectRecordingDevice()
-{
- WebRtc_Word16 nDevices = _audioDevice->RecordingDevices();
- WebRtc_Word8 name[kAdmMaxDeviceNameSize];
- WebRtc_Word8 guid[kAdmMaxGuidSize];
- WebRtc_Word32 ret(-1);
-
-#ifdef _WIN32
- TEST_LOG("\nSelect Recording Device\n \n");
- TEST_LOG(" (%d) Default\n", 0);
- TEST_LOG(" (%d) Default Communication [Win 7]\n", 1);
- TEST_LOG("- - - - - - - - - - - - - - - - - - - -\n");
- for (int i = 0; i < nDevices; i++)
- {
- TEST(_audioDevice->RecordingDeviceName(i, name, guid) == 0);
- TEST_LOG(" (%d) Device %d (%s)\n", i+10, i, name);
- }
- TEST_LOG("\n: ");
-
- int sel(0);
-
- scanf("%u", &sel);
-
- if (sel == 0)
- {
- TEST((ret = _audioDevice->SetRecordingDevice(AudioDeviceModule::kDefaultDevice)) == 0);
- }
- else if (sel == 1)
- {
- TEST((ret = _audioDevice->SetRecordingDevice(
- AudioDeviceModule::kDefaultCommunicationDevice)) == 0);
- }
- else if (sel < (nDevices+10))
- {
- TEST((ret = _audioDevice->SetRecordingDevice(sel-10)) == 0);
- }
- else
- {
- return -1;
- }
-#else
- TEST_LOG("\nSelect Recording Device\n \n");
- for (int i = 0; i < nDevices; i++)
- {
- TEST(_audioDevice->RecordingDeviceName(i, name, guid) == 0);
- TEST_LOG(" (%d) Device %d (%s)\n", i, i, name);
- }
- TEST_LOG("\n: ");
- int sel(0);
- TEST(scanf("%u", &sel) > 0);
- if (sel < (nDevices))
- {
- TEST((ret = _audioDevice->SetRecordingDevice(sel)) == 0);
- } else
- {
- return -1;
- }
-#endif
-
- return ret;
-}
-
-WebRtc_Word32 FuncTestManager::SelectPlayoutDevice()
-{
- WebRtc_Word16 nDevices = _audioDevice->PlayoutDevices();
- WebRtc_Word8 name[kAdmMaxDeviceNameSize];
- WebRtc_Word8 guid[kAdmMaxGuidSize];
-
-#ifdef _WIN32
- TEST_LOG("\nSelect Playout Device\n \n");
- TEST_LOG(" (%d) Default\n", 0);
- TEST_LOG(" (%d) Default Communication [Win 7]\n", 1);
- TEST_LOG("- - - - - - - - - - - - - - - - - - - -\n");
- for (int i = 0; i < nDevices; i++)
- {
- TEST(_audioDevice->PlayoutDeviceName(i, name, guid) == 0);
- TEST_LOG(" (%d) Device %d (%s)\n", i+10, i, name);
- }
- TEST_LOG("\n: ");
-
- int sel(0);
-
- scanf("%u", &sel);
-
- WebRtc_Word32 ret(0);
-
- if (sel == 0)
- {
- TEST((ret = _audioDevice->SetPlayoutDevice(
- AudioDeviceModule::kDefaultDevice)) == 0);
- }
- else if (sel == 1)
- {
- TEST((ret = _audioDevice->SetPlayoutDevice(
- AudioDeviceModule::kDefaultCommunicationDevice)) == 0);
- }
- else if (sel < (nDevices+10))
- {
- TEST((ret = _audioDevice->SetPlayoutDevice(sel-10)) == 0);
- }
- else
- {
- return -1;
- }
-#else
- TEST_LOG("\nSelect Playout Device\n \n");
- for (int i = 0; i < nDevices; i++)
- {
- TEST(_audioDevice->PlayoutDeviceName(i, name, guid) == 0);
- TEST_LOG(" (%d) Device %d (%s)\n", i, i, name);
- }
- TEST_LOG("\n: ");
- int sel(0);
- TEST(scanf("%u", &sel) > 0);
- WebRtc_Word32 ret(0);
- if (sel < (nDevices))
- {
- TEST((ret = _audioDevice->SetPlayoutDevice(sel)) == 0);
- } else
- {
- return -1;
- }
-#endif
-
- return ret;
-}
-
-WebRtc_Word32 FuncTestManager::TestAdvancedMBAPI()
-{
- TEST_LOG("\n=======================================\n");
- TEST_LOG(" Advanced mobile device API test:\n");
- TEST_LOG("=======================================\n");
-
- if (_audioDevice == NULL)
- {
- return -1;
- }
-
- RESET_TEST;
-
- AudioDeviceModule* audioDevice = _audioDevice;
-
- TEST(audioDevice->Init() == 0);
- TEST(audioDevice->Initialized() == true);
-
- if (SelectRecordingDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
- if (SelectPlayoutDevice() == -1)
- {
- TEST_LOG("\nERROR: Device selection failed!\n \n");
- return -1;
- }
- _audioTransport->SetFullDuplex(true);
- _audioTransport->SetLoopbackMeasurements(true);
-
- TEST(audioDevice->RegisterAudioCallback(_audioTransport) == 0);
- // Start recording
- TEST(audioDevice->InitRecording() == 0);
- TEST(audioDevice->StartRecording() == 0);
- // Start playout
- TEST(audioDevice->InitPlayout() == 0);
- TEST(audioDevice->StartPlayout() == 0);
-
- TEST(audioDevice->Recording() == true);
- TEST(audioDevice->Playing() == true);
-
-#if defined(_WIN32_WCE) || defined(MAC_IPHONE)
- TEST_LOG("\nResetAudioDevice\n \n");
- if (audioDevice->Recording() && audioDevice->Playing())
- {
- TEST_LOG("\n> Speak into the microphone and verify that the audio is good.\n\
-> Press any key to stop...\n \n");
- PAUSE(DEFAULT_PAUSE_TIME);
- }
- for (int p=0; p<=60; p+=20)
- {
- TEST_LOG("Resetting sound device several time with pause %d ms\n", p);
- for (int l=0; l<20; ++l)
- {
- TEST(audioDevice->ResetAudioDevice() == 0);
- AudioDeviceUtility::Sleep(p);
- }
- TEST_LOG("\n> Speak into the microphone and verify that the audio is good.\n");
- AudioDeviceUtility::Sleep(2000);
- }
-#endif
-
-#if defined(MAC_IPHONE)
- bool loudspeakerOn(false);
- TEST_LOG("\nSet playout spaker\n \n");
- if (audioDevice->Recording() && audioDevice->Playing())
- {
- TEST_LOG("\n> Speak into the microphone and verify that the audio is good.\n\
-> Press any key to stop...\n \n");
- PAUSE(DEFAULT_PAUSE_TIME);
- }
-
- TEST_LOG("Set to use speaker\n");
- TEST(audioDevice->SetLoudspeakerStatus(true) == 0);
- TEST_LOG("\n> Speak into the microphone and verify that the audio is"
- " from the loudspeaker.\n\
-> Press any key to stop...\n \n");
- PAUSE(DEFAULT_PAUSE_TIME);
- TEST(audioDevice->GetLoudspeakerStatus(loudspeakerOn) == 0);
- TEST(loudspeakerOn == true);
-
- TEST_LOG("Set to not use speaker\n");
- TEST(audioDevice->SetLoudspeakerStatus(false) == 0);
- TEST_LOG("\n> Speak into the microphone and verify that the audio is not"
- " from the loudspeaker.\n\
-> Press any key to stop...\n \n");
- PAUSE(DEFAULT_PAUSE_TIME);
- TEST(audioDevice->GetLoudspeakerStatus(loudspeakerOn) == 0);
- TEST(loudspeakerOn == false);
-#endif
-
- TEST(audioDevice->StopRecording() == 0);
- TEST(audioDevice->StopPlayout() == 0);
- TEST(audioDevice->RegisterAudioCallback(NULL) == 0);
-
- _audioTransport->SetFullDuplex(false);
-
- TEST_LOG("\n");
- PRINT_TEST_RESULTS;
-
- return 0;
-}
-
-} // namespace webrtc
-
-// EOF
diff --git a/trunk/src/modules/audio_device/main/test/func_test_manager.h b/trunk/src/modules/audio_device/main/test/func_test_manager.h
deleted file mode 100644
index 5efae50..0000000
--- a/trunk/src/modules/audio_device/main/test/func_test_manager.h
+++ /dev/null
@@ -1,228 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H
-#define WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H
-
-#include "../source/audio_device_utility.h"
-
-#include <string>
-
-#include "typedefs.h"
-#include "audio_device.h"
-#include "audio_device_test_defines.h"
-#include "file_wrapper.h"
-#include "list_wrapper.h"
-#include "resampler.h"
-
-#if defined(MAC_IPHONE) || defined(ANDROID)
-#define USE_SLEEP_AS_PAUSE
-#else
-//#define USE_SLEEP_AS_PAUSE
-#endif
-
-// Sets the default pause time if using sleep as pause
-#define DEFAULT_PAUSE_TIME 5000
-
-#if defined(USE_SLEEP_AS_PAUSE)
-#define PAUSE(a) AudioDeviceUtility::Sleep(a);
-#else
-#define PAUSE(a) AudioDeviceUtility::WaitForKey();
-#endif
-
-#define SLEEP(a) AudioDeviceUtility::Sleep(a);
-
-#define ADM_AUDIO_LAYER AudioDeviceModule::kPlatformDefaultAudio
-//#define ADM_AUDIO_LAYER AudioDeviceModule::kLinuxPulseAudio
-
-enum TestType
-{
- TTInvalid = -1,
- TTAll = 0,
- TTAudioLayerSelection = 1,
- TTDeviceEnumeration = 2,
- TTDeviceSelection = 3,
- TTAudioTransport = 4,
- TTSpeakerVolume = 5,
- TTMicrophoneVolume = 6,
- TTSpeakerMute = 7,
- TTMicrophoneMute = 8,
- TTMicrophoneBoost = 9,
- TTMicrophoneAGC = 10,
- TTLoopback = 11,
- TTDeviceRemoval = 13,
- TTMobileAPI = 14,
- TTTest = 66,
-};
-
-class ProcessThread;
-
-namespace webrtc
-{
-
-class AudioDeviceModule;
-class AudioEventObserver;
-class AudioTransport;
-
-// ----------------------------------------------------------------------------
-// AudioEventObserver
-// ----------------------------------------------------------------------------
-
-class AudioEventObserver: public AudioDeviceObserver
-{
-public:
- virtual void OnErrorIsReported(const ErrorCode error);
- virtual void OnWarningIsReported(const WarningCode warning);
- AudioEventObserver(AudioDeviceModule* audioDevice);
- ~AudioEventObserver();
-public:
- ErrorCode _error;
- WarningCode _warning;
-private:
- AudioDeviceModule* _audioDevice;
-};
-
-// ----------------------------------------------------------------------------
-// AudioTransport
-// ----------------------------------------------------------------------------
-
-class AudioTransportImpl: public AudioTransport
-{
-public:
- virtual WebRtc_Word32
- RecordedDataIsAvailable(const WebRtc_Word8* audioSamples,
- const WebRtc_UWord32 nSamples,
- const WebRtc_UWord8 nBytesPerSample,
- const WebRtc_UWord8 nChannels,
- const WebRtc_UWord32 samplesPerSec,
- const WebRtc_UWord32 totalDelayMS,
- const WebRtc_Word32 clockDrift,
- const WebRtc_UWord32 currentMicLevel,
- WebRtc_UWord32& newMicLevel);
-
- virtual WebRtc_Word32 NeedMorePlayData(const WebRtc_UWord32 nSamples,
- const WebRtc_UWord8 nBytesPerSample,
- const WebRtc_UWord8 nChannels,
- const WebRtc_UWord32 samplesPerSec,
- WebRtc_Word8* audioSamples,
- WebRtc_UWord32& nSamplesOut);
-
- AudioTransportImpl(AudioDeviceModule* audioDevice);
- ~AudioTransportImpl();
-
-public:
- WebRtc_Word32 SetFilePlayout(bool enable, const WebRtc_Word8* fileName =
- NULL);
- void SetFullDuplex(bool enable);
- void SetSpeakerVolume(bool enable)
- {
- _speakerVolume = enable;
- }
- ;
- void SetSpeakerMute(bool enable)
- {
- _speakerMute = enable;
- }
- ;
- void SetMicrophoneMute(bool enable)
- {
- _microphoneMute = enable;
- }
- ;
- void SetMicrophoneVolume(bool enable)
- {
- _microphoneVolume = enable;
- }
- ;
- void SetMicrophoneBoost(bool enable)
- {
- _microphoneBoost = enable;
- }
- ;
- void SetLoopbackMeasurements(bool enable)
- {
- _loopBackMeasurements = enable;
- }
- ;
- void SetMicrophoneAGC(bool enable)
- {
- _microphoneAGC = enable;
- }
- ;
-
-private:
- AudioDeviceModule* _audioDevice;
-
- bool _playFromFile;
- bool _fullDuplex;
- bool _speakerVolume;
- bool _speakerMute;
- bool _microphoneVolume;
- bool _microphoneMute;
- bool _microphoneBoost;
- bool _microphoneAGC;
- bool _loopBackMeasurements;
-
- FileWrapper& _playFile;
-
- WebRtc_UWord32 _recCount;
- WebRtc_UWord32 _playCount;
-
- ListWrapper _audioList;
-
- Resampler _resampler;
-};
-
-// ----------------------------------------------------------------------------
-// FuncTestManager
-// ----------------------------------------------------------------------------
-
-class FuncTestManager
-{
-public:
- FuncTestManager();
- ~FuncTestManager();
- WebRtc_Word32 Init();
- WebRtc_Word32 Close();
- WebRtc_Word32 DoTest(const TestType testType);
-private:
- WebRtc_Word32 TestAudioLayerSelection();
- WebRtc_Word32 TestDeviceEnumeration();
- WebRtc_Word32 TestDeviceSelection();
- WebRtc_Word32 TestAudioTransport();
- WebRtc_Word32 TestSpeakerVolume();
- WebRtc_Word32 TestMicrophoneVolume();
- WebRtc_Word32 TestSpeakerMute();
- WebRtc_Word32 TestMicrophoneMute();
- WebRtc_Word32 TestMicrophoneBoost();
- WebRtc_Word32 TestLoopback();
- WebRtc_Word32 TestDeviceRemoval();
- WebRtc_Word32 TestExtra();
- WebRtc_Word32 TestMicrophoneAGC();
- WebRtc_Word32 SelectPlayoutDevice();
- WebRtc_Word32 SelectRecordingDevice();
- WebRtc_Word32 TestAdvancedMBAPI();
-private:
- // Paths to where the resource files to be used for this test are located.
- std::string _resourcePath;
- std::string _playoutFile48;
- std::string _playoutFile44;
- std::string _playoutFile16;
- std::string _playoutFile8;
-
- ProcessThread* _processThread;
- AudioDeviceModule* _audioDevice;
- AudioEventObserver* _audioEventObserver;
- AudioTransportImpl* _audioTransport;
-};
-
-} // namespace webrtc
-
-#endif // #ifndef WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H
diff --git a/trunk/src/modules/audio_processing/Android.mk b/trunk/src/modules/audio_processing/Android.mk
deleted file mode 100644
index 2ab5bb6..0000000
--- a/trunk/src/modules/audio_processing/Android.mk
+++ /dev/null
@@ -1,143 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE := libwebrtc_apm
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := \
- $(call all-proto-files-under, .) \
- audio_buffer.cc \
- audio_processing_impl.cc \
- echo_cancellation_impl.cc \
- echo_control_mobile_impl.cc \
- gain_control_impl.cc \
- high_pass_filter_impl.cc \
- level_estimator_impl.cc \
- noise_suppression_impl.cc \
- splitting_filter.cc \
- processing_component.cc \
- voice_detection_impl.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS) \
- '-DWEBRTC_NS_FIXED'
-# floating point
-# -DWEBRTC_NS_FLOAT'
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include \
- $(LOCAL_PATH)/aec/include \
- $(LOCAL_PATH)/aecm/include \
- $(LOCAL_PATH)/agc/include \
- $(LOCAL_PATH)/ns/include \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../.. \
- $(LOCAL_PATH)/../../common_audio/signal_processing/include \
- $(LOCAL_PATH)/../../common_audio/vad/include \
- $(LOCAL_PATH)/../../system_wrappers/interface \
- external/protobuf/src
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
-
-# apm process test app
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_TAGS := tests
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES:= \
- $(call all-proto-files-under, .) \
- test/process_test.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../.. \
- $(LOCAL_PATH)/../../system_wrappers/interface \
- external/gtest/include
-
-LOCAL_STATIC_LIBRARIES := \
- libgtest \
- libprotobuf-cpp-2.3.0-lite
-
-LOCAL_SHARED_LIBRARIES := \
- libutils \
- libstlport \
- libwebrtc_audio_preprocessing
-
-LOCAL_MODULE:= webrtc_apm_process_test
-
-ifdef NDK_ROOT
-include $(BUILD_EXECUTABLE)
-else
-include external/stlport/libstlport.mk
-include $(BUILD_NATIVE_TEST)
-endif
-
-# apm unit test app
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_TAGS := tests
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES:= \
- $(call all-proto-files-under, test) \
- test/unit_test.cc \
- ../../../test/testsupport/fileutils.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS) \
- '-DWEBRTC_APM_UNIT_TEST_FIXED_PROFILE'
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../.. \
- $(LOCAL_PATH)/../../../test \
- $(LOCAL_PATH)/../../system_wrappers/interface \
- $(LOCAL_PATH)/../../common_audio/signal_processing/include \
- external/gtest/include \
- external/protobuf/src
-
-LOCAL_STATIC_LIBRARIES := \
- libgtest \
- libprotobuf-cpp-2.3.0-lite
-
-LOCAL_SHARED_LIBRARIES := \
- libstlport \
- libwebrtc_audio_preprocessing
-
-LOCAL_MODULE:= webrtc_apm_unit_test
-
-ifdef NDK_ROOT
-include $(BUILD_EXECUTABLE)
-else
-include external/stlport/libstlport.mk
-include $(BUILD_NATIVE_TEST)
-endif
diff --git a/trunk/src/modules/audio_processing/OWNERS b/trunk/src/modules/audio_processing/OWNERS
deleted file mode 100644
index 5a25634..0000000
--- a/trunk/src/modules/audio_processing/OWNERS
+++ /dev/null
@@ -1,2 +0,0 @@
-andrew@webrtc.org
-bjornv@webrtc.org
diff --git a/trunk/src/modules/audio_processing/aec/Android.mk b/trunk/src/modules/audio_processing/aec/Android.mk
deleted file mode 100644
index 26679d8..0000000
--- a/trunk/src/modules/audio_processing/aec/Android.mk
+++ /dev/null
@@ -1,48 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../android-webrtc.mk
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_aec
-LOCAL_MODULE_TAGS := optional
-LOCAL_SRC_FILES := \
- echo_cancellation.c \
- aec_resampler.c \
- aec_core.c \
- aec_rdft.c \
-
-ifeq ($(TARGET_ARCH),x86)
-LOCAL_SRC_FILES += \
- aec_core_sse2.c \
- aec_rdft_sse2.c
-endif
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include \
- $(LOCAL_PATH)/../utility \
- $(LOCAL_PATH)/../../.. \
- $(LOCAL_PATH)/../../../common_audio/signal_processing/include
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_processing/aec/aec.gypi b/trunk/src/modules/audio_processing/aec/aec.gypi
deleted file mode 100644
index 2e34c5b..0000000
--- a/trunk/src/modules/audio_processing/aec/aec.gypi
+++ /dev/null
@@ -1,74 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'aec',
- 'type': '<(library)',
- 'variables': {
- # Outputs some low-level debug files.
- 'aec_debug_dump%': 0,
- },
- 'dependencies': [
- 'apm_util',
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- ],
- 'include_dirs': [
- 'include',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- ],
- },
- 'sources': [
- 'include/echo_cancellation.h',
- 'echo_cancellation.c',
- 'aec_core.h',
- 'aec_core.c',
- 'aec_rdft.h',
- 'aec_rdft.c',
- 'aec_resampler.h',
- 'aec_resampler.c',
- ],
- 'conditions': [
- ['target_arch=="ia32" or target_arch=="x64"', {
- 'dependencies': [ 'aec_sse2', ],
- }],
- ['aec_debug_dump==1', {
- 'defines': [ 'WEBRTC_AEC_DEBUG_DUMP', ],
- }],
- ],
- },
- ],
- 'conditions': [
- ['target_arch=="ia32" or target_arch=="x64"', {
- 'targets': [
- {
- 'target_name': 'aec_sse2',
- 'type': '<(library)',
- 'sources': [
- 'aec_core_sse2.c',
- 'aec_rdft_sse2.c',
- ],
- 'conditions': [
- ['os_posix==1 and OS!="mac"', {
- 'cflags': [ '-msse2', ],
- }],
- ['OS=="mac"', {
- 'xcode_settings': {
- 'OTHER_CFLAGS': [ '-msse2', ],
- },
- }],
- ],
- },
- ],
- }],
- ],
-}
diff --git a/trunk/src/modules/audio_processing/aec/aec_core.c b/trunk/src/modules/audio_processing/aec/aec_core.c
deleted file mode 100644
index 1637e6f..0000000
--- a/trunk/src/modules/audio_processing/aec/aec_core.c
+++ /dev/null
@@ -1,1527 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * The core AEC algorithm, which is presented with time-aligned signals.
- */
-
-#include "aec_core.h"
-
-#include <assert.h>
-#include <math.h>
-#include <stddef.h> // size_t
-#include <stdlib.h>
-#include <string.h>
-
-#include "aec_rdft.h"
-#include "common_audio/signal_processing/include/signal_processing_library.h"
-#include "delay_estimator_wrapper.h"
-#include "ring_buffer.h"
-#include "system_wrappers/interface/cpu_features_wrapper.h"
-#include "typedefs.h"
-
-// Buffer size (samples)
-static const size_t kBufSizePartitions = 250; // 1 second of audio in 16 kHz.
-
-// Noise suppression
-static const int converged = 250;
-
-// Metrics
-static const int subCountLen = 4;
-static const int countLen = 50;
-
-// Quantities to control H band scaling for SWB input
-static const int flagHbandCn = 1; // flag for adding comfort noise in H band
-static const float cnScaleHband = (float)0.4; // scale for comfort noise in H band
-// Initial bin for averaging nlp gain in low band
-static const int freqAvgIc = PART_LEN / 2;
-
-// Matlab code to produce table:
-// win = sqrt(hanning(63)); win = [0 ; win(1:32)];
-// fprintf(1, '\t%.14f, %.14f, %.14f,\n', win);
-static const float sqrtHanning[65] = {
- 0.00000000000000f, 0.02454122852291f, 0.04906767432742f,
- 0.07356456359967f, 0.09801714032956f, 0.12241067519922f,
- 0.14673047445536f, 0.17096188876030f, 0.19509032201613f,
- 0.21910124015687f, 0.24298017990326f, 0.26671275747490f,
- 0.29028467725446f, 0.31368174039889f, 0.33688985339222f,
- 0.35989503653499f, 0.38268343236509f, 0.40524131400499f,
- 0.42755509343028f, 0.44961132965461f, 0.47139673682600f,
- 0.49289819222978f, 0.51410274419322f, 0.53499761988710f,
- 0.55557023301960f, 0.57580819141785f, 0.59569930449243f,
- 0.61523159058063f, 0.63439328416365f, 0.65317284295378f,
- 0.67155895484702f, 0.68954054473707f, 0.70710678118655f,
- 0.72424708295147f, 0.74095112535496f, 0.75720884650648f,
- 0.77301045336274f, 0.78834642762661f, 0.80320753148064f,
- 0.81758481315158f, 0.83146961230255f, 0.84485356524971f,
- 0.85772861000027f, 0.87008699110871f, 0.88192126434835f,
- 0.89322430119552f, 0.90398929312344f, 0.91420975570353f,
- 0.92387953251129f, 0.93299279883474f, 0.94154406518302f,
- 0.94952818059304f, 0.95694033573221f, 0.96377606579544f,
- 0.97003125319454f, 0.97570213003853f, 0.98078528040323f,
- 0.98527764238894f, 0.98917650996478f, 0.99247953459871f,
- 0.99518472667220f, 0.99729045667869f, 0.99879545620517f,
- 0.99969881869620f, 1.00000000000000f
-};
-
-// Matlab code to produce table:
-// weightCurve = [0 ; 0.3 * sqrt(linspace(0,1,64))' + 0.1];
-// fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', weightCurve);
-const float WebRtcAec_weightCurve[65] = {
- 0.0000f, 0.1000f, 0.1378f, 0.1535f, 0.1655f, 0.1756f,
- 0.1845f, 0.1926f, 0.2000f, 0.2069f, 0.2134f, 0.2195f,
- 0.2254f, 0.2309f, 0.2363f, 0.2414f, 0.2464f, 0.2512f,
- 0.2558f, 0.2604f, 0.2648f, 0.2690f, 0.2732f, 0.2773f,
- 0.2813f, 0.2852f, 0.2890f, 0.2927f, 0.2964f, 0.3000f,
- 0.3035f, 0.3070f, 0.3104f, 0.3138f, 0.3171f, 0.3204f,
- 0.3236f, 0.3268f, 0.3299f, 0.3330f, 0.3360f, 0.3390f,
- 0.3420f, 0.3449f, 0.3478f, 0.3507f, 0.3535f, 0.3563f,
- 0.3591f, 0.3619f, 0.3646f, 0.3673f, 0.3699f, 0.3726f,
- 0.3752f, 0.3777f, 0.3803f, 0.3828f, 0.3854f, 0.3878f,
- 0.3903f, 0.3928f, 0.3952f, 0.3976f, 0.4000f
-};
-
-// Matlab code to produce table:
-// overDriveCurve = [sqrt(linspace(0,1,65))' + 1];
-// fprintf(1, '\t%.4f, %.4f, %.4f, %.4f, %.4f, %.4f,\n', overDriveCurve);
-const float WebRtcAec_overDriveCurve[65] = {
- 1.0000f, 1.1250f, 1.1768f, 1.2165f, 1.2500f, 1.2795f,
- 1.3062f, 1.3307f, 1.3536f, 1.3750f, 1.3953f, 1.4146f,
- 1.4330f, 1.4507f, 1.4677f, 1.4841f, 1.5000f, 1.5154f,
- 1.5303f, 1.5449f, 1.5590f, 1.5728f, 1.5863f, 1.5995f,
- 1.6124f, 1.6250f, 1.6374f, 1.6495f, 1.6614f, 1.6731f,
- 1.6847f, 1.6960f, 1.7071f, 1.7181f, 1.7289f, 1.7395f,
- 1.7500f, 1.7603f, 1.7706f, 1.7806f, 1.7906f, 1.8004f,
- 1.8101f, 1.8197f, 1.8292f, 1.8385f, 1.8478f, 1.8570f,
- 1.8660f, 1.8750f, 1.8839f, 1.8927f, 1.9014f, 1.9100f,
- 1.9186f, 1.9270f, 1.9354f, 1.9437f, 1.9520f, 1.9601f,
- 1.9682f, 1.9763f, 1.9843f, 1.9922f, 2.0000f
-};
-
-// "Private" function prototypes.
-static void ProcessBlock(aec_t* aec);
-
-static void NonLinearProcessing(aec_t *aec, short *output, short *outputH);
-
-static void GetHighbandGain(const float *lambda, float *nlpGainHband);
-
-// Comfort_noise also computes noise for H band returned in comfortNoiseHband
-static void ComfortNoise(aec_t *aec, float efw[2][PART_LEN1],
- complex_t *comfortNoiseHband,
- const float *noisePow, const float *lambda);
-
-static void WebRtcAec_InitLevel(power_level_t *level);
-static void WebRtcAec_InitStats(stats_t *stats);
-static void UpdateLevel(power_level_t* level, float in[2][PART_LEN1]);
-static void UpdateMetrics(aec_t *aec);
-// Convert from time domain to frequency domain. Note that |time_data| are
-// overwritten.
-static void TimeToFrequency(float time_data[PART_LEN2],
- float freq_data[2][PART_LEN1],
- int window);
-
-__inline static float MulRe(float aRe, float aIm, float bRe, float bIm)
-{
- return aRe * bRe - aIm * bIm;
-}
-
-__inline static float MulIm(float aRe, float aIm, float bRe, float bIm)
-{
- return aRe * bIm + aIm * bRe;
-}
-
-static int CmpFloat(const void *a, const void *b)
-{
- const float *da = (const float *)a;
- const float *db = (const float *)b;
-
- return (*da > *db) - (*da < *db);
-}
-
-int WebRtcAec_CreateAec(aec_t **aecInst)
-{
- aec_t *aec = malloc(sizeof(aec_t));
- *aecInst = aec;
- if (aec == NULL) {
- return -1;
- }
-
- if (WebRtc_CreateBuffer(&aec->nearFrBuf,
- FRAME_LEN + PART_LEN,
- sizeof(int16_t)) == -1) {
- WebRtcAec_FreeAec(aec);
- aec = NULL;
- return -1;
- }
-
- if (WebRtc_CreateBuffer(&aec->outFrBuf,
- FRAME_LEN + PART_LEN,
- sizeof(int16_t)) == -1) {
- WebRtcAec_FreeAec(aec);
- aec = NULL;
- return -1;
- }
-
- if (WebRtc_CreateBuffer(&aec->nearFrBufH,
- FRAME_LEN + PART_LEN,
- sizeof(int16_t)) == -1) {
- WebRtcAec_FreeAec(aec);
- aec = NULL;
- return -1;
- }
-
- if (WebRtc_CreateBuffer(&aec->outFrBufH,
- FRAME_LEN + PART_LEN,
- sizeof(int16_t)) == -1) {
- WebRtcAec_FreeAec(aec);
- aec = NULL;
- return -1;
- }
-
- // Create far-end buffers.
- if (WebRtc_CreateBuffer(&aec->far_buf, kBufSizePartitions,
- sizeof(float) * 2 * PART_LEN1) == -1) {
- WebRtcAec_FreeAec(aec);
- aec = NULL;
- return -1;
- }
- if (WebRtc_CreateBuffer(&aec->far_buf_windowed, kBufSizePartitions,
- sizeof(float) * 2 * PART_LEN1) == -1) {
- WebRtcAec_FreeAec(aec);
- aec = NULL;
- return -1;
- }
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- if (WebRtc_CreateBuffer(&aec->far_time_buf, kBufSizePartitions,
- sizeof(int16_t) * PART_LEN) == -1) {
- WebRtcAec_FreeAec(aec);
- aec = NULL;
- return -1;
- }
-#endif
- if (WebRtc_CreateDelayEstimator(&aec->delay_estimator,
- PART_LEN1,
- kMaxDelayBlocks,
- kLookaheadBlocks) == -1) {
- WebRtcAec_FreeAec(aec);
- aec = NULL;
- return -1;
- }
-
- return 0;
-}
-
-int WebRtcAec_FreeAec(aec_t *aec)
-{
- if (aec == NULL) {
- return -1;
- }
-
- WebRtc_FreeBuffer(aec->nearFrBuf);
- WebRtc_FreeBuffer(aec->outFrBuf);
-
- WebRtc_FreeBuffer(aec->nearFrBufH);
- WebRtc_FreeBuffer(aec->outFrBufH);
-
- WebRtc_FreeBuffer(aec->far_buf);
- WebRtc_FreeBuffer(aec->far_buf_windowed);
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- WebRtc_FreeBuffer(aec->far_time_buf);
-#endif
- WebRtc_FreeDelayEstimator(aec->delay_estimator);
-
- free(aec);
- return 0;
-}
-
-static void FilterFar(aec_t *aec, float yf[2][PART_LEN1])
-{
- int i;
- for (i = 0; i < NR_PART; i++) {
- int j;
- int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
- int pos = i * PART_LEN1;
- // Check for wrap
- if (i + aec->xfBufBlockPos >= NR_PART) {
- xPos -= NR_PART*(PART_LEN1);
- }
-
- for (j = 0; j < PART_LEN1; j++) {
- yf[0][j] += MulRe(aec->xfBuf[0][xPos + j], aec->xfBuf[1][xPos + j],
- aec->wfBuf[0][ pos + j], aec->wfBuf[1][ pos + j]);
- yf[1][j] += MulIm(aec->xfBuf[0][xPos + j], aec->xfBuf[1][xPos + j],
- aec->wfBuf[0][ pos + j], aec->wfBuf[1][ pos + j]);
- }
- }
-}
-
-static void ScaleErrorSignal(aec_t *aec, float ef[2][PART_LEN1])
-{
- int i;
- float absEf;
- for (i = 0; i < (PART_LEN1); i++) {
- ef[0][i] /= (aec->xPow[i] + 1e-10f);
- ef[1][i] /= (aec->xPow[i] + 1e-10f);
- absEf = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
-
- if (absEf > aec->errThresh) {
- absEf = aec->errThresh / (absEf + 1e-10f);
- ef[0][i] *= absEf;
- ef[1][i] *= absEf;
- }
-
- // Stepsize factor
- ef[0][i] *= aec->mu;
- ef[1][i] *= aec->mu;
- }
-}
-
-// Time-unconstrined filter adaptation.
-// TODO(andrew): consider for a low-complexity mode.
-//static void FilterAdaptationUnconstrained(aec_t *aec, float *fft,
-// float ef[2][PART_LEN1]) {
-// int i, j;
-// for (i = 0; i < NR_PART; i++) {
-// int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
-// int pos;
-// // Check for wrap
-// if (i + aec->xfBufBlockPos >= NR_PART) {
-// xPos -= NR_PART * PART_LEN1;
-// }
-//
-// pos = i * PART_LEN1;
-//
-// for (j = 0; j < PART_LEN1; j++) {
-// aec->wfBuf[pos + j][0] += MulRe(aec->xfBuf[xPos + j][0],
-// -aec->xfBuf[xPos + j][1],
-// ef[j][0], ef[j][1]);
-// aec->wfBuf[pos + j][1] += MulIm(aec->xfBuf[xPos + j][0],
-// -aec->xfBuf[xPos + j][1],
-// ef[j][0], ef[j][1]);
-// }
-// }
-//}
-
-static void FilterAdaptation(aec_t *aec, float *fft, float ef[2][PART_LEN1]) {
- int i, j;
- for (i = 0; i < NR_PART; i++) {
- int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
- int pos;
- // Check for wrap
- if (i + aec->xfBufBlockPos >= NR_PART) {
- xPos -= NR_PART * PART_LEN1;
- }
-
- pos = i * PART_LEN1;
-
- for (j = 0; j < PART_LEN; j++) {
-
- fft[2 * j] = MulRe(aec->xfBuf[0][xPos + j],
- -aec->xfBuf[1][xPos + j],
- ef[0][j], ef[1][j]);
- fft[2 * j + 1] = MulIm(aec->xfBuf[0][xPos + j],
- -aec->xfBuf[1][xPos + j],
- ef[0][j], ef[1][j]);
- }
- fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN],
- -aec->xfBuf[1][xPos + PART_LEN],
- ef[0][PART_LEN], ef[1][PART_LEN]);
-
- aec_rdft_inverse_128(fft);
- memset(fft + PART_LEN, 0, sizeof(float) * PART_LEN);
-
- // fft scaling
- {
- float scale = 2.0f / PART_LEN2;
- for (j = 0; j < PART_LEN; j++) {
- fft[j] *= scale;
- }
- }
- aec_rdft_forward_128(fft);
-
- aec->wfBuf[0][pos] += fft[0];
- aec->wfBuf[0][pos + PART_LEN] += fft[1];
-
- for (j = 1; j < PART_LEN; j++) {
- aec->wfBuf[0][pos + j] += fft[2 * j];
- aec->wfBuf[1][pos + j] += fft[2 * j + 1];
- }
- }
-}
-
-static void OverdriveAndSuppress(aec_t *aec, float hNl[PART_LEN1],
- const float hNlFb,
- float efw[2][PART_LEN1]) {
- int i;
- for (i = 0; i < PART_LEN1; i++) {
- // Weight subbands
- if (hNl[i] > hNlFb) {
- hNl[i] = WebRtcAec_weightCurve[i] * hNlFb +
- (1 - WebRtcAec_weightCurve[i]) * hNl[i];
- }
- hNl[i] = powf(hNl[i], aec->overDriveSm * WebRtcAec_overDriveCurve[i]);
-
- // Suppress error signal
- efw[0][i] *= hNl[i];
- efw[1][i] *= hNl[i];
-
- // Ooura fft returns incorrect sign on imaginary component. It matters here
- // because we are making an additive change with comfort noise.
- efw[1][i] *= -1;
- }
-}
-
-WebRtcAec_FilterFar_t WebRtcAec_FilterFar;
-WebRtcAec_ScaleErrorSignal_t WebRtcAec_ScaleErrorSignal;
-WebRtcAec_FilterAdaptation_t WebRtcAec_FilterAdaptation;
-WebRtcAec_OverdriveAndSuppress_t WebRtcAec_OverdriveAndSuppress;
-
-int WebRtcAec_InitAec(aec_t *aec, int sampFreq)
-{
- int i;
-
- aec->sampFreq = sampFreq;
-
- if (sampFreq == 8000) {
- aec->mu = 0.6f;
- aec->errThresh = 2e-6f;
- }
- else {
- aec->mu = 0.5f;
- aec->errThresh = 1.5e-6f;
- }
-
- if (WebRtc_InitBuffer(aec->nearFrBuf) == -1) {
- return -1;
- }
-
- if (WebRtc_InitBuffer(aec->outFrBuf) == -1) {
- return -1;
- }
-
- if (WebRtc_InitBuffer(aec->nearFrBufH) == -1) {
- return -1;
- }
-
- if (WebRtc_InitBuffer(aec->outFrBufH) == -1) {
- return -1;
- }
-
- // Initialize far-end buffers.
- if (WebRtc_InitBuffer(aec->far_buf) == -1) {
- return -1;
- }
- if (WebRtc_InitBuffer(aec->far_buf_windowed) == -1) {
- return -1;
- }
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- if (WebRtc_InitBuffer(aec->far_time_buf) == -1) {
- return -1;
- }
-#endif
- aec->system_delay = 0;
-
- if (WebRtc_InitDelayEstimator(aec->delay_estimator) != 0) {
- return -1;
- }
- aec->delay_logging_enabled = 0;
- memset(aec->delay_histogram, 0, sizeof(aec->delay_histogram));
-
- // Default target suppression level
- aec->targetSupp = -11.5;
- aec->minOverDrive = 2.0;
-
- // Sampling frequency multiplier
- // SWB is processed as 160 frame size
- if (aec->sampFreq == 32000) {
- aec->mult = (short)aec->sampFreq / 16000;
- }
- else {
- aec->mult = (short)aec->sampFreq / 8000;
- }
-
- aec->farBufWritePos = 0;
- aec->farBufReadPos = 0;
-
- aec->inSamples = 0;
- aec->outSamples = 0;
- aec->knownDelay = 0;
-
- // Initialize buffers
- memset(aec->dBuf, 0, sizeof(aec->dBuf));
- memset(aec->eBuf, 0, sizeof(aec->eBuf));
- // For H band
- memset(aec->dBufH, 0, sizeof(aec->dBufH));
-
- memset(aec->xPow, 0, sizeof(aec->xPow));
- memset(aec->dPow, 0, sizeof(aec->dPow));
- memset(aec->dInitMinPow, 0, sizeof(aec->dInitMinPow));
- aec->noisePow = aec->dInitMinPow;
- aec->noiseEstCtr = 0;
-
- // Initial comfort noise power
- for (i = 0; i < PART_LEN1; i++) {
- aec->dMinPow[i] = 1.0e6f;
- }
-
- // Holds the last block written to
- aec->xfBufBlockPos = 0;
- // TODO: Investigate need for these initializations. Deleting them doesn't
- // change the output at all and yields 0.4% overall speedup.
- memset(aec->xfBuf, 0, sizeof(complex_t) * NR_PART * PART_LEN1);
- memset(aec->wfBuf, 0, sizeof(complex_t) * NR_PART * PART_LEN1);
- memset(aec->sde, 0, sizeof(complex_t) * PART_LEN1);
- memset(aec->sxd, 0, sizeof(complex_t) * PART_LEN1);
- memset(aec->xfwBuf, 0, sizeof(complex_t) * NR_PART * PART_LEN1);
- memset(aec->se, 0, sizeof(float) * PART_LEN1);
-
- // To prevent numerical instability in the first block.
- for (i = 0; i < PART_LEN1; i++) {
- aec->sd[i] = 1;
- }
- for (i = 0; i < PART_LEN1; i++) {
- aec->sx[i] = 1;
- }
-
- memset(aec->hNs, 0, sizeof(aec->hNs));
- memset(aec->outBuf, 0, sizeof(float) * PART_LEN);
-
- aec->hNlFbMin = 1;
- aec->hNlFbLocalMin = 1;
- aec->hNlXdAvgMin = 1;
- aec->hNlNewMin = 0;
- aec->hNlMinCtr = 0;
- aec->overDrive = 2;
- aec->overDriveSm = 2;
- aec->delayIdx = 0;
- aec->stNearState = 0;
- aec->echoState = 0;
- aec->divergeState = 0;
-
- aec->seed = 777;
- aec->delayEstCtr = 0;
-
- // Metrics disabled by default
- aec->metricsMode = 0;
- WebRtcAec_InitMetrics(aec);
-
- // Assembly optimization
- WebRtcAec_FilterFar = FilterFar;
- WebRtcAec_ScaleErrorSignal = ScaleErrorSignal;
- WebRtcAec_FilterAdaptation = FilterAdaptation;
- WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppress;
-
-#if defined(WEBRTC_ARCH_X86_FAMILY)
- if (WebRtc_GetCPUInfo(kSSE2)) {
- WebRtcAec_InitAec_SSE2();
- }
-#endif
-
- aec_rdft_init();
-
- return 0;
-}
-
-void WebRtcAec_InitMetrics(aec_t *aec)
-{
- aec->stateCounter = 0;
- WebRtcAec_InitLevel(&aec->farlevel);
- WebRtcAec_InitLevel(&aec->nearlevel);
- WebRtcAec_InitLevel(&aec->linoutlevel);
- WebRtcAec_InitLevel(&aec->nlpoutlevel);
-
- WebRtcAec_InitStats(&aec->erl);
- WebRtcAec_InitStats(&aec->erle);
- WebRtcAec_InitStats(&aec->aNlp);
- WebRtcAec_InitStats(&aec->rerl);
-}
-
-
-void WebRtcAec_BufferFarendPartition(aec_t *aec, const float* farend) {
- float fft[PART_LEN2];
- float xf[2][PART_LEN1];
-
- // Check if the buffer is full, and in that case flush the oldest data.
- if (WebRtc_available_write(aec->far_buf) < 1) {
- WebRtc_MoveReadPtr(aec->far_buf, 1);
- WebRtc_MoveReadPtr(aec->far_buf_windowed, 1);
- aec->system_delay -= PART_LEN;
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- WebRtc_MoveReadPtr(aec->far_time_buf, 1);
-#endif
- }
- // Convert far-end partition to the frequency domain without windowing.
- memcpy(fft, farend, sizeof(float) * PART_LEN2);
- TimeToFrequency(fft, xf, 0);
- WebRtc_WriteBuffer(aec->far_buf, &xf[0][0], 1);
-
- // Convert far-end partition to the frequency domain with windowing.
- memcpy(fft, farend, sizeof(float) * PART_LEN2);
- TimeToFrequency(fft, xf, 1);
- WebRtc_WriteBuffer(aec->far_buf_windowed, &xf[0][0], 1);
-}
-
-void WebRtcAec_ProcessFrame(aec_t *aec,
- const short *nearend,
- const short *nearendH,
- int knownDelay)
-{
- // For each frame the process is as follows:
- // 1) If the system_delay indicates on being too small for processing a
- // frame we stuff the buffer with enough data for 10 ms.
- // 2) Adjust the buffer to the system delay, by moving the read pointer.
- // 3) If we can't move read pointer due to buffer size limitations we
- // flush/stuff the buffer.
- // 4) Process as many partitions as possible.
- // 5) Update the |system_delay| with respect to a full frame of FRAME_LEN
- // samples. Even though we will have data left to process (we work with
- // partitions) we consider updating a whole frame, since that's the
- // amount of data we input and output in audio_processing.
-
- // TODO(bjornv): Investigate how we should round the delay difference; right
- // now we know that incoming |knownDelay| is underestimated when it's less
- // than |aec->knownDelay|. We therefore, round (-32) in that direction. In
- // the other direction, we don't have this situation, but might flush one
- // partition too little. This can cause non-causality, which should be
- // investigated. Maybe, allow for a non-symmetric rounding, like -16.
- int move_elements = (aec->knownDelay - knownDelay - 32) / PART_LEN;
- int moved_elements = 0;
-
- // TODO(bjornv): Change the near-end buffer handling to be the same as for
- // far-end, that is, with a near_pre_buf.
- // Buffer the near-end frame.
- WebRtc_WriteBuffer(aec->nearFrBuf, nearend, FRAME_LEN);
- // For H band
- if (aec->sampFreq == 32000) {
- WebRtc_WriteBuffer(aec->nearFrBufH, nearendH, FRAME_LEN);
- }
-
- // 1) At most we process |aec->mult|+1 partitions in 10 ms. Make sure we
- // have enough far-end data for that by stuffing the buffer if the
- // |system_delay| indicates others.
- if (aec->system_delay < FRAME_LEN) {
- // We don't have enough data so we rewind 10 ms.
- WebRtc_MoveReadPtr(aec->far_buf_windowed, -(aec->mult + 1));
- aec->system_delay -= WebRtc_MoveReadPtr(aec->far_buf, -(aec->mult + 1)) *
- PART_LEN;
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- WebRtc_MoveReadPtr(aec->far_time_buf, -(aec->mult + 1));
-#endif
- }
-
- // 2) Compensate for a possible change in the system delay.
-
- WebRtc_MoveReadPtr(aec->far_buf_windowed, move_elements);
- moved_elements = WebRtc_MoveReadPtr(aec->far_buf, move_elements);
- aec->knownDelay -= moved_elements * PART_LEN;
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- WebRtc_MoveReadPtr(aec->far_time_buf, move_elements);
-#endif
-
- // 4) Process as many blocks as possible.
- while (WebRtc_available_read(aec->nearFrBuf) >= PART_LEN) {
- ProcessBlock(aec);
- }
-
- // 5) Update system delay with respect to the entire frame.
- aec->system_delay -= FRAME_LEN;
-}
-
-static void ProcessBlock(aec_t* aec) {
- int i;
- float d[PART_LEN], y[PART_LEN], e[PART_LEN], dH[PART_LEN];
- float scale;
-
- float fft[PART_LEN2];
- float xf[2][PART_LEN1], yf[2][PART_LEN1], ef[2][PART_LEN1];
- float df[2][PART_LEN1];
- float far_spectrum = 0.0f;
- float near_spectrum = 0.0f;
- float abs_far_spectrum[PART_LEN1];
- float abs_near_spectrum[PART_LEN1];
-
- const float gPow[2] = {0.9f, 0.1f};
-
- // Noise estimate constants.
- const int noiseInitBlocks = 500 * aec->mult;
- const float step = 0.1f;
- const float ramp = 1.0002f;
- const float gInitNoise[2] = {0.999f, 0.001f};
-
- int16_t nearend[PART_LEN];
- int16_t* nearend_ptr = NULL;
- int16_t output[PART_LEN];
- int16_t outputH[PART_LEN];
-
- float* xf_ptr = NULL;
-
- memset(dH, 0, sizeof(dH));
- if (aec->sampFreq == 32000) {
- // Get the upper band first so we can reuse |nearend|.
- WebRtc_ReadBuffer(aec->nearFrBufH,
- (void**) &nearend_ptr,
- nearend,
- PART_LEN);
- for (i = 0; i < PART_LEN; i++) {
- dH[i] = (float) (nearend_ptr[i]);
- }
- memcpy(aec->dBufH + PART_LEN, dH, sizeof(float) * PART_LEN);
- }
- WebRtc_ReadBuffer(aec->nearFrBuf, (void**) &nearend_ptr, nearend, PART_LEN);
-
- // ---------- Ooura fft ----------
- // Concatenate old and new nearend blocks.
- for (i = 0; i < PART_LEN; i++) {
- d[i] = (float) (nearend_ptr[i]);
- }
- memcpy(aec->dBuf + PART_LEN, d, sizeof(float) * PART_LEN);
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- {
- int16_t farend[PART_LEN];
- int16_t* farend_ptr = NULL;
- WebRtc_ReadBuffer(aec->far_time_buf, (void**) &farend_ptr, farend, 1);
- fwrite(farend_ptr, sizeof(int16_t), PART_LEN, aec->farFile);
- fwrite(nearend_ptr, sizeof(int16_t), PART_LEN, aec->nearFile);
- }
-#endif
-
- // We should always have at least one element stored in |far_buf|.
- assert(WebRtc_available_read(aec->far_buf) > 0);
- WebRtc_ReadBuffer(aec->far_buf, (void**) &xf_ptr, &xf[0][0], 1);
-
- // Near fft
- memcpy(fft, aec->dBuf, sizeof(float) * PART_LEN2);
- TimeToFrequency(fft, df, 0);
-
- // Power smoothing
- for (i = 0; i < PART_LEN1; i++) {
- far_spectrum = (xf_ptr[i] * xf_ptr[i]) +
- (xf_ptr[PART_LEN1 + i] * xf_ptr[PART_LEN1 + i]);
- aec->xPow[i] = gPow[0] * aec->xPow[i] + gPow[1] * NR_PART * far_spectrum;
- // Calculate absolute spectra
- abs_far_spectrum[i] = sqrtf(far_spectrum);
-
- near_spectrum = df[0][i] * df[0][i] + df[1][i] * df[1][i];
- aec->dPow[i] = gPow[0] * aec->dPow[i] + gPow[1] * near_spectrum;
- // Calculate absolute spectra
- abs_near_spectrum[i] = sqrtf(near_spectrum);
- }
-
- // Estimate noise power. Wait until dPow is more stable.
- if (aec->noiseEstCtr > 50) {
- for (i = 0; i < PART_LEN1; i++) {
- if (aec->dPow[i] < aec->dMinPow[i]) {
- aec->dMinPow[i] = (aec->dPow[i] + step * (aec->dMinPow[i] -
- aec->dPow[i])) * ramp;
- }
- else {
- aec->dMinPow[i] *= ramp;
- }
- }
- }
-
- // Smooth increasing noise power from zero at the start,
- // to avoid a sudden burst of comfort noise.
- if (aec->noiseEstCtr < noiseInitBlocks) {
- aec->noiseEstCtr++;
- for (i = 0; i < PART_LEN1; i++) {
- if (aec->dMinPow[i] > aec->dInitMinPow[i]) {
- aec->dInitMinPow[i] = gInitNoise[0] * aec->dInitMinPow[i] +
- gInitNoise[1] * aec->dMinPow[i];
- }
- else {
- aec->dInitMinPow[i] = aec->dMinPow[i];
- }
- }
- aec->noisePow = aec->dInitMinPow;
- }
- else {
- aec->noisePow = aec->dMinPow;
- }
-
- // Block wise delay estimation used for logging
- if (aec->delay_logging_enabled) {
- int delay_estimate = 0;
- // Estimate the delay
- delay_estimate = WebRtc_DelayEstimatorProcessFloat(aec->delay_estimator,
- abs_far_spectrum,
- abs_near_spectrum,
- PART_LEN1);
- if (delay_estimate >= 0) {
- // Update delay estimate buffer.
- aec->delay_histogram[delay_estimate]++;
- }
- }
-
- // Update the xfBuf block position.
- aec->xfBufBlockPos--;
- if (aec->xfBufBlockPos == -1) {
- aec->xfBufBlockPos = NR_PART - 1;
- }
-
- // Buffer xf
- memcpy(aec->xfBuf[0] + aec->xfBufBlockPos * PART_LEN1, xf_ptr,
- sizeof(float) * PART_LEN1);
- memcpy(aec->xfBuf[1] + aec->xfBufBlockPos * PART_LEN1, &xf_ptr[PART_LEN1],
- sizeof(float) * PART_LEN1);
-
- memset(yf[0], 0, sizeof(float) * (PART_LEN1 * 2));
-
- // Filter far
- WebRtcAec_FilterFar(aec, yf);
-
- // Inverse fft to obtain echo estimate and error.
- fft[0] = yf[0][0];
- fft[1] = yf[0][PART_LEN];
- for (i = 1; i < PART_LEN; i++) {
- fft[2 * i] = yf[0][i];
- fft[2 * i + 1] = yf[1][i];
- }
- aec_rdft_inverse_128(fft);
-
- scale = 2.0f / PART_LEN2;
- for (i = 0; i < PART_LEN; i++) {
- y[i] = fft[PART_LEN + i] * scale; // fft scaling
- }
-
- for (i = 0; i < PART_LEN; i++) {
- e[i] = d[i] - y[i];
- }
-
- // Error fft
- memcpy(aec->eBuf + PART_LEN, e, sizeof(float) * PART_LEN);
- memset(fft, 0, sizeof(float) * PART_LEN);
- memcpy(fft + PART_LEN, e, sizeof(float) * PART_LEN);
- // TODO(bjornv): Change to use TimeToFrequency().
- aec_rdft_forward_128(fft);
-
- ef[1][0] = 0;
- ef[1][PART_LEN] = 0;
- ef[0][0] = fft[0];
- ef[0][PART_LEN] = fft[1];
- for (i = 1; i < PART_LEN; i++) {
- ef[0][i] = fft[2 * i];
- ef[1][i] = fft[2 * i + 1];
- }
-
- if (aec->metricsMode == 1) {
- // Note that the first PART_LEN samples in fft (before transformation) are
- // zero. Hence, the scaling by two in UpdateLevel() should not be
- // performed. That scaling is taken care of in UpdateMetrics() instead.
- UpdateLevel(&aec->linoutlevel, ef);
- }
-
- // Scale error signal inversely with far power.
- WebRtcAec_ScaleErrorSignal(aec, ef);
- WebRtcAec_FilterAdaptation(aec, fft, ef);
- NonLinearProcessing(aec, output, outputH);
-
- if (aec->metricsMode == 1) {
- // Update power levels and echo metrics
- UpdateLevel(&aec->farlevel, (float (*)[PART_LEN1]) xf_ptr);
- UpdateLevel(&aec->nearlevel, df);
- UpdateMetrics(aec);
- }
-
- // Store the output block.
- WebRtc_WriteBuffer(aec->outFrBuf, output, PART_LEN);
- // For H band
- if (aec->sampFreq == 32000) {
- WebRtc_WriteBuffer(aec->outFrBufH, outputH, PART_LEN);
- }
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- {
- int16_t eInt16[PART_LEN];
- for (i = 0; i < PART_LEN; i++) {
- eInt16[i] = (int16_t)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, e[i],
- WEBRTC_SPL_WORD16_MIN);
- }
-
- fwrite(eInt16, sizeof(int16_t), PART_LEN, aec->outLinearFile);
- fwrite(output, sizeof(int16_t), PART_LEN, aec->outFile);
- }
-#endif
-}
-
-static void NonLinearProcessing(aec_t *aec, short *output, short *outputH)
-{
- float efw[2][PART_LEN1], dfw[2][PART_LEN1], xfw[2][PART_LEN1];
- complex_t comfortNoiseHband[PART_LEN1];
- float fft[PART_LEN2];
- float scale, dtmp;
- float nlpGainHband;
- int i, j, pos;
-
- // Coherence and non-linear filter
- float cohde[PART_LEN1], cohxd[PART_LEN1];
- float hNlDeAvg, hNlXdAvg;
- float hNl[PART_LEN1];
- float hNlPref[PREF_BAND_SIZE];
- float hNlFb = 0, hNlFbLow = 0;
- const float prefBandQuant = 0.75f, prefBandQuantLow = 0.5f;
- const int prefBandSize = PREF_BAND_SIZE / aec->mult;
- const int minPrefBand = 4 / aec->mult;
-
- // Near and error power sums
- float sdSum = 0, seSum = 0;
-
- // Power estimate smoothing coefficients
- const float gCoh[2][2] = {{0.9f, 0.1f}, {0.93f, 0.07f}};
- const float *ptrGCoh = gCoh[aec->mult - 1];
-
- // Filter energy
- float wfEnMax = 0, wfEn = 0;
- const int delayEstInterval = 10 * aec->mult;
-
- float* xfw_ptr = NULL;
-
- aec->delayEstCtr++;
- if (aec->delayEstCtr == delayEstInterval) {
- aec->delayEstCtr = 0;
- }
-
- // initialize comfort noise for H band
- memset(comfortNoiseHband, 0, sizeof(comfortNoiseHband));
- nlpGainHband = (float)0.0;
- dtmp = (float)0.0;
-
- // Measure energy in each filter partition to determine delay.
- // TODO: Spread by computing one partition per block?
- if (aec->delayEstCtr == 0) {
- wfEnMax = 0;
- aec->delayIdx = 0;
- for (i = 0; i < NR_PART; i++) {
- pos = i * PART_LEN1;
- wfEn = 0;
- for (j = 0; j < PART_LEN1; j++) {
- wfEn += aec->wfBuf[0][pos + j] * aec->wfBuf[0][pos + j] +
- aec->wfBuf[1][pos + j] * aec->wfBuf[1][pos + j];
- }
-
- if (wfEn > wfEnMax) {
- wfEnMax = wfEn;
- aec->delayIdx = i;
- }
- }
- }
-
- // We should always have at least one element stored in |far_buf|.
- assert(WebRtc_available_read(aec->far_buf_windowed) > 0);
- // NLP
- WebRtc_ReadBuffer(aec->far_buf_windowed, (void**) &xfw_ptr, &xfw[0][0], 1);
-
- // TODO(bjornv): Investigate if we can reuse |far_buf_windowed| instead of
- // |xfwBuf|.
- // Buffer far.
- memcpy(aec->xfwBuf, xfw_ptr, sizeof(float) * 2 * PART_LEN1);
-
- // Use delayed far.
- memcpy(xfw, aec->xfwBuf + aec->delayIdx * PART_LEN1, sizeof(xfw));
-
- // Windowed near fft
- for (i = 0; i < PART_LEN; i++) {
- fft[i] = aec->dBuf[i] * sqrtHanning[i];
- fft[PART_LEN + i] = aec->dBuf[PART_LEN + i] * sqrtHanning[PART_LEN - i];
- }
- aec_rdft_forward_128(fft);
-
- dfw[1][0] = 0;
- dfw[1][PART_LEN] = 0;
- dfw[0][0] = fft[0];
- dfw[0][PART_LEN] = fft[1];
- for (i = 1; i < PART_LEN; i++) {
- dfw[0][i] = fft[2 * i];
- dfw[1][i] = fft[2 * i + 1];
- }
-
- // Windowed error fft
- for (i = 0; i < PART_LEN; i++) {
- fft[i] = aec->eBuf[i] * sqrtHanning[i];
- fft[PART_LEN + i] = aec->eBuf[PART_LEN + i] * sqrtHanning[PART_LEN - i];
- }
- aec_rdft_forward_128(fft);
- efw[1][0] = 0;
- efw[1][PART_LEN] = 0;
- efw[0][0] = fft[0];
- efw[0][PART_LEN] = fft[1];
- for (i = 1; i < PART_LEN; i++) {
- efw[0][i] = fft[2 * i];
- efw[1][i] = fft[2 * i + 1];
- }
-
- // Smoothed PSD
- for (i = 0; i < PART_LEN1; i++) {
- aec->sd[i] = ptrGCoh[0] * aec->sd[i] + ptrGCoh[1] *
- (dfw[0][i] * dfw[0][i] + dfw[1][i] * dfw[1][i]);
- aec->se[i] = ptrGCoh[0] * aec->se[i] + ptrGCoh[1] *
- (efw[0][i] * efw[0][i] + efw[1][i] * efw[1][i]);
- // We threshold here to protect against the ill-effects of a zero farend.
- // The threshold is not arbitrarily chosen, but balances protection and
- // adverse interaction with the algorithm's tuning.
- // TODO: investigate further why this is so sensitive.
- aec->sx[i] = ptrGCoh[0] * aec->sx[i] + ptrGCoh[1] *
- WEBRTC_SPL_MAX(xfw[0][i] * xfw[0][i] + xfw[1][i] * xfw[1][i], 15);
-
- aec->sde[i][0] = ptrGCoh[0] * aec->sde[i][0] + ptrGCoh[1] *
- (dfw[0][i] * efw[0][i] + dfw[1][i] * efw[1][i]);
- aec->sde[i][1] = ptrGCoh[0] * aec->sde[i][1] + ptrGCoh[1] *
- (dfw[0][i] * efw[1][i] - dfw[1][i] * efw[0][i]);
-
- aec->sxd[i][0] = ptrGCoh[0] * aec->sxd[i][0] + ptrGCoh[1] *
- (dfw[0][i] * xfw[0][i] + dfw[1][i] * xfw[1][i]);
- aec->sxd[i][1] = ptrGCoh[0] * aec->sxd[i][1] + ptrGCoh[1] *
- (dfw[0][i] * xfw[1][i] - dfw[1][i] * xfw[0][i]);
-
- sdSum += aec->sd[i];
- seSum += aec->se[i];
- }
-
- // Divergent filter safeguard.
- if (aec->divergeState == 0) {
- if (seSum > sdSum) {
- aec->divergeState = 1;
- }
- }
- else {
- if (seSum * 1.05f < sdSum) {
- aec->divergeState = 0;
- }
- }
-
- if (aec->divergeState == 1) {
- memcpy(efw, dfw, sizeof(efw));
- }
-
- // Reset if error is significantly larger than nearend (13 dB).
- if (seSum > (19.95f * sdSum)) {
- memset(aec->wfBuf, 0, sizeof(aec->wfBuf));
- }
-
- // Subband coherence
- for (i = 0; i < PART_LEN1; i++) {
- cohde[i] = (aec->sde[i][0] * aec->sde[i][0] + aec->sde[i][1] * aec->sde[i][1]) /
- (aec->sd[i] * aec->se[i] + 1e-10f);
- cohxd[i] = (aec->sxd[i][0] * aec->sxd[i][0] + aec->sxd[i][1] * aec->sxd[i][1]) /
- (aec->sx[i] * aec->sd[i] + 1e-10f);
- }
-
- hNlXdAvg = 0;
- for (i = minPrefBand; i < prefBandSize + minPrefBand; i++) {
- hNlXdAvg += cohxd[i];
- }
- hNlXdAvg /= prefBandSize;
- hNlXdAvg = 1 - hNlXdAvg;
-
- hNlDeAvg = 0;
- for (i = minPrefBand; i < prefBandSize + minPrefBand; i++) {
- hNlDeAvg += cohde[i];
- }
- hNlDeAvg /= prefBandSize;
-
- if (hNlXdAvg < 0.75f && hNlXdAvg < aec->hNlXdAvgMin) {
- aec->hNlXdAvgMin = hNlXdAvg;
- }
-
- if (hNlDeAvg > 0.98f && hNlXdAvg > 0.9f) {
- aec->stNearState = 1;
- }
- else if (hNlDeAvg < 0.95f || hNlXdAvg < 0.8f) {
- aec->stNearState = 0;
- }
-
- if (aec->hNlXdAvgMin == 1) {
- aec->echoState = 0;
- aec->overDrive = aec->minOverDrive;
-
- if (aec->stNearState == 1) {
- memcpy(hNl, cohde, sizeof(hNl));
- hNlFb = hNlDeAvg;
- hNlFbLow = hNlDeAvg;
- }
- else {
- for (i = 0; i < PART_LEN1; i++) {
- hNl[i] = 1 - cohxd[i];
- }
- hNlFb = hNlXdAvg;
- hNlFbLow = hNlXdAvg;
- }
- }
- else {
-
- if (aec->stNearState == 1) {
- aec->echoState = 0;
- memcpy(hNl, cohde, sizeof(hNl));
- hNlFb = hNlDeAvg;
- hNlFbLow = hNlDeAvg;
- }
- else {
- aec->echoState = 1;
- for (i = 0; i < PART_LEN1; i++) {
- hNl[i] = WEBRTC_SPL_MIN(cohde[i], 1 - cohxd[i]);
- }
-
- // Select an order statistic from the preferred bands.
- // TODO: Using quicksort now, but a selection algorithm may be preferred.
- memcpy(hNlPref, &hNl[minPrefBand], sizeof(float) * prefBandSize);
- qsort(hNlPref, prefBandSize, sizeof(float), CmpFloat);
- hNlFb = hNlPref[(int)floor(prefBandQuant * (prefBandSize - 1))];
- hNlFbLow = hNlPref[(int)floor(prefBandQuantLow * (prefBandSize - 1))];
- }
- }
-
- // Track the local filter minimum to determine suppression overdrive.
- if (hNlFbLow < 0.6f && hNlFbLow < aec->hNlFbLocalMin) {
- aec->hNlFbLocalMin = hNlFbLow;
- aec->hNlFbMin = hNlFbLow;
- aec->hNlNewMin = 1;
- aec->hNlMinCtr = 0;
- }
- aec->hNlFbLocalMin = WEBRTC_SPL_MIN(aec->hNlFbLocalMin + 0.0008f / aec->mult, 1);
- aec->hNlXdAvgMin = WEBRTC_SPL_MIN(aec->hNlXdAvgMin + 0.0006f / aec->mult, 1);
-
- if (aec->hNlNewMin == 1) {
- aec->hNlMinCtr++;
- }
- if (aec->hNlMinCtr == 2) {
- aec->hNlNewMin = 0;
- aec->hNlMinCtr = 0;
- aec->overDrive = WEBRTC_SPL_MAX(aec->targetSupp /
- ((float)log(aec->hNlFbMin + 1e-10f) + 1e-10f), aec->minOverDrive);
- }
-
- // Smooth the overdrive.
- if (aec->overDrive < aec->overDriveSm) {
- aec->overDriveSm = 0.99f * aec->overDriveSm + 0.01f * aec->overDrive;
- }
- else {
- aec->overDriveSm = 0.9f * aec->overDriveSm + 0.1f * aec->overDrive;
- }
-
- WebRtcAec_OverdriveAndSuppress(aec, hNl, hNlFb, efw);
-
- // Add comfort noise.
- ComfortNoise(aec, efw, comfortNoiseHband, aec->noisePow, hNl);
-
- // TODO(bjornv): Investigate how to take the windowing below into account if
- // needed.
- if (aec->metricsMode == 1) {
- // Note that we have a scaling by two in the time domain |eBuf|.
- // In addition the time domain signal is windowed before transformation,
- // losing half the energy on the average. We take care of the first
- // scaling only in UpdateMetrics().
- UpdateLevel(&aec->nlpoutlevel, efw);
- }
- // Inverse error fft.
- fft[0] = efw[0][0];
- fft[1] = efw[0][PART_LEN];
- for (i = 1; i < PART_LEN; i++) {
- fft[2*i] = efw[0][i];
- // Sign change required by Ooura fft.
- fft[2*i + 1] = -efw[1][i];
- }
- aec_rdft_inverse_128(fft);
-
- // Overlap and add to obtain output.
- scale = 2.0f / PART_LEN2;
- for (i = 0; i < PART_LEN; i++) {
- fft[i] *= scale; // fft scaling
- fft[i] = fft[i]*sqrtHanning[i] + aec->outBuf[i];
-
- // Saturation protection
- output[i] = (short)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, fft[i],
- WEBRTC_SPL_WORD16_MIN);
-
- fft[PART_LEN + i] *= scale; // fft scaling
- aec->outBuf[i] = fft[PART_LEN + i] * sqrtHanning[PART_LEN - i];
- }
-
- // For H band
- if (aec->sampFreq == 32000) {
-
- // H band gain
- // average nlp over low band: average over second half of freq spectrum
- // (4->8khz)
- GetHighbandGain(hNl, &nlpGainHband);
-
- // Inverse comfort_noise
- if (flagHbandCn == 1) {
- fft[0] = comfortNoiseHband[0][0];
- fft[1] = comfortNoiseHband[PART_LEN][0];
- for (i = 1; i < PART_LEN; i++) {
- fft[2*i] = comfortNoiseHband[i][0];
- fft[2*i + 1] = comfortNoiseHband[i][1];
- }
- aec_rdft_inverse_128(fft);
- scale = 2.0f / PART_LEN2;
- }
-
- // compute gain factor
- for (i = 0; i < PART_LEN; i++) {
- dtmp = (float)aec->dBufH[i];
- dtmp = (float)dtmp * nlpGainHband; // for variable gain
-
- // add some comfort noise where Hband is attenuated
- if (flagHbandCn == 1) {
- fft[i] *= scale; // fft scaling
- dtmp += cnScaleHband * fft[i];
- }
-
- // Saturation protection
- outputH[i] = (short)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX, dtmp,
- WEBRTC_SPL_WORD16_MIN);
- }
- }
-
- // Copy the current block to the old position.
- memcpy(aec->dBuf, aec->dBuf + PART_LEN, sizeof(float) * PART_LEN);
- memcpy(aec->eBuf, aec->eBuf + PART_LEN, sizeof(float) * PART_LEN);
-
- // Copy the current block to the old position for H band
- if (aec->sampFreq == 32000) {
- memcpy(aec->dBufH, aec->dBufH + PART_LEN, sizeof(float) * PART_LEN);
- }
-
- memmove(aec->xfwBuf + PART_LEN1, aec->xfwBuf, sizeof(aec->xfwBuf) -
- sizeof(complex_t) * PART_LEN1);
-}
-
-static void GetHighbandGain(const float *lambda, float *nlpGainHband)
-{
- int i;
-
- nlpGainHband[0] = (float)0.0;
- for (i = freqAvgIc; i < PART_LEN1 - 1; i++) {
- nlpGainHband[0] += lambda[i];
- }
- nlpGainHband[0] /= (float)(PART_LEN1 - 1 - freqAvgIc);
-}
-
-static void ComfortNoise(aec_t *aec, float efw[2][PART_LEN1],
- complex_t *comfortNoiseHband, const float *noisePow, const float *lambda)
-{
- int i, num;
- float rand[PART_LEN];
- float noise, noiseAvg, tmp, tmpAvg;
- WebRtc_Word16 randW16[PART_LEN];
- complex_t u[PART_LEN1];
-
- const float pi2 = 6.28318530717959f;
-
- // Generate a uniform random array on [0 1]
- WebRtcSpl_RandUArray(randW16, PART_LEN, &aec->seed);
- for (i = 0; i < PART_LEN; i++) {
- rand[i] = ((float)randW16[i]) / 32768;
- }
-
- // Reject LF noise
- u[0][0] = 0;
- u[0][1] = 0;
- for (i = 1; i < PART_LEN1; i++) {
- tmp = pi2 * rand[i - 1];
-
- noise = sqrtf(noisePow[i]);
- u[i][0] = noise * (float)cos(tmp);
- u[i][1] = -noise * (float)sin(tmp);
- }
- u[PART_LEN][1] = 0;
-
- for (i = 0; i < PART_LEN1; i++) {
- // This is the proper weighting to match the background noise power
- tmp = sqrtf(WEBRTC_SPL_MAX(1 - lambda[i] * lambda[i], 0));
- //tmp = 1 - lambda[i];
- efw[0][i] += tmp * u[i][0];
- efw[1][i] += tmp * u[i][1];
- }
-
- // For H band comfort noise
- // TODO: don't compute noise and "tmp" twice. Use the previous results.
- noiseAvg = 0.0;
- tmpAvg = 0.0;
- num = 0;
- if (aec->sampFreq == 32000 && flagHbandCn == 1) {
-
- // average noise scale
- // average over second half of freq spectrum (i.e., 4->8khz)
- // TODO: we shouldn't need num. We know how many elements we're summing.
- for (i = PART_LEN1 >> 1; i < PART_LEN1; i++) {
- num++;
- noiseAvg += sqrtf(noisePow[i]);
- }
- noiseAvg /= (float)num;
-
- // average nlp scale
- // average over second half of freq spectrum (i.e., 4->8khz)
- // TODO: we shouldn't need num. We know how many elements we're summing.
- num = 0;
- for (i = PART_LEN1 >> 1; i < PART_LEN1; i++) {
- num++;
- tmpAvg += sqrtf(WEBRTC_SPL_MAX(1 - lambda[i] * lambda[i], 0));
- }
- tmpAvg /= (float)num;
-
- // Use average noise for H band
- // TODO: we should probably have a new random vector here.
- // Reject LF noise
- u[0][0] = 0;
- u[0][1] = 0;
- for (i = 1; i < PART_LEN1; i++) {
- tmp = pi2 * rand[i - 1];
-
- // Use average noise for H band
- u[i][0] = noiseAvg * (float)cos(tmp);
- u[i][1] = -noiseAvg * (float)sin(tmp);
- }
- u[PART_LEN][1] = 0;
-
- for (i = 0; i < PART_LEN1; i++) {
- // Use average NLP weight for H band
- comfortNoiseHband[i][0] = tmpAvg * u[i][0];
- comfortNoiseHband[i][1] = tmpAvg * u[i][1];
- }
- }
-}
-
-static void WebRtcAec_InitLevel(power_level_t *level)
-{
- const float bigFloat = 1E17f;
-
- level->averagelevel = 0;
- level->framelevel = 0;
- level->minlevel = bigFloat;
- level->frsum = 0;
- level->sfrsum = 0;
- level->frcounter = 0;
- level->sfrcounter = 0;
-}
-
-static void WebRtcAec_InitStats(stats_t *stats)
-{
- stats->instant = offsetLevel;
- stats->average = offsetLevel;
- stats->max = offsetLevel;
- stats->min = offsetLevel * (-1);
- stats->sum = 0;
- stats->hisum = 0;
- stats->himean = offsetLevel;
- stats->counter = 0;
- stats->hicounter = 0;
-}
-
-static void UpdateLevel(power_level_t* level, float in[2][PART_LEN1]) {
- // Do the energy calculation in the frequency domain. The FFT is performed on
- // a segment of PART_LEN2 samples due to overlap, but we only want the energy
- // of half that data (the last PART_LEN samples). Parseval's relation states
- // that the energy is preserved according to
- //
- // \sum_{n=0}^{N-1} |x(n)|^2 = 1/N * \sum_{n=0}^{N-1} |X(n)|^2
- // = ENERGY,
- //
- // where N = PART_LEN2. Since we are only interested in calculating the energy
- // for the last PART_LEN samples we approximate by calculating ENERGY and
- // divide by 2,
- //
- // \sum_{n=N/2}^{N-1} |x(n)|^2 ~= ENERGY / 2
- //
- // Since we deal with real valued time domain signals we only store frequency
- // bins [0, PART_LEN], which is what |in| consists of. To calculate ENERGY we
- // need to add the contribution from the missing part in
- // [PART_LEN+1, PART_LEN2-1]. These values are, up to a phase shift, identical
- // with the values in [1, PART_LEN-1], hence multiply those values by 2. This
- // is the values in the for loop below, but multiplication by 2 and division
- // by 2 cancel.
-
- // TODO(bjornv): Investigate reusing energy calculations performed at other
- // places in the code.
- int k = 1;
- // Imaginary parts are zero at end points and left out of the calculation.
- float energy = (in[0][0] * in[0][0]) / 2;
- energy += (in[0][PART_LEN] * in[0][PART_LEN]) / 2;
-
- for (k = 1; k < PART_LEN; k++) {
- energy += (in[0][k] * in[0][k] + in[1][k] * in[1][k]);
- }
- energy /= PART_LEN2;
-
- level->sfrsum += energy;
- level->sfrcounter++;
-
- if (level->sfrcounter > subCountLen) {
- level->framelevel = level->sfrsum / (subCountLen * PART_LEN);
- level->sfrsum = 0;
- level->sfrcounter = 0;
- if (level->framelevel > 0) {
- if (level->framelevel < level->minlevel) {
- level->minlevel = level->framelevel; // New minimum.
- } else {
- level->minlevel *= (1 + 0.001f); // Small increase.
- }
- }
- level->frcounter++;
- level->frsum += level->framelevel;
- if (level->frcounter > countLen) {
- level->averagelevel = level->frsum / countLen;
- level->frsum = 0;
- level->frcounter = 0;
- }
- }
-}
-
-static void UpdateMetrics(aec_t *aec)
-{
- float dtmp, dtmp2;
-
- const float actThresholdNoisy = 8.0f;
- const float actThresholdClean = 40.0f;
- const float safety = 0.99995f;
- const float noisyPower = 300000.0f;
-
- float actThreshold;
- float echo, suppressedEcho;
-
- if (aec->echoState) { // Check if echo is likely present
- aec->stateCounter++;
- }
-
- if (aec->farlevel.frcounter == 0) {
-
- if (aec->farlevel.minlevel < noisyPower) {
- actThreshold = actThresholdClean;
- }
- else {
- actThreshold = actThresholdNoisy;
- }
-
- if ((aec->stateCounter > (0.5f * countLen * subCountLen))
- && (aec->farlevel.sfrcounter == 0)
-
- // Estimate in active far-end segments only
- && (aec->farlevel.averagelevel > (actThreshold * aec->farlevel.minlevel))
- ) {
-
- // Subtract noise power
- echo = aec->nearlevel.averagelevel - safety * aec->nearlevel.minlevel;
-
- // ERL
- dtmp = 10 * (float)log10(aec->farlevel.averagelevel /
- aec->nearlevel.averagelevel + 1e-10f);
- dtmp2 = 10 * (float)log10(aec->farlevel.averagelevel / echo + 1e-10f);
-
- aec->erl.instant = dtmp;
- if (dtmp > aec->erl.max) {
- aec->erl.max = dtmp;
- }
-
- if (dtmp < aec->erl.min) {
- aec->erl.min = dtmp;
- }
-
- aec->erl.counter++;
- aec->erl.sum += dtmp;
- aec->erl.average = aec->erl.sum / aec->erl.counter;
-
- // Upper mean
- if (dtmp > aec->erl.average) {
- aec->erl.hicounter++;
- aec->erl.hisum += dtmp;
- aec->erl.himean = aec->erl.hisum / aec->erl.hicounter;
- }
-
- // A_NLP
- dtmp = 10 * (float)log10(aec->nearlevel.averagelevel /
- (2 * aec->linoutlevel.averagelevel) + 1e-10f);
-
- // subtract noise power
- suppressedEcho = 2 * (aec->linoutlevel.averagelevel -
- safety * aec->linoutlevel.minlevel);
-
- dtmp2 = 10 * (float)log10(echo / suppressedEcho + 1e-10f);
-
- aec->aNlp.instant = dtmp2;
- if (dtmp > aec->aNlp.max) {
- aec->aNlp.max = dtmp;
- }
-
- if (dtmp < aec->aNlp.min) {
- aec->aNlp.min = dtmp;
- }
-
- aec->aNlp.counter++;
- aec->aNlp.sum += dtmp;
- aec->aNlp.average = aec->aNlp.sum / aec->aNlp.counter;
-
- // Upper mean
- if (dtmp > aec->aNlp.average) {
- aec->aNlp.hicounter++;
- aec->aNlp.hisum += dtmp;
- aec->aNlp.himean = aec->aNlp.hisum / aec->aNlp.hicounter;
- }
-
- // ERLE
-
- // subtract noise power
- suppressedEcho = 2 * (aec->nlpoutlevel.averagelevel -
- safety * aec->nlpoutlevel.minlevel);
-
- dtmp = 10 * (float)log10(aec->nearlevel.averagelevel /
- (2 * aec->nlpoutlevel.averagelevel) + 1e-10f);
- dtmp2 = 10 * (float)log10(echo / suppressedEcho + 1e-10f);
-
- dtmp = dtmp2;
- aec->erle.instant = dtmp;
- if (dtmp > aec->erle.max) {
- aec->erle.max = dtmp;
- }
-
- if (dtmp < aec->erle.min) {
- aec->erle.min = dtmp;
- }
-
- aec->erle.counter++;
- aec->erle.sum += dtmp;
- aec->erle.average = aec->erle.sum / aec->erle.counter;
-
- // Upper mean
- if (dtmp > aec->erle.average) {
- aec->erle.hicounter++;
- aec->erle.hisum += dtmp;
- aec->erle.himean = aec->erle.hisum / aec->erle.hicounter;
- }
- }
-
- aec->stateCounter = 0;
- }
-}
-
-static void TimeToFrequency(float time_data[PART_LEN2],
- float freq_data[2][PART_LEN1],
- int window) {
- int i = 0;
-
- // TODO(bjornv): Should we have a different function/wrapper for windowed FFT?
- if (window) {
- for (i = 0; i < PART_LEN; i++) {
- time_data[i] *= sqrtHanning[i];
- time_data[PART_LEN + i] *= sqrtHanning[PART_LEN - i];
- }
- }
-
- aec_rdft_forward_128(time_data);
- // Reorder.
- freq_data[1][0] = 0;
- freq_data[1][PART_LEN] = 0;
- freq_data[0][0] = time_data[0];
- freq_data[0][PART_LEN] = time_data[1];
- for (i = 1; i < PART_LEN; i++) {
- freq_data[0][i] = time_data[2 * i];
- freq_data[1][i] = time_data[2 * i + 1];
- }
-}
diff --git a/trunk/src/modules/audio_processing/aec/aec_core.h b/trunk/src/modules/audio_processing/aec/aec_core.h
deleted file mode 100644
index d326a68..0000000
--- a/trunk/src/modules/audio_processing/aec/aec_core.h
+++ /dev/null
@@ -1,180 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Specifies the interface for the AEC core.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-#include <stdio.h>
-#endif
-
-#include "typedefs.h"
-
-#define FRAME_LEN 80
-#define PART_LEN 64 // Length of partition
-#define PART_LEN1 (PART_LEN + 1) // Unique fft coefficients
-#define PART_LEN2 (PART_LEN * 2) // Length of partition * 2
-#define NR_PART 12 // Number of partitions in filter.
-#define PREF_BAND_SIZE 24
-
-// Delay estimator constants, used for logging.
-enum { kMaxDelayBlocks = 60 };
-enum { kLookaheadBlocks = 15 };
-enum { kHistorySizeBlocks = kMaxDelayBlocks + kLookaheadBlocks };
-
-typedef float complex_t[2];
-// For performance reasons, some arrays of complex numbers are replaced by twice
-// as long arrays of float, all the real parts followed by all the imaginary
-// ones (complex_t[SIZE] -> float[2][SIZE]). This allows SIMD optimizations and
-// is better than two arrays (one for the real parts and one for the imaginary
-// parts) as this other way would require two pointers instead of one and cause
-// extra register spilling. This also allows the offsets to be calculated at
-// compile time.
-
-// Metrics
-enum {offsetLevel = -100};
-
-typedef struct {
- float sfrsum;
- int sfrcounter;
- float framelevel;
- float frsum;
- int frcounter;
- float minlevel;
- float averagelevel;
-} power_level_t;
-
-typedef struct {
- float instant;
- float average;
- float min;
- float max;
- float sum;
- float hisum;
- float himean;
- int counter;
- int hicounter;
-} stats_t;
-
-typedef struct {
- int farBufWritePos, farBufReadPos;
-
- int knownDelay;
- int inSamples, outSamples;
- int delayEstCtr;
-
- void *nearFrBuf, *outFrBuf;
-
- void *nearFrBufH;
- void *outFrBufH;
-
- float dBuf[PART_LEN2]; // nearend
- float eBuf[PART_LEN2]; // error
-
- float dBufH[PART_LEN2]; // nearend
-
- float xPow[PART_LEN1];
- float dPow[PART_LEN1];
- float dMinPow[PART_LEN1];
- float dInitMinPow[PART_LEN1];
- float *noisePow;
-
- float xfBuf[2][NR_PART * PART_LEN1]; // farend fft buffer
- float wfBuf[2][NR_PART * PART_LEN1]; // filter fft
- complex_t sde[PART_LEN1]; // cross-psd of nearend and error
- complex_t sxd[PART_LEN1]; // cross-psd of farend and nearend
- complex_t xfwBuf[NR_PART * PART_LEN1]; // farend windowed fft buffer
-
- float sx[PART_LEN1], sd[PART_LEN1], se[PART_LEN1]; // far, near and error psd
- float hNs[PART_LEN1];
- float hNlFbMin, hNlFbLocalMin;
- float hNlXdAvgMin;
- int hNlNewMin, hNlMinCtr;
- float overDrive, overDriveSm;
- float targetSupp, minOverDrive;
- float outBuf[PART_LEN];
- int delayIdx;
-
- short stNearState, echoState;
- short divergeState;
-
- int xfBufBlockPos;
-
- void* far_buf;
- void* far_buf_windowed;
- int system_delay; // Current system delay buffered in AEC.
-
- int mult; // sampling frequency multiple
- int sampFreq;
- WebRtc_UWord32 seed;
-
- float mu; // stepsize
- float errThresh; // error threshold
-
- int noiseEstCtr;
-
- power_level_t farlevel;
- power_level_t nearlevel;
- power_level_t linoutlevel;
- power_level_t nlpoutlevel;
-
- int metricsMode;
- int stateCounter;
- stats_t erl;
- stats_t erle;
- stats_t aNlp;
- stats_t rerl;
-
- // Quantities to control H band scaling for SWB input
- int freq_avg_ic; //initial bin for averaging nlp gain
- int flag_Hband_cn; //for comfort noise
- float cn_scale_Hband; //scale for comfort noise in H band
-
- int delay_histogram[kHistorySizeBlocks];
- int delay_logging_enabled;
- void* delay_estimator;
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- void* far_time_buf;
- FILE *farFile;
- FILE *nearFile;
- FILE *outFile;
- FILE *outLinearFile;
-#endif
-} aec_t;
-
-typedef void (*WebRtcAec_FilterFar_t)(aec_t *aec, float yf[2][PART_LEN1]);
-extern WebRtcAec_FilterFar_t WebRtcAec_FilterFar;
-typedef void (*WebRtcAec_ScaleErrorSignal_t)(aec_t *aec, float ef[2][PART_LEN1]);
-extern WebRtcAec_ScaleErrorSignal_t WebRtcAec_ScaleErrorSignal;
-typedef void (*WebRtcAec_FilterAdaptation_t)
- (aec_t *aec, float *fft, float ef[2][PART_LEN1]);
-extern WebRtcAec_FilterAdaptation_t WebRtcAec_FilterAdaptation;
-typedef void (*WebRtcAec_OverdriveAndSuppress_t)
- (aec_t *aec, float hNl[PART_LEN1], const float hNlFb, float efw[2][PART_LEN1]);
-extern WebRtcAec_OverdriveAndSuppress_t WebRtcAec_OverdriveAndSuppress;
-
-int WebRtcAec_CreateAec(aec_t **aec);
-int WebRtcAec_FreeAec(aec_t *aec);
-int WebRtcAec_InitAec(aec_t *aec, int sampFreq);
-void WebRtcAec_InitAec_SSE2(void);
-
-void WebRtcAec_InitMetrics(aec_t *aec);
-void WebRtcAec_BufferFarendPartition(aec_t *aec, const float* farend);
-void WebRtcAec_ProcessFrame(aec_t* aec,
- const short *nearend,
- const short *nearendH,
- int knownDelay);
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_CORE_H_
diff --git a/trunk/src/modules/audio_processing/aec/aec_core_sse2.c b/trunk/src/modules/audio_processing/aec/aec_core_sse2.c
deleted file mode 100644
index 74a1c48..0000000
--- a/trunk/src/modules/audio_processing/aec/aec_core_sse2.c
+++ /dev/null
@@ -1,415 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * The core AEC algorithm, SSE2 version of speed-critical functions.
- */
-
-#include "aec_core.h"
-
-#include <emmintrin.h>
-#include <math.h>
-#include <string.h> // memset
-
-#include "aec_rdft.h"
-
-__inline static float MulRe(float aRe, float aIm, float bRe, float bIm)
-{
- return aRe * bRe - aIm * bIm;
-}
-
-__inline static float MulIm(float aRe, float aIm, float bRe, float bIm)
-{
- return aRe * bIm + aIm * bRe;
-}
-
-static void FilterFarSSE2(aec_t *aec, float yf[2][PART_LEN1])
-{
- int i;
- for (i = 0; i < NR_PART; i++) {
- int j;
- int xPos = (i + aec->xfBufBlockPos) * PART_LEN1;
- int pos = i * PART_LEN1;
- // Check for wrap
- if (i + aec->xfBufBlockPos >= NR_PART) {
- xPos -= NR_PART*(PART_LEN1);
- }
-
- // vectorized code (four at once)
- for (j = 0; j + 3 < PART_LEN1; j += 4) {
- const __m128 xfBuf_re = _mm_loadu_ps(&aec->xfBuf[0][xPos + j]);
- const __m128 xfBuf_im = _mm_loadu_ps(&aec->xfBuf[1][xPos + j]);
- const __m128 wfBuf_re = _mm_loadu_ps(&aec->wfBuf[0][pos + j]);
- const __m128 wfBuf_im = _mm_loadu_ps(&aec->wfBuf[1][pos + j]);
- const __m128 yf_re = _mm_loadu_ps(&yf[0][j]);
- const __m128 yf_im = _mm_loadu_ps(&yf[1][j]);
- const __m128 a = _mm_mul_ps(xfBuf_re, wfBuf_re);
- const __m128 b = _mm_mul_ps(xfBuf_im, wfBuf_im);
- const __m128 c = _mm_mul_ps(xfBuf_re, wfBuf_im);
- const __m128 d = _mm_mul_ps(xfBuf_im, wfBuf_re);
- const __m128 e = _mm_sub_ps(a, b);
- const __m128 f = _mm_add_ps(c, d);
- const __m128 g = _mm_add_ps(yf_re, e);
- const __m128 h = _mm_add_ps(yf_im, f);
- _mm_storeu_ps(&yf[0][j], g);
- _mm_storeu_ps(&yf[1][j], h);
- }
- // scalar code for the remaining items.
- for (; j < PART_LEN1; j++) {
- yf[0][j] += MulRe(aec->xfBuf[0][xPos + j], aec->xfBuf[1][xPos + j],
- aec->wfBuf[0][ pos + j], aec->wfBuf[1][ pos + j]);
- yf[1][j] += MulIm(aec->xfBuf[0][xPos + j], aec->xfBuf[1][xPos + j],
- aec->wfBuf[0][ pos + j], aec->wfBuf[1][ pos + j]);
- }
- }
-}
-
-static void ScaleErrorSignalSSE2(aec_t *aec, float ef[2][PART_LEN1])
-{
- const __m128 k1e_10f = _mm_set1_ps(1e-10f);
- const __m128 kThresh = _mm_set1_ps(aec->errThresh);
- const __m128 kMu = _mm_set1_ps(aec->mu);
-
- int i;
- // vectorized code (four at once)
- for (i = 0; i + 3 < PART_LEN1; i += 4) {
- const __m128 xPow = _mm_loadu_ps(&aec->xPow[i]);
- const __m128 ef_re_base = _mm_loadu_ps(&ef[0][i]);
- const __m128 ef_im_base = _mm_loadu_ps(&ef[1][i]);
-
- const __m128 xPowPlus = _mm_add_ps(xPow, k1e_10f);
- __m128 ef_re = _mm_div_ps(ef_re_base, xPowPlus);
- __m128 ef_im = _mm_div_ps(ef_im_base, xPowPlus);
- const __m128 ef_re2 = _mm_mul_ps(ef_re, ef_re);
- const __m128 ef_im2 = _mm_mul_ps(ef_im, ef_im);
- const __m128 ef_sum2 = _mm_add_ps(ef_re2, ef_im2);
- const __m128 absEf = _mm_sqrt_ps(ef_sum2);
- const __m128 bigger = _mm_cmpgt_ps(absEf, kThresh);
- __m128 absEfPlus = _mm_add_ps(absEf, k1e_10f);
- const __m128 absEfInv = _mm_div_ps(kThresh, absEfPlus);
- __m128 ef_re_if = _mm_mul_ps(ef_re, absEfInv);
- __m128 ef_im_if = _mm_mul_ps(ef_im, absEfInv);
- ef_re_if = _mm_and_ps(bigger, ef_re_if);
- ef_im_if = _mm_and_ps(bigger, ef_im_if);
- ef_re = _mm_andnot_ps(bigger, ef_re);
- ef_im = _mm_andnot_ps(bigger, ef_im);
- ef_re = _mm_or_ps(ef_re, ef_re_if);
- ef_im = _mm_or_ps(ef_im, ef_im_if);
- ef_re = _mm_mul_ps(ef_re, kMu);
- ef_im = _mm_mul_ps(ef_im, kMu);
-
- _mm_storeu_ps(&ef[0][i], ef_re);
- _mm_storeu_ps(&ef[1][i], ef_im);
- }
- // scalar code for the remaining items.
- for (; i < (PART_LEN1); i++) {
- float absEf;
- ef[0][i] /= (aec->xPow[i] + 1e-10f);
- ef[1][i] /= (aec->xPow[i] + 1e-10f);
- absEf = sqrtf(ef[0][i] * ef[0][i] + ef[1][i] * ef[1][i]);
-
- if (absEf > aec->errThresh) {
- absEf = aec->errThresh / (absEf + 1e-10f);
- ef[0][i] *= absEf;
- ef[1][i] *= absEf;
- }
-
- // Stepsize factor
- ef[0][i] *= aec->mu;
- ef[1][i] *= aec->mu;
- }
-}
-
-static void FilterAdaptationSSE2(aec_t *aec, float *fft, float ef[2][PART_LEN1]) {
- int i, j;
- for (i = 0; i < NR_PART; i++) {
- int xPos = (i + aec->xfBufBlockPos)*(PART_LEN1);
- int pos = i * PART_LEN1;
- // Check for wrap
- if (i + aec->xfBufBlockPos >= NR_PART) {
- xPos -= NR_PART * PART_LEN1;
- }
-
- // Process the whole array...
- for (j = 0; j < PART_LEN; j+= 4) {
- // Load xfBuf and ef.
- const __m128 xfBuf_re = _mm_loadu_ps(&aec->xfBuf[0][xPos + j]);
- const __m128 xfBuf_im = _mm_loadu_ps(&aec->xfBuf[1][xPos + j]);
- const __m128 ef_re = _mm_loadu_ps(&ef[0][j]);
- const __m128 ef_im = _mm_loadu_ps(&ef[1][j]);
- // Calculate the product of conjugate(xfBuf) by ef.
- // re(conjugate(a) * b) = aRe * bRe + aIm * bIm
- // im(conjugate(a) * b)= aRe * bIm - aIm * bRe
- const __m128 a = _mm_mul_ps(xfBuf_re, ef_re);
- const __m128 b = _mm_mul_ps(xfBuf_im, ef_im);
- const __m128 c = _mm_mul_ps(xfBuf_re, ef_im);
- const __m128 d = _mm_mul_ps(xfBuf_im, ef_re);
- const __m128 e = _mm_add_ps(a, b);
- const __m128 f = _mm_sub_ps(c, d);
- // Interleave real and imaginary parts.
- const __m128 g = _mm_unpacklo_ps(e, f);
- const __m128 h = _mm_unpackhi_ps(e, f);
- // Store
- _mm_storeu_ps(&fft[2*j + 0], g);
- _mm_storeu_ps(&fft[2*j + 4], h);
- }
- // ... and fixup the first imaginary entry.
- fft[1] = MulRe(aec->xfBuf[0][xPos + PART_LEN],
- -aec->xfBuf[1][xPos + PART_LEN],
- ef[0][PART_LEN], ef[1][PART_LEN]);
-
- aec_rdft_inverse_128(fft);
- memset(fft + PART_LEN, 0, sizeof(float)*PART_LEN);
-
- // fft scaling
- {
- float scale = 2.0f / PART_LEN2;
- const __m128 scale_ps = _mm_load_ps1(&scale);
- for (j = 0; j < PART_LEN; j+=4) {
- const __m128 fft_ps = _mm_loadu_ps(&fft[j]);
- const __m128 fft_scale = _mm_mul_ps(fft_ps, scale_ps);
- _mm_storeu_ps(&fft[j], fft_scale);
- }
- }
- aec_rdft_forward_128(fft);
-
- {
- float wt1 = aec->wfBuf[1][pos];
- aec->wfBuf[0][pos + PART_LEN] += fft[1];
- for (j = 0; j < PART_LEN; j+= 4) {
- __m128 wtBuf_re = _mm_loadu_ps(&aec->wfBuf[0][pos + j]);
- __m128 wtBuf_im = _mm_loadu_ps(&aec->wfBuf[1][pos + j]);
- const __m128 fft0 = _mm_loadu_ps(&fft[2 * j + 0]);
- const __m128 fft4 = _mm_loadu_ps(&fft[2 * j + 4]);
- const __m128 fft_re = _mm_shuffle_ps(fft0, fft4, _MM_SHUFFLE(2, 0, 2 ,0));
- const __m128 fft_im = _mm_shuffle_ps(fft0, fft4, _MM_SHUFFLE(3, 1, 3 ,1));
- wtBuf_re = _mm_add_ps(wtBuf_re, fft_re);
- wtBuf_im = _mm_add_ps(wtBuf_im, fft_im);
- _mm_storeu_ps(&aec->wfBuf[0][pos + j], wtBuf_re);
- _mm_storeu_ps(&aec->wfBuf[1][pos + j], wtBuf_im);
- }
- aec->wfBuf[1][pos] = wt1;
- }
- }
-}
-
-static __m128 mm_pow_ps(__m128 a, __m128 b)
-{
- // a^b = exp2(b * log2(a))
- // exp2(x) and log2(x) are calculated using polynomial approximations.
- __m128 log2_a, b_log2_a, a_exp_b;
-
- // Calculate log2(x), x = a.
- {
- // To calculate log2(x), we decompose x like this:
- // x = y * 2^n
- // n is an integer
- // y is in the [1.0, 2.0) range
- //
- // log2(x) = log2(y) + n
- // n can be evaluated by playing with float representation.
- // log2(y) in a small range can be approximated, this code uses an order
- // five polynomial approximation. The coefficients have been
- // estimated with the Remez algorithm and the resulting
- // polynomial has a maximum relative error of 0.00086%.
-
- // Compute n.
- // This is done by masking the exponent, shifting it into the top bit of
- // the mantissa, putting eight into the biased exponent (to shift/
- // compensate the fact that the exponent has been shifted in the top/
- // fractional part and finally getting rid of the implicit leading one
- // from the mantissa by substracting it out.
- static const ALIGN16_BEG int float_exponent_mask[4] ALIGN16_END =
- {0x7F800000, 0x7F800000, 0x7F800000, 0x7F800000};
- static const ALIGN16_BEG int eight_biased_exponent[4] ALIGN16_END =
- {0x43800000, 0x43800000, 0x43800000, 0x43800000};
- static const ALIGN16_BEG int implicit_leading_one[4] ALIGN16_END =
- {0x43BF8000, 0x43BF8000, 0x43BF8000, 0x43BF8000};
- static const int shift_exponent_into_top_mantissa = 8;
- const __m128 two_n = _mm_and_ps(a, *((__m128 *)float_exponent_mask));
- const __m128 n_1 = _mm_castsi128_ps(_mm_srli_epi32(_mm_castps_si128(two_n),
- shift_exponent_into_top_mantissa));
- const __m128 n_0 = _mm_or_ps(n_1, *((__m128 *)eight_biased_exponent));
- const __m128 n = _mm_sub_ps(n_0, *((__m128 *)implicit_leading_one));
-
- // Compute y.
- static const ALIGN16_BEG int mantissa_mask[4] ALIGN16_END =
- {0x007FFFFF, 0x007FFFFF, 0x007FFFFF, 0x007FFFFF};
- static const ALIGN16_BEG int zero_biased_exponent_is_one[4] ALIGN16_END =
- {0x3F800000, 0x3F800000, 0x3F800000, 0x3F800000};
- const __m128 mantissa = _mm_and_ps(a, *((__m128 *)mantissa_mask));
- const __m128 y = _mm_or_ps(
- mantissa, *((__m128 *)zero_biased_exponent_is_one));
-
- // Approximate log2(y) ~= (y - 1) * pol5(y).
- // pol5(y) = C5 * y^5 + C4 * y^4 + C3 * y^3 + C2 * y^2 + C1 * y + C0
- static const ALIGN16_BEG float ALIGN16_END C5[4] =
- {-3.4436006e-2f, -3.4436006e-2f, -3.4436006e-2f, -3.4436006e-2f};
- static const ALIGN16_BEG float ALIGN16_END C4[4] =
- {3.1821337e-1f, 3.1821337e-1f, 3.1821337e-1f, 3.1821337e-1f};
- static const ALIGN16_BEG float ALIGN16_END C3[4] =
- {-1.2315303f, -1.2315303f, -1.2315303f, -1.2315303f};
- static const ALIGN16_BEG float ALIGN16_END C2[4] =
- {2.5988452f, 2.5988452f, 2.5988452f, 2.5988452f};
- static const ALIGN16_BEG float ALIGN16_END C1[4] =
- {-3.3241990f, -3.3241990f, -3.3241990f, -3.3241990f};
- static const ALIGN16_BEG float ALIGN16_END C0[4] =
- {3.1157899f, 3.1157899f, 3.1157899f, 3.1157899f};
- const __m128 pol5_y_0 = _mm_mul_ps(y, *((__m128 *)C5));
- const __m128 pol5_y_1 = _mm_add_ps(pol5_y_0, *((__m128 *)C4));
- const __m128 pol5_y_2 = _mm_mul_ps(pol5_y_1, y);
- const __m128 pol5_y_3 = _mm_add_ps(pol5_y_2, *((__m128 *)C3));
- const __m128 pol5_y_4 = _mm_mul_ps(pol5_y_3, y);
- const __m128 pol5_y_5 = _mm_add_ps(pol5_y_4, *((__m128 *)C2));
- const __m128 pol5_y_6 = _mm_mul_ps(pol5_y_5, y);
- const __m128 pol5_y_7 = _mm_add_ps(pol5_y_6, *((__m128 *)C1));
- const __m128 pol5_y_8 = _mm_mul_ps(pol5_y_7, y);
- const __m128 pol5_y = _mm_add_ps(pol5_y_8, *((__m128 *)C0));
- const __m128 y_minus_one = _mm_sub_ps(
- y, *((__m128 *)zero_biased_exponent_is_one));
- const __m128 log2_y = _mm_mul_ps(y_minus_one , pol5_y);
-
- // Combine parts.
- log2_a = _mm_add_ps(n, log2_y);
- }
-
- // b * log2(a)
- b_log2_a = _mm_mul_ps(b, log2_a);
-
- // Calculate exp2(x), x = b * log2(a).
- {
- // To calculate 2^x, we decompose x like this:
- // x = n + y
- // n is an integer, the value of x - 0.5 rounded down, therefore
- // y is in the [0.5, 1.5) range
- //
- // 2^x = 2^n * 2^y
- // 2^n can be evaluated by playing with float representation.
- // 2^y in a small range can be approximated, this code uses an order two
- // polynomial approximation. The coefficients have been estimated
- // with the Remez algorithm and the resulting polynomial has a
- // maximum relative error of 0.17%.
-
- // To avoid over/underflow, we reduce the range of input to ]-127, 129].
- static const ALIGN16_BEG float max_input[4] ALIGN16_END =
- {129.f, 129.f, 129.f, 129.f};
- static const ALIGN16_BEG float min_input[4] ALIGN16_END =
- {-126.99999f, -126.99999f, -126.99999f, -126.99999f};
- const __m128 x_min = _mm_min_ps(b_log2_a, *((__m128 *)max_input));
- const __m128 x_max = _mm_max_ps(x_min, *((__m128 *)min_input));
- // Compute n.
- static const ALIGN16_BEG float half[4] ALIGN16_END =
- {0.5f, 0.5f, 0.5f, 0.5f};
- const __m128 x_minus_half = _mm_sub_ps(x_max, *((__m128 *)half));
- const __m128i x_minus_half_floor = _mm_cvtps_epi32(x_minus_half);
- // Compute 2^n.
- static const ALIGN16_BEG int float_exponent_bias[4] ALIGN16_END =
- {127, 127, 127, 127};
- static const int float_exponent_shift = 23;
- const __m128i two_n_exponent = _mm_add_epi32(
- x_minus_half_floor, *((__m128i *)float_exponent_bias));
- const __m128 two_n = _mm_castsi128_ps(_mm_slli_epi32(
- two_n_exponent, float_exponent_shift));
- // Compute y.
- const __m128 y = _mm_sub_ps(x_max, _mm_cvtepi32_ps(x_minus_half_floor));
- // Approximate 2^y ~= C2 * y^2 + C1 * y + C0.
- static const ALIGN16_BEG float C2[4] ALIGN16_END =
- {3.3718944e-1f, 3.3718944e-1f, 3.3718944e-1f, 3.3718944e-1f};
- static const ALIGN16_BEG float C1[4] ALIGN16_END =
- {6.5763628e-1f, 6.5763628e-1f, 6.5763628e-1f, 6.5763628e-1f};
- static const ALIGN16_BEG float C0[4] ALIGN16_END =
- {1.0017247f, 1.0017247f, 1.0017247f, 1.0017247f};
- const __m128 exp2_y_0 = _mm_mul_ps(y, *((__m128 *)C2));
- const __m128 exp2_y_1 = _mm_add_ps(exp2_y_0, *((__m128 *)C1));
- const __m128 exp2_y_2 = _mm_mul_ps(exp2_y_1, y);
- const __m128 exp2_y = _mm_add_ps(exp2_y_2, *((__m128 *)C0));
-
- // Combine parts.
- a_exp_b = _mm_mul_ps(exp2_y, two_n);
- }
- return a_exp_b;
-}
-
-extern const float WebRtcAec_weightCurve[65];
-extern const float WebRtcAec_overDriveCurve[65];
-
-static void OverdriveAndSuppressSSE2(aec_t *aec, float hNl[PART_LEN1],
- const float hNlFb,
- float efw[2][PART_LEN1]) {
- int i;
- const __m128 vec_hNlFb = _mm_set1_ps(hNlFb);
- const __m128 vec_one = _mm_set1_ps(1.0f);
- const __m128 vec_minus_one = _mm_set1_ps(-1.0f);
- const __m128 vec_overDriveSm = _mm_set1_ps(aec->overDriveSm);
- // vectorized code (four at once)
- for (i = 0; i + 3 < PART_LEN1; i+=4) {
- // Weight subbands
- __m128 vec_hNl = _mm_loadu_ps(&hNl[i]);
- const __m128 vec_weightCurve = _mm_loadu_ps(&WebRtcAec_weightCurve[i]);
- const __m128 bigger = _mm_cmpgt_ps(vec_hNl, vec_hNlFb);
- const __m128 vec_weightCurve_hNlFb = _mm_mul_ps(
- vec_weightCurve, vec_hNlFb);
- const __m128 vec_one_weightCurve = _mm_sub_ps(vec_one, vec_weightCurve);
- const __m128 vec_one_weightCurve_hNl = _mm_mul_ps(
- vec_one_weightCurve, vec_hNl);
- const __m128 vec_if0 = _mm_andnot_ps(bigger, vec_hNl);
- const __m128 vec_if1 = _mm_and_ps(
- bigger, _mm_add_ps(vec_weightCurve_hNlFb, vec_one_weightCurve_hNl));
- vec_hNl = _mm_or_ps(vec_if0, vec_if1);
-
- {
- const __m128 vec_overDriveCurve = _mm_loadu_ps(
- &WebRtcAec_overDriveCurve[i]);
- const __m128 vec_overDriveSm_overDriveCurve = _mm_mul_ps(
- vec_overDriveSm, vec_overDriveCurve);
- vec_hNl = mm_pow_ps(vec_hNl, vec_overDriveSm_overDriveCurve);
- _mm_storeu_ps(&hNl[i], vec_hNl);
- }
-
- // Suppress error signal
- {
- __m128 vec_efw_re = _mm_loadu_ps(&efw[0][i]);
- __m128 vec_efw_im = _mm_loadu_ps(&efw[1][i]);
- vec_efw_re = _mm_mul_ps(vec_efw_re, vec_hNl);
- vec_efw_im = _mm_mul_ps(vec_efw_im, vec_hNl);
-
- // Ooura fft returns incorrect sign on imaginary component. It matters
- // here because we are making an additive change with comfort noise.
- vec_efw_im = _mm_mul_ps(vec_efw_im, vec_minus_one);
- _mm_storeu_ps(&efw[0][i], vec_efw_re);
- _mm_storeu_ps(&efw[1][i], vec_efw_im);
- }
- }
- // scalar code for the remaining items.
- for (; i < PART_LEN1; i++) {
- // Weight subbands
- if (hNl[i] > hNlFb) {
- hNl[i] = WebRtcAec_weightCurve[i] * hNlFb +
- (1 - WebRtcAec_weightCurve[i]) * hNl[i];
- }
- hNl[i] = powf(hNl[i], aec->overDriveSm * WebRtcAec_overDriveCurve[i]);
-
- // Suppress error signal
- efw[0][i] *= hNl[i];
- efw[1][i] *= hNl[i];
-
- // Ooura fft returns incorrect sign on imaginary component. It matters
- // here because we are making an additive change with comfort noise.
- efw[1][i] *= -1;
- }
-}
-
-void WebRtcAec_InitAec_SSE2(void) {
- WebRtcAec_FilterFar = FilterFarSSE2;
- WebRtcAec_ScaleErrorSignal = ScaleErrorSignalSSE2;
- WebRtcAec_FilterAdaptation = FilterAdaptationSSE2;
- WebRtcAec_OverdriveAndSuppress = OverdriveAndSuppressSSE2;
-}
-
diff --git a/trunk/src/modules/audio_processing/aec/aec_rdft.c b/trunk/src/modules/audio_processing/aec/aec_rdft.c
deleted file mode 100644
index 19908d8..0000000
--- a/trunk/src/modules/audio_processing/aec/aec_rdft.c
+++ /dev/null
@@ -1,587 +0,0 @@
-/*
- * http://www.kurims.kyoto-u.ac.jp/~ooura/fft.html
- * Copyright Takuya OOURA, 1996-2001
- *
- * You may use, copy, modify and distribute this code for any purpose (include
- * commercial use) and without fee. Please refer to this package when you modify
- * this code.
- *
- * Changes by the WebRTC authors:
- * - Trivial type modifications.
- * - Minimal code subset to do rdft of length 128.
- * - Optimizations because of known length.
- *
- * All changes are covered by the WebRTC license and IP grant:
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "aec_rdft.h"
-
-#include <math.h>
-
-#include "system_wrappers/interface/cpu_features_wrapper.h"
-#include "typedefs.h"
-
-// constants shared by all paths (C, SSE2).
-float rdft_w[64];
-// constants used by the C path.
-float rdft_wk3ri_first[32];
-float rdft_wk3ri_second[32];
-// constants used by SSE2 but initialized in C path.
-ALIGN16_BEG float ALIGN16_END rdft_wk1r[32];
-ALIGN16_BEG float ALIGN16_END rdft_wk2r[32];
-ALIGN16_BEG float ALIGN16_END rdft_wk3r[32];
-ALIGN16_BEG float ALIGN16_END rdft_wk1i[32];
-ALIGN16_BEG float ALIGN16_END rdft_wk2i[32];
-ALIGN16_BEG float ALIGN16_END rdft_wk3i[32];
-ALIGN16_BEG float ALIGN16_END cftmdl_wk1r[4];
-
-static int ip[16];
-
-static void bitrv2_32or128(int n, int *ip, float *a) {
- // n is 32 or 128
- int j, j1, k, k1, m, m2;
- float xr, xi, yr, yi;
-
- ip[0] = 0;
- {
- int l = n;
- m = 1;
- while ((m << 3) < l) {
- l >>= 1;
- for (j = 0; j < m; j++) {
- ip[m + j] = ip[j] + l;
- }
- m <<= 1;
- }
- }
- m2 = 2 * m;
- for (k = 0; k < m; k++) {
- for (j = 0; j < k; j++) {
- j1 = 2 * j + ip[k];
- k1 = 2 * k + ip[j];
- xr = a[j1];
- xi = a[j1 + 1];
- yr = a[k1];
- yi = a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- j1 += m2;
- k1 += 2 * m2;
- xr = a[j1];
- xi = a[j1 + 1];
- yr = a[k1];
- yi = a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- j1 += m2;
- k1 -= m2;
- xr = a[j1];
- xi = a[j1 + 1];
- yr = a[k1];
- yi = a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- j1 += m2;
- k1 += 2 * m2;
- xr = a[j1];
- xi = a[j1 + 1];
- yr = a[k1];
- yi = a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- }
- j1 = 2 * k + m2 + ip[k];
- k1 = j1 + m2;
- xr = a[j1];
- xi = a[j1 + 1];
- yr = a[k1];
- yi = a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- }
-}
-
-static void makewt_32(void) {
- const int nw = 32;
- int j, nwh;
- float delta, x, y;
-
- ip[0] = nw;
- ip[1] = 1;
- nwh = nw >> 1;
- delta = atanf(1.0f) / nwh;
- rdft_w[0] = 1;
- rdft_w[1] = 0;
- rdft_w[nwh] = cosf(delta * nwh);
- rdft_w[nwh + 1] = rdft_w[nwh];
- for (j = 2; j < nwh; j += 2) {
- x = cosf(delta * j);
- y = sinf(delta * j);
- rdft_w[j] = x;
- rdft_w[j + 1] = y;
- rdft_w[nw - j] = y;
- rdft_w[nw - j + 1] = x;
- }
- bitrv2_32or128(nw, ip + 2, rdft_w);
-
- // pre-calculate constants used by cft1st_128 and cftmdl_128...
- cftmdl_wk1r[0] = rdft_w[2];
- cftmdl_wk1r[1] = rdft_w[2];
- cftmdl_wk1r[2] = rdft_w[2];
- cftmdl_wk1r[3] = -rdft_w[2];
- {
- int k1;
-
- for (k1 = 0, j = 0; j < 128; j += 16, k1 += 2) {
- const int k2 = 2 * k1;
- const float wk2r = rdft_w[k1 + 0];
- const float wk2i = rdft_w[k1 + 1];
- float wk1r, wk1i;
- // ... scalar version.
- wk1r = rdft_w[k2 + 0];
- wk1i = rdft_w[k2 + 1];
- rdft_wk3ri_first[k1 + 0] = wk1r - 2 * wk2i * wk1i;
- rdft_wk3ri_first[k1 + 1] = 2 * wk2i * wk1r - wk1i;
- wk1r = rdft_w[k2 + 2];
- wk1i = rdft_w[k2 + 3];
- rdft_wk3ri_second[k1 + 0] = wk1r - 2 * wk2r * wk1i;
- rdft_wk3ri_second[k1 + 1] = 2 * wk2r * wk1r - wk1i;
- // ... vector version.
- rdft_wk1r[k2 + 0] = rdft_w[k2 + 0];
- rdft_wk1r[k2 + 1] = rdft_w[k2 + 0];
- rdft_wk1r[k2 + 2] = rdft_w[k2 + 2];
- rdft_wk1r[k2 + 3] = rdft_w[k2 + 2];
- rdft_wk2r[k2 + 0] = rdft_w[k1 + 0];
- rdft_wk2r[k2 + 1] = rdft_w[k1 + 0];
- rdft_wk2r[k2 + 2] = -rdft_w[k1 + 1];
- rdft_wk2r[k2 + 3] = -rdft_w[k1 + 1];
- rdft_wk3r[k2 + 0] = rdft_wk3ri_first[k1 + 0];
- rdft_wk3r[k2 + 1] = rdft_wk3ri_first[k1 + 0];
- rdft_wk3r[k2 + 2] = rdft_wk3ri_second[k1 + 0];
- rdft_wk3r[k2 + 3] = rdft_wk3ri_second[k1 + 0];
- rdft_wk1i[k2 + 0] = -rdft_w[k2 + 1];
- rdft_wk1i[k2 + 1] = rdft_w[k2 + 1];
- rdft_wk1i[k2 + 2] = -rdft_w[k2 + 3];
- rdft_wk1i[k2 + 3] = rdft_w[k2 + 3];
- rdft_wk2i[k2 + 0] = -rdft_w[k1 + 1];
- rdft_wk2i[k2 + 1] = rdft_w[k1 + 1];
- rdft_wk2i[k2 + 2] = -rdft_w[k1 + 0];
- rdft_wk2i[k2 + 3] = rdft_w[k1 + 0];
- rdft_wk3i[k2 + 0] = -rdft_wk3ri_first[k1 + 1];
- rdft_wk3i[k2 + 1] = rdft_wk3ri_first[k1 + 1];
- rdft_wk3i[k2 + 2] = -rdft_wk3ri_second[k1 + 1];
- rdft_wk3i[k2 + 3] = rdft_wk3ri_second[k1 + 1];
- }
- }
-}
-
-static void makect_32(void) {
- float *c = rdft_w + 32;
- const int nc = 32;
- int j, nch;
- float delta;
-
- ip[1] = nc;
- nch = nc >> 1;
- delta = atanf(1.0f) / nch;
- c[0] = cosf(delta * nch);
- c[nch] = 0.5f * c[0];
- for (j = 1; j < nch; j++) {
- c[j] = 0.5f * cosf(delta * j);
- c[nc - j] = 0.5f * sinf(delta * j);
- }
-}
-
-static void cft1st_128_C(float *a) {
- const int n = 128;
- int j, k1, k2;
- float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
- float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
-
- x0r = a[0] + a[2];
- x0i = a[1] + a[3];
- x1r = a[0] - a[2];
- x1i = a[1] - a[3];
- x2r = a[4] + a[6];
- x2i = a[5] + a[7];
- x3r = a[4] - a[6];
- x3i = a[5] - a[7];
- a[0] = x0r + x2r;
- a[1] = x0i + x2i;
- a[4] = x0r - x2r;
- a[5] = x0i - x2i;
- a[2] = x1r - x3i;
- a[3] = x1i + x3r;
- a[6] = x1r + x3i;
- a[7] = x1i - x3r;
- wk1r = rdft_w[2];
- x0r = a[8] + a[10];
- x0i = a[9] + a[11];
- x1r = a[8] - a[10];
- x1i = a[9] - a[11];
- x2r = a[12] + a[14];
- x2i = a[13] + a[15];
- x3r = a[12] - a[14];
- x3i = a[13] - a[15];
- a[8] = x0r + x2r;
- a[9] = x0i + x2i;
- a[12] = x2i - x0i;
- a[13] = x0r - x2r;
- x0r = x1r - x3i;
- x0i = x1i + x3r;
- a[10] = wk1r * (x0r - x0i);
- a[11] = wk1r * (x0r + x0i);
- x0r = x3i + x1r;
- x0i = x3r - x1i;
- a[14] = wk1r * (x0i - x0r);
- a[15] = wk1r * (x0i + x0r);
- k1 = 0;
- for (j = 16; j < n; j += 16) {
- k1 += 2;
- k2 = 2 * k1;
- wk2r = rdft_w[k1 + 0];
- wk2i = rdft_w[k1 + 1];
- wk1r = rdft_w[k2 + 0];
- wk1i = rdft_w[k2 + 1];
- wk3r = rdft_wk3ri_first[k1 + 0];
- wk3i = rdft_wk3ri_first[k1 + 1];
- x0r = a[j + 0] + a[j + 2];
- x0i = a[j + 1] + a[j + 3];
- x1r = a[j + 0] - a[j + 2];
- x1i = a[j + 1] - a[j + 3];
- x2r = a[j + 4] + a[j + 6];
- x2i = a[j + 5] + a[j + 7];
- x3r = a[j + 4] - a[j + 6];
- x3i = a[j + 5] - a[j + 7];
- a[j + 0] = x0r + x2r;
- a[j + 1] = x0i + x2i;
- x0r -= x2r;
- x0i -= x2i;
- a[j + 4] = wk2r * x0r - wk2i * x0i;
- a[j + 5] = wk2r * x0i + wk2i * x0r;
- x0r = x1r - x3i;
- x0i = x1i + x3r;
- a[j + 2] = wk1r * x0r - wk1i * x0i;
- a[j + 3] = wk1r * x0i + wk1i * x0r;
- x0r = x1r + x3i;
- x0i = x1i - x3r;
- a[j + 6] = wk3r * x0r - wk3i * x0i;
- a[j + 7] = wk3r * x0i + wk3i * x0r;
- wk1r = rdft_w[k2 + 2];
- wk1i = rdft_w[k2 + 3];
- wk3r = rdft_wk3ri_second[k1 + 0];
- wk3i = rdft_wk3ri_second[k1 + 1];
- x0r = a[j + 8] + a[j + 10];
- x0i = a[j + 9] + a[j + 11];
- x1r = a[j + 8] - a[j + 10];
- x1i = a[j + 9] - a[j + 11];
- x2r = a[j + 12] + a[j + 14];
- x2i = a[j + 13] + a[j + 15];
- x3r = a[j + 12] - a[j + 14];
- x3i = a[j + 13] - a[j + 15];
- a[j + 8] = x0r + x2r;
- a[j + 9] = x0i + x2i;
- x0r -= x2r;
- x0i -= x2i;
- a[j + 12] = -wk2i * x0r - wk2r * x0i;
- a[j + 13] = -wk2i * x0i + wk2r * x0r;
- x0r = x1r - x3i;
- x0i = x1i + x3r;
- a[j + 10] = wk1r * x0r - wk1i * x0i;
- a[j + 11] = wk1r * x0i + wk1i * x0r;
- x0r = x1r + x3i;
- x0i = x1i - x3r;
- a[j + 14] = wk3r * x0r - wk3i * x0i;
- a[j + 15] = wk3r * x0i + wk3i * x0r;
- }
-}
-
-static void cftmdl_128_C(float *a) {
- const int l = 8;
- const int n = 128;
- const int m = 32;
- int j0, j1, j2, j3, k, k1, k2, m2;
- float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
- float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
-
- for (j0 = 0; j0 < l; j0 += 2) {
- j1 = j0 + 8;
- j2 = j0 + 16;
- j3 = j0 + 24;
- x0r = a[j0 + 0] + a[j1 + 0];
- x0i = a[j0 + 1] + a[j1 + 1];
- x1r = a[j0 + 0] - a[j1 + 0];
- x1i = a[j0 + 1] - a[j1 + 1];
- x2r = a[j2 + 0] + a[j3 + 0];
- x2i = a[j2 + 1] + a[j3 + 1];
- x3r = a[j2 + 0] - a[j3 + 0];
- x3i = a[j2 + 1] - a[j3 + 1];
- a[j0 + 0] = x0r + x2r;
- a[j0 + 1] = x0i + x2i;
- a[j2 + 0] = x0r - x2r;
- a[j2 + 1] = x0i - x2i;
- a[j1 + 0] = x1r - x3i;
- a[j1 + 1] = x1i + x3r;
- a[j3 + 0] = x1r + x3i;
- a[j3 + 1] = x1i - x3r;
- }
- wk1r = rdft_w[2];
- for (j0 = m; j0 < l + m; j0 += 2) {
- j1 = j0 + 8;
- j2 = j0 + 16;
- j3 = j0 + 24;
- x0r = a[j0 + 0] + a[j1 + 0];
- x0i = a[j0 + 1] + a[j1 + 1];
- x1r = a[j0 + 0] - a[j1 + 0];
- x1i = a[j0 + 1] - a[j1 + 1];
- x2r = a[j2 + 0] + a[j3 + 0];
- x2i = a[j2 + 1] + a[j3 + 1];
- x3r = a[j2 + 0] - a[j3 + 0];
- x3i = a[j2 + 1] - a[j3 + 1];
- a[j0 + 0] = x0r + x2r;
- a[j0 + 1] = x0i + x2i;
- a[j2 + 0] = x2i - x0i;
- a[j2 + 1] = x0r - x2r;
- x0r = x1r - x3i;
- x0i = x1i + x3r;
- a[j1 + 0] = wk1r * (x0r - x0i);
- a[j1 + 1] = wk1r * (x0r + x0i);
- x0r = x3i + x1r;
- x0i = x3r - x1i;
- a[j3 + 0] = wk1r * (x0i - x0r);
- a[j3 + 1] = wk1r * (x0i + x0r);
- }
- k1 = 0;
- m2 = 2 * m;
- for (k = m2; k < n; k += m2) {
- k1 += 2;
- k2 = 2 * k1;
- wk2r = rdft_w[k1 + 0];
- wk2i = rdft_w[k1 + 1];
- wk1r = rdft_w[k2 + 0];
- wk1i = rdft_w[k2 + 1];
- wk3r = rdft_wk3ri_first[k1 + 0];
- wk3i = rdft_wk3ri_first[k1 + 1];
- for (j0 = k; j0 < l + k; j0 += 2) {
- j1 = j0 + 8;
- j2 = j0 + 16;
- j3 = j0 + 24;
- x0r = a[j0 + 0] + a[j1 + 0];
- x0i = a[j0 + 1] + a[j1 + 1];
- x1r = a[j0 + 0] - a[j1 + 0];
- x1i = a[j0 + 1] - a[j1 + 1];
- x2r = a[j2 + 0] + a[j3 + 0];
- x2i = a[j2 + 1] + a[j3 + 1];
- x3r = a[j2 + 0] - a[j3 + 0];
- x3i = a[j2 + 1] - a[j3 + 1];
- a[j0 + 0] = x0r + x2r;
- a[j0 + 1] = x0i + x2i;
- x0r -= x2r;
- x0i -= x2i;
- a[j2 + 0] = wk2r * x0r - wk2i * x0i;
- a[j2 + 1] = wk2r * x0i + wk2i * x0r;
- x0r = x1r - x3i;
- x0i = x1i + x3r;
- a[j1 + 0] = wk1r * x0r - wk1i * x0i;
- a[j1 + 1] = wk1r * x0i + wk1i * x0r;
- x0r = x1r + x3i;
- x0i = x1i - x3r;
- a[j3 + 0] = wk3r * x0r - wk3i * x0i;
- a[j3 + 1] = wk3r * x0i + wk3i * x0r;
- }
- wk1r = rdft_w[k2 + 2];
- wk1i = rdft_w[k2 + 3];
- wk3r = rdft_wk3ri_second[k1 + 0];
- wk3i = rdft_wk3ri_second[k1 + 1];
- for (j0 = k + m; j0 < l + (k + m); j0 += 2) {
- j1 = j0 + 8;
- j2 = j0 + 16;
- j3 = j0 + 24;
- x0r = a[j0 + 0] + a[j1 + 0];
- x0i = a[j0 + 1] + a[j1 + 1];
- x1r = a[j0 + 0] - a[j1 + 0];
- x1i = a[j0 + 1] - a[j1 + 1];
- x2r = a[j2 + 0] + a[j3 + 0];
- x2i = a[j2 + 1] + a[j3 + 1];
- x3r = a[j2 + 0] - a[j3 + 0];
- x3i = a[j2 + 1] - a[j3 + 1];
- a[j0 + 0] = x0r + x2r;
- a[j0 + 1] = x0i + x2i;
- x0r -= x2r;
- x0i -= x2i;
- a[j2 + 0] = -wk2i * x0r - wk2r * x0i;
- a[j2 + 1] = -wk2i * x0i + wk2r * x0r;
- x0r = x1r - x3i;
- x0i = x1i + x3r;
- a[j1 + 0] = wk1r * x0r - wk1i * x0i;
- a[j1 + 1] = wk1r * x0i + wk1i * x0r;
- x0r = x1r + x3i;
- x0i = x1i - x3r;
- a[j3 + 0] = wk3r * x0r - wk3i * x0i;
- a[j3 + 1] = wk3r * x0i + wk3i * x0r;
- }
- }
-}
-
-static void cftfsub_128(float *a) {
- int j, j1, j2, j3, l;
- float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
-
- cft1st_128(a);
- cftmdl_128(a);
- l = 32;
- for (j = 0; j < l; j += 2) {
- j1 = j + l;
- j2 = j1 + l;
- j3 = j2 + l;
- x0r = a[j] + a[j1];
- x0i = a[j + 1] + a[j1 + 1];
- x1r = a[j] - a[j1];
- x1i = a[j + 1] - a[j1 + 1];
- x2r = a[j2] + a[j3];
- x2i = a[j2 + 1] + a[j3 + 1];
- x3r = a[j2] - a[j3];
- x3i = a[j2 + 1] - a[j3 + 1];
- a[j] = x0r + x2r;
- a[j + 1] = x0i + x2i;
- a[j2] = x0r - x2r;
- a[j2 + 1] = x0i - x2i;
- a[j1] = x1r - x3i;
- a[j1 + 1] = x1i + x3r;
- a[j3] = x1r + x3i;
- a[j3 + 1] = x1i - x3r;
- }
-}
-
-static void cftbsub_128(float *a) {
- int j, j1, j2, j3, l;
- float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
-
- cft1st_128(a);
- cftmdl_128(a);
- l = 32;
-
- for (j = 0; j < l; j += 2) {
- j1 = j + l;
- j2 = j1 + l;
- j3 = j2 + l;
- x0r = a[j] + a[j1];
- x0i = -a[j + 1] - a[j1 + 1];
- x1r = a[j] - a[j1];
- x1i = -a[j + 1] + a[j1 + 1];
- x2r = a[j2] + a[j3];
- x2i = a[j2 + 1] + a[j3 + 1];
- x3r = a[j2] - a[j3];
- x3i = a[j2 + 1] - a[j3 + 1];
- a[j] = x0r + x2r;
- a[j + 1] = x0i - x2i;
- a[j2] = x0r - x2r;
- a[j2 + 1] = x0i + x2i;
- a[j1] = x1r - x3i;
- a[j1 + 1] = x1i - x3r;
- a[j3] = x1r + x3i;
- a[j3 + 1] = x1i + x3r;
- }
-}
-
-static void rftfsub_128_C(float *a) {
- const float *c = rdft_w + 32;
- int j1, j2, k1, k2;
- float wkr, wki, xr, xi, yr, yi;
-
- for (j1 = 1, j2 = 2; j2 < 64; j1 += 1, j2 += 2) {
- k2 = 128 - j2;
- k1 = 32 - j1;
- wkr = 0.5f - c[k1];
- wki = c[j1];
- xr = a[j2 + 0] - a[k2 + 0];
- xi = a[j2 + 1] + a[k2 + 1];
- yr = wkr * xr - wki * xi;
- yi = wkr * xi + wki * xr;
- a[j2 + 0] -= yr;
- a[j2 + 1] -= yi;
- a[k2 + 0] += yr;
- a[k2 + 1] -= yi;
- }
-}
-
-static void rftbsub_128_C(float *a) {
- const float *c = rdft_w + 32;
- int j1, j2, k1, k2;
- float wkr, wki, xr, xi, yr, yi;
-
- a[1] = -a[1];
- for (j1 = 1, j2 = 2; j2 < 64; j1 += 1, j2 += 2) {
- k2 = 128 - j2;
- k1 = 32 - j1;
- wkr = 0.5f - c[k1];
- wki = c[j1];
- xr = a[j2 + 0] - a[k2 + 0];
- xi = a[j2 + 1] + a[k2 + 1];
- yr = wkr * xr + wki * xi;
- yi = wkr * xi - wki * xr;
- a[j2 + 0] = a[j2 + 0] - yr;
- a[j2 + 1] = yi - a[j2 + 1];
- a[k2 + 0] = yr + a[k2 + 0];
- a[k2 + 1] = yi - a[k2 + 1];
- }
- a[65] = -a[65];
-}
-
-void aec_rdft_forward_128(float *a) {
- const int n = 128;
- float xi;
-
- bitrv2_32or128(n, ip + 2, a);
- cftfsub_128(a);
- rftfsub_128(a);
- xi = a[0] - a[1];
- a[0] += a[1];
- a[1] = xi;
-}
-
-void aec_rdft_inverse_128(float *a) {
- const int n = 128;
-
- a[1] = 0.5f * (a[0] - a[1]);
- a[0] -= a[1];
- rftbsub_128(a);
- bitrv2_32or128(n, ip + 2, a);
- cftbsub_128(a);
-}
-
-// code path selection
-rft_sub_128_t cft1st_128;
-rft_sub_128_t cftmdl_128;
-rft_sub_128_t rftfsub_128;
-rft_sub_128_t rftbsub_128;
-
-void aec_rdft_init(void) {
- cft1st_128 = cft1st_128_C;
- cftmdl_128 = cftmdl_128_C;
- rftfsub_128 = rftfsub_128_C;
- rftbsub_128 = rftbsub_128_C;
-#if defined(WEBRTC_ARCH_X86_FAMILY)
- if (WebRtc_GetCPUInfo(kSSE2)) {
- aec_rdft_init_sse2();
- }
-#endif
- // init library constants.
- makewt_32();
- makect_32();
-}
diff --git a/trunk/src/modules/audio_processing/aec/aec_rdft.h b/trunk/src/modules/audio_processing/aec/aec_rdft.h
deleted file mode 100644
index 91bedc9..0000000
--- a/trunk/src/modules/audio_processing/aec/aec_rdft.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_RDFT_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_RDFT_H_
-
-// These intrinsics were unavailable before VS 2008.
-// TODO(andrew): move to a common file.
-#if defined(_MSC_VER) && _MSC_VER < 1500
-#include <emmintrin.h>
-static __inline __m128 _mm_castsi128_ps(__m128i a) { return *(__m128*)&a; }
-static __inline __m128i _mm_castps_si128(__m128 a) { return *(__m128i*)&a; }
-#endif
-
-#ifdef _MSC_VER /* visual c++ */
-# define ALIGN16_BEG __declspec(align(16))
-# define ALIGN16_END
-#else /* gcc or icc */
-# define ALIGN16_BEG
-# define ALIGN16_END __attribute__((aligned(16)))
-#endif
-
-// constants shared by all paths (C, SSE2).
-extern float rdft_w[64];
-// constants used by the C path.
-extern float rdft_wk3ri_first[32];
-extern float rdft_wk3ri_second[32];
-// constants used by SSE2 but initialized in C path.
-extern float rdft_wk1r[32];
-extern float rdft_wk2r[32];
-extern float rdft_wk3r[32];
-extern float rdft_wk1i[32];
-extern float rdft_wk2i[32];
-extern float rdft_wk3i[32];
-extern float cftmdl_wk1r[4];
-
-// code path selection function pointers
-typedef void (*rft_sub_128_t)(float *a);
-extern rft_sub_128_t rftfsub_128;
-extern rft_sub_128_t rftbsub_128;
-extern rft_sub_128_t cft1st_128;
-extern rft_sub_128_t cftmdl_128;
-
-// entry points
-void aec_rdft_init(void);
-void aec_rdft_init_sse2(void);
-void aec_rdft_forward_128(float *a);
-void aec_rdft_inverse_128(float *a);
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_MAIN_SOURCE_AEC_RDFT_H_
diff --git a/trunk/src/modules/audio_processing/aec/aec_rdft_sse2.c b/trunk/src/modules/audio_processing/aec/aec_rdft_sse2.c
deleted file mode 100644
index eeb3152..0000000
--- a/trunk/src/modules/audio_processing/aec/aec_rdft_sse2.c
+++ /dev/null
@@ -1,427 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "aec_rdft.h"
-
-#include <emmintrin.h>
-
-static const ALIGN16_BEG float ALIGN16_END k_swap_sign[4] =
- {-1.f, 1.f, -1.f, 1.f};
-
-static void cft1st_128_SSE2(float *a) {
- const __m128 mm_swap_sign = _mm_load_ps(k_swap_sign);
- int j, k2;
-
- for (k2 = 0, j = 0; j < 128; j += 16, k2 += 4) {
- __m128 a00v = _mm_loadu_ps(&a[j + 0]);
- __m128 a04v = _mm_loadu_ps(&a[j + 4]);
- __m128 a08v = _mm_loadu_ps(&a[j + 8]);
- __m128 a12v = _mm_loadu_ps(&a[j + 12]);
- __m128 a01v = _mm_shuffle_ps(a00v, a08v, _MM_SHUFFLE(1, 0, 1 ,0));
- __m128 a23v = _mm_shuffle_ps(a00v, a08v, _MM_SHUFFLE(3, 2, 3 ,2));
- __m128 a45v = _mm_shuffle_ps(a04v, a12v, _MM_SHUFFLE(1, 0, 1 ,0));
- __m128 a67v = _mm_shuffle_ps(a04v, a12v, _MM_SHUFFLE(3, 2, 3 ,2));
-
- const __m128 wk1rv = _mm_load_ps(&rdft_wk1r[k2]);
- const __m128 wk1iv = _mm_load_ps(&rdft_wk1i[k2]);
- const __m128 wk2rv = _mm_load_ps(&rdft_wk2r[k2]);
- const __m128 wk2iv = _mm_load_ps(&rdft_wk2i[k2]);
- const __m128 wk3rv = _mm_load_ps(&rdft_wk3r[k2]);
- const __m128 wk3iv = _mm_load_ps(&rdft_wk3i[k2]);
- __m128 x0v = _mm_add_ps(a01v, a23v);
- const __m128 x1v = _mm_sub_ps(a01v, a23v);
- const __m128 x2v = _mm_add_ps(a45v, a67v);
- const __m128 x3v = _mm_sub_ps(a45v, a67v);
- __m128 x0w;
- a01v = _mm_add_ps(x0v, x2v);
- x0v = _mm_sub_ps(x0v, x2v);
- x0w = _mm_shuffle_ps(x0v, x0v, _MM_SHUFFLE(2, 3, 0 ,1));
- {
- const __m128 a45_0v = _mm_mul_ps(wk2rv, x0v);
- const __m128 a45_1v = _mm_mul_ps(wk2iv, x0w);
- a45v = _mm_add_ps(a45_0v, a45_1v);
- }
- {
- __m128 a23_0v, a23_1v;
- const __m128 x3w = _mm_shuffle_ps(x3v, x3v, _MM_SHUFFLE(2, 3, 0 ,1));
- const __m128 x3s = _mm_mul_ps(mm_swap_sign, x3w);
- x0v = _mm_add_ps(x1v, x3s);
- x0w = _mm_shuffle_ps(x0v, x0v, _MM_SHUFFLE(2, 3, 0 ,1));
- a23_0v = _mm_mul_ps(wk1rv, x0v);
- a23_1v = _mm_mul_ps(wk1iv, x0w);
- a23v = _mm_add_ps(a23_0v, a23_1v);
-
- x0v = _mm_sub_ps(x1v, x3s);
- x0w = _mm_shuffle_ps(x0v, x0v, _MM_SHUFFLE(2, 3, 0 ,1));
- }
- {
- const __m128 a67_0v = _mm_mul_ps(wk3rv, x0v);
- const __m128 a67_1v = _mm_mul_ps(wk3iv, x0w);
- a67v = _mm_add_ps(a67_0v, a67_1v);
- }
-
- a00v = _mm_shuffle_ps(a01v, a23v, _MM_SHUFFLE(1, 0, 1 ,0));
- a04v = _mm_shuffle_ps(a45v, a67v, _MM_SHUFFLE(1, 0, 1 ,0));
- a08v = _mm_shuffle_ps(a01v, a23v, _MM_SHUFFLE(3, 2, 3 ,2));
- a12v = _mm_shuffle_ps(a45v, a67v, _MM_SHUFFLE(3, 2, 3 ,2));
- _mm_storeu_ps(&a[j + 0], a00v);
- _mm_storeu_ps(&a[j + 4], a04v);
- _mm_storeu_ps(&a[j + 8], a08v);
- _mm_storeu_ps(&a[j + 12], a12v);
- }
-}
-
-static void cftmdl_128_SSE2(float *a) {
- const int l = 8;
- const __m128 mm_swap_sign = _mm_load_ps(k_swap_sign);
- int j0;
-
- __m128 wk1rv = _mm_load_ps(cftmdl_wk1r);
- for (j0 = 0; j0 < l; j0 += 2) {
- const __m128i a_00 = _mm_loadl_epi64((__m128i*)&a[j0 + 0]);
- const __m128i a_08 = _mm_loadl_epi64((__m128i*)&a[j0 + 8]);
- const __m128i a_32 = _mm_loadl_epi64((__m128i*)&a[j0 + 32]);
- const __m128i a_40 = _mm_loadl_epi64((__m128i*)&a[j0 + 40]);
- const __m128 a_00_32 = _mm_shuffle_ps(_mm_castsi128_ps(a_00),
- _mm_castsi128_ps(a_32),
- _MM_SHUFFLE(1, 0, 1 ,0));
- const __m128 a_08_40 = _mm_shuffle_ps(_mm_castsi128_ps(a_08),
- _mm_castsi128_ps(a_40),
- _MM_SHUFFLE(1, 0, 1 ,0));
- __m128 x0r0_0i0_0r1_x0i1 = _mm_add_ps(a_00_32, a_08_40);
- const __m128 x1r0_1i0_1r1_x1i1 = _mm_sub_ps(a_00_32, a_08_40);
-
- const __m128i a_16 = _mm_loadl_epi64((__m128i*)&a[j0 + 16]);
- const __m128i a_24 = _mm_loadl_epi64((__m128i*)&a[j0 + 24]);
- const __m128i a_48 = _mm_loadl_epi64((__m128i*)&a[j0 + 48]);
- const __m128i a_56 = _mm_loadl_epi64((__m128i*)&a[j0 + 56]);
- const __m128 a_16_48 = _mm_shuffle_ps(_mm_castsi128_ps(a_16),
- _mm_castsi128_ps(a_48),
- _MM_SHUFFLE(1, 0, 1 ,0));
- const __m128 a_24_56 = _mm_shuffle_ps(_mm_castsi128_ps(a_24),
- _mm_castsi128_ps(a_56),
- _MM_SHUFFLE(1, 0, 1 ,0));
- const __m128 x2r0_2i0_2r1_x2i1 = _mm_add_ps(a_16_48, a_24_56);
- const __m128 x3r0_3i0_3r1_x3i1 = _mm_sub_ps(a_16_48, a_24_56);
-
- const __m128 xx0 = _mm_add_ps(x0r0_0i0_0r1_x0i1, x2r0_2i0_2r1_x2i1);
- const __m128 xx1 = _mm_sub_ps(x0r0_0i0_0r1_x0i1, x2r0_2i0_2r1_x2i1);
-
- const __m128 x3i0_3r0_3i1_x3r1 = _mm_castsi128_ps(
- _mm_shuffle_epi32(_mm_castps_si128(x3r0_3i0_3r1_x3i1),
- _MM_SHUFFLE(2, 3, 0, 1)));
- const __m128 x3_swapped = _mm_mul_ps(mm_swap_sign, x3i0_3r0_3i1_x3r1);
- const __m128 x1_x3_add = _mm_add_ps(x1r0_1i0_1r1_x1i1, x3_swapped);
- const __m128 x1_x3_sub = _mm_sub_ps(x1r0_1i0_1r1_x1i1, x3_swapped);
-
- const __m128 yy0 = _mm_shuffle_ps(x1_x3_add, x1_x3_sub,
- _MM_SHUFFLE(2, 2, 2 ,2));
- const __m128 yy1 = _mm_shuffle_ps(x1_x3_add, x1_x3_sub,
- _MM_SHUFFLE(3, 3, 3 ,3));
- const __m128 yy2 = _mm_mul_ps(mm_swap_sign, yy1);
- const __m128 yy3 = _mm_add_ps(yy0, yy2);
- const __m128 yy4 = _mm_mul_ps(wk1rv, yy3);
-
- _mm_storel_epi64((__m128i*)&a[j0 + 0], _mm_castps_si128(xx0));
- _mm_storel_epi64((__m128i*)&a[j0 + 32],
- _mm_shuffle_epi32(_mm_castps_si128(xx0),
- _MM_SHUFFLE(3, 2, 3, 2)));
-
- _mm_storel_epi64((__m128i*)&a[j0 + 16], _mm_castps_si128(xx1));
- _mm_storel_epi64((__m128i*)&a[j0 + 48],
- _mm_shuffle_epi32(_mm_castps_si128(xx1),
- _MM_SHUFFLE(2, 3, 2, 3)));
- a[j0 + 48] = -a[j0 + 48];
-
- _mm_storel_epi64((__m128i*)&a[j0 + 8], _mm_castps_si128(x1_x3_add));
- _mm_storel_epi64((__m128i*)&a[j0 + 24], _mm_castps_si128(x1_x3_sub));
-
- _mm_storel_epi64((__m128i*)&a[j0 + 40], _mm_castps_si128(yy4));
- _mm_storel_epi64((__m128i*)&a[j0 + 56],
- _mm_shuffle_epi32(_mm_castps_si128(yy4),
- _MM_SHUFFLE(2, 3, 2, 3)));
- }
-
- {
- int k = 64;
- int k1 = 2;
- int k2 = 2 * k1;
- const __m128 wk2rv = _mm_load_ps(&rdft_wk2r[k2+0]);
- const __m128 wk2iv = _mm_load_ps(&rdft_wk2i[k2+0]);
- const __m128 wk1iv = _mm_load_ps(&rdft_wk1i[k2+0]);
- const __m128 wk3rv = _mm_load_ps(&rdft_wk3r[k2+0]);
- const __m128 wk3iv = _mm_load_ps(&rdft_wk3i[k2+0]);
- wk1rv = _mm_load_ps(&rdft_wk1r[k2+0]);
- for (j0 = k; j0 < l + k; j0 += 2) {
- const __m128i a_00 = _mm_loadl_epi64((__m128i*)&a[j0 + 0]);
- const __m128i a_08 = _mm_loadl_epi64((__m128i*)&a[j0 + 8]);
- const __m128i a_32 = _mm_loadl_epi64((__m128i*)&a[j0 + 32]);
- const __m128i a_40 = _mm_loadl_epi64((__m128i*)&a[j0 + 40]);
- const __m128 a_00_32 = _mm_shuffle_ps(_mm_castsi128_ps(a_00),
- _mm_castsi128_ps(a_32),
- _MM_SHUFFLE(1, 0, 1 ,0));
- const __m128 a_08_40 = _mm_shuffle_ps(_mm_castsi128_ps(a_08),
- _mm_castsi128_ps(a_40),
- _MM_SHUFFLE(1, 0, 1 ,0));
- __m128 x0r0_0i0_0r1_x0i1 = _mm_add_ps(a_00_32, a_08_40);
- const __m128 x1r0_1i0_1r1_x1i1 = _mm_sub_ps(a_00_32, a_08_40);
-
- const __m128i a_16 = _mm_loadl_epi64((__m128i*)&a[j0 + 16]);
- const __m128i a_24 = _mm_loadl_epi64((__m128i*)&a[j0 + 24]);
- const __m128i a_48 = _mm_loadl_epi64((__m128i*)&a[j0 + 48]);
- const __m128i a_56 = _mm_loadl_epi64((__m128i*)&a[j0 + 56]);
- const __m128 a_16_48 = _mm_shuffle_ps(_mm_castsi128_ps(a_16),
- _mm_castsi128_ps(a_48),
- _MM_SHUFFLE(1, 0, 1 ,0));
- const __m128 a_24_56 = _mm_shuffle_ps(_mm_castsi128_ps(a_24),
- _mm_castsi128_ps(a_56),
- _MM_SHUFFLE(1, 0, 1 ,0));
- const __m128 x2r0_2i0_2r1_x2i1 = _mm_add_ps(a_16_48, a_24_56);
- const __m128 x3r0_3i0_3r1_x3i1 = _mm_sub_ps(a_16_48, a_24_56);
-
- const __m128 xx = _mm_add_ps(x0r0_0i0_0r1_x0i1, x2r0_2i0_2r1_x2i1);
- const __m128 xx1 = _mm_sub_ps(x0r0_0i0_0r1_x0i1, x2r0_2i0_2r1_x2i1);
- const __m128 xx2 = _mm_mul_ps(xx1 , wk2rv);
- const __m128 xx3 = _mm_mul_ps(wk2iv,
- _mm_castsi128_ps(_mm_shuffle_epi32(_mm_castps_si128(xx1),
- _MM_SHUFFLE(2, 3, 0, 1))));
- const __m128 xx4 = _mm_add_ps(xx2, xx3);
-
- const __m128 x3i0_3r0_3i1_x3r1 = _mm_castsi128_ps(
- _mm_shuffle_epi32(_mm_castps_si128(x3r0_3i0_3r1_x3i1),
- _MM_SHUFFLE(2, 3, 0, 1)));
- const __m128 x3_swapped = _mm_mul_ps(mm_swap_sign, x3i0_3r0_3i1_x3r1);
- const __m128 x1_x3_add = _mm_add_ps(x1r0_1i0_1r1_x1i1, x3_swapped);
- const __m128 x1_x3_sub = _mm_sub_ps(x1r0_1i0_1r1_x1i1, x3_swapped);
-
- const __m128 xx10 = _mm_mul_ps(x1_x3_add, wk1rv);
- const __m128 xx11 = _mm_mul_ps(wk1iv,
- _mm_castsi128_ps(_mm_shuffle_epi32(_mm_castps_si128(x1_x3_add),
- _MM_SHUFFLE(2, 3, 0, 1))));
- const __m128 xx12 = _mm_add_ps(xx10, xx11);
-
- const __m128 xx20 = _mm_mul_ps(x1_x3_sub, wk3rv);
- const __m128 xx21 = _mm_mul_ps(wk3iv,
- _mm_castsi128_ps(_mm_shuffle_epi32(_mm_castps_si128(x1_x3_sub),
- _MM_SHUFFLE(2, 3, 0, 1))));
- const __m128 xx22 = _mm_add_ps(xx20, xx21);
-
- _mm_storel_epi64((__m128i*)&a[j0 + 0], _mm_castps_si128(xx));
- _mm_storel_epi64((__m128i*)&a[j0 + 32],
- _mm_shuffle_epi32(_mm_castps_si128(xx),
- _MM_SHUFFLE(3, 2, 3, 2)));
-
- _mm_storel_epi64((__m128i*)&a[j0 + 16], _mm_castps_si128(xx4));
- _mm_storel_epi64((__m128i*)&a[j0 + 48],
- _mm_shuffle_epi32(_mm_castps_si128(xx4),
- _MM_SHUFFLE(3, 2, 3, 2)));
-
- _mm_storel_epi64((__m128i*)&a[j0 + 8], _mm_castps_si128(xx12));
- _mm_storel_epi64((__m128i*)&a[j0 + 40],
- _mm_shuffle_epi32(_mm_castps_si128(xx12),
- _MM_SHUFFLE(3, 2, 3, 2)));
-
- _mm_storel_epi64((__m128i*)&a[j0 + 24], _mm_castps_si128(xx22));
- _mm_storel_epi64((__m128i*)&a[j0 + 56],
- _mm_shuffle_epi32(_mm_castps_si128(xx22),
- _MM_SHUFFLE(3, 2, 3, 2)));
- }
- }
-}
-
-static void rftfsub_128_SSE2(float *a) {
- const float *c = rdft_w + 32;
- int j1, j2, k1, k2;
- float wkr, wki, xr, xi, yr, yi;
-
- static const ALIGN16_BEG float ALIGN16_END k_half[4] =
- {0.5f, 0.5f, 0.5f, 0.5f};
- const __m128 mm_half = _mm_load_ps(k_half);
-
- // Vectorized code (four at once).
- // Note: commented number are indexes for the first iteration of the loop.
- for (j1 = 1, j2 = 2; j2 + 7 < 64; j1 += 4, j2 += 8) {
- // Load 'wk'.
- const __m128 c_j1 = _mm_loadu_ps(&c[ j1]); // 1, 2, 3, 4,
- const __m128 c_k1 = _mm_loadu_ps(&c[29 - j1]); // 28, 29, 30, 31,
- const __m128 wkrt = _mm_sub_ps(mm_half, c_k1); // 28, 29, 30, 31,
- const __m128 wkr_ =
- _mm_shuffle_ps(wkrt, wkrt, _MM_SHUFFLE(0, 1, 2, 3)); // 31, 30, 29, 28,
- const __m128 wki_ = c_j1; // 1, 2, 3, 4,
- // Load and shuffle 'a'.
- const __m128 a_j2_0 = _mm_loadu_ps(&a[0 + j2]); // 2, 3, 4, 5,
- const __m128 a_j2_4 = _mm_loadu_ps(&a[4 + j2]); // 6, 7, 8, 9,
- const __m128 a_k2_0 = _mm_loadu_ps(&a[122 - j2]); // 120, 121, 122, 123,
- const __m128 a_k2_4 = _mm_loadu_ps(&a[126 - j2]); // 124, 125, 126, 127,
- const __m128 a_j2_p0 = _mm_shuffle_ps(a_j2_0, a_j2_4,
- _MM_SHUFFLE(2, 0, 2 ,0)); // 2, 4, 6, 8,
- const __m128 a_j2_p1 = _mm_shuffle_ps(a_j2_0, a_j2_4,
- _MM_SHUFFLE(3, 1, 3 ,1)); // 3, 5, 7, 9,
- const __m128 a_k2_p0 = _mm_shuffle_ps(a_k2_4, a_k2_0,
- _MM_SHUFFLE(0, 2, 0 ,2)); // 126, 124, 122, 120,
- const __m128 a_k2_p1 = _mm_shuffle_ps(a_k2_4, a_k2_0,
- _MM_SHUFFLE(1, 3, 1 ,3)); // 127, 125, 123, 121,
- // Calculate 'x'.
- const __m128 xr_ = _mm_sub_ps(a_j2_p0, a_k2_p0);
- // 2-126, 4-124, 6-122, 8-120,
- const __m128 xi_ = _mm_add_ps(a_j2_p1, a_k2_p1);
- // 3-127, 5-125, 7-123, 9-121,
- // Calculate product into 'y'.
- // yr = wkr * xr - wki * xi;
- // yi = wkr * xi + wki * xr;
- const __m128 a_ = _mm_mul_ps(wkr_, xr_);
- const __m128 b_ = _mm_mul_ps(wki_, xi_);
- const __m128 c_ = _mm_mul_ps(wkr_, xi_);
- const __m128 d_ = _mm_mul_ps(wki_, xr_);
- const __m128 yr_ = _mm_sub_ps(a_, b_); // 2-126, 4-124, 6-122, 8-120,
- const __m128 yi_ = _mm_add_ps(c_, d_); // 3-127, 5-125, 7-123, 9-121,
- // Update 'a'.
- // a[j2 + 0] -= yr;
- // a[j2 + 1] -= yi;
- // a[k2 + 0] += yr;
- // a[k2 + 1] -= yi;
- const __m128 a_j2_p0n = _mm_sub_ps(a_j2_p0, yr_); // 2, 4, 6, 8,
- const __m128 a_j2_p1n = _mm_sub_ps(a_j2_p1, yi_); // 3, 5, 7, 9,
- const __m128 a_k2_p0n = _mm_add_ps(a_k2_p0, yr_); // 126, 124, 122, 120,
- const __m128 a_k2_p1n = _mm_sub_ps(a_k2_p1, yi_); // 127, 125, 123, 121,
- // Shuffle in right order and store.
- const __m128 a_j2_0n = _mm_unpacklo_ps(a_j2_p0n, a_j2_p1n);
- // 2, 3, 4, 5,
- const __m128 a_j2_4n = _mm_unpackhi_ps(a_j2_p0n, a_j2_p1n);
- // 6, 7, 8, 9,
- const __m128 a_k2_0nt = _mm_unpackhi_ps(a_k2_p0n, a_k2_p1n);
- // 122, 123, 120, 121,
- const __m128 a_k2_4nt = _mm_unpacklo_ps(a_k2_p0n, a_k2_p1n);
- // 126, 127, 124, 125,
- const __m128 a_k2_0n = _mm_shuffle_ps(a_k2_0nt, a_k2_0nt,
- _MM_SHUFFLE(1, 0, 3 ,2)); // 120, 121, 122, 123,
- const __m128 a_k2_4n = _mm_shuffle_ps(a_k2_4nt, a_k2_4nt,
- _MM_SHUFFLE(1, 0, 3 ,2)); // 124, 125, 126, 127,
- _mm_storeu_ps(&a[0 + j2], a_j2_0n);
- _mm_storeu_ps(&a[4 + j2], a_j2_4n);
- _mm_storeu_ps(&a[122 - j2], a_k2_0n);
- _mm_storeu_ps(&a[126 - j2], a_k2_4n);
- }
- // Scalar code for the remaining items.
- for (; j2 < 64; j1 += 1, j2 += 2) {
- k2 = 128 - j2;
- k1 = 32 - j1;
- wkr = 0.5f - c[k1];
- wki = c[j1];
- xr = a[j2 + 0] - a[k2 + 0];
- xi = a[j2 + 1] + a[k2 + 1];
- yr = wkr * xr - wki * xi;
- yi = wkr * xi + wki * xr;
- a[j2 + 0] -= yr;
- a[j2 + 1] -= yi;
- a[k2 + 0] += yr;
- a[k2 + 1] -= yi;
- }
-}
-
-static void rftbsub_128_SSE2(float *a) {
- const float *c = rdft_w + 32;
- int j1, j2, k1, k2;
- float wkr, wki, xr, xi, yr, yi;
-
- static const ALIGN16_BEG float ALIGN16_END k_half[4] =
- {0.5f, 0.5f, 0.5f, 0.5f};
- const __m128 mm_half = _mm_load_ps(k_half);
-
- a[1] = -a[1];
- // Vectorized code (four at once).
- // Note: commented number are indexes for the first iteration of the loop.
- for (j1 = 1, j2 = 2; j2 + 7 < 64; j1 += 4, j2 += 8) {
- // Load 'wk'.
- const __m128 c_j1 = _mm_loadu_ps(&c[ j1]); // 1, 2, 3, 4,
- const __m128 c_k1 = _mm_loadu_ps(&c[29 - j1]); // 28, 29, 30, 31,
- const __m128 wkrt = _mm_sub_ps(mm_half, c_k1); // 28, 29, 30, 31,
- const __m128 wkr_ =
- _mm_shuffle_ps(wkrt, wkrt, _MM_SHUFFLE(0, 1, 2, 3)); // 31, 30, 29, 28,
- const __m128 wki_ = c_j1; // 1, 2, 3, 4,
- // Load and shuffle 'a'.
- const __m128 a_j2_0 = _mm_loadu_ps(&a[0 + j2]); // 2, 3, 4, 5,
- const __m128 a_j2_4 = _mm_loadu_ps(&a[4 + j2]); // 6, 7, 8, 9,
- const __m128 a_k2_0 = _mm_loadu_ps(&a[122 - j2]); // 120, 121, 122, 123,
- const __m128 a_k2_4 = _mm_loadu_ps(&a[126 - j2]); // 124, 125, 126, 127,
- const __m128 a_j2_p0 = _mm_shuffle_ps(a_j2_0, a_j2_4,
- _MM_SHUFFLE(2, 0, 2 ,0)); // 2, 4, 6, 8,
- const __m128 a_j2_p1 = _mm_shuffle_ps(a_j2_0, a_j2_4,
- _MM_SHUFFLE(3, 1, 3 ,1)); // 3, 5, 7, 9,
- const __m128 a_k2_p0 = _mm_shuffle_ps(a_k2_4, a_k2_0,
- _MM_SHUFFLE(0, 2, 0 ,2)); // 126, 124, 122, 120,
- const __m128 a_k2_p1 = _mm_shuffle_ps(a_k2_4, a_k2_0,
- _MM_SHUFFLE(1, 3, 1 ,3)); // 127, 125, 123, 121,
- // Calculate 'x'.
- const __m128 xr_ = _mm_sub_ps(a_j2_p0, a_k2_p0);
- // 2-126, 4-124, 6-122, 8-120,
- const __m128 xi_ = _mm_add_ps(a_j2_p1, a_k2_p1);
- // 3-127, 5-125, 7-123, 9-121,
- // Calculate product into 'y'.
- // yr = wkr * xr + wki * xi;
- // yi = wkr * xi - wki * xr;
- const __m128 a_ = _mm_mul_ps(wkr_, xr_);
- const __m128 b_ = _mm_mul_ps(wki_, xi_);
- const __m128 c_ = _mm_mul_ps(wkr_, xi_);
- const __m128 d_ = _mm_mul_ps(wki_, xr_);
- const __m128 yr_ = _mm_add_ps(a_, b_); // 2-126, 4-124, 6-122, 8-120,
- const __m128 yi_ = _mm_sub_ps(c_, d_); // 3-127, 5-125, 7-123, 9-121,
- // Update 'a'.
- // a[j2 + 0] = a[j2 + 0] - yr;
- // a[j2 + 1] = yi - a[j2 + 1];
- // a[k2 + 0] = yr + a[k2 + 0];
- // a[k2 + 1] = yi - a[k2 + 1];
- const __m128 a_j2_p0n = _mm_sub_ps(a_j2_p0, yr_); // 2, 4, 6, 8,
- const __m128 a_j2_p1n = _mm_sub_ps(yi_, a_j2_p1); // 3, 5, 7, 9,
- const __m128 a_k2_p0n = _mm_add_ps(a_k2_p0, yr_); // 126, 124, 122, 120,
- const __m128 a_k2_p1n = _mm_sub_ps(yi_, a_k2_p1); // 127, 125, 123, 121,
- // Shuffle in right order and store.
- const __m128 a_j2_0n = _mm_unpacklo_ps(a_j2_p0n, a_j2_p1n);
- // 2, 3, 4, 5,
- const __m128 a_j2_4n = _mm_unpackhi_ps(a_j2_p0n, a_j2_p1n);
- // 6, 7, 8, 9,
- const __m128 a_k2_0nt = _mm_unpackhi_ps(a_k2_p0n, a_k2_p1n);
- // 122, 123, 120, 121,
- const __m128 a_k2_4nt = _mm_unpacklo_ps(a_k2_p0n, a_k2_p1n);
- // 126, 127, 124, 125,
- const __m128 a_k2_0n = _mm_shuffle_ps(a_k2_0nt, a_k2_0nt,
- _MM_SHUFFLE(1, 0, 3 ,2)); // 120, 121, 122, 123,
- const __m128 a_k2_4n = _mm_shuffle_ps(a_k2_4nt, a_k2_4nt,
- _MM_SHUFFLE(1, 0, 3 ,2)); // 124, 125, 126, 127,
- _mm_storeu_ps(&a[0 + j2], a_j2_0n);
- _mm_storeu_ps(&a[4 + j2], a_j2_4n);
- _mm_storeu_ps(&a[122 - j2], a_k2_0n);
- _mm_storeu_ps(&a[126 - j2], a_k2_4n);
- }
- // Scalar code for the remaining items.
- for (; j2 < 64; j1 += 1, j2 += 2) {
- k2 = 128 - j2;
- k1 = 32 - j1;
- wkr = 0.5f - c[k1];
- wki = c[j1];
- xr = a[j2 + 0] - a[k2 + 0];
- xi = a[j2 + 1] + a[k2 + 1];
- yr = wkr * xr + wki * xi;
- yi = wkr * xi - wki * xr;
- a[j2 + 0] = a[j2 + 0] - yr;
- a[j2 + 1] = yi - a[j2 + 1];
- a[k2 + 0] = yr + a[k2 + 0];
- a[k2 + 1] = yi - a[k2 + 1];
- }
- a[65] = -a[65];
-}
-
-void aec_rdft_init_sse2(void) {
- cft1st_128 = cft1st_128_SSE2;
- cftmdl_128 = cftmdl_128_SSE2;
- rftfsub_128 = rftfsub_128_SSE2;
- rftbsub_128 = rftbsub_128_SSE2;
-}
-
diff --git a/trunk/src/modules/audio_processing/aec/aec_resampler.c b/trunk/src/modules/audio_processing/aec/aec_resampler.c
deleted file mode 100644
index ea980cd..0000000
--- a/trunk/src/modules/audio_processing/aec/aec_resampler.c
+++ /dev/null
@@ -1,233 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/* Resamples a signal to an arbitrary rate. Used by the AEC to compensate for clock
- * skew by resampling the farend signal.
- */
-
-#include "aec_resampler.h"
-
-#include <assert.h>
-#include <stdlib.h>
-#include <string.h>
-#include <math.h>
-
-#include "aec_core.h"
-
-enum { kEstimateLengthFrames = 400 };
-
-typedef struct {
- short buffer[kResamplerBufferSize];
- float position;
-
- int deviceSampleRateHz;
- int skewData[kEstimateLengthFrames];
- int skewDataIndex;
- float skewEstimate;
-} resampler_t;
-
-static int EstimateSkew(const int* rawSkew,
- int size,
- int absLimit,
- float *skewEst);
-
-int WebRtcAec_CreateResampler(void **resampInst)
-{
- resampler_t *obj = malloc(sizeof(resampler_t));
- *resampInst = obj;
- if (obj == NULL) {
- return -1;
- }
-
- return 0;
-}
-
-int WebRtcAec_InitResampler(void *resampInst, int deviceSampleRateHz)
-{
- resampler_t *obj = (resampler_t*) resampInst;
- memset(obj->buffer, 0, sizeof(obj->buffer));
- obj->position = 0.0;
-
- obj->deviceSampleRateHz = deviceSampleRateHz;
- memset(obj->skewData, 0, sizeof(obj->skewData));
- obj->skewDataIndex = 0;
- obj->skewEstimate = 0.0;
-
- return 0;
-}
-
-int WebRtcAec_FreeResampler(void *resampInst)
-{
- resampler_t *obj = (resampler_t*) resampInst;
- free(obj);
-
- return 0;
-}
-
-int WebRtcAec_ResampleLinear(void *resampInst,
- const short *inspeech,
- int size,
- float skew,
- short *outspeech)
-{
- resampler_t *obj = (resampler_t*) resampInst;
-
- short *y;
- float be, tnew, interp;
- int tn, outsize, mm;
-
- if (size < 0 || size > 2 * FRAME_LEN) {
- return -1;
- }
-
- // Add new frame data in lookahead
- memcpy(&obj->buffer[FRAME_LEN + kResamplingDelay],
- inspeech,
- size * sizeof(short));
-
- // Sample rate ratio
- be = 1 + skew;
-
- // Loop over input frame
- mm = 0;
- y = &obj->buffer[FRAME_LEN]; // Point at current frame
-
- tnew = be * mm + obj->position;
- tn = (int) tnew;
-
- while (tn < size) {
-
- // Interpolation
- interp = y[tn] + (tnew - tn) * (y[tn+1] - y[tn]);
-
- if (interp > 32767) {
- interp = 32767;
- }
- else if (interp < -32768) {
- interp = -32768;
- }
-
- outspeech[mm] = (short) interp;
- mm++;
-
- tnew = be * mm + obj->position;
- tn = (int) tnew;
- }
-
- outsize = mm;
- obj->position += outsize * be - size;
-
- // Shift buffer
- memmove(obj->buffer,
- &obj->buffer[size],
- (kResamplerBufferSize - size) * sizeof(short));
-
- return outsize;
-}
-
-int WebRtcAec_GetSkew(void *resampInst, int rawSkew, float *skewEst)
-{
- resampler_t *obj = (resampler_t*)resampInst;
- int err = 0;
-
- if (obj->skewDataIndex < kEstimateLengthFrames) {
- obj->skewData[obj->skewDataIndex] = rawSkew;
- obj->skewDataIndex++;
- }
- else if (obj->skewDataIndex == kEstimateLengthFrames) {
- err = EstimateSkew(obj->skewData,
- kEstimateLengthFrames,
- obj->deviceSampleRateHz,
- skewEst);
- obj->skewEstimate = *skewEst;
- obj->skewDataIndex++;
- }
- else {
- *skewEst = obj->skewEstimate;
- }
-
- return err;
-}
-
-int EstimateSkew(const int* rawSkew,
- int size,
- int deviceSampleRateHz,
- float *skewEst)
-{
- const int absLimitOuter = (int)(0.04f * deviceSampleRateHz);
- const int absLimitInner = (int)(0.0025f * deviceSampleRateHz);
- int i = 0;
- int n = 0;
- float rawAvg = 0;
- float err = 0;
- float rawAbsDev = 0;
- int upperLimit = 0;
- int lowerLimit = 0;
- float cumSum = 0;
- float x = 0;
- float x2 = 0;
- float y = 0;
- float xy = 0;
- float xAvg = 0;
- float denom = 0;
- float skew = 0;
-
- *skewEst = 0; // Set in case of error below.
- for (i = 0; i < size; i++) {
- if ((rawSkew[i] < absLimitOuter && rawSkew[i] > -absLimitOuter)) {
- n++;
- rawAvg += rawSkew[i];
- }
- }
-
- if (n == 0) {
- return -1;
- }
- assert(n > 0);
- rawAvg /= n;
-
- for (i = 0; i < size; i++) {
- if ((rawSkew[i] < absLimitOuter && rawSkew[i] > -absLimitOuter)) {
- err = rawSkew[i] - rawAvg;
- rawAbsDev += err >= 0 ? err : -err;
- }
- }
- assert(n > 0);
- rawAbsDev /= n;
- upperLimit = (int)(rawAvg + 5 * rawAbsDev + 1); // +1 for ceiling.
- lowerLimit = (int)(rawAvg - 5 * rawAbsDev - 1); // -1 for floor.
-
- n = 0;
- for (i = 0; i < size; i++) {
- if ((rawSkew[i] < absLimitInner && rawSkew[i] > -absLimitInner) ||
- (rawSkew[i] < upperLimit && rawSkew[i] > lowerLimit)) {
- n++;
- cumSum += rawSkew[i];
- x += n;
- x2 += n*n;
- y += cumSum;
- xy += n * cumSum;
- }
- }
-
- if (n == 0) {
- return -1;
- }
- assert(n > 0);
- xAvg = x / n;
- denom = x2 - xAvg*x;
-
- if (denom != 0) {
- skew = (xy - xAvg*y) / denom;
- }
-
- *skewEst = skew;
- return 0;
-}
diff --git a/trunk/src/modules/audio_processing/aec/aec_resampler.h b/trunk/src/modules/audio_processing/aec/aec_resampler.h
deleted file mode 100644
index ab4cc6e..0000000
--- a/trunk/src/modules/audio_processing/aec/aec_resampler.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
-
-#include "aec_core.h"
-
-enum { kResamplingDelay = 1 };
-enum { kResamplerBufferSize = FRAME_LEN * 4 };
-
-// Unless otherwise specified, functions return 0 on success and -1 on error
-int WebRtcAec_CreateResampler(void **resampInst);
-int WebRtcAec_InitResampler(void *resampInst, int deviceSampleRateHz);
-int WebRtcAec_FreeResampler(void *resampInst);
-
-// Estimates skew from raw measurement.
-int WebRtcAec_GetSkew(void *resampInst, int rawSkew, float *skewEst);
-
-// Resamples input using linear interpolation.
-// Returns size of resampled array.
-int WebRtcAec_ResampleLinear(void *resampInst,
- const short *inspeech,
- int size,
- float skew,
- short *outspeech);
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_AEC_RESAMPLER_H_
diff --git a/trunk/src/modules/audio_processing/aec/echo_cancellation.c b/trunk/src/modules/audio_processing/aec/echo_cancellation.c
deleted file mode 100644
index bde9c87..0000000
--- a/trunk/src/modules/audio_processing/aec/echo_cancellation.c
+++ /dev/null
@@ -1,924 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Contains the API functions for the AEC.
- */
-#include "echo_cancellation.h"
-
-#include <math.h>
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-#include <stdio.h>
-#endif
-#include <stdlib.h>
-#include <string.h>
-
-#include "aec_core.h"
-#include "aec_resampler.h"
-#include "common_audio/signal_processing/include/signal_processing_library.h"
-#include "ring_buffer.h"
-#include "typedefs.h"
-
-// Maximum length of resampled signal. Must be an integer multiple of frames
-// (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN
-// The factor of 2 handles wb, and the + 1 is as a safety margin
-// TODO(bjornv): Replace with kResamplerBufferSize
-#define MAX_RESAMP_LEN (5 * FRAME_LEN)
-
-static const int kMaxBufSizeStart = 62; // In partitions
-static const int sampMsNb = 8; // samples per ms in nb
-// Target suppression levels for nlp modes
-// log{0.001, 0.00001, 0.00000001}
-static const float targetSupp[3] = {-6.9f, -11.5f, -18.4f};
-static const float minOverDrive[3] = {1.0f, 2.0f, 5.0f};
-static const int initCheck = 42;
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
-static int instance_count = 0;
-#endif
-
-typedef struct {
- int delayCtr;
- int sampFreq;
- int splitSampFreq;
- int scSampFreq;
- float sampFactor; // scSampRate / sampFreq
- short nlpMode;
- short autoOnOff;
- short activity;
- short skewMode;
- int bufSizeStart;
- //short bufResetCtr; // counts number of noncausal frames
- int knownDelay;
-
- short initFlag; // indicates if AEC has been initialized
-
- // Variables used for averaging far end buffer size
- short counter;
- int sum;
- short firstVal;
- short checkBufSizeCtr;
-
- // Variables used for delay shifts
- short msInSndCardBuf;
- short filtDelay; // Filtered delay estimate.
- int timeForDelayChange;
- int ECstartup;
- int checkBuffSize;
- short lastDelayDiff;
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- void* far_pre_buf_s16; // Time domain far-end pre-buffer in int16_t.
- FILE *bufFile;
- FILE *delayFile;
- FILE *skewFile;
-#endif
-
- // Structures
- void *resampler;
-
- int skewFrCtr;
- int resample; // if the skew is small enough we don't resample
- int highSkewCtr;
- float skew;
-
- void* far_pre_buf; // Time domain far-end pre-buffer.
-
- int lastError;
-
- aec_t *aec;
-} aecpc_t;
-
-// Estimates delay to set the position of the far-end buffer read pointer
-// (controlled by knownDelay)
-static int EstBufDelay(aecpc_t *aecInst);
-
-WebRtc_Word32 WebRtcAec_Create(void **aecInst)
-{
- aecpc_t *aecpc;
- if (aecInst == NULL) {
- return -1;
- }
-
- aecpc = malloc(sizeof(aecpc_t));
- *aecInst = aecpc;
- if (aecpc == NULL) {
- return -1;
- }
-
- if (WebRtcAec_CreateAec(&aecpc->aec) == -1) {
- WebRtcAec_Free(aecpc);
- aecpc = NULL;
- return -1;
- }
-
- if (WebRtcAec_CreateResampler(&aecpc->resampler) == -1) {
- WebRtcAec_Free(aecpc);
- aecpc = NULL;
- return -1;
- }
- // Create far-end pre-buffer. The buffer size has to be large enough for
- // largest possible drift compensation (kResamplerBufferSize) + "almost" an
- // FFT buffer (PART_LEN2 - 1).
- if (WebRtc_CreateBuffer(&aecpc->far_pre_buf,
- PART_LEN2 + kResamplerBufferSize,
- sizeof(float)) == -1) {
- WebRtcAec_Free(aecpc);
- aecpc = NULL;
- return -1;
- }
-
- aecpc->initFlag = 0;
- aecpc->lastError = 0;
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- if (WebRtc_CreateBuffer(&aecpc->far_pre_buf_s16,
- PART_LEN2 + kResamplerBufferSize,
- sizeof(int16_t)) == -1) {
- WebRtcAec_Free(aecpc);
- aecpc = NULL;
- return -1;
- }
- {
- char filename[64];
- sprintf(filename, "aec_far%d.pcm", instance_count);
- aecpc->aec->farFile = fopen(filename, "wb");
- sprintf(filename, "aec_near%d.pcm", instance_count);
- aecpc->aec->nearFile = fopen(filename, "wb");
- sprintf(filename, "aec_out%d.pcm", instance_count);
- aecpc->aec->outFile = fopen(filename, "wb");
- sprintf(filename, "aec_out_linear%d.pcm", instance_count);
- aecpc->aec->outLinearFile = fopen(filename, "wb");
- sprintf(filename, "aec_buf%d.dat", instance_count);
- aecpc->bufFile = fopen(filename, "wb");
- sprintf(filename, "aec_skew%d.dat", instance_count);
- aecpc->skewFile = fopen(filename, "wb");
- sprintf(filename, "aec_delay%d.dat", instance_count);
- aecpc->delayFile = fopen(filename, "wb");
- instance_count++;
- }
-#endif
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAec_Free(void *aecInst)
-{
- aecpc_t *aecpc = aecInst;
-
- if (aecpc == NULL) {
- return -1;
- }
-
- WebRtc_FreeBuffer(aecpc->far_pre_buf);
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- WebRtc_FreeBuffer(aecpc->far_pre_buf_s16);
- fclose(aecpc->aec->farFile);
- fclose(aecpc->aec->nearFile);
- fclose(aecpc->aec->outFile);
- fclose(aecpc->aec->outLinearFile);
- fclose(aecpc->bufFile);
- fclose(aecpc->skewFile);
- fclose(aecpc->delayFile);
-#endif
-
- WebRtcAec_FreeAec(aecpc->aec);
- WebRtcAec_FreeResampler(aecpc->resampler);
- free(aecpc);
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAec_Init(void *aecInst, WebRtc_Word32 sampFreq, WebRtc_Word32 scSampFreq)
-{
- aecpc_t *aecpc = aecInst;
- AecConfig aecConfig;
-
- if (aecpc == NULL) {
- return -1;
- }
-
- if (sampFreq != 8000 && sampFreq != 16000 && sampFreq != 32000) {
- aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
- return -1;
- }
- aecpc->sampFreq = sampFreq;
-
- if (scSampFreq < 1 || scSampFreq > 96000) {
- aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
- return -1;
- }
- aecpc->scSampFreq = scSampFreq;
-
- // Initialize echo canceller core
- if (WebRtcAec_InitAec(aecpc->aec, aecpc->sampFreq) == -1) {
- aecpc->lastError = AEC_UNSPECIFIED_ERROR;
- return -1;
- }
-
- if (WebRtcAec_InitResampler(aecpc->resampler, aecpc->scSampFreq) == -1) {
- aecpc->lastError = AEC_UNSPECIFIED_ERROR;
- return -1;
- }
-
- if (WebRtc_InitBuffer(aecpc->far_pre_buf) == -1) {
- aecpc->lastError = AEC_UNSPECIFIED_ERROR;
- return -1;
- }
- WebRtc_MoveReadPtr(aecpc->far_pre_buf, -PART_LEN); // Start overlap.
-
- aecpc->initFlag = initCheck; // indicates that initialization has been done
-
- if (aecpc->sampFreq == 32000) {
- aecpc->splitSampFreq = 16000;
- }
- else {
- aecpc->splitSampFreq = sampFreq;
- }
-
- aecpc->skewFrCtr = 0;
- aecpc->activity = 0;
-
- aecpc->delayCtr = 0;
-
- aecpc->sum = 0;
- aecpc->counter = 0;
- aecpc->checkBuffSize = 1;
- aecpc->firstVal = 0;
-
- aecpc->ECstartup = 1;
- aecpc->bufSizeStart = 0;
- aecpc->checkBufSizeCtr = 0;
- aecpc->filtDelay = 0;
- aecpc->timeForDelayChange = 0;
- aecpc->knownDelay = 0;
- aecpc->lastDelayDiff = 0;
-
- aecpc->skew = 0;
- aecpc->resample = kAecFalse;
- aecpc->highSkewCtr = 0;
- aecpc->sampFactor = (aecpc->scSampFreq * 1.0f) / aecpc->splitSampFreq;
-
- // Default settings.
- aecConfig.nlpMode = kAecNlpModerate;
- aecConfig.skewMode = kAecFalse;
- aecConfig.metricsMode = kAecFalse;
- aecConfig.delay_logging = kAecFalse;
-
- if (WebRtcAec_set_config(aecpc, aecConfig) == -1) {
- aecpc->lastError = AEC_UNSPECIFIED_ERROR;
- return -1;
- }
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- if (WebRtc_InitBuffer(aecpc->far_pre_buf_s16) == -1) {
- aecpc->lastError = AEC_UNSPECIFIED_ERROR;
- return -1;
- }
- WebRtc_MoveReadPtr(aecpc->far_pre_buf_s16, -PART_LEN); // Start overlap.
-#endif
-
- return 0;
-}
-
-// only buffer L band for farend
-WebRtc_Word32 WebRtcAec_BufferFarend(void *aecInst, const WebRtc_Word16 *farend,
- WebRtc_Word16 nrOfSamples)
-{
- aecpc_t *aecpc = aecInst;
- WebRtc_Word32 retVal = 0;
- int newNrOfSamples = (int) nrOfSamples;
- short newFarend[MAX_RESAMP_LEN];
- const int16_t* farend_ptr = farend;
- float tmp_farend[MAX_RESAMP_LEN];
- const float* farend_float = tmp_farend;
- float skew;
- int i = 0;
-
- if (aecpc == NULL) {
- return -1;
- }
-
- if (farend == NULL) {
- aecpc->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (aecpc->initFlag != initCheck) {
- aecpc->lastError = AEC_UNINITIALIZED_ERROR;
- return -1;
- }
-
- // number of samples == 160 for SWB input
- if (nrOfSamples != 80 && nrOfSamples != 160) {
- aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
- return -1;
- }
-
- skew = aecpc->skew;
-
- if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
- // Resample and get a new number of samples
- newNrOfSamples = WebRtcAec_ResampleLinear(aecpc->resampler,
- farend,
- nrOfSamples,
- skew,
- newFarend);
- farend_ptr = (const int16_t*) newFarend;
- }
-
- aecpc->aec->system_delay += newNrOfSamples;
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- WebRtc_WriteBuffer(aecpc->far_pre_buf_s16, farend_ptr,
- (size_t) newNrOfSamples);
-#endif
- // Cast to float and write the time-domain data to |far_pre_buf|.
- for (i = 0; i < newNrOfSamples; i++) {
- tmp_farend[i] = (float) farend_ptr[i];
- }
- WebRtc_WriteBuffer(aecpc->far_pre_buf, farend_float,
- (size_t) newNrOfSamples);
-
- // Transform to frequency domain if we have enough data.
- while (WebRtc_available_read(aecpc->far_pre_buf) >= PART_LEN2) {
- // We have enough data to pass to the FFT, hence read PART_LEN2 samples.
- WebRtc_ReadBuffer(aecpc->far_pre_buf, (void**) &farend_float, tmp_farend,
- PART_LEN2);
-
- WebRtcAec_BufferFarendPartition(aecpc->aec, farend_float);
-
- // Rewind |far_pre_buf| PART_LEN samples for overlap before continuing.
- WebRtc_MoveReadPtr(aecpc->far_pre_buf, -PART_LEN);
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- WebRtc_ReadBuffer(aecpc->far_pre_buf_s16, (void**) &farend_ptr, newFarend,
- PART_LEN2);
- WebRtc_WriteBuffer(aecpc->aec->far_time_buf, &farend_ptr[PART_LEN], 1);
- WebRtc_MoveReadPtr(aecpc->far_pre_buf_s16, -PART_LEN);
-#endif
- }
-
- return retVal;
-}
-
-WebRtc_Word32 WebRtcAec_Process(void *aecInst, const WebRtc_Word16 *nearend,
- const WebRtc_Word16 *nearendH, WebRtc_Word16 *out, WebRtc_Word16 *outH,
- WebRtc_Word16 nrOfSamples, WebRtc_Word16 msInSndCardBuf, WebRtc_Word32 skew)
-{
- aecpc_t *aecpc = aecInst;
- WebRtc_Word32 retVal = 0;
- short i;
- short nBlocks10ms;
- short nFrames;
- // Limit resampling to doubling/halving of signal
- const float minSkewEst = -0.5f;
- const float maxSkewEst = 1.0f;
-
- if (aecpc == NULL) {
- return -1;
- }
-
- if (nearend == NULL) {
- aecpc->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (out == NULL) {
- aecpc->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (aecpc->initFlag != initCheck) {
- aecpc->lastError = AEC_UNINITIALIZED_ERROR;
- return -1;
- }
-
- // number of samples == 160 for SWB input
- if (nrOfSamples != 80 && nrOfSamples != 160) {
- aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
- return -1;
- }
-
- // Check for valid pointers based on sampling rate
- if (aecpc->sampFreq == 32000 && nearendH == NULL) {
- aecpc->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (msInSndCardBuf < 0) {
- msInSndCardBuf = 0;
- aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
- retVal = -1;
- }
- else if (msInSndCardBuf > 500) {
- msInSndCardBuf = 500;
- aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
- retVal = -1;
- }
- // TODO(andrew): we need to investigate if this +10 is really wanted.
- msInSndCardBuf += 10;
- aecpc->msInSndCardBuf = msInSndCardBuf;
-
- if (aecpc->skewMode == kAecTrue) {
- if (aecpc->skewFrCtr < 25) {
- aecpc->skewFrCtr++;
- }
- else {
- retVal = WebRtcAec_GetSkew(aecpc->resampler, skew, &aecpc->skew);
- if (retVal == -1) {
- aecpc->skew = 0;
- aecpc->lastError = AEC_BAD_PARAMETER_WARNING;
- }
-
- aecpc->skew /= aecpc->sampFactor*nrOfSamples;
-
- if (aecpc->skew < 1.0e-3 && aecpc->skew > -1.0e-3) {
- aecpc->resample = kAecFalse;
- }
- else {
- aecpc->resample = kAecTrue;
- }
-
- if (aecpc->skew < minSkewEst) {
- aecpc->skew = minSkewEst;
- }
- else if (aecpc->skew > maxSkewEst) {
- aecpc->skew = maxSkewEst;
- }
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- fwrite(&aecpc->skew, sizeof(aecpc->skew), 1, aecpc->skewFile);
-#endif
- }
- }
-
- nFrames = nrOfSamples / FRAME_LEN;
- nBlocks10ms = nFrames / aecpc->aec->mult;
-
- if (aecpc->ECstartup) {
- if (nearend != out) {
- // Only needed if they don't already point to the same place.
- memcpy(out, nearend, sizeof(short) * nrOfSamples);
- }
-
- // The AEC is in the start up mode
- // AEC is disabled until the system delay is OK
-
- // Mechanism to ensure that the system delay is reasonably stable.
- if (aecpc->checkBuffSize) {
- aecpc->checkBufSizeCtr++;
- // Before we fill up the far-end buffer we require the system delay
- // to be stable (+/-8 ms) compared to the first value. This
- // comparison is made during the following 6 consecutive 10 ms
- // blocks. If it seems to be stable then we start to fill up the
- // far-end buffer.
- if (aecpc->counter == 0) {
- aecpc->firstVal = aecpc->msInSndCardBuf;
- aecpc->sum = 0;
- }
-
- if (abs(aecpc->firstVal - aecpc->msInSndCardBuf) <
- WEBRTC_SPL_MAX(0.2 * aecpc->msInSndCardBuf, sampMsNb)) {
- aecpc->sum += aecpc->msInSndCardBuf;
- aecpc->counter++;
- }
- else {
- aecpc->counter = 0;
- }
-
- if (aecpc->counter * nBlocks10ms >= 6) {
- // The far-end buffer size is determined in partitions of
- // PART_LEN samples. Use 75% of the average value of the system
- // delay as buffer size to start with.
- aecpc->bufSizeStart = WEBRTC_SPL_MIN((3 * aecpc->sum *
- aecpc->aec->mult * 8) / (4 * aecpc->counter * PART_LEN),
- kMaxBufSizeStart);
- // Buffer size has now been determined.
- aecpc->checkBuffSize = 0;
- }
-
- if (aecpc->checkBufSizeCtr * nBlocks10ms > 50) {
- // For really bad systems, don't disable the echo canceller for
- // more than 0.5 sec.
- aecpc->bufSizeStart = WEBRTC_SPL_MIN((aecpc->msInSndCardBuf *
- aecpc->aec->mult * 3) / 40, kMaxBufSizeStart);
- aecpc->checkBuffSize = 0;
- }
- }
-
- // If |checkBuffSize| changed in the if-statement above.
- if (!aecpc->checkBuffSize) {
- // The system delay is now reasonably stable (or has been unstable
- // for too long). When the far-end buffer is filled with
- // approximately the same amount of data as reported by the system
- // we end the startup phase.
- int overhead_elements = aecpc->aec->system_delay / PART_LEN -
- aecpc->bufSizeStart;
- if (overhead_elements == 0) {
- // Enable the AEC
- aecpc->ECstartup = 0;
- } else if (overhead_elements > 0) {
- WebRtc_MoveReadPtr(aecpc->aec->far_buf_windowed,
- overhead_elements);
- WebRtc_MoveReadPtr(aecpc->aec->far_buf, overhead_elements);
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- WebRtc_MoveReadPtr(aecpc->aec->far_time_buf, overhead_elements);
-#endif
- // TODO(bjornv): Do we need a check on how much we actually
- // moved the read pointer? It should always be possible to move
- // the pointer |overhead_elements| since we have only added data
- // to the buffer and no delay compensation nor AEC processing
- // has been done.
- aecpc->aec->system_delay -= overhead_elements * PART_LEN;
-
- // Enable the AEC
- aecpc->ECstartup = 0;
- }
- }
- } else {
- // AEC is enabled.
-
- int out_elements = 0;
-
- EstBufDelay(aecpc);
-
- // Note that 1 frame is supported for NB and 2 frames for WB.
- for (i = 0; i < nFrames; i++) {
- int16_t* out_ptr = NULL;
- int16_t out_tmp[FRAME_LEN];
-
- // Call the AEC.
- WebRtcAec_ProcessFrame(aecpc->aec,
- &nearend[FRAME_LEN * i],
- &nearendH[FRAME_LEN * i],
- aecpc->knownDelay);
- // TODO(bjornv): Re-structure such that we don't have to pass
- // |aecpc->knownDelay| as input. Change name to something like
- // |system_buffer_diff|.
-
- // Stuff the out buffer if we have less than a frame to output.
- // This should only happen for the first frame.
- out_elements = (int) WebRtc_available_read(aecpc->aec->outFrBuf);
- if (out_elements < FRAME_LEN) {
- WebRtc_MoveReadPtr(aecpc->aec->outFrBuf,
- out_elements - FRAME_LEN);
- if (aecpc->sampFreq == 32000) {
- WebRtc_MoveReadPtr(aecpc->aec->outFrBufH,
- out_elements - FRAME_LEN);
- }
- }
-
- // Obtain an output frame.
- WebRtc_ReadBuffer(aecpc->aec->outFrBuf, (void**) &out_ptr,
- out_tmp, FRAME_LEN);
- memcpy(&out[FRAME_LEN * i], out_ptr, sizeof(int16_t) * FRAME_LEN);
- // For H band
- if (aecpc->sampFreq == 32000) {
- WebRtc_ReadBuffer(aecpc->aec->outFrBufH, (void**) &out_ptr,
- out_tmp, FRAME_LEN);
- memcpy(&outH[FRAME_LEN * i], out_ptr,
- sizeof(int16_t) * FRAME_LEN);
- }
- }
- }
-
-#ifdef WEBRTC_AEC_DEBUG_DUMP
- {
- int16_t far_buf_size_ms = (int16_t) (aecpc->aec->system_delay /
- (sampMsNb * aecpc->aec->mult));
- fwrite(&far_buf_size_ms, 2, 1, aecpc->bufFile);
- fwrite(&(aecpc->knownDelay), sizeof(aecpc->knownDelay), 1, aecpc->delayFile);
- }
-#endif
-
- return retVal;
-}
-
-WebRtc_Word32 WebRtcAec_set_config(void *aecInst, AecConfig config)
-{
- aecpc_t *aecpc = aecInst;
-
- if (aecpc == NULL) {
- return -1;
- }
-
- if (aecpc->initFlag != initCheck) {
- aecpc->lastError = AEC_UNINITIALIZED_ERROR;
- return -1;
- }
-
- if (config.skewMode != kAecFalse && config.skewMode != kAecTrue) {
- aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
- return -1;
- }
- aecpc->skewMode = config.skewMode;
-
- if (config.nlpMode != kAecNlpConservative && config.nlpMode !=
- kAecNlpModerate && config.nlpMode != kAecNlpAggressive) {
- aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
- return -1;
- }
- aecpc->nlpMode = config.nlpMode;
- aecpc->aec->targetSupp = targetSupp[aecpc->nlpMode];
- aecpc->aec->minOverDrive = minOverDrive[aecpc->nlpMode];
-
- if (config.metricsMode != kAecFalse && config.metricsMode != kAecTrue) {
- aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
- return -1;
- }
- aecpc->aec->metricsMode = config.metricsMode;
- if (aecpc->aec->metricsMode == kAecTrue) {
- WebRtcAec_InitMetrics(aecpc->aec);
- }
-
- if (config.delay_logging != kAecFalse && config.delay_logging != kAecTrue) {
- aecpc->lastError = AEC_BAD_PARAMETER_ERROR;
- return -1;
- }
- aecpc->aec->delay_logging_enabled = config.delay_logging;
- if (aecpc->aec->delay_logging_enabled == kAecTrue) {
- memset(aecpc->aec->delay_histogram, 0, sizeof(aecpc->aec->delay_histogram));
- }
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAec_get_config(void *aecInst, AecConfig *config)
-{
- aecpc_t *aecpc = aecInst;
-
- if (aecpc == NULL) {
- return -1;
- }
-
- if (config == NULL) {
- aecpc->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (aecpc->initFlag != initCheck) {
- aecpc->lastError = AEC_UNINITIALIZED_ERROR;
- return -1;
- }
-
- config->nlpMode = aecpc->nlpMode;
- config->skewMode = aecpc->skewMode;
- config->metricsMode = aecpc->aec->metricsMode;
- config->delay_logging = aecpc->aec->delay_logging_enabled;
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAec_get_echo_status(void *aecInst, WebRtc_Word16 *status)
-{
- aecpc_t *aecpc = aecInst;
-
- if (aecpc == NULL) {
- return -1;
- }
-
- if (status == NULL) {
- aecpc->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (aecpc->initFlag != initCheck) {
- aecpc->lastError = AEC_UNINITIALIZED_ERROR;
- return -1;
- }
-
- *status = aecpc->aec->echoState;
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAec_GetMetrics(void *aecInst, AecMetrics *metrics)
-{
- const float upweight = 0.7f;
- float dtmp;
- short stmp;
- aecpc_t *aecpc = aecInst;
-
- if (aecpc == NULL) {
- return -1;
- }
-
- if (metrics == NULL) {
- aecpc->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (aecpc->initFlag != initCheck) {
- aecpc->lastError = AEC_UNINITIALIZED_ERROR;
- return -1;
- }
-
- // ERL
- metrics->erl.instant = (short) aecpc->aec->erl.instant;
-
- if ((aecpc->aec->erl.himean > offsetLevel) && (aecpc->aec->erl.average > offsetLevel)) {
- // Use a mix between regular average and upper part average
- dtmp = upweight * aecpc->aec->erl.himean + (1 - upweight) * aecpc->aec->erl.average;
- metrics->erl.average = (short) dtmp;
- }
- else {
- metrics->erl.average = offsetLevel;
- }
-
- metrics->erl.max = (short) aecpc->aec->erl.max;
-
- if (aecpc->aec->erl.min < (offsetLevel * (-1))) {
- metrics->erl.min = (short) aecpc->aec->erl.min;
- }
- else {
- metrics->erl.min = offsetLevel;
- }
-
- // ERLE
- metrics->erle.instant = (short) aecpc->aec->erle.instant;
-
- if ((aecpc->aec->erle.himean > offsetLevel) && (aecpc->aec->erle.average > offsetLevel)) {
- // Use a mix between regular average and upper part average
- dtmp = upweight * aecpc->aec->erle.himean + (1 - upweight) * aecpc->aec->erle.average;
- metrics->erle.average = (short) dtmp;
- }
- else {
- metrics->erle.average = offsetLevel;
- }
-
- metrics->erle.max = (short) aecpc->aec->erle.max;
-
- if (aecpc->aec->erle.min < (offsetLevel * (-1))) {
- metrics->erle.min = (short) aecpc->aec->erle.min;
- } else {
- metrics->erle.min = offsetLevel;
- }
-
- // RERL
- if ((metrics->erl.average > offsetLevel) && (metrics->erle.average > offsetLevel)) {
- stmp = metrics->erl.average + metrics->erle.average;
- }
- else {
- stmp = offsetLevel;
- }
- metrics->rerl.average = stmp;
-
- // No other statistics needed, but returned for completeness
- metrics->rerl.instant = stmp;
- metrics->rerl.max = stmp;
- metrics->rerl.min = stmp;
-
- // A_NLP
- metrics->aNlp.instant = (short) aecpc->aec->aNlp.instant;
-
- if ((aecpc->aec->aNlp.himean > offsetLevel) && (aecpc->aec->aNlp.average > offsetLevel)) {
- // Use a mix between regular average and upper part average
- dtmp = upweight * aecpc->aec->aNlp.himean + (1 - upweight) * aecpc->aec->aNlp.average;
- metrics->aNlp.average = (short) dtmp;
- }
- else {
- metrics->aNlp.average = offsetLevel;
- }
-
- metrics->aNlp.max = (short) aecpc->aec->aNlp.max;
-
- if (aecpc->aec->aNlp.min < (offsetLevel * (-1))) {
- metrics->aNlp.min = (short) aecpc->aec->aNlp.min;
- }
- else {
- metrics->aNlp.min = offsetLevel;
- }
-
- return 0;
-}
-
-int WebRtcAec_GetDelayMetrics(void* handle, int* median, int* std) {
- aecpc_t* self = handle;
- int i = 0;
- int delay_values = 0;
- int num_delay_values = 0;
- int my_median = 0;
- const int kMsPerBlock = (PART_LEN * 1000) / self->splitSampFreq;
- float l1_norm = 0;
-
- if (self == NULL) {
- return -1;
- }
- if (median == NULL) {
- self->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
- if (std == NULL) {
- self->lastError = AEC_NULL_POINTER_ERROR;
- return -1;
- }
- if (self->initFlag != initCheck) {
- self->lastError = AEC_UNINITIALIZED_ERROR;
- return -1;
- }
- if (self->aec->delay_logging_enabled == 0) {
- // Logging disabled
- self->lastError = AEC_UNSUPPORTED_FUNCTION_ERROR;
- return -1;
- }
-
- // Get number of delay values since last update
- for (i = 0; i < kHistorySizeBlocks; i++) {
- num_delay_values += self->aec->delay_histogram[i];
- }
- if (num_delay_values == 0) {
- // We have no new delay value data. Even though -1 is a valid estimate, it
- // will practically never be used since multiples of |kMsPerBlock| will
- // always be returned.
- *median = -1;
- *std = -1;
- return 0;
- }
-
- delay_values = num_delay_values >> 1; // Start value for median count down
- // Get median of delay values since last update
- for (i = 0; i < kHistorySizeBlocks; i++) {
- delay_values -= self->aec->delay_histogram[i];
- if (delay_values < 0) {
- my_median = i;
- break;
- }
- }
- // Account for lookahead.
- *median = (my_median - kLookaheadBlocks) * kMsPerBlock;
-
- // Calculate the L1 norm, with median value as central moment
- for (i = 0; i < kHistorySizeBlocks; i++) {
- l1_norm += (float) (fabs(i - my_median) * self->aec->delay_histogram[i]);
- }
- *std = (int) (l1_norm / (float) num_delay_values + 0.5f) * kMsPerBlock;
-
- // Reset histogram
- memset(self->aec->delay_histogram, 0, sizeof(self->aec->delay_histogram));
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAec_get_error_code(void *aecInst)
-{
- aecpc_t *aecpc = aecInst;
-
- if (aecpc == NULL) {
- return -1;
- }
-
- return aecpc->lastError;
-}
-
-static int EstBufDelay(aecpc_t* aecpc) {
- int nSampSndCard = aecpc->msInSndCardBuf * sampMsNb * aecpc->aec->mult;
- int current_delay = nSampSndCard - aecpc->aec->system_delay;
- int delay_difference = 0;
-
- // Before we proceed with the delay estimate filtering we:
- // 1) Compensate for the frame that will be read.
- // 2) Compensate for drift resampling.
-
- // 1) Compensating for the frame(s) that will be read/processed.
- current_delay += FRAME_LEN * aecpc->aec->mult;
-
- // 2) Account for resampling frame delay.
- if (aecpc->skewMode == kAecTrue && aecpc->resample == kAecTrue) {
- current_delay -= kResamplingDelay;
- }
-
- aecpc->filtDelay = WEBRTC_SPL_MAX(0, (short) (0.8 * aecpc->filtDelay +
- 0.2 * current_delay));
-
- delay_difference = aecpc->filtDelay - aecpc->knownDelay;
- if (delay_difference > 224) {
- if (aecpc->lastDelayDiff < 96) {
- aecpc->timeForDelayChange = 0;
- } else {
- aecpc->timeForDelayChange++;
- }
- } else if (delay_difference < 96 && aecpc->knownDelay > 0) {
- if (aecpc->lastDelayDiff > 224) {
- aecpc->timeForDelayChange = 0;
- } else {
- aecpc->timeForDelayChange++;
- }
- } else {
- aecpc->timeForDelayChange = 0;
- }
- aecpc->lastDelayDiff = delay_difference;
-
- if (aecpc->timeForDelayChange > 25) {
- aecpc->knownDelay = WEBRTC_SPL_MAX((int) aecpc->filtDelay - 160, 0);
- }
-
- return 0;
-}
diff --git a/trunk/src/modules/audio_processing/aec/include/echo_cancellation.h b/trunk/src/modules/audio_processing/aec/include/echo_cancellation.h
deleted file mode 100644
index a266e84..0000000
--- a/trunk/src/modules/audio_processing/aec/include/echo_cancellation.h
+++ /dev/null
@@ -1,262 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AEC_INCLUDE_ECHO_CANCELLATION_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AEC_INCLUDE_ECHO_CANCELLATION_H_
-
-#include "typedefs.h"
-
-// Errors
-#define AEC_UNSPECIFIED_ERROR 12000
-#define AEC_UNSUPPORTED_FUNCTION_ERROR 12001
-#define AEC_UNINITIALIZED_ERROR 12002
-#define AEC_NULL_POINTER_ERROR 12003
-#define AEC_BAD_PARAMETER_ERROR 12004
-
-// Warnings
-#define AEC_BAD_PARAMETER_WARNING 12050
-
-enum {
- kAecNlpConservative = 0,
- kAecNlpModerate,
- kAecNlpAggressive
-};
-
-enum {
- kAecFalse = 0,
- kAecTrue
-};
-
-typedef struct {
- WebRtc_Word16 nlpMode; // default kAecNlpModerate
- WebRtc_Word16 skewMode; // default kAecFalse
- WebRtc_Word16 metricsMode; // default kAecFalse
- int delay_logging; // default kAecFalse
- //float realSkew;
-} AecConfig;
-
-typedef struct {
- WebRtc_Word16 instant;
- WebRtc_Word16 average;
- WebRtc_Word16 max;
- WebRtc_Word16 min;
-} AecLevel;
-
-typedef struct {
- AecLevel rerl;
- AecLevel erl;
- AecLevel erle;
- AecLevel aNlp;
-} AecMetrics;
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/*
- * Allocates the memory needed by the AEC. The memory needs to be initialized
- * separately using the WebRtcAec_Init() function.
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void **aecInst Pointer to the AEC instance to be created
- * and initialized
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAec_Create(void **aecInst);
-
-/*
- * This function releases the memory allocated by WebRtcAec_Create().
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void *aecInst Pointer to the AEC instance
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAec_Free(void *aecInst);
-
-/*
- * Initializes an AEC instance.
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void *aecInst Pointer to the AEC instance
- * WebRtc_Word32 sampFreq Sampling frequency of data
- * WebRtc_Word32 scSampFreq Soundcard sampling frequency
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAec_Init(void *aecInst,
- WebRtc_Word32 sampFreq,
- WebRtc_Word32 scSampFreq);
-
-/*
- * Inserts an 80 or 160 sample block of data into the farend buffer.
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void *aecInst Pointer to the AEC instance
- * WebRtc_Word16 *farend In buffer containing one frame of
- * farend signal for L band
- * WebRtc_Word16 nrOfSamples Number of samples in farend buffer
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAec_BufferFarend(void *aecInst,
- const WebRtc_Word16 *farend,
- WebRtc_Word16 nrOfSamples);
-
-/*
- * Runs the echo canceller on an 80 or 160 sample blocks of data.
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void *aecInst Pointer to the AEC instance
- * WebRtc_Word16 *nearend In buffer containing one frame of
- * nearend+echo signal for L band
- * WebRtc_Word16 *nearendH In buffer containing one frame of
- * nearend+echo signal for H band
- * WebRtc_Word16 nrOfSamples Number of samples in nearend buffer
- * WebRtc_Word16 msInSndCardBuf Delay estimate for sound card and
- * system buffers
- * WebRtc_Word16 skew Difference between number of samples played
- * and recorded at the soundcard (for clock skew
- * compensation)
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * WebRtc_Word16 *out Out buffer, one frame of processed nearend
- * for L band
- * WebRtc_Word16 *outH Out buffer, one frame of processed nearend
- * for H band
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAec_Process(void *aecInst,
- const WebRtc_Word16 *nearend,
- const WebRtc_Word16 *nearendH,
- WebRtc_Word16 *out,
- WebRtc_Word16 *outH,
- WebRtc_Word16 nrOfSamples,
- WebRtc_Word16 msInSndCardBuf,
- WebRtc_Word32 skew);
-
-/*
- * This function enables the user to set certain parameters on-the-fly.
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void *aecInst Pointer to the AEC instance
- * AecConfig config Config instance that contains all
- * properties to be set
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAec_set_config(void *aecInst, AecConfig config);
-
-/*
- * Gets the on-the-fly paramters.
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void *aecInst Pointer to the AEC instance
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * AecConfig *config Pointer to the config instance that
- * all properties will be written to
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAec_get_config(void *aecInst, AecConfig *config);
-
-/*
- * Gets the current echo status of the nearend signal.
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void *aecInst Pointer to the AEC instance
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * WebRtc_Word16 *status 0: Almost certainly nearend single-talk
- * 1: Might not be neared single-talk
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAec_get_echo_status(void *aecInst, WebRtc_Word16 *status);
-
-/*
- * Gets the current echo metrics for the session.
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void *aecInst Pointer to the AEC instance
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * AecMetrics *metrics Struct which will be filled out with the
- * current echo metrics.
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAec_GetMetrics(void *aecInst, AecMetrics *metrics);
-
-/*
- * Gets the current delay metrics for the session.
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void* handle Pointer to the AEC instance
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * int* median Delay median value.
- * int* std Delay standard deviation.
- *
- * int return 0: OK
- * -1: error
- */
-int WebRtcAec_GetDelayMetrics(void* handle, int* median, int* std);
-
-/*
- * Gets the last error code.
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void *aecInst Pointer to the AEC instance
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return 11000-11100: error code
- */
-WebRtc_Word32 WebRtcAec_get_error_code(void *aecInst);
-
-#ifdef __cplusplus
-}
-#endif
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AEC_INCLUDE_ECHO_CANCELLATION_H_
diff --git a/trunk/src/modules/audio_processing/aecm/Android.mk b/trunk/src/modules/audio_processing/aecm/Android.mk
deleted file mode 100644
index 2d64b85..0000000
--- a/trunk/src/modules/audio_processing/aecm/Android.mk
+++ /dev/null
@@ -1,78 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-#############################
-# Build the non-neon library.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_aecm
-LOCAL_MODULE_TAGS := optional
-LOCAL_SRC_FILES := \
- echo_control_mobile.c \
- aecm_core.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include \
- $(LOCAL_PATH)/../utility \
- $(LOCAL_PATH)/../../.. \
- $(LOCAL_PATH)/../../../common_audio/signal_processing/include \
- $(LOCAL_PATH)/../../../system_wrappers/interface
-
-LOCAL_STATIC_LIBRARIES += libwebrtc_system_wrappers
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
-
-#########################
-# Build the neon library.
-ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_aecm_neon
-LOCAL_MODULE_TAGS := optional
-
-LOCAL_SRC_FILES := aecm_core_neon.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS) \
- -mfpu=neon \
- -mfloat-abi=softfp \
- -flax-vector-conversions
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include \
- $(LOCAL_PATH)/../../.. \
- $(LOCAL_PATH)/../../../common_audio/signal_processing/include
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
-
-endif # ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
diff --git a/trunk/src/modules/audio_processing/aecm/aecm.gypi b/trunk/src/modules/audio_processing/aecm/aecm.gypi
deleted file mode 100644
index 3e4a113..0000000
--- a/trunk/src/modules/audio_processing/aecm/aecm.gypi
+++ /dev/null
@@ -1,35 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'aecm',
- 'type': '<(library)',
- 'dependencies': [
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- 'apm_util'
- ],
- 'include_dirs': [
- 'include',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- ],
- },
- 'sources': [
- 'include/echo_control_mobile.h',
- 'echo_control_mobile.c',
- 'aecm_core.c',
- 'aecm_core.h',
- ],
- },
- ],
-}
diff --git a/trunk/src/modules/audio_processing/aecm/aecm_core.c b/trunk/src/modules/audio_processing/aecm/aecm_core.c
deleted file mode 100644
index 4a51d56..0000000
--- a/trunk/src/modules/audio_processing/aecm/aecm_core.c
+++ /dev/null
@@ -1,2131 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "aecm_core.h"
-
-#include <assert.h>
-#include <stdlib.h>
-
-#include "cpu_features_wrapper.h"
-#include "delay_estimator_wrapper.h"
-#include "echo_control_mobile.h"
-#include "ring_buffer.h"
-#include "typedefs.h"
-
-#ifdef ARM_WINM_LOG
-#include <stdio.h>
-#include <windows.h>
-#endif
-
-#ifdef AEC_DEBUG
-FILE *dfile;
-FILE *testfile;
-#endif
-
-#ifdef _MSC_VER // visual c++
-#define ALIGN8_BEG __declspec(align(8))
-#define ALIGN8_END
-#else // gcc or icc
-#define ALIGN8_BEG
-#define ALIGN8_END __attribute__((aligned(8)))
-#endif
-
-#ifdef AECM_SHORT
-
-// Square root of Hanning window in Q14
-const WebRtc_Word16 WebRtcAecm_kSqrtHanning[] =
-{
- 0, 804, 1606, 2404, 3196, 3981, 4756, 5520,
- 6270, 7005, 7723, 8423, 9102, 9760, 10394, 11003,
- 11585, 12140, 12665, 13160, 13623, 14053, 14449, 14811,
- 15137, 15426, 15679, 15893, 16069, 16207, 16305, 16364,
- 16384
-};
-
-#else
-
-// Square root of Hanning window in Q14
-const ALIGN8_BEG WebRtc_Word16 WebRtcAecm_kSqrtHanning[] ALIGN8_END =
-{
- 0, 399, 798, 1196, 1594, 1990, 2386, 2780, 3172,
- 3562, 3951, 4337, 4720, 5101, 5478, 5853, 6224, 6591, 6954, 7313, 7668, 8019, 8364,
- 8705, 9040, 9370, 9695, 10013, 10326, 10633, 10933, 11227, 11514, 11795, 12068, 12335,
- 12594, 12845, 13089, 13325, 13553, 13773, 13985, 14189, 14384, 14571, 14749, 14918,
- 15079, 15231, 15373, 15506, 15631, 15746, 15851, 15947, 16034, 16111, 16179, 16237,
- 16286, 16325, 16354, 16373, 16384
-};
-
-#endif
-
-//Q15 alpha = 0.99439986968132 const Factor for magnitude approximation
-static const WebRtc_UWord16 kAlpha1 = 32584;
-//Q15 beta = 0.12967166976970 const Factor for magnitude approximation
-static const WebRtc_UWord16 kBeta1 = 4249;
-//Q15 alpha = 0.94234827210087 const Factor for magnitude approximation
-static const WebRtc_UWord16 kAlpha2 = 30879;
-//Q15 beta = 0.33787806009150 const Factor for magnitude approximation
-static const WebRtc_UWord16 kBeta2 = 11072;
-//Q15 alpha = 0.82247698684306 const Factor for magnitude approximation
-static const WebRtc_UWord16 kAlpha3 = 26951;
-//Q15 beta = 0.57762063060713 const Factor for magnitude approximation
-static const WebRtc_UWord16 kBeta3 = 18927;
-
-// Initialization table for echo channel in 8 kHz
-static const WebRtc_Word16 kChannelStored8kHz[PART_LEN1] = {
- 2040, 1815, 1590, 1498, 1405, 1395, 1385, 1418,
- 1451, 1506, 1562, 1644, 1726, 1804, 1882, 1918,
- 1953, 1982, 2010, 2025, 2040, 2034, 2027, 2021,
- 2014, 1997, 1980, 1925, 1869, 1800, 1732, 1683,
- 1635, 1604, 1572, 1545, 1517, 1481, 1444, 1405,
- 1367, 1331, 1294, 1270, 1245, 1239, 1233, 1247,
- 1260, 1282, 1303, 1338, 1373, 1407, 1441, 1470,
- 1499, 1524, 1549, 1565, 1582, 1601, 1621, 1649,
- 1676
-};
-
-// Initialization table for echo channel in 16 kHz
-static const WebRtc_Word16 kChannelStored16kHz[PART_LEN1] = {
- 2040, 1590, 1405, 1385, 1451, 1562, 1726, 1882,
- 1953, 2010, 2040, 2027, 2014, 1980, 1869, 1732,
- 1635, 1572, 1517, 1444, 1367, 1294, 1245, 1233,
- 1260, 1303, 1373, 1441, 1499, 1549, 1582, 1621,
- 1676, 1741, 1802, 1861, 1921, 1983, 2040, 2102,
- 2170, 2265, 2375, 2515, 2651, 2781, 2922, 3075,
- 3253, 3471, 3738, 3976, 4151, 4258, 4308, 4288,
- 4270, 4253, 4237, 4179, 4086, 3947, 3757, 3484,
- 3153
-};
-
-static const WebRtc_Word16 kCosTable[] = {
- 8192, 8190, 8187, 8180, 8172, 8160, 8147, 8130, 8112,
- 8091, 8067, 8041, 8012, 7982, 7948, 7912, 7874, 7834,
- 7791, 7745, 7697, 7647, 7595, 7540, 7483, 7424, 7362,
- 7299, 7233, 7164, 7094, 7021, 6947, 6870, 6791, 6710,
- 6627, 6542, 6455, 6366, 6275, 6182, 6087, 5991, 5892,
- 5792, 5690, 5586, 5481, 5374, 5265, 5155, 5043, 4930,
- 4815, 4698, 4580, 4461, 4341, 4219, 4096, 3971, 3845,
- 3719, 3591, 3462, 3331, 3200, 3068, 2935, 2801, 2667,
- 2531, 2395, 2258, 2120, 1981, 1842, 1703, 1563, 1422,
- 1281, 1140, 998, 856, 713, 571, 428, 285, 142,
- 0, -142, -285, -428, -571, -713, -856, -998, -1140,
- -1281, -1422, -1563, -1703, -1842, -1981, -2120, -2258, -2395,
- -2531, -2667, -2801, -2935, -3068, -3200, -3331, -3462, -3591,
- -3719, -3845, -3971, -4095, -4219, -4341, -4461, -4580, -4698,
- -4815, -4930, -5043, -5155, -5265, -5374, -5481, -5586, -5690,
- -5792, -5892, -5991, -6087, -6182, -6275, -6366, -6455, -6542,
- -6627, -6710, -6791, -6870, -6947, -7021, -7094, -7164, -7233,
- -7299, -7362, -7424, -7483, -7540, -7595, -7647, -7697, -7745,
- -7791, -7834, -7874, -7912, -7948, -7982, -8012, -8041, -8067,
- -8091, -8112, -8130, -8147, -8160, -8172, -8180, -8187, -8190,
- -8191, -8190, -8187, -8180, -8172, -8160, -8147, -8130, -8112,
- -8091, -8067, -8041, -8012, -7982, -7948, -7912, -7874, -7834,
- -7791, -7745, -7697, -7647, -7595, -7540, -7483, -7424, -7362,
- -7299, -7233, -7164, -7094, -7021, -6947, -6870, -6791, -6710,
- -6627, -6542, -6455, -6366, -6275, -6182, -6087, -5991, -5892,
- -5792, -5690, -5586, -5481, -5374, -5265, -5155, -5043, -4930,
- -4815, -4698, -4580, -4461, -4341, -4219, -4096, -3971, -3845,
- -3719, -3591, -3462, -3331, -3200, -3068, -2935, -2801, -2667,
- -2531, -2395, -2258, -2120, -1981, -1842, -1703, -1563, -1422,
- -1281, -1140, -998, -856, -713, -571, -428, -285, -142,
- 0, 142, 285, 428, 571, 713, 856, 998, 1140,
- 1281, 1422, 1563, 1703, 1842, 1981, 2120, 2258, 2395,
- 2531, 2667, 2801, 2935, 3068, 3200, 3331, 3462, 3591,
- 3719, 3845, 3971, 4095, 4219, 4341, 4461, 4580, 4698,
- 4815, 4930, 5043, 5155, 5265, 5374, 5481, 5586, 5690,
- 5792, 5892, 5991, 6087, 6182, 6275, 6366, 6455, 6542,
- 6627, 6710, 6791, 6870, 6947, 7021, 7094, 7164, 7233,
- 7299, 7362, 7424, 7483, 7540, 7595, 7647, 7697, 7745,
- 7791, 7834, 7874, 7912, 7948, 7982, 8012, 8041, 8067,
- 8091, 8112, 8130, 8147, 8160, 8172, 8180, 8187, 8190
-};
-
-static const WebRtc_Word16 kSinTable[] = {
- 0, 142, 285, 428, 571, 713, 856, 998,
- 1140, 1281, 1422, 1563, 1703, 1842, 1981, 2120,
- 2258, 2395, 2531, 2667, 2801, 2935, 3068, 3200,
- 3331, 3462, 3591, 3719, 3845, 3971, 4095, 4219,
- 4341, 4461, 4580, 4698, 4815, 4930, 5043, 5155,
- 5265, 5374, 5481, 5586, 5690, 5792, 5892, 5991,
- 6087, 6182, 6275, 6366, 6455, 6542, 6627, 6710,
- 6791, 6870, 6947, 7021, 7094, 7164, 7233, 7299,
- 7362, 7424, 7483, 7540, 7595, 7647, 7697, 7745,
- 7791, 7834, 7874, 7912, 7948, 7982, 8012, 8041,
- 8067, 8091, 8112, 8130, 8147, 8160, 8172, 8180,
- 8187, 8190, 8191, 8190, 8187, 8180, 8172, 8160,
- 8147, 8130, 8112, 8091, 8067, 8041, 8012, 7982,
- 7948, 7912, 7874, 7834, 7791, 7745, 7697, 7647,
- 7595, 7540, 7483, 7424, 7362, 7299, 7233, 7164,
- 7094, 7021, 6947, 6870, 6791, 6710, 6627, 6542,
- 6455, 6366, 6275, 6182, 6087, 5991, 5892, 5792,
- 5690, 5586, 5481, 5374, 5265, 5155, 5043, 4930,
- 4815, 4698, 4580, 4461, 4341, 4219, 4096, 3971,
- 3845, 3719, 3591, 3462, 3331, 3200, 3068, 2935,
- 2801, 2667, 2531, 2395, 2258, 2120, 1981, 1842,
- 1703, 1563, 1422, 1281, 1140, 998, 856, 713,
- 571, 428, 285, 142, 0, -142, -285, -428,
- -571, -713, -856, -998, -1140, -1281, -1422, -1563,
- -1703, -1842, -1981, -2120, -2258, -2395, -2531, -2667,
- -2801, -2935, -3068, -3200, -3331, -3462, -3591, -3719,
- -3845, -3971, -4095, -4219, -4341, -4461, -4580, -4698,
- -4815, -4930, -5043, -5155, -5265, -5374, -5481, -5586,
- -5690, -5792, -5892, -5991, -6087, -6182, -6275, -6366,
- -6455, -6542, -6627, -6710, -6791, -6870, -6947, -7021,
- -7094, -7164, -7233, -7299, -7362, -7424, -7483, -7540,
- -7595, -7647, -7697, -7745, -7791, -7834, -7874, -7912,
- -7948, -7982, -8012, -8041, -8067, -8091, -8112, -8130,
- -8147, -8160, -8172, -8180, -8187, -8190, -8191, -8190,
- -8187, -8180, -8172, -8160, -8147, -8130, -8112, -8091,
- -8067, -8041, -8012, -7982, -7948, -7912, -7874, -7834,
- -7791, -7745, -7697, -7647, -7595, -7540, -7483, -7424,
- -7362, -7299, -7233, -7164, -7094, -7021, -6947, -6870,
- -6791, -6710, -6627, -6542, -6455, -6366, -6275, -6182,
- -6087, -5991, -5892, -5792, -5690, -5586, -5481, -5374,
- -5265, -5155, -5043, -4930, -4815, -4698, -4580, -4461,
- -4341, -4219, -4096, -3971, -3845, -3719, -3591, -3462,
- -3331, -3200, -3068, -2935, -2801, -2667, -2531, -2395,
- -2258, -2120, -1981, -1842, -1703, -1563, -1422, -1281,
- -1140, -998, -856, -713, -571, -428, -285, -142
-};
-
-static const WebRtc_Word16 kNoiseEstQDomain = 15;
-static const WebRtc_Word16 kNoiseEstIncCount = 5;
-
-static void ComfortNoise(AecmCore_t* aecm,
- const WebRtc_UWord16* dfa,
- complex16_t* out,
- const WebRtc_Word16* lambda);
-
-static WebRtc_Word16 CalcSuppressionGain(AecmCore_t * const aecm);
-
-// Moves the pointer to the next entry and inserts |far_spectrum| and
-// corresponding Q-domain in its buffer.
-//
-// Inputs:
-// - self : Pointer to the delay estimation instance
-// - far_spectrum : Pointer to the far end spectrum
-// - far_q : Q-domain of far end spectrum
-//
-static void UpdateFarHistory(AecmCore_t* self,
- uint16_t* far_spectrum,
- int far_q) {
- // Get new buffer position
- self->far_history_pos++;
- if (self->far_history_pos >= MAX_DELAY) {
- self->far_history_pos = 0;
- }
- // Update Q-domain buffer
- self->far_q_domains[self->far_history_pos] = far_q;
- // Update far end spectrum buffer
- memcpy(&(self->far_history[self->far_history_pos * PART_LEN1]),
- far_spectrum,
- sizeof(uint16_t) * PART_LEN1);
-}
-
-// Returns a pointer to the far end spectrum aligned to current near end
-// spectrum. The function WebRtc_DelayEstimatorProcessFix(...) should have been
-// called before AlignedFarend(...). Otherwise, you get the pointer to the
-// previous frame. The memory is only valid until the next call of
-// WebRtc_DelayEstimatorProcessFix(...).
-//
-// Inputs:
-// - self : Pointer to the AECM instance.
-// - delay : Current delay estimate.
-//
-// Output:
-// - far_q : The Q-domain of the aligned far end spectrum
-//
-// Return value:
-// - far_spectrum : Pointer to the aligned far end spectrum
-// NULL - Error
-//
-static const uint16_t* AlignedFarend(AecmCore_t* self, int* far_q, int delay) {
- int buffer_position = 0;
- assert(self != NULL);
- buffer_position = self->far_history_pos - delay;
-
- // Check buffer position
- if (buffer_position < 0) {
- buffer_position += MAX_DELAY;
- }
- // Get Q-domain
- *far_q = self->far_q_domains[buffer_position];
- // Return far end spectrum
- return &(self->far_history[buffer_position * PART_LEN1]);
-}
-
-#ifdef ARM_WINM_LOG
-HANDLE logFile = NULL;
-#endif
-
-// Declare function pointers.
-CalcLinearEnergies WebRtcAecm_CalcLinearEnergies;
-StoreAdaptiveChannel WebRtcAecm_StoreAdaptiveChannel;
-ResetAdaptiveChannel WebRtcAecm_ResetAdaptiveChannel;
-WindowAndFFT WebRtcAecm_WindowAndFFT;
-InverseFFTAndWindow WebRtcAecm_InverseFFTAndWindow;
-
-int WebRtcAecm_CreateCore(AecmCore_t **aecmInst)
-{
- AecmCore_t *aecm = malloc(sizeof(AecmCore_t));
- *aecmInst = aecm;
- if (aecm == NULL)
- {
- return -1;
- }
-
- if (WebRtc_CreateBuffer(&aecm->farFrameBuf, FRAME_LEN + PART_LEN,
- sizeof(int16_t)) == -1)
- {
- WebRtcAecm_FreeCore(aecm);
- aecm = NULL;
- return -1;
- }
-
- if (WebRtc_CreateBuffer(&aecm->nearNoisyFrameBuf, FRAME_LEN + PART_LEN,
- sizeof(int16_t)) == -1)
- {
- WebRtcAecm_FreeCore(aecm);
- aecm = NULL;
- return -1;
- }
-
- if (WebRtc_CreateBuffer(&aecm->nearCleanFrameBuf, FRAME_LEN + PART_LEN,
- sizeof(int16_t)) == -1)
- {
- WebRtcAecm_FreeCore(aecm);
- aecm = NULL;
- return -1;
- }
-
- if (WebRtc_CreateBuffer(&aecm->outFrameBuf, FRAME_LEN + PART_LEN,
- sizeof(int16_t)) == -1)
- {
- WebRtcAecm_FreeCore(aecm);
- aecm = NULL;
- return -1;
- }
-
- if (WebRtc_CreateDelayEstimator(&aecm->delay_estimator,
- PART_LEN1,
- MAX_DELAY,
- 0) == -1) {
- WebRtcAecm_FreeCore(aecm);
- aecm = NULL;
- return -1;
- }
-
- // Init some aecm pointers. 16 and 32 byte alignment is only necessary
- // for Neon code currently.
- aecm->xBuf = (WebRtc_Word16*) (((uintptr_t)aecm->xBuf_buf + 31) & ~ 31);
- aecm->dBufClean = (WebRtc_Word16*) (((uintptr_t)aecm->dBufClean_buf + 31) & ~ 31);
- aecm->dBufNoisy = (WebRtc_Word16*) (((uintptr_t)aecm->dBufNoisy_buf + 31) & ~ 31);
- aecm->outBuf = (WebRtc_Word16*) (((uintptr_t)aecm->outBuf_buf + 15) & ~ 15);
- aecm->channelStored = (WebRtc_Word16*) (((uintptr_t)
- aecm->channelStored_buf + 15) & ~ 15);
- aecm->channelAdapt16 = (WebRtc_Word16*) (((uintptr_t)
- aecm->channelAdapt16_buf + 15) & ~ 15);
- aecm->channelAdapt32 = (WebRtc_Word32*) (((uintptr_t)
- aecm->channelAdapt32_buf + 31) & ~ 31);
-
- return 0;
-}
-
-void WebRtcAecm_InitEchoPathCore(AecmCore_t* aecm, const WebRtc_Word16* echo_path)
-{
- int i = 0;
-
- // Reset the stored channel
- memcpy(aecm->channelStored, echo_path, sizeof(WebRtc_Word16) * PART_LEN1);
- // Reset the adapted channels
- memcpy(aecm->channelAdapt16, echo_path, sizeof(WebRtc_Word16) * PART_LEN1);
- for (i = 0; i < PART_LEN1; i++)
- {
- aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
- (WebRtc_Word32)(aecm->channelAdapt16[i]), 16);
- }
-
- // Reset channel storing variables
- aecm->mseAdaptOld = 1000;
- aecm->mseStoredOld = 1000;
- aecm->mseThreshold = WEBRTC_SPL_WORD32_MAX;
- aecm->mseChannelCount = 0;
-}
-
-static void WindowAndFFTC(WebRtc_Word16* fft,
- const WebRtc_Word16* time_signal,
- complex16_t* freq_signal,
- int time_signal_scaling)
-{
- int i, j;
-
- memset(fft, 0, sizeof(WebRtc_Word16) * PART_LEN4);
- // FFT of signal
- for (i = 0, j = 0; i < PART_LEN; i++, j += 2)
- {
- // Window time domain signal and insert into real part of
- // transformation array |fft|
- fft[j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
- (time_signal[i] << time_signal_scaling),
- WebRtcAecm_kSqrtHanning[i],
- 14);
- fft[PART_LEN2 + j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
- (time_signal[i + PART_LEN] << time_signal_scaling),
- WebRtcAecm_kSqrtHanning[PART_LEN - i],
- 14);
- // Inserting zeros in imaginary parts not necessary since we
- // initialized the array with all zeros
- }
-
- WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
- WebRtcSpl_ComplexFFT(fft, PART_LEN_SHIFT, 1);
-
- // Take only the first PART_LEN2 samples
- for (i = 0, j = 0; j < PART_LEN2; i += 1, j += 2)
- {
- freq_signal[i].real = fft[j];
-
- // The imaginary part has to switch sign
- freq_signal[i].imag = - fft[j+1];
- }
-}
-
-static void InverseFFTAndWindowC(AecmCore_t* aecm,
- WebRtc_Word16* fft,
- complex16_t* efw,
- WebRtc_Word16* output,
- const WebRtc_Word16* nearendClean)
-{
- int i, j, outCFFT;
- WebRtc_Word32 tmp32no1;
-
- // Synthesis
- for (i = 1; i < PART_LEN; i++)
- {
- j = WEBRTC_SPL_LSHIFT_W32(i, 1);
- fft[j] = efw[i].real;
-
- // mirrored data, even
- fft[PART_LEN4 - j] = efw[i].real;
- fft[j + 1] = -efw[i].imag;
-
- //mirrored data, odd
- fft[PART_LEN4 - (j - 1)] = efw[i].imag;
- }
- fft[0] = efw[0].real;
- fft[1] = -efw[0].imag;
-
- fft[PART_LEN2] = efw[PART_LEN].real;
- fft[PART_LEN2 + 1] = -efw[PART_LEN].imag;
-
- // inverse FFT, result should be scaled with outCFFT
- WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
- outCFFT = WebRtcSpl_ComplexIFFT(fft, PART_LEN_SHIFT, 1);
-
- //take only the real values and scale with outCFFT
- for (i = 0; i < PART_LEN2; i++)
- {
- j = WEBRTC_SPL_LSHIFT_W32(i, 1);
- fft[i] = fft[j];
- }
-
- for (i = 0; i < PART_LEN; i++)
- {
- fft[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
- fft[i],
- WebRtcAecm_kSqrtHanning[i],
- 14);
- tmp32no1 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)fft[i],
- outCFFT - aecm->dfaCleanQDomain);
- fft[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
- tmp32no1 + aecm->outBuf[i],
- WEBRTC_SPL_WORD16_MIN);
- output[i] = fft[i];
-
- tmp32no1 = WEBRTC_SPL_MUL_16_16_RSFT(
- fft[PART_LEN + i],
- WebRtcAecm_kSqrtHanning[PART_LEN - i],
- 14);
- tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1,
- outCFFT - aecm->dfaCleanQDomain);
- aecm->outBuf[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(
- WEBRTC_SPL_WORD16_MAX,
- tmp32no1,
- WEBRTC_SPL_WORD16_MIN);
- }
-
-#ifdef ARM_WINM_LOG_
- // measure tick end
- QueryPerformanceCounter((LARGE_INTEGER*)&end);
- diff__ = ((end - start) * 1000) / (freq/1000);
- milliseconds = (unsigned int)(diff__ & 0xffffffff);
- WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
-#endif
-
- // Copy the current block to the old position (aecm->outBuf is shifted elsewhere)
- memcpy(aecm->xBuf, aecm->xBuf + PART_LEN, sizeof(WebRtc_Word16) * PART_LEN);
- memcpy(aecm->dBufNoisy, aecm->dBufNoisy + PART_LEN, sizeof(WebRtc_Word16) * PART_LEN);
- if (nearendClean != NULL)
- {
- memcpy(aecm->dBufClean, aecm->dBufClean + PART_LEN, sizeof(WebRtc_Word16) * PART_LEN);
- }
-}
-
-static void CalcLinearEnergiesC(AecmCore_t* aecm,
- const WebRtc_UWord16* far_spectrum,
- WebRtc_Word32* echo_est,
- WebRtc_UWord32* far_energy,
- WebRtc_UWord32* echo_energy_adapt,
- WebRtc_UWord32* echo_energy_stored)
-{
- int i;
-
- // Get energy for the delayed far end signal and estimated
- // echo using both stored and adapted channels.
- for (i = 0; i < PART_LEN1; i++)
- {
- echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
- far_spectrum[i]);
- (*far_energy) += (WebRtc_UWord32)(far_spectrum[i]);
- (*echo_energy_adapt) += WEBRTC_SPL_UMUL_16_16(aecm->channelAdapt16[i],
- far_spectrum[i]);
- (*echo_energy_stored) += (WebRtc_UWord32)echo_est[i];
- }
-}
-
-static void StoreAdaptiveChannelC(AecmCore_t* aecm,
- const WebRtc_UWord16* far_spectrum,
- WebRtc_Word32* echo_est)
-{
- int i;
-
- // During startup we store the channel every block.
- memcpy(aecm->channelStored, aecm->channelAdapt16, sizeof(WebRtc_Word16) * PART_LEN1);
- // Recalculate echo estimate
- for (i = 0; i < PART_LEN; i += 4)
- {
- echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
- far_spectrum[i]);
- echo_est[i + 1] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 1],
- far_spectrum[i + 1]);
- echo_est[i + 2] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 2],
- far_spectrum[i + 2]);
- echo_est[i + 3] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i + 3],
- far_spectrum[i + 3]);
- }
- echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i],
- far_spectrum[i]);
-}
-
-static void ResetAdaptiveChannelC(AecmCore_t* aecm)
-{
- int i;
-
- // The stored channel has a significantly lower MSE than the adaptive one for
- // two consecutive calculations. Reset the adaptive channel.
- memcpy(aecm->channelAdapt16, aecm->channelStored,
- sizeof(WebRtc_Word16) * PART_LEN1);
- // Restore the W32 channel
- for (i = 0; i < PART_LEN; i += 4)
- {
- aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
- (WebRtc_Word32)aecm->channelStored[i], 16);
- aecm->channelAdapt32[i + 1] = WEBRTC_SPL_LSHIFT_W32(
- (WebRtc_Word32)aecm->channelStored[i + 1], 16);
- aecm->channelAdapt32[i + 2] = WEBRTC_SPL_LSHIFT_W32(
- (WebRtc_Word32)aecm->channelStored[i + 2], 16);
- aecm->channelAdapt32[i + 3] = WEBRTC_SPL_LSHIFT_W32(
- (WebRtc_Word32)aecm->channelStored[i + 3], 16);
- }
- aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)aecm->channelStored[i], 16);
-}
-
-// WebRtcAecm_InitCore(...)
-//
-// This function initializes the AECM instant created with WebRtcAecm_CreateCore(...)
-// Input:
-// - aecm : Pointer to the Echo Suppression instance
-// - samplingFreq : Sampling Frequency
-//
-// Output:
-// - aecm : Initialized instance
-//
-// Return value : 0 - Ok
-// -1 - Error
-//
-int WebRtcAecm_InitCore(AecmCore_t * const aecm, int samplingFreq)
-{
- int i = 0;
- WebRtc_Word32 tmp32 = PART_LEN1 * PART_LEN1;
- WebRtc_Word16 tmp16 = PART_LEN1;
-
- if (samplingFreq != 8000 && samplingFreq != 16000)
- {
- samplingFreq = 8000;
- return -1;
- }
- // sanity check of sampling frequency
- aecm->mult = (WebRtc_Word16)samplingFreq / 8000;
-
- aecm->farBufWritePos = 0;
- aecm->farBufReadPos = 0;
- aecm->knownDelay = 0;
- aecm->lastKnownDelay = 0;
-
- WebRtc_InitBuffer(aecm->farFrameBuf);
- WebRtc_InitBuffer(aecm->nearNoisyFrameBuf);
- WebRtc_InitBuffer(aecm->nearCleanFrameBuf);
- WebRtc_InitBuffer(aecm->outFrameBuf);
-
- memset(aecm->xBuf_buf, 0, sizeof(aecm->xBuf_buf));
- memset(aecm->dBufClean_buf, 0, sizeof(aecm->dBufClean_buf));
- memset(aecm->dBufNoisy_buf, 0, sizeof(aecm->dBufNoisy_buf));
- memset(aecm->outBuf_buf, 0, sizeof(aecm->outBuf_buf));
-
- aecm->seed = 666;
- aecm->totCount = 0;
-
- if (WebRtc_InitDelayEstimator(aecm->delay_estimator) != 0) {
- return -1;
- }
- // Set far end histories to zero
- memset(aecm->far_history, 0, sizeof(uint16_t) * PART_LEN1 * MAX_DELAY);
- memset(aecm->far_q_domains, 0, sizeof(int) * MAX_DELAY);
- aecm->far_history_pos = MAX_DELAY;
-
- aecm->nlpFlag = 1;
- aecm->fixedDelay = -1;
-
- aecm->dfaCleanQDomain = 0;
- aecm->dfaCleanQDomainOld = 0;
- aecm->dfaNoisyQDomain = 0;
- aecm->dfaNoisyQDomainOld = 0;
-
- memset(aecm->nearLogEnergy, 0, sizeof(aecm->nearLogEnergy));
- aecm->farLogEnergy = 0;
- memset(aecm->echoAdaptLogEnergy, 0, sizeof(aecm->echoAdaptLogEnergy));
- memset(aecm->echoStoredLogEnergy, 0, sizeof(aecm->echoStoredLogEnergy));
-
- // Initialize the echo channels with a stored shape.
- if (samplingFreq == 8000)
- {
- WebRtcAecm_InitEchoPathCore(aecm, kChannelStored8kHz);
- }
- else
- {
- WebRtcAecm_InitEchoPathCore(aecm, kChannelStored16kHz);
- }
-
- memset(aecm->echoFilt, 0, sizeof(aecm->echoFilt));
- memset(aecm->nearFilt, 0, sizeof(aecm->nearFilt));
- aecm->noiseEstCtr = 0;
-
- aecm->cngMode = AecmTrue;
-
- memset(aecm->noiseEstTooLowCtr, 0, sizeof(aecm->noiseEstTooLowCtr));
- memset(aecm->noiseEstTooHighCtr, 0, sizeof(aecm->noiseEstTooHighCtr));
- // Shape the initial noise level to an approximate pink noise.
- for (i = 0; i < (PART_LEN1 >> 1) - 1; i++)
- {
- aecm->noiseEst[i] = (tmp32 << 8);
- tmp16--;
- tmp32 -= (WebRtc_Word32)((tmp16 << 1) + 1);
- }
- for (; i < PART_LEN1; i++)
- {
- aecm->noiseEst[i] = (tmp32 << 8);
- }
-
- aecm->farEnergyMin = WEBRTC_SPL_WORD16_MAX;
- aecm->farEnergyMax = WEBRTC_SPL_WORD16_MIN;
- aecm->farEnergyMaxMin = 0;
- aecm->farEnergyVAD = FAR_ENERGY_MIN; // This prevents false speech detection at the
- // beginning.
- aecm->farEnergyMSE = 0;
- aecm->currentVADValue = 0;
- aecm->vadUpdateCount = 0;
- aecm->firstVAD = 1;
-
- aecm->startupState = 0;
- aecm->supGain = SUPGAIN_DEFAULT;
- aecm->supGainOld = SUPGAIN_DEFAULT;
-
- aecm->supGainErrParamA = SUPGAIN_ERROR_PARAM_A;
- aecm->supGainErrParamD = SUPGAIN_ERROR_PARAM_D;
- aecm->supGainErrParamDiffAB = SUPGAIN_ERROR_PARAM_A - SUPGAIN_ERROR_PARAM_B;
- aecm->supGainErrParamDiffBD = SUPGAIN_ERROR_PARAM_B - SUPGAIN_ERROR_PARAM_D;
-
- assert(PART_LEN % 16 == 0);
-
- // Initialize function pointers.
- WebRtcAecm_WindowAndFFT = WindowAndFFTC;
- WebRtcAecm_InverseFFTAndWindow = InverseFFTAndWindowC;
- WebRtcAecm_CalcLinearEnergies = CalcLinearEnergiesC;
- WebRtcAecm_StoreAdaptiveChannel = StoreAdaptiveChannelC;
- WebRtcAecm_ResetAdaptiveChannel = ResetAdaptiveChannelC;
-
-#ifdef WEBRTC_DETECT_ARM_NEON
- uint64_t features = WebRtc_GetCPUFeaturesARM();
- if ((features & kCPUFeatureNEON) != 0)
- {
- WebRtcAecm_InitNeon();
- }
-#elif defined(WEBRTC_ARCH_ARM_NEON)
- WebRtcAecm_InitNeon();
-#endif
-
- return 0;
-}
-
-// TODO(bjornv): This function is currently not used. Add support for these
-// parameters from a higher level
-int WebRtcAecm_Control(AecmCore_t *aecm, int delay, int nlpFlag)
-{
- aecm->nlpFlag = nlpFlag;
- aecm->fixedDelay = delay;
-
- return 0;
-}
-
-int WebRtcAecm_FreeCore(AecmCore_t *aecm)
-{
- if (aecm == NULL)
- {
- return -1;
- }
-
- WebRtc_FreeBuffer(aecm->farFrameBuf);
- WebRtc_FreeBuffer(aecm->nearNoisyFrameBuf);
- WebRtc_FreeBuffer(aecm->nearCleanFrameBuf);
- WebRtc_FreeBuffer(aecm->outFrameBuf);
-
- WebRtc_FreeDelayEstimator(aecm->delay_estimator);
- free(aecm);
-
- return 0;
-}
-
-int WebRtcAecm_ProcessFrame(AecmCore_t * aecm,
- const WebRtc_Word16 * farend,
- const WebRtc_Word16 * nearendNoisy,
- const WebRtc_Word16 * nearendClean,
- WebRtc_Word16 * out)
-{
- WebRtc_Word16 outBlock_buf[PART_LEN + 8]; // Align buffer to 8-byte boundary.
- WebRtc_Word16* outBlock = (WebRtc_Word16*) (((uintptr_t) outBlock_buf + 15) & ~ 15);
-
- WebRtc_Word16 farFrame[FRAME_LEN];
- const int16_t* out_ptr = NULL;
- int size = 0;
-
- // Buffer the current frame.
- // Fetch an older one corresponding to the delay.
- WebRtcAecm_BufferFarFrame(aecm, farend, FRAME_LEN);
- WebRtcAecm_FetchFarFrame(aecm, farFrame, FRAME_LEN, aecm->knownDelay);
-
- // Buffer the synchronized far and near frames,
- // to pass the smaller blocks individually.
- WebRtc_WriteBuffer(aecm->farFrameBuf, farFrame, FRAME_LEN);
- WebRtc_WriteBuffer(aecm->nearNoisyFrameBuf, nearendNoisy, FRAME_LEN);
- if (nearendClean != NULL)
- {
- WebRtc_WriteBuffer(aecm->nearCleanFrameBuf, nearendClean, FRAME_LEN);
- }
-
- // Process as many blocks as possible.
- while (WebRtc_available_read(aecm->farFrameBuf) >= PART_LEN)
- {
- int16_t far_block[PART_LEN];
- const int16_t* far_block_ptr = NULL;
- int16_t near_noisy_block[PART_LEN];
- const int16_t* near_noisy_block_ptr = NULL;
-
- WebRtc_ReadBuffer(aecm->farFrameBuf, (void**) &far_block_ptr, far_block,
- PART_LEN);
- WebRtc_ReadBuffer(aecm->nearNoisyFrameBuf,
- (void**) &near_noisy_block_ptr,
- near_noisy_block,
- PART_LEN);
- if (nearendClean != NULL)
- {
- int16_t near_clean_block[PART_LEN];
- const int16_t* near_clean_block_ptr = NULL;
-
- WebRtc_ReadBuffer(aecm->nearCleanFrameBuf,
- (void**) &near_clean_block_ptr,
- near_clean_block,
- PART_LEN);
- if (WebRtcAecm_ProcessBlock(aecm,
- far_block_ptr,
- near_noisy_block_ptr,
- near_clean_block_ptr,
- outBlock) == -1)
- {
- return -1;
- }
- } else
- {
- if (WebRtcAecm_ProcessBlock(aecm,
- far_block_ptr,
- near_noisy_block_ptr,
- NULL,
- outBlock) == -1)
- {
- return -1;
- }
- }
-
- WebRtc_WriteBuffer(aecm->outFrameBuf, outBlock, PART_LEN);
- }
-
- // Stuff the out buffer if we have less than a frame to output.
- // This should only happen for the first frame.
- size = (int) WebRtc_available_read(aecm->outFrameBuf);
- if (size < FRAME_LEN)
- {
- WebRtc_MoveReadPtr(aecm->outFrameBuf, size - FRAME_LEN);
- }
-
- // Obtain an output frame.
- WebRtc_ReadBuffer(aecm->outFrameBuf, (void**) &out_ptr, out, FRAME_LEN);
- if (out_ptr != out) {
- // ReadBuffer() hasn't copied to |out| in this case.
- memcpy(out, out_ptr, FRAME_LEN * sizeof(int16_t));
- }
-
- return 0;
-}
-
-// WebRtcAecm_AsymFilt(...)
-//
-// Performs asymmetric filtering.
-//
-// Inputs:
-// - filtOld : Previous filtered value.
-// - inVal : New input value.
-// - stepSizePos : Step size when we have a positive contribution.
-// - stepSizeNeg : Step size when we have a negative contribution.
-//
-// Output:
-//
-// Return: - Filtered value.
-//
-WebRtc_Word16 WebRtcAecm_AsymFilt(const WebRtc_Word16 filtOld, const WebRtc_Word16 inVal,
- const WebRtc_Word16 stepSizePos,
- const WebRtc_Word16 stepSizeNeg)
-{
- WebRtc_Word16 retVal;
-
- if ((filtOld == WEBRTC_SPL_WORD16_MAX) | (filtOld == WEBRTC_SPL_WORD16_MIN))
- {
- return inVal;
- }
- retVal = filtOld;
- if (filtOld > inVal)
- {
- retVal -= WEBRTC_SPL_RSHIFT_W16(filtOld - inVal, stepSizeNeg);
- } else
- {
- retVal += WEBRTC_SPL_RSHIFT_W16(inVal - filtOld, stepSizePos);
- }
-
- return retVal;
-}
-
-// WebRtcAecm_CalcEnergies(...)
-//
-// This function calculates the log of energies for nearend, farend and estimated
-// echoes. There is also an update of energy decision levels, i.e. internal VAD.
-//
-//
-// @param aecm [i/o] Handle of the AECM instance.
-// @param far_spectrum [in] Pointer to farend spectrum.
-// @param far_q [in] Q-domain of farend spectrum.
-// @param nearEner [in] Near end energy for current block in
-// Q(aecm->dfaQDomain).
-// @param echoEst [out] Estimated echo in Q(xfa_q+RESOLUTION_CHANNEL16).
-//
-void WebRtcAecm_CalcEnergies(AecmCore_t * aecm,
- const WebRtc_UWord16* far_spectrum,
- const WebRtc_Word16 far_q,
- const WebRtc_UWord32 nearEner,
- WebRtc_Word32 * echoEst)
-{
- // Local variables
- WebRtc_UWord32 tmpAdapt = 0;
- WebRtc_UWord32 tmpStored = 0;
- WebRtc_UWord32 tmpFar = 0;
-
- int i;
-
- WebRtc_Word16 zeros, frac;
- WebRtc_Word16 tmp16;
- WebRtc_Word16 increase_max_shifts = 4;
- WebRtc_Word16 decrease_max_shifts = 11;
- WebRtc_Word16 increase_min_shifts = 11;
- WebRtc_Word16 decrease_min_shifts = 3;
- WebRtc_Word16 kLogLowValue = WEBRTC_SPL_LSHIFT_W16(PART_LEN_SHIFT, 7);
-
- // Get log of near end energy and store in buffer
-
- // Shift buffer
- memmove(aecm->nearLogEnergy + 1, aecm->nearLogEnergy,
- sizeof(WebRtc_Word16) * (MAX_BUF_LEN - 1));
-
- // Logarithm of integrated magnitude spectrum (nearEner)
- tmp16 = kLogLowValue;
- if (nearEner)
- {
- zeros = WebRtcSpl_NormU32(nearEner);
- frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(
- (WEBRTC_SPL_LSHIFT_U32(nearEner, zeros) & 0x7FFFFFFF),
- 23);
- // log2 in Q8
- tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
- tmp16 -= WEBRTC_SPL_LSHIFT_W16(aecm->dfaNoisyQDomain, 8);
- }
- aecm->nearLogEnergy[0] = tmp16;
- // END: Get log of near end energy
-
- WebRtcAecm_CalcLinearEnergies(aecm, far_spectrum, echoEst, &tmpFar, &tmpAdapt, &tmpStored);
-
- // Shift buffers
- memmove(aecm->echoAdaptLogEnergy + 1, aecm->echoAdaptLogEnergy,
- sizeof(WebRtc_Word16) * (MAX_BUF_LEN - 1));
- memmove(aecm->echoStoredLogEnergy + 1, aecm->echoStoredLogEnergy,
- sizeof(WebRtc_Word16) * (MAX_BUF_LEN - 1));
-
- // Logarithm of delayed far end energy
- tmp16 = kLogLowValue;
- if (tmpFar)
- {
- zeros = WebRtcSpl_NormU32(tmpFar);
- frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpFar, zeros)
- & 0x7FFFFFFF), 23);
- // log2 in Q8
- tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
- tmp16 -= WEBRTC_SPL_LSHIFT_W16(far_q, 8);
- }
- aecm->farLogEnergy = tmp16;
-
- // Logarithm of estimated echo energy through adapted channel
- tmp16 = kLogLowValue;
- if (tmpAdapt)
- {
- zeros = WebRtcSpl_NormU32(tmpAdapt);
- frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpAdapt, zeros)
- & 0x7FFFFFFF), 23);
- //log2 in Q8
- tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
- tmp16 -= WEBRTC_SPL_LSHIFT_W16(RESOLUTION_CHANNEL16 + far_q, 8);
- }
- aecm->echoAdaptLogEnergy[0] = tmp16;
-
- // Logarithm of estimated echo energy through stored channel
- tmp16 = kLogLowValue;
- if (tmpStored)
- {
- zeros = WebRtcSpl_NormU32(tmpStored);
- frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32((WEBRTC_SPL_LSHIFT_U32(tmpStored, zeros)
- & 0x7FFFFFFF), 23);
- //log2 in Q8
- tmp16 += WEBRTC_SPL_LSHIFT_W16((31 - zeros), 8) + frac;
- tmp16 -= WEBRTC_SPL_LSHIFT_W16(RESOLUTION_CHANNEL16 + far_q, 8);
- }
- aecm->echoStoredLogEnergy[0] = tmp16;
-
- // Update farend energy levels (min, max, vad, mse)
- if (aecm->farLogEnergy > FAR_ENERGY_MIN)
- {
- if (aecm->startupState == 0)
- {
- increase_max_shifts = 2;
- decrease_min_shifts = 2;
- increase_min_shifts = 8;
- }
-
- aecm->farEnergyMin = WebRtcAecm_AsymFilt(aecm->farEnergyMin, aecm->farLogEnergy,
- increase_min_shifts, decrease_min_shifts);
- aecm->farEnergyMax = WebRtcAecm_AsymFilt(aecm->farEnergyMax, aecm->farLogEnergy,
- increase_max_shifts, decrease_max_shifts);
- aecm->farEnergyMaxMin = (aecm->farEnergyMax - aecm->farEnergyMin);
-
- // Dynamic VAD region size
- tmp16 = 2560 - aecm->farEnergyMin;
- if (tmp16 > 0)
- {
- tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16, FAR_ENERGY_VAD_REGION, 9);
- } else
- {
- tmp16 = 0;
- }
- tmp16 += FAR_ENERGY_VAD_REGION;
-
- if ((aecm->startupState == 0) | (aecm->vadUpdateCount > 1024))
- {
- // In startup phase or VAD update halted
- aecm->farEnergyVAD = aecm->farEnergyMin + tmp16;
- } else
- {
- if (aecm->farEnergyVAD > aecm->farLogEnergy)
- {
- aecm->farEnergyVAD += WEBRTC_SPL_RSHIFT_W16(aecm->farLogEnergy +
- tmp16 -
- aecm->farEnergyVAD,
- 6);
- aecm->vadUpdateCount = 0;
- } else
- {
- aecm->vadUpdateCount++;
- }
- }
- // Put MSE threshold higher than VAD
- aecm->farEnergyMSE = aecm->farEnergyVAD + (1 << 8);
- }
-
- // Update VAD variables
- if (aecm->farLogEnergy > aecm->farEnergyVAD)
- {
- if ((aecm->startupState == 0) | (aecm->farEnergyMaxMin > FAR_ENERGY_DIFF))
- {
- // We are in startup or have significant dynamics in input speech level
- aecm->currentVADValue = 1;
- }
- } else
- {
- aecm->currentVADValue = 0;
- }
- if ((aecm->currentVADValue) && (aecm->firstVAD))
- {
- aecm->firstVAD = 0;
- if (aecm->echoAdaptLogEnergy[0] > aecm->nearLogEnergy[0])
- {
- // The estimated echo has higher energy than the near end signal.
- // This means that the initialization was too aggressive. Scale
- // down by a factor 8
- for (i = 0; i < PART_LEN1; i++)
- {
- aecm->channelAdapt16[i] >>= 3;
- }
- // Compensate the adapted echo energy level accordingly.
- aecm->echoAdaptLogEnergy[0] -= (3 << 8);
- aecm->firstVAD = 1;
- }
- }
-}
-
-// WebRtcAecm_CalcStepSize(...)
-//
-// This function calculates the step size used in channel estimation
-//
-//
-// @param aecm [in] Handle of the AECM instance.
-// @param mu [out] (Return value) Stepsize in log2(), i.e. number of shifts.
-//
-//
-WebRtc_Word16 WebRtcAecm_CalcStepSize(AecmCore_t * const aecm)
-{
-
- WebRtc_Word32 tmp32;
- WebRtc_Word16 tmp16;
- WebRtc_Word16 mu = MU_MAX;
-
- // Here we calculate the step size mu used in the
- // following NLMS based Channel estimation algorithm
- if (!aecm->currentVADValue)
- {
- // Far end energy level too low, no channel update
- mu = 0;
- } else if (aecm->startupState > 0)
- {
- if (aecm->farEnergyMin >= aecm->farEnergyMax)
- {
- mu = MU_MIN;
- } else
- {
- tmp16 = (aecm->farLogEnergy - aecm->farEnergyMin);
- tmp32 = WEBRTC_SPL_MUL_16_16(tmp16, MU_DIFF);
- tmp32 = WebRtcSpl_DivW32W16(tmp32, aecm->farEnergyMaxMin);
- mu = MU_MIN - 1 - (WebRtc_Word16)(tmp32);
- // The -1 is an alternative to rounding. This way we get a larger
- // stepsize, so we in some sense compensate for truncation in NLMS
- }
- if (mu < MU_MAX)
- {
- mu = MU_MAX; // Equivalent with maximum step size of 2^-MU_MAX
- }
- }
-
- return mu;
-}
-
-// WebRtcAecm_UpdateChannel(...)
-//
-// This function performs channel estimation. NLMS and decision on channel storage.
-//
-//
-// @param aecm [i/o] Handle of the AECM instance.
-// @param far_spectrum [in] Absolute value of the farend signal in Q(far_q)
-// @param far_q [in] Q-domain of the farend signal
-// @param dfa [in] Absolute value of the nearend signal (Q[aecm->dfaQDomain])
-// @param mu [in] NLMS step size.
-// @param echoEst [i/o] Estimated echo in Q(far_q+RESOLUTION_CHANNEL16).
-//
-void WebRtcAecm_UpdateChannel(AecmCore_t * aecm,
- const WebRtc_UWord16* far_spectrum,
- const WebRtc_Word16 far_q,
- const WebRtc_UWord16 * const dfa,
- const WebRtc_Word16 mu,
- WebRtc_Word32 * echoEst)
-{
-
- WebRtc_UWord32 tmpU32no1, tmpU32no2;
- WebRtc_Word32 tmp32no1, tmp32no2;
- WebRtc_Word32 mseStored;
- WebRtc_Word32 mseAdapt;
-
- int i;
-
- WebRtc_Word16 zerosFar, zerosNum, zerosCh, zerosDfa;
- WebRtc_Word16 shiftChFar, shiftNum, shift2ResChan;
- WebRtc_Word16 tmp16no1;
- WebRtc_Word16 xfaQ, dfaQ;
-
- // This is the channel estimation algorithm. It is base on NLMS but has a variable step
- // length, which was calculated above.
- if (mu)
- {
- for (i = 0; i < PART_LEN1; i++)
- {
- // Determine norm of channel and farend to make sure we don't get overflow in
- // multiplication
- zerosCh = WebRtcSpl_NormU32(aecm->channelAdapt32[i]);
- zerosFar = WebRtcSpl_NormU32((WebRtc_UWord32)far_spectrum[i]);
- if (zerosCh + zerosFar > 31)
- {
- // Multiplication is safe
- tmpU32no1 = WEBRTC_SPL_UMUL_32_16(aecm->channelAdapt32[i],
- far_spectrum[i]);
- shiftChFar = 0;
- } else
- {
- // We need to shift down before multiplication
- shiftChFar = 32 - zerosCh - zerosFar;
- tmpU32no1 = WEBRTC_SPL_UMUL_32_16(
- WEBRTC_SPL_RSHIFT_W32(aecm->channelAdapt32[i], shiftChFar),
- far_spectrum[i]);
- }
- // Determine Q-domain of numerator
- zerosNum = WebRtcSpl_NormU32(tmpU32no1);
- if (dfa[i])
- {
- zerosDfa = WebRtcSpl_NormU32((WebRtc_UWord32)dfa[i]);
- } else
- {
- zerosDfa = 32;
- }
- tmp16no1 = zerosDfa - 2 + aecm->dfaNoisyQDomain -
- RESOLUTION_CHANNEL32 - far_q + shiftChFar;
- if (zerosNum > tmp16no1 + 1)
- {
- xfaQ = tmp16no1;
- dfaQ = zerosDfa - 2;
- } else
- {
- xfaQ = zerosNum - 2;
- dfaQ = RESOLUTION_CHANNEL32 + far_q - aecm->dfaNoisyQDomain -
- shiftChFar + xfaQ;
- }
- // Add in the same Q-domain
- tmpU32no1 = WEBRTC_SPL_SHIFT_W32(tmpU32no1, xfaQ);
- tmpU32no2 = WEBRTC_SPL_SHIFT_W32((WebRtc_UWord32)dfa[i], dfaQ);
- tmp32no1 = (WebRtc_Word32)tmpU32no2 - (WebRtc_Word32)tmpU32no1;
- zerosNum = WebRtcSpl_NormW32(tmp32no1);
- if ((tmp32no1) && (far_spectrum[i] > (CHANNEL_VAD << far_q)))
- {
- //
- // Update is needed
- //
- // This is what we would like to compute
- //
- // tmp32no1 = dfa[i] - (aecm->channelAdapt[i] * far_spectrum[i])
- // tmp32norm = (i + 1)
- // aecm->channelAdapt[i] += (2^mu) * tmp32no1
- // / (tmp32norm * far_spectrum[i])
- //
-
- // Make sure we don't get overflow in multiplication.
- if (zerosNum + zerosFar > 31)
- {
- if (tmp32no1 > 0)
- {
- tmp32no2 = (WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(tmp32no1,
- far_spectrum[i]);
- } else
- {
- tmp32no2 = -(WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(-tmp32no1,
- far_spectrum[i]);
- }
- shiftNum = 0;
- } else
- {
- shiftNum = 32 - (zerosNum + zerosFar);
- if (tmp32no1 > 0)
- {
- tmp32no2 = (WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(
- WEBRTC_SPL_RSHIFT_W32(tmp32no1, shiftNum),
- far_spectrum[i]);
- } else
- {
- tmp32no2 = -(WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(
- WEBRTC_SPL_RSHIFT_W32(-tmp32no1, shiftNum),
- far_spectrum[i]);
- }
- }
- // Normalize with respect to frequency bin
- tmp32no2 = WebRtcSpl_DivW32W16(tmp32no2, i + 1);
- // Make sure we are in the right Q-domain
- shift2ResChan = shiftNum + shiftChFar - xfaQ - mu - ((30 - zerosFar) << 1);
- if (WebRtcSpl_NormW32(tmp32no2) < shift2ResChan)
- {
- tmp32no2 = WEBRTC_SPL_WORD32_MAX;
- } else
- {
- tmp32no2 = WEBRTC_SPL_SHIFT_W32(tmp32no2, shift2ResChan);
- }
- aecm->channelAdapt32[i] = WEBRTC_SPL_ADD_SAT_W32(aecm->channelAdapt32[i],
- tmp32no2);
- if (aecm->channelAdapt32[i] < 0)
- {
- // We can never have negative channel gain
- aecm->channelAdapt32[i] = 0;
- }
- aecm->channelAdapt16[i]
- = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(aecm->channelAdapt32[i], 16);
- }
- }
- }
- // END: Adaptive channel update
-
- // Determine if we should store or restore the channel
- if ((aecm->startupState == 0) & (aecm->currentVADValue))
- {
- // During startup we store the channel every block,
- // and we recalculate echo estimate
- WebRtcAecm_StoreAdaptiveChannel(aecm, far_spectrum, echoEst);
- } else
- {
- if (aecm->farLogEnergy < aecm->farEnergyMSE)
- {
- aecm->mseChannelCount = 0;
- } else
- {
- aecm->mseChannelCount++;
- }
- // Enough data for validation. Store channel if we can.
- if (aecm->mseChannelCount >= (MIN_MSE_COUNT + 10))
- {
- // We have enough data.
- // Calculate MSE of "Adapt" and "Stored" versions.
- // It is actually not MSE, but average absolute error.
- mseStored = 0;
- mseAdapt = 0;
- for (i = 0; i < MIN_MSE_COUNT; i++)
- {
- tmp32no1 = ((WebRtc_Word32)aecm->echoStoredLogEnergy[i]
- - (WebRtc_Word32)aecm->nearLogEnergy[i]);
- tmp32no2 = WEBRTC_SPL_ABS_W32(tmp32no1);
- mseStored += tmp32no2;
-
- tmp32no1 = ((WebRtc_Word32)aecm->echoAdaptLogEnergy[i]
- - (WebRtc_Word32)aecm->nearLogEnergy[i]);
- tmp32no2 = WEBRTC_SPL_ABS_W32(tmp32no1);
- mseAdapt += tmp32no2;
- }
- if (((mseStored << MSE_RESOLUTION) < (MIN_MSE_DIFF * mseAdapt))
- & ((aecm->mseStoredOld << MSE_RESOLUTION) < (MIN_MSE_DIFF
- * aecm->mseAdaptOld)))
- {
- // The stored channel has a significantly lower MSE than the adaptive one for
- // two consecutive calculations. Reset the adaptive channel.
- WebRtcAecm_ResetAdaptiveChannel(aecm);
- } else if (((MIN_MSE_DIFF * mseStored) > (mseAdapt << MSE_RESOLUTION)) & (mseAdapt
- < aecm->mseThreshold) & (aecm->mseAdaptOld < aecm->mseThreshold))
- {
- // The adaptive channel has a significantly lower MSE than the stored one.
- // The MSE for the adaptive channel has also been low for two consecutive
- // calculations. Store the adaptive channel.
- WebRtcAecm_StoreAdaptiveChannel(aecm, far_spectrum, echoEst);
-
- // Update threshold
- if (aecm->mseThreshold == WEBRTC_SPL_WORD32_MAX)
- {
- aecm->mseThreshold = (mseAdapt + aecm->mseAdaptOld);
- } else
- {
- aecm->mseThreshold += WEBRTC_SPL_MUL_16_16_RSFT(mseAdapt
- - WEBRTC_SPL_MUL_16_16_RSFT(aecm->mseThreshold, 5, 3), 205, 8);
- }
-
- }
-
- // Reset counter
- aecm->mseChannelCount = 0;
-
- // Store the MSE values.
- aecm->mseStoredOld = mseStored;
- aecm->mseAdaptOld = mseAdapt;
- }
- }
- // END: Determine if we should store or reset channel estimate.
-}
-
-// CalcSuppressionGain(...)
-//
-// This function calculates the suppression gain that is used in the Wiener filter.
-//
-//
-// @param aecm [i/n] Handle of the AECM instance.
-// @param supGain [out] (Return value) Suppression gain with which to scale the noise
-// level (Q14).
-//
-//
-static WebRtc_Word16 CalcSuppressionGain(AecmCore_t * const aecm)
-{
- WebRtc_Word32 tmp32no1;
-
- WebRtc_Word16 supGain = SUPGAIN_DEFAULT;
- WebRtc_Word16 tmp16no1;
- WebRtc_Word16 dE = 0;
-
- // Determine suppression gain used in the Wiener filter. The gain is based on a mix of far
- // end energy and echo estimation error.
- // Adjust for the far end signal level. A low signal level indicates no far end signal,
- // hence we set the suppression gain to 0
- if (!aecm->currentVADValue)
- {
- supGain = 0;
- } else
- {
- // Adjust for possible double talk. If we have large variations in estimation error we
- // likely have double talk (or poor channel).
- tmp16no1 = (aecm->nearLogEnergy[0] - aecm->echoStoredLogEnergy[0] - ENERGY_DEV_OFFSET);
- dE = WEBRTC_SPL_ABS_W16(tmp16no1);
-
- if (dE < ENERGY_DEV_TOL)
- {
- // Likely no double talk. The better estimation, the more we can suppress signal.
- // Update counters
- if (dE < SUPGAIN_EPC_DT)
- {
- tmp32no1 = WEBRTC_SPL_MUL_16_16(aecm->supGainErrParamDiffAB, dE);
- tmp32no1 += (SUPGAIN_EPC_DT >> 1);
- tmp16no1 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32no1, SUPGAIN_EPC_DT);
- supGain = aecm->supGainErrParamA - tmp16no1;
- } else
- {
- tmp32no1 = WEBRTC_SPL_MUL_16_16(aecm->supGainErrParamDiffBD,
- (ENERGY_DEV_TOL - dE));
- tmp32no1 += ((ENERGY_DEV_TOL - SUPGAIN_EPC_DT) >> 1);
- tmp16no1 = (WebRtc_Word16)WebRtcSpl_DivW32W16(tmp32no1, (ENERGY_DEV_TOL
- - SUPGAIN_EPC_DT));
- supGain = aecm->supGainErrParamD + tmp16no1;
- }
- } else
- {
- // Likely in double talk. Use default value
- supGain = aecm->supGainErrParamD;
- }
- }
-
- if (supGain > aecm->supGainOld)
- {
- tmp16no1 = supGain;
- } else
- {
- tmp16no1 = aecm->supGainOld;
- }
- aecm->supGainOld = supGain;
- if (tmp16no1 < aecm->supGain)
- {
- aecm->supGain += (WebRtc_Word16)((tmp16no1 - aecm->supGain) >> 4);
- } else
- {
- aecm->supGain += (WebRtc_Word16)((tmp16no1 - aecm->supGain) >> 4);
- }
-
- // END: Update suppression gain
-
- return aecm->supGain;
-}
-
-// Transforms a time domain signal into the frequency domain, outputting the
-// complex valued signal, absolute value and sum of absolute values.
-//
-// time_signal [in] Pointer to time domain signal
-// freq_signal_real [out] Pointer to real part of frequency domain array
-// freq_signal_imag [out] Pointer to imaginary part of frequency domain
-// array
-// freq_signal_abs [out] Pointer to absolute value of frequency domain
-// array
-// freq_signal_sum_abs [out] Pointer to the sum of all absolute values in
-// the frequency domain array
-// return value The Q-domain of current frequency values
-//
-static int TimeToFrequencyDomain(const WebRtc_Word16* time_signal,
- complex16_t* freq_signal,
- WebRtc_UWord16* freq_signal_abs,
- WebRtc_UWord32* freq_signal_sum_abs)
-{
- int i = 0;
- int time_signal_scaling = 0;
-
- WebRtc_Word32 tmp32no1;
- WebRtc_Word32 tmp32no2;
-
- // In fft_buf, +16 for 32-byte alignment.
- WebRtc_Word16 fft_buf[PART_LEN4 + 16];
- WebRtc_Word16 *fft = (WebRtc_Word16 *) (((uintptr_t) fft_buf + 31) & ~31);
-
- WebRtc_Word16 tmp16no1;
- WebRtc_Word16 tmp16no2;
-#ifdef AECM_WITH_ABS_APPROX
- WebRtc_Word16 max_value = 0;
- WebRtc_Word16 min_value = 0;
- WebRtc_UWord16 alpha = 0;
- WebRtc_UWord16 beta = 0;
-#endif
-
-#ifdef AECM_DYNAMIC_Q
- tmp16no1 = WebRtcSpl_MaxAbsValueW16(time_signal, PART_LEN2);
- time_signal_scaling = WebRtcSpl_NormW16(tmp16no1);
-#endif
-
- WebRtcAecm_WindowAndFFT(fft, time_signal, freq_signal, time_signal_scaling);
-
- // Extract imaginary and real part, calculate the magnitude for all frequency bins
- freq_signal[0].imag = 0;
- freq_signal[PART_LEN].imag = 0;
- freq_signal[PART_LEN].real = fft[PART_LEN2];
- freq_signal_abs[0] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(
- freq_signal[0].real);
- freq_signal_abs[PART_LEN] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(
- freq_signal[PART_LEN].real);
- (*freq_signal_sum_abs) = (WebRtc_UWord32)(freq_signal_abs[0]) +
- (WebRtc_UWord32)(freq_signal_abs[PART_LEN]);
-
- for (i = 1; i < PART_LEN; i++)
- {
- if (freq_signal[i].real == 0)
- {
- freq_signal_abs[i] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(
- freq_signal[i].imag);
- }
- else if (freq_signal[i].imag == 0)
- {
- freq_signal_abs[i] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(
- freq_signal[i].real);
- }
- else
- {
- // Approximation for magnitude of complex fft output
- // magn = sqrt(real^2 + imag^2)
- // magn ~= alpha * max(|imag|,|real|) + beta * min(|imag|,|real|)
- //
- // The parameters alpha and beta are stored in Q15
-
-#ifdef AECM_WITH_ABS_APPROX
- tmp16no1 = WEBRTC_SPL_ABS_W16(freq_signal[i].real);
- tmp16no2 = WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
-
- if(tmp16no1 > tmp16no2)
- {
- max_value = tmp16no1;
- min_value = tmp16no2;
- } else
- {
- max_value = tmp16no2;
- min_value = tmp16no1;
- }
-
- // Magnitude in Q(-6)
- if ((max_value >> 2) > min_value)
- {
- alpha = kAlpha1;
- beta = kBeta1;
- } else if ((max_value >> 1) > min_value)
- {
- alpha = kAlpha2;
- beta = kBeta2;
- } else
- {
- alpha = kAlpha3;
- beta = kBeta3;
- }
- tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(max_value,
- alpha,
- 15);
- tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(min_value,
- beta,
- 15);
- freq_signal_abs[i] = (WebRtc_UWord16)tmp16no1 +
- (WebRtc_UWord16)tmp16no2;
-#else
-#ifdef WEBRTC_ARCH_ARM_V7A
- __asm __volatile(
- "smulbb %[tmp32no1], %[real], %[real]\n\t"
- "smlabb %[tmp32no2], %[imag], %[imag], %[tmp32no1]\n\t"
- :[tmp32no1]"=r"(tmp32no1),
- [tmp32no2]"=r"(tmp32no2)
- :[real]"r"(freq_signal[i].real),
- [imag]"r"(freq_signal[i].imag)
- );
-#else
- tmp16no1 = WEBRTC_SPL_ABS_W16(freq_signal[i].real);
- tmp16no2 = WEBRTC_SPL_ABS_W16(freq_signal[i].imag);
- tmp32no1 = WEBRTC_SPL_MUL_16_16(tmp16no1, tmp16no1);
- tmp32no2 = WEBRTC_SPL_MUL_16_16(tmp16no2, tmp16no2);
- tmp32no2 = WEBRTC_SPL_ADD_SAT_W32(tmp32no1, tmp32no2);
-#endif // WEBRTC_ARCH_ARM_V7A
- tmp32no1 = WebRtcSpl_SqrtFloor(tmp32no2);
-
- freq_signal_abs[i] = (WebRtc_UWord16)tmp32no1;
-#endif // AECM_WITH_ABS_APPROX
- }
- (*freq_signal_sum_abs) += (WebRtc_UWord32)freq_signal_abs[i];
- }
-
- return time_signal_scaling;
-}
-
-int WebRtcAecm_ProcessBlock(AecmCore_t * aecm,
- const WebRtc_Word16 * farend,
- const WebRtc_Word16 * nearendNoisy,
- const WebRtc_Word16 * nearendClean,
- WebRtc_Word16 * output)
-{
- int i;
-
- WebRtc_UWord32 xfaSum;
- WebRtc_UWord32 dfaNoisySum;
- WebRtc_UWord32 dfaCleanSum;
- WebRtc_UWord32 echoEst32Gained;
- WebRtc_UWord32 tmpU32;
-
- WebRtc_Word32 tmp32no1;
-
- WebRtc_UWord16 xfa[PART_LEN1];
- WebRtc_UWord16 dfaNoisy[PART_LEN1];
- WebRtc_UWord16 dfaClean[PART_LEN1];
- WebRtc_UWord16* ptrDfaClean = dfaClean;
- const WebRtc_UWord16* far_spectrum_ptr = NULL;
-
- // 32 byte aligned buffers (with +8 or +16).
- // TODO (kma): define fft with complex16_t.
- WebRtc_Word16 fft_buf[PART_LEN4 + 2 + 16]; // +2 to make a loop safe.
- WebRtc_Word32 echoEst32_buf[PART_LEN1 + 8];
- WebRtc_Word32 dfw_buf[PART_LEN1 + 8];
- WebRtc_Word32 efw_buf[PART_LEN1 + 8];
-
- WebRtc_Word16* fft = (WebRtc_Word16*) (((uintptr_t) fft_buf + 31) & ~ 31);
- WebRtc_Word32* echoEst32 = (WebRtc_Word32*) (((uintptr_t) echoEst32_buf + 31) & ~ 31);
- complex16_t* dfw = (complex16_t*) (((uintptr_t) dfw_buf + 31) & ~ 31);
- complex16_t* efw = (complex16_t*) (((uintptr_t) efw_buf + 31) & ~ 31);
-
- WebRtc_Word16 hnl[PART_LEN1];
- WebRtc_Word16 numPosCoef = 0;
- WebRtc_Word16 nlpGain = ONE_Q14;
- int delay;
- WebRtc_Word16 tmp16no1;
- WebRtc_Word16 tmp16no2;
- WebRtc_Word16 mu;
- WebRtc_Word16 supGain;
- WebRtc_Word16 zeros32, zeros16;
- WebRtc_Word16 zerosDBufNoisy, zerosDBufClean, zerosXBuf;
- int far_q;
- WebRtc_Word16 resolutionDiff, qDomainDiff;
-
- const int kMinPrefBand = 4;
- const int kMaxPrefBand = 24;
- WebRtc_Word32 avgHnl32 = 0;
-
-#ifdef ARM_WINM_LOG_
- DWORD temp;
- static int flag0 = 0;
- __int64 freq, start, end, diff__;
- unsigned int milliseconds;
-#endif
-
- // Determine startup state. There are three states:
- // (0) the first CONV_LEN blocks
- // (1) another CONV_LEN blocks
- // (2) the rest
-
- if (aecm->startupState < 2)
- {
- aecm->startupState = (aecm->totCount >= CONV_LEN) + (aecm->totCount >= CONV_LEN2);
- }
- // END: Determine startup state
-
- // Buffer near and far end signals
- memcpy(aecm->xBuf + PART_LEN, farend, sizeof(WebRtc_Word16) * PART_LEN);
- memcpy(aecm->dBufNoisy + PART_LEN, nearendNoisy, sizeof(WebRtc_Word16) * PART_LEN);
- if (nearendClean != NULL)
- {
- memcpy(aecm->dBufClean + PART_LEN, nearendClean, sizeof(WebRtc_Word16) * PART_LEN);
- }
-
-#ifdef ARM_WINM_LOG_
- // measure tick start
- QueryPerformanceFrequency((LARGE_INTEGER*)&freq);
- QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
-
- // Transform far end signal from time domain to frequency domain.
- far_q = TimeToFrequencyDomain(aecm->xBuf,
- dfw,
- xfa,
- &xfaSum);
-
- // Transform noisy near end signal from time domain to frequency domain.
- zerosDBufNoisy = TimeToFrequencyDomain(aecm->dBufNoisy,
- dfw,
- dfaNoisy,
- &dfaNoisySum);
- aecm->dfaNoisyQDomainOld = aecm->dfaNoisyQDomain;
- aecm->dfaNoisyQDomain = (WebRtc_Word16)zerosDBufNoisy;
-
-
- if (nearendClean == NULL)
- {
- ptrDfaClean = dfaNoisy;
- aecm->dfaCleanQDomainOld = aecm->dfaNoisyQDomainOld;
- aecm->dfaCleanQDomain = aecm->dfaNoisyQDomain;
- dfaCleanSum = dfaNoisySum;
- } else
- {
- // Transform clean near end signal from time domain to frequency domain.
- zerosDBufClean = TimeToFrequencyDomain(aecm->dBufClean,
- dfw,
- dfaClean,
- &dfaCleanSum);
- aecm->dfaCleanQDomainOld = aecm->dfaCleanQDomain;
- aecm->dfaCleanQDomain = (WebRtc_Word16)zerosDBufClean;
- }
-
-#ifdef ARM_WINM_LOG_
- // measure tick end
- QueryPerformanceCounter((LARGE_INTEGER*)&end);
- diff__ = ((end - start) * 1000) / (freq/1000);
- milliseconds = (unsigned int)(diff__ & 0xffffffff);
- WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
- // measure tick start
- QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
-
- // Get the delay
- // Save far-end history and estimate delay
- UpdateFarHistory(aecm, xfa, far_q);
- delay = WebRtc_DelayEstimatorProcessFix(aecm->delay_estimator,
- xfa,
- dfaNoisy,
- PART_LEN1,
- far_q,
- zerosDBufNoisy);
- if (delay == -1)
- {
- return -1;
- }
- else if (delay == -2)
- {
- // If the delay is unknown, we assume zero.
- // NOTE: this will have to be adjusted if we ever add lookahead.
- delay = 0;
- }
-
- if (aecm->fixedDelay >= 0)
- {
- // Use fixed delay
- delay = aecm->fixedDelay;
- }
-
-#ifdef ARM_WINM_LOG_
- // measure tick end
- QueryPerformanceCounter((LARGE_INTEGER*)&end);
- diff__ = ((end - start) * 1000) / (freq/1000);
- milliseconds = (unsigned int)(diff__ & 0xffffffff);
- WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
- // measure tick start
- QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
- // Get aligned far end spectrum
- far_spectrum_ptr = AlignedFarend(aecm, &far_q, delay);
- zerosXBuf = (WebRtc_Word16) far_q;
- if (far_spectrum_ptr == NULL)
- {
- return -1;
- }
-
- // Calculate log(energy) and update energy threshold levels
- WebRtcAecm_CalcEnergies(aecm,
- far_spectrum_ptr,
- zerosXBuf,
- dfaNoisySum,
- echoEst32);
-
- // Calculate stepsize
- mu = WebRtcAecm_CalcStepSize(aecm);
-
- // Update counters
- aecm->totCount++;
-
- // This is the channel estimation algorithm.
- // It is base on NLMS but has a variable step length, which was calculated above.
- WebRtcAecm_UpdateChannel(aecm, far_spectrum_ptr, zerosXBuf, dfaNoisy, mu, echoEst32);
- supGain = CalcSuppressionGain(aecm);
-
-#ifdef ARM_WINM_LOG_
- // measure tick end
- QueryPerformanceCounter((LARGE_INTEGER*)&end);
- diff__ = ((end - start) * 1000) / (freq/1000);
- milliseconds = (unsigned int)(diff__ & 0xffffffff);
- WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
- // measure tick start
- QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
-
- // Calculate Wiener filter hnl[]
- for (i = 0; i < PART_LEN1; i++)
- {
- // Far end signal through channel estimate in Q8
- // How much can we shift right to preserve resolution
- tmp32no1 = echoEst32[i] - aecm->echoFilt[i];
- aecm->echoFilt[i] += WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(tmp32no1, 50), 8);
-
- zeros32 = WebRtcSpl_NormW32(aecm->echoFilt[i]) + 1;
- zeros16 = WebRtcSpl_NormW16(supGain) + 1;
- if (zeros32 + zeros16 > 16)
- {
- // Multiplication is safe
- // Result in Q(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN+aecm->xfaQDomainBuf[diff])
- echoEst32Gained = WEBRTC_SPL_UMUL_32_16((WebRtc_UWord32)aecm->echoFilt[i],
- (WebRtc_UWord16)supGain);
- resolutionDiff = 14 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
- resolutionDiff += (aecm->dfaCleanQDomain - zerosXBuf);
- } else
- {
- tmp16no1 = 17 - zeros32 - zeros16;
- resolutionDiff = 14 + tmp16no1 - RESOLUTION_CHANNEL16 - RESOLUTION_SUPGAIN;
- resolutionDiff += (aecm->dfaCleanQDomain - zerosXBuf);
- if (zeros32 > tmp16no1)
- {
- echoEst32Gained = WEBRTC_SPL_UMUL_32_16((WebRtc_UWord32)aecm->echoFilt[i],
- (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_W16(supGain,
- tmp16no1)); // Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN-16)
- } else
- {
- // Result in Q-(RESOLUTION_CHANNEL+RESOLUTION_SUPGAIN-16)
- echoEst32Gained = WEBRTC_SPL_UMUL_32_16(
- (WebRtc_UWord32)WEBRTC_SPL_RSHIFT_W32(aecm->echoFilt[i], tmp16no1),
- (WebRtc_UWord16)supGain);
- }
- }
-
- zeros16 = WebRtcSpl_NormW16(aecm->nearFilt[i]);
- if ((zeros16 < (aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld))
- & (aecm->nearFilt[i]))
- {
- tmp16no1 = WEBRTC_SPL_SHIFT_W16(aecm->nearFilt[i], zeros16);
- qDomainDiff = zeros16 - aecm->dfaCleanQDomain + aecm->dfaCleanQDomainOld;
- } else
- {
- tmp16no1 = WEBRTC_SPL_SHIFT_W16(aecm->nearFilt[i],
- aecm->dfaCleanQDomain - aecm->dfaCleanQDomainOld);
- qDomainDiff = 0;
- }
- tmp16no2 = WEBRTC_SPL_SHIFT_W16(ptrDfaClean[i], qDomainDiff);
- tmp32no1 = (WebRtc_Word32)(tmp16no2 - tmp16no1);
- tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 4);
- tmp16no2 += tmp16no1;
- zeros16 = WebRtcSpl_NormW16(tmp16no2);
- if ((tmp16no2) & (-qDomainDiff > zeros16))
- {
- aecm->nearFilt[i] = WEBRTC_SPL_WORD16_MAX;
- } else
- {
- aecm->nearFilt[i] = WEBRTC_SPL_SHIFT_W16(tmp16no2, -qDomainDiff);
- }
-
- // Wiener filter coefficients, resulting hnl in Q14
- if (echoEst32Gained == 0)
- {
- hnl[i] = ONE_Q14;
- } else if (aecm->nearFilt[i] == 0)
- {
- hnl[i] = 0;
- } else
- {
- // Multiply the suppression gain
- // Rounding
- echoEst32Gained += (WebRtc_UWord32)(aecm->nearFilt[i] >> 1);
- tmpU32 = WebRtcSpl_DivU32U16(echoEst32Gained, (WebRtc_UWord16)aecm->nearFilt[i]);
-
- // Current resolution is
- // Q-(RESOLUTION_CHANNEL + RESOLUTION_SUPGAIN - max(0, 17 - zeros16 - zeros32))
- // Make sure we are in Q14
- tmp32no1 = (WebRtc_Word32)WEBRTC_SPL_SHIFT_W32(tmpU32, resolutionDiff);
- if (tmp32no1 > ONE_Q14)
- {
- hnl[i] = 0;
- } else if (tmp32no1 < 0)
- {
- hnl[i] = ONE_Q14;
- } else
- {
- // 1-echoEst/dfa
- hnl[i] = ONE_Q14 - (WebRtc_Word16)tmp32no1;
- if (hnl[i] < 0)
- {
- hnl[i] = 0;
- }
- }
- }
- if (hnl[i])
- {
- numPosCoef++;
- }
- }
- // Only in wideband. Prevent the gain in upper band from being larger than
- // in lower band.
- if (aecm->mult == 2)
- {
- // TODO(bjornv): Investigate if the scaling of hnl[i] below can cause
- // speech distortion in double-talk.
- for (i = 0; i < PART_LEN1; i++)
- {
- hnl[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(hnl[i], hnl[i], 14);
- }
-
- for (i = kMinPrefBand; i <= kMaxPrefBand; i++)
- {
- avgHnl32 += (WebRtc_Word32)hnl[i];
- }
- assert(kMaxPrefBand - kMinPrefBand + 1 > 0);
- avgHnl32 /= (kMaxPrefBand - kMinPrefBand + 1);
-
- for (i = kMaxPrefBand; i < PART_LEN1; i++)
- {
- if (hnl[i] > (WebRtc_Word16)avgHnl32)
- {
- hnl[i] = (WebRtc_Word16)avgHnl32;
- }
- }
- }
-
-#ifdef ARM_WINM_LOG_
- // measure tick end
- QueryPerformanceCounter((LARGE_INTEGER*)&end);
- diff__ = ((end - start) * 1000) / (freq/1000);
- milliseconds = (unsigned int)(diff__ & 0xffffffff);
- WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
- // measure tick start
- QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
-
- // Calculate NLP gain, result is in Q14
- if (aecm->nlpFlag)
- {
- for (i = 0; i < PART_LEN1; i++)
- {
- // Truncate values close to zero and one.
- if (hnl[i] > NLP_COMP_HIGH)
- {
- hnl[i] = ONE_Q14;
- } else if (hnl[i] < NLP_COMP_LOW)
- {
- hnl[i] = 0;
- }
-
- // Remove outliers
- if (numPosCoef < 3)
- {
- nlpGain = 0;
- } else
- {
- nlpGain = ONE_Q14;
- }
-
- // NLP
- if ((hnl[i] == ONE_Q14) && (nlpGain == ONE_Q14))
- {
- hnl[i] = ONE_Q14;
- } else
- {
- hnl[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(hnl[i], nlpGain, 14);
- }
-
- // multiply with Wiener coefficients
- efw[i].real = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real,
- hnl[i], 14));
- efw[i].imag = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag,
- hnl[i], 14));
- }
- }
- else
- {
- // multiply with Wiener coefficients
- for (i = 0; i < PART_LEN1; i++)
- {
- efw[i].real = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].real,
- hnl[i], 14));
- efw[i].imag = (WebRtc_Word16)(WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(dfw[i].imag,
- hnl[i], 14));
- }
- }
-
- if (aecm->cngMode == AecmTrue)
- {
- ComfortNoise(aecm, ptrDfaClean, efw, hnl);
- }
-
-#ifdef ARM_WINM_LOG_
- // measure tick end
- QueryPerformanceCounter((LARGE_INTEGER*)&end);
- diff__ = ((end - start) * 1000) / (freq/1000);
- milliseconds = (unsigned int)(diff__ & 0xffffffff);
- WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
- // measure tick start
- QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#endif
-
- WebRtcAecm_InverseFFTAndWindow(aecm, fft, efw, output, nearendClean);
-
- return 0;
-}
-
-
-// Generate comfort noise and add to output signal.
-//
-// \param[in] aecm Handle of the AECM instance.
-// \param[in] dfa Absolute value of the nearend signal (Q[aecm->dfaQDomain]).
-// \param[in,out] outReal Real part of the output signal (Q[aecm->dfaQDomain]).
-// \param[in,out] outImag Imaginary part of the output signal (Q[aecm->dfaQDomain]).
-// \param[in] lambda Suppression gain with which to scale the noise level (Q14).
-//
-static void ComfortNoise(AecmCore_t* aecm,
- const WebRtc_UWord16* dfa,
- complex16_t* out,
- const WebRtc_Word16* lambda)
-{
- WebRtc_Word16 i;
- WebRtc_Word16 tmp16;
- WebRtc_Word32 tmp32;
-
- WebRtc_Word16 randW16[PART_LEN];
- WebRtc_Word16 uReal[PART_LEN1];
- WebRtc_Word16 uImag[PART_LEN1];
- WebRtc_Word32 outLShift32;
- WebRtc_Word16 noiseRShift16[PART_LEN1];
-
- WebRtc_Word16 shiftFromNearToNoise = kNoiseEstQDomain - aecm->dfaCleanQDomain;
- WebRtc_Word16 minTrackShift;
-
- assert(shiftFromNearToNoise >= 0);
- assert(shiftFromNearToNoise < 16);
-
- if (aecm->noiseEstCtr < 100)
- {
- // Track the minimum more quickly initially.
- aecm->noiseEstCtr++;
- minTrackShift = 6;
- } else
- {
- minTrackShift = 9;
- }
-
- // Estimate noise power.
- for (i = 0; i < PART_LEN1; i++)
- {
-
- // Shift to the noise domain.
- tmp32 = (WebRtc_Word32)dfa[i];
- outLShift32 = WEBRTC_SPL_LSHIFT_W32(tmp32, shiftFromNearToNoise);
-
- if (outLShift32 < aecm->noiseEst[i])
- {
- // Reset "too low" counter
- aecm->noiseEstTooLowCtr[i] = 0;
- // Track the minimum.
- if (aecm->noiseEst[i] < (1 << minTrackShift))
- {
- // For small values, decrease noiseEst[i] every
- // |kNoiseEstIncCount| block. The regular approach below can not
- // go further down due to truncation.
- aecm->noiseEstTooHighCtr[i]++;
- if (aecm->noiseEstTooHighCtr[i] >= kNoiseEstIncCount)
- {
- aecm->noiseEst[i]--;
- aecm->noiseEstTooHighCtr[i] = 0; // Reset the counter
- }
- }
- else
- {
- aecm->noiseEst[i] -= ((aecm->noiseEst[i] - outLShift32) >> minTrackShift);
- }
- } else
- {
- // Reset "too high" counter
- aecm->noiseEstTooHighCtr[i] = 0;
- // Ramp slowly upwards until we hit the minimum again.
- if ((aecm->noiseEst[i] >> 19) > 0)
- {
- // Avoid overflow.
- // Multiplication with 2049 will cause wrap around. Scale
- // down first and then multiply
- aecm->noiseEst[i] >>= 11;
- aecm->noiseEst[i] *= 2049;
- }
- else if ((aecm->noiseEst[i] >> 11) > 0)
- {
- // Large enough for relative increase
- aecm->noiseEst[i] *= 2049;
- aecm->noiseEst[i] >>= 11;
- }
- else
- {
- // Make incremental increases based on size every
- // |kNoiseEstIncCount| block
- aecm->noiseEstTooLowCtr[i]++;
- if (aecm->noiseEstTooLowCtr[i] >= kNoiseEstIncCount)
- {
- aecm->noiseEst[i] += (aecm->noiseEst[i] >> 9) + 1;
- aecm->noiseEstTooLowCtr[i] = 0; // Reset counter
- }
- }
- }
- }
-
- for (i = 0; i < PART_LEN1; i++)
- {
- tmp32 = WEBRTC_SPL_RSHIFT_W32(aecm->noiseEst[i], shiftFromNearToNoise);
- if (tmp32 > 32767)
- {
- tmp32 = 32767;
- aecm->noiseEst[i] = WEBRTC_SPL_LSHIFT_W32(tmp32, shiftFromNearToNoise);
- }
- noiseRShift16[i] = (WebRtc_Word16)tmp32;
-
- tmp16 = ONE_Q14 - lambda[i];
- noiseRShift16[i]
- = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16, noiseRShift16[i], 14);
- }
-
- // Generate a uniform random array on [0 2^15-1].
- WebRtcSpl_RandUArray(randW16, PART_LEN, &aecm->seed);
-
- // Generate noise according to estimated energy.
- uReal[0] = 0; // Reject LF noise.
- uImag[0] = 0;
- for (i = 1; i < PART_LEN1; i++)
- {
- // Get a random index for the cos and sin tables over [0 359].
- tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(359, randW16[i - 1], 15);
-
- // Tables are in Q13.
- uReal[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(noiseRShift16[i],
- kCosTable[tmp16], 13);
- uImag[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(-noiseRShift16[i],
- kSinTable[tmp16], 13);
- }
- uImag[PART_LEN] = 0;
-
-#if (!defined ARM_WINM) && (!defined ARM9E_GCC) && (!defined ANDROID_AECOPT)
- for (i = 0; i < PART_LEN1; i++)
- {
- out[i].real = WEBRTC_SPL_ADD_SAT_W16(out[i].real, uReal[i]);
- out[i].imag = WEBRTC_SPL_ADD_SAT_W16(out[i].imag, uImag[i]);
- }
-#else
- for (i = 0; i < PART_LEN1 -1; )
- {
- out[i].real = WEBRTC_SPL_ADD_SAT_W16(out[i].real, uReal[i]);
- out[i].imag = WEBRTC_SPL_ADD_SAT_W16(out[i].imag, uImag[i]);
- i++;
-
- out[i].real = WEBRTC_SPL_ADD_SAT_W16(out[i].real, uReal[i]);
- out[i].imag = WEBRTC_SPL_ADD_SAT_W16(out[i].imag, uImag[i]);
- i++;
- }
- out[i].real = WEBRTC_SPL_ADD_SAT_W16(out[i].real, uReal[i]);
- out[i].imag = WEBRTC_SPL_ADD_SAT_W16(out[i].imag, uImag[i]);
-#endif
-}
-
-void WebRtcAecm_BufferFarFrame(AecmCore_t* const aecm,
- const WebRtc_Word16* const farend,
- const int farLen)
-{
- int writeLen = farLen, writePos = 0;
-
- // Check if the write position must be wrapped
- while (aecm->farBufWritePos + writeLen > FAR_BUF_LEN)
- {
- // Write to remaining buffer space before wrapping
- writeLen = FAR_BUF_LEN - aecm->farBufWritePos;
- memcpy(aecm->farBuf + aecm->farBufWritePos, farend + writePos,
- sizeof(WebRtc_Word16) * writeLen);
- aecm->farBufWritePos = 0;
- writePos = writeLen;
- writeLen = farLen - writeLen;
- }
-
- memcpy(aecm->farBuf + aecm->farBufWritePos, farend + writePos,
- sizeof(WebRtc_Word16) * writeLen);
- aecm->farBufWritePos += writeLen;
-}
-
-void WebRtcAecm_FetchFarFrame(AecmCore_t * const aecm, WebRtc_Word16 * const farend,
- const int farLen, const int knownDelay)
-{
- int readLen = farLen;
- int readPos = 0;
- int delayChange = knownDelay - aecm->lastKnownDelay;
-
- aecm->farBufReadPos -= delayChange;
-
- // Check if delay forces a read position wrap
- while (aecm->farBufReadPos < 0)
- {
- aecm->farBufReadPos += FAR_BUF_LEN;
- }
- while (aecm->farBufReadPos > FAR_BUF_LEN - 1)
- {
- aecm->farBufReadPos -= FAR_BUF_LEN;
- }
-
- aecm->lastKnownDelay = knownDelay;
-
- // Check if read position must be wrapped
- while (aecm->farBufReadPos + readLen > FAR_BUF_LEN)
- {
-
- // Read from remaining buffer space before wrapping
- readLen = FAR_BUF_LEN - aecm->farBufReadPos;
- memcpy(farend + readPos, aecm->farBuf + aecm->farBufReadPos,
- sizeof(WebRtc_Word16) * readLen);
- aecm->farBufReadPos = 0;
- readPos = readLen;
- readLen = farLen - readLen;
- }
- memcpy(farend + readPos, aecm->farBuf + aecm->farBufReadPos,
- sizeof(WebRtc_Word16) * readLen);
- aecm->farBufReadPos += readLen;
-}
-
-
diff --git a/trunk/src/modules/audio_processing/aecm/aecm_core.h b/trunk/src/modules/audio_processing/aecm/aecm_core.h
deleted file mode 100644
index 0ec62ec..0000000
--- a/trunk/src/modules/audio_processing/aecm/aecm_core.h
+++ /dev/null
@@ -1,375 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Performs echo control (suppression) with fft routines in fixed-point
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_MAIN_SOURCE_AECM_CORE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_MAIN_SOURCE_AECM_CORE_H_
-
-#define AECM_DYNAMIC_Q // turn on/off dynamic Q-domain
-//#define AECM_WITH_ABS_APPROX
-//#define AECM_SHORT // for 32 sample partition length (otherwise 64)
-
-#include "typedefs.h"
-#include "signal_processing_library.h"
-
-// Algorithm parameters
-
-#define FRAME_LEN 80 // Total frame length, 10 ms
-#ifdef AECM_SHORT
-
-#define PART_LEN 32 // Length of partition
-#define PART_LEN_SHIFT 6 // Length of (PART_LEN * 2) in base 2
-
-#else
-
-#define PART_LEN 64 // Length of partition
-#define PART_LEN_SHIFT 7 // Length of (PART_LEN * 2) in base 2
-
-#endif
-
-#define PART_LEN1 (PART_LEN + 1) // Unique fft coefficients
-#define PART_LEN2 (PART_LEN << 1) // Length of partition * 2
-#define PART_LEN4 (PART_LEN << 2) // Length of partition * 4
-#define FAR_BUF_LEN PART_LEN4 // Length of buffers
-#define MAX_DELAY 100
-
-// Counter parameters
-#ifdef AECM_SHORT
-
-#define CONV_LEN 1024 // Convergence length used at startup
-#else
-
-#define CONV_LEN 512 // Convergence length used at startup
-#endif
-
-#define CONV_LEN2 (CONV_LEN << 1) // Convergence length * 2 used at startup
-// Energy parameters
-#define MAX_BUF_LEN 64 // History length of energy signals
-
-#define FAR_ENERGY_MIN 1025 // Lowest Far energy level: At least 2 in energy
-#define FAR_ENERGY_DIFF 929 // Allowed difference between max and min
-
-#define ENERGY_DEV_OFFSET 0 // The energy error offset in Q8
-#define ENERGY_DEV_TOL 400 // The energy estimation tolerance in Q8
-#define FAR_ENERGY_VAD_REGION 230 // Far VAD tolerance region
-// Stepsize parameters
-#define MU_MIN 10 // Min stepsize 2^-MU_MIN (far end energy dependent)
-#define MU_MAX 1 // Max stepsize 2^-MU_MAX (far end energy dependent)
-#define MU_DIFF 9 // MU_MIN - MU_MAX
-// Channel parameters
-#define MIN_MSE_COUNT 20 // Min number of consecutive blocks with enough far end
- // energy to compare channel estimates
-#define MIN_MSE_DIFF 29 // The ratio between adapted and stored channel to
- // accept a new storage (0.8 in Q-MSE_RESOLUTION)
-#define MSE_RESOLUTION 5 // MSE parameter resolution
-#define RESOLUTION_CHANNEL16 12 // W16 Channel in Q-RESOLUTION_CHANNEL16
-#define RESOLUTION_CHANNEL32 28 // W32 Channel in Q-RESOLUTION_CHANNEL
-#define CHANNEL_VAD 16 // Minimum energy in frequency band to update channel
-// Suppression gain parameters: SUPGAIN_ parameters in Q-(RESOLUTION_SUPGAIN)
-#define RESOLUTION_SUPGAIN 8 // Channel in Q-(RESOLUTION_SUPGAIN)
-#define SUPGAIN_DEFAULT (1 << RESOLUTION_SUPGAIN) // Default suppression gain
-#define SUPGAIN_ERROR_PARAM_A 3072 // Estimation error parameter (Maximum gain) (8 in Q8)
-#define SUPGAIN_ERROR_PARAM_B 1536 // Estimation error parameter (Gain before going down)
-#define SUPGAIN_ERROR_PARAM_D SUPGAIN_DEFAULT // Estimation error parameter
- // (Should be the same as Default) (1 in Q8)
-#define SUPGAIN_EPC_DT 200 // = SUPGAIN_ERROR_PARAM_C * ENERGY_DEV_TOL
-// Defines for "check delay estimation"
-#define CORR_WIDTH 31 // Number of samples to correlate over.
-#define CORR_MAX 16 // Maximum correlation offset
-#define CORR_MAX_BUF 63
-#define CORR_DEV 4
-#define CORR_MAX_LEVEL 20
-#define CORR_MAX_LOW 4
-#define CORR_BUF_LEN (CORR_MAX << 1) + 1
-// Note that CORR_WIDTH + 2*CORR_MAX <= MAX_BUF_LEN
-
-#define ONE_Q14 (1 << 14)
-
-// NLP defines
-#define NLP_COMP_LOW 3277 // 0.2 in Q14
-#define NLP_COMP_HIGH ONE_Q14 // 1 in Q14
-
-extern const WebRtc_Word16 WebRtcAecm_kSqrtHanning[];
-
-typedef struct {
- WebRtc_Word16 real;
- WebRtc_Word16 imag;
-} complex16_t;
-
-typedef struct
-{
- int farBufWritePos;
- int farBufReadPos;
- int knownDelay;
- int lastKnownDelay;
- int firstVAD; // Parameter to control poorly initialized channels
-
- void *farFrameBuf;
- void *nearNoisyFrameBuf;
- void *nearCleanFrameBuf;
- void *outFrameBuf;
-
- WebRtc_Word16 farBuf[FAR_BUF_LEN];
-
- WebRtc_Word16 mult;
- WebRtc_UWord32 seed;
-
- // Delay estimation variables
- void* delay_estimator;
- WebRtc_UWord16 currentDelay;
- // Far end history variables
- // TODO(bjornv): Replace |far_history| with ring_buffer.
- uint16_t far_history[PART_LEN1 * MAX_DELAY];
- int far_history_pos;
- int far_q_domains[MAX_DELAY];
-
- WebRtc_Word16 nlpFlag;
- WebRtc_Word16 fixedDelay;
-
- WebRtc_UWord32 totCount;
-
- WebRtc_Word16 dfaCleanQDomain;
- WebRtc_Word16 dfaCleanQDomainOld;
- WebRtc_Word16 dfaNoisyQDomain;
- WebRtc_Word16 dfaNoisyQDomainOld;
-
- WebRtc_Word16 nearLogEnergy[MAX_BUF_LEN];
- WebRtc_Word16 farLogEnergy;
- WebRtc_Word16 echoAdaptLogEnergy[MAX_BUF_LEN];
- WebRtc_Word16 echoStoredLogEnergy[MAX_BUF_LEN];
-
- // The extra 16 or 32 bytes in the following buffers are for alignment based Neon code.
- // It's designed this way since the current GCC compiler can't align a buffer in 16 or 32
- // byte boundaries properly.
- WebRtc_Word16 channelStored_buf[PART_LEN1 + 8];
- WebRtc_Word16 channelAdapt16_buf[PART_LEN1 + 8];
- WebRtc_Word32 channelAdapt32_buf[PART_LEN1 + 8];
- WebRtc_Word16 xBuf_buf[PART_LEN2 + 16]; // farend
- WebRtc_Word16 dBufClean_buf[PART_LEN2 + 16]; // nearend
- WebRtc_Word16 dBufNoisy_buf[PART_LEN2 + 16]; // nearend
- WebRtc_Word16 outBuf_buf[PART_LEN + 8];
-
- // Pointers to the above buffers
- WebRtc_Word16 *channelStored;
- WebRtc_Word16 *channelAdapt16;
- WebRtc_Word32 *channelAdapt32;
- WebRtc_Word16 *xBuf;
- WebRtc_Word16 *dBufClean;
- WebRtc_Word16 *dBufNoisy;
- WebRtc_Word16 *outBuf;
-
- WebRtc_Word32 echoFilt[PART_LEN1];
- WebRtc_Word16 nearFilt[PART_LEN1];
- WebRtc_Word32 noiseEst[PART_LEN1];
- int noiseEstTooLowCtr[PART_LEN1];
- int noiseEstTooHighCtr[PART_LEN1];
- WebRtc_Word16 noiseEstCtr;
- WebRtc_Word16 cngMode;
-
- WebRtc_Word32 mseAdaptOld;
- WebRtc_Word32 mseStoredOld;
- WebRtc_Word32 mseThreshold;
-
- WebRtc_Word16 farEnergyMin;
- WebRtc_Word16 farEnergyMax;
- WebRtc_Word16 farEnergyMaxMin;
- WebRtc_Word16 farEnergyVAD;
- WebRtc_Word16 farEnergyMSE;
- int currentVADValue;
- WebRtc_Word16 vadUpdateCount;
-
- WebRtc_Word16 startupState;
- WebRtc_Word16 mseChannelCount;
- WebRtc_Word16 supGain;
- WebRtc_Word16 supGainOld;
-
- WebRtc_Word16 supGainErrParamA;
- WebRtc_Word16 supGainErrParamD;
- WebRtc_Word16 supGainErrParamDiffAB;
- WebRtc_Word16 supGainErrParamDiffBD;
-
-#ifdef AEC_DEBUG
- FILE *farFile;
- FILE *nearFile;
- FILE *outFile;
-#endif
-} AecmCore_t;
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_CreateCore(...)
-//
-// Allocates the memory needed by the AECM. The memory needs to be
-// initialized separately using the WebRtcAecm_InitCore() function.
-//
-// Input:
-// - aecm : Instance that should be created
-//
-// Output:
-// - aecm : Created instance
-//
-// Return value : 0 - Ok
-// -1 - Error
-//
-int WebRtcAecm_CreateCore(AecmCore_t **aecm);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_InitCore(...)
-//
-// This function initializes the AECM instant created with WebRtcAecm_CreateCore(...)
-// Input:
-// - aecm : Pointer to the AECM instance
-// - samplingFreq : Sampling Frequency
-//
-// Output:
-// - aecm : Initialized instance
-//
-// Return value : 0 - Ok
-// -1 - Error
-//
-int WebRtcAecm_InitCore(AecmCore_t * const aecm, int samplingFreq);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_FreeCore(...)
-//
-// This function releases the memory allocated by WebRtcAecm_CreateCore()
-// Input:
-// - aecm : Pointer to the AECM instance
-//
-// Return value : 0 - Ok
-// -1 - Error
-// 11001-11016: Error
-//
-int WebRtcAecm_FreeCore(AecmCore_t *aecm);
-
-int WebRtcAecm_Control(AecmCore_t *aecm, int delay, int nlpFlag);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_InitEchoPathCore(...)
-//
-// This function resets the echo channel adaptation with the specified channel.
-// Input:
-// - aecm : Pointer to the AECM instance
-// - echo_path : Pointer to the data that should initialize the echo path
-//
-// Output:
-// - aecm : Initialized instance
-//
-void WebRtcAecm_InitEchoPathCore(AecmCore_t* aecm, const WebRtc_Word16* echo_path);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_ProcessFrame(...)
-//
-// This function processes frames and sends blocks to WebRtcAecm_ProcessBlock(...)
-//
-// Inputs:
-// - aecm : Pointer to the AECM instance
-// - farend : In buffer containing one frame of echo signal
-// - nearendNoisy : In buffer containing one frame of nearend+echo signal without NS
-// - nearendClean : In buffer containing one frame of nearend+echo signal with NS
-//
-// Output:
-// - out : Out buffer, one frame of nearend signal :
-//
-//
-int WebRtcAecm_ProcessFrame(AecmCore_t * aecm, const WebRtc_Word16 * farend,
- const WebRtc_Word16 * nearendNoisy,
- const WebRtc_Word16 * nearendClean,
- WebRtc_Word16 * out);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_ProcessBlock(...)
-//
-// This function is called for every block within one frame
-// This function is called by WebRtcAecm_ProcessFrame(...)
-//
-// Inputs:
-// - aecm : Pointer to the AECM instance
-// - farend : In buffer containing one block of echo signal
-// - nearendNoisy : In buffer containing one frame of nearend+echo signal without NS
-// - nearendClean : In buffer containing one frame of nearend+echo signal with NS
-//
-// Output:
-// - out : Out buffer, one block of nearend signal :
-//
-//
-int WebRtcAecm_ProcessBlock(AecmCore_t * aecm, const WebRtc_Word16 * farend,
- const WebRtc_Word16 * nearendNoisy,
- const WebRtc_Word16 * noisyClean,
- WebRtc_Word16 * out);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_BufferFarFrame()
-//
-// Inserts a frame of data into farend buffer.
-//
-// Inputs:
-// - aecm : Pointer to the AECM instance
-// - farend : In buffer containing one frame of farend signal
-// - farLen : Length of frame
-//
-void WebRtcAecm_BufferFarFrame(AecmCore_t * const aecm, const WebRtc_Word16 * const farend,
- const int farLen);
-
-///////////////////////////////////////////////////////////////////////////////////////////////
-// WebRtcAecm_FetchFarFrame()
-//
-// Read the farend buffer to account for known delay
-//
-// Inputs:
-// - aecm : Pointer to the AECM instance
-// - farend : In buffer containing one frame of farend signal
-// - farLen : Length of frame
-// - knownDelay : known delay
-//
-void WebRtcAecm_FetchFarFrame(AecmCore_t * const aecm, WebRtc_Word16 * const farend,
- const int farLen, const int knownDelay);
-
-///////////////////////////////////////////////////////////////////////////////
-// Some function pointers, for internal functions shared by ARM NEON and
-// generic C code.
-//
-typedef void (*CalcLinearEnergies)(
- AecmCore_t* aecm,
- const WebRtc_UWord16* far_spectrum,
- WebRtc_Word32* echoEst,
- WebRtc_UWord32* far_energy,
- WebRtc_UWord32* echo_energy_adapt,
- WebRtc_UWord32* echo_energy_stored);
-extern CalcLinearEnergies WebRtcAecm_CalcLinearEnergies;
-
-typedef void (*StoreAdaptiveChannel)(
- AecmCore_t* aecm,
- const WebRtc_UWord16* far_spectrum,
- WebRtc_Word32* echo_est);
-extern StoreAdaptiveChannel WebRtcAecm_StoreAdaptiveChannel;
-
-typedef void (*ResetAdaptiveChannel)(AecmCore_t* aecm);
-extern ResetAdaptiveChannel WebRtcAecm_ResetAdaptiveChannel;
-
-typedef void (*WindowAndFFT)(
- WebRtc_Word16* fft,
- const WebRtc_Word16* time_signal,
- complex16_t* freq_signal,
- int time_signal_scaling);
-extern WindowAndFFT WebRtcAecm_WindowAndFFT;
-
-typedef void (*InverseFFTAndWindow)(
- AecmCore_t* aecm,
- WebRtc_Word16* fft, complex16_t* efw,
- WebRtc_Word16* output,
- const WebRtc_Word16* nearendClean);
-extern InverseFFTAndWindow WebRtcAecm_InverseFFTAndWindow;
-
-// Initialization of the above function pointers for ARM Neon.
-void WebRtcAecm_InitNeon(void);
-
-
-#endif
diff --git a/trunk/src/modules/audio_processing/aecm/aecm_core_neon.c b/trunk/src/modules/audio_processing/aecm/aecm_core_neon.c
deleted file mode 100644
index ab448b4..0000000
--- a/trunk/src/modules/audio_processing/aecm/aecm_core_neon.c
+++ /dev/null
@@ -1,303 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "aecm_core.h"
-
-#include <arm_neon.h>
-#include <assert.h>
-
-
-// Square root of Hanning window in Q14.
-static const WebRtc_Word16 kSqrtHanningReversed[] __attribute__((aligned(8))) = {
- 16384, 16373, 16354, 16325,
- 16286, 16237, 16179, 16111,
- 16034, 15947, 15851, 15746,
- 15631, 15506, 15373, 15231,
- 15079, 14918, 14749, 14571,
- 14384, 14189, 13985, 13773,
- 13553, 13325, 13089, 12845,
- 12594, 12335, 12068, 11795,
- 11514, 11227, 10933, 10633,
- 10326, 10013, 9695, 9370,
- 9040, 8705, 8364, 8019,
- 7668, 7313, 6954, 6591,
- 6224, 5853, 5478, 5101,
- 4720, 4337, 3951, 3562,
- 3172, 2780, 2386, 1990,
- 1594, 1196, 798, 399
-};
-
-static void WindowAndFFTNeon(WebRtc_Word16* fft,
- const WebRtc_Word16* time_signal,
- complex16_t* freq_signal,
- int time_signal_scaling) {
- int i, j;
-
- int16x4_t tmp16x4_scaling = vdup_n_s16(time_signal_scaling);
- __asm__("vmov.i16 d21, #0" ::: "d21");
-
- for (i = 0, j = 0; i < PART_LEN; i += 4, j += 8) {
- int16x4_t tmp16x4_0;
- int16x4_t tmp16x4_1;
- int32x4_t tmp32x4_0;
-
- /* Window near end */
- // fft[j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT((time_signal[i]
- // << time_signal_scaling), WebRtcAecm_kSqrtHanning[i], 14);
- __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&time_signal[i]));
- tmp16x4_0 = vshl_s16(tmp16x4_0, tmp16x4_scaling);
-
- __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_1) : "r"(&WebRtcAecm_kSqrtHanning[i]));
- tmp32x4_0 = vmull_s16(tmp16x4_0, tmp16x4_1);
-
- __asm__("vshrn.i32 d20, %q0, #14" : : "w"(tmp32x4_0) : "d20");
- __asm__("vst2.16 {d20, d21}, [%0, :128]" : : "r"(&fft[j]) : "q10");
-
- // fft[PART_LEN2 + j] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
- // (time_signal[PART_LEN + i] << time_signal_scaling),
- // WebRtcAecm_kSqrtHanning[PART_LEN - i], 14);
- __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&time_signal[i + PART_LEN]));
- tmp16x4_0 = vshl_s16(tmp16x4_0, tmp16x4_scaling);
-
- __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_1) : "r"(&kSqrtHanningReversed[i]));
- tmp32x4_0 = vmull_s16(tmp16x4_0, tmp16x4_1);
-
- __asm__("vshrn.i32 d20, %q0, #14" : : "w"(tmp32x4_0) : "d20");
- __asm__("vst2.16 {d20, d21}, [%0, :128]" : : "r"(&fft[PART_LEN2 + j]) : "q10");
- }
-
- WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
- WebRtcSpl_ComplexFFT(fft, PART_LEN_SHIFT, 1);
-
- // Take only the first PART_LEN2 samples, and switch the sign of the imaginary part.
- for (i = 0, j = 0; j < PART_LEN2; i += 8, j += 16) {
- __asm__("vld2.16 {d20, d21, d22, d23}, [%0, :256]" : : "r"(&fft[j]) : "q10", "q11");
- __asm__("vneg.s16 d22, d22" : : : "q10");
- __asm__("vneg.s16 d23, d23" : : : "q11");
- __asm__("vst2.16 {d20, d21, d22, d23}, [%0, :256]" : :
- "r"(&freq_signal[i].real): "q10", "q11");
- }
-}
-
-static void InverseFFTAndWindowNeon(AecmCore_t* aecm,
- WebRtc_Word16* fft,
- complex16_t* efw,
- WebRtc_Word16* output,
- const WebRtc_Word16* nearendClean) {
- int i, j, outCFFT;
- WebRtc_Word32 tmp32no1;
-
- // Synthesis
- for (i = 0, j = 0; i < PART_LEN; i += 4, j += 8) {
- // We overwrite two more elements in fft[], but it's ok.
- __asm__("vld2.16 {d20, d21}, [%0, :128]" : : "r"(&(efw[i].real)) : "q10");
- __asm__("vmov q11, q10" : : : "q10", "q11");
-
- __asm__("vneg.s16 d23, d23" : : : "q11");
- __asm__("vst2.16 {d22, d23}, [%0, :128]" : : "r"(&fft[j]): "q11");
-
- __asm__("vrev64.16 q10, q10" : : : "q10");
- __asm__("vst2.16 {d20, d21}, [%0]" : : "r"(&fft[PART_LEN4 - j - 6]): "q10");
- }
-
- fft[PART_LEN2] = efw[PART_LEN].real;
- fft[PART_LEN2 + 1] = -efw[PART_LEN].imag;
-
- // Inverse FFT, result should be scaled with outCFFT.
- WebRtcSpl_ComplexBitReverse(fft, PART_LEN_SHIFT);
- outCFFT = WebRtcSpl_ComplexIFFT(fft, PART_LEN_SHIFT, 1);
-
- // Take only the real values and scale with outCFFT.
- for (i = 0, j = 0; i < PART_LEN2; i += 8, j += 16) {
- __asm__("vld2.16 {d20, d21, d22, d23}, [%0, :256]" : : "r"(&fft[j]) : "q10", "q11");
- __asm__("vst1.16 {d20, d21}, [%0, :128]" : : "r"(&fft[i]): "q10");
- }
-
- int32x4_t tmp32x4_2;
- __asm__("vdup.32 %q0, %1" : "=w"(tmp32x4_2) : "r"((WebRtc_Word32)
- (outCFFT - aecm->dfaCleanQDomain)));
- for (i = 0; i < PART_LEN; i += 4) {
- int16x4_t tmp16x4_0;
- int16x4_t tmp16x4_1;
- int32x4_t tmp32x4_0;
- int32x4_t tmp32x4_1;
-
- // fft[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
- // fft[i], WebRtcAecm_kSqrtHanning[i], 14);
- __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&fft[i]));
- __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_1) : "r"(&WebRtcAecm_kSqrtHanning[i]));
- __asm__("vmull.s16 %q0, %P1, %P2" : "=w"(tmp32x4_0) : "w"(tmp16x4_0), "w"(tmp16x4_1));
- __asm__("vrshr.s32 %q0, %q1, #14" : "=w"(tmp32x4_0) : "0"(tmp32x4_0));
-
- // tmp32no1 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)fft[i],
- // outCFFT - aecm->dfaCleanQDomain);
- __asm__("vshl.s32 %q0, %q1, %q2" : "=w"(tmp32x4_0) : "0"(tmp32x4_0), "w"(tmp32x4_2));
-
- // fft[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(WEBRTC_SPL_WORD16_MAX,
- // tmp32no1 + outBuf[i], WEBRTC_SPL_WORD16_MIN);
- // output[i] = fft[i];
- __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&aecm->outBuf[i]));
- __asm__("vmovl.s16 %q0, %P1" : "=w"(tmp32x4_1) : "w"(tmp16x4_0));
- __asm__("vadd.i32 %q0, %q1" : : "w"(tmp32x4_0), "w"(tmp32x4_1));
- __asm__("vqshrn.s32 %P0, %q1, #0" : "=w"(tmp16x4_0) : "w"(tmp32x4_0));
- __asm__("vst1.16 %P0, [%1, :64]" : : "w"(tmp16x4_0), "r"(&fft[i]));
- __asm__("vst1.16 %P0, [%1, :64]" : : "w"(tmp16x4_0), "r"(&output[i]));
-
- // tmp32no1 = WEBRTC_SPL_MUL_16_16_RSFT(
- // fft[PART_LEN + i], WebRtcAecm_kSqrtHanning[PART_LEN - i], 14);
- __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_0) : "r"(&fft[PART_LEN + i]));
- __asm__("vld1.16 %P0, [%1, :64]" : "=w"(tmp16x4_1) : "r"(&kSqrtHanningReversed[i]));
- __asm__("vmull.s16 %q0, %P1, %P2" : "=w"(tmp32x4_0) : "w"(tmp16x4_0), "w"(tmp16x4_1));
- __asm__("vshr.s32 %q0, %q1, #14" : "=w"(tmp32x4_0) : "0"(tmp32x4_0));
-
- // tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, outCFFT - aecm->dfaCleanQDomain);
- __asm__("vshl.s32 %q0, %q1, %q2" : "=w"(tmp32x4_0) : "0"(tmp32x4_0), "w"(tmp32x4_2));
- // outBuf[i] = (WebRtc_Word16)WEBRTC_SPL_SAT(
- // WEBRTC_SPL_WORD16_MAX, tmp32no1, WEBRTC_SPL_WORD16_MIN);
- __asm__("vqshrn.s32 %P0, %q1, #0" : "=w"(tmp16x4_0) : "w"(tmp32x4_0));
- __asm__("vst1.16 %P0, [%1, :64]" : : "w"(tmp16x4_0), "r"(&aecm->outBuf[i]));
- }
-
- // Copy the current block to the old position (outBuf is shifted elsewhere).
- for (i = 0; i < PART_LEN; i += 16) {
- __asm__("vld1.16 {d20, d21, d22, d23}, [%0, :256]" : :
- "r"(&aecm->xBuf[i + PART_LEN]) : "q10");
- __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : : "r"(&aecm->xBuf[i]): "q10");
- }
- for (i = 0; i < PART_LEN; i += 16) {
- __asm__("vld1.16 {d20, d21, d22, d23}, [%0, :256]" : :
- "r"(&aecm->dBufNoisy[i + PART_LEN]) : "q10");
- __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : :
- "r"(&aecm->dBufNoisy[i]): "q10");
- }
- if (nearendClean != NULL) {
- for (i = 0; i < PART_LEN; i += 16) {
- __asm__("vld1.16 {d20, d21, d22, d23}, [%0, :256]" : :
- "r"(&aecm->dBufClean[i + PART_LEN]) : "q10");
- __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : :
- "r"(&aecm->dBufClean[i]): "q10");
- }
- }
-}
-
-static void CalcLinearEnergiesNeon(AecmCore_t* aecm,
- const WebRtc_UWord16* far_spectrum,
- WebRtc_Word32* echo_est,
- WebRtc_UWord32* far_energy,
- WebRtc_UWord32* echo_energy_adapt,
- WebRtc_UWord32* echo_energy_stored) {
- int i;
-
- register WebRtc_UWord32 far_energy_r;
- register WebRtc_UWord32 echo_energy_stored_r;
- register WebRtc_UWord32 echo_energy_adapt_r;
- uint32x4_t tmp32x4_0;
-
- __asm__("vmov.i32 q14, #0" : : : "q14"); // far_energy
- __asm__("vmov.i32 q8, #0" : : : "q8"); // echo_energy_stored
- __asm__("vmov.i32 q9, #0" : : : "q9"); // echo_energy_adapt
-
- for (i = 0; i < PART_LEN - 7; i += 8) {
- // far_energy += (WebRtc_UWord32)(far_spectrum[i]);
- __asm__("vld1.16 {d26, d27}, [%0]" : : "r"(&far_spectrum[i]) : "q13");
- __asm__("vaddw.u16 q14, q14, d26" : : : "q14", "q13");
- __asm__("vaddw.u16 q14, q14, d27" : : : "q14", "q13");
-
- // Get estimated echo energies for adaptive channel and stored channel.
- // echoEst[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
- __asm__("vld1.16 {d24, d25}, [%0, :128]" : : "r"(&aecm->channelStored[i]) : "q12");
- __asm__("vmull.u16 q10, d26, d24" : : : "q12", "q13", "q10");
- __asm__("vmull.u16 q11, d27, d25" : : : "q12", "q13", "q11");
- __asm__("vst1.32 {d20, d21, d22, d23}, [%0, :256]" : : "r"(&echo_est[i]):
- "q10", "q11");
-
- // echo_energy_stored += (WebRtc_UWord32)echoEst[i];
- __asm__("vadd.u32 q8, q10" : : : "q10", "q8");
- __asm__("vadd.u32 q8, q11" : : : "q11", "q8");
-
- // echo_energy_adapt += WEBRTC_SPL_UMUL_16_16(
- // aecm->channelAdapt16[i], far_spectrum[i]);
- __asm__("vld1.16 {d24, d25}, [%0, :128]" : : "r"(&aecm->channelAdapt16[i]) : "q12");
- __asm__("vmull.u16 q10, d26, d24" : : : "q12", "q13", "q10");
- __asm__("vmull.u16 q11, d27, d25" : : : "q12", "q13", "q11");
- __asm__("vadd.u32 q9, q10" : : : "q9", "q15");
- __asm__("vadd.u32 q9, q11" : : : "q9", "q11");
- }
-
- __asm__("vadd.u32 d28, d29" : : : "q14");
- __asm__("vpadd.u32 d28, d28" : : : "q14");
- __asm__("vmov.32 %0, d28[0]" : "=r"(far_energy_r): : "q14");
-
- __asm__("vadd.u32 d18, d19" : : : "q9");
- __asm__("vpadd.u32 d18, d18" : : : "q9");
- __asm__("vmov.32 %0, d18[0]" : "=r"(echo_energy_adapt_r): : "q9");
-
- __asm__("vadd.u32 d16, d17" : : : "q8");
- __asm__("vpadd.u32 d16, d16" : : : "q8");
- __asm__("vmov.32 %0, d16[0]" : "=r"(echo_energy_stored_r): : "q8");
-
- // Get estimated echo energies for adaptive channel and stored channel.
- echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
- *echo_energy_stored = echo_energy_stored_r + (WebRtc_UWord32)echo_est[i];
- *far_energy = far_energy_r + (WebRtc_UWord32)(far_spectrum[i]);
- *echo_energy_adapt = echo_energy_adapt_r + WEBRTC_SPL_UMUL_16_16(
- aecm->channelAdapt16[i], far_spectrum[i]);
-}
-
-static void StoreAdaptiveChannelNeon(AecmCore_t* aecm,
- const WebRtc_UWord16* far_spectrum,
- WebRtc_Word32* echo_est) {
- int i;
-
- // During startup we store the channel every block.
- // Recalculate echo estimate.
- for (i = 0; i < PART_LEN - 7; i += 8) {
- // aecm->channelStored[i] = acem->channelAdapt16[i];
- // echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
- __asm__("vld1.16 {d26, d27}, [%0]" : : "r"(&far_spectrum[i]) : "q13");
- __asm__("vld1.16 {d24, d25}, [%0, :128]" : : "r"(&aecm->channelAdapt16[i]) : "q12");
- __asm__("vst1.16 {d24, d25}, [%0, :128]" : : "r"(&aecm->channelStored[i]) : "q12");
- __asm__("vmull.u16 q10, d26, d24" : : : "q12", "q13", "q10");
- __asm__("vmull.u16 q11, d27, d25" : : : "q12", "q13", "q11");
- __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : :
- "r"(&echo_est[i]) : "q10", "q11");
- }
- aecm->channelStored[i] = aecm->channelAdapt16[i];
- echo_est[i] = WEBRTC_SPL_MUL_16_U16(aecm->channelStored[i], far_spectrum[i]);
-}
-
-static void ResetAdaptiveChannelNeon(AecmCore_t* aecm) {
- int i;
-
- for (i = 0; i < PART_LEN - 7; i += 8) {
- // aecm->channelAdapt16[i] = aecm->channelStored[i];
- // aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)
- // aecm->channelStored[i], 16);
- __asm__("vld1.16 {d24, d25}, [%0, :128]" : :
- "r"(&aecm->channelStored[i]) : "q12");
- __asm__("vst1.16 {d24, d25}, [%0, :128]" : :
- "r"(&aecm->channelAdapt16[i]) : "q12");
- __asm__("vshll.s16 q10, d24, #16" : : : "q12", "q13", "q10");
- __asm__("vshll.s16 q11, d25, #16" : : : "q12", "q13", "q11");
- __asm__("vst1.16 {d20, d21, d22, d23}, [%0, :256]" : :
- "r"(&aecm->channelAdapt32[i]): "q10", "q11");
- }
- aecm->channelAdapt16[i] = aecm->channelStored[i];
- aecm->channelAdapt32[i] = WEBRTC_SPL_LSHIFT_W32(
- (WebRtc_Word32)aecm->channelStored[i], 16);
-}
-
-void WebRtcAecm_InitNeon(void) {
- WebRtcAecm_WindowAndFFT = WindowAndFFTNeon;
- WebRtcAecm_InverseFFTAndWindow = InverseFFTAndWindowNeon;
- WebRtcAecm_CalcLinearEnergies = CalcLinearEnergiesNeon;
- WebRtcAecm_StoreAdaptiveChannel = StoreAdaptiveChannelNeon;
- WebRtcAecm_ResetAdaptiveChannel = ResetAdaptiveChannelNeon;
-}
diff --git a/trunk/src/modules/audio_processing/aecm/echo_control_mobile.c b/trunk/src/modules/audio_processing/aecm/echo_control_mobile.c
deleted file mode 100644
index 566ae00..0000000
--- a/trunk/src/modules/audio_processing/aecm/echo_control_mobile.c
+++ /dev/null
@@ -1,792 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdlib.h>
-//#include <string.h>
-
-#include "echo_control_mobile.h"
-#include "aecm_core.h"
-#include "ring_buffer.h"
-#ifdef AEC_DEBUG
-#include <stdio.h>
-#endif
-#ifdef MAC_IPHONE_PRINT
-#include <time.h>
-#include <stdio.h>
-#elif defined ARM_WINM_LOG
-#include "windows.h"
-extern HANDLE logFile;
-#endif
-
-#define BUF_SIZE_FRAMES 50 // buffer size (frames)
-// Maximum length of resampled signal. Must be an integer multiple of frames
-// (ceil(1/(1 + MIN_SKEW)*2) + 1)*FRAME_LEN
-// The factor of 2 handles wb, and the + 1 is as a safety margin
-#define MAX_RESAMP_LEN (5 * FRAME_LEN)
-
-static const size_t kBufSizeSamp = BUF_SIZE_FRAMES * FRAME_LEN; // buffer size (samples)
-static const int kSampMsNb = 8; // samples per ms in nb
-// Target suppression levels for nlp modes
-// log{0.001, 0.00001, 0.00000001}
-static const int kInitCheck = 42;
-
-typedef struct
-{
- int sampFreq;
- int scSampFreq;
- short bufSizeStart;
- int knownDelay;
-
- // Stores the last frame added to the farend buffer
- short farendOld[2][FRAME_LEN];
- short initFlag; // indicates if AEC has been initialized
-
- // Variables used for averaging far end buffer size
- short counter;
- short sum;
- short firstVal;
- short checkBufSizeCtr;
-
- // Variables used for delay shifts
- short msInSndCardBuf;
- short filtDelay;
- int timeForDelayChange;
- int ECstartup;
- int checkBuffSize;
- int delayChange;
- short lastDelayDiff;
-
- WebRtc_Word16 echoMode;
-
-#ifdef AEC_DEBUG
- FILE *bufFile;
- FILE *delayFile;
- FILE *preCompFile;
- FILE *postCompFile;
-#endif // AEC_DEBUG
- // Structures
- void *farendBuf;
-
- int lastError;
-
- AecmCore_t *aecmCore;
-} aecmob_t;
-
-// Estimates delay to set the position of the farend buffer read pointer
-// (controlled by knownDelay)
-static int WebRtcAecm_EstBufDelay(aecmob_t *aecmInst, short msInSndCardBuf);
-
-// Stuffs the farend buffer if the estimated delay is too large
-static int WebRtcAecm_DelayComp(aecmob_t *aecmInst);
-
-WebRtc_Word32 WebRtcAecm_Create(void **aecmInst)
-{
- aecmob_t *aecm;
- if (aecmInst == NULL)
- {
- return -1;
- }
-
- aecm = malloc(sizeof(aecmob_t));
- *aecmInst = aecm;
- if (aecm == NULL)
- {
- return -1;
- }
-
- if (WebRtcAecm_CreateCore(&aecm->aecmCore) == -1)
- {
- WebRtcAecm_Free(aecm);
- aecm = NULL;
- return -1;
- }
-
- if (WebRtc_CreateBuffer(&aecm->farendBuf, kBufSizeSamp,
- sizeof(int16_t)) == -1)
- {
- WebRtcAecm_Free(aecm);
- aecm = NULL;
- return -1;
- }
-
- aecm->initFlag = 0;
- aecm->lastError = 0;
-
-#ifdef AEC_DEBUG
- aecm->aecmCore->farFile = fopen("aecFar.pcm","wb");
- aecm->aecmCore->nearFile = fopen("aecNear.pcm","wb");
- aecm->aecmCore->outFile = fopen("aecOut.pcm","wb");
- //aecm->aecmCore->outLpFile = fopen("aecOutLp.pcm","wb");
-
- aecm->bufFile = fopen("aecBuf.dat", "wb");
- aecm->delayFile = fopen("aecDelay.dat", "wb");
- aecm->preCompFile = fopen("preComp.pcm", "wb");
- aecm->postCompFile = fopen("postComp.pcm", "wb");
-#endif // AEC_DEBUG
- return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_Free(void *aecmInst)
-{
- aecmob_t *aecm = aecmInst;
-
- if (aecm == NULL)
- {
- return -1;
- }
-
-#ifdef AEC_DEBUG
- fclose(aecm->aecmCore->farFile);
- fclose(aecm->aecmCore->nearFile);
- fclose(aecm->aecmCore->outFile);
- //fclose(aecm->aecmCore->outLpFile);
-
- fclose(aecm->bufFile);
- fclose(aecm->delayFile);
- fclose(aecm->preCompFile);
- fclose(aecm->postCompFile);
-#endif // AEC_DEBUG
- WebRtcAecm_FreeCore(aecm->aecmCore);
- WebRtc_FreeBuffer(aecm->farendBuf);
- free(aecm);
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_Init(void *aecmInst, WebRtc_Word32 sampFreq)
-{
- aecmob_t *aecm = aecmInst;
- AecmConfig aecConfig;
-
- if (aecm == NULL)
- {
- return -1;
- }
-
- if (sampFreq != 8000 && sampFreq != 16000)
- {
- aecm->lastError = AECM_BAD_PARAMETER_ERROR;
- return -1;
- }
- aecm->sampFreq = sampFreq;
-
- // Initialize AECM core
- if (WebRtcAecm_InitCore(aecm->aecmCore, aecm->sampFreq) == -1)
- {
- aecm->lastError = AECM_UNSPECIFIED_ERROR;
- return -1;
- }
-
- // Initialize farend buffer
- if (WebRtc_InitBuffer(aecm->farendBuf) == -1)
- {
- aecm->lastError = AECM_UNSPECIFIED_ERROR;
- return -1;
- }
-
- aecm->initFlag = kInitCheck; // indicates that initialization has been done
-
- aecm->delayChange = 1;
-
- aecm->sum = 0;
- aecm->counter = 0;
- aecm->checkBuffSize = 1;
- aecm->firstVal = 0;
-
- aecm->ECstartup = 1;
- aecm->bufSizeStart = 0;
- aecm->checkBufSizeCtr = 0;
- aecm->filtDelay = 0;
- aecm->timeForDelayChange = 0;
- aecm->knownDelay = 0;
- aecm->lastDelayDiff = 0;
-
- memset(&aecm->farendOld[0][0], 0, 160);
-
- // Default settings.
- aecConfig.cngMode = AecmTrue;
- aecConfig.echoMode = 3;
-
- if (WebRtcAecm_set_config(aecm, aecConfig) == -1)
- {
- aecm->lastError = AECM_UNSPECIFIED_ERROR;
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_BufferFarend(void *aecmInst, const WebRtc_Word16 *farend,
- WebRtc_Word16 nrOfSamples)
-{
- aecmob_t *aecm = aecmInst;
- WebRtc_Word32 retVal = 0;
-
- if (aecm == NULL)
- {
- return -1;
- }
-
- if (farend == NULL)
- {
- aecm->lastError = AECM_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (aecm->initFlag != kInitCheck)
- {
- aecm->lastError = AECM_UNINITIALIZED_ERROR;
- return -1;
- }
-
- if (nrOfSamples != 80 && nrOfSamples != 160)
- {
- aecm->lastError = AECM_BAD_PARAMETER_ERROR;
- return -1;
- }
-
- // TODO: Is this really a good idea?
- if (!aecm->ECstartup)
- {
- WebRtcAecm_DelayComp(aecm);
- }
-
- WebRtc_WriteBuffer(aecm->farendBuf, farend, (size_t) nrOfSamples);
-
- return retVal;
-}
-
-WebRtc_Word32 WebRtcAecm_Process(void *aecmInst, const WebRtc_Word16 *nearendNoisy,
- const WebRtc_Word16 *nearendClean, WebRtc_Word16 *out,
- WebRtc_Word16 nrOfSamples, WebRtc_Word16 msInSndCardBuf)
-{
- aecmob_t *aecm = aecmInst;
- WebRtc_Word32 retVal = 0;
- short i;
- short nmbrOfFilledBuffers;
- short nBlocks10ms;
- short nFrames;
-#ifdef AEC_DEBUG
- short msInAECBuf;
-#endif
-
-#ifdef ARM_WINM_LOG
- __int64 freq, start, end, diff;
- unsigned int milliseconds;
- DWORD temp;
-#elif defined MAC_IPHONE_PRINT
- // double endtime = 0, starttime = 0;
- struct timeval starttime;
- struct timeval endtime;
- static long int timeused = 0;
- static int timecount = 0;
-#endif
-
- if (aecm == NULL)
- {
- return -1;
- }
-
- if (nearendNoisy == NULL)
- {
- aecm->lastError = AECM_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (out == NULL)
- {
- aecm->lastError = AECM_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (aecm->initFlag != kInitCheck)
- {
- aecm->lastError = AECM_UNINITIALIZED_ERROR;
- return -1;
- }
-
- if (nrOfSamples != 80 && nrOfSamples != 160)
- {
- aecm->lastError = AECM_BAD_PARAMETER_ERROR;
- return -1;
- }
-
- if (msInSndCardBuf < 0)
- {
- msInSndCardBuf = 0;
- aecm->lastError = AECM_BAD_PARAMETER_WARNING;
- retVal = -1;
- } else if (msInSndCardBuf > 500)
- {
- msInSndCardBuf = 500;
- aecm->lastError = AECM_BAD_PARAMETER_WARNING;
- retVal = -1;
- }
- msInSndCardBuf += 10;
- aecm->msInSndCardBuf = msInSndCardBuf;
-
- nFrames = nrOfSamples / FRAME_LEN;
- nBlocks10ms = nFrames / aecm->aecmCore->mult;
-
- if (aecm->ECstartup)
- {
- if (nearendClean == NULL)
- {
- if (out != nearendNoisy)
- {
- memcpy(out, nearendNoisy, sizeof(short) * nrOfSamples);
- }
- } else if (out != nearendClean)
- {
- memcpy(out, nearendClean, sizeof(short) * nrOfSamples);
- }
-
- nmbrOfFilledBuffers =
- (short) WebRtc_available_read(aecm->farendBuf) / FRAME_LEN;
- // The AECM is in the start up mode
- // AECM is disabled until the soundcard buffer and farend buffers are OK
-
- // Mechanism to ensure that the soundcard buffer is reasonably stable.
- if (aecm->checkBuffSize)
- {
- aecm->checkBufSizeCtr++;
- // Before we fill up the far end buffer we require the amount of data on the
- // sound card to be stable (+/-8 ms) compared to the first value. This
- // comparison is made during the following 4 consecutive frames. If it seems
- // to be stable then we start to fill up the far end buffer.
-
- if (aecm->counter == 0)
- {
- aecm->firstVal = aecm->msInSndCardBuf;
- aecm->sum = 0;
- }
-
- if (abs(aecm->firstVal - aecm->msInSndCardBuf)
- < WEBRTC_SPL_MAX(0.2 * aecm->msInSndCardBuf, kSampMsNb))
- {
- aecm->sum += aecm->msInSndCardBuf;
- aecm->counter++;
- } else
- {
- aecm->counter = 0;
- }
-
- if (aecm->counter * nBlocks10ms >= 6)
- {
- // The farend buffer size is determined in blocks of 80 samples
- // Use 75% of the average value of the soundcard buffer
- aecm->bufSizeStart
- = WEBRTC_SPL_MIN((3 * aecm->sum
- * aecm->aecmCore->mult) / (aecm->counter * 40), BUF_SIZE_FRAMES);
- // buffersize has now been determined
- aecm->checkBuffSize = 0;
- }
-
- if (aecm->checkBufSizeCtr * nBlocks10ms > 50)
- {
- // for really bad sound cards, don't disable echocanceller for more than 0.5 sec
- aecm->bufSizeStart = WEBRTC_SPL_MIN((3 * aecm->msInSndCardBuf
- * aecm->aecmCore->mult) / 40, BUF_SIZE_FRAMES);
- aecm->checkBuffSize = 0;
- }
- }
-
- // if checkBuffSize changed in the if-statement above
- if (!aecm->checkBuffSize)
- {
- // soundcard buffer is now reasonably stable
- // When the far end buffer is filled with approximately the same amount of
- // data as the amount on the sound card we end the start up phase and start
- // to cancel echoes.
-
- if (nmbrOfFilledBuffers == aecm->bufSizeStart)
- {
- aecm->ECstartup = 0; // Enable the AECM
- } else if (nmbrOfFilledBuffers > aecm->bufSizeStart)
- {
- WebRtc_MoveReadPtr(aecm->farendBuf,
- (int) WebRtc_available_read(aecm->farendBuf)
- - (int) aecm->bufSizeStart * FRAME_LEN);
- aecm->ECstartup = 0;
- }
- }
-
- } else
- {
- // AECM is enabled
-
- // Note only 1 block supported for nb and 2 blocks for wb
- for (i = 0; i < nFrames; i++)
- {
- int16_t farend[FRAME_LEN];
- const int16_t* farend_ptr = NULL;
-
- nmbrOfFilledBuffers =
- (short) WebRtc_available_read(aecm->farendBuf) / FRAME_LEN;
-
- // Check that there is data in the far end buffer
- if (nmbrOfFilledBuffers > 0)
- {
- // Get the next 80 samples from the farend buffer
- WebRtc_ReadBuffer(aecm->farendBuf, (void**) &farend_ptr, farend,
- FRAME_LEN);
-
- // Always store the last frame for use when we run out of data
- memcpy(&(aecm->farendOld[i][0]), farend_ptr,
- FRAME_LEN * sizeof(short));
- } else
- {
- // We have no data so we use the last played frame
- memcpy(farend, &(aecm->farendOld[i][0]), FRAME_LEN * sizeof(short));
- farend_ptr = farend;
- }
-
- // Call buffer delay estimator when all data is extracted,
- // i,e. i = 0 for NB and i = 1 for WB
- if ((i == 0 && aecm->sampFreq == 8000) || (i == 1 && aecm->sampFreq == 16000))
- {
- WebRtcAecm_EstBufDelay(aecm, aecm->msInSndCardBuf);
- }
-
-#ifdef ARM_WINM_LOG
- // measure tick start
- QueryPerformanceFrequency((LARGE_INTEGER*)&freq);
- QueryPerformanceCounter((LARGE_INTEGER*)&start);
-#elif defined MAC_IPHONE_PRINT
- // starttime = clock()/(double)CLOCKS_PER_SEC;
- gettimeofday(&starttime, NULL);
-#endif
- // Call the AECM
- /*WebRtcAecm_ProcessFrame(aecm->aecmCore, farend, &nearend[FRAME_LEN * i],
- &out[FRAME_LEN * i], aecm->knownDelay);*/
- if (nearendClean == NULL)
- {
- if (WebRtcAecm_ProcessFrame(aecm->aecmCore,
- farend_ptr,
- &nearendNoisy[FRAME_LEN * i],
- NULL,
- &out[FRAME_LEN * i]) == -1)
- {
- return -1;
- }
- } else
- {
- if (WebRtcAecm_ProcessFrame(aecm->aecmCore,
- farend_ptr,
- &nearendNoisy[FRAME_LEN * i],
- &nearendClean[FRAME_LEN * i],
- &out[FRAME_LEN * i]) == -1)
- {
- return -1;
- }
- }
-
-#ifdef ARM_WINM_LOG
-
- // measure tick end
- QueryPerformanceCounter((LARGE_INTEGER*)&end);
-
- if(end > start)
- {
- diff = ((end - start) * 1000) / (freq/1000);
- milliseconds = (unsigned int)(diff & 0xffffffff);
- WriteFile (logFile, &milliseconds, sizeof(unsigned int), &temp, NULL);
- }
-#elif defined MAC_IPHONE_PRINT
- // endtime = clock()/(double)CLOCKS_PER_SEC;
- // printf("%f\n", endtime - starttime);
-
- gettimeofday(&endtime, NULL);
-
- if( endtime.tv_usec > starttime.tv_usec)
- {
- timeused += endtime.tv_usec - starttime.tv_usec;
- } else
- {
- timeused += endtime.tv_usec + 1000000 - starttime.tv_usec;
- }
-
- if(++timecount == 1000)
- {
- timecount = 0;
- printf("AEC: %ld\n", timeused);
- timeused = 0;
- }
-#endif
-
- }
- }
-
-#ifdef AEC_DEBUG
- msInAECBuf = (short) WebRtc_available_read(aecm->farendBuf) /
- (kSampMsNb * aecm->aecmCore->mult);
- fwrite(&msInAECBuf, 2, 1, aecm->bufFile);
- fwrite(&(aecm->knownDelay), sizeof(aecm->knownDelay), 1, aecm->delayFile);
-#endif
-
- return retVal;
-}
-
-WebRtc_Word32 WebRtcAecm_set_config(void *aecmInst, AecmConfig config)
-{
- aecmob_t *aecm = aecmInst;
-
- if (aecm == NULL)
- {
- return -1;
- }
-
- if (aecm->initFlag != kInitCheck)
- {
- aecm->lastError = AECM_UNINITIALIZED_ERROR;
- return -1;
- }
-
- if (config.cngMode != AecmFalse && config.cngMode != AecmTrue)
- {
- aecm->lastError = AECM_BAD_PARAMETER_ERROR;
- return -1;
- }
- aecm->aecmCore->cngMode = config.cngMode;
-
- if (config.echoMode < 0 || config.echoMode > 4)
- {
- aecm->lastError = AECM_BAD_PARAMETER_ERROR;
- return -1;
- }
- aecm->echoMode = config.echoMode;
-
- if (aecm->echoMode == 0)
- {
- aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 3;
- aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 3;
- aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 3;
- aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 3;
- aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A >> 3)
- - (SUPGAIN_ERROR_PARAM_B >> 3);
- aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B >> 3)
- - (SUPGAIN_ERROR_PARAM_D >> 3);
- } else if (aecm->echoMode == 1)
- {
- aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 2;
- aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 2;
- aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 2;
- aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 2;
- aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A >> 2)
- - (SUPGAIN_ERROR_PARAM_B >> 2);
- aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B >> 2)
- - (SUPGAIN_ERROR_PARAM_D >> 2);
- } else if (aecm->echoMode == 2)
- {
- aecm->aecmCore->supGain = SUPGAIN_DEFAULT >> 1;
- aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT >> 1;
- aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A >> 1;
- aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D >> 1;
- aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A >> 1)
- - (SUPGAIN_ERROR_PARAM_B >> 1);
- aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B >> 1)
- - (SUPGAIN_ERROR_PARAM_D >> 1);
- } else if (aecm->echoMode == 3)
- {
- aecm->aecmCore->supGain = SUPGAIN_DEFAULT;
- aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT;
- aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A;
- aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D;
- aecm->aecmCore->supGainErrParamDiffAB = SUPGAIN_ERROR_PARAM_A - SUPGAIN_ERROR_PARAM_B;
- aecm->aecmCore->supGainErrParamDiffBD = SUPGAIN_ERROR_PARAM_B - SUPGAIN_ERROR_PARAM_D;
- } else if (aecm->echoMode == 4)
- {
- aecm->aecmCore->supGain = SUPGAIN_DEFAULT << 1;
- aecm->aecmCore->supGainOld = SUPGAIN_DEFAULT << 1;
- aecm->aecmCore->supGainErrParamA = SUPGAIN_ERROR_PARAM_A << 1;
- aecm->aecmCore->supGainErrParamD = SUPGAIN_ERROR_PARAM_D << 1;
- aecm->aecmCore->supGainErrParamDiffAB = (SUPGAIN_ERROR_PARAM_A << 1)
- - (SUPGAIN_ERROR_PARAM_B << 1);
- aecm->aecmCore->supGainErrParamDiffBD = (SUPGAIN_ERROR_PARAM_B << 1)
- - (SUPGAIN_ERROR_PARAM_D << 1);
- }
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_get_config(void *aecmInst, AecmConfig *config)
-{
- aecmob_t *aecm = aecmInst;
-
- if (aecm == NULL)
- {
- return -1;
- }
-
- if (config == NULL)
- {
- aecm->lastError = AECM_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (aecm->initFlag != kInitCheck)
- {
- aecm->lastError = AECM_UNINITIALIZED_ERROR;
- return -1;
- }
-
- config->cngMode = aecm->aecmCore->cngMode;
- config->echoMode = aecm->echoMode;
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_InitEchoPath(void* aecmInst,
- const void* echo_path,
- size_t size_bytes)
-{
- aecmob_t *aecm = aecmInst;
- const WebRtc_Word16* echo_path_ptr = echo_path;
-
- if ((aecm == NULL) || (echo_path == NULL))
- {
- aecm->lastError = AECM_NULL_POINTER_ERROR;
- return -1;
- }
- if (size_bytes != WebRtcAecm_echo_path_size_bytes())
- {
- // Input channel size does not match the size of AECM
- aecm->lastError = AECM_BAD_PARAMETER_ERROR;
- return -1;
- }
- if (aecm->initFlag != kInitCheck)
- {
- aecm->lastError = AECM_UNINITIALIZED_ERROR;
- return -1;
- }
-
- WebRtcAecm_InitEchoPathCore(aecm->aecmCore, echo_path_ptr);
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAecm_GetEchoPath(void* aecmInst,
- void* echo_path,
- size_t size_bytes)
-{
- aecmob_t *aecm = aecmInst;
- WebRtc_Word16* echo_path_ptr = echo_path;
-
- if ((aecm == NULL) || (echo_path == NULL))
- {
- aecm->lastError = AECM_NULL_POINTER_ERROR;
- return -1;
- }
- if (size_bytes != WebRtcAecm_echo_path_size_bytes())
- {
- // Input channel size does not match the size of AECM
- aecm->lastError = AECM_BAD_PARAMETER_ERROR;
- return -1;
- }
- if (aecm->initFlag != kInitCheck)
- {
- aecm->lastError = AECM_UNINITIALIZED_ERROR;
- return -1;
- }
-
- memcpy(echo_path_ptr, aecm->aecmCore->channelStored, size_bytes);
- return 0;
-}
-
-size_t WebRtcAecm_echo_path_size_bytes()
-{
- return (PART_LEN1 * sizeof(WebRtc_Word16));
-}
-
-WebRtc_Word32 WebRtcAecm_get_error_code(void *aecmInst)
-{
- aecmob_t *aecm = aecmInst;
-
- if (aecm == NULL)
- {
- return -1;
- }
-
- return aecm->lastError;
-}
-
-static int WebRtcAecm_EstBufDelay(aecmob_t *aecm, short msInSndCardBuf)
-{
- short delayNew, nSampSndCard;
- short nSampFar = (short) WebRtc_available_read(aecm->farendBuf);
- short diff;
-
- nSampSndCard = msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
-
- delayNew = nSampSndCard - nSampFar;
-
- if (delayNew < FRAME_LEN)
- {
- WebRtc_MoveReadPtr(aecm->farendBuf, FRAME_LEN);
- delayNew += FRAME_LEN;
- }
-
- aecm->filtDelay = WEBRTC_SPL_MAX(0, (8 * aecm->filtDelay + 2 * delayNew) / 10);
-
- diff = aecm->filtDelay - aecm->knownDelay;
- if (diff > 224)
- {
- if (aecm->lastDelayDiff < 96)
- {
- aecm->timeForDelayChange = 0;
- } else
- {
- aecm->timeForDelayChange++;
- }
- } else if (diff < 96 && aecm->knownDelay > 0)
- {
- if (aecm->lastDelayDiff > 224)
- {
- aecm->timeForDelayChange = 0;
- } else
- {
- aecm->timeForDelayChange++;
- }
- } else
- {
- aecm->timeForDelayChange = 0;
- }
- aecm->lastDelayDiff = diff;
-
- if (aecm->timeForDelayChange > 25)
- {
- aecm->knownDelay = WEBRTC_SPL_MAX((int)aecm->filtDelay - 160, 0);
- }
- return 0;
-}
-
-static int WebRtcAecm_DelayComp(aecmob_t *aecm)
-{
- int nSampFar = (int) WebRtc_available_read(aecm->farendBuf);
- int nSampSndCard, delayNew, nSampAdd;
- const int maxStuffSamp = 10 * FRAME_LEN;
-
- nSampSndCard = aecm->msInSndCardBuf * kSampMsNb * aecm->aecmCore->mult;
- delayNew = nSampSndCard - nSampFar;
-
- if (delayNew > FAR_BUF_LEN - FRAME_LEN * aecm->aecmCore->mult)
- {
- // The difference of the buffer sizes is larger than the maximum
- // allowed known delay. Compensate by stuffing the buffer.
- nSampAdd = (int)(WEBRTC_SPL_MAX(((nSampSndCard >> 1) - nSampFar),
- FRAME_LEN));
- nSampAdd = WEBRTC_SPL_MIN(nSampAdd, maxStuffSamp);
-
- WebRtc_MoveReadPtr(aecm->farendBuf, -nSampAdd);
- aecm->delayChange = 1; // the delay needs to be updated
- }
-
- return 0;
-}
diff --git a/trunk/src/modules/audio_processing/aecm/include/echo_control_mobile.h b/trunk/src/modules/audio_processing/aecm/include/echo_control_mobile.h
deleted file mode 100644
index da0ad86..0000000
--- a/trunk/src/modules/audio_processing/aecm/include/echo_control_mobile.h
+++ /dev/null
@@ -1,233 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AECM_INCLUDE_ECHO_CONTROL_MOBILE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AECM_INCLUDE_ECHO_CONTROL_MOBILE_H_
-
-#include "typedefs.h"
-
-enum {
- AecmFalse = 0,
- AecmTrue
-};
-
-// Errors
-#define AECM_UNSPECIFIED_ERROR 12000
-#define AECM_UNSUPPORTED_FUNCTION_ERROR 12001
-#define AECM_UNINITIALIZED_ERROR 12002
-#define AECM_NULL_POINTER_ERROR 12003
-#define AECM_BAD_PARAMETER_ERROR 12004
-
-// Warnings
-#define AECM_BAD_PARAMETER_WARNING 12100
-
-typedef struct {
- WebRtc_Word16 cngMode; // AECM_FALSE, AECM_TRUE (default)
- WebRtc_Word16 echoMode; // 0, 1, 2, 3 (default), 4
-} AecmConfig;
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/*
- * Allocates the memory needed by the AECM. The memory needs to be
- * initialized separately using the WebRtcAecm_Init() function.
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void **aecmInst Pointer to the AECM instance to be
- * created and initialized
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAecm_Create(void **aecmInst);
-
-/*
- * This function releases the memory allocated by WebRtcAecm_Create()
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void *aecmInst Pointer to the AECM instance
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAecm_Free(void *aecmInst);
-
-/*
- * Initializes an AECM instance.
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void *aecmInst Pointer to the AECM instance
- * WebRtc_Word32 sampFreq Sampling frequency of data
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAecm_Init(void* aecmInst,
- WebRtc_Word32 sampFreq);
-
-/*
- * Inserts an 80 or 160 sample block of data into the farend buffer.
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void *aecmInst Pointer to the AECM instance
- * WebRtc_Word16 *farend In buffer containing one frame of
- * farend signal
- * WebRtc_Word16 nrOfSamples Number of samples in farend buffer
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAecm_BufferFarend(void* aecmInst,
- const WebRtc_Word16* farend,
- WebRtc_Word16 nrOfSamples);
-
-/*
- * Runs the AECM on an 80 or 160 sample blocks of data.
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void *aecmInst Pointer to the AECM instance
- * WebRtc_Word16 *nearendNoisy In buffer containing one frame of
- * reference nearend+echo signal. If
- * noise reduction is active, provide
- * the noisy signal here.
- * WebRtc_Word16 *nearendClean In buffer containing one frame of
- * nearend+echo signal. If noise
- * reduction is active, provide the
- * clean signal here. Otherwise pass a
- * NULL pointer.
- * WebRtc_Word16 nrOfSamples Number of samples in nearend buffer
- * WebRtc_Word16 msInSndCardBuf Delay estimate for sound card and
- * system buffers
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * WebRtc_Word16 *out Out buffer, one frame of processed nearend
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAecm_Process(void* aecmInst,
- const WebRtc_Word16* nearendNoisy,
- const WebRtc_Word16* nearendClean,
- WebRtc_Word16* out,
- WebRtc_Word16 nrOfSamples,
- WebRtc_Word16 msInSndCardBuf);
-
-/*
- * This function enables the user to set certain parameters on-the-fly
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void *aecmInst Pointer to the AECM instance
- * AecmConfig config Config instance that contains all
- * properties to be set
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAecm_set_config(void* aecmInst,
- AecmConfig config);
-
-/*
- * This function enables the user to set certain parameters on-the-fly
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void *aecmInst Pointer to the AECM instance
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * AecmConfig *config Pointer to the config instance that
- * all properties will be written to
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAecm_get_config(void *aecmInst,
- AecmConfig *config);
-
-/*
- * This function enables the user to set the echo path on-the-fly.
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void* aecmInst Pointer to the AECM instance
- * void* echo_path Pointer to the echo path to be set
- * size_t size_bytes Size in bytes of the echo path
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAecm_InitEchoPath(void* aecmInst,
- const void* echo_path,
- size_t size_bytes);
-
-/*
- * This function enables the user to get the currently used echo path
- * on-the-fly
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void* aecmInst Pointer to the AECM instance
- * void* echo_path Pointer to echo path
- * size_t size_bytes Size in bytes of the echo path
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return 0: OK
- * -1: error
- */
-WebRtc_Word32 WebRtcAecm_GetEchoPath(void* aecmInst,
- void* echo_path,
- size_t size_bytes);
-
-/*
- * This function enables the user to get the echo path size in bytes
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * size_t return : size in bytes
- */
-size_t WebRtcAecm_echo_path_size_bytes();
-
-/*
- * Gets the last error code.
- *
- * Inputs Description
- * -------------------------------------------------------------------
- * void *aecmInst Pointer to the AECM instance
- *
- * Outputs Description
- * -------------------------------------------------------------------
- * WebRtc_Word32 return 11000-11100: error code
- */
-WebRtc_Word32 WebRtcAecm_get_error_code(void *aecmInst);
-
-#ifdef __cplusplus
-}
-#endif
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AECM_INCLUDE_ECHO_CONTROL_MOBILE_H_
diff --git a/trunk/src/modules/audio_processing/agc/Android.mk b/trunk/src/modules/audio_processing/agc/Android.mk
deleted file mode 100644
index 5eceb14..0000000
--- a/trunk/src/modules/audio_processing/agc/Android.mk
+++ /dev/null
@@ -1,40 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_agc
-LOCAL_MODULE_TAGS := optional
-LOCAL_SRC_FILES := \
- analog_agc.c \
- digital_agc.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include \
- $(LOCAL_PATH)/../../.. \
- $(LOCAL_PATH)/../../../common_audio/signal_processing/include
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_processing/agc/agc.gypi b/trunk/src/modules/audio_processing/agc/agc.gypi
deleted file mode 100644
index f96f237..0000000
--- a/trunk/src/modules/audio_processing/agc/agc.gypi
+++ /dev/null
@@ -1,34 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'agc',
- 'type': '<(library)',
- 'dependencies': [
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- ],
- 'include_dirs': [
- 'include',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- ],
- },
- 'sources': [
- 'include/gain_control.h',
- 'analog_agc.c',
- 'analog_agc.h',
- 'digital_agc.c',
- 'digital_agc.h',
- ],
- },
- ],
-}
diff --git a/trunk/src/modules/audio_processing/agc/analog_agc.c b/trunk/src/modules/audio_processing/agc/analog_agc.c
deleted file mode 100644
index 558a6cb..0000000
--- a/trunk/src/modules/audio_processing/agc/analog_agc.c
+++ /dev/null
@@ -1,1694 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/* analog_agc.c
- *
- * Using a feedback system, determines an appropriate analog volume level
- * given an input signal and current volume level. Targets a conservative
- * signal level and is intended for use with a digital AGC to apply
- * additional gain.
- *
- */
-
-#include <assert.h>
-#include <stdlib.h>
-#ifdef AGC_DEBUG //test log
-#include <stdio.h>
-#endif
-#include "analog_agc.h"
-
-/* The slope of in Q13*/
-static const WebRtc_Word16 kSlope1[8] = {21793, 12517, 7189, 4129, 2372, 1362, 472, 78};
-
-/* The offset in Q14 */
-static const WebRtc_Word16 kOffset1[8] = {25395, 23911, 22206, 20737, 19612, 18805, 17951,
- 17367};
-
-/* The slope of in Q13*/
-static const WebRtc_Word16 kSlope2[8] = {2063, 1731, 1452, 1218, 1021, 857, 597, 337};
-
-/* The offset in Q14 */
-static const WebRtc_Word16 kOffset2[8] = {18432, 18379, 18290, 18177, 18052, 17920, 17670,
- 17286};
-
-static const WebRtc_Word16 kMuteGuardTimeMs = 8000;
-static const WebRtc_Word16 kInitCheck = 42;
-
-/* Default settings if config is not used */
-#define AGC_DEFAULT_TARGET_LEVEL 3
-#define AGC_DEFAULT_COMP_GAIN 9
-/* This is the target level for the analog part in ENV scale. To convert to RMS scale you
- * have to add OFFSET_ENV_TO_RMS.
- */
-#define ANALOG_TARGET_LEVEL 11
-#define ANALOG_TARGET_LEVEL_2 5 // ANALOG_TARGET_LEVEL / 2
-/* Offset between RMS scale (analog part) and ENV scale (digital part). This value actually
- * varies with the FIXED_ANALOG_TARGET_LEVEL, hence we should in the future replace it with
- * a table.
- */
-#define OFFSET_ENV_TO_RMS 9
-/* The reference input level at which the digital part gives an output of targetLevelDbfs
- * (desired level) if we have no compression gain. This level should be set high enough not
- * to compress the peaks due to the dynamics.
- */
-#define DIGITAL_REF_AT_0_COMP_GAIN 4
-/* Speed of reference level decrease.
- */
-#define DIFF_REF_TO_ANALOG 5
-
-#ifdef MIC_LEVEL_FEEDBACK
-#define NUM_BLOCKS_IN_SAT_BEFORE_CHANGE_TARGET 7
-#endif
-/* Size of analog gain table */
-#define GAIN_TBL_LEN 32
-/* Matlab code:
- * fprintf(1, '\t%i, %i, %i, %i,\n', round(10.^(linspace(0,10,32)/20) * 2^12));
- */
-/* Q12 */
-static const WebRtc_UWord16 kGainTableAnalog[GAIN_TBL_LEN] = {4096, 4251, 4412, 4579, 4752,
- 4932, 5118, 5312, 5513, 5722, 5938, 6163, 6396, 6638, 6889, 7150, 7420, 7701, 7992,
- 8295, 8609, 8934, 9273, 9623, 9987, 10365, 10758, 11165, 11587, 12025, 12480, 12953};
-
-/* Gain/Suppression tables for virtual Mic (in Q10) */
-static const WebRtc_UWord16 kGainTableVirtualMic[128] = {1052, 1081, 1110, 1141, 1172, 1204,
- 1237, 1271, 1305, 1341, 1378, 1416, 1454, 1494, 1535, 1577, 1620, 1664, 1710, 1757,
- 1805, 1854, 1905, 1957, 2010, 2065, 2122, 2180, 2239, 2301, 2364, 2428, 2495, 2563,
- 2633, 2705, 2779, 2855, 2933, 3013, 3096, 3180, 3267, 3357, 3449, 3543, 3640, 3739,
- 3842, 3947, 4055, 4166, 4280, 4397, 4517, 4640, 4767, 4898, 5032, 5169, 5311, 5456,
- 5605, 5758, 5916, 6078, 6244, 6415, 6590, 6770, 6956, 7146, 7341, 7542, 7748, 7960,
- 8178, 8402, 8631, 8867, 9110, 9359, 9615, 9878, 10148, 10426, 10711, 11004, 11305,
- 11614, 11932, 12258, 12593, 12938, 13292, 13655, 14029, 14412, 14807, 15212, 15628,
- 16055, 16494, 16945, 17409, 17885, 18374, 18877, 19393, 19923, 20468, 21028, 21603,
- 22194, 22801, 23425, 24065, 24724, 25400, 26095, 26808, 27541, 28295, 29069, 29864,
- 30681, 31520, 32382};
-static const WebRtc_UWord16 kSuppressionTableVirtualMic[128] = {1024, 1006, 988, 970, 952,
- 935, 918, 902, 886, 870, 854, 839, 824, 809, 794, 780, 766, 752, 739, 726, 713, 700,
- 687, 675, 663, 651, 639, 628, 616, 605, 594, 584, 573, 563, 553, 543, 533, 524, 514,
- 505, 496, 487, 478, 470, 461, 453, 445, 437, 429, 421, 414, 406, 399, 392, 385, 378,
- 371, 364, 358, 351, 345, 339, 333, 327, 321, 315, 309, 304, 298, 293, 288, 283, 278,
- 273, 268, 263, 258, 254, 249, 244, 240, 236, 232, 227, 223, 219, 215, 211, 208, 204,
- 200, 197, 193, 190, 186, 183, 180, 176, 173, 170, 167, 164, 161, 158, 155, 153, 150,
- 147, 145, 142, 139, 137, 134, 132, 130, 127, 125, 123, 121, 118, 116, 114, 112, 110,
- 108, 106, 104, 102};
-
-/* Table for target energy levels. Values in Q(-7)
- * Matlab code
- * targetLevelTable = fprintf('%d,\t%d,\t%d,\t%d,\n', round((32767*10.^(-(0:63)'/20)).^2*16/2^7) */
-
-static const WebRtc_Word32 kTargetLevelTable[64] = {134209536, 106606424, 84680493, 67264106,
- 53429779, 42440782, 33711911, 26778323, 21270778, 16895980, 13420954, 10660642,
- 8468049, 6726411, 5342978, 4244078, 3371191, 2677832, 2127078, 1689598, 1342095,
- 1066064, 846805, 672641, 534298, 424408, 337119, 267783, 212708, 168960, 134210,
- 106606, 84680, 67264, 53430, 42441, 33712, 26778, 21271, 16896, 13421, 10661, 8468,
- 6726, 5343, 4244, 3371, 2678, 2127, 1690, 1342, 1066, 847, 673, 534, 424, 337, 268,
- 213, 169, 134, 107, 85, 67};
-
-int WebRtcAgc_AddMic(void *state, WebRtc_Word16 *in_mic, WebRtc_Word16 *in_mic_H,
- WebRtc_Word16 samples)
-{
- WebRtc_Word32 nrg, max_nrg, sample, tmp32;
- WebRtc_Word32 *ptr;
- WebRtc_UWord16 targetGainIdx, gain;
- WebRtc_Word16 i, n, L, M, subFrames, tmp16, tmp_speech[16];
- Agc_t *stt;
- stt = (Agc_t *)state;
-
- //default/initial values corresponding to 10ms for wb and swb
- M = 10;
- L = 16;
- subFrames = 160;
-
- if (stt->fs == 8000)
- {
- if (samples == 80)
- {
- subFrames = 80;
- M = 10;
- L = 8;
- } else if (samples == 160)
- {
- subFrames = 80;
- M = 20;
- L = 8;
- } else
- {
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt,
- "AGC->add_mic, frame %d: Invalid number of samples\n\n",
- (stt->fcount + 1));
-#endif
- return -1;
- }
- } else if (stt->fs == 16000)
- {
- if (samples == 160)
- {
- subFrames = 160;
- M = 10;
- L = 16;
- } else if (samples == 320)
- {
- subFrames = 160;
- M = 20;
- L = 16;
- } else
- {
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt,
- "AGC->add_mic, frame %d: Invalid number of samples\n\n",
- (stt->fcount + 1));
-#endif
- return -1;
- }
- } else if (stt->fs == 32000)
- {
- /* SWB is processed as 160 sample for L and H bands */
- if (samples == 160)
- {
- subFrames = 160;
- M = 10;
- L = 16;
- } else
- {
-#ifdef AGC_DEBUG
- fprintf(stt->fpt,
- "AGC->add_mic, frame %d: Invalid sample rate\n\n",
- (stt->fcount + 1));
-#endif
- return -1;
- }
- }
-
- /* Check for valid pointers based on sampling rate */
- if ((stt->fs == 32000) && (in_mic_H == NULL))
- {
- return -1;
- }
- /* Check for valid pointer for low band */
- if (in_mic == NULL)
- {
- return -1;
- }
-
- /* apply slowly varying digital gain */
- if (stt->micVol > stt->maxAnalog)
- {
- /* |maxLevel| is strictly >= |micVol|, so this condition should be
- * satisfied here, ensuring there is no divide-by-zero. */
- assert(stt->maxLevel > stt->maxAnalog);
-
- /* Q1 */
- tmp16 = (WebRtc_Word16)(stt->micVol - stt->maxAnalog);
- tmp32 = WEBRTC_SPL_MUL_16_16(GAIN_TBL_LEN - 1, tmp16);
- tmp16 = (WebRtc_Word16)(stt->maxLevel - stt->maxAnalog);
- targetGainIdx = (WebRtc_UWord16)WEBRTC_SPL_DIV(tmp32, tmp16);
- assert(targetGainIdx < GAIN_TBL_LEN);
-
- /* Increment through the table towards the target gain.
- * If micVol drops below maxAnalog, we allow the gain
- * to be dropped immediately. */
- if (stt->gainTableIdx < targetGainIdx)
- {
- stt->gainTableIdx++;
- } else if (stt->gainTableIdx > targetGainIdx)
- {
- stt->gainTableIdx--;
- }
-
- /* Q12 */
- gain = kGainTableAnalog[stt->gainTableIdx];
-
- for (i = 0; i < samples; i++)
- {
- // For lower band
- tmp32 = WEBRTC_SPL_MUL_16_U16(in_mic[i], gain);
- sample = WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
- if (sample > 32767)
- {
- in_mic[i] = 32767;
- } else if (sample < -32768)
- {
- in_mic[i] = -32768;
- } else
- {
- in_mic[i] = (WebRtc_Word16)sample;
- }
-
- // For higher band
- if (stt->fs == 32000)
- {
- tmp32 = WEBRTC_SPL_MUL_16_U16(in_mic_H[i], gain);
- sample = WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
- if (sample > 32767)
- {
- in_mic_H[i] = 32767;
- } else if (sample < -32768)
- {
- in_mic_H[i] = -32768;
- } else
- {
- in_mic_H[i] = (WebRtc_Word16)sample;
- }
- }
- }
- } else
- {
- stt->gainTableIdx = 0;
- }
-
- /* compute envelope */
- if ((M == 10) && (stt->inQueue > 0))
- {
- ptr = stt->env[1];
- } else
- {
- ptr = stt->env[0];
- }
-
- for (i = 0; i < M; i++)
- {
- /* iterate over samples */
- max_nrg = 0;
- for (n = 0; n < L; n++)
- {
- nrg = WEBRTC_SPL_MUL_16_16(in_mic[i * L + n], in_mic[i * L + n]);
- if (nrg > max_nrg)
- {
- max_nrg = nrg;
- }
- }
- ptr[i] = max_nrg;
- }
-
- /* compute energy */
- if ((M == 10) && (stt->inQueue > 0))
- {
- ptr = stt->Rxx16w32_array[1];
- } else
- {
- ptr = stt->Rxx16w32_array[0];
- }
-
- for (i = 0; i < WEBRTC_SPL_RSHIFT_W16(M, 1); i++)
- {
- if (stt->fs == 16000)
- {
- WebRtcSpl_DownsampleBy2(&in_mic[i * 32], 32, tmp_speech, stt->filterState);
- } else
- {
- memcpy(tmp_speech, &in_mic[i * 16], 16 * sizeof(short));
- }
- /* Compute energy in blocks of 16 samples */
- ptr[i] = WebRtcSpl_DotProductWithScale(tmp_speech, tmp_speech, 16, 4);
- }
-
- /* update queue information */
- if ((stt->inQueue == 0) && (M == 10))
- {
- stt->inQueue = 1;
- } else
- {
- stt->inQueue = 2;
- }
-
- /* call VAD (use low band only) */
- for (i = 0; i < samples; i += subFrames)
- {
- WebRtcAgc_ProcessVad(&stt->vadMic, &in_mic[i], subFrames);
- }
-
- return 0;
-}
-
-int WebRtcAgc_AddFarend(void *state, const WebRtc_Word16 *in_far, WebRtc_Word16 samples)
-{
- WebRtc_Word32 errHandle = 0;
- WebRtc_Word16 i, subFrames;
- Agc_t *stt;
- stt = (Agc_t *)state;
-
- if (stt == NULL)
- {
- return -1;
- }
-
- if (stt->fs == 8000)
- {
- if ((samples != 80) && (samples != 160))
- {
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt,
- "AGC->add_far_end, frame %d: Invalid number of samples\n\n",
- stt->fcount);
-#endif
- return -1;
- }
- subFrames = 80;
- } else if (stt->fs == 16000)
- {
- if ((samples != 160) && (samples != 320))
- {
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt,
- "AGC->add_far_end, frame %d: Invalid number of samples\n\n",
- stt->fcount);
-#endif
- return -1;
- }
- subFrames = 160;
- } else if (stt->fs == 32000)
- {
- if ((samples != 160) && (samples != 320))
- {
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt,
- "AGC->add_far_end, frame %d: Invalid number of samples\n\n",
- stt->fcount);
-#endif
- return -1;
- }
- subFrames = 160;
- } else
- {
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt,
- "AGC->add_far_end, frame %d: Invalid sample rate\n\n",
- stt->fcount + 1);
-#endif
- return -1;
- }
-
- for (i = 0; i < samples; i += subFrames)
- {
- errHandle += WebRtcAgc_AddFarendToDigital(&stt->digitalAgc, &in_far[i], subFrames);
- }
-
- return errHandle;
-}
-
-int WebRtcAgc_VirtualMic(void *agcInst, WebRtc_Word16 *in_near, WebRtc_Word16 *in_near_H,
- WebRtc_Word16 samples, WebRtc_Word32 micLevelIn,
- WebRtc_Word32 *micLevelOut)
-{
- WebRtc_Word32 tmpFlt, micLevelTmp, gainIdx;
- WebRtc_UWord16 gain;
- WebRtc_Word16 ii;
- Agc_t *stt;
-
- WebRtc_UWord32 nrg;
- WebRtc_Word16 sampleCntr;
- WebRtc_UWord32 frameNrg = 0;
- WebRtc_UWord32 frameNrgLimit = 5500;
- WebRtc_Word16 numZeroCrossing = 0;
- const WebRtc_Word16 kZeroCrossingLowLim = 15;
- const WebRtc_Word16 kZeroCrossingHighLim = 20;
-
- stt = (Agc_t *)agcInst;
-
- /*
- * Before applying gain decide if this is a low-level signal.
- * The idea is that digital AGC will not adapt to low-level
- * signals.
- */
- if (stt->fs != 8000)
- {
- frameNrgLimit = frameNrgLimit << 1;
- }
-
- frameNrg = WEBRTC_SPL_MUL_16_16(in_near[0], in_near[0]);
- for (sampleCntr = 1; sampleCntr < samples; sampleCntr++)
- {
-
- // increment frame energy if it is less than the limit
- // the correct value of the energy is not important
- if (frameNrg < frameNrgLimit)
- {
- nrg = WEBRTC_SPL_MUL_16_16(in_near[sampleCntr], in_near[sampleCntr]);
- frameNrg += nrg;
- }
-
- // Count the zero crossings
- numZeroCrossing += ((in_near[sampleCntr] ^ in_near[sampleCntr - 1]) < 0);
- }
-
- if ((frameNrg < 500) || (numZeroCrossing <= 5))
- {
- stt->lowLevelSignal = 1;
- } else if (numZeroCrossing <= kZeroCrossingLowLim)
- {
- stt->lowLevelSignal = 0;
- } else if (frameNrg <= frameNrgLimit)
- {
- stt->lowLevelSignal = 1;
- } else if (numZeroCrossing >= kZeroCrossingHighLim)
- {
- stt->lowLevelSignal = 1;
- } else
- {
- stt->lowLevelSignal = 0;
- }
-
- micLevelTmp = WEBRTC_SPL_LSHIFT_W32(micLevelIn, stt->scale);
- /* Set desired level */
- gainIdx = stt->micVol;
- if (stt->micVol > stt->maxAnalog)
- {
- gainIdx = stt->maxAnalog;
- }
- if (micLevelTmp != stt->micRef)
- {
- /* Something has happened with the physical level, restart. */
- stt->micRef = micLevelTmp;
- stt->micVol = 127;
- *micLevelOut = 127;
- stt->micGainIdx = 127;
- gainIdx = 127;
- }
- /* Pre-process the signal to emulate the microphone level. */
- /* Take one step at a time in the gain table. */
- if (gainIdx > 127)
- {
- gain = kGainTableVirtualMic[gainIdx - 128];
- } else
- {
- gain = kSuppressionTableVirtualMic[127 - gainIdx];
- }
- for (ii = 0; ii < samples; ii++)
- {
- tmpFlt = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_U16(in_near[ii], gain), 10);
- if (tmpFlt > 32767)
- {
- tmpFlt = 32767;
- gainIdx--;
- if (gainIdx >= 127)
- {
- gain = kGainTableVirtualMic[gainIdx - 127];
- } else
- {
- gain = kSuppressionTableVirtualMic[127 - gainIdx];
- }
- }
- if (tmpFlt < -32768)
- {
- tmpFlt = -32768;
- gainIdx--;
- if (gainIdx >= 127)
- {
- gain = kGainTableVirtualMic[gainIdx - 127];
- } else
- {
- gain = kSuppressionTableVirtualMic[127 - gainIdx];
- }
- }
- in_near[ii] = (WebRtc_Word16)tmpFlt;
- if (stt->fs == 32000)
- {
- tmpFlt = WEBRTC_SPL_MUL_16_U16(in_near_H[ii], gain);
- tmpFlt = WEBRTC_SPL_RSHIFT_W32(tmpFlt, 10);
- if (tmpFlt > 32767)
- {
- tmpFlt = 32767;
- }
- if (tmpFlt < -32768)
- {
- tmpFlt = -32768;
- }
- in_near_H[ii] = (WebRtc_Word16)tmpFlt;
- }
- }
- /* Set the level we (finally) used */
- stt->micGainIdx = gainIdx;
-// *micLevelOut = stt->micGainIdx;
- *micLevelOut = WEBRTC_SPL_RSHIFT_W32(stt->micGainIdx, stt->scale);
- /* Add to Mic as if it was the output from a true microphone */
- if (WebRtcAgc_AddMic(agcInst, in_near, in_near_H, samples) != 0)
- {
- return -1;
- }
- return 0;
-}
-
-void WebRtcAgc_UpdateAgcThresholds(Agc_t *stt)
-{
-
- WebRtc_Word16 tmp16;
-#ifdef MIC_LEVEL_FEEDBACK
- int zeros;
-
- if (stt->micLvlSat)
- {
- /* Lower the analog target level since we have reached its maximum */
- zeros = WebRtcSpl_NormW32(stt->Rxx160_LPw32);
- stt->targetIdxOffset = WEBRTC_SPL_RSHIFT_W16((3 * zeros) - stt->targetIdx - 2, 2);
- }
-#endif
-
- /* Set analog target level in envelope dBOv scale */
- tmp16 = (DIFF_REF_TO_ANALOG * stt->compressionGaindB) + ANALOG_TARGET_LEVEL_2;
- tmp16 = WebRtcSpl_DivW32W16ResW16((WebRtc_Word32)tmp16, ANALOG_TARGET_LEVEL);
- stt->analogTarget = DIGITAL_REF_AT_0_COMP_GAIN + tmp16;
- if (stt->analogTarget < DIGITAL_REF_AT_0_COMP_GAIN)
- {
- stt->analogTarget = DIGITAL_REF_AT_0_COMP_GAIN;
- }
- if (stt->agcMode == kAgcModeFixedDigital)
- {
- /* Adjust for different parameter interpretation in FixedDigital mode */
- stt->analogTarget = stt->compressionGaindB;
- }
-#ifdef MIC_LEVEL_FEEDBACK
- stt->analogTarget += stt->targetIdxOffset;
-#endif
- /* Since the offset between RMS and ENV is not constant, we should make this into a
- * table, but for now, we'll stick with a constant, tuned for the chosen analog
- * target level.
- */
- stt->targetIdx = ANALOG_TARGET_LEVEL + OFFSET_ENV_TO_RMS;
-#ifdef MIC_LEVEL_FEEDBACK
- stt->targetIdx += stt->targetIdxOffset;
-#endif
- /* Analog adaptation limits */
- /* analogTargetLevel = round((32767*10^(-targetIdx/20))^2*16/2^7) */
- stt->analogTargetLevel = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx]; /* ex. -20 dBov */
- stt->startUpperLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx - 1];/* -19 dBov */
- stt->startLowerLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx + 1];/* -21 dBov */
- stt->upperPrimaryLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx - 2];/* -18 dBov */
- stt->lowerPrimaryLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx + 2];/* -22 dBov */
- stt->upperSecondaryLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx - 5];/* -15 dBov */
- stt->lowerSecondaryLimit = RXX_BUFFER_LEN * kTargetLevelTable[stt->targetIdx + 5];/* -25 dBov */
- stt->upperLimit = stt->startUpperLimit;
- stt->lowerLimit = stt->startLowerLimit;
-}
-
-void WebRtcAgc_SaturationCtrl(Agc_t *stt, WebRtc_UWord8 *saturated, WebRtc_Word32 *env)
-{
- WebRtc_Word16 i, tmpW16;
-
- /* Check if the signal is saturated */
- for (i = 0; i < 10; i++)
- {
- tmpW16 = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(env[i], 20);
- if (tmpW16 > 875)
- {
- stt->envSum += tmpW16;
- }
- }
-
- if (stt->envSum > 25000)
- {
- *saturated = 1;
- stt->envSum = 0;
- }
-
- /* stt->envSum *= 0.99; */
- stt->envSum = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(stt->envSum,
- (WebRtc_Word16)32440, 15);
-}
-
-void WebRtcAgc_ZeroCtrl(Agc_t *stt, WebRtc_Word32 *inMicLevel, WebRtc_Word32 *env)
-{
- WebRtc_Word16 i;
- WebRtc_Word32 tmp32 = 0;
- WebRtc_Word32 midVal;
-
- /* Is the input signal zero? */
- for (i = 0; i < 10; i++)
- {
- tmp32 += env[i];
- }
-
- /* Each block is allowed to have a few non-zero
- * samples.
- */
- if (tmp32 < 500)
- {
- stt->msZero += 10;
- } else
- {
- stt->msZero = 0;
- }
-
- if (stt->muteGuardMs > 0)
- {
- stt->muteGuardMs -= 10;
- }
-
- if (stt->msZero > 500)
- {
- stt->msZero = 0;
-
- /* Increase microphone level only if it's less than 50% */
- midVal = WEBRTC_SPL_RSHIFT_W32(stt->maxAnalog + stt->minLevel + 1, 1);
- if (*inMicLevel < midVal)
- {
- /* *inMicLevel *= 1.1; */
- tmp32 = WEBRTC_SPL_MUL(1126, *inMicLevel);
- *inMicLevel = WEBRTC_SPL_RSHIFT_W32(tmp32, 10);
- /* Reduces risk of a muted mic repeatedly triggering excessive levels due
- * to zero signal detection. */
- *inMicLevel = WEBRTC_SPL_MIN(*inMicLevel, stt->zeroCtrlMax);
- stt->micVol = *inMicLevel;
- }
-
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt,
- "\t\tAGC->zeroCntrl, frame %d: 500 ms under threshold, micVol:\n",
- stt->fcount, stt->micVol);
-#endif
-
- stt->activeSpeech = 0;
- stt->Rxx16_LPw32Max = 0;
-
- /* The AGC has a tendency (due to problems with the VAD parameters), to
- * vastly increase the volume after a muting event. This timer prevents
- * upwards adaptation for a short period. */
- stt->muteGuardMs = kMuteGuardTimeMs;
- }
-}
-
-void WebRtcAgc_SpeakerInactiveCtrl(Agc_t *stt)
-{
- /* Check if the near end speaker is inactive.
- * If that is the case the VAD threshold is
- * increased since the VAD speech model gets
- * more sensitive to any sound after a long
- * silence.
- */
-
- WebRtc_Word32 tmp32;
- WebRtc_Word16 vadThresh;
-
- if (stt->vadMic.stdLongTerm < 2500)
- {
- stt->vadThreshold = 1500;
- } else
- {
- vadThresh = kNormalVadThreshold;
- if (stt->vadMic.stdLongTerm < 4500)
- {
- /* Scale between min and max threshold */
- vadThresh += WEBRTC_SPL_RSHIFT_W16(4500 - stt->vadMic.stdLongTerm, 1);
- }
-
- /* stt->vadThreshold = (31 * stt->vadThreshold + vadThresh) / 32; */
- tmp32 = (WebRtc_Word32)vadThresh;
- tmp32 += WEBRTC_SPL_MUL_16_16((WebRtc_Word16)31, stt->vadThreshold);
- stt->vadThreshold = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 5);
- }
-}
-
-void WebRtcAgc_ExpCurve(WebRtc_Word16 volume, WebRtc_Word16 *index)
-{
- // volume in Q14
- // index in [0-7]
- /* 8 different curves */
- if (volume > 5243)
- {
- if (volume > 7864)
- {
- if (volume > 12124)
- {
- *index = 7;
- } else
- {
- *index = 6;
- }
- } else
- {
- if (volume > 6554)
- {
- *index = 5;
- } else
- {
- *index = 4;
- }
- }
- } else
- {
- if (volume > 2621)
- {
- if (volume > 3932)
- {
- *index = 3;
- } else
- {
- *index = 2;
- }
- } else
- {
- if (volume > 1311)
- {
- *index = 1;
- } else
- {
- *index = 0;
- }
- }
- }
-}
-
-WebRtc_Word32 WebRtcAgc_ProcessAnalog(void *state, WebRtc_Word32 inMicLevel,
- WebRtc_Word32 *outMicLevel,
- WebRtc_Word16 vadLogRatio,
- WebRtc_Word16 echo, WebRtc_UWord8 *saturationWarning)
-{
- WebRtc_UWord32 tmpU32;
- WebRtc_Word32 Rxx16w32, tmp32;
- WebRtc_Word32 inMicLevelTmp, lastMicVol;
- WebRtc_Word16 i;
- WebRtc_UWord8 saturated = 0;
- Agc_t *stt;
-
- stt = (Agc_t *)state;
- inMicLevelTmp = WEBRTC_SPL_LSHIFT_W32(inMicLevel, stt->scale);
-
- if (inMicLevelTmp > stt->maxAnalog)
- {
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: micLvl > maxAnalog\n", stt->fcount);
-#endif
- return -1;
- } else if (inMicLevelTmp < stt->minLevel)
- {
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt, "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel\n", stt->fcount);
-#endif
- return -1;
- }
-
- if (stt->firstCall == 0)
- {
- WebRtc_Word32 tmpVol;
- stt->firstCall = 1;
- tmp32 = WEBRTC_SPL_RSHIFT_W32((stt->maxLevel - stt->minLevel) * (WebRtc_Word32)51, 9);
- tmpVol = (stt->minLevel + tmp32);
-
- /* If the mic level is very low at start, increase it! */
- if ((inMicLevelTmp < tmpVol) && (stt->agcMode == kAgcModeAdaptiveAnalog))
- {
- inMicLevelTmp = tmpVol;
- }
- stt->micVol = inMicLevelTmp;
- }
-
- /* Set the mic level to the previous output value if there is digital input gain */
- if ((inMicLevelTmp == stt->maxAnalog) && (stt->micVol > stt->maxAnalog))
- {
- inMicLevelTmp = stt->micVol;
- }
-
- /* If the mic level was manually changed to a very low value raise it! */
- if ((inMicLevelTmp != stt->micVol) && (inMicLevelTmp < stt->minOutput))
- {
- tmp32 = WEBRTC_SPL_RSHIFT_W32((stt->maxLevel - stt->minLevel) * (WebRtc_Word32)51, 9);
- inMicLevelTmp = (stt->minLevel + tmp32);
- stt->micVol = inMicLevelTmp;
-#ifdef MIC_LEVEL_FEEDBACK
- //stt->numBlocksMicLvlSat = 0;
-#endif
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt,
- "\tAGC->ProcessAnalog, frame %d: micLvl < minLevel by manual decrease, raise vol\n",
- stt->fcount);
-#endif
- }
-
- if (inMicLevelTmp != stt->micVol)
- {
- // Incoming level mismatch; update our level.
- // This could be the case if the volume is changed manually, or if the
- // sound device has a low volume resolution.
- stt->micVol = inMicLevelTmp;
- }
-
- if (inMicLevelTmp > stt->maxLevel)
- {
- // Always allow the user to raise the volume above the maxLevel.
- stt->maxLevel = inMicLevelTmp;
- }
-
- // Store last value here, after we've taken care of manual updates etc.
- lastMicVol = stt->micVol;
-
- /* Checks if the signal is saturated. Also a check if individual samples
- * are larger than 12000 is done. If they are the counter for increasing
- * the volume level is set to -100ms
- */
- WebRtcAgc_SaturationCtrl(stt, &saturated, stt->env[0]);
-
- /* The AGC is always allowed to lower the level if the signal is saturated */
- if (saturated == 1)
- {
- /* Lower the recording level
- * Rxx160_LP is adjusted down because it is so slow it could
- * cause the AGC to make wrong decisions. */
- /* stt->Rxx160_LPw32 *= 0.875; */
- stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 3), 7);
-
- stt->zeroCtrlMax = stt->micVol;
-
- /* stt->micVol *= 0.903; */
- tmp32 = inMicLevelTmp - stt->minLevel;
- tmpU32 = WEBRTC_SPL_UMUL(29591, (WebRtc_UWord32)(tmp32));
- stt->micVol = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(tmpU32, 15) + stt->minLevel;
- if (stt->micVol > lastMicVol - 2)
- {
- stt->micVol = lastMicVol - 2;
- }
- inMicLevelTmp = stt->micVol;
-
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt,
- "\tAGC->ProcessAnalog, frame %d: saturated, micVol = %d\n",
- stt->fcount, stt->micVol);
-#endif
-
- if (stt->micVol < stt->minOutput)
- {
- *saturationWarning = 1;
- }
-
- /* Reset counter for decrease of volume level to avoid
- * decreasing too much. The saturation control can still
- * lower the level if needed. */
- stt->msTooHigh = -100;
-
- /* Enable the control mechanism to ensure that our measure,
- * Rxx160_LP, is in the correct range. This must be done since
- * the measure is very slow. */
- stt->activeSpeech = 0;
- stt->Rxx16_LPw32Max = 0;
-
- /* Reset to initial values */
- stt->msecSpeechInnerChange = kMsecSpeechInner;
- stt->msecSpeechOuterChange = kMsecSpeechOuter;
- stt->changeToSlowMode = 0;
-
- stt->muteGuardMs = 0;
-
- stt->upperLimit = stt->startUpperLimit;
- stt->lowerLimit = stt->startLowerLimit;
-#ifdef MIC_LEVEL_FEEDBACK
- //stt->numBlocksMicLvlSat = 0;
-#endif
- }
-
- /* Check if the input speech is zero. If so the mic volume
- * is increased. On some computers the input is zero up as high
- * level as 17% */
- WebRtcAgc_ZeroCtrl(stt, &inMicLevelTmp, stt->env[0]);
-
- /* Check if the near end speaker is inactive.
- * If that is the case the VAD threshold is
- * increased since the VAD speech model gets
- * more sensitive to any sound after a long
- * silence.
- */
- WebRtcAgc_SpeakerInactiveCtrl(stt);
-
- for (i = 0; i < 5; i++)
- {
- /* Computed on blocks of 16 samples */
-
- Rxx16w32 = stt->Rxx16w32_array[0][i];
-
- /* Rxx160w32 in Q(-7) */
- tmp32 = WEBRTC_SPL_RSHIFT_W32(Rxx16w32 - stt->Rxx16_vectorw32[stt->Rxx16pos], 3);
- stt->Rxx160w32 = stt->Rxx160w32 + tmp32;
- stt->Rxx16_vectorw32[stt->Rxx16pos] = Rxx16w32;
-
- /* Circular buffer */
- stt->Rxx16pos++;
- if (stt->Rxx16pos == RXX_BUFFER_LEN)
- {
- stt->Rxx16pos = 0;
- }
-
- /* Rxx16_LPw32 in Q(-4) */
- tmp32 = WEBRTC_SPL_RSHIFT_W32(Rxx16w32 - stt->Rxx16_LPw32, kAlphaShortTerm);
- stt->Rxx16_LPw32 = (stt->Rxx16_LPw32) + tmp32;
-
- if (vadLogRatio > stt->vadThreshold)
- {
- /* Speech detected! */
-
- /* Check if Rxx160_LP is in the correct range. If
- * it is too high/low then we set it to the maximum of
- * Rxx16_LPw32 during the first 200ms of speech.
- */
- if (stt->activeSpeech < 250)
- {
- stt->activeSpeech += 2;
-
- if (stt->Rxx16_LPw32 > stt->Rxx16_LPw32Max)
- {
- stt->Rxx16_LPw32Max = stt->Rxx16_LPw32;
- }
- } else if (stt->activeSpeech == 250)
- {
- stt->activeSpeech += 2;
- tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx16_LPw32Max, 3);
- stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, RXX_BUFFER_LEN);
- }
-
- tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160w32 - stt->Rxx160_LPw32, kAlphaLongTerm);
- stt->Rxx160_LPw32 = stt->Rxx160_LPw32 + tmp32;
-
- if (stt->Rxx160_LPw32 > stt->upperSecondaryLimit)
- {
- stt->msTooHigh += 2;
- stt->msTooLow = 0;
- stt->changeToSlowMode = 0;
-
- if (stt->msTooHigh > stt->msecSpeechOuterChange)
- {
- stt->msTooHigh = 0;
-
- /* Lower the recording level */
- /* Multiply by 0.828125 which corresponds to decreasing ~0.8dB */
- tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6);
- stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 53);
-
- /* Reduce the max gain to avoid excessive oscillation
- * (but never drop below the maximum analog level).
- * stt->maxLevel = (15 * stt->maxLevel + stt->micVol) / 16;
- */
- tmp32 = (15 * stt->maxLevel) + stt->micVol;
- stt->maxLevel = WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
- stt->maxLevel = WEBRTC_SPL_MAX(stt->maxLevel, stt->maxAnalog);
-
- stt->zeroCtrlMax = stt->micVol;
-
- /* 0.95 in Q15 */
- tmp32 = inMicLevelTmp - stt->minLevel;
- tmpU32 = WEBRTC_SPL_UMUL(31130, (WebRtc_UWord32)(tmp32));
- stt->micVol = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(tmpU32, 15) + stt->minLevel;
- if (stt->micVol > lastMicVol - 1)
- {
- stt->micVol = lastMicVol - 1;
- }
- inMicLevelTmp = stt->micVol;
-
- /* Enable the control mechanism to ensure that our measure,
- * Rxx160_LP, is in the correct range.
- */
- stt->activeSpeech = 0;
- stt->Rxx16_LPw32Max = 0;
-#ifdef MIC_LEVEL_FEEDBACK
- //stt->numBlocksMicLvlSat = 0;
-#endif
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt,
- "\tAGC->ProcessAnalog, frame %d: measure > 2ndUpperLim, micVol = %d, maxLevel = %d\n",
- stt->fcount, stt->micVol, stt->maxLevel);
-#endif
- }
- } else if (stt->Rxx160_LPw32 > stt->upperLimit)
- {
- stt->msTooHigh += 2;
- stt->msTooLow = 0;
- stt->changeToSlowMode = 0;
-
- if (stt->msTooHigh > stt->msecSpeechInnerChange)
- {
- /* Lower the recording level */
- stt->msTooHigh = 0;
- /* Multiply by 0.828125 which corresponds to decreasing ~0.8dB */
- tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6);
- stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 53);
-
- /* Reduce the max gain to avoid excessive oscillation
- * (but never drop below the maximum analog level).
- * stt->maxLevel = (15 * stt->maxLevel + stt->micVol) / 16;
- */
- tmp32 = (15 * stt->maxLevel) + stt->micVol;
- stt->maxLevel = WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
- stt->maxLevel = WEBRTC_SPL_MAX(stt->maxLevel, stt->maxAnalog);
-
- stt->zeroCtrlMax = stt->micVol;
-
- /* 0.965 in Q15 */
- tmp32 = inMicLevelTmp - stt->minLevel;
- tmpU32 = WEBRTC_SPL_UMUL(31621, (WebRtc_UWord32)(inMicLevelTmp - stt->minLevel));
- stt->micVol = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(tmpU32, 15) + stt->minLevel;
- if (stt->micVol > lastMicVol - 1)
- {
- stt->micVol = lastMicVol - 1;
- }
- inMicLevelTmp = stt->micVol;
-
-#ifdef MIC_LEVEL_FEEDBACK
- //stt->numBlocksMicLvlSat = 0;
-#endif
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt,
- "\tAGC->ProcessAnalog, frame %d: measure > UpperLim, micVol = %d, maxLevel = %d\n",
- stt->fcount, stt->micVol, stt->maxLevel);
-#endif
- }
- } else if (stt->Rxx160_LPw32 < stt->lowerSecondaryLimit)
- {
- stt->msTooHigh = 0;
- stt->changeToSlowMode = 0;
- stt->msTooLow += 2;
-
- if (stt->msTooLow > stt->msecSpeechOuterChange)
- {
- /* Raise the recording level */
- WebRtc_Word16 index, weightFIX;
- WebRtc_Word16 volNormFIX = 16384; // =1 in Q14.
-
- stt->msTooLow = 0;
-
- /* Normalize the volume level */
- tmp32 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14);
- if (stt->maxInit != stt->minLevel)
- {
- volNormFIX = (WebRtc_Word16)WEBRTC_SPL_DIV(tmp32,
- (stt->maxInit - stt->minLevel));
- }
-
- /* Find correct curve */
- WebRtcAgc_ExpCurve(volNormFIX, &index);
-
- /* Compute weighting factor for the volume increase, 32^(-2*X)/2+1.05 */
- weightFIX = kOffset1[index]
- - (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(kSlope1[index],
- volNormFIX, 13);
-
- /* stt->Rxx160_LPw32 *= 1.047 [~0.2 dB]; */
- tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6);
- stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 67);
-
- tmp32 = inMicLevelTmp - stt->minLevel;
- tmpU32 = ((WebRtc_UWord32)weightFIX * (WebRtc_UWord32)(inMicLevelTmp - stt->minLevel));
- stt->micVol = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(tmpU32, 14) + stt->minLevel;
- if (stt->micVol < lastMicVol + 2)
- {
- stt->micVol = lastMicVol + 2;
- }
-
- inMicLevelTmp = stt->micVol;
-
-#ifdef MIC_LEVEL_FEEDBACK
- /* Count ms in level saturation */
- //if (stt->micVol > stt->maxAnalog) {
- if (stt->micVol > 150)
- {
- /* mic level is saturated */
- stt->numBlocksMicLvlSat++;
- fprintf(stderr, "Sat mic Level: %d\n", stt->numBlocksMicLvlSat);
- }
-#endif
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt,
- "\tAGC->ProcessAnalog, frame %d: measure < 2ndLowerLim, micVol = %d\n",
- stt->fcount, stt->micVol);
-#endif
- }
- } else if (stt->Rxx160_LPw32 < stt->lowerLimit)
- {
- stt->msTooHigh = 0;
- stt->changeToSlowMode = 0;
- stt->msTooLow += 2;
-
- if (stt->msTooLow > stt->msecSpeechInnerChange)
- {
- /* Raise the recording level */
- WebRtc_Word16 index, weightFIX;
- WebRtc_Word16 volNormFIX = 16384; // =1 in Q14.
-
- stt->msTooLow = 0;
-
- /* Normalize the volume level */
- tmp32 = WEBRTC_SPL_LSHIFT_W32(inMicLevelTmp - stt->minLevel, 14);
- if (stt->maxInit != stt->minLevel)
- {
- volNormFIX = (WebRtc_Word16)WEBRTC_SPL_DIV(tmp32,
- (stt->maxInit - stt->minLevel));
- }
-
- /* Find correct curve */
- WebRtcAgc_ExpCurve(volNormFIX, &index);
-
- /* Compute weighting factor for the volume increase, (3.^(-2.*X))/8+1 */
- weightFIX = kOffset2[index]
- - (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(kSlope2[index],
- volNormFIX, 13);
-
- /* stt->Rxx160_LPw32 *= 1.047 [~0.2 dB]; */
- tmp32 = WEBRTC_SPL_RSHIFT_W32(stt->Rxx160_LPw32, 6);
- stt->Rxx160_LPw32 = WEBRTC_SPL_MUL(tmp32, 67);
-
- tmp32 = inMicLevelTmp - stt->minLevel;
- tmpU32 = ((WebRtc_UWord32)weightFIX * (WebRtc_UWord32)(inMicLevelTmp - stt->minLevel));
- stt->micVol = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(tmpU32, 14) + stt->minLevel;
- if (stt->micVol < lastMicVol + 1)
- {
- stt->micVol = lastMicVol + 1;
- }
-
- inMicLevelTmp = stt->micVol;
-
-#ifdef MIC_LEVEL_FEEDBACK
- /* Count ms in level saturation */
- //if (stt->micVol > stt->maxAnalog) {
- if (stt->micVol > 150)
- {
- /* mic level is saturated */
- stt->numBlocksMicLvlSat++;
- fprintf(stderr, "Sat mic Level: %d\n", stt->numBlocksMicLvlSat);
- }
-#endif
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt,
- "\tAGC->ProcessAnalog, frame %d: measure < LowerLim, micVol = %d\n",
- stt->fcount, stt->micVol);
-#endif
-
- }
- } else
- {
- /* The signal is inside the desired range which is:
- * lowerLimit < Rxx160_LP/640 < upperLimit
- */
- if (stt->changeToSlowMode > 4000)
- {
- stt->msecSpeechInnerChange = 1000;
- stt->msecSpeechOuterChange = 500;
- stt->upperLimit = stt->upperPrimaryLimit;
- stt->lowerLimit = stt->lowerPrimaryLimit;
- } else
- {
- stt->changeToSlowMode += 2; // in milliseconds
- }
- stt->msTooLow = 0;
- stt->msTooHigh = 0;
-
- stt->micVol = inMicLevelTmp;
-
- }
-#ifdef MIC_LEVEL_FEEDBACK
- if (stt->numBlocksMicLvlSat > NUM_BLOCKS_IN_SAT_BEFORE_CHANGE_TARGET)
- {
- stt->micLvlSat = 1;
- fprintf(stderr, "target before = %d (%d)\n", stt->analogTargetLevel, stt->targetIdx);
- WebRtcAgc_UpdateAgcThresholds(stt);
- WebRtcAgc_CalculateGainTable(&(stt->digitalAgc.gainTable[0]),
- stt->compressionGaindB, stt->targetLevelDbfs, stt->limiterEnable,
- stt->analogTarget);
- stt->numBlocksMicLvlSat = 0;
- stt->micLvlSat = 0;
- fprintf(stderr, "target offset = %d\n", stt->targetIdxOffset);
- fprintf(stderr, "target after = %d (%d)\n", stt->analogTargetLevel, stt->targetIdx);
- }
-#endif
- }
- }
-
- /* Ensure gain is not increased in presence of echo or after a mute event
- * (but allow the zeroCtrl() increase on the frame of a mute detection).
- */
- if (echo == 1 || (stt->muteGuardMs > 0 && stt->muteGuardMs < kMuteGuardTimeMs))
- {
- if (stt->micVol > lastMicVol)
- {
- stt->micVol = lastMicVol;
- }
- }
-
- /* limit the gain */
- if (stt->micVol > stt->maxLevel)
- {
- stt->micVol = stt->maxLevel;
- } else if (stt->micVol < stt->minOutput)
- {
- stt->micVol = stt->minOutput;
- }
-
- *outMicLevel = WEBRTC_SPL_RSHIFT_W32(stt->micVol, stt->scale);
- if (*outMicLevel > WEBRTC_SPL_RSHIFT_W32(stt->maxAnalog, stt->scale))
- {
- *outMicLevel = WEBRTC_SPL_RSHIFT_W32(stt->maxAnalog, stt->scale);
- }
-
- return 0;
-}
-
-int WebRtcAgc_Process(void *agcInst, const WebRtc_Word16 *in_near,
- const WebRtc_Word16 *in_near_H, WebRtc_Word16 samples,
- WebRtc_Word16 *out, WebRtc_Word16 *out_H, WebRtc_Word32 inMicLevel,
- WebRtc_Word32 *outMicLevel, WebRtc_Word16 echo,
- WebRtc_UWord8 *saturationWarning)
-{
- Agc_t *stt;
- WebRtc_Word32 inMicLevelTmp;
- WebRtc_Word16 subFrames, i;
- WebRtc_UWord8 satWarningTmp = 0;
-
- stt = (Agc_t *)agcInst;
-
- //
- if (stt == NULL)
- {
- return -1;
- }
- //
-
-
- if (stt->fs == 8000)
- {
- if ((samples != 80) && (samples != 160))
- {
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt,
- "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
-#endif
- return -1;
- }
- subFrames = 80;
- } else if (stt->fs == 16000)
- {
- if ((samples != 160) && (samples != 320))
- {
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt,
- "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
-#endif
- return -1;
- }
- subFrames = 160;
- } else if (stt->fs == 32000)
- {
- if ((samples != 160) && (samples != 320))
- {
-#ifdef AGC_DEBUG //test log
- fprintf(stt->fpt,
- "AGC->Process, frame %d: Invalid number of samples\n\n", stt->fcount);
-#endif
- return -1;
- }
- subFrames = 160;
- } else
- {
-#ifdef AGC_DEBUG// test log
- fprintf(stt->fpt,
- "AGC->Process, frame %d: Invalid sample rate\n\n", stt->fcount);
-#endif
- return -1;
- }
-
- /* Check for valid pointers based on sampling rate */
- if (stt->fs == 32000 && in_near_H == NULL)
- {
- return -1;
- }
- /* Check for valid pointers for low band */
- if (in_near == NULL)
- {
- return -1;
- }
-
- *saturationWarning = 0;
- //TODO: PUT IN RANGE CHECKING FOR INPUT LEVELS
- *outMicLevel = inMicLevel;
- inMicLevelTmp = inMicLevel;
-
- // TODO(andrew): clearly we don't need input and output pointers...
- // Change the interface to take a shared input/output.
- if (in_near != out)
- {
- // Only needed if they don't already point to the same place.
- memcpy(out, in_near, samples * sizeof(WebRtc_Word16));
- }
- if (stt->fs == 32000)
- {
- if (in_near_H != out_H)
- {
- memcpy(out_H, in_near_H, samples * sizeof(WebRtc_Word16));
- }
- }
-
-#ifdef AGC_DEBUG//test log
- stt->fcount++;
-#endif
-
- for (i = 0; i < samples; i += subFrames)
- {
- if (WebRtcAgc_ProcessDigital(&stt->digitalAgc, &in_near[i], &in_near_H[i], &out[i], &out_H[i],
- stt->fs, stt->lowLevelSignal) == -1)
- {
-#ifdef AGC_DEBUG//test log
- fprintf(stt->fpt, "AGC->Process, frame %d: Error from DigAGC\n\n", stt->fcount);
-#endif
- return -1;
- }
- if ((stt->agcMode < kAgcModeFixedDigital) && ((stt->lowLevelSignal == 0)
- || (stt->agcMode != kAgcModeAdaptiveDigital)))
- {
- if (WebRtcAgc_ProcessAnalog(agcInst, inMicLevelTmp, outMicLevel,
- stt->vadMic.logRatio, echo, saturationWarning) == -1)
- {
- return -1;
- }
- }
-#ifdef AGC_DEBUG//test log
- fprintf(stt->agcLog, "%5d\t%d\t%d\t%d\n", stt->fcount, inMicLevelTmp, *outMicLevel, stt->maxLevel, stt->micVol);
-#endif
-
- /* update queue */
- if (stt->inQueue > 1)
- {
- memcpy(stt->env[0], stt->env[1], 10 * sizeof(WebRtc_Word32));
- memcpy(stt->Rxx16w32_array[0], stt->Rxx16w32_array[1], 5 * sizeof(WebRtc_Word32));
- }
-
- if (stt->inQueue > 0)
- {
- stt->inQueue--;
- }
-
- /* If 20ms frames are used the input mic level must be updated so that
- * the analog AGC does not think that there has been a manual volume
- * change. */
- inMicLevelTmp = *outMicLevel;
-
- /* Store a positive saturation warning. */
- if (*saturationWarning == 1)
- {
- satWarningTmp = 1;
- }
- }
-
- /* Trigger the saturation warning if displayed by any of the frames. */
- *saturationWarning = satWarningTmp;
-
- return 0;
-}
-
-int WebRtcAgc_set_config(void *agcInst, WebRtcAgc_config_t agcConfig)
-{
- Agc_t *stt;
- stt = (Agc_t *)agcInst;
-
- if (stt == NULL)
- {
- return -1;
- }
-
- if (stt->initFlag != kInitCheck)
- {
- stt->lastError = AGC_UNINITIALIZED_ERROR;
- return -1;
- }
-
- if (agcConfig.limiterEnable != kAgcFalse && agcConfig.limiterEnable != kAgcTrue)
- {
- stt->lastError = AGC_BAD_PARAMETER_ERROR;
- return -1;
- }
- stt->limiterEnable = agcConfig.limiterEnable;
- stt->compressionGaindB = agcConfig.compressionGaindB;
- if ((agcConfig.targetLevelDbfs < 0) || (agcConfig.targetLevelDbfs > 31))
- {
- stt->lastError = AGC_BAD_PARAMETER_ERROR;
- return -1;
- }
- stt->targetLevelDbfs = agcConfig.targetLevelDbfs;
-
- if (stt->agcMode == kAgcModeFixedDigital)
- {
- /* Adjust for different parameter interpretation in FixedDigital mode */
- stt->compressionGaindB += agcConfig.targetLevelDbfs;
- }
-
- /* Update threshold levels for analog adaptation */
- WebRtcAgc_UpdateAgcThresholds(stt);
-
- /* Recalculate gain table */
- if (WebRtcAgc_CalculateGainTable(&(stt->digitalAgc.gainTable[0]), stt->compressionGaindB,
- stt->targetLevelDbfs, stt->limiterEnable, stt->analogTarget) == -1)
- {
-#ifdef AGC_DEBUG//test log
- fprintf(stt->fpt, "AGC->set_config, frame %d: Error from calcGainTable\n\n", stt->fcount);
-#endif
- return -1;
- }
- /* Store the config in a WebRtcAgc_config_t */
- stt->usedConfig.compressionGaindB = agcConfig.compressionGaindB;
- stt->usedConfig.limiterEnable = agcConfig.limiterEnable;
- stt->usedConfig.targetLevelDbfs = agcConfig.targetLevelDbfs;
-
- return 0;
-}
-
-int WebRtcAgc_get_config(void *agcInst, WebRtcAgc_config_t *config)
-{
- Agc_t *stt;
- stt = (Agc_t *)agcInst;
-
- if (stt == NULL)
- {
- return -1;
- }
-
- if (config == NULL)
- {
- stt->lastError = AGC_NULL_POINTER_ERROR;
- return -1;
- }
-
- if (stt->initFlag != kInitCheck)
- {
- stt->lastError = AGC_UNINITIALIZED_ERROR;
- return -1;
- }
-
- config->limiterEnable = stt->usedConfig.limiterEnable;
- config->targetLevelDbfs = stt->usedConfig.targetLevelDbfs;
- config->compressionGaindB = stt->usedConfig.compressionGaindB;
-
- return 0;
-}
-
-int WebRtcAgc_Create(void **agcInst)
-{
- Agc_t *stt;
- if (agcInst == NULL)
- {
- return -1;
- }
- stt = (Agc_t *)malloc(sizeof(Agc_t));
-
- *agcInst = stt;
- if (stt == NULL)
- {
- return -1;
- }
-
-#ifdef AGC_DEBUG
- stt->fpt = fopen("./agc_test_log.txt", "wt");
- stt->agcLog = fopen("./agc_debug_log.txt", "wt");
- stt->digitalAgc.logFile = fopen("./agc_log.txt", "wt");
-#endif
-
- stt->initFlag = 0;
- stt->lastError = 0;
-
- return 0;
-}
-
-int WebRtcAgc_Free(void *state)
-{
- Agc_t *stt;
-
- stt = (Agc_t *)state;
-#ifdef AGC_DEBUG
- fclose(stt->fpt);
- fclose(stt->agcLog);
- fclose(stt->digitalAgc.logFile);
-#endif
- free(stt);
-
- return 0;
-}
-
-/* minLevel - Minimum volume level
- * maxLevel - Maximum volume level
- */
-int WebRtcAgc_Init(void *agcInst, WebRtc_Word32 minLevel, WebRtc_Word32 maxLevel,
- WebRtc_Word16 agcMode, WebRtc_UWord32 fs)
-{
- WebRtc_Word32 max_add, tmp32;
- WebRtc_Word16 i;
- int tmpNorm;
- Agc_t *stt;
-
- /* typecast state pointer */
- stt = (Agc_t *)agcInst;
-
- if (WebRtcAgc_InitDigital(&stt->digitalAgc, agcMode) != 0)
- {
- stt->lastError = AGC_UNINITIALIZED_ERROR;
- return -1;
- }
-
- /* Analog AGC variables */
- stt->envSum = 0;
-
- /* mode = 0 - Only saturation protection
- * 1 - Analog Automatic Gain Control [-targetLevelDbfs (default -3 dBOv)]
- * 2 - Digital Automatic Gain Control [-targetLevelDbfs (default -3 dBOv)]
- * 3 - Fixed Digital Gain [compressionGaindB (default 8 dB)]
- */
-#ifdef AGC_DEBUG//test log
- stt->fcount = 0;
- fprintf(stt->fpt, "AGC->Init\n");
-#endif
- if (agcMode < kAgcModeUnchanged || agcMode > kAgcModeFixedDigital)
- {
-#ifdef AGC_DEBUG//test log
- fprintf(stt->fpt, "AGC->Init: error, incorrect mode\n\n");
-#endif
- return -1;
- }
- stt->agcMode = agcMode;
- stt->fs = fs;
-
- /* initialize input VAD */
- WebRtcAgc_InitVad(&stt->vadMic);
-
- /* If the volume range is smaller than 0-256 then
- * the levels are shifted up to Q8-domain */
- tmpNorm = WebRtcSpl_NormU32((WebRtc_UWord32)maxLevel);
- stt->scale = tmpNorm - 23;
- if (stt->scale < 0)
- {
- stt->scale = 0;
- }
- // TODO(bjornv): Investigate if we really need to scale up a small range now when we have
- // a guard against zero-increments. For now, we do not support scale up (scale = 0).
- stt->scale = 0;
- maxLevel = WEBRTC_SPL_LSHIFT_W32(maxLevel, stt->scale);
- minLevel = WEBRTC_SPL_LSHIFT_W32(minLevel, stt->scale);
-
- /* Make minLevel and maxLevel static in AdaptiveDigital */
- if (stt->agcMode == kAgcModeAdaptiveDigital)
- {
- minLevel = 0;
- maxLevel = 255;
- stt->scale = 0;
- }
- /* The maximum supplemental volume range is based on a vague idea
- * of how much lower the gain will be than the real analog gain. */
- max_add = WEBRTC_SPL_RSHIFT_W32(maxLevel - minLevel, 2);
-
- /* Minimum/maximum volume level that can be set */
- stt->minLevel = minLevel;
- stt->maxAnalog = maxLevel;
- stt->maxLevel = maxLevel + max_add;
- stt->maxInit = stt->maxLevel;
-
- stt->zeroCtrlMax = stt->maxAnalog;
-
- /* Initialize micVol parameter */
- stt->micVol = stt->maxAnalog;
- if (stt->agcMode == kAgcModeAdaptiveDigital)
- {
- stt->micVol = 127; /* Mid-point of mic level */
- }
- stt->micRef = stt->micVol;
- stt->micGainIdx = 127;
-#ifdef MIC_LEVEL_FEEDBACK
- stt->numBlocksMicLvlSat = 0;
- stt->micLvlSat = 0;
-#endif
-#ifdef AGC_DEBUG//test log
- fprintf(stt->fpt,
- "AGC->Init: minLevel = %d, maxAnalog = %d, maxLevel = %d\n",
- stt->minLevel, stt->maxAnalog, stt->maxLevel);
-#endif
-
- /* Minimum output volume is 4% higher than the available lowest volume level */
- tmp32 = WEBRTC_SPL_RSHIFT_W32((stt->maxLevel - stt->minLevel) * (WebRtc_Word32)10, 8);
- stt->minOutput = (stt->minLevel + tmp32);
-
- stt->msTooLow = 0;
- stt->msTooHigh = 0;
- stt->changeToSlowMode = 0;
- stt->firstCall = 0;
- stt->msZero = 0;
- stt->muteGuardMs = 0;
- stt->gainTableIdx = 0;
-
- stt->msecSpeechInnerChange = kMsecSpeechInner;
- stt->msecSpeechOuterChange = kMsecSpeechOuter;
-
- stt->activeSpeech = 0;
- stt->Rxx16_LPw32Max = 0;
-
- stt->vadThreshold = kNormalVadThreshold;
- stt->inActive = 0;
-
- for (i = 0; i < RXX_BUFFER_LEN; i++)
- {
- stt->Rxx16_vectorw32[i] = (WebRtc_Word32)1000; /* -54dBm0 */
- }
- stt->Rxx160w32 = 125 * RXX_BUFFER_LEN; /* (stt->Rxx16_vectorw32[0]>>3) = 125 */
-
- stt->Rxx16pos = 0;
- stt->Rxx16_LPw32 = (WebRtc_Word32)16284; /* Q(-4) */
-
- for (i = 0; i < 5; i++)
- {
- stt->Rxx16w32_array[0][i] = 0;
- }
- for (i = 0; i < 20; i++)
- {
- stt->env[0][i] = 0;
- }
- stt->inQueue = 0;
-
-#ifdef MIC_LEVEL_FEEDBACK
- stt->targetIdxOffset = 0;
-#endif
-
- WebRtcSpl_MemSetW32(stt->filterState, 0, 8);
-
- stt->initFlag = kInitCheck;
- // Default config settings.
- stt->defaultConfig.limiterEnable = kAgcTrue;
- stt->defaultConfig.targetLevelDbfs = AGC_DEFAULT_TARGET_LEVEL;
- stt->defaultConfig.compressionGaindB = AGC_DEFAULT_COMP_GAIN;
-
- if (WebRtcAgc_set_config(stt, stt->defaultConfig) == -1)
- {
- stt->lastError = AGC_UNSPECIFIED_ERROR;
- return -1;
- }
- stt->Rxx160_LPw32 = stt->analogTargetLevel; // Initialize rms value
-
- stt->lowLevelSignal = 0;
-
- /* Only positive values are allowed that are not too large */
- if ((minLevel >= maxLevel) || (maxLevel & 0xFC000000))
- {
-#ifdef AGC_DEBUG//test log
- fprintf(stt->fpt, "minLevel, maxLevel value(s) are invalid\n\n");
-#endif
- return -1;
- } else
- {
-#ifdef AGC_DEBUG//test log
- fprintf(stt->fpt, "\n");
-#endif
- return 0;
- }
-}
diff --git a/trunk/src/modules/audio_processing/agc/analog_agc.h b/trunk/src/modules/audio_processing/agc/analog_agc.h
deleted file mode 100644
index b32ac65..0000000
--- a/trunk/src/modules/audio_processing/agc/analog_agc.h
+++ /dev/null
@@ -1,133 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
-
-#include "typedefs.h"
-#include "gain_control.h"
-#include "digital_agc.h"
-
-//#define AGC_DEBUG
-//#define MIC_LEVEL_FEEDBACK
-#ifdef AGC_DEBUG
-#include <stdio.h>
-#endif
-
-/* Analog Automatic Gain Control variables:
- * Constant declarations (inner limits inside which no changes are done)
- * In the beginning the range is narrower to widen as soon as the measure
- * 'Rxx160_LP' is inside it. Currently the starting limits are -22.2+/-1dBm0
- * and the final limits -22.2+/-2.5dBm0. These levels makes the speech signal
- * go towards -25.4dBm0 (-31.4dBov). Tuned with wbfile-31.4dBov.pcm
- * The limits are created by running the AGC with a file having the desired
- * signal level and thereafter plotting Rxx160_LP in the dBm0-domain defined
- * by out=10*log10(in/260537279.7); Set the target level to the average level
- * of our measure Rxx160_LP. Remember that the levels are in blocks of 16 in
- * Q(-7). (Example matlab code: round(db2pow(-21.2)*16/2^7) )
- */
-#define RXX_BUFFER_LEN 10
-
-static const WebRtc_Word16 kMsecSpeechInner = 520;
-static const WebRtc_Word16 kMsecSpeechOuter = 340;
-
-static const WebRtc_Word16 kNormalVadThreshold = 400;
-
-static const WebRtc_Word16 kAlphaShortTerm = 6; // 1 >> 6 = 0.0156
-static const WebRtc_Word16 kAlphaLongTerm = 10; // 1 >> 10 = 0.000977
-
-typedef struct
-{
- // Configurable parameters/variables
- WebRtc_UWord32 fs; // Sampling frequency
- WebRtc_Word16 compressionGaindB; // Fixed gain level in dB
- WebRtc_Word16 targetLevelDbfs; // Target level in -dBfs of envelope (default -3)
- WebRtc_Word16 agcMode; // Hard coded mode (adaptAna/adaptDig/fixedDig)
- WebRtc_UWord8 limiterEnable; // Enabling limiter (on/off (default off))
- WebRtcAgc_config_t defaultConfig;
- WebRtcAgc_config_t usedConfig;
-
- // General variables
- WebRtc_Word16 initFlag;
- WebRtc_Word16 lastError;
-
- // Target level parameters
- // Based on the above: analogTargetLevel = round((32767*10^(-22/20))^2*16/2^7)
- WebRtc_Word32 analogTargetLevel; // = RXX_BUFFER_LEN * 846805; -22 dBfs
- WebRtc_Word32 startUpperLimit; // = RXX_BUFFER_LEN * 1066064; -21 dBfs
- WebRtc_Word32 startLowerLimit; // = RXX_BUFFER_LEN * 672641; -23 dBfs
- WebRtc_Word32 upperPrimaryLimit; // = RXX_BUFFER_LEN * 1342095; -20 dBfs
- WebRtc_Word32 lowerPrimaryLimit; // = RXX_BUFFER_LEN * 534298; -24 dBfs
- WebRtc_Word32 upperSecondaryLimit;// = RXX_BUFFER_LEN * 2677832; -17 dBfs
- WebRtc_Word32 lowerSecondaryLimit;// = RXX_BUFFER_LEN * 267783; -27 dBfs
- WebRtc_UWord16 targetIdx; // Table index for corresponding target level
-#ifdef MIC_LEVEL_FEEDBACK
- WebRtc_UWord16 targetIdxOffset; // Table index offset for level compensation
-#endif
- WebRtc_Word16 analogTarget; // Digital reference level in ENV scale
-
- // Analog AGC specific variables
- WebRtc_Word32 filterState[8]; // For downsampling wb to nb
- WebRtc_Word32 upperLimit; // Upper limit for mic energy
- WebRtc_Word32 lowerLimit; // Lower limit for mic energy
- WebRtc_Word32 Rxx160w32; // Average energy for one frame
- WebRtc_Word32 Rxx16_LPw32; // Low pass filtered subframe energies
- WebRtc_Word32 Rxx160_LPw32; // Low pass filtered frame energies
- WebRtc_Word32 Rxx16_LPw32Max; // Keeps track of largest energy subframe
- WebRtc_Word32 Rxx16_vectorw32[RXX_BUFFER_LEN];// Array with subframe energies
- WebRtc_Word32 Rxx16w32_array[2][5];// Energy values of microphone signal
- WebRtc_Word32 env[2][10]; // Envelope values of subframes
-
- WebRtc_Word16 Rxx16pos; // Current position in the Rxx16_vectorw32
- WebRtc_Word16 envSum; // Filtered scaled envelope in subframes
- WebRtc_Word16 vadThreshold; // Threshold for VAD decision
- WebRtc_Word16 inActive; // Inactive time in milliseconds
- WebRtc_Word16 msTooLow; // Milliseconds of speech at a too low level
- WebRtc_Word16 msTooHigh; // Milliseconds of speech at a too high level
- WebRtc_Word16 changeToSlowMode; // Change to slow mode after some time at target
- WebRtc_Word16 firstCall; // First call to the process-function
- WebRtc_Word16 msZero; // Milliseconds of zero input
- WebRtc_Word16 msecSpeechOuterChange;// Min ms of speech between volume changes
- WebRtc_Word16 msecSpeechInnerChange;// Min ms of speech between volume changes
- WebRtc_Word16 activeSpeech; // Milliseconds of active speech
- WebRtc_Word16 muteGuardMs; // Counter to prevent mute action
- WebRtc_Word16 inQueue; // 10 ms batch indicator
-
- // Microphone level variables
- WebRtc_Word32 micRef; // Remember ref. mic level for virtual mic
- WebRtc_UWord16 gainTableIdx; // Current position in virtual gain table
- WebRtc_Word32 micGainIdx; // Gain index of mic level to increase slowly
- WebRtc_Word32 micVol; // Remember volume between frames
- WebRtc_Word32 maxLevel; // Max possible vol level, incl dig gain
- WebRtc_Word32 maxAnalog; // Maximum possible analog volume level
- WebRtc_Word32 maxInit; // Initial value of "max"
- WebRtc_Word32 minLevel; // Minimum possible volume level
- WebRtc_Word32 minOutput; // Minimum output volume level
- WebRtc_Word32 zeroCtrlMax; // Remember max gain => don't amp low input
-
- WebRtc_Word16 scale; // Scale factor for internal volume levels
-#ifdef MIC_LEVEL_FEEDBACK
- WebRtc_Word16 numBlocksMicLvlSat;
- WebRtc_UWord8 micLvlSat;
-#endif
- // Structs for VAD and digital_agc
- AgcVad_t vadMic;
- DigitalAgc_t digitalAgc;
-
-#ifdef AGC_DEBUG
- FILE* fpt;
- FILE* agcLog;
- WebRtc_Word32 fcount;
-#endif
-
- WebRtc_Word16 lowLevelSignal;
-} Agc_t;
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
diff --git a/trunk/src/modules/audio_processing/agc/digital_agc.c b/trunk/src/modules/audio_processing/agc/digital_agc.c
deleted file mode 100644
index 3b4b39b..0000000
--- a/trunk/src/modules/audio_processing/agc/digital_agc.c
+++ /dev/null
@@ -1,798 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/* digital_agc.c
- *
- */
-
-#include "digital_agc.h"
-
-#include <assert.h>
-#include <string.h>
-#ifdef AGC_DEBUG
-#include <stdio.h>
-#endif
-
-#include "gain_control.h"
-
-// To generate the gaintable, copy&paste the following lines to a Matlab window:
-// MaxGain = 6; MinGain = 0; CompRatio = 3; Knee = 1;
-// zeros = 0:31; lvl = 2.^(1-zeros);
-// A = -10*log10(lvl) * (CompRatio - 1) / CompRatio;
-// B = MaxGain - MinGain;
-// gains = round(2^16*10.^(0.05 * (MinGain + B * ( log(exp(-Knee*A)+exp(-Knee*B)) - log(1+exp(-Knee*B)) ) / log(1/(1+exp(Knee*B))))));
-// fprintf(1, '\t%i, %i, %i, %i,\n', gains);
-// % Matlab code for plotting the gain and input/output level characteristic (copy/paste the following 3 lines):
-// in = 10*log10(lvl); out = 20*log10(gains/65536);
-// subplot(121); plot(in, out); axis([-30, 0, -5, 20]); grid on; xlabel('Input (dB)'); ylabel('Gain (dB)');
-// subplot(122); plot(in, in+out); axis([-30, 0, -30, 5]); grid on; xlabel('Input (dB)'); ylabel('Output (dB)');
-// zoom on;
-
-// Generator table for y=log2(1+e^x) in Q8.
-enum { kGenFuncTableSize = 128 };
-static const WebRtc_UWord16 kGenFuncTable[kGenFuncTableSize] = {
- 256, 485, 786, 1126, 1484, 1849, 2217, 2586,
- 2955, 3324, 3693, 4063, 4432, 4801, 5171, 5540,
- 5909, 6279, 6648, 7017, 7387, 7756, 8125, 8495,
- 8864, 9233, 9603, 9972, 10341, 10711, 11080, 11449,
- 11819, 12188, 12557, 12927, 13296, 13665, 14035, 14404,
- 14773, 15143, 15512, 15881, 16251, 16620, 16989, 17359,
- 17728, 18097, 18466, 18836, 19205, 19574, 19944, 20313,
- 20682, 21052, 21421, 21790, 22160, 22529, 22898, 23268,
- 23637, 24006, 24376, 24745, 25114, 25484, 25853, 26222,
- 26592, 26961, 27330, 27700, 28069, 28438, 28808, 29177,
- 29546, 29916, 30285, 30654, 31024, 31393, 31762, 32132,
- 32501, 32870, 33240, 33609, 33978, 34348, 34717, 35086,
- 35456, 35825, 36194, 36564, 36933, 37302, 37672, 38041,
- 38410, 38780, 39149, 39518, 39888, 40257, 40626, 40996,
- 41365, 41734, 42104, 42473, 42842, 43212, 43581, 43950,
- 44320, 44689, 45058, 45428, 45797, 46166, 46536, 46905
-};
-
-static const WebRtc_Word16 kAvgDecayTime = 250; // frames; < 3000
-
-WebRtc_Word32 WebRtcAgc_CalculateGainTable(WebRtc_Word32 *gainTable, // Q16
- WebRtc_Word16 digCompGaindB, // Q0
- WebRtc_Word16 targetLevelDbfs,// Q0
- WebRtc_UWord8 limiterEnable,
- WebRtc_Word16 analogTarget) // Q0
-{
- // This function generates the compressor gain table used in the fixed digital part.
- WebRtc_UWord32 tmpU32no1, tmpU32no2, absInLevel, logApprox;
- WebRtc_Word32 inLevel, limiterLvl;
- WebRtc_Word32 tmp32, tmp32no1, tmp32no2, numFIX, den, y32;
- const WebRtc_UWord16 kLog10 = 54426; // log2(10) in Q14
- const WebRtc_UWord16 kLog10_2 = 49321; // 10*log10(2) in Q14
- const WebRtc_UWord16 kLogE_1 = 23637; // log2(e) in Q14
- WebRtc_UWord16 constMaxGain;
- WebRtc_UWord16 tmpU16, intPart, fracPart;
- const WebRtc_Word16 kCompRatio = 3;
- const WebRtc_Word16 kSoftLimiterLeft = 1;
- WebRtc_Word16 limiterOffset = 0; // Limiter offset
- WebRtc_Word16 limiterIdx, limiterLvlX;
- WebRtc_Word16 constLinApprox, zeroGainLvl, maxGain, diffGain;
- WebRtc_Word16 i, tmp16, tmp16no1;
- int zeros, zerosScale;
-
- // Constants
-// kLogE_1 = 23637; // log2(e) in Q14
-// kLog10 = 54426; // log2(10) in Q14
-// kLog10_2 = 49321; // 10*log10(2) in Q14
-
- // Calculate maximum digital gain and zero gain level
- tmp32no1 = WEBRTC_SPL_MUL_16_16(digCompGaindB - analogTarget, kCompRatio - 1);
- tmp16no1 = analogTarget - targetLevelDbfs;
- tmp16no1 += WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio);
- maxGain = WEBRTC_SPL_MAX(tmp16no1, (analogTarget - targetLevelDbfs));
- tmp32no1 = WEBRTC_SPL_MUL_16_16(maxGain, kCompRatio);
- zeroGainLvl = digCompGaindB;
- zeroGainLvl -= WebRtcSpl_DivW32W16ResW16(tmp32no1 + ((kCompRatio - 1) >> 1),
- kCompRatio - 1);
- if ((digCompGaindB <= analogTarget) && (limiterEnable))
- {
- zeroGainLvl += (analogTarget - digCompGaindB + kSoftLimiterLeft);
- limiterOffset = 0;
- }
-
- // Calculate the difference between maximum gain and gain at 0dB0v:
- // diffGain = maxGain + (compRatio-1)*zeroGainLvl/compRatio
- // = (compRatio-1)*digCompGaindB/compRatio
- tmp32no1 = WEBRTC_SPL_MUL_16_16(digCompGaindB, kCompRatio - 1);
- diffGain = WebRtcSpl_DivW32W16ResW16(tmp32no1 + (kCompRatio >> 1), kCompRatio);
- if (diffGain < 0 || diffGain >= kGenFuncTableSize)
- {
- assert(0);
- return -1;
- }
-
- // Calculate the limiter level and index:
- // limiterLvlX = analogTarget - limiterOffset
- // limiterLvl = targetLevelDbfs + limiterOffset/compRatio
- limiterLvlX = analogTarget - limiterOffset;
- limiterIdx = 2
- + WebRtcSpl_DivW32W16ResW16(WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)limiterLvlX, 13),
- WEBRTC_SPL_RSHIFT_U16(kLog10_2, 1));
- tmp16no1 = WebRtcSpl_DivW32W16ResW16(limiterOffset + (kCompRatio >> 1), kCompRatio);
- limiterLvl = targetLevelDbfs + tmp16no1;
-
- // Calculate (through table lookup):
- // constMaxGain = log2(1+2^(log2(e)*diffGain)); (in Q8)
- constMaxGain = kGenFuncTable[diffGain]; // in Q8
-
- // Calculate a parameter used to approximate the fractional part of 2^x with a
- // piecewise linear function in Q14:
- // constLinApprox = round(3/2*(4*(3-2*sqrt(2))/(log(2)^2)-0.5)*2^14);
- constLinApprox = 22817; // in Q14
-
- // Calculate a denominator used in the exponential part to convert from dB to linear scale:
- // den = 20*constMaxGain (in Q8)
- den = WEBRTC_SPL_MUL_16_U16(20, constMaxGain); // in Q8
-
- for (i = 0; i < 32; i++)
- {
- // Calculate scaled input level (compressor):
- // inLevel = fix((-constLog10_2*(compRatio-1)*(1-i)+fix(compRatio/2))/compRatio)
- tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(kCompRatio - 1, i - 1); // Q0
- tmp32 = WEBRTC_SPL_MUL_16_U16(tmp16, kLog10_2) + 1; // Q14
- inLevel = WebRtcSpl_DivW32W16(tmp32, kCompRatio); // Q14
-
- // Calculate diffGain-inLevel, to map using the genFuncTable
- inLevel = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)diffGain, 14) - inLevel; // Q14
-
- // Make calculations on abs(inLevel) and compensate for the sign afterwards.
- absInLevel = (WebRtc_UWord32)WEBRTC_SPL_ABS_W32(inLevel); // Q14
-
- // LUT with interpolation
- intPart = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_U32(absInLevel, 14);
- fracPart = (WebRtc_UWord16)(absInLevel & 0x00003FFF); // extract the fractional part
- tmpU16 = kGenFuncTable[intPart + 1] - kGenFuncTable[intPart]; // Q8
- tmpU32no1 = WEBRTC_SPL_UMUL_16_16(tmpU16, fracPart); // Q22
- tmpU32no1 += WEBRTC_SPL_LSHIFT_U32((WebRtc_UWord32)kGenFuncTable[intPart], 14); // Q22
- logApprox = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 8); // Q14
- // Compensate for negative exponent using the relation:
- // log2(1 + 2^-x) = log2(1 + 2^x) - x
- if (inLevel < 0)
- {
- zeros = WebRtcSpl_NormU32(absInLevel);
- zerosScale = 0;
- if (zeros < 15)
- {
- // Not enough space for multiplication
- tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(absInLevel, 15 - zeros); // Q(zeros-1)
- tmpU32no2 = WEBRTC_SPL_UMUL_32_16(tmpU32no2, kLogE_1); // Q(zeros+13)
- if (zeros < 9)
- {
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 9 - zeros); // Q(zeros+13)
- zerosScale = 9 - zeros;
- } else
- {
- tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, zeros - 9); // Q22
- }
- } else
- {
- tmpU32no2 = WEBRTC_SPL_UMUL_32_16(absInLevel, kLogE_1); // Q28
- tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 6); // Q22
- }
- logApprox = 0;
- if (tmpU32no2 < tmpU32no1)
- {
- logApprox = WEBRTC_SPL_RSHIFT_U32(tmpU32no1 - tmpU32no2, 8 - zerosScale); //Q14
- }
- }
- numFIX = WEBRTC_SPL_LSHIFT_W32(WEBRTC_SPL_MUL_16_U16(maxGain, constMaxGain), 6); // Q14
- numFIX -= WEBRTC_SPL_MUL_32_16((WebRtc_Word32)logApprox, diffGain); // Q14
-
- // Calculate ratio
- // Shift |numFIX| as much as possible.
- // Ensure we avoid wrap-around in |den| as well.
- if (numFIX > (den >> 8)) // |den| is Q8.
- {
- zeros = WebRtcSpl_NormW32(numFIX);
- } else
- {
- zeros = WebRtcSpl_NormW32(den) + 8;
- }
- numFIX = WEBRTC_SPL_LSHIFT_W32(numFIX, zeros); // Q(14+zeros)
-
- // Shift den so we end up in Qy1
- tmp32no1 = WEBRTC_SPL_SHIFT_W32(den, zeros - 8); // Q(zeros)
- if (numFIX < 0)
- {
- numFIX -= WEBRTC_SPL_RSHIFT_W32(tmp32no1, 1);
- } else
- {
- numFIX += WEBRTC_SPL_RSHIFT_W32(tmp32no1, 1);
- }
- y32 = WEBRTC_SPL_DIV(numFIX, tmp32no1); // in Q14
- if (limiterEnable && (i < limiterIdx))
- {
- tmp32 = WEBRTC_SPL_MUL_16_U16(i - 1, kLog10_2); // Q14
- tmp32 -= WEBRTC_SPL_LSHIFT_W32(limiterLvl, 14); // Q14
- y32 = WebRtcSpl_DivW32W16(tmp32 + 10, 20);
- }
- if (y32 > 39000)
- {
- tmp32 = WEBRTC_SPL_MUL(y32 >> 1, kLog10) + 4096; // in Q27
- tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 13); // in Q14
- } else
- {
- tmp32 = WEBRTC_SPL_MUL(y32, kLog10) + 8192; // in Q28
- tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 14); // in Q14
- }
- tmp32 += WEBRTC_SPL_LSHIFT_W32(16, 14); // in Q14 (Make sure final output is in Q16)
-
- // Calculate power
- if (tmp32 > 0)
- {
- intPart = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 14);
- fracPart = (WebRtc_UWord16)(tmp32 & 0x00003FFF); // in Q14
- if (WEBRTC_SPL_RSHIFT_W32(fracPart, 13))
- {
- tmp16 = WEBRTC_SPL_LSHIFT_W16(2, 14) - constLinApprox;
- tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - fracPart;
- tmp32no2 = WEBRTC_SPL_MUL_32_16(tmp32no2, tmp16);
- tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 13);
- tmp32no2 = WEBRTC_SPL_LSHIFT_W32(1, 14) - tmp32no2;
- } else
- {
- tmp16 = constLinApprox - WEBRTC_SPL_LSHIFT_W16(1, 14);
- tmp32no2 = WEBRTC_SPL_MUL_32_16(fracPart, tmp16);
- tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 13);
- }
- fracPart = (WebRtc_UWord16)tmp32no2;
- gainTable[i] = WEBRTC_SPL_LSHIFT_W32(1, intPart)
- + WEBRTC_SPL_SHIFT_W32(fracPart, intPart - 14);
- } else
- {
- gainTable[i] = 0;
- }
- }
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAgc_InitDigital(DigitalAgc_t *stt, WebRtc_Word16 agcMode)
-{
-
- if (agcMode == kAgcModeFixedDigital)
- {
- // start at minimum to find correct gain faster
- stt->capacitorSlow = 0;
- } else
- {
- // start out with 0 dB gain
- stt->capacitorSlow = 134217728; // (WebRtc_Word32)(0.125f * 32768.0f * 32768.0f);
- }
- stt->capacitorFast = 0;
- stt->gain = 65536;
- stt->gatePrevious = 0;
- stt->agcMode = agcMode;
-#ifdef AGC_DEBUG
- stt->frameCounter = 0;
-#endif
-
- // initialize VADs
- WebRtcAgc_InitVad(&stt->vadNearend);
- WebRtcAgc_InitVad(&stt->vadFarend);
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAgc_AddFarendToDigital(DigitalAgc_t *stt, const WebRtc_Word16 *in_far,
- WebRtc_Word16 nrSamples)
-{
- // Check for valid pointer
- if (&stt->vadFarend == NULL)
- {
- return -1;
- }
-
- // VAD for far end
- WebRtcAgc_ProcessVad(&stt->vadFarend, in_far, nrSamples);
-
- return 0;
-}
-
-WebRtc_Word32 WebRtcAgc_ProcessDigital(DigitalAgc_t *stt, const WebRtc_Word16 *in_near,
- const WebRtc_Word16 *in_near_H, WebRtc_Word16 *out,
- WebRtc_Word16 *out_H, WebRtc_UWord32 FS,
- WebRtc_Word16 lowlevelSignal)
-{
- // array for gains (one value per ms, incl start & end)
- WebRtc_Word32 gains[11];
-
- WebRtc_Word32 out_tmp, tmp32;
- WebRtc_Word32 env[10];
- WebRtc_Word32 nrg, max_nrg;
- WebRtc_Word32 cur_level;
- WebRtc_Word32 gain32, delta;
- WebRtc_Word16 logratio;
- WebRtc_Word16 lower_thr, upper_thr;
- WebRtc_Word16 zeros, zeros_fast, frac;
- WebRtc_Word16 decay;
- WebRtc_Word16 gate, gain_adj;
- WebRtc_Word16 k, n;
- WebRtc_Word16 L, L2; // samples/subframe
-
- // determine number of samples per ms
- if (FS == 8000)
- {
- L = 8;
- L2 = 3;
- } else if (FS == 16000)
- {
- L = 16;
- L2 = 4;
- } else if (FS == 32000)
- {
- L = 16;
- L2 = 4;
- } else
- {
- return -1;
- }
-
- // TODO(andrew): again, we don't need input and output pointers...
- if (in_near != out)
- {
- // Only needed if they don't already point to the same place.
- memcpy(out, in_near, 10 * L * sizeof(WebRtc_Word16));
- }
- if (FS == 32000)
- {
- if (in_near_H != out_H)
- {
- memcpy(out_H, in_near_H, 10 * L * sizeof(WebRtc_Word16));
- }
- }
- // VAD for near end
- logratio = WebRtcAgc_ProcessVad(&stt->vadNearend, out, L * 10);
-
- // Account for far end VAD
- if (stt->vadFarend.counter > 10)
- {
- tmp32 = WEBRTC_SPL_MUL_16_16(3, logratio);
- logratio = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 - stt->vadFarend.logRatio, 2);
- }
-
- // Determine decay factor depending on VAD
- // upper_thr = 1.0f;
- // lower_thr = 0.25f;
- upper_thr = 1024; // Q10
- lower_thr = 0; // Q10
- if (logratio > upper_thr)
- {
- // decay = -2^17 / DecayTime; -> -65
- decay = -65;
- } else if (logratio < lower_thr)
- {
- decay = 0;
- } else
- {
- // decay = (WebRtc_Word16)(((lower_thr - logratio)
- // * (2^27/(DecayTime*(upper_thr-lower_thr)))) >> 10);
- // SUBSTITUTED: 2^27/(DecayTime*(upper_thr-lower_thr)) -> 65
- tmp32 = WEBRTC_SPL_MUL_16_16((lower_thr - logratio), 65);
- decay = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 10);
- }
-
- // adjust decay factor for long silence (detected as low standard deviation)
- // This is only done in the adaptive modes
- if (stt->agcMode != kAgcModeFixedDigital)
- {
- if (stt->vadNearend.stdLongTerm < 4000)
- {
- decay = 0;
- } else if (stt->vadNearend.stdLongTerm < 8096)
- {
- // decay = (WebRtc_Word16)(((stt->vadNearend.stdLongTerm - 4000) * decay) >> 12);
- tmp32 = WEBRTC_SPL_MUL_16_16((stt->vadNearend.stdLongTerm - 4000), decay);
- decay = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
- }
-
- if (lowlevelSignal != 0)
- {
- decay = 0;
- }
- }
-#ifdef AGC_DEBUG
- stt->frameCounter++;
- fprintf(stt->logFile, "%5.2f\t%d\t%d\t%d\t", (float)(stt->frameCounter) / 100, logratio, decay, stt->vadNearend.stdLongTerm);
-#endif
- // Find max amplitude per sub frame
- // iterate over sub frames
- for (k = 0; k < 10; k++)
- {
- // iterate over samples
- max_nrg = 0;
- for (n = 0; n < L; n++)
- {
- nrg = WEBRTC_SPL_MUL_16_16(out[k * L + n], out[k * L + n]);
- if (nrg > max_nrg)
- {
- max_nrg = nrg;
- }
- }
- env[k] = max_nrg;
- }
-
- // Calculate gain per sub frame
- gains[0] = stt->gain;
- for (k = 0; k < 10; k++)
- {
- // Fast envelope follower
- // decay time = -131000 / -1000 = 131 (ms)
- stt->capacitorFast = AGC_SCALEDIFF32(-1000, stt->capacitorFast, stt->capacitorFast);
- if (env[k] > stt->capacitorFast)
- {
- stt->capacitorFast = env[k];
- }
- // Slow envelope follower
- if (env[k] > stt->capacitorSlow)
- {
- // increase capacitorSlow
- stt->capacitorSlow
- = AGC_SCALEDIFF32(500, (env[k] - stt->capacitorSlow), stt->capacitorSlow);
- } else
- {
- // decrease capacitorSlow
- stt->capacitorSlow
- = AGC_SCALEDIFF32(decay, stt->capacitorSlow, stt->capacitorSlow);
- }
-
- // use maximum of both capacitors as current level
- if (stt->capacitorFast > stt->capacitorSlow)
- {
- cur_level = stt->capacitorFast;
- } else
- {
- cur_level = stt->capacitorSlow;
- }
- // Translate signal level into gain, using a piecewise linear approximation
- // find number of leading zeros
- zeros = WebRtcSpl_NormU32((WebRtc_UWord32)cur_level);
- if (cur_level == 0)
- {
- zeros = 31;
- }
- tmp32 = (WEBRTC_SPL_LSHIFT_W32(cur_level, zeros) & 0x7FFFFFFF);
- frac = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 19); // Q12
- tmp32 = WEBRTC_SPL_MUL((stt->gainTable[zeros-1] - stt->gainTable[zeros]), frac);
- gains[k + 1] = stt->gainTable[zeros] + WEBRTC_SPL_RSHIFT_W32(tmp32, 12);
-#ifdef AGC_DEBUG
- if (k == 0)
- {
- fprintf(stt->logFile, "%d\t%d\t%d\t%d\t%d\n", env[0], cur_level, stt->capacitorFast, stt->capacitorSlow, zeros);
- }
-#endif
- }
-
- // Gate processing (lower gain during absence of speech)
- zeros = WEBRTC_SPL_LSHIFT_W16(zeros, 9) - WEBRTC_SPL_RSHIFT_W16(frac, 3);
- // find number of leading zeros
- zeros_fast = WebRtcSpl_NormU32((WebRtc_UWord32)stt->capacitorFast);
- if (stt->capacitorFast == 0)
- {
- zeros_fast = 31;
- }
- tmp32 = (WEBRTC_SPL_LSHIFT_W32(stt->capacitorFast, zeros_fast) & 0x7FFFFFFF);
- zeros_fast = WEBRTC_SPL_LSHIFT_W16(zeros_fast, 9);
- zeros_fast -= (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 22);
-
- gate = 1000 + zeros_fast - zeros - stt->vadNearend.stdShortTerm;
-
- if (gate < 0)
- {
- stt->gatePrevious = 0;
- } else
- {
- tmp32 = WEBRTC_SPL_MUL_16_16(stt->gatePrevious, 7);
- gate = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32((WebRtc_Word32)gate + tmp32, 3);
- stt->gatePrevious = gate;
- }
- // gate < 0 -> no gate
- // gate > 2500 -> max gate
- if (gate > 0)
- {
- if (gate < 2500)
- {
- gain_adj = WEBRTC_SPL_RSHIFT_W16(2500 - gate, 5);
- } else
- {
- gain_adj = 0;
- }
- for (k = 0; k < 10; k++)
- {
- if ((gains[k + 1] - stt->gainTable[0]) > 8388608)
- {
- // To prevent wraparound
- tmp32 = WEBRTC_SPL_RSHIFT_W32((gains[k+1] - stt->gainTable[0]), 8);
- tmp32 = WEBRTC_SPL_MUL(tmp32, (178 + gain_adj));
- } else
- {
- tmp32 = WEBRTC_SPL_MUL((gains[k+1] - stt->gainTable[0]), (178 + gain_adj));
- tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 8);
- }
- gains[k + 1] = stt->gainTable[0] + tmp32;
- }
- }
-
- // Limit gain to avoid overload distortion
- for (k = 0; k < 10; k++)
- {
- // To prevent wrap around
- zeros = 10;
- if (gains[k + 1] > 47453132)
- {
- zeros = 16 - WebRtcSpl_NormW32(gains[k + 1]);
- }
- gain32 = WEBRTC_SPL_RSHIFT_W32(gains[k+1], zeros) + 1;
- gain32 = WEBRTC_SPL_MUL(gain32, gain32);
- // check for overflow
- while (AGC_MUL32(WEBRTC_SPL_RSHIFT_W32(env[k], 12) + 1, gain32)
- > WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)32767, 2 * (1 - zeros + 10)))
- {
- // multiply by 253/256 ==> -0.1 dB
- if (gains[k + 1] > 8388607)
- {
- // Prevent wrap around
- gains[k + 1] = WEBRTC_SPL_MUL(WEBRTC_SPL_RSHIFT_W32(gains[k+1], 8), 253);
- } else
- {
- gains[k + 1] = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(gains[k+1], 253), 8);
- }
- gain32 = WEBRTC_SPL_RSHIFT_W32(gains[k+1], zeros) + 1;
- gain32 = WEBRTC_SPL_MUL(gain32, gain32);
- }
- }
- // gain reductions should be done 1 ms earlier than gain increases
- for (k = 1; k < 10; k++)
- {
- if (gains[k] > gains[k + 1])
- {
- gains[k] = gains[k + 1];
- }
- }
- // save start gain for next frame
- stt->gain = gains[10];
-
- // Apply gain
- // handle first sub frame separately
- delta = WEBRTC_SPL_LSHIFT_W32(gains[1] - gains[0], (4 - L2));
- gain32 = WEBRTC_SPL_LSHIFT_W32(gains[0], 4);
- // iterate over samples
- for (n = 0; n < L; n++)
- {
- // For lower band
- tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out[n], WEBRTC_SPL_RSHIFT_W32(gain32 + 127, 7));
- out_tmp = WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
- if (out_tmp > 4095)
- {
- out[n] = (WebRtc_Word16)32767;
- } else if (out_tmp < -4096)
- {
- out[n] = (WebRtc_Word16)-32768;
- } else
- {
- tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out[n], WEBRTC_SPL_RSHIFT_W32(gain32, 4));
- out[n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
- }
- // For higher band
- if (FS == 32000)
- {
- tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out_H[n],
- WEBRTC_SPL_RSHIFT_W32(gain32 + 127, 7));
- out_tmp = WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
- if (out_tmp > 4095)
- {
- out_H[n] = (WebRtc_Word16)32767;
- } else if (out_tmp < -4096)
- {
- out_H[n] = (WebRtc_Word16)-32768;
- } else
- {
- tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out_H[n],
- WEBRTC_SPL_RSHIFT_W32(gain32, 4));
- out_H[n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
- }
- }
- //
-
- gain32 += delta;
- }
- // iterate over subframes
- for (k = 1; k < 10; k++)
- {
- delta = WEBRTC_SPL_LSHIFT_W32(gains[k+1] - gains[k], (4 - L2));
- gain32 = WEBRTC_SPL_LSHIFT_W32(gains[k], 4);
- // iterate over samples
- for (n = 0; n < L; n++)
- {
- // For lower band
- tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out[k * L + n],
- WEBRTC_SPL_RSHIFT_W32(gain32, 4));
- out[k * L + n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
- // For higher band
- if (FS == 32000)
- {
- tmp32 = WEBRTC_SPL_MUL((WebRtc_Word32)out_H[k * L + n],
- WEBRTC_SPL_RSHIFT_W32(gain32, 4));
- out_H[k * L + n] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32 , 16);
- }
- gain32 += delta;
- }
- }
-
- return 0;
-}
-
-void WebRtcAgc_InitVad(AgcVad_t *state)
-{
- WebRtc_Word16 k;
-
- state->HPstate = 0; // state of high pass filter
- state->logRatio = 0; // log( P(active) / P(inactive) )
- // average input level (Q10)
- state->meanLongTerm = WEBRTC_SPL_LSHIFT_W16(15, 10);
-
- // variance of input level (Q8)
- state->varianceLongTerm = WEBRTC_SPL_LSHIFT_W32(500, 8);
-
- state->stdLongTerm = 0; // standard deviation of input level in dB
- // short-term average input level (Q10)
- state->meanShortTerm = WEBRTC_SPL_LSHIFT_W16(15, 10);
-
- // short-term variance of input level (Q8)
- state->varianceShortTerm = WEBRTC_SPL_LSHIFT_W32(500, 8);
-
- state->stdShortTerm = 0; // short-term standard deviation of input level in dB
- state->counter = 3; // counts updates
- for (k = 0; k < 8; k++)
- {
- // downsampling filter
- state->downState[k] = 0;
- }
-}
-
-WebRtc_Word16 WebRtcAgc_ProcessVad(AgcVad_t *state, // (i) VAD state
- const WebRtc_Word16 *in, // (i) Speech signal
- WebRtc_Word16 nrSamples) // (i) number of samples
-{
- WebRtc_Word32 out, nrg, tmp32, tmp32b;
- WebRtc_UWord16 tmpU16;
- WebRtc_Word16 k, subfr, tmp16;
- WebRtc_Word16 buf1[8];
- WebRtc_Word16 buf2[4];
- WebRtc_Word16 HPstate;
- WebRtc_Word16 zeros, dB;
-
- // process in 10 sub frames of 1 ms (to save on memory)
- nrg = 0;
- HPstate = state->HPstate;
- for (subfr = 0; subfr < 10; subfr++)
- {
- // downsample to 4 kHz
- if (nrSamples == 160)
- {
- for (k = 0; k < 8; k++)
- {
- tmp32 = (WebRtc_Word32)in[2 * k] + (WebRtc_Word32)in[2 * k + 1];
- tmp32 = WEBRTC_SPL_RSHIFT_W32(tmp32, 1);
- buf1[k] = (WebRtc_Word16)tmp32;
- }
- in += 16;
-
- WebRtcSpl_DownsampleBy2(buf1, 8, buf2, state->downState);
- } else
- {
- WebRtcSpl_DownsampleBy2(in, 8, buf2, state->downState);
- in += 8;
- }
-
- // high pass filter and compute energy
- for (k = 0; k < 4; k++)
- {
- out = buf2[k] + HPstate;
- tmp32 = WEBRTC_SPL_MUL(600, out);
- HPstate = (WebRtc_Word16)(WEBRTC_SPL_RSHIFT_W32(tmp32, 10) - buf2[k]);
- tmp32 = WEBRTC_SPL_MUL(out, out);
- nrg += WEBRTC_SPL_RSHIFT_W32(tmp32, 6);
- }
- }
- state->HPstate = HPstate;
-
- // find number of leading zeros
- if (!(0xFFFF0000 & nrg))
- {
- zeros = 16;
- } else
- {
- zeros = 0;
- }
- if (!(0xFF000000 & (nrg << zeros)))
- {
- zeros += 8;
- }
- if (!(0xF0000000 & (nrg << zeros)))
- {
- zeros += 4;
- }
- if (!(0xC0000000 & (nrg << zeros)))
- {
- zeros += 2;
- }
- if (!(0x80000000 & (nrg << zeros)))
- {
- zeros += 1;
- }
-
- // energy level (range {-32..30}) (Q10)
- dB = WEBRTC_SPL_LSHIFT_W16(15 - zeros, 11);
-
- // Update statistics
-
- if (state->counter < kAvgDecayTime)
- {
- // decay time = AvgDecTime * 10 ms
- state->counter++;
- }
-
- // update short-term estimate of mean energy level (Q10)
- tmp32 = (WEBRTC_SPL_MUL_16_16(state->meanShortTerm, 15) + (WebRtc_Word32)dB);
- state->meanShortTerm = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
-
- // update short-term estimate of variance in energy level (Q8)
- tmp32 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(dB, dB), 12);
- tmp32 += WEBRTC_SPL_MUL(state->varianceShortTerm, 15);
- state->varianceShortTerm = WEBRTC_SPL_RSHIFT_W32(tmp32, 4);
-
- // update short-term estimate of standard deviation in energy level (Q10)
- tmp32 = WEBRTC_SPL_MUL_16_16(state->meanShortTerm, state->meanShortTerm);
- tmp32 = WEBRTC_SPL_LSHIFT_W32(state->varianceShortTerm, 12) - tmp32;
- state->stdShortTerm = (WebRtc_Word16)WebRtcSpl_Sqrt(tmp32);
-
- // update long-term estimate of mean energy level (Q10)
- tmp32 = WEBRTC_SPL_MUL_16_16(state->meanLongTerm, state->counter) + (WebRtc_Word32)dB;
- state->meanLongTerm = WebRtcSpl_DivW32W16ResW16(tmp32,
- WEBRTC_SPL_ADD_SAT_W16(state->counter, 1));
-
- // update long-term estimate of variance in energy level (Q8)
- tmp32 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(dB, dB), 12);
- tmp32 += WEBRTC_SPL_MUL(state->varianceLongTerm, state->counter);
- state->varianceLongTerm = WebRtcSpl_DivW32W16(tmp32,
- WEBRTC_SPL_ADD_SAT_W16(state->counter, 1));
-
- // update long-term estimate of standard deviation in energy level (Q10)
- tmp32 = WEBRTC_SPL_MUL_16_16(state->meanLongTerm, state->meanLongTerm);
- tmp32 = WEBRTC_SPL_LSHIFT_W32(state->varianceLongTerm, 12) - tmp32;
- state->stdLongTerm = (WebRtc_Word16)WebRtcSpl_Sqrt(tmp32);
-
- // update voice activity measure (Q10)
- tmp16 = WEBRTC_SPL_LSHIFT_W16(3, 12);
- tmp32 = WEBRTC_SPL_MUL_16_16(tmp16, (dB - state->meanLongTerm));
- tmp32 = WebRtcSpl_DivW32W16(tmp32, state->stdLongTerm);
- tmpU16 = WEBRTC_SPL_LSHIFT_U16((WebRtc_UWord16)13, 12);
- tmp32b = WEBRTC_SPL_MUL_16_U16(state->logRatio, tmpU16);
- tmp32 += WEBRTC_SPL_RSHIFT_W32(tmp32b, 10);
-
- state->logRatio = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32, 6);
-
- // limit
- if (state->logRatio > 2048)
- {
- state->logRatio = 2048;
- }
- if (state->logRatio < -2048)
- {
- state->logRatio = -2048;
- }
-
- return state->logRatio; // Q10
-}
diff --git a/trunk/src/modules/audio_processing/agc/digital_agc.h b/trunk/src/modules/audio_processing/agc/digital_agc.h
deleted file mode 100644
index 240b220..0000000
--- a/trunk/src/modules/audio_processing/agc/digital_agc.h
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_DIGITAL_AGC_H_
-
-#ifdef AGC_DEBUG
-#include <stdio.h>
-#endif
-#include "typedefs.h"
-#include "signal_processing_library.h"
-
-// the 32 most significant bits of A(19) * B(26) >> 13
-#define AGC_MUL32(A, B) (((B)>>13)*(A) + ( ((0x00001FFF & (B))*(A)) >> 13 ))
-// C + the 32 most significant bits of A * B
-#define AGC_SCALEDIFF32(A, B, C) ((C) + ((B)>>16)*(A) + ( ((0x0000FFFF & (B))*(A)) >> 16 ))
-
-typedef struct
-{
- WebRtc_Word32 downState[8];
- WebRtc_Word16 HPstate;
- WebRtc_Word16 counter;
- WebRtc_Word16 logRatio; // log( P(active) / P(inactive) ) (Q10)
- WebRtc_Word16 meanLongTerm; // Q10
- WebRtc_Word32 varianceLongTerm; // Q8
- WebRtc_Word16 stdLongTerm; // Q10
- WebRtc_Word16 meanShortTerm; // Q10
- WebRtc_Word32 varianceShortTerm; // Q8
- WebRtc_Word16 stdShortTerm; // Q10
-} AgcVad_t; // total = 54 bytes
-
-typedef struct
-{
- WebRtc_Word32 capacitorSlow;
- WebRtc_Word32 capacitorFast;
- WebRtc_Word32 gain;
- WebRtc_Word32 gainTable[32];
- WebRtc_Word16 gatePrevious;
- WebRtc_Word16 agcMode;
- AgcVad_t vadNearend;
- AgcVad_t vadFarend;
-#ifdef AGC_DEBUG
- FILE* logFile;
- int frameCounter;
-#endif
-} DigitalAgc_t;
-
-WebRtc_Word32 WebRtcAgc_InitDigital(DigitalAgc_t *digitalAgcInst, WebRtc_Word16 agcMode);
-
-WebRtc_Word32 WebRtcAgc_ProcessDigital(DigitalAgc_t *digitalAgcInst, const WebRtc_Word16 *inNear,
- const WebRtc_Word16 *inNear_H, WebRtc_Word16 *out,
- WebRtc_Word16 *out_H, WebRtc_UWord32 FS,
- WebRtc_Word16 lowLevelSignal);
-
-WebRtc_Word32 WebRtcAgc_AddFarendToDigital(DigitalAgc_t *digitalAgcInst, const WebRtc_Word16 *inFar,
- WebRtc_Word16 nrSamples);
-
-void WebRtcAgc_InitVad(AgcVad_t *vadInst);
-
-WebRtc_Word16 WebRtcAgc_ProcessVad(AgcVad_t *vadInst, // (i) VAD state
- const WebRtc_Word16 *in, // (i) Speech signal
- WebRtc_Word16 nrSamples); // (i) number of samples
-
-WebRtc_Word32 WebRtcAgc_CalculateGainTable(WebRtc_Word32 *gainTable, // Q16
- WebRtc_Word16 compressionGaindB, // Q0 (in dB)
- WebRtc_Word16 targetLevelDbfs,// Q0 (in dB)
- WebRtc_UWord8 limiterEnable, WebRtc_Word16 analogTarget);
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_MAIN_SOURCE_ANALOG_AGC_H_
diff --git a/trunk/src/modules/audio_processing/agc/include/gain_control.h b/trunk/src/modules/audio_processing/agc/include/gain_control.h
deleted file mode 100644
index 8af5c71..0000000
--- a/trunk/src/modules/audio_processing/agc/include/gain_control.h
+++ /dev/null
@@ -1,259 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
-
-#include "typedefs.h"
-
-// Errors
-#define AGC_UNSPECIFIED_ERROR 18000
-#define AGC_UNSUPPORTED_FUNCTION_ERROR 18001
-#define AGC_UNINITIALIZED_ERROR 18002
-#define AGC_NULL_POINTER_ERROR 18003
-#define AGC_BAD_PARAMETER_ERROR 18004
-
-// Warnings
-#define AGC_BAD_PARAMETER_WARNING 18050
-
-enum
-{
- kAgcModeUnchanged,
- kAgcModeAdaptiveAnalog,
- kAgcModeAdaptiveDigital,
- kAgcModeFixedDigital
-};
-
-enum
-{
- kAgcFalse = 0,
- kAgcTrue
-};
-
-typedef struct
-{
- WebRtc_Word16 targetLevelDbfs; // default 3 (-3 dBOv)
- WebRtc_Word16 compressionGaindB; // default 9 dB
- WebRtc_UWord8 limiterEnable; // default kAgcTrue (on)
-} WebRtcAgc_config_t;
-
-#if defined(__cplusplus)
-extern "C"
-{
-#endif
-
-/*
- * This function processes a 10/20ms frame of far-end speech to determine
- * if there is active speech. Far-end speech length can be either 10ms or
- * 20ms. The length of the input speech vector must be given in samples
- * (80/160 when FS=8000, and 160/320 when FS=16000 or FS=32000).
- *
- * Input:
- * - agcInst : AGC instance.
- * - inFar : Far-end input speech vector (10 or 20ms)
- * - samples : Number of samples in input vector
- *
- * Return value:
- * : 0 - Normal operation.
- * : -1 - Error
- */
-int WebRtcAgc_AddFarend(void* agcInst,
- const WebRtc_Word16* inFar,
- WebRtc_Word16 samples);
-
-/*
- * This function processes a 10/20ms frame of microphone speech to determine
- * if there is active speech. Microphone speech length can be either 10ms or
- * 20ms. The length of the input speech vector must be given in samples
- * (80/160 when FS=8000, and 160/320 when FS=16000 or FS=32000). For very low
- * input levels, the input signal is increased in level by multiplying and
- * overwriting the samples in inMic[].
- *
- * This function should be called before any further processing of the
- * near-end microphone signal.
- *
- * Input:
- * - agcInst : AGC instance.
- * - inMic : Microphone input speech vector (10 or 20 ms) for
- * L band
- * - inMic_H : Microphone input speech vector (10 or 20 ms) for
- * H band
- * - samples : Number of samples in input vector
- *
- * Return value:
- * : 0 - Normal operation.
- * : -1 - Error
- */
-int WebRtcAgc_AddMic(void* agcInst,
- WebRtc_Word16* inMic,
- WebRtc_Word16* inMic_H,
- WebRtc_Word16 samples);
-
-/*
- * This function replaces the analog microphone with a virtual one.
- * It is a digital gain applied to the input signal and is used in the
- * agcAdaptiveDigital mode where no microphone level is adjustable.
- * Microphone speech length can be either 10ms or 20ms. The length of the
- * input speech vector must be given in samples (80/160 when FS=8000, and
- * 160/320 when FS=16000 or FS=32000).
- *
- * Input:
- * - agcInst : AGC instance.
- * - inMic : Microphone input speech vector for (10 or 20 ms)
- * L band
- * - inMic_H : Microphone input speech vector for (10 or 20 ms)
- * H band
- * - samples : Number of samples in input vector
- * - micLevelIn : Input level of microphone (static)
- *
- * Output:
- * - inMic : Microphone output after processing (L band)
- * - inMic_H : Microphone output after processing (H band)
- * - micLevelOut : Adjusted microphone level after processing
- *
- * Return value:
- * : 0 - Normal operation.
- * : -1 - Error
- */
-int WebRtcAgc_VirtualMic(void* agcInst,
- WebRtc_Word16* inMic,
- WebRtc_Word16* inMic_H,
- WebRtc_Word16 samples,
- WebRtc_Word32 micLevelIn,
- WebRtc_Word32* micLevelOut);
-
-/*
- * This function processes a 10/20ms frame and adjusts (normalizes) the gain
- * both analog and digitally. The gain adjustments are done only during
- * active periods of speech. The input speech length can be either 10ms or
- * 20ms and the output is of the same length. The length of the speech
- * vectors must be given in samples (80/160 when FS=8000, and 160/320 when
- * FS=16000 or FS=32000). The echo parameter can be used to ensure the AGC will
- * not adjust upward in the presence of echo.
- *
- * This function should be called after processing the near-end microphone
- * signal, in any case after any echo cancellation.
- *
- * Input:
- * - agcInst : AGC instance
- * - inNear : Near-end input speech vector (10 or 20 ms) for
- * L band
- * - inNear_H : Near-end input speech vector (10 or 20 ms) for
- * H band
- * - samples : Number of samples in input/output vector
- * - inMicLevel : Current microphone volume level
- * - echo : Set to 0 if the signal passed to add_mic is
- * almost certainly free of echo; otherwise set
- * to 1. If you have no information regarding echo
- * set to 0.
- *
- * Output:
- * - outMicLevel : Adjusted microphone volume level
- * - out : Gain-adjusted near-end speech vector (L band)
- * : May be the same vector as the input.
- * - out_H : Gain-adjusted near-end speech vector (H band)
- * - saturationWarning : A returned value of 1 indicates a saturation event
- * has occurred and the volume cannot be further
- * reduced. Otherwise will be set to 0.
- *
- * Return value:
- * : 0 - Normal operation.
- * : -1 - Error
- */
-int WebRtcAgc_Process(void* agcInst,
- const WebRtc_Word16* inNear,
- const WebRtc_Word16* inNear_H,
- WebRtc_Word16 samples,
- WebRtc_Word16* out,
- WebRtc_Word16* out_H,
- WebRtc_Word32 inMicLevel,
- WebRtc_Word32* outMicLevel,
- WebRtc_Word16 echo,
- WebRtc_UWord8* saturationWarning);
-
-/*
- * This function sets the config parameters (targetLevelDbfs,
- * compressionGaindB and limiterEnable).
- *
- * Input:
- * - agcInst : AGC instance
- * - config : config struct
- *
- * Output:
- *
- * Return value:
- * : 0 - Normal operation.
- * : -1 - Error
- */
-int WebRtcAgc_set_config(void* agcInst, WebRtcAgc_config_t config);
-
-/*
- * This function returns the config parameters (targetLevelDbfs,
- * compressionGaindB and limiterEnable).
- *
- * Input:
- * - agcInst : AGC instance
- *
- * Output:
- * - config : config struct
- *
- * Return value:
- * : 0 - Normal operation.
- * : -1 - Error
- */
-int WebRtcAgc_get_config(void* agcInst, WebRtcAgc_config_t* config);
-
-/*
- * This function creates an AGC instance, which will contain the state
- * information for one (duplex) channel.
- *
- * Return value : AGC instance if successful
- * : 0 (i.e., a NULL pointer) if unsuccessful
- */
-int WebRtcAgc_Create(void **agcInst);
-
-/*
- * This function frees the AGC instance created at the beginning.
- *
- * Input:
- * - agcInst : AGC instance.
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcAgc_Free(void *agcInst);
-
-/*
- * This function initializes an AGC instance.
- *
- * Input:
- * - agcInst : AGC instance.
- * - minLevel : Minimum possible mic level
- * - maxLevel : Maximum possible mic level
- * - agcMode : 0 - Unchanged
- * : 1 - Adaptive Analog Automatic Gain Control -3dBOv
- * : 2 - Adaptive Digital Automatic Gain Control -3dBOv
- * : 3 - Fixed Digital Gain 0dB
- * - fs : Sampling frequency
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcAgc_Init(void *agcInst,
- WebRtc_Word32 minLevel,
- WebRtc_Word32 maxLevel,
- WebRtc_Word16 agcMode,
- WebRtc_UWord32 fs);
-
-#if defined(__cplusplus)
-}
-#endif
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_AGC_INCLUDE_GAIN_CONTROL_H_
diff --git a/trunk/src/modules/audio_processing/apm_tests.gypi b/trunk/src/modules/audio_processing/apm_tests.gypi
deleted file mode 100644
index c00a6d2..0000000
--- a/trunk/src/modules/audio_processing/apm_tests.gypi
+++ /dev/null
@@ -1,75 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'audioproc_unittest',
- 'type': 'executable',
- 'conditions': [
- ['prefer_fixed_point==1', {
- 'defines': [ 'WEBRTC_AUDIOPROC_FIXED_PROFILE' ],
- }, {
- 'defines': [ 'WEBRTC_AUDIOPROC_FLOAT_PROFILE' ],
- }],
- ['enable_protobuf==1', {
- 'defines': [ 'WEBRTC_AUDIOPROC_DEBUG_DUMP' ],
- }],
- ],
- 'dependencies': [
- 'audio_processing',
- 'audioproc_unittest_proto',
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/../test/test.gyp:test_support',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'sources': [ 'test/unit_test.cc', ],
- },
- {
- 'target_name': 'audioproc_unittest_proto',
- 'type': 'static_library',
- 'sources': [ 'test/unittest.proto', ],
- 'variables': {
- 'proto_in_dir': 'test',
- # Workaround to protect against gyp's pathname relativization when this
- # file is included by modules.gyp.
- 'proto_out_protected': 'webrtc/audio_processing',
- 'proto_out_dir': '<(proto_out_protected)',
- },
- 'includes': [ '../../build/protoc.gypi', ],
- },
- ],
- 'conditions': [
- ['enable_protobuf==1', {
- 'targets': [
- {
- 'target_name': 'audioproc',
- 'type': 'executable',
- 'dependencies': [
- 'audio_processing',
- 'audioproc_debug_proto',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'sources': [ 'test/process_test.cc', ],
- },
- {
- 'target_name': 'unpack_aecdump',
- 'type': 'executable',
- 'dependencies': [
- 'audioproc_debug_proto',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/../third_party/google-gflags/google-gflags.gyp:google-gflags',
- ],
- 'sources': [ 'test/unpack.cc', ],
- },
- ],
- }],
- ],
-}
diff --git a/trunk/src/modules/audio_processing/audio_buffer.cc b/trunk/src/modules/audio_processing/audio_buffer.cc
deleted file mode 100644
index a7fb04d..0000000
--- a/trunk/src/modules/audio_processing/audio_buffer.cc
+++ /dev/null
@@ -1,306 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_buffer.h"
-
-#include "signal_processing_library.h"
-
-namespace webrtc {
-namespace {
-
-enum {
- kSamplesPer8kHzChannel = 80,
- kSamplesPer16kHzChannel = 160,
- kSamplesPer32kHzChannel = 320
-};
-
-void StereoToMono(const int16_t* left, const int16_t* right,
- int16_t* out, int samples_per_channel) {
- assert(left != NULL && right != NULL && out != NULL);
- for (int i = 0; i < samples_per_channel; i++) {
- int32_t data32 = (static_cast<int32_t>(left[i]) +
- static_cast<int32_t>(right[i])) >> 1;
-
- out[i] = WebRtcSpl_SatW32ToW16(data32);
- }
-}
-} // namespace
-
-struct AudioChannel {
- AudioChannel() {
- memset(data, 0, sizeof(data));
- }
-
- int16_t data[kSamplesPer32kHzChannel];
-};
-
-struct SplitAudioChannel {
- SplitAudioChannel() {
- memset(low_pass_data, 0, sizeof(low_pass_data));
- memset(high_pass_data, 0, sizeof(high_pass_data));
- memset(analysis_filter_state1, 0, sizeof(analysis_filter_state1));
- memset(analysis_filter_state2, 0, sizeof(analysis_filter_state2));
- memset(synthesis_filter_state1, 0, sizeof(synthesis_filter_state1));
- memset(synthesis_filter_state2, 0, sizeof(synthesis_filter_state2));
- }
-
- int16_t low_pass_data[kSamplesPer16kHzChannel];
- int16_t high_pass_data[kSamplesPer16kHzChannel];
-
- WebRtc_Word32 analysis_filter_state1[6];
- WebRtc_Word32 analysis_filter_state2[6];
- WebRtc_Word32 synthesis_filter_state1[6];
- WebRtc_Word32 synthesis_filter_state2[6];
-};
-
-// TODO(andrew): check range of input parameters?
-AudioBuffer::AudioBuffer(int max_num_channels,
- int samples_per_channel)
- : max_num_channels_(max_num_channels),
- num_channels_(0),
- num_mixed_channels_(0),
- num_mixed_low_pass_channels_(0),
- data_was_mixed_(false),
- samples_per_channel_(samples_per_channel),
- samples_per_split_channel_(samples_per_channel),
- reference_copied_(false),
- activity_(AudioFrame::kVadUnknown),
- is_muted_(false),
- data_(NULL),
- channels_(NULL),
- split_channels_(NULL),
- mixed_channels_(NULL),
- mixed_low_pass_channels_(NULL),
- low_pass_reference_channels_(NULL) {
- if (max_num_channels_ > 1) {
- channels_.reset(new AudioChannel[max_num_channels_]);
- mixed_channels_.reset(new AudioChannel[max_num_channels_]);
- mixed_low_pass_channels_.reset(new AudioChannel[max_num_channels_]);
- }
- low_pass_reference_channels_.reset(new AudioChannel[max_num_channels_]);
-
- if (samples_per_channel_ == kSamplesPer32kHzChannel) {
- split_channels_.reset(new SplitAudioChannel[max_num_channels_]);
- samples_per_split_channel_ = kSamplesPer16kHzChannel;
- }
-}
-
-AudioBuffer::~AudioBuffer() {}
-
-int16_t* AudioBuffer::data(int channel) const {
- assert(channel >= 0 && channel < num_channels_);
- if (data_ != NULL) {
- return data_;
- }
-
- return channels_[channel].data;
-}
-
-int16_t* AudioBuffer::low_pass_split_data(int channel) const {
- assert(channel >= 0 && channel < num_channels_);
- if (split_channels_.get() == NULL) {
- return data(channel);
- }
-
- return split_channels_[channel].low_pass_data;
-}
-
-int16_t* AudioBuffer::high_pass_split_data(int channel) const {
- assert(channel >= 0 && channel < num_channels_);
- if (split_channels_.get() == NULL) {
- return NULL;
- }
-
- return split_channels_[channel].high_pass_data;
-}
-
-int16_t* AudioBuffer::mixed_data(int channel) const {
- assert(channel >= 0 && channel < num_mixed_channels_);
-
- return mixed_channels_[channel].data;
-}
-
-int16_t* AudioBuffer::mixed_low_pass_data(int channel) const {
- assert(channel >= 0 && channel < num_mixed_low_pass_channels_);
-
- return mixed_low_pass_channels_[channel].data;
-}
-
-int16_t* AudioBuffer::low_pass_reference(int channel) const {
- assert(channel >= 0 && channel < num_channels_);
- if (!reference_copied_) {
- return NULL;
- }
-
- return low_pass_reference_channels_[channel].data;
-}
-
-WebRtc_Word32* AudioBuffer::analysis_filter_state1(int channel) const {
- assert(channel >= 0 && channel < num_channels_);
- return split_channels_[channel].analysis_filter_state1;
-}
-
-WebRtc_Word32* AudioBuffer::analysis_filter_state2(int channel) const {
- assert(channel >= 0 && channel < num_channels_);
- return split_channels_[channel].analysis_filter_state2;
-}
-
-WebRtc_Word32* AudioBuffer::synthesis_filter_state1(int channel) const {
- assert(channel >= 0 && channel < num_channels_);
- return split_channels_[channel].synthesis_filter_state1;
-}
-
-WebRtc_Word32* AudioBuffer::synthesis_filter_state2(int channel) const {
- assert(channel >= 0 && channel < num_channels_);
- return split_channels_[channel].synthesis_filter_state2;
-}
-
-void AudioBuffer::set_activity(AudioFrame::VADActivity activity) {
- activity_ = activity;
-}
-
-AudioFrame::VADActivity AudioBuffer::activity() const {
- return activity_;
-}
-
-bool AudioBuffer::is_muted() const {
- return is_muted_;
-}
-
-int AudioBuffer::num_channels() const {
- return num_channels_;
-}
-
-int AudioBuffer::samples_per_channel() const {
- return samples_per_channel_;
-}
-
-int AudioBuffer::samples_per_split_channel() const {
- return samples_per_split_channel_;
-}
-
-// TODO(andrew): Do deinterleaving and mixing in one step?
-void AudioBuffer::DeinterleaveFrom(AudioFrame* frame) {
- assert(frame->_audioChannel <= max_num_channels_);
- assert(frame->_payloadDataLengthInSamples == samples_per_channel_);
-
- num_channels_ = frame->_audioChannel;
- data_was_mixed_ = false;
- num_mixed_channels_ = 0;
- num_mixed_low_pass_channels_ = 0;
- reference_copied_ = false;
- activity_ = frame->_vadActivity;
- is_muted_ = false;
- if (frame->_energy == 0) {
- is_muted_ = true;
- }
-
- if (num_channels_ == 1) {
- // We can get away with a pointer assignment in this case.
- data_ = frame->_payloadData;
- return;
- }
-
- int16_t* interleaved = frame->_payloadData;
- for (int i = 0; i < num_channels_; i++) {
- int16_t* deinterleaved = channels_[i].data;
- int interleaved_idx = i;
- for (int j = 0; j < samples_per_channel_; j++) {
- deinterleaved[j] = interleaved[interleaved_idx];
- interleaved_idx += num_channels_;
- }
- }
-}
-
-void AudioBuffer::InterleaveTo(AudioFrame* frame, bool data_changed) const {
- assert(frame->_audioChannel == num_channels_);
- assert(frame->_payloadDataLengthInSamples == samples_per_channel_);
- frame->_vadActivity = activity_;
-
- if (!data_changed) {
- return;
- }
-
- if (num_channels_ == 1) {
- if (data_was_mixed_) {
- memcpy(frame->_payloadData,
- channels_[0].data,
- sizeof(int16_t) * samples_per_channel_);
- } else {
- // These should point to the same buffer in this case.
- assert(data_ == frame->_payloadData);
- }
-
- return;
- }
-
- int16_t* interleaved = frame->_payloadData;
- for (int i = 0; i < num_channels_; i++) {
- int16_t* deinterleaved = channels_[i].data;
- int interleaved_idx = i;
- for (int j = 0; j < samples_per_channel_; j++) {
- interleaved[interleaved_idx] = deinterleaved[j];
- interleaved_idx += num_channels_;
- }
- }
-}
-
-// TODO(andrew): would be good to support the no-mix case with pointer
-// assignment.
-// TODO(andrew): handle mixing to multiple channels?
-void AudioBuffer::Mix(int num_mixed_channels) {
- // We currently only support the stereo to mono case.
- assert(num_channels_ == 2);
- assert(num_mixed_channels == 1);
-
- StereoToMono(channels_[0].data,
- channels_[1].data,
- channels_[0].data,
- samples_per_channel_);
-
- num_channels_ = num_mixed_channels;
- data_was_mixed_ = true;
-}
-
-void AudioBuffer::CopyAndMix(int num_mixed_channels) {
- // We currently only support the stereo to mono case.
- assert(num_channels_ == 2);
- assert(num_mixed_channels == 1);
-
- StereoToMono(channels_[0].data,
- channels_[1].data,
- mixed_channels_[0].data,
- samples_per_channel_);
-
- num_mixed_channels_ = num_mixed_channels;
-}
-
-void AudioBuffer::CopyAndMixLowPass(int num_mixed_channels) {
- // We currently only support the stereo to mono case.
- assert(num_channels_ == 2);
- assert(num_mixed_channels == 1);
-
- StereoToMono(low_pass_split_data(0),
- low_pass_split_data(1),
- mixed_low_pass_channels_[0].data,
- samples_per_split_channel_);
-
- num_mixed_low_pass_channels_ = num_mixed_channels;
-}
-
-void AudioBuffer::CopyLowPassToReference() {
- reference_copied_ = true;
- for (int i = 0; i < num_channels_; i++) {
- memcpy(low_pass_reference_channels_[i].data,
- low_pass_split_data(i),
- sizeof(int16_t) * samples_per_split_channel_);
- }
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/audio_buffer.h b/trunk/src/modules/audio_processing/audio_buffer.h
deleted file mode 100644
index 87d6972..0000000
--- a/trunk/src/modules/audio_processing/audio_buffer.h
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
-
-#include "module_common_types.h"
-#include "scoped_ptr.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-struct AudioChannel;
-struct SplitAudioChannel;
-
-class AudioBuffer {
- public:
- AudioBuffer(int max_num_channels, int samples_per_channel);
- virtual ~AudioBuffer();
-
- int num_channels() const;
- int samples_per_channel() const;
- int samples_per_split_channel() const;
-
- int16_t* data(int channel) const;
- int16_t* low_pass_split_data(int channel) const;
- int16_t* high_pass_split_data(int channel) const;
- int16_t* mixed_data(int channel) const;
- int16_t* mixed_low_pass_data(int channel) const;
- int16_t* low_pass_reference(int channel) const;
-
- int32_t* analysis_filter_state1(int channel) const;
- int32_t* analysis_filter_state2(int channel) const;
- int32_t* synthesis_filter_state1(int channel) const;
- int32_t* synthesis_filter_state2(int channel) const;
-
- void set_activity(AudioFrame::VADActivity activity);
- AudioFrame::VADActivity activity() const;
-
- bool is_muted() const;
-
- void DeinterleaveFrom(AudioFrame* audioFrame);
- void InterleaveTo(AudioFrame* audioFrame) const;
- // If |data_changed| is false, only the non-audio data members will be copied
- // to |frame|.
- void InterleaveTo(AudioFrame* frame, bool data_changed) const;
- void Mix(int num_mixed_channels);
- void CopyAndMix(int num_mixed_channels);
- void CopyAndMixLowPass(int num_mixed_channels);
- void CopyLowPassToReference();
-
- private:
- const int max_num_channels_;
- int num_channels_;
- int num_mixed_channels_;
- int num_mixed_low_pass_channels_;
- // Whether the original data was replaced with mixed data.
- bool data_was_mixed_;
- const int samples_per_channel_;
- int samples_per_split_channel_;
- bool reference_copied_;
- AudioFrame::VADActivity activity_;
- bool is_muted_;
-
- int16_t* data_;
- scoped_array<AudioChannel> channels_;
- scoped_array<SplitAudioChannel> split_channels_;
- scoped_array<AudioChannel> mixed_channels_;
- // TODO(andrew): improve this, we don't need the full 32 kHz space here.
- scoped_array<AudioChannel> mixed_low_pass_channels_;
- scoped_array<AudioChannel> low_pass_reference_channels_;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_BUFFER_H_
diff --git a/trunk/src/modules/audio_processing/audio_processing.gypi b/trunk/src/modules/audio_processing/audio_processing.gypi
deleted file mode 100644
index 00078c4..0000000
--- a/trunk/src/modules/audio_processing/audio_processing.gypi
+++ /dev/null
@@ -1,91 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'audio_processing',
- 'type': '<(library)',
- 'conditions': [
- ['prefer_fixed_point==1', {
- 'dependencies': [ 'ns_fix' ],
- 'defines': [ 'WEBRTC_NS_FIXED' ],
- }, {
- 'dependencies': [ 'ns' ],
- 'defines': [ 'WEBRTC_NS_FLOAT' ],
- }],
- ['enable_protobuf==1', {
- 'dependencies': [ 'audioproc_debug_proto' ],
- 'defines': [ 'WEBRTC_AUDIOPROC_DEBUG_DUMP' ],
- }],
- ],
- 'dependencies': [
- 'aec',
- 'aecm',
- 'agc',
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- '<(webrtc_root)/common_audio/common_audio.gyp:vad',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- 'include',
- '../interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- '../interface',
- ],
- },
- 'sources': [
- 'include/audio_processing.h',
- 'audio_buffer.cc',
- 'audio_buffer.h',
- 'audio_processing_impl.cc',
- 'audio_processing_impl.h',
- 'echo_cancellation_impl.cc',
- 'echo_cancellation_impl.h',
- 'echo_control_mobile_impl.cc',
- 'echo_control_mobile_impl.h',
- 'gain_control_impl.cc',
- 'gain_control_impl.h',
- 'high_pass_filter_impl.cc',
- 'high_pass_filter_impl.h',
- 'level_estimator_impl.cc',
- 'level_estimator_impl.h',
- 'noise_suppression_impl.cc',
- 'noise_suppression_impl.h',
- 'splitting_filter.cc',
- 'splitting_filter.h',
- 'processing_component.cc',
- 'processing_component.h',
- 'voice_detection_impl.cc',
- 'voice_detection_impl.h',
- ],
- },
- ],
- 'conditions': [
- ['enable_protobuf==1', {
- 'targets': [
- {
- 'target_name': 'audioproc_debug_proto',
- 'type': 'static_library',
- 'sources': [ 'debug.proto', ],
- 'variables': {
- 'proto_in_dir': '.',
- # Workaround to protect against gyp's pathname relativization when
- # this file is included by modules.gyp.
- 'proto_out_protected': 'webrtc/audio_processing',
- 'proto_out_dir': '<(proto_out_protected)',
- },
- 'includes': [ '../../build/protoc.gypi', ],
- },
- ],
- }],
- ],
-}
diff --git a/trunk/src/modules/audio_processing/audio_processing_impl.cc b/trunk/src/modules/audio_processing/audio_processing_impl.cc
deleted file mode 100644
index 35d266d..0000000
--- a/trunk/src/modules/audio_processing/audio_processing_impl.cc
+++ /dev/null
@@ -1,643 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_processing_impl.h"
-
-#include <assert.h>
-
-#include "audio_buffer.h"
-#include "critical_section_wrapper.h"
-#include "echo_cancellation_impl.h"
-#include "echo_control_mobile_impl.h"
-#include "file_wrapper.h"
-#include "high_pass_filter_impl.h"
-#include "gain_control_impl.h"
-#include "level_estimator_impl.h"
-#include "module_common_types.h"
-#include "noise_suppression_impl.h"
-#include "processing_component.h"
-#include "splitting_filter.h"
-#include "voice_detection_impl.h"
-
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-// Files generated at build-time by the protobuf compiler.
-#ifdef WEBRTC_ANDROID
-#include "external/webrtc/src/modules/audio_processing/debug.pb.h"
-#else
-#include "webrtc/audio_processing/debug.pb.h"
-#endif
-#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
-
-namespace webrtc {
-AudioProcessing* AudioProcessing::Create(int id) {
-
- AudioProcessingImpl* apm = new AudioProcessingImpl(id);
- if (apm->Initialize() != kNoError) {
- delete apm;
- apm = NULL;
- }
-
- return apm;
-}
-
-void AudioProcessing::Destroy(AudioProcessing* apm) {
- delete static_cast<AudioProcessingImpl*>(apm);
-}
-
-AudioProcessingImpl::AudioProcessingImpl(int id)
- : id_(id),
- echo_cancellation_(NULL),
- echo_control_mobile_(NULL),
- gain_control_(NULL),
- high_pass_filter_(NULL),
- level_estimator_(NULL),
- noise_suppression_(NULL),
- voice_detection_(NULL),
- crit_(CriticalSectionWrapper::CreateCriticalSection()),
- render_audio_(NULL),
- capture_audio_(NULL),
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
- debug_file_(FileWrapper::Create()),
- event_msg_(new audioproc::Event()),
-#endif
- sample_rate_hz_(kSampleRate16kHz),
- split_sample_rate_hz_(kSampleRate16kHz),
- samples_per_channel_(sample_rate_hz_ / 100),
- stream_delay_ms_(0),
- was_stream_delay_set_(false),
- num_reverse_channels_(1),
- num_input_channels_(1),
- num_output_channels_(1) {
-
- echo_cancellation_ = new EchoCancellationImpl(this);
- component_list_.push_back(echo_cancellation_);
-
- echo_control_mobile_ = new EchoControlMobileImpl(this);
- component_list_.push_back(echo_control_mobile_);
-
- gain_control_ = new GainControlImpl(this);
- component_list_.push_back(gain_control_);
-
- high_pass_filter_ = new HighPassFilterImpl(this);
- component_list_.push_back(high_pass_filter_);
-
- level_estimator_ = new LevelEstimatorImpl(this);
- component_list_.push_back(level_estimator_);
-
- noise_suppression_ = new NoiseSuppressionImpl(this);
- component_list_.push_back(noise_suppression_);
-
- voice_detection_ = new VoiceDetectionImpl(this);
- component_list_.push_back(voice_detection_);
-}
-
-AudioProcessingImpl::~AudioProcessingImpl() {
- while (!component_list_.empty()) {
- ProcessingComponent* component = component_list_.front();
- component->Destroy();
- delete component;
- component_list_.pop_front();
- }
-
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
- if (debug_file_->Open()) {
- debug_file_->CloseFile();
- }
-#endif
-
- delete crit_;
- crit_ = NULL;
-
- if (render_audio_) {
- delete render_audio_;
- render_audio_ = NULL;
- }
-
- if (capture_audio_) {
- delete capture_audio_;
- capture_audio_ = NULL;
- }
-}
-
-CriticalSectionWrapper* AudioProcessingImpl::crit() const {
- return crit_;
-}
-
-int AudioProcessingImpl::split_sample_rate_hz() const {
- return split_sample_rate_hz_;
-}
-
-int AudioProcessingImpl::Initialize() {
- CriticalSectionScoped crit_scoped(crit_);
- return InitializeLocked();
-}
-
-int AudioProcessingImpl::InitializeLocked() {
- if (render_audio_ != NULL) {
- delete render_audio_;
- render_audio_ = NULL;
- }
-
- if (capture_audio_ != NULL) {
- delete capture_audio_;
- capture_audio_ = NULL;
- }
-
- render_audio_ = new AudioBuffer(num_reverse_channels_,
- samples_per_channel_);
- capture_audio_ = new AudioBuffer(num_input_channels_,
- samples_per_channel_);
-
- was_stream_delay_set_ = false;
-
- // Initialize all components.
- std::list<ProcessingComponent*>::iterator it;
- for (it = component_list_.begin(); it != component_list_.end(); it++) {
- int err = (*it)->Initialize();
- if (err != kNoError) {
- return err;
- }
- }
-
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
- if (debug_file_->Open()) {
- int err = WriteInitMessage();
- if (err != kNoError) {
- return err;
- }
- }
-#endif
-
- return kNoError;
-}
-
-int AudioProcessingImpl::set_sample_rate_hz(int rate) {
- CriticalSectionScoped crit_scoped(crit_);
- if (rate != kSampleRate8kHz &&
- rate != kSampleRate16kHz &&
- rate != kSampleRate32kHz) {
- return kBadParameterError;
- }
-
- sample_rate_hz_ = rate;
- samples_per_channel_ = rate / 100;
-
- if (sample_rate_hz_ == kSampleRate32kHz) {
- split_sample_rate_hz_ = kSampleRate16kHz;
- } else {
- split_sample_rate_hz_ = sample_rate_hz_;
- }
-
- return InitializeLocked();
-}
-
-int AudioProcessingImpl::sample_rate_hz() const {
- return sample_rate_hz_;
-}
-
-int AudioProcessingImpl::set_num_reverse_channels(int channels) {
- CriticalSectionScoped crit_scoped(crit_);
- // Only stereo supported currently.
- if (channels > 2 || channels < 1) {
- return kBadParameterError;
- }
-
- num_reverse_channels_ = channels;
-
- return InitializeLocked();
-}
-
-int AudioProcessingImpl::num_reverse_channels() const {
- return num_reverse_channels_;
-}
-
-int AudioProcessingImpl::set_num_channels(
- int input_channels,
- int output_channels) {
- CriticalSectionScoped crit_scoped(crit_);
- if (output_channels > input_channels) {
- return kBadParameterError;
- }
-
- // Only stereo supported currently.
- if (input_channels > 2 || input_channels < 1) {
- return kBadParameterError;
- }
-
- if (output_channels > 2 || output_channels < 1) {
- return kBadParameterError;
- }
-
- num_input_channels_ = input_channels;
- num_output_channels_ = output_channels;
-
- return InitializeLocked();
-}
-
-int AudioProcessingImpl::num_input_channels() const {
- return num_input_channels_;
-}
-
-int AudioProcessingImpl::num_output_channels() const {
- return num_output_channels_;
-}
-
-int AudioProcessingImpl::ProcessStream(AudioFrame* frame) {
- CriticalSectionScoped crit_scoped(crit_);
- int err = kNoError;
-
- if (frame == NULL) {
- return kNullPointerError;
- }
-
- if (frame->_frequencyInHz != sample_rate_hz_) {
- return kBadSampleRateError;
- }
-
- if (frame->_audioChannel != num_input_channels_) {
- return kBadNumberChannelsError;
- }
-
- if (frame->_payloadDataLengthInSamples != samples_per_channel_) {
- return kBadDataLengthError;
- }
-
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
- if (debug_file_->Open()) {
- event_msg_->set_type(audioproc::Event::STREAM);
- audioproc::Stream* msg = event_msg_->mutable_stream();
- const size_t data_size = sizeof(int16_t) *
- frame->_payloadDataLengthInSamples *
- frame->_audioChannel;
- msg->set_input_data(frame->_payloadData, data_size);
- msg->set_delay(stream_delay_ms_);
- msg->set_drift(echo_cancellation_->stream_drift_samples());
- msg->set_level(gain_control_->stream_analog_level());
- }
-#endif
-
- capture_audio_->DeinterleaveFrom(frame);
-
- // TODO(ajm): experiment with mixing and AEC placement.
- if (num_output_channels_ < num_input_channels_) {
- capture_audio_->Mix(num_output_channels_);
- frame->_audioChannel = num_output_channels_;
- }
-
- bool data_changed = stream_data_changed();
- if (analysis_needed(data_changed)) {
- for (int i = 0; i < num_output_channels_; i++) {
- // Split into a low and high band.
- SplittingFilterAnalysis(capture_audio_->data(i),
- capture_audio_->low_pass_split_data(i),
- capture_audio_->high_pass_split_data(i),
- capture_audio_->analysis_filter_state1(i),
- capture_audio_->analysis_filter_state2(i));
- }
- }
-
- err = high_pass_filter_->ProcessCaptureAudio(capture_audio_);
- if (err != kNoError) {
- return err;
- }
-
- err = gain_control_->AnalyzeCaptureAudio(capture_audio_);
- if (err != kNoError) {
- return err;
- }
-
- err = echo_cancellation_->ProcessCaptureAudio(capture_audio_);
- if (err != kNoError) {
- return err;
- }
-
- if (echo_control_mobile_->is_enabled() &&
- noise_suppression_->is_enabled()) {
- capture_audio_->CopyLowPassToReference();
- }
-
- err = noise_suppression_->ProcessCaptureAudio(capture_audio_);
- if (err != kNoError) {
- return err;
- }
-
- err = echo_control_mobile_->ProcessCaptureAudio(capture_audio_);
- if (err != kNoError) {
- return err;
- }
-
- err = voice_detection_->ProcessCaptureAudio(capture_audio_);
- if (err != kNoError) {
- return err;
- }
-
- err = gain_control_->ProcessCaptureAudio(capture_audio_);
- if (err != kNoError) {
- return err;
- }
-
- if (synthesis_needed(data_changed)) {
- for (int i = 0; i < num_output_channels_; i++) {
- // Recombine low and high bands.
- SplittingFilterSynthesis(capture_audio_->low_pass_split_data(i),
- capture_audio_->high_pass_split_data(i),
- capture_audio_->data(i),
- capture_audio_->synthesis_filter_state1(i),
- capture_audio_->synthesis_filter_state2(i));
- }
- }
-
- // The level estimator operates on the recombined data.
- err = level_estimator_->ProcessStream(capture_audio_);
- if (err != kNoError) {
- return err;
- }
-
- capture_audio_->InterleaveTo(frame, data_changed);
-
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
- if (debug_file_->Open()) {
- audioproc::Stream* msg = event_msg_->mutable_stream();
- const size_t data_size = sizeof(int16_t) *
- frame->_payloadDataLengthInSamples *
- frame->_audioChannel;
- msg->set_output_data(frame->_payloadData, data_size);
- err = WriteMessageToDebugFile();
- if (err != kNoError) {
- return err;
- }
- }
-#endif
-
- was_stream_delay_set_ = false;
- return kNoError;
-}
-
-int AudioProcessingImpl::AnalyzeReverseStream(AudioFrame* frame) {
- CriticalSectionScoped crit_scoped(crit_);
- int err = kNoError;
-
- if (frame == NULL) {
- return kNullPointerError;
- }
-
- if (frame->_frequencyInHz != sample_rate_hz_) {
- return kBadSampleRateError;
- }
-
- if (frame->_audioChannel != num_reverse_channels_) {
- return kBadNumberChannelsError;
- }
-
- if (frame->_payloadDataLengthInSamples != samples_per_channel_) {
- return kBadDataLengthError;
- }
-
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
- if (debug_file_->Open()) {
- event_msg_->set_type(audioproc::Event::REVERSE_STREAM);
- audioproc::ReverseStream* msg = event_msg_->mutable_reverse_stream();
- const size_t data_size = sizeof(int16_t) *
- frame->_payloadDataLengthInSamples *
- frame->_audioChannel;
- msg->set_data(frame->_payloadData, data_size);
- err = WriteMessageToDebugFile();
- if (err != kNoError) {
- return err;
- }
- }
-#endif
-
- render_audio_->DeinterleaveFrom(frame);
-
- // TODO(ajm): turn the splitting filter into a component?
- if (sample_rate_hz_ == kSampleRate32kHz) {
- for (int i = 0; i < num_reverse_channels_; i++) {
- // Split into low and high band.
- SplittingFilterAnalysis(render_audio_->data(i),
- render_audio_->low_pass_split_data(i),
- render_audio_->high_pass_split_data(i),
- render_audio_->analysis_filter_state1(i),
- render_audio_->analysis_filter_state2(i));
- }
- }
-
- // TODO(ajm): warnings possible from components?
- err = echo_cancellation_->ProcessRenderAudio(render_audio_);
- if (err != kNoError) {
- return err;
- }
-
- err = echo_control_mobile_->ProcessRenderAudio(render_audio_);
- if (err != kNoError) {
- return err;
- }
-
- err = gain_control_->ProcessRenderAudio(render_audio_);
- if (err != kNoError) {
- return err;
- }
-
- return err; // TODO(ajm): this is for returning warnings; necessary?
-}
-
-int AudioProcessingImpl::set_stream_delay_ms(int delay) {
- was_stream_delay_set_ = true;
- if (delay < 0) {
- return kBadParameterError;
- }
-
- // TODO(ajm): the max is rather arbitrarily chosen; investigate.
- if (delay > 500) {
- stream_delay_ms_ = 500;
- return kBadStreamParameterWarning;
- }
-
- stream_delay_ms_ = delay;
- return kNoError;
-}
-
-int AudioProcessingImpl::stream_delay_ms() const {
- return stream_delay_ms_;
-}
-
-bool AudioProcessingImpl::was_stream_delay_set() const {
- return was_stream_delay_set_;
-}
-
-int AudioProcessingImpl::StartDebugRecording(
- const char filename[AudioProcessing::kMaxFilenameSize]) {
- CriticalSectionScoped crit_scoped(crit_);
- assert(kMaxFilenameSize == FileWrapper::kMaxFileNameSize);
-
- if (filename == NULL) {
- return kNullPointerError;
- }
-
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
- // Stop any ongoing recording.
- if (debug_file_->Open()) {
- if (debug_file_->CloseFile() == -1) {
- return kFileError;
- }
- }
-
- if (debug_file_->OpenFile(filename, false) == -1) {
- debug_file_->CloseFile();
- return kFileError;
- }
-
- int err = WriteInitMessage();
- if (err != kNoError) {
- return err;
- }
- return kNoError;
-#else
- return kUnsupportedFunctionError;
-#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
-}
-
-int AudioProcessingImpl::StopDebugRecording() {
- CriticalSectionScoped crit_scoped(crit_);
-
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
- // We just return if recording hasn't started.
- if (debug_file_->Open()) {
- if (debug_file_->CloseFile() == -1) {
- return kFileError;
- }
- }
- return kNoError;
-#else
- return kUnsupportedFunctionError;
-#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
-}
-
-EchoCancellation* AudioProcessingImpl::echo_cancellation() const {
- return echo_cancellation_;
-}
-
-EchoControlMobile* AudioProcessingImpl::echo_control_mobile() const {
- return echo_control_mobile_;
-}
-
-GainControl* AudioProcessingImpl::gain_control() const {
- return gain_control_;
-}
-
-HighPassFilter* AudioProcessingImpl::high_pass_filter() const {
- return high_pass_filter_;
-}
-
-LevelEstimator* AudioProcessingImpl::level_estimator() const {
- return level_estimator_;
-}
-
-NoiseSuppression* AudioProcessingImpl::noise_suppression() const {
- return noise_suppression_;
-}
-
-VoiceDetection* AudioProcessingImpl::voice_detection() const {
- return voice_detection_;
-}
-
-WebRtc_Word32 AudioProcessingImpl::ChangeUniqueId(const WebRtc_Word32 id) {
- CriticalSectionScoped crit_scoped(crit_);
- id_ = id;
-
- return kNoError;
-}
-
-bool AudioProcessingImpl::stream_data_changed() const {
- int enabled_count = 0;
- std::list<ProcessingComponent*>::const_iterator it;
- for (it = component_list_.begin(); it != component_list_.end(); it++) {
- if ((*it)->is_component_enabled()) {
- enabled_count++;
- }
- }
-
- // Data is unchanged if no components are enabled, or if only level_estimator_
- // or voice_detection_ is enabled.
- if (enabled_count == 0) {
- return false;
- } else if (enabled_count == 1) {
- if (level_estimator_->is_enabled() || voice_detection_->is_enabled()) {
- return false;
- }
- } else if (enabled_count == 2) {
- if (level_estimator_->is_enabled() && voice_detection_->is_enabled()) {
- return false;
- }
- }
- return true;
-}
-
-bool AudioProcessingImpl::synthesis_needed(bool stream_data_changed) const {
- return (stream_data_changed && sample_rate_hz_ == kSampleRate32kHz);
-}
-
-bool AudioProcessingImpl::analysis_needed(bool stream_data_changed) const {
- if (!stream_data_changed && !voice_detection_->is_enabled()) {
- // Only level_estimator_ is enabled.
- return false;
- } else if (sample_rate_hz_ == kSampleRate32kHz) {
- // Something besides level_estimator_ is enabled, and we have super-wb.
- return true;
- }
- return false;
-}
-
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-int AudioProcessingImpl::WriteMessageToDebugFile() {
- int32_t size = event_msg_->ByteSize();
- if (size <= 0) {
- return kUnspecifiedError;
- }
-#if defined(WEBRTC_BIG_ENDIAN)
- // TODO(ajm): Use little-endian "on the wire". For the moment, we can be
- // pretty safe in assuming little-endian.
-#endif
-
- if (!event_msg_->SerializeToString(&event_str_)) {
- return kUnspecifiedError;
- }
-
- // Write message preceded by its size.
- if (!debug_file_->Write(&size, sizeof(int32_t))) {
- return kFileError;
- }
- if (!debug_file_->Write(event_str_.data(), event_str_.length())) {
- return kFileError;
- }
-
- event_msg_->Clear();
-
- return 0;
-}
-
-int AudioProcessingImpl::WriteInitMessage() {
- event_msg_->set_type(audioproc::Event::INIT);
- audioproc::Init* msg = event_msg_->mutable_init();
- msg->set_sample_rate(sample_rate_hz_);
- msg->set_device_sample_rate(echo_cancellation_->device_sample_rate_hz());
- msg->set_num_input_channels(num_input_channels_);
- msg->set_num_output_channels(num_output_channels_);
- msg->set_num_reverse_channels(num_reverse_channels_);
-
- int err = WriteMessageToDebugFile();
- if (err != kNoError) {
- return err;
- }
-
- return kNoError;
-}
-#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/audio_processing_impl.h b/trunk/src/modules/audio_processing/audio_processing_impl.h
deleted file mode 100644
index c1ab476..0000000
--- a/trunk/src/modules/audio_processing/audio_processing_impl.h
+++ /dev/null
@@ -1,126 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
-
-#include "audio_processing.h"
-
-#include <list>
-#include <string>
-
-#include "scoped_ptr.h"
-
-namespace webrtc {
-class AudioBuffer;
-class CriticalSectionWrapper;
-class EchoCancellationImpl;
-class EchoControlMobileImpl;
-class FileWrapper;
-class GainControlImpl;
-class HighPassFilterImpl;
-class LevelEstimatorImpl;
-class NoiseSuppressionImpl;
-class ProcessingComponent;
-class VoiceDetectionImpl;
-
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
-namespace audioproc {
-
-class Event;
-
-} // namespace audioproc
-#endif
-
-class AudioProcessingImpl : public AudioProcessing {
- public:
- enum {
- kSampleRate8kHz = 8000,
- kSampleRate16kHz = 16000,
- kSampleRate32kHz = 32000
- };
-
- explicit AudioProcessingImpl(int id);
- virtual ~AudioProcessingImpl();
-
- CriticalSectionWrapper* crit() const;
-
- int split_sample_rate_hz() const;
- bool was_stream_delay_set() const;
-
- // AudioProcessing methods.
- virtual int Initialize();
- virtual int InitializeLocked();
- virtual int set_sample_rate_hz(int rate);
- virtual int sample_rate_hz() const;
- virtual int set_num_channels(int input_channels, int output_channels);
- virtual int num_input_channels() const;
- virtual int num_output_channels() const;
- virtual int set_num_reverse_channels(int channels);
- virtual int num_reverse_channels() const;
- virtual int ProcessStream(AudioFrame* frame);
- virtual int AnalyzeReverseStream(AudioFrame* frame);
- virtual int set_stream_delay_ms(int delay);
- virtual int stream_delay_ms() const;
- virtual int StartDebugRecording(const char filename[kMaxFilenameSize]);
- virtual int StopDebugRecording();
- virtual EchoCancellation* echo_cancellation() const;
- virtual EchoControlMobile* echo_control_mobile() const;
- virtual GainControl* gain_control() const;
- virtual HighPassFilter* high_pass_filter() const;
- virtual LevelEstimator* level_estimator() const;
- virtual NoiseSuppression* noise_suppression() const;
- virtual VoiceDetection* voice_detection() const;
-
- // Module methods.
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- private:
- bool stream_data_changed() const;
- bool synthesis_needed(bool stream_data_changed) const;
- bool analysis_needed(bool stream_data_changed) const;
-
- int id_;
-
- EchoCancellationImpl* echo_cancellation_;
- EchoControlMobileImpl* echo_control_mobile_;
- GainControlImpl* gain_control_;
- HighPassFilterImpl* high_pass_filter_;
- LevelEstimatorImpl* level_estimator_;
- NoiseSuppressionImpl* noise_suppression_;
- VoiceDetectionImpl* voice_detection_;
-
- std::list<ProcessingComponent*> component_list_;
- CriticalSectionWrapper* crit_;
- AudioBuffer* render_audio_;
- AudioBuffer* capture_audio_;
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
- // TODO(andrew): make this more graceful. Ideally we would split this stuff
- // out into a separate class with an "enabled" and "disabled" implementation.
- int WriteMessageToDebugFile();
- int WriteInitMessage();
- scoped_ptr<FileWrapper> debug_file_;
- scoped_ptr<audioproc::Event> event_msg_; // Protobuf message.
- std::string event_str_; // Memory for protobuf serialization.
-#endif
-
- int sample_rate_hz_;
- int split_sample_rate_hz_;
- int samples_per_channel_;
- int stream_delay_ms_;
- bool was_stream_delay_set_;
-
- int num_reverse_channels_;
- int num_input_channels_;
- int num_output_channels_;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_AUDIO_PROCESSING_IMPL_H_
diff --git a/trunk/src/modules/audio_processing/debug.proto b/trunk/src/modules/audio_processing/debug.proto
deleted file mode 100644
index 4b3a163..0000000
--- a/trunk/src/modules/audio_processing/debug.proto
+++ /dev/null
@@ -1,37 +0,0 @@
-syntax = "proto2";
-option optimize_for = LITE_RUNTIME;
-package webrtc.audioproc;
-
-message Init {
- optional int32 sample_rate = 1;
- optional int32 device_sample_rate = 2;
- optional int32 num_input_channels = 3;
- optional int32 num_output_channels = 4;
- optional int32 num_reverse_channels = 5;
-}
-
-message ReverseStream {
- optional bytes data = 1;
-}
-
-message Stream {
- optional bytes input_data = 1;
- optional bytes output_data = 2;
- optional int32 delay = 3;
- optional sint32 drift = 4;
- optional int32 level = 5;
-}
-
-message Event {
- enum Type {
- INIT = 0;
- REVERSE_STREAM = 1;
- STREAM = 2;
- }
-
- required Type type = 1;
-
- optional Init init = 2;
- optional ReverseStream reverse_stream = 3;
- optional Stream stream = 4;
-}
diff --git a/trunk/src/modules/audio_processing/echo_cancellation_impl.cc b/trunk/src/modules/audio_processing/echo_cancellation_impl.cc
deleted file mode 100644
index d4c5523..0000000
--- a/trunk/src/modules/audio_processing/echo_cancellation_impl.cc
+++ /dev/null
@@ -1,371 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "echo_cancellation_impl.h"
-
-#include <cassert>
-#include <string.h>
-
-#include "critical_section_wrapper.h"
-#include "echo_cancellation.h"
-
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
-
-namespace webrtc {
-
-typedef void Handle;
-
-namespace {
-WebRtc_Word16 MapSetting(EchoCancellation::SuppressionLevel level) {
- switch (level) {
- case EchoCancellation::kLowSuppression:
- return kAecNlpConservative;
- case EchoCancellation::kModerateSuppression:
- return kAecNlpModerate;
- case EchoCancellation::kHighSuppression:
- return kAecNlpAggressive;
- }
- assert(false);
- return -1;
-}
-
-AudioProcessing::Error MapError(int err) {
- switch (err) {
- case AEC_UNSUPPORTED_FUNCTION_ERROR:
- return AudioProcessing::kUnsupportedFunctionError;
- case AEC_BAD_PARAMETER_ERROR:
- return AudioProcessing::kBadParameterError;
- case AEC_BAD_PARAMETER_WARNING:
- return AudioProcessing::kBadStreamParameterWarning;
- default:
- // AEC_UNSPECIFIED_ERROR
- // AEC_UNINITIALIZED_ERROR
- // AEC_NULL_POINTER_ERROR
- return AudioProcessing::kUnspecifiedError;
- }
-}
-} // namespace
-
-EchoCancellationImpl::EchoCancellationImpl(const AudioProcessingImpl* apm)
- : ProcessingComponent(apm),
- apm_(apm),
- drift_compensation_enabled_(false),
- metrics_enabled_(false),
- suppression_level_(kModerateSuppression),
- device_sample_rate_hz_(48000),
- stream_drift_samples_(0),
- was_stream_drift_set_(false),
- stream_has_echo_(false),
- delay_logging_enabled_(false) {}
-
-EchoCancellationImpl::~EchoCancellationImpl() {}
-
-int EchoCancellationImpl::ProcessRenderAudio(const AudioBuffer* audio) {
- if (!is_component_enabled()) {
- return apm_->kNoError;
- }
-
- assert(audio->samples_per_split_channel() <= 160);
- assert(audio->num_channels() == apm_->num_reverse_channels());
-
- int err = apm_->kNoError;
-
- // The ordering convention must be followed to pass to the correct AEC.
- size_t handle_index = 0;
- for (int i = 0; i < apm_->num_output_channels(); i++) {
- for (int j = 0; j < audio->num_channels(); j++) {
- Handle* my_handle = static_cast<Handle*>(handle(handle_index));
- err = WebRtcAec_BufferFarend(
- my_handle,
- audio->low_pass_split_data(j),
- static_cast<WebRtc_Word16>(audio->samples_per_split_channel()));
-
- if (err != apm_->kNoError) {
- return GetHandleError(my_handle); // TODO(ajm): warning possible?
- }
-
- handle_index++;
- }
- }
-
- return apm_->kNoError;
-}
-
-int EchoCancellationImpl::ProcessCaptureAudio(AudioBuffer* audio) {
- if (!is_component_enabled()) {
- return apm_->kNoError;
- }
-
- if (!apm_->was_stream_delay_set()) {
- return apm_->kStreamParameterNotSetError;
- }
-
- if (drift_compensation_enabled_ && !was_stream_drift_set_) {
- return apm_->kStreamParameterNotSetError;
- }
-
- assert(audio->samples_per_split_channel() <= 160);
- assert(audio->num_channels() == apm_->num_output_channels());
-
- int err = apm_->kNoError;
-
- // The ordering convention must be followed to pass to the correct AEC.
- size_t handle_index = 0;
- stream_has_echo_ = false;
- for (int i = 0; i < audio->num_channels(); i++) {
- for (int j = 0; j < apm_->num_reverse_channels(); j++) {
- Handle* my_handle = handle(handle_index);
- err = WebRtcAec_Process(
- my_handle,
- audio->low_pass_split_data(i),
- audio->high_pass_split_data(i),
- audio->low_pass_split_data(i),
- audio->high_pass_split_data(i),
- static_cast<WebRtc_Word16>(audio->samples_per_split_channel()),
- apm_->stream_delay_ms(),
- stream_drift_samples_);
-
- if (err != apm_->kNoError) {
- err = GetHandleError(my_handle);
- // TODO(ajm): Figure out how to return warnings properly.
- if (err != apm_->kBadStreamParameterWarning) {
- return err;
- }
- }
-
- WebRtc_Word16 status = 0;
- err = WebRtcAec_get_echo_status(my_handle, &status);
- if (err != apm_->kNoError) {
- return GetHandleError(my_handle);
- }
-
- if (status == 1) {
- stream_has_echo_ = true;
- }
-
- handle_index++;
- }
- }
-
- was_stream_drift_set_ = false;
- return apm_->kNoError;
-}
-
-int EchoCancellationImpl::Enable(bool enable) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- // Ensure AEC and AECM are not both enabled.
- if (enable && apm_->echo_control_mobile()->is_enabled()) {
- return apm_->kBadParameterError;
- }
-
- return EnableComponent(enable);
-}
-
-bool EchoCancellationImpl::is_enabled() const {
- return is_component_enabled();
-}
-
-int EchoCancellationImpl::set_suppression_level(SuppressionLevel level) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- if (MapSetting(level) == -1) {
- return apm_->kBadParameterError;
- }
-
- suppression_level_ = level;
- return Configure();
-}
-
-EchoCancellation::SuppressionLevel EchoCancellationImpl::suppression_level()
- const {
- return suppression_level_;
-}
-
-int EchoCancellationImpl::enable_drift_compensation(bool enable) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- drift_compensation_enabled_ = enable;
- return Configure();
-}
-
-bool EchoCancellationImpl::is_drift_compensation_enabled() const {
- return drift_compensation_enabled_;
-}
-
-int EchoCancellationImpl::set_device_sample_rate_hz(int rate) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- if (rate < 8000 || rate > 96000) {
- return apm_->kBadParameterError;
- }
-
- device_sample_rate_hz_ = rate;
- return Initialize();
-}
-
-int EchoCancellationImpl::device_sample_rate_hz() const {
- return device_sample_rate_hz_;
-}
-
-int EchoCancellationImpl::set_stream_drift_samples(int drift) {
- was_stream_drift_set_ = true;
- stream_drift_samples_ = drift;
- return apm_->kNoError;
-}
-
-int EchoCancellationImpl::stream_drift_samples() const {
- return stream_drift_samples_;
-}
-
-int EchoCancellationImpl::enable_metrics(bool enable) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- metrics_enabled_ = enable;
- return Configure();
-}
-
-bool EchoCancellationImpl::are_metrics_enabled() const {
- return metrics_enabled_;
-}
-
-// TODO(ajm): we currently just use the metrics from the first AEC. Think more
-// aboue the best way to extend this to multi-channel.
-int EchoCancellationImpl::GetMetrics(Metrics* metrics) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- if (metrics == NULL) {
- return apm_->kNullPointerError;
- }
-
- if (!is_component_enabled() || !metrics_enabled_) {
- return apm_->kNotEnabledError;
- }
-
- AecMetrics my_metrics;
- memset(&my_metrics, 0, sizeof(my_metrics));
- memset(metrics, 0, sizeof(Metrics));
-
- Handle* my_handle = static_cast<Handle*>(handle(0));
- int err = WebRtcAec_GetMetrics(my_handle, &my_metrics);
- if (err != apm_->kNoError) {
- return GetHandleError(my_handle);
- }
-
- metrics->residual_echo_return_loss.instant = my_metrics.rerl.instant;
- metrics->residual_echo_return_loss.average = my_metrics.rerl.average;
- metrics->residual_echo_return_loss.maximum = my_metrics.rerl.max;
- metrics->residual_echo_return_loss.minimum = my_metrics.rerl.min;
-
- metrics->echo_return_loss.instant = my_metrics.erl.instant;
- metrics->echo_return_loss.average = my_metrics.erl.average;
- metrics->echo_return_loss.maximum = my_metrics.erl.max;
- metrics->echo_return_loss.minimum = my_metrics.erl.min;
-
- metrics->echo_return_loss_enhancement.instant = my_metrics.erle.instant;
- metrics->echo_return_loss_enhancement.average = my_metrics.erle.average;
- metrics->echo_return_loss_enhancement.maximum = my_metrics.erle.max;
- metrics->echo_return_loss_enhancement.minimum = my_metrics.erle.min;
-
- metrics->a_nlp.instant = my_metrics.aNlp.instant;
- metrics->a_nlp.average = my_metrics.aNlp.average;
- metrics->a_nlp.maximum = my_metrics.aNlp.max;
- metrics->a_nlp.minimum = my_metrics.aNlp.min;
-
- return apm_->kNoError;
-}
-
-bool EchoCancellationImpl::stream_has_echo() const {
- return stream_has_echo_;
-}
-
-int EchoCancellationImpl::enable_delay_logging(bool enable) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- delay_logging_enabled_ = enable;
- return Configure();
-}
-
-bool EchoCancellationImpl::is_delay_logging_enabled() const {
- return delay_logging_enabled_;
-}
-
-// TODO(bjornv): How should we handle the multi-channel case?
-int EchoCancellationImpl::GetDelayMetrics(int* median, int* std) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- if (median == NULL) {
- return apm_->kNullPointerError;
- }
- if (std == NULL) {
- return apm_->kNullPointerError;
- }
-
- if (!is_component_enabled() || !delay_logging_enabled_) {
- return apm_->kNotEnabledError;
- }
-
- Handle* my_handle = static_cast<Handle*>(handle(0));
- if (WebRtcAec_GetDelayMetrics(my_handle, median, std) !=
- apm_->kNoError) {
- return GetHandleError(my_handle);
- }
-
- return apm_->kNoError;
-}
-
-int EchoCancellationImpl::Initialize() {
- int err = ProcessingComponent::Initialize();
- if (err != apm_->kNoError || !is_component_enabled()) {
- return err;
- }
-
- was_stream_drift_set_ = false;
-
- return apm_->kNoError;
-}
-
-void* EchoCancellationImpl::CreateHandle() const {
- Handle* handle = NULL;
- if (WebRtcAec_Create(&handle) != apm_->kNoError) {
- handle = NULL;
- } else {
- assert(handle != NULL);
- }
-
- return handle;
-}
-
-int EchoCancellationImpl::DestroyHandle(void* handle) const {
- assert(handle != NULL);
- return WebRtcAec_Free(static_cast<Handle*>(handle));
-}
-
-int EchoCancellationImpl::InitializeHandle(void* handle) const {
- assert(handle != NULL);
- return WebRtcAec_Init(static_cast<Handle*>(handle),
- apm_->sample_rate_hz(),
- device_sample_rate_hz_);
-}
-
-int EchoCancellationImpl::ConfigureHandle(void* handle) const {
- assert(handle != NULL);
- AecConfig config;
- config.metricsMode = metrics_enabled_;
- config.nlpMode = MapSetting(suppression_level_);
- config.skewMode = drift_compensation_enabled_;
- config.delay_logging = delay_logging_enabled_;
-
- return WebRtcAec_set_config(static_cast<Handle*>(handle), config);
-}
-
-int EchoCancellationImpl::num_handles_required() const {
- return apm_->num_output_channels() *
- apm_->num_reverse_channels();
-}
-
-int EchoCancellationImpl::GetHandleError(void* handle) const {
- assert(handle != NULL);
- return MapError(WebRtcAec_get_error_code(static_cast<Handle*>(handle)));
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/echo_cancellation_impl.h b/trunk/src/modules/audio_processing/echo_cancellation_impl.h
deleted file mode 100644
index 3c2198c..0000000
--- a/trunk/src/modules/audio_processing/echo_cancellation_impl.h
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
-
-#include "audio_processing.h"
-#include "processing_component.h"
-
-namespace webrtc {
-class AudioProcessingImpl;
-class AudioBuffer;
-
-class EchoCancellationImpl : public EchoCancellation,
- public ProcessingComponent {
- public:
- explicit EchoCancellationImpl(const AudioProcessingImpl* apm);
- virtual ~EchoCancellationImpl();
-
- int ProcessRenderAudio(const AudioBuffer* audio);
- int ProcessCaptureAudio(AudioBuffer* audio);
-
- // EchoCancellation implementation.
- virtual bool is_enabled() const;
- virtual int device_sample_rate_hz() const;
- virtual int stream_drift_samples() const;
-
- // ProcessingComponent implementation.
- virtual int Initialize();
-
- private:
- // EchoCancellation implementation.
- virtual int Enable(bool enable);
- virtual int enable_drift_compensation(bool enable);
- virtual bool is_drift_compensation_enabled() const;
- virtual int set_device_sample_rate_hz(int rate);
- virtual int set_stream_drift_samples(int drift);
- virtual int set_suppression_level(SuppressionLevel level);
- virtual SuppressionLevel suppression_level() const;
- virtual int enable_metrics(bool enable);
- virtual bool are_metrics_enabled() const;
- virtual bool stream_has_echo() const;
- virtual int GetMetrics(Metrics* metrics);
- virtual int enable_delay_logging(bool enable);
- virtual bool is_delay_logging_enabled() const;
- virtual int GetDelayMetrics(int* median, int* std);
-
- // ProcessingComponent implementation.
- virtual void* CreateHandle() const;
- virtual int InitializeHandle(void* handle) const;
- virtual int ConfigureHandle(void* handle) const;
- virtual int DestroyHandle(void* handle) const;
- virtual int num_handles_required() const;
- virtual int GetHandleError(void* handle) const;
-
- const AudioProcessingImpl* apm_;
- bool drift_compensation_enabled_;
- bool metrics_enabled_;
- SuppressionLevel suppression_level_;
- int device_sample_rate_hz_;
- int stream_drift_samples_;
- bool was_stream_drift_set_;
- bool stream_has_echo_;
- bool delay_logging_enabled_;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CANCELLATION_IMPL_H_
diff --git a/trunk/src/modules/audio_processing/echo_control_mobile_impl.cc b/trunk/src/modules/audio_processing/echo_control_mobile_impl.cc
deleted file mode 100644
index 9427789..0000000
--- a/trunk/src/modules/audio_processing/echo_control_mobile_impl.cc
+++ /dev/null
@@ -1,300 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "echo_control_mobile_impl.h"
-
-#include <cassert>
-#include <cstring>
-
-#include "critical_section_wrapper.h"
-#include "echo_control_mobile.h"
-
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
-
-namespace webrtc {
-
-typedef void Handle;
-
-namespace {
-WebRtc_Word16 MapSetting(EchoControlMobile::RoutingMode mode) {
- switch (mode) {
- case EchoControlMobile::kQuietEarpieceOrHeadset:
- return 0;
- case EchoControlMobile::kEarpiece:
- return 1;
- case EchoControlMobile::kLoudEarpiece:
- return 2;
- case EchoControlMobile::kSpeakerphone:
- return 3;
- case EchoControlMobile::kLoudSpeakerphone:
- return 4;
- }
- assert(false);
- return -1;
-}
-
-AudioProcessing::Error MapError(int err) {
- switch (err) {
- case AECM_UNSUPPORTED_FUNCTION_ERROR:
- return AudioProcessing::kUnsupportedFunctionError;
- case AECM_NULL_POINTER_ERROR:
- return AudioProcessing::kNullPointerError;
- case AECM_BAD_PARAMETER_ERROR:
- return AudioProcessing::kBadParameterError;
- case AECM_BAD_PARAMETER_WARNING:
- return AudioProcessing::kBadStreamParameterWarning;
- default:
- // AECM_UNSPECIFIED_ERROR
- // AECM_UNINITIALIZED_ERROR
- return AudioProcessing::kUnspecifiedError;
- }
-}
-} // namespace
-
-size_t EchoControlMobile::echo_path_size_bytes() {
- return WebRtcAecm_echo_path_size_bytes();
-}
-
-EchoControlMobileImpl::EchoControlMobileImpl(const AudioProcessingImpl* apm)
- : ProcessingComponent(apm),
- apm_(apm),
- routing_mode_(kSpeakerphone),
- comfort_noise_enabled_(true),
- external_echo_path_(NULL) {}
-
-EchoControlMobileImpl::~EchoControlMobileImpl() {
- if (external_echo_path_ != NULL) {
- delete [] external_echo_path_;
- external_echo_path_ = NULL;
- }
-}
-
-int EchoControlMobileImpl::ProcessRenderAudio(const AudioBuffer* audio) {
- if (!is_component_enabled()) {
- return apm_->kNoError;
- }
-
- assert(audio->samples_per_split_channel() <= 160);
- assert(audio->num_channels() == apm_->num_reverse_channels());
-
- int err = apm_->kNoError;
-
- // The ordering convention must be followed to pass to the correct AECM.
- size_t handle_index = 0;
- for (int i = 0; i < apm_->num_output_channels(); i++) {
- for (int j = 0; j < audio->num_channels(); j++) {
- Handle* my_handle = static_cast<Handle*>(handle(handle_index));
- err = WebRtcAecm_BufferFarend(
- my_handle,
- audio->low_pass_split_data(j),
- static_cast<WebRtc_Word16>(audio->samples_per_split_channel()));
-
- if (err != apm_->kNoError) {
- return GetHandleError(my_handle); // TODO(ajm): warning possible?
- }
-
- handle_index++;
- }
- }
-
- return apm_->kNoError;
-}
-
-int EchoControlMobileImpl::ProcessCaptureAudio(AudioBuffer* audio) {
- if (!is_component_enabled()) {
- return apm_->kNoError;
- }
-
- if (!apm_->was_stream_delay_set()) {
- return apm_->kStreamParameterNotSetError;
- }
-
- assert(audio->samples_per_split_channel() <= 160);
- assert(audio->num_channels() == apm_->num_output_channels());
-
- int err = apm_->kNoError;
-
- // The ordering convention must be followed to pass to the correct AECM.
- size_t handle_index = 0;
- for (int i = 0; i < audio->num_channels(); i++) {
- // TODO(ajm): improve how this works, possibly inside AECM.
- // This is kind of hacked up.
- WebRtc_Word16* noisy = audio->low_pass_reference(i);
- WebRtc_Word16* clean = audio->low_pass_split_data(i);
- if (noisy == NULL) {
- noisy = clean;
- clean = NULL;
- }
- for (int j = 0; j < apm_->num_reverse_channels(); j++) {
- Handle* my_handle = static_cast<Handle*>(handle(handle_index));
- err = WebRtcAecm_Process(
- my_handle,
- noisy,
- clean,
- audio->low_pass_split_data(i),
- static_cast<WebRtc_Word16>(audio->samples_per_split_channel()),
- apm_->stream_delay_ms());
-
- if (err != apm_->kNoError) {
- return GetHandleError(my_handle); // TODO(ajm): warning possible?
- }
-
- handle_index++;
- }
- }
-
- return apm_->kNoError;
-}
-
-int EchoControlMobileImpl::Enable(bool enable) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- // Ensure AEC and AECM are not both enabled.
- if (enable && apm_->echo_cancellation()->is_enabled()) {
- return apm_->kBadParameterError;
- }
-
- return EnableComponent(enable);
-}
-
-bool EchoControlMobileImpl::is_enabled() const {
- return is_component_enabled();
-}
-
-int EchoControlMobileImpl::set_routing_mode(RoutingMode mode) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- if (MapSetting(mode) == -1) {
- return apm_->kBadParameterError;
- }
-
- routing_mode_ = mode;
- return Configure();
-}
-
-EchoControlMobile::RoutingMode EchoControlMobileImpl::routing_mode()
- const {
- return routing_mode_;
-}
-
-int EchoControlMobileImpl::enable_comfort_noise(bool enable) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- comfort_noise_enabled_ = enable;
- return Configure();
-}
-
-bool EchoControlMobileImpl::is_comfort_noise_enabled() const {
- return comfort_noise_enabled_;
-}
-
-int EchoControlMobileImpl::SetEchoPath(const void* echo_path,
- size_t size_bytes) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- if (echo_path == NULL) {
- return apm_->kNullPointerError;
- }
- if (size_bytes != echo_path_size_bytes()) {
- // Size mismatch
- return apm_->kBadParameterError;
- }
-
- if (external_echo_path_ == NULL) {
- external_echo_path_ = new unsigned char[size_bytes];
- }
- memcpy(external_echo_path_, echo_path, size_bytes);
-
- return Initialize();
-}
-
-int EchoControlMobileImpl::GetEchoPath(void* echo_path,
- size_t size_bytes) const {
- CriticalSectionScoped crit_scoped(apm_->crit());
- if (echo_path == NULL) {
- return apm_->kNullPointerError;
- }
- if (size_bytes != echo_path_size_bytes()) {
- // Size mismatch
- return apm_->kBadParameterError;
- }
- if (!is_component_enabled()) {
- return apm_->kNotEnabledError;
- }
-
- // Get the echo path from the first channel
- Handle* my_handle = static_cast<Handle*>(handle(0));
- if (WebRtcAecm_GetEchoPath(my_handle, echo_path, size_bytes) != 0) {
- return GetHandleError(my_handle);
- }
-
- return apm_->kNoError;
-}
-
-int EchoControlMobileImpl::Initialize() {
- if (!is_component_enabled()) {
- return apm_->kNoError;
- }
-
- if (apm_->sample_rate_hz() == apm_->kSampleRate32kHz) {
- // AECM doesn't support super-wideband.
- return apm_->kBadSampleRateError;
- }
-
- return ProcessingComponent::Initialize();
-}
-
-void* EchoControlMobileImpl::CreateHandle() const {
- Handle* handle = NULL;
- if (WebRtcAecm_Create(&handle) != apm_->kNoError) {
- handle = NULL;
- } else {
- assert(handle != NULL);
- }
-
- return handle;
-}
-
-int EchoControlMobileImpl::DestroyHandle(void* handle) const {
- return WebRtcAecm_Free(static_cast<Handle*>(handle));
-}
-
-int EchoControlMobileImpl::InitializeHandle(void* handle) const {
- assert(handle != NULL);
- Handle* my_handle = static_cast<Handle*>(handle);
- if (WebRtcAecm_Init(my_handle, apm_->sample_rate_hz()) != 0) {
- return GetHandleError(my_handle);
- }
- if (external_echo_path_ != NULL) {
- if (WebRtcAecm_InitEchoPath(my_handle,
- external_echo_path_,
- echo_path_size_bytes()) != 0) {
- return GetHandleError(my_handle);
- }
- }
-
- return apm_->kNoError;
-}
-
-int EchoControlMobileImpl::ConfigureHandle(void* handle) const {
- AecmConfig config;
- config.cngMode = comfort_noise_enabled_;
- config.echoMode = MapSetting(routing_mode_);
-
- return WebRtcAecm_set_config(static_cast<Handle*>(handle), config);
-}
-
-int EchoControlMobileImpl::num_handles_required() const {
- return apm_->num_output_channels() *
- apm_->num_reverse_channels();
-}
-
-int EchoControlMobileImpl::GetHandleError(void* handle) const {
- assert(handle != NULL);
- return MapError(WebRtcAecm_get_error_code(static_cast<Handle*>(handle)));
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/echo_control_mobile_impl.h b/trunk/src/modules/audio_processing/echo_control_mobile_impl.h
deleted file mode 100644
index 6d9e369..0000000
--- a/trunk/src/modules/audio_processing/echo_control_mobile_impl.h
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
-
-#include "audio_processing.h"
-#include "processing_component.h"
-
-namespace webrtc {
-class AudioProcessingImpl;
-class AudioBuffer;
-
-class EchoControlMobileImpl : public EchoControlMobile,
- public ProcessingComponent {
- public:
- explicit EchoControlMobileImpl(const AudioProcessingImpl* apm);
- virtual ~EchoControlMobileImpl();
-
- int ProcessRenderAudio(const AudioBuffer* audio);
- int ProcessCaptureAudio(AudioBuffer* audio);
-
- // EchoControlMobile implementation.
- virtual bool is_enabled() const;
-
- // ProcessingComponent implementation.
- virtual int Initialize();
-
- private:
- // EchoControlMobile implementation.
- virtual int Enable(bool enable);
- virtual int set_routing_mode(RoutingMode mode);
- virtual RoutingMode routing_mode() const;
- virtual int enable_comfort_noise(bool enable);
- virtual bool is_comfort_noise_enabled() const;
- virtual int SetEchoPath(const void* echo_path, size_t size_bytes);
- virtual int GetEchoPath(void* echo_path, size_t size_bytes) const;
-
- // ProcessingComponent implementation.
- virtual void* CreateHandle() const;
- virtual int InitializeHandle(void* handle) const;
- virtual int ConfigureHandle(void* handle) const;
- virtual int DestroyHandle(void* handle) const;
- virtual int num_handles_required() const;
- virtual int GetHandleError(void* handle) const;
-
- const AudioProcessingImpl* apm_;
- RoutingMode routing_mode_;
- bool comfort_noise_enabled_;
- unsigned char* external_echo_path_;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_ECHO_CONTROL_MOBILE_IMPL_H_
diff --git a/trunk/src/modules/audio_processing/gain_control_impl.cc b/trunk/src/modules/audio_processing/gain_control_impl.cc
deleted file mode 100644
index a518ab5..0000000
--- a/trunk/src/modules/audio_processing/gain_control_impl.cc
+++ /dev/null
@@ -1,367 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "gain_control_impl.h"
-
-#include <cassert>
-
-#include "critical_section_wrapper.h"
-#include "gain_control.h"
-
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
-
-namespace webrtc {
-
-typedef void Handle;
-
-namespace {
-WebRtc_Word16 MapSetting(GainControl::Mode mode) {
- switch (mode) {
- case GainControl::kAdaptiveAnalog:
- return kAgcModeAdaptiveAnalog;
- case GainControl::kAdaptiveDigital:
- return kAgcModeAdaptiveDigital;
- case GainControl::kFixedDigital:
- return kAgcModeFixedDigital;
- }
- assert(false);
- return -1;
-}
-} // namespace
-
-GainControlImpl::GainControlImpl(const AudioProcessingImpl* apm)
- : ProcessingComponent(apm),
- apm_(apm),
- mode_(kAdaptiveAnalog),
- minimum_capture_level_(0),
- maximum_capture_level_(255),
- limiter_enabled_(true),
- target_level_dbfs_(3),
- compression_gain_db_(9),
- analog_capture_level_(0),
- was_analog_level_set_(false),
- stream_is_saturated_(false) {}
-
-GainControlImpl::~GainControlImpl() {}
-
-int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) {
- if (!is_component_enabled()) {
- return apm_->kNoError;
- }
-
- assert(audio->samples_per_split_channel() <= 160);
-
- WebRtc_Word16* mixed_data = audio->low_pass_split_data(0);
- if (audio->num_channels() > 1) {
- audio->CopyAndMixLowPass(1);
- mixed_data = audio->mixed_low_pass_data(0);
- }
-
- for (int i = 0; i < num_handles(); i++) {
- Handle* my_handle = static_cast<Handle*>(handle(i));
- int err = WebRtcAgc_AddFarend(
- my_handle,
- mixed_data,
- static_cast<WebRtc_Word16>(audio->samples_per_split_channel()));
-
- if (err != apm_->kNoError) {
- return GetHandleError(my_handle);
- }
- }
-
- return apm_->kNoError;
-}
-
-int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
- if (!is_component_enabled()) {
- return apm_->kNoError;
- }
-
- assert(audio->samples_per_split_channel() <= 160);
- assert(audio->num_channels() == num_handles());
-
- int err = apm_->kNoError;
-
- if (mode_ == kAdaptiveAnalog) {
- for (int i = 0; i < num_handles(); i++) {
- Handle* my_handle = static_cast<Handle*>(handle(i));
- err = WebRtcAgc_AddMic(
- my_handle,
- audio->low_pass_split_data(i),
- audio->high_pass_split_data(i),
- static_cast<WebRtc_Word16>(audio->samples_per_split_channel()));
-
- if (err != apm_->kNoError) {
- return GetHandleError(my_handle);
- }
- }
- } else if (mode_ == kAdaptiveDigital) {
-
- for (int i = 0; i < num_handles(); i++) {
- Handle* my_handle = static_cast<Handle*>(handle(i));
- WebRtc_Word32 capture_level_out = 0;
-
- err = WebRtcAgc_VirtualMic(
- my_handle,
- audio->low_pass_split_data(i),
- audio->high_pass_split_data(i),
- static_cast<WebRtc_Word16>(audio->samples_per_split_channel()),
- //capture_levels_[i],
- analog_capture_level_,
- &capture_level_out);
-
- capture_levels_[i] = capture_level_out;
-
- if (err != apm_->kNoError) {
- return GetHandleError(my_handle);
- }
-
- }
- }
-
- return apm_->kNoError;
-}
-
-int GainControlImpl::ProcessCaptureAudio(AudioBuffer* audio) {
- if (!is_component_enabled()) {
- return apm_->kNoError;
- }
-
- if (mode_ == kAdaptiveAnalog && !was_analog_level_set_) {
- return apm_->kStreamParameterNotSetError;
- }
-
- assert(audio->samples_per_split_channel() <= 160);
- assert(audio->num_channels() == num_handles());
-
- stream_is_saturated_ = false;
- for (int i = 0; i < num_handles(); i++) {
- Handle* my_handle = static_cast<Handle*>(handle(i));
- WebRtc_Word32 capture_level_out = 0;
- WebRtc_UWord8 saturation_warning = 0;
-
- int err = WebRtcAgc_Process(
- my_handle,
- audio->low_pass_split_data(i),
- audio->high_pass_split_data(i),
- static_cast<WebRtc_Word16>(audio->samples_per_split_channel()),
- audio->low_pass_split_data(i),
- audio->high_pass_split_data(i),
- capture_levels_[i],
- &capture_level_out,
- apm_->echo_cancellation()->stream_has_echo(),
- &saturation_warning);
-
- if (err != apm_->kNoError) {
- return GetHandleError(my_handle);
- }
-
- capture_levels_[i] = capture_level_out;
- if (saturation_warning == 1) {
- stream_is_saturated_ = true;
- }
- }
-
- if (mode_ == kAdaptiveAnalog) {
- // Take the analog level to be the average across the handles.
- analog_capture_level_ = 0;
- for (int i = 0; i < num_handles(); i++) {
- analog_capture_level_ += capture_levels_[i];
- }
-
- analog_capture_level_ /= num_handles();
- }
-
- was_analog_level_set_ = false;
- return apm_->kNoError;
-}
-
-// TODO(ajm): ensure this is called under kAdaptiveAnalog.
-int GainControlImpl::set_stream_analog_level(int level) {
- was_analog_level_set_ = true;
- if (level < minimum_capture_level_ || level > maximum_capture_level_) {
- return apm_->kBadParameterError;
- }
-
- if (mode_ == kAdaptiveAnalog) {
- if (level != analog_capture_level_) {
- // The analog level has been changed; update our internal levels.
- capture_levels_.assign(num_handles(), level);
- }
- }
- analog_capture_level_ = level;
-
- return apm_->kNoError;
-}
-
-int GainControlImpl::stream_analog_level() {
- // TODO(ajm): enable this assertion?
- //assert(mode_ == kAdaptiveAnalog);
-
- return analog_capture_level_;
-}
-
-int GainControlImpl::Enable(bool enable) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- return EnableComponent(enable);
-}
-
-bool GainControlImpl::is_enabled() const {
- return is_component_enabled();
-}
-
-int GainControlImpl::set_mode(Mode mode) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- if (MapSetting(mode) == -1) {
- return apm_->kBadParameterError;
- }
-
- mode_ = mode;
- return Initialize();
-}
-
-GainControl::Mode GainControlImpl::mode() const {
- return mode_;
-}
-
-int GainControlImpl::set_analog_level_limits(int minimum,
- int maximum) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- if (minimum < 0) {
- return apm_->kBadParameterError;
- }
-
- if (maximum > 65535) {
- return apm_->kBadParameterError;
- }
-
- if (maximum < minimum) {
- return apm_->kBadParameterError;
- }
-
- minimum_capture_level_ = minimum;
- maximum_capture_level_ = maximum;
-
- return Initialize();
-}
-
-int GainControlImpl::analog_level_minimum() const {
- return minimum_capture_level_;
-}
-
-int GainControlImpl::analog_level_maximum() const {
- return maximum_capture_level_;
-}
-
-bool GainControlImpl::stream_is_saturated() const {
- return stream_is_saturated_;
-}
-
-int GainControlImpl::set_target_level_dbfs(int level) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- if (level > 31 || level < 0) {
- return apm_->kBadParameterError;
- }
-
- target_level_dbfs_ = level;
- return Configure();
-}
-
-int GainControlImpl::target_level_dbfs() const {
- return target_level_dbfs_;
-}
-
-int GainControlImpl::set_compression_gain_db(int gain) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- if (gain < 0 || gain > 90) {
- return apm_->kBadParameterError;
- }
-
- compression_gain_db_ = gain;
- return Configure();
-}
-
-int GainControlImpl::compression_gain_db() const {
- return compression_gain_db_;
-}
-
-int GainControlImpl::enable_limiter(bool enable) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- limiter_enabled_ = enable;
- return Configure();
-}
-
-bool GainControlImpl::is_limiter_enabled() const {
- return limiter_enabled_;
-}
-
-int GainControlImpl::Initialize() {
- int err = ProcessingComponent::Initialize();
- if (err != apm_->kNoError || !is_component_enabled()) {
- return err;
- }
-
- analog_capture_level_ =
- (maximum_capture_level_ - minimum_capture_level_) >> 1;
- capture_levels_.assign(num_handles(), analog_capture_level_);
- was_analog_level_set_ = false;
-
- return apm_->kNoError;
-}
-
-void* GainControlImpl::CreateHandle() const {
- Handle* handle = NULL;
- if (WebRtcAgc_Create(&handle) != apm_->kNoError) {
- handle = NULL;
- } else {
- assert(handle != NULL);
- }
-
- return handle;
-}
-
-int GainControlImpl::DestroyHandle(void* handle) const {
- return WebRtcAgc_Free(static_cast<Handle*>(handle));
-}
-
-int GainControlImpl::InitializeHandle(void* handle) const {
- return WebRtcAgc_Init(static_cast<Handle*>(handle),
- minimum_capture_level_,
- maximum_capture_level_,
- MapSetting(mode_),
- apm_->sample_rate_hz());
-}
-
-int GainControlImpl::ConfigureHandle(void* handle) const {
- WebRtcAgc_config_t config;
- // TODO(ajm): Flip the sign here (since AGC expects a positive value) if we
- // change the interface.
- //assert(target_level_dbfs_ <= 0);
- //config.targetLevelDbfs = static_cast<WebRtc_Word16>(-target_level_dbfs_);
- config.targetLevelDbfs = static_cast<WebRtc_Word16>(target_level_dbfs_);
- config.compressionGaindB =
- static_cast<WebRtc_Word16>(compression_gain_db_);
- config.limiterEnable = limiter_enabled_;
-
- return WebRtcAgc_set_config(static_cast<Handle*>(handle), config);
-}
-
-int GainControlImpl::num_handles_required() const {
- return apm_->num_output_channels();
-}
-
-int GainControlImpl::GetHandleError(void* handle) const {
- // The AGC has no get_error() function.
- // (Despite listing errors in its interface...)
- assert(handle != NULL);
- return apm_->kUnspecifiedError;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/gain_control_impl.h b/trunk/src/modules/audio_processing/gain_control_impl.h
deleted file mode 100644
index 5915eeb..0000000
--- a/trunk/src/modules/audio_processing/gain_control_impl.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
-
-#include <vector>
-
-#include "audio_processing.h"
-#include "processing_component.h"
-
-namespace webrtc {
-class AudioProcessingImpl;
-class AudioBuffer;
-
-class GainControlImpl : public GainControl,
- public ProcessingComponent {
- public:
- explicit GainControlImpl(const AudioProcessingImpl* apm);
- virtual ~GainControlImpl();
-
- int ProcessRenderAudio(AudioBuffer* audio);
- int AnalyzeCaptureAudio(AudioBuffer* audio);
- int ProcessCaptureAudio(AudioBuffer* audio);
-
- // ProcessingComponent implementation.
- virtual int Initialize();
-
- // GainControl implementation.
- virtual bool is_enabled() const;
- virtual int stream_analog_level();
-
- private:
- // GainControl implementation.
- virtual int Enable(bool enable);
- virtual int set_stream_analog_level(int level);
- virtual int set_mode(Mode mode);
- virtual Mode mode() const;
- virtual int set_target_level_dbfs(int level);
- virtual int target_level_dbfs() const;
- virtual int set_compression_gain_db(int gain);
- virtual int compression_gain_db() const;
- virtual int enable_limiter(bool enable);
- virtual bool is_limiter_enabled() const;
- virtual int set_analog_level_limits(int minimum, int maximum);
- virtual int analog_level_minimum() const;
- virtual int analog_level_maximum() const;
- virtual bool stream_is_saturated() const;
-
- // ProcessingComponent implementation.
- virtual void* CreateHandle() const;
- virtual int InitializeHandle(void* handle) const;
- virtual int ConfigureHandle(void* handle) const;
- virtual int DestroyHandle(void* handle) const;
- virtual int num_handles_required() const;
- virtual int GetHandleError(void* handle) const;
-
- const AudioProcessingImpl* apm_;
- Mode mode_;
- int minimum_capture_level_;
- int maximum_capture_level_;
- bool limiter_enabled_;
- int target_level_dbfs_;
- int compression_gain_db_;
- std::vector<int> capture_levels_;
- int analog_capture_level_;
- bool was_analog_level_set_;
- bool stream_is_saturated_;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_GAIN_CONTROL_IMPL_H_
diff --git a/trunk/src/modules/audio_processing/high_pass_filter_impl.cc b/trunk/src/modules/audio_processing/high_pass_filter_impl.cc
deleted file mode 100644
index b20fed8..0000000
--- a/trunk/src/modules/audio_processing/high_pass_filter_impl.cc
+++ /dev/null
@@ -1,173 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "high_pass_filter_impl.h"
-
-#include <cassert>
-
-#include "critical_section_wrapper.h"
-#include "typedefs.h"
-#include "signal_processing_library.h"
-
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
-
-namespace webrtc {
-namespace {
-const WebRtc_Word16 kFilterCoefficients8kHz[5] =
- {3798, -7596, 3798, 7807, -3733};
-
-const WebRtc_Word16 kFilterCoefficients[5] =
- {4012, -8024, 4012, 8002, -3913};
-
-struct FilterState {
- WebRtc_Word16 y[4];
- WebRtc_Word16 x[2];
- const WebRtc_Word16* ba;
-};
-
-int InitializeFilter(FilterState* hpf, int sample_rate_hz) {
- assert(hpf != NULL);
-
- if (sample_rate_hz == AudioProcessingImpl::kSampleRate8kHz) {
- hpf->ba = kFilterCoefficients8kHz;
- } else {
- hpf->ba = kFilterCoefficients;
- }
-
- WebRtcSpl_MemSetW16(hpf->x, 0, 2);
- WebRtcSpl_MemSetW16(hpf->y, 0, 4);
-
- return AudioProcessing::kNoError;
-}
-
-int Filter(FilterState* hpf, WebRtc_Word16* data, int length) {
- assert(hpf != NULL);
-
- WebRtc_Word32 tmp_int32 = 0;
- WebRtc_Word16* y = hpf->y;
- WebRtc_Word16* x = hpf->x;
- const WebRtc_Word16* ba = hpf->ba;
-
- for (int i = 0; i < length; i++) {
- // y[i] = b[0] * x[i] + b[1] * x[i-1] + b[2] * x[i-2]
- // + -a[1] * y[i-1] + -a[2] * y[i-2];
-
- tmp_int32 =
- WEBRTC_SPL_MUL_16_16(y[1], ba[3]); // -a[1] * y[i-1] (low part)
- tmp_int32 +=
- WEBRTC_SPL_MUL_16_16(y[3], ba[4]); // -a[2] * y[i-2] (low part)
- tmp_int32 = (tmp_int32 >> 15);
- tmp_int32 +=
- WEBRTC_SPL_MUL_16_16(y[0], ba[3]); // -a[1] * y[i-1] (high part)
- tmp_int32 +=
- WEBRTC_SPL_MUL_16_16(y[2], ba[4]); // -a[2] * y[i-2] (high part)
- tmp_int32 = (tmp_int32 << 1);
-
- tmp_int32 += WEBRTC_SPL_MUL_16_16(data[i], ba[0]); // b[0]*x[0]
- tmp_int32 += WEBRTC_SPL_MUL_16_16(x[0], ba[1]); // b[1]*x[i-1]
- tmp_int32 += WEBRTC_SPL_MUL_16_16(x[1], ba[2]); // b[2]*x[i-2]
-
- // Update state (input part)
- x[1] = x[0];
- x[0] = data[i];
-
- // Update state (filtered part)
- y[2] = y[0];
- y[3] = y[1];
- y[0] = static_cast<WebRtc_Word16>(tmp_int32 >> 13);
- y[1] = static_cast<WebRtc_Word16>((tmp_int32 -
- WEBRTC_SPL_LSHIFT_W32(static_cast<WebRtc_Word32>(y[0]), 13)) << 2);
-
- // Rounding in Q12, i.e. add 2^11
- tmp_int32 += 2048;
-
- // Saturate (to 2^27) so that the HP filtered signal does not overflow
- tmp_int32 = WEBRTC_SPL_SAT(static_cast<WebRtc_Word32>(134217727),
- tmp_int32,
- static_cast<WebRtc_Word32>(-134217728));
-
- // Convert back to Q0 and use rounding
- data[i] = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp_int32, 12);
-
- }
-
- return AudioProcessing::kNoError;
-}
-} // namespace
-
-typedef FilterState Handle;
-
-HighPassFilterImpl::HighPassFilterImpl(const AudioProcessingImpl* apm)
- : ProcessingComponent(apm),
- apm_(apm) {}
-
-HighPassFilterImpl::~HighPassFilterImpl() {}
-
-int HighPassFilterImpl::ProcessCaptureAudio(AudioBuffer* audio) {
- int err = apm_->kNoError;
-
- if (!is_component_enabled()) {
- return apm_->kNoError;
- }
-
- assert(audio->samples_per_split_channel() <= 160);
-
- for (int i = 0; i < num_handles(); i++) {
- Handle* my_handle = static_cast<Handle*>(handle(i));
- err = Filter(my_handle,
- audio->low_pass_split_data(i),
- audio->samples_per_split_channel());
-
- if (err != apm_->kNoError) {
- return GetHandleError(my_handle);
- }
- }
-
- return apm_->kNoError;
-}
-
-int HighPassFilterImpl::Enable(bool enable) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- return EnableComponent(enable);
-}
-
-bool HighPassFilterImpl::is_enabled() const {
- return is_component_enabled();
-}
-
-void* HighPassFilterImpl::CreateHandle() const {
- return new FilterState;
-}
-
-int HighPassFilterImpl::DestroyHandle(void* handle) const {
- delete static_cast<Handle*>(handle);
- return apm_->kNoError;
-}
-
-int HighPassFilterImpl::InitializeHandle(void* handle) const {
- return InitializeFilter(static_cast<Handle*>(handle),
- apm_->sample_rate_hz());
-}
-
-int HighPassFilterImpl::ConfigureHandle(void* /*handle*/) const {
- return apm_->kNoError; // Not configurable.
-}
-
-int HighPassFilterImpl::num_handles_required() const {
- return apm_->num_output_channels();
-}
-
-int HighPassFilterImpl::GetHandleError(void* handle) const {
- // The component has no detailed errors.
- assert(handle != NULL);
- return apm_->kUnspecifiedError;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/high_pass_filter_impl.h b/trunk/src/modules/audio_processing/high_pass_filter_impl.h
deleted file mode 100644
index 94a9c89..0000000
--- a/trunk/src/modules/audio_processing/high_pass_filter_impl.h
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_IMPL_H_
-
-#include "audio_processing.h"
-#include "processing_component.h"
-
-namespace webrtc {
-class AudioProcessingImpl;
-class AudioBuffer;
-
-class HighPassFilterImpl : public HighPassFilter,
- public ProcessingComponent {
- public:
- explicit HighPassFilterImpl(const AudioProcessingImpl* apm);
- virtual ~HighPassFilterImpl();
-
- int ProcessCaptureAudio(AudioBuffer* audio);
-
- // HighPassFilter implementation.
- virtual bool is_enabled() const;
-
- private:
- // HighPassFilter implementation.
- virtual int Enable(bool enable);
-
- // ProcessingComponent implementation.
- virtual void* CreateHandle() const;
- virtual int InitializeHandle(void* handle) const;
- virtual int ConfigureHandle(void* handle) const;
- virtual int DestroyHandle(void* handle) const;
- virtual int num_handles_required() const;
- virtual int GetHandleError(void* handle) const;
-
- const AudioProcessingImpl* apm_;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_HIGH_PASS_FILTER_IMPL_H_
diff --git a/trunk/src/modules/audio_processing/include/audio_processing.h b/trunk/src/modules/audio_processing/include/audio_processing.h
deleted file mode 100644
index aefb824..0000000
--- a/trunk/src/modules/audio_processing/include/audio_processing.h
+++ /dev/null
@@ -1,595 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_INTERFACE_AUDIO_PROCESSING_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_INTERFACE_AUDIO_PROCESSING_H_
-
-#include <stddef.h> // size_t
-
-#include "typedefs.h"
-#include "module.h"
-
-namespace webrtc {
-
-class AudioFrame;
-class EchoCancellation;
-class EchoControlMobile;
-class GainControl;
-class HighPassFilter;
-class LevelEstimator;
-class NoiseSuppression;
-class VoiceDetection;
-
-// The Audio Processing Module (APM) provides a collection of voice processing
-// components designed for real-time communications software.
-//
-// APM operates on two audio streams on a frame-by-frame basis. Frames of the
-// primary stream, on which all processing is applied, are passed to
-// |ProcessStream()|. Frames of the reverse direction stream, which are used for
-// analysis by some components, are passed to |AnalyzeReverseStream()|. On the
-// client-side, this will typically be the near-end (capture) and far-end
-// (render) streams, respectively. APM should be placed in the signal chain as
-// close to the audio hardware abstraction layer (HAL) as possible.
-//
-// On the server-side, the reverse stream will normally not be used, with
-// processing occurring on each incoming stream.
-//
-// Component interfaces follow a similar pattern and are accessed through
-// corresponding getters in APM. All components are disabled at create-time,
-// with default settings that are recommended for most situations. New settings
-// can be applied without enabling a component. Enabling a component triggers
-// memory allocation and initialization to allow it to start processing the
-// streams.
-//
-// Thread safety is provided with the following assumptions to reduce locking
-// overhead:
-// 1. The stream getters and setters are called from the same thread as
-// ProcessStream(). More precisely, stream functions are never called
-// concurrently with ProcessStream().
-// 2. Parameter getters are never called concurrently with the corresponding
-// setter.
-//
-// APM accepts only 16-bit linear PCM audio data in frames of 10 ms. Multiple
-// channels should be interleaved.
-//
-// Usage example, omitting error checking:
-// AudioProcessing* apm = AudioProcessing::Create(0);
-// apm->set_sample_rate_hz(32000); // Super-wideband processing.
-//
-// // Mono capture and stereo render.
-// apm->set_num_channels(1, 1);
-// apm->set_num_reverse_channels(2);
-//
-// apm->high_pass_filter()->Enable(true);
-//
-// apm->echo_cancellation()->enable_drift_compensation(false);
-// apm->echo_cancellation()->Enable(true);
-//
-// apm->noise_reduction()->set_level(kHighSuppression);
-// apm->noise_reduction()->Enable(true);
-//
-// apm->gain_control()->set_analog_level_limits(0, 255);
-// apm->gain_control()->set_mode(kAdaptiveAnalog);
-// apm->gain_control()->Enable(true);
-//
-// apm->voice_detection()->Enable(true);
-//
-// // Start a voice call...
-//
-// // ... Render frame arrives bound for the audio HAL ...
-// apm->AnalyzeReverseStream(render_frame);
-//
-// // ... Capture frame arrives from the audio HAL ...
-// // Call required set_stream_ functions.
-// apm->set_stream_delay_ms(delay_ms);
-// apm->gain_control()->set_stream_analog_level(analog_level);
-//
-// apm->ProcessStream(capture_frame);
-//
-// // Call required stream_ functions.
-// analog_level = apm->gain_control()->stream_analog_level();
-// has_voice = apm->stream_has_voice();
-//
-// // Repeate render and capture processing for the duration of the call...
-// // Start a new call...
-// apm->Initialize();
-//
-// // Close the application...
-// AudioProcessing::Destroy(apm);
-// apm = NULL;
-//
-class AudioProcessing : public Module {
- public:
- // Creates a APM instance, with identifier |id|. Use one instance for every
- // primary audio stream requiring processing. On the client-side, this would
- // typically be one instance for the near-end stream, and additional instances
- // for each far-end stream which requires processing. On the server-side,
- // this would typically be one instance for every incoming stream.
- static AudioProcessing* Create(int id);
- virtual ~AudioProcessing() {};
-
- // TODO(andrew): remove this method. We now allow users to delete instances
- // directly, useful for scoped_ptr.
- // Destroys a |apm| instance.
- static void Destroy(AudioProcessing* apm);
-
- // Initializes internal states, while retaining all user settings. This
- // should be called before beginning to process a new audio stream. However,
- // it is not necessary to call before processing the first stream after
- // creation.
- virtual int Initialize() = 0;
-
- // Sets the sample |rate| in Hz for both the primary and reverse audio
- // streams. 8000, 16000 or 32000 Hz are permitted.
- virtual int set_sample_rate_hz(int rate) = 0;
- virtual int sample_rate_hz() const = 0;
-
- // Sets the number of channels for the primary audio stream. Input frames must
- // contain a number of channels given by |input_channels|, while output frames
- // will be returned with number of channels given by |output_channels|.
- virtual int set_num_channels(int input_channels, int output_channels) = 0;
- virtual int num_input_channels() const = 0;
- virtual int num_output_channels() const = 0;
-
- // Sets the number of channels for the reverse audio stream. Input frames must
- // contain a number of channels given by |channels|.
- virtual int set_num_reverse_channels(int channels) = 0;
- virtual int num_reverse_channels() const = 0;
-
- // Processes a 10 ms |frame| of the primary audio stream. On the client-side,
- // this is the near-end (or captured) audio.
- //
- // If needed for enabled functionality, any function with the set_stream_ tag
- // must be called prior to processing the current frame. Any getter function
- // with the stream_ tag which is needed should be called after processing.
- //
- // The |_frequencyInHz|, |_audioChannel|, and |_payloadDataLengthInSamples|
- // members of |frame| must be valid, and correspond to settings supplied
- // to APM.
- virtual int ProcessStream(AudioFrame* frame) = 0;
-
- // Analyzes a 10 ms |frame| of the reverse direction audio stream. The frame
- // will not be modified. On the client-side, this is the far-end (or to be
- // rendered) audio.
- //
- // It is only necessary to provide this if echo processing is enabled, as the
- // reverse stream forms the echo reference signal. It is recommended, but not
- // necessary, to provide if gain control is enabled. On the server-side this
- // typically will not be used. If you're not sure what to pass in here,
- // chances are you don't need to use it.
- //
- // The |_frequencyInHz|, |_audioChannel|, and |_payloadDataLengthInSamples|
- // members of |frame| must be valid.
- //
- // TODO(ajm): add const to input; requires an implementation fix.
- virtual int AnalyzeReverseStream(AudioFrame* frame) = 0;
-
- // This must be called if and only if echo processing is enabled.
- //
- // Sets the |delay| in ms between AnalyzeReverseStream() receiving a far-end
- // frame and ProcessStream() receiving a near-end frame containing the
- // corresponding echo. On the client-side this can be expressed as
- // delay = (t_render - t_analyze) + (t_process - t_capture)
- // where,
- // - t_analyze is the time a frame is passed to AnalyzeReverseStream() and
- // t_render is the time the first sample of the same frame is rendered by
- // the audio hardware.
- // - t_capture is the time the first sample of a frame is captured by the
- // audio hardware and t_pull is the time the same frame is passed to
- // ProcessStream().
- virtual int set_stream_delay_ms(int delay) = 0;
- virtual int stream_delay_ms() const = 0;
-
- // Starts recording debugging information to a file specified by |filename|,
- // a NULL-terminated string. If there is an ongoing recording, the old file
- // will be closed, and recording will continue in the newly specified file.
- // An already existing file will be overwritten without warning.
- static const size_t kMaxFilenameSize = 1024;
- virtual int StartDebugRecording(const char filename[kMaxFilenameSize]) = 0;
-
- // Stops recording debugging information, and closes the file. Recording
- // cannot be resumed in the same file (without overwriting it).
- virtual int StopDebugRecording() = 0;
-
- // These provide access to the component interfaces and should never return
- // NULL. The pointers will be valid for the lifetime of the APM instance.
- // The memory for these objects is entirely managed internally.
- virtual EchoCancellation* echo_cancellation() const = 0;
- virtual EchoControlMobile* echo_control_mobile() const = 0;
- virtual GainControl* gain_control() const = 0;
- virtual HighPassFilter* high_pass_filter() const = 0;
- virtual LevelEstimator* level_estimator() const = 0;
- virtual NoiseSuppression* noise_suppression() const = 0;
- virtual VoiceDetection* voice_detection() const = 0;
-
- struct Statistic {
- int instant; // Instantaneous value.
- int average; // Long-term average.
- int maximum; // Long-term maximum.
- int minimum; // Long-term minimum.
- };
-
- enum Error {
- // Fatal errors.
- kNoError = 0,
- kUnspecifiedError = -1,
- kCreationFailedError = -2,
- kUnsupportedComponentError = -3,
- kUnsupportedFunctionError = -4,
- kNullPointerError = -5,
- kBadParameterError = -6,
- kBadSampleRateError = -7,
- kBadDataLengthError = -8,
- kBadNumberChannelsError = -9,
- kFileError = -10,
- kStreamParameterNotSetError = -11,
- kNotEnabledError = -12,
-
- // Warnings are non-fatal.
- // This results when a set_stream_ parameter is out of range. Processing
- // will continue, but the parameter may have been truncated.
- kBadStreamParameterWarning = -13
- };
-
- // Inherited from Module.
- virtual WebRtc_Word32 TimeUntilNextProcess() { return -1; };
- virtual WebRtc_Word32 Process() { return -1; };
-};
-
-// The acoustic echo cancellation (AEC) component provides better performance
-// than AECM but also requires more processing power and is dependent on delay
-// stability and reporting accuracy. As such it is well-suited and recommended
-// for PC and IP phone applications.
-//
-// Not recommended to be enabled on the server-side.
-class EchoCancellation {
- public:
- // EchoCancellation and EchoControlMobile may not be enabled simultaneously.
- // Enabling one will disable the other.
- virtual int Enable(bool enable) = 0;
- virtual bool is_enabled() const = 0;
-
- // Differences in clock speed on the primary and reverse streams can impact
- // the AEC performance. On the client-side, this could be seen when different
- // render and capture devices are used, particularly with webcams.
- //
- // This enables a compensation mechanism, and requires that
- // |set_device_sample_rate_hz()| and |set_stream_drift_samples()| be called.
- virtual int enable_drift_compensation(bool enable) = 0;
- virtual bool is_drift_compensation_enabled() const = 0;
-
- // Provides the sampling rate of the audio devices. It is assumed the render
- // and capture devices use the same nominal sample rate. Required if and only
- // if drift compensation is enabled.
- virtual int set_device_sample_rate_hz(int rate) = 0;
- virtual int device_sample_rate_hz() const = 0;
-
- // Sets the difference between the number of samples rendered and captured by
- // the audio devices since the last call to |ProcessStream()|. Must be called
- // if and only if drift compensation is enabled, prior to |ProcessStream()|.
- virtual int set_stream_drift_samples(int drift) = 0;
- virtual int stream_drift_samples() const = 0;
-
- enum SuppressionLevel {
- kLowSuppression,
- kModerateSuppression,
- kHighSuppression
- };
-
- // Sets the aggressiveness of the suppressor. A higher level trades off
- // double-talk performance for increased echo suppression.
- virtual int set_suppression_level(SuppressionLevel level) = 0;
- virtual SuppressionLevel suppression_level() const = 0;
-
- // Returns false if the current frame almost certainly contains no echo
- // and true if it _might_ contain echo.
- virtual bool stream_has_echo() const = 0;
-
- // Enables the computation of various echo metrics. These are obtained
- // through |GetMetrics()|.
- virtual int enable_metrics(bool enable) = 0;
- virtual bool are_metrics_enabled() const = 0;
-
- // Each statistic is reported in dB.
- // P_far: Far-end (render) signal power.
- // P_echo: Near-end (capture) echo signal power.
- // P_out: Signal power at the output of the AEC.
- // P_a: Internal signal power at the point before the AEC's non-linear
- // processor.
- struct Metrics {
- // RERL = ERL + ERLE
- AudioProcessing::Statistic residual_echo_return_loss;
-
- // ERL = 10log_10(P_far / P_echo)
- AudioProcessing::Statistic echo_return_loss;
-
- // ERLE = 10log_10(P_echo / P_out)
- AudioProcessing::Statistic echo_return_loss_enhancement;
-
- // (Pre non-linear processing suppression) A_NLP = 10log_10(P_echo / P_a)
- AudioProcessing::Statistic a_nlp;
- };
-
- // TODO(ajm): discuss the metrics update period.
- virtual int GetMetrics(Metrics* metrics) = 0;
-
- // Enables computation and logging of delay values. Statistics are obtained
- // through |GetDelayMetrics()|.
- virtual int enable_delay_logging(bool enable) = 0;
- virtual bool is_delay_logging_enabled() const = 0;
-
- // The delay metrics consists of the delay |median| and the delay standard
- // deviation |std|. The values are averaged over the time period since the
- // last call to |GetDelayMetrics()|.
- virtual int GetDelayMetrics(int* median, int* std) = 0;
-
- protected:
- virtual ~EchoCancellation() {};
-};
-
-// The acoustic echo control for mobile (AECM) component is a low complexity
-// robust option intended for use on mobile devices.
-//
-// Not recommended to be enabled on the server-side.
-class EchoControlMobile {
- public:
- // EchoCancellation and EchoControlMobile may not be enabled simultaneously.
- // Enabling one will disable the other.
- virtual int Enable(bool enable) = 0;
- virtual bool is_enabled() const = 0;
-
- // Recommended settings for particular audio routes. In general, the louder
- // the echo is expected to be, the higher this value should be set. The
- // preferred setting may vary from device to device.
- enum RoutingMode {
- kQuietEarpieceOrHeadset,
- kEarpiece,
- kLoudEarpiece,
- kSpeakerphone,
- kLoudSpeakerphone
- };
-
- // Sets echo control appropriate for the audio routing |mode| on the device.
- // It can and should be updated during a call if the audio routing changes.
- virtual int set_routing_mode(RoutingMode mode) = 0;
- virtual RoutingMode routing_mode() const = 0;
-
- // Comfort noise replaces suppressed background noise to maintain a
- // consistent signal level.
- virtual int enable_comfort_noise(bool enable) = 0;
- virtual bool is_comfort_noise_enabled() const = 0;
-
- // A typical use case is to initialize the component with an echo path from a
- // previous call. The echo path is retrieved using |GetEchoPath()|, typically
- // at the end of a call. The data can then be stored for later use as an
- // initializer before the next call, using |SetEchoPath()|.
- //
- // Controlling the echo path this way requires the data |size_bytes| to match
- // the internal echo path size. This size can be acquired using
- // |echo_path_size_bytes()|. |SetEchoPath()| causes an entire reset, worth
- // noting if it is to be called during an ongoing call.
- //
- // It is possible that version incompatibilities may result in a stored echo
- // path of the incorrect size. In this case, the stored path should be
- // discarded.
- virtual int SetEchoPath(const void* echo_path, size_t size_bytes) = 0;
- virtual int GetEchoPath(void* echo_path, size_t size_bytes) const = 0;
-
- // The returned path size is guaranteed not to change for the lifetime of
- // the application.
- static size_t echo_path_size_bytes();
-
- protected:
- virtual ~EchoControlMobile() {};
-};
-
-// The automatic gain control (AGC) component brings the signal to an
-// appropriate range. This is done by applying a digital gain directly and, in
-// the analog mode, prescribing an analog gain to be applied at the audio HAL.
-//
-// Recommended to be enabled on the client-side.
-class GainControl {
- public:
- virtual int Enable(bool enable) = 0;
- virtual bool is_enabled() const = 0;
-
- // When an analog mode is set, this must be called prior to |ProcessStream()|
- // to pass the current analog level from the audio HAL. Must be within the
- // range provided to |set_analog_level_limits()|.
- virtual int set_stream_analog_level(int level) = 0;
-
- // When an analog mode is set, this should be called after |ProcessStream()|
- // to obtain the recommended new analog level for the audio HAL. It is the
- // users responsibility to apply this level.
- virtual int stream_analog_level() = 0;
-
- enum Mode {
- // Adaptive mode intended for use if an analog volume control is available
- // on the capture device. It will require the user to provide coupling
- // between the OS mixer controls and AGC through the |stream_analog_level()|
- // functions.
- //
- // It consists of an analog gain prescription for the audio device and a
- // digital compression stage.
- kAdaptiveAnalog,
-
- // Adaptive mode intended for situations in which an analog volume control
- // is unavailable. It operates in a similar fashion to the adaptive analog
- // mode, but with scaling instead applied in the digital domain. As with
- // the analog mode, it additionally uses a digital compression stage.
- kAdaptiveDigital,
-
- // Fixed mode which enables only the digital compression stage also used by
- // the two adaptive modes.
- //
- // It is distinguished from the adaptive modes by considering only a
- // short time-window of the input signal. It applies a fixed gain through
- // most of the input level range, and compresses (gradually reduces gain
- // with increasing level) the input signal at higher levels. This mode is
- // preferred on embedded devices where the capture signal level is
- // predictable, so that a known gain can be applied.
- kFixedDigital
- };
-
- virtual int set_mode(Mode mode) = 0;
- virtual Mode mode() const = 0;
-
- // Sets the target peak |level| (or envelope) of the AGC in dBFs (decibels
- // from digital full-scale). The convention is to use positive values. For
- // instance, passing in a value of 3 corresponds to -3 dBFs, or a target
- // level 3 dB below full-scale. Limited to [0, 31].
- //
- // TODO(ajm): use a negative value here instead, if/when VoE will similarly
- // update its interface.
- virtual int set_target_level_dbfs(int level) = 0;
- virtual int target_level_dbfs() const = 0;
-
- // Sets the maximum |gain| the digital compression stage may apply, in dB. A
- // higher number corresponds to greater compression, while a value of 0 will
- // leave the signal uncompressed. Limited to [0, 90].
- virtual int set_compression_gain_db(int gain) = 0;
- virtual int compression_gain_db() const = 0;
-
- // When enabled, the compression stage will hard limit the signal to the
- // target level. Otherwise, the signal will be compressed but not limited
- // above the target level.
- virtual int enable_limiter(bool enable) = 0;
- virtual bool is_limiter_enabled() const = 0;
-
- // Sets the |minimum| and |maximum| analog levels of the audio capture device.
- // Must be set if and only if an analog mode is used. Limited to [0, 65535].
- virtual int set_analog_level_limits(int minimum,
- int maximum) = 0;
- virtual int analog_level_minimum() const = 0;
- virtual int analog_level_maximum() const = 0;
-
- // Returns true if the AGC has detected a saturation event (period where the
- // signal reaches digital full-scale) in the current frame and the analog
- // level cannot be reduced.
- //
- // This could be used as an indicator to reduce or disable analog mic gain at
- // the audio HAL.
- virtual bool stream_is_saturated() const = 0;
-
- protected:
- virtual ~GainControl() {};
-};
-
-// A filtering component which removes DC offset and low-frequency noise.
-// Recommended to be enabled on the client-side.
-class HighPassFilter {
- public:
- virtual int Enable(bool enable) = 0;
- virtual bool is_enabled() const = 0;
-
- protected:
- virtual ~HighPassFilter() {};
-};
-
-// An estimation component used to retrieve level metrics.
-class LevelEstimator {
- public:
- virtual int Enable(bool enable) = 0;
- virtual bool is_enabled() const = 0;
-
- // Returns the root mean square (RMS) level in dBFs (decibels from digital
- // full-scale), or alternately dBov. It is computed over all primary stream
- // frames since the last call to RMS(). The returned value is positive but
- // should be interpreted as negative. It is constrained to [0, 127].
- //
- // The computation follows:
- // http://tools.ietf.org/html/draft-ietf-avtext-client-to-mixer-audio-level-05
- // with the intent that it can provide the RTP audio level indication.
- //
- // Frames passed to ProcessStream() with an |_energy| of zero are considered
- // to have been muted. The RMS of the frame will be interpreted as -127.
- virtual int RMS() = 0;
-
- protected:
- virtual ~LevelEstimator() {};
-};
-
-// The noise suppression (NS) component attempts to remove noise while
-// retaining speech. Recommended to be enabled on the client-side.
-//
-// Recommended to be enabled on the client-side.
-class NoiseSuppression {
- public:
- virtual int Enable(bool enable) = 0;
- virtual bool is_enabled() const = 0;
-
- // Determines the aggressiveness of the suppression. Increasing the level
- // will reduce the noise level at the expense of a higher speech distortion.
- enum Level {
- kLow,
- kModerate,
- kHigh,
- kVeryHigh
- };
-
- virtual int set_level(Level level) = 0;
- virtual Level level() const = 0;
-
- protected:
- virtual ~NoiseSuppression() {};
-};
-
-// The voice activity detection (VAD) component analyzes the stream to
-// determine if voice is present. A facility is also provided to pass in an
-// external VAD decision.
-//
-// In addition to |stream_has_voice()| the VAD decision is provided through the
-// |AudioFrame| passed to |ProcessStream()|. The |_vadActivity| member will be
-// modified to reflect the current decision.
-class VoiceDetection {
- public:
- virtual int Enable(bool enable) = 0;
- virtual bool is_enabled() const = 0;
-
- // Returns true if voice is detected in the current frame. Should be called
- // after |ProcessStream()|.
- virtual bool stream_has_voice() const = 0;
-
- // Some of the APM functionality requires a VAD decision. In the case that
- // a decision is externally available for the current frame, it can be passed
- // in here, before |ProcessStream()| is called.
- //
- // VoiceDetection does _not_ need to be enabled to use this. If it happens to
- // be enabled, detection will be skipped for any frame in which an external
- // VAD decision is provided.
- virtual int set_stream_has_voice(bool has_voice) = 0;
-
- // Specifies the likelihood that a frame will be declared to contain voice.
- // A higher value makes it more likely that speech will not be clipped, at
- // the expense of more noise being detected as voice.
- enum Likelihood {
- kVeryLowLikelihood,
- kLowLikelihood,
- kModerateLikelihood,
- kHighLikelihood
- };
-
- virtual int set_likelihood(Likelihood likelihood) = 0;
- virtual Likelihood likelihood() const = 0;
-
- // Sets the |size| of the frames in ms on which the VAD will operate. Larger
- // frames will improve detection accuracy, but reduce the frequency of
- // updates.
- //
- // This does not impact the size of frames passed to |ProcessStream()|.
- virtual int set_frame_size_ms(int size) = 0;
- virtual int frame_size_ms() const = 0;
-
- protected:
- virtual ~VoiceDetection() {};
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_INTERFACE_AUDIO_PROCESSING_H_
diff --git a/trunk/src/modules/audio_processing/level_estimator_impl.cc b/trunk/src/modules/audio_processing/level_estimator_impl.cc
deleted file mode 100644
index 42cac99..0000000
--- a/trunk/src/modules/audio_processing/level_estimator_impl.cc
+++ /dev/null
@@ -1,165 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "level_estimator_impl.h"
-
-#include <assert.h>
-#include <math.h>
-#include <string.h>
-
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
-#include "critical_section_wrapper.h"
-
-namespace webrtc {
-namespace {
-
-const double kMaxSquaredLevel = 32768.0 * 32768.0;
-
-class Level {
- public:
- static const int kMinLevel = 127;
-
- Level()
- : sum_square_(0.0),
- sample_count_(0) {}
- ~Level() {}
-
- void Init() {
- sum_square_ = 0.0;
- sample_count_ = 0;
- }
-
- void Process(int16_t* data, int length) {
- assert(data != NULL);
- assert(length > 0);
- sum_square_ += SumSquare(data, length);
- sample_count_ += length;
- }
-
- void ProcessMuted(int length) {
- assert(length > 0);
- sample_count_ += length;
- }
-
- int RMS() {
- if (sample_count_ == 0 || sum_square_ == 0.0) {
- Init();
- return kMinLevel;
- }
-
- // Normalize by the max level.
- double rms = sum_square_ / (sample_count_ * kMaxSquaredLevel);
- // 20log_10(x^0.5) = 10log_10(x)
- rms = 10 * log10(rms);
- if (rms > 0)
- rms = 0;
- else if (rms < -kMinLevel)
- rms = -kMinLevel;
-
- rms = -rms;
- Init();
- return static_cast<int>(rms + 0.5);
- }
-
- private:
- static double SumSquare(int16_t* data, int length) {
- double sum_square = 0.0;
- for (int i = 0; i < length; ++i) {
- double data_d = static_cast<double>(data[i]);
- sum_square += data_d * data_d;
- }
- return sum_square;
- }
-
- double sum_square_;
- int sample_count_;
-};
-} // namespace
-
-LevelEstimatorImpl::LevelEstimatorImpl(const AudioProcessingImpl* apm)
- : ProcessingComponent(apm),
- apm_(apm) {}
-
-LevelEstimatorImpl::~LevelEstimatorImpl() {}
-
-int LevelEstimatorImpl::ProcessStream(AudioBuffer* audio) {
- if (!is_component_enabled()) {
- return apm_->kNoError;
- }
-
- Level* level = static_cast<Level*>(handle(0));
- if (audio->is_muted()) {
- level->ProcessMuted(audio->samples_per_channel());
- return apm_->kNoError;
- }
-
- int16_t* mixed_data = audio->data(0);
- if (audio->num_channels() > 1) {
- audio->CopyAndMix(1);
- mixed_data = audio->mixed_data(0);
- }
-
- level->Process(mixed_data, audio->samples_per_channel());
-
- return apm_->kNoError;
-}
-
-int LevelEstimatorImpl::Enable(bool enable) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- return EnableComponent(enable);
-}
-
-bool LevelEstimatorImpl::is_enabled() const {
- return is_component_enabled();
-}
-
-int LevelEstimatorImpl::RMS() {
- if (!is_component_enabled()) {
- return apm_->kNotEnabledError;
- }
-
- Level* level = static_cast<Level*>(handle(0));
- return level->RMS();
-}
-
-void* LevelEstimatorImpl::CreateHandle() const {
- return new Level;
-}
-
-int LevelEstimatorImpl::DestroyHandle(void* handle) const {
- assert(handle != NULL);
- Level* level = static_cast<Level*>(handle);
- delete level;
- return apm_->kNoError;
-}
-
-int LevelEstimatorImpl::InitializeHandle(void* handle) const {
- assert(handle != NULL);
- Level* level = static_cast<Level*>(handle);
- level->Init();
-
- return apm_->kNoError;
-}
-
-int LevelEstimatorImpl::ConfigureHandle(void* /*handle*/) const {
- return apm_->kNoError;
-}
-
-int LevelEstimatorImpl::num_handles_required() const {
- return 1;
-}
-
-int LevelEstimatorImpl::GetHandleError(void* handle) const {
- // The component has no detailed errors.
- assert(handle != NULL);
- return apm_->kUnspecifiedError;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/level_estimator_impl.h b/trunk/src/modules/audio_processing/level_estimator_impl.h
deleted file mode 100644
index 1a06343..0000000
--- a/trunk/src/modules/audio_processing/level_estimator_impl.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
-
-#include "audio_processing.h"
-#include "processing_component.h"
-
-namespace webrtc {
-class AudioProcessingImpl;
-class AudioBuffer;
-
-class LevelEstimatorImpl : public LevelEstimator,
- public ProcessingComponent {
- public:
- explicit LevelEstimatorImpl(const AudioProcessingImpl* apm);
- virtual ~LevelEstimatorImpl();
-
- int ProcessStream(AudioBuffer* audio);
-
- // LevelEstimator implementation.
- virtual bool is_enabled() const;
-
- private:
- // LevelEstimator implementation.
- virtual int Enable(bool enable);
- virtual int RMS();
-
- // ProcessingComponent implementation.
- virtual void* CreateHandle() const;
- virtual int InitializeHandle(void* handle) const;
- virtual int ConfigureHandle(void* handle) const;
- virtual int DestroyHandle(void* handle) const;
- virtual int num_handles_required() const;
- virtual int GetHandleError(void* handle) const;
-
- const AudioProcessingImpl* apm_;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_LEVEL_ESTIMATOR_IMPL_H_
diff --git a/trunk/src/modules/audio_processing/noise_suppression_impl.cc b/trunk/src/modules/audio_processing/noise_suppression_impl.cc
deleted file mode 100644
index c44d3fe..0000000
--- a/trunk/src/modules/audio_processing/noise_suppression_impl.cc
+++ /dev/null
@@ -1,165 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "noise_suppression_impl.h"
-
-#include <cassert>
-
-#include "critical_section_wrapper.h"
-#if defined(WEBRTC_NS_FLOAT)
-#include "noise_suppression.h"
-#elif defined(WEBRTC_NS_FIXED)
-#include "noise_suppression_x.h"
-#endif
-
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
-
-namespace webrtc {
-
-#if defined(WEBRTC_NS_FLOAT)
-typedef NsHandle Handle;
-#elif defined(WEBRTC_NS_FIXED)
-typedef NsxHandle Handle;
-#endif
-
-namespace {
-int MapSetting(NoiseSuppression::Level level) {
- switch (level) {
- case NoiseSuppression::kLow:
- return 0;
- case NoiseSuppression::kModerate:
- return 1;
- case NoiseSuppression::kHigh:
- return 2;
- case NoiseSuppression::kVeryHigh:
- return 3;
- }
- assert(false);
- return -1;
-}
-} // namespace
-
-NoiseSuppressionImpl::NoiseSuppressionImpl(const AudioProcessingImpl* apm)
- : ProcessingComponent(apm),
- apm_(apm),
- level_(kModerate) {}
-
-NoiseSuppressionImpl::~NoiseSuppressionImpl() {}
-
-int NoiseSuppressionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
- int err = apm_->kNoError;
-
- if (!is_component_enabled()) {
- return apm_->kNoError;
- }
- assert(audio->samples_per_split_channel() <= 160);
- assert(audio->num_channels() == num_handles());
-
- for (int i = 0; i < num_handles(); i++) {
- Handle* my_handle = static_cast<Handle*>(handle(i));
-#if defined(WEBRTC_NS_FLOAT)
- err = WebRtcNs_Process(static_cast<Handle*>(handle(i)),
- audio->low_pass_split_data(i),
- audio->high_pass_split_data(i),
- audio->low_pass_split_data(i),
- audio->high_pass_split_data(i));
-#elif defined(WEBRTC_NS_FIXED)
- err = WebRtcNsx_Process(static_cast<Handle*>(handle(i)),
- audio->low_pass_split_data(i),
- audio->high_pass_split_data(i),
- audio->low_pass_split_data(i),
- audio->high_pass_split_data(i));
-#endif
-
- if (err != apm_->kNoError) {
- return GetHandleError(my_handle);
- }
- }
-
- return apm_->kNoError;
-}
-
-int NoiseSuppressionImpl::Enable(bool enable) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- return EnableComponent(enable);
-}
-
-bool NoiseSuppressionImpl::is_enabled() const {
- return is_component_enabled();
-}
-
-int NoiseSuppressionImpl::set_level(Level level) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- if (MapSetting(level) == -1) {
- return apm_->kBadParameterError;
- }
-
- level_ = level;
- return Configure();
-}
-
-NoiseSuppression::Level NoiseSuppressionImpl::level() const {
- return level_;
-}
-
-void* NoiseSuppressionImpl::CreateHandle() const {
- Handle* handle = NULL;
-#if defined(WEBRTC_NS_FLOAT)
- if (WebRtcNs_Create(&handle) != apm_->kNoError)
-#elif defined(WEBRTC_NS_FIXED)
- if (WebRtcNsx_Create(&handle) != apm_->kNoError)
-#endif
- {
- handle = NULL;
- } else {
- assert(handle != NULL);
- }
-
- return handle;
-}
-
-int NoiseSuppressionImpl::DestroyHandle(void* handle) const {
-#if defined(WEBRTC_NS_FLOAT)
- return WebRtcNs_Free(static_cast<Handle*>(handle));
-#elif defined(WEBRTC_NS_FIXED)
- return WebRtcNsx_Free(static_cast<Handle*>(handle));
-#endif
-}
-
-int NoiseSuppressionImpl::InitializeHandle(void* handle) const {
-#if defined(WEBRTC_NS_FLOAT)
- return WebRtcNs_Init(static_cast<Handle*>(handle), apm_->sample_rate_hz());
-#elif defined(WEBRTC_NS_FIXED)
- return WebRtcNsx_Init(static_cast<Handle*>(handle), apm_->sample_rate_hz());
-#endif
-}
-
-int NoiseSuppressionImpl::ConfigureHandle(void* handle) const {
-#if defined(WEBRTC_NS_FLOAT)
- return WebRtcNs_set_policy(static_cast<Handle*>(handle),
- MapSetting(level_));
-#elif defined(WEBRTC_NS_FIXED)
- return WebRtcNsx_set_policy(static_cast<Handle*>(handle),
- MapSetting(level_));
-#endif
-}
-
-int NoiseSuppressionImpl::num_handles_required() const {
- return apm_->num_output_channels();
-}
-
-int NoiseSuppressionImpl::GetHandleError(void* handle) const {
- // The NS has no get_error() function.
- assert(handle != NULL);
- return apm_->kUnspecifiedError;
-}
-} // namespace webrtc
-
diff --git a/trunk/src/modules/audio_processing/noise_suppression_impl.h b/trunk/src/modules/audio_processing/noise_suppression_impl.h
deleted file mode 100644
index 7b65b70..0000000
--- a/trunk/src/modules/audio_processing/noise_suppression_impl.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_
-
-#include "audio_processing.h"
-#include "processing_component.h"
-
-namespace webrtc {
-class AudioProcessingImpl;
-class AudioBuffer;
-
-class NoiseSuppressionImpl : public NoiseSuppression,
- public ProcessingComponent {
- public:
- explicit NoiseSuppressionImpl(const AudioProcessingImpl* apm);
- virtual ~NoiseSuppressionImpl();
-
- int ProcessCaptureAudio(AudioBuffer* audio);
-
- // NoiseSuppression implementation.
- virtual bool is_enabled() const;
-
- private:
- // NoiseSuppression implementation.
- virtual int Enable(bool enable);
- virtual int set_level(Level level);
- virtual Level level() const;
-
- // ProcessingComponent implementation.
- virtual void* CreateHandle() const;
- virtual int InitializeHandle(void* handle) const;
- virtual int ConfigureHandle(void* handle) const;
- virtual int DestroyHandle(void* handle) const;
- virtual int num_handles_required() const;
- virtual int GetHandleError(void* handle) const;
-
- const AudioProcessingImpl* apm_;
- Level level_;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NOISE_SUPPRESSION_IMPL_H_
diff --git a/trunk/src/modules/audio_processing/ns/Android.mk b/trunk/src/modules/audio_processing/ns/Android.mk
deleted file mode 100644
index 5e442f6..0000000
--- a/trunk/src/modules/audio_processing/ns/Android.mk
+++ /dev/null
@@ -1,79 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-#############################
-# Build the non-neon library.
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../android-webrtc.mk
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_ns
-LOCAL_MODULE_TAGS := optional
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := \
- noise_suppression_x.c \
- nsx_core.c
-
-# Files for floating point.
-# noise_suppression.c ns_core.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include \
- $(LOCAL_PATH)/../utility \
- $(LOCAL_PATH)/../../.. \
- $(LOCAL_PATH)/../../../common_audio/signal_processing/include \
- $(LOCAL_PATH)/../../../system_wrappers/interface
-
-LOCAL_STATIC_LIBRARIES += libwebrtc_system_wrappers
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
-
-#############################
-# Build the neon library.
-ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_ns_neon
-LOCAL_MODULE_TAGS := optional
-
-LOCAL_SRC_FILES := nsx_core_neon.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS) \
- -mfpu=neon \
- -mfloat-abi=softfp \
- -flax-vector-conversions
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include \
- $(LOCAL_PATH)/../../.. \
- $(LOCAL_PATH)/../../../common_audio/signal_processing/include
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
-endif # ifeq ($(WEBRTC_BUILD_NEON_LIBS),true)
diff --git a/trunk/src/modules/audio_processing/ns/defines.h b/trunk/src/modules/audio_processing/ns/defines.h
deleted file mode 100644
index d253967..0000000
--- a/trunk/src/modules/audio_processing/ns/defines.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_DEFINES_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_DEFINES_H_
-
-//#define PROCESS_FLOW_0 // Use the traditional method.
-//#define PROCESS_FLOW_1 // Use traditional with DD estimate of prior SNR.
-#define PROCESS_FLOW_2 // Use the new method of speech/noise classification.
-
-#define BLOCKL_MAX 160 // max processing block length: 160
-#define ANAL_BLOCKL_MAX 256 // max analysis block length: 256
-#define HALF_ANAL_BLOCKL 129 // half max analysis block length + 1
-
-#define QUANTILE (float)0.25
-
-#define SIMULT 3
-#define END_STARTUP_LONG 200
-#define END_STARTUP_SHORT 50
-#define FACTOR (float)40.0
-#define WIDTH (float)0.01
-
-#define SMOOTH (float)0.75 // filter smoothing
-// Length of fft work arrays.
-#define IP_LENGTH (ANAL_BLOCKL_MAX >> 1) // must be at least ceil(2 + sqrt(ANAL_BLOCKL_MAX/2))
-#define W_LENGTH (ANAL_BLOCKL_MAX >> 1)
-
-//PARAMETERS FOR NEW METHOD
-#define DD_PR_SNR (float)0.98 // DD update of prior SNR
-#define LRT_TAVG (float)0.50 // tavg parameter for LRT (previously 0.90)
-#define SPECT_FL_TAVG (float)0.30 // tavg parameter for spectral flatness measure
-#define SPECT_DIFF_TAVG (float)0.30 // tavg parameter for spectral difference measure
-#define PRIOR_UPDATE (float)0.10 // update parameter of prior model
-#define NOISE_UPDATE (float)0.90 // update parameter for noise
-#define SPEECH_UPDATE (float)0.99 // update parameter when likely speech
-#define WIDTH_PR_MAP (float)4.0 // width parameter in sigmoid map for prior model
-#define LRT_FEATURE_THR (float)0.5 // default threshold for LRT feature
-#define SF_FEATURE_THR (float)0.5 // default threshold for Spectral Flatness feature
-#define SD_FEATURE_THR (float)0.5 // default threshold for Spectral Difference feature
-#define PROB_RANGE (float)0.20 // probability threshold for noise state in
- // speech/noise likelihood
-#define HIST_PAR_EST 1000 // histogram size for estimation of parameters
-#define GAMMA_PAUSE (float)0.05 // update for conservative noise estimate
-//
-#define B_LIM (float)0.5 // threshold in final energy gain factor calculation
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_DEFINES_H_
diff --git a/trunk/src/modules/audio_processing/ns/include/noise_suppression.h b/trunk/src/modules/audio_processing/ns/include/noise_suppression.h
deleted file mode 100644
index 1f498c1..0000000
--- a/trunk/src/modules/audio_processing/ns/include/noise_suppression.h
+++ /dev/null
@@ -1,110 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_H_
-
-#include "typedefs.h"
-
-typedef struct NsHandleT NsHandle;
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/*
- * This function creates an instance to the noise reduction structure
- *
- * Input:
- * - NS_inst : Pointer to noise reduction instance that should be
- * created
- *
- * Output:
- * - NS_inst : Pointer to created noise reduction instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcNs_Create(NsHandle** NS_inst);
-
-
-/*
- * This function frees the dynamic memory of a specified Noise Reduction
- * instance.
- *
- * Input:
- * - NS_inst : Pointer to NS instance that should be freed
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcNs_Free(NsHandle* NS_inst);
-
-
-/*
- * This function initializes a NS instance
- *
- * Input:
- * - NS_inst : Instance that should be initialized
- * - fs : sampling frequency
- *
- * Output:
- * - NS_inst : Initialized instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcNs_Init(NsHandle* NS_inst, WebRtc_UWord32 fs);
-
-/*
- * This changes the aggressiveness of the noise suppression method.
- *
- * Input:
- * - NS_inst : Instance that should be initialized
- * - mode : 0: Mild, 1: Medium , 2: Aggressive
- *
- * Output:
- * - NS_inst : Initialized instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcNs_set_policy(NsHandle* NS_inst, int mode);
-
-
-/*
- * This functions does Noise Suppression for the inserted speech frame. The
- * input and output signals should always be 10ms (80 or 160 samples).
- *
- * Input
- * - NS_inst : NS Instance. Needs to be initiated before call.
- * - spframe : Pointer to speech frame buffer for L band
- * - spframe_H : Pointer to speech frame buffer for H band
- * - fs : sampling frequency
- *
- * Output:
- * - NS_inst : Updated NS instance
- * - outframe : Pointer to output frame for L band
- * - outframe_H : Pointer to output frame for H band
- *
- * Return value : 0 - OK
- * -1 - Error
- */
-int WebRtcNs_Process(NsHandle* NS_inst,
- short* spframe,
- short* spframe_H,
- short* outframe,
- short* outframe_H);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_H_
diff --git a/trunk/src/modules/audio_processing/ns/include/noise_suppression_x.h b/trunk/src/modules/audio_processing/ns/include/noise_suppression_x.h
deleted file mode 100644
index b6eef90..0000000
--- a/trunk/src/modules/audio_processing/ns/include/noise_suppression_x.h
+++ /dev/null
@@ -1,109 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_X_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_X_H_
-
-#include "typedefs.h"
-
-typedef struct NsxHandleT NsxHandle;
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/*
- * This function creates an instance to the noise reduction structure
- *
- * Input:
- * - nsxInst : Pointer to noise reduction instance that should be
- * created
- *
- * Output:
- * - nsxInst : Pointer to created noise reduction instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcNsx_Create(NsxHandle** nsxInst);
-
-
-/*
- * This function frees the dynamic memory of a specified Noise Suppression
- * instance.
- *
- * Input:
- * - nsxInst : Pointer to NS instance that should be freed
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcNsx_Free(NsxHandle* nsxInst);
-
-
-/*
- * This function initializes a NS instance
- *
- * Input:
- * - nsxInst : Instance that should be initialized
- * - fs : sampling frequency
- *
- * Output:
- * - nsxInst : Initialized instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcNsx_Init(NsxHandle* nsxInst, WebRtc_UWord32 fs);
-
-/*
- * This changes the aggressiveness of the noise suppression method.
- *
- * Input:
- * - nsxInst : Instance that should be initialized
- * - mode : 0: Mild, 1: Medium , 2: Aggressive
- *
- * Output:
- * - nsxInst : Initialized instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcNsx_set_policy(NsxHandle* nsxInst, int mode);
-
-/*
- * This functions does noise suppression for the inserted speech frame. The
- * input and output signals should always be 10ms (80 or 160 samples).
- *
- * Input
- * - nsxInst : NSx instance. Needs to be initiated before call.
- * - speechFrame : Pointer to speech frame buffer for L band
- * - speechFrameHB : Pointer to speech frame buffer for H band
- * - fs : sampling frequency
- *
- * Output:
- * - nsxInst : Updated NSx instance
- * - outFrame : Pointer to output frame for L band
- * - outFrameHB : Pointer to output frame for H band
- *
- * Return value : 0 - OK
- * -1 - Error
- */
-int WebRtcNsx_Process(NsxHandle* nsxInst,
- short* speechFrame,
- short* speechFrameHB,
- short* outFrame,
- short* outFrameHB);
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_INCLUDE_NOISE_SUPPRESSION_X_H_
diff --git a/trunk/src/modules/audio_processing/ns/noise_suppression.c b/trunk/src/modules/audio_processing/ns/noise_suppression.c
deleted file mode 100644
index a1e5ae8..0000000
--- a/trunk/src/modules/audio_processing/ns/noise_suppression.c
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdlib.h>
-#include <string.h>
-
-#include "noise_suppression.h"
-#include "ns_core.h"
-#include "defines.h"
-
-int WebRtcNs_Create(NsHandle** NS_inst) {
- *NS_inst = (NsHandle*) malloc(sizeof(NSinst_t));
- if (*NS_inst != NULL) {
- (*(NSinst_t**)NS_inst)->initFlag = 0;
- return 0;
- } else {
- return -1;
- }
-
-}
-
-int WebRtcNs_Free(NsHandle* NS_inst) {
- free(NS_inst);
- return 0;
-}
-
-
-int WebRtcNs_Init(NsHandle* NS_inst, WebRtc_UWord32 fs) {
- return WebRtcNs_InitCore((NSinst_t*) NS_inst, fs);
-}
-
-int WebRtcNs_set_policy(NsHandle* NS_inst, int mode) {
- return WebRtcNs_set_policy_core((NSinst_t*) NS_inst, mode);
-}
-
-
-int WebRtcNs_Process(NsHandle* NS_inst, short* spframe, short* spframe_H,
- short* outframe, short* outframe_H) {
- return WebRtcNs_ProcessCore(
- (NSinst_t*) NS_inst, spframe, spframe_H, outframe, outframe_H);
-}
diff --git a/trunk/src/modules/audio_processing/ns/noise_suppression_x.c b/trunk/src/modules/audio_processing/ns/noise_suppression_x.c
deleted file mode 100644
index 6d27d0e..0000000
--- a/trunk/src/modules/audio_processing/ns/noise_suppression_x.c
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdlib.h>
-#include <string.h>
-
-#include "noise_suppression_x.h"
-#include "nsx_core.h"
-#include "nsx_defines.h"
-
-int WebRtcNsx_Create(NsxHandle** nsxInst) {
- *nsxInst = (NsxHandle*)malloc(sizeof(NsxInst_t));
- if (*nsxInst != NULL) {
- (*(NsxInst_t**)nsxInst)->initFlag = 0;
- return 0;
- } else {
- return -1;
- }
-
-}
-
-int WebRtcNsx_Free(NsxHandle* nsxInst) {
- free(nsxInst);
- return 0;
-}
-
-int WebRtcNsx_Init(NsxHandle* nsxInst, WebRtc_UWord32 fs) {
- return WebRtcNsx_InitCore((NsxInst_t*)nsxInst, fs);
-}
-
-int WebRtcNsx_set_policy(NsxHandle* nsxInst, int mode) {
- return WebRtcNsx_set_policy_core((NsxInst_t*)nsxInst, mode);
-}
-
-int WebRtcNsx_Process(NsxHandle* nsxInst, short* speechFrame,
- short* speechFrameHB, short* outFrame,
- short* outFrameHB) {
- return WebRtcNsx_ProcessCore(
- (NsxInst_t*)nsxInst, speechFrame, speechFrameHB, outFrame, outFrameHB);
-}
-
diff --git a/trunk/src/modules/audio_processing/ns/ns.gypi b/trunk/src/modules/audio_processing/ns/ns.gypi
deleted file mode 100644
index 147f7ea..0000000
--- a/trunk/src/modules/audio_processing/ns/ns.gypi
+++ /dev/null
@@ -1,59 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'ns',
- 'type': '<(library)',
- 'dependencies': [
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- 'apm_util'
- ],
- 'include_dirs': [
- 'include',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- ],
- },
- 'sources': [
- 'include/noise_suppression.h',
- 'noise_suppression.c',
- 'windows_private.h',
- 'defines.h',
- 'ns_core.c',
- 'ns_core.h',
- ],
- },
- {
- 'target_name': 'ns_fix',
- 'type': '<(library)',
- 'dependencies': [
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- 'include',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- ],
- },
- 'sources': [
- 'include/noise_suppression_x.h',
- 'noise_suppression_x.c',
- 'nsx_defines.h',
- 'nsx_core.c',
- 'nsx_core.h',
- ],
- },
- ],
-}
diff --git a/trunk/src/modules/audio_processing/ns/ns_core.c b/trunk/src/modules/audio_processing/ns/ns_core.c
deleted file mode 100644
index e5aa4ff..0000000
--- a/trunk/src/modules/audio_processing/ns/ns_core.c
+++ /dev/null
@@ -1,1303 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string.h>
-#include <math.h>
-//#include <stdio.h>
-#include <stdlib.h>
-#include "noise_suppression.h"
-#include "ns_core.h"
-#include "windows_private.h"
-#include "fft4g.h"
-#include "signal_processing_library.h"
-
-// Set Feature Extraction Parameters
-void WebRtcNs_set_feature_extraction_parameters(NSinst_t* inst) {
- //bin size of histogram
- inst->featureExtractionParams.binSizeLrt = (float)0.1;
- inst->featureExtractionParams.binSizeSpecFlat = (float)0.05;
- inst->featureExtractionParams.binSizeSpecDiff = (float)0.1;
-
- //range of histogram over which lrt threshold is computed
- inst->featureExtractionParams.rangeAvgHistLrt = (float)1.0;
-
- //scale parameters: multiply dominant peaks of the histograms by scale factor to obtain
- // thresholds for prior model
- inst->featureExtractionParams.factor1ModelPars = (float)1.20; //for lrt and spectral diff
- inst->featureExtractionParams.factor2ModelPars = (float)0.9; //for spectral_flatness:
- // used when noise is flatter than speech
-
- //peak limit for spectral flatness (varies between 0 and 1)
- inst->featureExtractionParams.thresPosSpecFlat = (float)0.6;
-
- //limit on spacing of two highest peaks in histogram: spacing determined by bin size
- inst->featureExtractionParams.limitPeakSpacingSpecFlat =
- 2 * inst->featureExtractionParams.binSizeSpecFlat;
- inst->featureExtractionParams.limitPeakSpacingSpecDiff =
- 2 * inst->featureExtractionParams.binSizeSpecDiff;
-
- //limit on relevance of second peak:
- inst->featureExtractionParams.limitPeakWeightsSpecFlat = (float)0.5;
- inst->featureExtractionParams.limitPeakWeightsSpecDiff = (float)0.5;
-
- // fluctuation limit of lrt feature
- inst->featureExtractionParams.thresFluctLrt = (float)0.05;
-
- //limit on the max and min values for the feature thresholds
- inst->featureExtractionParams.maxLrt = (float)1.0;
- inst->featureExtractionParams.minLrt = (float)0.20;
-
- inst->featureExtractionParams.maxSpecFlat = (float)0.95;
- inst->featureExtractionParams.minSpecFlat = (float)0.10;
-
- inst->featureExtractionParams.maxSpecDiff = (float)1.0;
- inst->featureExtractionParams.minSpecDiff = (float)0.16;
-
- //criteria of weight of histogram peak to accept/reject feature
- inst->featureExtractionParams.thresWeightSpecFlat = (int)(0.3
- * (inst->modelUpdatePars[1])); //for spectral flatness
- inst->featureExtractionParams.thresWeightSpecDiff = (int)(0.3
- * (inst->modelUpdatePars[1])); //for spectral difference
-}
-
-// Initialize state
-int WebRtcNs_InitCore(NSinst_t* inst, WebRtc_UWord32 fs) {
- int i;
- //We only support 10ms frames
-
- //check for valid pointer
- if (inst == NULL) {
- return -1;
- }
-
- // Initialization of struct
- if (fs == 8000 || fs == 16000 || fs == 32000) {
- inst->fs = fs;
- } else {
- return -1;
- }
- inst->windShift = 0;
- if (fs == 8000) {
- // We only support 10ms frames
- inst->blockLen = 80;
- inst->blockLen10ms = 80;
- inst->anaLen = 128;
- inst->window = kBlocks80w128;
- inst->outLen = 0;
- } else if (fs == 16000) {
- // We only support 10ms frames
- inst->blockLen = 160;
- inst->blockLen10ms = 160;
- inst->anaLen = 256;
- inst->window = kBlocks160w256;
- inst->outLen = 0;
- } else if (fs == 32000) {
- // We only support 10ms frames
- inst->blockLen = 160;
- inst->blockLen10ms = 160;
- inst->anaLen = 256;
- inst->window = kBlocks160w256;
- inst->outLen = 0;
- }
- inst->magnLen = inst->anaLen / 2 + 1; // Number of frequency bins
-
- // Initialize fft work arrays.
- inst->ip[0] = 0; // Setting this triggers initialization.
- memset(inst->dataBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
- WebRtc_rdft(inst->anaLen, 1, inst->dataBuf, inst->ip, inst->wfft);
-
- memset(inst->dataBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
- memset(inst->syntBuf, 0, sizeof(float) * ANAL_BLOCKL_MAX);
-
- //for HB processing
- memset(inst->dataBufHB, 0, sizeof(float) * ANAL_BLOCKL_MAX);
-
- //for quantile noise estimation
- memset(inst->quantile, 0, sizeof(float) * HALF_ANAL_BLOCKL);
- for (i = 0; i < SIMULT * HALF_ANAL_BLOCKL; i++) {
- inst->lquantile[i] = (float)8.0;
- inst->density[i] = (float)0.3;
- }
-
- for (i = 0; i < SIMULT; i++) {
- inst->counter[i] = (int)floor((float)(END_STARTUP_LONG * (i + 1)) / (float)SIMULT);
- }
-
- inst->updates = 0;
-
- // Wiener filter initialization
- for (i = 0; i < HALF_ANAL_BLOCKL; i++) {
- inst->smooth[i] = (float)1.0;
- }
-
- // Set the aggressiveness: default
- inst->aggrMode = 0;
-
- //initialize variables for new method
- inst->priorSpeechProb = (float)0.5; //prior prob for speech/noise
- for (i = 0; i < HALF_ANAL_BLOCKL; i++) {
- inst->magnPrev[i] = (float)0.0; //previous mag spectrum
- inst->noisePrev[i] = (float)0.0; //previous noise-spectrum
- inst->logLrtTimeAvg[i] = LRT_FEATURE_THR; //smooth LR ratio (same as threshold)
- inst->magnAvgPause[i] = (float)0.0; //conservative noise spectrum estimate
- inst->speechProbHB[i] = (float)0.0; //for estimation of HB in second pass
- inst->initMagnEst[i] = (float)0.0; //initial average mag spectrum
- }
-
- //feature quantities
- inst->featureData[0] = SF_FEATURE_THR; //spectral flatness (start on threshold)
- inst->featureData[1] = (float)0.0; //spectral entropy: not used in this version
- inst->featureData[2] = (float)0.0; //spectral variance: not used in this version
- inst->featureData[3] = LRT_FEATURE_THR; //average lrt factor (start on threshold)
- inst->featureData[4] = SF_FEATURE_THR; //spectral template diff (start on threshold)
- inst->featureData[5] = (float)0.0; //normalization for spectral-diff
- inst->featureData[6] = (float)0.0; //window time-average of input magnitude spectrum
-
- //histogram quantities: used to estimate/update thresholds for features
- for (i = 0; i < HIST_PAR_EST; i++) {
- inst->histLrt[i] = 0;
- inst->histSpecFlat[i] = 0;
- inst->histSpecDiff[i] = 0;
- }
-
- inst->blockInd = -1; //frame counter
- inst->priorModelPars[0] = LRT_FEATURE_THR; //default threshold for lrt feature
- inst->priorModelPars[1] = (float)0.5; //threshold for spectral flatness:
- // determined on-line
- inst->priorModelPars[2] = (float)1.0; //sgn_map par for spectral measure:
- // 1 for flatness measure
- inst->priorModelPars[3] = (float)0.5; //threshold for template-difference feature:
- // determined on-line
- inst->priorModelPars[4] = (float)1.0; //default weighting parameter for lrt feature
- inst->priorModelPars[5] = (float)0.0; //default weighting parameter for
- // spectral flatness feature
- inst->priorModelPars[6] = (float)0.0; //default weighting parameter for
- // spectral difference feature
-
- inst->modelUpdatePars[0] = 2; //update flag for parameters:
- // 0 no update, 1=update once, 2=update every window
- inst->modelUpdatePars[1] = 500; //window for update
- inst->modelUpdatePars[2] = 0; //counter for update of conservative noise spectrum
- //counter if the feature thresholds are updated during the sequence
- inst->modelUpdatePars[3] = inst->modelUpdatePars[1];
-
- inst->signalEnergy = 0.0;
- inst->sumMagn = 0.0;
- inst->whiteNoiseLevel = 0.0;
- inst->pinkNoiseNumerator = 0.0;
- inst->pinkNoiseExp = 0.0;
-
- WebRtcNs_set_feature_extraction_parameters(inst); // Set feature configuration
-
- //default mode
- WebRtcNs_set_policy_core(inst, 0);
-
-
- memset(inst->outBuf, 0, sizeof(float) * 3 * BLOCKL_MAX);
-
- inst->initFlag = 1;
- return 0;
-}
-
-int WebRtcNs_set_policy_core(NSinst_t* inst, int mode) {
- // allow for modes:0,1,2,3
- if (mode < 0 || mode > 3) {
- return (-1);
- }
-
- inst->aggrMode = mode;
- if (mode == 0) {
- inst->overdrive = (float)1.0;
- inst->denoiseBound = (float)0.5;
- inst->gainmap = 0;
- } else if (mode == 1) {
- //inst->overdrive = (float)1.25;
- inst->overdrive = (float)1.0;
- inst->denoiseBound = (float)0.25;
- inst->gainmap = 1;
- } else if (mode == 2) {
- //inst->overdrive = (float)1.25;
- inst->overdrive = (float)1.1;
- inst->denoiseBound = (float)0.125;
- inst->gainmap = 1;
- } else if (mode == 3) {
- //inst->overdrive = (float)1.30;
- inst->overdrive = (float)1.25;
- inst->denoiseBound = (float)0.09;
- inst->gainmap = 1;
- }
- return 0;
-}
-
-// Estimate noise
-void WebRtcNs_NoiseEstimation(NSinst_t* inst, float* magn, float* noise) {
- int i, s, offset;
- float lmagn[HALF_ANAL_BLOCKL], delta;
-
- if (inst->updates < END_STARTUP_LONG) {
- inst->updates++;
- }
-
- for (i = 0; i < inst->magnLen; i++) {
- lmagn[i] = (float)log(magn[i]);
- }
-
- // loop over simultaneous estimates
- for (s = 0; s < SIMULT; s++) {
- offset = s * inst->magnLen;
-
- // newquantest(...)
- for (i = 0; i < inst->magnLen; i++) {
- // compute delta
- if (inst->density[offset + i] > 1.0) {
- delta = FACTOR * (float)1.0 / inst->density[offset + i];
- } else {
- delta = FACTOR;
- }
-
- // update log quantile estimate
- if (lmagn[i] > inst->lquantile[offset + i]) {
- inst->lquantile[offset + i] += QUANTILE * delta
- / (float)(inst->counter[s] + 1);
- } else {
- inst->lquantile[offset + i] -= ((float)1.0 - QUANTILE) * delta
- / (float)(inst->counter[s] + 1);
- }
-
- // update density estimate
- if (fabs(lmagn[i] - inst->lquantile[offset + i]) < WIDTH) {
- inst->density[offset + i] = ((float)inst->counter[s] * inst->density[offset
- + i] + (float)1.0 / ((float)2.0 * WIDTH)) / (float)(inst->counter[s] + 1);
- }
- } // end loop over magnitude spectrum
-
- if (inst->counter[s] >= END_STARTUP_LONG) {
- inst->counter[s] = 0;
- if (inst->updates >= END_STARTUP_LONG) {
- for (i = 0; i < inst->magnLen; i++) {
- inst->quantile[i] = (float)exp(inst->lquantile[offset + i]);
- }
- }
- }
-
- inst->counter[s]++;
- } // end loop over simultaneous estimates
-
- // Sequentially update the noise during startup
- if (inst->updates < END_STARTUP_LONG) {
- // Use the last "s" to get noise during startup that differ from zero.
- for (i = 0; i < inst->magnLen; i++) {
- inst->quantile[i] = (float)exp(inst->lquantile[offset + i]);
- }
- }
-
- for (i = 0; i < inst->magnLen; i++) {
- noise[i] = inst->quantile[i];
- }
-}
-
-// Extract thresholds for feature parameters
-// histograms are computed over some window_size (given by inst->modelUpdatePars[1])
-// thresholds and weights are extracted every window
-// flag 0 means update histogram only, flag 1 means compute the thresholds/weights
-// threshold and weights are returned in: inst->priorModelPars
-void WebRtcNs_FeatureParameterExtraction(NSinst_t* inst, int flag) {
- int i, useFeatureSpecFlat, useFeatureSpecDiff, numHistLrt;
- int maxPeak1, maxPeak2;
- int weightPeak1SpecFlat, weightPeak2SpecFlat, weightPeak1SpecDiff, weightPeak2SpecDiff;
-
- float binMid, featureSum;
- float posPeak1SpecFlat, posPeak2SpecFlat, posPeak1SpecDiff, posPeak2SpecDiff;
- float fluctLrt, avgHistLrt, avgSquareHistLrt, avgHistLrtCompl;
-
- //3 features: lrt, flatness, difference
- //lrt_feature = inst->featureData[3];
- //flat_feature = inst->featureData[0];
- //diff_feature = inst->featureData[4];
-
- //update histograms
- if (flag == 0) {
- // LRT
- if ((inst->featureData[3] < HIST_PAR_EST * inst->featureExtractionParams.binSizeLrt)
- && (inst->featureData[3] >= 0.0)) {
- i = (int)(inst->featureData[3] / inst->featureExtractionParams.binSizeLrt);
- inst->histLrt[i]++;
- }
- // Spectral flatness
- if ((inst->featureData[0] < HIST_PAR_EST
- * inst->featureExtractionParams.binSizeSpecFlat)
- && (inst->featureData[0] >= 0.0)) {
- i = (int)(inst->featureData[0] / inst->featureExtractionParams.binSizeSpecFlat);
- inst->histSpecFlat[i]++;
- }
- // Spectral difference
- if ((inst->featureData[4] < HIST_PAR_EST
- * inst->featureExtractionParams.binSizeSpecDiff)
- && (inst->featureData[4] >= 0.0)) {
- i = (int)(inst->featureData[4] / inst->featureExtractionParams.binSizeSpecDiff);
- inst->histSpecDiff[i]++;
- }
- }
-
- // extract parameters for speech/noise probability
- if (flag == 1) {
- //lrt feature: compute the average over inst->featureExtractionParams.rangeAvgHistLrt
- avgHistLrt = 0.0;
- avgHistLrtCompl = 0.0;
- avgSquareHistLrt = 0.0;
- numHistLrt = 0;
- for (i = 0; i < HIST_PAR_EST; i++) {
- binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeLrt;
- if (binMid <= inst->featureExtractionParams.rangeAvgHistLrt) {
- avgHistLrt += inst->histLrt[i] * binMid;
- numHistLrt += inst->histLrt[i];
- }
- avgSquareHistLrt += inst->histLrt[i] * binMid * binMid;
- avgHistLrtCompl += inst->histLrt[i] * binMid;
- }
- if (numHistLrt > 0) {
- avgHistLrt = avgHistLrt / ((float)numHistLrt);
- }
- avgHistLrtCompl = avgHistLrtCompl / ((float)inst->modelUpdatePars[1]);
- avgSquareHistLrt = avgSquareHistLrt / ((float)inst->modelUpdatePars[1]);
- fluctLrt = avgSquareHistLrt - avgHistLrt * avgHistLrtCompl;
- // get threshold for lrt feature:
- if (fluctLrt < inst->featureExtractionParams.thresFluctLrt) {
- //very low fluct, so likely noise
- inst->priorModelPars[0] = inst->featureExtractionParams.maxLrt;
- } else {
- inst->priorModelPars[0] = inst->featureExtractionParams.factor1ModelPars
- * avgHistLrt;
- // check if value is within min/max range
- if (inst->priorModelPars[0] < inst->featureExtractionParams.minLrt) {
- inst->priorModelPars[0] = inst->featureExtractionParams.minLrt;
- }
- if (inst->priorModelPars[0] > inst->featureExtractionParams.maxLrt) {
- inst->priorModelPars[0] = inst->featureExtractionParams.maxLrt;
- }
- }
- // done with lrt feature
-
- //
- // for spectral flatness and spectral difference: compute the main peaks of histogram
- maxPeak1 = 0;
- maxPeak2 = 0;
- posPeak1SpecFlat = 0.0;
- posPeak2SpecFlat = 0.0;
- weightPeak1SpecFlat = 0;
- weightPeak2SpecFlat = 0;
-
- // peaks for flatness
- for (i = 0; i < HIST_PAR_EST; i++) {
- binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeSpecFlat;
- if (inst->histSpecFlat[i] > maxPeak1) {
- // Found new "first" peak
- maxPeak2 = maxPeak1;
- weightPeak2SpecFlat = weightPeak1SpecFlat;
- posPeak2SpecFlat = posPeak1SpecFlat;
-
- maxPeak1 = inst->histSpecFlat[i];
- weightPeak1SpecFlat = inst->histSpecFlat[i];
- posPeak1SpecFlat = binMid;
- } else if (inst->histSpecFlat[i] > maxPeak2) {
- // Found new "second" peak
- maxPeak2 = inst->histSpecFlat[i];
- weightPeak2SpecFlat = inst->histSpecFlat[i];
- posPeak2SpecFlat = binMid;
- }
- }
-
- //compute two peaks for spectral difference
- maxPeak1 = 0;
- maxPeak2 = 0;
- posPeak1SpecDiff = 0.0;
- posPeak2SpecDiff = 0.0;
- weightPeak1SpecDiff = 0;
- weightPeak2SpecDiff = 0;
- // peaks for spectral difference
- for (i = 0; i < HIST_PAR_EST; i++) {
- binMid = ((float)i + (float)0.5) * inst->featureExtractionParams.binSizeSpecDiff;
- if (inst->histSpecDiff[i] > maxPeak1) {
- // Found new "first" peak
- maxPeak2 = maxPeak1;
- weightPeak2SpecDiff = weightPeak1SpecDiff;
- posPeak2SpecDiff = posPeak1SpecDiff;
-
- maxPeak1 = inst->histSpecDiff[i];
- weightPeak1SpecDiff = inst->histSpecDiff[i];
- posPeak1SpecDiff = binMid;
- } else if (inst->histSpecDiff[i] > maxPeak2) {
- // Found new "second" peak
- maxPeak2 = inst->histSpecDiff[i];
- weightPeak2SpecDiff = inst->histSpecDiff[i];
- posPeak2SpecDiff = binMid;
- }
- }
-
- // for spectrum flatness feature
- useFeatureSpecFlat = 1;
- // merge the two peaks if they are close
- if ((fabs(posPeak2SpecFlat - posPeak1SpecFlat)
- < inst->featureExtractionParams.limitPeakSpacingSpecFlat)
- && (weightPeak2SpecFlat
- > inst->featureExtractionParams.limitPeakWeightsSpecFlat
- * weightPeak1SpecFlat)) {
- weightPeak1SpecFlat += weightPeak2SpecFlat;
- posPeak1SpecFlat = (float)0.5 * (posPeak1SpecFlat + posPeak2SpecFlat);
- }
- //reject if weight of peaks is not large enough, or peak value too small
- if (weightPeak1SpecFlat < inst->featureExtractionParams.thresWeightSpecFlat
- || posPeak1SpecFlat < inst->featureExtractionParams.thresPosSpecFlat) {
- useFeatureSpecFlat = 0;
- }
- // if selected, get the threshold
- if (useFeatureSpecFlat == 1) {
- // compute the threshold
- inst->priorModelPars[1] = inst->featureExtractionParams.factor2ModelPars
- * posPeak1SpecFlat;
- //check if value is within min/max range
- if (inst->priorModelPars[1] < inst->featureExtractionParams.minSpecFlat) {
- inst->priorModelPars[1] = inst->featureExtractionParams.minSpecFlat;
- }
- if (inst->priorModelPars[1] > inst->featureExtractionParams.maxSpecFlat) {
- inst->priorModelPars[1] = inst->featureExtractionParams.maxSpecFlat;
- }
- }
- // done with flatness feature
-
- // for template feature
- useFeatureSpecDiff = 1;
- // merge the two peaks if they are close
- if ((fabs(posPeak2SpecDiff - posPeak1SpecDiff)
- < inst->featureExtractionParams.limitPeakSpacingSpecDiff)
- && (weightPeak2SpecDiff
- > inst->featureExtractionParams.limitPeakWeightsSpecDiff
- * weightPeak1SpecDiff)) {
- weightPeak1SpecDiff += weightPeak2SpecDiff;
- posPeak1SpecDiff = (float)0.5 * (posPeak1SpecDiff + posPeak2SpecDiff);
- }
- // get the threshold value
- inst->priorModelPars[3] = inst->featureExtractionParams.factor1ModelPars
- * posPeak1SpecDiff;
- //reject if weight of peaks is not large enough
- if (weightPeak1SpecDiff < inst->featureExtractionParams.thresWeightSpecDiff) {
- useFeatureSpecDiff = 0;
- }
- //check if value is within min/max range
- if (inst->priorModelPars[3] < inst->featureExtractionParams.minSpecDiff) {
- inst->priorModelPars[3] = inst->featureExtractionParams.minSpecDiff;
- }
- if (inst->priorModelPars[3] > inst->featureExtractionParams.maxSpecDiff) {
- inst->priorModelPars[3] = inst->featureExtractionParams.maxSpecDiff;
- }
- // done with spectral difference feature
-
- // don't use template feature if fluctuation of lrt feature is very low:
- // most likely just noise state
- if (fluctLrt < inst->featureExtractionParams.thresFluctLrt) {
- useFeatureSpecDiff = 0;
- }
-
- // select the weights between the features
- // inst->priorModelPars[4] is weight for lrt: always selected
- // inst->priorModelPars[5] is weight for spectral flatness
- // inst->priorModelPars[6] is weight for spectral difference
- featureSum = (float)(1 + useFeatureSpecFlat + useFeatureSpecDiff);
- inst->priorModelPars[4] = (float)1.0 / featureSum;
- inst->priorModelPars[5] = ((float)useFeatureSpecFlat) / featureSum;
- inst->priorModelPars[6] = ((float)useFeatureSpecDiff) / featureSum;
-
- // set hists to zero for next update
- if (inst->modelUpdatePars[0] >= 1) {
- for (i = 0; i < HIST_PAR_EST; i++) {
- inst->histLrt[i] = 0;
- inst->histSpecFlat[i] = 0;
- inst->histSpecDiff[i] = 0;
- }
- }
- } // end of flag == 1
-}
-
-// Compute spectral flatness on input spectrum
-// magnIn is the magnitude spectrum
-// spectral flatness is returned in inst->featureData[0]
-void WebRtcNs_ComputeSpectralFlatness(NSinst_t* inst, float* magnIn) {
- int i;
- int shiftLP = 1; //option to remove first bin(s) from spectral measures
- float avgSpectralFlatnessNum, avgSpectralFlatnessDen, spectralTmp;
-
- // comute spectral measures
- // for flatness
- avgSpectralFlatnessNum = 0.0;
- avgSpectralFlatnessDen = inst->sumMagn;
- for (i = 0; i < shiftLP; i++) {
- avgSpectralFlatnessDen -= magnIn[i];
- }
- // compute log of ratio of the geometric to arithmetic mean: check for log(0) case
- for (i = shiftLP; i < inst->magnLen; i++) {
- if (magnIn[i] > 0.0) {
- avgSpectralFlatnessNum += (float)log(magnIn[i]);
- } else {
- inst->featureData[0] -= SPECT_FL_TAVG * inst->featureData[0];
- return;
- }
- }
- //normalize
- avgSpectralFlatnessDen = avgSpectralFlatnessDen / inst->magnLen;
- avgSpectralFlatnessNum = avgSpectralFlatnessNum / inst->magnLen;
-
- //ratio and inverse log: check for case of log(0)
- spectralTmp = (float)exp(avgSpectralFlatnessNum) / avgSpectralFlatnessDen;
-
- //time-avg update of spectral flatness feature
- inst->featureData[0] += SPECT_FL_TAVG * (spectralTmp - inst->featureData[0]);
- // done with flatness feature
-}
-
-// Compute the difference measure between input spectrum and a template/learned noise spectrum
-// magnIn is the input spectrum
-// the reference/template spectrum is inst->magnAvgPause[i]
-// returns (normalized) spectral difference in inst->featureData[4]
-void WebRtcNs_ComputeSpectralDifference(NSinst_t* inst, float* magnIn) {
- // avgDiffNormMagn = var(magnIn) - cov(magnIn, magnAvgPause)^2 / var(magnAvgPause)
- int i;
- float avgPause, avgMagn, covMagnPause, varPause, varMagn, avgDiffNormMagn;
-
- avgPause = 0.0;
- avgMagn = inst->sumMagn;
- // compute average quantities
- for (i = 0; i < inst->magnLen; i++) {
- //conservative smooth noise spectrum from pause frames
- avgPause += inst->magnAvgPause[i];
- }
- avgPause = avgPause / ((float)inst->magnLen);
- avgMagn = avgMagn / ((float)inst->magnLen);
-
- covMagnPause = 0.0;
- varPause = 0.0;
- varMagn = 0.0;
- // compute variance and covariance quantities
- for (i = 0; i < inst->magnLen; i++) {
- covMagnPause += (magnIn[i] - avgMagn) * (inst->magnAvgPause[i] - avgPause);
- varPause += (inst->magnAvgPause[i] - avgPause) * (inst->magnAvgPause[i] - avgPause);
- varMagn += (magnIn[i] - avgMagn) * (magnIn[i] - avgMagn);
- }
- covMagnPause = covMagnPause / ((float)inst->magnLen);
- varPause = varPause / ((float)inst->magnLen);
- varMagn = varMagn / ((float)inst->magnLen);
- // update of average magnitude spectrum
- inst->featureData[6] += inst->signalEnergy;
-
- avgDiffNormMagn = varMagn - (covMagnPause * covMagnPause) / (varPause + (float)0.0001);
- // normalize and compute time-avg update of difference feature
- avgDiffNormMagn = (float)(avgDiffNormMagn / (inst->featureData[5] + (float)0.0001));
- inst->featureData[4] += SPECT_DIFF_TAVG * (avgDiffNormMagn - inst->featureData[4]);
-}
-
-// Compute speech/noise probability
-// speech/noise probability is returned in: probSpeechFinal
-//magn is the input magnitude spectrum
-//noise is the noise spectrum
-//snrLocPrior is the prior snr for each freq.
-//snr loc_post is the post snr for each freq.
-void WebRtcNs_SpeechNoiseProb(NSinst_t* inst, float* probSpeechFinal, float* snrLocPrior,
- float* snrLocPost) {
- int i, sgnMap;
- float invLrt, gainPrior, indPrior;
- float logLrtTimeAvgKsum, besselTmp;
- float indicator0, indicator1, indicator2;
- float tmpFloat1, tmpFloat2;
- float weightIndPrior0, weightIndPrior1, weightIndPrior2;
- float threshPrior0, threshPrior1, threshPrior2;
- float widthPrior, widthPrior0, widthPrior1, widthPrior2;
-
- widthPrior0 = WIDTH_PR_MAP;
- widthPrior1 = (float)2.0 * WIDTH_PR_MAP; //width for pause region:
- // lower range, so increase width in tanh map
- widthPrior2 = (float)2.0 * WIDTH_PR_MAP; //for spectral-difference measure
-
- //threshold parameters for features
- threshPrior0 = inst->priorModelPars[0];
- threshPrior1 = inst->priorModelPars[1];
- threshPrior2 = inst->priorModelPars[3];
-
- //sign for flatness feature
- sgnMap = (int)(inst->priorModelPars[2]);
-
- //weight parameters for features
- weightIndPrior0 = inst->priorModelPars[4];
- weightIndPrior1 = inst->priorModelPars[5];
- weightIndPrior2 = inst->priorModelPars[6];
-
- // compute feature based on average LR factor
- // this is the average over all frequencies of the smooth log lrt
- logLrtTimeAvgKsum = 0.0;
- for (i = 0; i < inst->magnLen; i++) {
- tmpFloat1 = (float)1.0 + (float)2.0 * snrLocPrior[i];
- tmpFloat2 = (float)2.0 * snrLocPrior[i] / (tmpFloat1 + (float)0.0001);
- besselTmp = (snrLocPost[i] + (float)1.0) * tmpFloat2;
- inst->logLrtTimeAvg[i] += LRT_TAVG * (besselTmp - (float)log(tmpFloat1)
- - inst->logLrtTimeAvg[i]);
- logLrtTimeAvgKsum += inst->logLrtTimeAvg[i];
- }
- logLrtTimeAvgKsum = (float)logLrtTimeAvgKsum / (inst->magnLen);
- inst->featureData[3] = logLrtTimeAvgKsum;
- // done with computation of LR factor
-
- //
- //compute the indicator functions
- //
-
- // average lrt feature
- widthPrior = widthPrior0;
- //use larger width in tanh map for pause regions
- if (logLrtTimeAvgKsum < threshPrior0) {
- widthPrior = widthPrior1;
- }
- // compute indicator function: sigmoid map
- indicator0 = (float)0.5 * ((float)tanh(widthPrior *
- (logLrtTimeAvgKsum - threshPrior0)) + (float)1.0);
-
- //spectral flatness feature
- tmpFloat1 = inst->featureData[0];
- widthPrior = widthPrior0;
- //use larger width in tanh map for pause regions
- if (sgnMap == 1 && (tmpFloat1 > threshPrior1)) {
- widthPrior = widthPrior1;
- }
- if (sgnMap == -1 && (tmpFloat1 < threshPrior1)) {
- widthPrior = widthPrior1;
- }
- // compute indicator function: sigmoid map
- indicator1 = (float)0.5 * ((float)tanh((float)sgnMap *
- widthPrior * (threshPrior1 - tmpFloat1)) + (float)1.0);
-
- //for template spectrum-difference
- tmpFloat1 = inst->featureData[4];
- widthPrior = widthPrior0;
- //use larger width in tanh map for pause regions
- if (tmpFloat1 < threshPrior2) {
- widthPrior = widthPrior2;
- }
- // compute indicator function: sigmoid map
- indicator2 = (float)0.5 * ((float)tanh(widthPrior * (tmpFloat1 - threshPrior2))
- + (float)1.0);
-
- //combine the indicator function with the feature weights
- indPrior = weightIndPrior0 * indicator0 + weightIndPrior1 * indicator1 + weightIndPrior2
- * indicator2;
- // done with computing indicator function
-
- //compute the prior probability
- inst->priorSpeechProb += PRIOR_UPDATE * (indPrior - inst->priorSpeechProb);
- // make sure probabilities are within range: keep floor to 0.01
- if (inst->priorSpeechProb > 1.0) {
- inst->priorSpeechProb = (float)1.0;
- }
- if (inst->priorSpeechProb < 0.01) {
- inst->priorSpeechProb = (float)0.01;
- }
-
- //final speech probability: combine prior model with LR factor:
- gainPrior = ((float)1.0 - inst->priorSpeechProb) / (inst->priorSpeechProb + (float)0.0001);
- for (i = 0; i < inst->magnLen; i++) {
- invLrt = (float)exp(-inst->logLrtTimeAvg[i]);
- invLrt = (float)gainPrior * invLrt;
- probSpeechFinal[i] = (float)1.0 / ((float)1.0 + invLrt);
- }
-}
-
-int WebRtcNs_ProcessCore(NSinst_t* inst,
- short* speechFrame,
- short* speechFrameHB,
- short* outFrame,
- short* outFrameHB) {
- // main routine for noise reduction
-
- int flagHB = 0;
- int i;
- const int kStartBand = 5; // Skip first frequency bins during estimation.
- int updateParsFlag;
-
- float energy1, energy2, gain, factor, factor1, factor2;
- float signalEnergy, sumMagn;
- float snrPrior, currentEstimateStsa;
- float tmpFloat1, tmpFloat2, tmpFloat3, probSpeech, probNonSpeech;
- float gammaNoiseTmp, gammaNoiseOld;
- float noiseUpdateTmp, fTmp, dTmp;
- float fin[BLOCKL_MAX], fout[BLOCKL_MAX];
- float winData[ANAL_BLOCKL_MAX];
- float magn[HALF_ANAL_BLOCKL], noise[HALF_ANAL_BLOCKL];
- float theFilter[HALF_ANAL_BLOCKL], theFilterTmp[HALF_ANAL_BLOCKL];
- float snrLocPost[HALF_ANAL_BLOCKL], snrLocPrior[HALF_ANAL_BLOCKL];
- float probSpeechFinal[HALF_ANAL_BLOCKL], previousEstimateStsa[HALF_ANAL_BLOCKL];
- float real[ANAL_BLOCKL_MAX], imag[HALF_ANAL_BLOCKL];
- // Variables during startup
- float sum_log_i = 0.0;
- float sum_log_i_square = 0.0;
- float sum_log_magn = 0.0;
- float sum_log_i_log_magn = 0.0;
- float parametric_noise = 0.0;
- float parametric_exp = 0.0;
- float parametric_num = 0.0;
-
- // SWB variables
- int deltaBweHB = 1;
- int deltaGainHB = 1;
- float decayBweHB = 1.0;
- float gainMapParHB = 1.0;
- float gainTimeDomainHB = 1.0;
- float avgProbSpeechHB, avgProbSpeechHBTmp, avgFilterGainHB, gainModHB;
-
- // Check that initiation has been done
- if (inst->initFlag != 1) {
- return (-1);
- }
- // Check for valid pointers based on sampling rate
- if (inst->fs == 32000) {
- if (speechFrameHB == NULL) {
- return -1;
- }
- flagHB = 1;
- // range for averaging low band quantities for H band gain
- deltaBweHB = (int)inst->magnLen / 4;
- deltaGainHB = deltaBweHB;
- }
- //
- updateParsFlag = inst->modelUpdatePars[0];
- //
-
- //for LB do all processing
- // convert to float
- for (i = 0; i < inst->blockLen10ms; i++) {
- fin[i] = (float)speechFrame[i];
- }
- // update analysis buffer for L band
- memcpy(inst->dataBuf, inst->dataBuf + inst->blockLen10ms,
- sizeof(float) * (inst->anaLen - inst->blockLen10ms));
- memcpy(inst->dataBuf + inst->anaLen - inst->blockLen10ms, fin,
- sizeof(float) * inst->blockLen10ms);
-
- if (flagHB == 1) {
- // convert to float
- for (i = 0; i < inst->blockLen10ms; i++) {
- fin[i] = (float)speechFrameHB[i];
- }
- // update analysis buffer for H band
- memcpy(inst->dataBufHB, inst->dataBufHB + inst->blockLen10ms,
- sizeof(float) * (inst->anaLen - inst->blockLen10ms));
- memcpy(inst->dataBufHB + inst->anaLen - inst->blockLen10ms, fin,
- sizeof(float) * inst->blockLen10ms);
- }
-
- // check if processing needed
- if (inst->outLen == 0) {
- // windowing
- energy1 = 0.0;
- for (i = 0; i < inst->anaLen; i++) {
- winData[i] = inst->window[i] * inst->dataBuf[i];
- energy1 += winData[i] * winData[i];
- }
- if (energy1 == 0.0) {
- // synthesize the special case of zero input
- // we want to avoid updating statistics in this case:
- // Updating feature statistics when we have zeros only will cause thresholds to
- // move towards zero signal situations. This in turn has the effect that once the
- // signal is "turned on" (non-zero values) everything will be treated as speech
- // and there is no noise suppression effect. Depending on the duration of the
- // inactive signal it takes a considerable amount of time for the system to learn
- // what is noise and what is speech.
-
- // read out fully processed segment
- for (i = inst->windShift; i < inst->blockLen + inst->windShift; i++) {
- fout[i - inst->windShift] = inst->syntBuf[i];
- }
- // update synthesis buffer
- memcpy(inst->syntBuf, inst->syntBuf + inst->blockLen,
- sizeof(float) * (inst->anaLen - inst->blockLen));
- memset(inst->syntBuf + inst->anaLen - inst->blockLen, 0,
- sizeof(float) * inst->blockLen);
-
- // out buffer
- inst->outLen = inst->blockLen - inst->blockLen10ms;
- if (inst->blockLen > inst->blockLen10ms) {
- for (i = 0; i < inst->outLen; i++) {
- inst->outBuf[i] = fout[i + inst->blockLen10ms];
- }
- }
- // convert to short
- for (i = 0; i < inst->blockLen10ms; i++) {
- dTmp = fout[i];
- if (dTmp < WEBRTC_SPL_WORD16_MIN) {
- dTmp = WEBRTC_SPL_WORD16_MIN;
- } else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
- dTmp = WEBRTC_SPL_WORD16_MAX;
- }
- outFrame[i] = (short)dTmp;
- }
-
- // for time-domain gain of HB
- if (flagHB == 1) {
- for (i = 0; i < inst->blockLen10ms; i++) {
- dTmp = inst->dataBufHB[i];
- if (dTmp < WEBRTC_SPL_WORD16_MIN) {
- dTmp = WEBRTC_SPL_WORD16_MIN;
- } else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
- dTmp = WEBRTC_SPL_WORD16_MAX;
- }
- outFrameHB[i] = (short)dTmp;
- }
- } // end of H band gain computation
- //
- return 0;
- }
-
- //
- inst->blockInd++; // Update the block index only when we process a block.
- // FFT
- WebRtc_rdft(inst->anaLen, 1, winData, inst->ip, inst->wfft);
-
- imag[0] = 0;
- real[0] = winData[0];
- magn[0] = (float)(fabs(real[0]) + 1.0f);
- imag[inst->magnLen - 1] = 0;
- real[inst->magnLen - 1] = winData[1];
- magn[inst->magnLen - 1] = (float)(fabs(real[inst->magnLen - 1]) + 1.0f);
- signalEnergy = (float)(real[0] * real[0]) +
- (float)(real[inst->magnLen - 1] * real[inst->magnLen - 1]);
- sumMagn = magn[0] + magn[inst->magnLen - 1];
- if (inst->blockInd < END_STARTUP_SHORT) {
- inst->initMagnEst[0] += magn[0];
- inst->initMagnEst[inst->magnLen - 1] += magn[inst->magnLen - 1];
- tmpFloat2 = log((float)(inst->magnLen - 1));
- sum_log_i = tmpFloat2;
- sum_log_i_square = tmpFloat2 * tmpFloat2;
- tmpFloat1 = log(magn[inst->magnLen - 1]);
- sum_log_magn = tmpFloat1;
- sum_log_i_log_magn = tmpFloat2 * tmpFloat1;
- }
- for (i = 1; i < inst->magnLen - 1; i++) {
- real[i] = winData[2 * i];
- imag[i] = winData[2 * i + 1];
- // magnitude spectrum
- fTmp = real[i] * real[i];
- fTmp += imag[i] * imag[i];
- signalEnergy += fTmp;
- magn[i] = ((float)sqrt(fTmp)) + 1.0f;
- sumMagn += magn[i];
- if (inst->blockInd < END_STARTUP_SHORT) {
- inst->initMagnEst[i] += magn[i];
- if (i >= kStartBand) {
- tmpFloat2 = log((float)i);
- sum_log_i += tmpFloat2;
- sum_log_i_square += tmpFloat2 * tmpFloat2;
- tmpFloat1 = log(magn[i]);
- sum_log_magn += tmpFloat1;
- sum_log_i_log_magn += tmpFloat2 * tmpFloat1;
- }
- }
- }
- signalEnergy = signalEnergy / ((float)inst->magnLen);
- inst->signalEnergy = signalEnergy;
- inst->sumMagn = sumMagn;
-
- //compute spectral flatness on input spectrum
- WebRtcNs_ComputeSpectralFlatness(inst, magn);
- // quantile noise estimate
- WebRtcNs_NoiseEstimation(inst, magn, noise);
- //compute simplified noise model during startup
- if (inst->blockInd < END_STARTUP_SHORT) {
- // Estimate White noise
- inst->whiteNoiseLevel += sumMagn / ((float)inst->magnLen) * inst->overdrive;
- // Estimate Pink noise parameters
- tmpFloat1 = sum_log_i_square * ((float)(inst->magnLen - kStartBand));
- tmpFloat1 -= (sum_log_i * sum_log_i);
- tmpFloat2 = (sum_log_i_square * sum_log_magn - sum_log_i * sum_log_i_log_magn);
- tmpFloat3 = tmpFloat2 / tmpFloat1;
- // Constrain the estimated spectrum to be positive
- if (tmpFloat3 < 0.0f) {
- tmpFloat3 = 0.0f;
- }
- inst->pinkNoiseNumerator += tmpFloat3;
- tmpFloat2 = (sum_log_i * sum_log_magn);
- tmpFloat2 -= ((float)(inst->magnLen - kStartBand)) * sum_log_i_log_magn;
- tmpFloat3 = tmpFloat2 / tmpFloat1;
- // Constrain the pink noise power to be in the interval [0, 1];
- if (tmpFloat3 < 0.0f) {
- tmpFloat3 = 0.0f;
- }
- if (tmpFloat3 > 1.0f) {
- tmpFloat3 = 1.0f;
- }
- inst->pinkNoiseExp += tmpFloat3;
-
- // Calculate frequency independent parts of parametric noise estimate.
- if (inst->pinkNoiseExp == 0.0f) {
- // Use white noise estimate
- parametric_noise = inst->whiteNoiseLevel;
- } else {
- // Use pink noise estimate
- parametric_num = exp(inst->pinkNoiseNumerator / (float)(inst->blockInd + 1));
- parametric_num *= (float)(inst->blockInd + 1);
- parametric_exp = inst->pinkNoiseExp / (float)(inst->blockInd + 1);
- parametric_noise = parametric_num / pow((float)kStartBand, parametric_exp);
- }
- for (i = 0; i < inst->magnLen; i++) {
- // Estimate the background noise using the white and pink noise parameters
- if ((inst->pinkNoiseExp > 0.0f) && (i >= kStartBand)) {
- // Use pink noise estimate
- parametric_noise = parametric_num / pow((float)i, parametric_exp);
- }
- theFilterTmp[i] = (inst->initMagnEst[i] - inst->overdrive * parametric_noise);
- theFilterTmp[i] /= (inst->initMagnEst[i] + (float)0.0001);
- // Weight quantile noise with modeled noise
- noise[i] *= (inst->blockInd);
- tmpFloat2 = parametric_noise * (END_STARTUP_SHORT - inst->blockInd);
- noise[i] += (tmpFloat2 / (float)(inst->blockInd + 1));
- noise[i] /= END_STARTUP_SHORT;
- }
- }
- //compute average signal during END_STARTUP_LONG time:
- // used to normalize spectral difference measure
- if (inst->blockInd < END_STARTUP_LONG) {
- inst->featureData[5] *= inst->blockInd;
- inst->featureData[5] += signalEnergy;
- inst->featureData[5] /= (inst->blockInd + 1);
- }
-
-#ifdef PROCESS_FLOW_0
- if (inst->blockInd > END_STARTUP_LONG) {
- //option: average the quantile noise: for check with AEC2
- for (i = 0; i < inst->magnLen; i++) {
- noise[i] = (float)0.6 * inst->noisePrev[i] + (float)0.4 * noise[i];
- }
- for (i = 0; i < inst->magnLen; i++) {
- // Wiener with over sub-substraction:
- theFilter[i] = (magn[i] - inst->overdrive * noise[i]) / (magn[i] + (float)0.0001);
- }
- }
-#else
- //start processing at frames == converged+1
- //
- // STEP 1: compute prior and post snr based on quantile noise est
- //
-
- // compute DD estimate of prior SNR: needed for new method
- for (i = 0; i < inst->magnLen; i++) {
- // post snr
- snrLocPost[i] = (float)0.0;
- if (magn[i] > noise[i]) {
- snrLocPost[i] = magn[i] / (noise[i] + (float)0.0001) - (float)1.0;
- }
- // previous post snr
- // previous estimate: based on previous frame with gain filter
- previousEstimateStsa[i] = inst->magnPrev[i] / (inst->noisePrev[i] + (float)0.0001)
- * (inst->smooth[i]);
- // DD estimate is sum of two terms: current estimate and previous estimate
- // directed decision update of snrPrior
- snrLocPrior[i] = DD_PR_SNR * previousEstimateStsa[i] + ((float)1.0 - DD_PR_SNR)
- * snrLocPost[i];
- // post and prior snr needed for step 2
- } // end of loop over freqs
-#ifdef PROCESS_FLOW_1
- for (i = 0; i < inst->magnLen; i++) {
- // gain filter
- tmpFloat1 = inst->overdrive + snrLocPrior[i];
- tmpFloat2 = (float)snrLocPrior[i] / tmpFloat1;
- theFilter[i] = (float)tmpFloat2;
- } // end of loop over freqs
-#endif
- // done with step 1: dd computation of prior and post snr
-
- //
- //STEP 2: compute speech/noise likelihood
- //
-#ifdef PROCESS_FLOW_2
- // compute difference of input spectrum with learned/estimated noise spectrum
- WebRtcNs_ComputeSpectralDifference(inst, magn);
- // compute histograms for parameter decisions (thresholds and weights for features)
- // parameters are extracted once every window time (=inst->modelUpdatePars[1])
- if (updateParsFlag >= 1) {
- // counter update
- inst->modelUpdatePars[3]--;
- // update histogram
- if (inst->modelUpdatePars[3] > 0) {
- WebRtcNs_FeatureParameterExtraction(inst, 0);
- }
- // compute model parameters
- if (inst->modelUpdatePars[3] == 0) {
- WebRtcNs_FeatureParameterExtraction(inst, 1);
- inst->modelUpdatePars[3] = inst->modelUpdatePars[1];
- // if wish to update only once, set flag to zero
- if (updateParsFlag == 1) {
- inst->modelUpdatePars[0] = 0;
- } else {
- // update every window:
- // get normalization for spectral difference for next window estimate
- inst->featureData[6] = inst->featureData[6]
- / ((float)inst->modelUpdatePars[1]);
- inst->featureData[5] = (float)0.5 * (inst->featureData[6]
- + inst->featureData[5]);
- inst->featureData[6] = (float)0.0;
- }
- }
- }
- // compute speech/noise probability
- WebRtcNs_SpeechNoiseProb(inst, probSpeechFinal, snrLocPrior, snrLocPost);
- // time-avg parameter for noise update
- gammaNoiseTmp = NOISE_UPDATE;
- for (i = 0; i < inst->magnLen; i++) {
- probSpeech = probSpeechFinal[i];
- probNonSpeech = (float)1.0 - probSpeech;
- // temporary noise update:
- // use it for speech frames if update value is less than previous
- noiseUpdateTmp = gammaNoiseTmp * inst->noisePrev[i] + ((float)1.0 - gammaNoiseTmp)
- * (probNonSpeech * magn[i] + probSpeech * inst->noisePrev[i]);
- //
- // time-constant based on speech/noise state
- gammaNoiseOld = gammaNoiseTmp;
- gammaNoiseTmp = NOISE_UPDATE;
- // increase gamma (i.e., less noise update) for frame likely to be speech
- if (probSpeech > PROB_RANGE) {
- gammaNoiseTmp = SPEECH_UPDATE;
- }
- // conservative noise update
- if (probSpeech < PROB_RANGE) {
- inst->magnAvgPause[i] += GAMMA_PAUSE * (magn[i] - inst->magnAvgPause[i]);
- }
- // noise update
- if (gammaNoiseTmp == gammaNoiseOld) {
- noise[i] = noiseUpdateTmp;
- } else {
- noise[i] = gammaNoiseTmp * inst->noisePrev[i] + ((float)1.0 - gammaNoiseTmp)
- * (probNonSpeech * magn[i] + probSpeech * inst->noisePrev[i]);
- // allow for noise update downwards:
- // if noise update decreases the noise, it is safe, so allow it to happen
- if (noiseUpdateTmp < noise[i]) {
- noise[i] = noiseUpdateTmp;
- }
- }
- } // end of freq loop
- // done with step 2: noise update
-
- //
- // STEP 3: compute dd update of prior snr and post snr based on new noise estimate
- //
- for (i = 0; i < inst->magnLen; i++) {
- // post and prior snr
- currentEstimateStsa = (float)0.0;
- if (magn[i] > noise[i]) {
- currentEstimateStsa = magn[i] / (noise[i] + (float)0.0001) - (float)1.0;
- }
- // DD estimate is sume of two terms: current estimate and previous estimate
- // directed decision update of snrPrior
- snrPrior = DD_PR_SNR * previousEstimateStsa[i] + ((float)1.0 - DD_PR_SNR)
- * currentEstimateStsa;
- // gain filter
- tmpFloat1 = inst->overdrive + snrPrior;
- tmpFloat2 = (float)snrPrior / tmpFloat1;
- theFilter[i] = (float)tmpFloat2;
- } // end of loop over freqs
- // done with step3
-#endif
-#endif
-
- for (i = 0; i < inst->magnLen; i++) {
- // flooring bottom
- if (theFilter[i] < inst->denoiseBound) {
- theFilter[i] = inst->denoiseBound;
- }
- // flooring top
- if (theFilter[i] > (float)1.0) {
- theFilter[i] = 1.0;
- }
- if (inst->blockInd < END_STARTUP_SHORT) {
- // flooring bottom
- if (theFilterTmp[i] < inst->denoiseBound) {
- theFilterTmp[i] = inst->denoiseBound;
- }
- // flooring top
- if (theFilterTmp[i] > (float)1.0) {
- theFilterTmp[i] = 1.0;
- }
- // Weight the two suppression filters
- theFilter[i] *= (inst->blockInd);
- theFilterTmp[i] *= (END_STARTUP_SHORT - inst->blockInd);
- theFilter[i] += theFilterTmp[i];
- theFilter[i] /= (END_STARTUP_SHORT);
- }
- // smoothing
-#ifdef PROCESS_FLOW_0
- inst->smooth[i] *= SMOOTH; // value set to 0.7 in define.h file
- inst->smooth[i] += ((float)1.0 - SMOOTH) * theFilter[i];
-#else
- inst->smooth[i] = theFilter[i];
-#endif
- real[i] *= inst->smooth[i];
- imag[i] *= inst->smooth[i];
- }
- // keep track of noise and magn spectrum for next frame
- for (i = 0; i < inst->magnLen; i++) {
- inst->noisePrev[i] = noise[i];
- inst->magnPrev[i] = magn[i];
- }
- // back to time domain
- winData[0] = real[0];
- winData[1] = real[inst->magnLen - 1];
- for (i = 1; i < inst->magnLen - 1; i++) {
- winData[2 * i] = real[i];
- winData[2 * i + 1] = imag[i];
- }
- WebRtc_rdft(inst->anaLen, -1, winData, inst->ip, inst->wfft);
-
- for (i = 0; i < inst->anaLen; i++) {
- real[i] = 2.0f * winData[i] / inst->anaLen; // fft scaling
- }
-
- //scale factor: only do it after END_STARTUP_LONG time
- factor = (float)1.0;
- if (inst->gainmap == 1 && inst->blockInd > END_STARTUP_LONG) {
- factor1 = (float)1.0;
- factor2 = (float)1.0;
-
- energy2 = 0.0;
- for (i = 0; i < inst->anaLen; i++) {
- energy2 += (float)real[i] * (float)real[i];
- }
- gain = (float)sqrt(energy2 / (energy1 + (float)1.0));
-
-#ifdef PROCESS_FLOW_2
- // scaling for new version
- if (gain > B_LIM) {
- factor1 = (float)1.0 + (float)1.3 * (gain - B_LIM);
- if (gain * factor1 > (float)1.0) {
- factor1 = (float)1.0 / gain;
- }
- }
- if (gain < B_LIM) {
- //don't reduce scale too much for pause regions:
- // attenuation here should be controlled by flooring
- if (gain <= inst->denoiseBound) {
- gain = inst->denoiseBound;
- }
- factor2 = (float)1.0 - (float)0.3 * (B_LIM - gain);
- }
- //combine both scales with speech/noise prob:
- // note prior (priorSpeechProb) is not frequency dependent
- factor = inst->priorSpeechProb * factor1 + ((float)1.0 - inst->priorSpeechProb)
- * factor2;
-#else
- if (gain > B_LIM) {
- factor = (float)1.0 + (float)1.3 * (gain - B_LIM);
- } else {
- factor = (float)1.0 + (float)2.0 * (gain - B_LIM);
- }
- if (gain * factor > (float)1.0) {
- factor = (float)1.0 / gain;
- }
-#endif
- } // out of inst->gainmap==1
-
- // synthesis
- for (i = 0; i < inst->anaLen; i++) {
- inst->syntBuf[i] += factor * inst->window[i] * (float)real[i];
- }
- // read out fully processed segment
- for (i = inst->windShift; i < inst->blockLen + inst->windShift; i++) {
- fout[i - inst->windShift] = inst->syntBuf[i];
- }
- // update synthesis buffer
- memcpy(inst->syntBuf, inst->syntBuf + inst->blockLen,
- sizeof(float) * (inst->anaLen - inst->blockLen));
- memset(inst->syntBuf + inst->anaLen - inst->blockLen, 0,
- sizeof(float) * inst->blockLen);
-
- // out buffer
- inst->outLen = inst->blockLen - inst->blockLen10ms;
- if (inst->blockLen > inst->blockLen10ms) {
- for (i = 0; i < inst->outLen; i++) {
- inst->outBuf[i] = fout[i + inst->blockLen10ms];
- }
- }
- } // end of if out.len==0
- else {
- for (i = 0; i < inst->blockLen10ms; i++) {
- fout[i] = inst->outBuf[i];
- }
- memcpy(inst->outBuf, inst->outBuf + inst->blockLen10ms,
- sizeof(float) * (inst->outLen - inst->blockLen10ms));
- memset(inst->outBuf + inst->outLen - inst->blockLen10ms, 0,
- sizeof(float) * inst->blockLen10ms);
- inst->outLen -= inst->blockLen10ms;
- }
-
- // convert to short
- for (i = 0; i < inst->blockLen10ms; i++) {
- dTmp = fout[i];
- if (dTmp < WEBRTC_SPL_WORD16_MIN) {
- dTmp = WEBRTC_SPL_WORD16_MIN;
- } else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
- dTmp = WEBRTC_SPL_WORD16_MAX;
- }
- outFrame[i] = (short)dTmp;
- }
-
- // for time-domain gain of HB
- if (flagHB == 1) {
- for (i = 0; i < inst->magnLen; i++) {
- inst->speechProbHB[i] = probSpeechFinal[i];
- }
- // average speech prob from low band
- // avg over second half (i.e., 4->8kHz) of freq. spectrum
- avgProbSpeechHB = 0.0;
- for (i = inst->magnLen - deltaBweHB - 1; i < inst->magnLen - 1; i++) {
- avgProbSpeechHB += inst->speechProbHB[i];
- }
- avgProbSpeechHB = avgProbSpeechHB / ((float)deltaBweHB);
- // average filter gain from low band
- // average over second half (i.e., 4->8kHz) of freq. spectrum
- avgFilterGainHB = 0.0;
- for (i = inst->magnLen - deltaGainHB - 1; i < inst->magnLen - 1; i++) {
- avgFilterGainHB += inst->smooth[i];
- }
- avgFilterGainHB = avgFilterGainHB / ((float)(deltaGainHB));
- avgProbSpeechHBTmp = (float)2.0 * avgProbSpeechHB - (float)1.0;
- // gain based on speech prob:
- gainModHB = (float)0.5 * ((float)1.0 + (float)tanh(gainMapParHB * avgProbSpeechHBTmp));
- //combine gain with low band gain
- gainTimeDomainHB = (float)0.5 * gainModHB + (float)0.5 * avgFilterGainHB;
- if (avgProbSpeechHB >= (float)0.5) {
- gainTimeDomainHB = (float)0.25 * gainModHB + (float)0.75 * avgFilterGainHB;
- }
- gainTimeDomainHB = gainTimeDomainHB * decayBweHB;
- //make sure gain is within flooring range
- // flooring bottom
- if (gainTimeDomainHB < inst->denoiseBound) {
- gainTimeDomainHB = inst->denoiseBound;
- }
- // flooring top
- if (gainTimeDomainHB > (float)1.0) {
- gainTimeDomainHB = 1.0;
- }
- //apply gain
- for (i = 0; i < inst->blockLen10ms; i++) {
- dTmp = gainTimeDomainHB * inst->dataBufHB[i];
- if (dTmp < WEBRTC_SPL_WORD16_MIN) {
- dTmp = WEBRTC_SPL_WORD16_MIN;
- } else if (dTmp > WEBRTC_SPL_WORD16_MAX) {
- dTmp = WEBRTC_SPL_WORD16_MAX;
- }
- outFrameHB[i] = (short)dTmp;
- }
- } // end of H band gain computation
- //
-
- return 0;
-}
diff --git a/trunk/src/modules/audio_processing/ns/ns_core.h b/trunk/src/modules/audio_processing/ns/ns_core.h
deleted file mode 100644
index 2f4c34f..0000000
--- a/trunk/src/modules/audio_processing/ns/ns_core.h
+++ /dev/null
@@ -1,179 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_
-
-#include "defines.h"
-
-typedef struct NSParaExtract_t_ {
-
- //bin size of histogram
- float binSizeLrt;
- float binSizeSpecFlat;
- float binSizeSpecDiff;
- //range of histogram over which lrt threshold is computed
- float rangeAvgHistLrt;
- //scale parameters: multiply dominant peaks of the histograms by scale factor to obtain
- //thresholds for prior model
- float factor1ModelPars; //for lrt and spectral difference
- float factor2ModelPars; //for spectral_flatness: used when noise is flatter than speech
- //peak limit for spectral flatness (varies between 0 and 1)
- float thresPosSpecFlat;
- //limit on spacing of two highest peaks in histogram: spacing determined by bin size
- float limitPeakSpacingSpecFlat;
- float limitPeakSpacingSpecDiff;
- //limit on relevance of second peak:
- float limitPeakWeightsSpecFlat;
- float limitPeakWeightsSpecDiff;
- //limit on fluctuation of lrt feature
- float thresFluctLrt;
- //limit on the max and min values for the feature thresholds
- float maxLrt;
- float minLrt;
- float maxSpecFlat;
- float minSpecFlat;
- float maxSpecDiff;
- float minSpecDiff;
- //criteria of weight of histogram peak to accept/reject feature
- int thresWeightSpecFlat;
- int thresWeightSpecDiff;
-
-} NSParaExtract_t;
-
-typedef struct NSinst_t_ {
-
- WebRtc_UWord32 fs;
- int blockLen;
- int blockLen10ms;
- int windShift;
- int outLen;
- int anaLen;
- int magnLen;
- int aggrMode;
- const float* window;
- float dataBuf[ANAL_BLOCKL_MAX];
- float syntBuf[ANAL_BLOCKL_MAX];
- float outBuf[3 * BLOCKL_MAX];
-
- int initFlag;
- // parameters for quantile noise estimation
- float density[SIMULT* HALF_ANAL_BLOCKL];
- float lquantile[SIMULT* HALF_ANAL_BLOCKL];
- float quantile[HALF_ANAL_BLOCKL];
- int counter[SIMULT];
- int updates;
- // parameters for Wiener filter
- float smooth[HALF_ANAL_BLOCKL];
- float overdrive;
- float denoiseBound;
- int gainmap;
- // fft work arrays.
- int ip[IP_LENGTH];
- float wfft[W_LENGTH];
-
- // parameters for new method: some not needed, will reduce/cleanup later
- WebRtc_Word32 blockInd; //frame index counter
- int modelUpdatePars[4]; //parameters for updating or estimating
- // thresholds/weights for prior model
- float priorModelPars[7]; //parameters for prior model
- float noisePrev[HALF_ANAL_BLOCKL]; //noise spectrum from previous frame
- float magnPrev[HALF_ANAL_BLOCKL]; //magnitude spectrum of previous frame
- float logLrtTimeAvg[HALF_ANAL_BLOCKL]; //log lrt factor with time-smoothing
- float priorSpeechProb; //prior speech/noise probability
- float featureData[7]; //data for features
- float magnAvgPause[HALF_ANAL_BLOCKL]; //conservative noise spectrum estimate
- float signalEnergy; //energy of magn
- float sumMagn; //sum of magn
- float whiteNoiseLevel; //initial noise estimate
- float initMagnEst[HALF_ANAL_BLOCKL]; //initial magnitude spectrum estimate
- float pinkNoiseNumerator; //pink noise parameter: numerator
- float pinkNoiseExp; //pink noise parameter: power of freq
- NSParaExtract_t featureExtractionParams; //parameters for feature extraction
- //histograms for parameter estimation
- int histLrt[HIST_PAR_EST];
- int histSpecFlat[HIST_PAR_EST];
- int histSpecDiff[HIST_PAR_EST];
- //quantities for high band estimate
- float speechProbHB[HALF_ANAL_BLOCKL]; //final speech/noise prob: prior + LRT
- float dataBufHB[ANAL_BLOCKL_MAX]; //buffering data for HB
-
-} NSinst_t;
-
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/****************************************************************************
- * WebRtcNs_InitCore(...)
- *
- * This function initializes a noise suppression instance
- *
- * Input:
- * - inst : Instance that should be initialized
- * - fs : Sampling frequency
- *
- * Output:
- * - inst : Initialized instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcNs_InitCore(NSinst_t* inst, WebRtc_UWord32 fs);
-
-/****************************************************************************
- * WebRtcNs_set_policy_core(...)
- *
- * This changes the aggressiveness of the noise suppression method.
- *
- * Input:
- * - inst : Instance that should be initialized
- * - mode : 0: Mild (6 dB), 1: Medium (10 dB), 2: Aggressive (15 dB)
- *
- * Output:
- * - NS_inst : Initialized instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcNs_set_policy_core(NSinst_t* inst, int mode);
-
-/****************************************************************************
- * WebRtcNs_ProcessCore
- *
- * Do noise suppression.
- *
- * Input:
- * - inst : Instance that should be initialized
- * - inFrameLow : Input speech frame for lower band
- * - inFrameHigh : Input speech frame for higher band
- *
- * Output:
- * - inst : Updated instance
- * - outFrameLow : Output speech frame for lower band
- * - outFrameHigh : Output speech frame for higher band
- *
- * Return value : 0 - OK
- * -1 - Error
- */
-
-
-int WebRtcNs_ProcessCore(NSinst_t* inst,
- short* inFrameLow,
- short* inFrameHigh,
- short* outFrameLow,
- short* outFrameHigh);
-
-
-#ifdef __cplusplus
-}
-#endif
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NS_CORE_H_
diff --git a/trunk/src/modules/audio_processing/ns/nsx_core.c b/trunk/src/modules/audio_processing/ns/nsx_core.c
deleted file mode 100644
index 51bde0c..0000000
--- a/trunk/src/modules/audio_processing/ns/nsx_core.c
+++ /dev/null
@@ -1,2444 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "noise_suppression_x.h"
-
-#include <assert.h>
-#include <math.h>
-#include <string.h>
-#include <stdlib.h>
-#include <stdio.h>
-
-#include "cpu_features_wrapper.h"
-#include "nsx_core.h"
-
-// Skip first frequency bins during estimation. (0 <= value < 64)
-static const int kStartBand = 5;
-
-// Constants to compensate for shifting signal log(2^shifts).
-const WebRtc_Word16 WebRtcNsx_kLogTable[9] = {
- 0, 177, 355, 532, 710, 887, 1065, 1242, 1420
-};
-
-const WebRtc_Word16 WebRtcNsx_kCounterDiv[201] = {
- 32767, 16384, 10923, 8192, 6554, 5461, 4681,
- 4096, 3641, 3277, 2979, 2731, 2521, 2341, 2185, 2048, 1928, 1820, 1725, 1638, 1560,
- 1489, 1425, 1365, 1311, 1260, 1214, 1170, 1130, 1092, 1057, 1024, 993, 964, 936, 910,
- 886, 862, 840, 819, 799, 780, 762, 745, 728, 712, 697, 683, 669, 655, 643, 630, 618,
- 607, 596, 585, 575, 565, 555, 546, 537, 529, 520, 512, 504, 496, 489, 482, 475, 468,
- 462, 455, 449, 443, 437, 431, 426, 420, 415, 410, 405, 400, 395, 390, 386, 381, 377,
- 372, 368, 364, 360, 356, 352, 349, 345, 341, 338, 334, 331, 328, 324, 321, 318, 315,
- 312, 309, 306, 303, 301, 298, 295, 293, 290, 287, 285, 282, 280, 278, 275, 273, 271,
- 269, 266, 264, 262, 260, 258, 256, 254, 252, 250, 248, 246, 245, 243, 241, 239, 237,
- 236, 234, 232, 231, 229, 228, 226, 224, 223, 221, 220, 218, 217, 216, 214, 213, 211,
- 210, 209, 207, 206, 205, 204, 202, 201, 200, 199, 197, 196, 195, 194, 193, 192, 191,
- 189, 188, 187, 186, 185, 184, 183, 182, 181, 180, 179, 178, 177, 176, 175, 174, 173,
- 172, 172, 171, 170, 169, 168, 167, 166, 165, 165, 164, 163
-};
-
-const WebRtc_Word16 WebRtcNsx_kLogTableFrac[256] = {
- 0, 1, 3, 4, 6, 7, 9, 10, 11, 13, 14, 16, 17, 18, 20, 21,
- 22, 24, 25, 26, 28, 29, 30, 32, 33, 34, 36, 37, 38, 40, 41, 42,
- 44, 45, 46, 47, 49, 50, 51, 52, 54, 55, 56, 57, 59, 60, 61, 62,
- 63, 65, 66, 67, 68, 69, 71, 72, 73, 74, 75, 77, 78, 79, 80, 81,
- 82, 84, 85, 86, 87, 88, 89, 90, 92, 93, 94, 95, 96, 97, 98, 99,
- 100, 102, 103, 104, 105, 106, 107, 108, 109, 110, 111, 112, 113, 114, 116, 117,
- 118, 119, 120, 121, 122, 123, 124, 125, 126, 127, 128, 129, 130, 131, 132, 133,
- 134, 135, 136, 137, 138, 139, 140, 141, 142, 143, 144, 145, 146, 147, 148, 149,
- 150, 151, 152, 153, 154, 155, 155, 156, 157, 158, 159, 160, 161, 162, 163, 164,
- 165, 166, 167, 168, 169, 169, 170, 171, 172, 173, 174, 175, 176, 177, 178, 178,
- 179, 180, 181, 182, 183, 184, 185, 185, 186, 187, 188, 189, 190, 191, 192, 192,
- 193, 194, 195, 196, 197, 198, 198, 199, 200, 201, 202, 203, 203, 204, 205, 206,
- 207, 208, 208, 209, 210, 211, 212, 212, 213, 214, 215, 216, 216, 217, 218, 219,
- 220, 220, 221, 222, 223, 224, 224, 225, 226, 227, 228, 228, 229, 230, 231, 231,
- 232, 233, 234, 234, 235, 236, 237, 238, 238, 239, 240, 241, 241, 242, 243, 244,
- 244, 245, 246, 247, 247, 248, 249, 249, 250, 251, 252, 252, 253, 254, 255, 255
-};
-
-static const WebRtc_Word16 kPowTableFrac[1024] = {
- 0, 1, 1, 2, 3, 3, 4, 5,
- 6, 6, 7, 8, 8, 9, 10, 10,
- 11, 12, 13, 13, 14, 15, 15, 16,
- 17, 17, 18, 19, 20, 20, 21, 22,
- 22, 23, 24, 25, 25, 26, 27, 27,
- 28, 29, 30, 30, 31, 32, 32, 33,
- 34, 35, 35, 36, 37, 37, 38, 39,
- 40, 40, 41, 42, 42, 43, 44, 45,
- 45, 46, 47, 48, 48, 49, 50, 50,
- 51, 52, 53, 53, 54, 55, 56, 56,
- 57, 58, 58, 59, 60, 61, 61, 62,
- 63, 64, 64, 65, 66, 67, 67, 68,
- 69, 69, 70, 71, 72, 72, 73, 74,
- 75, 75, 76, 77, 78, 78, 79, 80,
- 81, 81, 82, 83, 84, 84, 85, 86,
- 87, 87, 88, 89, 90, 90, 91, 92,
- 93, 93, 94, 95, 96, 96, 97, 98,
- 99, 100, 100, 101, 102, 103, 103, 104,
- 105, 106, 106, 107, 108, 109, 109, 110,
- 111, 112, 113, 113, 114, 115, 116, 116,
- 117, 118, 119, 119, 120, 121, 122, 123,
- 123, 124, 125, 126, 126, 127, 128, 129,
- 130, 130, 131, 132, 133, 133, 134, 135,
- 136, 137, 137, 138, 139, 140, 141, 141,
- 142, 143, 144, 144, 145, 146, 147, 148,
- 148, 149, 150, 151, 152, 152, 153, 154,
- 155, 156, 156, 157, 158, 159, 160, 160,
- 161, 162, 163, 164, 164, 165, 166, 167,
- 168, 168, 169, 170, 171, 172, 173, 173,
- 174, 175, 176, 177, 177, 178, 179, 180,
- 181, 181, 182, 183, 184, 185, 186, 186,
- 187, 188, 189, 190, 190, 191, 192, 193,
- 194, 195, 195, 196, 197, 198, 199, 200,
- 200, 201, 202, 203, 204, 205, 205, 206,
- 207, 208, 209, 210, 210, 211, 212, 213,
- 214, 215, 215, 216, 217, 218, 219, 220,
- 220, 221, 222, 223, 224, 225, 225, 226,
- 227, 228, 229, 230, 231, 231, 232, 233,
- 234, 235, 236, 237, 237, 238, 239, 240,
- 241, 242, 243, 243, 244, 245, 246, 247,
- 248, 249, 249, 250, 251, 252, 253, 254,
- 255, 255, 256, 257, 258, 259, 260, 261,
- 262, 262, 263, 264, 265, 266, 267, 268,
- 268, 269, 270, 271, 272, 273, 274, 275,
- 276, 276, 277, 278, 279, 280, 281, 282,
- 283, 283, 284, 285, 286, 287, 288, 289,
- 290, 291, 291, 292, 293, 294, 295, 296,
- 297, 298, 299, 299, 300, 301, 302, 303,
- 304, 305, 306, 307, 308, 308, 309, 310,
- 311, 312, 313, 314, 315, 316, 317, 318,
- 318, 319, 320, 321, 322, 323, 324, 325,
- 326, 327, 328, 328, 329, 330, 331, 332,
- 333, 334, 335, 336, 337, 338, 339, 339,
- 340, 341, 342, 343, 344, 345, 346, 347,
- 348, 349, 350, 351, 352, 352, 353, 354,
- 355, 356, 357, 358, 359, 360, 361, 362,
- 363, 364, 365, 366, 367, 367, 368, 369,
- 370, 371, 372, 373, 374, 375, 376, 377,
- 378, 379, 380, 381, 382, 383, 384, 385,
- 385, 386, 387, 388, 389, 390, 391, 392,
- 393, 394, 395, 396, 397, 398, 399, 400,
- 401, 402, 403, 404, 405, 406, 407, 408,
- 409, 410, 410, 411, 412, 413, 414, 415,
- 416, 417, 418, 419, 420, 421, 422, 423,
- 424, 425, 426, 427, 428, 429, 430, 431,
- 432, 433, 434, 435, 436, 437, 438, 439,
- 440, 441, 442, 443, 444, 445, 446, 447,
- 448, 449, 450, 451, 452, 453, 454, 455,
- 456, 457, 458, 459, 460, 461, 462, 463,
- 464, 465, 466, 467, 468, 469, 470, 471,
- 472, 473, 474, 475, 476, 477, 478, 479,
- 480, 481, 482, 483, 484, 485, 486, 487,
- 488, 489, 490, 491, 492, 493, 494, 495,
- 496, 498, 499, 500, 501, 502, 503, 504,
- 505, 506, 507, 508, 509, 510, 511, 512,
- 513, 514, 515, 516, 517, 518, 519, 520,
- 521, 522, 523, 525, 526, 527, 528, 529,
- 530, 531, 532, 533, 534, 535, 536, 537,
- 538, 539, 540, 541, 542, 544, 545, 546,
- 547, 548, 549, 550, 551, 552, 553, 554,
- 555, 556, 557, 558, 560, 561, 562, 563,
- 564, 565, 566, 567, 568, 569, 570, 571,
- 572, 574, 575, 576, 577, 578, 579, 580,
- 581, 582, 583, 584, 585, 587, 588, 589,
- 590, 591, 592, 593, 594, 595, 596, 597,
- 599, 600, 601, 602, 603, 604, 605, 606,
- 607, 608, 610, 611, 612, 613, 614, 615,
- 616, 617, 618, 620, 621, 622, 623, 624,
- 625, 626, 627, 628, 630, 631, 632, 633,
- 634, 635, 636, 637, 639, 640, 641, 642,
- 643, 644, 645, 646, 648, 649, 650, 651,
- 652, 653, 654, 656, 657, 658, 659, 660,
- 661, 662, 664, 665, 666, 667, 668, 669,
- 670, 672, 673, 674, 675, 676, 677, 678,
- 680, 681, 682, 683, 684, 685, 687, 688,
- 689, 690, 691, 692, 693, 695, 696, 697,
- 698, 699, 700, 702, 703, 704, 705, 706,
- 708, 709, 710, 711, 712, 713, 715, 716,
- 717, 718, 719, 720, 722, 723, 724, 725,
- 726, 728, 729, 730, 731, 732, 733, 735,
- 736, 737, 738, 739, 741, 742, 743, 744,
- 745, 747, 748, 749, 750, 751, 753, 754,
- 755, 756, 757, 759, 760, 761, 762, 763,
- 765, 766, 767, 768, 770, 771, 772, 773,
- 774, 776, 777, 778, 779, 780, 782, 783,
- 784, 785, 787, 788, 789, 790, 792, 793,
- 794, 795, 796, 798, 799, 800, 801, 803,
- 804, 805, 806, 808, 809, 810, 811, 813,
- 814, 815, 816, 818, 819, 820, 821, 823,
- 824, 825, 826, 828, 829, 830, 831, 833,
- 834, 835, 836, 838, 839, 840, 841, 843,
- 844, 845, 846, 848, 849, 850, 851, 853,
- 854, 855, 857, 858, 859, 860, 862, 863,
- 864, 866, 867, 868, 869, 871, 872, 873,
- 874, 876, 877, 878, 880, 881, 882, 883,
- 885, 886, 887, 889, 890, 891, 893, 894,
- 895, 896, 898, 899, 900, 902, 903, 904,
- 906, 907, 908, 909, 911, 912, 913, 915,
- 916, 917, 919, 920, 921, 923, 924, 925,
- 927, 928, 929, 931, 932, 933, 935, 936,
- 937, 938, 940, 941, 942, 944, 945, 946,
- 948, 949, 950, 952, 953, 955, 956, 957,
- 959, 960, 961, 963, 964, 965, 967, 968,
- 969, 971, 972, 973, 975, 976, 977, 979,
- 980, 981, 983, 984, 986, 987, 988, 990,
- 991, 992, 994, 995, 996, 998, 999, 1001,
- 1002, 1003, 1005, 1006, 1007, 1009, 1010, 1012,
- 1013, 1014, 1016, 1017, 1018, 1020, 1021, 1023
-};
-
-static const WebRtc_Word16 kIndicatorTable[17] = {
- 0, 2017, 3809, 5227, 6258, 6963, 7424, 7718,
- 7901, 8014, 8084, 8126, 8152, 8168, 8177, 8183, 8187
-};
-
-// hybrib Hanning & flat window
-static const WebRtc_Word16 kBlocks80w128x[128] = {
- 0, 536, 1072, 1606, 2139, 2669, 3196, 3720, 4240, 4756, 5266,
- 5771, 6270, 6762, 7246, 7723, 8192, 8652, 9102, 9543, 9974, 10394,
- 10803, 11200, 11585, 11958, 12318, 12665, 12998, 13318, 13623, 13913, 14189,
- 14449, 14694, 14924, 15137, 15334, 15515, 15679, 15826, 15956, 16069, 16165,
- 16244, 16305, 16349, 16375, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
- 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
- 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
- 16384, 16384, 16384, 16384, 16375, 16349, 16305, 16244, 16165, 16069, 15956,
- 15826, 15679, 15515, 15334, 15137, 14924, 14694, 14449, 14189, 13913, 13623,
- 13318, 12998, 12665, 12318, 11958, 11585, 11200, 10803, 10394, 9974, 9543,
- 9102, 8652, 8192, 7723, 7246, 6762, 6270, 5771, 5266, 4756, 4240,
- 3720, 3196, 2669, 2139, 1606, 1072, 536
-};
-
-// hybrib Hanning & flat window
-static const WebRtc_Word16 kBlocks160w256x[256] = {
- 0, 268, 536, 804, 1072, 1339, 1606, 1872,
- 2139, 2404, 2669, 2933, 3196, 3459, 3720, 3981,
- 4240, 4499, 4756, 5012, 5266, 5520, 5771, 6021,
- 6270, 6517, 6762, 7005, 7246, 7486, 7723, 7959,
- 8192, 8423, 8652, 8878, 9102, 9324, 9543, 9760,
- 9974, 10185, 10394, 10600, 10803, 11003, 11200, 11394,
- 11585, 11773, 11958, 12140, 12318, 12493, 12665, 12833,
- 12998, 13160, 13318, 13472, 13623, 13770, 13913, 14053,
- 14189, 14321, 14449, 14574, 14694, 14811, 14924, 15032,
- 15137, 15237, 15334, 15426, 15515, 15599, 15679, 15754,
- 15826, 15893, 15956, 16015, 16069, 16119, 16165, 16207,
- 16244, 16277, 16305, 16329, 16349, 16364, 16375, 16382,
- 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
- 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
- 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
- 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
- 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
- 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
- 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
- 16384, 16384, 16384, 16384, 16384, 16384, 16384, 16384,
- 16384, 16382, 16375, 16364, 16349, 16329, 16305, 16277,
- 16244, 16207, 16165, 16119, 16069, 16015, 15956, 15893,
- 15826, 15754, 15679, 15599, 15515, 15426, 15334, 15237,
- 15137, 15032, 14924, 14811, 14694, 14574, 14449, 14321,
- 14189, 14053, 13913, 13770, 13623, 13472, 13318, 13160,
- 12998, 12833, 12665, 12493, 12318, 12140, 11958, 11773,
- 11585, 11394, 11200, 11003, 10803, 10600, 10394, 10185,
- 9974, 9760, 9543, 9324, 9102, 8878, 8652, 8423,
- 8192, 7959, 7723, 7486, 7246, 7005, 6762, 6517,
- 6270, 6021, 5771, 5520, 5266, 5012, 4756, 4499,
- 4240, 3981, 3720, 3459, 3196, 2933, 2669, 2404,
- 2139, 1872, 1606, 1339, 1072, 804, 536, 268
-};
-
-// Gain factor1 table: Input value in Q8 and output value in Q13
-// original floating point code
-// if (gain > blim) {
-// factor1 = 1.0 + 1.3 * (gain - blim);
-// if (gain * factor1 > 1.0) {
-// factor1 = 1.0 / gain;
-// }
-// } else {
-// factor1 = 1.0;
-// }
-static const WebRtc_Word16 kFactor1Table[257] = {
- 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8233, 8274, 8315, 8355, 8396, 8436, 8475, 8515, 8554, 8592, 8631, 8669,
- 8707, 8745, 8783, 8820, 8857, 8894, 8931, 8967, 9003, 9039, 9075, 9111, 9146, 9181,
- 9216, 9251, 9286, 9320, 9354, 9388, 9422, 9456, 9489, 9523, 9556, 9589, 9622, 9655,
- 9687, 9719, 9752, 9784, 9816, 9848, 9879, 9911, 9942, 9973, 10004, 10035, 10066,
- 10097, 10128, 10158, 10188, 10218, 10249, 10279, 10308, 10338, 10368, 10397, 10426,
- 10456, 10485, 10514, 10543, 10572, 10600, 10629, 10657, 10686, 10714, 10742, 10770,
- 10798, 10826, 10854, 10882, 10847, 10810, 10774, 10737, 10701, 10666, 10631, 10596,
- 10562, 10527, 10494, 10460, 10427, 10394, 10362, 10329, 10297, 10266, 10235, 10203,
- 10173, 10142, 10112, 10082, 10052, 10023, 9994, 9965, 9936, 9908, 9879, 9851, 9824,
- 9796, 9769, 9742, 9715, 9689, 9662, 9636, 9610, 9584, 9559, 9534, 9508, 9484, 9459,
- 9434, 9410, 9386, 9362, 9338, 9314, 9291, 9268, 9245, 9222, 9199, 9176, 9154, 9132,
- 9110, 9088, 9066, 9044, 9023, 9002, 8980, 8959, 8939, 8918, 8897, 8877, 8857, 8836,
- 8816, 8796, 8777, 8757, 8738, 8718, 8699, 8680, 8661, 8642, 8623, 8605, 8586, 8568,
- 8550, 8532, 8514, 8496, 8478, 8460, 8443, 8425, 8408, 8391, 8373, 8356, 8339, 8323,
- 8306, 8289, 8273, 8256, 8240, 8224, 8208, 8192
-};
-
-// For Factor2 tables
-// original floating point code
-// if (gain > blim) {
-// factor2 = 1.0;
-// } else {
-// factor2 = 1.0 - 0.3 * (blim - gain);
-// if (gain <= inst->denoiseBound) {
-// factor2 = 1.0 - 0.3 * (blim - inst->denoiseBound);
-// }
-// }
-//
-// Gain factor table: Input value in Q8 and output value in Q13
-static const WebRtc_Word16 kFactor2Aggressiveness1[257] = {
- 7577, 7577, 7577, 7577, 7577, 7577,
- 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7577, 7596, 7614, 7632,
- 7650, 7667, 7683, 7699, 7715, 7731, 7746, 7761, 7775, 7790, 7804, 7818, 7832, 7845,
- 7858, 7871, 7884, 7897, 7910, 7922, 7934, 7946, 7958, 7970, 7982, 7993, 8004, 8016,
- 8027, 8038, 8049, 8060, 8070, 8081, 8091, 8102, 8112, 8122, 8132, 8143, 8152, 8162,
- 8172, 8182, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192
-};
-
-// Gain factor table: Input value in Q8 and output value in Q13
-static const WebRtc_Word16 kFactor2Aggressiveness2[257] = {
- 7270, 7270, 7270, 7270, 7270, 7306,
- 7339, 7369, 7397, 7424, 7448, 7472, 7495, 7517, 7537, 7558, 7577, 7596, 7614, 7632,
- 7650, 7667, 7683, 7699, 7715, 7731, 7746, 7761, 7775, 7790, 7804, 7818, 7832, 7845,
- 7858, 7871, 7884, 7897, 7910, 7922, 7934, 7946, 7958, 7970, 7982, 7993, 8004, 8016,
- 8027, 8038, 8049, 8060, 8070, 8081, 8091, 8102, 8112, 8122, 8132, 8143, 8152, 8162,
- 8172, 8182, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192
-};
-
-// Gain factor table: Input value in Q8 and output value in Q13
-static const WebRtc_Word16 kFactor2Aggressiveness3[257] = {
- 7184, 7184, 7184, 7229, 7270, 7306,
- 7339, 7369, 7397, 7424, 7448, 7472, 7495, 7517, 7537, 7558, 7577, 7596, 7614, 7632,
- 7650, 7667, 7683, 7699, 7715, 7731, 7746, 7761, 7775, 7790, 7804, 7818, 7832, 7845,
- 7858, 7871, 7884, 7897, 7910, 7922, 7934, 7946, 7958, 7970, 7982, 7993, 8004, 8016,
- 8027, 8038, 8049, 8060, 8070, 8081, 8091, 8102, 8112, 8122, 8132, 8143, 8152, 8162,
- 8172, 8182, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192,
- 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192, 8192
-};
-
-// sum of log2(i) from table index to inst->anaLen2 in Q5
-// Note that the first table value is invalid, since log2(0) = -infinity
-static const WebRtc_Word16 kSumLogIndex[66] = {
- 0, 22917, 22917, 22885, 22834, 22770, 22696, 22613,
- 22524, 22428, 22326, 22220, 22109, 21994, 21876, 21754,
- 21629, 21501, 21370, 21237, 21101, 20963, 20822, 20679,
- 20535, 20388, 20239, 20089, 19937, 19783, 19628, 19470,
- 19312, 19152, 18991, 18828, 18664, 18498, 18331, 18164,
- 17994, 17824, 17653, 17480, 17306, 17132, 16956, 16779,
- 16602, 16423, 16243, 16063, 15881, 15699, 15515, 15331,
- 15146, 14960, 14774, 14586, 14398, 14209, 14019, 13829,
- 13637, 13445
-};
-
-// sum of log2(i)^2 from table index to inst->anaLen2 in Q2
-// Note that the first table value is invalid, since log2(0) = -infinity
-static const WebRtc_Word16 kSumSquareLogIndex[66] = {
- 0, 16959, 16959, 16955, 16945, 16929, 16908, 16881,
- 16850, 16814, 16773, 16729, 16681, 16630, 16575, 16517,
- 16456, 16392, 16325, 16256, 16184, 16109, 16032, 15952,
- 15870, 15786, 15700, 15612, 15521, 15429, 15334, 15238,
- 15140, 15040, 14938, 14834, 14729, 14622, 14514, 14404,
- 14292, 14179, 14064, 13947, 13830, 13710, 13590, 13468,
- 13344, 13220, 13094, 12966, 12837, 12707, 12576, 12444,
- 12310, 12175, 12039, 11902, 11763, 11624, 11483, 11341,
- 11198, 11054
-};
-
-// log2(table index) in Q12
-// Note that the first table value is invalid, since log2(0) = -infinity
-static const WebRtc_Word16 kLogIndex[129] = {
- 0, 0, 4096, 6492, 8192, 9511, 10588, 11499,
- 12288, 12984, 13607, 14170, 14684, 15157, 15595, 16003,
- 16384, 16742, 17080, 17400, 17703, 17991, 18266, 18529,
- 18780, 19021, 19253, 19476, 19691, 19898, 20099, 20292,
- 20480, 20662, 20838, 21010, 21176, 21338, 21496, 21649,
- 21799, 21945, 22087, 22226, 22362, 22495, 22625, 22752,
- 22876, 22998, 23117, 23234, 23349, 23462, 23572, 23680,
- 23787, 23892, 23994, 24095, 24195, 24292, 24388, 24483,
- 24576, 24668, 24758, 24847, 24934, 25021, 25106, 25189,
- 25272, 25354, 25434, 25513, 25592, 25669, 25745, 25820,
- 25895, 25968, 26041, 26112, 26183, 26253, 26322, 26390,
- 26458, 26525, 26591, 26656, 26721, 26784, 26848, 26910,
- 26972, 27033, 27094, 27154, 27213, 27272, 27330, 27388,
- 27445, 27502, 27558, 27613, 27668, 27722, 27776, 27830,
- 27883, 27935, 27988, 28039, 28090, 28141, 28191, 28241,
- 28291, 28340, 28388, 28437, 28484, 28532, 28579, 28626,
- 28672
-};
-
-// determinant of estimation matrix in Q0 corresponding to the log2 tables above
-// Note that the first table value is invalid, since log2(0) = -infinity
-static const WebRtc_Word16 kDeterminantEstMatrix[66] = {
- 0, 29814, 25574, 22640, 20351, 18469, 16873, 15491,
- 14277, 13199, 12233, 11362, 10571, 9851, 9192, 8587,
- 8030, 7515, 7038, 6596, 6186, 5804, 5448, 5115,
- 4805, 4514, 4242, 3988, 3749, 3524, 3314, 3116,
- 2930, 2755, 2590, 2435, 2289, 2152, 2022, 1900,
- 1785, 1677, 1575, 1478, 1388, 1302, 1221, 1145,
- 1073, 1005, 942, 881, 825, 771, 721, 674,
- 629, 587, 547, 510, 475, 442, 411, 382,
- 355, 330
-};
-
-// Declare function pointers.
-NoiseEstimation WebRtcNsx_NoiseEstimation;
-PrepareSpectrum WebRtcNsx_PrepareSpectrum;
-SynthesisUpdate WebRtcNsx_SynthesisUpdate;
-AnalysisUpdate WebRtcNsx_AnalysisUpdate;
-Denormalize WebRtcNsx_Denormalize;
-CreateComplexBuffer WebRtcNsx_CreateComplexBuffer;
-
-// Update the noise estimation information.
-static void UpdateNoiseEstimate(NsxInst_t* inst, int offset) {
- WebRtc_Word32 tmp32no1 = 0;
- WebRtc_Word32 tmp32no2 = 0;
- WebRtc_Word16 tmp16 = 0;
- const WebRtc_Word16 kExp2Const = 11819; // Q13
-
- int i = 0;
-
- tmp16 = WebRtcSpl_MaxValueW16(inst->noiseEstLogQuantile + offset,
- inst->magnLen);
- // Guarantee a Q-domain as high as possible and still fit in int16
- inst->qNoise = 14 - (int) WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
- kExp2Const, tmp16, 21);
- for (i = 0; i < inst->magnLen; i++) {
- // inst->quantile[i]=exp(inst->lquantile[offset+i]);
- // in Q21
- tmp32no2 = WEBRTC_SPL_MUL_16_16(kExp2Const,
- inst->noiseEstLogQuantile[offset + i]);
- tmp32no1 = (0x00200000 | (tmp32no2 & 0x001FFFFF)); // 2^21 + frac
- tmp16 = (WebRtc_Word16) WEBRTC_SPL_RSHIFT_W32(tmp32no2, 21);
- tmp16 -= 21;// shift 21 to get result in Q0
- tmp16 += (WebRtc_Word16) inst->qNoise; //shift to get result in Q(qNoise)
- if (tmp16 < 0) {
- tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1, -tmp16);
- } else {
- tmp32no1 = WEBRTC_SPL_LSHIFT_W32(tmp32no1, tmp16);
- }
- inst->noiseEstQuantile[i] = WebRtcSpl_SatW32ToW16(tmp32no1);
- }
-}
-
-// Noise Estimation
-static void NoiseEstimationC(NsxInst_t* inst,
- uint16_t* magn,
- uint32_t* noise,
- int16_t* q_noise) {
- WebRtc_Word16 lmagn[HALF_ANAL_BLOCKL], counter, countDiv;
- WebRtc_Word16 countProd, delta, zeros, frac;
- WebRtc_Word16 log2, tabind, logval, tmp16, tmp16no1, tmp16no2;
- const int16_t log2_const = 22713; // Q15
- const int16_t width_factor = 21845;
-
- int i, s, offset;
-
- tabind = inst->stages - inst->normData;
- assert(tabind < 9);
- assert(tabind > -9);
- if (tabind < 0) {
- logval = -WebRtcNsx_kLogTable[-tabind];
- } else {
- logval = WebRtcNsx_kLogTable[tabind];
- }
-
- // lmagn(i)=log(magn(i))=log(2)*log2(magn(i))
- // magn is in Q(-stages), and the real lmagn values are:
- // real_lmagn(i)=log(magn(i)*2^stages)=log(magn(i))+log(2^stages)
- // lmagn in Q8
- for (i = 0; i < inst->magnLen; i++) {
- if (magn[i]) {
- zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magn[i]);
- frac = (WebRtc_Word16)((((WebRtc_UWord32)magn[i] << zeros)
- & 0x7FFFFFFF) >> 23);
- // log2(magn(i))
- assert(frac < 256);
- log2 = (WebRtc_Word16)(((31 - zeros) << 8)
- + WebRtcNsx_kLogTableFrac[frac]);
- // log2(magn(i))*log(2)
- lmagn[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(log2, log2_const, 15);
- // + log(2^stages)
- lmagn[i] += logval;
- } else {
- lmagn[i] = logval;//0;
- }
- }
-
- // loop over simultaneous estimates
- for (s = 0; s < SIMULT; s++) {
- offset = s * inst->magnLen;
-
- // Get counter values from state
- counter = inst->noiseEstCounter[s];
- assert(counter < 201);
- countDiv = WebRtcNsx_kCounterDiv[counter];
- countProd = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16(counter, countDiv);
-
- // quant_est(...)
- for (i = 0; i < inst->magnLen; i++) {
- // compute delta
- if (inst->noiseEstDensity[offset + i] > 512) {
- // Get the value for delta by shifting intead of dividing.
- int factor = WebRtcSpl_NormW16(inst->noiseEstDensity[offset + i]);
- delta = (int16_t)(FACTOR_Q16 >> (14 - factor));
- } else {
- delta = FACTOR_Q7;
- if (inst->blockIndex < END_STARTUP_LONG) {
- // Smaller step size during startup. This prevents from using
- // unrealistic values causing overflow.
- delta = FACTOR_Q7_STARTUP;
- }
- }
-
- // update log quantile estimate
- tmp16 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(delta, countDiv, 14);
- if (lmagn[i] > inst->noiseEstLogQuantile[offset + i]) {
- // +=QUANTILE*delta/(inst->counter[s]+1) QUANTILE=0.25, =1 in Q2
- // CounterDiv=1/(inst->counter[s]+1) in Q15
- tmp16 += 2;
- tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 2);
- inst->noiseEstLogQuantile[offset + i] += tmp16no1;
- } else {
- tmp16 += 1;
- tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 1);
- // *(1-QUANTILE), in Q2 QUANTILE=0.25, 1-0.25=0.75=3 in Q2
- tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, 3, 1);
- inst->noiseEstLogQuantile[offset + i] -= tmp16no2;
- if (inst->noiseEstLogQuantile[offset + i] < logval) {
- // This is the smallest fixed point representation we can
- // have, hence we limit the output.
- inst->noiseEstLogQuantile[offset + i] = logval;
- }
- }
-
- // update density estimate
- if (WEBRTC_SPL_ABS_W16(lmagn[i] - inst->noiseEstLogQuantile[offset + i])
- < WIDTH_Q8) {
- tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
- inst->noiseEstDensity[offset + i], countProd, 15);
- tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
- width_factor, countDiv, 15);
- inst->noiseEstDensity[offset + i] = tmp16no1 + tmp16no2;
- }
- } // end loop over magnitude spectrum
-
- if (counter >= END_STARTUP_LONG) {
- inst->noiseEstCounter[s] = 0;
- if (inst->blockIndex >= END_STARTUP_LONG) {
- UpdateNoiseEstimate(inst, offset);
- }
- }
- inst->noiseEstCounter[s]++;
-
- } // end loop over simultaneous estimates
-
- // Sequentially update the noise during startup
- if (inst->blockIndex < END_STARTUP_LONG) {
- UpdateNoiseEstimate(inst, offset);
- }
-
- for (i = 0; i < inst->magnLen; i++) {
- noise[i] = (WebRtc_UWord32)(inst->noiseEstQuantile[i]); // Q(qNoise)
- }
- (*q_noise) = (WebRtc_Word16)inst->qNoise;
-}
-
-// Filter the data in the frequency domain, and create spectrum.
-static void PrepareSpectrumC(NsxInst_t* inst, int16_t* freq_buf) {
- int i = 0, j = 0;
- int16_t tmp16 = 0;
-
- for (i = 0; i < inst->magnLen; i++) {
- inst->real[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->real[i],
- (WebRtc_Word16)(inst->noiseSupFilter[i]), 14); // Q(normData-stages)
- inst->imag[i] = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->imag[i],
- (WebRtc_Word16)(inst->noiseSupFilter[i]), 14); // Q(normData-stages)
- }
-
- freq_buf[0] = inst->real[0];
- freq_buf[1] = -inst->imag[0];
- for (i = 1, j = 2; i < inst->anaLen2; i += 1, j += 2) {
- tmp16 = (inst->anaLen << 1) - j;
- freq_buf[j] = inst->real[i];
- freq_buf[j + 1] = -inst->imag[i];
- freq_buf[tmp16] = inst->real[i];
- freq_buf[tmp16 + 1] = inst->imag[i];
- }
- freq_buf[inst->anaLen] = inst->real[inst->anaLen2];
- freq_buf[inst->anaLen + 1] = -inst->imag[inst->anaLen2];
-}
-
-// Denormalize the input buffer.
-static __inline void DenormalizeC(NsxInst_t* inst, int16_t* in, int factor) {
- int i = 0, j = 0;
- int32_t tmp32 = 0;
- for (i = 0, j = 0; i < inst->anaLen; i += 1, j += 2) {
- tmp32 = WEBRTC_SPL_SHIFT_W32((WebRtc_Word32)in[j],
- factor - inst->normData);
- inst->real[i] = WebRtcSpl_SatW32ToW16(tmp32); // Q0
- }
-}
-
-// For the noise supression process, synthesis, read out fully processed
-// segment, and update synthesis buffer.
-static void SynthesisUpdateC(NsxInst_t* inst,
- int16_t* out_frame,
- int16_t gain_factor) {
- int i = 0;
- int16_t tmp16a = 0;
- int16_t tmp16b = 0;
- int32_t tmp32 = 0;
-
- // synthesis
- for (i = 0; i < inst->anaLen; i++) {
- tmp16a = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
- inst->window[i], inst->real[i], 14); // Q0, window in Q14
- tmp32 = WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(tmp16a, gain_factor, 13); // Q0
- // Down shift with rounding
- tmp16b = WebRtcSpl_SatW32ToW16(tmp32); // Q0
- inst->synthesisBuffer[i] = WEBRTC_SPL_ADD_SAT_W16(inst->synthesisBuffer[i],
- tmp16b); // Q0
- }
-
- // read out fully processed segment
- for (i = 0; i < inst->blockLen10ms; i++) {
- out_frame[i] = inst->synthesisBuffer[i]; // Q0
- }
-
- // update synthesis buffer
- WEBRTC_SPL_MEMCPY_W16(inst->synthesisBuffer,
- inst->synthesisBuffer + inst->blockLen10ms,
- inst->anaLen - inst->blockLen10ms);
- WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer
- + inst->anaLen - inst->blockLen10ms, inst->blockLen10ms);
-}
-
-// Update analysis buffer for lower band, and window data before FFT.
-static void AnalysisUpdateC(NsxInst_t* inst,
- int16_t* out,
- int16_t* new_speech) {
- int i = 0;
-
- // For lower band update analysis buffer.
- WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer,
- inst->analysisBuffer + inst->blockLen10ms,
- inst->anaLen - inst->blockLen10ms);
- WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer
- + inst->anaLen - inst->blockLen10ms, new_speech, inst->blockLen10ms);
-
- // Window data before FFT.
- for (i = 0; i < inst->anaLen; i++) {
- out[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
- inst->window[i], inst->analysisBuffer[i], 14); // Q0
- }
-}
-
-// Create a complex number buffer (out[]) as the intput (in[]) interleaved with
-// zeros, and normalize it.
-static __inline void CreateComplexBufferC(NsxInst_t* inst,
- int16_t* in,
- int16_t* out) {
- int i = 0, j = 0;
- for (i = 0, j = 0; i < inst->anaLen; i += 1, j += 2) {
- out[j] = WEBRTC_SPL_LSHIFT_W16(in[i], inst->normData); // Q(normData)
- out[j + 1] = 0; // Insert zeros in imaginary part
- }
-}
-
-void WebRtcNsx_CalcParametricNoiseEstimate(NsxInst_t* inst,
- WebRtc_Word16 pink_noise_exp_avg,
- WebRtc_Word32 pink_noise_num_avg,
- int freq_index,
- WebRtc_UWord32* noise_estimate,
- WebRtc_UWord32* noise_estimate_avg) {
- WebRtc_Word32 tmp32no1 = 0;
- WebRtc_Word32 tmp32no2 = 0;
-
- WebRtc_Word16 int_part = 0;
- WebRtc_Word16 frac_part = 0;
-
- // Use pink noise estimate
- // noise_estimate = 2^(pinkNoiseNumerator + pinkNoiseExp * log2(j))
- assert(freq_index >= 0);
- assert(freq_index < 129);
- tmp32no2 = WEBRTC_SPL_MUL_16_16(pink_noise_exp_avg, kLogIndex[freq_index]); // Q26
- tmp32no2 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, 15); // Q11
- tmp32no1 = pink_noise_num_avg - tmp32no2; // Q11
-
- // Calculate output: 2^tmp32no1
- // Output in Q(minNorm-stages)
- tmp32no1 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)(inst->minNorm - inst->stages), 11);
- if (tmp32no1 > 0) {
- int_part = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 11);
- frac_part = (WebRtc_Word16)(tmp32no1 & 0x000007ff); // Q11
- // Piecewise linear approximation of 'b' in
- // 2^(int_part+frac_part) = 2^int_part * (1 + b)
- // 'b' is given in Q11 and below stored in frac_part.
- if (WEBRTC_SPL_RSHIFT_W16(frac_part, 10)) {
- // Upper fractional part
- tmp32no2 = WEBRTC_SPL_MUL_16_16(2048 - frac_part, 1244); // Q21
- tmp32no2 = 2048 - WEBRTC_SPL_RSHIFT_W32(tmp32no2, 10);
- } else {
- // Lower fractional part
- tmp32no2 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_16_16(frac_part, 804), 10);
- }
- // Shift fractional part to Q(minNorm-stages)
- tmp32no2 = WEBRTC_SPL_SHIFT_W32(tmp32no2, int_part - 11);
- *noise_estimate_avg = WEBRTC_SPL_LSHIFT_U32(1, int_part) + (WebRtc_UWord32)tmp32no2;
- // Scale up to initMagnEst, which is not block averaged
- *noise_estimate = (*noise_estimate_avg) * (WebRtc_UWord32)(inst->blockIndex + 1);
- }
-}
-
-// Initialize state
-WebRtc_Word32 WebRtcNsx_InitCore(NsxInst_t* inst, WebRtc_UWord32 fs) {
- int i;
-
- //check for valid pointer
- if (inst == NULL) {
- return -1;
- }
- //
-
- // Initialization of struct
- if (fs == 8000 || fs == 16000 || fs == 32000) {
- inst->fs = fs;
- } else {
- return -1;
- }
-
- if (fs == 8000) {
- inst->blockLen10ms = 80;
- inst->anaLen = 128;
- inst->stages = 7;
- inst->window = kBlocks80w128x;
- inst->thresholdLogLrt = 131072; //default threshold for LRT feature
- inst->maxLrt = 0x0040000;
- inst->minLrt = 52429;
- } else if (fs == 16000) {
- inst->blockLen10ms = 160;
- inst->anaLen = 256;
- inst->stages = 8;
- inst->window = kBlocks160w256x;
- inst->thresholdLogLrt = 212644; //default threshold for LRT feature
- inst->maxLrt = 0x0080000;
- inst->minLrt = 104858;
- } else if (fs == 32000) {
- inst->blockLen10ms = 160;
- inst->anaLen = 256;
- inst->stages = 8;
- inst->window = kBlocks160w256x;
- inst->thresholdLogLrt = 212644; //default threshold for LRT feature
- inst->maxLrt = 0x0080000;
- inst->minLrt = 104858;
- }
- inst->anaLen2 = WEBRTC_SPL_RSHIFT_W16(inst->anaLen, 1);
- inst->magnLen = inst->anaLen2 + 1;
-
- WebRtcSpl_ZerosArrayW16(inst->analysisBuffer, ANAL_BLOCKL_MAX);
- WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer, ANAL_BLOCKL_MAX);
-
- // for HB processing
- WebRtcSpl_ZerosArrayW16(inst->dataBufHBFX, ANAL_BLOCKL_MAX);
- // for quantile noise estimation
- WebRtcSpl_ZerosArrayW16(inst->noiseEstQuantile, HALF_ANAL_BLOCKL);
- for (i = 0; i < SIMULT * HALF_ANAL_BLOCKL; i++) {
- inst->noiseEstLogQuantile[i] = 2048; // Q8
- inst->noiseEstDensity[i] = 153; // Q9
- }
- for (i = 0; i < SIMULT; i++) {
- inst->noiseEstCounter[i] = (WebRtc_Word16)(END_STARTUP_LONG * (i + 1)) / SIMULT;
- }
-
- // Initialize suppression filter with ones
- WebRtcSpl_MemSetW16((WebRtc_Word16*)inst->noiseSupFilter, 16384, HALF_ANAL_BLOCKL);
-
- // Set the aggressiveness: default
- inst->aggrMode = 0;
-
- //initialize variables for new method
- inst->priorNonSpeechProb = 8192; // Q14(0.5) prior probability for speech/noise
- for (i = 0; i < HALF_ANAL_BLOCKL; i++) {
- inst->prevMagnU16[i] = 0;
- inst->prevNoiseU32[i] = 0; //previous noise-spectrum
- inst->logLrtTimeAvgW32[i] = 0; //smooth LR ratio
- inst->avgMagnPause[i] = 0; //conservative noise spectrum estimate
- inst->initMagnEst[i] = 0; //initial average magnitude spectrum
- }
-
- //feature quantities
- inst->thresholdSpecDiff = 50; //threshold for difference feature: determined on-line
- inst->thresholdSpecFlat = 20480; //threshold for flatness: determined on-line
- inst->featureLogLrt = inst->thresholdLogLrt; //average LRT factor (= threshold)
- inst->featureSpecFlat = inst->thresholdSpecFlat; //spectral flatness (= threshold)
- inst->featureSpecDiff = inst->thresholdSpecDiff; //spectral difference (= threshold)
- inst->weightLogLrt = 6; //default weighting par for LRT feature
- inst->weightSpecFlat = 0; //default weighting par for spectral flatness feature
- inst->weightSpecDiff = 0; //default weighting par for spectral difference feature
-
- inst->curAvgMagnEnergy = 0; //window time-average of input magnitude spectrum
- inst->timeAvgMagnEnergy = 0; //normalization for spectral difference
- inst->timeAvgMagnEnergyTmp = 0; //normalization for spectral difference
-
- //histogram quantities: used to estimate/update thresholds for features
- WebRtcSpl_ZerosArrayW16(inst->histLrt, HIST_PAR_EST);
- WebRtcSpl_ZerosArrayW16(inst->histSpecDiff, HIST_PAR_EST);
- WebRtcSpl_ZerosArrayW16(inst->histSpecFlat, HIST_PAR_EST);
-
- inst->blockIndex = -1; //frame counter
-
- //inst->modelUpdate = 500; //window for update
- inst->modelUpdate = (1 << STAT_UPDATES); //window for update
- inst->cntThresUpdate = 0; //counter feature thresholds updates
-
- inst->sumMagn = 0;
- inst->magnEnergy = 0;
- inst->prevQMagn = 0;
- inst->qNoise = 0;
- inst->prevQNoise = 0;
-
- inst->energyIn = 0;
- inst->scaleEnergyIn = 0;
-
- inst->whiteNoiseLevel = 0;
- inst->pinkNoiseNumerator = 0;
- inst->pinkNoiseExp = 0;
- inst->minNorm = 15; // Start with full scale
- inst->zeroInputSignal = 0;
-
- //default mode
- WebRtcNsx_set_policy_core(inst, 0);
-
-#ifdef NS_FILEDEBUG
- inst->infile = fopen("indebug.pcm", "wb");
- inst->outfile = fopen("outdebug.pcm", "wb");
- inst->file1 = fopen("file1.pcm", "wb");
- inst->file2 = fopen("file2.pcm", "wb");
- inst->file3 = fopen("file3.pcm", "wb");
- inst->file4 = fopen("file4.pcm", "wb");
- inst->file5 = fopen("file5.pcm", "wb");
-#endif
-
- // Initialize function pointers.
- WebRtcNsx_NoiseEstimation = NoiseEstimationC;
- WebRtcNsx_PrepareSpectrum = PrepareSpectrumC;
- WebRtcNsx_SynthesisUpdate = SynthesisUpdateC;
- WebRtcNsx_AnalysisUpdate = AnalysisUpdateC;
- WebRtcNsx_Denormalize = DenormalizeC;
- WebRtcNsx_CreateComplexBuffer = CreateComplexBufferC;
-
-#ifdef WEBRTC_DETECT_ARM_NEON
- uint64_t features = WebRtc_GetCPUFeaturesARM();
- if ((features & kCPUFeatureNEON) != 0)
- {
- WebRtcNsx_InitNeon();
- }
-#elif defined(WEBRTC_ARCH_ARM_NEON)
- WebRtcNsx_InitNeon();
-#endif
-
- inst->initFlag = 1;
-
- return 0;
-}
-
-int WebRtcNsx_set_policy_core(NsxInst_t* inst, int mode) {
- // allow for modes:0,1,2,3
- if (mode < 0 || mode > 3) {
- return -1;
- }
-
- inst->aggrMode = mode;
- if (mode == 0) {
- inst->overdrive = 256; // Q8(1.0)
- inst->denoiseBound = 8192; // Q14(0.5)
- inst->gainMap = 0; // No gain compensation
- } else if (mode == 1) {
- inst->overdrive = 256; // Q8(1.0)
- inst->denoiseBound = 4096; // Q14(0.25)
- inst->factor2Table = kFactor2Aggressiveness1;
- inst->gainMap = 1;
- } else if (mode == 2) {
- inst->overdrive = 282; // ~= Q8(1.1)
- inst->denoiseBound = 2048; // Q14(0.125)
- inst->factor2Table = kFactor2Aggressiveness2;
- inst->gainMap = 1;
- } else if (mode == 3) {
- inst->overdrive = 320; // Q8(1.25)
- inst->denoiseBound = 1475; // ~= Q14(0.09)
- inst->factor2Table = kFactor2Aggressiveness3;
- inst->gainMap = 1;
- }
- return 0;
-}
-
-// Extract thresholds for feature parameters
-// histograms are computed over some window_size (given by window_pars)
-// thresholds and weights are extracted every window
-// flag 0 means update histogram only, flag 1 means compute the thresholds/weights
-// threshold and weights are returned in: inst->priorModelPars
-void WebRtcNsx_FeatureParameterExtraction(NsxInst_t* inst, int flag) {
- WebRtc_UWord32 tmpU32;
- WebRtc_UWord32 histIndex;
- WebRtc_UWord32 posPeak1SpecFlatFX, posPeak2SpecFlatFX;
- WebRtc_UWord32 posPeak1SpecDiffFX, posPeak2SpecDiffFX;
-
- WebRtc_Word32 tmp32;
- WebRtc_Word32 fluctLrtFX, thresFluctLrtFX;
- WebRtc_Word32 avgHistLrtFX, avgSquareHistLrtFX, avgHistLrtComplFX;
-
- WebRtc_Word16 j;
- WebRtc_Word16 numHistLrt;
-
- int i;
- int useFeatureSpecFlat, useFeatureSpecDiff, featureSum;
- int maxPeak1, maxPeak2;
- int weightPeak1SpecFlat, weightPeak2SpecFlat;
- int weightPeak1SpecDiff, weightPeak2SpecDiff;
-
- //update histograms
- if (!flag) {
- // LRT
- // Type casting to UWord32 is safe since negative values will not be wrapped to larger
- // values than HIST_PAR_EST
- histIndex = (WebRtc_UWord32)(inst->featureLogLrt);
- if (histIndex < HIST_PAR_EST) {
- inst->histLrt[histIndex]++;
- }
- // Spectral flatness
- // (inst->featureSpecFlat*20)>>10 = (inst->featureSpecFlat*5)>>8
- histIndex = WEBRTC_SPL_RSHIFT_U32(inst->featureSpecFlat * 5, 8);
- if (histIndex < HIST_PAR_EST) {
- inst->histSpecFlat[histIndex]++;
- }
- // Spectral difference
- histIndex = HIST_PAR_EST;
- if (inst->timeAvgMagnEnergy > 0) {
- // Guard against division by zero
- // If timeAvgMagnEnergy == 0 we have no normalizing statistics and
- // therefore can't update the histogram
- histIndex = WEBRTC_SPL_UDIV((inst->featureSpecDiff * 5) >> inst->stages,
- inst->timeAvgMagnEnergy);
- }
- if (histIndex < HIST_PAR_EST) {
- inst->histSpecDiff[histIndex]++;
- }
- }
-
- // extract parameters for speech/noise probability
- if (flag) {
- useFeatureSpecDiff = 1;
- //for LRT feature:
- // compute the average over inst->featureExtractionParams.rangeAvgHistLrt
- avgHistLrtFX = 0;
- avgSquareHistLrtFX = 0;
- numHistLrt = 0;
- for (i = 0; i < BIN_SIZE_LRT; i++) {
- j = (2 * i + 1);
- tmp32 = WEBRTC_SPL_MUL_16_16(inst->histLrt[i], j);
- avgHistLrtFX += tmp32;
- numHistLrt += inst->histLrt[i];
- avgSquareHistLrtFX += WEBRTC_SPL_MUL_32_16(tmp32, j);
- }
- avgHistLrtComplFX = avgHistLrtFX;
- for (; i < HIST_PAR_EST; i++) {
- j = (2 * i + 1);
- tmp32 = WEBRTC_SPL_MUL_16_16(inst->histLrt[i], j);
- avgHistLrtComplFX += tmp32;
- avgSquareHistLrtFX += WEBRTC_SPL_MUL_32_16(tmp32, j);
- }
- fluctLrtFX = WEBRTC_SPL_MUL(avgSquareHistLrtFX, numHistLrt);
- fluctLrtFX -= WEBRTC_SPL_MUL(avgHistLrtFX, avgHistLrtComplFX);
- thresFluctLrtFX = THRES_FLUCT_LRT * numHistLrt;
- // get threshold for LRT feature:
- tmpU32 = (FACTOR_1_LRT_DIFF * (WebRtc_UWord32)avgHistLrtFX);
- if ((fluctLrtFX < thresFluctLrtFX) || (numHistLrt == 0) ||
- (tmpU32 > (WebRtc_UWord32)(100 * numHistLrt))) {
- //very low fluctuation, so likely noise
- inst->thresholdLogLrt = inst->maxLrt;
- } else {
- tmp32 = (WebRtc_Word32)((tmpU32 << (9 + inst->stages)) / numHistLrt /
- 25);
- // check if value is within min/max range
- inst->thresholdLogLrt = WEBRTC_SPL_SAT(inst->maxLrt,
- tmp32,
- inst->minLrt);
- }
- if (fluctLrtFX < thresFluctLrtFX) {
- // Do not use difference feature if fluctuation of LRT feature is very low:
- // most likely just noise state
- useFeatureSpecDiff = 0;
- }
-
- // for spectral flatness and spectral difference: compute the main peaks of histogram
- maxPeak1 = 0;
- maxPeak2 = 0;
- posPeak1SpecFlatFX = 0;
- posPeak2SpecFlatFX = 0;
- weightPeak1SpecFlat = 0;
- weightPeak2SpecFlat = 0;
-
- // peaks for flatness
- for (i = 0; i < HIST_PAR_EST; i++) {
- if (inst->histSpecFlat[i] > maxPeak1) {
- // Found new "first" peak
- maxPeak2 = maxPeak1;
- weightPeak2SpecFlat = weightPeak1SpecFlat;
- posPeak2SpecFlatFX = posPeak1SpecFlatFX;
-
- maxPeak1 = inst->histSpecFlat[i];
- weightPeak1SpecFlat = inst->histSpecFlat[i];
- posPeak1SpecFlatFX = (WebRtc_UWord32)(2 * i + 1);
- } else if (inst->histSpecFlat[i] > maxPeak2) {
- // Found new "second" peak
- maxPeak2 = inst->histSpecFlat[i];
- weightPeak2SpecFlat = inst->histSpecFlat[i];
- posPeak2SpecFlatFX = (WebRtc_UWord32)(2 * i + 1);
- }
- }
-
- // for spectral flatness feature
- useFeatureSpecFlat = 1;
- // merge the two peaks if they are close
- if ((posPeak1SpecFlatFX - posPeak2SpecFlatFX < LIM_PEAK_SPACE_FLAT_DIFF)
- && (weightPeak2SpecFlat * LIM_PEAK_WEIGHT_FLAT_DIFF > weightPeak1SpecFlat)) {
- weightPeak1SpecFlat += weightPeak2SpecFlat;
- posPeak1SpecFlatFX = (posPeak1SpecFlatFX + posPeak2SpecFlatFX) >> 1;
- }
- //reject if weight of peaks is not large enough, or peak value too small
- if (weightPeak1SpecFlat < THRES_WEIGHT_FLAT_DIFF || posPeak1SpecFlatFX
- < THRES_PEAK_FLAT) {
- useFeatureSpecFlat = 0;
- } else { // if selected, get the threshold
- // compute the threshold and check if value is within min/max range
- inst->thresholdSpecFlat = WEBRTC_SPL_SAT(MAX_FLAT_Q10, FACTOR_2_FLAT_Q10
- * posPeak1SpecFlatFX, MIN_FLAT_Q10); //Q10
- }
- // done with flatness feature
-
- if (useFeatureSpecDiff) {
- //compute two peaks for spectral difference
- maxPeak1 = 0;
- maxPeak2 = 0;
- posPeak1SpecDiffFX = 0;
- posPeak2SpecDiffFX = 0;
- weightPeak1SpecDiff = 0;
- weightPeak2SpecDiff = 0;
- // peaks for spectral difference
- for (i = 0; i < HIST_PAR_EST; i++) {
- if (inst->histSpecDiff[i] > maxPeak1) {
- // Found new "first" peak
- maxPeak2 = maxPeak1;
- weightPeak2SpecDiff = weightPeak1SpecDiff;
- posPeak2SpecDiffFX = posPeak1SpecDiffFX;
-
- maxPeak1 = inst->histSpecDiff[i];
- weightPeak1SpecDiff = inst->histSpecDiff[i];
- posPeak1SpecDiffFX = (WebRtc_UWord32)(2 * i + 1);
- } else if (inst->histSpecDiff[i] > maxPeak2) {
- // Found new "second" peak
- maxPeak2 = inst->histSpecDiff[i];
- weightPeak2SpecDiff = inst->histSpecDiff[i];
- posPeak2SpecDiffFX = (WebRtc_UWord32)(2 * i + 1);
- }
- }
-
- // merge the two peaks if they are close
- if ((posPeak1SpecDiffFX - posPeak2SpecDiffFX < LIM_PEAK_SPACE_FLAT_DIFF)
- && (weightPeak2SpecDiff * LIM_PEAK_WEIGHT_FLAT_DIFF > weightPeak1SpecDiff)) {
- weightPeak1SpecDiff += weightPeak2SpecDiff;
- posPeak1SpecDiffFX = (posPeak1SpecDiffFX + posPeak2SpecDiffFX) >> 1;
- }
- // get the threshold value and check if value is within min/max range
- inst->thresholdSpecDiff = WEBRTC_SPL_SAT(MAX_DIFF, FACTOR_1_LRT_DIFF
- * posPeak1SpecDiffFX, MIN_DIFF); //5x bigger
- //reject if weight of peaks is not large enough
- if (weightPeak1SpecDiff < THRES_WEIGHT_FLAT_DIFF) {
- useFeatureSpecDiff = 0;
- }
- // done with spectral difference feature
- }
-
- // select the weights between the features
- // inst->priorModelPars[4] is weight for LRT: always selected
- featureSum = 6 / (1 + useFeatureSpecFlat + useFeatureSpecDiff);
- inst->weightLogLrt = featureSum;
- inst->weightSpecFlat = useFeatureSpecFlat * featureSum;
- inst->weightSpecDiff = useFeatureSpecDiff * featureSum;
-
- // set histograms to zero for next update
- WebRtcSpl_ZerosArrayW16(inst->histLrt, HIST_PAR_EST);
- WebRtcSpl_ZerosArrayW16(inst->histSpecDiff, HIST_PAR_EST);
- WebRtcSpl_ZerosArrayW16(inst->histSpecFlat, HIST_PAR_EST);
- } // end of flag == 1
-}
-
-
-// Compute spectral flatness on input spectrum
-// magn is the magnitude spectrum
-// spectral flatness is returned in inst->featureSpecFlat
-void WebRtcNsx_ComputeSpectralFlatness(NsxInst_t* inst, WebRtc_UWord16* magn) {
- WebRtc_UWord32 tmpU32;
- WebRtc_UWord32 avgSpectralFlatnessNum, avgSpectralFlatnessDen;
-
- WebRtc_Word32 tmp32;
- WebRtc_Word32 currentSpectralFlatness, logCurSpectralFlatness;
-
- WebRtc_Word16 zeros, frac, intPart;
-
- int i;
-
- // for flatness
- avgSpectralFlatnessNum = 0;
- avgSpectralFlatnessDen = inst->sumMagn - (WebRtc_UWord32)magn[0]; // Q(normData-stages)
-
- // compute log of ratio of the geometric to arithmetic mean: check for log(0) case
- // flatness = exp( sum(log(magn[i]))/N - log(sum(magn[i])/N) )
- // = exp( sum(log(magn[i]))/N ) * N / sum(magn[i])
- // = 2^( sum(log2(magn[i]))/N - (log2(sum(magn[i])) - log2(N)) ) [This is used]
- for (i = 1; i < inst->magnLen; i++) {
- // First bin is excluded from spectrum measures. Number of bins is now a power of 2
- if (magn[i]) {
- zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magn[i]);
- frac = (WebRtc_Word16)(((WebRtc_UWord32)((WebRtc_UWord32)(magn[i]) << zeros)
- & 0x7FFFFFFF) >> 23);
- // log2(magn(i))
- assert(frac < 256);
- tmpU32 = (WebRtc_UWord32)(((31 - zeros) << 8)
- + WebRtcNsx_kLogTableFrac[frac]); // Q8
- avgSpectralFlatnessNum += tmpU32; // Q8
- } else {
- //if at least one frequency component is zero, treat separately
- tmpU32 = WEBRTC_SPL_UMUL_32_16(inst->featureSpecFlat, SPECT_FLAT_TAVG_Q14); // Q24
- inst->featureSpecFlat -= WEBRTC_SPL_RSHIFT_U32(tmpU32, 14); // Q10
- return;
- }
- }
- //ratio and inverse log: check for case of log(0)
- zeros = WebRtcSpl_NormU32(avgSpectralFlatnessDen);
- frac = (WebRtc_Word16)(((avgSpectralFlatnessDen << zeros) & 0x7FFFFFFF) >> 23);
- // log2(avgSpectralFlatnessDen)
- assert(frac < 256);
- tmp32 = (WebRtc_Word32)(((31 - zeros) << 8) + WebRtcNsx_kLogTableFrac[frac]); // Q8
- logCurSpectralFlatness = (WebRtc_Word32)avgSpectralFlatnessNum;
- logCurSpectralFlatness += ((WebRtc_Word32)(inst->stages - 1) << (inst->stages + 7)); // Q(8+stages-1)
- logCurSpectralFlatness -= (tmp32 << (inst->stages - 1));
- logCurSpectralFlatness = WEBRTC_SPL_LSHIFT_W32(logCurSpectralFlatness, 10 - inst->stages); // Q17
- tmp32 = (WebRtc_Word32)(0x00020000 | (WEBRTC_SPL_ABS_W32(logCurSpectralFlatness)
- & 0x0001FFFF)); //Q17
- intPart = -(WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(logCurSpectralFlatness, 17);
- intPart += 7; // Shift 7 to get the output in Q10 (from Q17 = -17+10)
- if (intPart > 0) {
- currentSpectralFlatness = WEBRTC_SPL_RSHIFT_W32(tmp32, intPart);
- } else {
- currentSpectralFlatness = WEBRTC_SPL_LSHIFT_W32(tmp32, -intPart);
- }
-
- //time average update of spectral flatness feature
- tmp32 = currentSpectralFlatness - (WebRtc_Word32)inst->featureSpecFlat; // Q10
- tmp32 = WEBRTC_SPL_MUL_32_16(SPECT_FLAT_TAVG_Q14, tmp32); // Q24
- inst->featureSpecFlat = (WebRtc_UWord32)((WebRtc_Word32)inst->featureSpecFlat
- + WEBRTC_SPL_RSHIFT_W32(tmp32, 14)); // Q10
- // done with flatness feature
-}
-
-
-// Compute the difference measure between input spectrum and a template/learned noise spectrum
-// magn_tmp is the input spectrum
-// the reference/template spectrum is inst->magn_avg_pause[i]
-// returns (normalized) spectral difference in inst->featureSpecDiff
-void WebRtcNsx_ComputeSpectralDifference(NsxInst_t* inst, WebRtc_UWord16* magnIn) {
- // This is to be calculated:
- // avgDiffNormMagn = var(magnIn) - cov(magnIn, magnAvgPause)^2 / var(magnAvgPause)
-
- WebRtc_UWord32 tmpU32no1, tmpU32no2;
- WebRtc_UWord32 varMagnUFX, varPauseUFX, avgDiffNormMagnUFX;
-
- WebRtc_Word32 tmp32no1, tmp32no2;
- WebRtc_Word32 avgPauseFX, avgMagnFX, covMagnPauseFX;
- WebRtc_Word32 maxPause, minPause;
-
- WebRtc_Word16 tmp16no1;
-
- int i, norm32, nShifts;
-
- avgPauseFX = 0;
- maxPause = 0;
- minPause = inst->avgMagnPause[0]; // Q(prevQMagn)
- // compute average quantities
- for (i = 0; i < inst->magnLen; i++) {
- // Compute mean of magn_pause
- avgPauseFX += inst->avgMagnPause[i]; // in Q(prevQMagn)
- maxPause = WEBRTC_SPL_MAX(maxPause, inst->avgMagnPause[i]);
- minPause = WEBRTC_SPL_MIN(minPause, inst->avgMagnPause[i]);
- }
- // normalize by replacing div of "inst->magnLen" with "inst->stages-1" shifts
- avgPauseFX = WEBRTC_SPL_RSHIFT_W32(avgPauseFX, inst->stages - 1);
- avgMagnFX = (WebRtc_Word32)WEBRTC_SPL_RSHIFT_U32(inst->sumMagn, inst->stages - 1);
- // Largest possible deviation in magnPause for (co)var calculations
- tmp32no1 = WEBRTC_SPL_MAX(maxPause - avgPauseFX, avgPauseFX - minPause);
- // Get number of shifts to make sure we don't get wrap around in varPause
- nShifts = WEBRTC_SPL_MAX(0, 10 + inst->stages - WebRtcSpl_NormW32(tmp32no1));
-
- varMagnUFX = 0;
- varPauseUFX = 0;
- covMagnPauseFX = 0;
- for (i = 0; i < inst->magnLen; i++) {
- // Compute var and cov of magn and magn_pause
- tmp16no1 = (WebRtc_Word16)((WebRtc_Word32)magnIn[i] - avgMagnFX);
- tmp32no2 = inst->avgMagnPause[i] - avgPauseFX;
- varMagnUFX += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(tmp16no1, tmp16no1); // Q(2*qMagn)
- tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no2, tmp16no1); // Q(prevQMagn+qMagn)
- covMagnPauseFX += tmp32no1; // Q(prevQMagn+qMagn)
- tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no2, nShifts); // Q(prevQMagn-minPause)
- varPauseUFX += (WebRtc_UWord32)WEBRTC_SPL_MUL(tmp32no1, tmp32no1); // Q(2*(prevQMagn-minPause))
- }
- //update of average magnitude spectrum: Q(-2*stages) and averaging replaced by shifts
- inst->curAvgMagnEnergy += WEBRTC_SPL_RSHIFT_U32(inst->magnEnergy, 2 * inst->normData
- + inst->stages - 1);
-
- avgDiffNormMagnUFX = varMagnUFX; // Q(2*qMagn)
- if ((varPauseUFX) && (covMagnPauseFX)) {
- tmpU32no1 = (WebRtc_UWord32)WEBRTC_SPL_ABS_W32(covMagnPauseFX); // Q(prevQMagn+qMagn)
- norm32 = WebRtcSpl_NormU32(tmpU32no1) - 16;
- if (norm32 > 0) {
- tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, norm32); // Q(prevQMagn+qMagn+norm32)
- } else {
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, -norm32); // Q(prevQMagn+qMagn+norm32)
- }
- tmpU32no2 = WEBRTC_SPL_UMUL(tmpU32no1, tmpU32no1); // Q(2*(prevQMagn+qMagn-norm32))
-
- nShifts += norm32;
- nShifts <<= 1;
- if (nShifts < 0) {
- varPauseUFX >>= (-nShifts); // Q(2*(qMagn+norm32+minPause))
- nShifts = 0;
- }
- if (varPauseUFX > 0) {
- // Q(2*(qMagn+norm32-16+minPause))
- tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no2, varPauseUFX);
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, nShifts);
-
- // Q(2*qMagn)
- avgDiffNormMagnUFX -= WEBRTC_SPL_MIN(avgDiffNormMagnUFX, tmpU32no1);
- } else {
- avgDiffNormMagnUFX = 0;
- }
- }
- //normalize and compute time average update of difference feature
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(avgDiffNormMagnUFX, 2 * inst->normData);
- if (inst->featureSpecDiff > tmpU32no1) {
- tmpU32no2 = WEBRTC_SPL_UMUL_32_16(inst->featureSpecDiff - tmpU32no1,
- SPECT_DIFF_TAVG_Q8); // Q(8-2*stages)
- inst->featureSpecDiff -= WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 8); // Q(-2*stages)
- } else {
- tmpU32no2 = WEBRTC_SPL_UMUL_32_16(tmpU32no1 - inst->featureSpecDiff,
- SPECT_DIFF_TAVG_Q8); // Q(8-2*stages)
- inst->featureSpecDiff += WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 8); // Q(-2*stages)
- }
-}
-
-// Compute speech/noise probability
-// speech/noise probability is returned in: probSpeechFinal
-//snrLocPrior is the prior SNR for each frequency (in Q11)
-//snrLocPost is the post SNR for each frequency (in Q11)
-void WebRtcNsx_SpeechNoiseProb(NsxInst_t* inst, WebRtc_UWord16* nonSpeechProbFinal,
- WebRtc_UWord32* priorLocSnr, WebRtc_UWord32* postLocSnr) {
- WebRtc_UWord32 zeros, num, den, tmpU32no1, tmpU32no2, tmpU32no3;
-
- WebRtc_Word32 invLrtFX, indPriorFX, tmp32, tmp32no1, tmp32no2, besselTmpFX32;
- WebRtc_Word32 frac32, logTmp;
- WebRtc_Word32 logLrtTimeAvgKsumFX;
-
- WebRtc_Word16 indPriorFX16;
- WebRtc_Word16 tmp16, tmp16no1, tmp16no2, tmpIndFX, tableIndex, frac, intPart;
-
- int i, normTmp, normTmp2, nShifts;
-
- // compute feature based on average LR factor
- // this is the average over all frequencies of the smooth log LRT
- logLrtTimeAvgKsumFX = 0;
- for (i = 0; i < inst->magnLen; i++) {
- besselTmpFX32 = (WebRtc_Word32)postLocSnr[i]; // Q11
- normTmp = WebRtcSpl_NormU32(postLocSnr[i]);
- num = WEBRTC_SPL_LSHIFT_U32(postLocSnr[i], normTmp); // Q(11+normTmp)
- if (normTmp > 10) {
- den = WEBRTC_SPL_LSHIFT_U32(priorLocSnr[i], normTmp - 11); // Q(normTmp)
- } else {
- den = WEBRTC_SPL_RSHIFT_U32(priorLocSnr[i], 11 - normTmp); // Q(normTmp)
- }
- if (den > 0) {
- besselTmpFX32 -= WEBRTC_SPL_UDIV(num, den); // Q11
- } else {
- besselTmpFX32 -= num; // Q11
- }
-
- // inst->logLrtTimeAvg[i] += LRT_TAVG * (besselTmp - log(snrLocPrior) - inst->logLrtTimeAvg[i]);
- // Here, LRT_TAVG = 0.5
- zeros = WebRtcSpl_NormU32(priorLocSnr[i]);
- frac32 = (WebRtc_Word32)(((priorLocSnr[i] << zeros) & 0x7FFFFFFF) >> 19);
- tmp32 = WEBRTC_SPL_MUL(frac32, frac32);
- tmp32 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(tmp32, -43), 19);
- tmp32 += WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16)frac32, 5412, 12);
- frac32 = tmp32 + 37;
- // tmp32 = log2(priorLocSnr[i])
- tmp32 = (WebRtc_Word32)(((31 - zeros) << 12) + frac32) - (11 << 12); // Q12
- logTmp = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL_32_16(tmp32, 178), 8); // log2(priorLocSnr[i])*log(2)
- tmp32no1 = WEBRTC_SPL_RSHIFT_W32(logTmp + inst->logLrtTimeAvgW32[i], 1); // Q12
- inst->logLrtTimeAvgW32[i] += (besselTmpFX32 - tmp32no1); // Q12
-
- logLrtTimeAvgKsumFX += inst->logLrtTimeAvgW32[i]; // Q12
- }
- inst->featureLogLrt = WEBRTC_SPL_RSHIFT_W32(logLrtTimeAvgKsumFX * 5, inst->stages + 10); // 5 = BIN_SIZE_LRT / 2
- // done with computation of LR factor
-
- //
- //compute the indicator functions
- //
-
- // average LRT feature
- // FLOAT code
- // indicator0 = 0.5 * (tanh(widthPrior * (logLrtTimeAvgKsum - threshPrior0)) + 1.0);
- tmpIndFX = 16384; // Q14(1.0)
- tmp32no1 = logLrtTimeAvgKsumFX - inst->thresholdLogLrt; // Q12
- nShifts = 7 - inst->stages; // WIDTH_PR_MAP_SHIFT - inst->stages + 5;
- //use larger width in tanh map for pause regions
- if (tmp32no1 < 0) {
- tmpIndFX = 0;
- tmp32no1 = -tmp32no1;
- //widthPrior = widthPrior * 2.0;
- nShifts++;
- }
- tmp32no1 = WEBRTC_SPL_SHIFT_W32(tmp32no1, nShifts); // Q14
- // compute indicator function: sigmoid map
- tableIndex = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 14);
- if ((tableIndex < 16) && (tableIndex >= 0)) {
- tmp16no2 = kIndicatorTable[tableIndex];
- tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
- frac = (WebRtc_Word16)(tmp32no1 & 0x00003fff); // Q14
- tmp16no2 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14);
- if (tmpIndFX == 0) {
- tmpIndFX = 8192 - tmp16no2; // Q14
- } else {
- tmpIndFX = 8192 + tmp16no2; // Q14
- }
- }
- indPriorFX = WEBRTC_SPL_MUL_16_16(inst->weightLogLrt, tmpIndFX); // 6*Q14
-
- //spectral flatness feature
- if (inst->weightSpecFlat) {
- tmpU32no1 = WEBRTC_SPL_UMUL(inst->featureSpecFlat, 400); // Q10
- tmpIndFX = 16384; // Q14(1.0)
- //use larger width in tanh map for pause regions
- tmpU32no2 = inst->thresholdSpecFlat - tmpU32no1; //Q10
- nShifts = 4;
- if (inst->thresholdSpecFlat < tmpU32no1) {
- tmpIndFX = 0;
- tmpU32no2 = tmpU32no1 - inst->thresholdSpecFlat;
- //widthPrior = widthPrior * 2.0;
- nShifts++;
- }
- tmp32no1 = (WebRtc_Word32)WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2,
- nShifts), 25); //Q14
- tmpU32no1 = WebRtcSpl_DivU32U16(WEBRTC_SPL_LSHIFT_U32(tmpU32no2, nShifts), 25); //Q14
- // compute indicator function: sigmoid map
- // FLOAT code
- // indicator1 = 0.5 * (tanh(sgnMap * widthPrior * (threshPrior1 - tmpFloat1)) + 1.0);
- tableIndex = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14);
- if (tableIndex < 16) {
- tmp16no2 = kIndicatorTable[tableIndex];
- tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
- frac = (WebRtc_Word16)(tmpU32no1 & 0x00003fff); // Q14
- tmp16no2 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, frac, 14);
- if (tmpIndFX) {
- tmpIndFX = 8192 + tmp16no2; // Q14
- } else {
- tmpIndFX = 8192 - tmp16no2; // Q14
- }
- }
- indPriorFX += WEBRTC_SPL_MUL_16_16(inst->weightSpecFlat, tmpIndFX); // 6*Q14
- }
-
- //for template spectral-difference
- if (inst->weightSpecDiff) {
- tmpU32no1 = 0;
- if (inst->featureSpecDiff) {
- normTmp = WEBRTC_SPL_MIN(20 - inst->stages,
- WebRtcSpl_NormU32(inst->featureSpecDiff));
- tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(inst->featureSpecDiff, normTmp); // Q(normTmp-2*stages)
- tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->timeAvgMagnEnergy, 20 - inst->stages
- - normTmp);
- if (tmpU32no2 > 0) {
- // Q(20 - inst->stages)
- tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2);
- } else {
- tmpU32no1 = (WebRtc_UWord32)(0x7fffffff);
- }
- }
- tmpU32no3 = WEBRTC_SPL_UDIV(WEBRTC_SPL_LSHIFT_U32(inst->thresholdSpecDiff, 17), 25);
- tmpU32no2 = tmpU32no1 - tmpU32no3;
- nShifts = 1;
- tmpIndFX = 16384; // Q14(1.0)
- //use larger width in tanh map for pause regions
- if (tmpU32no2 & 0x80000000) {
- tmpIndFX = 0;
- tmpU32no2 = tmpU32no3 - tmpU32no1;
- //widthPrior = widthPrior * 2.0;
- nShifts--;
- }
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, nShifts);
- // compute indicator function: sigmoid map
- /* FLOAT code
- indicator2 = 0.5 * (tanh(widthPrior * (tmpFloat1 - threshPrior2)) + 1.0);
- */
- tableIndex = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 14);
- if (tableIndex < 16) {
- tmp16no2 = kIndicatorTable[tableIndex];
- tmp16no1 = kIndicatorTable[tableIndex + 1] - kIndicatorTable[tableIndex];
- frac = (WebRtc_Word16)(tmpU32no1 & 0x00003fff); // Q14
- tmp16no2 += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
- tmp16no1, frac, 14);
- if (tmpIndFX) {
- tmpIndFX = 8192 + tmp16no2;
- } else {
- tmpIndFX = 8192 - tmp16no2;
- }
- }
- indPriorFX += WEBRTC_SPL_MUL_16_16(inst->weightSpecDiff, tmpIndFX); // 6*Q14
- }
-
- //combine the indicator function with the feature weights
- // FLOAT code
- // indPrior = 1 - (weightIndPrior0 * indicator0 + weightIndPrior1 * indicator1 + weightIndPrior2 * indicator2);
- indPriorFX16 = WebRtcSpl_DivW32W16ResW16(98307 - indPriorFX, 6); // Q14
- // done with computing indicator function
-
- //compute the prior probability
- // FLOAT code
- // inst->priorNonSpeechProb += PRIOR_UPDATE * (indPriorNonSpeech - inst->priorNonSpeechProb);
- tmp16 = indPriorFX16 - inst->priorNonSpeechProb; // Q14
- inst->priorNonSpeechProb += (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(
- PRIOR_UPDATE_Q14, tmp16, 14); // Q14
-
- //final speech probability: combine prior model with LR factor:
-
- memset(nonSpeechProbFinal, 0, sizeof(WebRtc_UWord16) * inst->magnLen);
-
- if (inst->priorNonSpeechProb > 0) {
- for (i = 0; i < inst->magnLen; i++) {
- // FLOAT code
- // invLrt = exp(inst->logLrtTimeAvg[i]);
- // invLrt = inst->priorSpeechProb * invLrt;
- // nonSpeechProbFinal[i] = (1.0 - inst->priorSpeechProb) / (1.0 - inst->priorSpeechProb + invLrt);
- // invLrt = (1.0 - inst->priorNonSpeechProb) * invLrt;
- // nonSpeechProbFinal[i] = inst->priorNonSpeechProb / (inst->priorNonSpeechProb + invLrt);
- if (inst->logLrtTimeAvgW32[i] < 65300) {
- tmp32no1 = WEBRTC_SPL_RSHIFT_W32(WEBRTC_SPL_MUL(inst->logLrtTimeAvgW32[i], 23637),
- 14); // Q12
- intPart = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_W32(tmp32no1, 12);
- if (intPart < -8) {
- intPart = -8;
- }
- frac = (WebRtc_Word16)(tmp32no1 & 0x00000fff); // Q12
-
- // Quadratic approximation of 2^frac
- tmp32no2 = WEBRTC_SPL_RSHIFT_W32(frac * frac * 44, 19); // Q12
- tmp32no2 += WEBRTC_SPL_MUL_16_16_RSFT(frac, 84, 7); // Q12
- invLrtFX = WEBRTC_SPL_LSHIFT_W32(1, 8 + intPart)
- + WEBRTC_SPL_SHIFT_W32(tmp32no2, intPart - 4); // Q8
-
- normTmp = WebRtcSpl_NormW32(invLrtFX);
- normTmp2 = WebRtcSpl_NormW16((16384 - inst->priorNonSpeechProb));
- if (normTmp + normTmp2 >= 7) {
- if (normTmp + normTmp2 < 15) {
- invLrtFX = WEBRTC_SPL_RSHIFT_W32(invLrtFX, 15 - normTmp2 - normTmp);
- // Q(normTmp+normTmp2-7)
- tmp32no1 = WEBRTC_SPL_MUL_32_16(invLrtFX, (16384 - inst->priorNonSpeechProb));
- // Q(normTmp+normTmp2+7)
- invLrtFX = WEBRTC_SPL_SHIFT_W32(tmp32no1, 7 - normTmp - normTmp2); // Q14
- } else {
- tmp32no1 = WEBRTC_SPL_MUL_32_16(invLrtFX, (16384 - inst->priorNonSpeechProb)); // Q22
- invLrtFX = WEBRTC_SPL_RSHIFT_W32(tmp32no1, 8); // Q14
- }
-
- tmp32no1 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)inst->priorNonSpeechProb, 8); // Q22
-
- nonSpeechProbFinal[i] = (WebRtc_UWord16)WEBRTC_SPL_DIV(tmp32no1,
- (WebRtc_Word32)inst->priorNonSpeechProb + invLrtFX); // Q8
- }
- }
- }
- }
-}
-
-// Transform input (speechFrame) to frequency domain magnitude (magnU16)
-void WebRtcNsx_DataAnalysis(NsxInst_t* inst, short* speechFrame, WebRtc_UWord16* magnU16) {
-
- WebRtc_UWord32 tmpU32no1, tmpU32no2;
-
- WebRtc_Word32 tmp_1_w32 = 0;
- WebRtc_Word32 tmp_2_w32 = 0;
- WebRtc_Word32 sum_log_magn = 0;
- WebRtc_Word32 sum_log_i_log_magn = 0;
-
- WebRtc_UWord16 sum_log_magn_u16 = 0;
- WebRtc_UWord16 tmp_u16 = 0;
-
- WebRtc_Word16 sum_log_i = 0;
- WebRtc_Word16 sum_log_i_square = 0;
- WebRtc_Word16 frac = 0;
- WebRtc_Word16 log2 = 0;
- WebRtc_Word16 matrix_determinant = 0;
- WebRtc_Word16 winData[ANAL_BLOCKL_MAX], maxWinData;
- WebRtc_Word16 realImag[ANAL_BLOCKL_MAX << 1];
-
- int i, j;
- int zeros;
- int net_norm = 0;
- int right_shifts_in_magnU16 = 0;
- int right_shifts_in_initMagnEst = 0;
-
- // Update analysis buffer for lower band, and window data before FFT.
- WebRtcNsx_AnalysisUpdate(inst, winData, speechFrame);
-
- // Get input energy
- inst->energyIn = WebRtcSpl_Energy(winData, (int)inst->anaLen, &(inst->scaleEnergyIn));
-
- // Reset zero input flag
- inst->zeroInputSignal = 0;
- // Acquire norm for winData
- maxWinData = WebRtcSpl_MaxAbsValueW16(winData, inst->anaLen);
- inst->normData = WebRtcSpl_NormW16(maxWinData);
- if (maxWinData == 0) {
- // Treat zero input separately.
- inst->zeroInputSignal = 1;
- return;
- }
-
- // Determine the net normalization in the frequency domain
- net_norm = inst->stages - inst->normData;
- // Track lowest normalization factor and use it to prevent wrap around in shifting
- right_shifts_in_magnU16 = inst->normData - inst->minNorm;
- right_shifts_in_initMagnEst = WEBRTC_SPL_MAX(-right_shifts_in_magnU16, 0);
- inst->minNorm -= right_shifts_in_initMagnEst;
- right_shifts_in_magnU16 = WEBRTC_SPL_MAX(right_shifts_in_magnU16, 0);
-
- // create realImag as winData interleaved with zeros (= imag. part), normalize it
- WebRtcNsx_CreateComplexBuffer(inst, winData, realImag);
-
- // bit-reverse position of elements in array and FFT the array
- WebRtcSpl_ComplexBitReverse(realImag, inst->stages); // Q(normData-stages)
- WebRtcSpl_ComplexFFT(realImag, inst->stages, 1);
-
- inst->imag[0] = 0; // Q(normData-stages)
- inst->imag[inst->anaLen2] = 0;
- inst->real[0] = realImag[0]; // Q(normData-stages)
- inst->real[inst->anaLen2] = realImag[inst->anaLen];
- // Q(2*(normData-stages))
- inst->magnEnergy = (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(inst->real[0], inst->real[0]);
- inst->magnEnergy += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(inst->real[inst->anaLen2],
- inst->real[inst->anaLen2]);
- magnU16[0] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(inst->real[0]); // Q(normData-stages)
- magnU16[inst->anaLen2] = (WebRtc_UWord16)WEBRTC_SPL_ABS_W16(inst->real[inst->anaLen2]);
- inst->sumMagn = (WebRtc_UWord32)magnU16[0]; // Q(normData-stages)
- inst->sumMagn += (WebRtc_UWord32)magnU16[inst->anaLen2];
-
- if (inst->blockIndex >= END_STARTUP_SHORT) {
- for (i = 1, j = 2; i < inst->anaLen2; i += 1, j += 2) {
- inst->real[i] = realImag[j];
- inst->imag[i] = -realImag[j + 1];
- // magnitude spectrum
- // energy in Q(2*(normData-stages))
- tmpU32no1 = (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(realImag[j], realImag[j]);
- tmpU32no1 += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(realImag[j + 1], realImag[j + 1]);
- inst->magnEnergy += tmpU32no1; // Q(2*(normData-stages))
-
- magnU16[i] = (WebRtc_UWord16)WebRtcSpl_SqrtFloor(tmpU32no1); // Q(normData-stages)
- inst->sumMagn += (WebRtc_UWord32)magnU16[i]; // Q(normData-stages)
- }
- } else {
- //
- // Gather information during startup for noise parameter estimation
- //
-
- // Switch initMagnEst to Q(minNorm-stages)
- inst->initMagnEst[0] = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[0],
- right_shifts_in_initMagnEst);
- inst->initMagnEst[inst->anaLen2] =
- WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[inst->anaLen2],
- right_shifts_in_initMagnEst); // Q(minNorm-stages)
-
- // Shift magnU16 to same domain as initMagnEst
- tmpU32no1 = WEBRTC_SPL_RSHIFT_W32((WebRtc_UWord32)magnU16[0],
- right_shifts_in_magnU16); // Q(minNorm-stages)
- tmpU32no2 = WEBRTC_SPL_RSHIFT_W32((WebRtc_UWord32)magnU16[inst->anaLen2],
- right_shifts_in_magnU16); // Q(minNorm-stages)
-
- // Update initMagnEst
- inst->initMagnEst[0] += tmpU32no1; // Q(minNorm-stages)
- inst->initMagnEst[inst->anaLen2] += tmpU32no2; // Q(minNorm-stages)
-
- log2 = 0;
- if (magnU16[inst->anaLen2]) {
- // Calculate log2(magnU16[inst->anaLen2])
- zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magnU16[inst->anaLen2]);
- frac = (WebRtc_Word16)((((WebRtc_UWord32)magnU16[inst->anaLen2] << zeros) &
- 0x7FFFFFFF) >> 23); // Q8
- // log2(magnU16(i)) in Q8
- assert(frac < 256);
- log2 = (WebRtc_Word16)(((31 - zeros) << 8) + WebRtcNsx_kLogTableFrac[frac]);
- }
-
- sum_log_magn = (WebRtc_Word32)log2; // Q8
- // sum_log_i_log_magn in Q17
- sum_log_i_log_magn = (WEBRTC_SPL_MUL_16_16(kLogIndex[inst->anaLen2], log2) >> 3);
-
- for (i = 1, j = 2; i < inst->anaLen2; i += 1, j += 2) {
- inst->real[i] = realImag[j];
- inst->imag[i] = -realImag[j + 1];
- // magnitude spectrum
- // energy in Q(2*(normData-stages))
- tmpU32no1 = (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(realImag[j], realImag[j]);
- tmpU32no1 += (WebRtc_UWord32)WEBRTC_SPL_MUL_16_16(realImag[j + 1], realImag[j + 1]);
- inst->magnEnergy += tmpU32no1; // Q(2*(normData-stages))
-
- magnU16[i] = (WebRtc_UWord16)WebRtcSpl_SqrtFloor(tmpU32no1); // Q(normData-stages)
- inst->sumMagn += (WebRtc_UWord32)magnU16[i]; // Q(normData-stages)
-
- // Switch initMagnEst to Q(minNorm-stages)
- inst->initMagnEst[i] = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[i],
- right_shifts_in_initMagnEst);
-
- // Shift magnU16 to same domain as initMagnEst, i.e., Q(minNorm-stages)
- tmpU32no1 = WEBRTC_SPL_RSHIFT_W32((WebRtc_UWord32)magnU16[i],
- right_shifts_in_magnU16);
- // Update initMagnEst
- inst->initMagnEst[i] += tmpU32no1; // Q(minNorm-stages)
-
- if (i >= kStartBand) {
- // For pink noise estimation. Collect data neglecting lower frequency band
- log2 = 0;
- if (magnU16[i]) {
- zeros = WebRtcSpl_NormU32((WebRtc_UWord32)magnU16[i]);
- frac = (WebRtc_Word16)((((WebRtc_UWord32)magnU16[i] << zeros) &
- 0x7FFFFFFF) >> 23);
- // log2(magnU16(i)) in Q8
- assert(frac < 256);
- log2 = (WebRtc_Word16)(((31 - zeros) << 8)
- + WebRtcNsx_kLogTableFrac[frac]);
- }
- sum_log_magn += (WebRtc_Word32)log2; // Q8
- // sum_log_i_log_magn in Q17
- sum_log_i_log_magn += (WEBRTC_SPL_MUL_16_16(kLogIndex[i], log2) >> 3);
- }
- }
-
- //
- //compute simplified noise model during startup
- //
-
- // Estimate White noise
-
- // Switch whiteNoiseLevel to Q(minNorm-stages)
- inst->whiteNoiseLevel = WEBRTC_SPL_RSHIFT_U32(inst->whiteNoiseLevel,
- right_shifts_in_initMagnEst);
-
- // Update the average magnitude spectrum, used as noise estimate.
- tmpU32no1 = WEBRTC_SPL_UMUL_32_16(inst->sumMagn, inst->overdrive);
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, inst->stages + 8);
-
- // Replacing division above with 'stages' shifts
- // Shift to same Q-domain as whiteNoiseLevel
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, right_shifts_in_magnU16);
- // This operation is safe from wrap around as long as END_STARTUP_SHORT < 128
- assert(END_STARTUP_SHORT < 128);
- inst->whiteNoiseLevel += tmpU32no1; // Q(minNorm-stages)
-
- // Estimate Pink noise parameters
- // Denominator used in both parameter estimates.
- // The value is only dependent on the size of the frequency band (kStartBand)
- // and to reduce computational complexity stored in a table (kDeterminantEstMatrix[])
- assert(kStartBand < 66);
- matrix_determinant = kDeterminantEstMatrix[kStartBand]; // Q0
- sum_log_i = kSumLogIndex[kStartBand]; // Q5
- sum_log_i_square = kSumSquareLogIndex[kStartBand]; // Q2
- if (inst->fs == 8000) {
- // Adjust values to shorter blocks in narrow band.
- tmp_1_w32 = (WebRtc_Word32)matrix_determinant;
- tmp_1_w32 += WEBRTC_SPL_MUL_16_16_RSFT(kSumLogIndex[65], sum_log_i, 9);
- tmp_1_w32 -= WEBRTC_SPL_MUL_16_16_RSFT(kSumLogIndex[65], kSumLogIndex[65], 10);
- tmp_1_w32 -= WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)sum_log_i_square, 4);
- tmp_1_w32 -= WEBRTC_SPL_MUL_16_16_RSFT((WebRtc_Word16)
- (inst->magnLen - kStartBand), kSumSquareLogIndex[65], 2);
- matrix_determinant = (WebRtc_Word16)tmp_1_w32;
- sum_log_i -= kSumLogIndex[65]; // Q5
- sum_log_i_square -= kSumSquareLogIndex[65]; // Q2
- }
-
- // Necessary number of shifts to fit sum_log_magn in a word16
- zeros = 16 - WebRtcSpl_NormW32(sum_log_magn);
- if (zeros < 0) {
- zeros = 0;
- }
- tmp_1_w32 = WEBRTC_SPL_LSHIFT_W32(sum_log_magn, 1); // Q9
- sum_log_magn_u16 = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_W32(tmp_1_w32, zeros);//Q(9-zeros)
-
- // Calculate and update pinkNoiseNumerator. Result in Q11.
- tmp_2_w32 = WEBRTC_SPL_MUL_16_U16(sum_log_i_square, sum_log_magn_u16); // Q(11-zeros)
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32((WebRtc_UWord32)sum_log_i_log_magn, 12); // Q5
-
- // Shift the largest value of sum_log_i and tmp32no3 before multiplication
- tmp_u16 = WEBRTC_SPL_LSHIFT_U16((WebRtc_UWord16)sum_log_i, 1); // Q6
- if ((WebRtc_UWord32)sum_log_i > tmpU32no1) {
- tmp_u16 = WEBRTC_SPL_RSHIFT_U16(tmp_u16, zeros);
- } else {
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, zeros);
- }
- tmp_2_w32 -= (WebRtc_Word32)WEBRTC_SPL_UMUL_32_16(tmpU32no1, tmp_u16); // Q(11-zeros)
- matrix_determinant = WEBRTC_SPL_RSHIFT_W16(matrix_determinant, zeros); // Q(-zeros)
- tmp_2_w32 = WebRtcSpl_DivW32W16(tmp_2_w32, matrix_determinant); // Q11
- tmp_2_w32 += WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)net_norm, 11); // Q11
- if (tmp_2_w32 < 0) {
- tmp_2_w32 = 0;
- }
- inst->pinkNoiseNumerator += tmp_2_w32; // Q11
-
- // Calculate and update pinkNoiseExp. Result in Q14.
- tmp_2_w32 = WEBRTC_SPL_MUL_16_U16(sum_log_i, sum_log_magn_u16); // Q(14-zeros)
- tmp_1_w32 = WEBRTC_SPL_RSHIFT_W32(sum_log_i_log_magn, 3 + zeros);
- tmp_1_w32 = WEBRTC_SPL_MUL((WebRtc_Word32)(inst->magnLen - kStartBand),
- tmp_1_w32);
- tmp_2_w32 -= tmp_1_w32; // Q(14-zeros)
- if (tmp_2_w32 > 0) {
- // If the exponential parameter is negative force it to zero, which means a
- // flat spectrum.
- tmp_1_w32 = WebRtcSpl_DivW32W16(tmp_2_w32, matrix_determinant); // Q14
- inst->pinkNoiseExp += WEBRTC_SPL_SAT(16384, tmp_1_w32, 0); // Q14
- }
- }
-}
-
-void WebRtcNsx_DataSynthesis(NsxInst_t* inst, short* outFrame) {
- WebRtc_Word32 energyOut;
-
- WebRtc_Word16 realImag[ANAL_BLOCKL_MAX << 1];
- WebRtc_Word16 tmp16no1, tmp16no2;
- WebRtc_Word16 energyRatio;
- WebRtc_Word16 gainFactor, gainFactor1, gainFactor2;
-
- int i;
- int outCIFFT;
- int scaleEnergyOut = 0;
-
- if (inst->zeroInputSignal) {
- // synthesize the special case of zero input
- // read out fully processed segment
- for (i = 0; i < inst->blockLen10ms; i++) {
- outFrame[i] = inst->synthesisBuffer[i]; // Q0
- }
- // update synthesis buffer
- WEBRTC_SPL_MEMCPY_W16(inst->synthesisBuffer,
- inst->synthesisBuffer + inst->blockLen10ms,
- inst->anaLen - inst->blockLen10ms);
- WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer + inst->anaLen - inst->blockLen10ms,
- inst->blockLen10ms);
- return;
- }
-
- // Filter the data in the frequency domain, and create spectrum.
- WebRtcNsx_PrepareSpectrum(inst, realImag);
-
- // bit-reverse position of elements in array and IFFT it
- WebRtcSpl_ComplexBitReverse(realImag, inst->stages);
- outCIFFT = WebRtcSpl_ComplexIFFT(realImag, inst->stages, 1);
-
- // Denormalize.
- WebRtcNsx_Denormalize(inst, realImag, outCIFFT);
-
- //scale factor: only do it after END_STARTUP_LONG time
- gainFactor = 8192; // 8192 = Q13(1.0)
- if (inst->gainMap == 1 &&
- inst->blockIndex > END_STARTUP_LONG &&
- inst->energyIn > 0) {
- energyOut = WebRtcSpl_Energy(inst->real, (int)inst->anaLen, &scaleEnergyOut); // Q(-scaleEnergyOut)
- if (scaleEnergyOut == 0 && !(energyOut & 0x7f800000)) {
- energyOut = WEBRTC_SPL_SHIFT_W32(energyOut, 8 + scaleEnergyOut
- - inst->scaleEnergyIn);
- } else {
- inst->energyIn = WEBRTC_SPL_RSHIFT_W32(inst->energyIn, 8 + scaleEnergyOut
- - inst->scaleEnergyIn); // Q(-8-scaleEnergyOut)
- }
-
- assert(inst->energyIn > 0);
- energyRatio = (WebRtc_Word16)WEBRTC_SPL_DIV(energyOut
- + WEBRTC_SPL_RSHIFT_W32(inst->energyIn, 1), inst->energyIn); // Q8
- // Limit the ratio to [0, 1] in Q8, i.e., [0, 256]
- energyRatio = WEBRTC_SPL_SAT(256, energyRatio, 0);
-
- // all done in lookup tables now
- assert(energyRatio < 257);
- gainFactor1 = kFactor1Table[energyRatio]; // Q8
- gainFactor2 = inst->factor2Table[energyRatio]; // Q8
-
- //combine both scales with speech/noise prob: note prior (priorSpeechProb) is not frequency dependent
-
- // factor = inst->priorSpeechProb*factor1 + (1.0-inst->priorSpeechProb)*factor2; // original code
- tmp16no1 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(16384 - inst->priorNonSpeechProb,
- gainFactor1, 14); // Q13 16384 = Q14(1.0)
- tmp16no2 = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(inst->priorNonSpeechProb,
- gainFactor2, 14); // Q13;
- gainFactor = tmp16no1 + tmp16no2; // Q13
- } // out of flag_gain_map==1
-
- // Synthesis, read out fully processed segment, and update synthesis buffer.
- WebRtcNsx_SynthesisUpdate(inst, outFrame, gainFactor);
-}
-
-int WebRtcNsx_ProcessCore(NsxInst_t* inst, short* speechFrame, short* speechFrameHB,
- short* outFrame, short* outFrameHB) {
- // main routine for noise suppression
-
- WebRtc_UWord32 tmpU32no1, tmpU32no2, tmpU32no3;
- WebRtc_UWord32 satMax, maxNoiseU32;
- WebRtc_UWord32 tmpMagnU32, tmpNoiseU32;
- WebRtc_UWord32 nearMagnEst;
- WebRtc_UWord32 noiseUpdateU32;
- WebRtc_UWord32 noiseU32[HALF_ANAL_BLOCKL];
- WebRtc_UWord32 postLocSnr[HALF_ANAL_BLOCKL];
- WebRtc_UWord32 priorLocSnr[HALF_ANAL_BLOCKL];
- WebRtc_UWord32 prevNearSnr[HALF_ANAL_BLOCKL];
- WebRtc_UWord32 curNearSnr;
- WebRtc_UWord32 priorSnr;
- WebRtc_UWord32 noise_estimate = 0;
- WebRtc_UWord32 noise_estimate_avg = 0;
- WebRtc_UWord32 numerator = 0;
-
- WebRtc_Word32 tmp32no1, tmp32no2;
- WebRtc_Word32 pink_noise_num_avg = 0;
-
- WebRtc_UWord16 tmpU16no1;
- WebRtc_UWord16 magnU16[HALF_ANAL_BLOCKL];
- WebRtc_UWord16 prevNoiseU16[HALF_ANAL_BLOCKL];
- WebRtc_UWord16 nonSpeechProbFinal[HALF_ANAL_BLOCKL];
- WebRtc_UWord16 gammaNoise, prevGammaNoise;
- WebRtc_UWord16 noiseSupFilterTmp[HALF_ANAL_BLOCKL];
-
- WebRtc_Word16 qMagn, qNoise;
- WebRtc_Word16 avgProbSpeechHB, gainModHB, avgFilterGainHB, gainTimeDomainHB;
- WebRtc_Word16 pink_noise_exp_avg = 0;
-
- int i;
- int nShifts, postShifts;
- int norm32no1, norm32no2;
- int flag, sign;
- int q_domain_to_use = 0;
-
- // Code for ARMv7-Neon platform assumes the following:
- assert(inst->anaLen % 16 == 0);
- assert(inst->anaLen2 % 8 == 0);
- assert(inst->blockLen10ms % 16 == 0);
- assert(inst->magnLen == inst->anaLen2 + 1);
-
-#ifdef NS_FILEDEBUG
- fwrite(spframe, sizeof(short), inst->blockLen10ms, inst->infile);
-#endif
-
- // Check that initialization has been done
- if (inst->initFlag != 1) {
- return -1;
- }
- // Check for valid pointers based on sampling rate
- if ((inst->fs == 32000) && (speechFrameHB == NULL)) {
- return -1;
- }
-
- // Store speechFrame and transform to frequency domain
- WebRtcNsx_DataAnalysis(inst, speechFrame, magnU16);
-
- if (inst->zeroInputSignal) {
- WebRtcNsx_DataSynthesis(inst, outFrame);
-
- if (inst->fs == 32000) {
- // update analysis buffer for H band
- // append new data to buffer FX
- WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX, inst->dataBufHBFX + inst->blockLen10ms,
- inst->anaLen - inst->blockLen10ms);
- WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX + inst->anaLen - inst->blockLen10ms,
- speechFrameHB, inst->blockLen10ms);
- for (i = 0; i < inst->blockLen10ms; i++) {
- outFrameHB[i] = inst->dataBufHBFX[i]; // Q0
- }
- } // end of H band gain computation
- return 0;
- }
-
- // Update block index when we have something to process
- inst->blockIndex++;
- //
-
- // Norm of magn
- qMagn = inst->normData - inst->stages;
-
- // Compute spectral flatness on input spectrum
- WebRtcNsx_ComputeSpectralFlatness(inst, magnU16);
-
- // quantile noise estimate
- WebRtcNsx_NoiseEstimation(inst, magnU16, noiseU32, &qNoise);
-
- //noise estimate from previous frame
- for (i = 0; i < inst->magnLen; i++) {
- prevNoiseU16[i] = (WebRtc_UWord16)WEBRTC_SPL_RSHIFT_U32(inst->prevNoiseU32[i], 11); // Q(prevQNoise)
- }
-
- if (inst->blockIndex < END_STARTUP_SHORT) {
- // Noise Q-domain to be used later; see description at end of section.
- q_domain_to_use = WEBRTC_SPL_MIN((int)qNoise, inst->minNorm - inst->stages);
-
- // Calculate frequency independent parts in parametric noise estimate and calculate
- // the estimate for the lower frequency band (same values for all frequency bins)
- if (inst->pinkNoiseExp) {
- pink_noise_exp_avg = (WebRtc_Word16)WebRtcSpl_DivW32W16(inst->pinkNoiseExp,
- (WebRtc_Word16)(inst->blockIndex + 1)); // Q14
- pink_noise_num_avg = WebRtcSpl_DivW32W16(inst->pinkNoiseNumerator,
- (WebRtc_Word16)(inst->blockIndex + 1)); // Q11
- WebRtcNsx_CalcParametricNoiseEstimate(inst,
- pink_noise_exp_avg,
- pink_noise_num_avg,
- kStartBand,
- &noise_estimate,
- &noise_estimate_avg);
- } else {
- // Use white noise estimate if we have poor pink noise parameter estimates
- noise_estimate = inst->whiteNoiseLevel; // Q(minNorm-stages)
- noise_estimate_avg = noise_estimate / (inst->blockIndex + 1); // Q(minNorm-stages)
- }
- for (i = 0; i < inst->magnLen; i++) {
- // Estimate the background noise using the pink noise parameters if permitted
- if ((inst->pinkNoiseExp) && (i >= kStartBand)) {
- // Reset noise_estimate
- noise_estimate = 0;
- noise_estimate_avg = 0;
- // Calculate the parametric noise estimate for current frequency bin
- WebRtcNsx_CalcParametricNoiseEstimate(inst,
- pink_noise_exp_avg,
- pink_noise_num_avg,
- i,
- &noise_estimate,
- &noise_estimate_avg);
- }
- // Calculate parametric Wiener filter
- noiseSupFilterTmp[i] = inst->denoiseBound;
- if (inst->initMagnEst[i]) {
- // numerator = (initMagnEst - noise_estimate * overdrive)
- // Result in Q(8+minNorm-stages)
- tmpU32no1 = WEBRTC_SPL_UMUL_32_16(noise_estimate, inst->overdrive);
- numerator = WEBRTC_SPL_LSHIFT_U32(inst->initMagnEst[i], 8);
- if (numerator > tmpU32no1) {
- // Suppression filter coefficient larger than zero, so calculate.
- numerator -= tmpU32no1;
-
- // Determine number of left shifts in numerator for best accuracy after
- // division
- nShifts = WebRtcSpl_NormU32(numerator);
- nShifts = WEBRTC_SPL_SAT(6, nShifts, 0);
-
- // Shift numerator to Q(nShifts+8+minNorm-stages)
- numerator = WEBRTC_SPL_LSHIFT_U32(numerator, nShifts);
-
- // Shift denominator to Q(nShifts-6+minNorm-stages)
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(inst->initMagnEst[i], 6 - nShifts);
- if (tmpU32no1 == 0) {
- // This is only possible if numerator = 0, in which case
- // we don't need any division.
- tmpU32no1 = 1;
- }
- tmpU32no2 = WEBRTC_SPL_UDIV(numerator, tmpU32no1); // Q14
- noiseSupFilterTmp[i] = (WebRtc_UWord16)WEBRTC_SPL_SAT(16384, tmpU32no2,
- (WebRtc_UWord32)(inst->denoiseBound)); // Q14
- }
- }
- // Weight quantile noise 'noiseU32' with modeled noise 'noise_estimate_avg'
- // 'noiseU32 is in Q(qNoise) and 'noise_estimate' in Q(minNorm-stages)
- // To guarantee that we do not get wrap around when shifting to the same domain
- // we use the lowest one. Furthermore, we need to save 6 bits for the weighting.
- // 'noise_estimate_avg' can handle this operation by construction, but 'noiseU32'
- // may not.
-
- // Shift 'noiseU32' to 'q_domain_to_use'
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], (int)qNoise - q_domain_to_use);
- // Shift 'noise_estimate_avg' to 'q_domain_to_use'
- tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(noise_estimate_avg, inst->minNorm - inst->stages
- - q_domain_to_use);
- // Make a simple check to see if we have enough room for weighting 'tmpU32no1'
- // without wrap around
- nShifts = 0;
- if (tmpU32no1 & 0xfc000000) {
- tmpU32no1 = WEBRTC_SPL_RSHIFT_U32(tmpU32no1, 6);
- tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpU32no2, 6);
- nShifts = 6;
- }
- tmpU32no1 *= inst->blockIndex;
- tmpU32no2 *= (END_STARTUP_SHORT - inst->blockIndex);
- // Add them together and divide by startup length
- noiseU32[i] = WebRtcSpl_DivU32U16(tmpU32no1 + tmpU32no2, END_STARTUP_SHORT);
- // Shift back if necessary
- noiseU32[i] = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], nShifts);
- }
- // Update new Q-domain for 'noiseU32'
- qNoise = q_domain_to_use;
- }
- // compute average signal during END_STARTUP_LONG time:
- // used to normalize spectral difference measure
- if (inst->blockIndex < END_STARTUP_LONG) {
- // substituting division with shift ending up in Q(-2*stages)
- inst->timeAvgMagnEnergyTmp
- += WEBRTC_SPL_RSHIFT_U32(inst->magnEnergy,
- 2 * inst->normData + inst->stages - 1);
- inst->timeAvgMagnEnergy = WebRtcSpl_DivU32U16(inst->timeAvgMagnEnergyTmp,
- inst->blockIndex + 1);
- }
-
- //start processing at frames == converged+1
- // STEP 1: compute prior and post SNR based on quantile noise estimates
-
- // compute direct decision (DD) estimate of prior SNR: needed for new method
- satMax = (WebRtc_UWord32)1048575;// Largest possible value without getting overflow despite shifting 12 steps
- postShifts = 6 + qMagn - qNoise;
- nShifts = 5 - inst->prevQMagn + inst->prevQNoise;
- for (i = 0; i < inst->magnLen; i++) {
- // FLOAT:
- // post SNR
- // postLocSnr[i] = 0.0;
- // if (magn[i] > noise[i])
- // {
- // postLocSnr[i] = magn[i] / (noise[i] + 0.0001);
- // }
- // // previous post SNR
- // // previous estimate: based on previous frame with gain filter (smooth is previous filter)
- //
- // prevNearSnr[i] = inst->prevMagnU16[i] / (inst->noisePrev[i] + 0.0001) * (inst->smooth[i]);
- //
- // // DD estimate is sum of two terms: current estimate and previous estimate
- // // directed decision update of priorSnr (or we actually store [2*priorSnr+1])
- //
- // priorLocSnr[i] = DD_PR_SNR * prevNearSnr[i] + (1.0 - DD_PR_SNR) * (postLocSnr[i] - 1.0);
-
- // calculate post SNR: output in Q11
- postLocSnr[i] = 2048; // 1.0 in Q11
- tmpU32no1 = WEBRTC_SPL_LSHIFT_U32((WebRtc_UWord32)magnU16[i], 6); // Q(6+qMagn)
- if (postShifts < 0) {
- tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], -postShifts); // Q(6+qMagn)
- } else {
- tmpU32no2 = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], postShifts); // Q(6+qMagn)
- }
- if (tmpU32no1 > tmpU32no2) {
- // Current magnitude larger than noise
- tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, 11); // Q(17+qMagn)
- if (tmpU32no2 > 0) {
- tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q11
- postLocSnr[i] = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
- } else {
- postLocSnr[i] = satMax;
- }
- }
-
- // calculate prevNearSnr[i] and save for later instead of recalculating it later
- nearMagnEst = WEBRTC_SPL_UMUL_16_16(inst->prevMagnU16[i], inst->noiseSupFilter[i]); // Q(prevQMagn+14)
- tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(nearMagnEst, 3); // Q(prevQMagn+17)
- tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(inst->prevNoiseU32[i], nShifts); // Q(prevQMagn+6)
-
- if (tmpU32no2 > 0) {
- tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q11
- tmpU32no1 = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
- } else {
- tmpU32no1 = satMax; // Q11
- }
- prevNearSnr[i] = tmpU32no1; // Q11
-
- //directed decision update of priorSnr
- tmpU32no1 = WEBRTC_SPL_UMUL_32_16(prevNearSnr[i], DD_PR_SNR_Q11); // Q22
- tmpU32no2 = WEBRTC_SPL_UMUL_32_16(postLocSnr[i] - 2048, ONE_MINUS_DD_PR_SNR_Q11); // Q22
- priorSnr = tmpU32no1 + tmpU32no2 + 512; // Q22 (added 512 for rounding)
- // priorLocSnr = 1 + 2*priorSnr
- priorLocSnr[i] = 2048 + WEBRTC_SPL_RSHIFT_U32(priorSnr, 10); // Q11
- } // end of loop over frequencies
- // done with step 1: DD computation of prior and post SNR
-
- // STEP 2: compute speech/noise likelihood
-
- //compute difference of input spectrum with learned/estimated noise spectrum
- WebRtcNsx_ComputeSpectralDifference(inst, magnU16);
- //compute histograms for determination of parameters (thresholds and weights for features)
- //parameters are extracted once every window time (=inst->modelUpdate)
- //counter update
- inst->cntThresUpdate++;
- flag = (int)(inst->cntThresUpdate == inst->modelUpdate);
- //update histogram
- WebRtcNsx_FeatureParameterExtraction(inst, flag);
- //compute model parameters
- if (flag) {
- inst->cntThresUpdate = 0; // Reset counter
- //update every window:
- // get normalization for spectral difference for next window estimate
-
- // Shift to Q(-2*stages)
- inst->curAvgMagnEnergy = WEBRTC_SPL_RSHIFT_U32(inst->curAvgMagnEnergy, STAT_UPDATES);
-
- tmpU32no1 = (inst->curAvgMagnEnergy + inst->timeAvgMagnEnergy + 1) >> 1; //Q(-2*stages)
- // Update featureSpecDiff
- if ((tmpU32no1 != inst->timeAvgMagnEnergy) && (inst->featureSpecDiff) &&
- (inst->timeAvgMagnEnergy > 0)) {
- norm32no1 = 0;
- tmpU32no3 = tmpU32no1;
- while (0xFFFF0000 & tmpU32no3) {
- tmpU32no3 >>= 1;
- norm32no1++;
- }
- tmpU32no2 = inst->featureSpecDiff;
- while (0xFFFF0000 & tmpU32no2) {
- tmpU32no2 >>= 1;
- norm32no1++;
- }
- tmpU32no3 = WEBRTC_SPL_UMUL(tmpU32no3, tmpU32no2);
- tmpU32no3 = WEBRTC_SPL_UDIV(tmpU32no3, inst->timeAvgMagnEnergy);
- if (WebRtcSpl_NormU32(tmpU32no3) < norm32no1) {
- inst->featureSpecDiff = 0x007FFFFF;
- } else {
- inst->featureSpecDiff = WEBRTC_SPL_MIN(0x007FFFFF,
- WEBRTC_SPL_LSHIFT_U32(tmpU32no3, norm32no1));
- }
- }
-
- inst->timeAvgMagnEnergy = tmpU32no1; // Q(-2*stages)
- inst->curAvgMagnEnergy = 0;
- }
-
- //compute speech/noise probability
- WebRtcNsx_SpeechNoiseProb(inst, nonSpeechProbFinal, priorLocSnr, postLocSnr);
-
- //time-avg parameter for noise update
- gammaNoise = NOISE_UPDATE_Q8; // Q8
-
- maxNoiseU32 = 0;
- postShifts = inst->prevQNoise - qMagn;
- nShifts = inst->prevQMagn - qMagn;
- for (i = 0; i < inst->magnLen; i++) {
- // temporary noise update: use it for speech frames if update value is less than previous
- // the formula has been rewritten into:
- // noiseUpdate = noisePrev[i] + (1 - gammaNoise) * nonSpeechProb * (magn[i] - noisePrev[i])
-
- if (postShifts < 0) {
- tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(magnU16[i], -postShifts); // Q(prevQNoise)
- } else {
- tmpU32no2 = WEBRTC_SPL_LSHIFT_U32(magnU16[i], postShifts); // Q(prevQNoise)
- }
- if (prevNoiseU16[i] > tmpU32no2) {
- sign = -1;
- tmpU32no1 = prevNoiseU16[i] - tmpU32no2;
- } else {
- sign = 1;
- tmpU32no1 = tmpU32no2 - prevNoiseU16[i];
- }
- noiseUpdateU32 = inst->prevNoiseU32[i]; // Q(prevQNoise+11)
- tmpU32no3 = 0;
- if ((tmpU32no1) && (nonSpeechProbFinal[i])) {
- // This value will be used later, if gammaNoise changes
- tmpU32no3 = WEBRTC_SPL_UMUL_32_16(tmpU32no1, nonSpeechProbFinal[i]); // Q(prevQNoise+8)
- if (0x7c000000 & tmpU32no3) {
- // Shifting required before multiplication
- tmpU32no2
- = WEBRTC_SPL_UMUL_32_16(WEBRTC_SPL_RSHIFT_U32(tmpU32no3, 5), gammaNoise); // Q(prevQNoise+11)
- } else {
- // We can do shifting after multiplication
- tmpU32no2
- = WEBRTC_SPL_RSHIFT_U32(WEBRTC_SPL_UMUL_32_16(tmpU32no3, gammaNoise), 5); // Q(prevQNoise+11)
- }
- if (sign > 0) {
- noiseUpdateU32 += tmpU32no2; // Q(prevQNoise+11)
- } else {
- // This operation is safe. We can never get wrap around, since worst
- // case scenario means magnU16 = 0
- noiseUpdateU32 -= tmpU32no2; // Q(prevQNoise+11)
- }
- }
-
- //increase gamma (i.e., less noise update) for frame likely to be speech
- prevGammaNoise = gammaNoise;
- gammaNoise = NOISE_UPDATE_Q8;
- //time-constant based on speech/noise state
- //increase gamma (i.e., less noise update) for frames likely to be speech
- if (nonSpeechProbFinal[i] < ONE_MINUS_PROB_RANGE_Q8) {
- gammaNoise = GAMMA_NOISE_TRANS_AND_SPEECH_Q8;
- }
-
- if (prevGammaNoise != gammaNoise) {
- // new noise update
- // this line is the same as above, only that the result is stored in a different variable and the gammaNoise
- // has changed
- //
- // noiseUpdate = noisePrev[i] + (1 - gammaNoise) * nonSpeechProb * (magn[i] - noisePrev[i])
-
- if (0x7c000000 & tmpU32no3) {
- // Shifting required before multiplication
- tmpU32no2
- = WEBRTC_SPL_UMUL_32_16(WEBRTC_SPL_RSHIFT_U32(tmpU32no3, 5), gammaNoise); // Q(prevQNoise+11)
- } else {
- // We can do shifting after multiplication
- tmpU32no2
- = WEBRTC_SPL_RSHIFT_U32(WEBRTC_SPL_UMUL_32_16(tmpU32no3, gammaNoise), 5); // Q(prevQNoise+11)
- }
- if (sign > 0) {
- tmpU32no1 = inst->prevNoiseU32[i] + tmpU32no2; // Q(prevQNoise+11)
- } else {
- tmpU32no1 = inst->prevNoiseU32[i] - tmpU32no2; // Q(prevQNoise+11)
- }
- if (noiseUpdateU32 > tmpU32no1) {
- noiseUpdateU32 = tmpU32no1; // Q(prevQNoise+11)
- }
- }
- noiseU32[i] = noiseUpdateU32; // Q(prevQNoise+11)
- if (noiseUpdateU32 > maxNoiseU32) {
- maxNoiseU32 = noiseUpdateU32;
- }
-
- // conservative noise update
- // // original FLOAT code
- // if (prob_speech < PROB_RANGE) {
- // inst->avgMagnPause[i] = inst->avgMagnPause[i] + (1.0 - gamma_pause)*(magn[i] - inst->avgMagnPause[i]);
- // }
-
- tmp32no2 = WEBRTC_SPL_SHIFT_W32(inst->avgMagnPause[i], -nShifts);
- if (nonSpeechProbFinal[i] > ONE_MINUS_PROB_RANGE_Q8) {
- if (nShifts < 0) {
- tmp32no1 = (WebRtc_Word32)magnU16[i] - tmp32no2; // Q(qMagn)
- tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no1, ONE_MINUS_GAMMA_PAUSE_Q8); // Q(8+prevQMagn+nShifts)
- tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1 + 128, 8); // Q(qMagn)
- } else {
- tmp32no1 = WEBRTC_SPL_LSHIFT_W32((WebRtc_Word32)magnU16[i], nShifts)
- - inst->avgMagnPause[i]; // Q(qMagn+nShifts)
- tmp32no1 = WEBRTC_SPL_MUL_32_16(tmp32no1, ONE_MINUS_GAMMA_PAUSE_Q8); // Q(8+prevQMagn+nShifts)
- tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1 + (128 << nShifts), 8 + nShifts); // Q(qMagn)
- }
- tmp32no2 += tmp32no1; // Q(qMagn)
- }
- inst->avgMagnPause[i] = tmp32no2;
- } // end of frequency loop
-
- norm32no1 = WebRtcSpl_NormU32(maxNoiseU32);
- qNoise = inst->prevQNoise + norm32no1 - 5;
- // done with step 2: noise update
-
- // STEP 3: compute dd update of prior snr and post snr based on new noise estimate
- nShifts = inst->prevQNoise + 11 - qMagn;
- for (i = 0; i < inst->magnLen; i++) {
- // FLOAT code
- // // post and prior SNR
- // curNearSnr = 0.0;
- // if (magn[i] > noise[i])
- // {
- // curNearSnr = magn[i] / (noise[i] + 0.0001) - 1.0;
- // }
- // // DD estimate is sum of two terms: current estimate and previous estimate
- // // directed decision update of snrPrior
- // snrPrior = DD_PR_SNR * prevNearSnr[i] + (1.0 - DD_PR_SNR) * curNearSnr;
- // // gain filter
- // tmpFloat1 = inst->overdrive + snrPrior;
- // tmpFloat2 = snrPrior / tmpFloat1;
- // theFilter[i] = tmpFloat2;
-
- // calculate curNearSnr again, this is necessary because a new noise estimate has been made since then. for the original
- curNearSnr = 0; // Q11
- if (nShifts < 0) {
- // This case is equivalent with magn < noise which implies curNearSnr = 0;
- tmpMagnU32 = (WebRtc_UWord32)magnU16[i]; // Q(qMagn)
- tmpNoiseU32 = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], -nShifts); // Q(qMagn)
- } else if (nShifts > 17) {
- tmpMagnU32 = WEBRTC_SPL_LSHIFT_U32(magnU16[i], 17); // Q(qMagn+17)
- tmpNoiseU32 = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], nShifts - 17); // Q(qMagn+17)
- } else {
- tmpMagnU32 = WEBRTC_SPL_LSHIFT_U32((WebRtc_UWord32)magnU16[i], nShifts); // Q(qNoise_prev+11)
- tmpNoiseU32 = noiseU32[i]; // Q(qNoise_prev+11)
- }
- if (tmpMagnU32 > tmpNoiseU32) {
- tmpU32no1 = tmpMagnU32 - tmpNoiseU32; // Q(qCur)
- norm32no2 = WEBRTC_SPL_MIN(11, WebRtcSpl_NormU32(tmpU32no1));
- tmpU32no1 = WEBRTC_SPL_LSHIFT_U32(tmpU32no1, norm32no2); // Q(qCur+norm32no2)
- tmpU32no2 = WEBRTC_SPL_RSHIFT_U32(tmpNoiseU32, 11 - norm32no2); // Q(qCur+norm32no2-11)
- if (tmpU32no2 > 0) {
- tmpU32no1 = WEBRTC_SPL_UDIV(tmpU32no1, tmpU32no2); // Q11
- }
- curNearSnr = WEBRTC_SPL_MIN(satMax, tmpU32no1); // Q11
- }
-
- //directed decision update of priorSnr
- // FLOAT
- // priorSnr = DD_PR_SNR * prevNearSnr + (1.0-DD_PR_SNR) * curNearSnr;
-
- tmpU32no1 = WEBRTC_SPL_UMUL_32_16(prevNearSnr[i], DD_PR_SNR_Q11); // Q22
- tmpU32no2 = WEBRTC_SPL_UMUL_32_16(curNearSnr, ONE_MINUS_DD_PR_SNR_Q11); // Q22
- priorSnr = tmpU32no1 + tmpU32no2; // Q22
-
- //gain filter
- tmpU32no1 = (WebRtc_UWord32)(inst->overdrive)
- + WEBRTC_SPL_RSHIFT_U32(priorSnr + 8192, 14); // Q8
- assert(inst->overdrive > 0);
- tmpU16no1 = (WebRtc_UWord16)WEBRTC_SPL_UDIV(priorSnr + (tmpU32no1 >> 1), tmpU32no1); // Q14
- inst->noiseSupFilter[i] = WEBRTC_SPL_SAT(16384, tmpU16no1, inst->denoiseBound); // 16384 = Q14(1.0) // Q14
-
- // Weight in the parametric Wiener filter during startup
- if (inst->blockIndex < END_STARTUP_SHORT) {
- // Weight the two suppression filters
- tmpU32no1 = WEBRTC_SPL_UMUL_16_16(inst->noiseSupFilter[i],
- (WebRtc_UWord16)inst->blockIndex);
- tmpU32no2 = WEBRTC_SPL_UMUL_16_16(noiseSupFilterTmp[i],
- (WebRtc_UWord16)(END_STARTUP_SHORT
- - inst->blockIndex));
- tmpU32no1 += tmpU32no2;
- inst->noiseSupFilter[i] = (WebRtc_UWord16)WebRtcSpl_DivU32U16(tmpU32no1,
- END_STARTUP_SHORT);
- }
- } // end of loop over frequencies
- //done with step3
-
- // save noise and magnitude spectrum for next frame
- inst->prevQNoise = qNoise;
- inst->prevQMagn = qMagn;
- if (norm32no1 > 5) {
- for (i = 0; i < inst->magnLen; i++) {
- inst->prevNoiseU32[i] = WEBRTC_SPL_LSHIFT_U32(noiseU32[i], norm32no1 - 5); // Q(qNoise+11)
- inst->prevMagnU16[i] = magnU16[i]; // Q(qMagn)
- }
- } else {
- for (i = 0; i < inst->magnLen; i++) {
- inst->prevNoiseU32[i] = WEBRTC_SPL_RSHIFT_U32(noiseU32[i], 5 - norm32no1); // Q(qNoise+11)
- inst->prevMagnU16[i] = magnU16[i]; // Q(qMagn)
- }
- }
-
- WebRtcNsx_DataSynthesis(inst, outFrame);
-#ifdef NS_FILEDEBUG
- fwrite(outframe, sizeof(short), inst->blockLen10ms, inst->outfile);
-#endif
-
- //for H band:
- // only update data buffer, then apply time-domain gain is applied derived from L band
- if (inst->fs == 32000) {
- // update analysis buffer for H band
- // append new data to buffer FX
- WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX, inst->dataBufHBFX + inst->blockLen10ms, inst->anaLen - inst->blockLen10ms);
- WEBRTC_SPL_MEMCPY_W16(inst->dataBufHBFX + inst->anaLen - inst->blockLen10ms, speechFrameHB, inst->blockLen10ms);
- // range for averaging low band quantities for H band gain
-
- gainTimeDomainHB = 16384; // 16384 = Q14(1.0)
- //average speech prob from low band
- //average filter gain from low band
- //avg over second half (i.e., 4->8kHz) of freq. spectrum
- tmpU32no1 = 0; // Q12
- tmpU16no1 = 0; // Q8
- for (i = inst->anaLen2 - (inst->anaLen2 >> 2); i < inst->anaLen2; i++) {
- tmpU16no1 += nonSpeechProbFinal[i]; // Q8
- tmpU32no1 += (WebRtc_UWord32)(inst->noiseSupFilter[i]); // Q14
- }
- avgProbSpeechHB = (WebRtc_Word16)(4096
- - WEBRTC_SPL_RSHIFT_U16(tmpU16no1, inst->stages - 7)); // Q12
- avgFilterGainHB = (WebRtc_Word16)WEBRTC_SPL_RSHIFT_U32(
- tmpU32no1, inst->stages - 3); // Q14
-
- // // original FLOAT code
- // // gain based on speech probability:
- // avg_prob_speech_tt=(float)2.0*avg_prob_speech-(float)1.0;
- // gain_mod=(float)0.5*((float)1.0+(float)tanh(avg_prob_speech_tt)); // between 0 and 1
-
- // gain based on speech probability:
- // original expression: "0.5 * (1 + tanh(2x-1))"
- // avgProbSpeechHB has been anyway saturated to a value between 0 and 1 so the other cases don't have to be dealt with
- // avgProbSpeechHB and gainModHB are in Q12, 3607 = Q12(0.880615234375) which is a zero point of
- // |0.5 * (1 + tanh(2x-1)) - x| - |0.5 * (1 + tanh(2x-1)) - 0.880615234375| meaning that from that point the error of approximating
- // the expression with f(x) = x would be greater than the error of approximating the expression with f(x) = 0.880615234375
- // error: "|0.5 * (1 + tanh(2x-1)) - x| from x=0 to 0.880615234375" -> http://www.wolframalpha.com/input/?i=|0.5+*+(1+%2B+tanh(2x-1))+-+x|+from+x%3D0+to+0.880615234375
- // and: "|0.5 * (1 + tanh(2x-1)) - 0.880615234375| from x=0.880615234375 to 1" -> http://www.wolframalpha.com/input/?i=+|0.5+*+(1+%2B+tanh(2x-1))+-+0.880615234375|+from+x%3D0.880615234375+to+1
- gainModHB = WEBRTC_SPL_MIN(avgProbSpeechHB, 3607);
-
- // // original FLOAT code
- // //combine gain with low band gain
- // if (avg_prob_speech < (float)0.5) {
- // gain_time_domain_HB=(float)0.5*gain_mod+(float)0.5*avg_filter_gain;
- // }
- // else {
- // gain_time_domain_HB=(float)0.25*gain_mod+(float)0.75*avg_filter_gain;
- // }
-
-
- //combine gain with low band gain
- if (avgProbSpeechHB < 2048) {
- // 2048 = Q12(0.5)
- // the next two lines in float are "gain_time_domain = 0.5 * gain_mod + 0.5 * avg_filter_gain"; Q2(0.5) = 2 equals one left shift
- gainTimeDomainHB = (gainModHB << 1) + (avgFilterGainHB >> 1); // Q14
- } else {
- // "gain_time_domain = 0.25 * gain_mod + 0.75 * agv_filter_gain;"
- gainTimeDomainHB = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(3, avgFilterGainHB, 2); // 3 = Q2(0.75); Q14
- gainTimeDomainHB += gainModHB; // Q14
- }
- //make sure gain is within flooring range
- gainTimeDomainHB
- = WEBRTC_SPL_SAT(16384, gainTimeDomainHB, (WebRtc_Word16)(inst->denoiseBound)); // 16384 = Q14(1.0)
-
-
- //apply gain
- for (i = 0; i < inst->blockLen10ms; i++) {
- outFrameHB[i]
- = (WebRtc_Word16)WEBRTC_SPL_MUL_16_16_RSFT(gainTimeDomainHB, inst->dataBufHBFX[i], 14); // Q0
- }
- } // end of H band gain computation
-
- return 0;
-}
-
-
diff --git a/trunk/src/modules/audio_processing/ns/nsx_core.h b/trunk/src/modules/audio_processing/ns/nsx_core.h
deleted file mode 100644
index 0a0faf9..0000000
--- a/trunk/src/modules/audio_processing/ns/nsx_core.h
+++ /dev/null
@@ -1,222 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_CORE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_CORE_H_
-
-#include "typedefs.h"
-#include "signal_processing_library.h"
-
-#include "nsx_defines.h"
-
-#ifdef NS_FILEDEBUG
-#include <stdio.h>
-#endif
-
-typedef struct NsxInst_t_ {
- WebRtc_UWord32 fs;
-
- const WebRtc_Word16* window;
- WebRtc_Word16 analysisBuffer[ANAL_BLOCKL_MAX];
- WebRtc_Word16 synthesisBuffer[ANAL_BLOCKL_MAX];
- WebRtc_UWord16 noiseSupFilter[HALF_ANAL_BLOCKL];
- WebRtc_UWord16 overdrive; /* Q8 */
- WebRtc_UWord16 denoiseBound; /* Q14 */
- const WebRtc_Word16* factor2Table;
- WebRtc_Word16 noiseEstLogQuantile[SIMULT* HALF_ANAL_BLOCKL];
- WebRtc_Word16 noiseEstDensity[SIMULT* HALF_ANAL_BLOCKL];
- WebRtc_Word16 noiseEstCounter[SIMULT];
- WebRtc_Word16 noiseEstQuantile[HALF_ANAL_BLOCKL];
-
- WebRtc_Word16 anaLen;
- int anaLen2;
- int magnLen;
- int aggrMode;
- int stages;
- int initFlag;
- int gainMap;
-
- WebRtc_Word32 maxLrt;
- WebRtc_Word32 minLrt;
- WebRtc_Word32 logLrtTimeAvgW32[HALF_ANAL_BLOCKL]; //log lrt factor with time-smoothing in Q8
- WebRtc_Word32 featureLogLrt;
- WebRtc_Word32 thresholdLogLrt;
- WebRtc_Word16 weightLogLrt;
-
- WebRtc_UWord32 featureSpecDiff;
- WebRtc_UWord32 thresholdSpecDiff;
- WebRtc_Word16 weightSpecDiff;
-
- WebRtc_UWord32 featureSpecFlat;
- WebRtc_UWord32 thresholdSpecFlat;
- WebRtc_Word16 weightSpecFlat;
-
- WebRtc_Word32 avgMagnPause[HALF_ANAL_BLOCKL]; //conservative estimate of noise spectrum
- WebRtc_UWord32 magnEnergy;
- WebRtc_UWord32 sumMagn;
- WebRtc_UWord32 curAvgMagnEnergy;
- WebRtc_UWord32 timeAvgMagnEnergy;
- WebRtc_UWord32 timeAvgMagnEnergyTmp;
-
- WebRtc_UWord32 whiteNoiseLevel; //initial noise estimate
- WebRtc_UWord32 initMagnEst[HALF_ANAL_BLOCKL];//initial magnitude spectrum estimate
- WebRtc_Word32 pinkNoiseNumerator; //pink noise parameter: numerator
- WebRtc_Word32 pinkNoiseExp; //pink noise parameter: power of freq
- int minNorm; //smallest normalization factor
- int zeroInputSignal; //zero input signal flag
-
- WebRtc_UWord32 prevNoiseU32[HALF_ANAL_BLOCKL]; //noise spectrum from previous frame
- WebRtc_UWord16 prevMagnU16[HALF_ANAL_BLOCKL]; //magnitude spectrum from previous frame
- WebRtc_Word16 priorNonSpeechProb; //prior speech/noise probability // Q14
-
- int blockIndex; //frame index counter
- int modelUpdate; //parameter for updating or estimating thresholds/weights for prior model
- int cntThresUpdate;
-
- //histograms for parameter estimation
- WebRtc_Word16 histLrt[HIST_PAR_EST];
- WebRtc_Word16 histSpecFlat[HIST_PAR_EST];
- WebRtc_Word16 histSpecDiff[HIST_PAR_EST];
-
- //quantities for high band estimate
- WebRtc_Word16 dataBufHBFX[ANAL_BLOCKL_MAX]; /* Q0 */
-
- int qNoise;
- int prevQNoise;
- int prevQMagn;
- int blockLen10ms;
-
- WebRtc_Word16 real[ANAL_BLOCKL_MAX];
- WebRtc_Word16 imag[ANAL_BLOCKL_MAX];
- WebRtc_Word32 energyIn;
- int scaleEnergyIn;
- int normData;
-
-} NsxInst_t;
-
-#ifdef __cplusplus
-extern "C"
-{
-#endif
-
-/****************************************************************************
- * WebRtcNsx_InitCore(...)
- *
- * This function initializes a noise suppression instance
- *
- * Input:
- * - inst : Instance that should be initialized
- * - fs : Sampling frequency
- *
- * Output:
- * - inst : Initialized instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-WebRtc_Word32 WebRtcNsx_InitCore(NsxInst_t* inst, WebRtc_UWord32 fs);
-
-/****************************************************************************
- * WebRtcNsx_set_policy_core(...)
- *
- * This changes the aggressiveness of the noise suppression method.
- *
- * Input:
- * - inst : Instance that should be initialized
- * - mode : 0: Mild (6 dB), 1: Medium (10 dB), 2: Aggressive (15 dB)
- *
- * Output:
- * - inst : Initialized instance
- *
- * Return value : 0 - Ok
- * -1 - Error
- */
-int WebRtcNsx_set_policy_core(NsxInst_t* inst, int mode);
-
-/****************************************************************************
- * WebRtcNsx_ProcessCore
- *
- * Do noise suppression.
- *
- * Input:
- * - inst : Instance that should be initialized
- * - inFrameLow : Input speech frame for lower band
- * - inFrameHigh : Input speech frame for higher band
- *
- * Output:
- * - inst : Updated instance
- * - outFrameLow : Output speech frame for lower band
- * - outFrameHigh : Output speech frame for higher band
- *
- * Return value : 0 - OK
- * -1 - Error
- */
-int WebRtcNsx_ProcessCore(NsxInst_t* inst,
- short* inFrameLow,
- short* inFrameHigh,
- short* outFrameLow,
- short* outFrameHigh);
-
-/****************************************************************************
- * Some function pointers, for internal functions shared by ARM NEON and
- * generic C code.
- */
-// Noise Estimation.
-typedef void (*NoiseEstimation)(NsxInst_t* inst,
- uint16_t* magn,
- uint32_t* noise,
- int16_t* q_noise);
-extern NoiseEstimation WebRtcNsx_NoiseEstimation;
-
-// Filter the data in the frequency domain, and create spectrum.
-typedef void (*PrepareSpectrum)(NsxInst_t* inst,
- int16_t* freq_buff);
-extern PrepareSpectrum WebRtcNsx_PrepareSpectrum;
-
-// For the noise supression process, synthesis, read out fully processed
-// segment, and update synthesis buffer.
-typedef void (*SynthesisUpdate)(NsxInst_t* inst,
- int16_t* out_frame,
- int16_t gain_factor);
-extern SynthesisUpdate WebRtcNsx_SynthesisUpdate;
-
-// Update analysis buffer for lower band, and window data before FFT.
-typedef void (*AnalysisUpdate)(NsxInst_t* inst,
- int16_t* out,
- int16_t* new_speech);
-extern AnalysisUpdate WebRtcNsx_AnalysisUpdate;
-
-// Denormalize the input buffer.
-typedef void (*Denormalize)(NsxInst_t* inst,
- int16_t* in,
- int factor);
-extern Denormalize WebRtcNsx_Denormalize;
-
-// Create a complex number buffer, as the intput interleaved with zeros,
-// and normalize it.
-typedef void (*CreateComplexBuffer)(NsxInst_t* inst,
- int16_t* in,
- int16_t* out);
-extern CreateComplexBuffer WebRtcNsx_CreateComplexBuffer;
-
-/****************************************************************************
- * Initialization of the above function pointers for ARM Neon.
- */
-void WebRtcNsx_InitNeon(void);
-
-extern const WebRtc_Word16 WebRtcNsx_kLogTable[9];
-extern const WebRtc_Word16 WebRtcNsx_kLogTableFrac[256];
-extern const WebRtc_Word16 WebRtcNsx_kCounterDiv[201];
-
-#ifdef __cplusplus
-}
-#endif
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_CORE_H_
diff --git a/trunk/src/modules/audio_processing/ns/nsx_core_neon.c b/trunk/src/modules/audio_processing/ns/nsx_core_neon.c
deleted file mode 100644
index 2f85abd..0000000
--- a/trunk/src/modules/audio_processing/ns/nsx_core_neon.c
+++ /dev/null
@@ -1,734 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "nsx_core.h"
-
-#include <arm_neon.h>
-#include <assert.h>
-
-// Update the noise estimation information.
-static void UpdateNoiseEstimateNeon(NsxInst_t* inst, int offset) {
- int i = 0;
- const int16_t kExp2Const = 11819; // Q13
- int16_t* ptr_noiseEstLogQuantile = NULL;
- int16_t* ptr_noiseEstQuantile = NULL;
- int16x4_t kExp2Const16x4 = vdup_n_s16(kExp2Const);
- int32x4_t twentyOne32x4 = vdupq_n_s32(21);
- int32x4_t constA32x4 = vdupq_n_s32(0x1fffff);
- int32x4_t constB32x4 = vdupq_n_s32(0x200000);
-
- int16_t tmp16 = WebRtcSpl_MaxValueW16(inst->noiseEstLogQuantile + offset,
- inst->magnLen);
-
- // Guarantee a Q-domain as high as possible and still fit in int16
- inst->qNoise = 14 - (int) WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(kExp2Const,
- tmp16,
- 21);
-
- int32x4_t qNoise32x4 = vdupq_n_s32(inst->qNoise);
-
- for (ptr_noiseEstLogQuantile = &inst->noiseEstLogQuantile[offset],
- ptr_noiseEstQuantile = &inst->noiseEstQuantile[0];
- ptr_noiseEstQuantile < &inst->noiseEstQuantile[inst->magnLen - 3];
- ptr_noiseEstQuantile += 4, ptr_noiseEstLogQuantile += 4) {
-
- // tmp32no2 = WEBRTC_SPL_MUL_16_16(kExp2Const,
- // inst->noiseEstLogQuantile[offset + i]);
- int16x4_t v16x4 = vld1_s16(ptr_noiseEstLogQuantile);
- int32x4_t v32x4B = vmull_s16(v16x4, kExp2Const16x4);
-
- // tmp32no1 = (0x00200000 | (tmp32no2 & 0x001FFFFF)); // 2^21 + frac
- int32x4_t v32x4A = vandq_s32(v32x4B, constA32x4);
- v32x4A = vorrq_s32(v32x4A, constB32x4);
-
- // tmp16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32no2, 21);
- v32x4B = vshrq_n_s32(v32x4B, 21);
-
- // tmp16 -= 21;// shift 21 to get result in Q0
- v32x4B = vsubq_s32(v32x4B, twentyOne32x4);
-
- // tmp16 += (int16_t) inst->qNoise;
- // shift to get result in Q(qNoise)
- v32x4B = vaddq_s32(v32x4B, qNoise32x4);
-
- // if (tmp16 < 0) {
- // tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1, -tmp16);
- // } else {
- // tmp32no1 = WEBRTC_SPL_LSHIFT_W32(tmp32no1, tmp16);
- // }
- v32x4B = vshlq_s32(v32x4A, v32x4B);
-
- // tmp16 = WebRtcSpl_SatW32ToW16(tmp32no1);
- v16x4 = vqmovn_s32(v32x4B);
-
- //inst->noiseEstQuantile[i] = tmp16;
- vst1_s16(ptr_noiseEstQuantile, v16x4);
- }
-
- // Last iteration:
-
- // inst->quantile[i]=exp(inst->lquantile[offset+i]);
- // in Q21
- int32_t tmp32no2 = WEBRTC_SPL_MUL_16_16(kExp2Const,
- *ptr_noiseEstLogQuantile);
- int32_t tmp32no1 = (0x00200000 | (tmp32no2 & 0x001FFFFF)); // 2^21 + frac
-
- tmp16 = (int16_t) WEBRTC_SPL_RSHIFT_W32(tmp32no2, 21);
- tmp16 -= 21;// shift 21 to get result in Q0
- tmp16 += (int16_t) inst->qNoise; //shift to get result in Q(qNoise)
- if (tmp16 < 0) {
- tmp32no1 = WEBRTC_SPL_RSHIFT_W32(tmp32no1, -tmp16);
- } else {
- tmp32no1 = WEBRTC_SPL_LSHIFT_W32(tmp32no1, tmp16);
- }
- *ptr_noiseEstQuantile = WebRtcSpl_SatW32ToW16(tmp32no1);
-}
-
-// Noise Estimation
-static void NoiseEstimationNeon(NsxInst_t* inst,
- uint16_t* magn,
- uint32_t* noise,
- int16_t* q_noise) {
- int16_t lmagn[HALF_ANAL_BLOCKL], counter, countDiv;
- int16_t countProd, delta, zeros, frac;
- int16_t log2, tabind, logval, tmp16, tmp16no1, tmp16no2;
- const int16_t log2_const = 22713;
- const int16_t width_factor = 21845;
-
- int i, s, offset;
-
- tabind = inst->stages - inst->normData;
- assert(tabind < 9);
- assert(tabind > -9);
- if (tabind < 0) {
- logval = -WebRtcNsx_kLogTable[-tabind];
- } else {
- logval = WebRtcNsx_kLogTable[tabind];
- }
-
- int16x8_t logval_16x8 = vdupq_n_s16(logval);
-
- // lmagn(i)=log(magn(i))=log(2)*log2(magn(i))
- // magn is in Q(-stages), and the real lmagn values are:
- // real_lmagn(i)=log(magn(i)*2^stages)=log(magn(i))+log(2^stages)
- // lmagn in Q8
- for (i = 0; i < inst->magnLen; i++) {
- if (magn[i]) {
- zeros = WebRtcSpl_NormU32((uint32_t)magn[i]);
- frac = (int16_t)((((uint32_t)magn[i] << zeros)
- & 0x7FFFFFFF) >> 23);
- assert(frac < 256);
- // log2(magn(i))
- log2 = (int16_t)(((31 - zeros) << 8)
- + WebRtcNsx_kLogTableFrac[frac]);
- // log2(magn(i))*log(2)
- lmagn[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(log2, log2_const, 15);
- // + log(2^stages)
- lmagn[i] += logval;
- } else {
- lmagn[i] = logval;
- }
- }
-
- int16x4_t Q3_16x4 = vdup_n_s16(3);
- int16x8_t WIDTHQ8_16x8 = vdupq_n_s16(WIDTH_Q8);
- int16x8_t WIDTHFACTOR_16x8 = vdupq_n_s16(width_factor);
-
- int16_t factor = FACTOR_Q7;
- if (inst->blockIndex < END_STARTUP_LONG)
- factor = FACTOR_Q7_STARTUP;
-
- // Loop over simultaneous estimates
- for (s = 0; s < SIMULT; s++) {
- offset = s * inst->magnLen;
-
- // Get counter values from state
- counter = inst->noiseEstCounter[s];
- assert(counter < 201);
- countDiv = WebRtcNsx_kCounterDiv[counter];
- countProd = (int16_t)WEBRTC_SPL_MUL_16_16(counter, countDiv);
-
- // quant_est(...)
- int16_t deltaBuff[8];
- int16x4_t tmp16x4_0;
- int16x4_t tmp16x4_1;
- int16x4_t countDiv_16x4 = vdup_n_s16(countDiv);
- int16x8_t countProd_16x8 = vdupq_n_s16(countProd);
- int16x8_t tmp16x8_0 = vdupq_n_s16(countDiv);
- int16x8_t prod16x8 = vqrdmulhq_s16(WIDTHFACTOR_16x8, tmp16x8_0);
- int16x8_t tmp16x8_1;
- int16x8_t tmp16x8_2;
- int16x8_t tmp16x8_3;
- int16x8_t tmp16x8_4;
- int16x8_t tmp16x8_5;
- int32x4_t tmp32x4;
-
- for (i = 0; i < inst->magnLen - 7; i += 8) {
- // Compute delta.
- // Smaller step size during startup. This prevents from using
- // unrealistic values causing overflow.
- tmp16x8_0 = vdupq_n_s16(factor);
- vst1q_s16(deltaBuff, tmp16x8_0);
-
- int j;
- for (j = 0; j < 8; j++) {
- if (inst->noiseEstDensity[offset + i + j] > 512) {
- // Get values for deltaBuff by shifting intead of dividing.
- int factor = WebRtcSpl_NormW16(inst->noiseEstDensity[offset + i + j]);
- deltaBuff[j] = (int16_t)(FACTOR_Q16 >> (14 - factor));
- }
- }
-
- // Update log quantile estimate
-
- // tmp16 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(delta, countDiv, 14);
- tmp32x4 = vmull_s16(vld1_s16(&deltaBuff[0]), countDiv_16x4);
- tmp16x4_1 = vshrn_n_s32(tmp32x4, 14);
- tmp32x4 = vmull_s16(vld1_s16(&deltaBuff[4]), countDiv_16x4);
- tmp16x4_0 = vshrn_n_s32(tmp32x4, 14);
- tmp16x8_0 = vcombine_s16(tmp16x4_1, tmp16x4_0); // Keep for several lines.
-
- // prepare for the "if" branch
- // tmp16 += 2;
- // tmp16_1 = (Word16)(tmp16>>2);
- tmp16x8_1 = vrshrq_n_s16(tmp16x8_0, 2);
-
- // inst->noiseEstLogQuantile[offset+i] + tmp16_1;
- tmp16x8_2 = vld1q_s16(&inst->noiseEstLogQuantile[offset + i]); // Keep
- tmp16x8_1 = vaddq_s16(tmp16x8_2, tmp16x8_1); // Keep for several lines
-
- // Prepare for the "else" branch
- // tmp16 += 1;
- // tmp16_1 = (Word16)(tmp16>>1);
- tmp16x8_0 = vrshrq_n_s16(tmp16x8_0, 1);
-
- // tmp16_2 = (Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16_1,3,1);
- tmp32x4 = vmull_s16(vget_low_s16(tmp16x8_0), Q3_16x4);
- tmp16x4_1 = vshrn_n_s32(tmp32x4, 1);
-
- // tmp16_2 = (Word16)WEBRTC_SPL_MUL_16_16_RSFT(tmp16_1,3,1);
- tmp32x4 = vmull_s16(vget_high_s16(tmp16x8_0), Q3_16x4);
- tmp16x4_0 = vshrn_n_s32(tmp32x4, 1);
-
- // inst->noiseEstLogQuantile[offset + i] - tmp16_2;
- tmp16x8_0 = vcombine_s16(tmp16x4_1, tmp16x4_0); // keep
- tmp16x8_0 = vsubq_s16(tmp16x8_2, tmp16x8_0);
-
- // logval is the smallest fixed point representation we can have. Values
- // below that will correspond to values in the interval [0, 1], which
- // can't possibly occur.
- tmp16x8_0 = vmaxq_s16(tmp16x8_0, logval_16x8);
-
- // Do the if-else branches:
- tmp16x8_3 = vld1q_s16(&lmagn[i]); // keep for several lines
- tmp16x8_5 = vsubq_s16(tmp16x8_3, tmp16x8_2);
- __asm__("vcgt.s16 %q0, %q1, #0"::"w"(tmp16x8_4), "w"(tmp16x8_5));
- __asm__("vbit %q0, %q1, %q2"::
- "w"(tmp16x8_2), "w"(tmp16x8_1), "w"(tmp16x8_4));
- __asm__("vbif %q0, %q1, %q2"::
- "w"(tmp16x8_2), "w"(tmp16x8_0), "w"(tmp16x8_4));
- vst1q_s16(&inst->noiseEstLogQuantile[offset + i], tmp16x8_2);
-
- // Update density estimate
- // tmp16_1 + tmp16_2
- tmp16x8_1 = vld1q_s16(&inst->noiseEstDensity[offset + i]);
- tmp16x8_0 = vqrdmulhq_s16(tmp16x8_1, countProd_16x8);
- tmp16x8_0 = vaddq_s16(tmp16x8_0, prod16x8);
-
- // lmagn[i] - inst->noiseEstLogQuantile[offset + i]
- tmp16x8_3 = vsubq_s16(tmp16x8_3, tmp16x8_2);
- tmp16x8_3 = vabsq_s16(tmp16x8_3);
- tmp16x8_4 = vcgtq_s16(WIDTHQ8_16x8, tmp16x8_3);
- __asm__("vbit %q0, %q1, %q2"::
- "w"(tmp16x8_1), "w"(tmp16x8_0), "w"(tmp16x8_4));
- vst1q_s16(&inst->noiseEstDensity[offset + i], tmp16x8_1);
- } // End loop over magnitude spectrum
-
- // Last iteration over magnitude spectrum:
- // compute delta
- if (inst->noiseEstDensity[offset + i] > 512) {
- // Get values for deltaBuff by shifting intead of dividing.
- int factor = WebRtcSpl_NormW16(inst->noiseEstDensity[offset + i]);
- delta = (int16_t)(FACTOR_Q16 >> (14 - factor));
- } else {
- delta = FACTOR_Q7;
- if (inst->blockIndex < END_STARTUP_LONG) {
- // Smaller step size during startup. This prevents from using
- // unrealistic values causing overflow.
- delta = FACTOR_Q7_STARTUP;
- }
- }
- // update log quantile estimate
- tmp16 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(delta, countDiv, 14);
- if (lmagn[i] > inst->noiseEstLogQuantile[offset + i]) {
- // +=QUANTILE*delta/(inst->counter[s]+1) QUANTILE=0.25, =1 in Q2
- // CounterDiv=1/(inst->counter[s]+1) in Q15
- tmp16 += 2;
- tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 2);
- inst->noiseEstLogQuantile[offset + i] += tmp16no1;
- } else {
- tmp16 += 1;
- tmp16no1 = WEBRTC_SPL_RSHIFT_W16(tmp16, 1);
- // *(1-QUANTILE), in Q2 QUANTILE=0.25, 1-0.25=0.75=3 in Q2
- tmp16no2 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(tmp16no1, 3, 1);
- inst->noiseEstLogQuantile[offset + i] -= tmp16no2;
- if (inst->noiseEstLogQuantile[offset + i] < logval) {
- // logval is the smallest fixed point representation we can have.
- // Values below that will correspond to values in the interval
- // [0, 1], which can't possibly occur.
- inst->noiseEstLogQuantile[offset + i] = logval;
- }
- }
-
- // update density estimate
- if (WEBRTC_SPL_ABS_W16(lmagn[i] - inst->noiseEstLogQuantile[offset + i])
- < WIDTH_Q8) {
- tmp16no1 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
- inst->noiseEstDensity[offset + i], countProd, 15);
- tmp16no2 = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
- width_factor, countDiv, 15);
- inst->noiseEstDensity[offset + i] = tmp16no1 + tmp16no2;
- }
-
-
- if (counter >= END_STARTUP_LONG) {
- inst->noiseEstCounter[s] = 0;
- if (inst->blockIndex >= END_STARTUP_LONG) {
- UpdateNoiseEstimateNeon(inst, offset);
- }
- }
- inst->noiseEstCounter[s]++;
-
- } // end loop over simultaneous estimates
-
- // Sequentially update the noise during startup
- if (inst->blockIndex < END_STARTUP_LONG) {
- UpdateNoiseEstimateNeon(inst, offset);
- }
-
- for (i = 0; i < inst->magnLen; i++) {
- noise[i] = (uint32_t)(inst->noiseEstQuantile[i]); // Q(qNoise)
- }
- (*q_noise) = (int16_t)inst->qNoise;
-}
-
-// Filter the data in the frequency domain, and create spectrum.
-static void PrepareSpectrumNeon(NsxInst_t* inst, int16_t* freq_buf) {
-
- // (1) Filtering.
-
- // Fixed point C code for the next block is as follows:
- // for (i = 0; i < inst->magnLen; i++) {
- // inst->real[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(inst->real[i],
- // (int16_t)(inst->noiseSupFilter[i]), 14); // Q(normData-stages)
- // inst->imag[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(inst->imag[i],
- // (int16_t)(inst->noiseSupFilter[i]), 14); // Q(normData-stages)
- // }
-
- int16_t* ptr_real = &inst->real[0];
- int16_t* ptr_imag = &inst->imag[0];
- uint16_t* ptr_noiseSupFilter = &inst->noiseSupFilter[0];
-
- // Filter the rest in the frequency domain.
- for (; ptr_real < &inst->real[inst->magnLen - 1];) {
- // Loop unrolled once. Both pointers are incremented by 4 twice.
- __asm__ __volatile__(
- "vld1.16 d20, [%[ptr_real]]\n\t"
- "vld1.16 d22, [%[ptr_imag]]\n\t"
- "vld1.16 d23, [%[ptr_noiseSupFilter]]!\n\t"
- "vmull.s16 q10, d20, d23\n\t"
- "vmull.s16 q11, d22, d23\n\t"
- "vshrn.s32 d20, q10, #14\n\t"
- "vshrn.s32 d22, q11, #14\n\t"
- "vst1.16 d20, [%[ptr_real]]!\n\t"
- "vst1.16 d22, [%[ptr_imag]]!\n\t"
-
- "vld1.16 d18, [%[ptr_real]]\n\t"
- "vld1.16 d24, [%[ptr_imag]]\n\t"
- "vld1.16 d25, [%[ptr_noiseSupFilter]]!\n\t"
- "vmull.s16 q9, d18, d25\n\t"
- "vmull.s16 q12, d24, d25\n\t"
- "vshrn.s32 d18, q9, #14\n\t"
- "vshrn.s32 d24, q12, #14\n\t"
- "vst1.16 d18, [%[ptr_real]]!\n\t"
- "vst1.16 d24, [%[ptr_imag]]!\n\t"
-
- // Specify constraints.
- :[ptr_imag]"+r"(ptr_imag),
- [ptr_real]"+r"(ptr_real),
- [ptr_noiseSupFilter]"+r"(ptr_noiseSupFilter)
- :
- :"d18", "d19", "d20", "d21", "d22", "d23", "d24", "d25",
- "q9", "q10", "q11", "q12"
- );
- }
-
- // Filter the last pair of elements in the frequency domain.
- *ptr_real = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(*ptr_real,
- (int16_t)(*ptr_noiseSupFilter), 14); // Q(normData-stages)
- *ptr_imag = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT(*ptr_imag,
- (int16_t)(*ptr_noiseSupFilter), 14); // Q(normData-stages)
-
- // (2) Create spectrum.
-
- // Fixed point C code for the rest of the function is as follows:
- // freq_buf[0] = inst->real[0];
- // freq_buf[1] = -inst->imag[0];
- // for (i = 1, j = 2; i < inst->anaLen2; i += 1, j += 2) {
- // tmp16 = (inst->anaLen << 1) - j;
- // freq_buf[j] = inst->real[i];
- // freq_buf[j + 1] = -inst->imag[i];
- // freq_buf[tmp16] = inst->real[i];
- // freq_buf[tmp16 + 1] = inst->imag[i];
- // }
- // freq_buf[inst->anaLen] = inst->real[inst->anaLen2];
- // freq_buf[inst->anaLen + 1] = -inst->imag[inst->anaLen2];
-
- freq_buf[0] = inst->real[0];
- freq_buf[1] = -inst->imag[0];
-
- int offset = -16;
- int16_t* ptr_realImag1 = &freq_buf[2];
- int16_t* ptr_realImag2 = ptr_realImag2 = &freq_buf[(inst->anaLen << 1) - 8];
- ptr_real = &inst->real[1];
- ptr_imag = &inst->imag[1];
- for (; ptr_real < &inst->real[inst->anaLen2 - 11];) {
- // Loop unrolled once. All pointers are incremented twice.
- __asm__ __volatile__(
- "vld1.16 d22, [%[ptr_real]]!\n\t"
- "vld1.16 d23, [%[ptr_imag]]!\n\t"
- // Negate and interleave:
- "vmov.s16 d20, d22\n\t"
- "vneg.s16 d21, d23\n\t"
- "vzip.16 d20, d21\n\t"
- // Write 8 elements to &freq_buf[j]
- "vst1.16 {d20, d21}, [%[ptr_realImag1]]!\n\t"
- // Interleave and reverse elements:
- "vzip.16 d22, d23\n\t"
- "vrev64.32 d18, d23\n\t"
- "vrev64.32 d19, d22\n\t"
- // Write 8 elements to &freq_buf[tmp16]
- "vst1.16 {d18, d19}, [%[ptr_realImag2]], %[offset]\n\t"
-
- "vld1.16 d22, [%[ptr_real]]!\n\t"
- "vld1.16 d23, [%[ptr_imag]]!\n\t"
- // Negate and interleave:
- "vmov.s16 d20, d22\n\t"
- "vneg.s16 d21, d23\n\t"
- "vzip.16 d20, d21\n\t"
- // Write 8 elements to &freq_buf[j]
- "vst1.16 {d20, d21}, [%[ptr_realImag1]]!\n\t"
- // Interleave and reverse elements:
- "vzip.16 d22, d23\n\t"
- "vrev64.32 d18, d23\n\t"
- "vrev64.32 d19, d22\n\t"
- // Write 8 elements to &freq_buf[tmp16]
- "vst1.16 {d18, d19}, [%[ptr_realImag2]], %[offset]\n\t"
-
- // Specify constraints.
- :[ptr_imag]"+r"(ptr_imag),
- [ptr_real]"+r"(ptr_real),
- [ptr_realImag1]"+r"(ptr_realImag1),
- [ptr_realImag2]"+r"(ptr_realImag2)
- :[offset]"r"(offset)
- :"d18", "d19", "d20", "d21", "d22", "d23"
- );
- }
- for (ptr_realImag2 += 6;
- ptr_real <= &inst->real[inst->anaLen2];
- ptr_real += 1, ptr_imag += 1, ptr_realImag1 += 2, ptr_realImag2 -= 2) {
- *ptr_realImag1 = *ptr_real;
- *(ptr_realImag1 + 1) = -(*ptr_imag);
- *ptr_realImag2 = *ptr_real;
- *(ptr_realImag2 + 1) = *ptr_imag;
- }
-
- freq_buf[inst->anaLen] = inst->real[inst->anaLen2];
- freq_buf[inst->anaLen + 1] = -inst->imag[inst->anaLen2];
-}
-
-// Denormalize the input buffer.
-static __inline void DenormalizeNeon(NsxInst_t* inst, int16_t* in, int factor) {
- int16_t* ptr_real = &inst->real[0];
- int16_t* ptr_in = &in[0];
-
- __asm__ __volatile__("vdup.32 q10, %0" ::
- "r"((int32_t)(factor - inst->normData)) : "q10");
- for (; ptr_real < &inst->real[inst->anaLen];) {
-
- // Loop unrolled once. Both pointers are incremented.
- __asm__ __volatile__(
- // tmp32 = WEBRTC_SPL_SHIFT_W32((int32_t)in[j],
- // factor - inst->normData);
- "vld2.16 {d24, d25}, [%[ptr_in]]!\n\t"
- "vmovl.s16 q12, d24\n\t"
- "vshl.s32 q12, q10\n\t"
- // inst->real[i] = WebRtcSpl_SatW32ToW16(tmp32); // Q0
- "vqmovn.s32 d24, q12\n\t"
- "vst1.16 d24, [%[ptr_real]]!\n\t"
-
- // tmp32 = WEBRTC_SPL_SHIFT_W32((int32_t)in[j],
- // factor - inst->normData);
- "vld2.16 {d22, d23}, [%[ptr_in]]!\n\t"
- "vmovl.s16 q11, d22\n\t"
- "vshl.s32 q11, q10\n\t"
- // inst->real[i] = WebRtcSpl_SatW32ToW16(tmp32); // Q0
- "vqmovn.s32 d22, q11\n\t"
- "vst1.16 d22, [%[ptr_real]]!\n\t"
-
- // Specify constraints.
- :[ptr_in]"+r"(ptr_in),
- [ptr_real]"+r"(ptr_real)
- :
- :"d22", "d23", "d24", "d25"
- );
- }
-}
-
-// For the noise supress process, synthesis, read out fully processed segment,
-// and update synthesis buffer.
-static void SynthesisUpdateNeon(NsxInst_t* inst,
- int16_t* out_frame,
- int16_t gain_factor) {
- int16_t* ptr_real = &inst->real[0];
- int16_t* ptr_syn = &inst->synthesisBuffer[0];
- int16_t* ptr_window = &inst->window[0];
-
- // synthesis
- __asm__ __volatile__("vdup.16 d24, %0" : : "r"(gain_factor) : "d24");
- // Loop unrolled once. All pointers are incremented in the assembly code.
- for (; ptr_syn < &inst->synthesisBuffer[inst->anaLen];) {
- __asm__ __volatile__(
- // Load variables.
- "vld1.16 d22, [%[ptr_real]]!\n\t"
- "vld1.16 d23, [%[ptr_window]]!\n\t"
- "vld1.16 d25, [%[ptr_syn]]\n\t"
- // tmp16a = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
- // inst->window[i], inst->real[i], 14); // Q0, window in Q14
- "vmull.s16 q11, d22, d23\n\t"
- "vrshrn.i32 d22, q11, #14\n\t"
- // tmp32 = WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(tmp16a, gain_factor, 13);
- "vmull.s16 q11, d24, d22\n\t"
- // tmp16b = WebRtcSpl_SatW32ToW16(tmp32); // Q0
- "vqrshrn.s32 d22, q11, #13\n\t"
- // inst->synthesisBuffer[i] = WEBRTC_SPL_ADD_SAT_W16(
- // inst->synthesisBuffer[i], tmp16b); // Q0
- "vqadd.s16 d25, d22\n\t"
- "vst1.16 d25, [%[ptr_syn]]!\n\t"
-
- // Load variables.
- "vld1.16 d26, [%[ptr_real]]!\n\t"
- "vld1.16 d27, [%[ptr_window]]!\n\t"
- "vld1.16 d28, [%[ptr_syn]]\n\t"
- // tmp16a = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
- // inst->window[i], inst->real[i], 14); // Q0, window in Q14
- "vmull.s16 q13, d26, d27\n\t"
- "vrshrn.i32 d26, q13, #14\n\t"
- // tmp32 = WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(tmp16a, gain_factor, 13);
- "vmull.s16 q13, d24, d26\n\t"
- // tmp16b = WebRtcSpl_SatW32ToW16(tmp32); // Q0
- "vqrshrn.s32 d26, q13, #13\n\t"
- // inst->synthesisBuffer[i] = WEBRTC_SPL_ADD_SAT_W16(
- // inst->synthesisBuffer[i], tmp16b); // Q0
- "vqadd.s16 d28, d26\n\t"
- "vst1.16 d28, [%[ptr_syn]]!\n\t"
-
- // Specify constraints.
- :[ptr_real]"+r"(ptr_real),
- [ptr_window]"+r"(ptr_window),
- [ptr_syn]"+r"(ptr_syn)
- :
- :"d22", "d23", "d24", "d25", "d26", "d27", "d28", "q11", "q12", "q13"
- );
- }
-
- int16_t* ptr_out = &out_frame[0];
- ptr_syn = &inst->synthesisBuffer[0];
- // read out fully processed segment
- for (; ptr_syn < &inst->synthesisBuffer[inst->blockLen10ms];) {
- // Loop unrolled once. Both pointers are incremented in the assembly code.
- __asm__ __volatile__(
- // out_frame[i] = inst->synthesisBuffer[i]; // Q0
- "vld1.16 {d22, d23}, [%[ptr_syn]]!\n\t"
- "vld1.16 {d24, d25}, [%[ptr_syn]]!\n\t"
- "vst1.16 {d22, d23}, [%[ptr_out]]!\n\t"
- "vst1.16 {d24, d25}, [%[ptr_out]]!\n\t"
- :[ptr_syn]"+r"(ptr_syn),
- [ptr_out]"+r"(ptr_out)
- :
- :"d22", "d23", "d24", "d25"
- );
- }
-
- // Update synthesis buffer.
- // C code:
- // WEBRTC_SPL_MEMCPY_W16(inst->synthesisBuffer,
- // inst->synthesisBuffer + inst->blockLen10ms,
- // inst->anaLen - inst->blockLen10ms);
- ptr_out = &inst->synthesisBuffer[0],
- ptr_syn = &inst->synthesisBuffer[inst->blockLen10ms];
- for (; ptr_syn < &inst->synthesisBuffer[inst->anaLen];) {
- // Loop unrolled once. Both pointers are incremented in the assembly code.
- __asm__ __volatile__(
- "vld1.16 {d22, d23}, [%[ptr_syn]]!\n\t"
- "vld1.16 {d24, d25}, [%[ptr_syn]]!\n\t"
- "vst1.16 {d22, d23}, [%[ptr_out]]!\n\t"
- "vst1.16 {d24, d25}, [%[ptr_out]]!\n\t"
- :[ptr_syn]"+r"(ptr_syn),
- [ptr_out]"+r"(ptr_out)
- :
- :"d22", "d23", "d24", "d25"
- );
- }
-
- // C code:
- // WebRtcSpl_ZerosArrayW16(inst->synthesisBuffer
- // + inst->anaLen - inst->blockLen10ms, inst->blockLen10ms);
- __asm__ __volatile__("vdup.16 q10, %0" : : "r"(0) : "q10");
- for (; ptr_out < &inst->synthesisBuffer[inst->anaLen];) {
- // Loop unrolled once. Pointer is incremented in the assembly code.
- __asm__ __volatile__(
- "vst1.16 {d20, d21}, [%[ptr_out]]!\n\t"
- "vst1.16 {d20, d21}, [%[ptr_out]]!\n\t"
- :[ptr_out]"+r"(ptr_out)
- :
- :"d20", "d21"
- );
- }
-}
-
-// Update analysis buffer for lower band, and window data before FFT.
-static void AnalysisUpdateNeon(NsxInst_t* inst,
- int16_t* out,
- int16_t* new_speech) {
-
- int16_t* ptr_ana = &inst->analysisBuffer[inst->blockLen10ms];
- int16_t* ptr_out = &inst->analysisBuffer[0];
-
- // For lower band update analysis buffer.
- // WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer,
- // inst->analysisBuffer + inst->blockLen10ms,
- // inst->anaLen - inst->blockLen10ms);
- for (; ptr_out < &inst->analysisBuffer[inst->anaLen - inst->blockLen10ms];) {
- // Loop unrolled once, so both pointers are incremented by 8 twice.
- __asm__ __volatile__(
- "vld1.16 {d20, d21}, [%[ptr_ana]]!\n\t"
- "vst1.16 {d20, d21}, [%[ptr_out]]!\n\t"
- "vld1.16 {d22, d23}, [%[ptr_ana]]!\n\t"
- "vst1.16 {d22, d23}, [%[ptr_out]]!\n\t"
- :[ptr_ana]"+r"(ptr_ana),
- [ptr_out]"+r"(ptr_out)
- :
- :"d20", "d21", "d22", "d23"
- );
- }
-
- // WEBRTC_SPL_MEMCPY_W16(inst->analysisBuffer
- // + inst->anaLen - inst->blockLen10ms, new_speech, inst->blockLen10ms);
- for (ptr_ana = new_speech; ptr_out < &inst->analysisBuffer[inst->anaLen];) {
- // Loop unrolled once, so both pointers are incremented by 8 twice.
- __asm__ __volatile__(
- "vld1.16 {d20, d21}, [%[ptr_ana]]!\n\t"
- "vst1.16 {d20, d21}, [%[ptr_out]]!\n\t"
- "vld1.16 {d22, d23}, [%[ptr_ana]]!\n\t"
- "vst1.16 {d22, d23}, [%[ptr_out]]!\n\t"
- :[ptr_ana]"+r"(ptr_ana),
- [ptr_out]"+r"(ptr_out)
- :
- :"d20", "d21", "d22", "d23"
- );
- }
-
- // Window data before FFT
- int16_t* ptr_window = &inst->window[0];
- ptr_out = &out[0];
- ptr_ana = &inst->analysisBuffer[0];
- for (; ptr_out < &out[inst->anaLen];) {
-
- // Loop unrolled once, so all pointers are incremented by 4 twice.
- __asm__ __volatile__(
- "vld1.16 d20, [%[ptr_ana]]!\n\t"
- "vld1.16 d21, [%[ptr_window]]!\n\t"
- // out[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
- // inst->window[i], inst->analysisBuffer[i], 14); // Q0
- "vmull.s16 q10, d20, d21\n\t"
- "vrshrn.i32 d20, q10, #14\n\t"
- "vst1.16 d20, [%[ptr_out]]!\n\t"
-
- "vld1.16 d22, [%[ptr_ana]]!\n\t"
- "vld1.16 d23, [%[ptr_window]]!\n\t"
- // out[i] = (int16_t)WEBRTC_SPL_MUL_16_16_RSFT_WITH_ROUND(
- // inst->window[i], inst->analysisBuffer[i], 14); // Q0
- "vmull.s16 q11, d22, d23\n\t"
- "vrshrn.i32 d22, q11, #14\n\t"
- "vst1.16 d22, [%[ptr_out]]!\n\t"
-
- // Specify constraints.
- :[ptr_ana]"+r"(ptr_ana),
- [ptr_window]"+r"(ptr_window),
- [ptr_out]"+r"(ptr_out)
- :
- :"d20", "d21", "d22", "d23", "q10", "q11"
- );
- }
-}
-
-// Create a complex number buffer (out[]) as the intput (in[]) interleaved with
-// zeros, and normalize it.
-static __inline void CreateComplexBufferNeon(NsxInst_t* inst,
- int16_t* in,
- int16_t* out) {
- int16_t* ptr_out = &out[0];
- int16_t* ptr_in = &in[0];
-
- __asm__ __volatile__("vdup.16 d25, %0" : : "r"(0) : "d25");
- __asm__ __volatile__("vdup.16 q10, %0" : : "r"(inst->normData) : "q10");
- for (; ptr_in < &in[inst->anaLen];) {
-
- // Loop unrolled once, so ptr_in is incremented by 8 twice,
- // and ptr_out is incremented by 8 four times.
- __asm__ __volatile__(
- // out[j] = WEBRTC_SPL_LSHIFT_W16(in[i], inst->normData); // Q(normData)
- "vld1.16 {d22, d23}, [%[ptr_in]]!\n\t"
- "vshl.s16 q11, q10\n\t"
- "vmov d24, d23\n\t"
-
- // out[j + 1] = 0; // Insert zeros in imaginary part
- "vmov d23, d25\n\t"
- "vst2.16 {d22, d23}, [%[ptr_out]]!\n\t"
- "vst2.16 {d24, d25}, [%[ptr_out]]!\n\t"
-
- // out[j] = WEBRTC_SPL_LSHIFT_W16(in[i], inst->normData); // Q(normData)
- "vld1.16 {d22, d23}, [%[ptr_in]]!\n\t"
- "vshl.s16 q11, q10\n\t"
- "vmov d24, d23\n\t"
-
- // out[j + 1] = 0; // Insert zeros in imaginary part
- "vmov d23, d25\n\t"
- "vst2.16 {d22, d23}, [%[ptr_out]]!\n\t"
- "vst2.16 {d24, d25}, [%[ptr_out]]!\n\t"
-
- // Specify constraints.
- :[ptr_in]"+r"(ptr_in),
- [ptr_out]"+r"(ptr_out)
- :
- :"d22", "d23", "d24", "d25", "q10", "q11"
- );
- }
-}
-
-void WebRtcNsx_InitNeon(void) {
- WebRtcNsx_NoiseEstimation = NoiseEstimationNeon;
- WebRtcNsx_PrepareSpectrum = PrepareSpectrumNeon;
- WebRtcNsx_SynthesisUpdate = SynthesisUpdateNeon;
- WebRtcNsx_AnalysisUpdate = AnalysisUpdateNeon;
- WebRtcNsx_Denormalize = DenormalizeNeon;
- WebRtcNsx_CreateComplexBuffer = CreateComplexBufferNeon;
-}
diff --git a/trunk/src/modules/audio_processing/ns/nsx_defines.h b/trunk/src/modules/audio_processing/ns/nsx_defines.h
deleted file mode 100644
index cd1e3bf..0000000
--- a/trunk/src/modules/audio_processing/ns/nsx_defines.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_DEFINES_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_DEFINES_H_
-
-#define ANAL_BLOCKL_MAX 256 // max analysis block length
-#define HALF_ANAL_BLOCKL 129 // half max analysis block length + 1
-#define SIMULT 3
-#define END_STARTUP_LONG 200
-#define END_STARTUP_SHORT 50
-#define FACTOR_Q16 (WebRtc_Word32)2621440 // 40 in Q16
-#define FACTOR_Q7 (WebRtc_Word16)5120 // 40 in Q7
-#define FACTOR_Q7_STARTUP (WebRtc_Word16)1024 // 8 in Q7
-#define WIDTH_Q8 3 // 0.01 in Q8 (or 25 )
-//PARAMETERS FOR NEW METHOD
-#define DD_PR_SNR_Q11 2007 // ~= Q11(0.98) DD update of prior SNR
-#define ONE_MINUS_DD_PR_SNR_Q11 41 // DD update of prior SNR
-#define SPECT_FLAT_TAVG_Q14 4915 // (0.30) tavg parameter for spectral flatness measure
-#define SPECT_DIFF_TAVG_Q8 77 // (0.30) tavg parameter for spectral flatness measure
-#define PRIOR_UPDATE_Q14 1638 // Q14(0.1) update parameter of prior model
-#define NOISE_UPDATE_Q8 26 // 26 ~= Q8(0.1) update parameter for noise
-// probability threshold for noise state in speech/noise likelihood
-#define ONE_MINUS_PROB_RANGE_Q8 205 // 205 ~= Q8(0.8)
-#define HIST_PAR_EST 1000 // histogram size for estimation of parameters
-//FEATURE EXTRACTION CONFIG
-//bin size of histogram
-#define BIN_SIZE_LRT 10
-//scale parameters: multiply dominant peaks of the histograms by scale factor to obtain
-// thresholds for prior model
-#define FACTOR_1_LRT_DIFF 6 //for LRT and spectral difference (5 times bigger)
-//for spectral_flatness: used when noise is flatter than speech (10 times bigger)
-#define FACTOR_2_FLAT_Q10 922
-//peak limit for spectral flatness (varies between 0 and 1)
-#define THRES_PEAK_FLAT 24 // * 2 * BIN_SIZE_FLAT_FX
-//limit on spacing of two highest peaks in histogram: spacing determined by bin size
-#define LIM_PEAK_SPACE_FLAT_DIFF 4 // * 2 * BIN_SIZE_DIFF_FX
-//limit on relevance of second peak:
-#define LIM_PEAK_WEIGHT_FLAT_DIFF 2
-#define THRES_FLUCT_LRT 10240 //=20 * inst->modelUpdate; fluctuation limit of LRT feat.
-//limit on the max and min values for the feature thresholds
-#define MAX_FLAT_Q10 38912 // * 2 * BIN_SIZE_FLAT_FX
-#define MIN_FLAT_Q10 4096 // * 2 * BIN_SIZE_FLAT_FX
-#define MAX_DIFF 100 // * 2 * BIN_SIZE_DIFF_FX
-#define MIN_DIFF 16 // * 2 * BIN_SIZE_DIFF_FX
-//criteria of weight of histogram peak to accept/reject feature
-#define THRES_WEIGHT_FLAT_DIFF 154//(int)(0.3*(inst->modelUpdate)) for flatness and difference
-//
-#define STAT_UPDATES 9 // Update every 512 = 1 << 9 block
-#define ONE_MINUS_GAMMA_PAUSE_Q8 13 // ~= Q8(0.05) update for conservative noise estimate
-#define GAMMA_NOISE_TRANS_AND_SPEECH_Q8 3 // ~= Q8(0.01) update for transition and noise region
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_NSX_DEFINES_H_
diff --git a/trunk/src/modules/audio_processing/ns/windows_private.h b/trunk/src/modules/audio_processing/ns/windows_private.h
deleted file mode 100644
index 44c2e84..0000000
--- a/trunk/src/modules/audio_processing/ns/windows_private.h
+++ /dev/null
@@ -1,574 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_WINDOWS_PRIVATE_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_WINDOWS_PRIVATE_H_
-
-// Hanning window for 4ms 16kHz
-static const float kHanning64w128[128] = {
- 0.00000000000000f, 0.02454122852291f, 0.04906767432742f,
- 0.07356456359967f, 0.09801714032956f, 0.12241067519922f,
- 0.14673047445536f, 0.17096188876030f, 0.19509032201613f,
- 0.21910124015687f, 0.24298017990326f, 0.26671275747490f,
- 0.29028467725446f, 0.31368174039889f, 0.33688985339222f,
- 0.35989503653499f, 0.38268343236509f, 0.40524131400499f,
- 0.42755509343028f, 0.44961132965461f, 0.47139673682600f,
- 0.49289819222978f, 0.51410274419322f, 0.53499761988710f,
- 0.55557023301960f, 0.57580819141785f, 0.59569930449243f,
- 0.61523159058063f, 0.63439328416365f, 0.65317284295378f,
- 0.67155895484702f, 0.68954054473707f, 0.70710678118655f,
- 0.72424708295147f, 0.74095112535496f, 0.75720884650648f,
- 0.77301045336274f, 0.78834642762661f, 0.80320753148064f,
- 0.81758481315158f, 0.83146961230255f, 0.84485356524971f,
- 0.85772861000027f, 0.87008699110871f, 0.88192126434835f,
- 0.89322430119552f, 0.90398929312344f, 0.91420975570353f,
- 0.92387953251129f, 0.93299279883474f, 0.94154406518302f,
- 0.94952818059304f, 0.95694033573221f, 0.96377606579544f,
- 0.97003125319454f, 0.97570213003853f, 0.98078528040323f,
- 0.98527764238894f, 0.98917650996478f, 0.99247953459871f,
- 0.99518472667220f, 0.99729045667869f, 0.99879545620517f,
- 0.99969881869620f, 1.00000000000000f,
- 0.99969881869620f, 0.99879545620517f, 0.99729045667869f,
- 0.99518472667220f, 0.99247953459871f, 0.98917650996478f,
- 0.98527764238894f, 0.98078528040323f, 0.97570213003853f,
- 0.97003125319454f, 0.96377606579544f, 0.95694033573221f,
- 0.94952818059304f, 0.94154406518302f, 0.93299279883474f,
- 0.92387953251129f, 0.91420975570353f, 0.90398929312344f,
- 0.89322430119552f, 0.88192126434835f, 0.87008699110871f,
- 0.85772861000027f, 0.84485356524971f, 0.83146961230255f,
- 0.81758481315158f, 0.80320753148064f, 0.78834642762661f,
- 0.77301045336274f, 0.75720884650648f, 0.74095112535496f,
- 0.72424708295147f, 0.70710678118655f, 0.68954054473707f,
- 0.67155895484702f, 0.65317284295378f, 0.63439328416365f,
- 0.61523159058063f, 0.59569930449243f, 0.57580819141785f,
- 0.55557023301960f, 0.53499761988710f, 0.51410274419322f,
- 0.49289819222978f, 0.47139673682600f, 0.44961132965461f,
- 0.42755509343028f, 0.40524131400499f, 0.38268343236509f,
- 0.35989503653499f, 0.33688985339222f, 0.31368174039889f,
- 0.29028467725446f, 0.26671275747490f, 0.24298017990326f,
- 0.21910124015687f, 0.19509032201613f, 0.17096188876030f,
- 0.14673047445536f, 0.12241067519922f, 0.09801714032956f,
- 0.07356456359967f, 0.04906767432742f, 0.02454122852291f
-};
-
-
-
-// hybrib Hanning & flat window
-static const float kBlocks80w128[128] = {
- (float)0.00000000, (float)0.03271908, (float)0.06540313, (float)0.09801714, (float)0.13052619,
- (float)0.16289547, (float)0.19509032, (float)0.22707626, (float)0.25881905, (float)0.29028468,
- (float)0.32143947, (float)0.35225005, (float)0.38268343, (float)0.41270703, (float)0.44228869,
- (float)0.47139674, (float)0.50000000, (float)0.52806785, (float)0.55557023, (float)0.58247770,
- (float)0.60876143, (float)0.63439328, (float)0.65934582, (float)0.68359230, (float)0.70710678,
- (float)0.72986407, (float)0.75183981, (float)0.77301045, (float)0.79335334, (float)0.81284668,
- (float)0.83146961, (float)0.84920218, (float)0.86602540, (float)0.88192126, (float)0.89687274,
- (float)0.91086382, (float)0.92387953, (float)0.93590593, (float)0.94693013, (float)0.95694034,
- (float)0.96592583, (float)0.97387698, (float)0.98078528, (float)0.98664333, (float)0.99144486,
- (float)0.99518473, (float)0.99785892, (float)0.99946459, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)0.99946459, (float)0.99785892, (float)0.99518473, (float)0.99144486,
- (float)0.98664333, (float)0.98078528, (float)0.97387698, (float)0.96592583, (float)0.95694034,
- (float)0.94693013, (float)0.93590593, (float)0.92387953, (float)0.91086382, (float)0.89687274,
- (float)0.88192126, (float)0.86602540, (float)0.84920218, (float)0.83146961, (float)0.81284668,
- (float)0.79335334, (float)0.77301045, (float)0.75183981, (float)0.72986407, (float)0.70710678,
- (float)0.68359230, (float)0.65934582, (float)0.63439328, (float)0.60876143, (float)0.58247770,
- (float)0.55557023, (float)0.52806785, (float)0.50000000, (float)0.47139674, (float)0.44228869,
- (float)0.41270703, (float)0.38268343, (float)0.35225005, (float)0.32143947, (float)0.29028468,
- (float)0.25881905, (float)0.22707626, (float)0.19509032, (float)0.16289547, (float)0.13052619,
- (float)0.09801714, (float)0.06540313, (float)0.03271908
-};
-
-// hybrib Hanning & flat window
-static const float kBlocks160w256[256] = {
- (float)0.00000000, (float)0.01636173, (float)0.03271908, (float)0.04906767, (float)0.06540313,
- (float)0.08172107, (float)0.09801714, (float)0.11428696, (float)0.13052619, (float)0.14673047,
- (float)0.16289547, (float)0.17901686, (float)0.19509032, (float)0.21111155, (float)0.22707626,
- (float)0.24298018, (float)0.25881905, (float)0.27458862, (float)0.29028468, (float)0.30590302,
- (float)0.32143947, (float)0.33688985, (float)0.35225005, (float)0.36751594, (float)0.38268343,
- (float)0.39774847, (float)0.41270703, (float)0.42755509, (float)0.44228869, (float)0.45690388,
- (float)0.47139674, (float)0.48576339, (float)0.50000000, (float)0.51410274, (float)0.52806785,
- (float)0.54189158, (float)0.55557023, (float)0.56910015, (float)0.58247770, (float)0.59569930,
- (float)0.60876143, (float)0.62166057, (float)0.63439328, (float)0.64695615, (float)0.65934582,
- (float)0.67155895, (float)0.68359230, (float)0.69544264, (float)0.70710678, (float)0.71858162,
- (float)0.72986407, (float)0.74095113, (float)0.75183981, (float)0.76252720, (float)0.77301045,
- (float)0.78328675, (float)0.79335334, (float)0.80320753, (float)0.81284668, (float)0.82226822,
- (float)0.83146961, (float)0.84044840, (float)0.84920218, (float)0.85772861, (float)0.86602540,
- (float)0.87409034, (float)0.88192126, (float)0.88951608, (float)0.89687274, (float)0.90398929,
- (float)0.91086382, (float)0.91749450, (float)0.92387953, (float)0.93001722, (float)0.93590593,
- (float)0.94154407, (float)0.94693013, (float)0.95206268, (float)0.95694034, (float)0.96156180,
- (float)0.96592583, (float)0.97003125, (float)0.97387698, (float)0.97746197, (float)0.98078528,
- (float)0.98384601, (float)0.98664333, (float)0.98917651, (float)0.99144486, (float)0.99344778,
- (float)0.99518473, (float)0.99665524, (float)0.99785892, (float)0.99879546, (float)0.99946459,
- (float)0.99986614, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)0.99986614, (float)0.99946459, (float)0.99879546, (float)0.99785892,
- (float)0.99665524, (float)0.99518473, (float)0.99344778, (float)0.99144486, (float)0.98917651,
- (float)0.98664333, (float)0.98384601, (float)0.98078528, (float)0.97746197, (float)0.97387698,
- (float)0.97003125, (float)0.96592583, (float)0.96156180, (float)0.95694034, (float)0.95206268,
- (float)0.94693013, (float)0.94154407, (float)0.93590593, (float)0.93001722, (float)0.92387953,
- (float)0.91749450, (float)0.91086382, (float)0.90398929, (float)0.89687274, (float)0.88951608,
- (float)0.88192126, (float)0.87409034, (float)0.86602540, (float)0.85772861, (float)0.84920218,
- (float)0.84044840, (float)0.83146961, (float)0.82226822, (float)0.81284668, (float)0.80320753,
- (float)0.79335334, (float)0.78328675, (float)0.77301045, (float)0.76252720, (float)0.75183981,
- (float)0.74095113, (float)0.72986407, (float)0.71858162, (float)0.70710678, (float)0.69544264,
- (float)0.68359230, (float)0.67155895, (float)0.65934582, (float)0.64695615, (float)0.63439328,
- (float)0.62166057, (float)0.60876143, (float)0.59569930, (float)0.58247770, (float)0.56910015,
- (float)0.55557023, (float)0.54189158, (float)0.52806785, (float)0.51410274, (float)0.50000000,
- (float)0.48576339, (float)0.47139674, (float)0.45690388, (float)0.44228869, (float)0.42755509,
- (float)0.41270703, (float)0.39774847, (float)0.38268343, (float)0.36751594, (float)0.35225005,
- (float)0.33688985, (float)0.32143947, (float)0.30590302, (float)0.29028468, (float)0.27458862,
- (float)0.25881905, (float)0.24298018, (float)0.22707626, (float)0.21111155, (float)0.19509032,
- (float)0.17901686, (float)0.16289547, (float)0.14673047, (float)0.13052619, (float)0.11428696,
- (float)0.09801714, (float)0.08172107, (float)0.06540313, (float)0.04906767, (float)0.03271908,
- (float)0.01636173
-};
-
-// hybrib Hanning & flat window: for 20ms
-static const float kBlocks320w512[512] = {
- (float)0.00000000, (float)0.00818114, (float)0.01636173, (float)0.02454123, (float)0.03271908,
- (float)0.04089475, (float)0.04906767, (float)0.05723732, (float)0.06540313, (float)0.07356456,
- (float)0.08172107, (float)0.08987211, (float)0.09801714, (float)0.10615561, (float)0.11428696,
- (float)0.12241068, (float)0.13052619, (float)0.13863297, (float)0.14673047, (float)0.15481816,
- (float)0.16289547, (float)0.17096189, (float)0.17901686, (float)0.18705985, (float)0.19509032,
- (float)0.20310773, (float)0.21111155, (float)0.21910124, (float)0.22707626, (float)0.23503609,
- (float)0.24298018, (float)0.25090801, (float)0.25881905, (float)0.26671276, (float)0.27458862,
- (float)0.28244610, (float)0.29028468, (float)0.29810383, (float)0.30590302, (float)0.31368174,
- (float)0.32143947, (float)0.32917568, (float)0.33688985, (float)0.34458148, (float)0.35225005,
- (float)0.35989504, (float)0.36751594, (float)0.37511224, (float)0.38268343, (float)0.39022901,
- (float)0.39774847, (float)0.40524131, (float)0.41270703, (float)0.42014512, (float)0.42755509,
- (float)0.43493645, (float)0.44228869, (float)0.44961133, (float)0.45690388, (float)0.46416584,
- (float)0.47139674, (float)0.47859608, (float)0.48576339, (float)0.49289819, (float)0.50000000,
- (float)0.50706834, (float)0.51410274, (float)0.52110274, (float)0.52806785, (float)0.53499762,
- (float)0.54189158, (float)0.54874927, (float)0.55557023, (float)0.56235401, (float)0.56910015,
- (float)0.57580819, (float)0.58247770, (float)0.58910822, (float)0.59569930, (float)0.60225052,
- (float)0.60876143, (float)0.61523159, (float)0.62166057, (float)0.62804795, (float)0.63439328,
- (float)0.64069616, (float)0.64695615, (float)0.65317284, (float)0.65934582, (float)0.66547466,
- (float)0.67155895, (float)0.67759830, (float)0.68359230, (float)0.68954054, (float)0.69544264,
- (float)0.70129818, (float)0.70710678, (float)0.71286806, (float)0.71858162, (float)0.72424708,
- (float)0.72986407, (float)0.73543221, (float)0.74095113, (float)0.74642045, (float)0.75183981,
- (float)0.75720885, (float)0.76252720, (float)0.76779452, (float)0.77301045, (float)0.77817464,
- (float)0.78328675, (float)0.78834643, (float)0.79335334, (float)0.79830715, (float)0.80320753,
- (float)0.80805415, (float)0.81284668, (float)0.81758481, (float)0.82226822, (float)0.82689659,
- (float)0.83146961, (float)0.83598698, (float)0.84044840, (float)0.84485357, (float)0.84920218,
- (float)0.85349396, (float)0.85772861, (float)0.86190585, (float)0.86602540, (float)0.87008699,
- (float)0.87409034, (float)0.87803519, (float)0.88192126, (float)0.88574831, (float)0.88951608,
- (float)0.89322430, (float)0.89687274, (float)0.90046115, (float)0.90398929, (float)0.90745693,
- (float)0.91086382, (float)0.91420976, (float)0.91749450, (float)0.92071783, (float)0.92387953,
- (float)0.92697940, (float)0.93001722, (float)0.93299280, (float)0.93590593, (float)0.93875641,
- (float)0.94154407, (float)0.94426870, (float)0.94693013, (float)0.94952818, (float)0.95206268,
- (float)0.95453345, (float)0.95694034, (float)0.95928317, (float)0.96156180, (float)0.96377607,
- (float)0.96592583, (float)0.96801094, (float)0.97003125, (float)0.97198664, (float)0.97387698,
- (float)0.97570213, (float)0.97746197, (float)0.97915640, (float)0.98078528, (float)0.98234852,
- (float)0.98384601, (float)0.98527764, (float)0.98664333, (float)0.98794298, (float)0.98917651,
- (float)0.99034383, (float)0.99144486, (float)0.99247953, (float)0.99344778, (float)0.99434953,
- (float)0.99518473, (float)0.99595331, (float)0.99665524, (float)0.99729046, (float)0.99785892,
- (float)0.99836060, (float)0.99879546, (float)0.99916346, (float)0.99946459, (float)0.99969882,
- (float)0.99986614, (float)0.99996653, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000, (float)1.00000000,
- (float)1.00000000, (float)0.99996653, (float)0.99986614, (float)0.99969882, (float)0.99946459,
- (float)0.99916346, (float)0.99879546, (float)0.99836060, (float)0.99785892, (float)0.99729046,
- (float)0.99665524, (float)0.99595331, (float)0.99518473, (float)0.99434953, (float)0.99344778,
- (float)0.99247953, (float)0.99144486, (float)0.99034383, (float)0.98917651, (float)0.98794298,
- (float)0.98664333, (float)0.98527764, (float)0.98384601, (float)0.98234852, (float)0.98078528,
- (float)0.97915640, (float)0.97746197, (float)0.97570213, (float)0.97387698, (float)0.97198664,
- (float)0.97003125, (float)0.96801094, (float)0.96592583, (float)0.96377607, (float)0.96156180,
- (float)0.95928317, (float)0.95694034, (float)0.95453345, (float)0.95206268, (float)0.94952818,
- (float)0.94693013, (float)0.94426870, (float)0.94154407, (float)0.93875641, (float)0.93590593,
- (float)0.93299280, (float)0.93001722, (float)0.92697940, (float)0.92387953, (float)0.92071783,
- (float)0.91749450, (float)0.91420976, (float)0.91086382, (float)0.90745693, (float)0.90398929,
- (float)0.90046115, (float)0.89687274, (float)0.89322430, (float)0.88951608, (float)0.88574831,
- (float)0.88192126, (float)0.87803519, (float)0.87409034, (float)0.87008699, (float)0.86602540,
- (float)0.86190585, (float)0.85772861, (float)0.85349396, (float)0.84920218, (float)0.84485357,
- (float)0.84044840, (float)0.83598698, (float)0.83146961, (float)0.82689659, (float)0.82226822,
- (float)0.81758481, (float)0.81284668, (float)0.80805415, (float)0.80320753, (float)0.79830715,
- (float)0.79335334, (float)0.78834643, (float)0.78328675, (float)0.77817464, (float)0.77301045,
- (float)0.76779452, (float)0.76252720, (float)0.75720885, (float)0.75183981, (float)0.74642045,
- (float)0.74095113, (float)0.73543221, (float)0.72986407, (float)0.72424708, (float)0.71858162,
- (float)0.71286806, (float)0.70710678, (float)0.70129818, (float)0.69544264, (float)0.68954054,
- (float)0.68359230, (float)0.67759830, (float)0.67155895, (float)0.66547466, (float)0.65934582,
- (float)0.65317284, (float)0.64695615, (float)0.64069616, (float)0.63439328, (float)0.62804795,
- (float)0.62166057, (float)0.61523159, (float)0.60876143, (float)0.60225052, (float)0.59569930,
- (float)0.58910822, (float)0.58247770, (float)0.57580819, (float)0.56910015, (float)0.56235401,
- (float)0.55557023, (float)0.54874927, (float)0.54189158, (float)0.53499762, (float)0.52806785,
- (float)0.52110274, (float)0.51410274, (float)0.50706834, (float)0.50000000, (float)0.49289819,
- (float)0.48576339, (float)0.47859608, (float)0.47139674, (float)0.46416584, (float)0.45690388,
- (float)0.44961133, (float)0.44228869, (float)0.43493645, (float)0.42755509, (float)0.42014512,
- (float)0.41270703, (float)0.40524131, (float)0.39774847, (float)0.39022901, (float)0.38268343,
- (float)0.37511224, (float)0.36751594, (float)0.35989504, (float)0.35225005, (float)0.34458148,
- (float)0.33688985, (float)0.32917568, (float)0.32143947, (float)0.31368174, (float)0.30590302,
- (float)0.29810383, (float)0.29028468, (float)0.28244610, (float)0.27458862, (float)0.26671276,
- (float)0.25881905, (float)0.25090801, (float)0.24298018, (float)0.23503609, (float)0.22707626,
- (float)0.21910124, (float)0.21111155, (float)0.20310773, (float)0.19509032, (float)0.18705985,
- (float)0.17901686, (float)0.17096189, (float)0.16289547, (float)0.15481816, (float)0.14673047,
- (float)0.13863297, (float)0.13052619, (float)0.12241068, (float)0.11428696, (float)0.10615561,
- (float)0.09801714, (float)0.08987211, (float)0.08172107, (float)0.07356456, (float)0.06540313,
- (float)0.05723732, (float)0.04906767, (float)0.04089475, (float)0.03271908, (float)0.02454123,
- (float)0.01636173, (float)0.00818114
-};
-
-
-// Hanning window: for 15ms at 16kHz with symmetric zeros
-static const float kBlocks240w512[512] = {
- (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
- (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
- (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
- (float)0.00000000, (float)0.00000000, (float)0.00654494, (float)0.01308960, (float)0.01963369,
- (float)0.02617695, (float)0.03271908, (float)0.03925982, (float)0.04579887, (float)0.05233596,
- (float)0.05887080, (float)0.06540313, (float)0.07193266, (float)0.07845910, (float)0.08498218,
- (float)0.09150162, (float)0.09801714, (float)0.10452846, (float)0.11103531, (float)0.11753740,
- (float)0.12403446, (float)0.13052620, (float)0.13701233, (float)0.14349262, (float)0.14996676,
- (float)0.15643448, (float)0.16289547, (float)0.16934951, (float)0.17579629, (float)0.18223552,
- (float)0.18866697, (float)0.19509032, (float)0.20150533, (float)0.20791170, (float)0.21430916,
- (float)0.22069745, (float)0.22707628, (float)0.23344538, (float)0.23980446, (float)0.24615330,
- (float)0.25249159, (float)0.25881904, (float)0.26513544, (float)0.27144045, (float)0.27773386,
- (float)0.28401536, (float)0.29028466, (float)0.29654160, (float)0.30278578, (float)0.30901700,
- (float)0.31523499, (float)0.32143945, (float)0.32763019, (float)0.33380687, (float)0.33996925,
- (float)0.34611708, (float)0.35225007, (float)0.35836795, (float)0.36447051, (float)0.37055743,
- (float)0.37662852, (float)0.38268346, (float)0.38872197, (float)0.39474389, (float)0.40074885,
- (float)0.40673664, (float)0.41270703, (float)0.41865975, (float)0.42459452, (float)0.43051112,
- (float)0.43640924, (float)0.44228873, (float)0.44814920, (float)0.45399052, (float)0.45981237,
- (float)0.46561453, (float)0.47139674, (float)0.47715878, (float)0.48290035, (float)0.48862126,
- (float)0.49432120, (float)0.50000000, (float)0.50565743, (float)0.51129311, (float)0.51690692,
- (float)0.52249855, (float)0.52806789, (float)0.53361452, (float)0.53913832, (float)0.54463905,
- (float)0.55011642, (float)0.55557024, (float)0.56100029, (float)0.56640625, (float)0.57178795,
- (float)0.57714522, (float)0.58247769, (float)0.58778524, (float)0.59306765, (float)0.59832460,
- (float)0.60355598, (float)0.60876143, (float)0.61394083, (float)0.61909395, (float)0.62422055,
- (float)0.62932038, (float)0.63439333, (float)0.63943899, (float)0.64445734, (float)0.64944810,
- (float)0.65441096, (float)0.65934587, (float)0.66425246, (float)0.66913062, (float)0.67398012,
- (float)0.67880076, (float)0.68359232, (float)0.68835455, (float)0.69308740, (float)0.69779050,
- (float)0.70246369, (float)0.70710677, (float)0.71171963, (float)0.71630198, (float)0.72085363,
- (float)0.72537440, (float)0.72986406, (float)0.73432255, (float)0.73874950, (float)0.74314487,
- (float)0.74750835, (float)0.75183982, (float)0.75613910, (float)0.76040596, (float)0.76464027,
- (float)0.76884186, (float)0.77301043, (float)0.77714598, (float)0.78124821, (float)0.78531694,
- (float)0.78935206, (float)0.79335338, (float)0.79732066, (float)0.80125386, (float)0.80515265,
- (float)0.80901700, (float)0.81284672, (float)0.81664157, (float)0.82040149, (float)0.82412618,
- (float)0.82781565, (float)0.83146966, (float)0.83508795, (float)0.83867061, (float)0.84221727,
- (float)0.84572780, (float)0.84920216, (float)0.85264021, (float)0.85604161, (float)0.85940641,
- (float)0.86273444, (float)0.86602545, (float)0.86927933, (float)0.87249607, (float)0.87567532,
- (float)0.87881714, (float)0.88192129, (float)0.88498765, (float)0.88801610, (float)0.89100653,
- (float)0.89395881, (float)0.89687276, (float)0.89974827, (float)0.90258533, (float)0.90538365,
- (float)0.90814316, (float)0.91086388, (float)0.91354549, (float)0.91618794, (float)0.91879123,
- (float)0.92135513, (float)0.92387950, (float)0.92636442, (float)0.92880958, (float)0.93121493,
- (float)0.93358046, (float)0.93590593, (float)0.93819135, (float)0.94043654, (float)0.94264150,
- (float)0.94480604, (float)0.94693011, (float)0.94901365, (float)0.95105654, (float)0.95305866,
- (float)0.95501995, (float)0.95694035, (float)0.95881975, (float)0.96065807, (float)0.96245527,
- (float)0.96421117, (float)0.96592581, (float)0.96759909, (float)0.96923089, (float)0.97082120,
- (float)0.97236991, (float)0.97387701, (float)0.97534233, (float)0.97676587, (float)0.97814763,
- (float)0.97948742, (float)0.98078531, (float)0.98204112, (float)0.98325491, (float)0.98442656,
- (float)0.98555607, (float)0.98664331, (float)0.98768836, (float)0.98869103, (float)0.98965138,
- (float)0.99056935, (float)0.99144489, (float)0.99227792, (float)0.99306846, (float)0.99381649,
- (float)0.99452192, (float)0.99518472, (float)0.99580491, (float)0.99638247, (float)0.99691731,
- (float)0.99740952, (float)0.99785894, (float)0.99826562, (float)0.99862951, (float)0.99895066,
- (float)0.99922901, (float)0.99946457, (float)0.99965733, (float)0.99980724, (float)0.99991435,
- (float)0.99997860, (float)1.00000000, (float)0.99997860, (float)0.99991435, (float)0.99980724,
- (float)0.99965733, (float)0.99946457, (float)0.99922901, (float)0.99895066, (float)0.99862951,
- (float)0.99826562, (float)0.99785894, (float)0.99740946, (float)0.99691731, (float)0.99638247,
- (float)0.99580491, (float)0.99518472, (float)0.99452192, (float)0.99381644, (float)0.99306846,
- (float)0.99227792, (float)0.99144489, (float)0.99056935, (float)0.98965138, (float)0.98869103,
- (float)0.98768836, (float)0.98664331, (float)0.98555607, (float)0.98442656, (float)0.98325491,
- (float)0.98204112, (float)0.98078525, (float)0.97948742, (float)0.97814757, (float)0.97676587,
- (float)0.97534227, (float)0.97387695, (float)0.97236991, (float)0.97082120, (float)0.96923089,
- (float)0.96759909, (float)0.96592581, (float)0.96421117, (float)0.96245521, (float)0.96065807,
- (float)0.95881969, (float)0.95694029, (float)0.95501995, (float)0.95305860, (float)0.95105648,
- (float)0.94901365, (float)0.94693011, (float)0.94480604, (float)0.94264150, (float)0.94043654,
- (float)0.93819129, (float)0.93590593, (float)0.93358046, (float)0.93121493, (float)0.92880952,
- (float)0.92636436, (float)0.92387950, (float)0.92135507, (float)0.91879123, (float)0.91618794,
- (float)0.91354543, (float)0.91086382, (float)0.90814310, (float)0.90538365, (float)0.90258527,
- (float)0.89974827, (float)0.89687276, (float)0.89395875, (float)0.89100647, (float)0.88801610,
- (float)0.88498759, (float)0.88192123, (float)0.87881714, (float)0.87567532, (float)0.87249595,
- (float)0.86927933, (float)0.86602539, (float)0.86273432, (float)0.85940641, (float)0.85604161,
- (float)0.85264009, (float)0.84920216, (float)0.84572780, (float)0.84221715, (float)0.83867055,
- (float)0.83508795, (float)0.83146954, (float)0.82781565, (float)0.82412612, (float)0.82040137,
- (float)0.81664157, (float)0.81284660, (float)0.80901700, (float)0.80515265, (float)0.80125374,
- (float)0.79732066, (float)0.79335332, (float)0.78935200, (float)0.78531694, (float)0.78124815,
- (float)0.77714586, (float)0.77301049, (float)0.76884180, (float)0.76464021, (float)0.76040596,
- (float)0.75613904, (float)0.75183970, (float)0.74750835, (float)0.74314481, (float)0.73874938,
- (float)0.73432249, (float)0.72986400, (float)0.72537428, (float)0.72085363, (float)0.71630186,
- (float)0.71171951, (float)0.70710677, (float)0.70246363, (float)0.69779032, (float)0.69308734,
- (float)0.68835449, (float)0.68359220, (float)0.67880070, (float)0.67398006, (float)0.66913044,
- (float)0.66425240, (float)0.65934575, (float)0.65441096, (float)0.64944804, (float)0.64445722,
- (float)0.63943905, (float)0.63439327, (float)0.62932026, (float)0.62422055, (float)0.61909389,
- (float)0.61394072, (float)0.60876143, (float)0.60355592, (float)0.59832448, (float)0.59306765,
- (float)0.58778518, (float)0.58247757, (float)0.57714522, (float)0.57178789, (float)0.56640613,
- (float)0.56100023, (float)0.55557019, (float)0.55011630, (float)0.54463905, (float)0.53913826,
- (float)0.53361434, (float)0.52806783, (float)0.52249849, (float)0.51690674, (float)0.51129305,
- (float)0.50565726, (float)0.50000006, (float)0.49432117, (float)0.48862115, (float)0.48290038,
- (float)0.47715873, (float)0.47139663, (float)0.46561456, (float)0.45981231, (float)0.45399037,
- (float)0.44814920, (float)0.44228864, (float)0.43640912, (float)0.43051112, (float)0.42459446,
- (float)0.41865960, (float)0.41270703, (float)0.40673658, (float)0.40074870, (float)0.39474386,
- (float)0.38872188, (float)0.38268328, (float)0.37662849, (float)0.37055734, (float)0.36447033,
- (float)0.35836792, (float)0.35224995, (float)0.34611690, (float)0.33996922, (float)0.33380675,
- (float)0.32763001, (float)0.32143945, (float)0.31523487, (float)0.30901679, (float)0.30278572,
- (float)0.29654145, (float)0.29028472, (float)0.28401530, (float)0.27773371, (float)0.27144048,
- (float)0.26513538, (float)0.25881892, (float)0.25249159, (float)0.24615324, (float)0.23980433,
- (float)0.23344538, (float)0.22707619, (float)0.22069728, (float)0.21430916, (float)0.20791161,
- (float)0.20150517, (float)0.19509031, (float)0.18866688, (float)0.18223536, (float)0.17579627,
- (float)0.16934940, (float)0.16289529, (float)0.15643445, (float)0.14996666, (float)0.14349243,
- (float)0.13701232, (float)0.13052608, (float)0.12403426, (float)0.11753736, (float)0.11103519,
- (float)0.10452849, (float)0.09801710, (float)0.09150149, (float)0.08498220, (float)0.07845904,
- (float)0.07193252, (float)0.06540315, (float)0.05887074, (float)0.05233581, (float)0.04579888,
- (float)0.03925974, (float)0.03271893, (float)0.02617695, (float)0.01963361, (float)0.01308943,
- (float)0.00654493, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
- (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
- (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
- (float)0.00000000, (float)0.00000000
-};
-
-
-// Hanning window: for 30ms with 1024 fft with symmetric zeros at 16kHz
-static const float kBlocks480w1024[1024] = {
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- (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
- (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
- (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
- (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
- (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
- (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00327249, (float)0.00654494,
- (float)0.00981732, (float)0.01308960, (float)0.01636173, (float)0.01963369, (float)0.02290544,
- (float)0.02617695, (float)0.02944817, (float)0.03271908, (float)0.03598964, (float)0.03925982,
- (float)0.04252957, (float)0.04579887, (float)0.04906768, (float)0.05233596, (float)0.05560368,
- (float)0.05887080, (float)0.06213730, (float)0.06540313, (float)0.06866825, (float)0.07193266,
- (float)0.07519628, (float)0.07845910, (float)0.08172107, (float)0.08498218, (float)0.08824237,
- (float)0.09150162, (float)0.09475989, (float)0.09801714, (float)0.10127335, (float)0.10452846,
- (float)0.10778246, (float)0.11103531, (float)0.11428697, (float)0.11753740, (float)0.12078657,
- (float)0.12403446, (float)0.12728101, (float)0.13052620, (float)0.13376999, (float)0.13701233,
- (float)0.14025325, (float)0.14349262, (float)0.14673047, (float)0.14996676, (float)0.15320145,
- (float)0.15643448, (float)0.15966582, (float)0.16289547, (float)0.16612339, (float)0.16934951,
- (float)0.17257382, (float)0.17579629, (float)0.17901687, (float)0.18223552, (float)0.18545224,
- (float)0.18866697, (float)0.19187967, (float)0.19509032, (float)0.19829889, (float)0.20150533,
- (float)0.20470962, (float)0.20791170, (float)0.21111156, (float)0.21430916, (float)0.21750447,
- (float)0.22069745, (float)0.22388805, (float)0.22707628, (float)0.23026206, (float)0.23344538,
- (float)0.23662618, (float)0.23980446, (float)0.24298020, (float)0.24615330, (float)0.24932377,
- (float)0.25249159, (float)0.25565669, (float)0.25881904, (float)0.26197866, (float)0.26513544,
- (float)0.26828939, (float)0.27144045, (float)0.27458861, (float)0.27773386, (float)0.28087610,
- (float)0.28401536, (float)0.28715158, (float)0.29028466, (float)0.29341471, (float)0.29654160,
- (float)0.29966527, (float)0.30278578, (float)0.30590302, (float)0.30901700, (float)0.31212768,
- (float)0.31523499, (float)0.31833893, (float)0.32143945, (float)0.32453656, (float)0.32763019,
- (float)0.33072028, (float)0.33380687, (float)0.33688986, (float)0.33996925, (float)0.34304500,
- (float)0.34611708, (float)0.34918544, (float)0.35225007, (float)0.35531089, (float)0.35836795,
- (float)0.36142117, (float)0.36447051, (float)0.36751595, (float)0.37055743, (float)0.37359497,
- (float)0.37662852, (float)0.37965801, (float)0.38268346, (float)0.38570479, (float)0.38872197,
- (float)0.39173502, (float)0.39474389, (float)0.39774847, (float)0.40074885, (float)0.40374491,
- (float)0.40673664, (float)0.40972406, (float)0.41270703, (float)0.41568562, (float)0.41865975,
- (float)0.42162940, (float)0.42459452, (float)0.42755508, (float)0.43051112, (float)0.43346250,
- (float)0.43640924, (float)0.43935132, (float)0.44228873, (float)0.44522133, (float)0.44814920,
- (float)0.45107228, (float)0.45399052, (float)0.45690390, (float)0.45981237, (float)0.46271592,
- (float)0.46561453, (float)0.46850815, (float)0.47139674, (float)0.47428030, (float)0.47715878,
- (float)0.48003215, (float)0.48290035, (float)0.48576337, (float)0.48862126, (float)0.49147385,
- (float)0.49432120, (float)0.49716330, (float)0.50000000, (float)0.50283140, (float)0.50565743,
- (float)0.50847799, (float)0.51129311, (float)0.51410276, (float)0.51690692, (float)0.51970553,
- (float)0.52249855, (float)0.52528602, (float)0.52806789, (float)0.53084403, (float)0.53361452,
- (float)0.53637928, (float)0.53913832, (float)0.54189163, (float)0.54463905, (float)0.54738063,
- (float)0.55011642, (float)0.55284631, (float)0.55557024, (float)0.55828828, (float)0.56100029,
- (float)0.56370628, (float)0.56640625, (float)0.56910014, (float)0.57178795, (float)0.57446963,
- (float)0.57714522, (float)0.57981455, (float)0.58247769, (float)0.58513463, (float)0.58778524,
- (float)0.59042960, (float)0.59306765, (float)0.59569931, (float)0.59832460, (float)0.60094351,
- (float)0.60355598, (float)0.60616195, (float)0.60876143, (float)0.61135441, (float)0.61394083,
- (float)0.61652070, (float)0.61909395, (float)0.62166059, (float)0.62422055, (float)0.62677383,
- (float)0.62932038, (float)0.63186020, (float)0.63439333, (float)0.63691956, (float)0.63943899,
- (float)0.64195162, (float)0.64445734, (float)0.64695615, (float)0.64944810, (float)0.65193301,
- (float)0.65441096, (float)0.65688187, (float)0.65934587, (float)0.66180271, (float)0.66425246,
- (float)0.66669512, (float)0.66913062, (float)0.67155898, (float)0.67398012, (float)0.67639405,
- (float)0.67880076, (float)0.68120021, (float)0.68359232, (float)0.68597710, (float)0.68835455,
- (float)0.69072467, (float)0.69308740, (float)0.69544262, (float)0.69779050, (float)0.70013082,
- (float)0.70246369, (float)0.70478904, (float)0.70710677, (float)0.70941699, (float)0.71171963,
- (float)0.71401459, (float)0.71630198, (float)0.71858168, (float)0.72085363, (float)0.72311789,
- (float)0.72537440, (float)0.72762316, (float)0.72986406, (float)0.73209721, (float)0.73432255,
- (float)0.73653996, (float)0.73874950, (float)0.74095118, (float)0.74314487, (float)0.74533057,
- (float)0.74750835, (float)0.74967808, (float)0.75183982, (float)0.75399351, (float)0.75613910,
- (float)0.75827658, (float)0.76040596, (float)0.76252723, (float)0.76464027, (float)0.76674515,
- (float)0.76884186, (float)0.77093029, (float)0.77301043, (float)0.77508241, (float)0.77714598,
- (float)0.77920127, (float)0.78124821, (float)0.78328675, (float)0.78531694, (float)0.78733873,
- (float)0.78935206, (float)0.79135692, (float)0.79335338, (float)0.79534125, (float)0.79732066,
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- (float)0.03925974, (float)0.03598953, (float)0.03271893, (float)0.02944798, (float)0.02617695,
- (float)0.02290541, (float)0.01963361, (float)0.01636161, (float)0.01308943, (float)0.00981712,
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- (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000,
- (float)0.00000000, (float)0.00000000, (float)0.00000000, (float)0.00000000
-};
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_NS_MAIN_SOURCE_WINDOWS_PRIVATE_H_
diff --git a/trunk/src/modules/audio_processing/processing_component.cc b/trunk/src/modules/audio_processing/processing_component.cc
deleted file mode 100644
index 9ac1257..0000000
--- a/trunk/src/modules/audio_processing/processing_component.cc
+++ /dev/null
@@ -1,112 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "processing_component.h"
-
-#include <cassert>
-
-#include "audio_processing_impl.h"
-
-namespace webrtc {
-
-ProcessingComponent::ProcessingComponent(const AudioProcessingImpl* apm)
- : apm_(apm),
- initialized_(false),
- enabled_(false),
- num_handles_(0) {}
-
-ProcessingComponent::~ProcessingComponent() {
- assert(initialized_ == false);
-}
-
-int ProcessingComponent::Destroy() {
- while (!handles_.empty()) {
- DestroyHandle(handles_.back());
- handles_.pop_back();
- }
- initialized_ = false;
-
- return apm_->kNoError;
-}
-
-int ProcessingComponent::EnableComponent(bool enable) {
- if (enable && !enabled_) {
- enabled_ = enable; // Must be set before Initialize() is called.
-
- int err = Initialize();
- if (err != apm_->kNoError) {
- enabled_ = false;
- return err;
- }
- } else {
- enabled_ = enable;
- }
-
- return apm_->kNoError;
-}
-
-bool ProcessingComponent::is_component_enabled() const {
- return enabled_;
-}
-
-void* ProcessingComponent::handle(int index) const {
- assert(index < num_handles_);
- return handles_[index];
-}
-
-int ProcessingComponent::num_handles() const {
- return num_handles_;
-}
-
-int ProcessingComponent::Initialize() {
- if (!enabled_) {
- return apm_->kNoError;
- }
-
- num_handles_ = num_handles_required();
- if (num_handles_ > static_cast<int>(handles_.size())) {
- handles_.resize(num_handles_, NULL);
- }
-
- assert(static_cast<int>(handles_.size()) >= num_handles_);
- for (int i = 0; i < num_handles_; i++) {
- if (handles_[i] == NULL) {
- handles_[i] = CreateHandle();
- if (handles_[i] == NULL) {
- return apm_->kCreationFailedError;
- }
- }
-
- int err = InitializeHandle(handles_[i]);
- if (err != apm_->kNoError) {
- return GetHandleError(handles_[i]);
- }
- }
-
- initialized_ = true;
- return Configure();
-}
-
-int ProcessingComponent::Configure() {
- if (!initialized_) {
- return apm_->kNoError;
- }
-
- assert(static_cast<int>(handles_.size()) >= num_handles_);
- for (int i = 0; i < num_handles_; i++) {
- int err = ConfigureHandle(handles_[i]);
- if (err != apm_->kNoError) {
- return GetHandleError(handles_[i]);
- }
- }
-
- return apm_->kNoError;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/processing_component.h b/trunk/src/modules/audio_processing/processing_component.h
deleted file mode 100644
index b3457b5..0000000
--- a/trunk/src/modules/audio_processing/processing_component.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_PROCESSING_COMPONENT_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_PROCESSING_COMPONENT_H_
-
-#include <vector>
-
-#include "audio_processing.h"
-
-namespace webrtc {
-class AudioProcessingImpl;
-
-class ProcessingComponent {
- public:
- explicit ProcessingComponent(const AudioProcessingImpl* apm);
- virtual ~ProcessingComponent();
-
- virtual int Initialize();
- virtual int Destroy();
-
- bool is_component_enabled() const;
-
- protected:
- virtual int Configure();
- int EnableComponent(bool enable);
- void* handle(int index) const;
- int num_handles() const;
-
- private:
- virtual void* CreateHandle() const = 0;
- virtual int InitializeHandle(void* handle) const = 0;
- virtual int ConfigureHandle(void* handle) const = 0;
- virtual int DestroyHandle(void* handle) const = 0;
- virtual int num_handles_required() const = 0;
- virtual int GetHandleError(void* handle) const = 0;
-
- const AudioProcessingImpl* apm_;
- std::vector<void*> handles_;
- bool initialized_;
- bool enabled_;
- int num_handles_;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_PROCESSING_COMPONENT_H__
diff --git a/trunk/src/modules/audio_processing/splitting_filter.cc b/trunk/src/modules/audio_processing/splitting_filter.cc
deleted file mode 100644
index 1526141..0000000
--- a/trunk/src/modules/audio_processing/splitting_filter.cc
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "splitting_filter.h"
-#include "signal_processing_library.h"
-
-namespace webrtc {
-
-void SplittingFilterAnalysis(const WebRtc_Word16* in_data,
- WebRtc_Word16* low_band,
- WebRtc_Word16* high_band,
- WebRtc_Word32* filter_state1,
- WebRtc_Word32* filter_state2)
-{
- WebRtcSpl_AnalysisQMF(in_data, low_band, high_band, filter_state1, filter_state2);
-}
-
-void SplittingFilterSynthesis(const WebRtc_Word16* low_band,
- const WebRtc_Word16* high_band,
- WebRtc_Word16* out_data,
- WebRtc_Word32* filt_state1,
- WebRtc_Word32* filt_state2)
-{
- WebRtcSpl_SynthesisQMF(low_band, high_band, out_data, filt_state1, filt_state2);
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/splitting_filter.h b/trunk/src/modules/audio_processing/splitting_filter.h
deleted file mode 100644
index 661bfb2..0000000
--- a/trunk/src/modules/audio_processing/splitting_filter.h
+++ /dev/null
@@ -1,63 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_SPLITTING_FILTER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_SPLITTING_FILTER_H_
-
-#include "typedefs.h"
-#include "signal_processing_library.h"
-
-namespace webrtc {
-/*
- * SplittingFilterbank_analysisQMF(...)
- *
- * Splits a super-wb signal into two subbands: 0-8 kHz and 8-16 kHz.
- *
- * Input:
- * - in_data : super-wb audio signal
- *
- * Input & Output:
- * - filt_state1: Filter state for first all-pass filter
- * - filt_state2: Filter state for second all-pass filter
- *
- * Output:
- * - low_band : The signal from the 0-4 kHz band
- * - high_band : The signal from the 4-8 kHz band
- */
-void SplittingFilterAnalysis(const WebRtc_Word16* in_data,
- WebRtc_Word16* low_band,
- WebRtc_Word16* high_band,
- WebRtc_Word32* filt_state1,
- WebRtc_Word32* filt_state2);
-
-/*
- * SplittingFilterbank_synthesisQMF(...)
- *
- * Combines the two subbands (0-8 and 8-16 kHz) into a super-wb signal.
- *
- * Input:
- * - low_band : The signal with the 0-8 kHz band
- * - high_band : The signal with the 8-16 kHz band
- *
- * Input & Output:
- * - filt_state1: Filter state for first all-pass filter
- * - filt_state2: Filter state for second all-pass filter
- *
- * Output:
- * - out_data : super-wb speech signal
- */
-void SplittingFilterSynthesis(const WebRtc_Word16* low_band,
- const WebRtc_Word16* high_band,
- WebRtc_Word16* out_data,
- WebRtc_Word32* filt_state1,
- WebRtc_Word32* filt_state2);
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_MAIN_SOURCE_SPLITTING_FILTER_H_
diff --git a/trunk/src/modules/audio_processing/test/android/apmtest/AndroidManifest.xml b/trunk/src/modules/audio_processing/test/android/apmtest/AndroidManifest.xml
deleted file mode 100644
index c6063b3..0000000
--- a/trunk/src/modules/audio_processing/test/android/apmtest/AndroidManifest.xml
+++ /dev/null
@@ -1,30 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<!-- BEGIN_INCLUDE(manifest) -->
-<manifest xmlns:android="http://schemas.android.com/apk/res/android"
- package="com.example.native_activity"
- android:versionCode="1"
- android:versionName="1.0">
-
- <!-- This is the platform API where NativeActivity was introduced. -->
- <uses-sdk android:minSdkVersion="8" />
-
- <!-- This .apk has no Java code itself, so set hasCode to false. -->
- <application android:label="@string/app_name" android:hasCode="false" android:debuggable="true">
-
- <!-- Our activity is the built-in NativeActivity framework class.
- This will take care of integrating with our NDK code. -->
- <activity android:name="android.app.NativeActivity"
- android:label="@string/app_name"
- android:configChanges="orientation|keyboardHidden">
- <!-- Tell NativeActivity the name of or .so -->
- <meta-data android:name="android.app.lib_name"
- android:value="apmtest-activity" />
- <intent-filter>
- <action android:name="android.intent.action.MAIN" />
- <category android:name="android.intent.category.LAUNCHER" />
- </intent-filter>
- </activity>
- </application>
-
-</manifest>
-<!-- END_INCLUDE(manifest) -->
diff --git a/trunk/src/modules/audio_processing/test/android/apmtest/default.properties b/trunk/src/modules/audio_processing/test/android/apmtest/default.properties
deleted file mode 100644
index 9a2c9f6..0000000
--- a/trunk/src/modules/audio_processing/test/android/apmtest/default.properties
+++ /dev/null
@@ -1,11 +0,0 @@
-# This file is automatically generated by Android Tools.
-# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
-#
-# This file must be checked in Version Control Systems.
-#
-# To customize properties used by the Ant build system use,
-# "build.properties", and override values to adapt the script to your
-# project structure.
-
-# Project target.
-target=android-9
diff --git a/trunk/src/modules/audio_processing/test/android/apmtest/jni/Android.mk b/trunk/src/modules/audio_processing/test/android/apmtest/jni/Android.mk
deleted file mode 100644
index eaf3c9d..0000000
--- a/trunk/src/modules/audio_processing/test/android/apmtest/jni/Android.mk
+++ /dev/null
@@ -1,26 +0,0 @@
-# Copyright (C) 2010 The Android Open Source Project
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-# http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-#
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE := apmtest-activity
-LOCAL_SRC_FILES := main.c
-LOCAL_LDLIBS := -llog -landroid -lEGL -lGLESv1_CM
-LOCAL_STATIC_LIBRARIES := android_native_app_glue
-
-include $(BUILD_SHARED_LIBRARY)
-
-$(call import-module,android/native_app_glue)
diff --git a/trunk/src/modules/audio_processing/test/android/apmtest/jni/Application.mk b/trunk/src/modules/audio_processing/test/android/apmtest/jni/Application.mk
deleted file mode 100644
index 22d188e..0000000
--- a/trunk/src/modules/audio_processing/test/android/apmtest/jni/Application.mk
+++ /dev/null
@@ -1 +0,0 @@
-APP_PLATFORM := android-9
diff --git a/trunk/src/modules/audio_processing/test/android/apmtest/jni/main.c b/trunk/src/modules/audio_processing/test/android/apmtest/jni/main.c
deleted file mode 100644
index 2e19635..0000000
--- a/trunk/src/modules/audio_processing/test/android/apmtest/jni/main.c
+++ /dev/null
@@ -1,307 +0,0 @@
-/*
- * Copyright (C) 2010 The Android Open Source Project
- *
- * Licensed under the Apache License, Version 2.0 (the "License");
- * you may not use this file except in compliance with the License.
- * You may obtain a copy of the License at
- *
- * http://www.apache.org/licenses/LICENSE-2.0
- *
- * Unless required by applicable law or agreed to in writing, software
- * distributed under the License is distributed on an "AS IS" BASIS,
- * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
- * See the License for the specific language governing permissions and
- * limitations under the License.
- *
- */
-
-//BEGIN_INCLUDE(all)
-#include <jni.h>
-#include <errno.h>
-
-#include <EGL/egl.h>
-#include <GLES/gl.h>
-
-#include <android/sensor.h>
-#include <android/log.h>
-#include <android_native_app_glue.h>
-
-#define LOGI(...) ((void)__android_log_print(ANDROID_LOG_INFO, "native-activity", __VA_ARGS__))
-#define LOGW(...) ((void)__android_log_print(ANDROID_LOG_WARN, "native-activity", __VA_ARGS__))
-
-/**
- * Our saved state data.
- */
-struct saved_state {
- float angle;
- int32_t x;
- int32_t y;
-};
-
-/**
- * Shared state for our app.
- */
-struct engine {
- struct android_app* app;
-
- ASensorManager* sensorManager;
- const ASensor* accelerometerSensor;
- ASensorEventQueue* sensorEventQueue;
-
- int animating;
- EGLDisplay display;
- EGLSurface surface;
- EGLContext context;
- int32_t width;
- int32_t height;
- struct saved_state state;
-};
-
-/**
- * Initialize an EGL context for the current display.
- */
-static int engine_init_display(struct engine* engine) {
- // initialize OpenGL ES and EGL
-
- /*
- * Here specify the attributes of the desired configuration.
- * Below, we select an EGLConfig with at least 8 bits per color
- * component compatible with on-screen windows
- */
- const EGLint attribs[] = {
- EGL_SURFACE_TYPE, EGL_WINDOW_BIT,
- EGL_BLUE_SIZE, 8,
- EGL_GREEN_SIZE, 8,
- EGL_RED_SIZE, 8,
- EGL_NONE
- };
- EGLint w, h, dummy, format;
- EGLint numConfigs;
- EGLConfig config;
- EGLSurface surface;
- EGLContext context;
-
- EGLDisplay display = eglGetDisplay(EGL_DEFAULT_DISPLAY);
-
- eglInitialize(display, 0, 0);
-
- /* Here, the application chooses the configuration it desires. In this
- * sample, we have a very simplified selection process, where we pick
- * the first EGLConfig that matches our criteria */
- eglChooseConfig(display, attribs, &config, 1, &numConfigs);
-
- /* EGL_NATIVE_VISUAL_ID is an attribute of the EGLConfig that is
- * guaranteed to be accepted by ANativeWindow_setBuffersGeometry().
- * As soon as we picked a EGLConfig, we can safely reconfigure the
- * ANativeWindow buffers to match, using EGL_NATIVE_VISUAL_ID. */
- eglGetConfigAttrib(display, config, EGL_NATIVE_VISUAL_ID, &format);
-
- ANativeWindow_setBuffersGeometry(engine->app->window, 0, 0, format);
-
- surface = eglCreateWindowSurface(display, config, engine->app->window, NULL);
- context = eglCreateContext(display, config, NULL, NULL);
-
- if (eglMakeCurrent(display, surface, surface, context) == EGL_FALSE) {
- LOGW("Unable to eglMakeCurrent");
- return -1;
- }
-
- eglQuerySurface(display, surface, EGL_WIDTH, &w);
- eglQuerySurface(display, surface, EGL_HEIGHT, &h);
-
- engine->display = display;
- engine->context = context;
- engine->surface = surface;
- engine->width = w;
- engine->height = h;
- engine->state.angle = 0;
-
- // Initialize GL state.
- glHint(GL_PERSPECTIVE_CORRECTION_HINT, GL_FASTEST);
- glEnable(GL_CULL_FACE);
- glShadeModel(GL_SMOOTH);
- glDisable(GL_DEPTH_TEST);
-
- return 0;
-}
-
-/**
- * Just the current frame in the display.
- */
-static void engine_draw_frame(struct engine* engine) {
- if (engine->display == NULL) {
- // No display.
- return;
- }
-
- // Just fill the screen with a color.
- glClearColor(((float)engine->state.x)/engine->width, engine->state.angle,
- ((float)engine->state.y)/engine->height, 1);
- glClear(GL_COLOR_BUFFER_BIT);
-
- eglSwapBuffers(engine->display, engine->surface);
-}
-
-/**
- * Tear down the EGL context currently associated with the display.
- */
-static void engine_term_display(struct engine* engine) {
- if (engine->display != EGL_NO_DISPLAY) {
- eglMakeCurrent(engine->display, EGL_NO_SURFACE, EGL_NO_SURFACE, EGL_NO_CONTEXT);
- if (engine->context != EGL_NO_CONTEXT) {
- eglDestroyContext(engine->display, engine->context);
- }
- if (engine->surface != EGL_NO_SURFACE) {
- eglDestroySurface(engine->display, engine->surface);
- }
- eglTerminate(engine->display);
- }
- engine->animating = 0;
- engine->display = EGL_NO_DISPLAY;
- engine->context = EGL_NO_CONTEXT;
- engine->surface = EGL_NO_SURFACE;
-}
-
-/**
- * Process the next input event.
- */
-static int32_t engine_handle_input(struct android_app* app, AInputEvent* event) {
- struct engine* engine = (struct engine*)app->userData;
- if (AInputEvent_getType(event) == AINPUT_EVENT_TYPE_MOTION) {
- engine->animating = 1;
- engine->state.x = AMotionEvent_getX(event, 0);
- engine->state.y = AMotionEvent_getY(event, 0);
- return 1;
- }
- return 0;
-}
-
-/**
- * Process the next main command.
- */
-static void engine_handle_cmd(struct android_app* app, int32_t cmd) {
- struct engine* engine = (struct engine*)app->userData;
- switch (cmd) {
- case APP_CMD_SAVE_STATE:
- // The system has asked us to save our current state. Do so.
- engine->app->savedState = malloc(sizeof(struct saved_state));
- *((struct saved_state*)engine->app->savedState) = engine->state;
- engine->app->savedStateSize = sizeof(struct saved_state);
- break;
- case APP_CMD_INIT_WINDOW:
- // The window is being shown, get it ready.
- if (engine->app->window != NULL) {
- engine_init_display(engine);
- engine_draw_frame(engine);
- }
- break;
- case APP_CMD_TERM_WINDOW:
- // The window is being hidden or closed, clean it up.
- engine_term_display(engine);
- break;
- case APP_CMD_GAINED_FOCUS:
- // When our app gains focus, we start monitoring the accelerometer.
- if (engine->accelerometerSensor != NULL) {
- ASensorEventQueue_enableSensor(engine->sensorEventQueue,
- engine->accelerometerSensor);
- // We'd like to get 60 events per second (in us).
- ASensorEventQueue_setEventRate(engine->sensorEventQueue,
- engine->accelerometerSensor, (1000L/60)*1000);
- }
- break;
- case APP_CMD_LOST_FOCUS:
- // When our app loses focus, we stop monitoring the accelerometer.
- // This is to avoid consuming battery while not being used.
- if (engine->accelerometerSensor != NULL) {
- ASensorEventQueue_disableSensor(engine->sensorEventQueue,
- engine->accelerometerSensor);
- }
- // Also stop animating.
- engine->animating = 0;
- engine_draw_frame(engine);
- break;
- }
-}
-
-/**
- * This is the main entry point of a native application that is using
- * android_native_app_glue. It runs in its own thread, with its own
- * event loop for receiving input events and doing other things.
- */
-void android_main(struct android_app* state) {
- struct engine engine;
-
- // Make sure glue isn't stripped.
- app_dummy();
-
- memset(&engine, 0, sizeof(engine));
- state->userData = &engine;
- state->onAppCmd = engine_handle_cmd;
- state->onInputEvent = engine_handle_input;
- engine.app = state;
-
- // Prepare to monitor accelerometer
- engine.sensorManager = ASensorManager_getInstance();
- engine.accelerometerSensor = ASensorManager_getDefaultSensor(engine.sensorManager,
- ASENSOR_TYPE_ACCELEROMETER);
- engine.sensorEventQueue = ASensorManager_createEventQueue(engine.sensorManager,
- state->looper, LOOPER_ID_USER, NULL, NULL);
-
- if (state->savedState != NULL) {
- // We are starting with a previous saved state; restore from it.
- engine.state = *(struct saved_state*)state->savedState;
- }
-
- // loop waiting for stuff to do.
-
- while (1) {
- // Read all pending events.
- int ident;
- int events;
- struct android_poll_source* source;
-
- // If not animating, we will block forever waiting for events.
- // If animating, we loop until all events are read, then continue
- // to draw the next frame of animation.
- while ((ident=ALooper_pollAll(engine.animating ? 0 : -1, NULL, &events,
- (void**)&source)) >= 0) {
-
- // Process this event.
- if (source != NULL) {
- source->process(state, source);
- }
-
- // If a sensor has data, process it now.
- if (ident == LOOPER_ID_USER) {
- if (engine.accelerometerSensor != NULL) {
- ASensorEvent event;
- while (ASensorEventQueue_getEvents(engine.sensorEventQueue,
- &event, 1) > 0) {
- LOGI("accelerometer: x=%f y=%f z=%f",
- event.acceleration.x, event.acceleration.y,
- event.acceleration.z);
- }
- }
- }
-
- // Check if we are exiting.
- if (state->destroyRequested != 0) {
- engine_term_display(&engine);
- return;
- }
- }
-
- if (engine.animating) {
- // Done with events; draw next animation frame.
- engine.state.angle += .01f;
- if (engine.state.angle > 1) {
- engine.state.angle = 0;
- }
-
- // Drawing is throttled to the screen update rate, so there
- // is no need to do timing here.
- engine_draw_frame(&engine);
- }
- }
-}
-//END_INCLUDE(all)
diff --git a/trunk/src/modules/audio_processing/test/android/apmtest/res/values/strings.xml b/trunk/src/modules/audio_processing/test/android/apmtest/res/values/strings.xml
deleted file mode 100644
index d0bd0f3..0000000
--- a/trunk/src/modules/audio_processing/test/android/apmtest/res/values/strings.xml
+++ /dev/null
@@ -1,4 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<resources>
- <string name="app_name">apmtest</string>
-</resources>
diff --git a/trunk/src/modules/audio_processing/test/apmtest.m b/trunk/src/modules/audio_processing/test/apmtest.m
deleted file mode 100644
index 3172cd1..0000000
--- a/trunk/src/modules/audio_processing/test/apmtest.m
+++ /dev/null
@@ -1,355 +0,0 @@
-function apmtest(task, testname, filepath, casenumber, legacy)
-%APMTEST is a tool to process APM file sets and easily display the output.
-% APMTEST(TASK, TESTNAME, CASENUMBER) performs one of several TASKs:
-% 'test' Processes the files to produce test output.
-% 'list' Prints a list of cases in the test set, preceded by their
-% CASENUMBERs.
-% 'show' Uses spclab to show the test case specified by the
-% CASENUMBER parameter.
-%
-% using a set of test files determined by TESTNAME:
-% 'all' All tests.
-% 'apm' The standard APM test set (default).
-% 'apmm' The mobile APM test set.
-% 'aec' The AEC test set.
-% 'aecm' The AECM test set.
-% 'agc' The AGC test set.
-% 'ns' The NS test set.
-% 'vad' The VAD test set.
-%
-% FILEPATH specifies the path to the test data files.
-%
-% CASENUMBER can be used to select a single test case. Omit CASENUMBER,
-% or set to zero, to use all test cases.
-%
-
-if nargin < 5 || isempty(legacy)
- % Set to true to run old VQE recordings.
- legacy = false;
-end
-
-if nargin < 4 || isempty(casenumber)
- casenumber = 0;
-end
-
-if nargin < 3 || isempty(filepath)
- filepath = 'data/';
-end
-
-if nargin < 2 || isempty(testname)
- testname = 'all';
-end
-
-if nargin < 1 || isempty(task)
- task = 'test';
-end
-
-if ~strcmp(task, 'test') && ~strcmp(task, 'list') && ~strcmp(task, 'show')
- error(['TASK ' task ' is not recognized']);
-end
-
-if casenumber == 0 && strcmp(task, 'show')
- error(['CASENUMBER must be specified for TASK ' task]);
-end
-
-inpath = [filepath 'input/'];
-outpath = [filepath 'output/'];
-refpath = [filepath 'reference/'];
-
-if strcmp(testname, 'all')
- tests = {'apm','apmm','aec','aecm','agc','ns','vad'};
-else
- tests = {testname};
-end
-
-if legacy
- progname = './test';
-else
- progname = './process_test';
-end
-
-global farFile;
-global nearFile;
-global eventFile;
-global delayFile;
-global driftFile;
-
-if legacy
- farFile = 'vqeFar.pcm';
- nearFile = 'vqeNear.pcm';
- eventFile = 'vqeEvent.dat';
- delayFile = 'vqeBuf.dat';
- driftFile = 'vqeDrift.dat';
-else
- farFile = 'apm_far.pcm';
- nearFile = 'apm_near.pcm';
- eventFile = 'apm_event.dat';
- delayFile = 'apm_delay.dat';
- driftFile = 'apm_drift.dat';
-end
-
-simulateMode = false;
-nErr = 0;
-nCases = 0;
-for i=1:length(tests)
- simulateMode = false;
-
- if strcmp(tests{i}, 'apm')
- testdir = ['apm/'];
- outfile = ['out'];
- if legacy
- opt = ['-ec 1 -agc 2 -nc 2 -vad 3'];
- else
- opt = ['--no_progress -hpf' ...
- ' -aec --drift_compensation -agc --fixed_digital' ...
- ' -ns --ns_moderate -vad'];
- end
-
- elseif strcmp(tests{i}, 'apm-swb')
- simulateMode = true;
- testdir = ['apm-swb/'];
- outfile = ['out'];
- if legacy
- opt = ['-fs 32000 -ec 1 -agc 2 -nc 2'];
- else
- opt = ['--no_progress -fs 32000 -hpf' ...
- ' -aec --drift_compensation -agc --adaptive_digital' ...
- ' -ns --ns_moderate -vad'];
- end
- elseif strcmp(tests{i}, 'apmm')
- testdir = ['apmm/'];
- outfile = ['out'];
- opt = ['-aec --drift_compensation -agc --fixed_digital -hpf -ns ' ...
- '--ns_moderate'];
-
- else
- error(['TESTNAME ' tests{i} ' is not recognized']);
- end
-
- inpathtest = [inpath testdir];
- outpathtest = [outpath testdir];
- refpathtest = [refpath testdir];
-
- if ~exist(inpathtest,'dir')
- error(['Input directory ' inpathtest ' does not exist']);
- end
-
- if ~exist(refpathtest,'dir')
- warning(['Reference directory ' refpathtest ' does not exist']);
- end
-
- [status, errMsg] = mkdir(outpathtest);
- if (status == 0)
- error(errMsg);
- end
-
- [nErr, nCases] = recurseDir(inpathtest, outpathtest, refpathtest, outfile, ...
- progname, opt, simulateMode, nErr, nCases, task, casenumber, legacy);
-
- if strcmp(task, 'test') || strcmp(task, 'show')
- system(['rm ' farFile]);
- system(['rm ' nearFile]);
- if simulateMode == false
- system(['rm ' eventFile]);
- system(['rm ' delayFile]);
- system(['rm ' driftFile]);
- end
- end
-end
-
-if ~strcmp(task, 'list')
- if nErr == 0
- fprintf(1, '\nAll files are bit-exact to reference\n', nErr);
- else
- fprintf(1, '\n%d files are NOT bit-exact to reference\n', nErr);
- end
-end
-
-
-function [nErrOut, nCases] = recurseDir(inpath, outpath, refpath, ...
- outfile, progname, opt, simulateMode, nErr, nCases, task, casenumber, ...
- legacy)
-
-global farFile;
-global nearFile;
-global eventFile;
-global delayFile;
-global driftFile;
-
-dirs = dir(inpath);
-nDirs = 0;
-nErrOut = nErr;
-for i=3:length(dirs) % skip . and ..
- nDirs = nDirs + dirs(i).isdir;
-end
-
-
-if nDirs == 0
- nCases = nCases + 1;
-
- if casenumber == nCases || casenumber == 0
-
- if strcmp(task, 'list')
- fprintf([num2str(nCases) '. ' outfile '\n'])
- else
- vadoutfile = ['vad_' outfile '.dat'];
- outfile = [outfile '.pcm'];
-
- % Check for VAD test
- vadTest = 0;
- if ~isempty(findstr(opt, '-vad'))
- vadTest = 1;
- if legacy
- opt = [opt ' ' outpath vadoutfile];
- else
- opt = [opt ' --vad_out_file ' outpath vadoutfile];
- end
- end
-
- if exist([inpath 'vqeFar.pcm'])
- system(['ln -s -f ' inpath 'vqeFar.pcm ' farFile]);
- elseif exist([inpath 'apm_far.pcm'])
- system(['ln -s -f ' inpath 'apm_far.pcm ' farFile]);
- end
-
- if exist([inpath 'vqeNear.pcm'])
- system(['ln -s -f ' inpath 'vqeNear.pcm ' nearFile]);
- elseif exist([inpath 'apm_near.pcm'])
- system(['ln -s -f ' inpath 'apm_near.pcm ' nearFile]);
- end
-
- if exist([inpath 'vqeEvent.dat'])
- system(['ln -s -f ' inpath 'vqeEvent.dat ' eventFile]);
- elseif exist([inpath 'apm_event.dat'])
- system(['ln -s -f ' inpath 'apm_event.dat ' eventFile]);
- end
-
- if exist([inpath 'vqeBuf.dat'])
- system(['ln -s -f ' inpath 'vqeBuf.dat ' delayFile]);
- elseif exist([inpath 'apm_delay.dat'])
- system(['ln -s -f ' inpath 'apm_delay.dat ' delayFile]);
- end
-
- if exist([inpath 'vqeSkew.dat'])
- system(['ln -s -f ' inpath 'vqeSkew.dat ' driftFile]);
- elseif exist([inpath 'vqeDrift.dat'])
- system(['ln -s -f ' inpath 'vqeDrift.dat ' driftFile]);
- elseif exist([inpath 'apm_drift.dat'])
- system(['ln -s -f ' inpath 'apm_drift.dat ' driftFile]);
- end
-
- if simulateMode == false
- command = [progname ' -o ' outpath outfile ' ' opt];
- else
- if legacy
- inputCmd = [' -in ' nearFile];
- else
- inputCmd = [' -i ' nearFile];
- end
-
- if exist([farFile])
- if legacy
- inputCmd = [' -if ' farFile inputCmd];
- else
- inputCmd = [' -ir ' farFile inputCmd];
- end
- end
- command = [progname inputCmd ' -o ' outpath outfile ' ' opt];
- end
- % This prevents MATLAB from using its own C libraries.
- shellcmd = ['bash -c "unset LD_LIBRARY_PATH;'];
- fprintf([command '\n']);
- [status, result] = system([shellcmd command '"']);
- fprintf(result);
-
- fprintf(['Reference file: ' refpath outfile '\n']);
-
- if vadTest == 1
- equal_to_ref = are_files_equal([outpath vadoutfile], ...
- [refpath vadoutfile], ...
- 'int8');
- if ~equal_to_ref
- nErr = nErr + 1;
- end
- end
-
- [equal_to_ref, diffvector] = are_files_equal([outpath outfile], ...
- [refpath outfile], ...
- 'int16');
- if ~equal_to_ref
- nErr = nErr + 1;
- end
-
- if strcmp(task, 'show')
- % Assume the last init gives the sample rate of interest.
- str_idx = strfind(result, 'Sample rate:');
- fs = str2num(result(str_idx(end) + 13:str_idx(end) + 17));
- fprintf('Using %d Hz\n', fs);
-
- if exist([farFile])
- spclab(fs, farFile, nearFile, [refpath outfile], ...
- [outpath outfile], diffvector);
- %spclab(fs, diffvector);
- else
- spclab(fs, nearFile, [refpath outfile], [outpath outfile], ...
- diffvector);
- %spclab(fs, diffvector);
- end
- end
- end
- end
-else
-
- for i=3:length(dirs)
- if dirs(i).isdir
- [nErr, nCases] = recurseDir([inpath dirs(i).name '/'], outpath, ...
- refpath,[outfile '_' dirs(i).name], progname, opt, ...
- simulateMode, nErr, nCases, task, casenumber, legacy);
- end
- end
-end
-nErrOut = nErr;
-
-function [are_equal, diffvector] = ...
- are_files_equal(newfile, reffile, precision, diffvector)
-
-are_equal = false;
-diffvector = 0;
-if ~exist(newfile,'file')
- warning(['Output file ' newfile ' does not exist']);
- return
-end
-
-if ~exist(reffile,'file')
- warning(['Reference file ' reffile ' does not exist']);
- return
-end
-
-fid = fopen(newfile,'rb');
-new = fread(fid,inf,precision);
-fclose(fid);
-
-fid = fopen(reffile,'rb');
-ref = fread(fid,inf,precision);
-fclose(fid);
-
-if length(new) ~= length(ref)
- warning('Reference is not the same length as output');
- minlength = min(length(new), length(ref));
- new = new(1:minlength);
- ref = ref(1:minlength);
-end
-diffvector = new - ref;
-
-if isequal(new, ref)
- fprintf([newfile ' is bit-exact to reference\n']);
- are_equal = true;
-else
- if isempty(new)
- warning([newfile ' is empty']);
- return
- end
- snr = snrseg(new,ref,80);
- fprintf('\n');
- are_equal = false;
-end
diff --git a/trunk/src/modules/audio_processing/test/process_test.cc b/trunk/src/modules/audio_processing/test/process_test.cc
deleted file mode 100644
index 2023ddb..0000000
--- a/trunk/src/modules/audio_processing/test/process_test.cc
+++ /dev/null
@@ -1,964 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <string.h>
-#ifdef WEBRTC_ANDROID
-#include <sys/stat.h>
-#endif
-
-#include "gtest/gtest.h"
-
-#include "audio_processing.h"
-#include "cpu_features_wrapper.h"
-#include "module_common_types.h"
-#include "scoped_ptr.h"
-#include "tick_util.h"
-#ifdef WEBRTC_ANDROID
-#include "external/webrtc/src/modules/audio_processing/debug.pb.h"
-#else
-#include "webrtc/audio_processing/debug.pb.h"
-#endif
-
-using webrtc::AudioFrame;
-using webrtc::AudioProcessing;
-using webrtc::EchoCancellation;
-using webrtc::GainControl;
-using webrtc::NoiseSuppression;
-using webrtc::scoped_array;
-using webrtc::TickInterval;
-using webrtc::TickTime;
-
-using webrtc::audioproc::Event;
-using webrtc::audioproc::Init;
-using webrtc::audioproc::ReverseStream;
-using webrtc::audioproc::Stream;
-
-namespace {
-// Returns true on success, false on error or end-of-file.
-bool ReadMessageFromFile(FILE* file,
- ::google::protobuf::MessageLite* msg) {
- // The "wire format" for the size is little-endian.
- // Assume process_test is running on a little-endian machine.
- int32_t size = 0;
- if (fread(&size, sizeof(int32_t), 1, file) != 1) {
- return false;
- }
- if (size <= 0) {
- return false;
- }
- const size_t usize = static_cast<size_t>(size);
-
- scoped_array<char> array(new char[usize]);
- if (fread(array.get(), sizeof(char), usize, file) != usize) {
- return false;
- }
-
- msg->Clear();
- return msg->ParseFromArray(array.get(), usize);
-}
-
-void PrintStat(const AudioProcessing::Statistic& stat) {
- printf("%d, %d, %d\n", stat.average,
- stat.maximum,
- stat.minimum);
-}
-
-void usage() {
- printf(
- "Usage: process_test [options] [-pb PROTOBUF_FILE]\n"
- " [-ir REVERSE_FILE] [-i PRIMARY_FILE] [-o OUT_FILE]\n");
- printf(
- "process_test is a test application for AudioProcessing.\n\n"
- "When a protobuf debug file is available, specify it with -pb.\n"
- "Alternately, when -ir or -i is used, the specified files will be\n"
- "processed directly in a simulation mode. Otherwise the full set of\n"
- "legacy test files is expected to be present in the working directory.\n");
- printf("\n");
- printf("Options\n");
- printf("General configuration (only used for the simulation mode):\n");
- printf(" -fs SAMPLE_RATE_HZ\n");
- printf(" -ch CHANNELS_IN CHANNELS_OUT\n");
- printf(" -rch REVERSE_CHANNELS\n");
- printf("\n");
- printf("Component configuration:\n");
- printf(
- "All components are disabled by default. Each block below begins with a\n"
- "flag to enable the component with default settings. The subsequent flags\n"
- "in the block are used to provide configuration settings.\n");
- printf("\n -aec Echo cancellation\n");
- printf(" --drift_compensation\n");
- printf(" --no_drift_compensation\n");
- printf(" --no_echo_metrics\n");
- printf(" --no_delay_logging\n");
- printf("\n -aecm Echo control mobile\n");
- printf(" --aecm_echo_path_in_file FILE\n");
- printf(" --aecm_echo_path_out_file FILE\n");
- printf("\n -agc Gain control\n");
- printf(" --analog\n");
- printf(" --adaptive_digital\n");
- printf(" --fixed_digital\n");
- printf(" --target_level LEVEL\n");
- printf(" --compression_gain GAIN\n");
- printf(" --limiter\n");
- printf(" --no_limiter\n");
- printf("\n -hpf High pass filter\n");
- printf("\n -ns Noise suppression\n");
- printf(" --ns_low\n");
- printf(" --ns_moderate\n");
- printf(" --ns_high\n");
- printf(" --ns_very_high\n");
- printf("\n -vad Voice activity detection\n");
- printf(" --vad_out_file FILE\n");
- printf("\n Level metrics (enabled by default)\n");
- printf(" --no_level_metrics\n");
- printf("\n");
- printf("Modifiers:\n");
- printf(" --noasm Disable SSE optimization.\n");
- printf(" --delay DELAY Add DELAY ms to input value.\n");
- printf(" --perf Measure performance.\n");
- printf(" --quiet Suppress text output.\n");
- printf(" --no_progress Suppress progress.\n");
- printf(" --debug_file FILE Dump a debug recording.\n");
-}
-
-// void function for gtest.
-void void_main(int argc, char* argv[]) {
- if (argc > 1 && strcmp(argv[1], "--help") == 0) {
- usage();
- return;
- }
-
- if (argc < 2) {
- printf("Did you mean to run without arguments?\n");
- printf("Try `process_test --help' for more information.\n\n");
- }
-
- AudioProcessing* apm = AudioProcessing::Create(0);
- ASSERT_TRUE(apm != NULL);
-
- const char* pb_filename = NULL;
- const char* far_filename = NULL;
- const char* near_filename = NULL;
- const char* out_filename = NULL;
- const char* vad_out_filename = NULL;
- const char* aecm_echo_path_in_filename = NULL;
- const char* aecm_echo_path_out_filename = NULL;
-
- int32_t sample_rate_hz = 16000;
- int32_t device_sample_rate_hz = 16000;
-
- int num_capture_input_channels = 1;
- int num_capture_output_channels = 1;
- int num_render_channels = 1;
-
- int samples_per_channel = sample_rate_hz / 100;
-
- bool simulating = false;
- bool perf_testing = false;
- bool verbose = true;
- bool progress = true;
- int extra_delay_ms = 0;
- //bool interleaved = true;
-
- ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(true));
- for (int i = 1; i < argc; i++) {
- if (strcmp(argv[i], "-pb") == 0) {
- i++;
- ASSERT_LT(i, argc) << "Specify protobuf filename after -pb";
- pb_filename = argv[i];
-
- } else if (strcmp(argv[i], "-ir") == 0) {
- i++;
- ASSERT_LT(i, argc) << "Specify filename after -ir";
- far_filename = argv[i];
- simulating = true;
-
- } else if (strcmp(argv[i], "-i") == 0) {
- i++;
- ASSERT_LT(i, argc) << "Specify filename after -i";
- near_filename = argv[i];
- simulating = true;
-
- } else if (strcmp(argv[i], "-o") == 0) {
- i++;
- ASSERT_LT(i, argc) << "Specify filename after -o";
- out_filename = argv[i];
-
- } else if (strcmp(argv[i], "-fs") == 0) {
- i++;
- ASSERT_LT(i, argc) << "Specify sample rate after -fs";
- ASSERT_EQ(1, sscanf(argv[i], "%d", &sample_rate_hz));
- samples_per_channel = sample_rate_hz / 100;
-
- ASSERT_EQ(apm->kNoError,
- apm->set_sample_rate_hz(sample_rate_hz));
-
- } else if (strcmp(argv[i], "-ch") == 0) {
- i++;
- ASSERT_LT(i + 1, argc) << "Specify number of channels after -ch";
- ASSERT_EQ(1, sscanf(argv[i], "%d", &num_capture_input_channels));
- i++;
- ASSERT_EQ(1, sscanf(argv[i], "%d", &num_capture_output_channels));
-
- ASSERT_EQ(apm->kNoError,
- apm->set_num_channels(num_capture_input_channels,
- num_capture_output_channels));
-
- } else if (strcmp(argv[i], "-rch") == 0) {
- i++;
- ASSERT_LT(i, argc) << "Specify number of channels after -rch";
- ASSERT_EQ(1, sscanf(argv[i], "%d", &num_render_channels));
-
- ASSERT_EQ(apm->kNoError,
- apm->set_num_reverse_channels(num_render_channels));
-
- } else if (strcmp(argv[i], "-aec") == 0) {
- ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->echo_cancellation()->enable_metrics(true));
- ASSERT_EQ(apm->kNoError,
- apm->echo_cancellation()->enable_delay_logging(true));
-
- } else if (strcmp(argv[i], "--drift_compensation") == 0) {
- ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
- // TODO(ajm): this is enabled in the VQE test app by default. Investigate
- // why it can give better performance despite passing zeros.
- ASSERT_EQ(apm->kNoError,
- apm->echo_cancellation()->enable_drift_compensation(true));
- } else if (strcmp(argv[i], "--no_drift_compensation") == 0) {
- ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->echo_cancellation()->enable_drift_compensation(false));
-
- } else if (strcmp(argv[i], "--no_echo_metrics") == 0) {
- ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->echo_cancellation()->enable_metrics(false));
-
- } else if (strcmp(argv[i], "--no_delay_logging") == 0) {
- ASSERT_EQ(apm->kNoError, apm->echo_cancellation()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->echo_cancellation()->enable_delay_logging(false));
-
- } else if (strcmp(argv[i], "--no_level_metrics") == 0) {
- ASSERT_EQ(apm->kNoError, apm->level_estimator()->Enable(false));
-
- } else if (strcmp(argv[i], "-aecm") == 0) {
- ASSERT_EQ(apm->kNoError, apm->echo_control_mobile()->Enable(true));
-
- } else if (strcmp(argv[i], "--aecm_echo_path_in_file") == 0) {
- i++;
- ASSERT_LT(i, argc) << "Specify filename after --aecm_echo_path_in_file";
- aecm_echo_path_in_filename = argv[i];
-
- } else if (strcmp(argv[i], "--aecm_echo_path_out_file") == 0) {
- i++;
- ASSERT_LT(i, argc) << "Specify filename after --aecm_echo_path_out_file";
- aecm_echo_path_out_filename = argv[i];
-
- } else if (strcmp(argv[i], "-agc") == 0) {
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
-
- } else if (strcmp(argv[i], "--analog") == 0) {
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
-
- } else if (strcmp(argv[i], "--adaptive_digital") == 0) {
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->set_mode(GainControl::kAdaptiveDigital));
-
- } else if (strcmp(argv[i], "--fixed_digital") == 0) {
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->set_mode(GainControl::kFixedDigital));
-
- } else if (strcmp(argv[i], "--target_level") == 0) {
- i++;
- int level;
- ASSERT_EQ(1, sscanf(argv[i], "%d", &level));
-
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->set_target_level_dbfs(level));
-
- } else if (strcmp(argv[i], "--compression_gain") == 0) {
- i++;
- int gain;
- ASSERT_EQ(1, sscanf(argv[i], "%d", &gain));
-
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->set_compression_gain_db(gain));
-
- } else if (strcmp(argv[i], "--limiter") == 0) {
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->enable_limiter(true));
-
- } else if (strcmp(argv[i], "--no_limiter") == 0) {
- ASSERT_EQ(apm->kNoError, apm->gain_control()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->enable_limiter(false));
-
- } else if (strcmp(argv[i], "-hpf") == 0) {
- ASSERT_EQ(apm->kNoError, apm->high_pass_filter()->Enable(true));
-
- } else if (strcmp(argv[i], "-ns") == 0) {
- ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
-
- } else if (strcmp(argv[i], "--ns_low") == 0) {
- ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->noise_suppression()->set_level(NoiseSuppression::kLow));
-
- } else if (strcmp(argv[i], "--ns_moderate") == 0) {
- ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->noise_suppression()->set_level(NoiseSuppression::kModerate));
-
- } else if (strcmp(argv[i], "--ns_high") == 0) {
- ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->noise_suppression()->set_level(NoiseSuppression::kHigh));
-
- } else if (strcmp(argv[i], "--ns_very_high") == 0) {
- ASSERT_EQ(apm->kNoError, apm->noise_suppression()->Enable(true));
- ASSERT_EQ(apm->kNoError,
- apm->noise_suppression()->set_level(NoiseSuppression::kVeryHigh));
-
- } else if (strcmp(argv[i], "-vad") == 0) {
- ASSERT_EQ(apm->kNoError, apm->voice_detection()->Enable(true));
-
- } else if (strcmp(argv[i], "--vad_out_file") == 0) {
- i++;
- ASSERT_LT(i, argc) << "Specify filename after --vad_out_file";
- vad_out_filename = argv[i];
-
- } else if (strcmp(argv[i], "--noasm") == 0) {
- WebRtc_GetCPUInfo = WebRtc_GetCPUInfoNoASM;
- // We need to reinitialize here if components have already been enabled.
- ASSERT_EQ(apm->kNoError, apm->Initialize());
-
- } else if (strcmp(argv[i], "--delay") == 0) {
- i++;
- ASSERT_EQ(1, sscanf(argv[i], "%d", &extra_delay_ms));
-
- } else if (strcmp(argv[i], "--perf") == 0) {
- perf_testing = true;
-
- } else if (strcmp(argv[i], "--quiet") == 0) {
- verbose = false;
- progress = false;
-
- } else if (strcmp(argv[i], "--no_progress") == 0) {
- progress = false;
-
- } else if (strcmp(argv[i], "--debug_file") == 0) {
- i++;
- ASSERT_LT(i, argc) << "Specify filename after --debug_file";
- ASSERT_EQ(apm->kNoError, apm->StartDebugRecording(argv[i]));
- } else {
- FAIL() << "Unrecognized argument " << argv[i];
- }
- }
- // If we're reading a protobuf file, ensure a simulation hasn't also
- // been requested (which makes no sense...)
- ASSERT_FALSE(pb_filename && simulating);
-
- if (verbose) {
- printf("Sample rate: %d Hz\n", sample_rate_hz);
- printf("Primary channels: %d (in), %d (out)\n",
- num_capture_input_channels,
- num_capture_output_channels);
- printf("Reverse channels: %d \n", num_render_channels);
- }
-
- const char far_file_default[] = "apm_far.pcm";
- const char near_file_default[] = "apm_near.pcm";
- const char out_file_default[] = "out.pcm";
- const char event_filename[] = "apm_event.dat";
- const char delay_filename[] = "apm_delay.dat";
- const char drift_filename[] = "apm_drift.dat";
- const char vad_file_default[] = "vad_out.dat";
-
- if (!simulating) {
- far_filename = far_file_default;
- near_filename = near_file_default;
- }
-
- if (!out_filename) {
- out_filename = out_file_default;
- }
-
- if (!vad_out_filename) {
- vad_out_filename = vad_file_default;
- }
-
- FILE* pb_file = NULL;
- FILE* far_file = NULL;
- FILE* near_file = NULL;
- FILE* out_file = NULL;
- FILE* event_file = NULL;
- FILE* delay_file = NULL;
- FILE* drift_file = NULL;
- FILE* vad_out_file = NULL;
- FILE* aecm_echo_path_in_file = NULL;
- FILE* aecm_echo_path_out_file = NULL;
-
- if (pb_filename) {
- pb_file = fopen(pb_filename, "rb");
- ASSERT_TRUE(NULL != pb_file) << "Unable to open protobuf file "
- << pb_filename;
- } else {
- if (far_filename) {
- far_file = fopen(far_filename, "rb");
- ASSERT_TRUE(NULL != far_file) << "Unable to open far-end audio file "
- << far_filename;
- }
-
- near_file = fopen(near_filename, "rb");
- ASSERT_TRUE(NULL != near_file) << "Unable to open near-end audio file "
- << near_filename;
- if (!simulating) {
- event_file = fopen(event_filename, "rb");
- ASSERT_TRUE(NULL != event_file) << "Unable to open event file "
- << event_filename;
-
- delay_file = fopen(delay_filename, "rb");
- ASSERT_TRUE(NULL != delay_file) << "Unable to open buffer file "
- << delay_filename;
-
- drift_file = fopen(drift_filename, "rb");
- ASSERT_TRUE(NULL != drift_file) << "Unable to open drift file "
- << drift_filename;
- }
- }
-
- out_file = fopen(out_filename, "wb");
- ASSERT_TRUE(NULL != out_file) << "Unable to open output audio file "
- << out_filename;
-
- int near_size_bytes = 0;
- if (pb_file) {
- struct stat st;
- stat(pb_filename, &st);
- // Crude estimate, but should be good enough.
- near_size_bytes = st.st_size / 3;
- } else {
- struct stat st;
- stat(near_filename, &st);
- near_size_bytes = st.st_size;
- }
-
- if (apm->voice_detection()->is_enabled()) {
- vad_out_file = fopen(vad_out_filename, "wb");
- ASSERT_TRUE(NULL != vad_out_file) << "Unable to open VAD output file "
- << vad_out_file;
- }
-
- if (aecm_echo_path_in_filename != NULL) {
- aecm_echo_path_in_file = fopen(aecm_echo_path_in_filename, "rb");
- ASSERT_TRUE(NULL != aecm_echo_path_in_file) << "Unable to open file "
- << aecm_echo_path_in_filename;
-
- const size_t path_size =
- apm->echo_control_mobile()->echo_path_size_bytes();
- scoped_array<char> echo_path(new char[path_size]);
- ASSERT_EQ(path_size, fread(echo_path.get(),
- sizeof(char),
- path_size,
- aecm_echo_path_in_file));
- EXPECT_EQ(apm->kNoError,
- apm->echo_control_mobile()->SetEchoPath(echo_path.get(),
- path_size));
- fclose(aecm_echo_path_in_file);
- aecm_echo_path_in_file = NULL;
- }
-
- if (aecm_echo_path_out_filename != NULL) {
- aecm_echo_path_out_file = fopen(aecm_echo_path_out_filename, "wb");
- ASSERT_TRUE(NULL != aecm_echo_path_out_file) << "Unable to open file "
- << aecm_echo_path_out_filename;
- }
-
- size_t read_count = 0;
- int reverse_count = 0;
- int primary_count = 0;
- int near_read_bytes = 0;
- TickInterval acc_ticks;
-
- AudioFrame far_frame;
- AudioFrame near_frame;
-
- int delay_ms = 0;
- int drift_samples = 0;
- int capture_level = 127;
- int8_t stream_has_voice = 0;
-
- TickTime t0 = TickTime::Now();
- TickTime t1 = t0;
- WebRtc_Word64 max_time_us = 0;
- WebRtc_Word64 max_time_reverse_us = 0;
- WebRtc_Word64 min_time_us = 1e6;
- WebRtc_Word64 min_time_reverse_us = 1e6;
-
- // TODO(ajm): Ideally we would refactor this block into separate functions,
- // but for now we want to share the variables.
- if (pb_file) {
- Event event_msg;
- while (ReadMessageFromFile(pb_file, &event_msg)) {
- std::ostringstream trace_stream;
- trace_stream << "Processed frames: " << reverse_count << " (reverse), "
- << primary_count << " (primary)";
- SCOPED_TRACE(trace_stream.str());
-
- if (event_msg.type() == Event::INIT) {
- ASSERT_TRUE(event_msg.has_init());
- const Init msg = event_msg.init();
-
- ASSERT_TRUE(msg.has_sample_rate());
- ASSERT_EQ(apm->kNoError,
- apm->set_sample_rate_hz(msg.sample_rate()));
-
- ASSERT_TRUE(msg.has_device_sample_rate());
- ASSERT_EQ(apm->kNoError,
- apm->echo_cancellation()->set_device_sample_rate_hz(
- msg.device_sample_rate()));
-
- ASSERT_TRUE(msg.has_num_input_channels());
- ASSERT_TRUE(msg.has_num_output_channels());
- ASSERT_EQ(apm->kNoError,
- apm->set_num_channels(msg.num_input_channels(),
- msg.num_output_channels()));
-
- ASSERT_TRUE(msg.has_num_reverse_channels());
- ASSERT_EQ(apm->kNoError,
- apm->set_num_reverse_channels(msg.num_reverse_channels()));
-
- samples_per_channel = msg.sample_rate() / 100;
- far_frame._frequencyInHz = msg.sample_rate();
- far_frame._payloadDataLengthInSamples = samples_per_channel;
- far_frame._audioChannel = msg.num_reverse_channels();
- near_frame._frequencyInHz = msg.sample_rate();
- near_frame._payloadDataLengthInSamples = samples_per_channel;
-
- if (verbose) {
- printf("Init at frame: %d (primary), %d (reverse)\n",
- primary_count, reverse_count);
- printf(" Sample rate: %d Hz\n", msg.sample_rate());
- printf(" Primary channels: %d (in), %d (out)\n",
- msg.num_input_channels(),
- msg.num_output_channels());
- printf(" Reverse channels: %d \n", msg.num_reverse_channels());
- }
-
- } else if (event_msg.type() == Event::REVERSE_STREAM) {
- ASSERT_TRUE(event_msg.has_reverse_stream());
- const ReverseStream msg = event_msg.reverse_stream();
- reverse_count++;
-
- ASSERT_TRUE(msg.has_data());
- ASSERT_EQ(sizeof(int16_t) * samples_per_channel *
- far_frame._audioChannel, msg.data().size());
- memcpy(far_frame._payloadData, msg.data().data(), msg.data().size());
-
- if (perf_testing) {
- t0 = TickTime::Now();
- }
-
- ASSERT_EQ(apm->kNoError,
- apm->AnalyzeReverseStream(&far_frame));
-
- if (perf_testing) {
- t1 = TickTime::Now();
- TickInterval tick_diff = t1 - t0;
- acc_ticks += tick_diff;
- if (tick_diff.Microseconds() > max_time_reverse_us) {
- max_time_reverse_us = tick_diff.Microseconds();
- }
- if (tick_diff.Microseconds() < min_time_reverse_us) {
- min_time_reverse_us = tick_diff.Microseconds();
- }
- }
-
- } else if (event_msg.type() == Event::STREAM) {
- ASSERT_TRUE(event_msg.has_stream());
- const Stream msg = event_msg.stream();
- primary_count++;
-
- // ProcessStream could have changed this for the output frame.
- near_frame._audioChannel = apm->num_input_channels();
-
- ASSERT_TRUE(msg.has_input_data());
- ASSERT_EQ(sizeof(int16_t) * samples_per_channel *
- near_frame._audioChannel, msg.input_data().size());
- memcpy(near_frame._payloadData,
- msg.input_data().data(),
- msg.input_data().size());
-
- near_read_bytes += msg.input_data().size();
- if (progress && primary_count % 100 == 0) {
- printf("%.0f%% complete\r",
- (near_read_bytes * 100.0) / near_size_bytes);
- fflush(stdout);
- }
-
- if (perf_testing) {
- t0 = TickTime::Now();
- }
-
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->set_stream_analog_level(msg.level()));
- ASSERT_EQ(apm->kNoError,
- apm->set_stream_delay_ms(msg.delay() + extra_delay_ms));
- ASSERT_EQ(apm->kNoError,
- apm->echo_cancellation()->set_stream_drift_samples(msg.drift()));
-
- int err = apm->ProcessStream(&near_frame);
- if (err == apm->kBadStreamParameterWarning) {
- printf("Bad parameter warning. %s\n", trace_stream.str().c_str());
- }
- ASSERT_TRUE(err == apm->kNoError ||
- err == apm->kBadStreamParameterWarning);
- ASSERT_TRUE(near_frame._audioChannel == apm->num_output_channels());
-
- capture_level = apm->gain_control()->stream_analog_level();
-
- stream_has_voice =
- static_cast<int8_t>(apm->voice_detection()->stream_has_voice());
- if (vad_out_file != NULL) {
- ASSERT_EQ(1u, fwrite(&stream_has_voice,
- sizeof(stream_has_voice),
- 1,
- vad_out_file));
- }
-
- if (apm->gain_control()->mode() != GainControl::kAdaptiveAnalog) {
- ASSERT_EQ(msg.level(), capture_level);
- }
-
- if (perf_testing) {
- t1 = TickTime::Now();
- TickInterval tick_diff = t1 - t0;
- acc_ticks += tick_diff;
- if (tick_diff.Microseconds() > max_time_us) {
- max_time_us = tick_diff.Microseconds();
- }
- if (tick_diff.Microseconds() < min_time_us) {
- min_time_us = tick_diff.Microseconds();
- }
- }
-
- size_t size = samples_per_channel * near_frame._audioChannel;
- ASSERT_EQ(size, fwrite(near_frame._payloadData,
- sizeof(int16_t),
- size,
- out_file));
- }
- }
-
- ASSERT_TRUE(feof(pb_file));
-
- } else {
- enum Events {
- kInitializeEvent,
- kRenderEvent,
- kCaptureEvent,
- kResetEventDeprecated
- };
- int16_t event = 0;
- while (simulating || feof(event_file) == 0) {
- std::ostringstream trace_stream;
- trace_stream << "Processed frames: " << reverse_count << " (reverse), "
- << primary_count << " (primary)";
- SCOPED_TRACE(trace_stream.str());
-
- if (simulating) {
- if (far_file == NULL) {
- event = kCaptureEvent;
- } else {
- if (event == kRenderEvent) {
- event = kCaptureEvent;
- } else {
- event = kRenderEvent;
- }
- }
- } else {
- read_count = fread(&event, sizeof(event), 1, event_file);
- if (read_count != 1) {
- break;
- }
- }
-
- far_frame._frequencyInHz = sample_rate_hz;
- far_frame._payloadDataLengthInSamples = samples_per_channel;
- far_frame._audioChannel = num_render_channels;
- near_frame._frequencyInHz = sample_rate_hz;
- near_frame._payloadDataLengthInSamples = samples_per_channel;
-
- if (event == kInitializeEvent || event == kResetEventDeprecated) {
- ASSERT_EQ(1u,
- fread(&sample_rate_hz, sizeof(sample_rate_hz), 1, event_file));
- samples_per_channel = sample_rate_hz / 100;
-
- ASSERT_EQ(1u,
- fread(&device_sample_rate_hz,
- sizeof(device_sample_rate_hz),
- 1,
- event_file));
-
- ASSERT_EQ(apm->kNoError,
- apm->set_sample_rate_hz(sample_rate_hz));
-
- ASSERT_EQ(apm->kNoError,
- apm->echo_cancellation()->set_device_sample_rate_hz(
- device_sample_rate_hz));
-
- far_frame._frequencyInHz = sample_rate_hz;
- far_frame._payloadDataLengthInSamples = samples_per_channel;
- far_frame._audioChannel = num_render_channels;
- near_frame._frequencyInHz = sample_rate_hz;
- near_frame._payloadDataLengthInSamples = samples_per_channel;
-
- if (verbose) {
- printf("Init at frame: %d (primary), %d (reverse)\n",
- primary_count, reverse_count);
- printf(" Sample rate: %d Hz\n", sample_rate_hz);
- }
-
- } else if (event == kRenderEvent) {
- reverse_count++;
-
- size_t size = samples_per_channel * num_render_channels;
- read_count = fread(far_frame._payloadData,
- sizeof(int16_t),
- size,
- far_file);
-
- if (simulating) {
- if (read_count != size) {
- // Read an equal amount from the near file to avoid errors due to
- // not reaching end-of-file.
- EXPECT_EQ(0, fseek(near_file, read_count * sizeof(int16_t),
- SEEK_CUR));
- break; // This is expected.
- }
- } else {
- ASSERT_EQ(size, read_count);
- }
-
- if (perf_testing) {
- t0 = TickTime::Now();
- }
-
- ASSERT_EQ(apm->kNoError,
- apm->AnalyzeReverseStream(&far_frame));
-
- if (perf_testing) {
- t1 = TickTime::Now();
- TickInterval tick_diff = t1 - t0;
- acc_ticks += tick_diff;
- if (tick_diff.Microseconds() > max_time_reverse_us) {
- max_time_reverse_us = tick_diff.Microseconds();
- }
- if (tick_diff.Microseconds() < min_time_reverse_us) {
- min_time_reverse_us = tick_diff.Microseconds();
- }
- }
-
- } else if (event == kCaptureEvent) {
- primary_count++;
- near_frame._audioChannel = num_capture_input_channels;
-
- size_t size = samples_per_channel * num_capture_input_channels;
- read_count = fread(near_frame._payloadData,
- sizeof(int16_t),
- size,
- near_file);
-
- near_read_bytes += read_count * sizeof(int16_t);
- if (progress && primary_count % 100 == 0) {
- printf("%.0f%% complete\r",
- (near_read_bytes * 100.0) / near_size_bytes);
- fflush(stdout);
- }
- if (simulating) {
- if (read_count != size) {
- break; // This is expected.
- }
-
- delay_ms = 0;
- drift_samples = 0;
- } else {
- ASSERT_EQ(size, read_count);
-
- // TODO(ajm): sizeof(delay_ms) for current files?
- ASSERT_EQ(1u,
- fread(&delay_ms, 2, 1, delay_file));
- ASSERT_EQ(1u,
- fread(&drift_samples, sizeof(drift_samples), 1, drift_file));
- }
-
- if (perf_testing) {
- t0 = TickTime::Now();
- }
-
- // TODO(ajm): fake an analog gain while simulating.
-
- int capture_level_in = capture_level;
- ASSERT_EQ(apm->kNoError,
- apm->gain_control()->set_stream_analog_level(capture_level));
- ASSERT_EQ(apm->kNoError,
- apm->set_stream_delay_ms(delay_ms + extra_delay_ms));
- ASSERT_EQ(apm->kNoError,
- apm->echo_cancellation()->set_stream_drift_samples(drift_samples));
-
- int err = apm->ProcessStream(&near_frame);
- if (err == apm->kBadStreamParameterWarning) {
- printf("Bad parameter warning. %s\n", trace_stream.str().c_str());
- }
- ASSERT_TRUE(err == apm->kNoError ||
- err == apm->kBadStreamParameterWarning);
- ASSERT_TRUE(near_frame._audioChannel == apm->num_output_channels());
-
- capture_level = apm->gain_control()->stream_analog_level();
-
- stream_has_voice =
- static_cast<int8_t>(apm->voice_detection()->stream_has_voice());
- if (vad_out_file != NULL) {
- ASSERT_EQ(1u, fwrite(&stream_has_voice,
- sizeof(stream_has_voice),
- 1,
- vad_out_file));
- }
-
- if (apm->gain_control()->mode() != GainControl::kAdaptiveAnalog) {
- ASSERT_EQ(capture_level_in, capture_level);
- }
-
- if (perf_testing) {
- t1 = TickTime::Now();
- TickInterval tick_diff = t1 - t0;
- acc_ticks += tick_diff;
- if (tick_diff.Microseconds() > max_time_us) {
- max_time_us = tick_diff.Microseconds();
- }
- if (tick_diff.Microseconds() < min_time_us) {
- min_time_us = tick_diff.Microseconds();
- }
- }
-
- size = samples_per_channel * near_frame._audioChannel;
- ASSERT_EQ(size, fwrite(near_frame._payloadData,
- sizeof(int16_t),
- size,
- out_file));
- }
- else {
- FAIL() << "Event " << event << " is unrecognized";
- }
- }
- }
- printf("100%% complete\r");
-
- if (aecm_echo_path_out_file != NULL) {
- const size_t path_size =
- apm->echo_control_mobile()->echo_path_size_bytes();
- scoped_array<char> echo_path(new char[path_size]);
- apm->echo_control_mobile()->GetEchoPath(echo_path.get(), path_size);
- ASSERT_EQ(path_size, fwrite(echo_path.get(),
- sizeof(char),
- path_size,
- aecm_echo_path_out_file));
- fclose(aecm_echo_path_out_file);
- aecm_echo_path_out_file = NULL;
- }
-
- if (verbose) {
- printf("\nProcessed frames: %d (primary), %d (reverse)\n",
- primary_count, reverse_count);
-
- if (apm->level_estimator()->is_enabled()) {
- printf("\n--Level metrics--\n");
- printf("RMS: %d dBFS\n", -apm->level_estimator()->RMS());
- }
- if (apm->echo_cancellation()->are_metrics_enabled()) {
- EchoCancellation::Metrics metrics;
- apm->echo_cancellation()->GetMetrics(&metrics);
- printf("\n--Echo metrics--\n");
- printf("(avg, max, min)\n");
- printf("ERL: ");
- PrintStat(metrics.echo_return_loss);
- printf("ERLE: ");
- PrintStat(metrics.echo_return_loss_enhancement);
- printf("ANLP: ");
- PrintStat(metrics.a_nlp);
- }
- if (apm->echo_cancellation()->is_delay_logging_enabled()) {
- int median = 0;
- int std = 0;
- apm->echo_cancellation()->GetDelayMetrics(&median, &std);
- printf("\n--Delay metrics--\n");
- printf("Median: %3d\n", median);
- printf("Standard deviation: %3d\n", std);
- }
- }
-
- if (!pb_file) {
- int8_t temp_int8;
- if (far_file) {
- read_count = fread(&temp_int8, sizeof(temp_int8), 1, far_file);
- EXPECT_NE(0, feof(far_file)) << "Far-end file not fully processed";
- }
-
- read_count = fread(&temp_int8, sizeof(temp_int8), 1, near_file);
- EXPECT_NE(0, feof(near_file)) << "Near-end file not fully processed";
-
- if (!simulating) {
- read_count = fread(&temp_int8, sizeof(temp_int8), 1, event_file);
- EXPECT_NE(0, feof(event_file)) << "Event file not fully processed";
- read_count = fread(&temp_int8, sizeof(temp_int8), 1, delay_file);
- EXPECT_NE(0, feof(delay_file)) << "Delay file not fully processed";
- read_count = fread(&temp_int8, sizeof(temp_int8), 1, drift_file);
- EXPECT_NE(0, feof(drift_file)) << "Drift file not fully processed";
- }
- }
-
- if (perf_testing) {
- if (primary_count > 0) {
- WebRtc_Word64 exec_time = acc_ticks.Milliseconds();
- printf("\nTotal time: %.3f s, file time: %.2f s\n",
- exec_time * 0.001, primary_count * 0.01);
- printf("Time per frame: %.3f ms (average), %.3f ms (max),"
- " %.3f ms (min)\n",
- (exec_time * 1.0) / primary_count,
- (max_time_us + max_time_reverse_us) / 1000.0,
- (min_time_us + min_time_reverse_us) / 1000.0);
- } else {
- printf("Warning: no capture frames\n");
- }
- }
-
- AudioProcessing::Destroy(apm);
- apm = NULL;
-}
-} // namespace
-
-int main(int argc, char* argv[])
-{
- void_main(argc, argv);
-
- // Optional, but removes memory leak noise from Valgrind.
- google::protobuf::ShutdownProtobufLibrary();
- return 0;
-}
diff --git a/trunk/src/modules/audio_processing/test/unit_test.cc b/trunk/src/modules/audio_processing/test/unit_test.cc
deleted file mode 100644
index 4fa5d7c..0000000
--- a/trunk/src/modules/audio_processing/test/unit_test.cc
+++ /dev/null
@@ -1,1322 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-
-#include "gtest/gtest.h"
-
-#include "audio_processing.h"
-#include "event_wrapper.h"
-#include "module_common_types.h"
-#include "scoped_ptr.h"
-#include "signal_processing_library.h"
-#include "testsupport/fileutils.h"
-#include "thread_wrapper.h"
-#include "trace.h"
-#ifdef WEBRTC_ANDROID
-#include "external/webrtc/src/modules/audio_processing/test/unittest.pb.h"
-#else
-#include "webrtc/audio_processing/unittest.pb.h"
-#endif
-
-#if (defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)) || \
- (defined(WEBRTC_LINUX) && defined(WEBRTC_ARCH_X86_64) && !defined(NDEBUG))
-# define WEBRTC_AUDIOPROC_BIT_EXACT
-#endif
-
-using webrtc::AudioProcessing;
-using webrtc::AudioFrame;
-using webrtc::GainControl;
-using webrtc::NoiseSuppression;
-using webrtc::EchoCancellation;
-using webrtc::EventWrapper;
-using webrtc::scoped_array;
-using webrtc::Trace;
-using webrtc::LevelEstimator;
-using webrtc::EchoCancellation;
-using webrtc::EchoControlMobile;
-using webrtc::VoiceDetection;
-
-namespace {
-// When false, this will compare the output data with the results stored to
-// file. This is the typical case. When the file should be updated, it can
-// be set to true with the command-line switch --write_ref_data.
-bool write_ref_data = false;
-
-class ApmTest : public ::testing::Test {
- protected:
- ApmTest();
- virtual void SetUp();
- virtual void TearDown();
-
- static void SetUpTestCase() {
- Trace::CreateTrace();
- std::string trace_filename = webrtc::test::OutputPath() +
- "audioproc_trace.txt";
- ASSERT_EQ(0, Trace::SetTraceFile(trace_filename.c_str()));
- }
-
- static void TearDownTestCase() {
- Trace::ReturnTrace();
- }
-
- void Init(int sample_rate_hz, int num_reverse_channels,
- int num_input_channels, int num_output_channels,
- bool open_output_file);
- std::string ResourceFilePath(std::string name, int sample_rate_hz);
- std::string OutputFilePath(std::string name,
- int sample_rate_hz,
- int num_reverse_channels,
- int num_input_channels,
- int num_output_channels);
-
- const std::string output_path_;
- const std::string ref_path_;
- const std::string ref_filename_;
- webrtc::AudioProcessing* apm_;
- webrtc::AudioFrame* frame_;
- webrtc::AudioFrame* revframe_;
- FILE* far_file_;
- FILE* near_file_;
- FILE* out_file_;
-};
-
-ApmTest::ApmTest()
- : output_path_(webrtc::test::OutputPath()),
- ref_path_(webrtc::test::ProjectRootPath() +
- "test/data/audio_processing/"),
-#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
- ref_filename_(ref_path_ + "output_data_fixed.pb"),
-#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
- ref_filename_(ref_path_ + "output_data_float.pb"),
-#endif
- apm_(NULL),
- frame_(NULL),
- revframe_(NULL),
- far_file_(NULL),
- near_file_(NULL),
- out_file_(NULL) {}
-
-void ApmTest::SetUp() {
- apm_ = AudioProcessing::Create(0);
- ASSERT_TRUE(apm_ != NULL);
-
- frame_ = new AudioFrame();
- revframe_ = new AudioFrame();
-
- Init(32000, 2, 2, 2, false);
-}
-
-void ApmTest::TearDown() {
- if (frame_) {
- delete frame_;
- }
- frame_ = NULL;
-
- if (revframe_) {
- delete revframe_;
- }
- revframe_ = NULL;
-
- if (far_file_) {
- ASSERT_EQ(0, fclose(far_file_));
- }
- far_file_ = NULL;
-
- if (near_file_) {
- ASSERT_EQ(0, fclose(near_file_));
- }
- near_file_ = NULL;
-
- if (out_file_) {
- ASSERT_EQ(0, fclose(out_file_));
- }
- out_file_ = NULL;
-
- if (apm_ != NULL) {
- AudioProcessing::Destroy(apm_);
- }
- apm_ = NULL;
-}
-
-std::string ApmTest::ResourceFilePath(std::string name, int sample_rate_hz) {
- std::ostringstream ss;
- // Resource files are all stereo.
- ss << name << sample_rate_hz / 1000 << "_stereo";
- return webrtc::test::ResourcePath(ss.str(), "pcm");
-}
-
-std::string ApmTest::OutputFilePath(std::string name,
- int sample_rate_hz,
- int num_reverse_channels,
- int num_input_channels,
- int num_output_channels) {
- std::ostringstream ss;
- ss << name << sample_rate_hz / 1000 << "_" << num_reverse_channels << "r" <<
- num_input_channels << "i" << "_";
- if (num_output_channels == 1) {
- ss << "mono";
- } else if (num_output_channels == 2) {
- ss << "stereo";
- } else {
- assert(false);
- return "";
- }
- ss << ".pcm";
-
- return output_path_ + ss.str();
-}
-
-
-void ApmTest::Init(int sample_rate_hz, int num_reverse_channels,
- int num_input_channels, int num_output_channels,
- bool open_output_file) {
- ASSERT_EQ(apm_->kNoError, apm_->Initialize());
-
- // Handles error checking of the parameters as well. No need to repeat it.
- ASSERT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(sample_rate_hz));
- ASSERT_EQ(apm_->kNoError, apm_->set_num_channels(num_input_channels,
- num_output_channels));
- ASSERT_EQ(apm_->kNoError,
- apm_->set_num_reverse_channels(num_reverse_channels));
-
- // We always use 10 ms frames.
- const int samples_per_channel = sample_rate_hz / 100;
- frame_->_payloadDataLengthInSamples = samples_per_channel;
- frame_->_audioChannel = num_input_channels;
- frame_->_frequencyInHz = sample_rate_hz;
- revframe_->_payloadDataLengthInSamples = samples_per_channel;
- revframe_->_audioChannel = num_reverse_channels;
- revframe_->_frequencyInHz = sample_rate_hz;
-
- if (far_file_) {
- ASSERT_EQ(0, fclose(far_file_));
- }
- std::string filename = ResourceFilePath("far", sample_rate_hz);
- far_file_ = fopen(filename.c_str(), "rb");
- ASSERT_TRUE(far_file_ != NULL) << "Could not open file " <<
- filename << "\n";
-
- if (near_file_) {
- ASSERT_EQ(0, fclose(near_file_));
- }
- filename = ResourceFilePath("near", sample_rate_hz);
- near_file_ = fopen(filename.c_str(), "rb");
- ASSERT_TRUE(near_file_ != NULL) << "Could not open file " <<
- filename << "\n";
-
- if (open_output_file) {
- if (out_file_) {
- ASSERT_EQ(0, fclose(out_file_));
- }
- filename = OutputFilePath("out", sample_rate_hz, num_reverse_channels,
- num_input_channels, num_output_channels);
- out_file_ = fopen(filename.c_str(), "wb");
- ASSERT_TRUE(out_file_ != NULL) << "Could not open file " <<
- filename << "\n";
- }
-}
-
-void MixStereoToMono(const int16_t* stereo,
- int16_t* mono,
- int samples_per_channel) {
- for (int i = 0; i < samples_per_channel; i++) {
- int32_t int32 = (static_cast<int32_t>(stereo[i * 2]) +
- static_cast<int32_t>(stereo[i * 2 + 1])) >> 1;
- mono[i] = static_cast<int16_t>(int32);
- }
-}
-
-template <class T>
-T MaxValue(T a, T b) {
- return a > b ? a : b;
-}
-
-template <class T>
-T AbsValue(T a) {
- return a > 0 ? a : -a;
-}
-
-void SetFrameTo(AudioFrame* frame, int16_t value) {
- for (int i = 0; i < frame->_payloadDataLengthInSamples * frame->_audioChannel;
- ++i) {
- frame->_payloadData[i] = value;
- }
-}
-
-int16_t MaxAudioFrame(const AudioFrame& frame) {
- const int length = frame._payloadDataLengthInSamples * frame._audioChannel;
- int16_t max = AbsValue(frame._payloadData[0]);
- for (int i = 1; i < length; i++) {
- max = MaxValue(max, AbsValue(frame._payloadData[i]));
- }
-
- return max;
-}
-
-bool FrameDataAreEqual(const AudioFrame& frame1, const AudioFrame& frame2) {
- if (frame1._payloadDataLengthInSamples !=
- frame2._payloadDataLengthInSamples) {
- return false;
- }
- if (frame1._audioChannel !=
- frame2._audioChannel) {
- return false;
- }
- if (memcmp(frame1._payloadData, frame2._payloadData,
- frame1._payloadDataLengthInSamples * frame1._audioChannel *
- sizeof(int16_t))) {
- return false;
- }
- return true;
-}
-
-void TestStats(const AudioProcessing::Statistic& test,
- const webrtc::audioproc::Test::Statistic& reference) {
- EXPECT_EQ(reference.instant(), test.instant);
- EXPECT_EQ(reference.average(), test.average);
- EXPECT_EQ(reference.maximum(), test.maximum);
- EXPECT_EQ(reference.minimum(), test.minimum);
-}
-
-void WriteStatsMessage(const AudioProcessing::Statistic& output,
- webrtc::audioproc::Test::Statistic* message) {
- message->set_instant(output.instant);
- message->set_average(output.average);
- message->set_maximum(output.maximum);
- message->set_minimum(output.minimum);
-}
-
-void WriteMessageLiteToFile(const std::string filename,
- const ::google::protobuf::MessageLite& message) {
- FILE* file = fopen(filename.c_str(), "wb");
- ASSERT_TRUE(file != NULL) << "Could not open " << filename;
- int size = message.ByteSize();
- ASSERT_GT(size, 0);
- unsigned char* array = new unsigned char[size];
- ASSERT_TRUE(message.SerializeToArray(array, size));
-
- ASSERT_EQ(1u, fwrite(&size, sizeof(int), 1, file));
- ASSERT_EQ(static_cast<size_t>(size),
- fwrite(array, sizeof(unsigned char), size, file));
-
- delete [] array;
- fclose(file);
-}
-
-void ReadMessageLiteFromFile(const std::string filename,
- ::google::protobuf::MessageLite* message) {
- assert(message != NULL);
-
- FILE* file = fopen(filename.c_str(), "rb");
- ASSERT_TRUE(file != NULL) << "Could not open " << filename;
- int size = 0;
- ASSERT_EQ(1u, fread(&size, sizeof(int), 1, file));
- ASSERT_GT(size, 0);
- unsigned char* array = new unsigned char[size];
- ASSERT_EQ(static_cast<size_t>(size),
- fread(array, sizeof(unsigned char), size, file));
-
- ASSERT_TRUE(message->ParseFromArray(array, size));
-
- delete [] array;
- fclose(file);
-}
-
-struct ThreadData {
- ThreadData(int thread_num_, AudioProcessing* ap_)
- : thread_num(thread_num_),
- error(false),
- ap(ap_) {}
- int thread_num;
- bool error;
- AudioProcessing* ap;
-};
-
-// Don't use GTest here; non-thread-safe on Windows (as of 1.5.0).
-bool DeadlockProc(void* thread_object) {
- ThreadData* thread_data = static_cast<ThreadData*>(thread_object);
- AudioProcessing* ap = thread_data->ap;
- int err = ap->kNoError;
-
- AudioFrame primary_frame;
- AudioFrame reverse_frame;
- primary_frame._payloadDataLengthInSamples = 320;
- primary_frame._audioChannel = 2;
- primary_frame._frequencyInHz = 32000;
- reverse_frame._payloadDataLengthInSamples = 320;
- reverse_frame._audioChannel = 2;
- reverse_frame._frequencyInHz = 32000;
-
- ap->echo_cancellation()->Enable(true);
- ap->gain_control()->Enable(true);
- ap->high_pass_filter()->Enable(true);
- ap->level_estimator()->Enable(true);
- ap->noise_suppression()->Enable(true);
- ap->voice_detection()->Enable(true);
-
- if (thread_data->thread_num % 2 == 0) {
- err = ap->AnalyzeReverseStream(&reverse_frame);
- if (err != ap->kNoError) {
- printf("Error in AnalyzeReverseStream(): %d\n", err);
- thread_data->error = true;
- return false;
- }
- }
-
- if (thread_data->thread_num % 2 == 1) {
- ap->set_stream_delay_ms(0);
- ap->echo_cancellation()->set_stream_drift_samples(0);
- ap->gain_control()->set_stream_analog_level(0);
- err = ap->ProcessStream(&primary_frame);
- if (err == ap->kStreamParameterNotSetError) {
- printf("Expected kStreamParameterNotSetError in ProcessStream(): %d\n",
- err);
- } else if (err != ap->kNoError) {
- printf("Error in ProcessStream(): %d\n", err);
- thread_data->error = true;
- return false;
- }
- ap->gain_control()->stream_analog_level();
- }
-
- EventWrapper* event = EventWrapper::Create();
- event->Wait(1);
- delete event;
- event = NULL;
-
- return true;
-}
-
-/*TEST_F(ApmTest, Deadlock) {
- const int num_threads = 16;
- std::vector<ThreadWrapper*> threads(num_threads);
- std::vector<ThreadData*> thread_data(num_threads);
-
- ASSERT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
- ASSERT_EQ(apm_->kNoError, apm_->set_num_channels(2, 2));
- ASSERT_EQ(apm_->kNoError, apm_->set_num_reverse_channels(2));
-
- for (int i = 0; i < num_threads; i++) {
- thread_data[i] = new ThreadData(i, apm_);
- threads[i] = ThreadWrapper::CreateThread(DeadlockProc,
- thread_data[i],
- kNormalPriority,
- 0);
- ASSERT_TRUE(threads[i] != NULL);
- unsigned int thread_id = 0;
- threads[i]->Start(thread_id);
- }
-
- EventWrapper* event = EventWrapper::Create();
- ASSERT_EQ(kEventTimeout, event->Wait(5000));
- delete event;
- event = NULL;
-
- for (int i = 0; i < num_threads; i++) {
- // This will return false if the thread has deadlocked.
- ASSERT_TRUE(threads[i]->Stop());
- ASSERT_FALSE(thread_data[i]->error);
- delete threads[i];
- threads[i] = NULL;
- delete thread_data[i];
- thread_data[i] = NULL;
- }
-}*/
-
-TEST_F(ApmTest, StreamParameters) {
- // No errors when the components are disabled.
- EXPECT_EQ(apm_->kNoError,
- apm_->ProcessStream(frame_));
-
- // -- Missing AGC level --
- EXPECT_EQ(apm_->kNoError, apm_->Initialize());
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
- EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
-
- // Resets after successful ProcessStream().
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_stream_analog_level(127));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
-
- // Other stream parameters set correctly.
- EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_drift_compensation(true));
- EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->set_stream_drift_samples(0));
- EXPECT_EQ(apm_->kStreamParameterNotSetError,
- apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_drift_compensation(false));
-
- // -- Missing delay --
- EXPECT_EQ(apm_->kNoError, apm_->Initialize());
- EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
- EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
-
- // Resets after successful ProcessStream().
- EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
-
- // Other stream parameters set correctly.
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_drift_compensation(true));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->set_stream_drift_samples(0));
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_stream_analog_level(127));
- EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
-
- // -- Missing drift --
- EXPECT_EQ(apm_->kNoError, apm_->Initialize());
- EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
-
- // Resets after successful ProcessStream().
- EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->set_stream_drift_samples(0));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
-
- // Other stream parameters set correctly.
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
- EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_stream_analog_level(127));
- EXPECT_EQ(apm_->kStreamParameterNotSetError, apm_->ProcessStream(frame_));
-
- // -- No stream parameters --
- EXPECT_EQ(apm_->kNoError, apm_->Initialize());
- EXPECT_EQ(apm_->kNoError,
- apm_->AnalyzeReverseStream(revframe_));
- EXPECT_EQ(apm_->kStreamParameterNotSetError,
- apm_->ProcessStream(frame_));
-
- // -- All there --
- EXPECT_EQ(apm_->kNoError, apm_->Initialize());
- EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(100));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->set_stream_drift_samples(0));
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_stream_analog_level(127));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
-}
-
-TEST_F(ApmTest, Channels) {
- // Testing number of invalid channels
- EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(0, 1));
- EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(1, 0));
- EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(3, 1));
- EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(1, 3));
- EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_reverse_channels(0));
- EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_reverse_channels(3));
- // Testing number of valid channels
- for (int i = 1; i < 3; i++) {
- for (int j = 1; j < 3; j++) {
- if (j > i) {
- EXPECT_EQ(apm_->kBadParameterError, apm_->set_num_channels(i, j));
- } else {
- EXPECT_EQ(apm_->kNoError, apm_->set_num_channels(i, j));
- EXPECT_EQ(j, apm_->num_output_channels());
- }
- }
- EXPECT_EQ(i, apm_->num_input_channels());
- EXPECT_EQ(apm_->kNoError, apm_->set_num_reverse_channels(i));
- EXPECT_EQ(i, apm_->num_reverse_channels());
- }
-}
-
-TEST_F(ApmTest, SampleRates) {
- // Testing invalid sample rates
- EXPECT_EQ(apm_->kBadParameterError, apm_->set_sample_rate_hz(10000));
- // Testing valid sample rates
- int fs[] = {8000, 16000, 32000};
- for (size_t i = 0; i < sizeof(fs) / sizeof(*fs); i++) {
- EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(fs[i]));
- EXPECT_EQ(fs[i], apm_->sample_rate_hz());
- }
-}
-
-
-TEST_F(ApmTest, EchoCancellation) {
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_drift_compensation(true));
- EXPECT_TRUE(apm_->echo_cancellation()->is_drift_compensation_enabled());
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_drift_compensation(false));
- EXPECT_FALSE(apm_->echo_cancellation()->is_drift_compensation_enabled());
-
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->echo_cancellation()->set_device_sample_rate_hz(4000));
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->echo_cancellation()->set_device_sample_rate_hz(100000));
-
- int rate[] = {16000, 44100, 48000};
- for (size_t i = 0; i < sizeof(rate)/sizeof(*rate); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->set_device_sample_rate_hz(rate[i]));
- EXPECT_EQ(rate[i],
- apm_->echo_cancellation()->device_sample_rate_hz());
- }
-
- EchoCancellation::SuppressionLevel level[] = {
- EchoCancellation::kLowSuppression,
- EchoCancellation::kModerateSuppression,
- EchoCancellation::kHighSuppression,
- };
- for (size_t i = 0; i < sizeof(level)/sizeof(*level); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->set_suppression_level(level[i]));
- EXPECT_EQ(level[i],
- apm_->echo_cancellation()->suppression_level());
- }
-
- EchoCancellation::Metrics metrics;
- EXPECT_EQ(apm_->kNotEnabledError,
- apm_->echo_cancellation()->GetMetrics(&metrics));
-
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_metrics(true));
- EXPECT_TRUE(apm_->echo_cancellation()->are_metrics_enabled());
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_metrics(false));
- EXPECT_FALSE(apm_->echo_cancellation()->are_metrics_enabled());
-
- int median = 0;
- int std = 0;
- EXPECT_EQ(apm_->kNotEnabledError,
- apm_->echo_cancellation()->GetDelayMetrics(&median, &std));
-
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_delay_logging(true));
- EXPECT_TRUE(apm_->echo_cancellation()->is_delay_logging_enabled());
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_delay_logging(false));
- EXPECT_FALSE(apm_->echo_cancellation()->is_delay_logging_enabled());
-
- EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
- EXPECT_TRUE(apm_->echo_cancellation()->is_enabled());
- EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(false));
- EXPECT_FALSE(apm_->echo_cancellation()->is_enabled());
-}
-
-TEST_F(ApmTest, EchoControlMobile) {
- // AECM won't use super-wideband.
- EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
- EXPECT_EQ(apm_->kBadSampleRateError, apm_->echo_control_mobile()->Enable(true));
- // Turn AECM on (and AEC off)
- Init(16000, 2, 2, 2, false);
- EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(true));
- EXPECT_TRUE(apm_->echo_control_mobile()->is_enabled());
-
- // Toggle routing modes
- EchoControlMobile::RoutingMode mode[] = {
- EchoControlMobile::kQuietEarpieceOrHeadset,
- EchoControlMobile::kEarpiece,
- EchoControlMobile::kLoudEarpiece,
- EchoControlMobile::kSpeakerphone,
- EchoControlMobile::kLoudSpeakerphone,
- };
- for (size_t i = 0; i < sizeof(mode)/sizeof(*mode); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_control_mobile()->set_routing_mode(mode[i]));
- EXPECT_EQ(mode[i],
- apm_->echo_control_mobile()->routing_mode());
- }
- // Turn comfort noise off/on
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_control_mobile()->enable_comfort_noise(false));
- EXPECT_FALSE(apm_->echo_control_mobile()->is_comfort_noise_enabled());
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_control_mobile()->enable_comfort_noise(true));
- EXPECT_TRUE(apm_->echo_control_mobile()->is_comfort_noise_enabled());
- // Set and get echo path
- const size_t echo_path_size =
- apm_->echo_control_mobile()->echo_path_size_bytes();
- scoped_array<char> echo_path_in(new char[echo_path_size]);
- scoped_array<char> echo_path_out(new char[echo_path_size]);
- EXPECT_EQ(apm_->kNullPointerError,
- apm_->echo_control_mobile()->SetEchoPath(NULL, echo_path_size));
- EXPECT_EQ(apm_->kNullPointerError,
- apm_->echo_control_mobile()->GetEchoPath(NULL, echo_path_size));
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->echo_control_mobile()->GetEchoPath(echo_path_out.get(), 1));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_control_mobile()->GetEchoPath(echo_path_out.get(),
- echo_path_size));
- for (size_t i = 0; i < echo_path_size; i++) {
- echo_path_in[i] = echo_path_out[i] + 1;
- }
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->echo_control_mobile()->SetEchoPath(echo_path_in.get(), 1));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_control_mobile()->SetEchoPath(echo_path_in.get(),
- echo_path_size));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_control_mobile()->GetEchoPath(echo_path_out.get(),
- echo_path_size));
- for (size_t i = 0; i < echo_path_size; i++) {
- EXPECT_EQ(echo_path_in[i], echo_path_out[i]);
- }
-
- // Process a few frames with NS in the default disabled state. This exercises
- // a different codepath than with it enabled.
- EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
-
- // Turn AECM off
- EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(false));
- EXPECT_FALSE(apm_->echo_control_mobile()->is_enabled());
-}
-
-TEST_F(ApmTest, GainControl) {
- // Testing gain modes
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_mode(
- apm_->gain_control()->mode()));
-
- GainControl::Mode mode[] = {
- GainControl::kAdaptiveAnalog,
- GainControl::kAdaptiveDigital,
- GainControl::kFixedDigital
- };
- for (size_t i = 0; i < sizeof(mode)/sizeof(*mode); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_mode(mode[i]));
- EXPECT_EQ(mode[i], apm_->gain_control()->mode());
- }
- // Testing invalid target levels
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_target_level_dbfs(-3));
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_target_level_dbfs(-40));
- // Testing valid target levels
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_target_level_dbfs(
- apm_->gain_control()->target_level_dbfs()));
-
- int level_dbfs[] = {0, 6, 31};
- for (size_t i = 0; i < sizeof(level_dbfs)/sizeof(*level_dbfs); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_target_level_dbfs(level_dbfs[i]));
- EXPECT_EQ(level_dbfs[i], apm_->gain_control()->target_level_dbfs());
- }
-
- // Testing invalid compression gains
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_compression_gain_db(-1));
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_compression_gain_db(100));
-
- // Testing valid compression gains
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_compression_gain_db(
- apm_->gain_control()->compression_gain_db()));
-
- int gain_db[] = {0, 10, 90};
- for (size_t i = 0; i < sizeof(gain_db)/sizeof(*gain_db); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_compression_gain_db(gain_db[i]));
- EXPECT_EQ(gain_db[i], apm_->gain_control()->compression_gain_db());
- }
-
- // Testing limiter off/on
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->enable_limiter(false));
- EXPECT_FALSE(apm_->gain_control()->is_limiter_enabled());
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->enable_limiter(true));
- EXPECT_TRUE(apm_->gain_control()->is_limiter_enabled());
-
- // Testing invalid level limits
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_analog_level_limits(-1, 512));
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_analog_level_limits(100000, 512));
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_analog_level_limits(512, -1));
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_analog_level_limits(512, 100000));
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->gain_control()->set_analog_level_limits(512, 255));
-
- // Testing valid level limits
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_analog_level_limits(
- apm_->gain_control()->analog_level_minimum(),
- apm_->gain_control()->analog_level_maximum()));
-
- int min_level[] = {0, 255, 1024};
- for (size_t i = 0; i < sizeof(min_level)/sizeof(*min_level); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_analog_level_limits(min_level[i], 1024));
- EXPECT_EQ(min_level[i], apm_->gain_control()->analog_level_minimum());
- }
-
- int max_level[] = {0, 1024, 65535};
- for (size_t i = 0; i < sizeof(min_level)/sizeof(*min_level); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_analog_level_limits(0, max_level[i]));
- EXPECT_EQ(max_level[i], apm_->gain_control()->analog_level_maximum());
- }
-
- // TODO(ajm): stream_is_saturated() and stream_analog_level()
-
- // Turn AGC off
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(false));
- EXPECT_FALSE(apm_->gain_control()->is_enabled());
-}
-
-TEST_F(ApmTest, NoiseSuppression) {
- // Test valid suppression levels.
- NoiseSuppression::Level level[] = {
- NoiseSuppression::kLow,
- NoiseSuppression::kModerate,
- NoiseSuppression::kHigh,
- NoiseSuppression::kVeryHigh
- };
- for (size_t i = 0; i < sizeof(level)/sizeof(*level); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->noise_suppression()->set_level(level[i]));
- EXPECT_EQ(level[i], apm_->noise_suppression()->level());
- }
-
- // Turn NS on/off
- EXPECT_EQ(apm_->kNoError, apm_->noise_suppression()->Enable(true));
- EXPECT_TRUE(apm_->noise_suppression()->is_enabled());
- EXPECT_EQ(apm_->kNoError, apm_->noise_suppression()->Enable(false));
- EXPECT_FALSE(apm_->noise_suppression()->is_enabled());
-}
-
-TEST_F(ApmTest, HighPassFilter) {
- // Turn HP filter on/off
- EXPECT_EQ(apm_->kNoError, apm_->high_pass_filter()->Enable(true));
- EXPECT_TRUE(apm_->high_pass_filter()->is_enabled());
- EXPECT_EQ(apm_->kNoError, apm_->high_pass_filter()->Enable(false));
- EXPECT_FALSE(apm_->high_pass_filter()->is_enabled());
-}
-
-TEST_F(ApmTest, LevelEstimator) {
- // Turn level estimator on/off
- EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(false));
- EXPECT_FALSE(apm_->level_estimator()->is_enabled());
-
- EXPECT_EQ(apm_->kNotEnabledError, apm_->level_estimator()->RMS());
-
- EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
- EXPECT_TRUE(apm_->level_estimator()->is_enabled());
-
- // Run this test in wideband; in super-wb, the splitting filter distorts the
- // audio enough to cause deviation from the expectation for small values.
- EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(16000));
- frame_->_payloadDataLengthInSamples = 160;
- frame_->_audioChannel = 2;
- frame_->_frequencyInHz = 16000;
-
- // Min value if no frames have been processed.
- EXPECT_EQ(127, apm_->level_estimator()->RMS());
-
- // Min value on zero frames.
- SetFrameTo(frame_, 0);
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(127, apm_->level_estimator()->RMS());
-
- // Try a few RMS values.
- // (These also test that the value resets after retrieving it.)
- SetFrameTo(frame_, 32767);
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(0, apm_->level_estimator()->RMS());
-
- SetFrameTo(frame_, 30000);
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(1, apm_->level_estimator()->RMS());
-
- SetFrameTo(frame_, 10000);
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(10, apm_->level_estimator()->RMS());
-
- SetFrameTo(frame_, 10);
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(70, apm_->level_estimator()->RMS());
-
- // Min value if _energy == 0.
- SetFrameTo(frame_, 10000);
- uint32_t energy = frame_->_energy; // Save default to restore below.
- frame_->_energy = 0;
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(127, apm_->level_estimator()->RMS());
- frame_->_energy = energy;
-
- // Verify reset after enable/disable.
- SetFrameTo(frame_, 32767);
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(false));
- EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
- SetFrameTo(frame_, 1);
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(90, apm_->level_estimator()->RMS());
-
- // Verify reset after initialize.
- SetFrameTo(frame_, 32767);
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->Initialize());
- SetFrameTo(frame_, 1);
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(90, apm_->level_estimator()->RMS());
-}
-
-TEST_F(ApmTest, VoiceDetection) {
- // Test external VAD
- EXPECT_EQ(apm_->kNoError,
- apm_->voice_detection()->set_stream_has_voice(true));
- EXPECT_TRUE(apm_->voice_detection()->stream_has_voice());
- EXPECT_EQ(apm_->kNoError,
- apm_->voice_detection()->set_stream_has_voice(false));
- EXPECT_FALSE(apm_->voice_detection()->stream_has_voice());
-
- // Test valid likelihoods
- VoiceDetection::Likelihood likelihood[] = {
- VoiceDetection::kVeryLowLikelihood,
- VoiceDetection::kLowLikelihood,
- VoiceDetection::kModerateLikelihood,
- VoiceDetection::kHighLikelihood
- };
- for (size_t i = 0; i < sizeof(likelihood)/sizeof(*likelihood); i++) {
- EXPECT_EQ(apm_->kNoError,
- apm_->voice_detection()->set_likelihood(likelihood[i]));
- EXPECT_EQ(likelihood[i], apm_->voice_detection()->likelihood());
- }
-
- /* TODO(bjornv): Enable once VAD supports other frame lengths than 10 ms
- // Test invalid frame sizes
- EXPECT_EQ(apm_->kBadParameterError,
- apm_->voice_detection()->set_frame_size_ms(12));
-
- // Test valid frame sizes
- for (int i = 10; i <= 30; i += 10) {
- EXPECT_EQ(apm_->kNoError,
- apm_->voice_detection()->set_frame_size_ms(i));
- EXPECT_EQ(i, apm_->voice_detection()->frame_size_ms());
- }
- */
-
- // Turn VAD on/off
- EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
- EXPECT_TRUE(apm_->voice_detection()->is_enabled());
- EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
- EXPECT_FALSE(apm_->voice_detection()->is_enabled());
-
- // Test that AudioFrame activity is maintained when VAD is disabled.
- EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
- AudioFrame::VADActivity activity[] = {
- AudioFrame::kVadActive,
- AudioFrame::kVadPassive,
- AudioFrame::kVadUnknown
- };
- for (size_t i = 0; i < sizeof(activity)/sizeof(*activity); i++) {
- frame_->_vadActivity = activity[i];
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(activity[i], frame_->_vadActivity);
- }
-
- // Test that AudioFrame activity is set when VAD is enabled.
- EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
- frame_->_vadActivity = AudioFrame::kVadUnknown;
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_NE(AudioFrame::kVadUnknown, frame_->_vadActivity);
-
- // TODO(bjornv): Add tests for streamed voice; stream_has_voice()
-}
-
-TEST_F(ApmTest, SplittingFilter) {
- // Verify the filter is not active through undistorted audio when:
- // 1. No components are enabled...
- SetFrameTo(frame_, 1000);
- AudioFrame frame_copy = *frame_;
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
-
- // 2. Only the level estimator is enabled...
- SetFrameTo(frame_, 1000);
- frame_copy = *frame_;
- EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
- EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(false));
-
- // 3. Only VAD is enabled...
- SetFrameTo(frame_, 1000);
- frame_copy = *frame_;
- EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
- EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
-
- // 4. Both VAD and the level estimator are enabled...
- SetFrameTo(frame_, 1000);
- frame_copy = *frame_;
- EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(true));
- EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(true));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
- EXPECT_EQ(apm_->kNoError, apm_->level_estimator()->Enable(false));
- EXPECT_EQ(apm_->kNoError, apm_->voice_detection()->Enable(false));
-
- // 5. Not using super-wb.
- EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(16000));
- frame_->_payloadDataLengthInSamples = 160;
- frame_->_audioChannel = 2;
- frame_->_frequencyInHz = 16000;
- // Enable AEC, which would require the filter in super-wb. We rely on the
- // first few frames of data being unaffected by the AEC.
- // TODO(andrew): This test, and the one below, rely rather tenuously on the
- // behavior of the AEC. Think of something more robust.
- EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
- SetFrameTo(frame_, 1000);
- frame_copy = *frame_;
- EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->set_stream_drift_samples(0));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->set_stream_drift_samples(0));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_TRUE(FrameDataAreEqual(*frame_, frame_copy));
-
- // Check the test is valid. We should have distortion from the filter
- // when AEC is enabled (which won't affect the audio).
- EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(32000));
- frame_->_payloadDataLengthInSamples = 320;
- frame_->_audioChannel = 2;
- frame_->_frequencyInHz = 32000;
- SetFrameTo(frame_, 1000);
- frame_copy = *frame_;
- EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->set_stream_drift_samples(0));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_FALSE(FrameDataAreEqual(*frame_, frame_copy));
-}
-
-// TODO(andrew): expand test to verify output.
-TEST_F(ApmTest, DebugDump) {
- const std::string filename = webrtc::test::OutputPath() + "debug.aec";
- EXPECT_EQ(apm_->kNullPointerError, apm_->StartDebugRecording(NULL));
-
-#ifdef WEBRTC_AUDIOPROC_DEBUG_DUMP
- // Stopping without having started should be OK.
- EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
-
- EXPECT_EQ(apm_->kNoError, apm_->StartDebugRecording(filename.c_str()));
- EXPECT_EQ(apm_->kNoError, apm_->AnalyzeReverseStream(revframe_));
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- EXPECT_EQ(apm_->kNoError, apm_->StopDebugRecording());
-
- // Verify the file has been written.
- FILE* fid = fopen(filename.c_str(), "r");
- ASSERT_TRUE(fid != NULL);
-
- // Clean it up.
- ASSERT_EQ(0, fclose(fid));
- ASSERT_EQ(0, remove(filename.c_str()));
-#else
- EXPECT_EQ(apm_->kUnsupportedFunctionError,
- apm_->StartDebugRecording(filename.c_str()));
- EXPECT_EQ(apm_->kUnsupportedFunctionError, apm_->StopDebugRecording());
-
- // Verify the file has NOT been written.
- ASSERT_TRUE(fopen(filename.c_str(), "r") == NULL);
-#endif // WEBRTC_AUDIOPROC_DEBUG_DUMP
-}
-
-// TODO(andrew): Add a test to process a few frames with different combinations
-// of enabled components.
-
-// TODO(andrew): Make this test more robust such that it can be run on multiple
-// platforms. It currently requires bit-exactness.
-#ifdef WEBRTC_AUDIOPROC_BIT_EXACT
-TEST_F(ApmTest, Process) {
- GOOGLE_PROTOBUF_VERIFY_VERSION;
- webrtc::audioproc::OutputData ref_data;
-
- if (!write_ref_data) {
- ReadMessageLiteFromFile(ref_filename_, &ref_data);
- } else {
- // Write the desired tests to the protobuf reference file.
- const int channels[] = {1, 2};
- const size_t channels_size = sizeof(channels) / sizeof(*channels);
-#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
- // AECM doesn't support super-wb.
- const int sample_rates[] = {8000, 16000};
-#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
- const int sample_rates[] = {8000, 16000, 32000};
-#endif
- const size_t sample_rates_size = sizeof(sample_rates) / sizeof(*sample_rates);
- for (size_t i = 0; i < channels_size; i++) {
- for (size_t j = 0; j < channels_size; j++) {
- // We can't have more output than input channels.
- for (size_t k = 0; k <= j; k++) {
- for (size_t l = 0; l < sample_rates_size; l++) {
- webrtc::audioproc::Test* test = ref_data.add_test();
- test->set_num_reverse_channels(channels[i]);
- test->set_num_input_channels(channels[j]);
- test->set_num_output_channels(channels[k]);
- test->set_sample_rate(sample_rates[l]);
- }
- }
- }
- }
- }
-
-#if defined(WEBRTC_AUDIOPROC_FIXED_PROFILE)
- EXPECT_EQ(apm_->kNoError, apm_->set_sample_rate_hz(16000));
- EXPECT_EQ(apm_->kNoError, apm_->echo_control_mobile()->Enable(true));
-
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_mode(GainControl::kAdaptiveDigital));
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
-#elif defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_drift_compensation(true));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_metrics(true));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->enable_delay_logging(true));
- EXPECT_EQ(apm_->kNoError, apm_->echo_cancellation()->Enable(true));
-
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_mode(GainControl::kAdaptiveAnalog));
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_analog_level_limits(0, 255));
- EXPECT_EQ(apm_->kNoError, apm_->gain_control()->Enable(true));
-#endif
-
- EXPECT_EQ(apm_->kNoError,
- apm_->high_pass_filter()->Enable(true));
-
- EXPECT_EQ(apm_->kNoError,
- apm_->level_estimator()->Enable(true));
-
- EXPECT_EQ(apm_->kNoError,
- apm_->noise_suppression()->Enable(true));
-
- EXPECT_EQ(apm_->kNoError,
- apm_->voice_detection()->Enable(true));
-
- for (int i = 0; i < ref_data.test_size(); i++) {
- printf("Running test %d of %d...\n", i + 1, ref_data.test_size());
-
- webrtc::audioproc::Test* test = ref_data.mutable_test(i);
- Init(test->sample_rate(), test->num_reverse_channels(),
- test->num_input_channels(), test->num_output_channels(), true);
-
- const int samples_per_channel = test->sample_rate() / 100;
- int frame_count = 0;
- int has_echo_count = 0;
- int has_voice_count = 0;
- int is_saturated_count = 0;
- int analog_level = 127;
- int analog_level_average = 0;
- int max_output_average = 0;
-
- while (1) {
- // Read far-end frame, always stereo.
- size_t frame_size = samples_per_channel * 2;
- size_t read_count = fread(revframe_->_payloadData,
- sizeof(int16_t),
- frame_size,
- far_file_);
- if (read_count != frame_size) {
- // Check that the file really ended.
- ASSERT_NE(0, feof(far_file_));
- break; // This is expected.
- }
-
- if (revframe_->_audioChannel == 1) {
- MixStereoToMono(revframe_->_payloadData, revframe_->_payloadData,
- samples_per_channel);
- }
-
- EXPECT_EQ(apm_->kNoError, apm_->AnalyzeReverseStream(revframe_));
-
- EXPECT_EQ(apm_->kNoError, apm_->set_stream_delay_ms(0));
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->set_stream_drift_samples(0));
- EXPECT_EQ(apm_->kNoError,
- apm_->gain_control()->set_stream_analog_level(analog_level));
-
- // Read near-end frame, always stereo.
- read_count = fread(frame_->_payloadData,
- sizeof(int16_t),
- frame_size,
- near_file_);
- if (read_count != frame_size) {
- // Check that the file really ended.
- ASSERT_NE(0, feof(near_file_));
- break; // This is expected.
- }
-
- if (frame_->_audioChannel == 1) {
- MixStereoToMono(frame_->_payloadData, frame_->_payloadData,
- samples_per_channel);
- }
- frame_->_vadActivity = AudioFrame::kVadUnknown;
-
- EXPECT_EQ(apm_->kNoError, apm_->ProcessStream(frame_));
- // Ensure the frame was downmixed properly.
- EXPECT_EQ(test->num_output_channels(), frame_->_audioChannel);
-
- max_output_average += MaxAudioFrame(*frame_);
-
- if (apm_->echo_cancellation()->stream_has_echo()) {
- has_echo_count++;
- }
-
- analog_level = apm_->gain_control()->stream_analog_level();
- analog_level_average += analog_level;
- if (apm_->gain_control()->stream_is_saturated()) {
- is_saturated_count++;
- }
- if (apm_->voice_detection()->stream_has_voice()) {
- has_voice_count++;
- EXPECT_EQ(AudioFrame::kVadActive, frame_->_vadActivity);
- } else {
- EXPECT_EQ(AudioFrame::kVadPassive, frame_->_vadActivity);
- }
-
- frame_size = samples_per_channel * frame_->_audioChannel;
- size_t write_count = fwrite(frame_->_payloadData,
- sizeof(int16_t),
- frame_size,
- out_file_);
- ASSERT_EQ(frame_size, write_count);
-
- // Reset in case of downmixing.
- frame_->_audioChannel = test->num_input_channels();
- frame_count++;
- }
- max_output_average /= frame_count;
- analog_level_average /= frame_count;
-
-#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
- EchoCancellation::Metrics echo_metrics;
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->GetMetrics(&echo_metrics));
- int median = 0;
- int std = 0;
- EXPECT_EQ(apm_->kNoError,
- apm_->echo_cancellation()->GetDelayMetrics(&median, &std));
-
- int rms_level = apm_->level_estimator()->RMS();
- EXPECT_LE(0, rms_level);
- EXPECT_GE(127, rms_level);
-#endif
-
- if (!write_ref_data) {
- EXPECT_EQ(test->has_echo_count(), has_echo_count);
- EXPECT_EQ(test->has_voice_count(), has_voice_count);
- EXPECT_EQ(test->is_saturated_count(), is_saturated_count);
-
- EXPECT_EQ(test->analog_level_average(), analog_level_average);
- EXPECT_EQ(test->max_output_average(), max_output_average);
-
-#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
- webrtc::audioproc::Test::EchoMetrics reference =
- test->echo_metrics();
- TestStats(echo_metrics.residual_echo_return_loss,
- reference.residual_echo_return_loss());
- TestStats(echo_metrics.echo_return_loss,
- reference.echo_return_loss());
- TestStats(echo_metrics.echo_return_loss_enhancement,
- reference.echo_return_loss_enhancement());
- TestStats(echo_metrics.a_nlp,
- reference.a_nlp());
-
- webrtc::audioproc::Test::DelayMetrics reference_delay =
- test->delay_metrics();
- EXPECT_EQ(reference_delay.median(), median);
- EXPECT_EQ(reference_delay.std(), std);
-
- EXPECT_EQ(test->rms_level(), rms_level);
-#endif
- } else {
- test->set_has_echo_count(has_echo_count);
- test->set_has_voice_count(has_voice_count);
- test->set_is_saturated_count(is_saturated_count);
-
- test->set_analog_level_average(analog_level_average);
- test->set_max_output_average(max_output_average);
-
-#if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
- webrtc::audioproc::Test::EchoMetrics* message =
- test->mutable_echo_metrics();
- WriteStatsMessage(echo_metrics.residual_echo_return_loss,
- message->mutable_residual_echo_return_loss());
- WriteStatsMessage(echo_metrics.echo_return_loss,
- message->mutable_echo_return_loss());
- WriteStatsMessage(echo_metrics.echo_return_loss_enhancement,
- message->mutable_echo_return_loss_enhancement());
- WriteStatsMessage(echo_metrics.a_nlp,
- message->mutable_a_nlp());
-
- webrtc::audioproc::Test::DelayMetrics* message_delay =
- test->mutable_delay_metrics();
- message_delay->set_median(median);
- message_delay->set_std(std);
-
- test->set_rms_level(rms_level);
-#endif
- }
-
- rewind(far_file_);
- rewind(near_file_);
- }
-
- if (write_ref_data) {
- WriteMessageLiteToFile(ref_filename_, ref_data);
- }
-}
-#endif // WEBRTC_AUDIOPROC_BIT_EXACT
-
-} // namespace
-
-int main(int argc, char** argv) {
- ::testing::InitGoogleTest(&argc, argv);
-
- for (int i = 1; i < argc; i++) {
- if (strcmp(argv[i], "--write_ref_data") == 0) {
- write_ref_data = true;
- }
- }
-
- int err = RUN_ALL_TESTS();
-
- // Optional, but removes memory leak noise from Valgrind.
- google::protobuf::ShutdownProtobufLibrary();
- return err;
-}
diff --git a/trunk/src/modules/audio_processing/test/unittest.proto b/trunk/src/modules/audio_processing/test/unittest.proto
deleted file mode 100644
index 67ba722..0000000
--- a/trunk/src/modules/audio_processing/test/unittest.proto
+++ /dev/null
@@ -1,52 +0,0 @@
-syntax = "proto2";
-option optimize_for = LITE_RUNTIME;
-package webrtc.audioproc;
-
-message Test {
- optional int32 num_reverse_channels = 1;
- optional int32 num_input_channels = 2;
- optional int32 num_output_channels = 3;
- optional int32 sample_rate = 4;
-
- message Frame {
- }
-
- repeated Frame frame = 5;
-
- optional int32 analog_level_average = 6;
- optional int32 max_output_average = 7;
-
- optional int32 has_echo_count = 8;
- optional int32 has_voice_count = 9;
- optional int32 is_saturated_count = 10;
-
- message Statistic {
- optional int32 instant = 1;
- optional int32 average = 2;
- optional int32 maximum = 3;
- optional int32 minimum = 4;
- }
-
- message EchoMetrics {
- optional Statistic residual_echo_return_loss = 1;
- optional Statistic echo_return_loss = 2;
- optional Statistic echo_return_loss_enhancement = 3;
- optional Statistic a_nlp = 4;
- }
-
- optional EchoMetrics echo_metrics = 11;
-
- message DelayMetrics {
- optional int32 median = 1;
- optional int32 std = 2;
- }
-
- optional DelayMetrics delay_metrics = 12;
-
- optional int32 rms_level = 13;
-}
-
-message OutputData {
- repeated Test test = 1;
-}
-
diff --git a/trunk/src/modules/audio_processing/test/unpack.cc b/trunk/src/modules/audio_processing/test/unpack.cc
deleted file mode 100644
index 2337131..0000000
--- a/trunk/src/modules/audio_processing/test/unpack.cc
+++ /dev/null
@@ -1,216 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Commandline tool to unpack audioproc debug files.
-//
-// The debug files are dumped as protobuf blobs. For analysis, it's necessary
-// to unpack the file into its component parts: audio and other data.
-
-#include <stdio.h>
-
-#include "google/gflags.h"
-#include "scoped_ptr.h"
-#include "typedefs.h"
-#include "webrtc/audio_processing/debug.pb.h"
-
-using webrtc::scoped_array;
-
-using webrtc::audioproc::Event;
-using webrtc::audioproc::ReverseStream;
-using webrtc::audioproc::Stream;
-using webrtc::audioproc::Init;
-
-// TODO(andrew): unpack more of the data.
-DEFINE_string(input_file, "input.pcm", "The name of the input stream file.");
-DEFINE_string(output_file, "ref_out.pcm",
- "The name of the reference output stream file.");
-DEFINE_string(reverse_file, "reverse.pcm",
- "The name of the reverse input stream file.");
-DEFINE_string(delay_file, "delay.int32", "The name of the delay file.");
-DEFINE_string(drift_file, "drift.int32", "The name of the drift file.");
-DEFINE_string(level_file, "level.int32", "The name of the level file.");
-DEFINE_string(settings_file, "settings.txt", "The name of the settings file.");
-DEFINE_bool(full, false,
- "Unpack the full set of files (normally not needed).");
-
-// TODO(andrew): move this to a helper class to share with process_test.cc?
-// Returns true on success, false on error or end-of-file.
-bool ReadMessageFromFile(FILE* file,
- ::google::protobuf::MessageLite* msg) {
- // The "wire format" for the size is little-endian.
- // Assume process_test is running on a little-endian machine.
- int32_t size = 0;
- if (fread(&size, sizeof(int32_t), 1, file) != 1) {
- return false;
- }
- if (size <= 0) {
- return false;
- }
- const size_t usize = static_cast<size_t>(size);
-
- scoped_array<char> array(new char[usize]);
- if (fread(array.get(), sizeof(char), usize, file) != usize) {
- return false;
- }
-
- msg->Clear();
- return msg->ParseFromArray(array.get(), usize);
-}
-
-int main(int argc, char* argv[]) {
- std::string program_name = argv[0];
- std::string usage = "Commandline tool to unpack audioproc debug files.\n"
- "Example usage:\n" + program_name + " debug_dump.pb\n";
- google::SetUsageMessage(usage);
- google::ParseCommandLineFlags(&argc, &argv, true);
-
- if (argc < 2) {
- printf("%s", google::ProgramUsage());
- return 1;
- }
-
- FILE* debug_file = fopen(argv[1], "rb");
- if (debug_file == NULL) {
- printf("Unable to open %s\n", argv[1]);
- return 1;
- }
- FILE* input_file = fopen(FLAGS_input_file.c_str(), "wb");
- if (input_file == NULL) {
- printf("Unable to open %s\n", FLAGS_input_file.c_str());
- return 1;
- }
- FILE* output_file = fopen(FLAGS_output_file.c_str(), "wb");
- if (output_file == NULL) {
- printf("Unable to open %s\n", FLAGS_output_file.c_str());
- return 1;
- }
- FILE* reverse_file = fopen(FLAGS_reverse_file.c_str(), "wb");
- if (reverse_file == NULL) {
- printf("Unable to open %s\n", FLAGS_reverse_file.c_str());
- return 1;
- }
- FILE* settings_file = fopen(FLAGS_settings_file.c_str(), "wb");
- if (settings_file == NULL) {
- printf("Unable to open %s\n", FLAGS_settings_file.c_str());
- return 1;
- }
-
- FILE* delay_file = NULL;
- FILE* drift_file = NULL;
- FILE* level_file = NULL;
- if (FLAGS_full) {
- delay_file = fopen(FLAGS_delay_file.c_str(), "wb");
- if (delay_file == NULL) {
- printf("Unable to open %s\n", FLAGS_delay_file.c_str());
- return 1;
- }
- drift_file = fopen(FLAGS_drift_file.c_str(), "wb");
- if (drift_file == NULL) {
- printf("Unable to open %s\n", FLAGS_drift_file.c_str());
- return 1;
- }
- level_file = fopen(FLAGS_level_file.c_str(), "wb");
- if (level_file == NULL) {
- printf("Unable to open %s\n", FLAGS_level_file.c_str());
- return 1;
- }
- }
-
- Event event_msg;
- int frame_count = 0;
- while (ReadMessageFromFile(debug_file, &event_msg)) {
- if (event_msg.type() == Event::REVERSE_STREAM) {
- if (!event_msg.has_reverse_stream()) {
- printf("Corrupted input file: ReverseStream missing.\n");
- return 1;
- }
-
- const ReverseStream msg = event_msg.reverse_stream();
- if (msg.has_data()) {
- if (fwrite(msg.data().data(), msg.data().size(), 1, reverse_file) !=
- 1) {
- printf("Error when writing to %s\n", FLAGS_reverse_file.c_str());
- return 1;
- }
- }
- } else if (event_msg.type() == Event::STREAM) {
- frame_count++;
- if (!event_msg.has_stream()) {
- printf("Corrupted input file: Stream missing.\n");
- return 1;
- }
-
- const Stream msg = event_msg.stream();
- if (msg.has_input_data()) {
- if (fwrite(msg.input_data().data(), msg.input_data().size(), 1,
- input_file) != 1) {
- printf("Error when writing to %s\n", FLAGS_input_file.c_str());
- return 1;
- }
- }
-
- if (msg.has_output_data()) {
- if (fwrite(msg.output_data().data(), msg.output_data().size(), 1,
- output_file) != 1) {
- printf("Error when writing to %s\n", FLAGS_output_file.c_str());
- return 1;
- }
- }
-
- if (FLAGS_full) {
- if (msg.has_delay()) {
- int32_t delay = msg.delay();
- if (fwrite(&delay, sizeof(int32_t), 1, delay_file) != 1) {
- printf("Error when writing to %s\n", FLAGS_delay_file.c_str());
- return 1;
- }
- }
-
- if (msg.has_drift()) {
- int32_t drift = msg.drift();
- if (fwrite(&drift, sizeof(int32_t), 1, drift_file) != 1) {
- printf("Error when writing to %s\n", FLAGS_drift_file.c_str());
- return 1;
- }
- }
-
- if (msg.has_level()) {
- int32_t level = msg.level();
- if (fwrite(&level, sizeof(int32_t), 1, level_file) != 1) {
- printf("Error when writing to %s\n", FLAGS_level_file.c_str());
- return 1;
- }
- }
- }
- } else if (event_msg.type() == Event::INIT) {
- if (!event_msg.has_init()) {
- printf("Corrupted input file: Init missing.\n");
- return 1;
- }
-
- const Init msg = event_msg.init();
- // These should print out zeros if they're missing.
- fprintf(settings_file, "Init at frame: %d\n", frame_count);
- fprintf(settings_file, " Sample rate: %d\n", msg.sample_rate());
- fprintf(settings_file, " Device sample rate: %d\n",
- msg.device_sample_rate());
- fprintf(settings_file, " Input channels: %d\n",
- msg.num_input_channels());
- fprintf(settings_file, " Output channels: %d\n",
- msg.num_output_channels());
- fprintf(settings_file, " Reverse channels: %d\n",
- msg.num_reverse_channels());
-
- fprintf(settings_file, "\n");
- }
- }
-
- return 0;
-}
diff --git a/trunk/src/modules/audio_processing/utility/Android.mk b/trunk/src/modules/audio_processing/utility/Android.mk
deleted file mode 100644
index bd3d039..0000000
--- a/trunk/src/modules/audio_processing/utility/Android.mk
+++ /dev/null
@@ -1,43 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_apm_utility
-LOCAL_MODULE_TAGS := optional
-LOCAL_SRC_FILES := \
- fft4g.c \
- ring_buffer.c \
- delay_estimator.c \
- delay_estimator_wrapper.c
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/../../.. \
- $(LOCAL_PATH)/../../../common_audio/signal_processing/include
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/audio_processing/utility/delay_estimator.c b/trunk/src/modules/audio_processing/utility/delay_estimator.c
deleted file mode 100644
index 24ee74d..0000000
--- a/trunk/src/modules/audio_processing/utility/delay_estimator.c
+++ /dev/null
@@ -1,319 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "delay_estimator.h"
-
-#include <assert.h>
-#include <stdlib.h>
-#include <string.h>
-
-// Number of right shifts for scaling is linearly depending on number of bits in
-// the far-end binary spectrum.
-static const int kShiftsAtZero = 13; // Right shifts at zero binary spectrum.
-static const int kShiftsLinearSlope = 3;
-
-static const int32_t kProbabilityOffset = 1024; // 2 in Q9.
-static const int32_t kProbabilityLowerLimit = 8704; // 17 in Q9.
-static const int32_t kProbabilityMinSpread = 2816; // 5.5 in Q9.
-
-// Counts and returns number of bits of a 32-bit word.
-static int BitCount(uint32_t u32) {
- uint32_t tmp = u32 - ((u32 >> 1) & 033333333333) -
- ((u32 >> 2) & 011111111111);
- tmp = ((tmp + (tmp >> 3)) & 030707070707);
- tmp = (tmp + (tmp >> 6));
- tmp = (tmp + (tmp >> 12) + (tmp >> 24)) & 077;
-
- return ((int) tmp);
-}
-
-// Compares the |binary_vector| with all rows of the |binary_matrix| and counts
-// per row the number of times they have the same value.
-//
-// Inputs:
-// - binary_vector : binary "vector" stored in a long
-// - binary_matrix : binary "matrix" stored as a vector of long
-// - matrix_size : size of binary "matrix"
-//
-// Output:
-// - bit_counts : "Vector" stored as a long, containing for each
-// row the number of times the matrix row and the
-// input vector have the same value
-//
-static void BitCountComparison(uint32_t binary_vector,
- const uint32_t* binary_matrix,
- int matrix_size,
- int32_t* bit_counts) {
- int n = 0;
-
- // Compare |binary_vector| with all rows of the |binary_matrix|
- for (; n < matrix_size; n++) {
- bit_counts[n] = (int32_t) BitCount(binary_vector ^ binary_matrix[n]);
- }
-}
-
-int WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* handle) {
- assert(handle != NULL);
-
- if (handle->mean_bit_counts != NULL) {
- free(handle->mean_bit_counts);
- handle->mean_bit_counts = NULL;
- }
- if (handle->bit_counts != NULL) {
- free(handle->bit_counts);
- handle->bit_counts = NULL;
- }
- if (handle->binary_far_history != NULL) {
- free(handle->binary_far_history);
- handle->binary_far_history = NULL;
- }
- if (handle->binary_near_history != NULL) {
- free(handle->binary_near_history);
- handle->binary_near_history = NULL;
- }
- if (handle->far_bit_counts != NULL) {
- free(handle->far_bit_counts);
- handle->far_bit_counts = NULL;
- }
-
- free(handle);
-
- return 0;
-}
-
-int WebRtc_CreateBinaryDelayEstimator(BinaryDelayEstimator** handle,
- int max_delay,
- int lookahead) {
- BinaryDelayEstimator* self = NULL;
- int history_size = max_delay + lookahead;
-
- if (handle == NULL) {
- return -1;
- }
- if (max_delay < 0) {
- return -1;
- }
- if (lookahead < 0) {
- return -1;
- }
- if (history_size < 2) {
- // Must be this large for buffer shifting.
- return -1;
- }
-
- self = malloc(sizeof(BinaryDelayEstimator));
- *handle = self;
- if (self == NULL) {
- return -1;
- }
-
- self->mean_bit_counts = NULL;
- self->bit_counts = NULL;
- self->binary_far_history = NULL;
- self->far_bit_counts = NULL;
-
- self->history_size = history_size;
- self->near_history_size = lookahead + 1;
-
- // Allocate memory for spectrum buffers.
- self->mean_bit_counts = malloc(history_size * sizeof(int32_t));
- if (self->mean_bit_counts == NULL) {
- WebRtc_FreeBinaryDelayEstimator(self);
- self = NULL;
- return -1;
- }
- self->bit_counts = malloc(history_size * sizeof(int32_t));
- if (self->bit_counts == NULL) {
- WebRtc_FreeBinaryDelayEstimator(self);
- self = NULL;
- return -1;
- }
- // Allocate memory for history buffers.
- self->binary_far_history = malloc(history_size * sizeof(uint32_t));
- if (self->binary_far_history == NULL) {
- WebRtc_FreeBinaryDelayEstimator(self);
- self = NULL;
- return -1;
- }
- self->binary_near_history = malloc(self->near_history_size *
- sizeof(uint32_t));
- if (self->binary_near_history == NULL) {
- WebRtc_FreeBinaryDelayEstimator(self);
- self = NULL;
- return -1;
- }
- self->far_bit_counts = malloc(history_size * sizeof(int));
- if (self->far_bit_counts == NULL) {
- WebRtc_FreeBinaryDelayEstimator(self);
- self = NULL;
- return -1;
- }
-
- return 0;
-}
-
-int WebRtc_InitBinaryDelayEstimator(BinaryDelayEstimator* handle) {
- int i = 0;
- assert(handle != NULL);
-
- memset(handle->bit_counts, 0, sizeof(int32_t) * handle->history_size);
- memset(handle->binary_far_history, 0,
- sizeof(uint32_t) * handle->history_size);
- memset(handle->binary_near_history, 0,
- sizeof(uint32_t) * handle->near_history_size);
- memset(handle->far_bit_counts, 0, sizeof(int) * handle->history_size);
- for (i = 0; i < handle->history_size; ++i) {
- handle->mean_bit_counts[i] = (20 << 9); // 20 in Q9.
- }
- handle->minimum_probability = (32 << 9); // 32 in Q9.
- handle->last_delay_probability = (32 << 9); // 32 in Q9.
-
- // Default return value if we're unable to estimate. -1 is used for errors.
- handle->last_delay = -2;
-
- return 0;
-}
-
-int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* handle,
- uint32_t binary_far_spectrum,
- uint32_t binary_near_spectrum) {
- int i = 0;
- int candidate_delay = -1;
-
- int32_t value_best_candidate = 16384; // 32 in Q9, (max |mean_bit_counts|).
- int32_t value_worst_candidate = 0;
-
- assert(handle != NULL);
- // Shift binary spectrum history and insert current |binary_far_spectrum|.
- memmove(&(handle->binary_far_history[1]), &(handle->binary_far_history[0]),
- (handle->history_size - 1) * sizeof(uint32_t));
- handle->binary_far_history[0] = binary_far_spectrum;
-
- // Shift history of far-end binary spectrum bit counts and insert bit count
- // of current |binary_far_spectrum|.
- memmove(&(handle->far_bit_counts[1]), &(handle->far_bit_counts[0]),
- (handle->history_size - 1) * sizeof(int));
- handle->far_bit_counts[0] = BitCount(binary_far_spectrum);
-
- if (handle->near_history_size > 1) {
- // If we apply lookahead, shift near-end binary spectrum history. Insert
- // current |binary_near_spectrum| and pull out the delayed one.
- memmove(&(handle->binary_near_history[1]),
- &(handle->binary_near_history[0]),
- (handle->near_history_size - 1) * sizeof(uint32_t));
- handle->binary_near_history[0] = binary_near_spectrum;
- binary_near_spectrum =
- handle->binary_near_history[handle->near_history_size - 1];
- }
-
- // Compare with delayed spectra and store the |bit_counts| for each delay.
- BitCountComparison(binary_near_spectrum,
- handle->binary_far_history,
- handle->history_size,
- handle->bit_counts);
-
- // Update |mean_bit_counts|, which is the smoothed version of |bit_counts|.
- for (i = 0; i < handle->history_size; i++) {
- // |bit_counts| is constrained to [0, 32], meaning we can smooth with a
- // factor up to 2^26. We use Q9.
- int32_t bit_count = (handle->bit_counts[i] << 9); // Q9.
-
- // Update |mean_bit_counts| only when far-end signal has something to
- // contribute. If |far_bit_counts| is zero the far-end signal is weak and
- // we likely have a poor echo condition, hence don't update.
- if (handle->far_bit_counts[i] > 0) {
- // Make number of right shifts piecewise linear w.r.t. |far_bit_counts|.
- int shifts = kShiftsAtZero;
- shifts -= (kShiftsLinearSlope * handle->far_bit_counts[i]) >> 4;
- WebRtc_MeanEstimatorFix(bit_count, shifts, &(handle->mean_bit_counts[i]));
- }
- }
-
- // Find |candidate_delay|, |value_best_candidate| and |value_worst_candidate|
- // of |mean_bit_counts|.
- for (i = 0; i < handle->history_size; i++) {
- if (handle->mean_bit_counts[i] < value_best_candidate) {
- value_best_candidate = handle->mean_bit_counts[i];
- candidate_delay = i;
- }
- if (handle->mean_bit_counts[i] > value_worst_candidate) {
- value_worst_candidate = handle->mean_bit_counts[i];
- }
- }
-
- // The |value_best_candidate| is a good indicator on the probability of
- // |candidate_delay| being an accurate delay (a small |value_best_candidate|
- // means a good binary match). In the following sections we make a decision
- // whether to update |last_delay| or not.
- // 1) If the difference bit counts between the best and the worst delay
- // candidates is too small we consider the situation to be unreliable and
- // don't update |last_delay|.
- // 2) If the situation is reliable we update |last_delay| if the value of the
- // best candidate delay has a value less than
- // i) an adaptive threshold |minimum_probability|, or
- // ii) this corresponding value |last_delay_probability|, but updated at
- // this time instant.
-
- // Update |minimum_probability|.
- if ((handle->minimum_probability > kProbabilityLowerLimit) &&
- (value_worst_candidate - value_best_candidate > kProbabilityMinSpread)) {
- // The "hard" threshold can't be lower than 17 (in Q9).
- // The valley in the curve also has to be distinct, i.e., the
- // difference between |value_worst_candidate| and |value_best_candidate| has
- // to be large enough.
- int32_t threshold = value_best_candidate + kProbabilityOffset;
- if (threshold < kProbabilityLowerLimit) {
- threshold = kProbabilityLowerLimit;
- }
- if (handle->minimum_probability > threshold) {
- handle->minimum_probability = threshold;
- }
- }
- // Update |last_delay_probability|.
- // We use a Markov type model, i.e., a slowly increasing level over time.
- handle->last_delay_probability++;
- if (value_worst_candidate > value_best_candidate + kProbabilityOffset) {
- // Reliable delay value for usage.
- if (value_best_candidate < handle->minimum_probability) {
- handle->last_delay = candidate_delay;
- }
- if (value_best_candidate < handle->last_delay_probability) {
- handle->last_delay = candidate_delay;
- // Reset |last_delay_probability|.
- handle->last_delay_probability = value_best_candidate;
- }
- }
-
- return handle->last_delay;
-}
-
-int WebRtc_binary_last_delay(BinaryDelayEstimator* handle) {
- assert(handle != NULL);
- return handle->last_delay;
-}
-
-int WebRtc_history_size(BinaryDelayEstimator* handle) {
- assert(handle != NULL);
- return handle->history_size;
-}
-
-void WebRtc_MeanEstimatorFix(int32_t new_value,
- int factor,
- int32_t* mean_value) {
- int32_t diff = new_value - *mean_value;
-
- // mean_new = mean_value + ((new_value - mean_value) >> factor);
- if (diff < 0) {
- diff = -((-diff) >> factor);
- } else {
- diff = (diff >> factor);
- }
- *mean_value += diff;
-}
diff --git a/trunk/src/modules/audio_processing/utility/delay_estimator.h b/trunk/src/modules/audio_processing/utility/delay_estimator.h
deleted file mode 100644
index a376dfe..0000000
--- a/trunk/src/modules/audio_processing/utility/delay_estimator.h
+++ /dev/null
@@ -1,128 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Performs delay estimation on binary converted spectra.
-// The return value is 0 - OK and -1 - Error, unless otherwise stated.
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_
-
-#include "typedefs.h"
-
-typedef struct {
- // Pointer to bit counts.
- int32_t* mean_bit_counts;
- int* far_bit_counts;
-
- // Array only used locally in ProcessBinarySpectrum() but whose size is
- // determined at run-time.
- int32_t* bit_counts;
-
- // Binary history variables.
- uint32_t* binary_far_history;
- uint32_t* binary_near_history;
-
- // Delay estimation variables.
- int32_t minimum_probability;
- int last_delay_probability;
-
- // Delay memory.
- int last_delay;
-
- // Buffer size.
- int history_size;
-
- // Near-end buffer size.
- int near_history_size;
-} BinaryDelayEstimator;
-
-// Releases the memory allocated by WebRtc_CreateBinaryDelayEstimator(...).
-// Input:
-// - handle : Pointer to the delay estimation instance.
-//
-int WebRtc_FreeBinaryDelayEstimator(BinaryDelayEstimator* handle);
-
-// Refer to WebRtc_CreateDelayEstimator() in delay_estimator_wrapper.h.
-int WebRtc_CreateBinaryDelayEstimator(BinaryDelayEstimator** handle,
- int max_delay,
- int lookahead);
-
-// Initializes the delay estimation instance created with
-// WebRtc_CreateBinaryDelayEstimator(...).
-// Input:
-// - handle : Pointer to the delay estimation instance.
-//
-// Output:
-// - handle : Initialized instance.
-//
-int WebRtc_InitBinaryDelayEstimator(BinaryDelayEstimator* handle);
-
-// Estimates and returns the delay between the binary far-end and binary near-
-// end spectra. The value will be offset by the lookahead (i.e. the lookahead
-// should be subtracted from the returned value).
-// Inputs:
-// - handle : Pointer to the delay estimation instance.
-// - binary_far_spectrum : Far-end binary spectrum.
-// - binary_near_spectrum : Near-end binary spectrum of the current block.
-//
-// Output:
-// - handle : Updated instance.
-//
-// Return value:
-// - delay : >= 0 - Calculated delay value.
-// -1 - Error.
-// -2 - Insufficient data for estimation.
-//
-int WebRtc_ProcessBinarySpectrum(BinaryDelayEstimator* handle,
- uint32_t binary_far_spectrum,
- uint32_t binary_near_spectrum);
-
-// Returns the last calculated delay updated by the function
-// WebRtc_ProcessBinarySpectrum(...).
-//
-// Input:
-// - handle : Pointer to the delay estimation instance.
-//
-// Return value:
-// - delay : >= 0 - Last calculated delay value
-// -1 - Error
-// -2 - Insufficient data for estimation.
-//
-int WebRtc_binary_last_delay(BinaryDelayEstimator* handle);
-
-// Returns the history size used in the far-end buffers to calculate the delay
-// over.
-//
-// Input:
-// - handle : Pointer to the delay estimation instance.
-//
-// Return value:
-// - history_size : > 0 - Far-end history size.
-// -1 - Error.
-//
-int WebRtc_history_size(BinaryDelayEstimator* handle);
-
-// Updates the |mean_value| recursively with a step size of 2^-|factor|. This
-// function is used internally in the Binary Delay Estimator as well as the
-// Fixed point wrapper.
-//
-// Inputs:
-// - new_value : The new value the mean should be updated with.
-// - factor : The step size, in number of right shifts.
-//
-// Input/Output:
-// - mean_value : Pointer to the mean value.
-//
-void WebRtc_MeanEstimatorFix(int32_t new_value,
- int factor,
- int32_t* mean_value);
-
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_H_
diff --git a/trunk/src/modules/audio_processing/utility/delay_estimator_wrapper.c b/trunk/src/modules/audio_processing/utility/delay_estimator_wrapper.c
deleted file mode 100644
index 438c95f..0000000
--- a/trunk/src/modules/audio_processing/utility/delay_estimator_wrapper.c
+++ /dev/null
@@ -1,336 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "delay_estimator_wrapper.h"
-
-#include <assert.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include "delay_estimator.h"
-
-typedef union {
- float float_;
- int32_t int32_;
-} SpectrumType;
-
-typedef struct {
- // Pointers to mean values of spectrum.
- SpectrumType* mean_far_spectrum;
- SpectrumType* mean_near_spectrum;
- // |mean_*_spectrum| initialization indicator.
- int far_spectrum_initialized;
- int near_spectrum_initialized;
-
- int spectrum_size;
-
- // Binary spectrum based delay estimator
- BinaryDelayEstimator* binary_handle;
-} DelayEstimator;
-
-// Only bit |kBandFirst| through bit |kBandLast| are processed and
-// |kBandFirst| - |kBandLast| must be < 32.
-static const int kBandFirst = 12;
-static const int kBandLast = 43;
-
-static __inline uint32_t SetBit(uint32_t in, int pos) {
- uint32_t mask = (1 << pos);
- uint32_t out = (in | mask);
-
- return out;
-}
-
-// Calculates the mean recursively. Same version as WebRtc_MeanEstimatorFix(),
-// but for float.
-//
-// Inputs:
-// - new_value : New additional value.
-// - scale : Scale for smoothing (should be less than 1.0).
-//
-// Input/Output:
-// - mean_value : Pointer to the mean value for updating.
-//
-static void MeanEstimatorFloat(float new_value,
- float scale,
- float* mean_value) {
- assert(scale < 1.0f);
- *mean_value += (new_value - *mean_value) * scale;
-}
-
-// Computes the binary spectrum by comparing the input |spectrum| with a
-// |threshold_spectrum|. Float and fixed point versions.
-//
-// Inputs:
-// - spectrum : Spectrum of which the binary spectrum should be
-// calculated.
-// - threshold_spectrum : Threshold spectrum with which the input
-// spectrum is compared.
-// Return:
-// - out : Binary spectrum.
-//
-static uint32_t BinarySpectrumFix(uint16_t* spectrum,
- SpectrumType* threshold_spectrum,
- int q_domain,
- int* threshold_initialized) {
- int i = kBandFirst;
- uint32_t out = 0;
-
- assert(q_domain < 16);
-
- if (!(*threshold_initialized)) {
- // Set the |threshold_spectrum| to half the input |spectrum| as starting
- // value. This speeds up the convergence.
- for (i = kBandFirst; i <= kBandLast; i++) {
- if (spectrum[i] > 0) {
- // Convert input spectrum from Q(|q_domain|) to Q15.
- int32_t spectrum_q15 = ((int32_t) spectrum[i]) << (15 - q_domain);
- threshold_spectrum[i].int32_ = (spectrum_q15 >> 1);
- *threshold_initialized = 1;
- }
- }
- }
- for (i = kBandFirst; i <= kBandLast; i++) {
- // Convert input spectrum from Q(|q_domain|) to Q15.
- int32_t spectrum_q15 = ((int32_t) spectrum[i]) << (15 - q_domain);
- // Update the |threshold_spectrum|.
- WebRtc_MeanEstimatorFix(spectrum_q15, 6, &(threshold_spectrum[i].int32_));
- // Convert |spectrum| at current frequency bin to a binary value.
- if (spectrum_q15 > threshold_spectrum[i].int32_) {
- out = SetBit(out, i - kBandFirst);
- }
- }
-
- return out;
-}
-
-static uint32_t BinarySpectrumFloat(float* spectrum,
- SpectrumType* threshold_spectrum,
- int* threshold_initialized) {
- int i = kBandFirst;
- uint32_t out = 0;
- const float kScale = 1 / 64.0;
-
- if (!(*threshold_initialized)) {
- // Set the |threshold_spectrum| to half the input |spectrum| as starting
- // value. This speeds up the convergence.
- for (i = kBandFirst; i <= kBandLast; i++) {
- if (spectrum[i] > 0.0f) {
- threshold_spectrum[i].float_ = (spectrum[i] / 2);
- *threshold_initialized = 1;
- }
- }
- }
-
- for (i = kBandFirst; i <= kBandLast; i++) {
- // Update the |threshold_spectrum|.
- MeanEstimatorFloat(spectrum[i], kScale, &(threshold_spectrum[i].float_));
- // Convert |spectrum| at current frequency bin to a binary value.
- if (spectrum[i] > threshold_spectrum[i].float_) {
- out = SetBit(out, i - kBandFirst);
- }
- }
-
- return out;
-}
-
-int WebRtc_FreeDelayEstimator(void* handle) {
- DelayEstimator* self = (DelayEstimator*) handle;
-
- if (self == NULL) {
- return -1;
- }
-
- if (self->mean_far_spectrum != NULL) {
- free(self->mean_far_spectrum);
- self->mean_far_spectrum = NULL;
- }
- if (self->mean_near_spectrum != NULL) {
- free(self->mean_near_spectrum);
- self->mean_near_spectrum = NULL;
- }
-
- WebRtc_FreeBinaryDelayEstimator(self->binary_handle);
-
- free(self);
-
- return 0;
-}
-
-int WebRtc_CreateDelayEstimator(void** handle,
- int spectrum_size,
- int max_delay,
- int lookahead) {
- DelayEstimator* self = NULL;
-
- // Check if the sub band used in the delay estimation is small enough to fit
- // the binary spectra in a uint32_t.
- assert(kBandLast - kBandFirst < 32);
-
- if (handle == NULL) {
- return -1;
- }
- if (spectrum_size < kBandLast) {
- return -1;
- }
-
- self = malloc(sizeof(DelayEstimator));
- *handle = self;
- if (self == NULL) {
- return -1;
- }
-
- self->mean_far_spectrum = NULL;
- self->mean_near_spectrum = NULL;
-
- // Create binary delay estimator.
- if (WebRtc_CreateBinaryDelayEstimator(&self->binary_handle,
- max_delay,
- lookahead) != 0) {
- WebRtc_FreeDelayEstimator(self);
- self = NULL;
- return -1;
- }
- // Allocate memory for spectrum buffers.
- self->mean_far_spectrum = malloc(spectrum_size * sizeof(SpectrumType));
- if (self->mean_far_spectrum == NULL) {
- WebRtc_FreeDelayEstimator(self);
- self = NULL;
- return -1;
- }
- self->mean_near_spectrum = malloc(spectrum_size * sizeof(SpectrumType));
- if (self->mean_near_spectrum == NULL) {
- WebRtc_FreeDelayEstimator(self);
- self = NULL;
- return -1;
- }
-
- self->spectrum_size = spectrum_size;
-
- return 0;
-}
-
-int WebRtc_InitDelayEstimator(void* handle) {
- DelayEstimator* self = (DelayEstimator*) handle;
-
- if (self == NULL) {
- return -1;
- }
-
- // Initialize binary delay estimator.
- if (WebRtc_InitBinaryDelayEstimator(self->binary_handle) != 0) {
- return -1;
- }
- // Set averaged far and near end spectra to zero.
- memset(self->mean_far_spectrum, 0,
- sizeof(SpectrumType) * self->spectrum_size);
- memset(self->mean_near_spectrum, 0,
- sizeof(SpectrumType) * self->spectrum_size);
- // Reset initialization indicators.
- self->far_spectrum_initialized = 0;
- self->near_spectrum_initialized = 0;
-
- return 0;
-}
-
-int WebRtc_DelayEstimatorProcessFix(void* handle,
- uint16_t* far_spectrum,
- uint16_t* near_spectrum,
- int spectrum_size,
- int far_q,
- int near_q) {
- DelayEstimator* self = (DelayEstimator*) handle;
- uint32_t binary_far_spectrum = 0;
- uint32_t binary_near_spectrum = 0;
-
- if (self == NULL) {
- return -1;
- }
- if (far_spectrum == NULL) {
- // Empty far end spectrum.
- return -1;
- }
- if (near_spectrum == NULL) {
- // Empty near end spectrum.
- return -1;
- }
- if (spectrum_size != self->spectrum_size) {
- // Data sizes don't match.
- return -1;
- }
- if (far_q > 15) {
- // If |far_q| is larger than 15 we cannot guarantee no wrap around.
- return -1;
- }
- if (near_q > 15) {
- // If |near_q| is larger than 15 we cannot guarantee no wrap around.
- return -1;
- }
-
- // Get binary spectra.
- binary_far_spectrum = BinarySpectrumFix(far_spectrum,
- self->mean_far_spectrum,
- far_q,
- &(self->far_spectrum_initialized));
- binary_near_spectrum = BinarySpectrumFix(near_spectrum,
- self->mean_near_spectrum,
- near_q,
- &(self->near_spectrum_initialized));
-
- return WebRtc_ProcessBinarySpectrum(self->binary_handle,
- binary_far_spectrum,
- binary_near_spectrum);
-}
-
-int WebRtc_DelayEstimatorProcessFloat(void* handle,
- float* far_spectrum,
- float* near_spectrum,
- int spectrum_size) {
- DelayEstimator* self = (DelayEstimator*) handle;
- uint32_t binary_far_spectrum = 0;
- uint32_t binary_near_spectrum = 0;
-
- if (self == NULL) {
- return -1;
- }
- if (far_spectrum == NULL) {
- // Empty far end spectrum.
- return -1;
- }
- if (near_spectrum == NULL) {
- // Empty near end spectrum.
- return -1;
- }
- if (spectrum_size != self->spectrum_size) {
- // Data sizes don't match.
- return -1;
- }
-
- // Get binary spectra.
- binary_far_spectrum = BinarySpectrumFloat(far_spectrum,
- self->mean_far_spectrum,
- &(self->far_spectrum_initialized));
- binary_near_spectrum = BinarySpectrumFloat(near_spectrum,
- self->mean_near_spectrum,
- &(self->near_spectrum_initialized));
-
- return WebRtc_ProcessBinarySpectrum(self->binary_handle,
- binary_far_spectrum,
- binary_near_spectrum);
-}
-
-int WebRtc_last_delay(void* handle) {
- DelayEstimator* self = (DelayEstimator*) handle;
-
- if (self == NULL) {
- return -1;
- }
-
- return WebRtc_binary_last_delay(self->binary_handle);
-}
diff --git a/trunk/src/modules/audio_processing/utility/delay_estimator_wrapper.h b/trunk/src/modules/audio_processing/utility/delay_estimator_wrapper.h
deleted file mode 100644
index 2a47b5d..0000000
--- a/trunk/src/modules/audio_processing/utility/delay_estimator_wrapper.h
+++ /dev/null
@@ -1,110 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Performs delay estimation on block by block basis.
-// The return value is 0 - OK and -1 - Error, unless otherwise stated.
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_
-
-#include "typedefs.h"
-
-// Releases the memory allocated by WebRtc_CreateDelayEstimator(...)
-// Input:
-// - handle : Pointer to the delay estimation instance.
-//
-int WebRtc_FreeDelayEstimator(void* handle);
-
-// Allocates the memory needed by the delay estimation. The memory needs to be
-// initialized separately through WebRtc_InitDelayEstimator(...).
-//
-// Inputs:
-// - handle : Instance that should be created.
-// - spectrum_size : Size of the spectrum used both in far-end and
-// near-end. Used to allocate memory for spectrum
-// specific buffers.
-// - max_delay : The maximum delay which can be estimated. Needed to
-// allocate memory for history buffers.
-// - lookahead : Amount of non-causal lookahead to use. This can
-// detect cases in which a near-end signal occurs before
-// the corresponding far-end signal. It will delay the
-// estimate for the current block by an equal amount,
-// and the returned values will be offset by it.
-//
-// A value of zero is the typical no-lookahead case.
-// This also represents the minimum delay which can be
-// estimated.
-//
-// Output:
-// - handle : Created instance.
-//
-int WebRtc_CreateDelayEstimator(void** handle,
- int spectrum_size,
- int max_delay,
- int lookahead);
-
-// Initializes the delay estimation instance created with
-// WebRtc_CreateDelayEstimator(...)
-// Input:
-// - handle : Pointer to the delay estimation instance.
-//
-// Output:
-// - handle : Initialized instance.
-//
-int WebRtc_InitDelayEstimator(void* handle);
-
-// Estimates and returns the delay between the far-end and near-end blocks. The
-// value will be offset by the lookahead (i.e. the lookahead should be
-// subtracted from the returned value).
-// Inputs:
-// - handle : Pointer to the delay estimation instance.
-// - far_spectrum : Pointer to the far-end spectrum data.
-// - near_spectrum : Pointer to the near-end spectrum data of the current
-// block.
-// - spectrum_size : The size of the data arrays (same for both far- and
-// near-end).
-// - far_q : The Q-domain of the far-end data.
-// - near_q : The Q-domain of the near-end data.
-//
-// Output:
-// - handle : Updated instance.
-//
-// Return value:
-// - delay : >= 0 - Calculated delay value.
-// -1 - Error.
-// -2 - Insufficient data for estimation.
-//
-int WebRtc_DelayEstimatorProcessFix(void* handle,
- uint16_t* far_spectrum,
- uint16_t* near_spectrum,
- int spectrum_size,
- int far_q,
- int near_q);
-
-// See WebRtc_DelayEstimatorProcessFix() for description.
-int WebRtc_DelayEstimatorProcessFloat(void* handle,
- float* far_spectrum,
- float* near_spectrum,
- int spectrum_size);
-
-// Returns the last calculated delay updated by the function
-// WebRtc_DelayEstimatorProcess(...).
-//
-// Input:
-// - handle : Pointer to the delay estimation instance.
-//
-// Return value:
-// - delay : >= 0 - Last calculated delay value.
-// -1 - Error.
-// -2 - Insufficient data for estimation.
-//
-int WebRtc_last_delay(void* handle);
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_DELAY_ESTIMATOR_WRAPPER_H_
diff --git a/trunk/src/modules/audio_processing/utility/fft4g.c b/trunk/src/modules/audio_processing/utility/fft4g.c
deleted file mode 100644
index cbc4dc3..0000000
--- a/trunk/src/modules/audio_processing/utility/fft4g.c
+++ /dev/null
@@ -1,1326 +0,0 @@
-/*
- * http://www.kurims.kyoto-u.ac.jp/~ooura/fft.html
- * Copyright Takuya OOURA, 1996-2001
- *
- * You may use, copy, modify and distribute this code for any purpose (include
- * commercial use) and without fee. Please refer to this package when you modify
- * this code.
- *
- * Changes:
- * Trivial type modifications by the WebRTC authors.
- */
-
-/*
-Fast Fourier/Cosine/Sine Transform
- dimension :one
- data length :power of 2
- decimation :frequency
- radix :4, 2
- data :inplace
- table :use
-functions
- cdft: Complex Discrete Fourier Transform
- rdft: Real Discrete Fourier Transform
- ddct: Discrete Cosine Transform
- ddst: Discrete Sine Transform
- dfct: Cosine Transform of RDFT (Real Symmetric DFT)
- dfst: Sine Transform of RDFT (Real Anti-symmetric DFT)
-function prototypes
- void cdft(int, int, float *, int *, float *);
- void rdft(int, int, float *, int *, float *);
- void ddct(int, int, float *, int *, float *);
- void ddst(int, int, float *, int *, float *);
- void dfct(int, float *, float *, int *, float *);
- void dfst(int, float *, float *, int *, float *);
-
-
--------- Complex DFT (Discrete Fourier Transform) --------
- [definition]
- <case1>
- X[k] = sum_j=0^n-1 x[j]*exp(2*pi*i*j*k/n), 0<=k<n
- <case2>
- X[k] = sum_j=0^n-1 x[j]*exp(-2*pi*i*j*k/n), 0<=k<n
- (notes: sum_j=0^n-1 is a summation from j=0 to n-1)
- [usage]
- <case1>
- ip[0] = 0; // first time only
- cdft(2*n, 1, a, ip, w);
- <case2>
- ip[0] = 0; // first time only
- cdft(2*n, -1, a, ip, w);
- [parameters]
- 2*n :data length (int)
- n >= 1, n = power of 2
- a[0...2*n-1] :input/output data (float *)
- input data
- a[2*j] = Re(x[j]),
- a[2*j+1] = Im(x[j]), 0<=j<n
- output data
- a[2*k] = Re(X[k]),
- a[2*k+1] = Im(X[k]), 0<=k<n
- ip[0...*] :work area for bit reversal (int *)
- length of ip >= 2+sqrt(n)
- strictly,
- length of ip >=
- 2+(1<<(int)(log(n+0.5)/log(2))/2).
- ip[0],ip[1] are pointers of the cos/sin table.
- w[0...n/2-1] :cos/sin table (float *)
- w[],ip[] are initialized if ip[0] == 0.
- [remark]
- Inverse of
- cdft(2*n, -1, a, ip, w);
- is
- cdft(2*n, 1, a, ip, w);
- for (j = 0; j <= 2 * n - 1; j++) {
- a[j] *= 1.0 / n;
- }
- .
-
-
--------- Real DFT / Inverse of Real DFT --------
- [definition]
- <case1> RDFT
- R[k] = sum_j=0^n-1 a[j]*cos(2*pi*j*k/n), 0<=k<=n/2
- I[k] = sum_j=0^n-1 a[j]*sin(2*pi*j*k/n), 0<k<n/2
- <case2> IRDFT (excluding scale)
- a[k] = (R[0] + R[n/2]*cos(pi*k))/2 +
- sum_j=1^n/2-1 R[j]*cos(2*pi*j*k/n) +
- sum_j=1^n/2-1 I[j]*sin(2*pi*j*k/n), 0<=k<n
- [usage]
- <case1>
- ip[0] = 0; // first time only
- rdft(n, 1, a, ip, w);
- <case2>
- ip[0] = 0; // first time only
- rdft(n, -1, a, ip, w);
- [parameters]
- n :data length (int)
- n >= 2, n = power of 2
- a[0...n-1] :input/output data (float *)
- <case1>
- output data
- a[2*k] = R[k], 0<=k<n/2
- a[2*k+1] = I[k], 0<k<n/2
- a[1] = R[n/2]
- <case2>
- input data
- a[2*j] = R[j], 0<=j<n/2
- a[2*j+1] = I[j], 0<j<n/2
- a[1] = R[n/2]
- ip[0...*] :work area for bit reversal (int *)
- length of ip >= 2+sqrt(n/2)
- strictly,
- length of ip >=
- 2+(1<<(int)(log(n/2+0.5)/log(2))/2).
- ip[0],ip[1] are pointers of the cos/sin table.
- w[0...n/2-1] :cos/sin table (float *)
- w[],ip[] are initialized if ip[0] == 0.
- [remark]
- Inverse of
- rdft(n, 1, a, ip, w);
- is
- rdft(n, -1, a, ip, w);
- for (j = 0; j <= n - 1; j++) {
- a[j] *= 2.0 / n;
- }
- .
-
-
--------- DCT (Discrete Cosine Transform) / Inverse of DCT --------
- [definition]
- <case1> IDCT (excluding scale)
- C[k] = sum_j=0^n-1 a[j]*cos(pi*j*(k+1/2)/n), 0<=k<n
- <case2> DCT
- C[k] = sum_j=0^n-1 a[j]*cos(pi*(j+1/2)*k/n), 0<=k<n
- [usage]
- <case1>
- ip[0] = 0; // first time only
- ddct(n, 1, a, ip, w);
- <case2>
- ip[0] = 0; // first time only
- ddct(n, -1, a, ip, w);
- [parameters]
- n :data length (int)
- n >= 2, n = power of 2
- a[0...n-1] :input/output data (float *)
- output data
- a[k] = C[k], 0<=k<n
- ip[0...*] :work area for bit reversal (int *)
- length of ip >= 2+sqrt(n/2)
- strictly,
- length of ip >=
- 2+(1<<(int)(log(n/2+0.5)/log(2))/2).
- ip[0],ip[1] are pointers of the cos/sin table.
- w[0...n*5/4-1] :cos/sin table (float *)
- w[],ip[] are initialized if ip[0] == 0.
- [remark]
- Inverse of
- ddct(n, -1, a, ip, w);
- is
- a[0] *= 0.5;
- ddct(n, 1, a, ip, w);
- for (j = 0; j <= n - 1; j++) {
- a[j] *= 2.0 / n;
- }
- .
-
-
--------- DST (Discrete Sine Transform) / Inverse of DST --------
- [definition]
- <case1> IDST (excluding scale)
- S[k] = sum_j=1^n A[j]*sin(pi*j*(k+1/2)/n), 0<=k<n
- <case2> DST
- S[k] = sum_j=0^n-1 a[j]*sin(pi*(j+1/2)*k/n), 0<k<=n
- [usage]
- <case1>
- ip[0] = 0; // first time only
- ddst(n, 1, a, ip, w);
- <case2>
- ip[0] = 0; // first time only
- ddst(n, -1, a, ip, w);
- [parameters]
- n :data length (int)
- n >= 2, n = power of 2
- a[0...n-1] :input/output data (float *)
- <case1>
- input data
- a[j] = A[j], 0<j<n
- a[0] = A[n]
- output data
- a[k] = S[k], 0<=k<n
- <case2>
- output data
- a[k] = S[k], 0<k<n
- a[0] = S[n]
- ip[0...*] :work area for bit reversal (int *)
- length of ip >= 2+sqrt(n/2)
- strictly,
- length of ip >=
- 2+(1<<(int)(log(n/2+0.5)/log(2))/2).
- ip[0],ip[1] are pointers of the cos/sin table.
- w[0...n*5/4-1] :cos/sin table (float *)
- w[],ip[] are initialized if ip[0] == 0.
- [remark]
- Inverse of
- ddst(n, -1, a, ip, w);
- is
- a[0] *= 0.5;
- ddst(n, 1, a, ip, w);
- for (j = 0; j <= n - 1; j++) {
- a[j] *= 2.0 / n;
- }
- .
-
-
--------- Cosine Transform of RDFT (Real Symmetric DFT) --------
- [definition]
- C[k] = sum_j=0^n a[j]*cos(pi*j*k/n), 0<=k<=n
- [usage]
- ip[0] = 0; // first time only
- dfct(n, a, t, ip, w);
- [parameters]
- n :data length - 1 (int)
- n >= 2, n = power of 2
- a[0...n] :input/output data (float *)
- output data
- a[k] = C[k], 0<=k<=n
- t[0...n/2] :work area (float *)
- ip[0...*] :work area for bit reversal (int *)
- length of ip >= 2+sqrt(n/4)
- strictly,
- length of ip >=
- 2+(1<<(int)(log(n/4+0.5)/log(2))/2).
- ip[0],ip[1] are pointers of the cos/sin table.
- w[0...n*5/8-1] :cos/sin table (float *)
- w[],ip[] are initialized if ip[0] == 0.
- [remark]
- Inverse of
- a[0] *= 0.5;
- a[n] *= 0.5;
- dfct(n, a, t, ip, w);
- is
- a[0] *= 0.5;
- a[n] *= 0.5;
- dfct(n, a, t, ip, w);
- for (j = 0; j <= n; j++) {
- a[j] *= 2.0 / n;
- }
- .
-
-
--------- Sine Transform of RDFT (Real Anti-symmetric DFT) --------
- [definition]
- S[k] = sum_j=1^n-1 a[j]*sin(pi*j*k/n), 0<k<n
- [usage]
- ip[0] = 0; // first time only
- dfst(n, a, t, ip, w);
- [parameters]
- n :data length + 1 (int)
- n >= 2, n = power of 2
- a[0...n-1] :input/output data (float *)
- output data
- a[k] = S[k], 0<k<n
- (a[0] is used for work area)
- t[0...n/2-1] :work area (float *)
- ip[0...*] :work area for bit reversal (int *)
- length of ip >= 2+sqrt(n/4)
- strictly,
- length of ip >=
- 2+(1<<(int)(log(n/4+0.5)/log(2))/2).
- ip[0],ip[1] are pointers of the cos/sin table.
- w[0...n*5/8-1] :cos/sin table (float *)
- w[],ip[] are initialized if ip[0] == 0.
- [remark]
- Inverse of
- dfst(n, a, t, ip, w);
- is
- dfst(n, a, t, ip, w);
- for (j = 1; j <= n - 1; j++) {
- a[j] *= 2.0 / n;
- }
- .
-
-
-Appendix :
- The cos/sin table is recalculated when the larger table required.
- w[] and ip[] are compatible with all routines.
-*/
-
-static void makewt(int nw, int *ip, float *w);
-static void makect(int nc, int *ip, float *c);
-static void bitrv2(int n, int *ip, float *a);
-static void bitrv2conj(int n, int *ip, float *a);
-static void cftfsub(int n, float *a, float *w);
-static void cftbsub(int n, float *a, float *w);
-static void cft1st(int n, float *a, float *w);
-static void cftmdl(int n, int l, float *a, float *w);
-static void rftfsub(int n, float *a, int nc, float *c);
-static void rftbsub(int n, float *a, int nc, float *c);
-#if 0 // Not used.
-static void dctsub(int n, float *a, int nc, float *c)
-static void dstsub(int n, float *a, int nc, float *c)
-#endif
-
-
-void WebRtc_cdft(int n, int isgn, float *a, int *ip, float *w)
-{
- if (n > (ip[0] << 2)) {
- makewt(n >> 2, ip, w);
- }
- if (n > 4) {
- if (isgn >= 0) {
- bitrv2(n, ip + 2, a);
- cftfsub(n, a, w);
- } else {
- bitrv2conj(n, ip + 2, a);
- cftbsub(n, a, w);
- }
- } else if (n == 4) {
- cftfsub(n, a, w);
- }
-}
-
-
-void WebRtc_rdft(int n, int isgn, float *a, int *ip, float *w)
-{
- int nw, nc;
- float xi;
-
- nw = ip[0];
- if (n > (nw << 2)) {
- nw = n >> 2;
- makewt(nw, ip, w);
- }
- nc = ip[1];
- if (n > (nc << 2)) {
- nc = n >> 2;
- makect(nc, ip, w + nw);
- }
- if (isgn >= 0) {
- if (n > 4) {
- bitrv2(n, ip + 2, a);
- cftfsub(n, a, w);
- rftfsub(n, a, nc, w + nw);
- } else if (n == 4) {
- cftfsub(n, a, w);
- }
- xi = a[0] - a[1];
- a[0] += a[1];
- a[1] = xi;
- } else {
- a[1] = 0.5f * (a[0] - a[1]);
- a[0] -= a[1];
- if (n > 4) {
- rftbsub(n, a, nc, w + nw);
- bitrv2(n, ip + 2, a);
- cftbsub(n, a, w);
- } else if (n == 4) {
- cftfsub(n, a, w);
- }
- }
-}
-
-#if 0 // Not used.
-static void ddct(int n, int isgn, float *a, int *ip, float *w)
-{
- int j, nw, nc;
- float xr;
-
- nw = ip[0];
- if (n > (nw << 2)) {
- nw = n >> 2;
- makewt(nw, ip, w);
- }
- nc = ip[1];
- if (n > nc) {
- nc = n;
- makect(nc, ip, w + nw);
- }
- if (isgn < 0) {
- xr = a[n - 1];
- for (j = n - 2; j >= 2; j -= 2) {
- a[j + 1] = a[j] - a[j - 1];
- a[j] += a[j - 1];
- }
- a[1] = a[0] - xr;
- a[0] += xr;
- if (n > 4) {
- rftbsub(n, a, nc, w + nw);
- bitrv2(n, ip + 2, a);
- cftbsub(n, a, w);
- } else if (n == 4) {
- cftfsub(n, a, w);
- }
- }
- dctsub(n, a, nc, w + nw);
- if (isgn >= 0) {
- if (n > 4) {
- bitrv2(n, ip + 2, a);
- cftfsub(n, a, w);
- rftfsub(n, a, nc, w + nw);
- } else if (n == 4) {
- cftfsub(n, a, w);
- }
- xr = a[0] - a[1];
- a[0] += a[1];
- for (j = 2; j < n; j += 2) {
- a[j - 1] = a[j] - a[j + 1];
- a[j] += a[j + 1];
- }
- a[n - 1] = xr;
- }
-}
-
-
-static void ddst(int n, int isgn, float *a, int *ip, float *w)
-{
- int j, nw, nc;
- float xr;
-
- nw = ip[0];
- if (n > (nw << 2)) {
- nw = n >> 2;
- makewt(nw, ip, w);
- }
- nc = ip[1];
- if (n > nc) {
- nc = n;
- makect(nc, ip, w + nw);
- }
- if (isgn < 0) {
- xr = a[n - 1];
- for (j = n - 2; j >= 2; j -= 2) {
- a[j + 1] = -a[j] - a[j - 1];
- a[j] -= a[j - 1];
- }
- a[1] = a[0] + xr;
- a[0] -= xr;
- if (n > 4) {
- rftbsub(n, a, nc, w + nw);
- bitrv2(n, ip + 2, a);
- cftbsub(n, a, w);
- } else if (n == 4) {
- cftfsub(n, a, w);
- }
- }
- dstsub(n, a, nc, w + nw);
- if (isgn >= 0) {
- if (n > 4) {
- bitrv2(n, ip + 2, a);
- cftfsub(n, a, w);
- rftfsub(n, a, nc, w + nw);
- } else if (n == 4) {
- cftfsub(n, a, w);
- }
- xr = a[0] - a[1];
- a[0] += a[1];
- for (j = 2; j < n; j += 2) {
- a[j - 1] = -a[j] - a[j + 1];
- a[j] -= a[j + 1];
- }
- a[n - 1] = -xr;
- }
-}
-
-
-static void dfct(int n, float *a, float *t, int *ip, float *w)
-{
- int j, k, l, m, mh, nw, nc;
- float xr, xi, yr, yi;
-
- nw = ip[0];
- if (n > (nw << 3)) {
- nw = n >> 3;
- makewt(nw, ip, w);
- }
- nc = ip[1];
- if (n > (nc << 1)) {
- nc = n >> 1;
- makect(nc, ip, w + nw);
- }
- m = n >> 1;
- yi = a[m];
- xi = a[0] + a[n];
- a[0] -= a[n];
- t[0] = xi - yi;
- t[m] = xi + yi;
- if (n > 2) {
- mh = m >> 1;
- for (j = 1; j < mh; j++) {
- k = m - j;
- xr = a[j] - a[n - j];
- xi = a[j] + a[n - j];
- yr = a[k] - a[n - k];
- yi = a[k] + a[n - k];
- a[j] = xr;
- a[k] = yr;
- t[j] = xi - yi;
- t[k] = xi + yi;
- }
- t[mh] = a[mh] + a[n - mh];
- a[mh] -= a[n - mh];
- dctsub(m, a, nc, w + nw);
- if (m > 4) {
- bitrv2(m, ip + 2, a);
- cftfsub(m, a, w);
- rftfsub(m, a, nc, w + nw);
- } else if (m == 4) {
- cftfsub(m, a, w);
- }
- a[n - 1] = a[0] - a[1];
- a[1] = a[0] + a[1];
- for (j = m - 2; j >= 2; j -= 2) {
- a[2 * j + 1] = a[j] + a[j + 1];
- a[2 * j - 1] = a[j] - a[j + 1];
- }
- l = 2;
- m = mh;
- while (m >= 2) {
- dctsub(m, t, nc, w + nw);
- if (m > 4) {
- bitrv2(m, ip + 2, t);
- cftfsub(m, t, w);
- rftfsub(m, t, nc, w + nw);
- } else if (m == 4) {
- cftfsub(m, t, w);
- }
- a[n - l] = t[0] - t[1];
- a[l] = t[0] + t[1];
- k = 0;
- for (j = 2; j < m; j += 2) {
- k += l << 2;
- a[k - l] = t[j] - t[j + 1];
- a[k + l] = t[j] + t[j + 1];
- }
- l <<= 1;
- mh = m >> 1;
- for (j = 0; j < mh; j++) {
- k = m - j;
- t[j] = t[m + k] - t[m + j];
- t[k] = t[m + k] + t[m + j];
- }
- t[mh] = t[m + mh];
- m = mh;
- }
- a[l] = t[0];
- a[n] = t[2] - t[1];
- a[0] = t[2] + t[1];
- } else {
- a[1] = a[0];
- a[2] = t[0];
- a[0] = t[1];
- }
-}
-
-static void dfst(int n, float *a, float *t, int *ip, float *w)
-{
- int j, k, l, m, mh, nw, nc;
- float xr, xi, yr, yi;
-
- nw = ip[0];
- if (n > (nw << 3)) {
- nw = n >> 3;
- makewt(nw, ip, w);
- }
- nc = ip[1];
- if (n > (nc << 1)) {
- nc = n >> 1;
- makect(nc, ip, w + nw);
- }
- if (n > 2) {
- m = n >> 1;
- mh = m >> 1;
- for (j = 1; j < mh; j++) {
- k = m - j;
- xr = a[j] + a[n - j];
- xi = a[j] - a[n - j];
- yr = a[k] + a[n - k];
- yi = a[k] - a[n - k];
- a[j] = xr;
- a[k] = yr;
- t[j] = xi + yi;
- t[k] = xi - yi;
- }
- t[0] = a[mh] - a[n - mh];
- a[mh] += a[n - mh];
- a[0] = a[m];
- dstsub(m, a, nc, w + nw);
- if (m > 4) {
- bitrv2(m, ip + 2, a);
- cftfsub(m, a, w);
- rftfsub(m, a, nc, w + nw);
- } else if (m == 4) {
- cftfsub(m, a, w);
- }
- a[n - 1] = a[1] - a[0];
- a[1] = a[0] + a[1];
- for (j = m - 2; j >= 2; j -= 2) {
- a[2 * j + 1] = a[j] - a[j + 1];
- a[2 * j - 1] = -a[j] - a[j + 1];
- }
- l = 2;
- m = mh;
- while (m >= 2) {
- dstsub(m, t, nc, w + nw);
- if (m > 4) {
- bitrv2(m, ip + 2, t);
- cftfsub(m, t, w);
- rftfsub(m, t, nc, w + nw);
- } else if (m == 4) {
- cftfsub(m, t, w);
- }
- a[n - l] = t[1] - t[0];
- a[l] = t[0] + t[1];
- k = 0;
- for (j = 2; j < m; j += 2) {
- k += l << 2;
- a[k - l] = -t[j] - t[j + 1];
- a[k + l] = t[j] - t[j + 1];
- }
- l <<= 1;
- mh = m >> 1;
- for (j = 1; j < mh; j++) {
- k = m - j;
- t[j] = t[m + k] + t[m + j];
- t[k] = t[m + k] - t[m + j];
- }
- t[0] = t[m + mh];
- m = mh;
- }
- a[l] = t[0];
- }
- a[0] = 0;
-}
-#endif // Not used.
-
-
-/* -------- initializing routines -------- */
-
-
-#include <math.h>
-
-static void makewt(int nw, int *ip, float *w)
-{
- int j, nwh;
- float delta, x, y;
-
- ip[0] = nw;
- ip[1] = 1;
- if (nw > 2) {
- nwh = nw >> 1;
- delta = (float)atan(1.0f) / nwh;
- w[0] = 1;
- w[1] = 0;
- w[nwh] = (float)cos(delta * nwh);
- w[nwh + 1] = w[nwh];
- if (nwh > 2) {
- for (j = 2; j < nwh; j += 2) {
- x = (float)cos(delta * j);
- y = (float)sin(delta * j);
- w[j] = x;
- w[j + 1] = y;
- w[nw - j] = y;
- w[nw - j + 1] = x;
- }
- bitrv2(nw, ip + 2, w);
- }
- }
-}
-
-
-static void makect(int nc, int *ip, float *c)
-{
- int j, nch;
- float delta;
-
- ip[1] = nc;
- if (nc > 1) {
- nch = nc >> 1;
- delta = (float)atan(1.0f) / nch;
- c[0] = (float)cos(delta * nch);
- c[nch] = 0.5f * c[0];
- for (j = 1; j < nch; j++) {
- c[j] = 0.5f * (float)cos(delta * j);
- c[nc - j] = 0.5f * (float)sin(delta * j);
- }
- }
-}
-
-
-/* -------- child routines -------- */
-
-
-static void bitrv2(int n, int *ip, float *a)
-{
- int j, j1, k, k1, l, m, m2;
- float xr, xi, yr, yi;
-
- ip[0] = 0;
- l = n;
- m = 1;
- while ((m << 3) < l) {
- l >>= 1;
- for (j = 0; j < m; j++) {
- ip[m + j] = ip[j] + l;
- }
- m <<= 1;
- }
- m2 = 2 * m;
- if ((m << 3) == l) {
- for (k = 0; k < m; k++) {
- for (j = 0; j < k; j++) {
- j1 = 2 * j + ip[k];
- k1 = 2 * k + ip[j];
- xr = a[j1];
- xi = a[j1 + 1];
- yr = a[k1];
- yi = a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- j1 += m2;
- k1 += 2 * m2;
- xr = a[j1];
- xi = a[j1 + 1];
- yr = a[k1];
- yi = a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- j1 += m2;
- k1 -= m2;
- xr = a[j1];
- xi = a[j1 + 1];
- yr = a[k1];
- yi = a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- j1 += m2;
- k1 += 2 * m2;
- xr = a[j1];
- xi = a[j1 + 1];
- yr = a[k1];
- yi = a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- }
- j1 = 2 * k + m2 + ip[k];
- k1 = j1 + m2;
- xr = a[j1];
- xi = a[j1 + 1];
- yr = a[k1];
- yi = a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- }
- } else {
- for (k = 1; k < m; k++) {
- for (j = 0; j < k; j++) {
- j1 = 2 * j + ip[k];
- k1 = 2 * k + ip[j];
- xr = a[j1];
- xi = a[j1 + 1];
- yr = a[k1];
- yi = a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- j1 += m2;
- k1 += m2;
- xr = a[j1];
- xi = a[j1 + 1];
- yr = a[k1];
- yi = a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- }
- }
- }
-}
-
-
-static void bitrv2conj(int n, int *ip, float *a)
-{
- int j, j1, k, k1, l, m, m2;
- float xr, xi, yr, yi;
-
- ip[0] = 0;
- l = n;
- m = 1;
- while ((m << 3) < l) {
- l >>= 1;
- for (j = 0; j < m; j++) {
- ip[m + j] = ip[j] + l;
- }
- m <<= 1;
- }
- m2 = 2 * m;
- if ((m << 3) == l) {
- for (k = 0; k < m; k++) {
- for (j = 0; j < k; j++) {
- j1 = 2 * j + ip[k];
- k1 = 2 * k + ip[j];
- xr = a[j1];
- xi = -a[j1 + 1];
- yr = a[k1];
- yi = -a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- j1 += m2;
- k1 += 2 * m2;
- xr = a[j1];
- xi = -a[j1 + 1];
- yr = a[k1];
- yi = -a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- j1 += m2;
- k1 -= m2;
- xr = a[j1];
- xi = -a[j1 + 1];
- yr = a[k1];
- yi = -a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- j1 += m2;
- k1 += 2 * m2;
- xr = a[j1];
- xi = -a[j1 + 1];
- yr = a[k1];
- yi = -a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- }
- k1 = 2 * k + ip[k];
- a[k1 + 1] = -a[k1 + 1];
- j1 = k1 + m2;
- k1 = j1 + m2;
- xr = a[j1];
- xi = -a[j1 + 1];
- yr = a[k1];
- yi = -a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- k1 += m2;
- a[k1 + 1] = -a[k1 + 1];
- }
- } else {
- a[1] = -a[1];
- a[m2 + 1] = -a[m2 + 1];
- for (k = 1; k < m; k++) {
- for (j = 0; j < k; j++) {
- j1 = 2 * j + ip[k];
- k1 = 2 * k + ip[j];
- xr = a[j1];
- xi = -a[j1 + 1];
- yr = a[k1];
- yi = -a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- j1 += m2;
- k1 += m2;
- xr = a[j1];
- xi = -a[j1 + 1];
- yr = a[k1];
- yi = -a[k1 + 1];
- a[j1] = yr;
- a[j1 + 1] = yi;
- a[k1] = xr;
- a[k1 + 1] = xi;
- }
- k1 = 2 * k + ip[k];
- a[k1 + 1] = -a[k1 + 1];
- a[k1 + m2 + 1] = -a[k1 + m2 + 1];
- }
- }
-}
-
-
-static void cftfsub(int n, float *a, float *w)
-{
- int j, j1, j2, j3, l;
- float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
-
- l = 2;
- if (n > 8) {
- cft1st(n, a, w);
- l = 8;
- while ((l << 2) < n) {
- cftmdl(n, l, a, w);
- l <<= 2;
- }
- }
- if ((l << 2) == n) {
- for (j = 0; j < l; j += 2) {
- j1 = j + l;
- j2 = j1 + l;
- j3 = j2 + l;
- x0r = a[j] + a[j1];
- x0i = a[j + 1] + a[j1 + 1];
- x1r = a[j] - a[j1];
- x1i = a[j + 1] - a[j1 + 1];
- x2r = a[j2] + a[j3];
- x2i = a[j2 + 1] + a[j3 + 1];
- x3r = a[j2] - a[j3];
- x3i = a[j2 + 1] - a[j3 + 1];
- a[j] = x0r + x2r;
- a[j + 1] = x0i + x2i;
- a[j2] = x0r - x2r;
- a[j2 + 1] = x0i - x2i;
- a[j1] = x1r - x3i;
- a[j1 + 1] = x1i + x3r;
- a[j3] = x1r + x3i;
- a[j3 + 1] = x1i - x3r;
- }
- } else {
- for (j = 0; j < l; j += 2) {
- j1 = j + l;
- x0r = a[j] - a[j1];
- x0i = a[j + 1] - a[j1 + 1];
- a[j] += a[j1];
- a[j + 1] += a[j1 + 1];
- a[j1] = x0r;
- a[j1 + 1] = x0i;
- }
- }
-}
-
-
-static void cftbsub(int n, float *a, float *w)
-{
- int j, j1, j2, j3, l;
- float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
-
- l = 2;
- if (n > 8) {
- cft1st(n, a, w);
- l = 8;
- while ((l << 2) < n) {
- cftmdl(n, l, a, w);
- l <<= 2;
- }
- }
- if ((l << 2) == n) {
- for (j = 0; j < l; j += 2) {
- j1 = j + l;
- j2 = j1 + l;
- j3 = j2 + l;
- x0r = a[j] + a[j1];
- x0i = -a[j + 1] - a[j1 + 1];
- x1r = a[j] - a[j1];
- x1i = -a[j + 1] + a[j1 + 1];
- x2r = a[j2] + a[j3];
- x2i = a[j2 + 1] + a[j3 + 1];
- x3r = a[j2] - a[j3];
- x3i = a[j2 + 1] - a[j3 + 1];
- a[j] = x0r + x2r;
- a[j + 1] = x0i - x2i;
- a[j2] = x0r - x2r;
- a[j2 + 1] = x0i + x2i;
- a[j1] = x1r - x3i;
- a[j1 + 1] = x1i - x3r;
- a[j3] = x1r + x3i;
- a[j3 + 1] = x1i + x3r;
- }
- } else {
- for (j = 0; j < l; j += 2) {
- j1 = j + l;
- x0r = a[j] - a[j1];
- x0i = -a[j + 1] + a[j1 + 1];
- a[j] += a[j1];
- a[j + 1] = -a[j + 1] - a[j1 + 1];
- a[j1] = x0r;
- a[j1 + 1] = x0i;
- }
- }
-}
-
-
-static void cft1st(int n, float *a, float *w)
-{
- int j, k1, k2;
- float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
- float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
-
- x0r = a[0] + a[2];
- x0i = a[1] + a[3];
- x1r = a[0] - a[2];
- x1i = a[1] - a[3];
- x2r = a[4] + a[6];
- x2i = a[5] + a[7];
- x3r = a[4] - a[6];
- x3i = a[5] - a[7];
- a[0] = x0r + x2r;
- a[1] = x0i + x2i;
- a[4] = x0r - x2r;
- a[5] = x0i - x2i;
- a[2] = x1r - x3i;
- a[3] = x1i + x3r;
- a[6] = x1r + x3i;
- a[7] = x1i - x3r;
- wk1r = w[2];
- x0r = a[8] + a[10];
- x0i = a[9] + a[11];
- x1r = a[8] - a[10];
- x1i = a[9] - a[11];
- x2r = a[12] + a[14];
- x2i = a[13] + a[15];
- x3r = a[12] - a[14];
- x3i = a[13] - a[15];
- a[8] = x0r + x2r;
- a[9] = x0i + x2i;
- a[12] = x2i - x0i;
- a[13] = x0r - x2r;
- x0r = x1r - x3i;
- x0i = x1i + x3r;
- a[10] = wk1r * (x0r - x0i);
- a[11] = wk1r * (x0r + x0i);
- x0r = x3i + x1r;
- x0i = x3r - x1i;
- a[14] = wk1r * (x0i - x0r);
- a[15] = wk1r * (x0i + x0r);
- k1 = 0;
- for (j = 16; j < n; j += 16) {
- k1 += 2;
- k2 = 2 * k1;
- wk2r = w[k1];
- wk2i = w[k1 + 1];
- wk1r = w[k2];
- wk1i = w[k2 + 1];
- wk3r = wk1r - 2 * wk2i * wk1i;
- wk3i = 2 * wk2i * wk1r - wk1i;
- x0r = a[j] + a[j + 2];
- x0i = a[j + 1] + a[j + 3];
- x1r = a[j] - a[j + 2];
- x1i = a[j + 1] - a[j + 3];
- x2r = a[j + 4] + a[j + 6];
- x2i = a[j + 5] + a[j + 7];
- x3r = a[j + 4] - a[j + 6];
- x3i = a[j + 5] - a[j + 7];
- a[j] = x0r + x2r;
- a[j + 1] = x0i + x2i;
- x0r -= x2r;
- x0i -= x2i;
- a[j + 4] = wk2r * x0r - wk2i * x0i;
- a[j + 5] = wk2r * x0i + wk2i * x0r;
- x0r = x1r - x3i;
- x0i = x1i + x3r;
- a[j + 2] = wk1r * x0r - wk1i * x0i;
- a[j + 3] = wk1r * x0i + wk1i * x0r;
- x0r = x1r + x3i;
- x0i = x1i - x3r;
- a[j + 6] = wk3r * x0r - wk3i * x0i;
- a[j + 7] = wk3r * x0i + wk3i * x0r;
- wk1r = w[k2 + 2];
- wk1i = w[k2 + 3];
- wk3r = wk1r - 2 * wk2r * wk1i;
- wk3i = 2 * wk2r * wk1r - wk1i;
- x0r = a[j + 8] + a[j + 10];
- x0i = a[j + 9] + a[j + 11];
- x1r = a[j + 8] - a[j + 10];
- x1i = a[j + 9] - a[j + 11];
- x2r = a[j + 12] + a[j + 14];
- x2i = a[j + 13] + a[j + 15];
- x3r = a[j + 12] - a[j + 14];
- x3i = a[j + 13] - a[j + 15];
- a[j + 8] = x0r + x2r;
- a[j + 9] = x0i + x2i;
- x0r -= x2r;
- x0i -= x2i;
- a[j + 12] = -wk2i * x0r - wk2r * x0i;
- a[j + 13] = -wk2i * x0i + wk2r * x0r;
- x0r = x1r - x3i;
- x0i = x1i + x3r;
- a[j + 10] = wk1r * x0r - wk1i * x0i;
- a[j + 11] = wk1r * x0i + wk1i * x0r;
- x0r = x1r + x3i;
- x0i = x1i - x3r;
- a[j + 14] = wk3r * x0r - wk3i * x0i;
- a[j + 15] = wk3r * x0i + wk3i * x0r;
- }
-}
-
-
-static void cftmdl(int n, int l, float *a, float *w)
-{
- int j, j1, j2, j3, k, k1, k2, m, m2;
- float wk1r, wk1i, wk2r, wk2i, wk3r, wk3i;
- float x0r, x0i, x1r, x1i, x2r, x2i, x3r, x3i;
-
- m = l << 2;
- for (j = 0; j < l; j += 2) {
- j1 = j + l;
- j2 = j1 + l;
- j3 = j2 + l;
- x0r = a[j] + a[j1];
- x0i = a[j + 1] + a[j1 + 1];
- x1r = a[j] - a[j1];
- x1i = a[j + 1] - a[j1 + 1];
- x2r = a[j2] + a[j3];
- x2i = a[j2 + 1] + a[j3 + 1];
- x3r = a[j2] - a[j3];
- x3i = a[j2 + 1] - a[j3 + 1];
- a[j] = x0r + x2r;
- a[j + 1] = x0i + x2i;
- a[j2] = x0r - x2r;
- a[j2 + 1] = x0i - x2i;
- a[j1] = x1r - x3i;
- a[j1 + 1] = x1i + x3r;
- a[j3] = x1r + x3i;
- a[j3 + 1] = x1i - x3r;
- }
- wk1r = w[2];
- for (j = m; j < l + m; j += 2) {
- j1 = j + l;
- j2 = j1 + l;
- j3 = j2 + l;
- x0r = a[j] + a[j1];
- x0i = a[j + 1] + a[j1 + 1];
- x1r = a[j] - a[j1];
- x1i = a[j + 1] - a[j1 + 1];
- x2r = a[j2] + a[j3];
- x2i = a[j2 + 1] + a[j3 + 1];
- x3r = a[j2] - a[j3];
- x3i = a[j2 + 1] - a[j3 + 1];
- a[j] = x0r + x2r;
- a[j + 1] = x0i + x2i;
- a[j2] = x2i - x0i;
- a[j2 + 1] = x0r - x2r;
- x0r = x1r - x3i;
- x0i = x1i + x3r;
- a[j1] = wk1r * (x0r - x0i);
- a[j1 + 1] = wk1r * (x0r + x0i);
- x0r = x3i + x1r;
- x0i = x3r - x1i;
- a[j3] = wk1r * (x0i - x0r);
- a[j3 + 1] = wk1r * (x0i + x0r);
- }
- k1 = 0;
- m2 = 2 * m;
- for (k = m2; k < n; k += m2) {
- k1 += 2;
- k2 = 2 * k1;
- wk2r = w[k1];
- wk2i = w[k1 + 1];
- wk1r = w[k2];
- wk1i = w[k2 + 1];
- wk3r = wk1r - 2 * wk2i * wk1i;
- wk3i = 2 * wk2i * wk1r - wk1i;
- for (j = k; j < l + k; j += 2) {
- j1 = j + l;
- j2 = j1 + l;
- j3 = j2 + l;
- x0r = a[j] + a[j1];
- x0i = a[j + 1] + a[j1 + 1];
- x1r = a[j] - a[j1];
- x1i = a[j + 1] - a[j1 + 1];
- x2r = a[j2] + a[j3];
- x2i = a[j2 + 1] + a[j3 + 1];
- x3r = a[j2] - a[j3];
- x3i = a[j2 + 1] - a[j3 + 1];
- a[j] = x0r + x2r;
- a[j + 1] = x0i + x2i;
- x0r -= x2r;
- x0i -= x2i;
- a[j2] = wk2r * x0r - wk2i * x0i;
- a[j2 + 1] = wk2r * x0i + wk2i * x0r;
- x0r = x1r - x3i;
- x0i = x1i + x3r;
- a[j1] = wk1r * x0r - wk1i * x0i;
- a[j1 + 1] = wk1r * x0i + wk1i * x0r;
- x0r = x1r + x3i;
- x0i = x1i - x3r;
- a[j3] = wk3r * x0r - wk3i * x0i;
- a[j3 + 1] = wk3r * x0i + wk3i * x0r;
- }
- wk1r = w[k2 + 2];
- wk1i = w[k2 + 3];
- wk3r = wk1r - 2 * wk2r * wk1i;
- wk3i = 2 * wk2r * wk1r - wk1i;
- for (j = k + m; j < l + (k + m); j += 2) {
- j1 = j + l;
- j2 = j1 + l;
- j3 = j2 + l;
- x0r = a[j] + a[j1];
- x0i = a[j + 1] + a[j1 + 1];
- x1r = a[j] - a[j1];
- x1i = a[j + 1] - a[j1 + 1];
- x2r = a[j2] + a[j3];
- x2i = a[j2 + 1] + a[j3 + 1];
- x3r = a[j2] - a[j3];
- x3i = a[j2 + 1] - a[j3 + 1];
- a[j] = x0r + x2r;
- a[j + 1] = x0i + x2i;
- x0r -= x2r;
- x0i -= x2i;
- a[j2] = -wk2i * x0r - wk2r * x0i;
- a[j2 + 1] = -wk2i * x0i + wk2r * x0r;
- x0r = x1r - x3i;
- x0i = x1i + x3r;
- a[j1] = wk1r * x0r - wk1i * x0i;
- a[j1 + 1] = wk1r * x0i + wk1i * x0r;
- x0r = x1r + x3i;
- x0i = x1i - x3r;
- a[j3] = wk3r * x0r - wk3i * x0i;
- a[j3 + 1] = wk3r * x0i + wk3i * x0r;
- }
- }
-}
-
-
-static void rftfsub(int n, float *a, int nc, float *c)
-{
- int j, k, kk, ks, m;
- float wkr, wki, xr, xi, yr, yi;
-
- m = n >> 1;
- ks = 2 * nc / m;
- kk = 0;
- for (j = 2; j < m; j += 2) {
- k = n - j;
- kk += ks;
- wkr = 0.5f - c[nc - kk];
- wki = c[kk];
- xr = a[j] - a[k];
- xi = a[j + 1] + a[k + 1];
- yr = wkr * xr - wki * xi;
- yi = wkr * xi + wki * xr;
- a[j] -= yr;
- a[j + 1] -= yi;
- a[k] += yr;
- a[k + 1] -= yi;
- }
-}
-
-
-static void rftbsub(int n, float *a, int nc, float *c)
-{
- int j, k, kk, ks, m;
- float wkr, wki, xr, xi, yr, yi;
-
- a[1] = -a[1];
- m = n >> 1;
- ks = 2 * nc / m;
- kk = 0;
- for (j = 2; j < m; j += 2) {
- k = n - j;
- kk += ks;
- wkr = 0.5f - c[nc - kk];
- wki = c[kk];
- xr = a[j] - a[k];
- xi = a[j + 1] + a[k + 1];
- yr = wkr * xr + wki * xi;
- yi = wkr * xi - wki * xr;
- a[j] -= yr;
- a[j + 1] = yi - a[j + 1];
- a[k] += yr;
- a[k + 1] = yi - a[k + 1];
- }
- a[m + 1] = -a[m + 1];
-}
-
-#if 0 // Not used.
-static void dctsub(int n, float *a, int nc, float *c)
-{
- int j, k, kk, ks, m;
- float wkr, wki, xr;
-
- m = n >> 1;
- ks = nc / n;
- kk = 0;
- for (j = 1; j < m; j++) {
- k = n - j;
- kk += ks;
- wkr = c[kk] - c[nc - kk];
- wki = c[kk] + c[nc - kk];
- xr = wki * a[j] - wkr * a[k];
- a[j] = wkr * a[j] + wki * a[k];
- a[k] = xr;
- }
- a[m] *= c[0];
-}
-
-
-static void dstsub(int n, float *a, int nc, float *c)
-{
- int j, k, kk, ks, m;
- float wkr, wki, xr;
-
- m = n >> 1;
- ks = nc / n;
- kk = 0;
- for (j = 1; j < m; j++) {
- k = n - j;
- kk += ks;
- wkr = c[kk] - c[nc - kk];
- wki = c[kk] + c[nc - kk];
- xr = wki * a[k] - wkr * a[j];
- a[k] = wkr * a[k] + wki * a[j];
- a[j] = xr;
- }
- a[m] *= c[0];
-}
-#endif // Not used.
diff --git a/trunk/src/modules/audio_processing/utility/fft4g.h b/trunk/src/modules/audio_processing/utility/fft4g.h
deleted file mode 100644
index 14a52a1..0000000
--- a/trunk/src/modules/audio_processing/utility/fft4g.h
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_FFT4G_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_FFT4G_H_
-
-void WebRtc_rdft(int, int, float *, int *, float *);
-void WebRtc_cdft(int, int, float *, int *, float *);
-
-#endif
diff --git a/trunk/src/modules/audio_processing/utility/ring_buffer.c b/trunk/src/modules/audio_processing/utility/ring_buffer.c
deleted file mode 100644
index 8b2b436..0000000
--- a/trunk/src/modules/audio_processing/utility/ring_buffer.c
+++ /dev/null
@@ -1,271 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// A ring buffer to hold arbitrary data. Provides no thread safety. Unless
-// otherwise specified, functions return 0 on success and -1 on error.
-
-#include "ring_buffer.h"
-
-#include <stddef.h> // size_t
-#include <stdlib.h>
-#include <string.h>
-
-enum Wrap {
- SAME_WRAP,
- DIFF_WRAP
-};
-
-typedef struct {
- size_t read_pos;
- size_t write_pos;
- size_t element_count;
- size_t element_size;
- enum Wrap rw_wrap;
- char* data;
-} buf_t;
-
-// Get address of region(s) from which we can read data.
-// If the region is contiguous, |data_ptr_bytes_2| will be zero.
-// If non-contiguous, |data_ptr_bytes_2| will be the size in bytes of the second
-// region. Returns room available to be read or |element_count|, whichever is
-// smaller.
-static size_t GetBufferReadRegions(buf_t* buf,
- size_t element_count,
- void** data_ptr_1,
- size_t* data_ptr_bytes_1,
- void** data_ptr_2,
- size_t* data_ptr_bytes_2) {
-
- const size_t readable_elements = WebRtc_available_read(buf);
- const size_t read_elements = (readable_elements < element_count ?
- readable_elements : element_count);
- const size_t margin = buf->element_count - buf->read_pos;
-
- // Check to see if read is not contiguous.
- if (read_elements > margin) {
- // Write data in two blocks that wrap the buffer.
- *data_ptr_1 = buf->data + buf->read_pos * buf->element_size;
- *data_ptr_bytes_1 = margin * buf->element_size;
- *data_ptr_2 = buf->data;
- *data_ptr_bytes_2 = (read_elements - margin) * buf->element_size;
- } else {
- *data_ptr_1 = buf->data + buf->read_pos * buf->element_size;
- *data_ptr_bytes_1 = read_elements * buf->element_size;
- *data_ptr_2 = NULL;
- *data_ptr_bytes_2 = 0;
- }
-
- return read_elements;
-}
-
-int WebRtc_CreateBuffer(void** handle,
- size_t element_count,
- size_t element_size) {
- buf_t* self = NULL;
-
- if (handle == NULL) {
- return -1;
- }
-
- self = malloc(sizeof(buf_t));
- if (self == NULL) {
- return -1;
- }
- *handle = self;
-
- self->data = malloc(element_count * element_size);
- if (self->data == NULL) {
- free(self);
- self = NULL;
- return -1;
- }
-
- self->element_count = element_count;
- self->element_size = element_size;
-
- return 0;
-}
-
-int WebRtc_InitBuffer(void* handle) {
- buf_t* self = (buf_t*) handle;
-
- if (self == NULL) {
- return -1;
- }
-
- self->read_pos = 0;
- self->write_pos = 0;
- self->rw_wrap = SAME_WRAP;
-
- // Initialize buffer to zeros
- memset(self->data, 0, self->element_count * self->element_size);
-
- return 0;
-}
-
-int WebRtc_FreeBuffer(void* handle) {
- buf_t* self = (buf_t*) handle;
-
- if (self == NULL) {
- return -1;
- }
-
- free(self->data);
- free(self);
-
- return 0;
-}
-
-size_t WebRtc_ReadBuffer(void* handle,
- void** data_ptr,
- void* data,
- size_t element_count) {
-
- buf_t* self = (buf_t*) handle;
-
- if (self == NULL) {
- return 0;
- }
- if (data == NULL) {
- return 0;
- }
- if (data_ptr == NULL) {
- return 0;
- }
-
- {
- void* buf_ptr_1 = NULL;
- void* buf_ptr_2 = NULL;
- size_t buf_ptr_bytes_1 = 0;
- size_t buf_ptr_bytes_2 = 0;
- const size_t read_count = GetBufferReadRegions(self,
- element_count,
- &buf_ptr_1,
- &buf_ptr_bytes_1,
- &buf_ptr_2,
- &buf_ptr_bytes_2);
-
- if (buf_ptr_bytes_2 > 0) {
- // We have a wrap around when reading the buffer. Copy the buffer data to
- // |data| and point to it.
- memcpy(data, buf_ptr_1, buf_ptr_bytes_1);
- memcpy(((char*) data) + buf_ptr_bytes_1, buf_ptr_2, buf_ptr_bytes_2);
- *data_ptr = data;
- } else {
- *data_ptr = buf_ptr_1;
- }
-
- // Update read position
- WebRtc_MoveReadPtr(handle, (int) read_count);
-
- return read_count;
- }
-}
-
-size_t WebRtc_WriteBuffer(void* handle,
- const void* data,
- size_t element_count) {
-
- buf_t* self = (buf_t*) handle;
-
- if (self == NULL) {
- return 0;
- }
- if (data == NULL) {
- return 0;
- }
-
- {
- const size_t free_elements = WebRtc_available_write(handle);
- const size_t write_elements = (free_elements < element_count ? free_elements
- : element_count);
- size_t n = write_elements;
- const size_t margin = self->element_count - self->write_pos;
-
- if (write_elements > margin) {
- // Buffer wrap around when writing.
- memcpy(self->data + self->write_pos * self->element_size,
- data, margin * self->element_size);
- self->write_pos = 0;
- n -= margin;
- self->rw_wrap = DIFF_WRAP;
- }
- memcpy(self->data + self->write_pos * self->element_size,
- ((const char*) data) + ((write_elements - n) * self->element_size),
- n * self->element_size);
- self->write_pos += n;
-
- return write_elements;
- }
-}
-
-int WebRtc_MoveReadPtr(void* handle, int element_count) {
-
- buf_t* self = (buf_t*) handle;
-
- if (self == NULL) {
- return 0;
- }
-
- {
- // We need to be able to take care of negative changes, hence use "int"
- // instead of "size_t".
- const int free_elements = (int) WebRtc_available_write(handle);
- const int readable_elements = (int) WebRtc_available_read(handle);
- int read_pos = (int) self->read_pos;
-
- if (element_count > readable_elements) {
- element_count = readable_elements;
- }
- if (element_count < -free_elements) {
- element_count = -free_elements;
- }
-
- read_pos += element_count;
- if (read_pos > (int) self->element_count) {
- // Buffer wrap around. Restart read position and wrap indicator.
- read_pos -= (int) self->element_count;
- self->rw_wrap = SAME_WRAP;
- }
- if (read_pos < 0) {
- // Buffer wrap around. Restart read position and wrap indicator.
- read_pos += (int) self->element_count;
- self->rw_wrap = DIFF_WRAP;
- }
-
- self->read_pos = (size_t) read_pos;
-
- return element_count;
- }
-}
-
-size_t WebRtc_available_read(const void* handle) {
- const buf_t* self = (buf_t*) handle;
-
- if (self == NULL) {
- return 0;
- }
-
- if (self->rw_wrap == SAME_WRAP) {
- return self->write_pos - self->read_pos;
- } else {
- return self->element_count - self->read_pos + self->write_pos;
- }
-}
-
-size_t WebRtc_available_write(const void* handle) {
- const buf_t* self = (buf_t*) handle;
-
- if (self == NULL) {
- return 0;
- }
-
- return self->element_count - WebRtc_available_read(handle);
-}
diff --git a/trunk/src/modules/audio_processing/utility/ring_buffer.h b/trunk/src/modules/audio_processing/utility/ring_buffer.h
deleted file mode 100644
index 3c44029..0000000
--- a/trunk/src/modules/audio_processing/utility/ring_buffer.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// A ring buffer to hold arbitrary data. Provides no thread safety. Unless
-// otherwise specified, functions return 0 on success and -1 on error.
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
-
-#include <stddef.h> // size_t
-
-int WebRtc_CreateBuffer(void** handle,
- size_t element_count,
- size_t element_size);
-int WebRtc_InitBuffer(void* handle);
-int WebRtc_FreeBuffer(void* handle);
-
-// Reads data from the buffer. The |data_ptr| will point to the address where
-// it is located. If all |element_count| data are feasible to read without
-// buffer wrap around |data_ptr| will point to the location in the buffer.
-// Otherwise, the data will be copied to |data| (memory allocation done by the
-// user) and |data_ptr| points to the address of |data|. |data_ptr| is only
-// guaranteed to be valid until the next call to WebRtc_WriteBuffer().
-// Returns number of elements read.
-size_t WebRtc_ReadBuffer(void* handle,
- void** data_ptr,
- void* data,
- size_t element_count);
-
-// Writes |data| to buffer and returns the number of elements written.
-size_t WebRtc_WriteBuffer(void* handle, const void* data, size_t element_count);
-
-// Moves the buffer read position and returns the number of elements moved.
-// Positive |element_count| moves the read position towards the write position,
-// that is, flushing the buffer. Negative |element_count| moves the read
-// position away from the the write position, that is, stuffing the buffer.
-// Returns number of elements moved.
-int WebRtc_MoveReadPtr(void* handle, int element_count);
-
-// Returns number of available elements to read.
-size_t WebRtc_available_read(const void* handle);
-
-// Returns number of available elements for write.
-size_t WebRtc_available_write(const void* handle);
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_UTILITY_RING_BUFFER_H_
diff --git a/trunk/src/modules/audio_processing/utility/util.gypi b/trunk/src/modules/audio_processing/utility/util.gypi
deleted file mode 100644
index 3c3024a..0000000
--- a/trunk/src/modules/audio_processing/utility/util.gypi
+++ /dev/null
@@ -1,40 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'apm_util',
- 'type': '<(library)',
- 'dependencies': [
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '.',
- ],
- },
- 'sources': [
- 'delay_estimator.c',
- 'delay_estimator.h',
- 'delay_estimator_wrapper.c',
- 'delay_estimator_wrapper.h',
- 'fft4g.c',
- 'fft4g.h',
- 'ring_buffer.c',
- 'ring_buffer.h',
- ],
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/audio_processing/voice_detection_impl.cc b/trunk/src/modules/audio_processing/voice_detection_impl.cc
deleted file mode 100644
index 50b99a0..0000000
--- a/trunk/src/modules/audio_processing/voice_detection_impl.cc
+++ /dev/null
@@ -1,188 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voice_detection_impl.h"
-
-#include <cassert>
-
-#include "critical_section_wrapper.h"
-#include "webrtc_vad.h"
-
-#include "audio_processing_impl.h"
-#include "audio_buffer.h"
-
-namespace webrtc {
-
-typedef VadInst Handle;
-
-namespace {
-int MapSetting(VoiceDetection::Likelihood likelihood) {
- switch (likelihood) {
- case VoiceDetection::kVeryLowLikelihood:
- return 3;
- case VoiceDetection::kLowLikelihood:
- return 2;
- case VoiceDetection::kModerateLikelihood:
- return 1;
- case VoiceDetection::kHighLikelihood:
- return 0;
- }
- assert(false);
- return -1;
-}
-} // namespace
-
-VoiceDetectionImpl::VoiceDetectionImpl(const AudioProcessingImpl* apm)
- : ProcessingComponent(apm),
- apm_(apm),
- stream_has_voice_(false),
- using_external_vad_(false),
- likelihood_(kLowLikelihood),
- frame_size_ms_(10),
- frame_size_samples_(0) {}
-
-VoiceDetectionImpl::~VoiceDetectionImpl() {}
-
-int VoiceDetectionImpl::ProcessCaptureAudio(AudioBuffer* audio) {
- if (!is_component_enabled()) {
- return apm_->kNoError;
- }
-
- if (using_external_vad_) {
- using_external_vad_ = false;
- return apm_->kNoError;
- }
- assert(audio->samples_per_split_channel() <= 160);
-
- WebRtc_Word16* mixed_data = audio->low_pass_split_data(0);
- if (audio->num_channels() > 1) {
- audio->CopyAndMixLowPass(1);
- mixed_data = audio->mixed_low_pass_data(0);
- }
-
- // TODO(ajm): concatenate data in frame buffer here.
-
- int vad_ret = WebRtcVad_Process(static_cast<Handle*>(handle(0)),
- apm_->split_sample_rate_hz(),
- mixed_data,
- frame_size_samples_);
- if (vad_ret == 0) {
- stream_has_voice_ = false;
- audio->set_activity(AudioFrame::kVadPassive);
- } else if (vad_ret == 1) {
- stream_has_voice_ = true;
- audio->set_activity(AudioFrame::kVadActive);
- } else {
- return apm_->kUnspecifiedError;
- }
-
- return apm_->kNoError;
-}
-
-int VoiceDetectionImpl::Enable(bool enable) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- return EnableComponent(enable);
-}
-
-bool VoiceDetectionImpl::is_enabled() const {
- return is_component_enabled();
-}
-
-int VoiceDetectionImpl::set_stream_has_voice(bool has_voice) {
- using_external_vad_ = true;
- stream_has_voice_ = has_voice;
- return apm_->kNoError;
-}
-
-bool VoiceDetectionImpl::stream_has_voice() const {
- // TODO(ajm): enable this assertion?
- //assert(using_external_vad_ || is_component_enabled());
- return stream_has_voice_;
-}
-
-int VoiceDetectionImpl::set_likelihood(VoiceDetection::Likelihood likelihood) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- if (MapSetting(likelihood) == -1) {
- return apm_->kBadParameterError;
- }
-
- likelihood_ = likelihood;
- return Configure();
-}
-
-VoiceDetection::Likelihood VoiceDetectionImpl::likelihood() const {
- return likelihood_;
-}
-
-int VoiceDetectionImpl::set_frame_size_ms(int size) {
- CriticalSectionScoped crit_scoped(apm_->crit());
- assert(size == 10); // TODO(ajm): remove when supported.
- if (size != 10 &&
- size != 20 &&
- size != 30) {
- return apm_->kBadParameterError;
- }
-
- frame_size_ms_ = size;
-
- return Initialize();
-}
-
-int VoiceDetectionImpl::frame_size_ms() const {
- return frame_size_ms_;
-}
-
-int VoiceDetectionImpl::Initialize() {
- int err = ProcessingComponent::Initialize();
- if (err != apm_->kNoError || !is_component_enabled()) {
- return err;
- }
-
- using_external_vad_ = false;
- frame_size_samples_ = frame_size_ms_ * (apm_->split_sample_rate_hz() / 1000);
- // TODO(ajm): intialize frame buffer here.
-
- return apm_->kNoError;
-}
-
-void* VoiceDetectionImpl::CreateHandle() const {
- Handle* handle = NULL;
- if (WebRtcVad_Create(&handle) != apm_->kNoError) {
- handle = NULL;
- } else {
- assert(handle != NULL);
- }
-
- return handle;
-}
-
-int VoiceDetectionImpl::DestroyHandle(void* handle) const {
- return WebRtcVad_Free(static_cast<Handle*>(handle));
-}
-
-int VoiceDetectionImpl::InitializeHandle(void* handle) const {
- return WebRtcVad_Init(static_cast<Handle*>(handle));
-}
-
-int VoiceDetectionImpl::ConfigureHandle(void* handle) const {
- return WebRtcVad_set_mode(static_cast<Handle*>(handle),
- MapSetting(likelihood_));
-}
-
-int VoiceDetectionImpl::num_handles_required() const {
- return 1;
-}
-
-int VoiceDetectionImpl::GetHandleError(void* handle) const {
- // The VAD has no get_error() function.
- assert(handle != NULL);
- return apm_->kUnspecifiedError;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/audio_processing/voice_detection_impl.h b/trunk/src/modules/audio_processing/voice_detection_impl.h
deleted file mode 100644
index 52d92e0..0000000
--- a/trunk/src/modules/audio_processing/voice_detection_impl.h
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
-#define WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
-
-#include "audio_processing.h"
-#include "processing_component.h"
-
-namespace webrtc {
-class AudioProcessingImpl;
-class AudioBuffer;
-
-class VoiceDetectionImpl : public VoiceDetection,
- public ProcessingComponent {
- public:
- explicit VoiceDetectionImpl(const AudioProcessingImpl* apm);
- virtual ~VoiceDetectionImpl();
-
- int ProcessCaptureAudio(AudioBuffer* audio);
-
- // VoiceDetection implementation.
- virtual bool is_enabled() const;
-
- // ProcessingComponent implementation.
- virtual int Initialize();
-
- private:
- // VoiceDetection implementation.
- virtual int Enable(bool enable);
- virtual int set_stream_has_voice(bool has_voice);
- virtual bool stream_has_voice() const;
- virtual int set_likelihood(Likelihood likelihood);
- virtual Likelihood likelihood() const;
- virtual int set_frame_size_ms(int size);
- virtual int frame_size_ms() const;
-
- // ProcessingComponent implementation.
- virtual void* CreateHandle() const;
- virtual int InitializeHandle(void* handle) const;
- virtual int ConfigureHandle(void* handle) const;
- virtual int DestroyHandle(void* handle) const;
- virtual int num_handles_required() const;
- virtual int GetHandleError(void* handle) const;
-
- const AudioProcessingImpl* apm_;
- bool stream_has_voice_;
- bool using_external_vad_;
- Likelihood likelihood_;
- int frame_size_ms_;
- int frame_size_samples_;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_AUDIO_PROCESSING_VOICE_DETECTION_IMPL_H_
diff --git a/trunk/src/modules/interface/module.h b/trunk/src/modules/interface/module.h
deleted file mode 100644
index 37e5027..0000000
--- a/trunk/src/modules/interface/module.h
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef MODULES_INTERFACE_MODULE_H_
-#define MODULES_INTERFACE_MODULE_H_
-
-#include <assert.h>
-
-#include "typedefs.h"
-
-namespace webrtc {
-
-class Module {
- public:
- // Change the unique identifier of this object.
- virtual int32_t ChangeUniqueId(const int32_t id) = 0;
-
- // Returns the number of milliseconds until the module want a worker
- // thread to call Process.
- virtual int32_t TimeUntilNextProcess() = 0;
-
- // Process any pending tasks such as timeouts.
- virtual int32_t Process() = 0;
-
- protected:
- virtual ~Module() {}
-};
-
-// Reference counted version of the module interface.
-class RefCountedModule : public Module {
- public:
- // Increase the reference count by one.
- // Returns the incremented reference count.
- // TODO(perkj): Make this pure virtual when Chromium have implemented
- // reference counting ADM and Video capture module.
- virtual int32_t AddRef() {
- assert(false && "Not implemented.");
- return 1;
- }
-
- // Decrease the reference count by one.
- // Returns the decreased reference count.
- // Returns 0 if the last reference was just released.
- // When the reference count reach 0 the object will self-destruct.
- // TODO(perkj): Make this pure virtual when Chromium have implemented
- // reference counting ADM and Video capture module.
- virtual int32_t Release() {
- assert(false && "Not implemented.");
- return 1;
- }
-
- protected:
- virtual ~RefCountedModule() {}
-};
-
-} // namespace webrtc
-
-#endif // MODULES_INTERFACE_MODULE_H_
diff --git a/trunk/src/modules/interface/module_common_types.h b/trunk/src/modules/interface/module_common_types.h
deleted file mode 100644
index 2e1beaa..0000000
--- a/trunk/src/modules/interface/module_common_types.h
+++ /dev/null
@@ -1,1047 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef MODULE_COMMON_TYPES_H
-#define MODULE_COMMON_TYPES_H
-
-#include <cstring> // memcpy
-#include <assert.h>
-
-#include "typedefs.h"
-#include "common_types.h"
-
-#ifdef _WIN32
- #pragma warning(disable:4351) // remove warning "new behavior: elements of array
- // 'array' will be default initialized"
-#endif
-
-namespace webrtc
-{
-struct RTPHeader
-{
- bool markerBit;
- WebRtc_UWord8 payloadType;
- WebRtc_UWord16 sequenceNumber;
- WebRtc_UWord32 timestamp;
- WebRtc_UWord32 ssrc;
- WebRtc_UWord8 numCSRCs;
- WebRtc_UWord32 arrOfCSRCs[kRtpCsrcSize];
- WebRtc_UWord8 paddingLength;
- WebRtc_UWord16 headerLength;
-};
-
-struct RTPHeaderExtension
-{
- WebRtc_Word32 transmissionTimeOffset;
-};
-
-struct RTPAudioHeader
-{
- WebRtc_UWord8 numEnergy; // number of valid entries in arrOfEnergy
- WebRtc_UWord8 arrOfEnergy[kRtpCsrcSize]; // one energy byte (0-9) per channel
- bool isCNG; // is this CNG
- WebRtc_UWord8 channel; // number of channels 2 = stereo
-};
-
-enum {kNoPictureId = -1};
-enum {kNoTl0PicIdx = -1};
-enum {kNoTemporalIdx = -1};
-enum {kNoKeyIdx = -1};
-enum {kNoSimulcastIdx = 0};
-
-struct RTPVideoHeaderVP8
-{
- void InitRTPVideoHeaderVP8()
- {
- nonReference = false;
- pictureId = kNoPictureId;
- tl0PicIdx = kNoTl0PicIdx;
- temporalIdx = kNoTemporalIdx;
- layerSync = false;
- keyIdx = kNoKeyIdx;
- partitionId = 0;
- beginningOfPartition = false;
- frameWidth = 0;
- frameHeight = 0;
- }
-
- bool nonReference; // Frame is discardable.
- WebRtc_Word16 pictureId; // Picture ID index, 15 bits;
- // kNoPictureId if PictureID does not exist.
- WebRtc_Word16 tl0PicIdx; // TL0PIC_IDX, 8 bits;
- // kNoTl0PicIdx means no value provided.
- WebRtc_Word8 temporalIdx; // Temporal layer index, or kNoTemporalIdx.
- bool layerSync; // This frame is a layer sync frame.
- // Disabled if temporalIdx == kNoTemporalIdx.
- int keyIdx; // 5 bits; kNoKeyIdx means not used.
- int partitionId; // VP8 partition ID
- bool beginningOfPartition; // True if this packet is the first
- // in a VP8 partition. Otherwise false
- int frameWidth; // Exists for key frames.
- int frameHeight; // Exists for key frames.
-};
-union RTPVideoTypeHeader
-{
- RTPVideoHeaderVP8 VP8;
-};
-
-enum RTPVideoCodecTypes
-{
- kRTPVideoGeneric = 0,
- kRTPVideoVP8 = 8,
- kRTPVideoNoVideo = 10,
- kRTPVideoFEC = 11,
- kRTPVideoI420 = 12
-};
-struct RTPVideoHeader
-{
- WebRtc_UWord16 width; // size
- WebRtc_UWord16 height;
-
- bool isFirstPacket; // first packet in frame
- WebRtc_UWord8 simulcastIdx; // Index if the simulcast encoder creating
- // this frame, 0 if not using simulcast.
- RTPVideoCodecTypes codec;
- RTPVideoTypeHeader codecHeader;
-};
-union RTPTypeHeader
-{
- RTPAudioHeader Audio;
- RTPVideoHeader Video;
-};
-
-struct WebRtcRTPHeader
-{
- RTPHeader header;
- FrameType frameType;
- RTPTypeHeader type;
- RTPHeaderExtension extension;
-};
-
-class RTPFragmentationHeader
-{
-public:
- RTPFragmentationHeader() :
- fragmentationVectorSize(0),
- fragmentationOffset(NULL),
- fragmentationLength(NULL),
- fragmentationTimeDiff(NULL),
- fragmentationPlType(NULL)
- {};
-
- ~RTPFragmentationHeader()
- {
- delete [] fragmentationOffset;
- delete [] fragmentationLength;
- delete [] fragmentationTimeDiff;
- delete [] fragmentationPlType;
- }
-
- RTPFragmentationHeader& operator=(const RTPFragmentationHeader& header)
- {
- if(this == &header)
- {
- return *this;
- }
-
- if(header.fragmentationVectorSize != fragmentationVectorSize)
- {
- // new size of vectors
-
- // delete old
- delete [] fragmentationOffset;
- fragmentationOffset = NULL;
- delete [] fragmentationLength;
- fragmentationLength = NULL;
- delete [] fragmentationTimeDiff;
- fragmentationTimeDiff = NULL;
- delete [] fragmentationPlType;
- fragmentationPlType = NULL;
-
- if(header.fragmentationVectorSize > 0)
- {
- // allocate new
- if(header.fragmentationOffset)
- {
- fragmentationOffset = new WebRtc_UWord32[header.fragmentationVectorSize];
- }
- if(header.fragmentationLength)
- {
- fragmentationLength = new WebRtc_UWord32[header.fragmentationVectorSize];
- }
- if(header.fragmentationTimeDiff)
- {
- fragmentationTimeDiff = new WebRtc_UWord16[header.fragmentationVectorSize];
- }
- if(header.fragmentationPlType)
- {
- fragmentationPlType = new WebRtc_UWord8[header.fragmentationVectorSize];
- }
- }
- // set new size
- fragmentationVectorSize = header.fragmentationVectorSize;
- }
-
- if(header.fragmentationVectorSize > 0)
- {
- // copy values
- if(header.fragmentationOffset)
- {
- memcpy(fragmentationOffset, header.fragmentationOffset,
- header.fragmentationVectorSize * sizeof(WebRtc_UWord32));
- }
- if(header.fragmentationLength)
- {
- memcpy(fragmentationLength, header.fragmentationLength,
- header.fragmentationVectorSize * sizeof(WebRtc_UWord32));
- }
- if(header.fragmentationTimeDiff)
- {
- memcpy(fragmentationTimeDiff, header.fragmentationTimeDiff,
- header.fragmentationVectorSize * sizeof(WebRtc_UWord16));
- }
- if(header.fragmentationPlType)
- {
- memcpy(fragmentationPlType, header.fragmentationPlType,
- header.fragmentationVectorSize * sizeof(WebRtc_UWord8));
- }
- }
- return *this;
- }
- void VerifyAndAllocateFragmentationHeader( const WebRtc_UWord16 size)
- {
- if( fragmentationVectorSize < size)
- {
- WebRtc_UWord16 oldVectorSize = fragmentationVectorSize;
- {
- // offset
- WebRtc_UWord32* oldOffsets = fragmentationOffset;
- fragmentationOffset = new WebRtc_UWord32[size];
- memset(fragmentationOffset+oldVectorSize, 0,
- sizeof(WebRtc_UWord32)*(size-oldVectorSize));
- // copy old values
- memcpy(fragmentationOffset,oldOffsets, sizeof(WebRtc_UWord32) * oldVectorSize);
- delete[] oldOffsets;
- }
- // length
- {
- WebRtc_UWord32* oldLengths = fragmentationLength;
- fragmentationLength = new WebRtc_UWord32[size];
- memset(fragmentationLength+oldVectorSize, 0,
- sizeof(WebRtc_UWord32) * (size- oldVectorSize));
- memcpy(fragmentationLength, oldLengths,
- sizeof(WebRtc_UWord32) * oldVectorSize);
- delete[] oldLengths;
- }
- // time diff
- {
- WebRtc_UWord16* oldTimeDiffs = fragmentationTimeDiff;
- fragmentationTimeDiff = new WebRtc_UWord16[size];
- memset(fragmentationTimeDiff+oldVectorSize, 0,
- sizeof(WebRtc_UWord16) * (size- oldVectorSize));
- memcpy(fragmentationTimeDiff, oldTimeDiffs,
- sizeof(WebRtc_UWord16) * oldVectorSize);
- delete[] oldTimeDiffs;
- }
- // payload type
- {
- WebRtc_UWord8* oldTimePlTypes = fragmentationPlType;
- fragmentationPlType = new WebRtc_UWord8[size];
- memset(fragmentationPlType+oldVectorSize, 0,
- sizeof(WebRtc_UWord8) * (size- oldVectorSize));
- memcpy(fragmentationPlType, oldTimePlTypes,
- sizeof(WebRtc_UWord8) * oldVectorSize);
- delete[] oldTimePlTypes;
- }
- fragmentationVectorSize = size;
- }
- }
-
- WebRtc_UWord16 fragmentationVectorSize; // Number of fragmentations
- WebRtc_UWord32* fragmentationOffset; // Offset of pointer to data for each fragm.
- WebRtc_UWord32* fragmentationLength; // Data size for each fragmentation
- WebRtc_UWord16* fragmentationTimeDiff; // Timestamp difference relative "now" for
- // each fragmentation
- WebRtc_UWord8* fragmentationPlType; // Payload type of each fragmentation
-};
-
-struct RTCPVoIPMetric
-{
- // RFC 3611 4.7
- WebRtc_UWord8 lossRate;
- WebRtc_UWord8 discardRate;
- WebRtc_UWord8 burstDensity;
- WebRtc_UWord8 gapDensity;
- WebRtc_UWord16 burstDuration;
- WebRtc_UWord16 gapDuration;
- WebRtc_UWord16 roundTripDelay;
- WebRtc_UWord16 endSystemDelay;
- WebRtc_UWord8 signalLevel;
- WebRtc_UWord8 noiseLevel;
- WebRtc_UWord8 RERL;
- WebRtc_UWord8 Gmin;
- WebRtc_UWord8 Rfactor;
- WebRtc_UWord8 extRfactor;
- WebRtc_UWord8 MOSLQ;
- WebRtc_UWord8 MOSCQ;
- WebRtc_UWord8 RXconfig;
- WebRtc_UWord16 JBnominal;
- WebRtc_UWord16 JBmax;
- WebRtc_UWord16 JBabsMax;
-};
-
-// class describing a complete, or parts of an encoded frame.
-class EncodedVideoData
-{
-public:
- EncodedVideoData() :
- completeFrame(false),
- missingFrame(false),
- payloadData(NULL),
- payloadSize(0),
- bufferSize(0)
- {};
-
- EncodedVideoData(const EncodedVideoData& data)
- {
- payloadType = data.payloadType;
- timeStamp = data.timeStamp;
- renderTimeMs = data.renderTimeMs;
- encodedWidth = data.encodedWidth;
- encodedHeight = data.encodedHeight;
- completeFrame = data.completeFrame;
- missingFrame = data.missingFrame;
- payloadSize = data.payloadSize;
- fragmentationHeader = data.fragmentationHeader;
- frameType = data.frameType;
- codec = data.codec;
- if (data.payloadSize > 0)
- {
- payloadData = new WebRtc_UWord8[data.payloadSize];
- memcpy(payloadData, data.payloadData, data.payloadSize);
- }
- else
- {
- payloadData = NULL;
- }
- }
-
-
- ~EncodedVideoData()
- {
- delete [] payloadData;
- };
-
- EncodedVideoData& operator=(const EncodedVideoData& data)
- {
- if (this == &data)
- {
- return *this;
- }
- payloadType = data.payloadType;
- timeStamp = data.timeStamp;
- renderTimeMs = data.renderTimeMs;
- encodedWidth = data.encodedWidth;
- encodedHeight = data.encodedHeight;
- completeFrame = data.completeFrame;
- missingFrame = data.missingFrame;
- payloadSize = data.payloadSize;
- fragmentationHeader = data.fragmentationHeader;
- frameType = data.frameType;
- codec = data.codec;
- if (data.payloadSize > 0)
- {
- delete [] payloadData;
- payloadData = new WebRtc_UWord8[data.payloadSize];
- memcpy(payloadData, data.payloadData, data.payloadSize);
- bufferSize = data.payloadSize;
- }
- return *this;
- };
- void VerifyAndAllocate( const WebRtc_UWord32 size)
- {
- if (bufferSize < size)
- {
- WebRtc_UWord8* oldPayload = payloadData;
- payloadData = new WebRtc_UWord8[size];
- memcpy(payloadData, oldPayload, sizeof(WebRtc_UWord8) * payloadSize);
-
- bufferSize = size;
- delete[] oldPayload;
- }
- }
-
- WebRtc_UWord8 payloadType;
- WebRtc_UWord32 timeStamp;
- WebRtc_Word64 renderTimeMs;
- WebRtc_UWord32 encodedWidth;
- WebRtc_UWord32 encodedHeight;
- bool completeFrame;
- bool missingFrame;
- WebRtc_UWord8* payloadData;
- WebRtc_UWord32 payloadSize;
- WebRtc_UWord32 bufferSize;
- RTPFragmentationHeader fragmentationHeader;
- FrameType frameType;
- VideoCodecType codec;
-};
-
-struct VideoContentMetrics {
- VideoContentMetrics()
- : motion_magnitude(0.0f),
- spatial_pred_err(0.0f),
- spatial_pred_err_h(0.0f),
- spatial_pred_err_v(0.0f) {
- }
-
- void Reset() {
- motion_magnitude = 0.0f;
- spatial_pred_err = 0.0f;
- spatial_pred_err_h = 0.0f;
- spatial_pred_err_v = 0.0f;
- }
- float motion_magnitude;
- float spatial_pred_err;
- float spatial_pred_err_h;
- float spatial_pred_err_v;
-};
-
-/*************************************************
- *
- * VideoFrame class
- *
- * The VideoFrame class allows storing and
- * handling of video frames.
- *
- *
- *************************************************/
-class VideoFrame
-{
-public:
- VideoFrame();
- ~VideoFrame();
- /**
- * Verifies that current allocated buffer size is larger than or equal to the input size.
- * If the current buffer size is smaller, a new allocation is made and the old buffer data
- * is copied to the new buffer.
- * Buffer size is updated to minimumSize.
- */
- WebRtc_Word32 VerifyAndAllocate(const WebRtc_UWord32 minimumSize);
- /**
- * Update length of data buffer in frame. Function verifies that new length is less or
- * equal to allocated size.
- */
- WebRtc_Word32 SetLength(const WebRtc_UWord32 newLength);
- /*
- * Swap buffer and size data
- */
- WebRtc_Word32 Swap(WebRtc_UWord8*& newMemory,
- WebRtc_UWord32& newLength,
- WebRtc_UWord32& newSize);
- /*
- * Swap buffer and size data
- */
- WebRtc_Word32 SwapFrame(VideoFrame& videoFrame);
- /**
- * Copy buffer: If newLength is bigger than allocated size, a new buffer of size length
- * is allocated.
- */
- WebRtc_Word32 CopyFrame(const VideoFrame& videoFrame);
- /**
- * Copy buffer: If newLength is bigger than allocated size, a new buffer of size length
- * is allocated.
- */
- WebRtc_Word32 CopyFrame(WebRtc_UWord32 length, const WebRtc_UWord8* sourceBuffer);
- /**
- * Delete VideoFrame and resets members to zero
- */
- void Free();
- /**
- * Set frame timestamp (90kHz)
- */
- void SetTimeStamp(const WebRtc_UWord32 timeStamp) {_timeStamp = timeStamp;}
- /**
- * Get pointer to frame buffer
- */
- WebRtc_UWord8* Buffer() const {return _buffer;}
-
- WebRtc_UWord8*& Buffer() {return _buffer;}
-
- /**
- * Get allocated buffer size
- */
- WebRtc_UWord32 Size() const {return _bufferSize;}
- /**
- * Get frame length
- */
- WebRtc_UWord32 Length() const {return _bufferLength;}
- /**
- * Get frame timestamp (90kHz)
- */
- WebRtc_UWord32 TimeStamp() const {return _timeStamp;}
- /**
- * Get frame width
- */
- WebRtc_UWord32 Width() const {return _width;}
- /**
- * Get frame height
- */
- WebRtc_UWord32 Height() const {return _height;}
- /**
- * Set frame width
- */
- void SetWidth(const WebRtc_UWord32 width) {_width = width;}
- /**
- * Set frame height
- */
- void SetHeight(const WebRtc_UWord32 height) {_height = height;}
- /**
- * Set render time in miliseconds
- */
- void SetRenderTime(const WebRtc_Word64 renderTimeMs) {_renderTimeMs = renderTimeMs;}
- /**
- * Get render time in miliseconds
- */
- WebRtc_Word64 RenderTimeMs() const {return _renderTimeMs;}
-
-private:
- void Set(WebRtc_UWord8* buffer,
- WebRtc_UWord32 size,
- WebRtc_UWord32 length,
- WebRtc_UWord32 timeStamp);
-
- WebRtc_UWord8* _buffer; // Pointer to frame buffer
- WebRtc_UWord32 _bufferSize; // Allocated buffer size
- WebRtc_UWord32 _bufferLength; // Length (in bytes) of buffer
- WebRtc_UWord32 _timeStamp; // Timestamp of frame (90kHz)
- WebRtc_UWord32 _width;
- WebRtc_UWord32 _height;
- WebRtc_Word64 _renderTimeMs;
-}; // end of VideoFrame class declaration
-
-// inline implementation of VideoFrame class:
-inline
-VideoFrame::VideoFrame():
- _buffer(0),
- _bufferSize(0),
- _bufferLength(0),
- _timeStamp(0),
- _width(0),
- _height(0),
- _renderTimeMs(0)
-{
- //
-}
-inline
-VideoFrame::~VideoFrame()
-{
- if(_buffer)
- {
- delete [] _buffer;
- _buffer = NULL;
- }
-}
-
-
-inline
-WebRtc_Word32
-VideoFrame::VerifyAndAllocate(const WebRtc_UWord32 minimumSize)
-{
- if (minimumSize < 1)
- {
- return -1;
- }
- if(minimumSize > _bufferSize)
- {
- // create buffer of sufficient size
- WebRtc_UWord8* newBufferBuffer = new WebRtc_UWord8[minimumSize];
- if(_buffer)
- {
- // copy old data
- memcpy(newBufferBuffer, _buffer, _bufferSize);
- delete [] _buffer;
- }
- else
- {
- memset(newBufferBuffer, 0, minimumSize * sizeof(WebRtc_UWord8));
- }
- _buffer = newBufferBuffer;
- _bufferSize = minimumSize;
- }
- return 0;
-}
-
-inline
-WebRtc_Word32
-VideoFrame::SetLength(const WebRtc_UWord32 newLength)
-{
- if (newLength >_bufferSize )
- { // can't accomodate new value
- return -1;
- }
- _bufferLength = newLength;
- return 0;
-}
-
-inline
-WebRtc_Word32
-VideoFrame::SwapFrame(VideoFrame& videoFrame)
-{
- WebRtc_UWord32 tmpTimeStamp = _timeStamp;
- WebRtc_UWord32 tmpWidth = _width;
- WebRtc_UWord32 tmpHeight = _height;
- WebRtc_Word64 tmpRenderTime = _renderTimeMs;
-
- _timeStamp = videoFrame._timeStamp;
- _width = videoFrame._width;
- _height = videoFrame._height;
- _renderTimeMs = videoFrame._renderTimeMs;
-
- videoFrame._timeStamp = tmpTimeStamp;
- videoFrame._width = tmpWidth;
- videoFrame._height = tmpHeight;
- videoFrame._renderTimeMs = tmpRenderTime;
-
- return Swap(videoFrame._buffer, videoFrame._bufferLength, videoFrame._bufferSize);
-}
-
-inline
-WebRtc_Word32
-VideoFrame::Swap(WebRtc_UWord8*& newMemory, WebRtc_UWord32& newLength, WebRtc_UWord32& newSize)
-{
- WebRtc_UWord8* tmpBuffer = _buffer;
- WebRtc_UWord32 tmpLength = _bufferLength;
- WebRtc_UWord32 tmpSize = _bufferSize;
- _buffer = newMemory;
- _bufferLength = newLength;
- _bufferSize = newSize;
- newMemory = tmpBuffer;
- newLength = tmpLength;
- newSize = tmpSize;
- return 0;
-}
-
-inline
-WebRtc_Word32
-VideoFrame::CopyFrame(WebRtc_UWord32 length, const WebRtc_UWord8* sourceBuffer)
-{
- if (length > _bufferSize)
- {
- WebRtc_Word32 ret = VerifyAndAllocate(length);
- if (ret < 0)
- {
- return ret;
- }
- }
- memcpy(_buffer, sourceBuffer, length);
- _bufferLength = length;
- return 0;
-}
-
-inline
-WebRtc_Word32
-VideoFrame::CopyFrame(const VideoFrame& videoFrame)
-{
- if(CopyFrame(videoFrame.Length(), videoFrame.Buffer()) != 0)
- {
- return -1;
- }
- _timeStamp = videoFrame._timeStamp;
- _width = videoFrame._width;
- _height = videoFrame._height;
- _renderTimeMs = videoFrame._renderTimeMs;
- return 0;
-}
-
-inline
-void
-VideoFrame::Free()
-{
- _timeStamp = 0;
- _bufferLength = 0;
- _bufferSize = 0;
- _height = 0;
- _width = 0;
- _renderTimeMs = 0;
-
- if(_buffer)
- {
- delete [] _buffer;
- _buffer = NULL;
- }
-}
-
-
-/*************************************************
- *
- * AudioFrame class
- *
- * The AudioFrame class holds up to 60 ms wideband
- * audio. It allows for adding and subtracting frames
- * while keeping track of the resulting states.
- *
- * Note
- * - The +operator assume that you would never add
- * exact opposite frames when deciding the resulting
- * state. To do this use the -operator.
- *
- * - _audioChannel of 1 indicated mono, and 2
- * indicates stereo.
- *
- * - _payloadDataLengthInSamples is the number of
- * samples per channel. Therefore, the total
- * number of samples in _payloadData is
- * (_payloadDataLengthInSamples * _audioChannel).
- *
- * - Stereo data is stored in interleaved fashion
- * starting with the left channel.
- *
- *************************************************/
-class AudioFrame
-{
-public:
- enum{kMaxAudioFrameSizeSamples = 3840}; // stereo 32KHz 60ms 2*32*60
-
- enum VADActivity
- {
- kVadActive = 0,
- kVadPassive = 1,
- kVadUnknown = 2
- };
- enum SpeechType
- {
- kNormalSpeech = 0,
- kPLC = 1,
- kCNG = 2,
- kPLCCNG = 3,
- kUndefined = 4
- };
-
- AudioFrame();
- virtual ~AudioFrame();
-
- WebRtc_Word32 UpdateFrame(
- const WebRtc_Word32 id,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_Word16* payloadData,
- const WebRtc_UWord16 payloadDataLengthInSamples,
- const int frequencyInHz,
- const SpeechType speechType,
- const VADActivity vadActivity,
- const WebRtc_UWord8 audioChannel = 1,
- const WebRtc_Word32 volume = -1,
- const WebRtc_Word32 energy = -1);
-
- AudioFrame& Append(const AudioFrame& rhs);
-
- void Mute();
-
- AudioFrame& operator=(const AudioFrame& rhs);
- AudioFrame& operator>>=(const WebRtc_Word32 rhs);
- AudioFrame& operator+=(const AudioFrame& rhs);
- AudioFrame& operator-=(const AudioFrame& rhs);
-
- WebRtc_Word32 _id;
- WebRtc_UWord32 _timeStamp;
-
- // Supporting Stereo, stereo samples are interleaved
- WebRtc_Word16 _payloadData[kMaxAudioFrameSizeSamples];
- WebRtc_UWord16 _payloadDataLengthInSamples;
- int _frequencyInHz;
- WebRtc_UWord8 _audioChannel;
- SpeechType _speechType;
- VADActivity _vadActivity;
-
- WebRtc_UWord32 _energy;
- WebRtc_Word32 _volume;
-};
-
-inline
-AudioFrame::AudioFrame()
- :
- _id(-1),
- _timeStamp(0),
- _payloadData(),
- _payloadDataLengthInSamples(0),
- _frequencyInHz(0),
- _audioChannel(1),
- _speechType(kUndefined),
- _vadActivity(kVadUnknown),
- _energy(0xffffffff),
- _volume(0xffffffff)
-{
-}
-
-inline
-AudioFrame::~AudioFrame()
-{
-}
-
-inline
-WebRtc_Word32
-AudioFrame::UpdateFrame(
- const WebRtc_Word32 id,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_Word16* payloadData,
- const WebRtc_UWord16 payloadDataLengthInSamples,
- const int frequencyInHz,
- const SpeechType speechType,
- const VADActivity vadActivity,
- const WebRtc_UWord8 audioChannel,
- const WebRtc_Word32 volume,
- const WebRtc_Word32 energy)
-{
- _id = id;
- _timeStamp = timeStamp;
- _frequencyInHz = frequencyInHz;
- _speechType = speechType;
- _vadActivity = vadActivity;
- _volume = volume;
- _audioChannel = audioChannel;
- _energy = energy;
-
- if((payloadDataLengthInSamples > kMaxAudioFrameSizeSamples) ||
- (audioChannel > 2) || (audioChannel < 1))
- {
- _payloadDataLengthInSamples = 0;
- return -1;
- }
- _payloadDataLengthInSamples = payloadDataLengthInSamples;
- if(payloadData != NULL)
- {
- memcpy(_payloadData, payloadData, sizeof(WebRtc_Word16) *
- payloadDataLengthInSamples * _audioChannel);
- }
- else
- {
- memset(_payloadData,0,sizeof(WebRtc_Word16) *
- payloadDataLengthInSamples * _audioChannel);
- }
- return 0;
-}
-
-inline
-void
-AudioFrame::Mute()
-{
- memset(_payloadData, 0, _payloadDataLengthInSamples * sizeof(WebRtc_Word16));
-}
-
-inline
-AudioFrame&
-AudioFrame::operator=(const AudioFrame& rhs)
-{
- // Sanity Check
- if((rhs._payloadDataLengthInSamples > kMaxAudioFrameSizeSamples) ||
- (rhs._audioChannel > 2) ||
- (rhs._audioChannel < 1))
- {
- return *this;
- }
- if(this == &rhs)
- {
- return *this;
- }
- _id = rhs._id;
- _timeStamp = rhs._timeStamp;
- _frequencyInHz = rhs._frequencyInHz;
- _speechType = rhs._speechType;
- _vadActivity = rhs._vadActivity;
- _volume = rhs._volume;
- _audioChannel = rhs._audioChannel;
- _energy = rhs._energy;
-
- _payloadDataLengthInSamples = rhs._payloadDataLengthInSamples;
- memcpy(_payloadData, rhs._payloadData,
- sizeof(WebRtc_Word16) * rhs._payloadDataLengthInSamples * _audioChannel);
-
- return *this;
-}
-
-inline
-AudioFrame&
-AudioFrame::operator>>=(const WebRtc_Word32 rhs)
-{
- assert((_audioChannel > 0) && (_audioChannel < 3));
- if((_audioChannel > 2) ||
- (_audioChannel < 1))
- {
- return *this;
- }
- for(WebRtc_UWord16 i = 0; i < _payloadDataLengthInSamples * _audioChannel; i++)
- {
- _payloadData[i] = WebRtc_Word16(_payloadData[i] >> rhs);
- }
- return *this;
-}
-
-inline
-AudioFrame&
-AudioFrame::Append(const AudioFrame& rhs)
-{
- // Sanity check
- assert((_audioChannel > 0) && (_audioChannel < 3));
- if((_audioChannel > 2) ||
- (_audioChannel < 1))
- {
- return *this;
- }
- if(_audioChannel != rhs._audioChannel)
- {
- return *this;
- }
- if((_vadActivity == kVadActive) ||
- rhs._vadActivity == kVadActive)
- {
- _vadActivity = kVadActive;
- }
- else if((_vadActivity == kVadUnknown) ||
- rhs._vadActivity == kVadUnknown)
- {
- _vadActivity = kVadUnknown;
- }
- if(_speechType != rhs._speechType)
- {
- _speechType = kUndefined;
- }
-
- WebRtc_UWord16 offset = _payloadDataLengthInSamples * _audioChannel;
- for(WebRtc_UWord16 i = 0;
- i < rhs._payloadDataLengthInSamples * rhs._audioChannel;
- i++)
- {
- _payloadData[offset+i] = rhs._payloadData[i];
- }
- _payloadDataLengthInSamples += rhs._payloadDataLengthInSamples;
- return *this;
-}
-
-// merge vectors
-inline
-AudioFrame&
-AudioFrame::operator+=(const AudioFrame& rhs)
-{
- // Sanity check
- assert((_audioChannel > 0) && (_audioChannel < 3));
- if((_audioChannel > 2) ||
- (_audioChannel < 1))
- {
- return *this;
- }
- if(_audioChannel != rhs._audioChannel)
- {
- return *this;
- }
- bool noPrevData = false;
- if(_payloadDataLengthInSamples != rhs._payloadDataLengthInSamples)
- {
- if(_payloadDataLengthInSamples == 0)
- {
- // special case we have no data to start with
- _payloadDataLengthInSamples = rhs._payloadDataLengthInSamples;
- noPrevData = true;
- } else
- {
- return *this;
- }
- }
-
- if((_vadActivity == kVadActive) ||
- rhs._vadActivity == kVadActive)
- {
- _vadActivity = kVadActive;
- }
- else if((_vadActivity == kVadUnknown) ||
- rhs._vadActivity == kVadUnknown)
- {
- _vadActivity = kVadUnknown;
- }
-
- if(_speechType != rhs._speechType)
- {
- _speechType = kUndefined;
- }
-
- if(noPrevData)
- {
- memcpy(_payloadData, rhs._payloadData,
- sizeof(WebRtc_Word16) * rhs._payloadDataLengthInSamples * _audioChannel);
- } else
- {
- // IMPROVEMENT this can be done very fast in assembly
- for(WebRtc_UWord16 i = 0; i < _payloadDataLengthInSamples * _audioChannel; i++)
- {
- WebRtc_Word32 wrapGuard = (WebRtc_Word32)_payloadData[i] +
- (WebRtc_Word32)rhs._payloadData[i];
- if(wrapGuard < -32768)
- {
- _payloadData[i] = -32768;
- }else if(wrapGuard > 32767)
- {
- _payloadData[i] = 32767;
- }else
- {
- _payloadData[i] = (WebRtc_Word16)wrapGuard;
- }
- }
- }
- _energy = 0xffffffff;
- _volume = 0xffffffff;
- return *this;
-}
-
-inline
-AudioFrame&
-AudioFrame::operator-=(const AudioFrame& rhs)
-{
- // Sanity check
- assert((_audioChannel > 0) && (_audioChannel < 3));
- if((_audioChannel > 2)||
- (_audioChannel < 1))
- {
- return *this;
- }
- if((_payloadDataLengthInSamples != rhs._payloadDataLengthInSamples) ||
- (_audioChannel != rhs._audioChannel))
- {
- return *this;
- }
- if((_vadActivity != kVadPassive) ||
- rhs._vadActivity != kVadPassive)
- {
- _vadActivity = kVadUnknown;
- }
- _speechType = kUndefined;
-
- for(WebRtc_UWord16 i = 0; i < _payloadDataLengthInSamples * _audioChannel; i++)
- {
- WebRtc_Word32 wrapGuard = (WebRtc_Word32)_payloadData[i] -
- (WebRtc_Word32)rhs._payloadData[i];
- if(wrapGuard < -32768)
- {
- _payloadData[i] = -32768;
- }
- else if(wrapGuard > 32767)
- {
- _payloadData[i] = 32767;
- }
- else
- {
- _payloadData[i] = (WebRtc_Word16)wrapGuard;
- }
- }
- _energy = 0xffffffff;
- _volume = 0xffffffff;
- return *this;
-}
-
-} // namespace webrtc
-
-#endif // MODULE_COMMON_TYPES_H
diff --git a/trunk/src/modules/media_file/OWNERS b/trunk/src/modules/media_file/OWNERS
deleted file mode 100644
index 2cc47e4..0000000
--- a/trunk/src/modules/media_file/OWNERS
+++ /dev/null
@@ -1,4 +0,0 @@
-pwestin@webrtc.org
-mflodman@webrtc.org
-perkj@webrtc.org
-niklas.enbom@webrtc.org
\ No newline at end of file
diff --git a/trunk/src/modules/media_file/interface/media_file.h b/trunk/src/modules/media_file/interface/media_file.h
deleted file mode 100644
index bc857ad..0000000
--- a/trunk/src/modules/media_file/interface/media_file.h
+++ /dev/null
@@ -1,227 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
-#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
-
-#include "common_types.h"
-#include "typedefs.h"
-#include "module.h"
-#include "module_common_types.h"
-#include "media_file_defines.h"
-
-namespace webrtc {
-class MediaFile : public Module
-{
-public:
- // Factory method. Constructor disabled. id is the identifier for the
- // MediaFile instance.
- static MediaFile* CreateMediaFile(const WebRtc_Word32 id);
- static void DestroyMediaFile(MediaFile* module);
-
- // Set the MediaFile instance identifier.
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0;
-
- // Put 10-60ms of audio data from file into the audioBuffer depending on
- // codec frame size. dataLengthInBytes is both an input and output
- // parameter. As input parameter it indicates the size of audioBuffer.
- // As output parameter it indicates the number of bytes written to
- // audioBuffer.
- // Note: This API only play mono audio but can be used on file containing
- // audio with more channels (in which case the audio will be converted to
- // mono).
- virtual WebRtc_Word32 PlayoutAudioData(
- WebRtc_Word8* audioBuffer,
- WebRtc_UWord32& dataLengthInBytes) = 0;
-
- // Put one video frame into videoBuffer. dataLengthInBytes is both an input
- // and output parameter. As input parameter it indicates the size of
- // videoBuffer. As output parameter it indicates the number of bytes written
- // to videoBuffer.
- virtual WebRtc_Word32 PlayoutAVIVideoData(
- WebRtc_Word8* videoBuffer,
- WebRtc_UWord32& dataLengthInBytes) = 0;
-
- // Put 10-60ms, depending on codec frame size, of audio data from file into
- // audioBufferLeft and audioBufferRight. The buffers contain the left and
- // right channel of played out stereo audio.
- // dataLengthInBytes is both an input and output parameter. As input
- // parameter it indicates the size of both audioBufferLeft and
- // audioBufferRight. As output parameter it indicates the number of bytes
- // written to both audio buffers.
- // Note: This API can only be successfully called for WAV files with stereo
- // audio.
- virtual WebRtc_Word32 PlayoutStereoData(
- WebRtc_Word8* audioBufferLeft,
- WebRtc_Word8* audioBufferRight,
- WebRtc_UWord32& dataLengthInBytes) = 0;
-
- // Open the file specified by fileName (relative path is allowed) for
- // reading. FileCallback::PlayNotification(..) will be called after
- // notificationTimeMs of the file has been played if notificationTimeMs is
- // greater than zero. If loop is true the file will be played until
- // StopPlaying() is called. When end of file is reached the file is read
- // from the start. format specifies the type of file fileName refers to.
- // codecInst specifies the encoding of the audio data. Note that
- // file formats that contain this information (like WAV files) don't need to
- // provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
- // specify what part of the file should be read. From startPointMs ms to
- // stopPointMs ms.
- // Note: codecInst.channels should be set to 2 for stereo (and 1 for
- // mono). Stereo audio is only supported for WAV files.
- virtual WebRtc_Word32 StartPlayingAudioFile(
- const WebRtc_Word8* fileName,
- const WebRtc_UWord32 notificationTimeMs = 0,
- const bool loop = false,
- const FileFormats format = kFileFormatPcm16kHzFile,
- const CodecInst* codecInst = NULL,
- const WebRtc_UWord32 startPointMs = 0,
- const WebRtc_UWord32 stopPointMs = 0) = 0;
-
- // Open the file specified by fileName for reading (relative path is
- // allowed). If loop is true the file will be played until StopPlaying() is
- // called. When end of file is reached the file is read from the start.
- // format specifies the type of file fileName refers to. Only video will be
- // read if videoOnly is true.
- virtual WebRtc_Word32 StartPlayingVideoFile(const WebRtc_Word8* fileName,
- const bool loop,
- bool videoOnly,
- const FileFormats format) = 0;
-
- // Prepare for playing audio from stream.
- // FileCallback::PlayNotification(..) will be called after
- // notificationTimeMs of the file has been played if notificationTimeMs is
- // greater than zero. format specifies the type of file fileName refers to.
- // codecInst specifies the encoding of the audio data. Note that
- // file formats that contain this information (like WAV files) don't need to
- // provide a non-NULL codecInst. startPointMs and stopPointMs, unless zero,
- // specify what part of the file should be read. From startPointMs ms to
- // stopPointMs ms.
- // Note: codecInst.channels should be set to 2 for stereo (and 1 for
- // mono). Stereo audio is only supported for WAV files.
- virtual WebRtc_Word32 StartPlayingAudioStream(
- InStream& stream,
- const WebRtc_UWord32 notificationTimeMs = 0,
- const FileFormats format = kFileFormatPcm16kHzFile,
- const CodecInst* codecInst = NULL,
- const WebRtc_UWord32 startPointMs = 0,
- const WebRtc_UWord32 stopPointMs = 0) = 0;
-
- // Stop playing from file or stream.
- virtual WebRtc_Word32 StopPlaying() = 0;
-
- // Return true if playing.
- virtual bool IsPlaying() = 0;
-
-
- // Set durationMs to the number of ms that has been played from file.
- virtual WebRtc_Word32 PlayoutPositionMs(
- WebRtc_UWord32& durationMs) const = 0;
-
- // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
- // to file. The audio frame size is determined by the codecInst.pacsize
- // parameter of the last sucessfull StartRecordingAudioFile(..) call.
- // Note: bufferLength must be exactly one frame.
- virtual WebRtc_Word32 IncomingAudioData(
- const WebRtc_Word8* audioBuffer,
- const WebRtc_UWord32 bufferLength) = 0;
-
- // Write one video frame, i.e. the bufferLength first bytes of videoBuffer,
- // to file.
- // Note: videoBuffer can contain encoded data. The codec used must be the
- // same as what was specified by videoCodecInst for the last successfull
- // StartRecordingVideoFile(..) call. The videoBuffer must contain exactly
- // one video frame.
- virtual WebRtc_Word32 IncomingAVIVideoData(
- const WebRtc_Word8* videoBuffer,
- const WebRtc_UWord32 bufferLength) = 0;
-
- // Open/creates file specified by fileName for writing (relative path is
- // allowed). FileCallback::RecordNotification(..) will be called after
- // notificationTimeMs of audio data has been recorded if
- // notificationTimeMs is greater than zero.
- // format specifies the type of file that should be created/opened.
- // codecInst specifies the encoding of the audio data. maxSizeBytes
- // specifies the number of bytes allowed to be written to file if it is
- // greater than zero.
- // Note: codecInst.channels should be set to 2 for stereo (and 1 for
- // mono). Stereo is only supported for WAV files.
- virtual WebRtc_Word32 StartRecordingAudioFile(
- const WebRtc_Word8* fileName,
- const FileFormats format,
- const CodecInst& codecInst,
- const WebRtc_UWord32 notificationTimeMs = 0,
- const WebRtc_UWord32 maxSizeBytes = 0) = 0;
-
- // Open/create the file specified by fileName for writing audio/video data
- // (relative path is allowed). format specifies the type of file fileName
- // should be. codecInst specifies the encoding of the audio data.
- // videoCodecInst specifies the encoding of the video data. Only video data
- // will be recorded if videoOnly is true.
- virtual WebRtc_Word32 StartRecordingVideoFile(
- const WebRtc_Word8* fileName,
- const FileFormats format,
- const CodecInst& codecInst,
- const VideoCodec& videoCodecInst,
- bool videoOnly = false) = 0;
-
- // Prepare for recording audio to stream.
- // FileCallback::RecordNotification(..) will be called after
- // notificationTimeMs of audio data has been recorded if
- // notificationTimeMs is greater than zero.
- // format specifies the type of file that stream should correspond to.
- // codecInst specifies the encoding of the audio data.
- // Note: codecInst.channels should be set to 2 for stereo (and 1 for
- // mono). Stereo is only supported for WAV files.
- virtual WebRtc_Word32 StartRecordingAudioStream(
- OutStream& stream,
- const FileFormats format,
- const CodecInst& codecInst,
- const WebRtc_UWord32 notificationTimeMs = 0) = 0;
-
- // Stop recording to file or stream.
- virtual WebRtc_Word32 StopRecording() = 0;
-
- // Return true if recording.
- virtual bool IsRecording() = 0;
-
- // Set durationMs to the number of ms that has been recorded to file.
- virtual WebRtc_Word32 RecordDurationMs(WebRtc_UWord32& durationMs) = 0;
-
- // Return true if recording or playing is stereo.
- virtual bool IsStereo() = 0;
-
- // Register callback to receive media file related notifications. Disables
- // callbacks if callback is NULL.
- virtual WebRtc_Word32 SetModuleFileCallback(FileCallback* callback) = 0;
-
- // Set durationMs to the size of the file (in ms) specified by fileName.
- // format specifies the type of file fileName refers to. freqInHz specifies
- // the sampling frequency of the file.
- virtual WebRtc_Word32 FileDurationMs(
- const WebRtc_Word8* fileName,
- WebRtc_UWord32& durationMs,
- const FileFormats format,
- const WebRtc_UWord32 freqInHz = 16000) = 0;
-
- // Update codecInst according to the current audio codec being used for
- // reading or writing.
- virtual WebRtc_Word32 codec_info(CodecInst& codecInst) const = 0;
-
- // Update videoCodecInst according to the current video codec being used for
- // reading or writing.
- virtual WebRtc_Word32 VideoCodecInst(VideoCodec& videoCodecInst) const = 0;
-
-protected:
- MediaFile() {}
- virtual ~MediaFile() {}
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_H_
diff --git a/trunk/src/modules/media_file/interface/media_file_defines.h b/trunk/src/modules/media_file/interface/media_file_defines.h
deleted file mode 100644
index 38af562..0000000
--- a/trunk/src/modules/media_file/interface/media_file_defines.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
-#define WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
-
-#include "engine_configurations.h"
-#include "module_common_types.h"
-#include "typedefs.h"
-
-namespace webrtc {
-// Callback class for the MediaFile class.
-class FileCallback
-{
-public:
- virtual ~FileCallback(){}
-
- // This function is called by MediaFile when a file has been playing for
- // durationMs ms. id is the identifier for the MediaFile instance calling
- // the callback.
- virtual void PlayNotification(const WebRtc_Word32 id,
- const WebRtc_UWord32 durationMs) = 0;
-
- // This function is called by MediaFile when a file has been recording for
- // durationMs ms. id is the identifier for the MediaFile instance calling
- // the callback.
- virtual void RecordNotification(const WebRtc_Word32 id,
- const WebRtc_UWord32 durationMs) = 0;
-
- // This function is called by MediaFile when a file has been stopped
- // playing. id is the identifier for the MediaFile instance calling the
- // callback.
- virtual void PlayFileEnded(const WebRtc_Word32 id) = 0;
-
- // This function is called by MediaFile when a file has been stopped
- // recording. id is the identifier for the MediaFile instance calling the
- // callback.
- virtual void RecordFileEnded(const WebRtc_Word32 id) = 0;
-
-protected:
- FileCallback() {}
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_INTERFACE_MEDIA_FILE_DEFINES_H_
diff --git a/trunk/src/modules/media_file/source/Android.mk b/trunk/src/modules/media_file/source/Android.mk
deleted file mode 100644
index 2c408c8..0000000
--- a/trunk/src/modules/media_file/source/Android.mk
+++ /dev/null
@@ -1,46 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_media_file
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := \
- avi_file.cc \
- media_file_impl.cc \
- media_file_utility.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS) \
- '-DWEBRTC_MODULE_UTILITY_VIDEO'
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../.. \
- $(LOCAL_PATH)/../../interface \
- $(LOCAL_PATH)/../../utility/interface \
- $(LOCAL_PATH)/../../audio_coding/main/interface \
- $(LOCAL_PATH)/../../../system_wrappers/interface
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/media_file/source/avi_file.cc b/trunk/src/modules/media_file/source/avi_file.cc
deleted file mode 100644
index 74f7908..0000000
--- a/trunk/src/modules/media_file/source/avi_file.cc
+++ /dev/null
@@ -1,1773 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// TODO(henrike): reassess the error handling in this class. Currently failure
-// is detected by asserts in many places. Also a refactoring of this class would
-// be beneficial.
-
-#include "avi_file.h"
-
-#include <assert.h>
-#include <string.h>
-
-#ifdef _WIN32
-#include <windows.h>
-#endif
-
-#include "critical_section_wrapper.h"
-#include "file_wrapper.h"
-#include "list_wrapper.h"
-#include "trace.h"
-
-// http://msdn2.microsoft.com/en-us/library/ms779636.aspx
-// A chunk has the following form:
-// ckID ckSize ckData
-// where ckID is a FOURCC that identifies the data contained in the
-// chunk, ckData is a 4-byte value giving the size of the data in
-// ckData, and ckData is zero or more bytes of data. The data is always
-// padded to nearest WORD boundary. ckSize gives the size of the valid
-// data in the chunk; it does not include the padding, the size of
-// ckID, or the size of ckSize.
-//http://msdn2.microsoft.com/en-us/library/ms779632.aspx
-//NOTE: Workaround to make MPEG4 files play on WMP. MPEG files
-// place the config parameters efter the BITMAPINFOHEADER and
-// *NOT* in the 'strd'!
-// http://msdn.microsoft.com/en-us/library/dd183375.aspx
-// http://msdn.microsoft.com/en-us/library/dd183376.aspx
-
-namespace webrtc {
-namespace {
-static const WebRtc_UWord32 kAvifHasindex = 0x00000010;
-static const WebRtc_UWord32 kAvifMustuseindex = 0x00000020;
-static const WebRtc_UWord32 kAvifIsinterleaved = 0x00000100;
-static const WebRtc_UWord32 kAvifTrustcktype = 0x00000800;
-static const WebRtc_UWord32 kAvifWascapturefile = 0x00010000;
-
-template <class T>
-T MinValue(T a, T b)
-{
- return a < b ? a : b;
-}
-} // namespace
-
-AviFile::AVIMAINHEADER::AVIMAINHEADER()
- : fcc( 0),
- cb( 0),
- dwMicroSecPerFrame( 0),
- dwMaxBytesPerSec( 0),
- dwPaddingGranularity( 0),
- dwFlags( 0),
- dwTotalFrames( 0),
- dwInitialFrames( 0),
- dwStreams( 0),
- dwSuggestedBufferSize(0),
- dwWidth( 0),
- dwHeight( 0)
-{
- dwReserved[0] = 0;
- dwReserved[1] = 0;
- dwReserved[2] = 0;
- dwReserved[3] = 0;
-}
-
-AVISTREAMHEADER::AVISTREAMHEADER()
- : fcc( 0),
- cb( 0),
- fccType( 0),
- fccHandler( 0),
- dwFlags( 0),
- wPriority( 0),
- wLanguage( 0),
- dwInitialFrames( 0),
- dwScale( 0),
- dwRate( 0),
- dwStart( 0),
- dwLength( 0),
- dwSuggestedBufferSize(0),
- dwQuality( 0),
- dwSampleSize( 0)
-{
- rcFrame.left = 0;
- rcFrame.top = 0;
- rcFrame.right = 0;
- rcFrame.bottom = 0;
-}
-
-BITMAPINFOHEADER::BITMAPINFOHEADER()
- : biSize( 0),
- biWidth( 0),
- biHeight( 0),
- biPlanes( 0),
- biBitCount( 0),
- biCompression( 0),
- biSizeImage( 0),
- biXPelsPerMeter(0),
- biYPelsPerMeter(0),
- biClrUsed( 0),
- biClrImportant( 0)
-{
-}
-
-WAVEFORMATEX::WAVEFORMATEX()
- : wFormatTag( 0),
- nChannels( 0),
- nSamplesPerSec( 0),
- nAvgBytesPerSec(0),
- nBlockAlign( 0),
- wBitsPerSample( 0),
- cbSize( 0)
-{
-}
-
-AviFile::AVIINDEXENTRY::AVIINDEXENTRY(WebRtc_UWord32 inckid,
- WebRtc_UWord32 indwFlags,
- WebRtc_UWord32 indwChunkOffset,
- WebRtc_UWord32 indwChunkLength)
- : ckid(inckid),
- dwFlags(indwFlags),
- dwChunkOffset(indwChunkOffset),
- dwChunkLength(indwChunkLength)
-{
-}
-
-AviFile::AviFile()
- : _crit(CriticalSectionWrapper::CreateCriticalSection()),
- _aviFile(NULL),
- _aviHeader(),
- _videoStreamHeader(),
- _audioStreamHeader(),
- _videoFormatHeader(),
- _audioFormatHeader(),
- _videoConfigParameters(),
- _videoConfigLength(0),
- _videoStreamName(),
- _videoStreamNameLength(0),
- _audioConfigParameters(),
- _audioStreamName(),
- _videoStream(),
- _audioStream(),
- _nrStreams(0),
- _aviLength(0),
- _dataLength(0),
- _bytesRead(0),
- _dataStartByte(0),
- _framesRead(0),
- _videoFrames(0),
- _audioFrames(0),
- _reading(false),
- _openedAs(AVI_AUDIO),
- _loop(false),
- _writing(false),
- _bytesWritten(0),
- _riffSizeMark(0),
- _moviSizeMark(0),
- _totNumFramesMark(0),
- _videoStreamLengthMark(0),
- _audioStreamLengthMark(0),
- _moviListOffset(0),
- _writeAudioStream(false),
- _writeVideoStream(false),
- _aviMode(NotSet),
- _videoCodecConfigParams(NULL),
- _videoCodecConfigParamsLength(0),
- _videoStreamDataChunkPrefix(0),
- _audioStreamDataChunkPrefix(0),
- _created(false),
- _indexList(new ListWrapper())
-{
- ResetComplexMembers();
-}
-
-AviFile::~AviFile()
-{
- Close();
-
- delete _indexList;
- delete[] _videoCodecConfigParams;
- delete _crit;
-}
-
-WebRtc_Word32 AviFile::Open(AVIStreamType streamType, const char* fileName,
- bool loop)
-{
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1, "OpenAVIFile(%s)",
- fileName);
- _crit->Enter();
-
- if (_aviMode != NotSet)
- {
- _crit->Leave();
- return -1;
- }
-
- _aviMode = Read;
-
- if (!fileName)
- {
- _crit->Leave();
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "\tfileName not valid!");
- return -1;
- }
-
-#ifdef _WIN32
- // fopen does not support wide characters on Windows, ergo _wfopen.
- wchar_t wideFileName[FileWrapper::kMaxFileNameSize];
- wideFileName[0] = 0;
- MultiByteToWideChar(CP_UTF8,0,fileName, -1, // convert the whole string
- wideFileName, FileWrapper::kMaxFileNameSize);
-
- _aviFile = _wfopen(wideFileName, L"rb");
-#else
- _aviFile = fopen(fileName, "rb");
-#endif
-
- if (!_aviFile)
- {
- _crit->Leave();
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Could not open file!");
- return -1;
- }
-
- // ReadRIFF verifies that the file is AVI and figures out the file length.
- WebRtc_Word32 err = ReadRIFF();
- if (err)
- {
- if (_aviFile)
- {
- fclose(_aviFile);
- _aviFile = NULL;
- }
- _crit->Leave();
- return -1;
- }
-
- err = ReadHeaders();
- if (err)
- {
- if (_aviFile)
- {
- fclose(_aviFile);
- _aviFile = NULL;
- }
- _crit->Leave();
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Unsupported or corrupt AVI format");
- return -1;
- }
-
- _dataStartByte = _bytesRead;
- _reading = true;
- _openedAs = streamType;
- _loop = loop;
- _crit->Leave();
- return 0;
-}
-
-WebRtc_Word32 AviFile::Close()
-{
- _crit->Enter();
- switch (_aviMode)
- {
- case Read:
- CloseRead();
- break;
- case Write:
- CloseWrite();
- break;
- default:
- break;
- }
-
- if (_videoCodecConfigParams)
- {
- delete [] _videoCodecConfigParams;
- _videoCodecConfigParams = 0;
- }
- ResetMembers();
- _crit->Leave();
- return 0;
-}
-
-WebRtc_UWord32 AviFile::MakeFourCc(WebRtc_UWord8 ch0, WebRtc_UWord8 ch1,
- WebRtc_UWord8 ch2, WebRtc_UWord8 ch3)
-{
- return ((WebRtc_UWord32)(WebRtc_UWord8)(ch0) |
- ((WebRtc_UWord32)(WebRtc_UWord8)(ch1) << 8) |
- ((WebRtc_UWord32)(WebRtc_UWord8)(ch2) << 16) |
- ((WebRtc_UWord32)(WebRtc_UWord8)(ch3) << 24 ));
-}
-
-WebRtc_Word32 AviFile::GetVideoStreamInfo(AVISTREAMHEADER& videoStreamHeader,
- BITMAPINFOHEADER& bitmapInfo,
- char* codecConfigParameters,
- WebRtc_Word32& configLength)
-{
- _crit->Enter();
- if (!_reading && !_created)
- {
- _crit->Leave();
- return -1;
- }
-
- memcpy(&videoStreamHeader, &_videoStreamHeader, sizeof(_videoStreamHeader));
- memcpy(&bitmapInfo, &_videoFormatHeader, sizeof(_videoFormatHeader));
-
- if (configLength <= _videoConfigLength)
- {
- memcpy(codecConfigParameters, _videoConfigParameters,
- _videoConfigLength);
- configLength = _videoConfigLength;
- }
- else
- {
- configLength = 0;
- }
- _crit->Leave();
- return 0;
-}
-
-WebRtc_Word32 AviFile::GetDuration(WebRtc_Word32& durationMs)
-{
- _crit->Enter();
- if (_videoStreamHeader.dwRate==0 || _videoStreamHeader.dwScale==0)
- {
- _crit->Leave();
- return -1;
- }
-
- durationMs = _videoStreamHeader.dwLength * 1000 /
- (_videoStreamHeader.dwRate/_videoStreamHeader.dwScale);
- _crit->Leave();
- return 0;
-}
-
-WebRtc_Word32 AviFile::GetAudioStreamInfo(WAVEFORMATEX& waveHeader)
-{
- _crit->Enter();
- if (_aviMode != Read)
- {
- _crit->Leave();
- return -1;
- }
- if (!_reading && !_created)
- {
- _crit->Leave();
- return -1;
- }
- memcpy(&waveHeader, &_audioFormatHeader, sizeof(_audioFormatHeader));
- _crit->Leave();
- return 0;
-}
-
-WebRtc_Word32 AviFile::WriteAudio(const WebRtc_UWord8* data,
- WebRtc_Word32 length)
-{
- _crit->Enter();
- size_t newBytesWritten = _bytesWritten;
-
- if (_aviMode != Write)
- {
- _crit->Leave();
- return -1;
- }
- if (!_created)
- {
- _crit->Leave();
- return -1;
- }
- if (!_writeAudioStream)
- {
- _crit->Leave();
- return -1;
- }
-
- // Start of chunk.
- const WebRtc_UWord32 chunkOffset = ftell(_aviFile) - _moviListOffset;
- _bytesWritten += PutLE32(_audioStreamDataChunkPrefix);
- // Size is unknown at this point. Update later.
- _bytesWritten += PutLE32(0);
- const size_t chunkSizeMark = _bytesWritten;
-
- _bytesWritten += PutBuffer(data, length);
-
- const long chunkSize = PutLE32LengthFromCurrent(
- static_cast<long>(chunkSizeMark));
-
- // Make sure that the chunk is aligned on 2 bytes (= 1 sample).
- if (chunkSize % 2)
- {
- _bytesWritten += PutByte(0);
- }
- // End of chunk
-
- // Save chunk information for use when closing file.
- AddChunkToIndexList(_audioStreamDataChunkPrefix, 0, // No flags.
- chunkOffset, chunkSize);
-
- ++_audioFrames;
- newBytesWritten = _bytesWritten - newBytesWritten;
- _crit->Leave();
- return static_cast<WebRtc_Word32>(newBytesWritten);
-}
-
-WebRtc_Word32 AviFile::WriteVideo(const WebRtc_UWord8* data,
- WebRtc_Word32 length)
-{
- _crit->Enter();
- size_t newBytesWritten = _bytesWritten;
- if (_aviMode != Write)
- {
- _crit->Leave();
- return -1;
- }
- if (!_created)
- {
- _crit->Leave();
- return -1;
- }
- if (!_writeVideoStream)
- {
- _crit->Leave();
- return -1;
- }
-
- // Start of chunk.
- const WebRtc_UWord32 chunkOffset = ftell(_aviFile) - _moviListOffset;
- _bytesWritten += PutLE32(_videoStreamDataChunkPrefix);
- // Size is unknown at this point. Update later.
- _bytesWritten += PutLE32(0);
- const size_t chunkSizeMark = _bytesWritten;
-
- _bytesWritten += PutBuffer(data, length);
-
- const long chunkSize = PutLE32LengthFromCurrent(
- static_cast<long>(chunkSizeMark));
-
- // Make sure that the chunk is aligned on 2 bytes (= 1 sample).
- if (chunkSize % 2)
- {
- //Pad one byte, to WORD align.
- _bytesWritten += PutByte(0);
- }
- //End chunk!
- AddChunkToIndexList(_videoStreamDataChunkPrefix, 0, // No flags.
- chunkOffset, static_cast<WebRtc_UWord32>(chunkSize));
-
- ++_videoFrames;
- newBytesWritten = _bytesWritten - newBytesWritten;
- _crit->Leave();
- return static_cast<WebRtc_Word32>(newBytesWritten);
-}
-
-WebRtc_Word32 AviFile::PrepareDataChunkHeaders()
-{
- // 00 video stream, 01 audio stream.
- // db uncompresses video, dc compressed video, wb WAV audio
- if (_writeVideoStream)
- {
- if (strncmp((const char*) &_videoStreamHeader.fccHandler, "I420", 4) ==
- 0)
- {
- _videoStreamDataChunkPrefix = MakeFourCc('0', '0', 'd', 'b');
- }
- else
- {
- _videoStreamDataChunkPrefix = MakeFourCc('0', '0', 'd', 'c');
- }
- _audioStreamDataChunkPrefix = MakeFourCc('0', '1', 'w', 'b');
- }
- else
- {
- _audioStreamDataChunkPrefix = MakeFourCc('0', '0', 'w', 'b');
- }
- return 0;
-}
-
-WebRtc_Word32 AviFile::ReadMoviSubChunk(WebRtc_UWord8* data,
- WebRtc_Word32& length,
- WebRtc_UWord32 tag1,
- WebRtc_UWord32 tag2)
-{
- if (!_reading)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceVideo, -1,
- "AviFile::ReadMoviSubChunk(): File not open!");
- length = 0;
- return -1;
- }
-
- WebRtc_UWord32 size;
- bool isEOFReached = false;
- // Try to read one data chunk header
- while (true)
- {
- // TODO (hellner): what happens if an empty AVI file is opened with
- // _loop set to true? Seems like this while-loop would never exit!
-
- // tag = db uncompresses video, dc compressed video or wb WAV audio.
- WebRtc_UWord32 tag;
- _bytesRead += GetLE32(tag);
- _bytesRead += GetLE32(size);
-
- const WebRtc_Word32 eof = feof(_aviFile);
- if (!eof)
- {
- if (tag == tag1)
- {
- // Supported tag found.
- break;
- }
- else if ((tag == tag2) && (tag2 != 0))
- {
- // Supported tag found.
- break;
- }
-
- // Jump to next chunk. The size is in bytes but chunks are aligned
- // on 2 byte boundaries.
- const WebRtc_UWord32 seekSize = (size % 2) ? size + 1 : size;
- const WebRtc_Word32 err = fseek(_aviFile, seekSize, SEEK_CUR);
-
- if (err)
- {
- isEOFReached = true;
- }
- }
- else
- {
- isEOFReached = true;
- }
-
- if (isEOFReached)
- {
- clearerr(_aviFile);
-
- if (_loop)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceVideo, -1,
- "AviFile::ReadMoviSubChunk(): Reached end of AVI\
- data file, starting from the beginning.");
-
- fseek(_aviFile, static_cast<long>(_dataStartByte), SEEK_SET);
-
- _bytesRead = _dataStartByte;
- _framesRead = 0;
- isEOFReached = false;
- }
- else
- {
- WEBRTC_TRACE(kTraceDebug, kTraceVideo, -1,
- "AviFile::ReadMoviSubChunk(): Reached end of AVI\
- file!");
- length = 0;
- return -1;
- }
- }
- _bytesRead += size;
- }
-
- if (static_cast<WebRtc_Word32>(size) > length)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceVideo, -1,
- "AviFile::ReadMoviSubChunk(): AVI read buffer too small!");
-
- // Jump to next chunk. The size is in bytes but chunks are aligned
- // on 2 byte boundaries.
- const WebRtc_UWord32 seekSize = (size % 2) ? size + 1 : size;
- fseek(_aviFile, seekSize, SEEK_CUR);
- _bytesRead += seekSize;
- length = 0;
- return -1;
- }
- _bytesRead += GetBuffer(data, size);
-
- // The size is in bytes but chunks are aligned on 2 byte boundaries.
- if (size % 2)
- {
- WebRtc_UWord8 dummy_byte;
- _bytesRead += GetByte(dummy_byte);
- }
- length = size;
- ++_framesRead;
- return 0;
-}
-
-WebRtc_Word32 AviFile::ReadAudio(WebRtc_UWord8* data, WebRtc_Word32& length)
-{
- _crit->Enter();
- WEBRTC_TRACE(kTraceDebug, kTraceVideo, -1, "AviFile::ReadAudio()");
-
- if (_aviMode != Read)
- {
- _crit->Leave();
- return -1;
- }
- if (_openedAs != AVI_AUDIO)
- {
- length = 0;
- _crit->Leave();
- WEBRTC_TRACE(kTraceDebug, kTraceVideo, -1, "File not open as audio!");
- return -1;
- }
-
- const WebRtc_Word32 ret = ReadMoviSubChunk(
- data,
- length,
- StreamAndTwoCharCodeToTag(_audioStream.streamNumber, "wb"));
-
- _crit->Leave();
- return ret;
-}
-
-WebRtc_Word32 AviFile::ReadVideo(WebRtc_UWord8* data, WebRtc_Word32& length)
-{
- WEBRTC_TRACE(kTraceDebug, kTraceVideo, -1, "AviFile::ReadVideo()");
-
- _crit->Enter();
- if (_aviMode != Read)
- {
- //Has to be Read!
- _crit->Leave();
- return -1;
- }
- if (_openedAs != AVI_VIDEO)
- {
- length = 0;
- _crit->Leave();
- WEBRTC_TRACE(kTraceDebug, kTraceVideo, -1, "File not open as video!");
- return -1;
- }
-
- const WebRtc_Word32 ret = ReadMoviSubChunk(
- data,
- length,
- StreamAndTwoCharCodeToTag(_videoStream.streamNumber, "dc"),
- StreamAndTwoCharCodeToTag(_videoStream.streamNumber, "db"));
- _crit->Leave();
- return ret;
-}
-
-WebRtc_Word32 AviFile::Create(const char* fileName)
-{
- _crit->Enter();
- if (_aviMode != Write)
- {
- _crit->Leave();
- return -1;
- }
-
- if (!_writeVideoStream && !_writeAudioStream)
- {
- _crit->Leave();
- return -1;
- }
- if (_created)
- {
- _crit->Leave();
- return -1;
- }
-
-#ifdef _WIN32
- // fopen does not support wide characters on Windows, ergo _wfopen.
- wchar_t wideFileName[FileWrapper::kMaxFileNameSize];
- wideFileName[0] = 0;
-
- MultiByteToWideChar(CP_UTF8,0,fileName, -1, // convert the whole string
- wideFileName, FileWrapper::kMaxFileNameSize);
-
- _aviFile = _wfopen(wideFileName, L"w+b");
- if (!_aviFile)
- {
- _crit->Leave();
- return -1;
- }
-#else
- _aviFile = fopen(fileName, "w+b");
- if (!_aviFile)
- {
- _crit->Leave();
- return -1;
- }
-#endif
-
- WriteRIFF();
- WriteHeaders();
-
- _created = true;
-
- PrepareDataChunkHeaders();
- ClearIndexList();
- WriteMoviStart();
- _aviMode = Write;
- _crit->Leave();
- return 0;
-}
-
-WebRtc_Word32 AviFile::CreateVideoStream(
- const AVISTREAMHEADER& videoStreamHeader,
- const BITMAPINFOHEADER& bitMapInfoHeader,
- const WebRtc_UWord8* codecConfigParams,
- WebRtc_Word32 codecConfigParamsLength)
-{
- _crit->Enter();
- if (_aviMode == Read)
- {
- _crit->Leave();
- return -1;
- }
-
- if (_created)
- {
- _crit->Leave();
- return -1;
- }
-
- _aviMode = Write;
- _writeVideoStream = true;
-
- _videoStreamHeader = videoStreamHeader;
- _videoFormatHeader = bitMapInfoHeader;
-
- if (codecConfigParams && codecConfigParamsLength > 0)
- {
- if (_videoCodecConfigParams)
- {
- delete [] _videoCodecConfigParams;
- _videoCodecConfigParams = 0;
- }
-
- _videoCodecConfigParams = new WebRtc_UWord8[codecConfigParamsLength];
- _videoCodecConfigParamsLength = codecConfigParamsLength;
-
- memcpy(_videoCodecConfigParams, codecConfigParams,
- _videoCodecConfigParamsLength);
- }
- _crit->Leave();
- return 0;
-}
-
-WebRtc_Word32 AviFile::CreateAudioStream(
- const AVISTREAMHEADER& audioStreamHeader,
- const WAVEFORMATEX& waveFormatHeader)
-{
- _crit->Enter();
-
- if (_aviMode == Read)
- {
- _crit->Leave();
- return -1;
- }
-
- if (_created)
- {
- _crit->Leave();
- return -1;
- }
-
- _aviMode = Write;
- _writeAudioStream = true;
- _audioStreamHeader = audioStreamHeader;
- _audioFormatHeader = waveFormatHeader;
- _crit->Leave();
- return 0;
-}
-
-WebRtc_Word32 AviFile::WriteRIFF()
-{
- const WebRtc_UWord32 riffTag = MakeFourCc('R', 'I', 'F', 'F');
- _bytesWritten += PutLE32(riffTag);
-
- // Size is unknown at this point. Update later.
- _bytesWritten += PutLE32(0);
- _riffSizeMark = _bytesWritten;
-
- const WebRtc_UWord32 aviTag = MakeFourCc('A', 'V', 'I', ' ');
- _bytesWritten += PutLE32(aviTag);
-
- return 0;
-}
-
-
-WebRtc_Word32 AviFile::WriteHeaders()
-{
- // Main AVI header list.
- const WebRtc_UWord32 listTag = MakeFourCc('L', 'I', 'S', 'T');
- _bytesWritten += PutLE32(listTag);
-
- // Size is unknown at this point. Update later.
- _bytesWritten += PutLE32(0);
- const size_t listhdrlSizeMark = _bytesWritten;
-
- const WebRtc_UWord32 hdrlTag = MakeFourCc('h', 'd', 'r', 'l');
- _bytesWritten += PutLE32(hdrlTag);
-
- WriteAVIMainHeader();
- WriteAVIStreamHeaders();
-
- const long hdrlLen = PutLE32LengthFromCurrent(
- static_cast<long>(listhdrlSizeMark));
-
- // Junk chunk to align on 2048 boundry (CD-ROM sector boundary).
- const WebRtc_UWord32 junkTag = MakeFourCc('J', 'U', 'N', 'K');
- _bytesWritten += PutLE32(junkTag);
- // Size is unknown at this point. Update later.
- _bytesWritten += PutLE32(0);
- const size_t junkSizeMark = _bytesWritten;
-
- const WebRtc_UWord32 junkBufferSize =
- 0x800 // 2048 byte alignment
- - 12 // RIFF SIZE 'AVI '
- - 8 // LIST SIZE
- - hdrlLen //
- - 8 // JUNK SIZE
- - 12; // LIST SIZE 'MOVI'
-
- // TODO (hellner): why not just fseek here?
- WebRtc_UWord8* junkBuffer = new WebRtc_UWord8[junkBufferSize];
- memset(junkBuffer, 0, junkBufferSize);
- _bytesWritten += PutBuffer(junkBuffer, junkBufferSize);
- delete [] junkBuffer;
-
- PutLE32LengthFromCurrent(static_cast<long>(junkSizeMark));
- // End of JUNK chunk.
- // End of main AVI header list.
- return 0;
-}
-
-WebRtc_Word32 AviFile::WriteAVIMainHeader()
-{
- const WebRtc_UWord32 avihTag = MakeFourCc('a', 'v', 'i', 'h');
- _bytesWritten += PutLE32(avihTag);
- _bytesWritten += PutLE32(14 * sizeof(WebRtc_UWord32));
-
- const WebRtc_UWord32 scale = _videoStreamHeader.dwScale ?
- _videoStreamHeader.dwScale : 1;
- const WebRtc_UWord32 microSecPerFrame = 1000000 /
- (_videoStreamHeader.dwRate / scale);
- _bytesWritten += PutLE32(microSecPerFrame);
- _bytesWritten += PutLE32(0);
- _bytesWritten += PutLE32(0);
-
- WebRtc_UWord32 numStreams = 0;
- if (_writeVideoStream)
- {
- ++numStreams;
- }
- if (_writeAudioStream)
- {
- ++numStreams;
- }
-
- if (numStreams == 1)
- {
- _bytesWritten += PutLE32(
- kAvifTrustcktype
- | kAvifHasindex
- | kAvifWascapturefile);
- }
- else
- {
- _bytesWritten += PutLE32(
- kAvifTrustcktype
- | kAvifHasindex
- | kAvifWascapturefile
- | kAvifIsinterleaved);
- }
-
- _totNumFramesMark = _bytesWritten;
- _bytesWritten += PutLE32(0);
- _bytesWritten += PutLE32(0);
- _bytesWritten += PutLE32(numStreams);
-
- if (_writeVideoStream)
- {
- _bytesWritten += PutLE32(
- _videoStreamHeader.dwSuggestedBufferSize);
- _bytesWritten += PutLE32(
- _videoStreamHeader.rcFrame.right-_videoStreamHeader.rcFrame.left);
- _bytesWritten += PutLE32(
- _videoStreamHeader.rcFrame.bottom-_videoStreamHeader.rcFrame.top);
- } else {
- _bytesWritten += PutLE32(0);
- _bytesWritten += PutLE32(0);
- _bytesWritten += PutLE32(0);
- }
- _bytesWritten += PutLE32(0);
- _bytesWritten += PutLE32(0);
- _bytesWritten += PutLE32(0);
- _bytesWritten += PutLE32(0);
- return 0;
-}
-
-WebRtc_Word32 AviFile::WriteAVIStreamHeaders()
-{
- if (_writeVideoStream)
- {
- WriteAVIVideoStreamHeaders();
- }
- if (_writeAudioStream)
- {
- WriteAVIAudioStreamHeaders();
- }
- return 0;
-}
-
-WebRtc_Word32 AviFile::WriteAVIVideoStreamHeaders()
-{
- const WebRtc_UWord32 listTag = MakeFourCc('L', 'I', 'S', 'T');
- _bytesWritten += PutLE32(listTag);
-
- // Size is unknown at this point. Update later.
- _bytesWritten += PutLE32(0);
- const size_t liststrlSizeMark = _bytesWritten;
-
- const WebRtc_UWord32 hdrlTag = MakeFourCc('s', 't', 'r', 'l');
- _bytesWritten += PutLE32(hdrlTag);
-
- WriteAVIVideoStreamHeaderChunks();
-
- PutLE32LengthFromCurrent(static_cast<long>(liststrlSizeMark));
-
- return 0;
-}
-
-WebRtc_Word32 AviFile::WriteAVIVideoStreamHeaderChunks()
-{
- // Start of strh
- const WebRtc_UWord32 strhTag = MakeFourCc('s', 't', 'r', 'h');
- _bytesWritten += PutLE32(strhTag);
-
- // Size is unknown at this point. Update later.
- _bytesWritten += PutLE32(0);
- const size_t strhSizeMark = _bytesWritten;
-
- _bytesWritten += PutLE32(_videoStreamHeader.fccType);
- _bytesWritten += PutLE32(_videoStreamHeader.fccHandler);
- _bytesWritten += PutLE32(_videoStreamHeader.dwFlags);
- _bytesWritten += PutLE16(_videoStreamHeader.wPriority);
- _bytesWritten += PutLE16(_videoStreamHeader.wLanguage);
- _bytesWritten += PutLE32(_videoStreamHeader.dwInitialFrames);
- _bytesWritten += PutLE32(_videoStreamHeader.dwScale);
- _bytesWritten += PutLE32(_videoStreamHeader.dwRate);
- _bytesWritten += PutLE32(_videoStreamHeader.dwStart);
-
- _videoStreamLengthMark = _bytesWritten;
- _bytesWritten += PutLE32(_videoStreamHeader.dwLength);
-
- _bytesWritten += PutLE32(_videoStreamHeader.dwSuggestedBufferSize);
- _bytesWritten += PutLE32(_videoStreamHeader.dwQuality);
- _bytesWritten += PutLE32(_videoStreamHeader.dwSampleSize);
- _bytesWritten += PutLE16(_videoStreamHeader.rcFrame.left);
- _bytesWritten += PutLE16(_videoStreamHeader.rcFrame.top);
- _bytesWritten += PutLE16(_videoStreamHeader.rcFrame.right);
- _bytesWritten += PutLE16(_videoStreamHeader.rcFrame.bottom);
-
- PutLE32LengthFromCurrent(static_cast<long>(strhSizeMark));
- // End of strh
-
- // Start of strf
- const WebRtc_UWord32 strfTag = MakeFourCc('s', 't', 'r', 'f');
- _bytesWritten += PutLE32(strfTag);
-
- // Size is unknown at this point. Update later.
- _bytesWritten += PutLE32(0);
- const size_t strfSizeMark = _bytesWritten;
-
- _bytesWritten += PutLE32(_videoFormatHeader.biSize);
- _bytesWritten += PutLE32(_videoFormatHeader.biWidth);
- _bytesWritten += PutLE32(_videoFormatHeader.biHeight);
- _bytesWritten += PutLE16(_videoFormatHeader.biPlanes);
- _bytesWritten += PutLE16(_videoFormatHeader.biBitCount);
- _bytesWritten += PutLE32(_videoFormatHeader.biCompression);
- _bytesWritten += PutLE32(_videoFormatHeader.biSizeImage);
- _bytesWritten += PutLE32(_videoFormatHeader.biXPelsPerMeter);
- _bytesWritten += PutLE32(_videoFormatHeader.biYPelsPerMeter);
- _bytesWritten += PutLE32(_videoFormatHeader.biClrUsed);
- _bytesWritten += PutLE32(_videoFormatHeader.biClrImportant);
-
- const bool isMpegFile = _videoStreamHeader.fccHandler ==
- AviFile::MakeFourCc('M','4','S','2');
- if (isMpegFile)
- {
- if (_videoCodecConfigParams && _videoCodecConfigParamsLength > 0)
- {
- _bytesWritten += PutBuffer(_videoCodecConfigParams,
- _videoCodecConfigParamsLength);
- }
- }
-
- PutLE32LengthFromCurrent(static_cast<long>(strfSizeMark));
- // End of strf
-
- if ( _videoCodecConfigParams
- && (_videoCodecConfigParamsLength > 0)
- && !isMpegFile)
- {
- // Write strd, unless it's an MPEG file
- const WebRtc_UWord32 strdTag = MakeFourCc('s', 't', 'r', 'd');
- _bytesWritten += PutLE32(strdTag);
-
- // Size is unknown at this point. Update later.
- _bytesWritten += PutLE32(0);
- const size_t strdSizeMark = _bytesWritten;
-
- _bytesWritten += PutBuffer(_videoCodecConfigParams,
- _videoCodecConfigParamsLength);
-
- PutLE32LengthFromCurrent(static_cast<long>(strdSizeMark));
- // End of strd
- }
-
- // Start of strn
- const WebRtc_UWord32 strnTag = MakeFourCc('s', 't', 'r', 'n');
- _bytesWritten += PutLE32(strnTag);
-
- // Size is unknown at this point. Update later.
- _bytesWritten += PutLE32(0);
- const size_t strnSizeMark = _bytesWritten;
-
- _bytesWritten += PutBufferZ("WebRtc.avi ");
-
- PutLE32LengthFromCurrent(static_cast<long>(strnSizeMark));
- // End of strd
-
- return 0;
-}
-
-WebRtc_Word32 AviFile::WriteAVIAudioStreamHeaders()
-{
- // Start of LIST
- WebRtc_UWord32 listTag = MakeFourCc('L', 'I', 'S', 'T');
- _bytesWritten += PutLE32(listTag);
-
- // Size is unknown at this point. Update later.
- _bytesWritten += PutLE32(0);
- const size_t liststrlSizeMark = _bytesWritten;
-
- WebRtc_UWord32 hdrlTag = MakeFourCc('s', 't', 'r', 'l');
- _bytesWritten += PutLE32(hdrlTag);
-
- WriteAVIAudioStreamHeaderChunks();
-
- PutLE32LengthFromCurrent(static_cast<long>(liststrlSizeMark));
- //End of LIST
- return 0;
-}
-
-WebRtc_Word32 AviFile::WriteAVIAudioStreamHeaderChunks()
-{
- // Start of strh
- const WebRtc_UWord32 strhTag = MakeFourCc('s', 't', 'r', 'h');
- _bytesWritten += PutLE32(strhTag);
-
- // Size is unknown at this point. Update later.
- _bytesWritten += PutLE32(0);
- const size_t strhSizeMark = _bytesWritten;
-
- _bytesWritten += PutLE32(_audioStreamHeader.fccType);
- _bytesWritten += PutLE32(_audioStreamHeader.fccHandler);
- _bytesWritten += PutLE32(_audioStreamHeader.dwFlags);
- _bytesWritten += PutLE16(_audioStreamHeader.wPriority);
- _bytesWritten += PutLE16(_audioStreamHeader.wLanguage);
- _bytesWritten += PutLE32(_audioStreamHeader.dwInitialFrames);
- _bytesWritten += PutLE32(_audioStreamHeader.dwScale);
- _bytesWritten += PutLE32(_audioStreamHeader.dwRate);
- _bytesWritten += PutLE32(_audioStreamHeader.dwStart);
-
- _audioStreamLengthMark = _bytesWritten;
- _bytesWritten += PutLE32(_audioStreamHeader.dwLength);
-
- _bytesWritten += PutLE32(_audioStreamHeader.dwSuggestedBufferSize);
- _bytesWritten += PutLE32(_audioStreamHeader.dwQuality);
- _bytesWritten += PutLE32(_audioStreamHeader.dwSampleSize);
- _bytesWritten += PutLE16(_audioStreamHeader.rcFrame.left);
- _bytesWritten += PutLE16(_audioStreamHeader.rcFrame.top);
- _bytesWritten += PutLE16(_audioStreamHeader.rcFrame.right);
- _bytesWritten += PutLE16(_audioStreamHeader.rcFrame.bottom);
-
- PutLE32LengthFromCurrent(static_cast<long>(strhSizeMark));
- // End of strh
-
- // Start of strf
- const WebRtc_UWord32 strfTag = MakeFourCc('s', 't', 'r', 'f');
- _bytesWritten += PutLE32(strfTag);
-
- // Size is unknown at this point. Update later.
- _bytesWritten += PutLE32(0);
- const size_t strfSizeMark = _bytesWritten;
-
- _bytesWritten += PutLE16(_audioFormatHeader.wFormatTag);
- _bytesWritten += PutLE16(_audioFormatHeader.nChannels);
- _bytesWritten += PutLE32(_audioFormatHeader.nSamplesPerSec);
- _bytesWritten += PutLE32(_audioFormatHeader.nAvgBytesPerSec);
- _bytesWritten += PutLE16(_audioFormatHeader.nBlockAlign);
- _bytesWritten += PutLE16(_audioFormatHeader.wBitsPerSample);
- _bytesWritten += PutLE16(_audioFormatHeader.cbSize);
-
- PutLE32LengthFromCurrent(static_cast<long>(strfSizeMark));
- // End end of strf.
-
- // Audio doesn't have strd.
-
- // Start of strn
- const WebRtc_UWord32 strnTag = MakeFourCc('s', 't', 'r', 'n');
- _bytesWritten += PutLE32(strnTag);
-
- // Size is unknown at this point. Update later.
- _bytesWritten += PutLE32(0);
- const size_t strnSizeMark = _bytesWritten;
-
- _bytesWritten += PutBufferZ("WebRtc.avi ");
-
- PutLE32LengthFromCurrent(static_cast<long>(strnSizeMark));
- // End of strd.
-
- return 0;
-}
-
-WebRtc_Word32 AviFile::WriteMoviStart()
-{
- // Create template movi list. Fill out size when known (i.e. when closing
- // file).
- const WebRtc_UWord32 listTag = MakeFourCc('L', 'I', 'S', 'T');
- _bytesWritten += PutLE32(listTag);
-
- _bytesWritten += PutLE32(0); //Size! Change later!
- _moviSizeMark = _bytesWritten;
- _moviListOffset = ftell(_aviFile);
-
- const WebRtc_UWord32 moviTag = MakeFourCc('m', 'o', 'v', 'i');
- _bytesWritten += PutLE32(moviTag);
-
- return 0;
-}
-
-size_t AviFile::PutByte(WebRtc_UWord8 byte)
-{
- return fwrite(&byte, sizeof(WebRtc_UWord8), sizeof(WebRtc_UWord8),
- _aviFile);
-}
-
-size_t AviFile::PutLE16(WebRtc_UWord16 word)
-{
- return fwrite(&word, sizeof(WebRtc_UWord8), sizeof(WebRtc_UWord16),
- _aviFile);
-}
-
-size_t AviFile::PutLE32(WebRtc_UWord32 word)
-{
- return fwrite(&word, sizeof(WebRtc_UWord8), sizeof(WebRtc_UWord32),
- _aviFile);
-}
-
-size_t AviFile::PutBuffer(const WebRtc_UWord8* str, size_t size)
-{
- return fwrite(str, sizeof(WebRtc_UWord8), size,
- _aviFile);
-}
-
-size_t AviFile::PutBufferZ(const char* str)
-{
- // Include NULL charachter, hence the + 1
- return PutBuffer(reinterpret_cast<const WebRtc_UWord8*>(str),
- strlen(str) + 1);
-}
-
-long AviFile::PutLE32LengthFromCurrent(long startPos)
-{
- const long endPos = ftell(_aviFile);
- if (endPos < 0) {
- return 0;
- }
- bool success = (0 == fseek(_aviFile, startPos - 4, SEEK_SET));
- if (!success) {
- assert(false);
- return 0;
- }
- const long len = endPos - startPos;
- if (endPos > startPos) {
- PutLE32(len);
- }
- else {
- assert(false);
- }
- success = (0 == fseek(_aviFile, endPos, SEEK_SET));
- assert(success);
- return len;
-}
-
-void AviFile::PutLE32AtPos(long pos, WebRtc_UWord32 word)
-{
- const long currPos = ftell(_aviFile);
- if (currPos < 0) {
- assert(false);
- return;
- }
- bool success = (0 == fseek(_aviFile, pos, SEEK_SET));
- if (!success) {
- assert(false);
- return;
- }
- PutLE32(word);
- success = (0 == fseek(_aviFile, currPos, SEEK_SET));
- assert(success);
-}
-
-void AviFile::CloseRead()
-{
- if (_aviFile)
- {
- fclose(_aviFile);
- _aviFile = NULL;
- }
-}
-
-void AviFile::CloseWrite()
-{
- if (_created)
- {
- // Update everything that isn't known until the file is closed. The
- // marks indicate where in the headers this update should be.
- PutLE32LengthFromCurrent(static_cast<long>(_moviSizeMark));
-
- PutLE32AtPos(static_cast<long>(_totNumFramesMark), _videoFrames);
-
- if (_writeVideoStream)
- {
- PutLE32AtPos(static_cast<long>(_videoStreamLengthMark),
- _videoFrames);
- }
-
- if (_writeAudioStream)
- {
- PutLE32AtPos(static_cast<long>(_audioStreamLengthMark),
- _audioFrames);
- }
-
- WriteIndex();
- PutLE32LengthFromCurrent(static_cast<long>(_riffSizeMark));
- ClearIndexList();
-
- if (_aviFile)
- {
- fclose(_aviFile);
- _aviFile = NULL;
- }
- }
-}
-
-void AviFile::ResetMembers()
-{
- ResetComplexMembers();
-
- _aviFile = NULL;
-
- _nrStreams = 0;
- _aviLength = 0;
- _dataLength = 0;
- _bytesRead = 0;
- _dataStartByte = 0;
- _framesRead = 0;
- _videoFrames = 0;
- _audioFrames = 0;
-
- _reading = false;
- _openedAs = AVI_AUDIO;
- _loop = false;
- _writing = false;
-
- _bytesWritten = 0;
-
- _riffSizeMark = 0;
- _moviSizeMark = 0;
- _totNumFramesMark = 0;
- _videoStreamLengthMark = 0;
- _audioStreamLengthMark = 0;
-
- _writeAudioStream = false;
- _writeVideoStream = false;
-
- _aviMode = NotSet;
- _videoCodecConfigParams = 0;
- _videoCodecConfigParamsLength = 0;
-
- _videoStreamDataChunkPrefix = 0;
- _audioStreamDataChunkPrefix = 0;
-
- _created = false;
-
- _moviListOffset = 0;
-
- _videoConfigLength = 0;
-}
-
-void AviFile::ResetComplexMembers()
-{
- memset(&_aviHeader, 0, sizeof(AVIMAINHEADER));
- memset(&_videoStreamHeader, 0, sizeof(AVISTREAMHEADER));
- memset(&_audioStreamHeader, 0, sizeof(AVISTREAMHEADER));
- memset(&_videoFormatHeader, 0, sizeof(BITMAPINFOHEADER));
- memset(&_audioFormatHeader, 0, sizeof(WAVEFORMATEX));
- memset(_videoConfigParameters, 0, CODEC_CONFIG_LENGTH);
- memset(_videoStreamName, 0, STREAM_NAME_LENGTH);
- memset(_audioStreamName, 0, STREAM_NAME_LENGTH);
- memset(&_videoStream, 0, sizeof(AVIStream));
- memset(&_audioStream, 0, sizeof(AVIStream));
-}
-
-size_t AviFile::GetByte(WebRtc_UWord8& word)
-{
- return fread(&word, sizeof(WebRtc_UWord8), sizeof(WebRtc_UWord8), _aviFile);
-}
-
-size_t AviFile::GetLE16(WebRtc_UWord16& word)
-{
- return fread(&word, sizeof(WebRtc_UWord8), sizeof(WebRtc_UWord16),
- _aviFile);
-}
-
-size_t AviFile::GetLE32(WebRtc_UWord32& word)
-{
- return fread(&word, sizeof(WebRtc_UWord8), sizeof(WebRtc_UWord32),
- _aviFile);
-}
-
-size_t AviFile::GetBuffer(WebRtc_UWord8* str, size_t size)
-{
- return fread(str, sizeof(WebRtc_UWord8), size, _aviFile);
-}
-
-WebRtc_Word32 AviFile::ReadRIFF()
-{
- WebRtc_UWord32 tag;
- _bytesRead = GetLE32(tag);
- if (tag != MakeFourCc('R', 'I', 'F', 'F'))
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not a RIFF file!");
- return -1;
- }
-
- WebRtc_UWord32 size;
- _bytesRead += GetLE32(size);
- _aviLength = size;
-
- _bytesRead += GetLE32(tag);
- if (tag != MakeFourCc('A', 'V', 'I', ' '))
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not an AVI file!");
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AviFile::ReadHeaders()
-{
- WebRtc_UWord32 tag;
- _bytesRead += GetLE32(tag);
- WebRtc_UWord32 size;
- _bytesRead += GetLE32(size);
-
- if (tag != MakeFourCc('L', 'I', 'S', 'T'))
- {
- return -1;
- }
-
- WebRtc_UWord32 listTag;
- _bytesRead += GetLE32(listTag);
- if (listTag != MakeFourCc('h', 'd', 'r', 'l'))
- {
- return -1;
- }
-
- WebRtc_Word32 err = ReadAVIMainHeader();
- if (err)
- {
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 AviFile::ReadAVIMainHeader()
-{
- _bytesRead += GetLE32(_aviHeader.fcc);
- _bytesRead += GetLE32(_aviHeader.cb);
- _bytesRead += GetLE32(_aviHeader.dwMicroSecPerFrame);
- _bytesRead += GetLE32(_aviHeader.dwMaxBytesPerSec);
- _bytesRead += GetLE32(_aviHeader.dwPaddingGranularity);
- _bytesRead += GetLE32(_aviHeader.dwFlags);
- _bytesRead += GetLE32(_aviHeader.dwTotalFrames);
- _bytesRead += GetLE32(_aviHeader.dwInitialFrames);
- _bytesRead += GetLE32(_aviHeader.dwStreams);
- _bytesRead += GetLE32(_aviHeader.dwSuggestedBufferSize);
- _bytesRead += GetLE32(_aviHeader.dwWidth);
- _bytesRead += GetLE32(_aviHeader.dwHeight);
- _bytesRead += GetLE32(_aviHeader.dwReserved[0]);
- _bytesRead += GetLE32(_aviHeader.dwReserved[1]);
- _bytesRead += GetLE32(_aviHeader.dwReserved[2]);
- _bytesRead += GetLE32(_aviHeader.dwReserved[3]);
-
- if (_aviHeader.fcc != MakeFourCc('a', 'v', 'i', 'h'))
- {
- return -1;
- }
-
- if (_aviHeader.dwFlags & kAvifMustuseindex)
- {
- return -1;
- }
-
- bool readVideoStreamHeader = false;
- bool readAudioStreamHeader = false;
- unsigned int streamsRead = 0;
- while (_aviHeader.dwStreams > streamsRead)
- {
- WebRtc_UWord32 strltag;
- _bytesRead += GetLE32(strltag);
- WebRtc_UWord32 strlsize;
- _bytesRead += GetLE32(strlsize);
- const long endSeekPos = ftell(_aviFile) +
- static_cast<WebRtc_Word32>(strlsize);
-
- if (strltag != MakeFourCc('L', 'I', 'S', 'T'))
- {
- return -1;
- }
-
- WebRtc_UWord32 listTag;
- _bytesRead += GetLE32(listTag);
- if (listTag != MakeFourCc('s', 't', 'r', 'l'))
- {
- return -1;
- }
-
- WebRtc_UWord32 chunktag;
- _bytesRead += GetLE32(chunktag);
- WebRtc_UWord32 chunksize;
- _bytesRead += GetLE32(chunksize);
-
- if (chunktag != MakeFourCc('s', 't', 'r', 'h'))
- {
- return -1;
- }
-
- AVISTREAMHEADER tmpStreamHeader;
- tmpStreamHeader.fcc = chunktag;
- tmpStreamHeader.cb = chunksize;
-
- _bytesRead += GetLE32(tmpStreamHeader.fccType);
- _bytesRead += GetLE32(tmpStreamHeader.fccHandler);
- _bytesRead += GetLE32(tmpStreamHeader.dwFlags);
- _bytesRead += GetLE16(tmpStreamHeader.wPriority);
- _bytesRead += GetLE16(tmpStreamHeader.wLanguage);
- _bytesRead += GetLE32(tmpStreamHeader.dwInitialFrames);
- _bytesRead += GetLE32(tmpStreamHeader.dwScale);
- _bytesRead += GetLE32(tmpStreamHeader.dwRate);
- _bytesRead += GetLE32(tmpStreamHeader.dwStart);
- _bytesRead += GetLE32(tmpStreamHeader.dwLength);
- _bytesRead += GetLE32(tmpStreamHeader.dwSuggestedBufferSize);
- _bytesRead += GetLE32(tmpStreamHeader.dwQuality);
- _bytesRead += GetLE32(tmpStreamHeader.dwSampleSize);
-
- WebRtc_UWord16 left;
- _bytesRead += GetLE16(left);
- tmpStreamHeader.rcFrame.left = left;
- WebRtc_UWord16 top;
- _bytesRead += GetLE16(top);
- tmpStreamHeader.rcFrame.top = top;
- WebRtc_UWord16 right;
- _bytesRead += GetLE16(right);
- tmpStreamHeader.rcFrame.right = right;
- WebRtc_UWord16 bottom;
- _bytesRead += GetLE16(bottom);
- tmpStreamHeader.rcFrame.bottom = bottom;
-
- if (!readVideoStreamHeader
- && (tmpStreamHeader.fccType == MakeFourCc('v', 'i', 'd', 's')))
- {
- _videoStreamHeader = tmpStreamHeader; //Bitwise copy is OK!
- const WebRtc_Word32 err = ReadAVIVideoStreamHeader(endSeekPos);
- if (err)
- {
- return -1;
- }
- // Make sure there actually is video data in the file...
- if (_videoStreamHeader.dwLength == 0)
- {
- return -1;
- }
- readVideoStreamHeader = true;
- } else if(!readAudioStreamHeader &&
- (tmpStreamHeader.fccType == MakeFourCc('a', 'u', 'd', 's'))) {
- _audioStreamHeader = tmpStreamHeader;
- const WebRtc_Word32 err = ReadAVIAudioStreamHeader(endSeekPos);
- if (err)
- {
- return -1;
- }
- readAudioStreamHeader = true;
- }
- else
- {
- fseek(_aviFile, endSeekPos, SEEK_SET);
- _bytesRead += endSeekPos;
- }
-
- ++streamsRead;
- }
-
- if (!readVideoStreamHeader && !readAudioStreamHeader)
- {
- return -1;
- }
-
- WebRtc_UWord32 tag;
- _bytesRead += GetLE32(tag);
- WebRtc_UWord32 size;
- _bytesRead += GetLE32(size);
-
- if (tag == MakeFourCc('J', 'U', 'N', 'K'))
- {
- fseek(_aviFile, size, SEEK_CUR);
- _bytesRead += size;
- _bytesRead += GetLE32(tag);
- _bytesRead += GetLE32(size);
- }
- if (tag != MakeFourCc('L', 'I', 'S', 'T'))
- {
- return -1;
- }
- WebRtc_UWord32 listTag;
- _bytesRead += GetLE32(listTag);
- if (listTag != MakeFourCc('m', 'o', 'v', 'i'))
- {
- return -1;
- }
- _dataLength = size;
- return 0;
-}
-
-WebRtc_Word32 AviFile::ReadAVIVideoStreamHeader(WebRtc_Word32 endpos)
-{
- WebRtc_UWord32 chunktag;
- _bytesRead += GetLE32(chunktag);
- WebRtc_UWord32 chunksize;
- _bytesRead += GetLE32(chunksize);
-
- if (chunktag != MakeFourCc('s', 't', 'r', 'f'))
- {
- return -1;
- }
-
- _bytesRead += GetLE32(_videoFormatHeader.biSize);
- _bytesRead += GetLE32(_videoFormatHeader.biWidth);
- _bytesRead += GetLE32(_videoFormatHeader.biHeight);
- _bytesRead += GetLE16(_videoFormatHeader.biPlanes);
- _bytesRead += GetLE16(_videoFormatHeader.biBitCount);
- _bytesRead += GetLE32(_videoFormatHeader.biCompression);
- _bytesRead += GetLE32(_videoFormatHeader.biSizeImage);
- _bytesRead += GetLE32(_videoFormatHeader.biXPelsPerMeter);
- _bytesRead += GetLE32(_videoFormatHeader.biYPelsPerMeter);
- _bytesRead += GetLE32(_videoFormatHeader.biClrUsed);
- _bytesRead += GetLE32(_videoFormatHeader.biClrImportant);
-
- if (chunksize > _videoFormatHeader.biSize)
- {
- const WebRtc_UWord32 size = chunksize - _videoFormatHeader.biSize;
- const WebRtc_UWord32 readSize = MinValue(size, CODEC_CONFIG_LENGTH);
- _bytesRead += GetBuffer(
- reinterpret_cast<WebRtc_UWord8*>(_videoConfigParameters), readSize);
- _videoConfigLength = readSize;
- WebRtc_Word32 skipSize = chunksize - _videoFormatHeader.biSize -
- readSize;
- if (skipSize > 0)
- {
- fseek(_aviFile, skipSize, SEEK_CUR);
- _bytesRead += skipSize;
- }
- }
-
- while (static_cast<long>(_bytesRead) < endpos)
- {
- WebRtc_UWord32 chunktag;
- _bytesRead += GetLE32(chunktag);
- WebRtc_UWord32 chunksize;
- _bytesRead += GetLE32(chunksize);
-
- if (chunktag == MakeFourCc('s', 't', 'r', 'n'))
- {
- const WebRtc_UWord32 size = MinValue(chunksize, STREAM_NAME_LENGTH);
- _bytesRead += GetBuffer(
- reinterpret_cast<WebRtc_UWord8*>(_videoStreamName), size);
- }
- else if (chunktag == MakeFourCc('s', 't', 'r', 'd'))
- {
- const WebRtc_UWord32 size = MinValue(chunksize,
- CODEC_CONFIG_LENGTH);
- _bytesRead += GetBuffer(
- reinterpret_cast<WebRtc_UWord8*>(_videoConfigParameters), size);
- _videoConfigLength = size;
- }
- else
- {
- fseek(_aviFile, chunksize, SEEK_CUR);
- _bytesRead += chunksize;
- }
-
- if (feof(_aviFile))
- {
- return -1;
- }
- }
- _videoStream.streamType = AviFile::AVI_VIDEO;
- _videoStream.streamNumber = _nrStreams++;
-
- return 0;
-}
-
-WebRtc_Word32 AviFile::ReadAVIAudioStreamHeader(WebRtc_Word32 endpos)
-{
- WebRtc_UWord32 chunktag;
- _bytesRead += GetLE32(chunktag);
- WebRtc_UWord32 chunksize;
- _bytesRead += GetLE32(chunksize);
-
- if (chunktag != MakeFourCc('s', 't', 'r', 'f'))
- {
- return -1;
- }
-
- const size_t startRead = _bytesRead;
- _bytesRead += GetLE16(_audioFormatHeader.wFormatTag);
- _bytesRead += GetLE16(_audioFormatHeader.nChannels);
- _bytesRead += GetLE32(_audioFormatHeader.nSamplesPerSec);
- _bytesRead += GetLE32(_audioFormatHeader.nAvgBytesPerSec);
- _bytesRead += GetLE16(_audioFormatHeader.nBlockAlign);
- _bytesRead += GetLE16(_audioFormatHeader.wBitsPerSample);
- _bytesRead += GetLE16(_audioFormatHeader.cbSize);
-
- const WebRtc_UWord32 diffRead = chunksize - (_bytesRead - startRead);
- if (diffRead > 0)
- {
- const WebRtc_UWord32 size = MinValue(diffRead, CODEC_CONFIG_LENGTH);
- _bytesRead += GetBuffer(
- reinterpret_cast<WebRtc_UWord8*>(_audioConfigParameters), size);
- }
-
- while (static_cast<long>(_bytesRead) < endpos)
- {
- WebRtc_UWord32 chunktag;
- _bytesRead += GetLE32(chunktag);
- WebRtc_UWord32 chunksize;
- _bytesRead += GetLE32(chunksize);
-
- if (chunktag == MakeFourCc('s', 't', 'r', 'n'))
- {
- const WebRtc_UWord32 size = MinValue(chunksize, STREAM_NAME_LENGTH);
- _bytesRead += GetBuffer(
- reinterpret_cast<WebRtc_UWord8*>(_audioStreamName), size);
- }
- else if (chunktag == MakeFourCc('s', 't', 'r', 'd'))
- {
- const WebRtc_UWord32 size = MinValue(chunksize,
- CODEC_CONFIG_LENGTH);
- _bytesRead += GetBuffer(
- reinterpret_cast<WebRtc_UWord8*>(_audioConfigParameters), size);
- }
- else
- {
- fseek(_aviFile, chunksize, SEEK_CUR);
- _bytesRead += chunksize;
- }
-
- if (feof(_aviFile))
- {
- return -1;
- }
- }
- _audioStream.streamType = AviFile::AVI_AUDIO;
- _audioStream.streamNumber = _nrStreams++;
- return 0;
-}
-
-WebRtc_UWord32 AviFile::StreamAndTwoCharCodeToTag(WebRtc_Word32 streamNum,
- const char* twoCharCode)
-{
- WebRtc_UWord8 a = '0';
- WebRtc_UWord8 b;
- switch (streamNum)
- {
- case 1:
- b = '1';
- break;
- case 2:
- b = '2';
- break;
- default:
- b = '0';
- }
- return MakeFourCc(a, b, twoCharCode[0], twoCharCode[1]);
-}
-
-void AviFile::ClearIndexList()
-{
- while (!_indexList->Empty())
- {
- ListItem* listItem = _indexList->First();
- if (listItem == 0)
- {
- break;
- }
-
- AVIINDEXENTRY* item = static_cast<AVIINDEXENTRY*>(listItem->GetItem());
- if (item != NULL)
- {
- delete item;
- }
- _indexList->PopFront();
- }
-}
-
-void AviFile::AddChunkToIndexList(WebRtc_UWord32 inChunkId,
- WebRtc_UWord32 inFlags,
- WebRtc_UWord32 inOffset,
- WebRtc_UWord32 inSize)
-{
- _indexList->PushBack(new AVIINDEXENTRY(inChunkId, inFlags, inOffset,
- inSize));
-}
-
-void AviFile::WriteIndex()
-{
- const WebRtc_UWord32 idxTag = MakeFourCc('i', 'd', 'x', '1');
- _bytesWritten += PutLE32(idxTag);
-
- // Size is unknown at this point. Update later.
- _bytesWritten += PutLE32(0);
- const size_t idxChunkSize = _bytesWritten;
-
- for (ListItem* listItem = _indexList->First();
- listItem != NULL;
- listItem = _indexList->Next(listItem))
- {
- const AVIINDEXENTRY* item =
- static_cast<AVIINDEXENTRY*>(listItem->GetItem());
- if (item != NULL)
- {
- _bytesWritten += PutLE32(item->ckid);
- _bytesWritten += PutLE32(item->dwFlags);
- _bytesWritten += PutLE32(item->dwChunkOffset);
- _bytesWritten += PutLE32(item->dwChunkLength);
- }
- }
- PutLE32LengthFromCurrent(static_cast<long>(idxChunkSize));
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/media_file/source/avi_file.h b/trunk/src/modules/media_file/source/avi_file.h
deleted file mode 100644
index 73a19af..0000000
--- a/trunk/src/modules/media_file/source/avi_file.h
+++ /dev/null
@@ -1,278 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Class for reading (x)or writing to an AVI file.
-// Note: the class cannot be used for reading and writing at the same time.
-#ifndef WEBRTC_MODULES_MEDIA_FILE_SOURCE_AVI_FILE_H_
-#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_AVI_FILE_H_
-
-#include <stdio.h>
-
-#include "typedefs.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class ListWrapper;
-
-struct AVISTREAMHEADER
-{
- AVISTREAMHEADER();
- WebRtc_UWord32 fcc;
- WebRtc_UWord32 cb;
- WebRtc_UWord32 fccType;
- WebRtc_UWord32 fccHandler;
- WebRtc_UWord32 dwFlags;
- WebRtc_UWord16 wPriority;
- WebRtc_UWord16 wLanguage;
- WebRtc_UWord32 dwInitialFrames;
- WebRtc_UWord32 dwScale;
- WebRtc_UWord32 dwRate;
- WebRtc_UWord32 dwStart;
- WebRtc_UWord32 dwLength;
- WebRtc_UWord32 dwSuggestedBufferSize;
- WebRtc_UWord32 dwQuality;
- WebRtc_UWord32 dwSampleSize;
- struct
- {
- WebRtc_Word16 left;
- WebRtc_Word16 top;
- WebRtc_Word16 right;
- WebRtc_Word16 bottom;
- } rcFrame;
-};
-
-struct BITMAPINFOHEADER
-{
- BITMAPINFOHEADER();
- WebRtc_UWord32 biSize;
- WebRtc_UWord32 biWidth;
- WebRtc_UWord32 biHeight;
- WebRtc_UWord16 biPlanes;
- WebRtc_UWord16 biBitCount;
- WebRtc_UWord32 biCompression;
- WebRtc_UWord32 biSizeImage;
- WebRtc_UWord32 biXPelsPerMeter;
- WebRtc_UWord32 biYPelsPerMeter;
- WebRtc_UWord32 biClrUsed;
- WebRtc_UWord32 biClrImportant;
-};
-
-struct WAVEFORMATEX
-{
- WAVEFORMATEX();
- WebRtc_UWord16 wFormatTag;
- WebRtc_UWord16 nChannels;
- WebRtc_UWord32 nSamplesPerSec;
- WebRtc_UWord32 nAvgBytesPerSec;
- WebRtc_UWord16 nBlockAlign;
- WebRtc_UWord16 wBitsPerSample;
- WebRtc_UWord16 cbSize;
-};
-
-class AviFile
-{
-public:
- enum AVIStreamType
- {
- AVI_AUDIO = 0,
- AVI_VIDEO = 1
- };
-
- // Unsigned, for comparison with must-be-unsigned types.
- static const unsigned int CODEC_CONFIG_LENGTH = 64;
- static const unsigned int STREAM_NAME_LENGTH = 32;
-
- AviFile();
- ~AviFile();
-
- WebRtc_Word32 Open(AVIStreamType streamType, const char* fileName,
- bool loop = false);
-
- WebRtc_Word32 CreateVideoStream(const AVISTREAMHEADER& videoStreamHeader,
- const BITMAPINFOHEADER& bitMapInfoHeader,
- const WebRtc_UWord8* codecConfigParams,
- WebRtc_Word32 codecConfigParamsLength);
-
- WebRtc_Word32 CreateAudioStream(const AVISTREAMHEADER& audioStreamHeader,
- const WAVEFORMATEX& waveFormatHeader);
- WebRtc_Word32 Create(const WebRtc_Word8* fileName);
-
- WebRtc_Word32 WriteAudio(const WebRtc_UWord8* data, WebRtc_Word32 length);
- WebRtc_Word32 WriteVideo(const WebRtc_UWord8* data, WebRtc_Word32 length);
-
- WebRtc_Word32 GetVideoStreamInfo(AVISTREAMHEADER& videoStreamHeader,
- BITMAPINFOHEADER& bitmapInfo,
- WebRtc_Word8* codecConfigParameters,
- WebRtc_Word32& configLength);
-
- WebRtc_Word32 GetDuration(WebRtc_Word32& durationMs);
-
- WebRtc_Word32 GetAudioStreamInfo(WAVEFORMATEX& waveHeader);
-
- WebRtc_Word32 ReadAudio(WebRtc_UWord8* data, WebRtc_Word32& length);
- WebRtc_Word32 ReadVideo(WebRtc_UWord8* data, WebRtc_Word32& length);
-
- WebRtc_Word32 Close();
-
- static WebRtc_UWord32 MakeFourCc(WebRtc_UWord8 ch0, WebRtc_UWord8 ch1,
- WebRtc_UWord8 ch2, WebRtc_UWord8 ch3);
-
-private:
- enum AVIFileMode
- {
- NotSet,
- Read,
- Write
- };
-
- struct AVIINDEXENTRY
- {
- AVIINDEXENTRY(WebRtc_UWord32 inckid, WebRtc_UWord32 indwFlags,
- WebRtc_UWord32 indwChunkOffset,
- WebRtc_UWord32 indwChunkLength);
- WebRtc_UWord32 ckid;
- WebRtc_UWord32 dwFlags;
- WebRtc_UWord32 dwChunkOffset;
- WebRtc_UWord32 dwChunkLength;
- };
-
- WebRtc_Word32 PrepareDataChunkHeaders();
-
- WebRtc_Word32 ReadMoviSubChunk(WebRtc_UWord8* data, WebRtc_Word32& length,
- WebRtc_UWord32 tag1,
- WebRtc_UWord32 tag2 = 0);
-
- WebRtc_Word32 WriteRIFF();
- WebRtc_Word32 WriteHeaders();
- WebRtc_Word32 WriteAVIMainHeader();
- WebRtc_Word32 WriteAVIStreamHeaders();
- WebRtc_Word32 WriteAVIVideoStreamHeaders();
- WebRtc_Word32 WriteAVIVideoStreamHeaderChunks();
- WebRtc_Word32 WriteAVIAudioStreamHeaders();
- WebRtc_Word32 WriteAVIAudioStreamHeaderChunks();
-
- WebRtc_Word32 WriteMoviStart();
-
- size_t PutByte(WebRtc_UWord8 byte);
- size_t PutLE16(WebRtc_UWord16 word);
- size_t PutLE32(WebRtc_UWord32 word);
- size_t PutBuffer(const WebRtc_UWord8* str, size_t size);
- size_t PutBufferZ(const char* str);
- long PutLE32LengthFromCurrent(long startPos);
- void PutLE32AtPos(long pos, WebRtc_UWord32 word);
-
- size_t GetByte(WebRtc_UWord8& word);
- size_t GetLE16(WebRtc_UWord16& word);
- size_t GetLE32(WebRtc_UWord32& word);
- size_t GetBuffer(WebRtc_UWord8* str, size_t size);
-
- void CloseRead();
- void CloseWrite();
-
- void ResetMembers();
- void ResetComplexMembers();
-
- WebRtc_Word32 ReadRIFF();
- WebRtc_Word32 ReadHeaders();
- WebRtc_Word32 ReadAVIMainHeader();
- WebRtc_Word32 ReadAVIVideoStreamHeader(WebRtc_Word32 endpos);
- WebRtc_Word32 ReadAVIAudioStreamHeader(WebRtc_Word32 endpos);
-
- WebRtc_UWord32 StreamAndTwoCharCodeToTag(WebRtc_Word32 streamNum,
- const char* twoCharCode);
-
- void ClearIndexList();
- void AddChunkToIndexList(WebRtc_UWord32 inChunkId, WebRtc_UWord32 inFlags,
- WebRtc_UWord32 inOffset, WebRtc_UWord32 inSize);
-
- void WriteIndex();
-
-private:
- struct AVIMAINHEADER
- {
- AVIMAINHEADER();
- WebRtc_UWord32 fcc;
- WebRtc_UWord32 cb;
- WebRtc_UWord32 dwMicroSecPerFrame;
- WebRtc_UWord32 dwMaxBytesPerSec;
- WebRtc_UWord32 dwPaddingGranularity;
- WebRtc_UWord32 dwFlags;
- WebRtc_UWord32 dwTotalFrames;
- WebRtc_UWord32 dwInitialFrames;
- WebRtc_UWord32 dwStreams;
- WebRtc_UWord32 dwSuggestedBufferSize;
- WebRtc_UWord32 dwWidth;
- WebRtc_UWord32 dwHeight;
- WebRtc_UWord32 dwReserved[4];
- };
-
- struct AVIStream
- {
- AVIStreamType streamType;
- int streamNumber;
- };
-
- CriticalSectionWrapper* _crit;
- FILE* _aviFile;
- AVIMAINHEADER _aviHeader;
- AVISTREAMHEADER _videoStreamHeader;
- AVISTREAMHEADER _audioStreamHeader;
- BITMAPINFOHEADER _videoFormatHeader;
- WAVEFORMATEX _audioFormatHeader;
-
- WebRtc_Word8 _videoConfigParameters[CODEC_CONFIG_LENGTH];
- WebRtc_Word32 _videoConfigLength;
- WebRtc_Word8 _videoStreamName[STREAM_NAME_LENGTH];
- WebRtc_Word32 _videoStreamNameLength;
- WebRtc_Word8 _audioConfigParameters[CODEC_CONFIG_LENGTH];
- WebRtc_Word8 _audioStreamName[STREAM_NAME_LENGTH];
-
- AVIStream _videoStream;
- AVIStream _audioStream;
-
- WebRtc_Word32 _nrStreams;
- WebRtc_Word32 _aviLength;
- WebRtc_Word32 _dataLength;
- size_t _bytesRead;
- size_t _dataStartByte;
- WebRtc_Word32 _framesRead;
- WebRtc_Word32 _videoFrames;
- WebRtc_Word32 _audioFrames;
-
- bool _reading;
- AVIStreamType _openedAs;
- bool _loop;
- bool _writing;
-
- size_t _bytesWritten;
-
- size_t _riffSizeMark;
- size_t _moviSizeMark;
- size_t _totNumFramesMark;
- size_t _videoStreamLengthMark;
- size_t _audioStreamLengthMark;
- WebRtc_Word32 _moviListOffset;
-
- bool _writeAudioStream;
- bool _writeVideoStream;
-
- AVIFileMode _aviMode;
- WebRtc_UWord8* _videoCodecConfigParams;
- WebRtc_Word32 _videoCodecConfigParamsLength;
-
- WebRtc_UWord32 _videoStreamDataChunkPrefix;
- WebRtc_UWord32 _audioStreamDataChunkPrefix;
- bool _created;
-
- ListWrapper* _indexList; // Elements are of type AVIINDEXENTRY.
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_AVI_FILE_H_
diff --git a/trunk/src/modules/media_file/source/media_file.gypi b/trunk/src/modules/media_file/source/media_file.gypi
deleted file mode 100644
index 2f9ffe9..0000000
--- a/trunk/src/modules/media_file/source/media_file.gypi
+++ /dev/null
@@ -1,67 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'media_file',
- 'type': '<(library)',
- 'dependencies': [
- 'webrtc_utility',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'defines': [
- 'WEBRTC_MODULE_UTILITY_VIDEO', # for compiling support for video recording
- ],
- 'include_dirs': [
- '../interface',
- '../../interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- '../../interface',
- ],
- },
- 'sources': [
- '../interface/media_file.h',
- '../interface/media_file_defines.h',
- 'avi_file.cc',
- 'avi_file.h',
- 'media_file_impl.cc',
- 'media_file_impl.h',
- 'media_file_utility.cc',
- 'media_file_utility.h',
- ], # source
- },
- ], # targets
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'media_file_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'media_file',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- ],
- 'sources': [
- 'media_file_unittest.cc',
- ],
- }, # media_file_unittests
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/media_file/source/media_file_impl.cc b/trunk/src/modules/media_file/source/media_file_impl.cc
deleted file mode 100644
index 6c42fa1..0000000
--- a/trunk/src/modules/media_file/source/media_file_impl.cc
+++ /dev/null
@@ -1,1372 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <assert.h>
-
-#include "critical_section_wrapper.h"
-#include "file_wrapper.h"
-#include "media_file_impl.h"
-#include "tick_util.h"
-#include "trace.h"
-
-#if (defined(WIN32) || defined(WINCE))
- #define STR_CASE_CMP _stricmp
- #define STR_NCASE_CMP _strnicmp
-#else
- #define STR_CASE_CMP strcasecmp
- #define STR_NCASE_CMP strncasecmp
-#endif
-
-namespace webrtc {
-MediaFile* MediaFile::CreateMediaFile(const WebRtc_Word32 id)
-{
- return new MediaFileImpl(id);
-}
-
-void MediaFile::DestroyMediaFile(MediaFile* module)
-{
- delete static_cast<MediaFileImpl*>(module);
-}
-
-MediaFileImpl::MediaFileImpl(const WebRtc_Word32 id)
- : _id(id),
- _crit(CriticalSectionWrapper::CreateCriticalSection()),
- _callbackCrit(CriticalSectionWrapper::CreateCriticalSection()),
- _ptrFileUtilityObj(NULL),
- codec_info_(),
- _ptrInStream(NULL),
- _ptrOutStream(NULL),
- _fileFormat((FileFormats)-1),
- _recordDurationMs(0),
- _playoutPositionMs(0),
- _notificationMs(0),
- _playingActive(false),
- _recordingActive(false),
- _isStereo(false),
- _openFile(false),
- _fileName(),
- _ptrCallback(NULL)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceFile, id, "Created");
-
- codec_info_.plname[0] = '\0';
- _fileName[0] = '\0';
-}
-
-
-MediaFileImpl::~MediaFileImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceFile, _id, "~MediaFileImpl()");
- {
- CriticalSectionScoped lock(_crit);
-
- if(_playingActive)
- {
- StopPlaying();
- }
-
- if(_recordingActive)
- {
- StopRecording();
- }
-
- delete _ptrFileUtilityObj;
-
- if(_openFile)
- {
- delete _ptrInStream;
- _ptrInStream = NULL;
- delete _ptrOutStream;
- _ptrOutStream = NULL;
- }
- }
-
- delete _crit;
- delete _callbackCrit;
-}
-
-WebRtc_Word32 MediaFileImpl::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
- return 0;
-}
-
-WebRtc_Word32 MediaFileImpl::TimeUntilNextProcess()
-{
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceFile,
- _id,
- "TimeUntilNextProcess: This method is not used by MediaFile class.");
- return -1;
-}
-
-WebRtc_Word32 MediaFileImpl::Process()
-{
- WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
- "Process: This method is not used by MediaFile class.");
- return -1;
-}
-
-WebRtc_Word32 MediaFileImpl::PlayoutAVIVideoData(
- WebRtc_Word8* buffer,
- WebRtc_UWord32& dataLengthInBytes)
-{
- return PlayoutData( buffer, dataLengthInBytes, true);
-}
-
-WebRtc_Word32 MediaFileImpl::PlayoutAudioData(WebRtc_Word8* buffer,
- WebRtc_UWord32& dataLengthInBytes)
-{
- return PlayoutData( buffer, dataLengthInBytes, false);
-}
-
-WebRtc_Word32 MediaFileImpl::PlayoutData(WebRtc_Word8* buffer,
- WebRtc_UWord32& dataLengthInBytes,
- bool video)
-{
- WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
- "MediaFileImpl::PlayoutData(buffer= 0x%x, bufLen= %ld)",
- buffer, dataLengthInBytes);
-
- const WebRtc_UWord32 bufferLengthInBytes = dataLengthInBytes;
- dataLengthInBytes = 0;
-
- if(buffer == NULL || bufferLengthInBytes == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Buffer pointer or length is NULL!");
- return -1;
- }
-
- WebRtc_Word32 bytesRead = 0;
- {
- CriticalSectionScoped lock(_crit);
-
- if(!_playingActive)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
- "Not currently playing!");
- return -1;
- }
-
- if(!_ptrFileUtilityObj)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Playing, but no FileUtility object!");
- StopPlaying();
- return -1;
- }
-
- switch(_fileFormat)
- {
- case kFileFormatPcm32kHzFile:
- case kFileFormatPcm16kHzFile:
- case kFileFormatPcm8kHzFile:
- bytesRead = _ptrFileUtilityObj->ReadPCMData(
- *_ptrInStream,
- buffer,
- bufferLengthInBytes);
- break;
- case kFileFormatCompressedFile:
- bytesRead = _ptrFileUtilityObj->ReadCompressedData(
- *_ptrInStream,
- buffer,
- bufferLengthInBytes);
- break;
- case kFileFormatWavFile:
- bytesRead = _ptrFileUtilityObj->ReadWavDataAsMono(
- *_ptrInStream,
- buffer,
- bufferLengthInBytes);
- break;
- case kFileFormatPreencodedFile:
- bytesRead = _ptrFileUtilityObj->ReadPreEncodedData(
- *_ptrInStream,
- buffer,
- bufferLengthInBytes);
- if(bytesRead > 0)
- {
- dataLengthInBytes = bytesRead;
- return 0;
- }
- break;
- case kFileFormatAviFile:
- {
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
- if(video)
- {
- bytesRead = _ptrFileUtilityObj->ReadAviVideoData(
- buffer,
- bufferLengthInBytes);
- }
- else
- {
- bytesRead = _ptrFileUtilityObj->ReadAviAudioData(
- buffer,
- bufferLengthInBytes);
- }
- break;
-#else
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Invalid file format: %d", kFileFormatAviFile);
- assert(false);
- break;
-#endif
- }
- }
-
- if( bytesRead > 0)
- {
- dataLengthInBytes =(WebRtc_UWord32) bytesRead;
- }
- }
- HandlePlayCallbacks(bytesRead);
- return 0;
-}
-
-void MediaFileImpl::HandlePlayCallbacks(WebRtc_Word32 bytesRead)
-{
- bool playEnded = false;
- WebRtc_UWord32 callbackNotifyMs = 0;
-
- if(bytesRead > 0)
- {
- // Check if it's time for PlayNotification(..).
- _playoutPositionMs = _ptrFileUtilityObj->PlayoutPositionMs();
- if(_notificationMs)
- {
- if(_playoutPositionMs >= _notificationMs)
- {
- _notificationMs = 0;
- callbackNotifyMs = _playoutPositionMs;
- }
- }
- }
- else
- {
- // If no bytes were read assume end of file.
- StopPlaying();
- playEnded = true;
- }
-
- // Only _callbackCrit may and should be taken when making callbacks.
- CriticalSectionScoped lock(_callbackCrit);
- if(_ptrCallback)
- {
- if(callbackNotifyMs)
- {
- _ptrCallback->PlayNotification(_id, callbackNotifyMs);
- }
- if(playEnded)
- {
- _ptrCallback->PlayFileEnded(_id);
- }
- }
-}
-
-WebRtc_Word32 MediaFileImpl::PlayoutStereoData(
- WebRtc_Word8* bufferLeft,
- WebRtc_Word8* bufferRight,
- WebRtc_UWord32& dataLengthInBytes)
-{
- WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
- "MediaFileImpl::PlayoutStereoData(Left = 0x%x, Right = 0x%x,\
- Len= %ld)",
- bufferLeft,
- bufferRight,
- dataLengthInBytes);
-
- const WebRtc_UWord32 bufferLengthInBytes = dataLengthInBytes;
- dataLengthInBytes = 0;
-
- if(bufferLeft == NULL || bufferRight == NULL || bufferLengthInBytes == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "A buffer pointer or the length is NULL!");
- return -1;
- }
-
- bool playEnded = false;
- WebRtc_UWord32 callbackNotifyMs = 0;
- {
- CriticalSectionScoped lock(_crit);
-
- if(!_playingActive || !_isStereo)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
- "Not currently playing stereo!");
- return -1;
- }
-
- if(!_ptrFileUtilityObj)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceFile,
- _id,
- "Playing stereo, but the FileUtility objects is NULL!");
- StopPlaying();
- return -1;
- }
-
- // Stereo playout only supported for WAV files.
- WebRtc_Word32 bytesRead = 0;
- switch(_fileFormat)
- {
- case kFileFormatWavFile:
- bytesRead = _ptrFileUtilityObj->ReadWavDataAsStereo(
- *_ptrInStream,
- bufferLeft,
- bufferRight,
- bufferLengthInBytes);
- break;
- default:
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Trying to read non-WAV as stereo audio\
- (not supported)");
- break;
- }
-
- if(bytesRead > 0)
- {
- dataLengthInBytes = bytesRead;
-
- // Check if it's time for PlayNotification(..).
- _playoutPositionMs = _ptrFileUtilityObj->PlayoutPositionMs();
- if(_notificationMs)
- {
- if(_playoutPositionMs >= _notificationMs)
- {
- _notificationMs = 0;
- callbackNotifyMs = _playoutPositionMs;
- }
- }
- }
- else
- {
- // If no bytes were read assume end of file.
- StopPlaying();
- playEnded = true;
- }
- }
-
- CriticalSectionScoped lock(_callbackCrit);
- if(_ptrCallback)
- {
- if(callbackNotifyMs)
- {
- _ptrCallback->PlayNotification(_id, callbackNotifyMs);
- }
- if(playEnded)
- {
- _ptrCallback->PlayFileEnded(_id);
- }
- }
- return 0;
-}
-
-WebRtc_Word32 MediaFileImpl::StartPlayingAudioFile(
- const WebRtc_Word8* fileName,
- const WebRtc_UWord32 notificationTimeMs,
- const bool loop,
- const FileFormats format,
- const CodecInst* codecInst,
- const WebRtc_UWord32 startPointMs,
- const WebRtc_UWord32 stopPointMs)
-{
- const bool videoOnly = false;
- return StartPlayingFile(fileName, notificationTimeMs, loop, videoOnly,
- format, codecInst, startPointMs, stopPointMs);
-}
-
-
-WebRtc_Word32 MediaFileImpl::StartPlayingVideoFile(const WebRtc_Word8* fileName,
- const bool loop,
- bool videoOnly,
- const FileFormats format)
-{
-
- const WebRtc_UWord32 notificationTimeMs = 0;
- const WebRtc_UWord32 startPointMs = 0;
- const WebRtc_UWord32 stopPointMs = 0;
- return StartPlayingFile(fileName, notificationTimeMs, loop, videoOnly,
- format, 0, startPointMs, stopPointMs);
-}
-
-WebRtc_Word32 MediaFileImpl::StartPlayingFile(
- const WebRtc_Word8* fileName,
- const WebRtc_UWord32 notificationTimeMs,
- const bool loop,
- bool videoOnly,
- const FileFormats format,
- const CodecInst* codecInst,
- const WebRtc_UWord32 startPointMs,
- const WebRtc_UWord32 stopPointMs)
-{
-
- if(!ValidFileName(fileName))
- {
- return -1;
- }
- if(!ValidFileFormat(format,codecInst))
- {
- return -1;
- }
- if(!ValidFilePositions(startPointMs,stopPointMs))
- {
- return -1;
- }
-
- // Check that the file will play longer than notificationTimeMs ms.
- if((startPointMs && stopPointMs && !loop) &&
- (notificationTimeMs > (stopPointMs - startPointMs)))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceFile,
- _id,
- "specified notification time is longer than amount of ms that will\
- be played");
- return -1;
- }
-
- FileWrapper* inputStream = FileWrapper::Create();
- if(inputStream == NULL)
- {
- WEBRTC_TRACE(kTraceMemory, kTraceFile, _id,
- "Failed to allocate input stream for file %s", fileName);
- return -1;
- }
-
- // TODO (hellner): make all formats support reading from stream.
- bool useStream = (format != kFileFormatAviFile);
- if( useStream)
- {
- if(inputStream->OpenFile(fileName, true, loop) != 0)
- {
- delete inputStream;
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Could not open input file %s", fileName);
- return -1;
- }
- }
-
- if(StartPlayingStream(*inputStream, fileName, loop, notificationTimeMs,
- format, codecInst, startPointMs, stopPointMs,
- videoOnly) == -1)
- {
- if( useStream)
- {
- inputStream->CloseFile();
- }
- delete inputStream;
- return -1;
- }
-
- CriticalSectionScoped lock(_crit);
- _openFile = true;
- strncpy(_fileName, fileName, sizeof(_fileName));
- _fileName[sizeof(_fileName) - 1] = '\0';
- return 0;
-}
-
-WebRtc_Word32 MediaFileImpl::StartPlayingAudioStream(
- InStream& stream,
- const WebRtc_UWord32 notificationTimeMs,
- const FileFormats format,
- const CodecInst* codecInst,
- const WebRtc_UWord32 startPointMs,
- const WebRtc_UWord32 stopPointMs)
-{
- return StartPlayingStream(stream, 0, false, notificationTimeMs, format,
- codecInst, startPointMs, stopPointMs);
-}
-
-WebRtc_Word32 MediaFileImpl::StartPlayingStream(
- InStream& stream,
- const WebRtc_Word8* filename,
- bool loop,
- const WebRtc_UWord32 notificationTimeMs,
- const FileFormats format,
- const CodecInst* codecInst,
- const WebRtc_UWord32 startPointMs,
- const WebRtc_UWord32 stopPointMs,
- bool videoOnly)
-{
- if(!ValidFileFormat(format,codecInst))
- {
- return -1;
- }
-
- if(!ValidFilePositions(startPointMs,stopPointMs))
- {
- return -1;
- }
-
- CriticalSectionScoped lock(_crit);
- if(_playingActive || _recordingActive)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceFile,
- _id,
- "StartPlaying called, but already playing or recording file %s",
- (_fileName[0] == '\0') ? "(name not set)" : _fileName);
- return -1;
- }
-
- if(_ptrFileUtilityObj != NULL)
- {
- WEBRTC_TRACE(kTraceError,
- kTraceFile,
- _id,
- "StartPlaying called, but FileUtilityObj already exists!");
- StopPlaying();
- return -1;
- }
-
- _ptrFileUtilityObj = new ModuleFileUtility(_id);
- if(_ptrFileUtilityObj == NULL)
- {
- WEBRTC_TRACE(kTraceMemory, kTraceFile, _id,
- "Failed to create FileUtilityObj!");
- return -1;
- }
-
- switch(format)
- {
- case kFileFormatWavFile:
- {
- if(_ptrFileUtilityObj->InitWavReading(stream, startPointMs,
- stopPointMs) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Not a valid WAV file!");
- StopPlaying();
- return -1;
- }
- _fileFormat = kFileFormatWavFile;
- break;
- }
- case kFileFormatCompressedFile:
- {
- if(_ptrFileUtilityObj->InitCompressedReading(stream, startPointMs,
- stopPointMs) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Not a valid Compressed file!");
- StopPlaying();
- return -1;
- }
- _fileFormat = kFileFormatCompressedFile;
- break;
- }
- case kFileFormatPcm8kHzFile:
- case kFileFormatPcm16kHzFile:
- case kFileFormatPcm32kHzFile:
- {
- // ValidFileFormat() called in the beginneing of this function
- // prevents codecInst from being NULL here.
- assert(codecInst != NULL);
- if(!ValidFrequency(codecInst->plfreq) ||
- _ptrFileUtilityObj->InitPCMReading(stream, startPointMs,
- stopPointMs,
- codecInst->plfreq) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Not a valid raw 8 or 16 KHz PCM file!");
- StopPlaying();
- return -1;
- }
-
- _fileFormat = format;
- break;
- }
- case kFileFormatPreencodedFile:
- {
- // ValidFileFormat() called in the beginneing of this function
- // prevents codecInst from being NULL here.
- assert(codecInst != NULL);
- if(_ptrFileUtilityObj->InitPreEncodedReading(stream, *codecInst) ==
- -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Not a valid PreEncoded file!");
- StopPlaying();
- return -1;
- }
-
- _fileFormat = kFileFormatPreencodedFile;
- break;
- }
- case kFileFormatAviFile:
- {
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
- if(_ptrFileUtilityObj->InitAviReading( filename, videoOnly, loop))
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Not a valid AVI file!");
- StopPlaying();
-
- return -1;
- }
-
- _ptrFileUtilityObj->codec_info(codec_info_);
-
- _fileFormat = kFileFormatAviFile;
- break;
-#else
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Invalid file format: %d", kFileFormatAviFile);
- assert(false);
- break;
-#endif
- }
- }
- if(_ptrFileUtilityObj->codec_info(codec_info_) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Failed to retrieve codec info!");
- StopPlaying();
- return -1;
- }
-
- _isStereo = (codec_info_.channels == 2);
- if(_isStereo && (_fileFormat != kFileFormatWavFile))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
- "Stereo is only allowed for WAV files");
- StopPlaying();
- return -1;
- }
- _playingActive = true;
- _playoutPositionMs = _ptrFileUtilityObj->PlayoutPositionMs();
- _ptrInStream = &stream;
- _notificationMs = notificationTimeMs;
-
- return 0;
-}
-
-WebRtc_Word32 MediaFileImpl::StopPlaying()
-{
-
- CriticalSectionScoped lock(_crit);
- _isStereo = false;
- if(_ptrFileUtilityObj)
- {
- delete _ptrFileUtilityObj;
- _ptrFileUtilityObj = NULL;
- }
- if(_ptrInStream)
- {
- // If MediaFileImpl opened the InStream it must be reclaimed here.
- if(_openFile)
- {
- delete _ptrInStream;
- _openFile = false;
- }
- _ptrInStream = NULL;
- }
-
- codec_info_.pltype = 0;
- codec_info_.plname[0] = '\0';
-
- if(!_playingActive)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
- "playing is not active!");
- return -1;
- }
-
- _playingActive = false;
- return 0;
-}
-
-bool MediaFileImpl::IsPlaying()
-{
- WEBRTC_TRACE(kTraceStream, kTraceFile, _id, "MediaFileImpl::IsPlaying()");
- CriticalSectionScoped lock(_crit);
- return _playingActive;
-}
-
-WebRtc_Word32 MediaFileImpl::IncomingAudioData(
- const WebRtc_Word8* buffer,
- const WebRtc_UWord32 bufferLengthInBytes)
-{
- return IncomingAudioVideoData( buffer, bufferLengthInBytes, false);
-}
-
-WebRtc_Word32 MediaFileImpl::IncomingAVIVideoData(
- const WebRtc_Word8* buffer,
- const WebRtc_UWord32 bufferLengthInBytes)
-{
- return IncomingAudioVideoData( buffer, bufferLengthInBytes, true);
-}
-
-WebRtc_Word32 MediaFileImpl::IncomingAudioVideoData(
- const WebRtc_Word8* buffer,
- const WebRtc_UWord32 bufferLengthInBytes,
- const bool video)
-{
- WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
- "MediaFile::IncomingData(buffer= 0x%x, bufLen= %hd",
- buffer, bufferLengthInBytes);
-
- if(buffer == NULL || bufferLengthInBytes == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Buffer pointer or length is NULL!");
- return -1;
- }
-
- bool recordingEnded = false;
- WebRtc_UWord32 callbackNotifyMs = 0;
- {
- CriticalSectionScoped lock(_crit);
-
- if(!_recordingActive)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
- "Not currently recording!");
- return -1;
- }
- if(_ptrOutStream == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Recording is active, but output stream is NULL!");
- assert(false);
- return -1;
- }
-
- WebRtc_Word32 bytesWritten = 0;
- WebRtc_UWord32 samplesWritten = codec_info_.pacsize;
- if(_ptrFileUtilityObj)
- {
- switch(_fileFormat)
- {
- case kFileFormatPcm8kHzFile:
- case kFileFormatPcm16kHzFile:
- case kFileFormatPcm32kHzFile:
- bytesWritten = _ptrFileUtilityObj->WritePCMData(
- *_ptrOutStream,
- buffer,
- bufferLengthInBytes);
-
- // Sample size is 2 bytes.
- if(bytesWritten > 0)
- {
- samplesWritten = bytesWritten/sizeof(WebRtc_Word16);
- }
- break;
- case kFileFormatCompressedFile:
- bytesWritten = _ptrFileUtilityObj->WriteCompressedData(
- *_ptrOutStream, buffer, bufferLengthInBytes);
- break;
- case kFileFormatWavFile:
- bytesWritten = _ptrFileUtilityObj->WriteWavData(
- *_ptrOutStream,
- buffer,
- bufferLengthInBytes);
- if(bytesWritten > 0 && STR_NCASE_CMP(codec_info_.plname,
- "L16", 4) == 0)
- {
- // Sample size is 2 bytes.
- samplesWritten = bytesWritten/sizeof(WebRtc_Word16);
- }
- break;
- case kFileFormatPreencodedFile:
- bytesWritten = _ptrFileUtilityObj->WritePreEncodedData(
- *_ptrOutStream, buffer, bufferLengthInBytes);
- break;
- case kFileFormatAviFile:
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
- if(video)
- {
- bytesWritten = _ptrFileUtilityObj->WriteAviVideoData(
- buffer, bufferLengthInBytes);
- }else
- {
- bytesWritten = _ptrFileUtilityObj->WriteAviAudioData(
- buffer, bufferLengthInBytes);
- }
- break;
-#else
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Invalid file format: %d", kFileFormatAviFile);
- assert(false);
- break;
-#endif
- }
- } else {
- // TODO (hellner): quick look at the code makes me think that this
- // code is never executed. Remove?
- if(_ptrOutStream)
- {
- if(_ptrOutStream->Write(buffer, bufferLengthInBytes))
- {
- bytesWritten = bufferLengthInBytes;
- }
- }
- }
-
- if(!video)
- {
- _recordDurationMs += samplesWritten / (codec_info_.plfreq / 1000);
- }
-
- // Check if it's time for RecordNotification(..).
- if(_notificationMs)
- {
- if(_recordDurationMs >= _notificationMs)
- {
- _notificationMs = 0;
- callbackNotifyMs = _recordDurationMs;
- }
- }
- if(bytesWritten < (WebRtc_Word32)bufferLengthInBytes)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
- "Failed to write all requested bytes!");
- StopRecording();
- recordingEnded = true;
- }
- }
-
- // Only _callbackCrit may and should be taken when making callbacks.
- CriticalSectionScoped lock(_callbackCrit);
- if(_ptrCallback)
- {
- if(callbackNotifyMs)
- {
- _ptrCallback->RecordNotification(_id, callbackNotifyMs);
- }
- if(recordingEnded)
- {
- _ptrCallback->RecordFileEnded(_id);
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_Word32 MediaFileImpl::StartRecordingAudioFile(
- const WebRtc_Word8* fileName,
- const FileFormats format,
- const CodecInst& codecInst,
- const WebRtc_UWord32 notificationTimeMs,
- const WebRtc_UWord32 maxSizeBytes)
-{
- VideoCodec dummyCodecInst;
- return StartRecordingFile(fileName, format, codecInst, dummyCodecInst,
- notificationTimeMs, maxSizeBytes);
-}
-
-
-WebRtc_Word32 MediaFileImpl::StartRecordingVideoFile(
- const WebRtc_Word8* fileName,
- const FileFormats format,
- const CodecInst& codecInst,
- const VideoCodec& videoCodecInst,
- bool videoOnly)
-{
- const WebRtc_UWord32 notificationTimeMs = 0;
- const WebRtc_UWord32 maxSizeBytes = 0;
-
- return StartRecordingFile(fileName, format, codecInst, videoCodecInst,
- notificationTimeMs, maxSizeBytes, videoOnly);
-}
-
-WebRtc_Word32 MediaFileImpl::StartRecordingFile(
- const WebRtc_Word8* fileName,
- const FileFormats format,
- const CodecInst& codecInst,
- const VideoCodec& videoCodecInst,
- const WebRtc_UWord32 notificationTimeMs,
- const WebRtc_UWord32 maxSizeBytes,
- bool videoOnly)
-{
-
- if(!ValidFileName(fileName))
- {
- return -1;
- }
- if(!ValidFileFormat(format,&codecInst))
- {
- return -1;
- }
-
- FileWrapper* outputStream = FileWrapper::Create();
- if(outputStream == NULL)
- {
- WEBRTC_TRACE(kTraceMemory, kTraceFile, _id,
- "Failed to allocate memory for output stream");
- return -1;
- }
-
- // TODO (hellner): make all formats support writing to stream.
- const bool useStream = ( format != kFileFormatAviFile);
- if( useStream)
- {
- if(outputStream->OpenFile(fileName, false) != 0)
- {
- delete outputStream;
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Could not open output file '%s' for writing!",
- fileName);
- return -1;
- }
- }
- if(maxSizeBytes)
- {
- outputStream->SetMaxFileSize(maxSizeBytes);
- }
-
- if(StartRecordingStream(*outputStream, fileName, format, codecInst,
- videoCodecInst, notificationTimeMs,
- videoOnly) == -1)
- {
- if( useStream)
- {
- outputStream->CloseFile();
- }
- delete outputStream;
- return -1;
- }
-
- CriticalSectionScoped lock(_crit);
- _openFile = true;
- strncpy(_fileName, fileName, sizeof(_fileName));
- _fileName[sizeof(_fileName) - 1] = '\0';
- return 0;
-}
-
-WebRtc_Word32 MediaFileImpl::StartRecordingAudioStream(
- OutStream& stream,
- const FileFormats format,
- const CodecInst& codecInst,
- const WebRtc_UWord32 notificationTimeMs)
-{
- VideoCodec dummyCodecInst;
- return StartRecordingStream(stream, 0, format, codecInst, dummyCodecInst,
- notificationTimeMs);
-}
-
-WebRtc_Word32 MediaFileImpl::StartRecordingStream(
- OutStream& stream,
- const WebRtc_Word8* fileName,
- const FileFormats format,
- const CodecInst& codecInst,
- const VideoCodec& videoCodecInst,
- const WebRtc_UWord32 notificationTimeMs,
- bool videoOnly)
-{
-
- // Check codec info
- if(!ValidFileFormat(format,&codecInst))
- {
- return -1;
- }
-
- CriticalSectionScoped lock(_crit);
- if(_recordingActive || _playingActive)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceFile,
- _id,
- "StartRecording called, but already recording or playing file %s!",
- _fileName);
- return -1;
- }
-
- if(_ptrFileUtilityObj != NULL)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceFile,
- _id,
- "StartRecording called, but fileUtilityObj already exists!");
- StopRecording();
- return -1;
- }
-
- _ptrFileUtilityObj = new ModuleFileUtility(_id);
- if(_ptrFileUtilityObj == NULL)
- {
- WEBRTC_TRACE(kTraceMemory, kTraceFile, _id,
- "Cannot allocate fileUtilityObj!");
- return -1;
- }
-
- CodecInst tmpAudioCodec;
- memcpy(&tmpAudioCodec, &codecInst, sizeof(CodecInst));
- switch(format)
- {
- case kFileFormatWavFile:
- {
- if(_ptrFileUtilityObj->InitWavWriting(stream, codecInst) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Failed to initialize WAV file!");
- delete _ptrFileUtilityObj;
- _ptrFileUtilityObj = NULL;
- return -1;
- }
- _fileFormat = kFileFormatWavFile;
- break;
- }
- case kFileFormatCompressedFile:
- {
- // Write compression codec name at beginning of file
- if(_ptrFileUtilityObj->InitCompressedWriting(stream, codecInst) ==
- -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Failed to initialize Compressed file!");
- delete _ptrFileUtilityObj;
- _ptrFileUtilityObj = NULL;
- return -1;
- }
- _fileFormat = kFileFormatCompressedFile;
- break;
- }
- case kFileFormatPcm8kHzFile:
- case kFileFormatPcm16kHzFile:
- {
- if(!ValidFrequency(codecInst.plfreq) ||
- _ptrFileUtilityObj->InitPCMWriting(stream, codecInst.plfreq) ==
- -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Failed to initialize 8 or 16KHz PCM file!");
- delete _ptrFileUtilityObj;
- _ptrFileUtilityObj = NULL;
- return -1;
- }
- _fileFormat = format;
- break;
- }
- case kFileFormatPreencodedFile:
- {
- if(_ptrFileUtilityObj->InitPreEncodedWriting(stream, codecInst) ==
- -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Failed to initialize Pre-Encoded file!");
- delete _ptrFileUtilityObj;
- _ptrFileUtilityObj = NULL;
- return -1;
- }
-
- _fileFormat = kFileFormatPreencodedFile;
- break;
- }
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
- case kFileFormatAviFile:
- {
- if( (_ptrFileUtilityObj->InitAviWriting(
- fileName,
- codecInst,
- videoCodecInst,videoOnly) == -1) ||
- (_ptrFileUtilityObj->codec_info(tmpAudioCodec) != 0))
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Failed to initialize AVI file!");
- delete _ptrFileUtilityObj;
- _ptrFileUtilityObj = NULL;
- return -1;
- }
- _fileFormat = kFileFormatAviFile;
- break;
- }
-#endif
- default:
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Invalid file format %d specified!", format);
- delete _ptrFileUtilityObj;
- _ptrFileUtilityObj = NULL;
- return -1;
- }
- }
- _isStereo = (tmpAudioCodec.channels == 2);
- if(_isStereo)
- {
- if(_fileFormat != kFileFormatWavFile)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
- "Stereo is only allowed for WAV files");
- StopRecording();
- return -1;
- }
- if((STR_NCASE_CMP(tmpAudioCodec.plname, "L16", 4) != 0) &&
- (STR_NCASE_CMP(tmpAudioCodec.plname, "PCMU", 5) != 0) &&
- (STR_NCASE_CMP(tmpAudioCodec.plname, "PCMA", 5) != 0))
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceFile,
- _id,
- "Stereo is only allowed for codec PCMU, PCMA and L16 ");
- StopRecording();
- return -1;
- }
- }
- memcpy(&codec_info_, &tmpAudioCodec, sizeof(CodecInst));
- _recordingActive = true;
- _ptrOutStream = &stream;
- _notificationMs = notificationTimeMs;
- _recordDurationMs = 0;
- return 0;
-}
-
-WebRtc_Word32 MediaFileImpl::StopRecording()
-{
-
- CriticalSectionScoped lock(_crit);
- if(!_recordingActive)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceFile, _id,
- "recording is not active!");
- return -1;
- }
-
- _isStereo = false;
-
- if(_ptrFileUtilityObj != NULL)
- {
- // Both AVI and WAV header has to be updated before closing the stream
- // because they contain size information.
- if((_fileFormat == kFileFormatWavFile) &&
- (_ptrOutStream != NULL))
- {
- _ptrFileUtilityObj->UpdateWavHeader(*_ptrOutStream);
- }
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
- else if( _fileFormat == kFileFormatAviFile)
- {
- _ptrFileUtilityObj->CloseAviFile( );
- }
-#endif
- delete _ptrFileUtilityObj;
- _ptrFileUtilityObj = NULL;
- }
-
- if(_ptrOutStream != NULL)
- {
- // If MediaFileImpl opened the OutStream it must be reclaimed here.
- if(_openFile)
- {
- delete _ptrOutStream;
- _openFile = false;
- }
- _ptrOutStream = NULL;
- }
-
- _recordingActive = false;
- codec_info_.pltype = 0;
- codec_info_.plname[0] = '\0';
-
- return 0;
-}
-
-bool MediaFileImpl::IsRecording()
-{
- WEBRTC_TRACE(kTraceStream, kTraceFile, _id, "MediaFileImpl::IsRecording()");
- CriticalSectionScoped lock(_crit);
- return _recordingActive;
-}
-
-WebRtc_Word32 MediaFileImpl::RecordDurationMs(WebRtc_UWord32& durationMs)
-{
-
- CriticalSectionScoped lock(_crit);
- if(!_recordingActive)
- {
- durationMs = 0;
- return -1;
- }
- durationMs = _recordDurationMs;
- return 0;
-}
-
-bool MediaFileImpl::IsStereo()
-{
- WEBRTC_TRACE(kTraceStream, kTraceFile, _id, "MediaFileImpl::IsStereo()");
- CriticalSectionScoped lock(_crit);
- return _isStereo;
-}
-
-WebRtc_Word32 MediaFileImpl::SetModuleFileCallback(FileCallback* callback)
-{
-
- CriticalSectionScoped lock(_callbackCrit);
-
- _ptrCallback = callback;
- return 0;
-}
-
-WebRtc_Word32 MediaFileImpl::FileDurationMs(const WebRtc_Word8* fileName,
- WebRtc_UWord32& durationMs,
- const FileFormats format,
- const WebRtc_UWord32 freqInHz)
-{
-
- if(!ValidFileName(fileName))
- {
- return -1;
- }
- if(!ValidFrequency(freqInHz))
- {
- return -1;
- }
-
- ModuleFileUtility* utilityObj = new ModuleFileUtility(_id);
- if(utilityObj == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "failed to allocate utility object!");
- return -1;
- }
-
- const WebRtc_Word32 duration = utilityObj->FileDurationMs(fileName, format,
- freqInHz);
- delete utilityObj;
- if(duration == -1)
- {
- durationMs = 0;
- return -1;
- }
-
- durationMs = duration;
- return 0;
-}
-
-WebRtc_Word32 MediaFileImpl::PlayoutPositionMs(WebRtc_UWord32& positionMs) const
-{
- CriticalSectionScoped lock(_crit);
- if(!_playingActive)
- {
- positionMs = 0;
- return -1;
- }
- positionMs = _playoutPositionMs;
- return 0;
-}
-
-WebRtc_Word32 MediaFileImpl::codec_info(CodecInst& codecInst) const
-{
- CriticalSectionScoped lock(_crit);
- if(!_playingActive && !_recordingActive)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Neither playout nor recording has been initialized!");
- return -1;
- }
- if (codec_info_.pltype == 0 && codec_info_.plname[0] == '\0')
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "The CodecInst for %s is unknown!",
- _playingActive ? "Playback" : "Recording");
- return -1;
- }
- memcpy(&codecInst,&codec_info_,sizeof(CodecInst));
- return 0;
-}
-
-WebRtc_Word32 MediaFileImpl::VideoCodecInst(VideoCodec& codecInst) const
-{
- CriticalSectionScoped lock(_crit);
- if(!_playingActive && !_recordingActive)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Neither playout nor recording has been initialized!");
- return -1;
- }
- if( _ptrFileUtilityObj == NULL)
- {
- return -1;
- }
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
- VideoCodec videoCodec;
- if( _ptrFileUtilityObj->VideoCodecInst( videoCodec) != 0)
- {
- return -1;
- }
- memcpy(&codecInst,&videoCodec,sizeof(VideoCodec));
- return 0;
-#else
- return -1;
-#endif
-}
-
-bool MediaFileImpl::ValidFileFormat(const FileFormats format,
- const CodecInst* codecInst)
-{
- if(codecInst == NULL)
- {
- if(format == kFileFormatPreencodedFile ||
- format == kFileFormatPcm8kHzFile ||
- format == kFileFormatPcm16kHzFile ||
- format == kFileFormatPcm32kHzFile)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, -1,
- "Codec info required for file format specified!");
- return false;
- }
- }
- return true;
-}
-
-bool MediaFileImpl::ValidFileName(const WebRtc_Word8* fileName)
-{
- if((fileName == NULL) ||(fileName[0] == '\0'))
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, -1, "FileName not specified!");
- return false;
- }
- return true;
-}
-
-
-bool MediaFileImpl::ValidFilePositions(const WebRtc_UWord32 startPointMs,
- const WebRtc_UWord32 stopPointMs)
-{
- if(startPointMs == 0 && stopPointMs == 0) // Default values
- {
- return true;
- }
- if(stopPointMs &&(startPointMs >= stopPointMs))
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, -1,
- "startPointMs must be less than stopPointMs!");
- return false;
- }
- if(stopPointMs &&((stopPointMs - startPointMs) < 20))
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, -1,
- "minimum play duration for files is 20 ms!");
- return false;
- }
- return true;
-}
-
-bool MediaFileImpl::ValidFrequency(const WebRtc_UWord32 frequency)
-{
- if((frequency == 8000) || (frequency == 16000)|| (frequency == 32000))
- {
- return true;
- }
- WEBRTC_TRACE(kTraceError, kTraceFile, -1,
- "Frequency should be 8000, 16000 or 32000 (Hz)");
- return false;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/media_file/source/media_file_impl.h b/trunk/src/modules/media_file/source/media_file_impl.h
deleted file mode 100644
index 06e20d0..0000000
--- a/trunk/src/modules/media_file/source/media_file_impl.h
+++ /dev/null
@@ -1,246 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
-#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
-
-#include "common_types.h"
-#include "media_file.h"
-#include "media_file_defines.h"
-#include "media_file_utility.h"
-#include "module_common_types.h"
-
-namespace webrtc {
-class MediaFileImpl : public MediaFile
-{
-
-public:
- MediaFileImpl(const WebRtc_Word32 id);
- ~MediaFileImpl();
-
- WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
- WebRtc_Word32 Process();
- WebRtc_Word32 TimeUntilNextProcess();
-
- // MediaFile functions
- WebRtc_Word32 PlayoutAudioData(WebRtc_Word8* audioBuffer,
- WebRtc_UWord32& dataLengthInBytes);
- WebRtc_Word32 PlayoutAVIVideoData(WebRtc_Word8* videoBuffer,
- WebRtc_UWord32& dataLengthInBytes);
- WebRtc_Word32 PlayoutStereoData(WebRtc_Word8* audioBufferLeft,
- WebRtc_Word8* audioBufferRight,
- WebRtc_UWord32& dataLengthInBytes);
- virtual WebRtc_Word32 StartPlayingAudioFile(
- const WebRtc_Word8* fileName,
- const WebRtc_UWord32 notificationTimeMs = 0,
- const bool loop = false,
- const FileFormats format = kFileFormatPcm16kHzFile,
- const CodecInst* codecInst = NULL,
- const WebRtc_UWord32 startPointMs = 0,
- const WebRtc_UWord32 stopPointMs = 0);
- WebRtc_Word32 StartPlayingVideoFile(const WebRtc_Word8* fileName,
- const bool loop,
- bool videoOnly,
- const FileFormats format);
- WebRtc_Word32 StartPlayingAudioStream(
- InStream& stream,
- const WebRtc_UWord32 notificationTimeMs = 0,
- const FileFormats format = kFileFormatPcm16kHzFile,
- const CodecInst* codecInst = NULL,
- const WebRtc_UWord32 startPointMs = 0,
- const WebRtc_UWord32 stopPointMs = 0);
- WebRtc_Word32 StopPlaying();
- bool IsPlaying();
- WebRtc_Word32 PlayoutPositionMs(WebRtc_UWord32& positionMs) const;
- WebRtc_Word32 IncomingAudioData(const WebRtc_Word8* audioBuffer,
- const WebRtc_UWord32 bufferLength);
- WebRtc_Word32 IncomingAVIVideoData(const WebRtc_Word8* audioBuffer,
- const WebRtc_UWord32 bufferLength);
- WebRtc_Word32 StartRecordingAudioFile(
- const WebRtc_Word8* fileName,
- const FileFormats format,
- const CodecInst& codecInst,
- const WebRtc_UWord32 notificationTimeMs = 0,
- const WebRtc_UWord32 maxSizeBytes = 0);
- WebRtc_Word32 StartRecordingVideoFile(
- const WebRtc_Word8* fileName,
- const FileFormats format,
- const CodecInst& codecInst,
- const VideoCodec& videoCodecInst,
- bool videoOnly = false);
- WebRtc_Word32 StartRecordingAudioStream(
- OutStream& stream,
- const FileFormats format,
- const CodecInst& codecInst,
- const WebRtc_UWord32 notificationTimeMs = 0);
- WebRtc_Word32 StopRecording();
- bool IsRecording();
- WebRtc_Word32 RecordDurationMs(WebRtc_UWord32& durationMs);
- bool IsStereo();
- WebRtc_Word32 SetModuleFileCallback(FileCallback* callback);
- WebRtc_Word32 FileDurationMs(
- const WebRtc_Word8* fileName,
- WebRtc_UWord32& durationMs,
- const FileFormats format,
- const WebRtc_UWord32 freqInHz = 16000);
- WebRtc_Word32 codec_info(CodecInst& codecInst) const;
- WebRtc_Word32 VideoCodecInst(VideoCodec& codecInst) const;
-
-private:
- // Returns true if the combination of format and codecInst is valid.
- static bool ValidFileFormat(const FileFormats format,
- const CodecInst* codecInst);
-
-
- // Returns true if the filename is valid
- static bool ValidFileName(const WebRtc_Word8* fileName);
-
- // Returns true if the combination of startPointMs and stopPointMs is valid.
- static bool ValidFilePositions(const WebRtc_UWord32 startPointMs,
- const WebRtc_UWord32 stopPointMs);
-
- // Open the file specified by fileName for reading (relative path is
- // allowed). FileCallback::PlayNotification(..) will be called after
- // notificationTimeMs of the file has been played if notificationTimeMs is
- // greater than zero. If loop is true the file will be played until
- // StopPlaying() is called. When end of file is reached the file is read
- // from the start. format specifies the type of file fileName refers to.
- // codecInst specifies the encoding of the audio data. Note that
- // file formats that contain this information (like WAV files) don't need to
- // provide a non-NULL codecInst. Only video will be read if videoOnly is
- // true. startPointMs and stopPointMs, unless zero,
- // specify what part of the file should be read. From startPointMs ms to
- // stopPointMs ms.
- WebRtc_Word32 StartPlayingFile(
- const WebRtc_Word8* fileName,
- const WebRtc_UWord32 notificationTimeMs = 0,
- const bool loop = false,
- bool videoOnly = false,
- const FileFormats format = kFileFormatPcm16kHzFile,
- const CodecInst* codecInst = NULL,
- const WebRtc_UWord32 startPointMs = 0,
- const WebRtc_UWord32 stopPointMs = 0);
-
- // Opens the file specified by fileName for reading (relative path is
- // allowed) if format is kFileFormatAviFile otherwise use stream for
- // reading. FileCallback::PlayNotification(..) will be called after
- // notificationTimeMs of the file has been played if notificationTimeMs is
- // greater than zero. If loop is true the file will be played until
- // StopPlaying() is called. When end of file is reached the file is read
- // from the start. format specifies the type of file fileName refers to.
- // codecInst specifies the encoding of the audio data. Note that
- // file formats that contain this information (like WAV files) don't need to
- // provide a non-NULL codecInst. Only video will be read if videoOnly is
- // true. startPointMs and stopPointMs, unless zero,
- // specify what part of the file should be read. From startPointMs ms to
- // stopPointMs ms.
- // TODO (hellner): there is no reason why fileName should be needed here.
- WebRtc_Word32 StartPlayingStream(
- InStream& stream,
- const WebRtc_Word8* fileName,
- bool loop,
- const WebRtc_UWord32 notificationTimeMs = 0,
- const FileFormats format = kFileFormatPcm16kHzFile,
- const CodecInst* codecInst = NULL,
- const WebRtc_UWord32 startPointMs = 0,
- const WebRtc_UWord32 stopPointMs = 0,
- bool videoOnly = true);
-
- // Writes one frame into dataBuffer. dataLengthInBytes is both an input and
- // output parameter. As input parameter it indicates the size of
- // audioBuffer. As output parameter it indicates the number of bytes
- // written to audioBuffer. If video is true the data written is a video
- // frame otherwise it is an audio frame.
- WebRtc_Word32 PlayoutData(WebRtc_Word8* dataBuffer,
- WebRtc_UWord32& dataLengthInBytes, bool video);
-
- // Write one frame, i.e. the bufferLength first bytes of audioBuffer,
- // to file. The frame is an audio frame if video is true otherwise it is an
- // audio frame.
- WebRtc_Word32 IncomingAudioVideoData(const WebRtc_Word8* buffer,
- const WebRtc_UWord32 bufferLength,
- const bool video);
-
- // Open/creates file specified by fileName for writing (relative path is
- // allowed) if format is kFileFormatAviFile otherwise use stream for
- // writing. FileCallback::RecordNotification(..) will be called after
- // notificationTimeMs of audio data has been recorded if
- // notificationTimeMs is greater than zero.
- // format specifies the type of file that should be created/opened.
- // codecInst specifies the encoding of the audio data. videoCodecInst
- // specifies the encoding of the video data. maxSizeBytes specifies the
- // number of bytes allowed to be written to file if it is greater than zero.
- // If format is kFileFormatAviFile and videoOnly is true the AVI file will
- // only contain video frames.
- // Note: codecInst.channels should be set to 2 for stereo (and 1 for
- // mono). Stereo is only supported for WAV files.
- WebRtc_Word32 StartRecordingFile(
- const WebRtc_Word8* fileName,
- const FileFormats format,
- const CodecInst& codecInst,
- const VideoCodec& videoCodecInst,
- const WebRtc_UWord32 notificationTimeMs = 0,
- const WebRtc_UWord32 maxSizeBytes = 0,
- bool videoOnly = false);
-
- // Open/creates file specified by fileName for writing (relative path is
- // allowed). FileCallback::RecordNotification(..) will be called after
- // notificationTimeMs of audio data has been recorded if
- // notificationTimeMs is greater than zero.
- // format specifies the type of file that should be created/opened.
- // codecInst specifies the encoding of the audio data. videoCodecInst
- // specifies the encoding of the video data. maxSizeBytes specifies the
- // number of bytes allowed to be written to file if it is greater than zero.
- // If format is kFileFormatAviFile and videoOnly is true the AVI file will
- // only contain video frames.
- // Note: codecInst.channels should be set to 2 for stereo (and 1 for
- // mono). Stereo is only supported for WAV files.
- // TODO (hellner): there is no reason why fileName should be needed here.
- WebRtc_Word32 StartRecordingStream(
- OutStream& stream,
- const WebRtc_Word8* fileName,
- const FileFormats format,
- const CodecInst& codecInst,
- const VideoCodec& videoCodecInst,
- const WebRtc_UWord32 notificationTimeMs = 0,
- const bool videoOnly = false);
-
- // Returns true if frequencyInHz is a supported frequency.
- static bool ValidFrequency(const WebRtc_UWord32 frequencyInHz);
-
- void HandlePlayCallbacks(WebRtc_Word32 bytesRead);
-
- WebRtc_Word32 _id;
- CriticalSectionWrapper* _crit;
- CriticalSectionWrapper* _callbackCrit;
-
- ModuleFileUtility* _ptrFileUtilityObj;
- CodecInst codec_info_;
-
- InStream* _ptrInStream;
- OutStream* _ptrOutStream;
-
- FileFormats _fileFormat;
- WebRtc_UWord32 _recordDurationMs;
- WebRtc_UWord32 _playoutPositionMs;
- WebRtc_UWord32 _notificationMs;
-
- bool _playingActive;
- bool _recordingActive;
- bool _isStereo;
- bool _openFile;
-
- WebRtc_Word8 _fileName[512];
-
- FileCallback* _ptrCallback;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_IMPL_H_
diff --git a/trunk/src/modules/media_file/source/media_file_unittest.cc b/trunk/src/modules/media_file/source/media_file_unittest.cc
deleted file mode 100644
index 86f37f3..0000000
--- a/trunk/src/modules/media_file/source/media_file_unittest.cc
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Empty test just to get code coverage metrics for this dir.
- */
-#include "media_file.h"
-#include "gtest/gtest.h"
-
-TEST(MediaFileTest, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/modules/media_file/source/media_file_utility.cc b/trunk/src/modules/media_file/source/media_file_utility.cc
deleted file mode 100644
index eeeb209..0000000
--- a/trunk/src/modules/media_file/source/media_file_utility.cc
+++ /dev/null
@@ -1,2563 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <assert.h>
-#include <sys/stat.h>
-#include <sys/types.h>
-
-#include "common_types.h"
-#include "engine_configurations.h"
-#include "file_wrapper.h"
-#include "media_file_utility.h"
-#include "module_common_types.h"
-#include "trace.h"
-
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
- #include "avi_file.h"
-#endif
-
-#if (defined(WIN32) || defined(WINCE))
- #define STR_CASE_CMP _stricmp
- #define STR_NCASE_CMP _strnicmp
-#else
- #define STR_CASE_CMP strcasecmp
- #define STR_NCASE_CMP strncasecmp
-#endif
-
-namespace {
-enum WaveFormats
-{
- kWaveFormatPcm = 0x0001,
- kWaveFormatALaw = 0x0006,
- kWaveFormatMuLaw = 0x0007
-};
-
-// First 16 bytes the WAVE header. ckID should be "RIFF", wave_ckID should be
-// "WAVE" and ckSize is the chunk size (4 + n)
-struct WAVE_RIFF_header
-{
- WebRtc_Word8 ckID[4];
- WebRtc_Word32 ckSize;
- WebRtc_Word8 wave_ckID[4];
-};
-
-// First 8 byte of the format chunk. fmt_ckID should be "fmt ". fmt_ckSize is
-// the chunk size (16, 18 or 40 byte)
-struct WAVE_CHUNK_header
-{
- WebRtc_Word8 fmt_ckID[4];
- WebRtc_Word32 fmt_ckSize;
-};
-} // unnamed namespace
-
-namespace webrtc {
-ModuleFileUtility::ModuleFileUtility(const WebRtc_Word32 id)
- : _wavFormatObj(),
- _dataSize(0),
- _readSizeBytes(0),
- _id(id),
- _stopPointInMs(0),
- _startPointInMs(0),
- _playoutPositionMs(0),
- _bytesWritten(0),
- codec_info_(),
- _codecId(kCodecNoCodec),
- _bytesPerSample(0),
- _readPos(0),
- _reading(false),
- _writing(false),
- _tempData()
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
- ,
- _aviAudioInFile(0),
- _aviVideoInFile(0),
- _aviOutFile(0)
-#endif
-{
- WEBRTC_TRACE(kTraceMemory, kTraceFile, _id,
- "ModuleFileUtility::ModuleFileUtility()");
- memset(&codec_info_,0,sizeof(CodecInst));
- codec_info_.pltype = -1;
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
- memset(&_videoCodec,0,sizeof(_videoCodec));
-#endif
-}
-
-ModuleFileUtility::~ModuleFileUtility()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceFile, _id,
- "ModuleFileUtility::~ModuleFileUtility()");
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
- delete _aviAudioInFile;
- delete _aviVideoInFile;
-#endif
-}
-
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
-WebRtc_Word32 ModuleFileUtility::InitAviWriting(
- const WebRtc_Word8* filename,
- const CodecInst& audioCodecInst,
- const VideoCodec& videoCodecInst,
- const bool videoOnly /*= false*/)
-{
- _writing = false;
-
- delete _aviOutFile;
- _aviOutFile = new AviFile( );
-
- AVISTREAMHEADER videoStreamHeader;
- videoStreamHeader.fccType = AviFile::MakeFourCc('v', 'i', 'd', 's');
-
-#ifdef VIDEOCODEC_I420
- if (strncmp(videoCodecInst.plName, "I420", 7) == 0)
- {
- videoStreamHeader.fccHandler = AviFile::MakeFourCc('I','4','2','0');
- }
-#endif
-#ifdef VIDEOCODEC_VP8
- if (strncmp(videoCodecInst.plName, "VP8", 7) == 0)
- {
- videoStreamHeader.fccHandler = AviFile::MakeFourCc('V','P','8','0');
- }
-#endif
- if (videoStreamHeader.fccHandler == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "InitAviWriting() Codec not supported");
-
- return -1;
- }
- videoStreamHeader.dwScale = 1;
- videoStreamHeader.dwRate = videoCodecInst.maxFramerate;
- videoStreamHeader.dwSuggestedBufferSize = videoCodecInst.height *
- (videoCodecInst.width >> 1) * 3;
- videoStreamHeader.dwQuality = (WebRtc_UWord32)-1;
- videoStreamHeader.dwSampleSize = 0;
- videoStreamHeader.rcFrame.top = 0;
- videoStreamHeader.rcFrame.bottom = videoCodecInst.height;
- videoStreamHeader.rcFrame.left = 0;
- videoStreamHeader.rcFrame.right = videoCodecInst.width;
-
- BITMAPINFOHEADER bitMapInfoHeader;
- bitMapInfoHeader.biSize = sizeof(BITMAPINFOHEADER);
- bitMapInfoHeader.biHeight = videoCodecInst.height;
- bitMapInfoHeader.biWidth = videoCodecInst.width;
- bitMapInfoHeader.biPlanes = 1;
- bitMapInfoHeader.biBitCount = 12;
- bitMapInfoHeader.biClrImportant = 0;
- bitMapInfoHeader.biClrUsed = 0;
- bitMapInfoHeader.biCompression = videoStreamHeader.fccHandler;
- bitMapInfoHeader.biSizeImage = bitMapInfoHeader.biWidth *
- bitMapInfoHeader.biHeight * bitMapInfoHeader.biBitCount / 8;
-
- if (_aviOutFile->CreateVideoStream(
- videoStreamHeader,
- bitMapInfoHeader,
- NULL,
- 0) != 0)
- {
- return -1;
- }
-
- if(!videoOnly)
- {
- AVISTREAMHEADER audioStreamHeader;
- audioStreamHeader.fccType = AviFile::MakeFourCc('a', 'u', 'd', 's');
- // fccHandler is the FOURCC of the codec for decoding the stream.
- // It's an optional parameter that is not used by audio streams.
- audioStreamHeader.fccHandler = 0;
- audioStreamHeader.dwScale = 1;
-
- WAVEFORMATEX waveFormatHeader;
- waveFormatHeader.cbSize = 0;
- waveFormatHeader.nChannels = 1;
-
- if (strncmp(audioCodecInst.plname, "PCMU", 4) == 0)
- {
- audioStreamHeader.dwSampleSize = 1;
- audioStreamHeader.dwRate = 8000;
- audioStreamHeader.dwQuality = (WebRtc_UWord32)-1;
- audioStreamHeader.dwSuggestedBufferSize = 80;
-
- waveFormatHeader.nAvgBytesPerSec = 8000;
- waveFormatHeader.nSamplesPerSec = 8000;
- waveFormatHeader.wBitsPerSample = 8;
- waveFormatHeader.nBlockAlign = 1;
- waveFormatHeader.wFormatTag = kWaveFormatMuLaw;
-
- } else if (strncmp(audioCodecInst.plname, "PCMA", 4) == 0)
- {
- audioStreamHeader.dwSampleSize = 1;
- audioStreamHeader.dwRate = 8000;
- audioStreamHeader.dwQuality = (WebRtc_UWord32)-1;
- audioStreamHeader.dwSuggestedBufferSize = 80;
-
- waveFormatHeader.nAvgBytesPerSec = 8000;
- waveFormatHeader.nSamplesPerSec = 8000;
- waveFormatHeader.wBitsPerSample = 8;
- waveFormatHeader.nBlockAlign = 1;
- waveFormatHeader.wFormatTag = kWaveFormatALaw;
-
- } else if (strncmp(audioCodecInst.plname, "L16", 3) == 0)
- {
- audioStreamHeader.dwSampleSize = 2;
- audioStreamHeader.dwRate = audioCodecInst.plfreq;
- audioStreamHeader.dwQuality = (WebRtc_UWord32)-1;
- audioStreamHeader.dwSuggestedBufferSize =
- (audioCodecInst.plfreq/100) * 2;
-
- waveFormatHeader.nAvgBytesPerSec = audioCodecInst.plfreq * 2;
- waveFormatHeader.nSamplesPerSec = audioCodecInst.plfreq;
- waveFormatHeader.wBitsPerSample = 16;
- waveFormatHeader.nBlockAlign = 2;
- waveFormatHeader.wFormatTag = kWaveFormatPcm;
- } else
- {
- return -1;
- }
-
- if(_aviOutFile->CreateAudioStream(
- audioStreamHeader,
- waveFormatHeader) != 0)
- {
- return -1;
- }
-
-
- if( InitWavCodec(waveFormatHeader.nSamplesPerSec,
- waveFormatHeader.nChannels,
- waveFormatHeader.wBitsPerSample,
- waveFormatHeader.wFormatTag) != 0)
- {
- return -1;
- }
- }
- _aviOutFile->Create(filename);
- _writing = true;
- return 0;
-}
-
-WebRtc_Word32 ModuleFileUtility::WriteAviAudioData(
- const WebRtc_Word8* buffer,
- WebRtc_UWord32 bufferLengthInBytes)
-{
- if( _aviOutFile != 0)
- {
- return _aviOutFile->WriteAudio(
- reinterpret_cast<const WebRtc_UWord8*>(buffer),
- bufferLengthInBytes);
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id, "AVI file not initialized");
- return -1;
- }
-}
-
-WebRtc_Word32 ModuleFileUtility::WriteAviVideoData(
- const WebRtc_Word8* buffer,
- WebRtc_UWord32 bufferLengthInBytes)
-{
- if( _aviOutFile != 0)
- {
- return _aviOutFile->WriteVideo(
- reinterpret_cast<const WebRtc_UWord8*>(buffer),
- bufferLengthInBytes);
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id, "AVI file not initialized");
- return -1;
- }
-}
-
-
-WebRtc_Word32 ModuleFileUtility::CloseAviFile( )
-{
- if( _reading && _aviAudioInFile)
- {
- delete _aviAudioInFile;
- _aviAudioInFile = 0;
- }
-
- if( _reading && _aviVideoInFile)
- {
- delete _aviVideoInFile;
- _aviVideoInFile = 0;
- }
-
- if( _writing && _aviOutFile)
- {
- delete _aviOutFile;
- _aviOutFile = 0;
- }
- return 0;
-}
-
-
-WebRtc_Word32 ModuleFileUtility::InitAviReading(const WebRtc_Word8* filename,
- bool videoOnly, bool loop)
-{
- _reading = false;
- delete _aviVideoInFile;
- _aviVideoInFile = new AviFile( );
-
- if ((_aviVideoInFile != 0) && _aviVideoInFile->Open(AviFile::AVI_VIDEO,
- filename, loop) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Unable to open AVI file (video)");
- return -1;
- }
-
-
- AVISTREAMHEADER videoInStreamHeader;
- BITMAPINFOHEADER bitmapInfo;
- char codecConfigParameters[AviFile::CODEC_CONFIG_LENGTH] = {};
- WebRtc_Word32 configLength = 0;
- if( _aviVideoInFile->GetVideoStreamInfo(videoInStreamHeader, bitmapInfo,
- codecConfigParameters,
- configLength) != 0)
- {
- return -1;
- }
- _videoCodec.width = static_cast<WebRtc_UWord16>(
- videoInStreamHeader.rcFrame.right);
- _videoCodec.height = static_cast<WebRtc_UWord16>(
- videoInStreamHeader.rcFrame.bottom);
- _videoCodec.maxFramerate = static_cast<WebRtc_UWord8>(
- videoInStreamHeader.dwRate);
-
- const size_t plnameLen = sizeof(_videoCodec.plName) / sizeof(char);
- if (bitmapInfo.biCompression == AviFile::MakeFourCc('I','4','2','0'))
- {
- strncpy(_videoCodec.plName, "I420", plnameLen);
- _videoCodec.codecType = kVideoCodecI420;
- }
- else if (bitmapInfo.biCompression ==
- AviFile::MakeFourCc('V', 'P', '8', '0'))
- {
- strncpy(_videoCodec.plName, "VP8", plnameLen);
- _videoCodec.codecType = kVideoCodecVP8;
- }
- else
- {
- return -1;
- }
-
- if(!videoOnly)
- {
- delete _aviAudioInFile;
- _aviAudioInFile = new AviFile();
-
- if ( (_aviAudioInFile != 0) &&
- _aviAudioInFile->Open(AviFile::AVI_AUDIO, filename, loop) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Unable to open AVI file (audio)");
- return -1;
- }
-
- WAVEFORMATEX waveHeader;
- if(_aviAudioInFile->GetAudioStreamInfo(waveHeader) != 0)
- {
- return -1;
- }
- if(InitWavCodec(waveHeader.nSamplesPerSec, waveHeader.nChannels,
- waveHeader.wBitsPerSample, waveHeader.wFormatTag) != 0)
- {
- return -1;
- }
- }
- _reading = true;
- return 0;
-}
-
-WebRtc_Word32 ModuleFileUtility::ReadAviAudioData(
- WebRtc_Word8* outBuffer,
- const WebRtc_UWord32 bufferLengthInBytes)
-{
- if(_aviAudioInFile == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "AVI file not opened.");
- return -1;
- }
-
- WebRtc_Word32 length = bufferLengthInBytes;
- if(_aviAudioInFile->ReadAudio(
- reinterpret_cast<WebRtc_UWord8*>(outBuffer),
- length) != 0)
- {
- return -1;
- }
- else
- {
- return length;
- }
-}
-
-WebRtc_Word32 ModuleFileUtility::ReadAviVideoData(
- WebRtc_Word8* outBuffer,
- const WebRtc_UWord32 bufferLengthInBytes)
-{
- if(_aviVideoInFile == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "AVI file not opened.");
- return -1;
- }
-
- WebRtc_Word32 length = bufferLengthInBytes;
- if( _aviVideoInFile->ReadVideo(
- reinterpret_cast<WebRtc_UWord8*>(outBuffer),
- length) != 0)
- {
- return -1;
- } else {
- return length;
- }
-}
-
-WebRtc_Word32 ModuleFileUtility::VideoCodecInst(VideoCodec& codecInst)
-{
- WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
- "ModuleFileUtility::CodecInst(codecInst= 0x%x)", &codecInst);
-
- if(!_reading)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "CodecInst: not currently reading audio file!");
- return -1;
- }
- memcpy(&codecInst,&_videoCodec,sizeof(VideoCodec));
- return 0;
-}
-#endif
-
-WebRtc_Word32 ModuleFileUtility::ReadWavHeader(InStream& wav)
-{
- WAVE_RIFF_header RIFFheaderObj;
- WAVE_CHUNK_header CHUNKheaderObj;
- // TODO (hellner): tmpStr and tmpStr2 seems unnecessary here.
- WebRtc_Word8 tmpStr[6] = "FOUR";
- WebRtc_UWord8 tmpStr2[4];
- WebRtc_Word32 i, len;
- bool dataFound = false;
- bool fmtFound = false;
- WebRtc_Word8 dummyRead;
-
-
- _dataSize = 0;
- len = wav.Read(&RIFFheaderObj, sizeof(WAVE_RIFF_header));
- if(len != sizeof(WAVE_RIFF_header))
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Not a wave file (too short)");
- return -1;
- }
-
- for (i = 0; i < 4; i++)
- {
- tmpStr[i] = RIFFheaderObj.ckID[i];
- }
- if(strcmp(tmpStr, "RIFF") != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Not a wave file (does not have RIFF)");
- return -1;
- }
- for (i = 0; i < 4; i++)
- {
- tmpStr[i] = RIFFheaderObj.wave_ckID[i];
- }
- if(strcmp(tmpStr, "WAVE") != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Not a wave file (does not have WAVE)");
- return -1;
- }
-
- len = wav.Read(&CHUNKheaderObj, sizeof(WAVE_CHUNK_header));
-
- // WAVE files are stored in little endian byte order. Make sure that the
- // data can be read on big endian as well.
- // TODO (hellner): little endian to system byte order should be done in
- // in a subroutine.
- memcpy(tmpStr2, &CHUNKheaderObj.fmt_ckSize, 4);
- CHUNKheaderObj.fmt_ckSize =
- (WebRtc_Word32) ((WebRtc_UWord32) tmpStr2[0] +
- (((WebRtc_UWord32)tmpStr2[1])<<8) +
- (((WebRtc_UWord32)tmpStr2[2])<<16) +
- (((WebRtc_UWord32)tmpStr2[3])<<24));
-
- memcpy(tmpStr, CHUNKheaderObj.fmt_ckID, 4);
-
- while ((len == sizeof(WAVE_CHUNK_header)) && (!fmtFound || !dataFound))
- {
- if(strcmp(tmpStr, "fmt ") == 0)
- {
- len = wav.Read(&_wavFormatObj, sizeof(WAVE_FMTINFO_header));
-
- memcpy(tmpStr2, &_wavFormatObj.formatTag, 2);
- _wavFormatObj.formatTag =
- (WaveFormats) ((WebRtc_UWord32)tmpStr2[0] +
- (((WebRtc_UWord32)tmpStr2[1])<<8));
- memcpy(tmpStr2, &_wavFormatObj.nChannels, 2);
- _wavFormatObj.nChannels =
- (WebRtc_Word16) ((WebRtc_UWord32)tmpStr2[0] +
- (((WebRtc_UWord32)tmpStr2[1])<<8));
- memcpy(tmpStr2, &_wavFormatObj.nSamplesPerSec, 4);
- _wavFormatObj.nSamplesPerSec =
- (WebRtc_Word32) ((WebRtc_UWord32)tmpStr2[0] +
- (((WebRtc_UWord32)tmpStr2[1])<<8) +
- (((WebRtc_UWord32)tmpStr2[2])<<16) +
- (((WebRtc_UWord32)tmpStr2[3])<<24));
- memcpy(tmpStr2, &_wavFormatObj.nAvgBytesPerSec, 4);
- _wavFormatObj.nAvgBytesPerSec =
- (WebRtc_Word32) ((WebRtc_UWord32)tmpStr2[0] +
- (((WebRtc_UWord32)tmpStr2[1])<<8) +
- (((WebRtc_UWord32)tmpStr2[2])<<16) +
- (((WebRtc_UWord32)tmpStr2[3])<<24));
- memcpy(tmpStr2, &_wavFormatObj.nBlockAlign, 2);
- _wavFormatObj.nBlockAlign =
- (WebRtc_Word16) ((WebRtc_UWord32)tmpStr2[0] +
- (((WebRtc_UWord32)tmpStr2[1])<<8));
- memcpy(tmpStr2, &_wavFormatObj.nBitsPerSample, 2);
- _wavFormatObj.nBitsPerSample =
- (WebRtc_Word16) ((WebRtc_UWord32)tmpStr2[0] +
- (((WebRtc_UWord32)tmpStr2[1])<<8));
-
- for (i = 0;
- i < (CHUNKheaderObj.fmt_ckSize -
- (WebRtc_Word32)sizeof(WAVE_FMTINFO_header));
- i++)
- {
- len = wav.Read(&dummyRead, 1);
- if(len != 1)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "File corrupted, reached EOF (reading fmt)");
- return -1;
- }
- }
- fmtFound = true;
- }
- else if(strcmp(tmpStr, "data") == 0)
- {
- _dataSize = CHUNKheaderObj.fmt_ckSize;
- dataFound = true;
- break;
- }
- else
- {
- for (i = 0; i < (CHUNKheaderObj.fmt_ckSize); i++)
- {
- len = wav.Read(&dummyRead, 1);
- if(len != 1)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "File corrupted, reached EOF (reading other)");
- return -1;
- }
- }
- }
-
- len = wav.Read(&CHUNKheaderObj, sizeof(WAVE_CHUNK_header));
-
- memcpy(tmpStr2, &CHUNKheaderObj.fmt_ckSize, 4);
- CHUNKheaderObj.fmt_ckSize =
- (WebRtc_Word32) ((WebRtc_UWord32)tmpStr2[0] +
- (((WebRtc_UWord32)tmpStr2[1])<<8) +
- (((WebRtc_UWord32)tmpStr2[2])<<16) +
- (((WebRtc_UWord32)tmpStr2[3])<<24));
-
- memcpy(tmpStr, CHUNKheaderObj.fmt_ckID, 4);
- }
-
- // Either a proper format chunk has been read or a data chunk was come
- // across.
- if( (_wavFormatObj.formatTag != kWaveFormatPcm) &&
- (_wavFormatObj.formatTag != kWaveFormatALaw) &&
- (_wavFormatObj.formatTag != kWaveFormatMuLaw))
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Coding formatTag value=%d not supported!",
- _wavFormatObj.formatTag);
- return -1;
- }
- if((_wavFormatObj.nChannels < 1) ||
- (_wavFormatObj.nChannels > 2))
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "nChannels value=%d not supported!",
- _wavFormatObj.nChannels);
- return -1;
- }
-
- if((_wavFormatObj.nBitsPerSample != 8) &&
- (_wavFormatObj.nBitsPerSample != 16))
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "nBitsPerSample value=%d not supported!",
- _wavFormatObj.nBitsPerSample);
- return -1;
- }
-
- // Calculate the number of bytes that 10 ms of audio data correspond to.
- if(_wavFormatObj.formatTag == kWaveFormatPcm)
- {
- // TODO (hellner): integer division for 22050 and 11025 would yield
- // the same result as the else statement. Remove those
- // special cases?
- if(_wavFormatObj.nSamplesPerSec == 44100)
- {
- _readSizeBytes = 440 * _wavFormatObj.nChannels *
- (_wavFormatObj.nBitsPerSample / 8);
- } else if(_wavFormatObj.nSamplesPerSec == 22050) {
- _readSizeBytes = 220 * _wavFormatObj.nChannels *
- (_wavFormatObj.nBitsPerSample / 8);
- } else if(_wavFormatObj.nSamplesPerSec == 11025) {
- _readSizeBytes = 110 * _wavFormatObj.nChannels *
- (_wavFormatObj.nBitsPerSample / 8);
- } else {
- _readSizeBytes = (_wavFormatObj.nSamplesPerSec/100) *
- _wavFormatObj.nChannels * (_wavFormatObj.nBitsPerSample / 8);
- }
-
- } else {
- _readSizeBytes = (_wavFormatObj.nSamplesPerSec/100) *
- _wavFormatObj.nChannels * (_wavFormatObj.nBitsPerSample / 8);
- }
- return 0;
-}
-
-WebRtc_Word32 ModuleFileUtility::InitWavCodec(WebRtc_UWord32 samplesPerSec,
- WebRtc_UWord32 channels,
- WebRtc_UWord32 bitsPerSample,
- WebRtc_UWord32 formatTag)
-{
- codec_info_.pltype = -1;
- codec_info_.plfreq = samplesPerSec;
- codec_info_.channels = channels;
- codec_info_.rate = bitsPerSample * samplesPerSec;
-
- // Calculate the packet size for 10ms frames
- switch(formatTag)
- {
- case kWaveFormatALaw:
- strcpy(codec_info_.plname, "PCMA");
- _codecId = kCodecPcma;
- codec_info_.pltype = 8;
- codec_info_.pacsize = codec_info_.plfreq / 100;
- break;
- case kWaveFormatMuLaw:
- strcpy(codec_info_.plname, "PCMU");
- _codecId = kCodecPcmu;
- codec_info_.pltype = 0;
- codec_info_.pacsize = codec_info_.plfreq / 100;
- break;
- case kWaveFormatPcm:
- codec_info_.pacsize = (bitsPerSample * (codec_info_.plfreq / 100)) / 8;
- if(samplesPerSec == 8000)
- {
- strcpy(codec_info_.plname, "L16");
- _codecId = kCodecL16_8Khz;
- }
- else if(samplesPerSec == 16000)
- {
- strcpy(codec_info_.plname, "L16");
- _codecId = kCodecL16_16kHz;
- }
- else if(samplesPerSec == 32000)
- {
- strcpy(codec_info_.plname, "L16");
- _codecId = kCodecL16_32Khz;
- }
- // Set the packet size for "odd" sampling frequencies so that it
- // properly corresponds to _readSizeBytes.
- else if(samplesPerSec == 11025)
- {
- strcpy(codec_info_.plname, "L16");
- _codecId = kCodecL16_16kHz;
- codec_info_.pacsize = 110;
- codec_info_.plfreq = 11000;
- }
- else if(samplesPerSec == 22050)
- {
- strcpy(codec_info_.plname, "L16");
- _codecId = kCodecL16_16kHz;
- codec_info_.pacsize = 220;
- codec_info_.plfreq = 22000;
- }
- else if(samplesPerSec == 44100)
- {
- strcpy(codec_info_.plname, "L16");
- _codecId = kCodecL16_16kHz;
- codec_info_.pacsize = 440;
- codec_info_.plfreq = 44000;
- }
- else if(samplesPerSec == 48000)
- {
- strcpy(codec_info_.plname, "L16");
- _codecId = kCodecL16_16kHz;
- codec_info_.pacsize = 480;
- codec_info_.plfreq = 48000;
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Unsupported PCM frequency!");
- return -1;
- }
- break;
- default:
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "unknown WAV format TAG!");
- return -1;
- break;
- }
- return 0;
-}
-
-WebRtc_Word32 ModuleFileUtility::InitWavReading(InStream& wav,
- const WebRtc_UWord32 start,
- const WebRtc_UWord32 stop)
-{
-
- _reading = false;
-
- if(ReadWavHeader(wav) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "failed to read WAV header!");
- return -1;
- }
-
- _playoutPositionMs = 0;
- _readPos = 0;
-
- if(start > 0)
- {
- WebRtc_UWord8 dummy[WAV_MAX_BUFFER_SIZE];
- WebRtc_Word32 readLength;
- if(_readSizeBytes <= WAV_MAX_BUFFER_SIZE)
- {
- while (_playoutPositionMs < start)
- {
- readLength = wav.Read(dummy, _readSizeBytes);
- if(readLength == _readSizeBytes)
- {
- _readPos += readLength;
- _playoutPositionMs += 10;
- }
- else // Must have reached EOF before start position!
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "InitWavReading(), EOF before start position");
- return -1;
- }
- }
- }
- else
- {
- return -1;
- }
- }
- if( InitWavCodec(_wavFormatObj.nSamplesPerSec, _wavFormatObj.nChannels,
- _wavFormatObj.nBitsPerSample,
- _wavFormatObj.formatTag) != 0)
- {
- return -1;
- }
- _bytesPerSample = _wavFormatObj.nBitsPerSample / 8;
-
-
- _startPointInMs = start;
- _stopPointInMs = stop;
- _reading = true;
- return 0;
-}
-
-WebRtc_Word32 ModuleFileUtility::ReadWavDataAsMono(
- InStream& wav,
- WebRtc_Word8* outData,
- const WebRtc_UWord32 bufferSize)
-{
- WEBRTC_TRACE(
- kTraceStream,
- kTraceFile,
- _id,
- "ModuleFileUtility::ReadWavDataAsMono(wav= 0x%x, outData= 0x%d,\
- bufSize= %ld)",
- &wav,
- outData,
- bufferSize);
-
- // The number of bytes that should be read from file.
- const WebRtc_UWord32 totalBytesNeeded = _readSizeBytes;
- // The number of bytes that will be written to outData.
- const WebRtc_UWord32 bytesRequested = (codec_info_.channels == 2) ?
- totalBytesNeeded >> 1 : totalBytesNeeded;
- if(bufferSize < bytesRequested)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "ReadWavDataAsMono: output buffer is too short!");
- return -1;
- }
- if(outData == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "ReadWavDataAsMono: output buffer NULL!");
- return -1;
- }
-
- if(!_reading)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "ReadWavDataAsMono: no longer reading file.");
- return -1;
- }
-
- WebRtc_Word32 bytesRead = ReadWavData(
- wav,
- (codec_info_.channels == 2) ? _tempData : (WebRtc_UWord8*)outData,
- totalBytesNeeded);
- if(bytesRead == 0)
- {
- return 0;
- }
- if(bytesRead < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "ReadWavDataAsMono: failed to read data from WAV file.");
- return -1;
- }
- // Output data is should be mono.
- if(codec_info_.channels == 2)
- {
- for (WebRtc_UWord32 i = 0; i < bytesRequested / _bytesPerSample; i++)
- {
- // Sample value is the average of left and right buffer rounded to
- // closest integer value. Note samples can be either 1 or 2 byte.
- if(_bytesPerSample == 1)
- {
- _tempData[i] = ((_tempData[2 * i] + _tempData[(2 * i) + 1] +
- 1) >> 1);
- }
- else
- {
- WebRtc_Word16* sampleData = (WebRtc_Word16*) _tempData;
- sampleData[i] = ((sampleData[2 * i] + sampleData[(2 * i) + 1] +
- 1) >> 1);
- }
- }
- memcpy(outData, _tempData, bytesRequested);
- }
- return bytesRequested;
-}
-
-WebRtc_Word32 ModuleFileUtility::ReadWavDataAsStereo(
- InStream& wav,
- WebRtc_Word8* outDataLeft,
- WebRtc_Word8* outDataRight,
- const WebRtc_UWord32 bufferSize)
-{
- WEBRTC_TRACE(
- kTraceStream,
- kTraceFile,
- _id,
- "ModuleFileUtility::ReadWavDataAsStereo(wav= 0x%x, outLeft= 0x%x,\
- outRight= 0x%x, bufSize= %ld)",
- &wav,
- outDataLeft,
- outDataRight,
- bufferSize);
-
- if((outDataLeft == NULL) ||
- (outDataRight == NULL))
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "ReadWavDataAsMono: an input buffer is NULL!");
- return -1;
- }
- if(codec_info_.channels != 2)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceFile,
- _id,
- "ReadWavDataAsStereo: WAV file does not contain stereo data!");
- return -1;
- }
- if(! _reading)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "ReadWavDataAsStereo: no longer reading file.");
- return -1;
- }
-
- // The number of bytes that should be read from file.
- const WebRtc_UWord32 totalBytesNeeded = _readSizeBytes;
- // The number of bytes that will be written to the left and the right
- // buffers.
- const WebRtc_UWord32 bytesRequested = totalBytesNeeded >> 1;
- if(bufferSize < bytesRequested)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "ReadWavData: Output buffers are too short!");
- assert(false);
- return -1;
- }
-
- WebRtc_Word32 bytesRead = ReadWavData(wav, _tempData, totalBytesNeeded);
- if(bytesRead <= 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "ReadWavDataAsStereo: failed to read data from WAV file.");
- return -1;
- }
-
- // Turn interleaved audio to left and right buffer. Note samples can be
- // either 1 or 2 bytes
- if(_bytesPerSample == 1)
- {
- for (WebRtc_UWord32 i = 0; i < bytesRequested; i++)
- {
- outDataLeft[i] = _tempData[2 * i];
- outDataRight[i] = _tempData[(2 * i) + 1];
- }
- }
- else if(_bytesPerSample == 2)
- {
- WebRtc_Word16* sampleData = reinterpret_cast<WebRtc_Word16*>(_tempData);
- WebRtc_Word16* outLeft = reinterpret_cast<WebRtc_Word16*>(outDataLeft);
- WebRtc_Word16* outRight = reinterpret_cast<WebRtc_Word16*>(
- outDataRight);
-
- // Bytes requested to samples requested.
- WebRtc_UWord32 sampleCount = bytesRequested >> 1;
- for (WebRtc_UWord32 i = 0; i < sampleCount; i++)
- {
- outLeft[i] = sampleData[2 * i];
- outRight[i] = sampleData[(2 * i) + 1];
- }
- } else {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "ReadWavStereoData: unsupported sample size %d!",
- _bytesPerSample);
- assert(false);
- return -1;
- }
- return bytesRequested;
-}
-
-WebRtc_Word32 ModuleFileUtility::ReadWavData(
- InStream& wav,
- WebRtc_UWord8* buffer,
- const WebRtc_UWord32 dataLengthInBytes)
-{
- WEBRTC_TRACE(
- kTraceStream,
- kTraceFile,
- _id,
- "ModuleFileUtility::ReadWavData(wav= 0x%x, buffer= 0x%x, dataLen= %ld)",
- &wav,
- buffer,
- dataLengthInBytes);
-
-
- if(buffer == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "ReadWavDataAsMono: output buffer NULL!");
- return -1;
- }
-
- // Make sure that a read won't return too few samples.
- // TODO (hellner): why not read the remaining bytes needed from the start
- // of the file?
- if((_dataSize - _readPos) < (WebRtc_Word32)dataLengthInBytes)
- {
- // Rewind() being -1 may be due to the file not supposed to be looped.
- if(wav.Rewind() == -1)
- {
- _reading = false;
- return 0;
- }
- if(InitWavReading(wav, _startPointInMs, _stopPointInMs) == -1)
- {
- _reading = false;
- return -1;
- }
- }
-
- WebRtc_Word32 bytesRead = wav.Read(buffer, dataLengthInBytes);
- if(bytesRead < 0)
- {
- _reading = false;
- return -1;
- }
-
- // This should never happen due to earlier sanity checks.
- // TODO (hellner): change to an assert and fail here since this should
- // never happen...
- if(bytesRead < (WebRtc_Word32)dataLengthInBytes)
- {
- if((wav.Rewind() == -1) ||
- (InitWavReading(wav, _startPointInMs, _stopPointInMs) == -1))
- {
- _reading = false;
- return -1;
- }
- else
- {
- bytesRead = wav.Read(buffer, dataLengthInBytes);
- if(bytesRead < (WebRtc_Word32)dataLengthInBytes)
- {
- _reading = false;
- return -1;
- }
- }
- }
-
- _readPos += bytesRead;
-
- // TODO (hellner): Why is dataLengthInBytes let dictate the number of bytes
- // to read when exactly 10ms should be read?!
- _playoutPositionMs += 10;
- if((_stopPointInMs > 0) &&
- (_playoutPositionMs >= _stopPointInMs))
- {
- if((wav.Rewind() == -1) ||
- (InitWavReading(wav, _startPointInMs, _stopPointInMs) == -1))
- {
- _reading = false;
- }
- }
- return bytesRead;
-}
-
-WebRtc_Word32 ModuleFileUtility::InitWavWriting(OutStream& wav,
- const CodecInst& codecInst)
-{
-
- if(set_codec_info(codecInst) != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "codecInst identifies unsupported codec!");
- return -1;
- }
- _writing = false;
- WebRtc_UWord32 channels = (codecInst.channels == 0) ?
- 1 : codecInst.channels;
-
- if(STR_CASE_CMP(codecInst.plname, "PCMU") == 0)
- {
- _bytesPerSample = 1;
- if(WriteWavHeader(wav, 8000, _bytesPerSample, channels,
- kWaveFormatMuLaw, 0) == -1)
- {
- return -1;
- }
- }else if(STR_CASE_CMP(codecInst.plname, "PCMA") == 0)
- {
- _bytesPerSample = 1;
- if(WriteWavHeader(wav, 8000, _bytesPerSample, channels, kWaveFormatALaw,
- 0) == -1)
- {
- return -1;
- }
- }
- else if(STR_CASE_CMP(codecInst.plname, "L16") == 0)
- {
- _bytesPerSample = 2;
- if(WriteWavHeader(wav, codecInst.plfreq, _bytesPerSample, channels,
- kWaveFormatPcm, 0) == -1)
- {
- return -1;
- }
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "codecInst identifies unsupported codec for WAV file!");
- return -1;
- }
- _writing = true;
- _bytesWritten = 0;
- return 0;
-}
-
-WebRtc_Word32 ModuleFileUtility::WriteWavData(OutStream& out,
- const WebRtc_Word8* buffer,
- const WebRtc_UWord32 dataLength)
-{
- WEBRTC_TRACE(
- kTraceStream,
- kTraceFile,
- _id,
- "ModuleFileUtility::WriteWavData(out= 0x%x, buf= 0x%x, dataLen= %d)",
- &out,
- buffer,
- dataLength);
-
- if(buffer == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "WriteWavData: input buffer NULL!");
- return -1;
- }
-
- if(!out.Write(buffer, dataLength))
- {
- return -1;
- }
- _bytesWritten += dataLength;
- return dataLength;
-}
-
-
-WebRtc_Word32 ModuleFileUtility::WriteWavHeader(
- OutStream& wav,
- const WebRtc_UWord32 freq,
- const WebRtc_UWord32 bytesPerSample,
- const WebRtc_UWord32 channels,
- const WebRtc_UWord32 format,
- const WebRtc_UWord32 lengthInBytes)
-{
-
- // Frame size in bytes for 10 ms of audio.
- // TODO (hellner): 44.1 kHz has 440 samples frame size. Doesn't seem to
- // be taken into consideration here!
- WebRtc_Word32 frameSize = (freq / 100) * bytesPerSample * channels;
-
- // Calculate the number of full frames that the wave file contain.
- const WebRtc_Word32 dataLengthInBytes = frameSize *
- (lengthInBytes / frameSize);
-
- WebRtc_Word8 tmpStr[4];
- WebRtc_Word8 tmpChar;
- WebRtc_UWord32 tmpLong;
-
- memcpy(tmpStr, "RIFF", 4);
- wav.Write(tmpStr, 4);
-
- tmpLong = dataLengthInBytes + 36;
- tmpChar = (WebRtc_Word8)(tmpLong);
- wav.Write(&tmpChar, 1);
- tmpChar = (WebRtc_Word8)(tmpLong >> 8);
- wav.Write(&tmpChar, 1);
- tmpChar = (WebRtc_Word8)(tmpLong >> 16);
- wav.Write(&tmpChar, 1);
- tmpChar = (WebRtc_Word8)(tmpLong >> 24);
- wav.Write(&tmpChar, 1);
-
- memcpy(tmpStr, "WAVE", 4);
- wav.Write(tmpStr, 4);
-
- memcpy(tmpStr, "fmt ", 4);
- wav.Write(tmpStr, 4);
-
- tmpChar = 16;
- wav.Write(&tmpChar, 1);
- tmpChar = 0;
- wav.Write(&tmpChar, 1);
- tmpChar = 0;
- wav.Write(&tmpChar, 1);
- tmpChar = 0;
- wav.Write(&tmpChar, 1);
-
- tmpChar = (WebRtc_Word8)(format);
- wav.Write(&tmpChar, 1);
- tmpChar = 0;
- wav.Write(&tmpChar, 1);
-
- tmpChar = (WebRtc_Word8)(channels);
- wav.Write(&tmpChar, 1);
- tmpChar = 0;
- wav.Write(&tmpChar, 1);
-
- tmpLong = freq;
- tmpChar = (WebRtc_Word8)(tmpLong);
- wav.Write(&tmpChar, 1);
- tmpChar = (WebRtc_Word8)(tmpLong >> 8);
- wav.Write(&tmpChar, 1);
- tmpChar = (WebRtc_Word8)(tmpLong >> 16);
- wav.Write(&tmpChar, 1);
- tmpChar = (WebRtc_Word8)(tmpLong >> 24);
- wav.Write(&tmpChar, 1);
-
- // nAverageBytesPerSec = Sample rate * Bytes per sample * Channels
- tmpLong = bytesPerSample * freq * channels;
- tmpChar = (WebRtc_Word8)(tmpLong);
- wav.Write(&tmpChar, 1);
- tmpChar = (WebRtc_Word8)(tmpLong >> 8);
- wav.Write(&tmpChar, 1);
- tmpChar = (WebRtc_Word8)(tmpLong >> 16);
- wav.Write(&tmpChar, 1);
- tmpChar = (WebRtc_Word8)(tmpLong >> 24);
- wav.Write(&tmpChar, 1);
-
- // nBlockAlign = Bytes per sample * Channels
- tmpChar = (WebRtc_Word8)(bytesPerSample * channels);
- wav.Write(&tmpChar, 1);
- tmpChar = 0;
- wav.Write(&tmpChar, 1);
-
- tmpChar = (WebRtc_Word8)(bytesPerSample*8);
- wav.Write(&tmpChar, 1);
- tmpChar = 0;
- wav.Write(&tmpChar, 1);
-
- memcpy(tmpStr, "data", 4);
- wav.Write(tmpStr, 4);
-
- tmpLong = dataLengthInBytes;
- tmpChar = (WebRtc_Word8)(tmpLong);
- wav.Write(&tmpChar, 1);
- tmpChar = (WebRtc_Word8)(tmpLong >> 8);
- wav.Write(&tmpChar, 1);
- tmpChar = (WebRtc_Word8)(tmpLong >> 16);
- wav.Write(&tmpChar, 1);
- tmpChar = (WebRtc_Word8)(tmpLong >> 24);
- wav.Write(&tmpChar, 1);
-
- return 0;
-}
-
-WebRtc_Word32 ModuleFileUtility::UpdateWavHeader(OutStream& wav)
-{
- WebRtc_Word32 res = -1;
- if(wav.Rewind() == -1)
- {
- return -1;
- }
- WebRtc_UWord32 channels = (codec_info_.channels == 0) ?
- 1 : codec_info_.channels;
-
- if(STR_CASE_CMP(codec_info_.plname, "L16") == 0)
- {
- res = WriteWavHeader(wav, codec_info_.plfreq, 2, channels,
- kWaveFormatPcm, _bytesWritten);
- } else if(STR_CASE_CMP(codec_info_.plname, "PCMU") == 0) {
- res = WriteWavHeader(wav, 8000, 1, channels, kWaveFormatMuLaw,
- _bytesWritten);
- } else if(STR_CASE_CMP(codec_info_.plname, "PCMA") == 0) {
- res = WriteWavHeader(wav, 8000, 1, channels, kWaveFormatALaw,
- _bytesWritten);
- } else {
- // Allow calling this API even if not writing to a WAVE file.
- // TODO (hellner): why?!
- return 0;
- }
- return res;
-}
-
-
-WebRtc_Word32 ModuleFileUtility::InitPreEncodedReading(InStream& in,
- const CodecInst& cinst)
-{
-
- WebRtc_UWord8 preEncodedID;
- in.Read(&preEncodedID, 1);
-
- MediaFileUtility_CodecType codecType =
- (MediaFileUtility_CodecType)preEncodedID;
-
- if(set_codec_info(cinst) != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Pre-encoded file send codec mismatch!");
- return -1;
- }
- if(codecType != _codecId)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "Pre-encoded file format codec mismatch!");
- return -1;
- }
- memcpy(&codec_info_,&cinst,sizeof(CodecInst));
- _reading = true;
- return 0;
-}
-
-WebRtc_Word32 ModuleFileUtility::ReadPreEncodedData(
- InStream& in,
- WebRtc_Word8* outData,
- const WebRtc_UWord32 bufferSize)
-{
- WEBRTC_TRACE(
- kTraceStream,
- kTraceFile,
- _id,
- "ModuleFileUtility::ReadPreEncodedData(in= 0x%x, outData= 0x%x,\
- bufferSize= %d)",
- &in,
- outData,
- bufferSize);
-
- if(outData == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id, "output buffer NULL");
- }
-
- WebRtc_UWord32 frameLen;
- WebRtc_UWord8 buf[64];
- // Each frame has a two byte header containing the frame length.
- WebRtc_Word32 res = in.Read(buf, 2);
- if(res != 2)
- {
- if(!in.Rewind())
- {
- // The first byte is the codec identifier.
- in.Read(buf, 1);
- res = in.Read(buf, 2);
- }
- else
- {
- return -1;
- }
- }
- frameLen = buf[0] + buf[1] * 256;
- if(bufferSize < frameLen)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceFile,
- _id,
- "buffer not large enough to read %d bytes of pre-encoded data!",
- frameLen);
- return -1;
- }
- return in.Read(outData, frameLen);
-}
-
-WebRtc_Word32 ModuleFileUtility::InitPreEncodedWriting(
- OutStream& out,
- const CodecInst& codecInst)
-{
-
- if(set_codec_info(codecInst) != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id, "CodecInst not recognized!");
- return -1;
- }
- _writing = true;
- _bytesWritten = 1;
- out.Write(&_codecId, 1);
- return 0;
-}
-
-WebRtc_Word32 ModuleFileUtility::WritePreEncodedData(
- OutStream& out,
- const WebRtc_Word8* buffer,
- const WebRtc_UWord32 dataLength)
-{
- WEBRTC_TRACE(
- kTraceStream,
- kTraceFile,
- _id,
- "ModuleFileUtility::WritePreEncodedData(out= 0x%x, inData= 0x%x,\
- dataLen= %d)",
- &out,
- buffer,
- dataLength);
-
- if(buffer == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,"buffer NULL");
- }
-
- WebRtc_Word32 bytesWritten = 0;
- // The first two bytes is the size of the frame.
- WebRtc_Word16 lengthBuf;
- lengthBuf = (WebRtc_Word16)dataLength;
- if(!out.Write(&lengthBuf, 2))
- {
- return -1;
- }
- bytesWritten = 2;
-
- if(!out.Write(buffer, dataLength))
- {
- return -1;
- }
- bytesWritten += dataLength;
- return bytesWritten;
-}
-
-WebRtc_Word32 ModuleFileUtility::InitCompressedReading(
- InStream& in,
- const WebRtc_UWord32 start,
- const WebRtc_UWord32 stop)
-{
- WEBRTC_TRACE(
- kTraceDebug,
- kTraceFile,
- _id,
- "ModuleFileUtility::InitCompressedReading(in= 0x%x, start= %d,\
- stop= %d)",
- &in,
- start,
- stop);
-
-#if defined(WEBRTC_CODEC_GSMAMR) || defined(WEBRTC_CODEC_GSMAMRWB) || \
- defined(WEBRTC_CODEC_ILBC)
- WebRtc_Word16 read_len = 0;
-#endif
- _codecId = kCodecNoCodec;
- _playoutPositionMs = 0;
- _reading = false;
-
- _startPointInMs = start;
- _stopPointInMs = stop;
-
-#ifdef WEBRTC_CODEC_GSMAMR
- WebRtc_Word32 AMRmode2bytes[9]={12,13,15,17,19,20,26,31,5};
-#endif
-#ifdef WEBRTC_CODEC_GSMAMRWB
- WebRtc_Word32 AMRWBmode2bytes[10]={17,23,32,36,40,46,50,58,60,6};
-#endif
-
- // Read the codec name
- WebRtc_Word32 cnt = 0;
- WebRtc_Word8 buf[64];
- do
- {
- in.Read(&buf[cnt++], 1);
- } while ((buf[cnt-1] != '\n') && (64 > cnt));
-
- if(cnt==64)
- {
- return -1;
- } else {
- buf[cnt]=0;
- }
-
-#ifdef WEBRTC_CODEC_GSMAMR
- if(!strcmp("#!AMR\n", buf))
- {
- strcpy(codec_info_.plname, "amr");
- codec_info_.pacsize = 160;
- _codecId = kCodecAmr;
- codec_info_.pltype = 112;
- codec_info_.rate = 12200;
- codec_info_.plfreq = 8000;
- codec_info_.channels = 1;
-
- WebRtc_Word16 mode = 0;
- if(_startPointInMs > 0)
- {
- while (_playoutPositionMs <= _startPointInMs)
- {
- // First read byte contain the AMR mode.
- read_len = in.Read(buf, 1);
- if(read_len != 1)
- {
- return -1;
- }
-
- mode = (buf[0]>>3)&0xF;
- if((mode < 0) || (mode > 8))
- {
- if(mode != 15)
- {
- return -1;
- }
- }
- if(mode != 15)
- {
- read_len = in.Read(&buf[1], AMRmode2bytes[mode]);
- if(read_len != AMRmode2bytes[mode])
- {
- return -1;
- }
- }
- _playoutPositionMs += 20;
- }
- }
- }
-#endif
-#ifdef WEBRTC_CODEC_GSMAMRWB
- if(!strcmp("#!AMRWB\n", buf))
- {
- strcpy(codec_info_.plname, "amr-wb");
- codec_info_.pacsize = 320;
- _codecId = kCodecAmrWb;
- codec_info_.pltype = 120;
- codec_info_.rate = 20000;
- codec_info_.plfreq = 16000;
- codec_info_.channels = 1;
-
- WebRtc_Word16 mode = 0;
- if(_startPointInMs > 0)
- {
- while (_playoutPositionMs <= _startPointInMs)
- {
- // First read byte contain the AMR mode.
- read_len = in.Read(buf, 1);
- if(read_len != 1)
- {
- return -1;
- }
-
- mode = (buf[0]>>3)&0xF;
- if((mode < 0) || (mode > 9))
- {
- if(mode != 15)
- {
- return -1;
- }
- }
- if(mode != 15)
- {
- read_len = in.Read(&buf[1], AMRWBmode2bytes[mode]);
- if(read_len != AMRWBmode2bytes[mode])
- {
- return -1;
- }
- }
- _playoutPositionMs += 20;
- }
- }
- }
-#endif
-#ifdef WEBRTC_CODEC_ILBC
- if(!strcmp("#!iLBC20\n", buf))
- {
- codec_info_.pltype = 102;
- strcpy(codec_info_.plname, "ilbc");
- codec_info_.plfreq = 8000;
- codec_info_.pacsize = 160;
- codec_info_.channels = 1;
- codec_info_.rate = 13300;
- _codecId = kCodecIlbc20Ms;
-
- if(_startPointInMs > 0)
- {
- while (_playoutPositionMs <= _startPointInMs)
- {
- read_len = in.Read(buf, 38);
- if(read_len == 38)
- {
- _playoutPositionMs += 20;
- }
- else
- {
- return -1;
- }
- }
- }
- }
-
- if(!strcmp("#!iLBC30\n", buf))
- {
- codec_info_.pltype = 102;
- strcpy(codec_info_.plname, "ilbc");
- codec_info_.plfreq = 8000;
- codec_info_.pacsize = 240;
- codec_info_.channels = 1;
- codec_info_.rate = 13300;
- _codecId = kCodecIlbc30Ms;
-
- if(_startPointInMs > 0)
- {
- while (_playoutPositionMs <= _startPointInMs)
- {
- read_len = in.Read(buf, 50);
- if(read_len == 50)
- {
- _playoutPositionMs += 20;
- }
- else
- {
- return -1;
- }
- }
- }
- }
-#endif
- if(_codecId == kCodecNoCodec)
- {
- return -1;
- }
- _reading = true;
- return 0;
-}
-
-WebRtc_Word32 ModuleFileUtility::ReadCompressedData(InStream& in,
- WebRtc_Word8* outData,
- WebRtc_UWord32 bufferSize)
-{
- WEBRTC_TRACE(
- kTraceStream,
- kTraceFile,
- _id,
- "ModuleFileUtility::ReadCompressedData(in=0x%x, outData=0x%x,\
- bytes=%ld)",
- &in,
- outData,
- bufferSize);
-
-#ifdef WEBRTC_CODEC_GSMAMR
- WebRtc_UWord32 AMRmode2bytes[9]={12,13,15,17,19,20,26,31,5};
-#endif
-#ifdef WEBRTC_CODEC_GSMAMRWB
- WebRtc_UWord32 AMRWBmode2bytes[10]={17,23,32,36,40,46,50,58,60,6};
-#endif
- WebRtc_UWord32 bytesRead = 0;
-
- if(! _reading)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id, "not currently reading!");
- return -1;
- }
-
-#ifdef WEBRTC_CODEC_GSMAMR
- if(_codecId == kCodecAmr)
- {
- WebRtc_Word32 res = in.Read(outData, 1);
- if(res != 1)
- {
- if(!in.Rewind())
- {
- InitCompressedReading(in, _startPointInMs, _stopPointInMs);
- res = in.Read(outData, 1);
- if(res != 1)
- {
- _reading = false;
- return -1;
- }
- }
- else
- {
- _reading = false;
- return -1;
- }
- }
- const WebRtc_Word16 mode = (outData[0]>>3)&0xF;
- if((mode < 0) ||
- (mode > 8))
- {
- if(mode != 15)
- {
- return -1;
- }
- }
- if(mode != 15)
- {
- if(bufferSize < AMRmode2bytes[mode] + 1)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceFile,
- _id,
- "output buffer is too short to read AMR compressed data.");
- assert(false);
- return -1;
- }
- bytesRead = in.Read(&outData[1], AMRmode2bytes[mode]);
- if(bytesRead != AMRmode2bytes[mode])
- {
- _reading = false;
- return -1;
- }
- // Count the mode byte to bytes read.
- bytesRead++;
- }
- else
- {
- bytesRead = 1;
- }
- }
-#endif
-#ifdef WEBRTC_CODEC_GSMAMRWB
- if(_codecId == kCodecAmrWb)
- {
- WebRtc_Word32 res = in.Read(outData, 1);
- if(res != 1)
- {
- if(!in.Rewind())
- {
- InitCompressedReading(in, _startPointInMs, _stopPointInMs);
- res = in.Read(outData, 1);
- if(res != 1)
- {
- _reading = false;
- return -1;
- }
- }
- else
- {
- _reading = false;
- return -1;
- }
- }
- WebRtc_Word16 mode = (outData[0]>>3)&0xF;
- if((mode < 0) ||
- (mode > 8))
- {
- if(mode != 15)
- {
- return -1;
- }
- }
- if(mode != 15)
- {
- if(bufferSize < AMRWBmode2bytes[mode] + 1)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "output buffer is too short to read AMRWB\
- compressed.");
- assert(false);
- return -1;
- }
- bytesRead = in.Read(&outData[1], AMRWBmode2bytes[mode]);
- if(bytesRead != AMRWBmode2bytes[mode])
- {
- _reading = false;
- return -1;
- }
- bytesRead++;
- }
- else
- {
- bytesRead = 1;
- }
- }
-#endif
-#ifdef WEBRTC_CODEC_ILBC
- if((_codecId == kCodecIlbc20Ms) ||
- (_codecId == kCodecIlbc30Ms))
- {
- WebRtc_UWord32 byteSize = 0;
- if(_codecId == kCodecIlbc30Ms)
- {
- byteSize = 50;
- }
- if(_codecId == kCodecIlbc20Ms)
- {
- byteSize = 38;
- }
- if(bufferSize < byteSize)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "output buffer is too short to read ILBC compressed\
- data.");
- assert(false);
- return -1;
- }
-
- bytesRead = in.Read(outData, byteSize);
- if(bytesRead != byteSize)
- {
- if(!in.Rewind())
- {
- InitCompressedReading(in, _startPointInMs, _stopPointInMs);
- bytesRead = in.Read(outData, byteSize);
- if(bytesRead != byteSize)
- {
- _reading = false;
- return -1;
- }
- }
- else
- {
- _reading = false;
- return -1;
- }
- }
- }
-#endif
- if(bytesRead == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "ReadCompressedData() no bytes read, codec not supported");
- return -1;
- }
-
- _playoutPositionMs += 20;
- if((_stopPointInMs > 0) &&
- (_playoutPositionMs >= _stopPointInMs))
- {
- if(!in.Rewind())
- {
- InitCompressedReading(in, _startPointInMs, _stopPointInMs);
- }
- else
- {
- _reading = false;
- }
- }
- return bytesRead;
-}
-
-WebRtc_Word32 ModuleFileUtility::InitCompressedWriting(
- OutStream& out,
- const CodecInst& codecInst)
-{
- WEBRTC_TRACE(kTraceDebug, kTraceFile, _id,
- "ModuleFileUtility::InitCompressedWriting(out= 0x%x,\
- codecName= %s)",
- &out, codecInst.plname);
-
- _writing = false;
-
-#ifdef WEBRTC_CODEC_GSMAMR
- if(STR_CASE_CMP(codecInst.plname, "amr") == 0)
- {
- if(codecInst.pacsize == 160)
- {
- memcpy(&codec_info_,&codecInst,sizeof(CodecInst));
- _codecId = kCodecAmr;
- out.Write("#!AMR\n",6);
- _writing = true;
- return 0;
- }
- }
-#endif
-#ifdef WEBRTC_CODEC_GSMAMRWB
- if(STR_CASE_CMP(codecInst.plname, "amr-wb") == 0)
- {
- if(codecInst.pacsize == 320)
- {
- memcpy(&codec_info_,&codecInst,sizeof(CodecInst));
- _codecId = kCodecAmrWb;
- out.Write("#!AMRWB\n",8);
- _writing = true;
- return 0;
- }
- }
-#endif
-#ifdef WEBRTC_CODEC_ILBC
- if(STR_CASE_CMP(codecInst.plname, "ilbc") == 0)
- {
- if(codecInst.pacsize == 160)
- {
- _codecId = kCodecIlbc20Ms;
- out.Write("#!iLBC20\n",9);
- }
- else if(codecInst.pacsize == 240)
- {
- _codecId = kCodecIlbc30Ms;
- out.Write("#!iLBC30\n",9);
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "codecInst defines unsupported compression codec!");
- return -1;
- }
- memcpy(&codec_info_,&codecInst,sizeof(CodecInst));
- _writing = true;
- return 0;
- }
-#endif
-
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "codecInst defines unsupported compression codec!");
- return -1;
-}
-
-WebRtc_Word32 ModuleFileUtility::WriteCompressedData(
- OutStream& out,
- const WebRtc_Word8* buffer,
- const WebRtc_UWord32 dataLength)
-{
- WEBRTC_TRACE(
- kTraceStream,
- kTraceFile,
- _id,
- "ModuleFileUtility::WriteCompressedData(out= 0x%x, buf= 0x%x,\
- dataLen= %d)",
- &out,
- buffer,
- dataLength);
-
- if(buffer == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,"buffer NULL");
- }
-
- if(!out.Write(buffer, dataLength))
- {
- return -1;
- }
- return dataLength;
-}
-
-WebRtc_Word32 ModuleFileUtility::InitPCMReading(InStream& pcm,
- const WebRtc_UWord32 start,
- const WebRtc_UWord32 stop,
- WebRtc_UWord32 freq)
-{
- WEBRTC_TRACE(
- kTraceInfo,
- kTraceFile,
- _id,
- "ModuleFileUtility::InitPCMReading(pcm= 0x%x, start=%d, stop=%d,\
- freq=%d)",
- &pcm,
- start,
- stop,
- freq);
-
- WebRtc_Word8 dummy[320];
- WebRtc_Word32 read_len;
-
- _playoutPositionMs = 0;
- _startPointInMs = start;
- _stopPointInMs = stop;
- _reading = false;
-
- if(freq == 8000)
- {
- strcpy(codec_info_.plname, "L16");
- codec_info_.pltype = -1;
- codec_info_.plfreq = 8000;
- codec_info_.pacsize = 160;
- codec_info_.channels = 1;
- codec_info_.rate = 128000;
- _codecId = kCodecL16_8Khz;
- }
- else if(freq == 16000)
- {
- strcpy(codec_info_.plname, "L16");
- codec_info_.pltype = -1;
- codec_info_.plfreq = 16000;
- codec_info_.pacsize = 320;
- codec_info_.channels = 1;
- codec_info_.rate = 256000;
- _codecId = kCodecL16_16kHz;
- }
- else if(freq == 32000)
- {
- strcpy(codec_info_.plname, "L16");
- codec_info_.pltype = -1;
- codec_info_.plfreq = 32000;
- codec_info_.pacsize = 320;
- codec_info_.channels = 1;
- codec_info_.rate = 512000;
- _codecId = kCodecL16_32Khz;
- }
-
- // Readsize for 10ms of audio data (2 bytes per sample).
- _readSizeBytes = 2 * codec_info_. plfreq / 100;
- if(_startPointInMs > 0)
- {
- while (_playoutPositionMs < _startPointInMs)
- {
- read_len = pcm.Read(dummy, _readSizeBytes);
- if(read_len == _readSizeBytes)
- {
- _playoutPositionMs += 10;
- }
- else // Must have reached EOF before start position!
- {
- return -1;
- }
- }
- }
- _reading = true;
- return 0;
-}
-
-WebRtc_Word32 ModuleFileUtility::ReadPCMData(InStream& pcm,
- WebRtc_Word8* outData,
- WebRtc_UWord32 bufferSize)
-{
- WEBRTC_TRACE(
- kTraceStream,
- kTraceFile,
- _id,
- "ModuleFileUtility::ReadPCMData(pcm= 0x%x, outData= 0x%x, bufSize= %d)",
- &pcm,
- outData,
- bufferSize);
-
- if(outData == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,"buffer NULL");
- }
-
- // Readsize for 10ms of audio data (2 bytes per sample).
- WebRtc_UWord32 bytesRequested = 2 * codec_info_.plfreq / 100;
- if(bufferSize < bytesRequested)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "ReadPCMData: buffer not long enough for a 10ms frame.");
- assert(false);
- return -1;
- }
-
- WebRtc_UWord32 bytesRead = pcm.Read(outData, bytesRequested);
- if(bytesRead < bytesRequested)
- {
- if(pcm.Rewind() == -1)
- {
- _reading = false;
- }
- else
- {
- if(InitPCMReading(pcm, _startPointInMs, _stopPointInMs,
- codec_info_.plfreq) == -1)
- {
- _reading = false;
- }
- else
- {
- WebRtc_Word32 rest = bytesRequested - bytesRead;
- WebRtc_Word32 len = pcm.Read(&(outData[bytesRead]), rest);
- if(len == rest)
- {
- bytesRead += len;
- }
- else
- {
- _reading = false;
- }
- }
- if(bytesRead <= 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "ReadPCMData: Failed to rewind audio file.");
- return -1;
- }
- }
- }
-
- if(bytesRead <= 0)
- {
- WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
- "ReadPCMData: end of file");
- return -1;
- }
- _playoutPositionMs += 10;
- if(_stopPointInMs && _playoutPositionMs >= _stopPointInMs)
- {
- if(!pcm.Rewind())
- {
- if(InitPCMReading(pcm, _startPointInMs, _stopPointInMs,
- codec_info_.plfreq) == -1)
- {
- _reading = false;
- }
- }
- }
- return bytesRead;
-}
-
-WebRtc_Word32 ModuleFileUtility::InitPCMWriting(OutStream& out,
- WebRtc_UWord32 freq)
-{
-
- if(freq == 8000)
- {
- strcpy(codec_info_.plname, "L16");
- codec_info_.pltype = -1;
- codec_info_.plfreq = 8000;
- codec_info_.pacsize = 160;
- codec_info_.channels = 1;
- codec_info_.rate = 128000;
-
- _codecId = kCodecL16_8Khz;
- }
- else if(freq == 16000)
- {
- strcpy(codec_info_.plname, "L16");
- codec_info_.pltype = -1;
- codec_info_.plfreq = 16000;
- codec_info_.pacsize = 320;
- codec_info_.channels = 1;
- codec_info_.rate = 256000;
-
- _codecId = kCodecL16_16kHz;
- }
- else if(freq == 32000)
- {
- strcpy(codec_info_.plname, "L16");
- codec_info_.pltype = -1;
- codec_info_.plfreq = 32000;
- codec_info_.pacsize = 320;
- codec_info_.channels = 1;
- codec_info_.rate = 512000;
-
- _codecId = kCodecL16_32Khz;
- }
- if((_codecId != kCodecL16_8Khz) &&
- (_codecId != kCodecL16_16kHz) &&
- (_codecId != kCodecL16_32Khz))
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "CodecInst is not 8KHz PCM or 16KHz PCM!");
- return -1;
- }
- _writing = true;
- _bytesWritten = 0;
- return 0;
-}
-
-WebRtc_Word32 ModuleFileUtility::WritePCMData(OutStream& out,
- const WebRtc_Word8* buffer,
- const WebRtc_UWord32 dataLength)
-{
- WEBRTC_TRACE(
- kTraceStream,
- kTraceFile,
- _id,
- "ModuleFileUtility::WritePCMData(out= 0x%x, buf= 0x%x, dataLen= %d)",
- &out,
- buffer,
- dataLength);
-
- if(buffer == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id, "buffer NULL");
- }
-
- if(!out.Write(buffer, dataLength))
- {
- return -1;
- }
-
- _bytesWritten += dataLength;
- return dataLength;
-}
-
-WebRtc_Word32 ModuleFileUtility::codec_info(CodecInst& codecInst)
-{
- WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
- "ModuleFileUtility::codec_info(codecInst= 0x%x)", &codecInst);
-
- if(!_reading && !_writing)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "CodecInst: not currently reading audio file!");
- return -1;
- }
- memcpy(&codecInst,&codec_info_,sizeof(CodecInst));
- return 0;
-}
-
-WebRtc_Word32 ModuleFileUtility::set_codec_info(const CodecInst& codecInst)
-{
-
- _codecId = kCodecNoCodec;
- if(STR_CASE_CMP(codecInst.plname, "PCMU") == 0)
- {
- _codecId = kCodecPcmu;
- }
- else if(STR_CASE_CMP(codecInst.plname, "PCMA") == 0)
- {
- _codecId = kCodecPcma;
- }
- else if(STR_CASE_CMP(codecInst.plname, "L16") == 0)
- {
- if(codecInst.plfreq == 8000)
- {
- _codecId = kCodecL16_8Khz;
- }
- else if(codecInst.plfreq == 16000)
- {
- _codecId = kCodecL16_16kHz;
- }
- else if(codecInst.plfreq == 32000)
- {
- _codecId = kCodecL16_32Khz;
- }
- }
-#ifdef WEBRTC_CODEC_GSMAMR
- else if(STR_CASE_CMP(codecInst.plname, "amr") == 0)
- {
- _codecId = kCodecAmr;
- }
-#endif
-#ifdef WEBRTC_CODEC_GSMAMRWB
- else if(STR_CASE_CMP(codecInst.plname, "amr-wb") == 0)
- {
- _codecId = kCodecAmrWb;
- }
-#endif
-#ifdef WEBRTC_CODEC_ILBC
- else if(STR_CASE_CMP(codecInst.plname, "ilbc") == 0)
- {
- if(codecInst.pacsize == 160)
- {
- _codecId = kCodecIlbc20Ms;
- }
- else if(codecInst.pacsize == 240)
- {
- _codecId = kCodecIlbc30Ms;
- }
- }
-#endif
-#if(defined(WEBRTC_CODEC_ISAC) || defined(WEBRTC_CODEC_ISACFX))
- else if(STR_CASE_CMP(codecInst.plname, "isac") == 0)
- {
- if(codecInst.plfreq == 16000)
- {
- _codecId = kCodecIsac;
- }
- else if(codecInst.plfreq == 32000)
- {
- _codecId = kCodecIsacSwb;
- }
- }
-#endif
-#ifdef WEBRTC_CODEC_ISACLC
- else if(STR_CASE_CMP(codecInst.plname, "isaclc") == 0)
- {
- _codecId = kCodecIsacLc;
- }
-#endif
-#ifdef WEBRTC_CODEC_G722
- else if(STR_CASE_CMP(codecInst.plname, "G722") == 0)
- {
- _codecId = kCodecG722;
- }
-#endif
- else if(STR_CASE_CMP(codecInst.plname, "G7221") == 0)
- {
-#ifdef WEBRTC_CODEC_G722_1
- if(codecInst.plfreq == 16000)
- {
- if(codecInst.rate == 16000)
- {
- _codecId = kCodecG722_1_16Kbps;
- }
- else if(codecInst.rate == 24000)
- {
- _codecId = kCodecG722_1_24Kbps;
- }
- else if(codecInst.rate == 32000)
- {
- _codecId = kCodecG722_1_32Kbps;
- }
- }
-#endif
-#ifdef WEBRTC_CODEC_G722_1C
- if(codecInst.plfreq == 32000)
- {
- if(codecInst.rate == 48000)
- {
- _codecId = kCodecG722_1c_48;
- }
- else if(codecInst.rate == 32000)
- {
- _codecId = kCodecG722_1c_32;
- }
- else if(codecInst.rate == 24000)
- {
- _codecId = kCodecG722_1c_24;
- }
- }
-#endif
- }
-#ifdef WEBRTC_CODEC_G726
- else if(STR_CASE_CMP(codecInst.plname, "G726-40") == 0)
- {
- _codecId = kCodecG726_40;
- }
- else if(STR_CASE_CMP(codecInst.plname, "G726-32") == 0)
- {
- _codecId = kCodecG726_24;
- }
- else if(STR_CASE_CMP(codecInst.plname, "G726-24") == 0)
- {
- _codecId = kCodecG726_32;
- }
- else if(STR_CASE_CMP(codecInst.plname, "G726-16") == 0)
- {
- _codecId = kCodecG726_16;
- }
-#endif
-#ifdef WEBRTC_CODEC_G729
- else if(STR_CASE_CMP(codecInst.plname, "G729") == 0)
- {
- _codecId = kCodecG729;
- }
-#endif
-#ifdef WEBRTC_CODEC_G729_1
- else if(STR_CASE_CMP(codecInst.plname, "G7291") == 0)
- {
- _codecId = kCodecG729_1;
- }
-#endif
-#ifdef WEBRTC_CODEC_SPEEX
- else if(STR_CASE_CMP(codecInst.plname, "speex") == 0)
- {
- if(codecInst.plfreq == 8000)
- {
- _codecId = kCodecSpeex8Khz;
- }
- else if(codecInst.plfreq == 16000)
- {
- _codecId = kCodecSpeex16Khz;
- }
- }
-#endif
- if(_codecId == kCodecNoCodec)
- {
- return -1;
- }
- memcpy(&codec_info_, &codecInst, sizeof(CodecInst));
- return 0;
-}
-
-WebRtc_Word32 ModuleFileUtility::FileDurationMs(const WebRtc_Word8* fileName,
- const FileFormats fileFormat,
- const WebRtc_UWord32 freqInHz)
-{
-
- if(fileName == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id, "filename NULL");
- return -1;
- }
-
- WebRtc_Word32 time_in_ms = -1;
- struct stat file_size;
- if(stat(fileName,&file_size) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "failed to retrieve file size with stat!");
- return -1;
- }
- FileWrapper* inStreamObj = FileWrapper::Create();
- if(inStreamObj == NULL)
- {
- WEBRTC_TRACE(kTraceMemory, kTraceFile, _id,
- "failed to create InStream object!");
- return -1;
- }
- if(inStreamObj->OpenFile(fileName, true) == -1)
- {
- delete inStreamObj;
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "failed to open file %s!", fileName);
- return -1;
- }
-
- switch (fileFormat)
- {
- case kFileFormatWavFile:
- {
- if(ReadWavHeader(*inStreamObj) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "failed to read WAV file header!");
- return -1;
- }
- time_in_ms = ((file_size.st_size - 44) /
- (_wavFormatObj.nAvgBytesPerSec/1000));
- break;
- }
- case kFileFormatPcm16kHzFile:
- {
- // 16 samples per ms. 2 bytes per sample.
- WebRtc_Word32 denominator = 16*2;
- time_in_ms = (file_size.st_size)/denominator;
- break;
- }
- case kFileFormatPcm8kHzFile:
- {
- // 8 samples per ms. 2 bytes per sample.
- WebRtc_Word32 denominator = 8*2;
- time_in_ms = (file_size.st_size)/denominator;
- break;
- }
- case kFileFormatCompressedFile:
- {
- WebRtc_Word32 cnt = 0;
- WebRtc_Word32 read_len = 0;
- WebRtc_Word8 buf[64];
- do
- {
- read_len = inStreamObj->Read(&buf[cnt++], 1);
- if(read_len != 1)
- {
- return -1;
- }
- } while ((buf[cnt-1] != '\n') && (64 > cnt));
-
- if(cnt == 64)
- {
- return -1;
- }
- else
- {
- buf[cnt] = 0;
- }
-#ifdef WEBRTC_CODEC_GSMAMR
- if(!strcmp("#!AMR\n", buf))
- {
- WebRtc_UWord8 dummy;
- read_len = inStreamObj->Read(&dummy, 1);
- if(read_len != 1)
- {
- return -1;
- }
-
- WebRtc_Word16 AMRMode = (dummy>>3)&0xF;
-
- // TODO (hellner): use tables instead of hardcoding like this!
- // Additionally, this calculation does not
- // take octet alignment into consideration.
- switch (AMRMode)
- {
- // Mode 0: 4.75 kbit/sec -> 95 bits per 20 ms frame.
- // 20 ms = 95 bits ->
- // file size in bytes * 8 / 95 is the number of
- // 20 ms frames in the file ->
- // time_in_ms = file size * 8 / 95 * 20
- case 0:
- time_in_ms = ((file_size.st_size)*160)/95;
- break;
- // Mode 1: 5.15 kbit/sec -> 103 bits per 20 ms frame.
- case 1:
- time_in_ms = ((file_size.st_size)*160)/103;
- break;
- // Mode 2: 5.90 kbit/sec -> 118 bits per 20 ms frame.
- case 2:
- time_in_ms = ((file_size.st_size)*160)/118;
- break;
- // Mode 3: 6.70 kbit/sec -> 134 bits per 20 ms frame.
- case 3:
- time_in_ms = ((file_size.st_size)*160)/134;
- break;
- // Mode 4: 7.40 kbit/sec -> 148 bits per 20 ms frame.
- case 4:
- time_in_ms = ((file_size.st_size)*160)/148;
- break;
- // Mode 5: 7.95 bit/sec -> 159 bits per 20 ms frame.
- case 5:
- time_in_ms = ((file_size.st_size)*160)/159;
- break;
- // Mode 6: 10.2 bit/sec -> 204 bits per 20 ms frame.
- case 6:
- time_in_ms = ((file_size.st_size)*160)/204;
- break;
- // Mode 7: 12.2 bit/sec -> 244 bits per 20 ms frame.
- case 7:
- time_in_ms = ((file_size.st_size)*160)/244;
- break;
- // Mode 8: SID Mode -> 39 bits per 20 ms frame.
- case 8:
- time_in_ms = ((file_size.st_size)*160)/39;
- break;
- default:
- break;
- }
- }
-#endif
-#ifdef WEBRTC_CODEC_GSMAMRWB
- if(!strcmp("#!AMRWB\n", buf))
- {
- WebRtc_UWord8 dummy;
- read_len = inStreamObj->Read(&dummy, 1);
- if(read_len != 1)
- {
- return -1;
- }
-
- // TODO (hellner): use tables instead of hardcoding like this!
- WebRtc_Word16 AMRWBMode = (dummy>>3)&0xF;
- switch(AMRWBMode)
- {
- // Mode 0: 6.6 kbit/sec -> 132 bits per 20 ms frame.
- case 0:
- time_in_ms = ((file_size.st_size)*160)/132;
- break;
- // Mode 1: 8.85 kbit/sec -> 177 bits per 20 ms frame.
- case 1:
- time_in_ms = ((file_size.st_size)*160)/177;
- break;
- // Mode 2: 12.65 kbit/sec -> 253 bits per 20 ms frame.
- case 2:
- time_in_ms = ((file_size.st_size)*160)/253;
- break;
- // Mode 3: 14.25 kbit/sec -> 285 bits per 20 ms frame.
- case 3:
- time_in_ms = ((file_size.st_size)*160)/285;
- break;
- // Mode 4: 15.85 kbit/sec -> 317 bits per 20 ms frame.
- case 4:
- time_in_ms = ((file_size.st_size)*160)/317;
- break;
- // Mode 5: 18.25 kbit/sec -> 365 bits per 20 ms frame.
- case 5:
- time_in_ms = ((file_size.st_size)*160)/365;
- break;
- // Mode 6: 19.85 kbit/sec -> 397 bits per 20 ms frame.
- case 6:
- time_in_ms = ((file_size.st_size)*160)/397;
- break;
- // Mode 7: 23.05 kbit/sec -> 461 bits per 20 ms frame.
- case 7:
- time_in_ms = ((file_size.st_size)*160)/461;
- break;
- // Mode 8: 23.85 kbit/sec -> 477 bits per 20 ms frame.
- case 8:
- time_in_ms = ((file_size.st_size)*160)/477;
- break;
- default:
- delete inStreamObj;
- return -1;
- }
- }
-#endif
-#ifdef WEBRTC_CODEC_ILBC
- if(!strcmp("#!iLBC20\n", buf))
- {
- // 20 ms is 304 bits
- time_in_ms = ((file_size.st_size)*160)/304;
- break;
- }
- if(!strcmp("#!iLBC30\n", buf))
- {
- // 30 ms takes 400 bits.
- // file size in bytes * 8 / 400 is the number of
- // 30 ms frames in the file ->
- // time_in_ms = file size * 8 / 400 * 30
- time_in_ms = ((file_size.st_size)*240)/400;
- break;
- }
-#endif
- }
- case kFileFormatPreencodedFile:
- {
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "cannot determine duration of Pre-Encoded file!");
- break;
- }
- default:
- WEBRTC_TRACE(kTraceError, kTraceFile, _id,
- "unsupported file format %d!", fileFormat);
- break;
- }
- inStreamObj->CloseFile();
- delete inStreamObj;
- return time_in_ms;
-}
-
-WebRtc_UWord32 ModuleFileUtility::PlayoutPositionMs()
-{
- WEBRTC_TRACE(kTraceStream, kTraceFile, _id,
- "ModuleFileUtility::PlayoutPosition()");
-
- if(_reading)
- {
- return _playoutPositionMs;
- }
- else
- {
- return 0;
- }
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/media_file/source/media_file_utility.h b/trunk/src/modules/media_file/source/media_file_utility.h
deleted file mode 100644
index cfe0314..0000000
--- a/trunk/src/modules/media_file/source/media_file_utility.h
+++ /dev/null
@@ -1,349 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Note: the class cannot be used for reading and writing at the same time.
-#ifndef WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
-#define WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
-
-#include <stdio.h>
-
-#include "common_types.h"
-#include "media_file_defines.h"
-
-namespace webrtc {
-class AviFile;
-class InStream;
-class OutStream;
-
-class ModuleFileUtility
-{
-public:
-
- ModuleFileUtility(const WebRtc_Word32 id);
- ~ModuleFileUtility();
-
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
- // Open the file specified by fileName for reading (relative path is
- // allowed). If loop is true the file will be played until StopPlaying() is
- // called. When end of file is reached the file is read from the start.
- // Only video will be read if videoOnly is true.
- WebRtc_Word32 InitAviReading(const WebRtc_Word8* fileName, bool videoOnly,
- bool loop);
-
- // Put 10-60ms of audio data from file into the outBuffer depending on
- // codec frame size. bufferLengthInBytes indicates the size of outBuffer.
- // The return value is the number of bytes written to audioBuffer.
- // Note: This API only play mono audio but can be used on file containing
- // audio with more channels (in which case the audio will be coverted to
- // mono).
- WebRtc_Word32 ReadAviAudioData(WebRtc_Word8* outBuffer,
- const WebRtc_UWord32 bufferLengthInBytes);
-
- // Put one video frame into outBuffer. bufferLengthInBytes indicates the
- // size of outBuffer.
- // The return value is the number of bytes written to videoBuffer.
- WebRtc_Word32 ReadAviVideoData(WebRtc_Word8* videoBuffer,
- const WebRtc_UWord32 bufferLengthInBytes);
-
- // Open/create the file specified by fileName for writing audio/video data
- // (relative path is allowed). codecInst specifies the encoding of the audio
- // data. videoCodecInst specifies the encoding of the video data. Only video
- // data will be recorded if videoOnly is true.
- WebRtc_Word32 InitAviWriting(const WebRtc_Word8* filename,
- const CodecInst& codecInst,
- const VideoCodec& videoCodecInst,
- const bool videoOnly);
-
- // Write one audio frame, i.e. the bufferLengthinBytes first bytes of
- // audioBuffer, to file. The audio frame size is determined by the
- // codecInst.pacsize parameter of the last sucessfull
- // InitAviWriting(..) call.
- // Note: bufferLength must be exactly one frame.
- WebRtc_Word32 WriteAviAudioData(const WebRtc_Word8* audioBuffer,
- WebRtc_UWord32 bufferLengthInBytes);
-
-
- // Write one video frame, i.e. the bufferLength first bytes of videoBuffer,
- // to file.
- // Note: videoBuffer can contain encoded data. The codec used must be the
- // same as what was specified by videoCodecInst for the last successfull
- // InitAviWriting(..) call. The videoBuffer must contain exactly
- // one video frame.
- WebRtc_Word32 WriteAviVideoData(const WebRtc_Word8* videoBuffer,
- WebRtc_UWord32 bufferLengthInBytes);
-
- // Stop recording to file or stream.
- WebRtc_Word32 CloseAviFile();
-
- WebRtc_Word32 VideoCodecInst(VideoCodec& codecInst);
-#endif // #ifdef WEBRTC_MODULE_UTILITY_VIDEO
-
- // Prepare for playing audio from stream.
- // startPointMs and stopPointMs, unless zero, specify what part of the file
- // should be read. From startPointMs ms to stopPointMs ms.
- WebRtc_Word32 InitWavReading(InStream& stream,
- const WebRtc_UWord32 startPointMs = 0,
- const WebRtc_UWord32 stopPointMs = 0);
-
- // Put 10-60ms of audio data from stream into the audioBuffer depending on
- // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
- // The return value is the number of bytes written to audioBuffer.
- // Note: This API only play mono audio but can be used on file containing
- // audio with more channels (in which case the audio will be converted to
- // mono).
- WebRtc_Word32 ReadWavDataAsMono(InStream& stream, WebRtc_Word8* audioBuffer,
- const WebRtc_UWord32 dataLengthInBytes);
-
- // Put 10-60ms, depending on codec frame size, of audio data from file into
- // audioBufferLeft and audioBufferRight. The buffers contain the left and
- // right channel of played out stereo audio.
- // dataLengthInBytes indicates the size of both audioBufferLeft and
- // audioBufferRight.
- // The return value is the number of bytes read for each buffer.
- // Note: This API can only be successfully called for WAV files with stereo
- // audio.
- WebRtc_Word32 ReadWavDataAsStereo(InStream& wav,
- WebRtc_Word8* audioBufferLeft,
- WebRtc_Word8* audioBufferRight,
- const WebRtc_UWord32 bufferLength);
-
- // Prepare for recording audio to stream.
- // codecInst specifies the encoding of the audio data.
- // Note: codecInst.channels should be set to 2 for stereo (and 1 for
- // mono). Stereo is only supported for WAV files.
- WebRtc_Word32 InitWavWriting(OutStream& stream, const CodecInst& codecInst);
-
- // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
- // to file. The audio frame size is determined by the codecInst.pacsize
- // parameter of the last sucessfull StartRecordingAudioFile(..) call.
- // The return value is the number of bytes written to audioBuffer.
- WebRtc_Word32 WriteWavData(OutStream& stream,
- const WebRtc_Word8* audioBuffer,
- const WebRtc_UWord32 bufferLength);
-
- // Finalizes the WAV header so that it is correct if nothing more will be
- // written to stream.
- // Note: this API must be called before closing stream to ensure that the
- // WAVE header is updated with the file size. Don't call this API
- // if more samples are to be written to stream.
- WebRtc_Word32 UpdateWavHeader(OutStream& stream);
-
- // Prepare for playing audio from stream.
- // startPointMs and stopPointMs, unless zero, specify what part of the file
- // should be read. From startPointMs ms to stopPointMs ms.
- // freqInHz is the PCM sampling frequency.
- // NOTE, allowed frequencies are 8000, 16000 and 32000 (Hz)
- WebRtc_Word32 InitPCMReading(InStream& stream,
- const WebRtc_UWord32 startPointMs = 0,
- const WebRtc_UWord32 stopPointMs = 0,
- const WebRtc_UWord32 freqInHz = 16000);
-
- // Put 10-60ms of audio data from stream into the audioBuffer depending on
- // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
- // The return value is the number of bytes written to audioBuffer.
- WebRtc_Word32 ReadPCMData(InStream& stream, WebRtc_Word8* audioBuffer,
- const WebRtc_UWord32 dataLengthInBytes);
-
- // Prepare for recording audio to stream.
- // freqInHz is the PCM sampling frequency.
- // NOTE, allowed frequencies are 8000, 16000 and 32000 (Hz)
- WebRtc_Word32 InitPCMWriting(OutStream& stream,
- const WebRtc_UWord32 freqInHz = 16000);
-
- // Write one 10ms audio frame, i.e. the bufferLength first bytes of
- // audioBuffer, to file. The audio frame size is determined by the freqInHz
- // parameter of the last sucessfull InitPCMWriting(..) call.
- // The return value is the number of bytes written to audioBuffer.
- WebRtc_Word32 WritePCMData(OutStream& stream,
- const WebRtc_Word8* audioBuffer,
- WebRtc_UWord32 bufferLength);
-
- // Prepare for playing audio from stream.
- // startPointMs and stopPointMs, unless zero, specify what part of the file
- // should be read. From startPointMs ms to stopPointMs ms.
- WebRtc_Word32 InitCompressedReading(InStream& stream,
- const WebRtc_UWord32 startPointMs = 0,
- const WebRtc_UWord32 stopPointMs = 0);
-
- // Put 10-60ms of audio data from stream into the audioBuffer depending on
- // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
- // The return value is the number of bytes written to audioBuffer.
- WebRtc_Word32 ReadCompressedData(InStream& stream,
- WebRtc_Word8* audioBuffer,
- const WebRtc_UWord32 dataLengthInBytes);
-
- // Prepare for recording audio to stream.
- // codecInst specifies the encoding of the audio data.
- WebRtc_Word32 InitCompressedWriting(OutStream& stream,
- const CodecInst& codecInst);
-
- // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
- // to file. The audio frame size is determined by the codecInst.pacsize
- // parameter of the last sucessfull InitCompressedWriting(..) call.
- // The return value is the number of bytes written to stream.
- // Note: bufferLength must be exactly one frame.
- WebRtc_Word32 WriteCompressedData(OutStream& stream,
- const WebRtc_Word8* audioBuffer,
- const WebRtc_UWord32 bufferLength);
-
- // Prepare for playing audio from stream.
- // codecInst specifies the encoding of the audio data.
- WebRtc_Word32 InitPreEncodedReading(InStream& stream,
- const CodecInst& codecInst);
-
- // Put 10-60ms of audio data from stream into the audioBuffer depending on
- // codec frame size. dataLengthInBytes indicates the size of audioBuffer.
- // The return value is the number of bytes written to audioBuffer.
- WebRtc_Word32 ReadPreEncodedData(InStream& stream,
- WebRtc_Word8* audioBuffer,
- const WebRtc_UWord32 dataLengthInBytes);
-
- // Prepare for recording audio to stream.
- // codecInst specifies the encoding of the audio data.
- WebRtc_Word32 InitPreEncodedWriting(OutStream& stream,
- const CodecInst& codecInst);
-
- // Write one audio frame, i.e. the bufferLength first bytes of audioBuffer,
- // to stream. The audio frame size is determined by the codecInst.pacsize
- // parameter of the last sucessfull InitPreEncodedWriting(..) call.
- // The return value is the number of bytes written to stream.
- // Note: bufferLength must be exactly one frame.
- WebRtc_Word32 WritePreEncodedData(OutStream& stream,
- const WebRtc_Word8* inData,
- const WebRtc_UWord32 dataLengthInBytes);
-
- // Set durationMs to the size of the file (in ms) specified by fileName.
- // freqInHz specifies the sampling frequency of the file.
- WebRtc_Word32 FileDurationMs(const WebRtc_Word8* fileName,
- const FileFormats fileFormat,
- const WebRtc_UWord32 freqInHz = 16000);
-
- // Return the number of ms that have been played so far.
- WebRtc_UWord32 PlayoutPositionMs();
-
- // Update codecInst according to the current audio codec being used for
- // reading or writing.
- WebRtc_Word32 codec_info(CodecInst& codecInst);
-
-private:
- // Biggest WAV frame supported is 10 ms at 48kHz of 2 channel, 16 bit audio.
- enum{WAV_MAX_BUFFER_SIZE = 480*2*2};
-
-
- WebRtc_Word32 InitWavCodec(WebRtc_UWord32 samplesPerSec,
- WebRtc_UWord32 channels,
- WebRtc_UWord32 bitsPerSample,
- WebRtc_UWord32 formatTag);
-
- // Parse the WAV header in stream.
- WebRtc_Word32 ReadWavHeader(InStream& stream);
-
- // Update the WAV header. freqInHz, bytesPerSample, channels, format,
- // lengthInBytes specify characterists of the audio data.
- // freqInHz is the sampling frequency. bytesPerSample is the sample size in
- // bytes. channels is the number of channels, e.g. 1 is mono and 2 is
- // stereo. format is the encode format (e.g. PCMU, PCMA, PCM etc).
- // lengthInBytes is the number of bytes the audio samples are using up.
- WebRtc_Word32 WriteWavHeader(OutStream& stream,
- const WebRtc_UWord32 freqInHz,
- const WebRtc_UWord32 bytesPerSample,
- const WebRtc_UWord32 channels,
- const WebRtc_UWord32 format,
- const WebRtc_UWord32 lengthInBytes);
-
- // Put dataLengthInBytes of audio data from stream into the audioBuffer.
- // The return value is the number of bytes written to audioBuffer.
- WebRtc_Word32 ReadWavData(InStream& stream, WebRtc_UWord8* audioBuffer,
- const WebRtc_UWord32 dataLengthInBytes);
-
- // Update the current audio codec being used for reading or writing
- // according to codecInst.
- WebRtc_Word32 set_codec_info(const CodecInst& codecInst);
-
- struct WAVE_FMTINFO_header
- {
- WebRtc_Word16 formatTag;
- WebRtc_Word16 nChannels;
- WebRtc_Word32 nSamplesPerSec;
- WebRtc_Word32 nAvgBytesPerSec;
- WebRtc_Word16 nBlockAlign;
- WebRtc_Word16 nBitsPerSample;
- };
- // Identifiers for preencoded files.
- enum MediaFileUtility_CodecType
- {
- kCodecNoCodec = 0,
- kCodecIsac,
- kCodecIsacSwb,
- kCodecIsacLc,
- kCodecL16_8Khz,
- kCodecL16_16kHz,
- kCodecL16_32Khz,
- kCodecPcmu,
- kCodecPcma,
- kCodecIlbc20Ms,
- kCodecIlbc30Ms,
- kCodecG722,
- kCodecG722_1_32Kbps,
- kCodecG722_1_24Kbps,
- kCodecG722_1_16Kbps,
- kCodecG722_1c_48,
- kCodecG722_1c_32,
- kCodecG722_1c_24,
- kCodecAmr,
- kCodecAmrWb,
- kCodecG729,
- kCodecG729_1,
- kCodecG726_40,
- kCodecG726_32,
- kCodecG726_24,
- kCodecG726_16,
- kCodecSpeex8Khz,
- kCodecSpeex16Khz
- };
-
- // TODO (hellner): why store multiple formats. Just store either codec_info_
- // or _wavFormatObj and supply conversion functions.
- WAVE_FMTINFO_header _wavFormatObj;
- WebRtc_Word32 _dataSize; // Chunk size if reading a WAV file
- // Number of bytes to read. I.e. frame size in bytes. May be multiple
- // chunks if reading WAV.
- WebRtc_Word32 _readSizeBytes;
-
- WebRtc_Word32 _id;
-
- WebRtc_UWord32 _stopPointInMs;
- WebRtc_UWord32 _startPointInMs;
- WebRtc_UWord32 _playoutPositionMs;
- WebRtc_UWord32 _bytesWritten;
-
- CodecInst codec_info_;
- MediaFileUtility_CodecType _codecId;
-
- // The amount of bytes, on average, used for one audio sample.
- WebRtc_Word32 _bytesPerSample;
- WebRtc_Word32 _readPos;
-
- // Only reading or writing can be enabled, not both.
- bool _reading;
- bool _writing;
-
- // Scratch buffer used for turning stereo audio to mono.
- WebRtc_UWord8 _tempData[WAV_MAX_BUFFER_SIZE];
-
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
- AviFile* _aviAudioInFile;
- AviFile* _aviVideoInFile;
- AviFile* _aviOutFile;
- VideoCodec _videoCodec;
-#endif
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_MEDIA_FILE_SOURCE_MEDIA_FILE_UTILITY_H_
diff --git a/trunk/src/modules/modules.gyp b/trunk/src/modules/modules.gyp
deleted file mode 100644
index 83b5cc6..0000000
--- a/trunk/src/modules/modules.gyp
+++ /dev/null
@@ -1,60 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'includes': [
- '../build/common.gypi',
- 'audio_coding/codecs/cng/cng.gypi',
- 'audio_coding/codecs/g711/g711.gypi',
- 'audio_coding/codecs/g722/g722.gypi',
- 'audio_coding/codecs/ilbc/ilbc.gypi',
- 'audio_coding/codecs/iSAC/main/source/isac.gypi',
- 'audio_coding/codecs/iSAC/fix/source/isacfix.gypi',
- 'audio_coding/codecs/pcm16b/pcm16b.gypi',
- 'audio_coding/main/source/audio_coding_module.gypi',
- 'audio_coding/neteq/neteq.gypi',
- 'audio_conference_mixer/source/audio_conference_mixer.gypi',
- 'audio_device/main/source/audio_device.gypi',
- 'audio_processing/audio_processing.gypi',
- 'audio_processing/aec/aec.gypi',
- 'audio_processing/aecm/aecm.gypi',
- 'audio_processing/agc/agc.gypi',
- 'audio_processing/ns/ns.gypi',
- 'audio_processing/utility/util.gypi',
- 'media_file/source/media_file.gypi',
- 'udp_transport/source/udp_transport.gypi',
- 'utility/source/utility.gypi',
- 'video_coding/codecs/i420/main/source/i420.gypi',
- 'video_coding/codecs/test_framework/test_framework.gypi',
- 'video_coding/codecs/vp8/main/source/vp8.gypi',
- 'video_coding/main/source/video_coding.gypi',
- 'video_capture/main/source/video_capture.gypi',
- 'video_processing/main/source/video_processing.gypi',
- 'video_render/main/source/video_render.gypi',
- 'rtp_rtcp/source/rtp_rtcp.gypi',
- ],
-
- # Test targets, excluded when building with Chromium.
- 'conditions': [
- ['build_with_chromium==0', {
- 'includes': [
- 'audio_coding/codecs/iSAC/isac_test.gypi',
- 'audio_coding/codecs/iSAC/isacfix_test.gypi',
- 'audio_processing/apm_tests.gypi',
- 'rtp_rtcp/source/rtp_rtcp_tests.gypi',
- 'rtp_rtcp/test/test_bwe/test_bwe.gypi',
- 'rtp_rtcp/test/testFec/test_fec.gypi',
- 'rtp_rtcp/test/testAPI/test_api.gypi',
- 'video_coding/main/source/video_coding_test.gypi',
- 'video_coding/codecs/test/video_codecs_test_framework.gypi',
- 'video_coding/codecs/tools/video_codecs_tools.gypi',
- 'video_processing/main/test/vpm_tests.gypi',
- ], # includes
- }], # build_with_chromium
- ], # conditions
-}
diff --git a/trunk/src/modules/rtp_rtcp/OWNERS b/trunk/src/modules/rtp_rtcp/OWNERS
deleted file mode 100644
index c968564..0000000
--- a/trunk/src/modules/rtp_rtcp/OWNERS
+++ /dev/null
@@ -1,5 +0,0 @@
-pwestin@webrtc.org
-stefan@webrtc.org
-henrik.lundin@webrtc.org
-mflodman@webrtc.org
-asapersson@webrtc.org
\ No newline at end of file
diff --git a/trunk/src/modules/rtp_rtcp/interface/rtp_rtcp.h b/trunk/src/modules/rtp_rtcp/interface/rtp_rtcp.h
deleted file mode 100644
index 6b55202..0000000
--- a/trunk/src/modules/rtp_rtcp/interface/rtp_rtcp.h
+++ /dev/null
@@ -1,1094 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
-
-#include <vector>
-
-#include "module.h"
-#include "rtp_rtcp_defines.h"
-
-namespace webrtc {
-// forward declaration
-class Transport;
-
-class RtpRtcp : public Module
-{
-public:
- /*
- * create a RTP/RTCP module object using the system clock
- *
- * id - unique identifier of this RTP/RTCP module object
- * audio - true for a audio version of the RTP/RTCP module object false will create a video version
- */
- static RtpRtcp* CreateRtpRtcp(const WebRtc_Word32 id,
- const bool audio);
-
- /*
- * create a RTP/RTCP module object
- *
- * id - unique identifier of this RTP/RTCP module object
- * audio - true for a audio version of the RTP/RTCP module object
- * false will create a video version
- * clock - the clock to use to read time; must not be NULL
- */
- static RtpRtcp* CreateRtpRtcp(const WebRtc_Word32 id,
- const bool audio,
- RtpRtcpClock* clock);
-
- /*
- * destroy a RTP/RTCP module object
- *
- * module - object to destroy
- */
- static void DestroyRtpRtcp(RtpRtcp* module);
-
- /*
- * Change the unique identifier of this object
- *
- * id - new unique identifier of this RTP/RTCP module object
- */
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0;
-
- /*
- * De-muxing functionality for conferencing
- *
- * register a module that will act as a default module for this module
- * used for feedback messages back to the encoder when one encoded stream
- * is sent to multiple destinations
- *
- * module - default module
- */
- virtual WebRtc_Word32 RegisterDefaultModule(RtpRtcp* module) = 0;
-
- /*
- * unregister the default module
- * will stop the demuxing feedback
- */
- virtual WebRtc_Word32 DeRegisterDefaultModule() = 0;
-
- /*
- * returns true if a default module is registered, false otherwise
- */
- virtual bool DefaultModuleRegistered() = 0;
-
- /*
- * returns number of registered child modules
- */
- virtual WebRtc_UWord32 NumberChildModules() = 0;
-
- /*
- * Lip-sync between voice-video
- *
- * module - audio module
- *
- * Note: only allowed on a video module
- */
- virtual WebRtc_Word32 RegisterSyncModule(RtpRtcp* module) = 0;
-
- /*
- * Turn off lip-sync between voice-video
- */
- virtual WebRtc_Word32 DeRegisterSyncModule() = 0;
-
- /**************************************************************************
- *
- * Receiver functions
- *
- ***************************************************************************/
-
- /*
- * Initialize receive side
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 InitReceiver() = 0;
-
- /*
- * Used by the module to deliver the incoming data to the codec module
- *
- * incomingDataCallback - callback object that will receive the incoming data
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RegisterIncomingDataCallback(RtpData* incomingDataCallback) = 0;
-
- /*
- * Used by the module to deliver messages to the codec module/appliation
- *
- * incomingMessagesCallback - callback object that will receive the incoming messages
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RegisterIncomingRTPCallback(RtpFeedback* incomingMessagesCallback) = 0;
-
- /*
- * configure a RTP packet timeout value
- *
- * RTPtimeoutMS - time in milliseconds after last received RTP packet
- * RTCPtimeoutMS - time in milliseconds after last received RTCP packet
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 RTPtimeoutMS,
- const WebRtc_UWord32 RTCPtimeoutMS) = 0;
-
- /*
- * Set periodic dead or alive notification
- *
- * enable - turn periodic dead or alive notification on/off
- * sampleTimeSeconds - sample interval in seconds for dead or alive notifications
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetPeriodicDeadOrAliveStatus(const bool enable,
- const WebRtc_UWord8 sampleTimeSeconds) = 0;
-
- /*
- * Get periodic dead or alive notification status
- *
- * enable - periodic dead or alive notification on/off
- * sampleTimeSeconds - sample interval in seconds for dead or alive notifications
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 PeriodicDeadOrAliveStatus(bool &enable,
- WebRtc_UWord8 &sampleTimeSeconds) = 0;
-
- /*
- * set voice codec name and payload type
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RegisterReceivePayload(
- const CodecInst& voiceCodec) = 0;
-
- /*
- * set video codec name and payload type
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RegisterReceivePayload(
- const VideoCodec& videoCodec) = 0;
-
- /*
- * get payload type for a voice codec
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 ReceivePayloadType(
- const CodecInst& voiceCodec,
- WebRtc_Word8* plType) = 0;
-
- /*
- * get payload type for a video codec
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 ReceivePayloadType(
- const VideoCodec& videoCodec,
- WebRtc_Word8* plType) = 0;
-
- /*
- * Remove a registered payload type from list of accepted payloads
- *
- * payloadType - payload type of codec
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 DeRegisterReceivePayload(
- const WebRtc_Word8 payloadType) = 0;
-
- /*
- * (De)register RTP header extension type and id.
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RegisterReceiveRtpHeaderExtension(
- const RTPExtensionType type,
- const WebRtc_UWord8 id) = 0;
-
- virtual WebRtc_Word32 DeregisterReceiveRtpHeaderExtension(
- const RTPExtensionType type) = 0;
-
- /*
- * Get last received remote timestamp
- */
- virtual WebRtc_UWord32 RemoteTimestamp() const = 0;
-
- /*
- * Get the current estimated remote timestamp
- *
- * timestamp - estimated timestamp
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 EstimatedRemoteTimeStamp(WebRtc_UWord32& timestamp) const = 0;
-
- /*
- * Get incoming SSRC
- */
- virtual WebRtc_UWord32 RemoteSSRC() const = 0;
-
- /*
- * Get remote CSRC
- *
- * arrOfCSRC - array that will receive the CSRCs
- *
- * return -1 on failure else the number of valid entries in the list
- */
- virtual WebRtc_Word32 RemoteCSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const = 0;
-
- /*
- * get the currently configured SSRC filter
- *
- * allowedSSRC - SSRC that will be allowed through
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowedSSRC) const = 0;
-
- /*
- * set a SSRC to be used as a filter for incoming RTP streams
- *
- * allowedSSRC - SSRC that will be allowed through
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetSSRCFilter(const bool enable,
- const WebRtc_UWord32 allowedSSRC) = 0;
-
- /*
- * Turn on/off receiving RTX (RFC 4588) on a specific SSRC.
- */
- virtual WebRtc_Word32 SetRTXReceiveStatus(const bool enable,
- const WebRtc_UWord32 SSRC) = 0;
-
- /*
- * Get status of receiving RTX (RFC 4588) on a specific SSRC.
- */
- virtual WebRtc_Word32 RTXReceiveStatus(bool* enable,
- WebRtc_UWord32* SSRC) const = 0;
-
- /*
- * called by the network module when we receive a packet
- *
- * incomingPacket - incoming packet buffer
- * packetLength - length of incoming buffer
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incomingPacket,
- const WebRtc_UWord16 packetLength) = 0;
-
-
- /*
- * Option when not using the RegisterSyncModule function
- *
- * Inform the module about the received audion NTP
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 IncomingAudioNTP(
- const WebRtc_UWord32 audioReceivedNTPsecs,
- const WebRtc_UWord32 audioReceivedNTPfrac,
- const WebRtc_UWord32 audioRTCPArrivalTimeSecs,
- const WebRtc_UWord32 audioRTCPArrivalTimeFrac) = 0;
-
- /**************************************************************************
- *
- * Sender
- *
- ***************************************************************************/
-
- /*
- * Initialize send side
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 InitSender() = 0;
-
- /*
- * Used by the module to send RTP and RTCP packet to the network module
- *
- * outgoingTransport - transport object that will be called when packets are ready to be sent out on the network
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RegisterSendTransport(Transport* outgoingTransport) = 0;
-
- /*
- * set MTU
- *
- * size - Max transfer unit in bytes, default is 1500
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetMaxTransferUnit(const WebRtc_UWord16 size) = 0;
-
- /*
- * set transtport overhead
- * default is IPv4 and UDP with no encryption
- *
- * TCP - true for TCP false UDP
- * IPv6 - true for IP version 6 false for version 4
- * authenticationOverhead - number of bytes to leave for an authentication header
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetTransportOverhead(const bool TCP,
- const bool IPV6,
- const WebRtc_UWord8 authenticationOverhead = 0) = 0;
-
- /*
- * Get max payload length
- *
- * A combination of the configuration MaxTransferUnit and TransportOverhead.
- * Does not account FEC/ULP/RED overhead if FEC is enabled.
- * Does not account for RTP headers
- */
- virtual WebRtc_UWord16 MaxPayloadLength() const = 0;
-
- /*
- * Get max data payload length
- *
- * A combination of the configuration MaxTransferUnit, headers and TransportOverhead.
- * Takes into account FEC/ULP/RED overhead if FEC is enabled.
- * Takes into account RTP headers
- */
- virtual WebRtc_UWord16 MaxDataPayloadLength() const = 0;
-
- /*
- * set RTPKeepaliveStatus
- *
- * enable - on/off
- * unknownPayloadType - payload type to use for RTP keepalive
- * deltaTransmitTimeMS - delta time between RTP keepalive packets
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetRTPKeepaliveStatus(
- const bool enable,
- const WebRtc_Word8 unknownPayloadType,
- const WebRtc_UWord16 deltaTransmitTimeMS) = 0;
-
- /*
- * Get RTPKeepaliveStatus
- *
- * enable - on/off
- * unknownPayloadType - payload type in use for RTP keepalive
- * deltaTransmitTimeMS - delta time between RTP keepalive packets
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RTPKeepaliveStatus(
- bool* enable,
- WebRtc_Word8* unknownPayloadType,
- WebRtc_UWord16* deltaTransmitTimeMS) const = 0;
-
- /*
- * check if RTPKeepaliveStatus is enabled
- */
- virtual bool RTPKeepalive() const = 0;
-
- /*
- * set codec name and payload type
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RegisterSendPayload(
- const CodecInst& voiceCodec) = 0;
-
- /*
- * set codec name and payload type
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RegisterSendPayload(
- const VideoCodec& videoCodec) = 0;
-
- /*
- * Unregister a send payload
- *
- * payloadType - payload type of codec
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 DeRegisterSendPayload(
- const WebRtc_Word8 payloadType) = 0;
-
- /*
- * (De)register RTP header extension type and id.
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RegisterSendRtpHeaderExtension(
- const RTPExtensionType type,
- const WebRtc_UWord8 id) = 0;
-
- virtual WebRtc_Word32 DeregisterSendRtpHeaderExtension(
- const RTPExtensionType type) = 0;
-
- /*
- * Enable/disable traffic smoothing of sending stream.
- */
- virtual void SetTransmissionSmoothingStatus(const bool enable) = 0;
-
- virtual bool TransmissionSmoothingStatus() const = 0;
-
- /*
- * get start timestamp
- */
- virtual WebRtc_UWord32 StartTimestamp() const = 0;
-
- /*
- * configure start timestamp, default is a random number
- *
- * timestamp - start timestamp
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetStartTimestamp(
- const WebRtc_UWord32 timestamp) = 0;
-
- /*
- * Get SequenceNumber
- */
- virtual WebRtc_UWord16 SequenceNumber() const = 0;
-
- /*
- * Set SequenceNumber, default is a random number
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetSequenceNumber(const WebRtc_UWord16 seq) = 0;
-
- /*
- * Get SSRC
- */
- virtual WebRtc_UWord32 SSRC() const = 0;
-
- /*
- * configure SSRC, default is a random number
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetSSRC(const WebRtc_UWord32 ssrc) = 0;
-
- /*
- * Get CSRC
- *
- * arrOfCSRC - array of CSRCs
- *
- * return -1 on failure else number of valid entries in the array
- */
- virtual WebRtc_Word32 CSRCs(
- WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const = 0;
-
- /*
- * Set CSRC
- *
- * arrOfCSRC - array of CSRCs
- * arrLength - number of valid entries in the array
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetCSRCs(
- const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
- const WebRtc_UWord8 arrLength) = 0;
-
- /*
- * includes CSRCs in RTP header if enabled
- *
- * include CSRC - on/off
- *
- * default:on
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetCSRCStatus(const bool include) = 0;
-
- /*
- * Turn on/off sending RTX (RFC 4588) on a specific SSRC.
- */
- virtual WebRtc_Word32 SetRTXSendStatus(const bool enable,
- const bool setSSRC,
- const WebRtc_UWord32 SSRC) = 0;
-
-
- /*
- * Get status of sending RTX (RFC 4588) on a specific SSRC.
- */
- virtual WebRtc_Word32 RTXSendStatus(bool* enable,
- WebRtc_UWord32* SSRC) const = 0;
-
- /*
- * sends kRtcpByeCode when going from true to false
- *
- * sending - on/off
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetSendingStatus(const bool sending) = 0;
-
- /*
- * get send status
- */
- virtual bool Sending() const = 0;
-
- /*
- * Starts/Stops media packets, on by default
- *
- * sending - on/off
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetSendingMediaStatus(const bool sending) = 0;
-
- /*
- * get send status
- */
- virtual bool SendingMedia() const = 0;
-
- /*
- * get sent bitrate in Kbit/s
- */
- virtual void BitrateSent(WebRtc_UWord32* totalRate,
- WebRtc_UWord32* videoRate,
- WebRtc_UWord32* fecRate,
- WebRtc_UWord32* nackRate) const = 0;
-
- /*
- * Get the send-side estimate of the available bandwidth.
- */
- virtual int EstimatedSendBandwidth(
- WebRtc_UWord32* available_bandwidth) const = 0;
-
- /*
- * Get the receive-side estimate of the available bandwidth.
- */
- virtual int EstimatedReceiveBandwidth(
- WebRtc_UWord32* available_bandwidth) const = 0;
-
- /*
- * Used by the codec module to deliver a video or audio frame for packetization
- *
- * frameType - type of frame to send
- * payloadType - payload type of frame to send
- * timestamp - timestamp of frame to send
- * payloadData - payload buffer of frame to send
- * payloadSize - size of payload buffer to send
- * fragmentation - fragmentation offset data for fragmented frames such as layers or RED
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SendOutgoingData(
- const FrameType frameType,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader* fragmentation = NULL,
- const RTPVideoHeader* rtpVideoHdr = NULL) = 0;
-
- /**************************************************************************
- *
- * RTCP
- *
- ***************************************************************************/
-
- /*
- * RegisterIncomingRTCPCallback
- *
- * incomingMessagesCallback - callback object that will receive messages from RTCP
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RegisterIncomingRTCPCallback(RtcpFeedback* incomingMessagesCallback) = 0;
-
- /*
- * Get RTCP status
- */
- virtual RTCPMethod RTCP() const = 0;
-
- /*
- * configure RTCP status i.e on(compound or non- compound)/off
- *
- * method - RTCP method to use
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetRTCPStatus(const RTCPMethod method) = 0;
-
- /*
- * Set RTCP CName (i.e unique identifier)
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetCNAME(const char cName[RTCP_CNAME_SIZE]) = 0;
-
- /*
- * Get RTCP CName (i.e unique identifier)
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 CNAME(char cName[RTCP_CNAME_SIZE]) = 0;
-
- /*
- * Get remote CName
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RemoteCNAME(
- const WebRtc_UWord32 remoteSSRC,
- char cName[RTCP_CNAME_SIZE]) const = 0;
-
- /*
- * Get remote NTP
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RemoteNTP(
- WebRtc_UWord32 *ReceivedNTPsecs,
- WebRtc_UWord32 *ReceivedNTPfrac,
- WebRtc_UWord32 *RTCPArrivalTimeSecs,
- WebRtc_UWord32 *RTCPArrivalTimeFrac) const = 0;
-
- /*
- * AddMixedCNAME
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 AddMixedCNAME(
- const WebRtc_UWord32 SSRC,
- const char cName[RTCP_CNAME_SIZE]) = 0;
-
- /*
- * RemoveMixedCNAME
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 SSRC) = 0;
-
- /*
- * Get RoundTripTime
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RTT(const WebRtc_UWord32 remoteSSRC,
- WebRtc_UWord16* RTT,
- WebRtc_UWord16* avgRTT,
- WebRtc_UWord16* minRTT,
- WebRtc_UWord16* maxRTT) const = 0 ;
-
- /*
- * Reset RoundTripTime statistics
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 ResetRTT(const WebRtc_UWord32 remoteSSRC)= 0 ;
-
- /*
- * Force a send of a RTCP packet
- * normal SR and RR are triggered via the process function
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SendRTCP(WebRtc_UWord32 rtcpPacketType = kRtcpReport) = 0;
-
- /*
- * Good state of RTP receiver inform sender
- */
- virtual WebRtc_Word32 SendRTCPReferencePictureSelection(const WebRtc_UWord64 pictureID) = 0;
-
- /*
- * Send a RTCP Slice Loss Indication (SLI)
- * 6 least significant bits of pictureID
- */
- virtual WebRtc_Word32 SendRTCPSliceLossIndication(const WebRtc_UWord8 pictureID) = 0;
-
- /*
- * Reset RTP statistics
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 ResetStatisticsRTP() = 0;
-
- /*
- * statistics of our localy created statistics of the received RTP stream
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 StatisticsRTP(WebRtc_UWord8 *fraction_lost, // scale 0 to 255
- WebRtc_UWord32 *cum_lost, // number of lost packets
- WebRtc_UWord32 *ext_max, // highest sequence number received
- WebRtc_UWord32 *jitter,
- WebRtc_UWord32 *max_jitter = NULL) const = 0;
-
- /*
- * Reset RTP data counters for the receiving side
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 ResetReceiveDataCountersRTP() = 0;
-
- /*
- * Reset RTP data counters for the sending side
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 ResetSendDataCountersRTP() = 0;
-
- /*
- * statistics of the amount of data sent and received
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 DataCountersRTP(WebRtc_UWord32 *bytesSent,
- WebRtc_UWord32 *packetsSent,
- WebRtc_UWord32 *bytesReceived,
- WebRtc_UWord32 *packetsReceived) const = 0;
- /*
- * Get received RTCP sender info
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RemoteRTCPStat(RTCPSenderInfo* senderInfo) = 0;
-
- /*
- * Get received RTCP report block
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RemoteRTCPStat(
- std::vector<RTCPReportBlock>* receiveBlocks) const = 0;
- /*
- * Set received RTCP report block
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 AddRTCPReportBlock(
- const WebRtc_UWord32 SSRC,
- const RTCPReportBlock* receiveBlock) = 0;
-
- /*
- * RemoveRTCPReportBlock
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RemoveRTCPReportBlock(const WebRtc_UWord32 SSRC) = 0;
-
- /*
- * (APP) Application specific data
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetRTCPApplicationSpecificData(const WebRtc_UWord8 subType,
- const WebRtc_UWord32 name,
- const WebRtc_UWord8* data,
- const WebRtc_UWord16 length) = 0;
- /*
- * (XR) VOIP metric
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric) = 0;
-
- /*
- * (REMB) Receiver Estimated Max Bitrate
- */
- virtual bool REMB() const = 0;
-
- virtual WebRtc_Word32 SetREMBStatus(const bool enable) = 0;
-
- virtual WebRtc_Word32 SetREMBData(const WebRtc_UWord32 bitrate,
- const WebRtc_UWord8 numberOfSSRC,
- const WebRtc_UWord32* SSRC) = 0;
-
- // Used to set maximum bitrate estimate received in a REMB packet.
- virtual WebRtc_Word32 SetMaximumBitrateEstimate(
- const WebRtc_UWord32 bitrate) = 0;
-
- // Registers an observer to call when the estimate of the incoming channel
- // changes.
- virtual bool SetRemoteBitrateObserver(
- RtpRemoteBitrateObserver* observer) = 0;
-
- /*
- * (IJ) Extended jitter report.
- */
- virtual bool IJ() const = 0;
-
- virtual WebRtc_Word32 SetIJStatus(const bool enable) = 0;
-
- /*
- * (TMMBR) Temporary Max Media Bit Rate
- */
- virtual bool TMMBR() const = 0;
-
- /*
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetTMMBRStatus(const bool enable) = 0;
-
- /*
- * local bw estimation changed
- *
- * for video called by internal estimator
- * for audio (iSAC) called by engine, geting the data from the decoder
- */
- virtual void OnBandwidthEstimateUpdate(WebRtc_UWord16 bandWidthKbit) = 0;
-
- /*
- * (NACK)
- */
- virtual NACKMethod NACK() const = 0;
-
- /*
- * Turn negative acknowledgement requests on/off
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method) = 0;
-
- /*
- * TODO(holmer): Propagate this API to VideoEngine.
- * Returns the currently configured selective retransmission settings.
- */
- virtual int SelectiveRetransmissions() const = 0;
-
- /*
- * TODO(holmer): Propagate this API to VideoEngine.
- * Sets the selective retransmission settings, which will decide which
- * packets will be retransmitted if NACKed. Settings are constructed by
- * combining the constants in enum RetransmissionMode with bitwise OR.
- * All packets are retransmitted if kRetransmitAllPackets is set, while no
- * packets are retransmitted if kRetransmitOff is set.
- * By default all packets except FEC packets are retransmitted. For VP8
- * with temporal scalability only base layer packets are retransmitted.
- *
- * Returns -1 on failure, otherwise 0.
- */
- virtual int SetSelectiveRetransmissions(uint8_t settings) = 0;
-
- /*
- * Send a Negative acknowledgement packet
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SendNACK(const WebRtc_UWord16* nackList,
- const WebRtc_UWord16 size) = 0;
-
- /*
- * Store the sent packets, needed to answer to a Negative acknowledgement requests
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetStorePacketsStatus(const bool enable, const WebRtc_UWord16 numberToStore = 200) = 0;
-
- /**************************************************************************
- *
- * Audio
- *
- ***************************************************************************/
-
- /*
- * RegisterAudioCallback
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RegisterAudioCallback(RtpAudioFeedback* messagesCallback) = 0;
-
- /*
- * set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples) = 0;
-
- /*
- * Outband TelephoneEvent(DTMF) detection
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetTelephoneEventStatus(const bool enable,
- const bool forwardToDecoder,
- const bool detectEndOfTone = false) = 0;
-
- /*
- * Is outband TelephoneEvent(DTMF) turned on/off?
- */
- virtual bool TelephoneEvent() const = 0;
-
- /*
- * Returns true if received DTMF events are forwarded to the decoder using
- * the OnPlayTelephoneEvent callback.
- */
- virtual bool TelephoneEventForwardToDecoder() const = 0;
-
- /*
- * SendTelephoneEventActive
- *
- * return true if we currently send a telephone event and 100 ms after an event is sent
- * used to prevent teh telephone event tone to be recorded by the microphone and send inband
- * just after the tone has ended
- */
- virtual bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const = 0;
-
- /*
- * Send a TelephoneEvent tone using RFC 2833 (4733)
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SendTelephoneEventOutband(const WebRtc_UWord8 key,
- const WebRtc_UWord16 time_ms,
- const WebRtc_UWord8 level) = 0;
-
- /*
- * Set payload type for Redundant Audio Data RFC 2198
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetSendREDPayloadType(const WebRtc_Word8 payloadType) = 0;
-
- /*
- * Get payload type for Redundant Audio Data RFC 2198
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SendREDPayloadType(WebRtc_Word8& payloadType) const = 0;
-
- /*
- * Set status and ID for header-extension-for-audio-level-indication.
- * See https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
- * for more details.
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetRTPAudioLevelIndicationStatus(const bool enable,
- const WebRtc_UWord8 ID) = 0;
-
- /*
- * Get status and ID for header-extension-for-audio-level-indication.
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 GetRTPAudioLevelIndicationStatus(bool& enable,
- WebRtc_UWord8& ID) const = 0;
-
- /*
- * Store the audio level in dBov for header-extension-for-audio-level-indication.
- * This API shall be called before transmision of an RTP packet to ensure
- * that the |level| part of the extended RTP header is updated.
- *
- * return -1 on failure else 0.
- */
- virtual WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov) = 0;
-
- /**************************************************************************
- *
- * Video
- *
- ***************************************************************************/
-
- /*
- * Register a callback object that will receive callbacks for video related events
- * such as an incoming key frame request.
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RegisterIncomingVideoCallback(RtpVideoFeedback* incomingMessagesCallback) = 0;
-
- /*
- * Set the estimated camera delay in MS
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delayMS) = 0;
-
- /*
- * Set the start and max send bitrate
- * used by the bandwidth management
- *
- * Not calling this or setting startBitrateKbit to 0 disables the bandwidth management
- *
- * minBitrateKbit = 0 equals no min bitrate
- * maxBitrateKbit = 0 equals no max bitrate
- *
- * return -1 on failure else 0
- */
- virtual void SetSendBitrate(const WebRtc_UWord32 startBitrate,
- const WebRtc_UWord16 minBitrateKbit,
- const WebRtc_UWord16 maxBitrateKbit) = 0;
-
- /*
- * Turn on/off generic FEC
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetGenericFECStatus(const bool enable,
- const WebRtc_UWord8 payloadTypeRED,
- const WebRtc_UWord8 payloadTypeFEC) = 0;
-
- /*
- * Get generic FEC setting
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 GenericFECStatus(bool& enable,
- WebRtc_UWord8& payloadTypeRED,
- WebRtc_UWord8& payloadTypeFEC) = 0;
-
-
- /*
- * Set FEC code rate of key and delta frames
- * codeRate on a scale of 0 to 255 where 255 is 100% added packets, hence protect up to 50% packet loss
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetFECCodeRate(const WebRtc_UWord8 keyFrameCodeRate,
- const WebRtc_UWord8 deltaFrameCodeRate) = 0;
-
-
- /*
- * Set FEC unequal protection (UEP) across packets,
- * for key and delta frames.
- *
- * If keyUseUepProtection is true UEP is enabled for key frames.
- * If deltaUseUepProtection is true UEP is enabled for delta frames.
- *
- * UEP skews the FEC protection towards being spent more on the
- * important packets, at the cost of less FEC protection for the
- * non-important packets.
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetFECUepProtection(const bool keyUseUepProtection,
- const bool deltaUseUepProtection) = 0;
-
-
- /*
- * Set method for requestion a new key frame
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 SetKeyFrameRequestMethod(const KeyFrameRequestMethod method) = 0;
-
- /*
- * send a request for a keyframe
- *
- * return -1 on failure else 0
- */
- virtual WebRtc_Word32 RequestKeyFrame() = 0;
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_H_
diff --git a/trunk/src/modules/rtp_rtcp/interface/rtp_rtcp_defines.h b/trunk/src/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
deleted file mode 100644
index b6a8a54..0000000
--- a/trunk/src/modules/rtp_rtcp/interface/rtp_rtcp_defines.h
+++ /dev/null
@@ -1,282 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
-#define WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
-
-#include "typedefs.h"
-#include "module_common_types.h"
-
-#ifndef NULL
- #define NULL 0
-#endif
-
-#define RTCP_CNAME_SIZE 256 // RFC 3550 page 44, including null termination
-#define IP_PACKET_SIZE 1500 // we assume ethernet
-#define MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS 10
-#define TIMEOUT_SEI_MESSAGES_MS 30000 // in milliseconds
-
-namespace webrtc{
-enum RTCPMethod
-{
- kRtcpOff = 0,
- kRtcpCompound = 1,
- kRtcpNonCompound = 2
-};
-
-enum RTPAliveType
-{
- kRtpDead = 0,
- kRtpNoRtp = 1,
- kRtpAlive = 2
-};
-
-enum StorageType {
- kDontStore,
- kDontRetransmit,
- kAllowRetransmission
-};
-
-enum RTPExtensionType
-{
- kRtpExtensionNone,
- kRtpExtensionTransmissionTimeOffset,
- kRtpExtensionAudioLevel,
-};
-
-enum RTCPAppSubTypes
-{
- kAppSubtypeBwe = 0x00
-};
-
-enum RTCPPacketType
-{
- kRtcpReport = 0x0001,
- kRtcpSr = 0x0002,
- kRtcpRr = 0x0004,
- kRtcpBye = 0x0008,
- kRtcpPli = 0x0010,
- kRtcpNack = 0x0020,
- kRtcpFir = 0x0040,
- kRtcpTmmbr = 0x0080,
- kRtcpTmmbn = 0x0100,
- kRtcpSrReq = 0x0200,
- kRtcpXrVoipMetric = 0x0400,
- kRtcpApp = 0x0800,
- kRtcpSli = 0x4000,
- kRtcpRpsi = 0x8000,
- kRtcpRemb = 0x10000,
- kRtcpTransmissionTimeOffset = 0x20000
-};
-
-enum KeyFrameRequestMethod
-{
- kKeyFrameReqFirRtp = 1,
- kKeyFrameReqPliRtcp = 2,
- kKeyFrameReqFirRtcp = 3
-};
-
-enum RtpRtcpPacketType
-{
- kPacketRtp = 0,
- kPacketKeepAlive = 1
-};
-
-enum NACKMethod
-{
- kNackOff = 0,
- kNackRtcp = 2
-};
-
-enum RetransmissionMode {
- kRetransmitOff = 0x0,
- kRetransmitFECPackets = 0x1,
- kRetransmitBaseLayer = 0x2,
- kRetransmitHigherLayers = 0x4,
- kRetransmitAllPackets = 0xFF
-};
-
-struct RTCPSenderInfo
-{
- WebRtc_UWord32 NTPseconds;
- WebRtc_UWord32 NTPfraction;
- WebRtc_UWord32 RTPtimeStamp;
- WebRtc_UWord32 sendPacketCount;
- WebRtc_UWord32 sendOctetCount;
-};
-
-struct RTCPReportBlock
-{
- // Fields as described by RFC 3550 6.4.2.
- WebRtc_UWord32 remoteSSRC; // SSRC of sender of this report.
- WebRtc_UWord32 sourceSSRC; // SSRC of the RTP packet sender.
- WebRtc_UWord8 fractionLost;
- WebRtc_UWord32 cumulativeLost; // 24 bits valid
- WebRtc_UWord32 extendedHighSeqNum;
- WebRtc_UWord32 jitter;
- WebRtc_UWord32 lastSR;
- WebRtc_UWord32 delaySinceLastSR;
-};
-
-class RtpData
-{
-public:
- virtual WebRtc_Word32 OnReceivedPayloadData(
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const WebRtcRTPHeader* rtpHeader) = 0;
-protected:
- virtual ~RtpData() {}
-};
-
-class RtcpFeedback
-{
-public:
- // if audioVideoOffset > 0 video is behind audio
- virtual void OnLipSyncUpdate(const WebRtc_Word32 /*id*/,
- const WebRtc_Word32 /*audioVideoOffset*/) {};
-
- virtual void OnApplicationDataReceived(const WebRtc_Word32 /*id*/,
- const WebRtc_UWord8 /*subType*/,
- const WebRtc_UWord32 /*name*/,
- const WebRtc_UWord16 /*length*/,
- const WebRtc_UWord8* /*data*/) {};
-
- virtual void OnXRVoIPMetricReceived(
- const WebRtc_Word32 /*id*/,
- const RTCPVoIPMetric* /*metric*/,
- const WebRtc_Word8 /*VoIPmetricBuffer*/[28]) {};
-
- virtual void OnRTCPPacketTimeout(const WebRtc_Word32 /*id*/) {};
-
- virtual void OnTMMBRReceived(const WebRtc_Word32 /*id*/,
- const WebRtc_UWord16 /*bwEstimateKbit*/) {};
-
- virtual void OnSLIReceived(const WebRtc_Word32 /*id*/,
- const WebRtc_UWord8 /*pictureId*/) {};
-
- virtual void OnRPSIReceived(const WebRtc_Word32 /*id*/,
- const WebRtc_UWord64 /*pictureId*/) {};
-
- virtual void OnReceiverEstimatedMaxBitrateReceived(
- const WebRtc_Word32 /*id*/,
- const WebRtc_UWord32 /*bitRate*/) {};
-
- virtual void OnSendReportReceived(const WebRtc_Word32 id,
- const WebRtc_UWord32 senderSSRC) {};
-
- virtual void OnReceiveReportReceived(const WebRtc_Word32 id,
- const WebRtc_UWord32 senderSSRC) {};
-
-protected:
- virtual ~RtcpFeedback() {}
-};
-
-class RtpFeedback
-{
-public:
- // Receiving payload change or SSRC change. (return success!)
- /*
- * channels - number of channels in codec (1 = mono, 2 = stereo)
- */
- virtual WebRtc_Word32 OnInitializeDecoder(
- const WebRtc_Word32 id,
- const WebRtc_Word8 payloadType,
- const char payloadName[RTP_PAYLOAD_NAME_SIZE],
- const int frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate) = 0;
-
- virtual void OnPacketTimeout(const WebRtc_Word32 id) = 0;
-
- virtual void OnReceivedPacket(const WebRtc_Word32 id,
- const RtpRtcpPacketType packetType) = 0;
-
- virtual void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
- const RTPAliveType alive) = 0;
-
- virtual void OnIncomingSSRCChanged( const WebRtc_Word32 id,
- const WebRtc_UWord32 SSRC) = 0;
-
- virtual void OnIncomingCSRCChanged( const WebRtc_Word32 id,
- const WebRtc_UWord32 CSRC,
- const bool added) = 0;
-
-protected:
- virtual ~RtpFeedback() {}
-};
-
-class RtpAudioFeedback
-{
-public:
- virtual void OnReceivedTelephoneEvent(const WebRtc_Word32 id,
- const WebRtc_UWord8 event,
- const bool endOfEvent) = 0;
-
- virtual void OnPlayTelephoneEvent(const WebRtc_Word32 id,
- const WebRtc_UWord8 event,
- const WebRtc_UWord16 lengthMs,
- const WebRtc_UWord8 volume) = 0;
-
-protected:
- virtual ~RtpAudioFeedback() {}
-};
-
-
-class RtpVideoFeedback
-{
-public:
- // this function should call codec module to inform it about the request
- virtual void OnReceivedIntraFrameRequest(const WebRtc_Word32 id,
- const FrameType type,
- const WebRtc_UWord8 streamIdx) = 0;
-
- virtual void OnNetworkChanged(const WebRtc_Word32 id,
- const WebRtc_UWord32 bitrateBps,
- const WebRtc_UWord8 fractionLost,
- const WebRtc_UWord16 roundTripTimeMs) = 0;
-
-protected:
- virtual ~RtpVideoFeedback() {}
-};
-
-// A clock interface that allows reading of absolute and relative
-// timestamps in an RTP/RTCP module.
-class RtpRtcpClock {
-public:
- virtual ~RtpRtcpClock() {}
-
- // Return a timestamp in milliseconds relative to some arbitrary
- // source; the source is fixed for this clock.
- virtual WebRtc_UWord32 GetTimeInMS() = 0;
-
- // Retrieve an NTP absolute timestamp.
- virtual void CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac) = 0;
-};
-
-// RtpReceiveBitrateUpdate is used to signal changes in bitrate estimates for
-// the incoming stream.
-class RtpRemoteBitrateObserver
-{
-public:
- // Called when a receive channel has a new bitrate estimate for the incoming
- // stream.
- virtual void OnReceiveBitrateChanged(unsigned int ssrc,
- unsigned int bitrate) = 0;
-
- // Called when a REMB packet has been received.
- virtual void OnReceivedRemb(unsigned int bitrate) = 0;
-
- virtual ~RtpRemoteBitrateObserver() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_INTERFACE_RTP_RTCP_DEFINES_H_
diff --git a/trunk/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h b/trunk/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
deleted file mode 100644
index cc71e7b..0000000
--- a/trunk/src/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h
+++ /dev/null
@@ -1,288 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "../testing/gmock/include/gmock/gmock.h"
-
-#include "modules/interface/module.h"
-#include "modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-
-namespace webrtc {
-
-class MockRtpRtcp : public RtpRtcp {
- public:
- MOCK_METHOD1(ChangeUniqueId,
- WebRtc_Word32(const WebRtc_Word32 id));
- MOCK_METHOD1(RegisterDefaultModule,
- WebRtc_Word32(RtpRtcp* module));
- MOCK_METHOD0(DeRegisterDefaultModule,
- WebRtc_Word32());
- MOCK_METHOD0(DefaultModuleRegistered,
- bool());
- MOCK_METHOD0(NumberChildModules,
- WebRtc_UWord32());
- MOCK_METHOD1(RegisterSyncModule,
- WebRtc_Word32(RtpRtcp* module));
- MOCK_METHOD0(DeRegisterSyncModule,
- WebRtc_Word32());
- MOCK_METHOD0(InitReceiver,
- WebRtc_Word32());
- MOCK_METHOD1(RegisterIncomingDataCallback,
- WebRtc_Word32(RtpData* incomingDataCallback));
- MOCK_METHOD1(RegisterIncomingRTPCallback,
- WebRtc_Word32(RtpFeedback* incomingMessagesCallback));
- MOCK_METHOD2(SetPacketTimeout,
- WebRtc_Word32(const WebRtc_UWord32 RTPtimeoutMS, const WebRtc_UWord32 RTCPtimeoutMS));
- MOCK_METHOD2(SetPeriodicDeadOrAliveStatus,
- WebRtc_Word32(const bool enable, const WebRtc_UWord8 sampleTimeSeconds));
- MOCK_METHOD2(PeriodicDeadOrAliveStatus,
- WebRtc_Word32(bool &enable, WebRtc_UWord8 &sampleTimeSeconds));
- MOCK_METHOD1(RegisterReceivePayload,
- WebRtc_Word32(const CodecInst& voiceCodec));
- MOCK_METHOD1(RegisterReceivePayload,
- WebRtc_Word32(const VideoCodec& videoCodec));
- MOCK_METHOD2(ReceivePayloadType,
- WebRtc_Word32(const CodecInst& voiceCodec, WebRtc_Word8* plType));
- MOCK_METHOD2(ReceivePayloadType,
- WebRtc_Word32(const VideoCodec& videoCodec, WebRtc_Word8* plType));
- MOCK_METHOD1(DeRegisterReceivePayload,
- WebRtc_Word32(const WebRtc_Word8 payloadType));
- MOCK_METHOD2(RegisterReceiveRtpHeaderExtension,
- WebRtc_Word32(const RTPExtensionType type, const WebRtc_UWord8 id));
- MOCK_METHOD1(DeregisterReceiveRtpHeaderExtension,
- WebRtc_Word32(const RTPExtensionType type));
- MOCK_CONST_METHOD0(RemoteTimestamp,
- WebRtc_UWord32());
- MOCK_CONST_METHOD1(EstimatedRemoteTimeStamp,
- WebRtc_Word32(WebRtc_UWord32& timestamp));
- MOCK_CONST_METHOD0(RemoteSSRC,
- WebRtc_UWord32());
- MOCK_CONST_METHOD1(RemoteCSRCs,
- WebRtc_Word32(WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]));
- MOCK_CONST_METHOD1(SSRCFilter,
- WebRtc_Word32(WebRtc_UWord32& allowedSSRC));
- MOCK_METHOD2(SetSSRCFilter,
- WebRtc_Word32(const bool enable, const WebRtc_UWord32 allowedSSRC));
- MOCK_METHOD2(SetRTXReceiveStatus,
- WebRtc_Word32(const bool enable, const WebRtc_UWord32 SSRC));
- MOCK_CONST_METHOD2(RTXReceiveStatus,
- WebRtc_Word32(bool* enable, WebRtc_UWord32* SSRC));
- MOCK_METHOD2(IncomingPacket,
- WebRtc_Word32(const WebRtc_UWord8* incomingPacket, const WebRtc_UWord16 packetLength));
- MOCK_METHOD4(IncomingAudioNTP,
- WebRtc_Word32(const WebRtc_UWord32 audioReceivedNTPsecs, const WebRtc_UWord32 audioReceivedNTPfrac, const WebRtc_UWord32 audioRTCPArrivalTimeSecs, const WebRtc_UWord32 audioRTCPArrivalTimeFrac));
- MOCK_METHOD0(InitSender,
- WebRtc_Word32());
- MOCK_METHOD1(RegisterSendTransport,
- WebRtc_Word32(Transport* outgoingTransport));
- MOCK_METHOD1(SetMaxTransferUnit,
- WebRtc_Word32(const WebRtc_UWord16 size));
- MOCK_METHOD3(SetTransportOverhead,
- WebRtc_Word32(const bool TCP, const bool IPV6, const WebRtc_UWord8 authenticationOverhead));
- MOCK_CONST_METHOD0(MaxPayloadLength,
- WebRtc_UWord16());
- MOCK_CONST_METHOD0(MaxDataPayloadLength,
- WebRtc_UWord16());
- MOCK_METHOD3(SetRTPKeepaliveStatus,
- WebRtc_Word32(const bool enable, const WebRtc_Word8 unknownPayloadType, const WebRtc_UWord16 deltaTransmitTimeMS));
- MOCK_CONST_METHOD3(RTPKeepaliveStatus,
- WebRtc_Word32(bool* enable, WebRtc_Word8* unknownPayloadType, WebRtc_UWord16* deltaTransmitTimeMS));
- MOCK_CONST_METHOD0(RTPKeepalive,
- bool());
- MOCK_METHOD1(RegisterSendPayload,
- WebRtc_Word32(const CodecInst& voiceCodec));
- MOCK_METHOD1(RegisterSendPayload,
- WebRtc_Word32(const VideoCodec& videoCodec));
- MOCK_METHOD1(DeRegisterSendPayload,
- WebRtc_Word32(const WebRtc_Word8 payloadType));
- MOCK_METHOD2(RegisterSendRtpHeaderExtension,
- WebRtc_Word32(const RTPExtensionType type, const WebRtc_UWord8 id));
- MOCK_METHOD1(DeregisterSendRtpHeaderExtension,
- WebRtc_Word32(const RTPExtensionType type));
- MOCK_METHOD1(SetTransmissionSmoothingStatus,
- void(const bool enable));
- MOCK_CONST_METHOD0(TransmissionSmoothingStatus,
- bool());
- MOCK_CONST_METHOD0(StartTimestamp,
- WebRtc_UWord32());
- MOCK_METHOD1(SetStartTimestamp,
- WebRtc_Word32(const WebRtc_UWord32 timestamp));
- MOCK_CONST_METHOD0(SequenceNumber,
- WebRtc_UWord16());
- MOCK_METHOD1(SetSequenceNumber,
- WebRtc_Word32(const WebRtc_UWord16 seq));
- MOCK_CONST_METHOD0(SSRC,
- WebRtc_UWord32());
- MOCK_METHOD1(SetSSRC,
- WebRtc_Word32(const WebRtc_UWord32 ssrc));
- MOCK_CONST_METHOD1(CSRCs,
- WebRtc_Word32(WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]));
- MOCK_METHOD2(SetCSRCs,
- WebRtc_Word32(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize], const WebRtc_UWord8 arrLength));
- MOCK_METHOD1(SetCSRCStatus,
- WebRtc_Word32(const bool include));
- MOCK_METHOD3(SetRTXSendStatus,
- WebRtc_Word32(const bool enable, const bool setSSRC, const WebRtc_UWord32 SSRC));
- MOCK_CONST_METHOD2(RTXSendStatus,
- WebRtc_Word32(bool* enable, WebRtc_UWord32* SSRC));
- MOCK_METHOD1(SetSendingStatus,
- WebRtc_Word32(const bool sending));
- MOCK_CONST_METHOD0(Sending,
- bool());
- MOCK_METHOD1(SetSendingMediaStatus,
- WebRtc_Word32(const bool sending));
- MOCK_CONST_METHOD0(SendingMedia,
- bool());
- MOCK_CONST_METHOD4(BitrateSent,
- void(WebRtc_UWord32* totalRate, WebRtc_UWord32* videoRate, WebRtc_UWord32* fecRate, WebRtc_UWord32* nackRate));
- MOCK_CONST_METHOD1(EstimatedSendBandwidth,
- int(WebRtc_UWord32* available_bandwidth));
- MOCK_CONST_METHOD1(EstimatedReceiveBandwidth,
- int(WebRtc_UWord32* available_bandwidth));
- MOCK_METHOD7(SendOutgoingData,
- WebRtc_Word32(const FrameType frameType, const WebRtc_Word8 payloadType, const WebRtc_UWord32 timeStamp, const WebRtc_UWord8* payloadData, const WebRtc_UWord32 payloadSize, const RTPFragmentationHeader* fragmentation, const RTPVideoHeader* rtpVideoHdr));
- MOCK_METHOD1(RegisterIncomingRTCPCallback,
- WebRtc_Word32(RtcpFeedback* incomingMessagesCallback));
- MOCK_CONST_METHOD0(RTCP,
- RTCPMethod());
- MOCK_METHOD1(SetRTCPStatus,
- WebRtc_Word32(const RTCPMethod method));
- MOCK_METHOD1(SetCNAME,
- WebRtc_Word32(const char cName[RTCP_CNAME_SIZE]));
- MOCK_METHOD1(CNAME,
- WebRtc_Word32(char cName[RTCP_CNAME_SIZE]));
- MOCK_CONST_METHOD2(RemoteCNAME,
- WebRtc_Word32(const WebRtc_UWord32 remoteSSRC,
- char cName[RTCP_CNAME_SIZE]));
- MOCK_CONST_METHOD4(RemoteNTP,
- WebRtc_Word32(WebRtc_UWord32 *ReceivedNTPsecs, WebRtc_UWord32 *ReceivedNTPfrac, WebRtc_UWord32 *RTCPArrivalTimeSecs, WebRtc_UWord32 *RTCPArrivalTimeFrac));
- MOCK_METHOD2(AddMixedCNAME,
- WebRtc_Word32(const WebRtc_UWord32 SSRC, const WebRtc_Word8 cName[RTCP_CNAME_SIZE]));
- MOCK_METHOD1(RemoveMixedCNAME,
- WebRtc_Word32(const WebRtc_UWord32 SSRC));
- MOCK_CONST_METHOD5(RTT,
- WebRtc_Word32(const WebRtc_UWord32 remoteSSRC, WebRtc_UWord16* RTT, WebRtc_UWord16* avgRTT, WebRtc_UWord16* minRTT, WebRtc_UWord16* maxRTT));
- MOCK_METHOD1(ResetRTT,
- WebRtc_Word32(const WebRtc_UWord32 remoteSSRC));
- MOCK_METHOD1(SendRTCP,
- WebRtc_Word32(WebRtc_UWord32 rtcpPacketType));
- MOCK_METHOD1(SendRTCPReferencePictureSelection,
- WebRtc_Word32(const WebRtc_UWord64 pictureID));
- MOCK_METHOD1(SendRTCPSliceLossIndication,
- WebRtc_Word32(const WebRtc_UWord8 pictureID));
- MOCK_METHOD0(ResetStatisticsRTP,
- WebRtc_Word32());
- MOCK_CONST_METHOD5(StatisticsRTP,
- WebRtc_Word32(WebRtc_UWord8 *fraction_lost, WebRtc_UWord32 *cum_lost, WebRtc_UWord32 *ext_max, WebRtc_UWord32 *jitter, WebRtc_UWord32 *max_jitter));
- MOCK_METHOD0(ResetReceiveDataCountersRTP,
- WebRtc_Word32());
- MOCK_METHOD0(ResetSendDataCountersRTP,
- WebRtc_Word32());
- MOCK_CONST_METHOD4(DataCountersRTP,
- WebRtc_Word32(WebRtc_UWord32 *bytesSent, WebRtc_UWord32 *packetsSent, WebRtc_UWord32 *bytesReceived, WebRtc_UWord32 *packetsReceived));
- MOCK_METHOD1(RemoteRTCPStat,
- WebRtc_Word32(RTCPSenderInfo* senderInfo));
- MOCK_CONST_METHOD1(RemoteRTCPStat,
- WebRtc_Word32(std::vector<RTCPReportBlock>* receiveBlocks));
- MOCK_METHOD2(AddRTCPReportBlock,
- WebRtc_Word32(const WebRtc_UWord32 SSRC, const RTCPReportBlock* receiveBlock));
- MOCK_METHOD1(RemoveRTCPReportBlock,
- WebRtc_Word32(const WebRtc_UWord32 SSRC));
- MOCK_METHOD4(SetRTCPApplicationSpecificData,
- WebRtc_Word32(const WebRtc_UWord8 subType, const WebRtc_UWord32 name, const WebRtc_UWord8* data, const WebRtc_UWord16 length));
- MOCK_METHOD1(SetRTCPVoIPMetrics,
- WebRtc_Word32(const RTCPVoIPMetric* VoIPMetric));
- MOCK_CONST_METHOD0(REMB,
- bool());
- MOCK_METHOD1(SetREMBStatus,
- WebRtc_Word32(const bool enable));
- MOCK_METHOD3(SetREMBData,
- WebRtc_Word32(const WebRtc_UWord32 bitrate, const WebRtc_UWord8 numberOfSSRC, const WebRtc_UWord32* SSRC));
- MOCK_METHOD1(SetMaximumBitrateEstimate,
- WebRtc_Word32(const WebRtc_UWord32 bitrate));
- MOCK_METHOD1(SetRemoteBitrateObserver,
- bool(RtpRemoteBitrateObserver*));
- MOCK_CONST_METHOD0(IJ,
- bool());
- MOCK_METHOD1(SetIJStatus,
- WebRtc_Word32(const bool));
- MOCK_CONST_METHOD0(TMMBR,
- bool());
- MOCK_METHOD1(SetTMMBRStatus,
- WebRtc_Word32(const bool enable));
- MOCK_METHOD1(OnBandwidthEstimateUpdate,
- void(WebRtc_UWord16 bandWidthKbit));
- MOCK_CONST_METHOD0(NACK,
- NACKMethod());
- MOCK_METHOD1(SetNACKStatus,
- WebRtc_Word32(const NACKMethod method));
- MOCK_CONST_METHOD0(SelectiveRetransmissions,
- int());
- MOCK_METHOD1(SetSelectiveRetransmissions,
- int(uint8_t settings));
- MOCK_METHOD2(SendNACK,
- WebRtc_Word32(const WebRtc_UWord16* nackList, const WebRtc_UWord16 size));
- MOCK_METHOD2(SetStorePacketsStatus,
- WebRtc_Word32(const bool enable, const WebRtc_UWord16 numberToStore));
- MOCK_METHOD1(RegisterAudioCallback,
- WebRtc_Word32(RtpAudioFeedback* messagesCallback));
- MOCK_METHOD1(SetAudioPacketSize,
- WebRtc_Word32(const WebRtc_UWord16 packetSizeSamples));
- MOCK_METHOD3(SetTelephoneEventStatus,
- WebRtc_Word32(const bool enable, const bool forwardToDecoder, const bool detectEndOfTone));
- MOCK_CONST_METHOD0(TelephoneEvent,
- bool());
- MOCK_CONST_METHOD0(TelephoneEventForwardToDecoder,
- bool());
- MOCK_CONST_METHOD1(SendTelephoneEventActive,
- bool(WebRtc_Word8& telephoneEvent));
- MOCK_METHOD3(SendTelephoneEventOutband,
- WebRtc_Word32(const WebRtc_UWord8 key, const WebRtc_UWord16 time_ms, const WebRtc_UWord8 level));
- MOCK_METHOD1(SetSendREDPayloadType,
- WebRtc_Word32(const WebRtc_Word8 payloadType));
- MOCK_CONST_METHOD1(SendREDPayloadType,
- WebRtc_Word32(WebRtc_Word8& payloadType));
- MOCK_METHOD2(SetRTPAudioLevelIndicationStatus,
- WebRtc_Word32(const bool enable, const WebRtc_UWord8 ID));
- MOCK_CONST_METHOD2(GetRTPAudioLevelIndicationStatus,
- WebRtc_Word32(bool& enable, WebRtc_UWord8& ID));
- MOCK_METHOD1(SetAudioLevel,
- WebRtc_Word32(const WebRtc_UWord8 level_dBov));
- MOCK_METHOD1(RegisterIncomingVideoCallback,
- WebRtc_Word32(RtpVideoFeedback* incomingMessagesCallback));
- MOCK_METHOD1(SetCameraDelay,
- WebRtc_Word32(const WebRtc_Word32 delayMS));
- MOCK_METHOD3(SetSendBitrate,
- void(const WebRtc_UWord32 startBitrate, const WebRtc_UWord16 minBitrateKbit, const WebRtc_UWord16 maxBitrateKbit));
- MOCK_METHOD3(SetGenericFECStatus,
- WebRtc_Word32(const bool enable, const WebRtc_UWord8 payloadTypeRED, const WebRtc_UWord8 payloadTypeFEC));
- MOCK_METHOD3(GenericFECStatus,
- WebRtc_Word32(bool& enable, WebRtc_UWord8& payloadTypeRED, WebRtc_UWord8& payloadTypeFEC));
- MOCK_METHOD2(SetFECCodeRate,
- WebRtc_Word32(const WebRtc_UWord8 keyFrameCodeRate, const WebRtc_UWord8 deltaFrameCodeRate));
- MOCK_METHOD2(SetFECUepProtection,
- WebRtc_Word32(const bool keyUseUepProtection, const bool deltaUseUepProtection));
- MOCK_METHOD1(SetKeyFrameRequestMethod,
- WebRtc_Word32(const KeyFrameRequestMethod method));
- MOCK_METHOD0(RequestKeyFrame,
- WebRtc_Word32());
-
- MOCK_CONST_METHOD3(Version,
- int32_t(char* version, uint32_t& remaining_buffer_in_bytes, uint32_t& position));
- MOCK_METHOD0(TimeUntilNextProcess,
- int32_t());
- MOCK_METHOD0(Process,
- int32_t());
-
- // Members.
- unsigned int remote_ssrc_;
-};
-
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/Android.mk b/trunk/src/modules/rtp_rtcp/source/Android.mk
deleted file mode 100644
index 68295be..0000000
--- a/trunk/src/modules/rtp_rtcp/source/Android.mk
+++ /dev/null
@@ -1,67 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../android-webrtc.mk
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_rtp_rtcp
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_GENERATED_SOURCES :=
-LOCAL_SRC_FILES := \
- bitrate.cc \
- rtp_rtcp_impl.cc \
- rtcp_receiver.cc \
- rtcp_receiver_help.cc \
- rtcp_sender.cc \
- rtcp_utility.cc \
- rtp_receiver.cc \
- rtp_sender.cc \
- rtp_utility.cc \
- rtp_header_extension.cc \
- ssrc_database.cc \
- tmmbr_help.cc \
- dtmf_queue.cc \
- rtp_receiver_audio.cc \
- rtp_sender_audio.cc \
- bandwidth_management.cc \
- forward_error_correction.cc \
- forward_error_correction_internal.cc \
- overuse_detector.cc \
- remote_rate_control.cc \
- rtp_packet_history.cc \
- receiver_fec.cc \
- rtp_receiver_video.cc \
- rtp_sender_video.cc \
- rtp_format_vp8.cc \
- transmission_bucket.cc \
- vp8_partition_aggregator.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../.. \
- $(LOCAL_PATH)/../../interface \
- $(LOCAL_PATH)/../../../system_wrappers/interface
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/rtp_rtcp/source/Bitrate.h b/trunk/src/modules/rtp_rtcp/source/Bitrate.h
deleted file mode 100644
index ab5637b..0000000
--- a/trunk/src/modules/rtp_rtcp/source/Bitrate.h
+++ /dev/null
@@ -1,89 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_BITRATE_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_BITRATE_H_
-
-#include "typedefs.h"
-#include "rtp_rtcp_config.h" // misc. defines (e.g. MAX_PACKET_LENGTH)
-#include "common_types.h" // Transport
-#include <stdio.h>
-#include <list>
-
-namespace webrtc {
-class RtpRtcpClock;
-
-class Bitrate
-{
-public:
- Bitrate(RtpRtcpClock* clock);
-
- // initialize members
- void Init();
-
- // calculate rates
- void Process();
-
- // update with a packet
- void Update(const WebRtc_Word32 bytes);
-
- // packet rate last second, updated roughly every 100 ms
- WebRtc_UWord32 PacketRate() const;
-
- // bitrate last second, updated roughly every 100 ms
- WebRtc_UWord32 BitrateLast() const;
-
- // bitrate last second, updated now
- WebRtc_UWord32 BitrateNow() const;
-
-protected:
- RtpRtcpClock& _clock;
-
-private:
- WebRtc_UWord32 _packetRate;
- WebRtc_UWord32 _bitrate;
- WebRtc_UWord8 _bitrateNextIdx;
- WebRtc_UWord32 _packetRateArray[10];
- WebRtc_UWord32 _bitrateArray[10];
- WebRtc_UWord32 _bitrateDiffMS[10];
- WebRtc_UWord32 _timeLastRateUpdate;
- WebRtc_UWord32 _bytesCount;
- WebRtc_UWord32 _packetCount;
-};
-
-struct DataTimeSizeTuple
-{
- DataTimeSizeTuple(WebRtc_UWord32 sizeBytes, WebRtc_Word64 timeCompleteMs) :
- _sizeBytes(sizeBytes),
- _timeCompleteMs(timeCompleteMs) {}
-
- WebRtc_UWord32 _sizeBytes;
- WebRtc_Word64 _timeCompleteMs;
-};
-
-class BitRateStats
-{
-public:
- BitRateStats();
- ~BitRateStats();
-
- void Init();
- void Update(WebRtc_UWord32 packetSizeBytes, WebRtc_Word64 nowMs);
- WebRtc_UWord32 BitRate(WebRtc_Word64 nowMs);
-
-private:
- void EraseOld(WebRtc_Word64 nowMs);
-
- std::list<DataTimeSizeTuple*> _dataSamples;
- WebRtc_UWord32 _accumulatedBytes;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_BITRATE_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/H264/bitstream_builder.cc b/trunk/src/modules/rtp_rtcp/source/H264/bitstream_builder.cc
deleted file mode 100644
index 05b7e2f..0000000
--- a/trunk/src/modules/rtp_rtcp/source/H264/bitstream_builder.cc
+++ /dev/null
@@ -1,580 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "bitstream_builder.h"
-
-#include <string.h>
-
-namespace webrtc {
-BitstreamBuilder::BitstreamBuilder(WebRtc_UWord8* data, const WebRtc_UWord32 dataSize) :
- _data(data),
- _dataSize(dataSize),
- _byteOffset(0),
- _bitOffset(0)
-{
- memset(data, 0, dataSize);
-}
-
-WebRtc_UWord32
-BitstreamBuilder::Length() const
-{
- return _byteOffset+ (_bitOffset?1:0);
-}
-
-WebRtc_Word32
-BitstreamBuilder::Add1Bit(const WebRtc_UWord8 bit)
-{
- // sanity
- if(_bitOffset + 1 > 8)
- {
- if(_dataSize < Length()+1)
- {
- // not enough space in buffer
- return -1;
- }
- }
- Add1BitWithoutSanity(bit);
- return 0;
-}
-
-void
-BitstreamBuilder::Add1BitWithoutSanity(const WebRtc_UWord8 bit)
-{
- if(bit & 0x1)
- {
- _data[_byteOffset] += (1 << (7-_bitOffset));
- }
-
- if(_bitOffset == 7)
- {
- // last bit in byte
- _bitOffset = 0;
- _byteOffset++;
- } else
- {
- _bitOffset++;
- }
-}
-
-WebRtc_Word32
-BitstreamBuilder::Add2Bits(const WebRtc_UWord8 bits)
-{
- // sanity
- if(_bitOffset + 2 > 8)
- {
- if(_dataSize < Length()+1)
- {
- // not enough space in buffer
- return -1;
- }
- }
- Add1BitWithoutSanity(bits >> 1);
- Add1BitWithoutSanity(bits);
- return 0;
-}
-
-WebRtc_Word32
-BitstreamBuilder::Add3Bits(const WebRtc_UWord8 bits)
-{
- // sanity
- if(_bitOffset + 3 > 8)
- {
- if(_dataSize < Length()+1)
- {
- // not enough space in buffer
- return -1;
- }
- }
- Add1BitWithoutSanity(bits >> 2);
- Add1BitWithoutSanity(bits >> 1);
- Add1BitWithoutSanity(bits);
- return 0;
-}
-
-WebRtc_Word32
-BitstreamBuilder::Add4Bits(const WebRtc_UWord8 bits)
-{
- // sanity
- if(_bitOffset + 4 > 8)
- {
- if(_dataSize < Length()+1)
- {
- // not enough space in buffer
- return -1;
- }
- }
- Add1BitWithoutSanity(bits >> 3);
- Add1BitWithoutSanity(bits >> 2);
- Add1BitWithoutSanity(bits >> 1);
- Add1BitWithoutSanity(bits);
- return 0;
-}
-
-WebRtc_Word32
-BitstreamBuilder::Add5Bits(const WebRtc_UWord8 bits)
-{
- // sanity
- if(_bitOffset + 5 > 8)
- {
- if(_dataSize < Length()+1)
- {
- // not enough space in buffer
- return -1;
- }
- }
- Add1BitWithoutSanity(bits >> 4);
- Add1BitWithoutSanity(bits >> 3);
- Add1BitWithoutSanity(bits >> 2);
- Add1BitWithoutSanity(bits >> 1);
- Add1BitWithoutSanity(bits);
- return 0;
-}
-
-WebRtc_Word32
-BitstreamBuilder::Add6Bits(const WebRtc_UWord8 bits)
-{
- // sanity
- if(_bitOffset + 6 > 8)
- {
- if(_dataSize < Length()+1)
- {
- // not enough space in buffer
- return -1;
- }
- }
- Add1BitWithoutSanity(bits >> 5);
- Add1BitWithoutSanity(bits >> 4);
- Add1BitWithoutSanity(bits >> 3);
- Add1BitWithoutSanity(bits >> 2);
- Add1BitWithoutSanity(bits >> 1);
- Add1BitWithoutSanity(bits);
- return 0;
-}
-
-WebRtc_Word32
-BitstreamBuilder::Add7Bits(const WebRtc_UWord8 bits)
-{
- // sanity
- if(_bitOffset + 7 > 8)
- {
- if(_dataSize < Length()+1)
- {
- // not enough space in buffer
- return -1;
- }
- }
- Add1BitWithoutSanity(bits >> 6);
- Add1BitWithoutSanity(bits >> 5);
- Add1BitWithoutSanity(bits >> 4);
- Add1BitWithoutSanity(bits >> 3);
- Add1BitWithoutSanity(bits >> 2);
- Add1BitWithoutSanity(bits >> 1);
- Add1BitWithoutSanity(bits);
- return 0;
-}
-
-WebRtc_Word32
-BitstreamBuilder::Add8Bits(const WebRtc_UWord8 bits)
-{
- // sanity
- if(_dataSize < Length()+1)
- {
- // not enough space in buffer
- return -1;
- }
- if(_bitOffset == 0)
- {
- _data[_byteOffset] = bits;
- } else
- {
- _data[_byteOffset] += (bits >> _bitOffset);
- _data[_byteOffset+1] += (bits << (8-_bitOffset));
- }
- _byteOffset++;
- return 0;
-}
-
-WebRtc_Word32
-BitstreamBuilder::Add16Bits(const WebRtc_UWord16 bits)
-{
- // sanity
- if(_dataSize < Length()+2)
- {
- // not enough space in buffer
- return -1;
- }
- if(_bitOffset == 0)
- {
- _data[_byteOffset] = (WebRtc_UWord8)(bits >> 8);
- _data[_byteOffset+1] = (WebRtc_UWord8)(bits);
- } else
- {
- _data[_byteOffset] += (WebRtc_UWord8)(bits >> (_bitOffset + 8));
- _data[_byteOffset+1] += (WebRtc_UWord8)(bits >> _bitOffset);
- _data[_byteOffset+2] += (WebRtc_UWord8)(bits << (8-_bitOffset));
- }
- _byteOffset += 2;
- return 0;
-}
-
-WebRtc_Word32
-BitstreamBuilder::Add24Bits(const WebRtc_UWord32 bits)
-{
- // sanity
- if(_dataSize < Length()+3)
- {
- // not enough space in buffer
- return -1;
- }
- if(_bitOffset == 0)
- {
- _data[_byteOffset] = (WebRtc_UWord8)(bits >> 16);
- _data[_byteOffset+1] = (WebRtc_UWord8)(bits >> 8);
- _data[_byteOffset+2] = (WebRtc_UWord8)(bits);
- } else
- {
- _data[_byteOffset] += (WebRtc_UWord8)(bits >> (_bitOffset+16));
- _data[_byteOffset+1] += (WebRtc_UWord8)(bits >> (_bitOffset+8));
- _data[_byteOffset+2] += (WebRtc_UWord8)(bits >> (_bitOffset));
- _data[_byteOffset+3] += (WebRtc_UWord8)(bits << (8-_bitOffset));
- }
- _byteOffset += 3;
- return 0;
-}
-
-WebRtc_Word32
-BitstreamBuilder::Add32Bits(const WebRtc_UWord32 bits)
-{
- // sanity
- if(_dataSize < Length()+4)
- {
- // not enough space in buffer
- return -1;
- }
- if(_bitOffset == 0)
- {
- _data[_byteOffset] = (WebRtc_UWord8)(bits >> 24);
- _data[_byteOffset+1] = (WebRtc_UWord8)(bits >> 16);
- _data[_byteOffset+2] = (WebRtc_UWord8)(bits >> 8);
- _data[_byteOffset+3] = (WebRtc_UWord8)(bits);
- } else
- {
- _data[_byteOffset] += (WebRtc_UWord8)(bits >> (_bitOffset+24));
- _data[_byteOffset+1] += (WebRtc_UWord8)(bits >> (_bitOffset+16));
- _data[_byteOffset+2] += (WebRtc_UWord8)(bits >> (_bitOffset+8));
- _data[_byteOffset+3] += (WebRtc_UWord8)(bits >> (_bitOffset));
- _data[_byteOffset+4] += (WebRtc_UWord8)(bits << (8-_bitOffset));
- }
- _byteOffset += 4;
- return 0;
-}
-
-// Exp-Golomb codes
-/*
- with "prefix" and "suffix" bits and assignment to codeNum ranges (informative)
- Bit string form Range of codeNum
- 1 0
- 0 1 x0 1..2 2bits-1
- 0 0 1 x1 x0 3..6 3bits-1
- 0 0 0 1 x2 x1 x0 7..14 4bits-1
- 0 0 0 0 1 x3 x2 x1 x0 15..30
- 0 0 0 0 0 1 x4 x3 x2 x1 x0 31..62
-*/
-WebRtc_Word32
-BitstreamBuilder::AddUE(const WebRtc_UWord32 value)
-{
- // un-rolled on 8 bit base to avoid too deep if else chain
- if(value < 0x0000ffff)
- {
- if(value < 0x000000ff)
- {
- if(value == 0)
- {
- if(AddPrefix(0) != 0)
- {
- return -1;
- }
- } else if(value < 3)
- {
- if(AddPrefix(1) != 0)
- {
- return -1;
- }
- AddSuffix(1, value-1);
- } else if(value < 7)
- {
- if(AddPrefix(2) != 0)
- {
- return -1;
- }
- AddSuffix(2, value-3);
- } else if(value < 15)
- {
- if(AddPrefix(3) != 0)
- {
- return -1;
- }
- AddSuffix(3, value-7);
- } else if(value < 31)
- {
- if(AddPrefix(4) != 0)
- {
- return -1;
- }
- AddSuffix(4, value-15);
- } else if(value < 63)
- {
- if(AddPrefix(5) != 0)
- {
- return -1;
- }
- AddSuffix(5, value-31);
- } else if(value < 127)
- {
- if(AddPrefix(6) != 0)
- {
- return -1;
- }
- AddSuffix(6, value-63);
- } else
- {
- if(AddPrefix(7) != 0)
- {
- return -1;
- }
- AddSuffix(7, value-127);
- }
- }else
- {
- if(value < 0x000001ff)
- {
- if(AddPrefix(8) != 0)
- {
- return -1;
- }
- AddSuffix(8, value-0x000000ff);
- } else if(value < 0x000003ff)
- {
- if(AddPrefix(9) != 0)
- {
- return -1;
- }
- AddSuffix(9, value-0x000001ff);
- } else if(value < 0x000007ff)
- {
- if(AddPrefix(10) != 0)
- {
- return -1;
- }
- AddSuffix(10, value-0x000003ff);
- } else if(value < 0x00000fff)
- {
- if(AddPrefix(11) != 0)
- {
- return -1;
- }
- AddSuffix(1, value-0x000007ff);
- } else if(value < 0x00001fff)
- {
- if(AddPrefix(12) != 0)
- {
- return -1;
- }
- AddSuffix(12, value-0x00000fff);
- } else if(value < 0x00003fff)
- {
- if(AddPrefix(13) != 0)
- {
- return -1;
- }
- AddSuffix(13, value-0x00001fff);
- } else if(value < 0x00007fff)
- {
- if(AddPrefix(14) != 0)
- {
- return -1;
- }
- AddSuffix(14, value-0x00003fff);
- } else
- {
- if(AddPrefix(15) != 0)
- {
- return -1;
- }
- AddSuffix(15, value-0x00007fff);
- }
- }
- }else
- {
- if(value < 0x00ffffff)
- {
- if(value < 0x0001ffff)
- {
- if(AddPrefix(16) != 0)
- {
- return -1;
- }
- AddSuffix(16, value-0x0000ffff);
- } else if(value < 0x0003ffff)
- {
- if(AddPrefix(17) != 0)
- {
- return -1;
- }
- AddSuffix(17, value-0x0001ffff);
- } else if(value < 0x0007ffff)
- {
- if(AddPrefix(18) != 0)
- {
- return -1;
- }
- AddSuffix(18, value-0x0003ffff);
- } else if(value < 0x000fffff)
- {
- if(AddPrefix(19) != 0)
- {
- return -1;
- }
- AddSuffix(19, value-0x0007ffff);
- } else if(value < 0x001fffff)
- {
- if(AddPrefix(20) != 0)
- {
- return -1;
- }
- AddSuffix(20, value-0x000fffff);
- } else if(value < 0x003fffff)
- {
- if(AddPrefix(21) != 0)
- {
- return -1;
- }
- AddSuffix(21, value-0x001fffff);
- } else if(value < 0x007fffff)
- {
- if(AddPrefix(22) != 0)
- {
- return -1;
- }
- AddSuffix(22, value-0x003fffff);
- } else
- {
- if(AddPrefix(23) != 0)
- {
- return -1;
- }
- AddSuffix(23, value-0x007fffff);
- }
- } else
- {
- if(value < 0x01ffffff)
- {
- if(AddPrefix(24) != 0)
- {
- return -1;
- }
- AddSuffix(24, value-0x00ffffff);
- } else if(value < 0x03ffffff)
- {
- if(AddPrefix(25) != 0)
- {
- return -1;
- }
- AddSuffix(25, value-0x01ffffff);
- } else if(value < 0x07ffffff)
- {
- if(AddPrefix(26) != 0)
- {
- return -1;
- }
- AddSuffix(26, value-0x03ffffff);
- } else if(value < 0x0fffffff)
- {
- if(AddPrefix(27) != 0)
- {
- return -1;
- }
- AddSuffix(27, value-0x07ffffff);
- } else if(value < 0x1fffffff)
- {
- if(AddPrefix(28) != 0)
- {
- return -1;
- }
- AddSuffix(28, value-0x0fffffff);
- } else if(value < 0x3fffffff)
- {
- if(AddPrefix(29) != 0)
- {
- return -1;
- }
- AddSuffix(29, value-0x1fffffff);
- } else if(value < 0x7fffffff)
- {
- if(AddPrefix(30) != 0)
- {
- return -1;
- }
- AddSuffix(30, value-0x3fffffff);
- } else if(value < 0xffffffff)
- {
- if(AddPrefix(31) != 0)
- {
- return -1;
- }
- AddSuffix(31, value-0x7ffffff);
- } else
- {
- if(AddPrefix(32) != 0)
- {
- return -1;
- }
- AddSuffix(32, 0); // exactly 0xffffffff
- }
- }
- }
- return 0;
-}
-
-WebRtc_Word32
-BitstreamBuilder::AddPrefix(const WebRtc_UWord8 numZeros)
-{
- // sanity for the sufix too
- WebRtc_UWord32 numBitsToAdd = numZeros * 2 + 1;
- if(((_dataSize - _byteOffset) *8 + 8-_bitOffset) < numBitsToAdd)
- {
- return -1;
- }
-
- // add numZeros
- for (WebRtc_UWord32 i = 0; i < numZeros; i++)
- {
- Add1Bit(0);
- }
- Add1Bit(1);
- return 0;
-}
-
-void
-BitstreamBuilder::AddSuffix(const WebRtc_UWord8 numBits, const WebRtc_UWord32 rest)
-{
- // most significant bit first
- for(WebRtc_Word32 i = numBits - 1; i >= 0; i--)
- {
- if(( rest >> i) & 0x1)
- {
- Add1Bit(1);
- }else
- {
- Add1Bit(0);
- }
- }
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/H264/bitstream_builder.h b/trunk/src/modules/rtp_rtcp/source/H264/bitstream_builder.h
deleted file mode 100644
index c88ef8f..0000000
--- a/trunk/src/modules/rtp_rtcp/source/H264/bitstream_builder.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_BUILDER_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_BUILDER_H_
-
-#include "typedefs.h"
-
-namespace webrtc {
-class BitstreamBuilder
-{
-public:
- BitstreamBuilder(WebRtc_UWord8* data, const WebRtc_UWord32 dataSize);
-
- WebRtc_UWord32 Length() const;
-
- WebRtc_Word32 Add1Bit(const WebRtc_UWord8 bit);
- WebRtc_Word32 Add2Bits(const WebRtc_UWord8 bits);
- WebRtc_Word32 Add3Bits(const WebRtc_UWord8 bits);
- WebRtc_Word32 Add4Bits(const WebRtc_UWord8 bits);
- WebRtc_Word32 Add5Bits(const WebRtc_UWord8 bits);
- WebRtc_Word32 Add6Bits(const WebRtc_UWord8 bits);
- WebRtc_Word32 Add7Bits(const WebRtc_UWord8 bits);
- WebRtc_Word32 Add8Bits(const WebRtc_UWord8 bits);
- WebRtc_Word32 Add16Bits(const WebRtc_UWord16 bits);
- WebRtc_Word32 Add24Bits(const WebRtc_UWord32 bits);
- WebRtc_Word32 Add32Bits(const WebRtc_UWord32 bits);
-
- // Exp-Golomb codes
- WebRtc_Word32 AddUE(const WebRtc_UWord32 value);
-
-private:
- WebRtc_Word32 AddPrefix(const WebRtc_UWord8 numZeros);
- void AddSuffix(const WebRtc_UWord8 numBits, const WebRtc_UWord32 rest);
- void Add1BitWithoutSanity(const WebRtc_UWord8 bit);
-
- WebRtc_UWord8* _data;
- WebRtc_UWord32 _dataSize;
-
- WebRtc_UWord32 _byteOffset;
- WebRtc_UWord8 _bitOffset;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_BUILDER_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/H264/bitstream_parser.cc b/trunk/src/modules/rtp_rtcp/source/H264/bitstream_parser.cc
deleted file mode 100644
index 79ec967..0000000
--- a/trunk/src/modules/rtp_rtcp/source/H264/bitstream_parser.cc
+++ /dev/null
@@ -1,217 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "bitstream_parser.h"
-
-namespace webrtc {
-BitstreamParser::BitstreamParser(const WebRtc_UWord8* data, const WebRtc_UWord32 dataLength) :
- _data(data),
- _dataLength(dataLength),
- _byteOffset(0),
- _bitOffset(0)
-{
-}
- // todo should we have any error codes from this?
-
-WebRtc_UWord8
-BitstreamParser::Get1Bit()
-{
- WebRtc_UWord8 retVal = 0x1 & (_data[_byteOffset] >> (7-_bitOffset++));
-
- // prepare next byte
- if(_bitOffset == 8)
- {
- _bitOffset = 0;
- _byteOffset++;
- }
- return retVal;
-}
-
-WebRtc_UWord8
-BitstreamParser::Get2Bits()
-{
- WebRtc_UWord8 retVal = (Get1Bit() << 1);
- retVal += Get1Bit();
- return retVal;
-}
-
-WebRtc_UWord8
-BitstreamParser::Get3Bits()
-{
- WebRtc_UWord8 retVal = (Get1Bit() << 2);
- retVal += (Get1Bit() << 1);
- retVal += Get1Bit();
- return retVal;
-}
-
-WebRtc_UWord8
-BitstreamParser::Get4Bits()
-{
- WebRtc_UWord8 retVal = (Get1Bit() << 3);
- retVal += (Get1Bit() << 2);
- retVal += (Get1Bit() << 1);
- retVal += Get1Bit();
- return retVal;
-}
-
-WebRtc_UWord8
-BitstreamParser::Get5Bits()
-{
- WebRtc_UWord8 retVal = (Get1Bit() << 4);
- retVal += (Get1Bit() << 3);
- retVal += (Get1Bit() << 2);
- retVal += (Get1Bit() << 1);
- retVal += Get1Bit();
- return retVal;
-}
-
-WebRtc_UWord8
-BitstreamParser::Get6Bits()
-{
- WebRtc_UWord8 retVal = (Get1Bit() << 5);
- retVal += (Get1Bit() << 4);
- retVal += (Get1Bit() << 3);
- retVal += (Get1Bit() << 2);
- retVal += (Get1Bit() << 1);
- retVal += Get1Bit();
- return retVal;
-}
-
-WebRtc_UWord8
-BitstreamParser::Get7Bits()
-{
- WebRtc_UWord8 retVal = (Get1Bit() << 6);
- retVal += (Get1Bit() << 5);
- retVal += (Get1Bit() << 4);
- retVal += (Get1Bit() << 3);
- retVal += (Get1Bit() << 2);
- retVal += (Get1Bit() << 1);
- retVal += Get1Bit();
- return retVal;
-}
-
-WebRtc_UWord8
-BitstreamParser::Get8Bits()
-{
- WebRtc_UWord16 retVal;
-
- if(_bitOffset != 0)
- {
- // read 16 bits
- retVal = (_data[_byteOffset] << 8)+ (_data[_byteOffset+1]) ;
- retVal = retVal >> (8-_bitOffset);
- } else
- {
- retVal = _data[_byteOffset];
- }
- _byteOffset++;
- return (WebRtc_UWord8)retVal;
-}
-
-WebRtc_UWord16
-BitstreamParser::Get16Bits()
-{
- WebRtc_UWord32 retVal;
-
- if(_bitOffset != 0)
- {
- // read 24 bits
- retVal = (_data[_byteOffset] << 16) + (_data[_byteOffset+1] << 8) + (_data[_byteOffset+2]);
- retVal = retVal >> (8-_bitOffset);
- }else
- {
- // read 16 bits
- retVal = (_data[_byteOffset] << 8) + (_data[_byteOffset+1]) ;
- }
- _byteOffset += 2;
- return (WebRtc_UWord16)retVal;
-}
-
-WebRtc_UWord32
-BitstreamParser::Get24Bits()
-{
- WebRtc_UWord32 retVal;
-
- if(_bitOffset != 0)
- {
- // read 32 bits
- retVal = (_data[_byteOffset] << 24) + (_data[_byteOffset+1] << 16) + (_data[_byteOffset+2] << 8) + (_data[_byteOffset+3]);
- retVal = retVal >> (8-_bitOffset);
- }else
- {
- // read 24 bits
- retVal = (_data[_byteOffset] << 16) + (_data[_byteOffset+1] << 8) + (_data[_byteOffset+2]) ;
- }
- _byteOffset += 3;
- return retVal & 0x00ffffff; // we need to clean up the high 8 bits
-}
-
-WebRtc_UWord32
-BitstreamParser::Get32Bits()
-{
- WebRtc_UWord32 retVal;
-
- if(_bitOffset != 0)
- {
- // read 40 bits
- WebRtc_UWord64 tempVal = _data[_byteOffset];
- tempVal <<= 8;
- tempVal += _data[_byteOffset+1];
- tempVal <<= 8;
- tempVal += _data[_byteOffset+2];
- tempVal <<= 8;
- tempVal += _data[_byteOffset+3];
- tempVal <<= 8;
- tempVal += _data[_byteOffset+4];
- tempVal >>= (8-_bitOffset);
-
- retVal = WebRtc_UWord32(tempVal);
- }else
- {
- // read 32 bits
- retVal = (_data[_byteOffset]<< 24) + (_data[_byteOffset+1] << 16) + (_data[_byteOffset+2] << 8) + (_data[_byteOffset+3]) ;
- }
- _byteOffset += 4;
- return retVal;
-}
-
-// Exp-Golomb codes
-/*
- with "prefix" and "suffix" bits and assignment to codeNum ranges (informative)
- Bit string form Range of codeNum
- 1 0
- 0 1 x0 1..2
- 0 0 1 x1 x0 3..6
- 0 0 0 1 x2 x1 x0 7..14
- 0 0 0 0 1 x3 x2 x1 x0 15..30
- 0 0 0 0 0 1 x4 x3 x2 x1 x0 31..62
-*/
-
-WebRtc_UWord32
-BitstreamParser::GetUE()
-{
- WebRtc_UWord32 retVal = 0;
- WebRtc_UWord8 numLeadingZeros = 0;
-
- while (Get1Bit() != 1)
- {
- numLeadingZeros++;
- }
- // prefix
- retVal = (1 << numLeadingZeros) - 1;
-
- // suffix
- while (numLeadingZeros)
- {
- retVal += (Get1Bit() << --numLeadingZeros);
- }
- return retVal;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/H264/bitstream_parser.h b/trunk/src/modules/rtp_rtcp/source/H264/bitstream_parser.h
deleted file mode 100644
index 3d8f9ef..0000000
--- a/trunk/src/modules/rtp_rtcp/source/H264/bitstream_parser.h
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
-
-#include "typedefs.h"
-
-namespace webrtc {
-class BitstreamParser
-{
-public:
- BitstreamParser(const WebRtc_UWord8* data, const WebRtc_UWord32 dataLength);
-
- WebRtc_UWord8 Get1Bit();
- WebRtc_UWord8 Get2Bits();
- WebRtc_UWord8 Get3Bits();
- WebRtc_UWord8 Get4Bits();
- WebRtc_UWord8 Get5Bits();
- WebRtc_UWord8 Get6Bits();
- WebRtc_UWord8 Get7Bits();
- WebRtc_UWord8 Get8Bits();
- WebRtc_UWord16 Get16Bits();
- WebRtc_UWord32 Get24Bits();
- WebRtc_UWord32 Get32Bits();
-
- // Exp-Golomb codes
- WebRtc_UWord32 GetUE();
-
-private:
- const WebRtc_UWord8* _data;
- const WebRtc_UWord32 _dataLength;
-
- WebRtc_UWord32 _byteOffset;
- WebRtc_UWord8 _bitOffset;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_BITSTREAM_PARSER_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/H264/h264_information.cc b/trunk/src/modules/rtp_rtcp/source/H264/h264_information.cc
deleted file mode 100644
index cf6b549..0000000
--- a/trunk/src/modules/rtp_rtcp/source/H264/h264_information.cc
+++ /dev/null
@@ -1,818 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string.h>
-#include "h264_information.h"
-
-//#define DEBUG_SEI_MESSAGE 1
-
-#ifdef DEBUG_SEI_MESSAGE
- #include "bitstream_parser.h"
- #include <stdio.h>
- #include <math.h>
-
- WebRtc_UWord32 BitRateBPS(WebRtc_UWord16 x )
- {
- return (x & 0x3fff) * WebRtc_UWord32(pow(10.0f,(2 + (x >> 14))));
- }
-
-#endif
-
-namespace webrtc {
-H264Information::H264Information(const bool SVC)
- : _SVC(SVC)
-
-{
-}
-
-H264Information::~H264Information()
-{
-
-}
-
-void
-H264Information::Reset()
-{
- _parsedLength = 0;
- _remLength = 0;
- _length = 0;
- _info.numNALUs = 0;
- _info.numLayers = 0;
-
- memset(_info.startCodeSize, 0, sizeof(_info.startCodeSize));
- memset(_info.payloadSize, 0, sizeof(_info.payloadSize));
- memset(_info.NRI, 0, sizeof(_info.NRI));
- memset(_info.type, 0, sizeof(_info.type));
- memset(_info.accLayerSize, 0, sizeof(_info.accLayerSize));
-
- for (WebRtc_Word32 i = 0; i < KMaxNumberOfNALUs; i++)
- {
- _info.SVCheader[i].idr = 0;
- _info.SVCheader[i].priorityID = 0;
- _info.SVCheader[i].interLayerPred = 0;
- _info.SVCheader[i].dependencyID = 0;
- _info.SVCheader[i].qualityID = 0;
- _info.SVCheader[i].temporalID = 0;
- _info.SVCheader[i].useRefBasePic = 0;
- _info.SVCheader[i].discardable = 0;
- _info.SVCheader[i].output = 0;
-
- _info.PACSI[i].X = 0;
- _info.PACSI[i].Y = 0;
-// _info.PACSI[i].T = 0;
- _info.PACSI[i].A = 0;
- _info.PACSI[i].P = 0;
- _info.PACSI[i].C = 0;
- _info.PACSI[i].S = 0;
- _info.PACSI[i].E = 0;
- _info.PACSI[i].TL0picIDx = 0;
- _info.PACSI[i].IDRpicID = 0;
- _info.PACSI[i].DONC = 0;
- _info.PACSI[i].numSEINALUs = 0;
- _info.PACSI[i].NALlength = 5;
- }
-}
-
-/*******************************************************************************
- * WebRtc_Word32 GetInfo(const WebRtc_UWord8* ptrEncodedBuffer,
- * const WebRtc_UWord32 length,
- * const H264Info*& ptrInfo);
- *
- * Gets information from an encoded stream.
- *
- * Input:
- * - ptrEncodedBuffer : Pointer to encoded stream.
- * - length : Length in bytes of encoded stream.
- *
- * Output:
- * - ptrInfo : Pointer to struct with H.264 info.
- *
- * Return value:
- * - 0 : ok
- * - (-1) : Error
- */
-WebRtc_Word32
-H264Information::GetInfo(const WebRtc_UWord8* ptrEncodedBuffer,
- const WebRtc_UWord32 length,
- const H264Info*& ptrInfo)
-{
- if (!ptrEncodedBuffer || length < 4)
- {
- return -1;
- }
-
- if (!HasInfo(length))
- {
- if (-1 == FindInfo(ptrEncodedBuffer, length))
- {
- Reset();
- return -1;
- }
- }
- ptrInfo = &_info;
- return 0;
-}
-
-RtpVideoCodecTypes
-H264Information::Type()
-{
- if(_SVC)
- {
- return RTP_H264_SVCVideo;
- }
- return RTP_H264Video;
-}
-
-
-/*******************************************************************************
- * bool HasInfo(const WebRtc_UWord32 length);
- *
- * Checks if information has already been stored for this encoded stream.
- *
- * Input:
- * - length : Length in bytes of encoded stream.
- *
- * Return value:
- * - true (false) : Information has (not) been stored.
- */
-
-bool
-H264Information::HasInfo(const WebRtc_UWord32 length)
-{
- if (!_info.numNALUs)
- {
- return false;
- }
-
- // has info, make sure current length matches info length
- if (length != _length)
- {
- Reset();
- return false;
- }
-
- return true;
-}
-
-/*******************************************************************************
- * WebRtc_Word32 FindInfo(const WebRtc_UWord8* ptrEncodedBuffer,
- * const WebRtc_UWord32 length);
- *
- * Parses the encoded stream.
- *
- * Input:
- * - ptrEncodedBuffer : Pointer to encoded stream.
- * - length : Length in bytes of encoded stream.
- *
- * Return value:
- * - 0 : ok
- * - (-1) : Error
- */
-WebRtc_Word32
-H264Information::FindInfo(const WebRtc_UWord8* ptrEncodedBuffer, const WebRtc_UWord32 length)
-{
- _ptrData = ptrEncodedBuffer;
- _length = length;
- _parsedLength = 0;
- _remLength = length;
-
- do
- {
- // Get start code length
- if (FindNALUStartCodeSize() == -1)
- {
- Reset();
- return -1;
- }
-
- // Get NAL unit payload size
- WebRtc_Word32 foundLast = FindNALU();
- if (foundLast == -1)
- {
- Reset();
- return -1;
- }
-
- // Validate parsed length
- if (_parsedLength > _length)
- {
- Reset();
- return -1;
- }
-
- // Get NRI
- GetNRI();
-
- // Get type
- if (FindNALUType() == -1)
- {
- Reset();
- return -1;
- }
-
- // Set layer start end bit
- SetLayerSEBit(foundLast);
-
-
- // Last NAL unit found?
- if (foundLast == 1)
- {
- if (_parsedLength != _length)
- {
- Reset();
- return -1;
- }
- _info.numNALUs++;
- return SetLayerLengths();
- }
-
- // Next NAL unit
- _ptrData += (_info.startCodeSize[_info.numNALUs] + _info.payloadSize[_info.numNALUs]);
- _remLength -= (_info.startCodeSize[_info.numNALUs] + _info.payloadSize[_info.numNALUs]);
- _info.numNALUs++;
-
- // Validate memory allocation
- if (_info.numNALUs >= KMaxNumberOfNALUs)
- {
- Reset();
- return -1;
- }
- }
- while(true);
-
- return 0;
-}
-
-/*******************************************************************************
- * WebRtc_Word32 FindNALUStartCodeSize();
- *
- * Finds the start code length of the current NAL unit.
- *
- * Output:
- * - _info.startCodeSize[currentNALU] : Start code length in bytes of NAL unit.
- *
- * Return value:
- * - 0 : ok
- * - (-1) : Error
- */
-WebRtc_Word32
-H264Information::FindNALUStartCodeSize()
-{
- // NAL unit start code. Ex. {0,0,1} or {0,0,0,1}
- for (WebRtc_UWord32 i = 2; i < _remLength; i++)
- {
- if (_ptrData[i] == 1 && _ptrData[i - 1] == 0 && _ptrData[i - 2] == 0)
- {
- _info.startCodeSize[_info.numNALUs] = WebRtc_UWord8(i + 1);
- return 0;
- }
- }
- return -1;
-}
-
-/*******************************************************************************
- * WebRtc_Word32 FindNALU();
- *
- * Finds the length of the current NAL unit.
- *
- * Output:
- * - _info.payloadSize[currentNALU] : Payload length in bytes of NAL unit
- * (start code length not included).
- * - _parsedLength : Current parsed length in bytes.
- *
- * Return value:
- * - 1 : ok. Last NAL unit found.
- * - 0 : ok
- * - (-1) : Error
- */
-WebRtc_Word32
-H264Information::FindNALU()
-{
- for (WebRtc_UWord32 i = _info.startCodeSize[_info.numNALUs]; i < _remLength - 2; i += 2)
- {
- if (_ptrData[i] == 0)
- {
- WebRtc_Word32 size = 0;
- if ((_ptrData[i + 1] == 1 && _ptrData[i - 1] == 0) ||
- (_ptrData[i + 2] == 1 && _ptrData[i + 1] == 0))
- {
- // Found a header
- // Reduce size by preceding zeroes
- while (_ptrData[i - 1] == 0)
- {
- i--;
- }
- size = i;
- }
- if (size > 0)
- {
- _info.payloadSize[_info.numNALUs] = size - _info.startCodeSize[_info.numNALUs];
- _parsedLength += _info.startCodeSize[_info.numNALUs] + _info.payloadSize[_info.numNALUs];
- return 0;
- }
- }
- }
- // Last NAL unit
- _info.payloadSize[_info.numNALUs] = _remLength - _info.startCodeSize[_info.numNALUs];
- if (_info.payloadSize[_info.numNALUs] > 0)
- {
- _parsedLength += _info.startCodeSize[_info.numNALUs] + _info.payloadSize[_info.numNALUs];
- return 1;
- }
- return -1;
-}
-
-/*******************************************************************************
- * void GetNRI();
- *
- * Finds the NRI of the current NAL unit.
- *
- * Output:
- * - _info.NRI[currentNALU] : NRI of NAL unit.
- *
- * Return value:
- * - 0 : ok
- * - (-1) : Error
- */
-void
-H264Information::GetNRI()
-{
- // NAL unit header (1 byte)
- // ---------------------------------
- // | start code |F|NRI| Type |
- // ---------------------------------
-
- // NRI (2 bits) - nal_ref_idc. '00' - the NAL unit is not used to reconstruct reference pictures.
- // >00 - the NAL unit is required to reconstruct reference pictures
- // in the same layer, or contains a parameter set.
-
-
- const WebRtc_UWord8 type = _ptrData[_info.startCodeSize[_info.numNALUs]] & 0x1f;
-
- // NALU type of 5, 7 and 8 shoud have NRI to b011
- if( type == 5 ||
- type == 7 ||
- type == 8)
- {
- _info.NRI[_info.numNALUs] = 0x60;
- }else
- {
- _info.NRI[_info.numNALUs] = _ptrData[_info.startCodeSize[_info.numNALUs]] & 0x60;
- }
-}
-
-
-/*******************************************************************************
- * WebRtc_Word32 FindNALUType();
- *
- * Finds the type of the current NAL unit.
- *
- * Output:
- * - _info.type[currentNALU] : Type of NAL unit
- *
- * Return value:
- * - 0 : ok
- * - (-1) : Error
- */
-WebRtc_Word32
-H264Information::FindNALUType()
-{
- // NAL unit header (1 byte)
- // ---------------------------------
- // | start code |F|NRI| Type |
- // ---------------------------------
-
- _info.type[_info.numNALUs] = _ptrData[_info.startCodeSize[_info.numNALUs]] & 0x1f;
-
- if (_info.type[_info.numNALUs] == 0)
- {
- return -1;
- }
-
- // SVC NAL units, extended header
- if (ParseSVCNALUHeader() == -1)
- {
- return -1;
- }
-
- return 0;
-}
-
-/*******************************************************************************
- * WebRtc_Word32 ParseSVCNALUHeader();
- *
- * Finds the extended header of the current NAL unit. Included for NAL unit types 14 and 20.
- *
- * Output:
- * - _info.SVCheader[currentNALU] : SVC header of NAL unit.
- *
- * Return value:
- * - 0 : ok
- * - (-1) : Error
- */
-WebRtc_Word32
-H264Information::ParseSVCNALUHeader()
-{
- if (_info.type[_info.numNALUs] == 5)
- {
- _info.SVCheader[_info.numNALUs].idr = 1;
- }
- if (_info.type[_info.numNALUs] == 6)
- {
- WebRtc_UWord32 seiPayloadSize;
- do
- {
- // SEI message
- seiPayloadSize = 0;
-
- WebRtc_UWord32 curByte = _info.startCodeSize[_info.numNALUs] + 1;
- const WebRtc_UWord32 seiStartOffset = curByte;
-
- WebRtc_UWord32 seiPayloadType = 0;
- while(_ptrData[curByte] == 0xff)
- {
- seiPayloadType += 255;
- curByte++;
- }
- seiPayloadType += _ptrData[curByte++];
-
- while(_ptrData[curByte] == 0xff)
- {
- seiPayloadSize += 255;
- curByte++;
- }
- seiPayloadSize += _ptrData[curByte++];
-
- if(_info.payloadSize[_info.numNALUs] < _info.startCodeSize[_info.numNALUs] + seiPayloadSize)
- {
- // sanity of remaining buffer
- // return 0 since no one "need" SEI messages
- assert(false);
- return 0;
- }
-
- if(seiPayloadType == 24)
- {
- // we add this to NALU 0 to be signaled in the first PACSI packet
- _info.PACSI[0].numSEINALUs = 1; // we allways add this to NALU 0 to send it in the first packet
- if(_info.PACSI[0].seiMessageLength[0] != seiPayloadSize)
- {
- _info.PACSI[0].seiMessageLength[0] = seiPayloadSize;
- delete [] _info.PACSI[0].seiMessageData[0];
- _info.PACSI[0].seiMessageData[0] = new WebRtc_UWord8[seiPayloadSize];
- }
- memcpy(_info.PACSI[0].seiMessageData[0], _ptrData+seiStartOffset, seiPayloadSize);
-
- _info.PACSI[0].NALlength += seiPayloadSize + 2; // additional 2 is the length
-
-#ifdef DEBUG_SEI_MESSAGE
- const WebRtc_UWord8 numberOfLayers = 10;
- WebRtc_UWord16 avgBitrate[numberOfLayers]= {0,0,0,0,0,0,0,0,0,0};
- WebRtc_UWord16 maxBitrateLayer[numberOfLayers]= {0,0,0,0,0,0,0,0,0,0};
- WebRtc_UWord16 maxBitrateLayerRepresentation[numberOfLayers] = {0,0,0,0,0,0,0,0,0,0};
- WebRtc_UWord16 maxBitrareCalcWindow[numberOfLayers] = {0,0,0,0,0,0,0,0,0,0};
-
- BitstreamParser parserScalabilityInfo(_ptrData+curByte, seiPayloadSize);
-
- parserScalabilityInfo.Get1Bit(); // not used in futher parsing
- const WebRtc_UWord8 priority_layer_info_present = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 priority_id_setting_flag = parserScalabilityInfo.Get1Bit();
-
- WebRtc_UWord32 numberOfLayersMinusOne = parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 j = 0; j<= numberOfLayersMinusOne; j++)
- {
- printf("\nLayer ID:%d \n",parserScalabilityInfo.GetUE());
- printf("Priority ID:%d \n", parserScalabilityInfo.Get6Bits());
- printf("Discardable:%d \n", parserScalabilityInfo.Get1Bit());
-
- printf("Dependency ID:%d \n", parserScalabilityInfo.Get3Bits());
- printf("Quality ID:%d \n", parserScalabilityInfo.Get4Bits());
- printf("Temporal ID:%d \n", parserScalabilityInfo.Get3Bits());
-
- const WebRtc_UWord8 sub_pic_layer_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 sub_region_layer_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 iroi_division_info_present_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 profile_level_info_present_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 bitrate_info_present_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 frm_rate_info_present_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 frm_size_info_present_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 layer_dependency_info_present_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 parameter_sets_info_present_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 bitstream_restriction_info_present_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 exact_inter_layer_pred_flag = parserScalabilityInfo.Get1Bit(); // not used in futher parsing
-
- if(sub_pic_layer_flag || iroi_division_info_present_flag)
- {
- parserScalabilityInfo.Get1Bit();
- }
- const WebRtc_UWord8 layer_conversion_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 layer_output_flag = parserScalabilityInfo.Get1Bit(); // not used in futher parsing
-
- if(profile_level_info_present_flag)
- {
- parserScalabilityInfo.Get24Bits();
- }
- if(bitrate_info_present_flag)
- {
- // this is what we want
- avgBitrate[j] = parserScalabilityInfo.Get16Bits();
- maxBitrateLayer[j] = parserScalabilityInfo.Get16Bits();
- maxBitrateLayerRepresentation[j] = parserScalabilityInfo.Get16Bits();
- maxBitrareCalcWindow[j] = parserScalabilityInfo.Get16Bits();
-
- printf("\tAvg:%d\n", BitRateBPS(avgBitrate[j]));
- printf("\tmaxBitrate:%d\n", BitRateBPS(maxBitrateLayer[j]));
- printf("\tmaxBitrate rep:%d\n", BitRateBPS(maxBitrateLayerRepresentation[j]));
- printf("\tCalcWindow:%d\n", maxBitrareCalcWindow[j]);
- }
- if(frm_rate_info_present_flag)
- {
- printf("\tFrame rate constant:%d\n", parserScalabilityInfo.Get2Bits()); // 0 = not constant, 1 = constant, 2 = maybe...
- printf("\tFrame rate avg:%d\n", parserScalabilityInfo.Get16Bits()/256);
- }
- if(frm_size_info_present_flag || iroi_division_info_present_flag)
- {
- printf("\tFrame Width:%d\n",(parserScalabilityInfo.GetUE()+1)*16);
- printf("\tFrame Height:%d\n",(parserScalabilityInfo.GetUE()+1)*16);
- }
- if(sub_region_layer_flag)
- {
- parserScalabilityInfo.GetUE();
- if(parserScalabilityInfo.Get1Bit())
- {
- parserScalabilityInfo.Get16Bits();
- parserScalabilityInfo.Get16Bits();
- parserScalabilityInfo.Get16Bits();
- parserScalabilityInfo.Get16Bits();
- }
- }
- if(sub_pic_layer_flag)
- {
- parserScalabilityInfo.GetUE();
- }
- if(iroi_division_info_present_flag)
- {
- if(parserScalabilityInfo.Get1Bit())
- {
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.GetUE();
- }else
- {
- const WebRtc_UWord32 numRoisMinusOne = parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 k = 0; k <= numRoisMinusOne; k++)
- {
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.GetUE();
- }
- }
- }
- if(layer_dependency_info_present_flag)
- {
- const WebRtc_UWord32 numDirectlyDependentLayers = parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 k = 0; k < numDirectlyDependentLayers; k++)
- {
- parserScalabilityInfo.GetUE();
- }
- } else
- {
- parserScalabilityInfo.GetUE();
- }
- if(parameter_sets_info_present_flag)
- {
- const WebRtc_UWord32 numSeqParameterSetMinusOne = parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 k = 0; k <= numSeqParameterSetMinusOne; k++)
- {
- parserScalabilityInfo.GetUE();
- }
- const WebRtc_UWord32 numSubsetSeqParameterSetMinusOne = parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 l = 0; l <= numSubsetSeqParameterSetMinusOne; l++)
- {
- parserScalabilityInfo.GetUE();
- }
- const WebRtc_UWord32 numPicParameterSetMinusOne = parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 m = 0; m <= numPicParameterSetMinusOne; m++)
- {
- parserScalabilityInfo.GetUE();
- }
- }else
- {
- parserScalabilityInfo.GetUE();
- }
- if(bitstream_restriction_info_present_flag)
- {
- parserScalabilityInfo.Get1Bit();
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.GetUE();
- }
- if(layer_conversion_flag)
- {
- parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 k = 0; k <2;k++)
- {
- if(parserScalabilityInfo.Get1Bit())
- {
- parserScalabilityInfo.Get24Bits();
- parserScalabilityInfo.Get16Bits();
- parserScalabilityInfo.Get16Bits();
- }
- }
- }
- }
- if(priority_layer_info_present)
- {
- const WebRtc_UWord32 prNumDidMinusOne = parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 k = 0; k <= prNumDidMinusOne;k++)
- {
- parserScalabilityInfo.Get3Bits();
- const WebRtc_UWord32 prNumMinusOne = parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 l = 0; l <= prNumMinusOne; l++)
- {
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.Get24Bits();
- parserScalabilityInfo.Get16Bits();
- parserScalabilityInfo.Get16Bits();
- }
- }
- }
- if(priority_id_setting_flag)
- {
- WebRtc_UWord8 priorityIdSettingUri;
- WebRtc_UWord32 priorityIdSettingUriIdx = 0;
- do
- {
- priorityIdSettingUri = parserScalabilityInfo.Get8Bits();
- } while (priorityIdSettingUri != 0);
- }
-#endif
- } else
- {
- // not seiPayloadType 24 ignore
- }
- //check if we have more SEI in NALU
- } while (_info.payloadSize[_info.numNALUs] > _info.startCodeSize[_info.numNALUs] + seiPayloadSize);
- }
-
- // Extended NAL unit header (3 bytes).
- // +---------------+---------------+---------------+
- // |0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7|
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // |R|I| PRID |N| DID | QID | TID |U|D|O| RR|
- // +---------------+---------------+---------------+
-
- // R - Reserved for future extensions (MUST be 1). Receivers SHOULD ignore the value of R.
- // I - Is layer representation an IDR layer (1) or not (0).
- // PRID - Priority identifier for the NAL unit.
- // N - Specifies whether inter-layer prediction may be used for decoding the coded slice (1) or not (0).
- // DID - Indicates the inter-layer coding dependency level of a layer representation.
- // QID - Indicates the quality level of an MGS layer representation.
- // TID - Indicates the temporal level of a layer representation.
- // U - Use only reference base pictures during the inter prediction process (1) or not (0).
- // D - Discardable flag.
- // O - Output_flag. Affects the decoded picture output process as defined in Annex C of [H.264].
- // RR - Reserved_three_2bits (MUST be '11'). Receivers SHOULD ignore the value of RR.
-
- if (_info.type[_info.numNALUs] == 14 ||
- _info.type[_info.numNALUs] == 20)
- {
- WebRtc_UWord32 curByte = _info.startCodeSize[_info.numNALUs] + 1;
-
- if (_remLength < curByte + 3)
- {
- return -1;
- }
-
- _info.SVCheader[_info.numNALUs].idr = (_ptrData[curByte] >> 6) & 0x01;
- _info.SVCheader[_info.numNALUs].priorityID = (_ptrData[curByte++] & 0x3F);
-
- _info.SVCheader[_info.numNALUs].interLayerPred = (_ptrData[curByte] >> 7) & 0x01;
- _info.SVCheader[_info.numNALUs].dependencyID = (_ptrData[curByte] >> 4) & 0x07;
- _info.SVCheader[_info.numNALUs].qualityID = (_ptrData[curByte++] & 0x0F);
-
- _info.SVCheader[_info.numNALUs].temporalID = (_ptrData[curByte] >> 5) & 0x07;
- _info.SVCheader[_info.numNALUs].useRefBasePic = (_ptrData[curByte] >> 4) & 0x01;
- _info.SVCheader[_info.numNALUs].discardable = (_ptrData[curByte] >> 3) & 0x01;
- _info.SVCheader[_info.numNALUs].output = (_ptrData[curByte] >> 2) & 0x01;
-
- if (_info.type[_info.numNALUs] == 14)
- {
- // inform the next NALU
- memcpy(&(_info.SVCheader[_info.numNALUs+1]), &(_info.SVCheader[_info.numNALUs]), sizeof(_H264_SVC_NALUHeader));
- }
- }
- return 0;
-}
-
-
-/*******************************************************************************
- * void SetLayerSEBit();
- *
- * Sets start and end bits for the current NAL unit.
- *
- * Output:
- * - _info.PACSI[currentNALU].S : First NAL unit in a layer (S = 1).
- * - _info.PACSI[currentNALU].E : Last NAL unit in a layer (E = 1).
- *
- */
-void
-H264Information::SetLayerSEBit(WebRtc_Word32 foundLast)
-{
- if (_info.numNALUs == 0)
- {
- // First NAL unit
- _info.PACSI[_info.numNALUs].S = 1;
- }
-
- if (_info.numNALUs > 0)
- {
- if (_info.type[_info.numNALUs] != _info.type[_info.numNALUs-1] &&
- (_info.type[_info.numNALUs] == 20))
- {
- // First layer in scalable extension
- _info.PACSI[_info.numNALUs].S = 1;
- _info.PACSI[_info.numNALUs-1].E = 1;
- }
-
- if (_info.type[_info.numNALUs] == 20 && _info.type[_info.numNALUs-1] == 20)
- {
- if (_info.SVCheader[_info.numNALUs].temporalID != _info.SVCheader[_info.numNALUs-1].temporalID ||
- _info.SVCheader[_info.numNALUs].dependencyID != _info.SVCheader[_info.numNALUs-1].dependencyID ||
- _info.SVCheader[_info.numNALUs].qualityID != _info.SVCheader[_info.numNALUs-1].qualityID)
- {
- // New layer in scalable extension
- _info.PACSI[_info.numNALUs].S = 1;
- _info.PACSI[_info.numNALUs-1].E = 1;
- }
- }
- }
-
- if (foundLast)
- {
- // Last NAL unit
- _info.PACSI[_info.numNALUs].E = 1;
- }
-
-}
-
-/*******************************************************************************
- * WebRtc_Word32 SetLayerLengths();
- *
- * Sets the accumulated layer length.
- *
- * Output:
- * - _info.accLayerSize[currentLayer] : Size in bytes of layer: 0 - currentLayer.
- *
- * Return value:
- * - 0 : ok
- * - (-1) : Error
- *
- */
-WebRtc_Word32
-H264Information::SetLayerLengths()
-{
- for (WebRtc_UWord32 curNALU = 0; curNALU < _info.numNALUs; curNALU++)
- {
- _info.accLayerSize[_info.numLayers] += _info.startCodeSize[curNALU] + _info.payloadSize[curNALU];
-
- if (_info.PACSI[curNALU].E == 1)
- {
- _info.numLayers++;
- if (curNALU == WebRtc_UWord32(_info.numNALUs - 1))
- {
- break;
- }
- if (_info.numLayers >= KMaxNumberOfLayers)
- {
- Reset();
- return -1;
- }
- _info.accLayerSize[_info.numLayers] += _info.accLayerSize[_info.numLayers - 1];
- }
- }
-
- if (_info.numLayers < 1 && _info.numLayers > KMaxNumberOfLayers)
- {
- Reset();
- return -1;
- }
-
- if (_info.accLayerSize[_info.numLayers - 1] != WebRtc_Word32(_length))
- {
- Reset();
- return -1;
- }
-
- return 0;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/H264/h264_information.h b/trunk/src/modules/rtp_rtcp/source/H264/h264_information.h
deleted file mode 100644
index c7f5214..0000000
--- a/trunk/src/modules/rtp_rtcp/source/H264/h264_information.h
+++ /dev/null
@@ -1,170 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_H264_INFORMATION_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_H264_INFORMATION_H_
-
-#include "VideoCodecInformation.h"
-#include "typedefs.h"
-
-namespace webrtc {
-enum
-{
- KMaxNumberOfNALUs = 128,
- KMaxNumberOfSEINALUs = 2,
- KMaxNumberOfLayers = 16
-};
-
-struct H264_SVC_NALUHeader
-{
- H264_SVC_NALUHeader()
- :
- r(1),
- idr(0),
- priorityID(0),
- interLayerPred(0),
- dependencyID(0),
- qualityID(0),
- temporalID(0),
- useRefBasePic(0),
- discardable(0),
- output(0),
- rr(3),
- length(3)
- {
- }
- const WebRtc_UWord8 r;
- WebRtc_UWord8 idr;
- WebRtc_UWord8 priorityID;
- WebRtc_UWord8 interLayerPred;
- WebRtc_UWord8 dependencyID;
- WebRtc_UWord8 qualityID;
- WebRtc_UWord8 temporalID;
- WebRtc_UWord8 useRefBasePic;
- WebRtc_UWord8 discardable;
- WebRtc_UWord8 output;
- const WebRtc_UWord8 rr;
- const WebRtc_UWord8 length;
-};
-
-class H264_PACSI_NALU
-{
-public:
- H264_PACSI_NALU() :
- NALlength(5),
- type(30),
- X(0),
- Y(0),
-// T(0),
- A(0),
- P(0),
- C(0),
- S(0),
- E(0),
- TL0picIDx(0),
- IDRpicID(0),
- DONC(0),
- numSEINALUs(0)
- {
- memset(seiMessageLength, 0, sizeof(seiMessageLength));
- memset(seiMessageData, 0, sizeof(seiMessageData));
- }
- ~H264_PACSI_NALU()
- {
- for(int i = 0; i<KMaxNumberOfSEINALUs; i++)
- {
- if(seiMessageData[i])
- {
- delete [] seiMessageData[i];
- }
- }
- }
-
- WebRtc_UWord32 NALlength;
- const WebRtc_UWord8 type;
- WebRtc_UWord8 X;
- WebRtc_UWord8 Y;
-// WebRtc_UWord8 T;
- WebRtc_UWord8 A;
- WebRtc_UWord8 P;
- WebRtc_UWord8 C;
- WebRtc_UWord8 S;
- WebRtc_UWord8 E;
- WebRtc_UWord8 TL0picIDx;
- WebRtc_UWord16 IDRpicID;
- WebRtc_UWord16 DONC;
- WebRtc_UWord32 numSEINALUs;
- WebRtc_UWord32 seiMessageLength[KMaxNumberOfSEINALUs]; // we allow KMaxNumberOfSEINALUs SEI messages
- WebRtc_UWord8* seiMessageData[KMaxNumberOfSEINALUs];
-};
-
-struct H264Info
-{
- H264Info()
- :
- numNALUs(0),
- numLayers(0)
- {
- memset(startCodeSize, 0, sizeof(startCodeSize));
- memset(payloadSize, 0, sizeof(payloadSize));
- memset(NRI, 0, sizeof(NRI));
- memset(type, 0, sizeof(type));
- memset(accLayerSize, 0, sizeof(accLayerSize));
- }
- WebRtc_UWord16 numNALUs;
- WebRtc_UWord8 numLayers;
- WebRtc_UWord8 startCodeSize[KMaxNumberOfNALUs];
- WebRtc_UWord32 payloadSize[KMaxNumberOfNALUs];
- WebRtc_UWord8 NRI[KMaxNumberOfNALUs];
- WebRtc_UWord8 type[KMaxNumberOfNALUs];
- H264_SVC_NALUHeader SVCheader[KMaxNumberOfNALUs];
- H264_PACSI_NALU PACSI[KMaxNumberOfNALUs];
- WebRtc_Word32 accLayerSize[KMaxNumberOfLayers];
-};
-
-
-class H264Information : public VideoCodecInformation
-{
-public:
- H264Information(const bool SVC);
- ~H264Information();
-
- virtual void Reset();
-
- virtual RtpVideoCodecTypes Type();
-
- virtual WebRtc_Word32 GetInfo(const WebRtc_UWord8* ptrEncodedBuffer, const WebRtc_UWord32 length, const H264Info*& ptrInfo);
-
-
-protected:
- bool HasInfo(const WebRtc_UWord32 length);
- WebRtc_Word32 FindInfo(const WebRtc_UWord8* ptrEncodedBuffer, const WebRtc_UWord32 length);
-
- void GetNRI();
- WebRtc_Word32 FindNALU();
- WebRtc_Word32 FindNALUStartCodeSize();
- WebRtc_Word32 FindNALUType();
-
- WebRtc_Word32 ParseSVCNALUHeader();
-
- void SetLayerSEBit(WebRtc_Word32 foundLast);
- WebRtc_Word32 SetLayerLengths();
-
-private:
- const bool _SVC;
- const WebRtc_UWord8* _ptrData;
- WebRtc_UWord32 _length;
- WebRtc_UWord32 _parsedLength;
- WebRtc_UWord32 _remLength;
- H264Info _info;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_H264_INFORMATION_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/H264/rtp_sender_h264.cc b/trunk/src/modules/rtp_rtcp/source/H264/rtp_sender_h264.cc
deleted file mode 100644
index 1f35526..0000000
--- a/trunk/src/modules/rtp_rtcp/source/H264/rtp_sender_h264.cc
+++ /dev/null
@@ -1,1280 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rtp_sender_h264.h"
-
-#include "rtp_utility.h"
-
-namespace webrtc {
-RTPSenderH264::RTPSenderH264(RTPSenderInterface* rtpSender) :
- // H264
- _rtpSender(*rtpSender),
- _h264Mode(H264_SINGLE_NAL_MODE),
- _h264SendPPS_SPS(true),
- _h264SVCPayloadType(-1),
- _h264SVCRelaySequenceNumber(0),
- _h264SVCRelayTimeStamp(0),
- _h264SVCRelayLayerComplete(false),
-
- _useHighestSendLayer(false),
- _highestDependencyLayerOld(MAX_NUMBER_OF_TEMPORAL_ID-1),
- _highestDependencyQualityIDOld(MAX_NUMBER_OF_DEPENDENCY_QUALITY_ID-1),
- _highestDependencyLayer(0),
- _highestDependencyQualityID(0),
- _highestTemporalLayer(0)
-{
-}
-
-RTPSenderH264::~RTPSenderH264()
-{
-}
-
-WebRtc_Word32
-RTPSenderH264::Init()
-{
- _h264SendPPS_SPS = true;
- _h264Mode = H264_SINGLE_NAL_MODE;
- return 0;
-}
-
-/*
- multi-session
- 3 modes supported
- NI-T timestamps
- NI-TC timestamps/CS-DON
- NI-C CS-DON
-
- Non-interleaved timestamp based mode (NI-T)
- Non-interleaved cross-session decoding order number (CS-DON) based mode (NI-C)
- Non-interleaved combined timestamp and CS-DON mode (NI-TC)
-
- NOT supported Interleaved CS-DON (I-C) mode.
-
- NI-T and NI-TC modes both use timestamps to recover the decoding
- order. In order to be able to do so, it is necessary for the RTP
- packet stream to contain data for all sampling instances of a given
- RTP session in all enhancement RTP sessions that depend on the given
- RTP session. The NI-C and I-C modes do not have this limitation,
- and use the CS-DON values as a means to explicitly indicate decoding
- order, either directly coded in PACSI NAL units, or inferred from
- them using the packetization rules. It is noted that the NI-TC mode
- offers both alternatives and it is up to the receiver to select
- which one to use.
-*/
-
-bool
-RTPSenderH264::AddH264SVCNALUHeader(const H264_SVC_NALUHeader& svc,
- WebRtc_UWord8* databuffer,
- WebRtc_Word32& curByte) const
-{
- // +---------------+---------------+---------------+
- // |0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7|0|1|2|3|4|5|6|7|
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // |R|I| PRID |N| DID | QID | TID |U|D|O| RR|
- // +---------------+---------------+---------------+
-
- // R - Reserved for future extensions (MUST be 1). Receivers SHOULD ignore the value of R.
- // I - Is layer representation an IDR layer (1) or not (0).
- // PRID - Priority identifier for the NAL unit.
- // N - Specifies whether inter-layer prediction may be used for decoding the coded slice (1) or not (0).
- // DID - Indicates the WebRtc_Word32er-layer coding dependency level of a layer representation.
- // QID - Indicates the quality level of an MGS layer representation.
- // TID - Indicates the temporal level of a layer representation.
- // U - Use only reference base pictures during the WebRtc_Word32er prediction process (1) or not (0).
- // D - Discardable flag.
- // O - Output_flag. Affects the decoded picture output process as defined in Annex C of [H.264].
- // RR - Reserved_three_2bits (MUST be '11'). Receivers SHOULD ignore the value of RR.
-
- // Add header data
- databuffer[curByte++] = (svc.r << 7) + (svc.idr << 6) + (svc.priorityID & 0x3F);
- databuffer[curByte++] = (svc.interLayerPred << 7) + (svc.dependencyID << 4) + (svc.qualityID & 0x0F);
- databuffer[curByte++] = (svc.temporalID << 5) + (svc.useRefBasePic << 4) + (svc.discardable << 3) +
- (svc.output << 2) + (svc.rr & 0x03);
- return true;
-}
-
-WebRtc_Word32
-RTPSenderH264::AddH264PACSINALU(const bool firstPacketInNALU,
- const bool lastPacketInNALU,
- const H264_PACSI_NALU& pacsi,
- const H264_SVC_NALUHeader& svc,
- const WebRtc_UWord16 DONC,
- WebRtc_UWord8* databuffer,
- WebRtc_Word32& curByte) const
-{
- // 0 1 2 3
- // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // |F|NRI|Type(30) | SVC NAL unit header |
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // |X|Y|T|A|P|C|S|E| TL0PICIDX (o.)| IDRPICID (o.) |
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // | DONC (o.) | NAL unit size 1 |
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // | |
- // | SEI NAL unit 1 |
- // | |
- // | +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // | | NAL unit size 2 | |
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+ |
- // | |
- // | SEI NAL unit 2 |
- // | +-+-+-+-+-+-+-+-+-+-+
- // | |
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-
- // If present, MUST be first NAL unit in aggregation packet + there MUST be at least
- // one additional unit in the same packet! The RTPHeader and payload header are set as if the 2nd NAL unit
- // (first non-PACSI NAL unit) is encapsulated in the same packet.
- // contains scalability info common for all remaining NAL units.
-
- // todo add API to configure this required for multisession
- const bool addDONC = false;
-
- if (svc.length == 0 || pacsi.NALlength == 0)
- {
- return 0;
- }
-
- WebRtc_Word32 startByte = curByte;
-
- // NAL unit header
- databuffer[curByte++] = 30; // NRI will be added later
-
- // Extended SVC header
- AddH264SVCNALUHeader(svc, databuffer, curByte);
-
- // Flags
- databuffer[curByte++] = (pacsi.X << 7) +
- (pacsi.Y << 6) +
- (addDONC << 5) +
- (pacsi.A << 4) +
- (pacsi.P << 3) +
- (pacsi.C << 2) +
- firstPacketInNALU?(pacsi.S << 1):0 +
- lastPacketInNALU?(pacsi.E):0;
-
- // Optional fields
- if (pacsi.Y)
- {
- databuffer[curByte++] = pacsi.TL0picIDx;
- databuffer[curByte++] = (WebRtc_UWord8)(pacsi.IDRpicID >> 8);
- databuffer[curByte++] = (WebRtc_UWord8)(pacsi.IDRpicID);
- }
- // Decoding order number
- if (addDONC) // pacsi.T
- {
- databuffer[curByte++] = (WebRtc_UWord8)(DONC >> 8);
- databuffer[curByte++] = (WebRtc_UWord8)(DONC);
- }
-
- // SEI NALU
- if(firstPacketInNALU) // IMPROVEMENT duplicate it to make sure it arrives...
- {
- // we only set this for NALU 0 to make sure we send it only once per frame
- for (WebRtc_UWord32 i = 0; i < pacsi.numSEINALUs; i++)
- {
- // NALU size
- databuffer[curByte++] = (WebRtc_UWord8)(pacsi.seiMessageLength[i] >> 8);
- databuffer[curByte++] = (WebRtc_UWord8)(pacsi.seiMessageLength[i]);
-
- // NALU data
- memcpy(databuffer + curByte, pacsi.seiMessageData[i], pacsi.seiMessageLength[i]);
- curByte += pacsi.seiMessageLength[i];
- }
- }
- return curByte - startByte;
-}
-
-WebRtc_Word32
-RTPSenderH264::SetH264RelaySequenceNumber(const WebRtc_UWord16 seqNum)
-{
- _h264SVCRelaySequenceNumber = seqNum;
- return 0;
-}
-
-WebRtc_Word32
-RTPSenderH264::SetH264RelayCompleteLayer(const bool complete)
-{
- _h264SVCRelayLayerComplete = complete;
- return 0;
-}
-
-/*
- 12 Filler data
-
- The only restriction of filler data NAL units within an
- access unit is that they shall not precede the first VCL
- NAL unit with the same access unit.
-*/
-WebRtc_Word32
-RTPSenderH264::SendH264FillerData(const WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord16 bytesToSend,
- const WebRtc_UWord32 ssrc)
-{
- WebRtc_UWord16 fillerLength = bytesToSend - 12 - 1;
-
- if (fillerLength > WEBRTC_IP_PACKET_SIZE - 12 - 1)
- {
- return 0;
- }
-
- if (fillerLength == 0)
- {
- // do not send an empty packet, will not reach JB
- fillerLength = 1;
- }
-
- // send codec valid data, H.264 has defined data which is binary 1111111
- WebRtc_UWord8 dataBuffer[WEBRTC_IP_PACKET_SIZE];
-
- dataBuffer[0] = static_cast<WebRtc_UWord8>(0x80); // version 2
- dataBuffer[1] = rtpHeader->header.payloadType;
- ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+2, _rtpSender.IncrementSequenceNumber()); // get the current SequenceNumber and add by 1 after returning
- ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+4, rtpHeader->header.timestamp);
- ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, rtpHeader->header.ssrc);
-
- // set filler NALU type
- dataBuffer[12] = 12; // NRI field = 0, type 12
-
- // fill with 0xff
- memset(dataBuffer + 12 + 1, 0xff, fillerLength);
-
- return _rtpSender.SendToNetwork(dataBuffer,
- fillerLength,
- 12 + 1);
-}
-
-WebRtc_Word32
-RTPSenderH264::SendH264FillerData(const WebRtc_UWord32 captureTimestamp,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 bytes
- )
-{
-
- const WebRtc_UWord16 rtpHeaderLength = _rtpSender.RTPHeaderLength();
- WebRtc_UWord16 maxLength = _rtpSender.MaxPayloadLength() - FECPacketOverhead() - _rtpSender.RTPHeaderLength();
-
- WebRtc_Word32 bytesToSend=bytes;
- WebRtc_UWord16 fillerLength=0;
-
- WebRtc_UWord8 dataBuffer[WEBRTC_IP_PACKET_SIZE];
-
- while(bytesToSend>0)
- {
- fillerLength=maxLength;
- if(fillerLength<maxLength)
- {
- fillerLength = (WebRtc_UWord16) bytesToSend;
- }
-
- bytesToSend-=fillerLength;
-
- if (fillerLength > WEBRTC_IP_PACKET_SIZE - 12 - 1)
- {
- return 0;
- }
-
- if (fillerLength == 0)
- {
- // do not send an empty packet, will not reach JB
- fillerLength = 1;
- }
-
- // send paded data
- // correct seq num, time stamp and payloadtype
- _rtpSender.BuildRTPheader(dataBuffer, payloadType, false,captureTimestamp, true, true);
-
- // set filler NALU type
- dataBuffer[12] = 12; // NRI field = 0, type 12
-
- // send codec valid data, H.264 has defined data which is binary 1111111
- // fill with 0xff
- memset(dataBuffer + 12 + 1, 0xff, fillerLength-1);
-
- if( _rtpSender.SendToNetwork(dataBuffer,
- fillerLength,
- 12)<0)
- {
-
- return -1;;
- }
- }
- return 0;
-}
-
-WebRtc_Word32
-RTPSenderH264::SendH264SVCRelayPacket(const WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord8* incomingRTPPacket,
- const WebRtc_UWord16 incomingRTPPacketSize,
- const WebRtc_UWord32 ssrc,
- const bool higestLayer)
-{
- if (rtpHeader->header.sequenceNumber != (WebRtc_UWord16)(_h264SVCRelaySequenceNumber + 1))
- {
- // not continous, signal loss
- _rtpSender.IncrementSequenceNumber();
- }
- _h264SVCRelaySequenceNumber = rtpHeader->header.sequenceNumber;
-
-
- if (rtpHeader->header.timestamp != _h264SVCRelayTimeStamp)
- {
- // new frame
- _h264SVCRelayLayerComplete = false;
- }
-
- if (rtpHeader->header.timestamp == _h264SVCRelayTimeStamp &&
- _h264SVCRelayLayerComplete)
- {
- // sanity, end of layer already sent
- // Could happened for fragmented packet with missing PACSI info (PACSI packet reorded and received after packet it belongs to)
- // fragmented packet has no layer info set (default info 0)
- return 0;
- }
- _h264SVCRelayTimeStamp = rtpHeader->header.timestamp;
-
- // re-packetize H.264-SVC packets
- // we keep the timestap unchanged
- // make a copy and only change the SSRC and seqNum
-
- WebRtc_UWord8 dataBuffer[WEBRTC_IP_PACKET_SIZE];
- memcpy(dataBuffer, incomingRTPPacket, incomingRTPPacketSize);
-
- // _sequenceNumber initiated in Init()
- // _ssrc initiated in constructor
-
- // re-write payload type
- if(_h264SVCPayloadType != -1)
- {
- dataBuffer[1] &= kRtpMarkerBitMask;
- dataBuffer[1] += _h264SVCPayloadType;
- }
-
- // _sequenceNumber will not work for re-ordering by NACK from original sender
- // engine responsible for this
- ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+2, _rtpSender.IncrementSequenceNumber()); // get the current SequenceNumber and add by 1 after returning
- //ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, ssrc);
-
- // how do we know it's the last relayed packet in a frame?
- // 1) packets arrive in order, the engine manages that
- // 2) highest layer that we relay
- // 3) the end bit is set for the highest layer
-
- if(higestLayer && rtpHeader->type.Video.codecHeader.H264.relayE)
- {
- // set marker bit
- dataBuffer[1] |= kRtpMarkerBitMask;
-
- // set relayed layer as complete
- _h264SVCRelayLayerComplete = true;
- }
- return _rtpSender.SendToNetwork(dataBuffer,
- incomingRTPPacketSize - rtpHeader->header.headerLength,
- rtpHeader->header.headerLength);
-}
-
-WebRtc_Word32
-RTPSenderH264::SendH264_STAP_A(const FrameType frameType,
- const H264Info* ptrH264Info,
- WebRtc_UWord16 &idxNALU,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- bool& switchToFUA,
- WebRtc_Word32 &payloadBytesToSend,
- const WebRtc_UWord8*& data,
- const WebRtc_UWord16 rtpHeaderLength)
-{
- const WebRtc_Word32 H264_NALU_LENGTH = 2;
-
- WebRtc_UWord16 h264HeaderLength = 1; // normal header length
- WebRtc_UWord16 maxPayloadLengthSTAP_A = _rtpSender.MaxPayloadLength() -
- FECPacketOverhead() - rtpHeaderLength -
- h264HeaderLength - H264_NALU_LENGTH;
-
- WebRtc_Word32 dataOffset = rtpHeaderLength + h264HeaderLength;
- WebRtc_UWord8 NRI = 0;
- WebRtc_UWord16 payloadBytesInPacket = 0;
- WebRtc_UWord8 dataBuffer[WEBRTC_IP_PACKET_SIZE];
-
- if (ptrH264Info->payloadSize[idxNALU] > maxPayloadLengthSTAP_A)
- {
- // we need to fragment NAL switch to mode FU-A
- switchToFUA = true;
- } else
- {
- // combine as many NAL units in every IP packet
- do
- {
- if(!_h264SendPPS_SPS)
- {
- // don't send NALU of type 7 and 8 SPS and PPS
- if(ptrH264Info->type[idxNALU] == 7 || ptrH264Info->type[idxNALU] == 8)
- {
- payloadBytesToSend -= ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
- data += ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
- idxNALU++;
- continue;
- }
- }
- if(ptrH264Info->payloadSize[idxNALU] + payloadBytesInPacket <= maxPayloadLengthSTAP_A)
- {
- if(ptrH264Info->NRI[idxNALU] > NRI)
- {
- NRI = ptrH264Info->NRI[idxNALU];
- }
- // put NAL size into packet
- dataBuffer[dataOffset] = (WebRtc_UWord8)(ptrH264Info->payloadSize[idxNALU] >> 8);
- dataOffset++;
- dataBuffer[dataOffset] = (WebRtc_UWord8)(ptrH264Info->payloadSize[idxNALU] & 0xff);
- dataOffset++;
- // Put payload in packet
- memcpy(&dataBuffer[dataOffset], &data[ptrH264Info->startCodeSize[idxNALU]], ptrH264Info->payloadSize[idxNALU]);
- dataOffset += ptrH264Info->payloadSize[idxNALU];
- data += ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
- payloadBytesInPacket += (WebRtc_UWord16)(ptrH264Info->payloadSize[idxNALU] + H264_NALU_LENGTH);
- payloadBytesToSend -= ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
- } else
- {
- // we don't fitt the next NALU in this packet
- break;
- }
- idxNALU++;
- }while(payloadBytesToSend);
- }
-
- // sanity
- // don't send empty packets
- if (payloadBytesInPacket)
- {
- // add RTP header
- _rtpSender.BuildRTPheader(dataBuffer, payloadType, (payloadBytesToSend==0)?true:false, captureTimeStamp);
- dataBuffer[rtpHeaderLength] = 24 + NRI; // STAP-A == 24
- WebRtc_UWord16 payloadLength = payloadBytesInPacket + h264HeaderLength;
-
- if(-1 == SendVideoPacket(frameType, dataBuffer, payloadLength, rtpHeaderLength))
- {
- return -1;
- }
- }
- return 0;
-} // end STAP-A
-
-// STAP-A for H.264 SVC
-WebRtc_Word32
-RTPSenderH264::SendH264_STAP_A_PACSI(const FrameType frameType,
- const H264Info* ptrH264Info,
- WebRtc_UWord16 &idxNALU,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- bool& switchToFUA,
- WebRtc_Word32 &payloadBytesToSend,
- const WebRtc_UWord8*& data,
- const WebRtc_UWord16 rtpHeaderLength,
- WebRtc_UWord16& decodingOrderNumber)
-{
- const WebRtc_Word32 H264_NALU_LENGTH = 2;
-
- WebRtc_UWord16 h264HeaderLength = 1; // normal header length
- WebRtc_UWord16 maxPayloadLengthSTAP_A = _rtpSender.MaxPayloadLength() - FECPacketOverhead() - rtpHeaderLength - h264HeaderLength - H264_NALU_LENGTH;
- WebRtc_Word32 dataOffset = rtpHeaderLength + h264HeaderLength;
- WebRtc_UWord8 NRI = 0;
- WebRtc_UWord16 payloadBytesInPacket = 0;
- WebRtc_UWord8 dataBuffer[WEBRTC_IP_PACKET_SIZE];
- bool firstNALUNotIDR = true; //delta
-
- // Put PACSI NAL unit into packet
- WebRtc_Word32 lengthPACSI = 0;
- WebRtc_UWord32 PACSI_NALlength = ptrH264Info->PACSI[idxNALU].NALlength;
- if (PACSI_NALlength > maxPayloadLengthSTAP_A)
- {
- return -1;
- }
- dataBuffer[dataOffset++] = (WebRtc_UWord8)(PACSI_NALlength >> 8);
- dataBuffer[dataOffset++] = (WebRtc_UWord8)(PACSI_NALlength & 0xff);
-
- // end bit will be updated later, since another NALU in this packet might be the last
- WebRtc_Word32 lengthPASCINALU = AddH264PACSINALU(true,
- false,
- ptrH264Info->PACSI[idxNALU],
- ptrH264Info->SVCheader[idxNALU],
- decodingOrderNumber,
- dataBuffer,
- dataOffset);
- if (lengthPASCINALU <= 0)
- {
- return -1;
- }
- decodingOrderNumber++;
-
- lengthPACSI = H264_NALU_LENGTH + lengthPASCINALU;
- maxPayloadLengthSTAP_A -= (WebRtc_UWord16)lengthPACSI;
- if (ptrH264Info->payloadSize[idxNALU] > maxPayloadLengthSTAP_A)
- {
- // we need to fragment NAL switch to mode FU-A
- switchToFUA = true;
- return 0;
- }
- if(!ptrH264Info->SVCheader[idxNALU].idr)
- {
- firstNALUNotIDR = true;
- }
-
- WebRtc_UWord32 layer = (ptrH264Info->SVCheader[idxNALU].dependencyID << 16)+
- (ptrH264Info->SVCheader[idxNALU].qualityID << 8) +
- ptrH264Info->SVCheader[idxNALU].temporalID;
-
- {
- // combine as many NAL units in every IP packet, with the same priorityID
- // Improvement we could allow several very small MGS NALU from different layers to be sent in one packet
-
- do
- {
- if(!_h264SendPPS_SPS)
- {
- // Don't send NALU of type 7 and 8 SPS and PPS,
- // they could be signaled outofband
- if(ptrH264Info->type[idxNALU] == 7 || ptrH264Info->type[idxNALU] == 8)
- {
- payloadBytesToSend -= ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
- data += ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
- idxNALU++;
- continue;
- }
- }
- // don't send NALU type 6 (SEI message) not allowed when we send it in PACSI
- if(ptrH264Info->type[idxNALU] == 6)
- {
- // SEI NALU Don't send, not allowed when we send it in PACSI
- payloadBytesToSend -= ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
- data += ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
- idxNALU++;
- continue;
- }
-
- const WebRtc_UWord32 layerNALU = (ptrH264Info->SVCheader[idxNALU].dependencyID << 16)+
- (ptrH264Info->SVCheader[idxNALU].qualityID << 8) +
- ptrH264Info->SVCheader[idxNALU].temporalID;
-
- // we need to break on a new layer
- if( ptrH264Info->payloadSize[idxNALU] + payloadBytesInPacket <= maxPayloadLengthSTAP_A &&
- layerNALU == layer)
- {
- if(ptrH264Info->NRI[idxNALU] > NRI)
- {
- NRI = ptrH264Info->NRI[idxNALU];
- }
- // put NAL size into packet
- dataBuffer[dataOffset] = (WebRtc_UWord8)(ptrH264Info->payloadSize[idxNALU] >> 8);
- dataOffset++;
- dataBuffer[dataOffset] = (WebRtc_UWord8)(ptrH264Info->payloadSize[idxNALU] & 0xff);
- dataOffset++;
- // Put payload in packet
- memcpy(&dataBuffer[dataOffset], &data[ptrH264Info->startCodeSize[idxNALU]], ptrH264Info->payloadSize[idxNALU]);
- dataOffset += ptrH264Info->payloadSize[idxNALU];
- data += ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
- payloadBytesInPacket += (WebRtc_UWord16)(ptrH264Info->payloadSize[idxNALU] + H264_NALU_LENGTH);
- payloadBytesToSend -= ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
- } else
- {
- // we don't fitt the next NALU in this packet or,
- // it's the next layer
-
- // check if we should send this NALU
- // based on the layer
-
- if(_useHighestSendLayer && layerNALU != layer)
- {
- // we don't send this NALU due to it's a new layer
- // check if we should send the next or if this is the last
- const WebRtc_UWord8 dependencyQualityID = (ptrH264Info->SVCheader[idxNALU].dependencyID << 4) + ptrH264Info->SVCheader[idxNALU].qualityID;
-
- bool highestLayer;
- if(SendH264SVCLayer(frameType,
- ptrH264Info->SVCheader[idxNALU].temporalID,
- dependencyQualityID,
- highestLayer) == false)
- {
- // will trigger markerbit and stop sending this frame
- payloadBytesToSend = 0;
- }
- }
- break;
- }
- idxNALU++;
-
- }while(payloadBytesToSend);
- }
-
- // sanity, don't send empty packets
- if (payloadBytesInPacket)
- {
- // add RTP header
- _rtpSender.BuildRTPheader(dataBuffer, payloadType, (payloadBytesToSend==0)?true:false, captureTimeStamp);
-
- dataBuffer[rtpHeaderLength] = 24 + NRI; // STAP-A == 24
-
- // NRI for PACSI
- dataBuffer[rtpHeaderLength + H264_NALU_LENGTH + 1] &= 0x1f; // zero out NRI field
- dataBuffer[rtpHeaderLength + H264_NALU_LENGTH + 1] |= NRI;
-
- if(ptrH264Info->PACSI[idxNALU-1].E)
- {
- // update end bit
- dataBuffer[rtpHeaderLength + H264_NALU_LENGTH + 5] |= 0x01;
- }
- if(firstNALUNotIDR)
- {
- // we have to check if any of the NALU in this packet is an IDR NALU
- bool setIBit = false;
- for(int i = 0; i < idxNALU; i++)
- {
- if(ptrH264Info->SVCheader[i].idr)
- {
- setIBit = true;
- break;
- }
- }
- if(setIBit)
- {
- // update I bit
- dataBuffer[rtpHeaderLength + H264_NALU_LENGTH + 2] |= 0x40;
- }
- }
- const WebRtc_UWord16 payloadLength = payloadBytesInPacket + h264HeaderLength + (WebRtc_UWord16)lengthPACSI;
- if(-1 == SendVideoPacket(frameType,
- dataBuffer,
- payloadLength,
- rtpHeaderLength,
- layer==0))
- {
- return -1;
- }
- }
- return 0;
-} // end STAP-A
-
-WebRtc_Word32
-RTPSenderH264::SendH264_FU_A(const FrameType frameType,
- const H264Info* ptrH264Info,
- WebRtc_UWord16 &idxNALU,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- WebRtc_Word32 &payloadBytesToSend,
- const WebRtc_UWord8*& data,
- const WebRtc_UWord16 rtpHeaderLength,
- WebRtc_UWord16& decodingOrderNumber,
- const bool sendSVCPACSI)
-{
-
- // FUA for the rest of the frame
- WebRtc_UWord16 maxPayloadLength = _rtpSender.MaxPayloadLength() - FECPacketOverhead() - rtpHeaderLength;
- WebRtc_UWord8 dataBuffer[WEBRTC_IP_PACKET_SIZE];
- WebRtc_UWord32 payloadBytesRemainingInNALU = ptrH264Info->payloadSize[idxNALU];
-
- bool isBaseLayer=false;
-
- if(payloadBytesRemainingInNALU > maxPayloadLength)
- {
- // we need to fragment NALU
- const WebRtc_UWord16 H264_FUA_LENGTH = 2; // FU-a H.264 header is 2 bytes
-
- if(sendSVCPACSI)
- {
- SendH264_SinglePACSI(frameType,
- ptrH264Info,
- idxNALU,
- payloadType,
- captureTimeStamp,
- true,
- false);
-
- WebRtc_UWord32 layer = (ptrH264Info->SVCheader[idxNALU].dependencyID << 16)+
- (ptrH264Info->SVCheader[idxNALU].qualityID << 8) +
- ptrH264Info->SVCheader[idxNALU].temporalID;
- isBaseLayer=(layer==0);
- }
-
- // First packet
- _rtpSender.BuildRTPheader(dataBuffer,payloadType, false, captureTimeStamp);
-
- WebRtc_UWord16 maxPayloadLengthFU_A = maxPayloadLength - H264_FUA_LENGTH ;
- WebRtc_UWord8 fuaIndc = 28 + ptrH264Info->NRI[idxNALU];
- dataBuffer[rtpHeaderLength] = fuaIndc; // FU-A indicator
- dataBuffer[rtpHeaderLength+1] = (WebRtc_UWord8)(ptrH264Info->type[idxNALU] + 0x80)/*start*/; // FU-A header
-
- memcpy(&dataBuffer[rtpHeaderLength + H264_FUA_LENGTH], &data[ptrH264Info->startCodeSize[idxNALU]+1], maxPayloadLengthFU_A);
- WebRtc_UWord16 payloadLength = maxPayloadLengthFU_A + H264_FUA_LENGTH;
- if(-1 == SendVideoPacket(frameType, dataBuffer, payloadLength, rtpHeaderLength, isBaseLayer))
- {
- return -1;
- }
-
- //+1 is from the type that is coded into the FU-a header
- data += maxPayloadLengthFU_A + 1 + ptrH264Info->startCodeSize[idxNALU]; // inc data ptr
- payloadBytesToSend -= maxPayloadLengthFU_A+1+ptrH264Info->startCodeSize[idxNALU];
- payloadBytesRemainingInNALU -= maxPayloadLengthFU_A+1;
-
- // all non first/last packets
- while(payloadBytesRemainingInNALU > maxPayloadLengthFU_A)
- {
- if(sendSVCPACSI)
- {
- SendH264_SinglePACSI(frameType,
- ptrH264Info,
- idxNALU,
- payloadType,
- captureTimeStamp,
- false,
- false);
- }
-
- // prepare next header
- _rtpSender.BuildRTPheader(dataBuffer, payloadType, false, captureTimeStamp);
-
- dataBuffer[rtpHeaderLength] = (WebRtc_UWord8)fuaIndc; // FU-A indicator
- dataBuffer[rtpHeaderLength+1] = ptrH264Info->type[idxNALU]; // FU-A header
-
- memcpy(&dataBuffer[rtpHeaderLength+H264_FUA_LENGTH], data, maxPayloadLengthFU_A);
- payloadLength = maxPayloadLengthFU_A + H264_FUA_LENGTH;
-
- if(-1 == SendVideoPacket(frameType, dataBuffer, payloadLength, rtpHeaderLength,isBaseLayer))
- {
- return -1;
- }
- data += maxPayloadLengthFU_A; // inc data ptr
- payloadBytesToSend -= maxPayloadLengthFU_A;
- payloadBytesRemainingInNALU -= maxPayloadLengthFU_A;
- dataBuffer[rtpHeaderLength] = fuaIndc; // FU-A indicator
- dataBuffer[rtpHeaderLength+1] = ptrH264Info->type[idxNALU]; // FU-A header
- }
- if(sendSVCPACSI)
- {
- SendH264_SinglePACSI(frameType,
- ptrH264Info,
- idxNALU,
- payloadType,
- captureTimeStamp,
- false,
- true); // last packet in NALU
-
- if(_useHighestSendLayer && idxNALU+1 < ptrH264Info->numNALUs)
- {
- // not last NALU in frame
- // check if it's the the next layer should not be sent
-
- // check if we should send the next or if this is the last
- const WebRtc_UWord8 dependencyQualityID = (ptrH264Info->SVCheader[idxNALU+1].dependencyID << 4) +
- ptrH264Info->SVCheader[idxNALU+1].qualityID;
-
- bool highestLayer;
- if(SendH264SVCLayer(frameType,
- ptrH264Info->SVCheader[idxNALU+1].temporalID,
- dependencyQualityID,
- highestLayer) == false)
- {
- // will trigger markerbit and stop sending this frame
- payloadBytesToSend = payloadBytesRemainingInNALU;
- }
- }
- }
- // last packet in NALU
- _rtpSender.BuildRTPheader(dataBuffer, payloadType,(payloadBytesToSend == (WebRtc_Word32)payloadBytesRemainingInNALU)?true:false, captureTimeStamp);
- dataBuffer[rtpHeaderLength+1] = ptrH264Info->type[idxNALU] + 0x40/*stop*/; // FU-A header
-
- memcpy(&dataBuffer[rtpHeaderLength+H264_FUA_LENGTH], data, payloadBytesRemainingInNALU);
- payloadLength = (WebRtc_UWord16)payloadBytesRemainingInNALU + H264_FUA_LENGTH;
- payloadBytesToSend -= payloadBytesRemainingInNALU;
- if(payloadBytesToSend != 0)
- {
- data += payloadBytesRemainingInNALU; // inc data ptr
- }
- idxNALU++;
- if(-1 == SendVideoPacket(frameType, dataBuffer, payloadLength, rtpHeaderLength,isBaseLayer))
- {
- return -1;
- }
- } else
- {
- // send NAL unit in singel mode
- return SendH264_SingleMode(frameType,
- ptrH264Info,
- idxNALU,
- payloadType,
- captureTimeStamp,
- payloadBytesToSend,
- data,
- rtpHeaderLength,
- sendSVCPACSI);
- }
- // end FU-a
- return 0;
-}
-
-WebRtc_Word32
-RTPSenderH264::SendH264_SingleMode(const FrameType frameType,
- const H264Info* ptrH264Info,
- WebRtc_UWord16 &idxNALU,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- WebRtc_Word32 &payloadBytesToSend,
- const WebRtc_UWord8*& data,
- const WebRtc_UWord16 rtpHeaderLength,
- WebRtc_UWord16& decodingOrderNumber,
- const bool sendSVCPACSI)
-{
- // no H.264 header lenght in single mode
- // we use WEBRTC_IP_PACKET_SIZE instead of the configured MTU since it's better to send fragmented UDP than not to send
- const WebRtc_UWord16 maxPayloadLength = WEBRTC_IP_PACKET_SIZE - _rtpSender.PacketOverHead() - FECPacketOverhead() - rtpHeaderLength;
- WebRtc_UWord8 dataBuffer[WEBRTC_IP_PACKET_SIZE];
- bool isBaseLayer=false;
-
- if(ptrH264Info->payloadSize[idxNALU] > maxPayloadLength)
- {
- return -3;
- }
- if(!_h264SendPPS_SPS)
- {
- // don't send NALU of type 7 and 8 SPS and PPS
- if(ptrH264Info->type[idxNALU] == 7 || ptrH264Info->type[idxNALU] == 8)
- {
- payloadBytesToSend -= ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
- data += ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
- idxNALU++;
- return 0;
- }
- }
- if(sendSVCPACSI)
- {
- SendH264_SinglePACSI(frameType,
- ptrH264Info,
- idxNALU,
- payloadType,
- captureTimeStamp,
- true,
- true);
-
- WebRtc_UWord32 layer = (ptrH264Info->SVCheader[idxNALU].dependencyID << 16)+
- (ptrH264Info->SVCheader[idxNALU].qualityID << 8) +
- ptrH264Info->SVCheader[idxNALU].temporalID;
- isBaseLayer=(layer==0);
- }
-
- // Put payload in packet
- memcpy(&dataBuffer[rtpHeaderLength], &data[ptrH264Info->startCodeSize[idxNALU]], ptrH264Info->payloadSize[idxNALU]);
-
- WebRtc_UWord16 payloadBytesInPacket = (WebRtc_UWord16)ptrH264Info->payloadSize[idxNALU];
- payloadBytesToSend -= ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU]; // left to send
-
- //
- _rtpSender.BuildRTPheader(dataBuffer,payloadType,(payloadBytesToSend ==0)?true:false, captureTimeStamp);
-
- dataBuffer[rtpHeaderLength] &= 0x1f; // zero out NRI field
- dataBuffer[rtpHeaderLength] |= ptrH264Info->NRI[idxNALU]; // nri
- if(payloadBytesToSend > 0)
- {
- data += ptrH264Info->payloadSize[idxNALU] + ptrH264Info->startCodeSize[idxNALU];
- }
- idxNALU++;
- if(-1 == SendVideoPacket(frameType, dataBuffer, payloadBytesInPacket, rtpHeaderLength,isBaseLayer))
- {
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32
-RTPSenderH264::SendH264_SinglePACSI(const FrameType frameType,
- const H264Info* ptrH264Info,
- const WebRtc_UWord16 idxNALU,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- const bool firstPacketInNALU,
- const bool lastPacketInNALU);
-{
- // Send PACSI in single mode
- WebRtc_UWord8 dataBuffer[WEBRTC_IP_PACKET_SIZE];
- WebRtc_UWord16 rtpHeaderLength = (WebRtc_UWord16)_rtpSender.BuildRTPheader(dataBuffer, payloadType,false, captureTimeStamp);
- WebRtc_Word32 dataOffset = rtpHeaderLength;
-
- WebRtc_Word32 lengthPASCINALU = AddH264PACSINALU(firstPacketInNALU,
- lastPacketInNALU,
- ptrH264Info->PACSI[idxNALU],
- ptrH264Info->SVCheader[idxNALU],
- decodingOrderNumber,
- dataBuffer,
- dataOffset);
-
- if (lengthPASCINALU <= 0)
- {
- return -1;
- }
- decodingOrderNumber++;
-
- WebRtc_UWord16 payloadBytesInPacket = (WebRtc_UWord16)lengthPASCINALU;
-
- // Set payload header (first payload byte co-serves as the payload header)
- dataBuffer[rtpHeaderLength] &= 0x1f; // zero out NRI field
- dataBuffer[rtpHeaderLength] |= ptrH264Info->NRI[idxNALU]; // nri
-
- const WebRtc_UWord32 layer = (ptrH264Info->SVCheader[idxNALU].dependencyID << 16)+
- (ptrH264Info->SVCheader[idxNALU].qualityID << 8) +
- ptrH264Info->SVCheader[idxNALU].temporalID;
-
- if (-1 == SendVideoPacket(frameType, dataBuffer, payloadBytesInPacket, rtpHeaderLength,layer==0))
- {
- return -1;
- }
- return 0;
-}
-
-
-
-
-WebRtc_Word32
-RTPSenderH264::SendH264SVC(const FrameType frameType,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- H264Information& h264Information,
- WebRtc_UWord16& decodingOrderNumber)
-{
- WebRtc_Word32 payloadBytesToSend = payloadSize;
- const WebRtc_UWord16 rtpHeaderLength = _rtpSender.RTPHeaderLength();
-
- const H264Info* ptrH264Info = NULL;
- if (h264Information.GetInfo(payloadData,payloadSize, ptrH264Info) == -1)
- {
- return -1;
- }
- if(_useHighestSendLayer)
- {
- // we need to check if we should drop the frame
- // it could be a temporal layer (aka a temporal frame)
- const WebRtc_UWord8 dependencyQualityID = (ptrH264Info->SVCheader[0].dependencyID << 4) + ptrH264Info->SVCheader[0].qualityID;
-
- bool dummyHighestLayer;
- if(SendH264SVCLayer(frameType,
- ptrH264Info->SVCheader[0].temporalID,
- dependencyQualityID,
- dummyHighestLayer) == false)
- {
- // skip send this frame
- return 0;
- }
- }
-
- WebRtc_UWord16 idxNALU = 0;
- while (payloadBytesToSend > 0)
- {
- bool switchToFUA = false;
- if (SendH264_STAP_A_PACSI(frameType,
- ptrH264Info,
- idxNALU,
- payloadType,
- captureTimeStamp,
- switchToFUA,
- payloadBytesToSend,
- payloadData,
- rtpHeaderLength,
- decodingOrderNumber) != 0)
- {
- return -1;
- }
- if(switchToFUA)
- {
- // FU_A for this NALU
- if (SendH264_FU_A(frameType,
- ptrH264Info,
- idxNALU,
- payloadType,
- captureTimeStamp,
- payloadBytesToSend,
- payloadData,
- rtpHeaderLength,
- true) != 0)
- {
- return -1;
- }
- }
- }
- return 0;
-}
-
-WebRtc_Word32
-RTPSenderH264::SetH264PacketizationMode(const H264PacketizationMode mode)
-{
- _h264Mode = mode;
- return 0;
-}
-
-WebRtc_Word32
-RTPSenderH264::SetH264SendModeNALU_PPS_SPS(const bool dontSend)
-{
- _h264SendPPS_SPS = !dontSend;
- return 0;
-}
-
-bool
-RTPSenderH264::SendH264SVCLayer(const FrameType frameType,
- const WebRtc_UWord8 temporalID,
- const WebRtc_UWord8 dependencyQualityID,
- bool& higestLayer)
-{
- WebRtc_UWord8 dependencyID = dependencyQualityID >> 4;
-
- // keyframe required to switch between dependency layers not quality and temporal
- if( _highestDependencyLayer != _highestDependencyLayerOld)
- {
- // we want to switch dependency layer
- if(frameType == kVideoFrameKey)
- {
- // key frame we can change layer if it's correct layer
- if(_highestDependencyLayer > _highestDependencyLayerOld)
- {
- // we want to switch up
- // does this packet belong to a new layer?
-
- if( dependencyID > _highestDependencyLayerOld &&
- dependencyID <= _highestDependencyLayer)
- {
- _highestDependencyLayerOld = dependencyID;
- _highestDependencyQualityIDOld = _highestDependencyQualityID;
-
- if( dependencyID == _highestDependencyLayer &&
- dependencyQualityID == _highestDependencyQualityID)
- {
- higestLayer = true;
- }
- // relay
- return true;
- }
- }
- if(_highestDependencyLayer < _highestDependencyLayerOld)
- {
- // we want to switch down
- // does this packet belong to a low layer?
- if( dependencyID <= _highestDependencyLayer)
- {
- _highestDependencyLayerOld = dependencyID;
- _highestDependencyQualityIDOld = _highestDependencyQualityID;
- if( dependencyID == _highestDependencyLayer &&
- dependencyQualityID == _highestDependencyQualityID)
- {
- higestLayer = true;
- }
- // relay
- return true;
- }
- }
- } else
- {
- // Delta frame and we are waiting to switch dependency layer
- if(_highestDependencyLayer > _highestDependencyLayerOld)
- {
- // we want to switch up to a higher dependency layer
- // use old setting until we get a key-frame
-
- // filter based on old dependency
- // we could have allowed to add a MGS layer lower than the dependency ID
- // but then we can't know the highest layer relayed we assume that the user
- // will add one layer at a time
- if( _highestTemporalLayer < temporalID ||
- _highestDependencyLayerOld < dependencyID ||
- _highestDependencyQualityIDOld < dependencyQualityID)
- {
- // drop
- return false;
- }
- // highest layer based on old
- if( dependencyID == _highestDependencyLayerOld &&
- dependencyQualityID == _highestDependencyQualityIDOld)
- {
- higestLayer = true;
- }
- } else
- {
- // we want to switch down to a lower dependency layer,
- // use old setting, done bellow
- // drop all temporal layers while waiting for the key-frame
- if(temporalID > 0)
- {
- // drop
- return false;
- }
- // we can't drop a lower MGS layer since this might depend on it
- // however we can drop MGS layers larger than dependecyQualityId
- // with dependency from old and quality 0
- if( _highestDependencyLayerOld < dependencyID ||
- (_highestDependencyQualityIDOld & 0xf0) < dependencyQualityID)
- {
- // drop
- return false;
- }
- if( dependencyID == _highestDependencyLayerOld &&
- dependencyQualityID == (_highestDependencyQualityIDOld & 0xf0))
- {
- higestLayer = true;
- }
- }
- }
- } else
- {
- // filter based on current state
- if( _highestTemporalLayer < temporalID ||
- _highestDependencyLayer < dependencyID ||
- _highestDependencyQualityID < dependencyQualityID)
- {
- // drop
- return false;
- }
- if( dependencyID == _highestDependencyLayer &&
- dependencyQualityID == _highestDependencyQualityID)
- {
- higestLayer = true;
- }
- }
- return true;
-}
-
-WebRtc_Word32
-RTPSenderH264::SetHighestSendLayer(const WebRtc_UWord8 dependencyQualityLayer,
- const WebRtc_UWord8 temporalLayer)
-{
- const WebRtc_UWord8 dependencyLayer = (dependencyQualityLayer >> 4);
-
- if(_highestDependencyLayerOld != _highestDependencyLayer)
- {
- // we have not switched to the new dependency yet
- } else
- {
- if(_highestDependencyLayer == dependencyLayer)
- {
- // no change of dependency
- // switch now _highestDependencyQualityIDOld
- _highestDependencyQualityIDOld = dependencyQualityLayer;
- }else
- {
- // change of dependency, update _highestDependencyQualityIDOld store as old
- _highestDependencyQualityIDOld = _highestDependencyQualityID;
- }
- }
- _useHighestSendLayer = true;
- _highestDependencyLayer = dependencyLayer;
- _highestDependencyQualityID = dependencyQualityLayer;
- _highestTemporalLayer = temporalLayer;
- return 0;
-}
-
-WebRtc_Word32
-RTPSenderH264::HighestSendLayer(WebRtc_UWord8& dependencyQualityLayer,
- WebRtc_UWord8& temporalLayer)
-{
- if (!_useHighestSendLayer)
- {
- // No information set
- return -1;
- }
- dependencyQualityLayer = _highestDependencyQualityID;
- temporalLayer = _highestTemporalLayer;
- return 0;
-}
-/*
-* H.264
-*/
-WebRtc_Word32
-RTPSenderH264::SendH264(const FrameType frameType,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- H264Information& h264Information)
-{
- WebRtc_Word32 payloadBytesToSend = payloadSize;
- const WebRtc_UWord8* data = payloadData;
- bool switchToFUA = false;
- const WebRtc_UWord16 rtpHeaderLength = _rtpSender.RTPHeaderLength();
-
- const H264Info* ptrH264Info = NULL;
- if (h264Information.GetInfo(payloadData,payloadSize, ptrH264Info) == -1)
- {
- return -1;
- }
- WebRtc_UWord16 idxNALU = 0;
- WebRtc_UWord16 DONCdummy = 0;
-
- while (payloadBytesToSend > 0)
- {
- switch(_h264Mode)
- {
- case H264_NON_INTERLEAVED_MODE:
-
- if(!switchToFUA)
- {
- if(SendH264_STAP_A(frameType,
- ptrH264Info,
- idxNALU,
- payloadType,
- captureTimeStamp,
- switchToFUA,
- payloadBytesToSend,
- data,
- rtpHeaderLength) != 0)
- {
- return -1;
- }
- }
- else
- {
- // FUA for the rest of the frame
- if(SendH264_FU_A(frameType,
- ptrH264Info,
- idxNALU,
- payloadType,
- captureTimeStamp,
- payloadBytesToSend,
- data,
- rtpHeaderLength,
- DONCdummy) != 0)
- {
- return -1;
- }
- // try to go back to STAP_A
- switchToFUA = false;
- }
- break;
- case H264_SINGLE_NAL_MODE:
- {
- // modeSingleU
- if(SendH264_SingleMode(frameType,
- ptrH264Info,
- idxNALU,
- payloadType,
- captureTimeStamp,
- payloadBytesToSend,
- data,
- rtpHeaderLength,
- DONCdummy) != 0)
- {
- return -1;
- }
- break;
- }
- case H264_INTERLEAVED_MODE:
- // not supported
- assert(false);
- return -1;
- }
- }
- return 0;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/H264/rtp_sender_h264.h b/trunk/src/modules/rtp_rtcp/source/H264/rtp_sender_h264.h
deleted file mode 100644
index 564b870..0000000
--- a/trunk/src/modules/rtp_rtcp/source/H264/rtp_sender_h264.h
+++ /dev/null
@@ -1,179 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_RTP_SENDER_H264_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_RTP_SENDER_H264_H_
-
-#include "typedefs.h"
-#include "ModuleRTPRTCPConfig.h"
-#include "rtp_rtcp_defines.h"
-#include "h264_information.h"
-
-#include "RTPSender.h"
-
-namespace webrtc {
-class RTPSenderH264
-{
-public:
- WebRtc_Word32 SendH264(const FrameType frameType,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- H264Information& h264Information);
-
- WebRtc_Word32 SendH264SVC(const FrameType frameType,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- H264Information& h264Information);
-
- // H.264 AVC
- WebRtc_Word32 SetH264PacketizationMode(const H264PacketizationMode mode);
-
- WebRtc_Word32 SetH264SendModeNALU_PPS_SPS(const bool dontSend);
-
- // H.264 SVC
- WebRtc_Word32 SetHighestSendLayer(const WebRtc_UWord8 dependencyQualityLayer,
- const WebRtc_UWord8 temporalLayer);
-
- WebRtc_Word32 HighestSendLayer(WebRtc_UWord8& dependencyQualityLayer,
- WebRtc_UWord8& temporalLayer);
-
-protected:
- RTPSenderH264(RTPSenderInterface* rtpSender);
- virtual ~RTPSenderH264();
-
- WebRtc_Word32 Init();
-
- virtual WebRtc_UWord16 FECPacketOverhead() const = 0;
- virtual RtpVideoCodecTypes VideoCodecType() const = 0;
-
- virtual WebRtc_Word32 SendVideoPacket(const FrameType frameType,
- const WebRtc_UWord8* dataBuffer,
- const WebRtc_UWord16 payloadLength,
- const WebRtc_UWord16 rtpHeaderLength,
- bool baseLayerVideoPacket=false) = 0;
-
-
- bool SendH264SVCLayer(const FrameType frameType,
- const WebRtc_UWord8 temporalID,
- const WebRtc_UWord8 dependencyQualityID,
- bool& higestLayer);
-
- // H.264 SVC
- WebRtc_Word32 AddH264PACSINALU(const bool firstPacketInNALU,
- const bool lastPacketInNALU,
- const H264_PACSI_NALU& paci,
- const H264_SVC_NALUHeader& svc,
- const WebRtc_UWord16 DONC,
- WebRtc_UWord8* databuffer,
- WebRtc_Word32& curByte) const;
-
- WebRtc_Word32 SendH264FillerData(const WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord16 bytesToSend,
- const WebRtc_UWord32 ssrc);
-
- WebRtc_Word32 SendH264FillerData(const WebRtc_UWord32 captureTimestamp,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 bytesToSend);
-
- WebRtc_Word32 SendH264SVCRelayPacket(const WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord8* incomingRTPPacket,
- const WebRtc_UWord16 incomingRTPPacketSize,
- const WebRtc_UWord32 ssrc,
- const bool higestLayer);
-
- WebRtc_Word32 SetH264RelaySequenceNumber(const WebRtc_UWord16 seqNum);
-
- WebRtc_Word32 SetH264RelayCompleteLayer(const bool complete);
-
- // H.264
- H264PacketizationMode _h264Mode;
- bool _h264SendPPS_SPS;
-
- // H.264-SVC
- WebRtc_Word8 _h264SVCPayloadType;
- WebRtc_UWord16 _h264SVCRelaySequenceNumber;
- WebRtc_UWord32 _h264SVCRelayTimeStamp;
- bool _h264SVCRelayLayerComplete;
-
-
-private:
- // H.264
- WebRtc_Word32 SendH264_SingleMode(const FrameType frameType,
- const H264Info* ptrH264Info,
- WebRtc_UWord16 &idxNALU,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- WebRtc_Word32 &payloadBytesToSend,
- const WebRtc_UWord8*& data,
- const WebRtc_UWord16 rtpHeaderLength,
- const bool sendSVCPACSI=false);
-
- WebRtc_Word32 SendH264_FU_A(const FrameType frameType,
- const H264Info* ptrH264Info,
- WebRtc_UWord16 &idxNALU,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- WebRtc_Word32 &payloadBytesToSend,
- const WebRtc_UWord8*& data,
- const WebRtc_UWord16 rtpHeaderLength,
- const bool sendSVCPACSI = false);
-
- WebRtc_Word32 SendH264_STAP_A(const FrameType frameType,
- const H264Info* ptrH264Info,
- WebRtc_UWord16 &idxNALU,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- bool& switchToFUA,
- WebRtc_Word32 &payloadBytesToSend,
- const WebRtc_UWord8*& data,
- const WebRtc_UWord16 rtpHeaderLength);
-
- WebRtc_Word32 SendH264_STAP_A_PACSI(const FrameType frameType,
- const H264Info* ptrH264Info,
- WebRtc_UWord16 &idxNALU,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- bool& switchToFUA,
- WebRtc_Word32 &payloadBytesToSend,
- const WebRtc_UWord8*& data,
- const WebRtc_UWord16 rtpHeaderLengh)
-
- WebRtc_Word32 SendH264_SinglePACSI(const FrameType frameType,
- const H264Info* ptrH264Info,
- const WebRtc_UWord16 idxNALU,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- const bool firstPacketInNALU,
- const bool lastPacketInNALU);
-
- bool AddH264SVCNALUHeader(const H264_SVC_NALUHeader& svc,
- WebRtc_UWord8* databuffer,
- WebRtc_Word32& curByte) const;
-
- RTPSenderInterface& _rtpSender;
-
- // relay
- bool _useHighestSendLayer;
- WebRtc_UWord8 _highestDependencyLayerOld;
- WebRtc_UWord8 _highestDependencyQualityIDOld;
- WebRtc_UWord8 _highestDependencyLayer;
- WebRtc_UWord8 _highestDependencyQualityID;
- WebRtc_UWord8 _highestTemporalLayer;
-
-
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_H264_RTP_SENDER_H264_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/bandwidth_management.cc b/trunk/src/modules/rtp_rtcp/source/bandwidth_management.cc
deleted file mode 100644
index a14a2ef..0000000
--- a/trunk/src/modules/rtp_rtcp/source/bandwidth_management.cc
+++ /dev/null
@@ -1,316 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "bandwidth_management.h"
-#include "trace.h"
-#include "rtp_utility.h"
-#include "rtp_rtcp_config.h"
-
-#include <math.h> // sqrt()
-
-namespace webrtc {
-
-BandwidthManagement::BandwidthManagement(const WebRtc_Word32 id) :
- _id(id),
- _critsect(CriticalSectionWrapper::CreateCriticalSection()),
- _lastPacketLossExtendedHighSeqNum(0),
- _lastReportAllLost(false),
- _lastLoss(0),
- _accumulateLostPacketsQ8(0),
- _accumulateExpectedPackets(0),
- _bitRate(0),
- _minBitRateConfigured(0),
- _maxBitRateConfigured(0),
- _last_fraction_loss(0),
- _last_round_trip_time(0),
- _bwEstimateIncoming(0),
- _smoothedFractionLostQ4(-1), // indicate uninitialized
- _sFLFactorQ4(14), // 0.875 in Q4
- _timeLastIncrease(0)
-{
-}
-
-BandwidthManagement::~BandwidthManagement()
-{
- delete _critsect;
-}
-
-void
-BandwidthManagement::SetSendBitrate(const WebRtc_UWord32 startBitrate,
- const WebRtc_UWord16 minBitrateKbit,
- const WebRtc_UWord16 maxBitrateKbit)
-{
- CriticalSectionScoped cs(_critsect);
-
- _bitRate = startBitrate;
- _minBitRateConfigured = minBitrateKbit*1000;
- if(maxBitrateKbit == 0)
- {
- // no max configured use 1Gbit/s
- _maxBitRateConfigured = 1000000000;
- } else
- {
- _maxBitRateConfigured = maxBitrateKbit*1000;
- }
-}
-
-WebRtc_Word32
-BandwidthManagement::MaxConfiguredBitrate(WebRtc_UWord16* maxBitrateKbit)
-{
- CriticalSectionScoped cs(_critsect);
-
- if(_maxBitRateConfigured == 0)
- {
- return -1;
- }
- *maxBitrateKbit = (WebRtc_UWord16)(_maxBitRateConfigured/1000);
- return 0;
-}
-
-WebRtc_Word32
-BandwidthManagement::UpdateBandwidthEstimate(const WebRtc_UWord16 bandWidthKbit,
- WebRtc_UWord32* newBitrate,
- WebRtc_UWord8* fractionLost,
- WebRtc_UWord16* roundTripTime)
-{
- *newBitrate = 0;
- CriticalSectionScoped cs(_critsect);
-
- _bwEstimateIncoming = bandWidthKbit*1000;
-
- if(_bitRate == 0)
- {
- // BandwidthManagement off
- return -1;
- }
- if (_bwEstimateIncoming > 0 && _bitRate > _bwEstimateIncoming)
- {
- _bitRate = _bwEstimateIncoming;
- } else
- {
- return -1;
- }
- *newBitrate = _bitRate;
- *fractionLost = _last_fraction_loss;
- *roundTripTime = _last_round_trip_time;
- return 0;
-}
-
-WebRtc_Word32 BandwidthManagement::UpdatePacketLoss(
- const WebRtc_UWord32 lastReceivedExtendedHighSeqNum,
- WebRtc_UWord32 sentBitrate,
- const WebRtc_UWord16 rtt,
- WebRtc_UWord8* loss,
- WebRtc_UWord32* newBitrate,
- WebRtc_Word64 nowMS)
-{
- CriticalSectionScoped cs(_critsect);
-
- _last_fraction_loss = *loss;
- _last_round_trip_time = rtt;
-
- if(_bitRate == 0)
- {
- // BandwidthManagement off
- return -1;
- }
-
- // Check sequence number diff and weight loss report
- if (_lastPacketLossExtendedHighSeqNum > 0 &&
- (lastReceivedExtendedHighSeqNum >= _lastPacketLossExtendedHighSeqNum))
- {
- // This is not the first loss report and the sequence number is
- // non-decreasing. Calculate sequence number diff.
- WebRtc_UWord32 seqNumDiff = lastReceivedExtendedHighSeqNum
- - _lastPacketLossExtendedHighSeqNum;
-
- // Check if this report and the last was 100% loss, then report
- // 100% loss even though seqNumDiff is small.
- // If not, go on with the checks.
- if (!(_lastReportAllLost && *loss == 255))
- {
- _lastReportAllLost = (*loss == 255);
-
- // Calculate number of lost packets.
- // loss = 256 * numLostPackets / expectedPackets.
- const int numLostPacketsQ8 = *loss * seqNumDiff;
-
- // Accumulate reports.
- _accumulateLostPacketsQ8 += numLostPacketsQ8;
- _accumulateExpectedPackets += seqNumDiff;
-
- // Report loss if the total report is based on sufficiently
- // many packets.
- const int limitNumPackets = 10;
- if (_accumulateExpectedPackets >= limitNumPackets)
- {
- *loss = _accumulateLostPacketsQ8 / _accumulateExpectedPackets;
-
- // Reset accumulators
- _accumulateLostPacketsQ8 = 0;
- _accumulateExpectedPackets = 0;
- }
- else
- {
- // Report zero loss until we have enough data to estimate
- // the loss rate.
- *loss = 0;
- }
- }
- }
- // Keep for next time.
- _lastLoss = *loss;
-
- // Remember the sequence number until next time
- _lastPacketLossExtendedHighSeqNum = lastReceivedExtendedHighSeqNum;
-
- WebRtc_UWord32 bitRate = ShapeSimple(*loss, rtt, sentBitrate, nowMS);
- if (bitRate == 0)
- {
- // no change
- return -1;
- }
- _bitRate = bitRate;
- *newBitrate = bitRate;
- return 0;
-}
-
-WebRtc_Word32 BandwidthManagement::AvailableBandwidth(
- WebRtc_UWord32* bandwidthKbit) const {
- CriticalSectionScoped cs(_critsect);
- if (_bitRate == 0) {
- return -1;
- }
- if (!bandwidthKbit) {
- return -1;
- }
- *bandwidthKbit = _bitRate;
- return 0;
-}
-
-/* Calculate the rate that TCP-Friendly Rate Control (TFRC) would apply.
- * The formula in RFC 3448, Section 3.1, is used.
- */
-
-// protected
-WebRtc_Word32 BandwidthManagement::CalcTFRCbps(WebRtc_Word16 avgPackSizeBytes,
- WebRtc_Word32 rttMs,
- WebRtc_Word32 packetLoss)
-{
- if (avgPackSizeBytes <= 0 || rttMs <= 0 || packetLoss <= 0)
- {
- // input variables out of range; return -1
- return -1;
- }
-
- double R = static_cast<double>(rttMs)/1000; // RTT in seconds
- int b = 1; // number of packets acknowledged by a single TCP acknowledgement; recommended = 1
- double t_RTO = 4.0 * R; // TCP retransmission timeout value in seconds; recommended = 4*R
- double p = static_cast<double>(packetLoss)/255; // packet loss rate in [0, 1)
- double s = static_cast<double>(avgPackSizeBytes);
-
- // calculate send rate in bytes/second
- double X = s / (R * sqrt(2 * b * p / 3) + (t_RTO * (3 * sqrt( 3 * b * p / 8) * p * (1 + 32 * p * p))));
-
- return (static_cast<WebRtc_Word32>(X*8)); // bits/second
-}
-
-/*
-* Simple bandwidth estimation. Depends a lot on bwEstimateIncoming and packetLoss.
-*/
-// protected
-WebRtc_UWord32 BandwidthManagement::ShapeSimple(WebRtc_Word32 packetLoss,
- WebRtc_Word32 rtt,
- WebRtc_UWord32 sentBitrate,
- WebRtc_Word64 nowMS)
-{
- WebRtc_UWord32 newBitRate = 0;
- bool reducing = false;
-
- // Limit the rate increases to once a second.
- if (packetLoss <= 5)
- {
- if ((nowMS - _timeLastIncrease) <
- kBWEUpdateIntervalMs)
- {
- return _bitRate;
- }
- _timeLastIncrease = nowMS;
- }
-
- if (packetLoss > 5 && packetLoss <= 26)
- {
- // 2% - 10%
- newBitRate = _bitRate;
- }
- else if (packetLoss > 26)
- {
- // 26/256 ~= 10%
- // reduce rate: newRate = rate * (1 - 0.5*lossRate)
- // packetLoss = 256*lossRate
- newBitRate = static_cast<WebRtc_UWord32>(
- (sentBitrate * static_cast<double>(512 - packetLoss)) / 512.0);
- reducing = true;
- }
- else
- {
- // increase rate by 8%
- newBitRate = static_cast<WebRtc_UWord32>(_bitRate * 1.08 + 0.5);
-
- // add 1 kbps extra, just to make sure that we do not get stuck
- // (gives a little extra increase at low rates, negligible at higher rates)
- newBitRate += 1000;
- }
-
- // Calculate smoothed loss number
- if (_smoothedFractionLostQ4 < 0)
- {
- // startup
- _smoothedFractionLostQ4 = static_cast<WebRtc_UWord16>(packetLoss);
- }
- else
- {
- _smoothedFractionLostQ4 = ((_sFLFactorQ4 * _smoothedFractionLostQ4 + 8) >> 4) // Q4*Q4 = Q8; down to Q4 again with proper rounding
- + (16 - _sFLFactorQ4) * static_cast<WebRtc_UWord16>(packetLoss); // Q4 * Q0 = Q4
- }
-
- // Calculate what rate TFRC would apply in this situation
- //WebRtc_Word32 tfrcRate = CalcTFRCbps(1000, rtt, _smoothedFractionLostQ4 >> 4); // scale loss to Q0 (back to [0, 255])
- WebRtc_Word32 tfrcRate = CalcTFRCbps(1000, rtt, packetLoss); // scale loss to Q0 (back to [0, 255])
-
- if (reducing &&
- tfrcRate > 0 &&
- static_cast<WebRtc_UWord32>(tfrcRate) > newBitRate)
- {
- // do not reduce further if rate is below TFRC rate
- newBitRate = _bitRate;
- }
-
- if (_bwEstimateIncoming > 0 && newBitRate > _bwEstimateIncoming)
- {
- newBitRate = _bwEstimateIncoming;
- }
- if (newBitRate > _maxBitRateConfigured)
- {
- newBitRate = _maxBitRateConfigured;
- }
- if (newBitRate < _minBitRateConfigured)
- {
- WEBRTC_TRACE(kTraceWarning,
- kTraceRtpRtcp,
- _id,
- "The configured min bitrate (%u kbps) is greater than the "
- "estimated available bandwidth (%u kbps).\n",
- _minBitRateConfigured / 1000, newBitRate / 1000);
- newBitRate = _minBitRateConfigured;
- }
- return newBitRate;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/bandwidth_management.h b/trunk/src/modules/rtp_rtcp/source/bandwidth_management.h
deleted file mode 100644
index 2516b86..0000000
--- a/trunk/src/modules/rtp_rtcp/source/bandwidth_management.h
+++ /dev/null
@@ -1,94 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_BANDWIDTH_MANAGEMENT_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_BANDWIDTH_MANAGEMENT_H_
-
-#include "typedefs.h"
-#include "rtp_rtcp_config.h"
-#include "critical_section_wrapper.h"
-
-/*
-* FEC and NACK added bitrate is handled outside class
-*/
-
-namespace webrtc {
-class BandwidthManagement
-{
-public:
- BandwidthManagement(const WebRtc_Word32 id);
- ~BandwidthManagement();
-
- // Call when we receive a RTCP message with TMMBR or REMB
- WebRtc_Word32 UpdateBandwidthEstimate(const WebRtc_UWord16 bandWidthKbit,
- WebRtc_UWord32* newBitrate,
- WebRtc_UWord8* fractionLost,
- WebRtc_UWord16* roundTripTime);
-
- // Call when we receive a RTCP message with a ReceiveBlock
- WebRtc_Word32 UpdatePacketLoss(
- const WebRtc_UWord32 lastReceivedExtendedHighSeqNum,
- WebRtc_UWord32 sentBitrate,
- const WebRtc_UWord16 rtt,
- WebRtc_UWord8* loss,
- WebRtc_UWord32* newBitrate,
- WebRtc_Word64 nowMS);
-
- // If no bandwidth estimate is available or if |bandwidthKbit| is NULL,
- // -1 is returned.
- WebRtc_Word32 AvailableBandwidth(WebRtc_UWord32* bandwidthKbit) const;
-
- void SetSendBitrate(const WebRtc_UWord32 startBitrate,
- const WebRtc_UWord16 minBitrateKbit,
- const WebRtc_UWord16 maxBitrateKbit);
-
- WebRtc_Word32 MaxConfiguredBitrate(WebRtc_UWord16* maxBitrateKbit);
-
-protected:
- WebRtc_UWord32 ShapeSimple(WebRtc_Word32 packetLoss,
- WebRtc_Word32 rtt,
- WebRtc_UWord32 sentBitrate,
- WebRtc_Word64 nowMS);
-
- WebRtc_Word32 CalcTFRCbps(WebRtc_Word16 avgPackSizeBytes,
- WebRtc_Word32 rttMs,
- WebRtc_Word32 packetLoss);
-
-private:
- enum { kBWEUpdateIntervalMs = 1000 };
-
- WebRtc_Word32 _id;
-
- CriticalSectionWrapper* _critsect;
-
- // incoming filters
- WebRtc_UWord32 _lastPacketLossExtendedHighSeqNum;
- bool _lastReportAllLost;
- WebRtc_UWord8 _lastLoss;
- int _accumulateLostPacketsQ8;
- int _accumulateExpectedPackets;
-
- // bitrate
- WebRtc_UWord32 _bitRate;
- WebRtc_UWord32 _minBitRateConfigured;
- WebRtc_UWord32 _maxBitRateConfigured;
-
- WebRtc_UWord8 _last_fraction_loss;
- WebRtc_UWord16 _last_round_trip_time;
-
- // bandwidth estimate
- WebRtc_UWord32 _bwEstimateIncoming;
- WebRtc_Word16 _smoothedFractionLostQ4;
- WebRtc_Word16 _sFLFactorQ4; // forgetting factor for _smoothedFractionLostQ4
- WebRtc_Word64 _timeLastIncrease;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_BANDWIDTH_MANAGEMENT_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/bitrate.cc b/trunk/src/modules/rtp_rtcp/source/bitrate.cc
deleted file mode 100644
index be0c7dd..0000000
--- a/trunk/src/modules/rtp_rtcp/source/bitrate.cc
+++ /dev/null
@@ -1,203 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "Bitrate.h"
-#include "rtp_utility.h"
-
-#define BITRATE_AVERAGE_WINDOW 2000
-
-namespace webrtc {
-Bitrate::Bitrate(RtpRtcpClock* clock) :
- _clock(*clock),
- _packetRate(0),
- _bitrate(0),
- _bitrateNextIdx(0),
- _timeLastRateUpdate(0),
- _bytesCount(0),
- _packetCount(0)
-{
- memset(_packetRateArray, 0, sizeof(_packetRateArray));
- memset(_bitrateDiffMS, 0, sizeof(_bitrateDiffMS));
- memset(_bitrateArray, 0, sizeof(_bitrateArray));
-}
-
-void
-Bitrate::Init()
-{
- _packetRate = 0;
- _bitrate = 0;
- _timeLastRateUpdate = 0;
- _bytesCount = 0;
- _packetCount = 0;
- _bitrateNextIdx = 0;
-
- memset(_packetRateArray, 0, sizeof(_packetRateArray));
- memset(_bitrateDiffMS, 0, sizeof(_bitrateDiffMS));
- memset(_bitrateArray, 0, sizeof(_bitrateArray));
-}
-
-void
-Bitrate::Update(const WebRtc_Word32 bytes)
-{
- _bytesCount += bytes;
- _packetCount++;
-}
-
-WebRtc_UWord32
-Bitrate::PacketRate() const
-{
- return _packetRate;
-}
-
-WebRtc_UWord32
-Bitrate::BitrateLast() const
-{
- return _bitrate;
-}
-
-WebRtc_UWord32
-Bitrate::BitrateNow() const
-{
- WebRtc_UWord32 now = _clock.GetTimeInMS();
- WebRtc_UWord32 diffMS = now -_timeLastRateUpdate;
-
- if(diffMS > 10000) // 10 sec
- {
- // too high diff ignore
- return _bitrate; // bits/s
- }
- WebRtc_UWord64 bitsSinceLastRateUpdate = 8*_bytesCount*1000;
-
- // have to consider the time when the measurement was done
- // ((bits/sec * sec) + (bits)) / sec
- WebRtc_UWord64 bitrate = (((WebRtc_UWord64)_bitrate * 1000) + bitsSinceLastRateUpdate)/(1000+diffMS);
- return (WebRtc_UWord32)bitrate;
-}
-
-void
-Bitrate::Process()
-{
- // triggered by timer
- WebRtc_UWord32 now = _clock.GetTimeInMS();
- WebRtc_UWord32 diffMS = now -_timeLastRateUpdate;
-
- if(diffMS > 100)
- {
- if(diffMS > 10000) // 10 sec
- {
- // too high diff ignore
- _timeLastRateUpdate = now;
- _bytesCount = 0;
- _packetCount = 0;
- return;
- }
- _packetRateArray[_bitrateNextIdx] = (_packetCount*1000)/diffMS;
- _bitrateArray[_bitrateNextIdx] = 8*((_bytesCount*1000)/diffMS);
- // will overflow at ~34 Mbit/s
- _bitrateDiffMS[_bitrateNextIdx] = diffMS;
- _bitrateNextIdx++;
- if(_bitrateNextIdx >= 10)
- {
- _bitrateNextIdx = 0;
- }
-
- WebRtc_UWord32 sumDiffMS = 0;
- WebRtc_UWord64 sumBitrateMS = 0;
- WebRtc_UWord32 sumPacketrateMS = 0;
- for(int i= 0; i <10; i++)
- {
- // sum of time
- sumDiffMS += _bitrateDiffMS[i];
- sumBitrateMS += _bitrateArray[i] * _bitrateDiffMS[i];
- sumPacketrateMS += _packetRateArray[i] * _bitrateDiffMS[i];
- }
- _timeLastRateUpdate = now;
- _bytesCount = 0;
- _packetCount = 0;
-
- _packetRate = sumPacketrateMS/sumDiffMS;
- _bitrate = WebRtc_UWord32(sumBitrateMS / sumDiffMS);
- }
-}
-
-BitRateStats::BitRateStats()
- :_dataSamples(), _accumulatedBytes(0)
-{
-}
-
-BitRateStats::~BitRateStats()
-{
- while (_dataSamples.size() > 0)
- {
- delete _dataSamples.front();
- _dataSamples.pop_front();
- }
-}
-
-void BitRateStats::Init()
-{
- _accumulatedBytes = 0;
- while (_dataSamples.size() > 0)
- {
- delete _dataSamples.front();
- _dataSamples.pop_front();
- }
-}
-
-void BitRateStats::Update(WebRtc_UWord32 packetSizeBytes, WebRtc_Word64 nowMs)
-{
- // Find an empty slot for storing the new sample and at the same time
- // accumulate the history.
- _dataSamples.push_back(new DataTimeSizeTuple(packetSizeBytes, nowMs));
- _accumulatedBytes += packetSizeBytes;
- EraseOld(nowMs);
-}
-
-void BitRateStats::EraseOld(WebRtc_Word64 nowMs)
-{
- while (_dataSamples.size() > 0)
- {
- if (nowMs - _dataSamples.front()->_timeCompleteMs >
- BITRATE_AVERAGE_WINDOW)
- {
- // Delete old sample
- _accumulatedBytes -= _dataSamples.front()->_sizeBytes;
- delete _dataSamples.front();
- _dataSamples.pop_front();
- }
- else
- {
- break;
- }
- }
-}
-
-WebRtc_UWord32 BitRateStats::BitRate(WebRtc_Word64 nowMs)
-{
- // Calculate the average bit rate the past BITRATE_AVERAGE_WINDOW ms.
- // Removes any old samples from the list.
- EraseOld(nowMs);
- WebRtc_Word64 timeOldest = nowMs;
- if (_dataSamples.size() > 0)
- {
- timeOldest = _dataSamples.front()->_timeCompleteMs;
- }
- // Update average bit rate
- float denom = static_cast<float>(nowMs - timeOldest);
- if (nowMs == timeOldest)
- {
- // Calculate with a one second window when we haven't
- // received more than one packet.
- denom = 1000.0;
- }
- return static_cast<WebRtc_UWord32>(_accumulatedBytes * 8.0f * 1000.0f /
- denom + 0.5f);
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/bwe_defines.h b/trunk/src/modules/rtp_rtcp/source/bwe_defines.h
deleted file mode 100644
index 8fdc985..0000000
--- a/trunk/src/modules/rtp_rtcp/source/bwe_defines.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
-
-#include "typedefs.h"
-
-#define BWE_MAX(a,b) ((a)>(b)?(a):(b))
-#define BWE_MIN(a,b) ((a)<(b)?(a):(b))
-
-namespace webrtc {
-enum BandwidthUsage
-{
- kBwNormal,
- kBwOverusing,
- kBwUnderUsing
-};
-
-enum RateControlState
-{
- kRcHold,
- kRcIncrease,
- kRcDecrease
-};
-
-enum RateControlRegion
-{
- kRcNearMax,
- kRcAboveMax,
- kRcMaxUnknown
-};
-
-class RateControlInput
-{
-public:
- RateControlInput(BandwidthUsage bwState,
- WebRtc_UWord32 incomingBitRate,
- double noiseVar) :
- _bwState(bwState), _incomingBitRate(incomingBitRate), _noiseVar(noiseVar)
- {};
-
- BandwidthUsage _bwState;
- WebRtc_UWord32 _incomingBitRate;
- double _noiseVar;
-};
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_BWE_DEFINES_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/dtmf_queue.cc b/trunk/src/modules/rtp_rtcp/source/dtmf_queue.cc
deleted file mode 100644
index 749309b..0000000
--- a/trunk/src/modules/rtp_rtcp/source/dtmf_queue.cc
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "dtmf_queue.h"
-
-#include <string.h> //memset
-
-namespace webrtc {
-DTMFqueue::DTMFqueue():
- _DTMFCritsect(CriticalSectionWrapper::CreateCriticalSection()),
- _nextEmptyIndex(0)
-{
- memset(_DTMFKey,0, sizeof(_DTMFKey));
- memset(_DTMFLen,0, sizeof(_DTMFLen));
- memset(_DTMFLevel,0, sizeof(_DTMFLevel));
-}
-
-DTMFqueue::~DTMFqueue()
-{
- delete _DTMFCritsect;
-}
-
-WebRtc_Word32
-DTMFqueue::AddDTMF(WebRtc_UWord8 key, WebRtc_UWord16 len, WebRtc_UWord8 level)
-{
- CriticalSectionScoped lock(_DTMFCritsect);
-
- if(_nextEmptyIndex >= DTMF_OUTBAND_MAX)
- {
- return -1;
- }
- WebRtc_Word32 index = _nextEmptyIndex;
- _DTMFKey[index] = key;
- _DTMFLen[index] = len;
- _DTMFLevel[index] = level;
- _nextEmptyIndex++;
- return 0;
-}
-
-WebRtc_Word8
-DTMFqueue::NextDTMF(WebRtc_UWord8* DTMFKey, WebRtc_UWord16* len, WebRtc_UWord8* level)
-{
- CriticalSectionScoped lock(_DTMFCritsect);
-
- if(!PendingDTMF())
- {
- return -1;
- }
- *DTMFKey=_DTMFKey[0];
- *len=_DTMFLen[0];
- *level=_DTMFLevel[0];
-
- memmove(&(_DTMFKey[0]), &(_DTMFKey[1]), _nextEmptyIndex*sizeof(WebRtc_UWord8));
- memmove(&(_DTMFLen[0]), &(_DTMFLen[1]), _nextEmptyIndex*sizeof(WebRtc_UWord16));
- memmove(&(_DTMFLevel[0]), &(_DTMFLevel[1]), _nextEmptyIndex*sizeof(WebRtc_UWord8));
-
- _nextEmptyIndex--;
- return 0;
-}
-
-bool
-DTMFqueue::PendingDTMF()
-{
- return(_nextEmptyIndex>0);
-}
-
-void
-DTMFqueue::ResetDTMF()
-{
- _nextEmptyIndex = 0;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/dtmf_queue.h b/trunk/src/modules/rtp_rtcp/source/dtmf_queue.h
deleted file mode 100644
index 8451a21..0000000
--- a/trunk/src/modules/rtp_rtcp/source/dtmf_queue.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
-
-#include "typedefs.h"
-#include "rtp_rtcp_config.h"
-
-#include "critical_section_wrapper.h"
-
-namespace webrtc {
-class DTMFqueue
-{
-public:
- DTMFqueue();
- virtual ~DTMFqueue();
-
- WebRtc_Word32 AddDTMF(WebRtc_UWord8 DTMFKey, WebRtc_UWord16 len, WebRtc_UWord8 level);
- WebRtc_Word8 NextDTMF(WebRtc_UWord8* DTMFKey, WebRtc_UWord16 * len, WebRtc_UWord8 * level);
- bool PendingDTMF();
- void ResetDTMF();
-
-private:
- CriticalSectionWrapper* _DTMFCritsect;
- WebRtc_UWord8 _nextEmptyIndex;
- WebRtc_UWord8 _DTMFKey[DTMF_OUTBAND_MAX];
- WebRtc_UWord16 _DTMFLen[DTMF_OUTBAND_MAX];
- WebRtc_UWord8 _DTMFLevel[DTMF_OUTBAND_MAX];
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_DTMF_QUEUE_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/fec_private_tables.h b/trunk/src/modules/rtp_rtcp/source/fec_private_tables.h
deleted file mode 100644
index 699ee5a..0000000
--- a/trunk/src/modules/rtp_rtcp/source/fec_private_tables.h
+++ /dev/null
@@ -1,25741 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_
-
-// Ensure the tables have internal linkage, as we only require them in one file.
-namespace
-{
- const WebRtc_UWord8 mask10_1[2] =
- {
- 0xff, 0xc0
- };
-
- const WebRtc_UWord8 mask10_10[20] =
- {
- 0x4c, 0x00,
- 0x51, 0x00,
- 0xa0, 0x40,
- 0x04, 0xc0,
- 0x03, 0x80,
- 0x86, 0x00,
- 0x29, 0x00,
- 0x42, 0x40,
- 0x98, 0x00,
- 0x30, 0x80
- };
-
- const WebRtc_UWord8 mask10_2[4] =
- {
- 0xaa, 0x80,
- 0xd5, 0x40
- };
-
- const WebRtc_UWord8 mask10_3[6] =
- {
- 0xa4, 0x40,
- 0xc9, 0x00,
- 0x52, 0x80
- };
-
- const WebRtc_UWord8 mask10_4[8] =
- {
- 0xca, 0x00,
- 0x32, 0x80,
- 0xa1, 0x40,
- 0x55, 0x00
- };
-
- const WebRtc_UWord8 mask10_5[10] =
- {
- 0xca, 0x00,
- 0x32, 0x80,
- 0xa1, 0x40,
- 0x55, 0x00,
- 0x08, 0xc0
- };
-
- const WebRtc_UWord8 mask10_6[12] =
- {
- 0x0e, 0x00,
- 0x33, 0x00,
- 0x10, 0xc0,
- 0x45, 0x40,
- 0x88, 0x80,
- 0xe0, 0x00
- };
-
- const WebRtc_UWord8 mask10_7[14] =
- {
- 0x46, 0x00,
- 0x33, 0x00,
- 0x80, 0xc0,
- 0x0c, 0x40,
- 0x28, 0x80,
- 0x94, 0x00,
- 0xc1, 0x00
- };
-
- const WebRtc_UWord8 mask10_8[16] =
- {
- 0x2c, 0x00,
- 0x81, 0x80,
- 0xa0, 0x40,
- 0x05, 0x40,
- 0x18, 0x80,
- 0xc2, 0x00,
- 0x22, 0x80,
- 0x50, 0x40
- };
-
- const WebRtc_UWord8 mask10_9[18] =
- {
- 0x4c, 0x00,
- 0x23, 0x00,
- 0x88, 0xc0,
- 0x21, 0x40,
- 0x52, 0x80,
- 0x94, 0x00,
- 0x26, 0x00,
- 0x48, 0x40,
- 0x91, 0x80
- };
-
- const WebRtc_UWord8 mask11_1[2] =
- {
- 0xff, 0xe0
- };
-
- const WebRtc_UWord8 mask11_10[20] =
- {
- 0x64, 0x40,
- 0x51, 0x40,
- 0xa9, 0x00,
- 0x04, 0xc0,
- 0xd0, 0x00,
- 0x82, 0x40,
- 0x21, 0x20,
- 0x0c, 0x20,
- 0x4a, 0x00,
- 0x12, 0xa0
- };
-
- const WebRtc_UWord8 mask11_11[22] =
- {
- 0x46, 0x40,
- 0x33, 0x20,
- 0x99, 0x00,
- 0x05, 0x80,
- 0x80, 0xa0,
- 0x84, 0x40,
- 0x40, 0x60,
- 0x0a, 0x80,
- 0x68, 0x00,
- 0x10, 0x20,
- 0x30, 0x40
- };
-
- const WebRtc_UWord8 mask11_2[4] =
- {
- 0xec, 0xc0,
- 0x9b, 0xa0
- };
-
- const WebRtc_UWord8 mask11_3[6] =
- {
- 0xca, 0xc0,
- 0xf1, 0x40,
- 0xb6, 0x20
- };
-
- const WebRtc_UWord8 mask11_4[8] =
- {
- 0xc4, 0xc0,
- 0x31, 0x60,
- 0x4b, 0x20,
- 0x2c, 0xa0
- };
-
- const WebRtc_UWord8 mask11_5[10] =
- {
- 0x86, 0x80,
- 0x23, 0x20,
- 0x16, 0x20,
- 0x4c, 0x20,
- 0x41, 0xc0
- };
-
- const WebRtc_UWord8 mask11_6[12] =
- {
- 0x64, 0x40,
- 0x51, 0x40,
- 0x0c, 0xa0,
- 0xa1, 0x20,
- 0x12, 0xa0,
- 0x8a, 0x40
- };
-
- const WebRtc_UWord8 mask11_7[14] =
- {
- 0x46, 0x40,
- 0x33, 0x20,
- 0x91, 0x80,
- 0xa4, 0x20,
- 0x50, 0xa0,
- 0x84, 0xc0,
- 0x09, 0x60
- };
-
- const WebRtc_UWord8 mask11_8[16] =
- {
- 0x0c, 0x80,
- 0x80, 0x60,
- 0xa0, 0x80,
- 0x05, 0x40,
- 0x43, 0x00,
- 0x1a, 0x00,
- 0x60, 0x20,
- 0x14, 0x20
- };
-
- const WebRtc_UWord8 mask11_9[18] =
- {
- 0x46, 0x40,
- 0x62, 0x60,
- 0x8c, 0x00,
- 0x01, 0x60,
- 0x07, 0x80,
- 0xa0, 0x80,
- 0x18, 0xa0,
- 0x91, 0x00,
- 0x78, 0x00
- };
-
- const WebRtc_UWord8 mask12_1[2] =
- {
- 0xff, 0xf0
- };
-
- const WebRtc_UWord8 mask12_10[20] =
- {
- 0x51, 0x40,
- 0x45, 0x10,
- 0x80, 0xd0,
- 0x24, 0x20,
- 0x0a, 0x20,
- 0x00, 0xe0,
- 0xb8, 0x00,
- 0x09, 0x10,
- 0x56, 0x00,
- 0xa2, 0x80
- };
-
- const WebRtc_UWord8 mask12_11[22] =
- {
- 0x53, 0x60,
- 0x21, 0x30,
- 0x10, 0x90,
- 0x00, 0x70,
- 0x0c, 0x10,
- 0x40, 0xc0,
- 0x6a, 0x00,
- 0x86, 0x00,
- 0x24, 0x80,
- 0x89, 0x00,
- 0xc0, 0x20
- };
-
- const WebRtc_UWord8 mask12_12[24] =
- {
- 0x10, 0x60,
- 0x02, 0x30,
- 0x40, 0x50,
- 0x21, 0x80,
- 0x81, 0x10,
- 0x14, 0x80,
- 0x98, 0x00,
- 0x08, 0x90,
- 0x62, 0x00,
- 0x24, 0x20,
- 0x8a, 0x00,
- 0x84, 0x40
- };
-
- const WebRtc_UWord8 mask12_2[4] =
- {
- 0xec, 0xc0,
- 0x93, 0xb0
- };
-
- const WebRtc_UWord8 mask12_3[6] =
- {
- 0x9b, 0x80,
- 0x4f, 0x10,
- 0x3c, 0x60
- };
-
- const WebRtc_UWord8 mask12_4[8] =
- {
- 0x8b, 0x20,
- 0x14, 0xb0,
- 0x22, 0xd0,
- 0x45, 0x50
- };
-
- const WebRtc_UWord8 mask12_5[10] =
- {
- 0x53, 0x60,
- 0x64, 0x20,
- 0x0c, 0xc0,
- 0x82, 0xa0,
- 0x09, 0x30
- };
-
- const WebRtc_UWord8 mask12_6[12] =
- {
- 0x51, 0x40,
- 0xc5, 0x10,
- 0x21, 0x80,
- 0x12, 0x30,
- 0x08, 0xe0,
- 0x2e, 0x00
- };
-
- const WebRtc_UWord8 mask12_7[14] =
- {
- 0x53, 0x60,
- 0x21, 0x30,
- 0x90, 0x90,
- 0x02, 0x50,
- 0x06, 0xa0,
- 0x2c, 0x00,
- 0x88, 0x60
- };
-
- const WebRtc_UWord8 mask12_8[16] =
- {
- 0x20, 0x60,
- 0x80, 0x30,
- 0x42, 0x40,
- 0x01, 0x90,
- 0x14, 0x10,
- 0x0a, 0x80,
- 0x38, 0x00,
- 0xc5, 0x00
- };
-
- const WebRtc_UWord8 mask12_9[18] =
- {
- 0x53, 0x60,
- 0xe4, 0x20,
- 0x24, 0x40,
- 0xa1, 0x10,
- 0x18, 0x30,
- 0x03, 0x90,
- 0x8a, 0x10,
- 0x04, 0x90,
- 0x00, 0xe0
- };
-
- const WebRtc_UWord8 mask13_1[2] =
- {
- 0xff, 0xf8
- };
-
- const WebRtc_UWord8 mask13_10[20] =
- {
- 0xd1, 0x00,
- 0x44, 0x50,
- 0x10, 0x98,
- 0xa0, 0x50,
- 0x4a, 0x08,
- 0x40, 0x30,
- 0x80, 0x28,
- 0x0c, 0x90,
- 0x05, 0x88,
- 0x62, 0x20
- };
-
- const WebRtc_UWord8 mask13_11[22] =
- {
- 0x51, 0x20,
- 0x22, 0x10,
- 0x13, 0x40,
- 0x25, 0x00,
- 0x18, 0x18,
- 0x0a, 0x20,
- 0x88, 0x88,
- 0x06, 0x80,
- 0xe0, 0x20,
- 0x84, 0x40,
- 0x44, 0x18
- };
-
- const WebRtc_UWord8 mask13_12[24] =
- {
- 0x28, 0x28,
- 0x84, 0x50,
- 0x60, 0x40,
- 0x05, 0x48,
- 0x02, 0x98,
- 0x01, 0x30,
- 0x48, 0x10,
- 0x24, 0x80,
- 0x94, 0x00,
- 0x8a, 0x00,
- 0x11, 0x80,
- 0x52, 0x20
- };
-
- const WebRtc_UWord8 mask13_13[26] =
- {
- 0x51, 0x20,
- 0x66, 0x40,
- 0x05, 0x48,
- 0x81, 0x20,
- 0x94, 0x00,
- 0x30, 0x80,
- 0x21, 0x10,
- 0x03, 0xc0,
- 0xe8, 0x00,
- 0x0a, 0x10,
- 0x80, 0x18,
- 0x04, 0x90,
- 0x08, 0xa8
- };
-
- const WebRtc_UWord8 mask13_2[4] =
- {
- 0xec, 0xc0,
- 0x1b, 0x38
- };
-
- const WebRtc_UWord8 mask13_3[6] =
- {
- 0x99, 0xb0,
- 0x46, 0xd8,
- 0x37, 0x28
- };
-
- const WebRtc_UWord8 mask13_4[8] =
- {
- 0x49, 0xb0,
- 0x26, 0xd0,
- 0x85, 0x68,
- 0x52, 0x58
- };
-
- const WebRtc_UWord8 mask13_5[10] =
- {
- 0x51, 0x30,
- 0x66, 0x40,
- 0x0c, 0x68,
- 0xa1, 0xc0,
- 0x22, 0x98
- };
-
- const WebRtc_UWord8 mask13_6[12] =
- {
- 0xd1, 0x20,
- 0x46, 0xd0,
- 0x15, 0x48,
- 0x21, 0x70,
- 0x28, 0xc8,
- 0xaa, 0x20
- };
-
- const WebRtc_UWord8 mask13_7[14] =
- {
- 0x59, 0x20,
- 0x26, 0x50,
- 0xb1, 0x40,
- 0x2b, 0x08,
- 0x14, 0xc8,
- 0xc8, 0x88,
- 0x84, 0xb0
- };
-
- const WebRtc_UWord8 mask13_8[16] =
- {
- 0x80, 0xa8,
- 0x30, 0x90,
- 0x16, 0x08,
- 0x03, 0x30,
- 0x44, 0x60,
- 0x08, 0x18,
- 0xd8, 0x00,
- 0xa1, 0x40
- };
-
- const WebRtc_UWord8 mask13_9[18] =
- {
- 0x59, 0x20,
- 0x66, 0x40,
- 0x14, 0x40,
- 0x21, 0x48,
- 0x02, 0xc8,
- 0x94, 0x10,
- 0x80, 0xa8,
- 0x0a, 0x90,
- 0x40, 0x18
- };
-
- const WebRtc_UWord8 mask14_1[2] =
- {
- 0xff, 0xfc
- };
-
- const WebRtc_UWord8 mask14_10[20] =
- {
- 0xc0, 0xd4,
- 0x1d, 0x40,
- 0xd4, 0x08,
- 0x02, 0x60,
- 0x04, 0x28,
- 0x20, 0x98,
- 0x40, 0x44,
- 0x08, 0x84,
- 0x68, 0x00,
- 0x23, 0x10
- };
-
- const WebRtc_UWord8 mask14_11[22] =
- {
- 0x62, 0xd0,
- 0x35, 0x20,
- 0x14, 0x14,
- 0xc5, 0x08,
- 0x22, 0x0c,
- 0x88, 0xb8,
- 0x42, 0x54,
- 0x28, 0xa4,
- 0x94, 0x20,
- 0x1b, 0x04,
- 0x22, 0xc0
- };
-
- const WebRtc_UWord8 mask14_12[24] =
- {
- 0x81, 0x04,
- 0x40, 0x68,
- 0x90, 0x24,
- 0x28, 0x28,
- 0x52, 0x10,
- 0x41, 0x88,
- 0x09, 0x30,
- 0x48, 0x44,
- 0x04, 0x44,
- 0x0e, 0x80,
- 0xa5, 0x90,
- 0x12, 0x0c
- };
-
- const WebRtc_UWord8 mask14_13[26] =
- {
- 0x62, 0x54,
- 0x34, 0x60,
- 0x48, 0x04,
- 0x00, 0xac,
- 0x28, 0x08,
- 0x81, 0x08,
- 0x23, 0x04,
- 0x06, 0x80,
- 0x80, 0x14,
- 0x30, 0x10,
- 0x8c, 0x20,
- 0x54, 0x00,
- 0x80, 0xc0
- };
-
- const WebRtc_UWord8 mask14_14[28] =
- {
- 0x40, 0x54,
- 0x15, 0x40,
- 0xc0, 0x04,
- 0x28, 0x10,
- 0x05, 0x0c,
- 0x64, 0x80,
- 0x81, 0x80,
- 0x10, 0x98,
- 0x84, 0x20,
- 0x12, 0x30,
- 0x62, 0x00,
- 0x28, 0x60,
- 0x0e, 0x08,
- 0x10, 0x84
- };
-
- const WebRtc_UWord8 mask14_2[4] =
- {
- 0xec, 0xe8,
- 0x3b, 0x9c
- };
-
- const WebRtc_UWord8 mask14_3[6] =
- {
- 0xac, 0xd8,
- 0x55, 0x6c,
- 0x27, 0xb4
- };
-
- const WebRtc_UWord8 mask14_4[8] =
- {
- 0x2c, 0xd8,
- 0x93, 0x68,
- 0x1a, 0xb4,
- 0x47, 0x2c
- };
-
- const WebRtc_UWord8 mask14_5[10] =
- {
- 0x64, 0xd8,
- 0xa5, 0x68,
- 0x52, 0xb4,
- 0x1d, 0xa8,
- 0x9c, 0x54
- };
-
- const WebRtc_UWord8 mask14_6[12] =
- {
- 0x4a, 0x54,
- 0x95, 0x48,
- 0x14, 0xb4,
- 0x51, 0xa8,
- 0x22, 0x6c,
- 0x88, 0x8c
- };
-
- const WebRtc_UWord8 mask14_7[14] =
- {
- 0x62, 0x54,
- 0xb9, 0x20,
- 0x18, 0xb4,
- 0x54, 0x98,
- 0x06, 0x6c,
- 0x85, 0x54,
- 0xaa, 0x88
- };
-
- const WebRtc_UWord8 mask14_8[16] =
- {
- 0xc0, 0x14,
- 0x41, 0x60,
- 0x88, 0x30,
- 0x20, 0xa4,
- 0x0a, 0x48,
- 0x04, 0x98,
- 0x94, 0x40,
- 0x72, 0x00
- };
-
- const WebRtc_UWord8 mask14_9[18] =
- {
- 0xa2, 0x54,
- 0x34, 0x60,
- 0x4a, 0x24,
- 0x20, 0xa8,
- 0x11, 0x84,
- 0x49, 0x08,
- 0x86, 0x0c,
- 0x20, 0xd4,
- 0x88, 0x48
- };
-
- const WebRtc_UWord8 mask15_1[2] =
- {
- 0xff, 0xfe
- };
-
- const WebRtc_UWord8 mask15_10[20] =
- {
- 0xc0, 0xa0,
- 0x15, 0x56,
- 0x74, 0x40,
- 0x00, 0x9c,
- 0x01, 0x2c,
- 0x44, 0x92,
- 0x88, 0x50,
- 0x20, 0xa4,
- 0xaa, 0x04,
- 0x02, 0x62
- };
-
- const WebRtc_UWord8 mask15_11[22] =
- {
- 0x62, 0x22,
- 0xf1, 0x10,
- 0x10, 0x0e,
- 0x10, 0xb0,
- 0x24, 0x24,
- 0x01, 0x12,
- 0x00, 0xc4,
- 0x04, 0xa2,
- 0x02, 0x58,
- 0x2b, 0x00,
- 0x98, 0x40
- };
-
- const WebRtc_UWord8 mask15_12[24] =
- {
- 0x88, 0x90,
- 0x40, 0x54,
- 0x82, 0x62,
- 0x21, 0xa4,
- 0x10, 0x64,
- 0x44, 0x0a,
- 0x10, 0xc8,
- 0x4d, 0x2a,
- 0x38, 0x02,
- 0x17, 0x48,
- 0x90, 0x84,
- 0x72, 0x14
- };
-
- const WebRtc_UWord8 mask15_13[26] =
- {
- 0x62, 0xa2,
- 0x34, 0x44,
- 0x40, 0x4a,
- 0xc4, 0x04,
- 0x08, 0x60,
- 0x94, 0x12,
- 0x88, 0xc0,
- 0x21, 0x32,
- 0xc1, 0x40,
- 0x10, 0x68,
- 0x06, 0x90,
- 0x59, 0x00,
- 0x0a, 0x0c
- };
-
- const WebRtc_UWord8 mask15_14[28] =
- {
- 0x40, 0x82,
- 0x15, 0x54,
- 0x88, 0x12,
- 0xc0, 0x10,
- 0x80, 0xa0,
- 0x01, 0x22,
- 0x40, 0x2c,
- 0x22, 0x02,
- 0x90, 0x04,
- 0x12, 0x40,
- 0x5d, 0x00,
- 0x20, 0x54,
- 0x86, 0x08,
- 0x28, 0x88
- };
-
- const WebRtc_UWord8 mask15_15[30] =
- {
- 0x62, 0x22,
- 0x31, 0x10,
- 0x58, 0x00,
- 0x01, 0x12,
- 0x88, 0x20,
- 0x44, 0x02,
- 0x29, 0x04,
- 0x82, 0xa0,
- 0x0a, 0x1a,
- 0x11, 0xe0,
- 0x84, 0x04,
- 0x86, 0x40,
- 0x00, 0x86,
- 0x44, 0x48,
- 0x10, 0x98
- };
-
- const WebRtc_UWord8 mask15_2[4] =
- {
- 0xec, 0xea,
- 0xbb, 0x9c
- };
-
- const WebRtc_UWord8 mask15_3[6] =
- {
- 0xac, 0x92,
- 0x55, 0x4a,
- 0x43, 0x36
- };
-
- const WebRtc_UWord8 mask15_4[8] =
- {
- 0x25, 0xaa,
- 0x95, 0x54,
- 0x1a, 0x6a,
- 0x43, 0xd4
- };
-
- const WebRtc_UWord8 mask15_5[10] =
- {
- 0x64, 0xa2,
- 0x25, 0x54,
- 0x49, 0x68,
- 0x53, 0x90,
- 0x8e, 0x30
- };
-
- const WebRtc_UWord8 mask15_6[12] =
- {
- 0x62, 0x8a,
- 0x15, 0x54,
- 0x4c, 0x46,
- 0x52, 0x94,
- 0x23, 0x64,
- 0x8a, 0x58
- };
-
- const WebRtc_UWord8 mask15_7[14] =
- {
- 0x62, 0xa2,
- 0xb1, 0x14,
- 0x18, 0x6a,
- 0x44, 0xd4,
- 0x13, 0x64,
- 0x49, 0x1a,
- 0x86, 0x8c
- };
-
- const WebRtc_UWord8 mask15_8[16] =
- {
- 0x90, 0x22,
- 0x09, 0x50,
- 0x00, 0x6a,
- 0x20, 0x34,
- 0x14, 0x44,
- 0xc2, 0x10,
- 0x00, 0xc6,
- 0x65, 0x80
- };
-
- const WebRtc_UWord8 mask15_9[18] =
- {
- 0x62, 0x22,
- 0x24, 0x44,
- 0xc0, 0x50,
- 0x03, 0x0c,
- 0x16, 0x28,
- 0x89, 0x00,
- 0x82, 0x90,
- 0x08, 0xa4,
- 0x90, 0x48
- };
-
- const WebRtc_UWord8 mask16_1[2] =
- {
- 0xff, 0xff
- };
-
- const WebRtc_UWord8 mask16_10[20] =
- {
- 0x45, 0x51,
- 0x10, 0xa2,
- 0x01, 0x25,
- 0x0b, 0x42,
- 0xd8, 0x20,
- 0x82, 0x8c,
- 0x24, 0x4a,
- 0x38, 0x18,
- 0x2a, 0x25,
- 0x84, 0x92
- };
-
- const WebRtc_UWord8 mask16_11[22] =
- {
- 0x55, 0x55,
- 0x2a, 0x22,
- 0x31, 0x11,
- 0x83, 0x42,
- 0x06, 0x98,
- 0x40, 0xe1,
- 0x2c, 0x44,
- 0xd8, 0x28,
- 0x92, 0x81,
- 0x84, 0x32,
- 0x68, 0x0c
- };
-
- const WebRtc_UWord8 mask16_12[24] =
- {
- 0x84, 0x31,
- 0x18, 0xa2,
- 0x4e, 0x01,
- 0x44, 0xc8,
- 0x0e, 0x90,
- 0x20, 0xcc,
- 0x93, 0x40,
- 0x2d, 0x10,
- 0x31, 0x44,
- 0xc0, 0x23,
- 0x11, 0x25,
- 0xe8, 0x80
- };
-
- const WebRtc_UWord8 mask16_13[26] =
- {
- 0x45, 0x15,
- 0x22, 0x22,
- 0x96, 0x0c,
- 0x0c, 0x50,
- 0x62, 0x04,
- 0x49, 0x06,
- 0x11, 0x82,
- 0x12, 0x38,
- 0x40, 0x71,
- 0xa8, 0x8a,
- 0x08, 0xa1,
- 0xa0, 0xc0,
- 0xc5, 0x10
- };
-
- const WebRtc_UWord8 mask16_14[28] =
- {
- 0x45, 0x51,
- 0x22, 0x0a,
- 0x84, 0xd0,
- 0x0c, 0x8a,
- 0x18, 0x06,
- 0x30, 0x03,
- 0x61, 0x08,
- 0x40, 0x11,
- 0x10, 0x2c,
- 0x09, 0x60,
- 0x00, 0x94,
- 0x52, 0x40,
- 0xa4, 0x24,
- 0x82, 0x88
- };
-
- const WebRtc_UWord8 mask16_15[30] =
- {
- 0x55, 0x11,
- 0x22, 0x22,
- 0x11, 0x11,
- 0x80, 0x45,
- 0x20, 0x1a,
- 0x08, 0x68,
- 0x22, 0x84,
- 0x48, 0x09,
- 0x07, 0x01,
- 0x94, 0x20,
- 0x82, 0x06,
- 0x60, 0x48,
- 0x89, 0x80,
- 0x00, 0x8e,
- 0x18, 0x22
- };
-
- const WebRtc_UWord8 mask16_16[32] =
- {
- 0xa4, 0x10,
- 0x01, 0x2a,
- 0x06, 0x42,
- 0x08, 0x68,
- 0x81, 0x90,
- 0x00, 0xf0,
- 0x50, 0x05,
- 0x20, 0x51,
- 0x43, 0x08,
- 0x68, 0x80,
- 0x80, 0x0b,
- 0x10, 0x4c,
- 0x12, 0x30,
- 0x40, 0x85,
- 0x0e, 0x04,
- 0x18, 0x12
- };
-
- const WebRtc_UWord8 mask16_2[4] =
- {
- 0xae, 0xae,
- 0x79, 0x79
- };
-
- const WebRtc_UWord8 mask16_3[6] =
- {
- 0xad, 0x2d,
- 0x76, 0x36,
- 0x26, 0xdb
- };
-
- const WebRtc_UWord8 mask16_4[8] =
- {
- 0x55, 0x55,
- 0xaa, 0xaa,
- 0x35, 0x35,
- 0xca, 0xca
- };
-
- const WebRtc_UWord8 mask16_5[10] =
- {
- 0x55, 0x55,
- 0x2a, 0x2a,
- 0x24, 0x25,
- 0x84, 0xc8,
- 0x10, 0xb6
- };
-
- const WebRtc_UWord8 mask16_6[12] =
- {
- 0x51, 0x51,
- 0x0a, 0x2a,
- 0xa2, 0x15,
- 0x84, 0x4a,
- 0x30, 0x92,
- 0x04, 0xac
- };
-
- const WebRtc_UWord8 mask16_7[14] =
- {
- 0x45, 0x51,
- 0x22, 0x2a,
- 0x91, 0x11,
- 0x2e, 0x08,
- 0x48, 0x34,
- 0x90, 0x29,
- 0x09, 0x86
- };
-
- const WebRtc_UWord8 mask16_8[16] =
- {
- 0x20, 0x54,
- 0x18, 0x88,
- 0x84, 0x07,
- 0x60, 0x48,
- 0x12, 0x82,
- 0x81, 0x41,
- 0x40, 0x62,
- 0x16, 0x30
- };
-
- const WebRtc_UWord8 mask16_9[18] =
- {
- 0x55, 0x51,
- 0x22, 0x2a,
- 0x05, 0x85,
- 0x09, 0x4a,
- 0x84, 0x32,
- 0xc0, 0x0d,
- 0x20, 0xa6,
- 0x1a, 0x09,
- 0x44, 0x64
- };
-
- const WebRtc_UWord8 mask17_1[6] =
- {
- 0xff, 0xff, 0x80, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask17_10[60] =
- {
- 0x55, 0x8c, 0x80, 0x00, 0x00, 0x00,
- 0xaa, 0x27, 0x00, 0x00, 0x00, 0x00,
- 0xa5, 0x32, 0x80, 0x00, 0x00, 0x00,
- 0x62, 0x61, 0x80, 0x00, 0x00, 0x00,
- 0x3c, 0x5c, 0x00, 0x00, 0x00, 0x00,
- 0x8e, 0xcc, 0x00, 0x00, 0x00, 0x00,
- 0x6a, 0x2b, 0x00, 0x00, 0x00, 0x00,
- 0x36, 0x32, 0x80, 0x00, 0x00, 0x00,
- 0xd1, 0x25, 0x80, 0x00, 0x00, 0x00,
- 0xc8, 0x02, 0x80, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask17_11[66] =
- {
- 0x55, 0x8c, 0x80, 0x00, 0x00, 0x00,
- 0xaa, 0x27, 0x00, 0x00, 0x00, 0x00,
- 0xa5, 0x32, 0x80, 0x00, 0x00, 0x00,
- 0x62, 0x61, 0x80, 0x00, 0x00, 0x00,
- 0x3c, 0x5c, 0x00, 0x00, 0x00, 0x00,
- 0x51, 0x84, 0x80, 0x00, 0x00, 0x00,
- 0xa2, 0x27, 0x00, 0x00, 0x00, 0x00,
- 0x95, 0x51, 0x80, 0x00, 0x00, 0x00,
- 0x4a, 0x1a, 0x00, 0x00, 0x00, 0x00,
- 0x30, 0x68, 0x00, 0x00, 0x00, 0x00,
- 0x2c, 0x89, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask17_12[72] =
- {
- 0x51, 0x84, 0x80, 0x00, 0x00, 0x00,
- 0xa2, 0x27, 0x00, 0x00, 0x00, 0x00,
- 0x95, 0x51, 0x80, 0x00, 0x00, 0x00,
- 0x4a, 0x1a, 0x00, 0x00, 0x00, 0x00,
- 0x30, 0x68, 0x00, 0x00, 0x00, 0x00,
- 0x2c, 0x89, 0x00, 0x00, 0x00, 0x00,
- 0x55, 0x8c, 0x80, 0x00, 0x00, 0x00,
- 0xaa, 0x27, 0x00, 0x00, 0x00, 0x00,
- 0xa5, 0x32, 0x80, 0x00, 0x00, 0x00,
- 0x62, 0x61, 0x80, 0x00, 0x00, 0x00,
- 0x3c, 0x5c, 0x00, 0x00, 0x00, 0x00,
- 0x51, 0x35, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask17_13[78] =
- {
- 0x51, 0x84, 0x80, 0x00, 0x00, 0x00,
- 0xa2, 0x27, 0x00, 0x00, 0x00, 0x00,
- 0x95, 0x51, 0x80, 0x00, 0x00, 0x00,
- 0x4a, 0x1a, 0x00, 0x00, 0x00, 0x00,
- 0x30, 0x68, 0x00, 0x00, 0x00, 0x00,
- 0x2c, 0x89, 0x00, 0x00, 0x00, 0x00,
- 0x15, 0x8c, 0x00, 0x00, 0x00, 0x00,
- 0x8a, 0x47, 0x00, 0x00, 0x00, 0x00,
- 0x25, 0x81, 0x80, 0x00, 0x00, 0x00,
- 0x62, 0x12, 0x80, 0x00, 0x00, 0x00,
- 0x58, 0x58, 0x00, 0x00, 0x00, 0x00,
- 0x0e, 0x28, 0x80, 0x00, 0x00, 0x00,
- 0x83, 0x34, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask17_14[84] =
- {
- 0x15, 0x8c, 0x00, 0x00, 0x00, 0x00,
- 0x8a, 0x47, 0x00, 0x00, 0x00, 0x00,
- 0x25, 0x81, 0x80, 0x00, 0x00, 0x00,
- 0x62, 0x12, 0x80, 0x00, 0x00, 0x00,
- 0x58, 0x58, 0x00, 0x00, 0x00, 0x00,
- 0x0e, 0x28, 0x80, 0x00, 0x00, 0x00,
- 0x83, 0x34, 0x00, 0x00, 0x00, 0x00,
- 0x51, 0x84, 0x80, 0x00, 0x00, 0x00,
- 0xa2, 0x27, 0x00, 0x00, 0x00, 0x00,
- 0x95, 0x51, 0x80, 0x00, 0x00, 0x00,
- 0x4a, 0x1a, 0x00, 0x00, 0x00, 0x00,
- 0x30, 0x68, 0x00, 0x00, 0x00, 0x00,
- 0x2c, 0x89, 0x00, 0x00, 0x00, 0x00,
- 0xb0, 0xde, 0x80, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask17_15[90] =
- {
- 0x15, 0x8c, 0x00, 0x00, 0x00, 0x00,
- 0x8a, 0x47, 0x00, 0x00, 0x00, 0x00,
- 0x25, 0x81, 0x80, 0x00, 0x00, 0x00,
- 0x62, 0x12, 0x80, 0x00, 0x00, 0x00,
- 0x58, 0x58, 0x00, 0x00, 0x00, 0x00,
- 0x0e, 0x28, 0x80, 0x00, 0x00, 0x00,
- 0x83, 0x34, 0x00, 0x00, 0x00, 0x00,
- 0x25, 0x2c, 0x00, 0x00, 0x00, 0x00,
- 0x8a, 0x91, 0x00, 0x00, 0x00, 0x00,
- 0x91, 0xc0, 0x80, 0x00, 0x00, 0x00,
- 0x68, 0x06, 0x80, 0x00, 0x00, 0x00,
- 0x32, 0xc8, 0x00, 0x00, 0x00, 0x00,
- 0x43, 0x45, 0x00, 0x00, 0x00, 0x00,
- 0xc4, 0x30, 0x80, 0x00, 0x00, 0x00,
- 0x1c, 0xa2, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask17_16[96] =
- {
- 0x25, 0x2c, 0x00, 0x00, 0x00, 0x00,
- 0x8a, 0x91, 0x00, 0x00, 0x00, 0x00,
- 0x91, 0xc0, 0x80, 0x00, 0x00, 0x00,
- 0x68, 0x06, 0x80, 0x00, 0x00, 0x00,
- 0x32, 0xc8, 0x00, 0x00, 0x00, 0x00,
- 0x43, 0x45, 0x00, 0x00, 0x00, 0x00,
- 0xc4, 0x30, 0x80, 0x00, 0x00, 0x00,
- 0x1c, 0xa2, 0x00, 0x00, 0x00, 0x00,
- 0x15, 0x8c, 0x00, 0x00, 0x00, 0x00,
- 0x8a, 0x47, 0x00, 0x00, 0x00, 0x00,
- 0x25, 0x81, 0x80, 0x00, 0x00, 0x00,
- 0x62, 0x12, 0x80, 0x00, 0x00, 0x00,
- 0x58, 0x58, 0x00, 0x00, 0x00, 0x00,
- 0x0e, 0x28, 0x80, 0x00, 0x00, 0x00,
- 0x83, 0x34, 0x00, 0x00, 0x00, 0x00,
- 0x0a, 0x1c, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask17_17[102] =
- {
- 0x25, 0x2c, 0x00, 0x00, 0x00, 0x00,
- 0x8a, 0x91, 0x00, 0x00, 0x00, 0x00,
- 0x91, 0xc0, 0x80, 0x00, 0x00, 0x00,
- 0x68, 0x06, 0x80, 0x00, 0x00, 0x00,
- 0x32, 0xc8, 0x00, 0x00, 0x00, 0x00,
- 0x43, 0x45, 0x00, 0x00, 0x00, 0x00,
- 0xc4, 0x30, 0x80, 0x00, 0x00, 0x00,
- 0x1c, 0xa2, 0x00, 0x00, 0x00, 0x00,
- 0x25, 0x4c, 0x00, 0x00, 0x00, 0x00,
- 0x8a, 0x66, 0x00, 0x00, 0x00, 0x00,
- 0x91, 0x91, 0x00, 0x00, 0x00, 0x00,
- 0x68, 0x42, 0x80, 0x00, 0x00, 0x00,
- 0x32, 0xa4, 0x00, 0x00, 0x00, 0x00,
- 0x43, 0x13, 0x00, 0x00, 0x00, 0x00,
- 0xc4, 0x30, 0x80, 0x00, 0x00, 0x00,
- 0x1c, 0x88, 0x80, 0x00, 0x00, 0x00,
- 0x3c, 0x09, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask17_2[12] =
- {
- 0xce, 0xce, 0x00, 0x00, 0x00, 0x00,
- 0xb9, 0x39, 0x80, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask17_3[18] =
- {
- 0xcd, 0xcc, 0x00, 0x00, 0x00, 0x00,
- 0x97, 0x27, 0x00, 0x00, 0x00, 0x00,
- 0xb8, 0xd1, 0x80, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask17_4[24] =
- {
- 0xca, 0xec, 0x00, 0x00, 0x00, 0x00,
- 0xa9, 0x67, 0x00, 0x00, 0x00, 0x00,
- 0x3a, 0xb1, 0x80, 0x00, 0x00, 0x00,
- 0x55, 0x5a, 0x80, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask17_5[30] =
- {
- 0x55, 0x44, 0x80, 0x00, 0x00, 0x00,
- 0x2a, 0x66, 0x00, 0x00, 0x00, 0x00,
- 0x25, 0xa1, 0x80, 0x00, 0x00, 0x00,
- 0xe2, 0x12, 0x80, 0x00, 0x00, 0x00,
- 0x99, 0x98, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask17_6[36] =
- {
- 0xd1, 0x4c, 0x00, 0x00, 0x00, 0x00,
- 0xa2, 0xc5, 0x00, 0x00, 0x00, 0x00,
- 0x95, 0x30, 0x80, 0x00, 0x00, 0x00,
- 0xca, 0x0a, 0x80, 0x00, 0x00, 0x00,
- 0xa4, 0xaa, 0x00, 0x00, 0x00, 0x00,
- 0x78, 0x15, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask17_7[42] =
- {
- 0x15, 0x44, 0x80, 0x00, 0x00, 0x00,
- 0x8a, 0x23, 0x00, 0x00, 0x00, 0x00,
- 0x85, 0x91, 0x00, 0x00, 0x00, 0x00,
- 0x32, 0x0a, 0x80, 0x00, 0x00, 0x00,
- 0x58, 0x34, 0x00, 0x00, 0x00, 0x00,
- 0x2c, 0x0d, 0x00, 0x00, 0x00, 0x00,
- 0x43, 0xc8, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask17_8[48] =
- {
- 0x64, 0x16, 0x00, 0x00, 0x00, 0x00,
- 0xa2, 0xc2, 0x00, 0x00, 0x00, 0x00,
- 0x51, 0x60, 0x80, 0x00, 0x00, 0x00,
- 0x4a, 0x85, 0x00, 0x00, 0x00, 0x00,
- 0x38, 0x4c, 0x00, 0x00, 0x00, 0x00,
- 0x89, 0x29, 0x00, 0x00, 0x00, 0x00,
- 0x07, 0x11, 0x80, 0x00, 0x00, 0x00,
- 0x94, 0xb0, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask17_9[54] =
- {
- 0x8e, 0xcc, 0x00, 0x00, 0x00, 0x00,
- 0x6a, 0x2b, 0x00, 0x00, 0x00, 0x00,
- 0x36, 0x32, 0x80, 0x00, 0x00, 0x00,
- 0xd1, 0x25, 0x80, 0x00, 0x00, 0x00,
- 0x55, 0x8c, 0x80, 0x00, 0x00, 0x00,
- 0xaa, 0x27, 0x00, 0x00, 0x00, 0x00,
- 0xa5, 0x32, 0x80, 0x00, 0x00, 0x00,
- 0x62, 0x61, 0x80, 0x00, 0x00, 0x00,
- 0x3c, 0x5c, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_1[6] =
- {
- 0xff, 0xff, 0xc0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_10[60] =
- {
- 0x8c, 0xc6, 0x40, 0x00, 0x00, 0x00,
- 0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
- 0x32, 0x99, 0x40, 0x00, 0x00, 0x00,
- 0x61, 0xb0, 0xc0, 0x00, 0x00, 0x00,
- 0x5c, 0x2e, 0x00, 0x00, 0x00, 0x00,
- 0xcc, 0x66, 0x00, 0x00, 0x00, 0x00,
- 0x2b, 0x15, 0x80, 0x00, 0x00, 0x00,
- 0x32, 0x99, 0x40, 0x00, 0x00, 0x00,
- 0x25, 0x92, 0xc0, 0x00, 0x00, 0x00,
- 0xfd, 0x9d, 0xc0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_11[66] =
- {
- 0x8c, 0xc6, 0x40, 0x00, 0x00, 0x00,
- 0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
- 0x32, 0x99, 0x40, 0x00, 0x00, 0x00,
- 0x61, 0xb0, 0xc0, 0x00, 0x00, 0x00,
- 0x5c, 0x2e, 0x00, 0x00, 0x00, 0x00,
- 0x84, 0xc2, 0x40, 0x00, 0x00, 0x00,
- 0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
- 0x51, 0xa8, 0xc0, 0x00, 0x00, 0x00,
- 0x1a, 0x0d, 0x00, 0x00, 0x00, 0x00,
- 0x68, 0x34, 0x00, 0x00, 0x00, 0x00,
- 0x89, 0x44, 0x80, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_12[72] =
- {
- 0x84, 0xc2, 0x40, 0x00, 0x00, 0x00,
- 0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
- 0x51, 0xa8, 0xc0, 0x00, 0x00, 0x00,
- 0x1a, 0x0d, 0x00, 0x00, 0x00, 0x00,
- 0x68, 0x34, 0x00, 0x00, 0x00, 0x00,
- 0x89, 0x44, 0x80, 0x00, 0x00, 0x00,
- 0x8c, 0xc6, 0x40, 0x00, 0x00, 0x00,
- 0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
- 0x32, 0x99, 0x40, 0x00, 0x00, 0x00,
- 0x61, 0xb0, 0xc0, 0x00, 0x00, 0x00,
- 0x5c, 0x2e, 0x00, 0x00, 0x00, 0x00,
- 0x5b, 0x0c, 0x40, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_13[78] =
- {
- 0x84, 0xc2, 0x40, 0x00, 0x00, 0x00,
- 0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
- 0x51, 0xa8, 0xc0, 0x00, 0x00, 0x00,
- 0x1a, 0x0d, 0x00, 0x00, 0x00, 0x00,
- 0x68, 0x34, 0x00, 0x00, 0x00, 0x00,
- 0x89, 0x44, 0x80, 0x00, 0x00, 0x00,
- 0x8c, 0x46, 0x00, 0x00, 0x00, 0x00,
- 0x47, 0x23, 0x80, 0x00, 0x00, 0x00,
- 0x81, 0xc0, 0xc0, 0x00, 0x00, 0x00,
- 0x12, 0x89, 0x40, 0x00, 0x00, 0x00,
- 0x58, 0x2c, 0x00, 0x00, 0x00, 0x00,
- 0x28, 0x94, 0x40, 0x00, 0x00, 0x00,
- 0x34, 0x1a, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_14[84] =
- {
- 0x8c, 0x46, 0x00, 0x00, 0x00, 0x00,
- 0x47, 0x23, 0x80, 0x00, 0x00, 0x00,
- 0x81, 0xc0, 0xc0, 0x00, 0x00, 0x00,
- 0x12, 0x89, 0x40, 0x00, 0x00, 0x00,
- 0x58, 0x2c, 0x00, 0x00, 0x00, 0x00,
- 0x28, 0x94, 0x40, 0x00, 0x00, 0x00,
- 0x34, 0x1a, 0x00, 0x00, 0x00, 0x00,
- 0x84, 0xc2, 0x40, 0x00, 0x00, 0x00,
- 0x27, 0x13, 0x80, 0x00, 0x00, 0x00,
- 0x51, 0xa8, 0xc0, 0x00, 0x00, 0x00,
- 0x1a, 0x0d, 0x00, 0x00, 0x00, 0x00,
- 0x68, 0x34, 0x00, 0x00, 0x00, 0x00,
- 0x89, 0x44, 0x80, 0x00, 0x00, 0x00,
- 0x7f, 0x4f, 0xc0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_15[90] =
- {
- 0x8c, 0x46, 0x00, 0x00, 0x00, 0x00,
- 0x47, 0x23, 0x80, 0x00, 0x00, 0x00,
- 0x81, 0xc0, 0xc0, 0x00, 0x00, 0x00,
- 0x12, 0x89, 0x40, 0x00, 0x00, 0x00,
- 0x58, 0x2c, 0x00, 0x00, 0x00, 0x00,
- 0x28, 0x94, 0x40, 0x00, 0x00, 0x00,
- 0x34, 0x1a, 0x00, 0x00, 0x00, 0x00,
- 0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
- 0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
- 0xc0, 0xe0, 0x40, 0x00, 0x00, 0x00,
- 0x06, 0x83, 0x40, 0x00, 0x00, 0x00,
- 0xc8, 0x64, 0x00, 0x00, 0x00, 0x00,
- 0x45, 0x22, 0x80, 0x00, 0x00, 0x00,
- 0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
- 0xa2, 0x51, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_16[96] =
- {
- 0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
- 0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
- 0xc0, 0xe0, 0x40, 0x00, 0x00, 0x00,
- 0x06, 0x83, 0x40, 0x00, 0x00, 0x00,
- 0xc8, 0x64, 0x00, 0x00, 0x00, 0x00,
- 0x45, 0x22, 0x80, 0x00, 0x00, 0x00,
- 0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
- 0xa2, 0x51, 0x00, 0x00, 0x00, 0x00,
- 0x8c, 0x46, 0x00, 0x00, 0x00, 0x00,
- 0x47, 0x23, 0x80, 0x00, 0x00, 0x00,
- 0x81, 0xc0, 0xc0, 0x00, 0x00, 0x00,
- 0x12, 0x89, 0x40, 0x00, 0x00, 0x00,
- 0x58, 0x2c, 0x00, 0x00, 0x00, 0x00,
- 0x28, 0x94, 0x40, 0x00, 0x00, 0x00,
- 0x34, 0x1a, 0x00, 0x00, 0x00, 0x00,
- 0xef, 0xf2, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_17[102] =
- {
- 0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
- 0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
- 0xc0, 0xe0, 0x40, 0x00, 0x00, 0x00,
- 0x06, 0x83, 0x40, 0x00, 0x00, 0x00,
- 0xc8, 0x64, 0x00, 0x00, 0x00, 0x00,
- 0x45, 0x22, 0x80, 0x00, 0x00, 0x00,
- 0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
- 0xa2, 0x51, 0x00, 0x00, 0x00, 0x00,
- 0x4c, 0x26, 0x00, 0x00, 0x00, 0x00,
- 0x66, 0x33, 0x00, 0x00, 0x00, 0x00,
- 0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
- 0x42, 0xa1, 0x40, 0x00, 0x00, 0x00,
- 0xa4, 0x52, 0x00, 0x00, 0x00, 0x00,
- 0x13, 0x09, 0x80, 0x00, 0x00, 0x00,
- 0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
- 0x88, 0xc4, 0x40, 0x00, 0x00, 0x00,
- 0x09, 0x04, 0x80, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_18[108] =
- {
- 0x4c, 0x26, 0x00, 0x00, 0x00, 0x00,
- 0x66, 0x33, 0x00, 0x00, 0x00, 0x00,
- 0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
- 0x42, 0xa1, 0x40, 0x00, 0x00, 0x00,
- 0xa4, 0x52, 0x00, 0x00, 0x00, 0x00,
- 0x13, 0x09, 0x80, 0x00, 0x00, 0x00,
- 0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
- 0x88, 0xc4, 0x40, 0x00, 0x00, 0x00,
- 0x09, 0x04, 0x80, 0x00, 0x00, 0x00,
- 0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
- 0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
- 0xc0, 0xe0, 0x40, 0x00, 0x00, 0x00,
- 0x06, 0x83, 0x40, 0x00, 0x00, 0x00,
- 0xc8, 0x64, 0x00, 0x00, 0x00, 0x00,
- 0x45, 0x22, 0x80, 0x00, 0x00, 0x00,
- 0x30, 0x98, 0x40, 0x00, 0x00, 0x00,
- 0xa2, 0x51, 0x00, 0x00, 0x00, 0x00,
- 0xd0, 0x03, 0x40, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_2[12] =
- {
- 0xce, 0x67, 0x00, 0x00, 0x00, 0x00,
- 0x39, 0x9c, 0xc0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_3[18] =
- {
- 0xcc, 0x66, 0x00, 0x00, 0x00, 0x00,
- 0x27, 0x15, 0x80, 0x00, 0x00, 0x00,
- 0x92, 0xc9, 0x40, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_4[24] =
- {
- 0xec, 0x76, 0x00, 0x00, 0x00, 0x00,
- 0x67, 0x33, 0x80, 0x00, 0x00, 0x00,
- 0xb1, 0xd8, 0xc0, 0x00, 0x00, 0x00,
- 0x5a, 0xad, 0x40, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_5[30] =
- {
- 0x4c, 0xa6, 0x40, 0x00, 0x00, 0x00,
- 0x66, 0x33, 0x00, 0x00, 0x00, 0x00,
- 0x19, 0xd0, 0xc0, 0x00, 0x00, 0x00,
- 0x9c, 0x89, 0x40, 0x00, 0x00, 0x00,
- 0xe3, 0x4c, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_6[36] =
- {
- 0xcc, 0x26, 0x00, 0x00, 0x00, 0x00,
- 0x45, 0x62, 0x80, 0x00, 0x00, 0x00,
- 0xb0, 0x98, 0x40, 0x00, 0x00, 0x00,
- 0x8a, 0x85, 0x40, 0x00, 0x00, 0x00,
- 0x29, 0x53, 0x00, 0x00, 0x00, 0x00,
- 0xa6, 0x0a, 0x80, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_7[42] =
- {
- 0x44, 0xa2, 0x40, 0x00, 0x00, 0x00,
- 0x23, 0x11, 0x80, 0x00, 0x00, 0x00,
- 0x91, 0x48, 0x80, 0x00, 0x00, 0x00,
- 0x0a, 0x85, 0x40, 0x00, 0x00, 0x00,
- 0x34, 0x1a, 0x00, 0x00, 0x00, 0x00,
- 0x0b, 0x06, 0x80, 0x00, 0x00, 0x00,
- 0xe0, 0x64, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_8[48] =
- {
- 0x16, 0x0b, 0x00, 0x00, 0x00, 0x00,
- 0xc2, 0x61, 0x00, 0x00, 0x00, 0x00,
- 0x60, 0xb0, 0x40, 0x00, 0x00, 0x00,
- 0x85, 0x42, 0x80, 0x00, 0x00, 0x00,
- 0x4c, 0x26, 0x00, 0x00, 0x00, 0x00,
- 0x29, 0x14, 0x80, 0x00, 0x00, 0x00,
- 0x11, 0x88, 0xc0, 0x00, 0x00, 0x00,
- 0xb0, 0x58, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask18_9[54] =
- {
- 0x44, 0xa2, 0x40, 0x00, 0x00, 0x00,
- 0x66, 0x26, 0x00, 0x00, 0x00, 0x00,
- 0x90, 0x49, 0x40, 0x00, 0x00, 0x00,
- 0x01, 0xa5, 0x80, 0x00, 0x00, 0x00,
- 0x0e, 0x12, 0x80, 0x00, 0x00, 0x00,
- 0x13, 0x0b, 0x00, 0x00, 0x00, 0x00,
- 0x20, 0xd0, 0x40, 0x00, 0x00, 0x00,
- 0xc2, 0x51, 0x00, 0x00, 0x00, 0x00,
- 0x29, 0x0c, 0x80, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_1[6] =
- {
- 0xff, 0xff, 0xe0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_10[60] =
- {
- 0x8c, 0xe3, 0x00, 0x00, 0x00, 0x00,
- 0x27, 0x11, 0xc0, 0x00, 0x00, 0x00,
- 0x32, 0x8d, 0x20, 0x00, 0x00, 0x00,
- 0x61, 0x92, 0x60, 0x00, 0x00, 0x00,
- 0x5c, 0x38, 0x80, 0x00, 0x00, 0x00,
- 0xcc, 0x75, 0x00, 0x00, 0x00, 0x00,
- 0x2b, 0x19, 0xc0, 0x00, 0x00, 0x00,
- 0x32, 0xd2, 0x60, 0x00, 0x00, 0x00,
- 0x25, 0x8e, 0xa0, 0x00, 0x00, 0x00,
- 0x50, 0x88, 0xc0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_11[66] =
- {
- 0x8c, 0xe3, 0x00, 0x00, 0x00, 0x00,
- 0x27, 0x11, 0xc0, 0x00, 0x00, 0x00,
- 0x32, 0x8d, 0x20, 0x00, 0x00, 0x00,
- 0x61, 0x92, 0x60, 0x00, 0x00, 0x00,
- 0x5c, 0x38, 0x80, 0x00, 0x00, 0x00,
- 0x84, 0x87, 0x00, 0x00, 0x00, 0x00,
- 0x27, 0x19, 0x80, 0x00, 0x00, 0x00,
- 0x51, 0x88, 0x60, 0x00, 0x00, 0x00,
- 0x1a, 0x22, 0xa0, 0x00, 0x00, 0x00,
- 0x68, 0x44, 0x40, 0x00, 0x00, 0x00,
- 0x89, 0x70, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_12[72] =
- {
- 0x84, 0x87, 0x00, 0x00, 0x00, 0x00,
- 0x27, 0x19, 0x80, 0x00, 0x00, 0x00,
- 0x51, 0x88, 0x60, 0x00, 0x00, 0x00,
- 0x1a, 0x22, 0xa0, 0x00, 0x00, 0x00,
- 0x68, 0x44, 0x40, 0x00, 0x00, 0x00,
- 0x89, 0x70, 0x00, 0x00, 0x00, 0x00,
- 0x8c, 0xe3, 0x00, 0x00, 0x00, 0x00,
- 0x27, 0x11, 0xc0, 0x00, 0x00, 0x00,
- 0x32, 0x8d, 0x20, 0x00, 0x00, 0x00,
- 0x61, 0x92, 0x60, 0x00, 0x00, 0x00,
- 0x5c, 0x38, 0x80, 0x00, 0x00, 0x00,
- 0x90, 0xc8, 0x80, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_13[78] =
- {
- 0x84, 0x87, 0x00, 0x00, 0x00, 0x00,
- 0x27, 0x19, 0x80, 0x00, 0x00, 0x00,
- 0x51, 0x88, 0x60, 0x00, 0x00, 0x00,
- 0x1a, 0x22, 0xa0, 0x00, 0x00, 0x00,
- 0x68, 0x44, 0x40, 0x00, 0x00, 0x00,
- 0x89, 0x70, 0x00, 0x00, 0x00, 0x00,
- 0x8c, 0x23, 0x00, 0x00, 0x00, 0x00,
- 0x47, 0x19, 0x80, 0x00, 0x00, 0x00,
- 0x81, 0x88, 0x60, 0x00, 0x00, 0x00,
- 0x12, 0x86, 0x20, 0x00, 0x00, 0x00,
- 0x58, 0x14, 0x40, 0x00, 0x00, 0x00,
- 0x28, 0xca, 0x00, 0x00, 0x00, 0x00,
- 0x34, 0x60, 0x80, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_14[84] =
- {
- 0x8c, 0x23, 0x00, 0x00, 0x00, 0x00,
- 0x47, 0x19, 0x80, 0x00, 0x00, 0x00,
- 0x81, 0x88, 0x60, 0x00, 0x00, 0x00,
- 0x12, 0x86, 0x20, 0x00, 0x00, 0x00,
- 0x58, 0x14, 0x40, 0x00, 0x00, 0x00,
- 0x28, 0xca, 0x00, 0x00, 0x00, 0x00,
- 0x34, 0x60, 0x80, 0x00, 0x00, 0x00,
- 0x84, 0x87, 0x00, 0x00, 0x00, 0x00,
- 0x27, 0x19, 0x80, 0x00, 0x00, 0x00,
- 0x51, 0x88, 0x60, 0x00, 0x00, 0x00,
- 0x1a, 0x22, 0xa0, 0x00, 0x00, 0x00,
- 0x68, 0x44, 0x40, 0x00, 0x00, 0x00,
- 0x89, 0x70, 0x00, 0x00, 0x00, 0x00,
- 0x6e, 0x27, 0x60, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_15[90] =
- {
- 0x8c, 0x23, 0x00, 0x00, 0x00, 0x00,
- 0x47, 0x19, 0x80, 0x00, 0x00, 0x00,
- 0x81, 0x88, 0x60, 0x00, 0x00, 0x00,
- 0x12, 0x86, 0x20, 0x00, 0x00, 0x00,
- 0x58, 0x14, 0x40, 0x00, 0x00, 0x00,
- 0x28, 0xca, 0x00, 0x00, 0x00, 0x00,
- 0x34, 0x60, 0x80, 0x00, 0x00, 0x00,
- 0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
- 0x91, 0x40, 0xc0, 0x00, 0x00, 0x00,
- 0xc0, 0xd0, 0x20, 0x00, 0x00, 0x00,
- 0x06, 0x82, 0xa0, 0x00, 0x00, 0x00,
- 0xc8, 0x0c, 0x40, 0x00, 0x00, 0x00,
- 0x45, 0x61, 0x00, 0x00, 0x00, 0x00,
- 0x30, 0x91, 0x40, 0x00, 0x00, 0x00,
- 0xa2, 0x28, 0x20, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_16[96] =
- {
- 0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
- 0x91, 0x40, 0xc0, 0x00, 0x00, 0x00,
- 0xc0, 0xd0, 0x20, 0x00, 0x00, 0x00,
- 0x06, 0x82, 0xa0, 0x00, 0x00, 0x00,
- 0xc8, 0x0c, 0x40, 0x00, 0x00, 0x00,
- 0x45, 0x61, 0x00, 0x00, 0x00, 0x00,
- 0x30, 0x91, 0x40, 0x00, 0x00, 0x00,
- 0xa2, 0x28, 0x20, 0x00, 0x00, 0x00,
- 0x8c, 0x23, 0x00, 0x00, 0x00, 0x00,
- 0x47, 0x19, 0x80, 0x00, 0x00, 0x00,
- 0x81, 0x88, 0x60, 0x00, 0x00, 0x00,
- 0x12, 0x86, 0x20, 0x00, 0x00, 0x00,
- 0x58, 0x14, 0x40, 0x00, 0x00, 0x00,
- 0x28, 0xca, 0x00, 0x00, 0x00, 0x00,
- 0x34, 0x60, 0x80, 0x00, 0x00, 0x00,
- 0x7e, 0x75, 0xe0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_17[102] =
- {
- 0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
- 0x91, 0x40, 0xc0, 0x00, 0x00, 0x00,
- 0xc0, 0xd0, 0x20, 0x00, 0x00, 0x00,
- 0x06, 0x82, 0xa0, 0x00, 0x00, 0x00,
- 0xc8, 0x0c, 0x40, 0x00, 0x00, 0x00,
- 0x45, 0x61, 0x00, 0x00, 0x00, 0x00,
- 0x30, 0x91, 0x40, 0x00, 0x00, 0x00,
- 0xa2, 0x28, 0x20, 0x00, 0x00, 0x00,
- 0x4c, 0x27, 0x00, 0x00, 0x00, 0x00,
- 0x66, 0x71, 0x80, 0x00, 0x00, 0x00,
- 0x91, 0x40, 0xe0, 0x00, 0x00, 0x00,
- 0x42, 0x90, 0xa0, 0x00, 0x00, 0x00,
- 0xa4, 0x29, 0x40, 0x00, 0x00, 0x00,
- 0x13, 0x5a, 0x00, 0x00, 0x00, 0x00,
- 0x30, 0x93, 0x40, 0x00, 0x00, 0x00,
- 0x88, 0xac, 0x20, 0x00, 0x00, 0x00,
- 0x09, 0x0c, 0xc0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_18[108] =
- {
- 0x4c, 0x27, 0x00, 0x00, 0x00, 0x00,
- 0x66, 0x71, 0x80, 0x00, 0x00, 0x00,
- 0x91, 0x40, 0xe0, 0x00, 0x00, 0x00,
- 0x42, 0x90, 0xa0, 0x00, 0x00, 0x00,
- 0xa4, 0x29, 0x40, 0x00, 0x00, 0x00,
- 0x13, 0x5a, 0x00, 0x00, 0x00, 0x00,
- 0x30, 0x93, 0x40, 0x00, 0x00, 0x00,
- 0x88, 0xac, 0x20, 0x00, 0x00, 0x00,
- 0x09, 0x0c, 0xc0, 0x00, 0x00, 0x00,
- 0x2c, 0x16, 0x00, 0x00, 0x00, 0x00,
- 0x91, 0x40, 0xc0, 0x00, 0x00, 0x00,
- 0xc0, 0xd0, 0x20, 0x00, 0x00, 0x00,
- 0x06, 0x82, 0xa0, 0x00, 0x00, 0x00,
- 0xc8, 0x0c, 0x40, 0x00, 0x00, 0x00,
- 0x45, 0x61, 0x00, 0x00, 0x00, 0x00,
- 0x30, 0x91, 0x40, 0x00, 0x00, 0x00,
- 0xa2, 0x28, 0x20, 0x00, 0x00, 0x00,
- 0x51, 0x97, 0x20, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_19[114] =
- {
- 0x4c, 0x27, 0x00, 0x00, 0x00, 0x00,
- 0x66, 0x71, 0x80, 0x00, 0x00, 0x00,
- 0x91, 0x40, 0xe0, 0x00, 0x00, 0x00,
- 0x42, 0x90, 0xa0, 0x00, 0x00, 0x00,
- 0xa4, 0x29, 0x40, 0x00, 0x00, 0x00,
- 0x13, 0x5a, 0x00, 0x00, 0x00, 0x00,
- 0x30, 0x93, 0x40, 0x00, 0x00, 0x00,
- 0x88, 0xac, 0x20, 0x00, 0x00, 0x00,
- 0x09, 0x0c, 0xc0, 0x00, 0x00, 0x00,
- 0x4c, 0x26, 0x00, 0x00, 0x00, 0x00,
- 0x66, 0x28, 0x80, 0x00, 0x00, 0x00,
- 0x91, 0x50, 0x20, 0x00, 0x00, 0x00,
- 0x42, 0x82, 0x60, 0x00, 0x00, 0x00,
- 0xa4, 0x01, 0xc0, 0x00, 0x00, 0x00,
- 0x13, 0x43, 0x00, 0x00, 0x00, 0x00,
- 0x30, 0x94, 0x80, 0x00, 0x00, 0x00,
- 0x88, 0xa1, 0x20, 0x00, 0x00, 0x00,
- 0x09, 0x4c, 0x00, 0x00, 0x00, 0x00,
- 0xcd, 0x98, 0x40, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_2[12] =
- {
- 0xce, 0x77, 0x00, 0x00, 0x00, 0x00,
- 0x39, 0xcc, 0xe0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_3[18] =
- {
- 0xcc, 0x67, 0x00, 0x00, 0x00, 0x00,
- 0x27, 0x2c, 0xc0, 0x00, 0x00, 0x00,
- 0x92, 0xd2, 0x60, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_4[24] =
- {
- 0xec, 0x73, 0x00, 0x00, 0x00, 0x00,
- 0x67, 0x19, 0xc0, 0x00, 0x00, 0x00,
- 0xb1, 0xcc, 0x60, 0x00, 0x00, 0x00,
- 0x5a, 0x96, 0xa0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_5[30] =
- {
- 0x4c, 0xe7, 0x00, 0x00, 0x00, 0x00,
- 0x66, 0x31, 0xc0, 0x00, 0x00, 0x00,
- 0xa1, 0xcc, 0x60, 0x00, 0x00, 0x00,
- 0x92, 0xa6, 0xa0, 0x00, 0x00, 0x00,
- 0xb8, 0x99, 0x80, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_6[36] =
- {
- 0x4c, 0x36, 0x00, 0x00, 0x00, 0x00,
- 0x45, 0x68, 0x80, 0x00, 0x00, 0x00,
- 0x30, 0xd0, 0x60, 0x00, 0x00, 0x00,
- 0x8a, 0x82, 0xa0, 0x00, 0x00, 0x00,
- 0x26, 0x0b, 0x40, 0x00, 0x00, 0x00,
- 0x95, 0x45, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_7[42] =
- {
- 0xc4, 0xa3, 0x00, 0x00, 0x00, 0x00,
- 0x23, 0x19, 0x80, 0x00, 0x00, 0x00,
- 0x91, 0x1c, 0x20, 0x00, 0x00, 0x00,
- 0x4a, 0x82, 0xa0, 0x00, 0x00, 0x00,
- 0x34, 0x49, 0x40, 0x00, 0x00, 0x00,
- 0x8b, 0x4a, 0x00, 0x00, 0x00, 0x00,
- 0xc8, 0x24, 0xc0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_8[48] =
- {
- 0x16, 0x13, 0x80, 0x00, 0x00, 0x00,
- 0xc2, 0x44, 0xc0, 0x00, 0x00, 0x00,
- 0x60, 0xe8, 0x20, 0x00, 0x00, 0x00,
- 0x85, 0x12, 0x60, 0x00, 0x00, 0x00,
- 0xcc, 0x21, 0x40, 0x00, 0x00, 0x00,
- 0x29, 0x63, 0x00, 0x00, 0x00, 0x00,
- 0x11, 0x98, 0xc0, 0x00, 0x00, 0x00,
- 0xb0, 0x0c, 0x60, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask19_9[54] =
- {
- 0x44, 0xa7, 0x00, 0x00, 0x00, 0x00,
- 0x66, 0x70, 0x80, 0x00, 0x00, 0x00,
- 0x12, 0xc0, 0xe0, 0x00, 0x00, 0x00,
- 0xc3, 0x10, 0xa0, 0x00, 0x00, 0x00,
- 0x8c, 0x29, 0x40, 0x00, 0x00, 0x00,
- 0x11, 0x5b, 0x00, 0x00, 0x00, 0x00,
- 0x21, 0x93, 0x40, 0x00, 0x00, 0x00,
- 0xa2, 0x2c, 0x00, 0x00, 0x00, 0x00,
- 0x18, 0x0c, 0xe0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask1_1[2] =
- {
- 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask20_1[6] =
- {
- 0xff, 0xff, 0xf0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_10[60] =
- {
- 0x4c, 0x13, 0x00, 0x00, 0x00, 0x00,
- 0x51, 0x14, 0x40, 0x00, 0x00, 0x00,
- 0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
- 0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
- 0x03, 0x80, 0xe0, 0x00, 0x00, 0x00,
- 0x86, 0x21, 0x80, 0x00, 0x00, 0x00,
- 0x29, 0x0a, 0x40, 0x00, 0x00, 0x00,
- 0x42, 0x50, 0x90, 0x00, 0x00, 0x00,
- 0x98, 0x26, 0x00, 0x00, 0x00, 0x00,
- 0x30, 0x8c, 0x20, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_11[66] =
- {
- 0xc6, 0x31, 0x80, 0x00, 0x00, 0x00,
- 0x23, 0x88, 0xe0, 0x00, 0x00, 0x00,
- 0x1a, 0x46, 0x90, 0x00, 0x00, 0x00,
- 0x24, 0xc9, 0x30, 0x00, 0x00, 0x00,
- 0x71, 0x1c, 0x40, 0x00, 0x00, 0x00,
- 0x0e, 0x03, 0x80, 0x00, 0x00, 0x00,
- 0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
- 0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
- 0x45, 0x51, 0x50, 0x00, 0x00, 0x00,
- 0x88, 0xa2, 0x20, 0x00, 0x00, 0x00,
- 0xe0, 0x38, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_12[72] =
- {
- 0x0e, 0x03, 0x80, 0x00, 0x00, 0x00,
- 0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
- 0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
- 0x45, 0x51, 0x50, 0x00, 0x00, 0x00,
- 0x88, 0xa2, 0x20, 0x00, 0x00, 0x00,
- 0xe0, 0x38, 0x00, 0x00, 0x00, 0x00,
- 0xc6, 0x31, 0x80, 0x00, 0x00, 0x00,
- 0x23, 0x88, 0xe0, 0x00, 0x00, 0x00,
- 0x1a, 0x46, 0x90, 0x00, 0x00, 0x00,
- 0x24, 0xc9, 0x30, 0x00, 0x00, 0x00,
- 0x71, 0x1c, 0x40, 0x00, 0x00, 0x00,
- 0xf5, 0xdc, 0x40, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_13[78] =
- {
- 0x0e, 0x03, 0x80, 0x00, 0x00, 0x00,
- 0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
- 0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
- 0x45, 0x51, 0x50, 0x00, 0x00, 0x00,
- 0x88, 0xa2, 0x20, 0x00, 0x00, 0x00,
- 0xe0, 0x38, 0x00, 0x00, 0x00, 0x00,
- 0x46, 0x11, 0x80, 0x00, 0x00, 0x00,
- 0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
- 0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
- 0x0c, 0x43, 0x10, 0x00, 0x00, 0x00,
- 0x28, 0x8a, 0x20, 0x00, 0x00, 0x00,
- 0x94, 0x25, 0x00, 0x00, 0x00, 0x00,
- 0xc1, 0x30, 0x40, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_14[84] =
- {
- 0x46, 0x11, 0x80, 0x00, 0x00, 0x00,
- 0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
- 0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
- 0x0c, 0x43, 0x10, 0x00, 0x00, 0x00,
- 0x28, 0x8a, 0x20, 0x00, 0x00, 0x00,
- 0x94, 0x25, 0x00, 0x00, 0x00, 0x00,
- 0xc1, 0x30, 0x40, 0x00, 0x00, 0x00,
- 0x0e, 0x03, 0x80, 0x00, 0x00, 0x00,
- 0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
- 0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
- 0x45, 0x51, 0x50, 0x00, 0x00, 0x00,
- 0x88, 0xa2, 0x20, 0x00, 0x00, 0x00,
- 0xe0, 0x38, 0x00, 0x00, 0x00, 0x00,
- 0x56, 0x3e, 0x20, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_15[90] =
- {
- 0x46, 0x11, 0x80, 0x00, 0x00, 0x00,
- 0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
- 0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
- 0x0c, 0x43, 0x10, 0x00, 0x00, 0x00,
- 0x28, 0x8a, 0x20, 0x00, 0x00, 0x00,
- 0x94, 0x25, 0x00, 0x00, 0x00, 0x00,
- 0xc1, 0x30, 0x40, 0x00, 0x00, 0x00,
- 0x2c, 0x0b, 0x00, 0x00, 0x00, 0x00,
- 0x81, 0xa0, 0x60, 0x00, 0x00, 0x00,
- 0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
- 0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
- 0x18, 0x86, 0x20, 0x00, 0x00, 0x00,
- 0xc2, 0x30, 0x80, 0x00, 0x00, 0x00,
- 0x22, 0x88, 0xa0, 0x00, 0x00, 0x00,
- 0x50, 0x54, 0x10, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_16[96] =
- {
- 0x2c, 0x0b, 0x00, 0x00, 0x00, 0x00,
- 0x81, 0xa0, 0x60, 0x00, 0x00, 0x00,
- 0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
- 0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
- 0x18, 0x86, 0x20, 0x00, 0x00, 0x00,
- 0xc2, 0x30, 0x80, 0x00, 0x00, 0x00,
- 0x22, 0x88, 0xa0, 0x00, 0x00, 0x00,
- 0x50, 0x54, 0x10, 0x00, 0x00, 0x00,
- 0x46, 0x11, 0x80, 0x00, 0x00, 0x00,
- 0x33, 0x0c, 0xc0, 0x00, 0x00, 0x00,
- 0x10, 0xc4, 0x30, 0x00, 0x00, 0x00,
- 0x0c, 0x43, 0x10, 0x00, 0x00, 0x00,
- 0x28, 0x8a, 0x20, 0x00, 0x00, 0x00,
- 0x94, 0x25, 0x00, 0x00, 0x00, 0x00,
- 0xc1, 0x30, 0x40, 0x00, 0x00, 0x00,
- 0x28, 0x1c, 0x60, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_17[102] =
- {
- 0x2c, 0x0b, 0x00, 0x00, 0x00, 0x00,
- 0x81, 0xa0, 0x60, 0x00, 0x00, 0x00,
- 0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
- 0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
- 0x18, 0x86, 0x20, 0x00, 0x00, 0x00,
- 0xc2, 0x30, 0x80, 0x00, 0x00, 0x00,
- 0x22, 0x88, 0xa0, 0x00, 0x00, 0x00,
- 0x50, 0x54, 0x10, 0x00, 0x00, 0x00,
- 0x4e, 0x13, 0x80, 0x00, 0x00, 0x00,
- 0xe3, 0x38, 0xc0, 0x00, 0x00, 0x00,
- 0x81, 0xe0, 0x70, 0x00, 0x00, 0x00,
- 0x21, 0x48, 0x50, 0x00, 0x00, 0x00,
- 0x52, 0x94, 0xa0, 0x00, 0x00, 0x00,
- 0xb4, 0x2d, 0x00, 0x00, 0x00, 0x00,
- 0x26, 0x89, 0xa0, 0x00, 0x00, 0x00,
- 0x58, 0x56, 0x10, 0x00, 0x00, 0x00,
- 0x19, 0x86, 0x60, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_18[108] =
- {
- 0x4e, 0x13, 0x80, 0x00, 0x00, 0x00,
- 0xe3, 0x38, 0xc0, 0x00, 0x00, 0x00,
- 0x81, 0xe0, 0x70, 0x00, 0x00, 0x00,
- 0x21, 0x48, 0x50, 0x00, 0x00, 0x00,
- 0x52, 0x94, 0xa0, 0x00, 0x00, 0x00,
- 0xb4, 0x2d, 0x00, 0x00, 0x00, 0x00,
- 0x26, 0x89, 0xa0, 0x00, 0x00, 0x00,
- 0x58, 0x56, 0x10, 0x00, 0x00, 0x00,
- 0x19, 0x86, 0x60, 0x00, 0x00, 0x00,
- 0x2c, 0x0b, 0x00, 0x00, 0x00, 0x00,
- 0x81, 0xa0, 0x60, 0x00, 0x00, 0x00,
- 0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
- 0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
- 0x18, 0x86, 0x20, 0x00, 0x00, 0x00,
- 0xc2, 0x30, 0x80, 0x00, 0x00, 0x00,
- 0x22, 0x88, 0xa0, 0x00, 0x00, 0x00,
- 0x50, 0x54, 0x10, 0x00, 0x00, 0x00,
- 0x21, 0x7b, 0xf0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_19[114] =
- {
- 0x4e, 0x13, 0x80, 0x00, 0x00, 0x00,
- 0xe3, 0x38, 0xc0, 0x00, 0x00, 0x00,
- 0x81, 0xe0, 0x70, 0x00, 0x00, 0x00,
- 0x21, 0x48, 0x50, 0x00, 0x00, 0x00,
- 0x52, 0x94, 0xa0, 0x00, 0x00, 0x00,
- 0xb4, 0x2d, 0x00, 0x00, 0x00, 0x00,
- 0x26, 0x89, 0xa0, 0x00, 0x00, 0x00,
- 0x58, 0x56, 0x10, 0x00, 0x00, 0x00,
- 0x19, 0x86, 0x60, 0x00, 0x00, 0x00,
- 0x4c, 0x13, 0x00, 0x00, 0x00, 0x00,
- 0x51, 0x14, 0x40, 0x00, 0x00, 0x00,
- 0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
- 0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
- 0x03, 0x80, 0xe0, 0x00, 0x00, 0x00,
- 0x86, 0x21, 0x80, 0x00, 0x00, 0x00,
- 0x29, 0x0a, 0x40, 0x00, 0x00, 0x00,
- 0x42, 0x50, 0x90, 0x00, 0x00, 0x00,
- 0x98, 0x26, 0x00, 0x00, 0x00, 0x00,
- 0x30, 0x8c, 0x20, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_2[12] =
- {
- 0xee, 0x3b, 0x80, 0x00, 0x00, 0x00,
- 0x99, 0xe6, 0x70, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_20[120] =
- {
- 0x4c, 0x13, 0x00, 0x00, 0x00, 0x00,
- 0x51, 0x14, 0x40, 0x00, 0x00, 0x00,
- 0xa0, 0x68, 0x10, 0x00, 0x00, 0x00,
- 0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
- 0x03, 0x80, 0xe0, 0x00, 0x00, 0x00,
- 0x86, 0x21, 0x80, 0x00, 0x00, 0x00,
- 0x29, 0x0a, 0x40, 0x00, 0x00, 0x00,
- 0x42, 0x50, 0x90, 0x00, 0x00, 0x00,
- 0x98, 0x26, 0x00, 0x00, 0x00, 0x00,
- 0x30, 0x8c, 0x20, 0x00, 0x00, 0x00,
- 0x4e, 0x13, 0x80, 0x00, 0x00, 0x00,
- 0xe3, 0x38, 0xc0, 0x00, 0x00, 0x00,
- 0x81, 0xe0, 0x70, 0x00, 0x00, 0x00,
- 0x21, 0x48, 0x50, 0x00, 0x00, 0x00,
- 0x52, 0x94, 0xa0, 0x00, 0x00, 0x00,
- 0xb4, 0x2d, 0x00, 0x00, 0x00, 0x00,
- 0x26, 0x89, 0xa0, 0x00, 0x00, 0x00,
- 0x58, 0x56, 0x10, 0x00, 0x00, 0x00,
- 0x19, 0x86, 0x60, 0x00, 0x00, 0x00,
- 0xf7, 0x8d, 0xa0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_3[18] =
- {
- 0xce, 0x33, 0x80, 0x00, 0x00, 0x00,
- 0x55, 0x95, 0x60, 0x00, 0x00, 0x00,
- 0xb1, 0x6a, 0x30, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_4[24] =
- {
- 0xe6, 0x39, 0x80, 0x00, 0x00, 0x00,
- 0x33, 0x8c, 0xe0, 0x00, 0x00, 0x00,
- 0x98, 0xe6, 0x30, 0x00, 0x00, 0x00,
- 0x2d, 0x4b, 0x50, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_5[30] =
- {
- 0xce, 0x33, 0x80, 0x00, 0x00, 0x00,
- 0x63, 0x98, 0xe0, 0x00, 0x00, 0x00,
- 0x98, 0xe5, 0x30, 0x00, 0x00, 0x00,
- 0x2b, 0x53, 0x50, 0x00, 0x00, 0x00,
- 0xb4, 0x5c, 0xa0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_6[36] =
- {
- 0x4c, 0x1b, 0x00, 0x00, 0x00, 0x00,
- 0x51, 0x34, 0x40, 0x00, 0x00, 0x00,
- 0x20, 0xe8, 0x30, 0x00, 0x00, 0x00,
- 0x85, 0x41, 0x50, 0x00, 0x00, 0x00,
- 0x06, 0x86, 0xa0, 0x00, 0x00, 0x00,
- 0x9a, 0x21, 0x80, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_7[42] =
- {
- 0x4e, 0x11, 0x80, 0x00, 0x00, 0x00,
- 0x33, 0x2c, 0x00, 0x00, 0x00, 0x00,
- 0x10, 0x0e, 0xb0, 0x00, 0x00, 0x00,
- 0x81, 0x51, 0x50, 0x00, 0x00, 0x00,
- 0x24, 0xc4, 0xa0, 0x00, 0x00, 0x00,
- 0xd4, 0x23, 0x00, 0x00, 0x00, 0x00,
- 0x0c, 0xa2, 0x60, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_8[48] =
- {
- 0x27, 0x09, 0xc0, 0x00, 0x00, 0x00,
- 0x89, 0xa2, 0x60, 0x00, 0x00, 0x00,
- 0xd0, 0x74, 0x10, 0x00, 0x00, 0x00,
- 0x24, 0xc9, 0x30, 0x00, 0x00, 0x00,
- 0xe2, 0x90, 0xa0, 0x00, 0x00, 0x00,
- 0xc6, 0x31, 0x80, 0x00, 0x00, 0x00,
- 0x31, 0x8c, 0x60, 0x00, 0x00, 0x00,
- 0x18, 0xc6, 0x30, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask20_9[54] =
- {
- 0x4e, 0x13, 0x80, 0x00, 0x00, 0x00,
- 0x62, 0x38, 0xc0, 0x00, 0x00, 0x00,
- 0x81, 0xe0, 0x70, 0x00, 0x00, 0x00,
- 0xe1, 0x48, 0x50, 0x00, 0x00, 0x00,
- 0x13, 0x94, 0xa0, 0x00, 0x00, 0x00,
- 0xb4, 0x2d, 0x00, 0x00, 0x00, 0x00,
- 0x26, 0x89, 0xa0, 0x00, 0x00, 0x00,
- 0x58, 0x56, 0x10, 0x00, 0x00, 0x00,
- 0x49, 0x86, 0x50, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_1[6] =
- {
- 0xff, 0xff, 0xf8, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_10[60] =
- {
- 0x4c, 0x19, 0x10, 0x00, 0x00, 0x00,
- 0x51, 0x14, 0x50, 0x00, 0x00, 0x00,
- 0xa0, 0x6a, 0x40, 0x00, 0x00, 0x00,
- 0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
- 0x03, 0xb4, 0x00, 0x00, 0x00, 0x00,
- 0x86, 0x20, 0x90, 0x00, 0x00, 0x00,
- 0x29, 0x08, 0x48, 0x00, 0x00, 0x00,
- 0x42, 0x43, 0x08, 0x00, 0x00, 0x00,
- 0x98, 0x12, 0x80, 0x00, 0x00, 0x00,
- 0x30, 0x84, 0xa8, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_11[66] =
- {
- 0xc6, 0x21, 0xa0, 0x00, 0x00, 0x00,
- 0x23, 0x88, 0xc8, 0x00, 0x00, 0x00,
- 0x1a, 0x45, 0x88, 0x00, 0x00, 0x00,
- 0x24, 0xd3, 0x08, 0x00, 0x00, 0x00,
- 0x71, 0x10, 0x70, 0x00, 0x00, 0x00,
- 0x0e, 0x19, 0x10, 0x00, 0x00, 0x00,
- 0x33, 0x14, 0x50, 0x00, 0x00, 0x00,
- 0x10, 0xc3, 0x28, 0x00, 0x00, 0x00,
- 0x45, 0x68, 0x48, 0x00, 0x00, 0x00,
- 0x88, 0x84, 0xa8, 0x00, 0x00, 0x00,
- 0xe0, 0x22, 0x90, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_12[72] =
- {
- 0x0e, 0x19, 0x10, 0x00, 0x00, 0x00,
- 0x33, 0x14, 0x50, 0x00, 0x00, 0x00,
- 0x10, 0xc3, 0x28, 0x00, 0x00, 0x00,
- 0x45, 0x68, 0x48, 0x00, 0x00, 0x00,
- 0x88, 0x84, 0xa8, 0x00, 0x00, 0x00,
- 0xe0, 0x22, 0x90, 0x00, 0x00, 0x00,
- 0xc6, 0x21, 0xa0, 0x00, 0x00, 0x00,
- 0x23, 0x88, 0xc8, 0x00, 0x00, 0x00,
- 0x1a, 0x45, 0x88, 0x00, 0x00, 0x00,
- 0x24, 0xd3, 0x08, 0x00, 0x00, 0x00,
- 0x71, 0x10, 0x70, 0x00, 0x00, 0x00,
- 0xa0, 0x65, 0x18, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_13[78] =
- {
- 0x0e, 0x19, 0x10, 0x00, 0x00, 0x00,
- 0x33, 0x14, 0x50, 0x00, 0x00, 0x00,
- 0x10, 0xc3, 0x28, 0x00, 0x00, 0x00,
- 0x45, 0x68, 0x48, 0x00, 0x00, 0x00,
- 0x88, 0x84, 0xa8, 0x00, 0x00, 0x00,
- 0xe0, 0x22, 0x90, 0x00, 0x00, 0x00,
- 0x46, 0x11, 0x90, 0x00, 0x00, 0x00,
- 0x33, 0x0c, 0xc8, 0x00, 0x00, 0x00,
- 0x10, 0xe4, 0x60, 0x00, 0x00, 0x00,
- 0x0c, 0x69, 0x08, 0x00, 0x00, 0x00,
- 0x28, 0x94, 0x28, 0x00, 0x00, 0x00,
- 0x94, 0x21, 0x30, 0x00, 0x00, 0x00,
- 0xc1, 0x02, 0x58, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_14[84] =
- {
- 0x46, 0x11, 0x90, 0x00, 0x00, 0x00,
- 0x33, 0x0c, 0xc8, 0x00, 0x00, 0x00,
- 0x10, 0xe4, 0x60, 0x00, 0x00, 0x00,
- 0x0c, 0x69, 0x08, 0x00, 0x00, 0x00,
- 0x28, 0x94, 0x28, 0x00, 0x00, 0x00,
- 0x94, 0x21, 0x30, 0x00, 0x00, 0x00,
- 0xc1, 0x02, 0x58, 0x00, 0x00, 0x00,
- 0x0e, 0x19, 0x10, 0x00, 0x00, 0x00,
- 0x33, 0x14, 0x50, 0x00, 0x00, 0x00,
- 0x10, 0xc3, 0x28, 0x00, 0x00, 0x00,
- 0x45, 0x68, 0x48, 0x00, 0x00, 0x00,
- 0x88, 0x84, 0xa8, 0x00, 0x00, 0x00,
- 0xe0, 0x22, 0x90, 0x00, 0x00, 0x00,
- 0x4d, 0xd0, 0xc0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_15[90] =
- {
- 0x46, 0x11, 0x90, 0x00, 0x00, 0x00,
- 0x33, 0x0c, 0xc8, 0x00, 0x00, 0x00,
- 0x10, 0xe4, 0x60, 0x00, 0x00, 0x00,
- 0x0c, 0x69, 0x08, 0x00, 0x00, 0x00,
- 0x28, 0x94, 0x28, 0x00, 0x00, 0x00,
- 0x94, 0x21, 0x30, 0x00, 0x00, 0x00,
- 0xc1, 0x02, 0x58, 0x00, 0x00, 0x00,
- 0x2c, 0x03, 0x20, 0x00, 0x00, 0x00,
- 0x81, 0xa0, 0x18, 0x00, 0x00, 0x00,
- 0xa0, 0x68, 0x20, 0x00, 0x00, 0x00,
- 0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
- 0x18, 0x90, 0xc0, 0x00, 0x00, 0x00,
- 0xc2, 0x06, 0x80, 0x00, 0x00, 0x00,
- 0x22, 0x98, 0x08, 0x00, 0x00, 0x00,
- 0x50, 0x45, 0x08, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_16[96] =
- {
- 0x2c, 0x03, 0x20, 0x00, 0x00, 0x00,
- 0x81, 0xa0, 0x18, 0x00, 0x00, 0x00,
- 0xa0, 0x68, 0x20, 0x00, 0x00, 0x00,
- 0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
- 0x18, 0x90, 0xc0, 0x00, 0x00, 0x00,
- 0xc2, 0x06, 0x80, 0x00, 0x00, 0x00,
- 0x22, 0x98, 0x08, 0x00, 0x00, 0x00,
- 0x50, 0x45, 0x08, 0x00, 0x00, 0x00,
- 0x46, 0x11, 0x90, 0x00, 0x00, 0x00,
- 0x33, 0x0c, 0xc8, 0x00, 0x00, 0x00,
- 0x10, 0xe4, 0x60, 0x00, 0x00, 0x00,
- 0x0c, 0x69, 0x08, 0x00, 0x00, 0x00,
- 0x28, 0x94, 0x28, 0x00, 0x00, 0x00,
- 0x94, 0x21, 0x30, 0x00, 0x00, 0x00,
- 0xc1, 0x02, 0x58, 0x00, 0x00, 0x00,
- 0x3b, 0xf5, 0x38, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_17[102] =
- {
- 0x2c, 0x03, 0x20, 0x00, 0x00, 0x00,
- 0x81, 0xa0, 0x18, 0x00, 0x00, 0x00,
- 0xa0, 0x68, 0x20, 0x00, 0x00, 0x00,
- 0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
- 0x18, 0x90, 0xc0, 0x00, 0x00, 0x00,
- 0xc2, 0x06, 0x80, 0x00, 0x00, 0x00,
- 0x22, 0x98, 0x08, 0x00, 0x00, 0x00,
- 0x50, 0x45, 0x08, 0x00, 0x00, 0x00,
- 0x4e, 0x11, 0x90, 0x00, 0x00, 0x00,
- 0xe3, 0x18, 0x98, 0x00, 0x00, 0x00,
- 0x81, 0xe3, 0x00, 0x00, 0x00, 0x00,
- 0x21, 0x40, 0x58, 0x00, 0x00, 0x00,
- 0x52, 0x81, 0xe0, 0x00, 0x00, 0x00,
- 0xb4, 0x28, 0x20, 0x00, 0x00, 0x00,
- 0x26, 0x86, 0x28, 0x00, 0x00, 0x00,
- 0x58, 0x64, 0x40, 0x00, 0x00, 0x00,
- 0x19, 0x9e, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_18[108] =
- {
- 0x4e, 0x11, 0x90, 0x00, 0x00, 0x00,
- 0xe3, 0x18, 0x98, 0x00, 0x00, 0x00,
- 0x81, 0xe3, 0x00, 0x00, 0x00, 0x00,
- 0x21, 0x40, 0x58, 0x00, 0x00, 0x00,
- 0x52, 0x81, 0xe0, 0x00, 0x00, 0x00,
- 0xb4, 0x28, 0x20, 0x00, 0x00, 0x00,
- 0x26, 0x86, 0x28, 0x00, 0x00, 0x00,
- 0x58, 0x64, 0x40, 0x00, 0x00, 0x00,
- 0x19, 0x9e, 0x00, 0x00, 0x00, 0x00,
- 0x2c, 0x03, 0x20, 0x00, 0x00, 0x00,
- 0x81, 0xa0, 0x18, 0x00, 0x00, 0x00,
- 0xa0, 0x68, 0x20, 0x00, 0x00, 0x00,
- 0x05, 0x41, 0x50, 0x00, 0x00, 0x00,
- 0x18, 0x90, 0xc0, 0x00, 0x00, 0x00,
- 0xc2, 0x06, 0x80, 0x00, 0x00, 0x00,
- 0x22, 0x98, 0x08, 0x00, 0x00, 0x00,
- 0x50, 0x45, 0x08, 0x00, 0x00, 0x00,
- 0x5a, 0x56, 0x58, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_19[114] =
- {
- 0x4e, 0x11, 0x90, 0x00, 0x00, 0x00,
- 0xe3, 0x18, 0x98, 0x00, 0x00, 0x00,
- 0x81, 0xe3, 0x00, 0x00, 0x00, 0x00,
- 0x21, 0x40, 0x58, 0x00, 0x00, 0x00,
- 0x52, 0x81, 0xe0, 0x00, 0x00, 0x00,
- 0xb4, 0x28, 0x20, 0x00, 0x00, 0x00,
- 0x26, 0x86, 0x28, 0x00, 0x00, 0x00,
- 0x58, 0x64, 0x40, 0x00, 0x00, 0x00,
- 0x19, 0x9e, 0x00, 0x00, 0x00, 0x00,
- 0x4c, 0x19, 0x10, 0x00, 0x00, 0x00,
- 0x51, 0x14, 0x50, 0x00, 0x00, 0x00,
- 0xa0, 0x6a, 0x40, 0x00, 0x00, 0x00,
- 0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
- 0x03, 0xb4, 0x00, 0x00, 0x00, 0x00,
- 0x86, 0x20, 0x90, 0x00, 0x00, 0x00,
- 0x29, 0x08, 0x48, 0x00, 0x00, 0x00,
- 0x42, 0x43, 0x08, 0x00, 0x00, 0x00,
- 0x98, 0x12, 0x80, 0x00, 0x00, 0x00,
- 0x30, 0x84, 0xa8, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_2[12] =
- {
- 0xee, 0x3b, 0x30, 0x00, 0x00, 0x00,
- 0x99, 0xe6, 0xe8, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_20[120] =
- {
- 0x4c, 0x19, 0x10, 0x00, 0x00, 0x00,
- 0x51, 0x14, 0x50, 0x00, 0x00, 0x00,
- 0xa0, 0x6a, 0x40, 0x00, 0x00, 0x00,
- 0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
- 0x03, 0xb4, 0x00, 0x00, 0x00, 0x00,
- 0x86, 0x20, 0x90, 0x00, 0x00, 0x00,
- 0x29, 0x08, 0x48, 0x00, 0x00, 0x00,
- 0x42, 0x43, 0x08, 0x00, 0x00, 0x00,
- 0x98, 0x12, 0x80, 0x00, 0x00, 0x00,
- 0x30, 0x84, 0xa8, 0x00, 0x00, 0x00,
- 0x4e, 0x11, 0x90, 0x00, 0x00, 0x00,
- 0xe3, 0x18, 0x98, 0x00, 0x00, 0x00,
- 0x81, 0xe3, 0x00, 0x00, 0x00, 0x00,
- 0x21, 0x40, 0x58, 0x00, 0x00, 0x00,
- 0x52, 0x81, 0xe0, 0x00, 0x00, 0x00,
- 0xb4, 0x28, 0x20, 0x00, 0x00, 0x00,
- 0x26, 0x86, 0x28, 0x00, 0x00, 0x00,
- 0x58, 0x64, 0x40, 0x00, 0x00, 0x00,
- 0x19, 0x9e, 0x00, 0x00, 0x00, 0x00,
- 0x2a, 0x03, 0x30, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_21[126] =
- {
- 0x4c, 0x19, 0x10, 0x00, 0x00, 0x00,
- 0x51, 0x14, 0x50, 0x00, 0x00, 0x00,
- 0xa0, 0x6a, 0x40, 0x00, 0x00, 0x00,
- 0x04, 0xc1, 0x30, 0x00, 0x00, 0x00,
- 0x03, 0xb4, 0x00, 0x00, 0x00, 0x00,
- 0x86, 0x20, 0x90, 0x00, 0x00, 0x00,
- 0x29, 0x08, 0x48, 0x00, 0x00, 0x00,
- 0x42, 0x43, 0x08, 0x00, 0x00, 0x00,
- 0x98, 0x12, 0x80, 0x00, 0x00, 0x00,
- 0x30, 0x84, 0xa8, 0x00, 0x00, 0x00,
- 0x4c, 0x11, 0x90, 0x00, 0x00, 0x00,
- 0x51, 0x0c, 0xc8, 0x00, 0x00, 0x00,
- 0xa0, 0x66, 0x40, 0x00, 0x00, 0x00,
- 0x04, 0xc1, 0x60, 0x00, 0x00, 0x00,
- 0x03, 0xa0, 0x28, 0x00, 0x00, 0x00,
- 0x86, 0x21, 0x10, 0x00, 0x00, 0x00,
- 0x29, 0x10, 0x18, 0x00, 0x00, 0x00,
- 0x42, 0x42, 0xa0, 0x00, 0x00, 0x00,
- 0x98, 0x1a, 0x00, 0x00, 0x00, 0x00,
- 0x30, 0x84, 0x08, 0x00, 0x00, 0x00,
- 0xdf, 0x4c, 0x10, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_3[18] =
- {
- 0xce, 0x32, 0xb0, 0x00, 0x00, 0x00,
- 0x55, 0xdc, 0x50, 0x00, 0x00, 0x00,
- 0xa8, 0xed, 0x88, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_4[24] =
- {
- 0xe6, 0x31, 0x30, 0x00, 0x00, 0x00,
- 0x33, 0x8c, 0x58, 0x00, 0x00, 0x00,
- 0x98, 0xd2, 0xc8, 0x00, 0x00, 0x00,
- 0x2d, 0x4b, 0x28, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_5[30] =
- {
- 0xce, 0x31, 0xb0, 0x00, 0x00, 0x00,
- 0x63, 0x98, 0xd8, 0x00, 0x00, 0x00,
- 0x98, 0xc7, 0x68, 0x00, 0x00, 0x00,
- 0x4d, 0x6b, 0x50, 0x00, 0x00, 0x00,
- 0xb2, 0x6c, 0xa8, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_6[36] =
- {
- 0x4c, 0x19, 0x10, 0x00, 0x00, 0x00,
- 0x51, 0x14, 0x50, 0x00, 0x00, 0x00,
- 0x20, 0xea, 0x08, 0x00, 0x00, 0x00,
- 0x85, 0x41, 0x28, 0x00, 0x00, 0x00,
- 0x06, 0x80, 0xd8, 0x00, 0x00, 0x00,
- 0x8a, 0x24, 0x30, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_7[42] =
- {
- 0xc6, 0x11, 0x90, 0x00, 0x00, 0x00,
- 0x33, 0x04, 0xc8, 0x00, 0x00, 0x00,
- 0x18, 0x67, 0x40, 0x00, 0x00, 0x00,
- 0x45, 0x42, 0xd0, 0x00, 0x00, 0x00,
- 0x12, 0xd4, 0x28, 0x00, 0x00, 0x00,
- 0xb4, 0x28, 0x30, 0x00, 0x00, 0x00,
- 0x29, 0x92, 0x18, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_8[48] =
- {
- 0x07, 0x0a, 0x70, 0x00, 0x00, 0x00,
- 0x49, 0xa8, 0x28, 0x00, 0x00, 0x00,
- 0xb0, 0x7a, 0x00, 0x00, 0x00, 0x00,
- 0x24, 0xc5, 0xc0, 0x00, 0x00, 0x00,
- 0x52, 0x80, 0xe8, 0x00, 0x00, 0x00,
- 0xc6, 0x31, 0x80, 0x00, 0x00, 0x00,
- 0x31, 0x94, 0x18, 0x00, 0x00, 0x00,
- 0x18, 0xc7, 0x08, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask21_9[54] =
- {
- 0x4e, 0x11, 0x10, 0x00, 0x00, 0x00,
- 0x62, 0x1a, 0x08, 0x00, 0x00, 0x00,
- 0x80, 0xe9, 0x40, 0x00, 0x00, 0x00,
- 0xa1, 0x50, 0x50, 0x00, 0x00, 0x00,
- 0x53, 0x00, 0x68, 0x00, 0x00, 0x00,
- 0xa4, 0x24, 0x30, 0x00, 0x00, 0x00,
- 0x16, 0xa0, 0x88, 0x00, 0x00, 0x00,
- 0x58, 0x45, 0x20, 0x00, 0x00, 0x00,
- 0x29, 0x86, 0x80, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_1[6] =
- {
- 0xff, 0xff, 0xfc, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_10[60] =
- {
- 0xc0, 0x38, 0x88, 0x00, 0x00, 0x00,
- 0x30, 0x0e, 0x28, 0x00, 0x00, 0x00,
- 0xe8, 0x07, 0x00, 0x00, 0x00, 0x00,
- 0x85, 0x08, 0xa8, 0x00, 0x00, 0x00,
- 0xd0, 0x92, 0x10, 0x00, 0x00, 0x00,
- 0x86, 0x50, 0x48, 0x00, 0x00, 0x00,
- 0x4a, 0x68, 0x0c, 0x00, 0x00, 0x00,
- 0x01, 0xa0, 0x74, 0x00, 0x00, 0x00,
- 0x4c, 0x81, 0x90, 0x00, 0x00, 0x00,
- 0x62, 0x24, 0x04, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_11[66] =
- {
- 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
- 0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
- 0x99, 0x13, 0x20, 0x00, 0x00, 0x00,
- 0x05, 0x80, 0xb0, 0x00, 0x00, 0x00,
- 0x80, 0xb0, 0x14, 0x00, 0x00, 0x00,
- 0x84, 0x50, 0x88, 0x00, 0x00, 0x00,
- 0x40, 0x68, 0x0c, 0x00, 0x00, 0x00,
- 0x0a, 0x81, 0x50, 0x00, 0x00, 0x00,
- 0x68, 0x0d, 0x00, 0x00, 0x00, 0x00,
- 0x10, 0x22, 0x04, 0x00, 0x00, 0x00,
- 0x30, 0x46, 0x08, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_12[72] =
- {
- 0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
- 0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
- 0x0c, 0xa1, 0x94, 0x00, 0x00, 0x00,
- 0xa1, 0x34, 0x24, 0x00, 0x00, 0x00,
- 0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
- 0x8a, 0x51, 0x48, 0x00, 0x00, 0x00,
- 0x86, 0x90, 0xd0, 0x00, 0x00, 0x00,
- 0x23, 0x24, 0x64, 0x00, 0x00, 0x00,
- 0x16, 0x22, 0xc4, 0x00, 0x00, 0x00,
- 0x4c, 0x29, 0x84, 0x00, 0x00, 0x00,
- 0x41, 0xc8, 0x38, 0x00, 0x00, 0x00,
- 0xf4, 0x18, 0x9c, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_13[78] =
- {
- 0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
- 0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
- 0x0c, 0xa1, 0x94, 0x00, 0x00, 0x00,
- 0xa1, 0x34, 0x24, 0x00, 0x00, 0x00,
- 0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
- 0x8a, 0x51, 0x48, 0x00, 0x00, 0x00,
- 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
- 0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
- 0x91, 0x92, 0x30, 0x00, 0x00, 0x00,
- 0xa4, 0x34, 0x84, 0x00, 0x00, 0x00,
- 0x50, 0xaa, 0x14, 0x00, 0x00, 0x00,
- 0x84, 0xd0, 0x98, 0x00, 0x00, 0x00,
- 0x09, 0x61, 0x2c, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_14[84] =
- {
- 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
- 0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
- 0x91, 0x92, 0x30, 0x00, 0x00, 0x00,
- 0xa4, 0x34, 0x84, 0x00, 0x00, 0x00,
- 0x50, 0xaa, 0x14, 0x00, 0x00, 0x00,
- 0x84, 0xd0, 0x98, 0x00, 0x00, 0x00,
- 0x09, 0x61, 0x2c, 0x00, 0x00, 0x00,
- 0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
- 0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
- 0x0c, 0xa1, 0x94, 0x00, 0x00, 0x00,
- 0xa1, 0x34, 0x24, 0x00, 0x00, 0x00,
- 0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
- 0x8a, 0x51, 0x48, 0x00, 0x00, 0x00,
- 0xc6, 0xca, 0xe8, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_15[90] =
- {
- 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
- 0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
- 0x91, 0x92, 0x30, 0x00, 0x00, 0x00,
- 0xa4, 0x34, 0x84, 0x00, 0x00, 0x00,
- 0x50, 0xaa, 0x14, 0x00, 0x00, 0x00,
- 0x84, 0xd0, 0x98, 0x00, 0x00, 0x00,
- 0x09, 0x61, 0x2c, 0x00, 0x00, 0x00,
- 0x0c, 0x81, 0x90, 0x00, 0x00, 0x00,
- 0x80, 0x70, 0x0c, 0x00, 0x00, 0x00,
- 0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
- 0x05, 0x40, 0xa8, 0x00, 0x00, 0x00,
- 0x43, 0x08, 0x60, 0x00, 0x00, 0x00,
- 0x1a, 0x03, 0x40, 0x00, 0x00, 0x00,
- 0x60, 0x2c, 0x04, 0x00, 0x00, 0x00,
- 0x14, 0x22, 0x84, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_16[96] =
- {
- 0x0c, 0x81, 0x90, 0x00, 0x00, 0x00,
- 0x80, 0x70, 0x0c, 0x00, 0x00, 0x00,
- 0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
- 0x05, 0x40, 0xa8, 0x00, 0x00, 0x00,
- 0x43, 0x08, 0x60, 0x00, 0x00, 0x00,
- 0x1a, 0x03, 0x40, 0x00, 0x00, 0x00,
- 0x60, 0x2c, 0x04, 0x00, 0x00, 0x00,
- 0x14, 0x22, 0x84, 0x00, 0x00, 0x00,
- 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
- 0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
- 0x91, 0x92, 0x30, 0x00, 0x00, 0x00,
- 0xa4, 0x34, 0x84, 0x00, 0x00, 0x00,
- 0x50, 0xaa, 0x14, 0x00, 0x00, 0x00,
- 0x84, 0xd0, 0x98, 0x00, 0x00, 0x00,
- 0x09, 0x61, 0x2c, 0x00, 0x00, 0x00,
- 0x86, 0xc1, 0x44, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_17[102] =
- {
- 0x0c, 0x81, 0x90, 0x00, 0x00, 0x00,
- 0x80, 0x70, 0x0c, 0x00, 0x00, 0x00,
- 0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
- 0x05, 0x40, 0xa8, 0x00, 0x00, 0x00,
- 0x43, 0x08, 0x60, 0x00, 0x00, 0x00,
- 0x1a, 0x03, 0x40, 0x00, 0x00, 0x00,
- 0x60, 0x2c, 0x04, 0x00, 0x00, 0x00,
- 0x14, 0x22, 0x84, 0x00, 0x00, 0x00,
- 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
- 0x62, 0x6c, 0x4c, 0x00, 0x00, 0x00,
- 0x8c, 0x11, 0x80, 0x00, 0x00, 0x00,
- 0x01, 0x60, 0x2c, 0x00, 0x00, 0x00,
- 0x07, 0x80, 0xf0, 0x00, 0x00, 0x00,
- 0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
- 0x18, 0xa3, 0x14, 0x00, 0x00, 0x00,
- 0x91, 0x12, 0x20, 0x00, 0x00, 0x00,
- 0x78, 0x0f, 0x00, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_18[108] =
- {
- 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
- 0x62, 0x6c, 0x4c, 0x00, 0x00, 0x00,
- 0x8c, 0x11, 0x80, 0x00, 0x00, 0x00,
- 0x01, 0x60, 0x2c, 0x00, 0x00, 0x00,
- 0x07, 0x80, 0xf0, 0x00, 0x00, 0x00,
- 0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
- 0x18, 0xa3, 0x14, 0x00, 0x00, 0x00,
- 0x91, 0x12, 0x20, 0x00, 0x00, 0x00,
- 0x78, 0x0f, 0x00, 0x00, 0x00, 0x00,
- 0x0c, 0x81, 0x90, 0x00, 0x00, 0x00,
- 0x80, 0x70, 0x0c, 0x00, 0x00, 0x00,
- 0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
- 0x05, 0x40, 0xa8, 0x00, 0x00, 0x00,
- 0x43, 0x08, 0x60, 0x00, 0x00, 0x00,
- 0x1a, 0x03, 0x40, 0x00, 0x00, 0x00,
- 0x60, 0x2c, 0x04, 0x00, 0x00, 0x00,
- 0x14, 0x22, 0x84, 0x00, 0x00, 0x00,
- 0xe4, 0xd4, 0x6c, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_19[114] =
- {
- 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
- 0x62, 0x6c, 0x4c, 0x00, 0x00, 0x00,
- 0x8c, 0x11, 0x80, 0x00, 0x00, 0x00,
- 0x01, 0x60, 0x2c, 0x00, 0x00, 0x00,
- 0x07, 0x80, 0xf0, 0x00, 0x00, 0x00,
- 0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
- 0x18, 0xa3, 0x14, 0x00, 0x00, 0x00,
- 0x91, 0x12, 0x20, 0x00, 0x00, 0x00,
- 0x78, 0x0f, 0x00, 0x00, 0x00, 0x00,
- 0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
- 0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
- 0xa9, 0x15, 0x20, 0x00, 0x00, 0x00,
- 0x04, 0xc0, 0x98, 0x00, 0x00, 0x00,
- 0xd0, 0x1a, 0x00, 0x00, 0x00, 0x00,
- 0x82, 0x50, 0x48, 0x00, 0x00, 0x00,
- 0x21, 0x24, 0x24, 0x00, 0x00, 0x00,
- 0x0c, 0x21, 0x84, 0x00, 0x00, 0x00,
- 0x4a, 0x09, 0x40, 0x00, 0x00, 0x00,
- 0x12, 0xa2, 0x54, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_2[12] =
- {
- 0xec, 0xdd, 0x98, 0x00, 0x00, 0x00,
- 0x9b, 0xb3, 0x74, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_20[120] =
- {
- 0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
- 0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
- 0xa9, 0x15, 0x20, 0x00, 0x00, 0x00,
- 0x04, 0xc0, 0x98, 0x00, 0x00, 0x00,
- 0xd0, 0x1a, 0x00, 0x00, 0x00, 0x00,
- 0x82, 0x50, 0x48, 0x00, 0x00, 0x00,
- 0x21, 0x24, 0x24, 0x00, 0x00, 0x00,
- 0x0c, 0x21, 0x84, 0x00, 0x00, 0x00,
- 0x4a, 0x09, 0x40, 0x00, 0x00, 0x00,
- 0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
- 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
- 0x62, 0x6c, 0x4c, 0x00, 0x00, 0x00,
- 0x8c, 0x11, 0x80, 0x00, 0x00, 0x00,
- 0x01, 0x60, 0x2c, 0x00, 0x00, 0x00,
- 0x07, 0x80, 0xf0, 0x00, 0x00, 0x00,
- 0xa0, 0x94, 0x10, 0x00, 0x00, 0x00,
- 0x18, 0xa3, 0x14, 0x00, 0x00, 0x00,
- 0x91, 0x12, 0x20, 0x00, 0x00, 0x00,
- 0x78, 0x0f, 0x00, 0x00, 0x00, 0x00,
- 0x3b, 0x48, 0xc4, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_21[126] =
- {
- 0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
- 0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
- 0xa9, 0x15, 0x20, 0x00, 0x00, 0x00,
- 0x04, 0xc0, 0x98, 0x00, 0x00, 0x00,
- 0xd0, 0x1a, 0x00, 0x00, 0x00, 0x00,
- 0x82, 0x50, 0x48, 0x00, 0x00, 0x00,
- 0x21, 0x24, 0x24, 0x00, 0x00, 0x00,
- 0x0c, 0x21, 0x84, 0x00, 0x00, 0x00,
- 0x4a, 0x09, 0x40, 0x00, 0x00, 0x00,
- 0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
- 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
- 0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
- 0x99, 0x13, 0x20, 0x00, 0x00, 0x00,
- 0x05, 0x80, 0xb0, 0x00, 0x00, 0x00,
- 0x80, 0xb0, 0x14, 0x00, 0x00, 0x00,
- 0x84, 0x50, 0x88, 0x00, 0x00, 0x00,
- 0x40, 0x68, 0x0c, 0x00, 0x00, 0x00,
- 0x0a, 0x81, 0x50, 0x00, 0x00, 0x00,
- 0x68, 0x0d, 0x00, 0x00, 0x00, 0x00,
- 0x10, 0x22, 0x04, 0x00, 0x00, 0x00,
- 0x30, 0x46, 0x08, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_22[132] =
- {
- 0x46, 0x48, 0xc8, 0x00, 0x00, 0x00,
- 0x33, 0x26, 0x64, 0x00, 0x00, 0x00,
- 0x99, 0x13, 0x20, 0x00, 0x00, 0x00,
- 0x05, 0x80, 0xb0, 0x00, 0x00, 0x00,
- 0x80, 0xb0, 0x14, 0x00, 0x00, 0x00,
- 0x84, 0x50, 0x88, 0x00, 0x00, 0x00,
- 0x40, 0x68, 0x0c, 0x00, 0x00, 0x00,
- 0x0a, 0x81, 0x50, 0x00, 0x00, 0x00,
- 0x68, 0x0d, 0x00, 0x00, 0x00, 0x00,
- 0x10, 0x22, 0x04, 0x00, 0x00, 0x00,
- 0x30, 0x46, 0x08, 0x00, 0x00, 0x00,
- 0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
- 0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
- 0xa9, 0x15, 0x20, 0x00, 0x00, 0x00,
- 0x04, 0xc0, 0x98, 0x00, 0x00, 0x00,
- 0xd0, 0x1a, 0x00, 0x00, 0x00, 0x00,
- 0x82, 0x50, 0x48, 0x00, 0x00, 0x00,
- 0x21, 0x24, 0x24, 0x00, 0x00, 0x00,
- 0x0c, 0x21, 0x84, 0x00, 0x00, 0x00,
- 0x4a, 0x09, 0x40, 0x00, 0x00, 0x00,
- 0x12, 0xa2, 0x54, 0x00, 0x00, 0x00,
- 0x9e, 0xce, 0x88, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_3[18] =
- {
- 0xca, 0xd9, 0x58, 0x00, 0x00, 0x00,
- 0xf1, 0x5e, 0x28, 0x00, 0x00, 0x00,
- 0xb6, 0x35, 0xc4, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_4[24] =
- {
- 0xc4, 0xd8, 0x98, 0x00, 0x00, 0x00,
- 0x31, 0x66, 0x2c, 0x00, 0x00, 0x00,
- 0x4b, 0x29, 0x64, 0x00, 0x00, 0x00,
- 0x2c, 0xa5, 0x94, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_5[30] =
- {
- 0xc6, 0xd8, 0xd8, 0x00, 0x00, 0x00,
- 0x63, 0x6c, 0x6c, 0x00, 0x00, 0x00,
- 0x1d, 0xa3, 0xb4, 0x00, 0x00, 0x00,
- 0xad, 0x55, 0xa8, 0x00, 0x00, 0x00,
- 0xb2, 0xb6, 0x54, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_6[36] =
- {
- 0x64, 0x4c, 0x88, 0x00, 0x00, 0x00,
- 0x51, 0x4a, 0x28, 0x00, 0x00, 0x00,
- 0xa8, 0x35, 0x04, 0x00, 0x00, 0x00,
- 0xc4, 0xa0, 0x94, 0x00, 0x00, 0x00,
- 0x03, 0x60, 0x6c, 0x00, 0x00, 0x00,
- 0x90, 0xd2, 0x18, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_7[42] =
- {
- 0xc6, 0x48, 0xc8, 0x00, 0x00, 0x00,
- 0x13, 0x26, 0x64, 0x00, 0x00, 0x00,
- 0x8d, 0x13, 0xa0, 0x00, 0x00, 0x00,
- 0x8b, 0x41, 0x68, 0x00, 0x00, 0x00,
- 0x52, 0xaa, 0x14, 0x00, 0x00, 0x00,
- 0xa2, 0xd4, 0x18, 0x00, 0x00, 0x00,
- 0x61, 0xa8, 0x2c, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_8[48] =
- {
- 0x28, 0x85, 0x38, 0x00, 0x00, 0x00,
- 0x21, 0xf4, 0x04, 0x00, 0x00, 0x00,
- 0xe9, 0x1d, 0x00, 0x00, 0x00, 0x00,
- 0x17, 0x02, 0xe0, 0x00, 0x00, 0x00,
- 0x83, 0xa0, 0x54, 0x00, 0x00, 0x00,
- 0x46, 0x18, 0xe8, 0x00, 0x00, 0x00,
- 0x50, 0x6a, 0x0c, 0x00, 0x00, 0x00,
- 0x1c, 0x23, 0x84, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask22_9[54] =
- {
- 0x44, 0x48, 0xc8, 0x00, 0x00, 0x00,
- 0x28, 0x2d, 0x0c, 0x00, 0x00, 0x00,
- 0x25, 0x14, 0xa0, 0x00, 0x00, 0x00,
- 0x59, 0x0a, 0x20, 0x00, 0x00, 0x00,
- 0x03, 0xa0, 0x34, 0x00, 0x00, 0x00,
- 0xc0, 0xd0, 0x18, 0x00, 0x00, 0x00,
- 0xa2, 0x30, 0x44, 0x00, 0x00, 0x00,
- 0x14, 0x82, 0xd0, 0x00, 0x00, 0x00,
- 0x9a, 0x03, 0x80, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_1[6] =
- {
- 0xff, 0xff, 0xfe, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_10[60] =
- {
- 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
- 0x51, 0x48, 0xa2, 0x00, 0x00, 0x00,
- 0xa9, 0x10, 0x1a, 0x00, 0x00, 0x00,
- 0x04, 0xc4, 0x84, 0x00, 0x00, 0x00,
- 0xd0, 0x01, 0x44, 0x00, 0x00, 0x00,
- 0x82, 0x40, 0x1c, 0x00, 0x00, 0x00,
- 0x21, 0x37, 0x00, 0x00, 0x00, 0x00,
- 0x0c, 0x21, 0x22, 0x00, 0x00, 0x00,
- 0x4a, 0x0a, 0xc0, 0x00, 0x00, 0x00,
- 0x12, 0xb4, 0x50, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_11[66] =
- {
- 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
- 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
- 0x99, 0x02, 0x12, 0x00, 0x00, 0x00,
- 0x05, 0x80, 0x0e, 0x00, 0x00, 0x00,
- 0x80, 0xa1, 0x82, 0x00, 0x00, 0x00,
- 0x84, 0x48, 0x18, 0x00, 0x00, 0x00,
- 0x40, 0x6d, 0x40, 0x00, 0x00, 0x00,
- 0x0a, 0x90, 0xc0, 0x00, 0x00, 0x00,
- 0x68, 0x04, 0x90, 0x00, 0x00, 0x00,
- 0x10, 0x31, 0x20, 0x00, 0x00, 0x00,
- 0x30, 0x58, 0x04, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_12[72] =
- {
- 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
- 0x51, 0x58, 0xa2, 0x00, 0x00, 0x00,
- 0x0c, 0xa4, 0x30, 0x00, 0x00, 0x00,
- 0xa1, 0x22, 0x46, 0x00, 0x00, 0x00,
- 0x12, 0xa1, 0x1c, 0x00, 0x00, 0x00,
- 0x8a, 0x45, 0xc0, 0x00, 0x00, 0x00,
- 0x86, 0x8a, 0x6c, 0x00, 0x00, 0x00,
- 0x23, 0x2c, 0x84, 0x00, 0x00, 0x00,
- 0x16, 0x21, 0x98, 0x00, 0x00, 0x00,
- 0x4c, 0x30, 0x54, 0x00, 0x00, 0x00,
- 0x41, 0xc1, 0x26, 0x00, 0x00, 0x00,
- 0x19, 0x56, 0xe4, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_13[78] =
- {
- 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
- 0x51, 0x58, 0xa2, 0x00, 0x00, 0x00,
- 0x0c, 0xa4, 0x30, 0x00, 0x00, 0x00,
- 0xa1, 0x22, 0x46, 0x00, 0x00, 0x00,
- 0x12, 0xa1, 0x1c, 0x00, 0x00, 0x00,
- 0x8a, 0x45, 0xc0, 0x00, 0x00, 0x00,
- 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
- 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
- 0x91, 0x92, 0x12, 0x00, 0x00, 0x00,
- 0xa4, 0x20, 0x4a, 0x00, 0x00, 0x00,
- 0x50, 0xa0, 0xd4, 0x00, 0x00, 0x00,
- 0x84, 0xc5, 0x80, 0x00, 0x00, 0x00,
- 0x09, 0x71, 0x0c, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_14[84] =
- {
- 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
- 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
- 0x91, 0x92, 0x12, 0x00, 0x00, 0x00,
- 0xa4, 0x20, 0x4a, 0x00, 0x00, 0x00,
- 0x50, 0xa0, 0xd4, 0x00, 0x00, 0x00,
- 0x84, 0xc5, 0x80, 0x00, 0x00, 0x00,
- 0x09, 0x71, 0x0c, 0x00, 0x00, 0x00,
- 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
- 0x51, 0x58, 0xa2, 0x00, 0x00, 0x00,
- 0x0c, 0xa4, 0x30, 0x00, 0x00, 0x00,
- 0xa1, 0x22, 0x46, 0x00, 0x00, 0x00,
- 0x12, 0xa1, 0x1c, 0x00, 0x00, 0x00,
- 0x8a, 0x45, 0xc0, 0x00, 0x00, 0x00,
- 0x9c, 0x3f, 0xb2, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_15[90] =
- {
- 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
- 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
- 0x91, 0x92, 0x12, 0x00, 0x00, 0x00,
- 0xa4, 0x20, 0x4a, 0x00, 0x00, 0x00,
- 0x50, 0xa0, 0xd4, 0x00, 0x00, 0x00,
- 0x84, 0xc5, 0x80, 0x00, 0x00, 0x00,
- 0x09, 0x71, 0x0c, 0x00, 0x00, 0x00,
- 0x0c, 0x84, 0x0c, 0x00, 0x00, 0x00,
- 0x80, 0x70, 0x06, 0x00, 0x00, 0x00,
- 0xa0, 0x88, 0x48, 0x00, 0x00, 0x00,
- 0x05, 0x40, 0x32, 0x00, 0x00, 0x00,
- 0x43, 0x02, 0x82, 0x00, 0x00, 0x00,
- 0x1a, 0x01, 0x50, 0x00, 0x00, 0x00,
- 0x60, 0x27, 0x00, 0x00, 0x00, 0x00,
- 0x14, 0x38, 0xa0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_16[96] =
- {
- 0x0c, 0x84, 0x0c, 0x00, 0x00, 0x00,
- 0x80, 0x70, 0x06, 0x00, 0x00, 0x00,
- 0xa0, 0x88, 0x48, 0x00, 0x00, 0x00,
- 0x05, 0x40, 0x32, 0x00, 0x00, 0x00,
- 0x43, 0x02, 0x82, 0x00, 0x00, 0x00,
- 0x1a, 0x01, 0x50, 0x00, 0x00, 0x00,
- 0x60, 0x27, 0x00, 0x00, 0x00, 0x00,
- 0x14, 0x38, 0xa0, 0x00, 0x00, 0x00,
- 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
- 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
- 0x91, 0x92, 0x12, 0x00, 0x00, 0x00,
- 0xa4, 0x20, 0x4a, 0x00, 0x00, 0x00,
- 0x50, 0xa0, 0xd4, 0x00, 0x00, 0x00,
- 0x84, 0xc5, 0x80, 0x00, 0x00, 0x00,
- 0x09, 0x71, 0x0c, 0x00, 0x00, 0x00,
- 0xfa, 0xd9, 0xf4, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_17[102] =
- {
- 0x0c, 0x84, 0x0c, 0x00, 0x00, 0x00,
- 0x80, 0x70, 0x06, 0x00, 0x00, 0x00,
- 0xa0, 0x88, 0x48, 0x00, 0x00, 0x00,
- 0x05, 0x40, 0x32, 0x00, 0x00, 0x00,
- 0x43, 0x02, 0x82, 0x00, 0x00, 0x00,
- 0x1a, 0x01, 0x50, 0x00, 0x00, 0x00,
- 0x60, 0x27, 0x00, 0x00, 0x00, 0x00,
- 0x14, 0x38, 0xa0, 0x00, 0x00, 0x00,
- 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
- 0x62, 0x7c, 0x84, 0x00, 0x00, 0x00,
- 0x8c, 0x04, 0x88, 0x00, 0x00, 0x00,
- 0x01, 0x74, 0x22, 0x00, 0x00, 0x00,
- 0x07, 0x83, 0x06, 0x00, 0x00, 0x00,
- 0xa0, 0x80, 0x72, 0x00, 0x00, 0x00,
- 0x18, 0xb1, 0x42, 0x00, 0x00, 0x00,
- 0x91, 0x00, 0x92, 0x00, 0x00, 0x00,
- 0x78, 0x00, 0x1c, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_18[108] =
- {
- 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
- 0x62, 0x7c, 0x84, 0x00, 0x00, 0x00,
- 0x8c, 0x04, 0x88, 0x00, 0x00, 0x00,
- 0x01, 0x74, 0x22, 0x00, 0x00, 0x00,
- 0x07, 0x83, 0x06, 0x00, 0x00, 0x00,
- 0xa0, 0x80, 0x72, 0x00, 0x00, 0x00,
- 0x18, 0xb1, 0x42, 0x00, 0x00, 0x00,
- 0x91, 0x00, 0x92, 0x00, 0x00, 0x00,
- 0x78, 0x00, 0x1c, 0x00, 0x00, 0x00,
- 0x0c, 0x84, 0x0c, 0x00, 0x00, 0x00,
- 0x80, 0x70, 0x06, 0x00, 0x00, 0x00,
- 0xa0, 0x88, 0x48, 0x00, 0x00, 0x00,
- 0x05, 0x40, 0x32, 0x00, 0x00, 0x00,
- 0x43, 0x02, 0x82, 0x00, 0x00, 0x00,
- 0x1a, 0x01, 0x50, 0x00, 0x00, 0x00,
- 0x60, 0x27, 0x00, 0x00, 0x00, 0x00,
- 0x14, 0x38, 0xa0, 0x00, 0x00, 0x00,
- 0x82, 0x32, 0x56, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_19[114] =
- {
- 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
- 0x62, 0x7c, 0x84, 0x00, 0x00, 0x00,
- 0x8c, 0x04, 0x88, 0x00, 0x00, 0x00,
- 0x01, 0x74, 0x22, 0x00, 0x00, 0x00,
- 0x07, 0x83, 0x06, 0x00, 0x00, 0x00,
- 0xa0, 0x80, 0x72, 0x00, 0x00, 0x00,
- 0x18, 0xb1, 0x42, 0x00, 0x00, 0x00,
- 0x91, 0x00, 0x92, 0x00, 0x00, 0x00,
- 0x78, 0x00, 0x1c, 0x00, 0x00, 0x00,
- 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
- 0x51, 0x48, 0xa2, 0x00, 0x00, 0x00,
- 0xa9, 0x10, 0x1a, 0x00, 0x00, 0x00,
- 0x04, 0xc4, 0x84, 0x00, 0x00, 0x00,
- 0xd0, 0x01, 0x44, 0x00, 0x00, 0x00,
- 0x82, 0x40, 0x1c, 0x00, 0x00, 0x00,
- 0x21, 0x37, 0x00, 0x00, 0x00, 0x00,
- 0x0c, 0x21, 0x22, 0x00, 0x00, 0x00,
- 0x4a, 0x0a, 0xc0, 0x00, 0x00, 0x00,
- 0x12, 0xb4, 0x50, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_2[12] =
- {
- 0xec, 0xdd, 0x98, 0x00, 0x00, 0x00,
- 0x9b, 0xb2, 0x76, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_20[120] =
- {
- 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
- 0x51, 0x48, 0xa2, 0x00, 0x00, 0x00,
- 0xa9, 0x10, 0x1a, 0x00, 0x00, 0x00,
- 0x04, 0xc4, 0x84, 0x00, 0x00, 0x00,
- 0xd0, 0x01, 0x44, 0x00, 0x00, 0x00,
- 0x82, 0x40, 0x1c, 0x00, 0x00, 0x00,
- 0x21, 0x37, 0x00, 0x00, 0x00, 0x00,
- 0x0c, 0x21, 0x22, 0x00, 0x00, 0x00,
- 0x4a, 0x0a, 0xc0, 0x00, 0x00, 0x00,
- 0x12, 0xb4, 0x50, 0x00, 0x00, 0x00,
- 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
- 0x62, 0x7c, 0x84, 0x00, 0x00, 0x00,
- 0x8c, 0x04, 0x88, 0x00, 0x00, 0x00,
- 0x01, 0x74, 0x22, 0x00, 0x00, 0x00,
- 0x07, 0x83, 0x06, 0x00, 0x00, 0x00,
- 0xa0, 0x80, 0x72, 0x00, 0x00, 0x00,
- 0x18, 0xb1, 0x42, 0x00, 0x00, 0x00,
- 0x91, 0x00, 0x92, 0x00, 0x00, 0x00,
- 0x78, 0x00, 0x1c, 0x00, 0x00, 0x00,
- 0xdb, 0x4a, 0x7a, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_21[126] =
- {
- 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
- 0x51, 0x48, 0xa2, 0x00, 0x00, 0x00,
- 0xa9, 0x10, 0x1a, 0x00, 0x00, 0x00,
- 0x04, 0xc4, 0x84, 0x00, 0x00, 0x00,
- 0xd0, 0x01, 0x44, 0x00, 0x00, 0x00,
- 0x82, 0x40, 0x1c, 0x00, 0x00, 0x00,
- 0x21, 0x37, 0x00, 0x00, 0x00, 0x00,
- 0x0c, 0x21, 0x22, 0x00, 0x00, 0x00,
- 0x4a, 0x0a, 0xc0, 0x00, 0x00, 0x00,
- 0x12, 0xb4, 0x50, 0x00, 0x00, 0x00,
- 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
- 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
- 0x99, 0x02, 0x12, 0x00, 0x00, 0x00,
- 0x05, 0x80, 0x0e, 0x00, 0x00, 0x00,
- 0x80, 0xa1, 0x82, 0x00, 0x00, 0x00,
- 0x84, 0x48, 0x18, 0x00, 0x00, 0x00,
- 0x40, 0x6d, 0x40, 0x00, 0x00, 0x00,
- 0x0a, 0x90, 0xc0, 0x00, 0x00, 0x00,
- 0x68, 0x04, 0x90, 0x00, 0x00, 0x00,
- 0x10, 0x31, 0x20, 0x00, 0x00, 0x00,
- 0x30, 0x58, 0x04, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_22[132] =
- {
- 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
- 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
- 0x99, 0x02, 0x12, 0x00, 0x00, 0x00,
- 0x05, 0x80, 0x0e, 0x00, 0x00, 0x00,
- 0x80, 0xa1, 0x82, 0x00, 0x00, 0x00,
- 0x84, 0x48, 0x18, 0x00, 0x00, 0x00,
- 0x40, 0x6d, 0x40, 0x00, 0x00, 0x00,
- 0x0a, 0x90, 0xc0, 0x00, 0x00, 0x00,
- 0x68, 0x04, 0x90, 0x00, 0x00, 0x00,
- 0x10, 0x31, 0x20, 0x00, 0x00, 0x00,
- 0x30, 0x58, 0x04, 0x00, 0x00, 0x00,
- 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
- 0x51, 0x48, 0xa2, 0x00, 0x00, 0x00,
- 0xa9, 0x10, 0x1a, 0x00, 0x00, 0x00,
- 0x04, 0xc4, 0x84, 0x00, 0x00, 0x00,
- 0xd0, 0x01, 0x44, 0x00, 0x00, 0x00,
- 0x82, 0x40, 0x1c, 0x00, 0x00, 0x00,
- 0x21, 0x37, 0x00, 0x00, 0x00, 0x00,
- 0x0c, 0x21, 0x22, 0x00, 0x00, 0x00,
- 0x4a, 0x0a, 0xc0, 0x00, 0x00, 0x00,
- 0x12, 0xb4, 0x50, 0x00, 0x00, 0x00,
- 0xea, 0x8d, 0x1a, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_23[138] =
- {
- 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
- 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
- 0x99, 0x02, 0x12, 0x00, 0x00, 0x00,
- 0x05, 0x80, 0x0e, 0x00, 0x00, 0x00,
- 0x80, 0xa1, 0x82, 0x00, 0x00, 0x00,
- 0x84, 0x48, 0x18, 0x00, 0x00, 0x00,
- 0x40, 0x6d, 0x40, 0x00, 0x00, 0x00,
- 0x0a, 0x90, 0xc0, 0x00, 0x00, 0x00,
- 0x68, 0x04, 0x90, 0x00, 0x00, 0x00,
- 0x10, 0x31, 0x20, 0x00, 0x00, 0x00,
- 0x30, 0x58, 0x04, 0x00, 0x00, 0x00,
- 0x46, 0x42, 0x0c, 0x00, 0x00, 0x00,
- 0x33, 0x20, 0x46, 0x00, 0x00, 0x00,
- 0x99, 0x08, 0x0a, 0x00, 0x00, 0x00,
- 0x05, 0x84, 0x30, 0x00, 0x00, 0x00,
- 0x80, 0xb0, 0x22, 0x00, 0x00, 0x00,
- 0x84, 0x42, 0x90, 0x00, 0x00, 0x00,
- 0x40, 0x73, 0x00, 0x00, 0x00, 0x00,
- 0x0a, 0x81, 0x12, 0x00, 0x00, 0x00,
- 0x68, 0x0c, 0x40, 0x00, 0x00, 0x00,
- 0x10, 0x24, 0x84, 0x00, 0x00, 0x00,
- 0x30, 0x51, 0x40, 0x00, 0x00, 0x00,
- 0x5f, 0x50, 0x88, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_3[18] =
- {
- 0xca, 0xd3, 0x64, 0x00, 0x00, 0x00,
- 0xf1, 0x49, 0x3a, 0x00, 0x00, 0x00,
- 0x76, 0x27, 0xd0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_4[24] =
- {
- 0xc4, 0xd1, 0x64, 0x00, 0x00, 0x00,
- 0x31, 0x62, 0x96, 0x00, 0x00, 0x00,
- 0x4b, 0x24, 0x5a, 0x00, 0x00, 0x00,
- 0x2c, 0xa8, 0xaa, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_5[30] =
- {
- 0xc6, 0xca, 0x6c, 0x00, 0x00, 0x00,
- 0x63, 0x6c, 0x96, 0x00, 0x00, 0x00,
- 0x1d, 0xa1, 0xdc, 0x00, 0x00, 0x00,
- 0xad, 0x55, 0x38, 0x00, 0x00, 0x00,
- 0xb2, 0xb7, 0x06, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_6[36] =
- {
- 0x64, 0x4a, 0x28, 0x00, 0x00, 0x00,
- 0x51, 0x58, 0xa2, 0x00, 0x00, 0x00,
- 0x0c, 0xa4, 0x30, 0x00, 0x00, 0x00,
- 0xa1, 0x22, 0x46, 0x00, 0x00, 0x00,
- 0x12, 0xa1, 0x1c, 0x00, 0x00, 0x00,
- 0x8a, 0x45, 0xc0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_7[42] =
- {
- 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
- 0x33, 0x24, 0x26, 0x00, 0x00, 0x00,
- 0x91, 0x92, 0x12, 0x00, 0x00, 0x00,
- 0xa4, 0x20, 0x4a, 0x00, 0x00, 0x00,
- 0x50, 0xa0, 0xd4, 0x00, 0x00, 0x00,
- 0x84, 0xc5, 0x80, 0x00, 0x00, 0x00,
- 0x09, 0x71, 0x0c, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_8[48] =
- {
- 0x0c, 0x84, 0x0c, 0x00, 0x00, 0x00,
- 0x80, 0x70, 0x06, 0x00, 0x00, 0x00,
- 0xa0, 0x88, 0x48, 0x00, 0x00, 0x00,
- 0x05, 0x40, 0x32, 0x00, 0x00, 0x00,
- 0x43, 0x02, 0x82, 0x00, 0x00, 0x00,
- 0x1a, 0x01, 0x50, 0x00, 0x00, 0x00,
- 0x60, 0x27, 0x00, 0x00, 0x00, 0x00,
- 0x14, 0x38, 0xa0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask23_9[54] =
- {
- 0x46, 0x4a, 0x6c, 0x00, 0x00, 0x00,
- 0x62, 0x7c, 0x84, 0x00, 0x00, 0x00,
- 0x8c, 0x04, 0x88, 0x00, 0x00, 0x00,
- 0x01, 0x74, 0x22, 0x00, 0x00, 0x00,
- 0x07, 0x83, 0x06, 0x00, 0x00, 0x00,
- 0xa0, 0x80, 0x72, 0x00, 0x00, 0x00,
- 0x18, 0xb1, 0x42, 0x00, 0x00, 0x00,
- 0x91, 0x00, 0x92, 0x00, 0x00, 0x00,
- 0x78, 0x00, 0x1c, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_1[6] =
- {
- 0xff, 0xff, 0xff, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_10[60] =
- {
- 0x11, 0x45, 0x14, 0x00, 0x00, 0x00,
- 0x45, 0x34, 0x53, 0x00, 0x00, 0x00,
- 0x00, 0x48, 0x05, 0x00, 0x00, 0x00,
- 0x10, 0x83, 0x09, 0x00, 0x00, 0x00,
- 0x4a, 0x14, 0xa1, 0x00, 0x00, 0x00,
- 0x40, 0xa4, 0x0a, 0x00, 0x00, 0x00,
- 0xa0, 0x6a, 0x02, 0x00, 0x00, 0x00,
- 0x88, 0x80, 0x8c, 0x00, 0x00, 0x00,
- 0x86, 0x08, 0x60, 0x00, 0x00, 0x00,
- 0x54, 0x0d, 0x40, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_11[66] =
- {
- 0x53, 0x65, 0x34, 0x00, 0x00, 0x00,
- 0xa0, 0x32, 0x11, 0x00, 0x00, 0x00,
- 0x15, 0x11, 0x41, 0x00, 0x00, 0x00,
- 0x03, 0x50, 0x15, 0x00, 0x00, 0x00,
- 0x8c, 0x88, 0xc8, 0x00, 0x00, 0x00,
- 0x28, 0x82, 0x88, 0x00, 0x00, 0x00,
- 0x08, 0x48, 0x84, 0x00, 0x00, 0x00,
- 0x99, 0x01, 0x90, 0x00, 0x00, 0x00,
- 0x22, 0x92, 0x29, 0x00, 0x00, 0x00,
- 0x46, 0x04, 0x60, 0x00, 0x00, 0x00,
- 0x8c, 0x2c, 0x02, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_12[72] =
- {
- 0x10, 0x61, 0x06, 0x00, 0x00, 0x00,
- 0x02, 0x30, 0x23, 0x00, 0x00, 0x00,
- 0x40, 0x54, 0x05, 0x00, 0x00, 0x00,
- 0x21, 0x82, 0x18, 0x00, 0x00, 0x00,
- 0x81, 0x18, 0x11, 0x00, 0x00, 0x00,
- 0x14, 0x81, 0x48, 0x00, 0x00, 0x00,
- 0x98, 0x09, 0x80, 0x00, 0x00, 0x00,
- 0x08, 0x90, 0x89, 0x00, 0x00, 0x00,
- 0x62, 0x06, 0x20, 0x00, 0x00, 0x00,
- 0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
- 0x8a, 0x08, 0xa0, 0x00, 0x00, 0x00,
- 0x84, 0x48, 0x44, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_13[78] =
- {
- 0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
- 0xc5, 0x1c, 0x51, 0x00, 0x00, 0x00,
- 0x21, 0x82, 0x18, 0x00, 0x00, 0x00,
- 0x12, 0x31, 0x23, 0x00, 0x00, 0x00,
- 0x08, 0xe0, 0x8e, 0x00, 0x00, 0x00,
- 0x2e, 0x02, 0xe0, 0x00, 0x00, 0x00,
- 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
- 0x90, 0x99, 0x09, 0x00, 0x00, 0x00,
- 0x02, 0x50, 0x25, 0x00, 0x00, 0x00,
- 0x06, 0xa0, 0x6a, 0x00, 0x00, 0x00,
- 0x2c, 0x02, 0xc0, 0x00, 0x00, 0x00,
- 0x88, 0x68, 0x86, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_14[84] =
- {
- 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
- 0x90, 0x99, 0x09, 0x00, 0x00, 0x00,
- 0x02, 0x50, 0x25, 0x00, 0x00, 0x00,
- 0x06, 0xa0, 0x6a, 0x00, 0x00, 0x00,
- 0x2c, 0x02, 0xc0, 0x00, 0x00, 0x00,
- 0x88, 0x68, 0x86, 0x00, 0x00, 0x00,
- 0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
- 0xc5, 0x1c, 0x51, 0x00, 0x00, 0x00,
- 0x21, 0x82, 0x18, 0x00, 0x00, 0x00,
- 0x12, 0x31, 0x23, 0x00, 0x00, 0x00,
- 0x08, 0xe0, 0x8e, 0x00, 0x00, 0x00,
- 0x2e, 0x02, 0xe0, 0x00, 0x00, 0x00,
- 0xf2, 0xd6, 0x8e, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_15[90] =
- {
- 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
- 0x90, 0x99, 0x09, 0x00, 0x00, 0x00,
- 0x02, 0x50, 0x25, 0x00, 0x00, 0x00,
- 0x06, 0xa0, 0x6a, 0x00, 0x00, 0x00,
- 0x2c, 0x02, 0xc0, 0x00, 0x00, 0x00,
- 0x88, 0x68, 0x86, 0x00, 0x00, 0x00,
- 0x20, 0x62, 0x06, 0x00, 0x00, 0x00,
- 0x80, 0x38, 0x03, 0x00, 0x00, 0x00,
- 0x42, 0x44, 0x24, 0x00, 0x00, 0x00,
- 0x01, 0x90, 0x19, 0x00, 0x00, 0x00,
- 0x14, 0x11, 0x41, 0x00, 0x00, 0x00,
- 0x0a, 0x80, 0xa8, 0x00, 0x00, 0x00,
- 0x38, 0x03, 0x80, 0x00, 0x00, 0x00,
- 0xc5, 0x0c, 0x50, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_16[96] =
- {
- 0x20, 0x62, 0x06, 0x00, 0x00, 0x00,
- 0x80, 0x38, 0x03, 0x00, 0x00, 0x00,
- 0x42, 0x44, 0x24, 0x00, 0x00, 0x00,
- 0x01, 0x90, 0x19, 0x00, 0x00, 0x00,
- 0x14, 0x11, 0x41, 0x00, 0x00, 0x00,
- 0x0a, 0x80, 0xa8, 0x00, 0x00, 0x00,
- 0x38, 0x03, 0x80, 0x00, 0x00, 0x00,
- 0xc5, 0x0c, 0x50, 0x00, 0x00, 0x00,
- 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
- 0x90, 0x99, 0x09, 0x00, 0x00, 0x00,
- 0x02, 0x50, 0x25, 0x00, 0x00, 0x00,
- 0x06, 0xa0, 0x6a, 0x00, 0x00, 0x00,
- 0x2c, 0x02, 0xc0, 0x00, 0x00, 0x00,
- 0x88, 0x68, 0x86, 0x00, 0x00, 0x00,
- 0xff, 0x6e, 0x0a, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_17[102] =
- {
- 0x20, 0x62, 0x06, 0x00, 0x00, 0x00,
- 0x80, 0x38, 0x03, 0x00, 0x00, 0x00,
- 0x42, 0x44, 0x24, 0x00, 0x00, 0x00,
- 0x01, 0x90, 0x19, 0x00, 0x00, 0x00,
- 0x14, 0x11, 0x41, 0x00, 0x00, 0x00,
- 0x0a, 0x80, 0xa8, 0x00, 0x00, 0x00,
- 0x38, 0x03, 0x80, 0x00, 0x00, 0x00,
- 0xc5, 0x0c, 0x50, 0x00, 0x00, 0x00,
- 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
- 0xe4, 0x2e, 0x42, 0x00, 0x00, 0x00,
- 0x24, 0x42, 0x44, 0x00, 0x00, 0x00,
- 0xa1, 0x1a, 0x11, 0x00, 0x00, 0x00,
- 0x18, 0x31, 0x83, 0x00, 0x00, 0x00,
- 0x03, 0x90, 0x39, 0x00, 0x00, 0x00,
- 0x8a, 0x18, 0xa1, 0x00, 0x00, 0x00,
- 0x04, 0x90, 0x49, 0x00, 0x00, 0x00,
- 0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_18[108] =
- {
- 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
- 0xe4, 0x2e, 0x42, 0x00, 0x00, 0x00,
- 0x24, 0x42, 0x44, 0x00, 0x00, 0x00,
- 0xa1, 0x1a, 0x11, 0x00, 0x00, 0x00,
- 0x18, 0x31, 0x83, 0x00, 0x00, 0x00,
- 0x03, 0x90, 0x39, 0x00, 0x00, 0x00,
- 0x8a, 0x18, 0xa1, 0x00, 0x00, 0x00,
- 0x04, 0x90, 0x49, 0x00, 0x00, 0x00,
- 0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
- 0x20, 0x62, 0x06, 0x00, 0x00, 0x00,
- 0x80, 0x38, 0x03, 0x00, 0x00, 0x00,
- 0x42, 0x44, 0x24, 0x00, 0x00, 0x00,
- 0x01, 0x90, 0x19, 0x00, 0x00, 0x00,
- 0x14, 0x11, 0x41, 0x00, 0x00, 0x00,
- 0x0a, 0x80, 0xa8, 0x00, 0x00, 0x00,
- 0x38, 0x03, 0x80, 0x00, 0x00, 0x00,
- 0xc5, 0x0c, 0x50, 0x00, 0x00, 0x00,
- 0x34, 0x50, 0xae, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_19[114] =
- {
- 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
- 0xe4, 0x2e, 0x42, 0x00, 0x00, 0x00,
- 0x24, 0x42, 0x44, 0x00, 0x00, 0x00,
- 0xa1, 0x1a, 0x11, 0x00, 0x00, 0x00,
- 0x18, 0x31, 0x83, 0x00, 0x00, 0x00,
- 0x03, 0x90, 0x39, 0x00, 0x00, 0x00,
- 0x8a, 0x18, 0xa1, 0x00, 0x00, 0x00,
- 0x04, 0x90, 0x49, 0x00, 0x00, 0x00,
- 0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
- 0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
- 0x45, 0x14, 0x51, 0x00, 0x00, 0x00,
- 0x80, 0xd8, 0x0d, 0x00, 0x00, 0x00,
- 0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
- 0x0a, 0x20, 0xa2, 0x00, 0x00, 0x00,
- 0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
- 0xb8, 0x0b, 0x80, 0x00, 0x00, 0x00,
- 0x09, 0x10, 0x91, 0x00, 0x00, 0x00,
- 0x56, 0x05, 0x60, 0x00, 0x00, 0x00,
- 0xa2, 0x8a, 0x28, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_2[12] =
- {
- 0xec, 0xce, 0xcc, 0x00, 0x00, 0x00,
- 0x93, 0xb9, 0x3b, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_20[120] =
- {
- 0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
- 0x45, 0x14, 0x51, 0x00, 0x00, 0x00,
- 0x80, 0xd8, 0x0d, 0x00, 0x00, 0x00,
- 0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
- 0x0a, 0x20, 0xa2, 0x00, 0x00, 0x00,
- 0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
- 0xb8, 0x0b, 0x80, 0x00, 0x00, 0x00,
- 0x09, 0x10, 0x91, 0x00, 0x00, 0x00,
- 0x56, 0x05, 0x60, 0x00, 0x00, 0x00,
- 0xa2, 0x8a, 0x28, 0x00, 0x00, 0x00,
- 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
- 0xe4, 0x2e, 0x42, 0x00, 0x00, 0x00,
- 0x24, 0x42, 0x44, 0x00, 0x00, 0x00,
- 0xa1, 0x1a, 0x11, 0x00, 0x00, 0x00,
- 0x18, 0x31, 0x83, 0x00, 0x00, 0x00,
- 0x03, 0x90, 0x39, 0x00, 0x00, 0x00,
- 0x8a, 0x18, 0xa1, 0x00, 0x00, 0x00,
- 0x04, 0x90, 0x49, 0x00, 0x00, 0x00,
- 0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
- 0x98, 0xa2, 0x95, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_21[126] =
- {
- 0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
- 0x45, 0x14, 0x51, 0x00, 0x00, 0x00,
- 0x80, 0xd8, 0x0d, 0x00, 0x00, 0x00,
- 0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
- 0x0a, 0x20, 0xa2, 0x00, 0x00, 0x00,
- 0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
- 0xb8, 0x0b, 0x80, 0x00, 0x00, 0x00,
- 0x09, 0x10, 0x91, 0x00, 0x00, 0x00,
- 0x56, 0x05, 0x60, 0x00, 0x00, 0x00,
- 0xa2, 0x8a, 0x28, 0x00, 0x00, 0x00,
- 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
- 0x10, 0x91, 0x09, 0x00, 0x00, 0x00,
- 0x00, 0x70, 0x07, 0x00, 0x00, 0x00,
- 0x0c, 0x10, 0xc1, 0x00, 0x00, 0x00,
- 0x40, 0xc4, 0x0c, 0x00, 0x00, 0x00,
- 0x6a, 0x06, 0xa0, 0x00, 0x00, 0x00,
- 0x86, 0x08, 0x60, 0x00, 0x00, 0x00,
- 0x24, 0x82, 0x48, 0x00, 0x00, 0x00,
- 0x89, 0x08, 0x90, 0x00, 0x00, 0x00,
- 0xc0, 0x2c, 0x02, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_22[132] =
- {
- 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
- 0x10, 0x91, 0x09, 0x00, 0x00, 0x00,
- 0x00, 0x70, 0x07, 0x00, 0x00, 0x00,
- 0x0c, 0x10, 0xc1, 0x00, 0x00, 0x00,
- 0x40, 0xc4, 0x0c, 0x00, 0x00, 0x00,
- 0x6a, 0x06, 0xa0, 0x00, 0x00, 0x00,
- 0x86, 0x08, 0x60, 0x00, 0x00, 0x00,
- 0x24, 0x82, 0x48, 0x00, 0x00, 0x00,
- 0x89, 0x08, 0x90, 0x00, 0x00, 0x00,
- 0xc0, 0x2c, 0x02, 0x00, 0x00, 0x00,
- 0x51, 0x45, 0x14, 0x00, 0x00, 0x00,
- 0x45, 0x14, 0x51, 0x00, 0x00, 0x00,
- 0x80, 0xd8, 0x0d, 0x00, 0x00, 0x00,
- 0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
- 0x0a, 0x20, 0xa2, 0x00, 0x00, 0x00,
- 0x00, 0xe0, 0x0e, 0x00, 0x00, 0x00,
- 0xb8, 0x0b, 0x80, 0x00, 0x00, 0x00,
- 0x09, 0x10, 0x91, 0x00, 0x00, 0x00,
- 0x56, 0x05, 0x60, 0x00, 0x00, 0x00,
- 0xa2, 0x8a, 0x28, 0x00, 0x00, 0x00,
- 0x1a, 0xaa, 0xee, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_23[138] =
- {
- 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
- 0x10, 0x91, 0x09, 0x00, 0x00, 0x00,
- 0x00, 0x70, 0x07, 0x00, 0x00, 0x00,
- 0x0c, 0x10, 0xc1, 0x00, 0x00, 0x00,
- 0x40, 0xc4, 0x0c, 0x00, 0x00, 0x00,
- 0x6a, 0x06, 0xa0, 0x00, 0x00, 0x00,
- 0x86, 0x08, 0x60, 0x00, 0x00, 0x00,
- 0x24, 0x82, 0x48, 0x00, 0x00, 0x00,
- 0x89, 0x08, 0x90, 0x00, 0x00, 0x00,
- 0xc0, 0x2c, 0x02, 0x00, 0x00, 0x00,
- 0x10, 0x61, 0x06, 0x00, 0x00, 0x00,
- 0x02, 0x30, 0x23, 0x00, 0x00, 0x00,
- 0x40, 0x54, 0x05, 0x00, 0x00, 0x00,
- 0x21, 0x82, 0x18, 0x00, 0x00, 0x00,
- 0x81, 0x18, 0x11, 0x00, 0x00, 0x00,
- 0x14, 0x81, 0x48, 0x00, 0x00, 0x00,
- 0x98, 0x09, 0x80, 0x00, 0x00, 0x00,
- 0x08, 0x90, 0x89, 0x00, 0x00, 0x00,
- 0x62, 0x06, 0x20, 0x00, 0x00, 0x00,
- 0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
- 0x8a, 0x08, 0xa0, 0x00, 0x00, 0x00,
- 0x84, 0x48, 0x44, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_24[144] =
- {
- 0x10, 0x61, 0x06, 0x00, 0x00, 0x00,
- 0x02, 0x30, 0x23, 0x00, 0x00, 0x00,
- 0x40, 0x54, 0x05, 0x00, 0x00, 0x00,
- 0x21, 0x82, 0x18, 0x00, 0x00, 0x00,
- 0x81, 0x18, 0x11, 0x00, 0x00, 0x00,
- 0x14, 0x81, 0x48, 0x00, 0x00, 0x00,
- 0x98, 0x09, 0x80, 0x00, 0x00, 0x00,
- 0x08, 0x90, 0x89, 0x00, 0x00, 0x00,
- 0x62, 0x06, 0x20, 0x00, 0x00, 0x00,
- 0x24, 0x22, 0x42, 0x00, 0x00, 0x00,
- 0x8a, 0x08, 0xa0, 0x00, 0x00, 0x00,
- 0x84, 0x48, 0x44, 0x00, 0x00, 0x00,
- 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x13, 0x00, 0x00, 0x00,
- 0x10, 0x91, 0x09, 0x00, 0x00, 0x00,
- 0x00, 0x70, 0x07, 0x00, 0x00, 0x00,
- 0x0c, 0x10, 0xc1, 0x00, 0x00, 0x00,
- 0x40, 0xc4, 0x0c, 0x00, 0x00, 0x00,
- 0x6a, 0x06, 0xa0, 0x00, 0x00, 0x00,
- 0x86, 0x08, 0x60, 0x00, 0x00, 0x00,
- 0x24, 0x82, 0x48, 0x00, 0x00, 0x00,
- 0x89, 0x08, 0x90, 0x00, 0x00, 0x00,
- 0xc0, 0x2c, 0x02, 0x00, 0x00, 0x00,
- 0x88, 0x32, 0x59, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_3[18] =
- {
- 0x9b, 0x29, 0xb2, 0x00, 0x00, 0x00,
- 0x49, 0xd4, 0x9d, 0x00, 0x00, 0x00,
- 0x3e, 0x83, 0xe8, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_4[24] =
- {
- 0x8b, 0x28, 0xb2, 0x00, 0x00, 0x00,
- 0x14, 0xb1, 0x4b, 0x00, 0x00, 0x00,
- 0x22, 0xd2, 0x2d, 0x00, 0x00, 0x00,
- 0x45, 0x54, 0x55, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_5[30] =
- {
- 0x53, 0x65, 0x36, 0x00, 0x00, 0x00,
- 0x64, 0xb6, 0x4b, 0x00, 0x00, 0x00,
- 0x0e, 0xe0, 0xee, 0x00, 0x00, 0x00,
- 0xa9, 0xca, 0x9c, 0x00, 0x00, 0x00,
- 0xb8, 0x3b, 0x83, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_6[36] =
- {
- 0xd1, 0x4d, 0x14, 0x00, 0x00, 0x00,
- 0x45, 0x34, 0x53, 0x00, 0x00, 0x00,
- 0x22, 0xd2, 0x2d, 0x00, 0x00, 0x00,
- 0x16, 0xc1, 0x6c, 0x00, 0x00, 0x00,
- 0x0b, 0xa0, 0xba, 0x00, 0x00, 0x00,
- 0xe8, 0x8e, 0x88, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_7[42] =
- {
- 0xd3, 0x65, 0x36, 0x00, 0x00, 0x00,
- 0x25, 0x32, 0x53, 0x00, 0x00, 0x00,
- 0x30, 0xd3, 0x05, 0x00, 0x00, 0x00,
- 0x06, 0x48, 0x6c, 0x00, 0x00, 0x00,
- 0xc0, 0xb8, 0x1b, 0x00, 0x00, 0x00,
- 0x2a, 0xa2, 0xaa, 0x00, 0x00, 0x00,
- 0xa8, 0x4e, 0x84, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_8[48] =
- {
- 0x81, 0x60, 0x16, 0x00, 0x00, 0x00,
- 0x40, 0x3c, 0x03, 0x00, 0x00, 0x00,
- 0x10, 0x91, 0x09, 0x00, 0x00, 0x00,
- 0x06, 0x50, 0x65, 0x00, 0x00, 0x00,
- 0x20, 0x4a, 0x84, 0x00, 0x00, 0x00,
- 0x8a, 0xa0, 0xaa, 0x00, 0x00, 0x00,
- 0x33, 0x03, 0x30, 0x00, 0x00, 0x00,
- 0x4c, 0x84, 0xc8, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask24_9[54] =
- {
- 0xd3, 0x65, 0x36, 0x00, 0x00, 0x00,
- 0x64, 0x26, 0x42, 0x00, 0x00, 0x00,
- 0x18, 0x41, 0xc4, 0x00, 0x00, 0x00,
- 0xa0, 0x4a, 0x04, 0x00, 0x00, 0x00,
- 0x81, 0x38, 0x13, 0x00, 0x00, 0x00,
- 0x22, 0xa2, 0x2a, 0x00, 0x00, 0x00,
- 0x08, 0x70, 0x87, 0x00, 0x00, 0x00,
- 0x04, 0x90, 0x49, 0x00, 0x00, 0x00,
- 0x01, 0xc0, 0x1c, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_1[6] =
- {
- 0xff, 0xff, 0xff, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_10[60] =
- {
- 0x51, 0x4d, 0x10, 0x00, 0x00, 0x00,
- 0x45, 0x14, 0x45, 0x00, 0x00, 0x00,
- 0x80, 0xd1, 0x09, 0x80, 0x00, 0x00,
- 0x24, 0x2a, 0x05, 0x00, 0x00, 0x00,
- 0x0a, 0x24, 0xa0, 0x80, 0x00, 0x00,
- 0x00, 0xe4, 0x03, 0x00, 0x00, 0x00,
- 0xb8, 0x08, 0x02, 0x80, 0x00, 0x00,
- 0x09, 0x10, 0xc9, 0x00, 0x00, 0x00,
- 0x56, 0x00, 0x58, 0x80, 0x00, 0x00,
- 0xa2, 0x86, 0x22, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_11[66] =
- {
- 0x53, 0x65, 0x12, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x21, 0x00, 0x00, 0x00,
- 0x10, 0x91, 0x34, 0x00, 0x00, 0x00,
- 0x00, 0x72, 0x50, 0x00, 0x00, 0x00,
- 0x0c, 0x11, 0x81, 0x80, 0x00, 0x00,
- 0x40, 0xc0, 0xa2, 0x00, 0x00, 0x00,
- 0x6a, 0x08, 0x88, 0x80, 0x00, 0x00,
- 0x86, 0x00, 0x68, 0x00, 0x00, 0x00,
- 0x24, 0x8e, 0x02, 0x00, 0x00, 0x00,
- 0x89, 0x08, 0x44, 0x00, 0x00, 0x00,
- 0xc0, 0x24, 0x41, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_12[72] =
- {
- 0x10, 0x62, 0x82, 0x80, 0x00, 0x00,
- 0x02, 0x38, 0x45, 0x00, 0x00, 0x00,
- 0x40, 0x56, 0x04, 0x00, 0x00, 0x00,
- 0x21, 0x80, 0x54, 0x80, 0x00, 0x00,
- 0x81, 0x10, 0x29, 0x80, 0x00, 0x00,
- 0x14, 0x80, 0x13, 0x00, 0x00, 0x00,
- 0x98, 0x04, 0x81, 0x00, 0x00, 0x00,
- 0x08, 0x92, 0x48, 0x00, 0x00, 0x00,
- 0x62, 0x09, 0x40, 0x00, 0x00, 0x00,
- 0x24, 0x28, 0xa0, 0x00, 0x00, 0x00,
- 0x8a, 0x01, 0x18, 0x00, 0x00, 0x00,
- 0x84, 0x45, 0x22, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_13[78] =
- {
- 0x51, 0x4d, 0x12, 0x00, 0x00, 0x00,
- 0xc5, 0x14, 0x6d, 0x00, 0x00, 0x00,
- 0x21, 0x81, 0x54, 0x80, 0x00, 0x00,
- 0x12, 0x32, 0x17, 0x00, 0x00, 0x00,
- 0x08, 0xe2, 0x8c, 0x80, 0x00, 0x00,
- 0x2e, 0x0a, 0xa2, 0x00, 0x00, 0x00,
- 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x65, 0x00, 0x00, 0x00,
- 0x90, 0x9b, 0x14, 0x00, 0x00, 0x00,
- 0x02, 0x52, 0xb0, 0x80, 0x00, 0x00,
- 0x06, 0xa1, 0x4c, 0x80, 0x00, 0x00,
- 0x2c, 0x0c, 0x88, 0x80, 0x00, 0x00,
- 0x88, 0x68, 0x4b, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_14[84] =
- {
- 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x65, 0x00, 0x00, 0x00,
- 0x90, 0x9b, 0x14, 0x00, 0x00, 0x00,
- 0x02, 0x52, 0xb0, 0x80, 0x00, 0x00,
- 0x06, 0xa1, 0x4c, 0x80, 0x00, 0x00,
- 0x2c, 0x0c, 0x88, 0x80, 0x00, 0x00,
- 0x88, 0x68, 0x4b, 0x00, 0x00, 0x00,
- 0x51, 0x4d, 0x12, 0x00, 0x00, 0x00,
- 0xc5, 0x14, 0x6d, 0x00, 0x00, 0x00,
- 0x21, 0x81, 0x54, 0x80, 0x00, 0x00,
- 0x12, 0x32, 0x17, 0x00, 0x00, 0x00,
- 0x08, 0xe2, 0x8c, 0x80, 0x00, 0x00,
- 0x2e, 0x0a, 0xa2, 0x00, 0x00, 0x00,
- 0x73, 0x76, 0x61, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_15[90] =
- {
- 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x65, 0x00, 0x00, 0x00,
- 0x90, 0x9b, 0x14, 0x00, 0x00, 0x00,
- 0x02, 0x52, 0xb0, 0x80, 0x00, 0x00,
- 0x06, 0xa1, 0x4c, 0x80, 0x00, 0x00,
- 0x2c, 0x0c, 0x88, 0x80, 0x00, 0x00,
- 0x88, 0x68, 0x4b, 0x00, 0x00, 0x00,
- 0x20, 0x68, 0x0a, 0x80, 0x00, 0x00,
- 0x80, 0x33, 0x09, 0x00, 0x00, 0x00,
- 0x42, 0x41, 0x60, 0x80, 0x00, 0x00,
- 0x01, 0x90, 0x33, 0x00, 0x00, 0x00,
- 0x14, 0x14, 0x46, 0x00, 0x00, 0x00,
- 0x0a, 0x80, 0x81, 0x80, 0x00, 0x00,
- 0x38, 0x0d, 0x80, 0x00, 0x00, 0x00,
- 0xc5, 0x0a, 0x14, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_16[96] =
- {
- 0x20, 0x68, 0x0a, 0x80, 0x00, 0x00,
- 0x80, 0x33, 0x09, 0x00, 0x00, 0x00,
- 0x42, 0x41, 0x60, 0x80, 0x00, 0x00,
- 0x01, 0x90, 0x33, 0x00, 0x00, 0x00,
- 0x14, 0x14, 0x46, 0x00, 0x00, 0x00,
- 0x0a, 0x80, 0x81, 0x80, 0x00, 0x00,
- 0x38, 0x0d, 0x80, 0x00, 0x00, 0x00,
- 0xc5, 0x0a, 0x14, 0x00, 0x00, 0x00,
- 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x65, 0x00, 0x00, 0x00,
- 0x90, 0x9b, 0x14, 0x00, 0x00, 0x00,
- 0x02, 0x52, 0xb0, 0x80, 0x00, 0x00,
- 0x06, 0xa1, 0x4c, 0x80, 0x00, 0x00,
- 0x2c, 0x0c, 0x88, 0x80, 0x00, 0x00,
- 0x88, 0x68, 0x4b, 0x00, 0x00, 0x00,
- 0x16, 0xe8, 0xdc, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_17[102] =
- {
- 0x20, 0x68, 0x0a, 0x80, 0x00, 0x00,
- 0x80, 0x33, 0x09, 0x00, 0x00, 0x00,
- 0x42, 0x41, 0x60, 0x80, 0x00, 0x00,
- 0x01, 0x90, 0x33, 0x00, 0x00, 0x00,
- 0x14, 0x14, 0x46, 0x00, 0x00, 0x00,
- 0x0a, 0x80, 0x81, 0x80, 0x00, 0x00,
- 0x38, 0x0d, 0x80, 0x00, 0x00, 0x00,
- 0xc5, 0x0a, 0x14, 0x00, 0x00, 0x00,
- 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
- 0xe4, 0x26, 0x64, 0x00, 0x00, 0x00,
- 0x24, 0x41, 0x44, 0x00, 0x00, 0x00,
- 0xa1, 0x12, 0x14, 0x80, 0x00, 0x00,
- 0x18, 0x30, 0x2c, 0x80, 0x00, 0x00,
- 0x03, 0x99, 0x41, 0x00, 0x00, 0x00,
- 0x8a, 0x18, 0x0a, 0x80, 0x00, 0x00,
- 0x04, 0x90, 0xa9, 0x00, 0x00, 0x00,
- 0x00, 0xe4, 0x01, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_18[108] =
- {
- 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
- 0xe4, 0x26, 0x64, 0x00, 0x00, 0x00,
- 0x24, 0x41, 0x44, 0x00, 0x00, 0x00,
- 0xa1, 0x12, 0x14, 0x80, 0x00, 0x00,
- 0x18, 0x30, 0x2c, 0x80, 0x00, 0x00,
- 0x03, 0x99, 0x41, 0x00, 0x00, 0x00,
- 0x8a, 0x18, 0x0a, 0x80, 0x00, 0x00,
- 0x04, 0x90, 0xa9, 0x00, 0x00, 0x00,
- 0x00, 0xe4, 0x01, 0x80, 0x00, 0x00,
- 0x20, 0x68, 0x0a, 0x80, 0x00, 0x00,
- 0x80, 0x33, 0x09, 0x00, 0x00, 0x00,
- 0x42, 0x41, 0x60, 0x80, 0x00, 0x00,
- 0x01, 0x90, 0x33, 0x00, 0x00, 0x00,
- 0x14, 0x14, 0x46, 0x00, 0x00, 0x00,
- 0x0a, 0x80, 0x81, 0x80, 0x00, 0x00,
- 0x38, 0x0d, 0x80, 0x00, 0x00, 0x00,
- 0xc5, 0x0a, 0x14, 0x00, 0x00, 0x00,
- 0xce, 0x9b, 0xe1, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_19[114] =
- {
- 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
- 0xe4, 0x26, 0x64, 0x00, 0x00, 0x00,
- 0x24, 0x41, 0x44, 0x00, 0x00, 0x00,
- 0xa1, 0x12, 0x14, 0x80, 0x00, 0x00,
- 0x18, 0x30, 0x2c, 0x80, 0x00, 0x00,
- 0x03, 0x99, 0x41, 0x00, 0x00, 0x00,
- 0x8a, 0x18, 0x0a, 0x80, 0x00, 0x00,
- 0x04, 0x90, 0xa9, 0x00, 0x00, 0x00,
- 0x00, 0xe4, 0x01, 0x80, 0x00, 0x00,
- 0x51, 0x4d, 0x10, 0x00, 0x00, 0x00,
- 0x45, 0x14, 0x45, 0x00, 0x00, 0x00,
- 0x80, 0xd1, 0x09, 0x80, 0x00, 0x00,
- 0x24, 0x2a, 0x05, 0x00, 0x00, 0x00,
- 0x0a, 0x24, 0xa0, 0x80, 0x00, 0x00,
- 0x00, 0xe4, 0x03, 0x00, 0x00, 0x00,
- 0xb8, 0x08, 0x02, 0x80, 0x00, 0x00,
- 0x09, 0x10, 0xc9, 0x00, 0x00, 0x00,
- 0x56, 0x00, 0x58, 0x80, 0x00, 0x00,
- 0xa2, 0x86, 0x22, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_2[12] =
- {
- 0xec, 0xce, 0xcc, 0x00, 0x00, 0x00,
- 0x93, 0xb1, 0xb3, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_20[120] =
- {
- 0x51, 0x4d, 0x10, 0x00, 0x00, 0x00,
- 0x45, 0x14, 0x45, 0x00, 0x00, 0x00,
- 0x80, 0xd1, 0x09, 0x80, 0x00, 0x00,
- 0x24, 0x2a, 0x05, 0x00, 0x00, 0x00,
- 0x0a, 0x24, 0xa0, 0x80, 0x00, 0x00,
- 0x00, 0xe4, 0x03, 0x00, 0x00, 0x00,
- 0xb8, 0x08, 0x02, 0x80, 0x00, 0x00,
- 0x09, 0x10, 0xc9, 0x00, 0x00, 0x00,
- 0x56, 0x00, 0x58, 0x80, 0x00, 0x00,
- 0xa2, 0x86, 0x22, 0x00, 0x00, 0x00,
- 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
- 0xe4, 0x26, 0x64, 0x00, 0x00, 0x00,
- 0x24, 0x41, 0x44, 0x00, 0x00, 0x00,
- 0xa1, 0x12, 0x14, 0x80, 0x00, 0x00,
- 0x18, 0x30, 0x2c, 0x80, 0x00, 0x00,
- 0x03, 0x99, 0x41, 0x00, 0x00, 0x00,
- 0x8a, 0x18, 0x0a, 0x80, 0x00, 0x00,
- 0x04, 0x90, 0xa9, 0x00, 0x00, 0x00,
- 0x00, 0xe4, 0x01, 0x80, 0x00, 0x00,
- 0x1b, 0x8a, 0xa0, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_21[126] =
- {
- 0x51, 0x4d, 0x10, 0x00, 0x00, 0x00,
- 0x45, 0x14, 0x45, 0x00, 0x00, 0x00,
- 0x80, 0xd1, 0x09, 0x80, 0x00, 0x00,
- 0x24, 0x2a, 0x05, 0x00, 0x00, 0x00,
- 0x0a, 0x24, 0xa0, 0x80, 0x00, 0x00,
- 0x00, 0xe4, 0x03, 0x00, 0x00, 0x00,
- 0xb8, 0x08, 0x02, 0x80, 0x00, 0x00,
- 0x09, 0x10, 0xc9, 0x00, 0x00, 0x00,
- 0x56, 0x00, 0x58, 0x80, 0x00, 0x00,
- 0xa2, 0x86, 0x22, 0x00, 0x00, 0x00,
- 0x53, 0x65, 0x12, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x21, 0x00, 0x00, 0x00,
- 0x10, 0x91, 0x34, 0x00, 0x00, 0x00,
- 0x00, 0x72, 0x50, 0x00, 0x00, 0x00,
- 0x0c, 0x11, 0x81, 0x80, 0x00, 0x00,
- 0x40, 0xc0, 0xa2, 0x00, 0x00, 0x00,
- 0x6a, 0x08, 0x88, 0x80, 0x00, 0x00,
- 0x86, 0x00, 0x68, 0x00, 0x00, 0x00,
- 0x24, 0x8e, 0x02, 0x00, 0x00, 0x00,
- 0x89, 0x08, 0x44, 0x00, 0x00, 0x00,
- 0xc0, 0x24, 0x41, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_22[132] =
- {
- 0x53, 0x65, 0x12, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x21, 0x00, 0x00, 0x00,
- 0x10, 0x91, 0x34, 0x00, 0x00, 0x00,
- 0x00, 0x72, 0x50, 0x00, 0x00, 0x00,
- 0x0c, 0x11, 0x81, 0x80, 0x00, 0x00,
- 0x40, 0xc0, 0xa2, 0x00, 0x00, 0x00,
- 0x6a, 0x08, 0x88, 0x80, 0x00, 0x00,
- 0x86, 0x00, 0x68, 0x00, 0x00, 0x00,
- 0x24, 0x8e, 0x02, 0x00, 0x00, 0x00,
- 0x89, 0x08, 0x44, 0x00, 0x00, 0x00,
- 0xc0, 0x24, 0x41, 0x80, 0x00, 0x00,
- 0x51, 0x4d, 0x10, 0x00, 0x00, 0x00,
- 0x45, 0x14, 0x45, 0x00, 0x00, 0x00,
- 0x80, 0xd1, 0x09, 0x80, 0x00, 0x00,
- 0x24, 0x2a, 0x05, 0x00, 0x00, 0x00,
- 0x0a, 0x24, 0xa0, 0x80, 0x00, 0x00,
- 0x00, 0xe4, 0x03, 0x00, 0x00, 0x00,
- 0xb8, 0x08, 0x02, 0x80, 0x00, 0x00,
- 0x09, 0x10, 0xc9, 0x00, 0x00, 0x00,
- 0x56, 0x00, 0x58, 0x80, 0x00, 0x00,
- 0xa2, 0x86, 0x22, 0x00, 0x00, 0x00,
- 0x15, 0xa2, 0x99, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_23[138] =
- {
- 0x53, 0x65, 0x12, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x21, 0x00, 0x00, 0x00,
- 0x10, 0x91, 0x34, 0x00, 0x00, 0x00,
- 0x00, 0x72, 0x50, 0x00, 0x00, 0x00,
- 0x0c, 0x11, 0x81, 0x80, 0x00, 0x00,
- 0x40, 0xc0, 0xa2, 0x00, 0x00, 0x00,
- 0x6a, 0x08, 0x88, 0x80, 0x00, 0x00,
- 0x86, 0x00, 0x68, 0x00, 0x00, 0x00,
- 0x24, 0x8e, 0x02, 0x00, 0x00, 0x00,
- 0x89, 0x08, 0x44, 0x00, 0x00, 0x00,
- 0xc0, 0x24, 0x41, 0x80, 0x00, 0x00,
- 0x10, 0x62, 0x82, 0x80, 0x00, 0x00,
- 0x02, 0x38, 0x45, 0x00, 0x00, 0x00,
- 0x40, 0x56, 0x04, 0x00, 0x00, 0x00,
- 0x21, 0x80, 0x54, 0x80, 0x00, 0x00,
- 0x81, 0x10, 0x29, 0x80, 0x00, 0x00,
- 0x14, 0x80, 0x13, 0x00, 0x00, 0x00,
- 0x98, 0x04, 0x81, 0x00, 0x00, 0x00,
- 0x08, 0x92, 0x48, 0x00, 0x00, 0x00,
- 0x62, 0x09, 0x40, 0x00, 0x00, 0x00,
- 0x24, 0x28, 0xa0, 0x00, 0x00, 0x00,
- 0x8a, 0x01, 0x18, 0x00, 0x00, 0x00,
- 0x84, 0x45, 0x22, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_24[144] =
- {
- 0x10, 0x62, 0x82, 0x80, 0x00, 0x00,
- 0x02, 0x38, 0x45, 0x00, 0x00, 0x00,
- 0x40, 0x56, 0x04, 0x00, 0x00, 0x00,
- 0x21, 0x80, 0x54, 0x80, 0x00, 0x00,
- 0x81, 0x10, 0x29, 0x80, 0x00, 0x00,
- 0x14, 0x80, 0x13, 0x00, 0x00, 0x00,
- 0x98, 0x04, 0x81, 0x00, 0x00, 0x00,
- 0x08, 0x92, 0x48, 0x00, 0x00, 0x00,
- 0x62, 0x09, 0x40, 0x00, 0x00, 0x00,
- 0x24, 0x28, 0xa0, 0x00, 0x00, 0x00,
- 0x8a, 0x01, 0x18, 0x00, 0x00, 0x00,
- 0x84, 0x45, 0x22, 0x00, 0x00, 0x00,
- 0x53, 0x65, 0x12, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x21, 0x00, 0x00, 0x00,
- 0x10, 0x91, 0x34, 0x00, 0x00, 0x00,
- 0x00, 0x72, 0x50, 0x00, 0x00, 0x00,
- 0x0c, 0x11, 0x81, 0x80, 0x00, 0x00,
- 0x40, 0xc0, 0xa2, 0x00, 0x00, 0x00,
- 0x6a, 0x08, 0x88, 0x80, 0x00, 0x00,
- 0x86, 0x00, 0x68, 0x00, 0x00, 0x00,
- 0x24, 0x8e, 0x02, 0x00, 0x00, 0x00,
- 0x89, 0x08, 0x44, 0x00, 0x00, 0x00,
- 0xc0, 0x24, 0x41, 0x80, 0x00, 0x00,
- 0xf9, 0x0c, 0x14, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_25[150] =
- {
- 0x10, 0x62, 0x82, 0x80, 0x00, 0x00,
- 0x02, 0x38, 0x45, 0x00, 0x00, 0x00,
- 0x40, 0x56, 0x04, 0x00, 0x00, 0x00,
- 0x21, 0x80, 0x54, 0x80, 0x00, 0x00,
- 0x81, 0x10, 0x29, 0x80, 0x00, 0x00,
- 0x14, 0x80, 0x13, 0x00, 0x00, 0x00,
- 0x98, 0x04, 0x81, 0x00, 0x00, 0x00,
- 0x08, 0x92, 0x48, 0x00, 0x00, 0x00,
- 0x62, 0x09, 0x40, 0x00, 0x00, 0x00,
- 0x24, 0x28, 0xa0, 0x00, 0x00, 0x00,
- 0x8a, 0x01, 0x18, 0x00, 0x00, 0x00,
- 0x84, 0x45, 0x22, 0x00, 0x00, 0x00,
- 0x10, 0x65, 0x12, 0x00, 0x00, 0x00,
- 0x02, 0x36, 0x64, 0x00, 0x00, 0x00,
- 0x40, 0x50, 0x54, 0x80, 0x00, 0x00,
- 0x21, 0x88, 0x12, 0x00, 0x00, 0x00,
- 0x81, 0x19, 0x40, 0x00, 0x00, 0x00,
- 0x14, 0x83, 0x08, 0x00, 0x00, 0x00,
- 0x98, 0x02, 0x11, 0x00, 0x00, 0x00,
- 0x08, 0x90, 0x3c, 0x00, 0x00, 0x00,
- 0x62, 0x0e, 0x80, 0x00, 0x00, 0x00,
- 0x24, 0x20, 0xa1, 0x00, 0x00, 0x00,
- 0x8a, 0x08, 0x01, 0x80, 0x00, 0x00,
- 0x84, 0x40, 0x49, 0x00, 0x00, 0x00,
- 0x1c, 0x20, 0x8a, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_3[18] =
- {
- 0x9b, 0x89, 0x9b, 0x00, 0x00, 0x00,
- 0x4f, 0x14, 0x6d, 0x80, 0x00, 0x00,
- 0x3c, 0x63, 0x72, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_4[24] =
- {
- 0x8b, 0x24, 0x9b, 0x00, 0x00, 0x00,
- 0x14, 0xb2, 0x6d, 0x00, 0x00, 0x00,
- 0x22, 0xd8, 0x56, 0x80, 0x00, 0x00,
- 0x45, 0x55, 0x25, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_5[30] =
- {
- 0x53, 0x65, 0x13, 0x00, 0x00, 0x00,
- 0x64, 0x26, 0x64, 0x00, 0x00, 0x00,
- 0x0c, 0xc0, 0xc6, 0x80, 0x00, 0x00,
- 0x82, 0xaa, 0x1c, 0x00, 0x00, 0x00,
- 0x09, 0x32, 0x29, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_6[36] =
- {
- 0x51, 0x4d, 0x12, 0x00, 0x00, 0x00,
- 0xc5, 0x14, 0x6d, 0x00, 0x00, 0x00,
- 0x21, 0x81, 0x54, 0x80, 0x00, 0x00,
- 0x12, 0x32, 0x17, 0x00, 0x00, 0x00,
- 0x08, 0xe2, 0x8c, 0x80, 0x00, 0x00,
- 0x2e, 0x0a, 0xa2, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_7[42] =
- {
- 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
- 0x21, 0x32, 0x65, 0x00, 0x00, 0x00,
- 0x90, 0x9b, 0x14, 0x00, 0x00, 0x00,
- 0x02, 0x52, 0xb0, 0x80, 0x00, 0x00,
- 0x06, 0xa1, 0x4c, 0x80, 0x00, 0x00,
- 0x2c, 0x0c, 0x88, 0x80, 0x00, 0x00,
- 0x88, 0x68, 0x4b, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_8[48] =
- {
- 0x20, 0x68, 0x0a, 0x80, 0x00, 0x00,
- 0x80, 0x33, 0x09, 0x00, 0x00, 0x00,
- 0x42, 0x41, 0x60, 0x80, 0x00, 0x00,
- 0x01, 0x90, 0x33, 0x00, 0x00, 0x00,
- 0x14, 0x14, 0x46, 0x00, 0x00, 0x00,
- 0x0a, 0x80, 0x81, 0x80, 0x00, 0x00,
- 0x38, 0x0d, 0x80, 0x00, 0x00, 0x00,
- 0xc5, 0x0a, 0x14, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask25_9[54] =
- {
- 0x53, 0x65, 0x92, 0x00, 0x00, 0x00,
- 0xe4, 0x26, 0x64, 0x00, 0x00, 0x00,
- 0x24, 0x41, 0x44, 0x00, 0x00, 0x00,
- 0xa1, 0x12, 0x14, 0x80, 0x00, 0x00,
- 0x18, 0x30, 0x2c, 0x80, 0x00, 0x00,
- 0x03, 0x99, 0x41, 0x00, 0x00, 0x00,
- 0x8a, 0x18, 0x0a, 0x80, 0x00, 0x00,
- 0x04, 0x90, 0xa9, 0x00, 0x00, 0x00,
- 0x00, 0xe4, 0x01, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_1[6] =
- {
- 0xff, 0xff, 0xff, 0xc0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_10[60] =
- {
- 0xd1, 0x06, 0x88, 0x00, 0x00, 0x00,
- 0x44, 0x52, 0x22, 0x80, 0x00, 0x00,
- 0x10, 0x98, 0x84, 0xc0, 0x00, 0x00,
- 0xa0, 0x55, 0x02, 0x80, 0x00, 0x00,
- 0x4a, 0x0a, 0x50, 0x40, 0x00, 0x00,
- 0x40, 0x32, 0x01, 0x80, 0x00, 0x00,
- 0x80, 0x2c, 0x01, 0x40, 0x00, 0x00,
- 0x0c, 0x90, 0x64, 0x80, 0x00, 0x00,
- 0x05, 0x88, 0x2c, 0x40, 0x00, 0x00,
- 0x62, 0x23, 0x11, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_11[66] =
- {
- 0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
- 0x22, 0x11, 0x10, 0x80, 0x00, 0x00,
- 0x13, 0x40, 0x9a, 0x00, 0x00, 0x00,
- 0x25, 0x01, 0x28, 0x00, 0x00, 0x00,
- 0x18, 0x18, 0xc0, 0xc0, 0x00, 0x00,
- 0x0a, 0x20, 0x51, 0x00, 0x00, 0x00,
- 0x88, 0x8c, 0x44, 0x40, 0x00, 0x00,
- 0x06, 0x80, 0x34, 0x00, 0x00, 0x00,
- 0xe0, 0x27, 0x01, 0x00, 0x00, 0x00,
- 0x84, 0x44, 0x22, 0x00, 0x00, 0x00,
- 0x44, 0x1a, 0x20, 0xc0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_12[72] =
- {
- 0x28, 0x29, 0x41, 0x40, 0x00, 0x00,
- 0x84, 0x54, 0x22, 0x80, 0x00, 0x00,
- 0x60, 0x43, 0x02, 0x00, 0x00, 0x00,
- 0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
- 0x02, 0x98, 0x14, 0xc0, 0x00, 0x00,
- 0x01, 0x30, 0x09, 0x80, 0x00, 0x00,
- 0x48, 0x12, 0x40, 0x80, 0x00, 0x00,
- 0x24, 0x81, 0x24, 0x00, 0x00, 0x00,
- 0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
- 0x8a, 0x04, 0x50, 0x00, 0x00, 0x00,
- 0x11, 0x80, 0x8c, 0x00, 0x00, 0x00,
- 0x52, 0x22, 0x91, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_13[78] =
- {
- 0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
- 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
- 0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
- 0x81, 0x24, 0x09, 0x00, 0x00, 0x00,
- 0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
- 0x30, 0x81, 0x84, 0x00, 0x00, 0x00,
- 0x21, 0x11, 0x08, 0x80, 0x00, 0x00,
- 0x03, 0xc0, 0x1e, 0x00, 0x00, 0x00,
- 0xe8, 0x07, 0x40, 0x00, 0x00, 0x00,
- 0x0a, 0x10, 0x50, 0x80, 0x00, 0x00,
- 0x80, 0x1c, 0x00, 0xc0, 0x00, 0x00,
- 0x04, 0x90, 0x24, 0x80, 0x00, 0x00,
- 0x08, 0xa8, 0x45, 0x40, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_14[84] =
- {
- 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
- 0x26, 0x51, 0x32, 0x80, 0x00, 0x00,
- 0xb1, 0x45, 0x8a, 0x00, 0x00, 0x00,
- 0x2b, 0x09, 0x58, 0x40, 0x00, 0x00,
- 0x14, 0xc8, 0xa6, 0x40, 0x00, 0x00,
- 0xc8, 0x8e, 0x44, 0x40, 0x00, 0x00,
- 0x84, 0xb4, 0x25, 0x80, 0x00, 0x00,
- 0xd1, 0x26, 0x89, 0x00, 0x00, 0x00,
- 0x46, 0xd2, 0x36, 0x80, 0x00, 0x00,
- 0x15, 0x48, 0xaa, 0x40, 0x00, 0x00,
- 0x21, 0x71, 0x0b, 0x80, 0x00, 0x00,
- 0x28, 0xc9, 0x46, 0x40, 0x00, 0x00,
- 0xaa, 0x25, 0x51, 0x00, 0x00, 0x00,
- 0x5d, 0xa7, 0x78, 0x40, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_15[90] =
- {
- 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
- 0x26, 0x51, 0x32, 0x80, 0x00, 0x00,
- 0xb1, 0x45, 0x8a, 0x00, 0x00, 0x00,
- 0x2b, 0x09, 0x58, 0x40, 0x00, 0x00,
- 0x14, 0xc8, 0xa6, 0x40, 0x00, 0x00,
- 0xc8, 0x8e, 0x44, 0x40, 0x00, 0x00,
- 0x84, 0xb4, 0x25, 0x80, 0x00, 0x00,
- 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
- 0x30, 0x91, 0x84, 0x80, 0x00, 0x00,
- 0x16, 0x08, 0xb0, 0x40, 0x00, 0x00,
- 0x03, 0x30, 0x19, 0x80, 0x00, 0x00,
- 0x44, 0x62, 0x23, 0x00, 0x00, 0x00,
- 0x08, 0x18, 0x40, 0xc0, 0x00, 0x00,
- 0xd8, 0x06, 0xc0, 0x00, 0x00, 0x00,
- 0xa1, 0x45, 0x0a, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_16[96] =
- {
- 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
- 0x30, 0x91, 0x84, 0x80, 0x00, 0x00,
- 0x16, 0x08, 0xb0, 0x40, 0x00, 0x00,
- 0x03, 0x30, 0x19, 0x80, 0x00, 0x00,
- 0x44, 0x62, 0x23, 0x00, 0x00, 0x00,
- 0x08, 0x18, 0x40, 0xc0, 0x00, 0x00,
- 0xd8, 0x06, 0xc0, 0x00, 0x00, 0x00,
- 0xa1, 0x45, 0x0a, 0x00, 0x00, 0x00,
- 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
- 0x26, 0x51, 0x32, 0x80, 0x00, 0x00,
- 0xb1, 0x45, 0x8a, 0x00, 0x00, 0x00,
- 0x2b, 0x09, 0x58, 0x40, 0x00, 0x00,
- 0x14, 0xc8, 0xa6, 0x40, 0x00, 0x00,
- 0xc8, 0x8e, 0x44, 0x40, 0x00, 0x00,
- 0x84, 0xb4, 0x25, 0x80, 0x00, 0x00,
- 0x3c, 0xaf, 0x88, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_17[102] =
- {
- 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
- 0x30, 0x91, 0x84, 0x80, 0x00, 0x00,
- 0x16, 0x08, 0xb0, 0x40, 0x00, 0x00,
- 0x03, 0x30, 0x19, 0x80, 0x00, 0x00,
- 0x44, 0x62, 0x23, 0x00, 0x00, 0x00,
- 0x08, 0x18, 0x40, 0xc0, 0x00, 0x00,
- 0xd8, 0x06, 0xc0, 0x00, 0x00, 0x00,
- 0xa1, 0x45, 0x0a, 0x00, 0x00, 0x00,
- 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
- 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
- 0x14, 0x40, 0xa2, 0x00, 0x00, 0x00,
- 0x21, 0x49, 0x0a, 0x40, 0x00, 0x00,
- 0x02, 0xc8, 0x16, 0x40, 0x00, 0x00,
- 0x94, 0x14, 0xa0, 0x80, 0x00, 0x00,
- 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
- 0x0a, 0x90, 0x54, 0x80, 0x00, 0x00,
- 0x40, 0x1a, 0x00, 0xc0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_18[108] =
- {
- 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
- 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
- 0x14, 0x40, 0xa2, 0x00, 0x00, 0x00,
- 0x21, 0x49, 0x0a, 0x40, 0x00, 0x00,
- 0x02, 0xc8, 0x16, 0x40, 0x00, 0x00,
- 0x94, 0x14, 0xa0, 0x80, 0x00, 0x00,
- 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
- 0x0a, 0x90, 0x54, 0x80, 0x00, 0x00,
- 0x40, 0x1a, 0x00, 0xc0, 0x00, 0x00,
- 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
- 0x30, 0x91, 0x84, 0x80, 0x00, 0x00,
- 0x16, 0x08, 0xb0, 0x40, 0x00, 0x00,
- 0x03, 0x30, 0x19, 0x80, 0x00, 0x00,
- 0x44, 0x62, 0x23, 0x00, 0x00, 0x00,
- 0x08, 0x18, 0x40, 0xc0, 0x00, 0x00,
- 0xd8, 0x06, 0xc0, 0x00, 0x00, 0x00,
- 0xa1, 0x45, 0x0a, 0x00, 0x00, 0x00,
- 0xaa, 0x0c, 0x83, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_19[114] =
- {
- 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
- 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
- 0x14, 0x40, 0xa2, 0x00, 0x00, 0x00,
- 0x21, 0x49, 0x0a, 0x40, 0x00, 0x00,
- 0x02, 0xc8, 0x16, 0x40, 0x00, 0x00,
- 0x94, 0x14, 0xa0, 0x80, 0x00, 0x00,
- 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
- 0x0a, 0x90, 0x54, 0x80, 0x00, 0x00,
- 0x40, 0x1a, 0x00, 0xc0, 0x00, 0x00,
- 0xd1, 0x06, 0x88, 0x00, 0x00, 0x00,
- 0x44, 0x52, 0x22, 0x80, 0x00, 0x00,
- 0x10, 0x98, 0x84, 0xc0, 0x00, 0x00,
- 0xa0, 0x55, 0x02, 0x80, 0x00, 0x00,
- 0x4a, 0x0a, 0x50, 0x40, 0x00, 0x00,
- 0x40, 0x32, 0x01, 0x80, 0x00, 0x00,
- 0x80, 0x2c, 0x01, 0x40, 0x00, 0x00,
- 0x0c, 0x90, 0x64, 0x80, 0x00, 0x00,
- 0x05, 0x88, 0x2c, 0x40, 0x00, 0x00,
- 0x62, 0x23, 0x11, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_2[12] =
- {
- 0xec, 0xc7, 0x66, 0x00, 0x00, 0x00,
- 0x1b, 0x38, 0xd9, 0xc0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_20[120] =
- {
- 0xd1, 0x06, 0x88, 0x00, 0x00, 0x00,
- 0x44, 0x52, 0x22, 0x80, 0x00, 0x00,
- 0x10, 0x98, 0x84, 0xc0, 0x00, 0x00,
- 0xa0, 0x55, 0x02, 0x80, 0x00, 0x00,
- 0x4a, 0x0a, 0x50, 0x40, 0x00, 0x00,
- 0x40, 0x32, 0x01, 0x80, 0x00, 0x00,
- 0x80, 0x2c, 0x01, 0x40, 0x00, 0x00,
- 0x0c, 0x90, 0x64, 0x80, 0x00, 0x00,
- 0x05, 0x88, 0x2c, 0x40, 0x00, 0x00,
- 0x62, 0x23, 0x11, 0x00, 0x00, 0x00,
- 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
- 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
- 0x14, 0x40, 0xa2, 0x00, 0x00, 0x00,
- 0x21, 0x49, 0x0a, 0x40, 0x00, 0x00,
- 0x02, 0xc8, 0x16, 0x40, 0x00, 0x00,
- 0x94, 0x14, 0xa0, 0x80, 0x00, 0x00,
- 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
- 0x0a, 0x90, 0x54, 0x80, 0x00, 0x00,
- 0x40, 0x1a, 0x00, 0xc0, 0x00, 0x00,
- 0xf4, 0x08, 0xec, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_21[126] =
- {
- 0xd1, 0x06, 0x88, 0x00, 0x00, 0x00,
- 0x44, 0x52, 0x22, 0x80, 0x00, 0x00,
- 0x10, 0x98, 0x84, 0xc0, 0x00, 0x00,
- 0xa0, 0x55, 0x02, 0x80, 0x00, 0x00,
- 0x4a, 0x0a, 0x50, 0x40, 0x00, 0x00,
- 0x40, 0x32, 0x01, 0x80, 0x00, 0x00,
- 0x80, 0x2c, 0x01, 0x40, 0x00, 0x00,
- 0x0c, 0x90, 0x64, 0x80, 0x00, 0x00,
- 0x05, 0x88, 0x2c, 0x40, 0x00, 0x00,
- 0x62, 0x23, 0x11, 0x00, 0x00, 0x00,
- 0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
- 0x22, 0x11, 0x10, 0x80, 0x00, 0x00,
- 0x13, 0x40, 0x9a, 0x00, 0x00, 0x00,
- 0x25, 0x01, 0x28, 0x00, 0x00, 0x00,
- 0x18, 0x18, 0xc0, 0xc0, 0x00, 0x00,
- 0x0a, 0x20, 0x51, 0x00, 0x00, 0x00,
- 0x88, 0x8c, 0x44, 0x40, 0x00, 0x00,
- 0x06, 0x80, 0x34, 0x00, 0x00, 0x00,
- 0xe0, 0x27, 0x01, 0x00, 0x00, 0x00,
- 0x84, 0x44, 0x22, 0x00, 0x00, 0x00,
- 0x44, 0x1a, 0x20, 0xc0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_22[132] =
- {
- 0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
- 0x22, 0x11, 0x10, 0x80, 0x00, 0x00,
- 0x13, 0x40, 0x9a, 0x00, 0x00, 0x00,
- 0x25, 0x01, 0x28, 0x00, 0x00, 0x00,
- 0x18, 0x18, 0xc0, 0xc0, 0x00, 0x00,
- 0x0a, 0x20, 0x51, 0x00, 0x00, 0x00,
- 0x88, 0x8c, 0x44, 0x40, 0x00, 0x00,
- 0x06, 0x80, 0x34, 0x00, 0x00, 0x00,
- 0xe0, 0x27, 0x01, 0x00, 0x00, 0x00,
- 0x84, 0x44, 0x22, 0x00, 0x00, 0x00,
- 0x44, 0x1a, 0x20, 0xc0, 0x00, 0x00,
- 0xd1, 0x06, 0x88, 0x00, 0x00, 0x00,
- 0x44, 0x52, 0x22, 0x80, 0x00, 0x00,
- 0x10, 0x98, 0x84, 0xc0, 0x00, 0x00,
- 0xa0, 0x55, 0x02, 0x80, 0x00, 0x00,
- 0x4a, 0x0a, 0x50, 0x40, 0x00, 0x00,
- 0x40, 0x32, 0x01, 0x80, 0x00, 0x00,
- 0x80, 0x2c, 0x01, 0x40, 0x00, 0x00,
- 0x0c, 0x90, 0x64, 0x80, 0x00, 0x00,
- 0x05, 0x88, 0x2c, 0x40, 0x00, 0x00,
- 0x62, 0x23, 0x11, 0x00, 0x00, 0x00,
- 0x13, 0xc6, 0x6b, 0x40, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_23[138] =
- {
- 0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
- 0x22, 0x11, 0x10, 0x80, 0x00, 0x00,
- 0x13, 0x40, 0x9a, 0x00, 0x00, 0x00,
- 0x25, 0x01, 0x28, 0x00, 0x00, 0x00,
- 0x18, 0x18, 0xc0, 0xc0, 0x00, 0x00,
- 0x0a, 0x20, 0x51, 0x00, 0x00, 0x00,
- 0x88, 0x8c, 0x44, 0x40, 0x00, 0x00,
- 0x06, 0x80, 0x34, 0x00, 0x00, 0x00,
- 0xe0, 0x27, 0x01, 0x00, 0x00, 0x00,
- 0x84, 0x44, 0x22, 0x00, 0x00, 0x00,
- 0x44, 0x1a, 0x20, 0xc0, 0x00, 0x00,
- 0x28, 0x29, 0x41, 0x40, 0x00, 0x00,
- 0x84, 0x54, 0x22, 0x80, 0x00, 0x00,
- 0x60, 0x43, 0x02, 0x00, 0x00, 0x00,
- 0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
- 0x02, 0x98, 0x14, 0xc0, 0x00, 0x00,
- 0x01, 0x30, 0x09, 0x80, 0x00, 0x00,
- 0x48, 0x12, 0x40, 0x80, 0x00, 0x00,
- 0x24, 0x81, 0x24, 0x00, 0x00, 0x00,
- 0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
- 0x8a, 0x04, 0x50, 0x00, 0x00, 0x00,
- 0x11, 0x80, 0x8c, 0x00, 0x00, 0x00,
- 0x52, 0x22, 0x91, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_24[144] =
- {
- 0x28, 0x29, 0x41, 0x40, 0x00, 0x00,
- 0x84, 0x54, 0x22, 0x80, 0x00, 0x00,
- 0x60, 0x43, 0x02, 0x00, 0x00, 0x00,
- 0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
- 0x02, 0x98, 0x14, 0xc0, 0x00, 0x00,
- 0x01, 0x30, 0x09, 0x80, 0x00, 0x00,
- 0x48, 0x12, 0x40, 0x80, 0x00, 0x00,
- 0x24, 0x81, 0x24, 0x00, 0x00, 0x00,
- 0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
- 0x8a, 0x04, 0x50, 0x00, 0x00, 0x00,
- 0x11, 0x80, 0x8c, 0x00, 0x00, 0x00,
- 0x52, 0x22, 0x91, 0x00, 0x00, 0x00,
- 0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
- 0x22, 0x11, 0x10, 0x80, 0x00, 0x00,
- 0x13, 0x40, 0x9a, 0x00, 0x00, 0x00,
- 0x25, 0x01, 0x28, 0x00, 0x00, 0x00,
- 0x18, 0x18, 0xc0, 0xc0, 0x00, 0x00,
- 0x0a, 0x20, 0x51, 0x00, 0x00, 0x00,
- 0x88, 0x8c, 0x44, 0x40, 0x00, 0x00,
- 0x06, 0x80, 0x34, 0x00, 0x00, 0x00,
- 0xe0, 0x27, 0x01, 0x00, 0x00, 0x00,
- 0x84, 0x44, 0x22, 0x00, 0x00, 0x00,
- 0x44, 0x1a, 0x20, 0xc0, 0x00, 0x00,
- 0xdb, 0x4d, 0xd8, 0x40, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_25[150] =
- {
- 0x28, 0x29, 0x41, 0x40, 0x00, 0x00,
- 0x84, 0x54, 0x22, 0x80, 0x00, 0x00,
- 0x60, 0x43, 0x02, 0x00, 0x00, 0x00,
- 0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
- 0x02, 0x98, 0x14, 0xc0, 0x00, 0x00,
- 0x01, 0x30, 0x09, 0x80, 0x00, 0x00,
- 0x48, 0x12, 0x40, 0x80, 0x00, 0x00,
- 0x24, 0x81, 0x24, 0x00, 0x00, 0x00,
- 0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
- 0x8a, 0x04, 0x50, 0x00, 0x00, 0x00,
- 0x11, 0x80, 0x8c, 0x00, 0x00, 0x00,
- 0x52, 0x22, 0x91, 0x00, 0x00, 0x00,
- 0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
- 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
- 0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
- 0x81, 0x24, 0x09, 0x00, 0x00, 0x00,
- 0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
- 0x30, 0x81, 0x84, 0x00, 0x00, 0x00,
- 0x21, 0x11, 0x08, 0x80, 0x00, 0x00,
- 0x03, 0xc0, 0x1e, 0x00, 0x00, 0x00,
- 0xe8, 0x07, 0x40, 0x00, 0x00, 0x00,
- 0x0a, 0x10, 0x50, 0x80, 0x00, 0x00,
- 0x80, 0x1c, 0x00, 0xc0, 0x00, 0x00,
- 0x04, 0x90, 0x24, 0x80, 0x00, 0x00,
- 0x08, 0xa8, 0x45, 0x40, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_26[156] =
- {
- 0x51, 0x22, 0x89, 0x00, 0x00, 0x00,
- 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
- 0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
- 0x81, 0x24, 0x09, 0x00, 0x00, 0x00,
- 0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
- 0x30, 0x81, 0x84, 0x00, 0x00, 0x00,
- 0x21, 0x11, 0x08, 0x80, 0x00, 0x00,
- 0x03, 0xc0, 0x1e, 0x00, 0x00, 0x00,
- 0xe8, 0x07, 0x40, 0x00, 0x00, 0x00,
- 0x0a, 0x10, 0x50, 0x80, 0x00, 0x00,
- 0x80, 0x1c, 0x00, 0xc0, 0x00, 0x00,
- 0x04, 0x90, 0x24, 0x80, 0x00, 0x00,
- 0x08, 0xa8, 0x45, 0x40, 0x00, 0x00,
- 0x28, 0x29, 0x41, 0x40, 0x00, 0x00,
- 0x84, 0x54, 0x22, 0x80, 0x00, 0x00,
- 0x60, 0x43, 0x02, 0x00, 0x00, 0x00,
- 0x05, 0x48, 0x2a, 0x40, 0x00, 0x00,
- 0x02, 0x98, 0x14, 0xc0, 0x00, 0x00,
- 0x01, 0x30, 0x09, 0x80, 0x00, 0x00,
- 0x48, 0x12, 0x40, 0x80, 0x00, 0x00,
- 0x24, 0x81, 0x24, 0x00, 0x00, 0x00,
- 0x94, 0x04, 0xa0, 0x00, 0x00, 0x00,
- 0x8a, 0x04, 0x50, 0x00, 0x00, 0x00,
- 0x11, 0x80, 0x8c, 0x00, 0x00, 0x00,
- 0x52, 0x22, 0x91, 0x00, 0x00, 0x00,
- 0xf9, 0x13, 0x51, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_3[18] =
- {
- 0x99, 0xb4, 0xcd, 0x80, 0x00, 0x00,
- 0x46, 0xda, 0x36, 0xc0, 0x00, 0x00,
- 0x37, 0x29, 0xb9, 0x40, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_4[24] =
- {
- 0x49, 0xb2, 0x4d, 0x80, 0x00, 0x00,
- 0x26, 0xd1, 0x36, 0x80, 0x00, 0x00,
- 0x85, 0x6c, 0x2b, 0x40, 0x00, 0x00,
- 0x52, 0x5a, 0x92, 0xc0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_5[30] =
- {
- 0x51, 0x32, 0x89, 0x80, 0x00, 0x00,
- 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
- 0x0c, 0x68, 0x63, 0x40, 0x00, 0x00,
- 0xa1, 0xc5, 0x0e, 0x00, 0x00, 0x00,
- 0x22, 0x99, 0x14, 0xc0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_6[36] =
- {
- 0xd1, 0x26, 0x89, 0x00, 0x00, 0x00,
- 0x46, 0xd2, 0x36, 0x80, 0x00, 0x00,
- 0x15, 0x48, 0xaa, 0x40, 0x00, 0x00,
- 0x21, 0x71, 0x0b, 0x80, 0x00, 0x00,
- 0x28, 0xc9, 0x46, 0x40, 0x00, 0x00,
- 0xaa, 0x25, 0x51, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_7[42] =
- {
- 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
- 0x26, 0x51, 0x32, 0x80, 0x00, 0x00,
- 0xb1, 0x45, 0x8a, 0x00, 0x00, 0x00,
- 0x2b, 0x09, 0x58, 0x40, 0x00, 0x00,
- 0x14, 0xc8, 0xa6, 0x40, 0x00, 0x00,
- 0xc8, 0x8e, 0x44, 0x40, 0x00, 0x00,
- 0x84, 0xb4, 0x25, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_8[48] =
- {
- 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
- 0x30, 0x91, 0x84, 0x80, 0x00, 0x00,
- 0x16, 0x08, 0xb0, 0x40, 0x00, 0x00,
- 0x03, 0x30, 0x19, 0x80, 0x00, 0x00,
- 0x44, 0x62, 0x23, 0x00, 0x00, 0x00,
- 0x08, 0x18, 0x40, 0xc0, 0x00, 0x00,
- 0xd8, 0x06, 0xc0, 0x00, 0x00, 0x00,
- 0xa1, 0x45, 0x0a, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask26_9[54] =
- {
- 0x59, 0x22, 0xc9, 0x00, 0x00, 0x00,
- 0x66, 0x43, 0x32, 0x00, 0x00, 0x00,
- 0x14, 0x40, 0xa2, 0x00, 0x00, 0x00,
- 0x21, 0x49, 0x0a, 0x40, 0x00, 0x00,
- 0x02, 0xc8, 0x16, 0x40, 0x00, 0x00,
- 0x94, 0x14, 0xa0, 0x80, 0x00, 0x00,
- 0x80, 0xac, 0x05, 0x40, 0x00, 0x00,
- 0x0a, 0x90, 0x54, 0x80, 0x00, 0x00,
- 0x40, 0x1a, 0x00, 0xc0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_1[6] =
- {
- 0xff, 0xff, 0xff, 0xe0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_10[60] =
- {
- 0xd1, 0x06, 0x06, 0xa0, 0x00, 0x00,
- 0x44, 0x50, 0xea, 0x00, 0x00, 0x00,
- 0x10, 0x9e, 0xa0, 0x40, 0x00, 0x00,
- 0xa0, 0x50, 0x13, 0x00, 0x00, 0x00,
- 0x4a, 0x08, 0x21, 0x40, 0x00, 0x00,
- 0x40, 0x31, 0x04, 0xc0, 0x00, 0x00,
- 0x80, 0x2a, 0x02, 0x20, 0x00, 0x00,
- 0x0c, 0x90, 0x44, 0x20, 0x00, 0x00,
- 0x05, 0x8b, 0x40, 0x00, 0x00, 0x00,
- 0x62, 0x21, 0x18, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_11[66] =
- {
- 0x51, 0x23, 0x16, 0x80, 0x00, 0x00,
- 0x22, 0x11, 0xa9, 0x00, 0x00, 0x00,
- 0x13, 0x40, 0xa0, 0xa0, 0x00, 0x00,
- 0x25, 0x06, 0x28, 0x40, 0x00, 0x00,
- 0x18, 0x19, 0x10, 0x60, 0x00, 0x00,
- 0x0a, 0x24, 0x45, 0xc0, 0x00, 0x00,
- 0x88, 0x8a, 0x12, 0xa0, 0x00, 0x00,
- 0x06, 0x81, 0x45, 0x20, 0x00, 0x00,
- 0xe0, 0x24, 0xa1, 0x00, 0x00, 0x00,
- 0x84, 0x40, 0xd8, 0x20, 0x00, 0x00,
- 0x44, 0x19, 0x16, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_12[72] =
- {
- 0x28, 0x2c, 0x08, 0x20, 0x00, 0x00,
- 0x84, 0x52, 0x03, 0x40, 0x00, 0x00,
- 0x60, 0x44, 0x81, 0x20, 0x00, 0x00,
- 0x05, 0x49, 0x41, 0x40, 0x00, 0x00,
- 0x02, 0x9a, 0x90, 0x80, 0x00, 0x00,
- 0x01, 0x32, 0x0c, 0x40, 0x00, 0x00,
- 0x48, 0x10, 0x49, 0x80, 0x00, 0x00,
- 0x24, 0x82, 0x42, 0x20, 0x00, 0x00,
- 0x94, 0x00, 0x22, 0x20, 0x00, 0x00,
- 0x8a, 0x00, 0x74, 0x00, 0x00, 0x00,
- 0x11, 0x85, 0x2c, 0x80, 0x00, 0x00,
- 0x52, 0x20, 0x90, 0x60, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_13[78] =
- {
- 0x51, 0x23, 0x12, 0xa0, 0x00, 0x00,
- 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
- 0x05, 0x4a, 0x40, 0x20, 0x00, 0x00,
- 0x81, 0x20, 0x05, 0x60, 0x00, 0x00,
- 0x94, 0x01, 0x40, 0x40, 0x00, 0x00,
- 0x30, 0x84, 0x08, 0x40, 0x00, 0x00,
- 0x21, 0x11, 0x18, 0x20, 0x00, 0x00,
- 0x03, 0xc0, 0x34, 0x00, 0x00, 0x00,
- 0xe8, 0x04, 0x00, 0xa0, 0x00, 0x00,
- 0x0a, 0x11, 0x80, 0x80, 0x00, 0x00,
- 0x80, 0x1c, 0x61, 0x00, 0x00, 0x00,
- 0x04, 0x92, 0xa0, 0x00, 0x00, 0x00,
- 0x08, 0xac, 0x06, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_14[84] =
- {
- 0x59, 0x23, 0x12, 0xa0, 0x00, 0x00,
- 0x26, 0x55, 0xc9, 0x00, 0x00, 0x00,
- 0xb1, 0x40, 0xc5, 0xa0, 0x00, 0x00,
- 0x2b, 0x0a, 0xa4, 0xc0, 0x00, 0x00,
- 0x14, 0xc8, 0x33, 0x60, 0x00, 0x00,
- 0xc8, 0x8c, 0x2a, 0xa0, 0x00, 0x00,
- 0x84, 0xb5, 0x54, 0x40, 0x00, 0x00,
- 0xd1, 0x22, 0x52, 0xa0, 0x00, 0x00,
- 0x46, 0xd4, 0xaa, 0x40, 0x00, 0x00,
- 0x15, 0x48, 0xa5, 0xa0, 0x00, 0x00,
- 0x21, 0x72, 0x8d, 0x40, 0x00, 0x00,
- 0x28, 0xc9, 0x13, 0x60, 0x00, 0x00,
- 0xaa, 0x24, 0x44, 0x60, 0x00, 0x00,
- 0x0a, 0xe7, 0x3b, 0x20, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_15[90] =
- {
- 0x59, 0x23, 0x12, 0xa0, 0x00, 0x00,
- 0x26, 0x55, 0xc9, 0x00, 0x00, 0x00,
- 0xb1, 0x40, 0xc5, 0xa0, 0x00, 0x00,
- 0x2b, 0x0a, 0xa4, 0xc0, 0x00, 0x00,
- 0x14, 0xc8, 0x33, 0x60, 0x00, 0x00,
- 0xc8, 0x8c, 0x2a, 0xa0, 0x00, 0x00,
- 0x84, 0xb5, 0x54, 0x40, 0x00, 0x00,
- 0x80, 0xae, 0x00, 0xa0, 0x00, 0x00,
- 0x30, 0x92, 0x0b, 0x00, 0x00, 0x00,
- 0x16, 0x0c, 0x41, 0x80, 0x00, 0x00,
- 0x03, 0x31, 0x05, 0x20, 0x00, 0x00,
- 0x44, 0x60, 0x52, 0x40, 0x00, 0x00,
- 0x08, 0x18, 0x24, 0xc0, 0x00, 0x00,
- 0xd8, 0x04, 0xa2, 0x00, 0x00, 0x00,
- 0xa1, 0x43, 0x90, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_16[96] =
- {
- 0x80, 0xae, 0x00, 0xa0, 0x00, 0x00,
- 0x30, 0x92, 0x0b, 0x00, 0x00, 0x00,
- 0x16, 0x0c, 0x41, 0x80, 0x00, 0x00,
- 0x03, 0x31, 0x05, 0x20, 0x00, 0x00,
- 0x44, 0x60, 0x52, 0x40, 0x00, 0x00,
- 0x08, 0x18, 0x24, 0xc0, 0x00, 0x00,
- 0xd8, 0x04, 0xa2, 0x00, 0x00, 0x00,
- 0xa1, 0x43, 0x90, 0x00, 0x00, 0x00,
- 0x59, 0x23, 0x12, 0xa0, 0x00, 0x00,
- 0x26, 0x55, 0xc9, 0x00, 0x00, 0x00,
- 0xb1, 0x40, 0xc5, 0xa0, 0x00, 0x00,
- 0x2b, 0x0a, 0xa4, 0xc0, 0x00, 0x00,
- 0x14, 0xc8, 0x33, 0x60, 0x00, 0x00,
- 0xc8, 0x8c, 0x2a, 0xa0, 0x00, 0x00,
- 0x84, 0xb5, 0x54, 0x40, 0x00, 0x00,
- 0x01, 0x50, 0xfb, 0xe0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_17[102] =
- {
- 0x80, 0xae, 0x00, 0xa0, 0x00, 0x00,
- 0x30, 0x92, 0x0b, 0x00, 0x00, 0x00,
- 0x16, 0x0c, 0x41, 0x80, 0x00, 0x00,
- 0x03, 0x31, 0x05, 0x20, 0x00, 0x00,
- 0x44, 0x60, 0x52, 0x40, 0x00, 0x00,
- 0x08, 0x18, 0x24, 0xc0, 0x00, 0x00,
- 0xd8, 0x04, 0xa2, 0x00, 0x00, 0x00,
- 0xa1, 0x43, 0x90, 0x00, 0x00, 0x00,
- 0x59, 0x25, 0x12, 0xa0, 0x00, 0x00,
- 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
- 0x14, 0x42, 0x51, 0x20, 0x00, 0x00,
- 0x21, 0x49, 0x05, 0x40, 0x00, 0x00,
- 0x02, 0xc8, 0x8c, 0x20, 0x00, 0x00,
- 0x94, 0x12, 0x48, 0x40, 0x00, 0x00,
- 0x80, 0xac, 0x30, 0x60, 0x00, 0x00,
- 0x0a, 0x91, 0x06, 0xa0, 0x00, 0x00,
- 0x40, 0x1c, 0x42, 0x40, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_18[108] =
- {
- 0x59, 0x25, 0x12, 0xa0, 0x00, 0x00,
- 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
- 0x14, 0x42, 0x51, 0x20, 0x00, 0x00,
- 0x21, 0x49, 0x05, 0x40, 0x00, 0x00,
- 0x02, 0xc8, 0x8c, 0x20, 0x00, 0x00,
- 0x94, 0x12, 0x48, 0x40, 0x00, 0x00,
- 0x80, 0xac, 0x30, 0x60, 0x00, 0x00,
- 0x0a, 0x91, 0x06, 0xa0, 0x00, 0x00,
- 0x40, 0x1c, 0x42, 0x40, 0x00, 0x00,
- 0x80, 0xae, 0x00, 0xa0, 0x00, 0x00,
- 0x30, 0x92, 0x0b, 0x00, 0x00, 0x00,
- 0x16, 0x0c, 0x41, 0x80, 0x00, 0x00,
- 0x03, 0x31, 0x05, 0x20, 0x00, 0x00,
- 0x44, 0x60, 0x52, 0x40, 0x00, 0x00,
- 0x08, 0x18, 0x24, 0xc0, 0x00, 0x00,
- 0xd8, 0x04, 0xa2, 0x00, 0x00, 0x00,
- 0xa1, 0x43, 0x90, 0x00, 0x00, 0x00,
- 0x53, 0xc3, 0x33, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_19[114] =
- {
- 0x59, 0x25, 0x12, 0xa0, 0x00, 0x00,
- 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
- 0x14, 0x42, 0x51, 0x20, 0x00, 0x00,
- 0x21, 0x49, 0x05, 0x40, 0x00, 0x00,
- 0x02, 0xc8, 0x8c, 0x20, 0x00, 0x00,
- 0x94, 0x12, 0x48, 0x40, 0x00, 0x00,
- 0x80, 0xac, 0x30, 0x60, 0x00, 0x00,
- 0x0a, 0x91, 0x06, 0xa0, 0x00, 0x00,
- 0x40, 0x1c, 0x42, 0x40, 0x00, 0x00,
- 0xd1, 0x06, 0x06, 0xa0, 0x00, 0x00,
- 0x44, 0x50, 0xea, 0x00, 0x00, 0x00,
- 0x10, 0x9e, 0xa0, 0x40, 0x00, 0x00,
- 0xa0, 0x50, 0x13, 0x00, 0x00, 0x00,
- 0x4a, 0x08, 0x21, 0x40, 0x00, 0x00,
- 0x40, 0x31, 0x04, 0xc0, 0x00, 0x00,
- 0x80, 0x2a, 0x02, 0x20, 0x00, 0x00,
- 0x0c, 0x90, 0x44, 0x20, 0x00, 0x00,
- 0x05, 0x8b, 0x40, 0x00, 0x00, 0x00,
- 0x62, 0x21, 0x18, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_2[12] =
- {
- 0xec, 0xc7, 0x67, 0x40, 0x00, 0x00,
- 0x1b, 0x39, 0xdc, 0xe0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_20[120] =
- {
- 0xd1, 0x06, 0x06, 0xa0, 0x00, 0x00,
- 0x44, 0x50, 0xea, 0x00, 0x00, 0x00,
- 0x10, 0x9e, 0xa0, 0x40, 0x00, 0x00,
- 0xa0, 0x50, 0x13, 0x00, 0x00, 0x00,
- 0x4a, 0x08, 0x21, 0x40, 0x00, 0x00,
- 0x40, 0x31, 0x04, 0xc0, 0x00, 0x00,
- 0x80, 0x2a, 0x02, 0x20, 0x00, 0x00,
- 0x0c, 0x90, 0x44, 0x20, 0x00, 0x00,
- 0x05, 0x8b, 0x40, 0x00, 0x00, 0x00,
- 0x62, 0x21, 0x18, 0x80, 0x00, 0x00,
- 0x59, 0x25, 0x12, 0xa0, 0x00, 0x00,
- 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
- 0x14, 0x42, 0x51, 0x20, 0x00, 0x00,
- 0x21, 0x49, 0x05, 0x40, 0x00, 0x00,
- 0x02, 0xc8, 0x8c, 0x20, 0x00, 0x00,
- 0x94, 0x12, 0x48, 0x40, 0x00, 0x00,
- 0x80, 0xac, 0x30, 0x60, 0x00, 0x00,
- 0x0a, 0x91, 0x06, 0xa0, 0x00, 0x00,
- 0x40, 0x1c, 0x42, 0x40, 0x00, 0x00,
- 0xcb, 0xff, 0x6f, 0xc0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_21[126] =
- {
- 0xd1, 0x06, 0x06, 0xa0, 0x00, 0x00,
- 0x44, 0x50, 0xea, 0x00, 0x00, 0x00,
- 0x10, 0x9e, 0xa0, 0x40, 0x00, 0x00,
- 0xa0, 0x50, 0x13, 0x00, 0x00, 0x00,
- 0x4a, 0x08, 0x21, 0x40, 0x00, 0x00,
- 0x40, 0x31, 0x04, 0xc0, 0x00, 0x00,
- 0x80, 0x2a, 0x02, 0x20, 0x00, 0x00,
- 0x0c, 0x90, 0x44, 0x20, 0x00, 0x00,
- 0x05, 0x8b, 0x40, 0x00, 0x00, 0x00,
- 0x62, 0x21, 0x18, 0x80, 0x00, 0x00,
- 0x51, 0x23, 0x16, 0x80, 0x00, 0x00,
- 0x22, 0x11, 0xa9, 0x00, 0x00, 0x00,
- 0x13, 0x40, 0xa0, 0xa0, 0x00, 0x00,
- 0x25, 0x06, 0x28, 0x40, 0x00, 0x00,
- 0x18, 0x19, 0x10, 0x60, 0x00, 0x00,
- 0x0a, 0x24, 0x45, 0xc0, 0x00, 0x00,
- 0x88, 0x8a, 0x12, 0xa0, 0x00, 0x00,
- 0x06, 0x81, 0x45, 0x20, 0x00, 0x00,
- 0xe0, 0x24, 0xa1, 0x00, 0x00, 0x00,
- 0x84, 0x40, 0xd8, 0x20, 0x00, 0x00,
- 0x44, 0x19, 0x16, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_22[132] =
- {
- 0x51, 0x23, 0x16, 0x80, 0x00, 0x00,
- 0x22, 0x11, 0xa9, 0x00, 0x00, 0x00,
- 0x13, 0x40, 0xa0, 0xa0, 0x00, 0x00,
- 0x25, 0x06, 0x28, 0x40, 0x00, 0x00,
- 0x18, 0x19, 0x10, 0x60, 0x00, 0x00,
- 0x0a, 0x24, 0x45, 0xc0, 0x00, 0x00,
- 0x88, 0x8a, 0x12, 0xa0, 0x00, 0x00,
- 0x06, 0x81, 0x45, 0x20, 0x00, 0x00,
- 0xe0, 0x24, 0xa1, 0x00, 0x00, 0x00,
- 0x84, 0x40, 0xd8, 0x20, 0x00, 0x00,
- 0x44, 0x19, 0x16, 0x00, 0x00, 0x00,
- 0xd1, 0x06, 0x06, 0xa0, 0x00, 0x00,
- 0x44, 0x50, 0xea, 0x00, 0x00, 0x00,
- 0x10, 0x9e, 0xa0, 0x40, 0x00, 0x00,
- 0xa0, 0x50, 0x13, 0x00, 0x00, 0x00,
- 0x4a, 0x08, 0x21, 0x40, 0x00, 0x00,
- 0x40, 0x31, 0x04, 0xc0, 0x00, 0x00,
- 0x80, 0x2a, 0x02, 0x20, 0x00, 0x00,
- 0x0c, 0x90, 0x44, 0x20, 0x00, 0x00,
- 0x05, 0x8b, 0x40, 0x00, 0x00, 0x00,
- 0x62, 0x21, 0x18, 0x80, 0x00, 0x00,
- 0xf5, 0x2d, 0x52, 0x40, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_23[138] =
- {
- 0x51, 0x23, 0x16, 0x80, 0x00, 0x00,
- 0x22, 0x11, 0xa9, 0x00, 0x00, 0x00,
- 0x13, 0x40, 0xa0, 0xa0, 0x00, 0x00,
- 0x25, 0x06, 0x28, 0x40, 0x00, 0x00,
- 0x18, 0x19, 0x10, 0x60, 0x00, 0x00,
- 0x0a, 0x24, 0x45, 0xc0, 0x00, 0x00,
- 0x88, 0x8a, 0x12, 0xa0, 0x00, 0x00,
- 0x06, 0x81, 0x45, 0x20, 0x00, 0x00,
- 0xe0, 0x24, 0xa1, 0x00, 0x00, 0x00,
- 0x84, 0x40, 0xd8, 0x20, 0x00, 0x00,
- 0x44, 0x19, 0x16, 0x00, 0x00, 0x00,
- 0x28, 0x2c, 0x08, 0x20, 0x00, 0x00,
- 0x84, 0x52, 0x03, 0x40, 0x00, 0x00,
- 0x60, 0x44, 0x81, 0x20, 0x00, 0x00,
- 0x05, 0x49, 0x41, 0x40, 0x00, 0x00,
- 0x02, 0x9a, 0x90, 0x80, 0x00, 0x00,
- 0x01, 0x32, 0x0c, 0x40, 0x00, 0x00,
- 0x48, 0x10, 0x49, 0x80, 0x00, 0x00,
- 0x24, 0x82, 0x42, 0x20, 0x00, 0x00,
- 0x94, 0x00, 0x22, 0x20, 0x00, 0x00,
- 0x8a, 0x00, 0x74, 0x00, 0x00, 0x00,
- 0x11, 0x85, 0x2c, 0x80, 0x00, 0x00,
- 0x52, 0x20, 0x90, 0x60, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_24[144] =
- {
- 0x28, 0x2c, 0x08, 0x20, 0x00, 0x00,
- 0x84, 0x52, 0x03, 0x40, 0x00, 0x00,
- 0x60, 0x44, 0x81, 0x20, 0x00, 0x00,
- 0x05, 0x49, 0x41, 0x40, 0x00, 0x00,
- 0x02, 0x9a, 0x90, 0x80, 0x00, 0x00,
- 0x01, 0x32, 0x0c, 0x40, 0x00, 0x00,
- 0x48, 0x10, 0x49, 0x80, 0x00, 0x00,
- 0x24, 0x82, 0x42, 0x20, 0x00, 0x00,
- 0x94, 0x00, 0x22, 0x20, 0x00, 0x00,
- 0x8a, 0x00, 0x74, 0x00, 0x00, 0x00,
- 0x11, 0x85, 0x2c, 0x80, 0x00, 0x00,
- 0x52, 0x20, 0x90, 0x60, 0x00, 0x00,
- 0x51, 0x23, 0x16, 0x80, 0x00, 0x00,
- 0x22, 0x11, 0xa9, 0x00, 0x00, 0x00,
- 0x13, 0x40, 0xa0, 0xa0, 0x00, 0x00,
- 0x25, 0x06, 0x28, 0x40, 0x00, 0x00,
- 0x18, 0x19, 0x10, 0x60, 0x00, 0x00,
- 0x0a, 0x24, 0x45, 0xc0, 0x00, 0x00,
- 0x88, 0x8a, 0x12, 0xa0, 0x00, 0x00,
- 0x06, 0x81, 0x45, 0x20, 0x00, 0x00,
- 0xe0, 0x24, 0xa1, 0x00, 0x00, 0x00,
- 0x84, 0x40, 0xd8, 0x20, 0x00, 0x00,
- 0x44, 0x19, 0x16, 0x00, 0x00, 0x00,
- 0xa2, 0x85, 0xdb, 0xa0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_25[150] =
- {
- 0x28, 0x2c, 0x08, 0x20, 0x00, 0x00,
- 0x84, 0x52, 0x03, 0x40, 0x00, 0x00,
- 0x60, 0x44, 0x81, 0x20, 0x00, 0x00,
- 0x05, 0x49, 0x41, 0x40, 0x00, 0x00,
- 0x02, 0x9a, 0x90, 0x80, 0x00, 0x00,
- 0x01, 0x32, 0x0c, 0x40, 0x00, 0x00,
- 0x48, 0x10, 0x49, 0x80, 0x00, 0x00,
- 0x24, 0x82, 0x42, 0x20, 0x00, 0x00,
- 0x94, 0x00, 0x22, 0x20, 0x00, 0x00,
- 0x8a, 0x00, 0x74, 0x00, 0x00, 0x00,
- 0x11, 0x85, 0x2c, 0x80, 0x00, 0x00,
- 0x52, 0x20, 0x90, 0x60, 0x00, 0x00,
- 0x51, 0x23, 0x12, 0xa0, 0x00, 0x00,
- 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
- 0x05, 0x4a, 0x40, 0x20, 0x00, 0x00,
- 0x81, 0x20, 0x05, 0x60, 0x00, 0x00,
- 0x94, 0x01, 0x40, 0x40, 0x00, 0x00,
- 0x30, 0x84, 0x08, 0x40, 0x00, 0x00,
- 0x21, 0x11, 0x18, 0x20, 0x00, 0x00,
- 0x03, 0xc0, 0x34, 0x00, 0x00, 0x00,
- 0xe8, 0x04, 0x00, 0xa0, 0x00, 0x00,
- 0x0a, 0x11, 0x80, 0x80, 0x00, 0x00,
- 0x80, 0x1c, 0x61, 0x00, 0x00, 0x00,
- 0x04, 0x92, 0xa0, 0x00, 0x00, 0x00,
- 0x08, 0xac, 0x06, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_26[156] =
- {
- 0x51, 0x23, 0x12, 0xa0, 0x00, 0x00,
- 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
- 0x05, 0x4a, 0x40, 0x20, 0x00, 0x00,
- 0x81, 0x20, 0x05, 0x60, 0x00, 0x00,
- 0x94, 0x01, 0x40, 0x40, 0x00, 0x00,
- 0x30, 0x84, 0x08, 0x40, 0x00, 0x00,
- 0x21, 0x11, 0x18, 0x20, 0x00, 0x00,
- 0x03, 0xc0, 0x34, 0x00, 0x00, 0x00,
- 0xe8, 0x04, 0x00, 0xa0, 0x00, 0x00,
- 0x0a, 0x11, 0x80, 0x80, 0x00, 0x00,
- 0x80, 0x1c, 0x61, 0x00, 0x00, 0x00,
- 0x04, 0x92, 0xa0, 0x00, 0x00, 0x00,
- 0x08, 0xac, 0x06, 0x00, 0x00, 0x00,
- 0x28, 0x2c, 0x08, 0x20, 0x00, 0x00,
- 0x84, 0x52, 0x03, 0x40, 0x00, 0x00,
- 0x60, 0x44, 0x81, 0x20, 0x00, 0x00,
- 0x05, 0x49, 0x41, 0x40, 0x00, 0x00,
- 0x02, 0x9a, 0x90, 0x80, 0x00, 0x00,
- 0x01, 0x32, 0x0c, 0x40, 0x00, 0x00,
- 0x48, 0x10, 0x49, 0x80, 0x00, 0x00,
- 0x24, 0x82, 0x42, 0x20, 0x00, 0x00,
- 0x94, 0x00, 0x22, 0x20, 0x00, 0x00,
- 0x8a, 0x00, 0x74, 0x00, 0x00, 0x00,
- 0x11, 0x85, 0x2c, 0x80, 0x00, 0x00,
- 0x52, 0x20, 0x90, 0x60, 0x00, 0x00,
- 0xcd, 0x41, 0xa2, 0x40, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_27[162] =
- {
- 0x51, 0x23, 0x12, 0xa0, 0x00, 0x00,
- 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
- 0x05, 0x4a, 0x40, 0x20, 0x00, 0x00,
- 0x81, 0x20, 0x05, 0x60, 0x00, 0x00,
- 0x94, 0x01, 0x40, 0x40, 0x00, 0x00,
- 0x30, 0x84, 0x08, 0x40, 0x00, 0x00,
- 0x21, 0x11, 0x18, 0x20, 0x00, 0x00,
- 0x03, 0xc0, 0x34, 0x00, 0x00, 0x00,
- 0xe8, 0x04, 0x00, 0xa0, 0x00, 0x00,
- 0x0a, 0x11, 0x80, 0x80, 0x00, 0x00,
- 0x80, 0x1c, 0x61, 0x00, 0x00, 0x00,
- 0x04, 0x92, 0xa0, 0x00, 0x00, 0x00,
- 0x08, 0xac, 0x06, 0x00, 0x00, 0x00,
- 0x51, 0x22, 0x02, 0xa0, 0x00, 0x00,
- 0x66, 0x40, 0xaa, 0x00, 0x00, 0x00,
- 0x05, 0x4e, 0x00, 0x20, 0x00, 0x00,
- 0x81, 0x21, 0x40, 0x80, 0x00, 0x00,
- 0x94, 0x00, 0x28, 0x60, 0x00, 0x00,
- 0x30, 0x83, 0x24, 0x00, 0x00, 0x00,
- 0x21, 0x14, 0x0c, 0x00, 0x00, 0x00,
- 0x03, 0xc0, 0x84, 0xc0, 0x00, 0x00,
- 0xe8, 0x04, 0x21, 0x00, 0x00, 0x00,
- 0x0a, 0x10, 0x91, 0x80, 0x00, 0x00,
- 0x80, 0x1b, 0x10, 0x00, 0x00, 0x00,
- 0x04, 0x91, 0x43, 0x00, 0x00, 0x00,
- 0x08, 0xa8, 0x70, 0x40, 0x00, 0x00,
- 0x9c, 0xc0, 0x84, 0x20, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_3[18] =
- {
- 0x99, 0xb5, 0x66, 0xc0, 0x00, 0x00,
- 0x46, 0xda, 0xab, 0x60, 0x00, 0x00,
- 0x37, 0x29, 0x3d, 0xa0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_4[24] =
- {
- 0x49, 0xb1, 0x66, 0xc0, 0x00, 0x00,
- 0x26, 0xd4, 0x9b, 0x40, 0x00, 0x00,
- 0x85, 0x68, 0xd5, 0xa0, 0x00, 0x00,
- 0x52, 0x5a, 0x39, 0x60, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_5[30] =
- {
- 0x51, 0x33, 0x26, 0xc0, 0x00, 0x00,
- 0x66, 0x45, 0x2b, 0x40, 0x00, 0x00,
- 0x0c, 0x6a, 0x95, 0xa0, 0x00, 0x00,
- 0xa1, 0xc0, 0xed, 0x40, 0x00, 0x00,
- 0x22, 0x9c, 0xe2, 0xa0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_6[36] =
- {
- 0xd1, 0x22, 0x52, 0xa0, 0x00, 0x00,
- 0x46, 0xd4, 0xaa, 0x40, 0x00, 0x00,
- 0x15, 0x48, 0xa5, 0xa0, 0x00, 0x00,
- 0x21, 0x72, 0x8d, 0x40, 0x00, 0x00,
- 0x28, 0xc9, 0x13, 0x60, 0x00, 0x00,
- 0xaa, 0x24, 0x44, 0x60, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_7[42] =
- {
- 0x59, 0x23, 0x12, 0xa0, 0x00, 0x00,
- 0x26, 0x55, 0xc9, 0x00, 0x00, 0x00,
- 0xb1, 0x40, 0xc5, 0xa0, 0x00, 0x00,
- 0x2b, 0x0a, 0xa4, 0xc0, 0x00, 0x00,
- 0x14, 0xc8, 0x33, 0x60, 0x00, 0x00,
- 0xc8, 0x8c, 0x2a, 0xa0, 0x00, 0x00,
- 0x84, 0xb5, 0x54, 0x40, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_8[48] =
- {
- 0x80, 0xae, 0x00, 0xa0, 0x00, 0x00,
- 0x30, 0x92, 0x0b, 0x00, 0x00, 0x00,
- 0x16, 0x0c, 0x41, 0x80, 0x00, 0x00,
- 0x03, 0x31, 0x05, 0x20, 0x00, 0x00,
- 0x44, 0x60, 0x52, 0x40, 0x00, 0x00,
- 0x08, 0x18, 0x24, 0xc0, 0x00, 0x00,
- 0xd8, 0x04, 0xa2, 0x00, 0x00, 0x00,
- 0xa1, 0x43, 0x90, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask27_9[54] =
- {
- 0x59, 0x25, 0x12, 0xa0, 0x00, 0x00,
- 0x66, 0x41, 0xa3, 0x00, 0x00, 0x00,
- 0x14, 0x42, 0x51, 0x20, 0x00, 0x00,
- 0x21, 0x49, 0x05, 0x40, 0x00, 0x00,
- 0x02, 0xc8, 0x8c, 0x20, 0x00, 0x00,
- 0x94, 0x12, 0x48, 0x40, 0x00, 0x00,
- 0x80, 0xac, 0x30, 0x60, 0x00, 0x00,
- 0x0a, 0x91, 0x06, 0xa0, 0x00, 0x00,
- 0x40, 0x1c, 0x42, 0x40, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_1[6] =
- {
- 0xff, 0xff, 0xff, 0xf0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_10[60] =
- {
- 0xc0, 0xd7, 0x03, 0x50, 0x00, 0x00,
- 0x1d, 0x40, 0x75, 0x00, 0x00, 0x00,
- 0xd4, 0x0b, 0x50, 0x20, 0x00, 0x00,
- 0x02, 0x60, 0x09, 0x80, 0x00, 0x00,
- 0x04, 0x28, 0x10, 0xa0, 0x00, 0x00,
- 0x20, 0x98, 0x82, 0x60, 0x00, 0x00,
- 0x40, 0x45, 0x01, 0x10, 0x00, 0x00,
- 0x08, 0x84, 0x22, 0x10, 0x00, 0x00,
- 0x68, 0x01, 0xa0, 0x00, 0x00, 0x00,
- 0x23, 0x10, 0x8c, 0x40, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_11[66] =
- {
- 0x62, 0xd1, 0x8b, 0x40, 0x00, 0x00,
- 0x35, 0x20, 0xd4, 0x80, 0x00, 0x00,
- 0x14, 0x14, 0x50, 0x50, 0x00, 0x00,
- 0xc5, 0x0b, 0x14, 0x20, 0x00, 0x00,
- 0x22, 0x0c, 0x88, 0x30, 0x00, 0x00,
- 0x88, 0xba, 0x22, 0xe0, 0x00, 0x00,
- 0x42, 0x55, 0x09, 0x50, 0x00, 0x00,
- 0x28, 0xa4, 0xa2, 0x90, 0x00, 0x00,
- 0x94, 0x22, 0x50, 0x80, 0x00, 0x00,
- 0x1b, 0x04, 0x6c, 0x10, 0x00, 0x00,
- 0x22, 0xc0, 0x8b, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_12[72] =
- {
- 0x81, 0x06, 0x04, 0x10, 0x00, 0x00,
- 0x40, 0x69, 0x01, 0xa0, 0x00, 0x00,
- 0x90, 0x26, 0x40, 0x90, 0x00, 0x00,
- 0x28, 0x28, 0xa0, 0xa0, 0x00, 0x00,
- 0x52, 0x11, 0x48, 0x40, 0x00, 0x00,
- 0x41, 0x89, 0x06, 0x20, 0x00, 0x00,
- 0x09, 0x30, 0x24, 0xc0, 0x00, 0x00,
- 0x48, 0x45, 0x21, 0x10, 0x00, 0x00,
- 0x04, 0x44, 0x11, 0x10, 0x00, 0x00,
- 0x0e, 0x80, 0x3a, 0x00, 0x00, 0x00,
- 0xa5, 0x92, 0x96, 0x40, 0x00, 0x00,
- 0x12, 0x0c, 0x48, 0x30, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_13[78] =
- {
- 0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
- 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
- 0x48, 0x05, 0x20, 0x10, 0x00, 0x00,
- 0x00, 0xac, 0x02, 0xb0, 0x00, 0x00,
- 0x28, 0x08, 0xa0, 0x20, 0x00, 0x00,
- 0x81, 0x0a, 0x04, 0x20, 0x00, 0x00,
- 0x23, 0x04, 0x8c, 0x10, 0x00, 0x00,
- 0x06, 0x80, 0x1a, 0x00, 0x00, 0x00,
- 0x80, 0x16, 0x00, 0x50, 0x00, 0x00,
- 0x30, 0x10, 0xc0, 0x40, 0x00, 0x00,
- 0x8c, 0x22, 0x30, 0x80, 0x00, 0x00,
- 0x54, 0x01, 0x50, 0x00, 0x00, 0x00,
- 0x80, 0xc2, 0x03, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_14[84] =
- {
- 0x40, 0x55, 0x01, 0x50, 0x00, 0x00,
- 0x15, 0x40, 0x55, 0x00, 0x00, 0x00,
- 0xc0, 0x07, 0x00, 0x10, 0x00, 0x00,
- 0x28, 0x10, 0xa0, 0x40, 0x00, 0x00,
- 0x05, 0x0c, 0x14, 0x30, 0x00, 0x00,
- 0x64, 0x81, 0x92, 0x00, 0x00, 0x00,
- 0x81, 0x82, 0x06, 0x00, 0x00, 0x00,
- 0x10, 0x98, 0x42, 0x60, 0x00, 0x00,
- 0x84, 0x22, 0x10, 0x80, 0x00, 0x00,
- 0x12, 0x30, 0x48, 0xc0, 0x00, 0x00,
- 0x62, 0x01, 0x88, 0x00, 0x00, 0x00,
- 0x28, 0x60, 0xa1, 0x80, 0x00, 0x00,
- 0x0e, 0x08, 0x38, 0x20, 0x00, 0x00,
- 0x10, 0x84, 0x42, 0x10, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_15[90] =
- {
- 0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
- 0xb9, 0x22, 0xe4, 0x80, 0x00, 0x00,
- 0x18, 0xb4, 0x62, 0xd0, 0x00, 0x00,
- 0x54, 0x99, 0x52, 0x60, 0x00, 0x00,
- 0x06, 0x6c, 0x19, 0xb0, 0x00, 0x00,
- 0x85, 0x56, 0x15, 0x50, 0x00, 0x00,
- 0xaa, 0x8a, 0xaa, 0x20, 0x00, 0x00,
- 0xc0, 0x17, 0x00, 0x50, 0x00, 0x00,
- 0x41, 0x61, 0x05, 0x80, 0x00, 0x00,
- 0x88, 0x32, 0x20, 0xc0, 0x00, 0x00,
- 0x20, 0xa4, 0x82, 0x90, 0x00, 0x00,
- 0x0a, 0x48, 0x29, 0x20, 0x00, 0x00,
- 0x04, 0x98, 0x12, 0x60, 0x00, 0x00,
- 0x94, 0x42, 0x51, 0x00, 0x00, 0x00,
- 0x72, 0x01, 0xc8, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_16[96] =
- {
- 0xc0, 0x17, 0x00, 0x50, 0x00, 0x00,
- 0x41, 0x61, 0x05, 0x80, 0x00, 0x00,
- 0x88, 0x32, 0x20, 0xc0, 0x00, 0x00,
- 0x20, 0xa4, 0x82, 0x90, 0x00, 0x00,
- 0x0a, 0x48, 0x29, 0x20, 0x00, 0x00,
- 0x04, 0x98, 0x12, 0x60, 0x00, 0x00,
- 0x94, 0x42, 0x51, 0x00, 0x00, 0x00,
- 0x72, 0x01, 0xc8, 0x00, 0x00, 0x00,
- 0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
- 0xb9, 0x22, 0xe4, 0x80, 0x00, 0x00,
- 0x18, 0xb4, 0x62, 0xd0, 0x00, 0x00,
- 0x54, 0x99, 0x52, 0x60, 0x00, 0x00,
- 0x06, 0x6c, 0x19, 0xb0, 0x00, 0x00,
- 0x85, 0x56, 0x15, 0x50, 0x00, 0x00,
- 0xaa, 0x8a, 0xaa, 0x20, 0x00, 0x00,
- 0xed, 0x76, 0x36, 0x50, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_17[102] =
- {
- 0xc0, 0x17, 0x00, 0x50, 0x00, 0x00,
- 0x41, 0x61, 0x05, 0x80, 0x00, 0x00,
- 0x88, 0x32, 0x20, 0xc0, 0x00, 0x00,
- 0x20, 0xa4, 0x82, 0x90, 0x00, 0x00,
- 0x0a, 0x48, 0x29, 0x20, 0x00, 0x00,
- 0x04, 0x98, 0x12, 0x60, 0x00, 0x00,
- 0x94, 0x42, 0x51, 0x00, 0x00, 0x00,
- 0x72, 0x01, 0xc8, 0x00, 0x00, 0x00,
- 0xa2, 0x56, 0x89, 0x50, 0x00, 0x00,
- 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
- 0x4a, 0x25, 0x28, 0x90, 0x00, 0x00,
- 0x20, 0xa8, 0x82, 0xa0, 0x00, 0x00,
- 0x11, 0x84, 0x46, 0x10, 0x00, 0x00,
- 0x49, 0x09, 0x24, 0x20, 0x00, 0x00,
- 0x86, 0x0e, 0x18, 0x30, 0x00, 0x00,
- 0x20, 0xd4, 0x83, 0x50, 0x00, 0x00,
- 0x88, 0x4a, 0x21, 0x20, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_18[108] =
- {
- 0xa2, 0x56, 0x89, 0x50, 0x00, 0x00,
- 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
- 0x4a, 0x25, 0x28, 0x90, 0x00, 0x00,
- 0x20, 0xa8, 0x82, 0xa0, 0x00, 0x00,
- 0x11, 0x84, 0x46, 0x10, 0x00, 0x00,
- 0x49, 0x09, 0x24, 0x20, 0x00, 0x00,
- 0x86, 0x0e, 0x18, 0x30, 0x00, 0x00,
- 0x20, 0xd4, 0x83, 0x50, 0x00, 0x00,
- 0x88, 0x4a, 0x21, 0x20, 0x00, 0x00,
- 0xc0, 0x17, 0x00, 0x50, 0x00, 0x00,
- 0x41, 0x61, 0x05, 0x80, 0x00, 0x00,
- 0x88, 0x32, 0x20, 0xc0, 0x00, 0x00,
- 0x20, 0xa4, 0x82, 0x90, 0x00, 0x00,
- 0x0a, 0x48, 0x29, 0x20, 0x00, 0x00,
- 0x04, 0x98, 0x12, 0x60, 0x00, 0x00,
- 0x94, 0x42, 0x51, 0x00, 0x00, 0x00,
- 0x72, 0x01, 0xc8, 0x00, 0x00, 0x00,
- 0x6e, 0x9f, 0x98, 0x10, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_19[114] =
- {
- 0xa2, 0x56, 0x89, 0x50, 0x00, 0x00,
- 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
- 0x4a, 0x25, 0x28, 0x90, 0x00, 0x00,
- 0x20, 0xa8, 0x82, 0xa0, 0x00, 0x00,
- 0x11, 0x84, 0x46, 0x10, 0x00, 0x00,
- 0x49, 0x09, 0x24, 0x20, 0x00, 0x00,
- 0x86, 0x0e, 0x18, 0x30, 0x00, 0x00,
- 0x20, 0xd4, 0x83, 0x50, 0x00, 0x00,
- 0x88, 0x4a, 0x21, 0x20, 0x00, 0x00,
- 0xc0, 0xd7, 0x03, 0x50, 0x00, 0x00,
- 0x1d, 0x40, 0x75, 0x00, 0x00, 0x00,
- 0xd4, 0x0b, 0x50, 0x20, 0x00, 0x00,
- 0x02, 0x60, 0x09, 0x80, 0x00, 0x00,
- 0x04, 0x28, 0x10, 0xa0, 0x00, 0x00,
- 0x20, 0x98, 0x82, 0x60, 0x00, 0x00,
- 0x40, 0x45, 0x01, 0x10, 0x00, 0x00,
- 0x08, 0x84, 0x22, 0x10, 0x00, 0x00,
- 0x68, 0x01, 0xa0, 0x00, 0x00, 0x00,
- 0x23, 0x10, 0x8c, 0x40, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_2[12] =
- {
- 0xec, 0xeb, 0xb3, 0xa0, 0x00, 0x00,
- 0x3b, 0x9c, 0xee, 0x70, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_20[120] =
- {
- 0xc0, 0xd7, 0x03, 0x50, 0x00, 0x00,
- 0x1d, 0x40, 0x75, 0x00, 0x00, 0x00,
- 0xd4, 0x0b, 0x50, 0x20, 0x00, 0x00,
- 0x02, 0x60, 0x09, 0x80, 0x00, 0x00,
- 0x04, 0x28, 0x10, 0xa0, 0x00, 0x00,
- 0x20, 0x98, 0x82, 0x60, 0x00, 0x00,
- 0x40, 0x45, 0x01, 0x10, 0x00, 0x00,
- 0x08, 0x84, 0x22, 0x10, 0x00, 0x00,
- 0x68, 0x01, 0xa0, 0x00, 0x00, 0x00,
- 0x23, 0x10, 0x8c, 0x40, 0x00, 0x00,
- 0xa2, 0x56, 0x89, 0x50, 0x00, 0x00,
- 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
- 0x4a, 0x25, 0x28, 0x90, 0x00, 0x00,
- 0x20, 0xa8, 0x82, 0xa0, 0x00, 0x00,
- 0x11, 0x84, 0x46, 0x10, 0x00, 0x00,
- 0x49, 0x09, 0x24, 0x20, 0x00, 0x00,
- 0x86, 0x0e, 0x18, 0x30, 0x00, 0x00,
- 0x20, 0xd4, 0x83, 0x50, 0x00, 0x00,
- 0x88, 0x4a, 0x21, 0x20, 0x00, 0x00,
- 0xea, 0x1b, 0x3a, 0x10, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_21[126] =
- {
- 0xc0, 0xd7, 0x03, 0x50, 0x00, 0x00,
- 0x1d, 0x40, 0x75, 0x00, 0x00, 0x00,
- 0xd4, 0x0b, 0x50, 0x20, 0x00, 0x00,
- 0x02, 0x60, 0x09, 0x80, 0x00, 0x00,
- 0x04, 0x28, 0x10, 0xa0, 0x00, 0x00,
- 0x20, 0x98, 0x82, 0x60, 0x00, 0x00,
- 0x40, 0x45, 0x01, 0x10, 0x00, 0x00,
- 0x08, 0x84, 0x22, 0x10, 0x00, 0x00,
- 0x68, 0x01, 0xa0, 0x00, 0x00, 0x00,
- 0x23, 0x10, 0x8c, 0x40, 0x00, 0x00,
- 0x62, 0xd1, 0x8b, 0x40, 0x00, 0x00,
- 0x35, 0x20, 0xd4, 0x80, 0x00, 0x00,
- 0x14, 0x14, 0x50, 0x50, 0x00, 0x00,
- 0xc5, 0x0b, 0x14, 0x20, 0x00, 0x00,
- 0x22, 0x0c, 0x88, 0x30, 0x00, 0x00,
- 0x88, 0xba, 0x22, 0xe0, 0x00, 0x00,
- 0x42, 0x55, 0x09, 0x50, 0x00, 0x00,
- 0x28, 0xa4, 0xa2, 0x90, 0x00, 0x00,
- 0x94, 0x22, 0x50, 0x80, 0x00, 0x00,
- 0x1b, 0x04, 0x6c, 0x10, 0x00, 0x00,
- 0x22, 0xc0, 0x8b, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_22[132] =
- {
- 0x62, 0xd1, 0x8b, 0x40, 0x00, 0x00,
- 0x35, 0x20, 0xd4, 0x80, 0x00, 0x00,
- 0x14, 0x14, 0x50, 0x50, 0x00, 0x00,
- 0xc5, 0x0b, 0x14, 0x20, 0x00, 0x00,
- 0x22, 0x0c, 0x88, 0x30, 0x00, 0x00,
- 0x88, 0xba, 0x22, 0xe0, 0x00, 0x00,
- 0x42, 0x55, 0x09, 0x50, 0x00, 0x00,
- 0x28, 0xa4, 0xa2, 0x90, 0x00, 0x00,
- 0x94, 0x22, 0x50, 0x80, 0x00, 0x00,
- 0x1b, 0x04, 0x6c, 0x10, 0x00, 0x00,
- 0x22, 0xc0, 0x8b, 0x00, 0x00, 0x00,
- 0xc0, 0xd7, 0x03, 0x50, 0x00, 0x00,
- 0x1d, 0x40, 0x75, 0x00, 0x00, 0x00,
- 0xd4, 0x0b, 0x50, 0x20, 0x00, 0x00,
- 0x02, 0x60, 0x09, 0x80, 0x00, 0x00,
- 0x04, 0x28, 0x10, 0xa0, 0x00, 0x00,
- 0x20, 0x98, 0x82, 0x60, 0x00, 0x00,
- 0x40, 0x45, 0x01, 0x10, 0x00, 0x00,
- 0x08, 0x84, 0x22, 0x10, 0x00, 0x00,
- 0x68, 0x01, 0xa0, 0x00, 0x00, 0x00,
- 0x23, 0x10, 0x8c, 0x40, 0x00, 0x00,
- 0x45, 0x05, 0x10, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_23[138] =
- {
- 0x62, 0xd1, 0x8b, 0x40, 0x00, 0x00,
- 0x35, 0x20, 0xd4, 0x80, 0x00, 0x00,
- 0x14, 0x14, 0x50, 0x50, 0x00, 0x00,
- 0xc5, 0x0b, 0x14, 0x20, 0x00, 0x00,
- 0x22, 0x0c, 0x88, 0x30, 0x00, 0x00,
- 0x88, 0xba, 0x22, 0xe0, 0x00, 0x00,
- 0x42, 0x55, 0x09, 0x50, 0x00, 0x00,
- 0x28, 0xa4, 0xa2, 0x90, 0x00, 0x00,
- 0x94, 0x22, 0x50, 0x80, 0x00, 0x00,
- 0x1b, 0x04, 0x6c, 0x10, 0x00, 0x00,
- 0x22, 0xc0, 0x8b, 0x00, 0x00, 0x00,
- 0x81, 0x06, 0x04, 0x10, 0x00, 0x00,
- 0x40, 0x69, 0x01, 0xa0, 0x00, 0x00,
- 0x90, 0x26, 0x40, 0x90, 0x00, 0x00,
- 0x28, 0x28, 0xa0, 0xa0, 0x00, 0x00,
- 0x52, 0x11, 0x48, 0x40, 0x00, 0x00,
- 0x41, 0x89, 0x06, 0x20, 0x00, 0x00,
- 0x09, 0x30, 0x24, 0xc0, 0x00, 0x00,
- 0x48, 0x45, 0x21, 0x10, 0x00, 0x00,
- 0x04, 0x44, 0x11, 0x10, 0x00, 0x00,
- 0x0e, 0x80, 0x3a, 0x00, 0x00, 0x00,
- 0xa5, 0x92, 0x96, 0x40, 0x00, 0x00,
- 0x12, 0x0c, 0x48, 0x30, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_24[144] =
- {
- 0x81, 0x06, 0x04, 0x10, 0x00, 0x00,
- 0x40, 0x69, 0x01, 0xa0, 0x00, 0x00,
- 0x90, 0x26, 0x40, 0x90, 0x00, 0x00,
- 0x28, 0x28, 0xa0, 0xa0, 0x00, 0x00,
- 0x52, 0x11, 0x48, 0x40, 0x00, 0x00,
- 0x41, 0x89, 0x06, 0x20, 0x00, 0x00,
- 0x09, 0x30, 0x24, 0xc0, 0x00, 0x00,
- 0x48, 0x45, 0x21, 0x10, 0x00, 0x00,
- 0x04, 0x44, 0x11, 0x10, 0x00, 0x00,
- 0x0e, 0x80, 0x3a, 0x00, 0x00, 0x00,
- 0xa5, 0x92, 0x96, 0x40, 0x00, 0x00,
- 0x12, 0x0c, 0x48, 0x30, 0x00, 0x00,
- 0x62, 0xd1, 0x8b, 0x40, 0x00, 0x00,
- 0x35, 0x20, 0xd4, 0x80, 0x00, 0x00,
- 0x14, 0x14, 0x50, 0x50, 0x00, 0x00,
- 0xc5, 0x0b, 0x14, 0x20, 0x00, 0x00,
- 0x22, 0x0c, 0x88, 0x30, 0x00, 0x00,
- 0x88, 0xba, 0x22, 0xe0, 0x00, 0x00,
- 0x42, 0x55, 0x09, 0x50, 0x00, 0x00,
- 0x28, 0xa4, 0xa2, 0x90, 0x00, 0x00,
- 0x94, 0x22, 0x50, 0x80, 0x00, 0x00,
- 0x1b, 0x04, 0x6c, 0x10, 0x00, 0x00,
- 0x22, 0xc0, 0x8b, 0x00, 0x00, 0x00,
- 0x6f, 0xd8, 0xee, 0xa0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_25[150] =
- {
- 0x81, 0x06, 0x04, 0x10, 0x00, 0x00,
- 0x40, 0x69, 0x01, 0xa0, 0x00, 0x00,
- 0x90, 0x26, 0x40, 0x90, 0x00, 0x00,
- 0x28, 0x28, 0xa0, 0xa0, 0x00, 0x00,
- 0x52, 0x11, 0x48, 0x40, 0x00, 0x00,
- 0x41, 0x89, 0x06, 0x20, 0x00, 0x00,
- 0x09, 0x30, 0x24, 0xc0, 0x00, 0x00,
- 0x48, 0x45, 0x21, 0x10, 0x00, 0x00,
- 0x04, 0x44, 0x11, 0x10, 0x00, 0x00,
- 0x0e, 0x80, 0x3a, 0x00, 0x00, 0x00,
- 0xa5, 0x92, 0x96, 0x40, 0x00, 0x00,
- 0x12, 0x0c, 0x48, 0x30, 0x00, 0x00,
- 0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
- 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
- 0x48, 0x05, 0x20, 0x10, 0x00, 0x00,
- 0x00, 0xac, 0x02, 0xb0, 0x00, 0x00,
- 0x28, 0x08, 0xa0, 0x20, 0x00, 0x00,
- 0x81, 0x0a, 0x04, 0x20, 0x00, 0x00,
- 0x23, 0x04, 0x8c, 0x10, 0x00, 0x00,
- 0x06, 0x80, 0x1a, 0x00, 0x00, 0x00,
- 0x80, 0x16, 0x00, 0x50, 0x00, 0x00,
- 0x30, 0x10, 0xc0, 0x40, 0x00, 0x00,
- 0x8c, 0x22, 0x30, 0x80, 0x00, 0x00,
- 0x54, 0x01, 0x50, 0x00, 0x00, 0x00,
- 0x80, 0xc2, 0x03, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_26[156] =
- {
- 0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
- 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
- 0x48, 0x05, 0x20, 0x10, 0x00, 0x00,
- 0x00, 0xac, 0x02, 0xb0, 0x00, 0x00,
- 0x28, 0x08, 0xa0, 0x20, 0x00, 0x00,
- 0x81, 0x0a, 0x04, 0x20, 0x00, 0x00,
- 0x23, 0x04, 0x8c, 0x10, 0x00, 0x00,
- 0x06, 0x80, 0x1a, 0x00, 0x00, 0x00,
- 0x80, 0x16, 0x00, 0x50, 0x00, 0x00,
- 0x30, 0x10, 0xc0, 0x40, 0x00, 0x00,
- 0x8c, 0x22, 0x30, 0x80, 0x00, 0x00,
- 0x54, 0x01, 0x50, 0x00, 0x00, 0x00,
- 0x80, 0xc2, 0x03, 0x00, 0x00, 0x00,
- 0x81, 0x06, 0x04, 0x10, 0x00, 0x00,
- 0x40, 0x69, 0x01, 0xa0, 0x00, 0x00,
- 0x90, 0x26, 0x40, 0x90, 0x00, 0x00,
- 0x28, 0x28, 0xa0, 0xa0, 0x00, 0x00,
- 0x52, 0x11, 0x48, 0x40, 0x00, 0x00,
- 0x41, 0x89, 0x06, 0x20, 0x00, 0x00,
- 0x09, 0x30, 0x24, 0xc0, 0x00, 0x00,
- 0x48, 0x45, 0x21, 0x10, 0x00, 0x00,
- 0x04, 0x44, 0x11, 0x10, 0x00, 0x00,
- 0x0e, 0x80, 0x3a, 0x00, 0x00, 0x00,
- 0xa5, 0x92, 0x96, 0x40, 0x00, 0x00,
- 0x12, 0x0c, 0x48, 0x30, 0x00, 0x00,
- 0xf1, 0x64, 0xbe, 0x40, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_27[162] =
- {
- 0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
- 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
- 0x48, 0x05, 0x20, 0x10, 0x00, 0x00,
- 0x00, 0xac, 0x02, 0xb0, 0x00, 0x00,
- 0x28, 0x08, 0xa0, 0x20, 0x00, 0x00,
- 0x81, 0x0a, 0x04, 0x20, 0x00, 0x00,
- 0x23, 0x04, 0x8c, 0x10, 0x00, 0x00,
- 0x06, 0x80, 0x1a, 0x00, 0x00, 0x00,
- 0x80, 0x16, 0x00, 0x50, 0x00, 0x00,
- 0x30, 0x10, 0xc0, 0x40, 0x00, 0x00,
- 0x8c, 0x22, 0x30, 0x80, 0x00, 0x00,
- 0x54, 0x01, 0x50, 0x00, 0x00, 0x00,
- 0x80, 0xc2, 0x03, 0x00, 0x00, 0x00,
- 0x40, 0x55, 0x01, 0x50, 0x00, 0x00,
- 0x15, 0x40, 0x55, 0x00, 0x00, 0x00,
- 0xc0, 0x07, 0x00, 0x10, 0x00, 0x00,
- 0x28, 0x10, 0xa0, 0x40, 0x00, 0x00,
- 0x05, 0x0c, 0x14, 0x30, 0x00, 0x00,
- 0x64, 0x81, 0x92, 0x00, 0x00, 0x00,
- 0x81, 0x82, 0x06, 0x00, 0x00, 0x00,
- 0x10, 0x98, 0x42, 0x60, 0x00, 0x00,
- 0x84, 0x22, 0x10, 0x80, 0x00, 0x00,
- 0x12, 0x30, 0x48, 0xc0, 0x00, 0x00,
- 0x62, 0x01, 0x88, 0x00, 0x00, 0x00,
- 0x28, 0x60, 0xa1, 0x80, 0x00, 0x00,
- 0x0e, 0x08, 0x38, 0x20, 0x00, 0x00,
- 0x10, 0x84, 0x42, 0x10, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_28[168] =
- {
- 0x40, 0x55, 0x01, 0x50, 0x00, 0x00,
- 0x15, 0x40, 0x55, 0x00, 0x00, 0x00,
- 0xc0, 0x07, 0x00, 0x10, 0x00, 0x00,
- 0x28, 0x10, 0xa0, 0x40, 0x00, 0x00,
- 0x05, 0x0c, 0x14, 0x30, 0x00, 0x00,
- 0x64, 0x81, 0x92, 0x00, 0x00, 0x00,
- 0x81, 0x82, 0x06, 0x00, 0x00, 0x00,
- 0x10, 0x98, 0x42, 0x60, 0x00, 0x00,
- 0x84, 0x22, 0x10, 0x80, 0x00, 0x00,
- 0x12, 0x30, 0x48, 0xc0, 0x00, 0x00,
- 0x62, 0x01, 0x88, 0x00, 0x00, 0x00,
- 0x28, 0x60, 0xa1, 0x80, 0x00, 0x00,
- 0x0e, 0x08, 0x38, 0x20, 0x00, 0x00,
- 0x10, 0x84, 0x42, 0x10, 0x00, 0x00,
- 0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
- 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
- 0x48, 0x05, 0x20, 0x10, 0x00, 0x00,
- 0x00, 0xac, 0x02, 0xb0, 0x00, 0x00,
- 0x28, 0x08, 0xa0, 0x20, 0x00, 0x00,
- 0x81, 0x0a, 0x04, 0x20, 0x00, 0x00,
- 0x23, 0x04, 0x8c, 0x10, 0x00, 0x00,
- 0x06, 0x80, 0x1a, 0x00, 0x00, 0x00,
- 0x80, 0x16, 0x00, 0x50, 0x00, 0x00,
- 0x30, 0x10, 0xc0, 0x40, 0x00, 0x00,
- 0x8c, 0x22, 0x30, 0x80, 0x00, 0x00,
- 0x54, 0x01, 0x50, 0x00, 0x00, 0x00,
- 0x80, 0xc2, 0x03, 0x00, 0x00, 0x00,
- 0x36, 0x4f, 0x1f, 0xb0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_3[18] =
- {
- 0xac, 0xda, 0xb3, 0x60, 0x00, 0x00,
- 0x55, 0x6d, 0x55, 0xb0, 0x00, 0x00,
- 0x27, 0xb4, 0x9e, 0xd0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_4[24] =
- {
- 0x2c, 0xd8, 0xb3, 0x60, 0x00, 0x00,
- 0x93, 0x6a, 0x4d, 0xa0, 0x00, 0x00,
- 0x1a, 0xb4, 0x6a, 0xd0, 0x00, 0x00,
- 0x47, 0x2d, 0x1c, 0xb0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_5[30] =
- {
- 0x64, 0xd9, 0x93, 0x60, 0x00, 0x00,
- 0xa5, 0x6a, 0x95, 0xa0, 0x00, 0x00,
- 0x52, 0xb5, 0x4a, 0xd0, 0x00, 0x00,
- 0x1d, 0xa8, 0x76, 0xa0, 0x00, 0x00,
- 0x9c, 0x56, 0x71, 0x50, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_6[36] =
- {
- 0x4a, 0x55, 0x29, 0x50, 0x00, 0x00,
- 0x95, 0x4a, 0x55, 0x20, 0x00, 0x00,
- 0x14, 0xb4, 0x52, 0xd0, 0x00, 0x00,
- 0x51, 0xa9, 0x46, 0xa0, 0x00, 0x00,
- 0x22, 0x6c, 0x89, 0xb0, 0x00, 0x00,
- 0x88, 0x8e, 0x22, 0x30, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_7[42] =
- {
- 0x62, 0x55, 0x89, 0x50, 0x00, 0x00,
- 0xb9, 0x22, 0xe4, 0x80, 0x00, 0x00,
- 0x18, 0xb4, 0x62, 0xd0, 0x00, 0x00,
- 0x54, 0x99, 0x52, 0x60, 0x00, 0x00,
- 0x06, 0x6c, 0x19, 0xb0, 0x00, 0x00,
- 0x85, 0x56, 0x15, 0x50, 0x00, 0x00,
- 0xaa, 0x8a, 0xaa, 0x20, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_8[48] =
- {
- 0xc0, 0x17, 0x00, 0x50, 0x00, 0x00,
- 0x41, 0x61, 0x05, 0x80, 0x00, 0x00,
- 0x88, 0x32, 0x20, 0xc0, 0x00, 0x00,
- 0x20, 0xa4, 0x82, 0x90, 0x00, 0x00,
- 0x0a, 0x48, 0x29, 0x20, 0x00, 0x00,
- 0x04, 0x98, 0x12, 0x60, 0x00, 0x00,
- 0x94, 0x42, 0x51, 0x00, 0x00, 0x00,
- 0x72, 0x01, 0xc8, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask28_9[54] =
- {
- 0xa2, 0x56, 0x89, 0x50, 0x00, 0x00,
- 0x34, 0x60, 0xd1, 0x80, 0x00, 0x00,
- 0x4a, 0x25, 0x28, 0x90, 0x00, 0x00,
- 0x20, 0xa8, 0x82, 0xa0, 0x00, 0x00,
- 0x11, 0x84, 0x46, 0x10, 0x00, 0x00,
- 0x49, 0x09, 0x24, 0x20, 0x00, 0x00,
- 0x86, 0x0e, 0x18, 0x30, 0x00, 0x00,
- 0x20, 0xd4, 0x83, 0x50, 0x00, 0x00,
- 0x88, 0x4a, 0x21, 0x20, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_1[6] =
- {
- 0xff, 0xff, 0xff, 0xf8, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_10[60] =
- {
- 0xc0, 0xd7, 0x02, 0x80, 0x00, 0x00,
- 0x1d, 0x40, 0x55, 0x58, 0x00, 0x00,
- 0xd4, 0x09, 0xd1, 0x00, 0x00, 0x00,
- 0x02, 0x60, 0x02, 0x70, 0x00, 0x00,
- 0x04, 0x28, 0x04, 0xb0, 0x00, 0x00,
- 0x20, 0x99, 0x12, 0x48, 0x00, 0x00,
- 0x40, 0x46, 0x21, 0x40, 0x00, 0x00,
- 0x08, 0x84, 0x82, 0x90, 0x00, 0x00,
- 0x68, 0x02, 0xa8, 0x10, 0x00, 0x00,
- 0x23, 0x10, 0x09, 0x88, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_11[66] =
- {
- 0x62, 0xd1, 0x88, 0x88, 0x00, 0x00,
- 0x35, 0x23, 0xc4, 0x40, 0x00, 0x00,
- 0x14, 0x14, 0x40, 0x38, 0x00, 0x00,
- 0xc5, 0x08, 0x42, 0xc0, 0x00, 0x00,
- 0x22, 0x0c, 0x90, 0x90, 0x00, 0x00,
- 0x88, 0xb8, 0x04, 0x48, 0x00, 0x00,
- 0x42, 0x54, 0x03, 0x10, 0x00, 0x00,
- 0x28, 0xa4, 0x12, 0x88, 0x00, 0x00,
- 0x94, 0x20, 0x09, 0x60, 0x00, 0x00,
- 0x1b, 0x04, 0xac, 0x00, 0x00, 0x00,
- 0x22, 0xc2, 0x61, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_12[72] =
- {
- 0x81, 0x06, 0x22, 0x40, 0x00, 0x00,
- 0x40, 0x69, 0x01, 0x50, 0x00, 0x00,
- 0x90, 0x26, 0x09, 0x88, 0x00, 0x00,
- 0x28, 0x28, 0x86, 0x90, 0x00, 0x00,
- 0x52, 0x10, 0x41, 0x90, 0x00, 0x00,
- 0x41, 0x89, 0x10, 0x28, 0x00, 0x00,
- 0x09, 0x30, 0x43, 0x20, 0x00, 0x00,
- 0x48, 0x45, 0x34, 0xa8, 0x00, 0x00,
- 0x04, 0x44, 0xe0, 0x08, 0x00, 0x00,
- 0x0e, 0x80, 0x5d, 0x20, 0x00, 0x00,
- 0xa5, 0x92, 0x42, 0x10, 0x00, 0x00,
- 0x12, 0x0d, 0xc8, 0x50, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_13[78] =
- {
- 0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
- 0x34, 0x60, 0xd1, 0x10, 0x00, 0x00,
- 0x48, 0x05, 0x01, 0x28, 0x00, 0x00,
- 0x00, 0xaf, 0x10, 0x10, 0x00, 0x00,
- 0x28, 0x08, 0x21, 0x80, 0x00, 0x00,
- 0x81, 0x0a, 0x50, 0x48, 0x00, 0x00,
- 0x23, 0x06, 0x23, 0x00, 0x00, 0x00,
- 0x06, 0x80, 0x84, 0xc8, 0x00, 0x00,
- 0x80, 0x17, 0x05, 0x00, 0x00, 0x00,
- 0x30, 0x10, 0x41, 0xa0, 0x00, 0x00,
- 0x8c, 0x20, 0x1a, 0x40, 0x00, 0x00,
- 0x54, 0x01, 0x64, 0x00, 0x00, 0x00,
- 0x80, 0xc0, 0x28, 0x30, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_14[84] =
- {
- 0x40, 0x55, 0x02, 0x08, 0x00, 0x00,
- 0x15, 0x40, 0x55, 0x50, 0x00, 0x00,
- 0xc0, 0x06, 0x20, 0x48, 0x00, 0x00,
- 0x28, 0x13, 0x00, 0x40, 0x00, 0x00,
- 0x05, 0x0e, 0x02, 0x80, 0x00, 0x00,
- 0x64, 0x80, 0x04, 0x88, 0x00, 0x00,
- 0x81, 0x81, 0x00, 0xb0, 0x00, 0x00,
- 0x10, 0x98, 0x88, 0x08, 0x00, 0x00,
- 0x84, 0x22, 0x40, 0x10, 0x00, 0x00,
- 0x12, 0x30, 0x49, 0x00, 0x00, 0x00,
- 0x62, 0x01, 0x74, 0x00, 0x00, 0x00,
- 0x28, 0x60, 0x81, 0x50, 0x00, 0x00,
- 0x0e, 0x0a, 0x18, 0x20, 0x00, 0x00,
- 0x10, 0x84, 0xa2, 0x20, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_15[90] =
- {
- 0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
- 0xb9, 0x22, 0xc4, 0x50, 0x00, 0x00,
- 0x18, 0xb4, 0x61, 0xa8, 0x00, 0x00,
- 0x54, 0x99, 0x13, 0x50, 0x00, 0x00,
- 0x06, 0x6c, 0x4d, 0x90, 0x00, 0x00,
- 0x85, 0x55, 0x24, 0x68, 0x00, 0x00,
- 0xaa, 0x8a, 0x1a, 0x30, 0x00, 0x00,
- 0xc0, 0x16, 0x40, 0x88, 0x00, 0x00,
- 0x41, 0x60, 0x25, 0x40, 0x00, 0x00,
- 0x88, 0x30, 0x01, 0xa8, 0x00, 0x00,
- 0x20, 0xa4, 0x80, 0xd0, 0x00, 0x00,
- 0x0a, 0x48, 0x51, 0x10, 0x00, 0x00,
- 0x04, 0x9b, 0x08, 0x40, 0x00, 0x00,
- 0x94, 0x40, 0x03, 0x18, 0x00, 0x00,
- 0x72, 0x01, 0x96, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_16[96] =
- {
- 0xc0, 0x16, 0x40, 0x88, 0x00, 0x00,
- 0x41, 0x60, 0x25, 0x40, 0x00, 0x00,
- 0x88, 0x30, 0x01, 0xa8, 0x00, 0x00,
- 0x20, 0xa4, 0x80, 0xd0, 0x00, 0x00,
- 0x0a, 0x48, 0x51, 0x10, 0x00, 0x00,
- 0x04, 0x9b, 0x08, 0x40, 0x00, 0x00,
- 0x94, 0x40, 0x03, 0x18, 0x00, 0x00,
- 0x72, 0x01, 0x96, 0x00, 0x00, 0x00,
- 0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
- 0xb9, 0x22, 0xc4, 0x50, 0x00, 0x00,
- 0x18, 0xb4, 0x61, 0xa8, 0x00, 0x00,
- 0x54, 0x99, 0x13, 0x50, 0x00, 0x00,
- 0x06, 0x6c, 0x4d, 0x90, 0x00, 0x00,
- 0x85, 0x55, 0x24, 0x68, 0x00, 0x00,
- 0xaa, 0x8a, 0x1a, 0x30, 0x00, 0x00,
- 0x0d, 0x2c, 0xf2, 0x20, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_17[102] =
- {
- 0xc0, 0x16, 0x40, 0x88, 0x00, 0x00,
- 0x41, 0x60, 0x25, 0x40, 0x00, 0x00,
- 0x88, 0x30, 0x01, 0xa8, 0x00, 0x00,
- 0x20, 0xa4, 0x80, 0xd0, 0x00, 0x00,
- 0x0a, 0x48, 0x51, 0x10, 0x00, 0x00,
- 0x04, 0x9b, 0x08, 0x40, 0x00, 0x00,
- 0x94, 0x40, 0x03, 0x18, 0x00, 0x00,
- 0x72, 0x01, 0x96, 0x00, 0x00, 0x00,
- 0xa2, 0x55, 0x88, 0x88, 0x00, 0x00,
- 0x34, 0x60, 0x91, 0x10, 0x00, 0x00,
- 0x4a, 0x27, 0x01, 0x40, 0x00, 0x00,
- 0x20, 0xa8, 0x0c, 0x30, 0x00, 0x00,
- 0x11, 0x84, 0x58, 0xa0, 0x00, 0x00,
- 0x49, 0x0a, 0x24, 0x00, 0x00, 0x00,
- 0x86, 0x0e, 0x0a, 0x40, 0x00, 0x00,
- 0x20, 0xd4, 0x22, 0x90, 0x00, 0x00,
- 0x88, 0x4a, 0x41, 0x20, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_18[108] =
- {
- 0xa2, 0x55, 0x88, 0x88, 0x00, 0x00,
- 0x34, 0x60, 0x91, 0x10, 0x00, 0x00,
- 0x4a, 0x27, 0x01, 0x40, 0x00, 0x00,
- 0x20, 0xa8, 0x0c, 0x30, 0x00, 0x00,
- 0x11, 0x84, 0x58, 0xa0, 0x00, 0x00,
- 0x49, 0x0a, 0x24, 0x00, 0x00, 0x00,
- 0x86, 0x0e, 0x0a, 0x40, 0x00, 0x00,
- 0x20, 0xd4, 0x22, 0x90, 0x00, 0x00,
- 0x88, 0x4a, 0x41, 0x20, 0x00, 0x00,
- 0xc0, 0x16, 0x40, 0x88, 0x00, 0x00,
- 0x41, 0x60, 0x25, 0x40, 0x00, 0x00,
- 0x88, 0x30, 0x01, 0xa8, 0x00, 0x00,
- 0x20, 0xa4, 0x80, 0xd0, 0x00, 0x00,
- 0x0a, 0x48, 0x51, 0x10, 0x00, 0x00,
- 0x04, 0x9b, 0x08, 0x40, 0x00, 0x00,
- 0x94, 0x40, 0x03, 0x18, 0x00, 0x00,
- 0x72, 0x01, 0x96, 0x00, 0x00, 0x00,
- 0x71, 0x36, 0xf2, 0xb0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_19[114] =
- {
- 0xa2, 0x55, 0x88, 0x88, 0x00, 0x00,
- 0x34, 0x60, 0x91, 0x10, 0x00, 0x00,
- 0x4a, 0x27, 0x01, 0x40, 0x00, 0x00,
- 0x20, 0xa8, 0x0c, 0x30, 0x00, 0x00,
- 0x11, 0x84, 0x58, 0xa0, 0x00, 0x00,
- 0x49, 0x0a, 0x24, 0x00, 0x00, 0x00,
- 0x86, 0x0e, 0x0a, 0x40, 0x00, 0x00,
- 0x20, 0xd4, 0x22, 0x90, 0x00, 0x00,
- 0x88, 0x4a, 0x41, 0x20, 0x00, 0x00,
- 0xc0, 0xd7, 0x02, 0x80, 0x00, 0x00,
- 0x1d, 0x40, 0x55, 0x58, 0x00, 0x00,
- 0xd4, 0x09, 0xd1, 0x00, 0x00, 0x00,
- 0x02, 0x60, 0x02, 0x70, 0x00, 0x00,
- 0x04, 0x28, 0x04, 0xb0, 0x00, 0x00,
- 0x20, 0x99, 0x12, 0x48, 0x00, 0x00,
- 0x40, 0x46, 0x21, 0x40, 0x00, 0x00,
- 0x08, 0x84, 0x82, 0x90, 0x00, 0x00,
- 0x68, 0x02, 0xa8, 0x10, 0x00, 0x00,
- 0x23, 0x10, 0x09, 0x88, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_2[12] =
- {
- 0xec, 0xeb, 0xb3, 0xa8, 0x00, 0x00,
- 0x3b, 0x9e, 0xee, 0x70, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_20[120] =
- {
- 0xc0, 0xd7, 0x02, 0x80, 0x00, 0x00,
- 0x1d, 0x40, 0x55, 0x58, 0x00, 0x00,
- 0xd4, 0x09, 0xd1, 0x00, 0x00, 0x00,
- 0x02, 0x60, 0x02, 0x70, 0x00, 0x00,
- 0x04, 0x28, 0x04, 0xb0, 0x00, 0x00,
- 0x20, 0x99, 0x12, 0x48, 0x00, 0x00,
- 0x40, 0x46, 0x21, 0x40, 0x00, 0x00,
- 0x08, 0x84, 0x82, 0x90, 0x00, 0x00,
- 0x68, 0x02, 0xa8, 0x10, 0x00, 0x00,
- 0x23, 0x10, 0x09, 0x88, 0x00, 0x00,
- 0xa2, 0x55, 0x88, 0x88, 0x00, 0x00,
- 0x34, 0x60, 0x91, 0x10, 0x00, 0x00,
- 0x4a, 0x27, 0x01, 0x40, 0x00, 0x00,
- 0x20, 0xa8, 0x0c, 0x30, 0x00, 0x00,
- 0x11, 0x84, 0x58, 0xa0, 0x00, 0x00,
- 0x49, 0x0a, 0x24, 0x00, 0x00, 0x00,
- 0x86, 0x0e, 0x0a, 0x40, 0x00, 0x00,
- 0x20, 0xd4, 0x22, 0x90, 0x00, 0x00,
- 0x88, 0x4a, 0x41, 0x20, 0x00, 0x00,
- 0xe7, 0xec, 0xdc, 0xb0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_21[126] =
- {
- 0xc0, 0xd7, 0x02, 0x80, 0x00, 0x00,
- 0x1d, 0x40, 0x55, 0x58, 0x00, 0x00,
- 0xd4, 0x09, 0xd1, 0x00, 0x00, 0x00,
- 0x02, 0x60, 0x02, 0x70, 0x00, 0x00,
- 0x04, 0x28, 0x04, 0xb0, 0x00, 0x00,
- 0x20, 0x99, 0x12, 0x48, 0x00, 0x00,
- 0x40, 0x46, 0x21, 0x40, 0x00, 0x00,
- 0x08, 0x84, 0x82, 0x90, 0x00, 0x00,
- 0x68, 0x02, 0xa8, 0x10, 0x00, 0x00,
- 0x23, 0x10, 0x09, 0x88, 0x00, 0x00,
- 0x62, 0xd1, 0x88, 0x88, 0x00, 0x00,
- 0x35, 0x23, 0xc4, 0x40, 0x00, 0x00,
- 0x14, 0x14, 0x40, 0x38, 0x00, 0x00,
- 0xc5, 0x08, 0x42, 0xc0, 0x00, 0x00,
- 0x22, 0x0c, 0x90, 0x90, 0x00, 0x00,
- 0x88, 0xb8, 0x04, 0x48, 0x00, 0x00,
- 0x42, 0x54, 0x03, 0x10, 0x00, 0x00,
- 0x28, 0xa4, 0x12, 0x88, 0x00, 0x00,
- 0x94, 0x20, 0x09, 0x60, 0x00, 0x00,
- 0x1b, 0x04, 0xac, 0x00, 0x00, 0x00,
- 0x22, 0xc2, 0x61, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_22[132] =
- {
- 0x62, 0xd1, 0x88, 0x88, 0x00, 0x00,
- 0x35, 0x23, 0xc4, 0x40, 0x00, 0x00,
- 0x14, 0x14, 0x40, 0x38, 0x00, 0x00,
- 0xc5, 0x08, 0x42, 0xc0, 0x00, 0x00,
- 0x22, 0x0c, 0x90, 0x90, 0x00, 0x00,
- 0x88, 0xb8, 0x04, 0x48, 0x00, 0x00,
- 0x42, 0x54, 0x03, 0x10, 0x00, 0x00,
- 0x28, 0xa4, 0x12, 0x88, 0x00, 0x00,
- 0x94, 0x20, 0x09, 0x60, 0x00, 0x00,
- 0x1b, 0x04, 0xac, 0x00, 0x00, 0x00,
- 0x22, 0xc2, 0x61, 0x00, 0x00, 0x00,
- 0xc0, 0xd7, 0x02, 0x80, 0x00, 0x00,
- 0x1d, 0x40, 0x55, 0x58, 0x00, 0x00,
- 0xd4, 0x09, 0xd1, 0x00, 0x00, 0x00,
- 0x02, 0x60, 0x02, 0x70, 0x00, 0x00,
- 0x04, 0x28, 0x04, 0xb0, 0x00, 0x00,
- 0x20, 0x99, 0x12, 0x48, 0x00, 0x00,
- 0x40, 0x46, 0x21, 0x40, 0x00, 0x00,
- 0x08, 0x84, 0x82, 0x90, 0x00, 0x00,
- 0x68, 0x02, 0xa8, 0x10, 0x00, 0x00,
- 0x23, 0x10, 0x09, 0x88, 0x00, 0x00,
- 0x1c, 0x90, 0xa9, 0xa0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_23[138] =
- {
- 0x62, 0xd1, 0x88, 0x88, 0x00, 0x00,
- 0x35, 0x23, 0xc4, 0x40, 0x00, 0x00,
- 0x14, 0x14, 0x40, 0x38, 0x00, 0x00,
- 0xc5, 0x08, 0x42, 0xc0, 0x00, 0x00,
- 0x22, 0x0c, 0x90, 0x90, 0x00, 0x00,
- 0x88, 0xb8, 0x04, 0x48, 0x00, 0x00,
- 0x42, 0x54, 0x03, 0x10, 0x00, 0x00,
- 0x28, 0xa4, 0x12, 0x88, 0x00, 0x00,
- 0x94, 0x20, 0x09, 0x60, 0x00, 0x00,
- 0x1b, 0x04, 0xac, 0x00, 0x00, 0x00,
- 0x22, 0xc2, 0x61, 0x00, 0x00, 0x00,
- 0x81, 0x06, 0x22, 0x40, 0x00, 0x00,
- 0x40, 0x69, 0x01, 0x50, 0x00, 0x00,
- 0x90, 0x26, 0x09, 0x88, 0x00, 0x00,
- 0x28, 0x28, 0x86, 0x90, 0x00, 0x00,
- 0x52, 0x10, 0x41, 0x90, 0x00, 0x00,
- 0x41, 0x89, 0x10, 0x28, 0x00, 0x00,
- 0x09, 0x30, 0x43, 0x20, 0x00, 0x00,
- 0x48, 0x45, 0x34, 0xa8, 0x00, 0x00,
- 0x04, 0x44, 0xe0, 0x08, 0x00, 0x00,
- 0x0e, 0x80, 0x5d, 0x20, 0x00, 0x00,
- 0xa5, 0x92, 0x42, 0x10, 0x00, 0x00,
- 0x12, 0x0d, 0xc8, 0x50, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_24[144] =
- {
- 0x81, 0x06, 0x22, 0x40, 0x00, 0x00,
- 0x40, 0x69, 0x01, 0x50, 0x00, 0x00,
- 0x90, 0x26, 0x09, 0x88, 0x00, 0x00,
- 0x28, 0x28, 0x86, 0x90, 0x00, 0x00,
- 0x52, 0x10, 0x41, 0x90, 0x00, 0x00,
- 0x41, 0x89, 0x10, 0x28, 0x00, 0x00,
- 0x09, 0x30, 0x43, 0x20, 0x00, 0x00,
- 0x48, 0x45, 0x34, 0xa8, 0x00, 0x00,
- 0x04, 0x44, 0xe0, 0x08, 0x00, 0x00,
- 0x0e, 0x80, 0x5d, 0x20, 0x00, 0x00,
- 0xa5, 0x92, 0x42, 0x10, 0x00, 0x00,
- 0x12, 0x0d, 0xc8, 0x50, 0x00, 0x00,
- 0x62, 0xd1, 0x88, 0x88, 0x00, 0x00,
- 0x35, 0x23, 0xc4, 0x40, 0x00, 0x00,
- 0x14, 0x14, 0x40, 0x38, 0x00, 0x00,
- 0xc5, 0x08, 0x42, 0xc0, 0x00, 0x00,
- 0x22, 0x0c, 0x90, 0x90, 0x00, 0x00,
- 0x88, 0xb8, 0x04, 0x48, 0x00, 0x00,
- 0x42, 0x54, 0x03, 0x10, 0x00, 0x00,
- 0x28, 0xa4, 0x12, 0x88, 0x00, 0x00,
- 0x94, 0x20, 0x09, 0x60, 0x00, 0x00,
- 0x1b, 0x04, 0xac, 0x00, 0x00, 0x00,
- 0x22, 0xc2, 0x61, 0x00, 0x00, 0x00,
- 0xbd, 0x86, 0x97, 0xc0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_25[150] =
- {
- 0x81, 0x06, 0x22, 0x40, 0x00, 0x00,
- 0x40, 0x69, 0x01, 0x50, 0x00, 0x00,
- 0x90, 0x26, 0x09, 0x88, 0x00, 0x00,
- 0x28, 0x28, 0x86, 0x90, 0x00, 0x00,
- 0x52, 0x10, 0x41, 0x90, 0x00, 0x00,
- 0x41, 0x89, 0x10, 0x28, 0x00, 0x00,
- 0x09, 0x30, 0x43, 0x20, 0x00, 0x00,
- 0x48, 0x45, 0x34, 0xa8, 0x00, 0x00,
- 0x04, 0x44, 0xe0, 0x08, 0x00, 0x00,
- 0x0e, 0x80, 0x5d, 0x20, 0x00, 0x00,
- 0xa5, 0x92, 0x42, 0x10, 0x00, 0x00,
- 0x12, 0x0d, 0xc8, 0x50, 0x00, 0x00,
- 0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
- 0x34, 0x60, 0xd1, 0x10, 0x00, 0x00,
- 0x48, 0x05, 0x01, 0x28, 0x00, 0x00,
- 0x00, 0xaf, 0x10, 0x10, 0x00, 0x00,
- 0x28, 0x08, 0x21, 0x80, 0x00, 0x00,
- 0x81, 0x0a, 0x50, 0x48, 0x00, 0x00,
- 0x23, 0x06, 0x23, 0x00, 0x00, 0x00,
- 0x06, 0x80, 0x84, 0xc8, 0x00, 0x00,
- 0x80, 0x17, 0x05, 0x00, 0x00, 0x00,
- 0x30, 0x10, 0x41, 0xa0, 0x00, 0x00,
- 0x8c, 0x20, 0x1a, 0x40, 0x00, 0x00,
- 0x54, 0x01, 0x64, 0x00, 0x00, 0x00,
- 0x80, 0xc0, 0x28, 0x30, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_26[156] =
- {
- 0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
- 0x34, 0x60, 0xd1, 0x10, 0x00, 0x00,
- 0x48, 0x05, 0x01, 0x28, 0x00, 0x00,
- 0x00, 0xaf, 0x10, 0x10, 0x00, 0x00,
- 0x28, 0x08, 0x21, 0x80, 0x00, 0x00,
- 0x81, 0x0a, 0x50, 0x48, 0x00, 0x00,
- 0x23, 0x06, 0x23, 0x00, 0x00, 0x00,
- 0x06, 0x80, 0x84, 0xc8, 0x00, 0x00,
- 0x80, 0x17, 0x05, 0x00, 0x00, 0x00,
- 0x30, 0x10, 0x41, 0xa0, 0x00, 0x00,
- 0x8c, 0x20, 0x1a, 0x40, 0x00, 0x00,
- 0x54, 0x01, 0x64, 0x00, 0x00, 0x00,
- 0x80, 0xc0, 0x28, 0x30, 0x00, 0x00,
- 0x81, 0x06, 0x22, 0x40, 0x00, 0x00,
- 0x40, 0x69, 0x01, 0x50, 0x00, 0x00,
- 0x90, 0x26, 0x09, 0x88, 0x00, 0x00,
- 0x28, 0x28, 0x86, 0x90, 0x00, 0x00,
- 0x52, 0x10, 0x41, 0x90, 0x00, 0x00,
- 0x41, 0x89, 0x10, 0x28, 0x00, 0x00,
- 0x09, 0x30, 0x43, 0x20, 0x00, 0x00,
- 0x48, 0x45, 0x34, 0xa8, 0x00, 0x00,
- 0x04, 0x44, 0xe0, 0x08, 0x00, 0x00,
- 0x0e, 0x80, 0x5d, 0x20, 0x00, 0x00,
- 0xa5, 0x92, 0x42, 0x10, 0x00, 0x00,
- 0x12, 0x0d, 0xc8, 0x50, 0x00, 0x00,
- 0xb5, 0x4c, 0xa9, 0x70, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_27[162] =
- {
- 0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
- 0x34, 0x60, 0xd1, 0x10, 0x00, 0x00,
- 0x48, 0x05, 0x01, 0x28, 0x00, 0x00,
- 0x00, 0xaf, 0x10, 0x10, 0x00, 0x00,
- 0x28, 0x08, 0x21, 0x80, 0x00, 0x00,
- 0x81, 0x0a, 0x50, 0x48, 0x00, 0x00,
- 0x23, 0x06, 0x23, 0x00, 0x00, 0x00,
- 0x06, 0x80, 0x84, 0xc8, 0x00, 0x00,
- 0x80, 0x17, 0x05, 0x00, 0x00, 0x00,
- 0x30, 0x10, 0x41, 0xa0, 0x00, 0x00,
- 0x8c, 0x20, 0x1a, 0x40, 0x00, 0x00,
- 0x54, 0x01, 0x64, 0x00, 0x00, 0x00,
- 0x80, 0xc0, 0x28, 0x30, 0x00, 0x00,
- 0x40, 0x55, 0x02, 0x08, 0x00, 0x00,
- 0x15, 0x40, 0x55, 0x50, 0x00, 0x00,
- 0xc0, 0x06, 0x20, 0x48, 0x00, 0x00,
- 0x28, 0x13, 0x00, 0x40, 0x00, 0x00,
- 0x05, 0x0e, 0x02, 0x80, 0x00, 0x00,
- 0x64, 0x80, 0x04, 0x88, 0x00, 0x00,
- 0x81, 0x81, 0x00, 0xb0, 0x00, 0x00,
- 0x10, 0x98, 0x88, 0x08, 0x00, 0x00,
- 0x84, 0x22, 0x40, 0x10, 0x00, 0x00,
- 0x12, 0x30, 0x49, 0x00, 0x00, 0x00,
- 0x62, 0x01, 0x74, 0x00, 0x00, 0x00,
- 0x28, 0x60, 0x81, 0x50, 0x00, 0x00,
- 0x0e, 0x0a, 0x18, 0x20, 0x00, 0x00,
- 0x10, 0x84, 0xa2, 0x20, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_28[168] =
- {
- 0x40, 0x55, 0x02, 0x08, 0x00, 0x00,
- 0x15, 0x40, 0x55, 0x50, 0x00, 0x00,
- 0xc0, 0x06, 0x20, 0x48, 0x00, 0x00,
- 0x28, 0x13, 0x00, 0x40, 0x00, 0x00,
- 0x05, 0x0e, 0x02, 0x80, 0x00, 0x00,
- 0x64, 0x80, 0x04, 0x88, 0x00, 0x00,
- 0x81, 0x81, 0x00, 0xb0, 0x00, 0x00,
- 0x10, 0x98, 0x88, 0x08, 0x00, 0x00,
- 0x84, 0x22, 0x40, 0x10, 0x00, 0x00,
- 0x12, 0x30, 0x49, 0x00, 0x00, 0x00,
- 0x62, 0x01, 0x74, 0x00, 0x00, 0x00,
- 0x28, 0x60, 0x81, 0x50, 0x00, 0x00,
- 0x0e, 0x0a, 0x18, 0x20, 0x00, 0x00,
- 0x10, 0x84, 0xa2, 0x20, 0x00, 0x00,
- 0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
- 0x34, 0x60, 0xd1, 0x10, 0x00, 0x00,
- 0x48, 0x05, 0x01, 0x28, 0x00, 0x00,
- 0x00, 0xaf, 0x10, 0x10, 0x00, 0x00,
- 0x28, 0x08, 0x21, 0x80, 0x00, 0x00,
- 0x81, 0x0a, 0x50, 0x48, 0x00, 0x00,
- 0x23, 0x06, 0x23, 0x00, 0x00, 0x00,
- 0x06, 0x80, 0x84, 0xc8, 0x00, 0x00,
- 0x80, 0x17, 0x05, 0x00, 0x00, 0x00,
- 0x30, 0x10, 0x41, 0xa0, 0x00, 0x00,
- 0x8c, 0x20, 0x1a, 0x40, 0x00, 0x00,
- 0x54, 0x01, 0x64, 0x00, 0x00, 0x00,
- 0x80, 0xc0, 0x28, 0x30, 0x00, 0x00,
- 0xbe, 0x1f, 0x99, 0xb0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_29[174] =
- {
- 0x40, 0x55, 0x02, 0x08, 0x00, 0x00,
- 0x15, 0x40, 0x55, 0x50, 0x00, 0x00,
- 0xc0, 0x06, 0x20, 0x48, 0x00, 0x00,
- 0x28, 0x13, 0x00, 0x40, 0x00, 0x00,
- 0x05, 0x0e, 0x02, 0x80, 0x00, 0x00,
- 0x64, 0x80, 0x04, 0x88, 0x00, 0x00,
- 0x81, 0x81, 0x00, 0xb0, 0x00, 0x00,
- 0x10, 0x98, 0x88, 0x08, 0x00, 0x00,
- 0x84, 0x22, 0x40, 0x10, 0x00, 0x00,
- 0x12, 0x30, 0x49, 0x00, 0x00, 0x00,
- 0x62, 0x01, 0x74, 0x00, 0x00, 0x00,
- 0x28, 0x60, 0x81, 0x50, 0x00, 0x00,
- 0x0e, 0x0a, 0x18, 0x20, 0x00, 0x00,
- 0x10, 0x84, 0xa2, 0x20, 0x00, 0x00,
- 0x40, 0x55, 0x88, 0x88, 0x00, 0x00,
- 0x15, 0x40, 0xc4, 0x40, 0x00, 0x00,
- 0xc0, 0x05, 0x60, 0x00, 0x00, 0x00,
- 0x28, 0x10, 0x04, 0x48, 0x00, 0x00,
- 0x05, 0x0e, 0x20, 0x80, 0x00, 0x00,
- 0x64, 0x81, 0x10, 0x08, 0x00, 0x00,
- 0x81, 0x80, 0xa4, 0x10, 0x00, 0x00,
- 0x10, 0x9a, 0x0a, 0x80, 0x00, 0x00,
- 0x84, 0x20, 0x28, 0x68, 0x00, 0x00,
- 0x12, 0x30, 0x47, 0x80, 0x00, 0x00,
- 0x62, 0x02, 0x10, 0x10, 0x00, 0x00,
- 0x28, 0x62, 0x19, 0x00, 0x00, 0x00,
- 0x0e, 0x08, 0x02, 0x18, 0x00, 0x00,
- 0x10, 0x85, 0x11, 0x20, 0x00, 0x00,
- 0x29, 0x50, 0x42, 0x60, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_3[18] =
- {
- 0xac, 0xda, 0xb2, 0x48, 0x00, 0x00,
- 0x55, 0x6d, 0x55, 0x28, 0x00, 0x00,
- 0x27, 0xb5, 0x0c, 0xd8, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_4[24] =
- {
- 0x2c, 0xd8, 0x96, 0xa8, 0x00, 0x00,
- 0x93, 0x6a, 0x55, 0x50, 0x00, 0x00,
- 0x1a, 0xb4, 0x69, 0xa8, 0x00, 0x00,
- 0x47, 0x2d, 0x0f, 0x50, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_5[30] =
- {
- 0x64, 0xd9, 0x92, 0x88, 0x00, 0x00,
- 0xa5, 0x68, 0x95, 0x50, 0x00, 0x00,
- 0x52, 0xb5, 0x25, 0xa0, 0x00, 0x00,
- 0x1d, 0xa9, 0x4e, 0x40, 0x00, 0x00,
- 0x9c, 0x56, 0x38, 0xc0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_6[36] =
- {
- 0x4a, 0x55, 0x8a, 0x28, 0x00, 0x00,
- 0x95, 0x48, 0x55, 0x50, 0x00, 0x00,
- 0x14, 0xb5, 0x31, 0x18, 0x00, 0x00,
- 0x51, 0xa9, 0x4a, 0x50, 0x00, 0x00,
- 0x22, 0x6c, 0x8d, 0x90, 0x00, 0x00,
- 0x88, 0x8e, 0x29, 0x60, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_7[42] =
- {
- 0x62, 0x55, 0x8a, 0x88, 0x00, 0x00,
- 0xb9, 0x22, 0xc4, 0x50, 0x00, 0x00,
- 0x18, 0xb4, 0x61, 0xa8, 0x00, 0x00,
- 0x54, 0x99, 0x13, 0x50, 0x00, 0x00,
- 0x06, 0x6c, 0x4d, 0x90, 0x00, 0x00,
- 0x85, 0x55, 0x24, 0x68, 0x00, 0x00,
- 0xaa, 0x8a, 0x1a, 0x30, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_8[48] =
- {
- 0xc0, 0x16, 0x40, 0x88, 0x00, 0x00,
- 0x41, 0x60, 0x25, 0x40, 0x00, 0x00,
- 0x88, 0x30, 0x01, 0xa8, 0x00, 0x00,
- 0x20, 0xa4, 0x80, 0xd0, 0x00, 0x00,
- 0x0a, 0x48, 0x51, 0x10, 0x00, 0x00,
- 0x04, 0x9b, 0x08, 0x40, 0x00, 0x00,
- 0x94, 0x40, 0x03, 0x18, 0x00, 0x00,
- 0x72, 0x01, 0x96, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask29_9[54] =
- {
- 0xa2, 0x55, 0x88, 0x88, 0x00, 0x00,
- 0x34, 0x60, 0x91, 0x10, 0x00, 0x00,
- 0x4a, 0x27, 0x01, 0x40, 0x00, 0x00,
- 0x20, 0xa8, 0x0c, 0x30, 0x00, 0x00,
- 0x11, 0x84, 0x58, 0xa0, 0x00, 0x00,
- 0x49, 0x0a, 0x24, 0x00, 0x00, 0x00,
- 0x86, 0x0e, 0x0a, 0x40, 0x00, 0x00,
- 0x20, 0xd4, 0x22, 0x90, 0x00, 0x00,
- 0x88, 0x4a, 0x41, 0x20, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask2_1[2] =
- {
- 0xc0, 0x00
- };
-
- const WebRtc_UWord8 mask2_2[4] =
- {
- 0xc0, 0x00,
- 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask30_1[6] =
- {
- 0xff, 0xff, 0xff, 0xfc, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_10[60] =
- {
- 0xc0, 0xa1, 0x81, 0x40, 0x00, 0x00,
- 0x15, 0x56, 0x2a, 0xac, 0x00, 0x00,
- 0x74, 0x40, 0xe8, 0x80, 0x00, 0x00,
- 0x00, 0x9c, 0x01, 0x38, 0x00, 0x00,
- 0x01, 0x2c, 0x02, 0x58, 0x00, 0x00,
- 0x44, 0x92, 0x89, 0x24, 0x00, 0x00,
- 0x88, 0x51, 0x10, 0xa0, 0x00, 0x00,
- 0x20, 0xa4, 0x41, 0x48, 0x00, 0x00,
- 0xaa, 0x05, 0x54, 0x08, 0x00, 0x00,
- 0x02, 0x62, 0x04, 0xc4, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_11[66] =
- {
- 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
- 0xf1, 0x11, 0xe2, 0x20, 0x00, 0x00,
- 0x10, 0x0e, 0x20, 0x1c, 0x00, 0x00,
- 0x10, 0xb0, 0x21, 0x60, 0x00, 0x00,
- 0x24, 0x24, 0x48, 0x48, 0x00, 0x00,
- 0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
- 0x00, 0xc4, 0x01, 0x88, 0x00, 0x00,
- 0x04, 0xa2, 0x09, 0x44, 0x00, 0x00,
- 0x02, 0x58, 0x04, 0xb0, 0x00, 0x00,
- 0x2b, 0x00, 0x56, 0x00, 0x00, 0x00,
- 0x98, 0x41, 0x30, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_12[72] =
- {
- 0x88, 0x91, 0x11, 0x20, 0x00, 0x00,
- 0x40, 0x54, 0x80, 0xa8, 0x00, 0x00,
- 0x82, 0x63, 0x04, 0xc4, 0x00, 0x00,
- 0x21, 0xa4, 0x43, 0x48, 0x00, 0x00,
- 0x10, 0x64, 0x20, 0xc8, 0x00, 0x00,
- 0x44, 0x0a, 0x88, 0x14, 0x00, 0x00,
- 0x10, 0xc8, 0x21, 0x90, 0x00, 0x00,
- 0x4d, 0x2a, 0x9a, 0x54, 0x00, 0x00,
- 0x38, 0x02, 0x70, 0x04, 0x00, 0x00,
- 0x17, 0x48, 0x2e, 0x90, 0x00, 0x00,
- 0x90, 0x85, 0x21, 0x08, 0x00, 0x00,
- 0x72, 0x14, 0xe4, 0x28, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_13[78] =
- {
- 0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
- 0x34, 0x44, 0x68, 0x88, 0x00, 0x00,
- 0x40, 0x4a, 0x80, 0x94, 0x00, 0x00,
- 0xc4, 0x05, 0x88, 0x08, 0x00, 0x00,
- 0x08, 0x60, 0x10, 0xc0, 0x00, 0x00,
- 0x94, 0x13, 0x28, 0x24, 0x00, 0x00,
- 0x88, 0xc1, 0x11, 0x80, 0x00, 0x00,
- 0x21, 0x32, 0x42, 0x64, 0x00, 0x00,
- 0xc1, 0x41, 0x82, 0x80, 0x00, 0x00,
- 0x10, 0x68, 0x20, 0xd0, 0x00, 0x00,
- 0x06, 0x90, 0x0d, 0x20, 0x00, 0x00,
- 0x59, 0x00, 0xb2, 0x00, 0x00, 0x00,
- 0x0a, 0x0c, 0x14, 0x18, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_14[84] =
- {
- 0x40, 0x82, 0x81, 0x04, 0x00, 0x00,
- 0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
- 0x88, 0x13, 0x10, 0x24, 0x00, 0x00,
- 0xc0, 0x11, 0x80, 0x20, 0x00, 0x00,
- 0x80, 0xa1, 0x01, 0x40, 0x00, 0x00,
- 0x01, 0x22, 0x02, 0x44, 0x00, 0x00,
- 0x40, 0x2c, 0x80, 0x58, 0x00, 0x00,
- 0x22, 0x02, 0x44, 0x04, 0x00, 0x00,
- 0x90, 0x05, 0x20, 0x08, 0x00, 0x00,
- 0x12, 0x40, 0x24, 0x80, 0x00, 0x00,
- 0x5d, 0x00, 0xba, 0x00, 0x00, 0x00,
- 0x20, 0x54, 0x40, 0xa8, 0x00, 0x00,
- 0x86, 0x09, 0x0c, 0x10, 0x00, 0x00,
- 0x28, 0x88, 0x51, 0x10, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_15[90] =
- {
- 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
- 0x31, 0x10, 0x62, 0x20, 0x00, 0x00,
- 0x58, 0x00, 0xb0, 0x00, 0x00, 0x00,
- 0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
- 0x88, 0x21, 0x10, 0x40, 0x00, 0x00,
- 0x44, 0x02, 0x88, 0x04, 0x00, 0x00,
- 0x29, 0x04, 0x52, 0x08, 0x00, 0x00,
- 0x82, 0xa1, 0x05, 0x40, 0x00, 0x00,
- 0x0a, 0x1a, 0x14, 0x34, 0x00, 0x00,
- 0x11, 0xe0, 0x23, 0xc0, 0x00, 0x00,
- 0x84, 0x05, 0x08, 0x08, 0x00, 0x00,
- 0x86, 0x41, 0x0c, 0x80, 0x00, 0x00,
- 0x00, 0x86, 0x01, 0x0c, 0x00, 0x00,
- 0x44, 0x48, 0x88, 0x90, 0x00, 0x00,
- 0x10, 0x98, 0x21, 0x30, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_16[96] =
- {
- 0x90, 0x23, 0x20, 0x44, 0x00, 0x00,
- 0x09, 0x50, 0x12, 0xa0, 0x00, 0x00,
- 0x00, 0x6a, 0x00, 0xd4, 0x00, 0x00,
- 0x20, 0x34, 0x40, 0x68, 0x00, 0x00,
- 0x14, 0x44, 0x28, 0x88, 0x00, 0x00,
- 0xc2, 0x11, 0x84, 0x20, 0x00, 0x00,
- 0x00, 0xc6, 0x01, 0x8c, 0x00, 0x00,
- 0x65, 0x80, 0xcb, 0x00, 0x00, 0x00,
- 0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
- 0xb1, 0x15, 0x62, 0x28, 0x00, 0x00,
- 0x18, 0x6a, 0x30, 0xd4, 0x00, 0x00,
- 0x44, 0xd4, 0x89, 0xa8, 0x00, 0x00,
- 0x13, 0x64, 0x26, 0xc8, 0x00, 0x00,
- 0x49, 0x1a, 0x92, 0x34, 0x00, 0x00,
- 0x86, 0x8d, 0x0d, 0x18, 0x00, 0x00,
- 0xce, 0x58, 0xa0, 0x14, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_17[102] =
- {
- 0x90, 0x23, 0x20, 0x44, 0x00, 0x00,
- 0x09, 0x50, 0x12, 0xa0, 0x00, 0x00,
- 0x00, 0x6a, 0x00, 0xd4, 0x00, 0x00,
- 0x20, 0x34, 0x40, 0x68, 0x00, 0x00,
- 0x14, 0x44, 0x28, 0x88, 0x00, 0x00,
- 0xc2, 0x11, 0x84, 0x20, 0x00, 0x00,
- 0x00, 0xc6, 0x01, 0x8c, 0x00, 0x00,
- 0x65, 0x80, 0xcb, 0x00, 0x00, 0x00,
- 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
- 0x24, 0x44, 0x48, 0x88, 0x00, 0x00,
- 0xc0, 0x51, 0x80, 0xa0, 0x00, 0x00,
- 0x03, 0x0c, 0x06, 0x18, 0x00, 0x00,
- 0x16, 0x28, 0x2c, 0x50, 0x00, 0x00,
- 0x89, 0x01, 0x12, 0x00, 0x00, 0x00,
- 0x82, 0x91, 0x05, 0x20, 0x00, 0x00,
- 0x08, 0xa4, 0x11, 0x48, 0x00, 0x00,
- 0x90, 0x49, 0x20, 0x90, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_18[108] =
- {
- 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
- 0x24, 0x44, 0x48, 0x88, 0x00, 0x00,
- 0xc0, 0x51, 0x80, 0xa0, 0x00, 0x00,
- 0x03, 0x0c, 0x06, 0x18, 0x00, 0x00,
- 0x16, 0x28, 0x2c, 0x50, 0x00, 0x00,
- 0x89, 0x01, 0x12, 0x00, 0x00, 0x00,
- 0x82, 0x91, 0x05, 0x20, 0x00, 0x00,
- 0x08, 0xa4, 0x11, 0x48, 0x00, 0x00,
- 0x90, 0x49, 0x20, 0x90, 0x00, 0x00,
- 0x90, 0x23, 0x20, 0x44, 0x00, 0x00,
- 0x09, 0x50, 0x12, 0xa0, 0x00, 0x00,
- 0x00, 0x6a, 0x00, 0xd4, 0x00, 0x00,
- 0x20, 0x34, 0x40, 0x68, 0x00, 0x00,
- 0x14, 0x44, 0x28, 0x88, 0x00, 0x00,
- 0xc2, 0x11, 0x84, 0x20, 0x00, 0x00,
- 0x00, 0xc6, 0x01, 0x8c, 0x00, 0x00,
- 0x65, 0x80, 0xcb, 0x00, 0x00, 0x00,
- 0x00, 0xb2, 0x47, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_19[114] =
- {
- 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
- 0x24, 0x44, 0x48, 0x88, 0x00, 0x00,
- 0xc0, 0x51, 0x80, 0xa0, 0x00, 0x00,
- 0x03, 0x0c, 0x06, 0x18, 0x00, 0x00,
- 0x16, 0x28, 0x2c, 0x50, 0x00, 0x00,
- 0x89, 0x01, 0x12, 0x00, 0x00, 0x00,
- 0x82, 0x91, 0x05, 0x20, 0x00, 0x00,
- 0x08, 0xa4, 0x11, 0x48, 0x00, 0x00,
- 0x90, 0x49, 0x20, 0x90, 0x00, 0x00,
- 0xc0, 0xa1, 0x81, 0x40, 0x00, 0x00,
- 0x15, 0x56, 0x2a, 0xac, 0x00, 0x00,
- 0x74, 0x40, 0xe8, 0x80, 0x00, 0x00,
- 0x00, 0x9c, 0x01, 0x38, 0x00, 0x00,
- 0x01, 0x2c, 0x02, 0x58, 0x00, 0x00,
- 0x44, 0x92, 0x89, 0x24, 0x00, 0x00,
- 0x88, 0x51, 0x10, 0xa0, 0x00, 0x00,
- 0x20, 0xa4, 0x41, 0x48, 0x00, 0x00,
- 0xaa, 0x05, 0x54, 0x08, 0x00, 0x00,
- 0x02, 0x62, 0x04, 0xc4, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_2[12] =
- {
- 0xec, 0xeb, 0xd9, 0xd4, 0x00, 0x00,
- 0xbb, 0x9d, 0x77, 0x38, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_20[120] =
- {
- 0xc0, 0xa1, 0x81, 0x40, 0x00, 0x00,
- 0x15, 0x56, 0x2a, 0xac, 0x00, 0x00,
- 0x74, 0x40, 0xe8, 0x80, 0x00, 0x00,
- 0x00, 0x9c, 0x01, 0x38, 0x00, 0x00,
- 0x01, 0x2c, 0x02, 0x58, 0x00, 0x00,
- 0x44, 0x92, 0x89, 0x24, 0x00, 0x00,
- 0x88, 0x51, 0x10, 0xa0, 0x00, 0x00,
- 0x20, 0xa4, 0x41, 0x48, 0x00, 0x00,
- 0xaa, 0x05, 0x54, 0x08, 0x00, 0x00,
- 0x02, 0x62, 0x04, 0xc4, 0x00, 0x00,
- 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
- 0x24, 0x44, 0x48, 0x88, 0x00, 0x00,
- 0xc0, 0x51, 0x80, 0xa0, 0x00, 0x00,
- 0x03, 0x0c, 0x06, 0x18, 0x00, 0x00,
- 0x16, 0x28, 0x2c, 0x50, 0x00, 0x00,
- 0x89, 0x01, 0x12, 0x00, 0x00, 0x00,
- 0x82, 0x91, 0x05, 0x20, 0x00, 0x00,
- 0x08, 0xa4, 0x11, 0x48, 0x00, 0x00,
- 0x90, 0x49, 0x20, 0x90, 0x00, 0x00,
- 0x51, 0x88, 0xd1, 0x78, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_21[126] =
- {
- 0xc0, 0xa1, 0x81, 0x40, 0x00, 0x00,
- 0x15, 0x56, 0x2a, 0xac, 0x00, 0x00,
- 0x74, 0x40, 0xe8, 0x80, 0x00, 0x00,
- 0x00, 0x9c, 0x01, 0x38, 0x00, 0x00,
- 0x01, 0x2c, 0x02, 0x58, 0x00, 0x00,
- 0x44, 0x92, 0x89, 0x24, 0x00, 0x00,
- 0x88, 0x51, 0x10, 0xa0, 0x00, 0x00,
- 0x20, 0xa4, 0x41, 0x48, 0x00, 0x00,
- 0xaa, 0x05, 0x54, 0x08, 0x00, 0x00,
- 0x02, 0x62, 0x04, 0xc4, 0x00, 0x00,
- 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
- 0xf1, 0x11, 0xe2, 0x20, 0x00, 0x00,
- 0x10, 0x0e, 0x20, 0x1c, 0x00, 0x00,
- 0x10, 0xb0, 0x21, 0x60, 0x00, 0x00,
- 0x24, 0x24, 0x48, 0x48, 0x00, 0x00,
- 0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
- 0x00, 0xc4, 0x01, 0x88, 0x00, 0x00,
- 0x04, 0xa2, 0x09, 0x44, 0x00, 0x00,
- 0x02, 0x58, 0x04, 0xb0, 0x00, 0x00,
- 0x2b, 0x00, 0x56, 0x00, 0x00, 0x00,
- 0x98, 0x41, 0x30, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_22[132] =
- {
- 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
- 0xf1, 0x11, 0xe2, 0x20, 0x00, 0x00,
- 0x10, 0x0e, 0x20, 0x1c, 0x00, 0x00,
- 0x10, 0xb0, 0x21, 0x60, 0x00, 0x00,
- 0x24, 0x24, 0x48, 0x48, 0x00, 0x00,
- 0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
- 0x00, 0xc4, 0x01, 0x88, 0x00, 0x00,
- 0x04, 0xa2, 0x09, 0x44, 0x00, 0x00,
- 0x02, 0x58, 0x04, 0xb0, 0x00, 0x00,
- 0x2b, 0x00, 0x56, 0x00, 0x00, 0x00,
- 0x98, 0x41, 0x30, 0x80, 0x00, 0x00,
- 0xc0, 0xa1, 0x81, 0x40, 0x00, 0x00,
- 0x15, 0x56, 0x2a, 0xac, 0x00, 0x00,
- 0x74, 0x40, 0xe8, 0x80, 0x00, 0x00,
- 0x00, 0x9c, 0x01, 0x38, 0x00, 0x00,
- 0x01, 0x2c, 0x02, 0x58, 0x00, 0x00,
- 0x44, 0x92, 0x89, 0x24, 0x00, 0x00,
- 0x88, 0x51, 0x10, 0xa0, 0x00, 0x00,
- 0x20, 0xa4, 0x41, 0x48, 0x00, 0x00,
- 0xaa, 0x05, 0x54, 0x08, 0x00, 0x00,
- 0x02, 0x62, 0x04, 0xc4, 0x00, 0x00,
- 0x03, 0x10, 0x18, 0x74, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_23[138] =
- {
- 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
- 0xf1, 0x11, 0xe2, 0x20, 0x00, 0x00,
- 0x10, 0x0e, 0x20, 0x1c, 0x00, 0x00,
- 0x10, 0xb0, 0x21, 0x60, 0x00, 0x00,
- 0x24, 0x24, 0x48, 0x48, 0x00, 0x00,
- 0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
- 0x00, 0xc4, 0x01, 0x88, 0x00, 0x00,
- 0x04, 0xa2, 0x09, 0x44, 0x00, 0x00,
- 0x02, 0x58, 0x04, 0xb0, 0x00, 0x00,
- 0x2b, 0x00, 0x56, 0x00, 0x00, 0x00,
- 0x98, 0x41, 0x30, 0x80, 0x00, 0x00,
- 0x88, 0x91, 0x11, 0x20, 0x00, 0x00,
- 0x40, 0x54, 0x80, 0xa8, 0x00, 0x00,
- 0x82, 0x63, 0x04, 0xc4, 0x00, 0x00,
- 0x21, 0xa4, 0x43, 0x48, 0x00, 0x00,
- 0x10, 0x64, 0x20, 0xc8, 0x00, 0x00,
- 0x44, 0x0a, 0x88, 0x14, 0x00, 0x00,
- 0x10, 0xc8, 0x21, 0x90, 0x00, 0x00,
- 0x4d, 0x2a, 0x9a, 0x54, 0x00, 0x00,
- 0x38, 0x02, 0x70, 0x04, 0x00, 0x00,
- 0x17, 0x48, 0x2e, 0x90, 0x00, 0x00,
- 0x90, 0x85, 0x21, 0x08, 0x00, 0x00,
- 0x72, 0x14, 0xe4, 0x28, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_24[144] =
- {
- 0x88, 0x91, 0x11, 0x20, 0x00, 0x00,
- 0x40, 0x54, 0x80, 0xa8, 0x00, 0x00,
- 0x82, 0x63, 0x04, 0xc4, 0x00, 0x00,
- 0x21, 0xa4, 0x43, 0x48, 0x00, 0x00,
- 0x10, 0x64, 0x20, 0xc8, 0x00, 0x00,
- 0x44, 0x0a, 0x88, 0x14, 0x00, 0x00,
- 0x10, 0xc8, 0x21, 0x90, 0x00, 0x00,
- 0x4d, 0x2a, 0x9a, 0x54, 0x00, 0x00,
- 0x38, 0x02, 0x70, 0x04, 0x00, 0x00,
- 0x17, 0x48, 0x2e, 0x90, 0x00, 0x00,
- 0x90, 0x85, 0x21, 0x08, 0x00, 0x00,
- 0x72, 0x14, 0xe4, 0x28, 0x00, 0x00,
- 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
- 0xf1, 0x11, 0xe2, 0x20, 0x00, 0x00,
- 0x10, 0x0e, 0x20, 0x1c, 0x00, 0x00,
- 0x10, 0xb0, 0x21, 0x60, 0x00, 0x00,
- 0x24, 0x24, 0x48, 0x48, 0x00, 0x00,
- 0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
- 0x00, 0xc4, 0x01, 0x88, 0x00, 0x00,
- 0x04, 0xa2, 0x09, 0x44, 0x00, 0x00,
- 0x02, 0x58, 0x04, 0xb0, 0x00, 0x00,
- 0x2b, 0x00, 0x56, 0x00, 0x00, 0x00,
- 0x98, 0x41, 0x30, 0x80, 0x00, 0x00,
- 0xf3, 0x4d, 0x1c, 0x70, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_25[150] =
- {
- 0x88, 0x91, 0x11, 0x20, 0x00, 0x00,
- 0x40, 0x54, 0x80, 0xa8, 0x00, 0x00,
- 0x82, 0x63, 0x04, 0xc4, 0x00, 0x00,
- 0x21, 0xa4, 0x43, 0x48, 0x00, 0x00,
- 0x10, 0x64, 0x20, 0xc8, 0x00, 0x00,
- 0x44, 0x0a, 0x88, 0x14, 0x00, 0x00,
- 0x10, 0xc8, 0x21, 0x90, 0x00, 0x00,
- 0x4d, 0x2a, 0x9a, 0x54, 0x00, 0x00,
- 0x38, 0x02, 0x70, 0x04, 0x00, 0x00,
- 0x17, 0x48, 0x2e, 0x90, 0x00, 0x00,
- 0x90, 0x85, 0x21, 0x08, 0x00, 0x00,
- 0x72, 0x14, 0xe4, 0x28, 0x00, 0x00,
- 0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
- 0x34, 0x44, 0x68, 0x88, 0x00, 0x00,
- 0x40, 0x4a, 0x80, 0x94, 0x00, 0x00,
- 0xc4, 0x05, 0x88, 0x08, 0x00, 0x00,
- 0x08, 0x60, 0x10, 0xc0, 0x00, 0x00,
- 0x94, 0x13, 0x28, 0x24, 0x00, 0x00,
- 0x88, 0xc1, 0x11, 0x80, 0x00, 0x00,
- 0x21, 0x32, 0x42, 0x64, 0x00, 0x00,
- 0xc1, 0x41, 0x82, 0x80, 0x00, 0x00,
- 0x10, 0x68, 0x20, 0xd0, 0x00, 0x00,
- 0x06, 0x90, 0x0d, 0x20, 0x00, 0x00,
- 0x59, 0x00, 0xb2, 0x00, 0x00, 0x00,
- 0x0a, 0x0c, 0x14, 0x18, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_26[156] =
- {
- 0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
- 0x34, 0x44, 0x68, 0x88, 0x00, 0x00,
- 0x40, 0x4a, 0x80, 0x94, 0x00, 0x00,
- 0xc4, 0x05, 0x88, 0x08, 0x00, 0x00,
- 0x08, 0x60, 0x10, 0xc0, 0x00, 0x00,
- 0x94, 0x13, 0x28, 0x24, 0x00, 0x00,
- 0x88, 0xc1, 0x11, 0x80, 0x00, 0x00,
- 0x21, 0x32, 0x42, 0x64, 0x00, 0x00,
- 0xc1, 0x41, 0x82, 0x80, 0x00, 0x00,
- 0x10, 0x68, 0x20, 0xd0, 0x00, 0x00,
- 0x06, 0x90, 0x0d, 0x20, 0x00, 0x00,
- 0x59, 0x00, 0xb2, 0x00, 0x00, 0x00,
- 0x0a, 0x0c, 0x14, 0x18, 0x00, 0x00,
- 0x88, 0x91, 0x11, 0x20, 0x00, 0x00,
- 0x40, 0x54, 0x80, 0xa8, 0x00, 0x00,
- 0x82, 0x63, 0x04, 0xc4, 0x00, 0x00,
- 0x21, 0xa4, 0x43, 0x48, 0x00, 0x00,
- 0x10, 0x64, 0x20, 0xc8, 0x00, 0x00,
- 0x44, 0x0a, 0x88, 0x14, 0x00, 0x00,
- 0x10, 0xc8, 0x21, 0x90, 0x00, 0x00,
- 0x4d, 0x2a, 0x9a, 0x54, 0x00, 0x00,
- 0x38, 0x02, 0x70, 0x04, 0x00, 0x00,
- 0x17, 0x48, 0x2e, 0x90, 0x00, 0x00,
- 0x90, 0x85, 0x21, 0x08, 0x00, 0x00,
- 0x72, 0x14, 0xe4, 0x28, 0x00, 0x00,
- 0x83, 0x11, 0xad, 0xe8, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_27[162] =
- {
- 0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
- 0x34, 0x44, 0x68, 0x88, 0x00, 0x00,
- 0x40, 0x4a, 0x80, 0x94, 0x00, 0x00,
- 0xc4, 0x05, 0x88, 0x08, 0x00, 0x00,
- 0x08, 0x60, 0x10, 0xc0, 0x00, 0x00,
- 0x94, 0x13, 0x28, 0x24, 0x00, 0x00,
- 0x88, 0xc1, 0x11, 0x80, 0x00, 0x00,
- 0x21, 0x32, 0x42, 0x64, 0x00, 0x00,
- 0xc1, 0x41, 0x82, 0x80, 0x00, 0x00,
- 0x10, 0x68, 0x20, 0xd0, 0x00, 0x00,
- 0x06, 0x90, 0x0d, 0x20, 0x00, 0x00,
- 0x59, 0x00, 0xb2, 0x00, 0x00, 0x00,
- 0x0a, 0x0c, 0x14, 0x18, 0x00, 0x00,
- 0x40, 0x82, 0x81, 0x04, 0x00, 0x00,
- 0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
- 0x88, 0x13, 0x10, 0x24, 0x00, 0x00,
- 0xc0, 0x11, 0x80, 0x20, 0x00, 0x00,
- 0x80, 0xa1, 0x01, 0x40, 0x00, 0x00,
- 0x01, 0x22, 0x02, 0x44, 0x00, 0x00,
- 0x40, 0x2c, 0x80, 0x58, 0x00, 0x00,
- 0x22, 0x02, 0x44, 0x04, 0x00, 0x00,
- 0x90, 0x05, 0x20, 0x08, 0x00, 0x00,
- 0x12, 0x40, 0x24, 0x80, 0x00, 0x00,
- 0x5d, 0x00, 0xba, 0x00, 0x00, 0x00,
- 0x20, 0x54, 0x40, 0xa8, 0x00, 0x00,
- 0x86, 0x09, 0x0c, 0x10, 0x00, 0x00,
- 0x28, 0x88, 0x51, 0x10, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_28[168] =
- {
- 0x40, 0x82, 0x81, 0x04, 0x00, 0x00,
- 0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
- 0x88, 0x13, 0x10, 0x24, 0x00, 0x00,
- 0xc0, 0x11, 0x80, 0x20, 0x00, 0x00,
- 0x80, 0xa1, 0x01, 0x40, 0x00, 0x00,
- 0x01, 0x22, 0x02, 0x44, 0x00, 0x00,
- 0x40, 0x2c, 0x80, 0x58, 0x00, 0x00,
- 0x22, 0x02, 0x44, 0x04, 0x00, 0x00,
- 0x90, 0x05, 0x20, 0x08, 0x00, 0x00,
- 0x12, 0x40, 0x24, 0x80, 0x00, 0x00,
- 0x5d, 0x00, 0xba, 0x00, 0x00, 0x00,
- 0x20, 0x54, 0x40, 0xa8, 0x00, 0x00,
- 0x86, 0x09, 0x0c, 0x10, 0x00, 0x00,
- 0x28, 0x88, 0x51, 0x10, 0x00, 0x00,
- 0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
- 0x34, 0x44, 0x68, 0x88, 0x00, 0x00,
- 0x40, 0x4a, 0x80, 0x94, 0x00, 0x00,
- 0xc4, 0x05, 0x88, 0x08, 0x00, 0x00,
- 0x08, 0x60, 0x10, 0xc0, 0x00, 0x00,
- 0x94, 0x13, 0x28, 0x24, 0x00, 0x00,
- 0x88, 0xc1, 0x11, 0x80, 0x00, 0x00,
- 0x21, 0x32, 0x42, 0x64, 0x00, 0x00,
- 0xc1, 0x41, 0x82, 0x80, 0x00, 0x00,
- 0x10, 0x68, 0x20, 0xd0, 0x00, 0x00,
- 0x06, 0x90, 0x0d, 0x20, 0x00, 0x00,
- 0x59, 0x00, 0xb2, 0x00, 0x00, 0x00,
- 0x0a, 0x0c, 0x14, 0x18, 0x00, 0x00,
- 0x94, 0x59, 0x03, 0x18, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_29[174] =
- {
- 0x40, 0x82, 0x81, 0x04, 0x00, 0x00,
- 0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
- 0x88, 0x13, 0x10, 0x24, 0x00, 0x00,
- 0xc0, 0x11, 0x80, 0x20, 0x00, 0x00,
- 0x80, 0xa1, 0x01, 0x40, 0x00, 0x00,
- 0x01, 0x22, 0x02, 0x44, 0x00, 0x00,
- 0x40, 0x2c, 0x80, 0x58, 0x00, 0x00,
- 0x22, 0x02, 0x44, 0x04, 0x00, 0x00,
- 0x90, 0x05, 0x20, 0x08, 0x00, 0x00,
- 0x12, 0x40, 0x24, 0x80, 0x00, 0x00,
- 0x5d, 0x00, 0xba, 0x00, 0x00, 0x00,
- 0x20, 0x54, 0x40, 0xa8, 0x00, 0x00,
- 0x86, 0x09, 0x0c, 0x10, 0x00, 0x00,
- 0x28, 0x88, 0x51, 0x10, 0x00, 0x00,
- 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
- 0x31, 0x10, 0x62, 0x20, 0x00, 0x00,
- 0x58, 0x00, 0xb0, 0x00, 0x00, 0x00,
- 0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
- 0x88, 0x21, 0x10, 0x40, 0x00, 0x00,
- 0x44, 0x02, 0x88, 0x04, 0x00, 0x00,
- 0x29, 0x04, 0x52, 0x08, 0x00, 0x00,
- 0x82, 0xa1, 0x05, 0x40, 0x00, 0x00,
- 0x0a, 0x1a, 0x14, 0x34, 0x00, 0x00,
- 0x11, 0xe0, 0x23, 0xc0, 0x00, 0x00,
- 0x84, 0x05, 0x08, 0x08, 0x00, 0x00,
- 0x86, 0x41, 0x0c, 0x80, 0x00, 0x00,
- 0x00, 0x86, 0x01, 0x0c, 0x00, 0x00,
- 0x44, 0x48, 0x88, 0x90, 0x00, 0x00,
- 0x10, 0x98, 0x21, 0x30, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_3[18] =
- {
- 0xac, 0x93, 0x59, 0x24, 0x00, 0x00,
- 0x55, 0x4a, 0xaa, 0x94, 0x00, 0x00,
- 0x43, 0x36, 0x86, 0x6c, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_30[180] =
- {
- 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
- 0x31, 0x10, 0x62, 0x20, 0x00, 0x00,
- 0x58, 0x00, 0xb0, 0x00, 0x00, 0x00,
- 0x01, 0x12, 0x02, 0x24, 0x00, 0x00,
- 0x88, 0x21, 0x10, 0x40, 0x00, 0x00,
- 0x44, 0x02, 0x88, 0x04, 0x00, 0x00,
- 0x29, 0x04, 0x52, 0x08, 0x00, 0x00,
- 0x82, 0xa1, 0x05, 0x40, 0x00, 0x00,
- 0x0a, 0x1a, 0x14, 0x34, 0x00, 0x00,
- 0x11, 0xe0, 0x23, 0xc0, 0x00, 0x00,
- 0x84, 0x05, 0x08, 0x08, 0x00, 0x00,
- 0x86, 0x41, 0x0c, 0x80, 0x00, 0x00,
- 0x00, 0x86, 0x01, 0x0c, 0x00, 0x00,
- 0x44, 0x48, 0x88, 0x90, 0x00, 0x00,
- 0x10, 0x98, 0x21, 0x30, 0x00, 0x00,
- 0x40, 0x82, 0x81, 0x04, 0x00, 0x00,
- 0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
- 0x88, 0x13, 0x10, 0x24, 0x00, 0x00,
- 0xc0, 0x11, 0x80, 0x20, 0x00, 0x00,
- 0x80, 0xa1, 0x01, 0x40, 0x00, 0x00,
- 0x01, 0x22, 0x02, 0x44, 0x00, 0x00,
- 0x40, 0x2c, 0x80, 0x58, 0x00, 0x00,
- 0x22, 0x02, 0x44, 0x04, 0x00, 0x00,
- 0x90, 0x05, 0x20, 0x08, 0x00, 0x00,
- 0x12, 0x40, 0x24, 0x80, 0x00, 0x00,
- 0x5d, 0x00, 0xba, 0x00, 0x00, 0x00,
- 0x20, 0x54, 0x40, 0xa8, 0x00, 0x00,
- 0x86, 0x09, 0x0c, 0x10, 0x00, 0x00,
- 0x28, 0x88, 0x51, 0x10, 0x00, 0x00,
- 0x46, 0xf1, 0xef, 0xec, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_4[24] =
- {
- 0x25, 0xaa, 0x4b, 0x54, 0x00, 0x00,
- 0x95, 0x55, 0x2a, 0xa8, 0x00, 0x00,
- 0x1a, 0x6a, 0x34, 0xd4, 0x00, 0x00,
- 0x43, 0xd4, 0x87, 0xa8, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_5[30] =
- {
- 0x64, 0xa2, 0xc9, 0x44, 0x00, 0x00,
- 0x25, 0x54, 0x4a, 0xa8, 0x00, 0x00,
- 0x49, 0x68, 0x92, 0xd0, 0x00, 0x00,
- 0x53, 0x90, 0xa7, 0x20, 0x00, 0x00,
- 0x8e, 0x31, 0x1c, 0x60, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_6[36] =
- {
- 0x62, 0x8a, 0xc5, 0x14, 0x00, 0x00,
- 0x15, 0x54, 0x2a, 0xa8, 0x00, 0x00,
- 0x4c, 0x46, 0x98, 0x8c, 0x00, 0x00,
- 0x52, 0x94, 0xa5, 0x28, 0x00, 0x00,
- 0x23, 0x64, 0x46, 0xc8, 0x00, 0x00,
- 0x8a, 0x59, 0x14, 0xb0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_7[42] =
- {
- 0x62, 0xa2, 0xc5, 0x44, 0x00, 0x00,
- 0xb1, 0x15, 0x62, 0x28, 0x00, 0x00,
- 0x18, 0x6a, 0x30, 0xd4, 0x00, 0x00,
- 0x44, 0xd4, 0x89, 0xa8, 0x00, 0x00,
- 0x13, 0x64, 0x26, 0xc8, 0x00, 0x00,
- 0x49, 0x1a, 0x92, 0x34, 0x00, 0x00,
- 0x86, 0x8d, 0x0d, 0x18, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_8[48] =
- {
- 0x90, 0x23, 0x20, 0x44, 0x00, 0x00,
- 0x09, 0x50, 0x12, 0xa0, 0x00, 0x00,
- 0x00, 0x6a, 0x00, 0xd4, 0x00, 0x00,
- 0x20, 0x34, 0x40, 0x68, 0x00, 0x00,
- 0x14, 0x44, 0x28, 0x88, 0x00, 0x00,
- 0xc2, 0x11, 0x84, 0x20, 0x00, 0x00,
- 0x00, 0xc6, 0x01, 0x8c, 0x00, 0x00,
- 0x65, 0x80, 0xcb, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask30_9[54] =
- {
- 0x62, 0x22, 0xc4, 0x44, 0x00, 0x00,
- 0x24, 0x44, 0x48, 0x88, 0x00, 0x00,
- 0xc0, 0x51, 0x80, 0xa0, 0x00, 0x00,
- 0x03, 0x0c, 0x06, 0x18, 0x00, 0x00,
- 0x16, 0x28, 0x2c, 0x50, 0x00, 0x00,
- 0x89, 0x01, 0x12, 0x00, 0x00, 0x00,
- 0x82, 0x91, 0x05, 0x20, 0x00, 0x00,
- 0x08, 0xa4, 0x11, 0x48, 0x00, 0x00,
- 0x90, 0x49, 0x20, 0x90, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_1[6] =
- {
- 0xff, 0xff, 0xff, 0xfe, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_10[60] =
- {
- 0xc0, 0xa0, 0x8a, 0xa2, 0x00, 0x00,
- 0x15, 0x56, 0x21, 0x44, 0x00, 0x00,
- 0x74, 0x40, 0x02, 0x4a, 0x00, 0x00,
- 0x00, 0x9c, 0x16, 0x84, 0x00, 0x00,
- 0x01, 0x2d, 0xb0, 0x40, 0x00, 0x00,
- 0x44, 0x93, 0x05, 0x18, 0x00, 0x00,
- 0x88, 0x50, 0x48, 0x94, 0x00, 0x00,
- 0x20, 0xa4, 0x70, 0x30, 0x00, 0x00,
- 0xaa, 0x04, 0x54, 0x4a, 0x00, 0x00,
- 0x02, 0x63, 0x09, 0x24, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_11[66] =
- {
- 0x62, 0x22, 0xaa, 0xaa, 0x00, 0x00,
- 0xf1, 0x10, 0x54, 0x44, 0x00, 0x00,
- 0x10, 0x0e, 0x62, 0x22, 0x00, 0x00,
- 0x10, 0xb1, 0x06, 0x84, 0x00, 0x00,
- 0x24, 0x24, 0x0d, 0x30, 0x00, 0x00,
- 0x01, 0x12, 0x81, 0xc2, 0x00, 0x00,
- 0x00, 0xc4, 0x58, 0x88, 0x00, 0x00,
- 0x04, 0xa3, 0xb0, 0x50, 0x00, 0x00,
- 0x02, 0x59, 0x25, 0x02, 0x00, 0x00,
- 0x2b, 0x01, 0x08, 0x64, 0x00, 0x00,
- 0x98, 0x40, 0xd0, 0x18, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_12[72] =
- {
- 0x88, 0x91, 0x08, 0x62, 0x00, 0x00,
- 0x40, 0x54, 0x31, 0x44, 0x00, 0x00,
- 0x82, 0x62, 0x9c, 0x02, 0x00, 0x00,
- 0x21, 0xa4, 0x89, 0x90, 0x00, 0x00,
- 0x10, 0x64, 0x1d, 0x20, 0x00, 0x00,
- 0x44, 0x0a, 0x41, 0x98, 0x00, 0x00,
- 0x10, 0xc9, 0x26, 0x80, 0x00, 0x00,
- 0x4d, 0x2a, 0x5a, 0x20, 0x00, 0x00,
- 0x38, 0x02, 0x62, 0x88, 0x00, 0x00,
- 0x17, 0x49, 0x80, 0x46, 0x00, 0x00,
- 0x90, 0x84, 0x22, 0x4a, 0x00, 0x00,
- 0x72, 0x15, 0xd1, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_13[78] =
- {
- 0x62, 0xa2, 0x8a, 0x2a, 0x00, 0x00,
- 0x34, 0x44, 0x44, 0x44, 0x00, 0x00,
- 0x40, 0x4b, 0x2c, 0x18, 0x00, 0x00,
- 0xc4, 0x04, 0x18, 0xa0, 0x00, 0x00,
- 0x08, 0x60, 0xc4, 0x08, 0x00, 0x00,
- 0x94, 0x12, 0x92, 0x0c, 0x00, 0x00,
- 0x88, 0xc0, 0x23, 0x04, 0x00, 0x00,
- 0x21, 0x32, 0x24, 0x70, 0x00, 0x00,
- 0xc1, 0x40, 0x80, 0xe2, 0x00, 0x00,
- 0x10, 0x69, 0x51, 0x14, 0x00, 0x00,
- 0x06, 0x90, 0x11, 0x42, 0x00, 0x00,
- 0x59, 0x01, 0x41, 0x80, 0x00, 0x00,
- 0x0a, 0x0d, 0x8a, 0x20, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_14[84] =
- {
- 0x40, 0x82, 0x8a, 0xa2, 0x00, 0x00,
- 0x15, 0x54, 0x44, 0x14, 0x00, 0x00,
- 0x88, 0x13, 0x09, 0xa0, 0x00, 0x00,
- 0xc0, 0x10, 0x19, 0x14, 0x00, 0x00,
- 0x80, 0xa0, 0x30, 0x0c, 0x00, 0x00,
- 0x01, 0x22, 0x60, 0x06, 0x00, 0x00,
- 0x40, 0x2c, 0xc2, 0x10, 0x00, 0x00,
- 0x22, 0x02, 0x80, 0x22, 0x00, 0x00,
- 0x90, 0x04, 0x20, 0x58, 0x00, 0x00,
- 0x12, 0x40, 0x12, 0xc0, 0x00, 0x00,
- 0x5d, 0x00, 0x01, 0x28, 0x00, 0x00,
- 0x20, 0x54, 0xa4, 0x80, 0x00, 0x00,
- 0x86, 0x09, 0x48, 0x48, 0x00, 0x00,
- 0x28, 0x89, 0x05, 0x10, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_15[90] =
- {
- 0x62, 0x22, 0xaa, 0x22, 0x00, 0x00,
- 0x31, 0x10, 0x44, 0x44, 0x00, 0x00,
- 0x58, 0x00, 0x22, 0x22, 0x00, 0x00,
- 0x01, 0x13, 0x00, 0x8a, 0x00, 0x00,
- 0x88, 0x20, 0x40, 0x34, 0x00, 0x00,
- 0x44, 0x02, 0x10, 0xd0, 0x00, 0x00,
- 0x29, 0x04, 0x45, 0x08, 0x00, 0x00,
- 0x82, 0xa0, 0x90, 0x12, 0x00, 0x00,
- 0x0a, 0x1a, 0x0e, 0x02, 0x00, 0x00,
- 0x11, 0xe1, 0x28, 0x40, 0x00, 0x00,
- 0x84, 0x05, 0x04, 0x0c, 0x00, 0x00,
- 0x86, 0x40, 0xc0, 0x90, 0x00, 0x00,
- 0x00, 0x87, 0x13, 0x00, 0x00, 0x00,
- 0x44, 0x48, 0x01, 0x1c, 0x00, 0x00,
- 0x10, 0x98, 0x30, 0x44, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_16[96] =
- {
- 0x90, 0x22, 0x40, 0xa8, 0x00, 0x00,
- 0x09, 0x50, 0x31, 0x10, 0x00, 0x00,
- 0x00, 0x6b, 0x08, 0x0e, 0x00, 0x00,
- 0x20, 0x34, 0xc0, 0x90, 0x00, 0x00,
- 0x14, 0x44, 0x25, 0x04, 0x00, 0x00,
- 0xc2, 0x11, 0x02, 0x82, 0x00, 0x00,
- 0x00, 0xc6, 0x80, 0xc4, 0x00, 0x00,
- 0x65, 0x80, 0x2c, 0x60, 0x00, 0x00,
- 0x62, 0xa2, 0x8a, 0xa2, 0x00, 0x00,
- 0xb1, 0x14, 0x44, 0x54, 0x00, 0x00,
- 0x18, 0x6b, 0x22, 0x22, 0x00, 0x00,
- 0x44, 0xd4, 0x5c, 0x10, 0x00, 0x00,
- 0x13, 0x64, 0x90, 0x68, 0x00, 0x00,
- 0x49, 0x1b, 0x20, 0x52, 0x00, 0x00,
- 0x86, 0x8c, 0x13, 0x0c, 0x00, 0x00,
- 0x8d, 0x94, 0xa9, 0xe0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_17[102] =
- {
- 0x90, 0x22, 0x40, 0xa8, 0x00, 0x00,
- 0x09, 0x50, 0x31, 0x10, 0x00, 0x00,
- 0x00, 0x6b, 0x08, 0x0e, 0x00, 0x00,
- 0x20, 0x34, 0xc0, 0x90, 0x00, 0x00,
- 0x14, 0x44, 0x25, 0x04, 0x00, 0x00,
- 0xc2, 0x11, 0x02, 0x82, 0x00, 0x00,
- 0x00, 0xc6, 0x80, 0xc4, 0x00, 0x00,
- 0x65, 0x80, 0x2c, 0x60, 0x00, 0x00,
- 0x62, 0x22, 0xaa, 0xa2, 0x00, 0x00,
- 0x24, 0x44, 0x44, 0x54, 0x00, 0x00,
- 0xc0, 0x50, 0x0b, 0x0a, 0x00, 0x00,
- 0x03, 0x0c, 0x12, 0x94, 0x00, 0x00,
- 0x16, 0x29, 0x08, 0x64, 0x00, 0x00,
- 0x89, 0x01, 0x80, 0x1a, 0x00, 0x00,
- 0x82, 0x90, 0x41, 0x4c, 0x00, 0x00,
- 0x08, 0xa4, 0x34, 0x12, 0x00, 0x00,
- 0x90, 0x48, 0x88, 0xc8, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_18[108] =
- {
- 0x62, 0x22, 0xaa, 0xa2, 0x00, 0x00,
- 0x24, 0x44, 0x44, 0x54, 0x00, 0x00,
- 0xc0, 0x50, 0x0b, 0x0a, 0x00, 0x00,
- 0x03, 0x0c, 0x12, 0x94, 0x00, 0x00,
- 0x16, 0x29, 0x08, 0x64, 0x00, 0x00,
- 0x89, 0x01, 0x80, 0x1a, 0x00, 0x00,
- 0x82, 0x90, 0x41, 0x4c, 0x00, 0x00,
- 0x08, 0xa4, 0x34, 0x12, 0x00, 0x00,
- 0x90, 0x48, 0x88, 0xc8, 0x00, 0x00,
- 0x90, 0x22, 0x40, 0xa8, 0x00, 0x00,
- 0x09, 0x50, 0x31, 0x10, 0x00, 0x00,
- 0x00, 0x6b, 0x08, 0x0e, 0x00, 0x00,
- 0x20, 0x34, 0xc0, 0x90, 0x00, 0x00,
- 0x14, 0x44, 0x25, 0x04, 0x00, 0x00,
- 0xc2, 0x11, 0x02, 0x82, 0x00, 0x00,
- 0x00, 0xc6, 0x80, 0xc4, 0x00, 0x00,
- 0x65, 0x80, 0x2c, 0x60, 0x00, 0x00,
- 0xe3, 0xd1, 0x2e, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_19[114] =
- {
- 0x62, 0x22, 0xaa, 0xa2, 0x00, 0x00,
- 0x24, 0x44, 0x44, 0x54, 0x00, 0x00,
- 0xc0, 0x50, 0x0b, 0x0a, 0x00, 0x00,
- 0x03, 0x0c, 0x12, 0x94, 0x00, 0x00,
- 0x16, 0x29, 0x08, 0x64, 0x00, 0x00,
- 0x89, 0x01, 0x80, 0x1a, 0x00, 0x00,
- 0x82, 0x90, 0x41, 0x4c, 0x00, 0x00,
- 0x08, 0xa4, 0x34, 0x12, 0x00, 0x00,
- 0x90, 0x48, 0x88, 0xc8, 0x00, 0x00,
- 0xc0, 0xa0, 0x8a, 0xa2, 0x00, 0x00,
- 0x15, 0x56, 0x21, 0x44, 0x00, 0x00,
- 0x74, 0x40, 0x02, 0x4a, 0x00, 0x00,
- 0x00, 0x9c, 0x16, 0x84, 0x00, 0x00,
- 0x01, 0x2d, 0xb0, 0x40, 0x00, 0x00,
- 0x44, 0x93, 0x05, 0x18, 0x00, 0x00,
- 0x88, 0x50, 0x48, 0x94, 0x00, 0x00,
- 0x20, 0xa4, 0x70, 0x30, 0x00, 0x00,
- 0xaa, 0x04, 0x54, 0x4a, 0x00, 0x00,
- 0x02, 0x63, 0x09, 0x24, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_2[12] =
- {
- 0xec, 0xeb, 0x5d, 0x5c, 0x00, 0x00,
- 0xbb, 0x9c, 0xf2, 0xf2, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_20[120] =
- {
- 0xc0, 0xa0, 0x8a, 0xa2, 0x00, 0x00,
- 0x15, 0x56, 0x21, 0x44, 0x00, 0x00,
- 0x74, 0x40, 0x02, 0x4a, 0x00, 0x00,
- 0x00, 0x9c, 0x16, 0x84, 0x00, 0x00,
- 0x01, 0x2d, 0xb0, 0x40, 0x00, 0x00,
- 0x44, 0x93, 0x05, 0x18, 0x00, 0x00,
- 0x88, 0x50, 0x48, 0x94, 0x00, 0x00,
- 0x20, 0xa4, 0x70, 0x30, 0x00, 0x00,
- 0xaa, 0x04, 0x54, 0x4a, 0x00, 0x00,
- 0x02, 0x63, 0x09, 0x24, 0x00, 0x00,
- 0x62, 0x22, 0xaa, 0xa2, 0x00, 0x00,
- 0x24, 0x44, 0x44, 0x54, 0x00, 0x00,
- 0xc0, 0x50, 0x0b, 0x0a, 0x00, 0x00,
- 0x03, 0x0c, 0x12, 0x94, 0x00, 0x00,
- 0x16, 0x29, 0x08, 0x64, 0x00, 0x00,
- 0x89, 0x01, 0x80, 0x1a, 0x00, 0x00,
- 0x82, 0x90, 0x41, 0x4c, 0x00, 0x00,
- 0x08, 0xa4, 0x34, 0x12, 0x00, 0x00,
- 0x90, 0x48, 0x88, 0xc8, 0x00, 0x00,
- 0x9a, 0xd4, 0x6a, 0x36, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_21[126] =
- {
- 0xc0, 0xa0, 0x8a, 0xa2, 0x00, 0x00,
- 0x15, 0x56, 0x21, 0x44, 0x00, 0x00,
- 0x74, 0x40, 0x02, 0x4a, 0x00, 0x00,
- 0x00, 0x9c, 0x16, 0x84, 0x00, 0x00,
- 0x01, 0x2d, 0xb0, 0x40, 0x00, 0x00,
- 0x44, 0x93, 0x05, 0x18, 0x00, 0x00,
- 0x88, 0x50, 0x48, 0x94, 0x00, 0x00,
- 0x20, 0xa4, 0x70, 0x30, 0x00, 0x00,
- 0xaa, 0x04, 0x54, 0x4a, 0x00, 0x00,
- 0x02, 0x63, 0x09, 0x24, 0x00, 0x00,
- 0x62, 0x22, 0xaa, 0xaa, 0x00, 0x00,
- 0xf1, 0x10, 0x54, 0x44, 0x00, 0x00,
- 0x10, 0x0e, 0x62, 0x22, 0x00, 0x00,
- 0x10, 0xb1, 0x06, 0x84, 0x00, 0x00,
- 0x24, 0x24, 0x0d, 0x30, 0x00, 0x00,
- 0x01, 0x12, 0x81, 0xc2, 0x00, 0x00,
- 0x00, 0xc4, 0x58, 0x88, 0x00, 0x00,
- 0x04, 0xa3, 0xb0, 0x50, 0x00, 0x00,
- 0x02, 0x59, 0x25, 0x02, 0x00, 0x00,
- 0x2b, 0x01, 0x08, 0x64, 0x00, 0x00,
- 0x98, 0x40, 0xd0, 0x18, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_22[132] =
- {
- 0x62, 0x22, 0xaa, 0xaa, 0x00, 0x00,
- 0xf1, 0x10, 0x54, 0x44, 0x00, 0x00,
- 0x10, 0x0e, 0x62, 0x22, 0x00, 0x00,
- 0x10, 0xb1, 0x06, 0x84, 0x00, 0x00,
- 0x24, 0x24, 0x0d, 0x30, 0x00, 0x00,
- 0x01, 0x12, 0x81, 0xc2, 0x00, 0x00,
- 0x00, 0xc4, 0x58, 0x88, 0x00, 0x00,
- 0x04, 0xa3, 0xb0, 0x50, 0x00, 0x00,
- 0x02, 0x59, 0x25, 0x02, 0x00, 0x00,
- 0x2b, 0x01, 0x08, 0x64, 0x00, 0x00,
- 0x98, 0x40, 0xd0, 0x18, 0x00, 0x00,
- 0xc0, 0xa0, 0x8a, 0xa2, 0x00, 0x00,
- 0x15, 0x56, 0x21, 0x44, 0x00, 0x00,
- 0x74, 0x40, 0x02, 0x4a, 0x00, 0x00,
- 0x00, 0x9c, 0x16, 0x84, 0x00, 0x00,
- 0x01, 0x2d, 0xb0, 0x40, 0x00, 0x00,
- 0x44, 0x93, 0x05, 0x18, 0x00, 0x00,
- 0x88, 0x50, 0x48, 0x94, 0x00, 0x00,
- 0x20, 0xa4, 0x70, 0x30, 0x00, 0x00,
- 0xaa, 0x04, 0x54, 0x4a, 0x00, 0x00,
- 0x02, 0x63, 0x09, 0x24, 0x00, 0x00,
- 0x32, 0x23, 0x73, 0x8e, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_23[138] =
- {
- 0x62, 0x22, 0xaa, 0xaa, 0x00, 0x00,
- 0xf1, 0x10, 0x54, 0x44, 0x00, 0x00,
- 0x10, 0x0e, 0x62, 0x22, 0x00, 0x00,
- 0x10, 0xb1, 0x06, 0x84, 0x00, 0x00,
- 0x24, 0x24, 0x0d, 0x30, 0x00, 0x00,
- 0x01, 0x12, 0x81, 0xc2, 0x00, 0x00,
- 0x00, 0xc4, 0x58, 0x88, 0x00, 0x00,
- 0x04, 0xa3, 0xb0, 0x50, 0x00, 0x00,
- 0x02, 0x59, 0x25, 0x02, 0x00, 0x00,
- 0x2b, 0x01, 0x08, 0x64, 0x00, 0x00,
- 0x98, 0x40, 0xd0, 0x18, 0x00, 0x00,
- 0x88, 0x91, 0x08, 0x62, 0x00, 0x00,
- 0x40, 0x54, 0x31, 0x44, 0x00, 0x00,
- 0x82, 0x62, 0x9c, 0x02, 0x00, 0x00,
- 0x21, 0xa4, 0x89, 0x90, 0x00, 0x00,
- 0x10, 0x64, 0x1d, 0x20, 0x00, 0x00,
- 0x44, 0x0a, 0x41, 0x98, 0x00, 0x00,
- 0x10, 0xc9, 0x26, 0x80, 0x00, 0x00,
- 0x4d, 0x2a, 0x5a, 0x20, 0x00, 0x00,
- 0x38, 0x02, 0x62, 0x88, 0x00, 0x00,
- 0x17, 0x49, 0x80, 0x46, 0x00, 0x00,
- 0x90, 0x84, 0x22, 0x4a, 0x00, 0x00,
- 0x72, 0x15, 0xd1, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_24[144] =
- {
- 0x88, 0x91, 0x08, 0x62, 0x00, 0x00,
- 0x40, 0x54, 0x31, 0x44, 0x00, 0x00,
- 0x82, 0x62, 0x9c, 0x02, 0x00, 0x00,
- 0x21, 0xa4, 0x89, 0x90, 0x00, 0x00,
- 0x10, 0x64, 0x1d, 0x20, 0x00, 0x00,
- 0x44, 0x0a, 0x41, 0x98, 0x00, 0x00,
- 0x10, 0xc9, 0x26, 0x80, 0x00, 0x00,
- 0x4d, 0x2a, 0x5a, 0x20, 0x00, 0x00,
- 0x38, 0x02, 0x62, 0x88, 0x00, 0x00,
- 0x17, 0x49, 0x80, 0x46, 0x00, 0x00,
- 0x90, 0x84, 0x22, 0x4a, 0x00, 0x00,
- 0x72, 0x15, 0xd1, 0x00, 0x00, 0x00,
- 0x62, 0x22, 0xaa, 0xaa, 0x00, 0x00,
- 0xf1, 0x10, 0x54, 0x44, 0x00, 0x00,
- 0x10, 0x0e, 0x62, 0x22, 0x00, 0x00,
- 0x10, 0xb1, 0x06, 0x84, 0x00, 0x00,
- 0x24, 0x24, 0x0d, 0x30, 0x00, 0x00,
- 0x01, 0x12, 0x81, 0xc2, 0x00, 0x00,
- 0x00, 0xc4, 0x58, 0x88, 0x00, 0x00,
- 0x04, 0xa3, 0xb0, 0x50, 0x00, 0x00,
- 0x02, 0x59, 0x25, 0x02, 0x00, 0x00,
- 0x2b, 0x01, 0x08, 0x64, 0x00, 0x00,
- 0x98, 0x40, 0xd0, 0x18, 0x00, 0x00,
- 0xf0, 0xdf, 0x91, 0xb6, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_25[150] =
- {
- 0x88, 0x91, 0x08, 0x62, 0x00, 0x00,
- 0x40, 0x54, 0x31, 0x44, 0x00, 0x00,
- 0x82, 0x62, 0x9c, 0x02, 0x00, 0x00,
- 0x21, 0xa4, 0x89, 0x90, 0x00, 0x00,
- 0x10, 0x64, 0x1d, 0x20, 0x00, 0x00,
- 0x44, 0x0a, 0x41, 0x98, 0x00, 0x00,
- 0x10, 0xc9, 0x26, 0x80, 0x00, 0x00,
- 0x4d, 0x2a, 0x5a, 0x20, 0x00, 0x00,
- 0x38, 0x02, 0x62, 0x88, 0x00, 0x00,
- 0x17, 0x49, 0x80, 0x46, 0x00, 0x00,
- 0x90, 0x84, 0x22, 0x4a, 0x00, 0x00,
- 0x72, 0x15, 0xd1, 0x00, 0x00, 0x00,
- 0x62, 0xa2, 0x8a, 0x2a, 0x00, 0x00,
- 0x34, 0x44, 0x44, 0x44, 0x00, 0x00,
- 0x40, 0x4b, 0x2c, 0x18, 0x00, 0x00,
- 0xc4, 0x04, 0x18, 0xa0, 0x00, 0x00,
- 0x08, 0x60, 0xc4, 0x08, 0x00, 0x00,
- 0x94, 0x12, 0x92, 0x0c, 0x00, 0x00,
- 0x88, 0xc0, 0x23, 0x04, 0x00, 0x00,
- 0x21, 0x32, 0x24, 0x70, 0x00, 0x00,
- 0xc1, 0x40, 0x80, 0xe2, 0x00, 0x00,
- 0x10, 0x69, 0x51, 0x14, 0x00, 0x00,
- 0x06, 0x90, 0x11, 0x42, 0x00, 0x00,
- 0x59, 0x01, 0x41, 0x80, 0x00, 0x00,
- 0x0a, 0x0d, 0x8a, 0x20, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_26[156] =
- {
- 0x62, 0xa2, 0x8a, 0x2a, 0x00, 0x00,
- 0x34, 0x44, 0x44, 0x44, 0x00, 0x00,
- 0x40, 0x4b, 0x2c, 0x18, 0x00, 0x00,
- 0xc4, 0x04, 0x18, 0xa0, 0x00, 0x00,
- 0x08, 0x60, 0xc4, 0x08, 0x00, 0x00,
- 0x94, 0x12, 0x92, 0x0c, 0x00, 0x00,
- 0x88, 0xc0, 0x23, 0x04, 0x00, 0x00,
- 0x21, 0x32, 0x24, 0x70, 0x00, 0x00,
- 0xc1, 0x40, 0x80, 0xe2, 0x00, 0x00,
- 0x10, 0x69, 0x51, 0x14, 0x00, 0x00,
- 0x06, 0x90, 0x11, 0x42, 0x00, 0x00,
- 0x59, 0x01, 0x41, 0x80, 0x00, 0x00,
- 0x0a, 0x0d, 0x8a, 0x20, 0x00, 0x00,
- 0x88, 0x91, 0x08, 0x62, 0x00, 0x00,
- 0x40, 0x54, 0x31, 0x44, 0x00, 0x00,
- 0x82, 0x62, 0x9c, 0x02, 0x00, 0x00,
- 0x21, 0xa4, 0x89, 0x90, 0x00, 0x00,
- 0x10, 0x64, 0x1d, 0x20, 0x00, 0x00,
- 0x44, 0x0a, 0x41, 0x98, 0x00, 0x00,
- 0x10, 0xc9, 0x26, 0x80, 0x00, 0x00,
- 0x4d, 0x2a, 0x5a, 0x20, 0x00, 0x00,
- 0x38, 0x02, 0x62, 0x88, 0x00, 0x00,
- 0x17, 0x49, 0x80, 0x46, 0x00, 0x00,
- 0x90, 0x84, 0x22, 0x4a, 0x00, 0x00,
- 0x72, 0x15, 0xd1, 0x00, 0x00, 0x00,
- 0xc5, 0x75, 0x48, 0xba, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_27[162] =
- {
- 0x62, 0xa2, 0x8a, 0x2a, 0x00, 0x00,
- 0x34, 0x44, 0x44, 0x44, 0x00, 0x00,
- 0x40, 0x4b, 0x2c, 0x18, 0x00, 0x00,
- 0xc4, 0x04, 0x18, 0xa0, 0x00, 0x00,
- 0x08, 0x60, 0xc4, 0x08, 0x00, 0x00,
- 0x94, 0x12, 0x92, 0x0c, 0x00, 0x00,
- 0x88, 0xc0, 0x23, 0x04, 0x00, 0x00,
- 0x21, 0x32, 0x24, 0x70, 0x00, 0x00,
- 0xc1, 0x40, 0x80, 0xe2, 0x00, 0x00,
- 0x10, 0x69, 0x51, 0x14, 0x00, 0x00,
- 0x06, 0x90, 0x11, 0x42, 0x00, 0x00,
- 0x59, 0x01, 0x41, 0x80, 0x00, 0x00,
- 0x0a, 0x0d, 0x8a, 0x20, 0x00, 0x00,
- 0x40, 0x82, 0x8a, 0xa2, 0x00, 0x00,
- 0x15, 0x54, 0x44, 0x14, 0x00, 0x00,
- 0x88, 0x13, 0x09, 0xa0, 0x00, 0x00,
- 0xc0, 0x10, 0x19, 0x14, 0x00, 0x00,
- 0x80, 0xa0, 0x30, 0x0c, 0x00, 0x00,
- 0x01, 0x22, 0x60, 0x06, 0x00, 0x00,
- 0x40, 0x2c, 0xc2, 0x10, 0x00, 0x00,
- 0x22, 0x02, 0x80, 0x22, 0x00, 0x00,
- 0x90, 0x04, 0x20, 0x58, 0x00, 0x00,
- 0x12, 0x40, 0x12, 0xc0, 0x00, 0x00,
- 0x5d, 0x00, 0x01, 0x28, 0x00, 0x00,
- 0x20, 0x54, 0xa4, 0x80, 0x00, 0x00,
- 0x86, 0x09, 0x48, 0x48, 0x00, 0x00,
- 0x28, 0x89, 0x05, 0x10, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_28[168] =
- {
- 0x40, 0x82, 0x8a, 0xa2, 0x00, 0x00,
- 0x15, 0x54, 0x44, 0x14, 0x00, 0x00,
- 0x88, 0x13, 0x09, 0xa0, 0x00, 0x00,
- 0xc0, 0x10, 0x19, 0x14, 0x00, 0x00,
- 0x80, 0xa0, 0x30, 0x0c, 0x00, 0x00,
- 0x01, 0x22, 0x60, 0x06, 0x00, 0x00,
- 0x40, 0x2c, 0xc2, 0x10, 0x00, 0x00,
- 0x22, 0x02, 0x80, 0x22, 0x00, 0x00,
- 0x90, 0x04, 0x20, 0x58, 0x00, 0x00,
- 0x12, 0x40, 0x12, 0xc0, 0x00, 0x00,
- 0x5d, 0x00, 0x01, 0x28, 0x00, 0x00,
- 0x20, 0x54, 0xa4, 0x80, 0x00, 0x00,
- 0x86, 0x09, 0x48, 0x48, 0x00, 0x00,
- 0x28, 0x89, 0x05, 0x10, 0x00, 0x00,
- 0x62, 0xa2, 0x8a, 0x2a, 0x00, 0x00,
- 0x34, 0x44, 0x44, 0x44, 0x00, 0x00,
- 0x40, 0x4b, 0x2c, 0x18, 0x00, 0x00,
- 0xc4, 0x04, 0x18, 0xa0, 0x00, 0x00,
- 0x08, 0x60, 0xc4, 0x08, 0x00, 0x00,
- 0x94, 0x12, 0x92, 0x0c, 0x00, 0x00,
- 0x88, 0xc0, 0x23, 0x04, 0x00, 0x00,
- 0x21, 0x32, 0x24, 0x70, 0x00, 0x00,
- 0xc1, 0x40, 0x80, 0xe2, 0x00, 0x00,
- 0x10, 0x69, 0x51, 0x14, 0x00, 0x00,
- 0x06, 0x90, 0x11, 0x42, 0x00, 0x00,
- 0x59, 0x01, 0x41, 0x80, 0x00, 0x00,
- 0x0a, 0x0d, 0x8a, 0x20, 0x00, 0x00,
- 0xbc, 0x0d, 0xca, 0x28, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_29[174] =
- {
- 0x40, 0x82, 0x8a, 0xa2, 0x00, 0x00,
- 0x15, 0x54, 0x44, 0x14, 0x00, 0x00,
- 0x88, 0x13, 0x09, 0xa0, 0x00, 0x00,
- 0xc0, 0x10, 0x19, 0x14, 0x00, 0x00,
- 0x80, 0xa0, 0x30, 0x0c, 0x00, 0x00,
- 0x01, 0x22, 0x60, 0x06, 0x00, 0x00,
- 0x40, 0x2c, 0xc2, 0x10, 0x00, 0x00,
- 0x22, 0x02, 0x80, 0x22, 0x00, 0x00,
- 0x90, 0x04, 0x20, 0x58, 0x00, 0x00,
- 0x12, 0x40, 0x12, 0xc0, 0x00, 0x00,
- 0x5d, 0x00, 0x01, 0x28, 0x00, 0x00,
- 0x20, 0x54, 0xa4, 0x80, 0x00, 0x00,
- 0x86, 0x09, 0x48, 0x48, 0x00, 0x00,
- 0x28, 0x89, 0x05, 0x10, 0x00, 0x00,
- 0x62, 0x22, 0xaa, 0x22, 0x00, 0x00,
- 0x31, 0x10, 0x44, 0x44, 0x00, 0x00,
- 0x58, 0x00, 0x22, 0x22, 0x00, 0x00,
- 0x01, 0x13, 0x00, 0x8a, 0x00, 0x00,
- 0x88, 0x20, 0x40, 0x34, 0x00, 0x00,
- 0x44, 0x02, 0x10, 0xd0, 0x00, 0x00,
- 0x29, 0x04, 0x45, 0x08, 0x00, 0x00,
- 0x82, 0xa0, 0x90, 0x12, 0x00, 0x00,
- 0x0a, 0x1a, 0x0e, 0x02, 0x00, 0x00,
- 0x11, 0xe1, 0x28, 0x40, 0x00, 0x00,
- 0x84, 0x05, 0x04, 0x0c, 0x00, 0x00,
- 0x86, 0x40, 0xc0, 0x90, 0x00, 0x00,
- 0x00, 0x87, 0x13, 0x00, 0x00, 0x00,
- 0x44, 0x48, 0x01, 0x1c, 0x00, 0x00,
- 0x10, 0x98, 0x30, 0x44, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_3[18] =
- {
- 0xac, 0x93, 0x5a, 0x5a, 0x00, 0x00,
- 0x55, 0x4a, 0xec, 0x6c, 0x00, 0x00,
- 0x43, 0x36, 0x4d, 0xb6, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_30[180] =
- {
- 0x62, 0x22, 0xaa, 0x22, 0x00, 0x00,
- 0x31, 0x10, 0x44, 0x44, 0x00, 0x00,
- 0x58, 0x00, 0x22, 0x22, 0x00, 0x00,
- 0x01, 0x13, 0x00, 0x8a, 0x00, 0x00,
- 0x88, 0x20, 0x40, 0x34, 0x00, 0x00,
- 0x44, 0x02, 0x10, 0xd0, 0x00, 0x00,
- 0x29, 0x04, 0x45, 0x08, 0x00, 0x00,
- 0x82, 0xa0, 0x90, 0x12, 0x00, 0x00,
- 0x0a, 0x1a, 0x0e, 0x02, 0x00, 0x00,
- 0x11, 0xe1, 0x28, 0x40, 0x00, 0x00,
- 0x84, 0x05, 0x04, 0x0c, 0x00, 0x00,
- 0x86, 0x40, 0xc0, 0x90, 0x00, 0x00,
- 0x00, 0x87, 0x13, 0x00, 0x00, 0x00,
- 0x44, 0x48, 0x01, 0x1c, 0x00, 0x00,
- 0x10, 0x98, 0x30, 0x44, 0x00, 0x00,
- 0x40, 0x82, 0x8a, 0xa2, 0x00, 0x00,
- 0x15, 0x54, 0x44, 0x14, 0x00, 0x00,
- 0x88, 0x13, 0x09, 0xa0, 0x00, 0x00,
- 0xc0, 0x10, 0x19, 0x14, 0x00, 0x00,
- 0x80, 0xa0, 0x30, 0x0c, 0x00, 0x00,
- 0x01, 0x22, 0x60, 0x06, 0x00, 0x00,
- 0x40, 0x2c, 0xc2, 0x10, 0x00, 0x00,
- 0x22, 0x02, 0x80, 0x22, 0x00, 0x00,
- 0x90, 0x04, 0x20, 0x58, 0x00, 0x00,
- 0x12, 0x40, 0x12, 0xc0, 0x00, 0x00,
- 0x5d, 0x00, 0x01, 0x28, 0x00, 0x00,
- 0x20, 0x54, 0xa4, 0x80, 0x00, 0x00,
- 0x86, 0x09, 0x48, 0x48, 0x00, 0x00,
- 0x28, 0x89, 0x05, 0x10, 0x00, 0x00,
- 0xe1, 0x4f, 0xe0, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_31[186] =
- {
- 0x62, 0x22, 0xaa, 0x22, 0x00, 0x00,
- 0x31, 0x10, 0x44, 0x44, 0x00, 0x00,
- 0x58, 0x00, 0x22, 0x22, 0x00, 0x00,
- 0x01, 0x13, 0x00, 0x8a, 0x00, 0x00,
- 0x88, 0x20, 0x40, 0x34, 0x00, 0x00,
- 0x44, 0x02, 0x10, 0xd0, 0x00, 0x00,
- 0x29, 0x04, 0x45, 0x08, 0x00, 0x00,
- 0x82, 0xa0, 0x90, 0x12, 0x00, 0x00,
- 0x0a, 0x1a, 0x0e, 0x02, 0x00, 0x00,
- 0x11, 0xe1, 0x28, 0x40, 0x00, 0x00,
- 0x84, 0x05, 0x04, 0x0c, 0x00, 0x00,
- 0x86, 0x40, 0xc0, 0x90, 0x00, 0x00,
- 0x00, 0x87, 0x13, 0x00, 0x00, 0x00,
- 0x44, 0x48, 0x01, 0x1c, 0x00, 0x00,
- 0x10, 0x98, 0x30, 0x44, 0x00, 0x00,
- 0x62, 0x23, 0x48, 0x20, 0x00, 0x00,
- 0x31, 0x10, 0x02, 0x54, 0x00, 0x00,
- 0x58, 0x00, 0x0c, 0x84, 0x00, 0x00,
- 0x01, 0x12, 0x10, 0xd0, 0x00, 0x00,
- 0x88, 0x21, 0x03, 0x20, 0x00, 0x00,
- 0x44, 0x02, 0x01, 0xe0, 0x00, 0x00,
- 0x29, 0x04, 0xa0, 0x0a, 0x00, 0x00,
- 0x82, 0xa0, 0x40, 0xa2, 0x00, 0x00,
- 0x0a, 0x1a, 0x86, 0x10, 0x00, 0x00,
- 0x11, 0xe0, 0xd1, 0x00, 0x00, 0x00,
- 0x84, 0x05, 0x00, 0x16, 0x00, 0x00,
- 0x86, 0x40, 0x20, 0x98, 0x00, 0x00,
- 0x00, 0x86, 0x24, 0x60, 0x00, 0x00,
- 0x44, 0x48, 0x81, 0x0a, 0x00, 0x00,
- 0x10, 0x98, 0x1c, 0x08, 0x00, 0x00,
- 0x87, 0x74, 0x30, 0x24, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_4[24] =
- {
- 0x25, 0xaa, 0xaa, 0xaa, 0x00, 0x00,
- 0x95, 0x55, 0x55, 0x54, 0x00, 0x00,
- 0x1a, 0x6a, 0x6a, 0x6a, 0x00, 0x00,
- 0x43, 0xd5, 0x95, 0x94, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_5[30] =
- {
- 0x64, 0xa2, 0xaa, 0xaa, 0x00, 0x00,
- 0x25, 0x54, 0x54, 0x54, 0x00, 0x00,
- 0x49, 0x68, 0x48, 0x4a, 0x00, 0x00,
- 0x53, 0x91, 0x09, 0x90, 0x00, 0x00,
- 0x8e, 0x30, 0x21, 0x6c, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_6[36] =
- {
- 0x62, 0x8a, 0xa2, 0xa2, 0x00, 0x00,
- 0x15, 0x54, 0x14, 0x54, 0x00, 0x00,
- 0x4c, 0x47, 0x44, 0x2a, 0x00, 0x00,
- 0x52, 0x95, 0x08, 0x94, 0x00, 0x00,
- 0x23, 0x64, 0x61, 0x24, 0x00, 0x00,
- 0x8a, 0x58, 0x09, 0x58, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_7[42] =
- {
- 0x62, 0xa2, 0x8a, 0xa2, 0x00, 0x00,
- 0xb1, 0x14, 0x44, 0x54, 0x00, 0x00,
- 0x18, 0x6b, 0x22, 0x22, 0x00, 0x00,
- 0x44, 0xd4, 0x5c, 0x10, 0x00, 0x00,
- 0x13, 0x64, 0x90, 0x68, 0x00, 0x00,
- 0x49, 0x1b, 0x20, 0x52, 0x00, 0x00,
- 0x86, 0x8c, 0x13, 0x0c, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_8[48] =
- {
- 0x90, 0x22, 0x40, 0xa8, 0x00, 0x00,
- 0x09, 0x50, 0x31, 0x10, 0x00, 0x00,
- 0x00, 0x6b, 0x08, 0x0e, 0x00, 0x00,
- 0x20, 0x34, 0xc0, 0x90, 0x00, 0x00,
- 0x14, 0x44, 0x25, 0x04, 0x00, 0x00,
- 0xc2, 0x11, 0x02, 0x82, 0x00, 0x00,
- 0x00, 0xc6, 0x80, 0xc4, 0x00, 0x00,
- 0x65, 0x80, 0x2c, 0x60, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask31_9[54] =
- {
- 0x62, 0x22, 0xaa, 0xa2, 0x00, 0x00,
- 0x24, 0x44, 0x44, 0x54, 0x00, 0x00,
- 0xc0, 0x50, 0x0b, 0x0a, 0x00, 0x00,
- 0x03, 0x0c, 0x12, 0x94, 0x00, 0x00,
- 0x16, 0x29, 0x08, 0x64, 0x00, 0x00,
- 0x89, 0x01, 0x80, 0x1a, 0x00, 0x00,
- 0x82, 0x90, 0x41, 0x4c, 0x00, 0x00,
- 0x08, 0xa4, 0x34, 0x12, 0x00, 0x00,
- 0x90, 0x48, 0x88, 0xc8, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_1[6] =
- {
- 0xff, 0xff, 0xff, 0xff, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_10[60] =
- {
- 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
- 0x10, 0xa2, 0x10, 0xa2, 0x00, 0x00,
- 0x01, 0x25, 0x01, 0x25, 0x00, 0x00,
- 0x0b, 0x42, 0x0b, 0x42, 0x00, 0x00,
- 0xd8, 0x20, 0xd8, 0x20, 0x00, 0x00,
- 0x82, 0x8c, 0x82, 0x8c, 0x00, 0x00,
- 0x24, 0x4a, 0x24, 0x4a, 0x00, 0x00,
- 0x38, 0x18, 0x38, 0x18, 0x00, 0x00,
- 0x2a, 0x25, 0x2a, 0x25, 0x00, 0x00,
- 0x84, 0x92, 0x84, 0x92, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_11[66] =
- {
- 0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
- 0x2a, 0x22, 0x2a, 0x22, 0x00, 0x00,
- 0x31, 0x11, 0x31, 0x11, 0x00, 0x00,
- 0x83, 0x42, 0x83, 0x42, 0x00, 0x00,
- 0x06, 0x98, 0x06, 0x98, 0x00, 0x00,
- 0x40, 0xe1, 0x40, 0xe1, 0x00, 0x00,
- 0x2c, 0x44, 0x2c, 0x44, 0x00, 0x00,
- 0xd8, 0x28, 0xd8, 0x28, 0x00, 0x00,
- 0x92, 0x81, 0x92, 0x81, 0x00, 0x00,
- 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
- 0x68, 0x0c, 0x68, 0x0c, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_12[72] =
- {
- 0x84, 0x31, 0x84, 0x31, 0x00, 0x00,
- 0x18, 0xa2, 0x18, 0xa2, 0x00, 0x00,
- 0x4e, 0x01, 0x4e, 0x01, 0x00, 0x00,
- 0x44, 0xc8, 0x44, 0xc8, 0x00, 0x00,
- 0x0e, 0x90, 0x0e, 0x90, 0x00, 0x00,
- 0x20, 0xcc, 0x20, 0xcc, 0x00, 0x00,
- 0x93, 0x40, 0x93, 0x40, 0x00, 0x00,
- 0x2d, 0x10, 0x2d, 0x10, 0x00, 0x00,
- 0x31, 0x44, 0x31, 0x44, 0x00, 0x00,
- 0xc0, 0x23, 0xc0, 0x23, 0x00, 0x00,
- 0x11, 0x25, 0x11, 0x25, 0x00, 0x00,
- 0xe8, 0x80, 0xe8, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_13[78] =
- {
- 0x45, 0x15, 0x45, 0x15, 0x00, 0x00,
- 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
- 0x96, 0x0c, 0x96, 0x0c, 0x00, 0x00,
- 0x0c, 0x50, 0x0c, 0x50, 0x00, 0x00,
- 0x62, 0x04, 0x62, 0x04, 0x00, 0x00,
- 0x49, 0x06, 0x49, 0x06, 0x00, 0x00,
- 0x11, 0x82, 0x11, 0x82, 0x00, 0x00,
- 0x12, 0x38, 0x12, 0x38, 0x00, 0x00,
- 0x40, 0x71, 0x40, 0x71, 0x00, 0x00,
- 0xa8, 0x8a, 0xa8, 0x8a, 0x00, 0x00,
- 0x08, 0xa1, 0x08, 0xa1, 0x00, 0x00,
- 0xa0, 0xc0, 0xa0, 0xc0, 0x00, 0x00,
- 0xc5, 0x10, 0xc5, 0x10, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_14[84] =
- {
- 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
- 0x22, 0x0a, 0x22, 0x0a, 0x00, 0x00,
- 0x84, 0xd0, 0x84, 0xd0, 0x00, 0x00,
- 0x0c, 0x8a, 0x0c, 0x8a, 0x00, 0x00,
- 0x18, 0x06, 0x18, 0x06, 0x00, 0x00,
- 0x30, 0x03, 0x30, 0x03, 0x00, 0x00,
- 0x61, 0x08, 0x61, 0x08, 0x00, 0x00,
- 0x40, 0x11, 0x40, 0x11, 0x00, 0x00,
- 0x10, 0x2c, 0x10, 0x2c, 0x00, 0x00,
- 0x09, 0x60, 0x09, 0x60, 0x00, 0x00,
- 0x00, 0x94, 0x00, 0x94, 0x00, 0x00,
- 0x52, 0x40, 0x52, 0x40, 0x00, 0x00,
- 0xa4, 0x24, 0xa4, 0x24, 0x00, 0x00,
- 0x82, 0x88, 0x82, 0x88, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_15[90] =
- {
- 0x55, 0x11, 0x55, 0x11, 0x00, 0x00,
- 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
- 0x11, 0x11, 0x11, 0x11, 0x00, 0x00,
- 0x80, 0x45, 0x80, 0x45, 0x00, 0x00,
- 0x20, 0x1a, 0x20, 0x1a, 0x00, 0x00,
- 0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
- 0x22, 0x84, 0x22, 0x84, 0x00, 0x00,
- 0x48, 0x09, 0x48, 0x09, 0x00, 0x00,
- 0x07, 0x01, 0x07, 0x01, 0x00, 0x00,
- 0x94, 0x20, 0x94, 0x20, 0x00, 0x00,
- 0x82, 0x06, 0x82, 0x06, 0x00, 0x00,
- 0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
- 0x89, 0x80, 0x89, 0x80, 0x00, 0x00,
- 0x00, 0x8e, 0x00, 0x8e, 0x00, 0x00,
- 0x18, 0x22, 0x18, 0x22, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_16[96] =
- {
- 0xa4, 0x10, 0xa4, 0x10, 0x00, 0x00,
- 0x01, 0x2a, 0x01, 0x2a, 0x00, 0x00,
- 0x06, 0x42, 0x06, 0x42, 0x00, 0x00,
- 0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
- 0x81, 0x90, 0x81, 0x90, 0x00, 0x00,
- 0x00, 0xf0, 0x00, 0xf0, 0x00, 0x00,
- 0x50, 0x05, 0x50, 0x05, 0x00, 0x00,
- 0x20, 0x51, 0x20, 0x51, 0x00, 0x00,
- 0x43, 0x08, 0x43, 0x08, 0x00, 0x00,
- 0x68, 0x80, 0x68, 0x80, 0x00, 0x00,
- 0x80, 0x0b, 0x80, 0x0b, 0x00, 0x00,
- 0x10, 0x4c, 0x10, 0x4c, 0x00, 0x00,
- 0x12, 0x30, 0x12, 0x30, 0x00, 0x00,
- 0x40, 0x85, 0x40, 0x85, 0x00, 0x00,
- 0x0e, 0x04, 0x0e, 0x04, 0x00, 0x00,
- 0x18, 0x12, 0x18, 0x12, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_17[102] =
- {
- 0x20, 0x54, 0x20, 0x54, 0x00, 0x00,
- 0x18, 0x88, 0x18, 0x88, 0x00, 0x00,
- 0x84, 0x07, 0x84, 0x07, 0x00, 0x00,
- 0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
- 0x12, 0x82, 0x12, 0x82, 0x00, 0x00,
- 0x81, 0x41, 0x81, 0x41, 0x00, 0x00,
- 0x40, 0x62, 0x40, 0x62, 0x00, 0x00,
- 0x16, 0x30, 0x16, 0x30, 0x00, 0x00,
- 0x55, 0x51, 0x55, 0x51, 0x00, 0x00,
- 0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
- 0x05, 0x85, 0x05, 0x85, 0x00, 0x00,
- 0x09, 0x4a, 0x09, 0x4a, 0x00, 0x00,
- 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
- 0xc0, 0x0d, 0xc0, 0x0d, 0x00, 0x00,
- 0x20, 0xa6, 0x20, 0xa6, 0x00, 0x00,
- 0x1a, 0x09, 0x1a, 0x09, 0x00, 0x00,
- 0x44, 0x64, 0x44, 0x64, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_18[108] =
- {
- 0x55, 0x51, 0x55, 0x51, 0x00, 0x00,
- 0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
- 0x05, 0x85, 0x05, 0x85, 0x00, 0x00,
- 0x09, 0x4a, 0x09, 0x4a, 0x00, 0x00,
- 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
- 0xc0, 0x0d, 0xc0, 0x0d, 0x00, 0x00,
- 0x20, 0xa6, 0x20, 0xa6, 0x00, 0x00,
- 0x1a, 0x09, 0x1a, 0x09, 0x00, 0x00,
- 0x44, 0x64, 0x44, 0x64, 0x00, 0x00,
- 0x20, 0x54, 0x20, 0x54, 0x00, 0x00,
- 0x18, 0x88, 0x18, 0x88, 0x00, 0x00,
- 0x84, 0x07, 0x84, 0x07, 0x00, 0x00,
- 0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
- 0x12, 0x82, 0x12, 0x82, 0x00, 0x00,
- 0x81, 0x41, 0x81, 0x41, 0x00, 0x00,
- 0x40, 0x62, 0x40, 0x62, 0x00, 0x00,
- 0x16, 0x30, 0x16, 0x30, 0x00, 0x00,
- 0x1e, 0xb2, 0xd8, 0x53, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_19[114] =
- {
- 0x55, 0x51, 0x55, 0x51, 0x00, 0x00,
- 0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
- 0x05, 0x85, 0x05, 0x85, 0x00, 0x00,
- 0x09, 0x4a, 0x09, 0x4a, 0x00, 0x00,
- 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
- 0xc0, 0x0d, 0xc0, 0x0d, 0x00, 0x00,
- 0x20, 0xa6, 0x20, 0xa6, 0x00, 0x00,
- 0x1a, 0x09, 0x1a, 0x09, 0x00, 0x00,
- 0x44, 0x64, 0x44, 0x64, 0x00, 0x00,
- 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
- 0x10, 0xa2, 0x10, 0xa2, 0x00, 0x00,
- 0x01, 0x25, 0x01, 0x25, 0x00, 0x00,
- 0x0b, 0x42, 0x0b, 0x42, 0x00, 0x00,
- 0xd8, 0x20, 0xd8, 0x20, 0x00, 0x00,
- 0x82, 0x8c, 0x82, 0x8c, 0x00, 0x00,
- 0x24, 0x4a, 0x24, 0x4a, 0x00, 0x00,
- 0x38, 0x18, 0x38, 0x18, 0x00, 0x00,
- 0x2a, 0x25, 0x2a, 0x25, 0x00, 0x00,
- 0x84, 0x92, 0x84, 0x92, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_2[12] =
- {
- 0xae, 0xae, 0xae, 0xae, 0x00, 0x00,
- 0x79, 0x79, 0x79, 0x79, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_20[120] =
- {
- 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
- 0x10, 0xa2, 0x10, 0xa2, 0x00, 0x00,
- 0x01, 0x25, 0x01, 0x25, 0x00, 0x00,
- 0x0b, 0x42, 0x0b, 0x42, 0x00, 0x00,
- 0xd8, 0x20, 0xd8, 0x20, 0x00, 0x00,
- 0x82, 0x8c, 0x82, 0x8c, 0x00, 0x00,
- 0x24, 0x4a, 0x24, 0x4a, 0x00, 0x00,
- 0x38, 0x18, 0x38, 0x18, 0x00, 0x00,
- 0x2a, 0x25, 0x2a, 0x25, 0x00, 0x00,
- 0x84, 0x92, 0x84, 0x92, 0x00, 0x00,
- 0x55, 0x51, 0x55, 0x51, 0x00, 0x00,
- 0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
- 0x05, 0x85, 0x05, 0x85, 0x00, 0x00,
- 0x09, 0x4a, 0x09, 0x4a, 0x00, 0x00,
- 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
- 0xc0, 0x0d, 0xc0, 0x0d, 0x00, 0x00,
- 0x20, 0xa6, 0x20, 0xa6, 0x00, 0x00,
- 0x1a, 0x09, 0x1a, 0x09, 0x00, 0x00,
- 0x44, 0x64, 0x44, 0x64, 0x00, 0x00,
- 0x96, 0xd3, 0xf6, 0xac, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_21[126] =
- {
- 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
- 0x10, 0xa2, 0x10, 0xa2, 0x00, 0x00,
- 0x01, 0x25, 0x01, 0x25, 0x00, 0x00,
- 0x0b, 0x42, 0x0b, 0x42, 0x00, 0x00,
- 0xd8, 0x20, 0xd8, 0x20, 0x00, 0x00,
- 0x82, 0x8c, 0x82, 0x8c, 0x00, 0x00,
- 0x24, 0x4a, 0x24, 0x4a, 0x00, 0x00,
- 0x38, 0x18, 0x38, 0x18, 0x00, 0x00,
- 0x2a, 0x25, 0x2a, 0x25, 0x00, 0x00,
- 0x84, 0x92, 0x84, 0x92, 0x00, 0x00,
- 0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
- 0x2a, 0x22, 0x2a, 0x22, 0x00, 0x00,
- 0x31, 0x11, 0x31, 0x11, 0x00, 0x00,
- 0x83, 0x42, 0x83, 0x42, 0x00, 0x00,
- 0x06, 0x98, 0x06, 0x98, 0x00, 0x00,
- 0x40, 0xe1, 0x40, 0xe1, 0x00, 0x00,
- 0x2c, 0x44, 0x2c, 0x44, 0x00, 0x00,
- 0xd8, 0x28, 0xd8, 0x28, 0x00, 0x00,
- 0x92, 0x81, 0x92, 0x81, 0x00, 0x00,
- 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
- 0x68, 0x0c, 0x68, 0x0c, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_22[132] =
- {
- 0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
- 0x2a, 0x22, 0x2a, 0x22, 0x00, 0x00,
- 0x31, 0x11, 0x31, 0x11, 0x00, 0x00,
- 0x83, 0x42, 0x83, 0x42, 0x00, 0x00,
- 0x06, 0x98, 0x06, 0x98, 0x00, 0x00,
- 0x40, 0xe1, 0x40, 0xe1, 0x00, 0x00,
- 0x2c, 0x44, 0x2c, 0x44, 0x00, 0x00,
- 0xd8, 0x28, 0xd8, 0x28, 0x00, 0x00,
- 0x92, 0x81, 0x92, 0x81, 0x00, 0x00,
- 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
- 0x68, 0x0c, 0x68, 0x0c, 0x00, 0x00,
- 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
- 0x10, 0xa2, 0x10, 0xa2, 0x00, 0x00,
- 0x01, 0x25, 0x01, 0x25, 0x00, 0x00,
- 0x0b, 0x42, 0x0b, 0x42, 0x00, 0x00,
- 0xd8, 0x20, 0xd8, 0x20, 0x00, 0x00,
- 0x82, 0x8c, 0x82, 0x8c, 0x00, 0x00,
- 0x24, 0x4a, 0x24, 0x4a, 0x00, 0x00,
- 0x38, 0x18, 0x38, 0x18, 0x00, 0x00,
- 0x2a, 0x25, 0x2a, 0x25, 0x00, 0x00,
- 0x84, 0x92, 0x84, 0x92, 0x00, 0x00,
- 0xeb, 0xb2, 0x22, 0x89, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_23[138] =
- {
- 0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
- 0x2a, 0x22, 0x2a, 0x22, 0x00, 0x00,
- 0x31, 0x11, 0x31, 0x11, 0x00, 0x00,
- 0x83, 0x42, 0x83, 0x42, 0x00, 0x00,
- 0x06, 0x98, 0x06, 0x98, 0x00, 0x00,
- 0x40, 0xe1, 0x40, 0xe1, 0x00, 0x00,
- 0x2c, 0x44, 0x2c, 0x44, 0x00, 0x00,
- 0xd8, 0x28, 0xd8, 0x28, 0x00, 0x00,
- 0x92, 0x81, 0x92, 0x81, 0x00, 0x00,
- 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
- 0x68, 0x0c, 0x68, 0x0c, 0x00, 0x00,
- 0x84, 0x31, 0x84, 0x31, 0x00, 0x00,
- 0x18, 0xa2, 0x18, 0xa2, 0x00, 0x00,
- 0x4e, 0x01, 0x4e, 0x01, 0x00, 0x00,
- 0x44, 0xc8, 0x44, 0xc8, 0x00, 0x00,
- 0x0e, 0x90, 0x0e, 0x90, 0x00, 0x00,
- 0x20, 0xcc, 0x20, 0xcc, 0x00, 0x00,
- 0x93, 0x40, 0x93, 0x40, 0x00, 0x00,
- 0x2d, 0x10, 0x2d, 0x10, 0x00, 0x00,
- 0x31, 0x44, 0x31, 0x44, 0x00, 0x00,
- 0xc0, 0x23, 0xc0, 0x23, 0x00, 0x00,
- 0x11, 0x25, 0x11, 0x25, 0x00, 0x00,
- 0xe8, 0x80, 0xe8, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_24[144] =
- {
- 0x84, 0x31, 0x84, 0x31, 0x00, 0x00,
- 0x18, 0xa2, 0x18, 0xa2, 0x00, 0x00,
- 0x4e, 0x01, 0x4e, 0x01, 0x00, 0x00,
- 0x44, 0xc8, 0x44, 0xc8, 0x00, 0x00,
- 0x0e, 0x90, 0x0e, 0x90, 0x00, 0x00,
- 0x20, 0xcc, 0x20, 0xcc, 0x00, 0x00,
- 0x93, 0x40, 0x93, 0x40, 0x00, 0x00,
- 0x2d, 0x10, 0x2d, 0x10, 0x00, 0x00,
- 0x31, 0x44, 0x31, 0x44, 0x00, 0x00,
- 0xc0, 0x23, 0xc0, 0x23, 0x00, 0x00,
- 0x11, 0x25, 0x11, 0x25, 0x00, 0x00,
- 0xe8, 0x80, 0xe8, 0x80, 0x00, 0x00,
- 0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
- 0x2a, 0x22, 0x2a, 0x22, 0x00, 0x00,
- 0x31, 0x11, 0x31, 0x11, 0x00, 0x00,
- 0x83, 0x42, 0x83, 0x42, 0x00, 0x00,
- 0x06, 0x98, 0x06, 0x98, 0x00, 0x00,
- 0x40, 0xe1, 0x40, 0xe1, 0x00, 0x00,
- 0x2c, 0x44, 0x2c, 0x44, 0x00, 0x00,
- 0xd8, 0x28, 0xd8, 0x28, 0x00, 0x00,
- 0x92, 0x81, 0x92, 0x81, 0x00, 0x00,
- 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
- 0x68, 0x0c, 0x68, 0x0c, 0x00, 0x00,
- 0xf3, 0x5a, 0x2f, 0x5d, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_25[150] =
- {
- 0x84, 0x31, 0x84, 0x31, 0x00, 0x00,
- 0x18, 0xa2, 0x18, 0xa2, 0x00, 0x00,
- 0x4e, 0x01, 0x4e, 0x01, 0x00, 0x00,
- 0x44, 0xc8, 0x44, 0xc8, 0x00, 0x00,
- 0x0e, 0x90, 0x0e, 0x90, 0x00, 0x00,
- 0x20, 0xcc, 0x20, 0xcc, 0x00, 0x00,
- 0x93, 0x40, 0x93, 0x40, 0x00, 0x00,
- 0x2d, 0x10, 0x2d, 0x10, 0x00, 0x00,
- 0x31, 0x44, 0x31, 0x44, 0x00, 0x00,
- 0xc0, 0x23, 0xc0, 0x23, 0x00, 0x00,
- 0x11, 0x25, 0x11, 0x25, 0x00, 0x00,
- 0xe8, 0x80, 0xe8, 0x80, 0x00, 0x00,
- 0x45, 0x15, 0x45, 0x15, 0x00, 0x00,
- 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
- 0x96, 0x0c, 0x96, 0x0c, 0x00, 0x00,
- 0x0c, 0x50, 0x0c, 0x50, 0x00, 0x00,
- 0x62, 0x04, 0x62, 0x04, 0x00, 0x00,
- 0x49, 0x06, 0x49, 0x06, 0x00, 0x00,
- 0x11, 0x82, 0x11, 0x82, 0x00, 0x00,
- 0x12, 0x38, 0x12, 0x38, 0x00, 0x00,
- 0x40, 0x71, 0x40, 0x71, 0x00, 0x00,
- 0xa8, 0x8a, 0xa8, 0x8a, 0x00, 0x00,
- 0x08, 0xa1, 0x08, 0xa1, 0x00, 0x00,
- 0xa0, 0xc0, 0xa0, 0xc0, 0x00, 0x00,
- 0xc5, 0x10, 0xc5, 0x10, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_26[156] =
- {
- 0x45, 0x15, 0x45, 0x15, 0x00, 0x00,
- 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
- 0x96, 0x0c, 0x96, 0x0c, 0x00, 0x00,
- 0x0c, 0x50, 0x0c, 0x50, 0x00, 0x00,
- 0x62, 0x04, 0x62, 0x04, 0x00, 0x00,
- 0x49, 0x06, 0x49, 0x06, 0x00, 0x00,
- 0x11, 0x82, 0x11, 0x82, 0x00, 0x00,
- 0x12, 0x38, 0x12, 0x38, 0x00, 0x00,
- 0x40, 0x71, 0x40, 0x71, 0x00, 0x00,
- 0xa8, 0x8a, 0xa8, 0x8a, 0x00, 0x00,
- 0x08, 0xa1, 0x08, 0xa1, 0x00, 0x00,
- 0xa0, 0xc0, 0xa0, 0xc0, 0x00, 0x00,
- 0xc5, 0x10, 0xc5, 0x10, 0x00, 0x00,
- 0x84, 0x31, 0x84, 0x31, 0x00, 0x00,
- 0x18, 0xa2, 0x18, 0xa2, 0x00, 0x00,
- 0x4e, 0x01, 0x4e, 0x01, 0x00, 0x00,
- 0x44, 0xc8, 0x44, 0xc8, 0x00, 0x00,
- 0x0e, 0x90, 0x0e, 0x90, 0x00, 0x00,
- 0x20, 0xcc, 0x20, 0xcc, 0x00, 0x00,
- 0x93, 0x40, 0x93, 0x40, 0x00, 0x00,
- 0x2d, 0x10, 0x2d, 0x10, 0x00, 0x00,
- 0x31, 0x44, 0x31, 0x44, 0x00, 0x00,
- 0xc0, 0x23, 0xc0, 0x23, 0x00, 0x00,
- 0x11, 0x25, 0x11, 0x25, 0x00, 0x00,
- 0xe8, 0x80, 0xe8, 0x80, 0x00, 0x00,
- 0x52, 0x15, 0x62, 0x0a, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_27[162] =
- {
- 0x45, 0x15, 0x45, 0x15, 0x00, 0x00,
- 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
- 0x96, 0x0c, 0x96, 0x0c, 0x00, 0x00,
- 0x0c, 0x50, 0x0c, 0x50, 0x00, 0x00,
- 0x62, 0x04, 0x62, 0x04, 0x00, 0x00,
- 0x49, 0x06, 0x49, 0x06, 0x00, 0x00,
- 0x11, 0x82, 0x11, 0x82, 0x00, 0x00,
- 0x12, 0x38, 0x12, 0x38, 0x00, 0x00,
- 0x40, 0x71, 0x40, 0x71, 0x00, 0x00,
- 0xa8, 0x8a, 0xa8, 0x8a, 0x00, 0x00,
- 0x08, 0xa1, 0x08, 0xa1, 0x00, 0x00,
- 0xa0, 0xc0, 0xa0, 0xc0, 0x00, 0x00,
- 0xc5, 0x10, 0xc5, 0x10, 0x00, 0x00,
- 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
- 0x22, 0x0a, 0x22, 0x0a, 0x00, 0x00,
- 0x84, 0xd0, 0x84, 0xd0, 0x00, 0x00,
- 0x0c, 0x8a, 0x0c, 0x8a, 0x00, 0x00,
- 0x18, 0x06, 0x18, 0x06, 0x00, 0x00,
- 0x30, 0x03, 0x30, 0x03, 0x00, 0x00,
- 0x61, 0x08, 0x61, 0x08, 0x00, 0x00,
- 0x40, 0x11, 0x40, 0x11, 0x00, 0x00,
- 0x10, 0x2c, 0x10, 0x2c, 0x00, 0x00,
- 0x09, 0x60, 0x09, 0x60, 0x00, 0x00,
- 0x00, 0x94, 0x00, 0x94, 0x00, 0x00,
- 0x52, 0x40, 0x52, 0x40, 0x00, 0x00,
- 0xa4, 0x24, 0xa4, 0x24, 0x00, 0x00,
- 0x82, 0x88, 0x82, 0x88, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_28[168] =
- {
- 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
- 0x22, 0x0a, 0x22, 0x0a, 0x00, 0x00,
- 0x84, 0xd0, 0x84, 0xd0, 0x00, 0x00,
- 0x0c, 0x8a, 0x0c, 0x8a, 0x00, 0x00,
- 0x18, 0x06, 0x18, 0x06, 0x00, 0x00,
- 0x30, 0x03, 0x30, 0x03, 0x00, 0x00,
- 0x61, 0x08, 0x61, 0x08, 0x00, 0x00,
- 0x40, 0x11, 0x40, 0x11, 0x00, 0x00,
- 0x10, 0x2c, 0x10, 0x2c, 0x00, 0x00,
- 0x09, 0x60, 0x09, 0x60, 0x00, 0x00,
- 0x00, 0x94, 0x00, 0x94, 0x00, 0x00,
- 0x52, 0x40, 0x52, 0x40, 0x00, 0x00,
- 0xa4, 0x24, 0xa4, 0x24, 0x00, 0x00,
- 0x82, 0x88, 0x82, 0x88, 0x00, 0x00,
- 0x45, 0x15, 0x45, 0x15, 0x00, 0x00,
- 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
- 0x96, 0x0c, 0x96, 0x0c, 0x00, 0x00,
- 0x0c, 0x50, 0x0c, 0x50, 0x00, 0x00,
- 0x62, 0x04, 0x62, 0x04, 0x00, 0x00,
- 0x49, 0x06, 0x49, 0x06, 0x00, 0x00,
- 0x11, 0x82, 0x11, 0x82, 0x00, 0x00,
- 0x12, 0x38, 0x12, 0x38, 0x00, 0x00,
- 0x40, 0x71, 0x40, 0x71, 0x00, 0x00,
- 0xa8, 0x8a, 0xa8, 0x8a, 0x00, 0x00,
- 0x08, 0xa1, 0x08, 0xa1, 0x00, 0x00,
- 0xa0, 0xc0, 0xa0, 0xc0, 0x00, 0x00,
- 0xc5, 0x10, 0xc5, 0x10, 0x00, 0x00,
- 0x7f, 0xe2, 0xbc, 0x01, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_29[174] =
- {
- 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
- 0x22, 0x0a, 0x22, 0x0a, 0x00, 0x00,
- 0x84, 0xd0, 0x84, 0xd0, 0x00, 0x00,
- 0x0c, 0x8a, 0x0c, 0x8a, 0x00, 0x00,
- 0x18, 0x06, 0x18, 0x06, 0x00, 0x00,
- 0x30, 0x03, 0x30, 0x03, 0x00, 0x00,
- 0x61, 0x08, 0x61, 0x08, 0x00, 0x00,
- 0x40, 0x11, 0x40, 0x11, 0x00, 0x00,
- 0x10, 0x2c, 0x10, 0x2c, 0x00, 0x00,
- 0x09, 0x60, 0x09, 0x60, 0x00, 0x00,
- 0x00, 0x94, 0x00, 0x94, 0x00, 0x00,
- 0x52, 0x40, 0x52, 0x40, 0x00, 0x00,
- 0xa4, 0x24, 0xa4, 0x24, 0x00, 0x00,
- 0x82, 0x88, 0x82, 0x88, 0x00, 0x00,
- 0x55, 0x11, 0x55, 0x11, 0x00, 0x00,
- 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
- 0x11, 0x11, 0x11, 0x11, 0x00, 0x00,
- 0x80, 0x45, 0x80, 0x45, 0x00, 0x00,
- 0x20, 0x1a, 0x20, 0x1a, 0x00, 0x00,
- 0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
- 0x22, 0x84, 0x22, 0x84, 0x00, 0x00,
- 0x48, 0x09, 0x48, 0x09, 0x00, 0x00,
- 0x07, 0x01, 0x07, 0x01, 0x00, 0x00,
- 0x94, 0x20, 0x94, 0x20, 0x00, 0x00,
- 0x82, 0x06, 0x82, 0x06, 0x00, 0x00,
- 0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
- 0x89, 0x80, 0x89, 0x80, 0x00, 0x00,
- 0x00, 0x8e, 0x00, 0x8e, 0x00, 0x00,
- 0x18, 0x22, 0x18, 0x22, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_3[18] =
- {
- 0xad, 0x2d, 0xad, 0x2d, 0x00, 0x00,
- 0x76, 0x36, 0x76, 0x36, 0x00, 0x00,
- 0x26, 0xdb, 0x26, 0xdb, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_30[180] =
- {
- 0x55, 0x11, 0x55, 0x11, 0x00, 0x00,
- 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
- 0x11, 0x11, 0x11, 0x11, 0x00, 0x00,
- 0x80, 0x45, 0x80, 0x45, 0x00, 0x00,
- 0x20, 0x1a, 0x20, 0x1a, 0x00, 0x00,
- 0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
- 0x22, 0x84, 0x22, 0x84, 0x00, 0x00,
- 0x48, 0x09, 0x48, 0x09, 0x00, 0x00,
- 0x07, 0x01, 0x07, 0x01, 0x00, 0x00,
- 0x94, 0x20, 0x94, 0x20, 0x00, 0x00,
- 0x82, 0x06, 0x82, 0x06, 0x00, 0x00,
- 0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
- 0x89, 0x80, 0x89, 0x80, 0x00, 0x00,
- 0x00, 0x8e, 0x00, 0x8e, 0x00, 0x00,
- 0x18, 0x22, 0x18, 0x22, 0x00, 0x00,
- 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
- 0x22, 0x0a, 0x22, 0x0a, 0x00, 0x00,
- 0x84, 0xd0, 0x84, 0xd0, 0x00, 0x00,
- 0x0c, 0x8a, 0x0c, 0x8a, 0x00, 0x00,
- 0x18, 0x06, 0x18, 0x06, 0x00, 0x00,
- 0x30, 0x03, 0x30, 0x03, 0x00, 0x00,
- 0x61, 0x08, 0x61, 0x08, 0x00, 0x00,
- 0x40, 0x11, 0x40, 0x11, 0x00, 0x00,
- 0x10, 0x2c, 0x10, 0x2c, 0x00, 0x00,
- 0x09, 0x60, 0x09, 0x60, 0x00, 0x00,
- 0x00, 0x94, 0x00, 0x94, 0x00, 0x00,
- 0x52, 0x40, 0x52, 0x40, 0x00, 0x00,
- 0xa4, 0x24, 0xa4, 0x24, 0x00, 0x00,
- 0x82, 0x88, 0x82, 0x88, 0x00, 0x00,
- 0x1e, 0x27, 0xe2, 0xd8, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_31[186] =
- {
- 0x55, 0x11, 0x55, 0x11, 0x00, 0x00,
- 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
- 0x11, 0x11, 0x11, 0x11, 0x00, 0x00,
- 0x80, 0x45, 0x80, 0x45, 0x00, 0x00,
- 0x20, 0x1a, 0x20, 0x1a, 0x00, 0x00,
- 0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
- 0x22, 0x84, 0x22, 0x84, 0x00, 0x00,
- 0x48, 0x09, 0x48, 0x09, 0x00, 0x00,
- 0x07, 0x01, 0x07, 0x01, 0x00, 0x00,
- 0x94, 0x20, 0x94, 0x20, 0x00, 0x00,
- 0x82, 0x06, 0x82, 0x06, 0x00, 0x00,
- 0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
- 0x89, 0x80, 0x89, 0x80, 0x00, 0x00,
- 0x00, 0x8e, 0x00, 0x8e, 0x00, 0x00,
- 0x18, 0x22, 0x18, 0x22, 0x00, 0x00,
- 0xa4, 0x10, 0xa4, 0x10, 0x00, 0x00,
- 0x01, 0x2a, 0x01, 0x2a, 0x00, 0x00,
- 0x06, 0x42, 0x06, 0x42, 0x00, 0x00,
- 0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
- 0x81, 0x90, 0x81, 0x90, 0x00, 0x00,
- 0x00, 0xf0, 0x00, 0xf0, 0x00, 0x00,
- 0x50, 0x05, 0x50, 0x05, 0x00, 0x00,
- 0x20, 0x51, 0x20, 0x51, 0x00, 0x00,
- 0x43, 0x08, 0x43, 0x08, 0x00, 0x00,
- 0x68, 0x80, 0x68, 0x80, 0x00, 0x00,
- 0x80, 0x0b, 0x80, 0x0b, 0x00, 0x00,
- 0x10, 0x4c, 0x10, 0x4c, 0x00, 0x00,
- 0x12, 0x30, 0x12, 0x30, 0x00, 0x00,
- 0x40, 0x85, 0x40, 0x85, 0x00, 0x00,
- 0x0e, 0x04, 0x0e, 0x04, 0x00, 0x00,
- 0x18, 0x12, 0x18, 0x12, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_32[192] =
- {
- 0xa4, 0x10, 0xa4, 0x10, 0x00, 0x00,
- 0x01, 0x2a, 0x01, 0x2a, 0x00, 0x00,
- 0x06, 0x42, 0x06, 0x42, 0x00, 0x00,
- 0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
- 0x81, 0x90, 0x81, 0x90, 0x00, 0x00,
- 0x00, 0xf0, 0x00, 0xf0, 0x00, 0x00,
- 0x50, 0x05, 0x50, 0x05, 0x00, 0x00,
- 0x20, 0x51, 0x20, 0x51, 0x00, 0x00,
- 0x43, 0x08, 0x43, 0x08, 0x00, 0x00,
- 0x68, 0x80, 0x68, 0x80, 0x00, 0x00,
- 0x80, 0x0b, 0x80, 0x0b, 0x00, 0x00,
- 0x10, 0x4c, 0x10, 0x4c, 0x00, 0x00,
- 0x12, 0x30, 0x12, 0x30, 0x00, 0x00,
- 0x40, 0x85, 0x40, 0x85, 0x00, 0x00,
- 0x0e, 0x04, 0x0e, 0x04, 0x00, 0x00,
- 0x18, 0x12, 0x18, 0x12, 0x00, 0x00,
- 0x55, 0x11, 0x55, 0x11, 0x00, 0x00,
- 0x22, 0x22, 0x22, 0x22, 0x00, 0x00,
- 0x11, 0x11, 0x11, 0x11, 0x00, 0x00,
- 0x80, 0x45, 0x80, 0x45, 0x00, 0x00,
- 0x20, 0x1a, 0x20, 0x1a, 0x00, 0x00,
- 0x08, 0x68, 0x08, 0x68, 0x00, 0x00,
- 0x22, 0x84, 0x22, 0x84, 0x00, 0x00,
- 0x48, 0x09, 0x48, 0x09, 0x00, 0x00,
- 0x07, 0x01, 0x07, 0x01, 0x00, 0x00,
- 0x94, 0x20, 0x94, 0x20, 0x00, 0x00,
- 0x82, 0x06, 0x82, 0x06, 0x00, 0x00,
- 0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
- 0x89, 0x80, 0x89, 0x80, 0x00, 0x00,
- 0x00, 0x8e, 0x00, 0x8e, 0x00, 0x00,
- 0x18, 0x22, 0x18, 0x22, 0x00, 0x00,
- 0x60, 0xc4, 0x02, 0x02, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_4[24] =
- {
- 0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
- 0xaa, 0xaa, 0xaa, 0xaa, 0x00, 0x00,
- 0x35, 0x35, 0x35, 0x35, 0x00, 0x00,
- 0xca, 0xca, 0xca, 0xca, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_5[30] =
- {
- 0x55, 0x55, 0x55, 0x55, 0x00, 0x00,
- 0x2a, 0x2a, 0x2a, 0x2a, 0x00, 0x00,
- 0x24, 0x25, 0x24, 0x25, 0x00, 0x00,
- 0x84, 0xc8, 0x84, 0xc8, 0x00, 0x00,
- 0x10, 0xb6, 0x10, 0xb6, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_6[36] =
- {
- 0x51, 0x51, 0x51, 0x51, 0x00, 0x00,
- 0x0a, 0x2a, 0x0a, 0x2a, 0x00, 0x00,
- 0xa2, 0x15, 0xa2, 0x15, 0x00, 0x00,
- 0x84, 0x4a, 0x84, 0x4a, 0x00, 0x00,
- 0x30, 0x92, 0x30, 0x92, 0x00, 0x00,
- 0x04, 0xac, 0x04, 0xac, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_7[42] =
- {
- 0x45, 0x51, 0x45, 0x51, 0x00, 0x00,
- 0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
- 0x91, 0x11, 0x91, 0x11, 0x00, 0x00,
- 0x2e, 0x08, 0x2e, 0x08, 0x00, 0x00,
- 0x48, 0x34, 0x48, 0x34, 0x00, 0x00,
- 0x90, 0x29, 0x90, 0x29, 0x00, 0x00,
- 0x09, 0x86, 0x09, 0x86, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_8[48] =
- {
- 0x20, 0x54, 0x20, 0x54, 0x00, 0x00,
- 0x18, 0x88, 0x18, 0x88, 0x00, 0x00,
- 0x84, 0x07, 0x84, 0x07, 0x00, 0x00,
- 0x60, 0x48, 0x60, 0x48, 0x00, 0x00,
- 0x12, 0x82, 0x12, 0x82, 0x00, 0x00,
- 0x81, 0x41, 0x81, 0x41, 0x00, 0x00,
- 0x40, 0x62, 0x40, 0x62, 0x00, 0x00,
- 0x16, 0x30, 0x16, 0x30, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask32_9[54] =
- {
- 0x55, 0x51, 0x55, 0x51, 0x00, 0x00,
- 0x22, 0x2a, 0x22, 0x2a, 0x00, 0x00,
- 0x05, 0x85, 0x05, 0x85, 0x00, 0x00,
- 0x09, 0x4a, 0x09, 0x4a, 0x00, 0x00,
- 0x84, 0x32, 0x84, 0x32, 0x00, 0x00,
- 0xc0, 0x0d, 0xc0, 0x0d, 0x00, 0x00,
- 0x20, 0xa6, 0x20, 0xa6, 0x00, 0x00,
- 0x1a, 0x09, 0x1a, 0x09, 0x00, 0x00,
- 0x44, 0x64, 0x44, 0x64, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_1[6] =
- {
- 0xff, 0xff, 0xff, 0xff, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask33_10[60] =
- {
- 0x45, 0x51, 0x55, 0x8c, 0x80, 0x00,
- 0x10, 0xa2, 0xaa, 0x27, 0x00, 0x00,
- 0x01, 0x25, 0xa5, 0x32, 0x80, 0x00,
- 0x0b, 0x42, 0x62, 0x61, 0x80, 0x00,
- 0xd8, 0x20, 0x3c, 0x5c, 0x00, 0x00,
- 0x82, 0x8c, 0x8e, 0xcc, 0x00, 0x00,
- 0x24, 0x4a, 0x6a, 0x2b, 0x00, 0x00,
- 0x38, 0x18, 0x36, 0x32, 0x80, 0x00,
- 0x2a, 0x25, 0xd1, 0x25, 0x80, 0x00,
- 0x84, 0x92, 0xc8, 0x02, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask33_11[66] =
- {
- 0x55, 0x55, 0x55, 0x8c, 0x80, 0x00,
- 0x2a, 0x22, 0xaa, 0x27, 0x00, 0x00,
- 0x31, 0x11, 0xa5, 0x32, 0x80, 0x00,
- 0x83, 0x42, 0x62, 0x61, 0x80, 0x00,
- 0x06, 0x98, 0x3c, 0x5c, 0x00, 0x00,
- 0x40, 0xe1, 0x51, 0x84, 0x80, 0x00,
- 0x2c, 0x44, 0xa2, 0x27, 0x00, 0x00,
- 0xd8, 0x28, 0x95, 0x51, 0x80, 0x00,
- 0x92, 0x81, 0x4a, 0x1a, 0x00, 0x00,
- 0x84, 0x32, 0x30, 0x68, 0x00, 0x00,
- 0x68, 0x0c, 0x2c, 0x89, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_12[72] =
- {
- 0x84, 0x31, 0x51, 0x84, 0x80, 0x00,
- 0x18, 0xa2, 0xa2, 0x27, 0x00, 0x00,
- 0x4e, 0x01, 0x95, 0x51, 0x80, 0x00,
- 0x44, 0xc8, 0x4a, 0x1a, 0x00, 0x00,
- 0x0e, 0x90, 0x30, 0x68, 0x00, 0x00,
- 0x20, 0xcc, 0x2c, 0x89, 0x00, 0x00,
- 0x93, 0x40, 0x55, 0x8c, 0x80, 0x00,
- 0x2d, 0x10, 0xaa, 0x27, 0x00, 0x00,
- 0x31, 0x44, 0xa5, 0x32, 0x80, 0x00,
- 0xc0, 0x23, 0x62, 0x61, 0x80, 0x00,
- 0x11, 0x25, 0x3c, 0x5c, 0x00, 0x00,
- 0xe8, 0x80, 0x51, 0x35, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_13[78] =
- {
- 0x45, 0x15, 0x51, 0x84, 0x80, 0x00,
- 0x22, 0x22, 0xa2, 0x27, 0x00, 0x00,
- 0x96, 0x0c, 0x95, 0x51, 0x80, 0x00,
- 0x0c, 0x50, 0x4a, 0x1a, 0x00, 0x00,
- 0x62, 0x04, 0x30, 0x68, 0x00, 0x00,
- 0x49, 0x06, 0x2c, 0x89, 0x00, 0x00,
- 0x11, 0x82, 0x15, 0x8c, 0x00, 0x00,
- 0x12, 0x38, 0x8a, 0x47, 0x00, 0x00,
- 0x40, 0x71, 0x25, 0x81, 0x80, 0x00,
- 0xa8, 0x8a, 0x62, 0x12, 0x80, 0x00,
- 0x08, 0xa1, 0x58, 0x58, 0x00, 0x00,
- 0xa0, 0xc0, 0x0e, 0x28, 0x80, 0x00,
- 0xc5, 0x10, 0x83, 0x34, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_14[84] =
- {
- 0x45, 0x51, 0x15, 0x8c, 0x00, 0x00,
- 0x22, 0x0a, 0x8a, 0x47, 0x00, 0x00,
- 0x84, 0xd0, 0x25, 0x81, 0x80, 0x00,
- 0x0c, 0x8a, 0x62, 0x12, 0x80, 0x00,
- 0x18, 0x06, 0x58, 0x58, 0x00, 0x00,
- 0x30, 0x03, 0x0e, 0x28, 0x80, 0x00,
- 0x61, 0x08, 0x83, 0x34, 0x00, 0x00,
- 0x40, 0x11, 0x51, 0x84, 0x80, 0x00,
- 0x10, 0x2c, 0xa2, 0x27, 0x00, 0x00,
- 0x09, 0x60, 0x95, 0x51, 0x80, 0x00,
- 0x00, 0x94, 0x4a, 0x1a, 0x00, 0x00,
- 0x52, 0x40, 0x30, 0x68, 0x00, 0x00,
- 0xa4, 0x24, 0x2c, 0x89, 0x00, 0x00,
- 0x82, 0x88, 0xb0, 0xde, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask33_15[90] =
- {
- 0x55, 0x11, 0x15, 0x8c, 0x00, 0x00,
- 0x22, 0x22, 0x8a, 0x47, 0x00, 0x00,
- 0x11, 0x11, 0x25, 0x81, 0x80, 0x00,
- 0x80, 0x45, 0x62, 0x12, 0x80, 0x00,
- 0x20, 0x1a, 0x58, 0x58, 0x00, 0x00,
- 0x08, 0x68, 0x0e, 0x28, 0x80, 0x00,
- 0x22, 0x84, 0x83, 0x34, 0x00, 0x00,
- 0x48, 0x09, 0x25, 0x2c, 0x00, 0x00,
- 0x07, 0x01, 0x8a, 0x91, 0x00, 0x00,
- 0x94, 0x20, 0x91, 0xc0, 0x80, 0x00,
- 0x82, 0x06, 0x68, 0x06, 0x80, 0x00,
- 0x60, 0x48, 0x32, 0xc8, 0x00, 0x00,
- 0x89, 0x80, 0x43, 0x45, 0x00, 0x00,
- 0x00, 0x8e, 0xc4, 0x30, 0x80, 0x00,
- 0x18, 0x22, 0x1c, 0xa2, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_16[96] =
- {
- 0xa4, 0x10, 0x25, 0x2c, 0x00, 0x00,
- 0x01, 0x2a, 0x8a, 0x91, 0x00, 0x00,
- 0x06, 0x42, 0x91, 0xc0, 0x80, 0x00,
- 0x08, 0x68, 0x68, 0x06, 0x80, 0x00,
- 0x81, 0x90, 0x32, 0xc8, 0x00, 0x00,
- 0x00, 0xf0, 0x43, 0x45, 0x00, 0x00,
- 0x50, 0x05, 0xc4, 0x30, 0x80, 0x00,
- 0x20, 0x51, 0x1c, 0xa2, 0x00, 0x00,
- 0x43, 0x08, 0x15, 0x8c, 0x00, 0x00,
- 0x68, 0x80, 0x8a, 0x47, 0x00, 0x00,
- 0x80, 0x0b, 0x25, 0x81, 0x80, 0x00,
- 0x10, 0x4c, 0x62, 0x12, 0x80, 0x00,
- 0x12, 0x30, 0x58, 0x58, 0x00, 0x00,
- 0x40, 0x85, 0x0e, 0x28, 0x80, 0x00,
- 0x0e, 0x04, 0x83, 0x34, 0x00, 0x00,
- 0x18, 0x12, 0x0a, 0x1c, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_17[102] =
- {
- 0x20, 0x54, 0x64, 0x16, 0x00, 0x00,
- 0x18, 0x88, 0xa2, 0xc2, 0x00, 0x00,
- 0x84, 0x07, 0x51, 0x60, 0x80, 0x00,
- 0x60, 0x48, 0x4a, 0x85, 0x00, 0x00,
- 0x12, 0x82, 0x38, 0x4c, 0x00, 0x00,
- 0x81, 0x41, 0x89, 0x29, 0x00, 0x00,
- 0x40, 0x62, 0x07, 0x11, 0x80, 0x00,
- 0x16, 0x30, 0x94, 0xb0, 0x00, 0x00,
- 0x55, 0x51, 0x8e, 0xcc, 0x00, 0x00,
- 0x22, 0x2a, 0x6a, 0x2b, 0x00, 0x00,
- 0x05, 0x85, 0x36, 0x32, 0x80, 0x00,
- 0x09, 0x4a, 0xd1, 0x25, 0x80, 0x00,
- 0x84, 0x32, 0x55, 0x8c, 0x80, 0x00,
- 0xc0, 0x0d, 0xaa, 0x27, 0x00, 0x00,
- 0x20, 0xa6, 0xa5, 0x32, 0x80, 0x00,
- 0x1a, 0x09, 0x62, 0x61, 0x80, 0x00,
- 0x44, 0x64, 0x3c, 0x5c, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_18[108] =
- {
- 0x55, 0x51, 0x8e, 0xcc, 0x00, 0x00,
- 0x22, 0x2a, 0x6a, 0x2b, 0x00, 0x00,
- 0x05, 0x85, 0x36, 0x32, 0x80, 0x00,
- 0x09, 0x4a, 0xd1, 0x25, 0x80, 0x00,
- 0x84, 0x32, 0x55, 0x8c, 0x80, 0x00,
- 0xc0, 0x0d, 0xaa, 0x27, 0x00, 0x00,
- 0x20, 0xa6, 0xa5, 0x32, 0x80, 0x00,
- 0x1a, 0x09, 0x62, 0x61, 0x80, 0x00,
- 0x44, 0x64, 0x3c, 0x5c, 0x00, 0x00,
- 0x20, 0x54, 0x64, 0x16, 0x00, 0x00,
- 0x18, 0x88, 0xa2, 0xc2, 0x00, 0x00,
- 0x84, 0x07, 0x51, 0x60, 0x80, 0x00,
- 0x60, 0x48, 0x4a, 0x85, 0x00, 0x00,
- 0x12, 0x82, 0x38, 0x4c, 0x00, 0x00,
- 0x81, 0x41, 0x89, 0x29, 0x00, 0x00,
- 0x40, 0x62, 0x07, 0x11, 0x80, 0x00,
- 0x16, 0x30, 0x94, 0xb0, 0x00, 0x00,
- 0x89, 0x53, 0x03, 0xad, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_19[114] =
- {
- 0x55, 0x51, 0x8e, 0xcc, 0x00, 0x00,
- 0x22, 0x2a, 0x6a, 0x2b, 0x00, 0x00,
- 0x05, 0x85, 0x36, 0x32, 0x80, 0x00,
- 0x09, 0x4a, 0xd1, 0x25, 0x80, 0x00,
- 0x84, 0x32, 0x55, 0x8c, 0x80, 0x00,
- 0xc0, 0x0d, 0xaa, 0x27, 0x00, 0x00,
- 0x20, 0xa6, 0xa5, 0x32, 0x80, 0x00,
- 0x1a, 0x09, 0x62, 0x61, 0x80, 0x00,
- 0x44, 0x64, 0x3c, 0x5c, 0x00, 0x00,
- 0x45, 0x51, 0x55, 0x8c, 0x80, 0x00,
- 0x10, 0xa2, 0xaa, 0x27, 0x00, 0x00,
- 0x01, 0x25, 0xa5, 0x32, 0x80, 0x00,
- 0x0b, 0x42, 0x62, 0x61, 0x80, 0x00,
- 0xd8, 0x20, 0x3c, 0x5c, 0x00, 0x00,
- 0x82, 0x8c, 0x8e, 0xcc, 0x00, 0x00,
- 0x24, 0x4a, 0x6a, 0x2b, 0x00, 0x00,
- 0x38, 0x18, 0x36, 0x32, 0x80, 0x00,
- 0x2a, 0x25, 0xd1, 0x25, 0x80, 0x00,
- 0x84, 0x92, 0xc8, 0x02, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask33_2[12] =
- {
- 0xae, 0xae, 0xce, 0xce, 0x00, 0x00,
- 0x79, 0x79, 0xb9, 0x39, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask33_20[120] =
- {
- 0x45, 0x51, 0x55, 0x8c, 0x80, 0x00,
- 0x10, 0xa2, 0xaa, 0x27, 0x00, 0x00,
- 0x01, 0x25, 0xa5, 0x32, 0x80, 0x00,
- 0x0b, 0x42, 0x62, 0x61, 0x80, 0x00,
- 0xd8, 0x20, 0x3c, 0x5c, 0x00, 0x00,
- 0x82, 0x8c, 0x8e, 0xcc, 0x00, 0x00,
- 0x24, 0x4a, 0x6a, 0x2b, 0x00, 0x00,
- 0x38, 0x18, 0x36, 0x32, 0x80, 0x00,
- 0x2a, 0x25, 0xd1, 0x25, 0x80, 0x00,
- 0x84, 0x92, 0xc8, 0x02, 0x80, 0x00,
- 0x55, 0x51, 0x8e, 0xcc, 0x00, 0x00,
- 0x22, 0x2a, 0x6a, 0x2b, 0x00, 0x00,
- 0x05, 0x85, 0x36, 0x32, 0x80, 0x00,
- 0x09, 0x4a, 0xd1, 0x25, 0x80, 0x00,
- 0x84, 0x32, 0x55, 0x8c, 0x80, 0x00,
- 0xc0, 0x0d, 0xaa, 0x27, 0x00, 0x00,
- 0x20, 0xa6, 0xa5, 0x32, 0x80, 0x00,
- 0x1a, 0x09, 0x62, 0x61, 0x80, 0x00,
- 0x44, 0x64, 0x3c, 0x5c, 0x00, 0x00,
- 0x73, 0x5f, 0x5b, 0x0e, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask33_21[126] =
- {
- 0x45, 0x51, 0x55, 0x8c, 0x80, 0x00,
- 0x10, 0xa2, 0xaa, 0x27, 0x00, 0x00,
- 0x01, 0x25, 0xa5, 0x32, 0x80, 0x00,
- 0x0b, 0x42, 0x62, 0x61, 0x80, 0x00,
- 0xd8, 0x20, 0x3c, 0x5c, 0x00, 0x00,
- 0x82, 0x8c, 0x8e, 0xcc, 0x00, 0x00,
- 0x24, 0x4a, 0x6a, 0x2b, 0x00, 0x00,
- 0x38, 0x18, 0x36, 0x32, 0x80, 0x00,
- 0x2a, 0x25, 0xd1, 0x25, 0x80, 0x00,
- 0x84, 0x92, 0xc8, 0x02, 0x80, 0x00,
- 0x55, 0x55, 0x55, 0x8c, 0x80, 0x00,
- 0x2a, 0x22, 0xaa, 0x27, 0x00, 0x00,
- 0x31, 0x11, 0xa5, 0x32, 0x80, 0x00,
- 0x83, 0x42, 0x62, 0x61, 0x80, 0x00,
- 0x06, 0x98, 0x3c, 0x5c, 0x00, 0x00,
- 0x40, 0xe1, 0x51, 0x84, 0x80, 0x00,
- 0x2c, 0x44, 0xa2, 0x27, 0x00, 0x00,
- 0xd8, 0x28, 0x95, 0x51, 0x80, 0x00,
- 0x92, 0x81, 0x4a, 0x1a, 0x00, 0x00,
- 0x84, 0x32, 0x30, 0x68, 0x00, 0x00,
- 0x68, 0x0c, 0x2c, 0x89, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_22[132] =
- {
- 0x55, 0x55, 0x55, 0x8c, 0x80, 0x00,
- 0x2a, 0x22, 0xaa, 0x27, 0x00, 0x00,
- 0x31, 0x11, 0xa5, 0x32, 0x80, 0x00,
- 0x83, 0x42, 0x62, 0x61, 0x80, 0x00,
- 0x06, 0x98, 0x3c, 0x5c, 0x00, 0x00,
- 0x40, 0xe1, 0x51, 0x84, 0x80, 0x00,
- 0x2c, 0x44, 0xa2, 0x27, 0x00, 0x00,
- 0xd8, 0x28, 0x95, 0x51, 0x80, 0x00,
- 0x92, 0x81, 0x4a, 0x1a, 0x00, 0x00,
- 0x84, 0x32, 0x30, 0x68, 0x00, 0x00,
- 0x68, 0x0c, 0x2c, 0x89, 0x00, 0x00,
- 0x45, 0x51, 0x55, 0x8c, 0x80, 0x00,
- 0x10, 0xa2, 0xaa, 0x27, 0x00, 0x00,
- 0x01, 0x25, 0xa5, 0x32, 0x80, 0x00,
- 0x0b, 0x42, 0x62, 0x61, 0x80, 0x00,
- 0xd8, 0x20, 0x3c, 0x5c, 0x00, 0x00,
- 0x82, 0x8c, 0x8e, 0xcc, 0x00, 0x00,
- 0x24, 0x4a, 0x6a, 0x2b, 0x00, 0x00,
- 0x38, 0x18, 0x36, 0x32, 0x80, 0x00,
- 0x2a, 0x25, 0xd1, 0x25, 0x80, 0x00,
- 0x84, 0x92, 0xc8, 0x02, 0x80, 0x00,
- 0xcc, 0xe3, 0x42, 0x6b, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask33_23[138] =
- {
- 0x55, 0x55, 0x55, 0x8c, 0x80, 0x00,
- 0x2a, 0x22, 0xaa, 0x27, 0x00, 0x00,
- 0x31, 0x11, 0xa5, 0x32, 0x80, 0x00,
- 0x83, 0x42, 0x62, 0x61, 0x80, 0x00,
- 0x06, 0x98, 0x3c, 0x5c, 0x00, 0x00,
- 0x40, 0xe1, 0x51, 0x84, 0x80, 0x00,
- 0x2c, 0x44, 0xa2, 0x27, 0x00, 0x00,
- 0xd8, 0x28, 0x95, 0x51, 0x80, 0x00,
- 0x92, 0x81, 0x4a, 0x1a, 0x00, 0x00,
- 0x84, 0x32, 0x30, 0x68, 0x00, 0x00,
- 0x68, 0x0c, 0x2c, 0x89, 0x00, 0x00,
- 0x84, 0x31, 0x51, 0x84, 0x80, 0x00,
- 0x18, 0xa2, 0xa2, 0x27, 0x00, 0x00,
- 0x4e, 0x01, 0x95, 0x51, 0x80, 0x00,
- 0x44, 0xc8, 0x4a, 0x1a, 0x00, 0x00,
- 0x0e, 0x90, 0x30, 0x68, 0x00, 0x00,
- 0x20, 0xcc, 0x2c, 0x89, 0x00, 0x00,
- 0x93, 0x40, 0x55, 0x8c, 0x80, 0x00,
- 0x2d, 0x10, 0xaa, 0x27, 0x00, 0x00,
- 0x31, 0x44, 0xa5, 0x32, 0x80, 0x00,
- 0xc0, 0x23, 0x62, 0x61, 0x80, 0x00,
- 0x11, 0x25, 0x3c, 0x5c, 0x00, 0x00,
- 0xe8, 0x80, 0x51, 0x35, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_24[144] =
- {
- 0x84, 0x31, 0x51, 0x84, 0x80, 0x00,
- 0x18, 0xa2, 0xa2, 0x27, 0x00, 0x00,
- 0x4e, 0x01, 0x95, 0x51, 0x80, 0x00,
- 0x44, 0xc8, 0x4a, 0x1a, 0x00, 0x00,
- 0x0e, 0x90, 0x30, 0x68, 0x00, 0x00,
- 0x20, 0xcc, 0x2c, 0x89, 0x00, 0x00,
- 0x93, 0x40, 0x55, 0x8c, 0x80, 0x00,
- 0x2d, 0x10, 0xaa, 0x27, 0x00, 0x00,
- 0x31, 0x44, 0xa5, 0x32, 0x80, 0x00,
- 0xc0, 0x23, 0x62, 0x61, 0x80, 0x00,
- 0x11, 0x25, 0x3c, 0x5c, 0x00, 0x00,
- 0xe8, 0x80, 0x51, 0x35, 0x00, 0x00,
- 0x55, 0x55, 0x55, 0x8c, 0x80, 0x00,
- 0x2a, 0x22, 0xaa, 0x27, 0x00, 0x00,
- 0x31, 0x11, 0xa5, 0x32, 0x80, 0x00,
- 0x83, 0x42, 0x62, 0x61, 0x80, 0x00,
- 0x06, 0x98, 0x3c, 0x5c, 0x00, 0x00,
- 0x40, 0xe1, 0x51, 0x84, 0x80, 0x00,
- 0x2c, 0x44, 0xa2, 0x27, 0x00, 0x00,
- 0xd8, 0x28, 0x95, 0x51, 0x80, 0x00,
- 0x92, 0x81, 0x4a, 0x1a, 0x00, 0x00,
- 0x84, 0x32, 0x30, 0x68, 0x00, 0x00,
- 0x68, 0x0c, 0x2c, 0x89, 0x00, 0x00,
- 0xdc, 0x4e, 0xfc, 0x70, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_25[150] =
- {
- 0x84, 0x31, 0x51, 0x84, 0x80, 0x00,
- 0x18, 0xa2, 0xa2, 0x27, 0x00, 0x00,
- 0x4e, 0x01, 0x95, 0x51, 0x80, 0x00,
- 0x44, 0xc8, 0x4a, 0x1a, 0x00, 0x00,
- 0x0e, 0x90, 0x30, 0x68, 0x00, 0x00,
- 0x20, 0xcc, 0x2c, 0x89, 0x00, 0x00,
- 0x93, 0x40, 0x55, 0x8c, 0x80, 0x00,
- 0x2d, 0x10, 0xaa, 0x27, 0x00, 0x00,
- 0x31, 0x44, 0xa5, 0x32, 0x80, 0x00,
- 0xc0, 0x23, 0x62, 0x61, 0x80, 0x00,
- 0x11, 0x25, 0x3c, 0x5c, 0x00, 0x00,
- 0xe8, 0x80, 0x51, 0x35, 0x00, 0x00,
- 0x45, 0x15, 0x51, 0x84, 0x80, 0x00,
- 0x22, 0x22, 0xa2, 0x27, 0x00, 0x00,
- 0x96, 0x0c, 0x95, 0x51, 0x80, 0x00,
- 0x0c, 0x50, 0x4a, 0x1a, 0x00, 0x00,
- 0x62, 0x04, 0x30, 0x68, 0x00, 0x00,
- 0x49, 0x06, 0x2c, 0x89, 0x00, 0x00,
- 0x11, 0x82, 0x15, 0x8c, 0x00, 0x00,
- 0x12, 0x38, 0x8a, 0x47, 0x00, 0x00,
- 0x40, 0x71, 0x25, 0x81, 0x80, 0x00,
- 0xa8, 0x8a, 0x62, 0x12, 0x80, 0x00,
- 0x08, 0xa1, 0x58, 0x58, 0x00, 0x00,
- 0xa0, 0xc0, 0x0e, 0x28, 0x80, 0x00,
- 0xc5, 0x10, 0x83, 0x34, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_26[156] =
- {
- 0x45, 0x15, 0x51, 0x84, 0x80, 0x00,
- 0x22, 0x22, 0xa2, 0x27, 0x00, 0x00,
- 0x96, 0x0c, 0x95, 0x51, 0x80, 0x00,
- 0x0c, 0x50, 0x4a, 0x1a, 0x00, 0x00,
- 0x62, 0x04, 0x30, 0x68, 0x00, 0x00,
- 0x49, 0x06, 0x2c, 0x89, 0x00, 0x00,
- 0x11, 0x82, 0x15, 0x8c, 0x00, 0x00,
- 0x12, 0x38, 0x8a, 0x47, 0x00, 0x00,
- 0x40, 0x71, 0x25, 0x81, 0x80, 0x00,
- 0xa8, 0x8a, 0x62, 0x12, 0x80, 0x00,
- 0x08, 0xa1, 0x58, 0x58, 0x00, 0x00,
- 0xa0, 0xc0, 0x0e, 0x28, 0x80, 0x00,
- 0xc5, 0x10, 0x83, 0x34, 0x00, 0x00,
- 0x84, 0x31, 0x51, 0x84, 0x80, 0x00,
- 0x18, 0xa2, 0xa2, 0x27, 0x00, 0x00,
- 0x4e, 0x01, 0x95, 0x51, 0x80, 0x00,
- 0x44, 0xc8, 0x4a, 0x1a, 0x00, 0x00,
- 0x0e, 0x90, 0x30, 0x68, 0x00, 0x00,
- 0x20, 0xcc, 0x2c, 0x89, 0x00, 0x00,
- 0x93, 0x40, 0x55, 0x8c, 0x80, 0x00,
- 0x2d, 0x10, 0xaa, 0x27, 0x00, 0x00,
- 0x31, 0x44, 0xa5, 0x32, 0x80, 0x00,
- 0xc0, 0x23, 0x62, 0x61, 0x80, 0x00,
- 0x11, 0x25, 0x3c, 0x5c, 0x00, 0x00,
- 0xe8, 0x80, 0x51, 0x35, 0x00, 0x00,
- 0xa4, 0xa4, 0xfc, 0x91, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_27[162] =
- {
- 0x45, 0x15, 0x51, 0x84, 0x80, 0x00,
- 0x22, 0x22, 0xa2, 0x27, 0x00, 0x00,
- 0x96, 0x0c, 0x95, 0x51, 0x80, 0x00,
- 0x0c, 0x50, 0x4a, 0x1a, 0x00, 0x00,
- 0x62, 0x04, 0x30, 0x68, 0x00, 0x00,
- 0x49, 0x06, 0x2c, 0x89, 0x00, 0x00,
- 0x11, 0x82, 0x15, 0x8c, 0x00, 0x00,
- 0x12, 0x38, 0x8a, 0x47, 0x00, 0x00,
- 0x40, 0x71, 0x25, 0x81, 0x80, 0x00,
- 0xa8, 0x8a, 0x62, 0x12, 0x80, 0x00,
- 0x08, 0xa1, 0x58, 0x58, 0x00, 0x00,
- 0xa0, 0xc0, 0x0e, 0x28, 0x80, 0x00,
- 0xc5, 0x10, 0x83, 0x34, 0x00, 0x00,
- 0x45, 0x51, 0x15, 0x8c, 0x00, 0x00,
- 0x22, 0x0a, 0x8a, 0x47, 0x00, 0x00,
- 0x84, 0xd0, 0x25, 0x81, 0x80, 0x00,
- 0x0c, 0x8a, 0x62, 0x12, 0x80, 0x00,
- 0x18, 0x06, 0x58, 0x58, 0x00, 0x00,
- 0x30, 0x03, 0x0e, 0x28, 0x80, 0x00,
- 0x61, 0x08, 0x83, 0x34, 0x00, 0x00,
- 0x40, 0x11, 0x51, 0x84, 0x80, 0x00,
- 0x10, 0x2c, 0xa2, 0x27, 0x00, 0x00,
- 0x09, 0x60, 0x95, 0x51, 0x80, 0x00,
- 0x00, 0x94, 0x4a, 0x1a, 0x00, 0x00,
- 0x52, 0x40, 0x30, 0x68, 0x00, 0x00,
- 0xa4, 0x24, 0x2c, 0x89, 0x00, 0x00,
- 0x82, 0x88, 0xb0, 0xde, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask33_28[168] =
- {
- 0x45, 0x51, 0x15, 0x8c, 0x00, 0x00,
- 0x22, 0x0a, 0x8a, 0x47, 0x00, 0x00,
- 0x84, 0xd0, 0x25, 0x81, 0x80, 0x00,
- 0x0c, 0x8a, 0x62, 0x12, 0x80, 0x00,
- 0x18, 0x06, 0x58, 0x58, 0x00, 0x00,
- 0x30, 0x03, 0x0e, 0x28, 0x80, 0x00,
- 0x61, 0x08, 0x83, 0x34, 0x00, 0x00,
- 0x40, 0x11, 0x51, 0x84, 0x80, 0x00,
- 0x10, 0x2c, 0xa2, 0x27, 0x00, 0x00,
- 0x09, 0x60, 0x95, 0x51, 0x80, 0x00,
- 0x00, 0x94, 0x4a, 0x1a, 0x00, 0x00,
- 0x52, 0x40, 0x30, 0x68, 0x00, 0x00,
- 0xa4, 0x24, 0x2c, 0x89, 0x00, 0x00,
- 0x82, 0x88, 0xb0, 0xde, 0x80, 0x00,
- 0x45, 0x15, 0x51, 0x84, 0x80, 0x00,
- 0x22, 0x22, 0xa2, 0x27, 0x00, 0x00,
- 0x96, 0x0c, 0x95, 0x51, 0x80, 0x00,
- 0x0c, 0x50, 0x4a, 0x1a, 0x00, 0x00,
- 0x62, 0x04, 0x30, 0x68, 0x00, 0x00,
- 0x49, 0x06, 0x2c, 0x89, 0x00, 0x00,
- 0x11, 0x82, 0x15, 0x8c, 0x00, 0x00,
- 0x12, 0x38, 0x8a, 0x47, 0x00, 0x00,
- 0x40, 0x71, 0x25, 0x81, 0x80, 0x00,
- 0xa8, 0x8a, 0x62, 0x12, 0x80, 0x00,
- 0x08, 0xa1, 0x58, 0x58, 0x00, 0x00,
- 0xa0, 0xc0, 0x0e, 0x28, 0x80, 0x00,
- 0xc5, 0x10, 0x83, 0x34, 0x00, 0x00,
- 0x1b, 0xf4, 0xaa, 0xec, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask33_29[174] =
- {
- 0x45, 0x51, 0x15, 0x8c, 0x00, 0x00,
- 0x22, 0x0a, 0x8a, 0x47, 0x00, 0x00,
- 0x84, 0xd0, 0x25, 0x81, 0x80, 0x00,
- 0x0c, 0x8a, 0x62, 0x12, 0x80, 0x00,
- 0x18, 0x06, 0x58, 0x58, 0x00, 0x00,
- 0x30, 0x03, 0x0e, 0x28, 0x80, 0x00,
- 0x61, 0x08, 0x83, 0x34, 0x00, 0x00,
- 0x40, 0x11, 0x51, 0x84, 0x80, 0x00,
- 0x10, 0x2c, 0xa2, 0x27, 0x00, 0x00,
- 0x09, 0x60, 0x95, 0x51, 0x80, 0x00,
- 0x00, 0x94, 0x4a, 0x1a, 0x00, 0x00,
- 0x52, 0x40, 0x30, 0x68, 0x00, 0x00,
- 0xa4, 0x24, 0x2c, 0x89, 0x00, 0x00,
- 0x82, 0x88, 0xb0, 0xde, 0x80, 0x00,
- 0x55, 0x11, 0x15, 0x8c, 0x00, 0x00,
- 0x22, 0x22, 0x8a, 0x47, 0x00, 0x00,
- 0x11, 0x11, 0x25, 0x81, 0x80, 0x00,
- 0x80, 0x45, 0x62, 0x12, 0x80, 0x00,
- 0x20, 0x1a, 0x58, 0x58, 0x00, 0x00,
- 0x08, 0x68, 0x0e, 0x28, 0x80, 0x00,
- 0x22, 0x84, 0x83, 0x34, 0x00, 0x00,
- 0x48, 0x09, 0x25, 0x2c, 0x00, 0x00,
- 0x07, 0x01, 0x8a, 0x91, 0x00, 0x00,
- 0x94, 0x20, 0x91, 0xc0, 0x80, 0x00,
- 0x82, 0x06, 0x68, 0x06, 0x80, 0x00,
- 0x60, 0x48, 0x32, 0xc8, 0x00, 0x00,
- 0x89, 0x80, 0x43, 0x45, 0x00, 0x00,
- 0x00, 0x8e, 0xc4, 0x30, 0x80, 0x00,
- 0x18, 0x22, 0x1c, 0xa2, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_3[18] =
- {
- 0xad, 0x2d, 0xcd, 0xcc, 0x00, 0x00,
- 0x76, 0x36, 0x97, 0x27, 0x00, 0x00,
- 0x26, 0xdb, 0xb8, 0xd1, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask33_30[180] =
- {
- 0x55, 0x11, 0x15, 0x8c, 0x00, 0x00,
- 0x22, 0x22, 0x8a, 0x47, 0x00, 0x00,
- 0x11, 0x11, 0x25, 0x81, 0x80, 0x00,
- 0x80, 0x45, 0x62, 0x12, 0x80, 0x00,
- 0x20, 0x1a, 0x58, 0x58, 0x00, 0x00,
- 0x08, 0x68, 0x0e, 0x28, 0x80, 0x00,
- 0x22, 0x84, 0x83, 0x34, 0x00, 0x00,
- 0x48, 0x09, 0x25, 0x2c, 0x00, 0x00,
- 0x07, 0x01, 0x8a, 0x91, 0x00, 0x00,
- 0x94, 0x20, 0x91, 0xc0, 0x80, 0x00,
- 0x82, 0x06, 0x68, 0x06, 0x80, 0x00,
- 0x60, 0x48, 0x32, 0xc8, 0x00, 0x00,
- 0x89, 0x80, 0x43, 0x45, 0x00, 0x00,
- 0x00, 0x8e, 0xc4, 0x30, 0x80, 0x00,
- 0x18, 0x22, 0x1c, 0xa2, 0x00, 0x00,
- 0x45, 0x51, 0x15, 0x8c, 0x00, 0x00,
- 0x22, 0x0a, 0x8a, 0x47, 0x00, 0x00,
- 0x84, 0xd0, 0x25, 0x81, 0x80, 0x00,
- 0x0c, 0x8a, 0x62, 0x12, 0x80, 0x00,
- 0x18, 0x06, 0x58, 0x58, 0x00, 0x00,
- 0x30, 0x03, 0x0e, 0x28, 0x80, 0x00,
- 0x61, 0x08, 0x83, 0x34, 0x00, 0x00,
- 0x40, 0x11, 0x51, 0x84, 0x80, 0x00,
- 0x10, 0x2c, 0xa2, 0x27, 0x00, 0x00,
- 0x09, 0x60, 0x95, 0x51, 0x80, 0x00,
- 0x00, 0x94, 0x4a, 0x1a, 0x00, 0x00,
- 0x52, 0x40, 0x30, 0x68, 0x00, 0x00,
- 0xa4, 0x24, 0x2c, 0x89, 0x00, 0x00,
- 0x82, 0x88, 0xb0, 0xde, 0x80, 0x00,
- 0x6d, 0xd2, 0x8c, 0x00, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_31[186] =
- {
- 0x55, 0x11, 0x15, 0x8c, 0x00, 0x00,
- 0x22, 0x22, 0x8a, 0x47, 0x00, 0x00,
- 0x11, 0x11, 0x25, 0x81, 0x80, 0x00,
- 0x80, 0x45, 0x62, 0x12, 0x80, 0x00,
- 0x20, 0x1a, 0x58, 0x58, 0x00, 0x00,
- 0x08, 0x68, 0x0e, 0x28, 0x80, 0x00,
- 0x22, 0x84, 0x83, 0x34, 0x00, 0x00,
- 0x48, 0x09, 0x25, 0x2c, 0x00, 0x00,
- 0x07, 0x01, 0x8a, 0x91, 0x00, 0x00,
- 0x94, 0x20, 0x91, 0xc0, 0x80, 0x00,
- 0x82, 0x06, 0x68, 0x06, 0x80, 0x00,
- 0x60, 0x48, 0x32, 0xc8, 0x00, 0x00,
- 0x89, 0x80, 0x43, 0x45, 0x00, 0x00,
- 0x00, 0x8e, 0xc4, 0x30, 0x80, 0x00,
- 0x18, 0x22, 0x1c, 0xa2, 0x00, 0x00,
- 0xa4, 0x10, 0x25, 0x2c, 0x00, 0x00,
- 0x01, 0x2a, 0x8a, 0x91, 0x00, 0x00,
- 0x06, 0x42, 0x91, 0xc0, 0x80, 0x00,
- 0x08, 0x68, 0x68, 0x06, 0x80, 0x00,
- 0x81, 0x90, 0x32, 0xc8, 0x00, 0x00,
- 0x00, 0xf0, 0x43, 0x45, 0x00, 0x00,
- 0x50, 0x05, 0xc4, 0x30, 0x80, 0x00,
- 0x20, 0x51, 0x1c, 0xa2, 0x00, 0x00,
- 0x43, 0x08, 0x15, 0x8c, 0x00, 0x00,
- 0x68, 0x80, 0x8a, 0x47, 0x00, 0x00,
- 0x80, 0x0b, 0x25, 0x81, 0x80, 0x00,
- 0x10, 0x4c, 0x62, 0x12, 0x80, 0x00,
- 0x12, 0x30, 0x58, 0x58, 0x00, 0x00,
- 0x40, 0x85, 0x0e, 0x28, 0x80, 0x00,
- 0x0e, 0x04, 0x83, 0x34, 0x00, 0x00,
- 0x18, 0x12, 0x0a, 0x1c, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_32[192] =
- {
- 0xa4, 0x10, 0x25, 0x2c, 0x00, 0x00,
- 0x01, 0x2a, 0x8a, 0x91, 0x00, 0x00,
- 0x06, 0x42, 0x91, 0xc0, 0x80, 0x00,
- 0x08, 0x68, 0x68, 0x06, 0x80, 0x00,
- 0x81, 0x90, 0x32, 0xc8, 0x00, 0x00,
- 0x00, 0xf0, 0x43, 0x45, 0x00, 0x00,
- 0x50, 0x05, 0xc4, 0x30, 0x80, 0x00,
- 0x20, 0x51, 0x1c, 0xa2, 0x00, 0x00,
- 0x43, 0x08, 0x15, 0x8c, 0x00, 0x00,
- 0x68, 0x80, 0x8a, 0x47, 0x00, 0x00,
- 0x80, 0x0b, 0x25, 0x81, 0x80, 0x00,
- 0x10, 0x4c, 0x62, 0x12, 0x80, 0x00,
- 0x12, 0x30, 0x58, 0x58, 0x00, 0x00,
- 0x40, 0x85, 0x0e, 0x28, 0x80, 0x00,
- 0x0e, 0x04, 0x83, 0x34, 0x00, 0x00,
- 0x18, 0x12, 0x0a, 0x1c, 0x00, 0x00,
- 0x55, 0x11, 0x15, 0x8c, 0x00, 0x00,
- 0x22, 0x22, 0x8a, 0x47, 0x00, 0x00,
- 0x11, 0x11, 0x25, 0x81, 0x80, 0x00,
- 0x80, 0x45, 0x62, 0x12, 0x80, 0x00,
- 0x20, 0x1a, 0x58, 0x58, 0x00, 0x00,
- 0x08, 0x68, 0x0e, 0x28, 0x80, 0x00,
- 0x22, 0x84, 0x83, 0x34, 0x00, 0x00,
- 0x48, 0x09, 0x25, 0x2c, 0x00, 0x00,
- 0x07, 0x01, 0x8a, 0x91, 0x00, 0x00,
- 0x94, 0x20, 0x91, 0xc0, 0x80, 0x00,
- 0x82, 0x06, 0x68, 0x06, 0x80, 0x00,
- 0x60, 0x48, 0x32, 0xc8, 0x00, 0x00,
- 0x89, 0x80, 0x43, 0x45, 0x00, 0x00,
- 0x00, 0x8e, 0xc4, 0x30, 0x80, 0x00,
- 0x18, 0x22, 0x1c, 0xa2, 0x00, 0x00,
- 0x73, 0x8e, 0x12, 0xca, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask33_33[198] =
- {
- 0xa4, 0x10, 0x25, 0x2c, 0x00, 0x00,
- 0x01, 0x2a, 0x8a, 0x91, 0x00, 0x00,
- 0x06, 0x42, 0x91, 0xc0, 0x80, 0x00,
- 0x08, 0x68, 0x68, 0x06, 0x80, 0x00,
- 0x81, 0x90, 0x32, 0xc8, 0x00, 0x00,
- 0x00, 0xf0, 0x43, 0x45, 0x00, 0x00,
- 0x50, 0x05, 0xc4, 0x30, 0x80, 0x00,
- 0x20, 0x51, 0x1c, 0xa2, 0x00, 0x00,
- 0x43, 0x08, 0x15, 0x8c, 0x00, 0x00,
- 0x68, 0x80, 0x8a, 0x47, 0x00, 0x00,
- 0x80, 0x0b, 0x25, 0x81, 0x80, 0x00,
- 0x10, 0x4c, 0x62, 0x12, 0x80, 0x00,
- 0x12, 0x30, 0x58, 0x58, 0x00, 0x00,
- 0x40, 0x85, 0x0e, 0x28, 0x80, 0x00,
- 0x0e, 0x04, 0x83, 0x34, 0x00, 0x00,
- 0x18, 0x12, 0x0a, 0x1c, 0x00, 0x00,
- 0xa4, 0x10, 0x25, 0x2c, 0x00, 0x00,
- 0x01, 0x2a, 0x8a, 0x91, 0x00, 0x00,
- 0x06, 0x42, 0x91, 0xc0, 0x80, 0x00,
- 0x08, 0x68, 0x68, 0x06, 0x80, 0x00,
- 0x81, 0x90, 0x32, 0xc8, 0x00, 0x00,
- 0x00, 0xf0, 0x43, 0x45, 0x00, 0x00,
- 0x50, 0x05, 0xc4, 0x30, 0x80, 0x00,
- 0x20, 0x51, 0x1c, 0xa2, 0x00, 0x00,
- 0x43, 0x08, 0x25, 0x4c, 0x00, 0x00,
- 0x68, 0x80, 0x8a, 0x66, 0x00, 0x00,
- 0x80, 0x0b, 0x91, 0x91, 0x00, 0x00,
- 0x10, 0x4c, 0x68, 0x42, 0x80, 0x00,
- 0x12, 0x30, 0x32, 0xa4, 0x00, 0x00,
- 0x40, 0x85, 0x43, 0x13, 0x00, 0x00,
- 0x0e, 0x04, 0xc4, 0x30, 0x80, 0x00,
- 0x18, 0x12, 0x1c, 0x88, 0x80, 0x00,
- 0xdb, 0x10, 0x3c, 0x09, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_4[24] =
- {
- 0x55, 0x55, 0xca, 0xec, 0x00, 0x00,
- 0xaa, 0xaa, 0xa9, 0x67, 0x00, 0x00,
- 0x35, 0x35, 0x3a, 0xb1, 0x80, 0x00,
- 0xca, 0xca, 0x55, 0x5a, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask33_5[30] =
- {
- 0x55, 0x55, 0x55, 0x44, 0x80, 0x00,
- 0x2a, 0x2a, 0x2a, 0x66, 0x00, 0x00,
- 0x24, 0x25, 0x25, 0xa1, 0x80, 0x00,
- 0x84, 0xc8, 0xe2, 0x12, 0x80, 0x00,
- 0x10, 0xb6, 0x99, 0x98, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_6[36] =
- {
- 0x51, 0x51, 0xd1, 0x4c, 0x00, 0x00,
- 0x0a, 0x2a, 0xa2, 0xc5, 0x00, 0x00,
- 0xa2, 0x15, 0x95, 0x30, 0x80, 0x00,
- 0x84, 0x4a, 0xca, 0x0a, 0x80, 0x00,
- 0x30, 0x92, 0xa4, 0xaa, 0x00, 0x00,
- 0x04, 0xac, 0x78, 0x15, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_7[42] =
- {
- 0x45, 0x51, 0x15, 0x44, 0x80, 0x00,
- 0x22, 0x2a, 0x8a, 0x23, 0x00, 0x00,
- 0x91, 0x11, 0x85, 0x91, 0x00, 0x00,
- 0x2e, 0x08, 0x32, 0x0a, 0x80, 0x00,
- 0x48, 0x34, 0x58, 0x34, 0x00, 0x00,
- 0x90, 0x29, 0x2c, 0x0d, 0x00, 0x00,
- 0x09, 0x86, 0x43, 0xc8, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_8[48] =
- {
- 0x20, 0x54, 0x64, 0x16, 0x00, 0x00,
- 0x18, 0x88, 0xa2, 0xc2, 0x00, 0x00,
- 0x84, 0x07, 0x51, 0x60, 0x80, 0x00,
- 0x60, 0x48, 0x4a, 0x85, 0x00, 0x00,
- 0x12, 0x82, 0x38, 0x4c, 0x00, 0x00,
- 0x81, 0x41, 0x89, 0x29, 0x00, 0x00,
- 0x40, 0x62, 0x07, 0x11, 0x80, 0x00,
- 0x16, 0x30, 0x94, 0xb0, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask33_9[54] =
- {
- 0x55, 0x51, 0x8e, 0xcc, 0x00, 0x00,
- 0x22, 0x2a, 0x6a, 0x2b, 0x00, 0x00,
- 0x05, 0x85, 0x36, 0x32, 0x80, 0x00,
- 0x09, 0x4a, 0xd1, 0x25, 0x80, 0x00,
- 0x84, 0x32, 0x55, 0x8c, 0x80, 0x00,
- 0xc0, 0x0d, 0xaa, 0x27, 0x00, 0x00,
- 0x20, 0xa6, 0xa5, 0x32, 0x80, 0x00,
- 0x1a, 0x09, 0x62, 0x61, 0x80, 0x00,
- 0x44, 0x64, 0x3c, 0x5c, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask34_1[6] =
- {
- 0xff, 0xff, 0xff, 0xff, 0xc0, 0x00
- };
-
- const WebRtc_UWord8 mask34_10[60] =
- {
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
- 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
- 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
- 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
- 0xc8, 0x02, 0xe4, 0x01, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask34_11[66] =
- {
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask34_12[72] =
- {
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x51, 0x35, 0x28, 0x9a, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask34_13[78] =
- {
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask34_14[84] =
- {
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0xb0, 0xde, 0xd8, 0x6f, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask34_15[90] =
- {
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
- 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
- 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
- 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
- 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
- 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
- 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
- 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask34_16[96] =
- {
- 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
- 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
- 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
- 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
- 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
- 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
- 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
- 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x0a, 0x1c, 0x05, 0x0e, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask34_17[102] =
- {
- 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
- 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
- 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
- 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
- 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
- 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
- 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
- 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
- 0x25, 0x4c, 0x12, 0xa6, 0x00, 0x00,
- 0x8a, 0x66, 0x45, 0x33, 0x00, 0x00,
- 0x91, 0x91, 0x48, 0xc8, 0x80, 0x00,
- 0x68, 0x42, 0xb4, 0x21, 0x40, 0x00,
- 0x32, 0xa4, 0x19, 0x52, 0x00, 0x00,
- 0x43, 0x13, 0x21, 0x89, 0x80, 0x00,
- 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
- 0x1c, 0x88, 0x8e, 0x44, 0x40, 0x00,
- 0x3c, 0x09, 0x1e, 0x04, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask34_18[108] =
- {
- 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
- 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
- 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
- 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x64, 0x16, 0x32, 0x0b, 0x00, 0x00,
- 0xa2, 0xc2, 0x51, 0x61, 0x00, 0x00,
- 0x51, 0x60, 0xa8, 0xb0, 0x40, 0x00,
- 0x4a, 0x85, 0x25, 0x42, 0x80, 0x00,
- 0x38, 0x4c, 0x1c, 0x26, 0x00, 0x00,
- 0x89, 0x29, 0x44, 0x94, 0x80, 0x00,
- 0x07, 0x11, 0x83, 0x88, 0xc0, 0x00,
- 0x94, 0xb0, 0x4a, 0x58, 0x00, 0x00,
- 0x89, 0x70, 0xf3, 0xf7, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask34_19[114] =
- {
- 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
- 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
- 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
- 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
- 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
- 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
- 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
- 0xc8, 0x02, 0xe4, 0x01, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask34_2[12] =
- {
- 0xce, 0xce, 0x67, 0x67, 0x00, 0x00,
- 0xb9, 0x39, 0xdc, 0x9c, 0xc0, 0x00
- };
-
- const WebRtc_UWord8 mask34_20[120] =
- {
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
- 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
- 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
- 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
- 0xc8, 0x02, 0xe4, 0x01, 0x40, 0x00,
- 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
- 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
- 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
- 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x5d, 0xc5, 0xfe, 0xd8, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask34_21[126] =
- {
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
- 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
- 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
- 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
- 0xc8, 0x02, 0xe4, 0x01, 0x40, 0x00,
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask34_22[132] =
- {
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
- 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
- 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
- 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
- 0xc8, 0x02, 0xe4, 0x01, 0x40, 0x00,
- 0x2a, 0xf7, 0x4f, 0xf5, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask34_23[138] =
- {
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x51, 0x35, 0x28, 0x9a, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask34_24[144] =
- {
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x51, 0x35, 0x28, 0x9a, 0x80, 0x00,
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0x4c, 0xb8, 0x04, 0x74, 0xc0, 0x00
- };
-
- const WebRtc_UWord8 mask34_25[150] =
- {
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x51, 0x35, 0x28, 0x9a, 0x80, 0x00,
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask34_26[156] =
- {
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00,
- 0x51, 0x35, 0x28, 0x9a, 0x80, 0x00,
- 0x95, 0x20, 0xe9, 0xef, 0xc0, 0x00
- };
-
- const WebRtc_UWord8 mask34_27[162] =
- {
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0xb0, 0xde, 0xd8, 0x6f, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask34_28[168] =
- {
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0xb0, 0xde, 0xd8, 0x6f, 0x40, 0x00,
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x10, 0x6c, 0xff, 0x60, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask34_29[174] =
- {
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0xb0, 0xde, 0xd8, 0x6f, 0x40, 0x00,
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
- 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
- 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
- 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
- 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
- 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
- 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
- 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask34_3[18] =
- {
- 0xcd, 0xcc, 0x66, 0xe6, 0x00, 0x00,
- 0x97, 0x27, 0x4b, 0x93, 0x80, 0x00,
- 0xb8, 0xd1, 0xdc, 0x68, 0xc0, 0x00
- };
-
- const WebRtc_UWord8 mask34_30[180] =
- {
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
- 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
- 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
- 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
- 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
- 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
- 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
- 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x51, 0x84, 0xa8, 0xc2, 0x40, 0x00,
- 0xa2, 0x27, 0x51, 0x13, 0x80, 0x00,
- 0x95, 0x51, 0xca, 0xa8, 0xc0, 0x00,
- 0x4a, 0x1a, 0x25, 0x0d, 0x00, 0x00,
- 0x30, 0x68, 0x18, 0x34, 0x00, 0x00,
- 0x2c, 0x89, 0x16, 0x44, 0x80, 0x00,
- 0xb0, 0xde, 0xd8, 0x6f, 0x40, 0x00,
- 0x87, 0x93, 0x96, 0xc7, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask34_31[186] =
- {
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
- 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
- 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
- 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
- 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
- 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
- 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
- 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
- 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
- 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
- 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
- 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
- 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
- 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
- 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
- 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x0a, 0x1c, 0x05, 0x0e, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask34_32[192] =
- {
- 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
- 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
- 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
- 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
- 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
- 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
- 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
- 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x0a, 0x1c, 0x05, 0x0e, 0x00, 0x00,
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
- 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
- 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
- 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
- 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
- 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
- 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
- 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
- 0xa6, 0x27, 0xa9, 0x4a, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask34_33[198] =
- {
- 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
- 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
- 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
- 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
- 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
- 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
- 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
- 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x0a, 0x1c, 0x05, 0x0e, 0x00, 0x00,
- 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
- 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
- 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
- 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
- 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
- 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
- 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
- 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
- 0x25, 0x4c, 0x12, 0xa6, 0x00, 0x00,
- 0x8a, 0x66, 0x45, 0x33, 0x00, 0x00,
- 0x91, 0x91, 0x48, 0xc8, 0x80, 0x00,
- 0x68, 0x42, 0xb4, 0x21, 0x40, 0x00,
- 0x32, 0xa4, 0x19, 0x52, 0x00, 0x00,
- 0x43, 0x13, 0x21, 0x89, 0x80, 0x00,
- 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
- 0x1c, 0x88, 0x8e, 0x44, 0x40, 0x00,
- 0x3c, 0x09, 0x1e, 0x04, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask34_34[204] =
- {
- 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
- 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
- 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
- 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
- 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
- 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
- 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
- 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
- 0x25, 0x4c, 0x12, 0xa6, 0x00, 0x00,
- 0x8a, 0x66, 0x45, 0x33, 0x00, 0x00,
- 0x91, 0x91, 0x48, 0xc8, 0x80, 0x00,
- 0x68, 0x42, 0xb4, 0x21, 0x40, 0x00,
- 0x32, 0xa4, 0x19, 0x52, 0x00, 0x00,
- 0x43, 0x13, 0x21, 0x89, 0x80, 0x00,
- 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
- 0x1c, 0x88, 0x8e, 0x44, 0x40, 0x00,
- 0x3c, 0x09, 0x1e, 0x04, 0x80, 0x00,
- 0x25, 0x2c, 0x12, 0x96, 0x00, 0x00,
- 0x8a, 0x91, 0x45, 0x48, 0x80, 0x00,
- 0x91, 0xc0, 0xc8, 0xe0, 0x40, 0x00,
- 0x68, 0x06, 0xb4, 0x03, 0x40, 0x00,
- 0x32, 0xc8, 0x19, 0x64, 0x00, 0x00,
- 0x43, 0x45, 0x21, 0xa2, 0x80, 0x00,
- 0xc4, 0x30, 0xe2, 0x18, 0x40, 0x00,
- 0x1c, 0xa2, 0x0e, 0x51, 0x00, 0x00,
- 0x15, 0x8c, 0x0a, 0xc6, 0x00, 0x00,
- 0x8a, 0x47, 0x45, 0x23, 0x80, 0x00,
- 0x25, 0x81, 0x92, 0xc0, 0xc0, 0x00,
- 0x62, 0x12, 0xb1, 0x09, 0x40, 0x00,
- 0x58, 0x58, 0x2c, 0x2c, 0x00, 0x00,
- 0x0e, 0x28, 0x87, 0x14, 0x40, 0x00,
- 0x83, 0x34, 0x41, 0x9a, 0x00, 0x00,
- 0x0a, 0x1c, 0x05, 0x0e, 0x00, 0x00,
- 0x30, 0x3c, 0xb3, 0xe6, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask34_4[24] =
- {
- 0xca, 0xec, 0x65, 0x76, 0x00, 0x00,
- 0xa9, 0x67, 0x54, 0xb3, 0x80, 0x00,
- 0x3a, 0xb1, 0x9d, 0x58, 0xc0, 0x00,
- 0x55, 0x5a, 0xaa, 0xad, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask34_5[30] =
- {
- 0x55, 0x44, 0xaa, 0xa2, 0x40, 0x00,
- 0x2a, 0x66, 0x15, 0x33, 0x00, 0x00,
- 0x25, 0xa1, 0x92, 0xd0, 0xc0, 0x00,
- 0xe2, 0x12, 0xf1, 0x09, 0x40, 0x00,
- 0x99, 0x98, 0x4c, 0xcc, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask34_6[36] =
- {
- 0xd1, 0x4c, 0x68, 0xa6, 0x00, 0x00,
- 0xa2, 0xc5, 0x51, 0x62, 0x80, 0x00,
- 0x95, 0x30, 0xca, 0x98, 0x40, 0x00,
- 0xca, 0x0a, 0xe5, 0x05, 0x40, 0x00,
- 0xa4, 0xaa, 0x52, 0x55, 0x00, 0x00,
- 0x78, 0x15, 0x3c, 0x0a, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask34_7[42] =
- {
- 0x15, 0x44, 0x8a, 0xa2, 0x40, 0x00,
- 0x8a, 0x23, 0x45, 0x11, 0x80, 0x00,
- 0x85, 0x91, 0x42, 0xc8, 0x80, 0x00,
- 0x32, 0x0a, 0x99, 0x05, 0x40, 0x00,
- 0x58, 0x34, 0x2c, 0x1a, 0x00, 0x00,
- 0x2c, 0x0d, 0x16, 0x06, 0x80, 0x00,
- 0x43, 0xc8, 0x21, 0xe4, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask34_8[48] =
- {
- 0x64, 0x16, 0x32, 0x0b, 0x00, 0x00,
- 0xa2, 0xc2, 0x51, 0x61, 0x00, 0x00,
- 0x51, 0x60, 0xa8, 0xb0, 0x40, 0x00,
- 0x4a, 0x85, 0x25, 0x42, 0x80, 0x00,
- 0x38, 0x4c, 0x1c, 0x26, 0x00, 0x00,
- 0x89, 0x29, 0x44, 0x94, 0x80, 0x00,
- 0x07, 0x11, 0x83, 0x88, 0xc0, 0x00,
- 0x94, 0xb0, 0x4a, 0x58, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask34_9[54] =
- {
- 0x8e, 0xcc, 0x47, 0x66, 0x00, 0x00,
- 0x6a, 0x2b, 0x35, 0x15, 0x80, 0x00,
- 0x36, 0x32, 0x9b, 0x19, 0x40, 0x00,
- 0xd1, 0x25, 0xe8, 0x92, 0xc0, 0x00,
- 0x55, 0x8c, 0xaa, 0xc6, 0x40, 0x00,
- 0xaa, 0x27, 0x55, 0x13, 0x80, 0x00,
- 0xa5, 0x32, 0xd2, 0x99, 0x40, 0x00,
- 0x62, 0x61, 0xb1, 0x30, 0xc0, 0x00,
- 0x3c, 0x5c, 0x1e, 0x2e, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask35_1[6] =
- {
- 0xff, 0xff, 0xff, 0xff, 0xe0, 0x00
- };
-
- const WebRtc_UWord8 mask35_10[60] =
- {
- 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
- 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
- 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
- 0x8e, 0xcc, 0x66, 0x33, 0x00, 0x00,
- 0x6a, 0x2b, 0x15, 0x8a, 0xc0, 0x00,
- 0x36, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0xd1, 0x25, 0x92, 0xc9, 0x60, 0x00,
- 0xc8, 0x02, 0xfe, 0xce, 0xe0, 0x00
- };
-
- const WebRtc_UWord8 mask35_11[66] =
- {
- 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
- 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
- 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask35_12[72] =
- {
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
- 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
- 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
- 0x51, 0x35, 0x2d, 0x86, 0x20, 0x00
- };
-
- const WebRtc_UWord8 mask35_13[78] =
- {
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask35_14[84] =
- {
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0xb0, 0xde, 0xbf, 0xa7, 0xe0, 0x00
- };
-
- const WebRtc_UWord8 mask35_15[90] =
- {
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
- 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
- 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
- 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask35_16[96] =
- {
- 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
- 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
- 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
- 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x0a, 0x1c, 0x77, 0xf9, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask35_17[102] =
- {
- 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
- 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
- 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
- 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
- 0x25, 0x4c, 0x26, 0x13, 0x00, 0x00,
- 0x8a, 0x66, 0x33, 0x19, 0x80, 0x00,
- 0x91, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x68, 0x42, 0xa1, 0x50, 0xa0, 0x00,
- 0x32, 0xa4, 0x52, 0x29, 0x00, 0x00,
- 0x43, 0x13, 0x09, 0x84, 0xc0, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0x88, 0xc4, 0x62, 0x20, 0x00,
- 0x3c, 0x09, 0x04, 0x82, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask35_18[108] =
- {
- 0x8e, 0xcc, 0x22, 0x51, 0x20, 0x00,
- 0x6a, 0x2b, 0x33, 0x13, 0x00, 0x00,
- 0x36, 0x32, 0xc8, 0x24, 0xa0, 0x00,
- 0xd1, 0x25, 0x80, 0xd2, 0xc0, 0x00,
- 0x55, 0x8c, 0x87, 0x09, 0x40, 0x00,
- 0xaa, 0x27, 0x09, 0x85, 0x80, 0x00,
- 0xa5, 0x32, 0x90, 0x68, 0x20, 0x00,
- 0x62, 0x61, 0xe1, 0x28, 0x80, 0x00,
- 0x3c, 0x5c, 0x14, 0x86, 0x40, 0x00,
- 0x64, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0xa2, 0xc2, 0x61, 0x30, 0x80, 0x00,
- 0x51, 0x60, 0xb0, 0x58, 0x20, 0x00,
- 0x4a, 0x85, 0x42, 0xa1, 0x40, 0x00,
- 0x38, 0x4c, 0x26, 0x13, 0x00, 0x00,
- 0x89, 0x29, 0x14, 0x8a, 0x40, 0x00,
- 0x07, 0x11, 0x88, 0xc4, 0x60, 0x00,
- 0x94, 0xb0, 0x58, 0x2c, 0x00, 0x00,
- 0x40, 0xc9, 0x65, 0xbe, 0xc0, 0x00
- };
-
- const WebRtc_UWord8 mask35_19[114] =
- {
- 0x8e, 0xcc, 0x22, 0x51, 0x20, 0x00,
- 0x6a, 0x2b, 0x33, 0x13, 0x00, 0x00,
- 0x36, 0x32, 0xc8, 0x24, 0xa0, 0x00,
- 0xd1, 0x25, 0x80, 0xd2, 0xc0, 0x00,
- 0x55, 0x8c, 0x87, 0x09, 0x40, 0x00,
- 0xaa, 0x27, 0x09, 0x85, 0x80, 0x00,
- 0xa5, 0x32, 0x90, 0x68, 0x20, 0x00,
- 0x62, 0x61, 0xe1, 0x28, 0x80, 0x00,
- 0x3c, 0x5c, 0x14, 0x86, 0x40, 0x00,
- 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
- 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
- 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
- 0x8e, 0xcc, 0x66, 0x33, 0x00, 0x00,
- 0x6a, 0x2b, 0x15, 0x8a, 0xc0, 0x00,
- 0x36, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0xd1, 0x25, 0x92, 0xc9, 0x60, 0x00,
- 0xc8, 0x02, 0xfe, 0xce, 0xe0, 0x00
- };
-
- const WebRtc_UWord8 mask35_2[12] =
- {
- 0xce, 0xce, 0x67, 0x33, 0x80, 0x00,
- 0xb9, 0x39, 0x9c, 0xce, 0x60, 0x00
- };
-
- const WebRtc_UWord8 mask35_20[120] =
- {
- 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
- 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
- 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
- 0x8e, 0xcc, 0x66, 0x33, 0x00, 0x00,
- 0x6a, 0x2b, 0x15, 0x8a, 0xc0, 0x00,
- 0x36, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0xd1, 0x25, 0x92, 0xc9, 0x60, 0x00,
- 0xc8, 0x02, 0xfe, 0xce, 0xe0, 0x00,
- 0x8e, 0xcc, 0x22, 0x51, 0x20, 0x00,
- 0x6a, 0x2b, 0x33, 0x13, 0x00, 0x00,
- 0x36, 0x32, 0xc8, 0x24, 0xa0, 0x00,
- 0xd1, 0x25, 0x80, 0xd2, 0xc0, 0x00,
- 0x55, 0x8c, 0x87, 0x09, 0x40, 0x00,
- 0xaa, 0x27, 0x09, 0x85, 0x80, 0x00,
- 0xa5, 0x32, 0x90, 0x68, 0x20, 0x00,
- 0x62, 0x61, 0xe1, 0x28, 0x80, 0x00,
- 0x3c, 0x5c, 0x14, 0x86, 0x40, 0x00,
- 0x63, 0x36, 0x5c, 0xd3, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask35_21[126] =
- {
- 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
- 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
- 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
- 0x8e, 0xcc, 0x66, 0x33, 0x00, 0x00,
- 0x6a, 0x2b, 0x15, 0x8a, 0xc0, 0x00,
- 0x36, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0xd1, 0x25, 0x92, 0xc9, 0x60, 0x00,
- 0xc8, 0x02, 0xfe, 0xce, 0xe0, 0x00,
- 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
- 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
- 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask35_22[132] =
- {
- 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
- 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
- 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
- 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
- 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
- 0x8e, 0xcc, 0x66, 0x33, 0x00, 0x00,
- 0x6a, 0x2b, 0x15, 0x8a, 0xc0, 0x00,
- 0x36, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0xd1, 0x25, 0x92, 0xc9, 0x60, 0x00,
- 0xc8, 0x02, 0xfe, 0xce, 0xe0, 0x00,
- 0x84, 0xc7, 0xbc, 0xcc, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask35_23[138] =
- {
- 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
- 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
- 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
- 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
- 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
- 0x51, 0x35, 0x2d, 0x86, 0x20, 0x00
- };
-
- const WebRtc_UWord8 mask35_24[144] =
- {
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
- 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
- 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
- 0x51, 0x35, 0x2d, 0x86, 0x20, 0x00,
- 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
- 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
- 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0x0d, 0xfb, 0x06, 0x89, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask35_25[150] =
- {
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
- 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
- 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
- 0x51, 0x35, 0x2d, 0x86, 0x20, 0x00,
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask35_26[156] =
- {
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0x55, 0x8c, 0xc6, 0x63, 0x20, 0x00,
- 0xaa, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0xa5, 0x32, 0x99, 0x4c, 0xa0, 0x00,
- 0x62, 0x61, 0xb0, 0xd8, 0x60, 0x00,
- 0x3c, 0x5c, 0x2e, 0x17, 0x00, 0x00,
- 0x51, 0x35, 0x2d, 0x86, 0x20, 0x00,
- 0xc4, 0x57, 0x70, 0x47, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask35_27[162] =
- {
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0xb0, 0xde, 0xbf, 0xa7, 0xe0, 0x00
- };
-
- const WebRtc_UWord8 mask35_28[168] =
- {
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0xb0, 0xde, 0xbf, 0xa7, 0xe0, 0x00,
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x34, 0x4a, 0x80, 0x94, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask35_29[174] =
- {
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0xb0, 0xde, 0xbf, 0xa7, 0xe0, 0x00,
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
- 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
- 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
- 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask35_3[18] =
- {
- 0xcd, 0xcc, 0x66, 0x33, 0x00, 0x00,
- 0x97, 0x27, 0x13, 0x8a, 0xc0, 0x00,
- 0xb8, 0xd1, 0xc9, 0x64, 0xa0, 0x00
- };
-
- const WebRtc_UWord8 mask35_30[180] =
- {
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
- 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
- 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
- 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x51, 0x84, 0xc2, 0x61, 0x20, 0x00,
- 0xa2, 0x27, 0x13, 0x89, 0xc0, 0x00,
- 0x95, 0x51, 0xa8, 0xd4, 0x60, 0x00,
- 0x4a, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x30, 0x68, 0x34, 0x1a, 0x00, 0x00,
- 0x2c, 0x89, 0x44, 0xa2, 0x40, 0x00,
- 0xb0, 0xde, 0xbf, 0xa7, 0xe0, 0x00,
- 0x32, 0x1b, 0x9f, 0x09, 0x20, 0x00
- };
-
- const WebRtc_UWord8 mask35_31[186] =
- {
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
- 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
- 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
- 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
- 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
- 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
- 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
- 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x0a, 0x1c, 0x77, 0xf9, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask35_32[192] =
- {
- 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
- 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
- 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
- 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x0a, 0x1c, 0x77, 0xf9, 0x00, 0x00,
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
- 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
- 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
- 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
- 0xeb, 0x31, 0x7b, 0x80, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask35_33[198] =
- {
- 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
- 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
- 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
- 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x0a, 0x1c, 0x77, 0xf9, 0x00, 0x00,
- 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
- 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
- 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
- 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
- 0x25, 0x4c, 0x26, 0x13, 0x00, 0x00,
- 0x8a, 0x66, 0x33, 0x19, 0x80, 0x00,
- 0x91, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x68, 0x42, 0xa1, 0x50, 0xa0, 0x00,
- 0x32, 0xa4, 0x52, 0x29, 0x00, 0x00,
- 0x43, 0x13, 0x09, 0x84, 0xc0, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0x88, 0xc4, 0x62, 0x20, 0x00,
- 0x3c, 0x09, 0x04, 0x82, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask35_34[204] =
- {
- 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
- 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
- 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
- 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
- 0x25, 0x4c, 0x26, 0x13, 0x00, 0x00,
- 0x8a, 0x66, 0x33, 0x19, 0x80, 0x00,
- 0x91, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x68, 0x42, 0xa1, 0x50, 0xa0, 0x00,
- 0x32, 0xa4, 0x52, 0x29, 0x00, 0x00,
- 0x43, 0x13, 0x09, 0x84, 0xc0, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0x88, 0xc4, 0x62, 0x20, 0x00,
- 0x3c, 0x09, 0x04, 0x82, 0x40, 0x00,
- 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
- 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
- 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
- 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
- 0x15, 0x8c, 0x46, 0x23, 0x00, 0x00,
- 0x8a, 0x47, 0x23, 0x91, 0xc0, 0x00,
- 0x25, 0x81, 0xc0, 0xe0, 0x60, 0x00,
- 0x62, 0x12, 0x89, 0x44, 0xa0, 0x00,
- 0x58, 0x58, 0x2c, 0x16, 0x00, 0x00,
- 0x0e, 0x28, 0x94, 0x4a, 0x20, 0x00,
- 0x83, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x0a, 0x1c, 0x77, 0xf9, 0x00, 0x00,
- 0x70, 0x07, 0xcd, 0x8c, 0xc0, 0x00
- };
-
- const WebRtc_UWord8 mask35_35[210] =
- {
- 0x25, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x8a, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x91, 0xc0, 0xe0, 0x70, 0x20, 0x00,
- 0x68, 0x06, 0x83, 0x41, 0xa0, 0x00,
- 0x32, 0xc8, 0x64, 0x32, 0x00, 0x00,
- 0x43, 0x45, 0x22, 0x91, 0x40, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0xa2, 0x51, 0x28, 0x80, 0x00,
- 0x25, 0x4c, 0x26, 0x13, 0x00, 0x00,
- 0x8a, 0x66, 0x33, 0x19, 0x80, 0x00,
- 0x91, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x68, 0x42, 0xa1, 0x50, 0xa0, 0x00,
- 0x32, 0xa4, 0x52, 0x29, 0x00, 0x00,
- 0x43, 0x13, 0x09, 0x84, 0xc0, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0x88, 0xc4, 0x62, 0x20, 0x00,
- 0x3c, 0x09, 0x04, 0x82, 0x40, 0x00,
- 0x25, 0x2c, 0x26, 0x13, 0x00, 0x00,
- 0x8a, 0x91, 0x33, 0x19, 0x80, 0x00,
- 0x91, 0xc0, 0xc8, 0xa4, 0x40, 0x00,
- 0x68, 0x06, 0xa1, 0x50, 0xa0, 0x00,
- 0x32, 0xc8, 0x52, 0x29, 0x00, 0x00,
- 0x43, 0x45, 0x09, 0x84, 0xc0, 0x00,
- 0xc4, 0x30, 0x98, 0x4c, 0x20, 0x00,
- 0x1c, 0xa2, 0x44, 0x62, 0x20, 0x00,
- 0x25, 0x4c, 0x04, 0x82, 0x40, 0x00,
- 0x8a, 0x66, 0x16, 0x0b, 0x00, 0x00,
- 0x91, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x68, 0x42, 0xe0, 0x70, 0x20, 0x00,
- 0x32, 0xa4, 0x03, 0x41, 0xa0, 0x00,
- 0x43, 0x13, 0x64, 0x32, 0x00, 0x00,
- 0xc4, 0x30, 0xa2, 0x91, 0x40, 0x00,
- 0x1c, 0x88, 0x98, 0x4c, 0x20, 0x00,
- 0x3c, 0x09, 0x51, 0x28, 0x80, 0x00,
- 0xc2, 0x1c, 0x68, 0x01, 0xa0, 0x00
- };
-
- const WebRtc_UWord8 mask35_4[24] =
- {
- 0xca, 0xec, 0x76, 0x3b, 0x00, 0x00,
- 0xa9, 0x67, 0x33, 0x99, 0xc0, 0x00,
- 0x3a, 0xb1, 0xd8, 0xec, 0x60, 0x00,
- 0x55, 0x5a, 0xad, 0x56, 0xa0, 0x00
- };
-
- const WebRtc_UWord8 mask35_5[30] =
- {
- 0x55, 0x44, 0xa6, 0x53, 0x20, 0x00,
- 0x2a, 0x66, 0x33, 0x19, 0x80, 0x00,
- 0x25, 0xa1, 0x8c, 0xe8, 0x60, 0x00,
- 0xe2, 0x12, 0xce, 0x44, 0xa0, 0x00,
- 0x99, 0x98, 0x71, 0xa6, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask35_6[36] =
- {
- 0xd1, 0x4c, 0x66, 0x13, 0x00, 0x00,
- 0xa2, 0xc5, 0x22, 0xb1, 0x40, 0x00,
- 0x95, 0x30, 0xd8, 0x4c, 0x20, 0x00,
- 0xca, 0x0a, 0xc5, 0x42, 0xa0, 0x00,
- 0xa4, 0xaa, 0x14, 0xa9, 0x80, 0x00,
- 0x78, 0x15, 0x53, 0x05, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask35_7[42] =
- {
- 0x15, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x8a, 0x23, 0x11, 0x88, 0xc0, 0x00,
- 0x85, 0x91, 0x48, 0xa4, 0x40, 0x00,
- 0x32, 0x0a, 0x85, 0x42, 0xa0, 0x00,
- 0x58, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x2c, 0x0d, 0x05, 0x83, 0x40, 0x00,
- 0x43, 0xc8, 0x70, 0x32, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask35_8[48] =
- {
- 0x64, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0xa2, 0xc2, 0x61, 0x30, 0x80, 0x00,
- 0x51, 0x60, 0xb0, 0x58, 0x20, 0x00,
- 0x4a, 0x85, 0x42, 0xa1, 0x40, 0x00,
- 0x38, 0x4c, 0x26, 0x13, 0x00, 0x00,
- 0x89, 0x29, 0x14, 0x8a, 0x40, 0x00,
- 0x07, 0x11, 0x88, 0xc4, 0x60, 0x00,
- 0x94, 0xb0, 0x58, 0x2c, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask35_9[54] =
- {
- 0x8e, 0xcc, 0x22, 0x51, 0x20, 0x00,
- 0x6a, 0x2b, 0x33, 0x13, 0x00, 0x00,
- 0x36, 0x32, 0xc8, 0x24, 0xa0, 0x00,
- 0xd1, 0x25, 0x80, 0xd2, 0xc0, 0x00,
- 0x55, 0x8c, 0x87, 0x09, 0x40, 0x00,
- 0xaa, 0x27, 0x09, 0x85, 0x80, 0x00,
- 0xa5, 0x32, 0x90, 0x68, 0x20, 0x00,
- 0x62, 0x61, 0xe1, 0x28, 0x80, 0x00,
- 0x3c, 0x5c, 0x14, 0x86, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask36_1[6] =
- {
- 0xff, 0xff, 0xff, 0xff, 0xf0, 0x00
- };
-
- const WebRtc_UWord8 mask36_10[60] =
- {
- 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
- 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
- 0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
- 0x2b, 0x15, 0x8a, 0xc5, 0x60, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x25, 0x92, 0xc9, 0x64, 0xb0, 0x00,
- 0xfd, 0x9d, 0xff, 0x67, 0x70, 0x00
- };
-
- const WebRtc_UWord8 mask36_11[66] =
- {
- 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
- 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00
- };
-
- const WebRtc_UWord8 mask36_12[72] =
- {
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
- 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
- 0x5b, 0x0c, 0x56, 0xc3, 0x10, 0x00
- };
-
- const WebRtc_UWord8 mask36_13[78] =
- {
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask36_14[84] =
- {
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x7f, 0x4f, 0xdf, 0xd3, 0xf0, 0x00
- };
-
- const WebRtc_UWord8 mask36_15[90] =
- {
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask36_16[96] =
- {
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0xef, 0xf2, 0x3b, 0xfc, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask36_17[102] =
- {
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
- 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
- 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
- 0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
- 0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
- 0x09, 0x04, 0x82, 0x41, 0x20, 0x00
- };
-
- const WebRtc_UWord8 mask36_18[108] =
- {
- 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
- 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
- 0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
- 0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
- 0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
- 0xd0, 0x03, 0x74, 0x00, 0xd0, 0x00
- };
-
- const WebRtc_UWord8 mask36_19[114] =
- {
- 0x44, 0xa2, 0x51, 0x28, 0x90, 0x00,
- 0x66, 0x26, 0x19, 0x89, 0x80, 0x00,
- 0x90, 0x49, 0x64, 0x12, 0x50, 0x00,
- 0x01, 0xa5, 0x80, 0x69, 0x60, 0x00,
- 0x0e, 0x12, 0x83, 0x84, 0xa0, 0x00,
- 0x13, 0x0b, 0x04, 0xc2, 0xc0, 0x00,
- 0x20, 0xd0, 0x48, 0x34, 0x10, 0x00,
- 0xc2, 0x51, 0x30, 0x94, 0x40, 0x00,
- 0x29, 0x0c, 0x8a, 0x43, 0x20, 0x00,
- 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
- 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
- 0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
- 0x2b, 0x15, 0x8a, 0xc5, 0x60, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x25, 0x92, 0xc9, 0x64, 0xb0, 0x00,
- 0xfd, 0x9d, 0xff, 0x67, 0x70, 0x00
- };
-
- const WebRtc_UWord8 mask36_2[12] =
- {
- 0xce, 0x67, 0x33, 0x99, 0xc0, 0x00,
- 0x39, 0x9c, 0xce, 0x67, 0x30, 0x00
- };
-
- const WebRtc_UWord8 mask36_20[120] =
- {
- 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
- 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
- 0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
- 0x2b, 0x15, 0x8a, 0xc5, 0x60, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x25, 0x92, 0xc9, 0x64, 0xb0, 0x00,
- 0xfd, 0x9d, 0xff, 0x67, 0x70, 0x00,
- 0x44, 0xa2, 0x51, 0x28, 0x90, 0x00,
- 0x66, 0x26, 0x19, 0x89, 0x80, 0x00,
- 0x90, 0x49, 0x64, 0x12, 0x50, 0x00,
- 0x01, 0xa5, 0x80, 0x69, 0x60, 0x00,
- 0x0e, 0x12, 0x83, 0x84, 0xa0, 0x00,
- 0x13, 0x0b, 0x04, 0xc2, 0xc0, 0x00,
- 0x20, 0xd0, 0x48, 0x34, 0x10, 0x00,
- 0xc2, 0x51, 0x30, 0x94, 0x40, 0x00,
- 0x29, 0x0c, 0x8a, 0x43, 0x20, 0x00,
- 0x45, 0xb9, 0x08, 0x16, 0x30, 0x00
- };
-
- const WebRtc_UWord8 mask36_21[126] =
- {
- 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
- 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
- 0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
- 0x2b, 0x15, 0x8a, 0xc5, 0x60, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x25, 0x92, 0xc9, 0x64, 0xb0, 0x00,
- 0xfd, 0x9d, 0xff, 0x67, 0x70, 0x00,
- 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
- 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00
- };
-
- const WebRtc_UWord8 mask36_22[132] =
- {
- 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
- 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
- 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
- 0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
- 0x2b, 0x15, 0x8a, 0xc5, 0x60, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x25, 0x92, 0xc9, 0x64, 0xb0, 0x00,
- 0xfd, 0x9d, 0xff, 0x67, 0x70, 0x00,
- 0x71, 0x04, 0xba, 0x7b, 0xe0, 0x00
- };
-
- const WebRtc_UWord8 mask36_23[138] =
- {
- 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
- 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
- 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
- 0x5b, 0x0c, 0x56, 0xc3, 0x10, 0x00
- };
-
- const WebRtc_UWord8 mask36_24[144] =
- {
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
- 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
- 0x5b, 0x0c, 0x56, 0xc3, 0x10, 0x00,
- 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
- 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x76, 0x3a, 0xeb, 0x17, 0xc0, 0x00
- };
-
- const WebRtc_UWord8 mask36_25[150] =
- {
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
- 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
- 0x5b, 0x0c, 0x56, 0xc3, 0x10, 0x00,
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask36_26[156] =
- {
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x8c, 0xc6, 0x63, 0x31, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x32, 0x99, 0x4c, 0xa6, 0x50, 0x00,
- 0x61, 0xb0, 0xd8, 0x6c, 0x30, 0x00,
- 0x5c, 0x2e, 0x17, 0x0b, 0x80, 0x00,
- 0x5b, 0x0c, 0x56, 0xc3, 0x10, 0x00,
- 0xec, 0x58, 0x0e, 0x6c, 0xe0, 0x00
- };
-
- const WebRtc_UWord8 mask36_27[162] =
- {
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x7f, 0x4f, 0xdf, 0xd3, 0xf0, 0x00
- };
-
- const WebRtc_UWord8 mask36_28[168] =
- {
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x7f, 0x4f, 0xdf, 0xd3, 0xf0, 0x00,
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x29, 0xfd, 0x91, 0x6f, 0xd0, 0x00
- };
-
- const WebRtc_UWord8 mask36_29[174] =
- {
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x7f, 0x4f, 0xdf, 0xd3, 0xf0, 0x00,
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask36_3[18] =
- {
- 0xcc, 0x66, 0x33, 0x19, 0x80, 0x00,
- 0x27, 0x15, 0x89, 0xc5, 0x60, 0x00,
- 0x92, 0xc9, 0x64, 0xb2, 0x50, 0x00
- };
-
- const WebRtc_UWord8 mask36_30[180] =
- {
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x84, 0xc2, 0x61, 0x30, 0x90, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0xe0, 0x00,
- 0x51, 0xa8, 0xd4, 0x6a, 0x30, 0x00,
- 0x1a, 0x0d, 0x06, 0x83, 0x40, 0x00,
- 0x68, 0x34, 0x1a, 0x0d, 0x00, 0x00,
- 0x89, 0x44, 0xa2, 0x51, 0x20, 0x00,
- 0x7f, 0x4f, 0xdf, 0xd3, 0xf0, 0x00,
- 0xc5, 0x38, 0xbb, 0x98, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask36_31[186] =
- {
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0xef, 0xf2, 0x3b, 0xfc, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask36_32[192] =
- {
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0xef, 0xf2, 0x3b, 0xfc, 0x80, 0x00,
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
- 0x3a, 0x28, 0x9c, 0x2f, 0xc0, 0x00
- };
-
- const WebRtc_UWord8 mask36_33[198] =
- {
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0xef, 0xf2, 0x3b, 0xfc, 0x80, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
- 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
- 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
- 0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
- 0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
- 0x09, 0x04, 0x82, 0x41, 0x20, 0x00
- };
-
- const WebRtc_UWord8 mask36_34[204] =
- {
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
- 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
- 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
- 0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
- 0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
- 0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
- 0x8c, 0x46, 0x23, 0x11, 0x80, 0x00,
- 0x47, 0x23, 0x91, 0xc8, 0xe0, 0x00,
- 0x81, 0xc0, 0xe0, 0x70, 0x30, 0x00,
- 0x12, 0x89, 0x44, 0xa2, 0x50, 0x00,
- 0x58, 0x2c, 0x16, 0x0b, 0x00, 0x00,
- 0x28, 0x94, 0x4a, 0x25, 0x10, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0xef, 0xf2, 0x3b, 0xfc, 0x80, 0x00,
- 0xf7, 0x5e, 0x66, 0x5b, 0x60, 0x00
- };
-
- const WebRtc_UWord8 mask36_35[210] =
- {
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
- 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
- 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
- 0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
- 0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
- 0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
- 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
- 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
- 0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
- 0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
- 0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
- 0xd0, 0x03, 0x74, 0x00, 0xd0, 0x00
- };
-
- const WebRtc_UWord8 mask36_36[216] =
- {
- 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
- 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
- 0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
- 0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
- 0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
- 0xd0, 0x03, 0x74, 0x00, 0xd0, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0xc0, 0xe0, 0x70, 0x38, 0x10, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xd0, 0x00,
- 0xc8, 0x64, 0x32, 0x19, 0x00, 0x00,
- 0x45, 0x22, 0x91, 0x48, 0xa0, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0xa2, 0x51, 0x28, 0x94, 0x40, 0x00,
- 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
- 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0x42, 0xa1, 0x50, 0xa8, 0x50, 0x00,
- 0xa4, 0x52, 0x29, 0x14, 0x80, 0x00,
- 0x13, 0x09, 0x84, 0xc2, 0x60, 0x00,
- 0x30, 0x98, 0x4c, 0x26, 0x10, 0x00,
- 0x88, 0xc4, 0x62, 0x31, 0x10, 0x00,
- 0x09, 0x04, 0x82, 0x41, 0x20, 0x00,
- 0xa4, 0x9c, 0x31, 0x13, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask36_4[24] =
- {
- 0xec, 0x76, 0x3b, 0x1d, 0x80, 0x00,
- 0x67, 0x33, 0x99, 0xcc, 0xe0, 0x00,
- 0xb1, 0xd8, 0xec, 0x76, 0x30, 0x00,
- 0x5a, 0xad, 0x56, 0xab, 0x50, 0x00
- };
-
- const WebRtc_UWord8 mask36_5[30] =
- {
- 0x4c, 0xa6, 0x53, 0x29, 0x90, 0x00,
- 0x66, 0x33, 0x19, 0x8c, 0xc0, 0x00,
- 0x19, 0xd0, 0xc6, 0x74, 0x30, 0x00,
- 0x9c, 0x89, 0x67, 0x22, 0x50, 0x00,
- 0xe3, 0x4c, 0x38, 0xd3, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask36_6[36] =
- {
- 0xcc, 0x26, 0x33, 0x09, 0x80, 0x00,
- 0x45, 0x62, 0x91, 0x58, 0xa0, 0x00,
- 0xb0, 0x98, 0x6c, 0x26, 0x10, 0x00,
- 0x8a, 0x85, 0x62, 0xa1, 0x50, 0x00,
- 0x29, 0x53, 0x0a, 0x54, 0xc0, 0x00,
- 0xa6, 0x0a, 0xa9, 0x82, 0xa0, 0x00
- };
-
- const WebRtc_UWord8 mask36_7[42] =
- {
- 0x44, 0xa2, 0x51, 0x28, 0x90, 0x00,
- 0x23, 0x11, 0x88, 0xc4, 0x60, 0x00,
- 0x91, 0x48, 0xa4, 0x52, 0x20, 0x00,
- 0x0a, 0x85, 0x42, 0xa1, 0x50, 0x00,
- 0x34, 0x1a, 0x0d, 0x06, 0x80, 0x00,
- 0x0b, 0x06, 0x82, 0xc1, 0xa0, 0x00,
- 0xe0, 0x64, 0x38, 0x19, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask36_8[48] =
- {
- 0x16, 0x0b, 0x05, 0x82, 0xc0, 0x00,
- 0xc2, 0x61, 0x30, 0x98, 0x40, 0x00,
- 0x60, 0xb0, 0x58, 0x2c, 0x10, 0x00,
- 0x85, 0x42, 0xa1, 0x50, 0xa0, 0x00,
- 0x4c, 0x26, 0x13, 0x09, 0x80, 0x00,
- 0x29, 0x14, 0x8a, 0x45, 0x20, 0x00,
- 0x11, 0x88, 0xc4, 0x62, 0x30, 0x00,
- 0xb0, 0x58, 0x2c, 0x16, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask36_9[54] =
- {
- 0x44, 0xa2, 0x51, 0x28, 0x90, 0x00,
- 0x66, 0x26, 0x19, 0x89, 0x80, 0x00,
- 0x90, 0x49, 0x64, 0x12, 0x50, 0x00,
- 0x01, 0xa5, 0x80, 0x69, 0x60, 0x00,
- 0x0e, 0x12, 0x83, 0x84, 0xa0, 0x00,
- 0x13, 0x0b, 0x04, 0xc2, 0xc0, 0x00,
- 0x20, 0xd0, 0x48, 0x34, 0x10, 0x00,
- 0xc2, 0x51, 0x30, 0x94, 0x40, 0x00,
- 0x29, 0x0c, 0x8a, 0x43, 0x20, 0x00
- };
-
- const WebRtc_UWord8 mask37_1[6] =
- {
- 0xff, 0xff, 0xff, 0xff, 0xf8, 0x00
- };
-
- const WebRtc_UWord8 mask37_10[60] =
- {
- 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
- 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
- 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
- 0xcc, 0x66, 0x33, 0x1d, 0x40, 0x00,
- 0x2b, 0x15, 0x8a, 0xc6, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xb4, 0x98, 0x00,
- 0x25, 0x92, 0xc9, 0x63, 0xa8, 0x00,
- 0xfd, 0x9d, 0xd4, 0x22, 0x30, 0x00
- };
-
- const WebRtc_UWord8 mask37_11[66] =
- {
- 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
- 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
- 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask37_12[72] =
- {
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
- 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
- 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
- 0x5b, 0x0c, 0x64, 0x32, 0x20, 0x00
- };
-
- const WebRtc_UWord8 mask37_13[78] =
- {
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00
- };
-
- const WebRtc_UWord8 mask37_14[84] =
- {
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0x7f, 0x4f, 0xdb, 0x89, 0xd8, 0x00
- };
-
- const WebRtc_UWord8 mask37_15[90] =
- {
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00
- };
-
- const WebRtc_UWord8 mask37_16[96] =
- {
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0xef, 0xf2, 0x1f, 0x9d, 0x78, 0x00
- };
-
- const WebRtc_UWord8 mask37_17[102] =
- {
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
- 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
- 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
- 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
- 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
- 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
- 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
- 0x09, 0x04, 0x82, 0x43, 0x30, 0x00
- };
-
- const WebRtc_UWord8 mask37_18[108] =
- {
- 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
- 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
- 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
- 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
- 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
- 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
- 0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
- 0xd0, 0x03, 0x54, 0x65, 0xc8, 0x00
- };
-
- const WebRtc_UWord8 mask37_19[114] =
- {
- 0x44, 0xa2, 0x51, 0x29, 0xc0, 0x00,
- 0x66, 0x26, 0x19, 0x9c, 0x20, 0x00,
- 0x90, 0x49, 0x44, 0xb0, 0x38, 0x00,
- 0x01, 0xa5, 0xb0, 0xc4, 0x28, 0x00,
- 0x0e, 0x12, 0xa3, 0x0a, 0x50, 0x00,
- 0x13, 0x0b, 0x04, 0x56, 0xc0, 0x00,
- 0x20, 0xd0, 0x48, 0x64, 0xd0, 0x00,
- 0xc2, 0x51, 0x28, 0x8b, 0x00, 0x00,
- 0x29, 0x0c, 0x86, 0x03, 0x38, 0x00,
- 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
- 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
- 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
- 0xcc, 0x66, 0x33, 0x1d, 0x40, 0x00,
- 0x2b, 0x15, 0x8a, 0xc6, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xb4, 0x98, 0x00,
- 0x25, 0x92, 0xc9, 0x63, 0xa8, 0x00,
- 0xfd, 0x9d, 0xd4, 0x22, 0x30, 0x00
- };
-
- const WebRtc_UWord8 mask37_2[12] =
- {
- 0xce, 0x67, 0x33, 0x9d, 0xc0, 0x00,
- 0x39, 0x9c, 0xce, 0x73, 0x38, 0x00
- };
-
- const WebRtc_UWord8 mask37_20[120] =
- {
- 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
- 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
- 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
- 0xcc, 0x66, 0x33, 0x1d, 0x40, 0x00,
- 0x2b, 0x15, 0x8a, 0xc6, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xb4, 0x98, 0x00,
- 0x25, 0x92, 0xc9, 0x63, 0xa8, 0x00,
- 0xfd, 0x9d, 0xd4, 0x22, 0x30, 0x00,
- 0x44, 0xa2, 0x51, 0x29, 0xc0, 0x00,
- 0x66, 0x26, 0x19, 0x9c, 0x20, 0x00,
- 0x90, 0x49, 0x44, 0xb0, 0x38, 0x00,
- 0x01, 0xa5, 0xb0, 0xc4, 0x28, 0x00,
- 0x0e, 0x12, 0xa3, 0x0a, 0x50, 0x00,
- 0x13, 0x0b, 0x04, 0x56, 0xc0, 0x00,
- 0x20, 0xd0, 0x48, 0x64, 0xd0, 0x00,
- 0xc2, 0x51, 0x28, 0x8b, 0x00, 0x00,
- 0x29, 0x0c, 0x86, 0x03, 0x38, 0x00,
- 0xe5, 0x44, 0xda, 0x3a, 0xc8, 0x00
- };
-
- const WebRtc_UWord8 mask37_21[126] =
- {
- 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
- 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
- 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
- 0xcc, 0x66, 0x33, 0x1d, 0x40, 0x00,
- 0x2b, 0x15, 0x8a, 0xc6, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xb4, 0x98, 0x00,
- 0x25, 0x92, 0xc9, 0x63, 0xa8, 0x00,
- 0xfd, 0x9d, 0xd4, 0x22, 0x30, 0x00,
- 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
- 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
- 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask37_22[132] =
- {
- 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
- 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
- 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
- 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
- 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
- 0xcc, 0x66, 0x33, 0x1d, 0x40, 0x00,
- 0x2b, 0x15, 0x8a, 0xc6, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xb4, 0x98, 0x00,
- 0x25, 0x92, 0xc9, 0x63, 0xa8, 0x00,
- 0xfd, 0x9d, 0xd4, 0x22, 0x30, 0x00,
- 0xe4, 0xd3, 0xff, 0x5a, 0x28, 0x00
- };
-
- const WebRtc_UWord8 mask37_23[138] =
- {
- 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
- 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
- 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
- 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
- 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
- 0x5b, 0x0c, 0x64, 0x32, 0x20, 0x00
- };
-
- const WebRtc_UWord8 mask37_24[144] =
- {
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
- 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
- 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
- 0x5b, 0x0c, 0x64, 0x32, 0x20, 0x00,
- 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
- 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
- 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0xad, 0x58, 0xb2, 0x36, 0x68, 0x00
- };
-
- const WebRtc_UWord8 mask37_25[150] =
- {
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
- 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
- 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
- 0x5b, 0x0c, 0x64, 0x32, 0x20, 0x00,
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00
- };
-
- const WebRtc_UWord8 mask37_26[156] =
- {
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0x8c, 0xc6, 0x63, 0x38, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc4, 0x70, 0x00,
- 0x32, 0x99, 0x4c, 0xa3, 0x48, 0x00,
- 0x61, 0xb0, 0xd8, 0x64, 0x98, 0x00,
- 0x5c, 0x2e, 0x17, 0x0e, 0x20, 0x00,
- 0x5b, 0x0c, 0x64, 0x32, 0x20, 0x00,
- 0x7f, 0xb2, 0x5a, 0xaa, 0x20, 0x00
- };
-
- const WebRtc_UWord8 mask37_27[162] =
- {
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0x7f, 0x4f, 0xdb, 0x89, 0xd8, 0x00
- };
-
- const WebRtc_UWord8 mask37_28[168] =
- {
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0x7f, 0x4f, 0xdb, 0x89, 0xd8, 0x00,
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0x7b, 0xc4, 0x24, 0xbf, 0x10, 0x00
- };
-
- const WebRtc_UWord8 mask37_29[174] =
- {
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0x7f, 0x4f, 0xdb, 0x89, 0xd8, 0x00,
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00
- };
-
- const WebRtc_UWord8 mask37_3[18] =
- {
- 0xcc, 0x66, 0x33, 0x19, 0xc0, 0x00,
- 0x27, 0x15, 0x89, 0xcb, 0x30, 0x00,
- 0x92, 0xc9, 0x64, 0xb4, 0x98, 0x00
- };
-
- const WebRtc_UWord8 mask37_30[180] =
- {
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0x84, 0xc2, 0x61, 0x21, 0xc0, 0x00,
- 0x27, 0x13, 0x89, 0xc6, 0x60, 0x00,
- 0x51, 0xa8, 0xd4, 0x62, 0x18, 0x00,
- 0x1a, 0x0d, 0x06, 0x88, 0xa8, 0x00,
- 0x68, 0x34, 0x1a, 0x11, 0x10, 0x00,
- 0x89, 0x44, 0xa2, 0x5c, 0x00, 0x00,
- 0x7f, 0x4f, 0xdb, 0x89, 0xd8, 0x00,
- 0x1d, 0x8e, 0x11, 0xb0, 0xe8, 0x00
- };
-
- const WebRtc_UWord8 mask37_31[186] =
- {
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0xef, 0xf2, 0x1f, 0x9d, 0x78, 0x00
- };
-
- const WebRtc_UWord8 mask37_32[192] =
- {
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0xef, 0xf2, 0x1f, 0x9d, 0x78, 0x00,
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
- 0xf7, 0x95, 0x57, 0x8c, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask37_33[198] =
- {
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0xef, 0xf2, 0x1f, 0x9d, 0x78, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
- 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
- 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
- 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
- 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
- 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
- 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
- 0x09, 0x04, 0x82, 0x43, 0x30, 0x00
- };
-
- const WebRtc_UWord8 mask37_34[204] =
- {
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
- 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
- 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
- 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
- 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
- 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
- 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
- 0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
- 0x8c, 0x46, 0x23, 0x08, 0xc0, 0x00,
- 0x47, 0x23, 0x91, 0xc6, 0x60, 0x00,
- 0x81, 0xc0, 0xe0, 0x62, 0x18, 0x00,
- 0x12, 0x89, 0x44, 0xa1, 0x88, 0x00,
- 0x58, 0x2c, 0x16, 0x05, 0x10, 0x00,
- 0x28, 0x94, 0x4a, 0x32, 0x80, 0x00,
- 0x34, 0x1a, 0x0d, 0x18, 0x20, 0x00,
- 0xef, 0xf2, 0x1f, 0x9d, 0x78, 0x00,
- 0x31, 0x9c, 0xfb, 0x37, 0xc0, 0x00
- };
-
- const WebRtc_UWord8 mask37_35[210] =
- {
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
- 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
- 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
- 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
- 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
- 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
- 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
- 0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
- 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
- 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
- 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
- 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
- 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
- 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
- 0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
- 0xd0, 0x03, 0x54, 0x65, 0xc8, 0x00
- };
-
- const WebRtc_UWord8 mask37_36[216] =
- {
- 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
- 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
- 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
- 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
- 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
- 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
- 0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
- 0xd0, 0x03, 0x54, 0x65, 0xc8, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
- 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
- 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
- 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
- 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
- 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
- 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
- 0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
- 0xc3, 0xc7, 0xce, 0xd8, 0x50, 0x00
- };
-
- const WebRtc_UWord8 mask37_37[222] =
- {
- 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
- 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
- 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
- 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
- 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
- 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
- 0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
- 0x2c, 0x16, 0x0b, 0x05, 0x80, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x30, 0x00,
- 0xc0, 0xe0, 0x70, 0x34, 0x08, 0x00,
- 0x06, 0x83, 0x41, 0xa0, 0xa8, 0x00,
- 0xc8, 0x64, 0x32, 0x03, 0x10, 0x00,
- 0x45, 0x22, 0x91, 0x58, 0x40, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0x50, 0x00,
- 0xa2, 0x51, 0x28, 0x8a, 0x08, 0x00,
- 0xd0, 0x03, 0x54, 0x65, 0xc8, 0x00,
- 0x4c, 0x26, 0x13, 0x09, 0xc0, 0x00,
- 0x66, 0x33, 0x19, 0x9c, 0x60, 0x00,
- 0x91, 0x48, 0xa4, 0x50, 0x38, 0x00,
- 0x42, 0xa1, 0x50, 0xa4, 0x28, 0x00,
- 0xa4, 0x52, 0x29, 0x0a, 0x50, 0x00,
- 0x13, 0x09, 0x84, 0xd6, 0x80, 0x00,
- 0x30, 0x98, 0x4c, 0x24, 0xd0, 0x00,
- 0x88, 0xc4, 0x62, 0x2b, 0x08, 0x00,
- 0x09, 0x04, 0x82, 0x43, 0x30, 0x00,
- 0x2c, 0x16, 0x13, 0x09, 0x80, 0x00,
- 0x91, 0x48, 0x99, 0x8a, 0x20, 0x00,
- 0xc0, 0xe0, 0x64, 0x54, 0x08, 0x00,
- 0x06, 0x83, 0x50, 0xa0, 0x98, 0x00,
- 0xc8, 0x64, 0x29, 0x00, 0x70, 0x00,
- 0x45, 0x22, 0x84, 0xd0, 0xc0, 0x00,
- 0x30, 0x98, 0x4c, 0x25, 0x20, 0x00,
- 0xa2, 0x51, 0x22, 0x28, 0x48, 0x00,
- 0xd0, 0x03, 0x42, 0x53, 0x00, 0x00,
- 0xee, 0xf5, 0xb3, 0x66, 0x10, 0x00
- };
-
- const WebRtc_UWord8 mask37_4[24] =
- {
- 0xec, 0x76, 0x3b, 0x1c, 0xc0, 0x00,
- 0x67, 0x33, 0x99, 0xc6, 0x70, 0x00,
- 0xb1, 0xd8, 0xec, 0x73, 0x18, 0x00,
- 0x5a, 0xad, 0x56, 0xa5, 0xa8, 0x00
- };
-
- const WebRtc_UWord8 mask37_5[30] =
- {
- 0x4c, 0xa6, 0x53, 0x39, 0xc0, 0x00,
- 0x66, 0x33, 0x19, 0x8c, 0x70, 0x00,
- 0x19, 0xd0, 0xe8, 0x73, 0x18, 0x00,
- 0x9c, 0x89, 0x64, 0xa9, 0xa8, 0x00,
- 0xe3, 0x4c, 0x2e, 0x26, 0x60, 0x00
- };
-
- const WebRtc_UWord8 mask37_6[36] =
- {
- 0xcc, 0x26, 0x13, 0x0d, 0x80, 0x00,
- 0x45, 0x62, 0x91, 0x5a, 0x20, 0x00,
- 0xb0, 0x98, 0x4c, 0x34, 0x18, 0x00,
- 0x8a, 0x85, 0x62, 0xa0, 0xa8, 0x00,
- 0x29, 0x53, 0x09, 0x82, 0xd0, 0x00,
- 0xa6, 0x0a, 0xa5, 0x51, 0x40, 0x00
- };
-
- const WebRtc_UWord8 mask37_7[42] =
- {
- 0x44, 0xa2, 0x71, 0x28, 0xc0, 0x00,
- 0x23, 0x11, 0x88, 0xc6, 0x60, 0x00,
- 0x91, 0x48, 0xa4, 0x47, 0x08, 0x00,
- 0x0a, 0x85, 0x52, 0xa0, 0xa8, 0x00,
- 0x34, 0x1a, 0x0d, 0x12, 0x50, 0x00,
- 0x0b, 0x06, 0xa2, 0xd2, 0x80, 0x00,
- 0xe0, 0x64, 0x32, 0x09, 0x30, 0x00
- };
-
- const WebRtc_UWord8 mask37_8[48] =
- {
- 0x16, 0x0b, 0x05, 0x84, 0xe0, 0x00,
- 0xc2, 0x61, 0x30, 0x91, 0x30, 0x00,
- 0x60, 0xb0, 0x58, 0x3a, 0x08, 0x00,
- 0x85, 0x42, 0xa1, 0x44, 0x98, 0x00,
- 0x4c, 0x26, 0x33, 0x08, 0x50, 0x00,
- 0x29, 0x14, 0x8a, 0x58, 0xc0, 0x00,
- 0x11, 0x88, 0xc4, 0x66, 0x30, 0x00,
- 0xb0, 0x58, 0x2c, 0x03, 0x18, 0x00
- };
-
- const WebRtc_UWord8 mask37_9[54] =
- {
- 0x44, 0xa2, 0x51, 0x29, 0xc0, 0x00,
- 0x66, 0x26, 0x19, 0x9c, 0x20, 0x00,
- 0x90, 0x49, 0x44, 0xb0, 0x38, 0x00,
- 0x01, 0xa5, 0xb0, 0xc4, 0x28, 0x00,
- 0x0e, 0x12, 0xa3, 0x0a, 0x50, 0x00,
- 0x13, 0x0b, 0x04, 0x56, 0xc0, 0x00,
- 0x20, 0xd0, 0x48, 0x64, 0xd0, 0x00,
- 0xc2, 0x51, 0x28, 0x8b, 0x00, 0x00,
- 0x29, 0x0c, 0x86, 0x03, 0x38, 0x00
- };
-
- const WebRtc_UWord8 mask38_1[6] =
- {
- 0xff, 0xff, 0xff, 0xff, 0xfc, 0x00
- };
-
- const WebRtc_UWord8 mask38_10[60] =
- {
- 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
- 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
- 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
- 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
- 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
- 0xcc, 0x75, 0x19, 0x8e, 0xa0, 0x00,
- 0x2b, 0x19, 0xc5, 0x63, 0x38, 0x00,
- 0x32, 0xd2, 0x66, 0x5a, 0x4c, 0x00,
- 0x25, 0x8e, 0xa4, 0xb1, 0xd4, 0x00,
- 0x50, 0x88, 0xca, 0x11, 0x18, 0x00
- };
-
- const WebRtc_UWord8 mask38_11[66] =
- {
- 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
- 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
- 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
- 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
- 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask38_12[72] =
- {
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
- 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
- 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
- 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
- 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
- 0x90, 0xc8, 0x92, 0x19, 0x10, 0x00
- };
-
- const WebRtc_UWord8 mask38_13[78] =
- {
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00
- };
-
- const WebRtc_UWord8 mask38_14[84] =
- {
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x6e, 0x27, 0x6d, 0xc4, 0xec, 0x00
- };
-
- const WebRtc_UWord8 mask38_15[90] =
- {
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00
- };
-
- const WebRtc_UWord8 mask38_16[96] =
- {
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x7e, 0x75, 0xef, 0xce, 0xbc, 0x00
- };
-
- const WebRtc_UWord8 mask38_17[102] =
- {
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
- 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
- 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
- 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
- 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
- 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
- 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
- 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
- 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00
- };
-
- const WebRtc_UWord8 mask38_18[108] =
- {
- 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
- 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
- 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
- 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
- 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
- 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
- 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
- 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
- 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x51, 0x97, 0x2a, 0x32, 0xe4, 0x00
- };
-
- const WebRtc_UWord8 mask38_19[114] =
- {
- 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
- 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
- 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
- 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
- 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
- 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
- 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
- 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
- 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
- 0x4c, 0x26, 0x09, 0x84, 0xc0, 0x00,
- 0x66, 0x28, 0x8c, 0xc5, 0x10, 0x00,
- 0x91, 0x50, 0x32, 0x2a, 0x04, 0x00,
- 0x42, 0x82, 0x68, 0x50, 0x4c, 0x00,
- 0xa4, 0x01, 0xd4, 0x80, 0x38, 0x00,
- 0x13, 0x43, 0x02, 0x68, 0x60, 0x00,
- 0x30, 0x94, 0x86, 0x12, 0x90, 0x00,
- 0x88, 0xa1, 0x31, 0x14, 0x24, 0x00,
- 0x09, 0x4c, 0x01, 0x29, 0x80, 0x00,
- 0xcd, 0x98, 0x59, 0xb3, 0x08, 0x00
- };
-
- const WebRtc_UWord8 mask38_2[12] =
- {
- 0xce, 0x77, 0x19, 0xce, 0xe0, 0x00,
- 0x39, 0xcc, 0xe7, 0x39, 0x9c, 0x00
- };
-
- const WebRtc_UWord8 mask38_20[120] =
- {
- 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
- 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
- 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
- 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
- 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
- 0xcc, 0x75, 0x19, 0x8e, 0xa0, 0x00,
- 0x2b, 0x19, 0xc5, 0x63, 0x38, 0x00,
- 0x32, 0xd2, 0x66, 0x5a, 0x4c, 0x00,
- 0x25, 0x8e, 0xa4, 0xb1, 0xd4, 0x00,
- 0x50, 0x88, 0xca, 0x11, 0x18, 0x00,
- 0x44, 0xa7, 0x08, 0x94, 0xe0, 0x00,
- 0x66, 0x70, 0x8c, 0xce, 0x10, 0x00,
- 0x12, 0xc0, 0xe2, 0x58, 0x1c, 0x00,
- 0xc3, 0x10, 0xb8, 0x62, 0x14, 0x00,
- 0x8c, 0x29, 0x51, 0x85, 0x28, 0x00,
- 0x11, 0x5b, 0x02, 0x2b, 0x60, 0x00,
- 0x21, 0x93, 0x44, 0x32, 0x68, 0x00,
- 0xa2, 0x2c, 0x14, 0x45, 0x80, 0x00,
- 0x18, 0x0c, 0xe3, 0x01, 0x9c, 0x00,
- 0xe6, 0xbc, 0x88, 0xe3, 0x78, 0x00
- };
-
- const WebRtc_UWord8 mask38_21[126] =
- {
- 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
- 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
- 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
- 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
- 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
- 0xcc, 0x75, 0x19, 0x8e, 0xa0, 0x00,
- 0x2b, 0x19, 0xc5, 0x63, 0x38, 0x00,
- 0x32, 0xd2, 0x66, 0x5a, 0x4c, 0x00,
- 0x25, 0x8e, 0xa4, 0xb1, 0xd4, 0x00,
- 0x50, 0x88, 0xca, 0x11, 0x18, 0x00,
- 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
- 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
- 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
- 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
- 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask38_22[132] =
- {
- 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
- 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
- 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
- 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
- 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
- 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
- 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
- 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
- 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
- 0xcc, 0x75, 0x19, 0x8e, 0xa0, 0x00,
- 0x2b, 0x19, 0xc5, 0x63, 0x38, 0x00,
- 0x32, 0xd2, 0x66, 0x5a, 0x4c, 0x00,
- 0x25, 0x8e, 0xa4, 0xb1, 0xd4, 0x00,
- 0x50, 0x88, 0xca, 0x11, 0x18, 0x00,
- 0x0c, 0x3c, 0x48, 0x3d, 0x58, 0x00
- };
-
- const WebRtc_UWord8 mask38_23[138] =
- {
- 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
- 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
- 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
- 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
- 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
- 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
- 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
- 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
- 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
- 0x90, 0xc8, 0x92, 0x19, 0x10, 0x00
- };
-
- const WebRtc_UWord8 mask38_24[144] =
- {
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
- 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
- 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
- 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
- 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
- 0x90, 0xc8, 0x92, 0x19, 0x10, 0x00,
- 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
- 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
- 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
- 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
- 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x93, 0xc8, 0xb3, 0xbe, 0x5c, 0x00
- };
-
- const WebRtc_UWord8 mask38_25[150] =
- {
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
- 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
- 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
- 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
- 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
- 0x90, 0xc8, 0x92, 0x19, 0x10, 0x00,
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00
- };
-
- const WebRtc_UWord8 mask38_26[156] =
- {
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x8c, 0xe3, 0x11, 0x9c, 0x60, 0x00,
- 0x27, 0x11, 0xc4, 0xe2, 0x38, 0x00,
- 0x32, 0x8d, 0x26, 0x51, 0xa4, 0x00,
- 0x61, 0x92, 0x6c, 0x32, 0x4c, 0x00,
- 0x5c, 0x38, 0x8b, 0x87, 0x10, 0x00,
- 0x90, 0xc8, 0x92, 0x19, 0x10, 0x00,
- 0x4b, 0xab, 0xfc, 0xe6, 0xe8, 0x00
- };
-
- const WebRtc_UWord8 mask38_27[162] =
- {
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x6e, 0x27, 0x6d, 0xc4, 0xec, 0x00
- };
-
- const WebRtc_UWord8 mask38_28[168] =
- {
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x6e, 0x27, 0x6d, 0xc4, 0xec, 0x00,
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x70, 0x1b, 0x5b, 0x2c, 0x0c, 0x00
- };
-
- const WebRtc_UWord8 mask38_29[174] =
- {
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x6e, 0x27, 0x6d, 0xc4, 0xec, 0x00,
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00
- };
-
- const WebRtc_UWord8 mask38_3[18] =
- {
- 0xcc, 0x67, 0x19, 0x8c, 0xe0, 0x00,
- 0x27, 0x2c, 0xc4, 0xe5, 0x98, 0x00,
- 0x92, 0xd2, 0x72, 0x5a, 0x4c, 0x00
- };
-
- const WebRtc_UWord8 mask38_30[180] =
- {
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x84, 0x87, 0x10, 0x90, 0xe0, 0x00,
- 0x27, 0x19, 0x84, 0xe3, 0x30, 0x00,
- 0x51, 0x88, 0x6a, 0x31, 0x0c, 0x00,
- 0x1a, 0x22, 0xa3, 0x44, 0x54, 0x00,
- 0x68, 0x44, 0x4d, 0x08, 0x88, 0x00,
- 0x89, 0x70, 0x11, 0x2e, 0x00, 0x00,
- 0x6e, 0x27, 0x6d, 0xc4, 0xec, 0x00,
- 0x5b, 0x16, 0xdf, 0xb8, 0xd0, 0x00
- };
-
- const WebRtc_UWord8 mask38_31[186] =
- {
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x7e, 0x75, 0xef, 0xce, 0xbc, 0x00
- };
-
- const WebRtc_UWord8 mask38_32[192] =
- {
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x7e, 0x75, 0xef, 0xce, 0xbc, 0x00,
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x33, 0x10, 0x02, 0x4e, 0x54, 0x00
- };
-
- const WebRtc_UWord8 mask38_33[198] =
- {
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x7e, 0x75, 0xef, 0xce, 0xbc, 0x00,
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
- 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
- 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
- 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
- 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
- 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
- 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
- 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
- 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00
- };
-
- const WebRtc_UWord8 mask38_34[204] =
- {
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
- 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
- 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
- 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
- 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
- 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
- 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
- 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
- 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x8c, 0x23, 0x11, 0x84, 0x60, 0x00,
- 0x47, 0x19, 0x88, 0xe3, 0x30, 0x00,
- 0x81, 0x88, 0x70, 0x31, 0x0c, 0x00,
- 0x12, 0x86, 0x22, 0x50, 0xc4, 0x00,
- 0x58, 0x14, 0x4b, 0x02, 0x88, 0x00,
- 0x28, 0xca, 0x05, 0x19, 0x40, 0x00,
- 0x34, 0x60, 0x86, 0x8c, 0x10, 0x00,
- 0x7e, 0x75, 0xef, 0xce, 0xbc, 0x00,
- 0x91, 0x48, 0xfa, 0xf0, 0xd8, 0x00
- };
-
- const WebRtc_UWord8 mask38_35[210] =
- {
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
- 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
- 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
- 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
- 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
- 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
- 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
- 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
- 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
- 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
- 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
- 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
- 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
- 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
- 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
- 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
- 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
- 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x51, 0x97, 0x2a, 0x32, 0xe4, 0x00
- };
-
- const WebRtc_UWord8 mask38_36[216] =
- {
- 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
- 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
- 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
- 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
- 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
- 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
- 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
- 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
- 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x51, 0x97, 0x2a, 0x32, 0xe4, 0x00,
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
- 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
- 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
- 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
- 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
- 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
- 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
- 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
- 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
- 0x80, 0x95, 0xc2, 0x68, 0x28, 0x00
- };
-
- const WebRtc_UWord8 mask38_37[222] =
- {
- 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
- 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
- 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
- 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
- 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
- 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
- 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
- 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
- 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x51, 0x97, 0x2a, 0x32, 0xe4, 0x00,
- 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
- 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
- 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
- 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
- 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
- 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
- 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
- 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
- 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
- 0x4c, 0x26, 0x09, 0x84, 0xc0, 0x00,
- 0x66, 0x28, 0x8c, 0xc5, 0x10, 0x00,
- 0x91, 0x50, 0x32, 0x2a, 0x04, 0x00,
- 0x42, 0x82, 0x68, 0x50, 0x4c, 0x00,
- 0xa4, 0x01, 0xd4, 0x80, 0x38, 0x00,
- 0x13, 0x43, 0x02, 0x68, 0x60, 0x00,
- 0x30, 0x94, 0x86, 0x12, 0x90, 0x00,
- 0x88, 0xa1, 0x31, 0x14, 0x24, 0x00,
- 0x09, 0x4c, 0x01, 0x29, 0x80, 0x00,
- 0xcd, 0x98, 0x59, 0xb3, 0x08, 0x00
- };
-
- const WebRtc_UWord8 mask38_38[228] =
- {
- 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
- 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
- 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
- 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
- 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
- 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
- 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
- 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
- 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
- 0x4c, 0x26, 0x09, 0x84, 0xc0, 0x00,
- 0x66, 0x28, 0x8c, 0xc5, 0x10, 0x00,
- 0x91, 0x50, 0x32, 0x2a, 0x04, 0x00,
- 0x42, 0x82, 0x68, 0x50, 0x4c, 0x00,
- 0xa4, 0x01, 0xd4, 0x80, 0x38, 0x00,
- 0x13, 0x43, 0x02, 0x68, 0x60, 0x00,
- 0x30, 0x94, 0x86, 0x12, 0x90, 0x00,
- 0x88, 0xa1, 0x31, 0x14, 0x24, 0x00,
- 0x09, 0x4c, 0x01, 0x29, 0x80, 0x00,
- 0xcd, 0x98, 0x59, 0xb3, 0x08, 0x00,
- 0x4c, 0x27, 0x09, 0x84, 0xe0, 0x00,
- 0x66, 0x71, 0x8c, 0xce, 0x30, 0x00,
- 0x91, 0x40, 0xf2, 0x28, 0x1c, 0x00,
- 0x42, 0x90, 0xa8, 0x52, 0x14, 0x00,
- 0xa4, 0x29, 0x54, 0x85, 0x28, 0x00,
- 0x13, 0x5a, 0x02, 0x6b, 0x40, 0x00,
- 0x30, 0x93, 0x46, 0x12, 0x68, 0x00,
- 0x88, 0xac, 0x31, 0x15, 0x84, 0x00,
- 0x09, 0x0c, 0xc1, 0x21, 0x98, 0x00,
- 0x2c, 0x16, 0x05, 0x82, 0xc0, 0x00,
- 0x91, 0x40, 0xd2, 0x28, 0x18, 0x00,
- 0xc0, 0xd0, 0x38, 0x1a, 0x04, 0x00,
- 0x06, 0x82, 0xa0, 0xd0, 0x54, 0x00,
- 0xc8, 0x0c, 0x59, 0x01, 0x88, 0x00,
- 0x45, 0x61, 0x08, 0xac, 0x20, 0x00,
- 0x30, 0x91, 0x46, 0x12, 0x28, 0x00,
- 0xa2, 0x28, 0x34, 0x45, 0x04, 0x00,
- 0x51, 0x97, 0x2a, 0x32, 0xe4, 0x00,
- 0x8c, 0xed, 0x11, 0x5f, 0x24, 0x00
- };
-
- const WebRtc_UWord8 mask38_4[24] =
- {
- 0xec, 0x73, 0x1d, 0x8e, 0x60, 0x00,
- 0x67, 0x19, 0xcc, 0xe3, 0x38, 0x00,
- 0xb1, 0xcc, 0x76, 0x39, 0x8c, 0x00,
- 0x5a, 0x96, 0xab, 0x52, 0xd4, 0x00
- };
-
- const WebRtc_UWord8 mask38_5[30] =
- {
- 0x4c, 0xe7, 0x09, 0x9c, 0xe0, 0x00,
- 0x66, 0x31, 0xcc, 0xc6, 0x38, 0x00,
- 0xa1, 0xcc, 0x74, 0x39, 0x8c, 0x00,
- 0x92, 0xa6, 0xb2, 0x54, 0xd4, 0x00,
- 0xb8, 0x99, 0x97, 0x13, 0x30, 0x00
- };
-
- const WebRtc_UWord8 mask38_6[36] =
- {
- 0x4c, 0x36, 0x09, 0x86, 0xc0, 0x00,
- 0x45, 0x68, 0x88, 0xad, 0x10, 0x00,
- 0x30, 0xd0, 0x66, 0x1a, 0x0c, 0x00,
- 0x8a, 0x82, 0xb1, 0x50, 0x54, 0x00,
- 0x26, 0x0b, 0x44, 0xc1, 0x68, 0x00,
- 0x95, 0x45, 0x12, 0xa8, 0xa0, 0x00
- };
-
- const WebRtc_UWord8 mask38_7[42] =
- {
- 0xc4, 0xa3, 0x18, 0x94, 0x60, 0x00,
- 0x23, 0x19, 0x84, 0x63, 0x30, 0x00,
- 0x91, 0x1c, 0x32, 0x23, 0x84, 0x00,
- 0x4a, 0x82, 0xa9, 0x50, 0x54, 0x00,
- 0x34, 0x49, 0x46, 0x89, 0x28, 0x00,
- 0x8b, 0x4a, 0x11, 0x69, 0x40, 0x00,
- 0xc8, 0x24, 0xd9, 0x04, 0x98, 0x00
- };
-
- const WebRtc_UWord8 mask38_8[48] =
- {
- 0x16, 0x13, 0x82, 0xc2, 0x70, 0x00,
- 0xc2, 0x44, 0xd8, 0x48, 0x98, 0x00,
- 0x60, 0xe8, 0x2c, 0x1d, 0x04, 0x00,
- 0x85, 0x12, 0x70, 0xa2, 0x4c, 0x00,
- 0xcc, 0x21, 0x59, 0x84, 0x28, 0x00,
- 0x29, 0x63, 0x05, 0x2c, 0x60, 0x00,
- 0x11, 0x98, 0xc2, 0x33, 0x18, 0x00,
- 0xb0, 0x0c, 0x76, 0x01, 0x8c, 0x00
- };
-
- const WebRtc_UWord8 mask38_9[54] =
- {
- 0x44, 0xa7, 0x08, 0x94, 0xe0, 0x00,
- 0x66, 0x70, 0x8c, 0xce, 0x10, 0x00,
- 0x12, 0xc0, 0xe2, 0x58, 0x1c, 0x00,
- 0xc3, 0x10, 0xb8, 0x62, 0x14, 0x00,
- 0x8c, 0x29, 0x51, 0x85, 0x28, 0x00,
- 0x11, 0x5b, 0x02, 0x2b, 0x60, 0x00,
- 0x21, 0x93, 0x44, 0x32, 0x68, 0x00,
- 0xa2, 0x2c, 0x14, 0x45, 0x80, 0x00,
- 0x18, 0x0c, 0xe3, 0x01, 0x9c, 0x00
- };
-
- const WebRtc_UWord8 mask39_1[6] =
- {
- 0xff, 0xff, 0xff, 0xff, 0xfe, 0x00
- };
-
- const WebRtc_UWord8 mask39_10[60] =
- {
- 0x8c, 0xe3, 0x09, 0x82, 0x60, 0x00,
- 0x27, 0x11, 0xca, 0x22, 0x88, 0x00,
- 0x32, 0x8d, 0x34, 0x0d, 0x02, 0x00,
- 0x61, 0x92, 0x60, 0x98, 0x26, 0x00,
- 0x5c, 0x38, 0x80, 0x70, 0x1c, 0x00,
- 0xcc, 0x75, 0x10, 0xc4, 0x30, 0x00,
- 0x2b, 0x19, 0xc5, 0x21, 0x48, 0x00,
- 0x32, 0xd2, 0x68, 0x4a, 0x12, 0x00,
- 0x25, 0x8e, 0xb3, 0x04, 0xc0, 0x00,
- 0x50, 0x88, 0xc6, 0x11, 0x84, 0x00
- };
-
- const WebRtc_UWord8 mask39_11[66] =
- {
- 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
- 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
- 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
- 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
- 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask39_12[72] =
- {
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
- 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
- 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
- 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
- 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
- 0x90, 0xc8, 0x9e, 0xbb, 0x88, 0x00
- };
-
- const WebRtc_UWord8 mask39_13[78] =
- {
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00
- };
-
- const WebRtc_UWord8 mask39_14[84] =
- {
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0x6e, 0x27, 0x6a, 0xc7, 0xc4, 0x00
- };
-
- const WebRtc_UWord8 mask39_15[90] =
- {
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00
- };
-
- const WebRtc_UWord8 mask39_16[96] =
- {
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x7e, 0x75, 0xe5, 0x03, 0x8c, 0x00
- };
-
- const WebRtc_UWord8 mask39_17[102] =
- {
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
- 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
- 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
- 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
- 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
- 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
- 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
- 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
- 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00
- };
-
- const WebRtc_UWord8 mask39_18[108] =
- {
- 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
- 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
- 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
- 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
- 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
- 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
- 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
- 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
- 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x51, 0x97, 0x24, 0x2f, 0x7e, 0x00
- };
-
- const WebRtc_UWord8 mask39_19[114] =
- {
- 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
- 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
- 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
- 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
- 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
- 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
- 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
- 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
- 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x4c, 0x26, 0x09, 0x82, 0x60, 0x00,
- 0x66, 0x28, 0x8a, 0x22, 0x88, 0x00,
- 0x91, 0x50, 0x34, 0x0d, 0x02, 0x00,
- 0x42, 0x82, 0x60, 0x98, 0x26, 0x00,
- 0xa4, 0x01, 0xc0, 0x70, 0x1c, 0x00,
- 0x13, 0x43, 0x10, 0xc4, 0x30, 0x00,
- 0x30, 0x94, 0x85, 0x21, 0x48, 0x00,
- 0x88, 0xa1, 0x28, 0x4a, 0x12, 0x00,
- 0x09, 0x4c, 0x13, 0x04, 0xc0, 0x00,
- 0xcd, 0x98, 0x46, 0x11, 0x84, 0x00
- };
-
- const WebRtc_UWord8 mask39_2[12] =
- {
- 0xce, 0x77, 0x1d, 0xc7, 0x70, 0x00,
- 0x39, 0xcc, 0xf3, 0x3c, 0xce, 0x00
- };
-
- const WebRtc_UWord8 mask39_20[120] =
- {
- 0x8c, 0xe3, 0x09, 0x82, 0x60, 0x00,
- 0x27, 0x11, 0xca, 0x22, 0x88, 0x00,
- 0x32, 0x8d, 0x34, 0x0d, 0x02, 0x00,
- 0x61, 0x92, 0x60, 0x98, 0x26, 0x00,
- 0x5c, 0x38, 0x80, 0x70, 0x1c, 0x00,
- 0xcc, 0x75, 0x10, 0xc4, 0x30, 0x00,
- 0x2b, 0x19, 0xc5, 0x21, 0x48, 0x00,
- 0x32, 0xd2, 0x68, 0x4a, 0x12, 0x00,
- 0x25, 0x8e, 0xb3, 0x04, 0xc0, 0x00,
- 0x50, 0x88, 0xc6, 0x11, 0x84, 0x00,
- 0x44, 0xa7, 0x09, 0xc2, 0x70, 0x00,
- 0x66, 0x70, 0x8c, 0x47, 0x18, 0x00,
- 0x12, 0xc0, 0xf0, 0x3c, 0x0e, 0x00,
- 0xc3, 0x10, 0xbc, 0x29, 0x0a, 0x00,
- 0x8c, 0x29, 0x42, 0x72, 0x94, 0x00,
- 0x11, 0x5b, 0x16, 0x85, 0xa0, 0x00,
- 0x21, 0x93, 0x44, 0xd1, 0x34, 0x00,
- 0xa2, 0x2c, 0x0b, 0x0a, 0xc2, 0x00,
- 0x18, 0x0c, 0xe9, 0x30, 0xca, 0x00,
- 0x0d, 0xba, 0x52, 0x38, 0xbc, 0x00
- };
-
- const WebRtc_UWord8 mask39_21[126] =
- {
- 0x8c, 0xe3, 0x09, 0x82, 0x60, 0x00,
- 0x27, 0x11, 0xca, 0x22, 0x88, 0x00,
- 0x32, 0x8d, 0x34, 0x0d, 0x02, 0x00,
- 0x61, 0x92, 0x60, 0x98, 0x26, 0x00,
- 0x5c, 0x38, 0x80, 0x70, 0x1c, 0x00,
- 0xcc, 0x75, 0x10, 0xc4, 0x30, 0x00,
- 0x2b, 0x19, 0xc5, 0x21, 0x48, 0x00,
- 0x32, 0xd2, 0x68, 0x4a, 0x12, 0x00,
- 0x25, 0x8e, 0xb3, 0x04, 0xc0, 0x00,
- 0x50, 0x88, 0xc6, 0x11, 0x84, 0x00,
- 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
- 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
- 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
- 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
- 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00
- };
-
- const WebRtc_UWord8 mask39_22[132] =
- {
- 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
- 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
- 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
- 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
- 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0x8c, 0xe3, 0x09, 0x82, 0x60, 0x00,
- 0x27, 0x11, 0xca, 0x22, 0x88, 0x00,
- 0x32, 0x8d, 0x34, 0x0d, 0x02, 0x00,
- 0x61, 0x92, 0x60, 0x98, 0x26, 0x00,
- 0x5c, 0x38, 0x80, 0x70, 0x1c, 0x00,
- 0xcc, 0x75, 0x10, 0xc4, 0x30, 0x00,
- 0x2b, 0x19, 0xc5, 0x21, 0x48, 0x00,
- 0x32, 0xd2, 0x68, 0x4a, 0x12, 0x00,
- 0x25, 0x8e, 0xb3, 0x04, 0xc0, 0x00,
- 0x50, 0x88, 0xc6, 0x11, 0x84, 0x00,
- 0xfc, 0x5a, 0xb2, 0x13, 0x12, 0x00
- };
-
- const WebRtc_UWord8 mask39_23[138] =
- {
- 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
- 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
- 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
- 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
- 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
- 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
- 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
- 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
- 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
- 0x90, 0xc8, 0x9e, 0xbb, 0x88, 0x00
- };
-
- const WebRtc_UWord8 mask39_24[144] =
- {
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
- 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
- 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
- 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
- 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
- 0x90, 0xc8, 0x9e, 0xbb, 0x88, 0x00,
- 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
- 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
- 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
- 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
- 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0xac, 0xbc, 0xf0, 0xff, 0x62, 0x00
- };
-
- const WebRtc_UWord8 mask39_25[150] =
- {
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
- 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
- 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
- 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
- 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
- 0x90, 0xc8, 0x9e, 0xbb, 0x88, 0x00,
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00
- };
-
- const WebRtc_UWord8 mask39_26[156] =
- {
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0x8c, 0xe3, 0x18, 0xc6, 0x30, 0x00,
- 0x27, 0x11, 0xc4, 0x71, 0x1c, 0x00,
- 0x32, 0x8d, 0x23, 0x48, 0xd2, 0x00,
- 0x61, 0x92, 0x64, 0x99, 0x26, 0x00,
- 0x5c, 0x38, 0x8e, 0x23, 0x88, 0x00,
- 0x90, 0xc8, 0x9e, 0xbb, 0x88, 0x00,
- 0x10, 0x17, 0x44, 0x72, 0xec, 0x00
- };
-
- const WebRtc_UWord8 mask39_27[162] =
- {
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0x6e, 0x27, 0x6a, 0xc7, 0xc4, 0x00
- };
-
- const WebRtc_UWord8 mask39_28[168] =
- {
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0x6e, 0x27, 0x6a, 0xc7, 0xc4, 0x00,
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x86, 0xb6, 0x04, 0xbc, 0x1e, 0x00
- };
-
- const WebRtc_UWord8 mask39_29[174] =
- {
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0x6e, 0x27, 0x6a, 0xc7, 0xc4, 0x00,
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00
- };
-
- const WebRtc_UWord8 mask39_3[18] =
- {
- 0xcc, 0x67, 0x19, 0xc6, 0x70, 0x00,
- 0x27, 0x2c, 0xca, 0xb2, 0xac, 0x00,
- 0x92, 0xd2, 0x76, 0x2d, 0x46, 0x00
- };
-
- const WebRtc_UWord8 mask39_30[180] =
- {
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x84, 0x87, 0x01, 0xc0, 0x70, 0x00,
- 0x27, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x51, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x1a, 0x22, 0xa8, 0xaa, 0x2a, 0x00,
- 0x68, 0x44, 0x51, 0x14, 0x44, 0x00,
- 0x89, 0x70, 0x1c, 0x07, 0x00, 0x00,
- 0x6e, 0x27, 0x6a, 0xc7, 0xc4, 0x00,
- 0xb3, 0x1d, 0x13, 0x03, 0x5a, 0x00
- };
-
- const WebRtc_UWord8 mask39_31[186] =
- {
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x7e, 0x75, 0xe5, 0x03, 0x8c, 0x00
- };
-
- const WebRtc_UWord8 mask39_32[192] =
- {
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x7e, 0x75, 0xe5, 0x03, 0x8c, 0x00,
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x83, 0x1a, 0x3c, 0x2a, 0x7a, 0x00
- };
-
- const WebRtc_UWord8 mask39_33[198] =
- {
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x7e, 0x75, 0xe5, 0x03, 0x8c, 0x00,
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
- 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
- 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
- 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
- 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
- 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
- 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
- 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
- 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00
- };
-
- const WebRtc_UWord8 mask39_34[204] =
- {
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
- 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
- 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
- 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
- 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
- 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
- 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
- 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
- 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x8c, 0x23, 0x08, 0xc2, 0x30, 0x00,
- 0x47, 0x19, 0x86, 0x61, 0x98, 0x00,
- 0x81, 0x88, 0x62, 0x18, 0x86, 0x00,
- 0x12, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0x58, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0x28, 0xca, 0x12, 0x84, 0xa0, 0x00,
- 0x34, 0x60, 0x98, 0x26, 0x08, 0x00,
- 0x7e, 0x75, 0xe5, 0x03, 0x8c, 0x00,
- 0xc6, 0xbb, 0x7e, 0xd9, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask39_35[210] =
- {
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
- 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
- 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
- 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
- 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
- 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
- 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
- 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
- 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
- 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
- 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
- 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
- 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
- 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
- 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
- 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
- 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x51, 0x97, 0x24, 0x2f, 0x7e, 0x00
- };
-
- const WebRtc_UWord8 mask39_36[216] =
- {
- 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
- 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
- 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
- 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
- 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
- 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
- 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
- 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
- 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x51, 0x97, 0x24, 0x2f, 0x7e, 0x00,
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
- 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
- 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
- 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
- 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
- 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
- 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
- 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
- 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x11, 0x78, 0xfe, 0x43, 0xd6, 0x00
- };
-
- const WebRtc_UWord8 mask39_37[222] =
- {
- 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
- 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
- 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
- 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
- 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
- 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
- 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
- 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
- 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x51, 0x97, 0x24, 0x2f, 0x7e, 0x00,
- 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
- 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
- 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
- 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
- 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
- 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
- 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
- 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
- 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x4c, 0x26, 0x09, 0x82, 0x60, 0x00,
- 0x66, 0x28, 0x8a, 0x22, 0x88, 0x00,
- 0x91, 0x50, 0x34, 0x0d, 0x02, 0x00,
- 0x42, 0x82, 0x60, 0x98, 0x26, 0x00,
- 0xa4, 0x01, 0xc0, 0x70, 0x1c, 0x00,
- 0x13, 0x43, 0x10, 0xc4, 0x30, 0x00,
- 0x30, 0x94, 0x85, 0x21, 0x48, 0x00,
- 0x88, 0xa1, 0x28, 0x4a, 0x12, 0x00,
- 0x09, 0x4c, 0x13, 0x04, 0xc0, 0x00,
- 0xcd, 0x98, 0x46, 0x11, 0x84, 0x00
- };
-
- const WebRtc_UWord8 mask39_38[228] =
- {
- 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
- 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
- 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
- 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
- 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
- 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
- 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
- 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
- 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x4c, 0x26, 0x09, 0x82, 0x60, 0x00,
- 0x66, 0x28, 0x8a, 0x22, 0x88, 0x00,
- 0x91, 0x50, 0x34, 0x0d, 0x02, 0x00,
- 0x42, 0x82, 0x60, 0x98, 0x26, 0x00,
- 0xa4, 0x01, 0xc0, 0x70, 0x1c, 0x00,
- 0x13, 0x43, 0x10, 0xc4, 0x30, 0x00,
- 0x30, 0x94, 0x85, 0x21, 0x48, 0x00,
- 0x88, 0xa1, 0x28, 0x4a, 0x12, 0x00,
- 0x09, 0x4c, 0x13, 0x04, 0xc0, 0x00,
- 0xcd, 0x98, 0x46, 0x11, 0x84, 0x00,
- 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
- 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
- 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
- 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
- 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
- 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
- 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
- 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
- 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x2c, 0x16, 0x05, 0x81, 0x60, 0x00,
- 0x91, 0x40, 0xd0, 0x34, 0x0c, 0x00,
- 0xc0, 0xd0, 0x34, 0x0d, 0x02, 0x00,
- 0x06, 0x82, 0xa0, 0xa8, 0x2a, 0x00,
- 0xc8, 0x0c, 0x43, 0x10, 0xc4, 0x00,
- 0x45, 0x61, 0x18, 0x46, 0x10, 0x00,
- 0x30, 0x91, 0x44, 0x51, 0x14, 0x00,
- 0xa2, 0x28, 0x2a, 0x0a, 0x82, 0x00,
- 0x51, 0x97, 0x24, 0x2f, 0x7e, 0x00,
- 0x9e, 0xd8, 0x3c, 0x7e, 0x2e, 0x00
- };
-
- const WebRtc_UWord8 mask39_39[234] =
- {
- 0x4c, 0x27, 0x09, 0xc2, 0x70, 0x00,
- 0x66, 0x71, 0x9c, 0x67, 0x18, 0x00,
- 0x91, 0x40, 0xf0, 0x3c, 0x0e, 0x00,
- 0x42, 0x90, 0xa4, 0x29, 0x0a, 0x00,
- 0xa4, 0x29, 0x4a, 0x52, 0x94, 0x00,
- 0x13, 0x5a, 0x16, 0x85, 0xa0, 0x00,
- 0x30, 0x93, 0x44, 0xd1, 0x34, 0x00,
- 0x88, 0xac, 0x2b, 0x0a, 0xc2, 0x00,
- 0x09, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x4c, 0x26, 0x09, 0x82, 0x60, 0x00,
- 0x66, 0x28, 0x8a, 0x22, 0x88, 0x00,
- 0x91, 0x50, 0x34, 0x0d, 0x02, 0x00,
- 0x42, 0x82, 0x60, 0x98, 0x26, 0x00,
- 0xa4, 0x01, 0xc0, 0x70, 0x1c, 0x00,
- 0x13, 0x43, 0x10, 0xc4, 0x30, 0x00,
- 0x30, 0x94, 0x85, 0x21, 0x48, 0x00,
- 0x88, 0xa1, 0x28, 0x4a, 0x12, 0x00,
- 0x09, 0x4c, 0x13, 0x04, 0xc0, 0x00,
- 0xcd, 0x98, 0x46, 0x11, 0x84, 0x00,
- 0x4c, 0x27, 0x09, 0x82, 0x60, 0x00,
- 0x66, 0x71, 0x8a, 0x22, 0x88, 0x00,
- 0x91, 0x40, 0xf4, 0x0d, 0x02, 0x00,
- 0x42, 0x90, 0xa0, 0x98, 0x26, 0x00,
- 0xa4, 0x29, 0x40, 0x70, 0x1c, 0x00,
- 0x13, 0x5a, 0x10, 0xc4, 0x30, 0x00,
- 0x30, 0x93, 0x45, 0x21, 0x48, 0x00,
- 0x88, 0xac, 0x28, 0x4a, 0x12, 0x00,
- 0x09, 0x0c, 0xd3, 0x04, 0xc0, 0x00,
- 0x4c, 0x26, 0x06, 0x11, 0x84, 0x00,
- 0x66, 0x28, 0x89, 0xc2, 0x70, 0x00,
- 0x91, 0x50, 0x3c, 0x67, 0x18, 0x00,
- 0x42, 0x82, 0x70, 0x3c, 0x0e, 0x00,
- 0xa4, 0x01, 0xc4, 0x29, 0x0a, 0x00,
- 0x13, 0x43, 0x0a, 0x52, 0x94, 0x00,
- 0x30, 0x94, 0x96, 0x85, 0xa0, 0x00,
- 0x88, 0xa1, 0x24, 0xd1, 0x34, 0x00,
- 0x09, 0x4c, 0x0b, 0x0a, 0xc2, 0x00,
- 0xcd, 0x98, 0x43, 0x30, 0xcc, 0x00,
- 0x1d, 0x04, 0x3e, 0xf1, 0xb4, 0x00
- };
-
- const WebRtc_UWord8 mask39_4[24] =
- {
- 0xec, 0x73, 0x1c, 0xc7, 0x30, 0x00,
- 0x67, 0x19, 0xc6, 0x71, 0x9c, 0x00,
- 0xb1, 0xcc, 0x73, 0x1c, 0xc6, 0x00,
- 0x5a, 0x96, 0xa5, 0xa9, 0x6a, 0x00
- };
-
- const WebRtc_UWord8 mask39_5[30] =
- {
- 0x4c, 0xe7, 0x19, 0xc6, 0x70, 0x00,
- 0x66, 0x31, 0xcc, 0x73, 0x1c, 0x00,
- 0xa1, 0xcc, 0x73, 0x1c, 0xa6, 0x00,
- 0x92, 0xa6, 0xa5, 0x6a, 0x6a, 0x00,
- 0xb8, 0x99, 0x96, 0x8b, 0x94, 0x00
- };
-
- const WebRtc_UWord8 mask39_6[36] =
- {
- 0x4c, 0x36, 0x09, 0x83, 0x60, 0x00,
- 0x45, 0x68, 0x8a, 0x26, 0x88, 0x00,
- 0x30, 0xd0, 0x64, 0x1d, 0x06, 0x00,
- 0x8a, 0x82, 0xb0, 0xa8, 0x2a, 0x00,
- 0x26, 0x0b, 0x40, 0xd0, 0xd4, 0x00,
- 0x95, 0x45, 0x13, 0x44, 0x30, 0x00
- };
-
- const WebRtc_UWord8 mask39_7[42] =
- {
- 0xc4, 0xa3, 0x09, 0xc2, 0x30, 0x00,
- 0x23, 0x19, 0x86, 0x65, 0x80, 0x00,
- 0x91, 0x1c, 0x22, 0x01, 0xd6, 0x00,
- 0x4a, 0x82, 0xb0, 0x2a, 0x2a, 0x00,
- 0x34, 0x49, 0x44, 0x98, 0x94, 0x00,
- 0x8b, 0x4a, 0x1a, 0x84, 0x60, 0x00,
- 0xc8, 0x24, 0xc1, 0x94, 0x4c, 0x00
- };
-
- const WebRtc_UWord8 mask39_8[48] =
- {
- 0x16, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xc2, 0x44, 0xd1, 0x34, 0x4c, 0x00,
- 0x60, 0xe8, 0x3a, 0x0e, 0x82, 0x00,
- 0x85, 0x12, 0x64, 0x99, 0x26, 0x00,
- 0xcc, 0x21, 0x5c, 0x52, 0x14, 0x00,
- 0x29, 0x63, 0x18, 0xc6, 0x30, 0x00,
- 0x11, 0x98, 0xc6, 0x31, 0x8c, 0x00,
- 0xb0, 0x0c, 0x63, 0x18, 0xc6, 0x00
- };
-
- const WebRtc_UWord8 mask39_9[54] =
- {
- 0x44, 0xa7, 0x09, 0xc2, 0x70, 0x00,
- 0x66, 0x70, 0x8c, 0x47, 0x18, 0x00,
- 0x12, 0xc0, 0xf0, 0x3c, 0x0e, 0x00,
- 0xc3, 0x10, 0xbc, 0x29, 0x0a, 0x00,
- 0x8c, 0x29, 0x42, 0x72, 0x94, 0x00,
- 0x11, 0x5b, 0x16, 0x85, 0xa0, 0x00,
- 0x21, 0x93, 0x44, 0xd1, 0x34, 0x00,
- 0xa2, 0x2c, 0x0b, 0x0a, 0xc2, 0x00,
- 0x18, 0x0c, 0xe9, 0x30, 0xca, 0x00
- };
-
- const WebRtc_UWord8 mask3_1[2] =
- {
- 0xe0, 0x00
- };
-
- const WebRtc_UWord8 mask3_2[4] =
- {
- 0xc0, 0x00,
- 0xa0, 0x00
- };
-
- const WebRtc_UWord8 mask3_3[6] =
- {
- 0xc0, 0x00,
- 0xa0, 0x00,
- 0x60, 0x00
- };
-
- const WebRtc_UWord8 mask40_1[6] =
- {
- 0xff, 0xff, 0xff, 0xff, 0xff, 0x00
- };
-
- const WebRtc_UWord8 mask40_10[60] =
- {
- 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
- 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
- 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00
- };
-
- const WebRtc_UWord8 mask40_11[66] =
- {
- 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
- 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
- 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
- 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask40_12[72] =
- {
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
- 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
- 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
- 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
- 0xf5, 0xdc, 0x4f, 0x5d, 0xc4, 0x00
- };
-
- const WebRtc_UWord8 mask40_13[78] =
- {
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00
- };
-
- const WebRtc_UWord8 mask40_14[84] =
- {
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0x56, 0x3e, 0x25, 0x63, 0xe2, 0x00
- };
-
- const WebRtc_UWord8 mask40_15[90] =
- {
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00
- };
-
- const WebRtc_UWord8 mask40_16[96] =
- {
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x28, 0x1c, 0x62, 0x81, 0xc6, 0x00
- };
-
- const WebRtc_UWord8 mask40_17[102] =
- {
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00
- };
-
- const WebRtc_UWord8 mask40_18[108] =
- {
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x21, 0x7b, 0xf2, 0x17, 0xbf, 0x00
- };
-
- const WebRtc_UWord8 mask40_19[114] =
- {
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
- 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
- 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
- 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00
- };
-
- const WebRtc_UWord8 mask40_2[12] =
- {
- 0xee, 0x3b, 0x8e, 0xe3, 0xb8, 0x00,
- 0x99, 0xe6, 0x79, 0x9e, 0x67, 0x00
- };
-
- const WebRtc_UWord8 mask40_20[120] =
- {
- 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
- 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
- 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
- 0xf7, 0x8d, 0xaf, 0x78, 0xda, 0x00
- };
-
- const WebRtc_UWord8 mask40_21[126] =
- {
- 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
- 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
- 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
- 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
- 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
- 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
- 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00
- };
-
- const WebRtc_UWord8 mask40_22[132] =
- {
- 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
- 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
- 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
- 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
- 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
- 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
- 0x89, 0xee, 0x1f, 0x38, 0xca, 0x00
- };
-
- const WebRtc_UWord8 mask40_23[138] =
- {
- 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
- 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
- 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
- 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
- 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
- 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
- 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
- 0xf5, 0xdc, 0x4f, 0x5d, 0xc4, 0x00
- };
-
- const WebRtc_UWord8 mask40_24[144] =
- {
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
- 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
- 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
- 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
- 0xf5, 0xdc, 0x4f, 0x5d, 0xc4, 0x00,
- 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
- 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
- 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
- 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0x68, 0xde, 0x83, 0xa9, 0xcf, 0x00
- };
-
- const WebRtc_UWord8 mask40_25[150] =
- {
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
- 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
- 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
- 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
- 0xf5, 0xdc, 0x4f, 0x5d, 0xc4, 0x00,
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00
- };
-
- const WebRtc_UWord8 mask40_26[156] =
- {
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8e, 0x00,
- 0x1a, 0x46, 0x91, 0xa4, 0x69, 0x00,
- 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
- 0x71, 0x1c, 0x47, 0x11, 0xc4, 0x00,
- 0xf5, 0xdc, 0x4f, 0x5d, 0xc4, 0x00,
- 0x06, 0x8e, 0x8c, 0x1a, 0xd2, 0x00
- };
-
- const WebRtc_UWord8 mask40_27[162] =
- {
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0x56, 0x3e, 0x25, 0x63, 0xe2, 0x00
- };
-
- const WebRtc_UWord8 mask40_28[168] =
- {
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0x56, 0x3e, 0x25, 0x63, 0xe2, 0x00,
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x68, 0x0e, 0x9b, 0x52, 0xb6, 0x00
- };
-
- const WebRtc_UWord8 mask40_29[174] =
- {
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0x56, 0x3e, 0x25, 0x63, 0xe2, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00
- };
-
- const WebRtc_UWord8 mask40_3[18] =
- {
- 0xce, 0x33, 0x8c, 0xe3, 0x38, 0x00,
- 0x55, 0x95, 0x65, 0x59, 0x56, 0x00,
- 0xb1, 0x6a, 0x3b, 0x16, 0xa3, 0x00
- };
-
- const WebRtc_UWord8 mask40_30[180] =
- {
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x0e, 0x03, 0x80, 0xe0, 0x38, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x45, 0x51, 0x54, 0x55, 0x15, 0x00,
- 0x88, 0xa2, 0x28, 0x8a, 0x22, 0x00,
- 0xe0, 0x38, 0x0e, 0x03, 0x80, 0x00,
- 0x56, 0x3e, 0x25, 0x63, 0xe2, 0x00,
- 0xe1, 0x47, 0x04, 0x05, 0x47, 0x00
- };
-
- const WebRtc_UWord8 mask40_31[186] =
- {
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x28, 0x1c, 0x62, 0x81, 0xc6, 0x00
- };
-
- const WebRtc_UWord8 mask40_32[192] =
- {
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x28, 0x1c, 0x62, 0x81, 0xc6, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x03, 0x0c, 0x46, 0x10, 0xc5, 0x00
- };
-
- const WebRtc_UWord8 mask40_33[198] =
- {
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x28, 0x1c, 0x62, 0x81, 0xc6, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00
- };
-
- const WebRtc_UWord8 mask40_34[204] =
- {
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x18, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x43, 0x00,
- 0x0c, 0x43, 0x10, 0xc4, 0x31, 0x00,
- 0x28, 0x8a, 0x22, 0x88, 0xa2, 0x00,
- 0x94, 0x25, 0x09, 0x42, 0x50, 0x00,
- 0xc1, 0x30, 0x4c, 0x13, 0x04, 0x00,
- 0x28, 0x1c, 0x62, 0x81, 0xc6, 0x00,
- 0x87, 0x3c, 0x08, 0x19, 0x31, 0x00
- };
-
- const WebRtc_UWord8 mask40_35[210] =
- {
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x21, 0x7b, 0xf2, 0x17, 0xbf, 0x00
- };
-
- const WebRtc_UWord8 mask40_36[216] =
- {
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x21, 0x7b, 0xf2, 0x17, 0xbf, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
- 0x1e, 0xb9, 0x3d, 0x25, 0xcc, 0x00
- };
-
- const WebRtc_UWord8 mask40_37[222] =
- {
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x21, 0x7b, 0xf2, 0x17, 0xbf, 0x00,
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
- 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
- 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
- 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00
- };
-
- const WebRtc_UWord8 mask40_38[228] =
- {
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
- 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
- 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
- 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0xb0, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x06, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x88, 0x62, 0x00,
- 0xc2, 0x30, 0x8c, 0x23, 0x08, 0x00,
- 0x22, 0x88, 0xa2, 0x28, 0x8a, 0x00,
- 0x50, 0x54, 0x15, 0x05, 0x41, 0x00,
- 0x21, 0x7b, 0xf2, 0x17, 0xbf, 0x00,
- 0xea, 0xaa, 0x20, 0xa2, 0x1b, 0x00
- };
-
- const WebRtc_UWord8 mask40_39[234] =
- {
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
- 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
- 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
- 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
- 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
- 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
- 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
- 0xf7, 0x8d, 0xaf, 0x78, 0xda, 0x00
- };
-
- const WebRtc_UWord8 mask40_4[24] =
- {
- 0xe6, 0x39, 0x8e, 0x63, 0x98, 0x00,
- 0x33, 0x8c, 0xe3, 0x38, 0xce, 0x00,
- 0x98, 0xe6, 0x39, 0x8e, 0x63, 0x00,
- 0x2d, 0x4b, 0x52, 0xd4, 0xb5, 0x00
- };
-
- const WebRtc_UWord8 mask40_40[240] =
- {
- 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
- 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
- 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
- 0xf7, 0x8d, 0xaf, 0x78, 0xda, 0x00,
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0xe3, 0x38, 0xce, 0x33, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x85, 0x00,
- 0x52, 0x94, 0xa5, 0x29, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x19, 0x86, 0x61, 0x98, 0x66, 0x00,
- 0x4c, 0x13, 0x04, 0xc1, 0x30, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x44, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0x81, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x38, 0x0e, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x18, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0xa4, 0x00,
- 0x42, 0x50, 0x94, 0x25, 0x09, 0x00,
- 0x98, 0x26, 0x09, 0x82, 0x60, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0xc2, 0x00,
- 0xa6, 0xf3, 0xab, 0x1b, 0x87, 0x00
- };
-
- const WebRtc_UWord8 mask40_5[30] =
- {
- 0xce, 0x33, 0x8c, 0xe3, 0x38, 0x00,
- 0x63, 0x98, 0xe6, 0x39, 0x8e, 0x00,
- 0x98, 0xe5, 0x39, 0x8e, 0x53, 0x00,
- 0x2b, 0x53, 0x52, 0xb5, 0x35, 0x00,
- 0xb4, 0x5c, 0xab, 0x45, 0xca, 0x00
- };
-
- const WebRtc_UWord8 mask40_6[36] =
- {
- 0x4c, 0x1b, 0x04, 0xc1, 0xb0, 0x00,
- 0x51, 0x34, 0x45, 0x13, 0x44, 0x00,
- 0x20, 0xe8, 0x32, 0x0e, 0x83, 0x00,
- 0x85, 0x41, 0x58, 0x54, 0x15, 0x00,
- 0x06, 0x86, 0xa0, 0x68, 0x6a, 0x00,
- 0x9a, 0x21, 0x89, 0xa2, 0x18, 0x00
- };
-
- const WebRtc_UWord8 mask40_7[42] =
- {
- 0x4e, 0x11, 0x84, 0xe1, 0x18, 0x00,
- 0x33, 0x2c, 0x03, 0x32, 0xc0, 0x00,
- 0x10, 0x0e, 0xb1, 0x00, 0xeb, 0x00,
- 0x81, 0x51, 0x58, 0x15, 0x15, 0x00,
- 0x24, 0xc4, 0xa2, 0x4c, 0x4a, 0x00,
- 0xd4, 0x23, 0x0d, 0x42, 0x30, 0x00,
- 0x0c, 0xa2, 0x60, 0xca, 0x26, 0x00
- };
-
- const WebRtc_UWord8 mask40_8[48] =
- {
- 0x27, 0x09, 0xc2, 0x70, 0x9c, 0x00,
- 0x89, 0xa2, 0x68, 0x9a, 0x26, 0x00,
- 0xd0, 0x74, 0x1d, 0x07, 0x41, 0x00,
- 0x24, 0xc9, 0x32, 0x4c, 0x93, 0x00,
- 0xe2, 0x90, 0xae, 0x29, 0x0a, 0x00,
- 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
- 0x31, 0x8c, 0x63, 0x18, 0xc6, 0x00,
- 0x18, 0xc6, 0x31, 0x8c, 0x63, 0x00
- };
-
- const WebRtc_UWord8 mask40_9[54] =
- {
- 0x4e, 0x13, 0x84, 0xe1, 0x38, 0x00,
- 0x62, 0x38, 0xc6, 0x23, 0x8c, 0x00,
- 0x81, 0xe0, 0x78, 0x1e, 0x07, 0x00,
- 0xe1, 0x48, 0x5e, 0x14, 0x85, 0x00,
- 0x13, 0x94, 0xa1, 0x39, 0x4a, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0xd0, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x9a, 0x00,
- 0x58, 0x56, 0x15, 0x85, 0x61, 0x00,
- 0x49, 0x86, 0x54, 0x98, 0x65, 0x00
- };
-
- const WebRtc_UWord8 mask41_1[6] =
- {
- 0xff, 0xff, 0xff, 0xff, 0xff, 0x80
- };
-
- const WebRtc_UWord8 mask41_10[60] =
- {
- 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
- 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
- 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80
- };
-
- const WebRtc_UWord8 mask41_11[66] =
- {
- 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
- 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
- 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
- 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00
- };
-
- const WebRtc_UWord8 mask41_12[72] =
- {
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
- 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
- 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
- 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
- 0xf5, 0xdc, 0x4a, 0x06, 0x51, 0x80
- };
-
- const WebRtc_UWord8 mask41_13[78] =
- {
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80
- };
-
- const WebRtc_UWord8 mask41_14[84] =
- {
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0x56, 0x3e, 0x24, 0xdd, 0x0c, 0x00
- };
-
- const WebRtc_UWord8 mask41_15[90] =
- {
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80
- };
-
- const WebRtc_UWord8 mask41_16[96] =
- {
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x28, 0x1c, 0x63, 0xbf, 0x53, 0x80
- };
-
- const WebRtc_UWord8 mask41_17[102] =
- {
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00
- };
-
- const WebRtc_UWord8 mask41_18[108] =
- {
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0x21, 0x7b, 0xf5, 0xa5, 0x65, 0x80
- };
-
- const WebRtc_UWord8 mask41_19[114] =
- {
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
- 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
- 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
- 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80
- };
-
- const WebRtc_UWord8 mask41_2[12] =
- {
- 0xee, 0x3b, 0x8e, 0xe3, 0xb3, 0x00,
- 0x99, 0xe6, 0x79, 0x9e, 0x6e, 0x80
- };
-
- const WebRtc_UWord8 mask41_20[120] =
- {
- 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
- 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
- 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
- 0xf7, 0x8d, 0xa2, 0xa0, 0x33, 0x00
- };
-
- const WebRtc_UWord8 mask41_21[126] =
- {
- 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
- 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
- 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
- 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
- 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
- 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
- 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00
- };
-
- const WebRtc_UWord8 mask41_22[132] =
- {
- 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
- 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
- 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
- 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
- 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
- 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
- 0x33, 0x09, 0x6e, 0x49, 0x6b, 0x80
- };
-
- const WebRtc_UWord8 mask41_23[138] =
- {
- 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
- 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
- 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
- 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
- 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
- 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
- 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
- 0xf5, 0xdc, 0x4a, 0x06, 0x51, 0x80
- };
-
- const WebRtc_UWord8 mask41_24[144] =
- {
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
- 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
- 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
- 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
- 0xf5, 0xdc, 0x4a, 0x06, 0x51, 0x80,
- 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
- 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
- 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
- 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0x45, 0xa6, 0xef, 0xc9, 0xc3, 0x00
- };
-
- const WebRtc_UWord8 mask41_25[150] =
- {
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
- 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
- 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
- 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
- 0xf5, 0xdc, 0x4a, 0x06, 0x51, 0x80,
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80
- };
-
- const WebRtc_UWord8 mask41_26[156] =
- {
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0xc6, 0x31, 0x8c, 0x62, 0x1a, 0x00,
- 0x23, 0x88, 0xe2, 0x38, 0x8c, 0x80,
- 0x1a, 0x46, 0x91, 0xa4, 0x58, 0x80,
- 0x24, 0xc9, 0x32, 0x4d, 0x30, 0x80,
- 0x71, 0x1c, 0x47, 0x11, 0x07, 0x00,
- 0xf5, 0xdc, 0x4a, 0x06, 0x51, 0x80,
- 0x6f, 0x72, 0xf1, 0xe7, 0x1a, 0x80
- };
-
- const WebRtc_UWord8 mask41_27[162] =
- {
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0x56, 0x3e, 0x24, 0xdd, 0x0c, 0x00
- };
-
- const WebRtc_UWord8 mask41_28[168] =
- {
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0x56, 0x3e, 0x24, 0xdd, 0x0c, 0x00,
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x61, 0x2c, 0xfa, 0x25, 0x38, 0x00
- };
-
- const WebRtc_UWord8 mask41_29[174] =
- {
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0x56, 0x3e, 0x24, 0xdd, 0x0c, 0x00,
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80
- };
-
- const WebRtc_UWord8 mask41_3[18] =
- {
- 0xce, 0x33, 0x8c, 0xe3, 0x2b, 0x00,
- 0x55, 0x95, 0x65, 0x5d, 0xc5, 0x00,
- 0xb1, 0x6a, 0x3a, 0x8e, 0xd8, 0x80
- };
-
- const WebRtc_UWord8 mask41_30[180] =
- {
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x0e, 0x03, 0x80, 0xe1, 0x91, 0x00,
- 0x33, 0x0c, 0xc3, 0x31, 0x45, 0x00,
- 0x10, 0xc4, 0x31, 0x0c, 0x32, 0x80,
- 0x45, 0x51, 0x54, 0x56, 0x84, 0x80,
- 0x88, 0xa2, 0x28, 0x88, 0x4a, 0x80,
- 0xe0, 0x38, 0x0e, 0x02, 0x29, 0x00,
- 0x56, 0x3e, 0x24, 0xdd, 0x0c, 0x00,
- 0x59, 0x53, 0x31, 0x62, 0x15, 0x00
- };
-
- const WebRtc_UWord8 mask41_31[186] =
- {
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x28, 0x1c, 0x63, 0xbf, 0x53, 0x80
- };
-
- const WebRtc_UWord8 mask41_32[192] =
- {
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x28, 0x1c, 0x63, 0xbf, 0x53, 0x80,
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0xca, 0xbb, 0xcb, 0x6d, 0xaa, 0x00
- };
-
- const WebRtc_UWord8 mask41_33[198] =
- {
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x28, 0x1c, 0x63, 0xbf, 0x53, 0x80,
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00
- };
-
- const WebRtc_UWord8 mask41_34[204] =
- {
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0x46, 0x11, 0x84, 0x61, 0x19, 0x00,
- 0x33, 0x0c, 0xc3, 0x30, 0xcc, 0x80,
- 0x10, 0xc4, 0x31, 0x0e, 0x46, 0x00,
- 0x0c, 0x43, 0x10, 0xc6, 0x90, 0x80,
- 0x28, 0x8a, 0x22, 0x89, 0x42, 0x80,
- 0x94, 0x25, 0x09, 0x42, 0x13, 0x00,
- 0xc1, 0x30, 0x4c, 0x10, 0x25, 0x80,
- 0x28, 0x1c, 0x63, 0xbf, 0x53, 0x80,
- 0xbd, 0x37, 0x3f, 0x75, 0x36, 0x80
- };
-
- const WebRtc_UWord8 mask41_35[210] =
- {
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0x21, 0x7b, 0xf5, 0xa5, 0x65, 0x80
- };
-
- const WebRtc_UWord8 mask41_36[216] =
- {
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0x21, 0x7b, 0xf5, 0xa5, 0x65, 0x80,
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
- 0xc1, 0xb1, 0x80, 0xbe, 0x3e, 0x00
- };
-
- const WebRtc_UWord8 mask41_37[222] =
- {
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0x21, 0x7b, 0xf5, 0xa5, 0x65, 0x80,
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
- 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
- 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
- 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80
- };
-
- const WebRtc_UWord8 mask41_38[228] =
- {
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
- 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
- 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
- 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
- 0x2c, 0x0b, 0x02, 0xc0, 0x32, 0x00,
- 0x81, 0xa0, 0x68, 0x1a, 0x01, 0x80,
- 0xa0, 0x68, 0x1a, 0x06, 0x82, 0x00,
- 0x05, 0x41, 0x50, 0x54, 0x15, 0x00,
- 0x18, 0x86, 0x21, 0x89, 0x0c, 0x00,
- 0xc2, 0x30, 0x8c, 0x20, 0x68, 0x00,
- 0x22, 0x88, 0xa2, 0x29, 0x80, 0x80,
- 0x50, 0x54, 0x15, 0x04, 0x50, 0x80,
- 0x21, 0x7b, 0xf5, 0xa5, 0x65, 0x80,
- 0xea, 0xc8, 0xbb, 0xd4, 0x5d, 0x00
- };
-
- const WebRtc_UWord8 mask41_39[234] =
- {
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
- 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
- 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
- 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
- 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
- 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
- 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
- 0xf7, 0x8d, 0xa2, 0xa0, 0x33, 0x00
- };
-
- const WebRtc_UWord8 mask41_4[24] =
- {
- 0xe6, 0x39, 0x8e, 0x63, 0x13, 0x00,
- 0x33, 0x8c, 0xe3, 0x38, 0xc5, 0x80,
- 0x98, 0xe6, 0x39, 0x8d, 0x2c, 0x80,
- 0x2d, 0x4b, 0x52, 0xd4, 0xb2, 0x80
- };
-
- const WebRtc_UWord8 mask41_40[240] =
- {
- 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
- 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
- 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
- 0xf7, 0x8d, 0xa2, 0xa0, 0x33, 0x00,
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
- 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
- 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
- 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
- 0xe8, 0x07, 0x18, 0x9a, 0x02, 0x00
- };
-
- const WebRtc_UWord8 mask41_41[246] =
- {
- 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
- 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
- 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
- 0x4e, 0x13, 0x84, 0xe1, 0x19, 0x00,
- 0xe3, 0x38, 0xce, 0x31, 0x89, 0x80,
- 0x81, 0xe0, 0x78, 0x1e, 0x30, 0x00,
- 0x21, 0x48, 0x52, 0x14, 0x05, 0x80,
- 0x52, 0x94, 0xa5, 0x28, 0x1e, 0x00,
- 0xb4, 0x2d, 0x0b, 0x42, 0x82, 0x00,
- 0x26, 0x89, 0xa2, 0x68, 0x62, 0x80,
- 0x58, 0x56, 0x15, 0x86, 0x44, 0x00,
- 0x19, 0x86, 0x61, 0x99, 0xe0, 0x00,
- 0xf7, 0x8d, 0xa2, 0xa0, 0x33, 0x00,
- 0x4c, 0x13, 0x04, 0xc1, 0x91, 0x00,
- 0x51, 0x14, 0x45, 0x11, 0x45, 0x00,
- 0xa0, 0x68, 0x1a, 0x06, 0xa4, 0x00,
- 0x04, 0xc1, 0x30, 0x4c, 0x13, 0x00,
- 0x03, 0x80, 0xe0, 0x3b, 0x40, 0x00,
- 0x86, 0x21, 0x88, 0x62, 0x09, 0x00,
- 0x29, 0x0a, 0x42, 0x90, 0x84, 0x80,
- 0x42, 0x50, 0x94, 0x24, 0x30, 0x80,
- 0x98, 0x26, 0x09, 0x81, 0x28, 0x00,
- 0x30, 0x8c, 0x23, 0x08, 0x4a, 0x80,
- 0x4e, 0x13, 0x84, 0xc1, 0x19, 0x00,
- 0xe3, 0x38, 0xc5, 0x10, 0xcc, 0x80,
- 0x81, 0xe0, 0x7a, 0x06, 0x64, 0x00,
- 0x21, 0x48, 0x50, 0x4c, 0x16, 0x00,
- 0x52, 0x94, 0xa0, 0x3a, 0x02, 0x80,
- 0xb4, 0x2d, 0x08, 0x62, 0x11, 0x00,
- 0x26, 0x89, 0xa2, 0x91, 0x01, 0x80,
- 0x58, 0x56, 0x14, 0x24, 0x2a, 0x00,
- 0x19, 0x86, 0x69, 0x81, 0xa0, 0x00,
- 0xf7, 0x8d, 0xa3, 0x08, 0x40, 0x80,
- 0x2b, 0xea, 0x4d, 0xf4, 0xc1, 0x00
- };
-
- const WebRtc_UWord8 mask41_5[30] =
- {
- 0xce, 0x33, 0x8c, 0xe3, 0x1b, 0x00,
- 0x63, 0x98, 0xe6, 0x39, 0x8d, 0x80,
- 0x98, 0xe5, 0x39, 0x8c, 0x76, 0x80,
- 0x2b, 0x53, 0x54, 0xd6, 0xb5, 0x00,
- 0xb4, 0x5c, 0xab, 0x26, 0xca, 0x80
- };
-
- const WebRtc_UWord8 mask41_6[36] =
- {
- 0x4c, 0x1b, 0x04, 0xc1, 0x91, 0x00,
- 0x51, 0x34, 0x45, 0x11, 0x45, 0x00,
- 0x20, 0xe8, 0x32, 0x0e, 0xa0, 0x80,
- 0x85, 0x41, 0x58, 0x54, 0x12, 0x80,
- 0x06, 0x86, 0xa0, 0x68, 0x0d, 0x80,
- 0x9a, 0x21, 0x88, 0xa2, 0x43, 0x00
- };
-
- const WebRtc_UWord8 mask41_7[42] =
- {
- 0x4e, 0x11, 0x8c, 0x61, 0x19, 0x00,
- 0x33, 0x2c, 0x03, 0x30, 0x4c, 0x80,
- 0x10, 0x0e, 0xb1, 0x86, 0x74, 0x00,
- 0x81, 0x51, 0x54, 0x54, 0x2d, 0x00,
- 0x24, 0xc4, 0xa1, 0x2d, 0x42, 0x80,
- 0xd4, 0x23, 0x0b, 0x42, 0x83, 0x00,
- 0x0c, 0xa2, 0x62, 0x99, 0x21, 0x80
- };
-
- const WebRtc_UWord8 mask41_8[48] =
- {
- 0x27, 0x09, 0xc0, 0x70, 0xa7, 0x00,
- 0x89, 0xa2, 0x64, 0x9a, 0x82, 0x80,
- 0xd0, 0x74, 0x1b, 0x07, 0xa0, 0x00,
- 0x24, 0xc9, 0x32, 0x4c, 0x5c, 0x00,
- 0xe2, 0x90, 0xa5, 0x28, 0x0e, 0x80,
- 0xc6, 0x31, 0x8c, 0x63, 0x18, 0x00,
- 0x31, 0x8c, 0x63, 0x19, 0x41, 0x80,
- 0x18, 0xc6, 0x31, 0x8c, 0x70, 0x80
- };
-
- const WebRtc_UWord8 mask41_9[54] =
- {
- 0x4e, 0x13, 0x84, 0xe1, 0x11, 0x00,
- 0x62, 0x38, 0xc6, 0x21, 0xa0, 0x80,
- 0x81, 0xe0, 0x78, 0x0e, 0x94, 0x00,
- 0xe1, 0x48, 0x5a, 0x15, 0x05, 0x00,
- 0x13, 0x94, 0xa5, 0x30, 0x06, 0x80,
- 0xb4, 0x2d, 0x0a, 0x42, 0x43, 0x00,
- 0x26, 0x89, 0xa1, 0x6a, 0x08, 0x80,
- 0x58, 0x56, 0x15, 0x84, 0x52, 0x00,
- 0x49, 0x86, 0x52, 0x98, 0x68, 0x00
- };
-
- const WebRtc_UWord8 mask42_1[6] =
- {
- 0xff, 0xff, 0xff, 0xff, 0xff, 0xc0
- };
-
- const WebRtc_UWord8 mask42_10[60] =
- {
- 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
- 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
- 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
- 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
- 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
- 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
- 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
- 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
- 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40
- };
-
- const WebRtc_UWord8 mask42_11[66] =
- {
- 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
- 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
- 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
- 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
- 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80
- };
-
- const WebRtc_UWord8 mask42_12[72] =
- {
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
- 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
- 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
- 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
- 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
- 0xa0, 0x65, 0x1d, 0x03, 0x28, 0xc0
- };
-
- const WebRtc_UWord8 mask42_13[78] =
- {
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0
- };
-
- const WebRtc_UWord8 mask42_14[84] =
- {
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0x4d, 0xd0, 0xc2, 0x6e, 0x86, 0x00
- };
-
- const WebRtc_UWord8 mask42_15[90] =
- {
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40
- };
-
- const WebRtc_UWord8 mask42_16[96] =
- {
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0x3b, 0xf5, 0x39, 0xdf, 0xa9, 0xc0
- };
-
- const WebRtc_UWord8 mask42_17[102] =
- {
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00
- };
-
- const WebRtc_UWord8 mask42_18[108] =
- {
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0x5a, 0x56, 0x5a, 0xd2, 0xb2, 0xc0
- };
-
- const WebRtc_UWord8 mask42_19[114] =
- {
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
- 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
- 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
- 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
- 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
- 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
- 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
- 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
- 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40
- };
-
- const WebRtc_UWord8 mask42_2[12] =
- {
- 0xee, 0x3b, 0x37, 0x71, 0xd9, 0x80,
- 0x99, 0xe6, 0xec, 0xcf, 0x37, 0x40
- };
-
- const WebRtc_UWord8 mask42_20[120] =
- {
- 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
- 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
- 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
- 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
- 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
- 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
- 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
- 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
- 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x2a, 0x03, 0x31, 0x50, 0x19, 0x80
- };
-
- const WebRtc_UWord8 mask42_21[126] =
- {
- 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
- 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
- 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
- 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
- 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
- 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
- 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
- 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
- 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
- 0x4c, 0x11, 0x92, 0x60, 0x8c, 0x80,
- 0x51, 0x0c, 0xca, 0x88, 0x66, 0x40,
- 0xa0, 0x66, 0x45, 0x03, 0x32, 0x00,
- 0x04, 0xc1, 0x60, 0x26, 0x0b, 0x00,
- 0x03, 0xa0, 0x28, 0x1d, 0x01, 0x40,
- 0x86, 0x21, 0x14, 0x31, 0x08, 0x80,
- 0x29, 0x10, 0x19, 0x48, 0x80, 0xc0,
- 0x42, 0x42, 0xa2, 0x12, 0x15, 0x00,
- 0x98, 0x1a, 0x04, 0xc0, 0xd0, 0x00,
- 0x30, 0x84, 0x09, 0x84, 0x20, 0x40,
- 0xdf, 0x4c, 0x16, 0xfa, 0x60, 0x80
- };
-
- const WebRtc_UWord8 mask42_22[132] =
- {
- 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
- 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
- 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
- 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
- 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
- 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
- 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
- 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
- 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
- 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
- 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
- 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
- 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
- 0xdb, 0x36, 0xb0, 0x33, 0x14, 0x80
- };
-
- const WebRtc_UWord8 mask42_23[138] =
- {
- 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
- 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
- 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
- 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
- 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
- 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
- 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
- 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
- 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
- 0xa0, 0x65, 0x1d, 0x03, 0x28, 0xc0
- };
-
- const WebRtc_UWord8 mask42_24[144] =
- {
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
- 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
- 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
- 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
- 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
- 0xa0, 0x65, 0x1d, 0x03, 0x28, 0xc0,
- 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
- 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
- 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
- 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
- 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0x2e, 0x1c, 0x92, 0xbb, 0x07, 0xc0
- };
-
- const WebRtc_UWord8 mask42_25[150] =
- {
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
- 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
- 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
- 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
- 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
- 0xa0, 0x65, 0x1d, 0x03, 0x28, 0xc0,
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0
- };
-
- const WebRtc_UWord8 mask42_26[156] =
- {
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0xc6, 0x21, 0xa6, 0x31, 0x0d, 0x00,
- 0x23, 0x88, 0xc9, 0x1c, 0x46, 0x40,
- 0x1a, 0x45, 0x88, 0xd2, 0x2c, 0x40,
- 0x24, 0xd3, 0x09, 0x26, 0x98, 0x40,
- 0x71, 0x10, 0x73, 0x88, 0x83, 0x80,
- 0xa0, 0x65, 0x1d, 0x03, 0x28, 0xc0,
- 0xb8, 0x41, 0xed, 0xa3, 0x77, 0xc0
- };
-
- const WebRtc_UWord8 mask42_27[162] =
- {
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0x4d, 0xd0, 0xc2, 0x6e, 0x86, 0x00
- };
-
- const WebRtc_UWord8 mask42_28[168] =
- {
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0x4d, 0xd0, 0xc2, 0x6e, 0x86, 0x00,
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0xc3, 0x3c, 0x56, 0xc2, 0x30, 0x40
- };
-
- const WebRtc_UWord8 mask42_29[174] =
- {
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0x4d, 0xd0, 0xc2, 0x6e, 0x86, 0x00,
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40
- };
-
- const WebRtc_UWord8 mask42_3[18] =
- {
- 0xce, 0x32, 0xb6, 0x71, 0x95, 0x80,
- 0x55, 0xdc, 0x52, 0xae, 0xe2, 0x80,
- 0xa8, 0xed, 0x8d, 0x47, 0x6c, 0x40
- };
-
- const WebRtc_UWord8 mask42_30[180] =
- {
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0x0e, 0x19, 0x10, 0x70, 0xc8, 0x80,
- 0x33, 0x14, 0x51, 0x98, 0xa2, 0x80,
- 0x10, 0xc3, 0x28, 0x86, 0x19, 0x40,
- 0x45, 0x68, 0x4a, 0x2b, 0x42, 0x40,
- 0x88, 0x84, 0xac, 0x44, 0x25, 0x40,
- 0xe0, 0x22, 0x97, 0x01, 0x14, 0x80,
- 0x4d, 0xd0, 0xc2, 0x6e, 0x86, 0x00,
- 0xf5, 0xdd, 0x0d, 0x58, 0xeb, 0x00
- };
-
- const WebRtc_UWord8 mask42_31[186] =
- {
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0x3b, 0xf5, 0x39, 0xdf, 0xa9, 0xc0
- };
-
- const WebRtc_UWord8 mask42_32[192] =
- {
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0x3b, 0xf5, 0x39, 0xdf, 0xa9, 0xc0,
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0xf9, 0x1f, 0xb6, 0xe1, 0x09, 0xc0
- };
-
- const WebRtc_UWord8 mask42_33[198] =
- {
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0x3b, 0xf5, 0x39, 0xdf, 0xa9, 0xc0,
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00
- };
-
- const WebRtc_UWord8 mask42_34[204] =
- {
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0x46, 0x11, 0x92, 0x30, 0x8c, 0x80,
- 0x33, 0x0c, 0xc9, 0x98, 0x66, 0x40,
- 0x10, 0xe4, 0x60, 0x87, 0x23, 0x00,
- 0x0c, 0x69, 0x08, 0x63, 0x48, 0x40,
- 0x28, 0x94, 0x29, 0x44, 0xa1, 0x40,
- 0x94, 0x21, 0x34, 0xa1, 0x09, 0x80,
- 0xc1, 0x02, 0x5e, 0x08, 0x12, 0xc0,
- 0x3b, 0xf5, 0x39, 0xdf, 0xa9, 0xc0,
- 0xf8, 0xbf, 0xf6, 0x76, 0x1b, 0x80
- };
-
- const WebRtc_UWord8 mask42_35[210] =
- {
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0x5a, 0x56, 0x5a, 0xd2, 0xb2, 0xc0
- };
-
- const WebRtc_UWord8 mask42_36[216] =
- {
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0x5a, 0x56, 0x5a, 0xd2, 0xb2, 0xc0,
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x57, 0xc7, 0x03, 0xf9, 0xc6, 0x00
- };
-
- const WebRtc_UWord8 mask42_37[222] =
- {
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0x5a, 0x56, 0x5a, 0xd2, 0xb2, 0xc0,
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
- 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
- 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
- 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
- 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
- 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
- 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
- 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
- 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40
- };
-
- const WebRtc_UWord8 mask42_38[228] =
- {
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
- 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
- 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
- 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
- 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
- 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
- 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
- 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
- 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x2c, 0x03, 0x21, 0x60, 0x19, 0x00,
- 0x81, 0xa0, 0x1c, 0x0d, 0x00, 0xc0,
- 0xa0, 0x68, 0x25, 0x03, 0x41, 0x00,
- 0x05, 0x41, 0x50, 0x2a, 0x0a, 0x80,
- 0x18, 0x90, 0xc0, 0xc4, 0x86, 0x00,
- 0xc2, 0x06, 0x86, 0x10, 0x34, 0x00,
- 0x22, 0x98, 0x09, 0x14, 0xc0, 0x40,
- 0x50, 0x45, 0x0a, 0x82, 0x28, 0x40,
- 0x5a, 0x56, 0x5a, 0xd2, 0xb2, 0xc0,
- 0x05, 0x19, 0x55, 0xee, 0xe2, 0xc0
- };
-
- const WebRtc_UWord8 mask42_39[234] =
- {
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
- 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
- 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
- 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
- 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
- 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
- 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
- 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
- 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
- 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
- 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
- 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
- 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
- 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
- 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
- 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
- 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
- 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x2a, 0x03, 0x31, 0x50, 0x19, 0x80
- };
-
- const WebRtc_UWord8 mask42_4[24] =
- {
- 0xe6, 0x31, 0x37, 0x31, 0x89, 0x80,
- 0x33, 0x8c, 0x59, 0x9c, 0x62, 0xc0,
- 0x98, 0xd2, 0xcc, 0xc6, 0x96, 0x40,
- 0x2d, 0x4b, 0x29, 0x6a, 0x59, 0x40
- };
-
- const WebRtc_UWord8 mask42_40[240] =
- {
- 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
- 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
- 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
- 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
- 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
- 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
- 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
- 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
- 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x2a, 0x03, 0x31, 0x50, 0x19, 0x80,
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
- 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
- 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
- 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
- 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
- 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
- 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
- 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
- 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
- 0xf9, 0xdb, 0x5d, 0x7a, 0xd4, 0x40
- };
-
- const WebRtc_UWord8 mask42_41[246] =
- {
- 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
- 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
- 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
- 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
- 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
- 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
- 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
- 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
- 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x2a, 0x03, 0x31, 0x50, 0x19, 0x80,
- 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
- 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
- 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
- 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
- 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
- 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
- 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
- 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
- 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
- 0x4c, 0x11, 0x92, 0x60, 0x8c, 0x80,
- 0x51, 0x0c, 0xca, 0x88, 0x66, 0x40,
- 0xa0, 0x66, 0x45, 0x03, 0x32, 0x00,
- 0x04, 0xc1, 0x60, 0x26, 0x0b, 0x00,
- 0x03, 0xa0, 0x28, 0x1d, 0x01, 0x40,
- 0x86, 0x21, 0x14, 0x31, 0x08, 0x80,
- 0x29, 0x10, 0x19, 0x48, 0x80, 0xc0,
- 0x42, 0x42, 0xa2, 0x12, 0x15, 0x00,
- 0x98, 0x1a, 0x04, 0xc0, 0xd0, 0x00,
- 0x30, 0x84, 0x09, 0x84, 0x20, 0x40,
- 0xdf, 0x4c, 0x16, 0xfa, 0x60, 0x80
- };
-
- const WebRtc_UWord8 mask42_42[252] =
- {
- 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
- 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
- 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
- 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
- 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
- 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
- 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
- 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
- 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
- 0x4c, 0x11, 0x92, 0x60, 0x8c, 0x80,
- 0x51, 0x0c, 0xca, 0x88, 0x66, 0x40,
- 0xa0, 0x66, 0x45, 0x03, 0x32, 0x00,
- 0x04, 0xc1, 0x60, 0x26, 0x0b, 0x00,
- 0x03, 0xa0, 0x28, 0x1d, 0x01, 0x40,
- 0x86, 0x21, 0x14, 0x31, 0x08, 0x80,
- 0x29, 0x10, 0x19, 0x48, 0x80, 0xc0,
- 0x42, 0x42, 0xa2, 0x12, 0x15, 0x00,
- 0x98, 0x1a, 0x04, 0xc0, 0xd0, 0x00,
- 0x30, 0x84, 0x09, 0x84, 0x20, 0x40,
- 0xdf, 0x4c, 0x16, 0xfa, 0x60, 0x80,
- 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
- 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
- 0xa0, 0x6a, 0x45, 0x03, 0x52, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x09, 0x80,
- 0x03, 0xb4, 0x00, 0x1d, 0xa0, 0x00,
- 0x86, 0x20, 0x94, 0x31, 0x04, 0x80,
- 0x29, 0x08, 0x49, 0x48, 0x42, 0x40,
- 0x42, 0x43, 0x0a, 0x12, 0x18, 0x40,
- 0x98, 0x12, 0x84, 0xc0, 0x94, 0x00,
- 0x30, 0x84, 0xa9, 0x84, 0x25, 0x40,
- 0x4e, 0x11, 0x92, 0x70, 0x8c, 0x80,
- 0xe3, 0x18, 0x9f, 0x18, 0xc4, 0xc0,
- 0x81, 0xe3, 0x04, 0x0f, 0x18, 0x00,
- 0x21, 0x40, 0x59, 0x0a, 0x02, 0xc0,
- 0x52, 0x81, 0xe2, 0x94, 0x0f, 0x00,
- 0xb4, 0x28, 0x25, 0xa1, 0x41, 0x00,
- 0x26, 0x86, 0x29, 0x34, 0x31, 0x40,
- 0x58, 0x64, 0x42, 0xc3, 0x22, 0x00,
- 0x19, 0x9e, 0x00, 0xcc, 0xf0, 0x00,
- 0x2a, 0x03, 0x31, 0x50, 0x19, 0x80,
- 0xea, 0x9e, 0x23, 0xb3, 0x65, 0x00
- };
-
- const WebRtc_UWord8 mask42_5[30] =
- {
- 0xce, 0x31, 0xb6, 0x71, 0x8d, 0x80,
- 0x63, 0x98, 0xdb, 0x1c, 0xc6, 0xc0,
- 0x98, 0xc7, 0x6c, 0xc6, 0x3b, 0x40,
- 0x4d, 0x6b, 0x52, 0x6b, 0x5a, 0x80,
- 0xb2, 0x6c, 0xad, 0x93, 0x65, 0x40
- };
-
- const WebRtc_UWord8 mask42_6[36] =
- {
- 0x4c, 0x19, 0x12, 0x60, 0xc8, 0x80,
- 0x51, 0x14, 0x52, 0x88, 0xa2, 0x80,
- 0x20, 0xea, 0x09, 0x07, 0x50, 0x40,
- 0x85, 0x41, 0x2c, 0x2a, 0x09, 0x40,
- 0x06, 0x80, 0xd8, 0x34, 0x06, 0xc0,
- 0x8a, 0x24, 0x34, 0x51, 0x21, 0x80
- };
-
- const WebRtc_UWord8 mask42_7[42] =
- {
- 0xc6, 0x11, 0x96, 0x30, 0x8c, 0x80,
- 0x33, 0x04, 0xc9, 0x98, 0x26, 0x40,
- 0x18, 0x67, 0x40, 0xc3, 0x3a, 0x00,
- 0x45, 0x42, 0xd2, 0x2a, 0x16, 0x80,
- 0x12, 0xd4, 0x28, 0x96, 0xa1, 0x40,
- 0xb4, 0x28, 0x35, 0xa1, 0x41, 0x80,
- 0x29, 0x92, 0x19, 0x4c, 0x90, 0xc0
- };
-
- const WebRtc_UWord8 mask42_8[48] =
- {
- 0x07, 0x0a, 0x70, 0x38, 0x53, 0x80,
- 0x49, 0xa8, 0x2a, 0x4d, 0x41, 0x40,
- 0xb0, 0x7a, 0x05, 0x83, 0xd0, 0x00,
- 0x24, 0xc5, 0xc1, 0x26, 0x2e, 0x00,
- 0x52, 0x80, 0xea, 0x94, 0x07, 0x40,
- 0xc6, 0x31, 0x86, 0x31, 0x8c, 0x00,
- 0x31, 0x94, 0x19, 0x8c, 0xa0, 0xc0,
- 0x18, 0xc7, 0x08, 0xc6, 0x38, 0x40
- };
-
- const WebRtc_UWord8 mask42_9[54] =
- {
- 0x4e, 0x11, 0x12, 0x70, 0x88, 0x80,
- 0x62, 0x1a, 0x0b, 0x10, 0xd0, 0x40,
- 0x80, 0xe9, 0x44, 0x07, 0x4a, 0x00,
- 0xa1, 0x50, 0x55, 0x0a, 0x82, 0x80,
- 0x53, 0x00, 0x6a, 0x98, 0x03, 0x40,
- 0xa4, 0x24, 0x35, 0x21, 0x21, 0x80,
- 0x16, 0xa0, 0x88, 0xb5, 0x04, 0x40,
- 0x58, 0x45, 0x22, 0xc2, 0x29, 0x00,
- 0x29, 0x86, 0x81, 0x4c, 0x34, 0x00
- };
-
- const WebRtc_UWord8 mask43_1[6] =
- {
- 0xff, 0xff, 0xff, 0xff, 0xff, 0xe0
- };
-
- const WebRtc_UWord8 mask43_10[60] =
- {
- 0x4c, 0x19, 0x16, 0x01, 0xc4, 0x40,
- 0x51, 0x14, 0x51, 0x80, 0x71, 0x40,
- 0xa0, 0x6a, 0x47, 0x40, 0x38, 0x00,
- 0x04, 0xc1, 0x34, 0x28, 0x45, 0x40,
- 0x03, 0xb4, 0x06, 0x84, 0x90, 0x80,
- 0x86, 0x20, 0x94, 0x32, 0x82, 0x40,
- 0x29, 0x08, 0x4a, 0x53, 0x40, 0x60,
- 0x42, 0x43, 0x08, 0x0d, 0x03, 0xa0,
- 0x98, 0x12, 0x82, 0x64, 0x0c, 0x80,
- 0x30, 0x84, 0xab, 0x11, 0x20, 0x20
- };
-
- const WebRtc_UWord8 mask43_11[66] =
- {
- 0xc6, 0x21, 0xa2, 0x32, 0x46, 0x40,
- 0x23, 0x88, 0xc9, 0x99, 0x33, 0x20,
- 0x1a, 0x45, 0x8c, 0xc8, 0x99, 0x00,
- 0x24, 0xd3, 0x08, 0x2c, 0x05, 0x80,
- 0x71, 0x10, 0x74, 0x05, 0x80, 0xa0,
- 0x0e, 0x19, 0x14, 0x22, 0x84, 0x40,
- 0x33, 0x14, 0x52, 0x03, 0x40, 0x60,
- 0x10, 0xc3, 0x28, 0x54, 0x0a, 0x80,
- 0x45, 0x68, 0x4b, 0x40, 0x68, 0x00,
- 0x88, 0x84, 0xa8, 0x81, 0x10, 0x20,
- 0xe0, 0x22, 0x91, 0x82, 0x30, 0x40
- };
-
- const WebRtc_UWord8 mask43_12[72] =
- {
- 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
- 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
- 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
- 0xc6, 0x21, 0xa4, 0x34, 0x86, 0x80,
- 0x23, 0x88, 0xc9, 0x19, 0x23, 0x20,
- 0x1a, 0x45, 0x88, 0xb1, 0x16, 0x20,
- 0x24, 0xd3, 0x0a, 0x61, 0x4c, 0x20,
- 0x71, 0x10, 0x72, 0x0e, 0x41, 0xc0,
- 0xa0, 0x65, 0x1f, 0xa0, 0xc4, 0xe0
- };
-
- const WebRtc_UWord8 mask43_13[78] =
- {
- 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
- 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
- 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60
- };
-
- const WebRtc_UWord8 mask43_14[84] =
- {
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
- 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
- 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
- 0x4d, 0xd0, 0xc6, 0x36, 0x57, 0x40
- };
-
- const WebRtc_UWord8 mask43_15[90] =
- {
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20
- };
-
- const WebRtc_UWord8 mask43_16[96] =
- {
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x3b, 0xf5, 0x3c, 0x36, 0x0a, 0x20
- };
-
- const WebRtc_UWord8 mask43_17[102] =
- {
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00
- };
-
- const WebRtc_UWord8 mask43_18[108] =
- {
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0x5a, 0x56, 0x5f, 0x26, 0xa3, 0x60
- };
-
- const WebRtc_UWord8 mask43_19[114] =
- {
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
- 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
- 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
- 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
- 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
- 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
- 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0
- };
-
- const WebRtc_UWord8 mask43_2[12] =
- {
- 0xee, 0x3b, 0x37, 0x66, 0xec, 0xc0,
- 0x99, 0xe6, 0xec, 0xdd, 0x9b, 0xa0
- };
-
- const WebRtc_UWord8 mask43_20[120] =
- {
- 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
- 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
- 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
- 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
- 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
- 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
- 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0x2a, 0x03, 0x31, 0xda, 0x46, 0x20
- };
-
- const WebRtc_UWord8 mask43_21[126] =
- {
- 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
- 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
- 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
- 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
- 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
- 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
- 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0x4c, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x51, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0xa0, 0x66, 0x44, 0xc8, 0x99, 0x00,
- 0x04, 0xc1, 0x60, 0x2c, 0x05, 0x80,
- 0x03, 0xa0, 0x2c, 0x05, 0x80, 0xa0,
- 0x86, 0x21, 0x14, 0x22, 0x84, 0x40,
- 0x29, 0x10, 0x1a, 0x03, 0x40, 0x60,
- 0x42, 0x42, 0xa0, 0x54, 0x0a, 0x80,
- 0x98, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x30, 0x84, 0x08, 0x81, 0x10, 0x20,
- 0xdf, 0x4c, 0x11, 0x82, 0x30, 0x40
- };
-
- const WebRtc_UWord8 mask43_22[132] =
- {
- 0xc6, 0x21, 0xa2, 0x32, 0x46, 0x40,
- 0x23, 0x88, 0xc9, 0x99, 0x33, 0x20,
- 0x1a, 0x45, 0x8c, 0xc8, 0x99, 0x00,
- 0x24, 0xd3, 0x08, 0x2c, 0x05, 0x80,
- 0x71, 0x10, 0x74, 0x05, 0x80, 0xa0,
- 0x0e, 0x19, 0x14, 0x22, 0x84, 0x40,
- 0x33, 0x14, 0x52, 0x03, 0x40, 0x60,
- 0x10, 0xc3, 0x28, 0x54, 0x0a, 0x80,
- 0x45, 0x68, 0x4b, 0x40, 0x68, 0x00,
- 0x88, 0x84, 0xa8, 0x81, 0x10, 0x20,
- 0xe0, 0x22, 0x91, 0x82, 0x30, 0x40,
- 0x4c, 0x19, 0x16, 0x01, 0xc4, 0x40,
- 0x51, 0x14, 0x51, 0x80, 0x71, 0x40,
- 0xa0, 0x6a, 0x47, 0x40, 0x38, 0x00,
- 0x04, 0xc1, 0x34, 0x28, 0x45, 0x40,
- 0x03, 0xb4, 0x06, 0x84, 0x90, 0x80,
- 0x86, 0x20, 0x94, 0x32, 0x82, 0x40,
- 0x29, 0x08, 0x4a, 0x53, 0x40, 0x60,
- 0x42, 0x43, 0x08, 0x0d, 0x03, 0xa0,
- 0x98, 0x12, 0x82, 0x64, 0x0c, 0x80,
- 0x30, 0x84, 0xab, 0x11, 0x20, 0x20,
- 0xfe, 0x2c, 0x85, 0xcc, 0x24, 0x80
- };
-
- const WebRtc_UWord8 mask43_23[138] =
- {
- 0xc6, 0x21, 0xa2, 0x32, 0x46, 0x40,
- 0x23, 0x88, 0xc9, 0x99, 0x33, 0x20,
- 0x1a, 0x45, 0x8c, 0xc8, 0x99, 0x00,
- 0x24, 0xd3, 0x08, 0x2c, 0x05, 0x80,
- 0x71, 0x10, 0x74, 0x05, 0x80, 0xa0,
- 0x0e, 0x19, 0x14, 0x22, 0x84, 0x40,
- 0x33, 0x14, 0x52, 0x03, 0x40, 0x60,
- 0x10, 0xc3, 0x28, 0x54, 0x0a, 0x80,
- 0x45, 0x68, 0x4b, 0x40, 0x68, 0x00,
- 0x88, 0x84, 0xa8, 0x81, 0x10, 0x20,
- 0xe0, 0x22, 0x91, 0x82, 0x30, 0x40,
- 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
- 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
- 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
- 0xc6, 0x21, 0xa4, 0x34, 0x86, 0x80,
- 0x23, 0x88, 0xc9, 0x19, 0x23, 0x20,
- 0x1a, 0x45, 0x88, 0xb1, 0x16, 0x20,
- 0x24, 0xd3, 0x0a, 0x61, 0x4c, 0x20,
- 0x71, 0x10, 0x72, 0x0e, 0x41, 0xc0,
- 0xa0, 0x65, 0x1f, 0xa0, 0xc4, 0xe0
- };
-
- const WebRtc_UWord8 mask43_24[144] =
- {
- 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
- 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
- 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
- 0xc6, 0x21, 0xa4, 0x34, 0x86, 0x80,
- 0x23, 0x88, 0xc9, 0x19, 0x23, 0x20,
- 0x1a, 0x45, 0x88, 0xb1, 0x16, 0x20,
- 0x24, 0xd3, 0x0a, 0x61, 0x4c, 0x20,
- 0x71, 0x10, 0x72, 0x0e, 0x41, 0xc0,
- 0xa0, 0x65, 0x1f, 0xa0, 0xc4, 0xe0,
- 0xc6, 0x21, 0xa2, 0x32, 0x46, 0x40,
- 0x23, 0x88, 0xc9, 0x99, 0x33, 0x20,
- 0x1a, 0x45, 0x8c, 0xc8, 0x99, 0x00,
- 0x24, 0xd3, 0x08, 0x2c, 0x05, 0x80,
- 0x71, 0x10, 0x74, 0x05, 0x80, 0xa0,
- 0x0e, 0x19, 0x14, 0x22, 0x84, 0x40,
- 0x33, 0x14, 0x52, 0x03, 0x40, 0x60,
- 0x10, 0xc3, 0x28, 0x54, 0x0a, 0x80,
- 0x45, 0x68, 0x4b, 0x40, 0x68, 0x00,
- 0x88, 0x84, 0xa8, 0x81, 0x10, 0x20,
- 0xe0, 0x22, 0x91, 0x82, 0x30, 0x40,
- 0xf9, 0xb1, 0x26, 0x6c, 0x51, 0xe0
- };
-
- const WebRtc_UWord8 mask43_25[150] =
- {
- 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
- 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
- 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
- 0xc6, 0x21, 0xa4, 0x34, 0x86, 0x80,
- 0x23, 0x88, 0xc9, 0x19, 0x23, 0x20,
- 0x1a, 0x45, 0x88, 0xb1, 0x16, 0x20,
- 0x24, 0xd3, 0x0a, 0x61, 0x4c, 0x20,
- 0x71, 0x10, 0x72, 0x0e, 0x41, 0xc0,
- 0xa0, 0x65, 0x1f, 0xa0, 0xc4, 0xe0,
- 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
- 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
- 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60
- };
-
- const WebRtc_UWord8 mask43_26[156] =
- {
- 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
- 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
- 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
- 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
- 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
- 0xc6, 0x21, 0xa4, 0x34, 0x86, 0x80,
- 0x23, 0x88, 0xc9, 0x19, 0x23, 0x20,
- 0x1a, 0x45, 0x88, 0xb1, 0x16, 0x20,
- 0x24, 0xd3, 0x0a, 0x61, 0x4c, 0x20,
- 0x71, 0x10, 0x72, 0x0e, 0x41, 0xc0,
- 0xa0, 0x65, 0x1f, 0xa0, 0xc4, 0xe0,
- 0xef, 0x84, 0x77, 0xca, 0x0d, 0x40
- };
-
- const WebRtc_UWord8 mask43_27[162] =
- {
- 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
- 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
- 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
- 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
- 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
- 0x4d, 0xd0, 0xc6, 0x36, 0x57, 0x40
- };
-
- const WebRtc_UWord8 mask43_28[168] =
- {
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
- 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
- 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
- 0x4d, 0xd0, 0xc6, 0x36, 0x57, 0x40,
- 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
- 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
- 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x16, 0xc9, 0x53, 0x1e, 0xc4, 0x00
- };
-
- const WebRtc_UWord8 mask43_29[174] =
- {
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
- 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
- 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
- 0x4d, 0xd0, 0xc6, 0x36, 0x57, 0x40,
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20
- };
-
- const WebRtc_UWord8 mask43_3[18] =
- {
- 0xce, 0x32, 0xb6, 0x56, 0xca, 0xc0,
- 0x55, 0xdc, 0x57, 0x8a, 0xf1, 0x40,
- 0xa8, 0xed, 0x8d, 0xb1, 0xae, 0x20
- };
-
- const WebRtc_UWord8 mask43_30[180] =
- {
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x0e, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x33, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0x10, 0xc3, 0x28, 0x65, 0x0c, 0xa0,
- 0x45, 0x68, 0x4d, 0x09, 0xa1, 0x20,
- 0x88, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0xe0, 0x22, 0x94, 0x52, 0x8a, 0x40,
- 0x4d, 0xd0, 0xc6, 0x36, 0x57, 0x40,
- 0x79, 0x4a, 0x8f, 0x42, 0x79, 0x40
- };
-
- const WebRtc_UWord8 mask43_31[186] =
- {
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x3b, 0xf5, 0x3c, 0x36, 0x0a, 0x20
- };
-
- const WebRtc_UWord8 mask43_32[192] =
- {
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x3b, 0xf5, 0x3c, 0x36, 0x0a, 0x20,
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0xd1, 0xd1, 0x11, 0xa4, 0xed, 0xc0
- };
-
- const WebRtc_UWord8 mask43_33[198] =
- {
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x3b, 0xf5, 0x3c, 0x36, 0x0a, 0x20,
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00
- };
-
- const WebRtc_UWord8 mask43_34[204] =
- {
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0x46, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x33, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0x10, 0xe4, 0x64, 0x8c, 0x91, 0x80,
- 0x0c, 0x69, 0x0d, 0x21, 0xa4, 0x20,
- 0x28, 0x94, 0x2a, 0x85, 0x50, 0xa0,
- 0x94, 0x21, 0x34, 0x26, 0x84, 0xc0,
- 0xc1, 0x02, 0x58, 0x4b, 0x09, 0x60,
- 0x3b, 0xf5, 0x3c, 0x36, 0x0a, 0x20,
- 0x76, 0x81, 0x4d, 0x33, 0x66, 0x00
- };
-
- const WebRtc_UWord8 mask43_35[210] =
- {
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0x5a, 0x56, 0x5f, 0x26, 0xa3, 0x60
- };
-
- const WebRtc_UWord8 mask43_36[216] =
- {
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0x5a, 0x56, 0x5f, 0x26, 0xa3, 0x60,
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0xa3, 0x85, 0x0a, 0xb5, 0x11, 0x60
- };
-
- const WebRtc_UWord8 mask43_37[222] =
- {
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0x5a, 0x56, 0x5f, 0x26, 0xa3, 0x60,
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
- 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
- 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
- 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
- 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
- 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
- 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0
- };
-
- const WebRtc_UWord8 mask43_38[228] =
- {
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
- 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
- 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
- 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
- 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
- 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
- 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0x2c, 0x03, 0x20, 0x64, 0x0c, 0x80,
- 0x81, 0xa0, 0x1c, 0x03, 0x80, 0x60,
- 0xa0, 0x68, 0x25, 0x04, 0xa0, 0x80,
- 0x05, 0x41, 0x50, 0x2a, 0x05, 0x40,
- 0x18, 0x90, 0xc2, 0x18, 0x43, 0x00,
- 0xc2, 0x06, 0x80, 0xd0, 0x1a, 0x00,
- 0x22, 0x98, 0x0b, 0x01, 0x60, 0x20,
- 0x50, 0x45, 0x08, 0xa1, 0x14, 0x20,
- 0x5a, 0x56, 0x5f, 0x26, 0xa3, 0x60,
- 0x9a, 0x16, 0x97, 0x21, 0xb9, 0x80
- };
-
- const WebRtc_UWord8 mask43_39[234] =
- {
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
- 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
- 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
- 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
- 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
- 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
- 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
- 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
- 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
- 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
- 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
- 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
- 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0x2a, 0x03, 0x31, 0xda, 0x46, 0x20
- };
-
- const WebRtc_UWord8 mask43_4[24] =
- {
- 0xe6, 0x31, 0x36, 0x26, 0xc4, 0xc0,
- 0x33, 0x8c, 0x59, 0x8b, 0x31, 0x60,
- 0x98, 0xd2, 0xca, 0x59, 0x4b, 0x20,
- 0x2d, 0x4b, 0x29, 0x65, 0x2c, 0xa0
- };
-
- const WebRtc_UWord8 mask43_40[240] =
- {
- 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
- 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
- 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
- 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
- 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
- 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
- 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0x2a, 0x03, 0x31, 0xda, 0x46, 0x20,
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
- 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
- 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
- 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
- 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
- 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
- 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0x3a, 0xab, 0x77, 0x63, 0xef, 0x60
- };
-
- const WebRtc_UWord8 mask43_41[246] =
- {
- 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
- 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
- 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
- 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
- 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
- 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
- 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0x2a, 0x03, 0x31, 0xda, 0x46, 0x20,
- 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
- 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
- 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
- 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
- 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
- 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
- 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0x4c, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x51, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0xa0, 0x66, 0x44, 0xc8, 0x99, 0x00,
- 0x04, 0xc1, 0x60, 0x2c, 0x05, 0x80,
- 0x03, 0xa0, 0x2c, 0x05, 0x80, 0xa0,
- 0x86, 0x21, 0x14, 0x22, 0x84, 0x40,
- 0x29, 0x10, 0x1a, 0x03, 0x40, 0x60,
- 0x42, 0x42, 0xa0, 0x54, 0x0a, 0x80,
- 0x98, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x30, 0x84, 0x08, 0x81, 0x10, 0x20,
- 0xdf, 0x4c, 0x11, 0x82, 0x30, 0x40
- };
-
- const WebRtc_UWord8 mask43_42[252] =
- {
- 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
- 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
- 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
- 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
- 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
- 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
- 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0x4c, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x51, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0xa0, 0x66, 0x44, 0xc8, 0x99, 0x00,
- 0x04, 0xc1, 0x60, 0x2c, 0x05, 0x80,
- 0x03, 0xa0, 0x2c, 0x05, 0x80, 0xa0,
- 0x86, 0x21, 0x14, 0x22, 0x84, 0x40,
- 0x29, 0x10, 0x1a, 0x03, 0x40, 0x60,
- 0x42, 0x42, 0xa0, 0x54, 0x0a, 0x80,
- 0x98, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x30, 0x84, 0x08, 0x81, 0x10, 0x20,
- 0xdf, 0x4c, 0x11, 0x82, 0x30, 0x40,
- 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
- 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
- 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
- 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
- 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
- 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
- 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0x4e, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0xe3, 0x18, 0x9b, 0x13, 0x62, 0x60,
- 0x81, 0xe3, 0x04, 0x60, 0x8c, 0x00,
- 0x21, 0x40, 0x58, 0x0b, 0x01, 0x60,
- 0x52, 0x81, 0xe0, 0x3c, 0x07, 0x80,
- 0xb4, 0x28, 0x25, 0x04, 0xa0, 0x80,
- 0x26, 0x86, 0x28, 0xc5, 0x18, 0xa0,
- 0x58, 0x64, 0x44, 0x88, 0x91, 0x00,
- 0x19, 0x9e, 0x03, 0xc0, 0x78, 0x00,
- 0x2a, 0x03, 0x31, 0xda, 0x46, 0x20,
- 0x26, 0x84, 0x10, 0xcd, 0xf7, 0x60
- };
-
- const WebRtc_UWord8 mask43_43[258] =
- {
- 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0xa0, 0x6a, 0x45, 0x48, 0xa9, 0x00,
- 0x04, 0xc1, 0x30, 0x26, 0x04, 0xc0,
- 0x03, 0xb4, 0x06, 0x80, 0xd0, 0x00,
- 0x86, 0x20, 0x94, 0x12, 0x82, 0x40,
- 0x29, 0x08, 0x49, 0x09, 0x21, 0x20,
- 0x42, 0x43, 0x08, 0x61, 0x0c, 0x20,
- 0x98, 0x12, 0x82, 0x50, 0x4a, 0x00,
- 0x30, 0x84, 0xa8, 0x95, 0x12, 0xa0,
- 0x4c, 0x11, 0x92, 0x32, 0x46, 0x40,
- 0x51, 0x0c, 0xc9, 0x99, 0x33, 0x20,
- 0xa0, 0x66, 0x44, 0xc8, 0x99, 0x00,
- 0x04, 0xc1, 0x60, 0x2c, 0x05, 0x80,
- 0x03, 0xa0, 0x2c, 0x05, 0x80, 0xa0,
- 0x86, 0x21, 0x14, 0x22, 0x84, 0x40,
- 0x29, 0x10, 0x1a, 0x03, 0x40, 0x60,
- 0x42, 0x42, 0xa0, 0x54, 0x0a, 0x80,
- 0x98, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x30, 0x84, 0x08, 0x81, 0x10, 0x20,
- 0xdf, 0x4c, 0x11, 0x82, 0x30, 0x40,
- 0x4c, 0x19, 0x12, 0x32, 0x46, 0x40,
- 0x51, 0x14, 0x51, 0x99, 0x33, 0x20,
- 0xa0, 0x6a, 0x44, 0xc8, 0x99, 0x00,
- 0x04, 0xc1, 0x30, 0x2c, 0x05, 0x80,
- 0x03, 0xb4, 0x04, 0x05, 0x80, 0xa0,
- 0x86, 0x20, 0x94, 0x22, 0x84, 0x40,
- 0x29, 0x08, 0x4a, 0x03, 0x40, 0x60,
- 0x42, 0x43, 0x08, 0x54, 0x0a, 0x80,
- 0x98, 0x12, 0x83, 0x40, 0x68, 0x00,
- 0x30, 0x84, 0xa8, 0x81, 0x10, 0x20,
- 0x4c, 0x11, 0x91, 0x82, 0x30, 0x40,
- 0x51, 0x0c, 0xcb, 0x22, 0x64, 0x40,
- 0xa0, 0x66, 0x42, 0x8a, 0x51, 0x40,
- 0x04, 0xc1, 0x65, 0x48, 0xa9, 0x00,
- 0x03, 0xa0, 0x28, 0x26, 0x04, 0xc0,
- 0x86, 0x21, 0x16, 0x80, 0xd0, 0x00,
- 0x29, 0x10, 0x1c, 0x12, 0x82, 0x40,
- 0x42, 0x42, 0xa1, 0x09, 0x21, 0x20,
- 0x98, 0x1a, 0x00, 0x61, 0x0c, 0x20,
- 0x30, 0x84, 0x0a, 0x50, 0x4a, 0x00,
- 0xdf, 0x4c, 0x10, 0x95, 0x12, 0xa0,
- 0x72, 0x06, 0x94, 0xf6, 0x74, 0x40
- };
-
- const WebRtc_UWord8 mask43_5[30] =
- {
- 0xce, 0x31, 0xb6, 0x36, 0xc6, 0xc0,
- 0x63, 0x98, 0xdb, 0x1b, 0x63, 0x60,
- 0x98, 0xc7, 0x68, 0xed, 0x1d, 0xa0,
- 0x4d, 0x6b, 0x55, 0x6a, 0xad, 0x40,
- 0xb2, 0x6c, 0xad, 0x95, 0xb2, 0xa0
- };
-
- const WebRtc_UWord8 mask43_6[36] =
- {
- 0x4c, 0x19, 0x13, 0x22, 0x64, 0x40,
- 0x51, 0x14, 0x52, 0x8a, 0x51, 0x40,
- 0x20, 0xea, 0x0d, 0x41, 0xa8, 0x20,
- 0x85, 0x41, 0x2e, 0x25, 0x04, 0xa0,
- 0x06, 0x80, 0xd8, 0x1b, 0x03, 0x60,
- 0x8a, 0x24, 0x34, 0x86, 0x90, 0xc0
- };
-
- const WebRtc_UWord8 mask43_7[42] =
- {
- 0xc6, 0x11, 0x96, 0x32, 0x46, 0x40,
- 0x33, 0x04, 0xc8, 0x99, 0x33, 0x20,
- 0x18, 0x67, 0x44, 0x68, 0x9d, 0x00,
- 0x45, 0x42, 0xd4, 0x5a, 0x0b, 0x40,
- 0x12, 0xd4, 0x2a, 0x95, 0x50, 0xa0,
- 0xb4, 0x28, 0x35, 0x16, 0xa0, 0xc0,
- 0x29, 0x92, 0x1b, 0x0d, 0x41, 0x60
- };
-
- const WebRtc_UWord8 mask43_8[48] =
- {
- 0x07, 0x0a, 0x71, 0x44, 0x29, 0xc0,
- 0x49, 0xa8, 0x29, 0x0f, 0xa0, 0x20,
- 0xb0, 0x7a, 0x07, 0x48, 0xe8, 0x00,
- 0x24, 0xc5, 0xc0, 0xb8, 0x17, 0x00,
- 0x52, 0x80, 0xec, 0x1d, 0x02, 0xa0,
- 0xc6, 0x31, 0x82, 0x30, 0xc7, 0x40,
- 0x31, 0x94, 0x1a, 0x83, 0x50, 0x60,
- 0x18, 0xc7, 0x08, 0xe1, 0x1c, 0x20
- };
-
- const WebRtc_UWord8 mask43_9[54] =
- {
- 0x4e, 0x11, 0x12, 0x22, 0x46, 0x40,
- 0x62, 0x1a, 0x09, 0x41, 0x68, 0x60,
- 0x80, 0xe9, 0x41, 0x28, 0xa5, 0x00,
- 0xa1, 0x50, 0x52, 0xc8, 0x51, 0x00,
- 0x53, 0x00, 0x68, 0x1d, 0x01, 0xa0,
- 0xa4, 0x24, 0x36, 0x06, 0x80, 0xc0,
- 0x16, 0xa0, 0x8d, 0x11, 0x82, 0x20,
- 0x58, 0x45, 0x20, 0xa4, 0x16, 0x80,
- 0x29, 0x86, 0x84, 0xd0, 0x1c, 0x00
- };
-
- const WebRtc_UWord8 mask44_1[6] =
- {
- 0xff, 0xff, 0xff, 0xff, 0xff, 0xf0
- };
-
- const WebRtc_UWord8 mask44_10[60] =
- {
- 0xc0, 0x38, 0x8b, 0x00, 0xe2, 0x20,
- 0x30, 0x0e, 0x28, 0xc0, 0x38, 0xa0,
- 0xe8, 0x07, 0x03, 0xa0, 0x1c, 0x00,
- 0x85, 0x08, 0xaa, 0x14, 0x22, 0xa0,
- 0xd0, 0x92, 0x13, 0x42, 0x48, 0x40,
- 0x86, 0x50, 0x4a, 0x19, 0x41, 0x20,
- 0x4a, 0x68, 0x0d, 0x29, 0xa0, 0x30,
- 0x01, 0xa0, 0x74, 0x06, 0x81, 0xd0,
- 0x4c, 0x81, 0x91, 0x32, 0x06, 0x40,
- 0x62, 0x24, 0x05, 0x88, 0x90, 0x10
- };
-
- const WebRtc_UWord8 mask44_11[66] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
- 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
- 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
- 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
- 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
- 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
- 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
- 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
- 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20
- };
-
- const WebRtc_UWord8 mask44_12[72] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
- 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
- 0x86, 0x90, 0xd2, 0x1a, 0x43, 0x40,
- 0x23, 0x24, 0x64, 0x8c, 0x91, 0x90,
- 0x16, 0x22, 0xc4, 0x58, 0x8b, 0x10,
- 0x4c, 0x29, 0x85, 0x30, 0xa6, 0x10,
- 0x41, 0xc8, 0x39, 0x07, 0x20, 0xe0,
- 0xf4, 0x18, 0x9f, 0xd0, 0x62, 0x70
- };
-
- const WebRtc_UWord8 mask44_13[78] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
- 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0
- };
-
- const WebRtc_UWord8 mask44_14[84] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
- 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
- 0xc6, 0xca, 0xeb, 0x1b, 0x2b, 0xa0
- };
-
- const WebRtc_UWord8 mask44_15[90] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10
- };
-
- const WebRtc_UWord8 mask44_16[96] =
- {
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x86, 0xc1, 0x46, 0x1b, 0x05, 0x10
- };
-
- const WebRtc_UWord8 mask44_17[102] =
- {
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00
- };
-
- const WebRtc_UWord8 mask44_18[108] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0xe4, 0xd4, 0x6f, 0x93, 0x51, 0xb0
- };
-
- const WebRtc_UWord8 mask44_19[114] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50
- };
-
- const WebRtc_UWord8 mask44_2[12] =
- {
- 0xec, 0xdd, 0x9b, 0xb3, 0x76, 0x60,
- 0x9b, 0xb3, 0x76, 0x6e, 0xcd, 0xd0
- };
-
- const WebRtc_UWord8 mask44_20[120] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0x3b, 0x48, 0xc4, 0xed, 0x23, 0x10
- };
-
- const WebRtc_UWord8 mask44_21[126] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
- 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
- 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
- 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
- 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
- 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
- 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
- 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
- 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20
- };
-
- const WebRtc_UWord8 mask44_22[132] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
- 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
- 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
- 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
- 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
- 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
- 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
- 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
- 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x9e, 0xce, 0x8a, 0x7b, 0x3a, 0x20
- };
-
- const WebRtc_UWord8 mask44_23[138] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
- 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
- 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
- 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
- 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
- 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
- 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
- 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
- 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
- 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
- 0x86, 0x90, 0xd2, 0x1a, 0x43, 0x40,
- 0x23, 0x24, 0x64, 0x8c, 0x91, 0x90,
- 0x16, 0x22, 0xc4, 0x58, 0x8b, 0x10,
- 0x4c, 0x29, 0x85, 0x30, 0xa6, 0x10,
- 0x41, 0xc8, 0x39, 0x07, 0x20, 0xe0,
- 0xf4, 0x18, 0x9f, 0xd0, 0x62, 0x70
- };
-
- const WebRtc_UWord8 mask44_24[144] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
- 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
- 0x86, 0x90, 0xd2, 0x1a, 0x43, 0x40,
- 0x23, 0x24, 0x64, 0x8c, 0x91, 0x90,
- 0x16, 0x22, 0xc4, 0x58, 0x8b, 0x10,
- 0x4c, 0x29, 0x85, 0x30, 0xa6, 0x10,
- 0x41, 0xc8, 0x39, 0x07, 0x20, 0xe0,
- 0xf4, 0x18, 0x9f, 0xd0, 0x62, 0x70,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
- 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
- 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
- 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
- 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
- 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
- 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
- 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
- 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
- 0x15, 0x0f, 0x44, 0x6d, 0x9d, 0xa0
- };
-
- const WebRtc_UWord8 mask44_25[150] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
- 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
- 0x86, 0x90, 0xd2, 0x1a, 0x43, 0x40,
- 0x23, 0x24, 0x64, 0x8c, 0x91, 0x90,
- 0x16, 0x22, 0xc4, 0x58, 0x8b, 0x10,
- 0x4c, 0x29, 0x85, 0x30, 0xa6, 0x10,
- 0x41, 0xc8, 0x39, 0x07, 0x20, 0xe0,
- 0xf4, 0x18, 0x9f, 0xd0, 0x62, 0x70,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
- 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0
- };
-
- const WebRtc_UWord8 mask44_26[156] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
- 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
- 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
- 0x86, 0x90, 0xd2, 0x1a, 0x43, 0x40,
- 0x23, 0x24, 0x64, 0x8c, 0x91, 0x90,
- 0x16, 0x22, 0xc4, 0x58, 0x8b, 0x10,
- 0x4c, 0x29, 0x85, 0x30, 0xa6, 0x10,
- 0x41, 0xc8, 0x39, 0x07, 0x20, 0xe0,
- 0xf4, 0x18, 0x9f, 0xd0, 0x62, 0x70,
- 0x02, 0xcb, 0x64, 0xb8, 0x55, 0x80
- };
-
- const WebRtc_UWord8 mask44_27[162] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
- 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
- 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
- 0xc6, 0xca, 0xeb, 0x1b, 0x2b, 0xa0
- };
-
- const WebRtc_UWord8 mask44_28[168] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
- 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
- 0xc6, 0xca, 0xeb, 0x1b, 0x2b, 0xa0,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
- 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x66, 0x26, 0x6c, 0x91, 0xc7, 0x20
- };
-
- const WebRtc_UWord8 mask44_29[174] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
- 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
- 0xc6, 0xca, 0xeb, 0x1b, 0x2b, 0xa0,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10
- };
-
- const WebRtc_UWord8 mask44_3[18] =
- {
- 0xca, 0xd9, 0x5b, 0x2b, 0x65, 0x60,
- 0xf1, 0x5e, 0x2b, 0xc5, 0x78, 0xa0,
- 0xb6, 0x35, 0xc6, 0xd8, 0xd7, 0x10
- };
-
- const WebRtc_UWord8 mask44_30[180] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0x0c, 0xa1, 0x94, 0x32, 0x86, 0x50,
- 0xa1, 0x34, 0x26, 0x84, 0xd0, 0x90,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x8a, 0x51, 0x4a, 0x29, 0x45, 0x20,
- 0xc6, 0xca, 0xeb, 0x1b, 0x2b, 0xa0,
- 0x60, 0xf4, 0x75, 0x84, 0x90, 0xc0
- };
-
- const WebRtc_UWord8 mask44_31[186] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x86, 0xc1, 0x46, 0x1b, 0x05, 0x10
- };
-
- const WebRtc_UWord8 mask44_32[192] =
- {
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x86, 0xc1, 0x46, 0x1b, 0x05, 0x10,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0x3e, 0x39, 0x86, 0x5c, 0xd9, 0xd0
- };
-
- const WebRtc_UWord8 mask44_33[198] =
- {
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x86, 0xc1, 0x46, 0x1b, 0x05, 0x10,
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00
- };
-
- const WebRtc_UWord8 mask44_34[204] =
- {
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0xc0,
- 0xa4, 0x34, 0x86, 0x90, 0xd2, 0x10,
- 0x50, 0xaa, 0x15, 0x42, 0xa8, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x42, 0x60,
- 0x09, 0x61, 0x2c, 0x25, 0x84, 0xb0,
- 0x86, 0xc1, 0x46, 0x1b, 0x05, 0x10,
- 0xb5, 0xc7, 0xe8, 0x0c, 0xb9, 0x90
- };
-
- const WebRtc_UWord8 mask44_35[210] =
- {
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0xe4, 0xd4, 0x6f, 0x93, 0x51, 0xb0
- };
-
- const WebRtc_UWord8 mask44_36[216] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0xe4, 0xd4, 0x6f, 0x93, 0x51, 0xb0,
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0xa6, 0x92, 0x01, 0x65, 0x91, 0x20
- };
-
- const WebRtc_UWord8 mask44_37[222] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0xe4, 0xd4, 0x6f, 0x93, 0x51, 0xb0,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50
- };
-
- const WebRtc_UWord8 mask44_38[228] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0x0c, 0x81, 0x90, 0x32, 0x06, 0x40,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x30,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x05, 0x40, 0xa8, 0x15, 0x02, 0xa0,
- 0x43, 0x08, 0x61, 0x0c, 0x21, 0x80,
- 0x1a, 0x03, 0x40, 0x68, 0x0d, 0x00,
- 0x60, 0x2c, 0x05, 0x80, 0xb0, 0x10,
- 0x14, 0x22, 0x84, 0x50, 0x8a, 0x10,
- 0xe4, 0xd4, 0x6f, 0x93, 0x51, 0xb0,
- 0x43, 0x64, 0xf2, 0xe5, 0x5d, 0x10
- };
-
- const WebRtc_UWord8 mask44_39[234] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0x3b, 0x48, 0xc4, 0xed, 0x23, 0x10
- };
-
- const WebRtc_UWord8 mask44_4[24] =
- {
- 0xc4, 0xd8, 0x9b, 0x13, 0x62, 0x60,
- 0x31, 0x66, 0x2c, 0xc5, 0x98, 0xb0,
- 0x4b, 0x29, 0x65, 0x2c, 0xa5, 0x90,
- 0x2c, 0xa5, 0x94, 0xb2, 0x96, 0x50
- };
-
- const WebRtc_UWord8 mask44_40[240] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0x3b, 0x48, 0xc4, 0xed, 0x23, 0x10,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0xd8, 0x2a, 0x16, 0x26, 0x51, 0x40
- };
-
- const WebRtc_UWord8 mask44_41[246] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0x3b, 0x48, 0xc4, 0xed, 0x23, 0x10,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
- 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
- 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
- 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
- 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
- 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
- 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
- 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
- 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20
- };
-
- const WebRtc_UWord8 mask44_42[252] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
- 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
- 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
- 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
- 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
- 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
- 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
- 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
- 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x62, 0x6c, 0x4d, 0x89, 0xb1, 0x30,
- 0x8c, 0x11, 0x82, 0x30, 0x46, 0x00,
- 0x01, 0x60, 0x2c, 0x05, 0x80, 0xb0,
- 0x07, 0x80, 0xf0, 0x1e, 0x03, 0xc0,
- 0xa0, 0x94, 0x12, 0x82, 0x50, 0x40,
- 0x18, 0xa3, 0x14, 0x62, 0x8c, 0x50,
- 0x91, 0x12, 0x22, 0x44, 0x48, 0x80,
- 0x78, 0x0f, 0x01, 0xe0, 0x3c, 0x00,
- 0x3b, 0x48, 0xc4, 0xed, 0x23, 0x10,
- 0xd9, 0xc1, 0x6f, 0xa8, 0x1c, 0x90
- };
-
- const WebRtc_UWord8 mask44_43[258] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
- 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
- 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
- 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
- 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
- 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
- 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
- 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
- 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
- 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
- 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
- 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
- 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
- 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
- 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
- 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
- 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x9e, 0xce, 0x8a, 0x7b, 0x3a, 0x20
- };
-
- const WebRtc_UWord8 mask44_44[264] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
- 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
- 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
- 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
- 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
- 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
- 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
- 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
- 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x9e, 0xce, 0x8a, 0x7b, 0x3a, 0x20,
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa9, 0x15, 0x22, 0xa4, 0x54, 0x80,
- 0x04, 0xc0, 0x98, 0x13, 0x02, 0x60,
- 0xd0, 0x1a, 0x03, 0x40, 0x68, 0x00,
- 0x82, 0x50, 0x4a, 0x09, 0x41, 0x20,
- 0x21, 0x24, 0x24, 0x84, 0x90, 0x90,
- 0x0c, 0x21, 0x84, 0x30, 0x86, 0x10,
- 0x4a, 0x09, 0x41, 0x28, 0x25, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0x89, 0x50,
- 0x46, 0x48, 0xc9, 0x19, 0x23, 0x20,
- 0x33, 0x26, 0x64, 0xcc, 0x99, 0x90,
- 0x99, 0x13, 0x22, 0x64, 0x4c, 0x80,
- 0x05, 0x80, 0xb0, 0x16, 0x02, 0xc0,
- 0x80, 0xb0, 0x16, 0x02, 0xc0, 0x50,
- 0x84, 0x50, 0x8a, 0x11, 0x42, 0x20,
- 0x40, 0x68, 0x0d, 0x01, 0xa0, 0x30,
- 0x0a, 0x81, 0x50, 0x2a, 0x05, 0x40,
- 0x68, 0x0d, 0x01, 0xa0, 0x34, 0x00,
- 0x10, 0x22, 0x04, 0x40, 0x88, 0x10,
- 0x30, 0x46, 0x08, 0xc1, 0x18, 0x20,
- 0xb5, 0x1c, 0x1c, 0x21, 0xac, 0xa0
- };
-
- const WebRtc_UWord8 mask44_5[30] =
- {
- 0xc6, 0xd8, 0xdb, 0x1b, 0x63, 0x60,
- 0x63, 0x6c, 0x6d, 0x8d, 0xb1, 0xb0,
- 0x1d, 0xa3, 0xb4, 0x76, 0x8e, 0xd0,
- 0xad, 0x55, 0xaa, 0xb5, 0x56, 0xa0,
- 0xb2, 0xb6, 0x56, 0xca, 0xd9, 0x50
- };
-
- const WebRtc_UWord8 mask44_6[36] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x32, 0x20,
- 0x51, 0x4a, 0x29, 0x45, 0x28, 0xa0,
- 0xa8, 0x35, 0x06, 0xa0, 0xd4, 0x10,
- 0xc4, 0xa0, 0x97, 0x12, 0x82, 0x50,
- 0x03, 0x60, 0x6c, 0x0d, 0x81, 0xb0,
- 0x90, 0xd2, 0x1a, 0x43, 0x48, 0x60
- };
-
- const WebRtc_UWord8 mask44_7[42] =
- {
- 0xc6, 0x48, 0xcb, 0x19, 0x23, 0x20,
- 0x13, 0x26, 0x64, 0x4c, 0x99, 0x90,
- 0x8d, 0x13, 0xa2, 0x34, 0x4e, 0x80,
- 0x8b, 0x41, 0x6a, 0x2d, 0x05, 0xa0,
- 0x52, 0xaa, 0x15, 0x4a, 0xa8, 0x50,
- 0xa2, 0xd4, 0x1a, 0x8b, 0x50, 0x60,
- 0x61, 0xa8, 0x2d, 0x86, 0xa0, 0xb0
- };
-
- const WebRtc_UWord8 mask44_8[48] =
- {
- 0x28, 0x85, 0x38, 0xa2, 0x14, 0xe0,
- 0x21, 0xf4, 0x04, 0x87, 0xd0, 0x10,
- 0xe9, 0x1d, 0x03, 0xa4, 0x74, 0x00,
- 0x17, 0x02, 0xe0, 0x5c, 0x0b, 0x80,
- 0x83, 0xa0, 0x56, 0x0e, 0x81, 0x50,
- 0x46, 0x18, 0xe9, 0x18, 0x63, 0xa0,
- 0x50, 0x6a, 0x0d, 0x41, 0xa8, 0x30,
- 0x1c, 0x23, 0x84, 0x70, 0x8e, 0x10
- };
-
- const WebRtc_UWord8 mask44_9[54] =
- {
- 0x44, 0x48, 0xc9, 0x11, 0x23, 0x20,
- 0x28, 0x2d, 0x0c, 0xa0, 0xb4, 0x30,
- 0x25, 0x14, 0xa0, 0x94, 0x52, 0x80,
- 0x59, 0x0a, 0x21, 0x64, 0x28, 0x80,
- 0x03, 0xa0, 0x34, 0x0e, 0x80, 0xd0,
- 0xc0, 0xd0, 0x1b, 0x03, 0x40, 0x60,
- 0xa2, 0x30, 0x46, 0x88, 0xc1, 0x10,
- 0x14, 0x82, 0xd0, 0x52, 0x0b, 0x40,
- 0x9a, 0x03, 0x82, 0x68, 0x0e, 0x00
- };
-
- const WebRtc_UWord8 mask45_1[6] =
- {
- 0xff, 0xff, 0xff, 0xff, 0xff, 0xf8
- };
-
- const WebRtc_UWord8 mask45_10[60] =
- {
- 0xc0, 0x38, 0x89, 0x91, 0x28, 0xa0,
- 0x30, 0x0e, 0x29, 0x45, 0x22, 0x88,
- 0xe8, 0x07, 0x02, 0xa4, 0x40, 0x68,
- 0x85, 0x08, 0xa8, 0x13, 0x12, 0x10,
- 0xd0, 0x92, 0x13, 0x40, 0x05, 0x10,
- 0x86, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x4a, 0x68, 0x0c, 0x84, 0xdc, 0x00,
- 0x01, 0xa0, 0x74, 0x30, 0x84, 0x88,
- 0x4c, 0x81, 0x91, 0x28, 0x2b, 0x00,
- 0x62, 0x24, 0x04, 0x4a, 0xd1, 0x40
- };
-
- const WebRtc_UWord8 mask45_11[66] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
- 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
- 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
- 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
- 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
- 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
- 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
- 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
- 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10
- };
-
- const WebRtc_UWord8 mask45_12[72] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
- 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
- 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
- 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
- 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
- 0x86, 0x90, 0xd2, 0x1a, 0x29, 0xb0,
- 0x23, 0x24, 0x64, 0x8c, 0xb2, 0x10,
- 0x16, 0x22, 0xc4, 0x58, 0x86, 0x60,
- 0x4c, 0x29, 0x85, 0x30, 0xc1, 0x50,
- 0x41, 0xc8, 0x39, 0x07, 0x04, 0x98,
- 0xf4, 0x18, 0x9c, 0x65, 0x5b, 0x90
- };
-
- const WebRtc_UWord8 mask45_13[78] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
- 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
- 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
- 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
- 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30
- };
-
- const WebRtc_UWord8 mask45_14[84] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
- 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
- 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
- 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
- 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
- 0xc6, 0xca, 0xea, 0x70, 0xfe, 0xc8
- };
-
- const WebRtc_UWord8 mask45_15[90] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80
- };
-
- const WebRtc_UWord8 mask45_16[96] =
- {
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x86, 0xc1, 0x47, 0xeb, 0x67, 0xd0
- };
-
- const WebRtc_UWord8 mask45_17[102] =
- {
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70
- };
-
- const WebRtc_UWord8 mask45_18[108] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0xe4, 0xd4, 0x6e, 0x08, 0xc9, 0x58
- };
-
- const WebRtc_UWord8 mask45_19[114] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40
- };
-
- const WebRtc_UWord8 mask45_2[12] =
- {
- 0xec, 0xdd, 0x9b, 0xb3, 0x76, 0x60,
- 0x9b, 0xb3, 0x76, 0x6e, 0xc9, 0xd8
- };
-
- const WebRtc_UWord8 mask45_20[120] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0x3b, 0x48, 0xc7, 0x6d, 0x29, 0xe8
- };
-
- const WebRtc_UWord8 mask45_21[126] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
- 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
- 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
- 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
- 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
- 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
- 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
- 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
- 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10
- };
-
- const WebRtc_UWord8 mask45_22[132] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
- 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
- 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
- 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
- 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
- 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
- 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
- 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
- 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
- 0x9e, 0xce, 0x8b, 0xaa, 0x34, 0x68
- };
-
- const WebRtc_UWord8 mask45_23[138] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
- 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
- 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
- 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
- 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
- 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
- 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
- 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
- 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
- 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
- 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
- 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
- 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
- 0x86, 0x90, 0xd2, 0x1a, 0x29, 0xb0,
- 0x23, 0x24, 0x64, 0x8c, 0xb2, 0x10,
- 0x16, 0x22, 0xc4, 0x58, 0x86, 0x60,
- 0x4c, 0x29, 0x85, 0x30, 0xc1, 0x50,
- 0x41, 0xc8, 0x39, 0x07, 0x04, 0x98,
- 0xf4, 0x18, 0x9c, 0x65, 0x5b, 0x90
- };
-
- const WebRtc_UWord8 mask45_24[144] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
- 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
- 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
- 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
- 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
- 0x86, 0x90, 0xd2, 0x1a, 0x29, 0xb0,
- 0x23, 0x24, 0x64, 0x8c, 0xb2, 0x10,
- 0x16, 0x22, 0xc4, 0x58, 0x86, 0x60,
- 0x4c, 0x29, 0x85, 0x30, 0xc1, 0x50,
- 0x41, 0xc8, 0x39, 0x07, 0x04, 0x98,
- 0xf4, 0x18, 0x9c, 0x65, 0x5b, 0x90,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
- 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
- 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
- 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
- 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
- 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
- 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
- 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
- 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
- 0x95, 0x91, 0xad, 0xd9, 0x86, 0x98
- };
-
- const WebRtc_UWord8 mask45_25[150] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
- 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
- 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
- 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
- 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
- 0x86, 0x90, 0xd2, 0x1a, 0x29, 0xb0,
- 0x23, 0x24, 0x64, 0x8c, 0xb2, 0x10,
- 0x16, 0x22, 0xc4, 0x58, 0x86, 0x60,
- 0x4c, 0x29, 0x85, 0x30, 0xc1, 0x50,
- 0x41, 0xc8, 0x39, 0x07, 0x04, 0x98,
- 0xf4, 0x18, 0x9c, 0x65, 0x5b, 0x90,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
- 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
- 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
- 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
- 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30
- };
-
- const WebRtc_UWord8 mask45_26[156] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
- 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
- 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
- 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
- 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
- 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
- 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
- 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
- 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
- 0x86, 0x90, 0xd2, 0x1a, 0x29, 0xb0,
- 0x23, 0x24, 0x64, 0x8c, 0xb2, 0x10,
- 0x16, 0x22, 0xc4, 0x58, 0x86, 0x60,
- 0x4c, 0x29, 0x85, 0x30, 0xc1, 0x50,
- 0x41, 0xc8, 0x39, 0x07, 0x04, 0x98,
- 0xf4, 0x18, 0x9c, 0x65, 0x5b, 0x90,
- 0xb0, 0xfd, 0xb2, 0xf3, 0x8a, 0xc0
- };
-
- const WebRtc_UWord8 mask45_27[162] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
- 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
- 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
- 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
- 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
- 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
- 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
- 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
- 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
- 0xc6, 0xca, 0xea, 0x70, 0xfe, 0xc8
- };
-
- const WebRtc_UWord8 mask45_28[168] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
- 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
- 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
- 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
- 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
- 0xc6, 0xca, 0xea, 0x70, 0xfe, 0xc8,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
- 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
- 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
- 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
- 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x44, 0x46, 0x28, 0xfb, 0x66, 0x80
- };
-
- const WebRtc_UWord8 mask45_29[174] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
- 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
- 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
- 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
- 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
- 0xc6, 0xca, 0xea, 0x70, 0xfe, 0xc8,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80
- };
-
- const WebRtc_UWord8 mask45_3[18] =
- {
- 0xca, 0xd9, 0x5b, 0x2b, 0x4d, 0x90,
- 0xf1, 0x5e, 0x2b, 0xc5, 0x24, 0xe8,
- 0xb6, 0x35, 0xc5, 0xd8, 0x9f, 0x40
- };
-
- const WebRtc_UWord8 mask45_30[180] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
- 0x0c, 0xa1, 0x94, 0x32, 0x90, 0xc0,
- 0xa1, 0x34, 0x26, 0x84, 0x89, 0x18,
- 0x12, 0xa2, 0x54, 0x4a, 0x84, 0x70,
- 0x8a, 0x51, 0x4a, 0x29, 0x17, 0x00,
- 0xc6, 0xca, 0xea, 0x70, 0xfe, 0xc8,
- 0x1c, 0xc9, 0x43, 0x25, 0xa7, 0x00
- };
-
- const WebRtc_UWord8 mask45_31[186] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x86, 0xc1, 0x47, 0xeb, 0x67, 0xd0
- };
-
- const WebRtc_UWord8 mask45_32[192] =
- {
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x86, 0xc1, 0x47, 0xeb, 0x67, 0xd0,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0x40, 0x7e, 0xc1, 0x30, 0x29, 0x50
- };
-
- const WebRtc_UWord8 mask45_33[198] =
- {
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x86, 0xc1, 0x47, 0xeb, 0x67, 0xd0,
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70
- };
-
- const WebRtc_UWord8 mask45_34[204] =
- {
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x91, 0x92, 0x32, 0x46, 0x48, 0x48,
- 0xa4, 0x34, 0x86, 0x90, 0x81, 0x28,
- 0x50, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0x84, 0xd0, 0x9a, 0x13, 0x16, 0x00,
- 0x09, 0x61, 0x2c, 0x25, 0xc4, 0x30,
- 0x86, 0xc1, 0x47, 0xeb, 0x67, 0xd0,
- 0x1f, 0x78, 0x45, 0x5e, 0x46, 0x50
- };
-
- const WebRtc_UWord8 mask45_35[210] =
- {
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0xe4, 0xd4, 0x6e, 0x08, 0xc9, 0x58
- };
-
- const WebRtc_UWord8 mask45_36[216] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0xe4, 0xd4, 0x6e, 0x08, 0xc9, 0x58,
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0xd0, 0x1a, 0xf0, 0x14, 0xf0, 0xe8
- };
-
- const WebRtc_UWord8 mask45_37[222] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0xe4, 0xd4, 0x6e, 0x08, 0xc9, 0x58,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40
- };
-
- const WebRtc_UWord8 mask45_38[228] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0x0c, 0x81, 0x90, 0x32, 0x10, 0x30,
- 0x80, 0x70, 0x0e, 0x01, 0xc0, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x21, 0x20,
- 0x05, 0x40, 0xa8, 0x15, 0x00, 0xc8,
- 0x43, 0x08, 0x61, 0x0c, 0x0a, 0x08,
- 0x1a, 0x03, 0x40, 0x68, 0x05, 0x40,
- 0x60, 0x2c, 0x05, 0x80, 0x9c, 0x00,
- 0x14, 0x22, 0x84, 0x50, 0xe2, 0x80,
- 0xe4, 0xd4, 0x6e, 0x08, 0xc9, 0x58,
- 0x04, 0x67, 0x1b, 0xba, 0x1d, 0xa0
- };
-
- const WebRtc_UWord8 mask45_39[234] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0x3b, 0x48, 0xc7, 0x6d, 0x29, 0xe8
- };
-
- const WebRtc_UWord8 mask45_4[24] =
- {
- 0xc4, 0xd8, 0x9b, 0x13, 0x45, 0x90,
- 0x31, 0x66, 0x2c, 0xc5, 0x8a, 0x58,
- 0x4b, 0x29, 0x65, 0x2c, 0x91, 0x68,
- 0x2c, 0xa5, 0x94, 0xb2, 0xa2, 0xa8
- };
-
- const WebRtc_UWord8 mask45_40[240] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0x3b, 0x48, 0xc7, 0x6d, 0x29, 0xe8,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
- 0xd9, 0x40, 0x46, 0xe6, 0x4f, 0xd8
- };
-
- const WebRtc_UWord8 mask45_41[246] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0x3b, 0x48, 0xc7, 0x6d, 0x29, 0xe8,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
- 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
- 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
- 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
- 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
- 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
- 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
- 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
- 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10
- };
-
- const WebRtc_UWord8 mask45_42[252] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
- 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
- 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
- 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
- 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
- 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
- 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
- 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
- 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x62, 0x6c, 0x4d, 0x89, 0xf2, 0x10,
- 0x8c, 0x11, 0x82, 0x30, 0x12, 0x20,
- 0x01, 0x60, 0x2c, 0x05, 0xd0, 0x88,
- 0x07, 0x80, 0xf0, 0x1e, 0x0c, 0x18,
- 0xa0, 0x94, 0x12, 0x82, 0x01, 0xc8,
- 0x18, 0xa3, 0x14, 0x62, 0xc5, 0x08,
- 0x91, 0x12, 0x22, 0x44, 0x02, 0x48,
- 0x78, 0x0f, 0x01, 0xe0, 0x00, 0x70,
- 0x3b, 0x48, 0xc7, 0x6d, 0x29, 0xe8,
- 0xac, 0xcc, 0x04, 0x41, 0x97, 0x30
- };
-
- const WebRtc_UWord8 mask45_43[258] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
- 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
- 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
- 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
- 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
- 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
- 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
- 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
- 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
- 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
- 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
- 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
- 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
- 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
- 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
- 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
- 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
- 0x9e, 0xce, 0x8b, 0xaa, 0x34, 0x68
- };
-
- const WebRtc_UWord8 mask45_44[264] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
- 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
- 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
- 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
- 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
- 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
- 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
- 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
- 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
- 0x9e, 0xce, 0x8b, 0xaa, 0x34, 0x68,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
- 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
- 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
- 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
- 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
- 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
- 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
- 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
- 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
- 0xf8, 0x40, 0xe3, 0x2e, 0x16, 0x00
- };
-
- const WebRtc_UWord8 mask45_45[270] =
- {
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
- 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
- 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
- 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
- 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
- 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
- 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
- 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
- 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x22, 0x88,
- 0xa9, 0x15, 0x22, 0xa4, 0x40, 0x68,
- 0x04, 0xc0, 0x98, 0x13, 0x12, 0x10,
- 0xd0, 0x1a, 0x03, 0x40, 0x05, 0x10,
- 0x82, 0x50, 0x4a, 0x09, 0x00, 0x70,
- 0x21, 0x24, 0x24, 0x84, 0xdc, 0x00,
- 0x0c, 0x21, 0x84, 0x30, 0x84, 0x88,
- 0x4a, 0x09, 0x41, 0x28, 0x2b, 0x00,
- 0x12, 0xa2, 0x54, 0x4a, 0xd1, 0x40,
- 0x9e, 0xce, 0x8b, 0xaa, 0x34, 0x68,
- 0x46, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x33, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x99, 0x13, 0x22, 0x64, 0x08, 0x48,
- 0x05, 0x80, 0xb0, 0x16, 0x00, 0x38,
- 0x80, 0xb0, 0x16, 0x02, 0x86, 0x08,
- 0x84, 0x50, 0x8a, 0x11, 0x20, 0x60,
- 0x40, 0x68, 0x0d, 0x01, 0xb5, 0x00,
- 0x0a, 0x81, 0x50, 0x2a, 0x43, 0x00,
- 0x68, 0x0d, 0x01, 0xa0, 0x12, 0x40,
- 0x10, 0x22, 0x04, 0x40, 0xc4, 0x80,
- 0x30, 0x46, 0x08, 0xc1, 0x60, 0x10,
- 0x64, 0x4c, 0x89, 0x19, 0x08, 0x30,
- 0x51, 0x4a, 0x28, 0xcc, 0x81, 0x18,
- 0xa9, 0x15, 0x22, 0x64, 0x20, 0x28,
- 0x04, 0xc0, 0x98, 0x16, 0x10, 0xc0,
- 0xd0, 0x1a, 0x02, 0x02, 0xc0, 0x88,
- 0x82, 0x50, 0x4a, 0x11, 0x0a, 0x40,
- 0x21, 0x24, 0x25, 0x01, 0xcc, 0x00,
- 0x0c, 0x21, 0x84, 0x2a, 0x04, 0x48,
- 0x4a, 0x09, 0x41, 0xa0, 0x31, 0x00,
- 0x12, 0xa2, 0x54, 0x40, 0x92, 0x10,
- 0x9e, 0xce, 0x88, 0xc1, 0x45, 0x00,
- 0xfb, 0x97, 0x5d, 0x7d, 0x42, 0x20
- };
-
- const WebRtc_UWord8 mask45_5[30] =
- {
- 0xc6, 0xd8, 0xdb, 0x1b, 0x29, 0xb0,
- 0x63, 0x6c, 0x6d, 0x8d, 0xb2, 0x58,
- 0x1d, 0xa3, 0xb4, 0x76, 0x87, 0x70,
- 0xad, 0x55, 0xaa, 0xb5, 0x54, 0xe0,
- 0xb2, 0xb6, 0x56, 0xca, 0xdc, 0x18
- };
-
- const WebRtc_UWord8 mask45_6[36] =
- {
- 0x64, 0x4c, 0x89, 0x91, 0x28, 0xa0,
- 0x51, 0x4a, 0x29, 0x45, 0x62, 0x88,
- 0xa8, 0x35, 0x04, 0x32, 0x90, 0xc0,
- 0xc4, 0xa0, 0x96, 0x84, 0x89, 0x18,
- 0x03, 0x60, 0x6c, 0x4a, 0x84, 0x70,
- 0x90, 0xd2, 0x1a, 0x29, 0x17, 0x00
- };
-
- const WebRtc_UWord8 mask45_7[42] =
- {
- 0xc6, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x13, 0x26, 0x64, 0xcc, 0x90, 0x98,
- 0x8d, 0x13, 0xa2, 0x46, 0x48, 0x48,
- 0x8b, 0x41, 0x6a, 0x90, 0x81, 0x28,
- 0x52, 0xaa, 0x15, 0x42, 0x83, 0x50,
- 0xa2, 0xd4, 0x1a, 0x13, 0x16, 0x00,
- 0x61, 0xa8, 0x2c, 0x25, 0xc4, 0x30
- };
-
- const WebRtc_UWord8 mask45_8[48] =
- {
- 0x28, 0x85, 0x38, 0x32, 0x10, 0x30,
- 0x21, 0xf4, 0x06, 0x01, 0xc0, 0x18,
- 0xe9, 0x1d, 0x02, 0x82, 0x21, 0x20,
- 0x17, 0x02, 0xe0, 0x15, 0x00, 0xc8,
- 0x83, 0xa0, 0x55, 0x0c, 0x0a, 0x08,
- 0x46, 0x18, 0xe8, 0x68, 0x05, 0x40,
- 0x50, 0x6a, 0x0d, 0x80, 0x9c, 0x00,
- 0x1c, 0x23, 0x84, 0x50, 0xe2, 0x80
- };
-
- const WebRtc_UWord8 mask45_9[54] =
- {
- 0x44, 0x48, 0xc9, 0x19, 0x29, 0xb0,
- 0x28, 0x2d, 0x0d, 0x89, 0xf2, 0x10,
- 0x25, 0x14, 0xa2, 0x30, 0x12, 0x20,
- 0x59, 0x0a, 0x20, 0x05, 0xd0, 0x88,
- 0x03, 0xa0, 0x34, 0x1e, 0x0c, 0x18,
- 0xc0, 0xd0, 0x1a, 0x82, 0x01, 0xc8,
- 0xa2, 0x30, 0x44, 0x62, 0xc5, 0x08,
- 0x14, 0x82, 0xd2, 0x44, 0x02, 0x48,
- 0x9a, 0x03, 0x81, 0xe0, 0x00, 0x70
- };
-
- const WebRtc_UWord8 mask46_1[6] =
- {
- 0xff, 0xff, 0xff, 0xff, 0xff, 0xfc
- };
-
- const WebRtc_UWord8 mask46_10[60] =
- {
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0
- };
-
- const WebRtc_UWord8 mask46_11[66] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
- 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
- 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
- 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
- 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
- 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
- 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
- 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
- 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08
- };
-
- const WebRtc_UWord8 mask46_12[72] =
- {
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
- 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
- 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
- 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
- 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
- 0x86, 0x8a, 0x6d, 0x0d, 0x14, 0xd8,
- 0x23, 0x2c, 0x84, 0x46, 0x59, 0x08,
- 0x16, 0x21, 0x98, 0x2c, 0x43, 0x30,
- 0x4c, 0x30, 0x54, 0x98, 0x60, 0xa8,
- 0x41, 0xc1, 0x26, 0x83, 0x82, 0x4c,
- 0x19, 0x56, 0xe4, 0x32, 0xad, 0xc8
- };
-
- const WebRtc_UWord8 mask46_13[78] =
- {
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
- 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
- 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
- 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
- 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18
- };
-
- const WebRtc_UWord8 mask46_14[84] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
- 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
- 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
- 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
- 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
- 0x9c, 0x3f, 0xb3, 0x38, 0x7f, 0x64
- };
-
- const WebRtc_UWord8 mask46_15[90] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40
- };
-
- const WebRtc_UWord8 mask46_16[96] =
- {
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0xfa, 0xd9, 0xf5, 0xf5, 0xb3, 0xe8
- };
-
- const WebRtc_UWord8 mask46_17[102] =
- {
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38
- };
-
- const WebRtc_UWord8 mask46_18[108] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x82, 0x32, 0x57, 0x04, 0x64, 0xac
- };
-
- const WebRtc_UWord8 mask46_19[114] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0
- };
-
- const WebRtc_UWord8 mask46_2[12] =
- {
- 0xec, 0xdd, 0x99, 0xd9, 0xbb, 0x30,
- 0x9b, 0xb2, 0x77, 0x37, 0x64, 0xec
- };
-
- const WebRtc_UWord8 mask46_20[120] =
- {
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0xdb, 0x4a, 0x7b, 0xb6, 0x94, 0xf4
- };
-
- const WebRtc_UWord8 mask46_21[126] =
- {
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
- 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
- 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
- 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
- 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
- 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
- 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
- 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
- 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08
- };
-
- const WebRtc_UWord8 mask46_22[132] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
- 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
- 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
- 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
- 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
- 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
- 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
- 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
- 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0xea, 0x8d, 0x1b, 0xd5, 0x1a, 0x34
- };
-
- const WebRtc_UWord8 mask46_23[138] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
- 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
- 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
- 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
- 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
- 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
- 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
- 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
- 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
- 0x46, 0x42, 0x0c, 0x8c, 0x84, 0x18,
- 0x33, 0x20, 0x46, 0x66, 0x40, 0x8c,
- 0x99, 0x08, 0x0b, 0x32, 0x10, 0x14,
- 0x05, 0x84, 0x30, 0x0b, 0x08, 0x60,
- 0x80, 0xb0, 0x23, 0x01, 0x60, 0x44,
- 0x84, 0x42, 0x91, 0x08, 0x85, 0x20,
- 0x40, 0x73, 0x00, 0x80, 0xe6, 0x00,
- 0x0a, 0x81, 0x12, 0x15, 0x02, 0x24,
- 0x68, 0x0c, 0x40, 0xd0, 0x18, 0x80,
- 0x10, 0x24, 0x84, 0x20, 0x49, 0x08,
- 0x30, 0x51, 0x40, 0x60, 0xa2, 0x80,
- 0x5f, 0x50, 0x88, 0xbe, 0xa1, 0x10
- };
-
- const WebRtc_UWord8 mask46_24[144] =
- {
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
- 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
- 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
- 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
- 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
- 0x86, 0x8a, 0x6d, 0x0d, 0x14, 0xd8,
- 0x23, 0x2c, 0x84, 0x46, 0x59, 0x08,
- 0x16, 0x21, 0x98, 0x2c, 0x43, 0x30,
- 0x4c, 0x30, 0x54, 0x98, 0x60, 0xa8,
- 0x41, 0xc1, 0x26, 0x83, 0x82, 0x4c,
- 0x19, 0x56, 0xe4, 0x32, 0xad, 0xc8,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
- 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
- 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
- 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
- 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
- 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
- 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
- 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
- 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
- 0x18, 0x8b, 0x03, 0xb4, 0x3b, 0x10
- };
-
- const WebRtc_UWord8 mask46_25[150] =
- {
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
- 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
- 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
- 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
- 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
- 0x86, 0x8a, 0x6d, 0x0d, 0x14, 0xd8,
- 0x23, 0x2c, 0x84, 0x46, 0x59, 0x08,
- 0x16, 0x21, 0x98, 0x2c, 0x43, 0x30,
- 0x4c, 0x30, 0x54, 0x98, 0x60, 0xa8,
- 0x41, 0xc1, 0x26, 0x83, 0x82, 0x4c,
- 0x19, 0x56, 0xe4, 0x32, 0xad, 0xc8,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
- 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
- 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
- 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
- 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18
- };
-
- const WebRtc_UWord8 mask46_26[156] =
- {
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
- 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
- 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
- 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
- 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
- 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
- 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
- 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
- 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
- 0x86, 0x8a, 0x6d, 0x0d, 0x14, 0xd8,
- 0x23, 0x2c, 0x84, 0x46, 0x59, 0x08,
- 0x16, 0x21, 0x98, 0x2c, 0x43, 0x30,
- 0x4c, 0x30, 0x54, 0x98, 0x60, 0xa8,
- 0x41, 0xc1, 0x26, 0x83, 0x82, 0x4c,
- 0x19, 0x56, 0xe4, 0x32, 0xad, 0xc8,
- 0x2d, 0x6d, 0xd2, 0x57, 0xd6, 0x2c
- };
-
- const WebRtc_UWord8 mask46_27[162] =
- {
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
- 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
- 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
- 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
- 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
- 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
- 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
- 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
- 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
- 0x9c, 0x3f, 0xb3, 0x38, 0x7f, 0x64
- };
-
- const WebRtc_UWord8 mask46_28[168] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
- 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
- 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
- 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
- 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
- 0x9c, 0x3f, 0xb3, 0x38, 0x7f, 0x64,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
- 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
- 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
- 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
- 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0xfa, 0x52, 0xf9, 0x72, 0xd9, 0x68
- };
-
- const WebRtc_UWord8 mask46_29[174] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
- 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
- 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
- 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
- 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
- 0x9c, 0x3f, 0xb3, 0x38, 0x7f, 0x64,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40
- };
-
- const WebRtc_UWord8 mask46_3[18] =
- {
- 0xca, 0xd3, 0x65, 0x95, 0xa6, 0xc8,
- 0xf1, 0x49, 0x3b, 0xe2, 0x92, 0x74,
- 0x76, 0x27, 0xd0, 0xec, 0x4f, 0xa0
- };
-
- const WebRtc_UWord8 mask46_30[180] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
- 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
- 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
- 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
- 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80,
- 0x9c, 0x3f, 0xb3, 0x38, 0x7f, 0x64,
- 0x99, 0xf6, 0x0a, 0xdd, 0x16, 0xb0
- };
-
- const WebRtc_UWord8 mask46_31[186] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0xfa, 0xd9, 0xf5, 0xf5, 0xb3, 0xe8
- };
-
- const WebRtc_UWord8 mask46_32[192] =
- {
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0xfa, 0xd9, 0xf5, 0xf5, 0xb3, 0xe8,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x69, 0xcd, 0xeb, 0x51, 0xc9, 0xa8
- };
-
- const WebRtc_UWord8 mask46_33[198] =
- {
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0xfa, 0xd9, 0xf5, 0xf5, 0xb3, 0xe8,
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38
- };
-
- const WebRtc_UWord8 mask46_34[204] =
- {
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18,
- 0xfa, 0xd9, 0xf5, 0xf5, 0xb3, 0xe8,
- 0x60, 0xf0, 0x13, 0xf0, 0x4d, 0xe0
- };
-
- const WebRtc_UWord8 mask46_35[210] =
- {
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x82, 0x32, 0x57, 0x04, 0x64, 0xac
- };
-
- const WebRtc_UWord8 mask46_36[216] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x82, 0x32, 0x57, 0x04, 0x64, 0xac,
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0x6c, 0x3a, 0x45, 0x70, 0xd7, 0x00
- };
-
- const WebRtc_UWord8 mask46_37[222] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x82, 0x32, 0x57, 0x04, 0x64, 0xac,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0
- };
-
- const WebRtc_UWord8 mask46_38[228] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40,
- 0x82, 0x32, 0x57, 0x04, 0x64, 0xac,
- 0x72, 0x2b, 0xa5, 0xd4, 0xb9, 0x30
- };
-
- const WebRtc_UWord8 mask46_39[234] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0xdb, 0x4a, 0x7b, 0xb6, 0x94, 0xf4
- };
-
- const WebRtc_UWord8 mask46_4[24] =
- {
- 0xc4, 0xd1, 0x65, 0x89, 0xa2, 0xc8,
- 0x31, 0x62, 0x96, 0x62, 0xc5, 0x2c,
- 0x4b, 0x24, 0x5a, 0x96, 0x48, 0xb4,
- 0x2c, 0xa8, 0xaa, 0x59, 0x51, 0x54
- };
-
- const WebRtc_UWord8 mask46_40[240] =
- {
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0xdb, 0x4a, 0x7b, 0xb6, 0x94, 0xf4,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0x7c, 0xc8, 0x93, 0x63, 0x3c, 0x80
- };
-
- const WebRtc_UWord8 mask46_41[246] =
- {
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0xdb, 0x4a, 0x7b, 0xb6, 0x94, 0xf4,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
- 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
- 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
- 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
- 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
- 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
- 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
- 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
- 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08
- };
-
- const WebRtc_UWord8 mask46_42[252] =
- {
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
- 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
- 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
- 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
- 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
- 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
- 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
- 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
- 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38,
- 0xdb, 0x4a, 0x7b, 0xb6, 0x94, 0xf4,
- 0xfc, 0x6e, 0x89, 0x54, 0x4f, 0x00
- };
-
- const WebRtc_UWord8 mask46_43[258] =
- {
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
- 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
- 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
- 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
- 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
- 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
- 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
- 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
- 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
- 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
- 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
- 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
- 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
- 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
- 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
- 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
- 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0xea, 0x8d, 0x1b, 0xd5, 0x1a, 0x34
- };
-
- const WebRtc_UWord8 mask46_44[264] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
- 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
- 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
- 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
- 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
- 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
- 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
- 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
- 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0xea, 0x8d, 0x1b, 0xd5, 0x1a, 0x34,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
- 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
- 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
- 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
- 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
- 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
- 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
- 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
- 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
- 0x1a, 0x8a, 0x00, 0x1c, 0x89, 0x54
- };
-
- const WebRtc_UWord8 mask46_45[270] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
- 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
- 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
- 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
- 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
- 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
- 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
- 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
- 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0xea, 0x8d, 0x1b, 0xd5, 0x1a, 0x34,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
- 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
- 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
- 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
- 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
- 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
- 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
- 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
- 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
- 0x46, 0x42, 0x0c, 0x8c, 0x84, 0x18,
- 0x33, 0x20, 0x46, 0x66, 0x40, 0x8c,
- 0x99, 0x08, 0x0b, 0x32, 0x10, 0x14,
- 0x05, 0x84, 0x30, 0x0b, 0x08, 0x60,
- 0x80, 0xb0, 0x23, 0x01, 0x60, 0x44,
- 0x84, 0x42, 0x91, 0x08, 0x85, 0x20,
- 0x40, 0x73, 0x00, 0x80, 0xe6, 0x00,
- 0x0a, 0x81, 0x12, 0x15, 0x02, 0x24,
- 0x68, 0x0c, 0x40, 0xd0, 0x18, 0x80,
- 0x10, 0x24, 0x84, 0x20, 0x49, 0x08,
- 0x30, 0x51, 0x40, 0x60, 0xa2, 0x80,
- 0x5f, 0x50, 0x88, 0xbe, 0xa1, 0x10
- };
-
- const WebRtc_UWord8 mask46_46[276] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
- 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
- 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
- 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
- 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
- 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
- 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
- 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
- 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
- 0x46, 0x42, 0x0c, 0x8c, 0x84, 0x18,
- 0x33, 0x20, 0x46, 0x66, 0x40, 0x8c,
- 0x99, 0x08, 0x0b, 0x32, 0x10, 0x14,
- 0x05, 0x84, 0x30, 0x0b, 0x08, 0x60,
- 0x80, 0xb0, 0x23, 0x01, 0x60, 0x44,
- 0x84, 0x42, 0x91, 0x08, 0x85, 0x20,
- 0x40, 0x73, 0x00, 0x80, 0xe6, 0x00,
- 0x0a, 0x81, 0x12, 0x15, 0x02, 0x24,
- 0x68, 0x0c, 0x40, 0xd0, 0x18, 0x80,
- 0x10, 0x24, 0x84, 0x20, 0x49, 0x08,
- 0x30, 0x51, 0x40, 0x60, 0xa2, 0x80,
- 0x5f, 0x50, 0x88, 0xbe, 0xa1, 0x10,
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x99, 0x02, 0x13, 0x32, 0x04, 0x24,
- 0x05, 0x80, 0x0e, 0x0b, 0x00, 0x1c,
- 0x80, 0xa1, 0x83, 0x01, 0x43, 0x04,
- 0x84, 0x48, 0x19, 0x08, 0x90, 0x30,
- 0x40, 0x6d, 0x40, 0x80, 0xda, 0x80,
- 0x0a, 0x90, 0xc0, 0x15, 0x21, 0x80,
- 0x68, 0x04, 0x90, 0xd0, 0x09, 0x20,
- 0x10, 0x31, 0x20, 0x20, 0x62, 0x40,
- 0x30, 0x58, 0x04, 0x60, 0xb0, 0x08,
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x48, 0xa2, 0xa2, 0x91, 0x44,
- 0xa9, 0x10, 0x1b, 0x52, 0x20, 0x34,
- 0x04, 0xc4, 0x84, 0x09, 0x89, 0x08,
- 0xd0, 0x01, 0x45, 0xa0, 0x02, 0x88,
- 0x82, 0x40, 0x1d, 0x04, 0x80, 0x38,
- 0x21, 0x37, 0x00, 0x42, 0x6e, 0x00,
- 0x0c, 0x21, 0x22, 0x18, 0x42, 0x44,
- 0x4a, 0x0a, 0xc0, 0x94, 0x15, 0x80,
- 0x12, 0xb4, 0x50, 0x25, 0x68, 0xa0,
- 0xea, 0x8d, 0x1b, 0xd5, 0x1a, 0x34,
- 0xd5, 0xdf, 0x59, 0xb9, 0xba, 0x10
- };
-
- const WebRtc_UWord8 mask46_5[30] =
- {
- 0xc6, 0xca, 0x6d, 0x8d, 0x94, 0xd8,
- 0x63, 0x6c, 0x96, 0xc6, 0xd9, 0x2c,
- 0x1d, 0xa1, 0xdc, 0x3b, 0x43, 0xb8,
- 0xad, 0x55, 0x39, 0x5a, 0xaa, 0x70,
- 0xb2, 0xb7, 0x07, 0x65, 0x6e, 0x0c
- };
-
- const WebRtc_UWord8 mask46_6[36] =
- {
- 0x64, 0x4a, 0x28, 0xc8, 0x94, 0x50,
- 0x51, 0x58, 0xa2, 0xa2, 0xb1, 0x44,
- 0x0c, 0xa4, 0x30, 0x19, 0x48, 0x60,
- 0xa1, 0x22, 0x47, 0x42, 0x44, 0x8c,
- 0x12, 0xa1, 0x1c, 0x25, 0x42, 0x38,
- 0x8a, 0x45, 0xc1, 0x14, 0x8b, 0x80
- };
-
- const WebRtc_UWord8 mask46_7[42] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x33, 0x24, 0x26, 0x66, 0x48, 0x4c,
- 0x91, 0x92, 0x13, 0x23, 0x24, 0x24,
- 0xa4, 0x20, 0x4b, 0x48, 0x40, 0x94,
- 0x50, 0xa0, 0xd4, 0xa1, 0x41, 0xa8,
- 0x84, 0xc5, 0x81, 0x09, 0x8b, 0x00,
- 0x09, 0x71, 0x0c, 0x12, 0xe2, 0x18
- };
-
- const WebRtc_UWord8 mask46_8[48] =
- {
- 0x0c, 0x84, 0x0c, 0x19, 0x08, 0x18,
- 0x80, 0x70, 0x07, 0x00, 0xe0, 0x0c,
- 0xa0, 0x88, 0x49, 0x41, 0x10, 0x90,
- 0x05, 0x40, 0x32, 0x0a, 0x80, 0x64,
- 0x43, 0x02, 0x82, 0x86, 0x05, 0x04,
- 0x1a, 0x01, 0x50, 0x34, 0x02, 0xa0,
- 0x60, 0x27, 0x00, 0xc0, 0x4e, 0x00,
- 0x14, 0x38, 0xa0, 0x28, 0x71, 0x40
- };
-
- const WebRtc_UWord8 mask46_9[54] =
- {
- 0x46, 0x4a, 0x6c, 0x8c, 0x94, 0xd8,
- 0x62, 0x7c, 0x84, 0xc4, 0xf9, 0x08,
- 0x8c, 0x04, 0x89, 0x18, 0x09, 0x10,
- 0x01, 0x74, 0x22, 0x02, 0xe8, 0x44,
- 0x07, 0x83, 0x06, 0x0f, 0x06, 0x0c,
- 0xa0, 0x80, 0x73, 0x41, 0x00, 0xe4,
- 0x18, 0xb1, 0x42, 0x31, 0x62, 0x84,
- 0x91, 0x00, 0x93, 0x22, 0x01, 0x24,
- 0x78, 0x00, 0x1c, 0xf0, 0x00, 0x38
- };
-
- const WebRtc_UWord8 mask47_1[6] =
- {
- 0xff, 0xff, 0xff, 0xff, 0xff, 0xfe
- };
-
- const WebRtc_UWord8 mask47_10[60] =
- {
- 0x64, 0x4a, 0x28, 0x22, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x68, 0xa6,
- 0xa9, 0x10, 0x1a, 0x00, 0x90, 0x0a,
- 0x04, 0xc4, 0x84, 0x21, 0x06, 0x12,
- 0xd0, 0x01, 0x44, 0x94, 0x29, 0x42,
- 0x82, 0x40, 0x1c, 0x81, 0x48, 0x14,
- 0x21, 0x37, 0x01, 0x40, 0xd4, 0x04,
- 0x0c, 0x21, 0x23, 0x11, 0x01, 0x18,
- 0x4a, 0x0a, 0xc1, 0x0c, 0x10, 0xc0,
- 0x12, 0xb4, 0x50, 0xa8, 0x1a, 0x80
- };
-
- const WebRtc_UWord8 mask47_11[66] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x68,
- 0x33, 0x24, 0x27, 0x40, 0x64, 0x22,
- 0x99, 0x02, 0x12, 0x2a, 0x22, 0x82,
- 0x05, 0x80, 0x0e, 0x06, 0xa0, 0x2a,
- 0x80, 0xa1, 0x83, 0x19, 0x11, 0x90,
- 0x84, 0x48, 0x18, 0x51, 0x05, 0x10,
- 0x40, 0x6d, 0x40, 0x10, 0x91, 0x08,
- 0x0a, 0x90, 0xc1, 0x32, 0x03, 0x20,
- 0x68, 0x04, 0x90, 0x45, 0x24, 0x52,
- 0x10, 0x31, 0x20, 0x8c, 0x08, 0xc0,
- 0x30, 0x58, 0x05, 0x18, 0x58, 0x04
- };
-
- const WebRtc_UWord8 mask47_12[72] =
- {
- 0x64, 0x4a, 0x28, 0x20, 0xc2, 0x0c,
- 0x51, 0x58, 0xa2, 0x04, 0x60, 0x46,
- 0x0c, 0xa4, 0x30, 0x80, 0xa8, 0x0a,
- 0xa1, 0x22, 0x46, 0x43, 0x04, 0x30,
- 0x12, 0xa1, 0x1d, 0x02, 0x30, 0x22,
- 0x8a, 0x45, 0xc0, 0x29, 0x02, 0x90,
- 0x86, 0x8a, 0x6d, 0x30, 0x13, 0x00,
- 0x23, 0x2c, 0x84, 0x11, 0x21, 0x12,
- 0x16, 0x21, 0x98, 0xc4, 0x0c, 0x40,
- 0x4c, 0x30, 0x54, 0x48, 0x44, 0x84,
- 0x41, 0xc1, 0x27, 0x14, 0x11, 0x40,
- 0x19, 0x56, 0xe5, 0x08, 0x90, 0x88
- };
-
- const WebRtc_UWord8 mask47_13[78] =
- {
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
- 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
- 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
- 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
- 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c
- };
-
- const WebRtc_UWord8 mask47_14[84] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
- 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
- 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
- 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
- 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
- 0x9c, 0x3f, 0xb3, 0xe5, 0xad, 0x1c
- };
-
- const WebRtc_UWord8 mask47_15[90] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0
- };
-
- const WebRtc_UWord8 mask47_16[96] =
- {
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0xfa, 0xd9, 0xf5, 0xfe, 0xdc, 0x14
- };
-
- const WebRtc_UWord8 mask47_17[102] =
- {
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c
- };
-
- const WebRtc_UWord8 mask47_18[108] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0x82, 0x32, 0x56, 0x68, 0xa1, 0x5c
- };
-
- const WebRtc_UWord8 mask47_19[114] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50
- };
-
- const WebRtc_UWord8 mask47_2[12] =
- {
- 0xec, 0xdd, 0x99, 0xd9, 0x9d, 0x98,
- 0x9b, 0xb2, 0x77, 0x27, 0x72, 0x76
- };
-
- const WebRtc_UWord8 mask47_20[120] =
- {
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0xdb, 0x4a, 0x7b, 0x31, 0x45, 0x2a
- };
-
- const WebRtc_UWord8 mask47_21[126] =
- {
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
- 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
- 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
- 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
- 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
- 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
- 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
- 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
- 0x30, 0x58, 0x05, 0x80, 0x58, 0x04
- };
-
- const WebRtc_UWord8 mask47_22[132] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
- 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
- 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
- 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
- 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
- 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
- 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
- 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
- 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0xea, 0x8d, 0x1a, 0x35, 0x55, 0xdc
- };
-
- const WebRtc_UWord8 mask47_23[138] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
- 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
- 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
- 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
- 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
- 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
- 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
- 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
- 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
- 0x46, 0x42, 0x0c, 0x20, 0xc2, 0x0c,
- 0x33, 0x20, 0x46, 0x04, 0x60, 0x46,
- 0x99, 0x08, 0x0a, 0x80, 0xa8, 0x0a,
- 0x05, 0x84, 0x30, 0x43, 0x04, 0x30,
- 0x80, 0xb0, 0x23, 0x02, 0x30, 0x22,
- 0x84, 0x42, 0x90, 0x29, 0x02, 0x90,
- 0x40, 0x73, 0x01, 0x30, 0x13, 0x00,
- 0x0a, 0x81, 0x12, 0x11, 0x21, 0x12,
- 0x68, 0x0c, 0x40, 0xc4, 0x0c, 0x40,
- 0x10, 0x24, 0x84, 0x48, 0x44, 0x84,
- 0x30, 0x51, 0x41, 0x14, 0x11, 0x40,
- 0x5f, 0x50, 0x89, 0x08, 0x90, 0x88
- };
-
- const WebRtc_UWord8 mask47_24[144] =
- {
- 0x64, 0x4a, 0x28, 0x20, 0xc2, 0x0c,
- 0x51, 0x58, 0xa2, 0x04, 0x60, 0x46,
- 0x0c, 0xa4, 0x30, 0x80, 0xa8, 0x0a,
- 0xa1, 0x22, 0x46, 0x43, 0x04, 0x30,
- 0x12, 0xa1, 0x1d, 0x02, 0x30, 0x22,
- 0x8a, 0x45, 0xc0, 0x29, 0x02, 0x90,
- 0x86, 0x8a, 0x6d, 0x30, 0x13, 0x00,
- 0x23, 0x2c, 0x84, 0x11, 0x21, 0x12,
- 0x16, 0x21, 0x98, 0xc4, 0x0c, 0x40,
- 0x4c, 0x30, 0x54, 0x48, 0x44, 0x84,
- 0x41, 0xc1, 0x27, 0x14, 0x11, 0x40,
- 0x19, 0x56, 0xe5, 0x08, 0x90, 0x88,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x68,
- 0x33, 0x24, 0x27, 0x40, 0x64, 0x22,
- 0x99, 0x02, 0x12, 0x2a, 0x22, 0x82,
- 0x05, 0x80, 0x0e, 0x06, 0xa0, 0x2a,
- 0x80, 0xa1, 0x83, 0x19, 0x11, 0x90,
- 0x84, 0x48, 0x18, 0x51, 0x05, 0x10,
- 0x40, 0x6d, 0x40, 0x10, 0x91, 0x08,
- 0x0a, 0x90, 0xc1, 0x32, 0x03, 0x20,
- 0x68, 0x04, 0x90, 0x45, 0x24, 0x52,
- 0x10, 0x31, 0x20, 0x8c, 0x08, 0xc0,
- 0x30, 0x58, 0x05, 0x18, 0x58, 0x04,
- 0x27, 0x41, 0x35, 0x57, 0x7e, 0x6a
- };
-
- const WebRtc_UWord8 mask47_25[150] =
- {
- 0x64, 0x4a, 0x28, 0x20, 0xc2, 0x0c,
- 0x51, 0x58, 0xa2, 0x04, 0x60, 0x46,
- 0x0c, 0xa4, 0x30, 0x80, 0xa8, 0x0a,
- 0xa1, 0x22, 0x46, 0x43, 0x04, 0x30,
- 0x12, 0xa1, 0x1d, 0x02, 0x30, 0x22,
- 0x8a, 0x45, 0xc0, 0x29, 0x02, 0x90,
- 0x86, 0x8a, 0x6d, 0x30, 0x13, 0x00,
- 0x23, 0x2c, 0x84, 0x11, 0x21, 0x12,
- 0x16, 0x21, 0x98, 0xc4, 0x0c, 0x40,
- 0x4c, 0x30, 0x54, 0x48, 0x44, 0x84,
- 0x41, 0xc1, 0x27, 0x14, 0x11, 0x40,
- 0x19, 0x56, 0xe5, 0x08, 0x90, 0x88,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
- 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
- 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
- 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
- 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c
- };
-
- const WebRtc_UWord8 mask47_26[156] =
- {
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
- 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
- 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
- 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
- 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0x64, 0x4a, 0x28, 0x20, 0xc2, 0x0c,
- 0x51, 0x58, 0xa2, 0x04, 0x60, 0x46,
- 0x0c, 0xa4, 0x30, 0x80, 0xa8, 0x0a,
- 0xa1, 0x22, 0x46, 0x43, 0x04, 0x30,
- 0x12, 0xa1, 0x1d, 0x02, 0x30, 0x22,
- 0x8a, 0x45, 0xc0, 0x29, 0x02, 0x90,
- 0x86, 0x8a, 0x6d, 0x30, 0x13, 0x00,
- 0x23, 0x2c, 0x84, 0x11, 0x21, 0x12,
- 0x16, 0x21, 0x98, 0xc4, 0x0c, 0x40,
- 0x4c, 0x30, 0x54, 0x48, 0x44, 0x84,
- 0x41, 0xc1, 0x27, 0x14, 0x11, 0x40,
- 0x19, 0x56, 0xe5, 0x08, 0x90, 0x88,
- 0x6c, 0xea, 0xc4, 0x42, 0x20, 0x9e
- };
-
- const WebRtc_UWord8 mask47_27[162] =
- {
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
- 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
- 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
- 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
- 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
- 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
- 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
- 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
- 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
- 0x9c, 0x3f, 0xb3, 0xe5, 0xad, 0x1c
- };
-
- const WebRtc_UWord8 mask47_28[168] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
- 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
- 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
- 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
- 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
- 0x9c, 0x3f, 0xb3, 0xe5, 0xad, 0x1c,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
- 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
- 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
- 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
- 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0x86, 0x1e, 0xa6, 0xaf, 0x3d, 0x04
- };
-
- const WebRtc_UWord8 mask47_29[174] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
- 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
- 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
- 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
- 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
- 0x9c, 0x3f, 0xb3, 0xe5, 0xad, 0x1c,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0
- };
-
- const WebRtc_UWord8 mask47_3[18] =
- {
- 0xca, 0xd3, 0x65, 0x36, 0x53, 0x64,
- 0xf1, 0x49, 0x3a, 0x93, 0xa9, 0x3a,
- 0x76, 0x27, 0xd0, 0x7d, 0x07, 0xd0
- };
-
- const WebRtc_UWord8 mask47_30[180] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x58, 0xa3, 0x8a, 0x38, 0xa2,
- 0x0c, 0xa4, 0x30, 0x43, 0x04, 0x30,
- 0xa1, 0x22, 0x46, 0x24, 0x62, 0x46,
- 0x12, 0xa1, 0x1c, 0x11, 0xc1, 0x1c,
- 0x8a, 0x45, 0xc0, 0x5c, 0x05, 0xc0,
- 0x9c, 0x3f, 0xb3, 0xe5, 0xad, 0x1c,
- 0x97, 0x43, 0x63, 0xc6, 0x09, 0x9c
- };
-
- const WebRtc_UWord8 mask47_31[186] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0xfa, 0xd9, 0xf5, 0xfe, 0xdc, 0x14
- };
-
- const WebRtc_UWord8 mask47_32[192] =
- {
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0xfa, 0xd9, 0xf5, 0xfe, 0xdc, 0x14,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0xe5, 0x50, 0x45, 0x63, 0xc2, 0xf4
- };
-
- const WebRtc_UWord8 mask47_33[198] =
- {
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0xfa, 0xd9, 0xf5, 0xfe, 0xdc, 0x14,
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c
- };
-
- const WebRtc_UWord8 mask47_34[204] =
- {
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x91, 0x92, 0x13, 0x21, 0x32, 0x12,
- 0xa4, 0x20, 0x4a, 0x04, 0xa0, 0x4a,
- 0x50, 0xa0, 0xd4, 0x0d, 0x40, 0xd4,
- 0x84, 0xc5, 0x80, 0x58, 0x05, 0x80,
- 0x09, 0x71, 0x0d, 0x10, 0xd1, 0x0c,
- 0xfa, 0xd9, 0xf5, 0xfe, 0xdc, 0x14,
- 0xef, 0xbb, 0xa6, 0x23, 0x5c, 0xbe
- };
-
- const WebRtc_UWord8 mask47_35[210] =
- {
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0x82, 0x32, 0x56, 0x68, 0xa1, 0x5c
- };
-
- const WebRtc_UWord8 mask47_36[216] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0x82, 0x32, 0x56, 0x68, 0xa1, 0x5c,
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0x0e, 0xd7, 0x38, 0x20, 0x87, 0x66
- };
-
- const WebRtc_UWord8 mask47_37[222] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0x82, 0x32, 0x56, 0x68, 0xa1, 0x5c,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50
- };
-
- const WebRtc_UWord8 mask47_38[228] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0x0c, 0x84, 0x0c, 0x40, 0xc4, 0x0c,
- 0x80, 0x70, 0x07, 0x00, 0x70, 0x06,
- 0xa0, 0x88, 0x48, 0x84, 0x88, 0x48,
- 0x05, 0x40, 0x32, 0x03, 0x20, 0x32,
- 0x43, 0x02, 0x82, 0x28, 0x22, 0x82,
- 0x1a, 0x01, 0x50, 0x15, 0x01, 0x50,
- 0x60, 0x27, 0x00, 0x70, 0x07, 0x00,
- 0x14, 0x38, 0xa1, 0x8a, 0x18, 0xa0,
- 0x82, 0x32, 0x56, 0x68, 0xa1, 0x5c,
- 0x7b, 0x47, 0xa5, 0xde, 0x9a, 0xd4
- };
-
- const WebRtc_UWord8 mask47_39[234] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0xdb, 0x4a, 0x7b, 0x31, 0x45, 0x2a
- };
-
- const WebRtc_UWord8 mask47_4[24] =
- {
- 0xc4, 0xd1, 0x65, 0x16, 0x51, 0x64,
- 0x31, 0x62, 0x96, 0x29, 0x62, 0x96,
- 0x4b, 0x24, 0x5a, 0x45, 0xa4, 0x5a,
- 0x2c, 0xa8, 0xaa, 0x8a, 0xa8, 0xaa
- };
-
- const WebRtc_UWord8 mask47_40[240] =
- {
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0xdb, 0x4a, 0x7b, 0x31, 0x45, 0x2a,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0xc4, 0xae, 0x5e, 0x33, 0xf5, 0x1a
- };
-
- const WebRtc_UWord8 mask47_41[246] =
- {
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0xdb, 0x4a, 0x7b, 0x31, 0x45, 0x2a,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
- 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
- 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
- 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
- 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
- 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
- 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
- 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
- 0x30, 0x58, 0x05, 0x80, 0x58, 0x04
- };
-
- const WebRtc_UWord8 mask47_42[252] =
- {
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
- 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
- 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
- 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
- 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
- 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
- 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
- 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
- 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x85, 0xc8, 0x5c, 0x84,
- 0x8c, 0x04, 0x88, 0x48, 0x84, 0x88,
- 0x01, 0x74, 0x23, 0x42, 0x34, 0x22,
- 0x07, 0x83, 0x06, 0x30, 0x63, 0x06,
- 0xa0, 0x80, 0x72, 0x07, 0x20, 0x72,
- 0x18, 0xb1, 0x43, 0x14, 0x31, 0x42,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x01, 0xc0, 0x1c,
- 0xdb, 0x4a, 0x7b, 0x31, 0x45, 0x2a,
- 0x3c, 0xb0, 0x36, 0x3b, 0x14, 0xa2
- };
-
- const WebRtc_UWord8 mask47_43[258] =
- {
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
- 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
- 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
- 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
- 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
- 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
- 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
- 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
- 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
- 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
- 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
- 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
- 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
- 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
- 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
- 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
- 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0xea, 0x8d, 0x1a, 0x35, 0x55, 0xdc
- };
-
- const WebRtc_UWord8 mask47_44[264] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
- 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
- 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
- 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
- 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
- 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
- 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
- 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
- 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0xea, 0x8d, 0x1a, 0x35, 0x55, 0xdc,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
- 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
- 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
- 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
- 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
- 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
- 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
- 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
- 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
- 0xd4, 0x8a, 0xd4, 0xd3, 0x3f, 0xe6
- };
-
- const WebRtc_UWord8 mask47_45[270] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
- 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
- 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
- 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
- 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
- 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
- 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
- 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
- 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0xea, 0x8d, 0x1a, 0x35, 0x55, 0xdc,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
- 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
- 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
- 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
- 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
- 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
- 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
- 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
- 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
- 0x46, 0x42, 0x0c, 0x20, 0xc2, 0x0c,
- 0x33, 0x20, 0x46, 0x04, 0x60, 0x46,
- 0x99, 0x08, 0x0a, 0x80, 0xa8, 0x0a,
- 0x05, 0x84, 0x30, 0x43, 0x04, 0x30,
- 0x80, 0xb0, 0x23, 0x02, 0x30, 0x22,
- 0x84, 0x42, 0x90, 0x29, 0x02, 0x90,
- 0x40, 0x73, 0x01, 0x30, 0x13, 0x00,
- 0x0a, 0x81, 0x12, 0x11, 0x21, 0x12,
- 0x68, 0x0c, 0x40, 0xc4, 0x0c, 0x40,
- 0x10, 0x24, 0x84, 0x48, 0x44, 0x84,
- 0x30, 0x51, 0x41, 0x14, 0x11, 0x40,
- 0x5f, 0x50, 0x89, 0x08, 0x90, 0x88
- };
-
- const WebRtc_UWord8 mask47_46[276] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
- 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
- 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
- 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
- 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
- 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
- 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
- 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
- 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
- 0x46, 0x42, 0x0c, 0x20, 0xc2, 0x0c,
- 0x33, 0x20, 0x46, 0x04, 0x60, 0x46,
- 0x99, 0x08, 0x0a, 0x80, 0xa8, 0x0a,
- 0x05, 0x84, 0x30, 0x43, 0x04, 0x30,
- 0x80, 0xb0, 0x23, 0x02, 0x30, 0x22,
- 0x84, 0x42, 0x90, 0x29, 0x02, 0x90,
- 0x40, 0x73, 0x01, 0x30, 0x13, 0x00,
- 0x0a, 0x81, 0x12, 0x11, 0x21, 0x12,
- 0x68, 0x0c, 0x40, 0xc4, 0x0c, 0x40,
- 0x10, 0x24, 0x84, 0x48, 0x44, 0x84,
- 0x30, 0x51, 0x41, 0x14, 0x11, 0x40,
- 0x5f, 0x50, 0x89, 0x08, 0x90, 0x88,
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
- 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
- 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
- 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
- 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
- 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
- 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
- 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
- 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
- 0x64, 0x4a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x48, 0xa2, 0x8a, 0x28, 0xa2,
- 0xa9, 0x10, 0x1b, 0x01, 0xb0, 0x1a,
- 0x04, 0xc4, 0x84, 0x48, 0x44, 0x84,
- 0xd0, 0x01, 0x44, 0x14, 0x41, 0x44,
- 0x82, 0x40, 0x1c, 0x01, 0xc0, 0x1c,
- 0x21, 0x37, 0x01, 0x70, 0x17, 0x00,
- 0x0c, 0x21, 0x22, 0x12, 0x21, 0x22,
- 0x4a, 0x0a, 0xc0, 0xac, 0x0a, 0xc0,
- 0x12, 0xb4, 0x51, 0x45, 0x14, 0x50,
- 0xea, 0x8d, 0x1a, 0x35, 0x55, 0xdc,
- 0x37, 0x9d, 0xcf, 0xe0, 0xe4, 0x20
- };
-
- const WebRtc_UWord8 mask47_47[282] =
- {
- 0x46, 0x4a, 0x6c, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x42, 0x64, 0x26,
- 0x99, 0x02, 0x12, 0x21, 0x22, 0x12,
- 0x05, 0x80, 0x0e, 0x00, 0xe0, 0x0e,
- 0x80, 0xa1, 0x82, 0x18, 0x21, 0x82,
- 0x84, 0x48, 0x18, 0x81, 0x88, 0x18,
- 0x40, 0x6d, 0x40, 0xd4, 0x0d, 0x40,
- 0x0a, 0x90, 0xc1, 0x0c, 0x10, 0xc0,
- 0x68, 0x04, 0x90, 0x49, 0x04, 0x90,
- 0x10, 0x31, 0x21, 0x12, 0x11, 0x20,
- 0x30, 0x58, 0x05, 0x80, 0x58, 0x04,
- 0x46, 0x42, 0x0c, 0x20, 0xc2, 0x0c,
- 0x33, 0x20, 0x46, 0x04, 0x60, 0x46,
- 0x99, 0x08, 0x0a, 0x80, 0xa8, 0x0a,
- 0x05, 0x84, 0x30, 0x43, 0x04, 0x30,
- 0x80, 0xb0, 0x23, 0x02, 0x30, 0x22,
- 0x84, 0x42, 0x90, 0x29, 0x02, 0x90,
- 0x40, 0x73, 0x01, 0x30, 0x13, 0x00,
- 0x0a, 0x81, 0x12, 0x11, 0x21, 0x12,
- 0x68, 0x0c, 0x40, 0xc4, 0x0c, 0x40,
- 0x10, 0x24, 0x84, 0x48, 0x44, 0x84,
- 0x30, 0x51, 0x41, 0x14, 0x11, 0x40,
- 0x5f, 0x50, 0x89, 0x08, 0x90, 0x88,
- 0x46, 0x4a, 0x6c, 0x20, 0xc2, 0x0c,
- 0x33, 0x24, 0x26, 0x04, 0x60, 0x46,
- 0x99, 0x02, 0x12, 0x80, 0xa8, 0x0a,
- 0x05, 0x80, 0x0e, 0x43, 0x04, 0x30,
- 0x80, 0xa1, 0x83, 0x02, 0x30, 0x22,
- 0x84, 0x48, 0x18, 0x29, 0x02, 0x90,
- 0x40, 0x6d, 0x41, 0x30, 0x13, 0x00,
- 0x0a, 0x90, 0xc0, 0x11, 0x21, 0x12,
- 0x68, 0x04, 0x90, 0xc4, 0x0c, 0x40,
- 0x10, 0x31, 0x20, 0x48, 0x44, 0x84,
- 0x30, 0x58, 0x05, 0x14, 0x11, 0x40,
- 0x46, 0x42, 0x0d, 0x08, 0x90, 0x88,
- 0x33, 0x20, 0x46, 0xa6, 0xca, 0x6c,
- 0x99, 0x08, 0x0a, 0x42, 0x64, 0x26,
- 0x05, 0x84, 0x30, 0x21, 0x22, 0x12,
- 0x80, 0xb0, 0x22, 0x00, 0xe0, 0x0e,
- 0x84, 0x42, 0x90, 0x18, 0x21, 0x82,
- 0x40, 0x73, 0x00, 0x81, 0x88, 0x18,
- 0x0a, 0x81, 0x12, 0xd4, 0x0d, 0x40,
- 0x68, 0x0c, 0x41, 0x0c, 0x10, 0xc0,
- 0x10, 0x24, 0x84, 0x49, 0x04, 0x90,
- 0x30, 0x51, 0x41, 0x12, 0x11, 0x20,
- 0x5f, 0x50, 0x89, 0x80, 0x58, 0x04,
- 0x1f, 0x2f, 0x63, 0x10, 0x64, 0xb2
- };
-
- const WebRtc_UWord8 mask47_5[30] =
- {
- 0xc6, 0xca, 0x6c, 0xa6, 0xca, 0x6c,
- 0x63, 0x6c, 0x96, 0xc9, 0x6c, 0x96,
- 0x1d, 0xa1, 0xdc, 0x1d, 0xc1, 0xdc,
- 0xad, 0x55, 0x39, 0x53, 0x95, 0x38,
- 0xb2, 0xb7, 0x07, 0x70, 0x77, 0x06
- };
-
- const WebRtc_UWord8 mask47_6[36] =
- {
- 0x64, 0x4a, 0x29, 0xa2, 0x9a, 0x28,
- 0x51, 0x58, 0xa2, 0x8a, 0x68, 0xa6,
- 0x0c, 0xa4, 0x30, 0x45, 0xa4, 0x5a,
- 0xa1, 0x22, 0x46, 0x2d, 0x82, 0xd8,
- 0x12, 0xa1, 0x1c, 0x17, 0x41, 0x74,
- 0x8a, 0x45, 0xc1, 0xd1, 0x1d, 0x10
- };
-
- const WebRtc_UWord8 mask47_7[42] =
- {
- 0x46, 0x4a, 0x6d, 0xa6, 0xca, 0x6c,
- 0x33, 0x24, 0x26, 0x4a, 0x64, 0xa6,
- 0x91, 0x92, 0x12, 0x61, 0xa6, 0x0a,
- 0xa4, 0x20, 0x4a, 0x0c, 0x90, 0xd8,
- 0x50, 0xa0, 0xd5, 0x81, 0x70, 0x36,
- 0x84, 0xc5, 0x80, 0x55, 0x45, 0x54,
- 0x09, 0x71, 0x0d, 0x50, 0x9d, 0x08
- };
-
- const WebRtc_UWord8 mask47_8[48] =
- {
- 0x0c, 0x84, 0x0d, 0x02, 0xc0, 0x2c,
- 0x80, 0x70, 0x06, 0x80, 0x78, 0x06,
- 0xa0, 0x88, 0x48, 0x21, 0x22, 0x12,
- 0x05, 0x40, 0x32, 0x0c, 0xa0, 0xca,
- 0x43, 0x02, 0x82, 0x40, 0x95, 0x08,
- 0x1a, 0x01, 0x51, 0x15, 0x41, 0x54,
- 0x60, 0x27, 0x00, 0x66, 0x06, 0x60,
- 0x14, 0x38, 0xa0, 0x99, 0x09, 0x90
- };
-
- const WebRtc_UWord8 mask47_9[54] =
- {
- 0x46, 0x4a, 0x6d, 0xa6, 0xca, 0x6c,
- 0x62, 0x7c, 0x84, 0xc8, 0x4c, 0x84,
- 0x8c, 0x04, 0x88, 0x30, 0x83, 0x88,
- 0x01, 0x74, 0x23, 0x40, 0x94, 0x08,
- 0x07, 0x83, 0x07, 0x02, 0x70, 0x26,
- 0xa0, 0x80, 0x72, 0x45, 0x44, 0x54,
- 0x18, 0xb1, 0x42, 0x10, 0xe1, 0x0e,
- 0x91, 0x00, 0x92, 0x09, 0x20, 0x92,
- 0x78, 0x00, 0x1c, 0x03, 0x80, 0x38
- };
-
- const WebRtc_UWord8 mask48_1[6] =
- {
- 0xff, 0xff, 0xff, 0xff, 0xff, 0xff
- };
-
- const WebRtc_UWord8 mask48_10[60] =
- {
- 0x11, 0x45, 0x14, 0x11, 0x45, 0x14,
- 0x45, 0x34, 0x53, 0x45, 0x34, 0x53,
- 0x00, 0x48, 0x05, 0x00, 0x48, 0x05,
- 0x10, 0x83, 0x09, 0x10, 0x83, 0x09,
- 0x4a, 0x14, 0xa1, 0x4a, 0x14, 0xa1,
- 0x40, 0xa4, 0x0a, 0x40, 0xa4, 0x0a,
- 0xa0, 0x6a, 0x02, 0xa0, 0x6a, 0x02,
- 0x88, 0x80, 0x8c, 0x88, 0x80, 0x8c,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x54, 0x0d, 0x40, 0x54, 0x0d, 0x40
- };
-
- const WebRtc_UWord8 mask48_11[66] =
- {
- 0x53, 0x65, 0x34, 0x53, 0x65, 0x34,
- 0xa0, 0x32, 0x11, 0xa0, 0x32, 0x11,
- 0x15, 0x11, 0x41, 0x15, 0x11, 0x41,
- 0x03, 0x50, 0x15, 0x03, 0x50, 0x15,
- 0x8c, 0x88, 0xc8, 0x8c, 0x88, 0xc8,
- 0x28, 0x82, 0x88, 0x28, 0x82, 0x88,
- 0x08, 0x48, 0x84, 0x08, 0x48, 0x84,
- 0x99, 0x01, 0x90, 0x99, 0x01, 0x90,
- 0x22, 0x92, 0x29, 0x22, 0x92, 0x29,
- 0x46, 0x04, 0x60, 0x46, 0x04, 0x60,
- 0x8c, 0x2c, 0x02, 0x8c, 0x2c, 0x02
- };
-
- const WebRtc_UWord8 mask48_12[72] =
- {
- 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
- 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
- 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
- 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
- 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
- 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
- 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
- 0x84, 0x48, 0x44, 0x84, 0x48, 0x44
- };
-
- const WebRtc_UWord8 mask48_13[78] =
- {
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
- 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
- 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86
- };
-
- const WebRtc_UWord8 mask48_14[84] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
- 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
- 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
- 0xf2, 0xd6, 0x8e, 0xf2, 0xd6, 0x8e
- };
-
- const WebRtc_UWord8 mask48_15[90] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50
- };
-
- const WebRtc_UWord8 mask48_16[96] =
- {
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0xff, 0x6e, 0x0a, 0xff, 0x6e, 0x0a
- };
-
- const WebRtc_UWord8 mask48_17[102] =
- {
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e
- };
-
- const WebRtc_UWord8 mask48_18[108] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0x34, 0x50, 0xae, 0x34, 0x50, 0xae
- };
-
- const WebRtc_UWord8 mask48_19[114] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28
- };
-
- const WebRtc_UWord8 mask48_2[12] =
- {
- 0xec, 0xce, 0xcc, 0xec, 0xce, 0xcc,
- 0x93, 0xb9, 0x3b, 0x93, 0xb9, 0x3b
- };
-
- const WebRtc_UWord8 mask48_20[120] =
- {
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x98, 0xa2, 0x95, 0x98, 0xa2, 0x95
- };
-
- const WebRtc_UWord8 mask48_21[126] =
- {
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02
- };
-
- const WebRtc_UWord8 mask48_22[132] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x1a, 0xaa, 0xee, 0x1a, 0xaa, 0xee
- };
-
- const WebRtc_UWord8 mask48_23[138] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
- 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
- 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
- 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
- 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
- 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
- 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
- 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
- 0x84, 0x48, 0x44, 0x84, 0x48, 0x44
- };
-
- const WebRtc_UWord8 mask48_24[144] =
- {
- 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
- 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
- 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
- 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
- 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
- 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
- 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
- 0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
- 0x88, 0x32, 0x59, 0x88, 0x32, 0x59
- };
-
- const WebRtc_UWord8 mask48_25[150] =
- {
- 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
- 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
- 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
- 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
- 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
- 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
- 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
- 0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
- 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
- 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86
- };
-
- const WebRtc_UWord8 mask48_26[156] =
- {
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
- 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
- 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
- 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
- 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
- 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
- 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
- 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
- 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
- 0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
- 0x3e, 0x20, 0x79, 0xe5, 0x55, 0x70
- };
-
- const WebRtc_UWord8 mask48_27[162] =
- {
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
- 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
- 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
- 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
- 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
- 0xf2, 0xd6, 0x8e, 0xf2, 0xd6, 0x8e
- };
-
- const WebRtc_UWord8 mask48_28[168] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
- 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
- 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
- 0xf2, 0xd6, 0x8e, 0xf2, 0xd6, 0x8e,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
- 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
- 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0x32, 0xe3, 0xc0, 0x4a, 0xf2, 0x2a
- };
-
- const WebRtc_UWord8 mask48_29[174] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
- 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
- 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
- 0xf2, 0xd6, 0x8e, 0xf2, 0xd6, 0x8e,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50
- };
-
- const WebRtc_UWord8 mask48_3[18] =
- {
- 0x9b, 0x29, 0xb2, 0x9b, 0x29, 0xb2,
- 0x49, 0xd4, 0x9d, 0x49, 0xd4, 0x9d,
- 0x3e, 0x83, 0xe8, 0x3e, 0x83, 0xe8
- };
-
- const WebRtc_UWord8 mask48_30[180] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0xc5, 0x1c, 0x51, 0xc5, 0x1c, 0x51,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x12, 0x31, 0x23, 0x12, 0x31, 0x23,
- 0x08, 0xe0, 0x8e, 0x08, 0xe0, 0x8e,
- 0x2e, 0x02, 0xe0, 0x2e, 0x02, 0xe0,
- 0xf2, 0xd6, 0x8e, 0xf2, 0xd6, 0x8e,
- 0x66, 0xf3, 0x9a, 0xdd, 0x68, 0x93
- };
-
- const WebRtc_UWord8 mask48_31[186] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0xff, 0x6e, 0x0a, 0xff, 0x6e, 0x0a
- };
-
- const WebRtc_UWord8 mask48_32[192] =
- {
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0xff, 0x6e, 0x0a, 0xff, 0x6e, 0x0a,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0xd5, 0x4a, 0x4f, 0x48, 0xb5, 0x31
- };
-
- const WebRtc_UWord8 mask48_33[198] =
- {
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0xff, 0x6e, 0x0a, 0xff, 0x6e, 0x0a,
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e
- };
-
- const WebRtc_UWord8 mask48_34[204] =
- {
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x90, 0x99, 0x09, 0x90, 0x99, 0x09,
- 0x02, 0x50, 0x25, 0x02, 0x50, 0x25,
- 0x06, 0xa0, 0x6a, 0x06, 0xa0, 0x6a,
- 0x2c, 0x02, 0xc0, 0x2c, 0x02, 0xc0,
- 0x88, 0x68, 0x86, 0x88, 0x68, 0x86,
- 0xff, 0x6e, 0x0a, 0xff, 0x6e, 0x0a,
- 0x40, 0x72, 0x4c, 0xe8, 0xf2, 0x42
- };
-
- const WebRtc_UWord8 mask48_35[210] =
- {
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0x34, 0x50, 0xae, 0x34, 0x50, 0xae
- };
-
- const WebRtc_UWord8 mask48_36[216] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0x34, 0x50, 0xae, 0x34, 0x50, 0xae,
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x71, 0xba, 0x8b, 0xf3, 0xfa, 0x9d
- };
-
- const WebRtc_UWord8 mask48_37[222] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0x34, 0x50, 0xae, 0x34, 0x50, 0xae,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28
- };
-
- const WebRtc_UWord8 mask48_38[228] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x20, 0x62, 0x06, 0x20, 0x62, 0x06,
- 0x80, 0x38, 0x03, 0x80, 0x38, 0x03,
- 0x42, 0x44, 0x24, 0x42, 0x44, 0x24,
- 0x01, 0x90, 0x19, 0x01, 0x90, 0x19,
- 0x14, 0x11, 0x41, 0x14, 0x11, 0x41,
- 0x0a, 0x80, 0xa8, 0x0a, 0x80, 0xa8,
- 0x38, 0x03, 0x80, 0x38, 0x03, 0x80,
- 0xc5, 0x0c, 0x50, 0xc5, 0x0c, 0x50,
- 0x34, 0x50, 0xae, 0x34, 0x50, 0xae,
- 0x2a, 0x7a, 0xf6, 0x8c, 0xde, 0x51
- };
-
- const WebRtc_UWord8 mask48_39[234] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x98, 0xa2, 0x95, 0x98, 0xa2, 0x95
- };
-
- const WebRtc_UWord8 mask48_4[24] =
- {
- 0x8b, 0x28, 0xb2, 0x8b, 0x28, 0xb2,
- 0x14, 0xb1, 0x4b, 0x14, 0xb1, 0x4b,
- 0x22, 0xd2, 0x2d, 0x22, 0xd2, 0x2d,
- 0x45, 0x54, 0x55, 0x45, 0x54, 0x55
- };
-
- const WebRtc_UWord8 mask48_40[240] =
- {
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x98, 0xa2, 0x95, 0x98, 0xa2, 0x95,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x20, 0x5f, 0x68, 0xd5, 0xa2, 0x1b
- };
-
- const WebRtc_UWord8 mask48_41[246] =
- {
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x98, 0xa2, 0x95, 0x98, 0xa2, 0x95,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02
- };
-
- const WebRtc_UWord8 mask48_42[252] =
- {
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0xe4, 0x2e, 0x42, 0xe4, 0x2e, 0x42,
- 0x24, 0x42, 0x44, 0x24, 0x42, 0x44,
- 0xa1, 0x1a, 0x11, 0xa1, 0x1a, 0x11,
- 0x18, 0x31, 0x83, 0x18, 0x31, 0x83,
- 0x03, 0x90, 0x39, 0x03, 0x90, 0x39,
- 0x8a, 0x18, 0xa1, 0x8a, 0x18, 0xa1,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0x98, 0xa2, 0x95, 0x98, 0xa2, 0x95,
- 0x66, 0xcf, 0xa3, 0x47, 0x69, 0x00
- };
-
- const WebRtc_UWord8 mask48_43[258] =
- {
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x1a, 0xaa, 0xee, 0x1a, 0xaa, 0xee
- };
-
- const WebRtc_UWord8 mask48_44[264] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x1a, 0xaa, 0xee, 0x1a, 0xaa, 0xee,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
- 0xc6, 0x40, 0x1f, 0x57, 0xc6, 0xe6
- };
-
- const WebRtc_UWord8 mask48_45[270] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x1a, 0xaa, 0xee, 0x1a, 0xaa, 0xee,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
- 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
- 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
- 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
- 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
- 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
- 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
- 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
- 0x84, 0x48, 0x44, 0x84, 0x48, 0x44
- };
-
- const WebRtc_UWord8 mask48_46[276] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
- 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
- 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
- 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
- 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
- 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
- 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
- 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
- 0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
- 0x51, 0x45, 0x14, 0x51, 0x45, 0x14,
- 0x45, 0x14, 0x51, 0x45, 0x14, 0x51,
- 0x80, 0xd8, 0x0d, 0x80, 0xd8, 0x0d,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x0a, 0x20, 0xa2, 0x0a, 0x20, 0xa2,
- 0x00, 0xe0, 0x0e, 0x00, 0xe0, 0x0e,
- 0xb8, 0x0b, 0x80, 0xb8, 0x0b, 0x80,
- 0x09, 0x10, 0x91, 0x09, 0x10, 0x91,
- 0x56, 0x05, 0x60, 0x56, 0x05, 0x60,
- 0xa2, 0x8a, 0x28, 0xa2, 0x8a, 0x28,
- 0x1a, 0xaa, 0xee, 0x1a, 0xaa, 0xee,
- 0x10, 0xf9, 0xab, 0x12, 0x14, 0xef
- };
-
- const WebRtc_UWord8 mask48_47[282] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
- 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
- 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
- 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
- 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
- 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
- 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
- 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
- 0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
- 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
- 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
- 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
- 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
- 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
- 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
- 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
- 0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
- 0x88, 0x32, 0x59, 0x88, 0x32, 0x59
- };
-
- const WebRtc_UWord8 mask48_48[288] =
- {
- 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
- 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
- 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
- 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
- 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
- 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
- 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
- 0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
- 0x88, 0x32, 0x59, 0x88, 0x32, 0x59,
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x21, 0x32, 0x13, 0x21, 0x32, 0x13,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x00, 0x70, 0x07, 0x00, 0x70, 0x07,
- 0x0c, 0x10, 0xc1, 0x0c, 0x10, 0xc1,
- 0x40, 0xc4, 0x0c, 0x40, 0xc4, 0x0c,
- 0x6a, 0x06, 0xa0, 0x6a, 0x06, 0xa0,
- 0x86, 0x08, 0x60, 0x86, 0x08, 0x60,
- 0x24, 0x82, 0x48, 0x24, 0x82, 0x48,
- 0x89, 0x08, 0x90, 0x89, 0x08, 0x90,
- 0xc0, 0x2c, 0x02, 0xc0, 0x2c, 0x02,
- 0x10, 0x61, 0x06, 0x10, 0x61, 0x06,
- 0x02, 0x30, 0x23, 0x02, 0x30, 0x23,
- 0x40, 0x54, 0x05, 0x40, 0x54, 0x05,
- 0x21, 0x82, 0x18, 0x21, 0x82, 0x18,
- 0x81, 0x18, 0x11, 0x81, 0x18, 0x11,
- 0x14, 0x81, 0x48, 0x14, 0x81, 0x48,
- 0x98, 0x09, 0x80, 0x98, 0x09, 0x80,
- 0x08, 0x90, 0x89, 0x08, 0x90, 0x89,
- 0x62, 0x06, 0x20, 0x62, 0x06, 0x20,
- 0x24, 0x22, 0x42, 0x24, 0x22, 0x42,
- 0x8a, 0x08, 0xa0, 0x8a, 0x08, 0xa0,
- 0x84, 0x48, 0x44, 0x84, 0x48, 0x44,
- 0xff, 0x9b, 0xdf, 0xec, 0xae, 0x0e
- };
-
- const WebRtc_UWord8 mask48_5[30] =
- {
- 0x53, 0x65, 0x36, 0x53, 0x65, 0x36,
- 0x64, 0xb6, 0x4b, 0x64, 0xb6, 0x4b,
- 0x0e, 0xe0, 0xee, 0x0e, 0xe0, 0xee,
- 0xa9, 0xca, 0x9c, 0xa9, 0xca, 0x9c,
- 0xb8, 0x3b, 0x83, 0xb8, 0x3b, 0x83
- };
-
- const WebRtc_UWord8 mask48_6[36] =
- {
- 0xd1, 0x4d, 0x14, 0xd1, 0x4d, 0x14,
- 0x45, 0x34, 0x53, 0x45, 0x34, 0x53,
- 0x22, 0xd2, 0x2d, 0x22, 0xd2, 0x2d,
- 0x16, 0xc1, 0x6c, 0x16, 0xc1, 0x6c,
- 0x0b, 0xa0, 0xba, 0x0b, 0xa0, 0xba,
- 0xe8, 0x8e, 0x88, 0xe8, 0x8e, 0x88
- };
-
- const WebRtc_UWord8 mask48_7[42] =
- {
- 0xd3, 0x65, 0x36, 0xd3, 0x65, 0x36,
- 0x25, 0x32, 0x53, 0x25, 0x32, 0x53,
- 0x30, 0xd3, 0x05, 0x30, 0xd3, 0x05,
- 0x06, 0x48, 0x6c, 0x06, 0x48, 0x6c,
- 0xc0, 0xb8, 0x1b, 0xc0, 0xb8, 0x1b,
- 0x2a, 0xa2, 0xaa, 0x2a, 0xa2, 0xaa,
- 0xa8, 0x4e, 0x84, 0xa8, 0x4e, 0x84
- };
-
- const WebRtc_UWord8 mask48_8[48] =
- {
- 0x81, 0x60, 0x16, 0x81, 0x60, 0x16,
- 0x40, 0x3c, 0x03, 0x40, 0x3c, 0x03,
- 0x10, 0x91, 0x09, 0x10, 0x91, 0x09,
- 0x06, 0x50, 0x65, 0x06, 0x50, 0x65,
- 0x20, 0x4a, 0x84, 0x20, 0x4a, 0x84,
- 0x8a, 0xa0, 0xaa, 0x8a, 0xa0, 0xaa,
- 0x33, 0x03, 0x30, 0x33, 0x03, 0x30,
- 0x4c, 0x84, 0xc8, 0x4c, 0x84, 0xc8
- };
-
- const WebRtc_UWord8 mask48_9[54] =
- {
- 0xd3, 0x65, 0x36, 0xd3, 0x65, 0x36,
- 0x64, 0x26, 0x42, 0x64, 0x26, 0x42,
- 0x18, 0x41, 0xc4, 0x18, 0x41, 0xc4,
- 0xa0, 0x4a, 0x04, 0xa0, 0x4a, 0x04,
- 0x81, 0x38, 0x13, 0x81, 0x38, 0x13,
- 0x22, 0xa2, 0x2a, 0x22, 0xa2, 0x2a,
- 0x08, 0x70, 0x87, 0x08, 0x70, 0x87,
- 0x04, 0x90, 0x49, 0x04, 0x90, 0x49,
- 0x01, 0xc0, 0x1c, 0x01, 0xc0, 0x1c
- };
-
- const WebRtc_UWord8 mask4_1[2] =
- {
- 0xf0, 0x00
- };
-
- const WebRtc_UWord8 mask4_2[4] =
- {
- 0xc0, 0x00,
- 0xb0, 0x00
- };
-
- const WebRtc_UWord8 mask4_3[6] =
- {
- 0xc0, 0x00,
- 0xb0, 0x00,
- 0x60, 0x00
- };
-
- const WebRtc_UWord8 mask4_4[8] =
- {
- 0xc0, 0x00,
- 0xa0, 0x00,
- 0x30, 0x00,
- 0x50, 0x00
- };
-
- const WebRtc_UWord8 mask5_1[2] =
- {
- 0xf8, 0x00
- };
-
- const WebRtc_UWord8 mask5_2[4] =
- {
- 0xa8, 0x00,
- 0xd0, 0x00
- };
-
- const WebRtc_UWord8 mask5_3[6] =
- {
- 0xb0, 0x00,
- 0xc8, 0x00,
- 0x50, 0x00
- };
-
- const WebRtc_UWord8 mask5_4[8] =
- {
- 0xc8, 0x00,
- 0xb0, 0x00,
- 0x50, 0x00,
- 0x28, 0x00
- };
-
- const WebRtc_UWord8 mask5_5[10] =
- {
- 0xc0, 0x00,
- 0x30, 0x00,
- 0x18, 0x00,
- 0xa0, 0x00,
- 0x48, 0x00
- };
-
- const WebRtc_UWord8 mask6_1[2] =
- {
- 0xfc, 0x00
- };
-
- const WebRtc_UWord8 mask6_2[4] =
- {
- 0xa8, 0x00,
- 0xd4, 0x00
- };
-
- const WebRtc_UWord8 mask6_3[6] =
- {
- 0xd0, 0x00,
- 0x68, 0x00,
- 0xa4, 0x00
- };
-
- const WebRtc_UWord8 mask6_4[8] =
- {
- 0xa8, 0x00,
- 0x58, 0x00,
- 0x64, 0x00,
- 0x94, 0x00
- };
-
- const WebRtc_UWord8 mask6_5[10] =
- {
- 0xa8, 0x00,
- 0x84, 0x00,
- 0x64, 0x00,
- 0x90, 0x00,
- 0x58, 0x00
- };
-
- const WebRtc_UWord8 mask6_6[12] =
- {
- 0x98, 0x00,
- 0x64, 0x00,
- 0x50, 0x00,
- 0x14, 0x00,
- 0xa8, 0x00,
- 0xe0, 0x00
- };
-
- const WebRtc_UWord8 mask7_1[2] =
- {
- 0xfe, 0x00
- };
-
- const WebRtc_UWord8 mask7_2[4] =
- {
- 0xd4, 0x00,
- 0xaa, 0x00
- };
-
- const WebRtc_UWord8 mask7_3[6] =
- {
- 0xd0, 0x00,
- 0xaa, 0x00,
- 0x64, 0x00
- };
-
- const WebRtc_UWord8 mask7_4[8] =
- {
- 0xd0, 0x00,
- 0xaa, 0x00,
- 0x64, 0x00,
- 0x1c, 0x00
- };
-
- const WebRtc_UWord8 mask7_5[10] =
- {
- 0x0c, 0x00,
- 0xb0, 0x00,
- 0x1a, 0x00,
- 0xc4, 0x00,
- 0x62, 0x00
- };
-
- const WebRtc_UWord8 mask7_6[12] =
- {
- 0x8c, 0x00,
- 0x4a, 0x00,
- 0x64, 0x00,
- 0xd0, 0x00,
- 0xa0, 0x00,
- 0x32, 0x00
- };
-
- const WebRtc_UWord8 mask7_7[14] =
- {
- 0x4a, 0x00,
- 0x94, 0x00,
- 0x1a, 0x00,
- 0xc4, 0x00,
- 0x28, 0x00,
- 0xc2, 0x00,
- 0x34, 0x00
- };
-
- const WebRtc_UWord8 mask8_1[2] =
- {
- 0xff, 0x00
- };
-
- const WebRtc_UWord8 mask8_2[4] =
- {
- 0xaa, 0x00,
- 0xd5, 0x00
- };
-
- const WebRtc_UWord8 mask8_3[6] =
- {
- 0xc5, 0x00,
- 0x92, 0x00,
- 0x6a, 0x00
- };
-
- const WebRtc_UWord8 mask8_4[8] =
- {
- 0x45, 0x00,
- 0xb4, 0x00,
- 0x6a, 0x00,
- 0x89, 0x00
- };
-
- const WebRtc_UWord8 mask8_5[10] =
- {
- 0x8c, 0x00,
- 0x92, 0x00,
- 0x2b, 0x00,
- 0x51, 0x00,
- 0x64, 0x00
- };
-
- const WebRtc_UWord8 mask8_6[12] =
- {
- 0xa1, 0x00,
- 0x52, 0x00,
- 0x91, 0x00,
- 0x2a, 0x00,
- 0xc4, 0x00,
- 0x4c, 0x00
- };
-
- const WebRtc_UWord8 mask8_7[14] =
- {
- 0x15, 0x00,
- 0xc2, 0x00,
- 0x25, 0x00,
- 0x62, 0x00,
- 0x58, 0x00,
- 0x8c, 0x00,
- 0xa3, 0x00
- };
-
- const WebRtc_UWord8 mask8_8[16] =
- {
- 0x25, 0x00,
- 0x8a, 0x00,
- 0x91, 0x00,
- 0x68, 0x00,
- 0x32, 0x00,
- 0x43, 0x00,
- 0xc4, 0x00,
- 0x1c, 0x00
- };
-
- const WebRtc_UWord8 mask9_1[2] =
- {
- 0xff, 0x80
- };
-
- const WebRtc_UWord8 mask9_2[4] =
- {
- 0xaa, 0x80,
- 0xd5, 0x00
- };
-
- const WebRtc_UWord8 mask9_3[6] =
- {
- 0xa5, 0x00,
- 0xc8, 0x00,
- 0x52, 0x80
- };
-
- const WebRtc_UWord8 mask9_4[8] =
- {
- 0xa2, 0x00,
- 0xc9, 0x00,
- 0x52, 0x80,
- 0x24, 0x80
- };
-
- const WebRtc_UWord8 mask9_5[10] =
- {
- 0x8c, 0x00,
- 0x25, 0x00,
- 0x92, 0x80,
- 0x41, 0x80,
- 0x58, 0x00
- };
-
- const WebRtc_UWord8 mask9_6[12] =
- {
- 0x84, 0x80,
- 0x27, 0x00,
- 0x51, 0x80,
- 0x1a, 0x00,
- 0x68, 0x00,
- 0x89, 0x00
- };
-
- const WebRtc_UWord8 mask9_7[14] =
- {
- 0x8c, 0x00,
- 0x47, 0x00,
- 0x81, 0x80,
- 0x12, 0x80,
- 0x58, 0x00,
- 0x28, 0x80,
- 0xb4, 0x00
- };
-
- const WebRtc_UWord8 mask9_8[16] =
- {
- 0x2c, 0x00,
- 0x91, 0x00,
- 0x40, 0x80,
- 0x06, 0x80,
- 0xc8, 0x00,
- 0x45, 0x00,
- 0x30, 0x80,
- 0xa2, 0x00
- };
-
- const WebRtc_UWord8 mask9_9[18] =
- {
- 0x4c, 0x00,
- 0x62, 0x00,
- 0x91, 0x00,
- 0x42, 0x80,
- 0xa4, 0x00,
- 0x13, 0x00,
- 0x30, 0x80,
- 0x88, 0x80,
- 0x09, 0x00
- };
-
- const WebRtc_UWord8* packetMask1[1] =
- {
- mask1_1
- };
-
- const WebRtc_UWord8* packetMask2[2] =
- {
- mask2_1,
- mask2_2
- };
-
- const WebRtc_UWord8* packetMask3[3] =
- {
- mask3_1,
- mask3_2,
- mask3_3
- };
-
- const WebRtc_UWord8* packetMask4[4] =
- {
- mask4_1,
- mask4_2,
- mask4_3,
- mask4_4
- };
-
- const WebRtc_UWord8* packetMask5[5] =
- {
- mask5_1,
- mask5_2,
- mask5_3,
- mask5_4,
- mask5_5
- };
-
- const WebRtc_UWord8* packetMask6[6] =
- {
- mask6_1,
- mask6_2,
- mask6_3,
- mask6_4,
- mask6_5,
- mask6_6
- };
-
- const WebRtc_UWord8* packetMask7[7] =
- {
- mask7_1,
- mask7_2,
- mask7_3,
- mask7_4,
- mask7_5,
- mask7_6,
- mask7_7
- };
-
- const WebRtc_UWord8* packetMask8[8] =
- {
- mask8_1,
- mask8_2,
- mask8_3,
- mask8_4,
- mask8_5,
- mask8_6,
- mask8_7,
- mask8_8
- };
-
- const WebRtc_UWord8* packetMask9[9] =
- {
- mask9_1,
- mask9_2,
- mask9_3,
- mask9_4,
- mask9_5,
- mask9_6,
- mask9_7,
- mask9_8,
- mask9_9
- };
-
- const WebRtc_UWord8* packetMask10[10] =
- {
- mask10_1,
- mask10_2,
- mask10_3,
- mask10_4,
- mask10_5,
- mask10_6,
- mask10_7,
- mask10_8,
- mask10_9,
- mask10_10
- };
-
- const WebRtc_UWord8* packetMask11[11] =
- {
- mask11_1,
- mask11_2,
- mask11_3,
- mask11_4,
- mask11_5,
- mask11_6,
- mask11_7,
- mask11_8,
- mask11_9,
- mask11_10,
- mask11_11
- };
-
- const WebRtc_UWord8* packetMask12[12] =
- {
- mask12_1,
- mask12_2,
- mask12_3,
- mask12_4,
- mask12_5,
- mask12_6,
- mask12_7,
- mask12_8,
- mask12_9,
- mask12_10,
- mask12_11,
- mask12_12
- };
-
- const WebRtc_UWord8* packetMask13[13] =
- {
- mask13_1,
- mask13_2,
- mask13_3,
- mask13_4,
- mask13_5,
- mask13_6,
- mask13_7,
- mask13_8,
- mask13_9,
- mask13_10,
- mask13_11,
- mask13_12,
- mask13_13
- };
-
- const WebRtc_UWord8* packetMask14[14] =
- {
- mask14_1,
- mask14_2,
- mask14_3,
- mask14_4,
- mask14_5,
- mask14_6,
- mask14_7,
- mask14_8,
- mask14_9,
- mask14_10,
- mask14_11,
- mask14_12,
- mask14_13,
- mask14_14
- };
-
- const WebRtc_UWord8* packetMask15[15] =
- {
- mask15_1,
- mask15_2,
- mask15_3,
- mask15_4,
- mask15_5,
- mask15_6,
- mask15_7,
- mask15_8,
- mask15_9,
- mask15_10,
- mask15_11,
- mask15_12,
- mask15_13,
- mask15_14,
- mask15_15
- };
-
- const WebRtc_UWord8* packetMask16[16] =
- {
- mask16_1,
- mask16_2,
- mask16_3,
- mask16_4,
- mask16_5,
- mask16_6,
- mask16_7,
- mask16_8,
- mask16_9,
- mask16_10,
- mask16_11,
- mask16_12,
- mask16_13,
- mask16_14,
- mask16_15,
- mask16_16
- };
-
- const WebRtc_UWord8* packetMask17[17] =
- {
- mask17_1,
- mask17_2,
- mask17_3,
- mask17_4,
- mask17_5,
- mask17_6,
- mask17_7,
- mask17_8,
- mask17_9,
- mask17_10,
- mask17_11,
- mask17_12,
- mask17_13,
- mask17_14,
- mask17_15,
- mask17_16,
- mask17_17
- };
-
- const WebRtc_UWord8* packetMask18[18] =
- {
- mask18_1,
- mask18_2,
- mask18_3,
- mask18_4,
- mask18_5,
- mask18_6,
- mask18_7,
- mask18_8,
- mask18_9,
- mask18_10,
- mask18_11,
- mask18_12,
- mask18_13,
- mask18_14,
- mask18_15,
- mask18_16,
- mask18_17,
- mask18_18
- };
-
- const WebRtc_UWord8* packetMask19[19] =
- {
- mask19_1,
- mask19_2,
- mask19_3,
- mask19_4,
- mask19_5,
- mask19_6,
- mask19_7,
- mask19_8,
- mask19_9,
- mask19_10,
- mask19_11,
- mask19_12,
- mask19_13,
- mask19_14,
- mask19_15,
- mask19_16,
- mask19_17,
- mask19_18,
- mask19_19
- };
-
- const WebRtc_UWord8* packetMask20[20] =
- {
- mask20_1,
- mask20_2,
- mask20_3,
- mask20_4,
- mask20_5,
- mask20_6,
- mask20_7,
- mask20_8,
- mask20_9,
- mask20_10,
- mask20_11,
- mask20_12,
- mask20_13,
- mask20_14,
- mask20_15,
- mask20_16,
- mask20_17,
- mask20_18,
- mask20_19,
- mask20_20
- };
-
- const WebRtc_UWord8* packetMask21[21] =
- {
- mask21_1,
- mask21_2,
- mask21_3,
- mask21_4,
- mask21_5,
- mask21_6,
- mask21_7,
- mask21_8,
- mask21_9,
- mask21_10,
- mask21_11,
- mask21_12,
- mask21_13,
- mask21_14,
- mask21_15,
- mask21_16,
- mask21_17,
- mask21_18,
- mask21_19,
- mask21_20,
- mask21_21
- };
-
- const WebRtc_UWord8* packetMask22[22] =
- {
- mask22_1,
- mask22_2,
- mask22_3,
- mask22_4,
- mask22_5,
- mask22_6,
- mask22_7,
- mask22_8,
- mask22_9,
- mask22_10,
- mask22_11,
- mask22_12,
- mask22_13,
- mask22_14,
- mask22_15,
- mask22_16,
- mask22_17,
- mask22_18,
- mask22_19,
- mask22_20,
- mask22_21,
- mask22_22
- };
-
- const WebRtc_UWord8* packetMask23[23] =
- {
- mask23_1,
- mask23_2,
- mask23_3,
- mask23_4,
- mask23_5,
- mask23_6,
- mask23_7,
- mask23_8,
- mask23_9,
- mask23_10,
- mask23_11,
- mask23_12,
- mask23_13,
- mask23_14,
- mask23_15,
- mask23_16,
- mask23_17,
- mask23_18,
- mask23_19,
- mask23_20,
- mask23_21,
- mask23_22,
- mask23_23
- };
-
- const WebRtc_UWord8* packetMask24[24] =
- {
- mask24_1,
- mask24_2,
- mask24_3,
- mask24_4,
- mask24_5,
- mask24_6,
- mask24_7,
- mask24_8,
- mask24_9,
- mask24_10,
- mask24_11,
- mask24_12,
- mask24_13,
- mask24_14,
- mask24_15,
- mask24_16,
- mask24_17,
- mask24_18,
- mask24_19,
- mask24_20,
- mask24_21,
- mask24_22,
- mask24_23,
- mask24_24
- };
-
- const WebRtc_UWord8* packetMask25[25] =
- {
- mask25_1,
- mask25_2,
- mask25_3,
- mask25_4,
- mask25_5,
- mask25_6,
- mask25_7,
- mask25_8,
- mask25_9,
- mask25_10,
- mask25_11,
- mask25_12,
- mask25_13,
- mask25_14,
- mask25_15,
- mask25_16,
- mask25_17,
- mask25_18,
- mask25_19,
- mask25_20,
- mask25_21,
- mask25_22,
- mask25_23,
- mask25_24,
- mask25_25
- };
-
- const WebRtc_UWord8* packetMask26[26] =
- {
- mask26_1,
- mask26_2,
- mask26_3,
- mask26_4,
- mask26_5,
- mask26_6,
- mask26_7,
- mask26_8,
- mask26_9,
- mask26_10,
- mask26_11,
- mask26_12,
- mask26_13,
- mask26_14,
- mask26_15,
- mask26_16,
- mask26_17,
- mask26_18,
- mask26_19,
- mask26_20,
- mask26_21,
- mask26_22,
- mask26_23,
- mask26_24,
- mask26_25,
- mask26_26
- };
-
- const WebRtc_UWord8* packetMask27[27] =
- {
- mask27_1,
- mask27_2,
- mask27_3,
- mask27_4,
- mask27_5,
- mask27_6,
- mask27_7,
- mask27_8,
- mask27_9,
- mask27_10,
- mask27_11,
- mask27_12,
- mask27_13,
- mask27_14,
- mask27_15,
- mask27_16,
- mask27_17,
- mask27_18,
- mask27_19,
- mask27_20,
- mask27_21,
- mask27_22,
- mask27_23,
- mask27_24,
- mask27_25,
- mask27_26,
- mask27_27
- };
-
- const WebRtc_UWord8* packetMask28[28] =
- {
- mask28_1,
- mask28_2,
- mask28_3,
- mask28_4,
- mask28_5,
- mask28_6,
- mask28_7,
- mask28_8,
- mask28_9,
- mask28_10,
- mask28_11,
- mask28_12,
- mask28_13,
- mask28_14,
- mask28_15,
- mask28_16,
- mask28_17,
- mask28_18,
- mask28_19,
- mask28_20,
- mask28_21,
- mask28_22,
- mask28_23,
- mask28_24,
- mask28_25,
- mask28_26,
- mask28_27,
- mask28_28
- };
-
- const WebRtc_UWord8* packetMask29[29] =
- {
- mask29_1,
- mask29_2,
- mask29_3,
- mask29_4,
- mask29_5,
- mask29_6,
- mask29_7,
- mask29_8,
- mask29_9,
- mask29_10,
- mask29_11,
- mask29_12,
- mask29_13,
- mask29_14,
- mask29_15,
- mask29_16,
- mask29_17,
- mask29_18,
- mask29_19,
- mask29_20,
- mask29_21,
- mask29_22,
- mask29_23,
- mask29_24,
- mask29_25,
- mask29_26,
- mask29_27,
- mask29_28,
- mask29_29
- };
-
- const WebRtc_UWord8* packetMask30[30] =
- {
- mask30_1,
- mask30_2,
- mask30_3,
- mask30_4,
- mask30_5,
- mask30_6,
- mask30_7,
- mask30_8,
- mask30_9,
- mask30_10,
- mask30_11,
- mask30_12,
- mask30_13,
- mask30_14,
- mask30_15,
- mask30_16,
- mask30_17,
- mask30_18,
- mask30_19,
- mask30_20,
- mask30_21,
- mask30_22,
- mask30_23,
- mask30_24,
- mask30_25,
- mask30_26,
- mask30_27,
- mask30_28,
- mask30_29,
- mask30_30
- };
-
- const WebRtc_UWord8* packetMask31[31] =
- {
- mask31_1,
- mask31_2,
- mask31_3,
- mask31_4,
- mask31_5,
- mask31_6,
- mask31_7,
- mask31_8,
- mask31_9,
- mask31_10,
- mask31_11,
- mask31_12,
- mask31_13,
- mask31_14,
- mask31_15,
- mask31_16,
- mask31_17,
- mask31_18,
- mask31_19,
- mask31_20,
- mask31_21,
- mask31_22,
- mask31_23,
- mask31_24,
- mask31_25,
- mask31_26,
- mask31_27,
- mask31_28,
- mask31_29,
- mask31_30,
- mask31_31
- };
-
- const WebRtc_UWord8* packetMask32[32] =
- {
- mask32_1,
- mask32_2,
- mask32_3,
- mask32_4,
- mask32_5,
- mask32_6,
- mask32_7,
- mask32_8,
- mask32_9,
- mask32_10,
- mask32_11,
- mask32_12,
- mask32_13,
- mask32_14,
- mask32_15,
- mask32_16,
- mask32_17,
- mask32_18,
- mask32_19,
- mask32_20,
- mask32_21,
- mask32_22,
- mask32_23,
- mask32_24,
- mask32_25,
- mask32_26,
- mask32_27,
- mask32_28,
- mask32_29,
- mask32_30,
- mask32_31,
- mask32_32
- };
-
- const WebRtc_UWord8* packetMask33[33] =
- {
- mask33_1,
- mask33_2,
- mask33_3,
- mask33_4,
- mask33_5,
- mask33_6,
- mask33_7,
- mask33_8,
- mask33_9,
- mask33_10,
- mask33_11,
- mask33_12,
- mask33_13,
- mask33_14,
- mask33_15,
- mask33_16,
- mask33_17,
- mask33_18,
- mask33_19,
- mask33_20,
- mask33_21,
- mask33_22,
- mask33_23,
- mask33_24,
- mask33_25,
- mask33_26,
- mask33_27,
- mask33_28,
- mask33_29,
- mask33_30,
- mask33_31,
- mask33_32,
- mask33_33
- };
-
- const WebRtc_UWord8* packetMask34[34] =
- {
- mask34_1,
- mask34_2,
- mask34_3,
- mask34_4,
- mask34_5,
- mask34_6,
- mask34_7,
- mask34_8,
- mask34_9,
- mask34_10,
- mask34_11,
- mask34_12,
- mask34_13,
- mask34_14,
- mask34_15,
- mask34_16,
- mask34_17,
- mask34_18,
- mask34_19,
- mask34_20,
- mask34_21,
- mask34_22,
- mask34_23,
- mask34_24,
- mask34_25,
- mask34_26,
- mask34_27,
- mask34_28,
- mask34_29,
- mask34_30,
- mask34_31,
- mask34_32,
- mask34_33,
- mask34_34
- };
-
- const WebRtc_UWord8* packetMask35[35] =
- {
- mask35_1,
- mask35_2,
- mask35_3,
- mask35_4,
- mask35_5,
- mask35_6,
- mask35_7,
- mask35_8,
- mask35_9,
- mask35_10,
- mask35_11,
- mask35_12,
- mask35_13,
- mask35_14,
- mask35_15,
- mask35_16,
- mask35_17,
- mask35_18,
- mask35_19,
- mask35_20,
- mask35_21,
- mask35_22,
- mask35_23,
- mask35_24,
- mask35_25,
- mask35_26,
- mask35_27,
- mask35_28,
- mask35_29,
- mask35_30,
- mask35_31,
- mask35_32,
- mask35_33,
- mask35_34,
- mask35_35
- };
-
- const WebRtc_UWord8* packetMask36[36] =
- {
- mask36_1,
- mask36_2,
- mask36_3,
- mask36_4,
- mask36_5,
- mask36_6,
- mask36_7,
- mask36_8,
- mask36_9,
- mask36_10,
- mask36_11,
- mask36_12,
- mask36_13,
- mask36_14,
- mask36_15,
- mask36_16,
- mask36_17,
- mask36_18,
- mask36_19,
- mask36_20,
- mask36_21,
- mask36_22,
- mask36_23,
- mask36_24,
- mask36_25,
- mask36_26,
- mask36_27,
- mask36_28,
- mask36_29,
- mask36_30,
- mask36_31,
- mask36_32,
- mask36_33,
- mask36_34,
- mask36_35,
- mask36_36
- };
-
- const WebRtc_UWord8* packetMask37[37] =
- {
- mask37_1,
- mask37_2,
- mask37_3,
- mask37_4,
- mask37_5,
- mask37_6,
- mask37_7,
- mask37_8,
- mask37_9,
- mask37_10,
- mask37_11,
- mask37_12,
- mask37_13,
- mask37_14,
- mask37_15,
- mask37_16,
- mask37_17,
- mask37_18,
- mask37_19,
- mask37_20,
- mask37_21,
- mask37_22,
- mask37_23,
- mask37_24,
- mask37_25,
- mask37_26,
- mask37_27,
- mask37_28,
- mask37_29,
- mask37_30,
- mask37_31,
- mask37_32,
- mask37_33,
- mask37_34,
- mask37_35,
- mask37_36,
- mask37_37
- };
-
- const WebRtc_UWord8* packetMask38[38] =
- {
- mask38_1,
- mask38_2,
- mask38_3,
- mask38_4,
- mask38_5,
- mask38_6,
- mask38_7,
- mask38_8,
- mask38_9,
- mask38_10,
- mask38_11,
- mask38_12,
- mask38_13,
- mask38_14,
- mask38_15,
- mask38_16,
- mask38_17,
- mask38_18,
- mask38_19,
- mask38_20,
- mask38_21,
- mask38_22,
- mask38_23,
- mask38_24,
- mask38_25,
- mask38_26,
- mask38_27,
- mask38_28,
- mask38_29,
- mask38_30,
- mask38_31,
- mask38_32,
- mask38_33,
- mask38_34,
- mask38_35,
- mask38_36,
- mask38_37,
- mask38_38
- };
-
- const WebRtc_UWord8* packetMask39[39] =
- {
- mask39_1,
- mask39_2,
- mask39_3,
- mask39_4,
- mask39_5,
- mask39_6,
- mask39_7,
- mask39_8,
- mask39_9,
- mask39_10,
- mask39_11,
- mask39_12,
- mask39_13,
- mask39_14,
- mask39_15,
- mask39_16,
- mask39_17,
- mask39_18,
- mask39_19,
- mask39_20,
- mask39_21,
- mask39_22,
- mask39_23,
- mask39_24,
- mask39_25,
- mask39_26,
- mask39_27,
- mask39_28,
- mask39_29,
- mask39_30,
- mask39_31,
- mask39_32,
- mask39_33,
- mask39_34,
- mask39_35,
- mask39_36,
- mask39_37,
- mask39_38,
- mask39_39
- };
-
- const WebRtc_UWord8* packetMask40[40] =
- {
- mask40_1,
- mask40_2,
- mask40_3,
- mask40_4,
- mask40_5,
- mask40_6,
- mask40_7,
- mask40_8,
- mask40_9,
- mask40_10,
- mask40_11,
- mask40_12,
- mask40_13,
- mask40_14,
- mask40_15,
- mask40_16,
- mask40_17,
- mask40_18,
- mask40_19,
- mask40_20,
- mask40_21,
- mask40_22,
- mask40_23,
- mask40_24,
- mask40_25,
- mask40_26,
- mask40_27,
- mask40_28,
- mask40_29,
- mask40_30,
- mask40_31,
- mask40_32,
- mask40_33,
- mask40_34,
- mask40_35,
- mask40_36,
- mask40_37,
- mask40_38,
- mask40_39,
- mask40_40
- };
-
- const WebRtc_UWord8* packetMask41[41] =
- {
- mask41_1,
- mask41_2,
- mask41_3,
- mask41_4,
- mask41_5,
- mask41_6,
- mask41_7,
- mask41_8,
- mask41_9,
- mask41_10,
- mask41_11,
- mask41_12,
- mask41_13,
- mask41_14,
- mask41_15,
- mask41_16,
- mask41_17,
- mask41_18,
- mask41_19,
- mask41_20,
- mask41_21,
- mask41_22,
- mask41_23,
- mask41_24,
- mask41_25,
- mask41_26,
- mask41_27,
- mask41_28,
- mask41_29,
- mask41_30,
- mask41_31,
- mask41_32,
- mask41_33,
- mask41_34,
- mask41_35,
- mask41_36,
- mask41_37,
- mask41_38,
- mask41_39,
- mask41_40,
- mask41_41
- };
-
- const WebRtc_UWord8* packetMask42[42] =
- {
- mask42_1,
- mask42_2,
- mask42_3,
- mask42_4,
- mask42_5,
- mask42_6,
- mask42_7,
- mask42_8,
- mask42_9,
- mask42_10,
- mask42_11,
- mask42_12,
- mask42_13,
- mask42_14,
- mask42_15,
- mask42_16,
- mask42_17,
- mask42_18,
- mask42_19,
- mask42_20,
- mask42_21,
- mask42_22,
- mask42_23,
- mask42_24,
- mask42_25,
- mask42_26,
- mask42_27,
- mask42_28,
- mask42_29,
- mask42_30,
- mask42_31,
- mask42_32,
- mask42_33,
- mask42_34,
- mask42_35,
- mask42_36,
- mask42_37,
- mask42_38,
- mask42_39,
- mask42_40,
- mask42_41,
- mask42_42
- };
-
- const WebRtc_UWord8* packetMask43[43] =
- {
- mask43_1,
- mask43_2,
- mask43_3,
- mask43_4,
- mask43_5,
- mask43_6,
- mask43_7,
- mask43_8,
- mask43_9,
- mask43_10,
- mask43_11,
- mask43_12,
- mask43_13,
- mask43_14,
- mask43_15,
- mask43_16,
- mask43_17,
- mask43_18,
- mask43_19,
- mask43_20,
- mask43_21,
- mask43_22,
- mask43_23,
- mask43_24,
- mask43_25,
- mask43_26,
- mask43_27,
- mask43_28,
- mask43_29,
- mask43_30,
- mask43_31,
- mask43_32,
- mask43_33,
- mask43_34,
- mask43_35,
- mask43_36,
- mask43_37,
- mask43_38,
- mask43_39,
- mask43_40,
- mask43_41,
- mask43_42,
- mask43_43
- };
-
- const WebRtc_UWord8* packetMask44[44] =
- {
- mask44_1,
- mask44_2,
- mask44_3,
- mask44_4,
- mask44_5,
- mask44_6,
- mask44_7,
- mask44_8,
- mask44_9,
- mask44_10,
- mask44_11,
- mask44_12,
- mask44_13,
- mask44_14,
- mask44_15,
- mask44_16,
- mask44_17,
- mask44_18,
- mask44_19,
- mask44_20,
- mask44_21,
- mask44_22,
- mask44_23,
- mask44_24,
- mask44_25,
- mask44_26,
- mask44_27,
- mask44_28,
- mask44_29,
- mask44_30,
- mask44_31,
- mask44_32,
- mask44_33,
- mask44_34,
- mask44_35,
- mask44_36,
- mask44_37,
- mask44_38,
- mask44_39,
- mask44_40,
- mask44_41,
- mask44_42,
- mask44_43,
- mask44_44
- };
-
- const WebRtc_UWord8* packetMask45[45] =
- {
- mask45_1,
- mask45_2,
- mask45_3,
- mask45_4,
- mask45_5,
- mask45_6,
- mask45_7,
- mask45_8,
- mask45_9,
- mask45_10,
- mask45_11,
- mask45_12,
- mask45_13,
- mask45_14,
- mask45_15,
- mask45_16,
- mask45_17,
- mask45_18,
- mask45_19,
- mask45_20,
- mask45_21,
- mask45_22,
- mask45_23,
- mask45_24,
- mask45_25,
- mask45_26,
- mask45_27,
- mask45_28,
- mask45_29,
- mask45_30,
- mask45_31,
- mask45_32,
- mask45_33,
- mask45_34,
- mask45_35,
- mask45_36,
- mask45_37,
- mask45_38,
- mask45_39,
- mask45_40,
- mask45_41,
- mask45_42,
- mask45_43,
- mask45_44,
- mask45_45
- };
-
- const WebRtc_UWord8* packetMask46[46] =
- {
- mask46_1,
- mask46_2,
- mask46_3,
- mask46_4,
- mask46_5,
- mask46_6,
- mask46_7,
- mask46_8,
- mask46_9,
- mask46_10,
- mask46_11,
- mask46_12,
- mask46_13,
- mask46_14,
- mask46_15,
- mask46_16,
- mask46_17,
- mask46_18,
- mask46_19,
- mask46_20,
- mask46_21,
- mask46_22,
- mask46_23,
- mask46_24,
- mask46_25,
- mask46_26,
- mask46_27,
- mask46_28,
- mask46_29,
- mask46_30,
- mask46_31,
- mask46_32,
- mask46_33,
- mask46_34,
- mask46_35,
- mask46_36,
- mask46_37,
- mask46_38,
- mask46_39,
- mask46_40,
- mask46_41,
- mask46_42,
- mask46_43,
- mask46_44,
- mask46_45,
- mask46_46
- };
-
- const WebRtc_UWord8* packetMask47[47] =
- {
- mask47_1,
- mask47_2,
- mask47_3,
- mask47_4,
- mask47_5,
- mask47_6,
- mask47_7,
- mask47_8,
- mask47_9,
- mask47_10,
- mask47_11,
- mask47_12,
- mask47_13,
- mask47_14,
- mask47_15,
- mask47_16,
- mask47_17,
- mask47_18,
- mask47_19,
- mask47_20,
- mask47_21,
- mask47_22,
- mask47_23,
- mask47_24,
- mask47_25,
- mask47_26,
- mask47_27,
- mask47_28,
- mask47_29,
- mask47_30,
- mask47_31,
- mask47_32,
- mask47_33,
- mask47_34,
- mask47_35,
- mask47_36,
- mask47_37,
- mask47_38,
- mask47_39,
- mask47_40,
- mask47_41,
- mask47_42,
- mask47_43,
- mask47_44,
- mask47_45,
- mask47_46,
- mask47_47
- };
-
- const WebRtc_UWord8* packetMask48[48] =
- {
- mask48_1,
- mask48_2,
- mask48_3,
- mask48_4,
- mask48_5,
- mask48_6,
- mask48_7,
- mask48_8,
- mask48_9,
- mask48_10,
- mask48_11,
- mask48_12,
- mask48_13,
- mask48_14,
- mask48_15,
- mask48_16,
- mask48_17,
- mask48_18,
- mask48_19,
- mask48_20,
- mask48_21,
- mask48_22,
- mask48_23,
- mask48_24,
- mask48_25,
- mask48_26,
- mask48_27,
- mask48_28,
- mask48_29,
- mask48_30,
- mask48_31,
- mask48_32,
- mask48_33,
- mask48_34,
- mask48_35,
- mask48_36,
- mask48_37,
- mask48_38,
- mask48_39,
- mask48_40,
- mask48_41,
- mask48_42,
- mask48_43,
- mask48_44,
- mask48_45,
- mask48_46,
- mask48_47,
- mask48_48
- };
-
- const WebRtc_UWord8** packetMaskTbl[48] =
- {
- packetMask1,
- packetMask2,
- packetMask3,
- packetMask4,
- packetMask5,
- packetMask6,
- packetMask7,
- packetMask8,
- packetMask9,
- packetMask10,
- packetMask11,
- packetMask12,
- packetMask13,
- packetMask14,
- packetMask15,
- packetMask16,
- packetMask17,
- packetMask18,
- packetMask19,
- packetMask20,
- packetMask21,
- packetMask22,
- packetMask23,
- packetMask24,
- packetMask25,
- packetMask26,
- packetMask27,
- packetMask28,
- packetMask29,
- packetMask30,
- packetMask31,
- packetMask32,
- packetMask33,
- packetMask34,
- packetMask35,
- packetMask36,
- packetMask37,
- packetMask38,
- packetMask39,
- packetMask40,
- packetMask41,
- packetMask42,
- packetMask43,
- packetMask44,
- packetMask45,
- packetMask46,
- packetMask47,
- packetMask48
- };
-} // Anonymous namespace
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_FEC_PRIVATE_TABLES_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/forward_error_correction.cc b/trunk/src/modules/rtp_rtcp/source/forward_error_correction.cc
deleted file mode 100644
index 22e1a38..0000000
--- a/trunk/src/modules/rtp_rtcp/source/forward_error_correction.cc
+++ /dev/null
@@ -1,699 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <algorithm>
-#include <cassert>
-#include <cstring>
-#include <iterator>
-
-#include "modules/rtp_rtcp/source/forward_error_correction.h"
-#include "modules/rtp_rtcp/source/forward_error_correction_internal.h"
-#include "modules/rtp_rtcp/source/rtp_utility.h"
-#include "system_wrappers/interface/trace.h"
-
-namespace webrtc {
-
-// Minimum RTP header size in bytes.
-const uint8_t kRtpHeaderSize = 12;
-
-// FEC header size in bytes.
-const uint8_t kFecHeaderSize = 10;
-
-// ULP header size in bytes (L bit is set).
-const uint8_t kUlpHeaderSizeLBitSet = (2 + kMaskSizeLBitSet);
-
-// ULP header size in bytes (L bit is cleared).
-const uint8_t kUlpHeaderSizeLBitClear = (2 + kMaskSizeLBitClear);
-
-// Transport header size in bytes. Assume UDP/IPv4 as a reasonable minimum.
-const uint8_t kTransportOverhead = 28;
-
-enum { kMaxFecPackets = ForwardErrorCorrection::kMaxMediaPackets };
-
-// Used to link media packets to their protecting FEC packets.
-//
-// TODO(holmer): Refactor into a proper class.
-class ProtectedPacket : public ForwardErrorCorrection::SortablePacket {
- public:
- scoped_refptr<ForwardErrorCorrection::Packet> pkt;
-};
-
-typedef std::list<ProtectedPacket*> ProtectedPacketList;
-
-//
-// Used for internal storage of FEC packets in a list.
-//
-// TODO(holmer): Refactor into a proper class.
-class FecPacket : public ForwardErrorCorrection::SortablePacket {
- public:
- ProtectedPacketList protectedPktList;
- uint32_t ssrc; // SSRC of the current frame.
- scoped_refptr<ForwardErrorCorrection::Packet> pkt;
-};
-
-bool ForwardErrorCorrection::SortablePacket::LessThan(
- const SortablePacket* first,
- const SortablePacket* second) {
- return (first->seqNum != second->seqNum &&
- LatestSequenceNumber(first->seqNum, second->seqNum) == second->seqNum);
-}
-
-ForwardErrorCorrection::ForwardErrorCorrection(int32_t id)
- : _id(id),
- _generatedFecPackets(kMaxMediaPackets),
- _fecPacketReceived(false) {
-}
-
-ForwardErrorCorrection::~ForwardErrorCorrection() {
-}
-
-// Input packet
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// | RTP Header (12 octets) |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// | RTP Payload |
-// | |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-// Output packet
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// | FEC Header (10 octets) |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// | FEC Level 0 Header |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// | FEC Level 0 Payload |
-// | |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-int32_t ForwardErrorCorrection::GenerateFEC(
- const PacketList& mediaPacketList,
- uint8_t protectionFactor,
- int numImportantPackets,
- bool useUnequalProtection,
- PacketList* fecPacketList) {
- if (mediaPacketList.empty()) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s media packet list is empty", __FUNCTION__);
- return -1;
- }
- if (!fecPacketList->empty()) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s FEC packet list is not empty", __FUNCTION__);
- return -1;
- }
- const uint16_t numMediaPackets = mediaPacketList.size();
- const uint8_t lBit = numMediaPackets > 16 ? 1 : 0;
- const uint16_t numMaskBytes = (lBit == 1)?
- kMaskSizeLBitSet : kMaskSizeLBitClear;
-
- if (numMediaPackets > kMaxMediaPackets) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s can only protect %d media packets per frame; %d requested",
- __FUNCTION__, kMaxMediaPackets, numMediaPackets);
- return -1;
- }
-
- // Error checking on the number of important packets.
- // Can't have more important packets than media packets.
- if (numImportantPackets > numMediaPackets) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "Number of important packets (%d) greater than number of media "
- "packets (%d)", numImportantPackets, numMediaPackets);
- return -1;
- }
- if (numImportantPackets < 0) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "Number of important packets (%d) less than zero",
- numImportantPackets);
- return -1;
- }
- // Do some error checking on the media packets.
- PacketList::const_iterator mediaListIt = mediaPacketList.begin();
- while (mediaListIt != mediaPacketList.end()) {
- Packet* mediaPacket = *mediaListIt;
- assert(mediaPacket);
-
- if (mediaPacket->length < kRtpHeaderSize) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s media packet (%d bytes) is smaller than RTP header",
- __FUNCTION__, mediaPacket->length);
- return -1;
- }
-
- // Ensure our FEC packets will fit in a typical MTU.
- if (mediaPacket->length + PacketOverhead() + kTransportOverhead >
- IP_PACKET_SIZE) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s media packet (%d bytes) with overhead is larger than MTU(%d)",
- __FUNCTION__, mediaPacket->length, IP_PACKET_SIZE);
- return -1;
- }
- mediaListIt++;
- }
- // Result in Q0 with an unsigned round.
- uint32_t numFecPackets = (numMediaPackets * protectionFactor + (1 << 7)) >> 8;
- // Generate at least one FEC packet if we need protection.
- if (protectionFactor > 0 && numFecPackets == 0) {
- numFecPackets = 1;
- }
- if (numFecPackets == 0) {
- return 0;
- }
- assert(numFecPackets <= numMediaPackets);
-
- // Prepare FEC packets by setting them to 0.
- for (uint32_t i = 0; i < numFecPackets; i++) {
- memset(_generatedFecPackets[i].data, 0, IP_PACKET_SIZE);
- _generatedFecPackets[i].length = 0; // Use this as a marker for untouched
- // packets.
- fecPacketList->push_back(&_generatedFecPackets[i]);
- }
-
- // -- Generate packet masks --
- uint8_t* packetMask = new uint8_t[numFecPackets * numMaskBytes];
- memset(packetMask, 0, numFecPackets * numMaskBytes);
- internal::GeneratePacketMasks(numMediaPackets, numFecPackets,
- numImportantPackets, useUnequalProtection,
- packetMask);
-
- GenerateFecBitStrings(mediaPacketList, packetMask, numFecPackets);
-
- GenerateFecUlpHeaders(mediaPacketList, packetMask, numFecPackets);
-
- delete [] packetMask;
- return 0;
-}
-
-void ForwardErrorCorrection::GenerateFecBitStrings(
- const PacketList& mediaPacketList,
- uint8_t* packetMask,
- uint32_t numFecPackets) {
- uint8_t mediaPayloadLength[2];
- const uint8_t lBit = mediaPacketList.size() > 16 ? 1 : 0;
- const uint16_t numMaskBytes = (lBit == 1) ?
- kMaskSizeLBitSet : kMaskSizeLBitClear;
- const uint16_t ulpHeaderSize = (lBit == 1) ?
- kUlpHeaderSizeLBitSet : kUlpHeaderSizeLBitClear;
- const uint16_t fecRtpOffset = kFecHeaderSize + ulpHeaderSize - kRtpHeaderSize;
-
- for (uint32_t i = 0; i < numFecPackets; i++) {
- PacketList::const_iterator mediaListIt = mediaPacketList.begin();
- uint32_t pktMaskIdx = i * numMaskBytes;
- uint32_t mediaPktIdx = 0;
- uint16_t fecPacketLength = 0;
- while (mediaListIt != mediaPacketList.end()) {
- // Each FEC packet has a multiple byte mask.
- if (packetMask[pktMaskIdx] & (1 << (7 - mediaPktIdx))) {
- Packet* mediaPacket = *mediaListIt;
-
- // Assign network-ordered media payload length.
- ModuleRTPUtility::AssignUWord16ToBuffer(
- mediaPayloadLength,
- mediaPacket->length - kRtpHeaderSize);
-
- fecPacketLength = mediaPacket->length + fecRtpOffset;
- // On the first protected packet, we don't need to XOR.
- if (_generatedFecPackets[i].length == 0) {
- // Copy the first 2 bytes of the RTP header.
- memcpy(_generatedFecPackets[i].data, mediaPacket->data, 2);
- // Copy the 5th to 8th bytes of the RTP header.
- memcpy(&_generatedFecPackets[i].data[4], &mediaPacket->data[4], 4);
- // Copy network-ordered payload size.
- memcpy(&_generatedFecPackets[i].data[8], mediaPayloadLength, 2);
-
- // Copy RTP payload, leaving room for the ULP header.
- memcpy(&_generatedFecPackets[i].data[kFecHeaderSize + ulpHeaderSize],
- &mediaPacket->data[kRtpHeaderSize],
- mediaPacket->length - kRtpHeaderSize);
- } else {
- // XOR with the first 2 bytes of the RTP header.
- _generatedFecPackets[i].data[0] ^= mediaPacket->data[0];
- _generatedFecPackets[i].data[1] ^= mediaPacket->data[1];
-
- // XOR with the 5th to 8th bytes of the RTP header.
- for (uint32_t j = 4; j < 8; j++) {
- _generatedFecPackets[i].data[j] ^= mediaPacket->data[j];
- }
-
- // XOR with the network-ordered payload size.
- _generatedFecPackets[i].data[8] ^= mediaPayloadLength[0];
- _generatedFecPackets[i].data[9] ^= mediaPayloadLength[1];
-
- // XOR with RTP payload, leaving room for the ULP header.
- for (int32_t j = kFecHeaderSize + ulpHeaderSize;
- j < fecPacketLength; j++) {
- _generatedFecPackets[i].data[j] ^=
- mediaPacket->data[j - fecRtpOffset];
- }
- }
- if (fecPacketLength > _generatedFecPackets[i].length) {
- _generatedFecPackets[i].length = fecPacketLength;
- }
- }
- mediaListIt++;
- mediaPktIdx++;
- if (mediaPktIdx == 8) {
- // Switch to the next mask byte.
- mediaPktIdx = 0;
- pktMaskIdx++;
- }
- }
- assert(_generatedFecPackets[i].length);
- //Note: This shouldn't happen: means packet mask is wrong or poorly designed
- }
-}
-
-void ForwardErrorCorrection::GenerateFecUlpHeaders(
- const PacketList& mediaPacketList,
- uint8_t* packetMask,
- uint32_t numFecPackets) {
- // -- Generate FEC and ULP headers --
- //
- // FEC Header, 10 bytes
- // 0 1 2 3
- // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // |E|L|P|X| CC |M| PT recovery | SN base |
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // | TS recovery |
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // | length recovery |
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- //
- // ULP Header, 4 bytes (for L = 0)
- // 0 1 2 3
- // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // | Protection Length | mask |
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // | mask cont. (present only when L = 1) |
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- PacketList::const_iterator mediaListIt = mediaPacketList.begin();
- Packet* mediaPacket = *mediaListIt;
- assert(mediaPacket != NULL);
- const uint8_t lBit = mediaPacketList.size() > 16 ? 1 : 0;
- const uint16_t numMaskBytes = (lBit == 1)?
- kMaskSizeLBitSet : kMaskSizeLBitClear;
- const uint16_t ulpHeaderSize = (lBit == 1)?
- kUlpHeaderSizeLBitSet : kUlpHeaderSizeLBitClear;
-
- for (uint32_t i = 0; i < numFecPackets; i++) {
- // -- FEC header --
- _generatedFecPackets[i].data[0] &= 0x7f; // Set E to zero.
- if (lBit == 0) {
- _generatedFecPackets[i].data[0] &= 0xbf; // Clear the L bit.
- } else {
- _generatedFecPackets[i].data[0] |= 0x40; // Set the L bit.
- }
- // Two byte sequence number from first RTP packet to SN base.
- // We use the same sequence number base for every FEC packet,
- // but that's not required in general.
- memcpy(&_generatedFecPackets[i].data[2], &mediaPacket->data[2], 2);
-
- // -- ULP header --
- // Copy the payload size to the protection length field.
- // (We protect the entire packet.)
- ModuleRTPUtility::AssignUWord16ToBuffer(&_generatedFecPackets[i].data[10],
- _generatedFecPackets[i].length - kFecHeaderSize - ulpHeaderSize);
-
- // Copy the packet mask.
- memcpy(&_generatedFecPackets[i].data[12], &packetMask[i * numMaskBytes],
- numMaskBytes);
- }
-}
-
-void ForwardErrorCorrection::ResetState(
- RecoveredPacketList* recoveredPacketList) {
- _fecPacketReceived = false;
-
- // Free the memory for any existing recovered packets, if the user hasn't.
- while (!recoveredPacketList->empty()) {
- delete recoveredPacketList->front();
- recoveredPacketList->pop_front();
- }
- assert(recoveredPacketList->empty());
-
- // Free the FEC packet list.
- while (!_fecPacketList.empty()) {
- FecPacketList::iterator fecPacketListIt = _fecPacketList.begin();
- FecPacket* fecPacket = *fecPacketListIt;
- ProtectedPacketList::iterator protectedPacketListIt;
- protectedPacketListIt = fecPacket->protectedPktList.begin();
- while (protectedPacketListIt != fecPacket->protectedPktList.end()) {
- delete *protectedPacketListIt;
- protectedPacketListIt =
- fecPacket->protectedPktList.erase(protectedPacketListIt);
- }
- assert(fecPacket->protectedPktList.empty());
- delete fecPacket;
- _fecPacketList.pop_front();
- }
- assert(_fecPacketList.empty());
-}
-
-void ForwardErrorCorrection::InsertMediaPacket(
- ReceivedPacket* rxPacket,
- RecoveredPacketList* recoveredPacketList) {
- RecoveredPacketList::iterator recoveredPacketListIt =
- recoveredPacketList->begin();
-
- // Search for duplicate packets.
- while (recoveredPacketListIt != recoveredPacketList->end()) {
- if (rxPacket->seqNum == (*recoveredPacketListIt)->seqNum) {
- // Duplicate packet, no need to add to list.
- // Delete duplicate media packet data.
- rxPacket->pkt = NULL;
- return;
- }
- recoveredPacketListIt++;
- }
- RecoveredPacket* recoverdPacketToInsert = new RecoveredPacket;
- recoverdPacketToInsert->wasRecovered = false;
- recoverdPacketToInsert->returned = false;
- recoverdPacketToInsert->seqNum = rxPacket->seqNum;
- recoverdPacketToInsert->pkt = rxPacket->pkt;
- recoverdPacketToInsert->pkt->length = rxPacket->pkt->length;
-
- // TODO(holmer): Consider replacing this with a binary search for the right
- // position, and then just insert the new packet. Would get rid of the sort.
- recoveredPacketList->push_back(recoverdPacketToInsert);
- recoveredPacketList->sort(SortablePacket::LessThan);
- UpdateCoveringFECPackets(recoverdPacketToInsert);
-}
-
-void ForwardErrorCorrection::UpdateCoveringFECPackets(RecoveredPacket* packet) {
- for (FecPacketList::iterator it = _fecPacketList.begin();
- it != _fecPacketList.end(); ++it) {
- // Is this FEC packet protecting the media packet |packet|?
- ProtectedPacketList::iterator protected_it = std::lower_bound(
- (*it)->protectedPktList.begin(),
- (*it)->protectedPktList.end(),
- packet,
- SortablePacket::LessThan);
- if (protected_it != (*it)->protectedPktList.end() &&
- (*protected_it)->seqNum == packet->seqNum) {
- // Found an FEC packet which is protecting |packet|.
- (*protected_it)->pkt = packet->pkt;
- }
- }
-}
-
-void ForwardErrorCorrection::InsertFECPacket(
- ReceivedPacket* rxPacket,
- const RecoveredPacketList* recoveredPacketList) {
- _fecPacketReceived = true;
-
- // Check for duplicate.
- FecPacketList::iterator fecPacketListIt = _fecPacketList.begin();
- while (fecPacketListIt != _fecPacketList.end()) {
- if (rxPacket->seqNum == (*fecPacketListIt)->seqNum) {
- // Delete duplicate FEC packet data.
- rxPacket->pkt = NULL;
- return;
- }
- fecPacketListIt++;
- }
- FecPacket* fecPacket = new FecPacket;
- fecPacket->pkt = rxPacket->pkt;
- fecPacket->seqNum = rxPacket->seqNum;
- fecPacket->ssrc = rxPacket->ssrc;
-
- const uint16_t seqNumBase = ModuleRTPUtility::BufferToUWord16(
- &fecPacket->pkt->data[2]);
- const uint16_t maskSizeBytes = (fecPacket->pkt->data[0] & 0x40) ?
- kMaskSizeLBitSet : kMaskSizeLBitClear; // L bit set?
-
- for (uint16_t byteIdx = 0; byteIdx < maskSizeBytes; byteIdx++) {
- uint8_t packetMask = fecPacket->pkt->data[12 + byteIdx];
- for (uint16_t bitIdx = 0; bitIdx < 8; bitIdx++) {
- if (packetMask & (1 << (7 - bitIdx))) {
- ProtectedPacket* protectedPacket = new ProtectedPacket;
- fecPacket->protectedPktList.push_back(protectedPacket);
- // This wraps naturally with the sequence number.
- protectedPacket->seqNum = static_cast<uint16_t>(seqNumBase +
- (byteIdx << 3) + bitIdx);
- protectedPacket->pkt = NULL;
- }
- }
- }
- if (fecPacket->protectedPktList.empty()) {
- // All-zero packet mask; we can discard this FEC packet.
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
- "FEC packet %u has an all-zero packet mask.",
- fecPacket->seqNum, __FUNCTION__);
- delete fecPacket;
- } else {
- AssignRecoveredPackets(fecPacket,
- recoveredPacketList);
- // TODO(holmer): Consider replacing this with a binary search for the right
- // position, and then just insert the new packet. Would get rid of the sort.
- _fecPacketList.push_back(fecPacket);
- _fecPacketList.sort(SortablePacket::LessThan);
- if (_fecPacketList.size() > kMaxFecPackets) {
- DiscardFECPacket(_fecPacketList.front());
- _fecPacketList.pop_front();
- }
- assert(_fecPacketList.size() <= kMaxFecPackets);
- }
-}
-
-void ForwardErrorCorrection::AssignRecoveredPackets(
- FecPacket* fec_packet,
- const RecoveredPacketList* recovered_packets) {
- // Search for missing packets which have arrived or have been recovered by
- // another FEC packet.
- ProtectedPacketList* not_recovered = &fec_packet->protectedPktList;
- RecoveredPacketList already_recovered;
- std::set_intersection(
- recovered_packets->begin(), recovered_packets->end(),
- not_recovered->begin(), not_recovered->end(),
- std::inserter(already_recovered, already_recovered.end()),
- SortablePacket::LessThan);
- // Set the FEC pointers to all recovered packets so that we don't have to
- // search for them when we are doing recovery.
- ProtectedPacketList::iterator not_recovered_it = not_recovered->begin();
- for (RecoveredPacketList::iterator it = already_recovered.begin();
- it != already_recovered.end(); ++it) {
- // Search for the next recovered packet in |not_recovered|.
- while ((*not_recovered_it)->seqNum != (*it)->seqNum)
- ++not_recovered_it;
- (*not_recovered_it)->pkt = (*it)->pkt;
- }
-}
-
-void ForwardErrorCorrection::InsertPackets(
- ReceivedPacketList* receivedPacketList,
- RecoveredPacketList* recoveredPacketList) {
-
- while (!receivedPacketList->empty()) {
- ReceivedPacket* rxPacket = receivedPacketList->front();
-
- if (rxPacket->isFec) {
- InsertFECPacket(rxPacket, recoveredPacketList);
- } else {
- // Insert packet at the end of |recoveredPacketList|.
- InsertMediaPacket(rxPacket, recoveredPacketList);
- }
- // Delete the received packet "wrapper", but not the packet data.
- delete rxPacket;
- receivedPacketList->pop_front();
- }
- assert(receivedPacketList->empty());
- DiscardOldPackets(recoveredPacketList);
-}
-
-void ForwardErrorCorrection::InitRecovery(
- const FecPacket* fec_packet,
- RecoveredPacket* recovered) {
- // This is the first packet which we try to recover with.
- const uint16_t ulpHeaderSize = fec_packet->pkt->data[0] & 0x40 ?
- kUlpHeaderSizeLBitSet : kUlpHeaderSizeLBitClear; // L bit set?
- recovered->pkt = new Packet;
- memset(recovered->pkt->data, 0, IP_PACKET_SIZE);
- recovered->returned = false;
- recovered->wasRecovered = true;
- uint8_t protectionLength[2];
- // Copy the protection length from the ULP header.
- memcpy(protectionLength, &fec_packet->pkt->data[10], 2);
- // Copy FEC payload, skipping the ULP header.
- memcpy(&recovered->pkt->data[kRtpHeaderSize],
- &fec_packet->pkt->data[kFecHeaderSize + ulpHeaderSize],
- ModuleRTPUtility::BufferToUWord16(protectionLength));
- // Copy the length recovery field.
- memcpy(recovered->length_recovery, &fec_packet->pkt->data[8], 2);
- // Copy the first 2 bytes of the FEC header.
- memcpy(recovered->pkt->data, fec_packet->pkt->data, 2);
- // Copy the 5th to 8th bytes of the FEC header.
- memcpy(&recovered->pkt->data[4], &fec_packet->pkt->data[4], 4);
- // Set the SSRC field.
- ModuleRTPUtility::AssignUWord32ToBuffer(&recovered->pkt->data[8],
- fec_packet->ssrc);
-}
-
-void ForwardErrorCorrection::FinishRecovery(RecoveredPacket* recovered) {
- // Set the RTP version to 2.
- recovered->pkt->data[0] |= 0x80; // Set the 1st bit.
- recovered->pkt->data[0] &= 0xbf; // Clear the 2nd bit.
-
- // Set the SN field.
- ModuleRTPUtility::AssignUWord16ToBuffer(&recovered->pkt->data[2],
- recovered->seqNum);
- // Recover the packet length.
- recovered->pkt->length = ModuleRTPUtility::BufferToUWord16(
- recovered->length_recovery) + kRtpHeaderSize;
-}
-
-void ForwardErrorCorrection::XorPackets(const Packet* src_packet,
- RecoveredPacket* dst_packet) {
- // XOR with the first 2 bytes of the RTP header.
- for (uint32_t i = 0; i < 2; i++) {
- dst_packet->pkt->data[i] ^= src_packet->data[i];
- }
- // XOR with the 5th to 8th bytes of the RTP header.
- for (uint32_t i = 4; i < 8; i++) {
- dst_packet->pkt->data[i] ^= src_packet->data[i];
- }
- // XOR with the network-ordered payload size.
- uint8_t mediaPayloadLength[2];
- ModuleRTPUtility::AssignUWord16ToBuffer(
- mediaPayloadLength,
- src_packet->length - kRtpHeaderSize);
- dst_packet->length_recovery[0] ^= mediaPayloadLength[0];
- dst_packet->length_recovery[1] ^= mediaPayloadLength[1];
-
- // XOR with RTP payload.
- // TODO(marpan/ajm): Are we doing more XORs than required here?
- for (int32_t i = kRtpHeaderSize; i < src_packet->length; i++) {
- dst_packet->pkt->data[i] ^= src_packet->data[i];
- }
-}
-
-void ForwardErrorCorrection::RecoverPacket(
- const FecPacket* fecPacket,
- RecoveredPacket* recPacketToInsert) {
- InitRecovery(fecPacket, recPacketToInsert);
- ProtectedPacketList::const_iterator protected_it =
- fecPacket->protectedPktList.begin();
- while (protected_it != fecPacket->protectedPktList.end()) {
- if ((*protected_it)->pkt == NULL) {
- // This is the packet we're recovering.
- recPacketToInsert->seqNum = (*protected_it)->seqNum;
- } else {
- XorPackets((*protected_it)->pkt, recPacketToInsert);
- }
- ++protected_it;
- }
- FinishRecovery(recPacketToInsert);
-}
-
-void ForwardErrorCorrection::AttemptRecover(
- RecoveredPacketList* recoveredPacketList) {
- FecPacketList::iterator fecPacketListIt = _fecPacketList.begin();
- while (fecPacketListIt != _fecPacketList.end()) {
- // Search for each FEC packet's protected media packets.
- int packets_missing = NumCoveredPacketsMissing(*fecPacketListIt);
-
- // We can only recover one packet with an FEC packet.
- if (packets_missing == 1) {
- // Recovery possible.
- RecoveredPacket* packetToInsert = new RecoveredPacket;
- packetToInsert->pkt = NULL;
- RecoverPacket(*fecPacketListIt, packetToInsert);
-
- // Add recovered packet to the list of recovered packets and update any
- // FEC packets covering this packet with a pointer to the data.
- // TODO(holmer): Consider replacing this with a binary search for the
- // right position, and then just insert the new packet. Would get rid of
- // the sort.
- recoveredPacketList->push_back(packetToInsert);
- recoveredPacketList->sort(SortablePacket::LessThan);
- UpdateCoveringFECPackets(packetToInsert);
- DiscardOldPackets(recoveredPacketList);
- DiscardFECPacket(*fecPacketListIt);
- fecPacketListIt = _fecPacketList.erase(fecPacketListIt);
-
- // A packet has been recovered. We need to check the FEC list again, as
- // this may allow additional packets to be recovered.
- // Restart for first FEC packet.
- fecPacketListIt = _fecPacketList.begin();
- } else if (packets_missing == 0) {
- // Either all protected packets arrived or have been recovered. We can
- // discard this FEC packet.
- DiscardFECPacket(*fecPacketListIt);
- fecPacketListIt = _fecPacketList.erase(fecPacketListIt);
- } else {
- fecPacketListIt++;
- }
- }
-}
-
-int ForwardErrorCorrection::NumCoveredPacketsMissing(
- const FecPacket* fec_packet) {
- int packets_missing = 0;
- ProtectedPacketList::const_iterator it = fec_packet->protectedPktList.begin();
- for (; it != fec_packet->protectedPktList.end(); ++it) {
- if ((*it)->pkt == NULL) {
- ++packets_missing;
- if (packets_missing > 1) {
- break; // We can't recover more than one packet.
- }
- }
- }
- return packets_missing;
-}
-
-void ForwardErrorCorrection::DiscardFECPacket(FecPacket* fec_packet) {
- while (!fec_packet->protectedPktList.empty()) {
- delete fec_packet->protectedPktList.front();
- fec_packet->protectedPktList.pop_front();
- }
- assert(fec_packet->protectedPktList.empty());
- delete fec_packet;
-}
-
-void ForwardErrorCorrection::DiscardOldPackets(
- RecoveredPacketList* recoveredPacketList) {
- while (recoveredPacketList->size() > kMaxMediaPackets) {
- ForwardErrorCorrection::RecoveredPacket* packet =
- recoveredPacketList->front();
- delete packet;
- recoveredPacketList->pop_front();
- }
- assert(recoveredPacketList->size() <= kMaxMediaPackets);
-}
-
-int32_t ForwardErrorCorrection::DecodeFEC(
- ReceivedPacketList* receivedPacketList,
- RecoveredPacketList* recoveredPacketList) {
- // TODO(marpan/ajm): can we check for multiple ULP headers, and return an
- // error?
- InsertPackets(receivedPacketList, recoveredPacketList);
- AttemptRecover(recoveredPacketList);
- return 0;
-}
-
-uint16_t ForwardErrorCorrection::PacketOverhead() {
- return kFecHeaderSize + kUlpHeaderSizeLBitSet;
-}
-
-uint16_t ForwardErrorCorrection::LatestSequenceNumber(uint16_t first,
- uint16_t second) {
- bool wrap = (first < 0x00ff && second > 0xff00) ||
- (first > 0xff00 && second < 0x00ff);
- if (second > first && !wrap)
- return second;
- else if (second <= first && !wrap)
- return first;
- else if (second < first && wrap)
- return second;
- else
- return first;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/forward_error_correction.h b/trunk/src/modules/rtp_rtcp/source/forward_error_correction.h
deleted file mode 100644
index a7588a8..0000000
--- a/trunk/src/modules/rtp_rtcp/source/forward_error_correction.h
+++ /dev/null
@@ -1,289 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_H_
-
-#include <list>
-#include <vector>
-
-#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "system_wrappers/interface/ref_count.h"
-#include "system_wrappers/interface/scoped_refptr.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-// Forward declaration.
-class FecPacket;
-
-/**
- * Performs codec-independent forward error correction (FEC), based on RFC 5109.
- * Option exists to enable unequal protection (UEP) across packets.
- * This is not to be confused with protection within packets
- * (referred to as uneven level protection (ULP) in RFC 5109).
- */
-class ForwardErrorCorrection {
- public:
- // Maximum number of media packets we can protect
- static const unsigned int kMaxMediaPackets = 48u;
-
- // TODO(holmer): As a next step all these struct-like packet classes should be
- // refactored into proper classes, and their members should be made private.
- // This will require parts of the functionality in forward_error_correction.cc
- // and receiver_fec.cc to be refactored into the packet classes.
- class Packet {
- public:
- Packet() : length(0), data(), ref_count_(0) {}
- virtual ~Packet() {}
-
- // Add a reference.
- virtual int32_t AddRef() {
- return ++ref_count_;
- }
-
- // Release a reference. Will delete the object if the reference count
- // reaches zero.
- virtual int32_t Release() {
- int32_t ref_count;
- ref_count = --ref_count_;
- if (ref_count == 0)
- delete this;
- return ref_count;
- }
-
- uint16_t length; // Length of packet in bytes.
- uint8_t data[IP_PACKET_SIZE]; // Packet data.
-
- private:
- int32_t ref_count_; // Counts the number of references to a packet.
- };
-
- // TODO(holmer): Refactor into a proper class.
- class SortablePacket {
- public:
- // True if first is <= than second.
- static bool LessThan(const SortablePacket* first,
- const SortablePacket* second);
-
- uint16_t seqNum;
- };
-
- /**
- * The received list parameter of #DecodeFEC() must reference structs of this
- * type. The lastMediaPktInFrame is not required to be used for correct
- * recovery, but will reduce delay by allowing #DecodeFEC() to pre-emptively
- * determine frame completion. If set, we assume a FEC stream, and the
- * following assumptions must hold:\n
- *
- * 1. The media packets in a frame have contiguous sequence numbers, i.e. the
- * frame's FEC packets have sequence numbers either lower than the first
- * media packet or higher than the last media packet.\n
- * 2. All FEC packets have a sequence number base equal to the first media
- * packet in the corresponding frame.\n
- *
- * The ssrc member is needed to ensure we can restore the SSRC field of
- * recovered packets. In most situations this could be retrieved from other
- * media packets, but in the case of an FEC packet protecting a single
- * missing media packet, we have no other means of obtaining it.
- */
- // TODO(holmer): Refactor into a proper class.
- class ReceivedPacket : public SortablePacket {
- public:
- uint32_t ssrc; // SSRC of the current frame. Must be set for FEC
- // packets, but not required for media packets.
- bool isFec; // Set to true if this is an FEC packet and false
- // otherwise.
- scoped_refptr<Packet> pkt; // Pointer to the packet storage.
- };
-
- /**
- * The recovered list parameter of #DecodeFEC() will reference structs of
- * this type.
- */
- // TODO(holmer): Refactor into a proper class.
- class RecoveredPacket : public SortablePacket {
- public:
- bool wasRecovered; // Will be true if this packet was recovered by
- // the FEC. Otherwise it was a media packet passed in
- // through the received packet list.
- bool returned; // True when the packet already has been returned to the
- // caller through the callback.
- uint8_t length_recovery[2]; // Two bytes used for recovering the packet
- // length with XOR operations.
- scoped_refptr<Packet> pkt; // Pointer to the packet storage.
- };
-
- typedef std::list<Packet*> PacketList;
- typedef std::list<ReceivedPacket*> ReceivedPacketList;
- typedef std::list<RecoveredPacket*> RecoveredPacketList;
-
- /**
- * \param[in] id Module ID
- */
- ForwardErrorCorrection(int32_t id);
-
- virtual ~ForwardErrorCorrection();
-
- /**
- * Generates a list of FEC packets from supplied media packets.
- *
- * \param[in] mediaPacketList List of media packets to protect, of type
- * #Packet. All packets must belong to the
- * same frame and the list must not be empty.
- * \param[in] protectionFactor FEC protection overhead in the [0, 255]
- * domain. To obtain 100% overhead, or an
- * equal number of FEC packets as media
- * packets, use 255.
- * \param[in] numImportantPackets The number of "important" packets in the
- * frame. These packets may receive greater
- * protection than the remaining packets. The
- * important packets must be located at the
- * start of the media packet list. For codecs
- * with data partitioning, the important
- * packets may correspond to first partition
- * packets.
- * \param[in] useUnequalProtection Parameter to enable/disable unequal
- * protection (UEP) across packets. Enabling
- * UEP will allocate more protection to the
- * numImportantPackets from the start of the
- * mediaPacketList.
- * \param[out] fecPacketList List of FEC packets, of type #Packet. Must
- * be empty on entry. The memory available
- * through the list will be valid until the
- * next call to GenerateFEC().
- *
- * \return 0 on success, -1 on failure.
- */
- int32_t GenerateFEC(const PacketList& mediaPacketList,
- uint8_t protectionFactor,
- int numImportantPackets,
- bool useUnequalProtection,
- PacketList* fecPacketList);
-
- /**
- * Decodes a list of media and FEC packets. It will parse the input received
- * packet list, storing FEC packets internally and inserting media packets to
- * the output recovered packet list. The recovered list will be sorted by
- * ascending sequence number and have duplicates removed. The function
- * should be called as new packets arrive, with the recovered list being
- * progressively assembled with each call. The received packet list will be
- * empty at output.\n
- *
- * The user will allocate packets submitted through the received list. The
- * function will handle allocation of recovered packets and optionally
- * deleting of all packet memory. The user may delete the recovered list
- * packets, in which case they must remove deleted packets from the
- * recovered list.\n
- *
- * \param[in] receivedPacketList List of new received packets, of type
- * #ReceivedPacket, belonging to a single
- * frame. At output the list will be empty,
- * with packets either stored internally,
- * or accessible through the recovered list.
- * \param[out] recoveredPacketList List of recovered media packets, of type
- * #RecoveredPacket, belonging to a single
- * frame. The memory available through the
- * list will be valid until the next call to
- * DecodeFEC().
- *
- * \return 0 on success, -1 on failure.
- */
- int32_t DecodeFEC(ReceivedPacketList* receivedPacketList,
- RecoveredPacketList* recoveredPacketList);
- /**
- * Gets the size in bytes of the FEC/ULP headers, which must be accounted for
- * as packet overhead.
- * \return Packet overhead in bytes.
- */
- static uint16_t PacketOverhead();
-
- // Reset internal states from last frame and clear the recoveredPacketList.
- // Frees all memory allocated by this class.
- void ResetState(RecoveredPacketList* recoveredPacketList);
-
- private:
- typedef std::list<FecPacket*> FecPacketList;
-
- void GenerateFecUlpHeaders(const PacketList& mediaPacketList,
- uint8_t* packetMask,
- uint32_t numFecPackets);
-
- void GenerateFecBitStrings(const PacketList& mediaPacketList,
- uint8_t* packetMask,
- uint32_t numFecPackets);
-
- // Insert received packets into FEC or recovered list.
- void InsertPackets(ReceivedPacketList* receivedPacketList,
- RecoveredPacketList* recoveredPacketList);
-
- // Insert media packet into recovered packet list. We delete duplicates.
- void InsertMediaPacket(ReceivedPacket* rxPacket,
- RecoveredPacketList* recoveredPacketList);
-
- // Assigns pointers to the recovered packet from all FEC packets which cover
- // it.
- // Note: This reduces the complexity when we want to try to recover a packet
- // since we don't have to find the intersection between recovered packets and
- // packets covered by the FEC packet.
- void UpdateCoveringFECPackets(RecoveredPacket* packet);
-
- // Insert packet into FEC list. We delete duplicates.
- void InsertFECPacket(ReceivedPacket* rxPacket,
- const RecoveredPacketList* recoveredPacketList);
-
- // Assigns pointers to already recovered packets covered by this FEC packet.
- static void AssignRecoveredPackets(
- FecPacket* fec_packet,
- const RecoveredPacketList* recovered_packets);
-
- // Insert into recovered list in correct position.
- void InsertRecoveredPacket(
- RecoveredPacket* recPacketToInsert,
- RecoveredPacketList* recoveredPacketList);
-
- // Attempt to recover missing packets.
- void AttemptRecover(RecoveredPacketList* recoveredPacketList);
-
- // Initializes the packet recovery using the FEC packet.
- static void InitRecovery(const FecPacket* fec_packet,
- RecoveredPacket* recovered);
-
- // Performs XOR between |src_packet| and |dst_packet| and stores the result
- // in |dst_packet|.
- static void XorPackets(const Packet* src_packet,
- RecoveredPacket* dst_packet);
-
- // Finish up the recovery of a packet.
- static void FinishRecovery(RecoveredPacket* recovered);
-
- // Recover a missing packet.
- void RecoverPacket(const FecPacket* fecPacket,
- RecoveredPacket* recPacketToInsert);
-
- // Get the number of missing media packets which are covered by this
- // FEC packet. An FEC packet can recover at most one packet, and if zero
- // packets are missing the FEC packet can be discarded.
- // This function returns 2 when two or more packets are missing.
- static int NumCoveredPacketsMissing(const FecPacket* fec_packet);
-
- static uint16_t LatestSequenceNumber(uint16_t first,
- uint16_t second);
-
- static void DiscardFECPacket(FecPacket* fec_packet);
- static void DiscardOldPackets(RecoveredPacketList* recoveredPacketList);
-
- int32_t _id;
- std::vector<Packet> _generatedFecPackets;
- FecPacketList _fecPacketList;
- bool _fecPacketReceived;
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_FORWARD_ERROR_CORRECTION_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/forward_error_correction_internal.cc b/trunk/src/modules/rtp_rtcp/source/forward_error_correction_internal.cc
deleted file mode 100644
index 5338174..0000000
--- a/trunk/src/modules/rtp_rtcp/source/forward_error_correction_internal.cc
+++ /dev/null
@@ -1,381 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "forward_error_correction_internal.h"
-#include "fec_private_tables.h"
-
-#include <cassert>
-#include <cstring>
-
-namespace {
-
-// Allow for different modes of protection for packets in UEP case.
-enum ProtectionMode
-{
- kModeNoOverlap,
- kModeOverlap,
- kModeBiasFirstPacket,
-};
-
-/**
- * Fits an input mask (subMask) to an output mask.
- * The mask is a matrix where the rows are the FEC packets,
- * and the columns are the source packets the FEC is applied to.
- * Each row of the mask is represented by a number of mask bytes.
- *
- * \param[in] numMaskBytes The number of mask bytes of output mask.
- * \param[in] numSubMaskBytes The number of mask bytes of input mask.
- * \param[in] numRows The number of rows of the input mask.
- * \param[in] subMask A pointer to hold the input mask, of size
- * [0, numRows * numSubMaskBytes]
- * \param[out] packetMask A pointer to hold the output mask, of size
- * [0, x * numMaskBytes], where x >= numRows.
- */
-void FitSubMask(int numMaskBytes,
- int numSubMaskBytes,
- int numRows,
- const uint8_t* subMask,
- uint8_t* packetMask)
-{
- if (numMaskBytes == numSubMaskBytes)
- {
- memcpy(packetMask, subMask, numRows * numSubMaskBytes);
- }
- else
- {
- for (int i = 0; i < numRows; i++)
- {
- int pktMaskIdx = i * numMaskBytes;
- int pktMaskIdx2 = i * numSubMaskBytes;
- for (int j = 0; j < numSubMaskBytes; j++)
- {
- packetMask[pktMaskIdx] = subMask[pktMaskIdx2];
- pktMaskIdx++;
- pktMaskIdx2++;
- }
- }
- }
-}
-
-/**
- * Shifts a mask by number of columns (bits), and fits it to an output mask.
- * The mask is a matrix where the rows are the FEC packets,
- * and the columns are the source packets the FEC is applied to.
- * Each row of the mask is represented by a number of mask bytes.
- *
- * \param[in] numMaskBytes The number of mask bytes of output mask.
- * \param[in] numSubMaskBytes The number of mask bytes of input mask.
- * \param[in] numColumnShift The number columns to be shifted, and
- * the starting row for the output mask.
- * \param[in] endRow The ending row for the output mask.
- * \param[in] subMask A pointer to hold the input mask, of size
- * [0, (endRowFEC - startRowFec) * numSubMaskBytes]
- * \param[out] packetMask A pointer to hold the output mask, of size
- * [0, x * numMaskBytes], where x >= endRowFEC.
- */
-// TODO (marpan): This function is doing three things at the same time:
-// shift within a byte, byte shift and resizing.
-// Split up into subroutines.
-void ShiftFitSubMask(int numMaskBytes,
- int resMaskBytes,
- int numColumnShift,
- int endRow,
- const uint8_t* subMask,
- uint8_t* packetMask)
-{
-
- // Number of bit shifts within a byte
- const int numBitShifts = (numColumnShift % 8);
- const int numByteShifts = numColumnShift >> 3;
-
- // Modify new mask with sub-mask21.
-
- // Loop over the remaining FEC packets.
- for (int i = numColumnShift; i < endRow; i++)
- {
- // Byte index of new mask, for row i and column resMaskBytes,
- // offset by the number of bytes shifts
- int pktMaskIdx = i * numMaskBytes + resMaskBytes - 1 + numByteShifts;
- // Byte index of subMask, for row i and column resMaskBytes
- int pktMaskIdx2 =
- (i - numColumnShift) * resMaskBytes + resMaskBytes - 1;
-
- uint8_t shiftRightCurrByte = 0;
- uint8_t shiftLeftPrevByte = 0;
- uint8_t combNewByte = 0;
-
- // Handle case of numMaskBytes > resMaskBytes:
- // For a given row, copy the rightmost "numBitShifts" bits
- // of the last byte of subMask into output mask.
- if (numMaskBytes > resMaskBytes)
- {
- shiftLeftPrevByte =
- (subMask[pktMaskIdx2] << (8 - numBitShifts));
- packetMask[pktMaskIdx + 1] = shiftLeftPrevByte;
- }
-
- // For each row i (FEC packet), shift the bit-mask of the subMask.
- // Each row of the mask contains "resMaskBytes" of bytes.
- // We start from the last byte of the subMask and move to first one.
- for (int j = resMaskBytes - 1; j > 0; j--)
- {
- // Shift current byte of sub21 to the right by "numBitShifts".
- shiftRightCurrByte =
- subMask[pktMaskIdx2] >> numBitShifts;
-
- // Fill in shifted bits with bits from the previous (left) byte:
- // First shift the previous byte to the left by "8-numBitShifts".
- shiftLeftPrevByte =
- (subMask[pktMaskIdx2 - 1] << (8 - numBitShifts));
-
- // Then combine both shifted bytes into new mask byte.
- combNewByte = shiftRightCurrByte | shiftLeftPrevByte;
-
- // Assign to new mask.
- packetMask[pktMaskIdx] = combNewByte;
- pktMaskIdx--;
- pktMaskIdx2--;
- }
- // For the first byte in the row (j=0 case).
- shiftRightCurrByte = subMask[pktMaskIdx2] >> numBitShifts;
- packetMask[pktMaskIdx] = shiftRightCurrByte;
-
- }
-}
-
-} //namespace
-
-namespace webrtc {
-namespace internal {
-
-// Remaining protection after important (first partition) packet protection
-void RemainingPacketProtection(int numMediaPackets,
- int numFecRemaining,
- int numFecForImpPackets,
- int numMaskBytes,
- ProtectionMode mode,
- uint8_t* packetMask)
-{
- if (mode == kModeNoOverlap)
- {
- // subMask21
-
- const int lBit =
- (numMediaPackets - numFecForImpPackets) > 16 ? 1 : 0;
-
- const int resMaskBytes =
- (lBit == 1) ? kMaskSizeLBitSet : kMaskSizeLBitClear;
-
- const uint8_t* packetMaskSub21 =
- packetMaskTbl[numMediaPackets - numFecForImpPackets - 1]
- [numFecRemaining - 1];
-
- ShiftFitSubMask(numMaskBytes, resMaskBytes, numFecForImpPackets,
- (numFecForImpPackets + numFecRemaining),
- packetMaskSub21, packetMask);
-
- }
- else if (mode == kModeOverlap || mode == kModeBiasFirstPacket)
- {
- // subMask22
-
- const uint8_t* packetMaskSub22 =
- packetMaskTbl[numMediaPackets - 1][numFecRemaining - 1];
-
- FitSubMask(numMaskBytes, numMaskBytes, numFecRemaining, packetMaskSub22,
- &packetMask[numFecForImpPackets * numMaskBytes]);
-
- if (mode == kModeBiasFirstPacket)
- {
- for (int i = 0; i < numFecRemaining; i++)
- {
- int pktMaskIdx = i * numMaskBytes;
- packetMask[pktMaskIdx] = packetMask[pktMaskIdx] | (1 << 7);
- }
- }
- }
- else
- {
- assert(false);
- }
-
-}
-
-// Protection for important (first partition) packets
-void ImportantPacketProtection(int numFecForImpPackets,
- int numImpPackets,
- int numMaskBytes,
- uint8_t* packetMask)
-{
- const int lBit = numImpPackets > 16 ? 1 : 0;
- const int numImpMaskBytes =
- (lBit == 1) ? kMaskSizeLBitSet : kMaskSizeLBitClear;
-
- // Get subMask1 from table
- const uint8_t* packetMaskSub1 =
- packetMaskTbl[numImpPackets - 1][numFecForImpPackets - 1];
-
- FitSubMask(numMaskBytes, numImpMaskBytes,
- numFecForImpPackets, packetMaskSub1, packetMask);
-
-}
-
-// This function sets the protection allocation: i.e., how many FEC packets
-// to use for numImp (1st partition) packets, given the: number of media
-// packets, number of FEC packets, and number of 1st partition packets.
-int SetProtectionAllocation(int numMediaPackets,
- int numFecPackets,
- int numImpPackets)
-{
-
- // TODO (marpan): test different cases for protection allocation:
-
- // Use at most (allocPar * numFecPackets) for important packets.
- float allocPar = 0.5;
- int maxNumFecForImp = allocPar * numFecPackets;
-
- int numFecForImpPackets = (numImpPackets < maxNumFecForImp) ?
- numImpPackets : maxNumFecForImp;
-
- // Fall back to equal protection in this case
- if (numFecPackets == 1 && (numMediaPackets > 2 * numImpPackets))
- {
- numFecForImpPackets = 0;
- }
-
- return numFecForImpPackets;
-}
-
-// Modification for UEP: reuse the off-line tables for the packet masks.
-// Note: these masks were designed for equal packet protection case,
-// assuming random packet loss.
-
-// Current version has 3 modes (options) to build UEP mask from existing ones.
-// Various other combinations may be added in future versions.
-// Longer-term, we may add another set of tables specifically for UEP cases.
-// TODO (marpan): also consider modification of masks for bursty loss cases.
-
-// Mask is characterized as (#packets_to_protect, #fec_for_protection).
-// Protection factor defined as: (#fec_for_protection / #packets_to_protect).
-
-// Let k=numMediaPackets, n=total#packets, (n-k)=numFecPackets, m=numImpPackets.
-
-// For ProtectionMode 0 and 1:
-// one mask (subMask1) is used for 1st partition packets,
-// the other mask (subMask21/22, for 0/1) is for the remaining FEC packets.
-
-// In both mode 0 and 1, the packets of 1st partition (numImpPackets) are
-// treated equally important, and are afforded more protection than the
-// residual partition packets.
-
-// For numImpPackets:
-// subMask1 = (m, t): protection = t/(m), where t=F(k,n-k,m).
-// t=F(k,n-k,m) is the number of packets used to protect first partition in
-// subMask1. This is determined from the function SetProtectionAllocation().
-
-// For the left-over protection:
-// Mode 0: subMask21 = (k-m,n-k-t): protection = (n-k-t)/(k-m)
-// mode 0 has no protection overlap between the two partitions.
-// For mode 0, we would typically set t = min(m, n-k).
-
-
-// Mode 1: subMask22 = (k, n-k-t), with protection (n-k-t)/(k)
-// mode 1 has protection overlap between the two partitions (preferred).
-
-// For ProtectionMode 2:
-// This gives 1st packet of list (which is 1st packet of 1st partition) more
-// protection. In mode 2, the equal protection mask (which is obtained from
-// mode 1 for t=0) is modified (more "1s" added in 1st column of packet mask)
-// to bias higher protection for the 1st source packet.
-
-// Protection Mode 2 may be extended for a sort of sliding protection
-// (i.e., vary the number/density of "1s" across columns) across packets.
-
-void UnequalProtectionMask(int numMediaPackets,
- int numFecPackets,
- int numImpPackets,
- int numMaskBytes,
- uint8_t* packetMask)
-{
-
- // Set Protection type and allocation
- // TODO (marpan): test/update for best mode and some combinations thereof.
-
- ProtectionMode mode = kModeOverlap;
- int numFecForImpPackets = 0;
-
- if (mode != kModeBiasFirstPacket)
- {
- numFecForImpPackets = SetProtectionAllocation(numMediaPackets,
- numFecPackets,
- numImpPackets);
- }
-
- int numFecRemaining = numFecPackets - numFecForImpPackets;
- // Done with setting protection type and allocation
-
- //
- // Generate subMask1
- //
- if (numFecForImpPackets > 0)
- {
- ImportantPacketProtection(numFecForImpPackets, numImpPackets,
- numMaskBytes, packetMask);
- }
-
- //
- // Generate subMask2
- //
- if (numFecRemaining > 0)
- {
- RemainingPacketProtection(numMediaPackets, numFecRemaining,
- numFecForImpPackets, numMaskBytes,
- mode, packetMask);
- }
-
-}
-
-void GeneratePacketMasks(int numMediaPackets,
- int numFecPackets,
- int numImpPackets,
- bool useUnequalProtection,
- uint8_t* packetMask)
-{
- assert(numMediaPackets <= static_cast<int>(sizeof(packetMaskTbl) /
- sizeof(*packetMaskTbl)));
- assert(numMediaPackets > 0);
- assert(numFecPackets <= numMediaPackets && numFecPackets > 0);
- assert(numImpPackets <= numMediaPackets && numImpPackets >= 0);
-
- int lBit = numMediaPackets > 16 ? 1 : 0;
- const int numMaskBytes =
- (lBit == 1) ? kMaskSizeLBitSet : kMaskSizeLBitClear;
-
- // Equal-protection for these cases
- if (!useUnequalProtection || numImpPackets == 0)
- {
- // Retrieve corresponding mask table directly:for equal-protection case.
- // Mask = (k,n-k), with protection factor = (n-k)/k,
- // where k = numMediaPackets, n=total#packets, (n-k)=numFecPackets.
- memcpy(packetMask,
- packetMaskTbl[numMediaPackets - 1][numFecPackets - 1],
- numFecPackets * numMaskBytes);
- }
- else //UEP case
- {
- UnequalProtectionMask(numMediaPackets, numFecPackets, numImpPackets,
- numMaskBytes, packetMask);
-
- } // End of UEP modification
-
-} //End of GetPacketMasks
-
-} // namespace internal
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/forward_error_correction_internal.h b/trunk/src/modules/rtp_rtcp/source/forward_error_correction_internal.h
deleted file mode 100644
index e137ef2..0000000
--- a/trunk/src/modules/rtp_rtcp/source/forward_error_correction_internal.h
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "typedefs.h"
-
-namespace webrtc {
-
-// Packet mask size in bytes (L bit is set).
-static const int kMaskSizeLBitSet = 6;
-// Packet mask size in bytes (L bit is cleared).
-static const int kMaskSizeLBitClear = 2;
-
-namespace internal {
-
- /**
- * Returns an array of packet masks. The mask of a single FEC packet
- * corresponds to a number of mask bytes. The mask indicates which
- * media packets should be protected by the FEC packet.
-
- * \param[in] numMediaPackets The number of media packets to protect.
- * [1, maxMediaPackets].
- * \param[in] numFecPackets The number of FEC packets which will
- * be generated. [1, numMediaPackets].
- * \param[in] numImpPackets The number of important packets.
- * [0, numMediaPackets].
- * numImpPackets = 0 is the equal
- * protection scenario.
- * \param[in] useUnequalProtection Enables unequal protection: allocates
- * more protection to the numImpPackets.
- * \param[out] packetMask A pointer to hold the packet mask array,
- * of size:
- * numFecPackets * "number of mask bytes".
- */
-void GeneratePacketMasks(int numMediaPackets,
- int numFecPackets,
- int numImpPackets,
- bool useUnequalProtection,
- uint8_t* packetMask);
-
-} // namespace internal
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h b/trunk/src/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h
deleted file mode 100644
index cb7ebba..0000000
--- a/trunk/src/modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/rtp_rtcp/source/rtp_receiver_video.h"
-
-namespace webrtc {
-
-class MockRTPReceiverVideo : public RTPReceiverVideo {
- public:
- MOCK_METHOD1(ChangeUniqueId,
- void(const WebRtc_Word32 id));
- MOCK_METHOD3(ReceiveRecoveredPacketCallback,
- WebRtc_Word32(WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadDataLength));
- MOCK_METHOD3(CallbackOfReceivedPayloadData,
- WebRtc_Word32(const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const WebRtcRTPHeader* rtpHeader));
- MOCK_CONST_METHOD0(TimeStamp,
- WebRtc_UWord32());
- MOCK_CONST_METHOD0(SequenceNumber,
- WebRtc_UWord16());
- MOCK_CONST_METHOD2(PayloadTypeToPayload,
- WebRtc_UWord32(const WebRtc_UWord8 payloadType,
- ModuleRTPUtility::Payload*& payload));
- MOCK_CONST_METHOD2(RetransmitOfOldPacket,
- bool(const WebRtc_UWord16 sequenceNumber,
- const WebRtc_UWord32 rtpTimeStamp));
- MOCK_CONST_METHOD0(REDPayloadType,
- WebRtc_Word8());
-};
-
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/overuse_detector.cc b/trunk/src/modules/rtp_rtcp/source/overuse_detector.cc
deleted file mode 100644
index 4cffb0f..0000000
--- a/trunk/src/modules/rtp_rtcp/source/overuse_detector.cc
+++ /dev/null
@@ -1,450 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#if _WIN32
-#include <windows.h>
-#endif
-
-#include "trace.h"
-#include "overuse_detector.h"
-#include "remote_rate_control.h"
-#include "rtp_utility.h"
-#include <math.h>
-#include <stdlib.h> //abs
-
-#ifdef WEBRTC_BWE_MATLAB
-extern MatlabEngine eng; // global variable defined elsewhere
-#endif
-
-#define INIT_CAPACITY_SLOPE 8.0/512.0
-#define DETECTOR_THRESHOLD 25.0
-#define OVER_USING_TIME_THRESHOLD 100
-#define MIN_FRAME_PERIOD_HISTORY_LEN 60
-
-namespace webrtc {
-OverUseDetector::OverUseDetector()
-:
-_firstPacket(true),
-_currentFrame(),
-_prevFrame(),
-_numOfDeltas(0),
-_slope(INIT_CAPACITY_SLOPE),
-_offset(0),
-_E(),
-_processNoise(),
-_avgNoise(0.0),
-_varNoise(500),
-_threshold(DETECTOR_THRESHOLD),
-_tsDeltaHist(),
-_prevOffset(0.0),
-_timeOverUsing(-1),
-_overUseCounter(0),
-_hypothesis(kBwNormal)
-#ifdef WEBRTC_BWE_MATLAB
-,_plot1(NULL),
-_plot2(NULL),
-_plot3(NULL),
-_plot4(NULL)
-#endif
-{
- _E[0][0] = 100;
- _E[1][1] = 1e-1;
- _E[0][1] = _E[1][0] = 0;
- _processNoise[0] = 1e-10;
- _processNoise[1] = 1e-2;
-}
-
-OverUseDetector::~OverUseDetector()
-{
-#ifdef WEBRTC_BWE_MATLAB
- if (_plot1)
- {
- eng.DeletePlot(_plot1);
- _plot1 = NULL;
- }
- if (_plot2)
- {
- eng.DeletePlot(_plot2);
- _plot2 = NULL;
- }
- if (_plot3)
- {
- eng.DeletePlot(_plot3);
- _plot3 = NULL;
- }
- if (_plot4)
- {
- eng.DeletePlot(_plot4);
- _plot4 = NULL;
- }
-#endif
-
- _tsDeltaHist.clear();
-}
-
-void OverUseDetector::Reset()
-{
- _firstPacket = true;
- _currentFrame._size = 0;
- _currentFrame._completeTimeMs = -1;
- _currentFrame._timestamp = -1;
- _prevFrame._size = 0;
- _prevFrame._completeTimeMs = -1;
- _prevFrame._timestamp = -1;
- _numOfDeltas = 0;
- _slope = INIT_CAPACITY_SLOPE;
- _offset = 0;
- _E[0][0] = 100;
- _E[1][1] = 1e-1;
- _E[0][1] = _E[1][0] = 0;
- _processNoise[0] = 1e-10;
- _processNoise[1] = 1e-2;
- _avgNoise = 0.0;
- _varNoise = 500;
- _threshold = DETECTOR_THRESHOLD;
- _prevOffset = 0.0;
- _timeOverUsing = -1;
- _overUseCounter = 0;
- _hypothesis = kBwNormal;
- _tsDeltaHist.clear();
-}
-
-bool OverUseDetector::Update(const WebRtcRTPHeader& rtpHeader,
- const WebRtc_UWord16 packetSize,
- const WebRtc_Word64 nowMS)
-{
-#ifdef WEBRTC_BWE_MATLAB
- // Create plots
- const WebRtc_Word64 startTimeMs = nowMS;
- if (_plot1 == NULL)
- {
- _plot1 = eng.NewPlot(new MatlabPlot());
- _plot1->AddLine(1000, "b.", "scatter");
- }
- if (_plot2 == NULL)
- {
- _plot2 = eng.NewPlot(new MatlabPlot());
- _plot2->AddTimeLine(30, "b", "offset", startTimeMs);
- _plot2->AddTimeLine(30, "r--", "limitPos", startTimeMs);
- _plot2->AddTimeLine(30, "k.", "trigger", startTimeMs);
- _plot2->AddTimeLine(30, "ko", "detection", startTimeMs);
- //_plot2->AddTimeLine(30, "g", "slowMean", startTimeMs);
- }
- if (_plot3 == NULL)
- {
- _plot3 = eng.NewPlot(new MatlabPlot());
- _plot3->AddTimeLine(30, "b", "noiseVar", startTimeMs);
- }
- if (_plot4 == NULL)
- {
- _plot4 = eng.NewPlot(new MatlabPlot());
- //_plot4->AddTimeLine(60, "b", "p11", startTimeMs);
- //_plot4->AddTimeLine(60, "r", "p12", startTimeMs);
- _plot4->AddTimeLine(60, "g", "p22", startTimeMs);
- //_plot4->AddTimeLine(60, "g--", "p22_hat", startTimeMs);
- //_plot4->AddTimeLine(30, "b.-", "deltaFs", startTimeMs);
- }
-
-#endif
-
- bool wrapped = false;
- bool completeFrame = false;
- if (_currentFrame._timestamp == -1)
- {
- _currentFrame._timestamp = rtpHeader.header.timestamp;
- }
- else if (ModuleRTPUtility::OldTimestamp(
- rtpHeader.header.timestamp,
- static_cast<WebRtc_UWord32>(_currentFrame._timestamp),
- &wrapped))
- {
- // Don't update with old data
- return completeFrame;
- }
- else if (rtpHeader.header.timestamp != _currentFrame._timestamp)
- {
- // First packet of a later frame, the previous frame sample is ready
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "Frame complete at %I64i", _currentFrame._completeTimeMs);
- if (_prevFrame._completeTimeMs >= 0) // This is our second frame
- {
- WebRtc_Word64 tDelta = 0;
- double tsDelta = 0;
- // Check for wrap
- ModuleRTPUtility::OldTimestamp(
- static_cast<WebRtc_UWord32>(_prevFrame._timestamp),
- static_cast<WebRtc_UWord32>(_currentFrame._timestamp),
- &wrapped);
- CompensatedTimeDelta(_currentFrame, _prevFrame, tDelta, tsDelta, wrapped);
- UpdateKalman(tDelta, tsDelta, _currentFrame._size, _prevFrame._size);
- }
- // The new timestamp is now the current frame,
- // and the old timestamp becomes the previous frame.
- _prevFrame = _currentFrame;
- _currentFrame._timestamp = rtpHeader.header.timestamp;
- _currentFrame._size = 0;
- _currentFrame._completeTimeMs = -1;
- completeFrame = true;
- }
- // Accumulate the frame size
- _currentFrame._size += packetSize;
- _currentFrame._completeTimeMs = nowMS;
- return completeFrame;
-}
-
-BandwidthUsage OverUseDetector::State() const
-{
- return _hypothesis;
-}
-
-double OverUseDetector::NoiseVar() const
-{
- return _varNoise;
-}
-
-void OverUseDetector::SetRateControlRegion(RateControlRegion region)
-{
- switch (region)
- {
- case kRcMaxUnknown:
- {
- _threshold = DETECTOR_THRESHOLD;
- break;
- }
- case kRcAboveMax:
- case kRcNearMax:
- {
- _threshold = DETECTOR_THRESHOLD / 2;
- break;
- }
- }
-}
-
-void OverUseDetector::CompensatedTimeDelta(const FrameSample& currentFrame, const FrameSample& prevFrame, WebRtc_Word64& tDelta,
- double& tsDelta, bool wrapped)
-{
- _numOfDeltas++;
- if (_numOfDeltas > 1000)
- {
- _numOfDeltas = 1000;
- }
- // Add wrap-around compensation
- WebRtc_Word64 wrapCompensation = 0;
- if (wrapped)
- {
- wrapCompensation = static_cast<WebRtc_Word64>(1)<<32;
- }
- tsDelta = (currentFrame._timestamp + wrapCompensation - prevFrame._timestamp) / 90.0;
- tDelta = currentFrame._completeTimeMs - prevFrame._completeTimeMs;
- assert(tsDelta > 0);
-}
-
-double OverUseDetector::CurrentDrift()
-{
- return 1.0;
-}
-
-void OverUseDetector::UpdateKalman(WebRtc_Word64 tDelta, double tsDelta, WebRtc_UWord32 frameSize, WebRtc_UWord32 prevFrameSize)
-{
- const double minFramePeriod = UpdateMinFramePeriod(tsDelta);
- const double drift = CurrentDrift();
- // Compensate for drift
- const double tTsDelta = tDelta - tsDelta / drift;
- double fsDelta = static_cast<double>(frameSize) - prevFrameSize;
-
- // Update the Kalman filter
- const double scaleFactor = minFramePeriod / (1000.0 / 30.0);
- _E[0][0] += _processNoise[0] * scaleFactor;
- _E[1][1] += _processNoise[1] * scaleFactor;
-
- if ((_hypothesis == kBwOverusing && _offset < _prevOffset) ||
- (_hypothesis == kBwUnderUsing && _offset > _prevOffset))
- {
- _E[1][1] += 10 * _processNoise[1] * scaleFactor;
- }
-
- const double h[2] = {fsDelta, 1.0};
- const double Eh[2] = {_E[0][0]*h[0] + _E[0][1]*h[1],
- _E[1][0]*h[0] + _E[1][1]*h[1]};
-
- const double residual = tTsDelta - _slope*h[0] - _offset;
-
- const bool stableState = (BWE_MIN(_numOfDeltas, 60) * abs(_offset) < _threshold);
- // We try to filter out very late frames. For instance periodic key
- // frames doesn't fit the Gaussian model well.
- if (abs(residual) < 3 * sqrt(_varNoise))
- {
- UpdateNoiseEstimate(residual, minFramePeriod, stableState);
- }
- else
- {
- UpdateNoiseEstimate(3 * sqrt(_varNoise), minFramePeriod, stableState);
- }
-
- const double denom = _varNoise + h[0]*Eh[0] + h[1]*Eh[1];
-
- const double K[2] = {Eh[0] / denom,
- Eh[1] / denom};
-
- const double IKh[2][2] = {{1.0 - K[0]*h[0], -K[0]*h[1]},
- {-K[1]*h[0], 1.0 - K[1]*h[1]}};
- const double e00 = _E[0][0];
- const double e01 = _E[0][1];
-
- // Update state
- _E[0][0] = e00 * IKh[0][0] + _E[1][0] * IKh[0][1];
- _E[0][1] = e01 * IKh[0][0] + _E[1][1] * IKh[0][1];
- _E[1][0] = e00 * IKh[1][0] + _E[1][0] * IKh[1][1];
- _E[1][1] = e01 * IKh[1][0] + _E[1][1] * IKh[1][1];
-
- // Covariance matrix, must be positive semi-definite
- assert(_E[0][0] + _E[1][1] >= 0 &&
- _E[0][0] * _E[1][1] - _E[0][1] * _E[1][0] >= 0 &&
- _E[0][0] >= 0);
-
-#ifdef WEBRTC_BWE_MATLAB
- //_plot4->Append("p11",_E[0][0]);
- //_plot4->Append("p12",_E[0][1]);
- _plot4->Append("p22",_E[1][1]);
- //_plot4->Append("p22_hat", 0.5*(_processNoise[1] +
- // sqrt(_processNoise[1]*(_processNoise[1] + 4*_varNoise))));
- //_plot4->Append("deltaFs", fsDelta);
- _plot4->Plot();
-#endif
- _slope = _slope + K[0] * residual;
- _prevOffset = _offset;
- _offset = _offset + K[1] * residual;
-
- Detect(tsDelta);
-
-#ifdef WEBRTC_BWE_MATLAB
- _plot1->Append("scatter", static_cast<double>(_currentFrame._size) - _prevFrame._size,
- static_cast<double>(tDelta-tsDelta));
- _plot1->MakeTrend("scatter", "slope", _slope, _offset, "k-");
- _plot1->MakeTrend("scatter", "thresholdPos", _slope, _offset + 2 * sqrt(_varNoise), "r-");
- _plot1->MakeTrend("scatter", "thresholdNeg", _slope, _offset - 2 * sqrt(_varNoise), "r-");
- _plot1->Plot();
-
- _plot2->Append("offset", _offset);
- _plot2->Append("limitPos", _threshold/BWE_MIN(_numOfDeltas, 60));
- _plot2->Plot();
-
- _plot3->Append("noiseVar", _varNoise);
- _plot3->Plot();
-#endif
-}
-
-double OverUseDetector::UpdateMinFramePeriod(double tsDelta) {
- double minFramePeriod = tsDelta;
- if (_tsDeltaHist.size() >= MIN_FRAME_PERIOD_HISTORY_LEN) {
- std::list<double>::iterator firstItem = _tsDeltaHist.begin();
- _tsDeltaHist.erase(firstItem);
- }
- std::list<double>::iterator it = _tsDeltaHist.begin();
- for (; it != _tsDeltaHist.end(); it++) {
- minFramePeriod = BWE_MIN(*it, minFramePeriod);
- }
- _tsDeltaHist.push_back(tsDelta);
- return minFramePeriod;
-}
-
-void OverUseDetector::UpdateNoiseEstimate(double residual, double tsDelta, bool stableState)
-{
- if (!stableState)
- {
- return;
- }
- // Faster filter during startup to faster adapt to the jitter level of the network
- // alpha is tuned for 30 frames per second, but
- double alpha = 0.01;
- if (_numOfDeltas > 10*30)
- {
- alpha = 0.002;
- }
- // Only update the noise estimate if we're not over-using
- // beta is a function of alpha and the time delta since
- // the previous update.
- const double beta = pow(1 - alpha, tsDelta * 30.0 / 1000.0);
- _avgNoise = beta * _avgNoise + (1 - beta) * residual;
- _varNoise = beta * _varNoise + (1 - beta) * (_avgNoise - residual) * (_avgNoise - residual);
- if (_varNoise < 1e-7)
- {
- _varNoise = 1e-7;
- }
-}
-
-BandwidthUsage OverUseDetector::Detect(double tsDelta)
-{
- if (_numOfDeltas < 2)
- {
- return kBwNormal;
- }
- const double T = BWE_MIN(_numOfDeltas, 60) * _offset;
- if (abs(T) > _threshold)
- {
- if (_offset > 0)
- {
- if (_timeOverUsing == -1)
- {
- // Initialize the timer. Assume that we've been
- // over-using half of the time since the previous
- // sample.
- _timeOverUsing = tsDelta / 2;
- }
- else
- {
- // Increment timer
- _timeOverUsing += tsDelta;
- }
- _overUseCounter++;
- if (_timeOverUsing > OVER_USING_TIME_THRESHOLD && _overUseCounter > 1)
- {
- if (_offset >= _prevOffset)
- {
-#ifdef _DEBUG
- if (_hypothesis != kBwOverusing)
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: kBwOverusing");
-#endif
- _timeOverUsing = 0;
- _overUseCounter = 0;
- _hypothesis = kBwOverusing;
-#ifdef WEBRTC_BWE_MATLAB
- _plot2->Append("detection",_offset); // plot it later
-#endif
- }
- }
-#ifdef WEBRTC_BWE_MATLAB
- _plot2->Append("trigger",_offset); // plot it later
-#endif
- }
- else
- {
-#ifdef _DEBUG
- if (_hypothesis != kBwUnderUsing)
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: kBwUnderUsing");
-#endif
- _timeOverUsing = -1;
- _overUseCounter = 0;
- _hypothesis = kBwUnderUsing;
- }
- }
- else
- {
-#ifdef _DEBUG
- if (_hypothesis != kBwNormal)
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: kBwNormal");
-#endif
- _timeOverUsing = -1;
- _overUseCounter = 0;
- _hypothesis = kBwNormal;
- }
- return _hypothesis;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/overuse_detector.h b/trunk/src/modules/rtp_rtcp/source/overuse_detector.h
deleted file mode 100644
index b250e01..0000000
--- a/trunk/src/modules/rtp_rtcp/source/overuse_detector.h
+++ /dev/null
@@ -1,90 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_OVERUSE_DETECTOR_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_OVERUSE_DETECTOR_H_
-
-#include <list>
-
-#include "bwe_defines.h"
-#include "module_common_types.h"
-#include "typedefs.h"
-
-#ifdef WEBRTC_BWE_MATLAB
-#include "../test/BWEStandAlone/MatlabPlot.h"
-#endif
-
-namespace webrtc {
-enum RateControlRegion;
-
-class OverUseDetector
-{
-public:
- OverUseDetector();
- ~OverUseDetector();
- bool Update(const WebRtcRTPHeader& rtpHeader,
- const WebRtc_UWord16 packetSize,
- const WebRtc_Word64 nowMS);
- BandwidthUsage State() const;
- void Reset();
- double NoiseVar() const;
- void SetRateControlRegion(RateControlRegion region);
-
-private:
- struct FrameSample
- {
- FrameSample() : _size(0), _completeTimeMs(-1), _timestamp(-1) {}
-
- WebRtc_UWord32 _size;
- WebRtc_Word64 _completeTimeMs;
- WebRtc_Word64 _timestamp;
- };
-
- void CompensatedTimeDelta(const FrameSample& currentFrame,
- const FrameSample& prevFrame,
- WebRtc_Word64& tDelta,
- double& tsDelta,
- bool wrapped);
- void UpdateKalman(WebRtc_Word64 tDelta,
- double tsDelta,
- WebRtc_UWord32 frameSize,
- WebRtc_UWord32 prevFrameSize);
- double UpdateMinFramePeriod(double tsDelta);
- void UpdateNoiseEstimate(double residual, double tsDelta, bool stableState);
- BandwidthUsage Detect(double tsDelta);
- double CurrentDrift();
-
- bool _firstPacket;
- FrameSample _currentFrame;
- FrameSample _prevFrame;
- WebRtc_UWord16 _numOfDeltas;
- double _slope;
- double _offset;
- double _E[2][2];
- double _processNoise[2];
- double _avgNoise;
- double _varNoise;
- double _threshold;
- std::list<double> _tsDeltaHist;
- double _prevOffset;
- double _timeOverUsing;
- WebRtc_UWord16 _overUseCounter;
- BandwidthUsage _hypothesis;
-
-#ifdef WEBRTC_BWE_MATLAB
- MatlabPlot* _plot1;
- MatlabPlot* _plot2;
- MatlabPlot* _plot3;
- MatlabPlot* _plot4;
-#endif
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_OVERUSE_DETECTOR_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/receiver_fec.cc b/trunk/src/modules/rtp_rtcp/source/receiver_fec.cc
deleted file mode 100644
index 77a3465..0000000
--- a/trunk/src/modules/rtp_rtcp/source/receiver_fec.cc
+++ /dev/null
@@ -1,262 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/rtp_rtcp/source/receiver_fec.h"
-
-#include <cassert>
-
-#include "modules/rtp_rtcp/source/rtp_receiver_video.h"
-#include "modules/rtp_rtcp/source/rtp_utility.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-#include "system_wrappers/interface/trace.h"
-
-// RFC 5109
-namespace webrtc {
-ReceiverFEC::ReceiverFEC(const WebRtc_Word32 id, RTPReceiverVideo* owner)
- : _id(id),
- _owner(owner),
- _fec(new ForwardErrorCorrection(id)),
- _payloadTypeFEC(-1) {
-}
-
-ReceiverFEC::~ReceiverFEC() {
- // Clean up DecodeFEC()
- while (!_receivedPacketList.empty()){
- ForwardErrorCorrection::ReceivedPacket* receivedPacket =
- _receivedPacketList.front();
- delete receivedPacket;
- _receivedPacketList.pop_front();
- }
- assert(_receivedPacketList.empty());
-
- if (_fec != NULL) {
- _fec->ResetState(&_recoveredPacketList);
- delete _fec;
- }
-}
-
-void ReceiverFEC::SetPayloadTypeFEC(const WebRtc_Word8 payloadType) {
- _payloadTypeFEC = payloadType;
-}
-
-/*
- 0 1 2 3
- 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- |F| block PT | timestamp offset | block length |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
-
-RFC 2198 RTP Payload for Redundant Audio Data September 1997
-
- The bits in the header are specified as follows:
-
- F: 1 bit First bit in header indicates whether another header block
- follows. If 1 further header blocks follow, if 0 this is the
- last header block.
- If 0 there is only 1 byte RED header
-
- block PT: 7 bits RTP payload type for this block.
-
- timestamp offset: 14 bits Unsigned offset of timestamp of this block
- relative to timestamp given in RTP header. The use of an unsigned
- offset implies that redundant data must be sent after the primary
- data, and is hence a time to be subtracted from the current
- timestamp to determine the timestamp of the data for which this
- block is the redundancy.
-
- block length: 10 bits Length in bytes of the corresponding data
- block excluding header.
-*/
-
-WebRtc_Word32 ReceiverFEC::AddReceivedFECPacket(
- const WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord8* incomingRtpPacket,
- const WebRtc_UWord16 payloadDataLength,
- bool& FECpacket) {
- if (_payloadTypeFEC == -1) {
- return -1;
- }
-
- WebRtc_UWord8 REDHeaderLength = 1;
-
- // Add to list without RED header, aka a virtual RTP packet
- // we remove the RED header
-
- ForwardErrorCorrection::ReceivedPacket* receivedPacket =
- new ForwardErrorCorrection::ReceivedPacket;
- receivedPacket->pkt = new ForwardErrorCorrection::Packet;
-
- // get payload type from RED header
- WebRtc_UWord8 payloadType =
- incomingRtpPacket[rtpHeader->header.headerLength] & 0x7f;
-
- // use the payloadType to decide if it's FEC or coded data
- if (_payloadTypeFEC == payloadType) {
- receivedPacket->isFec = true;
- FECpacket = true;
- } else {
- receivedPacket->isFec = false;
- FECpacket = false;
- }
- receivedPacket->seqNum = rtpHeader->header.sequenceNumber;
-
- WebRtc_UWord16 blockLength = 0;
- if(incomingRtpPacket[rtpHeader->header.headerLength] & 0x80) {
- // f bit set in RED header
- REDHeaderLength = 4;
- WebRtc_UWord16 timestampOffset =
- (incomingRtpPacket[rtpHeader->header.headerLength + 1]) << 8;
- timestampOffset += incomingRtpPacket[rtpHeader->header.headerLength+2];
- timestampOffset = timestampOffset >> 2;
- if(timestampOffset != 0) {
- // |timestampOffset| should be 0. However, it's possible this is the first
- // location a corrupt payload can be caught, so don't assert.
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
- "Corrupt payload found in %s", __FUNCTION__);
- delete receivedPacket;
- return -1;
- }
-
- blockLength =
- (0x03 & incomingRtpPacket[rtpHeader->header.headerLength + 2]) << 8;
- blockLength += (incomingRtpPacket[rtpHeader->header.headerLength + 3]);
-
- // check next RED header
- if(incomingRtpPacket[rtpHeader->header.headerLength+4] & 0x80) {
- // more than 2 blocks in packet not supported
- delete receivedPacket;
- assert(false);
- return -1;
- }
- if(blockLength > payloadDataLength - REDHeaderLength) {
- // block length longer than packet
- delete receivedPacket;
- assert(false);
- return -1;
- }
- }
-
- ForwardErrorCorrection::ReceivedPacket* secondReceivedPacket = NULL;
- if (blockLength > 0) {
- // handle block length, split into 2 packets
- REDHeaderLength = 5;
-
- // copy the RTP header
- memcpy(receivedPacket->pkt->data,
- incomingRtpPacket,
- rtpHeader->header.headerLength);
-
- // replace the RED payload type
- receivedPacket->pkt->data[1] &= 0x80; // reset the payload
- receivedPacket->pkt->data[1] += payloadType; // set the media payload type
-
- // copy the payload data
- memcpy(receivedPacket->pkt->data + rtpHeader->header.headerLength,
- incomingRtpPacket + rtpHeader->header.headerLength + REDHeaderLength,
- blockLength);
-
- receivedPacket->pkt->length = blockLength;
-
- secondReceivedPacket = new ForwardErrorCorrection::ReceivedPacket;
- secondReceivedPacket->pkt = new ForwardErrorCorrection::Packet;
-
- secondReceivedPacket->isFec = true;
- secondReceivedPacket->seqNum = rtpHeader->header.sequenceNumber;
-
- // copy the FEC payload data
- memcpy(secondReceivedPacket->pkt->data,
- incomingRtpPacket + rtpHeader->header.headerLength +
- REDHeaderLength + blockLength,
- payloadDataLength - REDHeaderLength - blockLength);
-
- secondReceivedPacket->pkt->length = payloadDataLength - REDHeaderLength -
- blockLength;
-
- } else if(receivedPacket->isFec) {
- // everything behind the RED header
- memcpy(receivedPacket->pkt->data,
- incomingRtpPacket + rtpHeader->header.headerLength + REDHeaderLength,
- payloadDataLength - REDHeaderLength);
- receivedPacket->pkt->length = payloadDataLength - REDHeaderLength;
- receivedPacket->ssrc =
- ModuleRTPUtility::BufferToUWord32(&incomingRtpPacket[8]);
-
- } else {
- // copy the RTP header
- memcpy(receivedPacket->pkt->data,
- incomingRtpPacket,
- rtpHeader->header.headerLength);
-
- // replace the RED payload type
- receivedPacket->pkt->data[1] &= 0x80; // reset the payload
- receivedPacket->pkt->data[1] += payloadType; // set the media payload type
-
- // copy the media payload data
- memcpy(receivedPacket->pkt->data + rtpHeader->header.headerLength,
- incomingRtpPacket + rtpHeader->header.headerLength + REDHeaderLength,
- payloadDataLength - REDHeaderLength);
-
- receivedPacket->pkt->length = rtpHeader->header.headerLength +
- payloadDataLength - REDHeaderLength;
- }
-
- if(receivedPacket->pkt->length == 0) {
- delete secondReceivedPacket;
- delete receivedPacket;
- return 0;
- }
-
- _receivedPacketList.push_back(receivedPacket);
- if (secondReceivedPacket) {
- _receivedPacketList.push_back(secondReceivedPacket);
- }
- return 0;
-}
-
-WebRtc_Word32 ReceiverFEC::ProcessReceivedFEC() {
- if (!_receivedPacketList.empty()) {
- if (_fec->DecodeFEC(&_receivedPacketList, &_recoveredPacketList) != 0) {
- return -1;
- }
- assert(_receivedPacketList.empty());
- }
- ForwardErrorCorrection::RecoveredPacketList::iterator it =
- _recoveredPacketList.begin();
- for (; it != _recoveredPacketList.end(); ++it) {
- if ((*it)->returned) // Already sent to the VCM and the jitter buffer.
- continue;
- if (ParseAndReceivePacket((*it)->pkt) != 0) {
- return -1;
- }
- (*it)->returned = true;
- }
- return 0;
-}
-
-int ReceiverFEC::ParseAndReceivePacket(
- const ForwardErrorCorrection::Packet* packet) {
- WebRtcRTPHeader header;
- memset(&header, 0, sizeof(header));
- ModuleRTPUtility::RTPHeaderParser parser(packet->data,
- packet->length);
- if (!parser.Parse(header)) {
- return -1;
- }
- if (_owner->ReceiveRecoveredPacketCallback(
- &header,
- &packet->data[header.header.headerLength],
- packet->length - header.header.headerLength) != 0) {
- return -1;
- }
- return 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/receiver_fec.h b/trunk/src/modules/rtp_rtcp/source/receiver_fec.h
deleted file mode 100644
index 63aaa72..0000000
--- a/trunk/src/modules/rtp_rtcp/source/receiver_fec.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVER_FEC_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVER_FEC_H_
-
-#include "rtp_rtcp_defines.h"
-// This header is included to get the nested declaration of Packet structure.
-#include "forward_error_correction.h"
-
-#include "typedefs.h"
-
-namespace webrtc {
-class RTPReceiverVideo;
-
-class ReceiverFEC
-{
-public:
- ReceiverFEC(const WebRtc_Word32 id, RTPReceiverVideo* owner);
- virtual ~ReceiverFEC();
-
- WebRtc_Word32 AddReceivedFECPacket(const WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord8* incomingRtpPacket,
- const WebRtc_UWord16 payloadDataLength,
- bool& FECpacket);
-
- WebRtc_Word32 ProcessReceivedFEC();
-
- void SetPayloadTypeFEC(const WebRtc_Word8 payloadType);
-
-private:
- int ParseAndReceivePacket(const ForwardErrorCorrection::Packet* packet);
-
- int _id;
- RTPReceiverVideo* _owner;
- ForwardErrorCorrection* _fec;
- // TODO(holmer): In the current version _receivedPacketList is never more
- // than one packet, since we process FEC every time a new packet
- // arrives. We should remove the list.
- ForwardErrorCorrection::ReceivedPacketList _receivedPacketList;
- ForwardErrorCorrection::RecoveredPacketList _recoveredPacketList;
- WebRtc_Word8 _payloadTypeFEC;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RECEIVER_FEC_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/receiver_fec_unittest.cc b/trunk/src/modules/rtp_rtcp/source/receiver_fec_unittest.cc
deleted file mode 100644
index 669018e..0000000
--- a/trunk/src/modules/rtp_rtcp/source/receiver_fec_unittest.cc
+++ /dev/null
@@ -1,523 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string.h>
-#include <list>
-
-#include "gmock/gmock.h"
-#include "gtest/gtest.h"
-#include "modules/rtp_rtcp/source/forward_error_correction.h"
-#include "modules/rtp_rtcp/source/mock/mock_rtp_receiver_video.h"
-#include "modules/rtp_rtcp/source/receiver_fec.h"
-
-using ::testing::_;
-using ::testing::Args;
-using ::testing::ElementsAreArray;
-using ::testing::InSequence;
-
-namespace webrtc {
-
-typedef ForwardErrorCorrection::Packet Packet;
-
-enum { kRtpHeaderSize = 12 };
-enum { kFecPayloadType = 96 };
-enum { kRedPayloadType = 97 };
-enum { kVp8PayloadType = 120 };
-
-struct RtpPacket : public Packet {
- WebRtcRTPHeader header;
-};
-
-class FrameGenerator {
- public:
- FrameGenerator() : num_packets_(0), seq_num_(0), timestamp_(0) {}
-
- void NewFrame(int num_packets) {
- num_packets_ = num_packets;
- timestamp_ += 3000;
- }
-
- RtpPacket* NextPacket(int offset, size_t length) {
- RtpPacket* rtp_packet = new RtpPacket;
- for (size_t i = 0; i < length; ++i)
- rtp_packet->data[i] = offset + i;
- rtp_packet->length = length;
- memset(&rtp_packet->header, 0, sizeof(WebRtcRTPHeader));
- rtp_packet->header.frameType = kVideoFrameDelta;
- rtp_packet->header.header.headerLength = kRtpHeaderSize;
- rtp_packet->header.header.markerBit = (num_packets_ == 1);
- rtp_packet->header.header.sequenceNumber = seq_num_;
- rtp_packet->header.header.timestamp = timestamp_;
- rtp_packet->header.header.payloadType = kVp8PayloadType;
- BuildRtpHeader(rtp_packet->data, rtp_packet->header.header);
- ++seq_num_;
- --num_packets_;
- return rtp_packet;
- }
-
- // Creates a new RtpPacket with the RED header added to the packet.
- RtpPacket* BuildMediaRedPacket(const RtpPacket* packet) {
- const int kHeaderLength = packet->header.header.headerLength;
- RtpPacket* red_packet = new RtpPacket;
- red_packet->header = packet->header;
- red_packet->length = packet->length + 1; // 1 byte RED header.
- memset(red_packet->data, 0, red_packet->length);
- // Copy RTP header.
- memcpy(red_packet->data, packet->data, kHeaderLength);
- SetRedHeader(red_packet, red_packet->data[1] & 0x7f, kHeaderLength);
- memcpy(red_packet->data + kHeaderLength + 1, packet->data + kHeaderLength,
- packet->length - kHeaderLength);
- return red_packet;
- }
-
- // Creates a new RtpPacket with FEC payload and red header. Does this by
- // creating a new fake media RtpPacket, clears the marker bit and adds a RED
- // header. Finally replaces the payload with the content of |packet->data|.
- RtpPacket* BuildFecRedPacket(const Packet* packet) {
- // Create a fake media packet to get a correct header. 1 byte RED header.
- ++num_packets_;
- RtpPacket* red_packet = NextPacket(0, packet->length + 1);
- red_packet->data[1] &= ~0x80; // Clear marker bit.
- const int kHeaderLength = red_packet->header.header.headerLength;
- SetRedHeader(red_packet, kFecPayloadType, kHeaderLength);
- memcpy(red_packet->data + kHeaderLength + 1, packet->data,
- packet->length);
- red_packet->length = kHeaderLength + 1 + packet->length;
- return red_packet;
- }
-
- void SetRedHeader(Packet* red_packet, uint8_t payload_type,
- int header_length) const {
- // Replace pltype.
- red_packet->data[1] &= 0x80; // Reset.
- red_packet->data[1] += kRedPayloadType; // Replace.
-
- // Add RED header, f-bit always 0.
- red_packet->data[header_length] = payload_type;
- }
-
- private:
- void BuildRtpHeader(uint8_t* data, RTPHeader header) {
- data[0] = 0x80; // Version 2.
- data[1] = header.payloadType;
- data[1] |= (header.markerBit ? kRtpMarkerBitMask : 0);
- ModuleRTPUtility::AssignUWord16ToBuffer(data+2, header.sequenceNumber);
- ModuleRTPUtility::AssignUWord32ToBuffer(data+4, header.timestamp);
- ModuleRTPUtility::AssignUWord32ToBuffer(data+8, header.ssrc);
- }
-
- int num_packets_;
- uint16_t seq_num_;
- uint32_t timestamp_;
-};
-
-class ReceiverFecTest : public ::testing::Test {
- protected:
- virtual void SetUp() {
- fec_ = new ForwardErrorCorrection(0);
- receiver_fec_ = new ReceiverFEC(0, &rtp_receiver_video_);
- generator_ = new FrameGenerator();
- receiver_fec_->SetPayloadTypeFEC(kFecPayloadType);
- }
-
- virtual void TearDown() {
- delete fec_;
- delete receiver_fec_;
- delete generator_;
- }
-
- void GenerateAndAddFrames(int num_frames,
- int num_packets_per_frame,
- std::list<RtpPacket*>* media_rtp_packets,
- std::list<Packet*>* media_packets) {
- for (int i = 0; i < num_frames; ++i) {
- GenerateFrame(num_packets_per_frame, i, media_rtp_packets,
- media_packets);
- }
- for (std::list<RtpPacket*>::iterator it = media_rtp_packets->begin();
- it != media_rtp_packets->end(); ++it) {
- BuildAndAddRedMediaPacket(*it);
- }
- }
-
- void GenerateFEC(std::list<Packet*>* media_packets,
- std::list<Packet*>* fec_packets,
- unsigned int num_fec_packets) {
- EXPECT_EQ(0, fec_->GenerateFEC(
- *media_packets,
- num_fec_packets * 255 / media_packets->size(),
- 0,
- false,
- fec_packets));
- ASSERT_EQ(num_fec_packets, fec_packets->size());
- }
-
- void GenerateFrame(int num_media_packets,
- int frame_offset,
- std::list<RtpPacket*>* media_rtp_packets,
- std::list<Packet*>* media_packets) {
- generator_->NewFrame(num_media_packets);
- for (int i = 0; i < num_media_packets; ++i) {
- media_rtp_packets->push_back(generator_->NextPacket(frame_offset + i,
- kRtpHeaderSize + 10));
- media_packets->push_back(media_rtp_packets->back());
- }
- }
-
- void VerifyReconstructedMediaPacket(const RtpPacket* packet, int times) {
- // Verify that the content of the reconstructed packet is equal to the
- // content of |packet|, and that the same content is received |times| number
- // of times in a row.
- EXPECT_CALL(rtp_receiver_video_,
- ReceiveRecoveredPacketCallback(_, _,
- packet->length - kRtpHeaderSize))
- .With(Args<1, 2>(ElementsAreArray(packet->data + kRtpHeaderSize,
- packet->length - kRtpHeaderSize)))
- .Times(times);
- }
-
- void BuildAndAddRedMediaPacket(RtpPacket* packet) {
- RtpPacket* red_packet = generator_->BuildMediaRedPacket(packet);
- bool is_fec = false;
- EXPECT_EQ(0, receiver_fec_->AddReceivedFECPacket(&red_packet->header,
- red_packet->data,
- red_packet->length -
- kRtpHeaderSize,
- is_fec));
- delete red_packet;
- EXPECT_FALSE(is_fec);
- }
-
- void BuildAndAddRedFecPacket(Packet* packet) {
- RtpPacket* red_packet = generator_->BuildFecRedPacket(packet);
- bool is_fec = false;
- EXPECT_EQ(0, receiver_fec_->AddReceivedFECPacket(&red_packet->header,
- red_packet->data,
- red_packet->length -
- kRtpHeaderSize,
- is_fec));
- delete red_packet;
- EXPECT_TRUE(is_fec);
- }
-
- ForwardErrorCorrection* fec_;
- MockRTPReceiverVideo rtp_receiver_video_;
- ReceiverFEC* receiver_fec_;
- FrameGenerator* generator_;
-};
-
-void DeletePackets(std::list<Packet*>* packets) {
- while (!packets->empty()) {
- delete packets->front();
- packets->pop_front();
- }
-}
-
-TEST_F(ReceiverFecTest, TwoMediaOneFec) {
- const unsigned int kNumFecPackets = 1u;
- std::list<RtpPacket*> media_rtp_packets;
- std::list<Packet*> media_packets;
- GenerateFrame(2, 0, &media_rtp_packets, &media_packets);
- std::list<Packet*> fec_packets;
- GenerateFEC(&media_packets, &fec_packets, kNumFecPackets);
-
- // Recovery
- std::list<RtpPacket*>::iterator media_it = media_rtp_packets.begin();
- BuildAndAddRedMediaPacket(*media_it);
- // Drop one media packet.
- std::list<Packet*>::iterator fec_it = fec_packets.begin();
- BuildAndAddRedFecPacket(*fec_it);
- {
- InSequence s;
- std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
- VerifyReconstructedMediaPacket(*it, 1);
- ++it;
- VerifyReconstructedMediaPacket(*it, 1);
- }
- EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
-
- DeletePackets(&media_packets);
-}
-
-TEST_F(ReceiverFecTest, TwoMediaTwoFec) {
- const unsigned int kNumFecPackets = 2u;
- std::list<RtpPacket*> media_rtp_packets;
- std::list<Packet*> media_packets;
- GenerateFrame(2, 0, &media_rtp_packets, &media_packets);
- std::list<Packet*> fec_packets;
- GenerateFEC(&media_packets, &fec_packets, kNumFecPackets);
-
- // Recovery
- // Drop both media packets.
- std::list<Packet*>::iterator fec_it = fec_packets.begin();
- BuildAndAddRedFecPacket(*fec_it);
- ++fec_it;
- BuildAndAddRedFecPacket(*fec_it);
- {
- InSequence s;
- std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
- VerifyReconstructedMediaPacket(*it, 1);
- ++it;
- VerifyReconstructedMediaPacket(*it, 1);
- }
- EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
-
- DeletePackets(&media_packets);
-}
-
-TEST_F(ReceiverFecTest, TwoFramesOneFec) {
- const unsigned int kNumFecPackets = 1u;
- std::list<RtpPacket*> media_rtp_packets;
- std::list<Packet*> media_packets;
- GenerateFrame(1, 0, &media_rtp_packets, &media_packets);
- GenerateFrame(1, 1, &media_rtp_packets, &media_packets);
- std::list<Packet*> fec_packets;
- GenerateFEC(&media_packets, &fec_packets, kNumFecPackets);
-
- // Recovery
- BuildAndAddRedMediaPacket(media_rtp_packets.front());
- // Drop one media packet.
- BuildAndAddRedFecPacket(fec_packets.front());
- {
- InSequence s;
- std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
- VerifyReconstructedMediaPacket(*it, 1);
- ++it;
- VerifyReconstructedMediaPacket(*it, 1);
- }
- EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
-
- DeletePackets(&media_packets);
-}
-
-TEST_F(ReceiverFecTest, OneCompleteOneUnrecoverableFrame) {
- const unsigned int kNumFecPackets = 1u;
- std::list<RtpPacket*> media_rtp_packets;
- std::list<Packet*> media_packets;
- GenerateFrame(1, 0, &media_rtp_packets, &media_packets);
- GenerateFrame(2, 1, &media_rtp_packets, &media_packets);
- std::list<Packet*> fec_packets;
- GenerateFEC(&media_packets, &fec_packets, kNumFecPackets);
-
- // Recovery
- std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
- BuildAndAddRedMediaPacket(*it); // First frame
- BuildAndAddRedMediaPacket(*it); // First packet of second frame.
- EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_, _, _))
- .Times(1);
- EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
-
- DeletePackets(&media_packets);
-}
-
-TEST_F(ReceiverFecTest, MaxFramesOneFec) {
- const unsigned int kNumFecPackets = 1u;
- const unsigned int kNumMediaPackets = 48u;
- std::list<RtpPacket*> media_rtp_packets;
- std::list<Packet*> media_packets;
- for (unsigned int i = 0; i < kNumMediaPackets; ++i)
- GenerateFrame(1, i, &media_rtp_packets, &media_packets);
- std::list<Packet*> fec_packets;
- GenerateFEC(&media_packets, &fec_packets, kNumFecPackets);
-
- // Recovery
- std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
- ++it; // Drop first packet.
- for (; it != media_rtp_packets.end(); ++it)
- BuildAndAddRedMediaPacket(*it);
- BuildAndAddRedFecPacket(fec_packets.front());
- {
- InSequence s;
- std::list<RtpPacket*>::iterator it = media_rtp_packets.begin();
- for (; it != media_rtp_packets.end(); ++it)
- VerifyReconstructedMediaPacket(*it, 1);
- }
- EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
-
- DeletePackets(&media_packets);
-}
-
-TEST_F(ReceiverFecTest, TooManyFrames) {
- const unsigned int kNumFecPackets = 1u;
- const unsigned int kNumMediaPackets = 49u;
- std::list<RtpPacket*> media_rtp_packets;
- std::list<Packet*> media_packets;
- for (unsigned int i = 0; i < kNumMediaPackets; ++i)
- GenerateFrame(1, i, &media_rtp_packets, &media_packets);
- std::list<Packet*> fec_packets;
- EXPECT_EQ(-1, fec_->GenerateFEC(media_packets,
- kNumFecPackets * 255 / kNumMediaPackets,
- 0,
- false,
- &fec_packets));
-
- DeletePackets(&media_packets);
-}
-
-TEST_F(ReceiverFecTest, PacketNotDroppedTooEarly) {
- // 1 frame with 2 media packets and one FEC packet. One media packet missing.
- // Delay the FEC packet.
- Packet* delayed_fec = NULL;
- const unsigned int kNumFecPacketsBatch1 = 1u;
- const unsigned int kNumMediaPacketsBatch1 = 2u;
- std::list<RtpPacket*> media_rtp_packets_batch1;
- std::list<Packet*> media_packets_batch1;
- GenerateFrame(kNumMediaPacketsBatch1, 0, &media_rtp_packets_batch1,
- &media_packets_batch1);
- std::list<Packet*> fec_packets;
- GenerateFEC(&media_packets_batch1, &fec_packets, kNumFecPacketsBatch1);
-
- BuildAndAddRedMediaPacket(media_rtp_packets_batch1.front());
- EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
- .Times(1);
- EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
- delayed_fec = fec_packets.front();
-
- // Fill the FEC decoder. No packets should be dropped.
- const unsigned int kNumMediaPacketsBatch2 = 47u;
- std::list<RtpPacket*> media_rtp_packets_batch2;
- std::list<Packet*> media_packets_batch2;
- GenerateAndAddFrames(kNumMediaPacketsBatch2, 1, &media_rtp_packets_batch2,
- &media_packets_batch2);
- EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
- .Times(media_packets_batch2.size());
- EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
-
- // Add the delayed FEC packet. One packet should be reconstructed.
- BuildAndAddRedFecPacket(delayed_fec);
- EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
- .Times(1);
- EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
-
- DeletePackets(&media_packets_batch1);
- DeletePackets(&media_packets_batch2);
-}
-
-TEST_F(ReceiverFecTest, PacketDroppedWhenTooOld) {
- // 1 frame with 2 media packets and one FEC packet. One media packet missing.
- // Delay the FEC packet.
- Packet* delayed_fec = NULL;
- const unsigned int kNumFecPacketsBatch1 = 1u;
- const unsigned int kNumMediaPacketsBatch1 = 2u;
- std::list<RtpPacket*> media_rtp_packets_batch1;
- std::list<Packet*> media_packets_batch1;
- GenerateFrame(kNumMediaPacketsBatch1, 0, &media_rtp_packets_batch1,
- &media_packets_batch1);
- std::list<Packet*> fec_packets;
- GenerateFEC(&media_packets_batch1, &fec_packets, kNumFecPacketsBatch1);
-
- BuildAndAddRedMediaPacket(media_rtp_packets_batch1.front());
- EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
- .Times(1);
- EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
- delayed_fec = fec_packets.front();
-
- // Fill the FEC decoder and force the last packet to be dropped.
- const unsigned int kNumMediaPacketsBatch2 = 48u;
- std::list<RtpPacket*> media_rtp_packets_batch2;
- std::list<Packet*> media_packets_batch2;
- GenerateAndAddFrames(kNumMediaPacketsBatch2, 1, &media_rtp_packets_batch2,
- &media_packets_batch2);
- EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
- .Times(media_packets_batch2.size());
- EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
-
- // Add the delayed FEC packet. No packet should be reconstructed since the
- // first media packet of that frame has been dropped due to being too old.
- BuildAndAddRedFecPacket(delayed_fec);
- EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
- .Times(0);
- EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
-
- DeletePackets(&media_packets_batch1);
- DeletePackets(&media_packets_batch2);
-}
-
-TEST_F(ReceiverFecTest, OldFecPacketDropped) {
- // 49 frames with 2 media packets and one FEC packet. All media packets
- // missing.
- const unsigned int kNumMediaPackets = 49 * 2;
- std::list<RtpPacket*> media_rtp_packets;
- std::list<Packet*> media_packets;
- for (unsigned int i = 0; i < kNumMediaPackets / 2; ++i) {
- std::list<RtpPacket*> frame_media_rtp_packets;
- std::list<Packet*> frame_media_packets;
- std::list<Packet*> fec_packets;
- GenerateFrame(2, 0, &frame_media_rtp_packets, &frame_media_packets);
- GenerateFEC(&frame_media_packets, &fec_packets, 1);
- for (std::list<Packet*>::iterator it = fec_packets.begin();
- it != fec_packets.end(); ++it) {
- BuildAndAddRedFecPacket(*it);
- }
- media_packets.insert(media_packets.end(),
- frame_media_packets.begin(),
- frame_media_packets.end());
- media_rtp_packets.insert(media_rtp_packets.end(),
- frame_media_rtp_packets.begin(),
- frame_media_rtp_packets.end());
- }
- // Don't insert any media packets.
- // Only FEC packets inserted. No packets should be recoverable at this time.
- EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
- .Times(0);
- EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
-
- // Insert the oldest media packet. The corresponding FEC packet is too old
- // and should've been dropped. Only the media packet we inserted will be
- // returned.
- BuildAndAddRedMediaPacket(media_rtp_packets.front());
- EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
- .Times(1);
- EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
-
- DeletePackets(&media_packets);
-}
-
-TEST_F(ReceiverFecTest, PacketsOnlyReturnedOnce) {
- const unsigned int kNumFecPackets = 1u;
- std::list<RtpPacket*> media_rtp_packets;
- std::list<Packet*> media_packets;
- GenerateFrame(1, 0, &media_rtp_packets, &media_packets);
- GenerateFrame(2, 1, &media_rtp_packets, &media_packets);
- std::list<Packet*> fec_packets;
- GenerateFEC(&media_packets, &fec_packets, kNumFecPackets);
-
- // Recovery
- std::list<RtpPacket*>::iterator media_it = media_rtp_packets.begin();
- BuildAndAddRedMediaPacket(*media_it); // First frame.
- {
- std::list<RtpPacket*>::iterator verify_it = media_rtp_packets.begin();
- VerifyReconstructedMediaPacket(*verify_it, 1); // First frame
- }
- EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
-
- ++media_it;
- BuildAndAddRedMediaPacket(*media_it); // 1st packet of 2nd frame.
- BuildAndAddRedFecPacket(fec_packets.front()); // Insert FEC packet.
- {
- InSequence s;
- std::list<RtpPacket*>::iterator verify_it = media_rtp_packets.begin();
- ++verify_it; // First frame has already been returned.
- VerifyReconstructedMediaPacket(*verify_it, 1); // 1st packet of 2nd frame.
- ++verify_it;
- VerifyReconstructedMediaPacket(*verify_it, 1); // 2nd packet of 2nd frame.
- }
- EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
-
- ++media_it;
- BuildAndAddRedMediaPacket(*media_it); // 2nd packet of 2nd frame.
- EXPECT_CALL(rtp_receiver_video_, ReceiveRecoveredPacketCallback(_,_,_))
- .Times(0);
- EXPECT_EQ(0, receiver_fec_->ProcessReceivedFEC());
-
- DeletePackets(&media_packets);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/remote_rate_control.cc b/trunk/src/modules/rtp_rtcp/source/remote_rate_control.cc
deleted file mode 100644
index fe8477c..0000000
--- a/trunk/src/modules/rtp_rtcp/source/remote_rate_control.cc
+++ /dev/null
@@ -1,483 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#if _WIN32
-#include <windows.h>
-#endif
-
-#include "remote_rate_control.h"
-#include "trace.h"
-#include <math.h>
-#include <string.h>
-
-#ifdef MATLAB
-extern MatlabEngine eng; // global variable defined elsewhere
-#endif
-
-namespace webrtc {
-RemoteRateControl::RemoteRateControl()
-:
-_minConfiguredBitRate(30000),
-_maxConfiguredBitRate(30000000),
-_currentBitRate(_maxConfiguredBitRate),
-_maxHoldRate(0),
-_avgMaxBitRate(-1.0f),
-_varMaxBitRate(0.4f),
-_rcState(kRcHold),
-_cameFromState(kRcDecrease),
-_rcRegion(kRcMaxUnknown),
-_lastBitRateChange(-1),
-_currentInput(kBwNormal, 0, 1.0),
-_updated(false),
-_timeFirstIncomingEstimate(-1),
-_initializedBitRate(false),
-_avgChangePeriod(1000.0f),
-_lastChangeMs(-1),
-_beta(0.9f)
-#ifdef MATLAB
-,_plot1(NULL),
-_plot2(NULL)
-#endif
-{
-}
-
-RemoteRateControl::~RemoteRateControl()
-{
-#ifdef MATLAB
- eng.DeletePlot(_plot1);
- eng.DeletePlot(_plot2);
-#endif
-}
-
-void RemoteRateControl::Reset()
-{
- _minConfiguredBitRate = 30000;
- _maxConfiguredBitRate = 30000000;
- _currentBitRate = _maxConfiguredBitRate;
- _maxHoldRate = 0;
- _avgMaxBitRate = -1.0f;
- _varMaxBitRate = 0.4f;
- _rcState = kRcHold;
- _cameFromState = kRcHold;
- _rcRegion = kRcMaxUnknown;
- _lastBitRateChange = -1;
- _avgChangePeriod = 1000.0f;
- _lastChangeMs = -1;
- _beta = 0.9f;
- _currentInput._bwState = kBwNormal;
- _currentInput._incomingBitRate = 0;
- _currentInput._noiseVar = 1.0;
- _updated = false;
- _timeFirstIncomingEstimate = -1;
- _initializedBitRate = false;
-}
-
-bool RemoteRateControl::ValidEstimate() const {
- return _initializedBitRate;
-}
-
-WebRtc_Word32 RemoteRateControl::SetConfiguredBitRates(WebRtc_UWord32 minBitRateBps, WebRtc_UWord32 maxBitRateBps)
-{
- if (minBitRateBps > maxBitRateBps)
- {
- return -1;
- }
- _minConfiguredBitRate = minBitRateBps;
- _maxConfiguredBitRate = maxBitRateBps;
- _currentBitRate = BWE_MIN(BWE_MAX(minBitRateBps, _currentBitRate), maxBitRateBps);
- return 0;
-}
-
-WebRtc_UWord32 RemoteRateControl::LatestEstimate() const {
- return _currentBitRate;
-}
-
-WebRtc_UWord32 RemoteRateControl::UpdateBandwidthEstimate(WebRtc_UWord32 RTT,
- WebRtc_Word64 nowMS)
-{
- _currentBitRate = ChangeBitRate(_currentBitRate, _currentInput._incomingBitRate,
- _currentInput._noiseVar, RTT, nowMS);
- return _currentBitRate;
-}
-
-RateControlRegion RemoteRateControl::Update(const RateControlInput& input,
- bool& firstOverUse,
- WebRtc_Word64 nowMS)
-{
-#ifdef MATLAB
- // Create plots
- if (_plot1 == NULL)
- {
- _plot1 = eng.NewPlot(new MatlabPlot());
-
- _plot1->AddTimeLine(30, "b", "current");
- _plot1->AddTimeLine(30, "r-", "avgMax");
- _plot1->AddTimeLine(30, "r--", "pStdMax");
- _plot1->AddTimeLine(30, "r--", "nStdMax");
- _plot1->AddTimeLine(30, "r+", "max");
- _plot1->AddTimeLine(30, "g", "incoming");
- _plot1->AddTimeLine(30, "b+", "recovery");
- }
- if (_plot2 == NULL)
- {
- _plot2 = eng.NewPlot(new MatlabPlot());
-
- _plot2->AddTimeLine(30, "b", "alpha");
- }
-#endif
-
- firstOverUse = (_currentInput._bwState != kBwOverusing &&
- input._bwState == kBwOverusing);
-
- // Set the initial bit rate value to what we're receiving the first second
- if (!_initializedBitRate)
- {
- if (_timeFirstIncomingEstimate < 0)
- {
- if (input._incomingBitRate > 0)
- {
- _timeFirstIncomingEstimate = nowMS;
- }
- }
- else if (nowMS - _timeFirstIncomingEstimate > 1000 &&
- input._incomingBitRate > 0)
- {
- _currentBitRate = input._incomingBitRate;
- _initializedBitRate = true;
- }
- }
-
- if (_updated && _currentInput._bwState == kBwOverusing)
- {
- // Only update delay factor and incoming bit rate. We always want to react on an over-use.
- _currentInput._noiseVar = input._noiseVar;
- _currentInput._incomingBitRate = input._incomingBitRate;
- return _rcRegion;
- }
- _updated = true;
- _currentInput = input;
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: Incoming rate = %u kbps", input._incomingBitRate/1000);
- return _rcRegion;
-}
-
-WebRtc_UWord32 RemoteRateControl::ChangeBitRate(WebRtc_UWord32 currentBitRate,
- WebRtc_UWord32 incomingBitRate,
- double noiseVar,
- WebRtc_UWord32 RTT,
- WebRtc_Word64 nowMS)
-{
- if (!_updated)
- {
- return _currentBitRate;
- }
- _updated = false;
- UpdateChangePeriod(nowMS);
- ChangeState(_currentInput, nowMS);
- // calculated here because it's used in multiple places
- const float incomingBitRateKbps = incomingBitRate / 1000.0f;
- // Calculate the max bit rate std dev given the normalized
- // variance and the current incoming bit rate.
- const float stdMaxBitRate = sqrt(_varMaxBitRate * _avgMaxBitRate);
- bool recovery = false;
- switch (_rcState)
- {
- case kRcHold:
- {
- _maxHoldRate = BWE_MAX(_maxHoldRate, incomingBitRate);
- break;
- }
- case kRcIncrease:
- {
- if (_avgMaxBitRate >= 0)
- {
- if (incomingBitRateKbps > _avgMaxBitRate + 3 * stdMaxBitRate)
- {
- ChangeRegion(kRcMaxUnknown);
- _avgMaxBitRate = -1.0;
- }
- else if (incomingBitRateKbps > _avgMaxBitRate + 2.5 * stdMaxBitRate)
- {
- ChangeRegion(kRcAboveMax);
- }
- }
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
- "BWE: Response time: %f + %i + 10*33\n",
- _avgChangePeriod, RTT);
- const WebRtc_UWord32 responseTime = static_cast<WebRtc_UWord32>(_avgChangePeriod + 0.5f) + RTT + 300;
- double alpha = RateIncreaseFactor(nowMS, _lastBitRateChange,
- responseTime, noiseVar);
-
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
- "BWE: _avgChangePeriod = %f ms; RTT = %u ms", _avgChangePeriod, RTT);
-
- currentBitRate = static_cast<WebRtc_UWord32>(currentBitRate * alpha) + 1000;
- if (_maxHoldRate > 0 && _beta * _maxHoldRate > currentBitRate)
- {
- currentBitRate = static_cast<WebRtc_UWord32>(_beta * _maxHoldRate);
- _avgMaxBitRate = _beta * _maxHoldRate / 1000.0f;
- ChangeRegion(kRcNearMax);
- recovery = true;
-#ifdef MATLAB
- _plot1->Append("recovery", _maxHoldRate/1000);
-#endif
- }
- _maxHoldRate = 0;
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
- "BWE: Increase rate to currentBitRate = %u kbps", currentBitRate/1000);
- _lastBitRateChange = nowMS;
- break;
- }
- case kRcDecrease:
- {
- if (incomingBitRate < _minConfiguredBitRate)
- {
- currentBitRate = _minConfiguredBitRate;
- }
- else
- {
- // Set bit rate to something slightly lower than max
- // to get rid of any self-induced delay.
- currentBitRate = static_cast<WebRtc_UWord32>(_beta * incomingBitRate + 0.5);
- if (currentBitRate > _currentBitRate)
- {
- // Avoid increasing the rate when over-using.
- if (_rcRegion != kRcMaxUnknown)
- {
- currentBitRate = static_cast<WebRtc_UWord32>(_beta * _avgMaxBitRate * 1000 + 0.5f);
- }
- currentBitRate = BWE_MIN(currentBitRate, _currentBitRate);
- }
- ChangeRegion(kRcNearMax);
-
- if (incomingBitRateKbps < _avgMaxBitRate - 3 * stdMaxBitRate)
- {
- _avgMaxBitRate = -1.0f;
- }
-
- UpdateMaxBitRateEstimate(incomingBitRateKbps);
-
-#ifdef MATLAB
- _plot1->Append("max", incomingBitRateKbps);
-#endif
-
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1, "BWE: Decrease rate to currentBitRate = %u kbps", currentBitRate/1000);
- }
- // Stay on hold until the pipes are cleared.
- ChangeState(kRcHold);
- _lastBitRateChange = nowMS;
- break;
- }
- }
- if (!recovery && (incomingBitRate > 100000 || currentBitRate > 150000) &&
- currentBitRate > 1.5 * incomingBitRate)
- {
- // Allow changing the bit rate if we are operating at very low rates
- // Don't change the bit rate if the send side is too far off
- currentBitRate = _currentBitRate;
- _lastBitRateChange = nowMS;
- }
-#ifdef MATLAB
- if (_avgMaxBitRate >= 0.0f)
- {
- _plot1->Append("avgMax", _avgMaxBitRate);
- _plot1->Append("pStdMax", _avgMaxBitRate + 3*stdMaxBitRate);
- _plot1->Append("nStdMax", _avgMaxBitRate - 3*stdMaxBitRate);
- }
- _plot1->Append("incoming", incomingBitRate/1000);
- _plot1->Append("current", currentBitRate/1000);
- _plot1->Plot();
-#endif
- return currentBitRate;
-}
-
-double RemoteRateControl::RateIncreaseFactor(WebRtc_Word64 nowMs, WebRtc_Word64 lastMs, WebRtc_UWord32 reactionTimeMs, double noiseVar) const
-{
- // alpha = 1.02 + B ./ (1 + exp(b*(tr - (c1*s2 + c2))))
- // Parameters
- const double B = 0.0407;
- const double b = 0.0025;
- const double c1 = -6700.0 / (33 * 33);
- const double c2 = 800.0;
- const double d = 0.85;
-
- double alpha = 1.005 + B / (1 + exp( b * (d * reactionTimeMs - (c1 * noiseVar + c2))));
-
- if (alpha < 1.005)
- {
- alpha = 1.005;
- }
- else if (alpha > 1.3)
- {
- alpha = 1.3;
- }
-
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
- "BWE: alpha = %f", alpha);
-#ifdef MATLAB
- _plot2->Append("alpha", alpha);
- _plot2->Plot();
-#endif
-
- if (lastMs > -1)
- {
- alpha = pow(alpha, (nowMs - lastMs) / 1000.0);
- }
-
- if (_rcRegion == kRcNearMax)
- {
- // We're close to our previous maximum. Try to stabilize the
- // bit rate in this region, by increasing in smaller steps.
- alpha = alpha - (alpha - 1.0) / 2.0;
- }
- else if (_rcRegion == kRcMaxUnknown)
- {
- alpha = alpha + (alpha - 1.0) * 2.0;
- }
-
- return alpha;
-}
-
-void RemoteRateControl::UpdateChangePeriod(WebRtc_Word64 nowMs)
-{
- WebRtc_Word64 changePeriod = 0;
- if (_lastChangeMs > -1)
- {
- changePeriod = nowMs - _lastChangeMs;
- }
- _lastChangeMs = nowMs;
- _avgChangePeriod = 0.9f * _avgChangePeriod + 0.1f * changePeriod;
-}
-
-void RemoteRateControl::UpdateMaxBitRateEstimate(float incomingBitRateKbps)
-{
- const float alpha = 0.05f;
- if (_avgMaxBitRate == -1.0f)
- {
- _avgMaxBitRate = incomingBitRateKbps;
- }
- else
- {
- _avgMaxBitRate = (1 - alpha) * _avgMaxBitRate +
- alpha * incomingBitRateKbps;
- }
- // Estimate the max bit rate variance and normalize the variance
- // with the average max bit rate.
- const float norm = BWE_MAX(_avgMaxBitRate, 1.0f);
- _varMaxBitRate = (1 - alpha) * _varMaxBitRate +
- alpha * (_avgMaxBitRate - incomingBitRateKbps) *
- (_avgMaxBitRate - incomingBitRateKbps) /
- norm;
- // 0.4 ~= 14 kbit/s at 500 kbit/s
- if (_varMaxBitRate < 0.4f)
- {
- _varMaxBitRate = 0.4f;
- }
- // 2.5f ~= 35 kbit/s at 500 kbit/s
- if (_varMaxBitRate > 2.5f)
- {
- _varMaxBitRate = 2.5f;
- }
-}
-
-void RemoteRateControl::ChangeState(const RateControlInput& input, WebRtc_Word64 nowMs)
-{
- switch (_currentInput._bwState)
- {
- case kBwNormal:
- {
- if (_rcState == kRcHold)
- {
- _lastBitRateChange = nowMs;
- ChangeState(kRcIncrease);
- }
- break;
- }
- case kBwOverusing:
- {
- if (_rcState != kRcDecrease)
- {
- ChangeState(kRcDecrease);
- }
- break;
- }
- case kBwUnderUsing:
- {
- ChangeState(kRcHold);
- break;
- }
- }
-}
-
-void RemoteRateControl::ChangeRegion(RateControlRegion region)
-{
- _rcRegion = region;
- switch (_rcRegion)
- {
- case kRcAboveMax:
- case kRcMaxUnknown:
- {
- _beta = 0.9f;
- break;
- }
- case kRcNearMax:
- {
- _beta = 0.95f;
- break;
- }
- }
-}
-
-void RemoteRateControl::ChangeState(RateControlState newState)
-{
- _cameFromState = _rcState;
- _rcState = newState;
- char state1[15];
- char state2[15];
- char state3[15];
- StateStr(_cameFromState, state1);
- StateStr(_rcState, state2);
- StateStr(_currentInput._bwState, state3);
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
- "\t%s => %s due to %s\n", state1, state2, state3);
-}
-
-void RemoteRateControl::StateStr(RateControlState state, char* str)
-{
- switch (state)
- {
- case kRcDecrease:
- strncpy(str, "DECREASE", 9);
- break;
- case kRcHold:
- strncpy(str, "HOLD", 5);
- break;
- case kRcIncrease:
- strncpy(str, "INCREASE", 9);
- break;
- }
-}
-
-void RemoteRateControl::StateStr(BandwidthUsage state, char* str)
-{
- switch (state)
- {
- case kBwNormal:
- strncpy(str, "NORMAL", 7);
- break;
- case kBwOverusing:
- strncpy(str, "OVER USING", 11);
- break;
- case kBwUnderUsing:
- strncpy(str, "UNDER USING", 12);
- break;
- }
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/remote_rate_control.h b/trunk/src/modules/rtp_rtcp/source/remote_rate_control.h
deleted file mode 100644
index 197bf22..0000000
--- a/trunk/src/modules/rtp_rtcp/source/remote_rate_control.h
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_REMOTE_RATE_CONTROL_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_REMOTE_RATE_CONTROL_H_
-
-#include "bwe_defines.h"
-#include "typedefs.h"
-
-#ifdef MATLAB
-#include "../test/BWEStandAlone/MatlabPlot.h"
-#endif
-
-namespace webrtc {
-class RemoteRateControl
-{
-public:
- RemoteRateControl();
- ~RemoteRateControl();
- WebRtc_Word32 SetConfiguredBitRates(WebRtc_UWord32 minBitRate,
- WebRtc_UWord32 maxBitRate);
- WebRtc_UWord32 LatestEstimate() const;
- WebRtc_UWord32 UpdateBandwidthEstimate(WebRtc_UWord32 RTT,
- WebRtc_Word64 nowMS);
- RateControlRegion Update(const RateControlInput& input, bool& firstOverUse,
- WebRtc_Word64 nowMS);
- void Reset();
-
- // Returns true if there is a valid estimate of the incoming bitrate, false
- // otherwise.
- bool ValidEstimate() const;
-
-private:
- WebRtc_UWord32 ChangeBitRate(WebRtc_UWord32 currentBitRate,
- WebRtc_UWord32 incomingBitRate,
- double delayFactor, WebRtc_UWord32 RTT,
- WebRtc_Word64 nowMS);
- double RateIncreaseFactor(WebRtc_Word64 nowMs,
- WebRtc_Word64 lastMs,
- WebRtc_UWord32 reactionTimeMs,
- double noiseVar) const;
- void UpdateChangePeriod(WebRtc_Word64 nowMs);
- void UpdateMaxBitRateEstimate(float incomingBitRateKbps);
- void ChangeState(const RateControlInput& input, WebRtc_Word64 nowMs);
- void ChangeState(RateControlState newState);
- void ChangeRegion(RateControlRegion region);
- static void StateStr(RateControlState state, char* str);
- static void StateStr(BandwidthUsage state, char* str);
-
- WebRtc_UWord32 _minConfiguredBitRate;
- WebRtc_UWord32 _maxConfiguredBitRate;
- WebRtc_UWord32 _currentBitRate;
- WebRtc_UWord32 _maxHoldRate;
- float _avgMaxBitRate;
- float _varMaxBitRate;
- RateControlState _rcState;
- RateControlState _cameFromState;
- RateControlRegion _rcRegion;
- WebRtc_Word64 _lastBitRateChange;
- RateControlInput _currentInput;
- bool _updated;
- WebRtc_Word64 _timeFirstIncomingEstimate;
- bool _initializedBitRate;
-
- float _avgChangePeriod;
- WebRtc_Word64 _lastChangeMs;
- float _beta;
-#ifdef MATLAB
- MatlabPlot *_plot1;
- MatlabPlot *_plot2;
-#endif
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_REMOTE_RATE_CONTROL_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc b/trunk/src/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
deleted file mode 100644
index 9fdd6ca..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtcp_format_remb_unittest.cc
+++ /dev/null
@@ -1,120 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <gtest/gtest.h>
-
-#include "typedefs.h"
-#include "common_types.h"
-#include "rtp_utility.h"
-#include "rtcp_sender.h"
-#include "rtcp_receiver.h"
-#include "rtp_rtcp_impl.h"
-#include "bwe_defines.h"
-
-namespace {
-
-using namespace webrtc;
-
-
-class TestTransport : public Transport {
- public:
- TestTransport(RTCPReceiver* rtcp_receiver) :
- rtcp_receiver_(rtcp_receiver) {
- }
-
- virtual int SendPacket(int /*channel*/, const void* /*data*/, int /*len*/) {
- return -1;
- }
- virtual int SendRTCPPacket(int /*channel*/,
- const void *packet,
- int packetLength) {
- RTCPUtility::RTCPParserV2 rtcpParser((WebRtc_UWord8*)packet,
- (WebRtc_Word32)packetLength,
- true); // Allow non-compound RTCP
-
- EXPECT_TRUE(rtcpParser.IsValid());
- RTCPHelp::RTCPPacketInformation rtcpPacketInformation;
- EXPECT_EQ(0, rtcp_receiver_->IncomingRTCPPacket(rtcpPacketInformation,
- &rtcpParser));
-
- EXPECT_EQ((WebRtc_UWord32)kRtcpRemb,
- rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb);
- EXPECT_EQ((WebRtc_UWord32)1234,
- rtcpPacketInformation.receiverEstimatedMaxBitrate);
- return packetLength;
- }
- private:
- RTCPReceiver* rtcp_receiver_;
-};
-
-
-class RtcpFormatRembTest : public ::testing::Test {
- protected:
- RtcpFormatRembTest() {};
- virtual void SetUp();
- virtual void TearDown();
-
- RtpRtcpClock* system_clock_;
- ModuleRtpRtcpImpl* dummy_rtp_rtcp_impl_;
- RTCPSender* rtcp_sender_;
- RTCPReceiver* rtcp_receiver_;
- TestTransport* test_transport_;
-};
-
-void RtcpFormatRembTest::SetUp() {
- system_clock_ = ModuleRTPUtility::GetSystemClock();
- dummy_rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(0, false, system_clock_);
- rtcp_sender_ = new RTCPSender(0, false, system_clock_, dummy_rtp_rtcp_impl_);
- rtcp_receiver_ = new RTCPReceiver(0, system_clock_, dummy_rtp_rtcp_impl_);
- test_transport_ = new TestTransport(rtcp_receiver_);
-
- EXPECT_EQ(0, rtcp_sender_->Init());
- EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(test_transport_));
-}
-
-void RtcpFormatRembTest::TearDown() {
- delete rtcp_sender_;
- delete rtcp_receiver_;
- delete dummy_rtp_rtcp_impl_;
- delete test_transport_;
- delete system_clock_;
-}
-
-TEST_F(RtcpFormatRembTest, TestBasicAPI) {
- EXPECT_FALSE(rtcp_sender_->REMB());
- EXPECT_EQ(0, rtcp_sender_->SetREMBStatus(true));
- EXPECT_TRUE(rtcp_sender_->REMB());
- EXPECT_EQ(0, rtcp_sender_->SetREMBStatus(false));
- EXPECT_FALSE(rtcp_sender_->REMB());
-
- EXPECT_EQ(0, rtcp_sender_->SetREMBData(1234, 0, NULL));
-}
-
-TEST_F(RtcpFormatRembTest, TestNonCompund) {
- WebRtc_UWord32 SSRC = 456789;
- EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpNonCompound));
- EXPECT_EQ(0, rtcp_sender_->SetREMBData(1234, 1, &SSRC));
- EXPECT_EQ(0, rtcp_sender_->SendRTCP(kRtcpRemb));
-}
-
-TEST_F(RtcpFormatRembTest, TestCompund) {
- WebRtc_UWord32 SSRCs[2] = {456789, 98765};
- EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpCompound));
- EXPECT_EQ(0, rtcp_sender_->SetREMBData(1234, 2, SSRCs));
- EXPECT_EQ(0, rtcp_sender_->SendRTCP(kRtcpRemb));
-}
-
-int main(int argc, char** argv) {
- ::testing::InitGoogleTest(&argc, argv);
-
- return RUN_ALL_TESTS();
-}
-
-} // namespace
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_receiver.cc b/trunk/src/modules/rtp_rtcp/source/rtcp_receiver.cc
deleted file mode 100644
index f4eca71..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtcp_receiver.cc
+++ /dev/null
@@ -1,1500 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rtcp_receiver.h"
-
-#include <string.h> //memset
-#include <cassert> //assert
-
-#include "trace.h"
-#include "critical_section_wrapper.h"
-#include "rtcp_utility.h"
-#include "rtp_rtcp_impl.h"
-
-namespace
-{
- const float FRAC = 4.294967296E9;
-}
-
-namespace webrtc {
-using namespace RTCPUtility;
-using namespace RTCPHelp;
-
-RTCPReceiver::RTCPReceiver(const WebRtc_Word32 id,
- RtpRtcpClock* clock,
- ModuleRtpRtcpImpl* owner) :
- _id(id),
- _clock(*clock),
- _method(kRtcpOff),
- _lastReceived(0),
- _rtpRtcp(*owner),
- _criticalSectionFeedbacks(CriticalSectionWrapper::CreateCriticalSection()),
- _cbRtcpFeedback(NULL),
- _cbVideoFeedback(NULL),
- _criticalSectionRTCPReceiver(
- CriticalSectionWrapper::CreateCriticalSection()),
- _SSRC(0),
- _remoteSSRC(0),
- _remoteSenderInfo(),
- _lastReceivedSRNTPsecs(0),
- _lastReceivedSRNTPfrac(0),
- _receivedInfoMap(),
- _packetTimeOutMS(0),
- _rtt(0)
-{
- memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo));
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
-}
-
-RTCPReceiver::~RTCPReceiver() {
- delete _criticalSectionRTCPReceiver;
- delete _criticalSectionFeedbacks;
-
- while (!_receivedReportBlockMap.empty()) {
- std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::iterator first =
- _receivedReportBlockMap.begin();
- delete first->second;
- _receivedReportBlockMap.erase(first);
- }
- while (!_receivedInfoMap.empty()) {
- std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator first =
- _receivedInfoMap.begin();
- delete first->second;
- _receivedInfoMap.erase(first);
- }
- while (!_receivedCnameMap.empty()) {
- std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator first =
- _receivedCnameMap.begin();
- delete first->second;
- _receivedCnameMap.erase(first);
- }
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id,
- "%s deleted", __FUNCTION__);
-}
-
-void
-RTCPReceiver::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
-}
-
-RTCPMethod
-RTCPReceiver::Status() const
-{
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
- return _method;
-}
-
-WebRtc_Word32
-RTCPReceiver::SetRTCPStatus(const RTCPMethod method)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
- _method = method;
- return 0;
-}
-
-WebRtc_UWord32
-RTCPReceiver::LastReceived()
-{
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
- return _lastReceived;
-}
-
-WebRtc_Word32
-RTCPReceiver::SetRemoteSSRC( const WebRtc_UWord32 ssrc)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-
- // new SSRC reset old reports
- memset(&_remoteSenderInfo, 0, sizeof(_remoteSenderInfo));
- _lastReceivedSRNTPsecs = 0;
- _lastReceivedSRNTPfrac = 0;
-
- _remoteSSRC = ssrc;
- return 0;
-}
-
-WebRtc_Word32
-RTCPReceiver::RegisterIncomingRTCPCallback(RtcpFeedback* incomingMessagesCallback)
-{
- CriticalSectionScoped lock(_criticalSectionFeedbacks);
- _cbRtcpFeedback = incomingMessagesCallback;
- return 0;
-}
-
-WebRtc_Word32
-RTCPReceiver::RegisterIncomingVideoCallback(RtpVideoFeedback* incomingMessagesCallback)
-{
- CriticalSectionScoped lock(_criticalSectionFeedbacks);
- _cbVideoFeedback = incomingMessagesCallback;
- return 0;
-}
-
-void
-RTCPReceiver::SetSSRC( const WebRtc_UWord32 ssrc)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
- _SSRC = ssrc;
-}
-
-WebRtc_Word32 RTCPReceiver::ResetRTT(const WebRtc_UWord32 remoteSSRC) {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
- RTCPReportBlockInformation* reportBlock =
- GetReportBlockInformation(remoteSSRC);
- if (reportBlock == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "\tfailed to GetReportBlockInformation(%u)", remoteSSRC);
- return -1;
- }
- reportBlock->RTT = 0;
- reportBlock->avgRTT = 0;
- reportBlock->minRTT = 0;
- reportBlock->maxRTT = 0;
-
- return 0;
-}
-
-WebRtc_Word32 RTCPReceiver::RTT(const WebRtc_UWord32 remoteSSRC,
- WebRtc_UWord16* RTT,
- WebRtc_UWord16* avgRTT,
- WebRtc_UWord16* minRTT,
- WebRtc_UWord16* maxRTT) const {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-
- RTCPReportBlockInformation* reportBlock =
- GetReportBlockInformation(remoteSSRC);
-
- if (reportBlock == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "\tfailed to GetReportBlockInformation(%u)",
- remoteSSRC);
- return -1;
- }
- if (RTT) {
- *RTT = reportBlock->RTT;
- }
- if (avgRTT) {
- *avgRTT = reportBlock->avgRTT;
- }
- if (minRTT) {
- *minRTT = reportBlock->minRTT;
- }
- if (maxRTT) {
- *maxRTT = reportBlock->maxRTT;
- }
- return 0;
-}
-
-WebRtc_UWord16 RTCPReceiver::RTT() const {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
- if (!_receivedReportBlockMap.empty()) {
- return 0;
- }
- return _rtt;
-}
-
-int RTCPReceiver::SetRTT(WebRtc_UWord16 rtt) {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
- if (!_receivedReportBlockMap.empty()) {
- return -1;
- }
- _rtt = rtt;
- return 0;
-}
-
-void
-RTCPReceiver::UpdateLipSync(const WebRtc_Word32 audioVideoOffset) const
-{
- CriticalSectionScoped lock(_criticalSectionFeedbacks);
- if(_cbRtcpFeedback)
- {
- _cbRtcpFeedback->OnLipSyncUpdate(_id,audioVideoOffset);
- }
-};
-
-WebRtc_Word32
-RTCPReceiver::NTP(WebRtc_UWord32 *ReceivedNTPsecs,
- WebRtc_UWord32 *ReceivedNTPfrac,
- WebRtc_UWord32 *RTCPArrivalTimeSecs,
- WebRtc_UWord32 *RTCPArrivalTimeFrac) const
-{
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
- if(ReceivedNTPsecs)
- {
- *ReceivedNTPsecs = _remoteSenderInfo.NTPseconds; // NTP from incoming SendReport
- }
- if(ReceivedNTPfrac)
- {
- *ReceivedNTPfrac = _remoteSenderInfo.NTPfraction;
- }
- if(RTCPArrivalTimeFrac)
- {
- *RTCPArrivalTimeFrac = _lastReceivedSRNTPfrac; // local NTP time when we received a RTCP packet with a send block
- }
- if(RTCPArrivalTimeSecs)
- {
- *RTCPArrivalTimeSecs = _lastReceivedSRNTPsecs;
- }
- return 0;
-}
-
-WebRtc_Word32
-RTCPReceiver::SenderInfoReceived(RTCPSenderInfo* senderInfo) const
-{
- if(senderInfo == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
- return -1;
- }
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
- if(_lastReceivedSRNTPsecs == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s No received SR", __FUNCTION__);
- return -1;
- }
- memcpy(senderInfo, &(_remoteSenderInfo), sizeof(RTCPSenderInfo));
- return 0;
-}
-
-// statistics
-// we can get multiple receive reports when we receive the report from a CE
-WebRtc_Word32 RTCPReceiver::StatisticsReceived(
- std::vector<RTCPReportBlock>* receiveBlocks) const {
- assert(receiveBlocks);
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-
- std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::const_iterator it =
- _receivedReportBlockMap.begin();
-
- while (it != _receivedReportBlockMap.end()) {
- receiveBlocks->push_back(it->second->remoteReceiveBlock);
- it++;
- }
- return 0;
-}
-
-WebRtc_Word32
-RTCPReceiver::IncomingRTCPPacket(RTCPPacketInformation& rtcpPacketInformation,
- RTCPUtility::RTCPParserV2* rtcpParser)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-
- _lastReceived = _clock.GetTimeInMS();
-
- RTCPUtility::RTCPPacketTypes pktType = rtcpParser->Begin();
- while (pktType != RTCPUtility::kRtcpNotValidCode)
- {
- // Each "case" is responsible for iterate the parser to the
- // next top level packet.
- switch (pktType)
- {
- case RTCPUtility::kRtcpSrCode:
- case RTCPUtility::kRtcpRrCode:
- HandleSenderReceiverReport(*rtcpParser, rtcpPacketInformation);
- break;
- case RTCPUtility::kRtcpSdesCode:
- HandleSDES(*rtcpParser);
- break;
- case RTCPUtility::kRtcpXrVoipMetricCode:
- HandleXRVOIPMetric(*rtcpParser, rtcpPacketInformation);
- break;
- case RTCPUtility::kRtcpByeCode:
- HandleBYE(*rtcpParser);
- break;
- case RTCPUtility::kRtcpRtpfbNackCode:
- HandleNACK(*rtcpParser, rtcpPacketInformation);
- break;
- case RTCPUtility::kRtcpRtpfbTmmbrCode:
- HandleTMMBR(*rtcpParser, rtcpPacketInformation);
- break;
- case RTCPUtility::kRtcpRtpfbTmmbnCode:
- HandleTMMBN(*rtcpParser);
- break;
- case RTCPUtility::kRtcpRtpfbSrReqCode:
- HandleSR_REQ(*rtcpParser, rtcpPacketInformation);
- break;
- case RTCPUtility::kRtcpPsfbPliCode:
- HandlePLI(*rtcpParser, rtcpPacketInformation);
- break;
- case RTCPUtility::kRtcpPsfbSliCode:
- HandleSLI(*rtcpParser, rtcpPacketInformation);
- break;
- case RTCPUtility::kRtcpPsfbRpsiCode:
- HandleRPSI(*rtcpParser, rtcpPacketInformation);
- break;
- case RTCPUtility::kRtcpExtendedIjCode:
- HandleIJ(*rtcpParser, rtcpPacketInformation);
- break;
- case RTCPUtility::kRtcpPsfbFirCode:
- HandleFIR(*rtcpParser, rtcpPacketInformation);
- break;
- case RTCPUtility::kRtcpPsfbAppCode:
- HandlePsfbApp(*rtcpParser, rtcpPacketInformation);
- break;
- case RTCPUtility::kRtcpAppCode:
- // generic application messages
- HandleAPP(*rtcpParser, rtcpPacketInformation);
- break;
- case RTCPUtility::kRtcpAppItemCode:
- // generic application messages
- HandleAPPItem(*rtcpParser, rtcpPacketInformation);
- break;
- default:
- rtcpParser->Iterate();
- break;
- }
- pktType = rtcpParser->PacketType();
- }
- return 0;
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandleSenderReceiverReport(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPPacketInformation& rtcpPacketInformation)
-{
- RTCPUtility::RTCPPacketTypes rtcpPacketType = rtcpParser.PacketType();
- const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
-
- assert((rtcpPacketType == RTCPUtility::kRtcpRrCode) || (rtcpPacketType == RTCPUtility::kRtcpSrCode));
-
- // SR.SenderSSRC
- // The synchronization source identifier for the originator of this SR packet
-
- // rtcpPacket.RR.SenderSSRC
- // The source of the packet sender, same as of SR? or is this a CE?
-
- const WebRtc_UWord32 remoteSSRC = (rtcpPacketType == RTCPUtility::kRtcpRrCode) ? rtcpPacket.RR.SenderSSRC:rtcpPacket.SR.SenderSSRC;
- const WebRtc_UWord8 numberOfReportBlocks = (rtcpPacketType == RTCPUtility::kRtcpRrCode) ? rtcpPacket.RR.NumberOfReportBlocks:rtcpPacket.SR.NumberOfReportBlocks;
-
- rtcpPacketInformation.remoteSSRC = remoteSSRC;
-
- RTCPReceiveInformation* ptrReceiveInfo = CreateReceiveInformation(remoteSSRC);
- if (!ptrReceiveInfo)
- {
- rtcpParser.Iterate();
- return;
- }
-
- if (rtcpPacketType == RTCPUtility::kRtcpSrCode)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, _id,
- "Received SR(%d). SSRC:0x%x, from SSRC:0x%x, to us %d.", _id, _SSRC, remoteSSRC, (_remoteSSRC == remoteSSRC)?1:0);
-
- if (_remoteSSRC == remoteSSRC) // have I received RTP packets from this party
- {
- // only signal that we have received a SR when we accept one
- rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpSr;
-
- // We will only store the send report from one source, but
- // we will store all the receive block
-
- // Save the NTP time of this report
- _remoteSenderInfo.NTPseconds = rtcpPacket.SR.NTPMostSignificant;
- _remoteSenderInfo.NTPfraction = rtcpPacket.SR.NTPLeastSignificant;
- _remoteSenderInfo.RTPtimeStamp = rtcpPacket.SR.RTPTimestamp;
- _remoteSenderInfo.sendPacketCount = rtcpPacket.SR.SenderPacketCount;
- _remoteSenderInfo.sendOctetCount = rtcpPacket.SR.SenderOctetCount;
-
- _clock.CurrentNTP(_lastReceivedSRNTPsecs, _lastReceivedSRNTPfrac);
- }
- else
- {
- rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRr;
- }
- } else
- {
- WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, _id,
- "Received RR(%d). SSRC:0x%x, from SSRC:0x%x", _id, _SSRC, remoteSSRC);
-
- rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRr;
- }
- UpdateReceiveInformation(*ptrReceiveInfo);
-
- rtcpPacketType = rtcpParser.Iterate();
-
- while (rtcpPacketType == RTCPUtility::kRtcpReportBlockItemCode)
- {
- HandleReportBlock(rtcpPacket, rtcpPacketInformation, remoteSSRC, numberOfReportBlocks);
- rtcpPacketType = rtcpParser.Iterate();
- }
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandleReportBlock(const RTCPUtility::RTCPPacket& rtcpPacket,
- RTCPPacketInformation& rtcpPacketInformation,
- const WebRtc_UWord32 remoteSSRC,
- const WebRtc_UWord8 numberOfReportBlocks) {
- // This will be called once per report block in the RTCP packet.
- // We filter out all report blocks that are not for us.
- // Each packet has max 31 RR blocks.
- //
- // We can calc RTT if we send a send report and get a report block back.
-
- // |rtcpPacket.ReportBlockItem.SSRC| is the SSRC identifier of the source to
- // which the information in this reception report block pertains.
-
- // Filter out all report blocks that are not for us.
- if (rtcpPacket.ReportBlockItem.SSRC != _SSRC) {
- // This block is not for us ignore it.
- return;
- }
-
- // To avoid problem with acquiring _criticalSectionRTCPSender while holding
- // _criticalSectionRTCPReceiver.
- _criticalSectionRTCPReceiver->Leave();
- WebRtc_UWord32 sendTimeMS =
- _rtpRtcp.SendTimeOfSendReport(rtcpPacket.ReportBlockItem.LastSR);
- _criticalSectionRTCPReceiver->Enter();
-
- RTCPReportBlockInformation* reportBlock =
- CreateReportBlockInformation(remoteSSRC);
- if (reportBlock == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "\tfailed to CreateReportBlockInformation(%u)", remoteSSRC);
- return;
- }
- const RTCPPacketReportBlockItem& rb = rtcpPacket.ReportBlockItem;
- reportBlock->remoteReceiveBlock.remoteSSRC = remoteSSRC;
- reportBlock->remoteReceiveBlock.sourceSSRC = rb.SSRC;
- reportBlock->remoteReceiveBlock.fractionLost = rb.FractionLost;
- reportBlock->remoteReceiveBlock.cumulativeLost =
- rb.CumulativeNumOfPacketsLost;
- reportBlock->remoteReceiveBlock.extendedHighSeqNum =
- rb.ExtendedHighestSequenceNumber;
- reportBlock->remoteReceiveBlock.jitter = rb.Jitter;
- reportBlock->remoteReceiveBlock.delaySinceLastSR = rb.DelayLastSR;
- reportBlock->remoteReceiveBlock.lastSR = rb.LastSR;
-
- if (rtcpPacket.ReportBlockItem.Jitter > reportBlock->remoteMaxJitter) {
- reportBlock->remoteMaxJitter = rtcpPacket.ReportBlockItem.Jitter;
- }
-
- WebRtc_UWord32 delaySinceLastSendReport =
- rtcpPacket.ReportBlockItem.DelayLastSR;
-
- // local NTP time when we received this
- WebRtc_UWord32 lastReceivedRRNTPsecs = 0;
- WebRtc_UWord32 lastReceivedRRNTPfrac = 0;
-
- _clock.CurrentNTP(lastReceivedRRNTPsecs, lastReceivedRRNTPfrac);
-
- // time when we received this in MS
- WebRtc_UWord32 receiveTimeMS = ModuleRTPUtility::ConvertNTPTimeToMS(
- lastReceivedRRNTPsecs, lastReceivedRRNTPfrac);
-
- // Estimate RTT
- WebRtc_UWord32 d = (delaySinceLastSendReport & 0x0000ffff) * 1000;
- d /= 65536;
- d += ((delaySinceLastSendReport & 0xffff0000) >> 16) * 1000;
-
- WebRtc_Word32 RTT = 0;
-
- if (sendTimeMS > 0) {
- RTT = receiveTimeMS - d - sendTimeMS;
- if (RTT <= 0) {
- RTT = 1;
- }
- if (RTT > reportBlock->maxRTT) {
- // store max RTT
- reportBlock->maxRTT = (WebRtc_UWord16) RTT;
- }
- if (reportBlock->minRTT == 0) {
- // first RTT
- reportBlock->minRTT = (WebRtc_UWord16) RTT;
- } else if (RTT < reportBlock->minRTT) {
- // Store min RTT
- reportBlock->minRTT = (WebRtc_UWord16) RTT;
- }
- // store last RTT
- reportBlock->RTT = (WebRtc_UWord16) RTT;
-
- // store average RTT
- if (reportBlock->numAverageCalcs != 0) {
- float ac = static_cast<float> (reportBlock->numAverageCalcs);
- float newAverage = ((ac / (ac + 1)) * reportBlock->avgRTT)
- + ((1 / (ac + 1)) * RTT);
- reportBlock->avgRTT = static_cast<int> (newAverage + 0.5f);
- } else {
- // first RTT
- reportBlock->avgRTT = (WebRtc_UWord16) RTT;
- }
- reportBlock->numAverageCalcs++;
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceRtpRtcp, _id,
- " -> Received report block(%d), from SSRC:0x%x, RTT:%d, loss:%d",
- _id, remoteSSRC, RTT, rtcpPacket.ReportBlockItem.FractionLost);
-
- // rtcpPacketInformation
- rtcpPacketInformation.AddReportInfo(
- reportBlock->remoteReceiveBlock.fractionLost, (WebRtc_UWord16) RTT,
- reportBlock->remoteReceiveBlock.extendedHighSeqNum,
- reportBlock->remoteReceiveBlock.jitter);
-}
-
-RTCPReportBlockInformation*
-RTCPReceiver::CreateReportBlockInformation(WebRtc_UWord32 remoteSSRC) {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-
- std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::iterator it =
- _receivedReportBlockMap.find(remoteSSRC);
-
- RTCPReportBlockInformation* ptrReportBlockInfo = NULL;
- if (it != _receivedReportBlockMap.end()) {
- ptrReportBlockInfo = it->second;
- } else {
- ptrReportBlockInfo = new RTCPReportBlockInformation;
- _receivedReportBlockMap[remoteSSRC] = ptrReportBlockInfo;
- }
- return ptrReportBlockInfo;
-}
-
-RTCPReportBlockInformation*
-RTCPReceiver::GetReportBlockInformation(WebRtc_UWord32 remoteSSRC) const {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-
- std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::const_iterator it =
- _receivedReportBlockMap.find(remoteSSRC);
-
- if (it == _receivedReportBlockMap.end()) {
- return NULL;
- }
- return it->second;
-}
-
-RTCPCnameInformation*
-RTCPReceiver::CreateCnameInformation(WebRtc_UWord32 remoteSSRC) {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-
- std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator it =
- _receivedCnameMap.find(remoteSSRC);
-
- if (it != _receivedCnameMap.end()) {
- return it->second;
- }
- RTCPCnameInformation* cnameInfo = new RTCPCnameInformation;
- memset(cnameInfo->name, 0, RTCP_CNAME_SIZE);
- _receivedCnameMap[remoteSSRC] = cnameInfo;
- return cnameInfo;
-}
-
-RTCPCnameInformation*
-RTCPReceiver::GetCnameInformation(WebRtc_UWord32 remoteSSRC) const {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-
- std::map<WebRtc_UWord32, RTCPCnameInformation*>::const_iterator it =
- _receivedCnameMap.find(remoteSSRC);
-
- if (it == _receivedCnameMap.end()) {
- return NULL;
- }
- return it->second;
-}
-
-RTCPReceiveInformation*
-RTCPReceiver::CreateReceiveInformation(WebRtc_UWord32 remoteSSRC) {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-
- std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator it =
- _receivedInfoMap.find(remoteSSRC);
-
- if (it != _receivedInfoMap.end()) {
- return it->second;
- }
- RTCPReceiveInformation* receiveInfo = new RTCPReceiveInformation;
- _receivedInfoMap[remoteSSRC] = receiveInfo;
- return receiveInfo;
-}
-
-RTCPReceiveInformation*
-RTCPReceiver::GetReceiveInformation(WebRtc_UWord32 remoteSSRC) {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-
- std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator it =
- _receivedInfoMap.find(remoteSSRC);
- if (it == _receivedInfoMap.end()) {
- return NULL;
- }
- return it->second;
-}
-
-void RTCPReceiver::UpdateReceiveInformation(
- RTCPReceiveInformation& receiveInformation) {
- // Update that this remote is alive
- receiveInformation.lastTimeReceived = _clock.GetTimeInMS();
-}
-
-bool RTCPReceiver::UpdateRTCPReceiveInformationTimers() {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-
- bool updateBoundingSet = false;
- WebRtc_UWord32 timeNow = _clock.GetTimeInMS();
-
- std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator receiveInfoIt =
- _receivedInfoMap.begin();
-
- while (receiveInfoIt != _receivedInfoMap.end()) {
- RTCPReceiveInformation* receiveInfo = receiveInfoIt->second;
- if (receiveInfo == NULL) {
- return updateBoundingSet;
- }
- // time since last received rtcp packet
- // when we dont have a lastTimeReceived and the object is marked
- // readyForDelete it's removed from the map
- if (receiveInfo->lastTimeReceived) {
- /// use audio define since we don't know what interval the remote peer is
- // using
- if ((timeNow - receiveInfo->lastTimeReceived) >
- 5 * RTCP_INTERVAL_AUDIO_MS) {
- // no rtcp packet for the last five regular intervals, reset limitations
- receiveInfo->TmmbrSet.lengthOfSet = 0;
- // prevent that we call this over and over again
- receiveInfo->lastTimeReceived = 0;
- // send new TMMBN to all channels using the default codec
- updateBoundingSet = true;
- }
- receiveInfoIt++;
- } else if (receiveInfo->readyForDelete) {
- // store our current receiveInfoItem
- std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator
- receiveInfoItemToBeErased = receiveInfoIt;
- receiveInfoIt++;
- delete receiveInfoItemToBeErased->second;
- _receivedInfoMap.erase(receiveInfoItemToBeErased);
- } else {
- receiveInfoIt++;
- }
- }
- return updateBoundingSet;
-}
-
-WebRtc_Word32 RTCPReceiver::BoundingSet(bool &tmmbrOwner,
- TMMBRSet*& boundingSetRec) {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-
- std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator receiveInfoIt =
- _receivedInfoMap.find(_remoteSSRC);
-
- if (receiveInfoIt == _receivedInfoMap.end()) {
- return -1;
- }
- RTCPReceiveInformation* receiveInfo = receiveInfoIt->second;
- if (receiveInfo == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s failed to get RTCPReceiveInformation",
- __FUNCTION__);
- return -1;
- }
- if (receiveInfo->TmmbnBoundingSet.lengthOfSet > 0) {
- boundingSetRec->VerifyAndAllocateSet(
- receiveInfo->TmmbnBoundingSet.lengthOfSet + 1);
- for(WebRtc_UWord32 i=0; i< receiveInfo->TmmbnBoundingSet.lengthOfSet; i++) {
- if(receiveInfo->TmmbnBoundingSet.ptrSsrcSet[i] == _SSRC) {
- // owner of bounding set
- tmmbrOwner = true;
- }
- boundingSetRec->ptrTmmbrSet[i] =
- receiveInfo->TmmbnBoundingSet.ptrTmmbrSet[i];
- boundingSetRec->ptrPacketOHSet[i] =
- receiveInfo->TmmbnBoundingSet.ptrPacketOHSet[i];
- boundingSetRec->ptrSsrcSet[i] =
- receiveInfo->TmmbnBoundingSet.ptrSsrcSet[i];
- }
- }
- return receiveInfo->TmmbnBoundingSet.lengthOfSet;
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandleSDES(RTCPUtility::RTCPParserV2& rtcpParser)
-{
- RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
- while (pktType == RTCPUtility::kRtcpSdesChunkCode)
- {
- HandleSDESChunk(rtcpParser);
- pktType = rtcpParser.Iterate();
- }
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void RTCPReceiver::HandleSDESChunk(RTCPUtility::RTCPParserV2& rtcpParser) {
- const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
- RTCPCnameInformation* cnameInfo =
- CreateCnameInformation(rtcpPacket.CName.SenderSSRC);
- assert(cnameInfo);
-
- cnameInfo->name[RTCP_CNAME_SIZE - 1] = 0;
- strncpy(cnameInfo->name, rtcpPacket.CName.CName, RTCP_CNAME_SIZE - 1);
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandleNACK(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPPacketInformation& rtcpPacketInformation)
-{
- const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
-
- RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.NACK.SenderSSRC);
- if (ptrReceiveInfo == NULL)
- {
- // This remote SSRC must be saved before.
- rtcpParser.Iterate();
- return;
- }
- if (_SSRC != rtcpPacket.NACK.MediaSSRC)
- {
- // Not to us.
- rtcpParser.Iterate();
- return;
- }
-
- rtcpPacketInformation.ResetNACKPacketIdArray();
-
- RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
- while (pktType == RTCPUtility::kRtcpRtpfbNackItemCode)
- {
- HandleNACKItem(rtcpPacket, rtcpPacketInformation);
- pktType = rtcpParser.Iterate();
- }
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandleNACKItem(const RTCPUtility::RTCPPacket& rtcpPacket,
- RTCPPacketInformation& rtcpPacketInformation)
-{
- rtcpPacketInformation.AddNACKPacket(rtcpPacket.NACKItem.PacketID);
-
- WebRtc_UWord16 bitMask = rtcpPacket.NACKItem.BitMask;
- if(bitMask)
- {
- for(int i=1; i <= 16; ++i)
- {
- if(bitMask & 0x01)
- {
- rtcpPacketInformation.AddNACKPacket(rtcpPacket.NACKItem.PacketID + i);
- }
- bitMask = bitMask >>1;
- }
- }
-
- rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpNack;
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void RTCPReceiver::HandleBYE(RTCPUtility::RTCPParserV2& rtcpParser) {
- const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
-
- // clear our lists
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
- std::map<WebRtc_UWord32, RTCPReportBlockInformation*>::iterator
- reportBlockInfoIt = _receivedReportBlockMap.find(
- rtcpPacket.BYE.SenderSSRC);
-
- if (reportBlockInfoIt != _receivedReportBlockMap.end()) {
- delete reportBlockInfoIt->second;
- _receivedReportBlockMap.erase(reportBlockInfoIt);
- }
- // we can't delete it due to TMMBR
- std::map<WebRtc_UWord32, RTCPReceiveInformation*>::iterator receiveInfoIt =
- _receivedInfoMap.find(rtcpPacket.BYE.SenderSSRC);
-
- if (receiveInfoIt != _receivedInfoMap.end()) {
- receiveInfoIt->second->readyForDelete = true;
- }
-
- std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator cnameInfoIt =
- _receivedCnameMap.find(rtcpPacket.BYE.SenderSSRC);
-
- if (cnameInfoIt != _receivedCnameMap.end()) {
- delete cnameInfoIt->second;
- _receivedCnameMap.erase(cnameInfoIt);
- }
- rtcpParser.Iterate();
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandleXRVOIPMetric(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPPacketInformation& rtcpPacketInformation)
-{
- const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
-
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-
- if(rtcpPacket.XRVOIPMetricItem.SSRC == _SSRC)
- {
- // Store VoIP metrics block if it's about me
- // from OriginatorSSRC do we filter it?
- // rtcpPacket.XR.OriginatorSSRC;
-
- RTCPVoIPMetric receivedVoIPMetrics;
- receivedVoIPMetrics.burstDensity = rtcpPacket.XRVOIPMetricItem.burstDensity;
- receivedVoIPMetrics.burstDuration = rtcpPacket.XRVOIPMetricItem.burstDuration;
- receivedVoIPMetrics.discardRate = rtcpPacket.XRVOIPMetricItem.discardRate;
- receivedVoIPMetrics.endSystemDelay = rtcpPacket.XRVOIPMetricItem.endSystemDelay;
- receivedVoIPMetrics.extRfactor = rtcpPacket.XRVOIPMetricItem.extRfactor;
- receivedVoIPMetrics.gapDensity = rtcpPacket.XRVOIPMetricItem.gapDensity;
- receivedVoIPMetrics.gapDuration = rtcpPacket.XRVOIPMetricItem.gapDuration;
- receivedVoIPMetrics.Gmin = rtcpPacket.XRVOIPMetricItem.Gmin;
- receivedVoIPMetrics.JBabsMax = rtcpPacket.XRVOIPMetricItem.JBabsMax;
- receivedVoIPMetrics.JBmax = rtcpPacket.XRVOIPMetricItem.JBmax;
- receivedVoIPMetrics.JBnominal = rtcpPacket.XRVOIPMetricItem.JBnominal;
- receivedVoIPMetrics.lossRate = rtcpPacket.XRVOIPMetricItem.lossRate;
- receivedVoIPMetrics.MOSCQ = rtcpPacket.XRVOIPMetricItem.MOSCQ;
- receivedVoIPMetrics.MOSLQ = rtcpPacket.XRVOIPMetricItem.MOSLQ;
- receivedVoIPMetrics.noiseLevel = rtcpPacket.XRVOIPMetricItem.noiseLevel;
- receivedVoIPMetrics.RERL = rtcpPacket.XRVOIPMetricItem.RERL;
- receivedVoIPMetrics.Rfactor = rtcpPacket.XRVOIPMetricItem.Rfactor;
- receivedVoIPMetrics.roundTripDelay = rtcpPacket.XRVOIPMetricItem.roundTripDelay;
- receivedVoIPMetrics.RXconfig = rtcpPacket.XRVOIPMetricItem.RXconfig;
- receivedVoIPMetrics.signalLevel = rtcpPacket.XRVOIPMetricItem.signalLevel;
-
- rtcpPacketInformation.AddVoIPMetric(&receivedVoIPMetrics);
-
- rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpXrVoipMetric; // received signal
- }
- rtcpParser.Iterate();
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandlePLI(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPPacketInformation& rtcpPacketInformation)
-{
- const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
-
- RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.PLI.SenderSSRC);
- if (ptrReceiveInfo == NULL)
- {
- // This remote SSRC must be saved before.
- rtcpParser.Iterate();
- return;
- }
- if (_SSRC != rtcpPacket.PLI.MediaSSRC)
- {
- // Not to us.
- rtcpParser.Iterate();
- return;
- }
- rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpPli; // received signal that we need to send a new key frame
- rtcpParser.Iterate();
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandleTMMBR(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPPacketInformation& rtcpPacketInformation)
-{
- const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
-
- WebRtc_UWord32 senderSSRC = rtcpPacket.TMMBR.SenderSSRC;
- RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(senderSSRC);
- if (ptrReceiveInfo == NULL)
- {
- // This remote SSRC must be saved before.
- rtcpParser.Iterate();
- return;
- }
- if(rtcpPacket.TMMBR.MediaSSRC)
- {
- // rtcpPacket.TMMBR.MediaSSRC SHOULD be 0 if same as SenderSSRC
- // in relay mode this is a valid number
- senderSSRC = rtcpPacket.TMMBR.MediaSSRC;
- }
-
- // Use packet length to calc max number of TMMBR blocks
- // each TMMBR block is 8 bytes
- ptrdiff_t maxNumOfTMMBRBlocks = rtcpParser.LengthLeft() / 8;
-
- // sanity
- if(maxNumOfTMMBRBlocks > 200) // we can't have more than what's in one packet
- {
- assert(false);
- rtcpParser.Iterate();
- return;
- }
- ptrReceiveInfo->VerifyAndAllocateTMMBRSet((WebRtc_UWord32)maxNumOfTMMBRBlocks);
-
- RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
- while (pktType == RTCPUtility::kRtcpRtpfbTmmbrItemCode)
- {
- HandleTMMBRItem(*ptrReceiveInfo, rtcpPacket, rtcpPacketInformation, senderSSRC);
- pktType = rtcpParser.Iterate();
- }
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandleTMMBRItem(RTCPReceiveInformation& receiveInfo,
- const RTCPUtility::RTCPPacket& rtcpPacket,
- RTCPPacketInformation& rtcpPacketInformation,
- const WebRtc_UWord32 senderSSRC)
-{
- if (_SSRC == rtcpPacket.TMMBRItem.SSRC &&
- rtcpPacket.TMMBRItem.MaxTotalMediaBitRate > 0)
- {
- receiveInfo.InsertTMMBRItem(senderSSRC, rtcpPacket.TMMBRItem,
- _clock.GetTimeInMS());
- rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpTmmbr;
- }
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandleTMMBN(RTCPUtility::RTCPParserV2& rtcpParser)
-{
- const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
- RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.TMMBN.SenderSSRC);
- if (ptrReceiveInfo == NULL)
- {
- // This remote SSRC must be saved before.
- rtcpParser.Iterate();
- return;
- }
- // Use packet length to calc max number of TMMBN blocks
- // each TMMBN block is 8 bytes
- ptrdiff_t maxNumOfTMMBNBlocks = rtcpParser.LengthLeft() / 8;
-
- // sanity
- if(maxNumOfTMMBNBlocks > 200) // we cant have more than what's in one packet
- {
- assert(false);
- rtcpParser.Iterate();
- return;
- }
-
- ptrReceiveInfo->VerifyAndAllocateBoundingSet((WebRtc_UWord32)maxNumOfTMMBNBlocks);
-
- RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
- while (pktType == RTCPUtility::kRtcpRtpfbTmmbnItemCode)
- {
- HandleTMMBNItem(*ptrReceiveInfo, rtcpPacket);
- pktType = rtcpParser.Iterate();
- }
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandleSR_REQ(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPPacketInformation& rtcpPacketInformation)
-{
- rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpSrReq;
- rtcpParser.Iterate();
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandleTMMBNItem(RTCPReceiveInformation& receiveInfo,
- const RTCPUtility::RTCPPacket& rtcpPacket)
-{
- const unsigned int idx = receiveInfo.TmmbnBoundingSet.lengthOfSet;
-
- receiveInfo.TmmbnBoundingSet.ptrTmmbrSet[idx] = rtcpPacket.TMMBNItem.MaxTotalMediaBitRate;
- receiveInfo.TmmbnBoundingSet.ptrPacketOHSet[idx] = rtcpPacket.TMMBNItem.MeasuredOverhead;
- receiveInfo.TmmbnBoundingSet.ptrSsrcSet[idx] = rtcpPacket.TMMBNItem.SSRC;
-
- ++receiveInfo.TmmbnBoundingSet.lengthOfSet;
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandleSLI(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPPacketInformation& rtcpPacketInformation)
-{
- const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
-
- RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.SLI.SenderSSRC);
- if (ptrReceiveInfo == NULL)
- {
- // This remote SSRC must be saved before.
- rtcpParser.Iterate();
- return;
- }
-
- RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
- while (pktType == RTCPUtility::kRtcpPsfbSliItemCode)
- {
- HandleSLIItem(rtcpPacket, rtcpPacketInformation);
- pktType = rtcpParser.Iterate();
- }
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandleSLIItem(const RTCPUtility::RTCPPacket& rtcpPacket,
- RTCPPacketInformation& rtcpPacketInformation)
-{
- // in theory there could be multiple slices lost
- rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpSli; // received signal that we need to refresh a slice
- rtcpPacketInformation.sliPictureId = rtcpPacket.SLIItem.PictureId;
-}
-
-void
-RTCPReceiver::HandleRPSI(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation)
-{
- const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
-
- RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.RPSI.SenderSSRC);
- if (ptrReceiveInfo == NULL)
- {
- // This remote SSRC must be saved before.
- rtcpParser.Iterate();
- return;
- }
- RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
- if(pktType == RTCPUtility::kRtcpPsfbRpsiCode)
- {
- rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRpsi; // received signal that we have a confirmed reference picture
- if(rtcpPacket.RPSI.NumberOfValidBits%8 != 0)
- {
- // to us unknown
- // continue
- rtcpParser.Iterate();
- return;
- }
- rtcpPacketInformation.rpsiPictureId = 0;
-
- // convert NativeBitString to rpsiPictureId
- WebRtc_UWord8 numberOfBytes = rtcpPacket.RPSI.NumberOfValidBits /8;
- for(WebRtc_UWord8 n = 0; n < (numberOfBytes-1); n++)
- {
- rtcpPacketInformation.rpsiPictureId += (rtcpPacket.RPSI.NativeBitString[n] & 0x7f);
- rtcpPacketInformation.rpsiPictureId <<= 7; // prepare next
- }
- rtcpPacketInformation.rpsiPictureId += (rtcpPacket.RPSI.NativeBitString[numberOfBytes-1] & 0x7f);
- }
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandlePsfbApp(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPPacketInformation& rtcpPacketInformation)
-{
- RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
- if (pktType == RTCPUtility::kRtcpPsfbRembItemCode)
- {
- HandleREMBItem(rtcpParser, rtcpPacketInformation);
- rtcpParser.Iterate();
- }
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandleIJ(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPPacketInformation& rtcpPacketInformation)
-{
- const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
-
- RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
- while (pktType == RTCPUtility::kRtcpExtendedIjItemCode)
- {
- HandleIJItem(rtcpPacket, rtcpPacketInformation);
- pktType = rtcpParser.Iterate();
- }
-}
-
-void
-RTCPReceiver::HandleIJItem(const RTCPUtility::RTCPPacket& rtcpPacket,
- RTCPPacketInformation& rtcpPacketInformation)
-{
- rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpTransmissionTimeOffset;
- rtcpPacketInformation.interArrivalJitter =
- rtcpPacket.ExtendedJitterReportItem.Jitter;
-}
-
-void
-RTCPReceiver::HandleREMBItem(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPPacketInformation& rtcpPacketInformation)
-{
- rtcpParser.Iterate();
- const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
-
- rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpRemb;
- rtcpPacketInformation.receiverEstimatedMaxBitrate = rtcpPacket.REMB.BitRate;
- // TODO(pwestin) send up SSRCs and do a sanity check
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandleFIR(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPPacketInformation& rtcpPacketInformation)
-{
- const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
-
- RTCPReceiveInformation* ptrReceiveInfo = GetReceiveInformation(rtcpPacket.FIR.SenderSSRC);
- if (ptrReceiveInfo == NULL)
- {
- // This remote SSRC must be saved before.
- rtcpParser.Iterate();
- return;
- }
-
- RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Iterate();
- while (pktType == RTCPUtility::kRtcpPsfbFirItemCode)
- {
- HandleFIRItem(*ptrReceiveInfo, rtcpPacket, rtcpPacketInformation);
- pktType = rtcpParser.Iterate();
- }
-}
-
-// no need for critsect we have _criticalSectionRTCPReceiver
-void
-RTCPReceiver::HandleFIRItem(RTCPReceiveInformation& receiveInfo,
- const RTCPUtility::RTCPPacket& rtcpPacket,
- RTCPPacketInformation& rtcpPacketInformation)
-{
- if (_SSRC == rtcpPacket.FIRItem.SSRC) // is it our sender that is requested to generate a new keyframe
- {
- // rtcpPacket.FIR.MediaSSRC SHOULD be 0 but we ignore to check it
- // we don't know who this originate from
-
- // check if we have reported this FIRSequenceNumber before
- if (rtcpPacket.FIRItem.CommandSequenceNumber != receiveInfo.lastFIRSequenceNumber)
- {
- //
- WebRtc_UWord32 now = _clock.GetTimeInMS();
-
- // extra sanity don't go crazy with the callbacks
- if( (now - receiveInfo.lastFIRRequest) > RTCP_MIN_FRAME_LENGTH_MS)
- {
- receiveInfo.lastFIRRequest = now;
- receiveInfo.lastFIRSequenceNumber = rtcpPacket.FIRItem.CommandSequenceNumber;
-
- rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpFir; // received signal that we need to send a new key frame
- }
- }
- }
-}
-
-void
-RTCPReceiver::HandleAPP(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPPacketInformation& rtcpPacketInformation)
-{
- const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
-
- rtcpPacketInformation.rtcpPacketTypeFlags |= kRtcpApp;
- rtcpPacketInformation.applicationSubType = rtcpPacket.APP.SubType;
- rtcpPacketInformation.applicationName = rtcpPacket.APP.Name;
-
- rtcpParser.Iterate();
-}
-
-void
-RTCPReceiver::HandleAPPItem(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPPacketInformation& rtcpPacketInformation)
-{
- const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
-
- rtcpPacketInformation.AddApplicationData(rtcpPacket.APP.Data, rtcpPacket.APP.Size);
-
- rtcpParser.Iterate();
-}
-
-void
-RTCPReceiver::OnReceivedIntraFrameRequest(const FrameType frameType,
- const WebRtc_UWord8 streamIdx) const
-{
- CriticalSectionScoped lock(_criticalSectionFeedbacks);
-
- if(_cbVideoFeedback)
- {
- _cbVideoFeedback->OnReceivedIntraFrameRequest(_id, frameType, streamIdx);
- }
-}
-
-void
-RTCPReceiver::OnReceivedSliceLossIndication(const WebRtc_UWord8 pitureID) const
-{
- CriticalSectionScoped lock(_criticalSectionFeedbacks);
-
- if(_cbRtcpFeedback)
- {
- _cbRtcpFeedback->OnSLIReceived(_id, pitureID);
- }
-}
-
-void
-RTCPReceiver::OnReceivedReferencePictureSelectionIndication(const WebRtc_UWord64 pitureID) const
-{
- CriticalSectionScoped lock(_criticalSectionFeedbacks);
-
- if(_cbRtcpFeedback)
- {
- _cbRtcpFeedback->OnRPSIReceived(_id, pitureID);
- }
-}
-
-// Holding no Critical section
-void RTCPReceiver::TriggerCallbacksFromRTCPPacket(
- RTCPPacketInformation& rtcpPacketInformation)
-{
- // Process TMMBR and REMB first to avoid multiple callbacks
- // to OnNetworkChanged.
- if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpTmmbr)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
- "SIG [RTCP] Incoming TMMBR to id:%d", _id);
-
- // Might trigger a OnReceivedBandwidthEstimateUpdate.
- _rtpRtcp.OnReceivedTMMBR();
- }
- if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
- "SIG [RTCP] Incoming REMB to id:%d", _id);
-
- // We need to bounce this to the default channel.
- _rtpRtcp.OnReceivedEstimatedMaxBitrate(
- rtcpPacketInformation.receiverEstimatedMaxBitrate);
- }
- if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr ||
- rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRr)
- {
- if (rtcpPacketInformation.reportBlock)
- {
- _rtpRtcp.OnPacketLossStatisticsUpdate(
- rtcpPacketInformation.fractionLost,
- rtcpPacketInformation.roundTripTime,
- rtcpPacketInformation.lastReceivedExtendedHighSeqNum);
- }
- }
- if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr)
- {
- _rtpRtcp.OnReceivedNTP();
- }
- if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSrReq)
- {
- _rtpRtcp.OnRequestSendReport();
- }
- if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpNack)
- {
- if (rtcpPacketInformation.nackSequenceNumbersLength > 0)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
- "SIG [RTCP] Incoming NACK to id:%d", _id);
- _rtpRtcp.OnReceivedNACK(
- rtcpPacketInformation.nackSequenceNumbersLength,
- rtcpPacketInformation.nackSequenceNumbers);
- }
- }
- if ((rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli) ||
- (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpFir))
- {
- if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpPli)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
- "SIG [RTCP] Incoming PLI to id:%d", _id);
- } else
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceRtpRtcp, _id,
- "SIG [RTCP] Incoming FIR to id:%d", _id);
- }
- _rtpRtcp.OnReceivedIntraFrameRequest(&_rtpRtcp);
- }
- if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSli)
- {
- // we need use a bounce it up to handle default channel
- _rtpRtcp.OnReceivedSliceLossIndication(
- rtcpPacketInformation.sliPictureId);
- }
- if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRpsi)
- {
- // we need use a bounce it up to handle default channel
- _rtpRtcp.OnReceivedReferencePictureSelectionIndication(
- rtcpPacketInformation.rpsiPictureId);
- }
- {
- CriticalSectionScoped lock(_criticalSectionFeedbacks);
-
- // we need a feedback that we have received a report block(s) so that we can generate a new packet
- // in a conference relay scenario, one received report can generate several RTCP packets, based
- // on number relayed/mixed
- // a send report block should go out to all receivers
- if(_cbRtcpFeedback)
- {
- if(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpSr)
- {
- _cbRtcpFeedback->OnSendReportReceived(_id, rtcpPacketInformation.remoteSSRC);
- } else
- {
- _cbRtcpFeedback->OnReceiveReportReceived(_id, rtcpPacketInformation.remoteSSRC);
- }
- if (rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpRemb)
- {
- _cbRtcpFeedback->OnReceiverEstimatedMaxBitrateReceived(_id,
- rtcpPacketInformation.receiverEstimatedMaxBitrate);
- }
- if(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpXrVoipMetric)
- {
- WebRtc_Word8 VoIPmetricBuffer[7*4];
- VoIPmetricBuffer[0] = rtcpPacketInformation.VoIPMetric->lossRate;
- VoIPmetricBuffer[1] = rtcpPacketInformation.VoIPMetric->discardRate;
- VoIPmetricBuffer[2] = rtcpPacketInformation.VoIPMetric->burstDensity;
- VoIPmetricBuffer[3] = rtcpPacketInformation.VoIPMetric->gapDensity;
-
- VoIPmetricBuffer[4] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->burstDuration >> 8);
- VoIPmetricBuffer[5] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->burstDuration);
- VoIPmetricBuffer[6] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->gapDuration >> 8);
- VoIPmetricBuffer[7] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->gapDuration);
-
- VoIPmetricBuffer[8] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->roundTripDelay >> 8);
- VoIPmetricBuffer[9] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->roundTripDelay);
- VoIPmetricBuffer[10] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->endSystemDelay >> 8);
- VoIPmetricBuffer[11] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->endSystemDelay);
-
- VoIPmetricBuffer[12] = rtcpPacketInformation.VoIPMetric->signalLevel;
- VoIPmetricBuffer[13] = rtcpPacketInformation.VoIPMetric->noiseLevel;
- VoIPmetricBuffer[14] = rtcpPacketInformation.VoIPMetric->RERL;
- VoIPmetricBuffer[15] = rtcpPacketInformation.VoIPMetric->Gmin;
-
- VoIPmetricBuffer[16] = rtcpPacketInformation.VoIPMetric->Rfactor;
- VoIPmetricBuffer[17] = rtcpPacketInformation.VoIPMetric->extRfactor;
- VoIPmetricBuffer[18] = rtcpPacketInformation.VoIPMetric->MOSLQ;
- VoIPmetricBuffer[19] = rtcpPacketInformation.VoIPMetric->MOSCQ;
-
- VoIPmetricBuffer[20] = rtcpPacketInformation.VoIPMetric->RXconfig;
- VoIPmetricBuffer[21] = 0; // reserved
- VoIPmetricBuffer[22] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBnominal >> 8);
- VoIPmetricBuffer[23] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBnominal);
-
- VoIPmetricBuffer[24] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBmax >> 8);
- VoIPmetricBuffer[25] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBmax);
- VoIPmetricBuffer[26] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBabsMax >> 8);
- VoIPmetricBuffer[27] = (WebRtc_UWord8)(rtcpPacketInformation.VoIPMetric->JBabsMax);
-
- _cbRtcpFeedback->OnXRVoIPMetricReceived(_id, rtcpPacketInformation.VoIPMetric, VoIPmetricBuffer);
- }
- if(rtcpPacketInformation.rtcpPacketTypeFlags & kRtcpApp)
- {
- _cbRtcpFeedback->OnApplicationDataReceived(_id,
- rtcpPacketInformation.applicationSubType,
- rtcpPacketInformation.applicationName,
- rtcpPacketInformation.applicationLength,
- rtcpPacketInformation.applicationData);
- }
- }
- }
-}
-
-void
-RTCPReceiver::UpdateBandwidthEstimate(const WebRtc_UWord16 bwEstimateKbit)
-{
- CriticalSectionScoped lock(_criticalSectionFeedbacks);
-
- if(_cbRtcpFeedback)
- {
- _cbRtcpFeedback->OnTMMBRReceived(_id, bwEstimateKbit);
- }
-
-}
-
-WebRtc_Word32 RTCPReceiver::CNAME(const WebRtc_UWord32 remoteSSRC,
- char cName[RTCP_CNAME_SIZE]) const {
- if (cName == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s invalid argument", __FUNCTION__);
- return -1;
- }
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
- RTCPCnameInformation* cnameInfo = GetCnameInformation(remoteSSRC);
- assert(cnameInfo);
-
- cName[RTCP_CNAME_SIZE - 1] = 0;
- strncpy(cName, cnameInfo->name, RTCP_CNAME_SIZE - 1);
- return 0;
-}
-
-// no callbacks allowed inside this function
-WebRtc_Word32 RTCPReceiver::TMMBRReceived(const WebRtc_UWord32 size,
- const WebRtc_UWord32 accNumCandidates,
- TMMBRSet* candidateSet) const {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
-
- std::map<WebRtc_UWord32, RTCPReceiveInformation*>::const_iterator
- receiveInfoIt = _receivedInfoMap.begin();
- if (receiveInfoIt == _receivedInfoMap.end()) {
- return -1;
- }
- WebRtc_UWord32 num = accNumCandidates;
- if (candidateSet) {
- while( num < size && receiveInfoIt != _receivedInfoMap.end()) {
- RTCPReceiveInformation* receiveInfo = receiveInfoIt->second;
- if (receiveInfo == NULL) {
- return 0;
- }
- for (WebRtc_UWord32 i = 0;
- (num < size) && (i < receiveInfo->TmmbrSet.lengthOfSet); i++) {
- if (receiveInfo->GetTMMBRSet(i, num, candidateSet,
- _clock.GetTimeInMS()) == 0) {
- num++;
- }
- }
- receiveInfoIt++;
- }
- } else {
- while (receiveInfoIt != _receivedInfoMap.end()) {
- RTCPReceiveInformation* receiveInfo = receiveInfoIt->second;
- if(receiveInfo == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s failed to get RTCPReceiveInformation",
- __FUNCTION__);
- return -1;
- }
- num += receiveInfo->TmmbrSet.lengthOfSet;
- receiveInfoIt++;
- }
- }
- return num;
-}
-
-WebRtc_Word32
-RTCPReceiver::SetPacketTimeout(const WebRtc_UWord32 timeoutMS)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
- _packetTimeOutMS = timeoutMS;
- return 0;
-}
-
-void RTCPReceiver::PacketTimeout()
-{
- if(_packetTimeOutMS == 0)
- {
- // not configured
- return;
- }
-
- bool packetTimeOut = false;
- {
- CriticalSectionScoped lock(_criticalSectionRTCPReceiver);
- if(_lastReceived == 0)
- {
- // not active
- return;
- }
-
- WebRtc_UWord32 now = _clock.GetTimeInMS();
-
- if(now - _lastReceived > _packetTimeOutMS)
- {
- packetTimeOut = true;
- _lastReceived = 0; // only one callback
- }
- }
- CriticalSectionScoped lock(_criticalSectionFeedbacks);
- if(packetTimeOut && _cbRtcpFeedback)
- {
- _cbRtcpFeedback->OnRTCPPacketTimeout(_id);
- }
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_receiver.h b/trunk/src/modules/rtp_rtcp/source/rtcp_receiver.h
deleted file mode 100644
index eb1b658..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtcp_receiver.h
+++ /dev/null
@@ -1,233 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_
-
-#include <map>
-#include <vector>
-
-#include "typedefs.h"
-#include "rtp_utility.h"
-#include "rtcp_utility.h"
-#include "rtp_rtcp_defines.h"
-#include "rtcp_receiver_help.h"
-
-namespace webrtc {
-class ModuleRtpRtcpImpl;
-
-class RTCPReceiver
-{
-public:
- RTCPReceiver(const WebRtc_Word32 id, RtpRtcpClock* clock,
- ModuleRtpRtcpImpl* owner);
- virtual ~RTCPReceiver();
-
- void ChangeUniqueId(const WebRtc_Word32 id);
-
- RTCPMethod Status() const;
- WebRtc_Word32 SetRTCPStatus(const RTCPMethod method);
-
- WebRtc_UWord32 LastReceived();
-
- void SetSSRC( const WebRtc_UWord32 ssrc);
- void SetRelaySSRC( const WebRtc_UWord32 ssrc);
- WebRtc_Word32 SetRemoteSSRC( const WebRtc_UWord32 ssrc);
-
- WebRtc_UWord32 RelaySSRC() const;
-
- WebRtc_Word32 RegisterIncomingRTCPCallback(RtcpFeedback* incomingMessagesCallback);
-
- WebRtc_Word32 RegisterIncomingVideoCallback(RtpVideoFeedback* incomingMessagesCallback);
-
- WebRtc_Word32 IncomingRTCPPacket(RTCPHelp::RTCPPacketInformation& rtcpPacketInformation,
- RTCPUtility::RTCPParserV2 *rtcpParser);
-
- void TriggerCallbacksFromRTCPPacket(RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- // get received cname
- WebRtc_Word32 CNAME(const WebRtc_UWord32 remoteSSRC,
- char cName[RTCP_CNAME_SIZE]) const;
-
- // get received NTP
- WebRtc_Word32 NTP(WebRtc_UWord32 *ReceivedNTPsecs,
- WebRtc_UWord32 *ReceivedNTPfrac,
- WebRtc_UWord32 *RTCPArrivalTimeSecs,
- WebRtc_UWord32 *RTCPArrivalTimeFrac) const;
-
- // get rtt
- WebRtc_Word32 RTT(const WebRtc_UWord32 remoteSSRC,
- WebRtc_UWord16* RTT,
- WebRtc_UWord16* avgRTT,
- WebRtc_UWord16* minRTT,
- WebRtc_UWord16* maxRTT) const;
-
- WebRtc_UWord16 RTT() const;
-
- int SetRTT(WebRtc_UWord16 rtt);
-
- WebRtc_Word32 ResetRTT(const WebRtc_UWord32 remoteSSRC);
-
- void UpdateLipSync(const WebRtc_Word32 audioVideoOffset) const;
-
- WebRtc_Word32 SenderInfoReceived(RTCPSenderInfo* senderInfo) const;
-
- void OnReceivedIntraFrameRequest(const FrameType frameType,
- const WebRtc_UWord8 streamIdx) const;
-
- void OnReceivedSliceLossIndication(const WebRtc_UWord8 pitureID) const;
- void OnReceivedReferencePictureSelectionIndication(
- const WebRtc_UWord64 pitureID) const;
-
- // get statistics
- WebRtc_Word32 StatisticsReceived(
- std::vector<RTCPReportBlock>* receiveBlocks) const;
-
- // Get TMMBR
- WebRtc_Word32 TMMBRReceived(const WebRtc_UWord32 size,
- const WebRtc_UWord32 accNumCandidates,
- TMMBRSet* candidateSet) const;
-
- bool UpdateRTCPReceiveInformationTimers();
-
- void UpdateBandwidthEstimate(const WebRtc_UWord16 bwEstimateKbit);
-
- WebRtc_Word32 BoundingSet(bool &tmmbrOwner,
- TMMBRSet*& boundingSetRec);
-
- WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 timeoutMS);
- void PacketTimeout();
-
-protected:
- RTCPHelp::RTCPReportBlockInformation* CreateReportBlockInformation(const WebRtc_UWord32 remoteSSRC);
- RTCPHelp::RTCPReportBlockInformation* GetReportBlockInformation(const WebRtc_UWord32 remoteSSRC) const;
-
- RTCPUtility::RTCPCnameInformation* CreateCnameInformation(const WebRtc_UWord32 remoteSSRC);
- RTCPUtility::RTCPCnameInformation* GetCnameInformation(const WebRtc_UWord32 remoteSSRC) const;
-
- RTCPHelp::RTCPReceiveInformation* CreateReceiveInformation(const WebRtc_UWord32 remoteSSRC);
- RTCPHelp::RTCPReceiveInformation* GetReceiveInformation(const WebRtc_UWord32 remoteSSRC);
-
- void UpdateReceiveInformation( RTCPHelp::RTCPReceiveInformation& receiveInformation);
-
- void HandleSenderReceiverReport(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- void HandleReportBlock(const RTCPUtility::RTCPPacket& rtcpPacket,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation,
- const WebRtc_UWord32 remoteSSRC,
- const WebRtc_UWord8 numberOfReportBlocks);
-
- void HandleSDES(RTCPUtility::RTCPParserV2& rtcpParser);
-
- void HandleSDESChunk(RTCPUtility::RTCPParserV2& rtcpParser);
-
- void HandleXRVOIPMetric(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- void HandleNACK(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- void HandleNACKItem(const RTCPUtility::RTCPPacket& rtcpPacket,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- void HandleBYE(RTCPUtility::RTCPParserV2& rtcpParser);
-
- void HandlePLI(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- void HandleSLI(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- void HandleSLIItem(const RTCPUtility::RTCPPacket& rtcpPacket,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- void HandleRPSI(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- void HandlePsfbApp(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- void HandleREMBItem(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- void HandleIJ(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- void HandleIJItem(const RTCPUtility::RTCPPacket& rtcpPacket,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- void HandleTMMBR(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- void HandleTMMBRItem(RTCPHelp::RTCPReceiveInformation& receiveInfo,
- const RTCPUtility::RTCPPacket& rtcpPacket,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation,
- const WebRtc_UWord32 senderSSRC);
-
- void HandleTMMBN(RTCPUtility::RTCPParserV2& rtcpParser);
-
- void HandleSR_REQ(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- void HandleTMMBNItem(RTCPHelp::RTCPReceiveInformation& receiveInfo,
- const RTCPUtility::RTCPPacket& rtcpPacket);
-
- void HandleFIR(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- void HandleFIRItem(RTCPHelp::RTCPReceiveInformation& receiveInfo,
- const RTCPUtility::RTCPPacket& rtcpPacket,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- void HandleAPP(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- void HandleAPPItem(RTCPUtility::RTCPParserV2& rtcpParser,
- RTCPHelp::RTCPPacketInformation& rtcpPacketInformation);
-
- private:
- WebRtc_Word32 _id;
- RtpRtcpClock& _clock;
- RTCPMethod _method;
- WebRtc_UWord32 _lastReceived;
- ModuleRtpRtcpImpl& _rtpRtcp;
-
- CriticalSectionWrapper* _criticalSectionFeedbacks;
- RtcpFeedback* _cbRtcpFeedback;
- RtpVideoFeedback* _cbVideoFeedback;
-
- CriticalSectionWrapper* _criticalSectionRTCPReceiver;
- WebRtc_UWord32 _SSRC;
- WebRtc_UWord32 _remoteSSRC;
-
- // Received send report
- RTCPSenderInfo _remoteSenderInfo;
- // when did we receive the last send report
- WebRtc_UWord32 _lastReceivedSRNTPsecs;
- WebRtc_UWord32 _lastReceivedSRNTPfrac;
-
- // Received report blocks.
- std::map<WebRtc_UWord32, RTCPHelp::RTCPReportBlockInformation*>
- _receivedReportBlockMap;
- std::map<WebRtc_UWord32, RTCPHelp::RTCPReceiveInformation*>
- _receivedInfoMap;
- std::map<WebRtc_UWord32, RTCPUtility::RTCPCnameInformation*>
- _receivedCnameMap;
-
- WebRtc_UWord32 _packetTimeOutMS;
-
- // Externally set RTT. This value can only be used if there are no valid
- // RTT estimates.
- WebRtc_UWord16 _rtt;
-
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_receiver_help.cc b/trunk/src/modules/rtp_rtcp/source/rtcp_receiver_help.cc
deleted file mode 100644
index 0058aca..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtcp_receiver_help.cc
+++ /dev/null
@@ -1,258 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rtcp_receiver_help.h"
-#include "rtp_utility.h"
-
-#include <string.h> //memset
-#include <cassert> //assert
-
-namespace webrtc {
-using namespace RTCPHelp;
-
-RTCPPacketInformation::RTCPPacketInformation()
- : rtcpPacketTypeFlags(0),
- remoteSSRC(0),
- nackSequenceNumbers(0),
- nackSequenceNumbersLength(0),
- applicationSubType(0),
- applicationName(0),
- applicationData(),
- applicationLength(0),
- reportBlock(false),
- fractionLost(0),
- roundTripTime(0),
- lastReceivedExtendedHighSeqNum(0),
- jitter(0),
- interArrivalJitter(0),
- sliPictureId(0),
- rpsiPictureId(0),
- receiverEstimatedMaxBitrate(0),
- VoIPMetric(NULL) {
-}
-
-RTCPPacketInformation::~RTCPPacketInformation()
-{
- delete [] nackSequenceNumbers;
- delete [] applicationData;
- delete VoIPMetric;
-}
-
-void
-RTCPPacketInformation::AddVoIPMetric(const RTCPVoIPMetric* metric)
-{
- VoIPMetric = new RTCPVoIPMetric();
- memcpy(VoIPMetric, metric, sizeof(RTCPVoIPMetric));
-}
-
-void
-RTCPPacketInformation::AddApplicationData(const WebRtc_UWord8* data, const WebRtc_UWord16 size)
-{
- WebRtc_UWord8* oldData = applicationData;
- WebRtc_UWord16 oldLength = applicationLength;
-
- // Don't copy more than kRtcpAppCode_DATA_SIZE bytes.
- WebRtc_UWord16 copySize = size;
- if (size > kRtcpAppCode_DATA_SIZE) {
- copySize = kRtcpAppCode_DATA_SIZE;
- }
-
- applicationLength += copySize;
- applicationData = new WebRtc_UWord8[applicationLength];
-
- if(oldData)
- {
- memcpy(applicationData, oldData, oldLength);
- memcpy(applicationData+oldLength, data, copySize);
- delete [] oldData;
- } else
- {
- memcpy(applicationData, data, copySize);
- }
-}
-
-void
-RTCPPacketInformation::ResetNACKPacketIdArray()
-{
- if(NULL == nackSequenceNumbers)
- {
- nackSequenceNumbers = new WebRtc_UWord16[NACK_PACKETS_MAX_SIZE];
- }
- nackSequenceNumbersLength = 0;
-}
-
-void
-RTCPPacketInformation::AddNACKPacket(const WebRtc_UWord16 packetID)
-{
- assert(nackSequenceNumbers);
-
- WebRtc_UWord16& idx = nackSequenceNumbersLength;
- if (idx < NACK_PACKETS_MAX_SIZE)
- {
- nackSequenceNumbers[idx++] = packetID;
- }
-}
-
-void
-RTCPPacketInformation::AddReportInfo(const WebRtc_UWord8 fraction,
- const WebRtc_UWord16 rtt,
- const WebRtc_UWord32 extendedHighSeqNum,
- const WebRtc_UWord32 j)
-{
- reportBlock = true;
- fractionLost = fraction;
- roundTripTime = rtt;
- jitter = j;
- lastReceivedExtendedHighSeqNum = extendedHighSeqNum;
-}
-
-RTCPReportBlockInformation::RTCPReportBlockInformation():
- remoteReceiveBlock(),
- remoteMaxJitter(0),
- RTT(0),
- minRTT(0),
- maxRTT(0),
- avgRTT(0),
- numAverageCalcs(0)
-{
- memset(&remoteReceiveBlock,0,sizeof(remoteReceiveBlock));
-}
-
-RTCPReportBlockInformation::~RTCPReportBlockInformation()
-{
-}
-
-RTCPReceiveInformation::RTCPReceiveInformation() :
-
- lastTimeReceived(0),
- lastFIRSequenceNumber(-1),
- lastFIRRequest(0),
- readyForDelete(false),
- _tmmbrSetTimeouts(NULL)
-{
-}
-
-RTCPReceiveInformation::~RTCPReceiveInformation()
-{
- if(_tmmbrSetTimeouts)
- {
- delete [] _tmmbrSetTimeouts;
- }
-}
-
-// don't use TmmbrSet.VerifyAndAllocate this version keeps the data
-void
-RTCPReceiveInformation::VerifyAndAllocateTMMBRSet(const WebRtc_UWord32 minimumSize)
-{
- if(minimumSize > TmmbrSet.sizeOfSet)
- {
- // make sure that our buffers are big enough
- WebRtc_UWord32* ptrTmmbrSet = new WebRtc_UWord32[minimumSize];
- WebRtc_UWord32* ptrTmmbrPacketOHSet = new WebRtc_UWord32[minimumSize];
- WebRtc_UWord32* ptrTmmbrSsrcSet = new WebRtc_UWord32[minimumSize];
- WebRtc_UWord32* tmmbrSetTimeouts = new WebRtc_UWord32[minimumSize];
-
- if(TmmbrSet.lengthOfSet > 0)
- {
- // copy old values
- memcpy(ptrTmmbrSet, TmmbrSet.ptrTmmbrSet, sizeof(WebRtc_UWord32) * TmmbrSet.lengthOfSet);
- memcpy(ptrTmmbrPacketOHSet, TmmbrSet.ptrPacketOHSet, sizeof(WebRtc_UWord32) * TmmbrSet.lengthOfSet);
- memcpy(ptrTmmbrSsrcSet, TmmbrSet.ptrSsrcSet, sizeof(WebRtc_UWord32) * TmmbrSet.lengthOfSet);
- memcpy(tmmbrSetTimeouts, _tmmbrSetTimeouts, sizeof(WebRtc_UWord32) * TmmbrSet.lengthOfSet);
- }
- if(TmmbrSet.ptrTmmbrSet)
- {
- delete [] TmmbrSet.ptrTmmbrSet;
- delete [] TmmbrSet.ptrPacketOHSet;
- delete [] TmmbrSet.ptrSsrcSet;
- }
- if(_tmmbrSetTimeouts)
- {
- delete [] _tmmbrSetTimeouts;
- }
- TmmbrSet.ptrTmmbrSet = ptrTmmbrSet;
- TmmbrSet.ptrPacketOHSet = ptrTmmbrPacketOHSet;
- TmmbrSet.ptrSsrcSet = ptrTmmbrSsrcSet;
- TmmbrSet.sizeOfSet = minimumSize;
- _tmmbrSetTimeouts = tmmbrSetTimeouts;
- }
-}
-
-void
-RTCPReceiveInformation::InsertTMMBRItem(const WebRtc_UWord32 senderSSRC,
- const RTCPUtility::RTCPPacketRTPFBTMMBRItem& TMMBRItem,
- const WebRtc_UWord32 currentTimeMS)
-{
- // serach to see if we have it in our list
- for(WebRtc_UWord32 i = 0; i < TmmbrSet.lengthOfSet; i++)
- {
- if(TmmbrSet.ptrSsrcSet[i] == senderSSRC)
- {
- // we already have this SSRC in our list
- // update it
- TmmbrSet.ptrPacketOHSet[i] = TMMBRItem.MeasuredOverhead;
- TmmbrSet.ptrTmmbrSet[i] = TMMBRItem.MaxTotalMediaBitRate;
- _tmmbrSetTimeouts[i] = currentTimeMS;
- return;
- }
- }
- VerifyAndAllocateTMMBRSet(TmmbrSet.lengthOfSet+1);
-
- const WebRtc_UWord32 idx = TmmbrSet.lengthOfSet;
- TmmbrSet.ptrPacketOHSet[idx] = TMMBRItem.MeasuredOverhead;
- TmmbrSet.ptrTmmbrSet[idx] = TMMBRItem.MaxTotalMediaBitRate;
- TmmbrSet.ptrSsrcSet[idx] = senderSSRC;
- _tmmbrSetTimeouts[idx] = currentTimeMS;
- TmmbrSet.lengthOfSet++;
-}
-
-WebRtc_Word32
-RTCPReceiveInformation::GetTMMBRSet(const WebRtc_UWord32 sourceIdx,
- const WebRtc_UWord32 targetIdx,
- TMMBRSet* candidateSet,
- const WebRtc_UWord32 currentTimeMS)
-{
- if(sourceIdx >= TmmbrSet.lengthOfSet)
- {
- return -1;
- }
- if(targetIdx >= candidateSet->sizeOfSet)
- {
- return -1;
- }
- WebRtc_UWord32 timeNow = currentTimeMS;
-
- // use audio define since we don't know what interval the remote peer is using
- if(timeNow - _tmmbrSetTimeouts[sourceIdx] > 5*RTCP_INTERVAL_AUDIO_MS)
- {
- // value timed out
- const WebRtc_UWord32 move = TmmbrSet.lengthOfSet - (sourceIdx + 1);
- if(move > 0)
- {
- memmove(&(TmmbrSet.ptrTmmbrSet[sourceIdx]), &(TmmbrSet.ptrTmmbrSet[sourceIdx+1]), move* sizeof(WebRtc_UWord32));
- memmove(&(TmmbrSet.ptrPacketOHSet[sourceIdx]),&(TmmbrSet.ptrPacketOHSet[sourceIdx+1]), move* sizeof(WebRtc_UWord32));
- memmove(&(TmmbrSet.ptrSsrcSet[sourceIdx]),&(TmmbrSet.ptrSsrcSet[sourceIdx+1]), move* sizeof(WebRtc_UWord32));
- memmove(&(_tmmbrSetTimeouts[sourceIdx]),&(_tmmbrSetTimeouts[sourceIdx+1]), move* sizeof(WebRtc_UWord32));
- }
- TmmbrSet.lengthOfSet--;
- return -1;
- }
-
- candidateSet->ptrTmmbrSet[targetIdx] = TmmbrSet.ptrTmmbrSet[sourceIdx];
- candidateSet->ptrPacketOHSet[targetIdx] = TmmbrSet.ptrPacketOHSet[sourceIdx];
- candidateSet->ptrSsrcSet[targetIdx] = TmmbrSet.ptrSsrcSet[sourceIdx];
- return 0;
-}
-
-void RTCPReceiveInformation::VerifyAndAllocateBoundingSet(const WebRtc_UWord32 minimumSize)
-{
- TmmbnBoundingSet.VerifyAndAllocateSet(minimumSize);
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_receiver_help.h b/trunk/src/modules/rtp_rtcp/source/rtcp_receiver_help.h
deleted file mode 100644
index 6dcaca8..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtcp_receiver_help.h
+++ /dev/null
@@ -1,127 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
-
-#include "typedefs.h"
-
-#include "rtp_rtcp_defines.h" // RTCPReportBlock
-#include "rtp_rtcp_config.h" // RTCP_MAX_REPORT_BLOCKS
-#include "rtcp_utility.h"
-#include "tmmbr_help.h"
-
-namespace webrtc {
-namespace RTCPHelp
-{
-
-class RTCPPacketInformation
-{
-public:
- RTCPPacketInformation();
- ~RTCPPacketInformation();
-
- void AddVoIPMetric(const RTCPVoIPMetric* metric);
-
- void AddApplicationData(const WebRtc_UWord8* data, const WebRtc_UWord16 size);
-
- void AddNACKPacket(const WebRtc_UWord16 packetID);
- void ResetNACKPacketIdArray();
-
- void AddReportInfo(const WebRtc_UWord8 fractionLost,
- const WebRtc_UWord16 rtt,
- const WebRtc_UWord32 extendedHighSeqNum,
- const WebRtc_UWord32 jitter);
-
- WebRtc_UWord32 rtcpPacketTypeFlags; // RTCPPacketTypeFlags bit field
- WebRtc_UWord32 remoteSSRC;
-
- WebRtc_UWord16* nackSequenceNumbers;
- WebRtc_UWord16 nackSequenceNumbersLength;
-
- WebRtc_UWord8 applicationSubType;
- WebRtc_UWord32 applicationName;
- WebRtc_UWord8* applicationData;
- WebRtc_UWord16 applicationLength;
-
- bool reportBlock;
- WebRtc_UWord8 fractionLost;
- WebRtc_UWord16 roundTripTime;
- WebRtc_UWord32 lastReceivedExtendedHighSeqNum;
- WebRtc_UWord32 jitter;
-
- WebRtc_UWord32 interArrivalJitter;
-
- WebRtc_UWord8 sliPictureId;
- WebRtc_UWord64 rpsiPictureId;
- WebRtc_UWord32 receiverEstimatedMaxBitrate;
-
- RTCPVoIPMetric* VoIPMetric;
-};
-
-
-class RTCPReportBlockInformation
-{
-public:
- RTCPReportBlockInformation();
- ~RTCPReportBlockInformation();
-
- // Statistics
- RTCPReportBlock remoteReceiveBlock;
- WebRtc_UWord32 remoteMaxJitter;
-
- // RTT
- WebRtc_UWord16 RTT;
- WebRtc_UWord16 minRTT;
- WebRtc_UWord16 maxRTT;
- WebRtc_UWord16 avgRTT;
- WebRtc_UWord32 numAverageCalcs;
-};
-
-class RTCPReceiveInformation
-{
-public:
- RTCPReceiveInformation();
- ~RTCPReceiveInformation();
-
- void VerifyAndAllocateBoundingSet(const WebRtc_UWord32 minimumSize);
- void VerifyAndAllocateTMMBRSet(const WebRtc_UWord32 minimumSize);
-
- void InsertTMMBRItem(const WebRtc_UWord32 senderSSRC,
- const RTCPUtility::RTCPPacketRTPFBTMMBRItem& TMMBRItem,
- const WebRtc_UWord32 currentTimeMS);
-
- // get
- WebRtc_Word32 GetTMMBRSet(const WebRtc_UWord32 sourceIdx,
- const WebRtc_UWord32 targetIdx,
- TMMBRSet* candidateSet,
- const WebRtc_UWord32 currentTimeMS);
-
- WebRtc_UWord32 lastTimeReceived;
-
- // FIR
- WebRtc_Word32 lastFIRSequenceNumber;
- WebRtc_UWord32 lastFIRRequest;
-
- // TMMBN
- TMMBRSet TmmbnBoundingSet;
-
- // TMMBR
- TMMBRSet TmmbrSet;
-
- bool readyForDelete;
-private:
- WebRtc_UWord32* _tmmbrSetTimeouts;
-};
-
-} // end namespace RTCPHelp
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_RECEIVER_HELP_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_sender.cc b/trunk/src/modules/rtp_rtcp/source/rtcp_sender.cc
deleted file mode 100644
index 58e849f..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtcp_sender.cc
+++ /dev/null
@@ -1,2189 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rtcp_sender.h"
-
-#include <string.h> // memcpy
-#include <cassert> // assert
-#include <cstdlib> // rand
-
-#include "trace.h"
-#include "common_types.h"
-#include "critical_section_wrapper.h"
-
-#include "rtp_rtcp_impl.h"
-
-namespace webrtc {
-
-using RTCPUtility::RTCPCnameInformation;
-
-RTCPSender::RTCPSender(const WebRtc_Word32 id,
- const bool audio,
- RtpRtcpClock* clock,
- ModuleRtpRtcpImpl* owner) :
- _id(id),
- _audio(audio),
- _clock(*clock),
- _method(kRtcpOff),
- _rtpRtcp(*owner),
- _criticalSectionTransport(CriticalSectionWrapper::CreateCriticalSection()),
- _cbTransport(NULL),
-
- _criticalSectionRTCPSender(CriticalSectionWrapper::CreateCriticalSection()),
- _usingNack(false),
- _sending(false),
- _sendTMMBN(false),
- _REMB(false),
- _sendREMB(false),
- _TMMBR(false),
- _IJ(false),
- _nextTimeToSendRTCP(0),
- _SSRC(0),
- _remoteSSRC(0),
- _CNAME(),
- _reportBlocks(),
- _csrcCNAMEs(),
-
- _cameraDelayMS(0),
-
- _lastSendReport(),
- _lastRTCPTime(),
-
- _CSRCs(0),
- _CSRC(),
- _includeCSRCs(true),
-
- _sequenceNumberFIR(0),
-
- _lengthRembSSRC(0),
- _sizeRembSSRC(0),
- _rembSSRC(NULL),
- _rembBitrate(0),
- _bitrate_observer(NULL),
-
- _tmmbrHelp(audio),
- _tmmbr_Send(0),
- _packetOH_Send(0),
- _remoteRateControl(),
-
- _appSend(false),
- _appSubType(0),
- _appName(),
- _appData(NULL),
- _appLength(0),
- _xrSendVoIPMetric(false),
- _xrVoIPMetric()
-{
- memset(_CNAME, 0, sizeof(_CNAME));
- memset(_lastSendReport, 0, sizeof(_lastSendReport));
- memset(_lastRTCPTime, 0, sizeof(_lastRTCPTime));
-
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
-}
-
-RTCPSender::~RTCPSender() {
- delete [] _rembSSRC;
- delete [] _appData;
-
- while (!_reportBlocks.empty()) {
- std::map<WebRtc_UWord32, RTCPReportBlock*>::iterator it =
- _reportBlocks.begin();
- delete it->second;
- _reportBlocks.erase(it);
- }
- while (!_csrcCNAMEs.empty()) {
- std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator it =
- _csrcCNAMEs.begin();
- delete it->second;
- _csrcCNAMEs.erase(it);
- }
- delete _criticalSectionTransport;
- delete _criticalSectionRTCPSender;
-
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__);
-}
-
-WebRtc_Word32
-RTCPSender::Init()
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
-
- _method = kRtcpOff;
- _cbTransport = NULL;
- _usingNack = false;
- _sending = false;
- _sendTMMBN = false;
- _TMMBR = false;
- _IJ = false;
- _REMB = false;
- _sendREMB = false;
- _SSRC = 0;
- _remoteSSRC = 0;
- _cameraDelayMS = 0;
- _sequenceNumberFIR = 0;
- _tmmbr_Send = 0;
- _packetOH_Send = 0;
- _remoteRateControl.Reset();
- _nextTimeToSendRTCP = 0;
- _CSRCs = 0;
- _appSend = false;
- _appSubType = 0;
-
- if(_appData)
- {
- delete [] _appData;
- _appData = NULL;
- }
- _appLength = 0;
-
- _xrSendVoIPMetric = false;
-
- memset(&_xrVoIPMetric, 0, sizeof(_xrVoIPMetric));
- memset(_CNAME, 0, sizeof(_CNAME));
- memset(_lastSendReport, 0, sizeof(_lastSendReport));
- memset(_lastRTCPTime, 0, sizeof(_lastRTCPTime));
- return 0;
-}
-
-void
-RTCPSender::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
-}
-
-WebRtc_Word32
-RTCPSender::RegisterSendTransport(Transport* outgoingTransport)
-{
- CriticalSectionScoped lock(_criticalSectionTransport);
- _cbTransport = outgoingTransport;
- return 0;
-}
-
-RTCPMethod
-RTCPSender::Status() const
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- return _method;
-}
-
-WebRtc_Word32
-RTCPSender::SetRTCPStatus(const RTCPMethod method)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- if(method != kRtcpOff)
- {
- if(_audio)
- {
- _nextTimeToSendRTCP = _clock.GetTimeInMS() + (RTCP_INTERVAL_AUDIO_MS/2);
- } else
- {
- _nextTimeToSendRTCP = _clock.GetTimeInMS() + (RTCP_INTERVAL_VIDEO_MS/2);
- }
- }
- _method = method;
- return 0;
-}
-
-bool
-RTCPSender::Sending() const
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- return _sending;
-}
-
-WebRtc_Word32
-RTCPSender::SetSendingStatus(const bool sending)
-{
- bool sendRTCPBye = false;
- {
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
-
- if(_method != kRtcpOff)
- {
- if(sending == false && _sending == true)
- {
- // Trigger RTCP bye
- sendRTCPBye = true;
- }
- }
- _sending = sending;
- }
- if(sendRTCPBye)
- {
- return SendRTCP(kRtcpBye);
- }
- return 0;
-}
-
-bool
-RTCPSender::REMB() const
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- return _REMB;
-}
-
-WebRtc_Word32
-RTCPSender::SetREMBStatus(const bool enable)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- _REMB = enable;
- return 0;
-}
-
-WebRtc_Word32
-RTCPSender::SetREMBData(const WebRtc_UWord32 bitrate,
- const WebRtc_UWord8 numberOfSSRC,
- const WebRtc_UWord32* SSRC)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- _rembBitrate = bitrate;
-
- if(_sizeRembSSRC < numberOfSSRC)
- {
- delete [] _rembSSRC;
- _rembSSRC = new WebRtc_UWord32[numberOfSSRC];
- _sizeRembSSRC = numberOfSSRC;
- }
-
- _lengthRembSSRC = numberOfSSRC;
- for (int i = 0; i < numberOfSSRC; i++)
- {
- _rembSSRC[i] = SSRC[i];
- }
- _sendREMB = true;
- return 0;
-}
-
-bool RTCPSender::SetRemoteBitrateObserver(RtpRemoteBitrateObserver* observer) {
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- if (observer && _bitrate_observer) {
- return false;
- }
- _bitrate_observer = observer;
- return true;
-}
-
-void RTCPSender::UpdateRemoteBitrateEstimate(unsigned int target_bitrate) {
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- if (_bitrate_observer) {
- _bitrate_observer->OnReceiveBitrateChanged(_remoteSSRC, target_bitrate);
- }
-}
-
-void RTCPSender::ReceivedRemb(unsigned int estimated_bitrate) {
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- if (_bitrate_observer) {
- _bitrate_observer->OnReceivedRemb(estimated_bitrate);
- }
-}
-
-bool
-RTCPSender::TMMBR() const
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- return _TMMBR;
-}
-
-WebRtc_Word32
-RTCPSender::SetTMMBRStatus(const bool enable)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- _TMMBR = enable;
- return 0;
-}
-
-bool
-RTCPSender::IJ() const
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- return _IJ;
-}
-
-WebRtc_Word32
-RTCPSender::SetIJStatus(const bool enable)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- _IJ = enable;
- return 0;
-}
-
-void
-RTCPSender::SetSSRC( const WebRtc_UWord32 ssrc)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
-
- if(_SSRC != 0)
- {
- // not first SetSSRC, probably due to a collision
- // schedule a new RTCP report
- // make sure that we send a RTP packet
- _nextTimeToSendRTCP = _clock.GetTimeInMS() + 100;
- }
- _SSRC = ssrc;
-}
-
-WebRtc_Word32
-RTCPSender::SetRemoteSSRC( const WebRtc_UWord32 ssrc)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- _remoteSSRC = ssrc;
- _remoteRateControl.Reset();
- return 0;
-}
-
-WebRtc_Word32
-RTCPSender::SetCameraDelay(const WebRtc_Word32 delayMS)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- if(delayMS > 1000 || delayMS < -1000)
- {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument, delay can't be larger than 1 sec", __FUNCTION__);
- return -1;
- }
- _cameraDelayMS = delayMS;
- return 0;
-}
-
-WebRtc_Word32 RTCPSender::CNAME(char cName[RTCP_CNAME_SIZE]) {
- assert(cName);
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- cName[RTCP_CNAME_SIZE - 1] = 0;
- strncpy(cName, _CNAME, RTCP_CNAME_SIZE - 1);
- return 0;
-}
-
-WebRtc_Word32 RTCPSender::SetCNAME(const char cName[RTCP_CNAME_SIZE]) {
- assert(cName);
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- _CNAME[RTCP_CNAME_SIZE - 1] = 0;
- strncpy(_CNAME, cName, RTCP_CNAME_SIZE - 1);
- return 0;
-}
-
-WebRtc_Word32 RTCPSender::AddMixedCNAME(const WebRtc_UWord32 SSRC,
- const char cName[RTCP_CNAME_SIZE]) {
- assert(cName);
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- if (_csrcCNAMEs.size() >= kRtpCsrcSize) {
- return -1;
- }
- RTCPCnameInformation* ptr = new RTCPCnameInformation();
- ptr->name[RTCP_CNAME_SIZE - 1] = 0;
- strncpy(ptr->name, cName, RTCP_CNAME_SIZE - 1);
- _csrcCNAMEs[SSRC] = ptr;
- return 0;
-}
-
-WebRtc_Word32 RTCPSender::RemoveMixedCNAME(const WebRtc_UWord32 SSRC) {
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- std::map<WebRtc_UWord32, RTCPCnameInformation*>::iterator it =
- _csrcCNAMEs.find(SSRC);
-
- if (it == _csrcCNAMEs.end()) {
- return -1;
- }
- delete it->second;
- _csrcCNAMEs.erase(it);
- return 0;
-}
-
-bool
-RTCPSender::TimeToSendRTCPReport(const bool sendKeyframeBeforeRTP) const
-{
-/*
- For audio we use a fix 5 sec interval
-
- For video we use 1 sec interval fo a BW smaller than 360 kbit/s,
- technicaly we break the max 5% RTCP BW for video below 10 kbit/s but that should be extreamly rare
-
-
-From RFC 3550
-
- MAX RTCP BW is 5% if the session BW
- A send report is approximately 65 bytes inc CNAME
- A report report is approximately 28 bytes
-
- The RECOMMENDED value for the reduced minimum in seconds is 360
- divided by the session bandwidth in kilobits/second. This minimum
- is smaller than 5 seconds for bandwidths greater than 72 kb/s.
-
- If the participant has not yet sent an RTCP packet (the variable
- initial is true), the constant Tmin is set to 2.5 seconds, else it
- is set to 5 seconds.
-
- The interval between RTCP packets is varied randomly over the
- range [0.5,1.5] times the calculated interval to avoid unintended
- synchronization of all participants
-
- if we send
- If the participant is a sender (we_sent true), the constant C is
- set to the average RTCP packet size (avg_rtcp_size) divided by 25%
- of the RTCP bandwidth (rtcp_bw), and the constant n is set to the
- number of senders.
-
- if we receive only
- If we_sent is not true, the constant C is set
- to the average RTCP packet size divided by 75% of the RTCP
- bandwidth. The constant n is set to the number of receivers
- (members - senders). If the number of senders is greater than
- 25%, senders and receivers are treated together.
-
- reconsideration NOT required for peer-to-peer
- "timer reconsideration" is
- employed. This algorithm implements a simple back-off mechanism
- which causes users to hold back RTCP packet transmission if the
- group sizes are increasing.
-
- n = number of members
- C = avg_size/(rtcpBW/4)
-
- 3. The deterministic calculated interval Td is set to max(Tmin, n*C).
-
- 4. The calculated interval T is set to a number uniformly distributed
- between 0.5 and 1.5 times the deterministic calculated interval.
-
- 5. The resulting value of T is divided by e-3/2=1.21828 to compensate
- for the fact that the timer reconsideration algorithm converges to
- a value of the RTCP bandwidth below the intended average
-*/
-
- WebRtc_UWord32 now = _clock.GetTimeInMS();
-
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
-
- if(_method == kRtcpOff)
- {
- return false;
- }
-
- if(!_audio && sendKeyframeBeforeRTP)
- {
- // for video key-frames we want to send the RTCP before the large key-frame
- // if we have a 100 ms margin
- now += RTCP_SEND_BEFORE_KEY_FRAME_MS;
- }
-
- if(now > _nextTimeToSendRTCP)
- {
- return true;
-
- } else if(now < 0x0000ffff && _nextTimeToSendRTCP > 0xffff0000) // 65 sec margin
- {
- // wrap
- return true;
- }
- return false;
-}
-
-WebRtc_UWord32
-RTCPSender::LastSendReport( WebRtc_UWord32& lastRTCPTime)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
-
- lastRTCPTime = _lastRTCPTime[0];
- return _lastSendReport[0];
-}
-
-WebRtc_UWord32
-RTCPSender::SendTimeOfSendReport(const WebRtc_UWord32 sendReport)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
-
- // This is only saved when we are the sender
- if((_lastSendReport[0] == 0) || (sendReport == 0))
- {
- return 0; // will be ignored
- } else
- {
- for(int i = 0; i < RTCP_NUMBER_OF_SR; ++i)
- {
- if( _lastSendReport[i] == sendReport)
- {
- return _lastRTCPTime[i];
- }
- }
- }
- return 0;
-}
-
-WebRtc_Word32 RTCPSender::AddReportBlock(const WebRtc_UWord32 SSRC,
- const RTCPReportBlock* reportBlock) {
- if (reportBlock == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s invalid argument", __FUNCTION__);
- return -1;
- }
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
-
- if (_reportBlocks.size() >= RTCP_MAX_REPORT_BLOCKS) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s invalid argument", __FUNCTION__);
- return -1;
- }
- RTCPReportBlock* copyReportBlock = new RTCPReportBlock();
- memcpy(copyReportBlock, reportBlock, sizeof(RTCPReportBlock));
- _reportBlocks[SSRC] = copyReportBlock;
- return 0;
-}
-
-WebRtc_Word32 RTCPSender::RemoveReportBlock(const WebRtc_UWord32 SSRC) {
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
-
- std::map<WebRtc_UWord32, RTCPReportBlock*>::iterator it =
- _reportBlocks.find(SSRC);
-
- if (it == _reportBlocks.end()) {
- return -1;
- }
- delete it->second;
- _reportBlocks.erase(it);
- return 0;
-}
-
-WebRtc_Word32
-RTCPSender::BuildSR(WebRtc_UWord8* rtcpbuffer,
- WebRtc_UWord32& pos,
- const WebRtc_UWord32 NTPsec,
- const WebRtc_UWord32 NTPfrac,
- const RTCPReportBlock* received)
-{
- // sanity
- if(pos + 52 >= IP_PACKET_SIZE)
- {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
- return -2;
- }
- WebRtc_UWord32 RTPtime;
- WebRtc_UWord32 BackTimedNTPsec;
- WebRtc_UWord32 BackTimedNTPfrac;
-
- WebRtc_UWord32 posNumberOfReportBlocks = pos;
- rtcpbuffer[pos++]=(WebRtc_UWord8)0x80;
-
- // Sender report
- rtcpbuffer[pos++]=(WebRtc_UWord8)200;
-
- for(int i = (RTCP_NUMBER_OF_SR-2); i >= 0; i--)
- {
- // shift old
- _lastSendReport[i+1] = _lastSendReport[i];
- _lastRTCPTime[i+1] =_lastRTCPTime[i];
- }
-
- _lastRTCPTime[0] = ModuleRTPUtility::ConvertNTPTimeToMS(NTPsec, NTPfrac); // before video cam compensation
-
- if(_cameraDelayMS >= 0)
- {
- // fraction of a second as an unsigned word32 4.294 967 296E9
- WebRtc_UWord32 cameraDelayFixFrac = (WebRtc_UWord32)_cameraDelayMS* 4294967; // note camera delay can't be larger than +/-1000ms
- if(NTPfrac > cameraDelayFixFrac)
- {
- // no problem just reduce the fraction part
- BackTimedNTPfrac = NTPfrac - cameraDelayFixFrac;
- BackTimedNTPsec = NTPsec;
- } else
- {
- // we need to reduce the sec and add that sec to the frac
- BackTimedNTPsec = NTPsec - 1;
- BackTimedNTPfrac = 0xffffffff - (cameraDelayFixFrac - NTPfrac);
- }
- } else
- {
- // fraction of a second as an unsigned word32 4.294 967 296E9
- WebRtc_UWord32 cameraDelayFixFrac = (WebRtc_UWord32)(-_cameraDelayMS)* 4294967; // note camera delay can't be larger than +/-1000ms
- if(NTPfrac > 0xffffffff - cameraDelayFixFrac)
- {
- // we need to add the sec and add that sec to the frac
- BackTimedNTPsec = NTPsec + 1;
- BackTimedNTPfrac = cameraDelayFixFrac + NTPfrac; // this will wrap but that is ok
- } else
- {
- // no problem just add the fraction part
- BackTimedNTPsec = NTPsec;
- BackTimedNTPfrac = NTPfrac + cameraDelayFixFrac;
- }
- }
- _lastSendReport[0] = (BackTimedNTPsec <<16) + (BackTimedNTPfrac >> 16);
-
- // RTP timestamp
- // This should have a ramdom start value added
- // RTP is counted from NTP not the acctual RTP
- // This reflects the perfect RTP time
- // we solve this by initiating RTP to our NTP :)
-
- WebRtc_UWord32 freqHz = 90000; // For video
- if(_audio)
- {
- freqHz = _rtpRtcp.CurrentSendFrequencyHz();
- RTPtime = ModuleRTPUtility::GetCurrentRTP(&_clock, freqHz);
- }
- else // video
- {
- // used to be (WebRtc_UWord32)(((float)BackTimedNTPfrac/(float)FRAC)* 90000)
- WebRtc_UWord32 tmp = 9*(BackTimedNTPfrac/429496);
- RTPtime = BackTimedNTPsec*freqHz + tmp;
- }
-
-
-
-
- // Add sender data
- // Save for our length field
- pos++;
- pos++;
-
- // Add our own SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
- pos += 4;
- // NTP
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, BackTimedNTPsec);
- pos += 4;
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, BackTimedNTPfrac);
- pos += 4;
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, RTPtime);
- pos += 4;
-
- //sender's packet count
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _rtpRtcp.PacketCountSent());
- pos += 4;
-
- //sender's octet count
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _rtpRtcp.ByteCountSent());
- pos += 4;
-
- WebRtc_UWord8 numberOfReportBlocks = 0;
- WebRtc_Word32 retVal = AddReportBlocks(rtcpbuffer, pos, numberOfReportBlocks, received, NTPsec, NTPfrac);
- if(retVal < 0)
- {
- //
- return retVal ;
- }
- rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks;
-
- WebRtc_UWord16 len = WebRtc_UWord16((pos/4) -1);
- ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+2, len);
- return 0;
-}
-
-
-WebRtc_Word32 RTCPSender::BuildSDEC(WebRtc_UWord8* rtcpbuffer,
- WebRtc_UWord32& pos) {
- size_t lengthCname = strlen(_CNAME);
- assert(lengthCname < RTCP_CNAME_SIZE);
-
- // sanity
- if(pos + 12 + lengthCname >= IP_PACKET_SIZE) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s invalid argument", __FUNCTION__);
- return -2;
- }
- // SDEC Source Description
-
- // We always need to add SDES CNAME
- rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(0x80 + 1 + _csrcCNAMEs.size());
- rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(202);
-
- // handle SDES length later on
- WebRtc_UWord32 SDESLengthPos = pos;
- pos++;
- pos++;
-
- // Add our own SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
- pos += 4;
-
- // CNAME = 1
- rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(1);
-
- //
- rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(lengthCname);
-
- WebRtc_UWord16 SDESLength = 10;
-
- memcpy(&rtcpbuffer[pos], _CNAME, lengthCname);
- pos += lengthCname;
- SDESLength += (WebRtc_UWord16)lengthCname;
-
- WebRtc_UWord16 padding = 0;
- // We must have a zero field even if we have an even multiple of 4 bytes
- if ((pos % 4) == 0) {
- padding++;
- rtcpbuffer[pos++]=0;
- }
- while ((pos % 4) != 0) {
- padding++;
- rtcpbuffer[pos++]=0;
- }
- SDESLength += padding;
-
- std::map<WebRtc_UWord32, RTCPUtility::RTCPCnameInformation*>::iterator it =
- _csrcCNAMEs.begin();
-
- for(; it != _csrcCNAMEs.end(); it++) {
- RTCPCnameInformation* cname = it->second;
- WebRtc_UWord32 SSRC = it->first;
-
- // Add SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, SSRC);
- pos += 4;
-
- // CNAME = 1
- rtcpbuffer[pos++] = static_cast<WebRtc_UWord8>(1);
-
- size_t length = strlen(cname->name);
- assert(length < RTCP_CNAME_SIZE);
-
- rtcpbuffer[pos++]= static_cast<WebRtc_UWord8>(length);
- SDESLength += 6;
-
- memcpy(&rtcpbuffer[pos],cname->name, length);
-
- pos += length;
- SDESLength += length;
- WebRtc_UWord16 padding = 0;
-
- // We must have a zero field even if we have an even multiple of 4 bytes
- if((pos % 4) == 0){
- padding++;
- rtcpbuffer[pos++]=0;
- }
- while((pos % 4) != 0){
- padding++;
- rtcpbuffer[pos++] = 0;
- }
- SDESLength += padding;
- }
- // in 32-bit words minus one and we don't count the header
- WebRtc_UWord16 buffer_length = (SDESLength / 4) - 1;
- ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer + SDESLengthPos,
- buffer_length);
- return 0;
-}
-
-WebRtc_Word32
-RTCPSender::BuildRR(WebRtc_UWord8* rtcpbuffer,
- WebRtc_UWord32& pos,
- const WebRtc_UWord32 NTPsec,
- const WebRtc_UWord32 NTPfrac,
- const RTCPReportBlock* received)
-{
- // sanity one block
- if(pos + 32 >= IP_PACKET_SIZE)
- {
- return -2;
- }
- WebRtc_UWord32 posNumberOfReportBlocks = pos;
-
- rtcpbuffer[pos++]=(WebRtc_UWord8)0x80;
- rtcpbuffer[pos++]=(WebRtc_UWord8)201;
-
- // Save for our length field
- pos++;
- pos++;
-
- // Add our own SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
- pos += 4;
-
- WebRtc_UWord8 numberOfReportBlocks = 0;
- WebRtc_Word32 retVal = AddReportBlocks(rtcpbuffer, pos, numberOfReportBlocks, received, NTPsec, NTPfrac);
- if(retVal < 0)
- {
- return retVal;
- }
- rtcpbuffer[posNumberOfReportBlocks] += numberOfReportBlocks;
-
- WebRtc_UWord16 len = WebRtc_UWord16((pos)/4 -1);
- ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+2, len);
- return 0;
-}
-
-// From RFC 5450: Transmission Time Offsets in RTP Streams.
-// 0 1 2 3
-// 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// hdr |V=2|P| RC | PT=IJ=195 | length |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// | inter-arrival jitter |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-// . .
-// . .
-// . .
-// | inter-arrival jitter |
-// +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-//
-// If present, this RTCP packet must be placed after a receiver report
-// (inside a compound RTCP packet), and MUST have the same value for RC
-// (reception report count) as the receiver report.
-
-WebRtc_Word32
-RTCPSender::BuildExtendedJitterReport(
- WebRtc_UWord8* rtcpbuffer,
- WebRtc_UWord32& pos,
- const WebRtc_UWord32 jitterTransmissionTimeOffset)
-{
- if (_reportBlocks.size() > 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "Not implemented.");
- return 0;
- }
-
- // sanity
- if(pos + 8 >= IP_PACKET_SIZE)
- {
- return -2;
- }
- // add picture loss indicator
- WebRtc_UWord8 RC = 1;
- rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + RC;
- rtcpbuffer[pos++]=(WebRtc_UWord8)195;
-
- // Used fixed length of 2
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- rtcpbuffer[pos++]=(WebRtc_UWord8)(1);
-
- // Add inter-arrival jitter
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer + pos,
- jitterTransmissionTimeOffset);
- pos += 4;
- return 0;
-}
-
-WebRtc_Word32
-RTCPSender::BuildPLI(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos)
-{
- // sanity
- if(pos + 12 >= IP_PACKET_SIZE)
- {
- return -2;
- }
- // add picture loss indicator
- WebRtc_UWord8 FMT = 1;
- rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT;
- rtcpbuffer[pos++]=(WebRtc_UWord8)206;
-
- //Used fixed length of 2
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- rtcpbuffer[pos++]=(WebRtc_UWord8)(2);
-
- // Add our own SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
- pos += 4;
-
- // Add the remote SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC);
- pos += 4;
- return 0;
-}
-
-WebRtc_Word32 RTCPSender::BuildFIR(WebRtc_UWord8* rtcpbuffer,
- WebRtc_UWord32& pos,
- bool repeat) {
- // sanity
- if(pos + 20 >= IP_PACKET_SIZE) {
- return -2;
- }
- if (!repeat) {
- _sequenceNumberFIR++; // do not increase if repetition
- }
-
- // add full intra request indicator
- WebRtc_UWord8 FMT = 4;
- rtcpbuffer[pos++] = (WebRtc_UWord8)0x80 + FMT;
- rtcpbuffer[pos++] = (WebRtc_UWord8)206;
-
- //Length of 4
- rtcpbuffer[pos++] = (WebRtc_UWord8)0;
- rtcpbuffer[pos++] = (WebRtc_UWord8)(4);
-
- // Add our own SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _SSRC);
- pos += 4;
-
- // RFC 5104 4.3.1.2. Semantics
- // SSRC of media source
- rtcpbuffer[pos++] = (WebRtc_UWord8)0;
- rtcpbuffer[pos++] = (WebRtc_UWord8)0;
- rtcpbuffer[pos++] = (WebRtc_UWord8)0;
- rtcpbuffer[pos++] = (WebRtc_UWord8)0;
-
- // Additional Feedback Control Information (FCI)
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer + pos, _remoteSSRC);
- pos += 4;
-
- rtcpbuffer[pos++] = (WebRtc_UWord8)(_sequenceNumberFIR);
- rtcpbuffer[pos++] = (WebRtc_UWord8)0;
- rtcpbuffer[pos++] = (WebRtc_UWord8)0;
- rtcpbuffer[pos++] = (WebRtc_UWord8)0;
- return 0;
-}
-
-/*
- 0 1 2 3
- 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | First | Number | PictureID |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-*/
-WebRtc_Word32
-RTCPSender::BuildSLI(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos, const WebRtc_UWord8 pictureID)
-{
- // sanity
- if(pos + 16 >= IP_PACKET_SIZE)
- {
- return -2;
- }
- // add slice loss indicator
- WebRtc_UWord8 FMT = 2;
- rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT;
- rtcpbuffer[pos++]=(WebRtc_UWord8)206;
-
- //Used fixed length of 3
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- rtcpbuffer[pos++]=(WebRtc_UWord8)(3);
-
- // Add our own SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
- pos += 4;
-
- // Add the remote SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC);
- pos += 4;
-
- // Add first, number & picture ID 6 bits
- // first = 0, 13 - bits
- // number = 0x1fff, 13 - bits only ones for now
- WebRtc_UWord32 sliField = (0x1fff << 6)+ (0x3f & pictureID);
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, sliField);
- pos += 4;
- return 0;
-}
-
-/*
- 0 1 2 3
- 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | PB |0| Payload Type| Native RPSI bit string |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | defined per codec ... | Padding (0) |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-*/
-/*
-* Note: not generic made for VP8
-*/
-WebRtc_Word32
-RTCPSender::BuildRPSI(WebRtc_UWord8* rtcpbuffer,
- WebRtc_UWord32& pos,
- const WebRtc_UWord64 pictureID,
- const WebRtc_UWord8 payloadType)
-{
- // sanity
- if(pos + 24 >= IP_PACKET_SIZE)
- {
- return -2;
- }
- // add Reference Picture Selection Indication
- WebRtc_UWord8 FMT = 3;
- rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT;
- rtcpbuffer[pos++]=(WebRtc_UWord8)206;
-
- // calc length
- WebRtc_UWord32 bitsRequired = 7;
- WebRtc_UWord8 bytesRequired = 1;
- while((pictureID>>bitsRequired) > 0)
- {
- bitsRequired += 7;
- bytesRequired++;
- }
-
- WebRtc_UWord8 size = 3;
- if(bytesRequired > 6)
- {
- size = 5;
- } else if(bytesRequired > 2)
- {
- size = 4;
- }
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- rtcpbuffer[pos++]=size;
-
- // Add our own SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
- pos += 4;
-
- // Add the remote SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC);
- pos += 4;
-
- // calc padding length
- WebRtc_UWord8 paddingBytes = 4-((2+bytesRequired)%4);
- if(paddingBytes == 4)
- {
- paddingBytes = 0;
- }
- // add padding length in bits
- rtcpbuffer[pos] = paddingBytes*8; // padding can be 0, 8, 16 or 24
- pos++;
-
- // add payload type
- rtcpbuffer[pos] = payloadType;
- pos++;
-
- // add picture ID
- for(int i = bytesRequired-1; i > 0; i--)
- {
- rtcpbuffer[pos] = 0x80 | WebRtc_UWord8(pictureID >> (i*7));
- pos++;
- }
- // add last byte of picture ID
- rtcpbuffer[pos] = WebRtc_UWord8(pictureID & 0x7f);
- pos++;
-
- // add padding
- for(int j = 0; j <paddingBytes; j++)
- {
- rtcpbuffer[pos] = 0;
- pos++;
- }
- return 0;
-}
-
-WebRtc_Word32
-RTCPSender::BuildREMB(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos)
-{
- // sanity
- if(pos + 20 + 4 * _lengthRembSSRC >= IP_PACKET_SIZE)
- {
- return -2;
- }
- // add application layer feedback
- WebRtc_UWord8 FMT = 15;
- rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT;
- rtcpbuffer[pos++]=(WebRtc_UWord8)206;
-
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- rtcpbuffer[pos++]=_lengthRembSSRC + 4;
-
- // Add our own SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
- pos += 4;
-
- // Remote SSRC must be 0
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, 0);
- pos += 4;
-
- rtcpbuffer[pos++]='R';
- rtcpbuffer[pos++]='E';
- rtcpbuffer[pos++]='M';
- rtcpbuffer[pos++]='B';
-
- rtcpbuffer[pos++] = _lengthRembSSRC;
- // 6 bit Exp
- // 18 bit mantissa
- WebRtc_UWord8 brExp = 0;
- for(WebRtc_UWord32 i=0; i<64; i++)
- {
- if(_rembBitrate <= ((WebRtc_UWord32)262143 << i))
- {
- brExp = i;
- break;
- }
- }
- const WebRtc_UWord32 brMantissa = (_rembBitrate >> brExp);
- rtcpbuffer[pos++]=(WebRtc_UWord8)((brExp << 2) + ((brMantissa >> 16) & 0x03));
- rtcpbuffer[pos++]=(WebRtc_UWord8)(brMantissa >> 8);
- rtcpbuffer[pos++]=(WebRtc_UWord8)(brMantissa);
-
- for (int i = 0; i < _lengthRembSSRC; i++)
- {
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _rembSSRC[i]);
- pos += 4;
- }
- return 0;
-}
-
-WebRtc_UWord32
-RTCPSender::CalculateNewTargetBitrate(WebRtc_UWord32 RTT)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- WebRtc_UWord32 target_bitrate =
- _remoteRateControl.UpdateBandwidthEstimate(RTT, _clock.GetTimeInMS());
- _tmmbr_Send = target_bitrate / 1000;
- return target_bitrate;
-}
-
-WebRtc_UWord32 RTCPSender::LatestBandwidthEstimate() const {
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- return _remoteRateControl.LatestEstimate();
-}
-
-bool
-RTCPSender::ValidBitrateEstimate() const {
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- return _remoteRateControl.ValidEstimate();
-}
-
-WebRtc_Word32
-RTCPSender::BuildTMMBR(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos)
-{
- // Before sending the TMMBR check the received TMMBN, only an owner is allowed to raise the bitrate
- // If the sender is an owner of the TMMBN -> send TMMBR
- // If not an owner but the TMMBR would enter the TMMBN -> send TMMBR
-
- // get current bounding set from RTCP receiver
- bool tmmbrOwner = false;
- TMMBRSet* candidateSet = _tmmbrHelp.CandidateSet(); // store in candidateSet, allocates one extra slot
-
- // holding _criticalSectionRTCPSender while calling RTCPreceiver which will accuire _criticalSectionRTCPReceiver
- // is a potental deadlock but since RTCPreceiver is not doing the revese we should be fine
- WebRtc_Word32 lengthOfBoundingSet = _rtpRtcp.BoundingSet(tmmbrOwner, candidateSet);
-
- if(lengthOfBoundingSet > 0)
- {
- for (WebRtc_Word32 i = 0; i < lengthOfBoundingSet; i++)
- {
- if( candidateSet->ptrTmmbrSet[i] == _tmmbr_Send &&
- candidateSet->ptrPacketOHSet[i] == _packetOH_Send)
- {
- // do not send the same tuple
- return 0;
- }
- }
- if(!tmmbrOwner)
- {
- // use received bounding set as candidate set
- // add current tuple
- candidateSet->ptrTmmbrSet[lengthOfBoundingSet] = _tmmbr_Send;
- candidateSet->ptrPacketOHSet[lengthOfBoundingSet] = _packetOH_Send;
- candidateSet->ptrSsrcSet[lengthOfBoundingSet] = _SSRC;
- int numCandidates = lengthOfBoundingSet+ 1;
-
- // find bounding set
- TMMBRSet* boundingSet = NULL;
- int numBoundingSet = _tmmbrHelp.FindTMMBRBoundingSet(boundingSet);
- if(numBoundingSet > 0 || numBoundingSet <= numCandidates)
- {
- tmmbrOwner = _tmmbrHelp.IsOwner(_SSRC, numBoundingSet);
- }
- if(!tmmbrOwner)
- {
- // did not enter bounding set, no meaning to send this request
- return 0;
- }
- }
- }
-
- if(_tmmbr_Send)
- {
- // sanity
- if(pos + 20 >= IP_PACKET_SIZE)
- {
- return -2;
- }
- // add TMMBR indicator
- WebRtc_UWord8 FMT = 3;
- rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT;
- rtcpbuffer[pos++]=(WebRtc_UWord8)205;
-
- //Length of 4
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- rtcpbuffer[pos++]=(WebRtc_UWord8)(4);
-
- // Add our own SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
- pos += 4;
-
- // RFC 5104 4.2.1.2. Semantics
-
- // SSRC of media source
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
-
- // Additional Feedback Control Information (FCI)
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC);
- pos += 4;
-
- WebRtc_UWord32 bitRate = _tmmbr_Send*1000;
- WebRtc_UWord32 mmbrExp = 0;
- for(WebRtc_UWord32 i=0;i<64;i++)
- {
- if(bitRate <= ((WebRtc_UWord32)131071 << i))
- {
- mmbrExp = i;
- break;
- }
- }
- WebRtc_UWord32 mmbrMantissa = (bitRate >> mmbrExp);
-
- rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrExp << 2) + ((mmbrMantissa >> 15) & 0x03));
- rtcpbuffer[pos++]=(WebRtc_UWord8)(mmbrMantissa >> 7);
- rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrMantissa << 1) + ((_packetOH_Send >> 8)& 0x01));
- rtcpbuffer[pos++]=(WebRtc_UWord8)(_packetOH_Send);
- }
- return 0;
-}
-
-WebRtc_Word32
-RTCPSender::BuildTMMBN(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos)
-{
- TMMBRSet* boundingSet = _tmmbrHelp.BoundingSetToSend();
- if(boundingSet == NULL)
- {
- return -1;
- }
- // sanity
- if(pos + 12 + boundingSet->lengthOfSet*8 >= IP_PACKET_SIZE)
- {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
- return -2;
- }
- WebRtc_UWord8 FMT = 4;
- // add TMMBN indicator
- rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT;
- rtcpbuffer[pos++]=(WebRtc_UWord8)205;
-
- //Add length later
- int posLength = pos;
- pos++;
- pos++;
-
- // Add our own SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
- pos += 4;
-
- // RFC 5104 4.2.2.2. Semantics
-
- // SSRC of media source
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
-
- // Additional Feedback Control Information (FCI)
- int numBoundingSet = 0;
- for(WebRtc_UWord32 n=0; n< boundingSet->lengthOfSet; n++)
- {
- if (boundingSet->ptrTmmbrSet[n] > 0)
- {
- WebRtc_UWord32 tmmbrSSRC = boundingSet->ptrSsrcSet[n];
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, tmmbrSSRC);
- pos += 4;
-
- WebRtc_UWord32 bitRate = boundingSet->ptrTmmbrSet[n] * 1000;
- WebRtc_UWord32 mmbrExp = 0;
- for(int i=0; i<64; i++)
- {
- if(bitRate <= ((WebRtc_UWord32)131071 << i))
- {
- mmbrExp = i;
- break;
- }
- }
- WebRtc_UWord32 mmbrMantissa = (bitRate >> mmbrExp);
- WebRtc_UWord32 measuredOH = boundingSet->ptrPacketOHSet[n];
-
- rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrExp << 2) + ((mmbrMantissa >> 15) & 0x03));
- rtcpbuffer[pos++]=(WebRtc_UWord8)(mmbrMantissa >> 7);
- rtcpbuffer[pos++]=(WebRtc_UWord8)((mmbrMantissa << 1) + ((measuredOH >> 8)& 0x01));
- rtcpbuffer[pos++]=(WebRtc_UWord8)(measuredOH);
- numBoundingSet++;
- }
- }
- WebRtc_UWord16 length= (WebRtc_UWord16)(2+2*numBoundingSet);
- rtcpbuffer[posLength++]=(WebRtc_UWord8)(length>>8);
- rtcpbuffer[posLength]=(WebRtc_UWord8)(length);
- return 0;
-}
-
-WebRtc_Word32
-RTCPSender::BuildAPP(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos)
-{
- // sanity
- if(_appData == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s invalid state", __FUNCTION__);
- return -1;
- }
- if(pos + 12 + _appLength >= IP_PACKET_SIZE)
- {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
- return -2;
- }
- rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + _appSubType;
-
- // Add APP ID
- rtcpbuffer[pos++]=(WebRtc_UWord8)204;
-
- WebRtc_UWord16 length = (_appLength>>2) + 2; // include SSRC and name
- rtcpbuffer[pos++]=(WebRtc_UWord8)(length>>8);
- rtcpbuffer[pos++]=(WebRtc_UWord8)(length);
-
- // Add our own SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
- pos += 4;
-
- // Add our application name
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _appName);
- pos += 4;
-
- // Add the data
- memcpy(rtcpbuffer +pos, _appData,_appLength);
- pos += _appLength;
- return 0;
-}
-
-WebRtc_Word32
-RTCPSender::BuildNACK(WebRtc_UWord8* rtcpbuffer,
- WebRtc_UWord32& pos,
- const WebRtc_Word32 nackSize,
- const WebRtc_UWord16* nackList)
-{
- // sanity
- if(pos + 16 >= IP_PACKET_SIZE)
- {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
- return -2;
- }
-
- // int size, WebRtc_UWord16* nackList
- // add nack list
- WebRtc_UWord8 FMT = 1;
- rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + FMT;
- rtcpbuffer[pos++]=(WebRtc_UWord8)205;
-
- rtcpbuffer[pos++]=(WebRtc_UWord8) 0;
- int nackSizePos = pos;
- rtcpbuffer[pos++]=(WebRtc_UWord8)(3); //setting it to one kNACK signal as default
-
- // Add our own SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
- pos += 4;
-
- // Add the remote SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC);
- pos += 4;
-
- // add the list
- int i = 0;
- int numOfNackFields = 0;
- while(nackSize > i && numOfNackFields < 253)
- {
- WebRtc_UWord16 nack = nackList[i];
- // put dow our sequence number
- ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+pos, nack);
- pos += 2;
-
- i++;
- numOfNackFields++;
- if(nackSize > i)
- {
- bool moreThan16Away = (WebRtc_UWord16(nack+16) < nackList[i])?true: false;
- if(!moreThan16Away)
- {
- // check for a wrap
- if(WebRtc_UWord16(nack+16) > 0xff00 && nackList[i] < 0x0fff)
- {
- // wrap
- moreThan16Away = true;
- }
- }
- if(moreThan16Away)
- {
- // next is more than 16 away
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- } else
- {
- // build our bitmask
- WebRtc_UWord16 bitmask = 0;
-
- bool within16Away = (WebRtc_UWord16(nack+16) > nackList[i])?true: false;
- if(within16Away)
- {
- // check for a wrap
- if(WebRtc_UWord16(nack+16) > 0xff00 && nackList[i] < 0x0fff)
- {
- // wrap
- within16Away = false;
- }
- }
-
- while( nackSize > i && within16Away)
- {
- WebRtc_Word16 shift = (nackList[i]-nack)-1;
- assert(!(shift > 15) && !(shift < 0));
-
- bitmask += (1<< shift);
- i++;
- if(nackSize > i)
- {
- within16Away = (WebRtc_UWord16(nack+16) > nackList[i])?true: false;
- if(within16Away)
- {
- // check for a wrap
- if(WebRtc_UWord16(nack+16) > 0xff00 && nackList[i] < 0x0fff)
- {
- // wrap
- within16Away = false;
- }
- }
- }
- }
- ModuleRTPUtility::AssignUWord16ToBuffer(rtcpbuffer+pos, bitmask);
- pos += 2;
- }
- // sanity do we have room from one more 4 byte block?
- if(pos + 4 >= IP_PACKET_SIZE)
- {
- return -2;
- }
- } else
- {
- // no more in the list
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- }
- }
- rtcpbuffer[nackSizePos]=(WebRtc_UWord8)(2+numOfNackFields);
- return 0;
-}
-
-WebRtc_Word32
-RTCPSender::BuildBYE(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos)
-{
- // sanity
- if(pos + 8 >= IP_PACKET_SIZE)
- {
- return -2;
- }
- if(_includeCSRCs)
- {
- // Add a bye packet
- rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + 1 + _CSRCs; // number of SSRC+CSRCs
- rtcpbuffer[pos++]=(WebRtc_UWord8)203;
-
- // length
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- rtcpbuffer[pos++]=(WebRtc_UWord8)(1 + _CSRCs);
-
- // Add our own SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
- pos += 4;
-
- // add CSRCs
- for(int i = 0; i < _CSRCs; i++)
- {
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _CSRC[i]);
- pos += 4;
- }
- } else
- {
- // Add a bye packet
- rtcpbuffer[pos++]=(WebRtc_UWord8)0x80 + 1; // number of SSRC+CSRCs
- rtcpbuffer[pos++]=(WebRtc_UWord8)203;
-
- // length
- rtcpbuffer[pos++]=(WebRtc_UWord8)0;
- rtcpbuffer[pos++]=(WebRtc_UWord8)1;
-
- // Add our own SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
- pos += 4;
- }
- return 0;
-}
-
-WebRtc_Word32
-RTCPSender::BuildVoIPMetric(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos)
-{
- // sanity
- if(pos + 44 >= IP_PACKET_SIZE)
- {
- return -2;
- }
-
- // Add XR header
- rtcpbuffer[pos++]=(WebRtc_UWord8)0x80;
- rtcpbuffer[pos++]=(WebRtc_UWord8)207;
-
- WebRtc_UWord32 XRLengthPos = pos;
-
- // handle length later on
- pos++;
- pos++;
-
- // Add our own SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _SSRC);
- pos += 4;
-
- // Add a VoIP metrics block
- rtcpbuffer[pos++]=7;
- rtcpbuffer[pos++]=0;
- rtcpbuffer[pos++]=0;
- rtcpbuffer[pos++]=8;
-
- // Add the remote SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC);
- pos += 4;
-
- rtcpbuffer[pos++] = _xrVoIPMetric.lossRate;
- rtcpbuffer[pos++] = _xrVoIPMetric.discardRate;
- rtcpbuffer[pos++] = _xrVoIPMetric.burstDensity;
- rtcpbuffer[pos++] = _xrVoIPMetric.gapDensity;
-
- rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.burstDuration >> 8);
- rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.burstDuration);
- rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.gapDuration >> 8);
- rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.gapDuration);
-
- rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.roundTripDelay >> 8);
- rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.roundTripDelay);
- rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.endSystemDelay >> 8);
- rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.endSystemDelay);
-
- rtcpbuffer[pos++] = _xrVoIPMetric.signalLevel;
- rtcpbuffer[pos++] = _xrVoIPMetric.noiseLevel;
- rtcpbuffer[pos++] = _xrVoIPMetric.RERL;
- rtcpbuffer[pos++] = _xrVoIPMetric.Gmin;
-
- rtcpbuffer[pos++] = _xrVoIPMetric.Rfactor;
- rtcpbuffer[pos++] = _xrVoIPMetric.extRfactor;
- rtcpbuffer[pos++] = _xrVoIPMetric.MOSLQ;
- rtcpbuffer[pos++] = _xrVoIPMetric.MOSCQ;
-
- rtcpbuffer[pos++] = _xrVoIPMetric.RXconfig;
- rtcpbuffer[pos++] = 0; // reserved
- rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBnominal >> 8);
- rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBnominal);
-
- rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBmax >> 8);
- rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBmax);
- rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBabsMax >> 8);
- rtcpbuffer[pos++] = (WebRtc_UWord8)(_xrVoIPMetric.JBabsMax);
-
- rtcpbuffer[XRLengthPos]=(WebRtc_UWord8)(0);
- rtcpbuffer[XRLengthPos+1]=(WebRtc_UWord8)(10);
- return 0;
-}
-
-WebRtc_Word32
-RTCPSender::SendRTCP(const WebRtc_UWord32 packetTypeFlags,
- const WebRtc_Word32 nackSize, // NACK
- const WebRtc_UWord16* nackList, // NACK
- const bool repeat, // FIR
- const WebRtc_UWord64 pictureID) // SLI & RPSI
-{
- WebRtc_UWord32 rtcpPacketTypeFlags = packetTypeFlags;
- WebRtc_UWord32 pos = 0;
- WebRtc_UWord8 rtcpbuffer[IP_PACKET_SIZE];
-
- do // only to be able to use break :) (and the critsect must be inside its own scope)
- {
- // collect the received information
- RTCPReportBlock received;
- bool hasReceived = false;
- WebRtc_UWord32 NTPsec = 0;
- WebRtc_UWord32 NTPfrac = 0;
- bool rtcpCompound = false;
- WebRtc_UWord32 jitterTransmissionOffset = 0;
-
- {
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- if(_method == kRtcpOff)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
- "%s invalid state", __FUNCTION__);
- return -1;
- }
- rtcpCompound = (_method == kRtcpCompound) ? true : false;
- }
-
- if (rtcpCompound ||
- rtcpPacketTypeFlags & kRtcpReport ||
- rtcpPacketTypeFlags & kRtcpSr ||
- rtcpPacketTypeFlags & kRtcpRr)
- {
- // get statistics from our RTPreceiver outside critsect
- if(_rtpRtcp.ReportBlockStatistics(&received.fractionLost,
- &received.cumulativeLost,
- &received.extendedHighSeqNum,
- &received.jitter,
- &jitterTransmissionOffset) == 0)
- {
- hasReceived = true;
-
- WebRtc_UWord32 lastReceivedRRNTPsecs = 0;
- WebRtc_UWord32 lastReceivedRRNTPfrac = 0;
- WebRtc_UWord32 remoteSR = 0;
-
- // ok even if we have not received a SR, we will send 0 in that case
- _rtpRtcp.LastReceivedNTP(lastReceivedRRNTPsecs,
- lastReceivedRRNTPfrac,
- remoteSR);
-
- // get our NTP as late as possible to avoid a race
- _clock.CurrentNTP(NTPsec, NTPfrac);
-
- // Delay since last received report
- WebRtc_UWord32 delaySinceLastReceivedSR = 0;
- if((lastReceivedRRNTPsecs !=0) || (lastReceivedRRNTPfrac !=0))
- {
- // get the 16 lowest bits of seconds and the 16 higest bits of fractions
- WebRtc_UWord32 now=NTPsec&0x0000FFFF;
- now <<=16;
- now += (NTPfrac&0xffff0000)>>16;
-
- WebRtc_UWord32 receiveTime = lastReceivedRRNTPsecs&0x0000FFFF;
- receiveTime <<=16;
- receiveTime += (lastReceivedRRNTPfrac&0xffff0000)>>16;
-
- delaySinceLastReceivedSR = now-receiveTime;
- }
- received.delaySinceLastSR = delaySinceLastReceivedSR;
- received.lastSR = remoteSR;
- } else
- {
- // we need to send our NTP even if we dont have received any reports
- _clock.CurrentNTP(NTPsec, NTPfrac);
- }
- }
-
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
-
- if(_TMMBR ) // attach TMMBR to send and receive reports
- {
- rtcpPacketTypeFlags |= kRtcpTmmbr;
- }
- if(_appSend)
- {
- rtcpPacketTypeFlags |= kRtcpApp;
- _appSend = false;
- }
- if(_REMB && _sendREMB)
- {
- // Always attach REMB to SR if that is configured. Note that REMB is
- // only sent on one of the RTP modules in the REMB group.
- rtcpPacketTypeFlags |= kRtcpRemb;
- }
- if(_xrSendVoIPMetric)
- {
- rtcpPacketTypeFlags |= kRtcpXrVoipMetric;
- _xrSendVoIPMetric = false;
- }
- if(_sendTMMBN) // set when having received a TMMBR
- {
- rtcpPacketTypeFlags |= kRtcpTmmbn;
- _sendTMMBN = false;
- }
-
- if(_method == kRtcpCompound)
- {
- if(_sending)
- {
- rtcpPacketTypeFlags |= kRtcpSr;
- } else
- {
- rtcpPacketTypeFlags |= kRtcpRr;
- }
- if (_IJ && hasReceived)
- {
- rtcpPacketTypeFlags |= kRtcpTransmissionTimeOffset;
- }
- } else if(_method == kRtcpNonCompound)
- {
- if(rtcpPacketTypeFlags & kRtcpReport)
- {
- if(_sending)
- {
- rtcpPacketTypeFlags |= kRtcpSr;
- } else
- {
- rtcpPacketTypeFlags |= kRtcpRr;
- }
- }
- }
- if( rtcpPacketTypeFlags & kRtcpRr ||
- rtcpPacketTypeFlags & kRtcpSr)
- {
- // generate next time to send a RTCP report
- // seeded from RTP constructor
- WebRtc_Word32 random = rand() % 1000;
- WebRtc_Word32 timeToNext = RTCP_INTERVAL_AUDIO_MS;
-
- if(_audio)
- {
- timeToNext = (RTCP_INTERVAL_AUDIO_MS/2) + (RTCP_INTERVAL_AUDIO_MS*random/1000);
- }else
- {
- WebRtc_UWord32 minIntervalMs = RTCP_INTERVAL_AUDIO_MS;
- if(_sending)
- {
- // calc bw for video 360/sendBW in kbit/s
- WebRtc_UWord32 sendBitrateKbit = 0;
- WebRtc_UWord32 videoRate = 0;
- WebRtc_UWord32 fecRate = 0;
- WebRtc_UWord32 nackRate = 0;
- _rtpRtcp.BitrateSent(&sendBitrateKbit,
- &videoRate,
- &fecRate,
- &nackRate);
- sendBitrateKbit /= 1000;
- if(sendBitrateKbit != 0)
- {
- minIntervalMs = 360000/sendBitrateKbit;
- }
- }
- if(minIntervalMs > RTCP_INTERVAL_VIDEO_MS)
- {
- minIntervalMs = RTCP_INTERVAL_VIDEO_MS;
- }
- timeToNext = (minIntervalMs/2) + (minIntervalMs*random/1000);
- }
- _nextTimeToSendRTCP = _clock.GetTimeInMS() + timeToNext;
- }
-
- // if the data does not fitt in the packet we fill it as much as possible
- WebRtc_Word32 buildVal = 0;
-
- if(rtcpPacketTypeFlags & kRtcpSr)
- {
- if(hasReceived)
- {
- buildVal = BuildSR(rtcpbuffer, pos, NTPsec, NTPfrac, &received);
- } else
- {
- buildVal = BuildSR(rtcpbuffer, pos, NTPsec, NTPfrac);
- }
- if(buildVal == -1)
- {
- return -1; // error
-
- }else if(buildVal == -2)
- {
- break; // out of buffer
- }
- buildVal = BuildSDEC(rtcpbuffer, pos);
- if(buildVal == -1)
- {
- return -1; // error
-
- }else if(buildVal == -2)
- {
- break; // out of buffer
- }
-
- }else if(rtcpPacketTypeFlags & kRtcpRr)
- {
- if(hasReceived)
- {
- buildVal = BuildRR(rtcpbuffer, pos, NTPsec, NTPfrac,&received);
- }else
- {
- buildVal = BuildRR(rtcpbuffer, pos, NTPsec, NTPfrac);
- }
- if(buildVal == -1)
- {
- return -1; // error
-
- }else if(buildVal == -2)
- {
- break; // out of buffer
- }
- // only of set
- if(_CNAME[0] != 0)
- {
- buildVal = BuildSDEC(rtcpbuffer, pos);
- if(buildVal == -1)
- {
- return -1; // error
- }
- }
- }
- if(rtcpPacketTypeFlags & kRtcpTransmissionTimeOffset)
- {
- // If present, this RTCP packet must be placed after a
- // receiver report.
- buildVal = BuildExtendedJitterReport(rtcpbuffer,
- pos,
- jitterTransmissionOffset);
- if(buildVal == -1)
- {
- return -1; // error
- }
- else if(buildVal == -2)
- {
- break; // out of buffer
- }
- }
- if(rtcpPacketTypeFlags & kRtcpPli)
- {
- buildVal = BuildPLI(rtcpbuffer, pos);
- if(buildVal == -1)
- {
- return -1; // error
-
- }else if(buildVal == -2)
- {
- break; // out of buffer
- }
- }
- if(rtcpPacketTypeFlags & kRtcpFir)
- {
- buildVal = BuildFIR(rtcpbuffer, pos, repeat);
- if(buildVal == -1)
- {
- return -1; // error
-
- }else if(buildVal == -2)
- {
- break; // out of buffer
- }
- }
- if(rtcpPacketTypeFlags & kRtcpSli)
- {
- buildVal = BuildSLI(rtcpbuffer, pos, (WebRtc_UWord8)pictureID);
- if(buildVal == -1)
- {
- return -1; // error
-
- }else if(buildVal == -2)
- {
- break; // out of buffer
- }
- }
- if(rtcpPacketTypeFlags & kRtcpRpsi)
- {
- const WebRtc_Word8 payloadType = _rtpRtcp.SendPayloadType();
- if(payloadType == -1)
- {
- return -1;
- }
- buildVal = BuildRPSI(rtcpbuffer, pos, pictureID, (WebRtc_UWord8)payloadType);
- if(buildVal == -1)
- {
- return -1; // error
-
- }else if(buildVal == -2)
- {
- break; // out of buffer
- }
- }
- if(rtcpPacketTypeFlags & kRtcpRemb)
- {
- buildVal = BuildREMB(rtcpbuffer, pos);
- if(buildVal == -1)
- {
- return -1; // error
-
- }else if(buildVal == -2)
- {
- break; // out of buffer
- }
- }
- if(rtcpPacketTypeFlags & kRtcpBye)
- {
- buildVal = BuildBYE(rtcpbuffer, pos);
- if(buildVal == -1)
- {
- return -1; // error
-
- }else if(buildVal == -2)
- {
- break; // out of buffer
- }
- }
- if(rtcpPacketTypeFlags & kRtcpApp)
- {
- buildVal = BuildAPP(rtcpbuffer, pos);
- if(buildVal == -1)
- {
- return -1; // error
-
- }else if(buildVal == -2)
- {
- break; // out of buffer
- }
- }
- if(rtcpPacketTypeFlags & kRtcpTmmbr)
- {
- buildVal = BuildTMMBR(rtcpbuffer, pos);
- if(buildVal == -1)
- {
- return -1; // error
-
- }else if(buildVal == -2)
- {
- break; // out of buffer
- }
- }
- if(rtcpPacketTypeFlags & kRtcpTmmbn)
- {
- buildVal = BuildTMMBN(rtcpbuffer, pos);
- if(buildVal == -1)
- {
- return -1; // error
-
- }else if(buildVal == -2)
- {
- break; // out of buffer
- }
- }
- if(rtcpPacketTypeFlags & kRtcpNack)
- {
- buildVal = BuildNACK(rtcpbuffer, pos, nackSize, nackList);
- if(buildVal == -1)
- {
- return -1; // error
-
- }else if(buildVal == -2)
- {
- break; // out of buffer
- }
- }
- if(rtcpPacketTypeFlags & kRtcpXrVoipMetric)
- {
- buildVal = BuildVoIPMetric(rtcpbuffer, pos);
- if(buildVal == -1)
- {
- return -1; // error
-
- }else if(buildVal == -2)
- {
- break; // out of buffer
- }
- }
- }while (false);
- // Sanity don't send empty packets.
- if (pos == 0)
- {
- return -1;
- }
- return SendToNetwork(rtcpbuffer, (WebRtc_UWord16)pos);
-}
-
-WebRtc_Word32
-RTCPSender::SendToNetwork(const WebRtc_UWord8* dataBuffer,
- const WebRtc_UWord16 length)
-{
- CriticalSectionScoped lock(_criticalSectionTransport);
- if(_cbTransport)
- {
- if(_cbTransport->SendRTCPPacket(_id, dataBuffer, length) > 0)
- {
- return 0;
- }
- }
- return -1;
-}
-
-WebRtc_Word32
-RTCPSender::SetCSRCStatus(const bool include)
-{
- _includeCSRCs = include;
- return 0;
-}
-
-WebRtc_Word32
-RTCPSender::SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
- const WebRtc_UWord8 arrLength)
-{
- if(arrLength > kRtpCsrcSize)
- {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
- assert(false);
- return -1;
- }
-
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
-
- for(int i = 0; i < arrLength;i++)
- {
- _CSRC[i] = arrOfCSRC[i];
- }
- _CSRCs = arrLength;
- return 0;
-}
-
-WebRtc_Word32
-RTCPSender::SetApplicationSpecificData(const WebRtc_UWord8 subType,
- const WebRtc_UWord32 name,
- const WebRtc_UWord8* data,
- const WebRtc_UWord16 length)
-{
- if(length %4 != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
- return -1;
- }
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
-
- if(_appData)
- {
- delete [] _appData;
- }
-
- _appSend = true;
- _appSubType = subType;
- _appName = name;
- _appData = new WebRtc_UWord8[length];
- _appLength = length;
- memcpy(_appData, data, length);
- return 0;
-}
-
-WebRtc_Word32
-RTCPSender::SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- memcpy(&_xrVoIPMetric, VoIPMetric, sizeof(RTCPVoIPMetric));
-
- _xrSendVoIPMetric = true;
- return 0;
-}
-
-// called under critsect _criticalSectionRTCPSender
-WebRtc_Word32 RTCPSender::AddReportBlocks(WebRtc_UWord8* rtcpbuffer,
- WebRtc_UWord32& pos,
- WebRtc_UWord8& numberOfReportBlocks,
- const RTCPReportBlock* received,
- const WebRtc_UWord32 NTPsec,
- const WebRtc_UWord32 NTPfrac) {
- // sanity one block
- if(pos + 24 >= IP_PACKET_SIZE) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s invalid argument", __FUNCTION__);
- return -1;
- }
- numberOfReportBlocks = _reportBlocks.size();
- if (received) {
- // add our multiple RR to numberOfReportBlocks
- numberOfReportBlocks++;
- }
- if (received) {
- // answer to the one that sends to me
- _lastRTCPTime[0] = ModuleRTPUtility::ConvertNTPTimeToMS(NTPsec, NTPfrac);
-
- // Remote SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, _remoteSSRC);
- pos += 4;
-
- // fraction lost
- rtcpbuffer[pos++]=received->fractionLost;
-
- // cumulative loss
- ModuleRTPUtility::AssignUWord24ToBuffer(rtcpbuffer+pos,
- received->cumulativeLost);
- pos += 3;
- // extended highest seq_no, contain the highest sequence number received
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos,
- received->extendedHighSeqNum);
- pos += 4;
-
- //Jitter
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, received->jitter);
- pos += 4;
-
- // Last SR timestamp, our NTP time when we received the last report
- // This is the value that we read from the send report packet not when we
- // received it...
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, received->lastSR);
- pos += 4;
-
- // Delay since last received report,time since we received the report
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos,
- received->delaySinceLastSR);
- pos += 4;
- }
- if ((pos + _reportBlocks.size() * 24) >= IP_PACKET_SIZE) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s invalid argument", __FUNCTION__);
- return -1;
- }
- std::map<WebRtc_UWord32, RTCPReportBlock*>::iterator it =
- _reportBlocks.begin();
-
- for (; it != _reportBlocks.end(); it++) {
- // we can have multiple report block in a conference
- WebRtc_UWord32 remoteSSRC = it->first;
- RTCPReportBlock* reportBlock = it->second;
- if (reportBlock) {
- // Remote SSRC
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos, remoteSSRC);
- pos += 4;
-
- // fraction lost
- rtcpbuffer[pos++] = reportBlock->fractionLost;
-
- // cumulative loss
- ModuleRTPUtility::AssignUWord24ToBuffer(rtcpbuffer+pos,
- reportBlock->cumulativeLost);
- pos += 3;
-
- // extended highest seq_no, contain the highest sequence number received
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos,
- reportBlock->extendedHighSeqNum);
- pos += 4;
-
- //Jitter
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos,
- reportBlock->jitter);
- pos += 4;
-
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos,
- reportBlock->lastSR);
- pos += 4;
-
- ModuleRTPUtility::AssignUWord32ToBuffer(rtcpbuffer+pos,
- reportBlock->delaySinceLastSR);
- pos += 4;
- }
- }
- return pos;
-}
-
-// no callbacks allowed inside this function
-WebRtc_Word32
-RTCPSender::SetTMMBN(const TMMBRSet* boundingSet,
- const WebRtc_UWord32 maxBitrateKbit)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
-
- if (0 == _tmmbrHelp.SetTMMBRBoundingSetToSend(boundingSet, maxBitrateKbit))
- {
- _sendTMMBN = true;
- return 0;
- }
- return -1;
-}
-
-WebRtc_Word32
-RTCPSender::RequestTMMBR(WebRtc_UWord32 estimatedBW, WebRtc_UWord32 packetOH)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- if(_TMMBR)
- {
- _tmmbr_Send = estimatedBW;
- _packetOH_Send = packetOH;
-
- return 0;
- }
- return -1;
-}
-
-RateControlRegion
-RTCPSender::UpdateOverUseState(const RateControlInput& rateControlInput, bool& firstOverUse)
-{
- CriticalSectionScoped lock(_criticalSectionRTCPSender);
- return _remoteRateControl.Update(rateControlInput, firstOverUse,
- _clock.GetTimeInMS());
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_sender.h b/trunk/src/modules/rtp_rtcp/source/rtcp_sender.h
deleted file mode 100644
index 966b64f..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtcp_sender.h
+++ /dev/null
@@ -1,268 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
-
-#include <map>
-
-#include "typedefs.h"
-#include "rtcp_utility.h"
-#include "rtp_utility.h"
-#include "rtp_rtcp_defines.h"
-#include "remote_rate_control.h"
-#include "tmmbr_help.h"
-
-namespace webrtc {
-
-class ModuleRtpRtcpImpl;
-
-class RTCPSender
-{
-public:
- RTCPSender(const WebRtc_Word32 id, const bool audio,
- RtpRtcpClock* clock, ModuleRtpRtcpImpl* owner);
- virtual ~RTCPSender();
-
- void ChangeUniqueId(const WebRtc_Word32 id);
-
- WebRtc_Word32 Init();
-
- WebRtc_Word32 RegisterSendTransport(Transport* outgoingTransport);
-
- RTCPMethod Status() const;
- WebRtc_Word32 SetRTCPStatus(const RTCPMethod method);
-
- bool Sending() const;
- WebRtc_Word32 SetSendingStatus(const bool enabled); // combine the functions
-
- WebRtc_Word32 SetNackStatus(const bool enable);
-
- void SetSSRC( const WebRtc_UWord32 ssrc);
-
- WebRtc_Word32 SetRemoteSSRC( const WebRtc_UWord32 ssrc);
-
- WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delayMS);
-
- WebRtc_Word32 CNAME(char cName[RTCP_CNAME_SIZE]);
- WebRtc_Word32 SetCNAME(const char cName[RTCP_CNAME_SIZE]);
-
- WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 SSRC,
- const char cName[RTCP_CNAME_SIZE]);
-
- WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 SSRC);
-
- WebRtc_UWord32 SendTimeOfSendReport(const WebRtc_UWord32 sendReport);
-
- bool TimeToSendRTCPReport(const bool sendKeyframeBeforeRTP = false) const;
-
- WebRtc_UWord32 LastSendReport(WebRtc_UWord32& lastRTCPTime);
-
- WebRtc_Word32 SendRTCP(const WebRtc_UWord32 rtcpPacketTypeFlags,
- const WebRtc_Word32 nackSize = 0,
- const WebRtc_UWord16* nackList = 0,
- const bool repeat = false,
- const WebRtc_UWord64 pictureID = 0);
-
- WebRtc_Word32 AddReportBlock(const WebRtc_UWord32 SSRC,
- const RTCPReportBlock* receiveBlock);
-
- WebRtc_Word32 RemoveReportBlock(const WebRtc_UWord32 SSRC);
-
- /*
- * REMB
- */
- bool REMB() const;
-
- WebRtc_Word32 SetREMBStatus(const bool enable);
-
- WebRtc_Word32 SetREMBData(const WebRtc_UWord32 bitrate,
- const WebRtc_UWord8 numberOfSSRC,
- const WebRtc_UWord32* SSRC);
-
- bool SetRemoteBitrateObserver(RtpRemoteBitrateObserver* observer);
-
- void UpdateRemoteBitrateEstimate(unsigned int target_bitrate);
-
- void ReceivedRemb(unsigned int estimated_bitrate);
-
- /*
- * TMMBR
- */
- bool TMMBR() const;
-
- WebRtc_Word32 SetTMMBRStatus(const bool enable);
-
- WebRtc_Word32 SetTMMBN(const TMMBRSet* boundingSet,
- const WebRtc_UWord32 maxBitrateKbit);
-
- WebRtc_Word32 RequestTMMBR(const WebRtc_UWord32 estimatedBW,
- const WebRtc_UWord32 packetOH);
-
- /*
- * Extended jitter report
- */
- bool IJ() const;
-
- WebRtc_Word32 SetIJStatus(const bool enable);
-
- /*
- *
- */
-
- WebRtc_Word32 SetApplicationSpecificData(const WebRtc_UWord8 subType,
- const WebRtc_UWord32 name,
- const WebRtc_UWord8* data,
- const WebRtc_UWord16 length);
-
- WebRtc_Word32 SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
-
- WebRtc_Word32 SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
- const WebRtc_UWord8 arrLength);
-
- WebRtc_Word32 SetCSRCStatus(const bool include);
-
- /*
- * New bandwidth estimation
- */
-
- RateControlRegion UpdateOverUseState(const RateControlInput& rateControlInput, bool& firstOverUse);
-
- WebRtc_UWord32 CalculateNewTargetBitrate(WebRtc_UWord32 RTT);
-
- WebRtc_UWord32 LatestBandwidthEstimate() const;
-
- // Returns true if there is a valid estimate of the incoming bitrate, false
- // otherwise.
- bool ValidBitrateEstimate() const;
-
-private:
- WebRtc_Word32 SendToNetwork(const WebRtc_UWord8* dataBuffer,
- const WebRtc_UWord16 length);
-
- void UpdatePacketRate();
-
- WebRtc_Word32 AddReportBlocks(WebRtc_UWord8* rtcpbuffer,
- WebRtc_UWord32& pos,
- WebRtc_UWord8& numberOfReportBlocks,
- const RTCPReportBlock* received,
- const WebRtc_UWord32 NTPsec,
- const WebRtc_UWord32 NTPfrac);
-
- WebRtc_Word32 BuildSR(WebRtc_UWord8* rtcpbuffer,
- WebRtc_UWord32& pos,
- const WebRtc_UWord32 NTPsec,
- const WebRtc_UWord32 NTPfrac,
- const RTCPReportBlock* received = NULL);
-
- WebRtc_Word32 BuildRR(WebRtc_UWord8* rtcpbuffer,
- WebRtc_UWord32& pos,
- const WebRtc_UWord32 NTPsec,
- const WebRtc_UWord32 NTPfrac,
- const RTCPReportBlock* received = NULL);
-
- WebRtc_Word32 BuildExtendedJitterReport(
- WebRtc_UWord8* rtcpbuffer,
- WebRtc_UWord32& pos,
- const WebRtc_UWord32 jitterTransmissionTimeOffset);
-
- WebRtc_Word32 BuildSDEC(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
- WebRtc_Word32 BuildPLI(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
- WebRtc_Word32 BuildREMB(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
- WebRtc_Word32 BuildTMMBR(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
- WebRtc_Word32 BuildTMMBN(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
- WebRtc_Word32 BuildAPP(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
- WebRtc_Word32 BuildVoIPMetric(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
- WebRtc_Word32 BuildBYE(WebRtc_UWord8* rtcpbuffer, WebRtc_UWord32& pos);
- WebRtc_Word32 BuildFIR(WebRtc_UWord8* rtcpbuffer,
- WebRtc_UWord32& pos,
- bool repeat);
- WebRtc_Word32 BuildSLI(WebRtc_UWord8* rtcpbuffer,
- WebRtc_UWord32& pos,
- const WebRtc_UWord8 pictureID);
- WebRtc_Word32 BuildRPSI(WebRtc_UWord8* rtcpbuffer,
- WebRtc_UWord32& pos,
- const WebRtc_UWord64 pictureID,
- const WebRtc_UWord8 payloadType);
-
- WebRtc_Word32 BuildNACK(WebRtc_UWord8* rtcpbuffer,
- WebRtc_UWord32& pos,
- const WebRtc_Word32 nackSize,
- const WebRtc_UWord16* nackList);
-
-private:
- WebRtc_Word32 _id;
- const bool _audio;
- RtpRtcpClock& _clock;
- RTCPMethod _method;
-
- ModuleRtpRtcpImpl& _rtpRtcp;
-
- CriticalSectionWrapper* _criticalSectionTransport;
- Transport* _cbTransport;
-
- CriticalSectionWrapper* _criticalSectionRTCPSender;
- bool _usingNack;
- bool _sending;
- bool _sendTMMBN;
- bool _REMB;
- bool _sendREMB;
- bool _TMMBR;
- bool _IJ;
-
- WebRtc_UWord32 _nextTimeToSendRTCP;
-
- WebRtc_UWord32 _SSRC;
- WebRtc_UWord32 _remoteSSRC; // SSRC that we receive on our RTP channel
- char _CNAME[RTCP_CNAME_SIZE];
-
- std::map<WebRtc_UWord32, RTCPReportBlock*> _reportBlocks;
- std::map<WebRtc_UWord32, RTCPUtility::RTCPCnameInformation*> _csrcCNAMEs;
-
- WebRtc_Word32 _cameraDelayMS;
-
- // Sent
- WebRtc_UWord32 _lastSendReport[RTCP_NUMBER_OF_SR]; // allow packet loss and RTT above 1 sec
- WebRtc_UWord32 _lastRTCPTime[RTCP_NUMBER_OF_SR];
-
- // send CSRCs
- WebRtc_UWord8 _CSRCs;
- WebRtc_UWord32 _CSRC[kRtpCsrcSize];
- bool _includeCSRCs;
-
- // Full intra request
- WebRtc_UWord8 _sequenceNumberFIR;
-
- // REMB
- WebRtc_UWord8 _lengthRembSSRC;
- WebRtc_UWord8 _sizeRembSSRC;
- WebRtc_UWord32* _rembSSRC;
- WebRtc_UWord32 _rembBitrate;
- RtpRemoteBitrateObserver* _bitrate_observer;
-
- TMMBRHelp _tmmbrHelp;
- WebRtc_UWord32 _tmmbr_Send;
- WebRtc_UWord32 _packetOH_Send;
- RemoteRateControl _remoteRateControl;
-
- // APP
- bool _appSend;
- WebRtc_UWord8 _appSubType;
- WebRtc_UWord32 _appName;
- WebRtc_UWord8* _appData;
- WebRtc_UWord16 _appLength;
-
- // XR VoIP metric
- bool _xrSendVoIPMetric;
- RTCPVoIPMetric _xrVoIPMetric;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_SENDER_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_sender_test.cc b/trunk/src/modules/rtp_rtcp/source/rtcp_sender_test.cc
deleted file mode 100644
index 7b5b057..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtcp_sender_test.cc
+++ /dev/null
@@ -1,178 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes unit tests for the RTCPSender.
- */
-
-#include <gtest/gtest.h>
-
-#include "common_types.h"
-#include "rtp_utility.h"
-#include "rtcp_sender.h"
-#include "rtcp_receiver.h"
-#include "rtp_rtcp_impl.h"
-
-namespace webrtc {
-
-void CreateRtpPacket(const bool marker_bit, const WebRtc_UWord8 payload,
- const WebRtc_UWord16 seq_num, const WebRtc_UWord32 timestamp,
- const WebRtc_UWord32 ssrc, WebRtc_UWord8* array,
- WebRtc_UWord16* cur_pos) {
- ASSERT_TRUE(payload <= 127);
- array[(*cur_pos)++] = 0x80;
- array[(*cur_pos)++] = payload | (marker_bit ? 0x80 : 0);
- array[(*cur_pos)++] = seq_num >> 8;
- array[(*cur_pos)++] = seq_num;
- array[(*cur_pos)++] = timestamp >> 24;
- array[(*cur_pos)++] = timestamp >> 16;
- array[(*cur_pos)++] = timestamp >> 8;
- array[(*cur_pos)++] = timestamp;
- array[(*cur_pos)++] = ssrc >> 24;
- array[(*cur_pos)++] = ssrc >> 16;
- array[(*cur_pos)++] = ssrc >> 8;
- array[(*cur_pos)++] = ssrc;
- // VP8 payload header
- array[(*cur_pos)++] = 0x90; // X bit = 1
- array[(*cur_pos)++] = 0x20; // T bit = 1
- array[(*cur_pos)++] = 0x00; // TID = 0
- array[(*cur_pos)++] = 0x00; // Key frame
- array[(*cur_pos)++] = 0x00;
- array[(*cur_pos)++] = 0x00;
- array[(*cur_pos)++] = 0x9d;
- array[(*cur_pos)++] = 0x01;
- array[(*cur_pos)++] = 0x2a;
- array[(*cur_pos)++] = 128;
- array[(*cur_pos)++] = 0;
- array[(*cur_pos)++] = 96;
- array[(*cur_pos)++] = 0;
-}
-
-class TestTransport : public Transport,
- public RtpData {
- public:
- TestTransport(RTCPReceiver* rtcp_receiver) :
- rtcp_receiver_(rtcp_receiver) {
- }
-
- virtual int SendPacket(int /*ch*/, const void* /*data*/, int /*len*/) {
- return -1;
- }
-
- virtual int SendRTCPPacket(int /*ch*/, const void *packet, int packet_len) {
- RTCPUtility::RTCPParserV2 rtcpParser((WebRtc_UWord8*)packet,
- (WebRtc_Word32)packet_len,
- true); // Allow non-compound RTCP
-
- EXPECT_TRUE(rtcpParser.IsValid());
- RTCPHelp::RTCPPacketInformation rtcpPacketInformation;
- EXPECT_EQ(0, rtcp_receiver_->IncomingRTCPPacket(rtcpPacketInformation,
- &rtcpParser));
- rtcp_packet_info_ = rtcpPacketInformation;
-
- return packet_len;
- }
-
- virtual int OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const WebRtcRTPHeader* rtpHeader)
- {return 0;}
- RTCPReceiver* rtcp_receiver_;
- RTCPHelp::RTCPPacketInformation rtcp_packet_info_;
-};
-
-class RtcpSenderTest : public ::testing::Test {
- protected:
- RtcpSenderTest() {
- system_clock_ = ModuleRTPUtility::GetSystemClock();
- rtp_rtcp_impl_ = new ModuleRtpRtcpImpl(0, false, system_clock_);
- rtcp_sender_ = new RTCPSender(0, false, system_clock_, rtp_rtcp_impl_);
- rtcp_receiver_ = new RTCPReceiver(0, system_clock_, rtp_rtcp_impl_);
- test_transport_ = new TestTransport(rtcp_receiver_);
- // Initialize
- EXPECT_EQ(0, rtcp_sender_->Init());
- EXPECT_EQ(0, rtcp_sender_->RegisterSendTransport(test_transport_));
- EXPECT_EQ(0, rtp_rtcp_impl_->RegisterIncomingDataCallback(test_transport_));
- }
- ~RtcpSenderTest() {
- delete rtcp_sender_;
- delete rtcp_receiver_;
- delete rtp_rtcp_impl_;
- delete test_transport_;
- delete system_clock_;
- }
-
- RtpRtcpClock* system_clock_;
- ModuleRtpRtcpImpl* rtp_rtcp_impl_;
- RTCPSender* rtcp_sender_;
- RTCPReceiver* rtcp_receiver_;
- TestTransport* test_transport_;
-
- enum {kMaxPacketLength = 1500};
- uint8_t packet_[kMaxPacketLength];
-};
-
-TEST_F(RtcpSenderTest, RtcpOff) {
- EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpOff));
- EXPECT_EQ(-1, rtcp_sender_->SendRTCP(kRtcpSr));
-}
-
-TEST_F(RtcpSenderTest, IJStatus) {
- ASSERT_FALSE(rtcp_sender_->IJ());
- EXPECT_EQ(0, rtcp_sender_->SetIJStatus(true));
- ASSERT_TRUE(rtcp_sender_->IJ());
-}
-
-TEST_F(RtcpSenderTest, TestCompound) {
- const bool marker_bit = false;
- const WebRtc_UWord8 payload = 100;
- const WebRtc_UWord16 seq_num = 11111;
- const WebRtc_UWord32 timestamp = 1234567;
- const WebRtc_UWord32 ssrc = 0x11111111;
- WebRtc_UWord16 packet_length = 0;
- CreateRtpPacket(marker_bit, payload, seq_num, timestamp, ssrc, packet_,
- &packet_length);
- EXPECT_EQ(25, packet_length);
-
- VideoCodec codec_inst;
- strncpy(codec_inst.plName, "VP8", webrtc::kPayloadNameSize - 1);
- codec_inst.codecType = webrtc::kVideoCodecVP8;
- codec_inst.plType = payload;
- EXPECT_EQ(0, rtp_rtcp_impl_->RegisterReceivePayload(codec_inst));
-
- // Make sure RTP packet has been received.
- EXPECT_EQ(0, rtp_rtcp_impl_->IncomingPacket(packet_, packet_length));
-
- EXPECT_EQ(0, rtcp_sender_->SetIJStatus(true));
- EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpCompound));
- EXPECT_EQ(0, rtcp_sender_->SendRTCP(kRtcpRr));
-
- // Transmission time offset packet should be received.
- ASSERT_TRUE(test_transport_->rtcp_packet_info_.rtcpPacketTypeFlags &
- kRtcpTransmissionTimeOffset);
-}
-
-TEST_F(RtcpSenderTest, TestCompound_NoRtpReceived) {
- EXPECT_EQ(0, rtcp_sender_->SetIJStatus(true));
- EXPECT_EQ(0, rtcp_sender_->SetRTCPStatus(kRtcpCompound));
- EXPECT_EQ(0, rtcp_sender_->SendRTCP(kRtcpRr));
-
- // Transmission time offset packet should not be received.
- ASSERT_FALSE(test_transport_->rtcp_packet_info_.rtcpPacketTypeFlags &
- kRtcpTransmissionTimeOffset);
-}
-
-int main(int argc, char** argv) {
- ::testing::InitGoogleTest(&argc, argv);
-
- return RUN_ALL_TESTS();
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_utility.cc b/trunk/src/modules/rtp_rtcp/source/rtcp_utility.cc
deleted file mode 100644
index f421672..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtcp_utility.cc
+++ /dev/null
@@ -1,1509 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rtcp_utility.h"
-
-#include <cstring> // memcpy
-#include <cmath> // ceil
-#include <cassert>
-
-
-namespace webrtc {
-// RTCPParserV2 : currently read only
-
-RTCPUtility::RTCPParserV2::RTCPParserV2(const WebRtc_UWord8* rtcpData,
- size_t rtcpDataLength,
- bool rtcpReducedSizeEnable)
- : _ptrRTCPDataBegin(rtcpData),
- _RTCPReducedSizeEnable(rtcpReducedSizeEnable),
- _ptrRTCPDataEnd(rtcpData + rtcpDataLength),
- _validPacket(false),
- _ptrRTCPData(rtcpData),
- _ptrRTCPBlockEnd(NULL),
- _state(State_TopLevel),
- _numberOfBlocks(0),
- _packetType(kRtcpNotValidCode) {
- Validate();
-}
-
-RTCPUtility::RTCPParserV2::~RTCPParserV2() {
-}
-
-ptrdiff_t
-RTCPUtility::RTCPParserV2::LengthLeft() const
-{
- return (_ptrRTCPDataEnd- _ptrRTCPData);
-}
-
-RTCPUtility::RTCPPacketTypes
-RTCPUtility::RTCPParserV2::PacketType() const
-{
- return _packetType;
-}
-
-const RTCPUtility::RTCPPacket&
-RTCPUtility::RTCPParserV2::Packet() const
-{
- return _packet;
-}
-
-RTCPUtility::RTCPPacketTypes
-RTCPUtility::RTCPParserV2::Begin()
-{
- _ptrRTCPData = _ptrRTCPDataBegin;
-
- return Iterate();
-}
-
-RTCPUtility::RTCPPacketTypes
-RTCPUtility::RTCPParserV2::Iterate()
-{
- // Reset packet type
- _packetType = kRtcpNotValidCode;
-
- if (IsValid())
- {
- switch (_state)
- {
- case State_TopLevel:
- IterateTopLevel();
- break;
- case State_ReportBlockItem:
- IterateReportBlockItem();
- break;
- case State_SDESChunk:
- IterateSDESChunk();
- break;
- case State_BYEItem:
- IterateBYEItem();
- break;
- case State_ExtendedJitterItem:
- IterateExtendedJitterItem();
- break;
- case State_RTPFB_NACKItem:
- IterateNACKItem();
- break;
- case State_RTPFB_TMMBRItem:
- IterateTMMBRItem();
- break;
- case State_RTPFB_TMMBNItem:
- IterateTMMBNItem();
- break;
- case State_PSFB_SLIItem:
- IterateSLIItem();
- break;
- case State_PSFB_RPSIItem:
- IterateRPSIItem();
- break;
- case State_PSFB_FIRItem:
- IterateFIRItem();
- break;
- case State_PSFB_AppItem:
- IteratePsfbAppItem();
- break;
- case State_PSFB_REMBItem:
- IteratePsfbREMBItem();
- break;
- case State_AppItem:
- IterateAppItem();
- break;
- default:
- assert(false); // Invalid state!
- break;
- }
- }
- return _packetType;
-}
-
-void
-RTCPUtility::RTCPParserV2::IterateTopLevel()
-{
- for (;;)
- {
- RTCPCommonHeader header;
-
- const bool success = RTCPParseCommonHeader(_ptrRTCPData,
- _ptrRTCPDataEnd,
- header);
-
- if (!success)
- {
- return;
- }
- _ptrRTCPBlockEnd = _ptrRTCPData + header.LengthInOctets;
- if (_ptrRTCPBlockEnd > _ptrRTCPDataEnd)
- {
- // Bad block!
- return;
- }
-
- switch (header.PT)
- {
- case PT_SR:
- {
- // number of Report blocks
- _numberOfBlocks = header.IC;
- ParseSR();
- return;
- }
- case PT_RR:
- {
- // number of Report blocks
- _numberOfBlocks = header.IC;
- ParseRR();
- return;
- }
- case PT_SDES:
- {
- // number of SDES blocks
- _numberOfBlocks = header.IC;
- const bool ok = ParseSDES();
- if (!ok)
- {
- // Nothing supported found, continue to next block!
- break;
- }
- return;
- }
- case PT_BYE:
- {
- _numberOfBlocks = header.IC;
- const bool ok = ParseBYE();
- if (!ok)
- {
- // Nothing supported found, continue to next block!
- break;
- }
- return;
- }
- case PT_IJ:
- {
- // number of Report blocks
- _numberOfBlocks = header.IC;
- ParseIJ();
- return;
- }
- case PT_RTPFB: // Fall through!
- case PT_PSFB:
- {
- const bool ok = ParseFBCommon(header);
- if (!ok)
- {
- // Nothing supported found, continue to next block!
- break;
- }
-
- return;
- }
- case PT_APP:
- {
- const bool ok = ParseAPP(header);
- if (!ok)
- {
- // Nothing supported found, continue to next block!
- break;
- }
- return;
- }
- case PT_XR:
- {
- const bool ok = ParseXR();
- if (!ok)
- {
- // Nothing supported found, continue to next block!
- break;
- }
- return;
- }
- default:
- // Not supported! Skip!
- EndCurrentBlock();
- break;
- }
- }
-}
-
-void
-RTCPUtility::RTCPParserV2::IterateReportBlockItem()
-{
- const bool success = ParseReportBlockItem();
- if (!success)
- {
- Iterate();
- }
-}
-
-void
-RTCPUtility::RTCPParserV2::IterateSDESChunk()
-{
- const bool success = ParseSDESChunk();
- if (!success)
- {
- Iterate();
- }
-}
-
-void
-RTCPUtility::RTCPParserV2::IterateBYEItem()
-{
- const bool success = ParseBYEItem();
- if (!success)
- {
- Iterate();
- }
-}
-
-void
-RTCPUtility::RTCPParserV2::IterateExtendedJitterItem()
-{
- const bool success = ParseIJItem();
- if (!success)
- {
- Iterate();
- }
-}
-
-void
-RTCPUtility::RTCPParserV2::IterateNACKItem()
-{
- const bool success = ParseNACKItem();
- if (!success)
- {
- Iterate();
- }
-}
-
-void
-RTCPUtility::RTCPParserV2::IterateTMMBRItem()
-{
- const bool success = ParseTMMBRItem();
- if (!success)
- {
- Iterate();
- }
-}
-
-void
-RTCPUtility::RTCPParserV2::IterateTMMBNItem()
-{
- const bool success = ParseTMMBNItem();
- if (!success)
- {
- Iterate();
- }
-}
-
-void
-RTCPUtility::RTCPParserV2::IterateSLIItem()
-{
- const bool success = ParseSLIItem();
- if (!success)
- {
- Iterate();
- }
-}
-
-void
-RTCPUtility::RTCPParserV2::IterateRPSIItem()
-{
- const bool success = ParseRPSIItem();
- if (!success)
- {
- Iterate();
- }
-}
-
-void
-RTCPUtility::RTCPParserV2::IterateFIRItem()
-{
- const bool success = ParseFIRItem();
- if (!success)
- {
- Iterate();
- }
-}
-
-void
-RTCPUtility::RTCPParserV2::IteratePsfbAppItem()
-{
- const bool success = ParsePsfbAppItem();
- if (!success)
- {
- Iterate();
- }
-}
-
-void
-RTCPUtility::RTCPParserV2::IteratePsfbREMBItem()
-{
- const bool success = ParsePsfbREMBItem();
- if (!success)
- {
- Iterate();
- }
-}
-
-void
-RTCPUtility::RTCPParserV2::IterateAppItem()
-{
- const bool success = ParseAPPItem();
- if (!success)
- {
- Iterate();
- }
-}
-
-void
-RTCPUtility::RTCPParserV2::Validate()
-{
- if (_ptrRTCPData == NULL)
- {
- return; // NOT VALID
- }
-
- RTCPCommonHeader header;
- const bool success = RTCPParseCommonHeader(_ptrRTCPDataBegin,
- _ptrRTCPDataEnd,
- header);
-
- if (!success)
- {
- return; // NOT VALID!
- }
-
- // * if (!reducedSize) : first packet must be RR or SR.
- //
- // * The padding bit (P) should be zero for the first packet of a
- // compound RTCP packet because padding should only be applied,
- // if it is needed, to the last packet. (NOT CHECKED!)
- //
- // * The length fields of the individual RTCP packets must add up
- // to the overall length of the compound RTCP packet as
- // received. This is a fairly strong check. (NOT CHECKED!)
-
- if (!_RTCPReducedSizeEnable)
- {
- if ((header.PT != PT_SR) && (header.PT != PT_RR))
- {
- return; // NOT VALID
- }
- }
-
- _validPacket = true;
-}
-
-bool
-RTCPUtility::RTCPParserV2::IsValid() const
-{
- return _validPacket;
-}
-
-void
-RTCPUtility::RTCPParserV2::EndCurrentBlock()
-{
- _ptrRTCPData = _ptrRTCPBlockEnd;
-}
-
-bool
-RTCPUtility::RTCPParseCommonHeader( const WebRtc_UWord8* ptrDataBegin,
- const WebRtc_UWord8* ptrDataEnd,
- RTCPCommonHeader& parsedHeader)
-{
- if (!ptrDataBegin || !ptrDataEnd)
- {
- return false;
- }
-
- // 0 1 2 3
- // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // |V=2|P| IC | PT | length |
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- //
- // Common header for all RTCP packets, 4 octets.
-
- if ((ptrDataEnd - ptrDataBegin) < 4)
- {
- return false;
- }
-
- parsedHeader.V = ptrDataBegin[0] >> 6;
- parsedHeader.P = ((ptrDataBegin[0] & 0x20) == 0) ? false : true;
- parsedHeader.IC = ptrDataBegin[0] & 0x1f;
- parsedHeader.PT = ptrDataBegin[1];
-
- parsedHeader.LengthInOctets = (ptrDataBegin[2] << 8) + ptrDataBegin[3] + 1;
- parsedHeader.LengthInOctets *= 4;
-
- if(parsedHeader.LengthInOctets == 0)
- {
- return false;
- }
- // Check if RTP version field == 2
- if (parsedHeader.V != 2)
- {
- return false;
- }
-
- return true;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseRR()
-{
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 8)
- {
- return false;
- }
-
-
- _ptrRTCPData += 4; // Skip header
-
- _packetType = kRtcpRrCode;
-
- _packet.RR.SenderSSRC = *_ptrRTCPData++ << 24;
- _packet.RR.SenderSSRC += *_ptrRTCPData++ << 16;
- _packet.RR.SenderSSRC += *_ptrRTCPData++ << 8;
- _packet.RR.SenderSSRC += *_ptrRTCPData++;
-
- _packet.RR.NumberOfReportBlocks = _numberOfBlocks;
-
- // State transition
- _state = State_ReportBlockItem;
-
- return true;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseSR()
-{
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 28)
- {
- EndCurrentBlock();
- return false;
- }
-
- _ptrRTCPData += 4; // Skip header
-
- _packetType = kRtcpSrCode;
-
- _packet.SR.SenderSSRC = *_ptrRTCPData++ << 24;
- _packet.SR.SenderSSRC += *_ptrRTCPData++ << 16;
- _packet.SR.SenderSSRC += *_ptrRTCPData++ << 8;
- _packet.SR.SenderSSRC += *_ptrRTCPData++;
-
- _packet.SR.NTPMostSignificant = *_ptrRTCPData++ << 24;
- _packet.SR.NTPMostSignificant += *_ptrRTCPData++ << 16;
- _packet.SR.NTPMostSignificant += *_ptrRTCPData++ << 8;
- _packet.SR.NTPMostSignificant += *_ptrRTCPData++;
-
- _packet.SR.NTPLeastSignificant = *_ptrRTCPData++ << 24;
- _packet.SR.NTPLeastSignificant += *_ptrRTCPData++ << 16;
- _packet.SR.NTPLeastSignificant += *_ptrRTCPData++ << 8;
- _packet.SR.NTPLeastSignificant += *_ptrRTCPData++;
-
- _packet.SR.RTPTimestamp = *_ptrRTCPData++ << 24;
- _packet.SR.RTPTimestamp += *_ptrRTCPData++ << 16;
- _packet.SR.RTPTimestamp += *_ptrRTCPData++ << 8;
- _packet.SR.RTPTimestamp += *_ptrRTCPData++;
-
- _packet.SR.SenderPacketCount = *_ptrRTCPData++ << 24;
- _packet.SR.SenderPacketCount += *_ptrRTCPData++ << 16;
- _packet.SR.SenderPacketCount += *_ptrRTCPData++ << 8;
- _packet.SR.SenderPacketCount += *_ptrRTCPData++;
-
- _packet.SR.SenderOctetCount = *_ptrRTCPData++ << 24;
- _packet.SR.SenderOctetCount += *_ptrRTCPData++ << 16;
- _packet.SR.SenderOctetCount += *_ptrRTCPData++ << 8;
- _packet.SR.SenderOctetCount += *_ptrRTCPData++;
-
- _packet.SR.NumberOfReportBlocks = _numberOfBlocks;
-
- // State transition
- if(_numberOfBlocks != 0)
- {
- _state = State_ReportBlockItem;
- }else
- {
- // don't go to state report block item if 0 report blocks
- _state = State_TopLevel;
- EndCurrentBlock();
- }
- return true;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseReportBlockItem()
-{
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 24 || _numberOfBlocks <= 0)
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
- _packet.ReportBlockItem.SSRC = *_ptrRTCPData++ << 24;
- _packet.ReportBlockItem.SSRC += *_ptrRTCPData++ << 16;
- _packet.ReportBlockItem.SSRC += *_ptrRTCPData++ << 8;
- _packet.ReportBlockItem.SSRC += *_ptrRTCPData++;
-
- _packet.ReportBlockItem.FractionLost = *_ptrRTCPData++;
-
- _packet.ReportBlockItem.CumulativeNumOfPacketsLost = *_ptrRTCPData++ << 16;
- _packet.ReportBlockItem.CumulativeNumOfPacketsLost += *_ptrRTCPData++ << 8;
- _packet.ReportBlockItem.CumulativeNumOfPacketsLost += *_ptrRTCPData++;
-
- _packet.ReportBlockItem.ExtendedHighestSequenceNumber = *_ptrRTCPData++ << 24;
- _packet.ReportBlockItem.ExtendedHighestSequenceNumber += *_ptrRTCPData++ << 16;
- _packet.ReportBlockItem.ExtendedHighestSequenceNumber += *_ptrRTCPData++ << 8;
- _packet.ReportBlockItem.ExtendedHighestSequenceNumber += *_ptrRTCPData++;
-
- _packet.ReportBlockItem.Jitter = *_ptrRTCPData++ << 24;
- _packet.ReportBlockItem.Jitter += *_ptrRTCPData++ << 16;
- _packet.ReportBlockItem.Jitter += *_ptrRTCPData++ << 8;
- _packet.ReportBlockItem.Jitter += *_ptrRTCPData++;
-
- _packet.ReportBlockItem.LastSR = *_ptrRTCPData++ << 24;
- _packet.ReportBlockItem.LastSR += *_ptrRTCPData++ << 16;
- _packet.ReportBlockItem.LastSR += *_ptrRTCPData++ << 8;
- _packet.ReportBlockItem.LastSR += *_ptrRTCPData++;
-
- _packet.ReportBlockItem.DelayLastSR = *_ptrRTCPData++ << 24;
- _packet.ReportBlockItem.DelayLastSR += *_ptrRTCPData++ << 16;
- _packet.ReportBlockItem.DelayLastSR += *_ptrRTCPData++ << 8;
- _packet.ReportBlockItem.DelayLastSR += *_ptrRTCPData++;
-
- _numberOfBlocks--;
- _packetType = kRtcpReportBlockItemCode;
- return true;
-}
-
-/* From RFC 5450: Transmission Time Offsets in RTP Streams.
- 0 1 2 3
- 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- hdr |V=2|P| RC | PT=IJ=195 | length |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | inter-arrival jitter |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- . .
- . .
- . .
- | inter-arrival jitter |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-*/
-
-bool
-RTCPUtility::RTCPParserV2::ParseIJ()
-{
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 4)
- {
- return false;
- }
-
- _ptrRTCPData += 4; // Skip header
-
- _packetType = kRtcpExtendedIjCode;
-
- // State transition
- _state = State_ExtendedJitterItem;
- return true;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseIJItem()
-{
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 4 || _numberOfBlocks <= 0)
- {
- _state = State_TopLevel;
- EndCurrentBlock();
- return false;
- }
-
- _packet.ExtendedJitterReportItem.Jitter = *_ptrRTCPData++ << 24;
- _packet.ExtendedJitterReportItem.Jitter += *_ptrRTCPData++ << 16;
- _packet.ExtendedJitterReportItem.Jitter += *_ptrRTCPData++ << 8;
- _packet.ExtendedJitterReportItem.Jitter += *_ptrRTCPData++;
-
- _numberOfBlocks--;
- _packetType = kRtcpExtendedIjItemCode;
- return true;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseSDES()
-{
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 8)
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
- _ptrRTCPData += 4; // Skip header
-
- _state = State_SDESChunk;
- _packetType = kRtcpSdesCode;
- return true;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseSDESChunk()
-{
- if(_numberOfBlocks <= 0)
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
- _numberOfBlocks--;
-
- // Find CName item in a SDES chunk.
- while (_ptrRTCPData < _ptrRTCPBlockEnd)
- {
- const ptrdiff_t dataLen = _ptrRTCPBlockEnd - _ptrRTCPData;
- if (dataLen < 4)
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
-
- WebRtc_UWord32 SSRC = *_ptrRTCPData++ << 24;
- SSRC += *_ptrRTCPData++ << 16;
- SSRC += *_ptrRTCPData++ << 8;
- SSRC += *_ptrRTCPData++;
-
- const bool foundCname = ParseSDESItem();
- if (foundCname)
- {
- _packet.CName.SenderSSRC = SSRC; // Add SSRC
- return true;
- }
- }
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseSDESItem()
-{
- // Find CName
- // Only the CNAME item is mandatory. RFC 3550 page 46
- bool foundCName = false;
-
- size_t itemOctetsRead = 0;
- while (_ptrRTCPData < _ptrRTCPBlockEnd)
- {
- const WebRtc_UWord8 tag = *_ptrRTCPData++;
- ++itemOctetsRead;
-
- if (tag == 0)
- {
- // End tag! 4 oct aligned
- while ((itemOctetsRead++ % 4) != 0)
- {
- ++_ptrRTCPData;
- }
- return foundCName;
- }
-
- if (_ptrRTCPData < _ptrRTCPBlockEnd)
- {
- const WebRtc_UWord8 len = *_ptrRTCPData++;
- ++itemOctetsRead;
-
- if (tag == 1)
- {
- // CNAME
-
- // Sanity
- if ((_ptrRTCPData + len) >= _ptrRTCPBlockEnd)
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
- WebRtc_UWord8 i = 0;
- for (; i < len; ++i)
- {
- const WebRtc_UWord8 c = _ptrRTCPData[i];
- if ((c < ' ') || (c > '{') || (c == '%') || (c == '\\'))
- {
- // Illegal char
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
- _packet.CName.CName[i] = c;
- }
- // Make sure we are null terminated.
- _packet.CName.CName[i] = 0;
- _packetType = kRtcpSdesChunkCode;
-
- foundCName = true;
- }
- _ptrRTCPData += len;
- itemOctetsRead += len;
- }
- }
-
- // No end tag found!
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseBYE()
-{
- _ptrRTCPData += 4; // Skip header
-
- _state = State_BYEItem;
-
- return ParseBYEItem();
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseBYEItem()
-{
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
- if (length < 4 || _numberOfBlocks == 0)
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
-
- _packetType = kRtcpByeCode;
-
- _packet.BYE.SenderSSRC = *_ptrRTCPData++ << 24;
- _packet.BYE.SenderSSRC += *_ptrRTCPData++ << 16;
- _packet.BYE.SenderSSRC += *_ptrRTCPData++ << 8;
- _packet.BYE.SenderSSRC += *_ptrRTCPData++;
-
- // we can have several CSRCs attached
-
- // sanity
- if(length >= 4*_numberOfBlocks)
- {
- _ptrRTCPData += (_numberOfBlocks -1)*4;
- }
- _numberOfBlocks = 0;
-
- return true;
-}
-/*
- 0 1 2 3
- 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- |V=2|P|reserved | PT=XR=207 | length |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | SSRC |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- : report blocks :
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-*/
-bool RTCPUtility::RTCPParserV2::ParseXR()
-{
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 8)
- {
- EndCurrentBlock();
- return false;
- }
-
- _ptrRTCPData += 4; // Skip header
-
- _packet.XR.OriginatorSSRC = *_ptrRTCPData++ << 24;
- _packet.XR.OriginatorSSRC += *_ptrRTCPData++ << 16;
- _packet.XR.OriginatorSSRC += *_ptrRTCPData++ << 8;
- _packet.XR.OriginatorSSRC += *_ptrRTCPData++;
-
- return ParseXRItem();
-}
-/*
- 0 1 2 3
- 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | BT | type-specific | block length |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- : type-specific block contents :
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-*/
-
-bool
-RTCPUtility::RTCPParserV2::ParseXRItem()
-{
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 4) //
- {
- EndCurrentBlock();
- return false;
- }
-
- WebRtc_UWord8 blockType = *_ptrRTCPData++;
- WebRtc_UWord8 typeSpecific = *_ptrRTCPData++;
-
- WebRtc_UWord16 blockLength = *_ptrRTCPData++ << 8;
- blockLength = *_ptrRTCPData++;
-
- if(blockType == 7 && typeSpecific == 0)
- {
- if(blockLength != 8)
- {
- EndCurrentBlock();
- return false;
- }
- return ParseXRVOIPMetricItem();
- }else
- {
- EndCurrentBlock();
- return false;
- }
-}
-/*
- 0 1 2 3
- 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | BT=7 | reserved | block length = 8 |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | SSRC of source |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | loss rate | discard rate | burst density | gap density |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | burst duration | gap duration |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | round trip delay | end system delay |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | signal level | noise level | RERL | Gmin |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | R factor | ext. R factor | MOS-LQ | MOS-CQ |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | RX config | reserved | JB nominal |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | JB maximum | JB abs max |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-*/
-bool
-RTCPUtility::RTCPParserV2::ParseXRVOIPMetricItem()
-{
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 28)
- {
- EndCurrentBlock();
- return false;
- }
- _packetType = kRtcpXrVoipMetricCode;
-
- _packet.XRVOIPMetricItem.SSRC = *_ptrRTCPData++ << 24;
- _packet.XRVOIPMetricItem.SSRC += *_ptrRTCPData++ << 16;
- _packet.XRVOIPMetricItem.SSRC += *_ptrRTCPData++ << 8;
- _packet.XRVOIPMetricItem.SSRC += *_ptrRTCPData++;
-
- _packet.XRVOIPMetricItem.lossRate = *_ptrRTCPData++;
- _packet.XRVOIPMetricItem.discardRate = *_ptrRTCPData++;
- _packet.XRVOIPMetricItem.burstDensity = *_ptrRTCPData++;
- _packet.XRVOIPMetricItem.gapDensity = *_ptrRTCPData++;
-
- _packet.XRVOIPMetricItem.burstDuration = *_ptrRTCPData++ << 8;
- _packet.XRVOIPMetricItem.burstDuration += *_ptrRTCPData++;
-
- _packet.XRVOIPMetricItem.gapDuration = *_ptrRTCPData++ << 8;
- _packet.XRVOIPMetricItem.gapDuration += *_ptrRTCPData++;
-
- _packet.XRVOIPMetricItem.roundTripDelay = *_ptrRTCPData++ << 8;
- _packet.XRVOIPMetricItem.roundTripDelay += *_ptrRTCPData++;
-
- _packet.XRVOIPMetricItem.endSystemDelay = *_ptrRTCPData++ << 8;
- _packet.XRVOIPMetricItem.endSystemDelay += *_ptrRTCPData++;
-
- _packet.XRVOIPMetricItem.signalLevel = *_ptrRTCPData++;
- _packet.XRVOIPMetricItem.noiseLevel = *_ptrRTCPData++;
- _packet.XRVOIPMetricItem.RERL = *_ptrRTCPData++;
- _packet.XRVOIPMetricItem.Gmin = *_ptrRTCPData++;
- _packet.XRVOIPMetricItem.Rfactor = *_ptrRTCPData++;
- _packet.XRVOIPMetricItem.extRfactor = *_ptrRTCPData++;
- _packet.XRVOIPMetricItem.MOSLQ = *_ptrRTCPData++;
- _packet.XRVOIPMetricItem.MOSCQ = *_ptrRTCPData++;
- _packet.XRVOIPMetricItem.RXconfig = *_ptrRTCPData++;
- _ptrRTCPData++; // skip reserved
-
- _packet.XRVOIPMetricItem.JBnominal = *_ptrRTCPData++ << 8;
- _packet.XRVOIPMetricItem.JBnominal += *_ptrRTCPData++;
-
- _packet.XRVOIPMetricItem.JBmax = *_ptrRTCPData++ << 8;
- _packet.XRVOIPMetricItem.JBmax += *_ptrRTCPData++;
-
- _packet.XRVOIPMetricItem.JBabsMax = *_ptrRTCPData++ << 8;
- _packet.XRVOIPMetricItem.JBabsMax += *_ptrRTCPData++;
-
- return true;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseFBCommon(const RTCPCommonHeader& header)
-{
- assert((header.PT == PT_RTPFB) || (header.PT == PT_PSFB)); // Parser logic check
-
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 12) // 4 * 3, RFC4585 section 6.1
- {
- EndCurrentBlock();
- return false;
- }
-
- _ptrRTCPData += 4; // Skip RTCP header
-
- WebRtc_UWord32 senderSSRC = *_ptrRTCPData++ << 24;
- senderSSRC += *_ptrRTCPData++ << 16;
- senderSSRC += *_ptrRTCPData++ << 8;
- senderSSRC += *_ptrRTCPData++;
-
- WebRtc_UWord32 mediaSSRC = *_ptrRTCPData++ << 24;
- mediaSSRC += *_ptrRTCPData++ << 16;
- mediaSSRC += *_ptrRTCPData++ << 8;
- mediaSSRC += *_ptrRTCPData++;
-
- if (header.PT == PT_RTPFB)
- {
- // Transport layer feedback
-
- switch (header.IC)
- {
- case 1:
- {
- // NACK
- _packetType = kRtcpRtpfbNackCode;
- _packet.NACK.SenderSSRC = senderSSRC;
- _packet.NACK.MediaSSRC = mediaSSRC;
-
- _state = State_RTPFB_NACKItem;
-
- return true;
- }
- case 2:
- {
- // used to be ACK is this code point, which is removed
- // conficts with http://tools.ietf.org/html/draft-levin-avt-rtcp-burst-00
- break;
- }
- case 3:
- {
- // TMMBR
- _packetType = kRtcpRtpfbTmmbrCode;
- _packet.TMMBR.SenderSSRC = senderSSRC;
- _packet.TMMBR.MediaSSRC = mediaSSRC;
-
- _state = State_RTPFB_TMMBRItem;
-
- return true;
- }
- case 4:
- {
- // TMMBN
- _packetType = kRtcpRtpfbTmmbnCode;
- _packet.TMMBN.SenderSSRC = senderSSRC;
- _packet.TMMBN.MediaSSRC = mediaSSRC;
-
- _state = State_RTPFB_TMMBNItem;
-
- return true;
- }
- case 5:
- {
- // RTCP-SR-REQ Rapid Synchronisation of RTP Flows
- // draft-perkins-avt-rapid-rtp-sync-03.txt
- // trigger a new RTCP SR
- _packetType = kRtcpRtpfbSrReqCode;
-
- // Note: No state transition, SR REQ is empty!
- return true;
- }
- default:
- break;
- }
- EndCurrentBlock();
- return false;
- }
- else if (header.PT == PT_PSFB)
- {
- // Payload specific feedback
- switch (header.IC)
- {
- case 1:
- // PLI
- _packetType = kRtcpPsfbPliCode;
- _packet.PLI.SenderSSRC = senderSSRC;
- _packet.PLI.MediaSSRC = mediaSSRC;
-
- // Note: No state transition, PLI FCI is empty!
- return true;
- case 2:
- // SLI
- _packetType = kRtcpPsfbSliCode;
- _packet.SLI.SenderSSRC = senderSSRC;
- _packet.SLI.MediaSSRC = mediaSSRC;
-
- _state = State_PSFB_SLIItem;
-
- return true;
- case 3:
- _packetType = kRtcpPsfbRpsiCode;
- _packet.RPSI.SenderSSRC = senderSSRC;
- _packet.RPSI.MediaSSRC = mediaSSRC;
-
- _state = State_PSFB_RPSIItem;
- return true;
- case 4:
- // FIR
- _packetType = kRtcpPsfbFirCode;
- _packet.FIR.SenderSSRC = senderSSRC;
- _packet.FIR.MediaSSRC = mediaSSRC;
-
- _state = State_PSFB_FIRItem;
- return true;
- case 15:
- _packetType = kRtcpPsfbAppCode;
-
- _state = State_PSFB_AppItem;
- return true;
- default:
- break;
- }
-
- EndCurrentBlock();
- return false;
- }
- else
- {
- assert(false);
-
- EndCurrentBlock();
- return false;
- }
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseRPSIItem()
-{
- // RFC 4585 6.3.3. Reference Picture Selection Indication (RPSI)
- /*
- 0 1 2 3
- 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | PB |0| Payload Type| Native RPSI bit string |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | defined per codec ... | Padding (0) |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- */
-
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 4)
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
- if(length > 2+RTCP_RPSI_DATA_SIZE)
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
-
- _packetType = kRtcpPsfbRpsiCode;
-
- WebRtc_UWord8 paddingBits = *_ptrRTCPData++;
- _packet.RPSI.PayloadType = *_ptrRTCPData++;
-
- memcpy(_packet.RPSI.NativeBitString, _ptrRTCPData, length-2);
-
- _packet.RPSI.NumberOfValidBits = WebRtc_UWord16(length-2)*8 - paddingBits;
- return true;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseNACKItem()
-{
- // RFC 4585 6.2.1. Generic NACK
-
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 4)
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
-
- _packetType = kRtcpRtpfbNackItemCode;
-
- _packet.NACKItem.PacketID = *_ptrRTCPData++ << 8;
- _packet.NACKItem.PacketID += *_ptrRTCPData++;
-
- _packet.NACKItem.BitMask = *_ptrRTCPData++ << 8;
- _packet.NACKItem.BitMask += *_ptrRTCPData++;
-
- return true;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParsePsfbAppItem()
-{
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 4)
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
- if(*_ptrRTCPData++ != 'R')
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
- if(*_ptrRTCPData++ != 'E')
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
- if(*_ptrRTCPData++ != 'M')
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
- if(*_ptrRTCPData++ != 'B')
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
- _packetType = kRtcpPsfbRembItemCode;
- _state = State_PSFB_REMBItem;
- return true;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParsePsfbREMBItem()
-{
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 4)
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
-
- const WebRtc_UWord8 numSSRC = *_ptrRTCPData++;
- const WebRtc_UWord8 brExp = (_ptrRTCPData[0] >> 2) & 0x3F;
-
- WebRtc_UWord32 brMantissa = (_ptrRTCPData[0] & 0x03) << 16;
- brMantissa += (_ptrRTCPData[1] << 8);
- brMantissa += (_ptrRTCPData[2]);
-
- _ptrRTCPData += 3; // Fwd read data
- _packet.REMB.BitRate = (brMantissa << brExp);
-
- _ptrRTCPData += 4 * numSSRC; // Ignore the SSRCs for now
- return true;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseTMMBRItem()
-{
- // RFC 5104 4.2.1. Temporary Maximum Media Stream Bit Rate Request (TMMBR)
-
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 8)
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
-
- _packetType = kRtcpRtpfbTmmbrItemCode;
-
- _packet.TMMBRItem.SSRC = *_ptrRTCPData++ << 24;
- _packet.TMMBRItem.SSRC += *_ptrRTCPData++ << 16;
- _packet.TMMBRItem.SSRC += *_ptrRTCPData++ << 8;
- _packet.TMMBRItem.SSRC += *_ptrRTCPData++;
-
- WebRtc_UWord8 mxtbrExp = (_ptrRTCPData[0] >> 2) & 0x3F;
-
- WebRtc_UWord32 mxtbrMantissa = (_ptrRTCPData[0] & 0x03) << 15;
- mxtbrMantissa += (_ptrRTCPData[1] << 7);
- mxtbrMantissa += (_ptrRTCPData[2] >> 1) & 0x7F;
-
- WebRtc_UWord32 measuredOH = (_ptrRTCPData[2] & 0x01) << 8;
- measuredOH += _ptrRTCPData[3];
-
- _ptrRTCPData += 4; // Fwd read data
-
- _packet.TMMBRItem.MaxTotalMediaBitRate = ((mxtbrMantissa << mxtbrExp) / 1000);
- _packet.TMMBRItem.MeasuredOverhead = measuredOH;
-
- return true;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseTMMBNItem()
-{
- // RFC 5104 4.2.2. Temporary Maximum Media Stream Bit Rate Notification (TMMBN)
-
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 8)
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
-
- _packetType = kRtcpRtpfbTmmbnItemCode;
-
- _packet.TMMBNItem.SSRC = *_ptrRTCPData++ << 24;
- _packet.TMMBNItem.SSRC += *_ptrRTCPData++ << 16;
- _packet.TMMBNItem.SSRC += *_ptrRTCPData++ << 8;
- _packet.TMMBNItem.SSRC += *_ptrRTCPData++;
-
- WebRtc_UWord8 mxtbrExp = (_ptrRTCPData[0] >> 2) & 0x3F;
-
- WebRtc_UWord32 mxtbrMantissa = (_ptrRTCPData[0] & 0x03) << 15;
- mxtbrMantissa += (_ptrRTCPData[1] << 7);
- mxtbrMantissa += (_ptrRTCPData[2] >> 1) & 0x7F;
-
- WebRtc_UWord32 measuredOH = (_ptrRTCPData[2] & 0x01) << 8;
- measuredOH += _ptrRTCPData[3];
-
- _ptrRTCPData += 4; // Fwd read data
-
- _packet.TMMBNItem.MaxTotalMediaBitRate = ((mxtbrMantissa << mxtbrExp) / 1000);
- _packet.TMMBNItem.MeasuredOverhead = measuredOH;
-
- return true;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseSLIItem()
-{
- // RFC 5104 6.3.2. Slice Loss Indication (SLI)
- /*
- 0 1 2 3
- 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | First | Number | PictureID |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- */
-
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 4)
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
- _packetType = kRtcpPsfbSliItemCode;
-
- WebRtc_UWord32 buffer;
- buffer = *_ptrRTCPData++ << 24;
- buffer += *_ptrRTCPData++ << 16;
- buffer += *_ptrRTCPData++ << 8;
- buffer += *_ptrRTCPData++;
-
- _packet.SLIItem.FirstMB = WebRtc_UWord16((buffer>>19) & 0x1fff);
- _packet.SLIItem.NumberOfMB = WebRtc_UWord16((buffer>>6) & 0x1fff);
- _packet.SLIItem.PictureId = WebRtc_UWord8(buffer & 0x3f);
-
- return true;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseFIRItem()
-{
- // RFC 5104 4.3.1. Full Intra Request (FIR)
-
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 8)
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
-
- _packetType = kRtcpPsfbFirItemCode;
-
- _packet.FIRItem.SSRC = *_ptrRTCPData++ << 24;
- _packet.FIRItem.SSRC += *_ptrRTCPData++ << 16;
- _packet.FIRItem.SSRC += *_ptrRTCPData++ << 8;
- _packet.FIRItem.SSRC += *_ptrRTCPData++;
-
- _packet.FIRItem.CommandSequenceNumber = *_ptrRTCPData++;
- _ptrRTCPData += 3; // Skip "Reserved" bytes.
- return true;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseAPP( const RTCPCommonHeader& header)
-{
- ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- if (length < 12) // 4 * 3, RFC 3550 6.7 APP: Application-Defined RTCP Packet
- {
- EndCurrentBlock();
- return false;
- }
-
- _ptrRTCPData += 4; // Skip RTCP header
-
- WebRtc_UWord32 senderSSRC = *_ptrRTCPData++ << 24;
- senderSSRC += *_ptrRTCPData++ << 16;
- senderSSRC += *_ptrRTCPData++ << 8;
- senderSSRC += *_ptrRTCPData++;
-
- WebRtc_UWord32 name = *_ptrRTCPData++ << 24;
- name += *_ptrRTCPData++ << 16;
- name += *_ptrRTCPData++ << 8;
- name += *_ptrRTCPData++;
-
- length = _ptrRTCPBlockEnd - _ptrRTCPData;
-
- _packetType = kRtcpAppCode;
-
- _packet.APP.SubType = header.IC;
- _packet.APP.Name = name;
-
- _state = State_AppItem;
- return true;
-}
-
-bool
-RTCPUtility::RTCPParserV2::ParseAPPItem()
-{
- const ptrdiff_t length = _ptrRTCPBlockEnd - _ptrRTCPData;
- if (length < 4)
- {
- _state = State_TopLevel;
-
- EndCurrentBlock();
- return false;
- }
- _packetType = kRtcpAppItemCode;
-
- if(length > kRtcpAppCode_DATA_SIZE)
- {
- memcpy(_packet.APP.Data, _ptrRTCPData, kRtcpAppCode_DATA_SIZE);
- _packet.APP.Size = kRtcpAppCode_DATA_SIZE;
- _ptrRTCPData += kRtcpAppCode_DATA_SIZE;
- }else
- {
- memcpy(_packet.APP.Data, _ptrRTCPData, length);
- _packet.APP.Size = (WebRtc_UWord16)length;
- _ptrRTCPData += length;
- }
- return true;
-}
-
-RTCPUtility::RTCPPacketIterator::RTCPPacketIterator(WebRtc_UWord8* rtcpData,
- size_t rtcpDataLength)
- : _ptrBegin(rtcpData),
- _ptrEnd(rtcpData + rtcpDataLength),
- _ptrBlock(NULL) {
- memset(&_header, 0, sizeof(_header));
-}
-
-RTCPUtility::RTCPPacketIterator::~RTCPPacketIterator() {
-}
-
-const RTCPUtility::RTCPCommonHeader*
-RTCPUtility::RTCPPacketIterator::Begin()
-{
- _ptrBlock = _ptrBegin;
-
- return Iterate();
-}
-
-const RTCPUtility::RTCPCommonHeader*
-RTCPUtility::RTCPPacketIterator::Iterate()
-{
- const bool success = RTCPParseCommonHeader(_ptrBlock, _ptrEnd, _header);
- if (!success)
- {
- _ptrBlock = NULL;
- return NULL;
- }
- _ptrBlock += _header.LengthInOctets;
-
- if (_ptrBlock > _ptrEnd)
- {
- _ptrBlock = NULL;
- return NULL;
- }
-
- return &_header;
-}
-
-const RTCPUtility::RTCPCommonHeader*
-RTCPUtility::RTCPPacketIterator::Current()
-{
- if (!_ptrBlock)
- {
- return NULL;
- }
-
- return &_header;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtcp_utility.h b/trunk/src/modules/rtp_rtcp/source/rtcp_utility.h
deleted file mode 100644
index faba2ef..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtcp_utility.h
+++ /dev/null
@@ -1,437 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_UTILITY_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_UTILITY_H_
-
-#include <cstddef> // size_t, ptrdiff_t
-
-#include "typedefs.h"
-#include "rtp_rtcp_config.h"
-#include "rtp_rtcp_defines.h"
-
-namespace webrtc {
-namespace RTCPUtility {
- // CNAME
- struct RTCPCnameInformation
- {
- char name[RTCP_CNAME_SIZE];
- };
- struct RTCPPacketRR
- {
- WebRtc_UWord32 SenderSSRC;
- WebRtc_UWord8 NumberOfReportBlocks;
- };
- struct RTCPPacketSR
- {
- WebRtc_UWord32 SenderSSRC;
- WebRtc_UWord8 NumberOfReportBlocks;
-
- // sender info
- WebRtc_UWord32 NTPMostSignificant;
- WebRtc_UWord32 NTPLeastSignificant;
- WebRtc_UWord32 RTPTimestamp;
- WebRtc_UWord32 SenderPacketCount;
- WebRtc_UWord32 SenderOctetCount;
- };
- struct RTCPPacketReportBlockItem
- {
- // report block
- WebRtc_UWord32 SSRC;
- WebRtc_UWord8 FractionLost;
- WebRtc_UWord32 CumulativeNumOfPacketsLost;
- WebRtc_UWord32 ExtendedHighestSequenceNumber;
- WebRtc_UWord32 Jitter;
- WebRtc_UWord32 LastSR;
- WebRtc_UWord32 DelayLastSR;
- };
- struct RTCPPacketSDESCName
- {
- // RFC3550
- WebRtc_UWord32 SenderSSRC;
- char CName[RTCP_CNAME_SIZE];
- };
-
- struct RTCPPacketExtendedJitterReportItem
- {
- // RFC 5450
- WebRtc_UWord32 Jitter;
- };
-
- struct RTCPPacketBYE
- {
- WebRtc_UWord32 SenderSSRC;
- };
- struct RTCPPacketXR
- {
- // RFC 3611
- WebRtc_UWord32 OriginatorSSRC;
- };
- struct RTCPPacketXRVOIPMetricItem
- {
- // RFC 3611 4.7
- WebRtc_UWord32 SSRC;
- WebRtc_UWord8 lossRate;
- WebRtc_UWord8 discardRate;
- WebRtc_UWord8 burstDensity;
- WebRtc_UWord8 gapDensity;
- WebRtc_UWord16 burstDuration;
- WebRtc_UWord16 gapDuration;
- WebRtc_UWord16 roundTripDelay;
- WebRtc_UWord16 endSystemDelay;
- WebRtc_UWord8 signalLevel;
- WebRtc_UWord8 noiseLevel;
- WebRtc_UWord8 RERL;
- WebRtc_UWord8 Gmin;
- WebRtc_UWord8 Rfactor;
- WebRtc_UWord8 extRfactor;
- WebRtc_UWord8 MOSLQ;
- WebRtc_UWord8 MOSCQ;
- WebRtc_UWord8 RXconfig;
- WebRtc_UWord16 JBnominal;
- WebRtc_UWord16 JBmax;
- WebRtc_UWord16 JBabsMax;
- };
-
- struct RTCPPacketRTPFBNACK
- {
- WebRtc_UWord32 SenderSSRC;
- WebRtc_UWord32 MediaSSRC;
- };
- struct RTCPPacketRTPFBNACKItem
- {
- // RFC4585
- WebRtc_UWord16 PacketID;
- WebRtc_UWord16 BitMask;
- };
-
- struct RTCPPacketRTPFBTMMBR
- {
- WebRtc_UWord32 SenderSSRC;
- WebRtc_UWord32 MediaSSRC; // zero!
- };
- struct RTCPPacketRTPFBTMMBRItem
- {
- // RFC5104
- WebRtc_UWord32 SSRC;
- WebRtc_UWord32 MaxTotalMediaBitRate; // In Kbit/s
- WebRtc_UWord32 MeasuredOverhead;
- };
-
- struct RTCPPacketRTPFBTMMBN
- {
- WebRtc_UWord32 SenderSSRC;
- WebRtc_UWord32 MediaSSRC; // zero!
- };
- struct RTCPPacketRTPFBTMMBNItem
- {
- // RFC5104
- WebRtc_UWord32 SSRC; // "Owner"
- WebRtc_UWord32 MaxTotalMediaBitRate;
- WebRtc_UWord32 MeasuredOverhead;
- };
-
- struct RTCPPacketPSFBFIR
- {
- WebRtc_UWord32 SenderSSRC;
- WebRtc_UWord32 MediaSSRC; // zero!
- };
- struct RTCPPacketPSFBFIRItem
- {
- // RFC5104
- WebRtc_UWord32 SSRC;
- WebRtc_UWord8 CommandSequenceNumber;
- };
-
- struct RTCPPacketPSFBPLI
- {
- // RFC4585
- WebRtc_UWord32 SenderSSRC;
- WebRtc_UWord32 MediaSSRC;
- };
-
- struct RTCPPacketPSFBSLI
- {
- // RFC4585
- WebRtc_UWord32 SenderSSRC;
- WebRtc_UWord32 MediaSSRC;
- };
- struct RTCPPacketPSFBSLIItem
- {
- // RFC4585
- WebRtc_UWord16 FirstMB;
- WebRtc_UWord16 NumberOfMB;
- WebRtc_UWord8 PictureId;
- };
- struct RTCPPacketPSFBRPSI
- {
- // RFC4585
- WebRtc_UWord32 SenderSSRC;
- WebRtc_UWord32 MediaSSRC;
- WebRtc_UWord8 PayloadType;
- WebRtc_UWord16 NumberOfValidBits;
- WebRtc_UWord8 NativeBitString[RTCP_RPSI_DATA_SIZE];
- };
- struct RTCPPacketPSFBREMB
- {
- WebRtc_UWord32 BitRate;
- };
- // generic name APP
- struct RTCPPacketAPP
- {
- WebRtc_UWord8 SubType;
- WebRtc_UWord32 Name;
- WebRtc_UWord8 Data[kRtcpAppCode_DATA_SIZE];
- WebRtc_UWord16 Size;
- };
-
- union RTCPPacket
- {
- RTCPPacketRR RR;
- RTCPPacketSR SR;
- RTCPPacketReportBlockItem ReportBlockItem;
-
- RTCPPacketSDESCName CName;
- RTCPPacketBYE BYE;
-
- RTCPPacketExtendedJitterReportItem ExtendedJitterReportItem;
-
- RTCPPacketRTPFBNACK NACK;
- RTCPPacketRTPFBNACKItem NACKItem;
-
- RTCPPacketPSFBPLI PLI;
- RTCPPacketPSFBSLI SLI;
- RTCPPacketPSFBSLIItem SLIItem;
- RTCPPacketPSFBRPSI RPSI;
- RTCPPacketPSFBREMB REMB;
-
- RTCPPacketRTPFBTMMBR TMMBR;
- RTCPPacketRTPFBTMMBRItem TMMBRItem;
- RTCPPacketRTPFBTMMBN TMMBN;
- RTCPPacketRTPFBTMMBNItem TMMBNItem;
- RTCPPacketPSFBFIR FIR;
- RTCPPacketPSFBFIRItem FIRItem;
-
- RTCPPacketXR XR;
- RTCPPacketXRVOIPMetricItem XRVOIPMetricItem;
-
- RTCPPacketAPP APP;
- };
-
- enum RTCPPacketTypes
- {
- kRtcpNotValidCode,
-
- // RFC3550
- kRtcpRrCode,
- kRtcpSrCode,
- kRtcpReportBlockItemCode,
-
- kRtcpSdesCode,
- kRtcpSdesChunkCode,
- kRtcpByeCode,
-
- // RFC5450
- kRtcpExtendedIjCode,
- kRtcpExtendedIjItemCode,
-
- // RFC4585
- kRtcpRtpfbNackCode,
- kRtcpRtpfbNackItemCode,
-
- kRtcpPsfbPliCode,
- kRtcpPsfbRpsiCode,
- kRtcpPsfbSliCode,
- kRtcpPsfbSliItemCode,
- kRtcpPsfbAppCode,
- kRtcpPsfbRembItemCode,
-
- // RFC5104
- kRtcpRtpfbTmmbrCode,
- kRtcpRtpfbTmmbrItemCode,
- kRtcpRtpfbTmmbnCode,
- kRtcpRtpfbTmmbnItemCode,
- kRtcpPsfbFirCode,
- kRtcpPsfbFirItemCode,
-
- // draft-perkins-avt-rapid-rtp-sync
- kRtcpRtpfbSrReqCode,
-
- // RFC 3611
- kRtcpXrVoipMetricCode,
-
- kRtcpAppCode,
- kRtcpAppItemCode,
- };
-
- struct RTCPRawPacket
- {
- const WebRtc_UWord8* _ptrPacketBegin;
- const WebRtc_UWord8* _ptrPacketEnd;
- };
-
- struct RTCPModRawPacket
- {
- WebRtc_UWord8* _ptrPacketBegin;
- WebRtc_UWord8* _ptrPacketEnd;
- };
-
- struct RTCPCommonHeader
- {
- WebRtc_UWord8 V; // Version
- bool P; // Padding
- WebRtc_UWord8 IC; // Item count/subtype
- WebRtc_UWord8 PT; // Packet Type
- WebRtc_UWord16 LengthInOctets;
- };
-
- enum RTCPPT
- {
- PT_IJ = 195,
- PT_SR = 200,
- PT_RR = 201,
- PT_SDES = 202,
- PT_BYE = 203,
- PT_APP = 204,
- PT_RTPFB = 205,
- PT_PSFB = 206,
- PT_XR = 207
- };
-
- bool RTCPParseCommonHeader( const WebRtc_UWord8* ptrDataBegin,
- const WebRtc_UWord8* ptrDataEnd,
- RTCPCommonHeader& parsedHeader);
-
- class RTCPParserV2
- {
- public:
- RTCPParserV2(const WebRtc_UWord8* rtcpData,
- size_t rtcpDataLength,
- bool rtcpReducedSizeEnable); // Set to true, to allow non-compound RTCP!
- ~RTCPParserV2();
-
- RTCPPacketTypes PacketType() const;
- const RTCPPacket& Packet() const;
- const RTCPRawPacket& RawPacket() const;
- ptrdiff_t LengthLeft() const;
-
- bool IsValid() const;
-
- RTCPPacketTypes Begin();
- RTCPPacketTypes Iterate();
-
- private:
- enum ParseState
- {
- State_TopLevel, // Top level packet
- State_ReportBlockItem, // SR/RR report block
- State_SDESChunk, // SDES chunk
- State_BYEItem, // BYE item
- State_ExtendedJitterItem, // Extended jitter report item
- State_RTPFB_NACKItem, // NACK FCI item
- State_RTPFB_TMMBRItem, // TMMBR FCI item
- State_RTPFB_TMMBNItem, // TMMBN FCI item
- State_PSFB_SLIItem, // SLI FCI item
- State_PSFB_RPSIItem, // RPSI FCI item
- State_PSFB_FIRItem, // FIR FCI item
- State_PSFB_AppItem, // Application specific FCI item
- State_PSFB_REMBItem, // Application specific REMB item
- State_XRItem,
- State_AppItem
- };
-
- private:
- void IterateTopLevel();
- void IterateReportBlockItem();
- void IterateSDESChunk();
- void IterateBYEItem();
- void IterateExtendedJitterItem();
- void IterateNACKItem();
- void IterateTMMBRItem();
- void IterateTMMBNItem();
- void IterateSLIItem();
- void IterateRPSIItem();
- void IterateFIRItem();
- void IteratePsfbAppItem();
- void IteratePsfbREMBItem();
- void IterateAppItem();
-
- void Validate();
- void EndCurrentBlock();
-
- bool ParseRR();
- bool ParseSR();
- bool ParseReportBlockItem();
-
- bool ParseSDES();
- bool ParseSDESChunk();
- bool ParseSDESItem();
-
- bool ParseBYE();
- bool ParseBYEItem();
-
- bool ParseIJ();
- bool ParseIJItem();
-
- bool ParseXR();
- bool ParseXRItem();
- bool ParseXRVOIPMetricItem();
-
- bool ParseFBCommon(const RTCPCommonHeader& header);
- bool ParseNACKItem();
- bool ParseTMMBRItem();
- bool ParseTMMBNItem();
- bool ParseSLIItem();
- bool ParseRPSIItem();
- bool ParseFIRItem();
- bool ParsePsfbAppItem();
- bool ParsePsfbREMBItem();
-
- bool ParseAPP(const RTCPCommonHeader& header);
- bool ParseAPPItem();
-
- private:
- const WebRtc_UWord8* const _ptrRTCPDataBegin;
- const bool _RTCPReducedSizeEnable;
- const WebRtc_UWord8* const _ptrRTCPDataEnd;
-
- bool _validPacket;
- const WebRtc_UWord8* _ptrRTCPData;
- const WebRtc_UWord8* _ptrRTCPBlockEnd;
-
- ParseState _state;
- WebRtc_UWord8 _numberOfBlocks;
-
- RTCPPacketTypes _packetType;
- RTCPPacket _packet;
- };
-
- class RTCPPacketIterator
- {
- public:
- RTCPPacketIterator(WebRtc_UWord8* rtcpData,
- size_t rtcpDataLength);
- ~RTCPPacketIterator();
-
- const RTCPCommonHeader* Begin();
- const RTCPCommonHeader* Iterate();
- const RTCPCommonHeader* Current();
-
- private:
- WebRtc_UWord8* const _ptrBegin;
- WebRtc_UWord8* const _ptrEnd;
-
- WebRtc_UWord8* _ptrBlock;
-
- RTCPCommonHeader _header;
- };
-} // RTCPUtility
-} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTCP_UTILITY_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_fec_unittest.cc b/trunk/src/modules/rtp_rtcp/source/rtp_fec_unittest.cc
deleted file mode 100644
index d794701..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_fec_unittest.cc
+++ /dev/null
@@ -1,550 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/rtp_rtcp/source/forward_error_correction.h"
-
-#include <gtest/gtest.h>
-#include <list>
-
-#include "rtp_utility.h"
-
-using webrtc::ForwardErrorCorrection;
-
-// Minimum RTP header size in bytes.
-const uint8_t kRtpHeaderSize = 12;
-
-// Transport header size in bytes. Assume UDP/IPv4 as a reasonable minimum.
-const uint8_t kTransportOverhead = 28;
-
-// Maximum number of media packets used in the FEC (RFC 5109).
-const uint8_t kMaxNumberMediaPackets = ForwardErrorCorrection::kMaxMediaPackets;
-
-template<typename T> void ClearList(std::list<T*>* my_list) {
- T* packet = NULL;
- while (!my_list->empty()) {
- packet = my_list->front();
- delete packet;
- my_list->pop_front();
- }
-}
-
-class RtpFecTest : public ::testing::Test {
- protected:
- RtpFecTest()
- : fec_(new ForwardErrorCorrection(0)),
- ssrc_(rand()),
- fec_seq_num_(0) {
- }
-
- ForwardErrorCorrection* fec_;
- int ssrc_;
- uint16_t fec_seq_num_;
-
- std::list<ForwardErrorCorrection::Packet*> media_packet_list_;
- std::list<ForwardErrorCorrection::Packet*> fec_packet_list_;
- std::list<ForwardErrorCorrection::ReceivedPacket*> received_packet_list_;
- std::list<ForwardErrorCorrection::RecoveredPacket*> recovered_packet_list_;
-
- // Media packet "i" is lost if media_loss_mask_[i] = 1,
- // received if media_loss_mask_[i] = 0.
- int media_loss_mask_[kMaxNumberMediaPackets];
-
- // FEC packet "i" is lost if fec_loss_mask_[i] = 1,
- // received if fec_loss_mask_[i] = 0.
- int fec_loss_mask_[kMaxNumberMediaPackets];
-
- // Construct the media packet list, up to |num_media_packets| packets.
- // Returns the next sequence number after the last media packet.
- // (this will be the sequence of the first FEC packet)
- int ConstructMediaPackets(int num_media_packets);
-
- // Construct the received packet list: a subset of the media and FEC packets.
- void NetworkReceivedPackets();
-
- // Add packet from |packet_list| to list of received packets, using the
- // |loss_mask|.
- // The |packet_list| may be a media packet list (is_fec = false), or a
- // FEC packet list (is_fec = true).
- void ReceivedPackets(
- const std::list<ForwardErrorCorrection::Packet*>& packet_list,
- int* loss_mask,
- bool is_fec);
-
- // Check for complete recovery after FEC decoding.
- bool IsRecoveryComplete();
-
- // Delete the received packets.
- void FreeRecoveredPacketList();
-
- // Delete the media and FEC packets.
- void TearDown();
-};
-
-TEST_F(RtpFecTest, HandleIncorrectInputs) {
- int num_important_packets = 0;
- bool use_unequal_protection = false;
- uint8_t protection_factor = 60;
-
- // Media packet list is empty.
- EXPECT_EQ(-1, fec_->GenerateFEC(media_packet_list_,
- protection_factor,
- num_important_packets,
- use_unequal_protection,
- &fec_packet_list_));
-
- int num_media_packets = 10;
- ConstructMediaPackets(num_media_packets);
-
- num_important_packets = -1;
- // Number of important packets below 0.
- EXPECT_EQ(-1, fec_->GenerateFEC(media_packet_list_,
- protection_factor,
- num_important_packets,
- use_unequal_protection,
- &fec_packet_list_));
-
- num_important_packets = 12;
- // Number of important packets greater than number of media packets.
- EXPECT_EQ(-1, fec_->GenerateFEC(media_packet_list_,
- protection_factor,
- num_important_packets,
- use_unequal_protection,
- &fec_packet_list_));
-
- num_media_packets = kMaxNumberMediaPackets + 1;
- ConstructMediaPackets(num_media_packets);
-
- num_important_packets = 0;
- // Number of media packet is above maximum allowed (kMaxNumberMediaPackets).
- EXPECT_EQ(-1, fec_->GenerateFEC(media_packet_list_,
- protection_factor,
- num_important_packets,
- use_unequal_protection,
- &fec_packet_list_));
-}
-
-TEST_F(RtpFecTest, FecRecoveryNoLoss) {
- const int num_important_packets = 0;
- const bool use_unequal_protection = false;
- const int num_media_packets = 4;
- uint8_t protection_factor = 60;
-
- fec_seq_num_ = ConstructMediaPackets(num_media_packets);
-
- EXPECT_EQ(0, fec_->GenerateFEC(media_packet_list_,
- protection_factor,
- num_important_packets,
- use_unequal_protection,
- &fec_packet_list_));
-
- // Expect 1 FEC packet.
- EXPECT_EQ(1, static_cast<int>(fec_packet_list_.size()));
-
- // No packets lost.
- memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
- memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
- NetworkReceivedPackets();
-
- EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
- &recovered_packet_list_));
-
- // No packets lost, expect complete recovery.
- EXPECT_TRUE(IsRecoveryComplete());
-}
-
-TEST_F(RtpFecTest, FecRecoveryWithLoss) {
- const int num_important_packets = 0;
- const bool use_unequal_protection = false;
- const int num_media_packets = 4;
- uint8_t protection_factor = 60;
-
- fec_seq_num_ = ConstructMediaPackets(num_media_packets);
-
- EXPECT_EQ(0, fec_->GenerateFEC(media_packet_list_,
- protection_factor,
- num_important_packets,
- use_unequal_protection,
- &fec_packet_list_));
-
- // Expect 1 FEC packet.
- EXPECT_EQ(1, static_cast<int>(fec_packet_list_.size()));
-
- // 1 media packet lost
- memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
- memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
- media_loss_mask_[3] = 1;
- NetworkReceivedPackets();
-
- EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
- &recovered_packet_list_));
-
- // One packet lost, one FEC packet, expect complete recovery.
- EXPECT_TRUE(IsRecoveryComplete());
- FreeRecoveredPacketList();
-
- // 2 media packets lost.
- memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
- memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
- media_loss_mask_[1] = 1;
- media_loss_mask_[3] = 1;
- NetworkReceivedPackets();
-
- EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
- &recovered_packet_list_));
-
- // 2 packets lost, one FEC packet, cannot get complete recovery.
- EXPECT_FALSE(IsRecoveryComplete());
-}
-
-TEST_F(RtpFecTest, FecRecoveryWithLoss50perc) {
- const int num_important_packets = 0;
- const bool use_unequal_protection = false;
- const int num_media_packets = 4;
- const uint8_t protection_factor = 255;
-
- // Packet Mask for (4,4,0) code:
- // (num_media_packets = 4; num_fec_packets = 4, num_important_packets = 0)
-
- // media#0 media#1 media#2 media#3
- // fec#0: 1 1 0 0
- // fec#1: 1 0 1 0
- // fec#2: 0 1 0 1
- // fec#3: 0 0 1 1
- //
-
- fec_seq_num_ = ConstructMediaPackets(num_media_packets);
-
- EXPECT_EQ(0, fec_->GenerateFEC(media_packet_list_,
- protection_factor,
- num_important_packets,
- use_unequal_protection,
- &fec_packet_list_));
-
- // Expect 4 FEC packets.
- EXPECT_EQ(4, static_cast<int>(fec_packet_list_.size()));
-
- // 4 packets lost: 3 media packets and one FEC packet#2 lost.
- memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
- memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
- fec_loss_mask_[2] = 1;
- media_loss_mask_[0] = 1;
- media_loss_mask_[2] = 1;
- media_loss_mask_[3] = 1;
-
- NetworkReceivedPackets();
-
- EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
- &recovered_packet_list_));
-
- // With media packet#1 and FEC packets #0, #1, #3, expect complete recovery.
- EXPECT_TRUE(IsRecoveryComplete());
- FreeRecoveredPacketList();
-
- // 4 packets lost: all media packets
- memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
- memset(fec_loss_mask_, 1, sizeof(fec_loss_mask_));
- media_loss_mask_[0] = 1;
- media_loss_mask_[1] = 1;
- media_loss_mask_[2] = 1;
- media_loss_mask_[3] = 1;
- NetworkReceivedPackets();
-
- EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
- &recovered_packet_list_));
-
- // Cannot get complete recovery for this loss configuration.
- EXPECT_FALSE(IsRecoveryComplete());
-}
-
-TEST_F(RtpFecTest, FecRecoveryNoLossUep) {
- const int num_important_packets = 2;
- const bool use_unequal_protection = true;
- const int num_media_packets = 4;
- const uint8_t protection_factor = 60;
-
- fec_seq_num_ = ConstructMediaPackets(num_media_packets);
-
- EXPECT_EQ(0, fec_->GenerateFEC(media_packet_list_,
- protection_factor,
- num_important_packets,
- use_unequal_protection,
- &fec_packet_list_));
-
- // Expect 1 FEC packet.
- EXPECT_EQ(1, static_cast<int>(fec_packet_list_.size()));
-
- // No packets lost.
- memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
- memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
- NetworkReceivedPackets();
-
- EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
- &recovered_packet_list_));
-
- // No packets lost, expect complete recovery.
- EXPECT_TRUE(IsRecoveryComplete());
-}
-
-TEST_F(RtpFecTest, FecRecoveryWithLossUep) {
- const int num_important_packets = 2;
- const bool use_unequal_protection = true;
- const int num_media_packets = 4;
- const uint8_t protection_factor = 60;
-
- fec_seq_num_ = ConstructMediaPackets(num_media_packets);
-
- EXPECT_EQ(0, fec_->GenerateFEC(media_packet_list_,
- protection_factor,
- num_important_packets,
- use_unequal_protection,
- &fec_packet_list_));
-
- // Expect 1 FEC packet.
- EXPECT_EQ(1, static_cast<int>(fec_packet_list_.size()));
-
- // 1 media packet lost.
- memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
- memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
- media_loss_mask_[3] = 1;
- NetworkReceivedPackets();
-
- EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
- &recovered_packet_list_));
-
- // One packet lost, one FEC packet, expect complete recovery.
- EXPECT_TRUE(IsRecoveryComplete());
- FreeRecoveredPacketList();
-
- // 2 media packets lost.
- memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
- memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
- media_loss_mask_[1] = 1;
- media_loss_mask_[3] = 1;
- NetworkReceivedPackets();
-
- EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
- &recovered_packet_list_));
-
- // 2 packets lost, one FEC packet, cannot get complete recovery.
- EXPECT_FALSE(IsRecoveryComplete());
-}
-
-TEST_F(RtpFecTest, FecRecoveryWithLoss50percUep) {
- const int num_important_packets = 1;
- const bool use_unequal_protection = true;
- const int num_media_packets = 4;
- const uint8_t protection_factor = 255;
-
- // Packet Mask for (4,4,1) code:
- // (num_media_packets = 4; num_fec_packets = 4, num_important_packets = 2)
-
- // media#0 media#1 media#2 media#3
- // fec#0: 1 0 0 0
- // fec#1: 1 1 0 0
- // fec#2: 1 0 1 1
- // fec#3: 0 1 1 0
- //
-
- fec_seq_num_ = ConstructMediaPackets(num_media_packets);
-
- EXPECT_EQ(0, fec_->GenerateFEC(media_packet_list_,
- protection_factor,
- num_important_packets,
- use_unequal_protection,
- &fec_packet_list_));
-
- // Expect 4 FEC packets.
- EXPECT_EQ(4, static_cast<int>(fec_packet_list_.size()));
-
- // 4 packets lost: 3 media packets and FEC packet#1 lost.
- memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
- memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
- fec_loss_mask_[1] = 1;
- media_loss_mask_[0] = 1;
- media_loss_mask_[2] = 1;
- media_loss_mask_[3] = 1;
- NetworkReceivedPackets();
-
- EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
- &recovered_packet_list_));
-
- // With media packet#1 and FEC packets #0, #2, #3, expect complete recovery.
- EXPECT_TRUE(IsRecoveryComplete());
- FreeRecoveredPacketList();
-
- // 4 packets lost: 3 media packets and one FEC packet#2 lost.
- memset(media_loss_mask_, 0, sizeof(media_loss_mask_));
- memset(fec_loss_mask_, 0, sizeof(fec_loss_mask_));
- fec_loss_mask_[2] = 1;
- media_loss_mask_[0] = 1;
- media_loss_mask_[2] = 1;
- media_loss_mask_[3] = 1;
- NetworkReceivedPackets();
-
- EXPECT_EQ(0, fec_->DecodeFEC(&received_packet_list_ ,
- &recovered_packet_list_));
-
- // Cannot get complete recovery for this loss configuration.
- EXPECT_FALSE(IsRecoveryComplete());
-}
-
-// TODO(marpan): Add more test cases.
-
-void RtpFecTest::TearDown() {
- fec_->ResetState(&recovered_packet_list_);
- delete fec_;
- FreeRecoveredPacketList();
- ClearList(&media_packet_list_);
- EXPECT_TRUE(media_packet_list_.empty());
-}
-
-void RtpFecTest::FreeRecoveredPacketList() {
- ClearList(&recovered_packet_list_);
-}
-
-bool RtpFecTest::IsRecoveryComplete() {
- // Check that the number of media and recovered packets are equal.
- if (media_packet_list_.size() != recovered_packet_list_.size()) {
- return false;
- }
-
- ForwardErrorCorrection::Packet* media_packet;
- ForwardErrorCorrection::RecoveredPacket* recovered_packet;
-
- bool recovery = true;
-
- std::list<ForwardErrorCorrection::Packet*>::iterator
- media_packet_list_item = media_packet_list_.begin();
- std::list<ForwardErrorCorrection::RecoveredPacket*>::iterator
- recovered_packet_list_item = recovered_packet_list_.begin();
- while (media_packet_list_item != media_packet_list_.end()) {
- if (recovered_packet_list_item == recovered_packet_list_.end()) {
- return false;
- }
- media_packet = *media_packet_list_item;
- recovered_packet = *recovered_packet_list_item;
- if (recovered_packet->pkt->length != media_packet->length) {
- return false;
- }
- if (memcmp(recovered_packet->pkt->data, media_packet->data,
- media_packet->length) != 0) {
- return false;
- }
- media_packet_list_item++;
- recovered_packet_list_item++;
- }
- return recovery;
-}
-
-void RtpFecTest::NetworkReceivedPackets() {
- const bool kFecPacket = true;
- ReceivedPackets(media_packet_list_, media_loss_mask_, !kFecPacket);
- ReceivedPackets(fec_packet_list_, fec_loss_mask_, kFecPacket);
-}
-
-void RtpFecTest:: ReceivedPackets(
- const std::list<ForwardErrorCorrection::Packet*>& packet_list,
- int* loss_mask,
- bool is_fec) {
- ForwardErrorCorrection::Packet* packet;
- ForwardErrorCorrection::ReceivedPacket* received_packet;
- int seq_num = fec_seq_num_;
- int packet_idx = 0;
-
- std::list<ForwardErrorCorrection::Packet*>::const_iterator
- packet_list_item = packet_list.begin();
-
- while (packet_list_item != packet_list.end()) {
- packet = *packet_list_item;
- if (loss_mask[packet_idx] == 0) {
- received_packet = new ForwardErrorCorrection::ReceivedPacket;
- received_packet->pkt = new ForwardErrorCorrection::Packet;
- received_packet_list_.push_back(received_packet);
- received_packet->pkt->length = packet->length;
- memcpy(received_packet->pkt->data, packet->data,
- packet->length);
- received_packet->isFec = is_fec;
- if (!is_fec) {
- // For media packets, the sequence number and marker bit is
- // obtained from RTP header. These were set in ConstructMediaPackets().
- received_packet->seqNum =
- webrtc::ModuleRTPUtility::BufferToUWord16(&packet->data[2]);
- }
- else {
- // The sequence number, marker bit, and ssrc number are defined in the
- // RTP header of the FEC packet, which is not constructed in this test.
- // So we set these values below based on the values generated in
- // ConstructMediaPackets().
- received_packet->seqNum = seq_num;
- // The ssrc value for FEC packets is set to the one used for the
- // media packets in ConstructMediaPackets().
- received_packet->ssrc = ssrc_;
- }
- }
- packet_idx++;
- packet_list_item ++;
- // Sequence number of FEC packets are defined as increment by 1 from
- // last media packet in frame.
- if (is_fec) seq_num++;
- }
-}
-
-int RtpFecTest::ConstructMediaPackets(int num_media_packets) {
- assert(num_media_packets > 0);
- ForwardErrorCorrection::Packet* media_packet = NULL;
- int sequence_number = rand();
- int time_stamp = rand();
-
- for (int i = 0; i < num_media_packets; i++) {
- media_packet = new ForwardErrorCorrection::Packet;
- media_packet_list_.push_back(media_packet);
- media_packet->length =
- static_cast<uint16_t>((static_cast<float>(rand()) / RAND_MAX) *
- (IP_PACKET_SIZE - kRtpHeaderSize - kTransportOverhead -
- ForwardErrorCorrection::PacketOverhead()));
-
- if (media_packet->length < kRtpHeaderSize) {
- media_packet->length = kRtpHeaderSize;
- }
- // Generate random values for the first 2 bytes
- media_packet->data[0] = static_cast<uint8_t>(rand() % 256);
- media_packet->data[1] = static_cast<uint8_t>(rand() % 256);
-
- // The first two bits are assumed to be 10 by the FEC encoder.
- // In fact the FEC decoder will set the two first bits to 10 regardless of
- // what they actually were. Set the first two bits to 10 so that a memcmp
- // can be performed for the whole restored packet.
- media_packet->data[0] |= 0x80;
- media_packet->data[0] &= 0xbf;
-
- // FEC is applied to a whole frame.
- // A frame is signaled by multiple packets without the marker bit set
- // followed by the last packet of the frame for which the marker bit is set.
- // Only push one (fake) frame to the FEC.
- media_packet->data[1] &= 0x7f;
-
- webrtc::ModuleRTPUtility::AssignUWord16ToBuffer(&media_packet->data[2],
- sequence_number);
- webrtc::ModuleRTPUtility::AssignUWord32ToBuffer(&media_packet->data[4],
- time_stamp);
- webrtc::ModuleRTPUtility::AssignUWord32ToBuffer(&media_packet->data[8],
- ssrc_);
-
- // Generate random values for payload.
- for (int j = 12; j < media_packet->length; j++) {
- media_packet->data[j] = static_cast<uint8_t> (rand() % 256);
- }
- sequence_number++;
- }
- // Last packet, set marker bit.
- assert(media_packet != NULL);
- media_packet->data[1] |= 0x80;
- return sequence_number;
-}
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8.cc b/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8.cc
deleted file mode 100644
index f066ae2..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8.cc
+++ /dev/null
@@ -1,469 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
-
-#include <string.h> // memcpy
-
-#include <cassert> // assert
-#include <vector>
-
-#include "modules/rtp_rtcp/source/vp8_partition_aggregator.h"
-
-namespace webrtc {
-
-// Define how the VP8PacketizerModes are implemented.
-// Modes are: kStrict, kAggregate, kEqualSize.
-const RtpFormatVp8::AggregationMode RtpFormatVp8::aggr_modes_[kNumModes] =
- { kAggrNone, kAggrPartitions, kAggrFragments };
-const bool RtpFormatVp8::balance_modes_[kNumModes] =
- { true, true, true };
-const bool RtpFormatVp8::separate_first_modes_[kNumModes] =
- { true, false, false };
-
-RtpFormatVp8::RtpFormatVp8(const WebRtc_UWord8* payload_data,
- WebRtc_UWord32 payload_size,
- const RTPVideoHeaderVP8& hdr_info,
- int max_payload_len,
- const RTPFragmentationHeader& fragmentation,
- VP8PacketizerMode mode)
- : payload_data_(payload_data),
- payload_size_(static_cast<int>(payload_size)),
- vp8_fixed_payload_descriptor_bytes_(1),
- aggr_mode_(aggr_modes_[mode]),
- balance_(balance_modes_[mode]),
- separate_first_(separate_first_modes_[mode]),
- hdr_info_(hdr_info),
- num_partitions_(fragmentation.fragmentationVectorSize),
- max_payload_len_(max_payload_len),
- packets_calculated_(false) {
- part_info_ = fragmentation;
-}
-
-RtpFormatVp8::RtpFormatVp8(const WebRtc_UWord8* payload_data,
- WebRtc_UWord32 payload_size,
- const RTPVideoHeaderVP8& hdr_info,
- int max_payload_len)
- : payload_data_(payload_data),
- payload_size_(static_cast<int>(payload_size)),
- part_info_(),
- vp8_fixed_payload_descriptor_bytes_(1),
- aggr_mode_(aggr_modes_[kEqualSize]),
- balance_(balance_modes_[kEqualSize]),
- separate_first_(separate_first_modes_[kEqualSize]),
- hdr_info_(hdr_info),
- num_partitions_(1),
- max_payload_len_(max_payload_len),
- packets_calculated_(false) {
- part_info_.VerifyAndAllocateFragmentationHeader(1);
- part_info_.fragmentationLength[0] = payload_size;
- part_info_.fragmentationOffset[0] = 0;
-}
-
-int RtpFormatVp8::NextPacket(WebRtc_UWord8* buffer,
- int* bytes_to_send,
- bool* last_packet) {
- if (!packets_calculated_) {
- int ret = 0;
- if (aggr_mode_ == kAggrPartitions && balance_) {
- ret = GeneratePacketsBalancedAggregates();
- } else {
- ret = GeneratePackets();
- }
- if (ret < 0) {
- return ret;
- }
- }
- if (packets_.empty()) {
- return -1;
- }
- InfoStruct packet_info = packets_.front();
- packets_.pop();
-
- *bytes_to_send = WriteHeaderAndPayload(packet_info, buffer, max_payload_len_);
- if (*bytes_to_send < 0) {
- return -1;
- }
-
- *last_packet = packets_.empty();
- return packet_info.first_partition_ix;
-}
-
-int RtpFormatVp8::CalcNextSize(int max_payload_len, int remaining_bytes,
- bool split_payload) const {
- if (max_payload_len == 0 || remaining_bytes == 0) {
- return 0;
- }
- if (!split_payload) {
- return max_payload_len >= remaining_bytes ? remaining_bytes : 0;
- }
-
- if (balance_) {
- // Balance payload sizes to produce (almost) equal size
- // fragments.
- // Number of fragments for remaining_bytes:
- int num_frags = remaining_bytes / max_payload_len + 1;
- // Number of bytes in this fragment:
- return static_cast<int>(static_cast<double>(remaining_bytes)
- / num_frags + 0.5);
- } else {
- return max_payload_len >= remaining_bytes ? remaining_bytes
- : max_payload_len;
- }
-}
-
-int RtpFormatVp8::GeneratePackets() {
- if (max_payload_len_ < vp8_fixed_payload_descriptor_bytes_
- + PayloadDescriptorExtraLength() + 1) {
- // The provided payload length is not long enough for the payload
- // descriptor and one payload byte. Return an error.
- return -1;
- }
- int total_bytes_processed = 0;
- bool start_on_new_fragment = true;
- bool beginning = true;
- int part_ix = 0;
- while (total_bytes_processed < payload_size_) {
- int packet_bytes = 0; // How much data to send in this packet.
- bool split_payload = true; // Splitting of partitions is initially allowed.
- int remaining_in_partition = part_info_.fragmentationOffset[part_ix] -
- total_bytes_processed + part_info_.fragmentationLength[part_ix];
- int rem_payload_len = max_payload_len_ -
- (vp8_fixed_payload_descriptor_bytes_ + PayloadDescriptorExtraLength());
- int first_partition_in_packet = part_ix;
-
- while (int next_size = CalcNextSize(rem_payload_len, remaining_in_partition,
- split_payload)) {
- packet_bytes += next_size;
- rem_payload_len -= next_size;
- remaining_in_partition -= next_size;
-
- if (remaining_in_partition == 0 && !(beginning && separate_first_)) {
- // Advance to next partition?
- // Check that there are more partitions; verify that we are either
- // allowed to aggregate fragments, or that we are allowed to
- // aggregate intact partitions and that we started this packet
- // with an intact partition (indicated by first_fragment_ == true).
- if (part_ix + 1 < num_partitions_ &&
- ((aggr_mode_ == kAggrFragments) ||
- (aggr_mode_ == kAggrPartitions && start_on_new_fragment))) {
- assert(part_ix < num_partitions_);
- remaining_in_partition = part_info_.fragmentationLength[++part_ix];
- // Disallow splitting unless kAggrFragments. In kAggrPartitions,
- // we can only aggregate intact partitions.
- split_payload = (aggr_mode_ == kAggrFragments);
- }
- } else if (balance_ && remaining_in_partition > 0) {
- break;
- }
- }
- if (remaining_in_partition == 0) {
- ++part_ix; // Advance to next partition.
- }
- assert(packet_bytes > 0);
-
- QueuePacket(total_bytes_processed, packet_bytes, first_partition_in_packet,
- start_on_new_fragment);
- total_bytes_processed += packet_bytes;
- start_on_new_fragment = (remaining_in_partition == 0);
- beginning = false; // Next packet cannot be first packet in frame.
- }
- packets_calculated_ = true;
- assert(total_bytes_processed == payload_size_);
- return 0;
-}
-
-int RtpFormatVp8::GeneratePacketsBalancedAggregates() {
- if (max_payload_len_ < vp8_fixed_payload_descriptor_bytes_
- + PayloadDescriptorExtraLength() + 1) {
- // The provided payload length is not long enough for the payload
- // descriptor and one payload byte. Return an error.
- return -1;
- }
- std::vector<int> partition_decision;
- const int overhead = vp8_fixed_payload_descriptor_bytes_ +
- PayloadDescriptorExtraLength();
- const uint32_t max_payload_len = max_payload_len_ - overhead;
- int min_size, max_size;
- AggregateSmallPartitions(&partition_decision, &min_size, &max_size);
-
- int total_bytes_processed = 0;
- int part_ix = 0;
- while (part_ix < num_partitions_) {
- if (partition_decision[part_ix] == -1) {
- // Split large partitions.
- int remaining_partition = part_info_.fragmentationLength[part_ix];
- int num_fragments = Vp8PartitionAggregator::CalcNumberOfFragments(
- remaining_partition, max_payload_len, overhead, min_size, max_size);
- const int packet_bytes =
- (remaining_partition + num_fragments - 1) / num_fragments;
- for (int n = 0; n < num_fragments; ++n) {
- const int this_packet_bytes = packet_bytes < remaining_partition ?
- packet_bytes : remaining_partition;
- QueuePacket(total_bytes_processed, this_packet_bytes, part_ix,
- (n == 0));
- remaining_partition -= this_packet_bytes;
- total_bytes_processed += this_packet_bytes;
- if (this_packet_bytes < min_size) {
- min_size = this_packet_bytes;
- }
- if (this_packet_bytes > max_size) {
- max_size = this_packet_bytes;
- }
- }
- assert(remaining_partition == 0);
- ++part_ix;
- } else {
- int this_packet_bytes = 0;
- const int first_partition_in_packet = part_ix;
- const int aggregation_index = partition_decision[part_ix];
- while (static_cast<size_t>(part_ix) < partition_decision.size() &&
- partition_decision[part_ix] == aggregation_index) {
- // Collect all partitions that were aggregated into the same packet.
- this_packet_bytes += part_info_.fragmentationLength[part_ix];
- ++part_ix;
- }
- QueuePacket(total_bytes_processed, this_packet_bytes,
- first_partition_in_packet, true);
- total_bytes_processed += this_packet_bytes;
- }
- }
- packets_calculated_ = true;
- return 0;
-}
-
-void RtpFormatVp8::AggregateSmallPartitions(std::vector<int>* partition_vec,
- int* min_size,
- int* max_size) {
- assert(min_size && max_size);
- *min_size = -1;
- *max_size = -1;
- assert(partition_vec);
- partition_vec->assign(num_partitions_, -1);
- const int overhead = vp8_fixed_payload_descriptor_bytes_ +
- PayloadDescriptorExtraLength();
- const uint32_t max_payload_len = max_payload_len_ - overhead;
- int first_in_set = 0;
- int last_in_set = 0;
- int num_aggregate_packets = 0;
- // Find sets of partitions smaller than max_payload_len_.
- while (first_in_set < num_partitions_) {
- if (part_info_.fragmentationLength[first_in_set] < max_payload_len) {
- // Found start of a set.
- last_in_set = first_in_set;
- while (last_in_set + 1 < num_partitions_ &&
- part_info_.fragmentationLength[last_in_set + 1] < max_payload_len) {
- ++last_in_set;
- }
- // Found end of a set. Run optimized aggregator. It is ok if start == end.
- Vp8PartitionAggregator aggregator(part_info_, first_in_set,
- last_in_set);
- if (*min_size >= 0 && *max_size >= 0) {
- aggregator.SetPriorMinMax(*min_size, *max_size);
- }
- Vp8PartitionAggregator::ConfigVec optimal_config =
- aggregator.FindOptimalConfiguration(max_payload_len, overhead);
- aggregator.CalcMinMax(optimal_config, min_size, max_size);
- for (int i = first_in_set, j = 0; i <= last_in_set; ++i, ++j) {
- // Transfer configuration for this set of partitions to the joint
- // partition vector representing all partitions in the frame.
- (*partition_vec)[i] = num_aggregate_packets + optimal_config[j];
- }
- num_aggregate_packets += optimal_config.back() + 1;
- first_in_set = last_in_set;
- }
- ++first_in_set;
- }
-}
-
-void RtpFormatVp8::QueuePacket(int start_pos,
- int packet_size,
- int first_partition_in_packet,
- bool start_on_new_fragment) {
- // Write info to packet info struct and store in packet info queue.
- InfoStruct packet_info;
- packet_info.payload_start_pos = start_pos;
- packet_info.size = packet_size;
- packet_info.first_partition_ix = first_partition_in_packet;
- packet_info.first_fragment = start_on_new_fragment;
- packets_.push(packet_info);
-}
-
-int RtpFormatVp8::WriteHeaderAndPayload(const InfoStruct& packet_info,
- WebRtc_UWord8* buffer,
- int buffer_length) const {
- // Write the VP8 payload descriptor.
- // 0
- // 0 1 2 3 4 5 6 7 8
- // +-+-+-+-+-+-+-+-+-+
- // |X| |N|S| PART_ID |
- // +-+-+-+-+-+-+-+-+-+
- // X: |I|L|T|K| | (mandatory if any of the below are used)
- // +-+-+-+-+-+-+-+-+-+
- // I: |PictureID (8/16b)| (optional)
- // +-+-+-+-+-+-+-+-+-+
- // L: | TL0PIC_IDX | (optional)
- // +-+-+-+-+-+-+-+-+-+
- // T/K: |TID:Y| KEYIDX | (optional)
- // +-+-+-+-+-+-+-+-+-+
-
- assert(packet_info.size > 0);
- buffer[0] = 0;
- if (XFieldPresent()) buffer[0] |= kXBit;
- if (hdr_info_.nonReference) buffer[0] |= kNBit;
- if (packet_info.first_fragment) buffer[0] |= kSBit;
- buffer[0] |= (packet_info.first_partition_ix & kPartIdField);
-
- const int extension_length = WriteExtensionFields(buffer, buffer_length);
-
- memcpy(&buffer[vp8_fixed_payload_descriptor_bytes_ + extension_length],
- &payload_data_[packet_info.payload_start_pos], packet_info.size);
-
- // Return total length of written data.
- return packet_info.size + vp8_fixed_payload_descriptor_bytes_
- + extension_length;
-}
-
-int RtpFormatVp8::WriteExtensionFields(WebRtc_UWord8* buffer,
- int buffer_length) const {
- int extension_length = 0;
- if (XFieldPresent()) {
- WebRtc_UWord8* x_field = buffer + vp8_fixed_payload_descriptor_bytes_;
- *x_field = 0;
- extension_length = 1; // One octet for the X field.
- if (PictureIdPresent()) {
- if (WritePictureIDFields(x_field, buffer, buffer_length,
- &extension_length) < 0) {
- return -1;
- }
- }
- if (TL0PicIdxFieldPresent()) {
- if (WriteTl0PicIdxFields(x_field, buffer, buffer_length,
- &extension_length) < 0) {
- return -1;
- }
- }
- if (TIDFieldPresent() || KeyIdxFieldPresent()) {
- if (WriteTIDAndKeyIdxFields(x_field, buffer, buffer_length,
- &extension_length) < 0) {
- return -1;
- }
- }
- assert(extension_length == PayloadDescriptorExtraLength());
- }
- return extension_length;
-}
-
-int RtpFormatVp8::WritePictureIDFields(WebRtc_UWord8* x_field,
- WebRtc_UWord8* buffer,
- int buffer_length,
- int* extension_length) const {
- *x_field |= kIBit;
- const int pic_id_length = WritePictureID(
- buffer + vp8_fixed_payload_descriptor_bytes_ + *extension_length,
- buffer_length - vp8_fixed_payload_descriptor_bytes_
- - *extension_length);
- if (pic_id_length < 0) return -1;
- *extension_length += pic_id_length;
- return 0;
-}
-
-int RtpFormatVp8::WritePictureID(WebRtc_UWord8* buffer,
- int buffer_length) const {
- const WebRtc_UWord16 pic_id =
- static_cast<WebRtc_UWord16> (hdr_info_.pictureId);
- int picture_id_len = PictureIdLength();
- if (picture_id_len > buffer_length) return -1;
- if (picture_id_len == 2) {
- buffer[0] = 0x80 | ((pic_id >> 8) & 0x7F);
- buffer[1] = pic_id & 0xFF;
- } else if (picture_id_len == 1) {
- buffer[0] = pic_id & 0x7F;
- }
- return picture_id_len;
-}
-
-int RtpFormatVp8::WriteTl0PicIdxFields(WebRtc_UWord8* x_field,
- WebRtc_UWord8* buffer,
- int buffer_length,
- int* extension_length) const {
- if (buffer_length < vp8_fixed_payload_descriptor_bytes_ + *extension_length
- + 1) {
- return -1;
- }
- *x_field |= kLBit;
- buffer[vp8_fixed_payload_descriptor_bytes_
- + *extension_length] = hdr_info_.tl0PicIdx;
- ++*extension_length;
- return 0;
-}
-
-int RtpFormatVp8::WriteTIDAndKeyIdxFields(WebRtc_UWord8* x_field,
- WebRtc_UWord8* buffer,
- int buffer_length,
- int* extension_length) const {
- if (buffer_length < vp8_fixed_payload_descriptor_bytes_ + *extension_length
- + 1) {
- return -1;
- }
- WebRtc_UWord8* data_field =
- &buffer[vp8_fixed_payload_descriptor_bytes_ + *extension_length];
- *data_field = 0;
- if (TIDFieldPresent()) {
- *x_field |= kTBit;
- assert(hdr_info_.temporalIdx >= 0 && hdr_info_.temporalIdx <= 3);
- *data_field |= hdr_info_.temporalIdx << 6;
- *data_field |= hdr_info_.layerSync ? kYBit : 0;
- }
- if (KeyIdxFieldPresent()) {
- *x_field |= kKBit;
- *data_field |= (hdr_info_.keyIdx & kKeyIdxField);
- }
- ++*extension_length;
- return 0;
-}
-
-int RtpFormatVp8::PayloadDescriptorExtraLength() const {
- int length_bytes = PictureIdLength();
- if (TL0PicIdxFieldPresent()) ++length_bytes;
- if (TIDFieldPresent() || KeyIdxFieldPresent()) ++length_bytes;
- if (length_bytes > 0) ++length_bytes; // Include the extension field.
- return length_bytes;
-}
-
-int RtpFormatVp8::PictureIdLength() const {
- if (hdr_info_.pictureId == kNoPictureId) {
- return 0;
- }
- if (hdr_info_.pictureId <= 0x7F) {
- return 1;
- }
- return 2;
-}
-
-bool RtpFormatVp8::XFieldPresent() const {
- return (TIDFieldPresent() || TL0PicIdxFieldPresent() || PictureIdPresent()
- || KeyIdxFieldPresent());
-}
-
-bool RtpFormatVp8::TIDFieldPresent() const {
- assert((hdr_info_.layerSync == false) ||
- (hdr_info_.temporalIdx != kNoTemporalIdx));
- return (hdr_info_.temporalIdx != kNoTemporalIdx);
-}
-
-bool RtpFormatVp8::KeyIdxFieldPresent() const {
- return (hdr_info_.keyIdx != kNoKeyIdx);
-}
-
-bool RtpFormatVp8::TL0PicIdxFieldPresent() const {
- return (hdr_info_.tl0PicIdx != kNoTl0PicIdx);
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8.h b/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8.h
deleted file mode 100644
index f568f4d..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8.h
+++ /dev/null
@@ -1,206 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the declaration of the VP8 packetizer class.
- * A packetizer object is created for each encoded video frame. The
- * constructor is called with the payload data and size,
- * together with the fragmentation information and a packetizer mode
- * of choice. Alternatively, if no fragmentation info is available, the
- * second constructor can be used with only payload data and size; in that
- * case the mode kEqualSize is used.
- *
- * After creating the packetizer, the method NextPacket is called
- * repeatedly to get all packets for the frame. The method returns
- * false as long as there are more packets left to fetch.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_H_
-
-#include <queue>
-#include <vector>
-
-#include "modules/interface/module_common_types.h"
-#include "system_wrappers/interface/constructor_magic.h"
-#include "typedefs.h" // NOLINT(build/include)
-
-namespace webrtc {
-
-enum VP8PacketizerMode {
- kStrict = 0, // Split partitions if too large;
- // never aggregate, balance size.
- kAggregate, // Split partitions if too large; aggregate whole partitions.
- kEqualSize, // Split entire payload without considering partition limits.
- // This will produce equal size packets for the whole frame.
- kNumModes,
-};
-
-// Packetizer for VP8.
-class RtpFormatVp8 {
- public:
- // Initialize with payload from encoder and fragmentation info.
- // The payload_data must be exactly one encoded VP8 frame.
- RtpFormatVp8(const WebRtc_UWord8* payload_data,
- WebRtc_UWord32 payload_size,
- const RTPVideoHeaderVP8& hdr_info,
- int max_payload_len,
- const RTPFragmentationHeader& fragmentation,
- VP8PacketizerMode mode);
-
- // Initialize without fragmentation info. Mode kEqualSize will be used.
- // The payload_data must be exactly one encoded VP8 frame.
- RtpFormatVp8(const WebRtc_UWord8* payload_data,
- WebRtc_UWord32 payload_size,
- const RTPVideoHeaderVP8& hdr_info,
- int max_payload_len);
-
- // Get the next payload with VP8 payload header.
- // max_payload_len limits the sum length of payload and VP8 payload header.
- // buffer is a pointer to where the output will be written.
- // bytes_to_send is an output variable that will contain number of bytes
- // written to buffer. Parameter last_packet is true for the last packet of
- // the frame, false otherwise (i.e., call the function again to get the
- // next packet).
- // For the kStrict and kAggregate mode: returns the partition index from which
- // the first payload byte in the packet is taken, with the first partition
- // having index 0; returns negative on error.
- // For the kEqualSize mode: returns 0 on success, return negative on error.
- int NextPacket(WebRtc_UWord8* buffer,
- int* bytes_to_send,
- bool* last_packet);
-
- private:
- typedef struct {
- int payload_start_pos;
- int size;
- bool first_fragment;
- int first_partition_ix;
- } InfoStruct;
- typedef std::queue<InfoStruct> InfoQueue;
- enum AggregationMode {
- kAggrNone = 0, // No aggregation.
- kAggrPartitions, // Aggregate intact partitions.
- kAggrFragments // Aggregate intact and fragmented partitions.
- };
-
- static const AggregationMode aggr_modes_[kNumModes];
- static const bool balance_modes_[kNumModes];
- static const bool separate_first_modes_[kNumModes];
- static const int kXBit = 0x80;
- static const int kNBit = 0x20;
- static const int kSBit = 0x10;
- static const int kPartIdField = 0x0F;
- static const int kKeyIdxField = 0x1F;
- static const int kIBit = 0x80;
- static const int kLBit = 0x40;
- static const int kTBit = 0x20;
- static const int kKBit = 0x10;
- static const int kYBit = 0x20;
-
- // Calculate size of next chunk to send. Returns 0 if none can be sent.
- int CalcNextSize(int max_payload_len, int remaining_bytes,
- bool split_payload) const;
-
- // Calculate all packet sizes and load to packet info queue.
- int GeneratePackets();
-
- // Calculate all packet sizes using Vp8PartitionAggregator and load to packet
- // info queue.
- int GeneratePacketsBalancedAggregates();
-
- // Helper function to GeneratePacketsBalancedAggregates(). Find all
- // continuous sets of partitions smaller than the max payload size (not
- // max_size), and aggregate them into balanced packets. The result is written
- // to partition_vec, which is of the same length as the number of partitions.
- // A value of -1 indicates that the partition is too large and must be split.
- // Aggregates are numbered 0, 1, 2, etc. For each set of small partitions,
- // the aggregate numbers restart at 0. Output values min_size and max_size
- // will hold the smallest and largest resulting aggregates (i.e., not counting
- // those that must be split).
- void AggregateSmallPartitions(std::vector<int>* partition_vec,
- int* min_size,
- int* max_size);
-
- // Insert packet into packet queue.
- void QueuePacket(int start_pos,
- int packet_size,
- int first_partition_in_packet,
- bool start_on_new_fragment);
-
- // Write the payload header and copy the payload to the buffer.
- // The info in packet_info determines which part of the payload is written
- // and what to write in the header fields.
- int WriteHeaderAndPayload(const InfoStruct& packet_info,
- WebRtc_UWord8* buffer,
- int buffer_length) const;
-
-
- // Write the X field and the appropriate extension fields to buffer.
- // The function returns the extension length (including X field), or -1
- // on error.
- int WriteExtensionFields(WebRtc_UWord8* buffer, int buffer_length) const;
-
- // Set the I bit in the x_field, and write PictureID to the appropriate
- // position in buffer. The function returns 0 on success, -1 otherwise.
- int WritePictureIDFields(WebRtc_UWord8* x_field, WebRtc_UWord8* buffer,
- int buffer_length, int* extension_length) const;
-
- // Set the L bit in the x_field, and write Tl0PicIdx to the appropriate
- // position in buffer. The function returns 0 on success, -1 otherwise.
- int WriteTl0PicIdxFields(WebRtc_UWord8* x_field, WebRtc_UWord8* buffer,
- int buffer_length, int* extension_length) const;
-
- // Set the T and K bits in the x_field, and write TID, Y and KeyIdx to the
- // appropriate position in buffer. The function returns 0 on success,
- // -1 otherwise.
- int WriteTIDAndKeyIdxFields(WebRtc_UWord8* x_field, WebRtc_UWord8* buffer,
- int buffer_length, int* extension_length) const;
-
- // Write the PictureID from codec_specific_info_ to buffer. One or two
- // bytes are written, depending on magnitude of PictureID. The function
- // returns the number of bytes written.
- int WritePictureID(WebRtc_UWord8* buffer, int buffer_length) const;
-
- // Calculate and return length (octets) of the variable header fields in
- // the next header (i.e., header length in addition to vp8_header_bytes_).
- int PayloadDescriptorExtraLength() const;
-
- // Calculate and return length (octets) of PictureID field in the next
- // header. Can be 0, 1, or 2.
- int PictureIdLength() const;
-
- // Check whether each of the optional fields will be included in the header.
- bool XFieldPresent() const;
- bool TIDFieldPresent() const;
- bool KeyIdxFieldPresent() const;
- bool TL0PicIdxFieldPresent() const;
- bool PictureIdPresent() const { return (PictureIdLength() > 0); }
-
- const WebRtc_UWord8* payload_data_;
- const int payload_size_;
- RTPFragmentationHeader part_info_;
- const int vp8_fixed_payload_descriptor_bytes_; // Length of VP8 payload
- // descriptors's fixed part.
- const AggregationMode aggr_mode_;
- const bool balance_;
- const bool separate_first_;
- const RTPVideoHeaderVP8 hdr_info_;
- const int num_partitions_;
- const int max_payload_len_;
- InfoQueue packets_;
- bool packets_calculated_;
-
- DISALLOW_COPY_AND_ASSIGN(RtpFormatVp8);
-};
-
-} // namespace
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc b/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
deleted file mode 100644
index 59e76ee..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.cc
+++ /dev/null
@@ -1,248 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-#include "modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h"
-
-#include "gtest/gtest.h"
-
-namespace webrtc {
-
-namespace test {
-
-RtpFormatVp8TestHelper::RtpFormatVp8TestHelper(const RTPVideoHeaderVP8* hdr)
- : payload_data_(NULL),
- buffer_(NULL),
- fragmentation_(NULL),
- hdr_info_(hdr),
- sloppy_partitioning_(false),
- inited_(false) {}
-
-RtpFormatVp8TestHelper::~RtpFormatVp8TestHelper() {
- delete fragmentation_;
- delete [] payload_data_;
- delete [] buffer_;
-}
-
-bool RtpFormatVp8TestHelper::Init(const int* partition_sizes,
- int num_partitions) {
- if (inited_) return false;
- fragmentation_ = new RTPFragmentationHeader;
- fragmentation_->VerifyAndAllocateFragmentationHeader(num_partitions);
- payload_size_ = 0;
- // Calculate sum payload size.
- for (int p = 0; p < num_partitions; ++p) {
- payload_size_ += partition_sizes[p];
- }
- buffer_size_ = payload_size_ + 6; // Add space for payload descriptor.
- payload_data_ = new WebRtc_UWord8[payload_size_];
- buffer_ = new WebRtc_UWord8[buffer_size_];
- int j = 0;
- // Loop through the partitions again.
- for (int p = 0; p < num_partitions; ++p) {
- fragmentation_->fragmentationLength[p] = partition_sizes[p];
- fragmentation_->fragmentationOffset[p] = j;
- for (int i = 0; i < partition_sizes[p]; ++i) {
- assert(j < payload_size_);
- payload_data_[j++] = p; // Set the payload value to the partition index.
- }
- }
- data_ptr_ = payload_data_;
- inited_ = true;
- return true;
-}
-
-void RtpFormatVp8TestHelper::GetAllPacketsAndCheck(
- RtpFormatVp8* packetizer,
- const int* expected_sizes,
- const int* expected_part,
- const bool* expected_frag_start,
- int expected_num_packets) {
- ASSERT_TRUE(inited_);
- int send_bytes = 0;
- bool last;
- for (int i = 0; i < expected_num_packets; ++i) {
- std::ostringstream ss;
- ss << "Checking packet " << i;
- SCOPED_TRACE(ss.str());
- EXPECT_EQ(expected_part[i],
- packetizer->NextPacket(buffer_, &send_bytes, &last));
- CheckPacket(send_bytes, expected_sizes[i], last,
- expected_frag_start[i]);
- }
- EXPECT_TRUE(last);
-}
-
-// Payload descriptor
-// 0 1 2 3 4 5 6 7
-// +-+-+-+-+-+-+-+-+
-// |X|R|N|S|PartID | (REQUIRED)
-// +-+-+-+-+-+-+-+-+
-// X: |I|L|T|K| RSV | (OPTIONAL)
-// +-+-+-+-+-+-+-+-+
-// I: | PictureID | (OPTIONAL)
-// +-+-+-+-+-+-+-+-+
-// L: | TL0PICIDX | (OPTIONAL)
-// +-+-+-+-+-+-+-+-+
-// T/K: | TID | KEYIDX | (OPTIONAL)
-// +-+-+-+-+-+-+-+-+
-
-// First octet tests.
-#define EXPECT_BIT_EQ(x, n, a) EXPECT_EQ((((x) >> (n)) & 0x1), a)
-
-#define EXPECT_RSV_ZERO(x) EXPECT_EQ(((x) & 0xE0), 0)
-
-#define EXPECT_BIT_X_EQ(x, a) EXPECT_BIT_EQ(x, 7, a)
-
-#define EXPECT_BIT_N_EQ(x, a) EXPECT_BIT_EQ(x, 5, a)
-
-#define EXPECT_BIT_S_EQ(x, a) EXPECT_BIT_EQ(x, 4, a)
-
-#define EXPECT_PART_ID_EQ(x, a) EXPECT_EQ(((x) & 0x0F), a)
-
-// Extension fields tests
-#define EXPECT_BIT_I_EQ(x, a) EXPECT_BIT_EQ(x, 7, a)
-
-#define EXPECT_BIT_L_EQ(x, a) EXPECT_BIT_EQ(x, 6, a)
-
-#define EXPECT_BIT_T_EQ(x, a) EXPECT_BIT_EQ(x, 5, a)
-
-#define EXPECT_BIT_K_EQ(x, a) EXPECT_BIT_EQ(x, 4, a)
-
-#define EXPECT_TID_EQ(x, a) EXPECT_EQ((((x) & 0xC0) >> 6), a)
-
-#define EXPECT_BIT_Y_EQ(x, a) EXPECT_BIT_EQ(x, 5, a)
-
-#define EXPECT_KEYIDX_EQ(x, a) EXPECT_EQ(((x) & 0x1F), a)
-
-void RtpFormatVp8TestHelper::CheckHeader(bool frag_start) {
- payload_start_ = 1;
- EXPECT_BIT_EQ(buffer_[0], 6, 0); // Check reserved bit.
-
- if (hdr_info_->pictureId != kNoPictureId ||
- hdr_info_->temporalIdx != kNoTemporalIdx ||
- hdr_info_->tl0PicIdx != kNoTl0PicIdx ||
- hdr_info_->keyIdx != kNoKeyIdx) {
- EXPECT_BIT_X_EQ(buffer_[0], 1);
- ++payload_start_;
- CheckPictureID();
- CheckTl0PicIdx();
- CheckTIDAndKeyIdx();
- } else {
- EXPECT_BIT_X_EQ(buffer_[0], 0);
- }
-
- EXPECT_BIT_N_EQ(buffer_[0], hdr_info_->nonReference ? 1 : 0);
- EXPECT_BIT_S_EQ(buffer_[0], frag_start ? 1 : 0);
-
- // Check partition index.
- if (!sloppy_partitioning_) {
- // The test payload data is constructed such that the payload value is the
- // same as the partition index.
- EXPECT_EQ(buffer_[0] & 0x0F, buffer_[payload_start_]);
- } else {
- // Partition should be set to 0.
- EXPECT_EQ(buffer_[0] & 0x0F, 0);
- }
-}
-
-// Verify that the I bit and the PictureID field are both set in accordance
-// with the information in hdr_info_->pictureId.
-void RtpFormatVp8TestHelper::CheckPictureID() {
- if (hdr_info_->pictureId != kNoPictureId) {
- EXPECT_BIT_I_EQ(buffer_[1], 1);
- if (hdr_info_->pictureId > 0x7F) {
- EXPECT_BIT_EQ(buffer_[payload_start_], 7, 1);
- EXPECT_EQ(buffer_[payload_start_] & 0x7F,
- (hdr_info_->pictureId >> 8) & 0x7F);
- EXPECT_EQ(buffer_[payload_start_ + 1],
- hdr_info_->pictureId & 0xFF);
- payload_start_ += 2;
- } else {
- EXPECT_BIT_EQ(buffer_[payload_start_], 7, 0);
- EXPECT_EQ(buffer_[payload_start_] & 0x7F,
- (hdr_info_->pictureId) & 0x7F);
- payload_start_ += 1;
- }
- } else {
- EXPECT_BIT_I_EQ(buffer_[1], 0);
- }
-}
-
-// Verify that the L bit and the TL0PICIDX field are both set in accordance
-// with the information in hdr_info_->tl0PicIdx.
-void RtpFormatVp8TestHelper::CheckTl0PicIdx() {
- if (hdr_info_->tl0PicIdx != kNoTl0PicIdx) {
- EXPECT_BIT_L_EQ(buffer_[1], 1);
- EXPECT_EQ(buffer_[payload_start_], hdr_info_->tl0PicIdx);
- ++payload_start_;
- } else {
- EXPECT_BIT_L_EQ(buffer_[1], 0);
- }
-}
-
-// Verify that the T bit and the TL0PICIDX field, and the K bit and KEYIDX
-// field are all set in accordance with the information in
-// hdr_info_->temporalIdx and hdr_info_->keyIdx, respectively.
-void RtpFormatVp8TestHelper::CheckTIDAndKeyIdx() {
- if (hdr_info_->temporalIdx == kNoTemporalIdx &&
- hdr_info_->keyIdx == kNoKeyIdx) {
- EXPECT_BIT_T_EQ(buffer_[1], 0);
- EXPECT_BIT_K_EQ(buffer_[1], 0);
- return;
- }
- if (hdr_info_->temporalIdx != kNoTemporalIdx) {
- EXPECT_BIT_T_EQ(buffer_[1], 1);
- EXPECT_TID_EQ(buffer_[payload_start_], hdr_info_->temporalIdx);
- EXPECT_BIT_Y_EQ(buffer_[payload_start_], hdr_info_->layerSync ? 1 : 0);
- } else {
- EXPECT_BIT_T_EQ(buffer_[1], 0);
- EXPECT_TID_EQ(buffer_[payload_start_], 0);
- EXPECT_BIT_Y_EQ(buffer_[payload_start_], 0);
- }
- if (hdr_info_->keyIdx != kNoKeyIdx) {
- EXPECT_BIT_K_EQ(buffer_[1], 1);
- EXPECT_KEYIDX_EQ(buffer_[payload_start_], hdr_info_->keyIdx);
- } else {
- EXPECT_BIT_K_EQ(buffer_[1], 0);
- EXPECT_KEYIDX_EQ(buffer_[payload_start_], 0);
- }
- ++payload_start_;
-}
-
-// Verify that the payload (i.e., after the headers) of the packet stored in
-// buffer_ is identical to the expected (as found in data_ptr_).
-void RtpFormatVp8TestHelper::CheckPayload(int payload_end) {
- for (int i = payload_start_; i < payload_end; ++i, ++data_ptr_)
- EXPECT_EQ(buffer_[i], *data_ptr_);
-}
-
-// Verify that the input variable "last" agrees with the position of data_ptr_.
-// If data_ptr_ has advanced payload_size_ bytes from the start (payload_data_)
-// we are at the end and last should be true. Otherwise, it should be false.
-void RtpFormatVp8TestHelper::CheckLast(bool last) const {
- EXPECT_EQ(last, data_ptr_ == payload_data_ + payload_size_);
-}
-
-// Verify the contents of a packet. Check the length versus expected_bytes,
-// the header, payload, and "last" flag.
-void RtpFormatVp8TestHelper::CheckPacket(int send_bytes,
- int expect_bytes,
- bool last,
- bool frag_start) {
- EXPECT_EQ(expect_bytes, send_bytes);
- CheckHeader(frag_start);
- CheckPayload(send_bytes);
- CheckLast(last);
-}
-
-} // namespace test
-
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h b/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
deleted file mode 100644
index f9bf686..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file contains the class RtpFormatVp8TestHelper. The class is
-// responsible for setting up a fake VP8 bitstream according to the
-// RTPVideoHeaderVP8 header, and partition information. After initialization,
-// an RTPFragmentationHeader is provided so that the tester can create a
-// packetizer. The packetizer can then be provided to this helper class, which
-// will then extract all packets and compare to the expected outcome.
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
-
-#include "modules/interface/module_common_types.h"
-#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
-#include "system_wrappers/interface/constructor_magic.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-namespace test {
-
-class RtpFormatVp8TestHelper {
- public:
- explicit RtpFormatVp8TestHelper(const RTPVideoHeaderVP8* hdr);
- ~RtpFormatVp8TestHelper();
- bool Init(const int* partition_sizes, int num_partitions);
- void GetAllPacketsAndCheck(RtpFormatVp8* packetizer,
- const int* expected_sizes,
- const int* expected_part,
- const bool* expected_frag_start,
- int expected_num_packets);
-
- uint8_t* payload_data() const { return payload_data_; }
- int payload_size() const { return payload_size_; }
- RTPFragmentationHeader* fragmentation() const { return fragmentation_; }
- int buffer_size() const { return buffer_size_; }
- void set_sloppy_partitioning(bool value) { sloppy_partitioning_ = value; }
-
- private:
- void CheckHeader(bool frag_start);
- void CheckPictureID();
- void CheckTl0PicIdx();
- void CheckTIDAndKeyIdx();
- void CheckPayload(int payload_end);
- void CheckLast(bool last) const;
- void CheckPacket(int send_bytes, int expect_bytes, bool last,
- bool frag_start);
-
- uint8_t* payload_data_;
- uint8_t* buffer_;
- uint8_t* data_ptr_;
- RTPFragmentationHeader* fragmentation_;
- const RTPVideoHeaderVP8* hdr_info_;
- int payload_start_;
- int payload_size_;
- int buffer_size_;
- bool sloppy_partitioning_;
- bool inited_;
-
- DISALLOW_COPY_AND_ASSIGN(RtpFormatVp8TestHelper);
-};
-
-} // namespace test
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_FORMAT_VP8_TEST_HELPER_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc b/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
deleted file mode 100644
index a6f9632..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_format_vp8_unittest.cc
+++ /dev/null
@@ -1,364 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes unit tests for the VP8 packetizer.
- */
-
-#include <gtest/gtest.h>
-
-#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
-#include "modules/rtp_rtcp/source/rtp_format_vp8_test_helper.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-template <bool>
-struct CompileAssert {
-};
-
-#undef COMPILE_ASSERT
-#define COMPILE_ASSERT(expr, msg) \
- typedef CompileAssert<(bool(expr))> msg[bool(expr) ? 1 : -1]
-
-class RtpFormatVp8Test : public ::testing::Test {
- protected:
- RtpFormatVp8Test() : helper_(NULL) {}
- virtual void TearDown() { delete helper_; }
- bool Init(const int* partition_sizes, int num_partitions) {
- hdr_info_.pictureId = kNoPictureId;
- hdr_info_.nonReference = false;
- hdr_info_.temporalIdx = kNoTemporalIdx;
- hdr_info_.layerSync = false;
- hdr_info_.tl0PicIdx = kNoTl0PicIdx;
- hdr_info_.keyIdx = kNoKeyIdx;
- if (helper_ != NULL) return false;
- helper_ = new test::RtpFormatVp8TestHelper(&hdr_info_);
- return helper_->Init(partition_sizes, num_partitions);
- }
-
- RTPVideoHeaderVP8 hdr_info_;
- test::RtpFormatVp8TestHelper* helper_;
-};
-
-TEST_F(RtpFormatVp8Test, TestStrictMode) {
- const int kSizeVector[] = {10, 8, 27};
- const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
- ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
-
- hdr_info_.pictureId = 200; // > 0x7F should produce 2-byte PictureID.
- const int kMaxSize = 13;
- RtpFormatVp8 packetizer(helper_->payload_data(),
- helper_->payload_size(),
- hdr_info_,
- kMaxSize,
- *(helper_->fragmentation()),
- kStrict);
-
- // The expected sizes are obtained by running a verified good implementation.
- const int kExpectedSizes[] = {9, 9, 12, 11, 11, 11, 10};
- const int kExpectedPart[] = {0, 0, 1, 2, 2, 2, 2};
- const bool kExpectedFragStart[] =
- {true, false, true, true, false, false, false};
- const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
- kExpectedPart_wrong_size);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
- kExpectedFragStart_wrong_size);
-
- helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
- kExpectedFragStart, kExpectedNum);
-}
-
-TEST_F(RtpFormatVp8Test, TestAggregateMode) {
- const int kSizeVector[] = {60, 10, 10};
- const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
- ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
-
- hdr_info_.pictureId = 20; // <= 0x7F should produce 1-byte PictureID.
- const int kMaxSize = 25;
- RtpFormatVp8 packetizer(helper_->payload_data(),
- helper_->payload_size(),
- hdr_info_,
- kMaxSize,
- *(helper_->fragmentation()),
- kAggregate);
-
- // The expected sizes are obtained by running a verified good implementation.
- const int kExpectedSizes[] = {23, 23, 23, 23};
- const int kExpectedPart[] = {0, 0, 0, 1};
- const bool kExpectedFragStart[] = {true, false, false, true};
- const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
- kExpectedPart_wrong_size);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
- kExpectedFragStart_wrong_size);
-
- helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
- kExpectedFragStart, kExpectedNum);
-}
-
-TEST_F(RtpFormatVp8Test, TestAggregateModeManyPartitions1) {
- const int kSizeVector[] = {1600, 200, 200, 200, 200, 200, 200, 200, 200};
- const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
- ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
-
- hdr_info_.pictureId = 20; // <= 0x7F should produce 1-byte PictureID.
- const int kMaxSize = 1500;
- RtpFormatVp8 packetizer(helper_->payload_data(),
- helper_->payload_size(),
- hdr_info_,
- kMaxSize,
- *(helper_->fragmentation()),
- kAggregate);
-
- // The expected sizes are obtained by running a verified good implementation.
- const int kExpectedSizes[] = {803, 803, 803, 803};
- const int kExpectedPart[] = {0, 0, 1, 5};
- const bool kExpectedFragStart[] = {true, false, true, true};
- const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
- kExpectedPart_wrong_size);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
- kExpectedFragStart_wrong_size);
-
- helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
- kExpectedFragStart, kExpectedNum);
-}
-
-TEST_F(RtpFormatVp8Test, TestAggregateModeManyPartitions2) {
- const int kSizeVector[] = {1599, 200, 200, 200, 1600, 200, 200, 200, 200};
- const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
- ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
-
- hdr_info_.pictureId = 20; // <= 0x7F should produce 1-byte PictureID.
- const int kMaxSize = 1500;
- RtpFormatVp8 packetizer(helper_->payload_data(),
- helper_->payload_size(),
- hdr_info_,
- kMaxSize,
- *(helper_->fragmentation()),
- kAggregate);
-
- // The expected sizes are obtained by running a verified good implementation.
- const int kExpectedSizes[] = {803, 802, 603, 803, 803, 803};
- const int kExpectedPart[] = {0, 0, 1, 4, 4, 5};
- const bool kExpectedFragStart[] = {true, false, true, true, false, true};
- const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
- kExpectedPart_wrong_size);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
- kExpectedFragStart_wrong_size);
-
- helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
- kExpectedFragStart, kExpectedNum);
-}
-
-TEST_F(RtpFormatVp8Test, TestAggregateModeTwoLargePartitions) {
- const int kSizeVector[] = {1654, 2268};
- const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
- ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
-
- hdr_info_.pictureId = 20; // <= 0x7F should produce 1-byte PictureID.
- const int kMaxSize = 1460;
- RtpFormatVp8 packetizer(helper_->payload_data(),
- helper_->payload_size(),
- hdr_info_,
- kMaxSize,
- *(helper_->fragmentation()),
- kAggregate);
-
- // The expected sizes are obtained by running a verified good implementation.
- const int kExpectedSizes[] = {830, 830, 1137, 1137};
- const int kExpectedPart[] = {0, 0, 1, 1};
- const bool kExpectedFragStart[] = {true, false, true, false};
- const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
- kExpectedPart_wrong_size);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
- kExpectedFragStart_wrong_size);
-
- helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
- kExpectedFragStart, kExpectedNum);
-}
-
-// Verify that EqualSize mode is forced if fragmentation info is missing.
-TEST_F(RtpFormatVp8Test, TestEqualSizeModeFallback) {
- const int kSizeVector[] = {10, 10, 10};
- const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
- ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
-
- hdr_info_.pictureId = 200; // > 0x7F should produce 2-byte PictureID
- const int kMaxSize = 12; // Small enough to produce 4 packets.
- RtpFormatVp8 packetizer(helper_->payload_data(),
- helper_->payload_size(),
- hdr_info_,
- kMaxSize);
-
- // Expecting three full packets, and one with the remainder.
- const int kExpectedSizes[] = {12, 11, 12, 11};
- const int kExpectedPart[] = {0, 0, 0, 0}; // Always 0 for equal size mode.
- // Frag start only true for first packet in equal size mode.
- const bool kExpectedFragStart[] = {true, false, false, false};
- const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
- kExpectedPart_wrong_size);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
- kExpectedFragStart_wrong_size);
-
- helper_->set_sloppy_partitioning(true);
- helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
- kExpectedFragStart, kExpectedNum);
-}
-
-// Verify that non-reference bit is set. EqualSize mode fallback is expected.
-TEST_F(RtpFormatVp8Test, TestNonReferenceBit) {
- const int kSizeVector[] = {10, 10, 10};
- const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
- ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
-
- hdr_info_.nonReference = true;
- const int kMaxSize = 25; // Small enough to produce two packets.
- RtpFormatVp8 packetizer(helper_->payload_data(),
- helper_->payload_size(),
- hdr_info_,
- kMaxSize);
-
- // EqualSize mode => First packet full; other not.
- const int kExpectedSizes[] = {16, 16};
- const int kExpectedPart[] = {0, 0}; // Always 0 for equal size mode.
- // Frag start only true for first packet in equal size mode.
- const bool kExpectedFragStart[] = {true, false};
- const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
- kExpectedPart_wrong_size);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
- kExpectedFragStart_wrong_size);
-
- helper_->set_sloppy_partitioning(true);
- helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
- kExpectedFragStart, kExpectedNum);
-}
-
-// Verify Tl0PicIdx and TID fields, and layerSync bit.
-TEST_F(RtpFormatVp8Test, TestTl0PicIdxAndTID) {
- const int kSizeVector[] = {10, 10, 10};
- const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
- ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
-
- hdr_info_.tl0PicIdx = 117;
- hdr_info_.temporalIdx = 2;
- hdr_info_.layerSync = true;
- // kMaxSize is only limited by allocated buffer size.
- const int kMaxSize = helper_->buffer_size();
- RtpFormatVp8 packetizer(helper_->payload_data(),
- helper_->payload_size(),
- hdr_info_,
- kMaxSize,
- *(helper_->fragmentation()),
- kAggregate);
-
- // Expect one single packet of payload_size() + 4 bytes header.
- const int kExpectedSizes[1] = {helper_->payload_size() + 4};
- const int kExpectedPart[1] = {0}; // Packet starts with partition 0.
- const bool kExpectedFragStart[1] = {true};
- const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
- kExpectedPart_wrong_size);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
- kExpectedFragStart_wrong_size);
-
- helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
- kExpectedFragStart, kExpectedNum);
-}
-
-// Verify KeyIdx field.
-TEST_F(RtpFormatVp8Test, TestKeyIdx) {
- const int kSizeVector[] = {10, 10, 10};
- const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
- ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
-
- hdr_info_.keyIdx = 17;
- // kMaxSize is only limited by allocated buffer size.
- const int kMaxSize = helper_->buffer_size();
- RtpFormatVp8 packetizer(helper_->payload_data(),
- helper_->payload_size(),
- hdr_info_,
- kMaxSize,
- *(helper_->fragmentation()),
- kAggregate);
-
- // Expect one single packet of payload_size() + 3 bytes header.
- const int kExpectedSizes[1] = {helper_->payload_size() + 3};
- const int kExpectedPart[1] = {0}; // Packet starts with partition 0.
- const bool kExpectedFragStart[1] = {true};
- const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
- kExpectedPart_wrong_size);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
- kExpectedFragStart_wrong_size);
-
- helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
- kExpectedFragStart, kExpectedNum);
-}
-
-// Verify TID field and KeyIdx field in combination.
-TEST_F(RtpFormatVp8Test, TestTIDAndKeyIdx) {
- const int kSizeVector[] = {10, 10, 10};
- const int kNumPartitions = sizeof(kSizeVector) / sizeof(kSizeVector[0]);
- ASSERT_TRUE(Init(kSizeVector, kNumPartitions));
-
- hdr_info_.temporalIdx = 1;
- hdr_info_.keyIdx = 5;
- // kMaxSize is only limited by allocated buffer size.
- const int kMaxSize = helper_->buffer_size();
- RtpFormatVp8 packetizer(helper_->payload_data(),
- helper_->payload_size(),
- hdr_info_,
- kMaxSize,
- *(helper_->fragmentation()),
- kAggregate);
-
- // Expect one single packet of payload_size() + 3 bytes header.
- const int kExpectedSizes[1] = {helper_->payload_size() + 3};
- const int kExpectedPart[1] = {0}; // Packet starts with partition 0.
- const bool kExpectedFragStart[1] = {true};
- const int kExpectedNum = sizeof(kExpectedSizes) / sizeof(kExpectedSizes[0]);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedPart) / sizeof(kExpectedPart[0]),
- kExpectedPart_wrong_size);
- COMPILE_ASSERT(kExpectedNum ==
- sizeof(kExpectedFragStart) / sizeof(kExpectedFragStart[0]),
- kExpectedFragStart_wrong_size);
-
- helper_->GetAllPacketsAndCheck(&packetizer, kExpectedSizes, kExpectedPart,
- kExpectedFragStart, kExpectedNum);
-}
-
-} // namespace
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_header_extension.cc b/trunk/src/modules/rtp_rtcp/source/rtp_header_extension.cc
deleted file mode 100644
index 37a87d7..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_header_extension.cc
+++ /dev/null
@@ -1,176 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cassert>
-
-#include "common_types.h"
-#include "rtp_header_extension.h"
-
-namespace webrtc {
-
-RtpHeaderExtensionMap::RtpHeaderExtensionMap() {
-}
-
-RtpHeaderExtensionMap::~RtpHeaderExtensionMap() {
- Erase();
-}
-
-void RtpHeaderExtensionMap::Erase() {
- while (!extensionMap_.empty()) {
- std::map<uint8_t, HeaderExtension*>::iterator it =
- extensionMap_.begin();
- delete it->second;
- extensionMap_.erase(it);
- }
-}
-
-int32_t RtpHeaderExtensionMap::Register(const RTPExtensionType type,
- const uint8_t id) {
- if (id < 1 || id > 14) {
- return -1;
- }
- std::map<uint8_t, HeaderExtension*>::iterator it =
- extensionMap_.find(id);
- if (it != extensionMap_.end()) {
- return -1;
- }
- extensionMap_[id] = new HeaderExtension(type);
- return 0;
-}
-
-int32_t RtpHeaderExtensionMap::Deregister(const RTPExtensionType type) {
- uint8_t id;
- if (GetId(type, &id) != 0) {
- return -1;
- }
- std::map<uint8_t, HeaderExtension*>::iterator it =
- extensionMap_.find(id);
- if (it == extensionMap_.end()) {
- return -1;
- }
- delete it->second;
- extensionMap_.erase(it);
- return 0;
-}
-
-int32_t RtpHeaderExtensionMap::GetType(const uint8_t id,
- RTPExtensionType* type) const {
- assert(type);
- std::map<uint8_t, HeaderExtension*>::const_iterator it =
- extensionMap_.find(id);
- if (it == extensionMap_.end()) {
- return -1;
- }
- HeaderExtension* extension = it->second;
- *type = extension->type;
- return 0;
-}
-
-int32_t RtpHeaderExtensionMap::GetId(const RTPExtensionType type,
- uint8_t* id) const {
- assert(id);
- std::map<uint8_t, HeaderExtension*>::const_iterator it =
- extensionMap_.begin();
-
- while (it != extensionMap_.end()) {
- HeaderExtension* extension = it->second;
- if (extension->type == type) {
- *id = it->first;
- return 0;
- }
- it++;
- }
- return -1;
-}
-
-uint16_t RtpHeaderExtensionMap::GetTotalLengthInBytes() const {
- // Get length for each extension block.
- uint16_t length = 0;
- std::map<uint8_t, HeaderExtension*>::const_iterator it =
- extensionMap_.begin();
- while (it != extensionMap_.end()) {
- HeaderExtension* extension = it->second;
- length += extension->length;
- it++;
- }
- // Add RTP extension header length.
- if (length > 0) {
- length += RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES;
- }
- return length;
-}
-
-int32_t RtpHeaderExtensionMap::GetLengthUntilBlockStartInBytes(
- const RTPExtensionType type) const {
- uint8_t id;
- if (GetId(type, &id) != 0) {
- // Not registered.
- return -1;
- }
- // Get length until start of extension block type.
- uint16_t length = RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES;
-
- std::map<uint8_t, HeaderExtension*>::const_iterator it =
- extensionMap_.begin();
- while (it != extensionMap_.end()) {
- HeaderExtension* extension = it->second;
- if (extension->type == type) {
- break;
- } else {
- length += extension->length;
- }
- it++;
- }
- return length;
-}
-
-int32_t RtpHeaderExtensionMap::Size() const {
- return extensionMap_.size();
-}
-
-RTPExtensionType RtpHeaderExtensionMap::First() const {
- std::map<uint8_t, HeaderExtension*>::const_iterator it =
- extensionMap_.begin();
- if (it == extensionMap_.end()) {
- return kRtpExtensionNone;
- }
- HeaderExtension* extension = it->second;
- return extension->type;
-}
-
-RTPExtensionType RtpHeaderExtensionMap::Next(RTPExtensionType type) const {
- uint8_t id;
- if (GetId(type, &id) != 0) {
- return kRtpExtensionNone;
- }
- std::map<uint8_t, HeaderExtension*>::const_iterator it =
- extensionMap_.find(id);
- if (it == extensionMap_.end()) {
- return kRtpExtensionNone;
- }
- it++;
- if (it == extensionMap_.end()) {
- return kRtpExtensionNone;
- }
- HeaderExtension* extension = it->second;
- return extension->type;
-}
-
-void RtpHeaderExtensionMap::GetCopy(RtpHeaderExtensionMap* map) const {
- assert(map);
- std::map<uint8_t, HeaderExtension*>::const_iterator it =
- extensionMap_.begin();
- while (it != extensionMap_.end()) {
- HeaderExtension* extension = it->second;
- map->Register(extension->type, it->first);
- it++;
- }
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_header_extension.h b/trunk/src/modules/rtp_rtcp/source/rtp_header_extension.h
deleted file mode 100644
index b1612cc..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_header_extension.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
-#define WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
-
-#include <map>
-
-#include "rtp_rtcp_defines.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-enum {RTP_ONE_BYTE_HEADER_EXTENSION = 0xbede};
-
-enum ExtensionLength {
- RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES = 4,
- TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES = 4
-};
-
-struct HeaderExtension {
- HeaderExtension(RTPExtensionType extension_type)
- : type(extension_type),
- length(0) {
- if (type == kRtpExtensionTransmissionTimeOffset) {
- length = TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES;
- }
- }
-
- const RTPExtensionType type;
- uint8_t length;
-};
-
-class RtpHeaderExtensionMap {
- public:
- RtpHeaderExtensionMap();
- ~RtpHeaderExtensionMap();
-
- void Erase();
-
- int32_t Register(const RTPExtensionType type, const uint8_t id);
-
- int32_t Deregister(const RTPExtensionType type);
-
- int32_t GetType(const uint8_t id, RTPExtensionType* type) const;
-
- int32_t GetId(const RTPExtensionType type, uint8_t* id) const;
-
- uint16_t GetTotalLengthInBytes() const;
-
- int32_t GetLengthUntilBlockStartInBytes(const RTPExtensionType type) const;
-
- void GetCopy(RtpHeaderExtensionMap* map) const;
-
- int32_t Size() const;
-
- RTPExtensionType First() const;
-
- RTPExtensionType Next(RTPExtensionType type) const;
-
- private:
- std::map<uint8_t, HeaderExtension*> extensionMap_;
-};
-}
-#endif // WEBRTC_MODULES_RTP_RTCP_RTP_HEADER_EXTENSION_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_header_extension_test.cc b/trunk/src/modules/rtp_rtcp/source/rtp_header_extension_test.cc
deleted file mode 100644
index 003e92c..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_header_extension_test.cc
+++ /dev/null
@@ -1,117 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes unit tests for the RtpHeaderExtensionMap.
- */
-
-#include <gtest/gtest.h>
-
-#include "rtp_header_extension.h"
-#include "rtp_rtcp_defines.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-class RtpHeaderExtensionTest : public ::testing::Test {
- protected:
- RtpHeaderExtensionTest() {}
- ~RtpHeaderExtensionTest() {}
-
- RtpHeaderExtensionMap map_;
- enum {kId = 3};
-};
-
-TEST_F(RtpHeaderExtensionTest, Register) {
- EXPECT_EQ(0, map_.Size());
- EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
- EXPECT_EQ(1, map_.Size());
- EXPECT_EQ(0, map_.Deregister(kRtpExtensionTransmissionTimeOffset));
- EXPECT_EQ(0, map_.Size());
-}
-
-TEST_F(RtpHeaderExtensionTest, RegisterIllegalArg) {
- // Valid range for id: [1-14].
- EXPECT_EQ(-1, map_.Register(kRtpExtensionTransmissionTimeOffset, 0));
- EXPECT_EQ(-1, map_.Register(kRtpExtensionTransmissionTimeOffset, 15));
-}
-
-TEST_F(RtpHeaderExtensionTest, DeregisterIllegalArg) {
- // Not registered.
- EXPECT_EQ(-1, map_.Deregister(kRtpExtensionTransmissionTimeOffset));
-}
-
-TEST_F(RtpHeaderExtensionTest, NonUniqueId) {
- EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
- EXPECT_EQ(-1, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
-}
-
-TEST_F(RtpHeaderExtensionTest, GetTotalLength) {
- EXPECT_EQ(0, map_.GetTotalLengthInBytes());
- EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
- EXPECT_EQ(RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES +
- TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES,
- map_.GetTotalLengthInBytes());
-}
-
-TEST_F(RtpHeaderExtensionTest, GetLengthUntilBlockStart) {
- EXPECT_EQ(-1, map_.GetLengthUntilBlockStartInBytes(
- kRtpExtensionTransmissionTimeOffset));
- EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
- EXPECT_EQ(RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES,
- map_.GetLengthUntilBlockStartInBytes(
- kRtpExtensionTransmissionTimeOffset));
-}
-
-TEST_F(RtpHeaderExtensionTest, GetType) {
- RTPExtensionType typeOut;
- EXPECT_EQ(-1, map_.GetType(kId, &typeOut));
-
- EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
- EXPECT_EQ(0, map_.GetType(kId, &typeOut));
- EXPECT_EQ(kRtpExtensionTransmissionTimeOffset, typeOut);
-}
-
-TEST_F(RtpHeaderExtensionTest, GetId) {
- uint8_t idOut;
- EXPECT_EQ(-1, map_.GetId(kRtpExtensionTransmissionTimeOffset, &idOut));
-
- EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
- EXPECT_EQ(0, map_.GetId(kRtpExtensionTransmissionTimeOffset, &idOut));
- EXPECT_EQ(kId, idOut);
-}
-
-TEST_F(RtpHeaderExtensionTest, IterateTypes) {
- EXPECT_EQ(kRtpExtensionNone, map_.First());
- EXPECT_EQ(kRtpExtensionNone, map_.Next(kRtpExtensionTransmissionTimeOffset));
-
- EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
-
- EXPECT_EQ(kRtpExtensionTransmissionTimeOffset, map_.First());
- EXPECT_EQ(kRtpExtensionNone, map_.Next(kRtpExtensionTransmissionTimeOffset));
-}
-
-TEST_F(RtpHeaderExtensionTest, GetCopy) {
- EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
-
- RtpHeaderExtensionMap mapOut;
- map_.GetCopy(&mapOut);
- EXPECT_EQ(1, mapOut.Size());
- EXPECT_EQ(kRtpExtensionTransmissionTimeOffset, mapOut.First());
-}
-
-TEST_F(RtpHeaderExtensionTest, Erase) {
- EXPECT_EQ(0, map_.Register(kRtpExtensionTransmissionTimeOffset, kId));
- EXPECT_EQ(1, map_.Size());
- map_.Erase();
- EXPECT_EQ(0, map_.Size());
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_packet_history.cc b/trunk/src/modules/rtp_rtcp/source/rtp_packet_history.cc
deleted file mode 100644
index 0de7553..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_packet_history.cc
+++ /dev/null
@@ -1,278 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Class for storing RTP packets.
- */
-
-#include "rtp_packet_history.h"
-
-#include <assert.h>
-#include <cstring> // memset
-
-#include "critical_section_wrapper.h"
-#include "rtp_utility.h"
-#include "trace.h"
-
-namespace webrtc {
-
-RTPPacketHistory::RTPPacketHistory(RtpRtcpClock* clock)
- : clock_(*clock),
- critsect_(CriticalSectionWrapper::CreateCriticalSection()),
- store_(false),
- prev_index_(0),
- max_packet_length_(0) {
-}
-
-RTPPacketHistory::~RTPPacketHistory() {
- Free();
- delete critsect_;
-}
-
-void RTPPacketHistory::SetStorePacketsStatus(bool enable,
- uint16_t number_to_store) {
- if (enable) {
- Allocate(number_to_store);
- } else {
- Free();
- }
-}
-
-void RTPPacketHistory::Allocate(uint16_t number_to_store) {
- assert(number_to_store > 0);
- webrtc::CriticalSectionScoped cs(*critsect_);
- if (store_) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "SetStorePacketsStatus already set, number: %d", number_to_store);
- return;
- }
-
- store_ = true;
- stored_packets_.resize(number_to_store);
- stored_seq_nums_.resize(number_to_store);
- stored_lengths_.resize(number_to_store);
- stored_times_.resize(number_to_store);
- stored_resend_times_.resize(number_to_store);
- stored_types_.resize(number_to_store);
-}
-
-void RTPPacketHistory::Free() {
- webrtc::CriticalSectionScoped cs(*critsect_);
- if (!store_) {
- return;
- }
-
- std::vector<std::vector<uint8_t> >::iterator it;
- for (it = stored_packets_.begin(); it != stored_packets_.end(); ++it) {
- it->clear();
- }
-
- stored_packets_.clear();
- stored_seq_nums_.clear();
- stored_lengths_.clear();
- stored_times_.clear();
- stored_resend_times_.clear();
- stored_types_.clear();
-
- store_ = false;
- prev_index_ = 0;
- max_packet_length_ = 0;
-}
-
-bool RTPPacketHistory::StorePackets() const {
- webrtc::CriticalSectionScoped cs(*critsect_);
- return store_;
-}
-
-// private, lock should already be taken
-void RTPPacketHistory::VerifyAndAllocatePacketLength(uint16_t packet_length) {
- assert(packet_length > 0);
- if (!store_) {
- return;
- }
-
- if (packet_length <= max_packet_length_) {
- return;
- }
-
- std::vector<std::vector<uint8_t> >::iterator it;
- for (it = stored_packets_.begin(); it != stored_packets_.end(); ++it) {
- it->resize(packet_length);
- }
- max_packet_length_ = packet_length;
-}
-
-int32_t RTPPacketHistory::PutRTPPacket(const uint8_t* packet,
- uint16_t packet_length,
- uint16_t max_packet_length,
- StorageType type) {
- if (type == kDontStore) {
- return 0;
- }
-
- webrtc::CriticalSectionScoped cs(*critsect_);
- if (!store_) {
- return 0;
- }
-
- assert(packet);
- assert(packet_length > 3);
-
- VerifyAndAllocatePacketLength(max_packet_length);
-
- if (packet_length > max_packet_length_) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, -1,
- "Failed to store RTP packet, length: %d", packet_length);
- return -1;
- }
-
- const uint16_t seq_num = (packet[2] << 8) + packet[3];
-
- // Store packet
- std::vector<std::vector<uint8_t> >::iterator it =
- stored_packets_.begin() + prev_index_;
- std::copy(packet, packet + packet_length, it->begin());
-
- stored_seq_nums_[prev_index_] = seq_num;
- stored_lengths_[prev_index_] = packet_length;
- stored_times_[prev_index_] = clock_.GetTimeInMS();
- stored_resend_times_[prev_index_] = 0; // packet not resent
- stored_types_[prev_index_] = type;
-
- ++prev_index_;
- if (prev_index_ >= stored_seq_nums_.size()) {
- prev_index_ = 0;
- }
- return 0;
-}
-
-bool RTPPacketHistory::HasRTPPacket(uint16_t sequence_number) const {
- webrtc::CriticalSectionScoped cs(*critsect_);
- if (!store_) {
- return false;
- }
-
- int32_t index = 0;
- bool found = FindSeqNum(sequence_number, &index);
- if (!found) {
- return false;
- }
-
- uint16_t length = stored_lengths_.at(index);
- if (length == 0 || length > max_packet_length_) {
- // Invalid length.
- return false;
- }
- return true;
-}
-
-bool RTPPacketHistory::GetRTPPacket(uint16_t sequence_number,
- uint32_t min_elapsed_time_ms,
- uint8_t* packet,
- uint16_t* packet_length,
- uint32_t* stored_time_ms,
- StorageType* type) const {
- webrtc::CriticalSectionScoped cs(*critsect_);
- if (!store_) {
- return false;
- }
-
- int32_t index = 0;
- bool found = FindSeqNum(sequence_number, &index);
- if (!found) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
- "No match for getting seqNum %u", sequence_number);
- return false;
- }
-
- uint16_t length = stored_lengths_.at(index);
- if (length == 0 || length > max_packet_length_) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
- "No match for getting seqNum %u, len %d", sequence_number, length);
- return false;
- }
-
- if (length > *packet_length) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "Input buffer too short for packet %u", sequence_number);
- return false;
- }
-
- // Verify elapsed time since last retrieve.
- uint32_t now = clock_.GetTimeInMS();
- if (min_elapsed_time_ms > 0 &&
- ((now - stored_resend_times_.at(index)) < min_elapsed_time_ms)) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
- "Skip getting packet %u, packet recently resent.", sequence_number);
- *packet_length = 0;
- return true;
- }
-
- // Get packet.
- std::vector<std::vector<uint8_t> >::const_iterator it_found_packet =
- stored_packets_.begin() + index;
- std::copy(it_found_packet->begin(), it_found_packet->begin() + length, packet);
- *packet_length = stored_lengths_.at(index);
- *stored_time_ms = stored_times_.at(index);
- *type = stored_types_.at(index);
- return true;
-}
-
-void RTPPacketHistory::UpdateResendTime(uint16_t sequence_number) {
- webrtc::CriticalSectionScoped cs(*critsect_);
- if (!store_) {
- return;
- }
-
- int32_t index = 0;
- bool found = FindSeqNum(sequence_number, &index);
- if (!found) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "Failed to update resend time, seq num: %u.", sequence_number);
- return;
- }
- stored_resend_times_[index] = clock_.GetTimeInMS();
-}
-
-// private, lock should already be taken
-bool RTPPacketHistory::FindSeqNum(uint16_t sequence_number,
- int32_t* index) const {
- uint16_t temp_sequence_number = 0;
- if (prev_index_ > 0) {
- *index = prev_index_ - 1;
- temp_sequence_number = stored_seq_nums_[*index];
- } else {
- *index = stored_seq_nums_.size() - 1;
- temp_sequence_number = stored_seq_nums_[*index]; // wrap
- }
-
- int32_t idx = (prev_index_ - 1) - (temp_sequence_number - sequence_number);
- if (idx >= 0 && idx < static_cast<int>(stored_seq_nums_.size())) {
- *index = idx;
- temp_sequence_number = stored_seq_nums_[*index];
- }
-
- if (temp_sequence_number != sequence_number) {
- // We did not found a match, search all.
- for (uint16_t m = 0; m < stored_seq_nums_.size(); m++) {
- if (stored_seq_nums_[m] == sequence_number) {
- *index = m;
- temp_sequence_number = stored_seq_nums_[*index];
- break;
- }
- }
- }
- if (temp_sequence_number == sequence_number) {
- // We found a match.
- return true;
- }
- return false;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_packet_history.h b/trunk/src/modules/rtp_rtcp/source/rtp_packet_history.h
deleted file mode 100644
index c61fd0b..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_packet_history.h
+++ /dev/null
@@ -1,87 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Class for storing RTP packets.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_
-#define WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_
-
-#include <vector>
-
-#include "module_common_types.h"
-#include "rtp_rtcp_defines.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-class RtpRtcpClock;
-class CriticalSectionWrapper;
-
-class RTPPacketHistory {
- public:
- RTPPacketHistory(RtpRtcpClock* clock);
- ~RTPPacketHistory();
-
- void SetStorePacketsStatus(bool enable, uint16_t number_to_store);
-
- bool StorePackets() const;
-
- // Stores RTP packet.
- int32_t PutRTPPacket(const uint8_t* packet,
- uint16_t packet_length,
- uint16_t max_packet_length,
- StorageType type);
-
- // Gets stored RTP packet corresponding to the input sequence number.
- // The packet is copied to the buffer pointed to by ptr_rtp_packet.
- // The rtp_packet_length should show the available buffer size.
- // Returns true if packet is found.
- // rtp_packet_length: returns the copied packet length on success.
- // min_elapsed_time_ms: the minimum time that must have elapsed since the last
- // time the packet was resent (parameter is ignored if set to zero).
- // If the packet is found but the minimum time has not elaped, no bytes are
- // copied.
- // stored_time_ms: returns the time when the packet was stored.
- // type: returns the storage type set in PutRTPPacket.
- bool GetRTPPacket(uint16_t sequence_number,
- uint32_t min_elapsed_time_ms,
- uint8_t* packet,
- uint16_t* packet_length,
- uint32_t* stored_time_ms,
- StorageType* type) const;
-
- bool HasRTPPacket(uint16_t sequence_number) const;
-
- void UpdateResendTime(uint16_t sequence_number);
-
- private:
- void Allocate(uint16_t number_to_store);
- void Free();
- void VerifyAndAllocatePacketLength(uint16_t packet_length);
- bool FindSeqNum(uint16_t sequence_number, int32_t* index) const;
-
- private:
- RtpRtcpClock& clock_;
- CriticalSectionWrapper* critsect_;
- bool store_;
- uint32_t prev_index_;
- uint16_t max_packet_length_;
-
- std::vector<std::vector<uint8_t> > stored_packets_;
- std::vector<uint16_t> stored_seq_nums_;
- std::vector<uint16_t> stored_lengths_;
- std::vector<uint32_t> stored_times_;
- std::vector<uint32_t> stored_resend_times_;
- std::vector<StorageType> stored_types_;
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_RTP_PACKET_HISTORY_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_packet_history_test.cc b/trunk/src/modules/rtp_rtcp/source/rtp_packet_history_test.cc
deleted file mode 100644
index c849bc2..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_packet_history_test.cc
+++ /dev/null
@@ -1,214 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file includes unit tests for the RTPPacketHistory.
- */
-
-#include <gtest/gtest.h>
-
-#include "rtp_packet_history.h"
-#include "rtp_rtcp_defines.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-class FakeClock : public RtpRtcpClock {
- public:
- FakeClock() {
- time_in_ms_ = 123456;
- }
- // Return a timestamp in milliseconds relative to some arbitrary
- // source; the source is fixed for this clock.
- virtual WebRtc_UWord32 GetTimeInMS() {
- return time_in_ms_;
- }
- // Retrieve an NTP absolute timestamp.
- virtual void CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac) {
- secs = time_in_ms_ / 1000;
- frac = (time_in_ms_ % 1000) * 4294967;
- }
- void IncrementTime(WebRtc_UWord32 time_increment_ms) {
- time_in_ms_ += time_increment_ms;
- }
- private:
- WebRtc_UWord32 time_in_ms_;
-};
-
-class RtpPacketHistoryTest : public ::testing::Test {
- protected:
- RtpPacketHistoryTest()
- : hist_(new RTPPacketHistory(&fake_clock_)) {
- }
- ~RtpPacketHistoryTest() {
- delete hist_;
- }
-
- FakeClock fake_clock_;
- RTPPacketHistory* hist_;
- enum {kPayload = 127};
- enum {kSsrc = 12345678};
- enum {kSeqNum = 88};
- enum {kTimestamp = 127};
- enum {kMaxPacketLength = 1500};
- uint8_t packet_[kMaxPacketLength];
- uint8_t packet_out_[kMaxPacketLength];
-
- void CreateRtpPacket(uint16_t seq_num, uint32_t ssrc, uint8_t payload,
- uint32_t timestamp, uint8_t* array, uint16_t* cur_pos) {
- array[(*cur_pos)++] = 0x80;
- array[(*cur_pos)++] = payload;
- array[(*cur_pos)++] = seq_num >> 8;
- array[(*cur_pos)++] = seq_num;
- array[(*cur_pos)++] = timestamp >> 24;
- array[(*cur_pos)++] = timestamp >> 16;
- array[(*cur_pos)++] = timestamp >> 8;
- array[(*cur_pos)++] = timestamp;
- array[(*cur_pos)++] = ssrc >> 24;
- array[(*cur_pos)++] = ssrc >> 16;
- array[(*cur_pos)++] = ssrc >> 8;
- array[(*cur_pos)++] = ssrc;
- }
-};
-
-TEST_F(RtpPacketHistoryTest, SetStoreStatus) {
- EXPECT_FALSE(hist_->StorePackets());
- hist_->SetStorePacketsStatus(true, 10);
- EXPECT_TRUE(hist_->StorePackets());
- hist_->SetStorePacketsStatus(false, 0);
- EXPECT_FALSE(hist_->StorePackets());
-}
-
-TEST_F(RtpPacketHistoryTest, NoStoreStatus) {
- EXPECT_FALSE(hist_->StorePackets());
- uint16_t len = 0;
- CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
- EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
- kAllowRetransmission));
- // Packet should not be stored.
- len = kMaxPacketLength;
- uint32_t time;
- StorageType type;
- EXPECT_FALSE(hist_->GetRTPPacket(kSeqNum, 0, packet_, &len, &time, &type));
-}
-
-TEST_F(RtpPacketHistoryTest, DontStore) {
- hist_->SetStorePacketsStatus(true, 10);
- uint16_t len = 0;
- CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
- EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength, kDontStore));
-
- // Packet should not be stored.
- len = kMaxPacketLength;
- uint32_t time;
- StorageType type;
- EXPECT_FALSE(hist_->GetRTPPacket(kSeqNum, 0, packet_, &len, &time, &type));
-}
-
-TEST_F(RtpPacketHistoryTest, PutRtpPacket_TooLargePacketLength) {
- hist_->SetStorePacketsStatus(true, 10);
- EXPECT_EQ(-1, hist_->PutRTPPacket(packet_,
- kMaxPacketLength + 1,
- kMaxPacketLength,
- kAllowRetransmission));
-}
-
-TEST_F(RtpPacketHistoryTest, GetRtpPacket_TooSmallBuffer) {
- hist_->SetStorePacketsStatus(true, 10);
- uint16_t len = 0;
- CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
- EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
- kAllowRetransmission));
- uint16_t len_out = len - 1;
- uint32_t time;
- StorageType type;
- EXPECT_FALSE(hist_->GetRTPPacket(kSeqNum, 0, packet_, &len_out, &time,
- &type));
-}
-
-TEST_F(RtpPacketHistoryTest, GetRtpPacket_NotStored) {
- hist_->SetStorePacketsStatus(true, 10);
- uint16_t len = kMaxPacketLength;
- uint32_t time;
- StorageType type;
- EXPECT_FALSE(hist_->GetRTPPacket(0, 0, packet_, &len, &time, &type));
-}
-
-TEST_F(RtpPacketHistoryTest, PutRtpPacket) {
- hist_->SetStorePacketsStatus(true, 10);
- uint16_t len = 0;
- CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
-
- EXPECT_FALSE(hist_->HasRTPPacket(kSeqNum));
- EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
- kAllowRetransmission));
- EXPECT_TRUE(hist_->HasRTPPacket(kSeqNum));
-}
-
-TEST_F(RtpPacketHistoryTest, GetRtpPacket) {
- hist_->SetStorePacketsStatus(true, 10);
- uint16_t len = 0;
- CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
- EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
- kAllowRetransmission));
-
- uint16_t len_out = kMaxPacketLength;
- uint32_t time;
- StorageType type;
- EXPECT_TRUE(hist_->GetRTPPacket(kSeqNum, 0, packet_out_, &len_out, &time,
- &type));
- EXPECT_EQ(len, len_out);
- EXPECT_EQ(kAllowRetransmission, type);
- for (int i = 0; i < len; i++) {
- EXPECT_EQ(packet_[i], packet_out_[i]);
- }
-}
-
-TEST_F(RtpPacketHistoryTest, DontRetransmit) {
- hist_->SetStorePacketsStatus(true, 10);
- uint16_t len = 0;
- CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
- EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
- kDontRetransmit));
-
- uint16_t len_out = kMaxPacketLength;
- uint32_t time;
- StorageType type;
- EXPECT_TRUE(hist_->GetRTPPacket(kSeqNum, 0, packet_out_, &len_out, &time,
- &type));
- EXPECT_EQ(len, len_out);
- EXPECT_EQ(kDontRetransmit, type);
-}
-
-TEST_F(RtpPacketHistoryTest, MinResendTime) {
- hist_->SetStorePacketsStatus(true, 10);
- WebRtc_UWord32 store_time = fake_clock_.GetTimeInMS();
- uint16_t len = 0;
- CreateRtpPacket(kSeqNum, kSsrc, kPayload, kTimestamp, packet_, &len);
- EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, kMaxPacketLength,
- kAllowRetransmission));
-
- hist_->UpdateResendTime(kSeqNum);
- fake_clock_.IncrementTime(100);
-
- // Time has elapsed.
- len = kMaxPacketLength;
- StorageType type;
- uint32_t time;
- EXPECT_TRUE(hist_->GetRTPPacket(kSeqNum, 100, packet_, &len, &time, &type));
- EXPECT_GT(len, 0);
- EXPECT_EQ(store_time, time);
-
- // Time has not elapsed. Packet should be found, but no bytes copied.
- len = kMaxPacketLength;
- EXPECT_TRUE(hist_->GetRTPPacket(kSeqNum, 101, packet_, &len, &time, &type));
- EXPECT_EQ(0, len);
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_receiver.cc b/trunk/src/modules/rtp_rtcp/source/rtp_receiver.cc
deleted file mode 100644
index f421966..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_receiver.cc
+++ /dev/null
@@ -1,1685 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "trace.h"
-#include "rtp_receiver.h"
-
-#include "rtp_rtcp_defines.h"
-#include "rtp_rtcp_impl.h"
-#include "critical_section_wrapper.h"
-
-#include <cassert>
-#include <string.h> //memcpy
-#include <math.h> // floor
-#include <stdlib.h> // abs
-
-namespace webrtc {
-
-using ModuleRTPUtility::AudioPayload;
-using ModuleRTPUtility::GetCurrentRTP;
-using ModuleRTPUtility::Payload;
-using ModuleRTPUtility::RTPPayloadParser;
-using ModuleRTPUtility::StringCompare;
-using ModuleRTPUtility::VideoPayload;
-
-RTPReceiver::RTPReceiver(const WebRtc_Word32 id,
- const bool audio,
- RtpRtcpClock* clock,
- ModuleRtpRtcpImpl* owner) :
- RTPReceiverAudio(id),
- RTPReceiverVideo(id, owner),
- Bitrate(clock),
- _id(id),
- _audio(audio),
- _rtpRtcp(*owner),
- _criticalSectionCbs(CriticalSectionWrapper::CreateCriticalSection()),
- _cbRtpFeedback(NULL),
- _cbRtpData(NULL),
-
- _criticalSectionRTPReceiver(
- CriticalSectionWrapper::CreateCriticalSection()),
- _lastReceiveTime(0),
- _lastReceivedPayloadLength(0),
- _lastReceivedPayloadType(-1),
- _lastReceivedMediaPayloadType(-1),
- _lastReceivedAudioSpecific(),
- _lastReceivedVideoSpecific(),
-
- _packetTimeOutMS(0),
-
- _redPayloadType(-1),
- _payloadTypeMap(),
- _rtpHeaderExtensionMap(),
- _SSRC(0),
- _numCSRCs(0),
- _currentRemoteCSRC(),
- _numEnergy(0),
- _currentRemoteEnergy(),
- _useSSRCFilter(false),
- _SSRCFilter(0),
-
- _jitterQ4(0),
- _jitterMaxQ4(0),
- _cumulativeLoss(0),
- _jitterQ4TransmissionTimeOffset(0),
- _localTimeLastReceivedTimestamp(0),
- _lastReceivedTimestamp(0),
- _lastReceivedSequenceNumber(0),
- _lastReceivedTransmissionTimeOffset(0),
-
- _receivedSeqFirst(0),
- _receivedSeqMax(0),
- _receivedSeqWraps(0),
-
- _receivedPacketOH(12), // RTP header
- _receivedByteCount(0),
- _receivedOldPacketCount(0),
- _receivedInorderPacketCount(0),
-
- _lastReportInorderPackets(0),
- _lastReportOldPackets(0),
- _lastReportSeqMax(0),
- _lastReportFractionLost(0),
- _lastReportCumulativeLost(0),
- _lastReportExtendedHighSeqNum(0),
- _lastReportJitter(0),
- _lastReportJitterTransmissionTimeOffset(0),
-
- _nackMethod(kNackOff),
- _RTX(false),
- _ssrcRTX(0) {
- memset(_currentRemoteCSRC, 0, sizeof(_currentRemoteCSRC));
- memset(_currentRemoteEnergy, 0, sizeof(_currentRemoteEnergy));
- memset(&_lastReceivedAudioSpecific, 0, sizeof(_lastReceivedAudioSpecific));
-
- _lastReceivedAudioSpecific.channels = 1;
- _lastReceivedVideoSpecific.maxRate = 0;
- _lastReceivedVideoSpecific.videoCodecType = kRtpNoVideo;
-
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
-}
-
-RTPReceiver::~RTPReceiver() {
- if (_cbRtpFeedback) {
- for (int i = 0; i < _numCSRCs; i++) {
- _cbRtpFeedback->OnIncomingCSRCChanged(_id,_currentRemoteCSRC[i], false);
- }
- }
- delete _criticalSectionCbs;
- delete _criticalSectionRTPReceiver;
-
- while (!_payloadTypeMap.empty()) {
- std::map<WebRtc_Word8, Payload*>::iterator it = _payloadTypeMap.begin();
- delete it->second;
- _payloadTypeMap.erase(it);
- }
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__);
-}
-
-WebRtc_Word32 RTPReceiver::Init() {
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
-
- _lastReceiveTime = 0;
- _lastReceivedPayloadLength = 0;
- _packetTimeOutMS = 0;
- _lastReceivedPayloadType = -1;
- _lastReceivedMediaPayloadType = -1;
- _redPayloadType = -1;
-
- memset(&_lastReceivedAudioSpecific, 0, sizeof(_lastReceivedAudioSpecific));
- _lastReceivedAudioSpecific.channels = 1;
-
- _lastReceivedVideoSpecific.videoCodecType = kRtpNoVideo;
- _lastReceivedVideoSpecific.maxRate = 0;
- _SSRC = 0;
- _numCSRCs = 0;
- _numEnergy = 0;
- _jitterQ4 = 0;
- _jitterMaxQ4 = 0;
- _cumulativeLoss = 0;
- _jitterQ4TransmissionTimeOffset = 0;
- _useSSRCFilter = false;
- _SSRCFilter = 0;
-
- _localTimeLastReceivedTimestamp = 0;
- _lastReceivedTimestamp = 0;
- _lastReceivedSequenceNumber = 0;
- _lastReceivedTransmissionTimeOffset = 0;
-
- _receivedSeqFirst = 0;
- _receivedSeqMax = 0;
- _receivedSeqWraps = 0;
-
- _receivedPacketOH = 12; // RTP header
- _receivedByteCount = 0;
- _receivedOldPacketCount = 0;
- _receivedInorderPacketCount = 0;
-
- _lastReportInorderPackets = 0;
- _lastReportOldPackets = 0;
- _lastReportSeqMax = 0;
- _lastReportFractionLost = 0;
- _lastReportCumulativeLost = 0;
- _lastReportExtendedHighSeqNum = 0;
- _lastReportJitter = 0;
- _lastReportJitterTransmissionTimeOffset = 0;
-
- _rtpHeaderExtensionMap.Erase();
-
- while (!_payloadTypeMap.empty()) {
- std::map<WebRtc_Word8, Payload*>::iterator it = _payloadTypeMap.begin();
- delete it->second;
- _payloadTypeMap.erase(it);
- }
-
- Bitrate::Init();
- RTPReceiverAudio::Init();
- return RTPReceiverVideo::Init();
-}
-
-void
-RTPReceiver::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
- RTPReceiverAudio::ChangeUniqueId(id);
- RTPReceiverVideo::ChangeUniqueId(id);
-}
-
-RtpVideoCodecTypes
-RTPReceiver::VideoCodecType() const
-{
- return _lastReceivedVideoSpecific.videoCodecType;
-}
-
-WebRtc_UWord32
-RTPReceiver::MaxConfiguredBitrate() const
-{
- return _lastReceivedVideoSpecific.maxRate;
-}
-
-bool
-RTPReceiver::REDPayloadType(const WebRtc_Word8 payloadType) const
-{
- return (_redPayloadType == payloadType)?true:false;
-}
-
-WebRtc_Word8
-RTPReceiver::REDPayloadType() const
-{
- return _redPayloadType;
-}
-
- // configure a timeout value
-WebRtc_Word32
-RTPReceiver::SetPacketTimeout(const WebRtc_UWord32 timeoutMS)
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
- _packetTimeOutMS = timeoutMS;
- return 0;
-}
-
-void RTPReceiver::PacketTimeout()
-{
- bool packetTimeOut = false;
- {
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
- if(_packetTimeOutMS == 0)
- {
- // not configured
- return;
- }
-
- if(_lastReceiveTime == 0)
- {
- // not active
- return;
- }
-
- WebRtc_UWord32 now = _clock.GetTimeInMS();
-
- if(now - _lastReceiveTime > _packetTimeOutMS)
- {
- packetTimeOut = true;
- _lastReceiveTime = 0; // only one callback
- _lastReceivedPayloadType = -1; // makes RemotePayload return -1, which we want
- _lastReceivedMediaPayloadType = -1;
- }
- }
- CriticalSectionScoped lock(_criticalSectionCbs);
- if(packetTimeOut && _cbRtpFeedback)
- {
- _cbRtpFeedback->OnPacketTimeout(_id);
- }
-}
-
-void
-RTPReceiver::ProcessDeadOrAlive(const bool RTCPalive, const WebRtc_UWord32 now)
-{
- if(_cbRtpFeedback == NULL)
- {
- // no callback
- return;
- }
- RTPAliveType alive = kRtpDead;
-
- if(_lastReceiveTime + 1000 > now)
- {
- // always alive if we have received a RTP packet the last sec
- alive = kRtpAlive;
-
- } else
- {
- if(RTCPalive)
- {
- if(_audio)
- {
- // alive depends on CNG
- // if last received size < 10 likely CNG
- if(_lastReceivedPayloadLength < 10) // our CNG is 9 bytes
- {
- // potential CNG
- // receiver need to check kRtpNoRtp against NetEq speechType kOutputPLCtoCNG
- alive = kRtpNoRtp;
- } else
- {
- // dead
- }
- } else
- {
- // dead for video
- }
- }else
- {
- // no RTP packet for 1 sec and no RTCP
- // dead
- }
- }
-
-
- CriticalSectionScoped lock(_criticalSectionCbs);
- if(_cbRtpFeedback)
- {
- _cbRtpFeedback->OnPeriodicDeadOrAlive(_id, alive);
- }
-}
-
-WebRtc_UWord16
-RTPReceiver::PacketOHReceived() const
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
- return _receivedPacketOH;
-}
-
-WebRtc_UWord32
-RTPReceiver::PacketCountReceived() const
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
- return _receivedInorderPacketCount;
-}
-
-WebRtc_UWord32
-RTPReceiver::ByteCountReceived() const
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
- return _receivedByteCount;
-}
-
-WebRtc_Word32
-RTPReceiver::RegisterIncomingRTPCallback(RtpFeedback* incomingMessagesCallback)
-{
- CriticalSectionScoped lock(_criticalSectionCbs);
- _cbRtpFeedback = incomingMessagesCallback;
- return 0;
-}
-
-WebRtc_Word32
-RTPReceiver::RegisterIncomingDataCallback(RtpData* incomingDataCallback)
-{
- CriticalSectionScoped lock(_criticalSectionCbs);
- _cbRtpData = incomingDataCallback;
- return 0;
-}
-
-WebRtc_Word32 RTPReceiver::RegisterReceivePayload(
- const char payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate) {
- assert(payloadName);
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
-
- // sanity
- switch (payloadType) {
- // reserved payload types to avoid RTCP conflicts when marker bit is set
- case 64: // 192 Full INTRA-frame request
- case 72: // 200 Sender report
- case 73: // 201 Receiver report
- case 74: // 202 Source description
- case 75: // 203 Goodbye
- case 76: // 204 Application-defined
- case 77: // 205 Transport layer FB message
- case 78: // 206 Payload-specific FB message
- case 79: // 207 Extended report
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s invalid payloadtype:%d",
- __FUNCTION__, payloadType);
- return -1;
- default:
- break;
- }
- size_t payloadNameLength = strlen(payloadName);
-
- std::map<WebRtc_Word8, Payload*>::iterator it =
- _payloadTypeMap.find(payloadType);
- if (it != _payloadTypeMap.end()) {
- // we already use this payload type
- Payload* payload = it->second;
- assert(payload);
-
- size_t nameLength = strlen(payload->name);
-
- // check if it's the same as we already have
- // if same ignore sending an error
- if (payloadNameLength == nameLength &&
- StringCompare(payload->name, payloadName, payloadNameLength)) {
- if (_audio &&
- payload->audio &&
- payload->typeSpecific.Audio.frequency == frequency &&
- payload->typeSpecific.Audio.channels == channels &&
- (payload->typeSpecific.Audio.rate == rate ||
- payload->typeSpecific.Audio.rate == 0 || rate == 0)) {
- payload->typeSpecific.Audio.rate = rate;
- // Ensure that we update the rate if new or old is zero
- return 0;
- }
- if (!_audio && !payload->audio) {
- // update maxBitrate for video
- payload->typeSpecific.Video.maxRate = rate;
- return 0;
- }
- }
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s invalid argument payloadType:%d already registered",
- __FUNCTION__, payloadType);
- return -1;
- }
- if (_audio) {
- // remove existing item, hence search for the name
- // only for audio, for video we allow a codecs to use multiple pltypes
- std::map<WebRtc_Word8, Payload*>::iterator audio_it =
- _payloadTypeMap.begin();
- while (audio_it != _payloadTypeMap.end()) {
- Payload* payload = audio_it->second;
- size_t nameLength = strlen(payload->name);
-
- if (payloadNameLength == nameLength &&
- StringCompare(payload->name, payloadName, payloadNameLength)) {
- // we found the payload name in the list
- // if audio check frequency and rate
- if (payload->audio) {
- if (payload->typeSpecific.Audio.frequency == frequency &&
- (payload->typeSpecific.Audio.rate == rate ||
- payload->typeSpecific.Audio.rate == 0 || rate == 0)) {
- // remove old setting
- delete payload;
- _payloadTypeMap.erase(audio_it);
- break;
- }
- } else if(StringCompare(payloadName,"red",3)) {
- delete payload;
- _payloadTypeMap.erase(audio_it);
- break;
- }
- }
- audio_it++;
- }
- }
- Payload* payload = NULL;
-
- // save the RED payload type
- // used in both audio and video
- if (StringCompare(payloadName,"red",3)) {
- _redPayloadType = payloadType;
- payload = new Payload;
- payload->audio = false;
- payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
- strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
- } else {
- if (_audio) {
- payload = RegisterReceiveAudioPayload(payloadName, payloadType,
- frequency, channels, rate);
- } else {
- payload = RegisterReceiveVideoPayload(payloadName, payloadType, rate);
- }
- }
- if (payload == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s filed to register payload",
- __FUNCTION__);
- return -1;
- }
- _payloadTypeMap[payloadType] = payload;
-
- // Successful set of payload type, clear the value of last receivedPT,
- // since it might mean something else
- _lastReceivedPayloadType = -1;
- _lastReceivedMediaPayloadType = -1;
- return 0;
-}
-
-WebRtc_Word32 RTPReceiver::DeRegisterReceivePayload(
- const WebRtc_Word8 payloadType) {
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
-
- std::map<WebRtc_Word8, Payload*>::iterator it =
- _payloadTypeMap.find(payloadType);
-
- if (it == _payloadTypeMap.end()) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s failed to find payloadType:%d",
- __FUNCTION__, payloadType);
- return -1;
- }
- delete it->second;
- _payloadTypeMap.erase(it);
- return 0;
-}
-
-WebRtc_Word32 RTPReceiver::ReceivePayloadType(
- const char payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_UWord32 frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate,
- WebRtc_Word8* payloadType) const {
- if (payloadType == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s invalid argument", __FUNCTION__);
- return -1;
- }
- size_t payloadNameLength = strlen(payloadName);
-
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
-
- std::map<WebRtc_Word8, Payload*>::const_iterator it =
- _payloadTypeMap.begin();
-
- while (it != _payloadTypeMap.end()) {
- Payload* payload = it->second;
- assert(payload);
-
- size_t nameLength = strlen(payload->name);
- if (payloadNameLength == nameLength &&
- StringCompare(payload->name, payloadName, payloadNameLength)) {
- // name match
- if( payload->audio) {
- if (rate == 0) {
- // [default] audio, check freq and channels
- if (payload->typeSpecific.Audio.frequency == frequency &&
- payload->typeSpecific.Audio.channels == channels) {
- *payloadType = it->first;
- return 0;
- }
- } else {
- // audio, check freq, channels and rate
- if( payload->typeSpecific.Audio.frequency == frequency &&
- payload->typeSpecific.Audio.channels == channels &&
- payload->typeSpecific.Audio.rate == rate) {
- // extra rate condition added
- *payloadType = it->first;
- return 0;
- }
- }
- } else {
- // video
- *payloadType = it->first;
- return 0;
- }
- }
- it++;
- }
- return -1;
-}
-
-WebRtc_Word32 RTPReceiver::ReceivePayload(
- const WebRtc_Word8 payloadType,
- char payloadName[RTP_PAYLOAD_NAME_SIZE],
- WebRtc_UWord32* frequency,
- WebRtc_UWord8* channels,
- WebRtc_UWord32* rate) const {
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
-
- std::map<WebRtc_Word8, Payload*>::const_iterator it =
- _payloadTypeMap.find(payloadType);
-
- if (it == _payloadTypeMap.end()) {
- return -1;
- }
- Payload* payload = it->second;
- assert(payload);
-
- if(frequency) {
- if(payload->audio) {
- *frequency = payload->typeSpecific.Audio.frequency;
- } else {
- *frequency = 90000;
- }
- }
- if (channels) {
- if(payload->audio) {
- *channels = payload->typeSpecific.Audio.channels;
- } else {
- *channels = 1;
- }
- }
- if (rate) {
- if(payload->audio) {
- *rate = payload->typeSpecific.Audio.rate;
- } else {
- assert(false);
- *rate = 0;
- }
- }
- if (payloadName) {
- payloadName[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
- strncpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
- }
- return 0;
-}
-
-WebRtc_Word32 RTPReceiver::RemotePayload(
- char payloadName[RTP_PAYLOAD_NAME_SIZE],
- WebRtc_Word8* payloadType,
- WebRtc_UWord32* frequency,
- WebRtc_UWord8* channels) const {
- if(_lastReceivedPayloadType == -1) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
- "%s invalid state", __FUNCTION__);
- return -1;
- }
- std::map<WebRtc_Word8, Payload*>::const_iterator it =
- _payloadTypeMap.find(_lastReceivedPayloadType);
-
- if (it == _payloadTypeMap.end()) {
- return -1;
- }
- Payload* payload = it->second;
- assert(payload);
- payloadName[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
- strncpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
-
- if (payloadType) {
- *payloadType = _lastReceivedPayloadType;
- }
- if (frequency) {
- if (payload->audio) {
- *frequency = payload->typeSpecific.Audio.frequency;
- } else {
- *frequency = 90000;
- }
- }
- if (channels) {
- if (payload->audio) {
- *channels = payload->typeSpecific.Audio.channels;
- } else {
- *channels = 1;
- }
- }
- return 0;
-}
-
-WebRtc_Word32
-RTPReceiver::RegisterRtpHeaderExtension(const RTPExtensionType type,
- const WebRtc_UWord8 id)
-{
- CriticalSectionScoped cs(_criticalSectionRTPReceiver);
- return _rtpHeaderExtensionMap.Register(type, id);
-}
-
-WebRtc_Word32
-RTPReceiver::DeregisterRtpHeaderExtension(const RTPExtensionType type)
-{
- CriticalSectionScoped cs(_criticalSectionRTPReceiver);
- return _rtpHeaderExtensionMap.Deregister(type);
-}
-
-void RTPReceiver::GetHeaderExtensionMapCopy(RtpHeaderExtensionMap* map) const
-{
- CriticalSectionScoped cs(_criticalSectionRTPReceiver);
- _rtpHeaderExtensionMap.GetCopy(map);
-}
-
-NACKMethod
-RTPReceiver::NACK() const
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
- return _nackMethod;
-}
-
- // Turn negative acknowledgement requests on/off
-WebRtc_Word32
-RTPReceiver::SetNACKStatus(const NACKMethod method)
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
- _nackMethod = method;
- return 0;
-}
-
-void RTPReceiver::SetRTXStatus(const bool enable,
- const WebRtc_UWord32 SSRC) {
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
- _RTX = enable;
- _ssrcRTX = SSRC;
-}
-
-void RTPReceiver::RTXStatus(bool* enable, WebRtc_UWord32* SSRC) const {
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
- *enable = _RTX;
- *SSRC = _ssrcRTX;
-}
-
-WebRtc_UWord32
-RTPReceiver::SSRC() const
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
- return _SSRC;
-}
-
- // Get remote CSRC
-WebRtc_Word32
-RTPReceiver::CSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
-
- assert(_numCSRCs <= kRtpCsrcSize);
-
- if(_numCSRCs >0)
- {
- memcpy(arrOfCSRC, _currentRemoteCSRC, sizeof(WebRtc_UWord32)*_numCSRCs);
- }
- return _numCSRCs;
-}
-
-WebRtc_Word32
-RTPReceiver::Energy( WebRtc_UWord8 arrOfEnergy[kRtpCsrcSize]) const
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
-
- assert(_numEnergy <= kRtpCsrcSize);
-
- if(_numEnergy >0)
- {
- memcpy(arrOfEnergy, _currentRemoteEnergy, sizeof(WebRtc_UWord8)*_numCSRCs);
- }
- return _numEnergy;
-}
-
-WebRtc_Word32 RTPReceiver::IncomingRTPPacket(
- WebRtcRTPHeader* rtp_header,
- const WebRtc_UWord8* packet,
- const WebRtc_UWord16 packet_length) {
- // rtp_header contains the parsed RTP header.
- // Adjust packet length w r t RTP padding.
- int length = packet_length - rtp_header->header.paddingLength;
-
- // length sanity
- if ((length - rtp_header->header.headerLength) < 0) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "%s invalid argument",
- __FUNCTION__);
- return -1;
- }
- if (_RTX) {
- if (_ssrcRTX == rtp_header->header.ssrc) {
- // Sanity check.
- if (rtp_header->header.headerLength + 2 > packet_length) {
- return -1;
- }
- rtp_header->header.ssrc = _SSRC;
- rtp_header->header.sequenceNumber =
- (packet[rtp_header->header.headerLength] << 8) +
- packet[1 + rtp_header->header.headerLength];
- // Count the RTX header as part of the RTP header.
- rtp_header->header.headerLength += 2;
- }
- }
- if (_useSSRCFilter) {
- if (rtp_header->header.ssrc != _SSRCFilter) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
- "%s drop packet due to SSRC filter",
- __FUNCTION__);
- return -1;
- }
- }
- if (_lastReceiveTime == 0) {
- // trigger only once
- CriticalSectionScoped lock(_criticalSectionCbs);
- if (_cbRtpFeedback) {
- if (length - rtp_header->header.headerLength == 0) {
- // keepalive packet
- _cbRtpFeedback->OnReceivedPacket(_id, kPacketKeepAlive);
- } else {
- _cbRtpFeedback->OnReceivedPacket(_id, kPacketRtp);
- }
- }
- }
- WebRtc_Word8 first_payload_byte = 0;
- if (length > 0) {
- first_payload_byte = packet[rtp_header->header.headerLength];
- }
- // trigger our callbacks
- CheckSSRCChanged(rtp_header);
-
- bool is_red = false;
- VideoPayload video_specific;
- video_specific.maxRate = 0;
- video_specific.videoCodecType = kRtpNoVideo;
-
- AudioPayload audio_specific;
- audio_specific.bitsPerSample = 0;
- audio_specific.channels = 0;
- audio_specific.frequency = 0;
-
- if (CheckPayloadChanged(rtp_header,
- first_payload_byte,
- is_red,
- audio_specific,
- video_specific) == -1) {
- if (length - rtp_header->header.headerLength == 0)
- {
- // ok keepalive packet
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
- "%s received keepalive",
- __FUNCTION__);
- return 0;
- }
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
- "%s received invalid payloadtype",
- __FUNCTION__);
- return -1;
- }
- CheckCSRC(rtp_header);
-
- const WebRtc_UWord8* payload_data =
- packet + rtp_header->header.headerLength;
-
- WebRtc_UWord16 payload_data_length =
- static_cast<WebRtc_UWord16>(length - rtp_header->header.headerLength);
-
- WebRtc_Word32 retVal = 0;
- if(_audio) {
- retVal = ParseAudioCodecSpecific(rtp_header,
- payload_data,
- payload_data_length,
- audio_specific,
- is_red);
- } else {
- retVal = ParseVideoCodecSpecific(rtp_header,
- payload_data,
- payload_data_length,
- video_specific.videoCodecType,
- is_red,
- packet,
- packet_length,
- _clock.GetTimeInMS());
- }
- if(retVal < 0) {
- return retVal;
- }
-
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
-
- // this compare to _receivedSeqMax
- // we store the last received after we have done the callback
- bool old_packet = RetransmitOfOldPacket(rtp_header->header.sequenceNumber,
- rtp_header->header.timestamp);
-
- // this updates _receivedSeqMax and other members
- UpdateStatistics(rtp_header, payload_data_length, old_packet);
-
- // Need to be updated after RetransmitOfOldPacket &
- // RetransmitOfOldPacketUpdateStatistics
- _lastReceiveTime = _clock.GetTimeInMS();
- _lastReceivedPayloadLength = payload_data_length;
-
- if (!old_packet) {
- if (_lastReceivedTimestamp != rtp_header->header.timestamp) {
- _lastReceivedTimestamp = rtp_header->header.timestamp;
- }
- _lastReceivedSequenceNumber = rtp_header->header.sequenceNumber;
- _lastReceivedTransmissionTimeOffset =
- rtp_header->extension.transmissionTimeOffset;
- }
- return retVal;
-}
-
-// must not have critsect when called
-WebRtc_Word32
-RTPReceiver::CallbackOfReceivedPayloadData(const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const WebRtcRTPHeader* rtpHeader)
-{
- CriticalSectionScoped lock(_criticalSectionCbs);
- if(_cbRtpData)
- {
- return _cbRtpData->OnReceivedPayloadData(payloadData, payloadSize, rtpHeader);
- }
- return -1;
-}
-
-// we already have the _criticalSectionRTPReceiver critsect when we call this
-void
-RTPReceiver::UpdateStatistics(const WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord16 bytes,
- const bool oldPacket)
-{
- WebRtc_UWord32 freq = 90000;
- if(_audio)
- {
- freq = AudioFrequency();
- }
-
- Bitrate::Update(bytes);
-
- _receivedByteCount += bytes;
-
- if (_receivedSeqMax == 0 && _receivedSeqWraps == 0)
- {
- // First received report
- _receivedSeqFirst = rtpHeader->header.sequenceNumber;
- _receivedSeqMax = rtpHeader->header.sequenceNumber;
- _receivedInorderPacketCount = 1;
- _localTimeLastReceivedTimestamp =
- GetCurrentRTP(&_clock, freq); //time in samples
- return;
- }
-
- // count only the new packets received
- if(InOrderPacket(rtpHeader->header.sequenceNumber))
- {
- const WebRtc_UWord32 RTPtime =
- GetCurrentRTP(&_clock, freq); //time in samples
- _receivedInorderPacketCount++;
-
- // wrong if we use RetransmitOfOldPacket
- WebRtc_Word32 seqDiff = rtpHeader->header.sequenceNumber - _receivedSeqMax;
- if (seqDiff < 0)
- {
- // Wrap around detected
- _receivedSeqWraps++;
- }
- // new max
- _receivedSeqMax = rtpHeader->header.sequenceNumber;
-
- if (rtpHeader->header.timestamp != _lastReceivedTimestamp &&
- _receivedInorderPacketCount > 1)
- {
- WebRtc_Word32 timeDiffSamples = (RTPtime - _localTimeLastReceivedTimestamp) -
- (rtpHeader->header.timestamp - _lastReceivedTimestamp);
-
- timeDiffSamples = abs(timeDiffSamples);
-
- // libJingle sometimes deliver crazy jumps in TS for the same stream
- // If this happen don't update jitter value
- if(timeDiffSamples < 450000) // Use 5 secs video frequency as border
- {
- // note we calculate in Q4 to avoid using float
- WebRtc_Word32 jitterDiffQ4 = (timeDiffSamples << 4) - _jitterQ4;
- _jitterQ4 += ((jitterDiffQ4 + 8) >> 4);
- }
-
- // Extended jitter report, RFC 5450.
- // Actual network jitter, excluding the source-introduced jitter.
- WebRtc_Word32 timeDiffSamplesExt =
- (RTPtime - _localTimeLastReceivedTimestamp) -
- ((rtpHeader->header.timestamp +
- rtpHeader->extension.transmissionTimeOffset) -
- (_lastReceivedTimestamp +
- _lastReceivedTransmissionTimeOffset));
-
- timeDiffSamplesExt = abs(timeDiffSamplesExt);
-
- if(timeDiffSamplesExt < 450000) // Use 5 secs video freq as border
- {
- // note we calculate in Q4 to avoid using float
- WebRtc_Word32 jitterDiffQ4TransmissionTimeOffset =
- (timeDiffSamplesExt << 4) - _jitterQ4TransmissionTimeOffset;
- _jitterQ4TransmissionTimeOffset +=
- ((jitterDiffQ4TransmissionTimeOffset + 8) >> 4);
- }
- }
- _localTimeLastReceivedTimestamp = RTPtime;
- } else
- {
- if(oldPacket)
- {
- _receivedOldPacketCount++;
- }else
- {
- _receivedInorderPacketCount++;
- }
- }
-
- WebRtc_UWord16 packetOH = rtpHeader->header.headerLength + rtpHeader->header.paddingLength;
-
- // our measured overhead
- // filter from RFC 5104 4.2.1.2
- // avg_OH (new) = 15/16*avg_OH (old) + 1/16*pckt_OH,
- _receivedPacketOH = (15*_receivedPacketOH + packetOH) >> 4;
-}
-
-// we already have the _criticalSectionRTPReceiver critsect when we call this
-bool RTPReceiver::RetransmitOfOldPacket(
- const WebRtc_UWord16 sequenceNumber,
- const WebRtc_UWord32 rtpTimeStamp) const {
- if (InOrderPacket(sequenceNumber)) {
- return false;
- }
- WebRtc_UWord32 frequencyKHz = 90; // Video frequency.
- if (_audio) {
- frequencyKHz = AudioFrequency() / 1000;
- }
- WebRtc_UWord32 timeDiffMS = _clock.GetTimeInMS() - _lastReceiveTime;
- // Diff in time stamp since last received in order.
- WebRtc_Word32 rtpTimeStampDiffMS = static_cast<WebRtc_Word32>(
- rtpTimeStamp - _lastReceivedTimestamp) / frequencyKHz;
-
- WebRtc_UWord16 minRTT = 0;
- WebRtc_Word32 maxDelayMs = 0;
- _rtpRtcp.RTT(_SSRC, NULL, NULL, &minRTT, NULL);
- if (minRTT == 0) {
- float jitter = _jitterQ4 >> 4; // Jitter variance in samples.
- // Jitter standard deviation in samples.
- float jitterStd = sqrt(jitter);
- // 2 times the std deviation => 95% confidence.
- // And transform to ms by dividing by the frequency in kHz.
- maxDelayMs = static_cast<WebRtc_Word32>((2 * jitterStd) / frequencyKHz);
-
- // Min maxDelayMs is 1.
- if (maxDelayMs == 0) {
- maxDelayMs = 1;
- }
- } else {
- maxDelayMs = (minRTT / 3) + 1;
- }
- if (static_cast<WebRtc_Word32>(timeDiffMS) >
- rtpTimeStampDiffMS + maxDelayMs) {
- return true;
- }
- return false;
-}
-
-bool
-RTPReceiver::InOrderPacket(const WebRtc_UWord16 sequenceNumber) const
-{
- if(_receivedSeqMax >= sequenceNumber)
- {
- if(!(_receivedSeqMax > 0xff00 && sequenceNumber < 0x0ff ))//detect wrap around
- {
- if(_receivedSeqMax - NACK_PACKETS_MAX_SIZE > sequenceNumber)
- {
- // we have a restart of the remote side
- }else
- {
- // we received a retransmit of a packet we already have
- return false;
- }
- }
- }else
- {
- // check for a wrap
- if(sequenceNumber > 0xff00 && _receivedSeqMax < 0x0ff )//detect wrap around
- {
- if(_receivedSeqMax - NACK_PACKETS_MAX_SIZE > sequenceNumber)
- {
- // we have a restart of the remote side
- }else
- {
- // we received a retransmit of a packet we already have
- return false;
- }
- }
- }
- return true;
-}
-
-WebRtc_UWord16
-RTPReceiver::SequenceNumber() const
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
- return _lastReceivedSequenceNumber;
-}
-
-WebRtc_UWord32
-RTPReceiver::TimeStamp() const
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
- return _lastReceivedTimestamp;
-}
-
-WebRtc_UWord32 RTPReceiver::PayloadTypeToPayload(
- const WebRtc_UWord8 payloadType,
- Payload*& payload) const {
-
- std::map<WebRtc_Word8, Payload*>::const_iterator it =
- _payloadTypeMap.find(payloadType);
-
- // check that this is a registered payload type
- if (it == _payloadTypeMap.end()) {
- return -1;
- }
- payload = it->second;
- return 0;
-}
-
-// timeStamp of the last incoming packet that is the first packet of its frame
-WebRtc_Word32
-RTPReceiver::EstimatedRemoteTimeStamp(WebRtc_UWord32& timestamp) const
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
- WebRtc_UWord32 freq = 90000;
- if(_audio)
- {
- freq = AudioFrequency();
- }
- if(_localTimeLastReceivedTimestamp == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s invalid state", __FUNCTION__);
- return -1;
- }
- //time in samples
- WebRtc_UWord32 diff = GetCurrentRTP(&_clock, freq)
- - _localTimeLastReceivedTimestamp;
-
- timestamp = _lastReceivedTimestamp + diff;
- return 0;
-}
-
- // get the currently configured SSRC filter
-WebRtc_Word32
-RTPReceiver::SSRCFilter(WebRtc_UWord32& allowedSSRC) const
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
- if(_useSSRCFilter)
- {
- allowedSSRC = _SSRCFilter;
- return 0;
- }
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "%s invalid state", __FUNCTION__);
- return -1;
-}
-
- // set a SSRC to be used as a filter for incoming RTP streams
-WebRtc_Word32
-RTPReceiver::SetSSRCFilter(const bool enable, const WebRtc_UWord32 allowedSSRC)
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
-
- _useSSRCFilter = enable;
- if(enable)
- {
- _SSRCFilter = allowedSSRC;
- } else
- {
- _SSRCFilter = 0;
- }
- return 0;
-}
-
-// no criticalsection when called
-void RTPReceiver::CheckSSRCChanged(const WebRtcRTPHeader* rtpHeader) {
- bool newSSRC = false;
- bool reInitializeDecoder = false;
- char payloadName[RTP_PAYLOAD_NAME_SIZE];
- WebRtc_UWord32 frequency = 90000; // default video freq
- WebRtc_UWord8 channels = 1;
- WebRtc_UWord32 rate = 0;
-
- {
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
-
- if (_SSRC != rtpHeader->header.ssrc ||
- (_lastReceivedPayloadType == -1 && _SSRC == 0)) {
- // we need the _payloadType to make the call if the remote SSRC is 0
- newSSRC = true;
-
- // reset last report
- ResetStatistics();
- RTPReceiverVideo::ResetOverUseDetector();
-
- _lastReceivedTimestamp = 0;
- _lastReceivedSequenceNumber = 0;
- _lastReceivedTransmissionTimeOffset = 0;
-
- if (_SSRC) { // do we have a SSRC? then the stream is restarted
- // if we have the same codec? reinit decoder
- if (rtpHeader->header.payloadType == _lastReceivedPayloadType) {
- reInitializeDecoder = true;
-
- std::map<WebRtc_Word8, Payload*>::iterator it =
- _payloadTypeMap.find(rtpHeader->header.payloadType);
-
- if (it == _payloadTypeMap.end()) {
- return;
- }
- Payload* payload = it->second;
- assert(payload);
- payloadName[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
- strncpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
- if(payload->audio) {
- frequency = payload->typeSpecific.Audio.frequency;
- channels = payload->typeSpecific.Audio.channels;
- rate = payload->typeSpecific.Audio.rate;
- } else {
- frequency = 90000;
- }
- }
- }
- _SSRC = rtpHeader->header.ssrc;
- }
- }
- if(newSSRC) {
- // we need to get this to our RTCP sender and receiver
- // need to do this outside critical section
- _rtpRtcp.SetRemoteSSRC(rtpHeader->header.ssrc);
- }
- CriticalSectionScoped lock(_criticalSectionCbs);
- if(_cbRtpFeedback) {
- if(newSSRC) {
- _cbRtpFeedback->OnIncomingSSRCChanged(_id, rtpHeader->header.ssrc);
- }
- if(reInitializeDecoder) {
- if (-1 == _cbRtpFeedback->OnInitializeDecoder(_id,
- rtpHeader->header.payloadType, payloadName, frequency, channels,
- rate)) { // new stream same codec
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "Failed to create decoder for payload type:%d",
- rtpHeader->header.payloadType);
- }
- }
- }
-}
-
-// no criticalsection when called
-WebRtc_Word32 RTPReceiver::CheckPayloadChanged(
- const WebRtcRTPHeader* rtpHeader,
- const WebRtc_Word8 firstPayloadByte,
- bool& isRED,
- AudioPayload& audioSpecificPayload,
- VideoPayload& videoSpecificPayload) {
- bool reInitializeDecoder = false;
-
- char payloadName[RTP_PAYLOAD_NAME_SIZE];
- WebRtc_Word8 payloadType = rtpHeader->header.payloadType;
-
- {
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
-
- if (payloadType != _lastReceivedPayloadType) {
- if (REDPayloadType(payloadType)) {
- // get the real codec payload type
- payloadType = firstPayloadByte & 0x7f;
- isRED = true;
-
- if (REDPayloadType(payloadType)) {
- // Invalid payload type, traced by caller. If we proceeded here,
- // this would be set as |_lastReceivedPayloadType|, and we would no
- // longer catch corrupt packets at this level.
- return -1;
- }
-
- //when we receive RED we need to check the real payload type
- if (payloadType == _lastReceivedPayloadType) {
- if(_audio)
- {
- memcpy(&audioSpecificPayload, &_lastReceivedAudioSpecific,
- sizeof(_lastReceivedAudioSpecific));
- } else {
- memcpy(&videoSpecificPayload, &_lastReceivedVideoSpecific,
- sizeof(_lastReceivedVideoSpecific));
- }
- return 0;
- }
- }
- if (_audio) {
- if (TelephoneEventPayloadType(payloadType)) {
- // don't do callbacks for DTMF packets
- isRED = false;
- return 0;
- }
- // frequency is updated for CNG
- if (CNGPayloadType(payloadType, audioSpecificPayload.frequency)) {
- // don't do callbacks for DTMF packets
- isRED = false;
- return 0;
- }
- }
- std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
- _payloadTypeMap.find(payloadType);
-
- // check that this is a registered payload type
- if (it == _payloadTypeMap.end()) {
- return -1;
- }
- Payload* payload = it->second;
- assert(payload);
- payloadName[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
- strncpy(payloadName, payload->name, RTP_PAYLOAD_NAME_SIZE - 1);
- _lastReceivedPayloadType = payloadType;
-
- reInitializeDecoder = true;
-
- if(payload->audio) {
- memcpy(&_lastReceivedAudioSpecific, &(payload->typeSpecific.Audio),
- sizeof(_lastReceivedAudioSpecific));
- memcpy(&audioSpecificPayload, &(payload->typeSpecific.Audio),
- sizeof(_lastReceivedAudioSpecific));
- } else {
- memcpy(&_lastReceivedVideoSpecific, &(payload->typeSpecific.Video),
- sizeof(_lastReceivedVideoSpecific));
- memcpy(&videoSpecificPayload, &(payload->typeSpecific.Video),
- sizeof(_lastReceivedVideoSpecific));
-
- if (_lastReceivedVideoSpecific.videoCodecType == kRtpFecVideo)
- {
- // Only reset the decoder on media packets.
- reInitializeDecoder = false;
- } else {
- if (_lastReceivedMediaPayloadType == _lastReceivedPayloadType) {
- // Only reset the decoder if the media codec type has changed.
- reInitializeDecoder = false;
- }
- _lastReceivedMediaPayloadType = _lastReceivedPayloadType;
- }
- }
- if (reInitializeDecoder) {
- // reset statistics
- ResetStatistics();
- }
- } else {
- if(_audio)
- {
- memcpy(&audioSpecificPayload, &_lastReceivedAudioSpecific,
- sizeof(_lastReceivedAudioSpecific));
- } else
- {
- memcpy(&videoSpecificPayload, &_lastReceivedVideoSpecific,
- sizeof(_lastReceivedVideoSpecific));
- }
- isRED = false;
- }
- } // end critsect
- if (reInitializeDecoder) {
- CriticalSectionScoped lock(_criticalSectionCbs);
- if (_cbRtpFeedback) {
- // create new decoder instance
- if(_audio) {
- if (-1 == _cbRtpFeedback->OnInitializeDecoder(_id, payloadType,
- payloadName, audioSpecificPayload.frequency,
- audioSpecificPayload.channels, audioSpecificPayload.rate)) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "Failed to create audio decoder for payload type:%d",
- payloadType);
- return -1; // Wrong payload type
- }
- } else {
- if (-1 == _cbRtpFeedback->OnInitializeDecoder(_id, payloadType,
- payloadName, 90000, 1, 0)) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "Failed to create video decoder for payload type:%d",
- payloadType);
- return -1; // Wrong payload type
- }
- }
- }
- }
- return 0;
-}
-
-// no criticalsection when called
-void RTPReceiver::CheckCSRC(const WebRtcRTPHeader* rtpHeader) {
- WebRtc_Word32 numCSRCsDiff = 0;
- WebRtc_UWord32 oldRemoteCSRC[kRtpCsrcSize];
- WebRtc_UWord8 oldNumCSRCs = 0;
-
- {
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
-
- if (TelephoneEventPayloadType(rtpHeader->header.payloadType)) {
- // Don't do this for DTMF packets
- return;
- }
- _numEnergy = rtpHeader->type.Audio.numEnergy;
- if (rtpHeader->type.Audio.numEnergy > 0 &&
- rtpHeader->type.Audio.numEnergy <= kRtpCsrcSize) {
- memcpy(_currentRemoteEnergy,
- rtpHeader->type.Audio.arrOfEnergy,
- rtpHeader->type.Audio.numEnergy);
- }
- oldNumCSRCs = _numCSRCs;
- if (oldNumCSRCs > 0) {
- // Make a copy of old.
- memcpy(oldRemoteCSRC, _currentRemoteCSRC,
- _numCSRCs * sizeof(WebRtc_UWord32));
- }
- const WebRtc_UWord8 numCSRCs = rtpHeader->header.numCSRCs;
- if ((numCSRCs > 0) && (numCSRCs <= kRtpCsrcSize)) {
- // Copy new
- memcpy(_currentRemoteCSRC,
- rtpHeader->header.arrOfCSRCs,
- numCSRCs * sizeof(WebRtc_UWord32));
- }
- if (numCSRCs > 0 || oldNumCSRCs > 0) {
- numCSRCsDiff = numCSRCs - oldNumCSRCs;
- _numCSRCs = numCSRCs; // Update stored CSRCs.
- } else {
- // No change.
- return;
- }
- } // End scoped CriticalSection.
-
- CriticalSectionScoped lock(_criticalSectionCbs);
- if (_cbRtpFeedback == NULL) {
- return;
- }
- bool haveCalledCallback = false;
- // Search for new CSRC in old array.
- for (WebRtc_UWord8 i = 0; i < rtpHeader->header.numCSRCs; ++i) {
- const WebRtc_UWord32 csrc = rtpHeader->header.arrOfCSRCs[i];
-
- bool foundMatch = false;
- for (WebRtc_UWord8 j = 0; j < oldNumCSRCs; ++j) {
- if (csrc == oldRemoteCSRC[j]) { // old list
- foundMatch = true;
- break;
- }
- }
- if (!foundMatch && csrc) {
- // Didn't find it, report it as new.
- haveCalledCallback = true;
- _cbRtpFeedback->OnIncomingCSRCChanged(_id, csrc, true);
- }
- }
- // Search for old CSRC in new array.
- for (WebRtc_UWord8 i = 0; i < oldNumCSRCs; ++i) {
- const WebRtc_UWord32 csrc = oldRemoteCSRC[i];
-
- bool foundMatch = false;
- for (WebRtc_UWord8 j = 0; j < rtpHeader->header.numCSRCs; ++j) {
- if (csrc == rtpHeader->header.arrOfCSRCs[j]) {
- foundMatch = true;
- break;
- }
- }
- if (!foundMatch && csrc) {
- // Did not find it, report as removed.
- haveCalledCallback = true;
- _cbRtpFeedback->OnIncomingCSRCChanged(_id, csrc, false);
- }
- }
- if (!haveCalledCallback) {
- // If the CSRC list contain non-unique entries we will endup here.
- // Using CSRC 0 to signal this event, not interop safe, other
- // implementations might have CSRC 0 as avalid value.
- if (numCSRCsDiff > 0) {
- _cbRtpFeedback->OnIncomingCSRCChanged(_id, 0, true);
- } else if (numCSRCsDiff < 0) {
- _cbRtpFeedback->OnIncomingCSRCChanged(_id, 0, false);
- }
- }
-}
-
-WebRtc_Word32
-RTPReceiver::ResetStatistics()
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
-
- _lastReportInorderPackets = 0;
- _lastReportOldPackets = 0;
- _lastReportSeqMax = 0;
- _lastReportFractionLost = 0;
- _lastReportCumulativeLost = 0;
- _lastReportExtendedHighSeqNum = 0;
- _lastReportJitter = 0;
- _lastReportJitterTransmissionTimeOffset = 0;
- _jitterQ4 = 0;
- _jitterMaxQ4 = 0;
- _cumulativeLoss = 0;
- _jitterQ4TransmissionTimeOffset = 0;
- _receivedSeqWraps = 0;
- _receivedSeqMax = 0;
- _receivedSeqFirst = 0;
- _receivedByteCount = 0;
- _receivedOldPacketCount = 0;
- _receivedInorderPacketCount = 0;
- return 0;
-}
-
-WebRtc_Word32
-RTPReceiver::ResetDataCounters()
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
-
- _receivedByteCount = 0;
- _receivedOldPacketCount = 0;
- _receivedInorderPacketCount = 0;
- _lastReportInorderPackets = 0;
-
- return 0;
-}
-
-WebRtc_Word32
-RTPReceiver::Statistics(WebRtc_UWord8 *fraction_lost,
- WebRtc_UWord32 *cum_lost,
- WebRtc_UWord32 *ext_max,
- WebRtc_UWord32 *jitter,
- WebRtc_UWord32 *max_jitter,
- WebRtc_UWord32 *jitter_transmission_time_offset,
- bool reset) const
-{
- WebRtc_Word32 missing;
- return Statistics(fraction_lost,
- cum_lost,
- ext_max,
- jitter,
- max_jitter,
- jitter_transmission_time_offset,
- &missing,
- reset);
-}
-
-WebRtc_Word32
-RTPReceiver::Statistics(WebRtc_UWord8 *fraction_lost,
- WebRtc_UWord32 *cum_lost,
- WebRtc_UWord32 *ext_max,
- WebRtc_UWord32 *jitter,
- WebRtc_UWord32 *max_jitter,
- WebRtc_UWord32 *jitter_transmission_time_offset,
- WebRtc_Word32 *missing,
- bool reset) const
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
-
- if (missing == NULL)
- {
- return -1;
- }
- if(_receivedSeqFirst == 0 && _receivedByteCount == 0)
- {
- // we have not received anything
- // -1 required by RTCP sender
- return -1;
- }
- if(!reset)
- {
- if(_lastReportInorderPackets == 0)
- {
- // no report
- return -1;
- }
- // just get last report
- if(fraction_lost)
- {
- *fraction_lost = _lastReportFractionLost;
- }
- if(cum_lost)
- {
- *cum_lost = _lastReportCumulativeLost; // 24 bits valid
- }
- if(ext_max)
- {
- *ext_max = _lastReportExtendedHighSeqNum;
- }
- if(jitter)
- {
- *jitter =_lastReportJitter;
- }
- if(max_jitter)
- {
- // note that the internal jitter value is in Q4
- // and needs to be scaled by 1/16
- *max_jitter = (_jitterMaxQ4 >> 4);
- }
- if(jitter_transmission_time_offset)
- {
- *jitter_transmission_time_offset =
- _lastReportJitterTransmissionTimeOffset;
- }
- return 0;
- }
-
- if (_lastReportInorderPackets == 0)
- {
- // First time we send a report
- _lastReportSeqMax = _receivedSeqFirst-1;
- }
- /*
- * calc fraction lost
- */
- WebRtc_UWord16 expSinceLast = (_receivedSeqMax - _lastReportSeqMax);
-
- if(_lastReportSeqMax > _receivedSeqMax)
- {
- // can we assume that the seqNum can't go decrease over a full RTCP period ?
- expSinceLast = 0;
- }
-
- // number of received RTP packets since last report, counts all packets but not re-transmissions
- WebRtc_UWord32 recSinceLast = _receivedInorderPacketCount - _lastReportInorderPackets;
-
- if(_nackMethod == kNackOff)
- {
- // this is needed for re-ordered packets
- WebRtc_UWord32 oldPackets = _receivedOldPacketCount - _lastReportOldPackets;
- recSinceLast += oldPackets;
- }else
- {
- // with NACK we don't know the expected retransmitions during the last second
- // we know how many "old" packets we have received we just count the numer of
- // old received to estimate the loss but it still does not guarantee an exact number
- // since we run this based on time triggered by sending of a RTP packet this
- // should have a minimum effect
-
- // with NACK we don't count old packets as received since they are re-transmitted
- // we use RTT to decide if a packet is re-ordered or re-transmitted
- }
-
- *missing = 0;
- if(expSinceLast > recSinceLast)
- {
- *missing = (expSinceLast - recSinceLast);
- }
- WebRtc_UWord8 fractionLost = 0;
- if(expSinceLast)
- {
- // scale 0 to 255, where 255 is 100% loss
- fractionLost = (WebRtc_UWord8) ((255 * (*missing)) / expSinceLast);
- }
- if(fraction_lost)
- {
- *fraction_lost = fractionLost;
- }
- // we need a counter for cumulative loss too
- _cumulativeLoss += *missing;
-
- if(_jitterQ4 > _jitterMaxQ4)
- {
- _jitterMaxQ4 = _jitterQ4;
- }
- if(cum_lost)
- {
- *cum_lost = _cumulativeLoss;
- }
- if(ext_max)
- {
- *ext_max = (_receivedSeqWraps<<16) + _receivedSeqMax;
- }
- if(jitter)
- {
- // note that the internal jitter value is in Q4
- // and needs to be scaled by 1/16
- *jitter = (_jitterQ4 >> 4);
- }
- if(max_jitter)
- {
- // note that the internal jitter value is in Q4
- // and needs to be scaled by 1/16
- *max_jitter = (_jitterMaxQ4 >> 4);
- }
- if(jitter_transmission_time_offset)
- {
- // note that the internal jitter value is in Q4
- // and needs to be scaled by 1/16
- *jitter_transmission_time_offset =
- (_jitterQ4TransmissionTimeOffset >> 4);
- }
- if(reset)
- {
- // store this report
- _lastReportFractionLost = fractionLost;
- _lastReportCumulativeLost = _cumulativeLoss; // 24 bits valid
- _lastReportExtendedHighSeqNum = (_receivedSeqWraps<<16) + _receivedSeqMax;
- _lastReportJitter = (_jitterQ4 >> 4);
- _lastReportJitterTransmissionTimeOffset =
- (_jitterQ4TransmissionTimeOffset >> 4);
-
- // only for report blocks in RTCP SR and RR
- _lastReportInorderPackets = _receivedInorderPacketCount;
- _lastReportOldPackets = _receivedOldPacketCount;
- _lastReportSeqMax = _receivedSeqMax;
- }
- return 0;
-}
-
-WebRtc_Word32
-RTPReceiver::DataCounters(WebRtc_UWord32 *bytesReceived,
- WebRtc_UWord32 *packetsReceived) const
-{
- CriticalSectionScoped lock(_criticalSectionRTPReceiver);
-
- if(bytesReceived)
- {
- *bytesReceived = _receivedByteCount;
- }
- if(packetsReceived)
- {
- *packetsReceived = _receivedOldPacketCount + _receivedInorderPacketCount;
- }
- return 0;
-}
-
-void
-RTPReceiver::ProcessBitrate()
-{
- CriticalSectionScoped cs(_criticalSectionRTPReceiver);
-
- Bitrate::Process();
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_receiver.h b/trunk/src/modules/rtp_rtcp/source/rtp_receiver.h
deleted file mode 100644
index 4a14c25..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_receiver.h
+++ /dev/null
@@ -1,264 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
-
-#include <map>
-
-#include "typedefs.h"
-#include "rtp_utility.h"
-
-#include "rtp_header_extension.h"
-#include "rtp_rtcp.h"
-#include "rtp_rtcp_defines.h"
-#include "rtp_receiver_audio.h"
-#include "rtp_receiver_video.h"
-#include "rtcp_receiver_help.h"
-#include "Bitrate.h"
-
-namespace webrtc {
-class RtpRtcpFeedback;
-class ModuleRtpRtcpImpl;
-class Trace;
-
-class RTPReceiver : public RTPReceiverAudio, public RTPReceiverVideo, public Bitrate
-{
-public:
- RTPReceiver(const WebRtc_Word32 id,
- const bool audio,
- RtpRtcpClock* clock,
- ModuleRtpRtcpImpl* owner);
-
- virtual ~RTPReceiver();
-
- virtual void ChangeUniqueId(const WebRtc_Word32 id);
-
- WebRtc_Word32 Init();
-
- RtpVideoCodecTypes VideoCodecType() const;
- WebRtc_UWord32 MaxConfiguredBitrate() const;
-
- WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 timeoutMS);
- void PacketTimeout();
-
- void ProcessDeadOrAlive(const bool RTCPalive, const WebRtc_UWord32 now);
-
- void ProcessBitrate();
-
- WebRtc_Word32 RegisterIncomingDataCallback(RtpData* incomingDataCallback);
- WebRtc_Word32 RegisterIncomingRTPCallback(RtpFeedback* incomingMessagesCallback);
-
- WebRtc_Word32 RegisterReceivePayload(
- const char payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate);
-
- WebRtc_Word32 DeRegisterReceivePayload(const WebRtc_Word8 payloadType);
-
- WebRtc_Word32 ReceivePayloadType(
- const char payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_UWord32 frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate,
- WebRtc_Word8* payloadType) const;
-
- WebRtc_Word32 ReceivePayload(const WebRtc_Word8 payloadType,
- char payloadName[RTP_PAYLOAD_NAME_SIZE],
- WebRtc_UWord32* frequency,
- WebRtc_UWord8* channels,
- WebRtc_UWord32* rate) const;
-
- WebRtc_Word32 RemotePayload(char payloadName[RTP_PAYLOAD_NAME_SIZE],
- WebRtc_Word8* payloadType,
- WebRtc_UWord32* frequency,
- WebRtc_UWord8* channels) const;
-
- WebRtc_Word32 IncomingRTPPacket(WebRtcRTPHeader* rtpheader,
- const WebRtc_UWord8* incomingRtpPacket,
- const WebRtc_UWord16 incomingRtpPacketLengt);
-
- NACKMethod NACK() const ;
-
- // Turn negative acknowledgement requests on/off
- WebRtc_Word32 SetNACKStatus(const NACKMethod method);
-
-
- // last received
- virtual WebRtc_UWord32 TimeStamp() const;
- virtual WebRtc_UWord16 SequenceNumber() const;
-
- WebRtc_Word32 EstimatedRemoteTimeStamp(WebRtc_UWord32& timestamp) const;
-
- WebRtc_UWord32 SSRC() const;
-
- WebRtc_Word32 CSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const;
-
- WebRtc_Word32 Energy( WebRtc_UWord8 arrOfEnergy[kRtpCsrcSize]) const;
-
- // get the currently configured SSRC filter
- WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowedSSRC) const;
-
- // set a SSRC to be used as a filter for incoming RTP streams
- WebRtc_Word32 SetSSRCFilter(const bool enable, const WebRtc_UWord32 allowedSSRC);
-
- WebRtc_Word32 Statistics(WebRtc_UWord8 *fraction_lost,
- WebRtc_UWord32 *cum_lost,
- WebRtc_UWord32 *ext_max,
- WebRtc_UWord32 *jitter, // will be moved from JB
- WebRtc_UWord32 *max_jitter,
- WebRtc_UWord32 *jitter_transmission_time_offset,
- bool reset) const;
-
- WebRtc_Word32 Statistics(WebRtc_UWord8 *fraction_lost,
- WebRtc_UWord32 *cum_lost,
- WebRtc_UWord32 *ext_max,
- WebRtc_UWord32 *jitter, // will be moved from JB
- WebRtc_UWord32 *max_jitter,
- WebRtc_UWord32 *jitter_transmission_time_offset,
- WebRtc_Word32 *missing,
- bool reset) const;
-
- WebRtc_Word32 DataCounters(WebRtc_UWord32 *bytesReceived,
- WebRtc_UWord32 *packetsReceived) const;
-
- WebRtc_Word32 ResetStatistics();
-
- WebRtc_Word32 ResetDataCounters();
-
- WebRtc_UWord16 PacketOHReceived() const;
-
- WebRtc_UWord32 PacketCountReceived() const;
-
- WebRtc_UWord32 ByteCountReceived() const;
-
- WebRtc_Word32 RegisterRtpHeaderExtension(const RTPExtensionType type,
- const WebRtc_UWord8 id);
-
- WebRtc_Word32 DeregisterRtpHeaderExtension(const RTPExtensionType type);
-
- void GetHeaderExtensionMapCopy(RtpHeaderExtensionMap* map) const;
-
- virtual WebRtc_UWord32 PayloadTypeToPayload(const WebRtc_UWord8 payloadType,
- ModuleRTPUtility::Payload*& payload) const;
- /*
- * RTX
- */
- void SetRTXStatus(const bool enable, const WebRtc_UWord32 SSRC);
-
- void RTXStatus(bool* enable, WebRtc_UWord32* SSRC) const;
-
-protected:
- virtual WebRtc_Word32 CallbackOfReceivedPayloadData(const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const WebRtcRTPHeader* rtpHeader);
-
- virtual bool RetransmitOfOldPacket(const WebRtc_UWord16 sequenceNumber,
- const WebRtc_UWord32 rtpTimeStamp) const;
-
-
- void UpdateStatistics(const WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord16 bytes,
- const bool oldPacket);
-
- virtual WebRtc_Word8 REDPayloadType() const;
-
-private:
- // Is RED configured with payload type payloadType
- bool REDPayloadType(const WebRtc_Word8 payloadType) const;
-
- bool InOrderPacket(const WebRtc_UWord16 sequenceNumber) const;
-
- void CheckSSRCChanged(const WebRtcRTPHeader* rtpHeader);
- void CheckCSRC(const WebRtcRTPHeader* rtpHeader);
- WebRtc_Word32 CheckPayloadChanged(const WebRtcRTPHeader* rtpHeader,
- const WebRtc_Word8 firstPayloadByte,
- bool& isRED,
- ModuleRTPUtility::AudioPayload& audioSpecific,
- ModuleRTPUtility::VideoPayload& videoSpecific);
-
- void UpdateNACKBitRate(WebRtc_Word32 bytes, WebRtc_UWord32 now);
- bool ProcessNACKBitRate(WebRtc_UWord32 now);
-
-private:
- WebRtc_Word32 _id;
- const bool _audio;
- ModuleRtpRtcpImpl& _rtpRtcp;
-
- CriticalSectionWrapper* _criticalSectionCbs;
- RtpFeedback* _cbRtpFeedback;
- RtpData* _cbRtpData;
-
- CriticalSectionWrapper* _criticalSectionRTPReceiver;
- mutable WebRtc_UWord32 _lastReceiveTime;
- WebRtc_UWord16 _lastReceivedPayloadLength;
- WebRtc_Word8 _lastReceivedPayloadType;
- WebRtc_Word8 _lastReceivedMediaPayloadType;
-
- ModuleRTPUtility::AudioPayload _lastReceivedAudioSpecific;
- ModuleRTPUtility::VideoPayload _lastReceivedVideoSpecific;
-
- WebRtc_UWord32 _packetTimeOutMS;
-
- WebRtc_Word8 _redPayloadType;
-
- std::map<WebRtc_Word8, ModuleRTPUtility::Payload*> _payloadTypeMap;
- RtpHeaderExtensionMap _rtpHeaderExtensionMap;
-
- // SSRCs
- WebRtc_UWord32 _SSRC;
- WebRtc_UWord8 _numCSRCs;
- WebRtc_UWord32 _currentRemoteCSRC[kRtpCsrcSize];
- WebRtc_UWord8 _numEnergy;
- WebRtc_UWord8 _currentRemoteEnergy[kRtpCsrcSize];
-
- bool _useSSRCFilter;
- WebRtc_UWord32 _SSRCFilter;
-
- // stats on received RTP packets
- WebRtc_UWord32 _jitterQ4;
- mutable WebRtc_UWord32 _jitterMaxQ4;
- mutable WebRtc_UWord32 _cumulativeLoss;
- WebRtc_UWord32 _jitterQ4TransmissionTimeOffset;
-
- WebRtc_UWord32 _localTimeLastReceivedTimestamp;
- WebRtc_UWord32 _lastReceivedTimestamp;
- WebRtc_UWord16 _lastReceivedSequenceNumber;
- WebRtc_Word32 _lastReceivedTransmissionTimeOffset;
- WebRtc_UWord16 _receivedSeqFirst;
- WebRtc_UWord16 _receivedSeqMax;
- WebRtc_UWord16 _receivedSeqWraps;
-
- // current counter values
- WebRtc_UWord16 _receivedPacketOH;
- WebRtc_UWord32 _receivedByteCount;
- WebRtc_UWord32 _receivedOldPacketCount;
- WebRtc_UWord32 _receivedInorderPacketCount;
-
- // counter values when we sent the last report
- mutable WebRtc_UWord32 _lastReportInorderPackets;
- mutable WebRtc_UWord32 _lastReportOldPackets;
- mutable WebRtc_UWord16 _lastReportSeqMax;
- mutable WebRtc_UWord8 _lastReportFractionLost;
- mutable WebRtc_UWord32 _lastReportCumulativeLost; // 24 bits valid
- mutable WebRtc_UWord32 _lastReportExtendedHighSeqNum;
- mutable WebRtc_UWord32 _lastReportJitter;
- mutable WebRtc_UWord32 _lastReportJitterTransmissionTimeOffset;
-
- NACKMethod _nackMethod;
-
- bool _RTX;
- WebRtc_UWord32 _ssrcRTX;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc b/trunk/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
deleted file mode 100644
index b61157d..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_receiver_audio.cc
+++ /dev/null
@@ -1,590 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rtp_receiver_audio.h"
-
-#include <cassert> //assert
-#include <cstring> // memcpy()
-#include <math.h> // pow()
-
-#include "critical_section_wrapper.h"
-
-namespace webrtc {
-RTPReceiverAudio::RTPReceiverAudio(const WebRtc_Word32 id):
- _id(id),
- _lastReceivedFrequency(8000),
- _telephoneEvent(false),
- _telephoneEventForwardToDecoder(false),
- _telephoneEventDetectEndOfTone(false),
- _telephoneEventPayloadType(-1),
- _cngNBPayloadType(-1),
- _cngWBPayloadType(-1),
- _cngSWBPayloadType(-1),
- _cngPayloadType(-1),
- _G722PayloadType(-1),
- _lastReceivedG722(false),
- _criticalSectionFeedback(CriticalSectionWrapper::CreateCriticalSection()),
- _cbAudioFeedback(NULL)
-{
-}
-
-RTPReceiverAudio::~RTPReceiverAudio()
-{
- delete _criticalSectionFeedback;
-}
-
-WebRtc_Word32 RTPReceiverAudio::Init() {
- _lastReceivedFrequency = 8000;
- _telephoneEvent = false;
- _telephoneEventForwardToDecoder = false;
- _telephoneEventDetectEndOfTone = false;
- _telephoneEventPayloadType = -1;
-
- _telephoneEventReported.clear();
-
- _cngNBPayloadType = -1;
- _cngWBPayloadType = -1;
- _cngSWBPayloadType = -1;
- _cngPayloadType = -1;
- _G722PayloadType = -1;
- _lastReceivedG722 = false;
- return 0;
-}
-
-void
-RTPReceiverAudio::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
-}
-
-WebRtc_Word32
-RTPReceiverAudio::RegisterIncomingAudioCallback(RtpAudioFeedback* incomingMessagesCallback)
-{
- CriticalSectionScoped lock(_criticalSectionFeedback);
- _cbAudioFeedback = incomingMessagesCallback;
- return 0;
-}
-
-WebRtc_UWord32
-RTPReceiverAudio::AudioFrequency() const
-{
- if(_lastReceivedG722)
- {
- return 8000;
- }
- return _lastReceivedFrequency;
-}
-
-// Outband TelephoneEvent(DTMF) detection
-WebRtc_Word32
-RTPReceiverAudio::SetTelephoneEventStatus(const bool enable,
- const bool forwardToDecoder,
- const bool detectEndOfTone)
-{
- _telephoneEvent= enable;
- _telephoneEventDetectEndOfTone = detectEndOfTone;
- _telephoneEventForwardToDecoder = forwardToDecoder;
- return 0;
-}
-
- // Is outband TelephoneEvent(DTMF) turned on/off?
-bool
-RTPReceiverAudio::TelephoneEvent() const
-{
- return _telephoneEvent;
-}
-
-// Is forwarding of outband telephone events turned on/off?
-bool
-RTPReceiverAudio::TelephoneEventForwardToDecoder() const
-{
- return _telephoneEventForwardToDecoder;
-}
-
-bool
-RTPReceiverAudio::TelephoneEventPayloadType(const WebRtc_Word8 payloadType) const
-{
- return (_telephoneEventPayloadType == payloadType)?true:false;
-}
-
-bool
-RTPReceiverAudio::CNGPayloadType(const WebRtc_Word8 payloadType,
- WebRtc_UWord32& frequency)
-{
- // we can have three CNG on 8000Hz, 16000Hz and 32000Hz
- if(_cngNBPayloadType == payloadType)
- {
- frequency = 8000;
- if ((_cngPayloadType != -1) &&(_cngPayloadType !=_cngNBPayloadType))
- {
- ResetStatistics();
- }
- _cngPayloadType = _cngNBPayloadType;
- return true;
- } else if(_cngWBPayloadType == payloadType)
- {
- // if last received codec is G.722 we must use frequency 8000
- if(_lastReceivedG722)
- {
- frequency = 8000;
- } else
- {
- frequency = 16000;
- }
- if ((_cngPayloadType != -1) &&(_cngPayloadType !=_cngWBPayloadType))
- {
- ResetStatistics();
- }
- _cngPayloadType = _cngWBPayloadType;
- return true;
- }else if(_cngSWBPayloadType == payloadType)
- {
- frequency = 32000;
- if ((_cngPayloadType != -1) &&(_cngPayloadType !=_cngSWBPayloadType))
- {
- ResetStatistics();
- }
- _cngPayloadType = _cngSWBPayloadType;
- return true;
- }else
- {
- // not CNG
- if(_G722PayloadType == payloadType)
- {
- _lastReceivedG722 = true;
- }else
- {
- _lastReceivedG722 = false;
- }
- }
- return false;
-}
-
-/*
- Sample based or frame based codecs based on RFC 3551
-
- NOTE! There is one error in the RFC, stating G.722 uses 8 bits/samples.
- The correct rate is 4 bits/sample.
-
- name of sampling default
- encoding sample/frame bits/sample rate ms/frame ms/packet
-
- Sample based audio codecs
- DVI4 sample 4 var. 20
- G722 sample 4 16,000 20
- G726-40 sample 5 8,000 20
- G726-32 sample 4 8,000 20
- G726-24 sample 3 8,000 20
- G726-16 sample 2 8,000 20
- L8 sample 8 var. 20
- L16 sample 16 var. 20
- PCMA sample 8 var. 20
- PCMU sample 8 var. 20
-
- Frame based audio codecs
- G723 frame N/A 8,000 30 30
- G728 frame N/A 8,000 2.5 20
- G729 frame N/A 8,000 10 20
- G729D frame N/A 8,000 10 20
- G729E frame N/A 8,000 10 20
- GSM frame N/A 8,000 20 20
- GSM-EFR frame N/A 8,000 20 20
- LPC frame N/A 8,000 20 20
- MPA frame N/A var. var.
-
- G7221 frame N/A
-*/
-
-ModuleRTPUtility::Payload* RTPReceiverAudio::RegisterReceiveAudioPayload(
- const char payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate) {
- if (ModuleRTPUtility::StringCompare(payloadName, "telephone-event", 15)) {
- _telephoneEventPayloadType = payloadType;
- }
- if (ModuleRTPUtility::StringCompare(payloadName, "cn", 2)) {
- // we can have three CNG on 8000Hz, 16000Hz and 32000Hz
- if(frequency == 8000){
- _cngNBPayloadType = payloadType;
- } else if(frequency == 16000) {
- _cngWBPayloadType = payloadType;
- } else if(frequency == 32000) {
- _cngSWBPayloadType = payloadType;
- } else {
- assert(false);
- return NULL;
- }
- }
- WebRtc_UWord8 bitsPerSample = 0; // zero implies frame based
- bool isTrueStereo = false; // Default value
- if (ModuleRTPUtility::StringCompare(payloadName, "DVI4", 4)) {
- bitsPerSample = 4;
- } else if(ModuleRTPUtility::StringCompare(payloadName, "G722", 4)) {
- if(ModuleRTPUtility::StringCompare(payloadName, "G7221", 5)) {
- // frame based
- } else {
- _G722PayloadType = payloadType;
- bitsPerSample = 4;
- }
- } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-40",7)) {
- bitsPerSample = 5;
- } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-32",7)) {
- bitsPerSample = 4;
- } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-24",7)) {
- bitsPerSample = 3;
- } else if(ModuleRTPUtility::StringCompare(payloadName,"G726-16",7)) {
- bitsPerSample = 2;
- } else if(ModuleRTPUtility::StringCompare(payloadName,"L8",2)) {
- bitsPerSample = 8;
- } else if(ModuleRTPUtility::StringCompare(payloadName,"L16",3)) {
- bitsPerSample = 16;
- } else if(ModuleRTPUtility::StringCompare(payloadName,"PCMU",4)) {
- bitsPerSample = 8;
- } else if(ModuleRTPUtility::StringCompare(payloadName,"PCMA",4)) {
- bitsPerSample = 8;
- } else if(ModuleRTPUtility::StringCompare(payloadName,"CELT",4))
- {
- isTrueStereo = true;
- }
- ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload;
- payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
- strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
- payload->typeSpecific.Audio.frequency = frequency;
- payload->typeSpecific.Audio.channels = channels;
- payload->typeSpecific.Audio.bitsPerSample = bitsPerSample;
- payload->typeSpecific.Audio.rate = rate;
- payload->typeSpecific.Audio.trueStereoCodec = isTrueStereo;
- payload->audio = true;
- return payload;
-}
-
-// we are not allowed to have any critsects when calling CallbackOfReceivedPayloadData
-WebRtc_Word32
-RTPReceiverAudio::ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadLength,
- const ModuleRTPUtility::AudioPayload& audioSpecific,
- const bool isRED)
-{
- WebRtc_UWord8 newEvents[MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS];
- WebRtc_UWord8 removedEvents[MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS];
- WebRtc_UWord8 numberOfNewEvents = 0;
- WebRtc_UWord8 numberOfRemovedEvents = 0;
- bool telephoneEventPacket = TelephoneEventPayloadType(rtpHeader->header.payloadType);
-
- if(payloadLength == 0)
- {
- return 0;
- }
-
- {
- CriticalSectionScoped lock(_criticalSectionFeedback);
-
- if(telephoneEventPacket)
- {
- // RFC 4733 2.3
- /*
- 0 1 2 3
- 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | event |E|R| volume | duration |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- */
- if(payloadLength % 4 != 0)
- {
- return -1;
- }
- WebRtc_UWord8 numberOfEvents = payloadLength / 4;
-
- // sanity
- if(numberOfEvents >= MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS)
- {
- numberOfEvents = MAX_NUMBER_OF_PARALLEL_TELEPHONE_EVENTS;
- }
- for (int n = 0; n < numberOfEvents; n++)
- {
- bool end = (payloadData[(4*n)+1] & 0x80)? true:false;
-
- std::set<WebRtc_UWord8>::iterator event =
- _telephoneEventReported.find(payloadData[4*n]);
-
- if(event != _telephoneEventReported.end())
- {
- // we have already seen this event
- if(end)
- {
- removedEvents[numberOfRemovedEvents]= payloadData[4*n];
- numberOfRemovedEvents++;
- _telephoneEventReported.erase(payloadData[4*n]);
- }
- }else
- {
- if(end)
- {
- // don't add if it's a end of a tone
- }else
- {
- newEvents[numberOfNewEvents] = payloadData[4*n];
- numberOfNewEvents++;
- _telephoneEventReported.insert(payloadData[4*n]);
- }
- }
- }
-
- // RFC 4733 2.5.1.3 & 2.5.2.3 Long-Duration Events
- // should not be a problem since we don't care about the duration
-
- // RFC 4733 See 2.5.1.5. & 2.5.2.4. Multiple Events in a Packet
- }
-
- if(_telephoneEvent && _cbAudioFeedback)
- {
- for (int n = 0; n < numberOfNewEvents; n++)
- {
- _cbAudioFeedback->OnReceivedTelephoneEvent(_id, newEvents[n], false);
- }
- if(_telephoneEventDetectEndOfTone)
- {
- for (int n = 0; n < numberOfRemovedEvents; n++)
- {
- _cbAudioFeedback->OnReceivedTelephoneEvent(_id, removedEvents[n], true);
- }
- }
- }
- }
- if(! telephoneEventPacket )
- {
- _lastReceivedFrequency = audioSpecific.frequency;
- }
-
- // Check if this is a CNG packet, receiver might want to know
- WebRtc_UWord32 dummy;
- if(CNGPayloadType(rtpHeader->header.payloadType, dummy))
- {
- rtpHeader->type.Audio.isCNG=true;
- rtpHeader->frameType = kAudioFrameCN;
- }else
- {
- rtpHeader->frameType = kAudioFrameSpeech;
- rtpHeader->type.Audio.isCNG=false;
- }
-
- // check if it's a DTMF event, hence something we can playout
- if(telephoneEventPacket)
- {
- if(!_telephoneEventForwardToDecoder)
- {
- // don't forward event to decoder
- return 0;
- }
- std::set<WebRtc_UWord8>::iterator first =
- _telephoneEventReported.begin();
- if(first != _telephoneEventReported.end() && *first > 15)
- {
- // don't forward non DTMF events
- return 0;
- }
- }
- if(isRED && !(payloadData[0] & 0x80))
- {
- // we recive only one frame packed in a RED packet remove the RED wrapper
- rtpHeader->header.payloadType = payloadData[0];
-
- // only one frame in the RED strip the one byte to help NetEq
- return CallbackOfReceivedPayloadData(payloadData+1,
- payloadLength-1,
- rtpHeader);
- }
- if(audioSpecific.channels > 1)
- {
- WebRtc_Word32 retVal = 0;
- WebRtc_UWord16 channelLength = payloadLength/audioSpecific.channels;
-
- if(audioSpecific.bitsPerSample > 0)
- {
- // sanity
- assert((payloadLength*8)%audioSpecific.bitsPerSample == 0);
-
- // sample based codec
-
- // build matrix
- WebRtc_UWord8 matrix[IP_PACKET_SIZE];
- WebRtc_UWord32 offsetBytes = 0;
- WebRtc_UWord32 offsetBytesInsert = 0;
- // initialize matrix to 0
- memset(matrix, 0, audioSpecific.channels*channelLength);
-
- switch(audioSpecific.bitsPerSample)
- {
- case 1:
- case 2:
- case 3:
- case 4:
- case 5:
- case 6:
- case 7:
- {
- WebRtc_UWord32 offsetSamples = 0;
- WebRtc_UWord32 offsetSamplesInsert = 0;
- WebRtc_UWord16 bitMask = (WebRtc_UWord16)ModuleRTPUtility::pow2(audioSpecific.bitsPerSample)-1;
- WebRtc_UWord16 samplesPerChannel =payloadLength*8/audioSpecific.bitsPerSample/audioSpecific.channels;
-
- for(WebRtc_UWord32 i = 0; i < samplesPerChannel; i++)
- {
- WebRtc_UWord8 insertShift = (WebRtc_UWord8)((offsetSamplesInsert+audioSpecific.bitsPerSample)%16);
- insertShift = 16 - insertShift; // inverse the calculation
-
- for(WebRtc_UWord32 j = 0; j < audioSpecific.channels; j++)
- {
- // get sample
- WebRtc_UWord16 s = payloadData[offsetBytes] << 8;
-
- // check that we don't read outside the memory
- if(offsetBytes < (WebRtc_UWord32)payloadLength -2)
- {
- s += payloadData[offsetBytes+1];
- }
-
- WebRtc_UWord8 readShift = (WebRtc_UWord8)((offsetSamples+audioSpecific.bitsPerSample)%16);
- readShift = 16 - readShift; // inverse the calculation
- s >>= readShift;
- s &= bitMask;
-
- // prepare for reading next sample
- offsetSamples += audioSpecific.bitsPerSample;
- if(readShift <= audioSpecific.bitsPerSample)
- {
- // next does not fitt
- // or fitt exactly
- offsetSamples -= 8;
- offsetBytes++;
- }
-
- // insert sample into matrix
- WebRtc_UWord32 columOffset = j*channelLength;
-
- WebRtc_UWord16 insert = s << insertShift;
-#if defined(WEBRTC_LITTLE_ENDIAN)
- matrix[columOffset+offsetBytesInsert] |= static_cast<WebRtc_UWord8>(insert>>8);
- matrix[columOffset+offsetBytesInsert+1] |= static_cast<WebRtc_UWord8>(insert);
-#else
- WebRtc_UWord16* matrixU16 = (WebRtc_UWord16*)&(matrix[columOffset+offsetBytesInsert]);
- matrixU16[0] |= (s << insertShift);
-#endif
- }
- // prepare for writing next sample
- offsetSamplesInsert += audioSpecific.bitsPerSample;
- if(insertShift <= audioSpecific.bitsPerSample)
- {
- // next does not fitt
- // or fitt exactly
- offsetSamplesInsert -= 8;
- offsetBytesInsert++;
- }
- }
- }
- break;
- case 8:
- {
- WebRtc_UWord32 sample = 0;
- for(WebRtc_UWord32 i = 0; i < channelLength; i++)
- {
- for(WebRtc_UWord32 j = 0; j < audioSpecific.channels; j++)
- {
- WebRtc_UWord32 columOffset = j*channelLength;
- matrix[columOffset + i] = payloadData[sample++];
- }
- }
- }
- break;
- case 16:
- {
- WebRtc_UWord32 sample = 0;
- for(WebRtc_UWord32 i = 0; i < channelLength; i +=2)
- {
- for(WebRtc_UWord32 j = 0; j < audioSpecific.channels; j++)
- {
- WebRtc_UWord32 columOffset = j*channelLength;
- matrix[columOffset + i] = payloadData[sample++];
- matrix[columOffset + i + 1] = payloadData[sample++];
- }
- }
- }
- break;
- default:
- assert(false);
- return -1;
- }
- // we support 16 bits sample
- // callback for all channels
- for(int channel = 0; channel < audioSpecific.channels && retVal == 0; channel++)
- {
- // one callback per channel
- rtpHeader->type.Audio.channel = channel+1;
-
- if(channel == 0)
- {
- // include the original packet only in the first callback
- retVal = CallbackOfReceivedPayloadData(&matrix[channel*channelLength],
- channelLength,
- rtpHeader);
- } else
- {
- retVal = CallbackOfReceivedPayloadData(&matrix[channel*channelLength],
- channelLength,
- rtpHeader);
- }
- }
- } else if (audioSpecific.trueStereoCodec)
- {
- // One callback with the whole payload for each channel.
- for(int channel = 1; (channel <= audioSpecific.channels) &&
- (retVal == 0); channel++)
- {
- // One callback per channel.
- rtpHeader->type.Audio.channel = channel;
- retVal = CallbackOfReceivedPayloadData(payloadData,
- payloadLength,
- rtpHeader);
- }
- } else
- {
- for(int channel = 1; channel <= audioSpecific.channels && retVal == 0; channel++)
- {
- // one callback per channel
- rtpHeader->type.Audio.channel = channel;
-
- if(channel == 1)
- {
- // include the original packet only in the first callback
- retVal = CallbackOfReceivedPayloadData(payloadData,
- channelLength,
- rtpHeader);
- } else
- {
- retVal = CallbackOfReceivedPayloadData(payloadData,
- channelLength,
- rtpHeader);
- }
- payloadData += channelLength;
- }
- }
- return retVal;
- }else
- {
- rtpHeader->type.Audio.channel = 1;
- return CallbackOfReceivedPayloadData(payloadData,
- payloadLength,
- rtpHeader);
- }
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_receiver_audio.h b/trunk/src/modules/rtp_rtcp/source/rtp_receiver_audio.h
deleted file mode 100644
index 89cd062..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_receiver_audio.h
+++ /dev/null
@@ -1,98 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
-
-#include <set>
-
-#include "rtp_rtcp_defines.h"
-#include "rtp_utility.h"
-
-#include "typedefs.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class RTPReceiverAudio
-{
-public:
- RTPReceiverAudio(const WebRtc_Word32 id);
- virtual ~RTPReceiverAudio();
-
- virtual void ChangeUniqueId(const WebRtc_Word32 id);
-
- WebRtc_Word32 Init();
-
- WebRtc_Word32 RegisterIncomingAudioCallback(RtpAudioFeedback* incomingMessagesCallback);
-
- ModuleRTPUtility::Payload* RegisterReceiveAudioPayload(
- const char payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate);
-
- WebRtc_UWord32 AudioFrequency() const;
-
- // Outband TelephoneEvent (DTMF) detection
- WebRtc_Word32 SetTelephoneEventStatus(const bool enable,
- const bool forwardToDecoder,
- const bool detectEndOfTone);
-
- // Is outband DTMF(AVT) turned on/off?
- bool TelephoneEvent() const ;
-
- // Is forwarding of outband telephone events turned on/off?
- bool TelephoneEventForwardToDecoder() const ;
-
- // Is TelephoneEvent configured with payload type payloadType
- bool TelephoneEventPayloadType(const WebRtc_Word8 payloadType) const;
-
- // Is CNG configured with payload type payloadType
- bool CNGPayloadType(const WebRtc_Word8 payloadType, WebRtc_UWord32& frequency);
-
- WebRtc_Word32 ParseAudioCodecSpecific(WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadLength,
- const ModuleRTPUtility::AudioPayload& audioSpecific,
- const bool isRED);
-
- virtual WebRtc_Word32 ResetStatistics() = 0;
-
-protected:
- virtual WebRtc_Word32 CallbackOfReceivedPayloadData(const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const WebRtcRTPHeader* rtpHeader) = 0;
-private:
- WebRtc_Word32 _id;
-
- WebRtc_UWord32 _lastReceivedFrequency;
-
- bool _telephoneEvent;
- bool _telephoneEventForwardToDecoder;
- bool _telephoneEventDetectEndOfTone;
- WebRtc_Word8 _telephoneEventPayloadType;
- std::set<WebRtc_UWord8> _telephoneEventReported;
-
- WebRtc_Word8 _cngNBPayloadType;
- WebRtc_Word8 _cngWBPayloadType;
- WebRtc_Word8 _cngSWBPayloadType;
- WebRtc_Word8 _cngPayloadType;
-
- // G722 is special since it use the wrong number of RTP samples in timestamp VS. number of samples in the frame
- WebRtc_Word8 _G722PayloadType;
- bool _lastReceivedG722;
-
- CriticalSectionWrapper* _criticalSectionFeedback;
- RtpAudioFeedback* _cbAudioFeedback;
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_AUDIO_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_receiver_video.cc b/trunk/src/modules/rtp_rtcp/source/rtp_receiver_video.cc
deleted file mode 100644
index d6a3c17..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_receiver_video.cc
+++ /dev/null
@@ -1,517 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rtp_receiver_video.h"
-
-#include <cassert> //assert
-#include <cstring> // memcpy()
-#include <math.h>
-
-#include "critical_section_wrapper.h"
-#include "receiver_fec.h"
-#include "rtp_rtcp_impl.h"
-#include "rtp_utility.h"
-
-namespace webrtc {
-WebRtc_UWord32 BitRateBPS(WebRtc_UWord16 x )
-{
- return (x & 0x3fff) * WebRtc_UWord32(pow(10.0f,(2 + (x >> 14))));
-}
-
-RTPReceiverVideo::RTPReceiverVideo():
- _id(0),
- _rtpRtcp(NULL),
- _criticalSectionFeedback(CriticalSectionWrapper::CreateCriticalSection()),
- _cbVideoFeedback(NULL),
- _criticalSectionReceiverVideo(
- CriticalSectionWrapper::CreateCriticalSection()),
- _completeFrame(false),
- _packetStartTimeMs(0),
- _receivedBW(),
- _estimatedBW(0),
- _currentFecFrameDecoded(false),
- _receiveFEC(NULL),
- _overUseDetector(),
- _videoBitRate(),
- _lastBitRateChange(0),
- _packetOverHead(28)
-{
- memset(_receivedBW, 0,sizeof(_receivedBW));
-}
-
-RTPReceiverVideo::RTPReceiverVideo(const WebRtc_Word32 id,
- ModuleRtpRtcpImpl* owner):
- _id(id),
- _rtpRtcp(owner),
- _criticalSectionFeedback(CriticalSectionWrapper::CreateCriticalSection()),
- _cbVideoFeedback(NULL),
- _criticalSectionReceiverVideo(
- CriticalSectionWrapper::CreateCriticalSection()),
- _completeFrame(false),
- _packetStartTimeMs(0),
- _receivedBW(),
- _estimatedBW(0),
- _currentFecFrameDecoded(false),
- _receiveFEC(NULL),
- _overUseDetector(),
- _videoBitRate(),
- _lastBitRateChange(0),
- _packetOverHead(28)
-{
- memset(_receivedBW, 0,sizeof(_receivedBW));
-}
-
-RTPReceiverVideo::~RTPReceiverVideo()
-{
- delete _criticalSectionFeedback;
- delete _criticalSectionReceiverVideo;
- delete _receiveFEC;
-}
-
-WebRtc_Word32
-RTPReceiverVideo::Init()
-{
- _completeFrame = false;
- _packetStartTimeMs = 0;
- _estimatedBW = 0;
- _currentFecFrameDecoded = false;
- _packetOverHead = 28;
- for (int i = 0; i < BW_HISTORY_SIZE; i++)
- {
- _receivedBW[i] = 0;
- }
- ResetOverUseDetector();
- return 0;
-}
-
-void
-RTPReceiverVideo::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
-}
-
-WebRtc_Word32
-RTPReceiverVideo::RegisterIncomingVideoCallback(RtpVideoFeedback* incomingMessagesCallback)
-{
- CriticalSectionScoped lock(_criticalSectionFeedback);
- _cbVideoFeedback = incomingMessagesCallback;
- return 0;
-}
-
-void
-RTPReceiverVideo::UpdateBandwidthManagement(const WebRtc_UWord32 bitrateBps,
- const WebRtc_UWord8 fractionLost,
- const WebRtc_UWord16 roundTripTimeMs)
-{
- CriticalSectionScoped lock(_criticalSectionFeedback);
- if(_cbVideoFeedback)
- {
- _cbVideoFeedback->OnNetworkChanged(_id,
- bitrateBps,
- fractionLost,
- roundTripTimeMs);
- }
-}
-
-ModuleRTPUtility::Payload* RTPReceiverVideo::RegisterReceiveVideoPayload(
- const char payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 maxRate) {
- RtpVideoCodecTypes videoType = kRtpNoVideo;
- if (ModuleRTPUtility::StringCompare(payloadName, "VP8", 3)) {
- videoType = kRtpVp8Video;
- } else if (ModuleRTPUtility::StringCompare(payloadName, "I420", 4)) {
- videoType = kRtpNoVideo;
- } else if (ModuleRTPUtility::StringCompare(payloadName, "ULPFEC", 6)) {
- // store this
- if (_receiveFEC == NULL) {
- _receiveFEC = new ReceiverFEC(_id, this);
- }
- _receiveFEC->SetPayloadTypeFEC(payloadType);
- videoType = kRtpFecVideo;
- } else {
- return NULL;
- }
- ModuleRTPUtility::Payload* payload = new ModuleRTPUtility::Payload;
-
- payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
- strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
- payload->typeSpecific.Video.videoCodecType = videoType;
- payload->typeSpecific.Video.maxRate = maxRate;
- payload->audio = false;
- return payload;
-}
-
-void RTPReceiverVideo::ResetOverUseDetector()
-{
- _overUseDetector.Reset();
- _videoBitRate.Init();
- _lastBitRateChange = 0;
-}
-
-// called under _criticalSectionReceiverVideo
-WebRtc_UWord16
-RTPReceiverVideo::EstimateBandwidth(const WebRtc_UWord16 bandwidth)
-{
- // received fragments
- // estimate BW
-
- WebRtc_UWord16 bwSort[BW_HISTORY_SIZE];
- for(int i = 0; i < BW_HISTORY_SIZE-1; i++)
- {
- _receivedBW[i] = _receivedBW[i+1];
- bwSort[i] = _receivedBW[i+1];
- }
- _receivedBW[BW_HISTORY_SIZE-1] = bandwidth;
- bwSort[BW_HISTORY_SIZE-1] = bandwidth;
-
- WebRtc_UWord16 temp;
- for (int i = BW_HISTORY_SIZE-1; i >= 0; i--)
- {
- for (int j = 1; j <= i; j++)
- {
- if (bwSort[j-1] > bwSort[j])
- {
- temp = bwSort[j-1];
- bwSort[j-1] = bwSort[j];
- bwSort[j] = temp;
- }
- }
- }
- int zeroCount = 0;
- for (; zeroCount < BW_HISTORY_SIZE; zeroCount++)
- {
- if (bwSort[zeroCount]!= 0)
- {
- break;
- }
- }
- WebRtc_UWord32 indexMedian = (BW_HISTORY_SIZE -1) - (BW_HISTORY_SIZE-zeroCount)/2;
- WebRtc_UWord16 bandwidthMedian = bwSort[indexMedian];
-
- if (bandwidthMedian > 0)
- {
- if (_estimatedBW == bandwidth)
- {
- // don't trigger a callback
- bandwidthMedian = 0;
- } else
- {
- _estimatedBW = bandwidthMedian;
- }
- } else
- {
- // can't be negative
- bandwidthMedian = 0;
- }
-
- return bandwidthMedian;
-}
-
-// we have no critext when calling this
-// we are not allowed to have any critsects when calling CallbackOfReceivedPayloadData
-WebRtc_Word32
-RTPReceiverVideo::ParseVideoCodecSpecific(WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadDataLength,
- const RtpVideoCodecTypes videoType,
- const bool isRED,
- const WebRtc_UWord8* incomingRtpPacket,
- const WebRtc_UWord16 incomingRtpPacketSize,
- const WebRtc_Word64 nowMS)
-{
- WebRtc_Word32 retVal = 0;
-
- _criticalSectionReceiverVideo->Enter();
-
- _videoBitRate.Update(payloadDataLength + rtpHeader->header.paddingLength,
- nowMS);
-
- // Add headers, ideally we would like to include for instance
- // Ethernet header here as well.
- const WebRtc_UWord16 packetSize = payloadDataLength + _packetOverHead +
- rtpHeader->header.headerLength + rtpHeader->header.paddingLength;
- _overUseDetector.Update(*rtpHeader, packetSize, nowMS);
-
- if (isRED)
- {
- if(_receiveFEC == NULL)
- {
- _criticalSectionReceiverVideo->Leave();
- return -1;
- }
- bool FECpacket = false;
- retVal = _receiveFEC->AddReceivedFECPacket(
- rtpHeader,
- incomingRtpPacket,
- payloadDataLength,
- FECpacket);
- if (retVal != -1)
- retVal = _receiveFEC->ProcessReceivedFEC();
- _criticalSectionReceiverVideo->Leave();
-
- if(retVal == 0 && FECpacket)
- {
- // Callback with the received FEC packet.
- // The normal packets are delivered after parsing.
- // This contains the original RTP packet header but with
- // empty payload and data length.
- rtpHeader->frameType = kFrameEmpty;
- // We need this for the routing.
- WebRtc_Word32 retVal = SetCodecType(videoType, rtpHeader);
- if(retVal != 0)
- {
- return retVal;
- }
- retVal = CallbackOfReceivedPayloadData(NULL, 0, rtpHeader);
- }
- }else
- {
- // will leave the _criticalSectionReceiverVideo critsect
- retVal = ParseVideoCodecSpecificSwitch(rtpHeader,
- payloadData,
- payloadDataLength,
- videoType);
- }
-
- // Update the remote rate control object and update the overuse
- // detector with the current rate control region.
- _criticalSectionReceiverVideo->Enter();
- const RateControlInput input(_overUseDetector.State(),
- _videoBitRate.BitRate(nowMS),
- _overUseDetector.NoiseVar());
- _criticalSectionReceiverVideo->Leave();
-
- // Call the callback outside critical section
- if (_rtpRtcp) {
- const RateControlRegion region = _rtpRtcp->OnOverUseStateUpdate(input);
-
- _criticalSectionReceiverVideo->Enter();
- _overUseDetector.SetRateControlRegion(region);
- _criticalSectionReceiverVideo->Leave();
- }
-
- return retVal;
-}
-
-WebRtc_Word32
-RTPReceiverVideo::BuildRTPheader(const WebRtcRTPHeader* rtpHeader,
- WebRtc_UWord8* dataBuffer) const
-{
- dataBuffer[0] = static_cast<WebRtc_UWord8>(0x80); // version 2
- dataBuffer[1] = static_cast<WebRtc_UWord8>(rtpHeader->header.payloadType);
- if (rtpHeader->header.markerBit)
- {
- dataBuffer[1] |= kRtpMarkerBitMask; // MarkerBit is 1
- }
-
- ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+2, rtpHeader->header.sequenceNumber);
- ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+4, rtpHeader->header.timestamp);
- ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, rtpHeader->header.ssrc);
-
- WebRtc_Word32 rtpHeaderLength = 12;
-
- // Add the CSRCs if any
- if (rtpHeader->header.numCSRCs > 0)
- {
- if(rtpHeader->header.numCSRCs > 16)
- {
- // error
- assert(false);
- }
- WebRtc_UWord8* ptr = &dataBuffer[rtpHeaderLength];
- for (WebRtc_UWord32 i = 0; i < rtpHeader->header.numCSRCs; ++i)
- {
- ModuleRTPUtility::AssignUWord32ToBuffer(ptr, rtpHeader->header.arrOfCSRCs[i]);
- ptr +=4;
- }
- dataBuffer[0] = (dataBuffer[0]&0xf0) | rtpHeader->header.numCSRCs;
-
- // Update length of header
- rtpHeaderLength += sizeof(WebRtc_UWord32)*rtpHeader->header.numCSRCs;
- }
- return rtpHeaderLength;
-}
-
-WebRtc_Word32
-RTPReceiverVideo::ReceiveRecoveredPacketCallback(WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadDataLength)
-{
- _criticalSectionReceiverVideo->Enter();
-
- _currentFecFrameDecoded = true;
-
- ModuleRTPUtility::Payload* payload = NULL;
- if (PayloadTypeToPayload(rtpHeader->header.payloadType, payload) != 0)
- {
- return -1;
- }
- // here we can re-create the original lost packet so that we can use it for the relay
- // we need to re-create the RED header too
- WebRtc_UWord8 recoveredPacket[IP_PACKET_SIZE];
- WebRtc_UWord16 rtpHeaderLength = (WebRtc_UWord16)BuildRTPheader(rtpHeader, recoveredPacket);
-
- const WebRtc_UWord8 REDForFECHeaderLength = 1;
-
- // replace pltype
- recoveredPacket[1] &= 0x80; // reset
- recoveredPacket[1] += REDPayloadType(); // replace with RED payload type
-
- // add RED header
- recoveredPacket[rtpHeaderLength] = rtpHeader->header.payloadType; // f-bit always 0
-
- memcpy(recoveredPacket + rtpHeaderLength + REDForFECHeaderLength, payloadData, payloadDataLength);
-
- return ParseVideoCodecSpecificSwitch(rtpHeader,
- payloadData,
- payloadDataLength,
- payload->typeSpecific.Video.videoCodecType);
-}
-
-WebRtc_Word32 RTPReceiverVideo::SetCodecType(const RtpVideoCodecTypes videoType,
- WebRtcRTPHeader* rtpHeader) const {
- switch (videoType) {
- case kRtpNoVideo:
- rtpHeader->type.Video.codec = kRTPVideoGeneric;
- break;
- case kRtpVp8Video:
- rtpHeader->type.Video.codec = kRTPVideoVP8;
- break;
- case kRtpFecVideo:
- rtpHeader->type.Video.codec = kRTPVideoFEC;
- break;
- }
- return 0;
-}
-
-WebRtc_Word32 RTPReceiverVideo::ParseVideoCodecSpecificSwitch(
- WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadDataLength,
- const RtpVideoCodecTypes videoType) {
- WebRtc_Word32 retVal = SetCodecType(videoType, rtpHeader);
- if (retVal != 0) {
- return retVal;
- }
- // All receive functions release _criticalSectionReceiverVideo before
- // returning.
- switch (videoType) {
- case kRtpNoVideo:
- return ReceiveGenericCodec(rtpHeader, payloadData, payloadDataLength);
- case kRtpVp8Video:
- return ReceiveVp8Codec(rtpHeader, payloadData, payloadDataLength);
- case kRtpFecVideo:
- break;
- }
- _criticalSectionReceiverVideo->Leave();
- return -1;
-}
-
-WebRtc_Word32
-RTPReceiverVideo::ReceiveVp8Codec(WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadDataLength)
-{
- bool success;
- ModuleRTPUtility::RTPPayload parsedPacket;
- if (payloadDataLength == 0)
- {
- success = true;
- parsedPacket.info.VP8.dataLength = 0;
- } else
- {
- ModuleRTPUtility::RTPPayloadParser rtpPayloadParser(kRtpVp8Video,
- payloadData,
- payloadDataLength,
- _id);
-
- success = rtpPayloadParser.Parse(parsedPacket);
- }
- // from here down we only work on local data
- _criticalSectionReceiverVideo->Leave();
-
- if (!success)
- {
- return -1;
- }
- if (parsedPacket.info.VP8.dataLength == 0)
- {
- // we have an "empty" VP8 packet, it's ok, could be one way video
- // Inform the jitter buffer about this packet.
- rtpHeader->frameType = kFrameEmpty;
- if (CallbackOfReceivedPayloadData(NULL, 0, rtpHeader) != 0)
- {
- return -1;
- }
- return 0;
- }
- rtpHeader->frameType = (parsedPacket.frameType == ModuleRTPUtility::kIFrame) ? kVideoFrameKey : kVideoFrameDelta;
-
- RTPVideoHeaderVP8 *toHeader = &rtpHeader->type.Video.codecHeader.VP8;
- ModuleRTPUtility::RTPPayloadVP8 *fromHeader = &parsedPacket.info.VP8;
-
- rtpHeader->type.Video.isFirstPacket = fromHeader->beginningOfPartition
- && (fromHeader->partitionID == 0);
- toHeader->pictureId = fromHeader->hasPictureID ? fromHeader->pictureID :
- kNoPictureId;
- toHeader->tl0PicIdx = fromHeader->hasTl0PicIdx ? fromHeader->tl0PicIdx :
- kNoTl0PicIdx;
- if (fromHeader->hasTID) {
- toHeader->temporalIdx = fromHeader->tID;
- toHeader->layerSync = fromHeader->layerSync;
- } else {
- toHeader->temporalIdx = kNoTemporalIdx;
- toHeader->layerSync = false;
- }
- toHeader->keyIdx = fromHeader->hasKeyIdx ? fromHeader->keyIdx : kNoKeyIdx;
-
- toHeader->frameWidth = fromHeader->frameWidth;
- toHeader->frameHeight = fromHeader->frameHeight;
-
- toHeader->partitionId = fromHeader->partitionID;
- toHeader->beginningOfPartition = fromHeader->beginningOfPartition;
-
- if(CallbackOfReceivedPayloadData(parsedPacket.info.VP8.data,
- parsedPacket.info.VP8.dataLength,
- rtpHeader) != 0)
- {
- return -1;
- }
- return 0;
-}
-
-
-WebRtc_Word32
-RTPReceiverVideo::ReceiveGenericCodec(WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadDataLength)
-{
- rtpHeader->frameType = kVideoFrameKey;
-
- if(((SequenceNumber() + 1) == rtpHeader->header.sequenceNumber) &&
- (TimeStamp() != rtpHeader->header.timestamp))
- {
- rtpHeader->type.Video.isFirstPacket = true;
- }
- _criticalSectionReceiverVideo->Leave();
-
- if(CallbackOfReceivedPayloadData(payloadData, payloadDataLength, rtpHeader) != 0)
- {
- return -1;
- }
- return 0;
-}
-
-void RTPReceiverVideo::SetPacketOverHead(WebRtc_UWord16 packetOverHead)
-{
- _packetOverHead = packetOverHead;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_receiver_video.h b/trunk/src/modules/rtp_rtcp/source/rtp_receiver_video.h
deleted file mode 100644
index 5e0138a..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_receiver_video.h
+++ /dev/null
@@ -1,139 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
-
-#include "rtp_rtcp_defines.h"
-#include "rtp_utility.h"
-
-#include "typedefs.h"
-
-#include "overuse_detector.h"
-#include "remote_rate_control.h"
-#include "Bitrate.h"
-
-namespace webrtc {
-class ReceiverFEC;
-class ModuleRtpRtcpImpl;
-class CriticalSectionWrapper;
-
-class RTPReceiverVideo
-{
-public:
- RTPReceiverVideo();
- RTPReceiverVideo(const WebRtc_Word32 id, ModuleRtpRtcpImpl* owner);
-
- virtual ~RTPReceiverVideo();
-
- virtual void ChangeUniqueId(const WebRtc_Word32 id);
-
- WebRtc_Word32 Init();
-
- WebRtc_Word32 RegisterIncomingVideoCallback(
- RtpVideoFeedback* incomingMessagesCallback);
-
- void UpdateBandwidthManagement(const WebRtc_UWord32 bitrateBps,
- const WebRtc_UWord8 fractionLost,
- const WebRtc_UWord16 roundTripTimeMs);
-
- ModuleRTPUtility::Payload* RegisterReceiveVideoPayload(
- const char payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 maxRate);
-
- WebRtc_Word32 ParseVideoCodecSpecific(
- WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadDataLength,
- const RtpVideoCodecTypes videoType,
- const bool isRED,
- const WebRtc_UWord8* incomingRtpPacket,
- const WebRtc_UWord16 incomingRtpPacketSize,
- const WebRtc_Word64 nowMS);
-
- virtual WebRtc_Word32 ReceiveRecoveredPacketCallback(
- WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadDataLength);
-
- void SetPacketOverHead(WebRtc_UWord16 packetOverHead);
-
-protected:
- void ResetOverUseDetector();
-
- WebRtc_UWord16 EstimateBandwidth( const WebRtc_UWord16 bufferLength);
-
- virtual WebRtc_Word32 CallbackOfReceivedPayloadData(
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const WebRtcRTPHeader* rtpHeader) = 0;
-
- virtual WebRtc_UWord32 TimeStamp() const = 0;
- virtual WebRtc_UWord16 SequenceNumber() const = 0;
-
- virtual WebRtc_UWord32 PayloadTypeToPayload(
- const WebRtc_UWord8 payloadType,
- ModuleRTPUtility::Payload*& payload) const = 0;
-
- virtual bool RetransmitOfOldPacket(
- const WebRtc_UWord16 sequenceNumber,
- const WebRtc_UWord32 rtpTimeStamp) const = 0;
-
- virtual WebRtc_Word8 REDPayloadType() const = 0;
-
- WebRtc_Word32 SetCodecType(const RtpVideoCodecTypes videoType,
- WebRtcRTPHeader* rtpHeader) const;
-
- WebRtc_Word32 ParseVideoCodecSpecificSwitch(
- WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadDataLength,
- const RtpVideoCodecTypes videoType);
-
- WebRtc_Word32 ReceiveGenericCodec(WebRtcRTPHeader *rtpHeader,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadDataLength);
-
- WebRtc_Word32 ReceiveVp8Codec(WebRtcRTPHeader *rtpHeader,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadDataLength);
-
- WebRtc_Word32 BuildRTPheader(const WebRtcRTPHeader* rtpHeader,
- WebRtc_UWord8* dataBuffer) const;
-
-private:
- WebRtc_Word32 _id;
- ModuleRtpRtcpImpl* _rtpRtcp;
-
- CriticalSectionWrapper* _criticalSectionFeedback;
- RtpVideoFeedback* _cbVideoFeedback;
-
- CriticalSectionWrapper* _criticalSectionReceiverVideo;
-
- // bandwidth
- bool _completeFrame;
- WebRtc_UWord32 _packetStartTimeMs;
- WebRtc_UWord16 _receivedBW[BW_HISTORY_SIZE];
- WebRtc_UWord16 _estimatedBW;
-
- // FEC
- bool _currentFecFrameDecoded;
- ReceiverFEC* _receiveFEC;
-
- // BWE
- OverUseDetector _overUseDetector;
- BitRateStats _videoBitRate;
- WebRtc_Word64 _lastBitRateChange;
- WebRtc_UWord16 _packetOverHead;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RECEIVER_VIDEO_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_rtcp.gypi b/trunk/src/modules/rtp_rtcp/source/rtp_rtcp.gypi
deleted file mode 100644
index 48fc05f..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_rtcp.gypi
+++ /dev/null
@@ -1,102 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'rtp_rtcp',
- 'type': '<(library)',
- 'dependencies': [
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '../interface',
- '../../interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- '../../interface',
- ],
- },
- 'sources': [
- # Common
- '../interface/rtp_rtcp.h',
- '../interface/rtp_rtcp_defines.h',
- 'bitrate.cc',
- 'Bitrate.h',
- 'rtp_rtcp_config.h',
- 'rtp_rtcp_impl.cc',
- 'rtp_rtcp_impl.h',
- 'rtcp_receiver.cc',
- 'rtcp_receiver.h',
- 'rtcp_receiver_help.cc',
- 'rtcp_receiver_help.h',
- 'rtcp_sender.cc',
- 'rtcp_sender.h',
- 'rtcp_utility.cc',
- 'rtcp_utility.h',
- 'rtp_header_extension.cc',
- 'rtp_header_extension.h',
- 'rtp_receiver.cc',
- 'rtp_receiver.h',
- 'rtp_sender.cc',
- 'rtp_sender.h',
- 'rtp_utility.cc',
- 'rtp_utility.h',
- 'ssrc_database.cc',
- 'ssrc_database.h',
- 'tmmbr_help.cc',
- 'tmmbr_help.h',
- # Audio Files
- 'dtmf_queue.cc',
- 'dtmf_queue.h',
- 'rtp_receiver_audio.cc',
- 'rtp_receiver_audio.h',
- 'rtp_sender_audio.cc',
- 'rtp_sender_audio.h',
- # Video Files
- 'bandwidth_management.cc',
- 'bandwidth_management.h',
- 'bwe_defines.h',
- 'fec_private_tables.h',
- 'forward_error_correction.cc',
- 'forward_error_correction.h',
- 'forward_error_correction_internal.cc',
- 'forward_error_correction_internal.h',
- 'overuse_detector.cc',
- 'overuse_detector.h',
- 'remote_rate_control.cc',
- 'remote_rate_control.h',
- 'rtp_packet_history.cc',
- 'rtp_packet_history.h',
- 'rtp_receiver_video.cc',
- 'rtp_receiver_video.h',
- 'rtp_sender_video.cc',
- 'rtp_sender_video.h',
- 'receiver_fec.cc',
- 'receiver_fec.h',
- 'video_codec_information.h',
- 'rtp_format_vp8.cc',
- 'rtp_format_vp8.h',
- 'transmission_bucket.cc',
- 'transmission_bucket.h',
- 'vp8_partition_aggregator.cc',
- 'vp8_partition_aggregator.h',
- # Mocks
- '../mocks/mock_rtp_rtcp.h',
- ], # source
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_config.h b/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_config.h
deleted file mode 100644
index a0eed88..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_config.h
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
-
-// Configuration file for RTP utilities (RTPSender, RTPReceiver ...)
-namespace webrtc {
-enum { kRtpRtcpMaxIdleTimeProcess = 5,
- kRtpRtcpBitrateProcessTimeMs = 10,
- kRtpRtcpPacketTimeoutProcessTimeMs = 100 };
-
-enum { NACK_PACKETS_MAX_SIZE = 256 }; // in packets
-enum { NACK_BYTECOUNT_SIZE = 60}; // size of our NACK history
-
-enum { RTCP_INTERVAL_VIDEO_MS = 1000 };
-enum { RTCP_INTERVAL_AUDIO_MS = 5000 };
-enum { RTCP_SEND_BEFORE_KEY_FRAME_MS= 100 };
-enum { RTCP_MAX_REPORT_BLOCKS = 31}; // RFC 3550 page 37
-enum { RTCP_MIN_FRAME_LENGTH_MS = 17};
-enum { kRtcpAppCode_DATA_SIZE = 32*4}; // multiple of 4, this is not a limitation of the size
-enum { RTCP_RPSI_DATA_SIZE = 30};
-enum { RTCP_NUMBER_OF_SR = 60 };
-
-enum { MAX_NUMBER_OF_TEMPORAL_ID = 8 }; // RFC
-enum { MAX_NUMBER_OF_DEPENDENCY_QUALITY_ID = 128 };// RFC
-
-enum { BW_HISTORY_SIZE = 35};
-
-#define MIN_AUDIO_BW_MANAGEMENT_BITRATE 6
-#define MIN_VIDEO_BW_MANAGEMENT_BITRATE 30
-
-enum { DTMF_OUTBAND_MAX = 20};
-
-enum { RTP_MAX_BURST_SLEEP_TIME = 500 };
-enum { RTP_AUDIO_LEVEL_UNIQUE_ID = 0xbede };
-enum { RTP_MAX_PACKETS_PER_FRAME= 512 }; // must be multiple of 32
-} // namespace webrtc
-
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_CONFIG_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc b/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
deleted file mode 100644
index 08da8f8..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_impl.cc
+++ /dev/null
@@ -1,2971 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "common_types.h"
-#include "rtp_rtcp_impl.h"
-#include "trace.h"
-
-#ifdef MATLAB
-#include "../test/BWEStandAlone/MatlabPlot.h"
-extern MatlabEngine eng; // global variable defined elsewhere
-#endif
-
-#include <string.h> //memcpy
-#include <cassert> //assert
-
-// local for this file
-namespace {
-
-const float FracMS = 4.294967296E6f;
-
-} // namepace
-
-#ifdef _WIN32
-// disable warning C4355: 'this' : used in base member initializer list
-#pragma warning(disable : 4355)
-#endif
-
-namespace webrtc {
-
-const WebRtc_UWord16 kDefaultRtt = 200;
-
-RtpRtcp* RtpRtcp::CreateRtpRtcp(const WebRtc_Word32 id,
- bool audio) {
- if(audio) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id, "CreateRtpRtcp(audio)");
- } else {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, id, "CreateRtpRtcp(video)");
- }
- // ModuleRTPUtility::GetSystemClock() creates a new instance of a system
- // clock implementation. The OwnsClock() function informs the module that
- // it is responsible for deleting the instance.
- ModuleRtpRtcpImpl* rtp_rtcp_instance = new ModuleRtpRtcpImpl(id,
- audio, ModuleRTPUtility::GetSystemClock());
- rtp_rtcp_instance->OwnsClock();
- return rtp_rtcp_instance;
-}
-
-RtpRtcp* RtpRtcp::CreateRtpRtcp(const WebRtc_Word32 id,
- const bool audio,
- RtpRtcpClock* clock) {
- if (audio) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- id,
- "CreateRtpRtcp(audio)");
- } else {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- id,
- "CreateRtpRtcp(video)");
- }
- return new ModuleRtpRtcpImpl(id, audio, clock);
-}
-
-void RtpRtcp::DestroyRtpRtcp(RtpRtcp* module) {
- if (module) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- static_cast<ModuleRtpRtcpImpl*>(module)->Id(),
- "DestroyRtpRtcp()");
- delete static_cast<ModuleRtpRtcpImpl*>(module);
- }
-}
-
-ModuleRtpRtcpImpl::ModuleRtpRtcpImpl(const WebRtc_Word32 id,
- const bool audio,
- RtpRtcpClock* clock):
- TMMBRHelp(audio),
- _rtpSender(id, audio, clock),
- _rtpReceiver(id, audio, clock, this),
- _rtcpSender(id, audio, clock, this),
- _rtcpReceiver(id, clock, this),
- _owns_clock(false),
- _clock(*clock),
- _id(id),
- _audio(audio),
- _collisionDetected(false),
- _lastProcessTime(clock->GetTimeInMS()),
- _lastBitrateProcessTime(clock->GetTimeInMS()),
- _lastPacketTimeoutProcessTime(clock->GetTimeInMS()),
- _packetOverHead(28), // IPV4 UDP
- _criticalSectionModulePtrs(CriticalSectionWrapper::CreateCriticalSection()),
- _criticalSectionModulePtrsFeedback(
- CriticalSectionWrapper::CreateCriticalSection()),
- _defaultModule(NULL),
- _audioModule(NULL),
- _videoModule(NULL),
- _deadOrAliveActive(false),
- _deadOrAliveTimeoutMS(0),
- _deadOrAliveLastTimer(0),
- _bandwidthManagement(id),
- _receivedNTPsecsAudio(0),
- _receivedNTPfracAudio(0),
- _RTCPArrivalTimeSecsAudio(0),
- _RTCPArrivalTimeFracAudio(0),
- _nackMethod(kNackOff),
- _nackLastTimeSent(0),
- _nackLastSeqNumberSent(0),
- _simulcast(false),
- _keyFrameReqMethod(kKeyFrameReqFirRtp)
-#ifdef MATLAB
- , _plot1(NULL)
-#endif
-{
- _sendVideoCodec.codecType = kVideoCodecUnknown;
- // make sure that RTCP objects are aware of our SSRC
- WebRtc_UWord32 SSRC = _rtpSender.SSRC();
- _rtcpSender.SetSSRC(SSRC);
-
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
-}
-
-ModuleRtpRtcpImpl::~ModuleRtpRtcpImpl() {
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__);
-
- // make sure to unregister this module from other modules
-
- const bool defaultInstance(_childModules.empty() ? false : true);
-
- if (defaultInstance) {
- // deregister for the default module
- // will go in to the child modules and remove it self
- std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
- while (it != _childModules.end()) {
- RtpRtcp* module = *it;
- _childModules.erase(it);
- if (module) {
- module->DeRegisterDefaultModule();
- }
- it = _childModules.begin();
- }
- } else {
- // deregister for the child modules
- // will go in to the default and remove it self
- DeRegisterDefaultModule();
- }
-
- if (_audio) {
- DeRegisterVideoModule();
- } else {
- DeRegisterSyncModule();
- }
-
-#ifdef MATLAB
- if (_plot1) {
- eng.DeletePlot(_plot1);
- _plot1 = NULL;
- }
-#endif
-
- delete _criticalSectionModulePtrs;
- delete _criticalSectionModulePtrsFeedback;
- if (_owns_clock) {
- delete &_clock;
- }
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::ChangeUniqueId(const WebRtc_Word32 id) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "ChangeUniqueId(new id:%d)", id);
-
- _id = id;
-
- _rtpReceiver.ChangeUniqueId(id);
- _rtcpReceiver.ChangeUniqueId(id);
- _rtpSender.ChangeUniqueId(id);
- _rtcpSender.ChangeUniqueId(id);
- return 0;
-}
-
-// default encoder that we need to multiplex out
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterDefaultModule(RtpRtcp* module) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterDefaultModule(module:0x%x)", module);
-
- if (module == NULL) {
- return -1;
- }
- if (module == this) {
- WEBRTC_TRACE(kTraceError,
- kTraceRtpRtcp,
- _id,
- "RegisterDefaultModule can't register self as default");
- return -1;
- }
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
-
- if (_defaultModule) {
- _defaultModule->DeRegisterChildModule(this);
- }
- _defaultModule = (ModuleRtpRtcpImpl*)module;
- _defaultModule->RegisterChildModule(this);
- return 0;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::DeRegisterDefaultModule() {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "DeRegisterDefaultModule()");
-
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- if (_defaultModule) {
- _defaultModule->DeRegisterChildModule(this);
- _defaultModule = NULL;
- }
- return 0;
-}
-
-bool ModuleRtpRtcpImpl::DefaultModuleRegistered() {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "DefaultModuleRegistered()");
-
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- if (_defaultModule) {
- return true;
- }
- return false;
-}
-
-WebRtc_UWord32 ModuleRtpRtcpImpl::NumberChildModules() {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "NumberChildModules");
-
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- CriticalSectionScoped doubleLock(_criticalSectionModulePtrsFeedback);
- // we use two locks for protecting _childModules one
- // (_criticalSectionModulePtrsFeedback) for incoming messages
- // (BitrateSent and UpdateTMMBR) and _criticalSectionModulePtrs for
- // all outgoing messages sending packets etc
-
- return _childModules.size();
-}
-
-void ModuleRtpRtcpImpl::RegisterChildModule(RtpRtcp* module) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterChildModule(module:0x%x)",
- module);
-
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
-
- CriticalSectionScoped doubleLock(_criticalSectionModulePtrsFeedback);
- // we use two locks for protecting _childModules one
- // (_criticalSectionModulePtrsFeedback) for incoming
- // messages (BitrateSent and UpdateTMMBR) and _criticalSectionModulePtrs
- // for all outgoing messages sending packets etc
- _childModules.push_back((ModuleRtpRtcpImpl*)module);
-}
-
-void ModuleRtpRtcpImpl::DeRegisterChildModule(RtpRtcp* removeModule) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "DeRegisterChildModule(module:0x%x)", removeModule);
-
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
-
- CriticalSectionScoped doubleLock(_criticalSectionModulePtrsFeedback);
-
- std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
- while (it != _childModules.end()) {
- RtpRtcp* module = *it;
- if (module == removeModule) {
- _childModules.erase(it);
- return;
- }
- it++;
- }
-}
-
-// Lip-sync between voice-video engine,
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterSyncModule(RtpRtcp* audioModule) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterSyncModule(module:0x%x)",
- audioModule);
-
- if (audioModule == NULL) {
- return -1;
- }
- if (_audio) {
- return -1;
- }
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- _audioModule = (ModuleRtpRtcpImpl*)audioModule;
- return _audioModule->RegisterVideoModule(this);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::DeRegisterSyncModule() {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "DeRegisterSyncModule()");
-
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- if (_audioModule) {
- ModuleRtpRtcpImpl* audioModule = _audioModule;
- _audioModule = NULL;
- _receivedNTPsecsAudio = 0;
- _receivedNTPfracAudio = 0;
- _RTCPArrivalTimeSecsAudio = 0;
- _RTCPArrivalTimeFracAudio = 0;
- audioModule->DeRegisterVideoModule();
- }
- return 0;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterVideoModule(RtpRtcp* videoModule) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterVideoModule(module:0x%x)",
- videoModule);
-
- if (videoModule == NULL) {
- return -1;
- }
- if (!_audio) {
- return -1;
- }
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- _videoModule = (ModuleRtpRtcpImpl*)videoModule;
- return 0;
-}
-
-void ModuleRtpRtcpImpl::DeRegisterVideoModule() {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "DeRegisterVideoModule()");
-
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- if (_videoModule) {
- ModuleRtpRtcpImpl* videoModule = _videoModule;
- _videoModule = NULL;
- videoModule->DeRegisterSyncModule();
- }
-}
-
-// returns the number of milliseconds until the module want a worker thread
-// to call Process
-WebRtc_Word32 ModuleRtpRtcpImpl::TimeUntilNextProcess() {
- const WebRtc_UWord32 now = _clock.GetTimeInMS();
- return kRtpRtcpMaxIdleTimeProcess - (now - _lastProcessTime);
-}
-
-// Process any pending tasks such as timeouts
-// non time critical events
-WebRtc_Word32 ModuleRtpRtcpImpl::Process() {
- const WebRtc_UWord32 now = _clock.GetTimeInMS();
- _lastProcessTime = now;
-
- _rtpSender.ProcessSendToNetwork();
-
- if (now >= _lastPacketTimeoutProcessTime +
- kRtpRtcpPacketTimeoutProcessTimeMs) {
- _rtpReceiver.PacketTimeout();
- _rtcpReceiver.PacketTimeout();
- _lastPacketTimeoutProcessTime = now;
- }
-
- if (now >= _lastBitrateProcessTime + kRtpRtcpBitrateProcessTimeMs) {
- _rtpSender.ProcessBitrate();
- _rtpReceiver.ProcessBitrate();
- _lastBitrateProcessTime = now;
- }
-
- ProcessDeadOrAliveTimer();
-
- const bool defaultInstance(_childModules.empty() ? false : true);
- if (!defaultInstance && _rtcpSender.TimeToSendRTCPReport()) {
- WebRtc_UWord16 max_rtt = 0;
- if (_rtcpSender.Sending()) {
- std::vector<RTCPReportBlock> receive_blocks;
- _rtcpReceiver.StatisticsReceived(&receive_blocks);
- for (std::vector<RTCPReportBlock>::iterator it = receive_blocks.begin();
- it != receive_blocks.end(); ++it) {
- WebRtc_UWord16 rtt = 0;
- _rtcpReceiver.RTT(it->remoteSSRC, &max_rtt, NULL, NULL, NULL);
- max_rtt = (rtt > max_rtt) ? rtt : max_rtt;
- }
- } else {
- // We're only receiving, i.e. this module doesn't have its own RTT
- // estimate. Use the RTT set by a sending channel using the same default
- // module.
- max_rtt = _rtcpReceiver.RTT();
- }
- if (max_rtt == 0) {
- // No valid estimate available, i.e. no sending channel using the same
- // default module or no RTCP received yet.
- max_rtt = kDefaultRtt;
- }
- if (REMB() && _rtcpSender.ValidBitrateEstimate()) {
- unsigned int target_bitrate =
- _rtcpSender.CalculateNewTargetBitrate(max_rtt);
- _rtcpSender.UpdateRemoteBitrateEstimate(target_bitrate);
- } else if (TMMBR()) {
- _rtcpSender.CalculateNewTargetBitrate(max_rtt);
- }
- _rtcpSender.SendRTCP(kRtcpReport);
- }
-
- if (_rtpSender.RTPKeepalive()) {
- // check time to send RTP keep alive
- if (_rtpSender.TimeToSendRTPKeepalive()) {
- _rtpSender.SendRTPKeepalivePacket();
- }
- }
-
- if (UpdateRTCPReceiveInformationTimers()) {
- // a receiver has timed out
- UpdateTMMBR();
- }
- return 0;
-}
-
-/**
-* Receiver
-*/
-
-WebRtc_Word32 ModuleRtpRtcpImpl::InitReceiver() {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "InitReceiver()");
-
- _packetOverHead = 28; // default is IPV4 UDP
- _receivedNTPsecsAudio = 0;
- _receivedNTPfracAudio = 0;
- _RTCPArrivalTimeSecsAudio = 0;
- _RTCPArrivalTimeFracAudio = 0;
-
- WebRtc_Word32 ret = _rtpReceiver.Init();
- if (ret < 0) {
- return ret;
- }
- _rtpReceiver.SetPacketOverHead(_packetOverHead);
- return ret;
-}
-
-void ModuleRtpRtcpImpl::ProcessDeadOrAliveTimer() {
- if (_deadOrAliveActive) {
- const WebRtc_UWord32 now = _clock.GetTimeInMS();
- if (now > _deadOrAliveTimeoutMS + _deadOrAliveLastTimer) {
- // RTCP is alive if we have received a report the last 12 seconds
- _deadOrAliveLastTimer += _deadOrAliveTimeoutMS;
-
- bool RTCPalive = false;
- if (_rtcpReceiver.LastReceived() + 12000 > now) {
- RTCPalive = true;
- }
- _rtpReceiver.ProcessDeadOrAlive(RTCPalive, now);
- }
- }
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetPeriodicDeadOrAliveStatus(
- const bool enable,
- const WebRtc_UWord8 sampleTimeSeconds) {
- if (enable) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetPeriodicDeadOrAliveStatus(enable, %d)",
- sampleTimeSeconds);
- } else {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetPeriodicDeadOrAliveStatus(disable)");
- }
- if (sampleTimeSeconds == 0) {
- return -1;
- }
- _deadOrAliveActive = enable;
- _deadOrAliveTimeoutMS = sampleTimeSeconds * 1000;
- // trigger the first after one period
- _deadOrAliveLastTimer = _clock.GetTimeInMS();
- return 0;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::PeriodicDeadOrAliveStatus(
- bool& enable,
- WebRtc_UWord8& sampleTimeSeconds) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "PeriodicDeadOrAliveStatus()");
-
- enable = _deadOrAliveActive;
- sampleTimeSeconds = (WebRtc_UWord8)(_deadOrAliveTimeoutMS / 1000);
- return 0;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetPacketTimeout(
- const WebRtc_UWord32 RTPtimeoutMS,
- const WebRtc_UWord32 RTCPtimeoutMS) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetPacketTimeout(%u,%u)",
- RTPtimeoutMS,
- RTCPtimeoutMS);
-
- if (_rtpReceiver.SetPacketTimeout(RTPtimeoutMS) == 0) {
- return _rtcpReceiver.SetPacketTimeout(RTCPtimeoutMS);
- }
- return -1;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterReceivePayload(
- const CodecInst& voiceCodec) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterReceivePayload(voiceCodec)");
-
- return _rtpReceiver.RegisterReceivePayload(
- voiceCodec.plname,
- voiceCodec.pltype,
- voiceCodec.plfreq,
- voiceCodec.channels,
- (voiceCodec.rate < 0) ? 0 : voiceCodec.rate);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterReceivePayload(
- const VideoCodec& videoCodec) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterReceivePayload(videoCodec)");
-
- return _rtpReceiver.RegisterReceivePayload(videoCodec.plName,
- videoCodec.plType,
- 90000,
- 0,
- videoCodec.maxBitrate);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::ReceivePayloadType(
- const CodecInst& voiceCodec,
- WebRtc_Word8* plType) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "ReceivePayloadType(voiceCodec)");
-
- return _rtpReceiver.ReceivePayloadType(
- voiceCodec.plname,
- voiceCodec.plfreq,
- voiceCodec.channels,
- (voiceCodec.rate < 0) ? 0 : voiceCodec.rate,
- plType);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::ReceivePayloadType(
- const VideoCodec& videoCodec,
- WebRtc_Word8* plType) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "ReceivePayloadType(videoCodec)");
-
- return _rtpReceiver.ReceivePayloadType(videoCodec.plName,
- 90000,
- 0,
- videoCodec.maxBitrate,
- plType);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::DeRegisterReceivePayload(
- const WebRtc_Word8 payloadType) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "DeRegisterReceivePayload(%d)",
- payloadType);
-
- return _rtpReceiver.DeRegisterReceivePayload(payloadType);
-}
-
-// get the currently configured SSRC filter
-WebRtc_Word32 ModuleRtpRtcpImpl::SSRCFilter(WebRtc_UWord32& allowedSSRC) const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SSRCFilter()");
-
- return _rtpReceiver.SSRCFilter(allowedSSRC);
-}
-
-// set a SSRC to be used as a filter for incoming RTP streams
-WebRtc_Word32 ModuleRtpRtcpImpl::SetSSRCFilter(
- const bool enable,
- const WebRtc_UWord32 allowedSSRC) {
- if (enable) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetSSRCFilter(enable, 0x%x)",
- allowedSSRC);
- } else {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetSSRCFilter(disable)");
- }
-
- return _rtpReceiver.SetSSRCFilter(enable, allowedSSRC);
-}
-
-// Get last received remote timestamp
-WebRtc_UWord32 ModuleRtpRtcpImpl::RemoteTimestamp() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoteTimestamp()");
-
- return _rtpReceiver.TimeStamp();
-}
-
-// Get the current estimated remote timestamp
-WebRtc_Word32 ModuleRtpRtcpImpl::EstimatedRemoteTimeStamp(
- WebRtc_UWord32& timestamp) const {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "EstimatedRemoteTimeStamp()");
-
- return _rtpReceiver.EstimatedRemoteTimeStamp(timestamp);
-}
-
-// Get incoming SSRC
-WebRtc_UWord32 ModuleRtpRtcpImpl::RemoteSSRC() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoteSSRC()");
-
- return _rtpReceiver.SSRC();
-}
-
-// Get remote CSRC
-WebRtc_Word32 ModuleRtpRtcpImpl::RemoteCSRCs(
- WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoteCSRCs()");
-
- return _rtpReceiver.CSRCs(arrOfCSRC);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetRTXSendStatus(
- const bool enable,
- const bool setSSRC,
- const WebRtc_UWord32 SSRC) {
- _rtpSender.SetRTXStatus(enable, setSSRC, SSRC);
- return 0;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RTXSendStatus(bool* enable,
- WebRtc_UWord32* SSRC) const {
- _rtpSender.RTXStatus(enable, SSRC);
- return 0;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetRTXReceiveStatus(
- const bool enable,
- const WebRtc_UWord32 SSRC) {
- _rtpReceiver.SetRTXStatus(enable, SSRC);
- return 0;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RTXReceiveStatus(bool* enable,
- WebRtc_UWord32* SSRC) const {
- _rtpReceiver.RTXStatus(enable, SSRC);
- return 0;
-}
-
-// called by the network module when we receive a packet
-WebRtc_Word32 ModuleRtpRtcpImpl::IncomingPacket(
- const WebRtc_UWord8* incomingPacket,
- const WebRtc_UWord16 incomingPacketLength) {
- WEBRTC_TRACE(kTraceStream,
- kTraceRtpRtcp,
- _id,
- "IncomingPacket(packetLength:%u)",
- incomingPacketLength);
-
- // minimum RTP is 12 bytes
- // minimum RTCP is 8 bytes (RTCP BYE)
- if (incomingPacketLength < 8 || incomingPacket == NULL) {
- WEBRTC_TRACE(kTraceDebug,
- kTraceRtpRtcp,
- _id,
- "IncomingPacket invalid buffer or length");
- return -1;
- }
- // check RTP version
- const WebRtc_UWord8 version = incomingPacket[0] >> 6 ;
- if (version != 2) {
- WEBRTC_TRACE(kTraceDebug,
- kTraceRtpRtcp,
- _id,
- "IncomingPacket invalid RTP version");
- return -1;
- }
-
- ModuleRTPUtility::RTPHeaderParser rtpParser(incomingPacket,
- incomingPacketLength);
-
- if (rtpParser.RTCP()) {
- // Allow receive of non-compound RTCP packets.
- RTCPUtility::RTCPParserV2 rtcpParser(incomingPacket,
- incomingPacketLength,
- true);
-
- const bool validRTCPHeader = rtcpParser.IsValid();
- if (!validRTCPHeader) {
- WEBRTC_TRACE(kTraceDebug,
- kTraceRtpRtcp,
- _id,
- "IncomingPacket invalid RTCP packet");
- return -1;
- }
- RTCPHelp::RTCPPacketInformation rtcpPacketInformation;
- WebRtc_Word32 retVal = _rtcpReceiver.IncomingRTCPPacket(
- rtcpPacketInformation,
- &rtcpParser);
- if (retVal == 0) {
- _rtcpReceiver.TriggerCallbacksFromRTCPPacket(rtcpPacketInformation);
- }
- return retVal;
-
- } else {
- WebRtcRTPHeader rtpHeader;
- memset(&rtpHeader, 0, sizeof(rtpHeader));
-
- RtpHeaderExtensionMap map;
- _rtpReceiver.GetHeaderExtensionMapCopy(&map);
-
- const bool validRTPHeader = rtpParser.Parse(rtpHeader, &map);
- if (!validRTPHeader) {
- WEBRTC_TRACE(kTraceDebug,
- kTraceRtpRtcp,
- _id,
- "IncomingPacket invalid RTP header");
- return -1;
- }
- return _rtpReceiver.IncomingRTPPacket(&rtpHeader,
- incomingPacket,
- incomingPacketLength);
- }
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::IncomingAudioNTP(
- const WebRtc_UWord32 audioReceivedNTPsecs,
- const WebRtc_UWord32 audioReceivedNTPfrac,
- const WebRtc_UWord32 audioRTCPArrivalTimeSecs,
- const WebRtc_UWord32 audioRTCPArrivalTimeFrac) {
- _receivedNTPsecsAudio = audioReceivedNTPsecs;
- _receivedNTPfracAudio = audioReceivedNTPfrac;
- _RTCPArrivalTimeSecsAudio = audioRTCPArrivalTimeSecs;
- _RTCPArrivalTimeFracAudio = audioRTCPArrivalTimeFrac;
- return 0;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterIncomingDataCallback(
- RtpData* incomingDataCallback) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterIncomingDataCallback(incomingDataCallback:0x%x)",
- incomingDataCallback);
-
- return _rtpReceiver.RegisterIncomingDataCallback(incomingDataCallback);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterIncomingRTPCallback(
- RtpFeedback* incomingMessagesCallback) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterIncomingRTPCallback(incomingMessagesCallback:0x%x)",
- incomingMessagesCallback);
-
- return _rtpReceiver.RegisterIncomingRTPCallback(incomingMessagesCallback);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterIncomingRTCPCallback(
- RtcpFeedback* incomingMessagesCallback) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterIncomingRTCPCallback(incomingMessagesCallback:0x%x)",
- incomingMessagesCallback);
-
- return _rtcpReceiver.RegisterIncomingRTCPCallback(incomingMessagesCallback);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterIncomingVideoCallback(
- RtpVideoFeedback* incomingMessagesCallback) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterIncomingVideoCallback(incomingMessagesCallback:0x%x)",
- incomingMessagesCallback);
-
- if (_rtcpReceiver.RegisterIncomingVideoCallback(incomingMessagesCallback)
- == 0) {
- return _rtpReceiver.RegisterIncomingVideoCallback(
- incomingMessagesCallback);
- }
- return -1;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterAudioCallback(
- RtpAudioFeedback* messagesCallback) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterAudioCallback(messagesCallback:0x%x)",
- messagesCallback);
-
- if (_rtpSender.RegisterAudioCallback(messagesCallback) == 0) {
- return _rtpReceiver.RegisterIncomingAudioCallback(messagesCallback);
- }
- return -1;
-}
-
-/**
-* Sender
-*/
-
-WebRtc_Word32 ModuleRtpRtcpImpl::InitSender() {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "InitSender()");
-
- _collisionDetected = false;
-
- // if we are already receiving inform our sender to avoid collision
- if (_rtpSender.Init(_rtpReceiver.SSRC()) != 0) {
- return -1;
- }
- WebRtc_Word32 retVal = _rtcpSender.Init();
-
- // make sure that RTCP objects are aware of our SSRC
- // (it could have changed due to collision)
- WebRtc_UWord32 SSRC = _rtpSender.SSRC();
- _rtcpReceiver.SetSSRC(SSRC);
- _rtcpSender.SetSSRC(SSRC);
- return retVal;
-}
-
-bool ModuleRtpRtcpImpl::RTPKeepalive() const {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, "RTPKeepalive()");
-
- return _rtpSender.RTPKeepalive();
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RTPKeepaliveStatus(
- bool* enable,
- WebRtc_Word8* unknownPayloadType,
- WebRtc_UWord16* deltaTransmitTimeMS) const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RTPKeepaliveStatus()");
-
- return _rtpSender.RTPKeepaliveStatus(enable,
- unknownPayloadType,
- deltaTransmitTimeMS);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetRTPKeepaliveStatus(
- bool enable,
- WebRtc_Word8 unknownPayloadType,
- WebRtc_UWord16 deltaTransmitTimeMS) {
- if (enable) {
- WEBRTC_TRACE(
- kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetRTPKeepaliveStatus(true, plType:%d deltaTransmitTimeMS:%u)",
- unknownPayloadType,
- deltaTransmitTimeMS);
-
- // check the transmit keepalive delta time [1,60]
- if (deltaTransmitTimeMS < 1000 || deltaTransmitTimeMS > 60000) {
- WEBRTC_TRACE(kTraceError,
- kTraceRtpRtcp,
- _id,
- "\tinvalid deltaTransmitTimeSeconds (%d)",
- deltaTransmitTimeMS);
- return -1;
- }
-
- // check the payload time [0,127]
- if (unknownPayloadType < 0) {
- WEBRTC_TRACE(kTraceError,
- kTraceRtpRtcp,
- _id,
- "\tinvalid unknownPayloadType (%d)",
- unknownPayloadType);
- return -1;
- }
- // enable RTP keepalive mechanism
- return _rtpSender.EnableRTPKeepalive(unknownPayloadType,
- deltaTransmitTimeMS);
- } else {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetRTPKeepaliveStatus(disable)");
- return _rtpSender.DisableRTPKeepalive();
- }
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterSendPayload(
- const CodecInst& voiceCodec) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterSendPayload(plName:%s plType:%d frequency:%u)",
- voiceCodec.plname,
- voiceCodec.pltype,
- voiceCodec.plfreq);
-
- return _rtpSender.RegisterPayload(
- voiceCodec.plname,
- voiceCodec.pltype,
- voiceCodec.plfreq,
- voiceCodec.channels,
- (voiceCodec.rate < 0) ? 0 : voiceCodec.rate);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterSendPayload(
- const VideoCodec& videoCodec) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterSendPayload(plName:%s plType:%d)",
- videoCodec.plName,
- videoCodec.plType);
-
- _sendVideoCodec = videoCodec;
- _simulcast = (videoCodec.numberOfSimulcastStreams > 1) ? true : false;
- return _rtpSender.RegisterPayload(videoCodec.plName,
- videoCodec.plType,
- 90000,
- 0,
- videoCodec.maxBitrate);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::DeRegisterSendPayload(
- const WebRtc_Word8 payloadType) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "DeRegisterSendPayload(%d)", payloadType);
-
- return _rtpSender.DeRegisterSendPayload(payloadType);
-}
-
-WebRtc_Word8 ModuleRtpRtcpImpl::SendPayloadType() const {
- return _rtpSender.SendPayloadType();
-}
-
-WebRtc_UWord32 ModuleRtpRtcpImpl::StartTimestamp() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "StartTimestamp()");
-
- return _rtpSender.StartTimestamp();
-}
-
-// configure start timestamp, default is a random number
-WebRtc_Word32 ModuleRtpRtcpImpl::SetStartTimestamp(
- const WebRtc_UWord32 timestamp) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetStartTimestamp(%d)",
- timestamp);
-
- return _rtpSender.SetStartTimestamp(timestamp, true);
-}
-
-WebRtc_UWord16 ModuleRtpRtcpImpl::SequenceNumber() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SequenceNumber()");
-
- return _rtpSender.SequenceNumber();
-}
-
-// Set SequenceNumber, default is a random number
-WebRtc_Word32 ModuleRtpRtcpImpl::SetSequenceNumber(
- const WebRtc_UWord16 seqNum) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetSequenceNumber(%d)",
- seqNum);
-
- return _rtpSender.SetSequenceNumber(seqNum);
-}
-
-WebRtc_UWord32 ModuleRtpRtcpImpl::SSRC() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SSRC()");
-
- return _rtpSender.SSRC();
-}
-
-// configure SSRC, default is a random number
-WebRtc_Word32 ModuleRtpRtcpImpl::SetSSRC(const WebRtc_UWord32 ssrc) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetSSRC(%d)", ssrc);
-
- if (_rtpSender.SetSSRC(ssrc) == 0) {
- _rtcpReceiver.SetSSRC(ssrc);
- _rtcpSender.SetSSRC(ssrc);
- return 0;
- }
- return -1;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetCSRCStatus(const bool include) {
- _rtcpSender.SetCSRCStatus(include);
- return _rtpSender.SetCSRCStatus(include);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::CSRCs(
- WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "CSRCs()");
-
- return _rtpSender.CSRCs(arrOfCSRC);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetCSRCs(
- const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
- const WebRtc_UWord8 arrLength) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetCSRCs(arrLength:%d)",
- arrLength);
-
- const bool defaultInstance(_childModules.empty() ? false : true);
-
- if (defaultInstance) {
- // for default we need to update all child modules too
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
-
- std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
- while (it != _childModules.end()) {
- RtpRtcp* module = *it;
- if (module) {
- module->SetCSRCs(arrOfCSRC, arrLength);
- }
- it++;
- }
- return 0;
-
- } else {
- for (int i = 0; i < arrLength; i++) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "\tidx:%d CSRC:%u", i,
- arrOfCSRC[i]);
- }
- _rtcpSender.SetCSRCs(arrOfCSRC, arrLength);
- return _rtpSender.SetCSRCs(arrOfCSRC, arrLength);
- }
-}
-
-WebRtc_UWord32 ModuleRtpRtcpImpl::PacketCountSent() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "PacketCountSent()");
-
- return _rtpSender.Packets();
-}
-
-WebRtc_UWord32 ModuleRtpRtcpImpl::ByteCountSent() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "ByteCountSent()");
-
- return _rtpSender.Bytes();
-}
-
-int ModuleRtpRtcpImpl::CurrentSendFrequencyHz() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "CurrentSendFrequencyHz()");
-
- return _rtpSender.SendPayloadFrequency();
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetSendingStatus(const bool sending) {
- if (sending) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "SetSendingStatus(sending)");
- } else {
- if (_rtpSender.RTPKeepalive()) {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceRtpRtcp,
- _id,
- "Can't SetSendingStatus(stopped) when RTP Keepalive is active");
- return -1;
- }
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "SetSendingStatus(stopped)");
- }
- if (_rtcpSender.Sending() != sending) {
- // sends RTCP BYE when going from true to false
- if (_rtcpSender.SetSendingStatus(sending) != 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
- "Failed to send RTCP BYE");
- }
-
- _collisionDetected = false;
-
- // generate a new timeStamp if true and not configured via API
- // generate a new SSRC for the next "call" if false
- _rtpSender.SetSendingStatus(sending);
-
- // make sure that RTCP objects are aware of our SSRC (it could have changed
- // due to collision)
- WebRtc_UWord32 SSRC = _rtpSender.SSRC();
- _rtcpReceiver.SetSSRC(SSRC);
- _rtcpSender.SetSSRC(SSRC);
- return 0;
- }
- return 0;
-}
-
-bool ModuleRtpRtcpImpl::Sending() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "Sending()");
-
- return _rtcpSender.Sending();
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetSendingMediaStatus(const bool sending) {
- if (sending) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "SetSendingMediaStatus(sending)");
- } else {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "SetSendingMediaStatus(stopped)");
- }
- _rtpSender.SetSendingMediaStatus(sending);
- return 0;
-}
-
-bool ModuleRtpRtcpImpl::SendingMedia() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "Sending()");
-
- const bool haveChildModules(_childModules.empty() ? false : true);
- if (!haveChildModules) {
- return _rtpSender.SendingMedia();
- }
-
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- std::list<ModuleRtpRtcpImpl*>::const_iterator it = _childModules.begin();
- while (it != _childModules.end()) {
- RTPSender& rtpSender = (*it)->_rtpSender;
- if (rtpSender.SendingMedia()) {
- return true;
- }
- it++;
- }
- return false;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterSendTransport(
- Transport* outgoingTransport) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RegisterSendTransport(0x%x)", outgoingTransport);
-
- if (_rtpSender.RegisterSendTransport(outgoingTransport) == 0) {
- return _rtcpSender.RegisterSendTransport(outgoingTransport);
- }
- return -1;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SendOutgoingData(
- FrameType frameType,
- WebRtc_Word8 payloadType,
- WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader* fragmentation,
- const RTPVideoHeader* rtpVideoHdr) {
- WEBRTC_TRACE(
- kTraceStream,
- kTraceRtpRtcp,
- _id,
- "SendOutgoingData(frameType:%d payloadType:%d timeStamp:%u size:%u)",
- frameType, payloadType, timeStamp, payloadSize);
-
- const bool haveChildModules(_childModules.empty() ? false : true);
- if (!haveChildModules) {
- // Don't sent RTCP from default module
- if (_rtcpSender.TimeToSendRTCPReport(kVideoFrameKey == frameType)) {
- _rtcpSender.SendRTCP(kRtcpReport);
- }
- return _rtpSender.SendOutgoingData(frameType,
- payloadType,
- timeStamp,
- payloadData,
- payloadSize,
- fragmentation,
- NULL,
- &(rtpVideoHdr->codecHeader));
- }
- WebRtc_Word32 retVal = -1;
- if (_simulcast) {
- if (rtpVideoHdr == NULL) {
- return -1;
- }
- int idx = 0;
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
- for (; idx < rtpVideoHdr->simulcastIdx; idx++) {
- it++;
- if (it == _childModules.end()) {
- return -1;
- }
- }
- RTPSender& rtpSender = (*it)->_rtpSender;
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SendOutgoingData(SimulcastIdx:%u size:%u, ssrc:0x%x)",
- idx, payloadSize, rtpSender.SSRC());
- return rtpSender.SendOutgoingData(frameType,
- payloadType,
- timeStamp,
- payloadData,
- payloadSize,
- fragmentation,
- NULL,
- &(rtpVideoHdr->codecHeader));
- } else {
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- // TODO(pwestin) remove codecInfo from SendOutgoingData
- VideoCodecInformation* codecInfo = NULL;
-
- std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
- if (it != _childModules.end()) {
- RTPSender& rtpSender = (*it)->_rtpSender;
- retVal = rtpSender.SendOutgoingData(frameType,
- payloadType,
- timeStamp,
- payloadData,
- payloadSize,
- fragmentation,
- NULL,
- &(rtpVideoHdr->codecHeader));
-
- it++;
- }
-
- // send to all remaining "child" modules
- while (it != _childModules.end()) {
- RTPSender& rtpSender = (*it)->_rtpSender;
- retVal = rtpSender.SendOutgoingData(frameType,
- payloadType,
- timeStamp,
- payloadData,
- payloadSize,
- fragmentation,
- codecInfo,
- &(rtpVideoHdr->codecHeader));
-
- it++;
- }
- }
- return retVal;
-}
-
-WebRtc_UWord16 ModuleRtpRtcpImpl::MaxPayloadLength() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "MaxPayloadLength()");
-
- return _rtpSender.MaxPayloadLength();
-}
-
-WebRtc_UWord16 ModuleRtpRtcpImpl::MaxDataPayloadLength() const {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "MaxDataPayloadLength()");
-
- WebRtc_UWord16 minDataPayloadLength = IP_PACKET_SIZE - 28; // Assuming IP/UDP
-
- const bool defaultInstance(_childModules.empty() ? false : true);
- if (defaultInstance) {
- // for default we need to update all child modules too
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- std::list<ModuleRtpRtcpImpl*>::const_iterator it =
- _childModules.begin();
- while (it != _childModules.end()) {
- RtpRtcp* module = *it;
- if (module) {
- WebRtc_UWord16 dataPayloadLength =
- module->MaxDataPayloadLength();
- if (dataPayloadLength < minDataPayloadLength) {
- minDataPayloadLength = dataPayloadLength;
- }
- }
- it++;
- }
- }
-
- WebRtc_UWord16 dataPayloadLength = _rtpSender.MaxDataPayloadLength();
- if (dataPayloadLength < minDataPayloadLength) {
- minDataPayloadLength = dataPayloadLength;
- }
- return minDataPayloadLength;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetTransportOverhead(
- const bool TCP,
- const bool IPV6,
- const WebRtc_UWord8 authenticationOverhead) {
- WEBRTC_TRACE(
- kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetTransportOverhead(TCP:%d, IPV6:%d authenticationOverhead:%u)",
- TCP, IPV6, authenticationOverhead);
-
- WebRtc_UWord16 packetOverHead = 0;
- if (IPV6) {
- packetOverHead = 40;
- } else {
- packetOverHead = 20;
- }
- if (TCP) {
- // TCP
- packetOverHead += 20;
- } else {
- // UDP
- packetOverHead += 8;
- }
- packetOverHead += authenticationOverhead;
-
- if (packetOverHead == _packetOverHead) {
- // ok same as before
- return 0;
- }
- // calc diff
- WebRtc_Word16 packetOverHeadDiff = packetOverHead - _packetOverHead;
-
- // store new
- _packetOverHead = packetOverHead;
-
- _rtpReceiver.SetPacketOverHead(_packetOverHead);
- WebRtc_UWord16 length = _rtpSender.MaxPayloadLength() - packetOverHeadDiff;
- return _rtpSender.SetMaxPayloadLength(length, _packetOverHead);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetMaxTransferUnit(const WebRtc_UWord16 MTU) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetMaxTransferUnit(%u)",
- MTU);
-
- if (MTU > IP_PACKET_SIZE) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
- "Invalid in argument to SetMaxTransferUnit(%u)", MTU);
- return -1;
- }
- return _rtpSender.SetMaxPayloadLength(MTU - _packetOverHead,
- _packetOverHead);
-}
-
-/*
-* RTCP
-*/
-RTCPMethod ModuleRtpRtcpImpl::RTCP() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RTCP()");
-
- if (_rtcpSender.Status() != kRtcpOff) {
- return _rtcpReceiver.Status();
- }
- return kRtcpOff;
-}
-
-// configure RTCP status i.e on/off
-WebRtc_Word32 ModuleRtpRtcpImpl::SetRTCPStatus(const RTCPMethod method) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetRTCPStatus(%d)",
- method);
-
- if (_rtcpSender.SetRTCPStatus(method) == 0) {
- return _rtcpReceiver.SetRTCPStatus(method);
- }
- return -1;
-}
-
-// only for internal test
-WebRtc_UWord32 ModuleRtpRtcpImpl::LastSendReport(WebRtc_UWord32& lastRTCPTime) {
- return _rtcpSender.LastSendReport(lastRTCPTime);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetCNAME(const char cName[RTCP_CNAME_SIZE]) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetCNAME(%s)", cName);
- return _rtcpSender.SetCNAME(cName);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::CNAME(char cName[RTCP_CNAME_SIZE]) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "CNAME()");
- return _rtcpSender.CNAME(cName);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::AddMixedCNAME(
- const WebRtc_UWord32 SSRC,
- const char cName[RTCP_CNAME_SIZE]) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "AddMixedCNAME(SSRC:%u)", SSRC);
-
- return _rtcpSender.AddMixedCNAME(SSRC, cName);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RemoveMixedCNAME(const WebRtc_UWord32 SSRC) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "RemoveMixedCNAME(SSRC:%u)", SSRC);
- return _rtcpSender.RemoveMixedCNAME(SSRC);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RemoteCNAME(
- const WebRtc_UWord32 remoteSSRC,
- char cName[RTCP_CNAME_SIZE]) const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "RemoteCNAME(SSRC:%u)", remoteSSRC);
-
- return _rtcpReceiver.CNAME(remoteSSRC, cName);
-}
-
-WebRtc_UWord16 ModuleRtpRtcpImpl::RemoteSequenceNumber() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoteSequenceNumber()");
-
- return _rtpReceiver.SequenceNumber();
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RemoteNTP(
- WebRtc_UWord32* receivedNTPsecs,
- WebRtc_UWord32* receivedNTPfrac,
- WebRtc_UWord32* RTCPArrivalTimeSecs,
- WebRtc_UWord32* RTCPArrivalTimeFrac) const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoteNTP()");
-
- return _rtcpReceiver.NTP(receivedNTPsecs,
- receivedNTPfrac,
- RTCPArrivalTimeSecs,
- RTCPArrivalTimeFrac);
-}
-
-// Get RoundTripTime
-WebRtc_Word32 ModuleRtpRtcpImpl::RTT(const WebRtc_UWord32 remoteSSRC,
- WebRtc_UWord16* RTT,
- WebRtc_UWord16* avgRTT,
- WebRtc_UWord16* minRTT,
- WebRtc_UWord16* maxRTT) const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RTT()");
-
- return _rtcpReceiver.RTT(remoteSSRC, RTT, avgRTT, minRTT, maxRTT);
-}
-
-// Reset RoundTripTime statistics
-WebRtc_Word32
-ModuleRtpRtcpImpl::ResetRTT(const WebRtc_UWord32 remoteSSRC) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "ResetRTT(SSRC:%u)",
- remoteSSRC);
-
- return _rtcpReceiver.ResetRTT(remoteSSRC);
-}
-
-// Reset RTP statistics
-WebRtc_Word32
-ModuleRtpRtcpImpl::ResetStatisticsRTP() {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "ResetStatisticsRTP()");
-
- return _rtpReceiver.ResetStatistics();
-}
-
-// Reset RTP data counters for the receiving side
-WebRtc_Word32 ModuleRtpRtcpImpl::ResetReceiveDataCountersRTP() {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "ResetReceiveDataCountersRTP()");
-
- return _rtpReceiver.ResetDataCounters();
-}
-
-// Reset RTP data counters for the sending side
-WebRtc_Word32 ModuleRtpRtcpImpl::ResetSendDataCountersRTP() {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "ResetSendDataCountersRTP()");
-
- return _rtpSender.ResetDataCounters();
-}
-
-// Force a send of an RTCP packet
-// normal SR and RR are triggered via the process function
-WebRtc_Word32 ModuleRtpRtcpImpl::SendRTCP(WebRtc_UWord32 rtcpPacketType) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SendRTCP(0x%x)",
- rtcpPacketType);
-
- return _rtcpSender.SendRTCP(rtcpPacketType);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetRTCPApplicationSpecificData(
- const WebRtc_UWord8 subType,
- const WebRtc_UWord32 name,
- const WebRtc_UWord8* data,
- const WebRtc_UWord16 length) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "SetRTCPApplicationSpecificData(subType:%d name:0x%x)", subType,
- name);
-
- return _rtcpSender.SetApplicationSpecificData(subType, name, data, length);
-}
-
-/*
-* (XR) VOIP metric
-*/
-WebRtc_Word32 ModuleRtpRtcpImpl::SetRTCPVoIPMetrics(
- const RTCPVoIPMetric* VoIPMetric) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetRTCPVoIPMetrics()");
-
- return _rtcpSender.SetRTCPVoIPMetrics(VoIPMetric);
-}
-
-// our localy created statistics of the received RTP stream
-WebRtc_Word32 ModuleRtpRtcpImpl::StatisticsRTP(
- WebRtc_UWord8* fraction_lost,
- WebRtc_UWord32* cum_lost,
- WebRtc_UWord32* ext_max,
- WebRtc_UWord32* jitter,
- WebRtc_UWord32* max_jitter) const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "StatisticsRTP()");
-
- WebRtc_UWord32 jitter_transmission_time_offset = 0;
-
- WebRtc_Word32 retVal = _rtpReceiver.Statistics(
- fraction_lost,
- cum_lost,
- ext_max,
- jitter,
- max_jitter,
- &jitter_transmission_time_offset,
- (_rtcpSender.Status() == kRtcpOff));
- if (retVal == -1) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
- "StatisticsRTP() no statisitics availble");
- }
- return retVal;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::DataCountersRTP(
- WebRtc_UWord32* bytesSent,
- WebRtc_UWord32* packetsSent,
- WebRtc_UWord32* bytesReceived,
- WebRtc_UWord32* packetsReceived) const {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, "DataCountersRTP()");
-
- if (bytesSent) {
- *bytesSent = _rtpSender.Bytes();
- }
- if (packetsSent) {
- *packetsSent = _rtpSender.Packets();
- }
- return _rtpReceiver.DataCounters(bytesReceived, packetsReceived);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::ReportBlockStatistics(
- WebRtc_UWord8* fraction_lost,
- WebRtc_UWord32* cum_lost,
- WebRtc_UWord32* ext_max,
- WebRtc_UWord32* jitter,
- WebRtc_UWord32* jitter_transmission_time_offset) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "ReportBlockStatistics()");
- WebRtc_Word32 missing = 0;
- WebRtc_Word32 ret = _rtpReceiver.Statistics(fraction_lost,
- cum_lost,
- ext_max,
- jitter,
- NULL,
- jitter_transmission_time_offset,
- &missing,
- true);
-
-#ifdef MATLAB
- if (_plot1 == NULL) {
- _plot1 = eng.NewPlot(new MatlabPlot());
- _plot1->AddTimeLine(30, "b", "lost", _clock.GetTimeInMS());
- }
- _plot1->Append("lost", missing);
- _plot1->Plot();
-#endif
-
- return ret;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RemoteRTCPStat(RTCPSenderInfo* senderInfo) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoteRTCPStat()");
-
- return _rtcpReceiver.SenderInfoReceived(senderInfo);
-}
-
-// received RTCP report
-WebRtc_Word32 ModuleRtpRtcpImpl::RemoteRTCPStat(
- std::vector<RTCPReportBlock>* receiveBlocks) const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoteRTCPStat()");
-
- return _rtcpReceiver.StatisticsReceived(receiveBlocks);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::AddRTCPReportBlock(
- const WebRtc_UWord32 SSRC,
- const RTCPReportBlock* reportBlock) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "AddRTCPReportBlock()");
-
- return _rtcpSender.AddReportBlock(SSRC, reportBlock);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RemoveRTCPReportBlock(
- const WebRtc_UWord32 SSRC) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RemoveRTCPReportBlock()");
-
- return _rtcpSender.RemoveReportBlock(SSRC);
-}
-
-/*
- * (REMB) Receiver Estimated Max Bitrate
- */
-bool ModuleRtpRtcpImpl::REMB() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "REMB()");
-
- return _rtcpSender.REMB();
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetREMBStatus(const bool enable) {
- if (enable) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetREMBStatus(enable)");
- } else {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetREMBStatus(disable)");
- }
- return _rtcpSender.SetREMBStatus(enable);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetREMBData(const WebRtc_UWord32 bitrate,
- const WebRtc_UWord8 numberOfSSRC,
- const WebRtc_UWord32* SSRC) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "SetREMBData(bitrate:%d,?,?)", bitrate);
- return _rtcpSender.SetREMBData(bitrate, numberOfSSRC, SSRC);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetMaximumBitrateEstimate(
- const WebRtc_UWord32 bitrate) {
- if(!_rtcpSender.REMB()) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "SetMaximumBitrateEstimate - REMB not enabled.");
- return -1;
- }
- OnReceivedEstimatedMaxBitrate(bitrate);
- return 0;
-}
-
-bool ModuleRtpRtcpImpl::SetRemoteBitrateObserver(
- RtpRemoteBitrateObserver* observer) {
- return _rtcpSender.SetRemoteBitrateObserver(observer);
-}
-
-/*
- * (IJ) Extended jitter report.
- */
-bool ModuleRtpRtcpImpl::IJ() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "IJ()");
-
- return _rtcpSender.IJ();
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetIJStatus(const bool enable) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetIJStatus(%s)", enable ? "true" : "false");
-
- return _rtcpSender.SetIJStatus(enable);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterSendRtpHeaderExtension(
- const RTPExtensionType type,
- const WebRtc_UWord8 id) {
- return _rtpSender.RegisterRtpHeaderExtension(type, id);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::DeregisterSendRtpHeaderExtension(
- const RTPExtensionType type) {
- return _rtpSender.DeregisterRtpHeaderExtension(type);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RegisterReceiveRtpHeaderExtension(
- const RTPExtensionType type,
- const WebRtc_UWord8 id) {
- return _rtpReceiver.RegisterRtpHeaderExtension(type, id);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::DeregisterReceiveRtpHeaderExtension(
- const RTPExtensionType type) {
- return _rtpReceiver.DeregisterRtpHeaderExtension(type);
-}
-
-void ModuleRtpRtcpImpl::SetTransmissionSmoothingStatus(const bool enable) {
- _rtpSender.SetTransmissionSmoothingStatus(enable);
-}
-
-bool ModuleRtpRtcpImpl::TransmissionSmoothingStatus() const {
- return _rtpSender.TransmissionSmoothingStatus();
-}
-
-/*
-* (TMMBR) Temporary Max Media Bit Rate
-*/
-bool ModuleRtpRtcpImpl::TMMBR() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "TMMBR()");
-
- return _rtcpSender.TMMBR();
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetTMMBRStatus(const bool enable) {
- if (enable) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "SetTMMBRStatus(enable)");
- } else {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "SetTMMBRStatus(disable)");
- }
- return _rtcpSender.SetTMMBRStatus(enable);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::TMMBRReceived(
- const WebRtc_UWord32 size,
- const WebRtc_UWord32 accNumCandidates,
- TMMBRSet* candidateSet) const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "TMMBRReceived()");
-
- return _rtcpReceiver.TMMBRReceived(size, accNumCandidates, candidateSet);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetTMMBN(const TMMBRSet* boundingSet,
- const WebRtc_UWord32 maxBitrateKbit) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SetTMMBN()");
-
- return _rtcpSender.SetTMMBN(boundingSet, maxBitrateKbit);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RequestTMMBR(const WebRtc_UWord32 estimatedBW,
- const WebRtc_UWord32 packetOH) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "RequestTMMBR()");
-
- return _rtcpSender.RequestTMMBR(estimatedBW, packetOH);
-}
-
-/*
-* (NACK) Negative acknowledgement
-*/
-
-// Is Negative acknowledgement requests on/off?
-NACKMethod ModuleRtpRtcpImpl::NACK() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "NACK()");
-
- NACKMethod childMethod = kNackOff;
- const bool defaultInstance(_childModules.empty() ? false : true);
- if (defaultInstance) {
- // for default we need to check all child modules too
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- std::list<ModuleRtpRtcpImpl*>::const_iterator it =
- _childModules.begin();
- while (it != _childModules.end()) {
- RtpRtcp* module = *it;
- if (module) {
- NACKMethod nackMethod = module->NACK();
- if (nackMethod != kNackOff) {
- childMethod = nackMethod;
- break;
- }
- }
- it++;
- }
- }
-
- NACKMethod method = _nackMethod;
- if (childMethod != kNackOff) {
- method = childMethod;
- }
- return method;
-}
-
-// Turn negative acknowledgement requests on/off
-WebRtc_Word32 ModuleRtpRtcpImpl::SetNACKStatus(NACKMethod method) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetNACKStatus(%u)", method);
-
- _nackMethod = method;
- _rtpReceiver.SetNACKStatus(method);
- return 0;
-}
-
-// Returns the currently configured retransmission mode.
-int ModuleRtpRtcpImpl::SelectiveRetransmissions() const {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SelectiveRetransmissions()");
- return _rtpSender.SelectiveRetransmissions();
-}
-
-// Enable or disable a retransmission mode, which decides which packets will
-// be retransmitted if NACKed.
-int ModuleRtpRtcpImpl::SetSelectiveRetransmissions(uint8_t settings) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetSelectiveRetransmissions(%u)",
- settings);
- return _rtpSender.SetSelectiveRetransmissions(settings);
-}
-
-// Send a Negative acknowledgement packet
-WebRtc_Word32 ModuleRtpRtcpImpl::SendNACK(const WebRtc_UWord16* nackList,
- const WebRtc_UWord16 size) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SendNACK(size:%u)", size);
-
- if (size > NACK_PACKETS_MAX_SIZE) {
- RequestKeyFrame();
- return -1;
- }
- WebRtc_UWord16 avgRTT = 0;
- _rtcpReceiver.RTT(_rtpReceiver.SSRC(), NULL, &avgRTT, NULL, NULL);
-
- WebRtc_UWord32 waitTime = 5 + ((avgRTT * 3) >> 1); // 5 + RTT*1.5
- if (waitTime == 5) {
- waitTime = 100; //During startup we don't have an RTT
- }
- const WebRtc_UWord32 now = _clock.GetTimeInMS();
- const WebRtc_UWord32 timeLimit = now - waitTime;
-
- if (_nackLastTimeSent < timeLimit) {
- // send list
- } else {
- // only send if extended list
- if (_nackLastSeqNumberSent == nackList[size - 1]) {
- // last seq num is the same don't send list
- return 0;
- } else {
- // send list
- }
- }
- _nackLastTimeSent = now;
- _nackLastSeqNumberSent = nackList[size - 1];
-
- switch (_nackMethod) {
- case kNackRtcp:
- return _rtcpSender.SendRTCP(kRtcpNack, size, nackList);
- case kNackOff:
- return -1;
- };
- return -1;
-}
-
-// Store the sent packets, needed to answer to a Negative acknowledgement
-// requests
-WebRtc_Word32 ModuleRtpRtcpImpl::SetStorePacketsStatus(
- const bool enable,
- const WebRtc_UWord16 numberToStore) {
- if (enable) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "SetStorePacketsStatus(enable, numberToStore:%d)",
- numberToStore);
- } else {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "SetStorePacketsStatus(disable)");
- }
- return _rtpSender.SetStorePacketsStatus(enable, numberToStore);
-}
-
-/*
-* Audio
-*/
-
-// Outband TelephoneEvent detection
-WebRtc_Word32 ModuleRtpRtcpImpl::SetTelephoneEventStatus(
- const bool enable,
- const bool forwardToDecoder,
- const bool detectEndOfTone) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "SetTelephoneEventStatus(enable:%d forwardToDecoder:%d"
- " detectEndOfTone:%d)", enable, forwardToDecoder,
- detectEndOfTone);
-
- return _rtpReceiver.SetTelephoneEventStatus(enable, forwardToDecoder,
- detectEndOfTone);
-}
-
-// Is outband TelephoneEvent turned on/off?
-bool ModuleRtpRtcpImpl::TelephoneEvent() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "TelephoneEvent()");
-
- return _rtpReceiver.TelephoneEvent();
-}
-
-// Is forwarding of outband telephone events turned on/off?
-bool ModuleRtpRtcpImpl::TelephoneEventForwardToDecoder() const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "TelephoneEventForwardToDecoder()");
-
- return _rtpReceiver.TelephoneEventForwardToDecoder();
-}
-
-// Send a TelephoneEvent tone using RFC 2833 (4733)
-WebRtc_Word32 ModuleRtpRtcpImpl::SendTelephoneEventOutband(
- const WebRtc_UWord8 key,
- const WebRtc_UWord16 timeMs,
- const WebRtc_UWord8 level) {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id,
- "SendTelephoneEventOutband(key:%u, timeMs:%u, level:%u)", key,
- timeMs, level);
-
- return _rtpSender.SendTelephoneEvent(key, timeMs, level);
-}
-
-bool ModuleRtpRtcpImpl::SendTelephoneEventActive(
- WebRtc_Word8& telephoneEvent) const {
-
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SendTelephoneEventActive()");
-
- return _rtpSender.SendTelephoneEventActive(telephoneEvent);
-}
-
-// set audio packet size, used to determine when it's time to send a DTMF
-// packet in silence (CNG)
-WebRtc_Word32 ModuleRtpRtcpImpl::SetAudioPacketSize(
- const WebRtc_UWord16 packetSizeSamples) {
-
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetAudioPacketSize(%u)",
- packetSizeSamples);
-
- return _rtpSender.SetAudioPacketSize(packetSizeSamples);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetRTPAudioLevelIndicationStatus(
- const bool enable,
- const WebRtc_UWord8 ID) {
-
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetRTPAudioLevelIndicationStatus(enable=%d, ID=%u)",
- enable,
- ID);
-
- if (enable) {
- _rtpReceiver.RegisterRtpHeaderExtension(kRtpExtensionAudioLevel, ID);
- } else {
- _rtpReceiver.DeregisterRtpHeaderExtension(kRtpExtensionAudioLevel);
- }
- return _rtpSender.SetAudioLevelIndicationStatus(enable, ID);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::GetRTPAudioLevelIndicationStatus(
- bool& enable,
- WebRtc_UWord8& ID) const {
-
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "GetRTPAudioLevelIndicationStatus()");
- return _rtpSender.AudioLevelIndicationStatus(enable, ID);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetAudioLevel(const WebRtc_UWord8 level_dBov) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetAudioLevel(level_dBov:%u)",
- level_dBov);
- return _rtpSender.SetAudioLevel(level_dBov);
-}
-
-// Set payload type for Redundant Audio Data RFC 2198
-WebRtc_Word32 ModuleRtpRtcpImpl::SetSendREDPayloadType(
- const WebRtc_Word8 payloadType) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetSendREDPayloadType(%d)",
- payloadType);
-
- return _rtpSender.SetRED(payloadType);
-}
-
-// Get payload type for Redundant Audio Data RFC 2198
-WebRtc_Word32 ModuleRtpRtcpImpl::SendREDPayloadType(
- WebRtc_Word8& payloadType) const {
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "SendREDPayloadType()");
-
- return _rtpSender.RED(payloadType);
-}
-
-
-/*
-* Video
-*/
-RtpVideoCodecTypes ModuleRtpRtcpImpl::ReceivedVideoCodec() const {
- return _rtpReceiver.VideoCodecType();
-}
-
-RtpVideoCodecTypes ModuleRtpRtcpImpl::SendVideoCodec() const {
- return _rtpSender.VideoCodecType();
-}
-
-void ModuleRtpRtcpImpl::SetSendBitrate(const WebRtc_UWord32 startBitrate,
- const WebRtc_UWord16 minBitrateKbit,
- const WebRtc_UWord16 maxBitrateKbit) {
-
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetSendBitrate start:%ubit/s min:%uKbit/s max:%uKbit/s",
- startBitrate, minBitrateKbit, maxBitrateKbit);
-
- const bool defaultInstance(_childModules.empty() ? false : true);
-
- if (defaultInstance) {
- // for default we need to update all child modules too
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
-
- std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
- while (it != _childModules.end()) {
- RtpRtcp* module = *it;
- if (module) {
- module->SetSendBitrate(startBitrate,
- minBitrateKbit,
- maxBitrateKbit);
- }
- it++;
- }
- }
- // TODO(henrike): this function also returns a value. It never fails so
- // make it return void.
- _rtpSender.SetTargetSendBitrate(startBitrate);
-
- _bandwidthManagement.SetSendBitrate(startBitrate, minBitrateKbit,
- maxBitrateKbit);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetKeyFrameRequestMethod(
- const KeyFrameRequestMethod method) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetKeyFrameRequestMethod(method:%u)",
- method);
-
- _keyFrameReqMethod = method;
- return 0;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::RequestKeyFrame() {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "RequestKeyFrame");
-
- switch (_keyFrameReqMethod) {
- case kKeyFrameReqFirRtp:
- return _rtpSender.SendRTPIntraRequest();
- case kKeyFrameReqPliRtcp:
- return _rtcpSender.SendRTCP(kRtcpPli);
- case kKeyFrameReqFirRtcp:
- return _rtcpSender.SendRTCP(kRtcpFir);
- }
- return -1;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SendRTCPSliceLossIndication(
- const WebRtc_UWord8 pictureID) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SendRTCPSliceLossIndication (pictureID:%d)",
- pictureID);
- return _rtcpSender.SendRTCP(kRtcpSli, 0, 0, false, pictureID);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetCameraDelay(const WebRtc_Word32 delayMS) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetCameraDelay(%d)",
- delayMS);
- const bool defaultInstance(_childModules.empty() ? false : true);
-
- if (defaultInstance) {
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
-
- std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
- while (it != _childModules.end()) {
- RtpRtcp* module = *it;
- if (module) {
- module->SetCameraDelay(delayMS);
- }
- it++;
- }
- return 0;
- }
- return _rtcpSender.SetCameraDelay(delayMS);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetGenericFECStatus(
- const bool enable,
- const WebRtc_UWord8 payloadTypeRED,
- const WebRtc_UWord8 payloadTypeFEC) {
- if (enable) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetGenericFECStatus(enable, %u)",
- payloadTypeRED);
- } else {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetGenericFECStatus(disable)");
- }
- return _rtpSender.SetGenericFECStatus(enable,
- payloadTypeRED,
- payloadTypeFEC);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::GenericFECStatus(
- bool& enable,
- WebRtc_UWord8& payloadTypeRED,
- WebRtc_UWord8& payloadTypeFEC) {
-
- WEBRTC_TRACE(kTraceModuleCall, kTraceRtpRtcp, _id, "GenericFECStatus()");
-
- bool childEnabled = false;
- const bool defaultInstance(_childModules.empty() ? false : true);
- if (defaultInstance) {
- // for default we need to check all child modules too
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
- while (it != _childModules.end()) {
- RtpRtcp* module = *it;
- if (module) {
- bool enabled = false;
- WebRtc_UWord8 dummyPTypeRED = 0;
- WebRtc_UWord8 dummyPTypeFEC = 0;
- if (module->GenericFECStatus(enabled,
- dummyPTypeRED,
- dummyPTypeFEC) == 0 && enabled) {
- childEnabled = true;
- break;
- }
- }
- it++;
- }
- }
- WebRtc_Word32 retVal = _rtpSender.GenericFECStatus(enable,
- payloadTypeRED,
- payloadTypeFEC);
- if (childEnabled) {
- // returns true if enabled for any child module
- enable = childEnabled;
- }
- return retVal;
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetFECCodeRate(
- const WebRtc_UWord8 keyFrameCodeRate,
- const WebRtc_UWord8 deltaFrameCodeRate) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp,
- _id,
- "SetFECCodeRate(%u, %u)",
- keyFrameCodeRate,
- deltaFrameCodeRate);
-
- const bool defaultInstance(_childModules.empty() ? false : true);
- if (defaultInstance) {
- // for default we need to update all child modules too
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
-
- std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
- while (it != _childModules.end()) {
- RtpRtcp* module = *it;
- if (module) {
- module->SetFECCodeRate(keyFrameCodeRate, deltaFrameCodeRate);
- }
- it++;
- }
- return 0;
- }
- return _rtpSender.SetFECCodeRate(keyFrameCodeRate, deltaFrameCodeRate);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SetFECUepProtection(
- const bool keyUseUepProtection,
- const bool deltaUseUepProtection) {
- WEBRTC_TRACE(kTraceModuleCall,
- kTraceRtpRtcp, _id,
- "SetFECUepProtection(%d, %d)",
- keyUseUepProtection,
- deltaUseUepProtection);
-
- const bool defaultInstance(_childModules.empty() ? false : true);
- if (defaultInstance) {
- // for default we need to update all child modules too
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
-
- std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
- while (it != _childModules.end()) {
- RtpRtcp* module = *it;
- if (module) {
- module->SetFECUepProtection(keyUseUepProtection,
- deltaUseUepProtection);
- }
- it++;
- }
- return 0;
- }
- return _rtpSender.SetFECUepProtection(keyUseUepProtection,
- deltaUseUepProtection);
-}
-
-void ModuleRtpRtcpImpl::SetRemoteSSRC(const WebRtc_UWord32 SSRC) {
- // inform about the incoming SSRC
- _rtcpSender.SetRemoteSSRC(SSRC);
- _rtcpReceiver.SetRemoteSSRC(SSRC);
-
- // check for a SSRC collision
- if (_rtpSender.SSRC() == SSRC && !_collisionDetected) {
- // if we detect a collision change the SSRC but only once
- _collisionDetected = true;
- WebRtc_UWord32 newSSRC = _rtpSender.GenerateNewSSRC();
- if (newSSRC == 0) {
- // configured via API ignore
- return;
- }
- if (kRtcpOff != _rtcpSender.Status()) {
- // send RTCP bye on the current SSRC
- _rtcpSender.SendRTCP(kRtcpBye);
- }
- // change local SSRC
-
- // inform all objects about the new SSRC
- _rtcpSender.SetSSRC(newSSRC);
- _rtcpReceiver.SetSSRC(newSSRC);
- }
-}
-
-WebRtc_UWord32 ModuleRtpRtcpImpl::BitrateReceivedNow() const {
- return _rtpReceiver.BitrateNow();
-}
-
-void ModuleRtpRtcpImpl::BitrateSent(WebRtc_UWord32* totalRate,
- WebRtc_UWord32* videoRate,
- WebRtc_UWord32* fecRate,
- WebRtc_UWord32* nackRate) const {
- const bool defaultInstance(_childModules.empty() ? false : true);
-
- if (defaultInstance) {
- // for default we need to update the send bitrate
- CriticalSectionScoped lock(_criticalSectionModulePtrsFeedback);
-
- if (totalRate != NULL)
- *totalRate = 0;
- if (videoRate != NULL)
- *videoRate = 0;
- if (fecRate != NULL)
- *fecRate = 0;
- if (nackRate != NULL)
- *nackRate = 0;
-
- std::list<ModuleRtpRtcpImpl*>::const_iterator it =
- _childModules.begin();
- while (it != _childModules.end()) {
- RtpRtcp* module = *it;
- if (module) {
- WebRtc_UWord32 childTotalRate = 0;
- WebRtc_UWord32 childVideoRate = 0;
- WebRtc_UWord32 childFecRate = 0;
- WebRtc_UWord32 childNackRate = 0;
- module->BitrateSent(&childTotalRate,
- &childVideoRate,
- &childFecRate,
- &childNackRate);
- if (totalRate != NULL && childTotalRate > *totalRate)
- *totalRate = childTotalRate;
- if (videoRate != NULL && childVideoRate > *videoRate)
- *videoRate = childVideoRate;
- if (fecRate != NULL && childFecRate > *fecRate)
- *fecRate = childFecRate;
- if (nackRate != NULL && childNackRate > *nackRate)
- *nackRate = childNackRate;
- }
- it++;
- }
- return;
- }
- if (totalRate != NULL)
- *totalRate = _rtpSender.BitrateLast();
- if (videoRate != NULL)
- *videoRate = _rtpSender.VideoBitrateSent();
- if (fecRate != NULL)
- *fecRate = _rtpSender.FecOverheadRate();
- if (nackRate != NULL)
- *nackRate = _rtpSender.NackOverheadRate();
-}
-
-int ModuleRtpRtcpImpl::EstimatedSendBandwidth(
- WebRtc_UWord32* available_bandwidth) const {
- return _bandwidthManagement.AvailableBandwidth(available_bandwidth);
-}
-
-int ModuleRtpRtcpImpl::EstimatedReceiveBandwidth(
- WebRtc_UWord32* available_bandwidth) const {
- if (!_rtcpSender.ValidBitrateEstimate())
- return -1;
- *available_bandwidth = _rtcpSender.LatestBandwidthEstimate();
- return 0;
-}
-
-// for lip sync
-void ModuleRtpRtcpImpl::OnReceivedNTP() {
- // don't do anything if we are the audio module
- // video module is responsible for sync
- if (!_audio) {
- WebRtc_Word32 diff = 0;
- WebRtc_UWord32 receivedNTPsecs = 0;
- WebRtc_UWord32 receivedNTPfrac = 0;
- WebRtc_UWord32 RTCPArrivalTimeSecs = 0;
- WebRtc_UWord32 RTCPArrivalTimeFrac = 0;
-
- if (0 == _rtcpReceiver.NTP(&receivedNTPsecs,
- &receivedNTPfrac,
- &RTCPArrivalTimeSecs,
- &RTCPArrivalTimeFrac)) {
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
-
- if (_audioModule) {
- if (0 != _audioModule->RemoteNTP(&_receivedNTPsecsAudio,
- &_receivedNTPfracAudio,
- &_RTCPArrivalTimeSecsAudio,
- &_RTCPArrivalTimeFracAudio)) {
- // failed ot get audio NTP
- return;
- }
- }
- if (_receivedNTPfracAudio != 0) {
- // ReceivedNTPxxx is NTP at sender side when sent.
- // RTCPArrivalTimexxx is NTP at receiver side when received.
- // can't use ConvertNTPTimeToMS since calculation can be
- // negative
-
- WebRtc_Word32 NTPdiff = (WebRtc_Word32)
- ((_receivedNTPsecsAudio - receivedNTPsecs) *
- 1000); // ms
- NTPdiff += (WebRtc_Word32)
- (_receivedNTPfracAudio / FracMS - receivedNTPfrac / FracMS);
-
- WebRtc_Word32 RTCPdiff =
- static_cast<WebRtc_Word32> ((_RTCPArrivalTimeSecsAudio -
- RTCPArrivalTimeSecs) * 1000);
- RTCPdiff += (WebRtc_Word32)
- (_RTCPArrivalTimeFracAudio / FracMS -
- RTCPArrivalTimeFrac / FracMS);
-
- diff = NTPdiff - RTCPdiff;
- // if diff is + video is behind
- if (diff < -1000 || diff > 1000) {
- // unresonable ignore value.
- diff = 0;
- return;
- }
- }
- }
- // export via callback
- // after release of critsect
- _rtcpReceiver.UpdateLipSync(diff);
- }
-}
-
-// our local BW estimate is updated
-void ModuleRtpRtcpImpl::OnBandwidthEstimateUpdate(
- WebRtc_UWord16 bandWidthKbit) {
-
- WebRtc_UWord32 maxBitrateKbit = _rtpReceiver.MaxConfiguredBitrate() / 1000;
- if (maxBitrateKbit) {
- // the app has set a max bitrate
- if (maxBitrateKbit < bandWidthKbit) {
- // cap TMMBR at max configured bitrate
- bandWidthKbit = (WebRtc_UWord16)maxBitrateKbit;
- }
- }
- if (_rtcpSender.TMMBR()) {
- /* Maximum total media bit rate:
- The upper limit on total media bit rate for a given media
- stream at a particular receiver and for its selected protocol
- layer. Note that this value cannot be measured on the
- received media stream. Instead, it needs to be calculated or
- determined through other means, such as quality of service
- (QoS) negotiations or local resource limitations. Also note
- that this value is an average (on a timescale that is
- reasonable for the application) and that it may be different
- from the instantaneous bit rate seen by packets in the media
- stream.
- */
- /* Overhead:
- All protocol header information required to convey a packet
- with media data from sender to receiver, from the application
- layer down to a pre-defined protocol level (for example, down
- to, and including, the IP header). Overhead may include, for
- example, IP, UDP, and RTP headers, any layer 2 headers, any
- Contributing Sources (CSRCs), RTP padding, and RTP header
- extensions. Overhead excludes any RTP payload headers and the
- payload itself.
- */
- _rtpReceiver.PacketOHReceived();
-
- // call RequestTMMBR when our localy created estimate changes
- _rtcpSender.RequestTMMBR(bandWidthKbit, 0);
- }
-}
-
-RateControlRegion ModuleRtpRtcpImpl::OnOverUseStateUpdate(
- const RateControlInput& rateControlInput) {
-
- bool firstOverUse = false;
- RateControlRegion region = _rtcpSender.UpdateOverUseState(rateControlInput,
- firstOverUse);
- if (firstOverUse) {
- // Send TMMBR or REMB immediately.
- WebRtc_UWord16 RTT = 0;
- _rtcpReceiver.RTT(_rtpReceiver.SSRC(), &RTT, NULL, NULL, NULL);
- // About to send TMMBR, first run remote rate control
- // to get a target bit rate.
- unsigned int target_bitrate =
- _rtcpSender.CalculateNewTargetBitrate(RTT);
- if (REMB()) {
- _rtcpSender.UpdateRemoteBitrateEstimate(target_bitrate);
- } else if (TMMBR()) {
- _rtcpSender.SendRTCP(kRtcpTmmbr);
- }
- }
- return region;
-}
-
-// bad state of RTP receiver request a keyframe
-void ModuleRtpRtcpImpl::OnRequestIntraFrame() {
- RequestKeyFrame();
-}
-
-void ModuleRtpRtcpImpl::OnReceivedIntraFrameRequest(const RtpRtcp* caller) {
- if (_defaultModule) {
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- if (_defaultModule) {
- // if we use a default module pass this info to the default module
- _defaultModule->OnReceivedIntraFrameRequest(caller);
- return;
- }
- }
-
- WebRtc_UWord8 streamIdx = 0;
- FrameType frameType = kVideoFrameKey;
- if (_simulcast) {
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- // loop though child modules and count idx
- std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
- while (it != _childModules.end()) {
- ModuleRtpRtcpImpl* childModule = *it;
- if (childModule == caller) {
- break;
- }
- streamIdx++;
- it++;
- }
- }
- _rtcpReceiver.OnReceivedIntraFrameRequest(frameType, streamIdx);
-}
-
-void ModuleRtpRtcpImpl::OnReceivedEstimatedMaxBitrate(
- const WebRtc_UWord32 maxBitrate) {
- // TODO(mflodman) Split this function in two parts. One for the child module
- // and one for the default module.
-
- // We received a REMB.
- if (_defaultModule) {
- // Send this update to the REMB instance to take actions.
- _rtcpSender.ReceivedRemb(maxBitrate);
- return;
- }
-
- WebRtc_UWord32 newBitrate = 0;
- WebRtc_UWord8 fractionLost = 0;
- WebRtc_UWord16 roundTripTime = 0;
- WebRtc_UWord16 bwEstimateKbit = WebRtc_UWord16(maxBitrate / 1000);
- if (_bandwidthManagement.UpdateBandwidthEstimate(bwEstimateKbit,
- &newBitrate,
- &fractionLost,
- &roundTripTime) == 0) {
- _rtpReceiver.UpdateBandwidthManagement(newBitrate,
- fractionLost,
- roundTripTime);
-
- // We've received a new bandwidth estimate lower than the current send
- // bitrate. For simulcast we need to update the sending bitrate for all
- // streams.
- if (_simulcast) {
- CriticalSectionScoped lock(_criticalSectionModulePtrsFeedback);
- WebRtc_UWord8 idx = 0;
- for (std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
- it != _childModules.end(); ++it) {
- // sanity
- if (idx >= (_sendVideoCodec.numberOfSimulcastStreams - 1)) {
- return;
- }
- ModuleRtpRtcpImpl* module = *it;
- if (newBitrate >= _sendVideoCodec.simulcastStream[idx].maxBitrate) {
- module->_bandwidthManagement.SetSendBitrate(
- _sendVideoCodec.simulcastStream[idx].maxBitrate, 0, 0);
- module->_rtpSender.SetTargetSendBitrate(
- _sendVideoCodec.simulcastStream[idx].maxBitrate);
-
- newBitrate -= _sendVideoCodec.simulcastStream[idx].maxBitrate;
- } else {
- module->_bandwidthManagement.SetSendBitrate(newBitrate, 0, 0);
- module->_rtpSender.SetTargetSendBitrate(newBitrate);
- newBitrate -= newBitrate;
- }
- idx++;
- }
- }
- }
- // For non-simulcast, update all child modules with the new bandwidth estimate
- // regardless of the new estimate.
- if (!_simulcast) {
- // Update all child modules with the new max bitrate before exiting.
- CriticalSectionScoped lock(_criticalSectionModulePtrsFeedback);
- for (std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
- it != _childModules.end(); ++it) {
- // Update all child modules with the maximum bitrate estimate.
- ModuleRtpRtcpImpl* module = *it;
- WebRtc_UWord32 ignoreBitrate = 0;
- WebRtc_UWord8 ignoreFractionLost = 0;
- WebRtc_UWord16 ignoreRoundTripTime = 0;
- module->_bandwidthManagement.UpdateBandwidthEstimate(
- bwEstimateKbit,
- &ignoreBitrate,
- &ignoreFractionLost,
- &ignoreRoundTripTime);
- // We don't need to take care of a possible lowered bitrate, that is
- // handled earlier in this function for the default module.
- }
- }
-}
-
-// received a request for a new SLI
-void ModuleRtpRtcpImpl::OnReceivedSliceLossIndication(
- const WebRtc_UWord8 pictureID) {
-
- if (_defaultModule) {
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- if (_defaultModule) {
- // if we use a default module pass this info to the default module
- _defaultModule->OnReceivedSliceLossIndication(pictureID);
- return;
- }
- }
- _rtcpReceiver.OnReceivedSliceLossIndication(pictureID);
-}
-
-// received a new refereence frame
-void ModuleRtpRtcpImpl::OnReceivedReferencePictureSelectionIndication(
- const WebRtc_UWord64 pictureID) {
-
- if (_defaultModule) {
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- if (_defaultModule) {
- // if we use a default module pass this info to the default module
- _defaultModule->OnReceivedReferencePictureSelectionIndication(
- pictureID);
- return;
- }
- }
- _rtcpReceiver.OnReceivedReferencePictureSelectionIndication(pictureID);
-}
-
-void ModuleRtpRtcpImpl::OnReceivedBandwidthEstimateUpdate(
- const WebRtc_UWord16 bwEstimateKbit) {
-
- // We received a TMMBR
- const bool defaultInstance(_childModules.empty() ? false : true);
- if (defaultInstance) {
- ProcessDefaultModuleBandwidth();
- return;
- }
- if (_audio) {
- _rtcpReceiver.UpdateBandwidthEstimate(bwEstimateKbit);
- } else {
- WebRtc_UWord32 newBitrate = 0;
- WebRtc_UWord8 fractionLost = 0;
- WebRtc_UWord16 roundTripTime = 0;
- if (_bandwidthManagement.UpdateBandwidthEstimate(bwEstimateKbit,
- &newBitrate,
- &fractionLost,
- &roundTripTime) == 0) {
- if (!_defaultModule) {
- // No default module check if we should trigger OnNetworkChanged
- // via video callback
- _rtpReceiver.UpdateBandwidthManagement(newBitrate,
- fractionLost,
- roundTripTime);
- }
- if (newBitrate > 0) {
- // update bitrate
- _rtpSender.SetTargetSendBitrate(newBitrate);
- }
- }
- }
- if (_defaultModule) {
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- if (_defaultModule) {
- // if we use a default module pass this info to the default module
- _defaultModule->OnReceivedBandwidthEstimateUpdate(bwEstimateKbit);
- return;
- }
- }
-}
-
-// bw estimation
-// We received a RTCP report block
-void ModuleRtpRtcpImpl::OnPacketLossStatisticsUpdate(
- const WebRtc_UWord8 fractionLost,
- const WebRtc_UWord16 roundTripTime,
- const WebRtc_UWord32 lastReceivedExtendedHighSeqNum) {
-
- const bool defaultInstance(_childModules.empty() ? false : true);
- if (!defaultInstance) {
- WebRtc_UWord32 newBitrate = 0;
- WebRtc_UWord8 loss = fractionLost; // local copy since it can change
- WebRtc_UWord32 videoRate = 0;
- WebRtc_UWord32 fecRate = 0;
- WebRtc_UWord32 nackRate = 0;
- BitrateSent(NULL, &videoRate, &fecRate, &nackRate);
- if (_bandwidthManagement.UpdatePacketLoss(
- lastReceivedExtendedHighSeqNum,
- videoRate + fecRate + nackRate,
- roundTripTime,
- &loss,
- &newBitrate,
- _clock.GetTimeInMS()) != 0) {
- // ignore this update
- return;
- }
- // We need to do update RTP sender before calling default module in
- // case we'll strip any layers.
- if (!_simulcast) {
- // the default module will inform all child modules about
- // their bitrate
- _rtpSender.SetTargetSendBitrate(newBitrate);
- }
- if (_defaultModule) {
- // if we have a default module update it
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- if (_defaultModule) { // we need to check again inside the critsect
- // if we use a default module pass this info to the
- // default module
- _defaultModule->OnPacketLossStatisticsUpdate(
- loss, // send in the filtered loss
- roundTripTime,
- lastReceivedExtendedHighSeqNum);
- }
- return;
- }
- _rtpReceiver.UpdateBandwidthManagement(newBitrate,
- fractionLost,
- roundTripTime);
- } else {
- if (!_simulcast) {
- ProcessDefaultModuleBandwidth();
- } else {
- // default and simulcast
- WebRtc_UWord32 newBitrate = 0;
- WebRtc_UWord8 loss = fractionLost; // local copy
- WebRtc_UWord32 videoRate = 0;
- WebRtc_UWord32 fecRate = 0;
- WebRtc_UWord32 nackRate = 0;
- BitrateSent(NULL, &videoRate, &fecRate, &nackRate);
- if (_bandwidthManagement.UpdatePacketLoss(0, // we can't use this
- videoRate + fecRate + nackRate,
- roundTripTime,
- &loss,
- &newBitrate,
- _clock.GetTimeInMS()) != 0) {
- // ignore this update
- return;
- }
- _rtpSender.SetTargetSendBitrate(newBitrate);
- _rtpReceiver.UpdateBandwidthManagement(newBitrate,
- loss,
- roundTripTime);
- // sanity
- if (_sendVideoCodec.codecType == kVideoCodecUnknown) {
- return;
- }
- CriticalSectionScoped lock(_criticalSectionModulePtrsFeedback);
- std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
- WebRtc_UWord8 idx = 0;
- while (it != _childModules.end()) {
- // sanity
- if (idx >= (_sendVideoCodec.numberOfSimulcastStreams - 1)) {
- return;
- }
- ModuleRtpRtcpImpl* module = *it;
- // update all child modules
- if (newBitrate >=
- _sendVideoCodec.simulcastStream[idx].maxBitrate) {
- module->_bandwidthManagement.SetSendBitrate(
- _sendVideoCodec.simulcastStream[idx].maxBitrate, 0, 0);
- module->_rtpSender.SetTargetSendBitrate(
- _sendVideoCodec.simulcastStream[idx].maxBitrate);
-
- newBitrate -=
- _sendVideoCodec.simulcastStream[idx].maxBitrate;
- } else {
- module->_bandwidthManagement.SetSendBitrate(newBitrate,
- 0,
- 0);
- module->_rtpSender.SetTargetSendBitrate(newBitrate);
- newBitrate -= newBitrate;
- }
- idx++;
- }
- }
- }
-}
-
-void ModuleRtpRtcpImpl::ProcessDefaultModuleBandwidth() {
-
- WebRtc_UWord32 minBitrateBps = 0xffffffff;
- WebRtc_UWord32 maxBitrateBps = 0;
- WebRtc_UWord32 count = 0;
- WebRtc_UWord32 fractionLostAcc = 0;
- WebRtc_UWord16 maxRoundTripTime = 0;
- {
- // get min and max for the sending channels
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- for (std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
- it != _childModules.end(); ++ it) {
- // Get child RTP sender and ask for bitrate estimate.
- ModuleRtpRtcpImpl* childModule = *it;
- if (childModule->Sending()) {
- RTPSender& childRtpSender = (*it)->_rtpSender;
- const WebRtc_UWord32 childEstimateBps =
- 1000 * childRtpSender.TargetSendBitrateKbit();
- if (childEstimateBps < minBitrateBps) {
- minBitrateBps = childEstimateBps;
- }
- if (childEstimateBps > maxBitrateBps) {
- maxBitrateBps = childEstimateBps;
- }
- RTCPReceiver& childRtcpReceiver = (*it)->_rtcpReceiver;
-
- std::vector<RTCPReportBlock> rtcp_blocks;
- childRtcpReceiver.StatisticsReceived(&rtcp_blocks);
- for (std::vector<RTCPReportBlock>::iterator rit = rtcp_blocks.begin();
- rit != rtcp_blocks.end(); ++rit) {
- count++;
- fractionLostAcc += rit->fractionLost;
- WebRtc_UWord16 RTT = 0;
- childRtcpReceiver.RTT(rit->remoteSSRC, &RTT, NULL, NULL, NULL);
- maxRoundTripTime = (RTT > maxRoundTripTime) ? RTT : maxRoundTripTime;
- }
- }
- }
- } // end critsect
-
- if (count == 0) {
- // No sending modules and no bitrate estimate.
- return;
- }
-
- // Update RTT to all receive only child modules, they won't have their own RTT
- // estimate. Assume the receive only channels are on similar links as the
- // sending channel and have approximately the same RTT.
- {
- CriticalSectionScoped lock(_criticalSectionModulePtrs);
- for (std::list<ModuleRtpRtcpImpl*>::iterator it = _childModules.begin();
- it != _childModules.end(); ++it) {
- if (!(*it)->Sending()) {
- (*it)->_rtcpReceiver.SetRTT(maxRoundTripTime);
- }
- }
- }
-
- _bandwidthManagement.SetSendBitrate(minBitrateBps, 0, 0);
-
- // Update default module bitrate. Don't care about min max.
- WebRtc_UWord8 fractionLostAvg = WebRtc_UWord8(fractionLostAcc / count);
- _rtpReceiver.UpdateBandwidthManagement(minBitrateBps,
- fractionLostAvg ,
- maxRoundTripTime);
-}
-
-void ModuleRtpRtcpImpl::OnRequestSendReport() {
- _rtcpSender.SendRTCP(kRtcpSr);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::SendRTCPReferencePictureSelection(
- const WebRtc_UWord64 pictureID) {
- return _rtcpSender.SendRTCP(kRtcpRpsi, 0, 0, false, pictureID);
-}
-
-WebRtc_UWord32 ModuleRtpRtcpImpl::SendTimeOfSendReport(
- const WebRtc_UWord32 sendReport) {
- return _rtcpSender.SendTimeOfSendReport(sendReport);
-}
-
-void ModuleRtpRtcpImpl::OnReceivedNACK(
- const WebRtc_UWord16 nackSequenceNumbersLength,
- const WebRtc_UWord16* nackSequenceNumbers) {
- if (!_rtpSender.StorePackets() ||
- nackSequenceNumbers == NULL ||
- nackSequenceNumbersLength == 0) {
- return;
- }
- WebRtc_UWord16 avgRTT = 0;
- _rtcpReceiver.RTT(_rtpReceiver.SSRC(), NULL, &avgRTT, NULL, NULL);
- _rtpSender.OnReceivedNACK(nackSequenceNumbersLength,
- nackSequenceNumbers,
- avgRTT);
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::LastReceivedNTP(
- WebRtc_UWord32& RTCPArrivalTimeSecs, // when we received the last report
- WebRtc_UWord32& RTCPArrivalTimeFrac,
- WebRtc_UWord32& remoteSR) {
- // remote SR: NTP inside the last received (mid 16 bits from sec and frac)
- WebRtc_UWord32 NTPsecs = 0;
- WebRtc_UWord32 NTPfrac = 0;
-
- if (-1 == _rtcpReceiver.NTP(&NTPsecs,
- &NTPfrac,
- &RTCPArrivalTimeSecs,
- &RTCPArrivalTimeFrac)) {
- return -1;
- }
- remoteSR = ((NTPsecs & 0x0000ffff) << 16) + ((NTPfrac & 0xffff0000) >> 16);
- return 0;
-}
-
-void ModuleRtpRtcpImpl::OnReceivedTMMBR() {
- // we received a TMMBR in a RTCP packet
- // answer with a TMMBN
- UpdateTMMBR();
-}
-
-bool ModuleRtpRtcpImpl::UpdateRTCPReceiveInformationTimers() {
- // if this returns true this channel has timed out
- // periodically check if this is true and if so call UpdateTMMBR
- return _rtcpReceiver.UpdateRTCPReceiveInformationTimers();
-}
-
-WebRtc_Word32 ModuleRtpRtcpImpl::UpdateTMMBR() {
- WebRtc_Word32 numBoundingSet = 0;
- WebRtc_Word32 newBitrates = 0;
- WebRtc_UWord32 minBitrateKbit = 0;
- WebRtc_UWord32 maxBitrateKbit = 0;
- WebRtc_UWord32 accNumCandidates = 0;
-
- if (!_childModules.empty()) {
- // Default module should not handle this
- return -1;
- }
-
- WebRtc_Word32 size = _rtcpReceiver.TMMBRReceived(0, 0, NULL);
- if (size > 0) {
- TMMBRSet* candidateSet = VerifyAndAllocateCandidateSet(size);
- // get candidate set from receiver
- accNumCandidates = _rtcpReceiver.TMMBRReceived(size,
- accNumCandidates,
- candidateSet);
- } else {
- // candidate set empty
- VerifyAndAllocateCandidateSet(0); // resets candidate set
- }
- // Find bounding set
- TMMBRSet* boundingSet = NULL;
- numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- if (numBoundingSet == -1) {
- WEBRTC_TRACE(kTraceWarning,
- kTraceRtpRtcp,
- _id,
- "Failed to find TMMBR bounding set.");
- return -1;
- }
- // Set bounding set
- // Inform remote clients about the new bandwidth
- // inform the remote client
- _rtcpSender.SetTMMBN(boundingSet,
- _rtpSender.MaxConfiguredBitrateVideo() / 1000);
- // might trigger a TMMBN
- if (numBoundingSet == 0) {
- // owner of max bitrate request has timed out
- // empty bounding set has been sent
- return 0;
- }
- // Get net bitrate from bounding set depending on sent packet rate
- newBitrates = CalcMinMaxBitRate(_rtpSender.PacketRate(),
- (WebRtc_UWord32)numBoundingSet,
- minBitrateKbit,
- maxBitrateKbit);
-
- // no critsect when calling out to "unknown" code
- if (newBitrates == 0) {
- // we have new bitrate
- // Set new max bitrate
- // we have a new bandwidth estimate on this channel
- OnReceivedBandwidthEstimateUpdate((WebRtc_UWord16)minBitrateKbit);
- WEBRTC_TRACE(kTraceStream,
- kTraceRtpRtcp,
- _id,
- "Set TMMBR request min:%d kbps max:%d kbps, channel: %d",
- minBitrateKbit, maxBitrateKbit, _id);
- }
- return 0;
-}
-
-// called from RTCPsender
-WebRtc_Word32 ModuleRtpRtcpImpl::BoundingSet(bool& tmmbrOwner,
- TMMBRSet*& boundingSet) {
- return _rtcpReceiver.BoundingSet(tmmbrOwner,
- boundingSet);
-}
-
-void ModuleRtpRtcpImpl::SendKeyFrame() {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id, "SendKeyFrame()");
- OnReceivedIntraFrameRequest(0);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h b/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
deleted file mode 100644
index e16bf71..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_impl.h
+++ /dev/null
@@ -1,622 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
-
-#include <list>
-
-#include "bandwidth_management.h"
-#include "rtcp_receiver.h"
-#include "rtcp_sender.h"
-#include "rtp_receiver.h"
-#include "rtp_rtcp.h"
-#include "rtp_sender.h"
-
-#ifdef MATLAB
-class MatlabPlot;
-#endif
-
-namespace webrtc {
-
-class ModuleRtpRtcpImpl : public RtpRtcp, private TMMBRHelp
-{
-public:
- ModuleRtpRtcpImpl(const WebRtc_Word32 id,
- const bool audio,
- RtpRtcpClock* clock);
-
- virtual ~ModuleRtpRtcpImpl();
-
- // get Module ID
- WebRtc_Word32 Id() {return _id;}
-
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- // De-muxing functionality for
- virtual WebRtc_Word32 RegisterDefaultModule(RtpRtcp* module);
- virtual WebRtc_Word32 DeRegisterDefaultModule();
- virtual bool DefaultModuleRegistered();
-
- virtual WebRtc_UWord32 NumberChildModules();
-
- // Lip-sync between voice-video
- virtual WebRtc_Word32 RegisterSyncModule(RtpRtcp* module);
- virtual WebRtc_Word32 DeRegisterSyncModule();
-
- virtual WebRtc_Word32 RegisterVideoModule(RtpRtcp* videoModule);
- virtual void DeRegisterVideoModule();
-
- // returns the number of milliseconds until the module want a worker thread to call Process
- virtual WebRtc_Word32 TimeUntilNextProcess();
-
- // Process any pending tasks such as timeouts
- virtual WebRtc_Word32 Process();
-
- /**
- * Receiver
- */
- virtual WebRtc_Word32 InitReceiver();
-
- // configure a timeout value
- virtual WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 RTPtimeoutMS,
- const WebRtc_UWord32 RTCPtimeoutMS);
-
- // Set periodic dead or alive notification
- virtual WebRtc_Word32 SetPeriodicDeadOrAliveStatus(
- const bool enable,
- const WebRtc_UWord8 sampleTimeSeconds);
-
- // Get periodic dead or alive notification status
- virtual WebRtc_Word32 PeriodicDeadOrAliveStatus(
- bool &enable,
- WebRtc_UWord8 &sampleTimeSeconds);
-
- virtual WebRtc_Word32 RegisterReceivePayload(const CodecInst& voiceCodec);
-
- virtual WebRtc_Word32 RegisterReceivePayload(const VideoCodec& videoCodec);
-
- virtual WebRtc_Word32 ReceivePayloadType(const CodecInst& voiceCodec,
- WebRtc_Word8* plType);
-
- virtual WebRtc_Word32 ReceivePayloadType(const VideoCodec& videoCodec,
- WebRtc_Word8* plType);
-
- virtual WebRtc_Word32 DeRegisterReceivePayload(
- const WebRtc_Word8 payloadType);
-
- // register RTP header extension
- virtual WebRtc_Word32 RegisterReceiveRtpHeaderExtension(
- const RTPExtensionType type,
- const WebRtc_UWord8 id);
-
- virtual WebRtc_Word32 DeregisterReceiveRtpHeaderExtension(
- const RTPExtensionType type);
-
- // get the currently configured SSRC filter
- virtual WebRtc_Word32 SSRCFilter(WebRtc_UWord32& allowedSSRC) const;
-
- // set a SSRC to be used as a filter for incoming RTP streams
- virtual WebRtc_Word32 SetSSRCFilter(const bool enable, const WebRtc_UWord32 allowedSSRC);
-
- // Get last received remote timestamp
- virtual WebRtc_UWord32 RemoteTimestamp() const;
-
- // Get the current estimated remote timestamp
- virtual WebRtc_Word32 EstimatedRemoteTimeStamp(WebRtc_UWord32& timestamp) const;
-
- virtual WebRtc_UWord32 RemoteSSRC() const;
-
- virtual WebRtc_Word32 RemoteCSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const ;
-
- virtual WebRtc_Word32 SetRTXReceiveStatus(const bool enable,
- const WebRtc_UWord32 SSRC);
-
- virtual WebRtc_Word32 RTXReceiveStatus(bool* enable,
- WebRtc_UWord32* SSRC) const;
-
- // called by the network module when we receive a packet
- virtual WebRtc_Word32 IncomingPacket( const WebRtc_UWord8* incomingPacket,
- const WebRtc_UWord16 packetLength);
-
- virtual WebRtc_Word32 IncomingAudioNTP(const WebRtc_UWord32 audioReceivedNTPsecs,
- const WebRtc_UWord32 audioReceivedNTPfrac,
- const WebRtc_UWord32 audioRTCPArrivalTimeSecs,
- const WebRtc_UWord32 audioRTCPArrivalTimeFrac);
-
- // Used by the module to deliver the incoming data to the codec module
- virtual WebRtc_Word32 RegisterIncomingDataCallback(RtpData* incomingDataCallback);
-
- // Used by the module to deliver messages to the codec module/appliation
- virtual WebRtc_Word32 RegisterIncomingRTPCallback(RtpFeedback* incomingMessagesCallback);
-
- virtual WebRtc_Word32 RegisterIncomingRTCPCallback(RtcpFeedback* incomingMessagesCallback);
-
- virtual WebRtc_Word32 RegisterIncomingVideoCallback(RtpVideoFeedback* incomingMessagesCallback);
-
- virtual WebRtc_Word32 RegisterAudioCallback(RtpAudioFeedback* messagesCallback);
-
- /**
- * Sender
- */
- virtual WebRtc_Word32 InitSender();
-
- virtual WebRtc_Word32 SetRTPKeepaliveStatus(const bool enable,
- const WebRtc_Word8 unknownPayloadType,
- const WebRtc_UWord16 deltaTransmitTimeMS);
-
- virtual WebRtc_Word32 RTPKeepaliveStatus(bool* enable,
- WebRtc_Word8* unknownPayloadType,
- WebRtc_UWord16* deltaTransmitTimeMS) const;
-
- virtual bool RTPKeepalive() const;
-
- virtual WebRtc_Word32 RegisterSendPayload(const CodecInst& voiceCodec);
-
- virtual WebRtc_Word32 RegisterSendPayload(const VideoCodec& videoCodec);
-
- virtual WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payloadType);
-
- virtual WebRtc_Word8 SendPayloadType() const;
-
- // register RTP header extension
- virtual WebRtc_Word32 RegisterSendRtpHeaderExtension(
- const RTPExtensionType type,
- const WebRtc_UWord8 id);
-
- virtual WebRtc_Word32 DeregisterSendRtpHeaderExtension(
- const RTPExtensionType type);
-
- virtual void SetTransmissionSmoothingStatus(const bool enable);
-
- virtual bool TransmissionSmoothingStatus() const;
-
- // get start timestamp
- virtual WebRtc_UWord32 StartTimestamp() const;
-
- // configure start timestamp, default is a random number
- virtual WebRtc_Word32 SetStartTimestamp(const WebRtc_UWord32 timestamp);
-
- virtual WebRtc_UWord16 SequenceNumber() const;
-
- // Set SequenceNumber, default is a random number
- virtual WebRtc_Word32 SetSequenceNumber(const WebRtc_UWord16 seq);
-
- virtual WebRtc_UWord32 SSRC() const;
-
- // configure SSRC, default is a random number
- virtual WebRtc_Word32 SetSSRC(const WebRtc_UWord32 ssrc);
-
- virtual WebRtc_Word32 CSRCs( WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const ;
-
- virtual WebRtc_Word32 SetCSRCs( const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
- const WebRtc_UWord8 arrLength);
-
- virtual WebRtc_Word32 SetCSRCStatus(const bool include);
-
- virtual WebRtc_UWord32 PacketCountSent() const;
-
- virtual int CurrentSendFrequencyHz() const;
-
- virtual WebRtc_UWord32 ByteCountSent() const;
-
- virtual WebRtc_Word32 SetRTXSendStatus(const bool enable,
- const bool setSSRC,
- const WebRtc_UWord32 SSRC);
-
- virtual WebRtc_Word32 RTXSendStatus(bool* enable,
- WebRtc_UWord32* SSRC) const;
-
- // sends kRtcpByeCode when going from true to false
- virtual WebRtc_Word32 SetSendingStatus(const bool sending);
-
- virtual bool Sending() const;
-
- // Drops or relays media packets
- virtual WebRtc_Word32 SetSendingMediaStatus(const bool sending);
-
- virtual bool SendingMedia() const;
-
- // Used by the module to send RTP and RTCP packet to the network module
- virtual WebRtc_Word32 RegisterSendTransport(Transport* outgoingTransport);
-
- // Used by the codec module to deliver a video or audio frame for packetization
- virtual WebRtc_Word32 SendOutgoingData(
- const FrameType frameType,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader* fragmentation = NULL,
- const RTPVideoHeader* rtpVideoHdr = NULL);
-
- /*
- * RTCP
- */
-
- // Get RTCP status
- virtual RTCPMethod RTCP() const;
-
- // configure RTCP status i.e on/off
- virtual WebRtc_Word32 SetRTCPStatus(const RTCPMethod method);
-
- // Set RTCP CName
- virtual WebRtc_Word32 SetCNAME(const char cName[RTCP_CNAME_SIZE]);
-
- // Get RTCP CName
- virtual WebRtc_Word32 CNAME(char cName[RTCP_CNAME_SIZE]);
-
- // Get remote CName
- virtual WebRtc_Word32 RemoteCNAME(const WebRtc_UWord32 remoteSSRC,
- char cName[RTCP_CNAME_SIZE]) const;
-
- // Get remote NTP
- virtual WebRtc_Word32 RemoteNTP(WebRtc_UWord32 *ReceivedNTPsecs,
- WebRtc_UWord32 *ReceivedNTPfrac,
- WebRtc_UWord32 *RTCPArrivalTimeSecs,
- WebRtc_UWord32 *RTCPArrivalTimeFrac) const ;
-
- virtual WebRtc_Word32 AddMixedCNAME(const WebRtc_UWord32 SSRC,
- const char cName[RTCP_CNAME_SIZE]);
-
- virtual WebRtc_Word32 RemoveMixedCNAME(const WebRtc_UWord32 SSRC);
-
- // Get RoundTripTime
- virtual WebRtc_Word32 RTT(const WebRtc_UWord32 remoteSSRC,
- WebRtc_UWord16* RTT,
- WebRtc_UWord16* avgRTT,
- WebRtc_UWord16* minRTT,
- WebRtc_UWord16* maxRTT) const;
-
- // Reset RoundTripTime statistics
- virtual WebRtc_Word32 ResetRTT(const WebRtc_UWord32 remoteSSRC);
-
- // Force a send of an RTCP packet
- // normal SR and RR are triggered via the process function
- virtual WebRtc_Word32 SendRTCP(WebRtc_UWord32 rtcpPacketType = kRtcpReport);
-
- // statistics of our localy created statistics of the received RTP stream
- virtual WebRtc_Word32 StatisticsRTP(WebRtc_UWord8 *fraction_lost,
- WebRtc_UWord32 *cum_lost,
- WebRtc_UWord32 *ext_max,
- WebRtc_UWord32 *jitter,
- WebRtc_UWord32 *max_jitter = NULL) const;
-
- // Reset RTP statistics
- virtual WebRtc_Word32 ResetStatisticsRTP();
-
- virtual WebRtc_Word32 ResetReceiveDataCountersRTP();
-
- virtual WebRtc_Word32 ResetSendDataCountersRTP();
-
- // statistics of the amount of data sent and received
- virtual WebRtc_Word32 DataCountersRTP(WebRtc_UWord32 *bytesSent,
- WebRtc_UWord32 *packetsSent,
- WebRtc_UWord32 *bytesReceived,
- WebRtc_UWord32 *packetsReceived) const;
-
- virtual WebRtc_Word32 ReportBlockStatistics(
- WebRtc_UWord8 *fraction_lost,
- WebRtc_UWord32 *cum_lost,
- WebRtc_UWord32 *ext_max,
- WebRtc_UWord32 *jitter,
- WebRtc_UWord32 *jitter_transmission_time_offset);
-
- // Get received RTCP report, sender info
- virtual WebRtc_Word32 RemoteRTCPStat( RTCPSenderInfo* senderInfo);
-
- // Get received RTCP report, report block
- virtual WebRtc_Word32 RemoteRTCPStat(
- std::vector<RTCPReportBlock>* receiveBlocks) const;
-
- // Set received RTCP report block
- virtual WebRtc_Word32 AddRTCPReportBlock(const WebRtc_UWord32 SSRC,
- const RTCPReportBlock* receiveBlock);
-
- virtual WebRtc_Word32 RemoveRTCPReportBlock(const WebRtc_UWord32 SSRC);
-
- /*
- * (REMB) Receiver Estimated Max Bitrate
- */
- virtual bool REMB() const;
-
- virtual WebRtc_Word32 SetREMBStatus(const bool enable);
-
- virtual WebRtc_Word32 SetREMBData(const WebRtc_UWord32 bitrate,
- const WebRtc_UWord8 numberOfSSRC,
- const WebRtc_UWord32* SSRC);
-
- virtual WebRtc_Word32 SetMaximumBitrateEstimate(
- const WebRtc_UWord32 bitrate);
-
- virtual bool SetRemoteBitrateObserver(RtpRemoteBitrateObserver* observer);
- /*
- * (IJ) Extended jitter report.
- */
- virtual bool IJ() const;
-
- virtual WebRtc_Word32 SetIJStatus(const bool enable);
-
- /*
- * (TMMBR) Temporary Max Media Bit Rate
- */
- virtual bool TMMBR() const ;
-
- virtual WebRtc_Word32 SetTMMBRStatus(const bool enable);
-
- virtual WebRtc_Word32 TMMBRReceived(const WebRtc_UWord32 size,
- const WebRtc_UWord32 accNumCandidates,
- TMMBRSet* candidateSet) const;
-
- virtual WebRtc_Word32 SetTMMBN(const TMMBRSet* boundingSet,
- const WebRtc_UWord32 maxBitrateKbit);
-
- virtual WebRtc_Word32 RequestTMMBR(const WebRtc_UWord32 estimatedBW,
- const WebRtc_UWord32 packetOH);
-
- virtual WebRtc_UWord16 MaxPayloadLength() const;
-
- virtual WebRtc_UWord16 MaxDataPayloadLength() const;
-
- virtual WebRtc_Word32 SetMaxTransferUnit(const WebRtc_UWord16 size);
-
- virtual WebRtc_Word32 SetTransportOverhead(const bool TCP,
- const bool IPV6,
- const WebRtc_UWord8 authenticationOverhead = 0);
-
- /*
- * (NACK) Negative acknowledgement
- */
-
- // Is Negative acknowledgement requests on/off?
- virtual NACKMethod NACK() const ;
-
- // Turn negative acknowledgement requests on/off
- virtual WebRtc_Word32 SetNACKStatus(const NACKMethod method);
-
- virtual int SelectiveRetransmissions() const;
-
- virtual int SetSelectiveRetransmissions(uint8_t settings);
-
- // Send a Negative acknowledgement packet
- virtual WebRtc_Word32 SendNACK(const WebRtc_UWord16* nackList,
- const WebRtc_UWord16 size);
-
- // Store the sent packets, needed to answer to a Negative acknowledgement requests
- virtual WebRtc_Word32 SetStorePacketsStatus(const bool enable, const WebRtc_UWord16 numberToStore = 200);
-
- /*
- * (APP) Application specific data
- */
- virtual WebRtc_Word32 SetRTCPApplicationSpecificData(const WebRtc_UWord8 subType,
- const WebRtc_UWord32 name,
- const WebRtc_UWord8* data,
- const WebRtc_UWord16 length);
- /*
- * (XR) VOIP metric
- */
- virtual WebRtc_Word32 SetRTCPVoIPMetrics(const RTCPVoIPMetric* VoIPMetric);
-
- /*
- * Audio
- */
-
- // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
- virtual WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples);
-
- // Outband DTMF detection
- virtual WebRtc_Word32 SetTelephoneEventStatus(const bool enable,
- const bool forwardToDecoder,
- const bool detectEndOfTone = false);
-
- // Is outband DTMF turned on/off?
- virtual bool TelephoneEvent() const;
-
- // Is forwarding of outband telephone events turned on/off?
- virtual bool TelephoneEventForwardToDecoder() const;
-
- virtual bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const;
-
- // Send a TelephoneEvent tone using RFC 2833 (4733)
- virtual WebRtc_Word32 SendTelephoneEventOutband(const WebRtc_UWord8 key,
- const WebRtc_UWord16 time_ms,
- const WebRtc_UWord8 level);
-
- // Set payload type for Redundant Audio Data RFC 2198
- virtual WebRtc_Word32 SetSendREDPayloadType(const WebRtc_Word8 payloadType);
-
- // Get payload type for Redundant Audio Data RFC 2198
- virtual WebRtc_Word32 SendREDPayloadType(WebRtc_Word8& payloadType) const;
-
- // Set status and ID for header-extension-for-audio-level-indication.
- virtual WebRtc_Word32 SetRTPAudioLevelIndicationStatus(const bool enable,
- const WebRtc_UWord8 ID);
-
- // Get status and ID for header-extension-for-audio-level-indication.
- virtual WebRtc_Word32 GetRTPAudioLevelIndicationStatus(bool& enable,
- WebRtc_UWord8& ID) const;
-
- // Store the audio level in dBov for header-extension-for-audio-level-indication.
- virtual WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov);
-
- /*
- * Video
- */
- virtual RtpVideoCodecTypes ReceivedVideoCodec() const;
-
- virtual RtpVideoCodecTypes SendVideoCodec() const;
-
- virtual WebRtc_Word32 SendRTCPSliceLossIndication(const WebRtc_UWord8 pictureID);
-
- // Set method for requestion a new key frame
- virtual WebRtc_Word32 SetKeyFrameRequestMethod(const KeyFrameRequestMethod method);
-
- // send a request for a keyframe
- virtual WebRtc_Word32 RequestKeyFrame();
-
- virtual WebRtc_Word32 SetCameraDelay(const WebRtc_Word32 delayMS);
-
- virtual void SetSendBitrate(const WebRtc_UWord32 startBitrate,
- const WebRtc_UWord16 minBitrateKbit,
- const WebRtc_UWord16 maxBitrateKbit);
-
- virtual WebRtc_Word32 SetGenericFECStatus(const bool enable,
- const WebRtc_UWord8 payloadTypeRED,
- const WebRtc_UWord8 payloadTypeFEC);
-
- virtual WebRtc_Word32 GenericFECStatus(bool& enable,
- WebRtc_UWord8& payloadTypeRED,
- WebRtc_UWord8& payloadTypeFEC);
-
-
- virtual WebRtc_Word32 SetFECCodeRate(const WebRtc_UWord8 keyFrameCodeRate,
- const WebRtc_UWord8 deltaFrameCodeRate);
-
- virtual WebRtc_Word32 SetFECUepProtection(const bool keyUseUepProtection,
- const bool deltaUseUepProtection);
-
- virtual WebRtc_Word32 LastReceivedNTP(WebRtc_UWord32& NTPsecs,
- WebRtc_UWord32& NTPfrac,
- WebRtc_UWord32& remoteSR);
-
- virtual WebRtc_Word32 BoundingSet(bool &tmmbrOwner,
- TMMBRSet*& boundingSetRec);
-
- virtual void BitrateSent(WebRtc_UWord32* totalRate,
- WebRtc_UWord32* videoRate,
- WebRtc_UWord32* fecRate,
- WebRtc_UWord32* nackRate) const;
-
- virtual int EstimatedSendBandwidth(
- WebRtc_UWord32* available_bandwidth) const;
-
- virtual int EstimatedReceiveBandwidth(
- WebRtc_UWord32* available_bandwidth) const;
-
- virtual void SetRemoteSSRC(const WebRtc_UWord32 SSRC);
-
- virtual WebRtc_UWord32 SendTimeOfSendReport(const WebRtc_UWord32 sendReport);
-
- virtual RateControlRegion OnOverUseStateUpdate(const RateControlInput& rateControlInput);
-
- // good state of RTP receiver inform sender
- virtual WebRtc_Word32 SendRTCPReferencePictureSelection(const WebRtc_UWord64 pictureID);
-
- virtual void OnBandwidthEstimateUpdate(WebRtc_UWord16 bandWidthKbit);
-
- void OnReceivedNTP() ;
-
- // bw estimation
- void OnPacketLossStatisticsUpdate(
- const WebRtc_UWord8 fractionLost,
- const WebRtc_UWord16 roundTripTime,
- const WebRtc_UWord32 lastReceivedExtendedHighSeqNum);
-
- void OnReceivedTMMBR();
-
- void OnReceivedEstimatedMaxBitrate(const WebRtc_UWord32 maxBitrate);
-
- void OnReceivedBandwidthEstimateUpdate(const WebRtc_UWord16 bwEstimateKbit);
-
- // bad state of RTP receiver request a keyframe
- void OnRequestIntraFrame();
-
- void OnReceivedIntraFrameRequest(const RtpRtcp* caller);
-
- // received a request for a new SLI
- void OnReceivedSliceLossIndication(const WebRtc_UWord8 pictureID);
-
- // received a new refereence frame
- void OnReceivedReferencePictureSelectionIndication(
- const WebRtc_UWord64 pitureID);
-
- void OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength,
- const WebRtc_UWord16* nackSequenceNumbers);
-
- void OnRequestSendReport();
-
- // Following function is only called when constructing the object so no
- // need to worry about data race.
- void OwnsClock() { _owns_clock = true; }
-
-protected:
- void RegisterChildModule(RtpRtcp* module);
-
- void DeRegisterChildModule(RtpRtcp* module);
-
- bool UpdateRTCPReceiveInformationTimers();
-
- void ProcessDeadOrAliveTimer();
-
- WebRtc_UWord32 BitrateReceivedNow() const;
-
- // Get remote SequenceNumber
- WebRtc_UWord16 RemoteSequenceNumber() const;
-
- WebRtc_Word32 UpdateTMMBR();
-
- // only for internal testing
- WebRtc_UWord32 LastSendReport(WebRtc_UWord32& lastRTCPTime);
-
- RTPSender _rtpSender;
- RTPReceiver _rtpReceiver;
-
- RTCPSender _rtcpSender;
- RTCPReceiver _rtcpReceiver;
-
- bool _owns_clock;
- RtpRtcpClock& _clock;
-private:
- void SendKeyFrame();
- void ProcessDefaultModuleBandwidth();
-
- WebRtc_Word32 _id;
- const bool _audio;
- bool _collisionDetected;
- WebRtc_UWord32 _lastProcessTime;
- WebRtc_UWord32 _lastBitrateProcessTime;
- WebRtc_UWord32 _lastPacketTimeoutProcessTime;
- WebRtc_UWord16 _packetOverHead;
-
- CriticalSectionWrapper* _criticalSectionModulePtrs;
- CriticalSectionWrapper* _criticalSectionModulePtrsFeedback;
- ModuleRtpRtcpImpl* _defaultModule;
- ModuleRtpRtcpImpl* _audioModule;
- ModuleRtpRtcpImpl* _videoModule;
- std::list<ModuleRtpRtcpImpl*> _childModules;
-
- // Dead or alive
- bool _deadOrAliveActive;
- WebRtc_UWord32 _deadOrAliveTimeoutMS;
- WebRtc_UWord32 _deadOrAliveLastTimer;
-
- // receive side
- BandwidthManagement _bandwidthManagement;
-
- WebRtc_UWord32 _receivedNTPsecsAudio;
- WebRtc_UWord32 _receivedNTPfracAudio;
- WebRtc_UWord32 _RTCPArrivalTimeSecsAudio;
- WebRtc_UWord32 _RTCPArrivalTimeFracAudio;
-
- // send side
- NACKMethod _nackMethod;
- WebRtc_UWord32 _nackLastTimeSent;
- WebRtc_UWord16 _nackLastSeqNumberSent;
-
- bool _simulcast;
- VideoCodec _sendVideoCodec;
- KeyFrameRequestMethod _keyFrameReqMethod;
-
-#ifdef MATLAB
- MatlabPlot* _plot1;
-#endif
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_RTCP_IMPL_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_tests.gypi b/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_tests.gypi
deleted file mode 100644
index ea705a8..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_rtcp_tests.gypi
+++ /dev/null
@@ -1,47 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'rtp_rtcp_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'rtp_rtcp',
- '<(webrtc_root)/../testing/gmock.gyp:gmock',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '../../../',
- ],
- 'sources': [
- 'receiver_fec_unittest.cc',
- 'rtp_fec_unittest.cc',
- 'rtp_format_vp8_unittest.cc',
- 'rtp_format_vp8_test_helper.cc',
- 'rtp_format_vp8_test_helper.h',
- 'rtcp_format_remb_unittest.cc',
- 'rtp_packet_history_test.cc',
- 'rtp_utility_test.cc',
- 'rtp_header_extension_test.cc',
- 'rtp_sender_test.cc',
- 'rtcp_sender_test.cc',
- 'transmission_bucket_test.cc',
- 'vp8_partition_aggregator_unittest.cc',
- ],
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_sender.cc b/trunk/src/modules/rtp_rtcp/source/rtp_sender.cc
deleted file mode 100644
index 374ce53..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_sender.cc
+++ /dev/null
@@ -1,1768 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cstdlib> // srand
-
-#include "rtp_sender.h"
-
-#include "critical_section_wrapper.h"
-#include "trace.h"
-
-#include "rtp_packet_history.h"
-#include "rtp_sender_audio.h"
-#include "rtp_sender_video.h"
-
-namespace webrtc {
-RTPSender::RTPSender(const WebRtc_Word32 id,
- const bool audio,
- RtpRtcpClock* clock) :
- Bitrate(clock),
- _id(id),
- _audioConfigured(audio),
- _audio(NULL),
- _video(NULL),
- _sendCritsect(CriticalSectionWrapper::CreateCriticalSection()),
- _transportCritsect(CriticalSectionWrapper::CreateCriticalSection()),
-
- _transport(NULL),
-
- _sendingMedia(true), // Default to sending media
-
- _maxPayloadLength(IP_PACKET_SIZE-28), // default is IP/UDP
- _targetSendBitrate(0),
- _packetOverHead(28),
-
- _payloadType(-1),
- _payloadTypeMap(),
-
- _rtpHeaderExtensionMap(),
- _transmissionTimeOffset(0),
-
- _keepAliveIsActive(false),
- _keepAlivePayloadType(-1),
- _keepAliveLastSent(0),
- _keepAliveDeltaTimeSend(0),
-
- // NACK
- _nackByteCountTimes(),
- _nackByteCount(),
- _nackBitrate(clock),
-
- _packetHistory(new RTPPacketHistory(clock)),
- _sendBucket(),
- _timeLastSendToNetworkUpdate(clock->GetTimeInMS()),
- _transmissionSmoothing(false),
-
- // statistics
- _packetsSent(0),
- _payloadBytesSent(0),
-
- // RTP variables
- _startTimeStampForced(false),
- _startTimeStamp(0),
- _ssrcDB(*SSRCDatabase::GetSSRCDatabase()),
- _remoteSSRC(0),
- _sequenceNumberForced(false),
- _sequenceNumber(0),
- _sequenceNumberRTX(0),
- _ssrcForced(false),
- _ssrc(0),
- _timeStamp(0),
- _CSRCs(0),
- _CSRC(),
- _includeCSRCs(true),
- _RTX(false),
- _ssrcRTX(0)
-{
- memset(_nackByteCountTimes, 0, sizeof(_nackByteCountTimes));
- memset(_nackByteCount, 0, sizeof(_nackByteCount));
-
- memset(_CSRC, 0, sizeof(_CSRC));
-
- // we need to seed the random generator, otherwise we get 26500 each time, hardly a random value :)
- srand( (WebRtc_UWord32)_clock.GetTimeInMS() );
-
- _ssrc = _ssrcDB.CreateSSRC(); // can't be 0
-
- if(audio)
- {
- _audio = new RTPSenderAudio(id, &_clock, this);
- } else
- {
- _video = new RTPSenderVideo(id, &_clock, this);
- }
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, id, "%s created", __FUNCTION__);
-}
-
-RTPSender::~RTPSender() {
- if(_remoteSSRC != 0) {
- _ssrcDB.ReturnSSRC(_remoteSSRC);
- }
- _ssrcDB.ReturnSSRC(_ssrc);
-
- SSRCDatabase::ReturnSSRCDatabase();
- delete _sendCritsect;
- delete _transportCritsect;
- while (!_payloadTypeMap.empty()) {
- std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
- _payloadTypeMap.begin();
- delete it->second;
- _payloadTypeMap.erase(it);
- }
- delete _packetHistory;
- delete _audio;
- delete _video;
-
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, _id, "%s deleted", __FUNCTION__);
-}
-
-WebRtc_Word32
-RTPSender::Init(const WebRtc_UWord32 remoteSSRC)
-{
- CriticalSectionScoped cs(_sendCritsect);
-
- // reset to default generation
- _ssrcForced = false;
- _startTimeStampForced = false;
-
- // register a remote SSRC if we have it to avoid collisions
- if(remoteSSRC != 0)
- {
- if(_ssrc == remoteSSRC)
- {
- // collision detected
- _ssrc = _ssrcDB.CreateSSRC(); // can't be 0
- }
- _remoteSSRC = remoteSSRC;
- _ssrcDB.RegisterSSRC(remoteSSRC);
- }
- _sequenceNumber = rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER);
- _sequenceNumberRTX = rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER);
- _packetsSent = 0;
- _payloadBytesSent = 0;
- _packetOverHead = 28;
-
- _keepAlivePayloadType = -1;
-
- _rtpHeaderExtensionMap.Erase();
-
- while (!_payloadTypeMap.empty()) {
- std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
- _payloadTypeMap.begin();
- delete it->second;
- _payloadTypeMap.erase(it);
- }
-
- memset(_CSRC, 0, sizeof(_CSRC));
-
- memset(_nackByteCount, 0, sizeof(_nackByteCount));
- memset(_nackByteCountTimes, 0, sizeof(_nackByteCountTimes));
- _nackBitrate.Init();
-
- SetStorePacketsStatus(false, 0);
- _sendBucket.Reset();
-
- Bitrate::Init();
-
- if(_audioConfigured)
- {
- _audio->Init();
- } else
- {
- _video->Init();
- }
- return(0);
-}
-
-void
-RTPSender::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
- if(_audioConfigured)
- {
- _audio->ChangeUniqueId(id);
- } else
- {
- _video->ChangeUniqueId(id);
- }
-}
-
-WebRtc_Word32
-RTPSender::SetTargetSendBitrate(const WebRtc_UWord32 bits)
-{
- _targetSendBitrate = (WebRtc_UWord16)(bits/1000);
- return 0;
-}
-
-WebRtc_UWord16
-RTPSender::TargetSendBitrateKbit() const
-{
- return _targetSendBitrate;
-}
-
-WebRtc_UWord16
-RTPSender::ActualSendBitrateKbit() const
-{
- return (WebRtc_UWord16) (Bitrate::BitrateNow()/1000);
-}
-
-WebRtc_UWord32
-RTPSender::VideoBitrateSent() const {
- if (_video)
- return _video->VideoBitrateSent();
- else
- return 0;
-}
-
-WebRtc_UWord32
-RTPSender::FecOverheadRate() const {
- if (_video)
- return _video->FecOverheadRate();
- else
- return 0;
-}
-
-WebRtc_UWord32
-RTPSender::NackOverheadRate() const {
- return _nackBitrate.BitrateLast();
-}
-
-WebRtc_Word32
-RTPSender::SetTransmissionTimeOffset(
- const WebRtc_Word32 transmissionTimeOffset)
-{
- if (transmissionTimeOffset > (0x800000 - 1) ||
- transmissionTimeOffset < -(0x800000 - 1)) // Word24
- {
- return -1;
- }
- CriticalSectionScoped cs(_sendCritsect);
- _transmissionTimeOffset = transmissionTimeOffset;
- return 0;
-}
-
-WebRtc_Word32
-RTPSender::RegisterRtpHeaderExtension(const RTPExtensionType type,
- const WebRtc_UWord8 id)
-{
- CriticalSectionScoped cs(_sendCritsect);
- return _rtpHeaderExtensionMap.Register(type, id);
-}
-
-WebRtc_Word32
-RTPSender::DeregisterRtpHeaderExtension(const RTPExtensionType type)
-{
- CriticalSectionScoped cs(_sendCritsect);
- return _rtpHeaderExtensionMap.Deregister(type);
-}
-
-WebRtc_UWord16
-RTPSender::RtpHeaderExtensionTotalLength() const
-{
- CriticalSectionScoped cs(_sendCritsect);
- return _rtpHeaderExtensionMap.GetTotalLengthInBytes();
-}
-
-//can be called multiple times
-WebRtc_Word32 RTPSender::RegisterPayload(
- const char payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadNumber,
- const WebRtc_UWord32 frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate) {
- assert(payloadName);
- CriticalSectionScoped cs(_sendCritsect);
-
- if (payloadNumber == _keepAlivePayloadType) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id, "invalid state",
- __FUNCTION__);
- return -1;
- }
- std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
- _payloadTypeMap.find(payloadNumber);
-
- if (_payloadTypeMap.end() != it) {
- // we already use this payload type
- ModuleRTPUtility::Payload* payload = it->second;
- assert(payload);
-
- // check if it's the same as we already have
- if (ModuleRTPUtility::StringCompare(payload->name, payloadName,
- RTP_PAYLOAD_NAME_SIZE - 1)) {
- if (_audioConfigured && payload->audio &&
- payload->typeSpecific.Audio.frequency == frequency &&
- (payload->typeSpecific.Audio.rate == rate ||
- payload->typeSpecific.Audio.rate == 0 || rate == 0)) {
- payload->typeSpecific.Audio.rate = rate;
- // Ensure that we update the rate if new or old is zero
- return 0;
- }
- if(!_audioConfigured && !payload->audio) {
- return 0;
- }
- }
- return -1;
- }
- WebRtc_Word32 retVal = -1;
- ModuleRTPUtility::Payload* payload = NULL;
- if (_audioConfigured) {
- retVal = _audio->RegisterAudioPayload(payloadName, payloadNumber, frequency,
- channels, rate, payload);
- } else {
- retVal = _video->RegisterVideoPayload(payloadName, payloadNumber, rate,
- payload);
- }
- if(payload) {
- _payloadTypeMap[payloadNumber] = payload;
- }
- return retVal;
-}
-
-WebRtc_Word32 RTPSender::DeRegisterSendPayload(const WebRtc_Word8 payloadType) {
- CriticalSectionScoped lock(_sendCritsect);
-
- std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
- _payloadTypeMap.find(payloadType);
-
- if (_payloadTypeMap.end() == it) return -1;
-
- ModuleRTPUtility::Payload* payload = it->second;
- delete payload;
- _payloadTypeMap.erase(it);
- return 0;
-}
-
-WebRtc_Word8 RTPSender::SendPayloadType() const
-{
- return _payloadType;
-}
-
-
-int RTPSender::SendPayloadFrequency() const
-{
- return _audio->AudioFrequency();
-}
-
-
-// See http://www.ietf.org/internet-drafts/draft-ietf-avt-app-rtp-keepalive-04.txt
-// for details about this method. Only Section 4.6 is implemented so far.
-bool
-RTPSender::RTPKeepalive() const
-{
- return _keepAliveIsActive;
-}
-
-WebRtc_Word32
-RTPSender::RTPKeepaliveStatus(bool* enable,
- WebRtc_Word8* unknownPayloadType,
- WebRtc_UWord16* deltaTransmitTimeMS) const
-{
- CriticalSectionScoped cs(_sendCritsect);
-
- if(enable)
- {
- *enable = _keepAliveIsActive;
- }
- if(unknownPayloadType)
- {
- *unknownPayloadType = _keepAlivePayloadType;
- }
- if(deltaTransmitTimeMS)
- {
- *deltaTransmitTimeMS =_keepAliveDeltaTimeSend;
- }
- return 0;
-}
-
-WebRtc_Word32 RTPSender::EnableRTPKeepalive(
- const WebRtc_Word8 unknownPayloadType,
- const WebRtc_UWord16 deltaTransmitTimeMS) {
- CriticalSectionScoped cs(_sendCritsect);
-
- std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
- _payloadTypeMap.find(unknownPayloadType);
-
- if (it != _payloadTypeMap.end()) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument",
- __FUNCTION__);
- return -1;
- }
- _keepAliveIsActive = true;
- _keepAlivePayloadType = unknownPayloadType;
- _keepAliveLastSent = _clock.GetTimeInMS();
- _keepAliveDeltaTimeSend = deltaTransmitTimeMS;
- return 0;
-}
-
-WebRtc_Word32
-RTPSender::DisableRTPKeepalive()
-{
- _keepAliveIsActive = false;
- return 0;
-}
-
-bool
-RTPSender::TimeToSendRTPKeepalive() const
-{
- CriticalSectionScoped cs(_sendCritsect);
-
- bool timeToSend(false);
-
- WebRtc_UWord32 dT = _clock.GetTimeInMS() - _keepAliveLastSent;
- if (dT > _keepAliveDeltaTimeSend)
- {
- timeToSend = true;
- }
- return timeToSend;
-}
-
-// ----------------------------------------------------------------------------
-// From the RFC draft:
-//
-// 4.6. RTP Packet with Unknown Payload Type
-//
-// The application sends an RTP packet of 0 length with a dynamic
-// payload type that has not been negotiated by the peers (e.g. not
-// negotiated within the SDP offer/answer, and thus not mapped to any
-// media format).
-//
-// The sequence number is incremented by one for each packet, as it is
-// sent within the same RTP session as the actual media. The timestamp
-// contains the same value a media packet would have at this time. The
-// marker bit is not significant for the keepalive packets and is thus
-// set to zero.
-//
-// Normally the peer will ignore this packet, as RTP [RFC3550] states
-// that "a receiver MUST ignore packets with payload types that it does
-// not understand".
-//
-// Cons:
-// o [RFC4566] and [RFC3264] mandate not to send media with inactive
-// and recvonly attributes, however this is mitigated as no real
-// media is sent with this mechanism.
-//
-// Recommendation:
-// o This method should be used for RTP keepalive.
-//
-// 7. Timing and Transport Considerations
-//
-// An application supporting this specification must transmit keepalive
-// packets every Tr seconds during the whole duration of the media
-// session. Tr SHOULD be configurable, and otherwise MUST default to 15
-// seconds.
-//
-// Keepalives packets within a particular RTP session MUST use the tuple
-// (source IP address, source TCP/UDP ports, target IP address, target
-// TCP/UDP Port) of the regular RTP packets.
-//
-// The agent SHOULD only send RTP keepalive when it does not send
-// regular RTP packets.
-//
-// http://www.ietf.org/internet-drafts/draft-ietf-avt-app-rtp-keepalive-04.txt
-// ----------------------------------------------------------------------------
-
-WebRtc_Word32
-RTPSender::SendRTPKeepalivePacket()
-{
- // RFC summary:
- //
- // - Send an RTP packet of 0 length;
- // - dynamic payload type has not been negotiated (not mapped to any media);
- // - sequence number is incremented by one for each packet;
- // - timestamp contains the same value a media packet would have at this time;
- // - marker bit is set to zero.
-
- WebRtc_UWord8 dataBuffer[IP_PACKET_SIZE];
- WebRtc_UWord16 rtpHeaderLength = 12;
- {
- CriticalSectionScoped cs(_sendCritsect);
-
- WebRtc_UWord32 now = _clock.GetTimeInMS();
- WebRtc_UWord32 dT = now -_keepAliveLastSent; // delta time in MS
-
- WebRtc_UWord32 freqKHz = 90; // video
- if(_audioConfigured)
- {
- freqKHz = _audio->AudioFrequency()/1000;
- }
- WebRtc_UWord32 dSamples = dT*freqKHz;
-
- // set timestamp
- _timeStamp += dSamples;
- _keepAliveLastSent = now;
-
- rtpHeaderLength = RTPHeaderLength();
-
- // correct seq num, time stamp and payloadtype
- BuildRTPheader(dataBuffer, _keepAlivePayloadType, false, 0, false);
- }
-
- return SendToNetwork(dataBuffer, 0, rtpHeaderLength, kAllowRetransmission);
-}
-
-WebRtc_Word32
-RTPSender::SetMaxPayloadLength(const WebRtc_UWord16 maxPayloadLength, const WebRtc_UWord16 packetOverHead)
-{
- // sanity check
- if(maxPayloadLength < 100 || maxPayloadLength > IP_PACKET_SIZE)
- {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument", __FUNCTION__);
- return -1;
- }
-
- CriticalSectionScoped cs(_sendCritsect);
- _maxPayloadLength = maxPayloadLength;
- _packetOverHead = packetOverHead;
-
- WEBRTC_TRACE(kTraceInfo, kTraceRtpRtcp, _id, "SetMaxPayloadLength to %d.", maxPayloadLength);
- return 0;
-}
-
-WebRtc_UWord16 RTPSender::MaxDataPayloadLength() const {
- if(_audioConfigured) {
- return _maxPayloadLength - RTPHeaderLength();
- } else {
- return _maxPayloadLength - RTPHeaderLength() -
- _video->FECPacketOverhead() - ((_RTX) ? 2 : 0);
- // Include the FEC/ULP/RED overhead.
- }
-}
-
-WebRtc_UWord16
-RTPSender::MaxPayloadLength() const
-{
- return _maxPayloadLength;
-}
-
-WebRtc_UWord16
-RTPSender::PacketOverHead() const
-{
- return _packetOverHead;
-}
-
-void RTPSender::SetTransmissionSmoothingStatus(const bool enable) {
- CriticalSectionScoped cs(_sendCritsect);
- _transmissionSmoothing = enable;
-}
-
-bool RTPSender::TransmissionSmoothingStatus() const {
- CriticalSectionScoped cs(_sendCritsect);
- return _transmissionSmoothing;
-}
-
-void RTPSender::SetRTXStatus(const bool enable,
- const bool setSSRC,
- const WebRtc_UWord32 SSRC) {
- CriticalSectionScoped cs(_sendCritsect);
- _RTX = enable;
- if (enable) {
- if (setSSRC) {
- _ssrcRTX = SSRC;
- } else {
- _ssrcRTX = _ssrcDB.CreateSSRC(); // can't be 0
- }
- }
-}
-
-void RTPSender::RTXStatus(bool* enable,
- WebRtc_UWord32* SSRC) const {
- CriticalSectionScoped cs(_sendCritsect);
- *enable = _RTX;
- *SSRC = _ssrcRTX;
-}
-
-WebRtc_Word32 RTPSender::CheckPayloadType(const WebRtc_Word8 payloadType,
- RtpVideoCodecTypes& videoType) {
- CriticalSectionScoped cs(_sendCritsect);
-
- if (payloadType < 0) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "\tinvalid payloadType (%d)", payloadType);
- return -1;
- }
- if (_audioConfigured) {
- WebRtc_Word8 redPlType = -1;
- if (_audio->RED(redPlType) == 0) {
- // We have configured RED.
- if(redPlType == payloadType) {
- // And it's a match...
- return 0;
- }
- }
- }
- if (_payloadType == payloadType) {
- if (!_audioConfigured) {
- videoType = _video->VideoCodecType();
- }
- return 0;
- }
- std::map<WebRtc_Word8, ModuleRTPUtility::Payload*>::iterator it =
- _payloadTypeMap.find(payloadType);
- if (it == _payloadTypeMap.end()) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "\tpayloadType:%d not registered", payloadType);
- return -1;
- }
- _payloadType = payloadType;
- ModuleRTPUtility::Payload* payload = it->second;
- assert(payload);
- if (payload->audio) {
- if (_audioConfigured) {
- // Extract payload frequency
- int payloadFreqHz;
- if (ModuleRTPUtility::StringCompare(payload->name,"g722",4)&&
- (payload->name[4] == 0)) {
- //Check that strings end there, g722.1...
- // Special case for G.722, bug in spec
- payloadFreqHz=8000;
- } else {
- payloadFreqHz=payload->typeSpecific.Audio.frequency;
- }
-
- //we don't do anything if it's CN
- if ((_audio->AudioFrequency() != payloadFreqHz)&&
- (!ModuleRTPUtility::StringCompare(payload->name,"cn",2))) {
- _audio->SetAudioFrequency(payloadFreqHz);
- // We need to correct the timestamp again,
- // since this might happen after we've set it
- WebRtc_UWord32 RTPtime =
- ModuleRTPUtility::GetCurrentRTP(&_clock, payloadFreqHz);
- SetStartTimestamp(RTPtime);
- // will be ignored if it's already configured via API
- }
- }
- } else {
- if(!_audioConfigured) {
- _video->SetVideoCodecType(payload->typeSpecific.Video.videoCodecType);
- videoType = payload->typeSpecific.Video.videoCodecType;
- _video->SetMaxConfiguredBitrateVideo(
- payload->typeSpecific.Video.maxRate);
- }
- }
- return 0;
-}
-
-WebRtc_Word32
-RTPSender::SendOutgoingData(const FrameType frameType,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader* fragmentation,
- VideoCodecInformation* codecInfo,
- const RTPVideoTypeHeader* rtpTypeHdr)
-{
- {
- // Drop this packet if we're not sending media packets
- CriticalSectionScoped cs(_sendCritsect);
- if (!_sendingMedia)
- {
- return 0;
- }
- }
- RtpVideoCodecTypes videoType = kRtpNoVideo;
- if(CheckPayloadType(payloadType, videoType) != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id, "%s invalid argument failed to find payloadType:%d", __FUNCTION__, payloadType);
- return -1;
- }
- // update keepalive so that we don't trigger keepalive messages while sending data
- _keepAliveLastSent = _clock.GetTimeInMS();
-
- if(_audioConfigured)
- {
- // assert video frameTypes
- assert(frameType == kAudioFrameSpeech ||
- frameType == kAudioFrameCN ||
- frameType == kFrameEmpty);
-
- return _audio->SendAudio(frameType, payloadType, captureTimeStamp, payloadData, payloadSize,fragmentation);
- } else
- {
- // assert audio frameTypes
- assert(frameType == kVideoFrameKey ||
- frameType == kVideoFrameDelta ||
- frameType == kVideoFrameGolden ||
- frameType == kVideoFrameAltRef);
-
- return _video->SendVideo(videoType,
- frameType,
- payloadType,
- captureTimeStamp,
- payloadData,
- payloadSize,
- fragmentation,
- codecInfo,
- rtpTypeHdr);
- }
-}
-
-WebRtc_Word32 RTPSender::SendPadData(WebRtc_Word8 payload_type,
- WebRtc_UWord32 capture_timestamp,
- WebRtc_Word32 bytes) {
- // Drop this packet if we're not sending media packets
- if (!_sendingMedia) {
- return 0;
- }
- // Max in the RFC 3550 is 255 bytes, we limit it to be modulus 32 for SRTP.
- int max_length = 224;
- WebRtc_UWord8 data_buffer[IP_PACKET_SIZE];
-
- for (; bytes > 0; bytes -= max_length) {
- WebRtc_Word32 header_length;
- {
- // Correct seq num, timestamp and payload type.
- header_length = BuildRTPheader(data_buffer,
- payload_type,
- false, // No markerbit.
- capture_timestamp,
- true, // Timestamp provided.
- true); // Increment sequence number.
- }
- data_buffer[0] |= 0x20; // Set padding bit.
- WebRtc_Word32* data =
- reinterpret_cast<WebRtc_Word32*>(&(data_buffer[header_length]));
-
- int padding_bytes_in_packet = max_length;
- if (bytes < max_length) {
- padding_bytes_in_packet = (bytes + 16) & 0xffe0; // Keep our modulus 32.
- }
- if (padding_bytes_in_packet < 32) {
- // Sanity don't send empty packets.
- break;
- }
- // Fill data buffer with random data.
- for(int j = 0; j < (padding_bytes_in_packet >> 2); j++) {
- data[j] = rand();
- }
- // Set number of padding bytes in the last byte of the packet.
- data_buffer[header_length + padding_bytes_in_packet - 1] =
- padding_bytes_in_packet;
- // Send the packet
- if (0 > SendToNetwork(data_buffer,
- padding_bytes_in_packet,
- header_length,
- kDontRetransmit)) {
- // Error sending the packet.
- break;
- }
- }
- if (bytes > 31) { // 31 due to our modulus 32.
- // We did not manage to send all bytes.
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 RTPSender::SetStorePacketsStatus(
- const bool enable,
- const WebRtc_UWord16 numberToStore) {
- _packetHistory->SetStorePacketsStatus(enable, numberToStore);
- return 0;
-}
-
-bool RTPSender::StorePackets() const {
- return _packetHistory->StorePackets();
-}
-
-WebRtc_Word32 RTPSender::ReSendPacket(WebRtc_UWord16 packet_id,
- WebRtc_UWord32 min_resend_time) {
-
- WebRtc_UWord16 length = IP_PACKET_SIZE;
- WebRtc_UWord8 data_buffer[IP_PACKET_SIZE];
- WebRtc_UWord8* buffer_to_send_ptr = data_buffer;
-
- WebRtc_UWord32 stored_time_in_ms;
- StorageType type;
- bool found = _packetHistory->GetRTPPacket(packet_id,
- min_resend_time, data_buffer, &length, &stored_time_in_ms, &type);
- if (!found) {
- // Packet not found.
- return -1;
- }
-
- if (length == 0 || type == kDontRetransmit) {
- // No bytes copied (packet recently resent, skip resending) or
- // packet should not be retransmitted.
- return 0;
- }
-
- WebRtc_UWord8 data_buffer_rtx[IP_PACKET_SIZE];
- if (_RTX) {
- buffer_to_send_ptr = data_buffer_rtx;
-
- CriticalSectionScoped cs(_sendCritsect);
- // Add RTX header.
- ModuleRTPUtility::RTPHeaderParser rtpParser(
- reinterpret_cast<const WebRtc_UWord8*>(data_buffer),
- length);
-
- WebRtcRTPHeader rtp_header;
- rtpParser.Parse(rtp_header);
-
- // Add original RTP header.
- memcpy(data_buffer_rtx, data_buffer, rtp_header.header.headerLength);
-
- // Replace sequence number.
- WebRtc_UWord8* ptr = data_buffer_rtx + 2;
- ModuleRTPUtility::AssignUWord16ToBuffer(ptr, _sequenceNumberRTX++);
-
- // Replace SSRC.
- ptr += 6;
- ModuleRTPUtility::AssignUWord32ToBuffer(ptr, _ssrcRTX);
-
- // Add OSN (original sequence number).
- ptr = data_buffer_rtx + rtp_header.header.headerLength;
- ModuleRTPUtility::AssignUWord16ToBuffer(
- ptr, rtp_header.header.sequenceNumber);
- ptr += 2;
-
- // Add original payload data.
- memcpy(ptr,
- data_buffer + rtp_header.header.headerLength,
- length - rtp_header.header.headerLength);
- length += 2;
- }
-
- WebRtc_Word32 bytes_sent = ReSendToNetwork(buffer_to_send_ptr, length);
- if (bytes_sent <= 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, _id,
- "Transport failed to resend packet_id %u", packet_id);
- return -1;
- }
-
- // Store the time when the packet was last resent.
- _packetHistory->UpdateResendTime(packet_id);
-
- return bytes_sent;
-}
-
-WebRtc_Word32 RTPSender::ReSendToNetwork(const WebRtc_UWord8* packet,
- const WebRtc_UWord32 size) {
- WebRtc_Word32 bytes_sent = -1;
- {
- CriticalSectionScoped lock(_transportCritsect);
- if (_transport) {
- bytes_sent = _transport->SendPacket(_id, packet, size);
- }
- }
-
- if (bytes_sent <= 0) {
- return -1;
- }
-
- // Update send statistics
- CriticalSectionScoped cs(_sendCritsect);
- Bitrate::Update(bytes_sent);
- _packetsSent++;
- // We on purpose don't add to _payloadBytesSent since this is a
- // re-transmit and not new payload data.
- return bytes_sent;
-}
-
-int RTPSender::SelectiveRetransmissions() const {
- if (!_video) return -1;
- return _video->SelectiveRetransmissions();
-}
-
-int RTPSender::SetSelectiveRetransmissions(uint8_t settings) {
- if (!_video) return -1;
- return _video->SetSelectiveRetransmissions(settings);
-}
-
-void
-RTPSender::OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength,
- const WebRtc_UWord16* nackSequenceNumbers,
- const WebRtc_UWord16 avgRTT) {
- const WebRtc_UWord32 now = _clock.GetTimeInMS();
- WebRtc_UWord32 bytesReSent = 0;
-
- // Enough bandwidth to send NACK?
- if (!ProcessNACKBitRate(now)) {
- WEBRTC_TRACE(kTraceStream,
- kTraceRtpRtcp,
- _id,
- "NACK bitrate reached. Skip sending NACK response. Target %d",
- TargetSendBitrateKbit());
- return;
- }
-
- for (WebRtc_UWord16 i = 0; i < nackSequenceNumbersLength; ++i) {
- const WebRtc_Word32 bytesSent = ReSendPacket(nackSequenceNumbers[i],
- 5+avgRTT);
- if (bytesSent > 0) {
- bytesReSent += bytesSent;
- } else if (bytesSent == 0) {
- // The packet has previously been resent.
- // Try resending next packet in the list.
- continue;
- } else if (bytesSent < 0) {
- // Failed to send one Sequence number. Give up the rest in this nack.
- WEBRTC_TRACE(kTraceWarning,
- kTraceRtpRtcp,
- _id,
- "Failed resending RTP packet %d, Discard rest of packets",
- nackSequenceNumbers[i]);
- break;
- }
- // delay bandwidth estimate (RTT * BW)
- if (TargetSendBitrateKbit() != 0 && avgRTT) {
- // kbits/s * ms = bits => bits/8 = bytes
- WebRtc_UWord32 targetBytes =
- (static_cast<WebRtc_UWord32>(TargetSendBitrateKbit()) * avgRTT) >> 3;
- if (bytesReSent > targetBytes) {
- break; // ignore the rest of the packets in the list
- }
- }
- }
- if (bytesReSent > 0) {
- // TODO(pwestin) consolidate these two methods.
- UpdateNACKBitRate(bytesReSent, now);
- _nackBitrate.Update(bytesReSent);
- }
-}
-
-/**
-* @return true if the nack bitrate is lower than the requested max bitrate
-*/
-bool RTPSender::ProcessNACKBitRate(const WebRtc_UWord32 now) {
- WebRtc_UWord32 num = 0;
- WebRtc_Word32 byteCount = 0;
- const WebRtc_UWord32 avgInterval=1000;
-
- CriticalSectionScoped cs(_sendCritsect);
-
- if (_targetSendBitrate == 0) {
- return true;
- }
- for (num = 0; num < NACK_BYTECOUNT_SIZE; num++) {
- if ((now - _nackByteCountTimes[num]) > avgInterval) {
- // don't use data older than 1sec
- break;
- } else {
- byteCount += _nackByteCount[num];
- }
- }
- WebRtc_Word32 timeInterval = avgInterval;
- if (num == NACK_BYTECOUNT_SIZE) {
- // More than NACK_BYTECOUNT_SIZE nack messages has been received
- // during the last msgInterval
- timeInterval = now - _nackByteCountTimes[num-1];
- if(timeInterval < 0) {
- timeInterval = avgInterval;
- }
- }
- return (byteCount*8) < (_targetSendBitrate * timeInterval);
-}
-
-void RTPSender::UpdateNACKBitRate(const WebRtc_UWord32 bytes,
- const WebRtc_UWord32 now) {
- CriticalSectionScoped cs(_sendCritsect);
-
- // save bitrate statistics
- if(bytes > 0) {
- if(now == 0) {
- // add padding length
- _nackByteCount[0] += bytes;
- } else {
- if(_nackByteCountTimes[0] == 0) {
- // first no shift
- } else {
- // shift
- for(int i = (NACK_BYTECOUNT_SIZE-2); i >= 0 ; i--) {
- _nackByteCount[i+1] = _nackByteCount[i];
- _nackByteCountTimes[i+1] = _nackByteCountTimes[i];
- }
- }
- _nackByteCount[0] = bytes;
- _nackByteCountTimes[0] = now;
- }
- }
-}
-
-void RTPSender::ProcessSendToNetwork() {
-
- // triggered by timer
- WebRtc_UWord32 delta_time_ms;
- {
- CriticalSectionScoped cs(_sendCritsect);
-
- if (!_transmissionSmoothing) {
- return;
- }
-
- WebRtc_UWord32 now = _clock.GetTimeInMS();
- delta_time_ms = now - _timeLastSendToNetworkUpdate;
- _timeLastSendToNetworkUpdate = now;
- }
-
- _sendBucket.UpdateBytesPerInterval(delta_time_ms, _targetSendBitrate);
-
- while (!_sendBucket.Empty()) {
-
- WebRtc_Word32 seq_num = _sendBucket.GetNextPacket();
- if (seq_num < 0) {
- break;
- }
-
- WebRtc_UWord8 data_buffer[IP_PACKET_SIZE];
- WebRtc_UWord16 length = IP_PACKET_SIZE;
- WebRtc_UWord32 stored_time_ms;
- StorageType type;
- bool found = _packetHistory->GetRTPPacket(seq_num, 0, data_buffer, &length,
- &stored_time_ms, &type);
- if (!found) {
- assert(false);
- return;
- }
- assert(length > 0);
-
- WebRtc_UWord32 diff_ms = _clock.GetTimeInMS() - stored_time_ms;
-
- ModuleRTPUtility::RTPHeaderParser rtpParser(data_buffer, length);
- WebRtcRTPHeader rtp_header;
- rtpParser.Parse(rtp_header);
-
- UpdateTransmissionTimeOffset(data_buffer, length, rtp_header, diff_ms);
-
- // Send packet
- WebRtc_Word32 bytes_sent = -1;
- {
- CriticalSectionScoped cs(_transportCritsect);
- if (_transport) {
- bytes_sent = _transport->SendPacket(_id, data_buffer, length);
- }
- }
-
- // Update send statistics
- if (bytes_sent > 0) {
- CriticalSectionScoped cs(_sendCritsect);
- Bitrate::Update(bytes_sent);
- _packetsSent++;
- if (bytes_sent > rtp_header.header.headerLength) {
- _payloadBytesSent += bytes_sent - rtp_header.header.headerLength;
- }
- }
- }
-}
-
-WebRtc_Word32
-RTPSender::SendToNetwork(const WebRtc_UWord8* buffer,
- const WebRtc_UWord16 length,
- const WebRtc_UWord16 rtpLength,
- const StorageType storage)
-{
- // Used for NACK or to spead out the transmission of packets.
- if (_packetHistory->PutRTPPacket(
- buffer, rtpLength + length, _maxPayloadLength, storage) != 0) {
- return -1;
- }
-
- if (_transmissionSmoothing) {
- const WebRtc_UWord16 sequenceNumber = (buffer[2] << 8) + buffer[3];
- _sendBucket.Fill(sequenceNumber, rtpLength + length);
- // Packet will be sent at a later time.
- return 0;
- }
-
- // Send packet
- WebRtc_Word32 bytes_sent = -1;
- {
- CriticalSectionScoped cs(_transportCritsect);
- if (_transport) {
- bytes_sent = _transport->SendPacket(_id, buffer, length + rtpLength);
- }
- }
-
- if (bytes_sent <= 0) {
- return -1;
- }
-
- // Update send statistics
- CriticalSectionScoped cs(_sendCritsect);
- Bitrate::Update(bytes_sent);
- _packetsSent++;
- if (bytes_sent > rtpLength) {
- _payloadBytesSent += bytes_sent - rtpLength;
- }
- return 0;
-}
-
-void
-RTPSender::ProcessBitrate()
-{
- CriticalSectionScoped cs(_sendCritsect);
-
- Bitrate::Process();
- _nackBitrate.Process();
-
- if (_audioConfigured)
- return;
- _video->ProcessBitrate();
-}
-
-WebRtc_UWord16
-RTPSender::RTPHeaderLength() const
-{
- WebRtc_UWord16 rtpHeaderLength = 12;
-
- if(_includeCSRCs)
- {
- rtpHeaderLength += sizeof(WebRtc_UWord32)*_CSRCs;
- }
- rtpHeaderLength += RtpHeaderExtensionTotalLength();
-
- return rtpHeaderLength;
-}
-
-WebRtc_UWord16
-RTPSender::IncrementSequenceNumber()
-{
- CriticalSectionScoped cs(_sendCritsect);
- return _sequenceNumber++;
-}
-
-WebRtc_Word32
-RTPSender::ResetDataCounters()
-{
- _packetsSent = 0;
- _payloadBytesSent = 0;
-
- return 0;
-}
-
-// number of sent RTP packets
-// dont use critsect to avoid potental deadlock
-WebRtc_UWord32
-RTPSender::Packets() const
-{
- return _packetsSent;
-}
-
-// number of sent RTP bytes
-// dont use critsect to avoid potental deadlock
-WebRtc_UWord32
-RTPSender::Bytes() const
-{
- return _payloadBytesSent;
-}
-
-WebRtc_Word32
-RTPSender::BuildRTPheader(WebRtc_UWord8* dataBuffer,
- const WebRtc_Word8 payloadType,
- const bool markerBit,
- const WebRtc_UWord32 captureTimeStamp,
- const bool timeStampProvided,
- const bool incSequenceNumber)
-{
- assert(payloadType>=0);
-
- CriticalSectionScoped cs(_sendCritsect);
-
- dataBuffer[0] = static_cast<WebRtc_UWord8>(0x80); // version 2
- dataBuffer[1] = static_cast<WebRtc_UWord8>(payloadType);
- if (markerBit)
- {
- dataBuffer[1] |= kRtpMarkerBitMask; // MarkerBit is set
- }
-
- if(timeStampProvided)
- {
- _timeStamp = _startTimeStamp + captureTimeStamp;
- } else
- {
- // make a unique time stamp
- // used for inband signaling
- // we can't inc by the actual time, since then we increase the risk of back timing
- _timeStamp++;
- }
-
- ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+2, _sequenceNumber);
- ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+4, _timeStamp);
- ModuleRTPUtility::AssignUWord32ToBuffer(dataBuffer+8, _ssrc);
-
- WebRtc_Word32 rtpHeaderLength = 12;
-
- // Add the CSRCs if any
- if (_includeCSRCs && _CSRCs > 0)
- {
- if(_CSRCs > kRtpCsrcSize)
- {
- // error
- assert(false);
- return -1;
- }
- WebRtc_UWord8* ptr = &dataBuffer[rtpHeaderLength];
- for (WebRtc_UWord32 i = 0; i < _CSRCs; ++i)
- {
- ModuleRTPUtility::AssignUWord32ToBuffer(ptr, _CSRC[i]);
- ptr +=4;
- }
- dataBuffer[0] = (dataBuffer[0]&0xf0) | _CSRCs;
-
- // Update length of header
- rtpHeaderLength += sizeof(WebRtc_UWord32)*_CSRCs;
- }
- {
- _sequenceNumber++; // prepare for next packet
- }
-
- WebRtc_UWord16 len = BuildRTPHeaderExtension(dataBuffer + rtpHeaderLength);
- if (len)
- {
- dataBuffer[0] |= 0x10; // set eXtension bit
- rtpHeaderLength += len;
- }
-
- return rtpHeaderLength;
-}
-
-WebRtc_UWord16
-RTPSender::BuildRTPHeaderExtension(WebRtc_UWord8* dataBuffer) const
-{
- if (_rtpHeaderExtensionMap.Size() <= 0) {
- return 0;
- }
-
- /* RTP header extension, RFC 3550.
- 0 1 2 3
- 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | defined by profile | length |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | header extension |
- | .... |
- */
-
- const WebRtc_UWord32 kPosLength = 2;
- const WebRtc_UWord32 kHeaderLength = RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES;
-
- // Add extension ID (0xBEDE).
- ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer,
- RTP_ONE_BYTE_HEADER_EXTENSION);
-
- // Add extensions.
- WebRtc_UWord16 total_block_length = 0;
-
- RTPExtensionType type = _rtpHeaderExtensionMap.First();
- while (type != kRtpExtensionNone)
- {
- WebRtc_UWord8 block_length = 0;
- if (type == kRtpExtensionTransmissionTimeOffset)
- {
- block_length = BuildTransmissionTimeOffsetExtension(
- dataBuffer + kHeaderLength + total_block_length);
- }
- total_block_length += block_length;
- type = _rtpHeaderExtensionMap.Next(type);
- }
-
- if (total_block_length == 0)
- {
- // No extension added.
- return 0;
- }
-
- // Set header length (in number of Word32, header excluded).
- assert(total_block_length % 4 == 0);
- ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer + kPosLength,
- total_block_length / 4);
-
- // Total added length.
- return kHeaderLength + total_block_length;
-}
-
-WebRtc_UWord8
-RTPSender::BuildTransmissionTimeOffsetExtension(WebRtc_UWord8* dataBuffer) const
-{
- // From RFC 5450: Transmission Time Offsets in RTP Streams.
- //
- // The transmission time is signaled to the receiver in-band using the
- // general mechanism for RTP header extensions [RFC5285]. The payload
- // of this extension (the transmitted value) is a 24-bit signed integer.
- // When added to the RTP timestamp of the packet, it represents the
- // "effective" RTP transmission time of the packet, on the RTP
- // timescale.
- //
- // The form of the transmission offset extension block:
- //
- // 0 1 2 3
- // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // | ID | len=2 | transmission offset |
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
- // Get id defined by user.
- WebRtc_UWord8 id;
- if (_rtpHeaderExtensionMap.GetId(kRtpExtensionTransmissionTimeOffset, &id)
- != 0) {
- // Not registered.
- return 0;
- }
-
- int pos = 0;
- const WebRtc_UWord8 len = 2;
- dataBuffer[pos++] = (id << 4) + len;
- ModuleRTPUtility::AssignUWord24ToBuffer(dataBuffer + pos,
- _transmissionTimeOffset);
- pos += 3;
- assert(pos == TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES);
- return TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES;
-}
-
-void RTPSender::UpdateTransmissionTimeOffset(
- WebRtc_UWord8* rtp_packet,
- const WebRtc_UWord16 rtp_packet_length,
- const WebRtcRTPHeader& rtp_header,
- const WebRtc_UWord32 time_ms) const {
- CriticalSectionScoped cs(_sendCritsect);
-
- // Get length until start of transmission block.
- int transmission_block_pos =
- _rtpHeaderExtensionMap.GetLengthUntilBlockStartInBytes(
- kRtpExtensionTransmissionTimeOffset);
- if (transmission_block_pos < 0) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
- "Failed to update transmission time offset, not registered.");
- return;
- }
-
- int block_pos = 12 + rtp_header.header.numCSRCs + transmission_block_pos;
- if ((rtp_packet_length < block_pos + 4)) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
- "Failed to update transmission time offset, invalid length.");
- return;
- }
-
- // Verify that header contains extension.
- if (!((rtp_packet[12 + rtp_header.header.numCSRCs] == 0xBE) &&
- (rtp_packet[12 + rtp_header.header.numCSRCs + 1] == 0xDE))) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
- "Failed to update transmission time offset, hdr extension not found.");
- return;
- }
-
- // Get id.
- WebRtc_UWord8 id = 0;
- if (_rtpHeaderExtensionMap.GetId(kRtpExtensionTransmissionTimeOffset,
- &id) != 0) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
- "Failed to update transmission time offset, no id.");
- return;
- }
-
- // Verify first byte in block.
- const WebRtc_UWord8 first_block_byte = (id << 4) + 2;
- if (rtp_packet[block_pos] != first_block_byte) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, _id,
- "Failed to update transmission time offset.");
- return;
- }
-
- // Update transmission offset field.
- ModuleRTPUtility::AssignUWord24ToBuffer(rtp_packet + block_pos + 1,
- time_ms * 90); // RTP timestamp
-}
-
-WebRtc_Word32
-RTPSender::RegisterSendTransport(Transport* transport)
-{
- CriticalSectionScoped cs(_transportCritsect);
- _transport = transport;
- return 0;
-}
-
-void
-RTPSender::SetSendingStatus(const bool enabled)
-{
- if(enabled)
- {
- WebRtc_UWord32 freq;
- if(_audioConfigured)
- {
- WebRtc_UWord32 frequency = _audio->AudioFrequency();
-
- // sanity
- switch(frequency)
- {
- case 8000:
- case 12000:
- case 16000:
- case 24000:
- case 32000:
- break;
- default:
- assert(false);
- return;
- }
- freq = frequency;
- } else
- {
- freq = 90000; // 90 KHz for all video
- }
- WebRtc_UWord32 RTPtime = ModuleRTPUtility::GetCurrentRTP(&_clock, freq);
-
- SetStartTimestamp(RTPtime); // will be ignored if it's already configured via API
-
- } else
- {
- if(!_ssrcForced)
- {
- // generate a new SSRC
- _ssrcDB.ReturnSSRC(_ssrc);
- _ssrc = _ssrcDB.CreateSSRC(); // can't be 0
-
- }
- if(!_sequenceNumberForced && !_ssrcForced) // don't initialize seq number if SSRC passed externally
- {
- // generate a new sequence number
- _sequenceNumber = rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER);
- }
- }
-}
-
-void
-RTPSender::SetSendingMediaStatus(const bool enabled)
-{
- CriticalSectionScoped cs(_sendCritsect);
- _sendingMedia = enabled;
-}
-
-bool
-RTPSender::SendingMedia() const
-{
- CriticalSectionScoped cs(_sendCritsect);
- return _sendingMedia;
-}
-
-WebRtc_UWord32
-RTPSender::Timestamp() const
-{
- CriticalSectionScoped cs(_sendCritsect);
- return _timeStamp;
-}
-
-
-WebRtc_Word32
-RTPSender::SetStartTimestamp( const WebRtc_UWord32 timestamp, const bool force)
-{
- CriticalSectionScoped cs(_sendCritsect);
- if(force)
- {
- _startTimeStampForced = force;
- _startTimeStamp = timestamp;
- } else
- {
- if(!_startTimeStampForced)
- {
- _startTimeStamp = timestamp;
- }
- }
- return 0;
-}
-
-WebRtc_UWord32
-RTPSender::StartTimestamp() const
-{
- CriticalSectionScoped cs(_sendCritsect);
- return _startTimeStamp;
-}
-
-WebRtc_UWord32
-RTPSender::GenerateNewSSRC()
-{
- // if configured via API, return 0
- CriticalSectionScoped cs(_sendCritsect);
-
- if(_ssrcForced)
- {
- return 0;
- }
- _ssrc = _ssrcDB.CreateSSRC(); // can't be 0
- return _ssrc;
-}
-
-WebRtc_Word32
-RTPSender::SetSSRC(WebRtc_UWord32 ssrc)
-{
- // this is configured via the API
- CriticalSectionScoped cs(_sendCritsect);
-
- if (_ssrc == ssrc && _ssrcForced)
- {
- return 0; // since it's same ssrc, don't reset anything
- }
-
- _ssrcForced = true;
-
- _ssrcDB.ReturnSSRC(_ssrc);
- _ssrcDB.RegisterSSRC(ssrc);
- _ssrc = ssrc;
-
- if(!_sequenceNumberForced)
- {
- _sequenceNumber = rand() / (RAND_MAX / MAX_INIT_RTP_SEQ_NUMBER);
- }
- return 0;
-}
-
-WebRtc_UWord32
-RTPSender::SSRC() const
-{
- CriticalSectionScoped cs(_sendCritsect);
- return _ssrc;
-}
-
-WebRtc_Word32
-RTPSender::SetCSRCStatus(const bool include)
-{
- _includeCSRCs = include;
- return 0;
-}
-
-WebRtc_Word32
-RTPSender::SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
- const WebRtc_UWord8 arrLength)
-{
- if(arrLength > kRtpCsrcSize)
- {
- assert(false);
- return -1;
- }
-
- CriticalSectionScoped cs(_sendCritsect);
-
- for(int i = 0; i < arrLength;i++)
- {
- _CSRC[i] = arrOfCSRC[i];
- }
- _CSRCs = arrLength;
- return 0;
-}
-
-WebRtc_Word32
-RTPSender::CSRCs(WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const
-{
- CriticalSectionScoped cs(_sendCritsect);
-
- if(arrOfCSRC == NULL)
- {
- assert(false);
- return -1;
- }
- for(int i = 0; i < _CSRCs && i < kRtpCsrcSize;i++)
- {
- arrOfCSRC[i] = _CSRC[i];
- }
- return _CSRCs;
-}
-
-WebRtc_Word32
-RTPSender::SetSequenceNumber(WebRtc_UWord16 seq)
-{
- CriticalSectionScoped cs(_sendCritsect);
- _sequenceNumberForced = true;
- _sequenceNumber = seq;
- return 0;
-}
-
-WebRtc_UWord16
-RTPSender::SequenceNumber() const
-{
- CriticalSectionScoped cs(_sendCritsect);
- return _sequenceNumber;
-}
-
-
- /*
- * Audio
- */
-WebRtc_Word32
-RTPSender::RegisterAudioCallback(RtpAudioFeedback* messagesCallback)
-{
- if(!_audioConfigured)
- {
- return -1;
- }
- return _audio->RegisterAudioCallback(messagesCallback);
-}
-
- // Send a DTMF tone, RFC 2833 (4733)
-WebRtc_Word32
-RTPSender::SendTelephoneEvent(const WebRtc_UWord8 key,
- const WebRtc_UWord16 time_ms,
- const WebRtc_UWord8 level)
-{
- if(!_audioConfigured)
- {
- return -1;
- }
- return _audio->SendTelephoneEvent(key, time_ms, level);
-}
-
-bool
-RTPSender::SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const
-{
- if(!_audioConfigured)
- {
- return false;
- }
- return _audio->SendTelephoneEventActive(telephoneEvent);
-}
-
- // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
-WebRtc_Word32
-RTPSender::SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples)
-{
- if(!_audioConfigured)
- {
- return -1;
- }
- return _audio->SetAudioPacketSize(packetSizeSamples);
-}
-
-WebRtc_Word32
-RTPSender::SetAudioLevelIndicationStatus(const bool enable,
- const WebRtc_UWord8 ID)
-{
- if(!_audioConfigured)
- {
- return -1;
- }
- return _audio->SetAudioLevelIndicationStatus(enable, ID);
-}
-
-WebRtc_Word32
-RTPSender::AudioLevelIndicationStatus(bool& enable,
- WebRtc_UWord8& ID) const
-{
- return _audio->AudioLevelIndicationStatus(enable, ID);
-}
-
-WebRtc_Word32
-RTPSender::SetAudioLevel(const WebRtc_UWord8 level_dBov)
-{
- return _audio->SetAudioLevel(level_dBov);
-}
-
- // Set payload type for Redundant Audio Data RFC 2198
-WebRtc_Word32
-RTPSender::SetRED(const WebRtc_Word8 payloadType)
-{
- if(!_audioConfigured)
- {
- return -1;
- }
- return _audio->SetRED(payloadType);
-}
-
- // Get payload type for Redundant Audio Data RFC 2198
-WebRtc_Word32
-RTPSender::RED(WebRtc_Word8& payloadType) const
-{
- if(!_audioConfigured)
- {
- return -1;
- }
- return _audio->RED(payloadType);
-}
-
- /*
- * Video
- */
-VideoCodecInformation*
-RTPSender::CodecInformationVideo()
-{
- if(_audioConfigured)
- {
- return NULL;
- }
- return _video->CodecInformationVideo();
-}
-
-RtpVideoCodecTypes
-RTPSender::VideoCodecType() const
-{
- if(_audioConfigured)
- {
- return kRtpNoVideo;
- }
- return _video->VideoCodecType();
-}
-
-WebRtc_UWord32
-RTPSender::MaxConfiguredBitrateVideo() const
-{
- if(_audioConfigured)
- {
- return 0;
- }
- return _video->MaxConfiguredBitrateVideo();
-}
-
-WebRtc_Word32
-RTPSender::SendRTPIntraRequest()
-{
- if(_audioConfigured)
- {
- return -1;
- }
- return _video->SendRTPIntraRequest();
-}
-
-// FEC
-WebRtc_Word32
-RTPSender::SetGenericFECStatus(const bool enable,
- const WebRtc_UWord8 payloadTypeRED,
- const WebRtc_UWord8 payloadTypeFEC)
-{
- if(_audioConfigured)
- {
- return -1;
- }
- return _video->SetGenericFECStatus(enable, payloadTypeRED, payloadTypeFEC);
-}
-
-WebRtc_Word32
-RTPSender::GenericFECStatus(bool& enable,
- WebRtc_UWord8& payloadTypeRED,
- WebRtc_UWord8& payloadTypeFEC) const
-{
- if(_audioConfigured)
- {
- return -1;
- }
- return _video->GenericFECStatus(enable, payloadTypeRED, payloadTypeFEC);
-}
-
-WebRtc_Word32
-RTPSender::SetFECCodeRate(const WebRtc_UWord8 keyFrameCodeRate,
- const WebRtc_UWord8 deltaFrameCodeRate)
-{
- if(_audioConfigured)
- {
- return -1;
- }
- return _video->SetFECCodeRate(keyFrameCodeRate, deltaFrameCodeRate);
-}
-
-WebRtc_Word32
-RTPSender::SetFECUepProtection(const bool keyUseUepProtection,
- const bool deltaUseUepProtection)
-
-{
- if(_audioConfigured)
- {
- return -1;
- }
- return _video->SetFECUepProtection(keyUseUepProtection,
- deltaUseUepProtection);
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_sender.h b/trunk/src/modules/rtp_rtcp/source/rtp_sender.h
deleted file mode 100644
index 00012e1..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_sender.h
+++ /dev/null
@@ -1,376 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
-
-#include <cassert>
-#include <cmath>
-#include <map>
-
-#include "rtp_rtcp_config.h" // misc. defines (e.g. MAX_PACKET_LENGTH)
-#include "rtp_rtcp_defines.h"
-#include "common_types.h" // Encryption
-#include "ssrc_database.h"
-#include "Bitrate.h"
-#include "rtp_header_extension.h"
-#include "video_codec_information.h"
-#include "transmission_bucket.h"
-
-#define MAX_INIT_RTP_SEQ_NUMBER 32767 // 2^15 -1
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class RTPPacketHistory;
-class RTPSenderAudio;
-class RTPSenderVideo;
-
-class RTPSenderInterface
-{
-public:
- RTPSenderInterface() {}
- virtual ~RTPSenderInterface() {}
-
- virtual WebRtc_UWord32 SSRC() const = 0;
- virtual WebRtc_UWord32 Timestamp() const = 0;
-
- virtual WebRtc_Word32 BuildRTPheader(WebRtc_UWord8* dataBuffer,
- const WebRtc_Word8 payloadType,
- const bool markerBit,
- const WebRtc_UWord32 captureTimeStamp,
- const bool timeStampProvided = true,
- const bool incSequenceNumber = true) = 0;
-
- virtual WebRtc_UWord16 RTPHeaderLength() const = 0;
- virtual WebRtc_UWord16 IncrementSequenceNumber() = 0;
- virtual WebRtc_UWord16 SequenceNumber() const = 0;
- virtual WebRtc_UWord16 MaxPayloadLength() const = 0;
- virtual WebRtc_UWord16 MaxDataPayloadLength() const = 0;
- virtual WebRtc_UWord16 PacketOverHead() const = 0;
- virtual WebRtc_UWord16 TargetSendBitrateKbit() const = 0;
- virtual WebRtc_UWord16 ActualSendBitrateKbit() const = 0;
-
- virtual WebRtc_Word32 SendToNetwork(const WebRtc_UWord8* dataBuffer,
- const WebRtc_UWord16 payloadLength,
- const WebRtc_UWord16 rtpHeaderLength,
- const StorageType storage) = 0;
-};
-
-class RTPSender : public Bitrate, public RTPSenderInterface
-{
-public:
- RTPSender(const WebRtc_Word32 id, const bool audio, RtpRtcpClock* clock);
- virtual ~RTPSender();
-
- WebRtc_Word32 Init(const WebRtc_UWord32 remoteSSRC);
- void ChangeUniqueId(const WebRtc_Word32 id);
-
- void ProcessBitrate();
- void ProcessSendToNetwork();
-
- WebRtc_UWord16 TargetSendBitrateKbit() const;
- WebRtc_UWord16 ActualSendBitrateKbit() const;
-
- WebRtc_UWord32 VideoBitrateSent() const;
- WebRtc_UWord32 FecOverheadRate() const;
- WebRtc_UWord32 NackOverheadRate() const;
-
- WebRtc_Word32 SetTargetSendBitrate(const WebRtc_UWord32 bits);
-
- WebRtc_UWord16 MaxDataPayloadLength() const; // with RTP and FEC headers
-
- // callback
- WebRtc_Word32 RegisterSendTransport(Transport* outgoingTransport);
-
- WebRtc_Word32 RegisterPayload(
- const char payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate);
-
- WebRtc_Word32 DeRegisterSendPayload(const WebRtc_Word8 payloadType);
-
- WebRtc_Word8 SendPayloadType() const;
-
- int SendPayloadFrequency() const;
-
- void SetSendingStatus(const bool enabled);
-
- void SetSendingMediaStatus(const bool enabled);
- bool SendingMedia() const;
-
- // number of sent RTP packets
- WebRtc_UWord32 Packets() const;
-
- // number of sent RTP bytes
- WebRtc_UWord32 Bytes() const;
-
- WebRtc_Word32 ResetDataCounters();
-
- WebRtc_UWord32 StartTimestamp() const;
- WebRtc_Word32 SetStartTimestamp(const WebRtc_UWord32 timestamp,
- const bool force = false);
-
- WebRtc_UWord32 GenerateNewSSRC();
- WebRtc_Word32 SetSSRC( const WebRtc_UWord32 ssrc);
-
- WebRtc_UWord16 SequenceNumber() const;
- WebRtc_Word32 SetSequenceNumber( WebRtc_UWord16 seq);
-
- WebRtc_Word32 CSRCs(WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize]) const;
-
- WebRtc_Word32 SetCSRCStatus(const bool include);
-
- WebRtc_Word32 SetCSRCs(const WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize],
- const WebRtc_UWord8 arrLength);
-
- WebRtc_Word32 SetMaxPayloadLength(const WebRtc_UWord16 length,
- const WebRtc_UWord16 packetOverHead);
-
- WebRtc_Word32 SendOutgoingData(const FrameType frameType,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader* fragmentation,
- VideoCodecInformation* codecInfo = NULL,
- const RTPVideoTypeHeader* rtpTypeHdr = NULL);
-
- WebRtc_Word32 SendPadData(WebRtc_Word8 payload_type,
- WebRtc_UWord32 capture_timestamp,
- WebRtc_Word32 bytes);
- /*
- * RTP header extension
- */
- WebRtc_Word32 SetTransmissionTimeOffset(
- const WebRtc_Word32 transmissionTimeOffset);
-
- WebRtc_Word32 RegisterRtpHeaderExtension(const RTPExtensionType type,
- const WebRtc_UWord8 id);
-
- WebRtc_Word32 DeregisterRtpHeaderExtension(const RTPExtensionType type);
-
- WebRtc_UWord16 RtpHeaderExtensionTotalLength() const;
-
- WebRtc_UWord16 BuildRTPHeaderExtension(WebRtc_UWord8* dataBuffer) const;
-
- WebRtc_UWord8 BuildTransmissionTimeOffsetExtension(
- WebRtc_UWord8* dataBuffer) const;
-
- void UpdateTransmissionTimeOffset(WebRtc_UWord8* rtp_packet,
- const WebRtc_UWord16 rtp_packet_length,
- const WebRtcRTPHeader& rtp_header,
- const WebRtc_UWord32 time_ms) const;
-
- void SetTransmissionSmoothingStatus(const bool enable);
-
- bool TransmissionSmoothingStatus() const;
-
- /*
- * NACK
- */
- int SelectiveRetransmissions() const;
- int SetSelectiveRetransmissions(uint8_t settings);
- void OnReceivedNACK(const WebRtc_UWord16 nackSequenceNumbersLength,
- const WebRtc_UWord16* nackSequenceNumbers,
- const WebRtc_UWord16 avgRTT);
-
- WebRtc_Word32 SetStorePacketsStatus(const bool enable,
- const WebRtc_UWord16 numberToStore);
-
- bool StorePackets() const;
-
- WebRtc_Word32 ReSendPacket(WebRtc_UWord16 packet_id,
- WebRtc_UWord32 min_resend_time = 0);
-
- WebRtc_Word32 ReSendToNetwork(const WebRtc_UWord8* packet,
- const WebRtc_UWord32 size);
-
- bool ProcessNACKBitRate(const WebRtc_UWord32 now);
-
- /*
- * Keep alive
- */
- WebRtc_Word32 EnableRTPKeepalive( const WebRtc_Word8 unknownPayloadType,
- const WebRtc_UWord16 deltaTransmitTimeMS);
-
- WebRtc_Word32 RTPKeepaliveStatus(bool* enable,
- WebRtc_Word8* unknownPayloadType,
- WebRtc_UWord16* deltaTransmitTimeMS) const;
-
- WebRtc_Word32 DisableRTPKeepalive();
-
- bool RTPKeepalive() const;
-
- bool TimeToSendRTPKeepalive() const;
-
- WebRtc_Word32 SendRTPKeepalivePacket();
-
- /*
- * RTX
- */
- void SetRTXStatus(const bool enable,
- const bool setSSRC,
- const WebRtc_UWord32 SSRC);
-
- void RTXStatus(bool* enable, WebRtc_UWord32* SSRC) const;
-
- /*
- * Functions wrapping RTPSenderInterface
- */
- virtual WebRtc_Word32 BuildRTPheader(WebRtc_UWord8* dataBuffer,
- const WebRtc_Word8 payloadType,
- const bool markerBit,
- const WebRtc_UWord32 captureTimeStamp,
- const bool timeStampProvided = true,
- const bool incSequenceNumber = true);
-
- virtual WebRtc_UWord16 RTPHeaderLength() const ;
- virtual WebRtc_UWord16 IncrementSequenceNumber();
- virtual WebRtc_UWord16 MaxPayloadLength() const;
- virtual WebRtc_UWord16 PacketOverHead() const;
-
- // current timestamp
- virtual WebRtc_UWord32 Timestamp() const;
- virtual WebRtc_UWord32 SSRC() const;
-
- virtual WebRtc_Word32 SendToNetwork(const WebRtc_UWord8* dataBuffer,
- const WebRtc_UWord16 payloadLength,
- const WebRtc_UWord16 rtpHeaderLength,
- const StorageType storage);
-
- /*
- * Audio
- */
- WebRtc_Word32 RegisterAudioCallback(RtpAudioFeedback* messagesCallback);
-
- // Send a DTMF tone using RFC 2833 (4733)
- WebRtc_Word32 SendTelephoneEvent(const WebRtc_UWord8 key,
- const WebRtc_UWord16 time_ms,
- const WebRtc_UWord8 level);
-
- bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const;
-
- // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
- WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples);
-
- // Set status and ID for header-extension-for-audio-level-indication.
- WebRtc_Word32 SetAudioLevelIndicationStatus(const bool enable,
- const WebRtc_UWord8 ID);
-
- // Get status and ID for header-extension-for-audio-level-indication.
- WebRtc_Word32 AudioLevelIndicationStatus(bool& enable,
- WebRtc_UWord8& ID) const;
-
- // Store the audio level in dBov for header-extension-for-audio-level-indication.
- WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov);
-
- // Set payload type for Redundant Audio Data RFC 2198
- WebRtc_Word32 SetRED(const WebRtc_Word8 payloadType);
-
- // Get payload type for Redundant Audio Data RFC 2198
- WebRtc_Word32 RED(WebRtc_Word8& payloadType) const;
-
- /*
- * Video
- */
- VideoCodecInformation* CodecInformationVideo();
-
- RtpVideoCodecTypes VideoCodecType() const;
-
- WebRtc_UWord32 MaxConfiguredBitrateVideo() const;
-
- WebRtc_Word32 SendRTPIntraRequest();
-
- // FEC
- WebRtc_Word32 SetGenericFECStatus(const bool enable,
- const WebRtc_UWord8 payloadTypeRED,
- const WebRtc_UWord8 payloadTypeFEC);
-
- WebRtc_Word32 GenericFECStatus(bool& enable,
- WebRtc_UWord8& payloadTypeRED,
- WebRtc_UWord8& payloadTypeFEC) const;
-
- WebRtc_Word32 SetFECCodeRate(const WebRtc_UWord8 keyFrameCodeRate,
- const WebRtc_UWord8 deltaFrameCodeRate);
-
- WebRtc_Word32 SetFECUepProtection(const bool keyUseUepProtection,
- const bool deltaUseUepProtection);
-
-protected:
- WebRtc_Word32 CheckPayloadType(const WebRtc_Word8 payloadType,
- RtpVideoCodecTypes& videoType);
-
-private:
- void UpdateNACKBitRate(const WebRtc_UWord32 bytes,
- const WebRtc_UWord32 now);
-
- WebRtc_Word32 _id;
- const bool _audioConfigured;
- RTPSenderAudio* _audio;
- RTPSenderVideo* _video;
-
- CriticalSectionWrapper* _sendCritsect;
-
- CriticalSectionWrapper* _transportCritsect;
- Transport* _transport;
-
- bool _sendingMedia;
-
- WebRtc_UWord16 _maxPayloadLength;
- WebRtc_UWord16 _targetSendBitrate;
- WebRtc_UWord16 _packetOverHead;
-
- WebRtc_Word8 _payloadType;
- std::map<WebRtc_Word8, ModuleRTPUtility::Payload*> _payloadTypeMap;
-
- RtpHeaderExtensionMap _rtpHeaderExtensionMap;
- WebRtc_Word32 _transmissionTimeOffset;
-
- bool _keepAliveIsActive;
- WebRtc_Word8 _keepAlivePayloadType;
- WebRtc_UWord32 _keepAliveLastSent;
- WebRtc_UWord16 _keepAliveDeltaTimeSend;
-
- // NACK
- WebRtc_UWord32 _nackByteCountTimes[NACK_BYTECOUNT_SIZE];
- WebRtc_Word32 _nackByteCount[NACK_BYTECOUNT_SIZE];
- Bitrate _nackBitrate;
-
- RTPPacketHistory* _packetHistory;
- TransmissionBucket _sendBucket;
- WebRtc_UWord32 _timeLastSendToNetworkUpdate;
- bool _transmissionSmoothing;
-
- // statistics
- WebRtc_UWord32 _packetsSent;
- WebRtc_UWord32 _payloadBytesSent;
-
- // RTP variables
- bool _startTimeStampForced;
- WebRtc_UWord32 _startTimeStamp;
- SSRCDatabase& _ssrcDB;
- WebRtc_UWord32 _remoteSSRC;
- bool _sequenceNumberForced;
- WebRtc_UWord16 _sequenceNumber;
- WebRtc_UWord16 _sequenceNumberRTX;
- bool _ssrcForced;
- WebRtc_UWord32 _ssrc;
- WebRtc_UWord32 _timeStamp;
- WebRtc_UWord8 _CSRCs;
- WebRtc_UWord32 _CSRC[kRtpCsrcSize];
- bool _includeCSRCs;
- bool _RTX;
- WebRtc_UWord32 _ssrcRTX;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_sender_audio.cc b/trunk/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
deleted file mode 100644
index ee3c414..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_sender_audio.cc
+++ /dev/null
@@ -1,627 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rtp_sender_audio.h"
-
-#include <string.h> //memcpy
-#include <cassert> //assert
-
-namespace webrtc {
-RTPSenderAudio::RTPSenderAudio(const WebRtc_Word32 id, RtpRtcpClock* clock,
- RTPSenderInterface* rtpSender) :
- _id(id),
- _clock(*clock),
- _rtpSender(rtpSender),
- _audioFeedbackCritsect(CriticalSectionWrapper::CreateCriticalSection()),
- _audioFeedback(NULL),
- _sendAudioCritsect(CriticalSectionWrapper::CreateCriticalSection()),
- _frequency(8000),
- _packetSizeSamples(160),
- _dtmfEventIsOn(false),
- _dtmfEventFirstPacketSent(false),
- _dtmfPayloadType(-1),
- _dtmfTimestamp(0),
- _dtmfKey(0),
- _dtmfLengthSamples(0),
- _dtmfLevel(0),
- _dtmfTimeLastSent(0),
- _dtmfTimestampLastSent(0),
- _REDPayloadType(-1),
- _inbandVADactive(false),
- _cngNBPayloadType(-1),
- _cngWBPayloadType(-1),
- _cngSWBPayloadType(-1),
- _lastPayloadType(-1),
- _includeAudioLevelIndication(false), // @TODO - reset at Init()?
- _audioLevelIndicationID(0),
- _audioLevel_dBov(0) {
-};
-
-RTPSenderAudio::~RTPSenderAudio()
-{
- delete _sendAudioCritsect;
- delete _audioFeedbackCritsect;
-}
-
-WebRtc_Word32
-RTPSenderAudio::Init()
-{
- CriticalSectionScoped cs(_sendAudioCritsect);
-
- _dtmfPayloadType = -1;
- _inbandVADactive = false;
- _cngNBPayloadType = -1;
- _cngWBPayloadType = -1;
- _cngSWBPayloadType = -1;
- _lastPayloadType = -1;
- _REDPayloadType = -1;
- _dtmfTimeLastSent = 0;
- _dtmfTimestampLastSent = 0;
- ResetDTMF();
- return 0;
-}
-
-void
-RTPSenderAudio::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
-}
-
-WebRtc_Word32
-RTPSenderAudio::RegisterAudioCallback(RtpAudioFeedback* messagesCallback)
-{
- CriticalSectionScoped cs(_audioFeedbackCritsect);
- _audioFeedback = messagesCallback;
- return 0;
-}
-
-void
-RTPSenderAudio::SetAudioFrequency(const WebRtc_UWord32 f)
-{
- CriticalSectionScoped cs(_sendAudioCritsect);
- _frequency = f;
-}
-
-int
-RTPSenderAudio::AudioFrequency() const
-{
- CriticalSectionScoped cs(_sendAudioCritsect);
- return _frequency;
-}
-
- // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
-WebRtc_Word32
-RTPSenderAudio::SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples)
-{
- CriticalSectionScoped cs(_sendAudioCritsect);
-
- _packetSizeSamples = packetSizeSamples;
- return 0;
-}
-
-WebRtc_Word32 RTPSenderAudio::RegisterAudioPayload(
- const char payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate,
- ModuleRTPUtility::Payload*& payload) {
- CriticalSectionScoped cs(_sendAudioCritsect);
-
- if (ModuleRTPUtility::StringCompare(payloadName, "cn", 2)) {
- // we can have multiple CNG payload types
- if (frequency == 8000) {
- _cngNBPayloadType = payloadType;
-
- } else if (frequency == 16000) {
- _cngWBPayloadType = payloadType;
-
- } else if (frequency == 32000) {
- _cngSWBPayloadType = payloadType;
- } else {
- return -1;
- }
- }
- if (ModuleRTPUtility::StringCompare(payloadName, "telephone-event", 15)) {
- // Don't add it to the list
- // we dont want to allow send with a DTMF payloadtype
- _dtmfPayloadType = payloadType;
- return 0;
- // The default timestamp rate is 8000 Hz, but other rates may be defined.
- }
- payload = new ModuleRTPUtility::Payload;
- payload->typeSpecific.Audio.frequency = frequency;
- payload->typeSpecific.Audio.channels = channels;
- payload->typeSpecific.Audio.rate = rate;
- payload->audio = true;
- payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
- strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
- return 0;
-}
-
-bool
-RTPSenderAudio::MarkerBit(const FrameType frameType,
- const WebRtc_Word8 payloadType)
-{
- CriticalSectionScoped cs(_sendAudioCritsect);
-
- // for audio true for first packet in a speech burst
- bool markerBit = false;
- if(_lastPayloadType != payloadType)
- {
- if(_cngNBPayloadType != -1)
- {
- // we have configured NB CNG
- if(_cngNBPayloadType == payloadType)
- {
- // only set a marker bit when we change payload type to a non CNG
- return false;
- }
- }
- if(_cngWBPayloadType != -1)
- {
- // we have configured WB CNG
- if(_cngWBPayloadType == payloadType)
- {
- // only set a marker bit when we change payload type to a non CNG
- return false;
- }
- }
- if(_cngSWBPayloadType != -1)
- {
- // we have configured SWB CNG
- if(_cngSWBPayloadType == payloadType)
- {
- // only set a marker bit when we change payload type to a non CNG
- return false;
- }
- }
- // payloadType differ
- if(_lastPayloadType == -1)
- {
- if(frameType != kAudioFrameCN)
- {
- // first packet and NOT CNG
- return true;
-
- }else
- {
- // first packet and CNG
- _inbandVADactive = true;
- return false;
- }
- }
- // not first packet AND
- // not CNG AND
- // payloadType changed
-
- // set a marker bit when we change payload type
- markerBit = true;
- }
-
- // For G.723 G.729, AMR etc we can have inband VAD
- if(frameType == kAudioFrameCN)
- {
- _inbandVADactive = true;
-
- } else if(_inbandVADactive)
- {
- _inbandVADactive = false;
- markerBit = true;
- }
- return markerBit;
-}
-
-bool
-RTPSenderAudio::SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const
-{
- if(_dtmfEventIsOn)
- {
- telephoneEvent = _dtmfKey;
- return true;
- }
- WebRtc_UWord32 delaySinceLastDTMF = (_clock.GetTimeInMS() - _dtmfTimeLastSent);
- if(delaySinceLastDTMF < 100)
- {
- telephoneEvent = _dtmfKey;
- return true;
- }
- telephoneEvent = -1;
- return false;
-}
-
-WebRtc_Word32 RTPSenderAudio::SendAudio(
- const FrameType frameType,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 dataSize,
- const RTPFragmentationHeader* fragmentation) {
- // TODO(pwestin) Breakup function in smaller functions.
- WebRtc_UWord16 payloadSize = static_cast<WebRtc_UWord16>(dataSize);
- WebRtc_UWord16 maxPayloadLength = _rtpSender->MaxPayloadLength();
- bool dtmfToneStarted = false;
- WebRtc_UWord16 dtmfLengthMS = 0;
- WebRtc_UWord8 key = 0;
-
- // Check if we have pending DTMFs to send
- if (!_dtmfEventIsOn && PendingDTMF()) {
- CriticalSectionScoped cs(_sendAudioCritsect);
-
- WebRtc_UWord32 delaySinceLastDTMF = _clock.GetTimeInMS() -
- _dtmfTimeLastSent;
-
- if (delaySinceLastDTMF > 100) {
- // New tone to play
- _dtmfTimestamp = captureTimeStamp;
- if (NextDTMF(&key, &dtmfLengthMS, &_dtmfLevel) >= 0) {
- _dtmfEventFirstPacketSent = false;
- _dtmfKey = key;
- _dtmfLengthSamples = (_frequency / 1000) * dtmfLengthMS;
- dtmfToneStarted = true;
- _dtmfEventIsOn = true;
- }
- }
- }
- if (dtmfToneStarted) {
- CriticalSectionScoped cs(_audioFeedbackCritsect);
- if (_audioFeedback) {
- _audioFeedback->OnPlayTelephoneEvent(_id, key, dtmfLengthMS, _dtmfLevel);
- }
- }
-
- // A source MAY send events and coded audio packets for the same time
- // but we don't support it
- {
- _sendAudioCritsect->Enter();
-
- if (_dtmfEventIsOn) {
- if (frameType == kFrameEmpty) {
- // kFrameEmpty is used to drive the DTMF when in CN mode
- // it can be triggered more frequently than we want to send the
- // DTMF packets.
- if (_packetSizeSamples > (captureTimeStamp - _dtmfTimestampLastSent)) {
- // not time to send yet
- _sendAudioCritsect->Leave();
- return 0;
- }
- }
- _dtmfTimestampLastSent = captureTimeStamp;
- WebRtc_UWord32 dtmfDurationSamples = captureTimeStamp - _dtmfTimestamp;
- bool ended = false;
- bool send = true;
-
- if (_dtmfLengthSamples > dtmfDurationSamples) {
- if (dtmfDurationSamples <= 0) {
- // Skip send packet at start, since we shouldn't use duration 0
- send = false;
- }
- } else {
- ended = true;
- _dtmfEventIsOn = false;
- _dtmfTimeLastSent = _clock.GetTimeInMS();
- }
- // don't hold the critsect while calling SendTelephoneEventPacket
- _sendAudioCritsect->Leave();
- if (send) {
- if (dtmfDurationSamples > 0xffff) {
- // RFC 4733 2.5.2.3 Long-Duration Events
- SendTelephoneEventPacket(ended, _dtmfTimestamp,
- static_cast<WebRtc_UWord16>(0xffff), false);
-
- // set new timestap for this segment
- _dtmfTimestamp = captureTimeStamp;
- dtmfDurationSamples -= 0xffff;
- _dtmfLengthSamples -= 0xffff;
-
- return SendTelephoneEventPacket(
- ended,
- _dtmfTimestamp,
- static_cast<WebRtc_UWord16>(dtmfDurationSamples),
- false);
- } else {
- // set markerBit on the first packet in the burst
- _dtmfEventFirstPacketSent = true;
- return SendTelephoneEventPacket(
- ended,
- _dtmfTimestamp,
- static_cast<WebRtc_UWord16>(dtmfDurationSamples),
- !_dtmfEventFirstPacketSent);
- }
- }
- return 0;
- }
- _sendAudioCritsect->Leave();
- }
- if (payloadSize == 0 || payloadData == NULL) {
- if (frameType == kFrameEmpty) {
- // we don't send empty audio RTP packets
- // no error since we use it to drive DTMF when we use VAD
- return 0;
- }
- return -1;
- }
- WebRtc_UWord8 dataBuffer[IP_PACKET_SIZE];
- bool markerBit = MarkerBit(frameType, payloadType);
-
- WebRtc_Word32 rtpHeaderLength = 0;
- WebRtc_UWord16 timestampOffset = 0;
-
- if (_REDPayloadType >= 0 && fragmentation && !markerBit &&
- fragmentation->fragmentationVectorSize > 1) {
- // have we configured RED? use its payload type
- // we need to get the current timestamp to calc the diff
- WebRtc_UWord32 oldTimeStamp = _rtpSender->Timestamp();
- rtpHeaderLength = _rtpSender->BuildRTPheader(dataBuffer, _REDPayloadType,
- markerBit, captureTimeStamp);
-
- timestampOffset = WebRtc_UWord16(_rtpSender->Timestamp() - oldTimeStamp);
- } else {
- rtpHeaderLength = _rtpSender->BuildRTPheader(dataBuffer, payloadType,
- markerBit, captureTimeStamp);
- }
- if (rtpHeaderLength <= 0) {
- return -1;
- }
- {
- CriticalSectionScoped cs(_sendAudioCritsect);
-
- // https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/
- if (_includeAudioLevelIndication) {
- dataBuffer[0] |= 0x10; // set eXtension bit
- /*
- 0 1 2 3
- 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | 0xBE | 0xDE | length=1 |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | ID | len=0 |V| level | 0x00 | 0x00 |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- */
- // add our ID (0xBEDE)
- ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+rtpHeaderLength,
- RTP_AUDIO_LEVEL_UNIQUE_ID);
- rtpHeaderLength += 2;
-
- // add the length (length=1) in number of word32
- const WebRtc_UWord8 length = 1;
- ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+rtpHeaderLength,
- length);
- rtpHeaderLength += 2;
-
- // add ID (defined by the user) and len(=0) byte
- const WebRtc_UWord8 id = _audioLevelIndicationID;
- const WebRtc_UWord8 len = 0;
- dataBuffer[rtpHeaderLength++] = (id << 4) + len;
-
- // add voice-activity flag (V) bit and the audio level (in dBov)
- const WebRtc_UWord8 V = (frameType == kAudioFrameSpeech);
- WebRtc_UWord8 level = _audioLevel_dBov;
- dataBuffer[rtpHeaderLength++] = (V << 7) + level;
-
- // add two bytes zero padding
- ModuleRTPUtility::AssignUWord16ToBuffer(dataBuffer+rtpHeaderLength, 0);
- rtpHeaderLength += 2;
- }
-
- if(maxPayloadLength < rtpHeaderLength + payloadSize ) {
- // too large payload buffer
- return -1;
- }
-
- if (_REDPayloadType >= 0 && // Have we configured RED?
- fragmentation &&
- fragmentation->fragmentationVectorSize > 1 &&
- !markerBit) {
- if (timestampOffset <= 0x3fff) {
- if(fragmentation->fragmentationVectorSize != 2) {
- // we only support 2 codecs when using RED
- return -1;
- }
- // only 0x80 if we have multiple blocks
- dataBuffer[rtpHeaderLength++] = 0x80 +
- fragmentation->fragmentationPlType[1];
- WebRtc_UWord32 blockLength = fragmentation->fragmentationLength[1];
-
- // sanity blockLength
- if(blockLength > 0x3ff) { // block length 10 bits 1023 bytes
- return -1;
- }
- WebRtc_UWord32 REDheader = (timestampOffset << 10) + blockLength;
- ModuleRTPUtility::AssignUWord24ToBuffer(dataBuffer + rtpHeaderLength,
- REDheader);
- rtpHeaderLength += 3;
-
- dataBuffer[rtpHeaderLength++] = fragmentation->fragmentationPlType[0];
- // copy the RED data
- memcpy(dataBuffer+rtpHeaderLength,
- payloadData + fragmentation->fragmentationOffset[1],
- fragmentation->fragmentationLength[1]);
-
- // copy the normal data
- memcpy(dataBuffer+rtpHeaderLength +
- fragmentation->fragmentationLength[1],
- payloadData + fragmentation->fragmentationOffset[0],
- fragmentation->fragmentationLength[0]);
-
- payloadSize = static_cast<WebRtc_UWord16>(
- fragmentation->fragmentationLength[0] +
- fragmentation->fragmentationLength[1]);
- } else {
- // silence for too long send only new data
- dataBuffer[rtpHeaderLength++] = static_cast<WebRtc_UWord8>(payloadType);
- memcpy(dataBuffer+rtpHeaderLength,
- payloadData + fragmentation->fragmentationOffset[0],
- fragmentation->fragmentationLength[0]);
-
- payloadSize = static_cast<WebRtc_UWord16>(
- fragmentation->fragmentationLength[0]);
- }
- } else {
- if (fragmentation && fragmentation->fragmentationVectorSize > 0) {
- // use the fragment info if we have one
- memcpy( dataBuffer+rtpHeaderLength,
- payloadData + fragmentation->fragmentationOffset[0],
- fragmentation->fragmentationLength[0]);
-
- payloadSize = static_cast<WebRtc_UWord16>(
- fragmentation->fragmentationLength[0]);
- } else {
- memcpy(dataBuffer+rtpHeaderLength, payloadData, payloadSize);
- }
- }
- _lastPayloadType = payloadType;
- } // end critical section
- return _rtpSender->SendToNetwork(dataBuffer,
- payloadSize,
- static_cast<WebRtc_UWord16>(rtpHeaderLength),
- kAllowRetransmission);
-}
-
-WebRtc_Word32
-RTPSenderAudio::SetAudioLevelIndicationStatus(const bool enable,
- const WebRtc_UWord8 ID)
-{
- if(ID < 1 || ID > 14)
- {
- return -1;
- }
- CriticalSectionScoped cs(_sendAudioCritsect);
-
- _includeAudioLevelIndication = enable;
- _audioLevelIndicationID = ID;
-
- return 0;
-}
-
-WebRtc_Word32
-RTPSenderAudio::AudioLevelIndicationStatus(bool& enable,
- WebRtc_UWord8& ID) const
-{
- CriticalSectionScoped cs(_sendAudioCritsect);
- enable = _includeAudioLevelIndication;
- ID = _audioLevelIndicationID;
- return 0;
-}
-
- // Audio level magnitude and voice activity flag are set for each RTP packet
-WebRtc_Word32
-RTPSenderAudio::SetAudioLevel(const WebRtc_UWord8 level_dBov)
-{
- if (level_dBov > 127)
- {
- return -1;
- }
- CriticalSectionScoped cs(_sendAudioCritsect);
- _audioLevel_dBov = level_dBov;
- return 0;
-}
-
- // Set payload type for Redundant Audio Data RFC 2198
-WebRtc_Word32
-RTPSenderAudio::SetRED(const WebRtc_Word8 payloadType)
-{
- if(payloadType < -1 )
- {
- return -1;
- }
- _REDPayloadType = payloadType;
- return 0;
-}
-
- // Get payload type for Redundant Audio Data RFC 2198
-WebRtc_Word32
-RTPSenderAudio::RED(WebRtc_Word8& payloadType) const
-{
- if(_REDPayloadType == -1)
- {
- // not configured
- return -1;
- }
- payloadType = _REDPayloadType;
- return 0;
-}
-
-// Send a TelephoneEvent tone using RFC 2833 (4733)
-WebRtc_Word32
-RTPSenderAudio::SendTelephoneEvent(const WebRtc_UWord8 key,
- const WebRtc_UWord16 time_ms,
- const WebRtc_UWord8 level)
-{
- // DTMF is protected by its own critsect
- if(_dtmfPayloadType < 0)
- {
- // TelephoneEvent payloadtype not configured
- return -1;
- }
- return AddDTMF(key, time_ms, level);
-}
-
-WebRtc_Word32
-RTPSenderAudio::SendTelephoneEventPacket(const bool ended,
- const WebRtc_UWord32 dtmfTimeStamp,
- const WebRtc_UWord16 duration,
- const bool markerBit)
-{
- WebRtc_UWord8 dtmfbuffer[IP_PACKET_SIZE];
- WebRtc_UWord8 sendCount = 1;
- WebRtc_Word32 retVal = 0;
-
- if(ended)
- {
- // resend last packet in an event 3 times
- sendCount = 3;
- }
- do
- {
- _sendAudioCritsect->Enter();
-
- //Send DTMF data
- _rtpSender->BuildRTPheader(dtmfbuffer, _dtmfPayloadType, markerBit, dtmfTimeStamp);
-
- // reset CSRC and X bit
- dtmfbuffer[0] &= 0xe0;
-
- //Create DTMF data
- /* From RFC 2833:
-
- 0 1 2 3
- 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | event |E|R| volume | duration |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- */
- // R bit always cleared
- WebRtc_UWord8 R = 0x00;
- WebRtc_UWord8 volume = _dtmfLevel;
-
- // First packet un-ended
- WebRtc_UWord8 E = 0x00;
-
- if(ended)
- {
- E = 0x80;
- }
-
- // First byte is Event number, equals key number
- dtmfbuffer[12] = _dtmfKey;
- dtmfbuffer[13] = E|R|volume;
- ModuleRTPUtility::AssignUWord16ToBuffer(dtmfbuffer+14, duration);
-
- _sendAudioCritsect->Leave();
- retVal = _rtpSender->SendToNetwork(dtmfbuffer, 4, 12,
- kAllowRetransmission);
- sendCount--;
-
- }while (sendCount > 0 && retVal == 0);
-
- return retVal;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_sender_audio.h b/trunk/src/modules/rtp_rtcp/source/rtp_sender_audio.h
deleted file mode 100644
index 5fda2ef..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_sender_audio.h
+++ /dev/null
@@ -1,133 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
-
-#include "rtp_rtcp_config.h" // misc. defines (e.g. MAX_PACKET_LENGTH)
-#include "common_types.h" // Transport
-#include "typedefs.h"
-
-#include "dtmf_queue.h"
-#include "rtp_utility.h"
-
-#include "rtp_sender.h"
-
-namespace webrtc {
-class RTPSenderAudio: public DTMFqueue
-{
-public:
- RTPSenderAudio(const WebRtc_Word32 id, RtpRtcpClock* clock,
- RTPSenderInterface* rtpSender);
- virtual ~RTPSenderAudio();
-
- void ChangeUniqueId(const WebRtc_Word32 id);
-
- WebRtc_Word32 Init();
-
- WebRtc_Word32 RegisterAudioPayload(
- const char payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate,
- ModuleRTPUtility::Payload*& payload);
-
- WebRtc_Word32 SendAudio(const FrameType frameType,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader* fragmentation);
-
- // set audio packet size, used to determine when it's time to send a DTMF packet in silence (CNG)
- WebRtc_Word32 SetAudioPacketSize(const WebRtc_UWord16 packetSizeSamples);
-
- // Set status and ID for header-extension-for-audio-level-indication.
- // Valid ID range is [1,14].
- WebRtc_Word32 SetAudioLevelIndicationStatus(const bool enable,
- const WebRtc_UWord8 ID);
-
- // Get status and ID for header-extension-for-audio-level-indication.
- WebRtc_Word32 AudioLevelIndicationStatus(bool& enable,
- WebRtc_UWord8& ID) const;
-
- // Store the audio level in dBov for header-extension-for-audio-level-indication.
- // Valid range is [0,100]. Actual value is negative.
- WebRtc_Word32 SetAudioLevel(const WebRtc_UWord8 level_dBov);
-
- // Send a DTMF tone using RFC 2833 (4733)
- WebRtc_Word32 SendTelephoneEvent(const WebRtc_UWord8 key,
- const WebRtc_UWord16 time_ms,
- const WebRtc_UWord8 level);
-
- bool SendTelephoneEventActive(WebRtc_Word8& telephoneEvent) const;
-
- void SetAudioFrequency(const WebRtc_UWord32 f);
-
- int AudioFrequency() const;
-
- // Set payload type for Redundant Audio Data RFC 2198
- WebRtc_Word32 SetRED(const WebRtc_Word8 payloadType);
-
- // Get payload type for Redundant Audio Data RFC 2198
- WebRtc_Word32 RED(WebRtc_Word8& payloadType) const;
-
- WebRtc_Word32 RegisterAudioCallback(RtpAudioFeedback* messagesCallback);
-
-protected:
- WebRtc_Word32 SendTelephoneEventPacket(const bool ended,
- const WebRtc_UWord32 dtmfTimeStamp,
- const WebRtc_UWord16 duration,
- const bool markerBit); // set on first packet in talk burst
-
- bool MarkerBit(const FrameType frameType,
- const WebRtc_Word8 payloadType);
-
-private:
- WebRtc_Word32 _id;
- RtpRtcpClock& _clock;
- RTPSenderInterface* _rtpSender;
- CriticalSectionWrapper* _audioFeedbackCritsect;
- RtpAudioFeedback* _audioFeedback;
-
- CriticalSectionWrapper* _sendAudioCritsect;
-
- WebRtc_UWord32 _frequency;
- WebRtc_UWord16 _packetSizeSamples;
-
- // DTMF
- bool _dtmfEventIsOn;
- bool _dtmfEventFirstPacketSent;
- WebRtc_Word8 _dtmfPayloadType;
- WebRtc_UWord32 _dtmfTimestamp;
- WebRtc_UWord8 _dtmfKey;
- WebRtc_UWord32 _dtmfLengthSamples;
- WebRtc_UWord8 _dtmfLevel;
- WebRtc_UWord32 _dtmfTimeLastSent;
- WebRtc_UWord32 _dtmfTimestampLastSent;
-
- WebRtc_Word8 _REDPayloadType;
-
- // VAD detection, used for markerbit
- bool _inbandVADactive;
- WebRtc_Word8 _cngNBPayloadType;
- WebRtc_Word8 _cngWBPayloadType;
- WebRtc_Word8 _cngSWBPayloadType;
- WebRtc_Word8 _lastPayloadType;
-
- // Audio level indication (https://datatracker.ietf.org/doc/draft-lennox-avt-rtp-audio-level-exthdr/)
- bool _includeAudioLevelIndication;
- WebRtc_UWord8 _audioLevelIndicationID;
- WebRtc_UWord8 _audioLevel_dBov;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_AUDIO_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_sender_test.cc b/trunk/src/modules/rtp_rtcp/source/rtp_sender_test.cc
deleted file mode 100644
index 4e8c78c..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_sender_test.cc
+++ /dev/null
@@ -1,227 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file includes unit tests for the RTPSender.
- */
-
-#include <gtest/gtest.h>
-
-#include "rtp_header_extension.h"
-#include "rtp_rtcp_defines.h"
-#include "rtp_sender.h"
-#include "rtp_utility.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-namespace {
-const int kId = 1;
-const int kTypeLength = TRANSMISSION_TIME_OFFSET_LENGTH_IN_BYTES;
-const int kPayload = 100;
-const uint32_t kTimestamp = 10;
-const uint16_t kSeqNum = 33;
-const int kTimeOffset = 22222;
-const int kMaxPacketLength = 1500;
-} // namespace
-
-class FakeClockTest : public RtpRtcpClock {
- public:
- FakeClockTest() {
- time_in_ms_ = 123456;
- }
- // Return a timestamp in milliseconds relative to some arbitrary
- // source; the source is fixed for this clock.
- virtual WebRtc_UWord32 GetTimeInMS() {
- return time_in_ms_;
- }
- // Retrieve an NTP absolute timestamp.
- virtual void CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac) {
- secs = time_in_ms_ / 1000;
- frac = (time_in_ms_ % 1000) * 4294967;
- }
- void IncrementTime(WebRtc_UWord32 time_increment_ms) {
- time_in_ms_ += time_increment_ms;
- }
- private:
- WebRtc_UWord32 time_in_ms_;
-};
-
-class LoopbackTransportTest : public webrtc::Transport {
- public:
- LoopbackTransportTest()
- : packets_sent_(0),
- last_sent_packet_len_(0) {
- }
- virtual int SendPacket(int channel, const void *data, int len) {
- packets_sent_++;
- memcpy(last_sent_packet_, data, len);
- last_sent_packet_len_ = len;
- return len;
- }
- virtual int SendRTCPPacket(int channel, const void *data, int len) {
- return -1;
- }
- int packets_sent_;
- int last_sent_packet_len_;
- uint8_t last_sent_packet_[kMaxPacketLength];
-};
-
-class RtpSenderTest : public ::testing::Test {
- protected:
- RtpSenderTest()
- : fake_clock_(),
- rtp_sender_(new RTPSender(0, false, &fake_clock_)),
- transport_(),
- kMarkerBit(true),
- kType(kRtpExtensionTransmissionTimeOffset),
- packet_() {
- EXPECT_EQ(0, rtp_sender_->SetSequenceNumber(kSeqNum));
- }
- ~RtpSenderTest() {
- delete rtp_sender_;
- }
-
- FakeClockTest fake_clock_;
- RTPSender* rtp_sender_;
- LoopbackTransportTest transport_;
- const bool kMarkerBit;
- RTPExtensionType kType;
- uint8_t packet_[kMaxPacketLength];
-
- void VerifyRTPHeaderCommon(const WebRtcRTPHeader& rtp_header) {
- EXPECT_EQ(kMarkerBit, rtp_header.header.markerBit);
- EXPECT_EQ(kPayload, rtp_header.header.payloadType);
- EXPECT_EQ(kSeqNum, rtp_header.header.sequenceNumber);
- EXPECT_EQ(kTimestamp, rtp_header.header.timestamp);
- EXPECT_EQ(rtp_sender_->SSRC(), rtp_header.header.ssrc);
- EXPECT_EQ(0, rtp_header.header.numCSRCs);
- EXPECT_EQ(0, rtp_header.header.paddingLength);
- }
-};
-
-TEST_F(RtpSenderTest, RegisterRtpHeaderExtension) {
- EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength());
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kType, kId));
- EXPECT_EQ(RTP_ONE_BYTE_HEADER_LENGTH_IN_BYTES + kTypeLength,
- rtp_sender_->RtpHeaderExtensionTotalLength());
- EXPECT_EQ(0, rtp_sender_->DeregisterRtpHeaderExtension(kType));
- EXPECT_EQ(0, rtp_sender_->RtpHeaderExtensionTotalLength());
-}
-
-TEST_F(RtpSenderTest, BuildRTPPacket) {
- WebRtc_Word32 length = rtp_sender_->BuildRTPheader(packet_,
- kPayload,
- kMarkerBit,
- kTimestamp);
- EXPECT_EQ(12, length);
-
- // Verify
- webrtc::ModuleRTPUtility::RTPHeaderParser rtpParser(packet_, length);
- webrtc::WebRtcRTPHeader rtp_header;
-
- RtpHeaderExtensionMap map;
- map.Register(kType, kId);
- const bool valid_rtp_header = rtpParser.Parse(rtp_header, &map);
-
- ASSERT_TRUE(valid_rtp_header);
- ASSERT_FALSE(rtpParser.RTCP());
- VerifyRTPHeaderCommon(rtp_header);
- EXPECT_EQ(length, rtp_header.header.headerLength);
- EXPECT_EQ(0, rtp_header.extension.transmissionTimeOffset);
-}
-
-TEST_F(RtpSenderTest, BuildRTPPacketWithExtension) {
- EXPECT_EQ(0, rtp_sender_->SetTransmissionTimeOffset(kTimeOffset));
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kType, kId));
-
- WebRtc_Word32 length = rtp_sender_->BuildRTPheader(packet_,
- kPayload,
- kMarkerBit,
- kTimestamp);
- EXPECT_EQ(12 + rtp_sender_->RtpHeaderExtensionTotalLength(), length);
-
- // Verify
- webrtc::ModuleRTPUtility::RTPHeaderParser rtpParser(packet_, length);
- webrtc::WebRtcRTPHeader rtp_header;
-
- RtpHeaderExtensionMap map;
- map.Register(kType, kId);
- const bool valid_rtp_header = rtpParser.Parse(rtp_header, &map);
-
- ASSERT_TRUE(valid_rtp_header);
- ASSERT_FALSE(rtpParser.RTCP());
- VerifyRTPHeaderCommon(rtp_header);
- EXPECT_EQ(length, rtp_header.header.headerLength);
- EXPECT_EQ(kTimeOffset, rtp_header.extension.transmissionTimeOffset);
-
- // Parse without map extension
- webrtc::WebRtcRTPHeader rtp_header2;
- const bool valid_rtp_header2 = rtpParser.Parse(rtp_header2, NULL);
-
- ASSERT_TRUE(valid_rtp_header2);
- VerifyRTPHeaderCommon(rtp_header2);
- EXPECT_EQ(length, rtp_header2.header.headerLength);
- EXPECT_EQ(0, rtp_header2.extension.transmissionTimeOffset);
-}
-
-TEST_F(RtpSenderTest, NoTrafficSmoothing) {
- EXPECT_EQ(0, rtp_sender_->RegisterSendTransport(&transport_));
-
- WebRtc_Word32 rtp_length = rtp_sender_->BuildRTPheader(packet_,
- kPayload,
- kMarkerBit,
- kTimestamp);
-
- // Packet should be sent immediately.
- EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
- kAllowRetransmission));
- EXPECT_EQ(1, transport_.packets_sent_);
- EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
-}
-
-TEST_F(RtpSenderTest, TrafficSmoothing) {
- rtp_sender_->SetTransmissionSmoothingStatus(true);
- EXPECT_EQ(0, rtp_sender_->SetStorePacketsStatus(true, 10));
- EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(kType, kId));
- EXPECT_EQ(0, rtp_sender_->RegisterSendTransport(&transport_));
-
- WebRtc_Word32 rtp_length = rtp_sender_->BuildRTPheader(packet_,
- kPayload,
- kMarkerBit,
- kTimestamp);
-
- // Packet should be stored in a send bucket.
- EXPECT_EQ(0, rtp_sender_->SendToNetwork(packet_, 0, rtp_length,
- kAllowRetransmission));
- EXPECT_EQ(0, transport_.packets_sent_);
-
- const int kStoredTimeInMs = 100;
- fake_clock_.IncrementTime(kStoredTimeInMs);
-
- // Process send bucket. Packet should now be sent.
- rtp_sender_->ProcessSendToNetwork();
- EXPECT_EQ(1, transport_.packets_sent_);
- EXPECT_EQ(rtp_length, transport_.last_sent_packet_len_);
-
- // Parse sent packet.
- webrtc::ModuleRTPUtility::RTPHeaderParser rtpParser(
- transport_.last_sent_packet_, rtp_length);
- webrtc::WebRtcRTPHeader rtp_header;
-
- RtpHeaderExtensionMap map;
- map.Register(kType, kId);
- const bool valid_rtp_header = rtpParser.Parse(rtp_header, &map);
- ASSERT_TRUE(valid_rtp_header);
-
- // Verify transmission time offset.
- EXPECT_EQ(kStoredTimeInMs * 90, rtp_header.extension.transmissionTimeOffset);
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_sender_video.cc b/trunk/src/modules/rtp_rtcp/source/rtp_sender_video.cc
deleted file mode 100644
index 7a15208..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_sender_video.cc
+++ /dev/null
@@ -1,619 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rtp_sender_video.h"
-
-#include "critical_section_wrapper.h"
-#include "trace.h"
-
-#include "rtp_utility.h"
-
-#include <string.h> // memcpy
-#include <cassert> // assert
-#include <cstdlib> // srand
-
-#include "rtp_format_vp8.h"
-
-namespace webrtc {
-enum { REDForFECHeaderLength = 1 };
-
-RTPSenderVideo::RTPSenderVideo(const WebRtc_Word32 id,
- RtpRtcpClock* clock,
- RTPSenderInterface* rtpSender) :
- _id(id),
- _rtpSender(*rtpSender),
- _sendVideoCritsect(CriticalSectionWrapper::CreateCriticalSection()),
-
- _videoType(kRtpNoVideo),
- _videoCodecInformation(NULL),
- _maxBitrate(0),
- _retransmissionSettings(kRetransmitBaseLayer),
-
- // Generic FEC
- _fec(id),
- _fecEnabled(false),
- _payloadTypeRED(-1),
- _payloadTypeFEC(-1),
- _codeRateKey(0),
- _codeRateDelta(0),
- _useUepProtectionKey(false),
- _useUepProtectionDelta(false),
- _fecProtectionFactor(0),
- _fecUseUepProtection(false),
- _numberFirstPartition(0),
- _fecOverheadRate(clock),
- _videoBitrate(clock) {
-}
-
-RTPSenderVideo::~RTPSenderVideo()
-{
- if(_videoCodecInformation)
- {
- delete _videoCodecInformation;
- }
- delete _sendVideoCritsect;
-}
-
-WebRtc_Word32
-RTPSenderVideo::Init()
-{
- CriticalSectionScoped cs(_sendVideoCritsect);
-
- _retransmissionSettings = kRetransmitBaseLayer;
- _fecEnabled = false;
- _payloadTypeRED = -1;
- _payloadTypeFEC = -1;
- _codeRateKey = 0;
- _codeRateDelta = 0;
- _useUepProtectionKey = false;
- _useUepProtectionDelta = false;
- _fecProtectionFactor = 0;
- _fecUseUepProtection = false;
- _numberFirstPartition = 0;
- _fecOverheadRate.Init();
- return 0;
-}
-
-void
-RTPSenderVideo::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
-}
-
-void
-RTPSenderVideo::SetVideoCodecType(RtpVideoCodecTypes videoType)
-{
- CriticalSectionScoped cs(_sendVideoCritsect);
- _videoType = videoType;
-}
-
-RtpVideoCodecTypes
-RTPSenderVideo::VideoCodecType() const
-{
- return _videoType;
-}
-
-WebRtc_Word32 RTPSenderVideo::RegisterVideoPayload(
- const char payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 maxBitRate,
- ModuleRTPUtility::Payload*& payload) {
- CriticalSectionScoped cs(_sendVideoCritsect);
-
- RtpVideoCodecTypes videoType = kRtpNoVideo;
- if (ModuleRTPUtility::StringCompare(payloadName, "VP8",3)) {
- videoType = kRtpVp8Video;
- } else if (ModuleRTPUtility::StringCompare(payloadName, "I420", 4)) {
- videoType = kRtpNoVideo;
- } else {
- return -1;
- }
- payload = new ModuleRTPUtility::Payload;
- payload->name[RTP_PAYLOAD_NAME_SIZE - 1] = 0;
- strncpy(payload->name, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
- payload->typeSpecific.Video.videoCodecType = videoType;
- payload->typeSpecific.Video.maxRate = maxBitRate;
- payload->audio = false;
- return 0;
-}
-
-struct RtpPacket
-{
- WebRtc_UWord16 rtpHeaderLength;
- ForwardErrorCorrection::Packet* pkt;
-};
-
-WebRtc_Word32
-RTPSenderVideo::SendVideoPacket(const FrameType frameType,
- const WebRtc_UWord8* dataBuffer,
- const WebRtc_UWord16 payloadLength,
- const WebRtc_UWord16 rtpHeaderLength,
- StorageType storage)
-{
- if(_fecEnabled)
- {
- WebRtc_Word32 retVal = 0;
-
- const bool markerBit = (dataBuffer[1] & kRtpMarkerBitMask)?true:false;
- RtpPacket* ptrGenericFEC = new RtpPacket;
- ptrGenericFEC->pkt = new ForwardErrorCorrection::Packet;
- ptrGenericFEC->pkt->length = payloadLength + rtpHeaderLength;
- ptrGenericFEC->rtpHeaderLength = rtpHeaderLength;
- memcpy(ptrGenericFEC->pkt->data, dataBuffer,
- ptrGenericFEC->pkt->length);
-
- // Add packet to FEC list
- _rtpPacketListFec.push_back(ptrGenericFEC);
- // FEC can only protect up to kMaxMediaPackets packets
- if (_mediaPacketListFec.size() <
- ForwardErrorCorrection::kMaxMediaPackets)
- {
- _mediaPacketListFec.push_back(ptrGenericFEC->pkt);
- }
-
- // Last packet in frame
- if (markerBit)
- {
-
- // Retain the RTP header of the last media packet to construct FEC
- // packet RTP headers.
- ForwardErrorCorrection::Packet lastMediaRtpHeader;
- memcpy(lastMediaRtpHeader.data,
- ptrGenericFEC->pkt->data,
- ptrGenericFEC->rtpHeaderLength);
-
- lastMediaRtpHeader.length = ptrGenericFEC->rtpHeaderLength;
- // Replace payload and clear marker bit.
- lastMediaRtpHeader.data[1] = _payloadTypeRED;
-
- // Number of first partition packets cannot exceed kMaxMediaPackets
- if (_numberFirstPartition >
- ForwardErrorCorrection::kMaxMediaPackets)
- {
- _numberFirstPartition =
- ForwardErrorCorrection::kMaxMediaPackets;
- }
-
- std::list<ForwardErrorCorrection::Packet*> fecPacketList;
- retVal = _fec.GenerateFEC(_mediaPacketListFec,
- _fecProtectionFactor,
- _numberFirstPartition,
- _fecUseUepProtection,
- &fecPacketList);
-
- int fecOverheadSent = 0;
- int videoSent = 0;
-
- while(!_rtpPacketListFec.empty())
- {
- WebRtc_UWord8 newDataBuffer[IP_PACKET_SIZE];
- memset(newDataBuffer, 0, sizeof(newDataBuffer));
-
- RtpPacket* packetToSend = _rtpPacketListFec.front();
-
- // Copy RTP header
- memcpy(newDataBuffer, packetToSend->pkt->data,
- packetToSend->rtpHeaderLength);
-
- // Get codec pltype
- WebRtc_UWord8 payloadType = newDataBuffer[1] & 0x7f;
-
- // Replace pltype
- newDataBuffer[1] &= 0x80; // reset
- newDataBuffer[1] += _payloadTypeRED; // replace
-
- // Add RED header
- // f-bit always 0
- newDataBuffer[packetToSend->rtpHeaderLength] = payloadType;
-
- // Copy payload data
- memcpy(newDataBuffer + packetToSend->rtpHeaderLength +
- REDForFECHeaderLength,
- packetToSend->pkt->data + packetToSend->rtpHeaderLength,
- packetToSend->pkt->length -
- packetToSend->rtpHeaderLength);
-
- _rtpPacketListFec.pop_front();
- // Check if _mediaPacketListFec is non-empty.
- // This list may be smaller than rtpPacketList, if the frame
- // has more than kMaxMediaPackets.
- if (!_mediaPacketListFec.empty()) {
- _mediaPacketListFec.pop_front();
- }
-
- // Send normal packet with RED header
- int packetSuccess = _rtpSender.SendToNetwork(
- newDataBuffer,
- packetToSend->pkt->length - packetToSend->rtpHeaderLength +
- REDForFECHeaderLength,
- packetToSend->rtpHeaderLength,
- storage);
-
- retVal |= packetSuccess;
-
- if (packetSuccess == 0)
- {
- videoSent += packetToSend->pkt->length +
- REDForFECHeaderLength;
- }
-
- delete packetToSend->pkt;
- delete packetToSend;
- packetToSend = NULL;
- }
- assert(_mediaPacketListFec.empty());
- assert(_rtpPacketListFec.empty());
-
- while(!fecPacketList.empty())
- {
- WebRtc_UWord8 newDataBuffer[IP_PACKET_SIZE];
-
- // Build FEC packets
- ForwardErrorCorrection::Packet* packetToSend = fecPacketList.front();
-
- // The returned FEC packets have no RTP headers.
- // Copy the last media packet's modified RTP header.
- memcpy(newDataBuffer, lastMediaRtpHeader.data,
- lastMediaRtpHeader.length);
-
- // Add sequence number
- ModuleRTPUtility::AssignUWord16ToBuffer(
- &newDataBuffer[2], _rtpSender.IncrementSequenceNumber());
-
- // Add RED header
- // f-bit always 0
- newDataBuffer[lastMediaRtpHeader.length] = _payloadTypeFEC;
-
- // Copy payload data
- memcpy(newDataBuffer + lastMediaRtpHeader.length +
- REDForFECHeaderLength,
- packetToSend->data,
- packetToSend->length);
-
- fecPacketList.pop_front();
-
- // Invalid FEC packet
- assert(packetToSend->length != 0);
-
- StorageType storage = kDontRetransmit;
- if (_retransmissionSettings & kRetransmitFECPackets) {
- storage = kAllowRetransmission;
- }
-
- // No marker bit on FEC packets, last media packet have the
- // marker send FEC packet with RED header
- int packetSuccess = _rtpSender.SendToNetwork(
- newDataBuffer,
- packetToSend->length + REDForFECHeaderLength,
- lastMediaRtpHeader.length,
- storage);
-
- retVal |= packetSuccess;
-
- if (packetSuccess == 0)
- {
- fecOverheadSent += packetToSend->length +
- REDForFECHeaderLength + lastMediaRtpHeader.length;
- }
- }
- _videoBitrate.Update(videoSent);
- _fecOverheadRate.Update(fecOverheadSent);
- }
- return retVal;
- }
- int retVal = _rtpSender.SendToNetwork(dataBuffer,
- payloadLength,
- rtpHeaderLength,
- storage);
- if (retVal == 0)
- {
- _videoBitrate.Update(payloadLength + rtpHeaderLength);
- }
- return retVal;
-}
-
-WebRtc_Word32
-RTPSenderVideo::SendRTPIntraRequest()
-{
- // RFC 2032
- // 5.2.1. Full intra-frame Request (FIR) packet
-
- WebRtc_UWord16 length = 8;
- WebRtc_UWord8 data[8];
- data[0] = 0x80;
- data[1] = 192;
- data[2] = 0;
- data[3] = 1; // length
-
- ModuleRTPUtility::AssignUWord32ToBuffer(data+4, _rtpSender.SSRC());
-
- return _rtpSender.SendToNetwork(data, 0, length, kAllowRetransmission);
-}
-
-WebRtc_Word32
-RTPSenderVideo::SetGenericFECStatus(const bool enable,
- const WebRtc_UWord8 payloadTypeRED,
- const WebRtc_UWord8 payloadTypeFEC)
-{
- _fecEnabled = enable;
- _payloadTypeRED = payloadTypeRED;
- _payloadTypeFEC = payloadTypeFEC;
- _codeRateKey = 0;
- _codeRateDelta = 0;
- _useUepProtectionKey = false;
- _useUepProtectionDelta = false;
- return 0;
-}
-
-WebRtc_Word32
-RTPSenderVideo::GenericFECStatus(bool& enable,
- WebRtc_UWord8& payloadTypeRED,
- WebRtc_UWord8& payloadTypeFEC) const
-{
- enable = _fecEnabled;
- payloadTypeRED = _payloadTypeRED;
- payloadTypeFEC = _payloadTypeFEC;
- return 0;
-}
-
-WebRtc_UWord16
-RTPSenderVideo::FECPacketOverhead() const
-{
- if (_fecEnabled)
- {
- return ForwardErrorCorrection::PacketOverhead() +
- REDForFECHeaderLength;
- }
- return 0;
-}
-
-WebRtc_Word32
-RTPSenderVideo::SetFECCodeRate(const WebRtc_UWord8 keyFrameCodeRate,
- const WebRtc_UWord8 deltaFrameCodeRate)
-{
- _codeRateKey = keyFrameCodeRate;
- _codeRateDelta = deltaFrameCodeRate;
- return 0;
-}
-
-WebRtc_Word32
-RTPSenderVideo::SetFECUepProtection(const bool keyUseUepProtection,
- const bool deltaUseUepProtection)
-{
- _useUepProtectionKey = keyUseUepProtection;
- _useUepProtectionDelta = deltaUseUepProtection;
- return 0;
-}
-
-WebRtc_Word32
-RTPSenderVideo::SendVideo(const RtpVideoCodecTypes videoType,
- const FrameType frameType,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader* fragmentation,
- VideoCodecInformation* codecInfo,
- const RTPVideoTypeHeader* rtpTypeHdr)
-{
- if( payloadSize == 0)
- {
- return -1;
- }
-
- if (frameType == kVideoFrameKey)
- {
- _fecProtectionFactor = _codeRateKey;
- _fecUseUepProtection = _useUepProtectionKey;
- } else if (videoType == kRtpVp8Video && rtpTypeHdr->VP8.temporalIdx > 0)
- {
- // In current version, we only apply FEC on the base layer.
- _fecProtectionFactor = 0;
- _fecUseUepProtection = false;
- } else
- {
- _fecProtectionFactor = _codeRateDelta;
- _fecUseUepProtection = _useUepProtectionDelta;
- }
-
- // Default setting for number of first partition packets:
- // Will be extracted in SendVP8 for VP8 codec; other codecs use 0
- _numberFirstPartition = 0;
-
- WebRtc_Word32 retVal = -1;
- switch(videoType)
- {
- case kRtpNoVideo:
- retVal = SendGeneric(payloadType,captureTimeStamp, payloadData,
- payloadSize);
- break;
- case kRtpVp8Video:
- retVal = SendVP8(frameType, payloadType, captureTimeStamp,
- payloadData, payloadSize, fragmentation, rtpTypeHdr);
- break;
- default:
- assert(false);
- break;
- }
- if(retVal <= 0)
- {
- return retVal;
- }
- return 0;
-}
-
-WebRtc_Word32
-RTPSenderVideo::SendGeneric(const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize)
-{
- WebRtc_UWord16 payloadBytesInPacket = 0;
- WebRtc_UWord32 bytesSent = 0;
- WebRtc_Word32 payloadBytesToSend = payloadSize;
-
- const WebRtc_UWord8* data = payloadData;
- WebRtc_UWord16 rtpHeaderLength = _rtpSender.RTPHeaderLength();
- WebRtc_UWord16 maxLength = _rtpSender.MaxPayloadLength() -
- FECPacketOverhead() - rtpHeaderLength;
- WebRtc_UWord8 dataBuffer[IP_PACKET_SIZE];
-
- // Fragment packet into packets of max MaxPayloadLength bytes payload.
- while (payloadBytesToSend > 0)
- {
- if (payloadBytesToSend > maxLength)
- {
- payloadBytesInPacket = maxLength;
- payloadBytesToSend -= payloadBytesInPacket;
- // MarkerBit is 0
- if(_rtpSender.BuildRTPheader(dataBuffer,
- payloadType,
- false,
- captureTimeStamp) != rtpHeaderLength)
- {
- return -1;
- }
- }
- else
- {
- payloadBytesInPacket = (WebRtc_UWord16)payloadBytesToSend;
- payloadBytesToSend = 0;
- // MarkerBit is 1
- if(_rtpSender.BuildRTPheader(dataBuffer, payloadType, true,
- captureTimeStamp) != rtpHeaderLength)
- {
- return -1;
- }
- }
-
- // Put payload in packet
- memcpy(&dataBuffer[rtpHeaderLength], &data[bytesSent],
- payloadBytesInPacket);
- bytesSent += payloadBytesInPacket;
-
- if(-1 == SendVideoPacket(kVideoFrameKey,
- dataBuffer,
- payloadBytesInPacket,
- rtpHeaderLength,
- kAllowRetransmission))
- {
- return -1;
- }
- }
- return 0;
-}
-
-VideoCodecInformation*
-RTPSenderVideo::CodecInformationVideo()
-{
- return _videoCodecInformation;
-}
-
-void
-RTPSenderVideo::SetMaxConfiguredBitrateVideo(const WebRtc_UWord32 maxBitrate)
-{
- _maxBitrate = maxBitrate;
-}
-
-WebRtc_UWord32
-RTPSenderVideo::MaxConfiguredBitrateVideo() const
-{
- return _maxBitrate;
-}
-
-WebRtc_Word32
-RTPSenderVideo::SendVP8(const FrameType frameType,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader* fragmentation,
- const RTPVideoTypeHeader* rtpTypeHdr)
-{
- const WebRtc_UWord16 rtpHeaderLength = _rtpSender.RTPHeaderLength();
-
- WebRtc_Word32 payloadBytesToSend = payloadSize;
- const WebRtc_UWord8* data = payloadData;
-
- WebRtc_UWord16 maxPayloadLengthVP8 = _rtpSender.MaxDataPayloadLength();
-
- assert(rtpTypeHdr);
- // Initialize disregarding partition boundaries: this will use kEqualSize
- // packetization mode, which produces ~equal size packets for each frame.
- RtpFormatVp8 packetizer(data, payloadBytesToSend, rtpTypeHdr->VP8,
- maxPayloadLengthVP8);
-
- StorageType storage = kAllowRetransmission;
- if (rtpTypeHdr->VP8.temporalIdx == 0 &&
- !(_retransmissionSettings & kRetransmitBaseLayer)) {
- storage = kDontRetransmit;
- }
- if (rtpTypeHdr->VP8.temporalIdx > 0 &&
- !(_retransmissionSettings & kRetransmitHigherLayers)) {
- storage = kDontRetransmit;
- }
-
- bool last = false;
- _numberFirstPartition = 0;
- while (!last)
- {
- // Write VP8 Payload Descriptor and VP8 payload.
- WebRtc_UWord8 dataBuffer[IP_PACKET_SIZE] = {0};
- int payloadBytesInPacket = 0;
- int packetStartPartition =
- packetizer.NextPacket(&dataBuffer[rtpHeaderLength],
- &payloadBytesInPacket, &last);
- if (packetStartPartition == 0)
- {
- ++_numberFirstPartition;
- }
- else if (packetStartPartition < 0)
- {
- return -1;
- }
-
- // Write RTP header.
- // Set marker bit true if this is the last packet in frame.
- _rtpSender.BuildRTPheader(dataBuffer, payloadType, last,
- captureTimeStamp);
- if (-1 == SendVideoPacket(frameType, dataBuffer, payloadBytesInPacket,
- rtpHeaderLength, storage))
- {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "RTPSenderVideo::SendVP8 failed to send packet number"
- " %d", _rtpSender.SequenceNumber());
- }
- }
- return 0;
-}
-
-void RTPSenderVideo::ProcessBitrate() {
- _videoBitrate.Process();
- _fecOverheadRate.Process();
-}
-
-WebRtc_UWord32 RTPSenderVideo::VideoBitrateSent() const {
- return _videoBitrate.BitrateLast();
-}
-
-WebRtc_UWord32 RTPSenderVideo::FecOverheadRate() const {
- return _fecOverheadRate.BitrateLast();
-}
-
-int RTPSenderVideo::SelectiveRetransmissions() const {
- return _retransmissionSettings;
-}
-
-int RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
- _retransmissionSettings = settings;
- return 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_sender_video.h b/trunk/src/modules/rtp_rtcp/source/rtp_sender_video.h
deleted file mode 100644
index 1bf7142..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_sender_video.h
+++ /dev/null
@@ -1,149 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
-
-#include <list>
-
-#include "typedefs.h"
-#include "common_types.h" // Transport
-#include "rtp_rtcp_config.h"
-
-#include "rtp_rtcp_defines.h"
-#include "rtp_utility.h"
-
-#include "video_codec_information.h"
-#include "forward_error_correction.h"
-#include "Bitrate.h"
-#include "rtp_sender.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-struct RtpPacket;
-
-class RTPSenderVideo
-{
-public:
- RTPSenderVideo(const WebRtc_Word32 id, RtpRtcpClock* clock,
- RTPSenderInterface* rtpSender);
- virtual ~RTPSenderVideo();
-
- WebRtc_Word32 Init();
-
- virtual void ChangeUniqueId(const WebRtc_Word32 id);
-
- virtual RtpVideoCodecTypes VideoCodecType() const;
-
- WebRtc_UWord16 FECPacketOverhead() const;
-
- WebRtc_Word32 RegisterVideoPayload(
- const char payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 maxBitRate,
- ModuleRTPUtility::Payload*& payload);
-
- WebRtc_Word32 SendVideo(const RtpVideoCodecTypes videoType,
- const FrameType frameType,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader* fragmentation,
- VideoCodecInformation* codecInfo,
- const RTPVideoTypeHeader* rtpTypeHdr);
-
- WebRtc_Word32 SendRTPIntraRequest();
-
- void SetVideoCodecType(RtpVideoCodecTypes type);
-
- VideoCodecInformation* CodecInformationVideo();
-
- void SetMaxConfiguredBitrateVideo(const WebRtc_UWord32 maxBitrate);
-
- WebRtc_UWord32 MaxConfiguredBitrateVideo() const;
-
- // FEC
- WebRtc_Word32 SetGenericFECStatus(const bool enable,
- const WebRtc_UWord8 payloadTypeRED,
- const WebRtc_UWord8 payloadTypeFEC);
-
- WebRtc_Word32 GenericFECStatus(bool& enable,
- WebRtc_UWord8& payloadTypeRED,
- WebRtc_UWord8& payloadTypeFEC) const;
-
- WebRtc_Word32 SetFECCodeRate(const WebRtc_UWord8 keyFrameCodeRate,
- const WebRtc_UWord8 deltaFrameCodeRate);
-
- WebRtc_Word32 SetFECUepProtection(const bool keyUseUepProtection,
- const bool deltaUseUepProtection);
-
- void ProcessBitrate();
-
- WebRtc_UWord32 VideoBitrateSent() const;
- WebRtc_UWord32 FecOverheadRate() const;
-
- int SelectiveRetransmissions() const;
- int SetSelectiveRetransmissions(uint8_t settings);
-
-protected:
- virtual WebRtc_Word32 SendVideoPacket(const FrameType frameType,
- const WebRtc_UWord8* dataBuffer,
- const WebRtc_UWord16 payloadLength,
- const WebRtc_UWord16 rtpHeaderLength,
- StorageType storage);
-
-private:
- WebRtc_Word32 SendGeneric(const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize);
-
- WebRtc_Word32 SendVP8(const FrameType frameType,
- const WebRtc_Word8 payloadType,
- const WebRtc_UWord32 captureTimeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader* fragmentation,
- const RTPVideoTypeHeader* rtpTypeHdr);
-
-private:
- WebRtc_Word32 _id;
- RTPSenderInterface& _rtpSender;
-
- CriticalSectionWrapper* _sendVideoCritsect;
- RtpVideoCodecTypes _videoType;
- VideoCodecInformation* _videoCodecInformation;
- WebRtc_UWord32 _maxBitrate;
- WebRtc_Word32 _retransmissionSettings;
-
- // FEC
- ForwardErrorCorrection _fec;
- bool _fecEnabled;
- WebRtc_Word8 _payloadTypeRED;
- WebRtc_Word8 _payloadTypeFEC;
- WebRtc_UWord8 _codeRateKey;
- WebRtc_UWord8 _codeRateDelta;
- bool _useUepProtectionKey;
- bool _useUepProtectionDelta;
- WebRtc_UWord8 _fecProtectionFactor;
- bool _fecUseUepProtection;
- unsigned int _numberFirstPartition;
- std::list<ForwardErrorCorrection::Packet*> _mediaPacketListFec;
- std::list<RtpPacket*> _rtpPacketListFec;
- // Bitrate used for FEC payload, RED headers, RTP headers for FEC packets
- // and any padding overhead.
- Bitrate _fecOverheadRate;
- // Bitrate used for video payload and RTP headers
- Bitrate _videoBitrate;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_utility.cc b/trunk/src/modules/rtp_rtcp/source/rtp_utility.cc
deleted file mode 100644
index 947796d..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_utility.cc
+++ /dev/null
@@ -1,914 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rtp_utility.h"
-
-#include <cassert>
-#include <cmath> // ceil
-#include <cstring> // memcpy
-
-#if defined(_WIN32)
-#include <Windows.h> // FILETIME
-#include <WinSock.h> // timeval
-#include <MMSystem.h> // timeGetTime
-#elif ((defined WEBRTC_LINUX) || (defined WEBRTC_MAC))
-#include <sys/time.h> // gettimeofday
-#include <time.h>
-#endif
-#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
-#include <stdio.h>
-#endif
-
-#include "trace.h"
-
-#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
-#define DEBUG_PRINT(...) \
- { \
- char msg[256]; \
- sprintf(msg, __VA_ARGS__); \
- OutputDebugString(msg); \
- }
-#else
-// special fix for visual 2003
-#define DEBUG_PRINT(exp) ((void)0)
-#endif // defined(_DEBUG) && defined(_WIN32)
-
-namespace webrtc {
-
-namespace ModuleRTPUtility {
-
-/*
- * Time routines.
- */
-
-#if defined(_WIN32)
-
-struct reference_point {
- FILETIME file_time;
- LARGE_INTEGER counterMS;
-};
-
-struct WindowsHelpTimer {
- volatile LONG _timeInMs;
- volatile LONG _numWrapTimeInMs;
- reference_point _ref_point;
-
- volatile LONG _sync_flag;
-};
-
-void Synchronize(WindowsHelpTimer* help_timer) {
- const LONG start_value = 0;
- const LONG new_value = 1;
- const LONG synchronized_value = 2;
-
- LONG compare_flag = new_value;
- while (help_timer->_sync_flag == start_value) {
- const LONG new_value = 1;
- compare_flag = InterlockedCompareExchange(
- &help_timer->_sync_flag, new_value, start_value);
- }
- if (compare_flag != start_value) {
- // This thread was not the one that incremented the sync flag.
- // Block until synchronization finishes.
- while (compare_flag != synchronized_value) {
- ::Sleep(0);
- }
- return;
- }
- // Only the synchronizing thread gets here so this part can be
- // considered single threaded.
-
- // set timer accuracy to 1 ms
- timeBeginPeriod(1);
- FILETIME ft0 = { 0, 0 },
- ft1 = { 0, 0 };
- //
- // Spin waiting for a change in system time. Get the matching
- // performance counter value for that time.
- //
- ::GetSystemTimeAsFileTime(&ft0);
- do {
- ::GetSystemTimeAsFileTime(&ft1);
-
- help_timer->_ref_point.counterMS.QuadPart = ::timeGetTime();
- ::Sleep(0);
- } while ((ft0.dwHighDateTime == ft1.dwHighDateTime) &&
- (ft0.dwLowDateTime == ft1.dwLowDateTime));
- help_timer->_ref_point.file_time = ft1;
-}
-
-void get_time(WindowsHelpTimer* help_timer, FILETIME& current_time) {
- // we can't use query performance counter due to speed stepping
- DWORD t = timeGetTime();
- // NOTE: we have a missmatch in sign between _timeInMs(LONG) and
- // (DWORD) however we only use it here without +- etc
- volatile LONG* timeInMsPtr = &help_timer->_timeInMs;
- // Make sure that we only inc wrapper once.
- DWORD old = InterlockedExchange(timeInMsPtr, t);
- if(old > t) {
- // wrap
- help_timer->_numWrapTimeInMs++;
- }
- LARGE_INTEGER elapsedMS;
- elapsedMS.HighPart = help_timer->_numWrapTimeInMs;
- elapsedMS.LowPart = t;
-
- elapsedMS.QuadPart = elapsedMS.QuadPart -
- help_timer->_ref_point.counterMS.QuadPart;
-
- // Translate to 100-nanoseconds intervals (FILETIME resolution)
- // and add to reference FILETIME to get current FILETIME.
- ULARGE_INTEGER filetime_ref_as_ul;
-
- filetime_ref_as_ul.HighPart =
- help_timer->_ref_point.file_time.dwHighDateTime;
- filetime_ref_as_ul.LowPart =
- help_timer->_ref_point.file_time.dwLowDateTime;
- filetime_ref_as_ul.QuadPart +=
- (ULONGLONG)((elapsedMS.QuadPart)*1000*10);
-
- // Copy to result
- current_time.dwHighDateTime = filetime_ref_as_ul.HighPart;
- current_time.dwLowDateTime = filetime_ref_as_ul.LowPart;
- }
-
- // A clock reading times from the Windows API.
- class WindowsSystemClock : public RtpRtcpClock {
- public:
- WindowsSystemClock(WindowsHelpTimer* helpTimer)
- : _helpTimer(helpTimer) {}
-
- virtual ~WindowsSystemClock() {}
-
- virtual WebRtc_UWord32 GetTimeInMS();
-
- virtual void CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac);
-
- private:
- WindowsHelpTimer* _helpTimer;
-};
-
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
-
-// A clock reading times from the POSIX API.
-class UnixSystemClock : public RtpRtcpClock {
-public:
- UnixSystemClock() {}
- virtual ~UnixSystemClock() {}
-
- virtual WebRtc_UWord32 GetTimeInMS();
-
- virtual void CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac);
-};
-#endif
-
-#if defined(_WIN32)
-WebRtc_UWord32 WindowsSystemClock::GetTimeInMS() {
- return timeGetTime();
-}
-
-// Use the system time (roughly synchronised to the tick, and
-// extrapolated using the system performance counter.
-void WindowsSystemClock::CurrentNTP(WebRtc_UWord32& secs,
- WebRtc_UWord32& frac) {
- const WebRtc_UWord64 FILETIME_1970 = 0x019db1ded53e8000;
-
- FILETIME StartTime;
- WebRtc_UWord64 Time;
- struct timeval tv;
-
- // We can't use query performance counter since they can change depending on
- // speed steping
- get_time(_helpTimer, StartTime);
-
- Time = (((WebRtc_UWord64) StartTime.dwHighDateTime) << 32) +
- (WebRtc_UWord64) StartTime.dwLowDateTime;
-
- // Convert the hecto-nano second time to tv format
- Time -= FILETIME_1970;
-
- tv.tv_sec = (WebRtc_UWord32)(Time / (WebRtc_UWord64)10000000);
- tv.tv_usec = (WebRtc_UWord32)((Time % (WebRtc_UWord64)10000000) / 10);
-
- double dtemp;
-
- secs = tv.tv_sec + NTP_JAN_1970;
- dtemp = tv.tv_usec / 1e6;
-
- if (dtemp >= 1) {
- dtemp -= 1;
- secs++;
- } else if (dtemp < -1) {
- dtemp += 1;
- secs--;
- }
- dtemp *= NTP_FRAC;
- frac = (WebRtc_UWord32)dtemp;
-}
-
-#elif ((defined WEBRTC_LINUX) || (defined WEBRTC_MAC))
-
-WebRtc_UWord32 UnixSystemClock::GetTimeInMS() {
- struct timeval tv;
- struct timezone tz;
- WebRtc_UWord32 val;
-
- gettimeofday(&tv, &tz);
- val = (WebRtc_UWord32)(tv.tv_sec * 1000 + tv.tv_usec / 1000);
- return val;
-}
-
-// Use the system time.
-void UnixSystemClock::CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac) {
- double dtemp;
- struct timeval tv;
- struct timezone tz;
- tz.tz_minuteswest = 0;
- tz.tz_dsttime = 0;
- gettimeofday(&tv, &tz);
-
- secs = tv.tv_sec + NTP_JAN_1970;
- dtemp = tv.tv_usec / 1e6;
- if (dtemp >= 1) {
- dtemp -= 1;
- secs++;
- } else if (dtemp < -1) {
- dtemp += 1;
- secs--;
- }
- dtemp *= NTP_FRAC;
- frac = (WebRtc_UWord32)dtemp;
-}
-#endif
-
-#if defined(_WIN32)
-// Keeps the global state for the Windows implementation of RtpRtcpClock.
-// Note that this is a POD. Only PODs are allowed to have static storage
-// duration according to the Google Style guide.
-static WindowsHelpTimer global_help_timer = {0, 0, {{ 0, 0}, 0}, 0};
-#endif
-
-RtpRtcpClock* GetSystemClock() {
-#if defined(_WIN32)
- return new WindowsSystemClock(&global_help_timer);
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
- return new UnixSystemClock();
-#else
- return NULL;
-#endif
-}
-
-WebRtc_UWord32 GetCurrentRTP(RtpRtcpClock* clock, WebRtc_UWord32 freq) {
- const bool use_global_clock = (clock == NULL);
- RtpRtcpClock* local_clock = clock;
- if (use_global_clock) {
- local_clock = GetSystemClock();
- }
- WebRtc_UWord32 secs = 0, frac = 0;
- local_clock->CurrentNTP(secs, frac);
- if (use_global_clock) {
- delete local_clock;
- }
- return ConvertNTPTimeToRTP(secs, frac, freq);
-}
-
-WebRtc_UWord32 ConvertNTPTimeToRTP(WebRtc_UWord32 NTPsec,
- WebRtc_UWord32 NTPfrac,
- WebRtc_UWord32 freq) {
- float ftemp = (float)NTPfrac / (float)NTP_FRAC;
- WebRtc_UWord32 tmp = (WebRtc_UWord32)(ftemp * freq);
- return NTPsec * freq + tmp;
-}
-
-WebRtc_UWord32 ConvertNTPTimeToMS(WebRtc_UWord32 NTPsec,
- WebRtc_UWord32 NTPfrac) {
- int freq = 1000;
- float ftemp = (float)NTPfrac / (float)NTP_FRAC;
- WebRtc_UWord32 tmp = (WebRtc_UWord32)(ftemp * freq);
- WebRtc_UWord32 MStime = NTPsec * freq + tmp;
- return MStime;
-}
-
-bool OldTimestamp(uint32_t newTimestamp,
- uint32_t existingTimestamp,
- bool* wrapped) {
- bool tmpWrapped =
- (newTimestamp < 0x0000ffff && existingTimestamp > 0xffff0000) ||
- (newTimestamp > 0xffff0000 && existingTimestamp < 0x0000ffff);
- *wrapped = tmpWrapped;
- if (existingTimestamp > newTimestamp && !tmpWrapped) {
- return true;
- } else if (existingTimestamp <= newTimestamp && !tmpWrapped) {
- return false;
- } else if (existingTimestamp < newTimestamp && tmpWrapped) {
- return true;
- } else {
- return false;
- }
-}
-
-/*
- * Misc utility routines
- */
-
-#if defined(_WIN32)
-bool StringCompare(const char* str1, const char* str2,
- const WebRtc_UWord32 length) {
- return (_strnicmp(str1, str2, length) == 0) ? true : false;
-}
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
-bool StringCompare(const char* str1, const char* str2,
- const WebRtc_UWord32 length) {
- return (strncasecmp(str1, str2, length) == 0) ? true : false;
-}
-#endif
-
-#if !defined(WEBRTC_LITTLE_ENDIAN) && !defined(WEBRTC_BIG_ENDIAN)
-#error Either WEBRTC_LITTLE_ENDIAN or WEBRTC_BIG_ENDIAN must be defined
-#endif
-
-/* for RTP/RTCP
- All integer fields are carried in network byte order, that is, most
- significant byte (octet) first. AKA big-endian.
-*/
-void AssignUWord32ToBuffer(WebRtc_UWord8* dataBuffer, WebRtc_UWord32 value) {
-#if defined(WEBRTC_LITTLE_ENDIAN)
- dataBuffer[0] = static_cast<WebRtc_UWord8>(value >> 24);
- dataBuffer[1] = static_cast<WebRtc_UWord8>(value >> 16);
- dataBuffer[2] = static_cast<WebRtc_UWord8>(value >> 8);
- dataBuffer[3] = static_cast<WebRtc_UWord8>(value);
-#else
- WebRtc_UWord32* ptr = reinterpret_cast<WebRtc_UWord32*>(dataBuffer);
- ptr[0] = value;
-#endif
-}
-
-void AssignUWord24ToBuffer(WebRtc_UWord8* dataBuffer, WebRtc_UWord32 value) {
-#if defined(WEBRTC_LITTLE_ENDIAN)
- dataBuffer[0] = static_cast<WebRtc_UWord8>(value >> 16);
- dataBuffer[1] = static_cast<WebRtc_UWord8>(value >> 8);
- dataBuffer[2] = static_cast<WebRtc_UWord8>(value);
-#else
- dataBuffer[0] = static_cast<WebRtc_UWord8>(value);
- dataBuffer[1] = static_cast<WebRtc_UWord8>(value >> 8);
- dataBuffer[2] = static_cast<WebRtc_UWord8>(value >> 16);
-#endif
-}
-
-void AssignUWord16ToBuffer(WebRtc_UWord8* dataBuffer, WebRtc_UWord16 value) {
-#if defined(WEBRTC_LITTLE_ENDIAN)
- dataBuffer[0] = static_cast<WebRtc_UWord8>(value >> 8);
- dataBuffer[1] = static_cast<WebRtc_UWord8>(value);
-#else
- WebRtc_UWord16* ptr = reinterpret_cast<WebRtc_UWord16*>(dataBuffer);
- ptr[0] = value;
-#endif
-}
-
-WebRtc_UWord16 BufferToUWord16(const WebRtc_UWord8* dataBuffer) {
-#if defined(WEBRTC_LITTLE_ENDIAN)
- return (dataBuffer[0] << 8) + dataBuffer[1];
-#else
- return *reinterpret_cast<const WebRtc_UWord16*>(dataBuffer);
-#endif
-}
-
-WebRtc_UWord32 BufferToUWord24(const WebRtc_UWord8* dataBuffer) {
- return (dataBuffer[0] << 16) + (dataBuffer[1] << 8) + dataBuffer[2];
-}
-
-WebRtc_UWord32 BufferToUWord32(const WebRtc_UWord8* dataBuffer) {
-#if defined(WEBRTC_LITTLE_ENDIAN)
- return (dataBuffer[0] << 24) + (dataBuffer[1] << 16) + (dataBuffer[2] << 8) +
- dataBuffer[3];
-#else
- return *reinterpret_cast<const WebRtc_UWord32*>(dataBuffer);
-#endif
-}
-
-WebRtc_UWord32 pow2(WebRtc_UWord8 exp) {
- return 1 << exp;
-}
-
-void RTPPayload::SetType(RtpVideoCodecTypes videoType) {
- type = videoType;
-
- switch (type) {
- case kRtpNoVideo:
- break;
- case kRtpVp8Video: {
- info.VP8.nonReferenceFrame = false;
- info.VP8.beginningOfPartition = false;
- info.VP8.partitionID = 0;
- info.VP8.hasPictureID = false;
- info.VP8.hasTl0PicIdx = false;
- info.VP8.hasTID = false;
- info.VP8.hasKeyIdx = false;
- info.VP8.pictureID = -1;
- info.VP8.tl0PicIdx = -1;
- info.VP8.tID = -1;
- info.VP8.layerSync = false;
- info.VP8.frameWidth = 0;
- info.VP8.frameHeight = 0;
- break;
- }
- default:
- break;
- }
-}
-
-RTPHeaderParser::RTPHeaderParser(const WebRtc_UWord8* rtpData,
- const WebRtc_UWord32 rtpDataLength)
- : _ptrRTPDataBegin(rtpData),
- _ptrRTPDataEnd(rtpData ? (rtpData + rtpDataLength) : NULL) {
-}
-
-RTPHeaderParser::~RTPHeaderParser() {
-}
-
-bool RTPHeaderParser::RTCP() const {
- // 72 to 76 is reserved for RTP
- // 77 to 79 is not reserver but they are not assigned we will block them
- // for RTCP 200 SR == marker bit + 72
- // for RTCP 204 APP == marker bit + 76
- /*
- * RTCP
- *
- * FIR full INTRA-frame request 192 [RFC2032] supported
- * NACK negative acknowledgement 193 [RFC2032]
- * IJ Extended inter-arrival jitter report 195 [RFC-ietf-avt-rtp-toff
- * set-07.txt] http://tools.ietf.org/html/draft-ietf-avt-rtp-toffset-07
- * SR sender report 200 [RFC3551] supported
- * RR receiver report 201 [RFC3551] supported
- * SDES source description 202 [RFC3551] supported
- * BYE goodbye 203 [RFC3551] supported
- * APP application-defined 204 [RFC3551] ignored
- * RTPFB Transport layer FB message 205 [RFC4585] supported
- * PSFB Payload-specific FB message 206 [RFC4585] supported
- * XR extended report 207 [RFC3611] supported
- */
-
- /* 205 RFC 5104
- * FMT 1 NACK supported
- * FMT 2 reserved
- * FMT 3 TMMBR supported
- * FMT 4 TMMBN supported
- */
-
- /* 206 RFC 5104
- * FMT 1: Picture Loss Indication (PLI) supported
- * FMT 2: Slice Lost Indication (SLI)
- * FMT 3: Reference Picture Selection Indication (RPSI)
- * FMT 4: Full Intra Request (FIR) Command supported
- * FMT 5: Temporal-Spatial Trade-off Request (TSTR)
- * FMT 6: Temporal-Spatial Trade-off Notification (TSTN)
- * FMT 7: Video Back Channel Message (VBCM)
- * FMT 15: Application layer FB message
- */
-
- const WebRtc_UWord8 payloadType = _ptrRTPDataBegin[1];
-
- bool RTCP = false;
-
- // check if this is a RTCP packet
- switch (payloadType) {
- case 192:
- RTCP = true;
- break;
- case 193:
- // not supported
- // pass through and check for a potential RTP packet
- break;
- case 195:
- case 200:
- case 201:
- case 202:
- case 203:
- case 204:
- case 205:
- case 206:
- case 207:
- RTCP = true;
- break;
- }
- return RTCP;
-}
-
-bool RTPHeaderParser::Parse(WebRtcRTPHeader& parsedPacket,
- RtpHeaderExtensionMap* ptrExtensionMap) const {
- const ptrdiff_t length = _ptrRTPDataEnd - _ptrRTPDataBegin;
-
- if (length < 12) {
- return false;
- }
-
- // Version
- const WebRtc_UWord8 V = _ptrRTPDataBegin[0] >> 6;
- // Padding
- const bool P = ((_ptrRTPDataBegin[0] & 0x20) == 0) ? false : true;
- // eXtension
- const bool X = ((_ptrRTPDataBegin[0] & 0x10) == 0) ? false : true;
- const WebRtc_UWord8 CC = _ptrRTPDataBegin[0] & 0x0f;
- const bool M = ((_ptrRTPDataBegin[1] & 0x80) == 0) ? false : true;
-
- const WebRtc_UWord8 PT = _ptrRTPDataBegin[1] & 0x7f;
-
- const WebRtc_UWord16 sequenceNumber = (_ptrRTPDataBegin[2] << 8) +
- _ptrRTPDataBegin[3];
-
- const WebRtc_UWord8* ptr = &_ptrRTPDataBegin[4];
-
- WebRtc_UWord32 RTPTimestamp = *ptr++ << 24;
- RTPTimestamp += *ptr++ << 16;
- RTPTimestamp += *ptr++ << 8;
- RTPTimestamp += *ptr++;
-
- WebRtc_UWord32 SSRC = *ptr++ << 24;
- SSRC += *ptr++ << 16;
- SSRC += *ptr++ << 8;
- SSRC += *ptr++;
-
- if (V != 2) {
- return false;
- }
-
- const WebRtc_UWord8 CSRCocts = CC * 4;
-
- if ((ptr + CSRCocts) > _ptrRTPDataEnd) {
- return false;
- }
-
- parsedPacket.header.markerBit = M;
- parsedPacket.header.payloadType = PT;
- parsedPacket.header.sequenceNumber = sequenceNumber;
- parsedPacket.header.timestamp = RTPTimestamp;
- parsedPacket.header.ssrc = SSRC;
- parsedPacket.header.numCSRCs = CC;
- parsedPacket.header.paddingLength = P ? *(_ptrRTPDataEnd - 1) : 0;
-
- for (unsigned int i = 0; i < CC; ++i) {
- WebRtc_UWord32 CSRC = *ptr++ << 24;
- CSRC += *ptr++ << 16;
- CSRC += *ptr++ << 8;
- CSRC += *ptr++;
- parsedPacket.header.arrOfCSRCs[i] = CSRC;
- }
- parsedPacket.type.Audio.numEnergy = parsedPacket.header.numCSRCs;
-
- parsedPacket.header.headerLength = 12 + CSRCocts;
-
- // If in effect, MAY be omitted for those packets for which the offset
- // is zero.
- parsedPacket.extension.transmissionTimeOffset = 0;
-
- if (X) {
- /* RTP header extension, RFC 3550.
- 0 1 2 3
- 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | defined by profile | length |
- +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- | header extension |
- | .... |
- */
- const ptrdiff_t remain = _ptrRTPDataEnd - ptr;
- if (remain < 4) {
- return false;
- }
-
- parsedPacket.header.headerLength += 4;
-
- WebRtc_UWord16 definedByProfile = *ptr++ << 8;
- definedByProfile += *ptr++;
-
- WebRtc_UWord16 XLen = *ptr++ << 8;
- XLen += *ptr++; // in 32 bit words
- XLen *= 4; // in octs
-
- if (remain < (4 + XLen)) {
- return false;
- }
- if (definedByProfile == RTP_ONE_BYTE_HEADER_EXTENSION) {
- const WebRtc_UWord8* ptrRTPDataExtensionEnd = ptr + XLen;
- ParseOneByteExtensionHeader(parsedPacket,
- ptrExtensionMap,
- ptrRTPDataExtensionEnd,
- ptr);
- }
- parsedPacket.header.headerLength += XLen;
- }
- return true;
-}
-
-void RTPHeaderParser::ParseOneByteExtensionHeader(
- WebRtcRTPHeader& parsedPacket,
- const RtpHeaderExtensionMap* ptrExtensionMap,
- const WebRtc_UWord8* ptrRTPDataExtensionEnd,
- const WebRtc_UWord8* ptr) const {
- if (!ptrExtensionMap) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1, "No extension map.");
- return;
- }
-
- while (ptrRTPDataExtensionEnd - ptr > 0) {
- // 0
- // 0 1 2 3 4 5 6 7
- // +-+-+-+-+-+-+-+-+
- // | ID | len |
- // +-+-+-+-+-+-+-+-+
-
- const WebRtc_UWord8 id = (*ptr & 0xf0) >> 4;
- const WebRtc_UWord8 len = (*ptr & 0x0f);
- ptr++;
-
- if (id == 15) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "Ext id: 15 encountered, parsing terminated.");
- return;
- }
-
- RTPExtensionType type;
- if (ptrExtensionMap->GetType(id, &type) != 0) {
- WEBRTC_TRACE(kTraceStream, kTraceRtpRtcp, -1,
- "Failed to find extension id: %d", id);
- return;
- }
-
- switch (type) {
- case kRtpExtensionTransmissionTimeOffset: {
- if (len != 2) {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "Incorrect transmission time offset len: %d", len);
- return;
- }
- // 0 1 2 3
- // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // | ID | len=2 | transmission offset |
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
-
- WebRtc_Word32 transmissionTimeOffset = *ptr++ << 16;
- transmissionTimeOffset += *ptr++ << 8;
- transmissionTimeOffset += *ptr++;
- parsedPacket.extension.transmissionTimeOffset = transmissionTimeOffset;
- break;
- }
- case kRtpExtensionAudioLevel: {
- // --- Only used for debugging ---
- // 0 1 2 3
- // 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1 2 3 4 5 6 7 8 9 0 1
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- // | ID | len=0 |V| level | 0x00 | 0x00 |
- // +-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+-+
- //
-
- // Parse out the fields but only use it for debugging for now.
- // const WebRtc_UWord8 V = (*ptr & 0x80) >> 7;
- // const WebRtc_UWord8 level = (*ptr & 0x7f);
- // DEBUG_PRINT("RTP_AUDIO_LEVEL_UNIQUE_ID: ID=%u, len=%u, V=%u,
- // level=%u", ID, len, V, level);
- break;
- }
- default: {
- WEBRTC_TRACE(kTraceWarning, kTraceRtpRtcp, -1,
- "Extension type not implemented.");
- return;
- }
- }
- WebRtc_UWord8 num_bytes = ParsePaddingBytes(ptrRTPDataExtensionEnd, ptr);
- ptr += num_bytes;
- }
-}
-
-WebRtc_UWord8 RTPHeaderParser::ParsePaddingBytes(
- const WebRtc_UWord8* ptrRTPDataExtensionEnd,
- const WebRtc_UWord8* ptr) const {
-
- WebRtc_UWord8 num_zero_bytes = 0;
- while (ptrRTPDataExtensionEnd - ptr > 0) {
- if (*ptr != 0) {
- return num_zero_bytes;
- }
- ptr++;
- num_zero_bytes++;
- }
- return num_zero_bytes;
-}
-
-// RTP payload parser
-RTPPayloadParser::RTPPayloadParser(const RtpVideoCodecTypes videoType,
- const WebRtc_UWord8* payloadData,
- WebRtc_UWord16 payloadDataLength,
- WebRtc_Word32 id)
- :
- _id(id),
- _dataPtr(payloadData),
- _dataLength(payloadDataLength),
- _videoType(videoType) {
-}
-
-RTPPayloadParser::~RTPPayloadParser() {
-}
-
-bool RTPPayloadParser::Parse(RTPPayload& parsedPacket) const {
- parsedPacket.SetType(_videoType);
-
- switch (_videoType) {
- case kRtpNoVideo:
- return ParseGeneric(parsedPacket);
- case kRtpVp8Video:
- return ParseVP8(parsedPacket);
- default:
- return false;
- }
-}
-
-bool RTPPayloadParser::ParseGeneric(RTPPayload& /*parsedPacket*/) const {
- return false;
-}
-
-//
-// VP8 format:
-//
-// Payload descriptor
-// 0 1 2 3 4 5 6 7
-// +-+-+-+-+-+-+-+-+
-// |X|R|N|S|PartID | (REQUIRED)
-// +-+-+-+-+-+-+-+-+
-// X: |I|L|T|K| RSV | (OPTIONAL)
-// +-+-+-+-+-+-+-+-+
-// I: | PictureID | (OPTIONAL)
-// +-+-+-+-+-+-+-+-+
-// L: | TL0PICIDX | (OPTIONAL)
-// +-+-+-+-+-+-+-+-+
-// T/K: |TID:Y| KEYIDX | (OPTIONAL)
-// +-+-+-+-+-+-+-+-+
-//
-// Payload header (considered part of the actual payload, sent to decoder)
-// 0 1 2 3 4 5 6 7
-// +-+-+-+-+-+-+-+-+
-// |Size0|H| VER |P|
-// +-+-+-+-+-+-+-+-+
-// | ... |
-// + +
-
-bool RTPPayloadParser::ParseVP8(RTPPayload& parsedPacket) const {
- RTPPayloadVP8* vp8 = &parsedPacket.info.VP8;
- const WebRtc_UWord8* dataPtr = _dataPtr;
- int dataLength = _dataLength;
-
- // Parse mandatory first byte of payload descriptor
- bool extension = (*dataPtr & 0x80) ? true : false; // X bit
- vp8->nonReferenceFrame = (*dataPtr & 0x20) ? true : false; // N bit
- vp8->beginningOfPartition = (*dataPtr & 0x10) ? true : false; // S bit
- vp8->partitionID = (*dataPtr & 0x0F); // PartID field
-
- if (vp8->partitionID > 8) {
- // Weak check for corrupt data: PartID MUST NOT be larger than 8.
- return false;
- }
-
- // Advance dataPtr and decrease remaining payload size
- dataPtr++;
- dataLength--;
-
- if (extension) {
- const int parsedBytes = ParseVP8Extension(vp8, dataPtr, dataLength);
- if (parsedBytes < 0) return false;
- dataPtr += parsedBytes;
- dataLength -= parsedBytes;
- }
-
- if (dataLength <= 0) {
- WEBRTC_TRACE(kTraceError, kTraceRtpRtcp, _id,
- "Error parsing VP8 payload descriptor; payload too short");
- return false;
- }
-
- // Read P bit from payload header (only at beginning of first partition)
- if (dataLength > 0 && vp8->beginningOfPartition && vp8->partitionID == 0) {
- parsedPacket.frameType = (*dataPtr & 0x01) ? kPFrame : kIFrame;
- } else {
- parsedPacket.frameType = kPFrame;
- }
- if (0 != ParseVP8FrameSize(parsedPacket, dataPtr, dataLength)) {
- return false;
- }
- parsedPacket.info.VP8.data = dataPtr;
- parsedPacket.info.VP8.dataLength = dataLength;
- return true;
-}
-
-int RTPPayloadParser::ParseVP8FrameSize(RTPPayload& parsedPacket,
- const WebRtc_UWord8* dataPtr,
- int dataLength) const {
- if (parsedPacket.frameType != kIFrame) {
- // Included in payload header for I-frames.
- return 0;
- }
- if (dataLength < 10) {
- // For an I-frame we should always have the uncompressed VP8 header
- // in the beginning of the partition.
- return -1;
- }
- RTPPayloadVP8* vp8 = &parsedPacket.info.VP8;
- vp8->frameWidth = ((dataPtr[7] << 8) + dataPtr[6]) & 0x3FFF;
- vp8->frameHeight = ((dataPtr[9] << 8) + dataPtr[8]) & 0x3FFF;
- return 0;
-}
-
-int RTPPayloadParser::ParseVP8Extension(RTPPayloadVP8* vp8,
- const WebRtc_UWord8* dataPtr,
- int dataLength) const {
- int parsedBytes = 0;
- if (dataLength <= 0) return -1;
- // Optional X field is present
- vp8->hasPictureID = (*dataPtr & 0x80) ? true : false; // I bit
- vp8->hasTl0PicIdx = (*dataPtr & 0x40) ? true : false; // L bit
- vp8->hasTID = (*dataPtr & 0x20) ? true : false; // T bit
- vp8->hasKeyIdx = (*dataPtr & 0x10) ? true : false; // K bit
-
- // Advance dataPtr and decrease remaining payload size
- dataPtr++;
- parsedBytes++;
- dataLength--;
-
- if (vp8->hasPictureID) {
- if (ParseVP8PictureID(vp8, &dataPtr, &dataLength, &parsedBytes) != 0) {
- return -1;
- }
- }
-
- if (vp8->hasTl0PicIdx) {
- if (ParseVP8Tl0PicIdx(vp8, &dataPtr, &dataLength, &parsedBytes) != 0) {
- return -1;
- }
- }
-
- if (vp8->hasTID || vp8->hasKeyIdx) {
- if (ParseVP8TIDAndKeyIdx(vp8, &dataPtr, &dataLength, &parsedBytes) != 0) {
- return -1;
- }
- }
- return parsedBytes;
-}
-
-int RTPPayloadParser::ParseVP8PictureID(RTPPayloadVP8* vp8,
- const WebRtc_UWord8** dataPtr,
- int* dataLength,
- int* parsedBytes) const {
- if (*dataLength <= 0) return -1;
- vp8->pictureID = (**dataPtr & 0x7F);
- if (**dataPtr & 0x80) {
- (*dataPtr)++;
- (*parsedBytes)++;
- if (--(*dataLength) <= 0) return -1;
- // PictureID is 15 bits
- vp8->pictureID = (vp8->pictureID << 8) +** dataPtr;
- }
- (*dataPtr)++;
- (*parsedBytes)++;
- (*dataLength)--;
- return 0;
-}
-
-int RTPPayloadParser::ParseVP8Tl0PicIdx(RTPPayloadVP8* vp8,
- const WebRtc_UWord8** dataPtr,
- int* dataLength,
- int* parsedBytes) const {
- if (*dataLength <= 0) return -1;
- vp8->tl0PicIdx = **dataPtr;
- (*dataPtr)++;
- (*parsedBytes)++;
- (*dataLength)--;
- return 0;
-}
-
-int RTPPayloadParser::ParseVP8TIDAndKeyIdx(RTPPayloadVP8* vp8,
- const WebRtc_UWord8** dataPtr,
- int* dataLength,
- int* parsedBytes) const {
- if (*dataLength <= 0) return -1;
- if (vp8->hasTID) {
- vp8->tID = ((**dataPtr >> 6) & 0x03);
- vp8->layerSync = (**dataPtr & 0x20) ? true : false; // Y bit
- }
- if (vp8->hasKeyIdx) {
- vp8->keyIdx = (**dataPtr & 0x1F);
- }
- (*dataPtr)++;
- (*parsedBytes)++;
- (*dataLength)--;
- return 0;
-}
-
-} // namespace ModuleRTPUtility
-
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_utility.h b/trunk/src/modules/rtp_rtcp/source/rtp_utility.h
deleted file mode 100644
index ad88f11..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_utility.h
+++ /dev/null
@@ -1,245 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
-
-#include <cstddef> // size_t, ptrdiff_t
-
-#include "typedefs.h"
-#include "rtp_header_extension.h"
-#include "rtp_rtcp_config.h"
-#include "rtp_rtcp_defines.h"
-
-namespace webrtc {
-enum RtpVideoCodecTypes
-{
- kRtpNoVideo = 0,
- kRtpFecVideo = 10,
- kRtpVp8Video = 11
-};
-
-const WebRtc_UWord8 kRtpMarkerBitMask = 0x80;
-
-namespace ModuleRTPUtility
-{
- // January 1970, in NTP seconds.
- const uint32_t NTP_JAN_1970 = 2208988800UL;
-
- // Magic NTP fractional unit.
- const double NTP_FRAC = 4.294967296E+9;
-
- struct AudioPayload
- {
- WebRtc_UWord32 frequency;
- WebRtc_UWord8 channels;
- WebRtc_UWord8 bitsPerSample;
- WebRtc_UWord32 rate;
- bool trueStereoCodec;
- };
- struct VideoPayload
- {
- RtpVideoCodecTypes videoCodecType;
- WebRtc_UWord32 maxRate;
- };
- union PayloadUnion
- {
- AudioPayload Audio;
- VideoPayload Video;
- };
- struct Payload
- {
- char name[RTP_PAYLOAD_NAME_SIZE];
- bool audio;
- PayloadUnion typeSpecific;
- };
-
- // Return a clock that reads the time as reported by the operating
- // system. The returned instances are guaranteed to read the same
- // times; in particular, they return relative times relative to
- // the same base.
- // Note that even though the instances returned by this function
- // read the same times a new object is created every time this
- // API is called. The ownership of this object belongs to the
- // caller.
- RtpRtcpClock* GetSystemClock();
-
- // Return the current RTP timestamp from the NTP timestamp
- // returned by the specified clock.
- WebRtc_UWord32 GetCurrentRTP(RtpRtcpClock* clock, WebRtc_UWord32 freq);
-
- // Return the current RTP absolute timestamp.
- WebRtc_UWord32 ConvertNTPTimeToRTP(WebRtc_UWord32 NTPsec,
- WebRtc_UWord32 NTPfrac,
- WebRtc_UWord32 freq);
-
- // Return the time in milliseconds corresponding to the specified
- // NTP timestamp.
- WebRtc_UWord32 ConvertNTPTimeToMS(WebRtc_UWord32 NTPsec,
- WebRtc_UWord32 NTPfrac);
-
- WebRtc_UWord32 pow2(WebRtc_UWord8 exp);
-
- // Returns true if |newTimestamp| is older than |existingTimestamp|.
- // |wrapped| will be set to true if there has been a wraparound between the
- // two timestamps.
- bool OldTimestamp(uint32_t newTimestamp,
- uint32_t existingTimestamp,
- bool* wrapped);
-
- bool StringCompare(const char* str1,
- const char* str2,
- const WebRtc_UWord32 length);
-
- void AssignUWord32ToBuffer(WebRtc_UWord8* dataBuffer, WebRtc_UWord32 value);
- void AssignUWord24ToBuffer(WebRtc_UWord8* dataBuffer, WebRtc_UWord32 value);
- void AssignUWord16ToBuffer(WebRtc_UWord8* dataBuffer, WebRtc_UWord16 value);
-
- /**
- * Converts a network-ordered two-byte input buffer to a host-ordered value.
- * \param[in] dataBuffer Network-ordered two-byte buffer to convert.
- * \return Host-ordered value.
- */
- WebRtc_UWord16 BufferToUWord16(const WebRtc_UWord8* dataBuffer);
-
- /**
- * Converts a network-ordered three-byte input buffer to a host-ordered value.
- * \param[in] dataBuffer Network-ordered three-byte buffer to convert.
- * \return Host-ordered value.
- */
- WebRtc_UWord32 BufferToUWord24(const WebRtc_UWord8* dataBuffer);
-
- /**
- * Converts a network-ordered four-byte input buffer to a host-ordered value.
- * \param[in] dataBuffer Network-ordered four-byte buffer to convert.
- * \return Host-ordered value.
- */
- WebRtc_UWord32 BufferToUWord32(const WebRtc_UWord8* dataBuffer);
-
- class RTPHeaderParser
- {
- public:
- RTPHeaderParser(const WebRtc_UWord8* rtpData,
- const WebRtc_UWord32 rtpDataLength);
- ~RTPHeaderParser();
-
- bool RTCP() const;
- bool Parse(WebRtcRTPHeader& parsedPacket,
- RtpHeaderExtensionMap* ptrExtensionMap = NULL) const;
-
- private:
- void ParseOneByteExtensionHeader(
- WebRtcRTPHeader& parsedPacket,
- const RtpHeaderExtensionMap* ptrExtensionMap,
- const WebRtc_UWord8* ptrRTPDataExtensionEnd,
- const WebRtc_UWord8* ptr) const;
-
- WebRtc_UWord8 ParsePaddingBytes(
- const WebRtc_UWord8* ptrRTPDataExtensionEnd,
- const WebRtc_UWord8* ptr) const;
-
- const WebRtc_UWord8* const _ptrRTPDataBegin;
- const WebRtc_UWord8* const _ptrRTPDataEnd;
- };
-
- enum FrameTypes
- {
- kIFrame, // key frame
- kPFrame // Delta frame
- };
-
- struct RTPPayloadVP8
- {
- bool nonReferenceFrame;
- bool beginningOfPartition;
- int partitionID;
- bool hasPictureID;
- bool hasTl0PicIdx;
- bool hasTID;
- bool hasKeyIdx;
- int pictureID;
- int tl0PicIdx;
- int tID;
- bool layerSync;
- int keyIdx;
- int frameWidth;
- int frameHeight;
-
- const WebRtc_UWord8* data;
- WebRtc_UWord16 dataLength;
- };
-
- union RTPPayloadUnion
- {
- RTPPayloadVP8 VP8;
- };
-
- struct RTPPayload
- {
- void SetType(RtpVideoCodecTypes videoType);
-
- RtpVideoCodecTypes type;
- FrameTypes frameType;
- RTPPayloadUnion info;
- };
-
- // RTP payload parser
- class RTPPayloadParser
- {
- public:
- RTPPayloadParser(const RtpVideoCodecTypes payloadType,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadDataLength, // Length w/o padding.
- const WebRtc_Word32 id);
-
- ~RTPPayloadParser();
-
- bool Parse(RTPPayload& parsedPacket) const;
-
- private:
- bool ParseGeneric(RTPPayload& parsedPacket) const;
-
- bool ParseVP8(RTPPayload& parsedPacket) const;
-
- int ParseVP8Extension(RTPPayloadVP8 *vp8,
- const WebRtc_UWord8 *dataPtr,
- int dataLength) const;
-
- int ParseVP8PictureID(RTPPayloadVP8 *vp8,
- const WebRtc_UWord8 **dataPtr,
- int *dataLength,
- int *parsedBytes) const;
-
- int ParseVP8Tl0PicIdx(RTPPayloadVP8 *vp8,
- const WebRtc_UWord8 **dataPtr,
- int *dataLength,
- int *parsedBytes) const;
-
- int ParseVP8TIDAndKeyIdx(RTPPayloadVP8 *vp8,
- const WebRtc_UWord8 **dataPtr,
- int *dataLength,
- int *parsedBytes) const;
-
- int ParseVP8FrameSize(RTPPayload& parsedPacket,
- const WebRtc_UWord8 *dataPtr,
- int dataLength) const;
-
- private:
- WebRtc_Word32 _id;
- const WebRtc_UWord8* _dataPtr;
- const WebRtc_UWord16 _dataLength;
- const RtpVideoCodecTypes _videoType;
- };
-
-} // namespace ModuleRTPUtility
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_UTILITY_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/rtp_utility_test.cc b/trunk/src/modules/rtp_rtcp/source/rtp_utility_test.cc
deleted file mode 100644
index eabc812..0000000
--- a/trunk/src/modules/rtp_rtcp/source/rtp_utility_test.cc
+++ /dev/null
@@ -1,288 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file conatins unit tests for the ModuleRTPUtility.
- */
-
-#include "gtest/gtest.h"
-#include "modules/rtp_rtcp/source/rtp_format_vp8.h"
-#include "modules/rtp_rtcp/source/rtp_utility.h"
-#include "typedefs.h" // NOLINT(build/include)
-
-namespace webrtc {
-
-using ModuleRTPUtility::RTPPayloadParser;
-using ModuleRTPUtility::RTPPayload;
-using ModuleRTPUtility::RTPPayloadVP8;
-
-// Payload descriptor
-// 0 1 2 3 4 5 6 7
-// +-+-+-+-+-+-+-+-+
-// |X|R|N|S|PartID | (REQUIRED)
-// +-+-+-+-+-+-+-+-+
-// X: |I|L|T|K| RSV | (OPTIONAL)
-// +-+-+-+-+-+-+-+-+
-// I: | PictureID | (OPTIONAL)
-// +-+-+-+-+-+-+-+-+
-// L: | TL0PICIDX | (OPTIONAL)
-// +-+-+-+-+-+-+-+-+
-// T/K: |TID:Y| KEYIDX | (OPTIONAL)
-// +-+-+-+-+-+-+-+-+
-//
-// Payload header
-// 0 1 2 3 4 5 6 7
-// +-+-+-+-+-+-+-+-+
-// |Size0|H| VER |P|
-// +-+-+-+-+-+-+-+-+
-// | Size1 |
-// +-+-+-+-+-+-+-+-+
-// | Size2 |
-// +-+-+-+-+-+-+-+-+
-// | Bytes 4..N of |
-// | VP8 payload |
-// : :
-// +-+-+-+-+-+-+-+-+
-// | OPTIONAL RTP |
-// | padding |
-// : :
-// +-+-+-+-+-+-+-+-+
-
-void VerifyBasicHeader(const RTPPayloadVP8 &header,
- bool N, bool S, int PartID) {
- EXPECT_EQ(N, header.nonReferenceFrame);
- EXPECT_EQ(S, header.beginningOfPartition);
- EXPECT_EQ(PartID, header.partitionID);
-}
-
-void VerifyExtensions(const RTPPayloadVP8 &header,
- bool I, bool L, bool T, bool K) {
- EXPECT_EQ(I, header.hasPictureID);
- EXPECT_EQ(L, header.hasTl0PicIdx);
- EXPECT_EQ(T, header.hasTID);
- EXPECT_EQ(K, header.hasKeyIdx);
-}
-
-TEST(ParseVP8Test, BasicHeader) {
- WebRtc_UWord8 payload[4] = {0};
- payload[0] = 0x14; // Binary 0001 0100; S = 1, PartID = 4.
- payload[1] = 0x01; // P frame.
-
- RTPPayloadParser rtpPayloadParser(kRtpVp8Video, payload, 4, 0);
-
- RTPPayload parsedPacket;
- ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
-
- EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType);
- EXPECT_EQ(kRtpVp8Video, parsedPacket.type);
-
- VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 1 /*S*/, 4 /*PartID*/);
- VerifyExtensions(parsedPacket.info.VP8, 0 /*I*/, 0 /*L*/, 0 /*T*/, 0 /*K*/);
-
- EXPECT_EQ(payload + 1, parsedPacket.info.VP8.data);
- EXPECT_EQ(4 - 1, parsedPacket.info.VP8.dataLength);
-}
-
-TEST(ParseVP8Test, PictureID) {
- WebRtc_UWord8 payload[10] = {0};
- payload[0] = 0xA0;
- payload[1] = 0x80;
- payload[2] = 17;
-
- RTPPayloadParser rtpPayloadParser(kRtpVp8Video, payload, 10, 0);
-
- RTPPayload parsedPacket;
- ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
-
- EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType);
- EXPECT_EQ(kRtpVp8Video, parsedPacket.type);
-
- VerifyBasicHeader(parsedPacket.info.VP8, 1 /*N*/, 0 /*S*/, 0 /*PartID*/);
- VerifyExtensions(parsedPacket.info.VP8, 1 /*I*/, 0 /*L*/, 0 /*T*/, 0 /*K*/);
-
- EXPECT_EQ(17, parsedPacket.info.VP8.pictureID);
-
- EXPECT_EQ(payload + 3, parsedPacket.info.VP8.data);
- EXPECT_EQ(10 - 3, parsedPacket.info.VP8.dataLength);
-
-
- // Re-use payload, but change to long PictureID.
- payload[2] = 0x80 | 17;
- payload[3] = 17;
- RTPPayloadParser rtpPayloadParser2(kRtpVp8Video, payload, 10, 0);
-
- ASSERT_TRUE(rtpPayloadParser2.Parse(parsedPacket));
-
- VerifyBasicHeader(parsedPacket.info.VP8, 1 /*N*/, 0 /*S*/, 0 /*PartID*/);
- VerifyExtensions(parsedPacket.info.VP8, 1 /*I*/, 0 /*L*/, 0 /*T*/, 0 /*K*/);
-
- EXPECT_EQ((17<<8) + 17, parsedPacket.info.VP8.pictureID);
-
- EXPECT_EQ(payload + 4, parsedPacket.info.VP8.data);
- EXPECT_EQ(10 - 4, parsedPacket.info.VP8.dataLength);
-}
-
-TEST(ParseVP8Test, Tl0PicIdx) {
- WebRtc_UWord8 payload[13] = {0};
- payload[0] = 0x90;
- payload[1] = 0x40;
- payload[2] = 17;
-
- RTPPayloadParser rtpPayloadParser(kRtpVp8Video, payload, 13, 0);
-
- RTPPayload parsedPacket;
- ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
-
- EXPECT_EQ(ModuleRTPUtility::kIFrame, parsedPacket.frameType);
- EXPECT_EQ(kRtpVp8Video, parsedPacket.type);
-
- VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 1 /*S*/, 0 /*PartID*/);
- VerifyExtensions(parsedPacket.info.VP8, 0 /*I*/, 1 /*L*/, 0 /*T*/, 0 /*K*/);
-
- EXPECT_EQ(17, parsedPacket.info.VP8.tl0PicIdx);
-
- EXPECT_EQ(payload + 3, parsedPacket.info.VP8.data);
- EXPECT_EQ(13 - 3, parsedPacket.info.VP8.dataLength);
-}
-
-TEST(ParseVP8Test, TIDAndLayerSync) {
- WebRtc_UWord8 payload[10] = {0};
- payload[0] = 0x88;
- payload[1] = 0x20;
- payload[2] = 0x80; // TID(2) + LayerSync(false)
-
- RTPPayloadParser rtpPayloadParser(kRtpVp8Video, payload, 10, 0);
-
- RTPPayload parsedPacket;
- ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
-
- EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType);
- EXPECT_EQ(kRtpVp8Video, parsedPacket.type);
-
- VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 0 /*S*/, 8 /*PartID*/);
- VerifyExtensions(parsedPacket.info.VP8, 0 /*I*/, 0 /*L*/, 1 /*T*/, 0 /*K*/);
-
- EXPECT_EQ(2, parsedPacket.info.VP8.tID);
- EXPECT_FALSE(parsedPacket.info.VP8.layerSync);
-
- EXPECT_EQ(payload + 3, parsedPacket.info.VP8.data);
- EXPECT_EQ(10 - 3, parsedPacket.info.VP8.dataLength);
-}
-
-TEST(ParseVP8Test, KeyIdx) {
- WebRtc_UWord8 payload[10] = {0};
- payload[0] = 0x88;
- payload[1] = 0x10; // K = 1.
- payload[2] = 0x11; // KEYIDX = 17 decimal.
-
- RTPPayloadParser rtpPayloadParser(kRtpVp8Video, payload, 10, 0);
-
- RTPPayload parsedPacket;
- ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
-
- EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType);
- EXPECT_EQ(kRtpVp8Video, parsedPacket.type);
-
- VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 0 /*S*/, 8 /*PartID*/);
- VerifyExtensions(parsedPacket.info.VP8, 0 /*I*/, 0 /*L*/, 0 /*T*/, 1 /*K*/);
-
- EXPECT_EQ(17, parsedPacket.info.VP8.keyIdx);
-
- EXPECT_EQ(payload + 3, parsedPacket.info.VP8.data);
- EXPECT_EQ(10 - 3, parsedPacket.info.VP8.dataLength);
-}
-
-TEST(ParseVP8Test, MultipleExtensions) {
- WebRtc_UWord8 payload[10] = {0};
- payload[0] = 0x88;
- payload[1] = 0x80 | 0x40 | 0x20 | 0x10;
- payload[2] = 0x80 | 17; // PictureID, high 7 bits.
- payload[3] = 17; // PictureID, low 8 bits.
- payload[4] = 42; // Tl0PicIdx.
- payload[5] = 0x40 | 0x20 | 0x11; // TID(1) + LayerSync(true) + KEYIDX(17).
-
- RTPPayloadParser rtpPayloadParser(kRtpVp8Video, payload, 10, 0);
-
- RTPPayload parsedPacket;
- ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
-
- EXPECT_EQ(ModuleRTPUtility::kPFrame, parsedPacket.frameType);
- EXPECT_EQ(kRtpVp8Video, parsedPacket.type);
-
- VerifyBasicHeader(parsedPacket.info.VP8, 0 /*N*/, 0 /*S*/, 8 /*PartID*/);
- VerifyExtensions(parsedPacket.info.VP8, 1 /*I*/, 1 /*L*/, 1 /*T*/, 1 /*K*/);
-
- EXPECT_EQ((17<<8) + 17, parsedPacket.info.VP8.pictureID);
- EXPECT_EQ(42, parsedPacket.info.VP8.tl0PicIdx);
- EXPECT_EQ(1, parsedPacket.info.VP8.tID);
- EXPECT_EQ(17, parsedPacket.info.VP8.keyIdx);
-
- EXPECT_EQ(payload + 6, parsedPacket.info.VP8.data);
- EXPECT_EQ(10 - 6, parsedPacket.info.VP8.dataLength);
-}
-
-TEST(ParseVP8Test, TooShortHeader) {
- WebRtc_UWord8 payload[4] = {0};
- payload[0] = 0x88;
- payload[1] = 0x80 | 0x40 | 0x20 | 0x10; // All extensions are enabled...
- payload[2] = 0x80 | 17; // ... but only 2 bytes PictureID is provided.
- payload[3] = 17; // PictureID, low 8 bits.
-
- RTPPayloadParser rtpPayloadParser(kRtpVp8Video, payload, 4, 0);
-
- RTPPayload parsedPacket;
- EXPECT_FALSE(rtpPayloadParser.Parse(parsedPacket));
-}
-
-TEST(ParseVP8Test, TestWithPacketizer) {
- WebRtc_UWord8 payload[10] = {0};
- WebRtc_UWord8 packet[20] = {0};
- RTPVideoHeaderVP8 inputHeader;
- inputHeader.nonReference = true;
- inputHeader.pictureId = 300;
- inputHeader.temporalIdx = 1;
- inputHeader.layerSync = false;
- inputHeader.tl0PicIdx = kNoTl0PicIdx; // Disable.
- inputHeader.keyIdx = 31;
- RtpFormatVp8 packetizer(payload, 10, inputHeader, 20);
- bool last;
- int send_bytes;
- ASSERT_EQ(0, packetizer.NextPacket(packet, &send_bytes, &last));
- ASSERT_TRUE(last);
-
- RTPPayloadParser rtpPayloadParser(kRtpVp8Video, packet, send_bytes, 0);
-
- RTPPayload parsedPacket;
- ASSERT_TRUE(rtpPayloadParser.Parse(parsedPacket));
-
- EXPECT_EQ(ModuleRTPUtility::kIFrame, parsedPacket.frameType);
- EXPECT_EQ(kRtpVp8Video, parsedPacket.type);
-
- VerifyBasicHeader(parsedPacket.info.VP8,
- inputHeader.nonReference /*N*/,
- 1 /*S*/,
- 0 /*PartID*/);
- VerifyExtensions(parsedPacket.info.VP8,
- 1 /*I*/,
- 0 /*L*/,
- 1 /*T*/,
- 1 /*K*/);
-
- EXPECT_EQ(inputHeader.pictureId, parsedPacket.info.VP8.pictureID);
- EXPECT_EQ(inputHeader.temporalIdx, parsedPacket.info.VP8.tID);
- EXPECT_EQ(inputHeader.layerSync, parsedPacket.info.VP8.layerSync);
- EXPECT_EQ(inputHeader.keyIdx, parsedPacket.info.VP8.keyIdx);
-
- EXPECT_EQ(packet + 5, parsedPacket.info.VP8.data);
- EXPECT_EQ(send_bytes - 5, parsedPacket.info.VP8.dataLength);
-}
-
-} // namespace
diff --git a/trunk/src/modules/rtp_rtcp/source/ssrc_database.cc b/trunk/src/modules/rtp_rtcp/source/ssrc_database.cc
deleted file mode 100644
index b3e9ab0..0000000
--- a/trunk/src/modules/rtp_rtcp/source/ssrc_database.cc
+++ /dev/null
@@ -1,217 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "ssrc_database.h"
-
-#include "critical_section_wrapper.h"
-#include "trace.h"
-
-#include <stdlib.h>
-#include <cassert>
-
-#ifdef _WIN32
- #include <windows.h>
- #include <MMSystem.h> //timeGetTime
-
-// TODO(hellner): investigate if it is necessary to disable these warnings.
- #pragma warning(disable:4311)
- #pragma warning(disable:4312)
-#else
- #include <stdio.h>
- #include <string.h>
- #include <time.h>
- #include <sys/time.h>
-#endif
-
-namespace webrtc {
-SSRCDatabase*
-SSRCDatabase::StaticInstance(CountOperation count_operation)
-{
- SSRCDatabase* impl =
- GetStaticInstance<SSRCDatabase>(count_operation);
- return impl;
-}
-
-SSRCDatabase*
-SSRCDatabase::GetSSRCDatabase()
-{
- return StaticInstance(kAddRef);
-}
-
-void
-SSRCDatabase::ReturnSSRCDatabase()
-{
- StaticInstance(kRelease);
-}
-
-WebRtc_UWord32
-SSRCDatabase::CreateSSRC()
-{
- CriticalSectionScoped lock(_critSect);
-
- WebRtc_UWord32 ssrc = GenerateRandom();
-
-#ifndef WEBRTC_NO_STL
-
- while(_ssrcMap.find(ssrc) != _ssrcMap.end())
- {
- ssrc = GenerateRandom();
- }
- _ssrcMap[ssrc] = 0;
-
-#else
- if(_sizeOfSSRC <= _numberOfSSRC)
- {
- // allocate more space
- const int newSize = _sizeOfSSRC + 10;
- WebRtc_UWord32* tempSSRCVector = new WebRtc_UWord32[newSize];
- memcpy(tempSSRCVector, _ssrcVector, _sizeOfSSRC*sizeof(WebRtc_UWord32));
- delete [] _ssrcVector;
-
- _ssrcVector = tempSSRCVector;
- _sizeOfSSRC = newSize;
- }
-
- // check if in DB
- if(_ssrcVector)
- {
- for (int i=0; i<_numberOfSSRC; i++)
- {
- if (_ssrcVector[i] == ssrc)
- {
- // we have a match
- i = 0; // start over with a new ssrc
- ssrc = GenerateRandom();
- }
-
- }
- // add to database
- _ssrcVector[_numberOfSSRC] = ssrc;
- _numberOfSSRC++;
- }
-#endif
- return ssrc;
-}
-
-WebRtc_Word32
-SSRCDatabase::RegisterSSRC(const WebRtc_UWord32 ssrc)
-{
- CriticalSectionScoped lock(_critSect);
-
-#ifndef WEBRTC_NO_STL
-
- _ssrcMap[ssrc] = 0;
-
-#else
- if(_sizeOfSSRC <= _numberOfSSRC)
- {
- // allocate more space
- const int newSize = _sizeOfSSRC + 10;
- WebRtc_UWord32* tempSSRCVector = new WebRtc_UWord32[newSize];
- memcpy(tempSSRCVector, _ssrcVector, _sizeOfSSRC*sizeof(WebRtc_UWord32));
- delete [] _ssrcVector;
-
- _ssrcVector = tempSSRCVector;
- _sizeOfSSRC = newSize;
- }
- // check if in DB
- if(_ssrcVector)
- {
- for (int i=0; i<_numberOfSSRC; i++)
- {
- if (_ssrcVector[i] == ssrc)
- {
- // we have a match
- return -1;
- }
- }
- // add to database
- _ssrcVector[_numberOfSSRC] = ssrc;
- _numberOfSSRC++;
- }
-#endif
- return 0;
-}
-
-WebRtc_Word32
-SSRCDatabase::ReturnSSRC(const WebRtc_UWord32 ssrc)
-{
- CriticalSectionScoped lock(_critSect);
-
-#ifndef WEBRTC_NO_STL
- _ssrcMap.erase(ssrc);
-
-#else
- if(_ssrcVector)
- {
- for (int i=0; i<_numberOfSSRC; i++)
- {
- if (_ssrcVector[i] == ssrc)
- {
- // we have a match
- // remove from database
- _ssrcVector[i] = _ssrcVector[_numberOfSSRC-1];
- _numberOfSSRC--;
- break;
- }
- }
- }
-#endif
- return 0;
-}
-
-SSRCDatabase::SSRCDatabase()
-{
- // we need to seed the random generator, otherwise we get 26500 each time, hardly a random value :)
-#ifdef _WIN32
- srand(timeGetTime());
-#else
- struct timeval tv;
- struct timezone tz;
- gettimeofday(&tv, &tz);
- srand(tv.tv_usec);
-#endif
-
-#ifdef WEBRTC_NO_STL
- _sizeOfSSRC = 10;
- _numberOfSSRC = 0;
- _ssrcVector = new WebRtc_UWord32[10];
-#endif
- _critSect = CriticalSectionWrapper::CreateCriticalSection();
-
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, -1, "%s created", __FUNCTION__);
-}
-
-SSRCDatabase::~SSRCDatabase()
-{
-#ifdef WEBRTC_NO_STL
- delete [] _ssrcVector;
-#else
- _ssrcMap.clear();
-#endif
- delete _critSect;
-
- WEBRTC_TRACE(kTraceMemory, kTraceRtpRtcp, -1, "%s deleted", __FUNCTION__);
-}
-
-WebRtc_UWord32 SSRCDatabase::GenerateRandom()
-{
- WebRtc_UWord32 ssrc = 0;
- do
- {
- ssrc = rand();
- ssrc = ssrc <<16;
- ssrc += rand();
-
- } while (ssrc == 0 || ssrc == 0xffffffff);
-
- return ssrc;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/ssrc_database.h b/trunk/src/modules/rtp_rtcp/source/ssrc_database.h
deleted file mode 100644
index 370e549..0000000
--- a/trunk/src/modules/rtp_rtcp/source/ssrc_database.h
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
-
-#ifndef WEBRTC_NO_STL
- #include <map>
-#endif
-
-#include "system_wrappers/interface/static_instance.h"
-#include "typedefs.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class SSRCDatabase
-{
-public:
- static SSRCDatabase* GetSSRCDatabase();
- static void ReturnSSRCDatabase();
-
- WebRtc_UWord32 CreateSSRC();
- WebRtc_Word32 RegisterSSRC(const WebRtc_UWord32 ssrc);
- WebRtc_Word32 ReturnSSRC(const WebRtc_UWord32 ssrc);
-
-protected:
- SSRCDatabase();
- virtual ~SSRCDatabase();
-
- static SSRCDatabase* CreateInstance() { return new SSRCDatabase(); }
-
-private:
- // Friend function to allow the SSRC destructor to be accessed from the
- // template class.
- friend SSRCDatabase* GetStaticInstance<SSRCDatabase>(
- CountOperation count_operation);
- static SSRCDatabase* StaticInstance(CountOperation count_operation);
-
- WebRtc_UWord32 GenerateRandom();
-
-#ifdef WEBRTC_NO_STL
- int _numberOfSSRC;
- int _sizeOfSSRC;
-
- WebRtc_UWord32* _ssrcVector;
-#else
- std::map<WebRtc_UWord32, WebRtc_UWord32> _ssrcMap;
-#endif
-
- CriticalSectionWrapper* _critSect;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_SSRC_DATABASE_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/tmmbr_help.cc b/trunk/src/modules/rtp_rtcp/source/tmmbr_help.cc
deleted file mode 100644
index cf34e08..0000000
--- a/trunk/src/modules/rtp_rtcp/source/tmmbr_help.cc
+++ /dev/null
@@ -1,507 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "tmmbr_help.h"
-
-#include "rtp_rtcp_config.h"
-
-namespace webrtc {
-TMMBRSet::TMMBRSet() :
- ptrTmmbrSet(0),
- ptrPacketOHSet(0),
- ptrSsrcSet(0),
- sizeOfSet(0),
- lengthOfSet(0)
-{
-}
-
-TMMBRSet::~TMMBRSet()
-{
- delete [] ptrTmmbrSet;
- delete [] ptrPacketOHSet;
- delete [] ptrSsrcSet;
- ptrTmmbrSet = 0;
- ptrPacketOHSet = 0;
- ptrSsrcSet = 0;
- sizeOfSet = 0;
- lengthOfSet = 0;
-}
-
-void
-TMMBRSet::VerifyAndAllocateSet(WebRtc_UWord32 minimumSize)
-{
- if(minimumSize > sizeOfSet)
- {
- // make sure that our buffers are big enough
- if(ptrTmmbrSet)
- {
- delete [] ptrTmmbrSet;
- delete [] ptrPacketOHSet;
- delete [] ptrSsrcSet;
- }
- ptrTmmbrSet = new WebRtc_UWord32[minimumSize];
- ptrPacketOHSet = new WebRtc_UWord32[minimumSize];
- ptrSsrcSet = new WebRtc_UWord32[minimumSize];
- sizeOfSet = minimumSize;
- }
- // reset memory
- for(WebRtc_UWord32 i = 0; i < sizeOfSet; i++)
- {
- ptrTmmbrSet[i] = 0;
- ptrPacketOHSet[i] = 0;
- ptrSsrcSet[i] = 0;
- }
- lengthOfSet = 0;
-}
-
-TMMBRHelp::TMMBRHelp(const bool audio) :
- _criticalSection(CriticalSectionWrapper::CreateCriticalSection()),
- _audio(audio),
- _candidateSet(),
- _boundingSet(),
- _boundingSetToSend(),
- _ptrIntersectionBoundingSet(NULL),
- _ptrMaxPRBoundingSet(NULL)
-{
-}
-
-TMMBRHelp::~TMMBRHelp()
-{
- delete [] _ptrIntersectionBoundingSet;
- delete [] _ptrMaxPRBoundingSet;
- _ptrIntersectionBoundingSet = 0;
- _ptrMaxPRBoundingSet = 0;
- delete _criticalSection;
-}
-
-TMMBRSet*
-TMMBRHelp::VerifyAndAllocateBoundingSet(WebRtc_UWord32 minimumSize)
-{
- CriticalSectionScoped lock(_criticalSection);
-
- if(minimumSize > _boundingSet.sizeOfSet)
- {
- // make sure that our buffers are big enough
- if(_ptrIntersectionBoundingSet)
- {
- delete [] _ptrIntersectionBoundingSet;
- delete [] _ptrMaxPRBoundingSet;
- }
- _ptrIntersectionBoundingSet = new float[minimumSize];
- _ptrMaxPRBoundingSet = new float[minimumSize];
- }
- _boundingSet.VerifyAndAllocateSet(minimumSize);
- return &_boundingSet;
-}
-
-TMMBRSet*
-TMMBRHelp::BoundingSet()
-{
- return &_boundingSet;
-}
-
-WebRtc_Word32
-TMMBRHelp::SetTMMBRBoundingSetToSend(const TMMBRSet* boundingSetToSend,
- const WebRtc_UWord32 maxBitrateKbit)
-{
- CriticalSectionScoped lock(_criticalSection);
-
- if (boundingSetToSend == NULL)
- {
- _boundingSetToSend.lengthOfSet = 0;
- return 0;
- }
-
- VerifyAndAllocateBoundingSetToSend(boundingSetToSend->lengthOfSet);
-
- for (WebRtc_UWord32 i = 0; i < boundingSetToSend->lengthOfSet; i++)
- {
- // cap at our configured max bitrate
- WebRtc_UWord32 bitrate = boundingSetToSend->ptrTmmbrSet[i];
- if(maxBitrateKbit)
- {
- // do we have a configured max bitrate?
- if(bitrate > maxBitrateKbit)
- {
- bitrate = maxBitrateKbit;
- }
- }
-
- _boundingSetToSend.ptrTmmbrSet[i] = bitrate;
- _boundingSetToSend.ptrPacketOHSet[i] = boundingSetToSend->ptrPacketOHSet[i];
- _boundingSetToSend.ptrSsrcSet[i] = boundingSetToSend->ptrSsrcSet[i];
- }
- _boundingSetToSend.lengthOfSet = boundingSetToSend->lengthOfSet;
- return 0;
-}
-
-WebRtc_Word32
-TMMBRHelp::VerifyAndAllocateBoundingSetToSend(WebRtc_UWord32 minimumSize)
-{
- CriticalSectionScoped lock(_criticalSection);
-
- _boundingSetToSend.VerifyAndAllocateSet(minimumSize);
- return 0;
-}
-
-TMMBRSet*
-TMMBRHelp::VerifyAndAllocateCandidateSet(WebRtc_UWord32 minimumSize)
-{
- CriticalSectionScoped lock(_criticalSection);
-
- _candidateSet.VerifyAndAllocateSet(minimumSize);
- return &_candidateSet;
-}
-
-TMMBRSet*
-TMMBRHelp::CandidateSet()
-{
- return &_candidateSet;
-}
-
-TMMBRSet*
-TMMBRHelp::BoundingSetToSend()
-{
- return &_boundingSetToSend;
-}
-
-WebRtc_Word32
-TMMBRHelp::FindTMMBRBoundingSet(TMMBRSet*& boundingSet)
-{
- CriticalSectionScoped lock(_criticalSection);
-
- // Work on local variable, will be modified
- TMMBRSet candidateSet;
- candidateSet.VerifyAndAllocateSet(_candidateSet.sizeOfSet);
-
- // Number of set candidates
- WebRtc_Word32 numSetCandidates = 0;
- for (WebRtc_UWord32 i = 0; i < _candidateSet.sizeOfSet; i++)
- {
- if(_candidateSet.ptrTmmbrSet[i])
- {
- numSetCandidates++;
- candidateSet.ptrTmmbrSet[i] = _candidateSet.ptrTmmbrSet[i];
- candidateSet.ptrPacketOHSet[i] = _candidateSet.ptrPacketOHSet[i];
- candidateSet.ptrSsrcSet[i] = _candidateSet.ptrSsrcSet[i];
- }
- else
- {
- // make sure this is zero if tmmbr = 0
- _candidateSet.ptrPacketOHSet[i] = 0;
- }
- }
- candidateSet.lengthOfSet = numSetCandidates;
-
- // Find bounding set
- WebRtc_UWord32 numBoundingSet = 0;
- if (numSetCandidates > 0)
- {
- numBoundingSet = FindTMMBRBoundingSet(numSetCandidates, candidateSet);
- if(numBoundingSet < 1 || (numBoundingSet > _candidateSet.sizeOfSet))
- {
- return -1;
- }
- boundingSet = &_boundingSet;
- }
- return numBoundingSet;
-}
-
-
-WebRtc_Word32
-TMMBRHelp::FindTMMBRBoundingSet(WebRtc_Word32 numCandidates, TMMBRSet& candidateSet)
-{
- CriticalSectionScoped lock(_criticalSection);
-
- WebRtc_UWord32 numBoundingSet = 0;
- VerifyAndAllocateBoundingSet(candidateSet.sizeOfSet);
-
- if (numCandidates == 1)
- {
- for (WebRtc_UWord32 i = 0; i < candidateSet.sizeOfSet; i++)
- {
- if(candidateSet.ptrTmmbrSet[i] > 0)
- {
- _boundingSet.ptrTmmbrSet[numBoundingSet] = candidateSet.ptrTmmbrSet[i];
- _boundingSet.ptrPacketOHSet[numBoundingSet] = candidateSet.ptrPacketOHSet[i];
- _boundingSet.ptrSsrcSet[numBoundingSet] = candidateSet.ptrSsrcSet[i];
- numBoundingSet++;
- }
- }
- if (numBoundingSet != 1)
- {
- numBoundingSet = -1;
- }
- } else
- {
- // 1. Sort by increasing packetOH
- WebRtc_UWord32 temp;
- for (int i = candidateSet.sizeOfSet - 1; i >= 0; i--)
- {
- for (int j = 1; j <= i; j++)
- {
- if (candidateSet.ptrPacketOHSet[j-1] > candidateSet.ptrPacketOHSet[j])
- {
- temp = candidateSet.ptrPacketOHSet[j-1];
- candidateSet.ptrPacketOHSet[j-1] = candidateSet.ptrPacketOHSet[j];
- candidateSet.ptrPacketOHSet[j] = temp;
- temp = candidateSet.ptrTmmbrSet[j-1];
- candidateSet.ptrTmmbrSet[j-1] = candidateSet.ptrTmmbrSet[j];
- candidateSet.ptrTmmbrSet[j] = temp;
- temp = candidateSet.ptrSsrcSet[j-1];
- candidateSet.ptrSsrcSet[j-1] = candidateSet.ptrSsrcSet[j];
- candidateSet.ptrSsrcSet[j] = temp;
- }
- }
- }
- // 2. For tuples with same OH, keep the one w/ the lowest bitrate
- for (WebRtc_UWord32 i = 0; i < candidateSet.sizeOfSet; i++)
- {
- if (candidateSet.ptrTmmbrSet[i] > 0)
- {
- // get min bitrate for packets w/ same OH
- WebRtc_UWord32 currentPacketOH = candidateSet.ptrPacketOHSet[i];
- WebRtc_UWord32 currentMinTMMBR = candidateSet.ptrTmmbrSet[i];
- WebRtc_UWord32 currentMinIndexTMMBR = i;
- for (WebRtc_UWord32 j = i+1; j < candidateSet.sizeOfSet; j++)
- {
- if(candidateSet.ptrPacketOHSet[j] == currentPacketOH)
- {
- if(candidateSet.ptrTmmbrSet[j] < currentMinTMMBR)
- {
- currentMinTMMBR = candidateSet.ptrTmmbrSet[j];
- currentMinIndexTMMBR = j;
- }
- }
- }
- // keep lowest bitrate
- for (WebRtc_UWord32 j = 0; j < candidateSet.sizeOfSet; j++)
- {
- if(candidateSet.ptrPacketOHSet[j] == currentPacketOH && j != currentMinIndexTMMBR)
- {
- candidateSet.ptrTmmbrSet[j] = 0;
- candidateSet.ptrPacketOHSet[j] = 0;
- candidateSet.ptrSsrcSet[j] = 0;
- numCandidates--;
- }
- }
- }
- }
- // 3. Select and remove tuple w/ lowest tmmbr. (If more than 1, choose the one w/ highest OH).
- WebRtc_UWord32 minTMMBR = 0;
- WebRtc_UWord32 minIndexTMMBR = 0;
- for (WebRtc_UWord32 i = 0; i < candidateSet.sizeOfSet; i++)
- {
- if (candidateSet.ptrTmmbrSet[i] > 0)
- {
- minTMMBR = candidateSet.ptrTmmbrSet[i];
- minIndexTMMBR = i;
- break;
- }
- }
-
- for (WebRtc_UWord32 i = 0; i < candidateSet.sizeOfSet; i++)
- {
- if (candidateSet.ptrTmmbrSet[i] > 0 && candidateSet.ptrTmmbrSet[i] <= minTMMBR)
- {
- // get min bitrate
- minTMMBR = candidateSet.ptrTmmbrSet[i];
- minIndexTMMBR = i;
- }
- }
- // first member of selected list
- _boundingSet.ptrTmmbrSet[numBoundingSet] = candidateSet.ptrTmmbrSet[minIndexTMMBR];
- _boundingSet.ptrPacketOHSet[numBoundingSet] = candidateSet.ptrPacketOHSet[minIndexTMMBR];
- _boundingSet.ptrSsrcSet[numBoundingSet] = candidateSet.ptrSsrcSet[minIndexTMMBR];
- // set intersection value
- _ptrIntersectionBoundingSet[numBoundingSet] = 0;
- // calculate its maximum packet rate (where its line crosses x-axis)
- _ptrMaxPRBoundingSet[numBoundingSet] = _boundingSet.ptrTmmbrSet[numBoundingSet]*1000 / float(8*_boundingSet.ptrPacketOHSet[numBoundingSet]);
- numBoundingSet++;
- // remove from candidate list
- candidateSet.ptrTmmbrSet[minIndexTMMBR] = 0;
- candidateSet.ptrPacketOHSet[minIndexTMMBR] = 0;
- candidateSet.ptrSsrcSet[minIndexTMMBR] = 0;
- numCandidates--;
-
- // 4. Discard from candidate list all tuple w/ lower OH (next tuple must be steeper)
- for (WebRtc_UWord32 i = 0; i < candidateSet.sizeOfSet; i++)
- {
- if(candidateSet.ptrTmmbrSet[i] > 0 && candidateSet.ptrPacketOHSet[i] < _boundingSet.ptrPacketOHSet[0])
- {
- candidateSet.ptrTmmbrSet[i] = 0;
- candidateSet.ptrPacketOHSet[i] = 0;
- candidateSet.ptrSsrcSet[i] = 0;
- numCandidates--;
- }
- }
-
- if (numCandidates == 0)
- {
- _boundingSet.lengthOfSet = numBoundingSet;
- return numBoundingSet;
- }
-
- bool getNewCandidate = true;
- int curCandidateTMMBR = 0;
- int curCandidateIndex = 0;
- int curCandidatePacketOH = 0;
- int curCandidateSSRC = 0;
- do
- {
- if (getNewCandidate)
- {
- // 5. Remove first remaining tuple from candidate list
- for (WebRtc_UWord32 i = 0; i < candidateSet.sizeOfSet; i++)
- {
- if (candidateSet.ptrTmmbrSet[i] > 0)
- {
- curCandidateTMMBR = candidateSet.ptrTmmbrSet[i];
- curCandidatePacketOH = candidateSet.ptrPacketOHSet[i];
- curCandidateSSRC = candidateSet.ptrSsrcSet[i];
- curCandidateIndex = i;
- candidateSet.ptrTmmbrSet[curCandidateIndex] = 0;
- candidateSet.ptrPacketOHSet[curCandidateIndex] = 0;
- candidateSet.ptrSsrcSet[curCandidateIndex] = 0;
- break;
- }
- }
- }
-
- // 6. Calculate packet rate and intersection of the current line with line of last tuple in selected list
- float packetRate = float(curCandidateTMMBR - _boundingSet.ptrTmmbrSet[numBoundingSet-1])*1000 / (8*(curCandidatePacketOH - _boundingSet.ptrPacketOHSet[numBoundingSet-1]));
-
- // 7. If the packet rate is equal or lower than intersection of last tuple in selected list,
- // remove last tuple in selected list & go back to step 6
- if(packetRate <= _ptrIntersectionBoundingSet[numBoundingSet-1])
- {
- // remove last tuple and goto step 6
- numBoundingSet--;
- _boundingSet.ptrTmmbrSet[numBoundingSet] = 0;
- _boundingSet.ptrPacketOHSet[numBoundingSet] = 0;
- _boundingSet.ptrSsrcSet[numBoundingSet] = 0;
- _ptrIntersectionBoundingSet[numBoundingSet] = 0;
- _ptrMaxPRBoundingSet[numBoundingSet] = 0;
- getNewCandidate = false;
- } else
- {
- // 8. If packet rate is lower than maximum packet rate of last tuple in selected list, add current tuple to selected list
- if (packetRate < _ptrMaxPRBoundingSet[numBoundingSet-1])
- {
- _boundingSet.ptrTmmbrSet[numBoundingSet] = curCandidateTMMBR;
- _boundingSet.ptrPacketOHSet[numBoundingSet] = curCandidatePacketOH;
- _boundingSet.ptrSsrcSet[numBoundingSet] = curCandidateSSRC;
- _ptrIntersectionBoundingSet[numBoundingSet] = packetRate;
- _ptrMaxPRBoundingSet[numBoundingSet] = _boundingSet.ptrTmmbrSet[numBoundingSet]*1000 / float(8*_boundingSet.ptrPacketOHSet[numBoundingSet]);
- numBoundingSet++;
- }
- numCandidates--;
- getNewCandidate = true;
- }
-
- // 9. Go back to step 5 if any tuple remains in candidate list
- } while (numCandidates > 0);
- }
- _boundingSet.lengthOfSet = numBoundingSet;
- return numBoundingSet;
-}
-
-bool
-TMMBRHelp::IsOwner(const WebRtc_UWord32 ssrc,
- const WebRtc_UWord32 length) const
-{
- CriticalSectionScoped lock(_criticalSection);
-
- if (length == 0)
- {
- // empty bounding set
- return false;
- }
-
- for(WebRtc_UWord32 i = 0; (i < length) && (i < _boundingSet.sizeOfSet); ++i)
- {
- if(_boundingSet.ptrSsrcSet[i] == ssrc)
- {
- return true;
- }
- }
- return false;
-}
-
-WebRtc_Word32
-TMMBRHelp::CalcMinMaxBitRate(const WebRtc_UWord32 totalPacketRate,
- const WebRtc_UWord32 lengthOfBoundingSet,
- WebRtc_UWord32& minBitrateKbit,
- WebRtc_UWord32& maxBitrateKbit) const
-{
- CriticalSectionScoped lock(_criticalSection);
-
- if (lengthOfBoundingSet <= 0 || _candidateSet.sizeOfSet == 0)
- {
- // empty bounding set
- return -1;
- }
-
- minBitrateKbit = 0xFFFFFFFF;
- maxBitrateKbit = 0;
-
- for (WebRtc_UWord32 i = 0; i < _candidateSet.sizeOfSet; ++i)
- {
- if (_candidateSet.ptrTmmbrSet[i])
- {
- WebRtc_Word32 curNetBitRate = static_cast<WebRtc_Word32>((_candidateSet.ptrTmmbrSet[i]*1000.0
- - (totalPacketRate * (_candidateSet.ptrPacketOHSet[i] << 3)))/1000 + 0.5);
-
- if (curNetBitRate < 0)
- {
- // could be negative for a large packet rate
- if(_audio)
- {
- curNetBitRate = MIN_AUDIO_BW_MANAGEMENT_BITRATE;
- }else
- {
- curNetBitRate = MIN_VIDEO_BW_MANAGEMENT_BITRATE;
- }
- }
- minBitrateKbit = (WebRtc_UWord32(curNetBitRate) < minBitrateKbit) ? curNetBitRate : minBitrateKbit;
- }
- }
- maxBitrateKbit = minBitrateKbit;
-
- if (maxBitrateKbit == 0 || maxBitrateKbit < minBitrateKbit)
- {
- return -1;
- }
-
- if(_audio)
- {
- if (minBitrateKbit < MIN_AUDIO_BW_MANAGEMENT_BITRATE)
- {
- minBitrateKbit = MIN_AUDIO_BW_MANAGEMENT_BITRATE;
- }
- if (maxBitrateKbit < MIN_AUDIO_BW_MANAGEMENT_BITRATE)
- {
- maxBitrateKbit = MIN_AUDIO_BW_MANAGEMENT_BITRATE;
- }
- }else
- {
- if (minBitrateKbit < MIN_VIDEO_BW_MANAGEMENT_BITRATE)
- {
- minBitrateKbit = MIN_VIDEO_BW_MANAGEMENT_BITRATE;
- }
- if (maxBitrateKbit < MIN_VIDEO_BW_MANAGEMENT_BITRATE)
- {
- maxBitrateKbit = MIN_VIDEO_BW_MANAGEMENT_BITRATE;
- }
- }
-
- return 0;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/tmmbr_help.h b/trunk/src/modules/rtp_rtcp/source/tmmbr_help.h
deleted file mode 100644
index 35704fe..0000000
--- a/trunk/src/modules/rtp_rtcp/source/tmmbr_help.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_TMMBR_HELP_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_TMMBR_HELP_H_
-
-#include "typedefs.h"
-
-#include "critical_section_wrapper.h"
-
-#ifndef NULL
- #define NULL 0
-#endif
-
-namespace webrtc {
-class TMMBRSet
-{
-public:
- TMMBRSet();
- ~TMMBRSet();
-
- void VerifyAndAllocateSet(WebRtc_UWord32 minimumSize);
-
- WebRtc_UWord32* ptrTmmbrSet;
- WebRtc_UWord32* ptrPacketOHSet;
- WebRtc_UWord32* ptrSsrcSet;
- WebRtc_UWord32 sizeOfSet;
- WebRtc_UWord32 lengthOfSet;
-};
-
-class TMMBRHelp
-{
-public:
- TMMBRHelp(const bool audio);
- virtual ~TMMBRHelp();
-
- TMMBRSet* BoundingSet(); // used for debuging
- TMMBRSet* CandidateSet();
- TMMBRSet* BoundingSetToSend();
-
- TMMBRSet* VerifyAndAllocateCandidateSet(const WebRtc_UWord32 minimumSize);
- WebRtc_Word32 FindTMMBRBoundingSet(TMMBRSet*& boundingSet);
- WebRtc_Word32 SetTMMBRBoundingSetToSend(const TMMBRSet* boundingSetToSend,
- const WebRtc_UWord32 maxBitrateKbit);
-
- bool IsOwner(const WebRtc_UWord32 ssrc,
- const WebRtc_UWord32 length) const;
-
- WebRtc_Word32 CalcMinMaxBitRate(const WebRtc_UWord32 totalPacketRate,
- const WebRtc_UWord32 lengthOfBoundingSet,
- WebRtc_UWord32& minBitrateKbit,
- WebRtc_UWord32& maxBitrateKbit) const;
-
-protected:
- TMMBRSet* VerifyAndAllocateBoundingSet(WebRtc_UWord32 minimumSize);
- WebRtc_Word32 VerifyAndAllocateBoundingSetToSend(WebRtc_UWord32 minimumSize);
-
- WebRtc_Word32 FindTMMBRBoundingSet(WebRtc_Word32 numCandidates, TMMBRSet& candidateSet);
-
-private:
- CriticalSectionWrapper* _criticalSection;
- const bool _audio;
- TMMBRSet _candidateSet;
- TMMBRSet _boundingSet;
- TMMBRSet _boundingSetToSend;
-
- float* _ptrIntersectionBoundingSet;
- float* _ptrMaxPRBoundingSet;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_TMMBR_HELP_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/transmission_bucket.cc b/trunk/src/modules/rtp_rtcp/source/transmission_bucket.cc
deleted file mode 100644
index e79d227..0000000
--- a/trunk/src/modules/rtp_rtcp/source/transmission_bucket.cc
+++ /dev/null
@@ -1,117 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "transmission_bucket.h"
-
-#include <assert.h>
-#include "critical_section_wrapper.h"
-
-namespace webrtc {
-
-TransmissionBucket::TransmissionBucket()
- : critsect_(CriticalSectionWrapper::CreateCriticalSection()),
- accumulator_(0),
- bytes_rem_total_(0),
- bytes_rem_interval_(0),
- packets_(),
- first_(true) {
-}
-
-TransmissionBucket::~TransmissionBucket() {
- packets_.clear();
- delete critsect_;
-}
-
-void TransmissionBucket::Reset() {
- webrtc::CriticalSectionScoped cs(*critsect_);
- accumulator_ = 0;
- bytes_rem_total_ = 0;
- bytes_rem_interval_ = 0;
- packets_.clear();
- first_ = true;
-}
-
-void TransmissionBucket::Fill(const uint16_t seq_num,
- const uint32_t num_bytes) {
- webrtc::CriticalSectionScoped cs(*critsect_);
- accumulator_ += num_bytes;
-
- Packet p(seq_num, num_bytes);
- packets_.push_back(p);
-}
-
-bool TransmissionBucket::Empty() {
- webrtc::CriticalSectionScoped cs(*critsect_);
- return packets_.empty();
-}
-
-void TransmissionBucket::UpdateBytesPerInterval(
- const uint32_t delta_time_ms,
- const uint16_t target_bitrate_kbps) {
- webrtc::CriticalSectionScoped cs(*critsect_);
-
- const float kMargin = 1.05f;
- uint32_t bytes_per_interval =
- kMargin * (target_bitrate_kbps * delta_time_ms / 8);
-
- if (bytes_rem_interval_ < 0) {
- bytes_rem_interval_ += bytes_per_interval;
- } else {
- bytes_rem_interval_ = bytes_per_interval;
- }
-
- if (accumulator_) {
- bytes_rem_total_ += bytes_per_interval;
- return;
- }
- bytes_rem_total_ = bytes_per_interval;
-}
-
-int32_t TransmissionBucket::GetNextPacket() {
- webrtc::CriticalSectionScoped cs(*critsect_);
-
- if (accumulator_ == 0) {
- // Empty.
- return -1;
- }
-
- std::vector<Packet>::const_iterator it_begin = packets_.begin();
- const uint16_t num_bytes = (*it_begin).length_;
- const uint16_t seq_num = (*it_begin).sequence_number_;
-
- if (first_) {
- // Ok to transmit first packet.
- first_ = false;
- packets_.erase(packets_.begin());
- return seq_num;
- }
-
- const float kFrameComplete = 0.80f;
- if (num_bytes * kFrameComplete > bytes_rem_total_) {
- // Packet does not fit.
- return -1;
- }
-
- if (bytes_rem_interval_ <= 0) {
- // All bytes consumed for this interval.
- return -1;
- }
-
- // Ok to transmit packet.
- bytes_rem_total_ -= num_bytes;
- bytes_rem_interval_ -= num_bytes;
-
- assert(accumulator_ >= num_bytes);
- accumulator_ -= num_bytes;
-
- packets_.erase(packets_.begin());
- return seq_num;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/transmission_bucket.h b/trunk/src/modules/rtp_rtcp/source/transmission_bucket.h
deleted file mode 100644
index 79e45d8..0000000
--- a/trunk/src/modules/rtp_rtcp/source/transmission_bucket.h
+++ /dev/null
@@ -1,63 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_TRANSMISSION_BUCKET_H_
-#define WEBRTC_MODULES_RTP_RTCP_TRANSMISSION_BUCKET_H_
-
-#include <vector>
-
-#include "typedefs.h"
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-
-class TransmissionBucket {
- public:
- TransmissionBucket();
- ~TransmissionBucket();
-
- // Resets members to initial state.
- void Reset();
-
- // Adds packet to be sent.
- void Fill(const uint16_t seq_num, const uint32_t num_bytes);
-
- // Returns true if there is no packet to be sent.
- bool Empty();
-
- // Updates the number of bytes that can be sent for the next time interval.
- void UpdateBytesPerInterval(const uint32_t delta_time_in_ms,
- const uint16_t target_bitrate_kbps);
-
- // Checks if next packet in line can be transmitted. Returns the sequence
- // number of the packet on success, -1 otherwise. The packet is removed from
- // the vector on success.
- int32_t GetNextPacket();
-
- private:
- struct Packet {
- Packet(uint16_t sequence_number, uint16_t length_in_bytes)
- : sequence_number_(sequence_number),
- length_(length_in_bytes) {
- }
- uint16_t sequence_number_;
- uint16_t length_;
- };
-
- CriticalSectionWrapper* critsect_;
- uint32_t accumulator_;
- int32_t bytes_rem_total_;
- int32_t bytes_rem_interval_;
- std::vector<Packet> packets_;
- bool first_;
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_RTP_RTCP_TRANSMISSION_BUCKET_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/transmission_bucket_test.cc b/trunk/src/modules/rtp_rtcp/source/transmission_bucket_test.cc
deleted file mode 100644
index a8c9247..0000000
--- a/trunk/src/modules/rtp_rtcp/source/transmission_bucket_test.cc
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file includes unit tests for the TransmissionBucket.
- */
-
-#include <gtest/gtest.h>
-
-#include "transmission_bucket.h"
-
-namespace webrtc {
-
-class TransmissionBucketTest : public ::testing::Test {
- protected:
- TransmissionBucket send_bucket_;
-};
-
-TEST_F(TransmissionBucketTest, Fill) {
- EXPECT_TRUE(send_bucket_.Empty());
- send_bucket_.Fill(1, 100);
- EXPECT_FALSE(send_bucket_.Empty());
-}
-
-TEST_F(TransmissionBucketTest, Reset) {
- send_bucket_.Fill(1, 100);
- EXPECT_FALSE(send_bucket_.Empty());
- send_bucket_.Reset();
- EXPECT_TRUE(send_bucket_.Empty());
-}
-
-TEST_F(TransmissionBucketTest, GetNextPacket) {
- EXPECT_EQ(-1, send_bucket_.GetNextPacket()); // empty
- send_bucket_.Fill(1234, 100);
- EXPECT_EQ(1234, send_bucket_.GetNextPacket()); // first packet ok
- send_bucket_.Fill(1235, 100);
- EXPECT_EQ(-1, send_bucket_.GetNextPacket()); // packet does not fit
-}
-
-TEST_F(TransmissionBucketTest, UpdateBytesPerInterval) {
- const int delta_time_ms = 1;
- const int target_bitrate_kbps = 800;
- send_bucket_.UpdateBytesPerInterval(delta_time_ms, target_bitrate_kbps);
-
- send_bucket_.Fill(1234, 50);
- send_bucket_.Fill(1235, 50);
- send_bucket_.Fill(1236, 50);
-
- EXPECT_EQ(1234, send_bucket_.GetNextPacket()); // first packet ok
- EXPECT_EQ(1235, send_bucket_.GetNextPacket()); // ok
- EXPECT_EQ(1236, send_bucket_.GetNextPacket()); // ok
- EXPECT_TRUE(send_bucket_.Empty());
-
- send_bucket_.Fill(1237, 50);
- EXPECT_EQ(-1, send_bucket_.GetNextPacket()); // packet does not fit
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/rtp_rtcp/source/video_codec_information.h b/trunk/src/modules/rtp_rtcp/source/video_codec_information.h
deleted file mode 100644
index 4364f0b..0000000
--- a/trunk/src/modules/rtp_rtcp/source/video_codec_information.h
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_
-
-#include "rtp_rtcp_config.h"
-#include "rtp_utility.h"
-
-namespace webrtc {
-class VideoCodecInformation
-{
-public:
- virtual void Reset() = 0;
-
- virtual RtpVideoCodecTypes Type() = 0;
- virtual ~VideoCodecInformation(){};
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_VIDEO_CODEC_INFORMATION_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator.cc b/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator.cc
deleted file mode 100644
index 6d5bb75..0000000
--- a/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator.cc
+++ /dev/null
@@ -1,268 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/rtp_rtcp/source/vp8_partition_aggregator.h"
-
-#include <assert.h>
-#include <stdlib.h> // NULL
-
-#include <algorithm>
-#include <limits>
-
-namespace webrtc {
-
-PartitionTreeNode::PartitionTreeNode(PartitionTreeNode* parent,
- const int* size_vector,
- int num_partitions,
- int this_size)
- : parent_(parent),
- this_size_(this_size),
- size_vector_(size_vector),
- num_partitions_(num_partitions),
- max_parent_size_(0),
- min_parent_size_(std::numeric_limits<int>::max()),
- packet_start_(false) {
- assert(num_partitions >= 0);
- children_[kLeftChild] = NULL;
- children_[kRightChild] = NULL;
-}
-
-PartitionTreeNode* PartitionTreeNode::CreateRootNode(const int* size_vector,
- int num_partitions) {
- PartitionTreeNode* root_node =
- new PartitionTreeNode(NULL, &size_vector[1], num_partitions - 1,
- size_vector[0]);
- root_node->set_packet_start(true);
- return root_node;
-}
-
-PartitionTreeNode::~PartitionTreeNode() {
- delete children_[kLeftChild];
- delete children_[kRightChild];
-}
-
-int PartitionTreeNode::Cost(int penalty) {
- assert(penalty >= 0);
- int cost = 0;
- if (num_partitions_ == 0) {
- // This is a solution node.
- cost = std::max(max_parent_size_, this_size_) -
- std::min(min_parent_size_, this_size_);
- } else {
- cost = std::max(max_parent_size_, this_size_) - min_parent_size_;
- }
- return cost + NumPackets() * penalty;
-}
-
-bool PartitionTreeNode::CreateChildren(int max_size) {
- assert(max_size > 0);
- bool children_created = false;
- if (num_partitions_ > 0) {
- if (this_size_ + size_vector_[0] <= max_size) {
- assert(!children_[kLeftChild]);
- children_[kLeftChild] =
- new PartitionTreeNode(this,
- &size_vector_[1],
- num_partitions_ - 1,
- this_size_ + size_vector_[0]);
- children_[kLeftChild]->set_max_parent_size(max_parent_size_);
- children_[kLeftChild]->set_min_parent_size(min_parent_size_);
- // "Left" child is continuation of same packet.
- children_[kLeftChild]->set_packet_start(false);
- children_created = true;
- }
- if (this_size_ > 0) {
- assert(!children_[kRightChild]);
- children_[kRightChild] = new PartitionTreeNode(this,
- &size_vector_[1],
- num_partitions_ - 1,
- size_vector_[0]);
- children_[kRightChild]->set_max_parent_size(
- std::max(max_parent_size_, this_size_));
- children_[kRightChild]->set_min_parent_size(
- std::min(min_parent_size_, this_size_));
- // "Right" child starts a new packet.
- children_[kRightChild]->set_packet_start(true);
- children_created = true;
- }
- }
- return children_created;
-}
-
-int PartitionTreeNode::NumPackets() {
- if (parent_ == NULL) {
- // Root node is a "right" child by definition.
- return 1;
- }
- if (parent_->children_[kLeftChild] == this) {
- // This is a "left" child.
- return parent_->NumPackets();
- } else {
- // This is a "right" child.
- return 1 + parent_->NumPackets();
- }
-}
-
-PartitionTreeNode* PartitionTreeNode::GetOptimalNode(int max_size,
- int penalty) {
- CreateChildren(max_size);
- PartitionTreeNode* left = children_[kLeftChild];
- PartitionTreeNode* right = children_[kRightChild];
- if ((left == NULL) && (right == NULL)) {
- // This is a solution node; return it.
- return this;
- } else if (left == NULL) {
- // One child empty, return the other.
- return right->GetOptimalNode(max_size, penalty);
- } else if (right == NULL) {
- // One child empty, return the other.
- return left->GetOptimalNode(max_size, penalty);
- } else {
- PartitionTreeNode* first;
- PartitionTreeNode* second;
- if (left->Cost(penalty) <= right->Cost(penalty)) {
- first = left;
- second = right;
- } else {
- first = right;
- second = left;
- }
- first = first->GetOptimalNode(max_size, penalty);
- if (second->Cost(penalty) <= first->Cost(penalty)) {
- second = second->GetOptimalNode(max_size, penalty);
- // Compare cost estimate for "second" with actual cost for "first".
- if (second->Cost(penalty) < first->Cost(penalty)) {
- return second;
- }
- }
- return first;
- }
-}
-
-Vp8PartitionAggregator::Vp8PartitionAggregator(
- const RTPFragmentationHeader& fragmentation,
- int first_partition_idx, int last_partition_idx)
- : root_(NULL),
- num_partitions_(last_partition_idx - first_partition_idx + 1),
- size_vector_(new int[num_partitions_]),
- largest_partition_size_(0) {
- assert(first_partition_idx >= 0);
- assert(last_partition_idx >= first_partition_idx);
- assert(last_partition_idx < fragmentation.fragmentationVectorSize);
- for (size_t i = 0; i < num_partitions_; ++i) {
- size_vector_[i] =
- fragmentation.fragmentationLength[i + first_partition_idx];
- largest_partition_size_ = std::max(largest_partition_size_,
- size_vector_[i]);
- }
- root_ = PartitionTreeNode::CreateRootNode(size_vector_, num_partitions_);
-}
-
-Vp8PartitionAggregator::~Vp8PartitionAggregator() {
- delete [] size_vector_;
- delete root_;
-}
-
-void Vp8PartitionAggregator::SetPriorMinMax(int min_size, int max_size) {
- assert(root_);
- assert(min_size >= 0);
- assert(max_size >= min_size);
- root_->set_min_parent_size(min_size);
- root_->set_max_parent_size(max_size);
-}
-
-Vp8PartitionAggregator::ConfigVec
-Vp8PartitionAggregator::FindOptimalConfiguration(int max_size, int penalty) {
- assert(root_);
- assert(max_size >= largest_partition_size_);
- PartitionTreeNode* opt = root_->GetOptimalNode(max_size, penalty);
- ConfigVec config_vector(num_partitions_, 0);
- PartitionTreeNode* temp_node = opt;
- int packet_index = opt->NumPackets() - 1;
- for (int i = num_partitions_ - 1; i >= 0; --i) {
- assert(packet_index >= 0);
- assert(temp_node != NULL);
- config_vector[i] = packet_index;
- if (temp_node->packet_start()) --packet_index;
- temp_node = temp_node->parent();
- }
- return config_vector;
-}
-
-void Vp8PartitionAggregator::CalcMinMax(const ConfigVec& config,
- int* min_size, int* max_size) const {
- if (*min_size < 0) {
- *min_size = std::numeric_limits<int>::max();
- }
- if (*max_size < 0) {
- *max_size = 0;
- }
- unsigned int i = 0;
- while (i < config.size()) {
- int this_size = 0;
- unsigned int j = i;
- while (j < config.size() && config[i] == config[j]) {
- this_size += size_vector_[j];
- ++j;
- }
- i = j;
- if (this_size < *min_size) {
- *min_size = this_size;
- }
- if (this_size > *max_size) {
- *max_size = this_size;
- }
- }
-}
-
-int Vp8PartitionAggregator::CalcNumberOfFragments(int large_partition_size,
- int max_payload_size,
- int penalty,
- int min_size,
- int max_size) {
- assert(max_size <= max_payload_size);
- assert(min_size <= max_size);
- assert(max_payload_size > 0);
- // Divisions with rounding up.
- const int min_number_of_fragments =
- (large_partition_size + max_payload_size - 1) / max_payload_size;
- if (min_size < 0 || max_size < 0) {
- // No aggregates produced, so we do not have any size boundaries.
- // Simply split in as few partitions as possible.
- return min_number_of_fragments;
- }
- const int max_number_of_fragments =
- (large_partition_size + min_size - 1) / min_size;
- int num_fragments = -1;
- int best_cost = std::numeric_limits<int>::max();
- for (int n = min_number_of_fragments; n <= max_number_of_fragments; ++n) {
- // Round up so that we use the largest fragment.
- int fragment_size = (large_partition_size + n - 1) / n;
- int cost = 0;
- if (fragment_size < min_size) {
- cost = min_size - fragment_size + n * penalty;
- } else if (fragment_size > max_size) {
- cost = fragment_size - max_size + n * penalty;
- } else {
- cost = n * penalty;
- }
- if (fragment_size <= max_payload_size && cost < best_cost) {
- num_fragments = n;
- best_cost = cost;
- }
- }
- assert(num_fragments > 0);
- // TODO(mflodman) Assert disabled since it's falsely triggered, see issue 293.
- //assert(large_partition_size / num_fragments + 1 <= max_payload_size);
- return num_fragments;
-}
-
-} // namespace
-
diff --git a/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator.h b/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator.h
deleted file mode 100644
index c5d47de..0000000
--- a/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator.h
+++ /dev/null
@@ -1,135 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_SOURCE_VP8_PARTITION_AGGREGATOR_H_
-#define WEBRTC_MODULES_RTP_RTCP_SOURCE_VP8_PARTITION_AGGREGATOR_H_
-
-#include <vector>
-
-#include "modules/interface/module_common_types.h"
-#include "system_wrappers/interface/constructor_magic.h"
-#include "typedefs.h" // NOLINT(build/include)
-
-namespace webrtc {
-
-// Class used to solve the VP8 aggregation problem.
-class PartitionTreeNode {
- public:
- // Create a tree node.
- PartitionTreeNode(PartitionTreeNode* parent,
- const int* size_vector,
- int num_partitions,
- int this_size);
-
- // Create a root node.
- static PartitionTreeNode* CreateRootNode(const int* size_vector,
- int num_partitions);
-
- ~PartitionTreeNode();
-
- // Calculate the cost for the node. If the node is a solution node, the cost
- // will be the actual cost associated with that solution. If not, the cost
- // will be the cost accumulated so far along the current branch (which is a
- // lower bound for any solution along the branch).
- int Cost(int penalty);
-
- // Create the two children for this node.
- bool CreateChildren(int max_size);
-
- // Get the number of packets for the configuration that this node represents.
- int NumPackets();
-
- // Find the optimal solution given a maximum packet size and a per-packet
- // penalty. The method will be recursively called while the solver is
- // probing down the tree of nodes.
- PartitionTreeNode* GetOptimalNode(int max_size, int penalty);
-
- // Setters and getters.
- void set_max_parent_size(int size) { max_parent_size_ = size; }
- void set_min_parent_size(int size) { min_parent_size_ = size; }
- PartitionTreeNode* parent() const { return parent_; }
- PartitionTreeNode* left_child() const { return children_[kLeftChild]; }
- PartitionTreeNode* right_child() const { return children_[kRightChild]; }
- int this_size() const { return this_size_; }
- bool packet_start() const { return packet_start_; }
-
- private:
- enum Children {
- kLeftChild = 0,
- kRightChild = 1
- };
-
- void set_packet_start(bool value) { packet_start_ = value; }
-
- PartitionTreeNode* parent_;
- PartitionTreeNode* children_[2];
- int this_size_;
- const int* size_vector_;
- int num_partitions_;
- int max_parent_size_;
- int min_parent_size_;
- bool packet_start_;
-
- DISALLOW_COPY_AND_ASSIGN(PartitionTreeNode);
-};
-
-// Class that calculates the optimal aggregation of VP8 partitions smaller than
-// the maximum packet size.
-class Vp8PartitionAggregator {
- public:
- typedef std::vector<int> ConfigVec;
-
- // Constructor. All partitions in the fragmentation header from index
- // first_partition_idx to last_partition_idx must be smaller than
- // maximum packet size to be used in FindOptimalConfiguration.
- Vp8PartitionAggregator(const RTPFragmentationHeader& fragmentation,
- int first_partition_idx, int last_partition_idx);
-
- ~Vp8PartitionAggregator();
-
- // Set the smallest and largest payload sizes produces so far.
- void SetPriorMinMax(int min_size, int max_size);
-
- // Find the aggregation of VP8 partitions that produces the smallest cost.
- // The result is given as a vector of the same length as the number of
- // partitions given to the constructor (i.e., last_partition_idx -
- // first_partition_idx + 1), where each element indicates the packet index
- // for that partition. Thus, the output vector starts at 0 and is increasing
- // up to the number of packets - 1.
- ConfigVec FindOptimalConfiguration(int max_size, int penalty);
-
- // Calculate minimum and maximum packet sizes for a given aggregation config.
- // The extreme packet sizes of the given aggregation are compared with the
- // values given in min_size and max_size, and if either of these are exceeded,
- // the new extreme value will be written to the corresponding variable.
- void CalcMinMax(const ConfigVec& config, int* min_size, int* max_size) const;
-
- // Calculate the number of fragments to divide a large partition into.
- // The large partition is of size large_partition_size. The payload must not
- // be larger than max_payload_size. Each fragment comes at an overhead cost
- // of penalty bytes. If the size of the fragments fall outside the range
- // [min_size, max_size], an extra cost is inflicted.
- static int CalcNumberOfFragments(int large_partition_size,
- int max_payload_size,
- int penalty,
- int min_size,
- int max_size);
-
- private:
- PartitionTreeNode* root_;
- size_t num_partitions_;
- int* size_vector_;
- int largest_partition_size_;
-
- DISALLOW_COPY_AND_ASSIGN(Vp8PartitionAggregator);
-};
-} // namespace
-
-#endif // WEBRTC_MODULES_RTP_RTCP_SOURCE_VP8_PARTITION_AGGREGATOR_H_
diff --git a/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc b/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc
deleted file mode 100644
index 3c274d1..0000000
--- a/trunk/src/modules/rtp_rtcp/source/vp8_partition_aggregator_unittest.cc
+++ /dev/null
@@ -1,215 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdlib.h> // NULL
-
-#include "gtest/gtest.h"
-#include "modules/rtp_rtcp/source/vp8_partition_aggregator.h"
-
-namespace webrtc {
-
-TEST(PartitionTreeNode, CreateAndDelete) {
- const int kVector[] = {1, 2, 3};
- const int kNumPartitions = sizeof(kVector) / sizeof(kVector[0]);
- PartitionTreeNode* node1 =
- PartitionTreeNode::CreateRootNode(kVector, kNumPartitions);
- PartitionTreeNode* node2 =
- new PartitionTreeNode(node1, kVector, kNumPartitions, 17);
- delete node1;
- delete node2;
-}
-
-TEST(PartitionTreeNode, CreateChildrenAndDelete) {
- const int kVector[] = {1, 2, 3};
- const int kNumPartitions = sizeof(kVector) / sizeof(kVector[0]);
- const int kMaxSize = 10;
- const int kPenalty = 5;
- PartitionTreeNode* root =
- PartitionTreeNode::CreateRootNode(kVector, kNumPartitions);
- EXPECT_TRUE(root->CreateChildren(kMaxSize));
- ASSERT_TRUE(NULL != root->left_child());
- ASSERT_TRUE(NULL != root->right_child());
- EXPECT_EQ(3, root->left_child()->this_size());
- EXPECT_EQ(2, root->right_child()->this_size());
- EXPECT_EQ(11, root->right_child()->Cost(kPenalty));
- EXPECT_FALSE(root->left_child()->packet_start());
- EXPECT_TRUE(root->right_child()->packet_start());
- delete root;
-}
-
-TEST(PartitionTreeNode, FindOptimalConfig) {
- const int kVector[] = {197, 194, 213, 215, 184, 199, 197, 207};
- const int kNumPartitions = sizeof(kVector) / sizeof(kVector[0]);
- const int kMaxSize = 1500;
- const int kPenalty = 1;
- PartitionTreeNode* root =
- PartitionTreeNode::CreateRootNode(kVector, kNumPartitions);
- root->set_max_parent_size(500);
- root->set_min_parent_size(300);
- PartitionTreeNode* opt = root->GetOptimalNode(kMaxSize, kPenalty);
- ASSERT_TRUE(opt != NULL);
- EXPECT_EQ(4, opt->NumPackets());
- // Expect optimal sequence to be {1, 0, 1, 0, 1, 0, 1, 0}, which corresponds
- // to (right)-left-right-left-right-left-right-left, where the root node is
- // implicitly a "right" node by definition.
- EXPECT_TRUE(opt->parent()->parent()->parent()->parent()->parent()->
- parent()->parent()->packet_start());
- EXPECT_FALSE(opt->parent()->parent()->parent()->parent()->parent()->
- parent()->packet_start());
- EXPECT_TRUE(opt->parent()->parent()->parent()->parent()->parent()->
- packet_start());
- EXPECT_FALSE(opt->parent()->parent()->parent()->parent()->packet_start());
- EXPECT_TRUE(opt->parent()->parent()->parent()->packet_start());
- EXPECT_FALSE(opt->parent()->parent()->packet_start());
- EXPECT_TRUE(opt->parent()->packet_start());
- EXPECT_FALSE(opt->packet_start());
- EXPECT_TRUE(opt == root->left_child()->right_child()->left_child()->
- right_child()->left_child()->right_child()->left_child());
- delete root;
-}
-
-TEST(PartitionTreeNode, FindOptimalConfigSinglePartition) {
- const int kVector[] = {17};
- const int kNumPartitions = sizeof(kVector) / sizeof(kVector[0]);
- const int kMaxSize = 1500;
- const int kPenalty = 1;
- PartitionTreeNode* root =
- PartitionTreeNode::CreateRootNode(kVector, kNumPartitions);
- PartitionTreeNode* opt = root->GetOptimalNode(kMaxSize, kPenalty);
- ASSERT_TRUE(opt != NULL);
- EXPECT_EQ(1, opt->NumPackets());
- EXPECT_TRUE(opt == root);
- delete root;
-}
-
-static void VerifyConfiguration(const int* expected_config,
- size_t expected_config_len,
- const std::vector<int>& opt_config,
- const RTPFragmentationHeader& fragmentation) {
- ASSERT_EQ(expected_config_len, fragmentation.fragmentationVectorSize);
- EXPECT_EQ(expected_config_len, opt_config.size());
- for (size_t i = 0; i < expected_config_len; ++i) {
- EXPECT_EQ(expected_config[i], opt_config[i]);
- }
-}
-
-static void VerifyMinMax(const Vp8PartitionAggregator& aggregator,
- const std::vector<int>& opt_config,
- int expected_min,
- int expected_max) {
- int min_size = -1;
- int max_size = -1;
- aggregator.CalcMinMax(opt_config, &min_size, &max_size);
- EXPECT_EQ(expected_min, min_size);
- EXPECT_EQ(expected_max, max_size);
-}
-
-TEST(Vp8PartitionAggregator, CreateAndDelete) {
- RTPFragmentationHeader fragmentation;
- fragmentation.VerifyAndAllocateFragmentationHeader(3);
- Vp8PartitionAggregator* aggregator =
- new Vp8PartitionAggregator(fragmentation, 0, 2);
- delete aggregator;
-}
-
-TEST(Vp8PartitionAggregator, FindOptimalConfig) {
- RTPFragmentationHeader fragmentation;
- fragmentation.VerifyAndAllocateFragmentationHeader(8);
- fragmentation.fragmentationLength[0] = 197;
- fragmentation.fragmentationLength[1] = 194;
- fragmentation.fragmentationLength[2] = 213;
- fragmentation.fragmentationLength[3] = 215;
- fragmentation.fragmentationLength[4] = 184;
- fragmentation.fragmentationLength[5] = 199;
- fragmentation.fragmentationLength[6] = 197;
- fragmentation.fragmentationLength[7] = 207;
- Vp8PartitionAggregator* aggregator =
- new Vp8PartitionAggregator(fragmentation, 0, 7);
- aggregator->SetPriorMinMax(300, 500);
- int kMaxSize = 1500;
- int kPenalty = 1;
- std::vector<int> opt_config = aggregator->FindOptimalConfiguration(kMaxSize,
- kPenalty);
- const int kExpectedConfig[] = {0, 0, 1, 1, 2, 2, 3, 3};
- const size_t kExpectedConfigSize =
- sizeof(kExpectedConfig) / sizeof(kExpectedConfig[0]);
- VerifyConfiguration(kExpectedConfig, kExpectedConfigSize, opt_config,
- fragmentation);
- VerifyMinMax(*aggregator, opt_config, 383, 428);
- // Change min and max and run method again. This time, we expect it to leave
- // the values unchanged.
- int min_size = 382;
- int max_size = 429;
- aggregator->CalcMinMax(opt_config, &min_size, &max_size);
- EXPECT_EQ(382, min_size);
- EXPECT_EQ(429, max_size);
- delete aggregator;
-}
-
-TEST(Vp8PartitionAggregator, FindOptimalConfigEqualFragments) {
- RTPFragmentationHeader fragmentation;
- fragmentation.VerifyAndAllocateFragmentationHeader(8);
- fragmentation.fragmentationLength[0] = 200;
- fragmentation.fragmentationLength[1] = 200;
- fragmentation.fragmentationLength[2] = 200;
- fragmentation.fragmentationLength[3] = 200;
- fragmentation.fragmentationLength[4] = 200;
- fragmentation.fragmentationLength[5] = 200;
- fragmentation.fragmentationLength[6] = 200;
- fragmentation.fragmentationLength[7] = 200;
- Vp8PartitionAggregator* aggregator =
- new Vp8PartitionAggregator(fragmentation, 0, 7);
- int kMaxSize = 1500;
- int kPenalty = 1;
- std::vector<int> opt_config = aggregator->FindOptimalConfiguration(kMaxSize,
- kPenalty);
- const int kExpectedConfig[] = {0, 0, 0, 0, 1, 1, 1, 1};
- const size_t kExpectedConfigSize =
- sizeof(kExpectedConfig) / sizeof(kExpectedConfig[0]);
- VerifyConfiguration(kExpectedConfig, kExpectedConfigSize, opt_config,
- fragmentation);
- VerifyMinMax(*aggregator, opt_config, 800, 800);
- delete aggregator;
-}
-
-TEST(Vp8PartitionAggregator, FindOptimalConfigSinglePartition) {
- RTPFragmentationHeader fragmentation;
- fragmentation.VerifyAndAllocateFragmentationHeader(1);
- fragmentation.fragmentationLength[0] = 17;
- Vp8PartitionAggregator* aggregator =
- new Vp8PartitionAggregator(fragmentation, 0, 0);
- int kMaxSize = 1500;
- int kPenalty = 1;
- std::vector<int> opt_config = aggregator->FindOptimalConfiguration(kMaxSize,
- kPenalty);
- const int kExpectedConfig[] = {0};
- const size_t kExpectedConfigSize =
- sizeof(kExpectedConfig) / sizeof(kExpectedConfig[0]);
- VerifyConfiguration(kExpectedConfig, kExpectedConfigSize, opt_config,
- fragmentation);
- VerifyMinMax(*aggregator, opt_config, 17, 17);
- delete aggregator;
-}
-
-TEST(Vp8PartitionAggregator, TestCalcNumberOfFragments) {
- const int kMTU = 1500;
- EXPECT_EQ(2,
- Vp8PartitionAggregator::CalcNumberOfFragments(
- 1600, kMTU, 1, 300, 900));
- EXPECT_EQ(3,
- Vp8PartitionAggregator::CalcNumberOfFragments(
- 1600, kMTU, 1, 300, 798));
- EXPECT_EQ(2,
- Vp8PartitionAggregator::CalcNumberOfFragments(
- 1600, kMTU, 1, 900, 1000));
-}
-
-} // namespace
-
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.cc
deleted file mode 100644
index 1a55e4e..0000000
--- a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.cc
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "BWEConvergenceTest.h"
-
-#include <fstream>
-#include <string>
-
-#include "TestSenderReceiver.h"
-#include "TestLoadGenerator.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "tick_util.h"
-
-
-BWEConvergenceTestUp::BWEConvergenceTestUp(std::string testName, int startRateKbps, int availBWkbps)
-:
-_availBWkbps(availBWkbps),
-BWEOneWayTest(testName, startRateKbps)
-{
-}
-
-
-BWEConvergenceTestUp::~BWEConvergenceTestUp()
-{
- if (_gen)
- {
- delete _gen;
- _gen = NULL;
- }
-}
-
-
-int BWEConvergenceTestUp::Init(std::string ip, WebRtc_UWord16 port)
-{
- // create the load generator object
- const int rtpSampleRate = 90000;
- const int frameRate = 30;
- const double spreadFactor = 0.2;
-
- if (_master)
- {
- _gen = new CBRFixFRGenerator(_sendrec, _startRateKbps, rtpSampleRate, frameRate, spreadFactor);
- if (!_gen)
- {
- return (-1);
- }
- }
-
- return BWEOneWayTest::Init(ip, port);
-}
-
-
-bool BWEConvergenceTestUp::StoppingCriterionMaster()
-{
- return ((_sendrec->BitrateSent() / 1000.0) > (0.9 * _availBWkbps));
-}
-
-
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.h b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.h
deleted file mode 100644
index b830d14..0000000
--- a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEConvergenceTest.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWECONVERGENCETEST_H_
-#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWECONVERGENCETEST_H_
-
-#include <string>
-
-#include "BWETestBase.h"
-
-#include "typedefs.h"
-
-#include "TestSenderReceiver.h"
-
-class BWEConvergenceTestUp : public BWEOneWayTest
-{
-public:
- BWEConvergenceTestUp(std::string testName, int startRateKbps, int availBWkbps);
- virtual ~BWEConvergenceTestUp();
-
- virtual int Init(std::string ip, WebRtc_UWord16 port);
-
-protected:
- virtual bool StoppingCriterionMaster();
-
-private:
- int _availBWkbps;
-};
-
-
-#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWECONVERGENCETEST_H_
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.cc
deleted file mode 100644
index 1fd19fe..0000000
--- a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.cc
+++ /dev/null
@@ -1,84 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <fstream>
-#include <math.h>
-
-#include "BWEStabilityTest.h"
-#include "TestLoadGenerator.h"
-#include "tick_util.h"
-#include "critical_section_wrapper.h"
-
-
-BWEStabilityTest::BWEStabilityTest(std::string testName, int rateKbps, int testDurationSeconds)
-:
-_testDurationSeconds(testDurationSeconds),
-BWEOneWayTest(testName, rateKbps)
-{
-}
-
-
-BWEStabilityTest::~BWEStabilityTest()
-{
- if (_gen)
- {
- delete _gen;
- _gen = NULL;
- }
-}
-
-
-int BWEStabilityTest::Init(std::string ip, WebRtc_UWord16 port)
-{
- // create the load generator object
- const int rtpSampleRate = 90000;
- const int frameRate = 30;
- const double spreadFactor = 0.2;
- const double keyToDeltaRatio = 7;
- const int keyFramePeriod = 300;
-
- if (_master)
- {
- _gen = new CBRFixFRGenerator(_sendrec, _startRateKbps, rtpSampleRate, frameRate, spreadFactor);
- //_gen = new PeriodicKeyFixFRGenerator(_sendrec, _startRateKbps, rtpSampleRate, frameRate,
- // spreadFactor, keyToDeltaRatio, keyFramePeriod);
- if (!_gen)
- {
- return (-1);
- }
-
- }
-
- return BWEOneWayTest::Init(ip, port);
-}
-
-
-void BWEStabilityTest::Report(std::fstream &log)
-{
- // cannot report on a running test
- if(_running) return;
-
- BWETest::Report(log);
-
- CriticalSectionScoped cs(_statCritSect);
-
- log << "Bitrate statistics\n";
- log << "\tAverage = " << _rateVecKbps.Mean() << " kbps\n";
- log << "\tMin = " << _rateVecKbps.Min() << " kbps\n";
- log << "\tMax = " << _rateVecKbps.Max() << " kbps\n";
- log << "\tStd = " << _rateVecKbps.Std() << " kbps\n";
-
-}
-
-
-bool BWEStabilityTest::StoppingCriterionMaster()
-{
- return (TickTime::MillisecondTimestamp() - _startTimeMs >= _testDurationSeconds * 1000);
-}
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.h b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.h
deleted file mode 100644
index 8f213b1..0000000
--- a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStabilityTest.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWESTABILITYTEST_H_
-#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWESTABILITYTEST_H_
-
-#include <string>
-
-#include "BWETestBase.h"
-
-#include "typedefs.h"
-
-#include "TestSenderReceiver.h"
-
-class BWEStabilityTest : public BWEOneWayTest
-{
-public:
- BWEStabilityTest(std::string testName, int rateKbps, int testDurationSeconds);
- virtual ~BWEStabilityTest();
-
- virtual int Init(std::string ip, WebRtc_UWord16 port);
- virtual void Report(std::fstream &log);
-
-protected:
- virtual bool StoppingCriterionMaster();
-
-private:
- int _testDurationSeconds;
-};
-
-
-#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWESTABILITYTEST_H_
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc
deleted file mode 100644
index 471ea5f..0000000
--- a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWEStandAlone.cc
+++ /dev/null
@@ -1,200 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// BWEStandAlone.cpp : Defines the entry point for the console application.
-//
-
-#include <string>
-#include <stdio.h>
-
-#include "event_wrapper.h"
-#include "udp_transport.h"
-#include "rtp_rtcp.h"
-#include "trace.h"
-
-#include "TestSenderReceiver.h"
-#include "TestLoadGenerator.h"
-
-#include "MatlabPlot.h"
-
-//#include "vld.h"
-
-class myTransportCB: public UdpTransportData
-{
-public:
- myTransportCB (RtpRtcp *rtpMod) : _rtpMod(rtpMod) {};
-protected:
- // Inherited from UdpTransportData
- virtual void IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
- const WebRtc_Word32 rtpPacketLength,
- const WebRtc_Word8* fromIP,
- const WebRtc_UWord16 fromPort);
-
- virtual void IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
- const WebRtc_Word32 rtcpPacketLength,
- const WebRtc_Word8* fromIP,
- const WebRtc_UWord16 fromPort);
-
-private:
- RtpRtcp *_rtpMod;
-};
-
-void myTransportCB::IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
- const WebRtc_Word32 rtpPacketLength,
- const WebRtc_Word8* fromIP,
- const WebRtc_UWord16 fromPort)
-{
- printf("Receiving RTP from IP %s, port %u\n", fromIP, fromPort);
- _rtpMod->IncomingPacket((WebRtc_UWord8 *) incomingRtpPacket, static_cast<WebRtc_UWord16>(rtpPacketLength));
-}
-
-void myTransportCB::IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
- const WebRtc_Word32 rtcpPacketLength,
- const WebRtc_Word8* fromIP,
- const WebRtc_UWord16 fromPort)
-{
- printf("Receiving RTCP from IP %s, port %u\n", fromIP, fromPort);
- _rtpMod->IncomingPacket((WebRtc_UWord8 *) incomingRtcpPacket, static_cast<WebRtc_UWord16>(rtcpPacketLength));
-}
-
-
-int main(int argc, char* argv[])
-{
- bool isSender = false;
- bool isReceiver = false;
- WebRtc_UWord16 port;
- std::string ip;
- TestSenderReceiver *sendrec = new TestSenderReceiver();
- TestLoadGenerator *gen;
-
- if (argc == 2)
- {
- // receiver only
- isReceiver = true;
-
- // read port
- port = atoi(argv[1]);
- }
- else if (argc == 3)
- {
- // sender and receiver
- isSender = true;
- isReceiver = true;
-
- // read IP
- ip = argv[1];
-
- // read port
- port = atoi(argv[2]);
- }
-
- Trace::CreateTrace();
- Trace::SetTraceFile("BWEStandAloneTrace.txt");
- Trace::SetLevelFilter(webrtc::kTraceAll);
-
- sendrec->InitReceiver(port);
-
- sendrec->Start();
-
- if (isSender)
- {
- const WebRtc_UWord32 startRateKbps = 1000;
- //gen = new CBRGenerator(sendrec, 1000, 500);
- gen = new CBRFixFRGenerator(sendrec, startRateKbps, 90000, 30, 0.2);
- //gen = new PeriodicKeyFixFRGenerator(sendrec, startRateKbps, 90000, 30, 0.2, 7, 300);
- //const WebRtc_UWord16 numFrameRates = 5;
- //const WebRtc_UWord8 frameRates[numFrameRates] = {30, 15, 20, 23, 25};
- //gen = new CBRVarFRGenerator(sendrec, 1000, frameRates, numFrameRates, 90000, 4.0, 0.1, 0.2);
- //gen = new CBRFrameDropGenerator(sendrec, startRateKbps, 90000, 0.2);
- sendrec->SetLoadGenerator(gen);
- sendrec->InitSender(startRateKbps, ip.c_str(), port);
- gen->Start();
- }
-
- while (1)
- {
- }
-
- if (isSender)
- {
- gen->Stop();
- delete gen;
- }
-
- delete sendrec;
-
- //WebRtc_UWord8 numberOfSocketThreads = 1;
- //UdpTransport* transport = UdpTransport::Create(0, numberOfSocketThreads);
-
- //RtpRtcp* rtp = RtpRtcp::CreateRtpRtcp(1, false);
- //if (rtp->InitSender() != 0)
- //{
- // exit(1);
- //}
- //if (rtp->RegisterSendTransport(transport) != 0)
- //{
- // exit(1);
- //}
-
-// transport->InitializeSendSockets("192.168.200.39", 8000);
- //transport->InitializeSendSockets("127.0.0.1", 10000);
- //transport->InitializeSourcePorts(8000);
-
-
- return(0);
- // myTransportCB *tp = new myTransportCB(rtp);
- // transport->InitializeReceiveSockets(tp, 10000, "0.0.0.0");
- // transport->StartReceiving(500);
-
- // WebRtc_Word8 data[100];
- // for (int i = 0; i < 100; data[i] = i++);
-
- // for (int i = 0; i < 100; i++)
- // {
- // transport->SendRaw(data, 100, false);
- // }
-
-
-
- // WebRtc_Word32 totTime = 0;
- // while (totTime < 10000)
- // {
- // transport->Process();
- // WebRtc_Word32 wTime = transport->TimeUntilNextProcess();
- // totTime += wTime;
- // Sleep(wTime);
- // }
-
-
- //if (transport)
- //{
- // // Destroy the Socket Transport module
- // transport->StopReceiving();
- // transport->InitializeReceiveSockets(NULL,0);// deregister callback
- // UdpTransport::Destroy(transport);
- // transport = NULL;
- // }
-
- // if (tp)
- // {
- // delete tp;
- // tp = NULL;
- // }
-
- // if (rtp)
- // {
- // RtpRtcp::DestroyRtpRtcp(rtp);
- // rtp = NULL;
- // }
-
-
- //return 0;
-}
-
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.cc
deleted file mode 100644
index 2940abd..0000000
--- a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.cc
+++ /dev/null
@@ -1,453 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "BWETestBase.h"
-
-#include <algorithm> // sort
-#include <fstream>
-#include <string>
-#include <vector>
-#include <math.h>
-
-#include "TestSenderReceiver.h"
-#include "TestLoadGenerator.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "tick_util.h"
-#include "critical_section_wrapper.h"
-
-
-double StatVec::Mean()
-{
- double sum = 0;
-
- // sanity
- if (size() <= 0) return (0);
-
- std::vector<double>::iterator it;
- for (it = begin(); it < end(); ++it)
- {
- sum += (*it);
- }
-
- return (sum / size());
-}
-
-double StatVec::Variance()
-{
- double sumSqaure = 0;
- double sum = 0;
-
- std::vector<double>::iterator it;
- for (it = begin(); it < end(); ++it)
- {
- sum += (*it);
- sumSqaure += (*it) * (*it);
- }
-
- // Normalizes by N-1. This produces the best unbiased estimate of the
- // variance if X is a sample from a normal distribution.
- int M = static_cast<int> (size() - 1);
-
- if (M > 0)
- {
- double var = (sumSqaure / M) - (sum / (M+1)) * (sum / M);
- assert(var >= 0);
- return (var);
- }
- else
- {
- return (0);
- }
-}
-
-double StatVec::Std()
-{
- return (sqrt(Variance()));
-}
-
-double StatVec::Max()
-{
- // sanity
- if (size() <= 0) return (0);
-
- std::vector<double>::iterator it = begin();
- double maxVal = (*it);
- ++it;
-
- for (; it < end(); ++it)
- {
- if ((*it) > maxVal) maxVal = (*it);
- }
-
- return (maxVal);
-}
-
-double StatVec::Min()
-{
- // sanity
- if (size() <= 0) return (0);
-
- std::vector<double>::iterator it = begin();
- double minVal = (*it);
- ++it;
-
- for (; it < end(); ++it)
- {
- if ((*it) < minVal) minVal = (*it);
- }
-
- return (minVal);
-}
-
-double StatVec::Median()
-{
- double median;
-
- // sanity
- if (size() <= 0) return (0);
-
- // sort the vector
- sort(begin(), end());
-
- if ((size() % 2) == 0)
- {
- // even size; use average of two center elements
- median = (at(size()/2 - 1) + at(size()/2)) / 2.0;
- }
- else
- {
- // odd size; take center element
- median = at(size()/2);
- }
-
- return (median);
-}
-
-double StatVec::Percentile(double p)
-{
- // sanity
- if (size() <= 0) return (0);
-
- // sort the vector
- sort(begin(), end());
-
- int rank = static_cast<int> (((size() - 1) * p) / 100 + 0.5); // between 1 and size()
- rank -= 1; // between 0 and size()-1
-
- assert(rank >= 0);
- assert(rank < static_cast<int>(size()));
-
- return (at(rank));
-}
-
-void StatVec::Export(std::fstream &file, bool colVec /*= false*/)
-{
- // sanity
- if (size() <= 0) return;
-
- std::string separator;
- if (colVec) separator = "\n";
- else separator = ", ";
-
- std::vector<double>::iterator it = begin();
- file << (*it);
- ++it;
-
- for (; it < end(); ++it)
- {
- file << separator << (*it);
- }
-
- file << std::endl;
-}
-
-
-bool BWETestProcThreadFunction(void *obj)
-{
- if (obj == NULL)
- {
- return false;
- }
- BWETest *theObj = static_cast<BWETest *>(obj);
-
- theObj->ProcLoop();
-
- theObj->Stop();
-
- return(true);
-}
-
-
-BWETest::BWETest(std::string testName, int startRateKbps):
-_testName(testName),
-_startRateKbps(startRateKbps),
-_master(false),
-_sendrec(NULL),
-_gen(NULL),
-_initialized(false),
-_started(false),
-_running(false),
-_eventPtr(NULL),
-_procThread(NULL),
-_startTimeMs(-1),
-_stopTimeMs(-1),
-_statCritSect(CriticalSectionWrapper::CreateCriticalSection())
-{
- _sendrec = new TestSenderReceiver();
-}
-
-
-BWETest::~BWETest()
-{
- if (_running)
- {
- Stop();
- }
-
- _statCritSect->Enter();
- delete &_statCritSect;
-
- if (_sendrec)
- {
- delete _sendrec;
- _sendrec = NULL;
- }
-}
-
-
-bool BWETest::SetMaster(bool isMaster /*= true*/)
-{
- if (!_initialized)
- {
- // Can only set status before initializing.
- _master = isMaster;
- }
-
- return (_master);
-}
-
-
-int BWETest::Init(std::string ip, WebRtc_UWord16 port)
-{
- if (_initialized)
- {
- // cannot init twice
- return (-1);
- }
-
- if (!_sendrec)
- {
- throw "SenderReceiver must be created";
- exit(1);
- }
-
- if (_started)
- {
- // cannot init after start
- return (-1);
- }
-
- // initialize receiver port (for feedback)
- _sendrec->InitReceiver(port);
-
- // initialize sender
- _sendrec->SetLoadGenerator(_gen);
- _sendrec->InitSender(_startRateKbps, ip.c_str(), port);
- //_gen->Start();
-
- _sendrec->SetCallback(this);
-
- _initialized = true;
-
- return 0;
-}
-
-
-bool BWETest::Start()
-{
- if (!_initialized)
- {
- // must init first
- return (false);
- }
- if (_started)
- {
- // already started, do nothing
- return (true);
- }
-
- if (_sendrec->Start() != 0)
- {
- // failed
- return (false);
- }
-
- if (_gen)
- {
- if (_gen->Start() != 0)
- {
- // failed
- return (false);
- }
- }
-
- _eventPtr = EventWrapper::Create();
-
- _startTimeMs = TickTime::MillisecondTimestamp();
- _started = true;
- _running = true;
-
- return (true);
-}
-
-
-bool BWETest::Stop()
-{
- if (_procThread)
- {
- _stopTimeMs = TickTime::MillisecondTimestamp();
- _procThread->SetNotAlive();
- _running = false;
- _eventPtr->Set();
-
- while (!_procThread->Stop())
- {
- ;
- }
-
- delete _procThread;
- _procThread = NULL;
-
- }
-
- if (_eventPtr)
- {
- delete _eventPtr;
- _eventPtr = NULL;
- }
-
- _procThread = NULL;
-
- if(_gen)
- {
- _gen->Stop();
- }
-
- return(true);
-}
-
-
-bool BWETest::ProcLoop(void)
-{
- bool receiving = false;
-
- // no critSect
- while (_running)
- {
-
- // check stopping criterions
- if (_master && StoppingCriterionMaster())
- {
- printf("StoppingCriterionMaster()\n");
- _stopTimeMs = TickTime::MillisecondTimestamp();
- _running = false;
- }
- else if (!_master && StoppingCriterionSlave())
- {
- printf("StoppingCriterionSlave()\n");
- _running = false;
- }
-
- // wait
- _eventPtr->Wait(1000); // 1000 ms
-
- }
-
- return true;
-}
-
-
-void BWETest::Report(std::fstream &log)
-{
- // cannot report on a running test
- if(_running) return;
-
- CriticalSectionScoped cs(_statCritSect);
-
- log << "\n\n*** Test name = " << _testName << "\n";
- log << "Execution time = " << static_cast<double>(_stopTimeMs - _startTimeMs) / 1000 << " s\n";
- log << "\n";
- log << "RTT statistics\n";
- log << "\tMin = " << _rttVecMs.Min() << " ms\n";
- log << "\tMax = " << _rttVecMs.Max() << " ms\n";
- log << "\n";
- log << "Loss statistics\n";
- log << "\tAverage = " << _lossVec.Mean() << "%\n";
- log << "\tMax = " << _lossVec.Max() << "%\n";
-
- log << "\n" << "Rates" << "\n";
- _rateVecKbps.Export(log);
-
- log << "\n" << "RTT" << "\n";
- _rttVecMs.Export(log);
-
-}
-
-
-// SenderReceiver callback
-void BWETest::OnOnNetworkChanged(const WebRtc_UWord32 bitrateTargetBps,
- const WebRtc_UWord8 fractionLost,
- const WebRtc_UWord16 roundTripTimeMs,
- const WebRtc_UWord32 jitterMS,
- const WebRtc_UWord16 bwEstimateKbitMin,
- const WebRtc_UWord16 bwEstimateKbitMax)
-{
- CriticalSectionScoped cs(_statCritSect);
-
- // bitrate statistics
- WebRtc_Word32 newBitrateKbps = bitrateTargetBps/1000;
-
- _rateVecKbps.push_back(newBitrateKbps);
- _rttVecMs.push_back(roundTripTimeMs);
- _lossVec.push_back(static_cast<double>(fractionLost) / 255.0);
-}
-
-
-int BWEOneWayTest::Init(std::string ip, WebRtc_UWord16 port)
-{
-
- if (!_master)
- {
- // Use timeout stopping criterion by default for receiver
- UseRecvTimeout();
- }
-
- return (BWETest::Init(ip, port));
-
-}
-
-
-bool BWEOneWayTest::Start()
-{
- bool ret = BWETest::Start();
-
- if (!_master)
- {
- // send one dummy RTP packet to enable RTT measurements
- const WebRtc_UWord8 dummy = 0;
- //_gen->sendPayload(TickTime::MillisecondTimestamp(), &dummy, 0);
- _sendrec->SendOutgoingData(
- static_cast<WebRtc_UWord32>(TickTime::MillisecondTimestamp()*90),
- &dummy, 1, webrtc::kVideoFrameDelta);
- }
-
- return ret;
-}
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h
deleted file mode 100644
index bab1b94..0000000
--- a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETestBase.h
+++ /dev/null
@@ -1,102 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETESTBASE_H_
-#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETESTBASE_H_
-
-#include <string>
-#include <vector>
-
-#include "typedefs.h"
-
-#include "TestSenderReceiver.h"
-
-
-class StatVec : public std::vector<double>
-{
-public:
- double Mean();
- double Variance();
- double Std();
- double Max();
- double Min();
- double Median();
- double Percentile(double p); // 0 <= p <= 100%
- void Export(std::fstream &file, bool colVec = false);
-};
-
-
-class BWETest : public SendRecCB
-{
-public:
- BWETest(std::string testName, int startRateKbps);
- virtual ~BWETest();
-
- bool SetMaster(bool isMaster = true);
- void UseRecvTimeout() { _sendrec->SetPacketTimeout(1000); };
- virtual int Init(std::string ip, WebRtc_UWord16 port);
- virtual bool Start();
- virtual bool Stop();
- bool ProcLoop(void);
- virtual void Report(std::fstream &log);
- std::string TestName() { return (_testName); };
-
- // SenderReceiver callback
- virtual void OnOnNetworkChanged(const WebRtc_UWord32 bitrateTargetBps,
- const WebRtc_UWord8 fractionLost,
- const WebRtc_UWord16 roundTripTimeMs,
- const WebRtc_UWord32 jitterMS,
- const WebRtc_UWord16 bwEstimateKbitMin,
- const WebRtc_UWord16 bwEstimateKbitMax);
-
-
-protected:
- virtual bool StoppingCriterionMaster() = 0;
- virtual bool StoppingCriterionSlave() { return (_sendrec->timeOutTriggered()); };
-
- TestSenderReceiver * _sendrec;
- TestLoadGenerator * _gen;
-
- std::string _testName;
- int _startRateKbps;
- bool _master;
- bool _initialized;
- bool _started;
- bool _running;
- EventWrapper *_eventPtr;
- ThreadWrapper* _procThread;
- WebRtc_Word64 _startTimeMs;
- WebRtc_Word64 _stopTimeMs;
-
- // Statistics, protected by separate CritSect
- CriticalSectionWrapper* _statCritSect;
- StatVec _rateVecKbps;
- StatVec _rttVecMs;
- StatVec _lossVec;
-};
-
-
-class BWEOneWayTest : public BWETest
-{
-public:
- BWEOneWayTest(std::string testName, int startRateKbps) :
- BWETest(testName, startRateKbps) {};
-
- virtual int Init(std::string ip, WebRtc_UWord16 port);
- virtual bool Start();
-
-protected:
- virtual bool StoppingCriterionSlave() {return ( _sendrec->timeOutTriggered()); };
-
-private:
-
-};
-
-#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETESTBASE_H_
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETester.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETester.cc
deleted file mode 100644
index f1d79fe..0000000
--- a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETester.cc
+++ /dev/null
@@ -1,274 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// BWETester.cpp : Defines the entry point for the console application.
-//
-
-#include <fstream>
-#include <string>
-#include <iostream>
-#include <ctime>
-
-#include "event_wrapper.h"
-#include "trace.h"
-
-#include "BWEStabilityTest.h"
-#include "BWEConvergenceTest.h"
-#include "BWETwoWayLimitFinding.h"
-
-#include "MatlabPlot.h"
-
-//#include "vld.h"
-
-#ifdef MATLAB
-MatlabEngine eng;
-#endif
-
-
-class testContainer
-{
-public:
- testContainer(BWETest *test, bool waitForKeyStroke, int delayStartSec,
- std::string instruction) :
- _test(test),
- _waitMaster(waitForKeyStroke),
- _waitSlave(waitForKeyStroke),
- _delayMaster(delayStartSec),
- _delaySlave(delayStartSec),
- _instr(instruction) {};
-
- testContainer(BWETest *test,
- bool waitForKeyStrokeMaster,
- bool waitForKeyStrokeSlave,
- int delayStartSecMaster,
- int delayStartSecSlave,
- std::string instruction) :
- _test(test),
- _waitMaster(waitForKeyStrokeMaster),
- _waitSlave(waitForKeyStrokeSlave),
- _delayMaster(delayStartSecMaster),
- _delaySlave(delayStartSecSlave),
- _instr(instruction) {};
-
- ~testContainer() { if(_test) delete _test; _test = NULL; };
-
- BWETest *_test;
- bool _waitMaster;
- bool _waitSlave;
- int _delayMaster;
- int _delaySlave;
- std::string _instr;
-};
-
-
-// This is were the test cases are created.
-// Syntax:
-// tests->push_back(new testContainer(
-// new _BWETestConstructor_, // constructor for the test case
-// _wait_, // wait for user key press before start
-// _delay_, // delay test start (after a key press if enabled)
-// "Intruction to user.")); // message to show in console before starting
-//
-// Or:
-// tests->push_back(new testContainer(
-// new _BWETestConstructor_, // constructor for the test case
-// _waitMaster_, // master will wait for user key press before start
-// _waitSlave_, // slave will wait for user key press before start
-// _delayMaster_, // delay master test start (after a key press if enabled)
-// _delaySlave_, // delay slave test start (after a key press if enabled)
-// "Intruction to user.")); // message to show in console before starting
-//
-// Valid test cases are:
-// BWEConvergenceTestUp
-// BWEStabilityTest
-// BWETwoWayLimitFinding
-
-
-void PopulateTests(std::vector<testContainer *>* tests, bool isMaster)
-{
- tests->push_back(new testContainer(
- new BWEStabilityTest("Stability", 400, 5*60),
- true, true,
- 0, 0,
- "Set bandwidth limit to 512 kbps"));
-
- tests->push_back(new testContainer(
- new BWEStabilityTest("Stability", 4000, 5*60),
- true, true,
- 0, 0,
- "Set bandwidth limit to 5120 kbps"));
-
- tests->push_back(new testContainer(
- new BWEStabilityTest("Stability", 400, 5*60),
- true, true,
- 0, 0,
- "Set bandwidth limit to 512 kbps and a normal distributed delay\
- with mean 100 ms and std dev 15 ms"));
-
- tests->push_back(new testContainer(
- new BWEConvergenceTestUp("Convergence 256->512", 256, 512),
- true,
- 0,
- "Set bandwith limit to 512 kbps"));
-
- tests->push_back(new testContainer(
- new BWEConvergenceTestUp("Convergence 1024->5120", 1024, 5120),
- true,
- 0,
- "Set bandwith limit to 5120 kbps"));
-
- tests->push_back(new testContainer(
- new BWETwoWayLimitFinding("Asymmetric limit finding {1024, 2048} kbps",
- 500, 1024,
- 500, 2048,
- isMaster),
- true,
- 0,
- "Set bandwith limit to {1024, 2048} kbps asymmetric"));
-
- tests->push_back(new testContainer(
- new BWETwoWayLimitFinding("Symmetric limit finding {1024, 1024} kbps",
- 500, 1024,
- 500, 1024,
- isMaster),
- true,
- 0,
- "Set bandwith limit to 1024 kbps symmetric"));
-}
-
-
-int main(int argc, char* argv[])
-{
-
- bool isMaster = false;
- WebRtc_UWord16 port;
- std::string ip;
- std::fstream log;
- log.open("TestLog.txt", std::fstream::out | std::fstream::app);
-
- log << "\n\nBWE TESTER\n";
-
- time_t t = time(0); // get time now
- struct tm * now = localtime( & t );
- log << (now->tm_year + 1900) << '-'
- << (now->tm_mon + 1) << '-'
- << now->tm_mday << " "
- << now->tm_hour << ":" << now->tm_min
- << "\n";
-
- if (argc == 4)
- {
- // read IP
- ip = argv[1];
-
- // read port
- port = atoi(argv[2]);
-
- // read master/slave
- isMaster = (atoi(argv[3]) != 0);
-
- std::cout << "Destination: " << ip << "\n";
- log << "Destination: " << ip << "\n";
- std::cout << "Port: " << port << "\n";
- log << "Port: " << port << "\n";
- if (isMaster)
- {
- std::cout << "Master\n";
- log << "Master\n";
- }
- else
- {
- std::cout << "Slave\n";
- log << "Slave\n";
- }
-
- }
- else
- {
- printf("Usage\nBWETester dstIP port master\n");
- exit(1);
- }
-
- std::vector<testContainer*> tests;
- PopulateTests(&tests, isMaster);
-
- int testIndex = 0;
- EventWrapper* event = EventWrapper::Create();
- std::vector<testContainer*>::iterator it;
- for (it=tests.begin() ; it < tests.end(); it++)
- {
- ++testIndex;
-
- BWETest *theTest = (*it)->_test;
-
- if (theTest)
- {
- std::cout << "\nTest " << testIndex << ": " << theTest->TestName() << "\n";
- }
-
- // Print instructions
- std::cout << "--> " << (*it)->_instr << std::endl;
-
- if ((isMaster && (*it)->_waitMaster)
- || (!isMaster && (*it)->_waitSlave))
- {
- // Wait for a key press
- std::cout << "Press enter to start test\n";
- getc(stdin);
- }
-
- if (isMaster)
- {
- if ((*it)->_delayMaster > 0)
- {
- // Wait
- std::cout << "Test starting in "
- << (*it)->_delayMaster
- << " seconds" << std::endl;
- event->Wait((*it)->_delayMaster * 1000);
- }
- }
- else
- {
- if ((*it)->_delaySlave > 0)
- {
- // Wait
- std::cout << "Test starting in "
- << (*it)->_delaySlave
- << " seconds" << std::endl;
- event->Wait((*it)->_delaySlave * 1000);
- }
- }
-
- // Start execution
- if (theTest)
- {
- theTest->SetMaster(isMaster);
- if (theTest->Init(ip, port) != 0)
- {
- throw "Error initializing sender";
- exit (1);
- }
-
- theTest->Start();
- theTest->ProcLoop();
- theTest->Stop();
- theTest->Report(log);
- log << std::flush;
- }
-
- delete (*it); // deletes the test too
- }
- delete event;
- event = NULL;
-
- log.close();
- return (0);
-}
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.cc
deleted file mode 100644
index 043c7b0..0000000
--- a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.cc
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "BWETwoWayLimitFinding.h"
-#include "TestLoadGenerator.h"
-
-
-BWETwoWayLimitFinding::BWETwoWayLimitFinding(
- std::string testName,
- int masterStartRateKbps, int masterAvailBWkbps,
- int slaveStartRateKbps, int slaveAvailBWkbps,
- bool isMaster /*= false*/)
- :
-BWETest(testName, (isMaster ? masterStartRateKbps : slaveStartRateKbps)),
-_availBWkbps(isMaster ? masterAvailBWkbps : slaveAvailBWkbps),
-_incomingAvailBWkbps(isMaster ? slaveAvailBWkbps : masterAvailBWkbps),
-_forwLimitReached(false),
-_revLimitReached(false)
-{
- _master = isMaster;
-}
-
-
-BWETwoWayLimitFinding::~BWETwoWayLimitFinding()
-{
- if (_gen)
- {
- delete _gen;
- _gen = NULL;
- }
-}
-
-
-int BWETwoWayLimitFinding::Init(std::string ip, WebRtc_UWord16 port)
-{
- // create the load generator object
- const int rtpSampleRate = 90000;
- const int frameRate = 30;
- const double spreadFactor = 0.2;
-
- _gen = new CBRFixFRGenerator(_sendrec, _startRateKbps, rtpSampleRate, frameRate, spreadFactor);
- if (!_gen)
- {
- return (-1);
- }
-
- if (!_master) UseRecvTimeout(); // slave shuts down when incoming stream dies
-
- return BWETest::Init(ip, port);
-}
-
-
-bool BWETwoWayLimitFinding::StoppingCriterionMaster()
-{
- if ((_sendrec->BitrateSent() / 1000.0) > (0.95 * _availBWkbps))
- {
- _forwLimitReached = true;
- }
-
- WebRtc_Word32 revRateKbps = _sendrec->ReceiveBitrateKbps();
- if (revRateKbps > (0.95 * _incomingAvailBWkbps))
- {
- _revLimitReached = true;
- }
-
- return (_forwLimitReached && _revLimitReached);
-}
-
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.h b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.h
deleted file mode 100644
index fc790e5..0000000
--- a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/BWETwoWayLimitFinding.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETWOWAYLIMITFINDING_H_
-#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETWOWAYLIMITFINDING_H_
-
-#include "BWETestBase.h"
-
-class BWETwoWayLimitFinding : public BWETest
-{
-public:
- BWETwoWayLimitFinding(std::string testName,
- int masterStartRateKbps, int masterAvailBWkbps,
- int slaveStartRateKbps, int slaveAvailBWkbps,
- bool isMaster = false);
-
- virtual ~BWETwoWayLimitFinding();
-
- virtual int Init(std::string ip, WebRtc_UWord16 port);
-
-protected:
- virtual bool StoppingCriterionMaster();
- //virtual bool StoppingCriterionSlave();
-
-private:
- int _availBWkbps;
- int _incomingAvailBWkbps;
- bool _forwLimitReached;
- bool _revLimitReached;
-
-};
-
-
-#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_BWETWOWAYLIMITFINDING_H_
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc
deleted file mode 100644
index 9c81fd0..0000000
--- a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.cc
+++ /dev/null
@@ -1,1071 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "MatlabPlot.h"
-#ifdef MATLAB
-#include "engine.h"
-#endif
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "critical_section_wrapper.h"
-#include "tick_util.h"
-
-#include <sstream>
-#include <algorithm>
-#include <math.h>
-#include <stdio.h>
-
-using namespace webrtc;
-
-#ifdef MATLAB
-MatlabEngine eng;
-
-MatlabLine::MatlabLine(int maxLen /*= -1*/, const char *plotAttrib /*= NULL*/, const char *name /*= NULL*/)
-:
-_xArray(NULL),
-_yArray(NULL),
-_maxLen(maxLen),
-_plotAttribute(),
-_name()
-{
- if (_maxLen > 0)
- {
- _xArray = mxCreateDoubleMatrix(1, _maxLen, mxREAL);
- _yArray = mxCreateDoubleMatrix(1, _maxLen, mxREAL);
- }
-
- if (plotAttrib)
- {
- _plotAttribute = plotAttrib;
- }
-
- if (name)
- {
- _name = name;
- }
-}
-
-MatlabLine::~MatlabLine()
-{
- if (_xArray != NULL)
- {
- mxDestroyArray(_xArray);
- }
- if (_yArray != NULL)
- {
- mxDestroyArray(_yArray);
- }
-}
-
-void MatlabLine::Append(double x, double y)
-{
- if (_maxLen > 0 && _xData.size() > static_cast<WebRtc_UWord32>(_maxLen))
- {
- _xData.resize(_maxLen);
- _yData.resize(_maxLen);
- }
-
- _xData.push_front(x);
- _yData.push_front(y);
-}
-
-
-// append y-data with running integer index as x-data
-void MatlabLine::Append(double y)
-{
- if (_xData.empty())
- {
- // first element is index 0
- Append(0, y);
- }
- else
- {
- // take last x-value and increment
- double temp = _xData.back(); // last x-value
- Append(temp + 1, y);
- }
-}
-
-
-void MatlabLine::SetMaxLen(int maxLen)
-{
- if (maxLen <= 0)
- {
- // means no maxLen
- _maxLen = -1;
- }
- else
- {
- _maxLen = maxLen;
-
- if (_xArray != NULL)
- {
- mxDestroyArray(_xArray);
- mxDestroyArray(_yArray);
- }
- _xArray = mxCreateDoubleMatrix(1, _maxLen, mxREAL);
- _yArray = mxCreateDoubleMatrix(1, _maxLen, mxREAL);
-
- maxLen = ((unsigned int)maxLen <= _xData.size()) ? maxLen : (int)_xData.size();
- _xData.resize(maxLen);
- _yData.resize(maxLen);
-
- //// reserve the right amount of memory
- //_xData.reserve(_maxLen);
- //_yData.reserve(_maxLen);
- }
-}
-
-void MatlabLine::SetAttribute(char *plotAttrib)
-{
- _plotAttribute = plotAttrib;
-}
-
-void MatlabLine::SetName(char *name)
-{
- _name = name;
-}
-
-void MatlabLine::GetPlotData(mxArray** xData, mxArray** yData)
-{
- // Make sure we have enough Matlab allocated memory.
- // Assuming both arrays (x and y) are of the same size.
- if (_xData.empty())
- {
- return; // No data
- }
- unsigned int size = 0;
- if (_xArray != NULL)
- {
- size = (unsigned int)mxGetNumberOfElements(_xArray);
- }
- if (size < _xData.size())
- {
- if (_xArray != NULL)
- {
- mxDestroyArray(_xArray);
- mxDestroyArray(_yArray);
- }
- _xArray = mxCreateDoubleMatrix(1, _xData.size(), mxREAL);
- _yArray = mxCreateDoubleMatrix(1, _yData.size(), mxREAL);
- }
-
- if (!_xData.empty())
- {
- double* x = mxGetPr(_xArray);
-
- std::list<double>::iterator it = _xData.begin();
-
- for (int i = 0; it != _xData.end(); it++, i++)
- {
- x[i] = *it;
- }
- }
-
- if (!_yData.empty())
- {
- double* y = mxGetPr(_yArray);
-
- std::list<double>::iterator it = _yData.begin();
-
- for (int i = 0; it != _yData.end(); it++, i++)
- {
- y[i] = *it;
- }
- }
- *xData = _xArray;
- *yData = _yArray;
-}
-
-std::string MatlabLine::GetXName()
-{
- std::ostringstream xString;
- xString << "x_" << _name;
- return xString.str();
-}
-
-std::string MatlabLine::GetYName()
-{
- std::ostringstream yString;
- yString << "y_" << _name;
- return yString.str();
-}
-
-std::string MatlabLine::GetPlotString()
-{
-
- std::ostringstream s;
-
- if (_xData.size() == 0)
- {
- s << "[0 1], [0 1]"; // To get an empty plot
- }
- else
- {
- s << GetXName() << "(1:" << _xData.size() << "),";
- s << GetYName() << "(1:" << _yData.size() << ")";
- }
-
- s << ", '";
- s << _plotAttribute;
- s << "'";
-
- return s.str();
-}
-
-std::string MatlabLine::GetRefreshString()
-{
- std::ostringstream s;
-
- if (_xData.size() > 0)
- {
- s << "set(h,'xdata',"<< GetXName() <<"(1:" << _xData.size() << "),'ydata',"<< GetYName() << "(1:" << _yData.size() << "));";
- }
- else
- {
- s << "set(h,'xdata',[NaN],'ydata',[NaN]);";
- }
- return s.str();
-}
-
-std::string MatlabLine::GetLegendString()
-{
- return ("'" + _name + "'");
-}
-
-bool MatlabLine::hasLegend()
-{
- return (!_name.empty());
-}
-
-
-// remove data points, but keep attributes
-void MatlabLine::Reset()
-{
- _xData.clear();
- _yData.clear();
-}
-
-
-void MatlabLine::UpdateTrendLine(MatlabLine * sourceData, double slope, double offset)
-{
- Reset(); // reset data, not attributes and name
-
- double thexMin = sourceData->xMin();
- double thexMax = sourceData->xMax();
- Append(thexMin, thexMin * slope + offset);
- Append(thexMax, thexMax * slope + offset);
-}
-
-double MatlabLine::xMin()
-{
- if (!_xData.empty())
- {
- std::list<double>::iterator theStart = _xData.begin();
- std::list<double>::iterator theEnd = _xData.end();
- return(*min_element(theStart, theEnd));
- }
- return (0.0);
-}
-
-double MatlabLine::xMax()
-{
- if (!_xData.empty())
- {
- std::list<double>::iterator theStart = _xData.begin();
- std::list<double>::iterator theEnd = _xData.end();
- return(*max_element(theStart, theEnd));
- }
- return (0.0);
-}
-
-double MatlabLine::yMin()
-{
- if (!_yData.empty())
- {
- std::list<double>::iterator theStart = _yData.begin();
- std::list<double>::iterator theEnd = _yData.end();
- return(*min_element(theStart, theEnd));
- }
- return (0.0);
-}
-
-double MatlabLine::yMax()
-{
- if (!_yData.empty())
- {
- std::list<double>::iterator theStart = _yData.begin();
- std::list<double>::iterator theEnd = _yData.end();
- return(*max_element(theStart, theEnd));
- }
- return (0.0);
-}
-
-
-
-MatlabTimeLine::MatlabTimeLine(int horizonSeconds /*= -1*/, const char *plotAttrib /*= NULL*/,
- const char *name /*= NULL*/,
- WebRtc_Word64 refTimeMs /* = -1*/)
- :
-_timeHorizon(horizonSeconds),
-MatlabLine(-1, plotAttrib, name) // infinite number of elements
-{
- if (refTimeMs < 0)
- _refTimeMs = TickTime::MillisecondTimestamp();
- else
- _refTimeMs = refTimeMs;
-}
-
-void MatlabTimeLine::Append(double y)
-{
- MatlabLine::Append(static_cast<double>(TickTime::MillisecondTimestamp() - _refTimeMs) / 1000.0, y);
-
- PurgeOldData();
-}
-
-
-void MatlabTimeLine::PurgeOldData()
-{
- if (_timeHorizon > 0)
- {
- // remove old data
- double historyLimit = static_cast<double>(TickTime::MillisecondTimestamp() - _refTimeMs) / 1000.0
- - _timeHorizon; // remove data points older than this
-
- std::list<double>::reverse_iterator ritx = _xData.rbegin();
- WebRtc_UWord32 removeCount = 0;
- while (ritx != _xData.rend())
- {
- if (*ritx >= historyLimit)
- {
- break;
- }
- ritx++;
- removeCount++;
- }
- if (removeCount == 0)
- {
- return;
- }
-
- // remove the range [begin, it).
- //if (removeCount > 10)
- //{
- // printf("Removing %lu elements\n", removeCount);
- //}
- _xData.resize(_xData.size() - removeCount);
- _yData.resize(_yData.size() - removeCount);
- }
-}
-
-
-WebRtc_Word64 MatlabTimeLine::GetRefTime()
-{
- return(_refTimeMs);
-}
-
-
-
-
-MatlabPlot::MatlabPlot()
-:
-_figHandle(-1),
-_smartAxis(false),
-_critSect(CriticalSectionWrapper::CreateCriticalSection()),
-_timeToPlot(false),
-_plotting(false),
-_enabled(true),
-_firstPlot(true),
-_legendEnabled(true),
-_donePlottingEvent(EventWrapper::Create())
-{
- CriticalSectionScoped cs(_critSect);
-
- _xlim[0] = 0;
- _xlim[1] = 0;
- _ylim[0] = 0;
- _ylim[1] = 0;
-
-#ifdef PLOT_TESTING
- _plotStartTime = -1;
- _plotDelay = 0;
-#endif
-
-}
-
-
-MatlabPlot::~MatlabPlot()
-{
- _critSect->Enter();
-
- // delete all line objects
- while (!_line.empty())
- {
- delete *(_line.end() - 1);
- _line.pop_back();
- }
-
- delete _critSect;
- delete _donePlottingEvent;
-}
-
-
-int MatlabPlot::AddLine(int maxLen /*= -1*/, const char *plotAttrib /*= NULL*/, const char *name /*= NULL*/)
-{
- CriticalSectionScoped cs(_critSect);
- if (!_enabled)
- {
- return -1;
- }
-
- MatlabLine *newLine = new MatlabLine(maxLen, plotAttrib, name);
- _line.push_back(newLine);
-
- return (static_cast<int>(_line.size() - 1)); // index of newly inserted line
-}
-
-
-int MatlabPlot::AddTimeLine(int maxLen /*= -1*/, const char *plotAttrib /*= NULL*/, const char *name /*= NULL*/,
- WebRtc_Word64 refTimeMs /*= -1*/)
-{
- CriticalSectionScoped cs(_critSect);
-
- if (!_enabled)
- {
- return -1;
- }
-
- MatlabTimeLine *newLine = new MatlabTimeLine(maxLen, plotAttrib, name, refTimeMs);
- _line.push_back(newLine);
-
- return (static_cast<int>(_line.size() - 1)); // index of newly inserted line
-}
-
-
-int MatlabPlot::GetLineIx(const char *name)
-{
- CriticalSectionScoped cs(_critSect);
-
- if (!_enabled)
- {
- return -1;
- }
-
- // search the list for a matching line name
- std::vector<MatlabLine*>::iterator it = _line.begin();
- bool matchFound = false;
- int lineIx = 0;
-
- for (; it != _line.end(); it++, lineIx++)
- {
- if ((*it)->_name == name)
- {
- matchFound = true;
- break;
- }
- }
-
- if (matchFound)
- {
- return (lineIx);
- }
- else
- {
- return (-1);
- }
-}
-
-
-void MatlabPlot::Append(int lineIndex, double x, double y)
-{
- CriticalSectionScoped cs(_critSect);
-
- if (!_enabled)
- {
- return;
- }
-
- // sanity for index
- if (lineIndex < 0 || lineIndex >= static_cast<int>(_line.size()))
- {
- throw "Line index out of range";
- exit(1);
- }
-
- return (_line[lineIndex]->Append(x, y));
-}
-
-
-void MatlabPlot::Append(int lineIndex, double y)
-{
- CriticalSectionScoped cs(_critSect);
-
- if (!_enabled)
- {
- return;
- }
-
- // sanity for index
- if (lineIndex < 0 || lineIndex >= static_cast<int>(_line.size()))
- {
- throw "Line index out of range";
- exit(1);
- }
-
- return (_line[lineIndex]->Append(y));
-}
-
-
-int MatlabPlot::Append(const char *name, double x, double y)
-{
- CriticalSectionScoped cs(_critSect);
-
- if (!_enabled)
- {
- return -1;
- }
-
- // search the list for a matching line name
- int lineIx = GetLineIx(name);
-
- if (lineIx < 0) //(!matchFound)
- {
- // no match; append new line
- lineIx = AddLine(-1, NULL, name);
- }
-
- // append data to line
- Append(lineIx, x, y);
- return (lineIx);
-}
-
-int MatlabPlot::Append(const char *name, double y)
-{
- CriticalSectionScoped cs(_critSect);
-
- if (!_enabled)
- {
- return -1;
- }
-
- // search the list for a matching line name
- int lineIx = GetLineIx(name);
-
- if (lineIx < 0) //(!matchFound)
- {
- // no match; append new line
- lineIx = AddLine(-1, NULL, name);
- }
-
- // append data to line
- Append(lineIx, y);
- return (lineIx);
-}
-
-int MatlabPlot::Length(char *name)
-{
- CriticalSectionScoped cs(_critSect);
-
- if (!_enabled)
- {
- return -1;
- }
-
- int ix = GetLineIx(name);
- if (ix >= 0)
- {
- return (static_cast<int>(_line[ix]->_xData.size()));
- }
- else
- {
- return (-1);
- }
-}
-
-
-void MatlabPlot::SetPlotAttribute(char *name, char *plotAttrib)
-{
- CriticalSectionScoped cs(_critSect);
-
- if (!_enabled)
- {
- return;
- }
-
- int lineIx = GetLineIx(name);
-
- if (lineIx >= 0)
- {
- _line[lineIx]->SetAttribute(plotAttrib);
- }
-}
-
-// Must be called under critical section _critSect
-void MatlabPlot::UpdateData(Engine* ep)
-{
- if (!_enabled)
- {
- return;
- }
-
- for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
- {
- mxArray* xData = NULL;
- mxArray* yData = NULL;
- (*it)->GetPlotData(&xData, &yData);
- if (xData != NULL)
- {
- std::string xName = (*it)->GetXName();
- std::string yName = (*it)->GetYName();
- _critSect->Leave();
-#ifdef MATLAB6
- mxSetName(xData, xName.c_str());
- mxSetName(yData, yName.c_str());
- engPutArray(ep, xData);
- engPutArray(ep, yData);
-#else
- int ret = engPutVariable(ep, xName.c_str(), xData);
- assert(ret == 0);
- ret = engPutVariable(ep, yName.c_str(), yData);
- assert(ret == 0);
-#endif
- _critSect->Enter();
- }
- }
-}
-
-bool MatlabPlot::GetPlotCmd(std::ostringstream & cmd, Engine* ep)
-{
- _critSect->Enter();
-
- if (!DataAvailable())
- {
- return false;
- }
-
- if (_firstPlot)
- {
- GetPlotCmd(cmd);
- _firstPlot = false;
- }
- else
- {
- GetRefreshCmd(cmd);
- }
-
- UpdateData(ep);
-
- _critSect->Leave();
-
- return true;
-}
-
-// Call inside critsect
-void MatlabPlot::GetPlotCmd(std::ostringstream & cmd)
-{
- // we have something to plot
- // empty the stream
- cmd.str(""); // (this seems to be the only way)
-
- cmd << "figure; h" << _figHandle << "= plot(";
-
- // first line
- std::vector<MatlabLine*>::iterator it = _line.begin();
- cmd << (*it)->GetPlotString();
-
- it++;
-
- // remaining lines
- for (; it != _line.end(); it++)
- {
- cmd << ", ";
- cmd << (*it)->GetPlotString();
- }
-
- cmd << "); ";
-
- if (_legendEnabled)
- {
- GetLegendCmd(cmd);
- }
-
- if (_smartAxis)
- {
- double xMin = _xlim[0];
- double xMax = _xlim[1];
- double yMax = _ylim[1];
- for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
- {
- xMax = std::max(xMax, (*it)->xMax());
- xMin = std::min(xMin, (*it)->xMin());
-
- yMax = std::max(yMax, (*it)->yMax());
- yMax = std::max(yMax, fabs((*it)->yMin()));
- }
- _xlim[0] = xMin;
- _xlim[1] = xMax;
- _ylim[0] = -yMax;
- _ylim[1] = yMax;
-
- cmd << "axis([" << _xlim[0] << ", " << _xlim[1] << ", " << _ylim[0] << ", " << _ylim[1] << "]);";
- }
-
- int i=1;
- for (it = _line.begin(); it != _line.end(); i++, it++)
- {
- cmd << "set(h" << _figHandle << "(" << i << "), 'Tag', " << (*it)->GetLegendString() << ");";
- }
-}
-
-// Call inside critsect
-void MatlabPlot::GetRefreshCmd(std::ostringstream & cmd)
-{
- cmd.str(""); // (this seems to be the only way)
- std::vector<MatlabLine*>::iterator it = _line.begin();
- for (it = _line.begin(); it != _line.end(); it++)
- {
- cmd << "h = findobj(0, 'Tag', " << (*it)->GetLegendString() << ");";
- cmd << (*it)->GetRefreshString();
- }
- //if (_legendEnabled)
- //{
- // GetLegendCmd(cmd);
- //}
-}
-
-void MatlabPlot::GetLegendCmd(std::ostringstream & cmd)
-{
- std::vector<MatlabLine*>::iterator it = _line.begin();
- bool anyLegend = false;
- for (; it != _line.end(); it++)
- {
- anyLegend = anyLegend || (*it)->hasLegend();
- }
- if (anyLegend)
- {
- // create the legend
-
- cmd << "legend(h" << _figHandle << ",{";
-
-
- // iterate lines
- int i = 0;
- for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
- {
- if (i > 0)
- {
- cmd << ", ";
- }
- cmd << (*it)->GetLegendString();
- i++;
- }
-
- cmd << "}, 2); "; // place legend in upper-left corner
- }
-}
-
-// Call inside critsect
-bool MatlabPlot::DataAvailable()
-{
- if (!_enabled)
- {
- return false;
- }
-
- for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
- {
- (*it)->PurgeOldData();
- }
-
- return true;
-}
-
-void MatlabPlot::Plot()
-{
- CriticalSectionScoped cs(_critSect);
-
- _timeToPlot = true;
-
-#ifdef PLOT_TESTING
- _plotStartTime = TickTime::MillisecondTimestamp();
-#endif
-}
-
-
-void MatlabPlot::Reset()
-{
- CriticalSectionScoped cs(_critSect);
-
- _enabled = true;
-
- for (std::vector<MatlabLine*>::iterator it = _line.begin(); it != _line.end(); it++)
- {
- (*it)->Reset();
- }
-
-}
-
-void MatlabPlot::SetFigHandle(int handle)
-{
- CriticalSectionScoped cs(_critSect);
-
- if (handle > 0)
- _figHandle = handle;
-}
-
-bool
-MatlabPlot::TimeToPlot()
-{
- CriticalSectionScoped cs(_critSect);
- return _enabled && _timeToPlot;
-}
-
-void
-MatlabPlot::Plotting()
-{
- CriticalSectionScoped cs(_critSect);
- _plotting = true;
-}
-
-void
-MatlabPlot::DonePlotting()
-{
- CriticalSectionScoped cs(_critSect);
- _timeToPlot = false;
- _plotting = false;
- _donePlottingEvent->Set();
-}
-
-void
-MatlabPlot::DisablePlot()
-{
- _critSect->Enter();
- while (_plotting)
- {
- _critSect->Leave();
- _donePlottingEvent->Wait(WEBRTC_EVENT_INFINITE);
- _critSect->Enter();
- }
- _enabled = false;
-}
-
-int MatlabPlot::MakeTrend(const char *sourceName, const char *trendName, double slope, double offset, const char *plotAttrib)
-{
- CriticalSectionScoped cs(_critSect);
-
- int sourceIx;
- int trendIx;
-
- sourceIx = GetLineIx(sourceName);
- if (sourceIx < 0)
- {
- // could not find source
- return (-1);
- }
-
- trendIx = GetLineIx(trendName);
- if (trendIx < 0)
- {
- // no trend found; add new line
- trendIx = AddLine(2 /*maxLen*/, plotAttrib, trendName);
- }
-
- _line[trendIx]->UpdateTrendLine(_line[sourceIx], slope, offset);
-
- return (trendIx);
-
-}
-
-
-MatlabEngine::MatlabEngine()
-:
-_critSect(CriticalSectionWrapper::CreateCriticalSection()),
-_eventPtr(NULL),
-_plotThread(NULL),
-_running(false),
-_numPlots(0)
-{
- _eventPtr = EventWrapper::Create();
-
- _plotThread = ThreadWrapper::CreateThread(MatlabEngine::PlotThread, this, kLowPriority, "MatlabPlot");
-
- if (_plotThread == NULL)
- {
- throw "Unable to start MatlabEngine thread";
- exit(1);
- }
-
- _running = true;
-
- unsigned int tid;
- _plotThread->Start(tid);
-
-}
-
-MatlabEngine::~MatlabEngine()
-{
- _critSect->Enter();
-
- if (_plotThread)
- {
- _plotThread->SetNotAlive();
- _running = false;
- _eventPtr->Set();
-
- while (!_plotThread->Stop())
- {
- ;
- }
-
- delete _plotThread;
- }
-
- _plots.clear();
-
- _plotThread = NULL;
-
- delete _eventPtr;
- _eventPtr = NULL;
-
- _critSect->Leave();
- delete _critSect;
-
-}
-
-MatlabPlot * MatlabEngine::NewPlot(MatlabPlot *newPlot)
-{
- CriticalSectionScoped cs(_critSect);
-
- //MatlabPlot *newPlot = new MatlabPlot();
-
- if (newPlot)
- {
- newPlot->SetFigHandle(++_numPlots); // first plot is number 1
- _plots.push_back(newPlot);
- }
-
- return (newPlot);
-
-}
-
-
-void MatlabEngine::DeletePlot(MatlabPlot *plot)
-{
- CriticalSectionScoped cs(_critSect);
-
- if (plot == NULL)
- {
- return;
- }
-
- std::vector<MatlabPlot *>::iterator it;
- for (it = _plots.begin(); it < _plots.end(); it++)
- {
- if (plot == *it)
- {
- break;
- }
- }
-
- assert (plot == *it);
-
- (*it)->DisablePlot();
-
- _plots.erase(it);
- --_numPlots;
-
- delete plot;
-}
-
-
-bool MatlabEngine::PlotThread(void *obj)
-{
- if (!obj)
- {
- return (false);
- }
-
- MatlabEngine *eng = (MatlabEngine *) obj;
-
- Engine *ep = engOpen(NULL);
- if (!ep)
- {
- throw "Cannot open Matlab engine";
- return (false);
- }
-
- engSetVisible(ep, true);
- engEvalString(ep, "close all;");
-
- while (eng->_running)
- {
- eng->_critSect->Enter();
-
- // iterate through all plots
- for (unsigned int ix = 0; ix < eng->_plots.size(); ix++)
- {
- MatlabPlot *plot = eng->_plots[ix];
- if (plot->TimeToPlot())
- {
- plot->Plotting();
- eng->_critSect->Leave();
- std::ostringstream cmd;
-
- if (engEvalString(ep, cmd.str().c_str()))
- {
- // engine dead
- return (false);
- }
-
- // empty the stream
- cmd.str(""); // (this seems to be the only way)
- if (plot->GetPlotCmd(cmd, ep))
- {
- // things to plot, we have already accessed what we need in the plot
- plot->DonePlotting();
-
- WebRtc_Word64 start = TickTime::MillisecondTimestamp();
- // plot it
- int ret = engEvalString(ep, cmd.str().c_str());
- printf("time=%I64i\n", TickTime::MillisecondTimestamp() - start);
- if (ret)
- {
- // engine dead
- return (false);
- }
-
-#ifdef PLOT_TESTING
- if(plot->_plotStartTime >= 0)
- {
- plot->_plotDelay = TickTime::MillisecondTimestamp() - plot->_plotStartTime;
- plot->_plotStartTime = -1;
- }
-#endif
- }
- eng->_critSect->Enter();
- }
- }
-
- eng->_critSect->Leave();
- // wait a while
- eng->_eventPtr->Wait(66); // 33 ms
- }
-
- if (ep)
- {
- engClose(ep);
- ep = NULL;
- }
-
- return (true);
-
-}
-
-#endif // MATLAB
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h
deleted file mode 100644
index 08c7006..0000000
--- a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/MatlabPlot.h
+++ /dev/null
@@ -1,170 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_MATLABPLOT_H_
-#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_MATLABPLOT_H_
-
-#include <list>
-#include <string>
-#include <vector>
-
-#include "typedefs.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-class ThreadWrapper;
-}
-
-//#define PLOT_TESTING
-
-#ifdef MATLAB
-
-typedef struct engine Engine;
-typedef struct mxArray_tag mxArray;
-
-class MatlabLine
-{
- friend class MatlabPlot;
-
-public:
- MatlabLine(int maxLen = -1, const char *plotAttrib = NULL, const char *name = NULL);
- ~MatlabLine();
- virtual void Append(double x, double y);
- virtual void Append(double y);
- void SetMaxLen(int maxLen);
- void SetAttribute(char *plotAttrib);
- void SetName(char *name);
- void Reset();
- virtual void PurgeOldData() {};
-
- void UpdateTrendLine(MatlabLine * sourceData, double slope, double offset);
-
- double xMin();
- double xMax();
- double yMin();
- double yMax();
-
-protected:
- void GetPlotData(mxArray** xData, mxArray** yData);
- std::string GetXName();
- std::string GetYName();
- std::string GetPlotString();
- std::string GetRefreshString();
- std::string GetLegendString();
- bool hasLegend();
- std::list<double> _xData;
- std::list<double> _yData;
- mxArray* _xArray;
- mxArray* _yArray;
- int _maxLen;
- std::string _plotAttribute;
- std::string _name;
-};
-
-
-class MatlabTimeLine : public MatlabLine
-{
-public:
- MatlabTimeLine(int horizonSeconds = -1, const char *plotAttrib = NULL, const char *name = NULL,
- WebRtc_Word64 refTimeMs = -1);
- ~MatlabTimeLine() {};
- void Append(double y);
- void PurgeOldData();
- WebRtc_Word64 GetRefTime();
-
-private:
- WebRtc_Word64 _refTimeMs;
- int _timeHorizon;
-};
-
-
-class MatlabPlot
-{
- friend class MatlabEngine;
-
-public:
- MatlabPlot();
- ~MatlabPlot();
-
- int AddLine(int maxLen = -1, const char *plotAttrib = NULL, const char *name = NULL);
- int AddTimeLine(int maxLen = -1, const char *plotAttrib = NULL, const char *name = NULL,
- WebRtc_Word64 refTimeMs = -1);
- int GetLineIx(const char *name);
- void Append(int lineIndex, double x, double y);
- void Append(int lineIndex, double y);
- int Append(const char *name, double x, double y);
- int Append(const char *name, double y);
- int Length(char *name);
- void SetPlotAttribute(char *name, char *plotAttrib);
- void Plot();
- void Reset();
- void SmartAxis(bool status = true) { _smartAxis = status; };
- void SetFigHandle(int handle);
- void EnableLegend(bool enable) { _legendEnabled = enable; };
-
- bool TimeToPlot();
- void Plotting();
- void DonePlotting();
- void DisablePlot();
-
- int MakeTrend(const char *sourceName, const char *trendName, double slope, double offset, const char *plotAttrib = NULL);
-
-#ifdef PLOT_TESTING
- WebRtc_Word64 _plotStartTime;
- WebRtc_Word64 _plotDelay;
-#endif
-
-private:
- void UpdateData(Engine* ep);
- bool GetPlotCmd(std::ostringstream & cmd, Engine* ep);
- void GetPlotCmd(std::ostringstream & cmd); // call inside crit sect
- void GetRefreshCmd(std::ostringstream & cmd); // call inside crit sect
- void GetLegendCmd(std::ostringstream & cmd);
- bool DataAvailable();
-
- std::vector<MatlabLine *> _line;
- int _figHandle;
- bool _smartAxis;
- double _xlim[2];
- double _ylim[2];
- webrtc::CriticalSectionWrapper *_critSect;
- bool _timeToPlot;
- bool _plotting;
- bool _enabled;
- bool _firstPlot;
- bool _legendEnabled;
- webrtc::EventWrapper* _donePlottingEvent;
-};
-
-
-class MatlabEngine
-{
-public:
- MatlabEngine();
- ~MatlabEngine();
-
- MatlabPlot * NewPlot(MatlabPlot *newPlot);
- void DeletePlot(MatlabPlot *plot);
-
-private:
- static bool PlotThread(void *obj);
-
- std::vector<MatlabPlot *> _plots;
- webrtc::CriticalSectionWrapper *_critSect;
- webrtc::EventWrapper *_eventPtr;
- webrtc::ThreadWrapper* _plotThread;
- bool _running;
- int _numPlots;
-};
-
-#endif //MATLAB
-
-#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_MATLABPLOT_H_
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc
deleted file mode 100644
index d322242..0000000
--- a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.cc
+++ /dev/null
@@ -1,438 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <algorithm> // for max function
-#include <stdio.h>
-
-#include "TestLoadGenerator.h"
-#include "TestSenderReceiver.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "critical_section_wrapper.h"
-#include "tick_util.h"
-
-
-bool SenderThreadFunction(void *obj)
-{
- if (obj == NULL)
- {
- return false;
- }
- TestLoadGenerator *_genObj = static_cast<TestLoadGenerator *>(obj);
-
- return _genObj->GeneratorLoop();
-}
-
-
-TestLoadGenerator::TestLoadGenerator(TestSenderReceiver *sender, WebRtc_Word32 rtpSampleRate)
-:
-_critSect(CriticalSectionWrapper::CreateCriticalSection()),
-_eventPtr(NULL),
-_genThread(NULL),
-_bitrateKbps(0),
-_sender(sender),
-_running(false),
-_rtpSampleRate(rtpSampleRate)
-{
-}
-
-TestLoadGenerator::~TestLoadGenerator ()
-{
- if (_running)
- {
- Stop();
- }
-
- delete _critSect;
-}
-
-WebRtc_Word32 TestLoadGenerator::SetBitrate (WebRtc_Word32 newBitrateKbps)
-{
- CriticalSectionScoped cs(_critSect);
-
- if (newBitrateKbps < 0)
- {
- return -1;
- }
-
- _bitrateKbps = newBitrateKbps;
-
- printf("New bitrate = %i kbps\n", _bitrateKbps);
-
- return _bitrateKbps;
-}
-
-
-WebRtc_Word32 TestLoadGenerator::Start (const char *threadName)
-{
- CriticalSectionScoped cs(_critSect);
-
- _eventPtr = EventWrapper::Create();
-
- _genThread = ThreadWrapper::CreateThread(SenderThreadFunction, this, kRealtimePriority, threadName);
- if (_genThread == NULL)
- {
- throw "Unable to start generator thread";
- exit(1);
- }
-
- _running = true;
-
- unsigned int tid;
- _genThread->Start(tid);
-
- return 0;
-}
-
-
-WebRtc_Word32 TestLoadGenerator::Stop ()
-{
- _critSect.Enter();
-
- if (_genThread)
- {
- _genThread->SetNotAlive();
- _running = false;
- _eventPtr->Set();
-
- while (!_genThread->Stop())
- {
- _critSect.Leave();
- _critSect.Enter();
- }
-
- delete _genThread;
- _genThread = NULL;
-
- delete _eventPtr;
- _eventPtr = NULL;
- }
-
- _genThread = NULL;
- _critSect.Leave();
- return (0);
-}
-
-
-int TestLoadGenerator::generatePayload ()
-{
- return(generatePayload( static_cast<WebRtc_UWord32>( TickTime::MillisecondTimestamp() * _rtpSampleRate / 1000 )));
-}
-
-
-int TestLoadGenerator::sendPayload (const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const webrtc::FrameType frameType /*= webrtc::kVideoFrameDelta*/)
-{
-
- return (_sender->SendOutgoingData(timeStamp, payloadData, payloadSize, frameType));
-}
-
-
-CBRGenerator::CBRGenerator (TestSenderReceiver *sender, WebRtc_Word32 payloadSizeBytes, WebRtc_Word32 bitrateKbps, WebRtc_Word32 rtpSampleRate)
-:
-//_eventPtr(NULL),
-_payloadSizeBytes(payloadSizeBytes),
-_payload(new WebRtc_UWord8[payloadSizeBytes]),
-TestLoadGenerator(sender, rtpSampleRate)
-{
- SetBitrate (bitrateKbps);
-}
-
-CBRGenerator::~CBRGenerator ()
-{
- if (_running)
- {
- Stop();
- }
-
- if (_payload)
- {
- delete [] _payload;
- }
-
-}
-
-bool CBRGenerator::GeneratorLoop ()
-{
- double periodMs;
- WebRtc_Word64 nextSendTime = TickTime::MillisecondTimestamp();
-
-
- // no critSect
- while (_running)
- {
- // send data (critSect inside)
- generatePayload( static_cast<WebRtc_UWord32>(nextSendTime * _rtpSampleRate / 1000) );
-
- // calculate wait time
- periodMs = 8.0 * _payloadSizeBytes / ( _bitrateKbps );
-
- nextSendTime = static_cast<WebRtc_Word64>(nextSendTime + periodMs);
-
- WebRtc_Word32 waitTime = static_cast<WebRtc_Word32>(nextSendTime - TickTime::MillisecondTimestamp());
- if (waitTime < 0)
- {
- waitTime = 0;
- }
- // wait
- _eventPtr->Wait(static_cast<WebRtc_Word32>(waitTime));
- }
-
- return true;
-}
-
-int CBRGenerator::generatePayload ( WebRtc_UWord32 timestamp )
-{
- CriticalSectionScoped cs(_critSect);
-
- //WebRtc_UWord8 *payload = new WebRtc_UWord8[_payloadSizeBytes];
-
- int ret = sendPayload(timestamp, _payload, _payloadSizeBytes);
-
- //delete [] payload;
- return ret;
-}
-
-
-
-
-/////////////////////
-
-CBRFixFRGenerator::CBRFixFRGenerator (TestSenderReceiver *sender, WebRtc_Word32 bitrateKbps,
- WebRtc_Word32 rtpSampleRate, WebRtc_Word32 frameRateFps /*= 30*/,
- double spread /*= 0.0*/)
-:
-//_eventPtr(NULL),
-_payloadSizeBytes(0),
-_payload(NULL),
-_payloadAllocLen(0),
-_frameRateFps(frameRateFps),
-_spreadFactor(spread),
-TestLoadGenerator(sender, rtpSampleRate)
-{
- SetBitrate (bitrateKbps);
-}
-
-CBRFixFRGenerator::~CBRFixFRGenerator ()
-{
- if (_running)
- {
- Stop();
- }
-
- if (_payload)
- {
- delete [] _payload;
- _payloadAllocLen = 0;
- }
-
-}
-
-bool CBRFixFRGenerator::GeneratorLoop ()
-{
- double periodMs;
- WebRtc_Word64 nextSendTime = TickTime::MillisecondTimestamp();
-
- _critSect.Enter();
-
- if (_frameRateFps <= 0)
- {
- return false;
- }
-
- _critSect.Leave();
-
- // no critSect
- while (_running)
- {
- _critSect.Enter();
-
- // calculate payload size
- _payloadSizeBytes = nextPayloadSize();
-
- if (_payloadSizeBytes > 0)
- {
-
- if (_payloadAllocLen < _payloadSizeBytes * (1 + _spreadFactor))
- {
- // re-allocate _payload
- if (_payload)
- {
- delete [] _payload;
- _payload = NULL;
- }
-
- _payloadAllocLen = static_cast<WebRtc_Word32>((_payloadSizeBytes * (1 + _spreadFactor) * 3) / 2 + .5); // 50% extra to avoid frequent re-alloc
- _payload = new WebRtc_UWord8[_payloadAllocLen];
- }
-
-
- // send data (critSect inside)
- generatePayload( static_cast<WebRtc_UWord32>(nextSendTime * _rtpSampleRate / 1000) );
- }
-
- _critSect.Leave();
-
- // calculate wait time
- periodMs = 1000.0 / _frameRateFps;
- nextSendTime = static_cast<WebRtc_Word64>(nextSendTime + periodMs + 0.5);
-
- WebRtc_Word32 waitTime = static_cast<WebRtc_Word32>(nextSendTime - TickTime::MillisecondTimestamp());
- if (waitTime < 0)
- {
- waitTime = 0;
- }
- // wait
- _eventPtr->Wait(waitTime);
- }
-
- return true;
-}
-
-WebRtc_Word32 CBRFixFRGenerator::nextPayloadSize()
-{
- const double periodMs = 1000.0 / _frameRateFps;
- return static_cast<WebRtc_Word32>(_bitrateKbps * periodMs / 8 + 0.5);
-}
-
-int CBRFixFRGenerator::generatePayload ( WebRtc_UWord32 timestamp )
-{
- CriticalSectionScoped cs(_critSect);
-
- double factor = ((double) rand() - RAND_MAX/2) / RAND_MAX; // [-0.5; 0.5]
- factor = 1 + 2 * _spreadFactor * factor; // [1 - _spreadFactor ; 1 + _spreadFactor]
-
- WebRtc_Word32 thisPayloadBytes = static_cast<WebRtc_Word32>(_payloadSizeBytes * factor);
- // sanity
- if (thisPayloadBytes > _payloadAllocLen)
- {
- thisPayloadBytes = _payloadAllocLen;
- }
-
- int ret = sendPayload(timestamp, _payload, thisPayloadBytes);
- return ret;
-}
-
-
-/////////////////////
-
-PeriodicKeyFixFRGenerator::PeriodicKeyFixFRGenerator (TestSenderReceiver *sender, WebRtc_Word32 bitrateKbps,
- WebRtc_Word32 rtpSampleRate, WebRtc_Word32 frameRateFps /*= 30*/,
- double spread /*= 0.0*/, double keyFactor /*= 4.0*/, WebRtc_UWord32 keyPeriod /*= 300*/)
-:
-_keyFactor(keyFactor),
-_keyPeriod(keyPeriod),
-_frameCount(0),
-CBRFixFRGenerator(sender, bitrateKbps, rtpSampleRate, frameRateFps, spread)
-{
-}
-
-WebRtc_Word32 PeriodicKeyFixFRGenerator::nextPayloadSize()
-{
- // calculate payload size for a delta frame
- WebRtc_Word32 payloadSizeBytes = static_cast<WebRtc_Word32>(1000 * _bitrateKbps / (8.0 * _frameRateFps * (1.0 + (_keyFactor - 1.0) / _keyPeriod)) + 0.5);
-
- if (_frameCount % _keyPeriod == 0)
- {
- // this is a key frame, scale the payload size
- payloadSizeBytes = static_cast<WebRtc_Word32>(_keyFactor * _payloadSizeBytes + 0.5);
- }
- _frameCount++;
-
- return payloadSizeBytes;
-}
-
-////////////////////
-
-CBRVarFRGenerator::CBRVarFRGenerator(TestSenderReceiver *sender, WebRtc_Word32 bitrateKbps, const WebRtc_UWord8* frameRates,
- WebRtc_UWord16 numFrameRates, WebRtc_Word32 rtpSampleRate, double avgFrPeriodMs,
- double frSpreadFactor, double spreadFactor)
-:
-_avgFrPeriodMs(avgFrPeriodMs),
-_frSpreadFactor(frSpreadFactor),
-_frameRates(NULL),
-_numFrameRates(numFrameRates),
-_frChangeTimeMs(TickTime::MillisecondTimestamp() + _avgFrPeriodMs),
-CBRFixFRGenerator(sender, bitrateKbps, rtpSampleRate, frameRates[0], spreadFactor)
-{
- _frameRates = new WebRtc_UWord8[_numFrameRates];
- memcpy(_frameRates, frameRates, _numFrameRates);
-}
-
-CBRVarFRGenerator::~CBRVarFRGenerator()
-{
- delete [] _frameRates;
-}
-
-void CBRVarFRGenerator::ChangeFrameRate()
-{
- const WebRtc_Word64 nowMs = TickTime::MillisecondTimestamp();
- if (nowMs < _frChangeTimeMs)
- {
- return;
- }
- // Time to change frame rate
- WebRtc_UWord16 frIndex = static_cast<WebRtc_UWord16>(static_cast<double>(rand()) / RAND_MAX
- * (_numFrameRates - 1) + 0.5) ;
- assert(frIndex < _numFrameRates);
- _frameRateFps = _frameRates[frIndex];
- // Update the next frame rate change time
- double factor = ((double) rand() - RAND_MAX/2) / RAND_MAX; // [-0.5; 0.5]
- factor = 1 + 2 * _frSpreadFactor * factor; // [1 - _frSpreadFactor ; 1 + _frSpreadFactor]
- _frChangeTimeMs = nowMs + static_cast<WebRtc_Word64>(1000.0 * factor *
- _avgFrPeriodMs + 0.5);
-
- printf("New frame rate: %d\n", _frameRateFps);
-}
-
-WebRtc_Word32 CBRVarFRGenerator::nextPayloadSize()
-{
- ChangeFrameRate();
- return CBRFixFRGenerator::nextPayloadSize();
-}
-
-////////////////////
-
-CBRFrameDropGenerator::CBRFrameDropGenerator(TestSenderReceiver *sender, WebRtc_Word32 bitrateKbps,
- WebRtc_Word32 rtpSampleRate, double spreadFactor)
-:
-_accBits(0),
-CBRFixFRGenerator(sender, bitrateKbps, rtpSampleRate, 30, spreadFactor)
-{
-}
-
-CBRFrameDropGenerator::~CBRFrameDropGenerator()
-{
-}
-
-WebRtc_Word32 CBRFrameDropGenerator::nextPayloadSize()
-{
- _accBits -= 1000 * _bitrateKbps / _frameRateFps;
- if (_accBits < 0)
- {
- _accBits = 0;
- }
- if (_accBits > 0.3 * _bitrateKbps * 1000)
- {
- //printf("drop\n");
- return 0;
- }
- else
- {
- //printf("keep\n");
- const double periodMs = 1000.0 / _frameRateFps;
- WebRtc_Word32 frameSize = static_cast<WebRtc_Word32>(_bitrateKbps * periodMs / 8 + 0.5);
- frameSize = std::max(frameSize, static_cast<WebRtc_Word32>(300 * periodMs / 8 + 0.5));
- _accBits += frameSize * 8;
- return frameSize;
- }
-}
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
deleted file mode 100644
index c22591c..0000000
--- a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestLoadGenerator.h
+++ /dev/null
@@ -1,146 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTLOADGENERATOR_H_
-#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTLOADGENERATOR_H_
-
-#include <stdlib.h>
-
-#include "typedefs.h"
-#include "module_common_types.h"
-
-class TestSenderReceiver;
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-class ThreadWrapper;
-}
-
-class TestLoadGenerator
-{
-public:
- TestLoadGenerator (TestSenderReceiver *sender, WebRtc_Word32 rtpSampleRate = 90000);
- virtual ~TestLoadGenerator ();
-
- WebRtc_Word32 SetBitrate (WebRtc_Word32 newBitrateKbps);
- virtual WebRtc_Word32 Start (const char *threadName = NULL);
- virtual WebRtc_Word32 Stop ();
- virtual bool GeneratorLoop () = 0;
-
-protected:
- virtual int generatePayload ( WebRtc_UWord32 timestamp ) = 0;
- int generatePayload ();
- int sendPayload (const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const webrtc::FrameType frameType = webrtc::kVideoFrameDelta);
-
- webrtc::CriticalSectionWrapper* _critSect;
- webrtc::EventWrapper *_eventPtr;
- webrtc::ThreadWrapper* _genThread;
- WebRtc_Word32 _bitrateKbps;
- TestSenderReceiver *_sender;
- bool _running;
- WebRtc_Word32 _rtpSampleRate;
-};
-
-
-class CBRGenerator : public TestLoadGenerator
-{
-public:
- CBRGenerator (TestSenderReceiver *sender, WebRtc_Word32 payloadSizeBytes, WebRtc_Word32 bitrateKbps, WebRtc_Word32 rtpSampleRate = 90000);
- virtual ~CBRGenerator ();
-
- virtual WebRtc_Word32 Start () {return (TestLoadGenerator::Start("CBRGenerator"));};
-
- virtual bool GeneratorLoop ();
-
-protected:
- virtual int generatePayload ( WebRtc_UWord32 timestamp );
-
- WebRtc_Word32 _payloadSizeBytes;
- WebRtc_UWord8 *_payload;
-};
-
-
-class CBRFixFRGenerator : public TestLoadGenerator // constant bitrate and fixed frame rate
-{
-public:
- CBRFixFRGenerator (TestSenderReceiver *sender, WebRtc_Word32 bitrateKbps, WebRtc_Word32 rtpSampleRate = 90000,
- WebRtc_Word32 frameRateFps = 30, double spread = 0.0);
- virtual ~CBRFixFRGenerator ();
-
- virtual WebRtc_Word32 Start () {return (TestLoadGenerator::Start("CBRFixFRGenerator"));};
-
- virtual bool GeneratorLoop ();
-
-protected:
- virtual WebRtc_Word32 nextPayloadSize ();
- virtual int generatePayload ( WebRtc_UWord32 timestamp );
-
- WebRtc_Word32 _payloadSizeBytes;
- WebRtc_UWord8 *_payload;
- WebRtc_Word32 _payloadAllocLen;
- WebRtc_Word32 _frameRateFps;
- double _spreadFactor;
-};
-
-class PeriodicKeyFixFRGenerator : public CBRFixFRGenerator // constant bitrate and fixed frame rate with periodically large frames
-{
-public:
- PeriodicKeyFixFRGenerator (TestSenderReceiver *sender, WebRtc_Word32 bitrateKbps, WebRtc_Word32 rtpSampleRate = 90000,
- WebRtc_Word32 frameRateFps = 30, double spread = 0.0, double keyFactor = 4.0, WebRtc_UWord32 keyPeriod = 300);
- virtual ~PeriodicKeyFixFRGenerator () {}
-
-protected:
- virtual WebRtc_Word32 nextPayloadSize ();
-
- double _keyFactor;
- WebRtc_UWord32 _keyPeriod;
- WebRtc_UWord32 _frameCount;
-};
-
-// Probably better to inherit CBRFixFRGenerator from CBRVarFRGenerator, but since
-// the fix FR version already existed this was easier.
-class CBRVarFRGenerator : public CBRFixFRGenerator // constant bitrate and variable frame rate
-{
-public:
- CBRVarFRGenerator(TestSenderReceiver *sender, WebRtc_Word32 bitrateKbps, const WebRtc_UWord8* frameRates,
- WebRtc_UWord16 numFrameRates, WebRtc_Word32 rtpSampleRate = 90000, double avgFrPeriodMs = 5.0,
- double frSpreadFactor = 0.05, double spreadFactor = 0.0);
-
- ~CBRVarFRGenerator();
-
-protected:
- virtual void ChangeFrameRate();
- virtual WebRtc_Word32 nextPayloadSize ();
-
- double _avgFrPeriodMs;
- double _frSpreadFactor;
- WebRtc_UWord8* _frameRates;
- WebRtc_UWord16 _numFrameRates;
- WebRtc_Word64 _frChangeTimeMs;
-};
-
-class CBRFrameDropGenerator : public CBRFixFRGenerator // constant bitrate and variable frame rate
-{
-public:
- CBRFrameDropGenerator(TestSenderReceiver *sender, WebRtc_Word32 bitrateKbps,
- WebRtc_Word32 rtpSampleRate = 90000, double spreadFactor = 0.0);
-
- ~CBRFrameDropGenerator();
-
-protected:
- virtual WebRtc_Word32 nextPayloadSize();
-
- double _accBits;
-};
-
-#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTLOADGENERATOR_H_
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
deleted file mode 100644
index 1fc0fd3..0000000
--- a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.cc
+++ /dev/null
@@ -1,442 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-
-#include "rtp_rtcp.h"
-#include "udp_transport.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "tick_util.h"
-#include "critical_section_wrapper.h"
-#include "TestSenderReceiver.h"
-#include "TestLoadGenerator.h"
-#include <stdlib.h>
-
-#define NR_OF_SOCKET_BUFFERS 500
-
-
-bool ProcThreadFunction(void *obj)
-{
- if (obj == NULL)
- {
- return false;
- }
- TestSenderReceiver *theObj = static_cast<TestSenderReceiver *>(obj);
-
- return theObj->ProcLoop();
-}
-
-
-TestSenderReceiver::TestSenderReceiver (void)
-:
-_critSect(CriticalSectionWrapper::CreateCriticalSection()),
-_eventPtr(NULL),
-_procThread(NULL),
-_running(false),
-_payloadType(0),
-_loadGenerator(NULL),
-_isSender(false),
-_isReceiver(false),
-_timeOut(false),
-_sendRecCB(NULL),
-_lastBytesReceived(0),
-_lastTime(-1)
-{
- // RTP/RTCP module
- _rtp = RtpRtcp::CreateRtpRtcp(0, false);
- if (!_rtp)
- {
- throw "Could not create RTP/RTCP module";
- exit(1);
- }
-
- if (_rtp->InitReceiver() != 0)
- {
- throw "_rtp->InitReceiver()";
- exit(1);
- }
-
- if (_rtp->InitSender() != 0)
- {
- throw "_rtp->InitSender()";
- exit(1);
- }
-
- // SocketTransport module
- WebRtc_UWord8 numberOfThreads = 1;
- _transport = UdpTransport::Create(0, numberOfThreads);
- if (!_transport)
- {
- throw "Could not create transport module";
- exit(1);
- }
-}
-
-TestSenderReceiver::~TestSenderReceiver (void)
-{
-
- Stop(); // N.B. without critSect
-
- _critSect->Enter();
-
- if (_rtp)
- {
- RtpRtcp::DestroyRtpRtcp(_rtp);
- _rtp = NULL;
- }
-
- if (_transport)
- {
- UdpTransport::Destroy(_transport);
- _transport = NULL;
- }
-
- delete _critSect;
-
-}
-
-
-WebRtc_Word32 TestSenderReceiver::InitReceiver (const WebRtc_UWord16 rtpPort,
- const WebRtc_UWord16 rtcpPort,
- const WebRtc_Word8 payloadType /*= 127*/)
-{
- CriticalSectionScoped cs(_critSect);
-
- // init transport
- if (_transport->InitializeReceiveSockets(this, rtpPort/*, 0, NULL, 0, true*/) != 0)
- {
- throw "_transport->InitializeReceiveSockets";
- exit(1);
- }
-
- if (_rtp->RegisterIncomingRTPCallback(this) != 0)
- {
- throw "_rtp->RegisterIncomingRTPCallback";
- exit(1);
- }
-
- if (_rtp->RegisterIncomingDataCallback(this) != 0)
- {
- throw "_rtp->RegisterIncomingRTPCallback";
- exit(1);
- }
-
- if (_rtp->SetRTCPStatus(kRtcpNonCompound) != 0)
- {
- throw "_rtp->SetRTCPStatus";
- exit(1);
- }
-
- if (_rtp->SetTMMBRStatus(true) != 0)
- {
- throw "_rtp->SetTMMBRStatus";
- exit(1);
- }
-
- if (_rtp->RegisterReceivePayload("I420", payloadType, 90000) != 0)
- {
- throw "_rtp->RegisterReceivePayload";
- exit(1);
- }
-
- _isReceiver = true;
-
- return (0);
-}
-
-
-WebRtc_Word32 TestSenderReceiver::Start()
-{
- CriticalSectionScoped cs(_critSect);
-
- _eventPtr = EventWrapper::Create();
-
- if (_rtp->SetSendingStatus(true) != 0)
- {
- throw "_rtp->SetSendingStatus";
- exit(1);
- }
-
- _procThread = ThreadWrapper::CreateThread(ProcThreadFunction, this, kRealtimePriority, "TestSenderReceiver");
- if (_procThread == NULL)
- {
- throw "Unable to create process thread";
- exit(1);
- }
-
- _running = true;
-
- if (_isReceiver)
- {
- if (_transport->StartReceiving(NR_OF_SOCKET_BUFFERS) != 0)
- {
- throw "_transport->StartReceiving";
- exit(1);
- }
- }
-
- unsigned int tid;
- _procThread->Start(tid);
-
- return 0;
-
-}
-
-
-WebRtc_Word32 TestSenderReceiver::Stop ()
-{
- CriticalSectionScoped cs(_critSect);
-
- _transport->StopReceiving();
-
- if (_procThread)
- {
- _procThread->SetNotAlive();
- _running = false;
- _eventPtr->Set();
-
- while (!_procThread->Stop())
- {
- ;
- }
-
- delete _eventPtr;
-
- delete _procThread;
- }
-
- _procThread = NULL;
-
- return (0);
-}
-
-
-bool TestSenderReceiver::ProcLoop(void)
-{
-
- // process RTP/RTCP module
- _rtp->Process();
-
- // process SocketTransport module
- _transport->Process();
-
- // no critSect
- while (_running)
- {
- // ask RTP/RTCP module for wait time
- WebRtc_Word32 rtpWait = _rtp->TimeUntilNextProcess();
-
- // ask SocketTransport module for wait time
- WebRtc_Word32 tpWait = _transport->TimeUntilNextProcess();
-
- WebRtc_Word32 minWait = (rtpWait < tpWait) ? rtpWait: tpWait;
- minWait = (minWait > 0) ? minWait : 0;
- // wait
- _eventPtr->Wait(minWait);
-
- // process RTP/RTCP module
- _rtp->Process();
-
- // process SocketTransport module
- _transport->Process();
-
- }
-
- return true;
-}
-
-
-WebRtc_Word32 TestSenderReceiver::ReceiveBitrateKbps ()
-{
- WebRtc_UWord32 bytesSent;
- WebRtc_UWord32 packetsSent;
- WebRtc_UWord32 bytesReceived;
- WebRtc_UWord32 packetsReceived;
-
- if (_rtp->DataCountersRTP(&bytesSent, &packetsSent, &bytesReceived, &packetsReceived) == 0)
- {
- WebRtc_Word64 now = TickTime::MillisecondTimestamp();
- WebRtc_Word32 kbps = 0;
- if (now > _lastTime)
- {
- if (_lastTime > 0)
- {
- // 8 * bytes / ms = kbps
- kbps = static_cast<WebRtc_Word32>(
- (8 * (bytesReceived - _lastBytesReceived)) / (now - _lastTime));
- }
- _lastTime = now;
- _lastBytesReceived = bytesReceived;
- }
- return (kbps);
- }
-
- return (-1);
-}
-
-
-WebRtc_Word32 TestSenderReceiver::SetPacketTimeout(const WebRtc_UWord32 timeoutMS)
-{
- return (_rtp->SetPacketTimeout(timeoutMS, 0 /* RTCP timeout */));
-}
-
-
-void TestSenderReceiver::OnPacketTimeout(const WebRtc_Word32 id)
-{
- CriticalSectionScoped lock(_critSect);
-
- _timeOut = true;
-}
-
-
-void TestSenderReceiver::OnReceivedPacket(const WebRtc_Word32 id,
- const RtpRtcpPacketType packetType)
-{
- // do nothing
- //printf("OnReceivedPacket\n");
-
-}
-
-WebRtc_Word32 TestSenderReceiver::OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const webrtc::WebRtcRTPHeader* rtpHeader)
-{
- //printf("OnReceivedPayloadData\n");
- return (0);
-}
-
-
-void TestSenderReceiver::IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
- const WebRtc_Word32 rtpPacketLength,
- const WebRtc_Word8* fromIP,
- const WebRtc_UWord16 fromPort)
-{
- _rtp->IncomingPacket((WebRtc_UWord8 *) incomingRtpPacket, static_cast<WebRtc_UWord16>(rtpPacketLength));
-}
-
-
-
-void TestSenderReceiver::IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
- const WebRtc_Word32 rtcpPacketLength,
- const WebRtc_Word8* fromIP,
- const WebRtc_UWord16 fromPort)
-{
- _rtp->IncomingPacket((WebRtc_UWord8 *) incomingRtcpPacket, static_cast<WebRtc_UWord16>(rtcpPacketLength));
-}
-
-
-
-
-
-///////////////////
-
-
-WebRtc_Word32 TestSenderReceiver::InitSender (const WebRtc_UWord32 startBitrateKbps,
- const WebRtc_Word8* ipAddr,
- const WebRtc_UWord16 rtpPort,
- const WebRtc_UWord16 rtcpPort /*= 0*/,
- const WebRtc_Word8 payloadType /*= 127*/)
-{
- CriticalSectionScoped cs(_critSect);
-
- _payloadType = payloadType;
-
- // check load generator valid
- if (_loadGenerator)
- {
- _loadGenerator->SetBitrate(startBitrateKbps);
- }
-
- if (_rtp->RegisterSendTransport(_transport) != 0)
- {
- throw "_rtp->RegisterSendTransport";
- exit(1);
- }
- if (_rtp->RegisterSendPayload("I420", _payloadType, 90000) != 0)
- {
- throw "_rtp->RegisterSendPayload";
- exit(1);
- }
-
- if (_rtp->RegisterIncomingVideoCallback(this) != 0)
- {
- throw "_rtp->RegisterIncomingVideoCallback";
- exit(1);
- }
-
- if (_rtp->SetRTCPStatus(kRtcpNonCompound) != 0)
- {
- throw "_rtp->SetRTCPStatus";
- exit(1);
- }
-
- if (_rtp->SetSendBitrate(startBitrateKbps*1000, 0, MAX_BITRATE_KBPS) != 0)
- {
- throw "_rtp->SetSendBitrate";
- exit(1);
- }
-
-
- // SocketTransport
- if (_transport->InitializeSendSockets(ipAddr, rtpPort, rtcpPort))
- {
- throw "_transport->InitializeSendSockets";
- exit(1);
- }
-
- _isSender = true;
-
- return (0);
-}
-
-
-
-WebRtc_Word32
-TestSenderReceiver::SendOutgoingData(const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const webrtc::FrameType frameType /*= webrtc::kVideoFrameDelta*/)
-{
- return (_rtp->SendOutgoingData(frameType, _payloadType, timeStamp, payloadData, payloadSize));
-}
-
-
-WebRtc_Word32 TestSenderReceiver::SetLoadGenerator(TestLoadGenerator *generator)
-{
- CriticalSectionScoped cs(_critSect);
-
- _loadGenerator = generator;
- return(0);
-
-}
-
-void TestSenderReceiver::OnNetworkChanged(const WebRtc_Word32 id,
- const WebRtc_UWord32 minBitrateBps,
- const WebRtc_UWord32 maxBitrateBps,
- const WebRtc_UWord8 fractionLost,
- const WebRtc_UWord16 roundTripTimeMs,
- const WebRtc_UWord16 bwEstimateKbitMin,
- const WebRtc_UWord16 bwEstimateKbitMax)
-{
- if (_loadGenerator)
- {
- _loadGenerator->SetBitrate(maxBitrateBps/1000);
- }
-
- if (_sendRecCB)
- {
- _sendRecCB->OnOnNetworkChanged(maxBitrateBps,
- fractionLost,
- roundTripTimeMs,
- bwEstimateKbitMin,
- bwEstimateKbitMax);
- }
-}
diff --git a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h b/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
deleted file mode 100644
index 7f7f2f0..0000000
--- a/trunk/src/modules/rtp_rtcp/test/BWEStandAlone/TestSenderReceiver.h
+++ /dev/null
@@ -1,166 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
-#define WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
-
-#include "typedefs.h"
-#include "rtp_rtcp.h"
-#include "rtp_rtcp_defines.h"
-#include "udp_transport.h"
-
-class TestLoadGenerator;
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-class ThreadWrapper;
-}
-
-using namespace webrtc;
-
-#define MAX_BITRATE_KBPS 50000
-
-
-class SendRecCB
-{
-public:
- virtual void OnOnNetworkChanged(const WebRtc_UWord32 bitrateTarget,
- const WebRtc_UWord8 fractionLost,
- const WebRtc_UWord16 roundTripTimeMs,
- const WebRtc_UWord16 bwEstimateKbitMin,
- const WebRtc_UWord16 bwEstimateKbitMax) = 0;
-
- virtual ~SendRecCB() {};
-};
-
-
-class TestSenderReceiver : public RtpFeedback, public RtpData, public UdpTransportData, public RtpVideoFeedback
-{
-
-public:
- TestSenderReceiver (void);
-
- ~TestSenderReceiver (void);
-
- void SetCallback (SendRecCB *cb) { _sendRecCB = cb; };
-
- WebRtc_Word32 Start();
-
- WebRtc_Word32 Stop();
-
- bool ProcLoop();
-
- /////////////////////////////////////////////
- // Receiver methods
-
- WebRtc_Word32 InitReceiver (const WebRtc_UWord16 rtpPort,
- const WebRtc_UWord16 rtcpPort = 0,
- const WebRtc_Word8 payloadType = 127);
-
- WebRtc_Word32 ReceiveBitrateKbps ();
-
- WebRtc_Word32 SetPacketTimeout(const WebRtc_UWord32 timeoutMS);
-
- bool timeOutTriggered () { return (_timeOut); };
-
- // Inherited from RtpFeedback
- virtual WebRtc_Word32 OnInitializeDecoder(const WebRtc_Word32 id,
- const WebRtc_Word8 payloadType,
- const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
- const WebRtc_UWord32 frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate) { return(0);};
-
- virtual void OnPacketTimeout(const WebRtc_Word32 id);
-
- virtual void OnReceivedPacket(const WebRtc_Word32 id,
- const RtpRtcpPacketType packetType);
-
- virtual void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
- const RTPAliveType alive) {};
-
- virtual void OnIncomingSSRCChanged( const WebRtc_Word32 id,
- const WebRtc_UWord32 SSRC) {};
-
- virtual void OnIncomingCSRCChanged( const WebRtc_Word32 id,
- const WebRtc_UWord32 CSRC,
- const bool added) {};
-
-
- // Inherited from RtpData
-
- virtual WebRtc_Word32 OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const webrtc::WebRtcRTPHeader* rtpHeader);
-
-
- // Inherited from UdpTransportData
- virtual void IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
- const WebRtc_Word32 rtpPacketLength,
- const WebRtc_Word8* fromIP,
- const WebRtc_UWord16 fromPort);
-
- virtual void IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
- const WebRtc_Word32 rtcpPacketLength,
- const WebRtc_Word8* fromIP,
- const WebRtc_UWord16 fromPort);
-
-
-
- /////////////////////////////////
- // Sender methods
-
- WebRtc_Word32 InitSender (const WebRtc_UWord32 startBitrateKbps,
- const WebRtc_Word8* ipAddr,
- const WebRtc_UWord16 rtpPort,
- const WebRtc_UWord16 rtcpPort = 0,
- const WebRtc_Word8 payloadType = 127);
-
- WebRtc_Word32 SendOutgoingData(const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const webrtc::FrameType frameType = webrtc::kVideoFrameDelta);
-
- WebRtc_Word32 SetLoadGenerator(TestLoadGenerator *generator);
-
- WebRtc_UWord32 BitrateSent() { return (_rtp->BitrateSent()); };
-
-
- // Inherited from RtpVideoFeedback
- virtual void OnReceivedIntraFrameRequest(const WebRtc_Word32 id,
- const WebRtc_UWord8 message = 0) {};
-
- virtual void OnNetworkChanged(const WebRtc_Word32 id,
- const WebRtc_UWord32 minBitrateBps,
- const WebRtc_UWord32 maxBitrateBps,
- const WebRtc_UWord8 fractionLost,
- const WebRtc_UWord16 roundTripTimeMs,
- const WebRtc_UWord16 bwEstimateKbitMin,
- const WebRtc_UWord16 bwEstimateKbitMax);
-
-private:
- RtpRtcp* _rtp;
- UdpTransport* _transport;
- webrtc::CriticalSectionWrapper* _critSect;
- webrtc::EventWrapper *_eventPtr;
- webrtc::ThreadWrapper* _procThread;
- bool _running;
- WebRtc_Word8 _payloadType;
- TestLoadGenerator* _loadGenerator;
- bool _isSender;
- bool _isReceiver;
- bool _timeOut;
- SendRecCB * _sendRecCB;
- WebRtc_UWord32 _lastBytesReceived;
- WebRtc_Word64 _lastTime;
-
-};
-
-#endif // WEBRTC_MODULES_RTP_RTCP_TEST_BWESTANDALONE_TESTSENDERRECEIVER_H_
diff --git a/trunk/src/modules/rtp_rtcp/test/bitstreamTest/bitstreamTest.cc b/trunk/src/modules/rtp_rtcp/test/bitstreamTest/bitstreamTest.cc
deleted file mode 100644
index 38b6e15..0000000
--- a/trunk/src/modules/rtp_rtcp/test/bitstreamTest/bitstreamTest.cc
+++ /dev/null
@@ -1,538 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "../../source/BitstreamBuilder.h"
-#include "../../source/BitstreamParser.h"
-
-#include <assert.h>
-#include <stdio.h>
-#include <math.h>
-#include <tchar.h>
-#include <windows.h>
-
-WebRtc_UWord32 BitRateBPS(WebRtc_UWord16 x )
-{
- return (x & 0x3fff) * WebRtc_UWord32(pow(10.0f,(2 + (x >> 14))));
-}
-
-WebRtc_UWord16 BitRateBPSInv(WebRtc_UWord32 x )
-{
- // 16383 0x3fff
- // 1 638 300 exp 0
- // 16 383 000 exp 1
- // 163 830 000 exp 2
- // 1 638 300 000 exp 3
- const float exp = log10(float(x>>14)) - 2;
- if(exp < 0.0)
- {
- return WebRtc_UWord16(x /100);
- }else if(exp < 1.0)
- {
- return 0x4000 + WebRtc_UWord16(x /1000);
- }else if(exp < 2.0)
- {
- return 0x8000 + WebRtc_UWord16(x /10000);
- }else if(exp < 3.0)
- {
- return 0xC000 + WebRtc_UWord16(x /100000);
- } else
- {
- assert(false);
- return 0;
- }
-}
-
-
-int _tmain(int argc, _TCHAR* argv[])
-{
- WebRtc_UWord8 dataBuffer[128];
- BitstreamBuilder builder(dataBuffer, sizeof(dataBuffer));
-
- // test 1 to 4 bits
- builder.Add1Bit(1);
- builder.Add1Bit(0);
- builder.Add1Bit(1);
-
- builder.Add2Bits(1);
- builder.Add2Bits(2);
- builder.Add2Bits(3);
-
- builder.Add3Bits(1);
- builder.Add3Bits(3);
- builder.Add3Bits(7);
-
- builder.Add4Bits(1);
- builder.Add4Bits(5);
- builder.Add4Bits(15);
-
- assert(4 == builder.Length());
-
- BitstreamParser parser(dataBuffer, sizeof(dataBuffer));
-
- assert(1 == parser.Get1Bit());
- assert(0 == parser.Get1Bit());
- assert(1 == parser.Get1Bit());
-
- assert(1 == parser.Get2Bits());
- assert(2 == parser.Get2Bits());
- assert(3 == parser.Get2Bits());
-
- assert(1 == parser.Get3Bits());
- assert(3 == parser.Get3Bits());
- assert(7 == parser.Get3Bits());
-
- assert(1 == parser.Get4Bits());
- assert(5 == parser.Get4Bits());
- assert(15 == parser.Get4Bits());
-
- printf("Test of 1 to 4 bits done\n");
-
- // test 5 to 7 bits
- builder.Add5Bits(1);
- builder.Add5Bits(15);
- builder.Add5Bits(30);
-
- builder.Add6Bits(1);
- builder.Add6Bits(30);
- builder.Add6Bits(60);
-
- builder.Add7Bits(1);
- builder.Add7Bits(60);
- builder.Add7Bits(120);
-
- assert(1 == parser.Get5Bits());
- assert(15 == parser.Get5Bits());
- assert(30 == parser.Get5Bits());
-
- assert(1 == parser.Get6Bits());
- assert(30 == parser.Get6Bits());
- assert(60 == parser.Get6Bits());
-
- assert(1 == parser.Get7Bits());
- assert(60 == parser.Get7Bits());
- assert(120 == parser.Get7Bits());
-
- printf("Test of 5 to 7 bits done\n");
-
- builder.Add8Bits(1);
- builder.Add1Bit(1);
- builder.Add8Bits(255);
- builder.Add1Bit(0);
- builder.Add8Bits(127);
- builder.Add1Bit(1);
- builder.Add8Bits(60);
- builder.Add1Bit(0);
- builder.Add8Bits(30);
- builder.Add1Bit(1);
- builder.Add8Bits(120);
- builder.Add1Bit(0);
- builder.Add8Bits(160);
- builder.Add1Bit(1);
- builder.Add8Bits(180);
-
- assert(1 == parser.Get8Bits());
- assert(1 == parser.Get1Bit());
- assert(255 == parser.Get8Bits());
- assert(0 == parser.Get1Bit());
- assert(127 == parser.Get8Bits());
- assert(1 == parser.Get1Bit());
- assert(60 == parser.Get8Bits());
- assert(0 == parser.Get1Bit());
- assert(30 == parser.Get8Bits());
- assert(1 == parser.Get1Bit());
- assert(120 == parser.Get8Bits());
- assert(0 == parser.Get1Bit());
- assert(160 == parser.Get8Bits());
- assert(1 == parser.Get1Bit());
- assert(180 == parser.Get8Bits());
-
- printf("Test of 8 bits done\n");
-
- builder.Add16Bits(1);
- builder.Add1Bit(1);
- builder.Add16Bits(255);
- builder.Add1Bit(0);
- builder.Add16Bits(12756);
- builder.Add1Bit(1);
- builder.Add16Bits(60);
- builder.Add1Bit(0);
- builder.Add16Bits(30);
- builder.Add1Bit(1);
- builder.Add16Bits(30120);
- builder.Add1Bit(0);
- builder.Add16Bits(160);
- builder.Add1Bit(1);
- builder.Add16Bits(180);
-
- assert(1 == parser.Get16Bits());
- assert(1 == parser.Get1Bit());
- assert(255 == parser.Get16Bits());
- assert(0 == parser.Get1Bit());
- assert(12756 == parser.Get16Bits());
- assert(1 == parser.Get1Bit());
- assert(60 == parser.Get16Bits());
- assert(0 == parser.Get1Bit());
- assert(30 == parser.Get16Bits());
- assert(1 == parser.Get1Bit());
- assert(30120 == parser.Get16Bits());
- assert(0 == parser.Get1Bit());
- assert(160 == parser.Get16Bits());
- assert(1 == parser.Get1Bit());
- assert(180 == parser.Get16Bits());
-
- printf("Test of 16 bits done\n");
-
- builder.Add24Bits(1);
- builder.Add1Bit(1);
- builder.Add24Bits(255);
- builder.Add1Bit(0);
- builder.Add24Bits(12756);
- builder.Add1Bit(1);
- builder.Add24Bits(60);
- builder.Add1Bit(0);
- builder.Add24Bits(303333);
- builder.Add1Bit(1);
- builder.Add24Bits(30120);
- builder.Add1Bit(0);
- builder.Add24Bits(160);
- builder.Add1Bit(1);
- builder.Add24Bits(8018018);
-
- assert(1 == parser.Get24Bits());
- assert(1 == parser.Get1Bit());
- assert(255 == parser.Get24Bits());
- assert(0 == parser.Get1Bit());
- assert(12756 == parser.Get24Bits());
- assert(1 == parser.Get1Bit());
- assert(60 == parser.Get24Bits());
- assert(0 == parser.Get1Bit());
- assert(303333 == parser.Get24Bits());
- assert(1 == parser.Get1Bit());
- assert(30120 == parser.Get24Bits());
- assert(0 == parser.Get1Bit());
- assert(160 == parser.Get24Bits());
- assert(1 == parser.Get1Bit());
- assert(8018018 == parser.Get24Bits());
-
- printf("Test of 24 bits done\n");
-
- builder.Add32Bits(1);
- builder.Add1Bit(1);
- builder.Add32Bits(255);
- builder.Add1Bit(0);
- builder.Add32Bits(12756);
- builder.Add1Bit(1);
- builder.Add32Bits(60);
- builder.Add1Bit(0);
- builder.Add32Bits(303333);
- builder.Add1Bit(1);
- builder.Add32Bits(3012000012);
- builder.Add1Bit(0);
- builder.Add32Bits(1601601601);
- builder.Add1Bit(1);
- builder.Add32Bits(8018018);
-
- assert(1 == parser.Get32Bits());
- assert(1 == parser.Get1Bit());
- assert(255 == parser.Get32Bits());
- assert(0 == parser.Get1Bit());
- assert(12756 == parser.Get32Bits());
- assert(1 == parser.Get1Bit());
- assert(60 == parser.Get32Bits());
- assert(0 == parser.Get1Bit());
- assert(303333 == parser.Get32Bits());
- assert(1 == parser.Get1Bit());
- assert(3012000012 == parser.Get32Bits());
- assert(0 == parser.Get1Bit());
- assert(1601601601 == parser.Get32Bits());
- assert(1 == parser.Get1Bit());
- assert(8018018 == parser.Get32Bits());
-
- printf("Test of 32 bits done\n");
-
- builder.AddUE(1);
- builder.AddUE(4);
- builder.AddUE(9809706);
- builder.AddUE(2);
- builder.AddUE(15);
- builder.AddUE(16998);
-
- assert( 106 == builder.Length());
-
- assert(1 == parser.GetUE());
- assert(4 == parser.GetUE());
- assert(9809706 == parser.GetUE());
- assert(2 == parser.GetUE());
- assert(15 == parser.GetUE());
- assert(16998 == parser.GetUE());
-
- printf("Test UE bits done\n");
-
- BitstreamBuilder builderScalabilityInfo(dataBuffer, sizeof(dataBuffer));
- BitstreamParser parserScalabilityInfo(dataBuffer, sizeof(dataBuffer));
-
- const WebRtc_UWord8 numberOfLayers = 4;
- const WebRtc_UWord8 layerId[numberOfLayers] = {0,1,2,3};
- const WebRtc_UWord8 priorityId[numberOfLayers] = {0,1,2,3};
- const WebRtc_UWord8 discardableId[numberOfLayers] = {0,1,1,1};
-
- const WebRtc_UWord8 dependencyId[numberOfLayers]= {0,1,1,1};
- const WebRtc_UWord8 qualityId[numberOfLayers]= {0,0,0,1};
- const WebRtc_UWord8 temporalId[numberOfLayers]= {0,0,1,1};
-
- const WebRtc_UWord16 avgBitrate[numberOfLayers]= {BitRateBPSInv(100000),
- BitRateBPSInv(200000),
- BitRateBPSInv(400000),
- BitRateBPSInv(800000)};
-
- // todo which one is the sum?
- const WebRtc_UWord16 maxBitrateLayer[numberOfLayers]= {BitRateBPSInv(150000),
- BitRateBPSInv(300000),
- BitRateBPSInv(500000),
- BitRateBPSInv(900000)};
-
- const WebRtc_UWord16 maxBitrateLayerRepresentation[numberOfLayers] = {BitRateBPSInv(150000),
- BitRateBPSInv(450000),
- BitRateBPSInv(950000),
- BitRateBPSInv(1850000)};
-
- assert( 16300 == BitRateBPS(BitRateBPSInv(16383)));
- assert( 163800 == BitRateBPS(BitRateBPSInv(163830)));
- assert( 1638300 == BitRateBPS(BitRateBPSInv(1638300)));
- assert( 1638000 == BitRateBPS(BitRateBPSInv(1638400)));
-
- assert( 18500 == BitRateBPS(BitRateBPSInv(18500)));
- assert( 185000 == BitRateBPS(BitRateBPSInv(185000)));
- assert( 1850000 == BitRateBPS(BitRateBPSInv(1850000)));
- assert( 18500000 == BitRateBPS(BitRateBPSInv(18500000)));
- assert( 185000000 == BitRateBPS(BitRateBPSInv(185000000)));
-
- const WebRtc_UWord16 maxBitrareCalcWindow[numberOfLayers] = {200, 200,200,200};// in 1/100 of second
-
- builderScalabilityInfo.Add1Bit(0); // temporal_id_nesting_flag
- builderScalabilityInfo.Add1Bit(0); // priority_layer_info_present_flag
- builderScalabilityInfo.Add1Bit(0); // priority_id_setting_flag
-
- builderScalabilityInfo.AddUE(numberOfLayers-1);
-
- for(int i = 0; i<= numberOfLayers-1; i++)
- {
- builderScalabilityInfo.AddUE(layerId[i]);
- builderScalabilityInfo.Add6Bits(priorityId[i]);
- builderScalabilityInfo.Add1Bit(discardableId[i]);
- builderScalabilityInfo.Add3Bits(dependencyId[i]);
- builderScalabilityInfo.Add4Bits(qualityId[i]);
- builderScalabilityInfo.Add3Bits(temporalId[i]);
-
- builderScalabilityInfo.Add1Bit(0);
- builderScalabilityInfo.Add1Bit(0);
- builderScalabilityInfo.Add1Bit(0);
- builderScalabilityInfo.Add1Bit(0);
-
- builderScalabilityInfo.Add1Bit(1); // bitrate_info_present_flag
-
- builderScalabilityInfo.Add1Bit(0);
- builderScalabilityInfo.Add1Bit(0);
- builderScalabilityInfo.Add1Bit(0);
- builderScalabilityInfo.Add1Bit(0);
- builderScalabilityInfo.Add1Bit(0);
- builderScalabilityInfo.Add1Bit(0);
- builderScalabilityInfo.Add1Bit(0);
- builderScalabilityInfo.Add1Bit(0);
-
- builderScalabilityInfo.Add16Bits(avgBitrate[i]);
- builderScalabilityInfo.Add16Bits(maxBitrateLayer[i]);
- builderScalabilityInfo.Add16Bits(maxBitrateLayerRepresentation[i]);
- builderScalabilityInfo.Add16Bits(maxBitrareCalcWindow[i]);
-
- builderScalabilityInfo.AddUE(0); // layer_dependency_info_src_layer_id_delta
- builderScalabilityInfo.AddUE(0); // parameter_sets_info_src_layer_id_delta
- }
-
- printf("Test builderScalabilityInfo done\n");
-
- // Scalability Info parser
- parserScalabilityInfo.Get1Bit(); // not used in futher parsing
- const WebRtc_UWord8 priority_layer_info_present = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 priority_id_setting_flag = parserScalabilityInfo.Get1Bit();
-
- WebRtc_UWord32 numberOfLayersMinusOne = parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 j = 0; j<= numberOfLayersMinusOne; j++)
- {
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.Get6Bits();
- parserScalabilityInfo.Get1Bit();
- parserScalabilityInfo.Get3Bits();
- parserScalabilityInfo.Get4Bits();
- parserScalabilityInfo.Get3Bits();
-
- const WebRtc_UWord8 sub_pic_layer_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 sub_region_layer_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 iroi_division_info_present_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 profile_level_info_present_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 bitrate_info_present_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 frm_rate_info_present_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 frm_size_info_present_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 layer_dependency_info_present_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 parameter_sets_info_present_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 bitstream_restriction_info_present_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 exact_inter_layer_pred_flag = parserScalabilityInfo.Get1Bit(); // not used in futher parsing
-
- if(sub_pic_layer_flag || iroi_division_info_present_flag)
- {
- parserScalabilityInfo.Get1Bit();
- }
- const WebRtc_UWord8 layer_conversion_flag = parserScalabilityInfo.Get1Bit();
- const WebRtc_UWord8 layer_output_flag = parserScalabilityInfo.Get1Bit(); // not used in futher parsing
-
- if(profile_level_info_present_flag)
- {
- parserScalabilityInfo.Get24Bits();
- }
- if(bitrate_info_present_flag)
- {
- // this is what we want
- assert(avgBitrate[j] == parserScalabilityInfo.Get16Bits());
- assert(maxBitrateLayer[j] == parserScalabilityInfo.Get16Bits());
- assert(maxBitrateLayerRepresentation[j] == parserScalabilityInfo.Get16Bits());
- assert(maxBitrareCalcWindow[j] == parserScalabilityInfo.Get16Bits());
- }else
- {
- assert(false);
- }
- if(frm_rate_info_present_flag)
- {
- parserScalabilityInfo.Get2Bits();
- parserScalabilityInfo.Get16Bits();
- }
- if(frm_size_info_present_flag || iroi_division_info_present_flag)
- {
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.GetUE();
- }
- if(sub_region_layer_flag)
- {
- parserScalabilityInfo.GetUE();
- if(parserScalabilityInfo.Get1Bit())
- {
- parserScalabilityInfo.Get16Bits();
- parserScalabilityInfo.Get16Bits();
- parserScalabilityInfo.Get16Bits();
- parserScalabilityInfo.Get16Bits();
- }
- }
- if(sub_pic_layer_flag)
- {
- parserScalabilityInfo.GetUE();
- }
- if(iroi_division_info_present_flag)
- {
- if(parserScalabilityInfo.Get1Bit())
- {
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.GetUE();
- }else
- {
- const WebRtc_UWord32 numRoisMinusOne = parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 k = 0; k <= numRoisMinusOne; k++)
- {
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.GetUE();
- }
- }
- }
- if(layer_dependency_info_present_flag)
- {
- const WebRtc_UWord32 numDirectlyDependentLayers = parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 k = 0; k < numDirectlyDependentLayers; k++)
- {
- parserScalabilityInfo.GetUE();
- }
- } else
- {
- parserScalabilityInfo.GetUE();
- }
- if(parameter_sets_info_present_flag)
- {
- const WebRtc_UWord32 numSeqParameterSetMinusOne = parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 k = 0; k <= numSeqParameterSetMinusOne; k++)
- {
- parserScalabilityInfo.GetUE();
- }
- const WebRtc_UWord32 numSubsetSeqParameterSetMinusOne = parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 l = 0; l <= numSubsetSeqParameterSetMinusOne; l++)
- {
- parserScalabilityInfo.GetUE();
- }
- const WebRtc_UWord32 numPicParameterSetMinusOne = parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 m = 0; m <= numPicParameterSetMinusOne; m++)
- {
- parserScalabilityInfo.GetUE();
- }
- }else
- {
- parserScalabilityInfo.GetUE();
- }
- if(bitstream_restriction_info_present_flag)
- {
- parserScalabilityInfo.Get1Bit();
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.GetUE();
- }
- if(layer_conversion_flag)
- {
- parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 k = 0; k <2;k++)
- {
- if(parserScalabilityInfo.Get1Bit())
- {
- parserScalabilityInfo.Get24Bits();
- parserScalabilityInfo.Get16Bits();
- parserScalabilityInfo.Get16Bits();
- }
- }
- }
- }
- if(priority_layer_info_present)
- {
- const WebRtc_UWord32 prNumDidMinusOne = parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 k = 0; k <= prNumDidMinusOne;k++)
- {
- parserScalabilityInfo.Get3Bits();
- const WebRtc_UWord32 prNumMinusOne = parserScalabilityInfo.GetUE();
- for(WebRtc_UWord32 l = 0; l <= prNumMinusOne; l++)
- {
- parserScalabilityInfo.GetUE();
- parserScalabilityInfo.Get24Bits();
- parserScalabilityInfo.Get16Bits();
- parserScalabilityInfo.Get16Bits();
- }
- }
- }
- if(priority_id_setting_flag)
- {
- WebRtc_UWord8 priorityIdSettingUri;
- WebRtc_UWord32 priorityIdSettingUriIdx = 0;
- do
- {
- priorityIdSettingUri = parserScalabilityInfo.Get8Bits();
- } while (priorityIdSettingUri != 0);
- }
- printf("Test parserScalabilityInfo done\n");
-
- printf("\nAPI test of parser for ScalabilityInfo done\n");
-
- ::Sleep(5000);
-}
diff --git a/trunk/src/modules/rtp_rtcp/test/bwe_standalone.gypi b/trunk/src/modules/rtp_rtcp/test/bwe_standalone.gypi
deleted file mode 100644
index 36a50de..0000000
--- a/trunk/src/modules/rtp_rtcp/test/bwe_standalone.gypi
+++ /dev/null
@@ -1,111 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'bwe_standalone',
- 'type': 'executable',
- 'dependencies': [
- 'matlab_plotting',
- 'rtp_rtcp',
- 'udp_transport',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '../interface',
- '../../interface',
- ],
- 'sources': [
- 'BWEStandAlone/BWEStandAlone.cc',
- 'BWEStandAlone/TestLoadGenerator.cc',
- 'BWEStandAlone/TestLoadGenerator.h',
- 'BWEStandAlone/TestSenderReceiver.cc',
- 'BWEStandAlone/TestSenderReceiver.h',
- ], # source
- 'conditions': [
- ['OS=="linux"', {
- 'cflags': [
- '-fexceptions', # enable exceptions
- ],
- },
- ],
- ],
-
- 'include_dirs': [
- ],
- 'link_settings': {
- },
- },
-
- {
- 'target_name': 'matlab_plotting',
- 'type': '<(library)',
- 'dependencies': [
- 'matlab_plotting_include',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '/opt/matlab2010a/extern/include',
- ],
- # 'direct_dependent_settings': {
- # 'defines': [
- # 'MATLAB',
- # ],
- # 'include_dirs': [
- # 'BWEStandAlone',
- # ],
- # },
- 'export_dependent_settings': [
- 'matlab_plotting_include',
- ],
- 'sources': [
- 'BWEStandAlone/MatlabPlot.cc',
- 'BWEStandAlone/MatlabPlot.h',
- ],
- 'link_settings': {
- 'ldflags' : [
- '-L/opt/matlab2010a/bin/glnxa64',
- '-leng',
- '-lmx',
- '-Wl,-rpath,/opt/matlab2010a/bin/glnxa64',
- ],
- },
- 'defines': [
- 'MATLAB',
- ],
- 'conditions': [
- ['OS=="linux"', {
- 'cflags': [
- '-fexceptions', # enable exceptions
- ],
- },
- ],
- ],
- },
-
- {
- 'target_name': 'matlab_plotting_include',
- 'type': 'none',
- 'direct_dependent_settings': {
- 'defines': [
-# 'MATLAB',
- ],
- 'include_dirs': [
- 'BWEStandAlone',
- ],
- },
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.cc b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.cc
deleted file mode 100644
index 3b5390e..0000000
--- a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.cc
+++ /dev/null
@@ -1,120 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <algorithm>
-#include <vector>
-#include <gtest/gtest.h>
-
-#include "test_api.h"
-
-#include "common_types.h"
-#include "rtp_rtcp.h"
-#include "rtp_rtcp_defines.h"
-
-using namespace webrtc;
-
-class RtpRtcpAPITest : public ::testing::Test {
- protected:
- RtpRtcpAPITest() {
- test_CSRC[0] = 1234;
- test_CSRC[2] = 2345;
- test_id = 123;
- test_ssrc = 3456;
- test_timestamp = 4567;
- test_sequence_number = 2345;
- }
- ~RtpRtcpAPITest() {}
-
- virtual void SetUp() {
- module = RtpRtcp::CreateRtpRtcp(test_id, true, &fake_clock);
- EXPECT_EQ(0, module->InitReceiver());
- EXPECT_EQ(0, module->InitSender());
- }
-
- virtual void TearDown() {
- RtpRtcp::DestroyRtpRtcp(module);
- }
-
- int test_id;
- RtpRtcp* module;
- WebRtc_UWord32 test_ssrc;
- WebRtc_UWord32 test_timestamp;
- WebRtc_UWord16 test_sequence_number;
- WebRtc_UWord32 test_CSRC[webrtc::kRtpCsrcSize];
- FakeRtpRtcpClock fake_clock;
-};
-
-TEST_F(RtpRtcpAPITest, Basic) {
- EXPECT_EQ(0, module->SetSequenceNumber(test_sequence_number));
- EXPECT_EQ(test_sequence_number, module->SequenceNumber());
-
- EXPECT_EQ(0, module->SetStartTimestamp(test_timestamp));
- EXPECT_EQ(test_timestamp, module->StartTimestamp());
-
- EXPECT_FALSE(module->Sending());
- EXPECT_EQ(0, module->SetSendingStatus(true));
- EXPECT_TRUE(module->Sending());
-}
-
-TEST_F(RtpRtcpAPITest, MTU) {
- EXPECT_EQ(-1, module->SetMaxTransferUnit(10));
- EXPECT_EQ(-1, module->SetMaxTransferUnit(IP_PACKET_SIZE + 1));
- EXPECT_EQ(0, module->SetMaxTransferUnit(1234));
- EXPECT_EQ(1234-20-8, module->MaxPayloadLength());
-
- EXPECT_EQ(0, module->SetTransportOverhead(true, true, 12));
- EXPECT_EQ(1234 - 20- 20 -20 - 12, module->MaxPayloadLength());
-
- EXPECT_EQ(0, module->SetTransportOverhead(false, false, 0));
- EXPECT_EQ(1234 - 20 - 8, module->MaxPayloadLength());
-}
-
-TEST_F(RtpRtcpAPITest, SSRC) {
- EXPECT_EQ(0, module->SetSSRC(test_ssrc));
- EXPECT_EQ(test_ssrc, module->SSRC());
-}
-
-TEST_F(RtpRtcpAPITest, CSRC) {
- EXPECT_EQ(0, module->SetCSRCs(test_CSRC, 2));
- WebRtc_UWord32 testOfCSRC[webrtc::kRtpCsrcSize];
- EXPECT_EQ(2, module->CSRCs(testOfCSRC));
- EXPECT_EQ(test_CSRC[0], testOfCSRC[0]);
- EXPECT_EQ(test_CSRC[1], testOfCSRC[1]);
-}
-
-TEST_F(RtpRtcpAPITest, TrafficSmoothing) {
- EXPECT_FALSE(module->TransmissionSmoothingStatus());
- module->SetTransmissionSmoothingStatus(true);
- EXPECT_TRUE(module->TransmissionSmoothingStatus());
-}
-
-TEST_F(RtpRtcpAPITest, RTCP) {
- EXPECT_EQ(kRtcpOff, module->RTCP());
- EXPECT_EQ(0, module->SetRTCPStatus(kRtcpCompound));
- EXPECT_EQ(kRtcpCompound, module->RTCP());
-
- EXPECT_EQ(0, module->SetCNAME("john.doe@test.test"));
- EXPECT_EQ(-1, module->SetCNAME(NULL));
-
- char cName[RTCP_CNAME_SIZE];
- EXPECT_EQ(0, module->CNAME(cName));
- EXPECT_STRCASEEQ(cName, "john.doe@test.test");
- EXPECT_EQ(-1, module->CNAME(NULL));
-
- EXPECT_FALSE(module->TMMBR());
- EXPECT_EQ(0, module->SetTMMBRStatus(true));
- EXPECT_TRUE(module->TMMBR());
- EXPECT_EQ(0, module->SetTMMBRStatus(false));
- EXPECT_FALSE(module->TMMBR());
-
- EXPECT_EQ(kNackOff, module->NACK());
- EXPECT_EQ(0, module->SetNACKStatus(kNackRtcp));
- EXPECT_EQ(kNackRtcp, module->NACK());
-}
diff --git a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.gypi b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.gypi
deleted file mode 100644
index d47e6d4..0000000
--- a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.gypi
+++ /dev/null
@@ -1,42 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'test_rtp_rtcp_api',
- 'type': 'executable',
- 'dependencies': [
- 'rtp_rtcp',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
-
- 'include_dirs': [
- '../../interface',
- '../../source',
- '../../../../system_wrappers/interface',
- ],
-
- 'sources': [
- 'test_api.cc',
- 'test_api_audio.cc',
- 'test_api_nack.cc',
- 'test_api_rtcp.cc',
- 'test_api_video.cc',
- ],
-
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.h b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.h
deleted file mode 100644
index c4ff916..0000000
--- a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api.h
+++ /dev/null
@@ -1,86 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "common_types.h"
-#include "rtp_rtcp.h"
-#include "rtp_rtcp_defines.h"
-
-namespace webrtc {
-
-class FakeRtpRtcpClock : public RtpRtcpClock {
- public:
- FakeRtpRtcpClock() {
- time_in_ms_ = 123456;
- }
- // Return a timestamp in milliseconds relative to some arbitrary
- // source; the source is fixed for this clock.
- virtual WebRtc_UWord32 GetTimeInMS() {
- return time_in_ms_;
- }
- // Retrieve an NTP absolute timestamp.
- virtual void CurrentNTP(WebRtc_UWord32& secs, WebRtc_UWord32& frac) {
- secs = time_in_ms_ / 1000;
- frac = (time_in_ms_ % 1000) * 4294967;
- }
- void IncrementTime(WebRtc_UWord32 time_increment_ms) {
- time_in_ms_ += time_increment_ms;
- }
- private:
- WebRtc_UWord32 time_in_ms_;
-};
-
-// This class sends all its packet straight to the provided RtpRtcp module.
-// with optional packet loss.
-class LoopBackTransport : public webrtc::Transport {
- public:
- LoopBackTransport(RtpRtcp* rtpRtcpModule)
- : _count(0),
- _packetLoss(0),
- _rtpRtcpModule(rtpRtcpModule) {
- }
- void DropEveryNthPacket(int n) {
- _packetLoss = n;
- }
- virtual int SendPacket(int channel, const void *data, int len) {
- _count++;
- if (_packetLoss > 0) {
- if ((_count % _packetLoss) == 0) {
- return len;
- }
- }
- if (_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len) == 0) {
- return len;
- }
- return -1;
- }
- virtual int SendRTCPPacket(int channel, const void *data, int len) {
- if (_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len) == 0) {
- return len;
- }
- return -1;
- }
- private:
- int _count;
- int _packetLoss;
- RtpRtcp* _rtpRtcpModule;
-};
-
-class RtpReceiver : public RtpData {
- public:
- virtual WebRtc_Word32 OnReceivedPayloadData(
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const webrtc::WebRtcRTPHeader* rtpHeader) {
- return 0;
- }
-};
-
-} // namespace webrtc
-
diff --git a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_audio.cc b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
deleted file mode 100644
index d066864..0000000
--- a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_audio.cc
+++ /dev/null
@@ -1,446 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <algorithm>
-#include <vector>
-#include <gtest/gtest.h>
-
-#include "test_api.h"
-
-#include "common_types.h"
-#include "rtp_rtcp.h"
-#include "rtp_rtcp_defines.h"
-
-using namespace webrtc;
-
-#define test_rate 64000u
-
-class VerifyingAudioReceiver : public RtpData {
- public:
- VerifyingAudioReceiver(RtpRtcp* rtpRtcpModule) {}
-
- virtual WebRtc_Word32 OnReceivedPayloadData(
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const webrtc::WebRtcRTPHeader* rtpHeader) {
- if (rtpHeader->header.payloadType == 98 ||
- rtpHeader->header.payloadType == 99) {
- EXPECT_EQ(4, payloadSize);
- char str[5];
- memcpy(str, payloadData, payloadSize);
- str[4] = 0;
- // All our test vectors for payload type 96 and 97 even the stereo is on
- // a per channel base equal to the 4 chars "test".
- // Note there is no null termination so we add that to use the
- // test EXPECT_STRCASEEQ.
- EXPECT_STRCASEEQ("test", str);
- return 0;
- }
- if (rtpHeader->header.payloadType == 100 ||
- rtpHeader->header.payloadType == 101 ||
- rtpHeader->header.payloadType == 102) {
- if (rtpHeader->type.Audio.channel == 1) {
- if (payloadData[0] == 0xff) {
- // All our test vectors for payload type 100, 101 and 102 have the
- // first channel data being equal to 0xff.
- return 0;
- }
- } else if (rtpHeader->type.Audio.channel == 2) {
- if (payloadData[0] == 0x0) {
- // All our test vectors for payload type 100, 101 and 102 have the
- // second channel data being equal to 0x00.
- return 0;
- }
- } else if (rtpHeader->type.Audio.channel == 3) {
- // All our test vectors for payload type 100, 101 and 102 have the
- // third channel data being equal to 0xaa.
- if (payloadData[0] == 0xaa) {
- return 0;
- }
- }
- ADD_FAILURE() << "This code path should never happen.";
- return -1;
- }
- return 0;
- }
-};
-
-class RTPCallback : public RtpFeedback {
- public:
- virtual WebRtc_Word32 OnInitializeDecoder(
- const WebRtc_Word32 id,
- const WebRtc_Word8 payloadType,
- const char payloadName[RTP_PAYLOAD_NAME_SIZE],
- const int frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate) {
- if (payloadType == 96) {
- EXPECT_EQ(test_rate, rate) <<
- "The rate should be 64K for this payloadType";
- }
- return 0;
- }
- virtual void OnPacketTimeout(const WebRtc_Word32 id) {
- }
- virtual void OnReceivedPacket(const WebRtc_Word32 id,
- const RtpRtcpPacketType packetType) {
- }
- virtual void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
- const RTPAliveType alive) {
- }
- virtual void OnIncomingSSRCChanged(const WebRtc_Word32 id,
- const WebRtc_UWord32 SSRC) {
- }
- virtual void OnIncomingCSRCChanged(const WebRtc_Word32 id,
- const WebRtc_UWord32 CSRC,
- const bool added) {
- }
-};
-
-class AudioFeedback : public RtpAudioFeedback {
- virtual void OnReceivedTelephoneEvent(const WebRtc_Word32 id,
- const WebRtc_UWord8 event,
- const bool end) {
- static WebRtc_UWord8 expectedEvent = 0;
-
- if (end) {
- WebRtc_UWord8 oldEvent = expectedEvent-1;
- if (expectedEvent == 32) {
- oldEvent = 15;
- }
- EXPECT_EQ(oldEvent, event);
- } else {
- EXPECT_EQ(expectedEvent, event);
- expectedEvent++;
- }
- if (expectedEvent == 16) {
- expectedEvent = 32;
- }
- }
- virtual void OnPlayTelephoneEvent(const WebRtc_Word32 id,
- const WebRtc_UWord8 event,
- const WebRtc_UWord16 lengthMs,
- const WebRtc_UWord8 volume) {
- };
-};
-
-class RtpRtcpAudioTest : public ::testing::Test {
- protected:
- RtpRtcpAudioTest() {
- test_CSRC[0] = 1234;
- test_CSRC[2] = 2345;
- test_id = 123;
- test_ssrc = 3456;
- test_timestamp = 4567;
- test_sequence_number = 2345;
- }
- ~RtpRtcpAudioTest() {}
-
- virtual void SetUp() {
- module1 = RtpRtcp::CreateRtpRtcp(test_id, true, &fake_clock);
- module2 = RtpRtcp::CreateRtpRtcp(test_id+1, true, &fake_clock);
-
- EXPECT_EQ(0, module1->InitReceiver());
- EXPECT_EQ(0, module1->InitSender());
- EXPECT_EQ(0, module2->InitReceiver());
- EXPECT_EQ(0, module2->InitSender());
- data_receiver1 = new VerifyingAudioReceiver(module1);
- EXPECT_EQ(0, module1->RegisterIncomingDataCallback(data_receiver1));
- data_receiver2 = new VerifyingAudioReceiver(module2);
- EXPECT_EQ(0, module2->RegisterIncomingDataCallback(data_receiver2));
- transport1 = new LoopBackTransport(module2);
- EXPECT_EQ(0, module1->RegisterSendTransport(transport1));
- transport2 = new LoopBackTransport(module1);
- EXPECT_EQ(0, module2->RegisterSendTransport(transport2));
- rtp_callback = new RTPCallback();
- EXPECT_EQ(0, module2->RegisterIncomingRTPCallback(rtp_callback));
- }
-
- virtual void TearDown() {
- RtpRtcp::DestroyRtpRtcp(module1);
- RtpRtcp::DestroyRtpRtcp(module2);
- delete transport1;
- delete transport2;
- delete data_receiver1;
- delete data_receiver2;
- delete rtp_callback;
- }
-
- int test_id;
- RtpRtcp* module1;
- RtpRtcp* module2;
- VerifyingAudioReceiver* data_receiver1;
- VerifyingAudioReceiver* data_receiver2;
- LoopBackTransport* transport1;
- LoopBackTransport* transport2;
- RTPCallback* rtp_callback;
- WebRtc_UWord32 test_ssrc;
- WebRtc_UWord32 test_timestamp;
- WebRtc_UWord16 test_sequence_number;
- WebRtc_UWord32 test_CSRC[webrtc::kRtpCsrcSize];
- FakeRtpRtcpClock fake_clock;
-};
-
-TEST_F(RtpRtcpAudioTest, Basic) {
- EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
- EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));
-
- EXPECT_FALSE(module1->TelephoneEvent());
-
- // Test detection at the end of a DTMF tone.
- EXPECT_EQ(0, module2->SetTelephoneEventStatus(true, true, true));
- EXPECT_EQ(true, module2->TelephoneEvent());
-
- EXPECT_EQ(0, module1->SetSendingStatus(true));
-
- // Start basic RTP test.
-
- // Send an empty RTP packet.
- // Should fail since we have not registerd the payload type.
- EXPECT_EQ(-1, module1->SendOutgoingData(webrtc::kAudioFrameSpeech,
- 96, 0, NULL, 0));
-
- CodecInst voiceCodec;
- voiceCodec.pltype = 96;
- voiceCodec.plfreq = 8000;
- memcpy(voiceCodec.plname, "PCMU", 5);
-
- EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
- EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
- EXPECT_EQ(0, module2->RegisterSendPayload(voiceCodec));
- voiceCodec.rate = test_rate;
- EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
- printf("4\n");
-
- const WebRtc_UWord8 test[5] = "test";
- EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
- 0, test, 4));
-
- EXPECT_EQ(test_ssrc, module2->RemoteSSRC());
- EXPECT_EQ(test_timestamp, module2->RemoteTimestamp());
-}
-
-TEST_F(RtpRtcpAudioTest, RED) {
- CodecInst voiceCodec;
- voiceCodec.pltype = 96;
- voiceCodec.plfreq = 8000;
- memcpy(voiceCodec.plname, "PCMU", 5);
-
- EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
- EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
- EXPECT_EQ(0, module2->RegisterSendPayload(voiceCodec));
- voiceCodec.rate = test_rate;
- EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
-
- EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
- EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));
- EXPECT_EQ(0, module1->SetSendingStatus(true));
-
- voiceCodec.pltype = 127;
- voiceCodec.plfreq = 8000;
- memcpy(voiceCodec.plname, "RED", 4);
-
- EXPECT_EQ(0, module1->SetSendREDPayloadType(voiceCodec.pltype));
- WebRtc_Word8 red = 0;
- EXPECT_EQ(0, module1->SendREDPayloadType(red));
- EXPECT_EQ(voiceCodec.pltype, red);
- EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
- EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
-
- RTPFragmentationHeader fragmentation;
- fragmentation.fragmentationVectorSize = 2;
- fragmentation.fragmentationLength = new WebRtc_UWord32[2];
- fragmentation.fragmentationLength[0] = 4;
- fragmentation.fragmentationLength[1] = 4;
- fragmentation.fragmentationOffset = new WebRtc_UWord32[2];
- fragmentation.fragmentationOffset[0] = 0;
- fragmentation.fragmentationOffset[1] = 4;
- fragmentation.fragmentationTimeDiff = new WebRtc_UWord16[2];
- fragmentation.fragmentationTimeDiff[0] = 0;
- fragmentation.fragmentationTimeDiff[1] = 0;
- fragmentation.fragmentationPlType = new WebRtc_UWord8[2];
- fragmentation.fragmentationPlType[0] = 96;
- fragmentation.fragmentationPlType[1] = 96;
-
- const WebRtc_UWord8 test[5] = "test";
- // Send a RTP packet.
- EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech,
- 96, 160, test, 4,
- &fragmentation));
-
- EXPECT_EQ(0, module1->SetSendREDPayloadType(-1));
- EXPECT_EQ(-1, module1->SendREDPayloadType(red));
-}
-
-TEST_F(RtpRtcpAudioTest, DTMF) {
- CodecInst voiceCodec;
- voiceCodec.pltype = 96;
- voiceCodec.plfreq = 8000;
- memcpy(voiceCodec.plname, "PCMU", 5);
-
- EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
- EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
- EXPECT_EQ(0, module2->RegisterSendPayload(voiceCodec));
- voiceCodec.rate = test_rate;
- EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
-
- EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
- EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));
- EXPECT_EQ(0, module1->SetSendingStatus(true));
-
- AudioFeedback* audioFeedback = new AudioFeedback();
- EXPECT_EQ(0, module2->RegisterAudioCallback(audioFeedback));
-
- // Prepare for DTMF.
- voiceCodec.pltype = 97;
- voiceCodec.plfreq = 8000;
- memcpy(voiceCodec.plname, "telephone-event", 16);
-
- EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
- EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
-
- // Start DTMF test.
- WebRtc_UWord32 timeStamp = 160;
-
- // Send a DTMF tone using RFC 2833 (4733).
- for (int i = 0; i < 16; i++) {
- EXPECT_EQ(0, module1->SendTelephoneEventOutband(i, timeStamp, 10));
- }
- timeStamp += 160; // Prepare for next packet.
-
- const WebRtc_UWord8 test[9] = "test";
-
- // Send RTP packets for 16 tones a 160 ms 100ms
- // pause between = 2560ms + 1600ms = 4160ms
- for (;timeStamp <= 250 * 160; timeStamp += 160) {
- EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
- timeStamp, test, 4));
- fake_clock.IncrementTime(20);
- module1->Process();
- }
- EXPECT_EQ(0, module1->SendTelephoneEventOutband(32, 9000, 10));
-
- for (;timeStamp <= 740 * 160; timeStamp += 160) {
- EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
- timeStamp, test, 4));
- fake_clock.IncrementTime(20);
- module1->Process();
- }
- delete audioFeedback;
-}
-
-TEST_F(RtpRtcpAudioTest, Stereo) {
- CodecInst voiceCodec;
- voiceCodec.pltype = 96;
- voiceCodec.plfreq = 8000;
- memcpy(voiceCodec.plname, "PCMU", 5);
-
- EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
- EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
- EXPECT_EQ(0, module2->RegisterSendPayload(voiceCodec));
- voiceCodec.rate = test_rate;
- EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
-
- EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
- EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));
- EXPECT_EQ(0, module1->SetSendingStatus(true));
-
- // Prepare for 3 channel audio 8 bits per sample.
- voiceCodec.pltype = 98;
- voiceCodec.channels = 3;
- memcpy(voiceCodec.plname, "PCMA", 5);
- EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
- EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
-
- // Prepare for 3 channel audio 16 bits per sample.
- voiceCodec.pltype = 99;
- memcpy(voiceCodec.plname, "L16", 4);
- EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
- EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
-
- // Prepare for 3 channel audio 5 bits per sample.
- voiceCodec.pltype = 100;
- memcpy(voiceCodec.plname, "G726-40",8);
- EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
- EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
-
- // Prepare for 3 channel audio 3 bits per sample.
- voiceCodec.pltype = 101;
- memcpy(voiceCodec.plname, "G726-24",8);
- EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
- EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
-
- // Prepare for 3 channel audio 2 bits per sample.
- voiceCodec.pltype = 102;
- memcpy(voiceCodec.plname, "G726-16",8);
- EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
- EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
-
- // Test sample based multi channel codec, 3 channels 8 bits.
- WebRtc_UWord8 test3channels[15] = "ttteeesssttt";
- WebRtc_UWord32 timeStamp = 160;
- EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 98,
- timeStamp, test3channels, 12));
- fake_clock.IncrementTime(20);
- module1->Process();
- timeStamp += 160; // Prepare for next packet.
-
- // Test sample based multi channel codec, 3 channels 16 bits.
- const WebRtc_UWord8 test3channels16[13] = "teteteststst";
- EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 99,
- timeStamp, test3channels16, 12));
- fake_clock.IncrementTime(20);
- module1->Process();
- timeStamp += 160; // Prepare for next packet.
-
- // Test sample based multi channel codec, 3 channels 5 bits.
- test3channels[0] = 0xf8; // 5 ones 3 zeros.
- test3channels[1] = 0x2b; // 2 zeros 5 10 1 one.
- test3channels[2] = 0xf0; // 4 ones 4 zeros.
- test3channels[3] = 0x2b; // 1 zero 5 01 2 ones.
- test3channels[4] = 0xe0; // 3 ones 5 zeros.
- test3channels[5] = 0x0;
- test3channels[6] = 0x0;
- test3channels[7] = 0x0;
- test3channels[8] = 0x0;
- test3channels[9] = 0x0;
- test3channels[10] = 0x0;
- test3channels[11] = 0x0;
- test3channels[12] = 0x0;
- test3channels[13] = 0x0;
- test3channels[14] = 0x0;
-
- EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 100,
- timeStamp, test3channels, 15));
- fake_clock.IncrementTime(20);
- module1->Process();
- timeStamp += 160; // Prepare for next packet.
-
- // Test sample based multi channel codec, 3 channels 3 bits.
- test3channels[0] = 0xe2; // 3 ones 3 zeros 2 10
- test3channels[1] = 0xf0; // 1 1 3 ones 3 zeros 1 0
- test3channels[2] = 0xb8; // 2 10 3 ones 3 zeros
- test3channels[3] = 0xa0; // 3 101 5 zeros
- test3channels[4] = 0x0;
- EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 101,
- timeStamp, test3channels, 15));
- fake_clock.IncrementTime(20);
- module1->Process();
- timeStamp += 160; // Prepare for next packet.
-
- // Test sample based multi channel codec, 3 channels 2 bits.
- test3channels[0] = 0xcb; // 2 ones 2 zeros 2 10 2 ones
- test3channels[1] = 0x2c; // 2 zeros 2 10 2 ones 2 zeros
- test3channels[2] = 0xb2; // 2 10 2 ones 2 zeros 2 10
- test3channels[3] = 0xcb; // 2 ones 2 zeros 2 10 2 ones
- test3channels[4] = 0x2c; // 2 zeros 2 10 2 ones 2 zeros
- EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 102,
- timeStamp, test3channels, 15));
-}
diff --git a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_nack.cc b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_nack.cc
deleted file mode 100644
index fd3c3df..0000000
--- a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_nack.cc
+++ /dev/null
@@ -1,249 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <algorithm>
-#include <vector>
-#include <gtest/gtest.h>
-
-#include "test_api.h"
-
-#include "common_types.h"
-#include "rtp_rtcp.h"
-#include "rtp_rtcp_defines.h"
-
-using namespace webrtc;
-
-const int kVideoNackListSize = 10;
-const int kTestId = 123;
-const WebRtc_UWord32 kTestSsrc = 3456;
-const WebRtc_UWord16 kTestSequenceNumber = 2345;
-const WebRtc_UWord32 kTestNumberOfPackets = 450;
-const int kTestNumberOfRtxPackets = 49;
-
-class VerifyingNackReceiver : public RtpData
-{
- public:
- VerifyingNackReceiver() {}
-
- virtual WebRtc_Word32 OnReceivedPayloadData(
- const WebRtc_UWord8* data,
- const WebRtc_UWord16 size,
- const webrtc::WebRtcRTPHeader* rtp_header) {
-
- EXPECT_EQ(kTestSsrc, rtp_header->header.ssrc);
- EXPECT_EQ(find(sequence_numbers_.begin(),
- sequence_numbers_.end(),
- rtp_header->header.sequenceNumber),
- sequence_numbers_.end());
- sequence_numbers_.push_back(rtp_header->header.sequenceNumber);
- return 0;
- }
- std::vector<WebRtc_UWord16 > sequence_numbers_;
-};
-
-class NackLoopBackTransport : public webrtc::Transport {
- public:
- NackLoopBackTransport(RtpRtcp* rtp_rtcp_module, uint32_t rtx_ssrc)
- : count_(0),
- packet_loss_(0),
- rtx_ssrc_(rtx_ssrc),
- count_rtx_ssrc_(0),
- module_(rtp_rtcp_module) {
- }
- void DropEveryNthPacket(int n) {
- packet_loss_ = n;
- }
- virtual int SendPacket(int channel, const void *data, int len) {
- count_++;
- const unsigned char* ptr = static_cast<const unsigned char*>(data);
- uint32_t ssrc = (ptr[8] << 24) + (ptr[9] << 16) + (ptr[10] << 8) + ptr[11];
- if (ssrc == rtx_ssrc_) count_rtx_ssrc_++;
-
- if (packet_loss_ > 0) {
- if ((count_ % packet_loss_) == 0) {
- return len;
- }
- }
- if (module_->IncomingPacket((const WebRtc_UWord8*)data, len) == 0) {
- return len;
- }
- return -1;
- }
- virtual int SendRTCPPacket(int channel, const void *data, int len) {
- if (module_->IncomingPacket((const WebRtc_UWord8*)data, len) == 0) {
- return len;
- }
- return -1;
- }
- int count_;
- int packet_loss_;
- uint32_t rtx_ssrc_;
- int count_rtx_ssrc_;
- RtpRtcp* module_;
-};
-
-class RtpRtcpNackTest : public ::testing::Test {
- protected:
- RtpRtcpNackTest() {}
- ~RtpRtcpNackTest() {}
-
- virtual void SetUp() {
- video_module_ = RtpRtcp::CreateRtpRtcp(kTestId, false, &fake_clock);
- EXPECT_EQ(0, video_module_->InitReceiver());
- EXPECT_EQ(0, video_module_->InitSender());
- EXPECT_EQ(0, video_module_->SetRTCPStatus(kRtcpCompound));
- EXPECT_EQ(0, video_module_->SetSSRC(kTestSsrc));
- EXPECT_EQ(0, video_module_->SetNACKStatus(kNackRtcp));
- EXPECT_EQ(0, video_module_->SetStorePacketsStatus(true));
- EXPECT_EQ(0, video_module_->SetSendingStatus(true));
- EXPECT_EQ(0, video_module_->SetSequenceNumber(kTestSequenceNumber));
- EXPECT_EQ(0, video_module_->SetStartTimestamp(111111));
-
- transport_ = new NackLoopBackTransport(video_module_, kTestSsrc + 1);
- EXPECT_EQ(0, video_module_->RegisterSendTransport(transport_));
-
- nack_receiver_ = new VerifyingNackReceiver();
- EXPECT_EQ(0, video_module_->RegisterIncomingDataCallback(nack_receiver_));
-
- VideoCodec video_codec;
- memset(&video_codec, 0, sizeof(video_codec));
- video_codec.plType = 123;
- memcpy(video_codec.plName, "I420", 5);
-
- EXPECT_EQ(0, video_module_->RegisterSendPayload(video_codec));
- EXPECT_EQ(0, video_module_->RegisterReceivePayload(video_codec));
-
- payload_data_length = sizeof(payload_data);
-
- for (int n = 0; n < payload_data_length; n++) {
- payload_data[n] = n % 10;
- }
- }
-
- virtual void TearDown() {
- RtpRtcp::DestroyRtpRtcp(video_module_);
- delete transport_;
- delete nack_receiver_;
- }
-
- RtpRtcp* video_module_;
- NackLoopBackTransport* transport_;
- VerifyingNackReceiver* nack_receiver_;
- WebRtc_UWord8 payload_data[65000];
- int payload_data_length;
- FakeRtpRtcpClock fake_clock;
-};
-
-TEST_F(RtpRtcpNackTest, RTCP) {
- WebRtc_UWord32 timestamp = 3000;
- WebRtc_UWord16 nack_list[kVideoNackListSize];
- transport_->DropEveryNthPacket(10);
-
- for (int frame = 0; frame < 10; ++frame) {
- EXPECT_EQ(0, video_module_->SendOutgoingData(webrtc::kVideoFrameDelta, 123,
- timestamp,
- payload_data,
- payload_data_length));
-
- std::sort(nack_receiver_->sequence_numbers_.begin(),
- nack_receiver_->sequence_numbers_.end());
-
- std::vector<WebRtc_UWord16> missing_sequence_numbers;
- std::vector<WebRtc_UWord16>::iterator it =
- nack_receiver_->sequence_numbers_.begin();
-
- while (it != nack_receiver_->sequence_numbers_.end()) {
- WebRtc_UWord16 sequence_number_1 = *it;
- ++it;
- if (it != nack_receiver_->sequence_numbers_.end()) {
- WebRtc_UWord16 sequence_number_2 = *it;
- // Add all missing sequence numbers to list
- for (WebRtc_UWord16 i = sequence_number_1 + 1; i < sequence_number_2;
- ++i) {
- missing_sequence_numbers.push_back(i);
- }
- }
- }
- int n = 0;
- for (it = missing_sequence_numbers.begin();
- it != missing_sequence_numbers.end(); ++it) {
- nack_list[n++] = (*it);
- }
- video_module_->SendNACK(nack_list, n);
- fake_clock.IncrementTime(33);
- video_module_->Process();
-
- // Prepare next frame.
- timestamp += 3000;
- }
- std::sort(nack_receiver_->sequence_numbers_.begin(),
- nack_receiver_->sequence_numbers_.end());
- EXPECT_EQ(kTestSequenceNumber, *(nack_receiver_->sequence_numbers_.begin()));
- EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
- *(nack_receiver_->sequence_numbers_.rbegin()));
- EXPECT_EQ(kTestNumberOfPackets, nack_receiver_->sequence_numbers_.size());
- EXPECT_EQ(0, transport_->count_rtx_ssrc_);
-}
-
-TEST_F(RtpRtcpNackTest, RTX) {
- EXPECT_EQ(0, video_module_->SetRTXReceiveStatus(true, kTestSsrc + 1));
- EXPECT_EQ(0, video_module_->SetRTXSendStatus(true, true, kTestSsrc + 1));
-
- transport_->DropEveryNthPacket(10);
-
- WebRtc_UWord32 timestamp = 3000;
- WebRtc_UWord16 nack_list[kVideoNackListSize];
-
- for (int frame = 0; frame < 10; ++frame) {
- EXPECT_EQ(0, video_module_->SendOutgoingData(webrtc::kVideoFrameDelta, 123,
- timestamp,
- payload_data,
- payload_data_length));
-
- std::sort(nack_receiver_->sequence_numbers_.begin(),
- nack_receiver_->sequence_numbers_.end());
-
- std::vector<WebRtc_UWord16> missing_sequence_numbers;
-
-
- std::vector<WebRtc_UWord16>::iterator it =
- nack_receiver_->sequence_numbers_.begin();
- while (it != nack_receiver_->sequence_numbers_.end()) {
- int sequence_number_1 = *it;
- ++it;
- if (it != nack_receiver_->sequence_numbers_.end()) {
- int sequence_number_2 = *it;
- // Add all missing sequence numbers to list.
- for (int i = sequence_number_1 + 1; i < sequence_number_2; ++i) {
- missing_sequence_numbers.push_back(i);
- }
- }
- }
- int n = 0;
- for (it = missing_sequence_numbers.begin();
- it != missing_sequence_numbers.end(); ++it) {
- nack_list[n++] = (*it);
- }
- video_module_->SendNACK(nack_list, n);
- fake_clock.IncrementTime(33);
- video_module_->Process();
-
- // Prepare next frame.
- timestamp += 3000;
- }
- std::sort(nack_receiver_->sequence_numbers_.begin(),
- nack_receiver_->sequence_numbers_.end());
- EXPECT_EQ(kTestSequenceNumber, *(nack_receiver_->sequence_numbers_.begin()));
- EXPECT_EQ(kTestSequenceNumber + kTestNumberOfPackets - 1,
- *(nack_receiver_->sequence_numbers_.rbegin()));
- EXPECT_EQ(kTestNumberOfPackets, nack_receiver_->sequence_numbers_.size());
- EXPECT_EQ(kTestNumberOfRtxPackets, transport_->count_rtx_ssrc_);
-}
-
diff --git a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
deleted file mode 100644
index 306d47f..0000000
--- a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_rtcp.cc
+++ /dev/null
@@ -1,333 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <algorithm>
-#include <vector>
-#include <gtest/gtest.h>
-
-#include "test_api.h"
-
-#include "common_types.h"
-#include "rtp_rtcp.h"
-#include "rtp_rtcp_defines.h"
-
-using namespace webrtc;
-
-const WebRtc_UWord64 kTestPictureId = 12345678;
-
-class RtcpCallback : public RtcpFeedback {
- public:
- RtcpCallback(RtpRtcp* module) {
- _rtpRtcpModule = module;
- };
- virtual void OnRTCPPacketTimeout(const WebRtc_Word32 id) {
- }
- virtual void OnLipSyncUpdate(const WebRtc_Word32 id,
- const WebRtc_Word32 audioVideoOffset) {
- };
- virtual void OnTMMBRReceived(const WebRtc_Word32 id,
- const WebRtc_UWord16 bwEstimateKbit) {
- };
- virtual void OnXRVoIPMetricReceived(
- const WebRtc_Word32 id,
- const RTCPVoIPMetric* metric,
- const WebRtc_Word8 VoIPmetricBuffer[28]) {
- };
- virtual void OnSLIReceived(const WebRtc_Word32 id,
- const WebRtc_UWord8 pictureId) {
- EXPECT_EQ(28, pictureId);
- };
-
- virtual void OnRPSIReceived(const WebRtc_Word32 id,
- const WebRtc_UWord64 pictureId) {
- EXPECT_EQ(kTestPictureId, pictureId);
- };
- virtual void OnApplicationDataReceived(const WebRtc_Word32 id,
- const WebRtc_UWord8 subType,
- const WebRtc_UWord32 name,
- const WebRtc_UWord16 length,
- const WebRtc_UWord8* data) {
- char print_name[5];
- print_name[0] = static_cast<char>(name >> 24);
- print_name[1] = static_cast<char>(name >> 16);
- print_name[2] = static_cast<char>(name >> 8);
- print_name[3] = static_cast<char>(name);
- print_name[4] = 0;
-
- EXPECT_STRCASEEQ("test", print_name);
- };
-
- virtual void OnSendReportReceived(const WebRtc_Word32 id,
- const WebRtc_UWord32 senderSSRC) {
- RTCPSenderInfo senderInfo;
- EXPECT_EQ(0, _rtpRtcpModule->RemoteRTCPStat(&senderInfo));
- };
-
- virtual void OnReceiveReportReceived(const WebRtc_Word32 id,
- const WebRtc_UWord32 senderSSRC) {
- };
- private:
- RtpRtcp* _rtpRtcpModule;
-};
-
-class RtpRtcpRtcpTest : public ::testing::Test {
- protected:
- RtpRtcpRtcpTest() {
- test_CSRC[0] = 1234;
- test_CSRC[2] = 2345;
- test_id = 123;
- test_ssrc = 3456;
- test_timestamp = 4567;
- test_sequence_number = 2345;
- }
- ~RtpRtcpRtcpTest() {}
-
- virtual void SetUp() {
- module1 = RtpRtcp::CreateRtpRtcp(test_id, true, &fake_clock);
- module2 = RtpRtcp::CreateRtpRtcp(test_id+1, true, &fake_clock);
-
- EXPECT_EQ(0, module1->InitReceiver());
- EXPECT_EQ(0, module1->InitSender());
- EXPECT_EQ(0, module2->InitReceiver());
- EXPECT_EQ(0, module2->InitSender());
- receiver = new RtpReceiver();
- EXPECT_EQ(0, module2->RegisterIncomingDataCallback(receiver));
- transport1 = new LoopBackTransport(module2);
- EXPECT_EQ(0, module1->RegisterSendTransport(transport1));
- transport2 = new LoopBackTransport(module1);
- EXPECT_EQ(0, module2->RegisterSendTransport(transport2));
- }
-
- virtual void TearDown() {
- RtpRtcp::DestroyRtpRtcp(module1);
- RtpRtcp::DestroyRtpRtcp(module2);
- delete transport1;
- delete transport2;
- delete receiver;
- }
-
- void SetUpCallFromModule1(RtcpCallback* feedback1, RtcpCallback* feedback2 ) {
- EXPECT_EQ(0, module1->RegisterIncomingRTCPCallback(feedback1));
- EXPECT_EQ(0, module2->RegisterIncomingRTCPCallback(feedback2));
-
- EXPECT_EQ(0, module1->SetRTCPStatus(kRtcpCompound));
- EXPECT_EQ(0, module2->SetRTCPStatus(kRtcpCompound));
-
- EXPECT_EQ(0, module2->SetSSRC(test_ssrc + 1));
- EXPECT_EQ(0, module1->SetSSRC(test_ssrc));
- EXPECT_EQ(0, module1->SetSequenceNumber(test_sequence_number));
- EXPECT_EQ(0, module1->SetStartTimestamp(test_timestamp));
- EXPECT_EQ(0, module1->SetCSRCs(test_CSRC, 2));
- EXPECT_EQ(0, module1->SetCNAME("john.doe@test.test"));
-
- EXPECT_EQ(0, module1->SetSendingStatus(true));
-
- CodecInst voiceCodec;
- voiceCodec.pltype = 96;
- voiceCodec.plfreq = 8000;
- voiceCodec.rate = 64000;
- memcpy(voiceCodec.plname, "PCMU", 5);
-
- EXPECT_EQ(0, module1->RegisterSendPayload(voiceCodec));
- EXPECT_EQ(0, module1->RegisterReceivePayload(voiceCodec));
- EXPECT_EQ(0, module2->RegisterSendPayload(voiceCodec));
- EXPECT_EQ(0, module2->RegisterReceivePayload(voiceCodec));
-
- // We need to send one RTP packet to get the RTCP packet to be accepted by
- // the receiving module.
- // send RTP packet with the data "testtest"
- const WebRtc_UWord8 test[9] = "testtest";
- EXPECT_EQ(0, module1->SendOutgoingData(webrtc::kAudioFrameSpeech, 96,
- 0, test, 8));
- }
-
- int test_id;
- RtpRtcp* module1;
- RtpRtcp* module2;
- RtpReceiver* receiver;
- LoopBackTransport* transport1;
- LoopBackTransport* transport2;
- WebRtc_UWord32 test_ssrc;
- WebRtc_UWord32 test_timestamp;
- WebRtc_UWord16 test_sequence_number;
- WebRtc_UWord32 test_CSRC[webrtc::kRtpCsrcSize];
- FakeRtpRtcpClock fake_clock;
-};
-
-TEST_F(RtpRtcpRtcpTest, RTCP) {
- RtcpCallback* myRTCPFeedback1 = new RtcpCallback(module1);
- RtcpCallback* myRTCPFeedback2 = new RtcpCallback(module2);
-
- SetUpCallFromModule1(myRTCPFeedback1, myRTCPFeedback2);
-
- EXPECT_EQ(0, module1->SendRTCPReferencePictureSelection(kTestPictureId));
- EXPECT_EQ(0, module1->SendRTCPSliceLossIndication(156));
-
- WebRtc_UWord32 testOfCSRC[webrtc::kRtpCsrcSize];
- EXPECT_EQ(2, module2->RemoteCSRCs(testOfCSRC));
- EXPECT_EQ(test_CSRC[0], testOfCSRC[0]);
- EXPECT_EQ(test_CSRC[1], testOfCSRC[1]);
-
- // Set cname of mixed.
- EXPECT_EQ(0, module1->AddMixedCNAME(test_CSRC[0], "john@192.168.0.1"));
- EXPECT_EQ(0, module1->AddMixedCNAME(test_CSRC[1], "jane@192.168.0.2"));
- EXPECT_EQ(-1, module1->AddMixedCNAME(test_CSRC[0], NULL));
-
- EXPECT_EQ(-1, module1->RemoveMixedCNAME(test_CSRC[0] + 1));
- EXPECT_EQ(0, module1->RemoveMixedCNAME(test_CSRC[1]));
- EXPECT_EQ(0, module1->AddMixedCNAME(test_CSRC[1], "jane@192.168.0.2"));
-
- RTCPReportBlock reportBlock;
- reportBlock.cumulativeLost = 1;
- reportBlock.delaySinceLastSR = 2;
- reportBlock.extendedHighSeqNum = 3;
- reportBlock.fractionLost= 4;
- reportBlock.jitter = 5;
- reportBlock.lastSR = 6;
-
- // Set report blocks.
- EXPECT_EQ(-1, module1->AddRTCPReportBlock(test_CSRC[0], NULL));
- EXPECT_EQ(0, module1->AddRTCPReportBlock(test_CSRC[0], &reportBlock));
-
- reportBlock.lastSR= 7;
- EXPECT_EQ(0, module1->AddRTCPReportBlock(test_CSRC[1], &reportBlock));
-
- WebRtc_UWord32 name = 't' << 24;
- name += 'e' << 16;
- name += 's' << 8;
- name += 't';
- EXPECT_EQ(0, module1->SetRTCPApplicationSpecificData(
- 3,
- name,
- (const WebRtc_UWord8 *)"test test test test test test test test test"\
- " test test test test test test test test test test test test test"\
- " test test test test test test test test test test test test test"\
- " test test test test test test test test test test test test test"\
- " test test test test test test test test test test test test ",
- 300));
-
- // send RTCP packet, triggered by timer
- fake_clock.IncrementTime(7500);
- module1->Process();
- fake_clock.IncrementTime(100);
- module2->Process();
-
- WebRtc_UWord32 receivedNTPsecs = 0;
- WebRtc_UWord32 receivedNTPfrac = 0;
- WebRtc_UWord32 RTCPArrivalTimeSecs = 0;
- WebRtc_UWord32 RTCPArrivalTimeFrac = 0;
- char cName[RTCP_CNAME_SIZE];
-
- EXPECT_EQ(0, module2->RemoteNTP(&receivedNTPsecs, &receivedNTPfrac,
- &RTCPArrivalTimeSecs, &RTCPArrivalTimeFrac));
-
- EXPECT_EQ(-1, module2->RemoteCNAME(module2->RemoteSSRC() + 1, cName));
- EXPECT_EQ(-1, module2->RemoteCNAME(module2->RemoteSSRC(), NULL));
-
- // Check multiple CNAME.
- EXPECT_EQ(0, module2->RemoteCNAME(module2->RemoteSSRC(), cName));
- EXPECT_EQ(0, strncmp(cName, "john.doe@test.test", RTCP_CNAME_SIZE));
-
- EXPECT_EQ(0, module2->RemoteCNAME(test_CSRC[0], cName));
- EXPECT_EQ(0, strncmp(cName, "john@192.168.0.1", RTCP_CNAME_SIZE));
-
- EXPECT_EQ(0, module2->RemoteCNAME(test_CSRC[1], cName));
- EXPECT_EQ(0, strncmp(cName, "jane@192.168.0.2", RTCP_CNAME_SIZE));
-
- // get all report blocks
- std::vector<RTCPReportBlock> report_blocks;
- EXPECT_EQ(-1, module1->RemoteRTCPStat(NULL));
- EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks));
- EXPECT_EQ(1u, report_blocks.size());
- const RTCPReportBlock& reportBlockReceived = report_blocks[0];
-
- float secSinceLastReport =
- static_cast<float>(reportBlockReceived.delaySinceLastSR) / 65536.0f;
- EXPECT_GE(0.101f, secSinceLastReport);
- EXPECT_LE(0.100f, secSinceLastReport);
- EXPECT_EQ(test_sequence_number, reportBlockReceived.extendedHighSeqNum);
- EXPECT_EQ(0, reportBlockReceived.fractionLost);
-
- EXPECT_EQ(static_cast<WebRtc_UWord32>(0),
- reportBlockReceived.cumulativeLost);
-
- WebRtc_UWord8 fraction_lost = 0; // scale 0 to 255
- WebRtc_UWord32 cum_lost = 0; // number of lost packets
- WebRtc_UWord32 ext_max = 0; // highest sequence number received
- WebRtc_UWord32 jitter = 0;
- WebRtc_UWord32 max_jitter = 0;
- EXPECT_EQ(0, module2->StatisticsRTP(&fraction_lost,
- &cum_lost,
- &ext_max,
- &jitter,
- &max_jitter));
- EXPECT_EQ(0, fraction_lost);
- EXPECT_EQ((WebRtc_UWord32)0, cum_lost);
- EXPECT_EQ(test_sequence_number, ext_max);
- EXPECT_EQ(reportBlockReceived.jitter, jitter);
-
- WebRtc_UWord16 RTT;
- WebRtc_UWord16 avgRTT;
- WebRtc_UWord16 minRTT;
- WebRtc_UWord16 maxRTT;
-
- // Get RoundTripTime.
- EXPECT_EQ(0, module1->RTT(test_ssrc + 1, &RTT, &avgRTT, &minRTT, &maxRTT));
- EXPECT_GE(10, RTT);
- EXPECT_GE(10, avgRTT);
- EXPECT_GE(10, minRTT);
- EXPECT_GE(10, maxRTT);
-
- // Set report blocks.
- EXPECT_EQ(0, module1->AddRTCPReportBlock(test_CSRC[0], &reportBlock));
-
- // Test receive report.
- EXPECT_EQ(0, module1->SetSendingStatus(false));
-
- // Test that BYE clears the CNAME
- EXPECT_EQ(-1, module2->RemoteCNAME(module2->RemoteSSRC(), cName));
-
- // Send RTCP packet, triggered by timer.
- fake_clock.IncrementTime(5000);
- module1->Process();
- module2->Process();
-
- delete myRTCPFeedback1;
- delete myRTCPFeedback2;
-}
-
-TEST_F(RtpRtcpRtcpTest, RemoteRTCPStatRemote) {
- std::vector<RTCPReportBlock> report_blocks;
-
- RtcpCallback feedback1(module1);
- RtcpCallback feedback2(module2);
-
- SetUpCallFromModule1(&feedback1, &feedback2);
- EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks));
- EXPECT_EQ(0u, report_blocks.size());
-
- // send RTCP packet, triggered by timer
- fake_clock.IncrementTime(7500);
- module1->Process();
- fake_clock.IncrementTime(100);
- module2->Process();
-
- EXPECT_EQ(0, module1->RemoteRTCPStat(&report_blocks));
- ASSERT_EQ(1u, report_blocks.size());
-
- // |test_ssrc+1| is the SSRC of module2 that send the report.
- EXPECT_EQ(test_ssrc+1, report_blocks[0].remoteSSRC);
- EXPECT_EQ(test_ssrc, report_blocks[0].sourceSSRC);
-
- EXPECT_EQ(0u, report_blocks[0].cumulativeLost);
- EXPECT_LT(0u, report_blocks[0].delaySinceLastSR);
- EXPECT_EQ(test_sequence_number, report_blocks[0].extendedHighSeqNum);
- EXPECT_EQ(0u, report_blocks[0].fractionLost);
-}
diff --git a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_video.cc b/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_video.cc
deleted file mode 100644
index 7a06ef7..0000000
--- a/trunk/src/modules/rtp_rtcp/test/testAPI/test_api_video.cc
+++ /dev/null
@@ -1,90 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <algorithm>
-#include <vector>
-#include <gtest/gtest.h>
-
-#include "test_api.h"
-
-#include "common_types.h"
-#include "rtp_rtcp.h"
-#include "rtp_rtcp_defines.h"
-
-using namespace webrtc;
-
-class RtpRtcpVideoTest : public ::testing::Test {
- protected:
- RtpRtcpVideoTest() {
- test_id = 123;
- test_ssrc = 3456;
- test_timestamp = 4567;
- test_sequence_number = 2345;
- }
- ~RtpRtcpVideoTest() {}
-
- virtual void SetUp() {
- video_module = RtpRtcp::CreateRtpRtcp(test_id, false, &fake_clock);
- EXPECT_EQ(0, video_module->InitReceiver());
- EXPECT_EQ(0, video_module->InitSender());
- EXPECT_EQ(0, video_module->SetRTCPStatus(kRtcpCompound));
- EXPECT_EQ(0, video_module->SetSSRC(test_ssrc));
- EXPECT_EQ(0, video_module->SetNACKStatus(kNackRtcp));
- EXPECT_EQ(0, video_module->SetStorePacketsStatus(true));
- EXPECT_EQ(0, video_module->SetSendingStatus(true));
-
- transport = new LoopBackTransport(video_module);
- EXPECT_EQ(0, video_module->RegisterSendTransport(transport));
-
- receiver = new RtpReceiver();
- EXPECT_EQ(0, video_module->RegisterIncomingDataCallback(receiver));
-
- VideoCodec video_codec;
- memset(&video_codec, 0, sizeof(video_codec));
- video_codec.plType = 123;
- memcpy(video_codec.plName, "I420", 5);
-
- EXPECT_EQ(0, video_module->RegisterSendPayload(video_codec));
- EXPECT_EQ(0, video_module->RegisterReceivePayload(video_codec));
-
- payload_data_length = sizeof(payload_data);
-
- for (int n = 0; n < payload_data_length; n++) {
- payload_data[n] = n%10;
- }
- }
-
- virtual void TearDown() {
- RtpRtcp::DestroyRtpRtcp(video_module);
- delete transport;
- delete receiver;
- }
-
- int test_id;
- RtpRtcp* video_module;
- LoopBackTransport* transport;
- RtpReceiver* receiver;
- WebRtc_UWord32 test_ssrc;
- WebRtc_UWord32 test_timestamp;
- WebRtc_UWord16 test_sequence_number;
- WebRtc_UWord8 payload_data[65000];
- int payload_data_length;
- FakeRtpRtcpClock fake_clock;
-};
-
-TEST_F(RtpRtcpVideoTest, BasicVideo) {
- WebRtc_UWord32 timestamp = 3000;
- EXPECT_EQ(0, video_module->SendOutgoingData(webrtc::kVideoFrameDelta, 123,
- timestamp,
- payload_data,
- payload_data_length));
-
-}
-
diff --git a/trunk/src/modules/rtp_rtcp/test/testFec/test_fec.cc b/trunk/src/modules/rtp_rtcp/test/testFec/test_fec.cc
deleted file mode 100644
index 867c502..0000000
--- a/trunk/src/modules/rtp_rtcp/test/testFec/test_fec.cc
+++ /dev/null
@@ -1,492 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Test application for core FEC algorithm. Calls encoding and decoding
- * functions in ForwardErrorCorrection directly.
- */
-
-#include <cassert>
-#include <cstdio>
-#include <cstdlib>
-#include <cstring>
-#include <ctime>
-#include <list>
-
-#include "forward_error_correction.h"
-#include "forward_error_correction_internal.h"
-
-#include "rtp_utility.h"
-#include "testsupport/fileutils.h"
-
-//#define VERBOSE_OUTPUT
-
-using namespace webrtc;
-
-void ReceivePackets(
- ForwardErrorCorrection::ReceivedPacketList* toDecodeList,
- ForwardErrorCorrection::ReceivedPacketList* receivedPacketList,
- WebRtc_UWord32 numPacketsToDecode, float reorderRate, float duplicateRate);
-
-int main() {
- enum { kMaxNumberMediaPackets = 48 };
- enum { kMaxNumberFecPackets = 48 };
-
- const WebRtc_UWord32 kNumMaskBytesL0 = 2;
- const WebRtc_UWord32 kNumMaskBytesL1 = 6;
-
- // FOR UEP
- const bool kUseUnequalProtection = true;
-
- WebRtc_UWord32 id = 0;
- ForwardErrorCorrection fec(id);
-
- ForwardErrorCorrection::PacketList mediaPacketList;
- ForwardErrorCorrection::PacketList fecPacketList;
- ForwardErrorCorrection::ReceivedPacketList toDecodeList;
- ForwardErrorCorrection::ReceivedPacketList receivedPacketList;
- ForwardErrorCorrection::RecoveredPacketList recoveredPacketList;
- std::list<WebRtc_UWord8*> fecMaskList;
-
- ForwardErrorCorrection::Packet* mediaPacket;
- // Running over only one loss rate to limit execution time.
- const float lossRate[] = {0.5f};
- const WebRtc_UWord32 lossRateSize = sizeof(lossRate)/sizeof(*lossRate);
- const float reorderRate = 0.1f;
- const float duplicateRate = 0.1f;
-
- WebRtc_UWord8 mediaLossMask[kMaxNumberMediaPackets];
- WebRtc_UWord8 fecLossMask[kMaxNumberFecPackets];
- WebRtc_UWord8 fecPacketMasks[kMaxNumberFecPackets][kMaxNumberMediaPackets];
-
- // Seed the random number generator, storing the seed to file in order to
- // reproduce past results.
- const unsigned int randomSeed = static_cast<unsigned int>(time(NULL));
- srand(randomSeed);
- std::string filename = webrtc::test::OutputPath() + "randomSeedLog.txt";
- FILE* randomSeedFile = fopen(filename.c_str(), "a");
- fprintf(randomSeedFile, "%u\n", randomSeed);
- fclose(randomSeedFile);
- randomSeedFile = NULL;
-
- WebRtc_UWord16 seqNum = static_cast<WebRtc_UWord16>(rand());
- WebRtc_UWord32 timeStamp = static_cast<WebRtc_UWord32>(rand());
- const WebRtc_UWord32 ssrc = static_cast<WebRtc_UWord32>(rand());
-
- for (WebRtc_UWord32 lossRateIdx = 0; lossRateIdx < lossRateSize;
- lossRateIdx++) {
- WebRtc_UWord8* packetMask =
- new WebRtc_UWord8[kMaxNumberMediaPackets * kNumMaskBytesL1];
-
- printf("Loss rate: %.2f\n", lossRate[lossRateIdx]);
- for (WebRtc_UWord32 numMediaPackets = 1;
- numMediaPackets <= kMaxNumberMediaPackets;
- numMediaPackets++) {
-
- for (WebRtc_UWord32 numFecPackets = 1;
- numFecPackets <= numMediaPackets &&
- numFecPackets <= kMaxNumberFecPackets;
- numFecPackets++) {
-
- // Loop over numImpPackets: these are usually <= (0.3*numMediaPackets).
- // For this test we check up to ~ (0.5*numMediaPackets).
- WebRtc_UWord32 maxNumImpPackets = numMediaPackets / 2 + 1;
- for (WebRtc_UWord32 numImpPackets = 0;
- numImpPackets <= maxNumImpPackets &&
- numImpPackets <= kMaxNumberMediaPackets;
- numImpPackets++) {
-
- WebRtc_UWord8 protectionFactor = static_cast<WebRtc_UWord8>
- (numFecPackets * 255 / numMediaPackets);
-
- const WebRtc_UWord32 maskBytesPerFecPacket =
- (numMediaPackets > 16) ? kNumMaskBytesL1 : kNumMaskBytesL0;
-
- memset(packetMask, 0, numMediaPackets * maskBytesPerFecPacket);
-
- // Transfer packet masks from bit-mask to byte-mask.
- internal::GeneratePacketMasks(numMediaPackets,
- numFecPackets,
- numImpPackets,
- kUseUnequalProtection,
- packetMask);
-
-#ifdef VERBOSE_OUTPUT
- printf("%u media packets, %u FEC packets, %u numImpPackets, "
- "loss rate = %.2f \n",
- numMediaPackets, numFecPackets, numImpPackets, lossRate[lossRateIdx]);
- printf("Packet mask matrix \n");
-#endif
-
- for (WebRtc_UWord32 i = 0; i < numFecPackets; i++) {
- for (WebRtc_UWord32 j = 0; j < numMediaPackets; j++) {
- const WebRtc_UWord8 byteMask =
- packetMask[i * maskBytesPerFecPacket + j / 8];
- const WebRtc_UWord32 bitPosition = (7 - j % 8);
- fecPacketMasks[i][j] =
- (byteMask & (1 << bitPosition)) >> bitPosition;
-#ifdef VERBOSE_OUTPUT
- printf("%u ", fecPacketMasks[i][j]);
-#endif
- }
-#ifdef VERBOSE_OUTPUT
- printf("\n");
-#endif
- }
-#ifdef VERBOSE_OUTPUT
- printf("\n");
-#endif
- // Check for all zero rows or columns: indicates incorrect mask
- WebRtc_UWord32 rowLimit = numMediaPackets;
- for (WebRtc_UWord32 i = 0; i < numFecPackets; i++) {
- WebRtc_UWord32 rowSum = 0;
- for (WebRtc_UWord32 j = 0; j < rowLimit; j++) {
- rowSum += fecPacketMasks[i][j];
- }
- if (rowSum == 0) {
- printf("ERROR: row is all zero %d \n",i);
- return -1;
- }
- }
- for (WebRtc_UWord32 j = 0; j < rowLimit; j++) {
- WebRtc_UWord32 columnSum = 0;
- for (WebRtc_UWord32 i = 0; i < numFecPackets; i++) {
- columnSum += fecPacketMasks[i][j];
- }
- if (columnSum == 0) {
- printf("ERROR: column is all zero %d \n",j);
- return -1;
- }
- }
- // Construct media packets.
- for (WebRtc_UWord32 i = 0; i < numMediaPackets; i++) {
- mediaPacket = new ForwardErrorCorrection::Packet;
- mediaPacketList.push_back(mediaPacket);
- mediaPacket->length =
- static_cast<WebRtc_UWord16>((static_cast<float>(rand()) /
- RAND_MAX) * (IP_PACKET_SIZE - 12 - 28 -
- ForwardErrorCorrection::PacketOverhead()));
- if (mediaPacket->length < 12) {
- mediaPacket->length = 12;
- }
- // Generate random values for the first 2 bytes
- mediaPacket->data[0] = static_cast<WebRtc_UWord8>(rand() % 256);
- mediaPacket->data[1] = static_cast<WebRtc_UWord8>(rand() % 256);
-
- // The first two bits are assumed to be 10 by the
- // FEC encoder. In fact the FEC decoder will set the
- // two first bits to 10 regardless of what they
- // actually were. Set the first two bits to 10
- // so that a memcmp can be performed for the
- // whole restored packet.
- mediaPacket->data[0] |= 0x80;
- mediaPacket->data[0] &= 0xbf;
-
- // FEC is applied to a whole frame.
- // A frame is signaled by multiple packets without
- // the marker bit set followed by the last packet of
- // the frame for which the marker bit is set.
- // Only push one (fake) frame to the FEC.
- mediaPacket->data[1] &= 0x7f;
-
- ModuleRTPUtility::AssignUWord16ToBuffer(&mediaPacket->data[2],
- seqNum);
- ModuleRTPUtility::AssignUWord32ToBuffer(&mediaPacket->data[4],
- timeStamp);
- ModuleRTPUtility::AssignUWord32ToBuffer(&mediaPacket->data[8],
- ssrc);
- // Generate random values for payload
- for (WebRtc_Word32 j = 12; j < mediaPacket->length; j++) {
- mediaPacket->data[j] =
- static_cast<WebRtc_UWord8> (rand() % 256);
- }
- seqNum++;
- }
- mediaPacket->data[1] |= 0x80;
-
- if (fec.GenerateFEC(mediaPacketList, protectionFactor,
- numImpPackets, kUseUnequalProtection,
- &fecPacketList) != 0) {
- printf("Error: GenerateFEC() failed\n");
- return -1;
- }
-
- if (fecPacketList.size() != numFecPackets) {
- printf("Error: we requested %u FEC packets, "
- "but GenerateFEC() produced %u\n",
- numFecPackets,
- static_cast<WebRtc_UWord32>(fecPacketList.size()));
- return -1;
- }
- memset(mediaLossMask, 0, sizeof(mediaLossMask));
- ForwardErrorCorrection::PacketList::iterator
- mediaPacketListItem = mediaPacketList.begin();
- ForwardErrorCorrection::ReceivedPacket* receivedPacket;
- WebRtc_UWord32 mediaPacketIdx = 0;
-
- while (mediaPacketListItem != mediaPacketList.end()) {
- mediaPacket = *mediaPacketListItem;
- const float lossRandomVariable = (static_cast<float>(rand()) /
- (RAND_MAX));
-
- if (lossRandomVariable >= lossRate[lossRateIdx])
- {
- mediaLossMask[mediaPacketIdx] = 1;
- receivedPacket =
- new ForwardErrorCorrection::ReceivedPacket;
- receivedPacket->pkt = new ForwardErrorCorrection::Packet;
- receivedPacketList.push_back(receivedPacket);
-
- receivedPacket->pkt->length = mediaPacket->length;
- memcpy(receivedPacket->pkt->data, mediaPacket->data,
- mediaPacket->length);
- receivedPacket->seqNum =
- ModuleRTPUtility::BufferToUWord16(&mediaPacket->data[2]);
- receivedPacket->isFec = false;
- }
- mediaPacketIdx++;
- ++mediaPacketListItem;
- }
- memset(fecLossMask, 0, sizeof(fecLossMask));
- ForwardErrorCorrection::PacketList::iterator
- fecPacketListItem = fecPacketList.begin();
- ForwardErrorCorrection::Packet* fecPacket;
- WebRtc_UWord32 fecPacketIdx = 0;
- while (fecPacketListItem != fecPacketList.end()) {
- fecPacket = *fecPacketListItem;
- const float lossRandomVariable =
- (static_cast<float>(rand()) / (RAND_MAX));
- if (lossRandomVariable >= lossRate[lossRateIdx]) {
- fecLossMask[fecPacketIdx] = 1;
- receivedPacket =
- new ForwardErrorCorrection::ReceivedPacket;
- receivedPacket->pkt = new ForwardErrorCorrection::Packet;
-
- receivedPacketList.push_back(receivedPacket);
-
- receivedPacket->pkt->length = fecPacket->length;
- memcpy(receivedPacket->pkt->data, fecPacket->data,
- fecPacket->length);
-
- receivedPacket->seqNum = seqNum;
- receivedPacket->isFec = true;
- receivedPacket->ssrc = ssrc;
-
- fecMaskList.push_back(fecPacketMasks[fecPacketIdx]);
- }
- fecPacketIdx++;
- seqNum++;
- ++fecPacketListItem;
- }
-
-#ifdef VERBOSE_OUTPUT
- printf("Media loss mask:\n");
- for (WebRtc_UWord32 i = 0; i < numMediaPackets; i++) {
- printf("%u ", mediaLossMask[i]);
- }
- printf("\n\n");
-
- printf("FEC loss mask:\n");
- for (WebRtc_UWord32 i = 0; i < numFecPackets; i++) {
- printf("%u ", fecLossMask[i]);
- }
- printf("\n\n");
-#endif
-
- std::list<WebRtc_UWord8*>::iterator fecMaskIt = fecMaskList.begin();
- WebRtc_UWord8* fecMask;
- while (fecMaskIt != fecMaskList.end()) {
- fecMask = *fecMaskIt;
- WebRtc_UWord32 hammingDist = 0;
- WebRtc_UWord32 recoveryPosition = 0;
- for (WebRtc_UWord32 i = 0; i < numMediaPackets; i++) {
- if (mediaLossMask[i] == 0 && fecMask[i] == 1) {
- recoveryPosition = i;
- hammingDist++;
- }
- }
- std::list<WebRtc_UWord8*>::iterator itemToDelete = fecMaskIt;
- ++fecMaskIt;
-
- if (hammingDist == 1) {
- // Recovery possible. Restart search.
- mediaLossMask[recoveryPosition] = 1;
- fecMaskIt = fecMaskList.begin();
- } else if (hammingDist == 0) {
- // FEC packet cannot provide further recovery.
- fecMaskList.erase(itemToDelete);
- }
- }
-#ifdef VERBOSE_OUTPUT
- printf("Recovery mask:\n");
- for (WebRtc_UWord32 i = 0; i < numMediaPackets; i++) {
- printf("%u ", mediaLossMask[i]);
- }
- printf("\n\n");
-#endif
- bool fecPacketReceived = false; // For error-checking frame completion.
- while (!receivedPacketList.empty()) {
- WebRtc_UWord32 numPacketsToDecode = static_cast<WebRtc_UWord32>
- ((static_cast<float>(rand()) / RAND_MAX) *
- receivedPacketList.size() + 0.5);
- if (numPacketsToDecode < 1) {
- numPacketsToDecode = 1;
- }
- ReceivePackets(&toDecodeList, &receivedPacketList,
- numPacketsToDecode, reorderRate, duplicateRate);
-
- if (fecPacketReceived == false) {
- ForwardErrorCorrection::ReceivedPacketList::iterator
- toDecodeIt = toDecodeList.begin();
- while (toDecodeIt != toDecodeList.end()) {
- receivedPacket = *toDecodeIt;
- if (receivedPacket->isFec) {
- fecPacketReceived = true;
- }
- ++toDecodeIt;
- }
- }
- if (fec.DecodeFEC(&toDecodeList, &recoveredPacketList)
- != 0) {
- printf("Error: DecodeFEC() failed\n");
- return -1;
- }
- if (!toDecodeList.empty()) {
- printf("Error: received packet list is not empty\n");
- return -1;
- }
- }
- mediaPacketListItem = mediaPacketList.begin();
- mediaPacketIdx = 0;
- while (mediaPacketListItem != mediaPacketList.end()) {
- if (mediaLossMask[mediaPacketIdx] == 1) {
- // Should have recovered this packet.
- ForwardErrorCorrection::RecoveredPacketList::iterator
- recoveredPacketListItem = recoveredPacketList.begin();
-
- if (recoveredPacketListItem == recoveredPacketList.end()) {
- printf("Error: insufficient number of recovered packets.\n");
- return -1;
- }
- mediaPacket = *mediaPacketListItem;
- ForwardErrorCorrection::RecoveredPacket* recoveredPacket =
- *recoveredPacketListItem;
-
- if (recoveredPacket->pkt->length != mediaPacket->length) {
- printf("Error: recovered packet length not identical to "
- "original media packet\n");
- return -1;
- }
- if (memcmp(recoveredPacket->pkt->data, mediaPacket->data,
- mediaPacket->length) != 0) {
- printf("Error: recovered packet payload not identical to "
- "original media packet\n");
- return -1;
- }
- delete recoveredPacket;
- recoveredPacketList.pop_front();
- }
- mediaPacketIdx++;
- ++mediaPacketListItem;
- }
- fec.ResetState(&recoveredPacketList);
- if (!recoveredPacketList.empty()) {
- printf("Error: excessive number of recovered packets.\n");
- printf("\t size is:%u\n",
- static_cast<WebRtc_UWord32>(recoveredPacketList.size()));
- return -1;
- }
- // -- Teardown --
- mediaPacketListItem = mediaPacketList.begin();
- while (mediaPacketListItem != mediaPacketList.end()) {
- delete *mediaPacketListItem;
- ++mediaPacketListItem;
- mediaPacketList.pop_front();
- }
- assert(mediaPacketList.empty());
-
- fecPacketListItem = fecPacketList.begin();
- while (fecPacketListItem != fecPacketList.end()) {
- ++fecPacketListItem;
- fecPacketList.pop_front();
- }
-
- // Delete received packets we didn't pass to DecodeFEC(), due to early
- // frame completion.
- ForwardErrorCorrection::ReceivedPacketList::iterator
- receivedPacketIt = receivedPacketList.begin();
- while (receivedPacketIt != receivedPacketList.end()) {
- receivedPacket = *receivedPacketIt;
- delete receivedPacket;
- ++receivedPacketIt;
- receivedPacketList.pop_front();
- }
- assert(receivedPacketList.empty());
-
- while (!fecMaskList.empty()) {
- fecMaskList.pop_front();
- }
- timeStamp += 90000 / 30;
- } //loop over numImpPackets
- } //loop over FecPackets
- } //loop over numMediaPackets
- delete [] packetMask;
- } // loop over loss rates
-
- // Have DecodeFEC free allocated memory.
- fec.ResetState(&recoveredPacketList);
- if (!recoveredPacketList.empty()) {
- printf("Error: recovered packet list is not empty\n");
- return -1;
- }
- printf("\nAll tests passed successfully\n");
- return 0;
-}
-
-void ReceivePackets(
- ForwardErrorCorrection::ReceivedPacketList* toDecodeList,
- ForwardErrorCorrection::ReceivedPacketList* receivedPacketList,
- WebRtc_UWord32 numPacketsToDecode, float reorderRate, float duplicateRate) {
- assert(toDecodeList->empty());
- assert(numPacketsToDecode <= receivedPacketList->size());
-
- ForwardErrorCorrection::ReceivedPacketList::iterator it;
- for (WebRtc_UWord32 i = 0; i < numPacketsToDecode; i++) {
- it = receivedPacketList->begin();
- // Reorder packets.
- float randomVariable = static_cast<float>(rand()) / RAND_MAX;
- while (randomVariable < reorderRate) {
- ++it;
- if (it == receivedPacketList->end()) {
- --it;
- break;
- }
- randomVariable = static_cast<float>(rand()) / RAND_MAX;
- }
- ForwardErrorCorrection::ReceivedPacket* receivedPacket = *it;
- toDecodeList->push_back(receivedPacket);
-
- // Duplicate packets.
- randomVariable = static_cast<float>(rand()) / RAND_MAX;
- while (randomVariable < duplicateRate) {
- ForwardErrorCorrection::ReceivedPacket* duplicatePacket =
- new ForwardErrorCorrection::ReceivedPacket;
- *duplicatePacket = *receivedPacket;
- duplicatePacket->pkt = new ForwardErrorCorrection::Packet;
- memcpy(duplicatePacket->pkt->data, receivedPacket->pkt->data,
- receivedPacket->pkt->length);
- duplicatePacket->pkt->length = receivedPacket->pkt->length;
-
- toDecodeList->push_back(duplicatePacket);
- randomVariable = static_cast<float>(rand()) / RAND_MAX;
- }
- receivedPacketList->erase(it);
- }
-}
diff --git a/trunk/src/modules/rtp_rtcp/test/testFec/test_fec.gypi b/trunk/src/modules/rtp_rtcp/test/testFec/test_fec.gypi
deleted file mode 100644
index b7f933b..0000000
--- a/trunk/src/modules/rtp_rtcp/test/testFec/test_fec.gypi
+++ /dev/null
@@ -1,36 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'test_fec',
- 'type': 'executable',
- 'dependencies': [
- 'rtp_rtcp',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- ],
-
- 'include_dirs': [
- '../../source',
- '../../../../system_wrappers/interface',
- ],
-
- 'sources': [
- 'test_fec.cc',
- ],
-
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/rtp_rtcp/test/testH263Parser/testH263Parser.cc b/trunk/src/modules/rtp_rtcp/test/testH263Parser/testH263Parser.cc
deleted file mode 100644
index e4c36d1..0000000
--- a/trunk/src/modules/rtp_rtcp/test/testH263Parser/testH263Parser.cc
+++ /dev/null
@@ -1,580 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cassert>
-#include <windows.h>
-#include <iostream>
-#include <tchar.h>
-
-#include "h263_information.h"
-
-#define TEST_STR "Test H263 parser."
-#define TEST_PASSED() std::cerr << TEST_STR << " : [OK]" << std::endl
-#define PRINT_LINE std::cout << "------------------------------------------" << std::endl;
-
-
-void PrintInfo(const H263Info* ptr)
-{
- std::cout << "info = GOBs: {";
- for (int i = 0; i < ptr->numOfGOBs; i++)
- {
- std::cout << ptr->ptrGOBbuffer[i] << " ";
- }
- std::cout << "}" << std::endl;
- std::cout << " sBit: {";
- for (int i = 0; i < ptr->numOfGOBs; i++)
- {
- std::cout << (int)ptr->ptrGOBbufferSBit[i] << " ";
- }
- std::cout << "}" << std::endl;
-
- std::cout << "uiH263PTypeFmt: " << (int)ptr->uiH263PTypeFmt << std::endl;
- std::cout << " codecBits: " << (int)ptr->codecBits << std::endl;
- std::cout << " pQuant: " << (int)ptr->pQuant << std::endl;
- std::cout << " fType: " << (int)ptr->fType << std::endl;
- std::cout << " cpmBit: " << (int)ptr->cpmBit << std::endl;
- std::cout << " numOfGOBs: " << (int)ptr->numOfGOBs << std::endl;
- std::cout << " numOfMBs: " << (int)ptr->totalNumOfMBs << std::endl;
- std::cout << " " << std::endl;
-};
-
-void PrintMBInfo(
- const H263Info* ptr,
- const H263MBInfo* ptrMB)
-{
- std::cout << " gQuant: {";
- for (int i = 0; i < ptr->numOfGOBs; i++)
- {
- std::cout << (int)ptr->ptrGQuant[i] << " ";
- }
- std::cout << "}" << std::endl;
- std::cout << " " << std::endl;
-
- std::cout << "MBs:{";
- int k = 0;
- for (int i = 0; i < ptr->numOfGOBs; i++)
- {
- std::cout << " {";
- for (int j = 0; j < ptr->ptrNumOfMBs[i]; j++)
- {
- std::cout << ptrMB->ptrBuffer[k++] << " ";
- }
- std::cout << "}" << std::endl;
- }
- std::cout << "}" << std::endl;
- PRINT_LINE;
-};
-
-void ValidateResults(
- const H263Info* ptr,
- const H263Info* ptrRef)
-{
- assert(ptr->uiH263PTypeFmt == ptrRef->uiH263PTypeFmt);
- assert(ptr->codecBits == ptrRef->codecBits);
- assert(ptr->pQuant == ptrRef->pQuant);
- assert(ptr->fType == ptrRef->fType);
- assert(ptr->numOfGOBs == ptrRef->numOfGOBs);
- assert(ptr->totalNumOfMBs == ptrRef->totalNumOfMBs);
- for (int i = 0; i < ptr->numOfGOBs; i++)
- {
- assert(ptr->ptrGOBbuffer[i] == ptrRef->ptrGOBbuffer[i]);
- assert(ptr->ptrGOBbufferSBit[i] == ptrRef->ptrGOBbufferSBit[i]);
- }
- PrintInfo(ptr);
-}
-
-void ValidateMBResults(
- const H263Info* ptr,
- const H263MBInfo* ptrMB,
- const H263Info* ptrRef,
- bool printRes = true)
-{
- int offset = 0;
- int numBytes = 0;
- for (int i = 0; i < ptr->numOfGOBs; i++)
- {
- offset = ptr->CalculateMBOffset(i+1);
- numBytes += ptrMB->ptrBuffer[offset - 1] / 8;
- int numBytesRem = ptrMB->ptrBuffer[offset - 1] % 8;
- if (numBytesRem)
- {
- numBytes++;
- }
- assert(ptr->ptrGQuant[i] == ptrRef->ptrGQuant[i]);
- }
- assert(ptr->ptrGOBbuffer[ptr->numOfGOBs] == numBytes);
- assert(unsigned int( ptr->totalNumOfMBs) <= ptrMB->bufferSize);
- if (printRes)
- {
- PrintMBInfo(ptr, ptrMB);
- }
-}
-
-
-int _tmain(int argc, _TCHAR* argv[])
-{
-
- std::string str;
- std::cout << "--------------------------------" << std::endl;
- std::cout << "------- Test H.263 Parser ------" << std::endl;
- std::cout << "--------------------------------" << std::endl;
- std::cout << " " << std::endl;
-
- // -----------------------------------------------------
- // Input data - H.263 encoded stream SQCIF (P-frame)
- // -----------------------------------------------------
- const int lengthRefSQCIF = 77;
- const unsigned char encodedStreamSQCIF[lengthRefSQCIF] = {
- 0x00, 0x00, 0x81, 0xf6, 0x06, 0x04, 0x3f, 0xb7, 0xbc, 0x00, 0x00, 0x86, 0x23,
- 0x5b, 0xdb, 0xdf, 0xb1, 0x93, 0xdb, 0xde, 0xd6, 0xf0, 0x00, 0x00, 0x8a, 0x27,
- 0xdb, 0xcf, 0xad, 0xbe, 0x00, 0x00, 0x8e, 0x27, 0xed, 0xef, 0x80, 0x00, 0x00,
- 0x92, 0x27, 0x6f, 0x7f, 0x80, 0x00, 0x00, 0x96, 0x20, 0xfc, 0xe2, 0xdb, 0xfe,
- 0xb7, 0x7d, 0xea, 0x5f, 0xf8, 0xab, 0xd2, 0xff, 0xf6, 0xc9, 0xe5, 0x5e, 0x97,
- 0xf7, 0xff, 0xad, 0x4f, 0x49, 0x3b, 0xff, 0xd6, 0xa6, 0x75, 0x82, 0x60};
-
- // Expected results
- H263Info infoRefSQCIF;
- infoRefSQCIF.uiH263PTypeFmt = 1;
- infoRefSQCIF.codecBits = 8;
- infoRefSQCIF.pQuant = 4;
- infoRefSQCIF.fType = 1;
- infoRefSQCIF.cpmBit = 0;
- infoRefSQCIF.numOfGOBs = 6;
- infoRefSQCIF.totalNumOfMBs = 8*6;
-
- infoRefSQCIF.ptrGOBbuffer[0] = 0; infoRefSQCIF.ptrGOBbufferSBit[0] = 0; infoRefSQCIF.ptrGQuant[0] = 0;
- infoRefSQCIF.ptrGOBbuffer[1] = 9; infoRefSQCIF.ptrGOBbufferSBit[1] = 0; infoRefSQCIF.ptrGQuant[1] = 4;
- infoRefSQCIF.ptrGOBbuffer[2] = 22; infoRefSQCIF.ptrGOBbufferSBit[2] = 0; infoRefSQCIF.ptrGQuant[2] = 4;
- infoRefSQCIF.ptrGOBbuffer[3] = 30; infoRefSQCIF.ptrGOBbufferSBit[3] = 0; infoRefSQCIF.ptrGQuant[3] = 4;
- infoRefSQCIF.ptrGOBbuffer[4] = 37; infoRefSQCIF.ptrGOBbufferSBit[4] = 0; infoRefSQCIF.ptrGQuant[4] = 4;
- infoRefSQCIF.ptrGOBbuffer[5] = 44; infoRefSQCIF.ptrGOBbufferSBit[5] = 0; infoRefSQCIF.ptrGQuant[5] = 4;
-
- // ----------------------------------------------------
- // Input data - H.263 encoded stream QCIF (P-frame)
- // ----------------------------------------------------
- const int lengthRefQCIF = 123;
- const unsigned char encodedStreamQCIF[lengthRefQCIF] = {
- 0x00, 0x00, 0x81, 0x02, 0x0a, 0x04, 0x3f, 0xf8, 0x00, 0x00, 0x86, 0x27, 0x8b,
- 0xc6, 0x9f, 0x17, 0x9c, 0x00, 0x00, 0x8a, 0x20, 0xbc, 0x22, 0xf8, 0x5f, 0x46,
- 0x03, 0xc1, 0x77, 0x15, 0xe0, 0xb8, 0x38, 0x3f, 0x05, 0xa0, 0xbf, 0x8f, 0x00,
- 0x00, 0x8e, 0x27, 0xfc, 0x5e, 0x5a, 0x33, 0x80, 0x00, 0x00, 0x92, 0x25, 0x8c,
- 0x1e, 0xbf, 0xfc, 0x7e, 0x35, 0xfc, 0x00, 0x00, 0x96, 0x27, 0xff, 0x00, 0x00,
- 0x9a, 0x20, 0xdb, 0x34, 0xef, 0xfc, 0x00, 0x00, 0x9e, 0x20, 0xaf, 0x17, 0x0d,
- 0x3e, 0xde, 0x0f, 0x8f, 0xff, 0x80, 0x00, 0x00, 0xa2, 0x22, 0xbb, 0x27, 0x81,
- 0xeb, 0xff, 0x5b, 0x07, 0xab, 0xff, 0xad, 0x9e, 0xd8, 0xc9, 0x6b, 0x75, 0x54,
- 0xbf, 0xbe, 0x8a, 0xbd, 0xf2, 0xfb, 0xfb, 0x3d, 0x3d, 0x25, 0xb7, 0xf7, 0xfc,
- 0x92, 0x4c, 0xdb, 0x6d, 0x69, 0xc0};
-
- // Expected results
- H263Info infoRefQCIF;
- infoRefQCIF.uiH263PTypeFmt = 2;
- infoRefQCIF.codecBits = 8;
- infoRefQCIF.pQuant = 4;
- infoRefQCIF.fType = 1;
- infoRefQCIF.cpmBit = 0;
- infoRefQCIF.numOfGOBs = 9;
- infoRefQCIF.totalNumOfMBs = 11*9;
-
- infoRefQCIF.ptrGOBbuffer[0] = 0; infoRefQCIF.ptrGOBbufferSBit[0] = 0; infoRefQCIF.ptrGQuant[0] = 0;
- infoRefQCIF.ptrGOBbuffer[1] = 8; infoRefQCIF.ptrGOBbufferSBit[1] = 0; infoRefQCIF.ptrGQuant[1] = 4;
- infoRefQCIF.ptrGOBbuffer[2] = 17; infoRefQCIF.ptrGOBbufferSBit[2] = 0; infoRefQCIF.ptrGQuant[2] = 4;
- infoRefQCIF.ptrGOBbuffer[3] = 38; infoRefQCIF.ptrGOBbufferSBit[3] = 0; infoRefQCIF.ptrGQuant[3] = 4;
- infoRefQCIF.ptrGOBbuffer[4] = 47; infoRefQCIF.ptrGOBbufferSBit[4] = 0; infoRefQCIF.ptrGQuant[4] = 4;
- infoRefQCIF.ptrGOBbuffer[5] = 58; infoRefQCIF.ptrGOBbufferSBit[5] = 0; infoRefQCIF.ptrGQuant[5] = 4;
- infoRefQCIF.ptrGOBbuffer[6] = 63; infoRefQCIF.ptrGOBbufferSBit[6] = 0; infoRefQCIF.ptrGQuant[6] = 4;
- infoRefQCIF.ptrGOBbuffer[7] = 71; infoRefQCIF.ptrGOBbufferSBit[7] = 0; infoRefQCIF.ptrGQuant[7] = 4;
- infoRefQCIF.ptrGOBbuffer[8] = 84; infoRefQCIF.ptrGOBbufferSBit[8] = 0; infoRefQCIF.ptrGQuant[8] = 4;
-
- // ---------------------------------------------------
- // Input data - H.263 encoded stream CIF (P-frame)
- // ---------------------------------------------------
- const int lengthRefCIF = 212;
- const unsigned char encodedStreamCIF[lengthRefCIF] = {
- 0x00, 0x00, 0x82, 0x9a, 0x0e, 0x04, 0x3f, 0xff, 0xff, 0x00, 0x00, 0x86, 0x27,
- 0xff, 0xff, 0xe0, 0x00, 0x00, 0x8a, 0x27, 0xff, 0xff, 0xe0, 0x00, 0x00, 0x8e,
- 0x27, 0xff, 0xff, 0x6b, 0x09, 0x70, 0x00, 0x00, 0x92, 0x27, 0xff, 0xff, 0xe0,
- 0x00, 0x00, 0x96, 0x27, 0xff, 0xff, 0xe0, 0x00, 0x00, 0x9a, 0x27, 0x6f, 0x7f,
- 0xff, 0xfe, 0x00, 0x00, 0x9e, 0x27, 0xff, 0xfe, 0xc6, 0x31, 0xe0, 0x00, 0x00,
- 0xa2, 0x27, 0xff, 0xff, 0xe0, 0x00, 0x00, 0xa6, 0x27, 0xf6, 0xb7, 0xfe, 0xa6,
- 0x14, 0x95, 0xb4, 0xc6, 0x41, 0x6b, 0x3a, 0x2e, 0x8d, 0x42, 0xef, 0xc0, 0x00,
- 0x00, 0xaa, 0x27, 0xff, 0xb1, 0x95, 0x05, 0x0c, 0xe3, 0x4a, 0x17, 0xff, 0x80,
- 0x00, 0x00, 0xae, 0x27, 0xff, 0xf6, 0xf7, 0xfe, 0x00, 0x00, 0xb2, 0x27, 0xff,
- 0x8b, 0xdf, 0xff, 0x00, 0x00, 0xb6, 0x27, 0xff, 0xff, 0xe0, 0x00, 0x00, 0xba,
- 0x26, 0x2f, 0x7f, 0xff, 0xfb, 0x58, 0x5b, 0x80, 0x00, 0x00, 0xbe, 0x20, 0xbc,
- 0xe4, 0x5e, 0x6f, 0xff, 0xfe, 0xce, 0xf1, 0x94, 0x00, 0x00, 0xc2, 0x23, 0x18,
- 0x4b, 0x17, 0x87, 0x0f, 0xff, 0xb3, 0xb6, 0x09, 0x63, 0x46, 0x73, 0x40, 0xb2,
- 0x5f, 0x0a, 0xc6, 0xbe, 0xd7, 0x7a, 0x61, 0xbc, 0x68, 0xde, 0xf0, 0x00, 0x00,
- 0xc6, 0x26, 0x2f, 0x0a, 0xff, 0xff, 0x17, 0x9a, 0x17, 0x8c, 0xb1, 0x37, 0x67,
- 0xaf, 0xec, 0xf7, 0xa5, 0xbf, 0xb6, 0x49, 0x27, 0x6d, 0xb7, 0x92, 0x49, 0x1b,
- 0xb4, 0x9a, 0xe0, 0x62};
-
- // Expected results
- H263Info infoRefCIF;
- infoRefCIF.uiH263PTypeFmt = 3;
- infoRefCIF.codecBits = 8;
- infoRefCIF.pQuant = 4;
- infoRefCIF.fType = 1;
- infoRefCIF.cpmBit = 0;
- infoRefCIF.numOfGOBs = 18;
- infoRefCIF.totalNumOfMBs = 22*18;
-
- infoRefCIF.ptrGOBbuffer[0] = 0; infoRefCIF.ptrGOBbufferSBit[0] = 0; infoRefCIF.ptrGQuant[0] = 0;
- infoRefCIF.ptrGOBbuffer[1] = 9; infoRefCIF.ptrGOBbufferSBit[1] = 0; infoRefCIF.ptrGQuant[1] = 4;
- infoRefCIF.ptrGOBbuffer[2] = 16; infoRefCIF.ptrGOBbufferSBit[2] = 0; infoRefCIF.ptrGQuant[2] = 4;
- infoRefCIF.ptrGOBbuffer[3] = 23; infoRefCIF.ptrGOBbufferSBit[3] = 0; infoRefCIF.ptrGQuant[3] = 4;
- infoRefCIF.ptrGOBbuffer[4] = 32; infoRefCIF.ptrGOBbufferSBit[4] = 0; infoRefCIF.ptrGQuant[4] = 4;
- infoRefCIF.ptrGOBbuffer[5] = 39; infoRefCIF.ptrGOBbufferSBit[5] = 0; infoRefCIF.ptrGQuant[5] = 4;
- infoRefCIF.ptrGOBbuffer[6] = 46; infoRefCIF.ptrGOBbufferSBit[6] = 0; infoRefCIF.ptrGQuant[6] = 4;
- infoRefCIF.ptrGOBbuffer[7] = 54; infoRefCIF.ptrGOBbufferSBit[7] = 0; infoRefCIF.ptrGQuant[7] = 4;
- infoRefCIF.ptrGOBbuffer[8] = 63; infoRefCIF.ptrGOBbufferSBit[8] = 0; infoRefCIF.ptrGQuant[8] = 4;
- infoRefCIF.ptrGOBbuffer[9] = 70; infoRefCIF.ptrGOBbufferSBit[9] = 0; infoRefCIF.ptrGQuant[9] = 4;
- infoRefCIF.ptrGOBbuffer[10] = 90; infoRefCIF.ptrGOBbufferSBit[10] = 0; infoRefCIF.ptrGQuant[10] = 4;
- infoRefCIF.ptrGOBbuffer[11] = 104; infoRefCIF.ptrGOBbufferSBit[11] = 0; infoRefCIF.ptrGQuant[11] = 4;
- infoRefCIF.ptrGOBbuffer[12] = 112; infoRefCIF.ptrGOBbufferSBit[12] = 0; infoRefCIF.ptrGQuant[12] = 4;
- infoRefCIF.ptrGOBbuffer[13] = 120; infoRefCIF.ptrGOBbufferSBit[13] = 0; infoRefCIF.ptrGQuant[13] = 4;
- infoRefCIF.ptrGOBbuffer[14] = 127; infoRefCIF.ptrGOBbufferSBit[14] = 0; infoRefCIF.ptrGQuant[14] = 4;
- infoRefCIF.ptrGOBbuffer[15] = 138; infoRefCIF.ptrGOBbufferSBit[15] = 0; infoRefCIF.ptrGQuant[15] = 4;
- infoRefCIF.ptrGOBbuffer[16] = 151; infoRefCIF.ptrGOBbufferSBit[16] = 0; infoRefCIF.ptrGQuant[16] = 4;
- infoRefCIF.ptrGOBbuffer[17] = 180; infoRefCIF.ptrGOBbufferSBit[17] = 0; infoRefCIF.ptrGQuant[17] = 4;
-
- // -----------------------------------------------------------------------
- // Input data - H.263 encoded stream QCIF (I-frame). Non byte aligned GOBs
- // -----------------------------------------------------------------------
- const int lengthRefQCIF_N = 2020;
- const unsigned char encodedStreamQCIF_N[lengthRefQCIF_N] = {
- 0x00,0x00,0x80,0x06,0x08,0x07,0x34,0xe4,0xf7,0x66,0x75,0x12,0x9b,0x64,0x83,0xe9,0x4c,0xc7,0x3c,0x77,0x83,0xcf,0x67,0x96,
- 0xe0,0x06,0x69,0x95,0x70,0x60,0x43,0x53,0x96,0x8a,0xa2,0x9e,0x96,0xf8,0x79,0xf0,0xf2,0xb8,0x30,0x21,0xb0,0x0c,0xc0,0x22,
- 0x0e,0x70,0x12,0xb0,0x30,0x21,0x9e,0x03,0x3e,0x02,0x22,0xa8,0x83,0xcf,0x7d,0xe8,0xf3,0x85,0x8d,0x01,0x47,0x29,0x03,0x02,
- 0x16,0x5c,0x06,0x7c,0x84,0x56,0x9c,0x0c,0x08,0x49,0x70,0x29,0xe1,0x68,0x39,0xbf,0xd0,0xef,0x6f,0x6e,0x8e,0x64,0x72,0x70,
- 0x30,0x21,0x03,0xe0,0x53,0x82,0xab,0xe2,0xd4,0xa8,0x3e,0x12,0x80,0xc8,0x2a,0x9f,0x07,0x23,0xdf,0x4f,0xaa,0x5b,0x72,0xaa,
- 0x22,0x81,0x21,0x7a,0x80,0x54,0x85,0x82,0x50,0x17,0x40,0x2a,0x7f,0x3c,0xfe,0xa5,0x49,0x39,0x08,0x18,0x20,0x65,0x95,0x8c,
- 0x8f,0x67,0xc7,0xc0,0x63,0xe1,0x1b,0xf1,0xef,0xfd,0x25,0x13,0x24,0x9c,0xa9,0x1f,0x02,0x9b,0xe1,0x19,0xae,0x97,0x02,0x9a,
- 0x84,0x6f,0xbf,0xfd,0x14,0xf2,0x09,0xe3,0x10,0x38,0x0a,0x61,0x99,0x6e,0xf8,0x14,0xce,0x7e,0xff,0xe7,0xa7,0x8e,0x4f,0x14,
- 0xa9,0x80,0xa6,0x71,0x3a,0x98,0x06,0x1c,0xfc,0xff,0xcd,0x8f,0x18,0x9e,0x24,0x52,0x05,0x9c,0x42,0xa5,0x33,0x9f,0xdf,0x80,
- 0x00,0x21,0x0e,0xf6,0x61,0xe2,0x7a,0x35,0x64,0x8e,0x61,0xa2,0x58,0x2a,0x0b,0x45,0x4c,0xe0,0xa1,0xa0,0xa6,0x83,0x58,0xf0,
- 0x08,0x77,0x3e,0xb5,0xc6,0x05,0x6b,0x6e,0x69,0x06,0x07,0xe6,0x82,0x9b,0xc8,0x7e,0x2d,0x6d,0x06,0x07,0xe3,0xe0,0xa6,0x2e,
- 0x05,0x58,0x39,0xbf,0xd1,0xc7,0x9e,0x15,0x29,0x39,0x31,0xbc,0x18,0x1f,0x62,0xf5,0x95,0x20,0xa0,0xe6,0xff,0x38,0x83,0x03,
- 0xea,0x3f,0x02,0xc8,0x05,0x71,0x31,0xef,0x79,0x49,0x48,0x73,0x06,0x07,0xd4,0x4b,0x5f,0xc0,0xe4,0x81,0x3b,0x03,0x03,0xea,
- 0x24,0x81,0x90,0x55,0x41,0x64,0x2b,0xea,0xd1,0xcc,0xa5,0x99,0x79,0x06,0x07,0xd8,0x49,0x05,0x39,0x64,0x16,0x3d,0x03,0x03,
- 0xee,0x24,0x83,0x0b,0xf9,0x46,0x50,0x27,0xdf,0xd4,0xa6,0xa6,0x3d,0x83,0x03,0xf0,0x25,0x83,0x0b,0xf8,0x19,0xbe,0x83,0x03,
- 0xf2,0x3f,0x06,0x17,0xf1,0x58,0xd6,0x04,0xff,0xfa,0x94,0xb4,0x87,0xe0,0x43,0x2f,0x06,0x17,0xf1,0x58,0x11,0x7b,0xe0,0x21,
- 0x97,0x82,0x9d,0x58,0x11,0x7b,0xf9,0xf8,0x15,0xa2,0x9f,0x40,0x3e,0xf4,0x07,0xd5,0x81,0x9f,0xa1,0x7b,0xca,0x9a,0xbb,0xdf,
- 0xcf,0xc0,0xf3,0xb0,0x09,0x37,0x00,0x97,0x63,0xee,0x47,0xff,0xff,0xe8,0x68,0xa5,0xc6,0x43,0xdf,0x2f,0x06,0xad,0xad,0x3f,
- 0xef,0xf6,0x30,0x03,0xea,0x80,0xcf,0x80,0x8b,0x8a,0xe2,0xb5,0x40,0xa7,0x2e,0x06,0x27,0xf7,0xc2,0xd6,0x47,0x2f,0xc1,0x3f,
- 0x80,0x00,0x44,0x1f,0xb5,0xbd,0xc5,0xed,0x4f,0x70,0x7c,0x43,0xec,0x8e,0xcf,0x6e,0xcf,0x7b,0x88,0x7d,0x9b,0xfc,0x13,0xfc,
- 0x24,0x3b,0xf4,0xe1,0x61,0xe1,0xe2,0x61,0x5f,0xa7,0x13,0x15,0x17,0x1b,0x08,0xff,0x38,0x98,0x98,0xb8,0xb8,0x2f,0xf9,0xc3,
- 0xc2,0xc5,0x43,0xc0,0xff,0xce,0x09,0xfa,0x16,0x06,0x07,0xfe,0x6f,0x4e,0xcf,0xaf,0x30,0x3f,0xf3,0x70,0x6c,0x74,0x6e,0x81,
- 0xff,0x9b,0x33,0x13,0x53,0x2c,0x0f,0xf8,0x00,0x04,0x61,0xf0,0xd4,0xf7,0x17,0xb6,0x3a,0x3e,0x1d,0xfa,0x6f,0x0f,0xaf,0x4f,
- 0xf0,0xef,0xd3,0x81,0x85,0x84,0x88,0x86,0x7e,0x9c,0x4c,0x64,0x64,0x84,0x2b,0xfc,0xe3,0xa4,0x24,0xa4,0xe0,0xdf,0xe7,0x1f,
- 0x1d,0x27,0x25,0x03,0xff,0x38,0xd8,0xb9,0x19,0x0f,0xff,0xf9,0xc4,0x42,0xc6,0xc4,0xff,0xc0,0xce,0x05,0xf6,0x16,0x05,0xfe,
- 0x06,0x6f,0x0e,0x4f,0x8e,0xef,0xf0,0x59,0x6d,0x68,0x72,0x3e,0xd9,0x1f,0xff,0x02,0x00,0x01,0x20,0x7c,0x3a,0x29,0x4c,0xf7,
- 0xa5,0x34,0x0c,0x3d,0xb5,0xae,0x3a,0x1d,0xf5,0x6f,0xc3,0xb0,0x30,0x33,0x82,0x03,0xa0,0x60,0x66,0xbf,0x01,0xc0,0x04,0x5b,
- 0x07,0x04,0x16,0x10,0xf3,0xf2,0xe1,0x94,0x17,0x81,0x82,0xd1,0x54,0x40,0xf0,0x7e,0xb4,0x2d,0x16,0x2d,0x83,0x82,0x0a,0xb8,
- 0x0e,0x08,0x2c,0x11,0xe7,0xf5,0xc5,0x02,0x10,0x96,0xb7,0x8b,0x45,0x91,0x60,0xc0,0x83,0x89,0x60,0xa8,0x07,0x2b,0xf9,0x70,
- 0x1c,0x10,0x65,0xb0,0x70,0x41,0xa0,0x8f,0x3f,0xae,0x28,0x18,0x10,0x91,0xf8,0x19,0x54,0x81,0xf1,0x20,0xc0,0x84,0x8f,0xc0,
- 0xcf,0x90,0x3d,0x6c,0x1c,0x10,0x55,0xa0,0x70,0x41,0x7f,0xcf,0x7f,0xf8,0x90,0x60,0x43,0x0b,0xc1,0x85,0xfb,0x80,0xc4,0x81,
- 0x8a,0xe1,0x81,0x81,0x0c,0x56,0x0c,0x2f,0xd0,0x31,0x40,0x62,0xb5,0x80,0x70,0x41,0x55,0xc1,0xc1,0x05,0x7e,0xff,0x5c,0x18,
- 0x30,0x21,0xbf,0x06,0x17,0xe8,0x18,0xa0,0x31,0x5b,0xec,0x06,0x00,0x34,0x18,0x6f,0xbf,0xa0,0x0c,0x95,0x01,0xc1,0x04,0x51,
- 0x07,0x04,0x11,0xf4,0xf4,0x0f,0x77,0x06,0x04,0x34,0x18,0x6f,0xc5,0x60,0xaa,0x07,0x34,0x09,0xd0,0x18,0x10,0xef,0x03,0x0b,
- 0xf8,0x5e,0x0a,0xa0,0x73,0x40,0x93,0xc7,0x8e,0x4d,0x51,0xe5,0xdc,0xf9,0x03,0xdc,0x01,0x81,0x0e,0x54,0x0a,0x71,0xf8,0x11,
- 0x82,0xc6,0x98,0x0c,0x00,0x69,0x72,0xe5,0xe0,0xe4,0xff,0x25,0x67,0x80,0xcf,0x90,0xb9,0x1f,0x15,0x01,0x9f,0x21,0x73,0xe4,
- 0x17,0x63,0x06,0x04,0x34,0x7c,0x05,0xc0,0x8b,0xd7,0x41,0x81,0x0c,0x12,0x00,0xc0,0x2a,0xfe,0x2d,0x42,0x03,0x80,0x66,0x21,
- 0x71,0xf0,0x1c,0x03,0x30,0x6a,0xfb,0x05,0xd5,0xc1,0x81,0x0c,0x12,0x01,0x4c,0x0a,0xbf,0x8b,0x54,0xa0,0x06,0x09,0x00,0xa6,
- 0x05,0x5f,0xc5,0xa7,0x40,0xa2,0x03,0x2e,0x38,0xd2,0xe2,0x77,0xe8,0x10,0x00,0x09,0x43,0xa6,0x72,0xbf,0x3c,0xea,0x0a,0x1c,
- 0x60,0xb4,0x54,0xee,0x5e,0x6a,0x16,0x21,0xcf,0xb4,0x75,0x06,0x07,0xdb,0xec,0x14,0x0b,0x1d,0xc1,0x81,0xf4,0x1f,0x87,0xc2,
- 0xc8,0x98,0xe8,0x37,0xfa,0x3d,0x83,0x03,0xe6,0x3f,0x27,0x7f,0x06,0x07,0xd4,0x4b,0x03,0x25,0x62,0xd9,0x19,0x38,0x27,0xfe,
- 0xfe,0x0c,0x0f,0xb0,0xfc,0x0c,0xd4,0x3f,0x16,0xbc,0x41,0x24,0xbf,0x04,0x7a,0x04,0x7e,0x2d,0x90,0x74,0x2c,0xfc,0xcc,0x4f,
- 0x28,0xc6,0x09,0xfd,0x6e,0x80,0xa1,0x2f,0x03,0x34,0x08,0xfc,0x5a,0xe2,0x07,0xd5,0x81,0x9a,0x85,0xf0,0x60,0x7c,0x32,0x82,
- 0x06,0x07,0xe4,0x32,0xff,0x77,0xff,0x6f,0x53,0xf2,0x76,0xdd,0x7c,0x08,0x30,0x3e,0xe0,0x15,0xfe,0x0c,0x0f,0xaa,0xa0,0xc1,
- 0xf3,0xff,0x2d,0x6d,0x0f,0xc0,0xc0,0xfa,0x2a,0x0c,0x1f,0x81,0x81,0xf3,0x54,0x20,0xb9,0xee,0x07,0x4c,0x86,0x18,0xdf,0x01,
- 0x81,0xf3,0xf0,0x9d,0xf0,0x18,0x1f,0x30,0xc9,0xe6,0x0b,0x2c,0x0b,0xaf,0x40,0xc0,0xf9,0x86,0x4f,0x00,0xc0,0xf9,0xc2,0x67,
- 0x98,0x3c,0xac,0xaa,0xba,0x83,0x03,0xe5,0x16,0x01,0xce,0x20,0xe0,0xf9,0xbd,0x41,0xed,0x45,0x2c,0xc3,0x68,0x38,0x3e,0x6d,
- 0x00,0xe0,0xfa,0x3d,0xc1,0x00,0x00,0x4c,0x1d,0xab,0xda,0xb3,0x51,0x2f,0xaa,0xcd,0x45,0xc5,0x39,0xf1,0xf3,0x8c,0x90,0x8e,
- 0x92,0x86,0x7f,0x9c,0xa4,0xbc,0xb4,0xcc,0x1f,0xfa,0xb9,0x48,0xf1,0xd0,0x8b,0xb8,0x9b,0x94,0xb6,0x51,0x84,0xbb,0xa6,0x2c,
- 0x52,0xaf,0xde,0xad,0x58,0x2b,0x82,0x32,0xfe,0xb9,0x50,0x60,0x81,0xc1,0x86,0xfb,0x06,0x27,0xf5,0x50,0x39,0xa0,0x93,0x20,
- 0xc1,0x04,0x01,0x81,0xf8,0x31,0x3f,0x7e,0x07,0x34,0x12,0x4a,0x84,0x21,0xf6,0x88,0xa0,0x46,0x0b,0x25,0xc1,0x81,0x08,0x2e,
- 0xe8,0x31,0xbf,0xc3,0x60,0x95,0xe5,0x1f,0x90,0x40,0x73,0x40,0xa0,0xd7,0x35,0xa1,0x00,0x18,0x50,0x42,0xf0,0x62,0x7e,0x81,
- 0xce,0x03,0x9a,0x52,0x01,0xc0,0xc2,0x83,0x2b,0x06,0x27,0xe2,0x83,0x9a,0x05,0x32,0x0c,0x08,0x41,0x72,0xd7,0x51,0xa4,0x2d,
- 0x60,0x59,0x36,0x0c,0x08,0x38,0x94,0xb1,0x6f,0xfa,0xca,0xed,0x62,0x33,0x6f,0x18,0x18,0x42,0xae,0x6c,0x7a,0x01,0x80,0xc2,
- 0x83,0xfc,0x18,0x9f,0x7f,0x8b,0x26,0x47,0xa0,0x80,0x0c,0x28,0x47,0xc1,0x89,0xf7,0x56,0x2c,0x9d,0x05,0x00,0x91,0xa2,0x35,
- 0x2d,0x1c,0xf0,0x5d,0x3a,0x0a,0x01,0x2b,0xa3,0xaa,0x04,0x41,0x3d,0xdc,0x2f,0x85,0x5c,0xb0,0x21,0x80,0x60,0x30,0xa1,0x00,
- 0xc5,0x7c,0x86,0x72,0x80,0xc0,0xfc,0x84,0x00,0x61,0x42,0x3c,0x0c,0x4f,0xd8,0x67,0x36,0x0c,0x08,0x38,0xfb,0x04,0x60,0x55,
- 0x83,0x45,0xfe,0x6a,0x5c,0x14,0x02,0x50,0x29,0x86,0x6e,0xfc,0x5c,0x5b,0x0c,0x79,0x10,0x60,0x7e,0x02,0x10,0x30,0xa1,0x05,
- 0xc0,0x45,0xf1,0xa0,0xc0,0xfb,0x89,0x00,0xc2,0x84,0x03,0x80,0x0d,0x25,0x04,0xa1,0xf0,0x29,0x86,0x31,0xc0,0x84,0x3e,0x05,
- 0x34,0x06,0xb0,0x04,0xd3,0xb4,0x36,0x85,0xcb,0x88,0x06,0x07,0xdc,0x4a,0x06,0x14,0x20,0x21,0x20,0x15,0xc2,0x03,0x03,0xec,
- 0x3e,0x06,0x14,0x20,0x20,0x01,0x01,0x5c,0x40,0x1e,0x1f,0x02,0x9b,0xc8,0x01,0x7d,0x07,0x15,0x0f,0x81,0x4d,0xe0,0x20,0x0b,
- 0x37,0x74,0x25,0xa0,0xb2,0x86,0x5b,0xf8,0x30,0x3e,0xc5,0xc0,0xc2,0x83,0x84,0x00,0x62,0x40,0xc5,0x6f,0x10,0x03,0x7c,0x0c,
- 0x28,0x28,0x91,0x01,0xc8,0x04,0x7e,0x50,0x5c,0x0a,0x65,0x48,0x1c,0xeb,0x15,0x17,0x01,0x8f,0x0d,0x1e,0x11,0xfd,0x08,0xae,
- 0x18,0x00,0x02,0x70,0xe5,0x5f,0x95,0x9a,0x8e,0x7e,0x56,0x23,0x3e,0x42,0x31,0xed,0xf3,0x90,0x94,0x92,0x96,0x87,0x7f,0x9c,
- 0xc4,0xdc,0xdc,0xf4,0x1f,0xfa,0x79,0xa9,0x78,0x0c,0x1f,0xd9,0x72,0x32,0xaa,0x1f,0x32,0x6a,0x76,0x7e,0xfe,0x79,0x59,0x75,
- 0xb7,0x83,0xb0,0x62,0xc0,0x9f,0x03,0x0d,0xc0,0xbe,0x19,0x91,0xbc,0x05,0xd4,0x6e,0xf3,0xe3,0x90,0x46,0x25,0x9a,0x06,0x07,
- 0xe0,0x7c,0x06,0x4b,0x28,0x34,0xdf,0xef,0xa2,0xa1,0x05,0x56,0x03,0x07,0xf9,0x00,0x82,0xb4,0x8c,0x26,0x1c,0x12,0xbc,0x23,
- 0xa5,0xa3,0x98,0x8c,0x9a,0x14,0x16,0x2f,0x92,0xad,0x25,0xa0,0x5c,0x3e,0xc4,0x13,0x63,0x6f,0x97,0x05,0x28,0xdb,0x8d,0x08,
- 0x48,0xcf,0x53,0x03,0x03,0xef,0xe1,0xd0,0x28,0xe0,0x31,0x5f,0x8b,0x32,0x8d,0xb4,0xfd,0x68,0xfd,0x58,0x30,0x3e,0xea,0x94,
- 0xe8,0xf6,0x25,0xc5,0x69,0x98,0xc4,0xa1,0x83,0xc6,0x21,0xe2,0xc2,0xb8,0x60,0x68,0x80,0x5c,0xf9,0x75,0x15,0x32,0x05,0xb4,
- 0x11,0x74,0x9a,0x55,0xa8,0x21,0x02,0x30,0x13,0x24,0xac,0x06,0x07,0xd9,0x50,0x19,0x2c,0xa0,0xd2,0xff,0x9f,0xae,0x06,0x07,
- 0xd5,0x56,0x08,0xe0,0xaa,0x4a,0x0a,0xe3,0x4e,0xa8,0x83,0x96,0x0f,0x43,0x83,0x43,0xfe,0xf5,0xc9,0x83,0x5a,0x03,0x18,0x07,
- 0x08,0x0a,0x44,0x1b,0x50,0x12,0x5a,0xe0,0x60,0x7d,0x4b,0x97,0xf8,0xd2,0xb0,0x18,0x1f,0x42,0xe5,0xfe,0x34,0x75,0x0a,0xe2,
- 0x3c,0x80,0x3e,0x7f,0xf1,0x80,0xd9,0x80,0x9d,0xac,0x06,0x07,0xd1,0x50,0x17,0x07,0xcf,0xfe,0xa4,0x18,0x1f,0x45,0x40,0x5c,
- 0x23,0x76,0x88,0xdc,0x3c,0x18,0x70,0x52,0x20,0x9a,0xa7,0x06,0x07,0xcd,0x52,0x70,0x8e,0x90,0x18,0x1f,0x32,0xe0,0x2e,0x58,
- 0x39,0x20,0x76,0x88,0x3b,0xf1,0xf7,0x7d,0x04,0x89,0xf0,0x60,0x7c,0x8b,0x81,0x52,0x82,0x03,0x1b,0xfc,0x1d,0x12,0xcc,0x03,
- 0x03,0xe3,0xe5,0x83,0x37,0x88,0x70,0xe0,0x56,0x00,0x01,0x40,0x73,0x4f,0xbf,0x11,0x9f,0x22,0xfe,0xc5,0x22,0x8f,0x20,0x8d,
- 0x7b,0x7e,0x65,0x26,0x64,0x78,0x4d,0x2d,0x6c,0x26,0x87,0x7f,0xf4,0x0f,0xa7,0x7c,0xe5,0xb1,0x68,0x5a,0x2d,0xa2,0xdf,0x01,
- 0x82,0xd7,0xc1,0x41,0x7a,0xf7,0xdd,0x7f,0xad,0xae,0x2e,0xa3,0x57,0x7c,0x06,0x23,0x40,0x5f,0x89,0x9b,0x46,0x43,0x6b,0xfc,
- 0xff,0x32,0x44,0x9a,0x53,0x5e,0xfb,0x0b,0xf2,0x6e,0x03,0x8b,0xfd,0xa3,0x6c,0x07,0x17,0xfb,0x1a,0xaa,0x76,0x60,0xf1,0xcf,
- 0x10,0xff,0x5b,0x60,0x38,0xbf,0xdb,0x3b,0x17,0x18,0xe7,0x8f,0x1d,0xa4,0x3c,0x0c,0x9d,0x22,0x3e,0xd7,0x00,0xc9,0x63,0xf7,
- 0x38,0xb8,0xc7,0x4c,0x78,0xed,0x31,0xe3,0x9d,0x22,0x3e,0xe7,0x17,0x19,0x6d,0xec,0x27,0xd0,0x8a,0xbc,0x3b,0xe3,0x05,0x2e,
- 0xd2,0x51,0xe2,0x67,0x58,0x8f,0xb5,0xb0,0x97,0x63,0x61,0x2e,0x82,0x88,0x4d,0x9a,0xa3,0xc4,0xce,0x91,0x3f,0x57,0x61,0x2e,
- 0x96,0xc2,0x5c,0x95,0x10,0x9b,0x13,0x21,0x33,0xa4,0x4f,0xcd,0x9f,0x84,0xd8,0x92,0xc5,0x86,0x77,0xd2,0x3a,0xc7,0xc2,0x32,
- 0x38,0x2f,0x73,0x89};
-
- // Expected results
- H263Info infoRefQCIF_N;
- infoRefQCIF_N.uiH263PTypeFmt = 2;
- infoRefQCIF_N.codecBits = 0;
- infoRefQCIF_N.pQuant = 7;
- infoRefQCIF_N.fType = 0;
- infoRefQCIF_N.cpmBit = 0;
- infoRefQCIF_N.numOfGOBs = 9;
- infoRefQCIF_N.totalNumOfMBs = 11*9;
-
- infoRefQCIF_N.ptrGOBbuffer[0] = 0; infoRefQCIF_N.ptrGOBbufferSBit[0] = 0; infoRefQCIF_N.ptrGQuant[0] = 0;
- infoRefQCIF_N.ptrGOBbuffer[1] = 215; infoRefQCIF_N.ptrGOBbufferSBit[1] = 2; infoRefQCIF_N.ptrGQuant[1] = 7;
- infoRefQCIF_N.ptrGOBbuffer[2] = 456; infoRefQCIF_N.ptrGOBbufferSBit[2] = 1; infoRefQCIF_N.ptrGQuant[2] = 7;
- infoRefQCIF_N.ptrGOBbuffer[3] = 535; infoRefQCIF_N.ptrGOBbufferSBit[3] = 5; infoRefQCIF_N.ptrGQuant[3] = 7;
- infoRefQCIF_N.ptrGOBbuffer[4] = 615; infoRefQCIF_N.ptrGOBbufferSBit[4] = 7; infoRefQCIF_N.ptrGQuant[4] = 7;
- infoRefQCIF_N.ptrGOBbuffer[5] = 925; infoRefQCIF_N.ptrGOBbufferSBit[5] = 4; infoRefQCIF_N.ptrGQuant[5] = 7;
- infoRefQCIF_N.ptrGOBbuffer[6] = 1133; infoRefQCIF_N.ptrGOBbufferSBit[6] = 1; infoRefQCIF_N.ptrGQuant[6] = 7;
- infoRefQCIF_N.ptrGOBbuffer[7] = 1512; infoRefQCIF_N.ptrGOBbufferSBit[7] = 6; infoRefQCIF_N.ptrGQuant[7] = 7;
- infoRefQCIF_N.ptrGOBbuffer[8] = 1832; infoRefQCIF_N.ptrGOBbufferSBit[8] = 7; infoRefQCIF_N.ptrGQuant[8] = 7;
-
- // --------------------------------------------------
- // Input data - H.263 encoded stream CIF (I-frame)
- // --------------------------------------------------
-
- FILE* openFile = fopen("H263Foreman_CIF_Iframe.bin", "rb");
-
- fseek(openFile, 0, SEEK_END);
- int lengthRefCIF_I = ftell(openFile);
- fseek(openFile, 0, SEEK_SET);
-
- unsigned char* encodedStreamCIF_I = new unsigned char[lengthRefCIF_I];
- fread(encodedStreamCIF_I, 1, lengthRefCIF_I, openFile);
- fclose(openFile);
-
- // Expected results
- H263Info infoRefCIF_I;
- infoRefCIF_I.uiH263PTypeFmt = 3;
- infoRefCIF_I.codecBits = 0;
- infoRefCIF_I.pQuant = 5;
- infoRefCIF_I.fType = 0;
- infoRefCIF_I.cpmBit = 0;
- infoRefCIF_I.numOfGOBs = 18;
- infoRefCIF_I.totalNumOfMBs = 22*18;
-
- infoRefCIF_I.ptrGOBbuffer[0] = 0; infoRefCIF_I.ptrGOBbufferSBit[0] = 0; infoRefCIF_I.ptrGQuant[0] = 0;
- infoRefCIF_I.ptrGOBbuffer[1] = 1607; infoRefCIF_I.ptrGOBbufferSBit[1] = 0; infoRefCIF_I.ptrGQuant[1] = 5;
- infoRefCIF_I.ptrGOBbuffer[2] = 2759; infoRefCIF_I.ptrGOBbufferSBit[2] = 0; infoRefCIF_I.ptrGQuant[2] = 5;
- infoRefCIF_I.ptrGOBbuffer[3] = 3699; infoRefCIF_I.ptrGOBbufferSBit[3] = 0; infoRefCIF_I.ptrGQuant[3] = 5;
- infoRefCIF_I.ptrGOBbuffer[4] = 4506; infoRefCIF_I.ptrGOBbufferSBit[4] = 0; infoRefCIF_I.ptrGQuant[4] = 5;
- infoRefCIF_I.ptrGOBbuffer[5] = 5260; infoRefCIF_I.ptrGOBbufferSBit[5] = 0; infoRefCIF_I.ptrGQuant[5] = 5;
- infoRefCIF_I.ptrGOBbuffer[6] = 6254; infoRefCIF_I.ptrGOBbufferSBit[6] = 0; infoRefCIF_I.ptrGQuant[6] = 5;
- infoRefCIF_I.ptrGOBbuffer[7] = 7117; infoRefCIF_I.ptrGOBbufferSBit[7] = 0; infoRefCIF_I.ptrGQuant[7] = 5;
- infoRefCIF_I.ptrGOBbuffer[8] = 7804; infoRefCIF_I.ptrGOBbufferSBit[8] = 0; infoRefCIF_I.ptrGQuant[8] = 5;
- infoRefCIF_I.ptrGOBbuffer[9] = 8600; infoRefCIF_I.ptrGOBbufferSBit[9] = 0; infoRefCIF_I.ptrGQuant[9] = 5;
- infoRefCIF_I.ptrGOBbuffer[10] = 9419; infoRefCIF_I.ptrGOBbufferSBit[10] = 0; infoRefCIF_I.ptrGQuant[10] = 5;
- infoRefCIF_I.ptrGOBbuffer[11] = 10126; infoRefCIF_I.ptrGOBbufferSBit[11] = 0; infoRefCIF_I.ptrGQuant[11] = 5;
- infoRefCIF_I.ptrGOBbuffer[12] = 10723; infoRefCIF_I.ptrGOBbufferSBit[12] = 0; infoRefCIF_I.ptrGQuant[12] = 5;
- infoRefCIF_I.ptrGOBbuffer[13] = 11280; infoRefCIF_I.ptrGOBbufferSBit[13] = 0; infoRefCIF_I.ptrGQuant[13] = 5;
- infoRefCIF_I.ptrGOBbuffer[14] = 11910; infoRefCIF_I.ptrGOBbufferSBit[14] = 0; infoRefCIF_I.ptrGQuant[14] = 5;
- infoRefCIF_I.ptrGOBbuffer[15] = 12430; infoRefCIF_I.ptrGOBbufferSBit[15] = 0; infoRefCIF_I.ptrGQuant[15] = 5;
- infoRefCIF_I.ptrGOBbuffer[16] = 12925; infoRefCIF_I.ptrGOBbufferSBit[16] = 0; infoRefCIF_I.ptrGQuant[16] = 5;
- infoRefCIF_I.ptrGOBbuffer[17] = 13506; infoRefCIF_I.ptrGOBbufferSBit[17] = 0; infoRefCIF_I.ptrGQuant[17] = 5;
-
- // --------------------------------------------------
- // Input data - H.263 encoded stream CIF (P-frame)
- // --------------------------------------------------
-
- openFile = fopen("H263Foreman_CIF_Pframe.bin", "rb");
-
- fseek(openFile, 0, SEEK_END);
- int lengthRefCIF_P = ftell(openFile);
- fseek(openFile, 0, SEEK_SET);
-
- unsigned char* encodedStreamCIF_P = new unsigned char[lengthRefCIF_P];
- fread(encodedStreamCIF_P, 1, lengthRefCIF_P, openFile);
- fclose(openFile);
-
- // Expected results
- H263Info infoRefCIF_P;
- infoRefCIF_P.uiH263PTypeFmt = 3;
- infoRefCIF_P.codecBits = 8;
- infoRefCIF_P.pQuant = 4;
- infoRefCIF_P.fType = 1;
- infoRefCIF_P.cpmBit = 0;
- infoRefCIF_P.numOfGOBs = 18;
- infoRefCIF_P.totalNumOfMBs = 22*18;
-
- infoRefCIF_P.ptrGOBbuffer[0] = 0; infoRefCIF_P.ptrGOBbufferSBit[0] = 0; infoRefCIF_P.ptrGQuant[0] = 0;
- infoRefCIF_P.ptrGOBbuffer[1] = 252; infoRefCIF_P.ptrGOBbufferSBit[1] = 0; infoRefCIF_P.ptrGQuant[1] = 5;
- infoRefCIF_P.ptrGOBbuffer[2] = 482; infoRefCIF_P.ptrGOBbufferSBit[2] = 0; infoRefCIF_P.ptrGQuant[2] = 6;
- infoRefCIF_P.ptrGOBbuffer[3] = 581; infoRefCIF_P.ptrGOBbufferSBit[3] = 0; infoRefCIF_P.ptrGQuant[3] = 6;
- infoRefCIF_P.ptrGOBbuffer[4] = 676; infoRefCIF_P.ptrGOBbufferSBit[4] = 0; infoRefCIF_P.ptrGQuant[4] = 7;
- infoRefCIF_P.ptrGOBbuffer[5] = 756; infoRefCIF_P.ptrGOBbufferSBit[5] = 0; infoRefCIF_P.ptrGQuant[5] = 7;
- infoRefCIF_P.ptrGOBbuffer[6] = 855; infoRefCIF_P.ptrGOBbufferSBit[6] = 0; infoRefCIF_P.ptrGQuant[6] = 8;
- infoRefCIF_P.ptrGOBbuffer[7] = 949; infoRefCIF_P.ptrGOBbufferSBit[7] = 0; infoRefCIF_P.ptrGQuant[7] = 9;
- infoRefCIF_P.ptrGOBbuffer[8] = 1004; infoRefCIF_P.ptrGOBbufferSBit[8] = 0; infoRefCIF_P.ptrGQuant[8] = 10;
- infoRefCIF_P.ptrGOBbuffer[9] = 1062; infoRefCIF_P.ptrGOBbufferSBit[9] = 0; infoRefCIF_P.ptrGQuant[9] = 11;
- infoRefCIF_P.ptrGOBbuffer[10] = 1115; infoRefCIF_P.ptrGOBbufferSBit[10] = 0; infoRefCIF_P.ptrGQuant[10] = 11;
- infoRefCIF_P.ptrGOBbuffer[11] = 1152; infoRefCIF_P.ptrGOBbufferSBit[11] = 0; infoRefCIF_P.ptrGQuant[11] = 13;
- infoRefCIF_P.ptrGOBbuffer[12] = 1183; infoRefCIF_P.ptrGOBbufferSBit[12] = 0; infoRefCIF_P.ptrGQuant[12] = 14;
- infoRefCIF_P.ptrGOBbuffer[13] = 1214; infoRefCIF_P.ptrGOBbufferSBit[13] = 0; infoRefCIF_P.ptrGQuant[13] = 15;
- infoRefCIF_P.ptrGOBbuffer[14] = 1257; infoRefCIF_P.ptrGOBbufferSBit[14] = 0; infoRefCIF_P.ptrGQuant[14] = 16;
- infoRefCIF_P.ptrGOBbuffer[15] = 1286; infoRefCIF_P.ptrGOBbufferSBit[15] = 0; infoRefCIF_P.ptrGQuant[15] = 16;
- infoRefCIF_P.ptrGOBbuffer[16] = 1321; infoRefCIF_P.ptrGOBbufferSBit[16] = 0; infoRefCIF_P.ptrGQuant[16] = 16;
- infoRefCIF_P.ptrGOBbuffer[17] = 1352; infoRefCIF_P.ptrGOBbufferSBit[17] = 0; infoRefCIF_P.ptrGQuant[17] = 14;
-
- //---------------------------------------------------------------
- //---------------------------------------------------------------
- //---------------------------------------------------------------
- // Start test
- const H263Info* ptrInfoSQCIF = NULL;
- const H263MBInfo* ptrMBInfoSQCIF = NULL;
- const H263Info* ptrInfoQCIF = NULL;
- const H263MBInfo* ptrMBInfoQCIF = NULL;
- const H263Info* ptrInfoCIF = NULL;
- const H263MBInfo* ptrMBInfoCIF = NULL;
- const H263Info* ptrInfoQCIF_N = NULL;
- const H263MBInfo* ptrMBInfoQCIF_N = NULL;
- const H263Info* ptrInfoCIF_I = NULL;
- const H263MBInfo* ptrMBInfoCIF_I = NULL;
- const H263Info* ptrInfoCIF_P = NULL;
- const H263MBInfo* ptrMBInfoCIF_P = NULL;
- H263Information h263Information;
-
- // Input buffer
- const int length = 3000;
- unsigned char* encodedBuffer = new unsigned char[lengthRefCIF_P];
-
- // Test invalid inputs
- assert(-1 == h263Information.GetInfo(NULL, length, ptrInfoSQCIF));
- assert(-1 == h263Information.GetInfo(encodedBuffer, 0, ptrInfoSQCIF));
- assert(-1 == h263Information.GetInfo(encodedBuffer, length, ptrInfoSQCIF)); // invalid H.263 stream
-// assert(-1 == h263Information.GetInfo(encodedStreamSQCIF, lengthRefSQCIF/2, ptrInfoSQCIF)); // invalid H.263 stream
-
- assert(-1 == h263Information.GetMBInfo(NULL, length, 0, ptrMBInfoSQCIF));
- assert(-1 == h263Information.GetMBInfo(encodedBuffer, 0, 0, ptrMBInfoSQCIF));
- assert(-1 == h263Information.GetMBInfo(encodedBuffer, length, 0, ptrMBInfoSQCIF));
- assert(-1 == h263Information.GetMBInfo(encodedBuffer, length, -1, ptrMBInfoSQCIF)); // incorrect group number
- assert(-1 == h263Information.GetMBInfo(encodedBuffer, length, 8, ptrMBInfoSQCIF)); // incorrect group number
-
- // ----------------------------------------------
- // Get info from encoded H.263 stream - SQCIF
- // ----------------------------------------------
- h263Information.Reset();
- assert(0 == h263Information.GetInfo(encodedStreamSQCIF, lengthRefSQCIF, ptrInfoSQCIF));
- ValidateResults(ptrInfoSQCIF, &infoRefSQCIF);
-
- // Get MB info
- for (int i = 0; i < ptrInfoSQCIF->numOfGOBs; i++)
- {
- assert(0 == h263Information.GetMBInfo(encodedStreamSQCIF, lengthRefSQCIF, i, ptrMBInfoSQCIF));
- }
- ValidateMBResults(ptrInfoSQCIF, ptrMBInfoSQCIF, &infoRefSQCIF);
-
- // ---------------------------------------------
- // Get info from encoded H.263 stream - QCIF
- // ---------------------------------------------
- h263Information.Reset();
- assert(0 == h263Information.GetInfo(encodedStreamQCIF, lengthRefQCIF, ptrInfoQCIF));
- ValidateResults(ptrInfoQCIF, &infoRefQCIF);
-
- // Get MB info
- for (int i = 0; i < ptrInfoQCIF->numOfGOBs; i++)
- {
- assert(0 == h263Information.GetMBInfo(encodedStreamQCIF, lengthRefQCIF, i, ptrMBInfoQCIF));
- }
- ValidateMBResults(ptrInfoQCIF, ptrMBInfoQCIF, &infoRefQCIF);
-
- // --------------------------------------------
- // Get info from encoded H.263 stream - CIF
- // --------------------------------------------
- h263Information.Reset();
- assert(0 == h263Information.GetInfo(encodedStreamCIF, lengthRefCIF, ptrInfoCIF));
- ValidateResults(ptrInfoCIF, &infoRefCIF);
-
- // Get MB info
- for (int i = 0; i < ptrInfoCIF->numOfGOBs; i++)
- {
- assert(0 == h263Information.GetMBInfo(encodedStreamCIF, lengthRefCIF, i, ptrMBInfoCIF));
- }
- ValidateMBResults(ptrInfoCIF, ptrMBInfoCIF, &infoRefCIF);
-
- // ----------------------------------------------------------------------
- // Get info from encoded H.263 stream - QCIF - non byte aligned GOBs
- // ----------------------------------------------------------------------
- h263Information.Reset();
- assert(0 == h263Information.GetInfo(encodedStreamQCIF_N, lengthRefQCIF_N, ptrInfoQCIF_N));
- ValidateResults(ptrInfoQCIF_N, &infoRefQCIF_N);
-
- // Get MB info
- for (int i = 0; i < ptrInfoQCIF_N->numOfGOBs; i++)
- {
- assert(0 == h263Information.GetMBInfo(encodedStreamQCIF_N, lengthRefQCIF_N, i, ptrMBInfoQCIF_N));
- }
- ValidateMBResults(ptrInfoQCIF_N, ptrMBInfoQCIF_N, &infoRefQCIF_N);
-
- // -------------------------------------------
- // Get info from encoded H.263 stream - CIF
- // -------------------------------------------
- h263Information.Reset();
- assert(0 == h263Information.GetInfo(encodedStreamCIF_I, lengthRefCIF_I, ptrInfoCIF_I));
- ValidateResults(ptrInfoCIF_I, &infoRefCIF_I);
-
- // Get MB info
- unsigned int start = timeGetTime();
- for (int i = 0; i < ptrInfoCIF_I->numOfGOBs; i++)
- {
- assert(0 == h263Information.GetMBInfo(encodedStreamCIF_I, lengthRefCIF_I, i, ptrMBInfoCIF_I));
- }
- unsigned int endMB = timeGetTime();
- ValidateMBResults(ptrInfoCIF_I, ptrMBInfoCIF_I, &infoRefCIF_I, false);
-
- std::cout << "I-frame, length: " << lengthRefCIF_I << " bytes. Time: " << endMB - start << " ms." << std::endl;
- PRINT_LINE;
-
- // -------------------------------------------
- // Get info from encoded H.263 stream - CIF
- // -------------------------------------------
- h263Information.Reset();
- assert(0 == h263Information.GetInfo(encodedStreamCIF_P, lengthRefCIF_P, ptrInfoCIF_P));
- ValidateResults(ptrInfoCIF_P, &infoRefCIF_P);
-
- // Get MB info
- start = timeGetTime();
- for (int i = 0; i < ptrInfoCIF_P->numOfGOBs; i++)
- {
- assert(0 == h263Information.GetMBInfo(encodedStreamCIF_P, lengthRefCIF_P, i, ptrMBInfoCIF_P));
- }
- endMB = timeGetTime();
- ValidateMBResults(ptrInfoCIF_P, ptrMBInfoCIF_P, &infoRefCIF_P, false);
-
- std::cout << "P-frame, length: " << lengthRefCIF_P << " bytes. Time: " << endMB - start << " ms." << std::endl;
- PRINT_LINE;
-
- delete [] encodedStreamCIF_I;
- delete [] encodedStreamCIF_P;
- delete [] encodedBuffer;
-
- TEST_PASSED();
- ::Sleep(5000);
- return 0;
-}
-
diff --git a/trunk/src/modules/rtp_rtcp/test/testRateControl/testRateControl.cc b/trunk/src/modules/rtp_rtcp/test/testRateControl/testRateControl.cc
deleted file mode 100644
index c282557..0000000
--- a/trunk/src/modules/rtp_rtcp/test/testRateControl/testRateControl.cc
+++ /dev/null
@@ -1,271 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cassert>
-#include <windows.h>
-#include <iostream>
-#include <tchar.h>
-
-#include "rtp_rtcp.h"
-#include "common_types.h"
-#include "RateControlDetector.h"
-/*#include "rtcp_utility.h"
-#include "tmmbr_help.h"*/
-
-#define TEST_STR "Test RateControl."
-#define TEST_PASSED() std::cerr << TEST_STR << " : [OK]" << std::endl
-#define PRINT_LINE std::cout << "------------------------------------------" << std::endl;
-
-
-const int maxFileLen = 200;
-WebRtc_UWord8* dataFile[maxFileLen];
-
-
-struct InputSet
-{
- WebRtc_UWord32 TMMBR;
- WebRtc_UWord32 packetOH;
- WebRtc_UWord32 SSRC;
-};
-
-const InputSet set0 = {220, 80, 11111}; // bitRate, packetOH, ssrc
-const InputSet set1 = {180, 90, 22222};
-const InputSet set2 = {100, 210, 33333};
-const InputSet set3 = { 35, 40, 44444};
-const InputSet set4 = { 40, 60, 55555};
-const InputSet set4_1 = {100, 60, 55555};
-const InputSet set4_2 = { 10, 60, 55555};
-const InputSet set5 = {200, 40, 66666};
-const InputSet set00 = { 0, 40, 66666};
-
-
-
-
-WebRtc_Word32 GetFile(char* fileName)
-{
- if (!fileName[0])
- {
- return 0;
- }
-
- FILE* openFile = fopen(fileName, "rb");
- assert(openFile != NULL);
- fseek(openFile, 0, SEEK_END);
- int len = (WebRtc_Word16)(ftell(openFile));
- rewind(openFile);
- assert(len > 0 && len < maxFileLen);
- fread(dataFile, 1, len, openFile);
- fclose(openFile);
- return len;
-};
-
-
-class LoopBackTransport2 : public webrtc::Transport
-{
-public:
- LoopBackTransport2(RtpRtcp* rtpRtcpModule) :
- _rtpRtcpModule(rtpRtcpModule),
- _cnt(0)
- {
- }
- virtual int SendPacket(int channel, const void *data, int len)
- {
- return _rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len);
- }
- virtual int SendRTCPPacket(int channel, const void *data, int len)
- {
- char fileName[256] = {0};
-
-
- // Get stored rtcp packet w/ TMMBR
- len = GetFile(fileName);
- if (len == 0)
- {
- return 0;
- }
-
- // Send in bitrate request
- return _rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)dataFile, len);
- }
- RtpRtcp* _rtpRtcpModule;
- WebRtc_UWord32 _cnt;
-};
-
-
-class LoopBackTransportVideo : public webrtc::Transport
-{
-public:
- LoopBackTransportVideo(RtpRtcp* rtpRtcpModule) :
- _rtpRtcpModule(rtpRtcpModule),
- _cnt(0)
- {
- }
- virtual int SendPacket(int channel, const void *data, int len)
- {
- return _rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len);
- }
- virtual int SendRTCPPacket(int channel, const void *data, int len)
- {
- char fileName[256] = {0};
-
- strcpy(fileName, "RTCPPacketTMMBR0.bin");
-
- ++_cnt;
-
- // Get stored rtcp packet w/ TMMBR
- len = GetFile(fileName);
- if (len == 0)
- {
- return 0;
- }
-
- // Send in bitrate request*/
- return _rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)dataFile, len);
- }
-
- RtpRtcp* _rtpRtcpModule;
- WebRtc_UWord32 _cnt;
-};
-
-class TestRateControl : private RateControlDetector
-{
-public:
- TestRateControl():RateControlDetector(0)
- {
- }
- ~TestRateControl()
- {
- }
- void Start()
- {
- //Test perfect conditions
- // But only one packet per frame
- SetLastUsedBitRate(500);
- WebRtc_UWord32 rtpTs=1234*90;
- WebRtc_UWord32 framePeriod=33; // In Ms
- WebRtc_UWord32 rtpDelta=framePeriod*90;
- WebRtc_UWord32 netWorkDelay=10;
- WebRtc_UWord32 arrivalTime=rtpTs/90+netWorkDelay;
- WebRtc_UWord32 newBitRate=0;
- for(WebRtc_UWord32 k=0;k<10;k++)
- {
- // Receive 10 packets
- for(WebRtc_UWord32 i=0;i<10;i++)
- {
- NotifyNewArrivedPacket(rtpTs,arrivalTime);
- rtpTs+=rtpDelta;
- arrivalTime=rtpTs/90+netWorkDelay;
- }
- newBitRate=RateControl(2*netWorkDelay);
- SetLastUsedBitRate(newBitRate);
- Sleep(10*framePeriod);
- std::cout << "RTCP Packet " << k << " new bitrate " << newBitRate << std::endl;
- }
- Reset();
-
-
- //Test increasing RTT
- std::cout << "Test increasing RTT - No Receive timing changes" << std::endl;
- SetLastUsedBitRate(500);
-
- for(WebRtc_UWord32 k=0;k<10;k++)
- {
- // Receive 10 packets
- for(WebRtc_UWord32 i=0;i<10;i++)
- {
- NotifyNewArrivedPacket(rtpTs,arrivalTime);
- rtpTs+=rtpDelta;
- arrivalTime=rtpTs/90+netWorkDelay;
- }
- WebRtc_UWord32 rtt=2*netWorkDelay+k*20;
- newBitRate=RateControl(rtt);
- Sleep(10*framePeriod);
- SetLastUsedBitRate(newBitRate);
- std::cout << "RTCP Packet " << k << " RTT "<< rtt << " new bitrate " << newBitRate << std::endl;
-
- }
-
- Reset();
-
-
- //Test increasing RTT
- std::cout << "Test increasing RTT - Changed receive timing" << std::endl;
- SetLastUsedBitRate(500);
-
- for(WebRtc_UWord32 k=0;k<10;k++)
- {
- // Receive 10 packets
- for(WebRtc_UWord32 i=0;i<10;i++)
- {
- NotifyNewArrivedPacket(rtpTs,arrivalTime);
- rtpTs+=rtpDelta;
- arrivalTime=rtpTs/90+netWorkDelay+i+(k*20);
- }
- WebRtc_UWord32 rtt=2*netWorkDelay+k*20;
- newBitRate=RateControl(rtt);
- Sleep(10*framePeriod);
- SetLastUsedBitRate(newBitRate);
- std::cout << "RTCP Packet " << k << " RTT "<< rtt << " new bitrate " << newBitRate << std::endl;
-
- }
-
-
-
- };
-};
-
-class NULLDataZink: public RtpData
-{
- virtual WebRtc_Word32 OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const webrtc::WebRtcRTPHeader* rtpHeader)
- {
- return 0;
- };
-};
-
-
-int _tmain(int argc, _TCHAR* argv[])
-{
-
- std::string str;
- std::cout << "------------------------" << std::endl;
- std::cout << "---Test RateControl ----" << std::endl;
- std::cout << "------------------------" << std::endl;
- std::cout << " " << std::endl;
-
- // --------------------
- // Test TMMBRHelp class
-
- // --------------------
- TestRateControl test;
- test.Start();
-
- printf("RateControl-class test done.\n");
-
- // ------------------------
- // Test RateControl single module
- // ------------------------
- RtpRtcp* rtpRtcpModuleVideo = RtpRtcp::CreateRtpRtcp(0, false);
-
- LoopBackTransportVideo* myLoopBackTransportVideo = new LoopBackTransportVideo(rtpRtcpModuleVideo);
- assert(0 == rtpRtcpModuleVideo->RegisterSendTransport(myLoopBackTransportVideo));
- printf("Multi module test done.\n");
-
-
- RtpRtcp::DestroyRtpRtcp(rtpRtcpModuleVideo);
- delete myLoopBackTransportVideo;
-
- TEST_PASSED();
- ::Sleep(5000);
-
- return 0;
-}
-
diff --git a/trunk/src/modules/rtp_rtcp/test/testTMMBR/testTMMBR.cc b/trunk/src/modules/rtp_rtcp/test/testTMMBR/testTMMBR.cc
deleted file mode 100644
index d1e1572..0000000
--- a/trunk/src/modules/rtp_rtcp/test/testTMMBR/testTMMBR.cc
+++ /dev/null
@@ -1,1034 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cassert>
-#include <windows.h>
-#include <iostream>
-#include <tchar.h>
-
-#include "rtp_rtcp.h"
-#include "common_types.h"
-#include "rtcp_utility.h"
-#include "tmmbr_help.h"
-
-#define TEST_STR "Test TMMBR."
-#define TEST_PASSED() std::cerr << TEST_STR << " : [OK]" << std::endl
-#define PRINT_LINE std::cout << "------------------------------------------" << std::endl;
-
-
-const int maxFileLen = 200;
-WebRtc_UWord8* dataFile[maxFileLen];
-
-
-struct InputSet
-{
- WebRtc_UWord32 TMMBR;
- WebRtc_UWord32 packetOH;
- WebRtc_UWord32 SSRC;
-};
-
-const InputSet set0 = {220, 80, 11111}; // bitRate, packetOH, ssrc
-const InputSet set1 = {180, 90, 22222};
-const InputSet set2 = {100, 210, 33333};
-const InputSet set3 = { 35, 40, 44444};
-const InputSet set4 = { 40, 60, 55555};
-const InputSet set4_1 = {100, 60, 55555};
-const InputSet set4_2 = { 10, 60, 55555};
-const InputSet set5 = {200, 40, 66666};
-const InputSet set00 = { 0, 40, 66666};
-
-const int maxBitrate = 230; // if this is lower than max in the list above test should fail
-
-void Verify(TMMBRSet* boundingSet, int index, InputSet set)
-{
- assert(boundingSet->ptrTmmbrSet[index] == set.TMMBR);
- assert(boundingSet->ptrPacketOHSet[index] == set.packetOH);
- assert(boundingSet->ptrSsrcSet[index] == set.SSRC);
-};
-
-int ParseRTCPPacket(const void *data, int len, TMMBRSet*& boundingSet)
-{
- int numItems = -1;
- RTCPUtility::RTCPParserV2 rtcpParser((const WebRtc_UWord8*)data, len, true);
- RTCPUtility::RTCPPacketTypes pktType = rtcpParser.Begin();
- while (pktType != RTCPUtility::kRtcpNotValidCode)
- {
- const RTCPUtility::RTCPPacket& rtcpPacket = rtcpParser.Packet();
- if (pktType == RTCPUtility::kRtcpRtpfbTmmbnCode)
- {
- assert(0 == rtcpPacket.TMMBN.SenderSSRC);
- assert(0 == rtcpPacket.TMMBN.MediaSSRC);
- numItems = 0;
- }
- if (pktType == RTCPUtility::kRtcpRtpfbTmmbnItemCode)
- {
- boundingSet->ptrTmmbrSet[numItems] = rtcpPacket.TMMBNItem.MaxTotalMediaBitRate;
- boundingSet->ptrPacketOHSet[numItems] = rtcpPacket.TMMBNItem.MeasuredOverhead;
- boundingSet->ptrSsrcSet[numItems] = rtcpPacket.TMMBNItem.SSRC;
- ++numItems;
- }
- pktType = rtcpParser.Iterate();
- }
- return numItems;
-};
-
-WebRtc_Word32 GetFile(char* fileName)
-{
- if (!fileName[0])
- {
- return 0;
- }
-
- FILE* openFile = fopen(fileName, "rb");
- assert(openFile != NULL);
- fseek(openFile, 0, SEEK_END);
- int len = (WebRtc_Word16)(ftell(openFile));
- rewind(openFile);
- assert(len > 0 && len < maxFileLen);
- fread(dataFile, 1, len, openFile);
- fclose(openFile);
- return len;
-};
-
-
-class LoopBackTransport2 : public webrtc::Transport, private TMMBRHelp
-{
-public:
- LoopBackTransport2(RtpRtcp* rtpRtcpModule) :
- TMMBRHelp(false),
- _rtpRtcpModule(rtpRtcpModule),
- _cnt(0)
- {
- }
- virtual int SendPacket(int channel, const void *data, int len)
- {
- if( 0 == _rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len))
- {
- return len;
- }
- return -1;
- }
- virtual int SendRTCPPacket(int channel, const void *data, int len)
- {
- char fileName[256] = {0};
- TMMBRSet* boundingSet = BoundingSet();
- boundingSet->VerifyAndAllocateSet(3);
-
- if (_cnt == 0)
- {
- // TMMBN {}
- // TMMBN {}
- // TMMBN {}
- // TMMBN {2,4,0} -> {4,2}
- assert(2 == ParseRTCPPacket(data, len, boundingSet));
- Verify(boundingSet, 0, set4);
- Verify(boundingSet, 1, set2);
-
- strcpy(fileName, "RTCPPacketTMMBR3.bin");
- }
-
- ++_cnt;
-
- // Get stored rtcp packet w/ TMMBR
- len = GetFile(fileName);
- if (len == 0)
- {
- return 1;
- }
-
- // Send in bitrate request
- if(_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)dataFile, len) == 0)
- {
- return len;
- }
- return -1;
- }
- RtpRtcp* _rtpRtcpModule;
- WebRtc_UWord32 _cnt;
-};
-
-
-class LoopBackTransportVideo : public webrtc::Transport, private TMMBRHelp
-{
-public:
- LoopBackTransportVideo(RtpRtcp* rtpRtcpModule) :
- TMMBRHelp(false),
- _rtpRtcpModule(rtpRtcpModule),
- _cnt(0)
- {
- }
- virtual int SendPacket(int channel, const void *data, int len)
- {
- if(_rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)data, len)== 0)
- {
- return len;
- }
- return -1;
- }
- virtual int SendRTCPPacket(int channel, const void *data, int len)
- {
- char fileName[256] = {0};
- TMMBRSet* boundingSet = BoundingSet();
- boundingSet->VerifyAndAllocateSet(3);
-
- if (_cnt == 0)
- {
- strcpy(fileName, "RTCPPacketTMMBR0.bin");
- }
- else if (_cnt == 1)
- {
- // TMMBN {0} -> {0}
- assert(1 == ParseRTCPPacket(data, len, boundingSet));
- Verify(boundingSet, 0, set0);
-
- strcpy(fileName, "RTCPPacketTMMBR1.bin");
- }
- else if (_cnt == 2)
- {
- // TMMBN {0,1} -> {1}
- assert(1 == ParseRTCPPacket(data, len, boundingSet));
- Verify(boundingSet, 0, set1);
-
- strcpy(fileName, "RTCPPacketTMMBR2.bin");
- }
- else if (_cnt == 3)
- {
- // TMMBN {0,1,2} -> {2}
- assert(1 == ParseRTCPPacket(data, len, boundingSet));
- Verify(boundingSet, 0, set2);
-
- strcpy(fileName, "RTCPPacketTMMBR3.bin");
- }
- else if (_cnt == 4)
- {
- // TMMBN {0,1,2,3} -> {3,2}
- assert(2 == ParseRTCPPacket(data, len, boundingSet));
- Verify(boundingSet, 0, set3);
- Verify(boundingSet, 1, set2);
-
- strcpy(fileName, "RTCPPacketTMMBR4.bin");
- }
- else if (_cnt == 5)
- {
- // TMMBN {0,1,2,3,4} -> {3,4,2}
- assert(3 == ParseRTCPPacket(data, len, boundingSet));
- Verify(boundingSet, 0, set3);
- Verify(boundingSet, 1, set4);
- Verify(boundingSet, 2, set2);
-
- strcpy(fileName, "RTCPPacketTMMBR5.bin");
- }
- else if (_cnt == 6)
- {
- // TMMBN {0,1,2,3,4,5} -> {3,4,2}
- assert(3 == ParseRTCPPacket(data, len, boundingSet));
- Verify(boundingSet, 0, set3);
- Verify(boundingSet, 1, set4);
- Verify(boundingSet, 2, set2);
-
- strcpy(fileName, "RTCPPacketTMMBR4_2.bin");
- }
- else if (_cnt == 7)
- {
- // TMMBN {0,1,2,3,4_2,5} -> {4_2}
- assert(1 == ParseRTCPPacket(data, len, boundingSet));
- Verify(boundingSet, 0, set4_2);
-
- ++_cnt;
- ::Sleep(5*RTCP_INTERVAL_AUDIO_MS + 1000); // time out receiver
- _rtpRtcpModule->Process(); // SendRTCP() (_cnt == 8)
- // a receiver has timed out -> UpdateTMMBR()
- }
- else if (_cnt == 8)
- {
- // No TMMBN in this packet
- assert(-1 == ParseRTCPPacket(data, len, boundingSet));
- }
- else if (_cnt == 10)
- {
- // TMMBN {} -> {}, empty set
- assert(0 == ParseRTCPPacket(data, len, boundingSet));
-
- strcpy(fileName, "RTCPPacketTMMBR2.bin");
- }
- else if (_cnt == 11)
- {
- // TMMBN {2} -> {2}
- assert(1 == ParseRTCPPacket(data, len, boundingSet));
- Verify(boundingSet, 0, set2);
- }
- else if (_cnt == 12) // ----- multi module -------------
- {
- // No TMMBN in this packet
- assert(-1 == ParseRTCPPacket(data, len, boundingSet));
-
- strcpy(fileName, "RTCPPacketTMMBR4.bin");
- }
- else if (_cnt == 13)
- {
- // TMMBN {}
- // TMMBN {}
- // TMMBN {}
- // TMMBN {2,4} -> {4,2}
- assert(2 == ParseRTCPPacket(data, len, boundingSet));
- Verify(boundingSet, 0, set4);
- Verify(boundingSet, 1, set2);
-
- strcpy(fileName, "RTCPPacketTMMBR0.bin");
- }
- else if (_cnt == 14)
- {
- // TMMBN {}
- // TMMBN {3}
- // TMMBN {}
- // TMMBN {2,4,0} -> {3,4,2}
- assert(3 == ParseRTCPPacket(data, len, boundingSet));
- Verify(boundingSet, 0, set3);
- Verify(boundingSet, 1, set4);
- Verify(boundingSet, 2, set2);
-
- strcpy(fileName, "RTCPPacketTMMBR1.bin");
- }
- //else if (_cnt == 15)
- //{
- // // TMMBN {}
- // // TMMBN {}
- // // TMMBN {}
- // // TMMBN {2,4,0,1} -> {4,2}
- // //assert(2 == ParseRTCPPacket(data, len, boundingSet));
- // //Verify(boundingSet, 0, set4);
- // //Verify(boundingSet, 1, set2);
- //}
- //else if (_cnt == 15)
- //{
- // // No TMMBN in this packet
- // assert(-1 == ParseRTCPPacket(data, len, boundingSet));
- //}
- else if (_cnt == 15)
- {
- // TMMBN {}
- // TMMBN {}
- // TMMBN {}
- // TMMBN {} -> {}, empty set
- assert(0 == ParseRTCPPacket(data, len, boundingSet));
- }
-
- ++_cnt;
-
- // Get stored rtcp packet w/ TMMBR
- len = GetFile(fileName);
- if (len == 0)
- {
- return 1;
- }
-
- // Send in bitrate request
- if( 0 == _rtpRtcpModule->IncomingPacket((const WebRtc_UWord8*)dataFile, len))
- {
- return len;
- }
- return -1;
- }
-
- RtpRtcp* _rtpRtcpModule;
- WebRtc_UWord32 _cnt;
-};
-
-class TestTMMBR : private TMMBRHelp
-{
-public:
- TestTMMBR() : TMMBRHelp(false) {};
-
- void Add(TMMBRSet* candidateSet, int index, InputSet set)
- {
- candidateSet->ptrTmmbrSet[index] = set.TMMBR;
- candidateSet->ptrPacketOHSet[index] = set.packetOH;
- candidateSet->ptrSsrcSet[index] = set.SSRC;
- };
-
- void Start()
- {
- // Get sets
- TMMBRSet* candidateSet = CandidateSet();
- assert(0 == candidateSet->sizeOfSet);
- TMMBRSet* boundingSet = BoundingSet();
- assert(0 == boundingSet->sizeOfSet);
- TMMBRSet* boundingSetToSend = BoundingSetToSend();
- assert(0 == boundingSetToSend->sizeOfSet);
-
- WebRtc_Word32 numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- assert(0 == numBoundingSet); // should be empty
-
- assert( 0 == SetTMMBRBoundingSetToSend(NULL,0)); // ok to send empty set
- assert( 0 == SetTMMBRBoundingSetToSend(boundingSet,0)); // ok to send empty set
-
- WebRtc_UWord32 minBitrateKbit = 0;
- WebRtc_UWord32 maxBitrateKbit = 0;
- assert(-1 == CalcMinMaxBitRate(0, 0, 1, false, minBitrateKbit, maxBitrateKbit)); // no bounding set
-
- // ---------------------------------
- // Test candidate set {0} -> {0}
- // ---------------------------------
- candidateSet = VerifyAndAllocateCandidateSet(1);
- assert(1 == candidateSet->sizeOfSet);
- Add(candidateSet, 0, set0);
-
- // Find bounding set
- numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- assert(1 == numBoundingSet);
- Verify(boundingSet, 0, set0);
-
- // Is owner of set
- assert(!IsOwner(set0.SSRC, 0)); // incorrect length
- assert(!IsOwner(set1.SSRC, 100)); // incorrect length
-
- assert( IsOwner(set0.SSRC, numBoundingSet));
- assert(!IsOwner(set1.SSRC, numBoundingSet));
- assert(!IsOwner(set2.SSRC, numBoundingSet));
-
- // Set boundingSet to send
- assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
-
- // Get boundingSet to send
- boundingSetToSend = BoundingSetToSend();
- assert(boundingSetToSend->sizeOfSet == numBoundingSet);
- Verify(boundingSetToSend, 0, set0);
-
- // Get net bitrate depending on packet rate
- assert( 0 == CalcMinMaxBitRate(0, numBoundingSet, false,0, minBitrateKbit, maxBitrateKbit));
- assert(set0.TMMBR == minBitrateKbit);
- assert(set0.TMMBR == maxBitrateKbit);
- assert(0 == CalcMinMaxBitRate(0, 100, false,0, minBitrateKbit, maxBitrateKbit)); // incorrect length
- assert(set0.TMMBR == minBitrateKbit);
- assert(set0.TMMBR == maxBitrateKbit);
-
- // ---------------------------------
- // Test candidate set {0,1} -> {1}
- // ---------------------------------
- candidateSet = VerifyAndAllocateCandidateSet(2);
- assert(2 == candidateSet->sizeOfSet);
- Add(candidateSet, 0, set0);
- Add(candidateSet, 1, set1);
-
- // Find bounding set
- numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- assert(1 == numBoundingSet);
- Verify(boundingSet, 0, set1);
-
- // Is owner of set
- assert(!IsOwner(set0.SSRC, numBoundingSet));
- assert( IsOwner(set1.SSRC, numBoundingSet));
- assert(!IsOwner(set2.SSRC, numBoundingSet));
-
- // Set boundingSet to send
- assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
-
- // Get boundingSet to send
- boundingSetToSend = BoundingSetToSend();
- assert(boundingSetToSend->sizeOfSet == numBoundingSet);
- Verify(boundingSetToSend, 0, set1);
-
- // Get net bitrate depending on packet rate
- assert(0 == CalcMinMaxBitRate(0, numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
- assert(set1.TMMBR == minBitrateKbit);
- assert(set0.TMMBR == maxBitrateKbit);
-
- // ---------------------------------
- // Test candidate set {0,1,2} -> {2}
- // ---------------------------------
- candidateSet = VerifyAndAllocateCandidateSet(3);
- assert(3 == candidateSet->sizeOfSet);
- Add(candidateSet, 0, set0);
- Add(candidateSet, 1, set1);
- Add(candidateSet, 2, set2);
-
- // Find bounding set
- numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- assert(1 == numBoundingSet);
- Verify(boundingSet, 0, set2);
-
- // Is owner of set
- assert(!IsOwner(set0.SSRC, numBoundingSet));
- assert(!IsOwner(set1.SSRC, numBoundingSet));
- assert( IsOwner(set2.SSRC, numBoundingSet));
-
- // Set boundingSet to send
- assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
-
- // Get boundingSet to send
- boundingSetToSend = BoundingSetToSend();
- assert(boundingSetToSend->sizeOfSet == numBoundingSet);
- Verify(boundingSetToSend, 0, set2);
-
- // Get net bitrate depending on packet rate
- assert(0 == CalcMinMaxBitRate(0, numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
- assert(set2.TMMBR == minBitrateKbit);
- assert(set0.TMMBR == maxBitrateKbit);
-
- // ---------------------------------
- // Test candidate set {0,1,2,3} -> {3,2}
- // ---------------------------------
- candidateSet = VerifyAndAllocateCandidateSet(4);
- assert(4 == candidateSet->sizeOfSet);
- Add(candidateSet, 0, set0);
- Add(candidateSet, 1, set1);
- Add(candidateSet, 2, set2);
- Add(candidateSet, 3, set3);
-
- // Find bounding set
- numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- assert(2 == numBoundingSet);
- Verify(boundingSet, 0, set3);
- Verify(boundingSet, 1, set2);
-
- // Is owner of set
- assert(!IsOwner(set0.SSRC, numBoundingSet));
- assert(!IsOwner(set1.SSRC, numBoundingSet));
- assert( IsOwner(set2.SSRC, numBoundingSet));
- assert( IsOwner(set3.SSRC, numBoundingSet));
-
- // Set boundingSet to send
- assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
-
- // Get boundingSet to send
- boundingSetToSend = BoundingSetToSend();
- assert(boundingSetToSend->sizeOfSet == numBoundingSet);
- Verify(boundingSetToSend, 0, set3);
- Verify(boundingSetToSend, 1, set2);
-
- // Get net bitrate depending on packet rate
- assert(0 == CalcMinMaxBitRate(0, numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
- assert(set3.TMMBR == minBitrateKbit);
- assert(set0.TMMBR == maxBitrateKbit);
-
- // ---------------------------------
- // Test candidate set {0,1,2,3,4} -> {3,4,2}
- // ---------------------------------
- candidateSet = VerifyAndAllocateCandidateSet(5);
- assert(5 == candidateSet->sizeOfSet);
- Add(candidateSet, 0, set0);
- Add(candidateSet, 1, set1);
- Add(candidateSet, 2, set2);
- Add(candidateSet, 3, set3);
- Add(candidateSet, 4, set4);
-
- // Find bounding set
- numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- assert(3 == numBoundingSet);
- Verify(boundingSet, 0, set3);
- Verify(boundingSet, 1, set4);
- Verify(boundingSet, 2, set2);
-
- // Is owner of set
- assert(!IsOwner(set0.SSRC, numBoundingSet));
- assert(!IsOwner(set1.SSRC, numBoundingSet));
- assert( IsOwner(set2.SSRC, numBoundingSet));
- assert( IsOwner(set3.SSRC, numBoundingSet));
- assert( IsOwner(set4.SSRC, numBoundingSet));
-
- // Set boundingSet to send
- assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
-
- // Get boundingSet to send
- boundingSetToSend = BoundingSetToSend();
- assert(boundingSetToSend->sizeOfSet == numBoundingSet);
- Verify(boundingSetToSend, 0, set3);
- Verify(boundingSetToSend, 1, set4);
- Verify(boundingSetToSend, 2, set2);
-
- // Get net bitrate depending on packet rate
- assert(0 == CalcMinMaxBitRate(0,numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
- assert(set3.TMMBR == minBitrateKbit);
- assert(set0.TMMBR == maxBitrateKbit);
-
- // ---------------------------------
- // Test candidate set {0,1,2,3,4,5} -> {3,4,2}
- // ---------------------------------
- candidateSet = VerifyAndAllocateCandidateSet(6);
- assert(6 == candidateSet->sizeOfSet);
- Add(candidateSet, 0, set0);
- Add(candidateSet, 1, set1);
- Add(candidateSet, 2, set2);
- Add(candidateSet, 3, set3);
- Add(candidateSet, 4, set4);
- Add(candidateSet, 5, set5);
-
- // Find bounding set
- numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- assert(3 == numBoundingSet);
- Verify(boundingSet, 0, set3);
- Verify(boundingSet, 1, set4);
- Verify(boundingSet, 2, set2);
-
- // Is owner of set
- assert(!IsOwner(set0.SSRC, numBoundingSet));
- assert(!IsOwner(set1.SSRC, numBoundingSet));
- assert( IsOwner(set2.SSRC, numBoundingSet));
- assert( IsOwner(set3.SSRC, numBoundingSet));
- assert( IsOwner(set4.SSRC, numBoundingSet));
- assert(!IsOwner(set5.SSRC, numBoundingSet));
-
- // Set boundingSet to send
- assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
-
- // Get boundingSet to send
- boundingSetToSend = BoundingSetToSend();
- assert(boundingSetToSend->sizeOfSet == numBoundingSet);
- Verify(boundingSetToSend, 0, set3);
- Verify(boundingSetToSend, 1, set4);
- Verify(boundingSetToSend, 2, set2);
-
- // Get net bitrate depending on packet rate
- assert(0 == CalcMinMaxBitRate(0,numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
- assert(set3.TMMBR == minBitrateKbit);
- assert(set0.TMMBR == maxBitrateKbit);
-
-
- // ---------------------------------
- // Test candidate set {1,2,3,4,5} -> {3,4,2}
- // ---------------------------------
- candidateSet = VerifyAndAllocateCandidateSet(5);
- assert(6 == candidateSet->sizeOfSet);
- Add(candidateSet, 0, set1);
- Add(candidateSet, 1, set2);
- Add(candidateSet, 2, set3);
- Add(candidateSet, 3, set4);
- Add(candidateSet, 4, set5);
-
- // Find bounding set
- numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- assert(3 == numBoundingSet);
- Verify(boundingSet, 0, set3);
- Verify(boundingSet, 1, set4);
- Verify(boundingSet, 2, set2);
-
- // Is owner of set
- assert(!IsOwner(set0.SSRC, numBoundingSet));
- assert(!IsOwner(set1.SSRC, numBoundingSet));
- assert( IsOwner(set2.SSRC, numBoundingSet));
- assert( IsOwner(set3.SSRC, numBoundingSet));
- assert( IsOwner(set4.SSRC, numBoundingSet));
- assert(!IsOwner(set5.SSRC, numBoundingSet));
-
- // Set boundingSet to send
- assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
-
- // Get boundingSet to send
- boundingSetToSend = BoundingSetToSend();
- assert(boundingSetToSend->sizeOfSet == numBoundingSet);
- Verify(boundingSetToSend, 0, set3);
- Verify(boundingSetToSend, 1, set4);
- Verify(boundingSetToSend, 2, set2);
-
- // Get net bitrate depending on packet rate
- assert(0 == CalcMinMaxBitRate(0,numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
- assert(set3.TMMBR == minBitrateKbit);
- assert(set5.TMMBR == maxBitrateKbit);
-
-
- // ---------------------------------
- // Test candidate set {1,3,4,5} -> {3,4}
- // ---------------------------------
- candidateSet = VerifyAndAllocateCandidateSet(4);
- assert(6 == candidateSet->sizeOfSet);
- Add(candidateSet, 0, set1);
- Add(candidateSet, 1, set3);
- Add(candidateSet, 2, set4);
- Add(candidateSet, 3, set5);
-
- // Find bounding set
- numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- assert(2 == numBoundingSet);
- Verify(boundingSet, 0, set3);
- Verify(boundingSet, 1, set4);
-
- // Is owner of set
- assert(!IsOwner(set0.SSRC, numBoundingSet));
- assert(!IsOwner(set1.SSRC, numBoundingSet));
- assert(!IsOwner(set2.SSRC, numBoundingSet));
- assert( IsOwner(set3.SSRC, numBoundingSet));
- assert( IsOwner(set4.SSRC, numBoundingSet));
- assert(!IsOwner(set5.SSRC, numBoundingSet));
-
- // Set boundingSet to send
- assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
-
- // Get boundingSet to send
- boundingSetToSend = BoundingSetToSend();
- Verify(boundingSetToSend, 0, set3);
- Verify(boundingSetToSend, 1, set4);
-
- // Get net bitrate depending on packet rate
- assert(0 == CalcMinMaxBitRate(0, numBoundingSet,true,0, minBitrateKbit, maxBitrateKbit));
- assert(set3.TMMBR == minBitrateKbit);
- assert(set5.TMMBR == maxBitrateKbit);
-
- // ---------------------------------
- // Test candidate set {1,2,4,5} -> {4,2}
- // ---------------------------------
- candidateSet = VerifyAndAllocateCandidateSet(4);
- assert(6 == candidateSet->sizeOfSet);
- Add(candidateSet, 0, set1);
- Add(candidateSet, 1, set2);
- Add(candidateSet, 2, set4);
- Add(candidateSet, 3, set5);
-
- // Find bounding set
- numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- assert(2 == numBoundingSet);
- Verify(boundingSet, 0, set4);
- Verify(boundingSet, 1, set2);
-
- // Is owner of set
- assert(!IsOwner(set0.SSRC, numBoundingSet));
- assert(!IsOwner(set1.SSRC, numBoundingSet));
- assert( IsOwner(set2.SSRC, numBoundingSet));
- assert(!IsOwner(set3.SSRC, numBoundingSet));
- assert( IsOwner(set4.SSRC, numBoundingSet));
- assert(!IsOwner(set5.SSRC, numBoundingSet));
-
- // Set boundingSet to send
- assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
-
- // Get boundingSet to send
- boundingSetToSend = BoundingSetToSend();
- Verify(boundingSetToSend, 0, set4);
- Verify(boundingSetToSend, 1, set2);
-
- // Get net bitrate depending on packet rate
- assert(0 == CalcMinMaxBitRate(0, numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
- assert(set4.TMMBR == minBitrateKbit);
- assert(set5.TMMBR == maxBitrateKbit);
-
- // ---------------------------------
- // Test candidate set {1,2,3,5} -> {3,2}
- // ---------------------------------
- candidateSet = VerifyAndAllocateCandidateSet(4);
- assert(6 == candidateSet->sizeOfSet);
- Add(candidateSet, 0, set1);
- Add(candidateSet, 1, set2);
- Add(candidateSet, 2, set3);
- Add(candidateSet, 3, set5);
-
- // Find bounding set
- numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- assert(2 == numBoundingSet);
- Verify(boundingSet, 0, set3);
- Verify(boundingSet, 1, set2);
-
- // Is owner of set
- assert(!IsOwner(set0.SSRC, numBoundingSet));
- assert(!IsOwner(set1.SSRC, numBoundingSet));
- assert( IsOwner(set2.SSRC, numBoundingSet));
- assert( IsOwner(set3.SSRC, numBoundingSet));
- assert(!IsOwner(set4.SSRC, numBoundingSet));
- assert(!IsOwner(set5.SSRC, numBoundingSet));
-
- // Set boundingSet to send
- assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
-
- // Get boundingSet to send
- boundingSetToSend = BoundingSetToSend();
- Verify(boundingSetToSend, 0, set3);
- Verify(boundingSetToSend, 1, set2);
-
- // Get net bitrate depending on packet rate
- assert(0 == CalcMinMaxBitRate(0, numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
- assert(set3.TMMBR == minBitrateKbit);
- assert(set5.TMMBR == maxBitrateKbit);
-
- // ---------------------------------
- // Test candidate set {1,2,3,4_1,5} -> {3,2}
- // ---------------------------------
- candidateSet = VerifyAndAllocateCandidateSet(5);
- assert(6 == candidateSet->sizeOfSet);
- Add(candidateSet, 0, set1);
- Add(candidateSet, 1, set2);
- Add(candidateSet, 2, set3);
- Add(candidateSet, 3, set4_1);
- Add(candidateSet, 4, set5);
-
- // Find bounding set
- numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- assert(2 == numBoundingSet);
- Verify(boundingSet, 0, set3);
- Verify(boundingSet, 1, set2);
-
- // Is owner of set
- assert(!IsOwner(set0.SSRC, numBoundingSet));
- assert(!IsOwner(set1.SSRC, numBoundingSet));
- assert( IsOwner(set2.SSRC, numBoundingSet));
- assert( IsOwner(set3.SSRC, numBoundingSet));
- assert(!IsOwner(set4.SSRC, numBoundingSet));
- assert(!IsOwner(set5.SSRC, numBoundingSet));
-
- // Set boundingSet to send
- assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
-
- // Get boundingSet to send
- boundingSetToSend = BoundingSetToSend();
- Verify(boundingSetToSend, 0, set3);
- Verify(boundingSetToSend, 1, set2);
-
- // Get net bitrate depending on packet rate
- assert(0 == CalcMinMaxBitRate(0, numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
- assert(set3.TMMBR == minBitrateKbit);
- assert(set5.TMMBR == maxBitrateKbit);
-
- // ---------------------------------
- // Test candidate set {1,2,3,4_2,5} -> {4_2}
- // ---------------------------------
- candidateSet = VerifyAndAllocateCandidateSet(5);
- assert(6 == candidateSet->sizeOfSet);
- Add(candidateSet, 0, set1);
- Add(candidateSet, 1, set2);
- Add(candidateSet, 2, set3);
- Add(candidateSet, 3, set4_2);
- Add(candidateSet, 4, set5);
-
- // Find bounding set
- numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- assert(1 == numBoundingSet);
- Verify(boundingSet, 0, set4_2);
-
- // Is owner of set
- assert(!IsOwner(set0.SSRC, numBoundingSet));
- assert(!IsOwner(set1.SSRC, numBoundingSet));
- assert(!IsOwner(set2.SSRC, numBoundingSet));
- assert(!IsOwner(set3.SSRC, numBoundingSet));
- assert( IsOwner(set4.SSRC, numBoundingSet));
- assert(!IsOwner(set5.SSRC, numBoundingSet));
-
- // Set boundingSet to send
- assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
-
- // Get boundingSet to send
- boundingSetToSend = BoundingSetToSend();
- Verify(boundingSetToSend, 0, set4_2);
-
- // Get net bitrate depending on packet rate
- assert(0 == CalcMinMaxBitRate(0, numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
- assert(MIN_VIDEO_BW_MANAGEMENT_BITRATE == minBitrateKbit);
- assert(set5.TMMBR == maxBitrateKbit);
-
- // ---------------------------------
- // Test candidate set {} -> {}
- // ---------------------------------
- candidateSet = VerifyAndAllocateCandidateSet(0);
- assert(6 == candidateSet->sizeOfSet);
-
- // Find bounding set
- numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- assert(0 == numBoundingSet);
-
- // Is owner of set
- assert(!IsOwner(set0.SSRC, numBoundingSet));
- assert(!IsOwner(set1.SSRC, numBoundingSet));
- assert(!IsOwner(set2.SSRC, numBoundingSet));
- assert(!IsOwner(set3.SSRC, numBoundingSet));
- assert(!IsOwner(set4.SSRC, numBoundingSet));
- assert(!IsOwner(set5.SSRC, numBoundingSet));
-
- // Set boundingSet to send
- assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
-
- // Get boundingSet to send
- boundingSetToSend = BoundingSetToSend();
-
- // Get net bitrate depending on packet rate
- assert(-1 == CalcMinMaxBitRate(0,numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
-
- // ---------------------------------
- // Test candidate set {x0,5} -> {5}
- // ---------------------------------
- candidateSet = VerifyAndAllocateCandidateSet(2);
- assert(6 == candidateSet->sizeOfSet);
- Add(candidateSet, 0, set00);
- Add(candidateSet, 1, set5);
-
- // Find bounding set
- numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- assert(1 == numBoundingSet);
- Verify(boundingSet, 0, set5);
-
- // Is owner of set
- assert(!IsOwner(set0.SSRC, numBoundingSet));
- assert(!IsOwner(set1.SSRC, numBoundingSet));
- assert(!IsOwner(set2.SSRC, numBoundingSet));
- assert(!IsOwner(set3.SSRC, numBoundingSet));
- assert(!IsOwner(set4.SSRC, numBoundingSet));
- assert( IsOwner(set5.SSRC, numBoundingSet));
-
- // Set boundingSet to send
- assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
-
- // Get boundingSet to send
- boundingSetToSend = BoundingSetToSend();
- Verify(boundingSetToSend, 0, set5);
-
- // Get net bitrate depending on packet rate
- assert(0 == CalcMinMaxBitRate(0,numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
- assert(set5.TMMBR == minBitrateKbit);
- assert(set5.TMMBR == maxBitrateKbit);
-
- // ---------------------------------
- // Test candidate set {x0,4,2} -> {4,2}
- // ---------------------------------
- candidateSet = VerifyAndAllocateCandidateSet(3);
- assert(6 == candidateSet->sizeOfSet);
- Add(candidateSet, 0, set00);
- Add(candidateSet, 1, set4);
- Add(candidateSet, 2, set2);
-
- // Find bounding set
- numBoundingSet = FindTMMBRBoundingSet(boundingSet);
- assert(2 == numBoundingSet);
- Verify(boundingSet, 0, set4);
- Verify(boundingSet, 1, set2);
-
- // Is owner of set
- assert(!IsOwner(set0.SSRC, numBoundingSet));
- assert(!IsOwner(set1.SSRC, numBoundingSet));
- assert( IsOwner(set2.SSRC, numBoundingSet));
- assert(!IsOwner(set3.SSRC, numBoundingSet));
- assert( IsOwner(set4.SSRC, numBoundingSet));
- assert(!IsOwner(set5.SSRC, numBoundingSet));
-
- // Set boundingSet to send
- assert(0 == SetTMMBRBoundingSetToSend(boundingSet, maxBitrate));
-
- // Get boundingSet to send
- boundingSetToSend = BoundingSetToSend();
- Verify(boundingSetToSend, 0, set4);
- Verify(boundingSetToSend, 1, set2);
-
- // Get net bitrate depending on packet rate
- assert(0 == CalcMinMaxBitRate(0,numBoundingSet, true,0, minBitrateKbit, maxBitrateKbit));
- assert(set4.TMMBR == minBitrateKbit);
- assert(set2.TMMBR == maxBitrateKbit);
- };
-};
-
-class NULLDataZink: public RtpData
-{
- virtual WebRtc_Word32 OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const webrtc::WebRtcRTPHeader* rtpHeader,
- const WebRtc_UWord8* incomingRtpPacket,
- const WebRtc_UWord16 incomingRtpPacketLengt)
- {
- return 0;
- };
-};
-
-
-int _tmain(int argc, _TCHAR* argv[])
-{
-
- std::string str;
- std::cout << "------------------------" << std::endl;
- std::cout << "------ Test TMMBR ------" << std::endl;
- std::cout << "------------------------" << std::endl;
- std::cout << " " << std::endl;
-
- // --------------------
- // Test TMMBRHelp class
- // --------------------
- TestTMMBR test;
- test.Start();
-
- printf("TMMBRHelp-class test done.\n");
-
- // ------------------------
- // Test TMMBR single module
- // ------------------------
- RtpRtcp* rtpRtcpModuleVideo = RtpRtcp::CreateRtpRtcp(0, false);
-
- LoopBackTransportVideo* myLoopBackTransportVideo = new LoopBackTransportVideo(rtpRtcpModuleVideo);
- assert(0 == rtpRtcpModuleVideo->RegisterSendTransport(myLoopBackTransportVideo));
-
- assert(false == rtpRtcpModuleVideo->TMMBR());
- rtpRtcpModuleVideo->SetTMMBRStatus(true);
- assert(true == rtpRtcpModuleVideo->TMMBR());
-
- assert(0 == rtpRtcpModuleVideo->RegisterSendPayload( "I420", 96));
- assert(0 == rtpRtcpModuleVideo->RegisterReceivePayload( "I420", 96));
-
- // send a RTP packet with SSRC 11111 to get 11111 as the received SSRC
- assert(0 == rtpRtcpModuleVideo->SetSSRC(11111));
- const WebRtc_UWord8 testStream[9] = "testtest";
- assert(0 == rtpRtcpModuleVideo->RegisterIncomingDataCallback(new NULLDataZink())); // needed to avoid error from parsing the incoming stream
- assert(0 == rtpRtcpModuleVideo->SendOutgoingData(webrtc::kVideoFrameKey,96, 0, testStream, 8));
-
- // set the SSRC to 0
- assert(0 == rtpRtcpModuleVideo->SetSSRC(0));
-
- //
- assert(0 == rtpRtcpModuleVideo->SetRTCPStatus(kRtcpCompound));
-
- assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {0} // should this make us remember a TMMBR?
- assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {1}, verify TMMBN {0}
- assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {2}, verify TMMBN {1}
- assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {3}, verify TMMBN {2}
- assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {4}, verify TMMBN {3,2}
- assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {5}, verify TMMBN {3,4,2}
- assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {4_2}, verify TMMBN {3,4,2}
- assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> time out receivers, verify TMMBN {4_2}
- assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {2}
- assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> verify TMMBN {2}
-
- printf("Single module test done.\n");
-
- // ------------------------
- // Test TMMBR multi module
- // ------------------------
- RtpRtcp* rtpRtcpModuleVideoDef = RtpRtcp::CreateRtpRtcp(10, false);
- assert(0 == rtpRtcpModuleVideo->RegisterDefaultModule(rtpRtcpModuleVideoDef));
-
- RtpRtcp* rtpRtcpModuleVideo1 = RtpRtcp::CreateRtpRtcp(1, false);
- assert(0 == rtpRtcpModuleVideo1->RegisterDefaultModule(rtpRtcpModuleVideoDef));
-
- RtpRtcp* rtpRtcpModuleVideo2 = RtpRtcp::CreateRtpRtcp(2, false);
- assert(0 == rtpRtcpModuleVideo2->RegisterDefaultModule(rtpRtcpModuleVideoDef));
-
- RtpRtcp* rtpRtcpModuleVideo3 = RtpRtcp::CreateRtpRtcp(3, false);
- assert(0 == rtpRtcpModuleVideo3->RegisterDefaultModule(rtpRtcpModuleVideoDef));
-
- LoopBackTransport2* myLoopBackTransport2 = new LoopBackTransport2(rtpRtcpModuleVideo2);
- assert(0 == rtpRtcpModuleVideo2->RegisterSendTransport(myLoopBackTransport2));
-
- assert(0 == rtpRtcpModuleVideo2->SetRTCPStatus(kRtcpCompound));
-
- // set the SSRC to 0
- assert(0 == rtpRtcpModuleVideo2->SetSSRC(0));
-
- assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {4}, verify no TMMBN in this packet
- assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {0}, verify TMMBN {4,2}
- assert(0 == rtpRtcpModuleVideo2->SendRTCP()); // -> incoming TMMBR {3}, verify TMMBN {4,2}
- assert(0 == rtpRtcpModuleVideo->SendRTCP()); // -> incoming TMMBR {1}, verify TMMBN {3,4,2}
- ::Sleep(5*RTCP_INTERVAL_AUDIO_MS + 1000);
- rtpRtcpModuleVideo2->Process(); // time out receiver2 -> UpdateTMMBR()
- assert(0 == rtpRtcpModuleVideo->SendRTCP()); // verify TMMBN {}
-
- printf("Multi module test done.\n");
-
-
- RtpRtcp::DestroyRtpRtcp(rtpRtcpModuleVideo);
- RtpRtcp::DestroyRtpRtcp(rtpRtcpModuleVideo1);
- RtpRtcp::DestroyRtpRtcp(rtpRtcpModuleVideo2);
- RtpRtcp::DestroyRtpRtcp(rtpRtcpModuleVideo3);
- RtpRtcp::DestroyRtpRtcp(rtpRtcpModuleVideoDef);
-
- TEST_PASSED();
- ::Sleep(5000);
-
- return 0;
-}
-
diff --git a/trunk/src/modules/rtp_rtcp/test/test_bwe/test_bwe.gypi b/trunk/src/modules/rtp_rtcp/test/test_bwe/test_bwe.gypi
deleted file mode 100644
index d165e7d..0000000
--- a/trunk/src/modules/rtp_rtcp/test/test_bwe/test_bwe.gypi
+++ /dev/null
@@ -1,35 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'test_bwe',
- 'type': 'executable',
- 'dependencies': [
- 'rtp_rtcp',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '../../source',
- ],
- 'sources': [
- 'unit_test.cc',
- '../../source/bitrate.cc',
- ],
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/rtp_rtcp/test/test_bwe/unit_test.cc b/trunk/src/modules/rtp_rtcp/test/test_bwe/unit_test.cc
deleted file mode 100644
index 3adb099..0000000
--- a/trunk/src/modules/rtp_rtcp/test/test_bwe/unit_test.cc
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-/*
- * This file includes unit tests for the bandwidth estimation and management
- */
-
-#include <gtest/gtest.h>
-
-#include "typedefs.h"
-#include "Bitrate.h"
-
-namespace {
-
-using webrtc::BitRateStats;
-
-class BitRateStatsTest : public ::testing::Test
-{
-protected:
- BitRateStatsTest() {};
- BitRateStats bitRate;
-};
-
-TEST_F(BitRateStatsTest, TestStrictMode)
-{
- WebRtc_Word64 nowMs = 0;
- // Should be initialized to 0.
- EXPECT_EQ(0u, bitRate.BitRate(nowMs));
- bitRate.Update(1500, nowMs);
- // Expecting 12 kbps given a 1000 window with one 1500 bytes packet.
- EXPECT_EQ(12000u, bitRate.BitRate(nowMs));
- bitRate.Init();
- // Expecting 0 after init.
- EXPECT_EQ(0u, bitRate.BitRate(nowMs));
- for (int i = 0; i < 100000; ++i)
- {
- if (nowMs % 10 == 0)
- bitRate.Update(1500, nowMs);
- // Approximately 1200 kbps expected. Not exact since when packets
- // are removed we will jump 10 ms to the next packet.
- if (nowMs > 0 && nowMs % 2000 == 0)
- EXPECT_NEAR(1200000u, bitRate.BitRate(nowMs), 6000u);
- nowMs += 1;
- }
- nowMs += 2000;
- // The window is 2 seconds. If nothing has been received for that time
- // the estimate should be 0.
- EXPECT_EQ(0u, bitRate.BitRate(nowMs));
-}
-
-}
diff --git a/trunk/src/modules/udp_transport/OWNERS b/trunk/src/modules/udp_transport/OWNERS
deleted file mode 100644
index 3b2a444..0000000
--- a/trunk/src/modules/udp_transport/OWNERS
+++ /dev/null
@@ -1,4 +0,0 @@
-pwestin@webrtc.org
-henrikg@webrtc.org
-mallinath@webrtc.org
-tomasl@webrtc.org
\ No newline at end of file
diff --git a/trunk/src/modules/udp_transport/interface/udp_transport.h b/trunk/src/modules/udp_transport/interface/udp_transport.h
deleted file mode 100644
index 6596fde..0000000
--- a/trunk/src/modules/udp_transport/interface/udp_transport.h
+++ /dev/null
@@ -1,386 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UDP_TRANSPORT_INTERFACE_UDP_TRANSPORT_H_
-#define WEBRTC_MODULES_UDP_TRANSPORT_INTERFACE_UDP_TRANSPORT_H_
-
-#include "common_types.h"
-#include "module.h"
-#include "typedefs.h"
-
-#define SS_MAXSIZE 128
-#define SS_ALIGNSIZE (sizeof (WebRtc_UWord64))
-#define SS_PAD1SIZE (SS_ALIGNSIZE - sizeof(WebRtc_Word16))
-#define SS_PAD2SIZE (SS_MAXSIZE - (sizeof(WebRtc_Word16) + SS_PAD1SIZE +\
- SS_ALIGNSIZE))
-
-// BSD requires use of HAVE_STRUCT_SOCKADDR_SA_LEN
-namespace webrtc {
-struct SocketAddressIn
-{
- // sin_family should be either AF_INET (IPv4) or AF_INET6 (IPv6)
-#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- WebRtc_Word8 sin_length;
- WebRtc_Word8 sin_family;
-#else
- WebRtc_Word16 sin_family;
-#endif
- WebRtc_UWord16 sin_port;
- WebRtc_UWord32 sin_addr;
- WebRtc_Word8 sin_zero[8];
-};
-
-struct Version6InAddress
-{
- union
- {
- WebRtc_UWord8 _s6_u8[16];
- WebRtc_UWord32 _s6_u32[4];
- WebRtc_UWord64 _s6_u64[2];
- } Version6AddressUnion;
-};
-
-struct SocketAddressInVersion6
-{
- // sin_family should be either AF_INET (IPv4) or AF_INET6 (IPv6)
-#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- WebRtc_Word8 sin_length;
- WebRtc_Word8 sin_family;
-#else
- WebRtc_Word16 sin_family;
-#endif
- // Transport layer port number.
- WebRtc_UWord16 sin6_port;
- // IPv6 traffic class and flow info or ip4 address.
- WebRtc_UWord32 sin6_flowinfo;
- // IPv6 address
- struct Version6InAddress sin6_addr;
- // Set of interfaces for a scope.
- WebRtc_UWord32 sin6_scope_id;
-};
-
-struct SocketAddressStorage
-{
- // sin_family should be either AF_INET (IPv4) or AF_INET6 (IPv6)
-#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- WebRtc_Word8 sin_length;
- WebRtc_Word8 sin_family;
-#else
- WebRtc_Word16 sin_family;
-#endif
- WebRtc_Word8 __ss_pad1[SS_PAD1SIZE];
- WebRtc_UWord64 __ss_align;
- WebRtc_Word8 __ss_pad2[SS_PAD2SIZE];
-};
-
-struct SocketAddress
-{
- union
- {
- struct SocketAddressIn _sockaddr_in;
- struct SocketAddressInVersion6 _sockaddr_in6;
- struct SocketAddressStorage _sockaddr_storage;
- };
-};
-
-// Callback class that receives packets from UdpTransport.
-class UdpTransportData
-{
-public:
- virtual ~UdpTransportData() {};
-
- virtual void IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
- const WebRtc_Word32 rtpPacketLength,
- const char* fromIP,
- const WebRtc_UWord16 fromPort) = 0;
-
- virtual void IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
- const WebRtc_Word32 rtcpPacketLength,
- const char* fromIP,
- const WebRtc_UWord16 fromPort) = 0;
-};
-
-
-class UdpTransport : public Module, public Transport
-{
-public:
- enum
- {
- kIpAddressVersion6Length = 64,
- kIpAddressVersion4Length = 16
- };
- enum ErrorCode
- {
- kNoSocketError = 0,
- kFailedToBindPort = 1,
- kIpAddressInvalid = 2,
- kAddressInvalid = 3,
- kSocketInvalid = 4,
- kPortInvalid = 5,
- kTosInvalid = 6,
- kMulticastAddressInvalid = 7,
- kQosError = 8,
- kSocketAlreadyInitialized = 9,
- kIpVersion6Error = 10,
- FILTER_ERROR = 11,
- kStartReceiveError = 12,
- kStopReceiveError = 13,
- kCannotFindLocalIp = 14,
- kTosError = 16,
- kNotInitialized = 17,
- kPcpError = 18
- };
-
- // Factory method. Constructor disabled.
- static UdpTransport* Create(const WebRtc_Word32 id,
- WebRtc_UWord8& numSocketThreads);
- static void Destroy(UdpTransport* module);
-
- // Prepares the class for sending RTP packets to ipAddr:rtpPort and RTCP
- // packets to ipAddr:rtpPort+1 if rtcpPort is zero. Otherwise to
- // ipAddr:rtcpPort.
- virtual WebRtc_Word32 InitializeSendSockets(
- const char* ipAddr,
- const WebRtc_UWord16 rtpPort,
- const WebRtc_UWord16 rtcpPort = 0) = 0;
-
- // Register packetCallback for receiving incoming packets. Set the local
- // RTP port to rtpPort. Bind local IP address to ipAddr. If ipAddr is NULL
- // bind to local IP ANY. Set the local rtcp port to rtcpPort or rtpPort + 1
- // if rtcpPort is 0.
- virtual WebRtc_Word32 InitializeReceiveSockets(
- UdpTransportData* const packetCallback,
- const WebRtc_UWord16 rtpPort,
- const char* ipAddr = NULL,
- const char* multicastIpAddr = NULL,
- const WebRtc_UWord16 rtcpPort = 0) = 0;
-
- // Set local RTP port to rtpPort and RTCP port to rtcpPort or rtpPort + 1 if
- // rtcpPort is 0. These ports will be used for sending instead of the local
- // ports set by InitializeReceiveSockets(..).
- virtual WebRtc_Word32 InitializeSourcePorts(
- const WebRtc_UWord16 rtpPort,
- const WebRtc_UWord16 rtcpPort = 0) = 0;
-
- // Retrieve local ports used for sending if other than the ports specified
- // by InitializeReceiveSockets(..). rtpPort is set to the RTP port.
- // rtcpPort is set to the RTCP port.
- virtual WebRtc_Word32 SourcePorts(WebRtc_UWord16& rtpPort,
- WebRtc_UWord16& rtcpPort) const = 0;
-
- // Set ipAddr to the IP address that is currently being listened on. rtpPort
- // to the RTP port listened to. rtcpPort to the RTCP port listened on.
- // multicastIpAddr to the multicast IP address group joined (the address
- // is NULL terminated).
- virtual WebRtc_Word32 ReceiveSocketInformation(
- char ipAddr[kIpAddressVersion6Length],
- WebRtc_UWord16& rtpPort,
- WebRtc_UWord16& rtcpPort,
- char multicastIpAddr[kIpAddressVersion6Length]) const = 0;
-
- // Set ipAddr to the IP address being sent from. rtpPort to the local RTP
- // port used for sending and rtcpPort to the local RTCP port used for
- // sending.
- virtual WebRtc_Word32 SendSocketInformation(
- char ipAddr[kIpAddressVersion6Length],
- WebRtc_UWord16& rtpPort,
- WebRtc_UWord16& rtcpPort) const = 0;
-
- // Put the IP address, RTP port and RTCP port from the last received packet
- // into ipAddr, rtpPort and rtcpPort respectively.
- virtual WebRtc_Word32 RemoteSocketInformation(
- char ipAddr[kIpAddressVersion6Length],
- WebRtc_UWord16& rtpPort,
- WebRtc_UWord16& rtcpPort) const = 0;
-
- // Enable/disable quality of service if QoS is true or false respectively.
- // Set the type of service to serviceType, max bitrate in kbit/s to
- // maxBitrate and override DSCP if overrideDSCP is not 0.
- // Note: Must be called both InitializeSendSockets() and
- // InitializeReceiveSockets() has been called.
- virtual WebRtc_Word32 SetQoS(const bool QoS,
- const WebRtc_Word32 serviceType,
- const WebRtc_UWord32 maxBitrate = 0,
- const WebRtc_Word32 overrideDSCP = 0,
- const bool audio = false) = 0;
-
- // Set QoS to true if quality of service has been turned on. If QoS is true,
- // also set serviceType to type of service and overrideDSCP to override
- // DSCP.
- virtual WebRtc_Word32 QoS(bool& QoS,
- WebRtc_Word32& serviceType,
- WebRtc_Word32& overrideDSCP) const = 0;
-
- // Set type of service.
- virtual WebRtc_Word32 SetToS(const WebRtc_Word32 DSCP,
- const bool useSetSockOpt = false) = 0;
-
- // Get type of service configuration.
- virtual WebRtc_Word32 ToS(WebRtc_Word32& DSCP,
- bool& useSetSockOpt) const = 0;
-
- // Set Priority Code Point (IEEE 802.1Q)
- // Note: for Linux this function will set the priority for the socket,
- // which then can be mapped to a PCP value with vconfig.
- virtual WebRtc_Word32 SetPCP(const WebRtc_Word32 PCP) = 0;
-
- // Get Priority Code Point
- virtual WebRtc_Word32 PCP(WebRtc_Word32& PCP) const = 0;
-
- // Enable IPv6.
- // Note: this API must be called before any call to
- // InitializeReceiveSockets() or InitializeSendSockets(). It is not
- // possible to go back to IPv4 (default) after this call.
- virtual WebRtc_Word32 EnableIpV6() = 0;
-
- // Return true if IPv6 has been enabled.
- virtual bool IpV6Enabled() const = 0;
-
- // Only allow packets received from filterIPAddress to be processed.
- // Note: must be called after EnableIPv6(), if IPv6 is used.
- virtual WebRtc_Word32 SetFilterIP(
- const char filterIPAddress[kIpAddressVersion6Length]) = 0;
-
- // Write the filter IP address (if any) to filterIPAddress.
- virtual WebRtc_Word32 FilterIP(
- char filterIPAddress[kIpAddressVersion6Length]) const = 0;
-
- // Only allow RTP packets from rtpFilterPort and RTCP packets from
- // rtcpFilterPort be processed.
- // Note: must be called after EnableIPv6(), if IPv6 is used.
- virtual WebRtc_Word32 SetFilterPorts(
- const WebRtc_UWord16 rtpFilterPort,
- const WebRtc_UWord16 rtcpFilterPort) = 0;
-
- // Set rtpFilterPort to the filter RTP port and rtcpFilterPort to the
- // filter RTCP port (if filtering based on port is enabled).
- virtual WebRtc_Word32 FilterPorts(WebRtc_UWord16& rtpFilterPort,
- WebRtc_UWord16& rtcpFilterPort) const = 0;
-
- // Set the number of buffers that the socket implementation may use for
- // receiving packets to numberOfSocketBuffers. I.e. the number of packets
- // that can be received in parallell.
- // Note: this API only has effect on Windows.
- virtual WebRtc_Word32 StartReceiving(
- const WebRtc_UWord32 numberOfSocketBuffers) = 0;
-
- // Stop receive incoming packets.
- virtual WebRtc_Word32 StopReceiving() = 0;
-
- // Return true incoming packets are received.
- virtual bool Receiving() const = 0;
-
- // Return true if send sockets have been initialized.
- virtual bool SendSocketsInitialized() const = 0;
-
- // Return true if local ports for sending has been set.
- virtual bool SourcePortsInitialized() const = 0;
-
- // Return true if receive sockets have been initialized.
- virtual bool ReceiveSocketsInitialized() const = 0;
-
- // Send data with size length to ip:portnr. The same port as the set
- // with InitializeSendSockets(..) is used if portnr is 0. The same IP
- // address as set with InitializeSendSockets(..) is used if ip is NULL.
- // If isRTCP is true the port used will be the RTCP port.
- virtual WebRtc_Word32 SendRaw(const WebRtc_Word8* data,
- WebRtc_UWord32 length,
- WebRtc_Word32 isRTCP,
- WebRtc_UWord16 portnr = 0,
- const char* ip = NULL) = 0;
-
- // Send RTP data with size length to the address specified by to.
- virtual WebRtc_Word32 SendRTPPacketTo(const WebRtc_Word8* data,
- WebRtc_UWord32 length,
- const SocketAddress& to) = 0;
-
-
- // Send RTCP data with size length to the address specified by to.
- virtual WebRtc_Word32 SendRTCPPacketTo(const WebRtc_Word8* data,
- WebRtc_UWord32 length,
- const SocketAddress& to) = 0;
-
- // Send RTP data with size length to ip:rtpPort where ip is the ip set by
- // the InitializeSendSockets(..) call.
- virtual WebRtc_Word32 SendRTPPacketTo(const WebRtc_Word8* data,
- WebRtc_UWord32 length,
- WebRtc_UWord16 rtpPort) = 0;
-
-
- // Send RTCP data with size length to ip:rtcpPort where ip is the ip set by
- // the InitializeSendSockets(..) call.
- virtual WebRtc_Word32 SendRTCPPacketTo(const WebRtc_Word8* data,
- WebRtc_UWord32 length,
- WebRtc_UWord16 rtcpPort) = 0;
-
- // Set the IP address to which packets are sent to ipaddr.
- virtual WebRtc_Word32 SetSendIP(
- const char ipaddr[kIpAddressVersion6Length]) = 0;
-
- // Set the send RTP and RTCP port to rtpPort and rtcpPort respectively.
- virtual WebRtc_Word32 SetSendPorts(const WebRtc_UWord16 rtpPort,
- const WebRtc_UWord16 rtcpPort = 0) = 0;
-
- // Retreive the last registered error code.
- virtual ErrorCode LastError() const = 0;
-
- // Put the local IPv4 address in localIP.
- // Note: this API is for IPv4 only.
- static WebRtc_Word32 LocalHostAddress(WebRtc_UWord32& localIP);
-
- // Put the local IP6 address in localIP.
- // Note: this API is for IPv6 only.
- static WebRtc_Word32 LocalHostAddressIPV6(char localIP[16]);
-
- // Return a copy of hostOrder (host order) in network order.
- static WebRtc_UWord16 Htons(WebRtc_UWord16 hostOrder);
-
- // Return a copy of hostOrder (host order) in network order.
- static WebRtc_UWord32 Htonl(WebRtc_UWord32 hostOrder);
-
- // Return IPv4 address in ip as 32 bit integer.
- static WebRtc_UWord32 InetAddrIPV4(const char* ip);
-
- // Convert the character string src into a network address structure in
- // the af address family and put it in dst.
- // Note: same functionality as inet_pton(..)
- static WebRtc_Word32 InetPresentationToNumeric(WebRtc_Word32 af,
- const char* src,
- void* dst);
-
- // Set ip and sourcePort according to address. As input parameter ipSize
- // is the length of ip. As output parameter it's the number of characters
- // written to ip (not counting the '\0' character).
- // Note: this API is only implemented on Windows and Linux.
- static WebRtc_Word32 IPAddress(const SocketAddress& address,
- char* ip,
- WebRtc_UWord32& ipSize,
- WebRtc_UWord16& sourcePort);
-
- // Set ip and sourcePort according to address. As input parameter ipSize
- // is the length of ip. As output parameter it's the number of characters
- // written to ip (not counting the '\0' character).
- // Note: this API is only implemented on Windows and Linux.
- // Additional note: this API caches the address of the last call to it. If
- // address is likley to be the same for multiple calls it may be beneficial
- // to call this API instead of IPAddress().
- virtual WebRtc_Word32 IPAddressCached(const SocketAddress& address,
- char* ip,
- WebRtc_UWord32& ipSize,
- WebRtc_UWord16& sourcePort) = 0;
-
- // Return true if ipaddr is a valid IP address.
- // If ipV6 is false ipaddr is interpreted as an IPv4 address otherwise it
- // is interptreted as IPv6.
- static bool IsIpAddressValid(const char* ipaddr, const bool ipV6);
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_UDP_TRANSPORT_INTERFACE_UDP_TRANSPORT_H_
diff --git a/trunk/src/modules/udp_transport/source/Android.mk b/trunk/src/modules/udp_transport/source/Android.mk
deleted file mode 100644
index f3cb0e8..0000000
--- a/trunk/src/modules/udp_transport/source/Android.mk
+++ /dev/null
@@ -1,45 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_udp_transport
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := \
- udp_transport_impl.cc \
- udp_socket_wrapper.cc \
- udp_socket_manager_wrapper.cc \
- udp_socket_manager_posix.cc \
- udp_socket_posix.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../.. \
- $(LOCAL_PATH)/../../interface \
- $(LOCAL_PATH)/../../../system_wrappers/interface
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/udp_transport/source/traffic_control_windows.cc b/trunk/src/modules/udp_transport/source/traffic_control_windows.cc
deleted file mode 100644
index 09038c0..0000000
--- a/trunk/src/modules/udp_transport/source/traffic_control_windows.cc
+++ /dev/null
@@ -1,253 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "traffic_control_windows.h"
-
-#include <assert.h>
-
-#include "trace.h"
-
-namespace webrtc {
-TrafficControlWindows* TrafficControlWindows::instance = NULL;
-WebRtc_UWord32 TrafficControlWindows::refCounter = 0;
-
-TrafficControlWindows::TrafficControlWindows(const WebRtc_Word32 id) : _id(id)
-{
-}
-
-TrafficControlWindows* TrafficControlWindows::GetInstance(
- const WebRtc_Word32 id)
-{
- if(instance != NULL)
- {
- WEBRTC_TRACE(
- kTraceDebug,
- kTraceTransport,
- id,
- "TrafficControlWindows - Returning already created object");
- refCounter++;
- return instance;
- }
-
- WEBRTC_TRACE(kTraceMemory, kTraceTransport, id,
- "TrafficControlWindows - Creating new object");
- instance = new TrafficControlWindows(id);
- if(instance == NULL)
- {
- WEBRTC_TRACE(kTraceMemory, kTraceTransport, id,
- "TrafficControlWindows - Error allocating memory");
- return NULL;
- }
-
- instance->tcRegister = NULL;
- instance->tcDeregister = NULL;
-
- instance->tcEnumerate = NULL;
- instance->tcOpenInterface = NULL;
- instance->tcCloseInterface = NULL;
-
- instance->tcAddFlow = NULL;
- instance->tcDeleteFlow = NULL;
-
- instance->tcAddFilter = NULL;
- instance->tcDeleteFilter = NULL;
-
- HMODULE trafficLib = LoadLibrary(TEXT("traffic.dll"));
- if(trafficLib == NULL)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceTransport,
- id,
- "TrafficControlWindows - No QOS support, LoadLibrary returned NULL,\
- last error: %d\n",
- GetLastError());
- delete instance;
- instance = NULL;
- return NULL;
- }
-
- instance->tcRegister = (registerFn)GetProcAddress(trafficLib,
- "TcRegisterClient");
- instance->tcDeregister = (deregisterFn)GetProcAddress(trafficLib,
- "TcDeregisterClient");
- instance->tcEnumerate = (enumerateFn)GetProcAddress(
- trafficLib,
- "TcEnumerateInterfaces");
- instance->tcOpenInterface = (openInterfaceFn)GetProcAddress(
- trafficLib,
- "TcOpenInterfaceW");
- instance->tcCloseInterface = (closeInterfaceFn)GetProcAddress(
- trafficLib,
- "TcCloseInterface");
- instance->tcAddFlow = (flowAddFn)GetProcAddress(trafficLib,
- "TcAddFlow");
- instance->tcDeleteFlow = (flowDeleteFn)GetProcAddress(trafficLib,
- "TcDeleteFlow");
-
- instance->tcAddFilter = (filterAddFn)GetProcAddress(trafficLib,
- "TcAddFilter");
- instance->tcDeleteFilter = (filterDeleteFn)GetProcAddress(trafficLib,
- "TcDeleteFilter");
-
- if(instance->tcRegister == NULL ||
- instance->tcDeregister == NULL ||
- instance->tcEnumerate == NULL ||
- instance->tcOpenInterface == NULL ||
- instance->tcCloseInterface == NULL ||
- instance->tcAddFlow == NULL ||
- instance->tcAddFilter == NULL ||
- instance->tcDeleteFlow == NULL ||
- instance->tcDeleteFilter == NULL)
- {
- delete instance;
- instance = NULL;
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- id,
- "TrafficControlWindows - Could not find function pointer for\
- traffic control functions");
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- id,
- "Tcregister : %x, tcDeregister: %x, tcEnumerate: %x,\
- tcOpenInterface: %x, tcCloseInterface: %x, tcAddFlow: %x, tcAddFilter: %x,\
- tcDeleteFlow: %x, tcDeleteFilter: %x",
- instance->tcRegister,
- instance->tcDeregister,
- instance->tcEnumerate,
- instance->tcOpenInterface,
- instance->tcCloseInterface,
- instance->tcAddFlow,
- instance->tcAddFilter,
- instance->tcDeleteFlow,
- instance->tcDeleteFilter );
- return NULL;
- }
- refCounter++;
- return instance;
-}
-
-void TrafficControlWindows::Release(TrafficControlWindows* gtc)
-{
- if (0 == refCounter)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, -1,
- "TrafficControlWindows - Cannot release, refCounter is 0");
- return;
- }
- if (NULL == gtc)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, -1,
- "TrafficControlWindows - Not releasing, gtc is NULL");
- return;
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, gtc->_id,
- "TrafficControlWindows - Releasing object");
- refCounter--;
- if ((0 == refCounter) && instance)
- {
- WEBRTC_TRACE(kTraceMemory, kTraceTransport, gtc->_id,
- "TrafficControlWindows - Deleting object");
- delete instance;
- instance = NULL;
- }
-}
-WebRtc_Word32 TrafficControlWindows::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
- return 0;
-}
-
-ULONG TrafficControlWindows::TcRegisterClient(
- ULONG TciVersion,
- HANDLE ClRegCtx,
- PTCI_CLIENT_FUNC_LIST ClientHandlerList,
- PHANDLE pClientHandle)
-{
- assert(tcRegister != NULL);
-
- return tcRegister(TciVersion, ClRegCtx, ClientHandlerList, pClientHandle);
-}
-
-ULONG TrafficControlWindows::TcDeregisterClient(HANDLE clientHandle)
-{
- assert(tcDeregister != NULL);
-
- return tcDeregister(clientHandle);
-}
-
-
-ULONG TrafficControlWindows::TcEnumerateInterfaces(
- HANDLE ClientHandle,
- PULONG pBufferSize,
- PTC_IFC_DESCRIPTOR interfaceBuffer)
-{
- assert(tcEnumerate != NULL);
-
- return tcEnumerate(ClientHandle, pBufferSize, interfaceBuffer);
-}
-
-
-ULONG TrafficControlWindows::TcOpenInterfaceW(LPWSTR pInterfaceName,
- HANDLE ClientHandle,
- HANDLE ClIfcCtx,
- PHANDLE pIfcHandle)
-{
- assert(tcOpenInterface != NULL);
-
- return tcOpenInterface(pInterfaceName, ClientHandle, ClIfcCtx, pIfcHandle);
-
-}
-
-ULONG TrafficControlWindows::TcCloseInterface(HANDLE IfcHandle)
-{
- assert(tcCloseInterface != NULL);
-
- return tcCloseInterface(IfcHandle);
-}
-
-ULONG TrafficControlWindows::TcAddFlow(HANDLE IfcHandle, HANDLE ClFlowCtx,
- ULONG Flags, PTC_GEN_FLOW pGenericFlow,
- PHANDLE pFlowHandle)
-{
- assert(tcAddFlow != NULL);
- return tcAddFlow(IfcHandle, ClFlowCtx, Flags, pGenericFlow, pFlowHandle);
-}
-
-ULONG TrafficControlWindows::TcAddFilter(HANDLE FlowHandle,
- PTC_GEN_FILTER pGenericFilter,
- PHANDLE pFilterHandle)
-{
- assert(tcAddFilter != NULL);
- return tcAddFilter(FlowHandle, pGenericFilter, pFilterHandle);
-}
-
-ULONG TrafficControlWindows::TcDeleteFlow(HANDLE FlowHandle)
-{
- assert(tcDeleteFlow != NULL);
- return tcDeleteFlow(FlowHandle);
-
-}
-
-ULONG TrafficControlWindows::TcDeleteFilter(HANDLE FilterHandle)
-{
- assert(tcDeleteFilter != NULL);
- return tcDeleteFilter(FilterHandle);
-}
-
-void MyClNotifyHandler(HANDLE ClRegCtx, HANDLE ClIfcCtx, ULONG Event,
- HANDLE SubCode, ULONG BufSize, PVOID Buffer)
-{
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/traffic_control_windows.h b/trunk/src/modules/udp_transport/source/traffic_control_windows.h
deleted file mode 100644
index cfa52ce..0000000
--- a/trunk/src/modules/udp_transport/source/traffic_control_windows.h
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_TRAFFIC_CONTROL_WINDOWS_H_
-#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_TRAFFIC_CONTROL_WINDOWS_H_
-
-#ifndef WIN32_LEAN_AND_MEAN
-#define WIN32_LEAN_AND_MEAN
-#endif
-
-// Disable deprication warning from traffic.h
-#pragma warning(disable : 4995)
-
-#include <windows.h>
-#include <qos.h>
-#include <ntddndis.h>
-#include <traffic.h>
-
-#include "trace.h"
-
-namespace webrtc {
-void MyClNotifyHandler(HANDLE ClRegCtx, HANDLE ClIfcCtx, ULONG Event,
- HANDLE SubCode, ULONG BufSize, PVOID Buffer);
-
-
-typedef ULONG (WINAPI *registerFn)(ULONG, HANDLE, PTCI_CLIENT_FUNC_LIST,
- PHANDLE);
-typedef ULONG (WINAPI *deregisterFn)(HANDLE);
-typedef ULONG (WINAPI *enumerateFn)(HANDLE, PULONG, PTC_IFC_DESCRIPTOR);
-typedef ULONG (WINAPI *openInterfaceFn)(LPWSTR, HANDLE, HANDLE, PHANDLE);
-typedef ULONG (WINAPI *closeInterfaceFn)(HANDLE);
-typedef ULONG (WINAPI *flowAddFn)(HANDLE, HANDLE, ULONG, PTC_GEN_FLOW, PHANDLE);
-typedef ULONG (WINAPI *filterAddFn)(HANDLE, PTC_GEN_FILTER, PHANDLE);
-typedef ULONG (WINAPI *flowDeleteFn)(HANDLE);
-typedef ULONG (WINAPI *filterDeleteFn)(HANDLE);
-
-class TrafficControlWindows
-{
- public:
- // Factory method. Constructor disabled.
- static TrafficControlWindows* GetInstance(const WebRtc_Word32 id);
- static void Release(TrafficControlWindows* gtc);
-
- WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- ULONG TcRegisterClient(ULONG TciVersion, HANDLE ClRegCtx,
- PTCI_CLIENT_FUNC_LIST ClientHandlerList,
- PHANDLE pClientHandle);
-
- ULONG TcDeregisterClient(HANDLE clientHandle);
-
- ULONG TcEnumerateInterfaces(HANDLE ClientHandle, PULONG pBufferSize,
- PTC_IFC_DESCRIPTOR interfaceBuffer);
-
- ULONG TcOpenInterfaceW(LPWSTR pInterfaceName, HANDLE ClientHandle,
- HANDLE ClIfcCtx, PHANDLE pIfcHandle);
-
- ULONG TcCloseInterface(HANDLE IfcHandle);
-
- ULONG TcAddFlow(HANDLE IfcHandle, HANDLE ClFlowCtx, ULONG Flags,
- PTC_GEN_FLOW pGenericFlow, PHANDLE pFlowHandle);
-
- ULONG TcAddFilter(HANDLE FlowHandle, PTC_GEN_FILTER pGenericFilter,
- PHANDLE pFilterHandle);
-
- ULONG TcDeleteFlow(HANDLE FlowHandle);
- ULONG TcDeleteFilter(HANDLE FilterHandle);
-private:
- TrafficControlWindows(const WebRtc_Word32 id);
- WebRtc_Word32 _id;
- TCI_CLIENT_FUNC_LIST QoSFunctions;
-
- static TrafficControlWindows* instance;
-
- registerFn tcRegister;
- deregisterFn tcDeregister;
-
- enumerateFn tcEnumerate;
- openInterfaceFn tcOpenInterface;
- closeInterfaceFn tcCloseInterface;
-
- flowAddFn tcAddFlow;
- flowDeleteFn tcDeleteFlow;
-
- filterAddFn tcAddFilter;
- filterDeleteFn tcDeleteFilter;
-
- static WebRtc_UWord32 refCounter;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_TRAFFIC_CONTROL_WINDOWS_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_socket2_manager_windows.cc b/trunk/src/modules/udp_transport/source/udp_socket2_manager_windows.cc
deleted file mode 100644
index 0356331..0000000
--- a/trunk/src/modules/udp_transport/source/udp_socket2_manager_windows.cc
+++ /dev/null
@@ -1,658 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "udp_socket2_manager_windows.h"
-
-#include <assert.h>
-#include <stdio.h>
-
-#include "aligned_malloc.h"
-#include "udp_socket2_windows.h"
-
-namespace webrtc {
-WebRtc_UWord32 UdpSocket2ManagerWindows::_numOfActiveManagers = 0;
-bool UdpSocket2ManagerWindows::_wsaInit = false;
-
-UdpSocket2ManagerWindows::UdpSocket2ManagerWindows()
- : UdpSocketManager(),
- _id(-1),
- _stopped(false),
- _init(false),
- _pCrit(CriticalSectionWrapper::CreateCriticalSection()),
- _ioCompletionHandle(NULL),
- _numActiveSockets(0),
- _event(EventWrapper::Create())
-{
- _managerNumber = _numOfActiveManagers++;
-
- if(_numOfActiveManagers == 1)
- {
- WORD wVersionRequested = MAKEWORD(2, 2);
- WSADATA wsaData;
- _wsaInit = WSAStartup(wVersionRequested, &wsaData) == 0;
- // TODO (hellner): seems safer to use RAII for this. E.g. what happens
- // if a UdpSocket2ManagerWindows() created and destroyed
- // without being initialized.
- }
-}
-
-UdpSocket2ManagerWindows::~UdpSocket2ManagerWindows()
-{
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocket2ManagerWindows(%d)::~UdpSocket2ManagerWindows()",
- _managerNumber);
-
- if(_init)
- {
- _pCrit->Enter();
- if(_numActiveSockets)
- {
- _pCrit->Leave();
- _event->Wait(INFINITE);
- }
- else
- {
- _pCrit->Leave();
- }
- StopWorkerThreads();
-
- // All threads are stopped. Safe to delete them.
- ListItem* pItem = NULL;
- while((pItem = _workerThreadsList.First()) != NULL)
- {
- delete static_cast<UdpSocket2WorkerWindows*>(pItem->GetItem());
- _workerThreadsList.PopFront();
- }
-
- _ioContextPool.Free();
-
- _numOfActiveManagers--;
- if(_ioCompletionHandle)
- {
- CloseHandle(_ioCompletionHandle);
- }
- if (_numOfActiveManagers == 0)
- {
- if(_wsaInit)
- {
- WSACleanup();
- }
- }
- }
- if(_pCrit)
- {
- delete _pCrit;
- }
- if(_event)
- {
- delete _event;
- }
-}
-
-bool UdpSocket2ManagerWindows::Init(WebRtc_Word32 id,
- WebRtc_UWord8& numOfWorkThreads) {
- CriticalSectionScoped cs(_pCrit);
- if ((_id != -1) || (_numOfWorkThreads != 0)) {
- assert(_id != -1);
- assert(_numOfWorkThreads != 0);
- return false;
- }
- _id = id;
- _numOfWorkThreads = numOfWorkThreads;
- return true;
-}
-
-WebRtc_Word32 UdpSocket2ManagerWindows::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
- return 0;
-}
-
-bool UdpSocket2ManagerWindows::Start()
-{
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocket2ManagerWindows(%d)::Start()",_managerNumber);
- if(!_init)
- {
- StartWorkerThreads();
- }
-
- if(!_init)
- {
- return false;
- }
- _pCrit->Enter();
- // Start worker threads.
- _stopped = false;
- WebRtc_Word32 i = 0;
- WebRtc_Word32 error = 0;
- ListItem* pItem = _workerThreadsList.First();
- UdpSocket2WorkerWindows* pWorker;
- while(pItem != NULL && !error)
- {
- pWorker = (UdpSocket2WorkerWindows*)pItem->GetItem();
- if(!pWorker->Start())
- error = 1;
- pItem = _workerThreadsList.Next(pItem);
- }
- if(error)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2ManagerWindows(%d)::Start() error starting worker\
- threads",
- _managerNumber);
- _pCrit->Leave();
- return false;
- }
- _pCrit->Leave();
- return true;
-}
-
-bool UdpSocket2ManagerWindows::StartWorkerThreads()
-{
- if(!_init)
- {
- _pCrit->Enter();
-
- _ioCompletionHandle = CreateIoCompletionPort(INVALID_HANDLE_VALUE, NULL,
- 0, 0);
- if(_ioCompletionHandle == NULL)
- {
- WebRtc_Word32 error = GetLastError();
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2ManagerWindows(%d)::StartWorkerThreads()"
- "_ioCompletioHandle == NULL: error:%d",
- _managerNumber,error);
- _pCrit->Leave();
- return false;
- }
-
- // Create worker threads.
- WebRtc_UWord32 i = 0;
- bool error = false;
- while(i < _numOfWorkThreads && !error)
- {
- UdpSocket2WorkerWindows* pWorker =
- new UdpSocket2WorkerWindows(_ioCompletionHandle);
- if(pWorker->Init() != 0)
- {
- error = true;
- delete pWorker;
- break;
- }
- _workerThreadsList.PushFront(pWorker);
- i++;
- }
- if(error)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2ManagerWindows(%d)::StartWorkerThreads() error "
- "creating work threads",
- _managerNumber);
- // Delete worker threads.
- ListItem* pItem = NULL;
- while((pItem = _workerThreadsList.First()) != NULL)
- {
- delete static_cast<UdpSocket2WorkerWindows*>(pItem->GetItem());
- _workerThreadsList.PopFront();
- }
- _pCrit->Leave();
- return false;
- }
- if(_ioContextPool.Init())
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2ManagerWindows(%d)::StartWorkerThreads() error "
- "initiating _ioContextPool",
- _managerNumber);
- _pCrit->Leave();
- return false;
- }
- _init = true;
- WEBRTC_TRACE(
- kTraceDebug,
- kTraceTransport,
- _id,
- "UdpSocket2ManagerWindows::StartWorkerThreads %d number of work "
- "threads created and initialized",
- _numOfWorkThreads);
- _pCrit->Leave();
- }
- return true;
-}
-
-bool UdpSocket2ManagerWindows::Stop()
-{
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocket2ManagerWindows(%d)::Stop()",_managerNumber);
-
- if(!_init)
- {
- return false;
- }
- _pCrit->Enter();
- _stopped = true;
- if(_numActiveSockets)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2ManagerWindows(%d)::Stop() there is still active\
- sockets",
- _managerNumber);
- _pCrit->Leave();
- return false;
- }
- // No active sockets. Stop all worker threads.
- bool result = StopWorkerThreads();
- _pCrit->Leave();
- return result;
-}
-
-bool UdpSocket2ManagerWindows::StopWorkerThreads()
-{
- WebRtc_Word32 error = 0;
- WEBRTC_TRACE(
- kTraceDebug,
- kTraceTransport,
- _id,
- "UdpSocket2ManagerWindows(%d)::StopWorkerThreads() Worker\
- threadsStoped, numActicve Sockets=%d",
- _managerNumber,
- _numActiveSockets);
- UdpSocket2WorkerWindows* pWorker;
- ListItem* pItem = _workerThreadsList.First();
-
- // Set worker threads to not alive so that they will stop calling
- // UdpSocket2WorkerWindows::Run().
- while(pItem != NULL)
- {
- pWorker = (UdpSocket2WorkerWindows*)pItem->GetItem();
- pWorker->SetNotAlive();
- pItem = _workerThreadsList.Next(pItem);
- }
- // Release all threads waiting for GetQueuedCompletionStatus(..).
- if(_ioCompletionHandle)
- {
- WebRtc_UWord32 i = 0;
- for(i = 0; i < _workerThreadsList.GetSize(); i++)
- {
- PostQueuedCompletionStatus(_ioCompletionHandle, 0 ,0 , NULL);
- }
- }
- pItem = _workerThreadsList.First();
-
- while(pItem != NULL)
- {
- pWorker = (UdpSocket2WorkerWindows*)pItem->GetItem();
- if(pWorker->Stop() == false)
- {
- error = -1;
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, -1,
- "failed to stop worker thread");
- }
- pItem = _workerThreadsList.Next(pItem);
- }
-
- if(error)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2ManagerWindows(%d)::StopWorkerThreads() error stopping\
- worker threads",
- _managerNumber);
- return false;
- }
- return true;
-}
-
-bool UdpSocket2ManagerWindows::AddSocketPrv(UdpSocket2Windows* s)
-{
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocket2ManagerWindows(%d)::AddSocketPrv()",_managerNumber);
- if(!_init)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2ManagerWindows(%d)::AddSocketPrv() manager not\
- initialized",
- _managerNumber);
- return false;
- }
- _pCrit->Enter();
- if(s == NULL)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2ManagerWindows(%d)::AddSocketPrv() socket == NULL",
- _managerNumber);
- _pCrit->Leave();
- return false;
- }
- if(s->GetFd() == NULL || s->GetFd() == INVALID_SOCKET)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2ManagerWindows(%d)::AddSocketPrv() socket->GetFd() ==\
- %d",
- _managerNumber,
- (WebRtc_Word32)s->GetFd());
- _pCrit->Leave();
- return false;
-
- }
- _ioCompletionHandle = CreateIoCompletionPort((HANDLE)s->GetFd(),
- _ioCompletionHandle,
- (ULONG_PTR)(s), 0);
- if(_ioCompletionHandle == NULL)
- {
- WebRtc_Word32 error = GetLastError();
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2ManagerWindows(%d)::AddSocketPrv() Error adding to IO\
- completion: %d",
- _managerNumber,
- error);
- _pCrit->Leave();
- return false;
- }
- _numActiveSockets++;
- _pCrit->Leave();
- return true;
-}
-bool UdpSocket2ManagerWindows::RemoveSocketPrv(UdpSocket2Windows* s)
-{
- if(!_init)
- {
- return false;
- }
- _pCrit->Enter();
- _numActiveSockets--;
- if(_numActiveSockets == 0)
- {
- _event->Set();
- }
- _pCrit->Leave();
- return true;
-}
-
-PerIoContext* UdpSocket2ManagerWindows::PopIoContext()
-{
- if(!_init)
- {
- return NULL;
- }
-
- PerIoContext* pIoC = NULL;
- if(!_stopped)
- {
- pIoC = _ioContextPool.PopIoContext();
- }else
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2ManagerWindows(%d)::PopIoContext() Manager Not started",
- _managerNumber);
- }
- return pIoC;
-}
-
-WebRtc_Word32 UdpSocket2ManagerWindows::PushIoContext(PerIoContext* pIoContext)
-{
- return _ioContextPool.PushIoContext(pIoContext);
-}
-
-IoContextPool::IoContextPool()
- : _pListHead(NULL),
- _init(false),
- _size(0),
- _inUse(0)
-{
-}
-
-IoContextPool::~IoContextPool()
-{
- Free();
- assert(_size.Value() == 0);
- AlignedFree(_pListHead);
-}
-
-WebRtc_Word32 IoContextPool::Init(WebRtc_UWord32 /*increaseSize*/)
-{
- if(_init)
- {
- return 0;
- }
-
- _pListHead = (PSLIST_HEADER)AlignedMalloc(sizeof(SLIST_HEADER),
- MEMORY_ALLOCATION_ALIGNMENT);
- if(_pListHead == NULL)
- {
- return -1;
- }
- InitializeSListHead(_pListHead);
- _init = true;
- return 0;
-}
-
-PerIoContext* IoContextPool::PopIoContext()
-{
- if(!_init)
- {
- return NULL;
- }
-
- PSLIST_ENTRY pListEntry = InterlockedPopEntrySList(_pListHead);
- if(pListEntry == NULL)
- {
- IoContextPoolItem* item = (IoContextPoolItem*)
- AlignedMalloc(
- sizeof(IoContextPoolItem),
- MEMORY_ALLOCATION_ALIGNMENT);
- if(item == NULL)
- {
- return NULL;
- }
- memset(&item->payload.ioContext,0,sizeof(PerIoContext));
- item->payload.base = item;
- pListEntry = &(item->itemEntry);
- ++_size;
- }
- ++_inUse;
- return &((IoContextPoolItem*)pListEntry)->payload.ioContext;
-}
-
-WebRtc_Word32 IoContextPool::PushIoContext(PerIoContext* pIoContext)
-{
- // TODO (hellner): Overlapped IO should be completed at this point. Perhaps
- // add an assert?
- const bool overlappedIOCompleted = HasOverlappedIoCompleted(
- (LPOVERLAPPED)pIoContext);
-
- IoContextPoolItem* item = ((IoContextPoolItemPayload*)pIoContext)->base;
-
- const WebRtc_Word32 usedItems = --_inUse;
- const WebRtc_Word32 totalItems = _size.Value();
- const WebRtc_Word32 freeItems = totalItems - usedItems;
- if(freeItems < 0)
- {
- assert(false);
- AlignedFree(item);
- return -1;
- }
- if((freeItems >= totalItems>>1) &&
- overlappedIOCompleted)
- {
- AlignedFree(item);
- --_size;
- return 0;
- }
- InterlockedPushEntrySList(_pListHead, &(item->itemEntry));
- return 0;
-}
-
-WebRtc_Word32 IoContextPool::Free()
-{
- if(!_init)
- {
- return 0;
- }
-
- WebRtc_Word32 itemsFreed = 0;
- PSLIST_ENTRY pListEntry = InterlockedPopEntrySList(_pListHead);
- while(pListEntry != NULL)
- {
- IoContextPoolItem* item = ((IoContextPoolItem*)pListEntry);
- AlignedFree(item);
- --_size;
- itemsFreed++;
- pListEntry = InterlockedPopEntrySList(_pListHead);
- }
- return itemsFreed;
-}
-
-WebRtc_Word32 UdpSocket2WorkerWindows::_numOfWorkers = 0;
-
-UdpSocket2WorkerWindows::UdpSocket2WorkerWindows(HANDLE ioCompletionHandle)
- : _ioCompletionHandle(ioCompletionHandle),
- _pThread(NULL),
- _init(false)
-{
- _workerNumber = _numOfWorkers++;
- WEBRTC_TRACE(kTraceMemory, kTraceTransport, -1,
- "UdpSocket2WorkerWindows created");
-}
-
-UdpSocket2WorkerWindows::~UdpSocket2WorkerWindows()
-{
- if(_pThread)
- {
- delete _pThread;
- }
- WEBRTC_TRACE(kTraceMemory, kTraceTransport, -1,
- "UdpSocket2WorkerWindows deleted");
-}
-
-bool UdpSocket2WorkerWindows::Start()
-{
- unsigned int id = 0;
- WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, -1,
- "Start UdpSocket2WorkerWindows");
- return _pThread->Start(id);
-}
-
-bool UdpSocket2WorkerWindows::Stop()
-{
- WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, -1,
- "Stop UdpSocket2WorkerWindows");
- return _pThread->Stop();
-}
-
-void UdpSocket2WorkerWindows::SetNotAlive()
-{
- WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, -1,
- "SetNotAlive UdpSocket2WorkerWindows");
- _pThread->SetNotAlive();
-}
-
-WebRtc_Word32 UdpSocket2WorkerWindows::Init()
-{
- if(!_init)
- {
- const WebRtc_Word8* threadName = "UdpSocket2ManagerWindows_thread";
- _pThread = ThreadWrapper::CreateThread(Run, this, kRealtimePriority,
- threadName);
- if(_pThread == NULL)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- -1,
- "UdpSocket2WorkerWindows(%d)::Init(), error creating thread!",
- _workerNumber);
- return -1;
- }
- _init = true;
- }
- return 0;
-}
-
-bool UdpSocket2WorkerWindows::Run(ThreadObj obj)
-{
- UdpSocket2WorkerWindows* pWorker =
- static_cast<UdpSocket2WorkerWindows*>(obj);
- return pWorker->Process();
-}
-
-// Process should always return true. Stopping the worker threads is done in
-// the UdpSocket2ManagerWindows::StopWorkerThreads() function.
-bool UdpSocket2WorkerWindows::Process()
-{
- WebRtc_Word32 success = 0;
- DWORD ioSize = 0;
- UdpSocket2Windows* pSocket = NULL;
- PerIoContext* pIOContext = 0;
- OVERLAPPED* pOverlapped = 0;
- success = GetQueuedCompletionStatus(_ioCompletionHandle,
- &ioSize,
- (ULONG_PTR*)&pSocket, &pOverlapped, 200);
-
- WebRtc_UWord32 error = 0;
- if(!success)
- {
- error = GetLastError();
- if(error == WAIT_TIMEOUT)
- {
- return true;
- }
- // This may happen if e.g. PostQueuedCompletionStatus() has been called.
- // The IO context still needs to be reclaimed or re-used which is done
- // in UdpSocket2Windows::IOCompleted(..).
- }
- if(pSocket == NULL)
- {
- WEBRTC_TRACE(
- kTraceDebug,
- kTraceTransport,
- -1,
- "UdpSocket2WorkerWindows(%d)::Process(), pSocket == 0, end thread",
- _workerNumber);
- return true;
- }
- pIOContext = (PerIoContext*)pOverlapped;
- pSocket->IOCompleted(pIOContext,ioSize,error);
- return true;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_socket2_manager_windows.h b/trunk/src/modules/udp_transport/source/udp_socket2_manager_windows.h
deleted file mode 100644
index 5adb293..0000000
--- a/trunk/src/modules/udp_transport/source/udp_socket2_manager_windows.h
+++ /dev/null
@@ -1,162 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET2_MANAGER_WINDOWS_H_
-#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET2_MANAGER_WINDOWS_H_
-
-#if _MSC_VER > 1000
-#pragma once
-#endif
-
-#include <winsock2.h>
-
-#include "atomic32_wrapper.h"
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "list_wrapper.h"
-#include "thread_wrapper.h"
-#include "udp_socket2_windows.h"
-#include "udp_socket_manager_wrapper.h"
-
-#define MAX_IO_BUFF_SIZE 1600
-
-namespace webrtc {
-enum IO_OPERATION {
- OP_READ,
- OP_WRITE
-};
-
-class UdpSocket2Windows;
-
-// Struct used for all socket I/O operations.
-struct PerIoContext {
- WSAOVERLAPPED overlapped;
- char buffer[MAX_IO_BUFF_SIZE];
- WSABUF wsabuf;
- int nTotalBytes;
- int nSentBytes;
- int bytes;
- IO_OPERATION ioOperation;
- SocketAddress from;
- int fromLen;
- // Should be set to true if the I/O context was passed to the system by
- // a thread not controlled by the socket implementation.
- bool ioInitiatedByThreadWrapper;
- // TODO (hellner): Not used. Delete it.
- PerIoContext* pNextFree;
-};
-
-struct IoContextPoolItem;
-struct IoContextPoolItemPayload
-{
- PerIoContext ioContext;
- IoContextPoolItem* base;
-};
-
-struct IoContextPoolItem
-{
- // Atomic single linked list entry header.
- SLIST_ENTRY itemEntry;
- // Atomic single linked list payload
- IoContextPoolItemPayload payload;
-};
-
-class IoContextPool
-{
-public:
- IoContextPool();
- virtual ~IoContextPool();
- virtual WebRtc_Word32 Init(WebRtc_UWord32 increaseSize = 128);
- // Re-use an old unused IO context or create a new one.
- virtual PerIoContext* PopIoContext();
- virtual WebRtc_Word32 PushIoContext(PerIoContext* pIoContext);
- virtual inline WebRtc_Word32 GetSize(WebRtc_UWord32* inUse = 0)
- {return _size.Value();}
- virtual WebRtc_Word32 Free();
-private:
- // Sample code for use of msfts single linked atomic list can be found here:
- // http://msdn.microsoft.com/en-us/library/ms686962(VS.85).aspx
-
- // Atomic single linked list head.
- PSLIST_HEADER _pListHead;
-
- bool _init;
- Atomic32Wrapper _size;
- Atomic32Wrapper _inUse;
-};
-
-
-class UdpSocket2ManagerWindows : public UdpSocketManager
-{
-public:
- UdpSocket2ManagerWindows();
- virtual ~UdpSocket2ManagerWindows();
-
- virtual bool Init(WebRtc_Word32 id, WebRtc_UWord8& numOfWorkThreads);
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- virtual bool Start();
- virtual bool Stop();
-
- virtual inline bool AddSocket(UdpSocketWrapper* s)
- {if(s) return AddSocketPrv(reinterpret_cast<UdpSocket2Windows*>(s));
- return false;}
- virtual bool RemoveSocket(UdpSocketWrapper* s)
- {if(s) return RemoveSocketPrv(reinterpret_cast<UdpSocket2Windows*>(s));
- return false;}
-
- PerIoContext* PopIoContext(void);
- WebRtc_Word32 PushIoContext(PerIoContext* pIoContext);
-
-private:
- bool StopWorkerThreads();
- bool StartWorkerThreads();
- bool AddSocketPrv(UdpSocket2Windows* s);
- bool RemoveSocketPrv(UdpSocket2Windows* s);
-
- static WebRtc_UWord32 _numOfActiveManagers;
- static bool _wsaInit;
-
- WebRtc_Word32 _id;
- CriticalSectionWrapper* _pCrit;
- WebRtc_Word32 _managerNumber;
- volatile bool _stopped;
- bool _init;
- WebRtc_Word32 _numActiveSockets;
- ListWrapper _workerThreadsList;
- EventWrapper* _event;
-
- HANDLE _ioCompletionHandle;
- IoContextPool _ioContextPool;
-};
-
-class UdpSocket2WorkerWindows
-{
-public:
- UdpSocket2WorkerWindows(HANDLE ioCompletionHandle);
- virtual ~UdpSocket2WorkerWindows();
-
- virtual bool Start();
- virtual bool Stop();
- virtual WebRtc_Word32 Init();
- virtual void SetNotAlive();
-protected:
- static bool Run(ThreadObj obj);
- bool Process();
-private:
- HANDLE _ioCompletionHandle;
- ThreadWrapper*_pThread;
- static WebRtc_Word32 _numOfWorkers;
- WebRtc_Word32 _workerNumber;
- volatile bool _stop;
- bool _init;
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET2_MANAGER_WINDOWS_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_socket2_windows.cc b/trunk/src/modules/udp_transport/source/udp_socket2_windows.cc
deleted file mode 100644
index 32e3598..0000000
--- a/trunk/src/modules/udp_transport/source/udp_socket2_windows.cc
+++ /dev/null
@@ -1,1411 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "udp_socket2_windows.h"
-
-#include <assert.h>
-#include <stdlib.h>
-#include <winsock2.h>
-
-#include "traffic_control_windows.h"
-#include "udp_socket2_manager_windows.h"
-
-#pragma warning(disable : 4311)
-
-namespace webrtc {
-typedef struct _QOS_DESTADDR
-{
- QOS_OBJECT_HDR ObjectHdr;
- const struct sockaddr* SocketAddress;
- ULONG SocketAddressLength;
-} QOS_DESTADDR, *LPQOS_DESTADDR;
-
-typedef const QOS_DESTADDR* LPCQOS_DESTADDR;
-
-// TODO (patrikw): seems to be defined in ws2ipdef.h as 3. How come it's
-// redefined here (as a different value)?
-#define IP_TOS 8
-
-#define QOS_GENERAL_ID_BASE 2000
-#define QOS_OBJECT_DESTADDR (0x00000004 + QOS_GENERAL_ID_BASE)
-
-UdpSocket2Windows::UdpSocket2Windows(const WebRtc_Word32 id,
- UdpSocketManager* mgr, bool ipV6Enable,
- bool disableGQOS)
- : _id(id),
- _qos(true),
- _iProtocol(0),
- _outstandingCalls(0),
- _outstandingCallComplete(0),
- _terminate(false),
- _addedToMgr(false),
- _safeTodelete(false),
- _outstandingCallsDisabled(0),
- _clientHandle(NULL),
- _flowHandle(NULL),
- _filterHandle(NULL),
- _flow(NULL),
- _gtc(NULL),
- _pcp(-2),
- _receiveBuffers(0)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceTransport, _id,
- "UdpSocket2Windows::UdpSocket2Windows()");
-
- _wantsIncoming = false;
- _mgr = static_cast<UdpSocket2ManagerWindows *>(mgr);
-
- _obj = NULL;
- _incomingCb = NULL;
- _socket = INVALID_SOCKET;
- _pCrit = CriticalSectionWrapper::CreateCriticalSection();
- _ptrCbRWLock = RWLockWrapper::CreateRWLock();
- _ptrDestRWLock = RWLockWrapper::CreateRWLock();
- _ptrSocketRWLock = RWLockWrapper::CreateRWLock();
- _ptrDeleteCrit = CriticalSectionWrapper::CreateCriticalSection();
- _ptrDeleteCond = ConditionVariableWrapper::CreateConditionVariable();
-
- // Check if QoS is supported.
- BOOL bProtocolFound = FALSE;
- WSAPROTOCOL_INFO *lpProtocolBuf = NULL;
- WSAPROTOCOL_INFO pProtocolInfo;
-
- if(!disableGQOS)
- {
- DWORD dwBufLen = 0;
- // Set dwBufLen to the size needed to retreive all the requested
- // information from WSAEnumProtocols.
- WebRtc_Word32 nRet = WSAEnumProtocols(NULL, lpProtocolBuf, &dwBufLen);
- lpProtocolBuf = (WSAPROTOCOL_INFO*)malloc(dwBufLen);
- nRet = WSAEnumProtocols(NULL, lpProtocolBuf, &dwBufLen);
-
- if (ipV6Enable)
- {
- _iProtocol=AF_INET6;
- } else {
- _iProtocol=AF_INET;
- }
-
- for (WebRtc_Word32 i=0; i<nRet; i++)
- {
- if (_iProtocol == lpProtocolBuf[i].iAddressFamily &&
- IPPROTO_UDP == lpProtocolBuf[i].iProtocol)
- {
- if ((XP1_QOS_SUPPORTED ==
- (XP1_QOS_SUPPORTED & lpProtocolBuf[i].dwServiceFlags1)))
- {
- pProtocolInfo = lpProtocolBuf[i];
- bProtocolFound = TRUE;
- break;
- }
- }
- }
- }
-
- if(!bProtocolFound)
- {
- free(lpProtocolBuf);
- _qos=false;
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2Windows::UdpSocket2Windows(), SOCKET_ERROR_NO_QOS,\
- !bProtocolFound");
- } else {
-
- _socket = WSASocket(FROM_PROTOCOL_INFO, FROM_PROTOCOL_INFO,
- FROM_PROTOCOL_INFO,&pProtocolInfo, 0,
- WSA_FLAG_OVERLAPPED);
- free(lpProtocolBuf);
-
- if (_socket != INVALID_SOCKET)
- {
- return;
- } else {
- _qos = false;
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2Windows::UdpSocket2Windows(), SOCKET_ERROR_NO_QOS");
- }
- }
- // QoS not supported.
- if(ipV6Enable)
- {
- _socket = WSASocket(AF_INET6, SOCK_DGRAM, IPPROTO_UDP, 0 , 0,
- WSA_FLAG_OVERLAPPED);
- }else
- {
- _socket = WSASocket(AF_INET, SOCK_DGRAM, IPPROTO_UDP, 0 , 0,
- WSA_FLAG_OVERLAPPED);
- }
- if (_socket == INVALID_SOCKET)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2Windows::UdpSocket2Windows(), INVALID_SOCKET,\
- WSAerror: %d",
- WSAGetLastError());
- }
-
- // Disable send buffering on the socket to improve CPU usage.
- // This is done by setting SO_SNDBUF to 0.
- WebRtc_Word32 nZero = 0;
- WebRtc_Word32 nRet = setsockopt(_socket, SOL_SOCKET, SO_SNDBUF,
- (WebRtc_Word8*)&nZero, sizeof(nZero));
- if( nRet == SOCKET_ERROR )
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2Windows::UdpSocket2Windows(), SOCKET_ERROR,\
- WSAerror: %d",
- WSAGetLastError());
- }
-}
-
-UdpSocket2Windows::~UdpSocket2Windows()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceTransport, _id,
- "UdpSocket2Windows::~UdpSocket2Windows()");
-
- WaitForOutstandingCalls();
-
- delete _ptrCbRWLock;
- delete _ptrDeleteCrit;
- delete _ptrDeleteCond;
- delete _ptrDestRWLock;
- delete _ptrSocketRWLock;
-
- if(_pCrit)
- delete _pCrit;
-
- if (_flow)
- {
- free(_flow);
- _flow = NULL;
- }
-
- if (_gtc)
- {
- if(_filterHandle)
- {
- _gtc->TcDeleteFilter(_filterHandle);
- }
- if(_flowHandle)
- {
- _gtc->TcDeleteFlow(_flowHandle);
- }
- TrafficControlWindows::Release( _gtc);
- }
-}
-
-WebRtc_Word32 UdpSocket2Windows::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
- if (_gtc)
- {
- _gtc->ChangeUniqueId(id);
- }
- return 0;
-}
-
-bool UdpSocket2Windows::ValidHandle()
-{
- return GetFd() != INVALID_SOCKET;
-}
-
-bool UdpSocket2Windows::SetCallback(CallbackObj obj, IncomingSocketCallback cb)
-{
- _ptrCbRWLock->AcquireLockExclusive();
- _obj = obj;
- _incomingCb = cb;
- _ptrCbRWLock->ReleaseLockExclusive();
-
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocket2Windows(%d)::SetCallback ",(WebRtc_Word32)this);
- if(_addedToMgr)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocket2Windows(%d)::SetCallback alreadey added",
- (WebRtc_Word32) this);
- return false;
-
- }
- if (_mgr->AddSocket(this))
- {
- WEBRTC_TRACE(
- kTraceDebug, kTraceTransport, _id,
- "UdpSocket2Windows(%d)::SetCallback socket added to manager",
- (WebRtc_Word32)this);
- _addedToMgr = true;
- return true;
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocket2Windows(%d)::SetCallback error adding me to mgr",
- (WebRtc_Word32) this);
- return false;
-}
-
-bool UdpSocket2Windows::SetSockopt(WebRtc_Word32 level, WebRtc_Word32 optname,
- const WebRtc_Word8* optval,
- WebRtc_Word32 optlen)
-{
- bool returnValue = true;
- if(!AquireSocket())
- {
- return false;
- }
- if(0 != setsockopt(_socket, level, optname, optval, optlen ))
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "UdpSocket2Windows::SetSockopt(), WSAerror:%d",
- WSAGetLastError());
- returnValue = false;
- }
- ReleaseSocket();
- return returnValue;
-}
-
-bool UdpSocket2Windows::StartReceiving(WebRtc_UWord32 receiveBuffers)
-{
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocket2Windows(%d)::StartReceiving(%d)",
- (WebRtc_Word32)this, receiveBuffers);
-
- _wantsIncoming = true;
-
- WebRtc_Word32 numberOfReceiveBuffersToCreate =
- receiveBuffers - _receiveBuffers.Value();
- numberOfReceiveBuffersToCreate = (numberOfReceiveBuffersToCreate < 0) ?
- 0 : numberOfReceiveBuffersToCreate;
-
- WebRtc_Word32 error = 0;
- for(WebRtc_Word32 i = 0;
- i < numberOfReceiveBuffersToCreate;
- i++)
- {
- if(PostRecv())
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "UdpSocket2Windows::StartReceiving() i=%d", i);
- error = -1;
- break;
- }
- ++_receiveBuffers;
- }
- if(error == -1)
- {
- return false;
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "Socket receiving using:%d number of buffers",
- _receiveBuffers.Value());
- return true;
-}
-
-bool UdpSocket2Windows::StopReceiving()
-{
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocket2Windows::StopReceiving()");
- _wantsIncoming = false;
- return true;
-}
-
-bool UdpSocket2Windows::Bind(const SocketAddress& name)
-{
- const struct sockaddr* addr =
- reinterpret_cast<const struct sockaddr*>(&name);
- bool returnValue = true;
- if(!AquireSocket())
- {
- return false;
- }
- if (0 != bind(_socket, addr, sizeof(SocketAddress)))
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "UdpSocket2Windows::Bind() WSAerror: %d",
- WSAGetLastError());
- returnValue = false;
- }
- ReleaseSocket();
- return returnValue;
-}
-
-WebRtc_Word32 UdpSocket2Windows::SendTo(const WebRtc_Word8* buf,
- WebRtc_Word32 len,
- const SocketAddress& to)
-{
- WebRtc_Word32 retVal = 0;
- WebRtc_Word32 error = 0;
- if(len < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "UdpSocket2Windows(%d)::SendTo(), len= %d < 0",
- (WebRtc_Word32)this, len);
- return -1;
- }
-
- PerIoContext* pIoContext = _mgr->PopIoContext();
- if(pIoContext == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "UdpSocket2Windows(%d)::SendTo(), pIoContext==0",
- (WebRtc_Word32) this);
- return -1;
- }
- // sizeof(pIoContext->buffer) is smaller than the highest number that
- // can be represented by a WebRtc_Word32.
- if(len >= (WebRtc_Word32) sizeof(pIoContext->buffer))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2Windows(%d)::SendTo(), len= %d > buffer_size = %d",
- (WebRtc_Word32) this,
- len,sizeof(pIoContext->buffer));
- len = sizeof(pIoContext->buffer);
- }
-
- memcpy(pIoContext->buffer,buf,len);
- pIoContext->wsabuf.buf = pIoContext->buffer;
- pIoContext->wsabuf.len = len;
- pIoContext->fromLen=sizeof(SocketAddress);
- pIoContext->ioOperation = OP_WRITE;
- pIoContext->nTotalBytes = len;
- pIoContext->nSentBytes=0;
-
- DWORD numOfbytesSent = 0;
- const struct sockaddr* addr = reinterpret_cast<const struct sockaddr*>(&to);
-
- if(!AquireSocket())
- {
- _mgr->PushIoContext(pIoContext);
- return -1;
- }
- // Assume that the WSASendTo call will be successfull to make sure that
- // _outstandingCalls is positive. Roll back if WSASendTo failed.
- if(!NewOutstandingCall())
- {
- _mgr->PushIoContext(pIoContext);
- ReleaseSocket();
- return -1;
- }
- retVal = WSASendTo(_socket, &pIoContext->wsabuf, 1, &numOfbytesSent,
- 0, addr, sizeof(SocketAddress),
- &(pIoContext->overlapped), 0);
- ReleaseSocket();
-
- if( retVal == SOCKET_ERROR )
- {
- error = WSAGetLastError();
- if(error != ERROR_IO_PENDING)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "UdpSocket2Windows::SendTo() WSAerror: %d",error);
- }
- }
- if(retVal == 0 || (retVal == SOCKET_ERROR && error == ERROR_IO_PENDING))
- {
- return len;
- }
- if((error = _mgr->PushIoContext(pIoContext)))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2Windows(%d)::SendTo(), error:%d pushing ioContext",
- (WebRtc_Word32)this, error);
- }
-
- // Roll back.
- OutstandingCallCompleted();
- return -1;
-}
-
-void UdpSocket2Windows::IOCompleted(PerIoContext* pIOContext,
- WebRtc_UWord32 ioSize, WebRtc_UWord32 error)
-{
- if(pIOContext == NULL || error == ERROR_OPERATION_ABORTED)
- {
- if ((pIOContext != NULL) &&
- !pIOContext->ioInitiatedByThreadWrapper &&
- (error == ERROR_OPERATION_ABORTED) &&
- (pIOContext->ioOperation == OP_READ) &&
- _outstandingCallsDisabled.Value() == 1)
- {
- // !pIOContext->initiatedIOByThreadWrapper indicate that the I/O
- // was not initiaded by a ThreadWrapper thread.
- // This may happen if the thread that initiated receiving (e.g.
- // by calling StartListen())) is deleted before any packets have
- // been received.
- // In this case there is no packet in the PerIoContext. Re-use it
- // to post a new PostRecv(..).
- // Note 1: the PerIoContext will henceforth be posted by a thread
- // that is controlled by the socket implementation.
- // Note 2: This is more likely to happen to RTCP packets as
- // they are less frequent than RTP packets.
- // Note 3: _outstandingCallsDisabled being false (= 1) indicates
- // that the socket isn't being shut down.
- // Note 4: This should only happen buffers set to recevie packets
- // (OP_READ).
- if (_outstandingCallsDisabled.Value() != 1)
- {
- WEBRTC_TRACE(
- kTraceDebug,
- kTraceTransport,
- _id,
- "UdpSocket2Windows::IOCompleted(pIOContext=%p,\
- ioSize=%.lu, error=%.lu) Received operation aborted but continuing since\
- pIOContext->ioInitiatedByThreadWrapper == false",
- pIOContext,
- ioSize,
- error);
- WebRtc_Word32 ioOp = pIOContext ?
- (WebRtc_Word32)pIOContext->ioOperation : -1;
- WebRtc_Word32 ioInit = pIOContext ?
- (WebRtc_Word32)pIOContext->ioInitiatedByThreadWrapper : -1;
- WEBRTC_TRACE(
- kTraceDebug,
- kTraceTransport,
- _id,
- "pIOContext->ioOperation=%d,\
- pIOContext->ioInitiatedByThreadWrapper=%d, _outstandingCallsDisabled=%d,\
- _incomingCb=%p, this=%p",
- ioOp,
- ioInit,
- (WebRtc_Word32)_outstandingCallsDisabled.Value(),
- _incomingCb,
- this);
- }
- } else {
- if(pIOContext == NULL)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2Windows::IOCompleted(%d,%d,%d), %d",
- (WebRtc_Word32)pIOContext,
- ioSize,
- error,
- pIOContext ? (WebRtc_Word32)pIOContext->ioOperation : -1);
- } else {
- WEBRTC_TRACE(
- kTraceDebug,
- kTraceTransport,
- _id,
- "UdpSocket2Windows::IOCompleted() Operation aborted");
- }
- if(pIOContext)
- {
- WebRtc_Word32 remainingReceiveBuffers = --_receiveBuffers;
- if(remainingReceiveBuffers < 0)
- {
- assert(false);
- }
- WebRtc_Word32 err = 0;
- if((err = _mgr->PushIoContext(pIOContext)))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2Windows::IOCompleted(), err = %d, when\
- pushing ioContext after error",
- err);
- }
- }
- OutstandingCallCompleted();
- return;
- }
- } // if (pIOContext == NULL || error == ERROR_OPERATION_ABORTED)
-
- if(pIOContext->ioOperation == OP_WRITE)
- {
- _mgr->PushIoContext(pIOContext);
- }
- else if(pIOContext->ioOperation == OP_READ)
- {
- if(!error && ioSize != 0)
- {
- _ptrCbRWLock->AcquireLockShared();
- if(_wantsIncoming && _incomingCb)
- {
- _incomingCb(_obj,pIOContext->wsabuf.buf, ioSize,
- &pIOContext->from);
- }
- _ptrCbRWLock->ReleaseLockShared();
- }
- WebRtc_Word32 err = PostRecv(pIOContext);
- if(err == 0)
- {
- // The PerIoContext was posted by a thread controlled by the socket
- // implementation.
- pIOContext->ioInitiatedByThreadWrapper = true;
- }
- OutstandingCallCompleted();
- return;
- } else {
- // Unknown operation. Should not happen. Return pIOContext to avoid
- // memory leak.
- assert(false);
- _mgr->PushIoContext(pIOContext);
- }
- OutstandingCallCompleted();
- // Don't touch any members after OutstandingCallCompleted() since the socket
- // may be deleted at this point.
-}
-
-WebRtc_Word32 UdpSocket2Windows::PostRecv()
-{
- PerIoContext* pIoContext=_mgr->PopIoContext();
- if(pIoContext == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "UdpSocket2Windows(%d)::PostRecv(), pIoContext == 0",
- (WebRtc_Word32)this);
- return -1;
- }
- // This function may have been called by thread not controlled by the socket
- // implementation.
- pIoContext->ioInitiatedByThreadWrapper = false;
- return PostRecv(pIoContext);
-}
-
-WebRtc_Word32 UdpSocket2Windows::PostRecv(PerIoContext* pIoContext)
-{
- if(pIoContext==0)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "UdpSocket2Windows(%d)::PostRecv(?), pIoContext==0",
- (WebRtc_Word32)this);
- return -1;
- }
-
- DWORD numOfRecivedBytes = 0;
- DWORD flags = 0;
- pIoContext->wsabuf.buf = pIoContext->buffer;
- pIoContext->wsabuf.len = sizeof(pIoContext->buffer);
- pIoContext->fromLen = sizeof(SocketAddress);
- pIoContext->ioOperation = OP_READ;
- WebRtc_Word32 rxError = 0;
- WebRtc_Word32 nRet = 0;
- WebRtc_Word32 postingSucessfull = false;
-
- if(!AquireSocket())
- {
- _mgr->PushIoContext(pIoContext);
- return -1;
- }
-
- // Assume that the WSARecvFrom() call will be successfull to make sure that
- // _outstandingCalls is positive. Roll back if WSARecvFrom() failed.
- if(!NewOutstandingCall())
- {
- _mgr->PushIoContext(pIoContext);
- ReleaseSocket();
- return -1;
- }
- for(WebRtc_Word32 tries = 0; tries < 10; tries++)
- {
- nRet = WSARecvFrom(
- _socket,
- &(pIoContext->wsabuf),
- 1,
- &numOfRecivedBytes,
- &flags,
- reinterpret_cast<struct sockaddr*>(&(pIoContext->from)),
- &(pIoContext->fromLen),
- &(pIoContext->overlapped),
- 0);
-
- if( nRet == SOCKET_ERROR)
- {
- rxError = WSAGetLastError();
- if(rxError != ERROR_IO_PENDING)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2Windows(%d)::PostRecv(?), WSAerror:%d when\
- posting new recieve,trie:%d",
- (WebRtc_Word32)this,
- rxError,
- tries);
- // Tell the OS that this is a good place to context switch if
- // it wants to.
- Sleep(0);
- }
- }
- if((rxError == ERROR_IO_PENDING) || (nRet == 0))
- {
- postingSucessfull = true;
- break;
- }
- }
- ReleaseSocket();
-
- if(postingSucessfull)
- {
- return 0;
- }
- WebRtc_Word32 remainingReceiveBuffers = --_receiveBuffers;
- if(remainingReceiveBuffers < 0)
- {
- assert(false);
- }
- WebRtc_Word32 error = 0;
- if((error = _mgr->PushIoContext(pIoContext)))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocket2Windows(%d)::PostRecv(?), error:%d when PushIoContext",
- (WebRtc_Word32)this,
- error);
- }
- // Roll back.
- OutstandingCallCompleted();
- return -1;
-}
-
-void UdpSocket2Windows::CloseBlocking()
-{
- LINGER lingerStruct;
-
- lingerStruct.l_onoff = 1;
- lingerStruct.l_linger = 0;
- if(AquireSocket())
- {
- setsockopt(_socket, SOL_SOCKET, SO_LINGER,
- (WebRtc_Word8 *)&lingerStruct, sizeof(lingerStruct));
- ReleaseSocket();
- }
-
- _wantsIncoming = false;
- // Reclaims the socket and prevents it from being used again.
- InvalidateSocket();
- DisableNewOutstandingCalls();
- WaitForOutstandingCalls();
- delete this;
-}
-
-bool UdpSocket2Windows::SetQos(WebRtc_Word32 serviceType,
- WebRtc_Word32 tokenRate,
- WebRtc_Word32 bucketSize,
- WebRtc_Word32 peekBandwith,
- WebRtc_Word32 minPolicedSize,
- WebRtc_Word32 maxSduSize,
- const SocketAddress &stRemName,
- WebRtc_Word32 overrideDSCP)
-{
- if(_qos == false)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "UdpSocket2Windows::SetQos(), socket not capable of QOS");
- return false;
- }
- if(overrideDSCP != 0)
- {
- FLOWSPEC f;
- WebRtc_Word32 err = CreateFlowSpec(serviceType, tokenRate, bucketSize,
- peekBandwith, minPolicedSize,
- maxSduSize, &f);
- if(err == -1)
- {
- return false;
- }
-
- SocketAddress socketName;
- struct sockaddr_in* name =
- reinterpret_cast<struct sockaddr_in*>(&socketName);
- int nameLength = sizeof(SocketAddress);
- if(AquireSocket())
- {
- getsockname(_socket, (struct sockaddr*)name, &nameLength);
- ReleaseSocket();
- }
-
- if(serviceType == 0)
- {
- // Disable TOS byte setting.
- return SetTrafficControl(0, -1, name, &f, &f) == 0;
- }
- return SetTrafficControl(overrideDSCP, -1, name, &f, &f) == 0;
- }
-
- QOS Qos;
- WebRtc_Word32 result ;
- DWORD BytesRet;
- QOS_DESTADDR QosDestaddr;
-
- memset (&Qos, QOS_NOT_SPECIFIED, sizeof(QOS));
-
- Qos.SendingFlowspec.ServiceType = serviceType;
- Qos.SendingFlowspec.TokenRate = tokenRate;
- Qos.SendingFlowspec.TokenBucketSize = QOS_NOT_SPECIFIED;
- Qos.SendingFlowspec.PeakBandwidth = QOS_NOT_SPECIFIED;
- Qos.SendingFlowspec.DelayVariation = QOS_NOT_SPECIFIED;
- Qos.SendingFlowspec.Latency = QOS_NOT_SPECIFIED;
- Qos.SendingFlowspec.MinimumPolicedSize = QOS_NOT_SPECIFIED;
- Qos.SendingFlowspec.MaxSduSize = QOS_NOT_SPECIFIED;
-
- // Only ServiceType is needed for receiving.
- Qos.ReceivingFlowspec.ServiceType = serviceType;
- Qos.ReceivingFlowspec.TokenRate = QOS_NOT_SPECIFIED;
- Qos.ReceivingFlowspec.TokenBucketSize = QOS_NOT_SPECIFIED;
- Qos.ReceivingFlowspec.PeakBandwidth = QOS_NOT_SPECIFIED;
- Qos.ReceivingFlowspec.Latency = QOS_NOT_SPECIFIED;
- Qos.ReceivingFlowspec.DelayVariation = QOS_NOT_SPECIFIED;
- Qos.ReceivingFlowspec.MinimumPolicedSize = QOS_NOT_SPECIFIED;
- Qos.ReceivingFlowspec.MaxSduSize = QOS_NOT_SPECIFIED;
-
- Qos.ProviderSpecific.len = 0;
-
- Qos.ProviderSpecific.buf = NULL;
-
- ZeroMemory((WebRtc_Word8 *)&QosDestaddr, sizeof(QosDestaddr));
-
- OSVERSIONINFOEX osvie;
- osvie.dwOSVersionInfoSize = sizeof(OSVERSIONINFOEX);
- GetVersionEx((LPOSVERSIONINFO)&osvie);
-
-// Operating system Version number dwMajorVersion dwMinorVersion
-// Windows 7 6.1 6 1
-// Windows Server 2008 R2 6.1 6 1
-// Windows Server 2008 6.0 6 0
-// Windows Vista 6.0 6 0
-// Windows Server 2003 R2 5.2 5 2
-// Windows Server 2003 5.2 5 2
-// Windows XP 5.1 5 1
-// Windows 2000 5.0 5 0
-
- // SERVICE_NO_QOS_SIGNALING and QOS_DESTADDR should not be used if version
- // is 6.0 or greater.
- if(osvie.dwMajorVersion >= 6)
- {
- Qos.SendingFlowspec.MinimumPolicedSize = QOS_NOT_SPECIFIED;
- Qos.ReceivingFlowspec.ServiceType = serviceType;
-
- } else {
- Qos.SendingFlowspec.MinimumPolicedSize =
- QOS_NOT_SPECIFIED | SERVICE_NO_QOS_SIGNALING;
- Qos.ReceivingFlowspec.ServiceType =
- serviceType | SERVICE_NO_QOS_SIGNALING;
-
- QosDestaddr.ObjectHdr.ObjectType = QOS_OBJECT_DESTADDR;
- QosDestaddr.ObjectHdr.ObjectLength = sizeof(QosDestaddr);
- QosDestaddr.SocketAddress = (SOCKADDR *)&stRemName;
- if (AF_INET6 == _iProtocol)
- {
- QosDestaddr.SocketAddressLength = sizeof(SocketAddressInVersion6);
- } else {
- QosDestaddr.SocketAddressLength = sizeof(SocketAddressIn);
- }
-
- Qos.ProviderSpecific.len = QosDestaddr.ObjectHdr.ObjectLength;
- Qos.ProviderSpecific.buf = (WebRtc_Word8*)&QosDestaddr;
- }
-
- if(AquireSocket())
- {
- // To set QoS with SIO_SET_QOS the socket must be locally bound first
- // or the call will fail with error code 10022.
- result = WSAIoctl(GetFd(), SIO_SET_QOS, &Qos, sizeof(QOS), NULL, 0,
- &BytesRet, NULL,NULL);
- ReleaseSocket();
- }
- if (result == SOCKET_ERROR)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "UdpSocket2Windows::SetQos() WSAerror : %d",
- WSAGetLastError());
- return false;
- }
- return true;
-}
-
-WebRtc_Word32 UdpSocket2Windows::SetTOS(WebRtc_Word32 serviceType)
-{
- SocketAddress socketName;
-
- struct sockaddr_in* name =
- reinterpret_cast<struct sockaddr_in*>(&socketName);
- int nameLength = sizeof(SocketAddress);
- if(AquireSocket())
- {
- getsockname(_socket, (struct sockaddr*)name, &nameLength);
- ReleaseSocket();
- }
-
- WebRtc_Word32 res = SetTrafficControl(serviceType, -1, name);
- if (res == -1)
- {
- OSVERSIONINFO OsVersion;
- OsVersion.dwOSVersionInfoSize = sizeof(OSVERSIONINFO);
- GetVersionEx (&OsVersion);
-
- if ((OsVersion.dwMajorVersion == 4)) // NT 4.0
- {
- if(SetSockopt(IPPROTO_IP,IP_TOS ,
- (WebRtc_Word8*)&serviceType, 4) != 0)
- {
- return -1;
- }
- }
- }
- return res;
-}
-
-WebRtc_Word32 UdpSocket2Windows::SetPCP(WebRtc_Word32 pcp)
-{
- SocketAddress socketName;
- struct sockaddr_in* name =
- reinterpret_cast<struct sockaddr_in*>(&socketName);
- int nameLength = sizeof(SocketAddress);
- if(AquireSocket())
- {
- getsockname(_socket, (struct sockaddr*)name, &nameLength);
- ReleaseSocket();
- }
- return SetTrafficControl(-1, pcp, name);
-}
-
-WebRtc_Word32 UdpSocket2Windows::SetTrafficControl(
- WebRtc_Word32 dscp,
- WebRtc_Word32 pcp,
- const struct sockaddr_in* name,
- FLOWSPEC* send, FLOWSPEC* recv)
-{
- if (pcp == _pcp)
- {
- // No change.
- pcp = -1;
- }
- if ((-1 == pcp) && (-1 == dscp))
- {
- return 0;
- }
- if (!_gtc)
- {
- _gtc = TrafficControlWindows::GetInstance(_id);
- }
- if (!_gtc)
- {
- return -1;
- }
- if(_filterHandle)
- {
- _gtc->TcDeleteFilter(_filterHandle);
- _filterHandle = NULL;
- }
- if(_flowHandle)
- {
- _gtc->TcDeleteFlow(_flowHandle);
- _flowHandle = NULL;
- }
- if(_clientHandle)
- {
- _gtc->TcDeregisterClient(_clientHandle);
- _clientHandle = NULL;
- }
- if ((0 == dscp) && (-2 == _pcp) && (-1 == pcp))
- {
- // TODO (pwestin): why is this not done before deleting old filter and
- // flow? This scenario should probably be documented in
- // the function declaration.
- return 0;
- }
-
- TCI_CLIENT_FUNC_LIST QoSFunctions;
- QoSFunctions.ClAddFlowCompleteHandler = NULL;
- QoSFunctions.ClDeleteFlowCompleteHandler = NULL;
- QoSFunctions.ClModifyFlowCompleteHandler = NULL;
- QoSFunctions.ClNotifyHandler = (TCI_NOTIFY_HANDLER)MyClNotifyHandler;
- // Register the client with Traffic control interface.
- HANDLE ClientHandle;
- ULONG result = _gtc->TcRegisterClient(CURRENT_TCI_VERSION, NULL,
- &QoSFunctions,&ClientHandle);
- if(result != NO_ERROR)
- {
- // This is likely caused by the application not being run as
- // administrator.
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "TcRegisterClient returned %d", result);
- return result;
- }
-
- // Find traffic control-enabled network interfaces that matches this
- // socket's IP address.
- ULONG BufferSize = 0;
- result = _gtc->TcEnumerateInterfaces(ClientHandle, &BufferSize, NULL);
-
- if(result != NO_ERROR && result != ERROR_INSUFFICIENT_BUFFER)
- {
- _gtc->TcDeregisterClient(ClientHandle);
- return result;
- }
-
- if(result != ERROR_INSUFFICIENT_BUFFER)
- {
- // Empty buffer contains all control-enabled network interfaces. I.e.
- // QoS is not enabled.
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "QOS faild since QOS is not installed on the interface");
-
- _gtc->TcDeregisterClient(ClientHandle);
- return -1;
- }
-
- PTC_IFC_DESCRIPTOR pInterfaceBuffer =
- (PTC_IFC_DESCRIPTOR)malloc(BufferSize);
- if(pInterfaceBuffer == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Out ot memory failure");
- _gtc->TcDeregisterClient(ClientHandle);
- return ERROR_NOT_ENOUGH_MEMORY;
- }
-
- result = _gtc->TcEnumerateInterfaces(ClientHandle, &BufferSize,
- pInterfaceBuffer);
-
- if(result != NO_ERROR)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "Critical: error enumerating interfaces when passing in correct\
- buffer size: %d", result);
- _gtc->TcDeregisterClient(ClientHandle);
- free(pInterfaceBuffer);
- return result;
- }
-
- PTC_IFC_DESCRIPTOR oneinterface;
- HANDLE ifcHandle, iFilterHandle, iflowHandle;
- bool addrFound = false;
- ULONG filterSourceAddress = ULONG_MAX;
-
- // Find the interface corresponding to the local address.
- for(oneinterface = pInterfaceBuffer;
- oneinterface != (PTC_IFC_DESCRIPTOR)
- (((WebRtc_Word8*)pInterfaceBuffer) + BufferSize);
- oneinterface = (PTC_IFC_DESCRIPTOR)
- ((WebRtc_Word8 *)oneinterface + oneinterface->Length))
- {
-
- WebRtc_Word8 interfaceName[500];
- WideCharToMultiByte(CP_ACP, 0, oneinterface->pInterfaceName, -1,
- interfaceName, sizeof(interfaceName), 0, 0 );
-
- PNETWORK_ADDRESS_LIST addresses =
- &(oneinterface->AddressListDesc.AddressList);
- for(LONG i = 0; i < addresses->AddressCount ; i++)
- {
- // Only look at TCP/IP addresses.
- if(addresses->Address[i].AddressType != NDIS_PROTOCOL_ID_TCP_IP)
- {
- continue;
- }
-
- NETWORK_ADDRESS_IP* pIpAddr =
- (NETWORK_ADDRESS_IP*)&(addresses->Address[i].Address);
- struct in_addr in;
- in.S_un.S_addr = pIpAddr->in_addr;
- if(pIpAddr->in_addr == name->sin_addr.S_un.S_addr)
- {
- filterSourceAddress = pIpAddr->in_addr;
- addrFound = true;
- }
- }
- if(!addrFound)
- {
- continue;
- } else
- {
- break;
- }
- }
- if(!addrFound)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "QOS faild since address is not found");
- _gtc->TcDeregisterClient(ClientHandle);
- free(pInterfaceBuffer);
- return -1;
- }
- result = _gtc->TcOpenInterfaceW(oneinterface->pInterfaceName, ClientHandle,
- NULL, &ifcHandle);
- if(result != NO_ERROR)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Error opening interface: %d", result);
- _gtc->TcDeregisterClient(ClientHandle);
- free(pInterfaceBuffer);
- return result;
- }
-
- // Create flow if one doesn't exist.
- if (!_flow)
- {
- bool addPCP = ((pcp >= 0) || ((-1 == pcp) && (_pcp >= 0)));
- int allocSize = sizeof(TC_GEN_FLOW) + sizeof(QOS_DS_CLASS) +
- (addPCP ? sizeof(QOS_TRAFFIC_CLASS) : 0);
- _flow = (PTC_GEN_FLOW)malloc(allocSize);
-
- _flow->SendingFlowspec.DelayVariation = QOS_NOT_SPECIFIED;
- _flow->SendingFlowspec.Latency = QOS_NOT_SPECIFIED;
- _flow->SendingFlowspec.MaxSduSize = QOS_NOT_SPECIFIED;
- _flow->SendingFlowspec.MinimumPolicedSize = QOS_NOT_SPECIFIED;
- _flow->SendingFlowspec.PeakBandwidth = QOS_NOT_SPECIFIED;
- _flow->SendingFlowspec.ServiceType = SERVICETYPE_BESTEFFORT;
- _flow->SendingFlowspec.TokenBucketSize = QOS_NOT_SPECIFIED;
- _flow->SendingFlowspec.TokenRate = QOS_NOT_SPECIFIED;
-
- _flow->ReceivingFlowspec.DelayVariation = QOS_NOT_SPECIFIED;
- _flow->ReceivingFlowspec.Latency = QOS_NOT_SPECIFIED;
- _flow->ReceivingFlowspec.MaxSduSize = QOS_NOT_SPECIFIED;
- _flow->ReceivingFlowspec.MinimumPolicedSize = QOS_NOT_SPECIFIED;
- _flow->ReceivingFlowspec.PeakBandwidth = QOS_NOT_SPECIFIED;
- _flow->ReceivingFlowspec.ServiceType = SERVICETYPE_BESTEFFORT;
- _flow->ReceivingFlowspec.TokenBucketSize = QOS_NOT_SPECIFIED;
- _flow->ReceivingFlowspec.TokenRate = QOS_NOT_SPECIFIED;
-
- QOS_DS_CLASS* dsClass = (QOS_DS_CLASS*)_flow->TcObjects;
- dsClass->DSField = 0;
- dsClass->ObjectHdr.ObjectType = QOS_OBJECT_DS_CLASS;
- dsClass->ObjectHdr.ObjectLength = sizeof(QOS_DS_CLASS);
-
- if (addPCP)
- {
- QOS_TRAFFIC_CLASS* trafficClass = (QOS_TRAFFIC_CLASS*)(dsClass + 1);
- trafficClass->TrafficClass = 0;
- trafficClass->ObjectHdr.ObjectType = QOS_OBJECT_TRAFFIC_CLASS;
- trafficClass->ObjectHdr.ObjectLength = sizeof(QOS_TRAFFIC_CLASS);
- }
-
- _flow->TcObjectsLength = sizeof(QOS_DS_CLASS) +
- (addPCP ? sizeof(QOS_TRAFFIC_CLASS) : 0);
- } else if (-1 != pcp) {
- // Reallocate memory since pcp has changed.
- PTC_GEN_FLOW oldFlow = _flow;
- bool addPCP = (pcp >= 0);
- int allocSize = sizeof(TC_GEN_FLOW) + sizeof(QOS_DS_CLASS) +
- (addPCP ? sizeof(QOS_TRAFFIC_CLASS) : 0);
- _flow = (PTC_GEN_FLOW)malloc(allocSize);
-
- // Copy old flow.
- _flow->ReceivingFlowspec = oldFlow->ReceivingFlowspec;
- _flow->SendingFlowspec = oldFlow->SendingFlowspec;
- // The DS info is always the first object.
- QOS_DS_CLASS* dsClass = (QOS_DS_CLASS*)_flow->TcObjects;
- QOS_DS_CLASS* oldDsClass = (QOS_DS_CLASS*)oldFlow->TcObjects;
- dsClass->DSField = oldDsClass->DSField;
- dsClass->ObjectHdr.ObjectType = oldDsClass->ObjectHdr.ObjectType;
- dsClass->ObjectHdr.ObjectLength = oldDsClass->ObjectHdr.ObjectLength;
-
- if (addPCP)
- {
- QOS_TRAFFIC_CLASS* trafficClass = (QOS_TRAFFIC_CLASS*)(dsClass + 1);
- trafficClass->TrafficClass = 0;
- trafficClass->ObjectHdr.ObjectType = QOS_OBJECT_TRAFFIC_CLASS;
- trafficClass->ObjectHdr.ObjectLength = sizeof(QOS_TRAFFIC_CLASS);
- }
-
- _flow->TcObjectsLength = sizeof(QOS_DS_CLASS) +
- (addPCP ? sizeof(QOS_TRAFFIC_CLASS) : 0);
- free(oldFlow);
- }
-
- // Setup send and receive flow and DS object.
- if (dscp >= 0)
- {
- if (!send || (0 == dscp))
- {
- _flow->SendingFlowspec.DelayVariation = QOS_NOT_SPECIFIED;
- _flow->SendingFlowspec.Latency = QOS_NOT_SPECIFIED;
- _flow->SendingFlowspec.MaxSduSize = QOS_NOT_SPECIFIED;
- _flow->SendingFlowspec.MinimumPolicedSize = QOS_NOT_SPECIFIED;
- _flow->SendingFlowspec.PeakBandwidth =
- (0 == dscp ? QOS_NOT_SPECIFIED : POSITIVE_INFINITY_RATE);
- _flow->SendingFlowspec.ServiceType = SERVICETYPE_BESTEFFORT;
- _flow->SendingFlowspec.TokenBucketSize = QOS_NOT_SPECIFIED;
- // 128000 * 10 is 10mbit/s.
- _flow->SendingFlowspec.TokenRate =
- (0 == dscp ? QOS_NOT_SPECIFIED : 128000 * 10);
- }
- else
- {
- _flow->SendingFlowspec.DelayVariation = send->DelayVariation;
- _flow->SendingFlowspec.Latency = send->Latency;
- _flow->SendingFlowspec.MaxSduSize = send->MaxSduSize;
- _flow->SendingFlowspec.MinimumPolicedSize =
- send->MinimumPolicedSize;
- _flow->SendingFlowspec.PeakBandwidth = send->PeakBandwidth;
- _flow->SendingFlowspec.PeakBandwidth = POSITIVE_INFINITY_RATE;
- _flow->SendingFlowspec.ServiceType = send->ServiceType;
- _flow->SendingFlowspec.TokenBucketSize = send->TokenBucketSize;
- _flow->SendingFlowspec.TokenRate = send->TokenRate;
- }
-
- if (!recv || (0 == dscp))
- {
- _flow->ReceivingFlowspec.DelayVariation =
- _flow->SendingFlowspec.DelayVariation;
- _flow->ReceivingFlowspec.Latency = _flow->SendingFlowspec.Latency;
- _flow->ReceivingFlowspec.MaxSduSize =
- _flow->SendingFlowspec.MaxSduSize;
- _flow->ReceivingFlowspec.MinimumPolicedSize =
- _flow->SendingFlowspec.MinimumPolicedSize;
- _flow->ReceivingFlowspec.PeakBandwidth = QOS_NOT_SPECIFIED;
- _flow->ReceivingFlowspec.ServiceType =
- 0 == dscp ? SERVICETYPE_BESTEFFORT : SERVICETYPE_CONTROLLEDLOAD;
- _flow->ReceivingFlowspec.TokenBucketSize =
- _flow->SendingFlowspec.TokenBucketSize;
- _flow->ReceivingFlowspec.TokenRate =
- _flow->SendingFlowspec.TokenRate;
- } else {
- _flow->ReceivingFlowspec.DelayVariation = recv->DelayVariation;
- _flow->ReceivingFlowspec.Latency = recv->Latency;
- _flow->ReceivingFlowspec.MaxSduSize = recv->MaxSduSize;
- _flow->ReceivingFlowspec.MinimumPolicedSize =
- recv->MinimumPolicedSize;
- _flow->ReceivingFlowspec.PeakBandwidth = recv->PeakBandwidth;
- _flow->ReceivingFlowspec.ServiceType = recv->ServiceType;
- _flow->ReceivingFlowspec.TokenBucketSize = recv->TokenBucketSize;
- _flow->ReceivingFlowspec.TokenRate = QOS_NOT_SPECIFIED;
- }
-
- // Setup DS (for DSCP value).
- // DS is always the first object.
- QOS_DS_CLASS* dsClass = (QOS_DS_CLASS*)_flow->TcObjects;
- dsClass->DSField = dscp;
- }
-
- // Setup PCP (802.1p priority in 802.1Q/VLAN tagging)
- if (pcp >= 0)
- {
- // DS is always first object.
- QOS_DS_CLASS* dsClass = (QOS_DS_CLASS*)_flow->TcObjects;
- QOS_TRAFFIC_CLASS* trafficClass = (QOS_TRAFFIC_CLASS*)(dsClass + 1);
- trafficClass->TrafficClass = pcp;
- }
-
- result = _gtc->TcAddFlow(ifcHandle, NULL, 0, _flow, &iflowHandle);
- if(result != NO_ERROR)
- {
- _gtc->TcCloseInterface(ifcHandle);
- _gtc->TcDeregisterClient(ClientHandle);
- free(pInterfaceBuffer);
- return -1;
- }
-
- IP_PATTERN filterPattern, mask;
-
- ZeroMemory((WebRtc_Word8*)&filterPattern, sizeof(IP_PATTERN));
- ZeroMemory((WebRtc_Word8*)&mask, sizeof(IP_PATTERN));
-
- filterPattern.ProtocolId = IPPROTO_UDP;
- // "name" fields already in network order.
- filterPattern.S_un.S_un_ports.s_srcport = name->sin_port;
- filterPattern.SrcAddr = filterSourceAddress;
-
- // Unsigned max of a type corresponds to a bitmask with all bits set to 1.
- // I.e. the filter should allow all ProtocolIds, any source port and any
- // IP address
- mask.ProtocolId = UCHAR_MAX;
- mask.S_un.S_un_ports.s_srcport = USHRT_MAX;
- mask.SrcAddr = ULONG_MAX;
-
- TC_GEN_FILTER filter;
-
- filter.AddressType = NDIS_PROTOCOL_ID_TCP_IP;
- filter.Mask = (LPVOID)&mask;
- filter.Pattern = (LPVOID)&filterPattern;
- filter.PatternSize = sizeof(IP_PATTERN);
-
- result = _gtc->TcAddFilter(iflowHandle, &filter, &iFilterHandle);
- if(result != NO_ERROR)
- {
- _gtc->TcDeleteFlow(iflowHandle);
- _gtc->TcCloseInterface(ifcHandle);
- _gtc->TcDeregisterClient(ClientHandle);
- free(pInterfaceBuffer);
- return result;
- }
-
- _flowHandle = iflowHandle;
- _filterHandle = iFilterHandle;
- _clientHandle = ClientHandle;
- if (-1 != pcp)
- {
- _pcp = pcp;
- }
-
- _gtc->TcCloseInterface(ifcHandle);
- free(pInterfaceBuffer);
-
- return 0;
-}
-
-WebRtc_Word32 UdpSocket2Windows::CreateFlowSpec(WebRtc_Word32 serviceType,
- WebRtc_Word32 tokenRate,
- WebRtc_Word32 bucketSize,
- WebRtc_Word32 peekBandwith,
- WebRtc_Word32 minPolicedSize,
- WebRtc_Word32 maxSduSize,
- FLOWSPEC* f)
-{
- if (!f)
- {
- return -1;
- }
-
- f->ServiceType = serviceType;
- f->TokenRate = tokenRate;
- f->TokenBucketSize = QOS_NOT_SPECIFIED;
- f->PeakBandwidth = QOS_NOT_SPECIFIED;
- f->DelayVariation = QOS_NOT_SPECIFIED;
- f->Latency = QOS_NOT_SPECIFIED;
- f->MaxSduSize = QOS_NOT_SPECIFIED;
- f->MinimumPolicedSize = QOS_NOT_SPECIFIED;
- return 0;
-}
-
-bool UdpSocket2Windows::NewOutstandingCall()
-{
- assert(_outstandingCallsDisabled.Value() == 0);
-
- const WebRtc_Word32 outstandingCalls = ++_outstandingCalls;
- return true;
-}
-
-void UdpSocket2Windows::OutstandingCallCompleted()
-{
- _ptrDestRWLock->AcquireLockShared();
- ++_outstandingCallComplete;
- if((--_outstandingCalls == 0) && (_outstandingCallsDisabled.Value() == 1))
- {
- // When there are no outstanding calls and new outstandning calls are
- // disabled it is time to terminate.
- _terminate = true;
- }
- _ptrDestRWLock->ReleaseLockShared();
-
- if((--_outstandingCallComplete == 0) &&
- (_terminate))
- {
- // Only one thread will enter here. The thread with the last outstanding
- // call.
- CriticalSectionScoped cs(_ptrDeleteCrit);
- _safeTodelete = true;
- _ptrDeleteCond->Wake();
- }
-}
-
-void UdpSocket2Windows::DisableNewOutstandingCalls()
-{
- _ptrDestRWLock->AcquireLockExclusive();
- if(_outstandingCallsDisabled.Value() == 1)
- {
- // Outstandning calls are already disabled.
- _ptrDestRWLock->ReleaseLockExclusive();
- return;
- }
- _outstandingCallsDisabled = 1;
- const bool noOutstandingCalls = (_outstandingCalls.Value() == 0);
- _ptrDestRWLock->ReleaseLockExclusive();
-
- RemoveSocketFromManager();
-
- if(noOutstandingCalls)
- {
- CriticalSectionScoped cs(_ptrDeleteCrit);
- _safeTodelete = true;
- _ptrDeleteCond->Wake();
- }
-}
-
-void UdpSocket2Windows::WaitForOutstandingCalls()
-{
- CriticalSectionScoped cs(_ptrDeleteCrit);
- while(!_safeTodelete)
- {
- _ptrDeleteCond->SleepCS(*_ptrDeleteCrit);
- }
-}
-
-void UdpSocket2Windows::RemoveSocketFromManager()
-{
- // New outstanding calls should be disabled at this point.
- assert(_outstandingCallsDisabled.Value() != 0);
-
- if(_addedToMgr)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "calling UdpSocketManager::RemoveSocket()");
- if(_mgr->RemoveSocket(this))
- {
- _addedToMgr=false;
- }
- }
-}
-
-bool UdpSocket2Windows::AquireSocket()
-{
- _ptrSocketRWLock->AcquireLockShared();
- const bool returnValue = _socket != INVALID_SOCKET;
- if(!returnValue)
- {
- _ptrSocketRWLock->ReleaseLockShared();
- }
- return returnValue;
-}
-
-void UdpSocket2Windows::ReleaseSocket()
-{
- _ptrSocketRWLock->ReleaseLockShared();
-}
-
-bool UdpSocket2Windows::InvalidateSocket()
-{
- _ptrSocketRWLock->AcquireLockExclusive();
- if(_socket == INVALID_SOCKET)
- {
- _ptrSocketRWLock->ReleaseLockExclusive();
- return true;
- }
- // Give the socket back to the system. All socket calls will fail from now
- // on.
- if(closesocket(_socket) == SOCKET_ERROR)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "UdpSocket2Windows(%d)::InvalidateSocket() WSAerror: %d",
- (WebRtc_Word32)this, WSAGetLastError());
- }
- _socket = INVALID_SOCKET;
- _ptrSocketRWLock->ReleaseLockExclusive();
- return true;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_socket2_windows.h b/trunk/src/modules/udp_transport/source/udp_socket2_windows.h
deleted file mode 100644
index c0d68bb..0000000
--- a/trunk/src/modules/udp_transport/source/udp_socket2_windows.h
+++ /dev/null
@@ -1,171 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET2_WINDOWS_H_
-#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET2_WINDOWS_H_
-
-// Disable deprication warning from traffic.h
-#pragma warning(disable : 4995)
-
-// Don't change include order for these header files.
-#include <Winsock2.h>
-#include <Ntddndis.h>
-#include <traffic.h>
-
-#include "atomic32_wrapper.h"
-#include "condition_variable_wrapper.h"
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "list_wrapper.h"
-#include "rw_lock_wrapper.h"
-#include "trace.h"
-#include "udp_socket_wrapper.h"
-#include "udp_socket2_manager_windows.h"
-
-namespace webrtc {
-class UdpSocket2ManagerWindows;
-class TrafficControlWindows;
-struct PerIoContext;
-
-class UdpSocket2Windows : public UdpSocketWrapper
-{
-public:
- UdpSocket2Windows(const WebRtc_Word32 id, UdpSocketManager* mgr,
- bool ipV6Enable = false, bool disableGQOS = false);
- virtual ~UdpSocket2Windows();
-
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- virtual bool ValidHandle();
-
- virtual bool SetCallback(CallbackObj, IncomingSocketCallback);
-
- virtual bool Bind(const SocketAddress& name);
- virtual bool SetSockopt(WebRtc_Word32 level, WebRtc_Word32 optname,
- const WebRtc_Word8* optval, WebRtc_Word32 optlen);
-
- virtual bool StartReceiving(const WebRtc_UWord32 receiveBuffers);
- virtual inline bool StartReceiving() {return StartReceiving(8);}
- virtual bool StopReceiving();
-
- virtual WebRtc_Word32 SendTo(const WebRtc_Word8* buf, WebRtc_Word32 len,
- const SocketAddress& to);
-
- virtual void CloseBlocking();
-
- virtual SOCKET GetFd() { return _socket;}
- virtual bool SetQos(WebRtc_Word32 serviceType, WebRtc_Word32 tokenRate,
- WebRtc_Word32 bucketSize, WebRtc_Word32 peekBandwith,
- WebRtc_Word32 minPolicedSize, WebRtc_Word32 maxSduSize,
- const SocketAddress &stRemName,
- WebRtc_Word32 overrideDSCP = 0);
-
- virtual WebRtc_Word32 SetTOS(const WebRtc_Word32 serviceType);
- virtual WebRtc_Word32 SetPCP(const WebRtc_Word32 pcp);
-
- virtual WebRtc_UWord32 ReceiveBuffers(){return _receiveBuffers.Value();}
-
-protected:
- void IOCompleted(PerIoContext* pIOContext, WebRtc_UWord32 ioSize,
- WebRtc_UWord32 error);
-
- WebRtc_Word32 PostRecv();
- // Use pIoContext to post a new WSARecvFrom(..).
- WebRtc_Word32 PostRecv(PerIoContext* pIoContext);
-
-private:
- friend class UdpSocket2WorkerWindows;
-
- // Set traffic control (TC) flow adding it the interface that matches this
- // sockets address.
- // A filter is created and added to the flow.
- // The flow consists of:
- // (1) QoS send and receive information (flow specifications).
- // (2) A DS object (for specifying exact DSCP value).
- // (3) Possibly a traffic object (for specifying exact 802.1p priority (PCP)
- // value).
- //
- // dscp values:
- // -1 don't change the current dscp value.
- // 0 don't add any flow to TC, unless pcp is specified.
- // 1-63 Add a flow to TC with the specified dscp value.
- // pcp values:
- // -2 Don't add pcp info to the flow, (3) will not be added.
- // -1 Don't change the current value.
- // 0-7 Add pcp info to the flow with the specified value,
- // (3) will be added.
- //
- // If both dscp and pcp are -1 no flow will be created or added to TC.
- // If dscp is 0 and pcp is 0-7 (1), (2) and (3) will be created.
- // Note: input parameter values are assumed to be in valid range, checks
- // must be done by caller.
- WebRtc_Word32 SetTrafficControl(WebRtc_Word32 dscp, WebRtc_Word32 pcp,
- const struct sockaddr_in* name,
- FLOWSPEC* send = NULL,
- FLOWSPEC* recv = NULL);
- WebRtc_Word32 CreateFlowSpec(WebRtc_Word32 serviceType,
- WebRtc_Word32 tokenRate,
- WebRtc_Word32 bucketSize,
- WebRtc_Word32 peekBandwith,
- WebRtc_Word32 minPolicedSize,
- WebRtc_Word32 maxSduSize, FLOWSPEC *f);
-
- WebRtc_Word32 _id;
- RWLockWrapper* _ptrCbRWLock;
- IncomingSocketCallback _incomingCb;
- CallbackObj _obj;
- bool _qos;
-
- SocketAddress _remoteAddr;
- SOCKET _socket;
- WebRtc_Word32 _iProtocol;
- UdpSocket2ManagerWindows* _mgr;
-
- CriticalSectionWrapper* _pCrit;
- Atomic32Wrapper _outstandingCalls;
- Atomic32Wrapper _outstandingCallComplete;
- volatile bool _terminate;
- volatile bool _addedToMgr;
-
- CriticalSectionWrapper* _ptrDeleteCrit;
- ConditionVariableWrapper* _ptrDeleteCond;
- bool _safeTodelete;
-
- RWLockWrapper* _ptrDestRWLock;
- Atomic32Wrapper _outstandingCallsDisabled; // 0 = false, 1 = true
- bool NewOutstandingCall();
- void OutstandingCallCompleted();
- void DisableNewOutstandingCalls();
- void WaitForOutstandingCalls();
-
- void RemoveSocketFromManager();
-
- // RWLockWrapper is used as a reference counter for the socket. Write lock
- // is used for creating and deleting socket. Read lock is used for
- // accessing the socket.
- RWLockWrapper* _ptrSocketRWLock;
- bool AquireSocket();
- void ReleaseSocket();
- bool InvalidateSocket();
-
- // Traffic control handles and structure pointers.
- HANDLE _clientHandle;
- HANDLE _flowHandle;
- HANDLE _filterHandle;
- PTC_GEN_FLOW _flow;
- // TrafficControlWindows implements TOS and PCP.
- TrafficControlWindows* _gtc;
- // Holds the current pcp value. Can be -2 or 0 - 7.
- int _pcp;
-
- Atomic32Wrapper _receiveBuffers;
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET2_WINDOWS_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_manager_posix.cc b/trunk/src/modules/udp_transport/source/udp_socket_manager_posix.cc
deleted file mode 100644
index 889fb2f..0000000
--- a/trunk/src/modules/udp_transport/source/udp_socket_manager_posix.cc
+++ /dev/null
@@ -1,429 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "udp_socket_manager_posix.h"
-
-#include <strings.h>
-#include <sys/time.h>
-#include <sys/types.h>
-#include <time.h>
-#include <unistd.h>
-
-#include "trace.h"
-#include "udp_socket_posix.h"
-
-namespace webrtc {
-UdpSocketManagerPosix::UdpSocketManagerPosix()
- : UdpSocketManager(),
- _id(-1),
- _critSect(CriticalSectionWrapper::CreateCriticalSection()),
- _numberOfSocketMgr(-1),
- _incSocketMgrNextTime(0),
- _nextSocketMgrToAssign(0),
- _socketMgr()
-{
-}
-
-bool UdpSocketManagerPosix::Init(WebRtc_Word32 id,
- WebRtc_UWord8& numOfWorkThreads) {
- CriticalSectionScoped cs(_critSect);
- if ((_id != -1) || (_numOfWorkThreads != 0)) {
- assert(_id != -1);
- assert(_numOfWorkThreads != 0);
- return false;
- }
-
- _id = id;
- _numberOfSocketMgr = numOfWorkThreads;
- _numOfWorkThreads = numOfWorkThreads;
-
- if(MAX_NUMBER_OF_SOCKET_MANAGERS_LINUX < _numberOfSocketMgr)
- {
- _numberOfSocketMgr = MAX_NUMBER_OF_SOCKET_MANAGERS_LINUX;
- }
- for(int i = 0;i < _numberOfSocketMgr; i++)
- {
- _socketMgr[i] = new UdpSocketManagerPosixImpl();
- }
- return true;
-}
-
-
-UdpSocketManagerPosix::~UdpSocketManagerPosix()
-{
- Stop();
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocketManagerPosix(%d)::UdpSocketManagerPosix()",
- _numberOfSocketMgr);
-
- for(int i = 0;i < _numberOfSocketMgr; i++)
- {
- delete _socketMgr[i];
- }
- delete _critSect;
-}
-
-WebRtc_Word32 UdpSocketManagerPosix::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
- return 0;
-}
-
-bool UdpSocketManagerPosix::Start()
-{
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocketManagerPosix(%d)::Start()",
- _numberOfSocketMgr);
-
- _critSect->Enter();
- bool retVal = true;
- for(int i = 0;i < _numberOfSocketMgr && retVal; i++)
- {
- retVal = _socketMgr[i]->Start();
- }
- if(!retVal)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocketManagerPosix(%d)::Start() error starting socket managers",
- _numberOfSocketMgr);
- }
- _critSect->Leave();
- return retVal;
-}
-
-bool UdpSocketManagerPosix::Stop()
-{
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocketManagerPosix(%d)::Stop()",_numberOfSocketMgr);
-
- _critSect->Enter();
- bool retVal = true;
- for(int i = 0; i < _numberOfSocketMgr && retVal; i++)
- {
- retVal = _socketMgr[i]->Stop();
- }
- if(!retVal)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocketManagerPosix(%d)::Stop() there are still active socket "
- "managers",
- _numberOfSocketMgr);
- }
- _critSect->Leave();
- return retVal;
-}
-
-bool UdpSocketManagerPosix::AddSocket(UdpSocketWrapper* s)
-{
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocketManagerPosix(%d)::AddSocket()",_numberOfSocketMgr);
-
- _critSect->Enter();
- bool retVal = _socketMgr[_nextSocketMgrToAssign]->AddSocket(s);
- if(!retVal)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocketManagerPosix(%d)::AddSocket() failed to add socket to\
- manager",
- _numberOfSocketMgr);
- }
-
- // Distribute sockets on UdpSocketManagerPosixImpls in a round-robin
- // fashion.
- if(_incSocketMgrNextTime == 0)
- {
- _incSocketMgrNextTime++;
- } else {
- _incSocketMgrNextTime = 0;
- _nextSocketMgrToAssign++;
- if(_nextSocketMgrToAssign >= _numberOfSocketMgr)
- {
- _nextSocketMgrToAssign = 0;
- }
- }
- _critSect->Leave();
- return retVal;
-}
-
-bool UdpSocketManagerPosix::RemoveSocket(UdpSocketWrapper* s)
-{
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocketManagerPosix(%d)::RemoveSocket()",
- _numberOfSocketMgr);
-
- _critSect->Enter();
- bool retVal = false;
- for(int i = 0;i < _numberOfSocketMgr && (retVal == false); i++)
- {
- retVal = _socketMgr[i]->RemoveSocket(s);
- }
- if(!retVal)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpSocketManagerPosix(%d)::RemoveSocket() failed to remove socket\
- from manager",
- _numberOfSocketMgr);
- }
- _critSect->Leave();
- return retVal;
-}
-
-
-UdpSocketManagerPosixImpl::UdpSocketManagerPosixImpl()
-{
- _critSectList = CriticalSectionWrapper::CreateCriticalSection();
- _thread = ThreadWrapper::CreateThread(UdpSocketManagerPosixImpl::Run, this,
- kRealtimePriority,
- "UdpSocketManagerPosixImplThread");
- FD_ZERO(&_readFds);
- WEBRTC_TRACE(kTraceMemory, kTraceTransport, -1,
- "UdpSocketManagerPosix created");
-}
-
-UdpSocketManagerPosixImpl::~UdpSocketManagerPosixImpl()
-{
- if(_thread != NULL)
- {
- delete _thread;
- }
-
- if (_critSectList != NULL)
- {
- UpdateSocketMap();
-
- _critSectList->Enter();
-
- MapItem* item = _socketMap.First();
- while(item)
- {
- UdpSocketPosix* s = static_cast<UdpSocketPosix*>(item->GetItem());
- _socketMap.Erase(item);
- item = _socketMap.First();
- delete s;
- }
- _critSectList->Leave();
-
- delete _critSectList;
- }
-
- WEBRTC_TRACE(kTraceMemory, kTraceTransport, -1,
- "UdpSocketManagerPosix deleted");
-}
-
-bool UdpSocketManagerPosixImpl::Start()
-{
- unsigned int id = 0;
- if (_thread == NULL)
- {
- return false;
- }
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, -1,
- "Start UdpSocketManagerPosix");
- return _thread->Start(id);
-}
-
-bool UdpSocketManagerPosixImpl::Stop()
-{
- if (_thread == NULL)
- {
- return true;
- }
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, -1,
- "Stop UdpSocketManagerPosix");
- return _thread->Stop();
-}
-
-bool UdpSocketManagerPosixImpl::Process()
-{
- bool doSelect = false;
- // Timeout = 1 second.
- struct timeval timeout;
- timeout.tv_sec = 0;
- timeout.tv_usec = 10000;
- MapItem* it;
-
- FD_ZERO(&_readFds);
-
- UpdateSocketMap();
-
- unsigned int maxFd = 0;
- for (it = _socketMap.First(); it != NULL; it=_socketMap.Next(it))
- {
- doSelect = true;
- maxFd = maxFd > it->GetUnsignedId() ? maxFd : it->GetUnsignedId();
- FD_SET(it->GetUnsignedId(), &_readFds);
-
- maxFd = maxFd > it->GetUnsignedId() ? maxFd : it->GetUnsignedId();
- doSelect = true;
- }
-
- int num = 0;
- if (doSelect)
- {
- num = select(maxFd+1, &_readFds, NULL, NULL, &timeout);
-
- if (num == SOCKET_ERROR)
- {
- // Timeout = 10 ms.
- timespec t;
- t.tv_sec = 0;
- t.tv_nsec = 10000*1000;
- nanosleep(&t, NULL);
- return true;
- }
- }else
- {
- // Timeout = 10 ms.
- timespec t;
- t.tv_sec = 0;
- t.tv_nsec = 10000*1000;
- nanosleep(&t, NULL);
- return true;
- }
-
- for (it = _socketMap.First(); it != NULL && num > 0;
- it = _socketMap.Next(it))
- {
- UdpSocketPosix* s = static_cast<UdpSocketPosix*>(it->GetItem());
- if (FD_ISSET(it->GetUnsignedId(), &_readFds))
- {
- s->HasIncoming();
- num--;
- }
- }
- return true;
-}
-
-bool UdpSocketManagerPosixImpl::Run(ThreadObj obj)
-{
- UdpSocketManagerPosixImpl* mgr =
- static_cast<UdpSocketManagerPosixImpl*>(obj);
- return mgr->Process();
-}
-
-bool UdpSocketManagerPosixImpl::AddSocket(UdpSocketWrapper* s)
-{
- UdpSocketPosix* sl = static_cast<UdpSocketPosix*>(s);
- if(sl->GetFd() == INVALID_SOCKET || !(sl->GetFd() < FD_SETSIZE))
- {
- return false;
- }
- _critSectList->Enter();
- _addList.PushBack(s);
- _critSectList->Leave();
- return true;
-}
-
-bool UdpSocketManagerPosixImpl::RemoveSocket(UdpSocketWrapper* s)
-{
- // Put in remove list if this is the correct UdpSocketManagerPosixImpl.
- _critSectList->Enter();
-
- // If the socket is in the add list it's safe to remove and delete it.
- ListItem* addListItem = _addList.First();
- while(addListItem)
- {
- UdpSocketPosix* addSocket = (UdpSocketPosix*)addListItem->GetItem();
- unsigned int addFD = addSocket->GetFd();
- unsigned int removeFD = static_cast<UdpSocketPosix*>(s)->GetFd();
- if(removeFD == addFD)
- {
- _removeList.PushBack(removeFD);
- _critSectList->Leave();
- return true;
- }
- addListItem = _addList.Next(addListItem);
- }
-
- // Checking the socket map is safe since all Erase and Insert calls to this
- // map are also protected by _critSectList.
- if(_socketMap.Find(static_cast<UdpSocketPosix*>(s)->GetFd()) != NULL)
- {
- _removeList.PushBack(static_cast<UdpSocketPosix*>(s)->GetFd());
- _critSectList->Leave();
- return true;
- }
- _critSectList->Leave();
- return false;
-}
-
-void UdpSocketManagerPosixImpl::UpdateSocketMap()
-{
- // Remove items in remove list.
- _critSectList->Enter();
- while(!_removeList.Empty())
- {
- UdpSocketPosix* deleteSocket = NULL;
- unsigned int removeFD = _removeList.First()->GetUnsignedItem();
-
- // If the socket is in the add list it hasn't been added to the socket
- // map yet. Just remove the socket from the add list.
- ListItem* addListItem = _addList.First();
- while(addListItem)
- {
- UdpSocketPosix* addSocket = (UdpSocketPosix*)addListItem->GetItem();
- unsigned int addFD = addSocket->GetFd();
- if(removeFD == addFD)
- {
- deleteSocket = addSocket;
- _addList.Erase(addListItem);
- break;
- }
- addListItem = _addList.Next(addListItem);
- }
-
- // Find and remove socket from _socketMap.
- MapItem* it = _socketMap.Find(removeFD);
- if(it != NULL)
- {
- UdpSocketPosix* socket =
- static_cast<UdpSocketPosix*>(it->GetItem());
- if(socket)
- {
- deleteSocket = socket;
- }
- _socketMap.Erase(it);
- }
- if(deleteSocket)
- {
- deleteSocket->ReadyForDeletion();
- delete deleteSocket;
- }
- _removeList.PopFront();
- }
-
- // Add sockets from add list.
- while(!_addList.Empty())
- {
- UdpSocketPosix* s =
- static_cast<UdpSocketPosix*>(_addList.First()->GetItem());
- if(s)
- {
- _socketMap.Insert(s->GetFd(), s);
- }
- _addList.PopFront();
- }
- _critSectList->Leave();
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_manager_posix.h b/trunk/src/modules/udp_transport/source/udp_socket_manager_posix.h
deleted file mode 100644
index c89aa13..0000000
--- a/trunk/src/modules/udp_transport/source/udp_socket_manager_posix.h
+++ /dev/null
@@ -1,85 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_POSIX_H_
-#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_POSIX_H_
-
-#include <sys/types.h>
-#include <unistd.h>
-
-#include "critical_section_wrapper.h"
-#include "list_wrapper.h"
-#include "map_wrapper.h"
-#include "thread_wrapper.h"
-#include "udp_socket_manager_wrapper.h"
-#include "udp_socket_wrapper.h"
-
-#define MAX_NUMBER_OF_SOCKET_MANAGERS_LINUX 8
-
-namespace webrtc {
-
-class ConditionVariableWrapper;
-class UdpSocketManagerPosixImpl;
-
-class UdpSocketManagerPosix : public UdpSocketManager
-{
-public:
- UdpSocketManagerPosix();
- virtual ~UdpSocketManagerPosix();
-
- virtual bool Init(WebRtc_Word32 id,
- WebRtc_UWord8& numOfWorkThreads);
-
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- virtual bool Start();
- virtual bool Stop();
-
- virtual bool AddSocket(UdpSocketWrapper* s);
- virtual bool RemoveSocket(UdpSocketWrapper* s);
-private:
- WebRtc_Word32 _id;
- CriticalSectionWrapper* _critSect;
- WebRtc_UWord8 _numberOfSocketMgr;
- WebRtc_UWord8 _incSocketMgrNextTime;
- WebRtc_UWord8 _nextSocketMgrToAssign;
- UdpSocketManagerPosixImpl* _socketMgr[MAX_NUMBER_OF_SOCKET_MANAGERS_LINUX];
-};
-
-class UdpSocketManagerPosixImpl
-{
-public:
- UdpSocketManagerPosixImpl();
- virtual ~UdpSocketManagerPosixImpl();
-
- virtual bool Start();
- virtual bool Stop();
-
- virtual bool AddSocket(UdpSocketWrapper* s);
- virtual bool RemoveSocket(UdpSocketWrapper* s);
-
-protected:
- static bool Run(ThreadObj obj);
- bool Process();
- void UpdateSocketMap();
-
-private:
- ThreadWrapper* _thread;
- CriticalSectionWrapper* _critSectList;
-
- fd_set _readFds;
-
- MapWrapper _socketMap;
- ListWrapper _addList;
- ListWrapper _removeList;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_POSIX_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_manager_windows.cc b/trunk/src/modules/udp_transport/source/udp_socket_manager_windows.cc
deleted file mode 100644
index d817d74..0000000
--- a/trunk/src/modules/udp_transport/source/udp_socket_manager_windows.cc
+++ /dev/null
@@ -1,283 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "udp_socket_manager_windows.h"
-#include "udp_socket_windows.h"
-
-namespace webrtc {
-WebRtc_UWord32 UdpSocketManagerWindows::_numOfActiveManagers = 0;
-
-UdpSocketManagerWindows::UdpSocketManagerWindows()
- : UdpSocketManager(),
- _id(-1)
-{
- const WebRtc_Word8* threadName = "UdpSocketManagerWindows_Thread";
- _critSectList = CriticalSectionWrapper::CreateCriticalSection();
- _thread = ThreadWrapper::CreateThread(UdpSocketManagerWindows::Run,
- this, kRealtimePriority, threadName);
- FD_ZERO(&_readFds);
- FD_ZERO(&_writeFds);
- FD_ZERO(&_exceptFds);
- _numOfActiveManagers++;
-}
-
-bool UdpSocketManagerWindows::Init(WebRtc_Word32 id,
- WebRtc_UWord8& numOfWorkThreads) {
- CriticalSectionScoped cs(_critSectList);
- if ((_id != -1) || (_numOfWorkThreads != 0)) {
- assert(_id == -1);
- assert(_numOfWorkThreads == 0);
- return false;
- }
- _id = id;
- _numOfWorkThreads = numOfWorkThreads;
- return true;
-}
-
-UdpSocketManagerWindows::~UdpSocketManagerWindows()
-{
- Stop();
- if(_thread != NULL)
- {
- delete _thread;
- }
-
- if (_critSectList != NULL)
- {
- _critSectList->Enter();
-
- while(!_socketMap.empty())
- {
- std::map<SOCKET, UdpSocketWindows*>::iterator it =
- _socketMap.begin();
- UdpSocketWindows* s = static_cast<UdpSocketWindows*>(it->second);
- _socketMap.erase(it);
- delete s;
- }
- _removeList.erase(_removeList.begin(), _removeList.end());
-
- while(!_addList.empty())
- {
- std::list<UdpSocketWindows*>::iterator it = _addList.begin();
- UdpSocketWindows* s = static_cast<UdpSocketWindows*>(*it);
- _addList.erase(it);
- delete s;
- }
- _critSectList->Leave();
-
- delete _critSectList;
- }
-
- _numOfActiveManagers--;
-
- if (_numOfActiveManagers == 0)
- WSACleanup();
-}
-
-WebRtc_Word32 UdpSocketManagerWindows::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
- return 0;
-}
-
-bool UdpSocketManagerWindows::Start()
-{
- unsigned int id;
- if (_thread == NULL)
- return false;
-
- return _thread->Start(id);
-}
-
-bool UdpSocketManagerWindows::Stop()
-{
- if (_thread == NULL)
- return true;
-
- return _thread->Stop();
-}
-
-bool UdpSocketManagerWindows::Process()
-{
- bool doSelect = false;
- // Timeout = 1 second.
- timeval timeout;
- timeout.tv_sec = 0;
- timeout.tv_usec = 10000;
-
- FD_ZERO(&_readFds);
- FD_ZERO(&_writeFds);
- FD_ZERO(&_exceptFds);
-
- _critSectList->Enter();
- // Remove sockets that have been registered for removal.
- while(!_removeList.empty())
- {
- SOCKET id = *_removeList.begin();
- std::map<SOCKET, UdpSocketWindows*>::iterator it = _socketMap.find(id);
- if(it != _socketMap.end())
- {
- UdpSocketWindows* s = static_cast<UdpSocketWindows*>(it->second);
- _socketMap.erase(it);
- _removeList.pop_front();
- delete s;
- }
- }
-
- // Add sockets that have been registered for being added.
- while (!_addList.empty())
- {
- UdpSocketWindows* s = *_addList.begin();
- if(s)
- {
- _socketMap[s->GetFd()] = s;
- }
- _addList.pop_front();
- }
- _critSectList->Leave();
-
- std::map<SOCKET, UdpSocketWindows*>::iterator it = _socketMap.begin();
- while(it != _socketMap.end())
- {
- UdpSocketWindows* s = it->second;
- if (s->WantsIncoming())
- {
- doSelect = true;
- FD_SET(it->first, &_readFds);
- }
- if(!s->IsWritable())
- {
- FD_SET(it->first, &_writeFds);
- doSelect = true;
- }
- it++;
- }
-
- WebRtc_Word32 num = 0;
- if (doSelect)
- {
- num = select(0, &_readFds, &_writeFds, &_exceptFds, &timeout);
- if (num == SOCKET_ERROR)
- {
- Sleep(10);
- return true;
- }
- }else
- {
- Sleep(10);
- return true;
- }
-
- it = _socketMap.begin();
- while (it != _socketMap.end() && num > 0)
- {
- if (FD_ISSET(it->first, &_readFds))
- {
- static_cast<UdpSocketWindows*>(it->second)->HasIncoming();
- num--;
- }
- if (FD_ISSET(it->first, &_writeFds))
- {
- // Socket available for writing.
- static_cast<UdpSocketWindows*>(it->second)->SetWritable();
- num--;
- }
- }
- return true;
-}
-
-bool UdpSocketManagerWindows::Run(ThreadObj obj)
-{
- UdpSocketManagerWindows* mgr = static_cast<UdpSocketManagerWindows*>(obj);
- return mgr->Process();
-};
-
-bool UdpSocketManagerWindows::AddSocket(UdpSocketWrapper* s)
-{
- UdpSocketWindows* winSock = static_cast<UdpSocketWindows*>(s);
-
- _critSectList->Enter();
- std::map<SOCKET, UdpSocketWindows*>::iterator it =
- _socketMap.find(winSock->GetFd());
- if (it != _socketMap.end())
- {
- if (!_removeList.empty())
- {
- // File descriptors are re-used so it's possible that a socket has
- // been added with the same file descriptor as a socket that is to
- // be removed. I.e. the socket that is to be removed is no longer
- // in use, delete it.
- // TODO (hellner): removing items from _socketMap may cause race
- // condition. Fix this.
- std::list<SOCKET>::iterator removeIt = _removeList.begin();
- while(removeIt != _removeList.end())
- {
- if (*removeIt == winSock->GetFd())
- {
- it = _socketMap.find(*removeIt);
- UdpSocketWindows* delete_socket = it->second;
- _socketMap.erase(it);
- _removeList.erase(removeIt);
- delete delete_socket;
- _addList.push_back(winSock);
- _critSectList->Leave();
- return true;
- }
- removeIt++;
- }
- }
- _critSectList->Leave();
- return false;
- }
-
- _addList.push_back(winSock);
- _critSectList->Leave();
- return true;
-}
-
-bool UdpSocketManagerWindows::RemoveSocket(UdpSocketWrapper* s)
-{
- UdpSocketWindows* winSock = static_cast<UdpSocketWindows*>(s);
-
- _critSectList->Enter();
- // If socket is in the add list its safe to just remove it from the list.
- if (!_addList.empty())
- {
- std::list<UdpSocketWindows*>::iterator it = _addList.begin();
- while(it != _addList.end())
- {
- UdpSocketWindows* tempSocket = (*it);
- if (tempSocket->GetFd() == winSock->GetFd())
- {
- _addList.erase(it);
- delete winSock;
- _critSectList->Leave();
- return true;
- }
- it++;
- }
- }
-
- // If the socket is not even added to the UdpSocketManagerWindows it's
- // safe to delete the socket.
- std::map<SOCKET, UdpSocketWindows*>::iterator findIt =
- _socketMap.find(winSock->GetFd());
- if (findIt == _socketMap.end())
- {
- delete winSock;
- _critSectList->Leave();
- return false;
- }
-
- _removeList.push_back(winSock->GetFd());
- _critSectList->Leave();
- return true;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_manager_windows.h b/trunk/src/modules/udp_transport/source/udp_socket_manager_windows.h
deleted file mode 100644
index 06d905a..0000000
--- a/trunk/src/modules/udp_transport/source/udp_socket_manager_windows.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_WINDOWS_H_
-#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_WINDOWS_H_
-
-#define FD_SETSIZE 1024
-
-#include <winsock2.h>
-#include <map>
-#include <list>
-
-// Don't change the include order.
-// TODO (hellner): all header files should be compilable separately. Fix this.
-#include "udp_socket_manager_wrapper.h"
-#include "thread_wrapper.h"
-#include "critical_section_wrapper.h"
-
-namespace webrtc {
-class UdpSocketWindows;
-
-class UdpSocketManagerWindows : public UdpSocketManager
-{
-public:
- UdpSocketManagerWindows();
- virtual ~UdpSocketManagerWindows();
-
- virtual bool Init(WebRtc_Word32 id,
- WebRtc_UWord8& numOfWorkThreads);
-
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- virtual bool Start();
- virtual bool Stop();
-
- virtual bool AddSocket(UdpSocketWrapper* s);
- virtual bool RemoveSocket(UdpSocketWrapper* s);
-
-protected:
- static bool Run(ThreadObj obj);
- bool Process();
-
-private:
- WebRtc_Word32 _id;
- ThreadWrapper* _thread;
-
- fd_set _readFds;
- fd_set _writeFds;
- fd_set _exceptFds;
-
- CriticalSectionWrapper* _critSectList;
-
- std::map<SOCKET, UdpSocketWindows*> _socketMap;
- std::list<UdpSocketWindows*> _addList;
- std::list<SOCKET> _removeList;
-
- static WebRtc_UWord32 _numOfActiveManagers;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_WINDOWS_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_manager_wrapper.cc b/trunk/src/modules/udp_transport/source/udp_socket_manager_wrapper.cc
deleted file mode 100644
index 9f1162e..0000000
--- a/trunk/src/modules/udp_transport/source/udp_socket_manager_wrapper.cc
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "udp_socket_manager_wrapper.h"
-
-#include <cassert>
-
-#ifdef _WIN32
-#include "fix_interlocked_exchange_pointer_win.h"
-#include "udp_socket_manager_windows.h"
-#include "udp_socket2_manager_windows.h"
-#else
-#include "udp_socket_manager_posix.h"
-#endif
-
-namespace webrtc {
-UdpSocketManager* UdpSocketManager::CreateInstance()
-{
-#if defined(_WIN32)
- #if (defined(USE_WINSOCK2))
- return static_cast<UdpSocketManager*>(
- new UdpSocket2ManagerWindows());
- #else
- numOfWorkThreads = 1;
- return static_cast<UdpSocketManager*>(
- new UdpSocketManagerWindows());
- #endif
-#else
- return new UdpSocketManagerPosix();
-#endif
-}
-
-UdpSocketManager* UdpSocketManager::StaticInstance(
- CountOperation count_operation,
- const WebRtc_Word32 id,
- WebRtc_UWord8& numOfWorkThreads)
-{
- UdpSocketManager* impl =
- GetStaticInstance<UdpSocketManager>(count_operation);
- if (count_operation == kAddRef && impl != NULL) {
- if (impl->Init(id, numOfWorkThreads)) {
- impl->Start();
- }
- }
- return impl;
-}
-
-UdpSocketManager* UdpSocketManager::Create(const WebRtc_Word32 id,
- WebRtc_UWord8& numOfWorkThreads)
-{
- return UdpSocketManager::StaticInstance(kAddRef, id,
- numOfWorkThreads);
-}
-
-void UdpSocketManager::Return()
-{
- WebRtc_UWord8 numOfWorkThreads = 0;
- UdpSocketManager::StaticInstance(kRelease, -1,
- numOfWorkThreads);
-}
-
-UdpSocketManager::UdpSocketManager() : _numOfWorkThreads(0)
-{
-}
-
-WebRtc_UWord8 UdpSocketManager::WorkThreads() const
-{
- return _numOfWorkThreads;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_manager_wrapper.h b/trunk/src/modules/udp_transport/source/udp_socket_manager_wrapper.h
deleted file mode 100644
index e7bd09e..0000000
--- a/trunk/src/modules/udp_transport/source/udp_socket_manager_wrapper.h
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_WRAPPER_H_
-#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_WRAPPER_H_
-
-#include "system_wrappers/interface/static_instance.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-class UdpSocketWrapper;
-
-class UdpSocketManager
-{
-public:
- static UdpSocketManager* Create(const WebRtc_Word32 id,
- WebRtc_UWord8& numOfWorkThreads);
- static void Return();
-
- // Initializes the socket manager. Returns true if the manager wasn't
- // already initialized.
- virtual bool Init(WebRtc_Word32 id,
- WebRtc_UWord8& numOfWorkThreads) = 0;
-
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0;
-
- // Start listening to sockets that have been registered via the
- // AddSocket(..) API.
- virtual bool Start() = 0;
- // Stop listening to sockets.
- virtual bool Stop() = 0;
-
- virtual WebRtc_UWord8 WorkThreads() const;
-
- // Register a socket with the socket manager.
- virtual bool AddSocket(UdpSocketWrapper* s) = 0;
- // Unregister a socket from the manager.
- virtual bool RemoveSocket(UdpSocketWrapper* s) = 0;
-
-protected:
- UdpSocketManager();
- virtual ~UdpSocketManager() {}
-
- WebRtc_UWord8 _numOfWorkThreads;
-
- // Factory method.
- static UdpSocketManager* CreateInstance();
-
-private:
- // Friend function to allow the UDP destructor to be accessed from the
- // instance template.
- friend UdpSocketManager*
- GetStaticInstance<UdpSocketManager>(CountOperation count_operation);
-
- static UdpSocketManager* StaticInstance(
- CountOperation count_operation,
- const WebRtc_Word32 id,
- WebRtc_UWord8& numOfWorkThreads);
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_MANAGER_WRAPPER_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_posix.cc b/trunk/src/modules/udp_transport/source/udp_socket_posix.cc
deleted file mode 100644
index 9d3564d..0000000
--- a/trunk/src/modules/udp_transport/source/udp_socket_posix.cc
+++ /dev/null
@@ -1,275 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "udp_socket_posix.h"
-
-#include <errno.h>
-#include <fcntl.h>
-#include <netdb.h>
-#include <string.h>
-#include <sys/ioctl.h>
-#include <sys/types.h>
-#include <time.h>
-#include <unistd.h>
-
-#include "trace.h"
-#include "udp_socket_manager_wrapper.h"
-#include "udp_socket_wrapper.h"
-
-namespace webrtc {
-UdpSocketPosix::UdpSocketPosix(const WebRtc_Word32 id, UdpSocketManager* mgr,
- bool ipV6Enable)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceTransport, id,
- "UdpSocketPosix::UdpSocketPosix()");
-
- _wantsIncoming = false;
- _error = 0;
- _mgr = mgr;
-
- _id = id;
- _obj = NULL;
- _incomingCb = NULL;
- _readyForDeletionCond = ConditionVariableWrapper::CreateConditionVariable();
- _closeBlockingCompletedCond =
- ConditionVariableWrapper::CreateConditionVariable();
- _cs = CriticalSectionWrapper::CreateCriticalSection();
- _readyForDeletion = false;
- _closeBlockingActive = false;
- _closeBlockingCompleted= false;
- if(ipV6Enable)
- {
- _socket = socket(AF_INET6, SOCK_DGRAM, IPPROTO_UDP);
- }
- else {
- _socket = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
- }
-
- // Set socket to nonblocking mode.
- int enable_non_blocking = 1;
- if(ioctl(_socket, FIONBIO, &enable_non_blocking) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, id,
- "Failed to make socket nonblocking");
- }
- // Enable close on fork for file descriptor so that it will not block until
- // forked process terminates.
- if(fcntl(_socket, F_SETFD, FD_CLOEXEC) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, id,
- "Failed to set FD_CLOEXEC for socket");
- }
-}
-
-UdpSocketPosix::~UdpSocketPosix()
-{
- if(_socket != INVALID_SOCKET)
- {
- close(_socket);
- _socket = INVALID_SOCKET;
- }
- if(_readyForDeletionCond)
- {
- delete _readyForDeletionCond;
- }
-
- if(_closeBlockingCompletedCond)
- {
- delete _closeBlockingCompletedCond;
- }
-
- if(_cs)
- {
- delete _cs;
- }
-}
-
-WebRtc_Word32 UdpSocketPosix::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
- return 0;
-}
-
-bool UdpSocketPosix::SetCallback(CallbackObj obj, IncomingSocketCallback cb)
-{
- _obj = obj;
- _incomingCb = cb;
-
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocketPosix(%p)::SetCallback", this);
-
- if (_mgr->AddSocket(this))
- {
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocketPosix(%p)::SetCallback socket added to manager",
- this);
- return true; // socket is now ready for action
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocketPosix(%p)::SetCallback error adding me to mgr",
- this);
- return false;
-}
-
-bool UdpSocketPosix::SetSockopt(WebRtc_Word32 level, WebRtc_Word32 optname,
- const WebRtc_Word8* optval, WebRtc_Word32 optlen)
-{
- if(0 == setsockopt(_socket, level, optname, optval, optlen ))
- {
- return true;
- }
-
- _error = errno;
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "UdpSocketPosix::SetSockopt(), error:%d", _error);
- return false;
-}
-
-WebRtc_Word32 UdpSocketPosix::SetTOS(WebRtc_Word32 serviceType)
-{
- if (SetSockopt(IPPROTO_IP, IP_TOS ,(WebRtc_Word8*)&serviceType ,4) != 0)
- {
- return -1;
- }
- return 0;
-}
-
-bool UdpSocketPosix::Bind(const SocketAddress& name)
-{
- int size = sizeof(sockaddr);
- if (0 == bind(_socket, reinterpret_cast<const sockaddr*>(&name),size))
- {
- return true;
- }
- _error = errno;
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "UdpSocketPosix::Bind() error: %d",_error);
- return false;
-}
-
-WebRtc_Word32 UdpSocketPosix::SendTo(const WebRtc_Word8* buf, WebRtc_Word32 len,
- const SocketAddress& to)
-{
- int size = sizeof(sockaddr);
- int retVal = sendto(_socket,buf, len, 0,
- reinterpret_cast<const sockaddr*>(&to), size);
- if(retVal == SOCKET_ERROR)
- {
- _error = errno;
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "UdpSocketPosix::SendTo() error: %d", _error);
- }
-
- return retVal;
-}
-
-bool UdpSocketPosix::ValidHandle()
-{
- return _socket != INVALID_SOCKET;
-}
-
-void UdpSocketPosix::HasIncoming()
-{
- char buf[2048];
- int retval;
- SocketAddress from;
-#if defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_MAC)
- sockaddr sockaddrfrom;
- memset(&from, 0, sizeof(from));
- memset(&sockaddrfrom, 0, sizeof(sockaddrfrom));
- socklen_t fromlen = sizeof(sockaddrfrom);
-#else
- memset(&from, 0, sizeof(from));
- socklen_t fromlen = sizeof(from);
-#endif
-
-#if defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_MAC)
- retval = recvfrom(_socket,buf, sizeof(buf), 0,
- reinterpret_cast<sockaddr*>(&sockaddrfrom), &fromlen);
- memcpy(&from, &sockaddrfrom, fromlen);
- from._sockaddr_storage.sin_family = sockaddrfrom.sa_family;
-#else
- retval = recvfrom(_socket,buf, sizeof(buf), 0,
- reinterpret_cast<sockaddr*>(&from), &fromlen);
-#endif
-
- switch(retval)
- {
- case 0:
- // The peer has performed an orderly shutdown.
- break;
- case SOCKET_ERROR:
- break;
- default:
- if(_wantsIncoming && _incomingCb)
- {
- _incomingCb(_obj,buf, retval, &from);
- }
- break;
- }
-}
-
-void UdpSocketPosix::CloseBlocking()
-{
- _cs->Enter();
- _closeBlockingActive = true;
- if(!CleanUp())
- {
- _closeBlockingActive = false;
- _cs->Leave();
- return;
- }
-
- while(!_readyForDeletion)
- {
- _readyForDeletionCond->SleepCS(*_cs);
- }
- _closeBlockingCompleted = true;
- _closeBlockingCompletedCond->Wake();
- _cs->Leave();
-}
-
-void UdpSocketPosix::ReadyForDeletion()
-{
- _cs->Enter();
- if(!_closeBlockingActive)
- {
- _cs->Leave();
- return;
- }
- close(_socket);
- _socket = INVALID_SOCKET;
- _readyForDeletion = true;
- _readyForDeletionCond->Wake();
- while(!_closeBlockingCompleted)
- {
- _closeBlockingCompletedCond->SleepCS(*_cs);
- }
- _cs->Leave();
-}
-
-bool UdpSocketPosix::CleanUp()
-{
- _wantsIncoming = false;
-
- if (_socket == INVALID_SOCKET)
- {
- return false;
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "calling UdpSocketManager::RemoveSocket()...");
- _mgr->RemoveSocket(this);
- // After this, the socket should may be or will be as deleted. Return
- // immediately.
- return true;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_posix.h b/trunk/src/modules/udp_transport/source/udp_socket_posix.h
deleted file mode 100644
index ee76abb..0000000
--- a/trunk/src/modules/udp_transport/source/udp_socket_posix.h
+++ /dev/null
@@ -1,91 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_POSIX_H_
-#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_POSIX_H_
-
-#include <arpa/inet.h>
-#include <netinet/in.h>
-#include <sys/types.h>
-#include <sys/socket.h>
-
-#include "condition_variable_wrapper.h"
-#include "critical_section_wrapper.h"
-#include "udp_socket_wrapper.h"
-
-#define SOCKET_ERROR -1
-
-namespace webrtc {
-class UdpSocketPosix : public UdpSocketWrapper
-{
-public:
- UdpSocketPosix(const WebRtc_Word32 id, UdpSocketManager* mgr,
- bool ipV6Enable = false);
-
- virtual ~UdpSocketPosix();
-
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- virtual bool SetCallback(CallbackObj obj, IncomingSocketCallback cb);
-
- virtual bool Bind(const SocketAddress& name);
-
- virtual bool SetSockopt(WebRtc_Word32 level, WebRtc_Word32 optname,
- const WebRtc_Word8* optval, WebRtc_Word32 optlen);
-
- virtual WebRtc_Word32 SetTOS(const WebRtc_Word32 serviceType);
-
- virtual WebRtc_Word32 SendTo(const WebRtc_Word8* buf, WebRtc_Word32 len,
- const SocketAddress& to);
-
- // Deletes socket in addition to closing it.
- // TODO (hellner): make destructor protected.
- virtual void CloseBlocking();
-
- virtual SOCKET GetFd() {return _socket;}
- virtual WebRtc_Word32 GetError() {return _error;}
-
- virtual bool ValidHandle();
-
- virtual bool SetQos(WebRtc_Word32 /*serviceType*/,
- WebRtc_Word32 /*tokenRate*/,
- WebRtc_Word32 /*bucketSize*/,
- WebRtc_Word32 /*peekBandwith*/,
- WebRtc_Word32 /*minPolicedSize*/,
- WebRtc_Word32 /*maxSduSize*/,
- const SocketAddress& /*stRemName*/,
- WebRtc_Word32 /*overrideDSCP*/) {return false;}
-
- bool CleanUp();
- void HasIncoming();
- bool WantsIncoming() {return _wantsIncoming;}
- void ReadyForDeletion();
-private:
- friend class UdpSocketManagerPosix;
-
- WebRtc_Word32 _id;
- IncomingSocketCallback _incomingCb;
- CallbackObj _obj;
- WebRtc_Word32 _error;
-
- SOCKET _socket;
- UdpSocketManager* _mgr;
- ConditionVariableWrapper* _closeBlockingCompletedCond;
- ConditionVariableWrapper* _readyForDeletionCond;
-
- bool _closeBlockingActive;
- bool _closeBlockingCompleted;
- bool _readyForDeletion;
-
- CriticalSectionWrapper* _cs;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_POSIX_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_windows.cc b/trunk/src/modules/udp_transport/source/udp_socket_windows.cc
deleted file mode 100644
index 05381e1..0000000
--- a/trunk/src/modules/udp_transport/source/udp_socket_windows.cc
+++ /dev/null
@@ -1,772 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "udp_socket_windows.h"
-
-// Disable deprication warning from traffic.h
-#pragma warning(disable : 4995)
-
-#ifndef WIN32_LEAN_AND_MEAN
-#define WIN32_LEAN_AND_MEAN
-#endif
-
-#include <assert.h>
-#include <stdlib.h>
-#include <stdio.h>
-#include <windows.h>
-#include <Qos.h>
-// Don't change include order for these header files.
-#include <Winsock2.h>
-#include <Ntddndis.h>
-#include <traffic.h>
-
-#include "traffic_control_windows.h"
-#include "udp_socket_manager_wrapper.h"
-
-namespace webrtc {
-typedef struct _QOS_DESTADDR
-{
- QOS_OBJECT_HDR ObjectHdr;
- const struct sockaddr* SocketAddress;
- ULONG SocketAddressLength;
-} QOS_DESTADDR, *LPQOS_DESTADDR;
-
-typedef const QOS_DESTADDR* LPCQOS_DESTADDR;
-
-#define QOS_GENERAL_ID_BASE 2000
-#define QOS_OBJECT_DESTADDR (0x00000004 + QOS_GENERAL_ID_BASE)
-
-#define MAX_PACKET_SIZE 2048
-
-class UDPPacket
-{
-public:
- UDPPacket()
- {
- _length = 0;
- }
- WebRtc_Word32 Set(const WebRtc_Word8* buf, WebRtc_Word32 length)
- {
- if(length > MAX_PACKET_SIZE)
- return 0;
-
- _length = length;
- memcpy(_buffer,buf,length);
- return length;
- }
- WebRtc_Word32 Set(const WebRtc_Word8* buf, WebRtc_Word32 length,
- const SocketAddress* addr)
- {
- if(length > MAX_PACKET_SIZE)
- {
- return 0;
- }
-
- _length = length;
- memcpy(&_remoteAddr,addr,sizeof(SocketAddress));
- memcpy(_buffer,buf,length);
- return length;
- }
-
- SocketAddress _remoteAddr;
- WebRtc_Word8 _buffer[MAX_PACKET_SIZE];
- WebRtc_Word32 _length;
-};
-
-UdpSocketWindows::UdpSocketWindows(const WebRtc_Word32 id,
- UdpSocketManager* mgr, bool ipV6Enable)
- : _id(id),
- _qos(true)
-{
- _wantsIncoming = false;
- _error = 0;
- _mgr = mgr;
- _addedToMgr = false;
-
- _obj = NULL;
- _incomingCb = NULL;
- _socket = INVALID_SOCKET;
- _terminate=false;
-
- _clientHandle = INVALID_HANDLE_VALUE;
- _flowHandle = INVALID_HANDLE_VALUE;
- _filterHandle = INVALID_HANDLE_VALUE;
-
- WEBRTC_TRACE(kTraceMemory, kTraceTransport, _id,
- "UdpSocketWindows::UdpSocketWindows()");
-
- _gtc = NULL;
-
- // Check if QoS is supported.
- WSAPROTOCOL_INFO pProtocolInfo;
- DWORD dwBufLen = 0;
- BOOL bProtocolFound = FALSE;
- WSAPROTOCOL_INFO* lpProtocolBuf = NULL;
-
- // Set dwBufLen to the size needed to retreive all the requested information
- // from WSAEnumProtocols.
- WebRtc_Word32 nRet = WSAEnumProtocols(NULL, lpProtocolBuf, &dwBufLen);
- lpProtocolBuf = (WSAPROTOCOL_INFO*)malloc(dwBufLen);
- nRet = WSAEnumProtocols(NULL, lpProtocolBuf, &dwBufLen);
-
- WebRtc_Word32 iProtocol;
- if (ipV6Enable)
- {
- iProtocol = AF_INET6;
- } else {
- iProtocol = AF_INET;
- }
-
- for (WebRtc_Word32 i = 0; i < nRet; i++)
- {
- if (iProtocol == lpProtocolBuf[i].iAddressFamily && IPPROTO_UDP ==
- lpProtocolBuf[i].iProtocol)
- {
- if ((XP1_QOS_SUPPORTED ==
- (XP1_QOS_SUPPORTED & lpProtocolBuf[i].dwServiceFlags1)))
- {
- pProtocolInfo = lpProtocolBuf[i];
- bProtocolFound = TRUE;
- break;
- }
- }
- }
-
- if(!bProtocolFound)
- {
- _socket = INVALID_SOCKET;
- _qos = false;
- free(lpProtocolBuf);
- _error = SOCKET_ERROR_NO_QOS;
- }else {
- _socket = WSASocket(FROM_PROTOCOL_INFO, FROM_PROTOCOL_INFO,
- FROM_PROTOCOL_INFO,&pProtocolInfo, 0,
- WSA_FLAG_OVERLAPPED);
- free(lpProtocolBuf);
- if (_socket != INVALID_SOCKET)
- {
- return;
- }else
- {
- _qos = false;
- _error = SOCKET_ERROR_NO_QOS;
- }
- }
- // QoS not supported.
- if(ipV6Enable)
- {
- _socket = socket(AF_INET6, SOCK_DGRAM, IPPROTO_UDP);
- }else
- {
- _socket = socket(AF_INET, SOCK_DGRAM, IPPROTO_UDP);
- }
- // Non-blocking mode.
- WebRtc_Word32 iMode = 1;
- ioctlsocket(_socket, FIONBIO, (u_long FAR*) &iMode);
-}
-
-UdpSocketWindows::~UdpSocketWindows()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceTransport, _id,
- "UdpSocketWindows::~UdpSocketWindows()");
- if (_gtc)
- {
- TrafficControlWindows::Release(_gtc);
- }
-}
-
-WebRtc_Word32 UdpSocketWindows::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
- if (_gtc)
- {
- _gtc->ChangeUniqueId(id);
- }
- return 0;
-}
-
-bool UdpSocketWindows::ValidHandle()
-{
- return GetFd() != INVALID_SOCKET;
-}
-
-bool UdpSocketWindows::SetCallback(CallbackObj obj, IncomingSocketCallback cb)
-{
- _obj = obj;
- _incomingCb = cb;
-
- if (_mgr->AddSocket(this))
- {
- _addedToMgr = true;
- return true;
- }
- return false;
-}
-
-bool UdpSocketWindows::SetSockopt(WebRtc_Word32 level, WebRtc_Word32 optname,
- const WebRtc_Word8* optval,
- WebRtc_Word32 optlen)
-{
- if(0 == setsockopt(_socket, level, optname, optval, optlen))
- {
- return true;
- }
- _error = WSAGetLastError();
- return false;
-}
-
-bool UdpSocketWindows::Bind(const SocketAddress& name)
-{
- const struct sockaddr* socketName =
- reinterpret_cast<const struct sockaddr*>(&name);
-
- if (0 == bind(_socket, socketName, sizeof(SocketAddress)))
- {
- _localAddr = name;
- return true;
- }
- _error = WSAGetLastError();
- return false;
-}
-
-WebRtc_Word32 UdpSocketWindows::SendTo(const WebRtc_Word8* buf,
- WebRtc_Word32 len,
- const SocketAddress& to)
-{
- // Don't try to send this packet if there are older packets queued up.
- if(!_notSentPackets.Empty())
- {
- UDPPacket* packet = new UDPPacket();
- packet->Set(buf, len, &to);
- if(!_notSentPackets.Empty())
- {
- _notSentPackets.PushBack(packet);
- return len;
- }else {
- // No old packets queued up. Free to try to send.
- delete packet;
- }
- }
-
- WebRtc_Word32 retVal;
- retVal = sendto(_socket, buf, len, 0,
- reinterpret_cast<const struct sockaddr*>(&to),
- sizeof(SocketAddress));
-
- if(retVal == SOCKET_ERROR)
- {
- _error = WSAGetLastError();
- if (_error == WSAEWOULDBLOCK)
- {
- UDPPacket* packet = new UDPPacket();
- packet->Set(buf,len, &to);
- _notSentPackets.PushBack(packet);
- return len;
- }
- }
- return retVal;
-}
-
-void UdpSocketWindows::HasIncoming()
-{
- WebRtc_Word8 buf[MAX_PACKET_SIZE];
- SocketAddress from;
- int fromlen = sizeof(from);
- WebRtc_Word32 retval = recvfrom(_socket,buf, sizeof(buf), 0,
- reinterpret_cast<struct sockaddr*>(&from),
- &fromlen);
-
- switch(retval)
- {
- case 0:
- // The connection has been gracefully closed.
- break;
- case SOCKET_ERROR:
- _error = WSAGetLastError();
- break;
- default:
- if(_wantsIncoming && _incomingCb)
- _incomingCb(_obj,buf, retval, &from);
- break;
- }
-}
-
-void UdpSocketWindows::CleanUp()
-{
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "UdpSocketWindows::CleanUp()");
- _wantsIncoming = false;
-
- if(_clientHandle != INVALID_HANDLE_VALUE)
- {
- assert(_filterHandle != INVALID_HANDLE_VALUE);
- assert(_flowHandle != INVALID_HANDLE_VALUE);
-
- if (_gtc)
- {
- _gtc->TcDeleteFilter(_filterHandle);
- _gtc->TcDeleteFlow(_flowHandle);
- _gtc->TcDeregisterClient(_clientHandle);
- }
-
- _clientHandle = INVALID_HANDLE_VALUE;
- _filterHandle = INVALID_HANDLE_VALUE;
- _flowHandle = INVALID_HANDLE_VALUE;
- }
-
- while(!_notSentPackets.Empty())
- {
- UDPPacket* packet = (UDPPacket*)_notSentPackets.First()->GetItem();
- if(!packet)
- {
- break;
- }
- delete packet;
- _notSentPackets.PopFront();
- }
-
- if (_socket != INVALID_SOCKET)
- {
- if (closesocket(_socket) == SOCKET_ERROR)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "closesocket() => error = %d", WSAGetLastError());
- }
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "WinSock::closesocket() done");
-
- if(_addedToMgr)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "calling UdpSocketManager::RemoveSocket()");
- _mgr->RemoveSocket(this);
- }
- }
-}
-
-void UdpSocketWindows::SetWritable()
-{
- // Try to send packets that have been queued up.
- while(!_notSentPackets.Empty())
- {
- UDPPacket* packet = (UDPPacket*)_notSentPackets.First()->GetItem();
- if(!packet)
- {
- break;
- }
- if(sendto(
- _socket,packet->_buffer,
- packet->_length,
- 0,
- reinterpret_cast<const struct sockaddr*>(
- &(packet->_remoteAddr)),
- sizeof(SocketAddress)) == SOCKET_ERROR)
- {
- _error = WSAGetLastError();
- if (_error == WSAEWOULDBLOCK)
- {
- return;
- }
- } else {
- delete packet;
- _notSentPackets.PopFront();
- }
- }
-}
-
-bool UdpSocketWindows::SetQos(WebRtc_Word32 serviceType,
- WebRtc_Word32 tokenRate,
- WebRtc_Word32 bucketSize,
- WebRtc_Word32 peekBandwith,
- WebRtc_Word32 minPolicedSize,
- WebRtc_Word32 maxSduSize,
- const SocketAddress &stRemName,
- WebRtc_Word32 overrideDSCP)
-{
- if(_qos == false)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "UdpSocket2Windows::SetQos(), socket not capable of QOS");
- return false;
- }
- QOS Qos;
- WebRtc_Word32 result;
- DWORD BytesRet;
-
- if(overrideDSCP != 0)
- {
- FLOWSPEC f;
- WebRtc_Word32 err = CreateFlowSpec(serviceType, tokenRate, bucketSize,
- peekBandwith, minPolicedSize,
- maxSduSize, &f);
- if(err == -1)
- {
- return false;
- }
- return SetTOSByte(overrideDSCP, &f, &f) == 0;
- }
- memset(&Qos, QOS_NOT_SPECIFIED, sizeof(QOS));
-
- Qos.SendingFlowspec.ServiceType = serviceType;
- Qos.SendingFlowspec.TokenRate = tokenRate;
- Qos.SendingFlowspec.TokenBucketSize = bucketSize;
- Qos.SendingFlowspec.PeakBandwidth = peekBandwith;
- Qos.SendingFlowspec.DelayVariation = QOS_NOT_SPECIFIED;
- Qos.SendingFlowspec.Latency = QOS_NOT_SPECIFIED;
- Qos.SendingFlowspec.MinimumPolicedSize = minPolicedSize;
- Qos.SendingFlowspec.MaxSduSize = maxSduSize;
-
- // Only ServiceType is needed for receiving.
- Qos.ReceivingFlowspec.ServiceType = serviceType;
- Qos.ReceivingFlowspec.TokenRate = QOS_NOT_SPECIFIED;
- Qos.ReceivingFlowspec.TokenBucketSize = QOS_NOT_SPECIFIED;
- Qos.ReceivingFlowspec.PeakBandwidth = QOS_NOT_SPECIFIED;
- Qos.ReceivingFlowspec.Latency = QOS_NOT_SPECIFIED;
- Qos.ReceivingFlowspec.DelayVariation = QOS_NOT_SPECIFIED;
- Qos.ReceivingFlowspec.MinimumPolicedSize = QOS_NOT_SPECIFIED;
- Qos.ReceivingFlowspec.MaxSduSize = QOS_NOT_SPECIFIED;
-
- Qos.ProviderSpecific.len = 0;
- Qos.ProviderSpecific.buf = NULL;
- WebRtc_Word8* p = (WebRtc_Word8*)malloc(sizeof(QOS_DESTADDR) +
- sizeof(QOS_DS_CLASS));
-
- QOS_DESTADDR* QosDestaddr = (QOS_DESTADDR*)p;
- ZeroMemory((WebRtc_Word8 *)QosDestaddr, sizeof(QOS_DESTADDR));
- QosDestaddr->ObjectHdr.ObjectType = QOS_OBJECT_DESTADDR;
- QosDestaddr->ObjectHdr.ObjectLength = sizeof(QOS_DESTADDR);
- QosDestaddr->SocketAddress = (SOCKADDR*)&stRemName;
- QosDestaddr->SocketAddressLength = sizeof(SocketAddress);
- Qos.ProviderSpecific.len = QosDestaddr->ObjectHdr.ObjectLength;
- Qos.ProviderSpecific.buf = (WebRtc_Word8*)p;
-
- // Socket must be bound for this call to be successfull. If socket is not
- // bound WSAGetLastError() will return 10022.
- result = WSAIoctl(GetFd(),SIO_SET_QOS, &Qos,sizeof(QOS),NULL, 0, &BytesRet,
- NULL,NULL);
- if (result == SOCKET_ERROR)
- {
- _error = WSAGetLastError();
- free(p);
- return false;
- }
- free(p);
- return true;
-}
-
-WebRtc_Word32 UdpSocketWindows::SetTOS(WebRtc_Word32 serviceType)
-{
- WebRtc_Word32 res = SetTOSByte(serviceType, NULL, NULL);
-
- if (res == -1)
- {
- OSVERSIONINFO OsVersion;
- OsVersion.dwOSVersionInfoSize = sizeof(OSVERSIONINFO);
- GetVersionEx (&OsVersion);
-
- if ((OsVersion.dwMajorVersion == 4))
- {
- return -1;
- }
- }
- return res;
-}
-
-WebRtc_Word32 UdpSocketWindows::CreateFlowSpec(WebRtc_Word32 serviceType,
- WebRtc_Word32 tokenRate,
- WebRtc_Word32 bucketSize,
- WebRtc_Word32 peekBandwith,
- WebRtc_Word32 minPolicedSize,
- WebRtc_Word32 maxSduSize,
- FLOWSPEC *f)
-{
- if(!f)
- {
- return -1;
- }
-
- f->ServiceType = serviceType;
- f->TokenRate = tokenRate;
- f->TokenBucketSize = bucketSize;
- f->PeakBandwidth = peekBandwith;
- f->DelayVariation = QOS_NOT_SPECIFIED;
- f->Latency = QOS_NOT_SPECIFIED;
- f->MinimumPolicedSize = minPolicedSize;
- f->MaxSduSize = maxSduSize;
- return 0;
-}
-
-WebRtc_Word32 UdpSocketWindows::SetTOSByte(WebRtc_Word32 serviceType,
- FLOWSPEC* send, FLOWSPEC* recv)
-{
- if(_socket == INVALID_SOCKET)
- {
- return -1;
- }
- if (!_gtc)
- {
- _gtc = TrafficControlWindows::GetInstance(_id);
- }
- if (!_gtc)
- {
- return -1;
- }
-
- TCI_CLIENT_FUNC_LIST QoSFunctions;
- QoSFunctions.ClAddFlowCompleteHandler = NULL;
- QoSFunctions.ClDeleteFlowCompleteHandler = NULL;
- QoSFunctions.ClModifyFlowCompleteHandler = NULL;
- QoSFunctions.ClNotifyHandler = (TCI_NOTIFY_HANDLER)MyClNotifyHandler;
- // Register the client with Traffic control interface.
- HANDLE ClientHandle;
- ULONG result = _gtc->TcRegisterClient(CURRENT_TCI_VERSION, NULL,
- &QoSFunctions,&ClientHandle);
- if(result != NO_ERROR)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "TcRegisterClient returned %d", result);
- return result;
- }
-
- // Find traffic control-enabled network interfaces.
- ULONG BufferSize = 0;
- result = _gtc->TcEnumerateInterfaces(ClientHandle, &BufferSize, NULL);
-
- if(result != NO_ERROR && result != ERROR_INSUFFICIENT_BUFFER)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Error enumerating interfaces, %d", result);
- _gtc->TcDeregisterClient(ClientHandle);
- return result;
- }
-
- if(result != ERROR_INSUFFICIENT_BUFFER)
- {
- // Empty buffer contains all control-enabled network interfaces. I.e.
- // ToS is not enabled.
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "Error enumerating interfaces: passed in 0 and received\
- NO_ERROR when expecting INSUFFICIENT_BUFFER, %d");
- _gtc->TcDeregisterClient(ClientHandle);
- return -1;
- }
-
- PTC_IFC_DESCRIPTOR pInterfaceBuffer =
- (PTC_IFC_DESCRIPTOR)malloc(BufferSize);
- if(pInterfaceBuffer == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Out ot memory failure");
- _gtc->TcDeregisterClient(ClientHandle);
- return ERROR_NOT_ENOUGH_MEMORY;
- }
-
- result = _gtc->TcEnumerateInterfaces(ClientHandle, &BufferSize,
- pInterfaceBuffer);
-
- if(result != NO_ERROR)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "Critical: error enumerating interfaces when passing in correct\
- buffer size: %d", result);
- _gtc->TcDeregisterClient(ClientHandle);
- free(pInterfaceBuffer);
- return result;
- }
-
-
- PTC_IFC_DESCRIPTOR oneinterface;
- HANDLE ifcHandle, iFilterHandle, iflowHandle;
- bool addrFound = false;
- ULONG filterSourceAddress = ULONG_MAX;
-
- const struct sockaddr_in* name;
- name = reinterpret_cast<const struct sockaddr_in*>(&_localAddr);
-
- // Find the interface corresponding to the local address.
- for(oneinterface = pInterfaceBuffer;
- oneinterface != (PTC_IFC_DESCRIPTOR)
- (((WebRtc_Word8*)pInterfaceBuffer) + BufferSize);
- oneinterface = (PTC_IFC_DESCRIPTOR)
- ((WebRtc_Word8*)oneinterface + oneinterface->Length))
- {
-
- WebRtc_Word8 interfaceName[500];
- WideCharToMultiByte(CP_ACP, 0, oneinterface->pInterfaceName, -1,
- interfaceName, sizeof(interfaceName), 0, 0);
-
- PNETWORK_ADDRESS_LIST addresses =
- &(oneinterface->AddressListDesc.AddressList);
- for(LONG i = 0; i < addresses->AddressCount; i++)
- {
- // Only look at TCP/IP addresses.
- if(addresses->Address[i].AddressType != NDIS_PROTOCOL_ID_TCP_IP)
- {
- continue;
- }
-
- NETWORK_ADDRESS_IP* pIpAddr =
- (NETWORK_ADDRESS_IP*)&(addresses->Address[i].Address);
-
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "Examining Interface %s", interfaceName);
- if(pIpAddr->in_addr == name->sin_addr.S_un.S_addr)
- {
- filterSourceAddress = pIpAddr->in_addr;
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "Found ip addr: %s", inet_ntoa(name->sin_addr));
- addrFound = true;
- }
- }
- if(!addrFound)
- {
- continue;
- }
- else
- {
- break;
- }
- }
-
- if(!addrFound)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id, "IP Address not found");
- _gtc->TcDeregisterClient(ClientHandle);
- free(pInterfaceBuffer);
- return -1;
- }
-
-
- result = _gtc->TcOpenInterfaceW(oneinterface->pInterfaceName, ClientHandle,
- NULL, &ifcHandle);
-
- if(result != NO_ERROR)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Error opening interface: %d", result);
- _gtc->TcDeregisterClient(ClientHandle);
- free(pInterfaceBuffer);
- return result;
- }
-
- FLOWSPEC defaultSend, defaultRecv;
- if(send == NULL)
- {
- defaultSend.DelayVariation = QOS_NOT_SPECIFIED;
- defaultSend.Latency = QOS_NOT_SPECIFIED;
- defaultSend.MaxSduSize = QOS_NOT_SPECIFIED;
- defaultSend.MinimumPolicedSize = QOS_NOT_SPECIFIED;
- defaultSend.PeakBandwidth = QOS_NOT_SPECIFIED;
- defaultSend.ServiceType = SERVICETYPE_BESTEFFORT;
- defaultSend.TokenBucketSize = QOS_NOT_SPECIFIED;
- defaultSend.TokenRate = 10000;
- }
- else
- {
- defaultSend = *send;
- }
- if(recv == NULL)
- {
- defaultRecv = defaultSend;
- defaultRecv.ServiceType = SERVICETYPE_CONTROLLEDLOAD;
- }
- else
- {
- defaultRecv = *recv;
- }
-
-
- PTC_GEN_FLOW flow =
- (PTC_GEN_FLOW)malloc(sizeof(TC_GEN_FLOW) + sizeof(QOS_DS_CLASS));
-
- flow->ReceivingFlowspec = defaultRecv;
- flow->SendingFlowspec = defaultSend;
-
- QOS_DS_CLASS dsClass;
-
- ZeroMemory((WebRtc_Word8*)&dsClass, sizeof(QOS_DS_CLASS));
-
- dsClass.DSField = serviceType;
-
- dsClass.ObjectHdr.ObjectType = QOS_OBJECT_DS_CLASS;
- dsClass.ObjectHdr.ObjectLength = sizeof(dsClass);
-
- memcpy(flow->TcObjects, (void*)&dsClass, sizeof(QOS_DS_CLASS));
- flow->TcObjectsLength = sizeof(dsClass);
-
- result = _gtc->TcAddFlow(ifcHandle, NULL, 0, flow, &iflowHandle);
- if(result != NO_ERROR)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Error adding flow: %d", result);
- _gtc->TcCloseInterface(ifcHandle);
- _gtc->TcDeregisterClient(ClientHandle);
- free(pInterfaceBuffer);
- return -1;
- }
-
- free(flow);
- IP_PATTERN filterPattern, mask;
-
- ZeroMemory((WebRtc_Word8*)&filterPattern, sizeof(IP_PATTERN));
- ZeroMemory((WebRtc_Word8*)&mask, sizeof(IP_PATTERN));
-
- filterPattern.ProtocolId = IPPROTO_UDP;
- // "name" fields are in network order.
- filterPattern.S_un.S_un_ports.s_srcport = name->sin_port;
- filterPattern.SrcAddr = filterSourceAddress;
-
- // Unsigned max of a type corresponds to a bitmask with all bits set to 1.
- // I.e. the filter should allow all ProtocolIds, any source port and any
- // IP address.
- mask.ProtocolId = UCHAR_MAX;
- mask.S_un.S_un_ports.s_srcport = USHRT_MAX;
- mask.SrcAddr = ULONG_MAX;
-
- TC_GEN_FILTER filter;
- filter.AddressType = NDIS_PROTOCOL_ID_TCP_IP;
- filter.Mask = (LPVOID)&mask;
- filter.Pattern = (LPVOID)&filterPattern;
- filter.PatternSize = sizeof(IP_PATTERN);
- if(_filterHandle != INVALID_HANDLE_VALUE)
- {
- _gtc->TcDeleteFilter(_filterHandle);
- }
-
- result = _gtc->TcAddFilter(iflowHandle, &filter, &iFilterHandle);
- if(result != NO_ERROR)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Error adding filter: %d", result);
- _gtc->TcDeleteFlow(iflowHandle);
- _gtc->TcCloseInterface(ifcHandle);
- _gtc->TcDeregisterClient(ClientHandle);
- free(pInterfaceBuffer);
- return result;
- }
-
- _flowHandle = iflowHandle;
- _filterHandle = iFilterHandle;
- _clientHandle = ClientHandle;
-
- _gtc->TcCloseInterface(ifcHandle);
- free(pInterfaceBuffer);
-
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "Successfully created flow and filter.");
- return 0;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_windows.h b/trunk/src/modules/udp_transport/source/udp_socket_windows.h
deleted file mode 100644
index 0bbd20d..0000000
--- a/trunk/src/modules/udp_transport/source/udp_socket_windows.h
+++ /dev/null
@@ -1,103 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_WINDOWS_H_
-#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_WINDOWS_H_
-
-#include <Winsock2.h>
-
-#include "list_wrapper.h"
-#include "udp_socket_manager_wrapper.h"
-#include "udp_socket_wrapper.h"
-
-namespace webrtc {
-class TrafficControlWindows;
-
-class UdpSocketWindows : public UdpSocketWrapper
-{
-public:
- UdpSocketWindows(const WebRtc_Word32 id, UdpSocketManager* mgr,
- bool ipV6Enable = false);
- virtual ~UdpSocketWindows();
-
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- virtual bool ValidHandle();
-
- virtual bool SetCallback(CallbackObj obj, IncomingSocketCallback cb);
-
- virtual bool Bind(const SocketAddress& name);
-
- virtual bool SetSockopt(WebRtc_Word32 level, WebRtc_Word32 optname,
- const WebRtc_Word8* optval, WebRtc_Word32 optlen);
-
- virtual WebRtc_Word32 SendTo(const WebRtc_Word8* buf, WebRtc_Word32 len,
- const SocketAddress& to);
-
- virtual SOCKET GetFd() {return _socket;}
- virtual WebRtc_Word32 GetError() {return _error;}
-
- virtual bool SetQos(WebRtc_Word32 serviceType, WebRtc_Word32 tokenRate,
- WebRtc_Word32 bucketSize, WebRtc_Word32 peekBandwith,
- WebRtc_Word32 minPolicedSize, WebRtc_Word32 maxSduSize,
- const SocketAddress& stRemName,
- WebRtc_Word32 overrideDSCP = 0);
-
- virtual WebRtc_Word32 SetTOS(const WebRtc_Word32 serviceType);
-
-protected:
- void CleanUp();
- void HasIncoming();
- // Socket is free to process pending packets.
- void SetWritable();
- bool IsWritable() {return _notSentPackets.Empty();}
- bool WantsIncoming() {return _wantsIncoming;}
-
-private:
- friend class UdpSocketManagerWindows;
-
- WebRtc_Word32 CreateFlowSpec(WebRtc_Word32 serviceType,
- WebRtc_Word32 tokenRate,
- WebRtc_Word32 bucketSize,
- WebRtc_Word32 peekBandwith,
- WebRtc_Word32 minPolicedSize,
- WebRtc_Word32 maxSduSize, FLOWSPEC* f);
-
- WebRtc_Word32 SetTOSByte(WebRtc_Word32 serviceType, FLOWSPEC* send,
- FLOWSPEC* recv);
-
- WebRtc_Word32 _id;
- IncomingSocketCallback _incomingCb;
-
- CallbackObj _obj;
- bool _qos;
- WebRtc_Word32 _error;
-
- volatile bool _addedToMgr;
-
- SocketAddress _remoteAddr;
- SocketAddress _localAddr;
- SOCKET _socket;
- UdpSocketManager* _mgr;
-
- ListWrapper _notSentPackets;
- bool _terminate;
-
- // QoS handles.
- HANDLE _clientHandle;
- HANDLE _flowHandle;
- HANDLE _filterHandle;
-
- // TOS implementation.
- TrafficControlWindows* _gtc;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_WINDOWS_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_wrapper.cc b/trunk/src/modules/udp_transport/source/udp_socket_wrapper.cc
deleted file mode 100644
index 53093a1..0000000
--- a/trunk/src/modules/udp_transport/source/udp_socket_wrapper.cc
+++ /dev/null
@@ -1,162 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "udp_socket_wrapper.h"
-
-#include <stdlib.h>
-#include <string.h>
-
-#include "event_wrapper.h"
-#include "trace.h"
-#include "udp_socket_manager_wrapper.h"
-
-#if defined(_WIN32)
- #include "udp_socket_windows.h"
- #include "udp_socket2_windows.h"
-#else
- #include "udp_socket_posix.h"
-#endif
-
-
-namespace webrtc {
-bool UdpSocketWrapper::_initiated = false;
-
-// Temporary Android hack. The value 1024 is taken from
-// <ndk>/build/platforms/android-1.5/arch-arm/usr/include/linux/posix_types.h
-// TODO (tomasl): can we remove this now?
-#ifndef FD_SETSIZE
-#define FD_SETSIZE 1024
-#endif
-
-UdpSocketWrapper::UdpSocketWrapper()
- : _wantsIncoming(false),
- _deleteEvent(NULL)
-{
-}
-
-UdpSocketWrapper::~UdpSocketWrapper()
-{
- if(_deleteEvent)
- {
- _deleteEvent->Set();
- _deleteEvent = NULL;
- }
-}
-
-void UdpSocketWrapper::SetEventToNull()
-{
- if (_deleteEvent)
- {
- _deleteEvent = NULL;
- }
-}
-
-#ifdef USE_WINSOCK2
-UdpSocketWrapper* UdpSocketWrapper::CreateSocket(const WebRtc_Word32 id,
- UdpSocketManager* mgr,
- CallbackObj obj,
- IncomingSocketCallback cb,
- bool ipV6Enable,
- bool disableGQOS)
-#else
-UdpSocketWrapper* UdpSocketWrapper::CreateSocket(const WebRtc_Word32 id,
- UdpSocketManager* mgr,
- CallbackObj obj,
- IncomingSocketCallback cb,
- bool ipV6Enable,
- bool /*disableGQOS*/)
-#endif
-
-{
- WEBRTC_TRACE(kTraceMemory, kTraceTransport, id,
- "UdpSocketWrapper::CreateSocket");
-
- UdpSocketWrapper* s = 0;
-
-#ifdef _WIN32
- if (!_initiated)
- {
- WSADATA wsaData;
- WORD wVersionRequested = MAKEWORD( 2, 2 );
- WebRtc_Word32 err = WSAStartup( wVersionRequested, &wsaData);
- if (err != 0)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- id,
- "UdpSocketWrapper::CreateSocket failed to initialize sockets\
- WSAStartup error:%d",
- err);
- return NULL;
- }
-
- _initiated = true;
- }
-
-#ifdef USE_WINSOCK2
- s = new UdpSocket2Windows(id, mgr, ipV6Enable, disableGQOS);
-#else
- #pragma message("Error: No non-Winsock2 implementation for WinCE")
- s = new UdpSocketWindows(id, mgr, ipV6Enable);
-#endif
-
-#else
- if (!_initiated)
- {
- _initiated = true;
- }
- s = new UdpSocketPosix(id, mgr, ipV6Enable);
- if (s)
- {
- UdpSocketPosix* sl = static_cast<UdpSocketPosix*>(s);
- if (sl->GetFd() != INVALID_SOCKET && sl->GetFd() < FD_SETSIZE)
- {
- // ok
- } else
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- id,
- "UdpSocketWrapper::CreateSocket failed to initialize socket");
- delete s;
- s = NULL;
- }
- }
-#endif
- if (s)
- {
- s->_deleteEvent = NULL;
- if (!s->SetCallback(obj, cb))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- id,
- "UdpSocketWrapper::CreateSocket failed to ser callback");
- return(NULL);
- }
- }
- return s;
-}
-
-bool UdpSocketWrapper::StartReceiving()
-{
- _wantsIncoming = true;
- return true;
-}
-
-bool UdpSocketWrapper::StopReceiving()
-{
- _wantsIncoming = false;
- return true;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_socket_wrapper.h b/trunk/src/modules/udp_transport/source/udp_socket_wrapper.h
deleted file mode 100644
index 9a3b585..0000000
--- a/trunk/src/modules/udp_transport/source/udp_socket_wrapper.h
+++ /dev/null
@@ -1,110 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_WRAPPER_H_
-#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_WRAPPER_H_
-
-#include "udp_transport.h"
-
-namespace webrtc {
-class EventWrapper;
-class UdpSocketManager;
-
-#define SOCKET_ERROR_NO_QOS -1000
-
-#ifndef _WIN32
-typedef int SOCKET;
-#endif
-
-#ifndef INVALID_SOCKET
-#define INVALID_SOCKET (SOCKET)(~0)
-
-#ifndef AF_INET
-#define AF_INET 2
-#endif
-
-#endif
-
-typedef void* CallbackObj;
-typedef void(*IncomingSocketCallback)(CallbackObj obj, const WebRtc_Word8* buf,
- WebRtc_Word32 len,
- const SocketAddress* from);
-
-class UdpSocketWrapper
-{
-public:
- virtual ~UdpSocketWrapper();
- static UdpSocketWrapper* CreateSocket(const WebRtc_Word32 id,
- UdpSocketManager* mgr,
- CallbackObj obj,
- IncomingSocketCallback cb,
- bool ipV6Enable = false,
- bool disableGQOS = false);
-
- // Set the unique identifier of this class to id.
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0;
-
- // Register cb for receiving callbacks when there are incoming packets.
- // Register obj so that it will be passed in calls to cb.
- virtual bool SetCallback(CallbackObj obj, IncomingSocketCallback cb) = 0;
-
- // Socket to local address specified by name.
- virtual bool Bind(const SocketAddress& name) = 0;
-
- // Start receiving UDP data.
- virtual bool StartReceiving();
- virtual inline bool StartReceiving(const WebRtc_UWord32 /*receiveBuffers*/)
- {return StartReceiving();}
- // Stop receiving UDP data.
- virtual bool StopReceiving();
-
- virtual bool ValidHandle() = 0;
-
- // Set socket options.
- virtual bool SetSockopt(WebRtc_Word32 level, WebRtc_Word32 optname,
- const WebRtc_Word8* optval,
- WebRtc_Word32 optlen) = 0;
-
- // Set TOS for outgoing packets.
- virtual WebRtc_Word32 SetTOS(const WebRtc_Word32 serviceType) = 0;
-
- // Set 802.1Q PCP field (802.1p) for outgoing VLAN traffic.
- virtual WebRtc_Word32 SetPCP(const WebRtc_Word32 /*pcp*/) {return -1;}
-
- // Send buf of length len to the address specified by to.
- virtual WebRtc_Word32 SendTo(const WebRtc_Word8* buf, WebRtc_Word32 len,
- const SocketAddress& to) = 0;
-
- virtual void SetEventToNull();
-
- // Close socket and don't return until completed.
- virtual void CloseBlocking() {}
-
- // tokenRate is in bit/s. peakBandwidt is in byte/s
- virtual bool SetQos(WebRtc_Word32 serviceType, WebRtc_Word32 tokenRate,
- WebRtc_Word32 bucketSize, WebRtc_Word32 peekBandwith,
- WebRtc_Word32 minPolicedSize, WebRtc_Word32 maxSduSize,
- const SocketAddress &stRemName,
- WebRtc_Word32 overrideDSCP = 0) = 0;
-
- virtual WebRtc_UWord32 ReceiveBuffers() {return 0;};
-
-protected:
- UdpSocketWrapper();
-
- bool _wantsIncoming;
- EventWrapper* _deleteEvent;
-
-private:
- static bool _initiated;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_SOCKET_WRAPPER_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_transport.gypi b/trunk/src/modules/udp_transport/source/udp_transport.gypi
deleted file mode 100644
index c30e167..0000000
--- a/trunk/src/modules/udp_transport/source/udp_transport.gypi
+++ /dev/null
@@ -1,120 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'udp_transport',
- 'type': '<(library)',
- 'dependencies': [
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '../interface',
- '../../interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- '../../interface',
- ],
- },
- 'sources': [
- # PLATFORM INDEPENDENT SOURCE FILES
- '../interface/udp_transport.h',
- 'udp_transport_impl.cc',
- 'udp_socket_wrapper.cc',
- 'udp_socket_manager_wrapper.cc',
- 'udp_transport_impl.h',
- 'udp_socket_wrapper.h',
- 'udp_socket_manager_wrapper.h',
- # PLATFORM SPECIFIC SOURCE FILES - Will be filtered below
- # Posix (Linux/Mac)
- 'udp_socket_posix.cc',
- 'udp_socket_posix.h',
- 'udp_socket_manager_posix.cc',
- 'udp_socket_manager_posix.h',
- # Windows
- 'udp_socket_manager_windows.cc',
- 'udp_socket_manager_windows.h',
- 'udp_socket2_manager_windows.cc',
- 'udp_socket2_manager_windows.h',
- 'udp_socket_windows.cc',
- 'udp_socket_windows.h',
- 'udp_socket2_windows.cc',
- 'udp_socket2_windows.h',
- 'traffic_control_windows.cc',
- 'traffic_control_windows.h',
- ], # source
- 'conditions': [
- # DEFINE PLATFORM SPECIFIC SOURCE FILES
- ['os_posix==0', {
- 'sources!': [
- 'udp_socket_posix.cc',
- 'udp_socket_posix.h',
- 'udp_socket_manager_posix.cc',
- 'udp_socket_manager_posix.h',
- ],
- }],
- ['OS!="win"', {
- 'sources!': [
- 'udp_socket_manager_windows.cc',
- 'udp_socket_manager_windows.h',
- 'udp_socket2_manager_windows.cc',
- 'udp_socket2_manager_windows.h',
- 'udp_socket_windows.cc',
- 'udp_socket_windows.h',
- 'udp_socket2_windows.cc',
- 'udp_socket2_windows.h',
- 'traffic_control_windows.cc',
- 'traffic_control_windows.h',
- ],
- }],
- ['OS=="linux"', {
- 'cflags': [
- '-fno-strict-aliasing',
- ],
- }],
- ['OS=="mac"', {
- 'xcode_settings': {
- 'OTHER_CPLUSPLUSFLAGS': '-fno-strict-aliasing',
- },
- }],
- ['OS=="win"', {
- 'defines': [
- 'USE_WINSOCK2',
- ],
- }],
- ] # conditions
- },
- ], # targets
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'udp_transport_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'udp_transport',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- ],
- 'sources': [
- 'udp_transport_unittest.cc',
- ],
- }, # udp_transport_unittests
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/udp_transport/source/udp_transport_impl.cc b/trunk/src/modules/udp_transport/source/udp_transport_impl.cc
deleted file mode 100644
index 9ce2d92..0000000
--- a/trunk/src/modules/udp_transport/source/udp_transport_impl.cc
+++ /dev/null
@@ -1,3009 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "udp_transport_impl.h"
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <time.h>
-
-#if defined(_WIN32)
- #include <winsock2.h>
- #include <ws2tcpip.h>
- // Disable warning for default initialized arrays on VS2005
- #pragma warning(disable:4351)
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
- #include <arpa/inet.h>
- #include <ctype.h>
- #include <fcntl.h>
- #include <netdb.h>
- #include <net/if.h>
- #include <netinet/in.h>
- #include <stdlib.h>
- #include <sys/ioctl.h>
- #include <sys/socket.h>
- #include <sys/time.h>
- #include <unistd.h>
-#ifndef MAC_IPHONE
- #include <net/if_arp.h>
-#endif
-#endif // defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
-
-#if defined(WEBRTC_MAC)
-#include <ifaddrs.h>
-#include <machine/types.h>
-#endif
-#if defined(WEBRTC_LINUX)
-#include <linux/netlink.h>
-#include <linux/rtnetlink.h>
-#endif
-
-#include "common_types.h"
-#include "critical_section_wrapper.h"
-#include "rw_lock_wrapper.h"
-#include "trace.h"
-#include "typedefs.h"
-#include "udp_socket_manager_wrapper.h"
-
-#if defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
-#define GetLastError() errno
-
-#define IFRSIZE ((int)(size * sizeof (struct ifreq)))
-
-#define NLMSG_OK_NO_WARNING(nlh,len) \
- ((len) >= (int)sizeof(struct nlmsghdr) && \
- (int)(nlh)->nlmsg_len >= (int)sizeof(struct nlmsghdr) && \
- (int)(nlh)->nlmsg_len <= (len))
-
-#endif // defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
-
-namespace webrtc {
-UdpTransport* UdpTransport::Create(const WebRtc_Word32 id,
- WebRtc_UWord8& numSocketThreads)
-{
- return new UdpTransportImpl(id, numSocketThreads);
-}
-
-void UdpTransport::Destroy(UdpTransport* module)
-{
- if(module)
- {
- delete module;
- }
-}
-
-UdpTransportImpl::UdpTransportImpl(const WebRtc_Word32 id,
- WebRtc_UWord8& numSocketThreads)
- : _id(id),
- _crit(CriticalSectionWrapper::CreateCriticalSection()),
- _critFilter(CriticalSectionWrapper::CreateCriticalSection()),
- _critPacketCallback(CriticalSectionWrapper::CreateCriticalSection()),
- _mgr(UdpSocketManager::Create(id, numSocketThreads)),
- _lastError(kNoSocketError),
- _destPort(0),
- _destPortRTCP(0),
- _localPort(0),
- _localPortRTCP(0),
- _srcPort(0),
- _srcPortRTCP(0),
- _fromPort(0),
- _fromPortRTCP(0),
- _fromIP(),
- _destIP(),
- _localIP(),
- _localMulticastIP(),
- _ptrRtpSocket(NULL),
- _ptrRtcpSocket(NULL),
- _ptrSendRtpSocket(NULL),
- _ptrSendRtcpSocket(NULL),
- _remoteRTPAddr(),
- _remoteRTCPAddr(),
- _localRTPAddr(),
- _localRTCPAddr(),
- _tos(0),
- _receiving(false),
- _useSetSockOpt(false),
- _qos(false),
- _pcp(0),
- _ipV6Enabled(false),
- _serviceType(0),
- _overrideDSCP(0),
- _maxBitrate(0),
- _cachLock(RWLockWrapper::CreateRWLock()),
- _previousAddress(),
- _previousIP(),
- _previousIPSize(0),
- _previousSourcePort(0),
- _filterIPAddress(),
- _rtpFilterPort(0),
- _rtcpFilterPort(0),
- _packetCallback(0)
-{
- memset(&_remoteRTPAddr, 0, sizeof(_remoteRTPAddr));
- memset(&_remoteRTCPAddr, 0, sizeof(_remoteRTCPAddr));
- memset(&_localRTPAddr, 0, sizeof(_localRTPAddr));
- memset(&_localRTCPAddr, 0, sizeof(_localRTCPAddr));
-
- memset(_fromIP, 0, sizeof(_fromIP));
- memset(_destIP, 0, sizeof(_destIP));
- memset(_localIP, 0, sizeof(_localIP));
- memset(_localMulticastIP, 0, sizeof(_localMulticastIP));
-
- memset(&_filterIPAddress, 0, sizeof(_filterIPAddress));
- if(_mgr == NULL)
- {
- _mgr = UdpSocketManager::Create(id, numSocketThreads);
- }
-
- WEBRTC_TRACE(kTraceMemory, kTraceTransport, id, "%s created", __FUNCTION__);
-}
-
-UdpTransportImpl::~UdpTransportImpl()
-{
- CloseSendSockets();
- CloseReceiveSockets();
- delete _crit;
- delete _critFilter;
- delete _critPacketCallback;
- delete _cachLock;
-
- UdpSocketManager::Return();
- WEBRTC_TRACE(kTraceMemory, kTraceTransport, _id, "%s deleted",
- __FUNCTION__);
-}
-
-WebRtc_Word32 UdpTransportImpl::ChangeUniqueId(const WebRtc_Word32 id)
-{
-
- CriticalSectionScoped cs(_crit);
- _id = id;
- if(_mgr)
- {
- _mgr->ChangeUniqueId(id);
- }
- if(_ptrRtpSocket)
- {
- _ptrRtpSocket->ChangeUniqueId(id);
- }
- if(_ptrRtcpSocket)
- {
- _ptrRtcpSocket->ChangeUniqueId(id);
- }
- if(_ptrSendRtpSocket)
- {
- _ptrSendRtpSocket->ChangeUniqueId(id);
- }
- if(_ptrSendRtcpSocket)
- {
- _ptrSendRtcpSocket->ChangeUniqueId(id);
- }
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::TimeUntilNextProcess()
-{
- return 100;
-}
-
-WebRtc_Word32 UdpTransportImpl::Process()
-{
- return 0;
-}
-
-UdpTransport::ErrorCode UdpTransportImpl::LastError() const
-{
- return _lastError;
-}
-
-bool SameAddress(const SocketAddress& address1, const SocketAddress& address2)
-{
- return (memcmp(&address1,&address2,sizeof(address1)) == 0);
-}
-
-void UdpTransportImpl::GetCachedAddress(char* ip,
- WebRtc_UWord32& ipSize,
- WebRtc_UWord16& sourcePort)
-{
- const WebRtc_UWord32 originalIPSize = ipSize;
- // If the incoming string is too small, fill it as much as there is room
- // for. Make sure that there is room for the '\0' character.
- ipSize = (ipSize - 1 < _previousIPSize) ? ipSize - 1 : _previousIPSize;
- memcpy(ip,_previousIP,sizeof(WebRtc_Word8)*(ipSize + 1));
- ip[originalIPSize - 1] = '\0';
- sourcePort = _previousSourcePort;
-}
-
-WebRtc_Word32 UdpTransportImpl::IPAddressCached(const SocketAddress& address,
- char* ip,
- WebRtc_UWord32& ipSize,
- WebRtc_UWord16& sourcePort)
-{
- {
- ReadLockScoped rl(*_cachLock);
- // Check if the old address can be re-used (is the same).
- if(SameAddress(address,_previousAddress))
- {
- GetCachedAddress(ip,ipSize,sourcePort);
- return 0;
- }
- }
- // Get the new address and store it.
- WriteLockScoped wl(*_cachLock);
- ipSize = kIpAddressVersion6Length;
- if(IPAddress(address,_previousIP,ipSize,_previousSourcePort) != 0)
- {
- return -1;
- }
- _previousIPSize = ipSize;
- memcpy(&_previousAddress, &address, sizeof(address));
- // Address has been cached at this point.
- GetCachedAddress(ip,ipSize,sourcePort);
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::InitializeReceiveSockets(
- UdpTransportData* const packetCallback,
- const WebRtc_UWord16 portnr,
- const char* ip,
- const char* multicastIpAddr,
- const WebRtc_UWord16 rtcpPort)
-{
-
- {
- CriticalSectionScoped cs(_critPacketCallback);
- _packetCallback = packetCallback;
-
- if(packetCallback == NULL)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, _id,
- "Closing down receive sockets");
- return 0;
- }
- }
-
- CriticalSectionScoped cs(_crit);
- CloseReceiveSockets();
-
- if(portnr == 0)
- {
- // TODO (hellner): why not just fail here?
- if(_destPort == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "InitializeReceiveSockets port 0 not allowed");
- _lastError = kPortInvalid;
- return -1;
- }
- _localPort = _destPort;
- } else {
- _localPort = portnr;
- }
- if(rtcpPort)
- {
- _localPortRTCP = rtcpPort;
- }else {
- _localPortRTCP = _localPort + 1;
- WEBRTC_TRACE(
- kTraceStateInfo,
- kTraceTransport,
- _id,
- "InitializeReceiveSockets RTCP port not configured using RTP\
- port+1=%d",
- _localPortRTCP);
- }
-
- if(ip)
- {
- if(IsIpAddressValid(ip,IpV6Enabled()))
- {
- strncpy(_localIP, ip,kIpAddressVersion6Length);
- } else
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "InitializeReceiveSockets invalid IP address");
- _lastError = kIpAddressInvalid;
- return -1;
- }
- }else
- {
- // Don't bind to a specific IP address.
- if(! IpV6Enabled())
- {
- strncpy(_localIP, "0.0.0.0",16);
- } else
- {
- strncpy(_localIP, "0000:0000:0000:0000:0000:0000:0000:0000",
- kIpAddressVersion6Length);
- }
- }
- if(multicastIpAddr && !IpV6Enabled())
- {
- if(IsIpAddressValid(multicastIpAddr,IpV6Enabled()))
- {
- strncpy(_localMulticastIP, multicastIpAddr,
- kIpAddressVersion6Length);
- } else
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "InitializeReceiveSockets invalid IP address");
- _lastError = kIpAddressInvalid;
- return -1;
- }
- }
- if(_mgr == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "InitializeReceiveSockets no socket manager");
- return -1;
- }
-
- _useSetSockOpt=false;
- _tos=0;
- _pcp=0;
-
- _ptrRtpSocket = UdpSocketWrapper::CreateSocket(_id, _mgr, this,
- IncomingRTPCallback,
- IpV6Enabled());
-
- _ptrRtcpSocket = UdpSocketWrapper::CreateSocket(_id, _mgr, this,
- IncomingRTCPCallback,
- IpV6Enabled());
-
- ErrorCode retVal = BindLocalRTPSocket();
- if(retVal != kNoSocketError)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "InitializeReceiveSockets faild to bind RTP socket");
- _lastError = retVal;
- CloseReceiveSockets();
- return -1;
- }
- retVal = BindLocalRTCPSocket();
- if(retVal != kNoSocketError)
- {
- _lastError = retVal;
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "InitializeReceiveSockets faild to bind RTCP socket");
- CloseReceiveSockets();
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::ReceiveSocketInformation(
- char ipAddr[kIpAddressVersion6Length],
- WebRtc_UWord16& rtpPort,
- WebRtc_UWord16& rtcpPort,
- char multicastIpAddr[kIpAddressVersion6Length]) const
-{
- CriticalSectionScoped cs(_crit);
- rtpPort = _localPort;
- rtcpPort = _localPortRTCP;
- if (ipAddr)
- {
- strncpy(ipAddr, _localIP, IpV6Enabled() ?
- UdpTransport::kIpAddressVersion6Length :
- UdpTransport::kIpAddressVersion4Length);
- }
- if (multicastIpAddr)
- {
- strncpy(multicastIpAddr, _localMulticastIP, IpV6Enabled() ?
- UdpTransport::kIpAddressVersion6Length :
- UdpTransport::kIpAddressVersion4Length);
- }
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::SendSocketInformation(
- char ipAddr[kIpAddressVersion6Length],
- WebRtc_UWord16& rtpPort,
- WebRtc_UWord16& rtcpPort) const
-{
- CriticalSectionScoped cs(_crit);
- rtpPort = _destPort;
- rtcpPort = _destPortRTCP;
- strncpy(ipAddr, _destIP, IpV6Enabled() ?
- UdpTransport::kIpAddressVersion6Length :
- UdpTransport::kIpAddressVersion4Length);
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::RemoteSocketInformation(
- char ipAddr[kIpAddressVersion6Length],
- WebRtc_UWord16& rtpPort,
- WebRtc_UWord16& rtcpPort) const
-{
- CriticalSectionScoped cs(_crit);
- rtpPort = _fromPort;
- rtcpPort = _fromPortRTCP;
- if(ipAddr)
- {
- strncpy(ipAddr, _fromIP, IpV6Enabled() ?
- kIpAddressVersion6Length :
- kIpAddressVersion4Length);
- }
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::FilterPorts(
- WebRtc_UWord16& rtpFilterPort,
- WebRtc_UWord16& rtcpFilterPort) const
-{
- CriticalSectionScoped cs(_critFilter);
- rtpFilterPort = _rtpFilterPort;
- rtcpFilterPort = _rtcpFilterPort;
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::SetQoS(bool QoS, WebRtc_Word32 serviceType,
- WebRtc_UWord32 maxBitrate,
- WebRtc_Word32 overrideDSCP, bool audio)
-{
- if(QoS)
- {
- return EnableQoS(serviceType, audio, maxBitrate, overrideDSCP);
- }else
- {
- return DisableQoS();
- }
-}
-
-WebRtc_Word32 UdpTransportImpl::EnableQoS(WebRtc_Word32 serviceType,
- bool audio, WebRtc_UWord32 maxBitrate,
- WebRtc_Word32 overrideDSCP)
-{
- if (_ipV6Enabled)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "QOS is enabled but will be ignored since IPv6 is enabled");
- _lastError = kQosError;
- return -1;
- }
- if (_tos)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "TOS already enabled, can't use TOS and QoS at the same time");
- _lastError = kQosError;
- return -1;
- }
- if (_pcp)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "PCP already enabled, can't use PCP and QoS at the same time");
- _lastError = kQosError;
- return -1;
- }
- if(_destPort == 0)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "QOS is enabled but not started since we have not yet configured\
- the send destination");
- return -1;
- }
- if(_qos)
- {
- if(_overrideDSCP == 0 && overrideDSCP != 0)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "QOS is already enabled and overrideDSCP differs, not allowed");
- return -1;
- }
- }
- CriticalSectionScoped cs(_crit);
-
- UdpSocketWrapper* rtpSock = _ptrSendRtpSocket ?
- _ptrSendRtpSocket :
- _ptrRtpSocket;
- if (!rtpSock || !rtpSock->ValidHandle())
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "QOS is enabled but not started since we have not yet created the\
- RTP socket");
- return -1;
- }
- UdpSocketWrapper* rtcpSock = _ptrSendRtcpSocket ?
- _ptrSendRtcpSocket :
- _ptrRtcpSocket;
- if (!rtcpSock || !rtcpSock->ValidHandle())
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "QOS is enabled but not started since we have not yet created the\
- RTCP socket");
- return -1;
- }
-
- // Minimum packet size in bytes for which the requested quality of service
- // will be provided. The smallest RTP header is 12 byte.
- const WebRtc_Word32 min_policed_size = 12;
- // Max SDU, maximum packet size permitted or used in the traffic flow, in
- // bytes.
- const WebRtc_Word32 max_sdu_size = 1500;
-
- // Enable QoS for RTP sockets.
- if(maxBitrate)
- {
- // Note: 1 kbit is 125 bytes.
- // Token Rate is typically set to the average bit rate from peak to
- // peak.
- // Bucket size is normally set to the largest average frame size.
- if(audio)
- {
- WEBRTC_TRACE(kTraceStateInfo,
- kTraceTransport,
- _id,
- "Enable QOS for audio with max bitrate:%d",
- maxBitrate);
-
- const WebRtc_Word32 token_rate = maxBitrate*125;
- // The largest audio packets are 60ms frames. This is a fraction
- // more than 16 packets/second. These 16 frames are sent, at max,
- // at a bitrate of maxBitrate*125 -> 1 frame is maxBitrate*125/16 ~
- // maxBitrate * 8.
- const WebRtc_Word32 bucket_size = maxBitrate * 8;
- const WebRtc_Word32 peek_bandwith = maxBitrate * 125;
- if (!rtpSock->SetQos(serviceType, token_rate, bucket_size,
- peek_bandwith, min_policed_size,
- max_sdu_size, _remoteRTPAddr, overrideDSCP))
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "QOS failed on the RTP socket");
- _lastError = kQosError;
- return -1;
- }
- }else
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, _id,
- "Enable QOS for video with max bitrate:%d",
- maxBitrate);
-
- // Allow for a token rate that is twice that of the maximum bitrate
- // (in bytes).
- const WebRtc_Word32 token_rate = maxBitrate*250;
- // largest average frame size (key frame size). Assuming that a
- // keyframe is 25% of the bitrate during the second its sent
- // Assume that a key frame is 25% of the bitrate the second that it
- // is sent. The largest frame size is then maxBitrate* 125 * 0.25 ~
- // 31.
- const WebRtc_Word32 bucket_size = maxBitrate*31;
- const WebRtc_Word32 peek_bandwith = maxBitrate*125;
- if (!rtpSock->SetQos(serviceType, token_rate, bucket_size,
- peek_bandwith, min_policed_size, max_sdu_size,
- _remoteRTPAddr, overrideDSCP))
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "QOS failed on the RTP socket");
- _lastError = kQosError;
- return -1;
- }
- }
- } else if(audio)
- {
- // No max bitrate set. Audio.
- WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, _id,
- "Enable QOS for audio with default max bitrate");
-
- // Let max bitrate be 240kbit/s.
- const WebRtc_Word32 token_rate = 30000;
- const WebRtc_Word32 bucket_size = 2000;
- const WebRtc_Word32 peek_bandwith = 30000;
- if (!rtpSock->SetQos(serviceType, token_rate, bucket_size,
- peek_bandwith, min_policed_size, max_sdu_size,
- _remoteRTPAddr, overrideDSCP))
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "QOS failed on the RTP socket");
- _lastError = kQosError;
- return -1;
- }
- }else
- {
- // No max bitrate set. Video.
- WEBRTC_TRACE(kTraceStateInfo, kTraceTransport, _id,
- "Enable QOS for video with default max bitrate");
-
- // Let max bitrate be 10mbit/s.
- const WebRtc_Word32 token_rate = 128000*10;
- const WebRtc_Word32 bucket_size = 32000;
- const WebRtc_Word32 peek_bandwith = 256000;
- if (!rtpSock->SetQos(serviceType, token_rate, bucket_size,
- peek_bandwith, min_policed_size, max_sdu_size,
- _remoteRTPAddr, overrideDSCP))
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "QOS failed on the RTP socket");
- _lastError = kQosError;
- return -1;
- }
- }
-
- // Enable QoS for RTCP sockets.
- // TODO (hellner): shouldn't RTCP be based on 5% of the maximum bandwidth?
- if(audio)
- {
- const WebRtc_Word32 token_rate = 200;
- const WebRtc_Word32 bucket_size = 200;
- const WebRtc_Word32 peek_bandwith = 400;
- if (!rtcpSock->SetQos(serviceType, token_rate, bucket_size,
- peek_bandwith, min_policed_size, max_sdu_size,
- _remoteRTCPAddr, overrideDSCP))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
- "QOS failed on the RTCP socket");
- _lastError = kQosError;
- }
- }else
- {
- const WebRtc_Word32 token_rate = 5000;
- const WebRtc_Word32 bucket_size = 100;
- const WebRtc_Word32 peek_bandwith = 10000;
- if (!rtcpSock->SetQos(serviceType, token_rate, bucket_size,
- peek_bandwith, min_policed_size, max_sdu_size,
- _remoteRTCPAddr, _overrideDSCP))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
- "QOS failed on the RTCP socket");
- _lastError = kQosError;
- }
- }
- _qos = true;
- _serviceType = serviceType;
- _maxBitrate = maxBitrate;
- _overrideDSCP = overrideDSCP;
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::DisableQoS()
-{
- if(_qos == false)
- {
- return 0;
- }
- CriticalSectionScoped cs(_crit);
-
- UdpSocketWrapper* rtpSock = (_ptrSendRtpSocket ?
- _ptrSendRtpSocket : _ptrRtpSocket);
- if (!rtpSock || !rtpSock->ValidHandle())
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "QOS is enabled but not started since we have not yet created the\
- RTP socket");
- return -1;
- }
- UdpSocketWrapper* rtcpSock = (_ptrSendRtcpSocket ?
- _ptrSendRtcpSocket : _ptrRtcpSocket);
- if (!rtcpSock || !rtcpSock->ValidHandle())
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "QOS is enabled but not started since we have not yet created the\
- RTCP socket");
- return -1;
- }
-
- const WebRtc_Word32 service_type = 0; // = SERVICETYPE_NOTRAFFIC
- const WebRtc_Word32 not_specified = -1;
- if (!rtpSock->SetQos(service_type, not_specified, not_specified,
- not_specified, not_specified, not_specified,
- _remoteRTPAddr, _overrideDSCP))
- {
- _lastError = kQosError;
- return -1;
- }
- if (!rtcpSock->SetQos(service_type, not_specified, not_specified,
- not_specified, not_specified, not_specified,
- _remoteRTCPAddr,_overrideDSCP))
- {
- _lastError = kQosError;
- }
- _qos = false;
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::QoS(bool& QoS, WebRtc_Word32& serviceType,
- WebRtc_Word32& overrideDSCP) const
-{
- CriticalSectionScoped cs(_crit);
- QoS = _qos;
- serviceType = _serviceType;
- overrideDSCP = _overrideDSCP;
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::SetToS(WebRtc_Word32 DSCP, bool useSetSockOpt)
-{
- if (_qos)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id, "QoS already enabled");
- _lastError = kQosError;
- return -1;
- }
- if (DSCP < 0 || DSCP > 63)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id, "Invalid DSCP");
- _lastError = kTosInvalid;
- return -1;
- }
- if(_tos)
- {
- if(useSetSockOpt != _useSetSockOpt)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "Can't switch SetSockOpt method without disabling TOS first");
- _lastError = kTosInvalid;
- return -1;
- }
- }
- CriticalSectionScoped cs(_crit);
- UdpSocketWrapper* rtpSock = NULL;
- UdpSocketWrapper* rtcpSock = NULL;
- if(_ptrSendRtpSocket)
- {
- rtpSock = _ptrSendRtpSocket;
- }else
- {
- rtpSock = _ptrRtpSocket;
- }
- if (rtpSock == NULL)
- {
- _lastError = kSocketInvalid;
- return -1;
- }
- if(!rtpSock->ValidHandle())
- {
- _lastError = kSocketInvalid;
- return -1;
- }
- if(_ptrSendRtcpSocket)
- {
- rtcpSock = _ptrSendRtcpSocket;
- }else
- {
- rtcpSock = _ptrRtcpSocket;
- }
- if (rtcpSock == NULL)
- {
- _lastError = kSocketInvalid;
- return -1;
- }
- if(!rtcpSock->ValidHandle())
- {
- _lastError = kSocketInvalid;
- return -1;
- }
-
- if (useSetSockOpt)
- {
-#ifdef _WIN32
- OSVERSIONINFO OsVersion;
- OsVersion.dwOSVersionInfoSize = sizeof(OSVERSIONINFO);
- GetVersionEx(&OsVersion);
- // Disable QoS before setting ToS on Windows XP. This is done by closing
- // and re-opening the sockets.
- // TODO (hellner): why not just fail here and force the user to
- // re-initialize sockets? Doing this may trick the user
- // into thinking that the sockets are in a state which
- // they aren't.
- if (OsVersion.dwMajorVersion == 5 &&
- OsVersion.dwMinorVersion == 1)
- {
- if(!_useSetSockOpt)
- {
- if(_ptrSendRtpSocket)
- {
- CloseSendSockets();
- _ptrSendRtpSocket =
- UdpSocketWrapper::CreateSocket(_id, _mgr, NULL,
- NULL, IpV6Enabled(),
- true);
- _ptrSendRtcpSocket =
- UdpSocketWrapper::CreateSocket(_id, _mgr, NULL,
- NULL, IpV6Enabled(),
- true);
- rtpSock=_ptrSendRtpSocket;
- rtcpSock=_ptrSendRtcpSocket;
- ErrorCode retVal = BindRTPSendSocket();
- if(retVal != kNoSocketError)
- {
- _lastError = retVal;
- return -1;
- }
- retVal = BindRTCPSendSocket();
- if(retVal != kNoSocketError)
- {
- _lastError = retVal;
- return -1;
- }
- }
- else
- {
- bool receiving=_receiving;
- WebRtc_UWord32 noOfReceiveBuffers = 0;
- if(receiving)
- {
- noOfReceiveBuffers=_ptrRtpSocket->ReceiveBuffers();
- if(StopReceiving()!=0)
- {
- return -1;
- }
- }
- CloseReceiveSockets();
- _ptrRtpSocket = UdpSocketWrapper::CreateSocket(
- _id, _mgr, this, IncomingRTPCallback,
- IpV6Enabled(), true);
- _ptrRtcpSocket = UdpSocketWrapper::CreateSocket(
- _id, _mgr, this, IncomingRTCPCallback,
- IpV6Enabled(),true);
- rtpSock=_ptrRtpSocket;
- rtcpSock=_ptrRtcpSocket;
- ErrorCode retVal = BindLocalRTPSocket();
- if(retVal != kNoSocketError)
- {
- _lastError = retVal;
- return -1;
- }
- retVal = BindLocalRTCPSocket();
- if(retVal != kNoSocketError)
- {
- _lastError = retVal;
- return -1;
- }
- if(receiving)
- {
- if(StartReceiving(noOfReceiveBuffers) !=
- kNoSocketError)
- {
- return -1;
- }
- }
- }
- }
- }
-#endif // #ifdef _WIN32
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "Setting TOS using SetSockopt");
- WebRtc_Word32 TOSShifted = DSCP << 2;
- if (!rtpSock->SetSockopt(IPPROTO_IP, IP_TOS,
- (WebRtc_Word8*) &TOSShifted, 4))
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Could not SetSockopt tos value on RTP socket");
- _lastError = kTosInvalid;
- return -1;
- }
- if (!rtcpSock->SetSockopt(IPPROTO_IP, IP_TOS,
- (WebRtc_Word8*) &TOSShifted, 4))
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Could not sSetSockopt tos value on RTCP socket");
- _lastError = kTosInvalid;
- return -1;
- }
- } else
- {
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id,
- "Setting TOS NOT using SetSockopt");
- if (rtpSock->SetTOS(DSCP) != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Could not set tos value on RTP socket");
- _lastError = kTosError;
- return -1;
- }
- if (rtcpSock->SetTOS(DSCP) != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Could not set tos value on RTCP socket");
- _lastError = kTosError;
- return -1;
- }
- }
- _useSetSockOpt = useSetSockOpt;
- _tos = DSCP;
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::ToS(WebRtc_Word32& DSCP,
- bool& useSetSockOpt) const
-{
- CriticalSectionScoped cs(_crit);
- DSCP = _tos;
- useSetSockOpt = _useSetSockOpt;
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::SetPCP(WebRtc_Word32 PCP)
-{
-
- if (_qos)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id, "QoS already enabled");
- _lastError = kQosError;
- return -1;
- }
- if ((PCP < 0) || (PCP > 7))
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id, "Invalid PCP");
- _lastError = kPcpError;
- return -1;
- }
-
- CriticalSectionScoped cs(_crit);
- UdpSocketWrapper* rtpSock = NULL;
- UdpSocketWrapper* rtcpSock = NULL;
- if(_ptrSendRtpSocket)
- {
- rtpSock = _ptrSendRtpSocket;
- }else
- {
- rtpSock = _ptrRtpSocket;
- }
- if (rtpSock == NULL)
- {
- _lastError = kSocketInvalid;
- return -1;
- }
- if(!rtpSock->ValidHandle())
- {
- _lastError = kSocketInvalid;
- return -1;
- }
- if(_ptrSendRtcpSocket)
- {
- rtcpSock = _ptrSendRtcpSocket;
- }else
- {
- rtcpSock = _ptrRtcpSocket;
- }
- if (rtcpSock == NULL)
- {
- _lastError = kSocketInvalid;
- return -1;
- }
- if(!rtcpSock->ValidHandle())
- {
- _lastError = kSocketInvalid;
- return -1;
- }
-
-#if defined(_WIN32)
- if (rtpSock->SetPCP(PCP) != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Could not set PCP value on RTP socket");
- _lastError = kPcpError;
- return -1;
- }
- if (rtcpSock->SetPCP(PCP) != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Could not set PCP value on RTCP socket");
- _lastError = kPcpError;
- return -1;
- }
-
-#elif defined(WEBRTC_LINUX)
- if (!rtpSock->SetSockopt(SOL_SOCKET, SO_PRIORITY, (WebRtc_Word8*) &PCP,
- sizeof(PCP)))
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Could not SetSockopt PCP value on RTP socket");
- _lastError = kPcpError;
- return -1;
- }
- if (!rtcpSock->SetSockopt(SOL_SOCKET, SO_PRIORITY, (WebRtc_Word8*) &PCP,
- sizeof(PCP)))
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Could not SetSockopt PCP value on RTCP socket");
- _lastError = kPcpError;
- return -1;
- }
-#else
- // Not supported on other platforms (WEBRTC_MAC)
- _lastError = kPcpError;
- return -1;
-#endif
- _pcp = PCP;
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::PCP(WebRtc_Word32& PCP) const
-{
- CriticalSectionScoped cs(_crit);
- PCP = _pcp;
- return 0;
-}
-
-bool UdpTransportImpl::SetSockOptUsed()
-{
- return _useSetSockOpt;
-}
-
-WebRtc_Word32 UdpTransportImpl::EnableIpV6() {
-
- CriticalSectionScoped cs(_crit);
- const bool initialized = (_ptrSendRtpSocket || _ptrRtpSocket);
-
- if (_ipV6Enabled) {
- return 0;
- }
- if (initialized) {
- _lastError = kIpVersion6Error;
- return -1;
- }
- _ipV6Enabled = true;
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::FilterIP(
- char filterIPAddress[kIpAddressVersion6Length]) const
-{
-
- if(filterIPAddress == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "FilterIP: Invalid argument");
- return -1;
- }
- if(_filterIPAddress._sockaddr_storage.sin_family == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id, "No Filter configured");
- return -1;
- }
- CriticalSectionScoped cs(_critFilter);
- WebRtc_UWord32 ipSize = kIpAddressVersion6Length;
- WebRtc_UWord16 sourcePort;
- return IPAddress(_filterIPAddress, filterIPAddress, ipSize, sourcePort);
-}
-
-WebRtc_Word32 UdpTransportImpl::SetFilterIP(
- const char filterIPAddress[kIpAddressVersion6Length])
-{
- if(filterIPAddress == NULL)
- {
- memset(&_filterIPAddress, 0, sizeof(_filterIPAddress));
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id, "Filter IP reset");
- return 0;
- }
- CriticalSectionScoped cs(_critFilter);
- if (_ipV6Enabled)
- {
- _filterIPAddress._sockaddr_storage.sin_family = AF_INET6;
-
- if (InetPresentationToNumeric(
- AF_INET6,
- filterIPAddress,
- &_filterIPAddress._sockaddr_in6.sin6_addr) < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id, "Failed to set\
- filter IP for IPv6");
- _lastError = FILTER_ERROR;
- return -1;
- }
- }
- else
- {
- _filterIPAddress._sockaddr_storage.sin_family = AF_INET;
-
- if(InetPresentationToNumeric(
- AF_INET,
- filterIPAddress,
- &_filterIPAddress._sockaddr_in.sin_addr) < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Failed to set filter IP for IPv4");
- _lastError = FILTER_ERROR;
- return -1;
- }
- }
- WEBRTC_TRACE(kTraceDebug, kTraceTransport, _id, "Filter IP set");
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::SetFilterPorts(WebRtc_UWord16 rtpFilterPort,
- WebRtc_UWord16 rtcpFilterPort)
-{
- CriticalSectionScoped cs(_critFilter);
- _rtpFilterPort = rtpFilterPort;
- _rtcpFilterPort = rtcpFilterPort;
- return 0;
-}
-
-bool UdpTransportImpl::SendSocketsInitialized() const
-{
- CriticalSectionScoped cs(_crit);
- if(_ptrSendRtpSocket)
- {
- return true;
- }
- if(_destPort !=0)
- {
- return true;
- }
- return false;
-}
-
-bool UdpTransportImpl::ReceiveSocketsInitialized() const
-{
- if(_ptrRtpSocket)
- {
- return true;
- }
- return false;
-}
-
-bool UdpTransportImpl::SourcePortsInitialized() const
-{
- if(_ptrSendRtpSocket)
- {
- return true;
- }
- return false;
-}
-
-bool UdpTransportImpl::IpV6Enabled() const
-{
- WEBRTC_TRACE(kTraceStream, kTraceTransport, _id, "%s", __FUNCTION__);
- return _ipV6Enabled;
-}
-
-void UdpTransportImpl::BuildRemoteRTPAddr()
-{
- if(_ipV6Enabled)
- {
-#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- _remoteRTPAddr.sin_length = 0;
- _remoteRTPAddr.sin_family = PF_INET6;
-#else
- _remoteRTPAddr._sockaddr_storage.sin_family = PF_INET6;
-#endif
-
- _remoteRTPAddr._sockaddr_in6.sin6_flowinfo=0;
- _remoteRTPAddr._sockaddr_in6.sin6_scope_id=0;
- _remoteRTPAddr._sockaddr_in6.sin6_port = Htons(_destPort);
- InetPresentationToNumeric(AF_INET6,_destIP,
- &_remoteRTPAddr._sockaddr_in6.sin6_addr);
- } else
- {
-#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- _remoteRTPAddr.sin_length = 0;
- _remoteRTPAddr.sin_family = PF_INET;
-#else
- _remoteRTPAddr._sockaddr_storage.sin_family = PF_INET;
-#endif
- _remoteRTPAddr._sockaddr_in.sin_port = Htons(_destPort);
- _remoteRTPAddr._sockaddr_in.sin_addr = InetAddrIPV4(_destIP);
- }
-}
-
-void UdpTransportImpl::BuildRemoteRTCPAddr()
-{
- if(_ipV6Enabled)
- {
-#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- _remoteRTCPAddr.sin_length = 0;
- _remoteRTCPAddr.sin_family = PF_INET6;
-#else
- _remoteRTCPAddr._sockaddr_storage.sin_family = PF_INET6;
-#endif
-
- _remoteRTCPAddr._sockaddr_in6.sin6_flowinfo=0;
- _remoteRTCPAddr._sockaddr_in6.sin6_scope_id=0;
- _remoteRTCPAddr._sockaddr_in6.sin6_port = Htons(_destPortRTCP);
- InetPresentationToNumeric(AF_INET6,_destIP,
- &_remoteRTCPAddr._sockaddr_in6.sin6_addr);
-
- } else
- {
-#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- _remoteRTCPAddr.sin_length = 0;
- _remoteRTCPAddr.sin_family = PF_INET;
-#else
- _remoteRTCPAddr._sockaddr_storage.sin_family = PF_INET;
-#endif
- _remoteRTCPAddr._sockaddr_in.sin_port = Htons(_destPortRTCP);
- _remoteRTCPAddr._sockaddr_in.sin_addr= InetAddrIPV4(_destIP);
- }
-}
-
-UdpTransportImpl::ErrorCode UdpTransportImpl::BindRTPSendSocket()
-{
- if(!_ptrSendRtpSocket)
- {
- return kSocketInvalid;
- }
- if(!_ptrSendRtpSocket->ValidHandle())
- {
- return kIpAddressInvalid;
- }
- if(_ipV6Enabled)
- {
-#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- _localRTPAddr.sin_length = 0;
- _localRTPAddr.sin_family = PF_INET6;
-#else
- _localRTPAddr._sockaddr_storage.sin_family = PF_INET6;
-#endif
- _localRTPAddr._sockaddr_in6.sin6_flowinfo=0;
- _localRTPAddr._sockaddr_in6.sin6_scope_id=0;
- _localRTPAddr._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[0] =
- 0; // = INADDR_ANY
- _localRTPAddr._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[1] =
- 0;
- _localRTPAddr._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[2] =
- 0;
- _localRTPAddr._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[3] =
- 0;
- _localRTPAddr._sockaddr_in6.sin6_port = Htons(_srcPort);
- if(_ptrSendRtpSocket->Bind(_localRTPAddr) == false)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
- "Failed to bind to port:%d ", _srcPort);
- return kFailedToBindPort;
- }
- } else {
-#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- _localRTPAddr.sin_length = 0;
- _localRTPAddr.sin_family = PF_INET;
-#else
- _localRTPAddr._sockaddr_storage.sin_family = PF_INET;
-#endif
- _localRTPAddr._sockaddr_in.sin_addr = 0;
- _localRTPAddr._sockaddr_in.sin_port = Htons(_srcPort);
- if(_ptrSendRtpSocket->Bind(_localRTPAddr) == false)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
- "Failed to bind to port:%d ", _srcPort);
- return kFailedToBindPort;
- }
- }
- return kNoSocketError;
-}
-
-UdpTransportImpl::ErrorCode UdpTransportImpl::BindRTCPSendSocket()
-{
- if(!_ptrSendRtcpSocket)
- {
- return kSocketInvalid;
- }
-
- if(_ipV6Enabled)
- {
-#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- _localRTCPAddr.sin_length = 0;
- _localRTCPAddr.sin_family = PF_INET6;
-#else
- _localRTCPAddr._sockaddr_storage.sin_family = PF_INET6;
-#endif
- _localRTCPAddr._sockaddr_in6.sin6_flowinfo=0;
- _localRTCPAddr._sockaddr_in6.sin6_scope_id=0;
- _localRTCPAddr._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[0] =
- 0; // = INADDR_ANY
- _localRTCPAddr._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[1] =
- 0;
- _localRTCPAddr._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[2] =
- 0;
- _localRTCPAddr._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[3] =
- 0;
- _localRTCPAddr._sockaddr_in6.sin6_port = Htons(_srcPortRTCP);
- if(_ptrSendRtcpSocket->Bind(_localRTCPAddr) == false)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
- "Failed to bind to port:%d ", _srcPortRTCP);
- return kFailedToBindPort;
- }
- } else {
-#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- _localRTCPAddr.sin_length = 0;
- _localRTCPAddr.sin_family = PF_INET;
-#else
- _localRTCPAddr._sockaddr_storage.sin_family = PF_INET;
-#endif
- _localRTCPAddr._sockaddr_in.sin_addr= 0;
- _localRTCPAddr._sockaddr_in.sin_port = Htons(_srcPortRTCP);
- if(_ptrSendRtcpSocket->Bind(_localRTCPAddr) == false)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
- "Failed to bind to port:%d ", _srcPortRTCP);
- return kFailedToBindPort;
- }
- }
- return kNoSocketError;
-}
-
-UdpTransportImpl::ErrorCode UdpTransportImpl::BindLocalRTPSocket()
-{
- if(!_ptrRtpSocket)
- {
- return kSocketInvalid;
- }
- if(!IpV6Enabled())
- {
- SocketAddress recAddr;
- memset(&recAddr, 0, sizeof(SocketAddress));
- recAddr._sockaddr_storage.sin_family = AF_INET;
-#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- recAddr.sin_length = 0;
- recAddr.sin_family = PF_INET;
-#else
- recAddr._sockaddr_storage.sin_family = PF_INET;
-#endif
- recAddr._sockaddr_in.sin_addr = InetAddrIPV4(_localIP);
- recAddr._sockaddr_in.sin_port = Htons(_localPort);
-
- if (!_ptrRtpSocket->Bind(recAddr))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
- "Failed to bind to port:%d ", _localPort);
- return kFailedToBindPort;
- }
- }
- else
- {
- SocketAddress stLclName;
-#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- stLclName.sin_lenght = 0;
- stLclName.sin_family = PF_INET6;
-#else
- stLclName._sockaddr_storage.sin_family = PF_INET6;
-#endif
- InetPresentationToNumeric(AF_INET6,_localIP,
- &stLclName._sockaddr_in6.sin6_addr);
- stLclName._sockaddr_in6.sin6_port = Htons(_localPort);
- stLclName._sockaddr_in6.sin6_flowinfo = 0;
- stLclName._sockaddr_in6.sin6_scope_id = 0;
-
- if (!_ptrRtpSocket->Bind(stLclName))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
- "Failed to bind to port:%d ", _localPort);
- return kFailedToBindPort;
- }
- }
-
- if(_localMulticastIP[0] != 0)
- {
- // Join the multicast group from which to receive datagrams.
- struct ip_mreq mreq;
- mreq.imr_multiaddr.s_addr = InetAddrIPV4(_localMulticastIP);
- mreq.imr_interface.s_addr = INADDR_ANY;
-
- if (!_ptrRtpSocket->SetSockopt(IPPROTO_IP,IP_ADD_MEMBERSHIP,
- (WebRtc_Word8*)&mreq,sizeof (mreq)))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "setsockopt() for multicast failed, not closing socket");
- }else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceTransport, _id,
- "multicast group successfully joined");
- }
- }
- return kNoSocketError;
-}
-
-UdpTransportImpl::ErrorCode UdpTransportImpl::BindLocalRTCPSocket()
-{
- if(!_ptrRtcpSocket)
- {
- return kSocketInvalid;
- }
- if(! IpV6Enabled())
- {
- SocketAddress recAddr;
- memset(&recAddr, 0, sizeof(SocketAddress));
-#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- recAddr.sin_length = 0;
- recAddr.sin_family = AF_INET;
-#else
- recAddr._sockaddr_storage.sin_family = AF_INET;
-#endif
- recAddr._sockaddr_in.sin_addr = InetAddrIPV4(_localIP);
- recAddr._sockaddr_in.sin_port = Htons(_localPortRTCP);
-
- if (!_ptrRtcpSocket->Bind(recAddr))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
- "Failed to bind to port:%d ", _localPortRTCP);
- return kFailedToBindPort;
- }
- }
- else
- {
- SocketAddress stLclName;
-#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- stLclName.sin_length = 0;
- stLclName.sin_family = PF_INET6;
-#else
- stLclName._sockaddr_storage.sin_family = PF_INET6;
-#endif
- stLclName._sockaddr_in6.sin6_flowinfo = 0;
- stLclName._sockaddr_in6.sin6_scope_id = 0;
- stLclName._sockaddr_in6.sin6_port = Htons(_localPortRTCP);
-
- InetPresentationToNumeric(AF_INET6,_localIP,
- &stLclName._sockaddr_in6.sin6_addr);
- if (!_ptrRtcpSocket->Bind(stLclName))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, _id,
- "Failed to bind to port:%d ", _localPortRTCP);
- return kFailedToBindPort;
- }
- }
- if(_localMulticastIP[0] != 0)
- {
- // Join the multicast group from which to receive datagrams.
- struct ip_mreq mreq;
- mreq.imr_multiaddr.s_addr = InetAddrIPV4(_localMulticastIP);
- mreq.imr_interface.s_addr = INADDR_ANY;
-
- if (!_ptrRtcpSocket->SetSockopt(IPPROTO_IP,IP_ADD_MEMBERSHIP,
- (WebRtc_Word8*)&mreq,sizeof (mreq)))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "setsockopt() for multicast failed, not closing socket");
- }else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceTransport, _id,
- "multicast group successfully joined");
- }
- }
- return kNoSocketError;
-}
-
-WebRtc_Word32 UdpTransportImpl::InitializeSourcePorts(WebRtc_UWord16 rtpPort,
- WebRtc_UWord16 rtcpPort)
-{
-
- if(rtpPort == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "InitializeSourcePorts port 0 not allowed");
- _lastError = kPortInvalid;
- return -1;
- }
-
- CriticalSectionScoped cs(_crit);
-
- CloseSendSockets();
-
- if(_mgr == NULL)
- {
- return -1;
- }
-
- _srcPort = rtpPort;
- if(rtcpPort == 0)
- {
- _srcPortRTCP = rtpPort+1;
- } else
- {
- _srcPortRTCP = rtcpPort;
- }
- _useSetSockOpt =false;
- _tos=0;
- _pcp=0;
-
- _ptrSendRtpSocket = UdpSocketWrapper::CreateSocket(_id, _mgr, NULL, NULL,
- IpV6Enabled());
- _ptrSendRtcpSocket = UdpSocketWrapper::CreateSocket(_id, _mgr, NULL, NULL,
- IpV6Enabled());
-
- ErrorCode retVal = BindRTPSendSocket();
- if(retVal != kNoSocketError)
- {
- _lastError = retVal;
- return -1;
- }
- retVal = BindRTCPSendSocket();
- if(retVal != kNoSocketError)
- {
- _lastError = retVal;
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::SourcePorts(WebRtc_UWord16& rtpPort,
- WebRtc_UWord16& rtcpPort) const
-{
- CriticalSectionScoped cs(_crit);
-
- rtpPort = (_srcPort != 0) ? _srcPort : _localPort;
- rtcpPort = (_srcPortRTCP != 0) ? _srcPortRTCP : _localPortRTCP;
- return 0;
-}
-
-
-#ifdef _WIN32
-WebRtc_Word32 UdpTransportImpl::StartReceiving(
- WebRtc_UWord32 numberOfSocketBuffers)
-#else
-WebRtc_Word32 UdpTransportImpl::StartReceiving(
- WebRtc_UWord32 /*numberOfSocketBuffers*/)
-#endif
-{
- CriticalSectionScoped cs(_crit);
- if(_receiving)
- {
- return 0;
- }
- if(_ptrRtpSocket)
- {
-#ifdef _WIN32
- if(!_ptrRtpSocket->StartReceiving(numberOfSocketBuffers))
-#else
- if(!_ptrRtpSocket->StartReceiving())
-#endif
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Failed to start receive on RTP socket");
- _lastError = kStartReceiveError;
- return -1;
- }
- }
- if(_ptrRtcpSocket)
- {
- if(!_ptrRtcpSocket->StartReceiving())
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Failed to start receive on RTCP socket");
- _lastError = kStartReceiveError;
- return -1;
- }
- }
- if( _ptrRtpSocket == NULL &&
- _ptrRtcpSocket == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Failed to StartReceiving, no socket initialized");
- _lastError = kStartReceiveError;
- return -1;
- }
- _receiving = true;
- return 0;
-}
-
-bool UdpTransportImpl::Receiving() const
-{
- return _receiving;
-}
-
-WebRtc_Word32 UdpTransportImpl::StopReceiving()
-{
-
- CriticalSectionScoped cs(_crit);
-
- _receiving = false;
-
- if (_ptrRtpSocket)
- {
- if (!_ptrRtpSocket->StopReceiving())
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Failed to stop receiving on RTP socket");
- _lastError = kStopReceiveError;
- return -1;
- }
- }
- if (_ptrRtcpSocket)
- {
- if (!_ptrRtcpSocket->StopReceiving())
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "Failed to stop receiving on RTCP socket");
- _lastError = kStopReceiveError;
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::InitializeSendSockets(
- const char* ipaddr,
- const WebRtc_UWord16 rtpPort,
- const WebRtc_UWord16 rtcpPort)
-{
- {
- CriticalSectionScoped cs(_crit);
- _destPort = rtpPort;
- if(rtcpPort == 0)
- {
- _destPortRTCP = _destPort+1;
- } else
- {
- _destPortRTCP = rtcpPort;
- }
-
- if(ipaddr == NULL)
- {
- if (!IsIpAddressValid(_destIP, IpV6Enabled()))
- {
- _destPort = 0;
- _destPortRTCP = 0;
- _lastError = kIpAddressInvalid;
- return -1;
- }
- } else
- {
- if (IsIpAddressValid(ipaddr, IpV6Enabled()))
- {
- strncpy(
- _destIP,
- ipaddr,
- IpV6Enabled() ? kIpAddressVersion6Length :
- kIpAddressVersion4Length);
- } else {
- _destPort = 0;
- _destPortRTCP = 0;
- _lastError = kIpAddressInvalid;
- return -1;
- }
- }
- BuildRemoteRTPAddr();
- BuildRemoteRTCPAddr();
- }
-
- if (_ipV6Enabled)
- {
- if (_qos)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceTransport,
- _id,
- "QOS is enabled but will be ignored since IPv6 is enabled");
- }
- }else
- {
- // TODO (grunell): Multicast support is experimantal.
-
- // Put the first digit of the remote address in val.
- WebRtc_Word32 val = ntohl(_remoteRTPAddr._sockaddr_in.sin_addr)>> 24;
-
- if((val > 223) && (val < 240))
- {
- // Multicast address.
- CriticalSectionScoped cs(_crit);
-
- UdpSocketWrapper* rtpSock = (_ptrSendRtpSocket ?
- _ptrSendRtpSocket : _ptrRtpSocket);
- if (!rtpSock || !rtpSock->ValidHandle())
- {
- _lastError = kSocketInvalid;
- return -1;
- }
- UdpSocketWrapper* rtcpSock = (_ptrSendRtcpSocket ?
- _ptrSendRtcpSocket : _ptrRtcpSocket);
- if (!rtcpSock || !rtcpSock->ValidHandle())
- {
- _lastError = kSocketInvalid;
- return -1;
- }
-
- // Set Time To Live to same region
- WebRtc_Word32 iOptVal = 64;
- if (!rtpSock->SetSockopt(IPPROTO_IP, IP_MULTICAST_TTL,
- (WebRtc_Word8*)&iOptVal,
- sizeof (WebRtc_Word32)))
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "setsockopt for multicast error on RTP socket");
- _ptrRtpSocket->CloseBlocking();
- _ptrRtpSocket = NULL;
- _lastError = kMulticastAddressInvalid;
- return -1;
- }
- if (!rtcpSock->SetSockopt(IPPROTO_IP, IP_MULTICAST_TTL,
- (WebRtc_Word8*)&iOptVal,
- sizeof (WebRtc_Word32)))
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "setsockopt for multicast error on RTCP socket");
- _ptrRtpSocket->CloseBlocking();
- _ptrRtpSocket = NULL;
- _lastError = kMulticastAddressInvalid;
- return -1;
- }
- }
- }
- return 0;
-}
-
-void UdpTransportImpl::BuildSockaddrIn(WebRtc_UWord16 portnr,
- const char* ip,
- SocketAddress& remoteAddr) const
-{
- if(_ipV6Enabled)
- {
-#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- remoteAddr.sin_length = 0;
- remoteAddr.sin_family = PF_INET6;
-#else
- remoteAddr._sockaddr_storage.sin_family = PF_INET6;
-#endif
- remoteAddr._sockaddr_in6.sin6_port = Htons(portnr);
- InetPresentationToNumeric(AF_INET6, ip,
- &remoteAddr._sockaddr_in6.sin6_addr);
- remoteAddr._sockaddr_in6.sin6_flowinfo=0;
- remoteAddr._sockaddr_in6.sin6_scope_id=0;
- } else
- {
-#ifdef HAVE_STRUCT_SOCKADDR_SA_LEN
- remoteAddr.sin_length = 0;
- remoteAddr.sin_family = PF_INET;
-#else
- remoteAddr._sockaddr_storage.sin_family = PF_INET;
-#endif
- remoteAddr._sockaddr_in.sin_port = Htons(portnr);
- remoteAddr._sockaddr_in.sin_addr= InetAddrIPV4(
- const_cast<char*>(ip));
- }
-}
-
-WebRtc_Word32 UdpTransportImpl::SendRaw(const WebRtc_Word8 *data,
- WebRtc_UWord32 length,
- WebRtc_Word32 isRTCP,
- WebRtc_UWord16 portnr,
- const char* ip)
-{
- CriticalSectionScoped cs(_crit);
- if(isRTCP)
- {
- UdpSocketWrapper* rtcpSock = NULL;
- if(_ptrSendRtcpSocket)
- {
- rtcpSock = _ptrSendRtcpSocket;
- } else if(_ptrRtcpSocket)
- {
- rtcpSock = _ptrRtcpSocket;
- } else
- {
- return -1;
- }
- if(portnr == 0 && ip == NULL)
- {
- return rtcpSock->SendTo(data,length,_remoteRTCPAddr);
-
- } else if(portnr != 0 && ip != NULL)
- {
- SocketAddress remoteAddr;
- BuildSockaddrIn(portnr, ip, remoteAddr);
- return rtcpSock->SendTo(data,length,remoteAddr);
- } else if(ip != NULL)
- {
- SocketAddress remoteAddr;
- BuildSockaddrIn(_destPortRTCP, ip, remoteAddr);
- return rtcpSock->SendTo(data,length,remoteAddr);
- } else
- {
- SocketAddress remoteAddr;
- BuildSockaddrIn(portnr, _destIP, remoteAddr);
- return rtcpSock->SendTo(data,length,remoteAddr);
- }
- } else {
- UdpSocketWrapper* rtpSock = NULL;
- if(_ptrSendRtpSocket)
- {
- rtpSock = _ptrSendRtpSocket;
-
- } else if(_ptrRtpSocket)
- {
- rtpSock = _ptrRtpSocket;
- } else
- {
- return -1;
- }
- if(portnr == 0 && ip == NULL)
- {
- return rtpSock->SendTo(data,length,_remoteRTPAddr);
-
- } else if(portnr != 0 && ip != NULL)
- {
- SocketAddress remoteAddr;
- BuildSockaddrIn(portnr, ip, remoteAddr);
- return rtpSock->SendTo(data,length,remoteAddr);
- } else if(ip != NULL)
- {
- SocketAddress remoteAddr;
- BuildSockaddrIn(_destPort, ip, remoteAddr);
- return rtpSock->SendTo(data,length,remoteAddr);
- } else
- {
- SocketAddress remoteAddr;
- BuildSockaddrIn(portnr, _destIP, remoteAddr);
- return rtpSock->SendTo(data,length,remoteAddr);
- }
- }
-}
-
-WebRtc_Word32 UdpTransportImpl::SendRTPPacketTo(const WebRtc_Word8* data,
- WebRtc_UWord32 length,
- const SocketAddress& to)
-{
- CriticalSectionScoped cs(_crit);
- if(_ptrSendRtpSocket)
- {
- return _ptrSendRtpSocket->SendTo(data,length,to);
-
- } else if(_ptrRtpSocket)
- {
- return _ptrRtpSocket->SendTo(data,length,to);
- }
- return -1;
-}
-
-WebRtc_Word32 UdpTransportImpl::SendRTCPPacketTo(const WebRtc_Word8* data,
- WebRtc_UWord32 length,
- const SocketAddress& to)
-{
-
- CriticalSectionScoped cs(_crit);
-
- if(_ptrSendRtcpSocket)
- {
- return _ptrSendRtcpSocket->SendTo(data,length,to);
-
- } else if(_ptrRtcpSocket)
- {
- return _ptrRtcpSocket->SendTo(data,length,to);
- }
- return -1;
-}
-
-WebRtc_Word32 UdpTransportImpl::SendRTPPacketTo(const WebRtc_Word8* data,
- WebRtc_UWord32 length,
- const WebRtc_UWord16 rtpPort)
-{
-
- CriticalSectionScoped cs(_crit);
- // Use the current SocketAdress but update it with rtpPort.
- SocketAddress to;
- memcpy(&to, &_remoteRTPAddr, sizeof(SocketAddress));
-
- if(_ipV6Enabled)
- {
- to._sockaddr_in6.sin6_port = Htons(rtpPort);
- } else
- {
- to._sockaddr_in.sin_port = Htons(rtpPort);
- }
-
- if(_ptrSendRtpSocket)
- {
- return _ptrSendRtpSocket->SendTo(data,length,to);
-
- } else if(_ptrRtpSocket)
- {
- return _ptrRtpSocket->SendTo(data,length,to);
- }
- return -1;
-}
-
-WebRtc_Word32 UdpTransportImpl::SendRTCPPacketTo(const WebRtc_Word8* data,
- WebRtc_UWord32 length,
- const WebRtc_UWord16 rtcpPort)
-{
- CriticalSectionScoped cs(_crit);
-
- // Use the current SocketAdress but update it with rtcpPort.
- SocketAddress to;
- memcpy(&to, &_remoteRTCPAddr, sizeof(SocketAddress));
-
- if(_ipV6Enabled)
- {
- to._sockaddr_in6.sin6_port = Htons(rtcpPort);
- } else
- {
- to._sockaddr_in.sin_port = Htons(rtcpPort);
- }
-
- if(_ptrSendRtcpSocket)
- {
- return _ptrSendRtcpSocket->SendTo(data,length,to);
-
- } else if(_ptrRtcpSocket)
- {
- return _ptrRtcpSocket->SendTo(data,length,to);
- }
- return -1;
-}
-
-int UdpTransportImpl::SendPacket(int /*channel*/, const void* data, int length)
-{
- WEBRTC_TRACE(kTraceStream, kTraceTransport, _id, "%s", __FUNCTION__);
-
- CriticalSectionScoped cs(_crit);
-
- if(_destIP[0] == 0)
- {
- return -1;
- }
- if(_destPort == 0)
- {
- return -1;
- }
-
- // Create socket if it hasn't been set up already.
- // TODO (hellner): why not fail here instead. Sockets not being initialized
- // indicates that there is a problem somewhere.
- if( _ptrSendRtpSocket == NULL &&
- _ptrRtpSocket == NULL)
- {
- WEBRTC_TRACE(
- kTraceStateInfo,
- kTraceTransport,
- _id,
- "Creating RTP socket since no receive or source socket is\
- configured");
-
- _ptrRtpSocket = UdpSocketWrapper::CreateSocket(_id, _mgr, this,
- IncomingRTPCallback,
- IpV6Enabled());
-
- // Don't bind to a specific IP address.
- if(! IpV6Enabled())
- {
- strncpy(_localIP, "0.0.0.0",16);
- } else
- {
- strncpy(_localIP, "0000:0000:0000:0000:0000:0000:0000:0000",
- kIpAddressVersion6Length);
- }
- _localPort = _destPort;
-
- ErrorCode retVal = BindLocalRTPSocket();
- if(retVal != kNoSocketError)
- {
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "SendPacket() failed to bind RTP socket");
- _lastError = retVal;
- CloseReceiveSockets();
- return -1;
- }
- }
-
- if(_ptrSendRtpSocket)
- {
- return _ptrSendRtpSocket->SendTo((const WebRtc_Word8*)data, length,
- _remoteRTPAddr);
-
- } else if(_ptrRtpSocket)
- {
- return _ptrRtpSocket->SendTo((const WebRtc_Word8*)data, length,
- _remoteRTPAddr);
- }
- return -1;
-}
-
-int UdpTransportImpl::SendRTCPPacket(int /*channel*/, const void* data,
- int length)
-{
-
- CriticalSectionScoped cs(_crit);
- if(_destIP[0] == 0)
- {
- return -1;
- }
- if(_destPortRTCP == 0)
- {
- return -1;
- }
-
- // Create socket if it hasn't been set up already.
- // TODO (hellner): why not fail here instead. Sockets not being initialized
- // indicates that there is a problem somewhere.
- if( _ptrSendRtcpSocket == NULL &&
- _ptrRtcpSocket == NULL)
- {
- WEBRTC_TRACE(
- kTraceStateInfo,
- kTraceTransport,
- _id,
- "Creating RTCP socket since no receive or source socket is\
- configured");
-
- _ptrRtcpSocket = UdpSocketWrapper::CreateSocket(_id, _mgr, this,
- IncomingRTCPCallback,
- IpV6Enabled());
-
- // Don't bind to a specific IP address.
- if(! IpV6Enabled())
- {
- strncpy(_localIP, "0.0.0.0",16);
- } else
- {
- strncpy(_localIP, "0000:0000:0000:0000:0000:0000:0000:0000",
- kIpAddressVersion6Length);
- }
- _localPortRTCP = _destPortRTCP;
-
- ErrorCode retVal = BindLocalRTCPSocket();
- if(retVal != kNoSocketError)
- {
- _lastError = retVal;
- WEBRTC_TRACE(kTraceError, kTraceTransport, _id,
- "SendRTCPPacket() failed to bind RTCP socket");
- CloseReceiveSockets();
- return -1;
- }
- }
-
- if(_ptrSendRtcpSocket)
- {
- return _ptrSendRtcpSocket->SendTo((const WebRtc_Word8*)data, length,
- _remoteRTCPAddr);
- } else if(_ptrRtcpSocket)
- {
- return _ptrRtcpSocket->SendTo((const WebRtc_Word8*)data, length,
- _remoteRTCPAddr);
- }
- return -1;
-}
-
-WebRtc_Word32 UdpTransportImpl::SetSendIP(const char* ipaddr)
-{
- if(!IsIpAddressValid(ipaddr,IpV6Enabled()))
- {
- return kIpAddressInvalid;
- }
- CriticalSectionScoped cs(_crit);
- strncpy(_destIP, ipaddr,kIpAddressVersion6Length);
- BuildRemoteRTPAddr();
- BuildRemoteRTCPAddr();
- return 0;
-}
-
-WebRtc_Word32 UdpTransportImpl::SetSendPorts(WebRtc_UWord16 rtpPort,
- WebRtc_UWord16 rtcpPort)
-{
- CriticalSectionScoped cs(_crit);
- _destPort = rtpPort;
- if(rtcpPort == 0)
- {
- _destPortRTCP = _destPort+1;
- } else
- {
- _destPortRTCP = rtcpPort;
- }
- BuildRemoteRTPAddr();
- BuildRemoteRTCPAddr();
- return 0;
-}
-
-void UdpTransportImpl::IncomingRTPCallback(CallbackObj obj,
- const WebRtc_Word8* rtpPacket,
- WebRtc_Word32 rtpPacketLength,
- const SocketAddress* from)
-{
- if (rtpPacket && rtpPacketLength > 0)
- {
- UdpTransportImpl* socketTransport = (UdpTransportImpl*) obj;
- socketTransport->IncomingRTPFunction(rtpPacket, rtpPacketLength, from);
- }
-}
-
-void UdpTransportImpl::IncomingRTCPCallback(CallbackObj obj,
- const WebRtc_Word8* rtcpPacket,
- WebRtc_Word32 rtcpPacketLength,
- const SocketAddress* from)
-{
- if (rtcpPacket && rtcpPacketLength > 0)
- {
- UdpTransportImpl* socketTransport = (UdpTransportImpl*) obj;
- socketTransport->IncomingRTCPFunction(rtcpPacket, rtcpPacketLength,
- from);
- }
-}
-
-void UdpTransportImpl::IncomingRTPFunction(const WebRtc_Word8* rtpPacket,
- WebRtc_Word32 rtpPacketLength,
- const SocketAddress* fromSocket)
-{
- char ipAddress[kIpAddressVersion6Length];
- WebRtc_UWord32 ipAddressLength = kIpAddressVersion6Length;
- WebRtc_UWord16 portNr = 0;
-
- {
- CriticalSectionScoped cs(_critFilter);
- if (FilterIPAddress(fromSocket) == false)
- {
- // Packet should be filtered out. Drop it.
- WEBRTC_TRACE(kTraceStream, kTraceTransport, _id,
- "Incoming RTP packet blocked by IP filter");
- return;
- }
-
- if (IPAddressCached(*fromSocket, ipAddress, ipAddressLength, portNr) <
- 0)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpTransportImpl::IncomingRTPFunction - Cannot get sender\
- information");
- }else
- {
- strncpy(_fromIP, ipAddress, kIpAddressVersion6Length);
- }
-
- // Filter based on port.
- if (_rtpFilterPort != 0 &&
- _rtpFilterPort != portNr)
- {
- // Drop packet.
- memset(_fromIP, 0, sizeof(_fromIP));
- WEBRTC_TRACE(
- kTraceStream,
- kTraceTransport,
- _id,
- "Incoming RTP packet blocked by filter incoming from port:%d\
- allowed port:%d",
- portNr,
- _rtpFilterPort);
- return;
- }
- _fromPort = portNr;
- }
-
- CriticalSectionScoped cs(_critPacketCallback);
- if (_packetCallback)
- {
- WEBRTC_TRACE(kTraceStream, kTraceTransport, _id,
- "Incoming RTP packet from ip:%s port:%d", ipAddress, portNr);
- _packetCallback->IncomingRTPPacket(rtpPacket, rtpPacketLength,
- ipAddress, portNr);
- }
-}
-
-void UdpTransportImpl::IncomingRTCPFunction(const WebRtc_Word8* rtcpPacket,
- WebRtc_Word32 rtcpPacketLength,
- const SocketAddress* fromSocket)
-{
- char ipAddress[kIpAddressVersion6Length];
- WebRtc_UWord32 ipAddressLength = kIpAddressVersion6Length;
- WebRtc_UWord16 portNr = 0;
-
- {
- CriticalSectionScoped cs(_critFilter);
- if (FilterIPAddress(fromSocket) == false)
- {
- // Packet should be filtered out. Drop it.
- WEBRTC_TRACE(kTraceStream, kTraceTransport, _id,
- "Incoming RTCP packet blocked by IP filter");
- return;
- }
- if (IPAddress(*fromSocket, ipAddress, ipAddressLength, portNr) < 0)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpTransportImpl::IncomingRTCPFunction - Cannot get sender\
- information");
- }else {
- strncpy(_fromIP, ipAddress, kIpAddressVersion6Length);
- }
-
- // Filter based on port.
- if (_rtcpFilterPort != 0 &&
- _rtcpFilterPort != portNr)
- {
- // Drop packet.
- WEBRTC_TRACE(
- kTraceStream,
- kTraceTransport,
- _id,
- "Incoming RTCP packet blocked by filter incoming from port:%d\
- allowed port:%d",
- portNr,
- _rtpFilterPort);
- return;
- }
- _fromPortRTCP = portNr;
- }
-
- CriticalSectionScoped cs(_critPacketCallback);
- if (_packetCallback)
- {
- WEBRTC_TRACE(kTraceStream, kTraceTransport, _id,
- "Incoming RTCP packet from ip:%s port:%d", ipAddress,
- portNr);
- _packetCallback->IncomingRTCPPacket(rtcpPacket, rtcpPacketLength,
- ipAddress, portNr);
- }
-}
-
-bool UdpTransportImpl::FilterIPAddress(const SocketAddress* fromAddress)
-{
- if(fromAddress->_sockaddr_storage.sin_family == AF_INET)
- {
- if (_filterIPAddress._sockaddr_storage.sin_family == AF_INET)
- {
- // IP is stored in sin_addr.
- if (_filterIPAddress._sockaddr_in.sin_addr != 0 &&
- (_filterIPAddress._sockaddr_in.sin_addr !=
- fromAddress->_sockaddr_in.sin_addr))
- {
- return false;
- }
- }
- }
- else if(fromAddress->_sockaddr_storage.sin_family == AF_INET6)
- {
- if (_filterIPAddress._sockaddr_storage.sin_family == AF_INET6)
- {
- // IP is stored in sin_6addr.
- for (WebRtc_Word32 i = 0; i < 4; i++)
- {
- if (_filterIPAddress._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[i] != 0 &&
- _filterIPAddress._sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[i] != fromAddress->_sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u32[i])
- {
- return false;
- }
- }
- }
- }
- else
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- _id,
- "UdpTransportImpl::FilterIPAddress() unknown address family");
- return false;
- }
- return true;
-}
-
-void UdpTransportImpl::CloseReceiveSockets()
-{
- if(_ptrRtpSocket)
- {
- _ptrRtpSocket->CloseBlocking();
- _ptrRtpSocket = NULL;
- }
- if(_ptrRtcpSocket)
- {
- _ptrRtcpSocket->CloseBlocking();
- _ptrRtcpSocket = NULL;
- }
- _receiving = false;
-}
-
-void UdpTransportImpl::CloseSendSockets()
-{
- if(_ptrSendRtpSocket)
- {
- _ptrSendRtpSocket->CloseBlocking();
- _ptrSendRtpSocket = 0;
- }
- if(_ptrSendRtcpSocket)
- {
- _ptrSendRtcpSocket->CloseBlocking();
- _ptrSendRtcpSocket = 0;
- }
-}
-
-WebRtc_UWord16 UdpTransport::Htons(const WebRtc_UWord16 port)
-{
- return htons(port);
-}
-
-WebRtc_UWord32 UdpTransport::Htonl(const WebRtc_UWord32 a)
-{
- return htonl(a);
-}
-
-WebRtc_UWord32 UdpTransport::InetAddrIPV4(const char* ip)
-{
- return ::inet_addr(ip);
-}
-
-WebRtc_Word32 UdpTransport::InetPresentationToNumeric(WebRtc_Word32 af,
- const char* src,
- void* dst)
-{
-#if defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
- const WebRtc_Word32 result = inet_pton(af, src, dst);
- return result > 0 ? 0 : -1;
-
-#elif defined(_WIN32)
- SocketAddress temp;
- int length=sizeof(SocketAddress);
-
- if(af == AF_INET)
- {
- WebRtc_Word32 result = WSAStringToAddressA(
- (const LPSTR)src,
- af,
- 0,
- reinterpret_cast<struct sockaddr*>(&temp),
- &length);
- if(result != 0)
- {
- return -1;
- }
- memcpy(dst,&(temp._sockaddr_in.sin_addr),
- sizeof(temp._sockaddr_in.sin_addr));
- return 0;
- }
- else if(af == AF_INET6)
- {
- WebRtc_Word32 result = WSAStringToAddressA(
- (const LPSTR)src,
- af,
- 0,
- reinterpret_cast<struct sockaddr*>(&temp),
- &length);
- if(result !=0)
- {
- return -1;
- }
- memcpy(dst,&(temp._sockaddr_in6.sin6_addr),
- sizeof(temp._sockaddr_in6.sin6_addr));
- return 0;
-
- }else
- {
- return -1;
- }
-#else
- return -1;
-#endif
-}
-
-WebRtc_Word32 UdpTransport::LocalHostAddressIPV6(char n_localIP[16])
-{
-
-#if defined(_WIN32)
- struct addrinfo *result = NULL;
- struct addrinfo *ptr = NULL;
- struct addrinfo hints;
-
- ZeroMemory(&hints, sizeof(hints));
- hints.ai_family = AF_INET6;
-
- char szHostName[256] = "";
- if(::gethostname(szHostName, sizeof(szHostName) - 1))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, -1, "gethostname failed");
- return -1;
- }
-
- DWORD dwRetval = getaddrinfo(szHostName, NULL, &hints, &result);
- if ( dwRetval != 0 )
- {
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, -1,
- "getaddrinfo failed, error:%d", dwRetval);
- return -1;
- }
- for(ptr=result; ptr != NULL ;ptr=ptr->ai_next)
- {
- switch (ptr->ai_family)
- {
- case AF_INET6:
- {
- for(int i = 0; i< 16; i++)
- {
- n_localIP[i] = (*(SocketAddress*)ptr->ai_addr).
- _sockaddr_in6.sin6_addr.Version6AddressUnion._s6_u8[i];
- }
- bool islocalIP = true;
-
- for(int n = 0; n< 15; n++)
- {
- if(n_localIP[n] != 0)
- {
- islocalIP = false;
- break;
- }
- }
-
- if(islocalIP && n_localIP[15] != 1)
- {
- islocalIP = false;
- }
-
- if(islocalIP && ptr->ai_next)
- {
- continue;
- }
- if(n_localIP[0] == 0xfe &&
- n_localIP[1] == 0x80 && ptr->ai_next)
- {
- continue;
- }
- freeaddrinfo(result);
- }
- return 0;
- default:
- break;
- };
- }
- freeaddrinfo(result);
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, -1,
- "getaddrinfo failed to find address");
- return -1;
-
-#elif defined(WEBRTC_MAC)
- struct ifaddrs* ptrIfAddrs = NULL;
- struct ifaddrs* ptrIfAddrsStart = NULL;
-
- getifaddrs(&ptrIfAddrsStart);
- ptrIfAddrs = ptrIfAddrsStart;
- while(ptrIfAddrs)
- {
- if(ptrIfAddrs->ifa_addr->sa_family == AF_INET6)
- {
- const struct sockaddr_in6* sock_in6 =
- reinterpret_cast<struct sockaddr_in6*>(ptrIfAddrs->ifa_addr);
- const struct in6_addr* sin6_addr = &sock_in6->sin6_addr;
-
- if (IN6_IS_ADDR_LOOPBACK(sin6_addr) ||
- IN6_IS_ADDR_LINKLOCAL(sin6_addr)) {
- ptrIfAddrs = ptrIfAddrs->ifa_next;
- continue;
- }
- memcpy(n_localIP, sin6_addr->s6_addr, sizeof(sin6_addr->s6_addr));
- freeifaddrs(ptrIfAddrsStart);
- return 0;
- }
- ptrIfAddrs = ptrIfAddrs->ifa_next;
- }
- freeifaddrs(ptrIfAddrsStart);
- return -1;
-#elif defined(WEBRTC_ANDROID)
- return -1;
-#else // WEBRTC_LINUX
- struct
- {
- struct nlmsghdr n;
- struct ifaddrmsg r;
- } req;
-
- struct rtattr* rta = NULL;
- int status;
- char buf[16384]; // = 16 * 1024 (16 kB)
- struct nlmsghdr* nlmp;
- struct ifaddrmsg* rtmp;
- struct rtattr* rtatp;
- int rtattrlen;
- struct in6_addr* in6p;
-
- int fd = socket(PF_NETLINK, SOCK_DGRAM, NETLINK_ROUTE);
- if (fd == -1)
- {
- return -1;
- }
-
- // RTM_GETADDR is used to fetch the ip address from the kernel interface
- // table. Populate the msg structure (req) the size of the message buffer
- // is specified to netlinkmessage header, and flags values are set as
- // NLM_F_ROOT | NLM_F_REQUEST.
- // The request flag must be set for all messages requesting the data from
- // kernel. The root flag is used to notify the kernel to return the full
- // tabel. Another flag (not used) is NLM_F_MATCH. This is used to get only
- // specified entries in the table. At the time of writing this program this
- // flag is not implemented in kernel
-
- memset(&req, 0, sizeof(req));
- req.n.nlmsg_len = NLMSG_LENGTH(sizeof(struct ifaddrmsg));
- req.n.nlmsg_flags = NLM_F_REQUEST | NLM_F_ROOT;
- req.n.nlmsg_type = RTM_GETADDR;
- req.r.ifa_family = AF_INET6;
-
- // Fill up all the attributes for the rtnetlink header.
- // The lenght is very important. 16 signifies the ipv6 address.
- rta = (struct rtattr*)(((char*)&req) + NLMSG_ALIGN(req.n.nlmsg_len));
- rta->rta_len = RTA_LENGTH(16);
-
- status = send(fd, &req, req.n.nlmsg_len, 0);
- if (status < 0)
- {
- close(fd);
- return -1;
- }
- status = recv(fd, buf, sizeof(buf), 0);
- if (status < 0)
- {
- close(fd);
- return -1;
- }
- if(status == 0)
- {
- close(fd);
- return -1;
- }
- close(fd);
-
- // The message is stored in buff. Parse the message to get the requested
- // data.
- {
- nlmp = (struct nlmsghdr*)buf;
- int len = nlmp->nlmsg_len;
- int req_len = len - sizeof(*nlmp);
-
- if (req_len < 0 || len > status)
- {
- return -1;
- }
- if (!NLMSG_OK_NO_WARNING(nlmp, status))
- {
- return -1;
- }
- rtmp = (struct ifaddrmsg*)NLMSG_DATA(nlmp);
- rtatp = (struct rtattr*)IFA_RTA(rtmp);
-
- rtattrlen = IFA_PAYLOAD(nlmp);
-
- for (; RTA_OK(rtatp, rtattrlen); rtatp = RTA_NEXT(rtatp, rtattrlen))
- {
-
- // Here we hit the fist chunk of the message. Time to validate the
- // type. For more info on the different types see
- // "man(7) rtnetlink" The table below is taken from man pages.
- // Attributes
- // rta_type value type description
- // -------------------------------------------------------------
- // IFA_UNSPEC - unspecified.
- // IFA_ADDRESS raw protocol address interface address
- // IFA_LOCAL raw protocol address local address
- // IFA_LABEL asciiz string name of the interface
- // IFA_BROADCAST raw protocol address broadcast address.
- // IFA_ANYCAST raw protocol address anycast address
- // IFA_CACHEINFO struct ifa_cacheinfo Address information.
-
- if(rtatp->rta_type == IFA_ADDRESS)
- {
- bool islocalIP = true;
- in6p = (struct in6_addr*)RTA_DATA(rtatp);
- for(int n = 0; n< 15; n++)
- {
- if(in6p->s6_addr[n] != 0)
- {
- islocalIP = false;
- break;
- }
- }
- if(islocalIP && in6p->s6_addr[15] != 1)
- {
- islocalIP = false;
- }
- if(!islocalIP)
- {
- for(int i = 0; i< 16; i++)
- {
- n_localIP[i] = in6p->s6_addr[i];
- }
- if(n_localIP[0] == static_cast<char> (0xfe)
- && n_localIP[1] == static_cast<char>(0x80) )
- {
- // Auto configured IP.
- continue;
- }
- break;
- }
- }
- }
- }
- return 0;
-#endif
-}
-
-WebRtc_Word32 UdpTransport::LocalHostAddress(WebRtc_UWord32& localIP)
-{
- #if defined(_WIN32)
- hostent* localHost;
- localHost = gethostbyname( "" );
- if(localHost)
- {
- if(localHost->h_addrtype != AF_INET)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- -1,
- "LocalHostAddress can only get local IP for IP Version 4");
- return -1;
- }
- localIP= Htonl(
- (*(struct in_addr *)localHost->h_addr_list[0]).S_un.S_addr);
- return 0;
- }
- else
- {
- WebRtc_Word32 error = WSAGetLastError();
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, -1,
- "gethostbyname failed, error:%d", error);
- return -1;
- }
-#elif (defined(WEBRTC_MAC))
- char localname[255];
- if (gethostname(localname, 255) != -1)
- {
- hostent* localHost;
- localHost = gethostbyname(localname);
- if(localHost)
- {
- if(localHost->h_addrtype != AF_INET)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceTransport,
- -1,
- "LocalHostAddress can only get local IP for IP Version 4");
- return -1;
- }
- localIP = Htonl((*(struct in_addr*)*localHost->h_addr_list).s_addr);
- return 0;
- }
- }
- WEBRTC_TRACE(kTraceWarning, kTraceTransport, -1, "gethostname failed");
- return -1;
-#else // WEBRTC_LINUX
- int sockfd, size = 1;
- struct ifreq* ifr;
- struct ifconf ifc;
-
- if (0 > (sockfd = socket(AF_INET, SOCK_DGRAM, IPPROTO_IP)))
- {
- return -1;
- }
- ifc.ifc_len = IFRSIZE;
- ifc.ifc_req = NULL;
- do
- {
- ++size;
- // Buffer size needed is unknown. Try increasing it until no overflow
- // occurs.
- if (NULL == (ifc.ifc_req = (ifreq*)realloc(ifc.ifc_req, IFRSIZE))) {
- fprintf(stderr, "Out of memory.\n");
- exit(EXIT_FAILURE);
- }
- ifc.ifc_len = IFRSIZE;
- if (ioctl(sockfd, SIOCGIFCONF, &ifc))
- {
- close(sockfd);
- return -1;
- }
- } while (IFRSIZE <= ifc.ifc_len);
-
- ifr = ifc.ifc_req;
- for (;(char *) ifr < (char *) ifc.ifc_req + ifc.ifc_len; ++ifr)
- {
- if (ifr->ifr_addr.sa_data == (ifr+1)->ifr_addr.sa_data)
- {
- continue; // duplicate, skip it
- }
- if (ioctl(sockfd, SIOCGIFFLAGS, ifr))
- {
- continue; // failed to get flags, skip it
- }
- if(strncmp(ifr->ifr_name, "lo",3) == 0)
- {
- continue;
- }else
- {
- struct sockaddr* saddr = &(ifr->ifr_addr);
- SocketAddress* socket_addess = reinterpret_cast<SocketAddress*>(
- saddr);
- localIP = Htonl(socket_addess->_sockaddr_in.sin_addr);
- close(sockfd);
- return 0;
- }
- }
- close(sockfd);
- return -1;
-#endif
-}
-
-WebRtc_Word32 UdpTransport::IPAddress(const SocketAddress& address,
- char* ip,
- WebRtc_UWord32& ipSize,
- WebRtc_UWord16& sourcePort)
-{
- #if defined(_WIN32)
- DWORD dwIPSize = ipSize;
- WebRtc_Word32 returnvalue = WSAAddressToStringA((LPSOCKADDR)(&address),
- sizeof(SocketAddress),
- NULL,
- ip,
- &dwIPSize);
- if(returnvalue == -1)
- {
- return -1;
- }
-
- WebRtc_UWord16 source_port = 0;
- if(address._sockaddr_storage.sin_family == AF_INET)
- {
- // Parse IP assuming format "a.b.c.d:port".
- WebRtc_Word8* ipEnd = strchr(ip,':');
- if(ipEnd != NULL)
- {
- *ipEnd = '\0';
- }
- ipSize = (WebRtc_Word32)strlen(ip);
- if(ipSize == 0)
- {
- return -1;
- }
- source_port = address._sockaddr_in.sin_port;
- }
- else
- {
- // Parse IP assuming format "[address]:port".
- WebRtc_Word8* ipEnd = strchr(ip,']');
- if(ipEnd != NULL)
- {
- // Calculate length
- WebRtc_Word32 adrSize = WebRtc_Word32(ipEnd - ip) - 1;
- memmove(ip, &ip[1], adrSize); // Remove '['
- *(ipEnd - 1) = '\0';
- }
- ipSize = (WebRtc_Word32)strlen(ip);
- if(ipSize == 0)
- {
- return -1;
- }
-
- source_port = address._sockaddr_in6.sin6_port;
- }
- // Convert port number to network byte order.
- sourcePort = htons(source_port);
- return 0;
-
- #elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
- WebRtc_Word32 ipFamily = address._sockaddr_storage.sin_family;
- const void* ptrNumericIP = NULL;
-
- if(ipFamily == AF_INET)
- {
- ptrNumericIP = &(address._sockaddr_in.sin_addr);
- }
- else if(ipFamily == AF_INET6)
- {
- ptrNumericIP = &(address._sockaddr_in6.sin6_addr);
- }
- else
- {
- return -1;
- }
- if(inet_ntop(ipFamily, ptrNumericIP, ip, ipSize) == NULL)
- {
- return -1;
- }
- WebRtc_UWord16 source_port;
- if(ipFamily == AF_INET)
- {
- source_port = address._sockaddr_in.sin_port;
- } else
- {
- source_port = address._sockaddr_in6.sin6_port;
- }
- // Convert port number to network byte order.
- sourcePort = htons(source_port);
- return 0;
- #else
- return -1;
- #endif
-}
-
-bool UdpTransport::IsIpAddressValid(const char* ipadr, const bool ipV6)
-{
- if(ipV6)
- {
- WebRtc_Word32 len = (WebRtc_Word32)strlen(ipadr);
- if( len>39 || len == 0)
- {
- return false;
- }
-
- WebRtc_Word32 i;
- WebRtc_Word32 colonPos[7] = {0,0,0,0,0,0,0};
- WebRtc_Word32 lastColonPos = -2;
- WebRtc_Word32 nColons = 0;
- WebRtc_Word32 nDubbleColons = 0;
- WebRtc_Word32 nDots = 0;
- WebRtc_Word32 error = 0;
- char c;
- for(i = 0; i < len ; i++)
- {
- c=ipadr[i];
- if(isxdigit(c))
- ;
- else if(c == ':')
- {
- if(nColons < 7)
- colonPos[nColons] = i;
- if((i-lastColonPos)==1)
- nDubbleColons++;
- lastColonPos=i;
- if(nDots != 0)
- {
- error = 1;
- }
- nColons++;
- }
- else if(c == '.')
- {
- nDots++;
- }
- else
- {
- error = 1;
- }
-
- }
- if(error)
- {
- return false;
- }
- if(nDubbleColons > 1)
- {
- return false;
- }
- if(nColons > 7 || nColons < 2)
- {
- return false;
- }
- if(!(nDots == 3 || nDots == 0))
- {
- return false;
- }
- lastColonPos = -1;
- WebRtc_Word32 charsBeforeColon = 0;
- for(i = 0; i < nColons; i++)
- {
- charsBeforeColon=colonPos[i]-lastColonPos-1;
- if(charsBeforeColon > 4)
- {
- return false;
- }
- lastColonPos=colonPos[i];
- }
- WebRtc_Word32 lengthAfterLastColon = len - lastColonPos - 1;
- if(nDots == 0)
- {
- if(lengthAfterLastColon > 4)
- return false;
- }
- if(nDots == 3 && lengthAfterLastColon > 0)
- {
- return IsIpAddressValid((ipadr+lastColonPos+1),false);
- }
-
- }
- else
- {
- WebRtc_Word32 len = (WebRtc_Word32)strlen(ipadr);
- if((len>15)||(len==0))
- {
- return false;
- }
-
- // IPv4 should be [0-255].[0-255].[0-255].[0-255]
- WebRtc_Word32 i;
- WebRtc_Word32 nDots = 0;
- WebRtc_Word32 iDotPos[4] = {0,0,0,0};
-
- for (i = 0; (i < len) && (nDots < 4); i++)
- {
- if (ipadr[i] == (char)'.')
- {
- // Store index of dots and count number of dots.
- iDotPos[nDots++] = i;
- }
- }
-
- bool allUnder256 = false;
- // TODO (hellner): while loop seems to be abused here to get
- // label like functionality. Fix later to avoid introducing bugs now.
-
- // Check that all numbers are smaller than 256.
- do
- {
- if (nDots != 3 )
- {
- break;
- }
-
- if (iDotPos[0] <= 3)
- {
- char nr[4];
- memset(nr,0,4);
- strncpy(nr,&ipadr[0],iDotPos[0]);
- WebRtc_Word32 num = atoi(nr);
- if (num > 255)
- {
- break;
- }
- } else {
- break;
- }
-
- if (iDotPos[1] - iDotPos[0] <= 4)
- {
- char nr[4];
- memset(nr,0,4);
- strncpy(nr,&ipadr[iDotPos[0]+1], iDotPos[1] - iDotPos[0] - 1);
- WebRtc_Word32 num = atoi(nr);
- if (num > 255)
- break;
- } else {
- break;
- }
-
- if (iDotPos[2] - iDotPos[1] <= 4)
- {
- char nr[4];
- memset(nr,0,4);
- strncpy(nr,&ipadr[iDotPos[1]+1], iDotPos[1] - iDotPos[0] - 1);
- WebRtc_Word32 num = atoi(nr);
- if (num > 255)
- break;
-
- memset(nr,0,4);
- strncpy(nr,&ipadr[iDotPos[2]+1], len - iDotPos[2] -1);
- num = atoi(nr);
- if (num > 255)
- break;
- else
- allUnder256 = true;
- } else
- break;
- } while(false);
-
- if (nDots != 3 || !allUnder256)
- {
- return false;
- }
- }
- return true;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/udp_transport/source/udp_transport_impl.h b/trunk/src/modules/udp_transport/source/udp_transport_impl.h
deleted file mode 100644
index c5a73f9..0000000
--- a/trunk/src/modules/udp_transport/source/udp_transport_impl.h
+++ /dev/null
@@ -1,246 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_TRANSPORT_IMPL_H_
-#define WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_TRANSPORT_IMPL_H_
-
-#include "udp_transport.h"
-#include "udp_socket_wrapper.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class RWLockWrapper;
-class UdpSocketManager;
-
-class UdpTransportImpl : public UdpTransport
-{
-public:
- // Factory method. Constructor disabled.
- UdpTransportImpl(const WebRtc_Word32 id, WebRtc_UWord8& numSocketThreads);
- virtual ~UdpTransportImpl();
-
- // Module functions
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
- virtual WebRtc_Word32 TimeUntilNextProcess();
- virtual WebRtc_Word32 Process();
-
- // UdpTransport functions
- virtual WebRtc_Word32 InitializeSendSockets(
- const char* ipAddr,
- const WebRtc_UWord16 rtpPort,
- const WebRtc_UWord16 rtcpPort = 0);
- virtual WebRtc_Word32 InitializeReceiveSockets(
- UdpTransportData* const packetCallback,
- const WebRtc_UWord16 rtpPort,
- const char* ipAddr = NULL,
- const char* multicastIpAddr = NULL,
- const WebRtc_UWord16 rtcpPort = 0);
- virtual WebRtc_Word32 InitializeSourcePorts(
- const WebRtc_UWord16 rtpPort,
- const WebRtc_UWord16 rtcpPort = 0);
- virtual WebRtc_Word32 SourcePorts(WebRtc_UWord16& rtpPort,
- WebRtc_UWord16& rtcpPort) const;
- virtual WebRtc_Word32 ReceiveSocketInformation(
- char ipAddr[kIpAddressVersion6Length],
- WebRtc_UWord16& rtpPort,
- WebRtc_UWord16& rtcpPort,
- char multicastIpAddr[kIpAddressVersion6Length]) const;
- virtual WebRtc_Word32 SendSocketInformation(
- char ipAddr[kIpAddressVersion6Length],
- WebRtc_UWord16& rtpPort,
- WebRtc_UWord16& rtcpPort) const;
- virtual WebRtc_Word32 RemoteSocketInformation(
- char ipAddr[kIpAddressVersion6Length],
- WebRtc_UWord16& rtpPort,
- WebRtc_UWord16& rtcpPort) const;
- virtual WebRtc_Word32 SetQoS(const bool QoS,
- const WebRtc_Word32 serviceType,
- const WebRtc_UWord32 maxBitrate = 0,
- const WebRtc_Word32 overrideDSCP = 0,
- const bool audio = false);
- virtual WebRtc_Word32 QoS(bool& QoS, WebRtc_Word32& serviceType,
- WebRtc_Word32& overrideDSCP) const;
- virtual WebRtc_Word32 SetToS(const WebRtc_Word32 DSCP,
- const bool useSetSockOpt = false);
- virtual WebRtc_Word32 ToS(WebRtc_Word32& DSCP,
- bool& useSetSockOpt) const;
- virtual WebRtc_Word32 SetPCP(const WebRtc_Word32 PCP);
- virtual WebRtc_Word32 PCP(WebRtc_Word32& PCP) const;
- virtual WebRtc_Word32 EnableIpV6();
- virtual bool IpV6Enabled() const;
- virtual WebRtc_Word32 SetFilterIP(
- const char filterIPAddress[kIpAddressVersion6Length]);
- virtual WebRtc_Word32 FilterIP(
- char filterIPAddress[kIpAddressVersion6Length]) const;
- virtual WebRtc_Word32 SetFilterPorts(const WebRtc_UWord16 rtpFilterPort,
- const WebRtc_UWord16 rtcpFilterPort);
- virtual WebRtc_Word32 FilterPorts(WebRtc_UWord16& rtpFilterPort,
- WebRtc_UWord16& rtcpFilterPort) const;
- virtual WebRtc_Word32 StartReceiving(
- const WebRtc_UWord32 numberOfSocketBuffers);
- virtual WebRtc_Word32 StopReceiving();
- virtual bool Receiving() const;
- virtual bool SendSocketsInitialized() const;
- virtual bool SourcePortsInitialized() const;
- virtual bool ReceiveSocketsInitialized() const;
- virtual WebRtc_Word32 SendRaw(const WebRtc_Word8* data,
- WebRtc_UWord32 length, WebRtc_Word32 isRTCP,
- WebRtc_UWord16 portnr = 0,
- const char* ip = NULL);
- virtual WebRtc_Word32 SendRTPPacketTo(const WebRtc_Word8 *data,
- WebRtc_UWord32 length,
- const SocketAddress& to);
- virtual WebRtc_Word32 SendRTCPPacketTo(const WebRtc_Word8 *data,
- WebRtc_UWord32 length,
- const SocketAddress& to);
- virtual WebRtc_Word32 SendRTPPacketTo(const WebRtc_Word8 *data,
- WebRtc_UWord32 length,
- WebRtc_UWord16 rtpPort);
- virtual WebRtc_Word32 SendRTCPPacketTo(const WebRtc_Word8 *data,
- WebRtc_UWord32 length,
- WebRtc_UWord16 rtcpPort);
- // Transport functions
- virtual int SendPacket(int channel, const void* data, int length);
- virtual int SendRTCPPacket(int channel, const void* data, int length);
-
- // UdpTransport functions continue.
- virtual WebRtc_Word32 SetSendIP(const char* ipaddr);
- virtual WebRtc_Word32 SetSendPorts(const WebRtc_UWord16 rtpPort,
- const WebRtc_UWord16 rtcpPort = 0);
-
- virtual ErrorCode LastError() const;
-
- virtual WebRtc_Word32 IPAddressCached(const SocketAddress& address,
- char* ip,
- WebRtc_UWord32& ipSize,
- WebRtc_UWord16& sourcePort);
-
- WebRtc_Word32 Id() const {return _id;}
-protected:
- // IncomingSocketCallback signature functions for receiving callbacks from
- // UdpSocketWrapper.
- static void IncomingRTPCallback(CallbackObj obj,
- const WebRtc_Word8* rtpPacket,
- WebRtc_Word32 rtpPacketLength,
- const SocketAddress* from);
- static void IncomingRTCPCallback(CallbackObj obj,
- const WebRtc_Word8* rtcpPacket,
- WebRtc_Word32 rtcpPacketLength,
- const SocketAddress* from);
-
- void CloseSendSockets();
- void CloseReceiveSockets();
-
- // Update _remoteRTPAddr according to _destPort and _destIP
- void BuildRemoteRTPAddr();
- // Update _remoteRTCPAddr according to _destPortRTCP and _destIP
- void BuildRemoteRTCPAddr();
-
- void BuildSockaddrIn(WebRtc_UWord16 portnr, const char* ip,
- SocketAddress& remoteAddr) const;
-
- ErrorCode BindLocalRTPSocket();
- ErrorCode BindLocalRTCPSocket();
-
- ErrorCode BindRTPSendSocket();
- ErrorCode BindRTCPSendSocket();
-
- void IncomingRTPFunction(const WebRtc_Word8* rtpPacket,
- WebRtc_Word32 rtpPacketLength,
- const SocketAddress* from);
- void IncomingRTCPFunction(const WebRtc_Word8* rtcpPacket,
- WebRtc_Word32 rtcpPacketLength,
- const SocketAddress* from);
-
- bool FilterIPAddress(const SocketAddress* fromAddress);
-
- bool SetSockOptUsed();
-
- WebRtc_Word32 EnableQoS(WebRtc_Word32 serviceType, bool audio,
- WebRtc_UWord32 maxBitrate,
- WebRtc_Word32 overrideDSCP);
-
- WebRtc_Word32 DisableQoS();
-
-private:
- void GetCachedAddress(char* ip, WebRtc_UWord32& ipSize,
- WebRtc_UWord16& sourcePort);
-
- WebRtc_Word32 _id;
- // Protects the sockets from being re-configured while receiving packets.
- CriticalSectionWrapper* _crit;
- CriticalSectionWrapper* _critFilter;
- // _packetCallback's critical section.
- CriticalSectionWrapper* _critPacketCallback;
- UdpSocketManager* _mgr;
- ErrorCode _lastError;
-
- // Remote RTP and RTCP ports.
- WebRtc_UWord16 _destPort;
- WebRtc_UWord16 _destPortRTCP;
-
- // Local RTP and RTCP ports.
- WebRtc_UWord16 _localPort;
- WebRtc_UWord16 _localPortRTCP;
-
- // Local port number when the local port for receiving and local port number
- // for sending are not the same.
- WebRtc_UWord16 _srcPort;
- WebRtc_UWord16 _srcPortRTCP;
-
- // Remote port from which last received packet was sent.
- WebRtc_UWord16 _fromPort;
- WebRtc_UWord16 _fromPortRTCP;
-
- char _fromIP[kIpAddressVersion6Length];
- char _destIP[kIpAddressVersion6Length];
- char _localIP[kIpAddressVersion6Length];
- char _localMulticastIP[kIpAddressVersion6Length];
-
- UdpSocketWrapper* _ptrRtpSocket;
- UdpSocketWrapper* _ptrRtcpSocket;
-
- // Local port when the local port for receiving and local port for sending
- // are not the same.
- UdpSocketWrapper* _ptrSendRtpSocket;
- UdpSocketWrapper* _ptrSendRtcpSocket;
-
- SocketAddress _remoteRTPAddr;
- SocketAddress _remoteRTCPAddr;
-
- SocketAddress _localRTPAddr;
- SocketAddress _localRTCPAddr;
-
- WebRtc_Word32 _tos;
- bool _receiving;
- bool _useSetSockOpt;
- bool _qos;
- WebRtc_Word32 _pcp;
- bool _ipV6Enabled;
- WebRtc_Word32 _serviceType;
- WebRtc_Word32 _overrideDSCP;
- WebRtc_UWord32 _maxBitrate;
-
- // Cache used by GetCachedAddress(..).
- RWLockWrapper* _cachLock;
- SocketAddress _previousAddress;
- char _previousIP[kIpAddressVersion6Length];
- WebRtc_UWord32 _previousIPSize;
- WebRtc_UWord16 _previousSourcePort;
-
- SocketAddress _filterIPAddress;
- WebRtc_UWord16 _rtpFilterPort;
- WebRtc_UWord16 _rtcpFilterPort;
-
- UdpTransportData* _packetCallback;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_UDP_TRANSPORT_SOURCE_UDP_TRANSPORT_IMPL_H_
diff --git a/trunk/src/modules/udp_transport/source/udp_transport_unittest.cc b/trunk/src/modules/udp_transport/source/udp_transport_unittest.cc
deleted file mode 100644
index b894435..0000000
--- a/trunk/src/modules/udp_transport/source/udp_transport_unittest.cc
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Empty test just to get code coverage metrics for this dir.
- */
-#include "udp_transport.h"
-#include "gtest/gtest.h"
-
-TEST(UDPTransportTest, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/modules/udp_transport/test/SocketManagerTest.cc b/trunk/src/modules/udp_transport/test/SocketManagerTest.cc
deleted file mode 100644
index 03119be..0000000
--- a/trunk/src/modules/udp_transport/test/SocketManagerTest.cc
+++ /dev/null
@@ -1,449 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cassert>
-#include <iostream>
-
-#ifdef _WIN32
-#include <windows.h>
-#include <tchar.h>
-#else
-#include <stdio.h>
-#define Sleep(x) usleep(x*1000)
-#endif
-
-#include "udp_transport.h"
-#include "common_types.h"
-#include "trace.h"
-
-//#define QOS_TEST
-//#define QOS_TEST_WITH_OVERRIDE // require admin on Win7
-//#define TOS_TEST // require admin on Win7
-//#define TOS_TEST_USING_SETSOCKOPT
-//#define PCP_TEST
-
-class UdpTransportDataA: public UdpTransportData
-{
-public:
- UdpTransportDataA() :
- _counterRTP(0),
- _counterRTCP(0)
- {
- };
- virtual void IncomingRTPPacket(const WebRtc_Word8* incommingRtpPacket,
- const WebRtc_Word32 rtpPacketLength,
- const char* fromIP,
- const WebRtc_UWord16 fromPort)
- {
- _counterRTP++;
- };
-
- virtual void IncomingRTCPPacket(const WebRtc_Word8* incommingRtcpPacket,
- const WebRtc_Word32 rtcpPacketLength,
- const char* fromIP,
- const WebRtc_UWord16 fromPort)
- {
- _counterRTCP++;
- };
- WebRtc_UWord32 _counterRTP;
- WebRtc_UWord32 _counterRTCP;
-};
-
-class UdpTransportDataB: public UdpTransportData
-{
-public:
- UdpTransportDataB() :
- _counterRTP(0),
- _counterRTCP(0)
- {
- };
- virtual void IncomingRTPPacket(const WebRtc_Word8* incommingRtpPacket,
- const WebRtc_Word32 rtpPacketLength,
- const char* fromIP,
- const WebRtc_UWord16 fromPort)
- {
- _counterRTP++;
- };
-
- virtual void IncomingRTCPPacket(const WebRtc_Word8* incommingRtcpPacket,
- const WebRtc_Word32 rtcpPacketLength,
- const char* fromIP,
- const WebRtc_UWord16 fromPort)
- {
- _counterRTCP++;
- };
- WebRtc_UWord32 _counterRTP;
- WebRtc_UWord32 _counterRTCP;
-};
-
-#ifdef _WIN32
-int _tmain(int argc, _TCHAR* argv[])
-#else
-int main(int argc, char* argv[])
-#endif
-{
- Trace::CreateTrace();
- Trace::SetTraceFile("testTrace.txt");
- Trace::SetEncryptedTraceFile("testTraceDebug.txt");
- Trace::SetLevelFilter(webrtc::kTraceAll);
-
- printf("Start UdpTransport test\n");
-
- WebRtc_UWord8 numberOfSocketThreads = 5;
- UdpTransport* client1 = UdpTransport::Create(1,numberOfSocketThreads,NULL);
- numberOfSocketThreads = 0;
- UdpTransport* client2 = UdpTransport::Create(2,numberOfSocketThreads,NULL);
- assert(5 == numberOfSocketThreads);
-
- UdpTransportDataA* client1Callback = new UdpTransportDataA();
- UdpTransportDataB* client2Callback = new UdpTransportDataB();
-
- WebRtc_UWord32 localIP = 0;
- char localIPAddr[64];
- assert( 0 == client1->LocalHostAddress(localIP)); // network host order aka big-endian
-
- sprintf(localIPAddr,"%lu.%lu.%lu.%lu",(localIP>>24)& 0x0ff,(localIP>>16)& 0x0ff ,(localIP>>8)& 0x0ff, localIP & 0x0ff);
- printf("\tLocal IP:%s\n", localIPAddr);
-
- char localIPV6[16];
- char localIPAddrV6[128];
- if( 0 == client1->LocalHostAddressIPV6(localIPV6))
- {
- sprintf(localIPAddrV6,"%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x", localIPV6[0],localIPV6[1],localIPV6[2],localIPV6[3],localIPV6[4],localIPV6[5],localIPV6[6],localIPV6[7], localIPV6[8],localIPV6[9],localIPV6[10],localIPV6[11],localIPV6[12],localIPV6[13],localIPV6[14],localIPV6[15]);
- printf("\tLocal IPV6:%s\n", localIPAddrV6);
- }
-
- char test[9] = "testtest";
- assert( 0 == client1->InitializeReceiveSockets(client1Callback,1234, localIPAddr));
-
-#if defined QOS_TEST_WITH_OVERRIDE || defined QOS_TEST || defined TOS_TEST || defined TOS_TEST_USING_SETSOCKOPT
- assert( -1 == client1->SetQoS(true, 3, 1000)); // should fail
- assert( 0 == client1->InitializeSendSockets("192.168.200.1", 1236,1237));
-#else
- assert( 0 == client1->InitializeSendSockets(localIPAddr, 1236,1237));
-#endif
- assert( 0 == client1->StartReceiving(20));
-
- assert( 0 == client2->InitializeReceiveSockets(client2Callback,1236));
- assert( 0 == client2->InitializeSendSockets(localIPAddr, 1234,1235));
- assert( 0 == client2->StartReceiving(20));
-
- Sleep(10);
-
-#ifdef TOS_TEST
- // note: you need to have QOS installed on your interface for this test
- // test by doing a ethereal sniff and filter out packets with ip.dst == 192.168.200.1
- assert( 0 == client1->SetToS(2));
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- Sleep(10);
- assert( 0 == client1->SetToS(3));
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- Sleep(10);
- assert( 0 == client1->SetToS(0));
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
-
- printf("Tested TOS \n");
- Sleep(5000);
- return 0;
-#endif
-
-#ifdef TOS_TEST_USING_SETSOCKOPT
- // note: you need to have QOS installed on your interface for this test
- // test by doing a ethereal sniff and filter out packets with ip.dst == 192.168.200.1
- assert( 0 == client1->SetToS(2, true));
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- Sleep(10);
- assert( 0 == client1->SetToS(3, true));
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- Sleep(10);
- assert( 0 == client1->SetToS(0, true));
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
-
- printf("Tested TOS using setsockopt \n");
- Sleep(5000);
- return 0;
-#endif
-
-#ifdef QOS_TEST
- // note: you need to have QOS installed on your interface for this test
- // test by doing a ethereal sniff and filter out packets with ip.dst == 192.168.200.1
- assert( 0 == client1->SetQoS(true, 2, 1000)); // SERVICETYPE_CONTROLLEDLOAD 2
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- Sleep(10);
- assert( 0 == client1->SetQoS(true, 3, 1000)); // SERVICETYPE_GUARANTEED 3
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- Sleep(10);
- assert( 0 == client1->SetQoS(false, 0)); //
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
-
- printf("Tested QOS \n");
- Sleep(5000);
- return 0;
-#endif
-
-#ifdef QOS_TEST_WITH_OVERRIDE
- // note: you need to have QOS installed on your interface for this test
- // test by doing a ethereal sniff and filter out packets with ip.dst == 192.168.200.1
- assert( 0 == client1->SetQoS(true, 2, 1000, 1)); // SERVICETYPE_CONTROLLEDLOAD 2
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- Sleep(10);
- assert( 0 == client1->SetQoS(true, 2, 1000, 2)); // SERVICETYPE_GUARANTEED 3
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- Sleep(10);
- assert( 0 == client1->SetQoS(false, 0)); //
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- printf("Tested QOS with override \n");
- Sleep(5000);
- return 0;
-#endif
-
-#ifdef PCP_TEST
- // Note: We currently don't know how to test that the bits are actually set in the frame,
- // this test simply tests the API and that we can send a packet after setting PCP.
- assert( -1 == client1->SetPCP(-1)); // should fail
- assert( -1 == client1->SetPCP(8)); // should fail
- printf("Setting PCP to 7 returned %d \n", client1->SetPCP(7));
- printf("(Failing is normal, requires the CAP_NET_ADMIN capability to succeed.) \n");
- Sleep(10);
- for (int pcp = 6; pcp >= 0; --pcp)
- {
- assert( 0 == client1->SetPCP(pcp));
- Sleep(10);
- assert( 9 == client1->SendPacket(-1, test, 9));
- }
- printf("Tested PCP \n");
- Sleep(5000);
- return 0;
-#endif
-
- Sleep(10);
-
- assert( 9 == client1->SendPacket(-1, test, 9));
-
- // test start rec after a socket has revceived data
- // result: packets received before first startReceive is saved by the OS
-/*
- for(int i = 0; i < 100; i++)
- {
- assert( 9 == client1->SendPacket(-1, test, 9));
- }
- Sleep(10);
- assert( 0 == client2->StartReceiving(20));
-
-// assert( 0 == client2->StopReceiving());
-
- Sleep(10);
- for(int i = 0; i < 100; i++)
- {
- assert( 9 == client1->SendPacket(-1, test, 9));
- }
-
- assert( 0 == client2->StartReceiving(20));
-
- for(int i = 0; i < 100; i++)
- {
- assert( 9 == client1->SendPacket(-1, test, 9));
- }
-*/
- Sleep(10);
-
- assert( 0 == client1Callback->_counterRTP);
- assert( 1 == client2Callback->_counterRTP);
- assert( 0 == client1Callback->_counterRTCP);
- assert( 0 == client2Callback->_counterRTCP);
-
- printf("Sent 1 packet on one socket \n");
-
- char ipAddr[64];
- char tempIpAddr[64];
- char ipMulticastAddr[64];
- WebRtc_UWord16 rtpPort = 0;
- WebRtc_UWord16 rtcpPort = 0;
- bool reusableSocket = true;
- assert( 0 == client2->RemoteSocketInformation(ipAddr, rtpPort, rtcpPort));
- assert( rtpPort == 1234);
- assert( strncmp(ipAddr, localIPAddr, 16) == 0);
-
- assert( 0 == client2->ReceiveSocketInformation(ipAddr, rtpPort, rtcpPort, ipMulticastAddr, reusableSocket));
- assert( rtpPort == 1236);
- assert( rtcpPort == 1237);
- assert( strncmp(ipAddr, "0.0.0.0", 16) == 0);
- assert( ipMulticastAddr[0] == 0);
- assert( reusableSocket == false);
-
- assert( 0 == client2->SendSocketInformation(ipAddr, rtpPort, rtcpPort));
- assert( rtpPort == 1234);
- assert( rtcpPort == 1235);
- assert( strncmp(ipAddr,localIPAddr, 16) == 0);
-
- const int numberOfPackets = 1000;
- int n = 0;
- while(n < numberOfPackets)
- {
- assert( 9 == client1->SendPacket(-1, test, 9));
- assert( 9 == client2->SendPacket(-1, test, 9));
- assert( 9 == client1->SendRTCPPacket(-1, test, 9));
- assert( 9 == client2->SendRTCPPacket(-1, test, 9));
- n++;
- }
- int loops = 0;
- for(; loops < 100 &&
- !(client1Callback->_counterRTP == numberOfPackets &&
- client1Callback->_counterRTCP == numberOfPackets &&
- client2Callback->_counterRTP == numberOfPackets+1 &&
- client2Callback->_counterRTCP == numberOfPackets);
- loops++)
- {
- Sleep(10);
- }
- printf("\tSent %d packets on 4 sockets in:%d ms\n", numberOfPackets, loops*10);
-
- assert( numberOfPackets == client1Callback->_counterRTP);
- assert( numberOfPackets+1 == client2Callback->_counterRTP);
- assert( numberOfPackets == client1Callback->_counterRTCP);
- assert( numberOfPackets == client2Callback->_counterRTCP);
-
- assert( 0 == client1->StopReceiving());
- assert( 0 == client2->StopReceiving());
-
- printf("Tear down client 2\n");
-
- // configure that fail
- assert( -1 == client2->InitializeReceiveSockets(client2Callback,1234, localIPAddr)); // port in use
- assert( !client2->ReceiveSocketsInitialized());
- assert( 0 == client2->InitializeReceiveSockets(client2Callback,1236));
- assert( 0 == client2->StartReceiving(20));
-
- printf("Client 2 re-configured\n");
-
- assert( client1->SendSocketsInitialized());
- assert( client1->ReceiveSocketsInitialized());
- assert( client2->SendSocketsInitialized());
- assert( client2->ReceiveSocketsInitialized());
-
- assert( 9 == client1->SendPacket(-1, test, 9));
-
- // this should not be received since we dont receive in client 1
- assert( 9 == client2->SendPacket(-1, test, 9));
-
- Sleep(10);
-
- assert( numberOfPackets == client1Callback->_counterRTP);
- assert( numberOfPackets+2 == client2Callback->_counterRTP);
- assert( numberOfPackets == client1Callback->_counterRTCP);
- assert( numberOfPackets == client2Callback->_counterRTCP);
- printf("\tSent 1 packet on one socket \n");
-
- printf("Start filter test\n");
-
- assert( 0 == client1->StartReceiving(20));
-
- assert( 0 == client1->SetFilterPorts(1234, 1235)); // should filter out what we send
- assert( 0 == client1->SetFilterIP(localIPAddr));
-
- assert( 0 == client1->FilterIP(tempIpAddr));
- assert( strncmp(tempIpAddr, localIPAddr, 16) == 0);
-
- assert( 9 == client2->SendPacket(-1, test, 9));
- assert( 9 == client2->SendRTCPPacket(-1, test, 9));
-
- Sleep(10);
-
- assert( numberOfPackets == client1Callback->_counterRTP);
- assert( numberOfPackets+2 == client2Callback->_counterRTP);
- assert( numberOfPackets == client1Callback->_counterRTCP);
- assert( numberOfPackets == client2Callback->_counterRTCP);
-
- assert( 0 == client1->SetFilterPorts(1236, 1237)); // should pass through
-
- assert( 9 == client2->SendPacket(-1, test, 9));
- assert( 9 == client2->SendRTCPPacket(-1, test, 9));
- printf("\tSent 1 packet on two sockets \n");
-
- Sleep(10);
-
- assert( numberOfPackets+1 == client1Callback->_counterRTP);
- assert( numberOfPackets+2 == client2Callback->_counterRTP);
- assert( numberOfPackets+1 == client1Callback->_counterRTCP);
- assert( numberOfPackets == client2Callback->_counterRTCP);
-
- assert( 0 == client1->SetFilterIP("127.0.0.2"));
-
- assert( 9 == client2->SendPacket(-1, test, 9));
- assert( 9 == client2->SendRTCPPacket(-1, test, 9));
- printf("\tSent 1 packet on two sockets \n");
-
- Sleep(10);
-
- assert( numberOfPackets+1 == client1Callback->_counterRTP);
- assert( numberOfPackets+2 == client2Callback->_counterRTP);
- assert( numberOfPackets+1 == client1Callback->_counterRTCP);
- assert( numberOfPackets == client2Callback->_counterRTCP);
-
- assert( 0 == client1->SetFilterIP(NULL));
- assert( 0 == client1->SetFilterPorts(0, 0));
-
- printf("Tested filter \n");
-
- assert( 0 == client2->InitializeSourcePorts(1238, 1239));
- assert( 9 == client2->SendPacket(-1, test, 9));
- assert( 9 == client2->SendRTCPPacket(-1, test, 9));
- printf("\tSent 1 packet on two sockets \n");
-
- Sleep(10);
-
- assert( numberOfPackets+2 == client1Callback->_counterRTP);
- assert( numberOfPackets+2 == client2Callback->_counterRTP);
- assert( numberOfPackets+2 == client1Callback->_counterRTCP);
- assert( numberOfPackets == client2Callback->_counterRTCP);
-
- assert( 0 == client1->RemoteSocketInformation(ipAddr, rtpPort, rtcpPort));
- assert( rtpPort == 1238);
- assert( rtcpPort == 1239);
- assert( strncmp(ipAddr, localIPAddr, 16) == 0);
-
- printf("Tested source port \n");
-
- assert( 0 == client2->InitializeSourcePorts(1240 ));
- assert( 9 == client2->SendPacket(-1, test, 9));
- assert( 9 == client2->SendRTCPPacket(-1, test, 9));
- printf("\tSent 1 packet on two sockets \n");
-
- Sleep(10);
-
- assert( 0 == client1->RemoteSocketInformation(ipAddr, rtpPort, rtcpPort));
- assert( rtpPort == 1240);
- assert( rtcpPort == 1241);
-
- printf("Tested SetSendPorts source port \n");
-
- UdpTransport::Destroy(client1);
- UdpTransport::Destroy(client2);
-
- printf("\n\nUdpTransport test done\n");
-
- delete client1Callback;
- delete client2Callback;
-
- Sleep(5000);
- Trace::ReturnTrace();
-};
diff --git a/trunk/src/modules/utility/OWNERS b/trunk/src/modules/utility/OWNERS
deleted file mode 100644
index 674c738..0000000
--- a/trunk/src/modules/utility/OWNERS
+++ /dev/null
@@ -1,4 +0,0 @@
-henrike@webrtc.org
-pwestin@webrtc.org
-asapersson@webrtc.org
-perkj@webrtc.org
\ No newline at end of file
diff --git a/trunk/src/modules/utility/interface/file_player.h b/trunk/src/modules/utility/interface/file_player.h
deleted file mode 100644
index 29da8f1..0000000
--- a/trunk/src/modules/utility/interface/file_player.h
+++ /dev/null
@@ -1,107 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
-
-#include "common_types.h"
-#include "engine_configurations.h"
-#include "module_common_types.h"
-#include "typedefs.h"
-
-namespace webrtc {
-class FileCallback;
-
-class FilePlayer
-{
-public:
- // The largest decoded frame size in samples (60ms with 32kHz sample rate).
- enum {MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32};
- enum {MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES*2};
-
- // Note: will return NULL for video file formats (e.g. AVI) if the flag
- // WEBRTC_MODULE_UTILITY_VIDEO is not defined.
- static FilePlayer* CreateFilePlayer(const WebRtc_UWord32 instanceID,
- const FileFormats fileFormat);
-
- static void DestroyFilePlayer(FilePlayer* player);
-
- virtual WebRtc_Word32 Get10msAudioFromFile(
- WebRtc_Word16* decodedDataBuffer,
- WebRtc_UWord32& decodedDataLengthInSamples,
- const WebRtc_UWord32 frequencyInHz) = 0;
-
- // Register callback for receiving file playing notifications.
- virtual WebRtc_Word32 RegisterModuleFileCallback(
- FileCallback* callback) = 0;
-
- // API for playing audio from fileName to channel.
- // Note: codecInst is used for pre-encoded files.
- virtual WebRtc_Word32 StartPlayingFile(
- const char* fileName,
- bool loop,
- WebRtc_UWord32 startPosition,
- float volumeScaling,
- WebRtc_UWord32 notification,
- WebRtc_UWord32 stopPosition = 0,
- const CodecInst* codecInst = NULL) = 0;
-
- // Note: codecInst is used for pre-encoded files.
- virtual WebRtc_Word32 StartPlayingFile(
- InStream& sourceStream,
- WebRtc_UWord32 startPosition,
- float volumeScaling,
- WebRtc_UWord32 notification,
- WebRtc_UWord32 stopPosition = 0,
- const CodecInst* codecInst = NULL) = 0;
-
- virtual WebRtc_Word32 StopPlayingFile() = 0;
-
- virtual bool IsPlayingFile() const = 0;
-
- virtual WebRtc_Word32 GetPlayoutPosition(WebRtc_UWord32& durationMs) = 0;
-
- // Set audioCodec to the currently used audio codec.
- virtual WebRtc_Word32 AudioCodec(CodecInst& audioCodec) const = 0;
-
- virtual WebRtc_Word32 Frequency() const = 0;
-
- // Note: scaleFactor is in the range [0.0 - 2.0]
- virtual WebRtc_Word32 SetAudioScaling(float scaleFactor) = 0;
-
- // Return the time in ms until next video frame should be pulled (by
- // calling GetVideoFromFile(..)).
- // Note: this API reads one video frame from file. This means that it should
- // be called exactly once per GetVideoFromFile(..) API call.
- virtual WebRtc_Word32 TimeUntilNextVideoFrame() { return -1;}
-
- virtual WebRtc_Word32 StartPlayingVideoFile(
- const char* /*fileName*/,
- bool /*loop*/,
- bool /*videoOnly*/) { return -1;}
-
- virtual WebRtc_Word32 video_codec_info(VideoCodec& /*videoCodec*/) const
- {return -1;}
-
- virtual WebRtc_Word32 GetVideoFromFile(VideoFrame& /*videoFrame*/)
- { return -1;}
-
- // Same as GetVideoFromFile(). videoFrame will have the resolution specified
- // by the width outWidth and height outHeight in pixels.
- virtual WebRtc_Word32 GetVideoFromFile(VideoFrame& /*videoFrame*/,
- const WebRtc_UWord32 /*outWidth*/,
- const WebRtc_UWord32 /*outHeight*/)
- {return -1;}
-protected:
- virtual ~FilePlayer() {}
-
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_PLAYER_H_
diff --git a/trunk/src/modules/utility/interface/file_recorder.h b/trunk/src/modules/utility/interface/file_recorder.h
deleted file mode 100644
index eb460ae..0000000
--- a/trunk/src/modules/utility/interface/file_recorder.h
+++ /dev/null
@@ -1,88 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
-
-#include "audio_coding_module_typedefs.h"
-#include "common_types.h"
-#include "engine_configurations.h"
-#include "media_file_defines.h"
-#include "module_common_types.h"
-#include "tick_util.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-class FileRecorder
-{
-public:
-
- // Note: will return NULL for video file formats (e.g. AVI) if the flag
- // WEBRTC_MODULE_UTILITY_VIDEO is not defined.
- static FileRecorder* CreateFileRecorder(const WebRtc_UWord32 instanceID,
- const FileFormats fileFormat);
-
- static void DestroyFileRecorder(FileRecorder* recorder);
-
- virtual WebRtc_Word32 RegisterModuleFileCallback(
- FileCallback* callback) = 0;
-
- virtual FileFormats RecordingFileFormat() const = 0;
-
- virtual WebRtc_Word32 StartRecordingAudioFile(
- const char* fileName,
- const CodecInst& codecInst,
- WebRtc_UWord32 notification,
- ACMAMRPackingFormat amrFormat = AMRFileStorage) = 0;
-
- virtual WebRtc_Word32 StartRecordingAudioFile(
- OutStream& destStream,
- const CodecInst& codecInst,
- WebRtc_UWord32 notification,
- ACMAMRPackingFormat amrFormat = AMRFileStorage) = 0;
-
- // Stop recording.
- // Note: this API is for both audio and video.
- virtual WebRtc_Word32 StopRecording() = 0;
-
- // Return true if recording.
- // Note: this API is for both audio and video.
- virtual bool IsRecording() const = 0;
-
- virtual WebRtc_Word32 codec_info(CodecInst& codecInst) const = 0;
-
- // Write frame to file. Frame should contain 10ms of un-ecoded audio data.
- virtual WebRtc_Word32 RecordAudioToFile(
- const AudioFrame& frame,
- const TickTime* playoutTS = NULL) = 0;
-
- // Open/create the file specified by fileName for writing audio/video data
- // (relative path is allowed). audioCodecInst specifies the encoding of the
- // audio data. videoCodecInst specifies the encoding of the video data.
- // Only video data will be recorded if videoOnly is true. amrFormat
- // specifies the amr/amrwb storage format.
- // Note: the file format is AVI.
- virtual WebRtc_Word32 StartRecordingVideoFile(
- const char* fileName,
- const CodecInst& audioCodecInst,
- const VideoCodec& videoCodecInst,
- ACMAMRPackingFormat amrFormat = AMRFileStorage,
- bool videoOnly = false) = 0;
-
- // Record the video frame in videoFrame to AVI file.
- virtual WebRtc_Word32 RecordVideoToFile(const VideoFrame& videoFrame) = 0;
-
-protected:
- virtual ~FileRecorder() {}
-
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_FILE_RECORDER_H_
diff --git a/trunk/src/modules/utility/interface/process_thread.h b/trunk/src/modules/utility/interface/process_thread.h
deleted file mode 100644
index 6c51404..0000000
--- a/trunk/src/modules/utility/interface/process_thread.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
-
-#include "typedefs.h"
-
-namespace webrtc {
-class Module;
-
-class ProcessThread
-{
-public:
- static ProcessThread* CreateProcessThread();
- static void DestroyProcessThread(ProcessThread* module);
-
- virtual WebRtc_Word32 Start() = 0;
- virtual WebRtc_Word32 Stop() = 0;
-
- virtual WebRtc_Word32 RegisterModule(const Module* module) = 0;
- virtual WebRtc_Word32 DeRegisterModule(const Module* module) = 0;
-protected:
- virtual ~ProcessThread();
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_PROCESS_THREAD_H_
diff --git a/trunk/src/modules/utility/interface/rtp_dump.h b/trunk/src/modules/utility/interface/rtp_dump.h
deleted file mode 100644
index 9291a1c..0000000
--- a/trunk/src/modules/utility/interface/rtp_dump.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file implements a class that writes a stream of RTP and RTCP packets
-// to a file according to the format specified by rtpplay. See
-// http://www.cs.columbia.edu/irt/software/rtptools/.
-// Notes: supported platforms are Windows, Linux and Mac OSX
-
-#ifndef WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_
-#define WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_
-
-#include "typedefs.h"
-#include "file_wrapper.h"
-
-namespace webrtc {
-class RtpDump
-{
-public:
- // Factory method.
- static RtpDump* CreateRtpDump();
-
- // Delete function. Destructor disabled.
- static void DestroyRtpDump(RtpDump* object);
-
- // Open the file fileNameUTF8 for writing RTP/RTCP packets.
- // Note: this API also adds the rtpplay header.
- virtual WebRtc_Word32 Start(const char* fileNameUTF8) = 0;
-
- // Close the existing file. No more packets will be recorded.
- virtual WebRtc_Word32 Stop() = 0;
-
- // Return true if a file is open for recording RTP/RTCP packets.
- virtual bool IsActive() const = 0;
-
- // Writes the RTP/RTCP packet in packet with length packetLength in bytes.
- // Note: packet should contain the RTP/RTCP part of the packet. I.e. the
- // first bytes of packet should be the RTP/RTCP header.
- virtual WebRtc_Word32 DumpPacket(const WebRtc_UWord8* packet,
- WebRtc_UWord16 packetLength) = 0;
-
-protected:
- virtual ~RtpDump();
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_INTERFACE_RTP_DUMP_H_
diff --git a/trunk/src/modules/utility/source/Android.mk b/trunk/src/modules/utility/source/Android.mk
deleted file mode 100644
index 452a95b..0000000
--- a/trunk/src/modules/utility/source/Android.mk
+++ /dev/null
@@ -1,55 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../android-webrtc.mk
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_utility
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := coder.cc \
- file_player_impl.cc \
- file_recorder_impl.cc \
- process_thread_impl.cc \
- rtp_dump_impl.cc \
- frame_scaler.cc \
- video_coder.cc \
- video_frames_queue.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS) \
- '-DWEBRTC_MODULE_UTILITY_VIDEO'
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../interface \
- $(LOCAL_PATH)/../../audio_coding/main/interface \
- $(LOCAL_PATH)/../../media_file/interface \
- $(LOCAL_PATH)/../../video_coding/main/interface \
- $(LOCAL_PATH)/../../video_coding/codecs/interface \
- $(LOCAL_PATH)/../../.. \
- $(LOCAL_PATH)/../../../common_video/vplib/main/interface \
- $(LOCAL_PATH)/../../../common_audio/resampler/include \
- $(LOCAL_PATH)/../../../system_wrappers/interface
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
-
-
diff --git a/trunk/src/modules/utility/source/coder.cc b/trunk/src/modules/utility/source/coder.cc
deleted file mode 100644
index b858da1..0000000
--- a/trunk/src/modules/utility/source/coder.cc
+++ /dev/null
@@ -1,128 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "coder.h"
-#include "common_types.h"
-#include "module_common_types.h"
-
-// OS independent case insensitive string comparison.
-#ifdef WIN32
- #define STR_CASE_CMP(x,y) ::_stricmp(x,y)
-#else
- #define STR_CASE_CMP(x,y) ::strcasecmp(x,y)
-#endif
-
-namespace webrtc {
-AudioCoder::AudioCoder(WebRtc_UWord32 instanceID)
- : _instanceID(instanceID),
- _acm(AudioCodingModule::Create(instanceID)),
- _receiveCodec(),
- _encodeTimestamp(0),
- _encodedData(NULL),
- _encodedLengthInBytes(0),
- _decodeTimestamp(0)
-{
- _acm->InitializeSender();
- _acm->InitializeReceiver();
- _acm->RegisterTransportCallback(this);
-}
-
-AudioCoder::~AudioCoder()
-{
- AudioCodingModule::Destroy(_acm);
-}
-
-WebRtc_Word32 AudioCoder::SetEncodeCodec(const CodecInst& codecInst,
- ACMAMRPackingFormat amrFormat)
-{
- if(_acm->RegisterSendCodec((CodecInst&)codecInst) == -1)
- {
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 AudioCoder::SetDecodeCodec(const CodecInst& codecInst,
- ACMAMRPackingFormat amrFormat)
-{
- if(_acm->RegisterReceiveCodec((CodecInst&)codecInst) == -1)
- {
- return -1;
- }
- memcpy(&_receiveCodec,&codecInst,sizeof(CodecInst));
- return 0;
-}
-
-WebRtc_Word32 AudioCoder::Decode(AudioFrame& decodedAudio,
- WebRtc_UWord32 sampFreqHz,
- const WebRtc_Word8* incomingPayload,
- WebRtc_Word32 payloadLength)
-{
- if (payloadLength > 0)
- {
- const WebRtc_UWord8 payloadType = _receiveCodec.pltype;
- _decodeTimestamp += _receiveCodec.pacsize;
- if(_acm->IncomingPayload(incomingPayload,
- payloadLength,
- payloadType,
- _decodeTimestamp) == -1)
- {
- return -1;
- }
- }
- return _acm->PlayoutData10Ms((WebRtc_UWord16)sampFreqHz,
- (AudioFrame&)decodedAudio);
-}
-
-WebRtc_Word32 AudioCoder::PlayoutData(AudioFrame& decodedAudio,
- WebRtc_UWord16& sampFreqHz)
-{
- return _acm->PlayoutData10Ms(sampFreqHz, (AudioFrame&)decodedAudio);
-}
-
-WebRtc_Word32 AudioCoder::Encode(const AudioFrame& audio,
- WebRtc_Word8* encodedData,
- WebRtc_UWord32& encodedLengthInBytes)
-{
- // Fake a timestamp in case audio doesn't contain a correct timestamp.
- // Make a local copy of the audio frame since audio is const
- AudioFrame audioFrame = audio;
- audioFrame._timeStamp = _encodeTimestamp;
- _encodeTimestamp += audioFrame._payloadDataLengthInSamples;
-
- // For any codec with a frame size that is longer than 10 ms the encoded
- // length in bytes should be zero until a a full frame has been encoded.
- _encodedLengthInBytes = 0;
- if(_acm->Add10MsData((AudioFrame&)audioFrame) == -1)
- {
- return -1;
- }
- _encodedData = encodedData;
- if(_acm->Process() == -1)
- {
- return -1;
- }
- encodedLengthInBytes = _encodedLengthInBytes;
- return 0;
-}
-
-WebRtc_Word32 AudioCoder::SendData(
- FrameType /* frameType */,
- WebRtc_UWord8 /* payloadType */,
- WebRtc_UWord32 /* timeStamp */,
- const WebRtc_UWord8* payloadData,
- WebRtc_UWord16 payloadSize,
- const RTPFragmentationHeader* /* fragmentation*/)
-{
- memcpy(_encodedData,payloadData,sizeof(WebRtc_UWord8) * payloadSize);
- _encodedLengthInBytes = payloadSize;
- return 0;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/utility/source/coder.h b/trunk/src/modules/utility/source/coder.h
deleted file mode 100644
index e96f455..0000000
--- a/trunk/src/modules/utility/source/coder.h
+++ /dev/null
@@ -1,69 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
-#define WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
-
-#include "audio_coding_module.h"
-#include "common_types.h"
-#include "typedefs.h"
-
-namespace webrtc {
-class AudioFrame;
-
-class AudioCoder : public AudioPacketizationCallback
-{
-public:
- AudioCoder(WebRtc_UWord32 instanceID);
- ~AudioCoder();
-
- WebRtc_Word32 SetEncodeCodec(
- const CodecInst& codecInst,
- ACMAMRPackingFormat amrFormat = AMRBandwidthEfficient);
-
- WebRtc_Word32 SetDecodeCodec(
- const CodecInst& codecInst,
- ACMAMRPackingFormat amrFormat = AMRBandwidthEfficient);
-
- WebRtc_Word32 Decode(AudioFrame& decodedAudio, WebRtc_UWord32 sampFreqHz,
- const WebRtc_Word8* incomingPayload,
- WebRtc_Word32 payloadLength);
-
- WebRtc_Word32 PlayoutData(AudioFrame& decodedAudio,
- WebRtc_UWord16& sampFreqHz);
-
- WebRtc_Word32 Encode(const AudioFrame& audio,
- WebRtc_Word8* encodedData,
- WebRtc_UWord32& encodedLengthInBytes);
-
-protected:
- virtual WebRtc_Word32 SendData(FrameType frameType,
- WebRtc_UWord8 payloadType,
- WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- WebRtc_UWord16 payloadSize,
- const RTPFragmentationHeader* fragmentation);
-
-private:
- WebRtc_UWord32 _instanceID;
-
- AudioCodingModule* _acm;
-
- CodecInst _receiveCodec;
-
- WebRtc_UWord32 _encodeTimestamp;
- WebRtc_Word8* _encodedData;
- WebRtc_UWord32 _encodedLengthInBytes;
-
- WebRtc_UWord32 _decodeTimestamp;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_UTILITY_SOURCE_CODER_H_
diff --git a/trunk/src/modules/utility/source/file_player_impl.cc b/trunk/src/modules/utility/source/file_player_impl.cc
deleted file mode 100644
index 0e22872..0000000
--- a/trunk/src/modules/utility/source/file_player_impl.cc
+++ /dev/null
@@ -1,723 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "file_player_impl.h"
-#include "trace.h"
-
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
- #include "cpu_wrapper.h"
- #include "frame_scaler.h"
- #include "tick_util.h"
- #include "video_coder.h"
-#endif
-
-// OS independent case insensitive string comparison.
-#ifdef WIN32
- #define STR_CASE_CMP(x,y) ::_stricmp(x,y)
-#else
- #define STR_CASE_CMP(x,y) ::strcasecmp(x,y)
-#endif
-
-namespace webrtc {
-FilePlayer* FilePlayer::CreateFilePlayer(WebRtc_UWord32 instanceID,
- FileFormats fileFormat)
-{
- switch(fileFormat)
- {
- case kFileFormatWavFile:
- case kFileFormatCompressedFile:
- case kFileFormatPreencodedFile:
- case kFileFormatPcm16kHzFile:
- case kFileFormatPcm8kHzFile:
- case kFileFormatPcm32kHzFile:
- // audio formats
- return new FilePlayerImpl(instanceID, fileFormat);
- case kFileFormatAviFile:
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
- return new VideoFilePlayerImpl(instanceID, fileFormat);
-#else
- WEBRTC_TRACE(kTraceError, kTraceFile, -1,
- "Invalid file format: %d", kFileFormatAviFile);
- assert(false);
- return NULL;
-#endif
- }
- assert(false);
- return NULL;
-}
-
-void FilePlayer::DestroyFilePlayer(FilePlayer* player)
-{
- delete player;
-}
-
-FilePlayerImpl::FilePlayerImpl(const WebRtc_UWord32 instanceID,
- const FileFormats fileFormat)
- : _instanceID(instanceID),
- _fileFormat(fileFormat),
- _fileModule(*MediaFile::CreateMediaFile(instanceID)),
- _decodedLengthInMS(0),
- _decodedAudioBuffer(),
- _audioDecoder(instanceID),
- _codec(),
- _numberOf10MsPerFrame(0),
- _numberOf10MsInDecoder(0),
- _resampler(),
- _scaling(1.0)
-{
- _codec.plfreq = 0;
-}
-
-FilePlayerImpl::~FilePlayerImpl()
-{
- MediaFile::DestroyMediaFile(&_fileModule);
-}
-
-WebRtc_Word32 FilePlayerImpl::Frequency() const
-{
- if(_codec.plfreq == 0)
- {
- return -1;
- }
- // Make sure that sample rate is 8,16 or 32 kHz. E.g. WAVE files may have
- // other sampling rates.
- if(_codec.plfreq == 11000)
- {
- return 16000;
- }
- else if(_codec.plfreq == 22000)
- {
- return 32000;
- }
- else if(_codec.plfreq == 44000)
- {
- return 32000;
- }
- else if(_codec.plfreq == 48000)
- {
- return 32000;
- }
- else
- {
- return _codec.plfreq;
- }
-}
-
-WebRtc_Word32 FilePlayerImpl::AudioCodec(CodecInst& audioCodec) const
-{
- audioCodec = _codec;
- return 0;
-}
-
-WebRtc_Word32 FilePlayerImpl::Get10msAudioFromFile(
- WebRtc_Word16* outBuffer,
- WebRtc_UWord32& lengthInSamples,
- WebRtc_UWord32 frequencyInHz)
-{
- if(_codec.plfreq == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, _instanceID,
- "FilePlayerImpl::Get10msAudioFromFile() playing not started!\
- codecFreq = %d, wantedFreq = %d",
- _codec.plfreq, frequencyInHz);
- return -1;
- }
-
- AudioFrame unresampledAudioFrame;
- if(STR_CASE_CMP(_codec.plname, "L16") == 0)
- {
- unresampledAudioFrame._frequencyInHz = _codec.plfreq;
-
- // L16 is un-encoded data. Just pull 10 ms.
- WebRtc_UWord32 lengthInBytes =
- sizeof(unresampledAudioFrame._payloadData);
- if (_fileModule.PlayoutAudioData(
- (WebRtc_Word8*)unresampledAudioFrame._payloadData,
- lengthInBytes) == -1)
- {
- // End of file reached.
- return -1;
- }
- if(lengthInBytes == 0)
- {
- lengthInSamples = 0;
- return 0;
- }
- // One sample is two bytes.
- unresampledAudioFrame._payloadDataLengthInSamples =
- (WebRtc_UWord16)lengthInBytes >> 1;
-
- }else {
- // Decode will generate 10 ms of audio data. PlayoutAudioData(..)
- // expects a full frame. If the frame size is larger than 10 ms,
- // PlayoutAudioData(..) data should be called proportionally less often.
- WebRtc_Word16 encodedBuffer[MAX_AUDIO_BUFFER_IN_SAMPLES];
- WebRtc_UWord32 encodedLengthInBytes = 0;
- if(++_numberOf10MsInDecoder >= _numberOf10MsPerFrame)
- {
- _numberOf10MsInDecoder = 0;
- WebRtc_UWord32 bytesFromFile = sizeof(encodedBuffer);
- if (_fileModule.PlayoutAudioData((WebRtc_Word8*)encodedBuffer,
- bytesFromFile) == -1)
- {
- // End of file reached.
- return -1;
- }
- encodedLengthInBytes = bytesFromFile;
- }
- if(_audioDecoder.Decode(unresampledAudioFrame,frequencyInHz,
- (WebRtc_Word8*)encodedBuffer,
- encodedLengthInBytes) == -1)
- {
- return -1;
- }
- }
-
- int outLen = 0;
- if(_resampler.ResetIfNeeded(unresampledAudioFrame._frequencyInHz,
- frequencyInHz, kResamplerSynchronous))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, _instanceID,
- "FilePlayerImpl::Get10msAudioFromFile() unexpected codec");
-
- // New sampling frequency. Update state.
- outLen = frequencyInHz / 100;
- memset(outBuffer, 0, outLen * sizeof(WebRtc_Word16));
- return 0;
- }
- _resampler.Push(unresampledAudioFrame._payloadData,
- unresampledAudioFrame._payloadDataLengthInSamples,
- outBuffer,
- MAX_AUDIO_BUFFER_IN_SAMPLES,
- outLen);
-
- lengthInSamples = outLen;
-
- if(_scaling != 1.0)
- {
- for (int i = 0;i < outLen; i++)
- {
- outBuffer[i] = (WebRtc_Word16)(outBuffer[i] * _scaling);
- }
- }
- _decodedLengthInMS += 10;
- return 0;
-}
-
-WebRtc_Word32 FilePlayerImpl::RegisterModuleFileCallback(FileCallback* callback)
-{
- return _fileModule.SetModuleFileCallback(callback);
-}
-
-WebRtc_Word32 FilePlayerImpl::SetAudioScaling(float scaleFactor)
-{
- if((scaleFactor >= 0)&&(scaleFactor <= 2.0))
- {
- _scaling = scaleFactor;
- return 0;
- }
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, _instanceID,
- "FilePlayerImpl::SetAudioScaling() not allowed scale factor");
- return -1;
-}
-
-WebRtc_Word32 FilePlayerImpl::StartPlayingFile(const char* fileName,
- bool loop,
- WebRtc_UWord32 startPosition,
- float volumeScaling,
- WebRtc_UWord32 notification,
- WebRtc_UWord32 stopPosition,
- const CodecInst* codecInst)
-{
- if (_fileFormat == kFileFormatPcm16kHzFile ||
- _fileFormat == kFileFormatPcm8kHzFile||
- _fileFormat == kFileFormatPcm32kHzFile )
- {
- CodecInst codecInstL16;
- strncpy(codecInstL16.plname,"L16",32);
- codecInstL16.pltype = 93;
- codecInstL16.channels = 1;
-
- if (_fileFormat == kFileFormatPcm8kHzFile)
- {
- codecInstL16.rate = 128000;
- codecInstL16.plfreq = 8000;
- codecInstL16.pacsize = 80;
-
- } else if(_fileFormat == kFileFormatPcm16kHzFile)
- {
- codecInstL16.rate = 256000;
- codecInstL16.plfreq = 16000;
- codecInstL16.pacsize = 160;
-
- }else if(_fileFormat == kFileFormatPcm32kHzFile)
- {
- codecInstL16.rate = 512000;
- codecInstL16.plfreq = 32000;
- codecInstL16.pacsize = 160;
- } else
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, _instanceID,
- "FilePlayerImpl::StartPlayingFile() sample frequency\
- specifed not supported for PCM format.");
- return -1;
- }
-
- if (_fileModule.StartPlayingAudioFile(fileName, notification, loop,
- _fileFormat, &codecInstL16,
- startPosition,
- stopPosition) == -1)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVoice,
- _instanceID,
- "FilePlayerImpl::StartPlayingFile() failed to initialize file\
- %s playout.", fileName);
- return -1;
- }
- SetAudioScaling(volumeScaling);
- }else if(_fileFormat == kFileFormatPreencodedFile)
- {
- if (_fileModule.StartPlayingAudioFile(fileName, notification, loop,
- _fileFormat, codecInst) == -1)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVoice,
- _instanceID,
- "FilePlayerImpl::StartPlayingPreEncodedFile() failed to\
- initialize pre-encoded file %s playout.",
- fileName);
- return -1;
- }
- } else
- {
- CodecInst* no_inst = NULL;
- if (_fileModule.StartPlayingAudioFile(fileName, notification, loop,
- _fileFormat, no_inst,
- startPosition,
- stopPosition) == -1)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVoice,
- _instanceID,
- "FilePlayerImpl::StartPlayingFile() failed to initialize file\
- %s playout.", fileName);
- return -1;
- }
- SetAudioScaling(volumeScaling);
- }
- if (SetUpAudioDecoder() == -1)
- {
- StopPlayingFile();
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 FilePlayerImpl::StartPlayingFile(InStream& sourceStream,
- WebRtc_UWord32 startPosition,
- float volumeScaling,
- WebRtc_UWord32 notification,
- WebRtc_UWord32 stopPosition,
- const CodecInst* codecInst)
-{
- if (_fileFormat == kFileFormatPcm16kHzFile ||
- _fileFormat == kFileFormatPcm32kHzFile ||
- _fileFormat == kFileFormatPcm8kHzFile)
- {
- CodecInst codecInstL16;
- strncpy(codecInstL16.plname,"L16",32);
- codecInstL16.pltype = 93;
- codecInstL16.channels = 1;
-
- if (_fileFormat == kFileFormatPcm8kHzFile)
- {
- codecInstL16.rate = 128000;
- codecInstL16.plfreq = 8000;
- codecInstL16.pacsize = 80;
-
- }else if (_fileFormat == kFileFormatPcm16kHzFile)
- {
- codecInstL16.rate = 256000;
- codecInstL16.plfreq = 16000;
- codecInstL16.pacsize = 160;
-
- }else if (_fileFormat == kFileFormatPcm32kHzFile)
- {
- codecInstL16.rate = 512000;
- codecInstL16.plfreq = 32000;
- codecInstL16.pacsize = 160;
- }else
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVoice,
- _instanceID,
- "FilePlayerImpl::StartPlayingFile() sample frequency specifed\
- not supported for PCM format.");
- return -1;
- }
- if (_fileModule.StartPlayingAudioStream(sourceStream, notification,
- _fileFormat, &codecInstL16,
- startPosition,
- stopPosition) == -1)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVoice,
- _instanceID,
- "FilePlayerImpl::StartPlayingFile() failed to initialize stream\
- playout.");
- return -1;
- }
-
- }else if(_fileFormat == kFileFormatPreencodedFile)
- {
- if (_fileModule.StartPlayingAudioStream(sourceStream, notification,
- _fileFormat, codecInst) == -1)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVoice,
- _instanceID,
- "FilePlayerImpl::StartPlayingFile() failed to initialize stream\
- playout.");
- return -1;
- }
- } else {
- CodecInst* no_inst = NULL;
- if (_fileModule.StartPlayingAudioStream(sourceStream, notification,
- _fileFormat, no_inst,
- startPosition,
- stopPosition) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, _instanceID,
- "FilePlayerImpl::StartPlayingFile() failed to initialize\
- stream playout.");
- return -1;
- }
- }
- SetAudioScaling(volumeScaling);
-
- if (SetUpAudioDecoder() == -1)
- {
- StopPlayingFile();
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 FilePlayerImpl::StopPlayingFile()
-{
- memset(&_codec, 0, sizeof(CodecInst));
- _numberOf10MsPerFrame = 0;
- _numberOf10MsInDecoder = 0;
- return _fileModule.StopPlaying();
-}
-
-bool FilePlayerImpl::IsPlayingFile() const
-{
- return _fileModule.IsPlaying();
-}
-
-WebRtc_Word32 FilePlayerImpl::GetPlayoutPosition(WebRtc_UWord32& durationMs)
-{
- return _fileModule.PlayoutPositionMs(durationMs);
-}
-
-WebRtc_Word32 FilePlayerImpl::SetUpAudioDecoder()
-{
- if ((_fileModule.codec_info(_codec) == -1))
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVoice,
- _instanceID,
- "FilePlayerImpl::StartPlayingFile() failed to retrieve Codec info\
- of file data.");
- return -1;
- }
- if( STR_CASE_CMP(_codec.plname, "L16") != 0 &&
- _audioDecoder.SetDecodeCodec(_codec,AMRFileStorage) == -1)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVoice,
- _instanceID,
- "FilePlayerImpl::StartPlayingFile() codec %s not supported",
- _codec.plname);
- return -1;
- }
- _numberOf10MsPerFrame = _codec.pacsize / (_codec.plfreq / 100);
- _numberOf10MsInDecoder = 0;
- return 0;
-}
-
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
-VideoFilePlayerImpl::VideoFilePlayerImpl(WebRtc_UWord32 instanceID,
- FileFormats fileFormat)
- : FilePlayerImpl(instanceID,fileFormat),
- _videoDecoder(*new VideoCoder(instanceID)),
- video_codec_info_(),
- _decodedVideoFrames(0),
- _encodedData(*new EncodedVideoData()),
- _frameScaler(*new FrameScaler()),
- _critSec(CriticalSectionWrapper::CreateCriticalSection()),
- _startTime(),
- _accumulatedRenderTimeMs(0),
- _frameLengthMS(0),
- _numberOfFramesRead(0),
- _videoOnly(false)
-{
- memset(&video_codec_info_, 0, sizeof(video_codec_info_));
-}
-
-VideoFilePlayerImpl::~VideoFilePlayerImpl()
-{
- delete _critSec;
- delete &_frameScaler;
- delete &_videoDecoder;
- delete &_encodedData;
-}
-
-WebRtc_Word32 VideoFilePlayerImpl::StartPlayingVideoFile(
- const char* fileName,
- bool loop,
- bool videoOnly)
-{
- CriticalSectionScoped lock( _critSec);
-
- if(_fileModule.StartPlayingVideoFile(fileName, loop, videoOnly,
- _fileFormat) != 0)
- {
- return -1;
- }
-
- _decodedVideoFrames = 0;
- _accumulatedRenderTimeMs = 0;
- _frameLengthMS = 0;
- _numberOfFramesRead = 0;
- _videoOnly = videoOnly;
-
- // Set up video_codec_info_ according to file,
- if(SetUpVideoDecoder() != 0)
- {
- StopPlayingFile();
- return -1;
- }
- if(!videoOnly)
- {
- // Set up _codec according to file,
- if(SetUpAudioDecoder() != 0)
- {
- StopPlayingFile();
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_Word32 VideoFilePlayerImpl::StopPlayingFile()
-{
- CriticalSectionScoped lock( _critSec);
-
- _decodedVideoFrames = 0;
- _videoDecoder.ResetDecoder();
-
- return FilePlayerImpl::StopPlayingFile();
-}
-
-WebRtc_Word32 VideoFilePlayerImpl::GetVideoFromFile(VideoFrame& videoFrame,
- WebRtc_UWord32 outWidth,
- WebRtc_UWord32 outHeight)
-{
- CriticalSectionScoped lock( _critSec);
-
- WebRtc_Word32 retVal = GetVideoFromFile(videoFrame);
- if(retVal != 0)
- {
- return retVal;
- }
- if( videoFrame.Length() > 0)
- {
- retVal = _frameScaler.ResizeFrameIfNeeded(&videoFrame, outWidth,
- outHeight);
- }
- return retVal;
-}
-
-WebRtc_Word32 VideoFilePlayerImpl::GetVideoFromFile(VideoFrame& videoFrame)
-{
- CriticalSectionScoped lock( _critSec);
- // No new video data read from file.
- if(_encodedData.payloadSize == 0)
- {
- videoFrame.SetLength(0);
- return -1;
- }
- WebRtc_Word32 retVal = 0;
- if(strncmp(video_codec_info_.plName, "I420", 5) == 0)
- {
- videoFrame.CopyFrame(_encodedData.payloadSize,_encodedData.payloadData);
- videoFrame.SetLength(_encodedData.payloadSize);
- videoFrame.SetWidth(video_codec_info_.width);
- videoFrame.SetHeight(video_codec_info_.height);
- }else
- {
- // Set the timestamp manually since there is no timestamp in the file.
- // Update timestam according to 90 kHz stream.
- _encodedData.timeStamp += (90000 / video_codec_info_.maxFramerate);
- retVal = _videoDecoder.Decode(videoFrame, _encodedData);
- }
-
- WebRtc_Word64 renderTimeMs = TickTime::MillisecondTimestamp();
- videoFrame.SetRenderTime(renderTimeMs);
-
- // Indicate that the current frame in the encoded buffer is old/has
- // already been read.
- _encodedData.payloadSize = 0;
- if( retVal == 0)
- {
- _decodedVideoFrames++;
- }
- return retVal;
-}
-
-WebRtc_Word32 VideoFilePlayerImpl::video_codec_info(
- VideoCodec& videoCodec) const
-{
- if(video_codec_info_.plName[0] == 0)
- {
- return -1;
- }
- memcpy(&videoCodec, &video_codec_info_, sizeof(VideoCodec));
- return 0;
-}
-
-WebRtc_Word32 VideoFilePlayerImpl::TimeUntilNextVideoFrame()
-{
- if(_fileFormat != kFileFormatAviFile)
- {
- return -1;
- }
- if(!_fileModule.IsPlaying())
- {
- return -1;
- }
- if(_encodedData.payloadSize <= 0)
- {
- // Read next frame from file.
- CriticalSectionScoped lock( _critSec);
-
- if(_fileFormat == kFileFormatAviFile)
- {
- // Get next video frame
- WebRtc_UWord32 encodedBufferLengthInBytes = _encodedData.bufferSize;
- if(_fileModule.PlayoutAVIVideoData(
- reinterpret_cast< WebRtc_Word8*>(_encodedData.payloadData),
- encodedBufferLengthInBytes) != 0)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVideo,
- _instanceID,
- "FilePlayerImpl::TimeUntilNextVideoFrame() error reading\
- video data");
- return -1;
- }
- _encodedData.payloadSize = encodedBufferLengthInBytes;
- _encodedData.codec = video_codec_info_.codecType;
- _numberOfFramesRead++;
-
- if(_accumulatedRenderTimeMs == 0)
- {
- _startTime = TickTime::Now();
- // This if-statement should only trigger once.
- _accumulatedRenderTimeMs = 1;
- } else {
- // A full seconds worth of frames have been read.
- if(_numberOfFramesRead % video_codec_info_.maxFramerate == 0)
- {
- // Frame rate is in frames per seconds. Frame length is
- // calculated as an integer division which means it may
- // be rounded down. Compensate for this every second.
- WebRtc_UWord32 rest = 1000%_frameLengthMS;
- _accumulatedRenderTimeMs += rest;
- }
- _accumulatedRenderTimeMs += _frameLengthMS;
- }
- }
- }
-
- WebRtc_Word64 timeToNextFrame;
- if(_videoOnly)
- {
- timeToNextFrame = _accumulatedRenderTimeMs -
- (TickTime::Now() - _startTime).Milliseconds();
-
- } else {
- // Synchronize with the audio stream instead of system clock.
- timeToNextFrame = _accumulatedRenderTimeMs - _decodedLengthInMS;
- }
- if(timeToNextFrame < 0)
- {
- return 0;
-
- } else if(timeToNextFrame > 0x0fffffff)
- {
- // Wraparound or audio stream has gone to far ahead of the video stream.
- return -1;
- }
- return static_cast<WebRtc_Word32>(timeToNextFrame);
-}
-
-WebRtc_Word32 VideoFilePlayerImpl::SetUpVideoDecoder()
-{
- if (_fileModule.VideoCodecInst(video_codec_info_) != 0)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVideo,
- _instanceID,
- "FilePlayerImpl::SetVideoDecoder() failed to retrieve Codec info of\
- file data.");
- return -1;
- }
-
- WebRtc_Word32 useNumberOfCores = 1;
- if(_videoDecoder.SetDecodeCodec(video_codec_info_, useNumberOfCores) != 0)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVideo,
- _instanceID,
- "FilePlayerImpl::SetUpVideoDecoder() codec %s not supported",
- video_codec_info_.plName);
- return -1;
- }
-
- _frameLengthMS = 1000/video_codec_info_.maxFramerate;
-
- // Size of unencoded data (I420) should be the largest possible frame size
- // in a file.
- const WebRtc_UWord32 KReadBufferSize = 3 * video_codec_info_.width *
- video_codec_info_.height / 2;
- _encodedData.VerifyAndAllocate(KReadBufferSize);
- _encodedData.encodedHeight = video_codec_info_.height;
- _encodedData.encodedWidth = video_codec_info_.width;
- _encodedData.payloadType = video_codec_info_.plType;
- _encodedData.timeStamp = 0;
- return 0;
-}
-#endif // WEBRTC_MODULE_UTILITY_VIDEO
-} // namespace webrtc
diff --git a/trunk/src/modules/utility/source/file_player_impl.h b/trunk/src/modules/utility/source/file_player_impl.h
deleted file mode 100644
index e15cb09..0000000
--- a/trunk/src/modules/utility/source/file_player_impl.h
+++ /dev/null
@@ -1,124 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
-#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
-
-#include "coder.h"
-#include "common_types.h"
-#include "critical_section_wrapper.h"
-#include "engine_configurations.h"
-#include "file_player.h"
-#include "media_file_defines.h"
-#include "media_file.h"
-#include "resampler.h"
-#include "tick_util.h"
-#include "typedefs.h"
-
-namespace webrtc {
-class VideoCoder;
-class FrameScaler;
-
-class FilePlayerImpl : public FilePlayer
-{
-public:
- FilePlayerImpl(WebRtc_UWord32 instanceID, FileFormats fileFormat);
- ~FilePlayerImpl();
-
- // FilePlayer functions.
- virtual WebRtc_Word32 Get10msAudioFromFile(
- WebRtc_Word16* decodedDataBuffer,
- WebRtc_UWord32& decodedDataLengthInSamples,
- const WebRtc_UWord32 frequencyInHz);
- virtual WebRtc_Word32 RegisterModuleFileCallback(FileCallback* callback);
- virtual WebRtc_Word32 StartPlayingFile(
- const char* fileName,
- bool loop,
- WebRtc_UWord32 startPosition,
- float volumeScaling,
- WebRtc_UWord32 notification,
- WebRtc_UWord32 stopPosition = 0,
- const CodecInst* codecInst = NULL);
- virtual WebRtc_Word32 StartPlayingFile(
- InStream& sourceStream,
- WebRtc_UWord32 startPosition,
- float volumeScaling,
- WebRtc_UWord32 notification,
- WebRtc_UWord32 stopPosition = 0,
- const CodecInst* codecInst = NULL);
- virtual WebRtc_Word32 StopPlayingFile();
- virtual bool IsPlayingFile() const;
- virtual WebRtc_Word32 GetPlayoutPosition(WebRtc_UWord32& durationMs);
- virtual WebRtc_Word32 AudioCodec(CodecInst& audioCodec) const;
- virtual WebRtc_Word32 Frequency() const;
- virtual WebRtc_Word32 SetAudioScaling(float scaleFactor);
-
-protected:
- WebRtc_Word32 SetUpAudioDecoder();
-
- WebRtc_UWord32 _instanceID;
- const FileFormats _fileFormat;
- MediaFile& _fileModule;
-
- WebRtc_UWord32 _decodedLengthInMS;
-
-private:
- WebRtc_Word16 _decodedAudioBuffer[MAX_AUDIO_BUFFER_IN_SAMPLES];
- AudioCoder _audioDecoder;
-
- CodecInst _codec;
- WebRtc_Word32 _numberOf10MsPerFrame;
- WebRtc_Word32 _numberOf10MsInDecoder;
-
- Resampler _resampler;
- float _scaling;
-};
-
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
-class VideoFilePlayerImpl: public FilePlayerImpl
-{
-public:
- VideoFilePlayerImpl(WebRtc_UWord32 instanceID, FileFormats fileFormat);
- ~VideoFilePlayerImpl();
-
- // FilePlayer functions.
- virtual WebRtc_Word32 TimeUntilNextVideoFrame();
- virtual WebRtc_Word32 StartPlayingVideoFile(const char* fileName,
- bool loop,
- bool videoOnly);
- virtual WebRtc_Word32 StopPlayingFile();
- virtual WebRtc_Word32 video_codec_info(VideoCodec& videoCodec) const;
- virtual WebRtc_Word32 GetVideoFromFile(VideoFrame& videoFrame);
- virtual WebRtc_Word32 GetVideoFromFile(VideoFrame& videoFrame,
- const WebRtc_UWord32 outWidth,
- const WebRtc_UWord32 outHeight);
-
-private:
- WebRtc_Word32 SetUpVideoDecoder();
-
- VideoCoder& _videoDecoder;
- VideoCodec video_codec_info_;
- WebRtc_Word32 _decodedVideoFrames;
-
- EncodedVideoData& _encodedData;
-
- FrameScaler& _frameScaler;
- CriticalSectionWrapper* _critSec;
- TickTime _startTime;
- WebRtc_Word64 _accumulatedRenderTimeMs;
- WebRtc_UWord32 _frameLengthMS;
-
- WebRtc_Word32 _numberOfFramesRead;
- bool _videoOnly;
-};
-#endif //WEBRTC_MODULE_UTILITY_VIDEO
-
-} // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_PLAYER_IMPL_H_
diff --git a/trunk/src/modules/utility/source/file_player_unittest.cc b/trunk/src/modules/utility/source/file_player_unittest.cc
deleted file mode 100644
index 2e76905..0000000
--- a/trunk/src/modules/utility/source/file_player_unittest.cc
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Empty test just to get code coverage metrics for this dir.
- */
-#include "file_player.h"
-#include "gtest/gtest.h"
-
-TEST(FilePlayerTest, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/modules/utility/source/file_recorder_impl.cc b/trunk/src/modules/utility/source/file_recorder_impl.cc
deleted file mode 100644
index 07d16aa..0000000
--- a/trunk/src/modules/utility/source/file_recorder_impl.cc
+++ /dev/null
@@ -1,794 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "engine_configurations.h"
-#include "file_recorder_impl.h"
-#include "media_file.h"
-#include "trace.h"
-
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
- #include "cpu_wrapper.h"
- #include "critical_section_wrapper.h"
- #include "frame_scaler.h"
- #include "video_coder.h"
- #include "video_frames_queue.h"
-#endif
-
-// OS independent case insensitive string comparison.
-#ifdef WIN32
- #define STR_CASE_CMP(x,y) ::_stricmp(x,y)
-#else
- #define STR_CASE_CMP(x,y) ::strcasecmp(x,y)
-#endif
-
-namespace webrtc {
-FileRecorder* FileRecorder::CreateFileRecorder(WebRtc_UWord32 instanceID,
- FileFormats fileFormat)
-{
- switch(fileFormat)
- {
- case kFileFormatWavFile:
- case kFileFormatCompressedFile:
- case kFileFormatPreencodedFile:
- case kFileFormatPcm16kHzFile:
- case kFileFormatPcm8kHzFile:
- case kFileFormatPcm32kHzFile:
- return new FileRecorderImpl(instanceID, fileFormat);
- case kFileFormatAviFile:
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
- return new AviRecorder(instanceID, fileFormat);
-#else
- WEBRTC_TRACE(kTraceError, kTraceFile, -1,
- "Invalid file format: %d", kFileFormatAviFile);
- assert(false);
- return NULL;
-#endif
- }
- assert(false);
- return NULL;
-}
-
-void FileRecorder::DestroyFileRecorder(FileRecorder* recorder)
-{
- delete recorder;
-}
-
-FileRecorderImpl::FileRecorderImpl(WebRtc_UWord32 instanceID,
- FileFormats fileFormat)
- : _instanceID(instanceID),
- _fileFormat(fileFormat),
- _moduleFile(MediaFile::CreateMediaFile(_instanceID)),
- _stream(NULL),
- codec_info_(),
- _amrFormat(AMRFileStorage),
- _audioBuffer(),
- _audioEncoder(instanceID),
- _audioResampler()
-{
-}
-
-FileRecorderImpl::~FileRecorderImpl()
-{
- MediaFile::DestroyMediaFile(_moduleFile);
-}
-
-FileFormats FileRecorderImpl::RecordingFileFormat() const
-{
- return _fileFormat;
-}
-
-WebRtc_Word32 FileRecorderImpl::RegisterModuleFileCallback(
- FileCallback* callback)
-{
- if(_moduleFile == NULL)
- {
- return -1;
- }
- return _moduleFile->SetModuleFileCallback(callback);
-}
-
-WebRtc_Word32 FileRecorderImpl::StartRecordingAudioFile(
- const char* fileName,
- const CodecInst& codecInst,
- WebRtc_UWord32 notificationTimeMs,
- ACMAMRPackingFormat amrFormat)
-{
- if(_moduleFile == NULL)
- {
- return -1;
- }
- codec_info_ = codecInst;
- _amrFormat = amrFormat;
-
- WebRtc_Word32 retVal = 0;
- if(_fileFormat != kFileFormatAviFile)
- {
- // AVI files should be started using StartRecordingVideoFile(..) all
- // other formats should use this API.
- retVal =_moduleFile->StartRecordingAudioFile(fileName, _fileFormat,
- codecInst,
- notificationTimeMs);
- }
-
- if( retVal == 0)
- {
- retVal = SetUpAudioEncoder();
- }
- if( retVal != 0)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVoice,
- _instanceID,
- "FileRecorder::StartRecording() failed to initialize file %s for\
- recording.",
- fileName);
-
- if(IsRecording())
- {
- StopRecording();
- }
- }
- return retVal;
-}
-
-WebRtc_Word32 FileRecorderImpl::StartRecordingAudioFile(
- OutStream& destStream,
- const CodecInst& codecInst,
- WebRtc_UWord32 notificationTimeMs,
- ACMAMRPackingFormat amrFormat)
-{
- codec_info_ = codecInst;
- _amrFormat = amrFormat;
-
- WebRtc_Word32 retVal = _moduleFile->StartRecordingAudioStream(
- destStream,
- _fileFormat,
- codecInst,
- notificationTimeMs);
-
- if( retVal == 0)
- {
- retVal = SetUpAudioEncoder();
- }
- if( retVal != 0)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVoice,
- _instanceID,
- "FileRecorder::StartRecording() failed to initialize outStream for\
- recording.");
-
- if(IsRecording())
- {
- StopRecording();
- }
- }
- return retVal;
-}
-
-WebRtc_Word32 FileRecorderImpl::StopRecording()
-{
- memset(&codec_info_, 0, sizeof(CodecInst));
- return _moduleFile->StopRecording();
-}
-
-bool FileRecorderImpl::IsRecording() const
-{
- return _moduleFile->IsRecording();
-}
-
-WebRtc_Word32 FileRecorderImpl::RecordAudioToFile(
- const AudioFrame& incomingAudioFrame,
- const TickTime* playoutTS)
-{
- if (codec_info_.plfreq == 0)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVoice,
- _instanceID,
- "FileRecorder::RecordAudioToFile() recording audio is not turned\
- on");
- return -1;
- }
- AudioFrame tempAudioFrame;
- tempAudioFrame._payloadDataLengthInSamples = 0;
- if( incomingAudioFrame._audioChannel == 2 &&
- !_moduleFile->IsStereo())
- {
- // Recording mono but incoming audio is (interleaved) stereo.
- tempAudioFrame._audioChannel = 1;
- tempAudioFrame._frequencyInHz = incomingAudioFrame._frequencyInHz;
- tempAudioFrame._payloadDataLengthInSamples =
- incomingAudioFrame._payloadDataLengthInSamples;
- for (WebRtc_UWord16 i = 0;
- i < (incomingAudioFrame._payloadDataLengthInSamples); i++)
- {
- // Sample value is the average of left and right buffer rounded to
- // closest integer value. Note samples can be either 1 or 2 byte.
- tempAudioFrame._payloadData[i] =
- ((incomingAudioFrame._payloadData[2 * i] +
- incomingAudioFrame._payloadData[(2 * i) + 1] + 1) >> 1);
- }
- }
- else if( incomingAudioFrame._audioChannel == 1 &&
- _moduleFile->IsStereo())
- {
- // Recording stereo but incoming audio is mono.
- tempAudioFrame._audioChannel = 2;
- tempAudioFrame._frequencyInHz = incomingAudioFrame._frequencyInHz;
- tempAudioFrame._payloadDataLengthInSamples =
- incomingAudioFrame._payloadDataLengthInSamples;
- for (WebRtc_UWord16 i = 0;
- i < (incomingAudioFrame._payloadDataLengthInSamples); i++)
- {
- // Duplicate sample to both channels
- tempAudioFrame._payloadData[2*i] =
- incomingAudioFrame._payloadData[i];
- tempAudioFrame._payloadData[2*i+1] =
- incomingAudioFrame._payloadData[i];
- }
- }
-
- const AudioFrame* ptrAudioFrame = &incomingAudioFrame;
- if(tempAudioFrame._payloadDataLengthInSamples != 0)
- {
- // If ptrAudioFrame is not empty it contains the audio to be recorded.
- ptrAudioFrame = &tempAudioFrame;
- }
-
- // Encode the audio data before writing to file. Don't encode if the codec
- // is PCM.
- // NOTE: stereo recording is only supported for WAV files.
- // TODO (hellner): WAV expect PCM in little endian byte order. Not
- // "encoding" with PCM coder should be a problem for big endian systems.
- WebRtc_UWord32 encodedLenInBytes = 0;
- if (_fileFormat == kFileFormatPreencodedFile ||
- STR_CASE_CMP(codec_info_.plname, "L16") != 0)
- {
- if (_audioEncoder.Encode(*ptrAudioFrame, _audioBuffer,
- encodedLenInBytes) == -1)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVoice,
- _instanceID,
- "FileRecorder::RecordAudioToFile() codec %s not supported or\
- failed to encode stream",
- codec_info_.plname);
- return -1;
- }
- } else {
- int outLen = 0;
- if(ptrAudioFrame->_audioChannel == 2)
- {
- // ptrAudioFrame contains interleaved stereo audio.
- _audioResampler.ResetIfNeeded(ptrAudioFrame->_frequencyInHz,
- codec_info_.plfreq,
- kResamplerSynchronousStereo);
- _audioResampler.Push(ptrAudioFrame->_payloadData,
- ptrAudioFrame->_payloadDataLengthInSamples *
- ptrAudioFrame->_audioChannel,
- (WebRtc_Word16*)_audioBuffer,
- MAX_AUDIO_BUFFER_IN_BYTES, outLen);
- } else {
- _audioResampler.ResetIfNeeded(ptrAudioFrame->_frequencyInHz,
- codec_info_.plfreq,
- kResamplerSynchronous);
- _audioResampler.Push(ptrAudioFrame->_payloadData,
- ptrAudioFrame->_payloadDataLengthInSamples,
- (WebRtc_Word16*)_audioBuffer,
- MAX_AUDIO_BUFFER_IN_BYTES, outLen);
- }
- encodedLenInBytes = outLen * sizeof(WebRtc_Word16);
- }
-
- // Codec may not be operating at a frame rate of 10 ms. Whenever enough
- // 10 ms chunks of data has been pushed to the encoder an encoded frame
- // will be available. Wait until then.
- if (encodedLenInBytes)
- {
- WebRtc_UWord16 msOfData =
- ptrAudioFrame->_payloadDataLengthInSamples /
- WebRtc_UWord16(ptrAudioFrame->_frequencyInHz / 1000);
- if (WriteEncodedAudioData(_audioBuffer,
- (WebRtc_UWord16)encodedLenInBytes,
- msOfData, playoutTS) == -1)
- {
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_Word32 FileRecorderImpl::SetUpAudioEncoder()
-{
- if (_fileFormat == kFileFormatPreencodedFile ||
- STR_CASE_CMP(codec_info_.plname, "L16") != 0)
- {
- if(_audioEncoder.SetEncodeCodec(codec_info_,_amrFormat) == -1)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVoice,
- _instanceID,
- "FileRecorder::StartRecording() codec %s not supported",
- codec_info_.plname);
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_Word32 FileRecorderImpl::codec_info(CodecInst& codecInst) const
-{
- if(codec_info_.plfreq == 0)
- {
- return -1;
- }
- codecInst = codec_info_;
- return 0;
-}
-
-WebRtc_Word32 FileRecorderImpl::WriteEncodedAudioData(
- const WebRtc_Word8* audioBuffer,
- WebRtc_UWord16 bufferLength,
- WebRtc_UWord16 /*millisecondsOfData*/,
- const TickTime* /*playoutTS*/)
-{
- return _moduleFile->IncomingAudioData(audioBuffer, bufferLength);
-}
-
-
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
-class AudioFrameFileInfo
-{
- public:
- AudioFrameFileInfo(const WebRtc_Word8* audioData,
- const WebRtc_UWord16 audioSize,
- const WebRtc_UWord16 audioMS,
- const TickTime& playoutTS)
- : _audioData(), _audioSize(audioSize), _audioMS(audioMS),
- _playoutTS(playoutTS)
- {
- if(audioSize > MAX_AUDIO_BUFFER_IN_BYTES)
- {
- assert(false);
- _audioSize = 0;
- return;
- }
- memcpy(_audioData, audioData, audioSize);
- };
- // TODO (hellner): either turn into a struct or provide get/set functions.
- WebRtc_Word8 _audioData[MAX_AUDIO_BUFFER_IN_BYTES];
- WebRtc_UWord16 _audioSize;
- WebRtc_UWord16 _audioMS;
- TickTime _playoutTS;
-};
-
-AviRecorder::AviRecorder(WebRtc_UWord32 instanceID, FileFormats fileFormat)
- : FileRecorderImpl(instanceID, fileFormat),
- _videoOnly(false),
- _thread( 0),
- _timeEvent(*EventWrapper::Create()),
- _critSec(CriticalSectionWrapper::CreateCriticalSection()),
- _writtenVideoFramesCounter(0),
- _writtenAudioMS(0),
- _writtenVideoMS(0)
-{
- _videoEncoder = new VideoCoder(instanceID);
- _frameScaler = new FrameScaler();
- _videoFramesQueue = new VideoFramesQueue();
- _thread = ThreadWrapper::CreateThread(Run, this, kNormalPriority,
- "AviRecorder()");
-}
-
-AviRecorder::~AviRecorder( )
-{
- StopRecording( );
-
- delete _videoEncoder;
- delete _frameScaler;
- delete _videoFramesQueue;
- delete _thread;
- delete &_timeEvent;
- delete _critSec;
-}
-
-WebRtc_Word32 AviRecorder::StartRecordingVideoFile(
- const char* fileName,
- const CodecInst& audioCodecInst,
- const VideoCodec& videoCodecInst,
- ACMAMRPackingFormat amrFormat,
- bool videoOnly)
-{
- _firstAudioFrameReceived = false;
- _videoCodecInst = videoCodecInst;
- _videoOnly = videoOnly;
-
- if(_moduleFile->StartRecordingVideoFile(fileName, _fileFormat,
- audioCodecInst, videoCodecInst,
- videoOnly) != 0)
- {
- return -1;
- }
-
- if(!videoOnly)
- {
- if(FileRecorderImpl::StartRecordingAudioFile(fileName,audioCodecInst, 0,
- amrFormat) !=0)
- {
- StopRecording();
- return -1;
- }
- }
- if( SetUpVideoEncoder() != 0)
- {
- StopRecording();
- return -1;
- }
- if(_videoOnly)
- {
- // Writing to AVI file is non-blocking.
- // Start non-blocking timer if video only. If recording both video and
- // audio let the pushing of audio frames be the timer.
- _timeEvent.StartTimer(true, 1000 / _videoCodecInst.maxFramerate);
- }
- StartThread();
- return 0;
-}
-
-WebRtc_Word32 AviRecorder::StopRecording()
-{
- _timeEvent.StopTimer();
-
- StopThread();
- return FileRecorderImpl::StopRecording();
-}
-
-WebRtc_Word32 AviRecorder::CalcI420FrameSize( ) const
-{
- return 3 * _videoCodecInst.width * _videoCodecInst.height / 2;
-}
-
-WebRtc_Word32 AviRecorder::SetUpVideoEncoder()
-{
- // Size of unencoded data (I420) should be the largest possible frame size
- // in a file.
- _videoMaxPayloadSize = CalcI420FrameSize();
- _videoEncodedData.VerifyAndAllocate(_videoMaxPayloadSize);
-
- _videoCodecInst.plType = _videoEncoder->DefaultPayloadType(
- _videoCodecInst.plName);
-
- WebRtc_Word32 useNumberOfCores = 1;
- // Set the max payload size to 16000. This means that the codec will try to
- // create slices that will fit in 16000 kByte packets. However, the
- // Encode() call will still generate one full frame.
- if(_videoEncoder->SetEncodeCodec(_videoCodecInst, useNumberOfCores,
- 16000))
- {
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 AviRecorder::RecordVideoToFile(const VideoFrame& videoFrame)
-{
- CriticalSectionScoped lock(_critSec);
-
- if(!IsRecording() || ( videoFrame.Length() == 0))
- {
- return -1;
- }
- // The frame is written to file in AviRecorder::Process().
- WebRtc_Word32 retVal = _videoFramesQueue->AddFrame(videoFrame);
- if(retVal != 0)
- {
- StopRecording();
- }
- return retVal;
-}
-
-bool AviRecorder::StartThread()
-{
- unsigned int id;
- if( _thread == 0)
- {
- return false;
- }
-
- return _thread->Start(id);
-}
-
-bool AviRecorder::StopThread()
-{
- _critSec->Enter();
-
- if(_thread)
- {
- _thread->SetNotAlive();
-
- ThreadWrapper* thread = _thread;
- _thread = NULL;
-
- _timeEvent.Set();
-
- _critSec->Leave();
-
- if(thread->Stop())
- {
- delete thread;
- } else {
- return false;
- }
- } else {
- _critSec->Leave();
- }
- return true;
-}
-
-bool AviRecorder::Run( ThreadObj threadObj)
-{
- return static_cast<AviRecorder*>( threadObj)->Process();
-}
-
-WebRtc_Word32 AviRecorder::ProcessAudio()
-{
- if (_writtenVideoFramesCounter == 0)
- {
- // Get the most recent frame that is due for writing to file. Since
- // frames are unencoded it's safe to throw away frames if necessary
- // for synchronizing audio and video.
- VideoFrame* frameToProcess = _videoFramesQueue->FrameToRecord();
- if(frameToProcess)
- {
- // Syncronize audio to the current frame to process by throwing away
- // audio samples with older timestamp than the video frame.
- WebRtc_UWord32 numberOfAudioElements =
- _audioFramesToWrite.GetSize();
- for (WebRtc_UWord32 i = 0; i < numberOfAudioElements; ++i)
- {
- AudioFrameFileInfo* frameInfo =
- (AudioFrameFileInfo*)_audioFramesToWrite.First()->GetItem();
- if(frameInfo)
- {
- if(TickTime::TicksToMilliseconds(
- frameInfo->_playoutTS.Ticks()) <
- frameToProcess->RenderTimeMs())
- {
- delete frameInfo;
- _audioFramesToWrite.PopFront();
- } else
- {
- break;
- }
- }
- }
- }
- }
- // Write all audio up to current timestamp.
- WebRtc_Word32 error = 0;
- WebRtc_UWord32 numberOfAudioElements = _audioFramesToWrite.GetSize();
- for (WebRtc_UWord32 i = 0; i < numberOfAudioElements; ++i)
- {
- AudioFrameFileInfo* frameInfo =
- (AudioFrameFileInfo*)_audioFramesToWrite.First()->GetItem();
- if(frameInfo)
- {
- if((TickTime::Now() - frameInfo->_playoutTS).Milliseconds() > 0)
- {
- _moduleFile->IncomingAudioData(frameInfo->_audioData,
- frameInfo->_audioSize);
- _writtenAudioMS += frameInfo->_audioMS;
- delete frameInfo;
- _audioFramesToWrite.PopFront();
- } else {
- break;
- }
- } else {
- _audioFramesToWrite.PopFront();
- }
- }
- return error;
-}
-
-bool AviRecorder::Process()
-{
- switch(_timeEvent.Wait(500))
- {
- case kEventSignaled:
- if(_thread == NULL)
- {
- return false;
- }
- break;
- case kEventError:
- return false;
- case kEventTimeout:
- // No events triggered. No work to do.
- return true;
- }
- CriticalSectionScoped lock( _critSec);
-
- // Get the most recent frame to write to file (if any). Synchronize it with
- // the audio stream (if any). Synchronization the video based on its render
- // timestamp (i.e. VideoFrame::RenderTimeMS())
- VideoFrame* frameToProcess = _videoFramesQueue->FrameToRecord();
- if( frameToProcess == NULL)
- {
- return true;
- }
- WebRtc_Word32 error = 0;
- if(!_videoOnly)
- {
- if(!_firstAudioFrameReceived)
- {
- // Video and audio can only be synchronized if both have been
- // received.
- return true;
- }
- error = ProcessAudio();
-
- while (_writtenAudioMS > _writtenVideoMS)
- {
- error = EncodeAndWriteVideoToFile( *frameToProcess);
- if( error != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, _instanceID,
- "AviRecorder::Process() error writing to file.");
- break;
- } else {
- WebRtc_UWord32 frameLengthMS = 1000 /
- _videoCodecInst.maxFramerate;
- _writtenVideoFramesCounter++;
- _writtenVideoMS += frameLengthMS;
- // A full seconds worth of frames have been written.
- if(_writtenVideoFramesCounter%_videoCodecInst.maxFramerate == 0)
- {
- // Frame rate is in frames per seconds. Frame length is
- // calculated as an integer division which means it may
- // be rounded down. Compensate for this every second.
- WebRtc_UWord32 rest = 1000 % frameLengthMS;
- _writtenVideoMS += rest;
- }
- }
- }
- } else {
- // Frame rate is in frames per seconds. Frame length is calculated as an
- // integer division which means it may be rounded down. This introduces
- // drift. Once a full frame worth of drift has happened, skip writing
- // one frame. Note that frame rate is in frames per second so the
- // drift is completely compensated for.
- WebRtc_UWord32 frameLengthMS = 1000/_videoCodecInst.maxFramerate;
- WebRtc_UWord32 restMS = 1000 % frameLengthMS;
- WebRtc_UWord32 frameSkip = (_videoCodecInst.maxFramerate *
- frameLengthMS) / restMS;
-
- _writtenVideoFramesCounter++;
- if(_writtenVideoFramesCounter % frameSkip == 0)
- {
- _writtenVideoMS += frameLengthMS;
- return true;
- }
-
- error = EncodeAndWriteVideoToFile( *frameToProcess);
- if(error != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, _instanceID,
- "AviRecorder::Process() error writing to file.");
- } else {
- _writtenVideoMS += frameLengthMS;
- }
- }
- return error == 0;
-}
-
-WebRtc_Word32 AviRecorder::EncodeAndWriteVideoToFile(VideoFrame& videoFrame)
-{
- if(!IsRecording() || (videoFrame.Length() == 0))
- {
- return -1;
- }
-
- if(_frameScaler->ResizeFrameIfNeeded(&videoFrame, _videoCodecInst.width,
- _videoCodecInst.height) != 0)
- {
- return -1;
- }
-
- _videoEncodedData.payloadSize = 0;
-
- if( STR_CASE_CMP(_videoCodecInst.plName, "I420") == 0)
- {
- _videoEncodedData.VerifyAndAllocate(videoFrame.Length());
-
- // I420 is raw data. No encoding needed (each sample is represented by
- // 1 byte so there is no difference depending on endianness).
- memcpy(_videoEncodedData.payloadData, videoFrame.Buffer(),
- videoFrame.Length());
-
- _videoEncodedData.payloadSize = videoFrame.Length();
- _videoEncodedData.frameType = kVideoFrameKey;
- }else {
- if( _videoEncoder->Encode(videoFrame, _videoEncodedData) != 0)
- {
- return -1;
- }
- }
-
- if(_videoEncodedData.payloadSize > 0)
- {
- if(_moduleFile->IncomingAVIVideoData(
- (WebRtc_Word8*)(_videoEncodedData.payloadData),
- _videoEncodedData.payloadSize))
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, _instanceID,
- "Error writing AVI file");
- return -1;
- }
- } else {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- _instanceID,
- "FileRecorder::RecordVideoToFile() frame dropped by encoder bitrate\
- likely to low.");
- }
- return 0;
-}
-
-// Store audio frame in the _audioFramesToWrite buffer. The writing to file
-// happens in AviRecorder::Process().
-WebRtc_Word32 AviRecorder::WriteEncodedAudioData(
- const WebRtc_Word8* audioBuffer,
- WebRtc_UWord16 bufferLength,
- WebRtc_UWord16 millisecondsOfData,
- const TickTime* playoutTS)
-{
- if (!IsRecording())
- {
- return -1;
- }
- if (bufferLength > MAX_AUDIO_BUFFER_IN_BYTES)
- {
- return -1;
- }
- if (_videoOnly)
- {
- return -1;
- }
- if (_audioFramesToWrite.GetSize() > kMaxAudioBufferQueueLength)
- {
- StopRecording();
- return -1;
- }
- _firstAudioFrameReceived = true;
-
- if(playoutTS)
- {
- _audioFramesToWrite.PushBack(new AudioFrameFileInfo(audioBuffer,
- bufferLength,
- millisecondsOfData,
- *playoutTS));
- } else {
- _audioFramesToWrite.PushBack(new AudioFrameFileInfo(audioBuffer,
- bufferLength,
- millisecondsOfData,
- TickTime::Now()));
- }
- _timeEvent.Set();
- return 0;
-}
-
-#endif // WEBRTC_MODULE_UTILITY_VIDEO
-} // namespace webrtc
diff --git a/trunk/src/modules/utility/source/file_recorder_impl.h b/trunk/src/modules/utility/source/file_recorder_impl.h
deleted file mode 100644
index 3684166..0000000
--- a/trunk/src/modules/utility/source/file_recorder_impl.h
+++ /dev/null
@@ -1,164 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file contains a class that can write audio and/or video to file in
-// multiple file formats. The unencoded input data is written to file in the
-// encoded format specified.
-
-#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
-#define WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
-
-#include "coder.h"
-#include "common_types.h"
-#include "engine_configurations.h"
-#include "event_wrapper.h"
-#include "file_recorder.h"
-#include "media_file_defines.h"
-#include "media_file.h"
-#include "module_common_types.h"
-#include "resampler.h"
-#include "thread_wrapper.h"
-#include "tick_util.h"
-#include "typedefs.h"
-
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
- #include "frame_scaler.h"
- #include "video_coder.h"
- #include "video_frames_queue.h"
-#endif
-
-namespace webrtc {
-// The largest decoded frame size in samples (60ms with 32kHz sample rate).
-enum { MAX_AUDIO_BUFFER_IN_SAMPLES = 60*32};
-enum { MAX_AUDIO_BUFFER_IN_BYTES = MAX_AUDIO_BUFFER_IN_SAMPLES*2};
-enum { kMaxAudioBufferQueueLength = 100 };
-
-class FileRecorderImpl : public FileRecorder
-{
-public:
- FileRecorderImpl(WebRtc_UWord32 instanceID, FileFormats fileFormat);
- virtual ~FileRecorderImpl();
-
- // FileRecorder functions.
- virtual WebRtc_Word32 RegisterModuleFileCallback(FileCallback* callback);
- virtual FileFormats RecordingFileFormat() const;
- virtual WebRtc_Word32 StartRecordingAudioFile(
- const char* fileName,
- const CodecInst& codecInst,
- WebRtc_UWord32 notificationTimeMs,
- ACMAMRPackingFormat amrFormat = AMRFileStorage);
- virtual WebRtc_Word32 StartRecordingAudioFile(
- OutStream& destStream,
- const CodecInst& codecInst,
- WebRtc_UWord32 notificationTimeMs,
- ACMAMRPackingFormat amrFormat = AMRFileStorage);
- virtual WebRtc_Word32 StopRecording();
- virtual bool IsRecording() const;
- virtual WebRtc_Word32 codec_info(CodecInst& codecInst) const;
- virtual WebRtc_Word32 RecordAudioToFile(
- const AudioFrame& frame,
- const TickTime* playoutTS = NULL);
- virtual WebRtc_Word32 StartRecordingVideoFile(
- const char* fileName,
- const CodecInst& audioCodecInst,
- const VideoCodec& videoCodecInst,
- ACMAMRPackingFormat amrFormat = AMRFileStorage,
- bool videoOnly = false)
- {
- return -1;
- }
- virtual WebRtc_Word32 RecordVideoToFile(const VideoFrame& videoFrame)
- {
- return -1;
- }
-
-protected:
- virtual WebRtc_Word32 WriteEncodedAudioData(
- const WebRtc_Word8* audioBuffer,
- WebRtc_UWord16 bufferLength,
- WebRtc_UWord16 millisecondsOfData,
- const TickTime* playoutTS);
-
- WebRtc_Word32 SetUpAudioEncoder();
-
- WebRtc_UWord32 _instanceID;
- FileFormats _fileFormat;
- MediaFile* _moduleFile;
-
-private:
- OutStream* _stream;
- CodecInst codec_info_;
- ACMAMRPackingFormat _amrFormat;
-
- WebRtc_Word8 _audioBuffer[MAX_AUDIO_BUFFER_IN_BYTES];
- AudioCoder _audioEncoder;
- Resampler _audioResampler;
-};
-
-
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
-class AviRecorder : public FileRecorderImpl
-{
-public:
- AviRecorder(WebRtc_UWord32 instanceID, FileFormats fileFormat);
- virtual ~AviRecorder();
-
- // FileRecorder functions.
- virtual WebRtc_Word32 StartRecordingVideoFile(
- const char* fileName,
- const CodecInst& audioCodecInst,
- const VideoCodec& videoCodecInst,
- ACMAMRPackingFormat amrFormat = AMRFileStorage,
- bool videoOnly = false);
- virtual WebRtc_Word32 StopRecording();
- virtual WebRtc_Word32 RecordVideoToFile(const VideoFrame& videoFrame);
-
-protected:
- virtual WebRtc_Word32 WriteEncodedAudioData(
- const WebRtc_Word8* audioBuffer,
- WebRtc_UWord16 bufferLength,
- WebRtc_UWord16 millisecondsOfData,
- const TickTime* playoutTS);
-private:
- static bool Run(ThreadObj threadObj);
- bool Process();
-
- bool StartThread();
- bool StopThread();
-
- WebRtc_Word32 EncodeAndWriteVideoToFile(VideoFrame& videoFrame);
- WebRtc_Word32 ProcessAudio();
-
- WebRtc_Word32 CalcI420FrameSize() const;
- WebRtc_Word32 SetUpVideoEncoder();
-
- VideoCodec _videoCodecInst;
- bool _videoOnly;
-
- ListWrapper _audioFramesToWrite;
- bool _firstAudioFrameReceived;
-
- VideoFramesQueue* _videoFramesQueue;
-
- FrameScaler* _frameScaler;
- VideoCoder* _videoEncoder;
- WebRtc_Word32 _videoMaxPayloadSize;
- EncodedVideoData _videoEncodedData;
-
- ThreadWrapper* _thread;
- EventWrapper& _timeEvent;
- CriticalSectionWrapper* _critSec;
- WebRtc_Word64 _writtenVideoFramesCounter;
- WebRtc_Word64 _writtenAudioMS;
- WebRtc_Word64 _writtenVideoMS;
-};
-#endif // WEBRTC_MODULE_UTILITY_VIDEO
-} // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_SOURCE_FILE_RECORDER_IMPL_H_
diff --git a/trunk/src/modules/utility/source/frame_scaler.cc b/trunk/src/modules/utility/source/frame_scaler.cc
deleted file mode 100644
index c012e89..0000000
--- a/trunk/src/modules/utility/source/frame_scaler.cc
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/utility/source/frame_scaler.h"
-
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
-
-#include "common_video/libyuv/include/scaler.h"
-#include "system_wrappers/interface/trace.h"
-
-namespace webrtc {
-
-FrameScaler::FrameScaler()
- : scaler_(new Scaler()),
- scaled_frame_() {}
-
-FrameScaler::~FrameScaler() {}
-
-int FrameScaler::ResizeFrameIfNeeded(VideoFrame* video_frame,
- WebRtc_UWord32 out_width,
- WebRtc_UWord32 out_height) {
- if (video_frame->Length() == 0) {
- return -1;
- }
-
- if ((video_frame->Width() != out_width) ||
- (video_frame->Height() != out_height)) {
- // Set correct scale settings and scale |video_frame| into |scaled_frame_|.
- scaler_->Set(video_frame->Width(), video_frame->Height(), out_width,
- out_height, kI420, kI420, kScaleBox);
- int out_length = CalcBufferSize(kI420, out_width, out_height);
- scaled_frame_.VerifyAndAllocate(out_length);
- int ret = scaler_->Scale(video_frame->Buffer(), scaled_frame_.Buffer(),
- out_length);
- if (ret < 0) {
- return ret;
- }
-
- scaled_frame_.SetWidth(out_width);
- scaled_frame_.SetHeight(out_height);
- scaled_frame_.SetLength(out_length);
- scaled_frame_.SetRenderTime(video_frame->RenderTimeMs());
- scaled_frame_.SetTimeStamp(video_frame->TimeStamp());
- video_frame->SwapFrame(scaled_frame_);
- }
- return 0;
-}
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULE_UTILITY_VIDEO
diff --git a/trunk/src/modules/utility/source/frame_scaler.h b/trunk/src/modules/utility/source/frame_scaler.h
deleted file mode 100644
index f86a933..0000000
--- a/trunk/src/modules/utility/source/frame_scaler.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file implements a class that can be used for scaling frames.
-
-#ifndef WEBRTC_MODULES_UTILITY_SOURCE_FRAME_SCALER_H_
-#define WEBRTC_MODULES_UTILITY_SOURCE_FRAME_SCALER_H_
-
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
-
-#include "engine_configurations.h"
-#include "modules/interface/module_common_types.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-
-namespace webrtc {
-
-class Scaler;
-class VideoFrame;
-
-class FrameScaler {
- public:
- FrameScaler();
- ~FrameScaler();
-
- // Re-sizes |video_frame| so that it has the width |out_width| and height
- // |out_height|.
- int ResizeFrameIfNeeded(VideoFrame* video_frame,
- WebRtc_UWord32 out_width,
- WebRtc_UWord32 out_height);
-
- private:
- scoped_ptr<Scaler> scaler_;
- VideoFrame scaled_frame_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULE_UTILITY_VIDEO
-
-#endif // WEBRTC_MODULES_UTILITY_SOURCE_FRAME_SCALER_H_
diff --git a/trunk/src/modules/utility/source/process_thread_impl.cc b/trunk/src/modules/utility/source/process_thread_impl.cc
deleted file mode 100644
index 9028c44..0000000
--- a/trunk/src/modules/utility/source/process_thread_impl.cc
+++ /dev/null
@@ -1,188 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "process_thread_impl.h"
-#include "module.h"
-#include "trace.h"
-
-namespace webrtc {
-ProcessThread::~ProcessThread()
-{
-}
-
-ProcessThread* ProcessThread::CreateProcessThread()
-{
- return new ProcessThreadImpl();
-}
-
-void ProcessThread::DestroyProcessThread(ProcessThread* module)
-{
- delete module;
-}
-
-ProcessThreadImpl::ProcessThreadImpl()
- : _timeEvent(*EventWrapper::Create()),
- _critSectModules(CriticalSectionWrapper::CreateCriticalSection()),
- _thread(NULL)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1, "%s created", __FUNCTION__);
-}
-
-ProcessThreadImpl::~ProcessThreadImpl()
-{
- delete _critSectModules;
- delete &_timeEvent;
- WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1, "%s deleted", __FUNCTION__);
-}
-
-WebRtc_Word32 ProcessThreadImpl::Start()
-{
- CriticalSectionScoped lock(_critSectModules);
- if(_thread)
- {
- return -1;
- }
- _thread = ThreadWrapper::CreateThread(Run, this, kNormalPriority,
- "ProcessThread");
- unsigned int id;
- WebRtc_Word32 retVal = _thread->Start(id);
- if(retVal >= 0)
- {
- return 0;
- }
- delete _thread;
- _thread = NULL;
- return -1;
-}
-
-WebRtc_Word32 ProcessThreadImpl::Stop()
-{
- _critSectModules->Enter();
- if(_thread)
- {
- _thread->SetNotAlive();
-
- ThreadWrapper* thread = _thread;
- _thread = NULL;
-
- _timeEvent.Set();
- _critSectModules->Leave();
-
- if(thread->Stop())
- {
- delete thread;
- } else {
- return -1;
- }
- } else {
- _critSectModules->Leave();
- }
- return 0;
-}
-
-WebRtc_Word32 ProcessThreadImpl::RegisterModule(const Module* module)
-{
- CriticalSectionScoped lock(_critSectModules);
-
- // Only allow module to be registered once.
- ListItem* item = _modules.First();
- for(WebRtc_UWord32 i = 0; i < _modules.GetSize() && item; i++)
- {
- if(module == item->GetItem())
- {
- return -1;
- }
- item = _modules.Next(item);
- }
-
- _modules.PushFront(module);
- WEBRTC_TRACE(kTraceInfo, kTraceUtility, -1,
- "number of registered modules has increased to %d",
- _modules.GetSize());
- // Wake the thread calling ProcessThreadImpl::Process() to update the
- // waiting time. The waiting time for the just registered module may be
- // shorter than all other registered modules.
- _timeEvent.Set();
- return 0;
-}
-
-WebRtc_Word32 ProcessThreadImpl::DeRegisterModule(const Module* module)
-{
- CriticalSectionScoped lock(_critSectModules);
-
- ListItem* item = _modules.First();
- for(WebRtc_UWord32 i = 0; i < _modules.GetSize() && item; i++)
- {
- if(module == item->GetItem())
- {
- int res = _modules.Erase(item);
- WEBRTC_TRACE(kTraceInfo, kTraceUtility, -1,
- "number of registered modules has decreased to %d",
- _modules.GetSize());
- return res;
- }
- item = _modules.Next(item);
- }
- return -1;
-}
-
-bool ProcessThreadImpl::Run(void* obj)
-{
- return static_cast<ProcessThreadImpl*>(obj)->Process();
-}
-
-bool ProcessThreadImpl::Process()
-{
- // Wait for the module that should be called next, but don't block thread
- // longer than 100 ms.
- WebRtc_Word32 minTimeToNext = 100;
- {
- CriticalSectionScoped lock(_critSectModules);
- ListItem* item = _modules.First();
- for(WebRtc_UWord32 i = 0; i < _modules.GetSize() && item; i++)
- {
- WebRtc_Word32 timeToNext =
- static_cast<Module*>(item->GetItem())->TimeUntilNextProcess();
- if(minTimeToNext > timeToNext)
- {
- minTimeToNext = timeToNext;
- }
- item = _modules.Next(item);
- }
- }
-
- if(minTimeToNext > 0)
- {
- if(kEventError == _timeEvent.Wait(minTimeToNext))
- {
- return true;
- }
- if(!_thread)
- {
- return false;
- }
- }
- {
- CriticalSectionScoped lock(_critSectModules);
- ListItem* item = _modules.First();
- for(WebRtc_UWord32 i = 0; i < _modules.GetSize() && item; i++)
- {
- WebRtc_Word32 timeToNext =
- static_cast<Module*>(item->GetItem())->TimeUntilNextProcess();
- if(timeToNext < 1)
- {
- static_cast<Module*>(item->GetItem())->Process();
- }
- item = _modules.Next(item);
- }
- }
- return true;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/utility/source/process_thread_impl.h b/trunk/src/modules/utility/source/process_thread_impl.h
deleted file mode 100644
index 79b1272..0000000
--- a/trunk/src/modules/utility/source/process_thread_impl.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UTILITY_SOURCE_PROCESS_THREAD_IMPL_H_
-#define WEBRTC_MODULES_UTILITY_SOURCE_PROCESS_THREAD_IMPL_H_
-
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "list_wrapper.h"
-#include "process_thread.h"
-#include "thread_wrapper.h"
-#include "typedefs.h"
-
-namespace webrtc {
-class ProcessThreadImpl : public ProcessThread
-{
-public:
- ProcessThreadImpl();
- virtual ~ProcessThreadImpl();
-
- virtual WebRtc_Word32 Start();
- virtual WebRtc_Word32 Stop();
-
- virtual WebRtc_Word32 RegisterModule(const Module* module);
- virtual WebRtc_Word32 DeRegisterModule(const Module* module);
-
-protected:
- static bool Run(void* obj);
-
- bool Process();
-
-private:
- EventWrapper& _timeEvent;
- CriticalSectionWrapper* _critSectModules;
- ListWrapper _modules;
- ThreadWrapper* _thread;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_UTILITY_SOURCE_PROCESS_THREAD_IMPL_H_
diff --git a/trunk/src/modules/utility/source/rtp_dump_impl.cc b/trunk/src/modules/utility/source/rtp_dump_impl.cc
deleted file mode 100644
index 69a52ec..0000000
--- a/trunk/src/modules/utility/source/rtp_dump_impl.cc
+++ /dev/null
@@ -1,282 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rtp_dump_impl.h"
-
-#include <cassert>
-#include <stdio.h>
-
-#include "critical_section_wrapper.h"
-#include "trace.h"
-
-#if defined(_WIN32)
-#include <Windows.h>
-#include <mmsystem.h>
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
-#include <string.h>
-#include <sys/time.h>
-#include <time.h>
-#endif
-
-#if (defined(_DEBUG) && defined(_WIN32))
-#define DEBUG_PRINT(expr) OutputDebugString(##expr)
-#define DEBUG_PRINTP(expr, p) \
-{ \
- char msg[128]; \
- sprintf(msg, ##expr, p); \
- OutputDebugString(msg); \
-}
-#else
-#define DEBUG_PRINT(expr) ((void)0)
-#define DEBUG_PRINTP(expr,p) ((void)0)
-#endif // defined(_DEBUG) && defined(_WIN32)
-
-namespace webrtc {
-const char RTPFILE_VERSION[] = "1.0";
-const WebRtc_UWord32 MAX_UWORD32 = 0xffffffff;
-
-// This stucture is specified in the rtpdump documentation.
-// This struct corresponds to RD_packet_t in
-// http://www.cs.columbia.edu/irt/software/rtptools/
-typedef struct
-{
- // Length of packet, including this header (may be smaller than plen if not
- // whole packet recorded).
- WebRtc_UWord16 length;
- // Actual header+payload length for RTP, 0 for RTCP.
- WebRtc_UWord16 plen;
- // Milliseconds since the start of recording.
- WebRtc_UWord32 offset;
-} rtpDumpPktHdr_t;
-
-RtpDump* RtpDump::CreateRtpDump()
-{
- return new RtpDumpImpl();
-}
-
-void RtpDump::DestroyRtpDump(RtpDump* object)
-{
- delete object;
-}
-
-RtpDumpImpl::RtpDumpImpl()
- : _critSect(CriticalSectionWrapper::CreateCriticalSection()),
- _file(*FileWrapper::Create()),
- _startTime(0)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1, "%s created", __FUNCTION__);
-}
-
-RtpDump::~RtpDump()
-{
-}
-
-RtpDumpImpl::~RtpDumpImpl()
-{
- _file.Flush();
- _file.CloseFile();
- delete &_file;
- delete _critSect;
- WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1, "%s deleted", __FUNCTION__);
-}
-
-WebRtc_Word32 RtpDumpImpl::Start(const char* fileNameUTF8)
-{
-
- if (fileNameUTF8 == NULL)
- {
- return -1;
- }
-
- CriticalSectionScoped lock(_critSect);
- _file.Flush();
- _file.CloseFile();
- if (_file.OpenFile(fileNameUTF8, false, false, false) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
- "failed to open the specified file");
- return -1;
- }
-
- // Store start of RTP dump (to be used for offset calculation later).
- _startTime = GetTimeInMS();
-
- // All rtp dump files start with #!rtpplay.
- char magic[16];
- sprintf(magic, "#!rtpplay%s \n", RTPFILE_VERSION);
- if (_file.WriteText(magic) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
- "error writing to file");
- return -1;
- }
-
- // The header according to the rtpdump documentation is sizeof(RD_hdr_t)
- // which is 8 + 4 + 2 = 14 bytes for 32-bit architecture (and 22 bytes on
- // 64-bit architecture). However, Wireshark use 16 bytes for the header
- // regardless of if the binary is 32-bit or 64-bit. Go by the same approach
- // as Wireshark since it makes more sense.
- // http://wiki.wireshark.org/rtpdump explains that an additional 2 bytes
- // of padding should be added to the header.
- char dummyHdr[16];
- memset(dummyHdr, 0, 16);
- if (!_file.Write(dummyHdr, sizeof(dummyHdr)))
- {
- WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
- "error writing to file");
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 RtpDumpImpl::Stop()
-{
- CriticalSectionScoped lock(_critSect);
- _file.Flush();
- _file.CloseFile();
- return 0;
-}
-
-bool RtpDumpImpl::IsActive() const
-{
- CriticalSectionScoped lock(_critSect);
- return _file.Open();
-}
-
-WebRtc_Word32 RtpDumpImpl::DumpPacket(const WebRtc_UWord8* packet,
- WebRtc_UWord16 packetLength)
-{
- CriticalSectionScoped lock(_critSect);
- if (!IsActive())
- {
- return 0;
- }
-
- if (packet == NULL)
- {
- return -1;
- }
-
- if (packetLength < 1)
- {
- return -1;
- }
-
- // If the packet doesn't contain a valid RTCP header the packet will be
- // considered RTP (without further verification).
- bool isRTCP = RTCP(packet);
-
- rtpDumpPktHdr_t hdr;
- WebRtc_UWord32 offset;
-
- // Offset is relative to when recording was started.
- offset = GetTimeInMS();
- if (offset < _startTime)
- {
- // Compensate for wraparound.
- offset += MAX_UWORD32 - _startTime + 1;
- } else {
- offset -= _startTime;
- }
- hdr.offset = RtpDumpHtonl(offset);
-
- hdr.length = RtpDumpHtons((WebRtc_UWord16)(packetLength + sizeof(hdr)));
- if (isRTCP)
- {
- hdr.plen = 0;
- }
- else
- {
- hdr.plen = RtpDumpHtons((WebRtc_UWord16)packetLength);
- }
-
- if (!_file.Write(&hdr, sizeof(hdr)))
- {
- WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
- "error writing to file");
- return -1;
- }
- if (!_file.Write(packet, packetLength))
- {
- WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
- "error writing to file");
- return -1;
- }
-
- return 0;
-}
-
-bool RtpDumpImpl::RTCP(const WebRtc_UWord8* packet) const
-{
- const WebRtc_UWord8 payloadType = packet[1];
- bool is_rtcp = false;
-
- switch(payloadType)
- {
- case 192:
- is_rtcp = true;
- break;
- case 193: case 195:
- break;
- case 200: case 201: case 202: case 203:
- case 204: case 205: case 206: case 207:
- is_rtcp = true;
- break;
- }
- return is_rtcp;
-}
-
-// TODO (hellner): why is TickUtil not used here?
-inline WebRtc_UWord32 RtpDumpImpl::GetTimeInMS() const
-{
-#if defined(_WIN32)
- return timeGetTime();
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
- struct timeval tv;
- struct timezone tz;
- unsigned long val;
-
- gettimeofday(&tv, &tz);
- val = tv.tv_sec * 1000 + tv.tv_usec / 1000;
- return val;
-#else
- #error Either _WIN32 or LINUX or WEBRTC_MAC has to be defined!
- assert(false);
- return 0;
-#endif
-}
-
-inline WebRtc_UWord32 RtpDumpImpl::RtpDumpHtonl(WebRtc_UWord32 x) const
-{
-#if defined(WEBRTC_BIG_ENDIAN)
- return x;
-#elif defined(WEBRTC_LITTLE_ENDIAN)
- return (x >> 24) + ((((x >> 16) & 0xFF) << 8) + ((((x >> 8) & 0xFF) << 16) +
- ((x & 0xFF) << 24)));
-#else
-#error Either WEBRTC_BIG_ENDIAN or WEBRTC_LITTLE_ENDIAN has to be defined!
- assert(false);
- return 0;
-#endif
-}
-
-inline WebRtc_UWord16 RtpDumpImpl::RtpDumpHtons(WebRtc_UWord16 x) const
-{
-#if defined(WEBRTC_BIG_ENDIAN)
- return x;
-#elif defined(WEBRTC_LITTLE_ENDIAN)
- return (x >> 8) + ((x & 0xFF) << 8);
-#else
- #error Either WEBRTC_BIG_ENDIAN or WEBRTC_LITTLE_ENDIAN has to be defined!
- assert(false);
- return 0;
-#endif
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/utility/source/rtp_dump_impl.h b/trunk/src/modules/utility/source/rtp_dump_impl.h
deleted file mode 100644
index 9715c35..0000000
--- a/trunk/src/modules/utility/source/rtp_dump_impl.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_
-#define WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_
-
-#include "rtp_dump.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class FileWrapper;
-class RtpDumpImpl : public RtpDump
-{
-public:
- RtpDumpImpl();
- virtual ~RtpDumpImpl();
-
- virtual WebRtc_Word32 Start(const char* fileNameUTF8);
- virtual WebRtc_Word32 Stop();
- virtual bool IsActive() const;
- virtual WebRtc_Word32 DumpPacket(const WebRtc_UWord8* packet,
- WebRtc_UWord16 packetLength);
-private:
- // Return the system time in ms.
- inline WebRtc_UWord32 GetTimeInMS() const;
- // Return x in network byte order (big endian).
- inline WebRtc_UWord32 RtpDumpHtonl(WebRtc_UWord32 x) const;
- // Return x in network byte order (big endian).
- inline WebRtc_UWord16 RtpDumpHtons(WebRtc_UWord16 x) const;
-
- // Return true if the packet starts with a valid RTCP header.
- // Note: See ModuleRTPUtility::RTPHeaderParser::RTCP() for details on how
- // to determine if the packet is an RTCP packet.
- bool RTCP(const WebRtc_UWord8* packet) const;
-
-private:
- CriticalSectionWrapper* _critSect;
- FileWrapper& _file;
- WebRtc_UWord32 _startTime;
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_UTILITY_SOURCE_RTP_DUMP_IMPL_H_
diff --git a/trunk/src/modules/utility/source/utility.gypi b/trunk/src/modules/utility/source/utility.gypi
deleted file mode 100644
index 7dcce8e..0000000
--- a/trunk/src/modules/utility/source/utility.gypi
+++ /dev/null
@@ -1,92 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'webrtc_utility',
- 'type': '<(library)',
- 'dependencies': [
- 'audio_coding_module',
- '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '../interface',
- '../../interface',
- '../../media_file/interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- '../../interface',
- '../../audio_coding/main/interface',
- ],
- },
- 'sources': [
- '../interface/file_player.h',
- '../interface/file_recorder.h',
- '../interface/process_thread.h',
- '../interface/rtp_dump.h',
- 'coder.cc',
- 'coder.h',
- 'file_player_impl.cc',
- 'file_player_impl.h',
- 'file_recorder_impl.cc',
- 'file_recorder_impl.h',
- 'process_thread_impl.cc',
- 'process_thread_impl.h',
- 'rtp_dump_impl.cc',
- 'rtp_dump_impl.h',
- ],
- 'conditions': [
- ['enable_video==1', {
- # Adds support for video recording.
- 'defines': [
- 'WEBRTC_MODULE_UTILITY_VIDEO',
- ],
- 'dependencies': [
- 'webrtc_video_coding',
- ],
- 'include_dirs': [
- '../../video_coding/main/interface',
- ],
- 'sources': [
- 'frame_scaler.cc',
- 'video_coder.cc',
- 'video_frames_queue.cc',
- ],
- }],
- ],
- },
- ], # targets
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'webrtc_utility_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'webrtc_utility',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- ],
- 'sources': [
- 'file_player_unittest.cc',
- ],
- }, # webrtc_utility_unittests
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/utility/source/video_coder.cc b/trunk/src/modules/utility/source/video_coder.cc
deleted file mode 100644
index e17c3e0..0000000
--- a/trunk/src/modules/utility/source/video_coder.cc
+++ /dev/null
@@ -1,152 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
-
-#include "video_coder.h"
-
-namespace webrtc {
-VideoCoder::VideoCoder(WebRtc_UWord32 instanceID)
- : _instanceID( instanceID),
- _vcm(VideoCodingModule::Create(instanceID)),
- _decodedVideo(0)
-{
- _vcm->InitializeSender();
- _vcm->InitializeReceiver();
-
- _vcm->RegisterTransportCallback(this);
- _vcm->RegisterReceiveCallback(this);
-}
-
-VideoCoder::~VideoCoder()
-{
- VideoCodingModule::Destroy(_vcm);
-}
-
-WebRtc_Word32 VideoCoder::ResetDecoder()
-{
- _vcm->ResetDecoder();
-
- _vcm->InitializeSender();
- _vcm->InitializeReceiver();
-
- _vcm->RegisterTransportCallback(this);
- _vcm->RegisterReceiveCallback(this);
- return 0;
-}
-
-WebRtc_Word32 VideoCoder::SetEncodeCodec(VideoCodec& videoCodecInst,
- WebRtc_UWord32 numberOfCores,
- WebRtc_UWord32 maxPayloadSize)
-{
- if(_vcm->RegisterSendCodec(&videoCodecInst, numberOfCores,
- maxPayloadSize) != VCM_OK)
- {
- return -1;
- }
- return 0;
-}
-
-
-WebRtc_Word32 VideoCoder::SetDecodeCodec(VideoCodec& videoCodecInst,
- WebRtc_Word32 numberOfCores)
-{
- if (videoCodecInst.plType == 0)
- {
- WebRtc_Word8 plType = DefaultPayloadType(videoCodecInst.plName);
- if (plType == -1)
- {
- return -1;
- }
- videoCodecInst.plType = plType;
- }
-
- if(_vcm->RegisterReceiveCodec(&videoCodecInst, numberOfCores) != VCM_OK)
- {
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 VideoCoder::Decode(VideoFrame& decodedVideo,
- const EncodedVideoData& encodedData)
-{
- decodedVideo.SetLength(0);
- if(encodedData.payloadSize <= 0)
- {
- return -1;
- }
-
- _decodedVideo = &decodedVideo;
- if(_vcm->DecodeFromStorage(encodedData) != VCM_OK)
- {
- return -1;
- }
- return 0;
-}
-
-
-WebRtc_Word32 VideoCoder::Encode(const VideoFrame& videoFrame,
- EncodedVideoData& videoEncodedData)
-{
- // The AddVideoFrame(..) call will (indirectly) call SendData(). Store a
- // pointer to videoFrame so that it can be updated.
- _videoEncodedData = &videoEncodedData;
- videoEncodedData.payloadSize = 0;
- if(_vcm->AddVideoFrame(videoFrame) != VCM_OK)
- {
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word8 VideoCoder::DefaultPayloadType(const char* plName)
-{
- VideoCodec tmpCodec;
- WebRtc_Word32 numberOfCodecs = _vcm->NumberOfCodecs();
- for (WebRtc_UWord8 i = 0; i < numberOfCodecs; i++)
- {
- _vcm->Codec(i, &tmpCodec);
- if(strncmp(tmpCodec.plName, plName, kPayloadNameSize) == 0)
- {
- return tmpCodec.plType;
- }
- }
- return -1;
-}
-
-WebRtc_Word32 VideoCoder::FrameToRender(VideoFrame& videoFrame)
-{
- return _decodedVideo->CopyFrame(videoFrame);
-}
-
-WebRtc_Word32 VideoCoder::SendData(
- FrameType frameType,
- WebRtc_UWord8 payloadType,
- WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader& fragmentationHeader,
- const RTPVideoHeader* /*rtpVideoHdr*/)
-{
- // Store the data in _videoEncodedData which is a pointer to videoFrame in
- // Encode(..)
- _videoEncodedData->VerifyAndAllocate(payloadSize);
- _videoEncodedData->frameType = frameType;
- _videoEncodedData->payloadType = payloadType;
- _videoEncodedData->timeStamp = timeStamp;
- _videoEncodedData->fragmentationHeader = fragmentationHeader;
- memcpy(_videoEncodedData->payloadData, payloadData,
- sizeof(WebRtc_UWord8) * payloadSize);
- _videoEncodedData->payloadSize = payloadSize;
- return 0;
-}
-} // namespace webrtc
-#endif // WEBRTC_MODULE_UTILITY_VIDEO
diff --git a/trunk/src/modules/utility/source/video_coder.h b/trunk/src/modules/utility/source/video_coder.h
deleted file mode 100644
index 5c4b0ae..0000000
--- a/trunk/src/modules/utility/source/video_coder.h
+++ /dev/null
@@ -1,69 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UTILITY_SOURCE_VIDEO_CODER_H_
-#define WEBRTC_MODULES_UTILITY_SOURCE_VIDEO_CODER_H_
-
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
-
-#include "engine_configurations.h"
-#include "video_coding.h"
-
-namespace webrtc {
-class VideoCoder : public VCMPacketizationCallback, public VCMReceiveCallback
-{
-public:
- VideoCoder(WebRtc_UWord32 instanceID);
- ~VideoCoder();
-
- WebRtc_Word32 ResetDecoder();
-
- WebRtc_Word32 SetEncodeCodec(VideoCodec& videoCodecInst,
- WebRtc_UWord32 numberOfCores,
- WebRtc_UWord32 maxPayloadSize);
-
-
- // Select the codec that should be used for decoding. videoCodecInst.plType
- // will be set to the codec's default payload type.
- WebRtc_Word32 SetDecodeCodec(VideoCodec& videoCodecInst,
- WebRtc_Word32 numberOfCores);
-
- WebRtc_Word32 Decode(VideoFrame& decodedVideo,
- const EncodedVideoData& encodedData);
-
- WebRtc_Word32 Encode(const VideoFrame& videoFrame,
- EncodedVideoData& videoEncodedData);
-
- WebRtc_Word8 DefaultPayloadType(const char* plName);
-
-private:
- // VCMReceiveCallback function.
- // Note: called by VideoCodingModule when decoding finished.
- WebRtc_Word32 FrameToRender(VideoFrame& videoFrame);
-
- // VCMPacketizationCallback function.
- // Note: called by VideoCodingModule when encoding finished.
- WebRtc_Word32 SendData(
- const FrameType /*frameType*/,
- const WebRtc_UWord8 /*payloadType*/,
- const WebRtc_UWord32 /*timeStamp*/,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader& /* fragmentationHeader*/,
- const RTPVideoHeader* rtpTypeHdr);
-
- WebRtc_UWord32 _instanceID;
- VideoCodingModule* _vcm;
- VideoFrame* _decodedVideo;
- EncodedVideoData* _videoEncodedData;
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULE_UTILITY_VIDEO
-#endif // WEBRTC_MODULES_UTILITY_SOURCE_VIDEO_CODER_H_
diff --git a/trunk/src/modules/utility/source/video_frames_queue.cc b/trunk/src/modules/utility/source/video_frames_queue.cc
deleted file mode 100644
index ab590c4..0000000
--- a/trunk/src/modules/utility/source/video_frames_queue.cc
+++ /dev/null
@@ -1,150 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_frames_queue.h"
-
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
-
-#include <cassert>
-
-#include "module_common_types.h"
-#include "tick_util.h"
-#include "trace.h"
-
-namespace webrtc {
-VideoFramesQueue::VideoFramesQueue()
- : _incomingFrames(),
- _renderDelayMs(10)
-{
-}
-
-VideoFramesQueue::~VideoFramesQueue()
-{
- while (!_incomingFrames.Empty())
- {
- ListItem* item = _incomingFrames.First();
- if (item)
- {
- VideoFrame* ptrFrame = static_cast<VideoFrame*>(item->GetItem());
- assert(ptrFrame != NULL);
- ptrFrame->Free();
- delete ptrFrame;
- }
- _incomingFrames.Erase(item);
- }
- while (!_emptyFrames.Empty())
- {
- ListItem* item = _emptyFrames.First();
- if (item)
- {
- VideoFrame* ptrFrame = static_cast<VideoFrame*>(item->GetItem());
- assert(ptrFrame != NULL);
- ptrFrame->Free();
- delete ptrFrame;
- }
- _emptyFrames.Erase(item);
- }
-}
-
-WebRtc_Word32 VideoFramesQueue::AddFrame(const VideoFrame& newFrame)
-{
- VideoFrame* ptrFrameToAdd = NULL;
- // Try to re-use a VideoFrame. Only allocate new memory if it is necessary.
- if (!_emptyFrames.Empty())
- {
- ListItem* item = _emptyFrames.First();
- if (item)
- {
- ptrFrameToAdd = static_cast<VideoFrame*>(item->GetItem());
- _emptyFrames.Erase(item);
- }
- }
- if (!ptrFrameToAdd)
- {
- if (_emptyFrames.GetSize() + _incomingFrames.GetSize() >
- KMaxNumberOfFrames)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, -1,
- "%s: too many frames, limit: %d", __FUNCTION__,
- KMaxNumberOfFrames);
- return -1;
- }
-
- WEBRTC_TRACE(kTraceMemory, kTraceVideoRenderer, -1,
- "%s: allocating buffer %d", __FUNCTION__,
- _emptyFrames.GetSize() + _incomingFrames.GetSize());
-
- ptrFrameToAdd = new VideoFrame();
- if (!ptrFrameToAdd)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
- "%s: could not create new frame for", __FUNCTION__);
- return -1;
- }
- }
- ptrFrameToAdd->CopyFrame(newFrame);
- _incomingFrames.PushBack(ptrFrameToAdd);
- return 0;
-}
-
-// Find the most recent frame that has a VideoFrame::RenderTimeMs() that is
-// lower than current time in ms (TickTime::MillisecondTimestamp()).
-// Note _incomingFrames is sorted so that the oldest frame is first.
-// Recycle all frames that are older than the most recent frame.
-VideoFrame* VideoFramesQueue::FrameToRecord()
-{
- VideoFrame* ptrRenderFrame = NULL;
- ListItem* item = _incomingFrames.First();
- while(item)
- {
- VideoFrame* ptrOldestFrameInList =
- static_cast<VideoFrame*>(item->GetItem());
- if (ptrOldestFrameInList->RenderTimeMs() <=
- TickTime::MillisecondTimestamp() + _renderDelayMs)
- {
- if (ptrRenderFrame)
- {
- // List is traversed beginning to end. If ptrRenderFrame is not
- // NULL it must be the first, and thus oldest, VideoFrame in the
- // queue. It can be recycled.
- ReturnFrame(ptrRenderFrame);
- _incomingFrames.PopFront();
- }
- item = _incomingFrames.Next(item);
- ptrRenderFrame = ptrOldestFrameInList;
- }else
- {
- // All VideoFrames following this one will be even newer. No match
- // will be found.
- break;
- }
- }
- return ptrRenderFrame;
-}
-
-WebRtc_Word32 VideoFramesQueue::ReturnFrame(VideoFrame* ptrOldFrame)
-{
- ptrOldFrame->SetTimeStamp(0);
- ptrOldFrame->SetWidth(0);
- ptrOldFrame->SetHeight(0);
- ptrOldFrame->SetRenderTime(0);
- ptrOldFrame->SetLength(0);
- _emptyFrames.PushBack(ptrOldFrame);
- return 0;
-}
-
-//
-WebRtc_Word32 VideoFramesQueue::SetRenderDelay(WebRtc_UWord32 renderDelay)
-{
- _renderDelayMs = renderDelay;
- return 0;
-}
-} // namespace webrtc
-#endif // WEBRTC_MODULE_UTILITY_VIDEO
diff --git a/trunk/src/modules/utility/source/video_frames_queue.h b/trunk/src/modules/utility/source/video_frames_queue.h
deleted file mode 100644
index 6c9be1c..0000000
--- a/trunk/src/modules/utility/source/video_frames_queue.h
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_UTILITY_SOURCE_VIDEO_FRAMES_QUEUE_H_
-#define WEBRTC_MODULES_UTILITY_SOURCE_VIDEO_FRAMES_QUEUE_H_
-
-#ifdef WEBRTC_MODULE_UTILITY_VIDEO
-
-#include "engine_configurations.h"
-#include "list_wrapper.h"
-#include "typedefs.h"
-
-namespace webrtc {
-class VideoFrame;
-
-class VideoFramesQueue
-{
-public:
- VideoFramesQueue();
- ~VideoFramesQueue();
-
- // Put newFrame (last) in the queue.
- WebRtc_Word32 AddFrame(const VideoFrame& newFrame);
-
- // Return the most current frame. I.e. the frame with the highest
- // VideoFrame::RenderTimeMs() that is lower than
- // TickTime::MillisecondTimestamp().
- VideoFrame* FrameToRecord();
-
- // Set the render delay estimate to renderDelay ms.
- WebRtc_Word32 SetRenderDelay(WebRtc_UWord32 renderDelay);
-
-protected:
- // Make ptrOldFrame available for re-use. I.e. put it in the empty frames
- // queue.
- WebRtc_Word32 ReturnFrame(VideoFrame* ptrOldFrame);
-
-private:
- // Don't allow the buffer to expand beyond KMaxNumberOfFrames VideoFrames.
- // 300 frames correspond to 10 seconds worth of frames at 30 fps.
- enum {KMaxNumberOfFrames = 300};
-
- // List of VideoFrame pointers. The list is sorted in the order of when the
- // VideoFrame was inserted into the list. The first VideoFrame in the list
- // was inserted first.
- ListWrapper _incomingFrames;
- // A list of frames that are free to be re-used.
- ListWrapper _emptyFrames;
-
- // Estimated render delay.
- WebRtc_UWord32 _renderDelayMs;
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULE_UTILITY_VIDEO
-#endif // WEBRTC_MODULES_UTILITY_SOURCE_VIDEO_FRAMES_QUEUE_H_
diff --git a/trunk/src/modules/utility/test/testAPI.cc b/trunk/src/modules/utility/test/testAPI.cc
deleted file mode 100644
index 96664a3..0000000
--- a/trunk/src/modules/utility/test/testAPI.cc
+++ /dev/null
@@ -1,368 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// testAPI.cpp : Defines the entry point for the console application.
-//
-// NOTES:
-// 1. MediaFile library and testAPI.cpp must be built in DEBUG mode for testing.
-//
-
-#include <iostream>
-#include <stdio.h>
-#include <assert.h>
-
-#ifdef WIN32
- #include <windows.h>
- #include <tchar.h>
-#endif
-
-#include "common_types.h"
-#include "trace.h"
-
-#include "Engineconfigurations.h"
-#include "media_file.h"
-#include "file_player.h"
-#include "file_recorder.h"
-
-
-bool notify = false, playing = false, recording = false;
-
-// callback class for FileModule
-class MyFileModuleCallback : public FileCallback
-{
-public:
- virtual void PlayNotification( const WebRtc_Word32 id,
- const WebRtc_UWord32 durationMs )
- {
- printf("\tReceived PlayNotification from module %ld, durationMs = %ld\n",
- id, durationMs);
- notify = true;
- };
-
- virtual void RecordNotification( const WebRtc_Word32 id,
- const WebRtc_UWord32 durationMs )
- {
- printf("\tReceived RecordNotification from module %ld, durationMs = %ld\n",
- id, durationMs);
- notify = true;
- };
-
- virtual void PlayFileEnded(const WebRtc_Word32 id)
- {
- printf("\tReceived PlayFileEnded notification from module %ld.\n", id);
- playing = false;
- };
-
- virtual void RecordFileEnded(const WebRtc_Word32 id)
- {
- printf("\tReceived RecordFileEnded notification from module %ld.\n", id);
- recording = false;
- }
-};
-
-// main test app
-#ifdef WIN32
-int _tmain(int argc, _TCHAR* argv[])
-#else
-int main(int /*argc*/, char** /*argv*/)
-#endif
-{
- Trace::CreateTrace();
- Trace::SetTraceFile("testTrace.txt");
- Trace::SetEncryptedTraceFile("testTraceDebug.txt");
-
- int playId = 1;
- int recordId = 2;
-
- printf("Welcome to test of FilePlayer and FileRecorder\n");
-
-
- ///////////////////////////////////////////////
- //
- // avi test case 1
- //
- ///////////////////////////////////////////////
-
-
- // todo PW we need more AVI tests Mp4
-
- {
- FilePlayer& filePlayer(*FilePlayer::CreateFilePlayer(1, webrtc::kFileFormatAviFile));
- FileRecorder& fileRecorder(*FileRecorder::CreateFileRecorder(1, webrtc::kFileFormatAviFile));
-
- const char* KFileName = "./tmpAviFileTestCase1_audioI420CIF30fps.avi";
-
- printf("\tReading from an avi file and writing the information to another \n");
- printf("\tin the same format (I420 CIF 30fps) \n");
- printf("\t\t check file named %s\n", KFileName);
-
- assert(filePlayer.StartPlayingVideoFile(
- "../../../MediaFile/main/test/files/aviTestCase1_audioI420CIF30fps.avi",
- false, false) == 0);
-
- // init codecs
- webrtc::VideoCodec videoCodec;
- webrtc::VideoCodec recVideoCodec;
- webrtc::CodecInst audioCodec;
- assert(filePlayer.VideoCodec( videoCodec ) == 0);
- assert(filePlayer.AudioCodec( audioCodec) == 0);
-
- recVideoCodec = videoCodec;
-
- assert( fileRecorder.StartRecordingVideoFile(KFileName,
- audioCodec,
- recVideoCodec) == 0);
-
- assert(fileRecorder.IsRecording());
-
- WebRtc_UWord32 videoReadSize = static_cast<WebRtc_UWord32>( (videoCodec.width * videoCodec.height * 3.0) / 2.0);
-
- webrtc::VideoFrame videoFrame;
- videoFrame.VerifyAndAllocate(videoReadSize);
-
- int frameCount = 0;
- bool audioNotDone = true;
- bool videoNotDone = true;
- AudioFrame audioFrame;
-
- while( audioNotDone || videoNotDone)
- {
- if(filePlayer.TimeUntilNextVideoFrame() <= 0)
- {
- if(filePlayer.GetVideoFromFile( videoFrame) != 0)
- {
- // no more video frames
- break;
- }
- frameCount++;
- videoNotDone = ( videoFrame.Length() > 0);
- videoFrame.SetRenderTime(TickTime::MillisecondTimestamp());
- if( videoNotDone)
- {
- assert(fileRecorder.RecordVideoToFile(videoFrame) == 0);
- ::Sleep(10);
- }
- }
- WebRtc_UWord32 decodedDataLengthInSamples;
- if( 0 != filePlayer.Get10msAudioFromFile( audioFrame._payloadData, decodedDataLengthInSamples, audioCodec.plfreq))
- {
- audioNotDone = false;
- } else
- {
- audioFrame._frequencyInHz = filePlayer.Frequency();
- audioFrame._payloadDataLengthInSamples = (WebRtc_UWord16)decodedDataLengthInSamples;
- fileRecorder.RecordAudioToFile(audioFrame, &TickTime::Now());
- }
- }
- ::Sleep(100);
- assert(fileRecorder.StopRecording() == 0);
- assert( !fileRecorder.IsRecording());
- assert(frameCount == 135);
- printf("\tGenerated %s\n\n", KFileName);
- }
- ///////////////////////////////////////////////
- //
- // avi test case 2
- //
- ///////////////////////////////////////////////
- {
- FilePlayer& filePlayer(*FilePlayer::CreateFilePlayer(2, webrtc::kFileFormatAviFile));
- FileRecorder& fileRecorder(*FileRecorder::CreateFileRecorder(2, webrtc::kFileFormatAviFile));
-
- const char* KFileName = "./tmpAviFileTestCase2_audioI420CIF20fps.avi";
-
- printf("\tWriting information to a avi file and check the written file by \n");
- printf("\treopening it and control codec information.\n");
- printf("\t\t check file named %s all frames should be light green.\n", KFileName);
- // init codecs
- webrtc::VideoCodec videoCodec;
- webrtc::CodecInst audioCodec;
-
- memset(&videoCodec, 0, sizeof(videoCodec));
-
- const char* KVideoCodecName = "I420";
- strcpy(videoCodec.plName, KVideoCodecName);
- videoCodec.plType = 124;
- videoCodec.maxFramerate = 20;
- videoCodec.height = 288;
- videoCodec.width = 352;
-
- const char* KAudioCodecName = "PCMU";
- strcpy(audioCodec.plname, KAudioCodecName);
- audioCodec.pltype = 0;
- audioCodec.plfreq = 8000;
- audioCodec.pacsize = 80;
- audioCodec.channels = 1;
- audioCodec.rate = 64000;
-
- assert( fileRecorder.StartRecordingVideoFile(
- KFileName,
- audioCodec,
- videoCodec) == 0);
-
- assert(fileRecorder.IsRecording());
-
- const WebRtc_UWord32 KVideoWriteSize = static_cast< WebRtc_UWord32>( (videoCodec.width * videoCodec.height * 3) / 2);
- webrtc::VideoFrame videoFrame;
-
- // 10 ms
- AudioFrame audioFrame;
- audioFrame._payloadDataLengthInSamples = audioCodec.plfreq/100;
- memset(audioFrame._payloadData, 0, 2*audioFrame._payloadDataLengthInSamples);
- audioFrame._frequencyInHz = 8000;
-
- // prepare the video frame
- videoFrame.VerifyAndAllocate(KVideoWriteSize);
- memset(videoFrame.Buffer(), 127, videoCodec.width * videoCodec.height);
- memset(videoFrame.Buffer() +(videoCodec.width * videoCodec.height), 0, videoCodec.width * videoCodec.height/2);
- videoFrame.SetLength(KVideoWriteSize);
- videoFrame.SetHeight(videoCodec.height);
- videoFrame.SetWidth(videoCodec.width);
-
- // write avi file, with 20 video frames
- const int KWriteNumFrames = 20;
- int writeFrameCount = 0;
- while(writeFrameCount < KWriteNumFrames)
- {
- // add a video frame
- assert(fileRecorder.RecordVideoToFile(videoFrame) == 0);
-
- // add 50 ms of audio
- for(int i=0; i<5; i++)
- {
- assert( fileRecorder.RecordAudioToFile(audioFrame) == 0);
- }// for i
- writeFrameCount++;
- }
- ::Sleep(10); // enough tim eto write the queued data to the file
- assert(writeFrameCount == 20);
- assert(fileRecorder.StopRecording() == 0);
- assert( ! fileRecorder.IsRecording());
-
- assert(filePlayer.StartPlayingVideoFile(KFileName,false, false) == 0);
- assert(filePlayer.IsPlayingFile( ));
-
- // compare codecs read from file to the ones used when writing the file
- webrtc::VideoCodec readVideoCodec;
- assert(filePlayer.VideoCodec( readVideoCodec ) == 0);
- assert(strcmp(readVideoCodec.plName, videoCodec.plName) == 0);
- assert(readVideoCodec.width == videoCodec.width);
- assert(readVideoCodec.height == videoCodec.height);
- assert(readVideoCodec.maxFramerate == videoCodec.maxFramerate);
-
- webrtc::CodecInst readAudioCodec;
- assert(filePlayer.AudioCodec( readAudioCodec) == 0);
- assert(strcmp(readAudioCodec.plname, audioCodec.plname) == 0);
- assert(readAudioCodec.pltype == audioCodec.pltype);
- assert(readAudioCodec.plfreq == audioCodec.plfreq);
- assert(readAudioCodec.pacsize == audioCodec.pacsize);
- assert(readAudioCodec.channels == audioCodec.channels);
- assert(readAudioCodec.rate == audioCodec.rate);
-
- assert(filePlayer.StopPlayingFile() == 0);
- assert( ! filePlayer.IsPlayingFile());
- printf("\tGenerated %s\n\n", KFileName);
- }
- ///////////////////////////////////////////////
- //
- // avi test case 3
- //
- ///////////////////////////////////////////////
-
- {
- FilePlayer& filePlayer(*FilePlayer::CreateFilePlayer(2, webrtc::kFileFormatAviFile));
- FileRecorder& fileRecorder(*FileRecorder::CreateFileRecorder(3, webrtc::kFileFormatAviFile));
-
- printf("\tReading from an avi file and writing the information to another \n");
- printf("\tin a different format (H.263 CIF 30fps) \n");
- printf("\t\t check file named tmpAviFileTestCase1_audioH263CIF30fps.avi\n");
-
- assert(filePlayer.StartPlayingVideoFile(
- "../../../MediaFile/main/test/files/aviTestCase1_audioI420CIF30fps.avi",
- false,
- false) == 0);
-
- // init codecs
- webrtc::VideoCodec videoCodec;
- webrtc::VideoCodec recVideoCodec;
- webrtc::CodecInst audioCodec;
- assert(filePlayer.VideoCodec( videoCodec ) == 0);
- assert(filePlayer.AudioCodec( audioCodec) == 0);
- recVideoCodec = videoCodec;
-
- memcpy(recVideoCodec.plName, "H263",5);
- recVideoCodec.startBitrate = 1000;
- recVideoCodec.codecSpecific.H263.quality = 1;
- recVideoCodec.plType = 34;
- recVideoCodec.codecType = webrtc::kVideoCodecH263;
-
- assert( fileRecorder.StartRecordingVideoFile(
- "./tmpAviFileTestCase1_audioH263CIF30fps.avi",
- audioCodec,
- recVideoCodec) == 0);
-
- assert(fileRecorder.IsRecording());
-
- WebRtc_UWord32 videoReadSize = static_cast<WebRtc_UWord32>( (videoCodec.width * videoCodec.height * 3.0) / 2.0);
-
- webrtc::VideoFrame videoFrame;
- videoFrame.VerifyAndAllocate(videoReadSize);
-
- int videoFrameCount = 0;
- int audioFrameCount = 0;
- bool audioNotDone = true;
- bool videoNotDone = true;
- AudioFrame audioFrame;
-
- while( audioNotDone || videoNotDone)
- {
- if(filePlayer.TimeUntilNextVideoFrame() <= 0)
- {
- if(filePlayer.GetVideoFromFile( videoFrame) != 0)
- {
- break;
- }
- videoFrameCount++;
- videoNotDone = ( videoFrame.Length() > 0);
- if( videoNotDone)
- {
- assert(fileRecorder.RecordVideoToFile(videoFrame) == 0);
- }
- }
-
- WebRtc_UWord32 decodedDataLengthInSamples;
- if( 0 != filePlayer.Get10msAudioFromFile( audioFrame._payloadData, decodedDataLengthInSamples, audioCodec.plfreq))
- {
- audioNotDone = false;
-
- } else
- {
- ::Sleep(5);
- audioFrame._frequencyInHz = filePlayer.Frequency();
- audioFrame._payloadDataLengthInSamples = (WebRtc_UWord16)decodedDataLengthInSamples;
- assert(0 == fileRecorder.RecordAudioToFile(audioFrame));
-
- audioFrameCount++;
- }
- }
- assert(videoFrameCount == 135);
- assert(audioFrameCount == 446); // we will start & stop with a video frame
-
- assert(fileRecorder.StopRecording() == 0);
- assert( !fileRecorder.IsRecording());
- printf("\tGenerated ./tmpAviFileTestCase1_audioH263CIF30fps.avi\n\n");
- }
-
-
- printf("\nTEST completed.\n");
-
- Trace::ReturnTrace();
- return 0;
-}
diff --git a/trunk/src/modules/video_capture/OWNERS b/trunk/src/modules/video_capture/OWNERS
deleted file mode 100644
index 9034747..0000000
--- a/trunk/src/modules/video_capture/OWNERS
+++ /dev/null
@@ -1,4 +0,0 @@
-mallinath@webrtc.org
-mflodman@webrtc.org
-perkj@webrtc.org
-wu@webrtc.org
\ No newline at end of file
diff --git a/trunk/src/modules/video_capture/main/interface/video_capture.h b/trunk/src/modules/video_capture/main/interface/video_capture.h
deleted file mode 100644
index e2f2f23..0000000
--- a/trunk/src/modules/video_capture/main/interface/video_capture.h
+++ /dev/null
@@ -1,160 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_H_
-
-#include "module.h"
-#include "video_capture_defines.h"
-
-namespace webrtc {
-
-class VideoCaptureModule: public RefCountedModule {
- public:
- // Interface for receiving information about available camera devices.
- class DeviceInfo {
- public:
- virtual WebRtc_UWord32 NumberOfDevices() = 0;
-
- // Returns the available capture devices.
- // deviceNumber - Index of capture device.
- // deviceNameUTF8 - Friendly name of the capture device.
- // deviceUniqueIdUTF8 - Unique name of the capture device if it exist.
- // Otherwise same as deviceNameUTF8.
- // productUniqueIdUTF8 - Unique product id if it exist.
- // Null terminated otherwise.
- virtual WebRtc_Word32 GetDeviceName(
- WebRtc_UWord32 deviceNumber,
- WebRtc_UWord8* deviceNameUTF8,
- WebRtc_UWord32 deviceNameLength,
- WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord32 deviceUniqueIdUTF8Length,
- WebRtc_UWord8* productUniqueIdUTF8 = 0,
- WebRtc_UWord32 productUniqueIdUTF8Length = 0) = 0;
-
-
- // Returns the number of capabilities this device.
- virtual WebRtc_Word32 NumberOfCapabilities(
- const WebRtc_UWord8* deviceUniqueIdUTF8) = 0;
-
- // Gets the capabilities of the named device.
- virtual WebRtc_Word32 GetCapability(
- const WebRtc_UWord8* deviceUniqueIdUTF8,
- const WebRtc_UWord32 deviceCapabilityNumber,
- VideoCaptureCapability& capability) = 0;
-
- // Gets clockwise angle the captured frames should be rotated in order
- // to be displayed correctly on a normally rotated display.
- virtual WebRtc_Word32 GetOrientation(
- const WebRtc_UWord8* deviceUniqueIdUTF8,
- VideoCaptureRotation& orientation) = 0;
-
- // Gets the capability that best matches the requested width, height and
- // frame rate.
- // Returns the deviceCapabilityNumber on success.
- virtual WebRtc_Word32 GetBestMatchedCapability(
- const WebRtc_UWord8*deviceUniqueIdUTF8,
- const VideoCaptureCapability& requested,
- VideoCaptureCapability& resulting) = 0;
-
- // Display OS /capture device specific settings dialog
- virtual WebRtc_Word32 DisplayCaptureSettingsDialogBox(
- const WebRtc_UWord8* deviceUniqueIdUTF8,
- const WebRtc_UWord8* dialogTitleUTF8,
- void* parentWindow,
- WebRtc_UWord32 positionX,
- WebRtc_UWord32 positionY) = 0;
-
- virtual ~DeviceInfo() {}
- };
-
- class VideoCaptureEncodeInterface {
- public:
- virtual WebRtc_Word32 ConfigureEncoder(const VideoCodec& codec,
- WebRtc_UWord32 maxPayloadSize) = 0;
- // Inform the encoder about the new target bit rate.
- // - newBitRate : New target bit rate in Kbit/s.
- // - frameRate : The target frame rate.
- virtual WebRtc_Word32 SetRates(WebRtc_Word32 newBitRate,
- WebRtc_Word32 frameRate) = 0;
- // Inform the encoder about the packet loss and the round-trip time.
- // - packetLoss : Fraction lost
- // (loss rate in percent = 100 * packetLoss / 255).
- // - rtt : Round-trip time in milliseconds.
- virtual WebRtc_Word32 SetChannelParameters(WebRtc_UWord32 packetLoss,
- int rtt) = 0;
-
- // Encode the next frame as key frame.
- virtual WebRtc_Word32 EncodeFrameType(const FrameType type) = 0;
- protected:
- virtual ~VideoCaptureEncodeInterface() {
- }
- };
-
- // Register capture data callback
- virtual WebRtc_Word32 RegisterCaptureDataCallback(
- VideoCaptureDataCallback& dataCallback) = 0;
-
- // Remove capture data callback
- virtual WebRtc_Word32 DeRegisterCaptureDataCallback() = 0;
-
- // Register capture callback.
- virtual WebRtc_Word32 RegisterCaptureCallback(
- VideoCaptureFeedBack& callBack) = 0;
-
- // Remove capture callback.
- virtual WebRtc_Word32 DeRegisterCaptureCallback() = 0;
-
- // Start capture device
- virtual WebRtc_Word32 StartCapture(
- const VideoCaptureCapability& capability) = 0;
-
- virtual WebRtc_Word32 StopCapture() = 0;
-
- // Send an image when the capture device is not running.
- virtual WebRtc_Word32 StartSendImage(const VideoFrame& videoFrame,
- WebRtc_Word32 frameRate = 1) = 0;
-
- virtual WebRtc_Word32 StopSendImage() = 0;
-
- // Returns the name of the device used by this module.
- virtual const WebRtc_UWord8* CurrentDeviceName() const = 0;
-
- // Returns true if the capture device is running
- virtual bool CaptureStarted() = 0;
-
- // Gets the current configuration.
- virtual WebRtc_Word32 CaptureSettings(VideoCaptureCapability& settings) = 0;
-
- virtual WebRtc_Word32 SetCaptureDelay(WebRtc_Word32 delayMS) = 0;
-
- // Returns the current CaptureDelay. Only valid when the camera is running.
- virtual WebRtc_Word32 CaptureDelay() = 0;
-
- // Set the rotation of the captured frames.
- // If the rotation is set to the same as returned by
- // DeviceInfo::GetOrientation the captured frames are
- // displayed correctly if rendered.
- virtual WebRtc_Word32 SetCaptureRotation(VideoCaptureRotation rotation) = 0;
-
- // Gets a pointer to an encode interface if the capture device supports the
- // requested type and size. NULL otherwise.
- virtual VideoCaptureEncodeInterface* GetEncodeInterface(
- const VideoCodec& codec) = 0;
-
- virtual WebRtc_Word32 EnableFrameRateCallback(const bool enable) = 0;
- virtual WebRtc_Word32 EnableNoPictureAlarm(const bool enable) = 0;
-
-protected:
- virtual ~VideoCaptureModule() {};
-};
-
-} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_H_
diff --git a/trunk/src/modules/video_capture/main/interface/video_capture_defines.h b/trunk/src/modules/video_capture/main/interface/video_capture_defines.h
deleted file mode 100644
index 2a3408b..0000000
--- a/trunk/src/modules/video_capture/main/interface/video_capture_defines.h
+++ /dev/null
@@ -1,155 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_DEFINES_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_DEFINES_H_
-
-// Includes
-#include "typedefs.h"
-#include "module_common_types.h"
-
-namespace webrtc
-{
-// Defines
-#ifndef NULL
- #define NULL 0
-#endif
-
-enum {kVideoCaptureUniqueNameLength =1024}; //Max unique capture device name lenght
-enum {kVideoCaptureDeviceNameLength =256}; //Max capture device name lenght
-enum {kVideoCaptureProductIdLength =128}; //Max product id length
-
-// Enums
-enum VideoCaptureRotation
-{
- kCameraRotate0 = 0,
- kCameraRotate90 = 5,
- kCameraRotate180 = 10,
- kCameraRotate270 = 15
-};
-
-struct VideoCaptureCapability
-{
- WebRtc_Word32 width;
- WebRtc_Word32 height;
- WebRtc_Word32 maxFPS;
- WebRtc_Word32 expectedCaptureDelay;
- RawVideoType rawType;
- VideoCodecType codecType;
- bool interlaced;
-
- VideoCaptureCapability()
- {
- width = 0;
- height = 0;
- maxFPS = 0;
- expectedCaptureDelay = 0;
- rawType = kVideoUnknown;
- codecType = kVideoCodecUnknown;
- interlaced = false;
- }
- ;
- bool operator!=(const VideoCaptureCapability &other) const
- {
- if (width != other.width)
- return true;
- if (height != other.height)
- return true;
- if (maxFPS != other.maxFPS)
- return true;
- if (rawType != other.rawType)
- return true;
- if (codecType != other.codecType)
- return true;
- if (interlaced != other.interlaced)
- return true;
- return false;
- }
- bool operator==(const VideoCaptureCapability &other) const
- {
- return !operator!=(other);
- }
-};
-
-enum VideoCaptureAlarm
-{
- Raised = 0,
- Cleared = 1
-};
-
-// VideoFrameI420 doesn't take the ownership of the buffer.
-// It's mostly used to group the parameters for external capture.
-struct VideoFrameI420
-{
- VideoFrameI420() {
- y_plane = NULL;
- u_plane = NULL;
- v_plane = NULL;
- y_pitch = 0;
- u_pitch = 0;
- v_pitch = 0;
- width = 0;
- height = 0;
- }
-
- unsigned char* y_plane;
- unsigned char* u_plane;
- unsigned char* v_plane;
-
- int y_pitch;
- int u_pitch;
- int v_pitch;
-
- unsigned short width;
- unsigned short height;
-};
-
-/* External Capture interface. Returned by Create
- and implemented by the capture module.
- */
-class VideoCaptureExternal
-{
-public:
- virtual WebRtc_Word32 IncomingFrame(WebRtc_UWord8* videoFrame,
- WebRtc_Word32 videoFrameLength,
- const VideoCaptureCapability& frameInfo,
- WebRtc_Word64 captureTime = 0) = 0;
- virtual WebRtc_Word32 IncomingFrameI420(const VideoFrameI420& video_frame,
- WebRtc_Word64 captureTime = 0) = 0;
-protected:
- ~VideoCaptureExternal() {}
-};
-
-// Callback class to be implemented by module user
-class VideoCaptureDataCallback
-{
-public:
- virtual void OnIncomingCapturedFrame(const WebRtc_Word32 id,
- VideoFrame& videoFrame,
- VideoCodecType codecType) = 0;
- virtual void OnCaptureDelayChanged(const WebRtc_Word32 id,
- const WebRtc_Word32 delay) = 0;
-protected:
- virtual ~VideoCaptureDataCallback(){}
-};
-
-class VideoCaptureFeedBack
-{
-public:
- virtual void OnCaptureFrameRate(const WebRtc_Word32 id,
- const WebRtc_UWord32 frameRate) = 0;
- virtual void OnNoPictureAlarm(const WebRtc_Word32 id,
- const VideoCaptureAlarm alarm) = 0;
-protected:
- virtual ~VideoCaptureFeedBack(){}
-};
-
-} //namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_DEFINES_H_
diff --git a/trunk/src/modules/video_capture/main/interface/video_capture_factory.h b/trunk/src/modules/video_capture/main/interface/video_capture_factory.h
deleted file mode 100644
index 6cb7a77..0000000
--- a/trunk/src/modules/video_capture/main/interface/video_capture_factory.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file contains interfaces used for creating the VideoCaptureModule
-// and DeviceInfo.
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_FACTORY_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_FACTORY_H_
-
-#include "video_capture.h"
-
-namespace webrtc {
-
-class VideoCaptureFactory {
- public:
- // Create a video capture module object
- // id - unique identifier of this video capture module object.
- // deviceUniqueIdUTF8 - name of the device.
- // Available names can be found by using GetDeviceName
- static VideoCaptureModule* Create(const WebRtc_Word32 id,
- const WebRtc_UWord8* deviceUniqueIdUTF8);
-
- // Create a video capture module object used for external capture.
- // id - unique identifier of this video capture module object
- // externalCapture - [out] interface to call when a new frame is captured.
- static VideoCaptureModule* Create(const WebRtc_Word32 id,
- VideoCaptureExternal*& externalCapture);
-
- static VideoCaptureModule::DeviceInfo* CreateDeviceInfo(
- const WebRtc_Word32 id);
-
-#ifdef WEBRTC_ANDROID
- static WebRtc_Word32 SetAndroidObjects(void* javaVM, void* javaContext);
-#endif
-
- private:
- ~VideoCaptureFactory();
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_INTERFACE_VIDEO_CAPTURE_FACTORY_H_
diff --git a/trunk/src/modules/video_capture/main/source/Android.mk b/trunk/src/modules/video_capture/main/source/Android.mk
deleted file mode 100644
index 8976f19..0000000
--- a/trunk/src/modules/video_capture/main/source/Android.mk
+++ /dev/null
@@ -1,50 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_video_capture
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := \
- video_capture_impl.cc \
- device_info_impl.cc \
- video_capture_factory.cc \
- Android/video_capture_android.cc \
- Android/device_info_android.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/Android \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../source \
- $(LOCAL_PATH)/../../../interface \
- $(LOCAL_PATH)/../../../utility/interface \
- $(LOCAL_PATH)/../../../audio_coding/main/interface \
- $(LOCAL_PATH)/../../../.. \
- $(LOCAL_PATH)/../../../../common_video/vplib/main/interface \
- $(LOCAL_PATH)/../../../../system_wrappers/interface
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/video_capture/main/source/Android/device_info_android.cc b/trunk/src/modules/video_capture/main/source/Android/device_info_android.cc
deleted file mode 100644
index 3c5e052..0000000
--- a/trunk/src/modules/video_capture/main/source/Android/device_info_android.cc
+++ /dev/null
@@ -1,363 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "device_info_android.h"
-
-#include <stdio.h>
-
-#include "ref_count.h"
-#include "trace.h"
-#include "video_capture_android.h"
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-VideoCaptureModule::DeviceInfo* VideoCaptureImpl::CreateDeviceInfo (
- const WebRtc_Word32 id)
-{
- videocapturemodule::DeviceInfoAndroid *deviceInfo =
- new videocapturemodule::DeviceInfoAndroid(id);
- if (deviceInfo && deviceInfo->Init() != 0) // Failed to init
- {
- delete deviceInfo;
- deviceInfo = NULL;
- }
- return deviceInfo;
-}
-
-DeviceInfoAndroid::DeviceInfoAndroid(const WebRtc_Word32 id) :
- DeviceInfoImpl(id)
-{
-}
-
-WebRtc_Word32 DeviceInfoAndroid::Init()
-{
- return 0;
-}
-
-DeviceInfoAndroid::~DeviceInfoAndroid()
-{
-}
-
-WebRtc_UWord32 DeviceInfoAndroid::NumberOfDevices()
-{
-
- JNIEnv *env;
- jclass javaCmDevInfoClass;
- jobject javaCmDevInfoObject;
- bool attached = false;
- if (VideoCaptureAndroid::AttachAndUseAndroidDeviceInfoObjects(
- env,
- javaCmDevInfoClass,
- javaCmDevInfoObject,
- attached) != 0)
- {
- return 0;
- }
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- "%s GetMethodId", __FUNCTION__);
- // get the method ID for the Android Java GetDeviceUniqueName name.
- jmethodID cid = env->GetMethodID(javaCmDevInfoClass,
- "NumberOfDevices",
- "()I");
-
- jint numberOfDevices = 0;
- if (cid != NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- "%s Calling Number of devices", __FUNCTION__);
- numberOfDevices = env->CallIntMethod(javaCmDevInfoObject, cid);
- }
- VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
-
- if (numberOfDevices > 0)
- return numberOfDevices;
- return 0;
-}
-
-WebRtc_Word32 DeviceInfoAndroid::GetDeviceName(
- WebRtc_UWord32 deviceNumber,
- WebRtc_UWord8* deviceNameUTF8,
- WebRtc_UWord32 deviceNameLength,
- WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord32 deviceUniqueIdUTF8Length,
- WebRtc_UWord8* /*productUniqueIdUTF8*/,
- WebRtc_UWord32 /*productUniqueIdUTF8Length*/)
-{
-
- JNIEnv *env;
- jclass javaCmDevInfoClass;
- jobject javaCmDevInfoObject;
- WebRtc_Word32 result = 0;
- bool attached = false;
- if (VideoCaptureAndroid::AttachAndUseAndroidDeviceInfoObjects(
- env,
- javaCmDevInfoClass,
- javaCmDevInfoObject,
- attached)!= 0)
- {
- return -1;
- }
-
- // get the method ID for the Android Java GetDeviceUniqueName name.
- jmethodID cid = env->GetMethodID(javaCmDevInfoClass, "GetDeviceUniqueName",
- "(I)Ljava/lang/String;");
- if (cid != NULL)
- {
-
- jobject javaDeviceNameObj = env->CallObjectMethod(javaCmDevInfoObject,
- cid, deviceNumber);
- if (javaDeviceNameObj == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s: Failed to get device name for device %d.",
- __FUNCTION__, (int) deviceNumber);
- result = -1;
- }
- else
- {
- jboolean isCopy;
- const char* javaDeviceNameChar = env->GetStringUTFChars(
- (jstring) javaDeviceNameObj
- ,&isCopy);
- const jsize javaDeviceNameCharLength = env->GetStringUTFLength(
- (jstring) javaDeviceNameObj);
- if ((WebRtc_UWord32) javaDeviceNameCharLength < deviceUniqueIdUTF8Length)
- {
- memcpy(deviceUniqueIdUTF8,
- javaDeviceNameChar,
- javaDeviceNameCharLength + 1);
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
- _id, "%s: deviceUniqueIdUTF8 to short.",
- __FUNCTION__);
- result = -1;
- }
- if ((WebRtc_UWord32) javaDeviceNameCharLength < deviceNameLength)
- {
- memcpy(deviceNameUTF8,
- javaDeviceNameChar,
- javaDeviceNameCharLength + 1);
- }
- env->ReleaseStringUTFChars((jstring) javaDeviceNameObj,
- javaDeviceNameChar);
- }//javaDeviceNameObj==NULL
-
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "%s: Failed to find GetDeviceUniqueName function id",
- __FUNCTION__);
- result = -1;
- }
-
- VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
-
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
- "%s: result %d", __FUNCTION__, (int) result);
- return result;
-
-}
-
-WebRtc_Word32 DeviceInfoAndroid::CreateCapabilityMap(
- const WebRtc_UWord8* deviceUniqueIdUTF8)
-{
-
- MapItem* item = NULL;
- while ((item = _captureCapabilities.Last()))
- {
- delete (VideoCaptureCapability*) item->GetItem();
- _captureCapabilities.Erase(item);
- }
-
- JNIEnv *env;
- jclass javaCmDevInfoClass;
- jobject javaCmDevInfoObject;
- bool attached = false;
- if (VideoCaptureAndroid::AttachAndUseAndroidDeviceInfoObjects(
- env,
- javaCmDevInfoClass,
- javaCmDevInfoObject,
- attached) != 0)
- {
- return -1;
- }
-
- // Find the capability class
- jclass javaCapClassLocal = env->FindClass(AndroidJavaCaptureCapabilityClass);
- if (javaCapClassLocal == NULL)
- {
- VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s: Can't find java class VideoCaptureCapabilityAndroid.",
- __FUNCTION__);
- return -1;
- }
-
- // get the method ID for the Android Java GetCapabilityArray .
- char signature[256];
- sprintf(signature,
- "(Ljava/lang/String;)[L%s;",
- AndroidJavaCaptureCapabilityClass);
- jmethodID cid = env->GetMethodID(javaCmDevInfoClass,
- "GetCapabilityArray",
- signature);
- if (cid == NULL)
- {
- VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s: Can't find method GetCapabilityArray.", __FUNCTION__);
- return -1;
- }
- // Create a jstring so we can pass the deviceUniquName to the java method.
- jstring capureIdString = env->NewStringUTF((char*) deviceUniqueIdUTF8);
-
- if (capureIdString == NULL)
- {
- VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s: Can't create string for method GetCapabilityArray.",
- __FUNCTION__);
- return -1;
- }
- // Call the java class and get an array with capabilities back.
- jobject javaCapabilitiesObj = env->CallObjectMethod(javaCmDevInfoObject,
- cid, capureIdString);
- if (!javaCapabilitiesObj)
- {
- VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s: Failed to call java GetCapabilityArray.",
- __FUNCTION__);
- return -1;
- }
-
- jfieldID widthField = env->GetFieldID(javaCapClassLocal, "width", "I");
- jfieldID heigtField = env->GetFieldID(javaCapClassLocal, "height", "I");
- jfieldID maxFpsField = env->GetFieldID(javaCapClassLocal, "maxFPS", "I");
- if (widthField == NULL || heigtField == NULL || maxFpsField == NULL)
- {
- VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s: Failed to get field Id.", __FUNCTION__);
- return -1;
- }
-
- const jsize numberOfCapabilities =
- env->GetArrayLength((jarray) javaCapabilitiesObj);
-
- for (jsize i = 0; i < numberOfCapabilities; ++i)
- {
- VideoCaptureCapability *cap = new VideoCaptureCapability();
- jobject capabilityElement = env->GetObjectArrayElement(
- (jobjectArray) javaCapabilitiesObj,
- i);
-
- cap->width = env->GetIntField(capabilityElement, widthField);
- cap->height = env->GetIntField(capabilityElement, heigtField);
- cap->expectedCaptureDelay = _expectedCaptureDelay;
- cap->rawType = kVideoNV21;
- cap->maxFPS = env->GetIntField(capabilityElement, maxFpsField);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s: Cap width %d, height %d, fps %d", __FUNCTION__,
- cap->width, cap->height, cap->maxFPS);
- _captureCapabilities.Insert(i, cap);
- }
-
- _lastUsedDeviceNameLength = strlen((char*) deviceUniqueIdUTF8);
- _lastUsedDeviceName = (WebRtc_UWord8*) realloc(_lastUsedDeviceName,
- _lastUsedDeviceNameLength + 1);
- memcpy(_lastUsedDeviceName,
- deviceUniqueIdUTF8,
- _lastUsedDeviceNameLength + 1);
-
- VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "CreateCapabilityMap %d", _captureCapabilities.Size());
-
- return _captureCapabilities.Size();
-}
-
-WebRtc_Word32 DeviceInfoAndroid::GetOrientation(
- const WebRtc_UWord8* deviceUniqueIdUTF8,
- VideoCaptureRotation& orientation)
-{
-
- JNIEnv *env;
- jclass javaCmDevInfoClass;
- jobject javaCmDevInfoObject;
- WebRtc_Word32 result = 0;
- bool attached = false;
- if (VideoCaptureAndroid::AttachAndUseAndroidDeviceInfoObjects(
- env,
- javaCmDevInfoClass,
- javaCmDevInfoObject,
- attached) != 0)
- {
- return -1;
- }
-
- // get the method ID for the Android Java GetOrientation .
- jmethodID cid = env->GetMethodID(javaCmDevInfoClass, "GetOrientation",
- "(Ljava/lang/String;)I");
- if (cid == NULL)
- {
- VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s: Can't find method GetOrientation.", __FUNCTION__);
- return -1;
- }
- // Create a jstring so we can pass the deviceUniquName to the java method.
- jstring capureIdString = env->NewStringUTF((char*) deviceUniqueIdUTF8);
- if (capureIdString == NULL)
- {
- VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s: Can't create string for method GetCapabilityArray.",
- __FUNCTION__);
- return -1;
- }
- // Call the java class and get the orientation.
- jint jorientation = env->CallIntMethod(javaCmDevInfoObject, cid,
- capureIdString);
- VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(attached);
-
- WebRtc_Word32 retValue = 0;
- switch (jorientation)
- {
- case -1: //Error
- orientation = kCameraRotate0;
- retValue = -1;
- break;
- case 0:
- orientation = kCameraRotate0;
- break;
- case 90:
- orientation = kCameraRotate90;
- break;
- case 180:
- orientation = kCameraRotate180;
- break;
- case 270:
- orientation = kCameraRotate270;
- break;
- case 360:
- orientation = kCameraRotate0;
- break;
- }
- return retValue;
-}
-} // namespace videocapturemodule
-} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Android/device_info_android.h b/trunk/src/modules/video_capture/main/source/Android/device_info_android.h
deleted file mode 100644
index 33b22c3..0000000
--- a/trunk/src/modules/video_capture/main/source/Android/device_info_android.h
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_ANDROID_DEVICE_INFO_ANDROID_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_ANDROID_DEVICE_INFO_ANDROID_H_
-
-#include <jni.h>
-#include "video_capture_impl.h"
-#include "device_info_impl.h"
-
-#define AndroidJavaCaptureDeviceInfoClass "org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid"
-#define AndroidJavaCaptureCapabilityClass "org/webrtc/videoengine/CaptureCapabilityAndroid"
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-
-// Android logging, uncomment to print trace to logcat instead of trace file/callback
-//#include <android/log.h>
-//#define WEBRTC_TRACE(a,b,c,...) __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTCN*", __VA_ARGS__)
-
-class DeviceInfoAndroid: public DeviceInfoImpl
-{
-public:
-
- DeviceInfoAndroid(const WebRtc_Word32 id);
- WebRtc_Word32 Init();
- virtual ~DeviceInfoAndroid();
- virtual WebRtc_UWord32 NumberOfDevices();
- virtual WebRtc_Word32 GetDeviceName(WebRtc_UWord32 deviceNumber,
- WebRtc_UWord8* deviceNameUTF8,
- WebRtc_UWord32 deviceNameLength,
- WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord32 deviceUniqueIdUTF8Length,
- WebRtc_UWord8* productUniqueIdUTF8 = 0,
- WebRtc_UWord32 productUniqueIdUTF8Length = 0);
- virtual WebRtc_Word32 CreateCapabilityMap(const WebRtc_UWord8* deviceUniqueIdUTF8);
-
- virtual WebRtc_Word32 DisplayCaptureSettingsDialogBox(
- const WebRtc_UWord8* /*deviceUniqueIdUTF8*/,
- const WebRtc_UWord8* /*dialogTitleUTF8*/,
- void* /*parentWindow*/,
- WebRtc_UWord32 /*positionX*/,
- WebRtc_UWord32 /*positionY*/){return -1;}
- virtual WebRtc_Word32 GetOrientation(const WebRtc_UWord8* deviceUniqueIdUTF8,
- VideoCaptureRotation& orientation);
-private:
- bool IsDeviceNameMatches(const char* name, const char* deviceUniqueIdUTF8);
- enum {_expectedCaptureDelay = 190};
-};
-} // namespace videocapturemodule
-} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_ANDROID_DEVICE_INFO_ANDROID_H_
diff --git a/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/CaptureCapabilityAndroid.java b/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/CaptureCapabilityAndroid.java
deleted file mode 100644
index 0443953..0000000
--- a/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/CaptureCapabilityAndroid.java
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-package org.webrtc.videoengine;
-
-public class CaptureCapabilityAndroid {
- public int width = 0;
- public int height = 0;
- public int maxFPS = 0;
-}
diff --git a/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/VideoCaptureAndroid.java b/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/VideoCaptureAndroid.java
deleted file mode 100644
index a4c39a8..0000000
--- a/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/VideoCaptureAndroid.java
+++ /dev/null
@@ -1,262 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-package org.webrtc.videoengine;
-
-import java.io.IOException;
-import java.util.Locale;
-import java.util.concurrent.locks.ReentrantLock;
-
-import org.webrtc.videoengine.CaptureCapabilityAndroid;
-import org.webrtc.videoengine.VideoCaptureDeviceInfoAndroid.AndroidVideoCaptureDevice;
-
-import android.graphics.ImageFormat;
-import android.graphics.PixelFormat;
-import android.hardware.Camera;
-import android.hardware.Camera.PreviewCallback;
-import android.util.Log;
-import android.view.SurfaceHolder;
-import android.view.SurfaceHolder.Callback;
-
-public class VideoCaptureAndroid implements PreviewCallback, Callback {
-
- private Camera camera;
- private AndroidVideoCaptureDevice currentDevice = null;
- public ReentrantLock previewBufferLock = new ReentrantLock();
- private int PIXEL_FORMAT = ImageFormat.NV21;
- PixelFormat pixelFormat = new PixelFormat();
- // True when the C++ layer has ordered the camera to be started.
- private boolean isRunning=false;
-
- private final int numCaptureBuffers = 3;
- private int expectedFrameSize = 0;
- private int orientation = 0;
- private int id = 0;
- // C++ callback context variable.
- private long context = 0;
- private SurfaceHolder localPreview = null;
- // True if this class owns the preview video buffers.
- private boolean ownsBuffers = false;
-
- // Set this to 2 for VERBOSE logging. 1 for DEBUG
- private static int LOGLEVEL = 0;
- private static boolean VERBOSE = LOGLEVEL > 2;
- private static boolean DEBUG = LOGLEVEL > 1;
-
- CaptureCapabilityAndroid currentCapability = null;
-
- public static
- void DeleteVideoCaptureAndroid(VideoCaptureAndroid captureAndroid) {
- if(DEBUG) Log.d("*WEBRTC*", "DeleteVideoCaptureAndroid");
-
- captureAndroid.StopCapture();
- captureAndroid.camera.release();
- captureAndroid.camera = null;
- captureAndroid.context = 0;
-
- if(DEBUG) Log.v("*WEBRTC*", "DeleteVideoCaptureAndroid ended");
-
- }
-
- public VideoCaptureAndroid(int in_id,
- long in_context,
- Camera in_camera,
- AndroidVideoCaptureDevice in_device) {
- id = in_id;
- context = in_context;
- camera = in_camera;
- currentDevice = in_device;
- }
-
- public int StartCapture(int width, int height, int frameRate) {
- if(DEBUG) Log.d("*WEBRTC*", "StartCapture width" + width +
- " height " + height +" frame rate " + frameRate);
- try {
- if (camera == null) {
- Log.e("*WEBRTC*",
- String.format(Locale.US,"Camera not initialized %d",id));
- return -1;
- }
- currentCapability = new CaptureCapabilityAndroid();
- currentCapability.width = width;
- currentCapability.height = height;
- currentCapability.maxFPS = frameRate;
- PixelFormat.getPixelFormatInfo(PIXEL_FORMAT, pixelFormat);
-
- Camera.Parameters parameters = camera.getParameters();
- parameters.setPreviewSize(currentCapability.width,
- currentCapability.height);
- parameters.setPreviewFormat(PIXEL_FORMAT );
- parameters.setPreviewFrameRate(currentCapability.maxFPS);
- camera.setParameters(parameters);
-
- // Get the local preview SurfaceHolder from the static render class
- localPreview = ViERenderer.GetLocalRenderer();
- if(localPreview != null) {
- localPreview.addCallback(this);
- }
-
- int bufSize = width * height * pixelFormat.bitsPerPixel / 8;
- if(android.os.Build.VERSION.SDK_INT >= 7) {
- // According to Doc addCallbackBuffer belongs to API level 8.
- // But it seems like it works on Android 2.1 as well.
- // At least SE X10 and Milestone
- byte[] buffer = null;
- for (int i = 0; i < numCaptureBuffers; i++) {
- buffer = new byte[bufSize];
- camera.addCallbackBuffer(buffer);
- }
-
- camera.setPreviewCallbackWithBuffer(this);
- ownsBuffers = true;
- }
- else {
- camera.setPreviewCallback(this);
- }
-
- camera.startPreview();
- previewBufferLock.lock();
- expectedFrameSize = bufSize;
- isRunning = true;
- previewBufferLock.unlock();
- }
- catch (Exception ex) {
- Log.e("*WEBRTC*", "Failed to start camera");
- return -1;
- }
- return 0;
- }
-
- public int StopCapture() {
- if(DEBUG) Log.d("*WEBRTC*", "StopCapture");
- try {
- previewBufferLock.lock();
- isRunning = false;
- previewBufferLock.unlock();
-
- camera.stopPreview();
-
- if(android.os.Build.VERSION.SDK_INT > 7) {
- camera.setPreviewCallbackWithBuffer(null);
- }
- else {
- camera.setPreviewCallback(null);
- }
- }
- catch (Exception ex) {
- Log.e("*WEBRTC*", "Failed to stop camera");
- return -1;
- }
-
- if(DEBUG) {
- Log.d("*WEBRTC*", "StopCapture ended");
- }
- return 0;
- }
-
- native void ProvideCameraFrame(byte[] data,int length, long captureObject);
-
- public void onPreviewFrame(byte[] data, Camera camera) {
- previewBufferLock.lock();
-
- if(VERBOSE) {
- Log.v("*WEBRTC*",
- String.format(Locale.US, "preview frame length %d context %x",
- data.length, context));
- }
- if(isRunning) {
- // If StartCapture has been called but not StopCapture
- // Call the C++ layer with the captured frame
- if (data.length == expectedFrameSize) {
- ProvideCameraFrame(data, expectedFrameSize, context);
- if (VERBOSE) {
- Log.v("*WEBRTC*", String.format(Locale.US, "frame delivered"));
- }
- if(ownsBuffers) {
- // Give the video buffer to the camera service again.
- camera.addCallbackBuffer(data);
- }
- }
- }
- previewBufferLock.unlock();
- }
-
-
- public void surfaceChanged(SurfaceHolder holder,
- int format, int width, int height) {
-
- try {
- if(camera != null) {
- camera.setPreviewDisplay(localPreview);
- }
- } catch (IOException e) {
- Log.e("*WEBRTC*",
- String.format(Locale.US,
- "Failed to set Local preview. " + e.getMessage()));
- }
- }
-
- // Sets the rotation of the preview render window.
- // Does not affect the captured video image.
- public void SetPreviewRotation(int rotation) {
- if(camera != null) {
- previewBufferLock.lock();
- final boolean running = isRunning;
- int width = 0;
- int height = 0;
- int framerate = 0;
-
- if(running) {
- width = currentCapability.width;
- height = currentCapability.height;
- framerate = currentCapability.maxFPS;
-
- StopCapture();
- }
-
- int resultRotation = 0;
- if(currentDevice.frontCameraType ==
- VideoCaptureDeviceInfoAndroid.FrontFacingCameraType.Android23) {
- // this is a 2.3 or later front facing camera.
- // SetDisplayOrientation will flip the image horizontally
- // before doing the rotation.
- resultRotation=(360-rotation) % 360; // compensate the mirror
- }
- else {
- // Back facing or 2.2 or previous front camera
- resultRotation=rotation;
- }
- if(android.os.Build.VERSION.SDK_INT>7) {
- camera.setDisplayOrientation(resultRotation);
- }
- else {
- // Android 2.1 and previous
- // This rotation unfortunately does not seems to work.
- // http://code.google.com/p/android/issues/detail?id=1193
- Camera.Parameters parameters = camera.getParameters();
- parameters.setRotation(resultRotation);
- camera.setParameters(parameters);
- }
-
- if(running) {
- StartCapture(width, height, framerate);
- }
- previewBufferLock.unlock();
- }
- }
-
- public void surfaceCreated(SurfaceHolder holder) {
- }
-
-
- public void surfaceDestroyed(SurfaceHolder holder) {
- }
-
-}
diff --git a/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java b/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java
deleted file mode 100644
index 4ccf060..0000000
--- a/trunk/src/modules/video_capture/main/source/Android/java/org/webrtc/videoengine/VideoCaptureDeviceInfoAndroid.java
+++ /dev/null
@@ -1,432 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-package org.webrtc.videoengine;
-
-import java.io.File;
-import java.lang.reflect.InvocationTargetException;
-import java.lang.reflect.Method;
-import java.util.ArrayList;
-import java.util.List;
-import java.util.Locale;
-
-import dalvik.system.DexClassLoader;
-
-import android.content.Context;
-import android.hardware.Camera;
-import android.hardware.Camera.Size;
-import android.util.Log;
-
-public class VideoCaptureDeviceInfoAndroid {
-
- //Context
- Context context;
-
- // Set this to 2 for VERBOSE logging. 1 for DEBUG
- private static int LOGLEVEL = 0;
- private static boolean VERBOSE = LOGLEVEL > 2;
- private static boolean DEBUG = LOGLEVEL > 1;
-
- // Private class with info about all available cameras and the capabilities
- public class AndroidVideoCaptureDevice {
- AndroidVideoCaptureDevice() {
- frontCameraType = FrontFacingCameraType.None;
- index = 0;
- }
-
- public String deviceUniqueName;
- public CaptureCapabilityAndroid captureCapabilies[];
- public FrontFacingCameraType frontCameraType;
-
- // Orientation of camera as described in
- // android.hardware.Camera.CameraInfo.Orientation
- public int orientation;
- // Camera index used in Camera.Open on Android 2.3 and onwards
- public int index;
- }
-
- public enum FrontFacingCameraType {
- None, // This is not a front facing camera
- GalaxyS, // Galaxy S front facing camera.
- HTCEvo, // HTC Evo front facing camera
- Android23, // Android 2.3 front facing camera.
- }
-
- String currentDeviceUniqueId;
- int id;
- List<AndroidVideoCaptureDevice> deviceList;
-
- public static VideoCaptureDeviceInfoAndroid
- CreateVideoCaptureDeviceInfoAndroid(int in_id, Context in_context) {
- if(DEBUG) {
- Log.d("*WEBRTC*",
- String.format(Locale.US, "VideoCaptureDeviceInfoAndroid"));
- }
-
- VideoCaptureDeviceInfoAndroid self =
- new VideoCaptureDeviceInfoAndroid(in_id, in_context);
- if(self != null && self.Init() == 0) {
- return self;
- }
- else {
- if(DEBUG) {
- Log.d("*WEBRTC*", "Failed to create VideoCaptureDeviceInfoAndroid.");
- }
- }
- return null;
- }
-
- private VideoCaptureDeviceInfoAndroid(int in_id,
- Context in_context) {
- id = in_id;
- context = in_context;
- deviceList = new ArrayList<AndroidVideoCaptureDevice>();
- }
-
- private int Init() {
- // Populate the deviceList with available cameras and their capabilities.
- Camera camera = null;
- try{
- if(android.os.Build.VERSION.SDK_INT > 8) {
- // From Android 2.3 and onwards
- for(int i = 0; i < Camera.getNumberOfCameras(); ++i) {
- AndroidVideoCaptureDevice newDevice = new AndroidVideoCaptureDevice();
-
- Camera.CameraInfo info = new Camera.CameraInfo();
- Camera.getCameraInfo(i, info);
- newDevice.index = i;
- newDevice.orientation=info.orientation;
- if(info.facing == Camera.CameraInfo.CAMERA_FACING_BACK) {
- newDevice.deviceUniqueName =
- "Camera " + i +", Facing back, Orientation "+ info.orientation;
- }
- else {
- newDevice.deviceUniqueName =
- "Camera " + i +", Facing front, Orientation "+ info.orientation;
- newDevice.frontCameraType = FrontFacingCameraType.Android23;
- }
-
- camera = Camera.open(i);
- Camera.Parameters parameters = camera.getParameters();
- AddDeviceInfo(newDevice, parameters);
- camera.release();
- camera = null;
- deviceList.add(newDevice);
- }
- }
- else {
- // Prior to Android 2.3
- AndroidVideoCaptureDevice newDevice;
- Camera.Parameters parameters;
-
- newDevice = new AndroidVideoCaptureDevice();
- camera = Camera.open();
- parameters = camera.getParameters();
- newDevice.deviceUniqueName = "Camera 1, Facing back";
- newDevice.orientation = 90;
- AddDeviceInfo(newDevice, parameters);
-
- deviceList.add(newDevice);
- camera.release();
- camera=null;
-
- newDevice = new AndroidVideoCaptureDevice();
- newDevice.deviceUniqueName = "Camera 2, Facing front";
- parameters = SearchOldFrontFacingCameras(newDevice);
- if(parameters != null) {
- AddDeviceInfo(newDevice, parameters);
- deviceList.add(newDevice);
- }
- }
- }
- catch (Exception ex) {
- Log.e("*WEBRTC*", "Failed to init VideoCaptureDeviceInfo ex" +
- ex.getLocalizedMessage());
- return -1;
- }
- VerifyCapabilities();
- return 0;
- }
-
- // Adds the capture capabilities of the currently opened device
- private void AddDeviceInfo(AndroidVideoCaptureDevice newDevice,
- Camera.Parameters parameters) {
-
- List<Size> sizes = parameters.getSupportedPreviewSizes();
- List<Integer> frameRates = parameters.getSupportedPreviewFrameRates();
- int maxFPS=0;
- for(Integer frameRate:frameRates) {
- if(VERBOSE) {
- Log.v("*WEBRTC*",
- "VideoCaptureDeviceInfoAndroid:frameRate " + frameRate);
- }
- if(frameRate > maxFPS) {
- maxFPS = frameRate;
- }
- }
-
- newDevice.captureCapabilies = new CaptureCapabilityAndroid[sizes.size()];
- for(int i = 0; i < sizes.size(); ++i) {
- Size s = sizes.get(i);
- newDevice.captureCapabilies[i] = new CaptureCapabilityAndroid();
- newDevice.captureCapabilies[i].height = s.height;
- newDevice.captureCapabilies[i].width = s.width;
- newDevice.captureCapabilies[i].maxFPS = maxFPS;
- }
- }
-
- // Function that make sure device specific capabilities are
- // in the capability list.
- // Ie Galaxy S supports CIF but does not list CIF as a supported capability.
- // Motorola Droid Camera does not work with frame rate above 15fps.
- // http://code.google.com/p/android/issues/detail?id=5514#c0
- private void VerifyCapabilities() {
- // Nexus S or Galaxy S
- if(android.os.Build.DEVICE.equals("GT-I9000") ||
- android.os.Build.DEVICE.equals("crespo")) {
- CaptureCapabilityAndroid specificCapability =
- new CaptureCapabilityAndroid();
- specificCapability.width = 352;
- specificCapability.height = 288;
- specificCapability.maxFPS = 15;
- AddDeviceSpecificCapability(specificCapability);
-
- specificCapability = new CaptureCapabilityAndroid();
- specificCapability.width = 176;
- specificCapability.height = 144;
- specificCapability.maxFPS = 15;
- AddDeviceSpecificCapability(specificCapability);
-
- specificCapability = new CaptureCapabilityAndroid();
- specificCapability.width = 320;
- specificCapability.height = 240;
- specificCapability.maxFPS = 15;
- AddDeviceSpecificCapability(specificCapability);
- }
- // Motorola Milestone Camera server does not work at 30fps
- // even though it reports that it can
- if(android.os.Build.MANUFACTURER.equals("motorola") &&
- android.os.Build.DEVICE.equals("umts_sholes")) {
- for(AndroidVideoCaptureDevice device:deviceList) {
- for(CaptureCapabilityAndroid capability:device.captureCapabilies) {
- capability.maxFPS=15;
- }
- }
- }
- }
-
- private void AddDeviceSpecificCapability(
- CaptureCapabilityAndroid specificCapability) {
- for(AndroidVideoCaptureDevice device:deviceList) {
- boolean foundCapability = false;
- for(CaptureCapabilityAndroid capability:device.captureCapabilies) {
- if(capability.width == specificCapability.width &&
- capability.height == specificCapability.height) {
- foundCapability = true;
- break;
- }
- }
- if(foundCapability==false) {
- CaptureCapabilityAndroid newCaptureCapabilies[]=
- new CaptureCapabilityAndroid[device.captureCapabilies.length+1];
- for(int i = 0; i < device.captureCapabilies.length; ++i) {
- newCaptureCapabilies[i+1] = device.captureCapabilies[i];
- }
- newCaptureCapabilies[0] = specificCapability;
- device.captureCapabilies = newCaptureCapabilies;
- }
- }
- }
-
- // Returns the number of Capture devices that is supported
- public int NumberOfDevices() {
- return deviceList.size();
- }
-
- public String GetDeviceUniqueName(int deviceNumber) {
- if(deviceNumber < 0 || deviceNumber >= deviceList.size()) {
- return null;
- }
- return deviceList.get(deviceNumber).deviceUniqueName;
- }
-
- public CaptureCapabilityAndroid[] GetCapabilityArray (String deviceUniqueId)
- {
- for (AndroidVideoCaptureDevice device: deviceList) {
- if(device.deviceUniqueName.equals(deviceUniqueId)) {
- return (CaptureCapabilityAndroid[]) device.captureCapabilies;
- }
- }
- return null;
- }
-
- // Returns the camera orientation as described by
- // android.hardware.Camera.CameraInfo.orientation
- public int GetOrientation(String deviceUniqueId) {
- for (AndroidVideoCaptureDevice device: deviceList) {
- if(device.deviceUniqueName.equals(deviceUniqueId)) {
- return device.orientation;
- }
- }
- return -1;
- }
-
- // Returns an instance of VideoCaptureAndroid.
- public VideoCaptureAndroid AllocateCamera(int id, long context,
- String deviceUniqueId) {
- try {
- if(DEBUG) Log.d("*WEBRTC*", "AllocateCamera " + deviceUniqueId);
-
- Camera camera = null;
- AndroidVideoCaptureDevice deviceToUse = null;
- for (AndroidVideoCaptureDevice device: deviceList) {
- if(device.deviceUniqueName.equals(deviceUniqueId)) {
- // Found the wanted camera
- deviceToUse = device;
- switch(device.frontCameraType) {
- case GalaxyS:
- camera = AllocateGalaxySFrontCamera();
- break;
- case HTCEvo:
- camera = AllocateEVOFrontFacingCamera();
- break;
- default:
- // From Android 2.3 and onwards)
- if(android.os.Build.VERSION.SDK_INT>8)
- camera=Camera.open(device.index);
- else
- camera=Camera.open(); // Default camera
- }
- }
- }
-
- if(camera == null) {
- return null;
- }
- if(VERBOSE) {
- Log.v("*WEBRTC*", "AllocateCamera - creating VideoCaptureAndroid");
- }
-
- return new VideoCaptureAndroid(id,context,camera,deviceToUse);
-
- }catch (Exception ex) {
- Log.e("*WEBRTC*", "AllocateCamera Failed to open camera- ex " +
- ex.getLocalizedMessage());
- }
- return null;
- }
-
- // Searches for a front facing camera device. This is device specific code.
- private Camera.Parameters
- SearchOldFrontFacingCameras(AndroidVideoCaptureDevice newDevice)
- throws SecurityException, IllegalArgumentException,
- NoSuchMethodException, ClassNotFoundException,
- IllegalAccessException, InvocationTargetException {
- // Check the id of the opened camera device
- // Returns null on X10 and 1 on Samsung Galaxy S.
- Camera camera = Camera.open();
- Camera.Parameters parameters = camera.getParameters();
- String cameraId = parameters.get("camera-id");
- if(cameraId != null && cameraId.equals("1")) {
- // This might be a Samsung Galaxy S with a front facing camera.
- try {
- parameters.set("camera-id", 2);
- camera.setParameters(parameters);
- parameters = camera.getParameters();
- newDevice.frontCameraType = FrontFacingCameraType.GalaxyS;
- newDevice.orientation = 0;
- camera.release();
- return parameters;
- }
- catch (Exception ex) {
- //Nope - it did not work.
- Log.e("*WEBRTC*", "Init Failed to open front camera camera - ex " +
- ex.getLocalizedMessage());
- }
- }
- camera.release();
-
- //Check for Evo front facing camera
- File file =
- new File("/system/framework/com.htc.hardware.twinCamDevice.jar");
- boolean exists = file.exists();
- if (!exists){
- file =
- new File("/system/framework/com.sprint.hardware.twinCamDevice.jar");
- exists = file.exists();
- }
- if(exists) {
- newDevice.frontCameraType = FrontFacingCameraType.HTCEvo;
- newDevice.orientation = 0;
- Camera evCamera = AllocateEVOFrontFacingCamera();
- parameters = evCamera.getParameters();
- evCamera.release();
- return parameters;
- }
- return null;
- }
-
- // Returns a handle to HTC front facing camera.
- // The caller is responsible to release it on completion.
- private Camera AllocateEVOFrontFacingCamera()
- throws SecurityException, NoSuchMethodException,
- ClassNotFoundException, IllegalArgumentException,
- IllegalAccessException, InvocationTargetException {
- String classPath = null;
- File file =
- new File("/system/framework/com.htc.hardware.twinCamDevice.jar");
- classPath = "com.htc.hardware.twinCamDevice.FrontFacingCamera";
- boolean exists = file.exists();
- if (!exists){
- file =
- new File("/system/framework/com.sprint.hardware.twinCamDevice.jar");
- classPath = "com.sprint.hardware.twinCamDevice.FrontFacingCamera";
- exists = file.exists();
- }
- if(!exists) {
- return null;
- }
-
- String dexOutputDir = "";
- if(context != null) {
- dexOutputDir = context.getFilesDir().getAbsolutePath();
- File mFilesDir = new File(dexOutputDir, "dexfiles");
- if(!mFilesDir.exists()){
- //Log.e("*WEBRTCN*", "Directory doesn't exists");
- if(!mFilesDir.mkdirs()) {
- //Log.e("*WEBRTCN*", "Unable to create files directory");
- }
- }
- }
-
- dexOutputDir += "/dexfiles";
-
- DexClassLoader loader =
- new DexClassLoader(file.getAbsolutePath(), dexOutputDir,
- null, ClassLoader.getSystemClassLoader());
-
- Method method = loader.loadClass(classPath).getDeclaredMethod(
- "getFrontFacingCamera", (Class[]) null);
- Camera camera = (Camera) method.invoke((Object[])null,(Object[]) null);
- return camera;
- }
-
- // Returns a handle to Galaxy S front camera.
- // The caller is responsible to release it on completion.
- private Camera AllocateGalaxySFrontCamera()
- {
- Camera camera = Camera.open();
- Camera.Parameters parameters = camera.getParameters();
- parameters.set("camera-id",2);
- camera.setParameters(parameters);
- return camera;
- }
-
-}
diff --git a/trunk/src/modules/video_capture/main/source/Android/video_capture_android.cc b/trunk/src/modules/video_capture/main/source/Android/video_capture_android.cc
deleted file mode 100644
index acf7e3b..0000000
--- a/trunk/src/modules/video_capture/main/source/Android/video_capture_android.cc
+++ /dev/null
@@ -1,703 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_capture_android.h"
-
-#include <stdio.h>
-
-#include "critical_section_wrapper.h"
-#include "ref_count.h"
-#include "trace.h"
-namespace webrtc
-{
-namespace videocapturemodule
-{
-VideoCaptureModule* VideoCaptureImpl::Create(
- const WebRtc_Word32 id,
- const WebRtc_UWord8* deviceUniqueIdUTF8)
-{
-
- RefCountImpl<videocapturemodule::VideoCaptureAndroid>* implementation =
- new RefCountImpl<videocapturemodule::VideoCaptureAndroid>(id);
-
- if (!implementation || implementation->Init(id, deviceUniqueIdUTF8) != 0)
- {
- delete implementation;
- implementation = NULL;
- }
- return implementation;
-}
-
-// Android logging, uncomment to print trace to logcat instead of trace file/callback
-//#include <android/log.h>
-//#undef WEBRTC_TRACE
-//#define WEBRTC_TRACE(a,b,c,...) __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTCN*", __VA_ARGS__)
-
-JavaVM* VideoCaptureAndroid::g_jvm = NULL;
-jclass VideoCaptureAndroid::g_javaCmClass = NULL; //VideoCaptureAndroid.java
-jclass VideoCaptureAndroid::g_javaCmDevInfoClass = NULL; //VideoCaptureDeviceInfoAndroid.java
-jobject VideoCaptureAndroid::g_javaCmDevInfoObject = NULL; //static instance of VideoCaptureDeviceInfoAndroid.java
-jobject VideoCaptureAndroid::g_javaContext = NULL;
-
-/*
- * Register references to Java Capture class.
- */
-WebRtc_Word32 VideoCaptureAndroid::SetAndroidObjects(void* javaVM,
- void* javaContext)
-{
-
- g_jvm = static_cast<JavaVM*> (javaVM);
- g_javaContext = static_cast<jobject> (javaContext);
-
- if (javaVM)
- {
- JNIEnv* env = NULL;
- if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "%s: could not get Java environment", __FUNCTION__);
- return -1;
- }
- // get java capture class type (note path to class packet)
- jclass javaCmClassLocal = env->FindClass(AndroidJavaCaptureClass);
- if (!javaCmClassLocal)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "%s: could not find java class", __FUNCTION__);
- return -1;
- }
- // create a global reference to the class (to tell JNI that we are referencing it
- // after this function has returned)
- g_javaCmClass = static_cast<jclass>
- (env->NewGlobalRef(javaCmClassLocal));
- if (!g_javaCmClass)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "%s: InitVideoEngineJava(): could not create"
- " Java Camera class reference",
- __FUNCTION__);
- return -1;
- }
- // Delete local class ref, we only use the global ref
- env->DeleteLocalRef(javaCmClassLocal);
- JNINativeMethod nativeFunctions = { "ProvideCameraFrame", "([BIJ)V",
- (void*) &VideoCaptureAndroid::ProvideCameraFrame };
- if (env->RegisterNatives(g_javaCmClass, &nativeFunctions, 1) == 0)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
- "%s: Registered native functions", __FUNCTION__);
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "%s: Failed to register native functions",
- __FUNCTION__);
- return -1;
- }
-
- // get java capture class type (note path to class packet)
- jclass javaCmDevInfoClassLocal = env->FindClass(
- AndroidJavaCaptureDeviceInfoClass);
- if (!javaCmDevInfoClassLocal)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "%s: could not find java class", __FUNCTION__);
- return -1;
- }
-
- // create a global reference to the class (to tell JNI that we are referencing it
- // after this function has returned)
- g_javaCmDevInfoClass = static_cast<jclass>
- (env->NewGlobalRef(javaCmDevInfoClassLocal));
- if (!g_javaCmDevInfoClass)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "%s: InitVideoEngineJava(): could not create Java "
- "Camera Device info class reference",
- __FUNCTION__);
- return -1;
- }
- // Delete local class ref, we only use the global ref
- env->DeleteLocalRef(javaCmDevInfoClassLocal);
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
- "VideoCaptureDeviceInfoAndroid get method id");
-
- // get the method ID for the Android Java CaptureClass static
- //CreateVideoCaptureAndroid factory method.
- jmethodID cid = env->GetStaticMethodID(g_javaCmDevInfoClass,
- "CreateVideoCaptureDeviceInfoAndroid",
- "(ILandroid/content/Context;)"
- "Lorg/webrtc/videoengine/VideoCaptureDeviceInfoAndroid;");
- if (cid == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "%s: could not get java VideoCaptureDeviceInfoAndroid constructor ID",
- __FUNCTION__);
- return -1;
- }
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
- "%s: construct static java device object", __FUNCTION__);
-
- // construct the object by calling the static constructor object
- jobject javaCameraDeviceInfoObjLocal = env->CallStaticObjectMethod(
- g_javaCmDevInfoClass,
- cid, (int) -1,
- g_javaContext);
- if (!javaCameraDeviceInfoObjLocal)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, -1,
- "%s: could not create Java Capture Device info object",
- __FUNCTION__);
- return -1;
- }
- // create a reference to the object (to tell JNI that we are referencing it
- // after this function has returned)
- g_javaCmDevInfoObject = env->NewGlobalRef(javaCameraDeviceInfoObjLocal);
- if (!g_javaCmDevInfoObject)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioDevice, -1,
- "%s: could not create Java cameradevinceinfo object reference",
- __FUNCTION__);
- return -1;
- }
- // Delete local object ref, we only use the global ref
- env->DeleteLocalRef(javaCameraDeviceInfoObjLocal);
- return 0;
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
- "%s: JVM is NULL, assuming deinit", __FUNCTION__);
- if (!g_jvm)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "%s: SetAndroidObjects not called with a valid JVM.",
- __FUNCTION__);
- return -1;
- }
- JNIEnv* env = NULL;
- bool attached = false;
- if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = g_jvm->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
- -1, "%s: Could not attach thread to JVM (%d, %p)",
- __FUNCTION__, res, env);
- return -1;
- }
- attached = true;
- }
- env->DeleteGlobalRef(g_javaCmDevInfoObject);
- env->DeleteGlobalRef(g_javaCmDevInfoClass);
- env->DeleteGlobalRef(g_javaCmClass);
- if (attached && g_jvm->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, -1,
- "%s: Could not detach thread from JVM", __FUNCTION__);
- return -1;
- }
- return 0;
- env = (JNIEnv *) NULL;
- }
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureAndroid::AttachAndUseAndroidDeviceInfoObjects(
- JNIEnv*& env,
- jclass& javaCmDevInfoClass,
- jobject& javaCmDevInfoObject,
- bool& attached)
-{
- // get the JNI env for this thread
- if (!g_jvm)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "%s: SetAndroidObjects not called with a valid JVM.",
- __FUNCTION__);
- return -1;
- }
- attached = false;
- if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = g_jvm->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "%s: Could not attach thread to JVM (%d, %p)",
- __FUNCTION__, res, env);
- return -1;
- }
- attached = true;
- }
- javaCmDevInfoClass = g_javaCmDevInfoClass;
- javaCmDevInfoObject = g_javaCmDevInfoObject;
- return 0;
-
-}
-
-WebRtc_Word32 VideoCaptureAndroid::ReleaseAndroidDeviceInfoObjects(bool attached)
-{
- if (attached && g_jvm->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, -1,
- "%s: Could not detach thread from JVM", __FUNCTION__);
- return -1;
- }
- return 0;
-}
-
-/*
- * JNI callback from Java class. Called when the camera has a new frame to deliver
- * Class: org_webrtc_capturemodule_VideoCaptureAndroid
- * Method: ProvideCameraFrame
- * Signature: ([BIJ)V
- */
-void JNICALL VideoCaptureAndroid::ProvideCameraFrame(JNIEnv * env,
- jobject,
- jbyteArray javaCameraFrame,
- jint length,
- jlong context)
-{
- VideoCaptureAndroid* captureModule=reinterpret_cast<VideoCaptureAndroid*>(context);
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture,
- -1, "%s: IncomingFrame %d", __FUNCTION__,length);
- jbyte* cameraFrame= env->GetByteArrayElements(javaCameraFrame,NULL);
- captureModule->IncomingFrame((WebRtc_UWord8*) cameraFrame,
- length,captureModule->_frameInfo,0);
- env->ReleaseByteArrayElements(javaCameraFrame,cameraFrame,JNI_ABORT);
-}
-
-
-
-VideoCaptureAndroid::VideoCaptureAndroid(const WebRtc_Word32 id)
- : VideoCaptureImpl(id), _capInfo(id), _javaCaptureObj(NULL),
- _captureStarted(false)
-
-{
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
- "%s: context %x", __FUNCTION__, (int) this);
-}
-// ----------------------------------------------------------------------------
-// Init
-//
-// Initializes needed Java resources like the JNI interface to
-// VideoCaptureAndroid.java
-// ----------------------------------------------------------------------------
-WebRtc_Word32 VideoCaptureAndroid::Init(const WebRtc_Word32 id,
- const WebRtc_UWord8* deviceUniqueIdUTF8)
-{
- const int nameLength = strlen((char*) deviceUniqueIdUTF8);
- if (nameLength >= kVideoCaptureUniqueNameLength)
- {
- return -1;
- }
-
- // Store the device name
- _deviceUniqueId = new WebRtc_UWord8[nameLength + 1];
- memcpy(_deviceUniqueId, deviceUniqueIdUTF8, nameLength + 1);
-
- if (_capInfo.Init() != 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s: Failed to initialize CaptureDeviceInfo", __FUNCTION__);
- return -1;
- }
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1, "%s:",
- __FUNCTION__);
- // use the jvm that has been set
- if (!g_jvm)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s: Not a valid Java VM pointer", __FUNCTION__);
- return -1;
- }
- // get the JNI env for this thread
- JNIEnv *env;
- bool isAttached = false;
-
- // get the JNI env for this thread
- if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = g_jvm->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s: Could not attach thread to JVM (%d, %p)",
- __FUNCTION__, res, env);
- return -1;
- }
- isAttached = true;
- }
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- "get method id");
-
- // get the method ID for the Android Java CaptureDeviceInfoClass AllocateCamera factory method.
- char signature[256];
- sprintf(signature, "(IJLjava/lang/String;)L%s;", AndroidJavaCaptureClass);
-
- jmethodID cid = env->GetMethodID(g_javaCmDevInfoClass, "AllocateCamera",
- signature);
- if (cid == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s: could not get constructor ID", __FUNCTION__);
- return -1; /* exception thrown */
- }
-
- jstring capureIdString = env->NewStringUTF((char*) deviceUniqueIdUTF8);
- // construct the object by calling the static constructor object
- jobject javaCameraObjLocal = env->CallObjectMethod(g_javaCmDevInfoObject,
- cid, (jint) id,
- (jlong) this,
- capureIdString);
- if (!javaCameraObjLocal)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
- "%s: could not create Java Capture object", __FUNCTION__);
- return -1;
- }
-
- // create a reference to the object (to tell JNI that we are referencing it
- // after this function has returned)
- _javaCaptureObj = env->NewGlobalRef(javaCameraObjLocal);
- if (!_javaCaptureObj)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceAudioDevice, _id,
- "%s: could not create Java camera object reference",
- __FUNCTION__);
- return -1;
- }
-
- // Delete local object ref, we only use the global ref
- env->DeleteLocalRef(javaCameraObjLocal);
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (g_jvm->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioDevice, _id,
- "%s: Could not detach thread from JVM", __FUNCTION__);
- }
- }
-
- return 0;
-}
-
-VideoCaptureAndroid::~VideoCaptureAndroid()
-{
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1, "%s:",
- __FUNCTION__);
- if (_javaCaptureObj == NULL || g_jvm == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "%s: Nothing to clean", __FUNCTION__);
- }
- else
- {
- bool isAttached = false;
- // get the JNI env for this thread
- JNIEnv *env;
- if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = g_jvm->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
- _id,
- "%s: Could not attach thread to JVM (%d, %p)",
- __FUNCTION__, res, env);
- }
- else
- {
- isAttached = true;
- }
- }
-
- // get the method ID for the Android Java CaptureClass static
- // DeleteVideoCaptureAndroid method. Call this to release the camera so
- // another application can use it.
- jmethodID cid = env->GetStaticMethodID(g_javaCmClass,
- "DeleteVideoCaptureAndroid",
- "(Lorg/webrtc/videoengine/VideoCaptureAndroid;)V");
- if (cid != NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
- "%s: Call DeleteVideoCaptureAndroid", __FUNCTION__);
- // Close the camera by calling the static destruct function.
- env->CallStaticVoidMethod(g_javaCmClass, cid, _javaCaptureObj);
-
- // Delete global object ref to the camera.
- env->DeleteGlobalRef(_javaCaptureObj);
- _javaCaptureObj = NULL;
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "%s: Failed to find DeleteVideoCaptureAndroid id",
- __FUNCTION__);
- }
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (g_jvm->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioDevice,
- _id, "%s: Could not detach thread from JVM",
- __FUNCTION__);
- }
- }
- }
-}
-
-WebRtc_Word32 VideoCaptureAndroid::StartCapture(
- const VideoCaptureCapability& capability)
-{
- CriticalSectionScoped cs(_apiCs);
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
- "%s: ", __FUNCTION__);
-
- bool isAttached = false;
- WebRtc_Word32 result = 0;
- // get the JNI env for this thread
- JNIEnv *env;
- if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = g_jvm->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s: Could not attach thread to JVM (%d, %p)",
- __FUNCTION__, res, env);
- }
- else
- {
- isAttached = true;
- }
- }
-
- if (_capInfo.GetBestMatchedCapability(_deviceUniqueId, capability,
- _frameInfo) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "%s: GetBestMatchedCapability failed. Req cap w%d h%d",
- __FUNCTION__, capability.width, capability.height);
- return -1;
- }
-
- // Store the new expected capture delay
- _captureDelay = _frameInfo.expectedCaptureDelay;
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
- "%s: _frameInfo w%d h%d", __FUNCTION__, _frameInfo.width,
- _frameInfo.height);
-
- // get the method ID for the Android Java CaptureClass static StartCapture method.
- jmethodID cid = env->GetMethodID(g_javaCmClass, "StartCapture", "(III)I");
- if (cid != NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
- "%s: Call StartCapture", __FUNCTION__);
- // Close the camera by calling the static destruct function.
- result = env->CallIntMethod(_javaCaptureObj, cid, _frameInfo.width,
- _frameInfo.height, _frameInfo.maxFPS);
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "%s: Failed to find StartCapture id", __FUNCTION__);
- }
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (g_jvm->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioDevice, _id,
- "%s: Could not detach thread from JVM", __FUNCTION__);
- }
- }
- if (result == 0)
- {
- _requestedCapability = capability;
- _captureStarted = true;
- }
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
- "%s: result %d", __FUNCTION__, result);
- return result;
-}
-WebRtc_Word32 VideoCaptureAndroid::StopCapture()
-{
- CriticalSectionScoped cs(_apiCs);
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
- "%s: ", __FUNCTION__);
-
- bool isAttached = false;
- WebRtc_Word32 result = 0;
- // get the JNI env for this thread
- JNIEnv *env = NULL;
- if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = g_jvm->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s: Could not attach thread to JVM (%d, %p)",
- __FUNCTION__, res, env);
- }
- else
- {
- isAttached = true;
- }
- }
-
- memset(&_requestedCapability, 0, sizeof(_requestedCapability));
- memset(&_frameInfo, 0, sizeof(_frameInfo));
-
- // get the method ID for the Android Java CaptureClass StopCapture method.
- jmethodID cid = env->GetMethodID(g_javaCmClass, "StopCapture", "()I");
- if (cid != NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, -1,
- "%s: Call StopCapture", __FUNCTION__);
- // Close the camera by calling the static destruct function.
- result = env->CallIntMethod(_javaCaptureObj, cid);
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "%s: Failed to find StopCapture id", __FUNCTION__);
- }
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (g_jvm->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioDevice, _id,
- "%s: Could not detach thread from JVM", __FUNCTION__);
- }
- }
- _captureStarted = false;
-
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
- "%s: result %d", __FUNCTION__, result);
- return result;
-}
-
-bool VideoCaptureAndroid::CaptureStarted()
-{
- CriticalSectionScoped cs(_apiCs);
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
- "%s: ", __FUNCTION__);
- return _captureStarted;
-}
-WebRtc_Word32 VideoCaptureAndroid::CaptureSettings(
- VideoCaptureCapability& settings)
-{
- CriticalSectionScoped cs(_apiCs);
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, -1,
- "%s: ", __FUNCTION__);
- settings = _requestedCapability;
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureAndroid::SetCaptureRotation(
- VideoCaptureRotation rotation)
-{
- CriticalSectionScoped cs(_apiCs);
- if (VideoCaptureImpl::SetCaptureRotation(rotation) == 0)
- {
- if (!g_jvm)
- return -1;
-
- // get the JNI env for this thread
- JNIEnv *env;
- bool isAttached = false;
-
- // get the JNI env for this thread
- if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = g_jvm->AttachCurrentThread(&env, NULL);
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
- _id,
- "%s: Could not attach thread to JVM (%d, %p)",
- __FUNCTION__, res, env);
- return -1;
- }
- isAttached = true;
- }
-
- jmethodID cid = env->GetMethodID(g_javaCmClass, "SetPreviewRotation",
- "(I)V");
- if (cid == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "%s: could not get java SetPreviewRotation ID",
- __FUNCTION__);
- return -1;
- }
- jint rotateFrame = 0;
- switch (rotation)
- {
- case kCameraRotate0:
- rotateFrame = 0;
- break;
- case kCameraRotate90:
- rotateFrame = 90;
- break;
- case kCameraRotate180:
- rotateFrame = 180;
- break;
- case kCameraRotate270:
- rotateFrame = 270;
- break;
- }
- env->CallVoidMethod(_javaCaptureObj, cid, rotateFrame);
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (g_jvm->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceAudioDevice,
- _id, "%s: Could not detach thread from JVM",
- __FUNCTION__);
- }
- }
-
- }
- return 0;
-}
-} //namespace videocapturemodule
-} //namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Android/video_capture_android.h b/trunk/src/modules/video_capture/main/source/Android/video_capture_android.h
deleted file mode 100644
index abb4dc9..0000000
--- a/trunk/src/modules/video_capture/main/source/Android/video_capture_android.h
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_ANDROID_VIDEO_CAPTURE_ANDROID_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_ANDROID_VIDEO_CAPTURE_ANDROID_H_
-
-#include <jni.h>
-#include "device_info_android.h"
-#include "video_capture_impl.h"
-
-#define AndroidJavaCaptureClass "org/webrtc/videoengine/VideoCaptureAndroid"
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-class VideoCaptureAndroid: public VideoCaptureImpl
-{
-public:
- static WebRtc_Word32 SetAndroidObjects(void* javaVM, void* javaContext);
- static WebRtc_Word32 AttachAndUseAndroidDeviceInfoObjects(JNIEnv*& env,
- jclass& javaCmDevInfoClass,
- jobject& javaCmDevInfoObject,
- bool& attached);
- static WebRtc_Word32 ReleaseAndroidDeviceInfoObjects(bool attached);
-
- VideoCaptureAndroid(const WebRtc_Word32 id);
- virtual WebRtc_Word32 Init(const WebRtc_Word32 id,
- const WebRtc_UWord8* deviceUniqueIdUTF8);
-
-
- virtual WebRtc_Word32 StartCapture(const VideoCaptureCapability& capability);
- virtual WebRtc_Word32 StopCapture();
- virtual bool CaptureStarted();
- virtual WebRtc_Word32 CaptureSettings(VideoCaptureCapability& settings);
- virtual WebRtc_Word32 SetCaptureRotation(VideoCaptureRotation rotation);
-
-protected:
- virtual ~VideoCaptureAndroid();
- static void JNICALL ProvideCameraFrame (JNIEnv * env,
- jobject,
- jbyteArray javaCameraFrame,
- jint length, jlong context);
- DeviceInfoAndroid _capInfo;
- jobject _javaCaptureObj; // Java Camera object.
- VideoCaptureCapability _frameInfo;
- bool _captureStarted;
-
- static JavaVM* g_jvm;
- static jclass g_javaCmClass;
- static jclass g_javaCmDevInfoClass;
- static jobject g_javaCmDevInfoObject; //Static java object implementing the needed device info functions;
- static jobject g_javaContext; // Java Application context
-};
-} // namespace videocapturemodule
-} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_ANDROID_VIDEO_CAPTURE_ANDROID_H_
diff --git a/trunk/src/modules/video_capture/main/source/External/device_info_external.cc b/trunk/src/modules/video_capture/main/source/External/device_info_external.cc
deleted file mode 100644
index 2aad081..0000000
--- a/trunk/src/modules/video_capture/main/source/External/device_info_external.cc
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "../device_info_impl.h"
-#include "../video_capture_impl.h"
-
-namespace webrtc {
-
-namespace videocapturemodule {
-
-class ExternalDeviceInfo : public DeviceInfoImpl {
- public:
- ExternalDeviceInfo(const WebRtc_Word32 id)
- : DeviceInfoImpl(id) {
- }
- virtual ~ExternalDeviceInfo() {}
- virtual WebRtc_UWord32 NumberOfDevices() { return 0; }
- virtual WebRtc_Word32 DisplayCaptureSettingsDialogBox(
- const WebRtc_UWord8* /*deviceUniqueIdUTF8*/,
- const WebRtc_UWord8* /*dialogTitleUTF8*/,
- void* /*parentWindow*/,
- WebRtc_UWord32 /*positionX*/,
- WebRtc_UWord32 /*positionY*/) { return -1; }
- virtual WebRtc_Word32 GetDeviceName(
- WebRtc_UWord32 deviceNumber,
- WebRtc_UWord8* deviceNameUTF8,
- WebRtc_UWord32 deviceNameLength,
- WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord32 deviceUniqueIdUTF8Length,
- WebRtc_UWord8* productUniqueIdUTF8=0,
- WebRtc_UWord32 productUniqueIdUTF8Length=0) {
- return -1;
- }
- virtual WebRtc_Word32 CreateCapabilityMap(
- const WebRtc_UWord8* deviceUniqueIdUTF8) { return 0; }
- virtual WebRtc_Word32 Init() { return 0; }
-};
-
-VideoCaptureModule::DeviceInfo* VideoCaptureImpl::CreateDeviceInfo(
- const WebRtc_Word32 id) {
- return new ExternalDeviceInfo(id);
-}
-
-} // namespace videocapturemodule
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/External/video_capture_external.cc b/trunk/src/modules/video_capture/main/source/External/video_capture_external.cc
deleted file mode 100644
index e8cbccb..0000000
--- a/trunk/src/modules/video_capture/main/source/External/video_capture_external.cc
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "../video_capture_impl.h"
-#include "ref_count.h"
-
-namespace webrtc {
-
-namespace videocapturemodule {
-
-VideoCaptureModule* VideoCaptureImpl::Create(
- const WebRtc_Word32 id,
- const WebRtc_UWord8* deviceUniqueIdUTF8) {
- RefCountImpl<VideoCaptureImpl>* implementation =
- new RefCountImpl<VideoCaptureImpl>(id);
- return implementation;
-}
-
-} // namespace videocapturemodule
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Linux/device_info_linux.cc b/trunk/src/modules/video_capture/main/source/Linux/device_info_linux.cc
deleted file mode 100644
index c67885f..0000000
--- a/trunk/src/modules/video_capture/main/source/Linux/device_info_linux.cc
+++ /dev/null
@@ -1,330 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "device_info_linux.h"
-
-#include <errno.h>
-#include <unistd.h>
-#include <sys/ioctl.h>
-#include <sys/stat.h>
-#include <fcntl.h>
-#include <stdio.h>
-#include <stdlib.h>
-
-//v4l includes
-#include <linux/videodev2.h>
-
-#include "ref_count.h"
-#include "trace.h"
-
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-VideoCaptureModule::DeviceInfo*
-VideoCaptureImpl::CreateDeviceInfo(const WebRtc_Word32 id)
-{
- videocapturemodule::DeviceInfoLinux *deviceInfo =
- new videocapturemodule::DeviceInfoLinux(id);
- if (!deviceInfo)
- {
- deviceInfo = NULL;
- }
-
- return deviceInfo;
-}
-
-DeviceInfoLinux::DeviceInfoLinux(const WebRtc_Word32 id)
- : DeviceInfoImpl(id)
-{
-}
-
-WebRtc_Word32 DeviceInfoLinux::Init()
-{
- return 0;
-}
-
-DeviceInfoLinux::~DeviceInfoLinux()
-{
-}
-
-WebRtc_UWord32 DeviceInfoLinux::NumberOfDevices()
-{
- WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideoCapture, _id, "%s", __FUNCTION__);
-
- WebRtc_UWord32 count = 0;
- char device[20];
- int fd = -1;
-
- /* detect /dev/video [0-63]VideoCaptureModule entries */
- for (int n = 0; n < 64; n++)
- {
- sprintf(device, "/dev/video%d", n);
- if ((fd = open(device, O_RDONLY)) != -1)
- {
- close(fd);
- count++;
- }
- }
-
- return count;
-}
-
-WebRtc_Word32 DeviceInfoLinux::GetDeviceName(
- WebRtc_UWord32 deviceNumber,
- WebRtc_UWord8* deviceNameUTF8,
- WebRtc_UWord32 deviceNameLength,
- WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord32 deviceUniqueIdUTF8Length,
- WebRtc_UWord8* /*productUniqueIdUTF8*/,
- WebRtc_UWord32 /*productUniqueIdUTF8Length*/)
-{
- WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideoCapture, _id, "%s", __FUNCTION__);
-
- // Travel through /dev/video [0-63]
- WebRtc_UWord32 count = 0;
- char device[20];
- int fd = -1;
- bool found = false;
- for (int n = 0; n < 64; n++)
- {
- sprintf(device, "/dev/video%d", n);
- if ((fd = open(device, O_RDONLY)) != -1)
- {
- if (count == deviceNumber) {
- // Found the device
- found = true;
- break;
- } else {
- close(fd);
- count++;
- }
- }
- }
-
- if (!found)
- return -1;
-
- // query device capabilities
- struct v4l2_capability cap;
- if (ioctl(fd, VIDIOC_QUERYCAP, &cap) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "error in querying the device capability for device %s. errno = %d",
- device, errno);
- close(fd);
- return -1;
- }
-
- close(fd);
-
- char cameraName[64];
- memset(deviceNameUTF8, 0, deviceNameLength);
- memcpy(cameraName, cap.card, sizeof(cap.card));
-
- if (deviceNameLength >= strlen(cameraName))
- {
- memcpy(deviceNameUTF8, cameraName, strlen(cameraName));
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, "buffer passed is too small");
- return -1;
- }
-
- if (cap.bus_info[0] != 0) // may not available in all drivers
- {
- // copy device id
- if (deviceUniqueIdUTF8Length >= strlen((const char*) cap.bus_info))
- {
- memset(deviceUniqueIdUTF8, 0, deviceUniqueIdUTF8Length);
- memcpy(deviceUniqueIdUTF8, cap.bus_info,
- strlen((const char*) cap.bus_info));
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "buffer passed is too small");
- return -1;
- }
- }
-
- return 0;
-}
-
-WebRtc_Word32 DeviceInfoLinux::CreateCapabilityMap(
- const WebRtc_UWord8* deviceUniqueIdUTF8)
-{
- int fd;
- char device[32];
- bool found = false;
-
- const WebRtc_Word32 deviceUniqueIdUTF8Length =
- (WebRtc_Word32) strlen((char*) deviceUniqueIdUTF8);
- if (deviceUniqueIdUTF8Length > kVideoCaptureUniqueNameLength)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, "Device name too long");
- return -1;
- }
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "CreateCapabilityMap called for device %s", deviceUniqueIdUTF8);
-
- /* detect /dev/video [0-63] entries */
- for (int n = 0; n < 64; ++n)
- {
- sprintf(device, "/dev/video%d", n);
- fd = open(device, O_RDONLY);
- if (fd == -1)
- continue;
-
- // query device capabilities
- struct v4l2_capability cap;
- if (ioctl(fd, VIDIOC_QUERYCAP, &cap) == 0)
- {
- if (cap.bus_info[0] != 0)
- {
- if (strncmp((const char*) cap.bus_info,
- (const char*) deviceUniqueIdUTF8,
- strlen((const char*) deviceUniqueIdUTF8)) == 0) //match with device id
- {
- found = true;
- break; // fd matches with device unique id supplied
- }
- }
- else //match for device name
- {
- if (IsDeviceNameMatches((const char*) cap.card,
- (const char*) deviceUniqueIdUTF8))
- {
- found = true;
- break;
- }
- }
- }
- close(fd); // close since this is not the matching device
- }
-
- if (!found)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, "no matching device found");
- return -1;
- }
-
- // now fd will point to the matching device
- // reset old capability map
- MapItem* item = NULL;
- while ((item = _captureCapabilities.Last()))
- {
- delete static_cast<VideoCaptureCapability*> (item->GetItem());
- _captureCapabilities.Erase(item);
- }
-
- int size = FillCapabilityMap(fd);
- close(fd);
-
- // Store the new used device name
- _lastUsedDeviceNameLength = deviceUniqueIdUTF8Length;
- _lastUsedDeviceName = (WebRtc_UWord8*) realloc(_lastUsedDeviceName,
- _lastUsedDeviceNameLength + 1);
- memcpy(_lastUsedDeviceName, deviceUniqueIdUTF8, _lastUsedDeviceNameLength + 1);
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id, "CreateCapabilityMap %d",
- _captureCapabilities.Size());
-
- return size;
-}
-
-bool DeviceInfoLinux::IsDeviceNameMatches(const char* name,
- const char* deviceUniqueIdUTF8)
-{
- if (strncmp(deviceUniqueIdUTF8, name, strlen(name)) == 0)
- return true;
- return false;
-}
-
-WebRtc_Word32 DeviceInfoLinux::FillCapabilityMap(int fd)
-{
-
- // set image format
- struct v4l2_format video_fmt;
- memset(&video_fmt, 0, sizeof(struct v4l2_format));
-
- video_fmt.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
- video_fmt.fmt.pix.sizeimage = 0;
-
- int totalFmts = 3;
- unsigned int videoFormats[] = {
- V4L2_PIX_FMT_MJPEG,
- V4L2_PIX_FMT_YUV420,
- V4L2_PIX_FMT_YUYV };
-
- int sizes = 13;
- unsigned int size[][2] = { { 128, 96 }, { 160, 120 }, { 176, 144 },
- { 320, 240 }, { 352, 288 }, { 640, 480 },
- { 704, 576 }, { 800, 600 }, { 960, 720 },
- { 1280, 720 }, { 1024, 768 }, { 1440, 1080 },
- { 1920, 1080 } };
-
- int index = 0;
- for (int fmts = 0; fmts < totalFmts; fmts++)
- {
- for (int i = 0; i < sizes; i++)
- {
- video_fmt.fmt.pix.pixelformat = videoFormats[fmts];
- video_fmt.fmt.pix.width = size[i][0];
- video_fmt.fmt.pix.height = size[i][1];
-
- if (ioctl(fd, VIDIOC_TRY_FMT, &video_fmt) >= 0)
- {
- if ((video_fmt.fmt.pix.width == size[i][0])
- && (video_fmt.fmt.pix.height == size[i][1]))
- {
- VideoCaptureCapability *cap = new VideoCaptureCapability();
- cap->width = video_fmt.fmt.pix.width;
- cap->height = video_fmt.fmt.pix.height;
- cap->expectedCaptureDelay = 120;
- if (videoFormats[fmts] == V4L2_PIX_FMT_YUYV)
- {
- cap->rawType = kVideoYUY2;
- }
- else if (videoFormats[fmts] == V4L2_PIX_FMT_MJPEG)
- {
- cap->rawType = kVideoMJPEG;
- }
-
- // get fps of current camera mode
- // V4l2 does not have a stable method of knowing so we just guess.
- if(cap->width >= 800 && cap->rawType != kVideoMJPEG)
- {
- cap->maxFPS = 15;
- }
- else
- {
- cap->maxFPS = 30;
- }
-
- _captureCapabilities.Insert(index, cap);
- index++;
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "Camera capability, width:%d height:%d type:%d fps:%d",
- cap->width, cap->height, cap->rawType, cap->maxFPS);
- }
- }
- }
- }
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id, "CreateCapabilityMap %d",
- _captureCapabilities.Size());
- return _captureCapabilities.Size();
-}
-
-} // namespace videocapturemodule
-} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Linux/device_info_linux.h b/trunk/src/modules/video_capture/main/source/Linux/device_info_linux.h
deleted file mode 100644
index 3d0e031..0000000
--- a/trunk/src/modules/video_capture/main/source/Linux/device_info_linux.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_DEVICE_INFO_LINUX_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_DEVICE_INFO_LINUX_H_
-
-#include "../video_capture_impl.h"
-#include "../device_info_impl.h"
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-class DeviceInfoLinux: public DeviceInfoImpl
-{
-public:
- DeviceInfoLinux(const WebRtc_Word32 id);
- virtual ~DeviceInfoLinux();
- virtual WebRtc_UWord32 NumberOfDevices();
- virtual WebRtc_Word32 GetDeviceName(WebRtc_UWord32 deviceNumber,
- WebRtc_UWord8* deviceNameUTF8,
- WebRtc_UWord32 deviceNameLength,
- WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord32 deviceUniqueIdUTF8Length,
- WebRtc_UWord8* productUniqueIdUTF8=0,
- WebRtc_UWord32 productUniqueIdUTF8Length=0);
- /*
- * Fills the membervariable _captureCapabilities with capabilites for the given device name.
- */
- virtual WebRtc_Word32 CreateCapabilityMap (const WebRtc_UWord8* deviceUniqueIdUTF8);
- virtual WebRtc_Word32 DisplayCaptureSettingsDialogBox(const WebRtc_UWord8* /*deviceUniqueIdUTF8*/,
- const WebRtc_UWord8* /*dialogTitleUTF8*/,
- void* /*parentWindow*/,
- WebRtc_UWord32 /*positionX*/,
- WebRtc_UWord32 /*positionY*/) { return -1;}
- WebRtc_Word32 FillCapabilityMap(int fd);
- WebRtc_Word32 Init();
-private:
-
- bool IsDeviceNameMatches(const char* name, const char* deviceUniqueIdUTF8);
-};
-} // namespace videocapturemodule
-} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_DEVICE_INFO_LINUX_H_
diff --git a/trunk/src/modules/video_capture/main/source/Linux/video_capture_linux.cc b/trunk/src/modules/video_capture/main/source/Linux/video_capture_linux.cc
deleted file mode 100644
index 6b47f24..0000000
--- a/trunk/src/modules/video_capture/main/source/Linux/video_capture_linux.cc
+++ /dev/null
@@ -1,464 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <sys/ioctl.h>
-#include <unistd.h>
-#include <sys/stat.h>
-#include <fcntl.h>
-#include <linux/videodev2.h>
-#include <errno.h>
-#include <stdio.h>
-#include <sys/mman.h>
-#include <string.h>
-
-#include <iostream>
-#include <new>
-
-#include "ref_count.h"
-#include "trace.h"
-#include "thread_wrapper.h"
-#include "critical_section_wrapper.h"
-#include "video_capture_linux.h"
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-VideoCaptureModule* VideoCaptureImpl::Create(const WebRtc_Word32 id,
- const WebRtc_UWord8* deviceUniqueId)
-{
- RefCountImpl<videocapturemodule::VideoCaptureModuleV4L2>* implementation =
- new RefCountImpl<videocapturemodule::VideoCaptureModuleV4L2>(id);
-
- if (!implementation || implementation->Init(deviceUniqueId) != 0)
- {
- delete implementation;
- implementation = NULL;
- }
-
- return implementation;
-}
-
-VideoCaptureModuleV4L2::VideoCaptureModuleV4L2(const WebRtc_Word32 id)
- : VideoCaptureImpl(id),
- _captureThread(NULL),
- _captureCritSect(CriticalSectionWrapper::CreateCriticalSection()),
- _deviceId(-1),
- _deviceFd(-1),
- _buffersAllocatedByDevice(-1),
- _currentWidth(-1),
- _currentHeight(-1),
- _currentFrameRate(-1),
- _captureStarted(false),
- _captureVideoType(kVideoI420),
- _pool(NULL)
-{
-}
-
-WebRtc_Word32 VideoCaptureModuleV4L2::Init(const WebRtc_UWord8* deviceUniqueIdUTF8)
-{
- int len = strlen((const char*) deviceUniqueIdUTF8);
- _deviceUniqueId = new (std::nothrow) WebRtc_UWord8[len + 1];
- if (_deviceUniqueId)
- {
- memcpy(_deviceUniqueId, deviceUniqueIdUTF8, len + 1);
- }
-
- int fd;
- char device[32];
- bool found = false;
-
- /* detect /dev/video [0-63] entries */
- int n;
- for (n = 0; n < 64; n++)
- {
- sprintf(device, "/dev/video%d", n);
- if ((fd = open(device, O_RDONLY)) != -1)
- {
- // query device capabilities
- struct v4l2_capability cap;
- if (ioctl(fd, VIDIOC_QUERYCAP, &cap) == 0)
- {
- if (cap.bus_info[0] != 0)
- {
- if (strncmp((const char*) cap.bus_info,
- (const char*) deviceUniqueIdUTF8,
- strlen((const char*) deviceUniqueIdUTF8)) == 0) //match with device id
- {
- close(fd);
- found = true;
- break; // fd matches with device unique id supplied
- }
- }
- }
- close(fd); // close since this is not the matching device
- }
- }
- if (!found)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id, "no matching device found");
- return -1;
- }
- _deviceId = n; //store the device id
- return 0;
-}
-
-VideoCaptureModuleV4L2::~VideoCaptureModuleV4L2()
-{
- StopCapture();
- if (_captureCritSect)
- {
- delete _captureCritSect;
- }
- if (_deviceFd != -1)
- close(_deviceFd);
-}
-
-WebRtc_Word32 VideoCaptureModuleV4L2::StartCapture(
- const VideoCaptureCapability& capability)
-{
- if (_captureStarted)
- {
- if (capability.width == _currentWidth &&
- capability.height == _currentHeight &&
- _captureVideoType == capability.rawType)
- {
- return 0;
- }
- else
- {
- StopCapture();
- }
- }
-
- CriticalSectionScoped cs(*_captureCritSect);
- //first open /dev/video device
- char device[20];
- sprintf(device, "/dev/video%d", (int) _deviceId);
-
- if ((_deviceFd = open(device, O_RDWR | O_NONBLOCK, 0)) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "error in opening %s errono = %d", device, errno);
- return -1;
- }
-
- // Supported video formats in preferred order.
- // If the requested resolution is larger than VGA, we prefer MJPEG. Go for
- // I420 otherwise.
- const int nFormats = 3;
- unsigned int fmts[nFormats];
- if (capability.width > 640 || capability.height > 480) {
- fmts[0] = V4L2_PIX_FMT_MJPEG;
- fmts[1] = V4L2_PIX_FMT_YUV420;
- fmts[2] = V4L2_PIX_FMT_YUYV;
- } else {
- fmts[0] = V4L2_PIX_FMT_YUV420;
- fmts[1] = V4L2_PIX_FMT_YUYV;
- fmts[2] = V4L2_PIX_FMT_MJPEG;
- }
-
- struct v4l2_format video_fmt;
- memset(&video_fmt, 0, sizeof(struct v4l2_format));
- video_fmt.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
- video_fmt.fmt.pix.sizeimage = 0;
- video_fmt.fmt.pix.width = capability.width;
- video_fmt.fmt.pix.height = capability.height;
-
- bool formatMatch = false;
- for (int i = 0; i < nFormats; i++)
- {
- video_fmt.fmt.pix.pixelformat = fmts[i];
- if (ioctl(_deviceFd, VIDIOC_TRY_FMT, &video_fmt) < 0)
- {
- continue;
- }
- else
- {
- formatMatch = true;
- break;
- }
- }
- if (!formatMatch)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "no supporting video formats found");
- return -1;
- }
-
- if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_YUYV)
- _captureVideoType = kVideoYUY2;
- else if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_YUV420)
- _captureVideoType = kVideoI420;
- else if (video_fmt.fmt.pix.pixelformat == V4L2_PIX_FMT_MJPEG)
- _captureVideoType = kVideoMJPEG;
-
- //set format and frame size now
- if (ioctl(_deviceFd, VIDIOC_S_FMT, &video_fmt) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "error in VIDIOC_S_FMT, errno = %d", errno);
- return -1;
- }
-
- // initialize current width and height
- _currentWidth = video_fmt.fmt.pix.width;
- _currentHeight = video_fmt.fmt.pix.height;
- _captureDelay = 120;
- // No way of knowing frame rate, make a guess.
- if(_currentWidth >= 800 && _captureVideoType != kVideoMJPEG)
- _currentFrameRate = 15;
- else
- _currentFrameRate = 30;
-
- if (!AllocateVideoBuffers())
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "failed to allocate video capture buffers");
- return -1;
- }
-
- //start capture thread;
- if (!_captureThread)
- {
- _captureThread = ThreadWrapper::CreateThread(
- VideoCaptureModuleV4L2::CaptureThread, this, kHighPriority);
- unsigned int id;
- _captureThread->Start(id);
- }
-
- // Needed to start UVC camera - from the uvcview application
- enum v4l2_buf_type type;
- type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
- if (ioctl(_deviceFd, VIDIOC_STREAMON, &type) == -1)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to turn on stream");
- return -1;
- }
-
- _captureStarted = true;
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureModuleV4L2::StopCapture()
-{
- if (_captureThread)
- _captureThread->SetNotAlive();// Make sure the capture thread stop stop using the critsect.
-
-
- CriticalSectionScoped cs(*_captureCritSect);
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, -1, "StopCapture(), was running: %d",
- _captureStarted);
-
- if (!_captureStarted)
- {
- // we were not capturing!
- return 0;
- }
-
- _captureStarted = false;
-
- // stop the capture thread
- // Delete capture update thread and event
- if (_captureThread)
- {
- ThreadWrapper* temp = _captureThread;
- _captureThread = NULL;
- temp->SetNotAlive();
- if (temp->Stop())
- {
- delete temp;
- }
- }
-
- DeAllocateVideoBuffers();
- close(_deviceFd);
- _deviceFd = -1;
-
- return 0;
-}
-
-//critical section protected by the caller
-
-bool VideoCaptureModuleV4L2::AllocateVideoBuffers()
-{
- struct v4l2_requestbuffers rbuffer;
- memset(&rbuffer, 0, sizeof(v4l2_requestbuffers));
-
- rbuffer.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
- rbuffer.memory = V4L2_MEMORY_MMAP;
- rbuffer.count = kNoOfV4L2Bufffers;
-
- if (ioctl(_deviceFd, VIDIOC_REQBUFS, &rbuffer) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Could not get buffers from device. errno = %d", errno);
- return false;
- }
-
- if (rbuffer.count > kNoOfV4L2Bufffers)
- rbuffer.count = kNoOfV4L2Bufffers;
-
- _buffersAllocatedByDevice = rbuffer.count;
-
- //Map the buffers
- _pool = new Buffer[rbuffer.count];
-
- for (unsigned int i = 0; i < rbuffer.count; i++)
- {
- struct v4l2_buffer buffer;
- memset(&buffer, 0, sizeof(v4l2_buffer));
- buffer.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
- buffer.memory = V4L2_MEMORY_MMAP;
- buffer.index = i;
-
- if (ioctl(_deviceFd, VIDIOC_QUERYBUF, &buffer) < 0)
- {
- return false;
- }
-
- _pool[i].start = mmap(NULL, buffer.length, PROT_READ | PROT_WRITE, MAP_SHARED,
- _deviceFd, buffer.m.offset);
-
- if (MAP_FAILED == _pool[i].start)
- {
- for (unsigned int j = 0; j < i; j++)
- munmap(_pool[j].start, _pool[j].length);
- return false;
- }
-
- _pool[i].length = buffer.length;
-
- if (ioctl(_deviceFd, VIDIOC_QBUF, &buffer) < 0)
- {
- return false;
- }
- }
- return true;
-}
-
-bool VideoCaptureModuleV4L2::DeAllocateVideoBuffers()
-{
- // unmap buffers
- for (int i = 0; i < _buffersAllocatedByDevice; i++)
- munmap(_pool[i].start, _pool[i].length);
-
- delete[] _pool;
-
- // turn off stream
- enum v4l2_buf_type type;
- type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
- if (ioctl(_deviceFd, VIDIOC_STREAMOFF, &type) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "VIDIOC_STREAMOFF error. errno: %d", errno);
- }
-
- return true;
-}
-
-bool VideoCaptureModuleV4L2::CaptureStarted()
-{
- return _captureStarted;
-}
-
-bool VideoCaptureModuleV4L2::CaptureThread(void* obj)
-{
- return static_cast<VideoCaptureModuleV4L2*> (obj)->CaptureProcess();
-}
-bool VideoCaptureModuleV4L2::CaptureProcess()
-{
- int retVal = 0;
- fd_set rSet;
- struct timeval timeout;
-
- _captureCritSect->Enter();
- if (!_captureThread)
- {
- // terminating
- _captureCritSect->Leave();
- return false;
- }
-
- FD_ZERO(&rSet);
- FD_SET(_deviceFd, &rSet);
- timeout.tv_sec = 1;
- timeout.tv_usec = 0;
-
- retVal = select(_deviceFd + 1, &rSet, NULL, NULL, &timeout);
- if (retVal < 0 && errno != EINTR) // continue if interrupted
- {
- // select failed
- _captureCritSect->Leave();
- return false;
- }
- else if (retVal == 0)
- {
- // select timed out
- _captureCritSect->Leave();
- return true;
- }
- else if (!FD_ISSET(_deviceFd, &rSet))
- {
- // not event on camera handle
- _captureCritSect->Leave();
- return true;
- }
-
- if (_captureStarted)
- {
- struct v4l2_buffer buf;
- memset(&buf, 0, sizeof(struct v4l2_buffer));
- buf.type = V4L2_BUF_TYPE_VIDEO_CAPTURE;
- buf.memory = V4L2_MEMORY_MMAP;
- // dequeue a buffer - repeat until dequeued properly!
- while (ioctl(_deviceFd, VIDIOC_DQBUF, &buf) < 0)
- {
- if (errno != EINTR)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "could not sync on a buffer on device %s", strerror(errno));
- _captureCritSect->Leave();
- return true;
- }
- }
- VideoCaptureCapability frameInfo;
- frameInfo.width = _currentWidth;
- frameInfo.height = _currentHeight;
- frameInfo.rawType = _captureVideoType;
-
- // convert to to I420 if needed
- IncomingFrame((unsigned char*) _pool[buf.index].start,
- buf.bytesused, frameInfo);
- // enqueue the buffer again
- if (ioctl(_deviceFd, VIDIOC_QBUF, &buf) == -1)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
- "Failed to enqueue capture buffer");
- }
- }
- _captureCritSect->Leave();
- usleep(0);
- return true;
-}
-
-WebRtc_Word32 VideoCaptureModuleV4L2::CaptureSettings(VideoCaptureCapability& settings)
-{
- settings.width = _currentWidth;
- settings.height = _currentHeight;
- settings.maxFPS = _currentFrameRate;
- settings.rawType=_captureVideoType;
-
- return 0;
-}
-} // namespace videocapturemodule
-} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Linux/video_capture_linux.h b/trunk/src/modules/video_capture/main/source/Linux/video_capture_linux.h
deleted file mode 100644
index 1cb6702..0000000
--- a/trunk/src/modules/video_capture/main/source/Linux/video_capture_linux.h
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_VIDEO_CAPTURE_LINUX_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_VIDEO_CAPTURE_LINUX_H_
-
-#include "common_types.h"
-#include "../video_capture_impl.h"
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-class ThreadWrapper;
-namespace videocapturemodule
-{
-class VideoCaptureModuleV4L2: public VideoCaptureImpl
-{
-public:
- VideoCaptureModuleV4L2(WebRtc_Word32 id);
- virtual ~VideoCaptureModuleV4L2();
- virtual WebRtc_Word32 Init(const WebRtc_UWord8* deviceUniqueId);
- virtual WebRtc_Word32 StartCapture(const VideoCaptureCapability& capability);
- virtual WebRtc_Word32 StopCapture();
- virtual bool CaptureStarted();
- virtual WebRtc_Word32 CaptureSettings(VideoCaptureCapability& settings);
-
-private:
- enum {kNoOfV4L2Bufffers=4};
-
- static bool CaptureThread(void*);
- bool CaptureProcess();
- bool AllocateVideoBuffers();
- bool DeAllocateVideoBuffers();
-
- ThreadWrapper* _captureThread;
- CriticalSectionWrapper* _captureCritSect;
-
- WebRtc_Word32 _deviceId;
- WebRtc_Word32 _deviceFd;
-
- WebRtc_Word32 _buffersAllocatedByDevice;
- WebRtc_Word32 _currentWidth;
- WebRtc_Word32 _currentHeight;
- WebRtc_Word32 _currentFrameRate;
- bool _captureStarted;
- RawVideoType _captureVideoType;
- struct Buffer
- {
- void *start;
- size_t length;
- };
- Buffer *_pool;
-};
-} // namespace videocapturemodule
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_LINUX_VIDEO_CAPTURE_LINUX_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit.h b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit.h
deleted file mode 100644
index 14ca344..0000000
--- a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit.h
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_H_
-
-#import <QTKit/QTKit.h>
-
-#include <stdio.h>
-
-#include "../../video_capture_impl.h"
-#include "video_capture_qtkit_utility.h"
-#include "../../device_info_impl.h"
-
-
-// Forward declaraion
-@class VideoCaptureMacQTKitObjC;
-@class VideoCaptureMacQTKitInfoObjC;
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-
-class VideoCaptureMacQTKit : public VideoCaptureImpl
-{
-public:
- VideoCaptureMacQTKit(const WebRtc_Word32 id);
- virtual ~VideoCaptureMacQTKit();
-
- /*
- * Create a video capture module object
- *
- * id - unique identifier of this video capture module object
- * deviceUniqueIdUTF8 - name of the device. Available names can be found
- * by using GetDeviceName
- * deviceUniqueIdUTF8Length - length of deviceUniqueIdUTF8
- */
- static void Destroy(VideoCaptureModule* module);
-
- WebRtc_Word32 Init(const WebRtc_Word32 id,
- const WebRtc_UWord8* deviceUniqueIdUTF8);
-
-
- // Start/Stop
- virtual WebRtc_Word32 StartCapture(
- const VideoCaptureCapability& capability);
- virtual WebRtc_Word32 StopCapture();
-
- // Properties of the set device
-
- virtual bool CaptureStarted();
-
- WebRtc_Word32 CaptureSettings(VideoCaptureCapability& settings);
-
-protected:
- // Help functions
- WebRtc_Word32 SetCameraOutput();
-
-private:
- VideoCaptureMacQTKitObjC* _captureDevice;
- VideoCaptureMacQTKitInfoObjC* _captureInfo;
- bool _isCapturing;
- WebRtc_Word32 _id;
- WebRtc_Word32 _captureWidth;
- WebRtc_Word32 _captureHeight;
- WebRtc_Word32 _captureFrameRate;
- WebRtc_UWord8 _currentDeviceNameUTF8[MAX_NAME_LENGTH];
- WebRtc_UWord8 _currentDeviceUniqueIdUTF8[MAX_NAME_LENGTH];
- WebRtc_UWord8 _currentDeviceProductUniqueIDUTF8[MAX_NAME_LENGTH];
- WebRtc_Word32 _frameCount;
-};
-} // namespace videocapturemodule
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit.mm b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit.mm
deleted file mode 100644
index c4a02c9..0000000
--- a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit.mm
+++ /dev/null
@@ -1,223 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_capture_qtkit.h"
-#import "video_capture_qtkit_objc.h"
-#import "video_capture_qtkit_info_objc.h"
-#include "trace.h"
-#include "critical_section_wrapper.h"
-#include "../../video_capture_config.h"
-
-namespace webrtc
-{
-
-namespace videocapturemodule
-{
-
-VideoCaptureMacQTKit::VideoCaptureMacQTKit(const WebRtc_Word32 id) :
- VideoCaptureImpl(id),
- _captureDevice(NULL),
- _captureInfo(NULL),
- _isCapturing(false),
- _id(id),
- _captureWidth(QTKIT_DEFAULT_WIDTH),
- _captureHeight(QTKIT_DEFAULT_HEIGHT),
- _captureFrameRate(QTKIT_DEFAULT_FRAME_RATE),
- _frameCount(0)
-{
-
- memset(_currentDeviceNameUTF8, 0, MAX_NAME_LENGTH);
- memset(_currentDeviceUniqueIdUTF8, 0, MAX_NAME_LENGTH);
- memset(_currentDeviceProductUniqueIDUTF8, 0, MAX_NAME_LENGTH);
-}
-
-VideoCaptureMacQTKit::~VideoCaptureMacQTKit()
-{
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- "~VideoCaptureMacQTKit() called");
- if(_captureDevice)
- {
- [_captureDevice stopCapture];
- [_captureDevice release];
- }
-
- if(_captureInfo)
- {
- [_captureInfo release];
- }
-}
-
-WebRtc_Word32 VideoCaptureMacQTKit::Init(
- const WebRtc_Word32 id, const WebRtc_UWord8* iDeviceUniqueIdUTF8)
-{
- CriticalSectionScoped cs(_apiCs);
-
-
- const WebRtc_Word32 nameLength =
- (WebRtc_Word32) strlen((char*)iDeviceUniqueIdUTF8);
- if(nameLength>kVideoCaptureUniqueNameLength)
- return -1;
-
- // Store the device name
- _deviceUniqueId = new WebRtc_UWord8[nameLength+1];
- memcpy(_deviceUniqueId, iDeviceUniqueIdUTF8,nameLength+1);
-
- _captureDevice = [[VideoCaptureMacQTKitObjC alloc] init];
- if(NULL == _captureDevice)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, id,
- "Failed to create an instance of "
- "VideoCaptureMacQTKitObjC");
- return -1;
- }
-
- if(-1 == [[_captureDevice registerOwner:this]intValue])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, id,
- "Failed to register owner for _captureDevice");
- return -1;
- }
-
- if(0 == strcmp((char*)iDeviceUniqueIdUTF8, ""))
- {
- // the user doesn't want to set a capture device at this time
- return 0;
- }
-
- _captureInfo = [[VideoCaptureMacQTKitInfoObjC alloc]init];
- if(nil == _captureInfo)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, id, "Failed to create an instance of VideoCaptureMacQTKitInfoObjC");
- return -1;
- }
-
- int captureDeviceCount = [[_captureInfo getCaptureDeviceCount]intValue];
- if(captureDeviceCount < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, id,
- "No Capture Devices Present");
- return -1;
- }
-
- const int NAME_LENGTH = 1024;
- WebRtc_UWord8 deviceNameUTF8[1024] = "";
- WebRtc_UWord8 deviceUniqueIdUTF8[1024] = "";
- WebRtc_UWord8 deviceProductUniqueIDUTF8[1024] = "";
-
- bool captureDeviceFound = false;
- for(int index = 0; index < captureDeviceCount; index++){
-
- memset(deviceNameUTF8, 0, NAME_LENGTH);
- memset(deviceUniqueIdUTF8, 0, NAME_LENGTH);
- memset(deviceProductUniqueIDUTF8, 0, NAME_LENGTH);
- if(-1 == [[_captureInfo getDeviceNamesFromIndex:index
- DefaultName:deviceNameUTF8 WithLength:NAME_LENGTH
- AndUniqueID:deviceUniqueIdUTF8 WithLength:NAME_LENGTH
- AndProductID:deviceProductUniqueIDUTF8
- WithLength:NAME_LENGTH]intValue])
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "GetDeviceName returned -1 for index %d", index);
- return -1;
- }
- if(0 == strcmp((const char*)iDeviceUniqueIdUTF8,
- (char*)deviceUniqueIdUTF8))
- {
- // we have a match
- captureDeviceFound = true;
- break;
- }
- }
-
- if(false == captureDeviceFound)
- {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "Failed to find capture device unique ID %s",
- iDeviceUniqueIdUTF8);
- return -1;
- }
-
- // at this point we know that the user has passed in a valid camera. Let's
- // set it as the current.
- if(-1 == [[_captureDevice
- setCaptureDeviceById:(char*)deviceUniqueIdUTF8]intValue])
- {
- strcpy((char*)_deviceUniqueId, (char*)deviceUniqueIdUTF8);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to set capture device %s (unique ID %s) even "
- "though it was a valid return from "
- "VideoCaptureMacQTKitInfo");
- return -1;
- }
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "successfully Init VideoCaptureMacQTKit" );
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureMacQTKit::StartCapture(
- const VideoCaptureCapability& capability)
-{
-
- _captureWidth = capability.width;
- _captureHeight = capability.height;
- _captureFrameRate = capability.maxFPS;
-
- if(-1 == [[_captureDevice setCaptureHeight:_captureHeight
- AndWidth:_captureWidth AndFrameRate:_captureFrameRate]intValue])
- {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "Could not set width=%d height=%d frameRate=%d",
- _captureWidth, _captureHeight, _captureFrameRate);
- return -1;
- }
-
- if(-1 == [[_captureDevice startCapture]intValue])
- {
- return -1;
- }
- _isCapturing = true;
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureMacQTKit::StopCapture()
-{
- [_captureDevice stopCapture];
-
- _isCapturing = false;
- return 0;
-}
-
-bool VideoCaptureMacQTKit::CaptureStarted()
-{
- return _isCapturing;
-}
-
-WebRtc_Word32 VideoCaptureMacQTKit::CaptureSettings(VideoCaptureCapability& settings)
-{
- settings.width = _captureWidth;
- settings.height = _captureHeight;
- settings.maxFPS = _captureFrameRate;
- return 0;
-}
-
-
-// ********** begin functions inherited from DeviceInfoImpl **********
-
-struct VideoCaptureCapabilityMacQTKit:public VideoCaptureCapability
-{
- VideoCaptureCapabilityMacQTKit()
- {
- }
-};
-} // namespace videocapturemodule
-} // namespace webrtc
-
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info.h b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info.h
deleted file mode 100644
index b6d05ad..0000000
--- a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info.h
+++ /dev/null
@@ -1,95 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_INFO_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_INFO_H_
-
-#include "../../video_capture_impl.h"
-#include "../../device_info_impl.h"
-#include "video_capture_qtkit_utility.h"
-
-#include "map_wrapper.h"
-
-
-@class VideoCaptureMacQTKitInfoObjC;
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-
-class VideoCaptureMacQTKitInfo: public DeviceInfoImpl
-{
-public:
-
- VideoCaptureMacQTKitInfo(const WebRtc_Word32 id);
- virtual ~VideoCaptureMacQTKitInfo();
-
- WebRtc_Word32 Init();
-
- virtual WebRtc_UWord32 NumberOfDevices();
-
- /*
- * Returns the available capture devices.
- * deviceNumber -[in] index of capture device
- * deviceNameUTF8 - friendly name of the capture device
- * deviceUniqueIdUTF8 - unique name of the capture device if it exist.
- * Otherwise same as deviceNameUTF8
- * productUniqueIdUTF8 - unique product id if it exist. Null terminated
- * otherwise.
- */
- virtual WebRtc_Word32 GetDeviceName(
- WebRtc_UWord32 deviceNumber, WebRtc_UWord8* deviceNameUTF8,
- WebRtc_UWord32 deviceNameLength, WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord32 deviceUniqueIdUTF8Length,
- WebRtc_UWord8* productUniqueIdUTF8 = 0,
- WebRtc_UWord32 productUniqueIdUTF8Length = 0);
-
- /*
- * Returns the number of capabilities for this device
- */
- virtual WebRtc_Word32 NumberOfCapabilities(
- const WebRtc_UWord8* deviceUniqueIdUTF8);
-
- /*
- * Gets the capabilities of the named device
- */
- virtual WebRtc_Word32 GetCapability(
- const WebRtc_UWord8* deviceUniqueIdUTF8,
- const WebRtc_UWord32 deviceCapabilityNumber,
- VideoCaptureCapability& capability);
-
- /*
- * Gets the capability that best matches the requested width, height and frame rate.
- * Returns the deviceCapabilityNumber on success.
- */
- virtual WebRtc_Word32 GetBestMatchedCapability(
- const WebRtc_UWord8*deviceUniqueIdUTF8,
- const VideoCaptureCapability& requested,
- VideoCaptureCapability& resulting);
-
- /*
- * Display OS /capture device specific settings dialog
- */
- virtual WebRtc_Word32 DisplayCaptureSettingsDialogBox(
- const WebRtc_UWord8* deviceUniqueIdUTF8,
- const WebRtc_UWord8* dialogTitleUTF8, void* parentWindow,
- WebRtc_UWord32 positionX, WebRtc_UWord32 positionY);
-
-protected:
- virtual WebRtc_Word32 CreateCapabilityMap(
- const WebRtc_UWord8* deviceUniqueIdUTF8);
-
- VideoCaptureMacQTKitInfoObjC* _captureInfo;
-};
-} // namespace videocapturemodule
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_INFO_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info.mm b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info.mm
deleted file mode 100644
index b13a8ab..0000000
--- a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info.mm
+++ /dev/null
@@ -1,126 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "trace.h"
-#include "../../video_capture_config.h"
-#import "video_capture_qtkit_info_objc.h"
-
-#include "video_capture.h"
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-
-VideoCaptureMacQTKitInfo::VideoCaptureMacQTKitInfo(const WebRtc_Word32 id) :
- DeviceInfoImpl(id)
-{
- _captureInfo = [[VideoCaptureMacQTKitInfoObjC alloc] init];
-}
-
-VideoCaptureMacQTKitInfo::~VideoCaptureMacQTKitInfo()
-{
- [_captureInfo release];
-
-}
-
-WebRtc_Word32 VideoCaptureMacQTKitInfo::Init()
-{
-
- return 0;
-}
-
-WebRtc_UWord32 VideoCaptureMacQTKitInfo::NumberOfDevices()
-{
-
- WebRtc_UWord32 captureDeviceCount =
- [[_captureInfo getCaptureDeviceCount]intValue];
- return captureDeviceCount;
-
-}
-
-WebRtc_Word32 VideoCaptureMacQTKitInfo::GetDeviceName(
- WebRtc_UWord32 deviceNumber, WebRtc_UWord8* deviceNameUTF8,
- WebRtc_UWord32 deviceNameLength, WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord32 deviceUniqueIdUTF8Length, WebRtc_UWord8* productUniqueIdUTF8,
- WebRtc_UWord32 productUniqueIdUTF8Length)
-{
- int errNum = [[_captureInfo getDeviceNamesFromIndex:deviceNumber
- DefaultName:deviceNameUTF8 WithLength:deviceNameLength
- AndUniqueID:deviceUniqueIdUTF8
- WithLength:deviceUniqueIdUTF8Length
- AndProductID:productUniqueIdUTF8
- WithLength:productUniqueIdUTF8Length]intValue];
- return errNum;
-}
-
-WebRtc_Word32 VideoCaptureMacQTKitInfo::NumberOfCapabilities(
- const WebRtc_UWord8* deviceUniqueIdUTF8)
-{
- // Not implemented. Mac doesn't use discrete steps in capabilities, rather
- // "analog". QTKit will do it's best to convert frames to what ever format
- // you ask for.
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "NumberOfCapabilities is not supported on the Mac platform.");
- return -1;
-}
-
-
-WebRtc_Word32 VideoCaptureMacQTKitInfo::GetCapability(
- const WebRtc_UWord8* deviceUniqueIdUTF8,
- const WebRtc_UWord32 deviceCapabilityNumber,
- VideoCaptureCapability& capability)
-{
- // Not implemented. Mac doesn't use discrete steps in capabilities, rather
- // "analog". QTKit will do it's best to convert frames to what ever format
- // you ask for.
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "NumberOfCapabilities is not supported on the Mac platform.");
- return -1;
-}
-
-
-WebRtc_Word32 VideoCaptureMacQTKitInfo::GetBestMatchedCapability(
- const WebRtc_UWord8*deviceUniqueIdUTF8,
- const VideoCaptureCapability& requested, VideoCaptureCapability& resulting)
-{
- // Not implemented. Mac doesn't use discrete steps in capabilities, rather
- // "analog". QTKit will do it's best to convert frames to what ever format
- // you ask for.
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "NumberOfCapabilities is not supported on the Mac platform.");
- return -1;
-}
-
-WebRtc_Word32 VideoCaptureMacQTKitInfo::DisplayCaptureSettingsDialogBox(
- const WebRtc_UWord8* deviceUniqueIdUTF8,
- const WebRtc_UWord8* dialogTitleUTF8, void* parentWindow,
- WebRtc_UWord32 positionX, WebRtc_UWord32 positionY)
-{
-
- return [[_captureInfo
- displayCaptureSettingsDialogBoxWithDevice:deviceUniqueIdUTF8
- AndTitle:dialogTitleUTF8
- AndParentWindow:parentWindow AtX:positionX AndY:positionY]
- intValue];
-}
-
-WebRtc_Word32 VideoCaptureMacQTKitInfo::CreateCapabilityMap(
- const WebRtc_UWord8* deviceUniqueIdUTF8)
-{
- // Not implemented. Mac doesn't use discrete steps in capabilities, rather
- // "analog". QTKit will do it's best to convert frames to what ever format
- // you ask for.
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "NumberOfCapabilities is not supported on the Mac platform.");
- return -1;
-}
-} // namespace videocapturemodule
-} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info_objc.h b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info_objc.h
deleted file mode 100644
index 1d3ac86..0000000
--- a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info_objc.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// video_capture_qtkit_info_objc.h
-//
-//
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_INFO_OBJC_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_INFO_OBJC_H_
-
-#import <QTKit/QTKit.h>
-#import <Foundation/Foundation.h>
-#include "video_capture_qtkit_utility.h"
-#include "video_capture_qtkit_info.h"
-
-@interface VideoCaptureMacQTKitInfoObjC : NSObject{
- bool _OSSupportedInfo;
- NSArray* _captureDevicesInfo;
- NSAutoreleasePool* _poolInfo;
- int _captureDeviceCountInfo;
-
-}
-
-/**************************************************************************
- *
- * The following functions are considered to be private
- *
- ***************************************************************************/
-
-- (NSNumber*)getCaptureDevices;
-- (NSNumber*)initializeVariables;
-- (void)checkOSSupported;
-
-
-/**************************************************************************
- *
- * The following functions are considered to be public and called by VideoCaptureMacQTKitInfo class
- *
- ***************************************************************************/
-
-- (NSNumber*)getCaptureDeviceCount;
-
-- (NSNumber*)getDeviceNamesFromIndex:(WebRtc_UWord32)index
- DefaultName:(WebRtc_UWord8*)deviceName
- WithLength:(WebRtc_UWord32)deviceNameLength
- AndUniqueID:(WebRtc_UWord8*)deviceUniqueID
- WithLength:(WebRtc_UWord32)deviceUniqueIDLength
- AndProductID:(WebRtc_UWord8*)deviceProductID
- WithLength:(WebRtc_UWord32)deviceProductIDLength;
-
-- (NSNumber*)displayCaptureSettingsDialogBoxWithDevice:
- (const WebRtc_UWord8*)deviceUniqueIdUTF8
- AndTitle:(const WebRtc_UWord8*)dialogTitleUTF8
- AndParentWindow:(void*) parentWindow AtX:(WebRtc_UWord32)positionX
- AndY:(WebRtc_UWord32) positionY;
-@end
-
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_INFO_OBJC_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info_objc.mm b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info_objc.mm
deleted file mode 100644
index 57ab3d0..0000000
--- a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_info_objc.mm
+++ /dev/null
@@ -1,188 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#pragma mark **** imports/includes
-
-#import "video_capture_qtkit_info_objc.h"
-
-#include "trace.h"
-
-using namespace webrtc;
-
-#pragma mark **** hidden class interface
-
-@implementation VideoCaptureMacQTKitInfoObjC
-
-// ****************** over-written OS methods ***********************
-#pragma mark **** over-written OS methods
-
-/// ***** Objective-C. Similar to C++ constructor, although invoked manually
-/// ***** Potentially returns an instance of self
--(id)init{
- self = [super init];
- if(nil != self){
- [self checkOSSupported];
- [self initializeVariables];
- }
- else
- {
- return nil;
- }
- return self;
-}
-
-/// ***** Objective-C. Similar to C++ destructor
-/// ***** Returns nothing
-- (void)dealloc {
- [super dealloc];
-}
-
-// ****************** public methods ******************
-#pragma mark **** public method implementations
-
-/// ***** Creates a message box with Cocoa framework
-/// ***** Returns 0 on success, -1 otherwise.
-- (NSNumber*)displayCaptureSettingsDialogBoxWithDevice:(const WebRtc_UWord8*)deviceUniqueIdUTF8
- AndTitle:(const WebRtc_UWord8*)dialogTitleUTF8
- AndParentWindow:(void*) parentWindow
- AtX:(WebRtc_UWord32)positionX
- AndY:(WebRtc_UWord32) positionY
-{
- NSString* strTitle = [NSString stringWithFormat:@"%s", dialogTitleUTF8];
- NSString* strButton = @"Alright";
- NSString* strMessage = [NSString stringWithFormat:@"Device %s is capturing:\nWidth:%d\n:Height:%d\n@%dfps", deviceUniqueIdUTF8];
- NSAlert* alert = [NSAlert alertWithMessageText:strTitle
- defaultButton:strButton
- alternateButton:nil otherButton:nil
- informativeTextWithFormat:strMessage];
- [alert setAlertStyle:NSInformationalAlertStyle];
- [alert runModal];
- return [NSNumber numberWithInt:0];
-}
-
-- (NSNumber*)getCaptureDeviceCount{
- [self getCaptureDevices];
- return [NSNumber numberWithInt:_captureDeviceCountInfo];
-}
-
-
-- (NSNumber*)getDeviceNamesFromIndex:(WebRtc_UWord32)index
- DefaultName:(WebRtc_UWord8*)deviceName
- WithLength:(WebRtc_UWord32)deviceNameLength
- AndUniqueID:(WebRtc_UWord8*)deviceUniqueID
- WithLength:(WebRtc_UWord32)deviceUniqueIDLength
- AndProductID:(WebRtc_UWord8*)deviceProductID
- WithLength:(WebRtc_UWord32)deviceProductIDLength
-{
- if(NO == _OSSupportedInfo)
- {
- return [NSNumber numberWithInt:0];
- }
-
- if(index > (WebRtc_UWord32)_captureDeviceCountInfo)
- {
- return [NSNumber numberWithInt:-1];
- }
-
- QTCaptureDevice* tempCaptureDevice =
- (QTCaptureDevice*)[_captureDevicesInfo objectAtIndex:index];
- if(!tempCaptureDevice)
- {
- return [NSNumber numberWithInt:-1];
- }
-
- memset(deviceName, 0, deviceNameLength);
- memset(deviceUniqueID, 0, deviceUniqueIDLength);
-
- bool successful = NO;
-
- NSString* tempString = [tempCaptureDevice localizedDisplayName];
- successful = [tempString getCString:(char*)deviceName
- maxLength:deviceNameLength encoding:NSUTF8StringEncoding];
- if(NO == successful)
- {
- memset(deviceName, 0, deviceNameLength);
- return [NSNumber numberWithInt:-1];
- }
-
- tempString = [tempCaptureDevice uniqueID];
- successful = [tempString getCString:(char*)deviceUniqueID
- maxLength:deviceUniqueIDLength encoding:NSUTF8StringEncoding];
- if(NO == successful)
- {
- memset(deviceUniqueID, 0, deviceNameLength);
- return [NSNumber numberWithInt:-1];
- }
-
- return [NSNumber numberWithInt:0];
-
-}
-
-// ****************** "private" category functions below here ******************
-#pragma mark **** "private" method implementations
-
-- (NSNumber*)initializeVariables
-{
- if(NO == _OSSupportedInfo)
- {
- return [NSNumber numberWithInt:0];
- }
-
- _poolInfo = [[NSAutoreleasePool alloc]init];
- _captureDeviceCountInfo = 0;
- [self getCaptureDevices];
-
- return [NSNumber numberWithInt:0];
-}
-
-// ***** Checks to see if the QTCaptureSession framework is available in the OS
-// ***** If it is not, isOSSupprted = NO
-// ***** Throughout the rest of the class isOSSupprted is checked and functions
-// ***** are/aren't called depending
-// ***** The user can use weak linking to the QTKit framework and run on older
-// ***** versions of the OS
-// ***** I.E. Backwards compaitibility
-// ***** Returns nothing. Sets member variable
-- (void)checkOSSupported
-{
- Class osSupportedTest = NSClassFromString(@"QTCaptureSession");
- _OSSupportedInfo = NO;
- if(nil == osSupportedTest)
- {
- }
- _OSSupportedInfo = YES;
-}
-
-/// ***** Retrieves the number of capture devices currently available
-/// ***** Stores them in an NSArray instance
-/// ***** Returns 0 on success, -1 otherwise.
-- (NSNumber*)getCaptureDevices
-{
- if(NO == _OSSupportedInfo)
- {
- return [NSNumber numberWithInt:0];
- }
-
- if(_captureDevicesInfo)
- {
- [_captureDevicesInfo release];
- }
- _captureDevicesInfo = [[NSArray alloc]
- initWithArray:[QTCaptureDevice
- inputDevicesWithMediaType:QTMediaTypeVideo]];
-
- _captureDeviceCountInfo = _captureDevicesInfo.count;
- if(_captureDeviceCountInfo < 1){
- return [NSNumber numberWithInt:0];
- }
- return [NSNumber numberWithInt:0];
-}
-
-@end
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_objc.h b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_objc.h
deleted file mode 100644
index d48dbf1..0000000
--- a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_objc.h
+++ /dev/null
@@ -1,93 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// video_capture_qtkit_objc.h
-//
-//
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_OBJC_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_OBJC_H_
-
-#import <Foundation/Foundation.h>
-#import <QTKit/QTKit.h>
-#import <AppKit/AppKit.h>
-#import <CoreData/CoreData.h>
-#import <CoreFoundation/CoreFoundation.h>
-#import <CoreVideo/CoreVideo.h>
-
-#import "video_capture_recursive_lock.h"
-
-#include "video_capture_qtkit.h"
-
-@interface VideoCaptureMacQTKitObjC : NSObject{
- // class properties
- bool _capturing;
- int _counter;
- int _frameRate;
- int _frameWidth;
- int _frameHeight;
- int _framesDelivered;
- int _framesRendered;
- bool _OSSupported;
- bool _captureInitialized;
-
- // WebRTC Custom classes
- webrtc::videocapturemodule::VideoCaptureMacQTKit* _owner;
- VideoCaptureRecursiveLock* _rLock;
-
- // QTKit variables
- QTCaptureSession* _captureSession;
- QTCaptureDeviceInput* _captureVideoDeviceInput;
- QTCaptureDecompressedVideoOutput* _captureDecompressedVideoOutput;
- NSArray* _captureDevices;
- int _captureDeviceCount;
- int _captureDeviceIndex;
- NSString* _captureDeviceName;
- char _captureDeviceNameUTF8[1024];
- char _captureDeviceNameUniqueID[1024];
- char _captureDeviceNameProductID[1024];
- NSString* _key;
- NSNumber* _val;
- NSDictionary* _videoSettings;
- NSString* _captureQuality;
-
- // other
- NSAutoreleasePool* _pool;
-
-}
-/**************************************************************************
- *
- * The following functions are considered to be private.
- *
- ***************************************************************************/
-
-- (NSNumber*)getCaptureDevices;
-- (NSNumber*)initializeVideoCapture;
-- (NSNumber*)initializeVariables;
-- (void)checkOSSupported;
-
-
-/**************************************************************************
- *
- * The following functions are considered public and to be called by the VideoCaptureMacQTKit class.
- *
- ***************************************************************************/
-
-
-- (NSNumber*)registerOwner:(webrtc::videocapturemodule::VideoCaptureMacQTKit*)owner;
-- (NSNumber*)setCaptureDeviceById:(char*)uniqueId;
-- (NSNumber*)setCaptureHeight:(int)height AndWidth:(int)width AndFrameRate:(int)frameRate;
-- (NSNumber*)startCapture;
-- (NSNumber*)stopCapture;
-
-@end
-
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_OBJC_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_objc.mm b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_objc.mm
deleted file mode 100644
index 48f734c..0000000
--- a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_objc.mm
+++ /dev/null
@@ -1,462 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#define DEFAULT_CAPTURE_DEVICE_INDEX 1
-#define DEFAULT_FRAME_RATE 30
-#define DEFAULT_FRAME_WIDTH 352
-#define DEFAULT_FRAME_HEIGHT 288
-#define ROTATE_CAPTURED_FRAME 1
-#define LOW_QUALITY 1
-
-#import "video_capture_qtkit_objc.h"
-#include "video_capture_qtkit_utility.h"
-#include "trace.h"
-
-using namespace webrtc;
-using namespace videocapturemodule;
-
-@implementation VideoCaptureMacQTKitObjC
-
-#pragma mark **** over-written OS methods
-
-/// ***** Objective-C. Similar to C++ constructor, although must be invoked
-/// manually.
-/// ***** Potentially returns an instance of self
--(id)init{
- self = [super init];
- if(nil != self)
- {
- [self checkOSSupported];
- [self initializeVariables];
- }
- else
- {
- return nil;
- }
- return self;
-}
-
-/// ***** Objective-C. Similar to C++ destructor
-/// ***** Returns nothing
-- (void)dealloc {
- if(_captureSession)
- {
- [_captureSession stopRunning];
- [_captureSession release];
- }
- [super dealloc];
-}
-
-#pragma mark **** public methods
-
-
-
-/// ***** Registers the class's owner, which is where the delivered frames are
-/// sent
-/// ***** Returns 0 on success, -1 otherwise.
-- (NSNumber*)registerOwner:(VideoCaptureMacQTKit*)owner{
- if(!owner){
- return [NSNumber numberWithInt:-1];
- }
- _owner = owner;
- return [NSNumber numberWithInt:0];
-}
-
-/// ***** Sets the QTCaptureSession's input device from a char*
-/// ***** Sets several member variables. Can signal the error system if one has
-/// occurred
-/// ***** Returns 0 on success, -1 otherwise.
-- (NSNumber*)setCaptureDeviceById:(char*)uniqueId{
- if(NO == _OSSupported)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideoCapture, 0,
- "%s:%d OS version does not support necessary APIs",
- __FUNCTION__, __LINE__);
- return [NSNumber numberWithInt:0];
- }
-
- if(!uniqueId || (0 == strcmp("", uniqueId)))
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideoCapture, 0,
- "%s:%d \"\" was passed in for capture device name",
- __FUNCTION__, __LINE__);
- memset(_captureDeviceNameUTF8, 0, 1024);
- return [NSNumber numberWithInt:0];
- }
-
- if(0 == strcmp(uniqueId, _captureDeviceNameUniqueID))
- {
- // camera already set
- WEBRTC_TRACE(kTraceInfo, kTraceVideoCapture, 0,
- "%s:%d Capture device is already set to %s", __FUNCTION__,
- __LINE__, _captureDeviceNameUTF8);
- return [NSNumber numberWithInt:0];
- }
-
- bool success = NO;
- QTCaptureDevice* tempCaptureDevice;
- for(int index = 0; index < _captureDeviceCount; index++)
- {
- tempCaptureDevice = (QTCaptureDevice*)[_captureDevices
- objectAtIndex:index];
- char tempCaptureDeviceId[1024] = "";
- [[tempCaptureDevice uniqueID]
- getCString:tempCaptureDeviceId maxLength:1024
- encoding:NSUTF8StringEncoding];
- if(0 == strcmp(uniqueId, tempCaptureDeviceId))
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideoCapture, 0,
- "%s:%d Found capture device id %s as index %d",
- __FUNCTION__, __LINE__, tempCaptureDeviceId, index);
- success = YES;
- [[tempCaptureDevice localizedDisplayName]
- getCString:_captureDeviceNameUTF8
- maxLength:1024
- encoding:NSUTF8StringEncoding];
- [[tempCaptureDevice uniqueID]
- getCString:_captureDeviceNameUniqueID
- maxLength:1024
- encoding:NSUTF8StringEncoding];
- break;
- }
-
- }
-
- if(NO == success)
- {
- // camera not found
- // nothing has been changed yet, so capture device will stay in it's
- // state
- WEBRTC_TRACE(kTraceInfo, kTraceVideoCapture, 0,
- "%s:%d Capture device id %s was not found in list of "
- "available devices.", __FUNCTION__, __LINE__, uniqueId);
- return [NSNumber numberWithInt:0];
- }
-
- NSError* error;
- success = [tempCaptureDevice open:&error];
- if(!success)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoCapture, 0,
- "%s:%d Failed to open capture device: %s",
- __FUNCTION__, __LINE__, _captureDeviceNameUTF8);
- return [NSNumber numberWithInt:-1];
- }
-
- if(_captureVideoDeviceInput)
- {
- [_captureVideoDeviceInput release];
- }
- _captureVideoDeviceInput = [[QTCaptureDeviceInput alloc]
- initWithDevice:tempCaptureDevice];
-
- success = [_captureSession addInput:_captureVideoDeviceInput error:&error];
- if(!success)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoCapture, 0,
- "%s:%d Failed to add input from %s to the capture session",
- __FUNCTION__, __LINE__, _captureDeviceNameUTF8);
- return [NSNumber numberWithInt:-1];
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceVideoCapture, 0,
- "%s:%d successfully added capture device: %s", __FUNCTION__,
- __LINE__, _captureDeviceNameUTF8);
- return [NSNumber numberWithInt:0];
-}
-
-
-/// ***** Updates the capture devices size and frequency
-/// ***** Sets member variables _frame* and _captureDecompressedVideoOutput
-/// ***** Returns 0 on success, -1 otherwise.
-- (NSNumber*)setCaptureHeight:(int)height AndWidth:(int)width
- AndFrameRate:(int)frameRate{
- if(NO == _OSSupported)
- {
- return [NSNumber numberWithInt:0];
- }
-
- _frameWidth = width;
- _frameHeight = height;
- _frameRate = frameRate;
-
- // TODO(mflodman) Check fps settings.
- // [_captureDecompressedVideoOutput
- // setMinimumVideoFrameInterval:(NSTimeInterval)1/(float)_frameRate];
- NSDictionary* captureDictionary = [NSDictionary dictionaryWithObjectsAndKeys:
- [NSNumber numberWithDouble:_frameWidth], (id)kCVPixelBufferWidthKey,
- [NSNumber numberWithDouble:_frameHeight], (id)kCVPixelBufferHeightKey,
- [NSNumber numberWithUnsignedInt:kCVPixelFormatType_32ARGB],
- (id)kCVPixelBufferPixelFormatTypeKey, nil];
- [_captureDecompressedVideoOutput performSelectorOnMainThread:@selector(setPixelBufferAttributes:) withObject:captureDictionary waitUntilDone:NO];
-// [_captureDecompressedVideoOutput setPixelBufferAttributes:captureDictionary];
-
-
- // these methods return type void so there isn't much we can do about
- // checking success
- return [NSNumber numberWithInt:0];
-}
-
-/// ***** Starts the QTCaptureSession, assuming correct state. Also ensures that
-/// an NSRunLoop is running
-/// ***** Without and NSRunLoop to process events, the OS doesn't check for a
-/// new frame.
-/// ***** Sets member variables _capturing
-/// ***** Returns 0 on success, -1 otherwise.
-- (NSNumber*)startCapture{
- if(NO == _OSSupported)
- {
- return [NSNumber numberWithInt:0];
- }
-
- if(YES == _capturing)
- {
- return [NSNumber numberWithInt:0];
- }
-
-// NSLog(@"--------------- before ---------------");
- [[NSRunLoop mainRunLoop] runUntilDate:[NSDate distantFuture]];
-// NSLog(@"--------------- after ---------------");
-
- if(NO == _captureInitialized)
- {
- // this should never be called..... it is initialized on class init
- [self initializeVideoCapture];
- }
- [_captureSession startRunning];
-
-
- _capturing = YES;
-
- return [NSNumber numberWithInt:0];
-}
-
-/// ***** Stops the QTCaptureSession, assuming correct state
-/// ***** Sets member variables _capturing
-/// ***** Returns 0 on success, -1 otherwise.
-- (NSNumber*)stopCapture{
-
- if(NO == _OSSupported)
- {
- return [NSNumber numberWithInt:0];
- }
-
- if(nil == _captureSession)
- {
- return [NSNumber numberWithInt:0];
- }
-
- if(NO == _capturing)
- {
- return [NSNumber numberWithInt:0];
- }
-
- if(YES == _capturing)
- {
- [_captureSession stopRunning];
- }
-
- _capturing = NO;
- return [NSNumber numberWithInt:0];
-}
-
-// ********** "private" functions below here **********
-#pragma mark **** "private" methods
-
-/// ***** Class member variables are initialized here
-/// ***** Returns 0 on success, -1 otherwise.
-- (NSNumber*)initializeVariables{
-
- if(NO == _OSSupported)
- {
- return [NSNumber numberWithInt:0];
- }
-
- _pool = [[NSAutoreleasePool alloc]init];
-
- memset(_captureDeviceNameUTF8, 0, 1024);
- _counter = 0;
- _framesDelivered = 0;
- _framesRendered = 0;
- _captureDeviceCount = 0;
- _capturing = NO;
- _captureInitialized = NO;
- _frameRate = DEFAULT_FRAME_RATE;
- _frameWidth = DEFAULT_FRAME_WIDTH;
- _frameHeight = DEFAULT_FRAME_HEIGHT;
- _captureDeviceName = [[NSString alloc] initWithFormat:@""];
- _rLock = [[VideoCaptureRecursiveLock alloc] init];
- _captureSession = [[QTCaptureSession alloc] init];
- _captureDecompressedVideoOutput = [[QTCaptureDecompressedVideoOutput alloc]
- init];
- [_captureDecompressedVideoOutput setDelegate:self];
-
- [self getCaptureDevices];
- [self initializeVideoCapture];
-
- return [NSNumber numberWithInt:0];
-
-}
-
-// Checks to see if the QTCaptureSession framework is available in the OS
-// If it is not, isOSSupprted = NO.
-// Throughout the rest of the class isOSSupprted is checked and functions
-// are/aren't called depending
-// The user can use weak linking to the QTKit framework and run on older
-// versions of the OS. I.E. Backwards compaitibility
-// Returns nothing. Sets member variable
-- (void)checkOSSupported{
-
- Class osSupportedTest = NSClassFromString(@"QTCaptureSession");
- _OSSupported = NO;
- if(nil == osSupportedTest)
- {
- }
- _OSSupported = YES;
-}
-
-/// ***** Retrieves the number of capture devices currently available
-/// ***** Stores them in an NSArray instance
-/// ***** Returns 0 on success, -1 otherwise.
-- (NSNumber*)getCaptureDevices{
-
- if(NO == _OSSupported)
- {
- return [NSNumber numberWithInt:0];
- }
-
- if(_captureDevices)
- {
- [_captureDevices release];
- }
- _captureDevices = [[NSArray alloc] initWithArray:
- [QTCaptureDevice inputDevicesWithMediaType:QTMediaTypeVideo]];
-
- _captureDeviceCount = _captureDevices.count;
- if(_captureDeviceCount < 1)
- {
- return [NSNumber numberWithInt:0];
- }
- return [NSNumber numberWithInt:0];
-}
-
-// Initializes a QTCaptureSession (member variable) to deliver frames via
-// callback
-// QTCapture* member variables affected
-// The image format and frequency are setup here
-// Returns 0 on success, -1 otherwise.
-- (NSNumber*)initializeVideoCapture{
-
- if(YES == _captureInitialized)
- {
- return [NSNumber numberWithInt:-1];
- }
-
- QTCaptureDevice* videoDevice =
- (QTCaptureDevice*)[_captureDevices objectAtIndex:0];
-
- bool success = NO;
- NSError* error;
-
- success = [videoDevice open:&error];
- if(!success)
- {
- return [NSNumber numberWithInt:-1];
- }
-
- [_captureDecompressedVideoOutput setPixelBufferAttributes:
- [NSDictionary dictionaryWithObjectsAndKeys:
- [NSNumber numberWithDouble:_frameWidth], (id)kCVPixelBufferWidthKey,
- [NSNumber numberWithDouble:_frameHeight], (id)kCVPixelBufferHeightKey,
- [NSNumber numberWithUnsignedInt:kCVPixelFormatType_32ARGB],
- (id)kCVPixelBufferPixelFormatTypeKey, nil]];
-
- // TODO(mflodman) Check fps settings.
- //[_captureDecompressedVideoOutput setMinimumVideoFrameInterval:
- // (NSTimeInterval)1/(float)_frameRate];
- //[_captureDecompressedVideoOutput setAutomaticallyDropsLateVideoFrames:YES];
-
- success = [_captureSession addOutput:_captureDecompressedVideoOutput
- error:&error];
-
- if(!success)
- {
- return [NSNumber numberWithInt:-1];
- }
-
- _captureInitialized = YES;
-
- return [NSNumber numberWithInt:0];
-}
-
-// This is the callback that is called when the OS has a frame to deliver to us.
-// Starts being called when [_captureSession startRunning] is called. Stopped
-// similarly.
-// Parameter videoFrame contains the image. The format, size, and frequency
-// were setup earlier.
-// Returns 0 on success, -1 otherwise.
-- (void)captureOutput:(QTCaptureOutput *)captureOutput
- didOutputVideoFrame:(CVImageBufferRef)videoFrame
- withSampleBuffer:(QTSampleBuffer *)sampleBuffer
- fromConnection:(QTCaptureConnection *)connection{
-
- if(YES == [_rLock tryLock])
- {
- [_rLock lock];
- }
- else
- {
- return;
- }
-
- if(NO == _OSSupported)
- {
- return;
- }
-
- const int LOCK_FLAGS = 0; // documentation says to pass 0
-
- // get size of the frame
- CVPixelBufferLockBaseAddress(videoFrame, LOCK_FLAGS);
- void* baseAddress = CVPixelBufferGetBaseAddress(videoFrame);
- size_t bytesPerRow = CVPixelBufferGetBytesPerRow(videoFrame);
- int frameHeight = CVPixelBufferGetHeight(videoFrame);
- CVPixelBufferUnlockBaseAddress(videoFrame, LOCK_FLAGS);
-
- if(_owner)
- {
-
- int frameSize = bytesPerRow * frameHeight; // 32 bit ARGB format
- CVBufferRetain(videoFrame);
- VideoCaptureCapability tempCaptureCapability;
- tempCaptureCapability.width = _frameWidth;
- tempCaptureCapability.height = _frameHeight;
- tempCaptureCapability.maxFPS = _frameRate;
- // TODO(wu) : Update actual type and not hard-coded value.
- tempCaptureCapability.rawType = kVideoBGRA;
-
- _owner->IncomingFrame((unsigned char*)baseAddress,
- frameSize,
- tempCaptureCapability,
- 0);
-
- CVBufferRelease(videoFrame);
- }
-
- _framesDelivered++;
- _framesRendered++;
-
- if(YES == [_rLock locked])
- {
- [_rLock unlock];
- }
-}
-
-@end
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_utility.h b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_utility.h
deleted file mode 100644
index 5ef0b96..0000000
--- a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_qtkit_utility.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * video_capture_qtkit_utility.h
- *
- */
-
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_UTILITY_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_UTILITY_H_
-
-#define MAX_NAME_LENGTH 1024
-
-#define QTKIT_MIN_WIDTH 0
-#define QTKIT_MAX_WIDTH 2560
-#define QTKIT_DEFAULT_WIDTH 352
-
-#define QTKIT_MIN_HEIGHT 0
-#define QTKIT_MAX_HEIGHT 1440
-#define QTKIT_DEFAULT_HEIGHT 288
-
-#define QTKIT_MIN_FRAME_RATE 1
-#define QTKIT_MAX_FRAME_RATE 60
-#define QTKIT_DEFAULT_FRAME_RATE 30
-
-#define RELEASE_AND_CLEAR(p) if (p) { (p) -> Release () ; (p) = NULL ; }
-
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_QTKIT_UTILITY_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_recursive_lock.h b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_recursive_lock.h
deleted file mode 100644
index f4008a4..0000000
--- a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_recursive_lock.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// video_capture_recursive_lock.h
-//
-//
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_RECURSIVE_LOCK_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_RECURSIVE_LOCK_H_
-
-#import <Foundation/Foundation.h>
-
-@interface VideoCaptureRecursiveLock : NSRecursiveLock <NSLocking> {
- BOOL _locked;
-}
-
-@property BOOL locked;
-
-- (void)lock;
-- (void)unlock;
-
-@end
-
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QTKIT_VIDEO_CAPTURE_RECURSIVE_LOCK_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_recursive_lock.mm b/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_recursive_lock.mm
deleted file mode 100644
index d9df5cb..0000000
--- a/trunk/src/modules/video_capture/main/source/Mac/QTKit/video_capture_recursive_lock.mm
+++ /dev/null
@@ -1,33 +0,0 @@
-//
-// video_capture_recursive_lock.mm
-//
-//
-
-#import "video_capture_recursive_lock.h"
-
-@implementation VideoCaptureRecursiveLock
-
-@synthesize locked = _locked;
-
-- (id)init{
- self = [super init];
- if(nil == self){
- return nil;
- }
-
- [self setLocked:NO];
- return self;
-}
-
-- (void)lock{
- [self setLocked:YES];
- [super lock];
-}
-
-- (void)unlock{
- [self setLocked:NO];
- [super unlock];
-}
-
-
-@end
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time.cc b/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time.cc
deleted file mode 100644
index f70923e..0000000
--- a/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time.cc
+++ /dev/null
@@ -1,1387 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * video_capture_quick_time.cc
- *
- */
-
-
-#include "video_capture_quick_time.h"
-
-#include "CriticalSectionWrapper.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "tick_util.h"
-#include "trace.h"
-#include <unistd.h>
-
-namespace webrtc
-{
-
-VideoCaptureMacQuickTime::VideoCaptureMacQuickTime(WebRtc_Word32 iID) :
- VideoCaptureImpl(iID), // super class constructor
- _id(iID),
- _isCapturing(false),
- _captureCapability(),
- _grabberCritsect(CriticalSectionWrapper::CreateCriticalSection()),
- _videoMacCritsect(CriticalSectionWrapper::CreateCriticalSection()),
- _terminated(true), _grabberUpdateThread(NULL),
- _grabberUpdateEvent(NULL), _captureGrabber(NULL), _captureDevice(NULL),
- _captureVideoType(kVideoUnknown), _captureIsInitialized(false),
- _gWorld(NULL), _captureChannel(0), _captureSequence(NULL),
- _sgPrepared(false), _sgStarted(false), _trueCaptureWidth(0),
- _trueCaptureHeight(0), _captureDeviceList(),
- _captureDeviceListTime(0), _captureCapabilityList()
-
-{
- _captureCapability.width = START_CODEC_WIDTH;
- _captureCapability.height = START_CODEC_HEIGHT;
- memset(_captureDeviceDisplayName, 0, sizeof(_captureDeviceDisplayName));
-}
-
-VideoCaptureMacQuickTime::~VideoCaptureMacQuickTime()
-{
-
-
- VideoCaptureTerminate();
-
- if (_videoMacCritsect)
- {
- delete _videoMacCritsect;
- }
- if (_grabberCritsect)
- {
- delete _grabberCritsect;
- }
-
-}
-
-WebRtc_Word32 VideoCaptureMacQuickTime::Init(
- const WebRtc_Word32 id, const WebRtc_UWord8* deviceUniqueIdUTF8)
-{
-
- const WebRtc_Word32 nameLength =
- (WebRtc_Word32) strlen((char*) deviceUniqueIdUTF8);
- if (nameLength > kVideoCaptureUniqueNameLength)
- return -1;
-
- // Store the device name
- _deviceUniqueId = new WebRtc_UWord8[nameLength + 1];
- memset(_deviceUniqueId, 0, nameLength + 1);
- memcpy(_deviceUniqueId, deviceUniqueIdUTF8, nameLength + 1);
-
- // Check OSX version
- OSErr err = noErr;
- long version;
-
- _videoMacCritsect->Enter();
- if (!_terminated)
- {
- _videoMacCritsect->Leave();
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Already Initialized", __FUNCTION__, __LINE__);
- return -1;
- }
-
- err = Gestalt(gestaltSystemVersion, &version);
- if (err != noErr)
- {
- _videoMacCritsect->Leave();
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Could not retrieve OS version", __FUNCTION__,
- __LINE__);
- return -1;
- }
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "%s:%d OS X version: %x,", __FUNCTION__, __LINE__, version);
- if (version < 0x00001040) // Older version than Mac OSX 10.4
- {
- _videoMacCritsect->Leave();
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d OS version not supported", __FUNCTION__, __LINE__);
- return -1;
- }
-
- err = Gestalt(gestaltQuickTime, &version);
- if (err != noErr)
- {
- _videoMacCritsect->Leave();
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Could not retrieve QuickTime version",
- __FUNCTION__, __LINE__);
- return -1;
- }
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "%s:%d QuickTime version: %x", __FUNCTION__, __LINE__,
- version);
- if (version < 0x07000000) // QT v. 7.x or newer (QT 5.0.2 0x05020000)
- {
- _videoMacCritsect->Leave();
- return -1;
- }
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- "%s:%d EnterMovies()", __FUNCTION__, __LINE__);
- EnterMovies();
-
- if (VideoCaptureSetCaptureDevice((char*) deviceUniqueIdUTF8,
- kVideoCaptureProductIdLength) == -1)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d failed to set capture device: %s", __FUNCTION__,
- __LINE__, deviceUniqueIdUTF8);
- _videoMacCritsect->Leave();
- return -1;
- }
-
- _terminated = false;
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "%s:%d successful initialization", __FUNCTION__, __LINE__);
- _videoMacCritsect->Leave();
-
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureMacQuickTime::StartCapture(
- const VideoCaptureCapability& capability)
-{
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id, "%s:%d "
- "capability.width=%d, capability.height=%d ,capability.maxFPS=%d "
- "capability.expectedCaptureDelay=%d, capability.interlaced=%d",
- __FUNCTION__, __LINE__, capability.width, capability.height,
- capability.maxFPS, capability.expectedCaptureDelay,
- capability.interlaced);
-
- _captureCapability.width = capability.width;
- _captureCapability.height = capability.height;
-
- if (VideoCaptureRun() == -1)
- {
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureMacQuickTime::StopCapture()
-{
-
- if (VideoCaptureStop() == -1)
- {
- return -1;
- }
-
- return 0;
-}
-
-bool VideoCaptureMacQuickTime::CaptureStarted()
-{
- return _isCapturing;
-}
-
-WebRtc_Word32 VideoCaptureMacQuickTime::CaptureSettings(
- VideoCaptureCapability& settings)
-{
- settings.width = _captureCapability.width;
- settings.height = _captureCapability.height;
- settings.maxFPS = 0;
- return 0;
-}
-
-int VideoCaptureMacQuickTime::VideoCaptureTerminate()
-{
- VideoCaptureStop();
-
- _videoMacCritsect->Enter();
- if (_terminated)
- {
- _videoMacCritsect->Leave();
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Already terminated", __FUNCTION__, __LINE__);
- return -1;
- }
-
- _grabberCritsect->Enter();
-
- // Stop the camera/sequence grabber
- // Resets: _captureSequence, _sgStarted
- StopQuickTimeCapture();
-
- // Remove local video settings
- // Resets: _gWorld, _captureCapability.width, _captureCapability.height
- RemoveLocalGWorld();
- DisconnectCaptureDevice();
-
- if (_grabberUpdateThread)
- _grabberUpdateThread->SetNotAlive();
-
- _grabberCritsect->Leave();
-
- if (_grabberUpdateEvent)
- _grabberUpdateEvent->Set();
-
- SLEEP(1);
- _grabberCritsect->Enter();
-
- if (_grabberUpdateThread)
- {
- _grabberUpdateThread->Stop();
- delete _grabberUpdateThread;
- _grabberUpdateThread = NULL;
- }
- if (_grabberUpdateEvent)
- {
- delete _grabberUpdateEvent;
- _grabberUpdateEvent = NULL;
- }
-
- // Close the sequence grabber
- if (_captureGrabber)
- {
- SGRelease(_captureGrabber);
- _captureGrabber = NULL;
- CloseComponent(_captureGrabber);
- _captureDevice = NULL;
- }
- _captureVideoType = kVideoUnknown;
-
- // Delete capture device list
- ListItem* item = _captureDeviceList.First();
- while (item)
- {
- delete static_cast<unsigned char*> (item->GetItem());
- _captureDeviceList.Erase(item);
- item = _captureDeviceList.First();
- }
- _captureDeviceListTime = 0;
-
- _terminated = true;
-
- _grabberCritsect->Leave();
- _videoMacCritsect->Leave();
-
- return 0;
-}
-
-int VideoCaptureMacQuickTime::UpdateCaptureSettings(int channel,
- webrtc::VideoCodec& inst,
- bool def)
-{
-
- if (channel < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Invalid channel number: %d", __FUNCTION__,
- __LINE__, channel);
- return -1;
- }
-
- // the size has changed, we need to change our setup
- _videoMacCritsect->Enter();
-
- // Stop capturing, if we are...
- _grabberCritsect->Enter();
-
- bool wasCapturing = false;
- StopQuickTimeCapture(&wasCapturing);
-
- // Create a new offline GWorld to receive captured frames
- RemoveLocalGWorld();
-
- if (CreateLocalGWorld(inst.width, inst.height) == -1)
- {
- _grabberCritsect->Leave();
- _videoMacCritsect->Leave();
- // Error already logged
- return -1;
- }
- _captureCapability.width = inst.width;
- _captureCapability.height = inst.height;
-
- // Connect the capture device to our offline GWorld
- // if we already have a capture device selected.
- if (_captureDevice)
- {
- DisconnectCaptureDevice();
- if (ConnectCaptureDevice() == -1)
- {
- // Error already logged
- _grabberCritsect->Leave();
- _videoMacCritsect->Leave();
- return -1;
- }
- }
-
- // Start capture if we did before
- if (wasCapturing)
- {
- if (StartQuickTimeCapture() == -1)
- {
- _grabberCritsect->Leave();
- _videoMacCritsect->Leave();
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Failed to start capturing", __FUNCTION__,
- __LINE__);
- return -1;
- }
- }
- _grabberCritsect->Leave();
- _videoMacCritsect->Leave();
-
- return 0;
-}
-
-// Creates an off screen graphics world used for converting
-// captured video frames if we can't get a format we want.
-// Assumed protected by critsects
-int VideoCaptureMacQuickTime::CreateLocalGWorld(int width, int height)
-{
- if (_gWorld)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
- "%s:%d GWorld already created", __FUNCTION__, __LINE__);
- return -1;
- }
- if (width == 0 || height == 0)
- {
- return -1;
- }
-
- Rect captureRect;
- captureRect.left = 0;
- captureRect.top = 0;
- captureRect.right = width;
- captureRect.bottom = height;
-
- // Create a GWorld in same size as we want to send to the codec
- if (QTNewGWorld(&(_gWorld), k2vuyPixelFormat, &captureRect, 0, NULL, 0)
- != noErr)
- {
- return -1;
- }
- _captureCapability.width = width;
- _captureCapability.height = height;
-
- if (!LockPixels(GetGWorldPixMap(_gWorld)))
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
- "%s:%d Could not lock pixmap. Continuing anyhow",
- __FUNCTION__, __LINE__);
- }
-
- CGrafPtr theOldPort;
- GDHandle theOldDevice;
- GetGWorld(&theOldPort, &theOldDevice); // Gets the result from QTGetNewGWorld
- SetGWorld(_gWorld, NULL); // Sets the new GWorld
- BackColor( blackColor); // Changes the color on the graphic port
- ForeColor( whiteColor);
- EraseRect(&captureRect);
- SetGWorld(theOldPort, theOldDevice);
-
- return 0;
-}
-
-// Assumed critsect protected
-int VideoCaptureMacQuickTime::RemoveLocalGWorld()
-{
- if (!_gWorld)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
- "%s:%d !gWorld", __FUNCTION__, __LINE__);
- return -1;
- }
-
- DisposeGWorld(_gWorld);
- _gWorld = NULL;
- _captureCapability.width = START_CODEC_WIDTH;
- _captureCapability.height = START_CODEC_HEIGHT;
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "%s:%d GWorld has been removed", __FUNCTION__, __LINE__);
- return 0;
-}
-
-// ConnectCaptureDevice
-// This function prepares the capture device
-// with the wanted settings, but the capture
-// device isn't started.
-//
-// Assumed critsect protected
-int VideoCaptureMacQuickTime::ConnectCaptureDevice()
-{
- // Prepare the capture grabber if a capture device is already set
- if (!_captureGrabber)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d No capture device is selected", __FUNCTION__,
- __LINE__);
- return -1;
- }
- if (_captureIsInitialized)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
- "%s:%d Capture device is already initialized",
- __FUNCTION__, __LINE__);
- return -1;
- }
- if (!_gWorld)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d No GWorld is created", __FUNCTION__, __LINE__);
- return -1;
- }
-
- OSErr err = noErr;
- long flags = 0;
-
- // Connect the camera to our offline GWorld
- // We won't use the GWorld if we get the format we want
- // from the camera.
- if (SGSetGWorld(_captureGrabber, _gWorld, NULL ) != noErr)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Could not connect capture device", __FUNCTION__,
- __LINE__);
- return -1;
- }
- if (SGSetDataRef(_captureGrabber, 0, 0, seqGrabDontMakeMovie) != noErr)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Could not configure capture device", __FUNCTION__,
- __LINE__);
- return -1;
- }
-
- // Set our capture callback
- if (SGSetDataProc(_captureGrabber, NewSGDataUPP(SendProcess), (long) this)
- != noErr)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Could not set capture callback. Unable to receive "
- "frames", __FUNCTION__, __LINE__);
- return -1;
- }
-
- // Create a video channel to the sequence grabber
- if (SGNewChannel(_captureGrabber, VideoMediaType, &_captureChannel)
- != noErr) // Takes time!!!
- {
- return -1;
- }
-
- // Get a list with all capture devices to choose the one we want.
- SGDeviceList deviceList = NULL;
- if (SGGetChannelDeviceList(_captureChannel, sgDeviceListIncludeInputs,
- &deviceList) != noErr)
- {
-
- }
-
- int numDevicesTypes = (*deviceList)->count;
- bool captureDeviceFound = false;
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- "%s:%d Found %d channel devices", __FUNCTION__, __LINE__,
- numDevicesTypes);
-
- // Loop through all devices to get the one we want.
- for (int i = 0; i < numDevicesTypes; i++)
- {
- SGDeviceName deviceTypeName = (*deviceList)->entry[i];
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- "%s:%d Inspecting device number: %d", __FUNCTION__,
- __LINE__, i);
- // Get the list with input devices
- if (deviceTypeName.inputs)
- {
- SGDeviceInputList inputList = deviceTypeName.inputs;
- int numInputDev = (*inputList)->count;
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- "%s:%d Device has %d inputs", __FUNCTION__, __LINE__,
- numInputDev);
- for (int inputDevIndex = 0;
- inputDevIndex < numInputDev;
- inputDevIndex++)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture,
- _id, "%s:%d Inspecting input number: %d",
- __FUNCTION__, __LINE__, inputDevIndex);
- SGDeviceInputName deviceInputName =
- (*inputList)->entry[inputDevIndex];
- char devInName[64];
- memset(devInName, 0, 64);
-
- // SGDeviceInputName::name is a Str63, defined as a Pascal string.
- // (Refer to MacTypes.h)
- CFIndex devInNameLength =
- PascalStringToCString(deviceInputName.name, devInName,
- sizeof(devInName));
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture,
- _id,
- "%s:%d Converted pascal string with length:%d "
- "to: %s", __FUNCTION__, __LINE__,
- sizeof(devInName), devInName);
- if (devInNameLength < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
- _id,
- "%s:%d Failed to convert device name from "
- "pascal string to c string", __FUNCTION__,
- __LINE__);
- return -1;
- }
-
- if (!strcmp(devInName, _captureDeviceDisplayName))
- {
- WEBRTC_TRACE(webrtc::kTraceDebug,
- webrtc::kTraceVideoCapture, _id,
- "%s:%d We have found our device: %s",
- __FUNCTION__, __LINE__,
- _captureDeviceDisplayName);
-
- if (SGSetChannelDevice(_captureChannel, deviceTypeName.name)
- != noErr)
- {
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceVideoCapture, _id,
- "%s:%d Could not set capture device type: "
- "%s",__FUNCTION__, __LINE__,
- deviceTypeName.name);
- return -1;
- }
-
- WEBRTC_TRACE(webrtc::kTraceInfo,
- webrtc::kTraceVideoCapture, _id,
- "%s:%d Capture device type is: %s",
- __FUNCTION__, __LINE__, deviceTypeName.name);
- if (SGSetChannelDeviceInput(_captureChannel, inputDevIndex)
- != noErr)
- {
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceVideoCapture, _id,
- "%s:%d Could not set SG device",
- __FUNCTION__, __LINE__);
- return -1;
- }
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture,
- _id,
- "%s:%d Capture device: %s has successfully "
- "been set", __FUNCTION__, __LINE__,
- _captureDeviceDisplayName);
- captureDeviceFound = true;
- break;
- }
- }
- if (captureDeviceFound)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture,
- _id,
- "%s:%d Capture device found, breaking from loops",
- __FUNCTION__, __LINE__);
- break;
- }
- }
- }
- err = SGDisposeDeviceList(_captureGrabber, deviceList);
-
- if (!captureDeviceFound)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Failed to find capture device: %s. Returning -1",
- __FUNCTION__, __LINE__, _captureDeviceDisplayName);
- return -1;
- }
-
- // Set the size we want from the capture device
- Rect captureSize;
- captureSize.left = 0;
- captureSize.top = 0;
- captureSize.right = _captureCapability.width;
- captureSize.bottom = _captureCapability.height;
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "%s:%d Using capture rect: l:%d t:%d r:%d b:%d", __FUNCTION__,
- __LINE__, captureSize.left, captureSize.top,
- captureSize.right, captureSize.bottom);
-
- err = SGSetChannelBounds(_captureChannel, &captureSize);
- if (err == noErr)
- {
- err = SGSetChannelUsage(_captureChannel, flags | seqGrabRecord);
- }
- if (err != noErr)
- {
- SGDisposeChannel(_captureGrabber, _captureChannel);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Error setting SG channel to device", __FUNCTION__,
- __LINE__);
- return -1;
- }
-
- // Find out what video format we'll get from the capture device.
- OSType compType;
- err = SGGetVideoCompressorType(_captureChannel, &compType);
-
- // Convert the Apple video format name to a VideoCapture name.
- if (compType == k2vuyPixelFormat)
- {
- _captureVideoType = kVideoUYVY;
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "%s:%d Device delivers UYUV formatted frames",
- __FUNCTION__, __LINE__);
- }
- else if (compType == kYUVSPixelFormat)
- {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "%s:%d Device delivers YUY2 formatted frames",
- __FUNCTION__, __LINE__);
- _captureVideoType = kVideoYUY2;
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- "%s:%d Device delivers frames in an unknown format: 0x%x. "
- "Consult QuickdrawTypes.h",
- __FUNCTION__, __LINE__, compType);
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "%s:%d Device delivers frames in an unknown format.",
- __FUNCTION__, __LINE__);
- _captureVideoType = kVideoUnknown;
- }
-
- if (SGPrepare(_captureGrabber, false, true) != noErr)
- {
- _grabberCritsect->Leave();
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Error starting sequence grabber", __FUNCTION__,
- __LINE__);
- return -1;
- }
-
- // Try to set the codec size as capture size.
- err = SGSetChannelBounds(_captureChannel, &captureSize);
-
- // Check if we really will get the size we asked for.
- ImageDescriptionHandle imageDesc = (ImageDescriptionHandle) NewHandle(0);
- err = SGGetChannelSampleDescription(_captureChannel, (Handle) imageDesc);
-
- _trueCaptureWidth = (**imageDesc).width;
- _trueCaptureHeight = (**imageDesc).height;
-
- DisposeHandle((Handle) imageDesc);
-
- _captureIsInitialized = true;
- _sgPrepared = true;
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "%s:%d Success starting sequence grabber", __FUNCTION__,
- __LINE__);
-
- return 0;
-}
-
-// Assumed critsect protected
-int VideoCaptureMacQuickTime::DisconnectCaptureDevice()
-{
- if (_sgStarted)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Capture device is still running. Returning -1",
- __FUNCTION__, __LINE__);
- return -1;
- }
- if (!_sgPrepared)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d No capture device connected", __FUNCTION__,
- __LINE__);
- return -1;
- }
-
- // Close the capture channel
- SGStop(_captureGrabber);
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- "%s:%d !!!! releasing sg stuff", __FUNCTION__, __LINE__);
- SGDisposeChannel(_captureGrabber, _captureChannel);
- SGRelease(_captureGrabber);
- CloseComponent(_captureGrabber);
-
- // Reset all values
- _captureChannel = NULL;
- _captureVideoType = kVideoUnknown;
- _trueCaptureWidth = 0;
- _trueCaptureHeight = 0;
- _captureIsInitialized = false;
- _sgPrepared = false;
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "%s:%d Sequence grabber removed", __FUNCTION__, __LINE__);
-
- return 0;
-}
-
-// StartQuickTimeCapture
-//
-// Actually starts the camera
-//
-int VideoCaptureMacQuickTime::StartQuickTimeCapture()
-{
- _grabberCritsect->Enter();
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "%s:%d Attempting to start sequence grabber", __FUNCTION__,
- __LINE__);
-
- if (_sgStarted)
- {
- _grabberCritsect->Leave();
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Sequence grabber already started", __FUNCTION__,
- __LINE__);
- return 0;
- }
- if (!_sgPrepared)
- {
- _grabberCritsect->Leave();
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Sequence grabber not prepared properly",
- __FUNCTION__, __LINE__);
- return 0;
- }
-
- if (SGStartRecord(_captureGrabber) != noErr)
- {
- _grabberCritsect->Leave();
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Error starting sequence grabber", __FUNCTION__,
- __LINE__);
- return -1;
- }
-
- Rect captureRect = { 0, 0, 0, 0 };
- MatrixRecord scaleMatrix;
- ImageDescriptionHandle imageDesc = (ImageDescriptionHandle) NewHandle(0);
-
- // Get the sample description for the channel, which is the same as for the
- // capture device
- if (SGGetChannelSampleDescription(_captureChannel, (Handle) imageDesc)
- != noErr)
- {
- _grabberCritsect->Leave();
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Error accessing device properties", __FUNCTION__,
- __LINE__);
- return -1;
- }
-
- // Create a scale matrix to scale the captured image
- // Needed if we don't get the size wanted from the camera
- captureRect.right = (**imageDesc).width;
- captureRect.bottom = (**imageDesc).height;
-
- Rect codecRect;
- codecRect.left = 0;
- codecRect.top = 0;
- codecRect.right = _captureCapability.width;
- codecRect.bottom = _captureCapability.height;
- RectMatrix(&scaleMatrix, &captureRect, &codecRect);
-
- // Start grabbing images from the capture device to _gWorld
- if (DecompressSequenceBegin(&_captureSequence, imageDesc, _gWorld, NULL,
- NULL, &scaleMatrix, srcCopy, (RgnHandle) NULL,
- NULL, codecNormalQuality, bestSpeedCodec)
- != noErr)
- {
- _grabberCritsect->Leave();
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "%s:%d Error starting decompress sequence", __FUNCTION__,
- __LINE__);
- return -1;
- }
- DisposeHandle((Handle) imageDesc);
- _sgStarted = true;
- _grabberCritsect->Leave();
- return 0;
-}
-
-int VideoCaptureMacQuickTime::StopQuickTimeCapture(bool* wasCapturing)
-{
- _grabberCritsect->Enter();
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "%s:%d wasCapturing=%d", __FUNCTION__, __LINE__, wasCapturing);
-
- if (!_sgStarted)
- {
- if (wasCapturing)
- *wasCapturing = false;
-
- _grabberCritsect->Leave();
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
- "%s:%d Sequence grabber was never started", __FUNCTION__,
- __LINE__);
- return 0;
- }
-
- if (wasCapturing)
- *wasCapturing = true;
-
- OSErr error = noErr;
- error = SGStop(_captureGrabber);
- CDSequenceEnd(_captureSequence);
- _captureSequence = NULL;
- _sgStarted = false;
-
- _grabberCritsect->Leave();
- if (error != noErr)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
- "%s:%d Could not stop sequence grabber", __FUNCTION__,
- __LINE__);
- return -1;
- }
-
- return 0;
-}
-
-//-------------------------------------------------
-//
-// Thread/function to keep capture device working
-//
-//-------------------------------------------------
-
-
-//
-// GrabberUpdateThread / GrabberUpdateProcess
-//
-// Called at a certain time interval to tell
-// the capture device / SequenceGrabber to
-// actually work.
-bool VideoCaptureMacQuickTime::GrabberUpdateThread(void* obj)
-{
- return static_cast<VideoCaptureMacQuickTime*> (obj)->GrabberUpdateProcess();
-}
-
-bool VideoCaptureMacQuickTime::GrabberUpdateProcess()
-{
- _grabberUpdateEvent->Wait(30);
-
- if (_isCapturing == false)
- return false;
-
- _grabberCritsect->Enter();
- if (_captureGrabber)
- {
- if (SGIdle(_captureGrabber) != noErr)
- {
- }
- }
- _grabberCritsect->Leave();
- return true;
-}
-
-//
-// VideoCaptureStop
-//
-// Stops the capture device
-//
-int VideoCaptureMacQuickTime::VideoCaptureStop()
-{
- if (_grabberUpdateThread)
- {
- _grabberUpdateThread->Stop();
- }
-
- _videoMacCritsect->Enter();
- _grabberCritsect->Enter();
- int retVal = StopQuickTimeCapture();
- _grabberCritsect->Leave();
- _videoMacCritsect->Leave();
- if (retVal == -1)
- {
- return -1;
- }
-
- _isCapturing = false;
- return 0;
-}
-
-//
-// VideoCaptureRun
-//
-// Starts the capture device and creates
-// the update thread.
-//
-int VideoCaptureMacQuickTime::VideoCaptureRun()
-{
- _videoMacCritsect->Enter();
- _grabberCritsect->Enter();
-
- int res = StartQuickTimeCapture();
-
- // Create the thread for updating sequence grabber if not created earlier
- if (!_grabberUpdateThread)
- {
- _grabberUpdateEvent = EventWrapper::Create();
- _grabberUpdateThread = ThreadWrapper::CreateThread(
- VideoCaptureMacQuickTime::GrabberUpdateThread, this, kHighPriority);
- unsigned int id;
- _grabberUpdateThread->Start(id);
- }
- else
- {
- unsigned int id;
- _grabberUpdateThread->Start(id);
- }
-
- _grabberCritsect->Leave();
- _videoMacCritsect->Leave();
-
- _isCapturing = true;
- return res;
-}
-
-// ----------------------------------------------------------------------
-//
-// SendProcess
-// sequence grabber data procedure
-//
-// This function is called by the capture device as soon as a new
-// frame is available.
-//
-//
-// SendFrame
-//
-// The non-static function used by the capture device callback
-//
-// Input:
-// sgChannel: the capture device channel generating the callback
-// data: the video frame
-// length: the data length in bytes
-// grabTime: time stamp generated by the capture device / sequece grabber
-//
-// ----------------------------------------------------------------------
-
-OSErr VideoCaptureMacQuickTime::SendProcess(SGChannel sgChannel, Ptr p,
- long len, long* /*offset*/,
- long /*chRefCon*/, TimeValue time,
- short /*writeType*/, long refCon)
-{
- VideoCaptureMacQuickTime* videoEngine =
- reinterpret_cast<VideoCaptureMacQuickTime*> (refCon);
- return videoEngine->SendFrame(sgChannel, (char*) p, len, time);
-}
-
-int VideoCaptureMacQuickTime::SendFrame(SGChannel /*sgChannel*/, char* data,
- long length, TimeValue /*grabTime*/)
-{
- if (!_sgPrepared)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
- "%s:%d Sequence Grabber is not initialized", __FUNCTION__,
- __LINE__);
- return 0;
- }
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- "%s:%d Frame has been delivered\n", __FUNCTION__, __LINE__);
-
- CodecFlags ignore;
- _grabberCritsect->Enter();
- if (_gWorld)
- {
- // Will be set to true if we don't recognize the size and/or video
- // format.
- bool convertFrame = false;
- WebRtc_Word32 width = 352;
- WebRtc_Word32 height = 288;
- WebRtc_Word32 frameSize = 0;
-
- VideoCaptureCapability captureCapability;
- captureCapability.width = width;
- captureCapability.height = height;
- captureCapability.maxFPS = 30;
-
- switch (_captureVideoType)
- {
- case kVideoUYVY:
- captureCapability.rawType = kVideoUYVY;
- break;
- case kVideoYUY2:
- captureCapability.rawType = kVideoYUY2;
- break;
- case kVideoI420:
- captureCapability.rawType = kVideoI420;
- break;
- default:
- captureCapability.rawType = kVideoI420;
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture,
- _id, "%s:%d raw = I420 by default\n",
- __FUNCTION__, __LINE__);
- break;
- }
-
- // Convert the camera video type to something VideoEngine can work with
- // Check if we need to downsample the incomming frame.
- switch (_captureVideoType)
- {
- case kVideoUYVY:
- case kVideoYUY2:
- frameSize = (width * height * 16) >> 3; // 16 is for YUY2 format
- if (width == _captureCapability.width || height
- == _captureCapability.height)
- {
- // Ok format and size, send the frame to super class
- IncomingFrame((WebRtc_UWord8*) data,
- (WebRtc_Word32) frameSize, captureCapability,
- TickTime::MillisecondTimestamp());
-
- }
- else if (width == _trueCaptureWidth && height
- == _trueCaptureHeight)
- {
- // We need to scale the picture to correct size...
- // This happens for cameras not supporting all sizes.
- // E.g. older built-in iSight doesn't support QCIF.
-
- // Convert the incoming frame into our GWorld.
- int res =
- DecompressSequenceFrameS(_captureSequence, data,
- length, 0, &ignore, NULL);
- if (res != noErr && res != -8976) // 8796 == black frame
- {
- WEBRTC_TRACE(webrtc::kTraceWarning,
- webrtc::kTraceVideoCapture, _id,
- "%s:%d Captured black frame. Not "
- "processing it", __FUNCTION__, __LINE__);
- _grabberCritsect->Leave();
- return 0;
- }
-
- // Copy the frame from the PixMap to our video buffer
- PixMapHandle pixMap = GetGWorldPixMap(_gWorld);
-
- // Lock the image data in the GWorld.
- LockPixels(pixMap);
-
- // Get a pointer to the pixel data.
- Ptr capturedFrame = GetPixBaseAddr(pixMap);
-
- // Send the converted frame out to super class
- IncomingFrame((WebRtc_UWord8*) data,
- (WebRtc_Word32) frameSize, captureCapability,
- TickTime::MillisecondTimestamp());
-
- // Unlock the image data to get ready for the next frame.
- UnlockPixels(pixMap);
- }
- else
- {
- // Not a size we recognize, use the Mac internal scaling...
- convertFrame = true;
- WEBRTC_TRACE(webrtc::kTraceDebug,
- webrtc::kTraceVideoCapture, _id,
- "%s:%d Not correct incoming stream size for "
- "the format and configured size",
- __FUNCTION__, __LINE__);
- }
- break;
- default:
-
- // Not a video format we recognize, use the Mac internal scaling
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture,
- _id, "%s:%d Unknown video frame format (default)",
- __FUNCTION__, __LINE__);
- convertFrame = true;
- break;
- }
-
- if (convertFrame)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- "%s:%d Unrecognized frame format. Converting frame",
- __FUNCTION__, __LINE__);
-
- // We don't recognise the input format. Convert to UYVY, I420 is not
- // supported on osx. Decompress the grabbed frame into the GWorld,
- // i.e from webcam format to ARGB (RGB24), and extract the frame.
- int res = DecompressSequenceFrameS(_captureSequence, data, length,
- 0, &ignore, NULL);
- if (res != noErr && res != -8976) // 8796 means a black frame
- {
- _grabberCritsect->Leave();
- return 0;
- }
-
- // Copy the frame from the PixMap to our video buffer
- PixMapHandle rgbPixMap = GetGWorldPixMap(_gWorld);
- LockPixels(rgbPixMap);
- Ptr capturedFrame = GetPixBaseAddr(rgbPixMap);
-
- // Get the picture size
- int width = (*rgbPixMap)->bounds.right;
- int height = (*rgbPixMap)->bounds.bottom;
-
- // 16 is for YUY2 format.
- WebRtc_Word32 frameSize = (width * height * 16) >> 3;
-
- // Ok format and size, send the frame to super class
- IncomingFrame((WebRtc_UWord8*) data, (WebRtc_Word32) frameSize,
- captureCapability, TickTime::MillisecondTimestamp());
-
- UnlockPixels(rgbPixMap);
- }
-
- // Tell the capture device it's ok to update.
- SGUpdate(_captureGrabber, NULL);
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
- "%s:%d No GWorld created, but frames are being delivered",
- __FUNCTION__, __LINE__);
- }
-
- _grabberCritsect->Leave();
- return 0;
-}
-
-int VideoCaptureMacQuickTime::VideoCaptureInitThreadContext()
-{
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- "%s:%d ", __FUNCTION__, __LINE__);
- _videoMacCritsect->Enter();
- EnterMoviesOnThread( kQTEnterMoviesFlagDontSetComponentsThreadMode);
- _videoMacCritsect->Leave();
- return 0;
-}
-
-//
-//
-// Functions for handling capture devices
-//
-//
-
-VideoCaptureMacQuickTime::VideoCaptureMacName::VideoCaptureMacName() :
- _size(0)
-{
- memset(_name, 0, kVideoCaptureMacNameMaxSize);
-}
-
-int VideoCaptureMacQuickTime::VideoCaptureSetCaptureDevice(
- const char* deviceName, int size)
-{
-
-
- _videoMacCritsect->Enter();
- bool wasCapturing = false;
-
- _grabberCritsect->Enter();
- if (_captureGrabber)
- {
- // Stop grabbing, disconnect and close the old capture device
- StopQuickTimeCapture(&wasCapturing);
- DisconnectCaptureDevice();
- CloseComponent(_captureGrabber);
- _captureDevice = NULL;
- _captureGrabber = NULL;
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Old capture device removed", __FUNCTION__,
- __LINE__);
- }
-
- if (deviceName == NULL || size == 0)
- {
- _grabberCritsect->Leave();
- _videoMacCritsect->Leave();
- return 0;
- }
-
- if (size < 0)
- {
- _grabberCritsect->Leave();
- _videoMacCritsect->Leave();
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d 'size' is not valid", __FUNCTION__, __LINE__);
- return 0;
- }
-
- ComponentDescription compCaptureType;
-
- // Define the component we want to open
- compCaptureType.componentType = SeqGrabComponentType;
- compCaptureType.componentSubType = 0;
- compCaptureType.componentManufacturer = 0;
- compCaptureType.componentFlags = 0;
- compCaptureType.componentFlagsMask = 0;
-
- long numSequenceGrabbers = CountComponents(&compCaptureType);
-
- // loop through the available grabbers and open the first possible
- for (int i = 0; i < numSequenceGrabbers; i++)
- {
- _captureDevice = FindNextComponent(0, &compCaptureType);
- _captureGrabber = OpenComponent(_captureDevice);
- if (_captureGrabber != NULL)
- {
- // We've found a sequencegrabber that we could open
- if (SGInitialize(_captureGrabber) != noErr)
- {
- _grabberCritsect->Leave();
- _videoMacCritsect->Leave();
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
- _id,
- "%s:%d Could not initialize sequence grabber",
- __FUNCTION__, __LINE__);
- return -1;
- }
- break;
- }
- if (i == numSequenceGrabbers - 1)
- {
- // Couldn't open a sequence grabber
- _grabberCritsect->Leave();
- _videoMacCritsect->Leave();
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Could not open a sequence grabber",
- __FUNCTION__, __LINE__);
- return -1;
- }
- }
-
- if (!_gWorld)
- {
- // We don't have a GWorld. Create one to enable early preview
- // without calling SetSendCodec
- if (CreateLocalGWorld(_captureCapability.width,
- _captureCapability.height) == -1)
- {
- // Error already logged
- _grabberCritsect->Leave();
- _videoMacCritsect->Leave();
- return -1;
- }
- }
- // Connect the camera with our GWorld
- int cpySize = size;
- if ((unsigned int) size > sizeof(_captureDeviceDisplayName))
- {
- cpySize = sizeof(_captureDeviceDisplayName);
- }
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- "%s:%d Copying %d chars from deviceName to "
- "_captureDeviceDisplayName (size=%d)\n",
- __FUNCTION__, __LINE__, cpySize, size);
- memcpy(_captureDeviceDisplayName, deviceName, cpySize);
- if (ConnectCaptureDevice() == -1)
- {
- // Error already logged
- _grabberCritsect->Leave();
- _videoMacCritsect->Leave();
- return -1;
- }
-
- if (StartQuickTimeCapture() == -1)
- {
- // Error already logged
- _grabberCritsect->Leave();
- _videoMacCritsect->Leave();
- return -1;
- }
- _grabberCritsect->Leave();
- _videoMacCritsect->Leave();
- return 0;
-}
-
-bool VideoCaptureMacQuickTime::IsCaptureDeviceSelected()
-{
- _grabberCritsect->Leave();
- return (_captureIsInitialized) ? true : false;
- _grabberCritsect->Leave();
-}
-
-/**
- Convert a Pascal string to a C string.
-
- \param[in] pascalString
- Pascal string to convert. Pascal strings contain the number of
- characters in the first byte and are not null-terminated.
-
- \param[out] cString
- The C string buffer into which to copy the converted string.
-
- \param[in] bufferSize
- The size of the C string buffer in bytes.
-
- \return The number of characters in the string on success and -1 on failure.
- */
-CFIndex VideoCaptureMacQuickTime::PascalStringToCString(
- const unsigned char* pascalString, char* cString, CFIndex bufferSize)
-{
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, 0,
- "%s:%d Converting pascal string to c string", __FUNCTION__,
- __LINE__);
- if (pascalString == NULL)
- {
- return -1;
- }
-
- if (cString == NULL)
- {
- return -1;
- }
-
- if (bufferSize == 0)
- {
- return -1;
- }
-
- CFIndex cStringLength = 0;
- CFIndex maxStringLength = bufferSize - 1;
-
- CFStringRef cfString = CFStringCreateWithPascalString(
- NULL, pascalString, kCFStringEncodingMacRoman);
- if (cfString == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, 0,
- "%s:%d Error in CFStringCreateWithPascalString()",
- __FUNCTION__, __LINE__);
- CFRelease(cfString);
- return -1;
- }
-
- CFIndex cfLength = CFStringGetLength(cfString);
- cStringLength = cfLength;
- if (cfLength > maxStringLength)
- {
- cStringLength = maxStringLength;
- }
-
- Boolean success = CFStringGetCString(cfString, cString, bufferSize,
- kCFStringEncodingMacRoman);
-
- // Ensure the problem isn't insufficient buffer length.
- // This is fine; we will return a partial string.
- if (success == false && cfLength <= maxStringLength)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, 0,
- "%s:%d Error in CFStringGetCString()", __FUNCTION__,
- __LINE__);
- CFRelease(cfString);
- return -1;
- }
-
- CFRelease(cfString);
- return cStringLength;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time.h b/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time.h
deleted file mode 100644
index 84e0667..0000000
--- a/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time.h
+++ /dev/null
@@ -1,133 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * video_capture_quick_time.h
- *
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QUICKTIME_VIDEO_CAPTURE_QUICK_TIME_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QUICKTIME_VIDEO_CAPTURE_QUICK_TIME_H_
-
-#include <QuickTime/QuickTime.h>
-
-
-#include "../../device_info_impl.h"
-#include "../../video_capture_impl.h"
-#include "list_wrapper.h"
-
-
-#define START_CODEC_WIDTH 352
-#define START_CODEC_HEIGHT 288
-#define SLEEP(x) usleep(x * 1000);
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-class EventWrapper;
-class ThreadWrapper;
-
-class VideoCaptureMacQuickTime : public VideoCaptureImpl
-{
-
-public:
- VideoCaptureMacQuickTime(const WebRtc_Word32 id);
- virtual ~VideoCaptureMacQuickTime();
-
- static void Destroy(VideoCaptureModule* module);
-
- WebRtc_Word32 Init(const WebRtc_Word32 id,
- const WebRtc_UWord8* deviceUniqueIdUTF8);
- virtual WebRtc_Word32 StartCapture(
- const VideoCaptureCapability& capability);
- virtual WebRtc_Word32 StopCapture();
- virtual bool CaptureStarted();
- virtual WebRtc_Word32 CaptureSettings(VideoCaptureCapability& settings);
-
- // TODO: remove?
- int VideoCaptureInitThreadContext();
- int VideoCaptureTerminate();
- int VideoCaptureSetCaptureDevice(const char* deviceName, int size);
- int UpdateCaptureSettings(int channel, webrtc::VideoCodec& inst, bool def);
- int VideoCaptureRun();
- int VideoCaptureStop();
-
-protected:
-
-private: // functions
-
- struct VideoCaptureMacName
- {
- VideoCaptureMacName();
-
- enum { kVideoCaptureMacNameMaxSize = 64};
- char _name[kVideoCaptureMacNameMaxSize];
- CFIndex _size;
- };
-
- // Timeout value [ms] if we want to create a new device list or not
- enum { kVideoCaptureDeviceListTimeout = 5000};
- // Temporary constant allowing this size from builtin iSight webcams.
- enum { kYuy2_1280_1024_length = 2621440};
-
-private:
-
- // Capture device callback
- static OSErr SendProcess(SGChannel sgChannel, Ptr p, long len, long *offset,
- long chRefCon, TimeValue time, short writeType,
- long refCon);
- int SendFrame(SGChannel sgChannel, char* data, long length, TimeValue time);
-
- // Capture device functions
- int CreateLocalGWorld(int width, int height);
- int RemoveLocalGWorld();
- int ConnectCaptureDevice();
- int DisconnectCaptureDevice();
- virtual bool IsCaptureDeviceSelected();
-
- // Process to make sure the capture device won't stop
- static bool GrabberUpdateThread(void*);
- bool GrabberUpdateProcess();
-
- // Starts and stops the capture
- int StartQuickTimeCapture();
- int StopQuickTimeCapture(bool* wasCapturing = NULL);
-
- static CFIndex PascalStringToCString(const unsigned char* pascalString,
- char* cString,
- CFIndex bufferSize);
-
-private: // variables
- WebRtc_Word32 _id;
- bool _isCapturing;
- VideoCaptureCapability _captureCapability;
- CriticalSectionWrapper* _grabberCritsect;
- CriticalSectionWrapper* _videoMacCritsect;
- bool _terminated;
- webrtc::ThreadWrapper* _grabberUpdateThread;
- webrtc::EventWrapper* _grabberUpdateEvent;
- SeqGrabComponent _captureGrabber;
- Component _captureDevice;
- char _captureDeviceDisplayName[64];
- RawVideoType _captureVideoType;
- bool _captureIsInitialized;
- GWorldPtr _gWorld;
- SGChannel _captureChannel;
- ImageSequence _captureSequence;
- bool _sgPrepared;
- bool _sgStarted;
- int _trueCaptureWidth;
- int _trueCaptureHeight;
- ListWrapper _captureDeviceList;
- unsigned long _captureDeviceListTime;
- ListWrapper _captureCapabilityList;
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QUICKTIME_VIDEO_CAPTURE_QUICK_TIME_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time_info.cc b/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time_info.cc
deleted file mode 100644
index 4f3cc1a..0000000
--- a/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time_info.cc
+++ /dev/null
@@ -1,391 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * video_capture_quick_time_info.cc
- *
- */
-
-#include "../../video_capture_config.h"
-#include "video_capture_quick_time_info.h"
-
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "trace.h"
-#include "video_capture.h"
-
-namespace webrtc
-{
-
-VideoCaptureMacQuickTimeInfo::VideoCaptureMacQuickTimeInfo(
- const WebRtc_Word32 iID) :
- DeviceInfoImpl(iID), _id(iID),
- _grabberCritsect(CriticalSectionWrapper::CreateCriticalSection())
-{
-}
-
-VideoCaptureMacQuickTimeInfo::~VideoCaptureMacQuickTimeInfo()
-{
-}
-
-WebRtc_Word32 VideoCaptureMacQuickTimeInfo::Init()
-{
-
- return 0;
-}
-
-WebRtc_UWord32 VideoCaptureMacQuickTimeInfo::NumberOfDevices()
-{
- int numOfDevices = 0;
-
- // don't care about these variables... dummy vars to call GetCaptureDevices
- const int kNameLength = 1024;
- WebRtc_UWord8 deviceNameUTF8[kNameLength] = "";
- WebRtc_UWord8 deviceUniqueIdUTF8[kNameLength] = "";
- WebRtc_UWord8 productUniqueIdUTF8[kNameLength] = "";
-
- if (GetCaptureDevices(0, deviceNameUTF8, kNameLength, deviceUniqueIdUTF8,
- kNameLength, productUniqueIdUTF8, kNameLength,
- numOfDevices) != 0)
- {
- return 0;
- }
-
- return numOfDevices;
-}
-
-WebRtc_Word32 VideoCaptureMacQuickTimeInfo::GetDeviceName(
- WebRtc_UWord32 deviceNumber, WebRtc_UWord8* deviceNameUTF8,
- WebRtc_UWord32 deviceNameUTF8Length, WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord32 deviceUniqueIdUTF8Length, WebRtc_UWord8* productUniqueIdUTF8,
- WebRtc_UWord32 productUniqueIdUTF8Length)
-{
-
- int numOfDevices = 0; // not needed for this function
- return GetCaptureDevices(deviceNumber, deviceNameUTF8,
- deviceNameUTF8Length, deviceUniqueIdUTF8,
- deviceUniqueIdUTF8Length, productUniqueIdUTF8,
- productUniqueIdUTF8Length, numOfDevices);
-}
-
-WebRtc_Word32 VideoCaptureMacQuickTimeInfo::NumberOfCapabilities(
- const WebRtc_UWord8* deviceUniqueIdUTF8)
-{
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "NumberOfCapabilities is not supported on the Mac platform.");
- return -1;
-}
-
-WebRtc_Word32 VideoCaptureMacQuickTimeInfo::GetCapability(
- const WebRtc_UWord8* deviceUniqueIdUTF8,
- const WebRtc_UWord32 deviceCapabilityNumber,
- VideoCaptureCapability& capability)
-{
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "NumberOfCapabilities is not supported on the Mac platform.");
- return -1;
-}
-
-WebRtc_Word32 VideoCaptureMacQuickTimeInfo::GetBestMatchedCapability(
- const WebRtc_UWord8*deviceUniqueIdUTF8,
- const VideoCaptureCapability& requested, VideoCaptureCapability& resulting)
-{
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "NumberOfCapabilities is not supported on the Mac platform.");
- return -1;
-}
-
-WebRtc_Word32 VideoCaptureMacQuickTimeInfo::DisplayCaptureSettingsDialogBox(
- const WebRtc_UWord8* deviceUniqueIdUTF8,
- const WebRtc_UWord8* dialogTitleUTF8, void* parentWindow,
- WebRtc_UWord32 positionX, WebRtc_UWord32 positionY)
-{
- return -1;
-}
-
-WebRtc_Word32 VideoCaptureMacQuickTimeInfo::CreateCapabilityMap(
- const WebRtc_UWord8* deviceUniqueIdUTF8)
-{
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "NumberOfCapabilities is not supported on the Mac platform.");
- return -1;
-}
-
-int VideoCaptureMacQuickTimeInfo::GetCaptureDevices(
- WebRtc_UWord32 deviceNumber, WebRtc_UWord8* deviceNameUTF8,
- WebRtc_UWord32 deviceNameUTF8Length, WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord32 deviceUniqueIdUTF8Length, WebRtc_UWord8* productUniqueIdUTF8,
- WebRtc_UWord32 productUniqueIdUTF8Length, int& numberOfDevices)
-{
-
-
- numberOfDevices = 0;
- memset(deviceNameUTF8, 0, deviceNameUTF8Length);
- memset(deviceUniqueIdUTF8, 0, deviceUniqueIdUTF8Length);
- memset(productUniqueIdUTF8, 0, productUniqueIdUTF8Length);
-
- if (deviceNumber < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Invalid deviceNumber", __FUNCTION__, __LINE__);
- return -1;
- }
-
- Component captureDevice = NULL;
- SeqGrabComponent captureGrabber = NULL;
- SGChannel captureChannel = NULL;
- bool closeChannel = false;
-
- ComponentDescription compCaptureType;
-
- compCaptureType.componentType = SeqGrabComponentType;
- compCaptureType.componentSubType = 0;
- compCaptureType.componentManufacturer = 0;
- compCaptureType.componentFlags = 0;
- compCaptureType.componentFlagsMask = 0;
-
- // Get the number of sequence grabbers
- long numSequenceGrabbers = CountComponents(&compCaptureType);
-
- if (deviceNumber > numSequenceGrabbers)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Invalid deviceNumber", __FUNCTION__, __LINE__);
- return -1;
- }
-
- if (numSequenceGrabbers <= 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d No sequence grabbers available", __FUNCTION__,
- __LINE__);
- return -1;
- }
-
- // Open a sequence grabber
- for (int seqGrabberIndex = 0;
- seqGrabberIndex < numSequenceGrabbers;
- seqGrabberIndex++)
- {
- captureDevice = FindNextComponent(0, &compCaptureType);
- captureGrabber = OpenComponent(captureDevice);
- if (captureGrabber != NULL)
- {
- // We've found a sequencegrabber
- if (SGInitialize(captureGrabber) != noErr)
- {
- CloseComponent(captureGrabber);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
- _id, "%s:%d Could not init the sequence grabber",
- __FUNCTION__, __LINE__);
- return -1;
- }
- break;
- }
- if (seqGrabberIndex == numSequenceGrabbers - 1)
- {
- // Couldn't open a sequence grabber
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Could not open a sequence grabber",
- __FUNCTION__, __LINE__);
- return -1;
- }
- }
-
- // Create a temporary channel to get the names of the capture devices.
- // Takes time, make this in a nother way...
- if (SGNewChannel(captureGrabber, VideoMediaType, &captureChannel) != noErr)
- {
- // Could not create a video channel...
- SGRelease(captureGrabber);
- CloseComponent(captureGrabber);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Could not create a sequence grabber video channel",
- __FUNCTION__, __LINE__);
- return -1;
- }
- closeChannel = true;
-
- // Find the type of capture devices, e.g. USB-devices, Firewire, DV, ...
- SGDeviceList deviceList = NULL;
- if (SGGetChannelDeviceList(captureChannel, sgDeviceListIncludeInputs,
- &deviceList) != noErr)
- {
- if (closeChannel)
- SGDisposeChannel(captureGrabber, captureChannel);
- if (captureGrabber)
- {
- SGRelease(captureGrabber);
- CloseComponent(captureGrabber);
- }
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "%s:%d Could not create a device list", __FUNCTION__,
- __LINE__);
- return -1;
- }
-
- // Loop through all device types and all devices for each type
- // and store in a list.
- int numDevices = (*deviceList)->count;
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- "%s:%d Found %d devices", __FUNCTION__, __LINE__, numDevices);
-
- for (int i = 0; i < numDevices; i++)
- {
-
- SGDeviceName sgDeviceName = (*deviceList)->entry[i];
- // Get the list with input devices for this type of device
- if (sgDeviceName.inputs)
- {
- SGDeviceInputList inputList = sgDeviceName.inputs;
- int numInputDev = (*inputList)->count;
-
- for (int inputDevIndex = 0;
- inputDevIndex < numInputDev;
- inputDevIndex++)
- {
- // Get the name for this capture device
- SGDeviceInputName deviceInputName =
- (*inputList)->entry[inputDevIndex];
-
- VideoCaptureMacName* deviceName = new VideoCaptureMacName();
-
- deviceName->_size = PascalStringToCString(
- deviceInputName.name, deviceName->_name,
- sizeof(deviceName->_name));
-
- if (deviceName->_size > 0)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug,webrtc::kTraceVideoCapture,
- _id,
- "%s:%d Capture device %d: %s was successfully "
- "set", __FUNCTION__, __LINE__, numberOfDevices,
- deviceName->_name);
-
- if (numberOfDevices == deviceNumber)
- {
- strcpy((char*) deviceNameUTF8, deviceName->_name);
- strcpy((char*) deviceUniqueIdUTF8, deviceName->_name);
- }
- numberOfDevices++;
- }
- else
- {
- delete deviceName;
-
- if (deviceName->_size < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceVideoCapture, _id,
- "%s:%d Error in PascalStringToCString",
- __FUNCTION__, __LINE__);
- return -1;
- }
- }
- }
- }
- }
-
- // clean up
- SGDisposeDeviceList(captureGrabber, deviceList);
- if (closeChannel)
- {
- SGDisposeChannel(captureGrabber, captureChannel);
- }
- if (captureGrabber)
- {
- SGRelease(captureGrabber);
- CloseComponent(captureGrabber);
- }
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- "%s:%d End function successfully", __FUNCTION__, __LINE__);
- return 0;
-}
-
-/**
- Convert a Pascal string to a C string.
-
- \param[in] pascalString
- Pascal string to convert. Pascal strings contain the number of
- characters in the first byte and are not null-terminated.
-
- \param[out] cString
- The C string buffer into which to copy the converted string.
-
- \param[in] bufferSize
- The size of the C string buffer in bytes.
-
- \return The number of characters in the string on success and -1 on failure.
- */
-CFIndex VideoCaptureMacQuickTimeInfo::PascalStringToCString(
- const unsigned char* pascalString, char* cString, CFIndex bufferSize)
-{
- if (pascalString == NULL)
- {
- return -1;
- }
-
- if (cString == NULL)
- {
- return -1;
- }
-
- if (bufferSize == 0)
- {
- return -1;
- }
-
- CFIndex cStringLength = 0;
- CFIndex maxStringLength = bufferSize - 1;
-
- CFStringRef cfString = CFStringCreateWithPascalString(
- NULL, pascalString, kCFStringEncodingMacRoman);
- if (cfString == NULL)
- {
- CFRelease(cfString);
- return -1;
- }
-
- CFIndex cfLength = CFStringGetLength(cfString);
- cStringLength = cfLength;
- if (cfLength > maxStringLength)
- {
- cStringLength = maxStringLength;
- }
-
- Boolean success = CFStringGetCString(cfString, cString, bufferSize,
- kCFStringEncodingMacRoman);
-
- // Ensure the problem isn't insufficient buffer length.
- // This is fine; we will return a partial string.
- if (success == false && cfLength <= maxStringLength)
- {
- CFRelease(cfString);
- return -1;
- }
-
- CFRelease(cfString);
- return cStringLength;
-}
-
-//
-//
-// Functions for handling capture devices
-//
-//
-
-VideoCaptureMacQuickTimeInfo::VideoCaptureMacName::VideoCaptureMacName() :
- _size(0)
-{
- memset(_name, 0, kVideoCaptureMacNameMaxSize);
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time_info.h b/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time_info.h
deleted file mode 100644
index 5114d11..0000000
--- a/trunk/src/modules/video_capture/main/source/Mac/QuickTime/video_capture_quick_time_info.h
+++ /dev/null
@@ -1,166 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * video_capture_quick_time_info.h
- *
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QUICKTIME_VIDEO_CAPTURE_QUICK_TIME_INFO_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QUICKTIME_VIDEO_CAPTURE_QUICK_TIME_INFO_H_
-
-#include <QuickTime/QuickTime.h>
-
-#include "../../video_capture_impl.h"
-#include "../../device_info_impl.h"
-#include "list_wrapper.h"
-#include "map_wrapper.h"
-
-class VideoRenderCallback;
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-class EventWrapper;
-class ThreadWrapper;
-class Trace;
-
-class VideoCaptureMacQuickTimeInfo: public DeviceInfoImpl
-{
-public:
-
- static DeviceInfo* Create(const WebRtc_Word32 id);
- static void Destroy(DeviceInfo* deviceInfo);
-
- VideoCaptureMacQuickTimeInfo(const WebRtc_Word32 id);
- virtual ~VideoCaptureMacQuickTimeInfo();
-
- WebRtc_Word32 Init();
-
- virtual WebRtc_UWord32 NumberOfDevices();
-
- /*
- * Returns the available capture devices.
- * deviceNumber -[in] index of capture device
- * deviceNameUTF8 - friendly name of the capture device
- * deviceUniqueIdUTF8 - unique name of the capture device if it exist.
- * Otherwise same as deviceNameUTF8
- * productUniqueIdUTF8 - unique product id if it exist. Null terminated
- * otherwise.
- */
- virtual WebRtc_Word32 GetDeviceName(
- WebRtc_UWord32 deviceNumber, WebRtc_UWord8* deviceNameUTF8,
- WebRtc_UWord32 deviceNameLength, WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord32 deviceUniqueIdUTF8Length,
- WebRtc_UWord8* productUniqueIdUTF8 = 0,
- WebRtc_UWord32 productUniqueIdUTF8Length = 0);
-
-
- // ************** The remaining public functions are not supported on Mac
-
- /*
- * Returns the number of capabilities for this device
- */
- virtual WebRtc_Word32 NumberOfCapabilities(const WebRtc_UWord8* deviceUniqueIdUTF8);
-
- /*
- * Gets the capabilities of the named device
- */
- virtual WebRtc_Word32 GetCapability(
- const WebRtc_UWord8* deviceUniqueIdUTF8,
- const WebRtc_UWord32 deviceCapabilityNumber,
- VideoCaptureCapability& capability);
-
- /*
- * Gets the capability that best matches the requested width, height and frame rate.
- * Returns the deviceCapabilityNumber on success.
- */
- virtual WebRtc_Word32 GetBestMatchedCapability(
- const WebRtc_UWord8*deviceUniqueIdUTF8,
- const VideoCaptureCapability& requested,
- VideoCaptureCapability& resulting);
-
- /*
- * Display OS /capture device specific settings dialog
- */
- virtual WebRtc_Word32 DisplayCaptureSettingsDialogBox(
- const WebRtc_UWord8* deviceUniqueIdUTF8,
- const WebRtc_UWord8* dialogTitleUTF8, void* parentWindow,
- WebRtc_UWord32 positionX, WebRtc_UWord32 positionY);
-
-protected:
- virtual WebRtc_Word32 CreateCapabilityMap(
- const WebRtc_UWord8* deviceUniqueIdUTF8);
-
-private:
-
- struct VideoCaptureMacName
- {
- VideoCaptureMacName();
-
- enum
- {
- kVideoCaptureMacNameMaxSize = 64
- };
- char _name[kVideoCaptureMacNameMaxSize];
- CFIndex _size;
- };
-
- enum
- {
- kVideoCaptureMacDeviceListTimeout = 5000
- }; // Timeout value [ms] if we want to create a new device list or not
- enum
- {
- kYuy2_1280_1024_length = 2621440
- }; // Temporary constant allowing this size from built-in iSight webcams.
-
-private:
- // private methods
-
- int GetCaptureDevices(WebRtc_UWord32 deviceNumber,
- WebRtc_UWord8* deviceNameUTF8,
- WebRtc_UWord32 deviceNameUTF8Length,
- WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord32 deviceUniqueIdUTF8Length,
- WebRtc_UWord8* productUniqueIdUTF8,
- WebRtc_UWord32 productUniqueIdUTF8Length,
- int& numberOfDevices);
-
- static CFIndex PascalStringToCString(const unsigned char* pascalString,
- char* cString, CFIndex bufferSize);
-
-private:
- // member vars
- WebRtc_Word32 _id;
- bool _terminated;
- CriticalSectionWrapper* _grabberCritsect;
- webrtc::Trace* _trace;
- webrtc::ThreadWrapper* _grabberUpdateThread;
- webrtc::EventWrapper* _grabberUpdateEvent;
- SeqGrabComponent _captureGrabber;
- Component _captureDevice;
- char _captureDeviceDisplayName[64];
- bool _captureIsInitialized;
- GWorldPtr _gWorld;
- SGChannel _captureChannel;
- ImageSequence _captureSequence;
- bool _sgPrepared;
- bool _sgStarted;
- int _codecWidth;
- int _codecHeight;
- int _trueCaptureWidth;
- int _trueCaptureHeight;
- ListWrapper _captureDeviceList;
- WebRtc_Word64 _captureDeviceListTime;
- ListWrapper _captureCapabilityList;
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_MAC_QUICKTIME_VIDEO_CAPTURE_QUICK_TIME_INFO_H_
diff --git a/trunk/src/modules/video_capture/main/source/Mac/video_capture_mac.h b/trunk/src/modules/video_capture/main/source/Mac/video_capture_mac.h
deleted file mode 100644
index e69de29..0000000
--- a/trunk/src/modules/video_capture/main/source/Mac/video_capture_mac.h
+++ /dev/null
diff --git a/trunk/src/modules/video_capture/main/source/Mac/video_capture_mac.mm b/trunk/src/modules/video_capture/main/source/Mac/video_capture_mac.mm
deleted file mode 100644
index d329936..0000000
--- a/trunk/src/modules/video_capture/main/source/Mac/video_capture_mac.mm
+++ /dev/null
@@ -1,275 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * video_capture_mac.cc
- *
- */
-
-
-// super class stuff
-#include "../video_capture_impl.h"
-#include "../device_info_impl.h"
-#include "../video_capture_config.h"
-#include "ref_count.h"
-
-#include "trace.h"
-
-#include <QuickTime/QuickTime.h>
-
-// 10.4 support must be decided runtime. We will just decide which framework to
-// use at compile time "work" classes. One for QTKit, one for QuickTime
-#if __MAC_OS_X_VERSION_MIN_REQUIRED == __MAC_10_4 // QuickTime version
-#include "QuickTime/video_capture_quick_time.h"
-#include "QuickTime/video_capture_quick_time_info.h"
-#else
-#include "QTKit/video_capture_qtkit.h"
-#include "QTKit/video_capture_qtkit_info.h"
-#endif
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-
-// static
-bool CheckOSVersion()
-{
- // Check OSX version
- OSErr err = noErr;
-
- SInt32 version;
-
- err = Gestalt(gestaltSystemVersion, &version);
- if (err != noErr)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, 0,
- "Could not get OS version");
- return false;
- }
-
- if (version < 0x00001040) // Older version than Mac OSX 10.4
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, 0,
- "OS version too old: 0x%x", version);
- return false;
- }
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, 0,
- "OS version compatible: 0x%x", version);
-
- return true;
-}
-
-// static
-bool CheckQTVersion()
-{
- // Check OSX version
- OSErr err = noErr;
-
- SInt32 version;
-
- err = Gestalt(gestaltQuickTime, &version);
- if (err != noErr)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, 0,
- "Could not get QuickTime version");
- return false;
- }
-
- if (version < 0x07000000) // QT v. 7.x or newer (QT 5.0.2 0x05020000)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, 0,
- "QuickTime version too old: 0x%x", version);
- return false;
- }
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, 0,
- "QuickTime version compatible: 0x%x", version);
- return true;
-}
-
-/**************************************************************************
- *
- * Create/Destroy a VideoCaptureModule
- *
- ***************************************************************************/
-
-/*
- * Returns version of the module and its components
- *
- * version - buffer to which the version will be written
- * remainingBufferInBytes - remaining number of WebRtc_Word8 in the version
- * buffer
- * position - position of the next empty WebRtc_Word8 in the
- * version buffer
- */
-
-VideoCaptureModule* VideoCaptureImpl::Create(
- const WebRtc_Word32 id, const WebRtc_UWord8* deviceUniqueIdUTF8)
-{
-
- if (webrtc::videocapturemodule::CheckOSVersion() == false)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, id,
- "OS version is too old. Could not create video capture "
- "module. Returning NULL");
- return NULL;
- }
-
-#if __MAC_OS_X_VERSION_MIN_REQUIRED == __MAC_10_4 // QuickTime version
- if (webrtc::videocapturemodule::CheckQTVersion() == false)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, id,
- "QuickTime version is too old. Could not create video "
- "capture module. Returning NULL");
- return NULL;
- }
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, id,
- "%s line %d. QTKit is not supported on this machine. Using "
- "QuickTime framework to capture video",
- __FILE__, __LINE__);
-
- RefCountImpl<videocapturemodule::VideoCaptureMacQuickTime>*
- newCaptureModule =
- new RefCountImpl<videocapturemodule::VideoCaptureMacQuickTime>(id);
-
- if (!newCaptureModule)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, id,
- "could not Create for unique device %s, !newCaptureModule",
- deviceUniqueIdUTF8);
- return NULL;
- }
-
- if (newCaptureModule->Init(id, deviceUniqueIdUTF8) != 0)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, id,
- "could not Create for unique device %s, "
- "newCaptureModule->Init()!=0",
- deviceUniqueIdUTF8);
- delete newCaptureModule;
- return NULL;
- }
-
- // Successfully created VideoCaptureMacQuicktime. Return it
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, id,
- "Module created for unique device %s. Will use QuickTime "
- "framework to capture",
- deviceUniqueIdUTF8);
- return newCaptureModule;
-
-#else // QTKit version
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, id,
- "Using QTKit framework to capture video", id);
-
- RefCountImpl<videocapturemodule::VideoCaptureMacQTKit>* newCaptureModule =
- new RefCountImpl<videocapturemodule::VideoCaptureMacQTKit>(id);
-
- if(!newCaptureModule)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, id,
- "could not Create for unique device %s, !newCaptureModule",
- deviceUniqueIdUTF8);
- return NULL;
- }
- if(newCaptureModule->Init(id, deviceUniqueIdUTF8) != 0)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, id,
- "could not Create for unique device %s, "
- "newCaptureModule->Init()!=0", deviceUniqueIdUTF8);
- delete newCaptureModule;
- return NULL;
- }
-
- // Successfully created VideoCaptureMacQuicktime. Return it
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, id,
- "Module created for unique device %s, will use QTKit "
- "framework",deviceUniqueIdUTF8);
- return newCaptureModule;
-#endif
-}
-
-/**************************************************************************
- *
- * Create/Destroy a DeviceInfo
- *
- ***************************************************************************/
-
-VideoCaptureModule::DeviceInfo*
-VideoCaptureImpl::CreateDeviceInfo(const WebRtc_Word32 id)
-{
-
-
- if (webrtc::videocapturemodule::CheckOSVersion() == false)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, id,
- "OS version is too old. Could not create video capture "
- "module. Returning NULL");
- return NULL;
- }
-
-#if __MAC_OS_X_VERSION_MIN_REQUIRED == __MAC_10_4 // QuickTime version
- if (webrtc::videocapturemodule::CheckQTVersion() == false)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, id,
- "QuickTime version is too old. Could not create video "
- "capture module. Returning NULL");
- return NULL;
- }
-
- webrtc::videocapturemodule::VideoCaptureMacQuickTimeInfo* newCaptureInfoModule =
- new webrtc::videocapturemodule::VideoCaptureMacQuickTimeInfo(id);
-
- if (!newCaptureInfoModule || newCaptureInfoModule->Init() != 0)
- {
- Destroy(newCaptureInfoModule);
- newCaptureInfoModule = NULL;
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, id,
- "Failed to Init newCaptureInfoModule created with id %d "
- "and device \"\" ", id);
- return NULL;
- }
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, id,
- "VideoCaptureModule created for id", id);
- return newCaptureInfoModule;
-
-#else // QTKit version
- webrtc::videocapturemodule::VideoCaptureMacQTKitInfo* newCaptureInfoModule =
- new webrtc::videocapturemodule::VideoCaptureMacQTKitInfo(id);
-
- if(!newCaptureInfoModule || newCaptureInfoModule->Init() != 0)
- {
- //Destroy(newCaptureInfoModule);
- delete newCaptureInfoModule;
- newCaptureInfoModule = NULL;
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, id,
- "Failed to Init newCaptureInfoModule created with id %d "
- "and device \"\" ", id);
- return NULL;
- }
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, id,
- "VideoCaptureModule created for id", id);
- return newCaptureInfoModule;
-
-#endif
-
-}
-
-/**************************************************************************
- *
- * End Create/Destroy VideoCaptureModule
- *
- ***************************************************************************/
-} // namespace videocapturemodule
-} // namespace webrtc
-
diff --git a/trunk/src/modules/video_capture/main/source/Windows/capture_delay_values_windows.h b/trunk/src/modules/video_capture/main/source/Windows/capture_delay_values_windows.h
deleted file mode 100644
index a4d9da6..0000000
--- a/trunk/src/modules/video_capture/main/source/Windows/capture_delay_values_windows.h
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_CAPTURE_DELAY_VALUES_WINDOWS_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_CAPTURE_DELAY_VALUES_WINDOWS_H_
-
-#include "../video_capture_delay.h"
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-const WebRtc_Word32 NoWindowsCaptureDelays=1;
-const DelayValues WindowsCaptureDelays[NoWindowsCaptureDelays]=
-{
- "Microsoft LifeCam Cinema","usb#vid_045e&pid_075d",{{640,480,125},{640,360,117},{424,240,111},{352,288,111},{320,240,116},{176,144,101},{160,120,109},{1280,720,166},{960,544,126},{800,448,120},{800,600,127}},
-};
-
-} // namespace videocapturemodule
-} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_CAPTURE_DELAY_VALUES_WINDOWS_H_
diff --git a/trunk/src/modules/video_capture/main/source/Windows/device_info_windows.cc b/trunk/src/modules/video_capture/main/source/Windows/device_info_windows.cc
deleted file mode 100644
index 5adcc96..0000000
--- a/trunk/src/modules/video_capture/main/source/Windows/device_info_windows.cc
+++ /dev/null
@@ -1,792 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "device_info_windows.h"
-
-#include "../video_capture_config.h"
-#include "help_functions_windows.h"
-#include "capture_delay_values_windows.h"
-#include "ref_count.h"
-#include "trace.h"
-
-#include <Streams.h>
-#include <Dvdmedia.h>
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-VideoCaptureModule::DeviceInfo* VideoCaptureImpl::CreateDeviceInfo(
- const WebRtc_Word32 id)
-{
- videocapturemodule::DeviceInfoWindows* dsInfo =
- new videocapturemodule::DeviceInfoWindows(id);
-
- if (!dsInfo || dsInfo->Init() != 0)
- {
- delete dsInfo;
- dsInfo = NULL;
- }
- return dsInfo;
-}
-
-DeviceInfoWindows::DeviceInfoWindows(const WebRtc_Word32 id)
- : DeviceInfoImpl(id), _dsDevEnum(NULL), _dsMonikerDevEnum(NULL),
- _CoUninitializeIsRequired(true)
-{
- // 1) Initialize the COM library (make Windows load the DLLs).
- //
- // CoInitializeEx must be called at least once, and is usually called only once,
- // for each thread that uses the COM library. Multiple calls to CoInitializeEx
- // by the same thread are allowed as long as they pass the same concurrency flag,
- // but subsequent valid calls return S_FALSE.
- // To close the COM library gracefully on a thread, each successful call to
- // CoInitializeEx, including any call that returns S_FALSE, must be balanced
- // by a corresponding call to CoUninitialize.
- //
-
- /*Apartment-threading, while allowing for multiple threads of execution,
- serializes all incoming calls by requiring that calls to methods of objects created by this thread always run on the same thread
- the apartment/thread that created them. In addition, calls can arrive only at message-queue boundaries (i.e., only during a
- PeekMessage, SendMessage, DispatchMessage, etc.). Because of this serialization, it is not typically necessary to write concurrency control into
- the code for the object, other than to avoid calls to PeekMessage and SendMessage during processing that must not be interrupted by other method
- invocations or calls to other objects in the same apartment/thread.*/
-
- ///CoInitializeEx(NULL, COINIT_APARTMENTTHREADED ); //| COINIT_SPEED_OVER_MEMORY
- HRESULT hr = CoInitializeEx(NULL, COINIT_MULTITHREADED); // Use COINIT_MULTITHREADED since Voice Engine uses COINIT_MULTITHREADED
- if (FAILED(hr))
- {
- // Avoid calling CoUninitialize() since CoInitializeEx() failed.
- _CoUninitializeIsRequired = FALSE;
-
- if (hr == RPC_E_CHANGED_MODE)
- {
- // Calling thread has already initialized COM to be used in a single-threaded
- // apartment (STA). We are then prevented from using STA.
- // Details: hr = 0x80010106 <=> "Cannot change thread mode after it is set".
- //
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
- "VideoCaptureWindowsDSInfo::VideoCaptureWindowsDSInfo "
- "CoInitializeEx(NULL, COINIT_APARTMENTTHREADED) => "
- "RPC_E_CHANGED_MODE, error 0x%x",
- hr);
- }
- }
-}
-
-DeviceInfoWindows::~DeviceInfoWindows()
-{
- RELEASE_AND_CLEAR(_dsMonikerDevEnum);
- RELEASE_AND_CLEAR(_dsDevEnum);
- if (_CoUninitializeIsRequired)
- {
- CoUninitialize();
- }
-}
-
-WebRtc_Word32 DeviceInfoWindows::Init()
-{
- HRESULT hr = CoCreateInstance(CLSID_SystemDeviceEnum, NULL, CLSCTX_INPROC,
- IID_ICreateDevEnum, (void **) &_dsDevEnum);
- if (hr != NOERROR)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to create CLSID_SystemDeviceEnum, error 0x%x", hr);
- return -1;
- }
- return 0;
-}
-WebRtc_UWord32 DeviceInfoWindows::NumberOfDevices()
-{
- ReadLockScoped cs(_apiLock);
- return GetDeviceInfo(0, 0, 0, 0, 0, 0, 0);
-
-}
-WebRtc_Word32 DeviceInfoWindows::GetDeviceName(
- WebRtc_UWord32 deviceNumber,
- WebRtc_UWord8* deviceNameUTF8,
- WebRtc_UWord32 deviceNameLength,
- WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord32 deviceUniqueIdUTF8Length,
- WebRtc_UWord8* productUniqueIdUTF8,
- WebRtc_UWord32 productUniqueIdUTF8Length)
-{
- ReadLockScoped cs(_apiLock);
- const WebRtc_Word32 result = GetDeviceInfo(deviceNumber, deviceNameUTF8,
- deviceNameLength,
- deviceUniqueIdUTF8,
- deviceUniqueIdUTF8Length,
- productUniqueIdUTF8,
- productUniqueIdUTF8Length);
- return result > (WebRtc_Word32) deviceNumber ? 0 : -1;
-}
-
-WebRtc_Word32 DeviceInfoWindows::GetDeviceInfo(
- WebRtc_UWord32 deviceNumber,
- WebRtc_UWord8* deviceNameUTF8,
- WebRtc_UWord32 deviceNameLength,
- WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord32 deviceUniqueIdUTF8Length,
- WebRtc_UWord8* productUniqueIdUTF8,
- WebRtc_UWord32 productUniqueIdUTF8Length)
-
-{
-
- // enumerate all video capture devices
- RELEASE_AND_CLEAR(_dsMonikerDevEnum);
- HRESULT hr =
- _dsDevEnum->CreateClassEnumerator(CLSID_VideoInputDeviceCategory,
- &_dsMonikerDevEnum, 0);
- if (hr != NOERROR)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to enumerate CLSID_SystemDeviceEnum, error 0x%x."
- " No webcam exist?", hr);
- return 0;
- }
-
- _dsMonikerDevEnum->Reset();
- ULONG cFetched;
- IMoniker *pM;
- int index = 0;
- while (S_OK == _dsMonikerDevEnum->Next(1, &pM, &cFetched))
- {
- IPropertyBag *pBag;
- hr = pM->BindToStorage(0, 0, IID_IPropertyBag, (void **) &pBag);
- if (S_OK == hr)
- {
- // Find the description or friendly name.
- VARIANT varName;
- VariantInit(&varName);
- hr = pBag->Read(L"Description", &varName, 0);
- if (FAILED(hr))
- {
- hr = pBag->Read(L"FriendlyName", &varName, 0);
- }
- if (SUCCEEDED(hr))
- {
- // ignore all VFW drivers
- if ((wcsstr(varName.bstrVal, (L"(VFW)")) == NULL) &&
- (_wcsnicmp(varName.bstrVal, (L"Google Camera Adapter"),21)
- != 0))
- {
- // Found a valid device
- if (index == deviceNumber) // This is the device we are interested in.
- {
- int convResult = 0;
- if (deviceNameLength > 0)
- {
- convResult = WideCharToMultiByte(CP_UTF8, 0,
- varName.bstrVal, -1,
- (char*) deviceNameUTF8,
- deviceNameLength, NULL,
- NULL);
- if (convResult == 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceVideoCapture, _id,
- "Failed to convert device name to UTF8. %d",
- GetLastError());
- return -1;
- }
- }
- if (deviceUniqueIdUTF8Length > 0)
- {
- hr = pBag->Read(L"DevicePath", &varName, 0);
- if (FAILED(hr))
- {
- strncpy_s((char *) deviceUniqueIdUTF8,
- deviceUniqueIdUTF8Length,
- (char *) deviceNameUTF8, convResult);
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceVideoCapture, _id,
- "Failed to get deviceUniqueIdUTF8 using deviceNameUTF8");
- }
- else
- {
- convResult = WideCharToMultiByte(
- CP_UTF8,
- 0,
- varName.bstrVal,
- -1,
- (char*) deviceUniqueIdUTF8,
- deviceUniqueIdUTF8Length,
- NULL, NULL);
- if (convResult == 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceVideoCapture, _id,
- "Failed to convert device name to UTF8. %d",
- GetLastError());
- return -1;
- }
- if (productUniqueIdUTF8
- && productUniqueIdUTF8Length > 0)
- {
- GetProductId(deviceUniqueIdUTF8,
- productUniqueIdUTF8,
- productUniqueIdUTF8Length);
- }
- }
- }
-
- }
- ++index; // increase the number of valid devices
- }
- }
- VariantClear(&varName);
- pBag->Release();
- pM->Release();
- }
-
- }
- if (deviceNameLength)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id, "%s %s",
- __FUNCTION__, deviceNameUTF8);
- }
- return index;
-}
-
-IBaseFilter * DeviceInfoWindows::GetDeviceFilter(
- const WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord8* productUniqueIdUTF8,
- WebRtc_UWord32 productUniqueIdUTF8Length)
-{
-
- const WebRtc_Word32 deviceUniqueIdUTF8Length =
- (WebRtc_Word32) strlen((char*) deviceUniqueIdUTF8); // UTF8 is also NULL terminated
- if (deviceUniqueIdUTF8Length > kVideoCaptureUniqueNameLength)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Device name too long");
- return NULL;
- }
-
- // enumerate all video capture devices
- RELEASE_AND_CLEAR(_dsMonikerDevEnum);
- HRESULT hr = _dsDevEnum->CreateClassEnumerator(CLSID_VideoInputDeviceCategory,
- &_dsMonikerDevEnum, 0);
- if (hr != NOERROR)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to enumerate CLSID_SystemDeviceEnum, error 0x%x."
- " No webcam exist?", hr);
- return 0;
- }
- _dsMonikerDevEnum->Reset();
- ULONG cFetched;
- IMoniker *pM;
-
- IBaseFilter *captureFilter = NULL;
- bool deviceFound = false;
- while (S_OK == _dsMonikerDevEnum->Next(1, &pM, &cFetched) && !deviceFound)
- {
- IPropertyBag *pBag;
- hr = pM->BindToStorage(0, 0, IID_IPropertyBag, (void **) &pBag);
- if (S_OK == hr)
- {
- // Find the description or friendly name.
- VARIANT varName;
- VariantInit(&varName);
- if (deviceUniqueIdUTF8Length > 0)
- {
- hr = pBag->Read(L"DevicePath", &varName, 0);
- if (FAILED(hr))
- {
- hr = pBag->Read(L"Description", &varName, 0);
- if (FAILED(hr))
- {
- hr = pBag->Read(L"FriendlyName", &varName, 0);
- }
- }
- if (SUCCEEDED(hr))
- {
- char tempDevicePathUTF8[256];
- tempDevicePathUTF8[0] = 0;
- const int compresult =
- WideCharToMultiByte(CP_UTF8, 0, varName.bstrVal, -1,
- tempDevicePathUTF8,
- sizeof(tempDevicePathUTF8), NULL,
- NULL);
- if (strncmp(tempDevicePathUTF8,
- (const char*) deviceUniqueIdUTF8,
- deviceUniqueIdUTF8Length) == 0)
- {
- // We have found the requested device
- deviceFound = true;
- hr = pM->BindToObject(0, 0, IID_IBaseFilter,
- (void**) &captureFilter);
- if FAILED(hr)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
- _id, "Failed to bind to the selected capture device %d",hr);
- }
-
- if (productUniqueIdUTF8
- && productUniqueIdUTF8Length > 0) // Get the device name
- {
-
- GetProductId(deviceUniqueIdUTF8,
- productUniqueIdUTF8,
- productUniqueIdUTF8Length);
- }
-
- }
- }
- }
- VariantClear(&varName);
- pBag->Release();
- pM->Release();
- }
- }
- return captureFilter;
-}
-
-WebRtc_Word32 DeviceInfoWindows::GetWindowsCapability(
- const WebRtc_Word32 capabilityIndex,
- VideoCaptureCapabilityWindows& windowsCapability)
-
-{
- ReadLockScoped cs(_apiLock);
- // Make sure the number is valid
- if (capabilityIndex >= _captureCapabilities.Size() || capabilityIndex < 0)
- return -1;
-
- MapItem* item = _captureCapabilities.Find(capabilityIndex);
- if (!item)
- return -1;
-
- VideoCaptureCapabilityWindows* capPointer =
- static_cast<VideoCaptureCapabilityWindows*> (item->GetItem());
- windowsCapability = *capPointer;
- return 0;
-}
-
-WebRtc_Word32 DeviceInfoWindows::CreateCapabilityMap(
- const WebRtc_UWord8* deviceUniqueIdUTF8)
-
-{
- // Reset old capability list
- MapItem* item = NULL;
- while (item = _captureCapabilities.Last())
- {
- delete item->GetItem();
- _captureCapabilities.Erase(item);
- }
-
- const WebRtc_Word32 deviceUniqueIdUTF8Length =
- (WebRtc_Word32) strlen((char*) deviceUniqueIdUTF8);
- if (deviceUniqueIdUTF8Length > kVideoCaptureUniqueNameLength)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Device name too long");
- return -1;
- }
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "CreateCapabilityMap called for device %s", deviceUniqueIdUTF8);
-
-
- WebRtc_UWord8 productId[kVideoCaptureProductIdLength];
- IBaseFilter* captureDevice = DeviceInfoWindows::GetDeviceFilter(
- deviceUniqueIdUTF8,
- productId,
- kVideoCaptureProductIdLength);
- if (!captureDevice)
- return -1;
- IPin* outputCapturePin = GetOutputPin(captureDevice);
- if (!outputCapturePin)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to get capture device output pin");
- RELEASE_AND_CLEAR(captureDevice);
- return -1;
- }
- IAMExtDevice* extDevice = NULL;
- HRESULT hr = captureDevice->QueryInterface(IID_IAMExtDevice,
- (void **) &extDevice);
- if (SUCCEEDED(hr) && extDevice)
- {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "This is an external device");
- extDevice->Release();
- }
-
- IAMStreamConfig* streamConfig = NULL;
- hr = outputCapturePin->QueryInterface(IID_IAMStreamConfig,
- (void**) &streamConfig);
- if (FAILED(hr))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to get IID_IAMStreamConfig interface from capture device");
- return -1;
- }
-
- // this gets the FPS
- IAMVideoControl* videoControlConfig = NULL;
- HRESULT hrVC = captureDevice->QueryInterface(IID_IAMVideoControl,
- (void**) &videoControlConfig);
- if (FAILED(hrVC))
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
- "IID_IAMVideoControl Interface NOT SUPPORTED");
- }
-
- AM_MEDIA_TYPE *pmt = NULL;
- VIDEO_STREAM_CONFIG_CAPS caps;
- int count, size;
-
- hr = streamConfig->GetNumberOfCapabilities(&count, &size);
- if (FAILED(hr))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to GetNumberOfCapabilities");
- RELEASE_AND_CLEAR(videoControlConfig);
- RELEASE_AND_CLEAR(streamConfig);
- RELEASE_AND_CLEAR(outputCapturePin);
- RELEASE_AND_CLEAR(captureDevice);
- return -1;
- }
-
- WebRtc_Word32 index = 0; // Index in created _capabilities map
- // Check if the device support formattype == FORMAT_VideoInfo2 and FORMAT_VideoInfo.
- // Prefer FORMAT_VideoInfo since some cameras (ZureCam) has been seen having problem with MJPEG and FORMAT_VideoInfo2
- // Interlace flag is only supported in FORMAT_VideoInfo2
- bool supportFORMAT_VideoInfo2 = false;
- bool supportFORMAT_VideoInfo = false;
- bool foundInterlacedFormat = false;
- GUID preferedVideoFormat = FORMAT_VideoInfo;
- for (WebRtc_Word32 tmp = 0; tmp < count; ++tmp)
- {
- hr = streamConfig->GetStreamCaps(tmp, &pmt,
- reinterpret_cast<BYTE*> (&caps));
- if (!FAILED(hr))
- {
- if (pmt->majortype == MEDIATYPE_Video
- && pmt->formattype == FORMAT_VideoInfo2)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- " Device support FORMAT_VideoInfo2");
- supportFORMAT_VideoInfo2 = true;
- VIDEOINFOHEADER2* h =
- reinterpret_cast<VIDEOINFOHEADER2*> (pmt->pbFormat);
- assert(h);
- foundInterlacedFormat |= h->dwInterlaceFlags
- & (AMINTERLACE_IsInterlaced
- | AMINTERLACE_DisplayModeBobOnly);
- }
- if (pmt->majortype == MEDIATYPE_Video
- && pmt->formattype == FORMAT_VideoInfo)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCapture, _id,
- " Device support FORMAT_VideoInfo2");
- supportFORMAT_VideoInfo = true;
- }
- }
- }
- if (supportFORMAT_VideoInfo2)
- {
- if (supportFORMAT_VideoInfo && !foundInterlacedFormat)
- {
- preferedVideoFormat = FORMAT_VideoInfo;
- }
- else
- {
- preferedVideoFormat = FORMAT_VideoInfo2;
- }
- }
-
- for (WebRtc_Word32 tmp = 0; tmp < count; ++tmp)
- {
- hr = streamConfig->GetStreamCaps(tmp, &pmt,
- reinterpret_cast<BYTE*> (&caps));
- if (FAILED(hr))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to GetStreamCaps");
- RELEASE_AND_CLEAR(videoControlConfig);
- RELEASE_AND_CLEAR(streamConfig);
- RELEASE_AND_CLEAR(outputCapturePin);
- RELEASE_AND_CLEAR(captureDevice);
- return -1;
- }
-
- if (pmt->majortype == MEDIATYPE_Video
- && pmt->formattype == preferedVideoFormat)
- {
-
- VideoCaptureCapabilityWindows* capability =
- new VideoCaptureCapabilityWindows();
- WebRtc_Word64 avgTimePerFrame = 0;
- bool interlaced = false;
-
- if (pmt->formattype == FORMAT_VideoInfo)
- {
- VIDEOINFOHEADER* h =
- reinterpret_cast<VIDEOINFOHEADER*> (pmt->pbFormat);
- assert(h);
- capability->directShowCapabilityIndex = tmp;
- capability->width = h->bmiHeader.biWidth;
- capability->height = h->bmiHeader.biHeight;
- avgTimePerFrame = h->AvgTimePerFrame;
- }
- if (pmt->formattype == FORMAT_VideoInfo2)
- {
- VIDEOINFOHEADER2* h =
- reinterpret_cast<VIDEOINFOHEADER2*> (pmt->pbFormat);
- assert(h);
- capability->directShowCapabilityIndex = tmp;
- capability->width = h->bmiHeader.biWidth;
- capability->height = h->bmiHeader.biHeight;
- capability->interlaced = h->dwInterlaceFlags
- & (AMINTERLACE_IsInterlaced
- | AMINTERLACE_DisplayModeBobOnly);
- avgTimePerFrame = h->AvgTimePerFrame;
- }
-
- if (hrVC == S_OK)
- {
- LONGLONG *maxFps; // array
- long listSize;
- SIZE size;
- size.cx = capability->width;
- size.cy = capability->height;
-
- // GetMaxAvailableFrameRate doesn't return max frame rate always
- // eg: Logitech Notebook. This may be due to a bug in that API
- // because GetFrameRateList array is reversed in the above camera. So
- // a util method written. Can't assume the first value will return
- // the max fps.
- hrVC = videoControlConfig->GetFrameRateList(outputCapturePin,
- tmp, size,
- &listSize,
- &maxFps);
-
- if (hrVC == S_OK && listSize > 0)
- {
- LONGLONG maxFPS = GetMaxOfFrameArray(maxFps, listSize);
- capability->maxFPS = static_cast<int> (10000000
- / maxFPS);
- capability->supportFrameRateControl = true;
- }
- else // use existing method
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture,
- _id,
- "GetMaxAvailableFrameRate NOT SUPPORTED");
- if (avgTimePerFrame > 0)
- capability->maxFPS = static_cast<int> (10000000
- / avgTimePerFrame);
- else
- capability->maxFPS = 0;
- }
- }
- else // use existing method in case IAMVideoControl is not supported
- {
- if (avgTimePerFrame > 0)
- capability->maxFPS = static_cast<int> (10000000
- / avgTimePerFrame);
- else
- capability->maxFPS = 0;
- }
-
- // can't switch MEDIATYPE :~(
- if (pmt->subtype == MEDIASUBTYPE_I420)
- {
- capability->rawType = kVideoI420;
- }
- else if (pmt->subtype == MEDIASUBTYPE_IYUV)
- {
- capability->rawType = kVideoIYUV;
- }
- else if (pmt->subtype == MEDIASUBTYPE_RGB24)
- {
- capability->rawType = kVideoRGB24;
- }
- else if (pmt->subtype == MEDIASUBTYPE_YUY2)
- {
- capability->rawType = kVideoYUY2;
- }
- else if (pmt->subtype == MEDIASUBTYPE_RGB565)
- {
- capability->rawType = kVideoRGB565;
- }
- else if (pmt->subtype == MEDIASUBTYPE_MJPG)
- {
- capability->rawType = kVideoMJPEG;
- }
- else if (pmt->subtype == MEDIASUBTYPE_dvsl
- || pmt->subtype == MEDIASUBTYPE_dvsd
- || pmt->subtype == MEDIASUBTYPE_dvhd) // If this is an external DV camera
- {
- capability->rawType = kVideoYUY2;// MS DV filter seems to create this type
- }
- else if (pmt->subtype == MEDIASUBTYPE_UYVY) // Seen used by Declink capture cards
- {
- capability->rawType = kVideoUYVY;
- }
- else if (pmt->subtype == MEDIASUBTYPE_HDYC) // Seen used by Declink capture cards. Uses BT. 709 color. Not entiry correct to use UYVY. http://en.wikipedia.org/wiki/YCbCr
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
- "Device support HDYC.");
- capability->rawType = kVideoUYVY;
- }
- else
- {
- WCHAR strGuid[39];
- StringFromGUID2(pmt->subtype, strGuid, 39);
- WEBRTC_TRACE( webrtc::kTraceWarning,
- webrtc::kTraceVideoCapture, _id,
- "Device support unknown media type %ls, width %d, height %d",
- strGuid);
- delete capability;
- continue;
- }
-
- // Get the expected capture delay from the static list
- capability->expectedCaptureDelay
- = GetExpectedCaptureDelay(WindowsCaptureDelays,
- NoWindowsCaptureDelays,
- productId,
- capability->width,
- capability->height);
- _captureCapabilities.Insert(index++, capability);
- WEBRTC_TRACE( webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "Camera capability, width:%d height:%d type:%d fps:%d",
- capability->width, capability->height,
- capability->rawType, capability->maxFPS);
- }
- DeleteMediaType(pmt);
- pmt = NULL;
- }
- RELEASE_AND_CLEAR(streamConfig);
- RELEASE_AND_CLEAR(videoControlConfig);
- RELEASE_AND_CLEAR(outputCapturePin);
- RELEASE_AND_CLEAR(captureDevice); // Release the capture device
-
- // Store the new used device name
- _lastUsedDeviceNameLength = deviceUniqueIdUTF8Length;
- _lastUsedDeviceName = (WebRtc_UWord8*) realloc(_lastUsedDeviceName,
- _lastUsedDeviceNameLength
- + 1);
- memcpy(_lastUsedDeviceName, deviceUniqueIdUTF8, _lastUsedDeviceNameLength+ 1);
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "CreateCapabilityMap %d", _captureCapabilities.Size());
-
- return _captureCapabilities.Size();
-}
-
-/* Constructs a product ID from the Windows DevicePath. on a USB device the devicePath contains product id and vendor id.
- This seems to work for firewire as well
- /* Example of device path
- "\\?\usb#vid_0408&pid_2010&mi_00#7&258e7aaf&0&0000#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\global"
- "\\?\avc#sony&dv-vcr&camcorder&dv#65b2d50301460008#{65e8773d-8f56-11d0-a3b9-00a0c9223196}\global"
- */
-void DeviceInfoWindows::GetProductId(const WebRtc_UWord8* devicePath,
- WebRtc_UWord8* productUniqueIdUTF8,
- WebRtc_UWord32 productUniqueIdUTF8Length)
-{
- *productUniqueIdUTF8 = '\0';
- char* startPos = strstr((char*) devicePath, "\\\\?\\");
- if (!startPos)
- {
- strncpy_s((char*) productUniqueIdUTF8, productUniqueIdUTF8Length, "", 1);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "Failed to get the product Id");
- return;
- }
- startPos += 4;
-
- char* pos = strchr(startPos, '&');
- if (!pos || pos >= (char*) devicePath + strlen((char*) devicePath))
- {
- strncpy_s((char*) productUniqueIdUTF8, productUniqueIdUTF8Length, "", 1);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "Failed to get the product Id");
- return;
- }
- // Find the second occurence
- pos = strchr(pos + 1, '&');
- WebRtc_UWord32 bytesToCopy = (WebRtc_UWord32)(pos - startPos);
- if (pos && (bytesToCopy <= productUniqueIdUTF8Length) && bytesToCopy
- <= kVideoCaptureProductIdLength)
- {
- strncpy_s((char*) productUniqueIdUTF8, productUniqueIdUTF8Length,
- (char*) startPos, bytesToCopy);
- }
- else
- {
- strncpy_s((char*) productUniqueIdUTF8, productUniqueIdUTF8Length, "", 1);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, -1,
- "Failed to get the product Id");
- }
-}
-
-WebRtc_Word32 DeviceInfoWindows::DisplayCaptureSettingsDialogBox(
- const WebRtc_UWord8* deviceUniqueIdUTF8,
- const WebRtc_UWord8* dialogTitleUTF8,
- void* parentWindow,
- WebRtc_UWord32 positionX,
- WebRtc_UWord32 positionY)
-{
- ReadLockScoped cs(_apiLock);
- HWND window = (HWND) parentWindow;
-
- IBaseFilter* filter = GetDeviceFilter(deviceUniqueIdUTF8, NULL, 0);
- if (!filter)
- return -1;
-
- ISpecifyPropertyPages* pPages = NULL;
- CAUUID uuid;
- HRESULT hr = S_OK;
-
- hr = filter->QueryInterface(IID_ISpecifyPropertyPages, (LPVOID*) &pPages);
- if (!SUCCEEDED(hr))
- {
- filter->Release();
- return -1;
- }
- hr = pPages->GetPages(&uuid);
- if (!SUCCEEDED(hr))
- {
- filter->Release();
- return -1;
- }
-
- WCHAR tempDialogTitleWide[256];
- tempDialogTitleWide[0] = 0;
- int size = 255;
-
- // UTF-8 to wide char
- MultiByteToWideChar(CP_UTF8, 0, (char*) dialogTitleUTF8, -1,
- tempDialogTitleWide, size);
-
- // Invoke a dialog box to display.
-
- hr = OleCreatePropertyFrame(window, // You must create the parent window.
- positionX, // Horizontal position for the dialog box.
- positionY, // Vertical position for the dialog box.
- tempDialogTitleWide,// String used for the dialog box caption.
- 1, // Number of pointers passed in pPlugin.
- (LPUNKNOWN*) &filter, // Pointer to the filter.
- uuid.cElems, // Number of property pages.
- uuid.pElems, // Array of property page CLSIDs.
- LOCALE_USER_DEFAULT, // Locale ID for the dialog box.
- 0, NULL); // Reserved
- // Release memory.
- if (uuid.pElems)
- {
- CoTaskMemFree(uuid.pElems);
- }
- filter->Release();
- return 0;
-}
-} // namespace videocapturemodule
-} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Windows/device_info_windows.h b/trunk/src/modules/video_capture/main/source/Windows/device_info_windows.h
deleted file mode 100644
index b05a63c..0000000
--- a/trunk/src/modules/video_capture/main/source/Windows/device_info_windows.h
+++ /dev/null
@@ -1,105 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_DEVICE_INFO_WINDOWS_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_DEVICE_INFO_WINDOWS_H_
-
-#include "../video_capture_impl.h"
-#include "../device_info_impl.h"
-
-#include <Dshow.h>
-#include "map_wrapper.h"
-
-// forward declarations
-namespace webrtc
-{
-namespace videocapturemodule
-{
-struct VideoCaptureCapabilityWindows: public VideoCaptureCapability
-{
- WebRtc_UWord32 directShowCapabilityIndex;
- bool supportFrameRateControl;
- VideoCaptureCapabilityWindows()
- {
- directShowCapabilityIndex = 0;
- supportFrameRateControl = false;
- }
-
-};
-class DeviceInfoWindows: public DeviceInfoImpl
-{
-public:
- DeviceInfoWindows(const WebRtc_Word32 id);
- virtual ~DeviceInfoWindows();
-
- WebRtc_Word32 Init();
- virtual WebRtc_UWord32 NumberOfDevices();
-
- /*
- * Returns the available capture devices.
- */
- virtual WebRtc_Word32
- GetDeviceName(WebRtc_UWord32 deviceNumber,
- WebRtc_UWord8* deviceNameUTF8,
- WebRtc_UWord32 deviceNameLength,
- WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord32 deviceUniqueIdUTF8Length,
- WebRtc_UWord8* productUniqueIdUTF8,
- WebRtc_UWord32 productUniqueIdUTF8Length);
-
- /*
- * Display OS /capture device specific settings dialog
- */
- virtual WebRtc_Word32
- DisplayCaptureSettingsDialogBox(
- const WebRtc_UWord8* deviceUniqueIdUTF8,
- const WebRtc_UWord8* dialogTitleUTF8,
- void* parentWindow,
- WebRtc_UWord32 positionX,
- WebRtc_UWord32 positionY);
-
- // Windows specific
-
- /* Gets a capture device filter
- The user of this API is responsible for releasing the filter when it not needed.
- */
- IBaseFilter * GetDeviceFilter(const WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord8* productUniqueIdUTF8 = NULL,
- WebRtc_UWord32 productUniqueIdUTF8Length = 0);
-
- WebRtc_Word32
- GetWindowsCapability(const WebRtc_Word32 capabilityIndex,
- VideoCaptureCapabilityWindows& windowsCapability);
-
- static void GetProductId(const WebRtc_UWord8* devicePath,
- WebRtc_UWord8* productUniqueIdUTF8,
- WebRtc_UWord32 productUniqueIdUTF8Length);
-protected:
-
- WebRtc_Word32 GetDeviceInfo(WebRtc_UWord32 deviceNumber,
- WebRtc_UWord8* deviceNameUTF8,
- WebRtc_UWord32 deviceNameLength,
- WebRtc_UWord8* deviceUniqueIdUTF8,
- WebRtc_UWord32 deviceUniqueIdUTF8Length,
- WebRtc_UWord8* productUniqueIdUTF8,
- WebRtc_UWord32 productUniqueIdUTF8Length);
-
- virtual WebRtc_Word32
- CreateCapabilityMap(const WebRtc_UWord8* deviceUniqueIdUTF8);
-
-private:
- ICreateDevEnum* _dsDevEnum;
- IEnumMoniker* _dsMonikerDevEnum;
- bool _CoUninitializeIsRequired;
-
-};
-} // namespace videocapturemodule
-} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_DEVICE_INFO_WINDOWS_H_
diff --git a/trunk/src/modules/video_capture/main/source/Windows/help_functions_windows.cc b/trunk/src/modules/video_capture/main/source/Windows/help_functions_windows.cc
deleted file mode 100644
index dd533a3..0000000
--- a/trunk/src/modules/video_capture/main/source/Windows/help_functions_windows.cc
+++ /dev/null
@@ -1,116 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "help_functions_windows.h"
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-// This returns minimum :), which will give max frame rate...
-LONGLONG GetMaxOfFrameArray(LONGLONG *maxFps, long size)
-{
- LONGLONG maxFPS = maxFps[0];
- for (int i = 0; i < size; i++)
- {
- if (maxFPS > maxFps[i])
- maxFPS = maxFps[i];
- }
- return maxFPS;
-}
-
-IPin* GetInputPin(IBaseFilter* filter)
-{
- HRESULT hr;
- IPin* pin = NULL;
- IEnumPins* pPinEnum = NULL;
- filter->EnumPins(&pPinEnum);
- if (pPinEnum == NULL)
- {
- return NULL;
- }
-
- // get first unconnected pin
- hr = pPinEnum->Reset(); // set to first pin
-
- int count = 0;
- while (S_OK == pPinEnum->Next(1, &pin, NULL))
- {
- PIN_DIRECTION pPinDir;
- pin->QueryDirection(&pPinDir);
- if (PINDIR_INPUT == pPinDir) // This is an input pin
- {
- IPin* tempPin = NULL;
- if (S_OK != pin->ConnectedTo(&tempPin)) // The pint is not connected
- {
- pPinEnum->Release();
- return pin;
- }
- }
- pin->Release();
- }
- pPinEnum->Release();
- return NULL;
-}
-
-IPin* GetOutputPin(IBaseFilter* filter, REFGUID Category)
-{
- HRESULT hr;
- IPin* pin = NULL;
- IEnumPins* pPinEnum = NULL;
- filter->EnumPins(&pPinEnum);
- if (pPinEnum == NULL)
- {
- return NULL;
- }
- // get first unconnected pin
- hr = pPinEnum->Reset(); // set to first pin
- int count = 0;
- while (S_OK == pPinEnum->Next(1, &pin, NULL))
- {
- PIN_DIRECTION pPinDir;
- pin->QueryDirection(&pPinDir);
- if (PINDIR_OUTPUT == pPinDir) // This is an output pin
- {
- if (Category == GUID_NULL || PinMatchesCategory(pin, Category))
- {
- pPinEnum->Release();
- return pin;
- }
- }
- pin->Release();
- pin = NULL;
- }
- pPinEnum->Release();
- return NULL;
-}
-
-BOOL PinMatchesCategory(IPin *pPin, REFGUID Category)
-{
- BOOL bFound = FALSE;
- IKsPropertySet *pKs = NULL;
- HRESULT hr = pPin->QueryInterface(IID_PPV_ARGS(&pKs));
- if (SUCCEEDED(hr))
- {
- GUID PinCategory;
- DWORD cbReturned;
- hr = pKs->Get(AMPROPSETID_Pin, AMPROPERTY_PIN_CATEGORY, NULL, 0, &PinCategory,
- sizeof(GUID), &cbReturned);
- if (SUCCEEDED(hr) && (cbReturned == sizeof(GUID)))
- {
- bFound = (PinCategory == Category);
- }
- pKs->Release();
- }
- return bFound;
-}
-} // namespace videocapturemodule
-} // namespace webrtc
-
diff --git a/trunk/src/modules/video_capture/main/source/Windows/help_functions_windows.h b/trunk/src/modules/video_capture/main/source/Windows/help_functions_windows.h
deleted file mode 100644
index 0020877..0000000
--- a/trunk/src/modules/video_capture/main/source/Windows/help_functions_windows.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_HELP_FUNCTIONS_WINDOWS_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_HELP_FUNCTIONS_WINDOWS_H_
-
-#include <dshow.h>
-#include <initguid.h>
-
-DEFINE_GUID(MEDIASUBTYPE_I420, 0x30323449, 0x0000, 0x0010, 0x80, 0x00, 0x00,
- 0xAA, 0x00, 0x38, 0x9B, 0x71);
-DEFINE_GUID(MEDIASUBTYPE_V210, 0x30313276, 0x0000, 0x0010, 0x80, 0x00, 0x00,
- 0xAA, 0x00, 0x38, 0x9B, 0x71);
-DEFINE_GUID(MEDIASUBTYPE_HDYC, 0x43594448, 0x0000, 0x0010, 0x80, 0x00, 0x00,
- 0xAA, 0x00, 0x38, 0x9B, 0x71);
-
-#define RELEASE_AND_CLEAR(p) if (p) { (p) -> Release () ; (p) = NULL ; }
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-LONGLONG GetMaxOfFrameArray(LONGLONG *maxFps, long size);
-
-IPin* GetInputPin(IBaseFilter* filter);
-IPin* GetOutputPin(IBaseFilter* filter, REFGUID Category = GUID_NULL);
-BOOL PinMatchesCategory(IPin *pPin, REFGUID Category);
-
-} // namespace videocapturemodule
-} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_HELP_FUNCTIONS_WINDOWS_H_
diff --git a/trunk/src/modules/video_capture/main/source/Windows/sink_filter_windows.cc b/trunk/src/modules/video_capture/main/source/Windows/sink_filter_windows.cc
deleted file mode 100644
index 88e47a0..0000000
--- a/trunk/src/modules/video_capture/main/source/Windows/sink_filter_windows.cc
+++ /dev/null
@@ -1,506 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "sink_filter_windows.h"
-
-#include "trace.h"
-#include "help_functions_windows.h"
-
-#include <Dvdmedia.h> // VIDEOINFOHEADER2
-#include <initguid.h>
-
-#define DELETE_RESET(p) { delete (p) ; (p) = NULL ;}
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-
-typedef struct tagTHREADNAME_INFO
-{
- DWORD dwType; // must be 0x1000
- LPCSTR szName; // pointer to name (in user addr space)
- DWORD dwThreadID; // thread ID (-1=caller thread)
- DWORD dwFlags; // reserved for future use, must be zero
-} THREADNAME_INFO;
-
-DEFINE_GUID(CLSID_SINKFILTER, 0x88cdbbdc, 0xa73b, 0x4afa, 0xac, 0xbf, 0x15, 0xd5,
- 0xe2, 0xce, 0x12, 0xc3);
-
-CaptureInputPin::CaptureInputPin (WebRtc_Word32 moduleId,
- IN TCHAR * szName,
- IN CaptureSinkFilter* pFilter,
- IN CCritSec * pLock,
- OUT HRESULT * pHr,
- IN LPCWSTR pszName)
- : CBaseInputPin (szName, pFilter, pLock, pHr, pszName),
- _requestedCapability(),
- _resultingCapability()
-{
- _moduleId=moduleId;
- _threadHandle = NULL;
-}
-
-CaptureInputPin::~CaptureInputPin()
-{
-}
-
-HRESULT
-CaptureInputPin::GetMediaType (IN int iPosition, OUT CMediaType * pmt)
-{
- // reset the thread handle
- _threadHandle = NULL;
-
- if(iPosition < 0)
- return E_INVALIDARG;
-
- VIDEOINFOHEADER* pvi = (VIDEOINFOHEADER*) pmt->AllocFormatBuffer(
- sizeof(VIDEOINFOHEADER));
- if(NULL == pvi)
- {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _moduleId,
- "CheckMediaType VIDEOINFOHEADER is NULL. Returning...Line:%d\n", __LINE__);
- return(E_OUTOFMEMORY);
- }
-
- ZeroMemory(pvi, sizeof(VIDEOINFOHEADER));
- pvi->bmiHeader.biSize = sizeof(BITMAPINFOHEADER);
- pvi->bmiHeader.biPlanes = 1;
- pvi->bmiHeader.biClrImportant = 0;
- pvi->bmiHeader.biClrUsed = 0;
- pvi->AvgTimePerFrame = 10000000/_requestedCapability.maxFPS;
-
- SetRectEmpty(&(pvi->rcSource)); // we want the whole image area rendered.
- SetRectEmpty(&(pvi->rcTarget)); // no particular destination rectangle
-
- pmt->SetType(&MEDIATYPE_Video);
- pmt->SetFormatType(&FORMAT_VideoInfo);
- pmt->SetTemporalCompression(FALSE);
-
- WebRtc_Word32 positionOffset=1;
- if(_requestedCapability.codecType!=kVideoCodecUnknown)
- {
- positionOffset=0;
- }
-
- switch (iPosition+positionOffset)
- {
- case 0:
- {
- pvi->bmiHeader.biCompression = MAKEFOURCC('I','4','2','0');
- pvi->bmiHeader.biBitCount = 12; //bit per pixel
- pvi->bmiHeader.biWidth = _requestedCapability.width;
- pvi->bmiHeader.biHeight = _requestedCapability.height;
- pvi->bmiHeader.biSizeImage = 3*_requestedCapability.height
- *_requestedCapability.width/2;
- pmt->SetSubtype(&MEDIASUBTYPE_I420);
- }
- break;
- case 1:
- {
- pvi->bmiHeader.biCompression = MAKEFOURCC('Y','U','Y','2');;
- pvi->bmiHeader.biBitCount = 16; //bit per pixel
- pvi->bmiHeader.biWidth = _requestedCapability.width;
- pvi->bmiHeader.biHeight = _requestedCapability.height;
- pvi->bmiHeader.biSizeImage = 2*_requestedCapability.width
- *_requestedCapability.height;
- pmt->SetSubtype(&MEDIASUBTYPE_YUY2);
- }
- break;
- case 2:
- {
- pvi->bmiHeader.biCompression = BI_RGB;
- pvi->bmiHeader.biBitCount = 24; //bit per pixel
- pvi->bmiHeader.biWidth = _requestedCapability.width;
- pvi->bmiHeader.biHeight = _requestedCapability.height;
- pvi->bmiHeader.biSizeImage = 3*_requestedCapability.height
- *_requestedCapability.width;
- pmt->SetSubtype(&MEDIASUBTYPE_RGB24);
- }
- break;
- case 3:
- {
- pvi->bmiHeader.biCompression = MAKEFOURCC('U','Y','V','Y');
- pvi->bmiHeader.biBitCount = 16; //bit per pixel
- pvi->bmiHeader.biWidth = _requestedCapability.width;
- pvi->bmiHeader.biHeight = _requestedCapability.height;
- pvi->bmiHeader.biSizeImage = 2*_requestedCapability.height
- *_requestedCapability.width;
- pmt->SetSubtype(&MEDIASUBTYPE_UYVY);
- }
- break;
- case 4:
- {
- pvi->bmiHeader.biCompression = MAKEFOURCC('M','J','P','G');
- pvi->bmiHeader.biBitCount = 12; //bit per pixel
- pvi->bmiHeader.biWidth = _requestedCapability.width;
- pvi->bmiHeader.biHeight = _requestedCapability.height;
- pvi->bmiHeader.biSizeImage = 3*_requestedCapability.height
- *_requestedCapability.width/2;
- pmt->SetSubtype(&MEDIASUBTYPE_MJPG);
- }
- break;
- default :
- return VFW_S_NO_MORE_ITEMS;
- }
- pmt->SetSampleSize(pvi->bmiHeader.biSizeImage);
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _moduleId,
- "GetMediaType position %d, width %d, height %d, biCompression 0x%x",
- iPosition, _requestedCapability.width,
- _requestedCapability.height,pvi->bmiHeader.biCompression);
- return NOERROR;
-}
-
-HRESULT
-CaptureInputPin::CheckMediaType ( IN const CMediaType * pMediaType)
-{
- // reset the thread handle
- _threadHandle = NULL;
-
- const GUID *type = pMediaType->Type();
- if (*type != MEDIATYPE_Video)
- return E_INVALIDARG;
-
- const GUID *formatType = pMediaType->FormatType();
-
- // Check for the subtypes we support
- const GUID *SubType = pMediaType->Subtype();
- if (SubType == NULL)
- {
- return E_INVALIDARG;
- }
-
- if(*formatType == FORMAT_VideoInfo)
- {
- VIDEOINFOHEADER *pvi = (VIDEOINFOHEADER *) pMediaType->Format();
- if(pvi == NULL)
- {
- return E_INVALIDARG;
- }
-
- // Store the incoming width and height
- _resultingCapability.width = pvi->bmiHeader.biWidth;
- _resultingCapability.height = abs(pvi->bmiHeader.biHeight);
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _moduleId,
- "CheckMediaType width:%d height:%d Compression:0x%x\n",
- pvi->bmiHeader.biWidth,pvi->bmiHeader.biHeight,
- pvi->bmiHeader.biCompression);
-
- if(*SubType == MEDIASUBTYPE_MJPG
- && pvi->bmiHeader.biCompression == MAKEFOURCC('M','J','P','G'))
- {
- _resultingCapability.rawType = kVideoMJPEG;
- return S_OK; // This format is acceptable.
- }
- if(*SubType == MEDIASUBTYPE_I420
- && pvi->bmiHeader.biCompression == MAKEFOURCC('I','4','2','0'))
- {
- _resultingCapability.rawType = kVideoI420;
- return S_OK; // This format is acceptable.
- }
- if(*SubType == MEDIASUBTYPE_YUY2
- && pvi->bmiHeader.biCompression == MAKEFOURCC('Y','U','Y','2'))
- {
- _resultingCapability.rawType = kVideoYUY2;
- ::Sleep(60); // workaround for bad driver
- return S_OK; // This format is acceptable.
- }
- if(*SubType == MEDIASUBTYPE_UYVY
- && pvi->bmiHeader.biCompression == MAKEFOURCC('U','Y','V','Y'))
- {
- _resultingCapability.rawType = kVideoUYVY;
- return S_OK; // This format is acceptable.
- }
-
- if(*SubType == MEDIASUBTYPE_HDYC)
- {
- _resultingCapability.rawType = kVideoUYVY;
- return S_OK; // This format is acceptable.
- }
- if(*SubType == MEDIASUBTYPE_RGB24
- && pvi->bmiHeader.biCompression == BI_RGB)
- {
- _resultingCapability.rawType = kVideoRGB24;
- return S_OK; // This format is acceptable.
- }
- }
- if(*formatType == FORMAT_VideoInfo2)
- {
- // VIDEOINFOHEADER2 that has dwInterlaceFlags
- VIDEOINFOHEADER2 *pvi = (VIDEOINFOHEADER2 *) pMediaType->Format();
-
- if(pvi == NULL)
- {
- return E_INVALIDARG;
- }
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _moduleId,
- "CheckMediaType width:%d height:%d Compression:0x%x\n",
- pvi->bmiHeader.biWidth,pvi->bmiHeader.biHeight,
- pvi->bmiHeader.biCompression);
-
- _resultingCapability.width = pvi->bmiHeader.biWidth;
- _resultingCapability.height = abs(pvi->bmiHeader.biHeight);
-
- if(*SubType == MEDIASUBTYPE_MJPG
- && pvi->bmiHeader.biCompression == MAKEFOURCC('M','J','P','G'))
- {
- _resultingCapability.rawType = kVideoMJPEG;
- return S_OK; // This format is acceptable.
- }
- if(*SubType == MEDIASUBTYPE_I420
- && pvi->bmiHeader.biCompression == MAKEFOURCC('I','4','2','0'))
- {
- _resultingCapability.rawType = kVideoI420;
- return S_OK; // This format is acceptable.
- }
- if(*SubType == MEDIASUBTYPE_YUY2
- && pvi->bmiHeader.biCompression == MAKEFOURCC('Y','U','Y','2'))
- {
- _resultingCapability.rawType = kVideoYUY2;
- return S_OK; // This format is acceptable.
- }
- if(*SubType == MEDIASUBTYPE_UYVY
- && pvi->bmiHeader.biCompression == MAKEFOURCC('U','Y','V','Y'))
- {
- _resultingCapability.rawType = kVideoUYVY;
- return S_OK; // This format is acceptable.
- }
-
- if(*SubType == MEDIASUBTYPE_HDYC)
- {
- _resultingCapability.rawType = kVideoUYVY;
- return S_OK; // This format is acceptable.
- }
- if(*SubType == MEDIASUBTYPE_RGB24
- && pvi->bmiHeader.biCompression == BI_RGB)
- {
- _resultingCapability.rawType = kVideoRGB24;
- return S_OK; // This format is acceptable.
- }
- }
- return E_INVALIDARG;
-}
-
-HRESULT
-CaptureInputPin::Receive ( IN IMediaSample * pIMediaSample )
-{
- HRESULT hr = S_OK;
-
- ASSERT (m_pFilter);
- ASSERT (pIMediaSample);
-
- // get the thread handle of the delivering thread inc its priority
- if( _threadHandle == NULL)
- {
- HANDLE handle= GetCurrentThread();
- SetThreadPriority(handle, THREAD_PRIORITY_HIGHEST);
- _threadHandle = handle;
- // See http://msdn.microsoft.com/en-us/library/xcb2z8hs(VS.71).aspx for details on the code
- // in this function. Name od article is "Setting a Thread Name (Unmanaged)".
-
- THREADNAME_INFO info;
- info.dwType = 0x1000;
- info.szName = "capture_thread";
- info.dwThreadID = (DWORD)-1;
- info.dwFlags = 0;
-
- __try
- {
- RaiseException( 0x406D1388, 0, sizeof(info)/sizeof(DWORD),
- (DWORD_PTR*)&info );
- }
- __except (EXCEPTION_CONTINUE_EXECUTION)
- {
- }
-
- }
-
- reinterpret_cast <CaptureSinkFilter *>(m_pFilter)->LockReceive();
- hr = CBaseInputPin::Receive (pIMediaSample);
-
- if (SUCCEEDED (hr))
- {
- const WebRtc_Word32 length = pIMediaSample->GetActualDataLength();
-
- unsigned char* pBuffer = NULL;
- if(S_OK != pIMediaSample->GetPointer(&pBuffer))
- {
- reinterpret_cast <CaptureSinkFilter *>(m_pFilter)->UnlockReceive();
- return S_FALSE;
- }
-
- // NOTE: filter unlocked within Send call
- reinterpret_cast <CaptureSinkFilter *> (m_pFilter)->ProcessCapturedFrame(
- pBuffer,length,_resultingCapability);
- }
- else
- {
- reinterpret_cast <CaptureSinkFilter *>(m_pFilter)->UnlockReceive();
- }
-
- return hr;
-}
-
-// called under LockReceive
-HRESULT CaptureInputPin::SetMatchingMediaType(
- const VideoCaptureCapability& capability)
-{
-
- _requestedCapability = capability;
- _resultingCapability = VideoCaptureCapability();
- return S_OK;
-}
-// ----------------------------------------------------------------------------
-CaptureSinkFilter::CaptureSinkFilter (IN TCHAR * tszName,
- IN LPUNKNOWN punk,
- OUT HRESULT * phr,
- VideoCaptureExternal& captureObserver,
- WebRtc_Word32 moduleId)
- : CBaseFilter(tszName,punk,& m_crtFilter,CLSID_SINKFILTER),
- m_pInput(NULL),
- _captureObserver(captureObserver),
- _moduleId(moduleId)
-{
- (* phr) = S_OK;
- m_pInput = new CaptureInputPin(moduleId,NAME ("VideoCaptureInputPin"),
- this,
- & m_crtFilter,
- phr, L"VideoCapture");
- if (m_pInput == NULL || FAILED (* phr))
- {
- (* phr) = FAILED (* phr) ? (* phr) : E_OUTOFMEMORY;
- goto cleanup;
- }
- cleanup :
- return;
-}
-
-CaptureSinkFilter::~CaptureSinkFilter()
-{
- delete m_pInput;
-}
-
-int CaptureSinkFilter::GetPinCount()
-{
- return 1;
-}
-
-CBasePin *
-CaptureSinkFilter::GetPin(IN int Index)
-{
- CBasePin * pPin;
- LockFilter ();
- if (Index == 0)
- {
- pPin = m_pInput;
- }
- else
- {
- pPin = NULL;
- }
- UnlockFilter ();
- return pPin;
-}
-
-STDMETHODIMP CaptureSinkFilter::Pause()
-{
- HRESULT hr = S_OK;
- LockFilter();
- if (m_State == State_Stopped)
- {
- // change the state, THEN activate the input pin
- m_State = State_Paused;
- if (m_pInput && m_pInput->IsConnected())
- {
- m_pInput->Active();
- }
- if (m_pInput && !m_pInput->IsConnected())
- {
- m_State = State_Running;
- }
- }
- else if (m_State == State_Running)
- {
- m_State = State_Paused;
- }
- UnlockFilter();
- return S_OK;
-}
-
-STDMETHODIMP CaptureSinkFilter::Stop()
-{
- LockReceive();
- LockFilter();
-
- // set the state
- m_State = State_Stopped;
-
- // inactivate the pins
- if (m_pInput)
- m_pInput->Inactive();
-
- UnlockFilter();
- UnlockReceive();
- return S_OK;
-}
-
-void CaptureSinkFilter::SetFilterGraph(IGraphBuilder* graph)
-{
- LockFilter();
- m_pGraph = graph;
- UnlockFilter();
-}
-
-void CaptureSinkFilter::ProcessCapturedFrame(unsigned char* pBuffer,
- WebRtc_Word32 length,
- const VideoCaptureCapability& frameInfo)
-{
- // we have the receiver lock
- if (m_State == State_Running)
- {
- _captureObserver.IncomingFrame(pBuffer, length, frameInfo);
-
- // trying to hold it since it's only a memcpy
- // IMPROVEMENT if this work move critsect
- UnlockReceive();
- return;
- }
- UnlockReceive();
- return;
-}
-
-STDMETHODIMP CaptureSinkFilter::SetMatchingMediaType(
- const VideoCaptureCapability& capability)
-{
- LockReceive();
- LockFilter();
- HRESULT hr;
- if (m_pInput)
- {
- hr = m_pInput->SetMatchingMediaType(capability);
- }
- else
- {
- hr = E_UNEXPECTED;
- }
- UnlockFilter();
- UnlockReceive();
- return hr;
-}
-
-STDMETHODIMP CaptureSinkFilter::GetClassID( OUT CLSID * pCLSID )
-{
- (* pCLSID) = CLSID_SINKFILTER;
- return S_OK;
-}
-
-} // namespace videocapturemodule
-} //namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Windows/sink_filter_windows.h b/trunk/src/modules/video_capture/main/source/Windows/sink_filter_windows.h
deleted file mode 100644
index efeb9c7..0000000
--- a/trunk/src/modules/video_capture/main/source/Windows/sink_filter_windows.h
+++ /dev/null
@@ -1,100 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_SINK_FILTER_WINDOWS_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_SINK_FILTER_WINDOWS_H_
-
-#include <Streams.h> // Include base DS filter header files
-
-#include "video_capture_defines.h"
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-//forward declaration
-
-class CaptureSinkFilter;
-/**
- * input pin for camera input
- *
- */
-class CaptureInputPin: public CBaseInputPin
-{
-public:
- WebRtc_Word32 _moduleId;
-
- VideoCaptureCapability _requestedCapability;
- VideoCaptureCapability _resultingCapability;
- HANDLE _threadHandle;
-
- CaptureInputPin ( WebRtc_Word32 moduleId,
- IN TCHAR* szName,
- IN CaptureSinkFilter* pFilter,
- IN CCritSec * pLock,
- OUT HRESULT * pHr,
- IN LPCWSTR pszName);
- virtual ~CaptureInputPin();
-
- HRESULT GetMediaType (IN int iPos, OUT CMediaType * pmt);
- HRESULT CheckMediaType (IN const CMediaType * pmt);
- STDMETHODIMP Receive (IN IMediaSample *);
- HRESULT SetMatchingMediaType(const VideoCaptureCapability& capability);
-};
-
-class CaptureSinkFilter: public CBaseFilter
-{
-
-public:
- CaptureSinkFilter (IN TCHAR * tszName,
- IN LPUNKNOWN punk,
- OUT HRESULT * phr,
- VideoCaptureExternal& captureObserver,
- WebRtc_Word32 moduleId);
- virtual ~CaptureSinkFilter();
-
- // --------------------------------------------------------------------
- // class methods
-
- void ProcessCapturedFrame(unsigned char* pBuffer, WebRtc_Word32 length,
- const VideoCaptureCapability& frameInfo);
- // explicit receiver lock aquisition and release
- void LockReceive() { m_crtRecv.Lock();}
- void UnlockReceive() {m_crtRecv.Unlock();}
- // explicit filter lock aquisition and release
- void LockFilter() {m_crtFilter.Lock();}
- void UnlockFilter() { m_crtFilter.Unlock(); }
- void SetFilterGraph(IGraphBuilder* graph); // Used if EVR
-
- // --------------------------------------------------------------------
- // COM interfaces
-DECLARE_IUNKNOWN ;
- STDMETHODIMP SetMatchingMediaType(const VideoCaptureCapability& capability);
-
- // --------------------------------------------------------------------
- // CBaseFilter methods
- int GetPinCount ();
- CBasePin * GetPin ( IN int Index);
- STDMETHODIMP Pause ();
- STDMETHODIMP Stop ();
- STDMETHODIMP GetClassID ( OUT CLSID * pCLSID);
- // --------------------------------------------------------------------
- // class factory calls this
- static CUnknown * CreateInstance (IN LPUNKNOWN punk, OUT HRESULT * phr);
-private:
- CCritSec m_crtFilter; // filter lock
- CCritSec m_crtRecv; // receiver lock; always acquire before filter lock
- CaptureInputPin * m_pInput;
- VideoCaptureExternal& _captureObserver;
- WebRtc_Word32 _moduleId;
-};
-} // namespace videocapturemodule
-} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_SINK_FILTER_WINDOWS_H_
diff --git a/trunk/src/modules/video_capture/main/source/Windows/video_capture_factory_windows.cc b/trunk/src/modules/video_capture/main/source/Windows/video_capture_factory_windows.cc
deleted file mode 100644
index bfc7a30..0000000
--- a/trunk/src/modules/video_capture/main/source/Windows/video_capture_factory_windows.cc
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "ref_count.h"
-#include "video_capture_windows.h"
-#include "trace.h"
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-VideoCaptureModule* VideoCaptureImpl::Create(
- const WebRtc_Word32 id,
- const WebRtc_UWord8* deviceUniqueIdUTF8)
-{
-
- if (deviceUniqueIdUTF8 == NULL)
- {
- return NULL;
- }
-
- WebRtc_UWord8 productId[kVideoCaptureProductIdLength];
- videocapturemodule::DeviceInfoWindows::GetProductId(deviceUniqueIdUTF8,
- productId,
- sizeof(productId));
-
- RefCountImpl<videocapturemodule::VideoCaptureDS>* newCaptureModule =
- new RefCountImpl<videocapturemodule::VideoCaptureDS>(id);
-
- if (newCaptureModule->Init(id, deviceUniqueIdUTF8) != 0)
- {
- delete newCaptureModule;
- newCaptureModule = NULL;
- }
- return newCaptureModule;
-}
-} //namespace videocapturemodule
-} //namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Windows/video_capture_windows.cc b/trunk/src/modules/video_capture/main/source/Windows/video_capture_windows.cc
deleted file mode 100644
index 889b7ef..0000000
--- a/trunk/src/modules/video_capture/main/source/Windows/video_capture_windows.cc
+++ /dev/null
@@ -1,413 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_capture_windows.h"
-
-#include "../video_capture_config.h"
-#include "critical_section_wrapper.h"
-#include "help_functions_windows.h"
-#include "sink_filter_windows.h"
-#include "trace.h"
-
-#include <Dvdmedia.h> // VIDEOINFOHEADER2
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-VideoCaptureDS::VideoCaptureDS(const WebRtc_Word32 id)
- : VideoCaptureImpl(id), _dsInfo(id), _captureFilter(NULL),
- _graphBuilder(NULL), _mediaControl(NULL), _sinkFilter(NULL),
- _inputSendPin(NULL), _outputCapturePin(NULL), _dvFilter(NULL),
- _inputDvPin(NULL), _outputDvPin(NULL)
-{
-}
-
-VideoCaptureDS::~VideoCaptureDS()
-{
- if (_mediaControl)
- {
- _mediaControl->Stop();
- }
- if (_graphBuilder)
- {
- if (_sinkFilter)
- _graphBuilder->RemoveFilter(_sinkFilter);
- if (_captureFilter)
- _graphBuilder->RemoveFilter(_captureFilter);
- if (_dvFilter)
- _graphBuilder->RemoveFilter(_dvFilter);
- }
- RELEASE_AND_CLEAR(_captureFilter); // release the capture device
- RELEASE_AND_CLEAR(_sinkFilter);
- RELEASE_AND_CLEAR(_dvFilter);
-
- RELEASE_AND_CLEAR(_mediaControl);
- RELEASE_AND_CLEAR(_inputSendPin);
- RELEASE_AND_CLEAR(_outputCapturePin);
-
- RELEASE_AND_CLEAR(_inputDvPin);
- RELEASE_AND_CLEAR(_outputDvPin);
-
- RELEASE_AND_CLEAR(_graphBuilder);
-}
-
-WebRtc_Word32 VideoCaptureDS::Init(const WebRtc_Word32 id,
- const WebRtc_UWord8* deviceUniqueIdUTF8)
-{
- WebRtc_Word32 result = 0;
-
- const WebRtc_Word32 nameLength =
- (WebRtc_Word32) strlen((char*) deviceUniqueIdUTF8);
- if (nameLength > kVideoCaptureUniqueNameLength)
- return -1;
-
- // Store the device name
- _deviceUniqueId = new (std::nothrow) WebRtc_UWord8[nameLength + 1];
- memcpy(_deviceUniqueId, deviceUniqueIdUTF8, nameLength + 1);
-
- if (_dsInfo.Init() != 0)
- return -1;
-
- _captureFilter = _dsInfo.GetDeviceFilter(deviceUniqueIdUTF8);
- if (!_captureFilter)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to create capture filter.");
- return -1;
- }
-
- // Get the interface for DirectShow's GraphBuilder
- HRESULT hr = CoCreateInstance(CLSID_FilterGraph, NULL,
- CLSCTX_INPROC_SERVER, IID_IGraphBuilder,
- (void **) &_graphBuilder);
- if (FAILED(hr))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to create graph builder.");
- return -1;
- }
-
- hr = _graphBuilder->QueryInterface(IID_IMediaControl,
- (void **) &_mediaControl);
- if (FAILED(hr))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to create media control builder.");
- return -1;
- }
- hr = _graphBuilder->AddFilter(_captureFilter, CAPTURE_FILTER_NAME);
- if (FAILED(hr))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to add the capture device to the graph.");
- return -1;
- }
-
- _outputCapturePin = GetOutputPin(_captureFilter, PIN_CATEGORY_CAPTURE);
-
- // Create the sink filte used for receiving Captured frames.
- _sinkFilter = new CaptureSinkFilter(SINK_FILTER_NAME, NULL, &hr,
- *this, _id);
- if (hr != S_OK)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to create send filter");
- return -1;
- }
- _sinkFilter->AddRef();
-
- hr = _graphBuilder->AddFilter(_sinkFilter, SINK_FILTER_NAME);
- if (FAILED(hr))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to add the send filter to the graph.");
- return -1;
- }
- _inputSendPin = GetInputPin(_sinkFilter);
-
- // Temporary connect here.
- // This is done so that no one else can use the capture device.
- if (SetCameraOutput(_requestedCapability) != 0)
- {
- return -1;
- }
- hr = _mediaControl->Pause();
- if (FAILED(hr))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to Pause the Capture device. Is it already occupied? %d.",
- hr);
- return -1;
- }
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCapture, _id,
- "Capture device '%s' initialized.", deviceUniqueIdUTF8);
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureDS::StartCapture(
- const VideoCaptureCapability& capability)
-{
- CriticalSectionScoped cs(_apiCs);
-
- if (capability != _requestedCapability)
- {
- DisconnectGraph();
-
- if (SetCameraOutput(capability) != 0)
- {
- return -1;
- }
- }
- HRESULT hr = _mediaControl->Run();
- if (FAILED(hr))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to start the Capture device.");
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureDS::StopCapture()
-{
- CriticalSectionScoped cs(_apiCs);
-
- HRESULT hr = _mediaControl->Pause();
- if (FAILED(hr))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to stop the capture graph. %d", hr);
- return -1;
- }
- return 0;
-}
-bool VideoCaptureDS::CaptureStarted()
-{
- OAFilterState state = 0;
- HRESULT hr = _mediaControl->GetState(1000, &state);
- if (hr != S_OK && hr != VFW_S_CANT_CUE)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to get the CaptureStarted status");
- }
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "CaptureStarted %d", state);
- return state == State_Running;
-
-}
-WebRtc_Word32 VideoCaptureDS::CaptureSettings(
- VideoCaptureCapability& settings)
-{
- settings = _requestedCapability;
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureDS::SetCameraOutput(
- const VideoCaptureCapability& requestedCapability)
-{
-
- // Get the best matching capability
- VideoCaptureCapability capability;
- WebRtc_Word32 capabilityIndex;
-
- // Store the new requested size
- _requestedCapability = requestedCapability;
- // Match the requested capability with the supported.
- if ((capabilityIndex = _dsInfo.GetBestMatchedCapability(_deviceUniqueId,
- _requestedCapability,
- capability)) < 0)
- {
- return -1;
- }
- //Reduce the frame rate if possible.
- if (capability.maxFPS > requestedCapability.maxFPS)
- {
- capability.maxFPS = requestedCapability.maxFPS;
- }
- // Store the new expected capture delay
- _captureDelay = capability.expectedCaptureDelay;
-
- // Convert it to the windows capability index since they are not nexessary
- // the same
- VideoCaptureCapabilityWindows windowsCapability;
- if (_dsInfo.GetWindowsCapability(capabilityIndex, windowsCapability) != 0)
- {
- return -1;
- }
-
- IAMStreamConfig* streamConfig = NULL;
- AM_MEDIA_TYPE *pmt = NULL;
- VIDEO_STREAM_CONFIG_CAPS caps;
-
- HRESULT hr = _outputCapturePin->QueryInterface(IID_IAMStreamConfig,
- (void**) &streamConfig);
- if (hr)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Can't get the Capture format settings.");
- return -1;
- }
-
- //Get the windows capability from the capture device
- bool isDVCamera = false;
- hr = streamConfig->GetStreamCaps(
- windowsCapability.directShowCapabilityIndex,
- &pmt, reinterpret_cast<BYTE*> (&caps));
- if (!FAILED(hr))
- {
- if (pmt->formattype == FORMAT_VideoInfo2)
- {
- VIDEOINFOHEADER2* h =
- reinterpret_cast<VIDEOINFOHEADER2*> (pmt->pbFormat);
- if (capability.maxFPS > 0
- && windowsCapability.supportFrameRateControl)
- {
- h->AvgTimePerFrame = REFERENCE_TIME(10000000.0
- / capability.maxFPS);
- }
- }
- else
- {
- VIDEOINFOHEADER* h = reinterpret_cast<VIDEOINFOHEADER*>
- (pmt->pbFormat);
- if (capability.maxFPS > 0
- && windowsCapability.supportFrameRateControl)
- {
- h->AvgTimePerFrame = REFERENCE_TIME(10000000.0
- / capability.maxFPS);
- }
-
- }
-
- // Set the sink filter to request this capability
- _sinkFilter->SetMatchingMediaType(capability);
- //Order the capture device to use this capability
- hr += streamConfig->SetFormat(pmt);
-
- //Check if this is a DV camera and we need to add MS DV Filter
- if (pmt->subtype == MEDIASUBTYPE_dvsl
- || pmt->subtype == MEDIASUBTYPE_dvsd
- || pmt->subtype == MEDIASUBTYPE_dvhd)
- isDVCamera = true; // This is a DV camera. Use MS DV filter
- }
- RELEASE_AND_CLEAR(streamConfig);
-
- if (FAILED(hr))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to set capture device output format");
- return -1;
- }
-
- if (isDVCamera)
- {
- hr = ConnectDVCamera();
- }
- else
- {
- hr = _graphBuilder->ConnectDirect(_outputCapturePin, _inputSendPin,
- NULL);
- }
- if (hr != S_OK)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to connect the Capture graph %d", hr);
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureDS::DisconnectGraph()
-{
- HRESULT hr = _mediaControl->Stop();
- hr += _graphBuilder->Disconnect(_outputCapturePin);
- hr += _graphBuilder->Disconnect(_inputSendPin);
-
- //if the DV camera filter exist
- if (_dvFilter)
- {
- _graphBuilder->Disconnect(_inputDvPin);
- _graphBuilder->Disconnect(_outputDvPin);
- }
- if (hr != S_OK)
- {
- WEBRTC_TRACE( webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to Stop the Capture device for reconfiguration %d",
- hr);
- return -1;
- }
- return 0;
-}
-HRESULT VideoCaptureDS::ConnectDVCamera()
-{
- HRESULT hr = S_OK;
-
- if (!_dvFilter)
- {
- hr = CoCreateInstance(CLSID_DVVideoCodec, NULL, CLSCTX_INPROC,
- IID_IBaseFilter, (void **) &_dvFilter);
- if (hr != S_OK)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to create the dv decoder: %x", hr);
- return hr;
- }
- hr = _graphBuilder->AddFilter(_dvFilter, L"VideoDecoderDV");
- if (hr != S_OK)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to add the dv decoder to the graph: %x", hr);
- return hr;
- }
- _inputDvPin = GetInputPin(_dvFilter);
- if (_inputDvPin == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to get input pin from DV decoder");
- return -1;
- }
- _outputDvPin = GetOutputPin(_dvFilter);
- if (_outputDvPin == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to get output pin from DV decoder");
- return -1;
- }
- }
- hr = _graphBuilder->ConnectDirect(_outputCapturePin, _inputDvPin, NULL);
- if (hr != S_OK)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to connect capture device to the dv devoder: %x",
- hr);
- return hr;
- }
-
- hr = _graphBuilder->ConnectDirect(_outputDvPin, _inputSendPin, NULL);
- if (hr != S_OK)
- {
- if (hr == 0x80070004)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to connect the capture device, busy");
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to connect capture device to the send graph: 0x%x",
- hr);
- }
- return hr;
- }
- return hr;
-}
-} // namespace videocapturemodule
-} //namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/Windows/video_capture_windows.h b/trunk/src/modules/video_capture/main/source/Windows/video_capture_windows.h
deleted file mode 100644
index 69e2db5..0000000
--- a/trunk/src/modules/video_capture/main/source/Windows/video_capture_windows.h
+++ /dev/null
@@ -1,86 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_VIDEO_CAPTURE_WINDOWS_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_VIDEO_CAPTURE_WINDOWS_H_
-
-#include "../video_capture_impl.h"
-#include <tchar.h>
-
-#include "device_info_windows.h"
-
-#define CAPTURE_FILTER_NAME L"VideoCaptureFilter"
-#define SINK_FILTER_NAME L"SinkFilter"
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-// Forward declaraion
-class CaptureSinkFilter;
-
-class VideoCaptureDS: public VideoCaptureImpl
-{
-public:
-
- static VideoCaptureModule* Create(const WebRtc_Word32 id,
- const WebRtc_UWord8* deviceUniqueIdUTF8);
-
- VideoCaptureDS(const WebRtc_Word32 id);
-
- virtual WebRtc_Word32 Init(const WebRtc_Word32 id,
- const WebRtc_UWord8* deviceUniqueIdUTF8);
-
- /*************************************************************************
- *
- * Start/Stop
- *
- *************************************************************************/
- virtual WebRtc_Word32
- StartCapture(const VideoCaptureCapability& capability);
- virtual WebRtc_Word32 StopCapture();
-
- /**************************************************************************
- *
- * Properties of the set device
- *
- **************************************************************************/
-
- virtual bool CaptureStarted();
- virtual WebRtc_Word32 CaptureSettings(VideoCaptureCapability& settings);
-
-protected:
- virtual ~VideoCaptureDS();
-
- // Help functions
-
- WebRtc_Word32
- SetCameraOutput(const VideoCaptureCapability& requestedCapability);
- WebRtc_Word32 DisconnectGraph();
- HRESULT VideoCaptureDS::ConnectDVCamera();
-
- DeviceInfoWindows _dsInfo;
-
- IBaseFilter* _captureFilter;
- IGraphBuilder* _graphBuilder;
- IMediaControl* _mediaControl;
- CaptureSinkFilter* _sinkFilter;
- IPin* _inputSendPin;
- IPin* _outputCapturePin;
-
- // Microsoft DV interface (external DV cameras)
- IBaseFilter* _dvFilter;
- IPin* _inputDvPin;
- IPin* _outputDvPin;
-
-};
-} // namespace videocapturemodule
-} //namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_WINDOWS_VIDEO_CAPTURE_WINDOWS_H_
diff --git a/trunk/src/modules/video_capture/main/source/device_info_impl.cc b/trunk/src/modules/video_capture/main/source/device_info_impl.cc
deleted file mode 100644
index e41e0bb..0000000
--- a/trunk/src/modules/video_capture/main/source/device_info_impl.cc
+++ /dev/null
@@ -1,400 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "device_info_impl.h"
-#include "video_capture_config.h"
-#include "trace.h"
-#include <stdlib.h>
-
-#ifndef abs
-#define abs(a) (a>=0?a:-a)
-#endif
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-DeviceInfoImpl::DeviceInfoImpl(const WebRtc_Word32 id)
- : _id(id), _apiLock(*RWLockWrapper::CreateRWLock()), _lastUsedDeviceName(NULL),
- _lastUsedDeviceNameLength(0)
-{
-}
-
-DeviceInfoImpl::~DeviceInfoImpl(void)
-{
- _apiLock.AcquireLockExclusive();
- // Reset old capability list
- MapItem* item = NULL;
- while ((item = _captureCapabilities.Last()))
- {
- delete (VideoCaptureCapability*) item->GetItem();
- _captureCapabilities.Erase(item);
- }
- free(_lastUsedDeviceName);
- _apiLock.ReleaseLockExclusive();
-
- delete &_apiLock;
-}
-WebRtc_Word32 DeviceInfoImpl::NumberOfCapabilities(
- const WebRtc_UWord8* deviceUniqueIdUTF8)
-{
-
- if (!deviceUniqueIdUTF8)
- return -1;
-
- _apiLock.AcquireLockShared();
-
- if (_lastUsedDeviceNameLength == strlen((char*) deviceUniqueIdUTF8))
- {
- // Is it the same device that is asked for again.
-#if defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_LINUX)
- if(strncasecmp((char*)_lastUsedDeviceName,
- (char*) deviceUniqueIdUTF8,
- _lastUsedDeviceNameLength)==0)
-#else
- if (_strnicmp((char*) _lastUsedDeviceName,
- (char*) deviceUniqueIdUTF8,
- _lastUsedDeviceNameLength) == 0)
-#endif
- {
- //yes
- _apiLock.ReleaseLockShared();
- return _captureCapabilities.Size();
- }
- }
- // Need to get exclusive rights to create the new capability map.
- _apiLock.ReleaseLockShared();
- WriteLockScoped cs2(_apiLock);
-
- WebRtc_Word32 ret = CreateCapabilityMap(deviceUniqueIdUTF8);
- return ret;
-}
-
-WebRtc_Word32 DeviceInfoImpl::GetCapability(const WebRtc_UWord8* deviceUniqueIdUTF8,
- const WebRtc_UWord32 deviceCapabilityNumber,
- VideoCaptureCapability& capability)
-{
-
- if (!deviceUniqueIdUTF8)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "deviceUniqueIdUTF8 parameter not set in call to GetCapability");
- return -1;
- }
- ReadLockScoped cs(_apiLock);
-
- if ((_lastUsedDeviceNameLength != strlen((char*) deviceUniqueIdUTF8))
-#if defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_LINUX)
- || (strncasecmp((char*)_lastUsedDeviceName,
- (char*) deviceUniqueIdUTF8,
- _lastUsedDeviceNameLength)!=0))
-#else
- || (_strnicmp((char*) _lastUsedDeviceName,
- (char*) deviceUniqueIdUTF8,
- _lastUsedDeviceNameLength) != 0))
-#endif
-
- {
- _apiLock.ReleaseLockShared();
- _apiLock.AcquireLockExclusive();
- if (-1 == CreateCapabilityMap(deviceUniqueIdUTF8))
- {
- _apiLock.ReleaseLockExclusive();
- _apiLock.AcquireLockShared();
- return -1;
- }
- _apiLock.ReleaseLockExclusive();
- _apiLock.AcquireLockShared();
- }
-
- // Make sure the number is valid
- if (deviceCapabilityNumber >= (unsigned int) _captureCapabilities.Size())
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "deviceCapabilityNumber %d is invalid in call to GetCapability",
- deviceCapabilityNumber);
- return -1;
- }
-
- MapItem* item = _captureCapabilities.Find(deviceCapabilityNumber);
- if (!item)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to find capability number %d of %d possible",
- deviceCapabilityNumber, _captureCapabilities.Size());
- return -1;
- }
-
- VideoCaptureCapability* capPointer = static_cast<VideoCaptureCapability*>
- (item->GetItem());
- if (!capPointer)
- {
- return -1;
- }
-
- capability = *capPointer;
- return 0;
-}
-
-WebRtc_Word32 DeviceInfoImpl::GetBestMatchedCapability(
- const WebRtc_UWord8*deviceUniqueIdUTF8,
- const VideoCaptureCapability& requested,
- VideoCaptureCapability& resulting)
-{
-
-
- if (!deviceUniqueIdUTF8)
- return -1;
-
- ReadLockScoped cs(_apiLock);
- if ((_lastUsedDeviceNameLength != strlen((char*) deviceUniqueIdUTF8))
-#if defined(WEBRTC_MAC_INTEL) || defined(WEBRTC_LINUX)
- || (strncasecmp((char*)_lastUsedDeviceName,
- (char*) deviceUniqueIdUTF8,
- _lastUsedDeviceNameLength)!=0))
-#else
- || (_strnicmp((char*) _lastUsedDeviceName,
- (char*) deviceUniqueIdUTF8,
- _lastUsedDeviceNameLength) != 0))
-#endif
- {
- _apiLock.ReleaseLockShared();
- _apiLock.AcquireLockExclusive();
- if (-1 == CreateCapabilityMap(deviceUniqueIdUTF8))
- {
- return -1;
- }
- _apiLock.ReleaseLockExclusive();
- _apiLock.AcquireLockShared();
- }
-
- WebRtc_Word32 bestformatIndex = -1;
- WebRtc_Word32 bestWidth = 0;
- WebRtc_Word32 bestHeight = 0;
- WebRtc_Word32 bestFrameRate = 0;
- RawVideoType bestRawType = kVideoUnknown;
- webrtc::VideoCodecType bestCodecType = webrtc::kVideoCodecUnknown;
-
- const WebRtc_Word32 numberOfCapabilies = _captureCapabilities.Size();
-
- for (WebRtc_Word32 tmp = 0; tmp < numberOfCapabilies; ++tmp) // Loop through all capabilities
- {
- MapItem* item = _captureCapabilities.Find(tmp);
- if (!item)
- return -1;
-
- VideoCaptureCapability& capability = *static_cast<VideoCaptureCapability*>
- (item->GetItem());
-
- const WebRtc_Word32 diffWidth = capability.width - requested.width;
- const WebRtc_Word32 diffHeight = capability.height - requested.height;
- const WebRtc_Word32 diffFrameRate = capability.maxFPS - requested.maxFPS;
-
- const WebRtc_Word32 currentbestDiffWith = bestWidth - requested.width;
- const WebRtc_Word32 currentbestDiffHeight = bestHeight - requested.height;
- const WebRtc_Word32 currentbestDiffFrameRate = bestFrameRate - requested.maxFPS;
-
- if ((diffHeight >= 0 && diffHeight <= abs(currentbestDiffHeight)) // Height better or equalt that previouse.
- || (currentbestDiffHeight < 0 && diffHeight >= currentbestDiffHeight))
- {
-
- if (diffHeight == currentbestDiffHeight) // Found best height. Care about the width)
- {
- if ((diffWidth >= 0 && diffWidth <= abs(currentbestDiffWith)) // Width better or equal
- || (currentbestDiffWith < 0 && diffWidth >= currentbestDiffWith))
- {
- if (diffWidth == currentbestDiffWith && diffHeight
- == currentbestDiffHeight) // Same size as previously
- {
- //Also check the best frame rate if the diff is the same as previouse
- if (((diffFrameRate >= 0 &&
- diffFrameRate <= currentbestDiffFrameRate) // Frame rate to high but better match than previouse and we have not selected IUV
- ||
- (currentbestDiffFrameRate < 0 &&
- diffFrameRate >= currentbestDiffFrameRate)) // Current frame rate is lower than requested. This is better.
- )
- {
- if ((currentbestDiffFrameRate == diffFrameRate) // Same frame rate as previous or frame rate allready good enough
- || (currentbestDiffFrameRate >= 0))
- {
- if (bestRawType != requested.rawType
- && requested.rawType != kVideoUnknown
- && (capability.rawType == requested.rawType
- || capability.rawType == kVideoI420
- || capability.rawType == kVideoYUY2
- || capability.rawType == kVideoYV12))
- {
- bestCodecType = capability.codecType;
- bestRawType = capability.rawType;
- bestformatIndex = tmp;
- }
- // If width height and frame rate is full filled we can use the camera for encoding if it is supported.
- if (capability.height == requested.height
- && capability.width == requested.width
- && capability.maxFPS >= requested.maxFPS)
- {
- if (capability.codecType == requested.codecType
- && bestCodecType != requested.codecType)
- {
- bestCodecType = capability.codecType;
- bestformatIndex = tmp;
- }
- }
- }
- else // Better frame rate
- {
- if (requested.codecType == capability.codecType)
- {
-
- bestWidth = capability.width;
- bestHeight = capability.height;
- bestFrameRate = capability.maxFPS;
- bestCodecType = capability.codecType;
- bestRawType = capability.rawType;
- bestformatIndex = tmp;
- }
- }
- }
- }
- else // Better width than previously
- {
- if (requested.codecType == capability.codecType)
- {
- bestWidth = capability.width;
- bestHeight = capability.height;
- bestFrameRate = capability.maxFPS;
- bestCodecType = capability.codecType;
- bestRawType = capability.rawType;
- bestformatIndex = tmp;
- }
- }
- }// else width no good
- }
- else // Better height
- {
- if (requested.codecType == capability.codecType)
- {
- bestWidth = capability.width;
- bestHeight = capability.height;
- bestFrameRate = capability.maxFPS;
- bestCodecType = capability.codecType;
- bestRawType = capability.rawType;
- bestformatIndex = tmp;
- }
- }
- }// else height not good
- }//end for
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideoCapture, _id,
- "Best camera format: Width %d, Height %d, Frame rate %d, Color format %d",
- bestWidth, bestHeight, bestFrameRate, bestRawType);
-
- // Copy the capability
- MapItem* item = _captureCapabilities.Find(bestformatIndex);
- if (!item)
- return -1;
- VideoCaptureCapability* capPointer =
- static_cast<VideoCaptureCapability*> (item->GetItem());
- if (!capPointer)
- return -1;
-
- resulting = *capPointer;
-
- return bestformatIndex;
-}
-
-/* Returns the expected Capture delay*/
-WebRtc_Word32 DeviceInfoImpl::GetExpectedCaptureDelay(
- const DelayValues delayValues[],
- const WebRtc_UWord32 sizeOfDelayValues,
- const WebRtc_UWord8* productId,
- const WebRtc_UWord32 width,
- const WebRtc_UWord32 height)
-{
- WebRtc_Word32 bestDelay = kDefaultCaptureDelay;
-
- for (WebRtc_UWord32 device = 0; device < sizeOfDelayValues; ++device)
- {
- if (delayValues[device].productId && strncmp((char*) productId,
- (char*) delayValues[device].productId,
- kVideoCaptureProductIdLength) == 0)
- {
- // We have found the camera
-
- WebRtc_Word32 bestWidth = 0;
- WebRtc_Word32 bestHeight = 0;
-
- //Loop through all tested sizes and find one that seems fitting
- for (WebRtc_UWord32 delayIndex = 0; delayIndex < NoOfDelayValues; ++delayIndex)
- {
- const DelayValue& currentValue = delayValues[device].delayValues[delayIndex];
-
- const WebRtc_Word32 diffWidth = currentValue.width - width;
- const WebRtc_Word32 diffHeight = currentValue.height - height;
-
- const WebRtc_Word32 currentbestDiffWith = bestWidth - width;
- const WebRtc_Word32 currentbestDiffHeight = bestHeight - height;
-
- if ((diffHeight >= 0 && diffHeight <= abs(currentbestDiffHeight)) // Height better or equal than previous.
- || (currentbestDiffHeight < 0 && diffHeight >= currentbestDiffHeight))
- {
-
- if (diffHeight == currentbestDiffHeight) // Found best height. Care about the width)
- {
- if ((diffWidth >= 0 && diffWidth <= abs(currentbestDiffWith)) // Width better or equal
- || (currentbestDiffWith < 0 && diffWidth >= currentbestDiffWith))
- {
- if (diffWidth == currentbestDiffWith && diffHeight
- == currentbestDiffHeight) // Same size as previous
- {
- }
- else // Better width than previously
- {
- bestWidth = currentValue.width;
- bestHeight = currentValue.height;
- bestDelay = currentValue.delay;
- }
- }// else width no good
- }
- else // Better height
- {
- bestWidth = currentValue.width;
- bestHeight = currentValue.height;
- bestDelay = currentValue.delay;
- }
- }// else height not good
- }//end for
- break;
- }
- }
- if (bestDelay > kMaxCaptureDelay)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
- "Expected capture delay too high. %dms, will use %d", bestDelay,
- kMaxCaptureDelay);
- bestDelay = kMaxCaptureDelay;
-
- }
-
- return bestDelay;
-
-}
-
-//Default implementation. This should be overridden by Mobile implementations.
-WebRtc_Word32 DeviceInfoImpl::GetOrientation(const WebRtc_UWord8* deviceUniqueIdUTF8,
- VideoCaptureRotation& orientation)
-{
- orientation = kCameraRotate0;
- return -1;
-}
-} //namespace videocapturemodule
-} // namespace webrtc
-
-
diff --git a/trunk/src/modules/video_capture/main/source/device_info_impl.h b/trunk/src/modules/video_capture/main/source/device_info_impl.h
deleted file mode 100644
index 34ef65d..0000000
--- a/trunk/src/modules/video_capture/main/source/device_info_impl.h
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_DEVICE_INFO_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_DEVICE_INFO_IMPL_H_
-
-#include "video_capture.h"
-
-#include "map_wrapper.h"
-#include "rw_lock_wrapper.h"
-#include "video_capture_delay.h"
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-class DeviceInfoImpl: public VideoCaptureModule::DeviceInfo
-{
-public:
- DeviceInfoImpl(const WebRtc_Word32 id);
- virtual ~DeviceInfoImpl(void);
- virtual WebRtc_Word32 NumberOfCapabilities(const WebRtc_UWord8* deviceUniqueIdUTF8);
- virtual WebRtc_Word32 GetCapability(const WebRtc_UWord8* deviceUniqueIdUTF8,
- const WebRtc_UWord32 deviceCapabilityNumber,
- VideoCaptureCapability& capability);
-
- virtual WebRtc_Word32 GetBestMatchedCapability(const WebRtc_UWord8*deviceUniqueIdUTF8,
- const VideoCaptureCapability& requested,
- VideoCaptureCapability& resulting);
- virtual WebRtc_Word32 GetOrientation(const WebRtc_UWord8* deviceUniqueIdUTF8,
- VideoCaptureRotation& orientation);
-
-protected:
- /* Initialize this object*/
-
- virtual WebRtc_Word32 Init()=0;
- /*
- * Fills the member variable _captureCapabilities with capabilities for the given device name.
- */
- virtual WebRtc_Word32 CreateCapabilityMap(const WebRtc_UWord8* deviceUniqueIdUTF8)=0;
-
- /* Returns the expected Capture delay*/
- WebRtc_Word32 GetExpectedCaptureDelay(const DelayValues delayValues[],
- const WebRtc_UWord32 sizeOfDelayValues,
- const WebRtc_UWord8* productId,
- const WebRtc_UWord32 width,
- const WebRtc_UWord32 height);
-protected:
- // Data members
- WebRtc_Word32 _id;
- MapWrapper _captureCapabilities;
- RWLockWrapper& _apiLock;
- WebRtc_UWord8* _lastUsedDeviceName;
- WebRtc_UWord32 _lastUsedDeviceNameLength;
-};
-} //namespace videocapturemodule
-} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_DEVICE_INFO_IMPL_H_
diff --git a/trunk/src/modules/video_capture/main/source/video_capture.gypi b/trunk/src/modules/video_capture/main/source/video_capture.gypi
deleted file mode 100644
index de696ae..0000000
--- a/trunk/src/modules/video_capture/main/source/video_capture.gypi
+++ /dev/null
@@ -1,259 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'video_capture_module',
- 'type': '<(library)',
- 'dependencies': [
- 'webrtc_utility',
- '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '../interface',
- '../../../interface',
- '<(webrtc_root)/common_video/libyuv/include',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- '../../../interface',
- '<(webrtc_root)/common_video/libyuv/include',
- ],
- },
- 'sources': [
- # interfaces
- '../interface/video_capture.h',
- '../interface/video_capture_defines.h',
- '../interface/video_capture_factory.h',
- # headers
- 'video_capture_config.h',
- 'video_capture_delay.h',
- 'video_capture_impl.h',
- 'device_info_impl.h',
-
- # DEFINE PLATFORM INDEPENDENT SOURCE FILES
- 'video_capture_factory.cc',
- 'video_capture_impl.cc',
- 'device_info_impl.cc',
- ],
- 'conditions': [
- ['include_internal_video_capture==0', {
- 'sources': [
- 'External/device_info_external.cc',
- 'External/video_capture_external.cc',
- ],
- },{ # include_internal_video_capture == 1
- 'conditions': [
- # DEFINE PLATFORM SPECIFIC SOURCE FILES
- ['OS=="linux"', {
- 'include_dirs': [
- 'Linux',
- ],
- 'sources': [
- 'Linux/device_info_linux.h',
- 'Linux/video_capture_linux.h',
- 'Linux/device_info_linux.cc',
- 'Linux/video_capture_linux.cc',
- ],
- }], # linux
- ['OS=="mac"', {
- 'sources': [
- 'Mac/QTKit/video_capture_recursive_lock.h',
- 'Mac/QTKit/video_capture_qtkit.h',
- 'Mac/QTKit/video_capture_qtkit_info.h',
- 'Mac/QTKit/video_capture_qtkit_info_objc.h',
- 'Mac/QTKit/video_capture_qtkit_objc.h',
- 'Mac/QTKit/video_capture_qtkit_utility.h',
- 'Mac/video_capture_mac.mm',
- 'Mac/QTKit/video_capture_qtkit.mm',
- 'Mac/QTKit/video_capture_qtkit_objc.mm',
- 'Mac/QTKit/video_capture_recursive_lock.mm',
- 'Mac/QTKit/video_capture_qtkit_info.mm',
- 'Mac/QTKit/video_capture_qtkit_info_objc.mm',
- ],
- 'include_dirs': [
- 'Mac',
- ],
- 'link_settings': {
- 'xcode_settings': {
- 'OTHER_LDFLAGS': [
- '-framework QTKit',
- ],
- },
- },
- }], # mac
- ['OS=="win"', {
- 'include_dirs': [
- 'Windows',
- '<(direct_show_base_classes)',
- ],
- 'defines!': [
- 'NOMINMAX',
- ],
- 'sources': [
- 'Windows/help_functions_windows.h',
- 'Windows/sink_filter_windows.h',
- 'Windows/video_capture_windows.h',
- 'Windows/device_info_windows.h',
- 'Windows/capture_delay_values_windows.h',
- 'Windows/help_functions_windows.cc',
- 'Windows/sink_filter_windows.cc',
- 'Windows/video_capture_windows.cc',
- 'Windows/device_info_windows.cc',
- 'Windows/video_capture_factory_windows.cc',
- '<(direct_show_base_classes)amextra.cpp',
- '<(direct_show_base_classes)amextra.h',
- '<(direct_show_base_classes)amfilter.cpp',
- '<(direct_show_base_classes)amfilter.h',
- '<(direct_show_base_classes)amvideo.cpp',
- '<(direct_show_base_classes)arithutil.cpp',
- '<(direct_show_base_classes)cache.h',
- '<(direct_show_base_classes)checkbmi.h',
- '<(direct_show_base_classes)combase.cpp',
- '<(direct_show_base_classes)combase.h',
- '<(direct_show_base_classes)cprop.cpp',
- '<(direct_show_base_classes)cprop.h',
- '<(direct_show_base_classes)ctlutil.cpp',
- '<(direct_show_base_classes)ctlutil.h',
- '<(direct_show_base_classes)ddmm.cpp',
- '<(direct_show_base_classes)ddmm.h',
- '<(direct_show_base_classes)dllentry.cpp',
- '<(direct_show_base_classes)dllsetup.cpp',
- '<(direct_show_base_classes)dllsetup.h',
- '<(direct_show_base_classes)dxmperf.h',
- '<(direct_show_base_classes)fourcc.h',
- '<(direct_show_base_classes)measure.h',
- '<(direct_show_base_classes)msgthrd.h',
- '<(direct_show_base_classes)mtype.cpp',
- '<(direct_show_base_classes)mtype.h',
- '<(direct_show_base_classes)outputq.cpp',
- '<(direct_show_base_classes)outputq.h',
- '<(direct_show_base_classes)perflog.cpp',
- '<(direct_show_base_classes)perflog.h',
- '<(direct_show_base_classes)perfstruct.h',
- '<(direct_show_base_classes)pstream.cpp',
- '<(direct_show_base_classes)pstream.h',
- '<(direct_show_base_classes)pullpin.cpp',
- '<(direct_show_base_classes)pullpin.h',
- '<(direct_show_base_classes)refclock.cpp',
- '<(direct_show_base_classes)refclock.h',
- '<(direct_show_base_classes)reftime.h',
- '<(direct_show_base_classes)renbase.cpp',
- '<(direct_show_base_classes)renbase.h',
- '<(direct_show_base_classes)schedule.cpp',
- '<(direct_show_base_classes)schedule.h',
- '<(direct_show_base_classes)seekpt.cpp',
- '<(direct_show_base_classes)seekpt.h',
- '<(direct_show_base_classes)source.cpp',
- '<(direct_show_base_classes)source.h',
- '<(direct_show_base_classes)streams.h',
- '<(direct_show_base_classes)strmctl.cpp',
- '<(direct_show_base_classes)strmctl.h',
- '<(direct_show_base_classes)sysclock.cpp',
- '<(direct_show_base_classes)sysclock.h',
- '<(direct_show_base_classes)transfrm.cpp',
- '<(direct_show_base_classes)transfrm.h',
- '<(direct_show_base_classes)transip.cpp',
- '<(direct_show_base_classes)transip.h',
- '<(direct_show_base_classes)videoctl.cpp',
- '<(direct_show_base_classes)videoctl.h',
- '<(direct_show_base_classes)vtrans.cpp',
- '<(direct_show_base_classes)vtrans.h',
- '<(direct_show_base_classes)winctrl.cpp',
- '<(direct_show_base_classes)winctrl.h',
- '<(direct_show_base_classes)winutil.cpp',
- '<(direct_show_base_classes)winutil.h',
- '<(direct_show_base_classes)wxdebug.cpp',
- '<(direct_show_base_classes)wxdebug.h',
- '<(direct_show_base_classes)wxlist.cpp',
- '<(direct_show_base_classes)wxlist.h',
- '<(direct_show_base_classes)wxutil.cpp',
- '<(direct_show_base_classes)wxutil.h',
- ],
- 'msvs_settings': {
- 'VCLibrarianTool': {
- 'AdditionalDependencies': 'Strmiids.lib',
- },
- },
- }], # win
- ['OS=="android"', {
- 'include_dirs': [
- 'Android',
- ],
- 'sources': [
- 'Android/device_info_android.cc',
- 'Android/device_info_android.h',
- 'Android/video_capture_android.cc',
- 'Android/video_capture_android.h',
- ],
- }], # android
- ], # conditions
- }], # include_internal_video_capture
- ], # conditions
- },
- ],
- # Exclude the test targets when building with chromium.
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'video_capture_module_test',
- 'type': 'executable',
- 'dependencies': [
- 'video_capture_module',
- 'webrtc_utility',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- ],
- 'include_dirs': [
- '../interface',
- ],
- 'sources': [
- '../test/video_capture_unittest.cc',
- ],
- 'conditions': [
- # DEFINE PLATFORM SPECIFIC INCLUDE AND CFLAGS
- ['OS=="mac" or OS=="linux"', {
- 'cflags': [
- '-Wno-write-strings',
- ],
- 'ldflags': [
- '-lpthread -lm',
- ],
- }],
- ['OS=="linux"', {
- 'libraries': [
- '-lrt',
- '-lXext',
- '-lX11',
- ],
- }],
- ['OS=="mac"', {
- 'xcode_settings': {
- # TODO(andrew): CoreAudio and AudioToolbox shouldn't be needed.
- 'OTHER_LDFLAGS': [
- '-framework Foundation -framework AppKit -framework Cocoa -framework OpenGL -framework CoreVideo -framework CoreAudio -framework AudioToolbox',
- ],
- },
- }],
- ] # conditions
- },
- ],
- }],
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/video_capture/main/source/video_capture_config.h b/trunk/src/modules/video_capture/main/source/video_capture_config.h
deleted file mode 100644
index ab4010c..0000000
--- a/trunk/src/modules/video_capture/main/source/video_capture_config.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_CONFIG_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_CONFIG_H_
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-enum {kDefaultWidth = 640}; // Start width
-enum {kDefaultHeight = 480}; // Start heigt
-enum {kDefaultFrameRate = 30}; // Start frame rate
-
-enum {kMaxFrameRate =60}; // Max allowed frame rate of the start image
-
-enum {kDefaultCaptureDelay = 120};
-enum {kMaxCaptureDelay = 270}; // Max capture delay allowed in the precompiled capture delay values.
-
-enum {kProcessInterval = 300};
-enum {kFrameRateCallbackInterval = 1000};
-enum {kFrameRateCountHistorySize = 90};
-enum {kFrameRateHistoryWindowMs = 2000};
-} // namespace videocapturemodule
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_CONFIG_H_
diff --git a/trunk/src/modules/video_capture/main/source/video_capture_delay.h b/trunk/src/modules/video_capture/main/source/video_capture_delay.h
deleted file mode 100644
index 9f5b76e..0000000
--- a/trunk/src/modules/video_capture/main/source/video_capture_delay.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_DELAY_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_DELAY_H_
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-
-struct DelayValue
-{
- WebRtc_Word32 width;
- WebRtc_Word32 height;
- WebRtc_Word32 delay;
-};
-
-enum { NoOfDelayValues = 40 };
-struct DelayValues
-{
- char * deviceName;
- char* productId;
- DelayValue delayValues[NoOfDelayValues];
-};
-
-} //namespace videocapturemodule
-} //namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_DELAY_H_
diff --git a/trunk/src/modules/video_capture/main/source/video_capture_factory.cc b/trunk/src/modules/video_capture/main/source/video_capture_factory.cc
deleted file mode 100644
index ac0e641..0000000
--- a/trunk/src/modules/video_capture/main/source/video_capture_factory.cc
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_capture_factory.h"
-#include "video_capture_impl.h"
-#ifdef WEBRTC_ANDROID
-#include "video_capture_android.h"
-#endif
-
-namespace webrtc
-{
-
-VideoCaptureModule* VideoCaptureFactory::Create(const WebRtc_Word32 id,
- const WebRtc_UWord8* deviceUniqueIdUTF8) {
- return videocapturemodule::VideoCaptureImpl::Create(id, deviceUniqueIdUTF8);
-}
-
-VideoCaptureModule* VideoCaptureFactory::Create(const WebRtc_Word32 id,
- VideoCaptureExternal*& externalCapture) {
- return videocapturemodule::VideoCaptureImpl::Create(id, externalCapture);
-}
-
-VideoCaptureModule::DeviceInfo* VideoCaptureFactory::CreateDeviceInfo(
- const WebRtc_Word32 id) {
- return videocapturemodule::VideoCaptureImpl::CreateDeviceInfo(id);
-}
-
-#ifdef WEBRTC_ANDROID
-WebRtc_Word32 VideoCaptureFactory::SetAndroidObjects(void* javaVM,
- void* javaContext) {
- return videocapturemodule::VideoCaptureAndroid::SetAndroidObjects(javaVM,
- javaContext);
-}
-#endif
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/video_capture_impl.cc b/trunk/src/modules/video_capture/main/source/video_capture_impl.cc
deleted file mode 100644
index f976d10..0000000
--- a/trunk/src/modules/video_capture/main/source/video_capture_impl.cc
+++ /dev/null
@@ -1,501 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_capture_impl.h"
-
-#include "common_video/libyuv/include/libyuv.h"
-#include "critical_section_wrapper.h"
-#include "module_common_types.h"
-#include "ref_count.h"
-#include "tick_util.h"
-#include "trace.h"
-#include "video_capture_config.h"
-
-#ifdef WEBRTC_ANDROID
-#include "video_capture_android.h" // Need inclusion here to set Java environment.
-#endif
-
-namespace webrtc
-{
-namespace videocapturemodule
-{
-VideoCaptureModule* VideoCaptureImpl::Create(
- const WebRtc_Word32 id,
- VideoCaptureExternal*& externalCapture)
-{
- RefCountImpl<VideoCaptureImpl>* implementation =
- new RefCountImpl<VideoCaptureImpl>(id);
- externalCapture = implementation;
- return implementation;
-}
-
-const WebRtc_UWord8* VideoCaptureImpl::CurrentDeviceName() const
-{
- return _deviceUniqueId;
-}
-
-WebRtc_Word32 VideoCaptureImpl::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
- return 0;
-}
-
-// returns the number of milliseconds until the module want a worker thread to call Process
-WebRtc_Word32 VideoCaptureImpl::TimeUntilNextProcess()
-{
- TickTime timeNow = TickTime::Now();
-
- WebRtc_Word32 timeToNormalProcess = kProcessInterval
- - (WebRtc_Word32)((TickTime::Now() - _lastProcessTime).Milliseconds());
- WebRtc_Word32 timeToStartImage = timeToNormalProcess;
- if (_startImageFrameIntervall)
- {
- timeToStartImage = _startImageFrameIntervall
- - (WebRtc_Word32)((timeNow - _lastSentStartImageTime).Milliseconds());
- if (timeToStartImage < 0)
- {
- timeToStartImage = 0;
- }
- }
- return (timeToStartImage < timeToNormalProcess)
- ? timeToStartImage : timeToNormalProcess;
-}
-
-// Process any pending tasks such as timeouts
-WebRtc_Word32 VideoCaptureImpl::Process()
-{
- CriticalSectionScoped cs(_callBackCs);
-
- const TickTime now = TickTime::Now();
- _lastProcessTime = TickTime::Now();
-
- // Handle No picture alarm
-
- if (_lastProcessFrameCount.Ticks() == _incomingFrameTimes[0].Ticks() &&
- _captureAlarm != Raised)
- {
- if (_noPictureAlarmCallBack && _captureCallBack)
- {
- _captureAlarm = Raised;
- _captureCallBack->OnNoPictureAlarm(_id, _captureAlarm);
- }
- }
- else if (_lastProcessFrameCount.Ticks() != _incomingFrameTimes[0].Ticks() &&
- _captureAlarm != Cleared)
- {
- if (_noPictureAlarmCallBack && _captureCallBack)
- {
- _captureAlarm = Cleared;
- _captureCallBack->OnNoPictureAlarm(_id, _captureAlarm);
-
- }
- }
-
- // Handle frame rate callback
- if ((now - _lastFrameRateCallbackTime).Milliseconds()
- > kFrameRateCallbackInterval)
- {
- if (_frameRateCallBack && _captureCallBack)
- {
- const WebRtc_UWord32 frameRate = CalculateFrameRate(now);
- _captureCallBack->OnCaptureFrameRate(_id, frameRate);
- }
- _lastFrameRateCallbackTime = now; // Can be set by EnableFrameRateCallback
-
- }
-
- _lastProcessFrameCount = _incomingFrameTimes[0];
-
- // Handle start image frame rates.
- if (_startImageFrameIntervall
- && (now - _lastSentStartImageTime).Milliseconds() >= _startImageFrameIntervall)
- {
- _lastSentStartImageTime = now;
- if (_dataCallBack)
- {
- _captureFrame.CopyFrame(_startImage);
- _captureFrame.SetRenderTime(TickTime::MillisecondTimestamp());
- _dataCallBack->OnIncomingCapturedFrame(_id, _captureFrame,
- kVideoCodecUnknown);
- }
- }
- return 0;
-}
-
-VideoCaptureImpl::VideoCaptureImpl(const WebRtc_Word32 id)
- : _id(id), _deviceUniqueId(NULL), _apiCs(*CriticalSectionWrapper::CreateCriticalSection()),
- _captureDelay(0), _requestedCapability(),
- _callBackCs(*CriticalSectionWrapper::CreateCriticalSection()),
- _lastProcessTime(TickTime::Now()),
- _lastFrameRateCallbackTime(TickTime::Now()), _frameRateCallBack(false),
- _noPictureAlarmCallBack(false), _captureAlarm(Cleared), _setCaptureDelay(0),
- _dataCallBack(NULL), _captureCallBack(NULL),
- _startImage(), _startImageFrameIntervall(0),
- _lastSentStartImageTime(TickTime::Now()),
- _lastProcessFrameCount(TickTime::Now()), _rotateFrame(kRotateNone),
- last_capture_time_(TickTime::MillisecondTimestamp())
-
-{
- _requestedCapability.width = kDefaultWidth;
- _requestedCapability.height = kDefaultHeight;
- _requestedCapability.maxFPS = 30;
- _requestedCapability.rawType = kVideoI420;
- _requestedCapability.codecType = kVideoCodecUnknown;
- memset(_incomingFrameTimes, 0, sizeof(_incomingFrameTimes));
-}
-
-VideoCaptureImpl::~VideoCaptureImpl()
-{
- DeRegisterCaptureDataCallback();
- DeRegisterCaptureCallback();
- delete &_callBackCs;
- delete &_apiCs;
-
- if (_deviceUniqueId)
- delete[] _deviceUniqueId;
-}
-
-WebRtc_Word32 VideoCaptureImpl::RegisterCaptureDataCallback(
- VideoCaptureDataCallback& dataCallBack)
-{
- CriticalSectionScoped cs(_apiCs);
- CriticalSectionScoped cs2(_callBackCs);
- _dataCallBack = &dataCallBack;
-
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureImpl::DeRegisterCaptureDataCallback()
-{
- CriticalSectionScoped cs(_apiCs);
- CriticalSectionScoped cs2(_callBackCs);
- _dataCallBack = NULL;
- return 0;
-}
-WebRtc_Word32 VideoCaptureImpl::RegisterCaptureCallback(VideoCaptureFeedBack& callBack)
-{
-
- CriticalSectionScoped cs(_apiCs);
- CriticalSectionScoped cs2(_callBackCs);
- _captureCallBack = &callBack;
- return 0;
-}
-WebRtc_Word32 VideoCaptureImpl::DeRegisterCaptureCallback()
-{
-
- CriticalSectionScoped cs(_apiCs);
- CriticalSectionScoped cs2(_callBackCs);
- _captureCallBack = NULL;
- return 0;
-
-}
-WebRtc_Word32 VideoCaptureImpl::SetCaptureDelay(WebRtc_Word32 delayMS)
-{
- CriticalSectionScoped cs(_apiCs);
- _captureDelay = delayMS;
- return 0;
-}
-WebRtc_Word32 VideoCaptureImpl::CaptureDelay()
-{
- CriticalSectionScoped cs(_apiCs);
- return _setCaptureDelay;
-}
-
-WebRtc_Word32 VideoCaptureImpl::DeliverCapturedFrame(VideoFrame& captureFrame,
- WebRtc_Word32 width, WebRtc_Word32 height, WebRtc_Word64 capture_time,
- VideoCodecType codec_type) {
- UpdateFrameCount();// frame count used for local frame rate callback.
- _startImageFrameIntervall = 0; // prevent the start image to be displayed.
-
- const bool callOnCaptureDelayChanged = _setCaptureDelay != _captureDelay;
- // Capture delay changed
- if (_setCaptureDelay != _captureDelay) {
- _setCaptureDelay = _captureDelay;
- }
-
- // Set the capture time
- if (capture_time != 0) {
- captureFrame.SetRenderTime(capture_time);
- }
- else {
- captureFrame.SetRenderTime(TickTime::MillisecondTimestamp());
- }
-
- if (captureFrame.RenderTimeMs() == last_capture_time_) {
- // We don't allow the same capture time for two frames, drop this one.
- return -1;
- }
- last_capture_time_ = captureFrame.RenderTimeMs();
-
- captureFrame.SetHeight(height);
- captureFrame.SetWidth(width);
-
- if (_dataCallBack) {
- if (callOnCaptureDelayChanged) {
- _dataCallBack->OnCaptureDelayChanged(_id, _captureDelay);
- }
- _dataCallBack->OnIncomingCapturedFrame(_id, captureFrame, codec_type);
- }
-
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureImpl::IncomingFrame(
- WebRtc_UWord8* videoFrame,
- WebRtc_Word32 videoFrameLength,
- const VideoCaptureCapability& frameInfo,
- WebRtc_Word64 captureTime/*=0*/)
-{
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideoCapture, _id,
- "IncomingFrame width %d, height %d", (int) frameInfo.width,
- (int) frameInfo.height);
-
- TickTime startProcessTime = TickTime::Now();
-
- CriticalSectionScoped cs(_callBackCs);
-
- const WebRtc_Word32 width = frameInfo.width;
- const WebRtc_Word32 height = frameInfo.height;
-
- if (frameInfo.codecType == kVideoCodecUnknown)
- {
- // Not encoded, convert to I420.
- const VideoType commonVideoType =
- RawVideoTypeToCommonVideoVideoType(frameInfo.rawType);
-
- if (frameInfo.rawType != kVideoMJPEG &&
- CalcBufferSize(commonVideoType, width, height) != videoFrameLength)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Wrong incoming frame length.");
- return -1;
- }
-
- // Allocate I420 buffer.
- int requiredLength = CalcBufferSize(kI420, width, height);
- _captureFrame.VerifyAndAllocate(requiredLength);
- if (!_captureFrame.Buffer())
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to allocate frame buffer.");
- return -1;
- }
-
- memset(_captureFrame.Buffer(), 0, _captureFrame.Size());
- // Keeping stride = width for I420 destination.
- int dstStride = width;
- const int conversionResult = ConvertToI420(commonVideoType,
- videoFrame,
- 0, 0, // No cropping
- width, height,
- videoFrameLength,
- width, height, dstStride,
- _rotateFrame,
- _captureFrame.Buffer());
- if (conversionResult < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to convert capture frame from type %d to I420",
- frameInfo.rawType);
- return -1;
- }
- _captureFrame.SetLength(requiredLength);
- }
- else // Encoded format
- {
- if (_captureFrame.CopyFrame(videoFrameLength, videoFrame) != 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to copy captured frame of length %d", (int) videoFrameLength);
- }
- }
-
- DeliverCapturedFrame(_captureFrame, width, height, captureTime, frameInfo.codecType);
-
-
- const WebRtc_UWord32 processTime =
- (WebRtc_UWord32)(TickTime::Now() - startProcessTime).Milliseconds();
- if (processTime > 10) // If the process time is too long MJPG will not work well.
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCapture, _id,
- "Too long processing time of Incoming frame: %ums",
- (unsigned int) processTime);
- }
-
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureImpl::IncomingFrameI420(
- const VideoFrameI420& video_frame, WebRtc_Word64 captureTime) {
-
- CriticalSectionScoped cs(_callBackCs);
-
- // Allocate I420 buffer
- int frame_size = CalcBufferSize(kI420,
- video_frame.width,
- video_frame.height);
- _captureFrame.VerifyAndAllocate(frame_size);
- if (!_captureFrame.Buffer()) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "Failed to allocate frame buffer.");
- return -1;
- }
-
- // Copy planes to the _captureFrame
- int y_width = video_frame.width;
- int uv_width = video_frame.width / 2;
- int y_rows = video_frame.height;
- int uv_rows = video_frame.height / 2; // I420
- unsigned char* current_pointer = _captureFrame.Buffer();
- unsigned char* y_plane = video_frame.y_plane;
- unsigned char* u_plane = video_frame.u_plane;
- unsigned char* v_plane = video_frame.v_plane;
- // Copy Y
- for (int i = 0; i < y_rows; ++i) {
- memcpy(current_pointer, y_plane, y_width);
- current_pointer += video_frame.y_pitch;
- y_plane += video_frame.y_pitch;
- }
- // Copy U
- for (int i = 0; i < uv_rows; ++i) {
- memcpy(current_pointer, u_plane, uv_width);
- current_pointer += video_frame.u_pitch;
- u_plane += video_frame.u_pitch;
- }
- // Copy V
- for (int i = 0; i < uv_rows; ++i) {
- memcpy(current_pointer, v_plane, uv_width);
- current_pointer += video_frame.v_pitch;
- v_plane += video_frame.v_pitch;
- }
- _captureFrame.SetLength(frame_size);
-
- DeliverCapturedFrame(_captureFrame,
- video_frame.width,
- video_frame.height,
- captureTime,
- kVideoCodecUnknown);
-
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureImpl::SetCaptureRotation(VideoCaptureRotation rotation)
-{
- CriticalSectionScoped cs(_apiCs);
- CriticalSectionScoped cs2(_callBackCs);
- switch (rotation)
- {
- case kCameraRotate0:
- _rotateFrame = kRotateNone;
- break;
- case kCameraRotate90:
- _rotateFrame = kRotate90;
- break;
- case kCameraRotate180:
- _rotateFrame = kRotate180;
- break;
- case kCameraRotate270:
- _rotateFrame = kRotate270;
- break;
- }
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureImpl::StartSendImage(const VideoFrame& videoFrame,
- WebRtc_Word32 frameRate)
-{
- CriticalSectionScoped cs(_apiCs);
- CriticalSectionScoped cs2(_callBackCs);
- if (frameRate < 1 || frameRate > kMaxFrameRate)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCapture, _id,
- "StartSendImage Invalid parameter. frameRate %d", (int) frameRate);
- return -1;;
- }
- _startImage.CopyFrame(videoFrame);
- _startImageFrameIntervall = 1000 / frameRate;
- _lastSentStartImageTime = TickTime::Now();
- return 0;
-
-}
-WebRtc_Word32 VideoCaptureImpl::StopSendImage()
-{
- CriticalSectionScoped cs(_apiCs);
- CriticalSectionScoped cs2(_callBackCs);
- _startImageFrameIntervall = 0;
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureImpl::EnableFrameRateCallback(const bool enable)
-{
- CriticalSectionScoped cs(_apiCs);
- CriticalSectionScoped cs2(_callBackCs);
- _frameRateCallBack = enable;
- if (enable)
- {
- _lastFrameRateCallbackTime = TickTime::Now();
- }
- return 0;
-}
-
-WebRtc_Word32 VideoCaptureImpl::EnableNoPictureAlarm(const bool enable)
-{
- CriticalSectionScoped cs(_apiCs);
- CriticalSectionScoped cs2(_callBackCs);
- _noPictureAlarmCallBack = enable;
- return 0;
-}
-
-void VideoCaptureImpl::UpdateFrameCount()
-{
- if (_incomingFrameTimes[0].MicrosecondTimestamp() == 0)
- {
- // first no shift
- }
- else
- {
- // shift
- for (int i = (kFrameRateCountHistorySize - 2); i >= 0; i--)
- {
- _incomingFrameTimes[i + 1] = _incomingFrameTimes[i];
- }
- }
- _incomingFrameTimes[0] = TickTime::Now();
-}
-
-WebRtc_UWord32 VideoCaptureImpl::CalculateFrameRate(const TickTime& now)
-{
- WebRtc_Word32 num = 0;
- WebRtc_Word32 nrOfFrames = 0;
- for (num = 1; num < (kFrameRateCountHistorySize - 1); num++)
- {
- if (_incomingFrameTimes[num].Ticks() <= 0
- || (now - _incomingFrameTimes[num]).Milliseconds() > kFrameRateHistoryWindowMs) // don't use data older than 2sec
- {
- break;
- }
- else
- {
- nrOfFrames++;
- }
- }
- if (num > 1)
- {
- WebRtc_Word64 diff = (now - _incomingFrameTimes[num - 1]).Milliseconds();
- if (diff > 0)
- {
- return WebRtc_UWord32((nrOfFrames * 1000.0f / diff) + 0.5f);
- }
- }
-
- return nrOfFrames;
-}
-} // namespace videocapturemodule
-} // namespace webrtc
diff --git a/trunk/src/modules/video_capture/main/source/video_capture_impl.h b/trunk/src/modules/video_capture/main/source/video_capture_impl.h
deleted file mode 100644
index eae3f39..0000000
--- a/trunk/src/modules/video_capture/main/source/video_capture_impl.h
+++ /dev/null
@@ -1,143 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_IMPL_H_
-
-/*
- * video_capture_impl.h
- */
-
-#include "video_capture.h"
-#include "video_capture_config.h"
-#include "tick_util.h"
-#include "common_video/libyuv/include/libyuv.h"
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-
-namespace videocapturemodule {
-// Class definitions
-class VideoCaptureImpl: public VideoCaptureModule, public VideoCaptureExternal
-{
-public:
-
- /*
- * Create a video capture module object
- *
- * id - unique identifier of this video capture module object
- * deviceUniqueIdUTF8 - name of the device. Available names can be found by using GetDeviceName
- */
- static VideoCaptureModule* Create(const WebRtc_Word32 id,
- const WebRtc_UWord8* deviceUniqueIdUTF8);
-
- /*
- * Create a video capture module object used for external capture.
- *
- * id - unique identifier of this video capture module object
- * externalCapture - [out] interface to call when a new frame is captured.
- */
- static VideoCaptureModule* Create(const WebRtc_Word32 id,
- VideoCaptureExternal*& externalCapture);
-
- static DeviceInfo* CreateDeviceInfo(const WebRtc_Word32 id);
-
- // Implements Module declared functions.
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- //Call backs
- virtual WebRtc_Word32 RegisterCaptureDataCallback(VideoCaptureDataCallback& dataCallback);
- virtual WebRtc_Word32 DeRegisterCaptureDataCallback();
- virtual WebRtc_Word32 RegisterCaptureCallback(VideoCaptureFeedBack& callBack);
- virtual WebRtc_Word32 DeRegisterCaptureCallback();
-
- virtual WebRtc_Word32 StartSendImage(const VideoFrame& videoFrame,
- WebRtc_Word32 frameRate = 1);
- virtual WebRtc_Word32 StopSendImage();
-
- virtual WebRtc_Word32 SetCaptureDelay(WebRtc_Word32 delayMS);
- virtual WebRtc_Word32 CaptureDelay();
- virtual WebRtc_Word32 SetCaptureRotation(VideoCaptureRotation rotation);
-
- virtual WebRtc_Word32 EnableFrameRateCallback(const bool enable);
- virtual WebRtc_Word32 EnableNoPictureAlarm(const bool enable);
-
- virtual const WebRtc_UWord8* CurrentDeviceName() const;
-
- // Module handling
- virtual WebRtc_Word32 TimeUntilNextProcess();
- virtual WebRtc_Word32 Process();
-
- // Implement VideoCaptureExternal
- virtual WebRtc_Word32 IncomingFrame(WebRtc_UWord8* videoFrame,
- WebRtc_Word32 videoFrameLength,
- const VideoCaptureCapability& frameInfo,
- WebRtc_Word64 captureTime = 0);
- virtual WebRtc_Word32 IncomingFrameI420(
- const VideoFrameI420& video_frame,
- WebRtc_Word64 captureTime = 0);
-
- // Platform dependent
- virtual WebRtc_Word32 StartCapture(const VideoCaptureCapability& capability)
- {
- _requestedCapability = capability;
- return -1;
- }
- virtual WebRtc_Word32 StopCapture() { return -1; }
- virtual bool CaptureStarted() {return false; }
- virtual WebRtc_Word32 CaptureSettings(VideoCaptureCapability& /*settings*/)
- { return -1; }
- VideoCaptureEncodeInterface* GetEncodeInterface(const VideoCodec& /*codec*/)
- { return NULL; }
-
-protected:
- VideoCaptureImpl(const WebRtc_Word32 id);
- virtual ~VideoCaptureImpl();
- WebRtc_Word32 DeliverCapturedFrame(
- VideoFrame& captureFrame, WebRtc_Word32 width, WebRtc_Word32 height,
- WebRtc_Word64 capture_time, VideoCodecType codec_type);
-
- WebRtc_Word32 _id; // Module ID
- WebRtc_UWord8* _deviceUniqueId; // current Device unique name;
- CriticalSectionWrapper& _apiCs;
- WebRtc_Word32 _captureDelay; // Current capture delay. May be changed of platform dependent parts.
- VideoCaptureCapability _requestedCapability; // Should be set by platform dependent code in StartCapture.
-private:
- void UpdateFrameCount();
- WebRtc_UWord32 CalculateFrameRate(const TickTime& now);
-
- CriticalSectionWrapper& _callBackCs;
-
- TickTime _lastProcessTime; // last time the module process function was called.
- TickTime _lastFrameRateCallbackTime; // last time the frame rate callback function was called.
- bool _frameRateCallBack; // true if EnableFrameRateCallback
- bool _noPictureAlarmCallBack; //true if EnableNoPictureAlarm
- VideoCaptureAlarm _captureAlarm; // current value of the noPictureAlarm
-
- WebRtc_Word32 _setCaptureDelay; // The currently used capture delay
- VideoCaptureDataCallback* _dataCallBack;
- VideoCaptureFeedBack* _captureCallBack;
-
- VideoFrame _startImage;
- WebRtc_Word32 _startImageFrameIntervall;
- TickTime _lastSentStartImageTime; // last time the start image was sent
- TickTime _lastProcessFrameCount;
- TickTime _incomingFrameTimes[kFrameRateCountHistorySize];// timestamp for local captured frames
- VideoRotationMode _rotateFrame; //Set if the frame should be rotated by the capture module.
-
- VideoFrame _captureFrame;
-
- // Used to make sure incoming timestamp is increasing for every frame.
- WebRtc_Word64 last_capture_time_;
-};
-} // namespace videocapturemodule
-} //namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CAPTURE_MAIN_SOURCE_VIDEO_CAPTURE_IMPL_H_
diff --git a/trunk/src/modules/video_capture/main/test/android/.classpath b/trunk/src/modules/video_capture/main/test/android/.classpath
deleted file mode 100644
index 841ac51..0000000
--- a/trunk/src/modules/video_capture/main/test/android/.classpath
+++ /dev/null
@@ -1,9 +0,0 @@
-<?xml version="1.0" encoding="UTF-8"?>
-<classpath>
- <classpathentry kind="src" path="src"/>
- <classpathentry kind="src" path="renderer"/>
- <classpathentry kind="src" path="java"/>
- <classpathentry kind="src" path="gen"/>
- <classpathentry kind="con" path="com.android.ide.eclipse.adt.ANDROID_FRAMEWORK"/>
- <classpathentry kind="output" path="bin"/>
-</classpath>
diff --git a/trunk/src/modules/video_capture/main/test/android/AndroidManifest.xml b/trunk/src/modules/video_capture/main/test/android/AndroidManifest.xml
deleted file mode 100644
index 1a4ec9b..0000000
--- a/trunk/src/modules/video_capture/main/test/android/AndroidManifest.xml
+++ /dev/null
@@ -1,31 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<manifest xmlns:android="http://schemas.android.com/apk/res/android"
- package="org.webrtc.capturemoduleandroidtest"
- android:versionCode="1"
- android:versionName="1.0">
- <application android:icon="@drawable/icon"
- android:label="@string/app_name"
- android:debuggable="true">
- <activity android:label="@string/app_name"
- android:name="VideoCaptureModuleTest"
- android:configChanges="orientation|keyboardHidden"
- android:launchMode="singleTask"
- android:multiprocess="false">
- <intent-filter>
- <action android:name="android.intent.action.MAIN" />
- <category android:name="android.intent.category.LAUNCHER" />
- </intent-filter>
- </activity>
- </application>
-
- <uses-feature android:required="true"
- android:name="android.hardware.camera">
- </uses-feature>
-
- <uses-permission android:name="android.permission.CAMERA">
- </uses-permission>
- <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/>
- <uses-sdk android:targetSdkVersion="7"
- android:minSdkVersion="7">
- </uses-sdk>
-</manifest>
diff --git a/trunk/src/modules/video_capture/main/test/android/default.properties b/trunk/src/modules/video_capture/main/test/android/default.properties
deleted file mode 100644
index 2ad44a4..0000000
--- a/trunk/src/modules/video_capture/main/test/android/default.properties
+++ /dev/null
@@ -1,11 +0,0 @@
-# This file is automatically generated by Android Tools.
-# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
-#
-# This file must be checked in Version Control Systems.
-#
-# To customize properties used by the Ant build system use,
-# "build.properties", and override values to adapt the script to your
-# project structure.
-
-# Project target.
-target=android-9
diff --git a/trunk/src/modules/video_capture/main/test/android/gen/org/webrtc/capturemoduleandroidtest/R.java b/trunk/src/modules/video_capture/main/test/android/gen/org/webrtc/capturemoduleandroidtest/R.java
deleted file mode 100644
index 94bf93b..0000000
--- a/trunk/src/modules/video_capture/main/test/android/gen/org/webrtc/capturemoduleandroidtest/R.java
+++ /dev/null
@@ -1,33 +0,0 @@
-/* AUTO-GENERATED FILE. DO NOT MODIFY.
- *
- * This class was automatically generated by the
- * aapt tool from the resource data it found. It
- * should not be modified by hand.
- */
-
-package org.webrtc.capturemoduleandroidtest;
-
-public final class R {
- public static final class attr {
- }
- public static final class drawable {
- public static final int icon=0x7f020000;
- }
- public static final class id {
- public static final int Button01=0x7f050000;
- public static final int Button02=0x7f050001;
- public static final int Button03=0x7f050002;
- public static final int Button04=0x7f050003;
- public static final int renderView=0x7f050004;
- }
- public static final class layout {
- public static final int main=0x7f030000;
- }
- public static final class string {
- public static final int app_name=0x7f040000;
- public static final int run_button=0x7f040001;
- public static final int run_button2=0x7f040002;
- public static final int run_button3=0x7f040003;
- public static final int run_button4=0x7f040004;
- }
-}
diff --git a/trunk/src/modules/video_capture/main/test/android/jni/Android.mk b/trunk/src/modules/video_capture/main/test/android/jni/Android.mk
deleted file mode 100644
index 7eba756..0000000
--- a/trunk/src/modules/video_capture/main/test/android/jni/Android.mk
+++ /dev/null
@@ -1,19 +0,0 @@
-# Android makefile for VideoCapture Module
-
-LOCAL_PATH := $(call my-dir)
-
-WEBRTC_INTERFACES_PATH := $(LOCAL_PATH)/../../../../../../../build/interface
-WEBRTC_LIBS_PATH := $(LOCAL_PATH)/../../../../../../../build/libraries
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE := VideoCaptureModuleAndroidTestJniAPI
-LOCAL_SRC_FILES := video_capture_module_android_test_jni.cc
-LOCAL_CFLAGS := -DWEBRTC_TARGET_PC # For typedefs.h
-LOCAL_C_INCLUDES := $(WEBRTC_INTERFACES_PATH)
-LOCAL_LDLIBS := \
- $(WEBRTC_LIBS_PATH)/testVideoCaptureAPI_android_gcc.a \
- $(WEBRTC_LIBS_PATH)/VideoCaptureModuleTestApiLib_android_gcc.a \
- -llog -lgcc -lGLESv2
-include $(BUILD_SHARED_LIBRARY)
-
diff --git a/trunk/src/modules/video_capture/main/test/android/jni/org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest.h b/trunk/src/modules/video_capture/main/test/android/jni/org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest.h
deleted file mode 100644
index 0320df0..0000000
--- a/trunk/src/modules/video_capture/main/test/android/jni/org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/* DO NOT EDIT THIS FILE - it is machine generated */
-#include <jni.h>
-/* Header for class org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest */
-
-#ifndef _Included_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest
-#define _Included_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest
-#ifdef __cplusplus
-extern "C" {
-#endif
-/*
- * Class: org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest
- * Method: RunTest
- * Signature: (Landroid/content/Context;)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest_RunTest
- (JNIEnv *, jobject, jobject);
-
-JNIEXPORT jint JNICALL Java_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest_RenderInit
-(JNIEnv * env, jobject context,jobject surface);
-
-JNIEXPORT jint JNICALL Java_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest_StartCapture
-(JNIEnv *, jobject);
-
-JNIEXPORT jint JNICALL Java_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest_StopCapture
-(JNIEnv *, jobject);
-
-#ifdef __cplusplus
-}
-#endif
-#endif
diff --git a/trunk/src/modules/video_capture/main/test/android/jni/video_capture_module_android_test_jni.cc b/trunk/src/modules/video_capture/main/test/android/jni/video_capture_module_android_test_jni.cc
deleted file mode 100644
index ac2f6e1..0000000
--- a/trunk/src/modules/video_capture/main/test/android/jni/video_capture_module_android_test_jni.cc
+++ /dev/null
@@ -1,149 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string.h> // memset
-#include <android/log.h>
-
-#include "org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest.h"
-#include "../../../interface/video_capture_factory.h"
-#include "../../../../../video_render/main/interface/video_render.h"
-#include "../../testAPI/testPlatformDependent.h"
-#include "../../testAPI/testPlatformDependent.h"
-#ifdef RENDER_PREVIEW
-#include "../../testAPI/Renderer.h"
-#endif
-
-using namespace webrtc;
-#define WEBRTC_LOG_TAG "*WEBRTCN*" // As in WEBRTC Native...
-// ADM data struct
-typedef struct
-{
- // Other
- JavaVM* jvm;
- Renderer* renderer;
- VideoCaptureModule* _videoCapture;
- VideoCaptureModule::DeviceInfo*_captureInfo;
-} JniData;
-
-// Global variables visible in this file
-static JniData jniData;
-
-//////////////////////////////////////////////////////////////////
-// General functions
-//////////////////////////////////////////////////////////////////
-
-/////////////////////////////////////////////
-// JNI_OnLoad
-//
-jint JNI_OnLoad(JavaVM* vm, void* /*reserved*/)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "JNI_OnLoad");
- if (!vm)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "JNI_OnLoad did not receive a valid VM pointer");
- return -1;
- }
-
- // Get JNI
- JNIEnv* env;
- if (JNI_OK != vm->GetEnv(reinterpret_cast<void**> (&env), JNI_VERSION_1_4))
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "JNI_OnLoad could not get JNI env");
- return -1;
- }
-
- // Init JniData data
- memset(&jniData, 0, sizeof(jniData));
-
- // Store the JVM
- jniData.jvm = vm;
-
- return JNI_VERSION_1_4;
-}
-
-/////////////////////////////////////////////
-// Run Test
-//
-JNIEXPORT jint JNICALL
-Java_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest_RunTest(
- JNIEnv * env,
- jobject context,
- jobject surface)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "Run test");
- // Set instance independent Java objects
- VideoCaptureModule::SetAndroidObjects(jniData.jvm, context);
-
- // Start test
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "Create testPlatformDependent");
- testPlatformDependent testPlatformDependent;
- testPlatformDependent.SetRenderer(jniData.renderer);
- testPlatformDependent.DoTest();
-
- // Clear instance independent Java objects
- VideoCaptureModule::SetAndroidObjects(NULL, NULL);
-
- return 0;
-}
-
-JNIEXPORT jint JNICALL
-Java_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest_RenderInit(
- JNIEnv * env,
- jobject context,
- jobject surface)
-{
- VideoRender::SetAndroidObjects(jniData.jvm);
-#ifdef RENDER_PREVIEW
- Renderer::SetRenderWindow(surface);
- jniData.renderer=new Renderer(true);
-#endif
-}
-
-JNIEXPORT jint JNICALL
-Java_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest_StartCapture(
- JNIEnv * env,
- jobject context)
-{
- if (!jniData._captureInfo) {
- VideoCaptureModule::SetAndroidObjects(jniData.jvm, context);
- jniData._captureInfo = VideoCaptureFactory::CreateDeviceInfo(5);
- WebRtc_UWord8 id[256];
- WebRtc_UWord8 name[256];
- jniData._captureInfo->GetDeviceName(0, name, 256, id, 256);
- jniData._videoCapture = VideoCaptureFactory::Create(0, id);
- VideoCaptureCapability capability;
-
- jniData._captureInfo->GetCapability(id, 0, capability);
- capability.width = 176;
- capability.height = 144;
- capability.maxFPS = 15;
-
- jniData._videoCapture->StartCapture(capability);
- }
- return 0;
-}
-
-JNIEXPORT jint JNICALL
-Java_org_webrtc_capturemoduleandroidtest_VideoCaptureModuleTest_StopCapture(
- JNIEnv * env,
- jobject context)
-{
- if (jniData._videoCapture) {
- jniData._videoCapture->StopCapture();
- delete jniData._captureInfo;
- VideoCaptureModule::Destroy(jniData._videoCapture);
- jniData._videoCapture = NULL;
- jniData._captureInfo = NULL;
- }
- return 0;
-}
diff --git a/trunk/src/modules/video_capture/main/test/android/res/drawable-hdpi/icon.png b/trunk/src/modules/video_capture/main/test/android/res/drawable-hdpi/icon.png
deleted file mode 100644
index 8074c4c..0000000
--- a/trunk/src/modules/video_capture/main/test/android/res/drawable-hdpi/icon.png
+++ /dev/null
Binary files differ
diff --git a/trunk/src/modules/video_capture/main/test/android/res/drawable-ldpi/icon.png b/trunk/src/modules/video_capture/main/test/android/res/drawable-ldpi/icon.png
deleted file mode 100644
index 1095584..0000000
--- a/trunk/src/modules/video_capture/main/test/android/res/drawable-ldpi/icon.png
+++ /dev/null
Binary files differ
diff --git a/trunk/src/modules/video_capture/main/test/android/res/drawable-mdpi/icon.png b/trunk/src/modules/video_capture/main/test/android/res/drawable-mdpi/icon.png
deleted file mode 100644
index a07c69f..0000000
--- a/trunk/src/modules/video_capture/main/test/android/res/drawable-mdpi/icon.png
+++ /dev/null
Binary files differ
diff --git a/trunk/src/modules/video_capture/main/test/android/res/layout/main.xml b/trunk/src/modules/video_capture/main/test/android/res/layout/main.xml
deleted file mode 100644
index 3642733..0000000
--- a/trunk/src/modules/video_capture/main/test/android/res/layout/main.xml
+++ /dev/null
@@ -1,32 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android"
- android:orientation="vertical"
- android:layout_width="fill_parent"
- android:layout_height="fill_parent">
- <Button android:text="@string/run_button"
- android:id="@+id/Button01"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </Button>
- <Button android:text="@string/run_button2"
- android:id="@+id/Button02"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </Button>
- <Button android:text="@string/run_button3"
- android:id="@+id/Button03"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </Button>
- <Button android:text="@string/run_button4"
- android:id="@+id/Button04"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </Button>
- <LinearLayout
- android:id="@+id/renderView"
- android:layout_width="fill_parent"
- android:layout_height="fill_parent"
- android:layout_weight="1">
- </LinearLayout>
-</LinearLayout>
diff --git a/trunk/src/modules/video_capture/main/test/android/res/values/strings.xml b/trunk/src/modules/video_capture/main/test/android/res/values/strings.xml
deleted file mode 100644
index bd9a752..0000000
--- a/trunk/src/modules/video_capture/main/test/android/res/values/strings.xml
+++ /dev/null
@@ -1,11 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<resources>
-
- <string name="app_name">VideoCaptureModuleAndroidTest</string>
-<string name="run_button">Run Test</string>
-
-<string name="run_button2">Run Java test</string>
-<string name="run_button3">Start c++ Capture</string>
-<string name="run_button4">Stop c++ Capture</string>
-
-</resources>
diff --git a/trunk/src/modules/video_capture/main/test/android/src/org/webrtc/capturemoduleandroidtest/VideoCaptureJavaTest.java b/trunk/src/modules/video_capture/main/test/android/src/org/webrtc/capturemoduleandroidtest/VideoCaptureJavaTest.java
deleted file mode 100644
index 7a92c48..0000000
--- a/trunk/src/modules/video_capture/main/test/android/src/org/webrtc/capturemoduleandroidtest/VideoCaptureJavaTest.java
+++ /dev/null
@@ -1,61 +0,0 @@
-package org.webrtc.capturemoduleandroidtest;
-
-import java.util.List;
-
-import android.content.Context;
-import android.util.Log;
-
-import org.webrtc.videoengine.CaptureCapabilityAndroid;
-import org.webrtc.videoengine.VideoCaptureAndroid;
-import org.webrtc.videoengine.VideoCaptureDeviceInfoAndroid;
-
-public class VideoCaptureJavaTest {
- void DoTest(Context context)
- {
- VideoCaptureDeviceInfoAndroid videoCaptureDeviceInfo =
- VideoCaptureDeviceInfoAndroid.CreateVideoCaptureDeviceInfoAndroid(
- 5,context);
- for(int i = 0; i < videoCaptureDeviceInfo.NumberOfDevices(); i++) {
- String deviceUniqueId=videoCaptureDeviceInfo.GetDeviceUniqueName(i);
- VideoCaptureAndroid videoCapture =
- videoCaptureDeviceInfo.AllocateCamera(i,0,deviceUniqueId);
-
- CaptureCapabilityAndroid capArray[] =
- videoCaptureDeviceInfo.GetCapabilityArray(deviceUniqueId);
- for(CaptureCapabilityAndroid cap: capArray) {
- Log.d("*WEBRTC*", "Capability widht" + cap.width +
- " height " +cap.height+ " frameRate " +cap.maxFPS);
- int result=videoCapture.StartCapture(cap.width,
- cap.height,
- cap.maxFPS);
- try{
- Thread.sleep(2000);//sleep for 2000 ms
- }
- catch(InterruptedException ie){
- //If this thread was interrupted by another thread
- }
- result+=videoCapture.StopCapture();
- Log.d("*WEBRTC*", "Start stop result " + result);
- }
- VideoCaptureAndroid.DeleteVideoCaptureAndroid(videoCapture);
- videoCapture=null;
- }
- Log.d("*WEBRTC*", "Test complete");
- }
-
- VideoCaptureDeviceInfoAndroid _videoCaptureDeviceInfo;
- VideoCaptureAndroid _videoCapture;
- void StartCapture(Context context) {
- _videoCaptureDeviceInfo =
- VideoCaptureDeviceInfoAndroid.CreateVideoCaptureDeviceInfoAndroid(
- 5,context);
- String deviceUniqueId=_videoCaptureDeviceInfo.GetDeviceUniqueName(0);
- _videoCapture=_videoCaptureDeviceInfo.AllocateCamera(5,0,deviceUniqueId);
- _videoCapture.StartCapture(176,144,15);
- }
- void StopCapture() {
- _videoCapture.StopCapture();
- VideoCaptureAndroid.DeleteVideoCaptureAndroid(_videoCapture);
- }
-
-}
diff --git a/trunk/src/modules/video_capture/main/test/android/src/org/webrtc/capturemoduleandroidtest/VideoCaptureModuleTest.java b/trunk/src/modules/video_capture/main/test/android/src/org/webrtc/capturemoduleandroidtest/VideoCaptureModuleTest.java
deleted file mode 100644
index 7191b1c..0000000
--- a/trunk/src/modules/video_capture/main/test/android/src/org/webrtc/capturemoduleandroidtest/VideoCaptureModuleTest.java
+++ /dev/null
@@ -1,142 +0,0 @@
-package org.webrtc.capturemoduleandroidtest;
-
-import javax.microedition.khronos.egl.EGLConfig;
-import javax.microedition.khronos.opengles.GL10;
-
-import org.webrtc.capturemoduleandroidtest.R;
-import org.webrtc.videoengine.ViERenderer;
-
-import android.app.Activity;
-import android.opengl.GLSurfaceView;
-import android.os.Bundle;
-import android.util.Log;
-import android.view.SurfaceHolder;
-import android.view.SurfaceView;
-import android.view.View;
-import android.view.View.OnClickListener;
-import android.widget.Button;
-import android.widget.LinearLayout;
-
-public class VideoCaptureModuleTest
- extends Activity implements OnClickListener {
- // Set to 1 if OpenGL shall be used. 0 Otherwise
- private final int _useOpenGL=1;
- private Thread _testThread;
- private SurfaceView _view=null;
- private VideoCaptureModuleTest _thisPointer;
- private VideoCaptureJavaTest _videoCaptureJavaTest;
- /** Called when the activity is first created. */
- @Override
- public void onCreate(Bundle savedInstanceState) {
- super.onCreate(savedInstanceState);
- setContentView(R.layout.main);
-
- final Button buttonStartCP = (Button) findViewById(R.id.Button01);
- buttonStartCP.setOnClickListener(this);
- final Button buttonStartJava = (Button) findViewById(R.id.Button02);
- buttonStartJava.setOnClickListener(this);
- final Button buttonStartCPP = (Button) findViewById(R.id.Button03);
- buttonStartCPP.setOnClickListener(this);
- final Button buttonStopCPP = (Button) findViewById(R.id.Button04);
- buttonStopCPP.setOnClickListener(this);
- }
-
- private Runnable _testProc = new Runnable() {
- public void run() {
- // TODO: choose test from GUI
- // Select test here, 0 for API test, 1-> for Func tests
- RunTest(_view);
- }
- };
-
- @Override
- protected void onStart()
- {
- super.onStart();
- }
- @Override
- protected void onRestart()
- {
- super.onRestart();
- }
- @Override
- protected void onPause()
- {
- super.onPause();
- }
- @Override
- protected void onStop()
- {
- super.onStop();
- }
-
- // Function used to call test
- private native int RunTest(Object view);
- private native int RenderInit(Object view);
-
- private native int StartCapture();
- private native int StopCapture();
-
- static {
- Log.d("*WEBRTC*",
- "Loading ModuleVideoCaptureModuleAndroidTest...");
- System.loadLibrary(
- "ModuleVideoCaptureModuleAndroidTestJniAPI");
- }
-
- public void onClick(View v) {
- //get the handle to the layout
- LinearLayout renderLayout=(LinearLayout) findViewById(R.id.renderView);
- switch(v.getId())
- {
- case R.id.Button01:
- renderLayout.removeAllViews();
- _view=ViERenderer.CreateLocalRenderer(this);
- if(_useOpenGL==1)
- {
- _view= ViERenderer.CreateRenderer(this, true);
- }
- else
- {
- _view= new SurfaceView(this);
- }
- // add the surfaceview to the layout,
- // the surfaceview will be the same size as the layout (container)
- renderLayout.addView(_view);
- RenderInit(_view);
- _testThread = new Thread(_testProc);
- _testThread.start();
- break;
- case R.id.Button02:
- _view=ViERenderer.CreateLocalRenderer(this);
- renderLayout.removeAllViews();
- if(_videoCaptureJavaTest==null)
- {
- _videoCaptureJavaTest=new VideoCaptureJavaTest();
- _videoCaptureJavaTest.StartCapture(this);
- // add the surfaceview to the layout,
- // the surfaceview will be the same size as the layout (container)
- renderLayout.addView(_view);
- }
- else
- {
- _videoCaptureJavaTest.StopCapture();
- _videoCaptureJavaTest=null;
- }
- break;
-
- case R.id.Button03:
- _view=ViERenderer.CreateLocalRenderer(this);
- renderLayout.removeAllViews();
- StartCapture();
- // add the surfaceview to the layout,
- // the surfaceview will be the same size as the layout (container)
- renderLayout.addView(_view);
- break;
- case R.id.Button04:
- renderLayout.removeAllViews();
- StopCapture();
- break;
- }
- }
-}
\ No newline at end of file
diff --git a/trunk/src/modules/video_capture/main/test/video_capture_unittest.cc b/trunk/src/modules/video_capture/main/test/video_capture_unittest.cc
deleted file mode 100644
index ac49601..0000000
--- a/trunk/src/modules/video_capture/main/test/video_capture_unittest.cc
+++ /dev/null
@@ -1,455 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-
-#include "gtest/gtest.h"
-#include "process_thread.h"
-#include "scoped_ptr.h"
-#include "scoped_refptr.h"
-#include "tick_util.h"
-#include "video_capture.h"
-#include "video_capture_factory.h"
-
-using webrtc::TickTime;
-using webrtc::VideoCaptureAlarm;
-using webrtc::VideoCaptureCapability;
-using webrtc::VideoCaptureDataCallback;
-using webrtc::VideoCaptureFactory;
-using webrtc::VideoCaptureFeedBack;
-using webrtc::VideoCaptureModule;
-
-#if defined(_WIN32)
-#define SLEEP(x) Sleep(x)
-#elif defined(WEBRTC_ANDROID)
-#define SLEEP(x) usleep(x*1000)
-#else
-#include <unistd.h>
-#define SLEEP(x) usleep(x * 1000)
-#endif
-
-#define WAIT_(ex, timeout, res) \
- do { \
- res = (ex); \
- WebRtc_Word64 start = TickTime::MillisecondTimestamp(); \
- while (!res && TickTime::MillisecondTimestamp() < start + timeout) { \
- SLEEP(5); \
- res = (ex); \
- } \
- } while (0);\
-
-#define EXPECT_TRUE_WAIT(ex, timeout) \
- do { \
- bool res; \
- WAIT_(ex, timeout, res); \
- if (!res) EXPECT_TRUE(ex); \
- } while (0);
-
-
-static const int kTimeOut = 5000;
-static const int kTestHeight = 288;
-static const int kTestWidth = 352;
-static const int kTestFramerate = 30;
-
-// Compares the content of two video frames.
-static bool CompareFrames(const webrtc::VideoFrame& frame1,
- const webrtc::VideoFrame& frame2) {
- bool result =
- (frame1.Length() == frame2.Length()) &&
- (frame1.Width() == frame2.Width()) &&
- (frame1.Height() == frame2.Height());
-
- for (unsigned int i = 0; i < frame1.Length() && result; ++i)
- result = (*(frame1.Buffer()+i) == *(frame2.Buffer()+i));
- return result;
-}
-
-// Compares the content of a I420 frame in planar form and video frame.
-static bool CompareFrames(const webrtc::VideoFrameI420& frame1,
- const webrtc::VideoFrame& frame2) {
- if (frame1.width != frame2.Width() ||
- frame1.height != frame2.Height()) {
- return false;
- }
-
- // Compare Y
- unsigned char* y_plane = frame1.y_plane;
- for (unsigned int i = 0; i < frame2.Height(); ++i) {
- for (unsigned int j = 0; j < frame2.Width(); ++j) {
- if (*y_plane != *(frame2.Buffer()+i*frame2.Width() +j))
- return false;
- ++y_plane;
- }
- y_plane += frame1.y_pitch - frame1.width;
- }
-
- // Compare U
- unsigned char* u_plane = frame1.u_plane;
- for (unsigned int i = 0; i < frame2.Height() /2; ++i) {
- for (unsigned int j = 0; j < frame2.Width() /2; ++j) {
- if (*u_plane !=*(
- frame2.Buffer()+frame2.Width() * frame2.Height() +
- i*frame2.Width() / 2 + j)) {
- return false;
- }
- ++u_plane;
- }
- u_plane += frame1.u_pitch - frame1.width / 2;
- }
-
- // Compare V
- unsigned char* v_plane = frame1.v_plane;
- for (unsigned int i = 0; i < frame2.Height() /2; ++i) {
- for (unsigned int j = 0; j < frame2.Width() /2; ++j) {
- if (*v_plane != *(
- frame2.Buffer()+frame2.Width() * frame2.Height()* 5 / 4 +
- i*frame2.Width() / 2 + j)) {
- return false;
- }
- ++v_plane;
- }
- v_plane += frame1.v_pitch - frame1.width / 2;
- }
- return true;
-}
-
-class TestVideoCaptureCallback : public VideoCaptureDataCallback {
- public:
- TestVideoCaptureCallback()
- : capture_delay(0),
- last_render_time_ms(0),
- incoming_frames(0),
- timing_warnings(0) {
- }
-
- ~TestVideoCaptureCallback() {
- if (timing_warnings > 0)
- printf("No of timing warnings %d\n", timing_warnings);
- }
-
- virtual void OnIncomingCapturedFrame(const WebRtc_Word32 id,
- webrtc::VideoFrame& videoFrame,
- webrtc::VideoCodecType codecType) {
- int height = static_cast<int>(videoFrame.Height());
- int width = static_cast<int>(videoFrame.Width());
- EXPECT_EQ(height, capability.height);
- EXPECT_EQ(width, capability.width);
- // RenderTimstamp should be the time now.
- EXPECT_TRUE(
- videoFrame.RenderTimeMs() >= TickTime::MillisecondTimestamp()-30 &&
- videoFrame.RenderTimeMs() <= TickTime::MillisecondTimestamp());
-
- if ((videoFrame.RenderTimeMs() >
- last_render_time_ms + (1000 * 1.1) / capability.maxFPS &&
- last_render_time_ms > 0) ||
- (videoFrame.RenderTimeMs() <
- last_render_time_ms + (1000 * 0.9) / capability.maxFPS &&
- last_render_time_ms > 0)) {
- timing_warnings++;
- }
-
- incoming_frames++;
- last_render_time_ms = videoFrame.RenderTimeMs();
- last_frame.CopyFrame(videoFrame);
- }
-
- virtual void OnCaptureDelayChanged(const WebRtc_Word32 id,
- const WebRtc_Word32 delay) {
- capture_delay = delay;
- }
-
- VideoCaptureCapability capability;
- int capture_delay;
- WebRtc_Word64 last_render_time_ms;
- int incoming_frames;
- int timing_warnings;
- webrtc::VideoFrame last_frame;
-};
-
-class TestVideoCaptureFeedBack : public VideoCaptureFeedBack {
- public:
- TestVideoCaptureFeedBack() : frame_rate(0), alarm(webrtc::Cleared) {}
-
- virtual void OnCaptureFrameRate(const WebRtc_Word32 id,
- const WebRtc_UWord32 frameRate) {
- frame_rate = frameRate;
- }
-
- virtual void OnNoPictureAlarm(const WebRtc_Word32 id,
- const VideoCaptureAlarm reported_alarm) {
- alarm = reported_alarm;
- }
- unsigned int frame_rate;
- VideoCaptureAlarm alarm;
-};
-
-class VideoCaptureTest : public testing::Test {
- public:
- VideoCaptureTest() : number_of_devices_(0) {}
-
- void SetUp() {
- device_info_.reset(VideoCaptureFactory::CreateDeviceInfo(5));
- number_of_devices_ = device_info_->NumberOfDevices();
- ASSERT_GT(number_of_devices_, 0u);
- }
-
- webrtc::scoped_refptr<VideoCaptureModule> OpenVideoCaptureDevice(
- unsigned int device,
- VideoCaptureDataCallback* callback) {
- WebRtc_UWord8 device_name[256];
- WebRtc_UWord8 unique_name[256];
-
- EXPECT_EQ(0, device_info_->GetDeviceName(
- device, device_name, 256, unique_name, 256));
-
- webrtc::scoped_refptr<VideoCaptureModule> module(
- VideoCaptureFactory::Create(device, unique_name));
- if (module.get() == NULL)
- return NULL;
-
- EXPECT_FALSE(module->CaptureStarted());
-
- EXPECT_EQ(0, module->RegisterCaptureDataCallback(*callback));
- return module;
- }
-
- void StartCapture(VideoCaptureModule* capture_module,
- VideoCaptureCapability capability) {
- EXPECT_EQ(0, capture_module->StartCapture(capability));
- EXPECT_TRUE(capture_module->CaptureStarted());
-
- VideoCaptureCapability resulting_capability;
- EXPECT_EQ(0, capture_module->CaptureSettings(resulting_capability));
- EXPECT_EQ(capability, resulting_capability);
- }
-
- webrtc::scoped_ptr<VideoCaptureModule::DeviceInfo> device_info_;
- unsigned int number_of_devices_;
-};
-
-TEST_F(VideoCaptureTest, CreateDelete) {
- for (int i = 0; i < 5; ++i) {
- WebRtc_Word64 start_time = TickTime::MillisecondTimestamp();
- TestVideoCaptureCallback capture_observer;
- webrtc::scoped_refptr<VideoCaptureModule> module(OpenVideoCaptureDevice(
- 0, &capture_observer));
- ASSERT_TRUE(module.get() != NULL);
-
-#ifndef WEBRTC_MAC
- device_info_->GetCapability(module->CurrentDeviceName(), 0,
- capture_observer.capability);
-#else
- capture_observer.capability.width = kTestWidth;
- capture_observer.capability.height = kTestHeight;
- capture_observer.capability.maxFPS = kTestFramerate;
- capture_observer.capability.rawType = webrtc::kVideoUnknown;
-#endif
-
- StartCapture(module.get(), capture_observer.capability);
-
- // Less than 4s to start the camera.
- EXPECT_LE(TickTime::MillisecondTimestamp() - start_time, 4000);
-
- // Make sure 5 frames are captured.
- EXPECT_TRUE_WAIT(capture_observer.incoming_frames >= 5, kTimeOut);
-
- EXPECT_GT(capture_observer.capture_delay, 0);
-
- WebRtc_Word64 stop_time = TickTime::MillisecondTimestamp();
- EXPECT_EQ(0, module->StopCapture());
- EXPECT_FALSE(module->CaptureStarted());
-
- // Less than 3s to stop the camera.
- EXPECT_LE(TickTime::MillisecondTimestamp() - stop_time, 3000);
- }
-}
-
-TEST_F(VideoCaptureTest, Capabilities) {
-#ifdef WEBRTC_MAC
- printf("Video capture capabilities are not supported on Mac.\n");
- return;
-#endif
-
- TestVideoCaptureCallback capture_observer;
-
- webrtc::scoped_refptr<VideoCaptureModule> module(OpenVideoCaptureDevice(
- 0, &capture_observer));
- ASSERT_TRUE(module.get() != NULL);
-
- int number_of_capabilities = device_info_->NumberOfCapabilities(
- module->CurrentDeviceName());
- EXPECT_GT(number_of_capabilities, 0);
- for (int i = 0; i < number_of_capabilities; ++i) {
- device_info_->GetCapability(module->CurrentDeviceName(), i,
- capture_observer.capability);
- StartCapture(module.get(), capture_observer.capability);
- // Make sure 5 frames are captured.
- EXPECT_TRUE_WAIT(capture_observer.incoming_frames >= 5, kTimeOut);
- capture_observer.incoming_frames = 0;
- EXPECT_EQ(0, module->StopCapture());
- }
-}
-
-TEST_F(VideoCaptureTest, TestTwoCameras) {
- if (number_of_devices_ < 2) {
- printf("There are not two cameras available. Aborting test. \n");
- return;
- }
-
- TestVideoCaptureCallback capture_observer1;
- webrtc::scoped_refptr<VideoCaptureModule> module1(OpenVideoCaptureDevice(
- 0, &capture_observer1));
- ASSERT_TRUE(module1.get() != NULL);
-
-#ifndef WEBRTC_MAC
- device_info_->GetCapability(module1->CurrentDeviceName(), 0,
- capture_observer1.capability);
-#else
- capture_observer1.capability.width = kTestWidth;
- capture_observer1.capability.height = kTestHeight;
- capture_observer1.capability.maxFPS = kTestFramerate;
- capture_observer1.capability.rawType = webrtc::kVideoUnknown;
-#endif
-
- TestVideoCaptureCallback capture_observer2;
- webrtc::scoped_refptr<VideoCaptureModule> module2(OpenVideoCaptureDevice(
- 1, &capture_observer2));
- ASSERT_TRUE(module1.get() != NULL);
-
-
-#ifndef WEBRTC_MAC
- device_info_->GetCapability(module2->CurrentDeviceName(), 0,
- capture_observer2.capability);
-#else
- capture_observer2.capability.width = kTestWidth;
- capture_observer2.capability.height = kTestHeight;
- capture_observer2.capability.maxFPS = kTestFramerate;
- capture_observer2.capability.rawType = webrtc::kVideoUnknown;
-#endif
-
- StartCapture(module1.get(), capture_observer1.capability);
- StartCapture(module2.get(), capture_observer2.capability);
- EXPECT_TRUE_WAIT(capture_observer1.incoming_frames >= 5, kTimeOut);
- EXPECT_TRUE_WAIT(capture_observer2.incoming_frames >= 5, kTimeOut);
-}
-
-// Test class for testing external capture and capture feedback information
-// such as frame rate and picture alarm.
-class VideoCaptureExternalTest : public testing::Test {
- public:
- void SetUp() {
- capture_module_ = VideoCaptureFactory::Create(0, capture_input_interface_);
- process_module_ = webrtc::ProcessThread::CreateProcessThread();
- process_module_->Start();
- process_module_->RegisterModule(capture_module_);
-
- capture_callback_.capability.width = kTestWidth;
- capture_callback_.capability.height = kTestHeight;
- capture_callback_.capability.rawType = webrtc::kVideoYV12;
- capture_callback_.capability.maxFPS = kTestFramerate;
-
- test_frame_.VerifyAndAllocate(kTestWidth * kTestHeight * 3 / 2);
- test_frame_.SetLength(kTestWidth * kTestHeight * 3 / 2);
- test_frame_.SetHeight(kTestHeight);
- test_frame_.SetWidth(kTestWidth);
- SLEEP(1); // Wait 1ms so that two tests can't have the same timestamp.
- memset(test_frame_.Buffer(), 127, test_frame_.Length());
-
- EXPECT_EQ(0, capture_module_->RegisterCaptureDataCallback(
- capture_callback_));
- EXPECT_EQ(0, capture_module_->RegisterCaptureCallback(capture_feedback_));
- EXPECT_EQ(0, capture_module_->EnableFrameRateCallback(true));
- EXPECT_EQ(0, capture_module_->EnableNoPictureAlarm(true));
- }
-
- void TearDown() {
- process_module_->Stop();
- webrtc::ProcessThread::DestroyProcessThread(process_module_);
- }
-
- webrtc::VideoCaptureExternal* capture_input_interface_;
- webrtc::scoped_refptr<VideoCaptureModule> capture_module_;
- webrtc::ProcessThread* process_module_;
- webrtc::VideoFrame test_frame_;
- TestVideoCaptureCallback capture_callback_;
- TestVideoCaptureFeedBack capture_feedback_;
-};
-
-// Test input of external video frames.
-TEST_F(VideoCaptureExternalTest , TestExternalCapture) {
- EXPECT_EQ(0, capture_input_interface_->IncomingFrame(
- test_frame_.Buffer(), test_frame_.Length(), capture_callback_.capability,
- 0));
- EXPECT_TRUE(CompareFrames(test_frame_, capture_callback_.last_frame));
-}
-
-// Test input of planar I420 frames.
-TEST_F(VideoCaptureExternalTest , TestExternalCaptureI420) {
- webrtc::VideoFrameI420 frame_i420;
- frame_i420.width = kTestWidth;
- frame_i420.height = kTestHeight;
- frame_i420.y_plane = test_frame_.Buffer();
- frame_i420.u_plane = frame_i420.y_plane + (kTestWidth * kTestHeight);
- frame_i420.v_plane = frame_i420.u_plane + ((kTestWidth * kTestHeight) >> 2);
- frame_i420.y_pitch = kTestWidth;
- frame_i420.u_pitch = kTestWidth / 2;
- frame_i420.v_pitch = kTestWidth / 2;
- EXPECT_EQ(0, capture_input_interface_->IncomingFrameI420(frame_i420, 0));
-
- EXPECT_TRUE(CompareFrames(frame_i420, capture_callback_.last_frame));
-}
-
-// Test frame rate and no picture alarm.
-TEST_F(VideoCaptureExternalTest , FrameRate) {
- WebRtc_Word64 testTime = 3;
- TickTime startTime = TickTime::Now();
- capture_callback_.capability.maxFPS = 10;
- while ((TickTime::Now() - startTime).Milliseconds() < testTime * 1000) {
- EXPECT_EQ(0, capture_input_interface_->IncomingFrame(
- test_frame_.Buffer(), test_frame_.Length(),
- capture_callback_.capability, 0));
- SLEEP(1000 / capture_callback_.capability.maxFPS);
- }
- EXPECT_TRUE(capture_feedback_.frame_rate >= 8 &&
- capture_feedback_.frame_rate <= 10);
- SLEEP(500);
- EXPECT_EQ(webrtc::Raised, capture_feedback_.alarm);
-
- startTime = TickTime::Now();
- capture_callback_.capability.maxFPS = 30;
- while ((TickTime::Now() - startTime).Milliseconds() < testTime * 1000) {
- EXPECT_EQ(0, capture_input_interface_->IncomingFrame(
- test_frame_.Buffer(), test_frame_.Length(),
- capture_callback_.capability, 0));
- SLEEP(1000 / capture_callback_.capability.maxFPS);
- }
- EXPECT_EQ(webrtc::Cleared, capture_feedback_.alarm);
- // Frame rate might be less than 33 since we have paused providing
- // frames for a while.
- EXPECT_TRUE(capture_feedback_.frame_rate >= 25 &&
- capture_feedback_.frame_rate <= 33);
-}
-
-// Test start image
-TEST_F(VideoCaptureExternalTest , StartImage) {
- capture_callback_.capability.maxFPS = 10;
- EXPECT_EQ(0, capture_module_->StartSendImage(
- test_frame_, capture_callback_.capability.maxFPS));
-
- EXPECT_TRUE_WAIT(capture_callback_.incoming_frames == 5, kTimeOut);
- EXPECT_EQ(0, capture_module_->StopSendImage());
-
- SLEEP(200);
- // Test that no more start images have arrived.
- EXPECT_TRUE(capture_callback_.incoming_frames >= 4 &&
- capture_callback_.incoming_frames <= 5);
- EXPECT_TRUE(CompareFrames(test_frame_, capture_callback_.last_frame));
-}
-
diff --git a/trunk/src/modules/video_coding/codecs/OWNERS b/trunk/src/modules/video_coding/codecs/OWNERS
deleted file mode 100644
index 7183cf2..0000000
--- a/trunk/src/modules/video_coding/codecs/OWNERS
+++ /dev/null
@@ -1,4 +0,0 @@
-stefan@webrtc.org
-mikhal@webrtc.org
-marpan@webrtc.org
-henrik.lundin@webrtc.org
diff --git a/trunk/src/modules/video_coding/codecs/i420/main/interface/i420.h b/trunk/src/modules/video_coding/codecs/i420/main/interface/i420.h
deleted file mode 100644
index fb4ba8c..0000000
--- a/trunk/src/modules/video_coding/codecs/i420/main/interface/i420.h
+++ /dev/null
@@ -1,156 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_H_
-
-#include "video_codec_interface.h"
-#include "typedefs.h"
-
-namespace webrtc
-{
-
-class I420Encoder : public VideoEncoder
-{
-public:
-
- I420Encoder();
-
- virtual ~I420Encoder();
-
-// Initialize the encoder with the information from the VideoCodec
-//
-// Input:
-// - codecSettings : Codec settings
-// - numberOfCores : Number of cores available for the encoder
-// - maxPayloadSize : The maximum size each payload is allowed
-// to have. Usually MTU - overhead.
-//
-// Return value : WEBRTC_VIDEO_CODEC_OK if OK
-// <0 - Error
- virtual WebRtc_Word32 InitEncode(const VideoCodec* codecSettings, WebRtc_Word32 /*numberOfCores*/, WebRtc_UWord32 /*maxPayloadSize*/);
-
-// "Encode" an I420 image (as a part of a video stream). The encoded image
-// will be returned to the user via the encode complete callback.
-//
-// Input:
-// - inputImage : Image to be encoded
-// - codecSpecificInfo : Pointer to codec specific data
-// - frameType : Frame type to be sent (Key /Delta) .
-//
-// Return value : WEBRTC_VIDEO_CODEC_OK if OK
-// <0 - Error
- virtual WebRtc_Word32
- Encode(const RawImage& inputImage,
- const CodecSpecificInfo* /*codecSpecificInfo*/,
- const VideoFrameType* /*frameTypes*/);
-
-// Register an encode complete callback object.
-//
-// Input:
-// - callback : Callback object which handles encoded images.
-//
-// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual WebRtc_Word32 RegisterEncodeCompleteCallback(EncodedImageCallback* callback);
-
-// Free encoder memory.
-//
-// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual WebRtc_Word32 Release();
-
- virtual WebRtc_Word32 SetRates(WebRtc_UWord32 /*newBitRate*/,
- WebRtc_UWord32 /*frameRate*/)
- {return WEBRTC_VIDEO_CODEC_OK;}
-
- virtual WebRtc_Word32 SetChannelParameters(WebRtc_UWord32 /*packetLoss*/,
- int /*rtt*/)
- {return WEBRTC_VIDEO_CODEC_OK;}
-
- virtual WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* /*buffer*/,
- WebRtc_Word32 /*size*/)
- {return WEBRTC_VIDEO_CODEC_OK;}
-
-private:
- bool _inited;
- EncodedImage _encodedImage;
- EncodedImageCallback* _encodedCompleteCallback;
-
-}; // end of WebRtcI420DEncoder class
-
-class I420Decoder : public VideoDecoder
-{
-public:
-
- I420Decoder();
-
- virtual ~I420Decoder();
-
-// Initialize the decoder.
-// The user must notify the codec of width and height values.
-//
-// Return value : WEBRTC_VIDEO_CODEC_OK.
-// <0 - Errors
- virtual WebRtc_Word32 InitDecode(const VideoCodec* codecSettings, WebRtc_Word32 /*numberOfCores*/);
-
- virtual WebRtc_Word32 SetCodecConfigParameters(const WebRtc_UWord8* /*buffer*/, WebRtc_Word32 /*size*/){return WEBRTC_VIDEO_CODEC_OK;};
-
-// Decode encoded image (as a part of a video stream). The decoded image
-// will be returned to the user through the decode complete callback.
-//
-// Input:
-// - inputImage : Encoded image to be decoded
-// - missingFrames : True if one or more frames have been lost
-// since the previous decode call.
-// - codecSpecificInfo : pointer to specific codec data
-// - renderTimeMs : Render time in Ms
-//
-// Return value : WEBRTC_VIDEO_CODEC_OK if OK
-// <0 - Error
- virtual WebRtc_Word32 Decode(
- const EncodedImage& inputImage,
- bool missingFrames,
- const RTPFragmentationHeader* /*fragmentation*/,
- const CodecSpecificInfo* /*codecSpecificInfo*/,
- WebRtc_Word64 /*renderTimeMs*/);
-
-// Register a decode complete callback object.
-//
-// Input:
-// - callback : Callback object which handles decoded images.
-//
-// Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual WebRtc_Word32 RegisterDecodeCompleteCallback(DecodedImageCallback* callback);
-
-// Free decoder memory.
-//
-// Return value : WEBRTC_VIDEO_CODEC_OK if OK
-// <0 - Error
- virtual WebRtc_Word32 Release();
-
-// Reset decoder state and prepare for a new call.
-//
-// Return value : WEBRTC_VIDEO_CODEC_OK.
-// <0 - Error
- virtual WebRtc_Word32 Reset();
-
-private:
-
- RawImage _decodedImage;
- WebRtc_Word32 _width;
- WebRtc_Word32 _height;
- bool _inited;
- DecodedImageCallback* _decodeCompleteCallback;
-
-
-}; // end of WebRtcI420Decoder class
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_I420_H_
diff --git a/trunk/src/modules/video_coding/codecs/i420/main/source/Android.mk b/trunk/src/modules/video_coding/codecs/i420/main/source/Android.mk
deleted file mode 100644
index d73e77d..0000000
--- a/trunk/src/modules/video_coding/codecs/i420/main/source/Android.mk
+++ /dev/null
@@ -1,41 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../../../android-webrtc.mk
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_i420
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := i420.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-# Include paths placed before CFLAGS/CPPFLAGS
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../../interface \
- $(LOCAL_PATH)/../../../../../.. \
- $(LOCAL_PATH)/../../../../../../common_video/interface \
- $(LOCAL_PATH)/../../../../../../system_wrappers/interface
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/video_coding/codecs/i420/main/source/i420.cc b/trunk/src/modules/video_coding/codecs/i420/main/source/i420.cc
deleted file mode 100644
index b73a00a..0000000
--- a/trunk/src/modules/video_coding/codecs/i420/main/source/i420.cc
+++ /dev/null
@@ -1,253 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "i420.h"
-#include <string.h>
-
-namespace webrtc
-{
-
-I420Encoder::I420Encoder():
-_inited(false),
-_encodedImage(),
-_encodedCompleteCallback(NULL)
-{
- //
-}
-
-I420Encoder::~I420Encoder()
-{
- _inited = false;
- if (_encodedImage._buffer != NULL)
- {
- delete [] _encodedImage._buffer;
- _encodedImage._buffer = NULL;
- }
-}
-
-WebRtc_Word32
-I420Encoder::Release()
-{
- // should allocate an encoded frame and then release it here, for that we actaully need an init flag
- if (_encodedImage._buffer != NULL)
- {
- delete [] _encodedImage._buffer;
- _encodedImage._buffer = NULL;
- }
- _inited = false;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-WebRtc_Word32
-I420Encoder::InitEncode(const VideoCodec* codecSettings,
- WebRtc_Word32 /*numberOfCores*/,
- WebRtc_UWord32 /*maxPayloadSize */)
-{
- if (codecSettings == NULL)
- {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- if (codecSettings->width < 1 || codecSettings->height < 1)
- {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
-
- // allocating encoded memory
-
- if (_encodedImage._buffer != NULL)
- {
- delete [] _encodedImage._buffer;
- _encodedImage._buffer = NULL;
- _encodedImage._size = 0;
- }
- const WebRtc_UWord32 newSize = (3 * codecSettings->width *
- codecSettings->height) >> 1;
- WebRtc_UWord8* newBuffer = new WebRtc_UWord8[newSize];
- if (newBuffer == NULL)
- {
- return WEBRTC_VIDEO_CODEC_MEMORY;
- }
- _encodedImage._size = newSize;
- _encodedImage._buffer = newBuffer;
-
- // if no memory allocation, no point to init
- _inited = true;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-
-
-WebRtc_Word32
-I420Encoder::Encode(const RawImage& inputImage,
- const CodecSpecificInfo* /*codecSpecificInfo*/,
- const VideoFrameType* /*frameTypes*/)
-{
- if (!_inited)
- {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
- if (_encodedCompleteCallback == NULL)
- {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
-
- _encodedImage._frameType = kKeyFrame; // no coding
- _encodedImage._timeStamp = inputImage._timeStamp;
- _encodedImage._encodedHeight = inputImage._height;
- _encodedImage._encodedWidth = inputImage._width;
- if (inputImage._length > _encodedImage._size)
- {
-
- // allocating encoded memory
- if (_encodedImage._buffer != NULL)
- {
- delete [] _encodedImage._buffer;
- _encodedImage._buffer = NULL;
- _encodedImage._size = 0;
- }
- const WebRtc_UWord32 newSize = (3 * _encodedImage._encodedWidth * _encodedImage._encodedHeight) >> 1;
- WebRtc_UWord8* newBuffer = new WebRtc_UWord8[newSize];
- if (newBuffer == NULL)
- {
- return WEBRTC_VIDEO_CODEC_MEMORY;
- }
- _encodedImage._size = newSize;
- _encodedImage._buffer = newBuffer;
- }
- memcpy(_encodedImage._buffer, inputImage._buffer, inputImage._length);
- _encodedImage._length = inputImage._length;
- _encodedCompleteCallback->Encoded(_encodedImage);
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-
-WebRtc_Word32
-I420Encoder::RegisterEncodeCompleteCallback(EncodedImageCallback* callback)
-{
- _encodedCompleteCallback = callback;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-
-I420Decoder::I420Decoder():
-_decodedImage(),
-_width(0),
-_height(0),
-_inited(false),
-_decodeCompleteCallback(NULL)
-{
- //
-}
-
-I420Decoder::~I420Decoder()
-{
- Release();
-}
-
-WebRtc_Word32
-I420Decoder::Reset()
-{
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-
-WebRtc_Word32
-I420Decoder::InitDecode(const VideoCodec* codecSettings, WebRtc_Word32 /*numberOfCores */)
-{
- if (codecSettings == NULL)
- {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- else if (codecSettings->width < 1 || codecSettings->height < 1)
- {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- _width = codecSettings->width;
- _height = codecSettings->height;
- _inited = true;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-WebRtc_Word32
-I420Decoder::Decode(const EncodedImage& inputImage,
- bool /*missingFrames*/,
- const RTPFragmentationHeader* /*fragmentation*/,
- const CodecSpecificInfo* /*codecSpecificInfo*/,
- WebRtc_Word64 /*renderTimeMs*/)
-{
- if (inputImage._buffer == NULL)
- {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- if (_decodeCompleteCallback == NULL)
- {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
- if (inputImage._length <= 0)
- {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- if (!_inited)
- {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
-
- //Allocate memory for decoded image
-
- if (_decodedImage._buffer != NULL)
- {
- delete [] _decodedImage._buffer;
- _decodedImage._buffer = NULL;
- _decodedImage._size = 0;
- }
- if (_decodedImage._buffer == NULL)
- {
- const WebRtc_UWord32 newSize = (3*_width*_height) >> 1;
- WebRtc_UWord8* newBuffer = new WebRtc_UWord8[newSize];
- if (newBuffer == NULL)
- {
- return WEBRTC_VIDEO_CODEC_MEMORY;
- }
- _decodedImage._size = newSize;
- _decodedImage._buffer = newBuffer;
- }
-
- // Set decoded image parameters
- _decodedImage._height = _height;
- _decodedImage._width = _width;
- _decodedImage._timeStamp = inputImage._timeStamp;
- memcpy(_decodedImage._buffer, inputImage._buffer, inputImage._length);
- _decodedImage._length = inputImage._length;
- //_decodedImage._buffer = inputImage._buffer;
-
- _decodeCompleteCallback->Decoded(_decodedImage);
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-WebRtc_Word32
-I420Decoder::RegisterDecodeCompleteCallback(DecodedImageCallback* callback)
-{
- _decodeCompleteCallback = callback;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-WebRtc_Word32
-I420Decoder::Release()
-{
- if (_decodedImage._buffer != NULL)
- {
- delete [] _decodedImage._buffer;
- _decodedImage._buffer = NULL;
- }
- _inited = false;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-}
diff --git a/trunk/src/modules/video_coding/codecs/i420/main/source/i420.gypi b/trunk/src/modules/video_coding/codecs/i420/main/source/i420.gypi
deleted file mode 100644
index af13f8d..0000000
--- a/trunk/src/modules/video_coding/codecs/i420/main/source/i420.gypi
+++ /dev/null
@@ -1,40 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'webrtc_i420',
- 'type': '<(library)',
- 'dependencies': [
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '../interface',
- '../../../interface',
- '../../../../../../common_video/interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- '../../../../../../common_video/interface',
- ],
- },
- 'sources': [
- '../interface/i420.h',
- 'i420.cc',
- ],
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h b/trunk/src/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h
deleted file mode 100644
index f61fce5..0000000
--- a/trunk/src/modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h
+++ /dev/null
@@ -1,89 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
-
-#include <string>
-
-#include "gmock/gmock.h"
-#include "modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-class MockEncodedImageCallback : public EncodedImageCallback {
- public:
- MOCK_METHOD3(Encoded,
- WebRtc_Word32(EncodedImage& encodedImage,
- const CodecSpecificInfo* codecSpecificInfo,
- const RTPFragmentationHeader* fragmentation));
-};
-
-class MockVideoEncoder : public VideoEncoder {
- public:
- MOCK_CONST_METHOD2(Version,
- WebRtc_Word32(WebRtc_Word8 *version,
- WebRtc_Word32 length));
- MOCK_METHOD3(InitEncode,
- WebRtc_Word32(const VideoCodec* codecSettings,
- WebRtc_Word32 numberOfCores,
- WebRtc_UWord32 maxPayloadSize));
- MOCK_METHOD3(Encode,
- WebRtc_Word32(const RawImage& inputImage,
- const CodecSpecificInfo* codecSpecificInfo,
- const VideoFrameType* frameType));
- MOCK_METHOD1(RegisterEncodeCompleteCallback,
- WebRtc_Word32(EncodedImageCallback* callback));
- MOCK_METHOD0(Release, WebRtc_Word32());
- MOCK_METHOD0(Reset, WebRtc_Word32());
- MOCK_METHOD2(SetChannelParameters, WebRtc_Word32(WebRtc_UWord32 packetLoss,
- int rtt));
- MOCK_METHOD2(SetRates,
- WebRtc_Word32(WebRtc_UWord32 newBitRate,
- WebRtc_UWord32 frameRate));
- MOCK_METHOD1(SetPeriodicKeyFrames, WebRtc_Word32(bool enable));
- MOCK_METHOD2(CodecConfigParameters,
- WebRtc_Word32(WebRtc_UWord8* /*buffer*/, WebRtc_Word32));
-};
-
-class MockDecodedImageCallback : public DecodedImageCallback {
- public:
- MOCK_METHOD1(Decoded,
- WebRtc_Word32(RawImage& decodedImage));
- MOCK_METHOD1(ReceivedDecodedReferenceFrame,
- WebRtc_Word32(const WebRtc_UWord64 pictureId));
- MOCK_METHOD1(ReceivedDecodedFrame,
- WebRtc_Word32(const WebRtc_UWord64 pictureId));
-};
-
-class MockVideoDecoder : public VideoDecoder {
- public:
- MOCK_METHOD2(InitDecode,
- WebRtc_Word32(const VideoCodec* codecSettings,
- WebRtc_Word32 numberOfCores));
- MOCK_METHOD5(Decode,
- WebRtc_Word32(const EncodedImage& inputImage,
- bool missingFrames,
- const RTPFragmentationHeader* fragmentation,
- const CodecSpecificInfo* codecSpecificInfo,
- WebRtc_Word64 renderTimeMs));
- MOCK_METHOD1(RegisterDecodeCompleteCallback,
- WebRtc_Word32(DecodedImageCallback* callback));
- MOCK_METHOD0(Release, WebRtc_Word32());
- MOCK_METHOD0(Reset, WebRtc_Word32());
- MOCK_METHOD2(SetCodecConfigParameters,
- WebRtc_Word32(const WebRtc_UWord8* /*buffer*/, WebRtc_Word32));
- MOCK_METHOD0(Copy, VideoDecoder*());
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_MOCK_MOCK_VIDEO_CODEC_INTERFACE_H_
diff --git a/trunk/src/modules/video_coding/codecs/interface/video_codec_interface.h b/trunk/src/modules/video_coding/codecs/interface/video_codec_interface.h
deleted file mode 100644
index 9aa5cb7..0000000
--- a/trunk/src/modules/video_coding/codecs/interface/video_codec_interface.h
+++ /dev/null
@@ -1,247 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H
-
-#include "common_types.h"
-#include "common_video/interface/video_image.h"
-#include "modules/video_coding/codecs/interface/video_error_codes.h"
-#include "typedefs.h"
-
-namespace webrtc
-{
-
-class RTPFragmentationHeader; // forward declaration
-
-// Note: if any pointers are added to this struct, it must be fitted
-// with a copy-constructor. See below.
-struct CodecSpecificInfoVP8
-{
- bool hasReceivedSLI;
- WebRtc_UWord8 pictureIdSLI;
- bool hasReceivedRPSI;
- WebRtc_UWord64 pictureIdRPSI;
- WebRtc_Word16 pictureId; // negative value to skip pictureId
- bool nonReference;
- WebRtc_UWord8 simulcastIdx;
- WebRtc_UWord8 temporalIdx;
- bool layerSync;
- int tl0PicIdx; // Negative value to skip tl0PicIdx
- WebRtc_Word8 keyIdx; // negative value to skip keyIdx
-};
-
-union CodecSpecificInfoUnion
-{
- CodecSpecificInfoVP8 VP8;
-};
-
-// Note: if any pointers are added to this struct or its sub-structs, it
-// must be fitted with a copy-constructor. This is because it is copied
-// in the copy-constructor of VCMEncodedFrame.
-struct CodecSpecificInfo
-{
- VideoCodecType codecType;
- CodecSpecificInfoUnion codecSpecific;
-};
-
-class EncodedImageCallback
-{
-public:
- virtual ~EncodedImageCallback() {};
-
- // Callback function which is called when an image has been encoded.
- //
- // Input:
- // - encodedImage : The encoded image
- //
- // Return value : > 0, signals to the caller that one or more future frames
- // should be dropped to keep bit rate or frame rate.
- // = 0, if OK.
- // < 0, on error.
- virtual WebRtc_Word32
- Encoded(EncodedImage& encodedImage,
- const CodecSpecificInfo* codecSpecificInfo = NULL,
- const RTPFragmentationHeader* fragmentation = NULL) = 0;
-};
-
-class VideoEncoder
-{
-public:
- virtual ~VideoEncoder() {};
-
- // Initialize the encoder with the information from the VideoCodec.
- //
- // Input:
- // - codecSettings : Codec settings
- // - numberOfCores : Number of cores available for the encoder
- // - maxPayloadSize : The maximum size each payload is allowed
- // to have. Usually MTU - overhead.
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual WebRtc_Word32 InitEncode(const VideoCodec* codecSettings, WebRtc_Word32 numberOfCores, WebRtc_UWord32 maxPayloadSize) = 0;
-
- // Encode an I420 image (as a part of a video stream). The encoded image
- // will be returned to the user through the encode complete callback.
- //
- // Input:
- // - inputImage : Image to be encoded
- // - codecSpecificInfo : Pointer to codec specific data
- // - frameType : The frame type to encode
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual WebRtc_Word32
- Encode(const RawImage& inputImage,
- const CodecSpecificInfo* codecSpecificInfo,
- const VideoFrameType* frameTypes) = 0;
-
- // Register an encode complete callback object.
- //
- // Input:
- // - callback : Callback object which handles encoded images.
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual WebRtc_Word32 RegisterEncodeCompleteCallback(EncodedImageCallback* callback) = 0;
-
- // Free encoder memory.
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual WebRtc_Word32 Release() = 0;
-
- // Inform the encoder about the packet loss and round trip time on the
- // network used to decide the best pattern and signaling.
- //
- // - packetLoss : Fraction lost (loss rate in percent =
- // 100 * packetLoss / 255)
- // - rtt : Round-trip time in milliseconds
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual WebRtc_Word32 SetChannelParameters(WebRtc_UWord32 packetLoss,
- int rtt) = 0;
-
- // Inform the encoder about the new target bit rate.
- //
- // - newBitRate : New target bit rate
- // - frameRate : The target frame rate
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual WebRtc_Word32 SetRates(WebRtc_UWord32 newBitRate, WebRtc_UWord32 frameRate) = 0;
-
- // Use this function to enable or disable periodic key frames. Can be useful for codecs
- // which have other ways of stopping error propagation.
- //
- // - enable : Enable or disable periodic key frames
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual WebRtc_Word32 SetPeriodicKeyFrames(bool enable) { return WEBRTC_VIDEO_CODEC_ERROR; }
-
- // Codec configuration data to send out-of-band, i.e. in SIP call setup
- //
- // - buffer : Buffer pointer to where the configuration data
- // should be stored
- // - size : The size of the buffer in bytes
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* /*buffer*/, WebRtc_Word32 /*size*/) { return WEBRTC_VIDEO_CODEC_ERROR; }
-};
-
-class DecodedImageCallback
-{
-public:
- virtual ~DecodedImageCallback() {};
-
- // Callback function which is called when an image has been decoded.
- //
- // Input:
- // - decodedImage : The decoded image
- //
- // Return value : 0 if OK, < 0 otherwise.
- virtual WebRtc_Word32 Decoded(RawImage& decodedImage) = 0;
-
- virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId) {return -1;}
-
- virtual WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 pictureId) {return -1;}
-};
-
-class VideoDecoder
-{
-public:
- virtual ~VideoDecoder() {};
-
- // Initialize the decoder with the information from the VideoCodec.
- //
- // Input:
- // - inst : Codec settings
- // - numberOfCores : Number of cores available for the decoder
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual WebRtc_Word32 InitDecode(const VideoCodec* codecSettings, WebRtc_Word32 numberOfCores) = 0;
-
- // Decode encoded image (as a part of a video stream). The decoded image
- // will be returned to the user through the decode complete callback.
- //
- // Input:
- // - inputImage : Encoded image to be decoded
- // - missingFrames : True if one or more frames have been lost
- // since the previous decode call.
- // - fragmentation : Specifies where the encoded frame can be
- // split into separate fragments. The meaning
- // of fragment is codec specific, but often
- // means that each fragment is decodable by
- // itself.
- // - codecSpecificInfo : Pointer to codec specific data
- // - renderTimeMs : System time to render in milliseconds. Only
- // used by decoders with internal rendering.
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual WebRtc_Word32
- Decode(const EncodedImage& inputImage,
- bool missingFrames,
- const RTPFragmentationHeader* fragmentation,
- const CodecSpecificInfo* codecSpecificInfo = NULL,
- WebRtc_Word64 renderTimeMs = -1) = 0;
-
- // Register an decode complete callback object.
- //
- // Input:
- // - callback : Callback object which handles decoded images.
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual WebRtc_Word32 RegisterDecodeCompleteCallback(DecodedImageCallback* callback) = 0;
-
- // Free decoder memory.
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual WebRtc_Word32 Release() = 0;
-
- // Reset decoder state and prepare for a new call.
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual WebRtc_Word32 Reset() = 0;
-
- // Codec configuration data sent out-of-band, i.e. in SIP call setup
- //
- // Input/Output:
- // - buffer : Buffer pointer to the configuration data
- // - size : The size of the configuration data in
- // bytes
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual WebRtc_Word32 SetCodecConfigParameters(const WebRtc_UWord8* /*buffer*/, WebRtc_Word32 /*size*/) { return WEBRTC_VIDEO_CODEC_ERROR; }
-
- // Create a copy of the codec and its internal state.
- //
- // Return value : A copy of the instance if OK, NULL otherwise.
- virtual VideoDecoder* Copy() { return NULL; }
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_CODEC_INTERFACE_H
diff --git a/trunk/src/modules/video_coding/codecs/interface/video_error_codes.h b/trunk/src/modules/video_coding/codecs/interface/video_error_codes.h
deleted file mode 100644
index dfa3f53..0000000
--- a/trunk/src/modules/video_coding/codecs/interface/video_error_codes.h
+++ /dev/null
@@ -1,30 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H
-
-// NOTE: in sync with video_coding_module_defines.h
-
-// Define return values
-
-#define WEBRTC_VIDEO_CODEC_REQUEST_SLI 2
-#define WEBRTC_VIDEO_CODEC_NO_OUTPUT 1
-#define WEBRTC_VIDEO_CODEC_OK 0
-#define WEBRTC_VIDEO_CODEC_ERROR -1
-#define WEBRTC_VIDEO_CODEC_LEVEL_EXCEEDED -2
-#define WEBRTC_VIDEO_CODEC_MEMORY -3
-#define WEBRTC_VIDEO_CODEC_ERR_PARAMETER -4
-#define WEBRTC_VIDEO_CODEC_ERR_SIZE -5
-#define WEBRTC_VIDEO_CODEC_TIMEOUT -6
-#define WEBRTC_VIDEO_CODEC_UNINITIALIZED -7
-#define WEBRTC_VIDEO_CODEC_ERR_REQUEST_SLI -12
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_INTERFACE_VIDEO_ERROR_CODES_H
diff --git a/trunk/src/modules/video_coding/codecs/test/mock/mock_packet_manipulator.h b/trunk/src/modules/video_coding/codecs/test/mock/mock_packet_manipulator.h
deleted file mode 100644
index 57d21ca..0000000
--- a/trunk/src/modules/video_coding/codecs/test/mock/mock_packet_manipulator.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_MOCK_MOCK_PACKET_MANIPULATOR_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_MOCK_MOCK_PACKET_MANIPULATOR_H_
-
-#include "modules/video_coding/codecs/test/packet_manipulator.h"
-
-#include <string>
-
-#include "common_video/interface/video_image.h"
-#include "gmock/gmock.h"
-#include "typedefs.h"
-
-namespace webrtc {
-namespace test {
-
-class MockPacketManipulator : public PacketManipulator {
- public:
- MOCK_METHOD1(ManipulatePackets, int(webrtc::EncodedImage* encoded_image));
-};
-
-} // namespace test
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_MOCK_MOCK_PACKET_MANIPULATOR_H_
diff --git a/trunk/src/modules/video_coding/codecs/test/packet_manipulator.cc b/trunk/src/modules/video_coding/codecs/test/packet_manipulator.cc
deleted file mode 100644
index acdb2e5..0000000
--- a/trunk/src/modules/video_coding/codecs/test/packet_manipulator.cc
+++ /dev/null
@@ -1,111 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/video_coding/codecs/test/packet_manipulator.h"
-
-#include <cassert>
-#include <cstdio>
-
-namespace webrtc {
-namespace test {
-
-PacketManipulatorImpl::PacketManipulatorImpl(PacketReader* packet_reader,
- const NetworkingConfig& config,
- bool verbose)
- : packet_reader_(packet_reader),
- config_(config),
- active_burst_packets_(0),
- critsect_(CriticalSectionWrapper::CreateCriticalSection()),
- random_seed_(1),
- verbose_(verbose) {
- assert(packet_reader);
-}
-
-PacketManipulatorImpl::~PacketManipulatorImpl() {
- delete critsect_;
-}
-
-int PacketManipulatorImpl::ManipulatePackets(
- webrtc::EncodedImage* encoded_image) {
- assert(encoded_image);
- int nbr_packets_dropped = 0;
- // There's no need to build a copy of the image data since viewing an
- // EncodedImage object, setting the length to a new lower value represents
- // that everything is dropped after that position in the byte array.
- // EncodedImage._size is the allocated bytes.
- // EncodedImage._length is how many that are filled with data.
- int new_length = 0;
- packet_reader_->InitializeReading(encoded_image->_buffer,
- encoded_image->_length,
- config_.packet_size_in_bytes);
- WebRtc_UWord8* packet = NULL;
- int nbr_bytes_to_read;
- // keep track of if we've lost any packets, since then we shall loose
- // the remains of the current frame:
- bool packet_loss_has_occurred = false;
- while ((nbr_bytes_to_read = packet_reader_->NextPacket(&packet)) > 0) {
- // Check if we're currently in a packet loss burst that is not completed:
- if (active_burst_packets_ > 0) {
- active_burst_packets_--;
- nbr_packets_dropped++;
- } else if (RandomUniform() < config_.packet_loss_probability ||
- packet_loss_has_occurred) {
- packet_loss_has_occurred = true;
- nbr_packets_dropped++;
- if (config_.packet_loss_mode == kBurst) {
- // Initiate a new burst
- active_burst_packets_ = config_.packet_loss_burst_length - 1;
- }
- } else {
- new_length += nbr_bytes_to_read;
- }
- }
- encoded_image->_length = new_length;
- if (nbr_packets_dropped > 0) {
- // Must set completeFrame to false to inform the decoder about this:
- encoded_image->_completeFrame = false;
- if (verbose_) {
- printf("Dropped %d packets for frame %d (frame length: %d)\n",
- nbr_packets_dropped, encoded_image->_timeStamp,
- encoded_image->_length);
- }
- }
- return nbr_packets_dropped;
-}
-
-void PacketManipulatorImpl::InitializeRandomSeed(unsigned int seed) {
- random_seed_ = seed;
-}
-
-inline double PacketManipulatorImpl::RandomUniform() {
- // Use the previous result as new seed before each rand() call. Doing this
- // it doesn't matter if other threads are calling rand() since we'll always
- // get the same behavior as long as we're using a fixed initial seed.
- critsect_->Enter();
- srand(random_seed_);
- random_seed_ = std::rand();
- critsect_->Leave();
- return (random_seed_ + 1.0)/(RAND_MAX + 1.0);
-}
-
-const char* PacketLossModeToStr(PacketLossMode e) {
- switch (e) {
- case kUniform:
- return "Uniform";
- case kBurst:
- return "Burst";
- default:
- assert(false);
- return "Unknown";
- }
-}
-
-} // namespace test
-} // namespace webrtcc
diff --git a/trunk/src/modules/video_coding/codecs/test/packet_manipulator.h b/trunk/src/modules/video_coding/codecs/test/packet_manipulator.h
deleted file mode 100644
index e3891cb..0000000
--- a/trunk/src/modules/video_coding/codecs/test/packet_manipulator.h
+++ /dev/null
@@ -1,113 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_
-
-#include <cstdlib>
-
-#include "modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "system_wrappers/interface/critical_section_wrapper.h"
-#include "testsupport/packet_reader.h"
-
-namespace webrtc {
-namespace test {
-
-// Which mode the packet loss shall be performed according to.
-enum PacketLossMode {
- // Drops packets with a configured probability independently for each packet
- kUniform,
- // Drops packets similar to uniform but when a packet is being dropped,
- // the number of lost packets in a row is equal to the configured burst
- // length.
- kBurst
-};
-// Returns a string representation of the enum value.
-const char* PacketLossModeToStr(PacketLossMode e);
-
-// Contains configurations related to networking and simulation of
-// scenarios caused by network interference.
-struct NetworkingConfig {
- NetworkingConfig()
- : packet_size_in_bytes(1500), max_payload_size_in_bytes(1440),
- packet_loss_mode(kUniform), packet_loss_probability(0.0),
- packet_loss_burst_length(1) {
- }
-
- // Packet size in bytes. Default: 1500 bytes.
- int packet_size_in_bytes;
-
- // Encoder specific setting of maximum size in bytes of each payload.
- // Default: 1440 bytes.
- int max_payload_size_in_bytes;
-
- // Packet loss mode. Two different packet loss models are supported:
- // uniform or burst. This setting has no effect unless
- // packet_loss_probability is >0.
- // Default: uniform.
- PacketLossMode packet_loss_mode;
-
- // Packet loss probability. A value between 0.0 and 1.0 that defines the
- // probability of a packet being lost. 0.1 means 10% and so on.
- // Default: 0 (no loss).
- double packet_loss_probability;
-
- // Packet loss burst length. Defines how many packets will be lost in a burst
- // when a packet has been decided to be lost. Must be >=1. Default: 1.
- int packet_loss_burst_length;
-};
-
-// Class for simulating packet loss on the encoded frame data.
-// When a packet loss has occurred in a frame, the remaining data in that
-// frame is lost (even if burst length is only a single packet).
-// TODO(kjellander): Support discarding only individual packets in the frame
-// when CL 172001 has been submitted. This also requires a correct
-// fragmentation header to be passed to the decoder.
-//
-// To get a repeatable packet drop pattern, re-initialize the random seed
-// using InitializeRandomSeed before each test run.
-class PacketManipulator {
- public:
- virtual ~PacketManipulator() {}
-
- // Manipulates the data of the encoded_image to simulate parts being lost
- // during transport.
- // If packets are dropped from frame data, the completedFrame field will be
- // set to false.
- // Returns the number of packets being dropped.
- virtual int
- ManipulatePackets(webrtc::EncodedImage* encoded_image) = 0;
-};
-
-class PacketManipulatorImpl : public PacketManipulator {
- public:
- PacketManipulatorImpl(PacketReader* packet_reader,
- const NetworkingConfig& config,
- bool verbose);
- virtual ~PacketManipulatorImpl();
- virtual int ManipulatePackets(webrtc::EncodedImage* encoded_image);
- virtual void InitializeRandomSeed(unsigned int seed);
- protected:
- // Returns a uniformly distributed random value between 0.0 and 1.0
- virtual double RandomUniform();
- private:
- PacketReader* packet_reader_;
- const NetworkingConfig& config_;
- // Used to simulate a burst over several frames.
- int active_burst_packets_;
- CriticalSectionWrapper* critsect_;
- unsigned int random_seed_;
- bool verbose_;
-};
-
-} // namespace test
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PACKET_MANIPULATOR_H_
diff --git a/trunk/src/modules/video_coding/codecs/test/packet_manipulator_unittest.cc b/trunk/src/modules/video_coding/codecs/test/packet_manipulator_unittest.cc
deleted file mode 100644
index a5d8bc3..0000000
--- a/trunk/src/modules/video_coding/codecs/test/packet_manipulator_unittest.cc
+++ /dev/null
@@ -1,153 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/video_coding/codecs/test/packet_manipulator.h"
-
-#include <queue>
-
-#include "gtest/gtest.h"
-#include "modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "modules/video_coding/codecs/test/predictive_packet_manipulator.h"
-#include "testsupport/unittest_utils.h"
-#include "typedefs.h"
-
-namespace webrtc {
-namespace test {
-
-const double kNeverDropProbability = 0.0;
-const double kAlwaysDropProbability = 1.0;
-const int kBurstLength = 1;
-
-class PacketManipulatorTest: public PacketRelatedTest {
- protected:
- PacketReader packet_reader_;
- EncodedImage image_;
- NetworkingConfig drop_config_;
- NetworkingConfig no_drop_config_;
-
- PacketManipulatorTest() {
- image_._buffer = packet_data_;
- image_._length = kPacketDataLength;
- image_._size = kPacketDataLength;
-
- drop_config_.packet_size_in_bytes = kPacketSizeInBytes;
- drop_config_.packet_loss_probability = kAlwaysDropProbability;
- drop_config_.packet_loss_burst_length = kBurstLength;
- drop_config_.packet_loss_mode = kUniform;
-
- no_drop_config_.packet_size_in_bytes = kPacketSizeInBytes;
- no_drop_config_.packet_loss_probability = kNeverDropProbability;
- no_drop_config_.packet_loss_burst_length = kBurstLength;
- no_drop_config_.packet_loss_mode = kUniform;
- }
-
- virtual ~PacketManipulatorTest() {}
-
- void SetUp() {
- PacketRelatedTest::SetUp();
- }
-
- void TearDown() {
- PacketRelatedTest::TearDown();
- }
-
- void VerifyPacketLoss(int expected_nbr_packets_dropped,
- int actual_nbr_packets_dropped,
- int expected_packet_data_length,
- WebRtc_UWord8* expected_packet_data,
- EncodedImage& actual_image) {
- EXPECT_EQ(expected_nbr_packets_dropped, actual_nbr_packets_dropped);
- EXPECT_EQ(expected_packet_data_length, static_cast<int>(image_._length));
- EXPECT_EQ(0, memcmp(expected_packet_data, actual_image._buffer,
- expected_packet_data_length));
- }
-};
-
-TEST_F(PacketManipulatorTest, Constructor) {
- PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
-}
-
-TEST_F(PacketManipulatorTest, DropNone) {
- PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
- int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
- VerifyPacketLoss(0, nbr_packets_dropped, kPacketDataLength,
- packet_data_, image_);
-}
-
-TEST_F(PacketManipulatorTest, UniformDropNoneSmallFrame) {
- int data_length = 400; // smaller than the packet size
- image_._length = data_length;
- PacketManipulatorImpl manipulator(&packet_reader_, no_drop_config_, false);
- int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
-
- VerifyPacketLoss(0, nbr_packets_dropped, data_length,
- packet_data_, image_);
-}
-
-TEST_F(PacketManipulatorTest, UniformDropAll) {
- PacketManipulatorImpl manipulator(&packet_reader_, drop_config_, false);
- int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
- VerifyPacketLoss(kPacketDataNumberOfPackets, nbr_packets_dropped,
- 0, packet_data_, image_);
-}
-
-// Use our customized test class to make the second packet being lost
-TEST_F(PacketManipulatorTest, UniformDropSinglePacket) {
- drop_config_.packet_loss_probability = 0.5;
- PredictivePacketManipulator manipulator(&packet_reader_, drop_config_);
- manipulator.AddRandomResult(1.0);
- manipulator.AddRandomResult(0.3); // less than 0.5 will cause packet loss
- manipulator.AddRandomResult(1.0);
-
- // Execute the test target method:
- int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
-
- // Since we setup the predictive packet manipulator, it will throw away the
- // second packet. The third packet is also lost because when we have lost one,
- // the remains shall also be discarded (in the current implementation).
- VerifyPacketLoss(2, nbr_packets_dropped, kPacketSizeInBytes, packet1_,
- image_);
-}
-
-// Use our customized test class to make the second packet being lost
-TEST_F(PacketManipulatorTest, BurstDropNinePackets) {
- // Create a longer packet data structure (10 packets)
- const int kNbrPackets = 10;
- const int kDataLength = kPacketSizeInBytes * kNbrPackets;
- WebRtc_UWord8 data[kDataLength];
- WebRtc_UWord8* data_pointer = data;
- // Fill with 0s, 1s and so on to be able to easily verify which were dropped:
- for (int i = 0; i < kNbrPackets; ++i) {
- memset(data_pointer + i * kPacketSizeInBytes, i, kPacketSizeInBytes);
- }
- // Overwrite the defaults from the test fixture:
- image_._buffer = data;
- image_._length = kDataLength;
- image_._size = kDataLength;
-
- drop_config_.packet_loss_probability = 0.5;
- drop_config_.packet_loss_burst_length = 5;
- drop_config_.packet_loss_mode = kBurst;
- PredictivePacketManipulator manipulator(&packet_reader_, drop_config_);
- manipulator.AddRandomResult(1.0);
- manipulator.AddRandomResult(0.3); // less than 0.5 will cause packet loss
- for (int i = 0; i < kNbrPackets - 2; ++i) {
- manipulator.AddRandomResult(1.0);
- }
-
- // Execute the test target method:
- int nbr_packets_dropped = manipulator.ManipulatePackets(&image_);
-
- // Should discard every packet after the first one.
- VerifyPacketLoss(9, nbr_packets_dropped, kPacketSizeInBytes, data, image_);
-}
-
-} // namespace test
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/test/predictive_packet_manipulator.cc b/trunk/src/modules/video_coding/codecs/test/predictive_packet_manipulator.cc
deleted file mode 100644
index 5668378..0000000
--- a/trunk/src/modules/video_coding/codecs/test/predictive_packet_manipulator.cc
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/video_coding/codecs/test/predictive_packet_manipulator.h"
-
-#include <cassert>
-#include <cstdio>
-
-#include "testsupport/packet_reader.h"
-
-namespace webrtc {
-namespace test {
-
-PredictivePacketManipulator::PredictivePacketManipulator(
- PacketReader* packet_reader, const NetworkingConfig& config)
- : PacketManipulatorImpl(packet_reader, config, false) {
-}
-
-PredictivePacketManipulator::~PredictivePacketManipulator() {
-}
-
-
-void PredictivePacketManipulator::AddRandomResult(double result) {
- assert(result >= 0.0 && result <= 1.0);
- random_results_.push(result);
-}
-
-double PredictivePacketManipulator::RandomUniform() {
- if(random_results_.size() == 0u) {
- fprintf(stderr, "No more stored results, please make sure AddRandomResult()"
- "is called same amount of times you're going to invoke the "
- "RandomUniform() function, i.e. once per packet.\n");
- assert(false);
- }
- double result = random_results_.front();
- random_results_.pop();
- return result;
-}
-
-} // namespace test
-} // namespace webrtcc
diff --git a/trunk/src/modules/video_coding/codecs/test/predictive_packet_manipulator.h b/trunk/src/modules/video_coding/codecs/test/predictive_packet_manipulator.h
deleted file mode 100644
index 22a0ce9..0000000
--- a/trunk/src/modules/video_coding/codecs/test/predictive_packet_manipulator.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PREDICTIVE_PACKET_MANIPULATOR_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PREDICTIVE_PACKET_MANIPULATOR_H_
-
-#include <queue>
-
-#include "modules/video_coding/codecs/test/packet_manipulator.h"
-#include "testsupport/packet_reader.h"
-
-namespace webrtc {
-namespace test {
-
-// Predictive packet manipulator that allows for setup of the result of
-// the random invocations.
-class PredictivePacketManipulator : public PacketManipulatorImpl {
- public:
- PredictivePacketManipulator(PacketReader* packet_reader,
- const NetworkingConfig& config);
- virtual ~PredictivePacketManipulator();
- // Adds a result. You must add at least the same number of results as the
- // expected calls to the RandomUniform method. The results are added to a
- // FIFO queue so they will be returned in the same order they were added.
- // Result parameter must be 0.0 to 1.0.
- void AddRandomResult(double result);
- protected:
- // Returns a uniformly distributed random value between 0.0 and 1.0
- virtual double RandomUniform();
-
- private:
- std::queue<double> random_results_;
-};
-
-} // namespace test
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_PREDICTIVE_PACKET_MANIPULATOR_H_
diff --git a/trunk/src/modules/video_coding/codecs/test/stats.cc b/trunk/src/modules/video_coding/codecs/test/stats.cc
deleted file mode 100644
index 5cdf364..0000000
--- a/trunk/src/modules/video_coding/codecs/test/stats.cc
+++ /dev/null
@@ -1,172 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/video_coding/codecs/test/stats.h"
-
-#include <algorithm> // min_element, max_element
-#include <cassert>
-#include <cstdio>
-
-namespace webrtc {
-namespace test {
-
-Stats::Stats() {}
-
-Stats::~Stats() {}
-
-bool LessForEncodeTime(const FrameStatistic& s1, const FrameStatistic& s2) {
- return s1.encode_time_in_us < s2.encode_time_in_us;
-}
-
-bool LessForDecodeTime(const FrameStatistic& s1, const FrameStatistic& s2) {
- return s1.decode_time_in_us < s2.decode_time_in_us;
-}
-
-bool LessForEncodedSize(const FrameStatistic& s1, const FrameStatistic& s2) {
- return s1.encoded_frame_length_in_bytes < s2.encoded_frame_length_in_bytes;
-}
-
-bool LessForBitRate(const FrameStatistic& s1, const FrameStatistic& s2) {
- return s1.bit_rate_in_kbps < s2.bit_rate_in_kbps;
-}
-
-FrameStatistic& Stats::NewFrame(int frame_number) {
- assert(frame_number >= 0);
- FrameStatistic stat;
- stat.frame_number = frame_number;
- stats_.push_back(stat);
- return stats_[frame_number];
-}
-
-void Stats::PrintSummary() {
- printf("Processing summary:\n");
- if (stats_.size() == 0) {
- printf("No frame statistics have been logged yet.\n");
- return;
- }
-
- // Calculate min, max, average and total encoding time
- int total_encoding_time_in_us = 0;
- int total_decoding_time_in_us = 0;
- int total_encoded_frames_lengths = 0;
- int total_encoded_key_frames_lengths = 0;
- int total_encoded_nonkey_frames_lengths = 0;
- int nbr_keyframes = 0;
- int nbr_nonkeyframes = 0;
-
- for (FrameStatisticsIterator it = stats_.begin();
- it != stats_.end(); ++it) {
- total_encoding_time_in_us += it->encode_time_in_us;
- total_decoding_time_in_us += it->decode_time_in_us;
- total_encoded_frames_lengths += it->encoded_frame_length_in_bytes;
- if (it->frame_type == webrtc::kKeyFrame) {
- total_encoded_key_frames_lengths += it->encoded_frame_length_in_bytes;
- nbr_keyframes++;
- } else {
- total_encoded_nonkey_frames_lengths += it->encoded_frame_length_in_bytes;
- nbr_nonkeyframes++;
- }
- }
-
- FrameStatisticsIterator frame;
-
- // ENCODING
- printf("Encoding time:\n");
- frame = min_element(stats_.begin(),
- stats_.end(), LessForEncodeTime);
- printf(" Min : %7d us (frame %d)\n",
- frame->encode_time_in_us, frame->frame_number);
-
- frame = max_element(stats_.begin(),
- stats_.end(), LessForEncodeTime);
- printf(" Max : %7d us (frame %d)\n",
- frame->encode_time_in_us, frame->frame_number);
-
- printf(" Average : %7d us\n",
- static_cast<int>(total_encoding_time_in_us / stats_.size()));
-
- // DECODING
- printf("Decoding time:\n");
- // only consider frames that were successfully decoded (packet loss may cause
- // failures)
- std::vector<FrameStatistic> decoded_frames;
- for (std::vector<FrameStatistic>::iterator it = stats_.begin();
- it != stats_.end(); ++it) {
- if (it->decoding_successful) {
- decoded_frames.push_back(*it);
- }
- }
- if (decoded_frames.size() == 0) {
- printf("No successfully decoded frames exist in this statistics.\n");
- } else {
- frame = min_element(decoded_frames.begin(),
- decoded_frames.end(), LessForDecodeTime);
- printf(" Min : %7d us (frame %d)\n",
- frame->decode_time_in_us, frame->frame_number);
-
- frame = max_element(decoded_frames.begin(),
- decoded_frames.end(), LessForDecodeTime);
- printf(" Max : %7d us (frame %d)\n",
- frame->decode_time_in_us, frame->frame_number);
-
- printf(" Average : %7d us\n",
- static_cast<int>(total_decoding_time_in_us / decoded_frames.size()));
- printf(" Failures: %d frames failed to decode.\n",
- static_cast<int>(stats_.size() - decoded_frames.size()));
- }
-
- // SIZE
- printf("Frame sizes:\n");
- frame = min_element(stats_.begin(),
- stats_.end(), LessForEncodedSize);
- printf(" Min : %7d bytes (frame %d)\n",
- frame->encoded_frame_length_in_bytes, frame->frame_number);
-
- frame = max_element(stats_.begin(),
- stats_.end(), LessForEncodedSize);
- printf(" Max : %7d bytes (frame %d)\n",
- frame->encoded_frame_length_in_bytes, frame->frame_number);
-
- printf(" Average : %7d bytes\n",
- static_cast<int>(total_encoded_frames_lengths / stats_.size()));
- if (nbr_keyframes > 0) {
- printf(" Average key frame size : %7d bytes (%d keyframes)\n",
- total_encoded_key_frames_lengths / nbr_keyframes,
- nbr_keyframes);
- }
- if (nbr_nonkeyframes > 0) {
- printf(" Average non-key frame size: %7d bytes (%d frames)\n",
- total_encoded_nonkey_frames_lengths / nbr_nonkeyframes,
- nbr_nonkeyframes);
- }
-
- // BIT RATE
- printf("Bit rates:\n");
- frame = min_element(stats_.begin(),
- stats_.end(), LessForBitRate);
- printf(" Min bit rate: %7d kbps (frame %d)\n",
- frame->bit_rate_in_kbps, frame->frame_number);
-
- frame = max_element(stats_.begin(),
- stats_.end(), LessForBitRate);
- printf(" Max bit rate: %7d kbps (frame %d)\n",
- frame->bit_rate_in_kbps, frame->frame_number);
-
- printf("\n");
- printf("Total encoding time : %7d ms.\n",
- total_encoding_time_in_us / 1000);
- printf("Total decoding time : %7d ms.\n",
- total_decoding_time_in_us / 1000);
- printf("Total processing time: %7d ms.\n",
- (total_encoding_time_in_us + total_decoding_time_in_us) / 1000);
-}
-
-} // namespace test
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/test/stats.h b/trunk/src/modules/video_coding/codecs/test/stats.h
deleted file mode 100644
index ec2bb9d..0000000
--- a/trunk/src/modules/video_coding/codecs/test/stats.h
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_
-
-#include <vector>
-
-#include "common_video/interface/video_image.h"
-
-namespace webrtc {
-namespace test {
-
-// Contains statistics of a single frame that has been processed.
-struct FrameStatistic {
- FrameStatistic() :
- encoding_successful(false), decoding_successful(false),
- encode_return_code(0), decode_return_code(0),
- encode_time_in_us(0), decode_time_in_us(0),
- frame_number(0), packets_dropped(0), total_packets(0),
- bit_rate_in_kbps(0), encoded_frame_length_in_bytes(0),
- frame_type(kDeltaFrame) {
- };
- bool encoding_successful;
- bool decoding_successful;
- int encode_return_code;
- int decode_return_code;
- int encode_time_in_us;
- int decode_time_in_us;
- int frame_number;
- // How many packets were discarded of the encoded frame data (if any)
- int packets_dropped;
- int total_packets;
-
- // Current bit rate. Calculated out of the size divided with the time
- // interval per frame.
- int bit_rate_in_kbps;
-
- // Copied from EncodedImage
- int encoded_frame_length_in_bytes;
- webrtc::VideoFrameType frame_type;
-};
-
-// Handles statistics from a single video processing run.
-// Contains calculation methods for interesting metrics from these stats.
-class Stats {
- public:
- typedef std::vector<FrameStatistic>::iterator FrameStatisticsIterator;
-
- Stats();
- virtual ~Stats();
-
- // Add a new statistic data object.
- // The frame number must be incrementing and start at zero in order to use
- // it as an index for the frame_statistics_ vector.
- // Returns the newly created statistic object.
- FrameStatistic& NewFrame(int frame_number);
-
- // Prints a summary of all the statistics that have been gathered during the
- // processing
- void PrintSummary();
-
- std::vector<FrameStatistic> stats_;
-};
-
-} // namespace test
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_STATS_H_
diff --git a/trunk/src/modules/video_coding/codecs/test/stats_unittest.cc b/trunk/src/modules/video_coding/codecs/test/stats_unittest.cc
deleted file mode 100644
index 53a50d7..0000000
--- a/trunk/src/modules/video_coding/codecs/test/stats_unittest.cc
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/video_coding/codecs/test/stats.h"
-
-#include "gtest/gtest.h"
-#include "typedefs.h"
-
-namespace webrtc {
-namespace test {
-
-class StatsTest: public testing::Test {
- protected:
- StatsTest() {
- }
-
- virtual ~StatsTest() {
- }
-
- void SetUp() {
- stats_ = new Stats();
- }
-
- void TearDown() {
- delete stats_;
- }
-
- Stats* stats_;
-};
-
-// Test empty object
-TEST_F(StatsTest, Uninitialized) {
- EXPECT_EQ(0u, stats_->stats_.size());
- stats_->PrintSummary(); // should not crash
-}
-
-// Add single frame stats and verify
-TEST_F(StatsTest, AddOne) {
- stats_->NewFrame(0u);
- FrameStatistic* frameStat = &stats_->stats_[0];
- EXPECT_EQ(0, frameStat->frame_number);
-}
-
-// Add multiple frame stats and verify
-TEST_F(StatsTest, AddMany) {
- int nbr_of_frames = 1000;
- for (int i = 0; i < nbr_of_frames; ++i) {
- FrameStatistic& frameStat = stats_->NewFrame(i);
- EXPECT_EQ(i, frameStat.frame_number);
- }
- EXPECT_EQ(nbr_of_frames, static_cast<int>(stats_->stats_.size()));
-
- stats_->PrintSummary(); // should not crash
-}
-
-} // namespace test
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/test/video_codecs_test_framework.gypi b/trunk/src/modules/video_coding/codecs/test/video_codecs_test_framework.gypi
deleted file mode 100644
index 662d09d..0000000
--- a/trunk/src/modules/video_coding/codecs/test/video_codecs_test_framework.gypi
+++ /dev/null
@@ -1,72 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- # Exclude the test target when building with chromium.
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'video_codecs_test_framework',
- 'type': '<(library)',
- 'dependencies': [
- '<(webrtc_root)/../test/test.gyp:test_support',
- ],
- 'sources': [
- 'mock/mock_packet_manipulator.h',
- 'packet_manipulator.h',
- 'packet_manipulator.cc',
- 'predictive_packet_manipulator.h',
- 'predictive_packet_manipulator.cc',
- 'stats.h',
- 'stats.cc',
- 'videoprocessor.h',
- 'videoprocessor.cc',
- ],
- },
- {
- 'target_name': 'video_codecs_test_framework_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'video_codecs_test_framework',
- 'webrtc_video_coding',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/../testing/gmock.gyp:gmock',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- ],
- 'sources': [
- 'packet_manipulator_unittest.cc',
- 'stats_unittest.cc',
- 'videoprocessor_unittest.cc',
- ],
- },
- {
- 'target_name': 'video_codecs_test_framework_integrationtests',
- 'type': 'executable',
- 'dependencies': [
- 'video_codecs_test_framework',
- 'webrtc_video_coding',
- 'webrtc_vp8',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../test/metrics.gyp:metrics',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- ],
- 'sources': [
- 'videoprocessor_integrationtest.cc',
- ],
- },
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
\ No newline at end of file
diff --git a/trunk/src/modules/video_coding/codecs/test/videoprocessor.cc b/trunk/src/modules/video_coding/codecs/test/videoprocessor.cc
deleted file mode 100644
index 8b13513..0000000
--- a/trunk/src/modules/video_coding/codecs/test/videoprocessor.cc
+++ /dev/null
@@ -1,287 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/video_coding/codecs/test/videoprocessor.h"
-
-#include <cassert>
-#include <cstring>
-#include <limits>
-
-#include "system_wrappers/interface/cpu_info.h"
-
-namespace webrtc {
-namespace test {
-
-VideoProcessorImpl::VideoProcessorImpl(webrtc::VideoEncoder* encoder,
- webrtc::VideoDecoder* decoder,
- FrameReader* frame_reader,
- FrameWriter* frame_writer,
- PacketManipulator* packet_manipulator,
- const TestConfig& config,
- Stats* stats)
- : encoder_(encoder),
- decoder_(decoder),
- frame_reader_(frame_reader),
- frame_writer_(frame_writer),
- packet_manipulator_(packet_manipulator),
- config_(config),
- stats_(stats),
- encode_callback_(NULL),
- decode_callback_(NULL),
- source_buffer_(NULL),
- first_key_frame_has_been_excluded_(false),
- last_frame_missing_(false),
- initialized_(false) {
- assert(encoder);
- assert(decoder);
- assert(frame_reader);
- assert(frame_writer);
- assert(packet_manipulator);
- assert(stats);
-}
-
-bool VideoProcessorImpl::Init() {
- // Calculate a factor used for bit rate calculations:
- bit_rate_factor_ = config_.codec_settings->maxFramerate * 0.001 * 8; // bits
-
- int frame_length_in_bytes = frame_reader_->FrameLength();
-
- // Initialize data structures used by the encoder/decoder APIs
- source_buffer_ = new WebRtc_UWord8[frame_length_in_bytes];
- last_successful_frame_buffer_ = new WebRtc_UWord8[frame_length_in_bytes];
-
- // Set fixed properties common for all frames:
- source_frame_._width = config_.codec_settings->width;
- source_frame_._height = config_.codec_settings->height;
- source_frame_._length = frame_length_in_bytes;
- source_frame_._size = frame_length_in_bytes;
-
- // Setup required callbacks for the encoder/decoder:
- encode_callback_ = new VideoProcessorEncodeCompleteCallback(this);
- decode_callback_ = new VideoProcessorDecodeCompleteCallback(this);
- WebRtc_Word32 register_result =
- encoder_->RegisterEncodeCompleteCallback(encode_callback_);
- if (register_result != WEBRTC_VIDEO_CODEC_OK) {
- fprintf(stderr, "Failed to register encode complete callback, return code: "
- "%d\n", register_result);
- return false;
- }
- register_result = decoder_->RegisterDecodeCompleteCallback(decode_callback_);
- if (register_result != WEBRTC_VIDEO_CODEC_OK) {
- fprintf(stderr, "Failed to register decode complete callback, return code: "
- "%d\n", register_result);
- return false;
- }
- // Init the encoder and decoder
- WebRtc_UWord32 nbr_of_cores = 1;
- if (!config_.use_single_core) {
- nbr_of_cores = CpuInfo::DetectNumberOfCores();
- }
- WebRtc_Word32 init_result =
- encoder_->InitEncode(config_.codec_settings, nbr_of_cores,
- config_.networking_config.max_payload_size_in_bytes);
- if (init_result != WEBRTC_VIDEO_CODEC_OK) {
- fprintf(stderr, "Failed to initialize VideoEncoder, return code: %d\n",
- init_result);
- return false;
- }
- init_result = decoder_->InitDecode(config_.codec_settings, nbr_of_cores);
- if (init_result != WEBRTC_VIDEO_CODEC_OK) {
- fprintf(stderr, "Failed to initialize VideoDecoder, return code: %d\n",
- init_result);
- return false;
- }
-
- if (config_.verbose) {
- printf("Video Processor:\n");
- printf(" #CPU cores used : %d\n", nbr_of_cores);
- printf(" Total # of frames: %d\n", frame_reader_->NumberOfFrames());
- printf(" Codec settings:\n");
- printf(" Start bitrate : %d kbps\n",
- config_.codec_settings->startBitrate);
- printf(" Width : %d\n", config_.codec_settings->width);
- printf(" Height : %d\n", config_.codec_settings->height);
- }
- initialized_ = true;
- return true;
-}
-
-VideoProcessorImpl::~VideoProcessorImpl() {
- delete[] source_buffer_;
- delete[] last_successful_frame_buffer_;
- encoder_->RegisterEncodeCompleteCallback(NULL);
- delete encode_callback_;
- decoder_->RegisterDecodeCompleteCallback(NULL);
- delete decode_callback_;
-}
-
-bool VideoProcessorImpl::ProcessFrame(int frame_number) {
- assert(frame_number >=0);
- if (!initialized_) {
- fprintf(stderr, "Attempting to use uninitialized VideoProcessor!\n");
- return false;
- }
- if (frame_reader_->ReadFrame(source_buffer_)) {
- // point the source frame buffer to the newly read frame data:
- source_frame_._buffer = source_buffer_;
-
- // Ensure we have a new statistics data object we can fill:
- FrameStatistic& stat = stats_->NewFrame(frame_number);
-
- encode_start_ = TickTime::Now();
- // Use the frame number as "timestamp" to identify frames
- source_frame_._timeStamp = frame_number;
-
- // Decide if we're going to force a keyframe:
- VideoFrameType frame_type = kDeltaFrame;
- if (config_.keyframe_interval > 0 &&
- frame_number % config_.keyframe_interval == 0) {
- frame_type = kKeyFrame;
- }
- WebRtc_Word32 encode_result = encoder_->Encode(source_frame_, NULL,
- &frame_type);
- if (encode_result != WEBRTC_VIDEO_CODEC_OK) {
- fprintf(stderr, "Failed to encode frame %d, return code: %d\n",
- frame_number, encode_result);
- }
- stat.encode_return_code = encode_result;
- return true;
- } else {
- return false; // we've reached the last frame
- }
-}
-
-void VideoProcessorImpl::FrameEncoded(EncodedImage* encoded_image) {
- TickTime encode_stop = TickTime::Now();
- int frame_number = encoded_image->_timeStamp;
- FrameStatistic& stat = stats_->stats_[frame_number];
- stat.encode_time_in_us = GetElapsedTimeMicroseconds(encode_start_,
- encode_stop);
- stat.encoding_successful = true;
- stat.encoded_frame_length_in_bytes = encoded_image->_length;
- stat.frame_number = encoded_image->_timeStamp;
- stat.frame_type = encoded_image->_frameType;
- stat.bit_rate_in_kbps = encoded_image->_length * bit_rate_factor_;
- stat.total_packets = encoded_image->_length /
- config_.networking_config.packet_size_in_bytes + 1;
-
- // Perform packet loss if criteria is fullfilled:
- bool exclude_this_frame = false;
- // Only keyframes can be excluded
- if (encoded_image->_frameType == kKeyFrame) {
- switch (config_.exclude_frame_types) {
- case kExcludeOnlyFirstKeyFrame:
- if (!first_key_frame_has_been_excluded_) {
- first_key_frame_has_been_excluded_ = true;
- exclude_this_frame = true;
- }
- break;
- case kExcludeAllKeyFrames:
- exclude_this_frame = true;
- break;
- default:
- assert(false);
- }
- }
- if (!exclude_this_frame) {
- stat.packets_dropped =
- packet_manipulator_->ManipulatePackets(encoded_image);
- }
-
- // Keep track of if frames are lost due to packet loss so we can tell
- // this to the encoder (this is handled by the RTP logic in the full stack)
- decode_start_ = TickTime::Now();
- // TODO(kjellander): Pass fragmentation header to the decoder when
- // CL 172001 has been submitted and PacketManipulator supports this.
- WebRtc_Word32 decode_result = decoder_->Decode(*encoded_image,
- last_frame_missing_, NULL);
- stat.decode_return_code = decode_result;
- if (decode_result != WEBRTC_VIDEO_CODEC_OK) {
- // Write the last successful frame the output file to avoid getting it out
- // of sync with the source file for SSIM and PSNR comparisons:
- frame_writer_->WriteFrame(last_successful_frame_buffer_);
- }
- // save status for losses so we can inform the decoder for the next frame:
- last_frame_missing_ = encoded_image->_length == 0;
-}
-
-void VideoProcessorImpl::FrameDecoded(const RawImage& image) {
- TickTime decode_stop = TickTime::Now();
- int frame_number = image._timeStamp;
- // Report stats
- FrameStatistic& stat = stats_->stats_[frame_number];
- stat.decode_time_in_us = GetElapsedTimeMicroseconds(decode_start_,
- decode_stop);
- stat.decoding_successful = true;
- // Update our copy of the last successful frame:
- memcpy(last_successful_frame_buffer_, image._buffer, image._length);
-
- bool write_success = frame_writer_->WriteFrame(image._buffer);
- if (!write_success) {
- fprintf(stderr, "Failed to write frame %d to disk!", frame_number);
- }
-}
-
-int VideoProcessorImpl::GetElapsedTimeMicroseconds(
- const webrtc::TickTime& start, const webrtc::TickTime& stop) {
- WebRtc_UWord64 encode_time = (stop - start).Microseconds();
- assert(encode_time <
- static_cast<unsigned int>(std::numeric_limits<int>::max()));
- return static_cast<int>(encode_time);
-}
-
-const char* ExcludeFrameTypesToStr(ExcludeFrameTypes e) {
- switch (e) {
- case kExcludeOnlyFirstKeyFrame:
- return "ExcludeOnlyFirstKeyFrame";
- case kExcludeAllKeyFrames:
- return "ExcludeAllKeyFrames";
- default:
- assert(false);
- return "Unknown";
- }
-}
-
-const char* VideoCodecTypeToStr(webrtc::VideoCodecType e) {
- switch (e) {
- case kVideoCodecVP8:
- return "VP8";
- case kVideoCodecI420:
- return "I420";
- case kVideoCodecRED:
- return "RED";
- case kVideoCodecULPFEC:
- return "ULPFEC";
- case kVideoCodecUnknown:
- return "Unknown";
- default:
- assert(false);
- return "Unknown";
- }
-}
-
-// Callbacks
-WebRtc_Word32
-VideoProcessorImpl::VideoProcessorEncodeCompleteCallback::Encoded(
- EncodedImage& encoded_image,
- const webrtc::CodecSpecificInfo* codec_specific_info,
- const webrtc::RTPFragmentationHeader* fragmentation) {
- video_processor_->FrameEncoded(&encoded_image); // forward to parent class
- return 0;
-}
-WebRtc_Word32
-VideoProcessorImpl::VideoProcessorDecodeCompleteCallback::Decoded(
- RawImage& image) {
- video_processor_->FrameDecoded(image); // forward to parent class
- return 0;
-}
-
-} // namespace test
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/test/videoprocessor.h b/trunk/src/modules/video_coding/codecs/test/videoprocessor.h
deleted file mode 100644
index 7c2c14f..0000000
--- a/trunk/src/modules/video_coding/codecs/test/videoprocessor.h
+++ /dev/null
@@ -1,229 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_
-
-#include <string>
-
-#include "modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "modules/video_coding/codecs/test/packet_manipulator.h"
-#include "modules/video_coding/codecs/test/stats.h"
-#include "system_wrappers/interface/tick_util.h"
-#include "testsupport/frame_reader.h"
-#include "testsupport/frame_writer.h"
-
-namespace webrtc {
-namespace test {
-
-// Defines which frame types shall be excluded from packet loss and when.
-enum ExcludeFrameTypes {
- // Will exclude the first keyframe in the video sequence from packet loss.
- // Following keyframes will be targeted for packet loss.
- kExcludeOnlyFirstKeyFrame,
- // Exclude all keyframes from packet loss, no matter where in the video
- // sequence they occur.
- kExcludeAllKeyFrames
-};
-// Returns a string representation of the enum value.
-const char* ExcludeFrameTypesToStr(ExcludeFrameTypes e);
-
-// Test configuration for a test run
-struct TestConfig {
- TestConfig()
- : name(""), description(""), test_number(0),
- input_filename(""), output_filename(""), output_dir("out"),
- networking_config(), exclude_frame_types(kExcludeOnlyFirstKeyFrame),
- frame_length_in_bytes(-1), use_single_core(false), keyframe_interval(0),
- codec_settings(NULL), verbose(true) {
- };
-
- // Name of the test. This is purely metadata and does not affect
- // the test in any way.
- std::string name;
-
- // More detailed description of the test. This is purely metadata and does
- // not affect the test in any way.
- std::string description;
-
- // Number of this test. Useful if multiple runs of the same test with
- // different configurations shall be managed.
- int test_number;
-
- // File to process for the test. This must be a video file in the YUV format.
- std::string input_filename;
-
- // File to write to during processing for the test. Will be a video file
- // in the YUV format.
- std::string output_filename;
-
- // Path to the directory where encoded files will be put
- // (absolute or relative to the executable). Default: "out".
- std::string output_dir;
-
- // Configurations related to networking.
- NetworkingConfig networking_config;
-
- // Decides how the packet loss simulations shall exclude certain frames
- // from packet loss. Default: kExcludeOnlyFirstKeyFrame.
- ExcludeFrameTypes exclude_frame_types;
-
- // The length of a single frame of the input video file. This value is
- // calculated out of the width and height according to the video format
- // specification. Must be set before processing.
- int frame_length_in_bytes;
-
- // Force the encoder and decoder to use a single core for processing.
- // Using a single core is necessary to get a deterministic behavior for the
- // encoded frames - using multiple cores will produce different encoded frames
- // since multiple cores are competing to consume the byte budget for each
- // frame in parallel.
- // If set to false, the maximum number of available cores will be used.
- // Default: false.
- bool use_single_core;
-
- // If set to a value >0 this setting forces the encoder to create a keyframe
- // every Nth frame. Note that the encoder may create a keyframe in other
- // locations in addition to the interval that is set using this parameter.
- // Forcing key frames may also affect encoder planning optimizations in
- // a negative way, since it will suddenly be forced to produce an expensive
- // key frame.
- // Default: 0.
- int keyframe_interval;
-
- // The codec settings to use for the test (target bitrate, video size,
- // framerate and so on). This struct must be created and filled in using
- // the VideoCodingModule::Codec() method.
- webrtc::VideoCodec* codec_settings;
-
- // If printing of information to stdout shall be performed during processing.
- bool verbose;
-};
-
-// Returns a string representation of the enum value.
-const char* VideoCodecTypeToStr(webrtc::VideoCodecType e);
-
-// Handles encoding/decoding of video using the VideoEncoder/VideoDecoder
-// interfaces. This is done in a sequential manner in order to be able to
-// measure times properly.
-// The class processes a frame at the time for the configured input file.
-// It maintains state of where in the source input file the processing is at.
-//
-// Regarding packet loss: Note that keyframes are excluded (first or all
-// depending on the ExcludeFrameTypes setting). This is because if key frames
-// would be altered, all the following delta frames would be pretty much
-// worthless. VP8 has an error-resilience feature that makes it able to handle
-// packet loss in key non-first keyframes, which is why only the first is
-// excluded by default.
-// Packet loss in such important frames is handled on a higher level in the
-// Video Engine, where signaling would request a retransmit of the lost packets,
-// since they're so important.
-//
-// Note this class is not thread safe in any way and is meant for simple testing
-// purposes.
-class VideoProcessor {
- public:
- virtual ~VideoProcessor() {}
-
- // Performs initial calculations about frame size, sets up callbacks etc.
- // Returns false if an error has occurred, in addition to printing to stderr.
- virtual bool Init() = 0;
-
- // Processes a single frame. Returns true as long as there's more frames
- // available in the source clip.
- // Frame number must be an integer >=0.
- virtual bool ProcessFrame(int frame_number) = 0;
-};
-
-class VideoProcessorImpl : public VideoProcessor {
- public:
- VideoProcessorImpl(webrtc::VideoEncoder* encoder,
- webrtc::VideoDecoder* decoder,
- FrameReader* frame_reader,
- FrameWriter* frame_writer,
- PacketManipulator* packet_manipulator,
- const TestConfig& config,
- Stats* stats);
- virtual ~VideoProcessorImpl();
- virtual bool Init();
- virtual bool ProcessFrame(int frame_number);
-
- private:
- // Invoked by the callback when a frame has completed encoding.
- void FrameEncoded(EncodedImage* encodedImage);
- // Invoked by the callback when a frame has completed decoding.
- void FrameDecoded(const RawImage& image);
- // Used for getting a 32-bit integer representing time
- // (checks the size is within signed 32-bit bounds before casting it)
- int GetElapsedTimeMicroseconds(const webrtc::TickTime& start,
- const webrtc::TickTime& stop);
-
- webrtc::VideoEncoder* encoder_;
- webrtc::VideoDecoder* decoder_;
- FrameReader* frame_reader_;
- FrameWriter* frame_writer_;
- PacketManipulator* packet_manipulator_;
- const TestConfig& config_;
- Stats* stats_;
-
- EncodedImageCallback* encode_callback_;
- DecodedImageCallback* decode_callback_;
- // Buffer used for reading the source video file:
- WebRtc_UWord8* source_buffer_;
- // Keep track of the last successful frame, since we need to write that
- // when decoding fails:
- WebRtc_UWord8* last_successful_frame_buffer_;
- webrtc::RawImage source_frame_;
- // To keep track of if we have excluded the first key frame from packet loss:
- bool first_key_frame_has_been_excluded_;
- // To tell the decoder previous frame have been dropped due to packet loss:
- bool last_frame_missing_;
- // If Init() has executed successfully.
- bool initialized_;
-
- // Statistics
- double bit_rate_factor_; // multiply frame length with this to get bit rate
- webrtc::TickTime encode_start_;
- webrtc::TickTime decode_start_;
-
- // Callback class required to implement according to the VideoEncoder API.
- class VideoProcessorEncodeCompleteCallback
- : public webrtc::EncodedImageCallback {
- public:
- explicit VideoProcessorEncodeCompleteCallback(VideoProcessorImpl* vp)
- : video_processor_(vp) {
- }
- WebRtc_Word32 Encoded(
- webrtc::EncodedImage& encoded_image,
- const webrtc::CodecSpecificInfo* codec_specific_info = NULL,
- const webrtc::RTPFragmentationHeader* fragmentation = NULL);
-
- private:
- VideoProcessorImpl* video_processor_;
- };
-
- // Callback class required to implement according to the VideoDecoder API.
- class VideoProcessorDecodeCompleteCallback
- : public webrtc::DecodedImageCallback {
- public:
- explicit VideoProcessorDecodeCompleteCallback(VideoProcessorImpl* vp)
- : video_processor_(vp) {
- }
- WebRtc_Word32 Decoded(webrtc::RawImage& image);
-
- private:
- VideoProcessorImpl* video_processor_;
- };
-};
-
-} // namespace test
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_VIDEOPROCESSOR_H_
diff --git a/trunk/src/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc b/trunk/src/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
deleted file mode 100644
index 19c40ad..0000000
--- a/trunk/src/modules/video_coding/codecs/test/videoprocessor_integrationtest.cc
+++ /dev/null
@@ -1,173 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "gtest/gtest.h"
-#include "modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "modules/video_coding/codecs/test/packet_manipulator.h"
-#include "modules/video_coding/codecs/test/videoprocessor.h"
-#include "modules/video_coding/codecs/vp8/main/interface/vp8.h"
-#include "modules/video_coding/main/interface/video_coding.h"
-#include "testsupport/fileutils.h"
-#include "testsupport/frame_reader.h"
-#include "testsupport/frame_writer.h"
-#include "testsupport/metrics/video_metrics.h"
-#include "testsupport/packet_reader.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-const int kNbrFrames = 61; // foreman_cif_short.yuv
-const int kCIFWidth = 352;
-const int kCIFHeight = 288;
-const int kBitRateKbps = 500;
-
-// Integration test for video processor. Encodes+decodes a small clip and
-// writes it to the output directory. After completion, PSNR and SSIM
-// measurements are performed on the original and the processed clip to verify
-// the quality is acceptable.
-// The limits for the PSNR and SSIM values must be set quite low, since we have
-// no control over the random function used for packet loss in this test.
-class VideoProcessorIntegrationTest: public testing::Test {
- protected:
- VideoEncoder* encoder_;
- VideoDecoder* decoder_;
- webrtc::test::FrameReader* frame_reader_;
- webrtc::test::FrameWriter* frame_writer_;
- webrtc::test::PacketReader packet_reader_;
- webrtc::test::PacketManipulator* packet_manipulator_;
- webrtc::test::Stats stats_;
- webrtc::test::TestConfig config_;
- VideoCodec codec_settings_;
- webrtc::test::VideoProcessor* processor_;
-
- VideoProcessorIntegrationTest() {}
- virtual ~VideoProcessorIntegrationTest() {}
-
- void SetUp() {
- encoder_ = VP8Encoder::Create();
- decoder_ = VP8Decoder::Create();
-
- // Setup the TestConfig struct for processing of a clip in CIF resolution.
- config_.input_filename =
- webrtc::test::ResourcePath("foreman_cif_short", "yuv");
- config_.output_filename = webrtc::test::OutputPath() +
- "foreman_cif_short_video_codecs_test_framework_integrationtests.yuv";
- config_.frame_length_in_bytes = 3 * kCIFWidth * kCIFHeight / 2;
- config_.verbose = false;
- // Only allow encoder/decoder to use single core, for predictability.
- config_.use_single_core = true;
-
- // Get a codec configuration struct and configure it.
- VideoCodingModule::Codec(kVideoCodecVP8, &codec_settings_);
- config_.codec_settings = &codec_settings_;
- config_.codec_settings->startBitrate = kBitRateKbps;
- config_.codec_settings->width = kCIFWidth;
- config_.codec_settings->height = kCIFHeight;
-
- frame_reader_ =
- new webrtc::test::FrameReaderImpl(config_.input_filename,
- config_.frame_length_in_bytes);
- frame_writer_ =
- new webrtc::test::FrameWriterImpl(config_.output_filename,
- config_.frame_length_in_bytes);
- ASSERT_TRUE(frame_reader_->Init());
- ASSERT_TRUE(frame_writer_->Init());
-
- packet_manipulator_ = new webrtc::test::PacketManipulatorImpl(
- &packet_reader_, config_.networking_config, config_.verbose);
- processor_ = new webrtc::test::VideoProcessorImpl(encoder_, decoder_,
- frame_reader_,
- frame_writer_,
- packet_manipulator_,
- config_, &stats_);
- ASSERT_TRUE(processor_->Init());
- }
-
- void TearDown() {
- delete processor_;
- delete packet_manipulator_;
- delete frame_writer_;
- delete frame_reader_;
- delete decoder_;
- delete encoder_;
- }
-
- // Processes all frames in the clip and verifies the result.
- void ProcessFramesAndVerify(double minimum_avg_psnr,
- double minimum_min_psnr,
- double minimum_avg_ssim,
- double minimum_min_ssim) {
- int frame_number = 0;
- while (processor_->ProcessFrame(frame_number)) {
- frame_number++;
- }
- EXPECT_EQ(kNbrFrames, frame_number);
- EXPECT_EQ(kNbrFrames, static_cast<int>(stats_.stats_.size()));
-
- // Release encoder and decoder to make sure they have finished processing:
- EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, encoder_->Release());
- EXPECT_EQ(WEBRTC_VIDEO_CODEC_OK, decoder_->Release());
- // Close the files before we start using them for SSIM/PSNR calculations.
- frame_reader_->Close();
- frame_writer_->Close();
-
- webrtc::test::QualityMetricsResult psnr_result, ssim_result;
- EXPECT_EQ(0, webrtc::test::I420MetricsFromFiles(
- config_.input_filename.c_str(),
- config_.output_filename.c_str(),
- config_.codec_settings->width,
- config_.codec_settings->height,
- &psnr_result,
- &ssim_result));
- printf("PSNR avg: %f, min: %f SSIM avg: %f, min: %f\n",
- psnr_result.average, psnr_result.min,
- ssim_result.average, ssim_result.min);
- EXPECT_GT(psnr_result.average, minimum_avg_psnr);
- EXPECT_GT(psnr_result.min, minimum_min_psnr);
- EXPECT_GT(ssim_result.average, minimum_avg_ssim);
- EXPECT_GT(ssim_result.min, minimum_min_ssim);
- }
-};
-
-// Run with no packet loss. Quality should be very high.
-TEST_F(VideoProcessorIntegrationTest, ProcessZeroPacketLoss) {
- config_.networking_config.packet_loss_probability = 0;
- double minimum_avg_psnr = 36;
- double minimum_min_psnr = 34;
- double minimum_avg_ssim = 0.9;
- double minimum_min_ssim = 0.9;
- ProcessFramesAndVerify(minimum_avg_psnr, minimum_min_psnr,
- minimum_avg_ssim, minimum_min_ssim);
-}
-
-// Run with 5% packet loss. Quality should be a bit lower.
-// TODO(mflodman): Reenable this once it's not flaky.
-TEST_F(VideoProcessorIntegrationTest, Process5PercentPacketLoss) {
- config_.networking_config.packet_loss_probability = 0.05;
- double minimum_avg_psnr = 21;
- double minimum_min_psnr = 17;
- double minimum_avg_ssim = 0.6;
- double minimum_min_ssim = 0.4;
- ProcessFramesAndVerify(minimum_avg_psnr, minimum_min_psnr,
- minimum_avg_ssim, minimum_min_ssim);
-}
-
-// Run with 10% packet loss. Quality should be even lower.
-TEST_F(VideoProcessorIntegrationTest, Process10PercentPacketLoss) {
- config_.networking_config.packet_loss_probability = 0.10;
- double minimum_avg_psnr = 19;
- double minimum_min_psnr = 16;
- double minimum_avg_ssim = 0.6;
- double minimum_min_ssim = 0.4;
- ProcessFramesAndVerify(minimum_avg_psnr, minimum_min_psnr,
- minimum_avg_ssim, minimum_min_ssim);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/test/videoprocessor_unittest.cc b/trunk/src/modules/video_coding/codecs/test/videoprocessor_unittest.cc
deleted file mode 100644
index d51ef6b..0000000
--- a/trunk/src/modules/video_coding/codecs/test/videoprocessor_unittest.cc
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "gtest/gtest.h"
-#include "gmock/gmock.h"
-#include "modules/video_coding/codecs/test/mock/mock_packet_manipulator.h"
-#include "modules/video_coding/codecs/test/videoprocessor.h"
-#include "modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h"
-#include "modules/video_coding/main/interface/video_coding.h"
-#include "testsupport/mock/mock_frame_reader.h"
-#include "testsupport/mock/mock_frame_writer.h"
-#include "testsupport/packet_reader.h"
-#include "testsupport/unittest_utils.h"
-#include "typedefs.h"
-
-using ::testing::_;
-using ::testing::AtLeast;
-using ::testing::Return;
-
-namespace webrtc {
-namespace test {
-
-// Very basic testing for VideoProcessor. It's mostly tested by running the
-// video_quality_measurement program.
-class VideoProcessorTest: public testing::Test {
- protected:
- MockVideoEncoder encoder_mock_;
- MockVideoDecoder decoder_mock_;
- MockFrameReader frame_reader_mock_;
- MockFrameWriter frame_writer_mock_;
- MockPacketManipulator packet_manipulator_mock_;
- Stats stats_;
- TestConfig config_;
- VideoCodec codec_settings_;
-
- VideoProcessorTest() {}
- virtual ~VideoProcessorTest() {}
- void SetUp() {
- // Get a codec configuration struct and configure it.
- VideoCodingModule::Codec(kVideoCodecVP8, &codec_settings_);
- config_.codec_settings = &codec_settings_;
- config_.codec_settings->startBitrate = 100;
- config_.codec_settings->width = 352;
- config_.codec_settings->height = 288;
- }
- void TearDown() {}
-
- void ExpectInit() {
- EXPECT_CALL(encoder_mock_, InitEncode(_, _, _))
- .Times(1);
- EXPECT_CALL(encoder_mock_, RegisterEncodeCompleteCallback(_))
- .Times(AtLeast(1));
- EXPECT_CALL(decoder_mock_, InitDecode(_, _))
- .Times(1);
- EXPECT_CALL(decoder_mock_, RegisterDecodeCompleteCallback(_))
- .Times(AtLeast(1));
- EXPECT_CALL(frame_reader_mock_, NumberOfFrames())
- .WillOnce(Return(1));
- EXPECT_CALL(frame_reader_mock_, FrameLength())
- .WillOnce(Return(150000));
- }
-};
-
-TEST_F(VideoProcessorTest, Init) {
- ExpectInit();
- VideoProcessorImpl video_processor(&encoder_mock_, &decoder_mock_,
- &frame_reader_mock_,
- &frame_writer_mock_,
- &packet_manipulator_mock_, config_,
- &stats_);
- ASSERT_TRUE(video_processor.Init());
-}
-
-TEST_F(VideoProcessorTest, ProcessFrame) {
- ExpectInit();
- EXPECT_CALL(encoder_mock_, Encode(_, _, _))
- .Times(1);
- EXPECT_CALL(frame_reader_mock_, ReadFrame(_))
- .WillOnce(Return(true));
- // Since we don't return any callback from the mock, the decoder will not
- // be more than initialized...
- VideoProcessorImpl video_processor(&encoder_mock_, &decoder_mock_,
- &frame_reader_mock_,
- &frame_writer_mock_,
- &packet_manipulator_mock_, config_,
- &stats_);
- ASSERT_TRUE(video_processor.Init());
- video_processor.ProcessFrame(0);
-}
-
-} // namespace test
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/benchmark.cc b/trunk/src/modules/video_coding/codecs/test_framework/benchmark.cc
deleted file mode 100644
index 883330c..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/benchmark.cc
+++ /dev/null
@@ -1,310 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "benchmark.h"
-
-#include <cassert>
-#include <iostream>
-#include <sstream>
-#include <vector>
-#if defined(_WIN32)
- #include <windows.h>
-#endif
-
-#include "common_video/libyuv/include/libyuv.h"
-#include "system_wrappers/interface/event_wrapper.h"
-#include "modules/video_coding/codecs/test_framework/video_source.h"
-#include "testsupport/fileutils.h"
-#include "testsupport/metrics/video_metrics.h"
-
-#define SSIM_CALC 0 // by default, don't compute SSIM
-
-using namespace webrtc;
-
-Benchmark::Benchmark()
-:
-NormalAsyncTest("Benchmark", "Codec benchmark over a range of test cases", 6),
-_resultsFileName(webrtc::test::OutputPath() + "benchmark.txt"),
-_codecName("Default")
-{
-}
-
-Benchmark::Benchmark(std::string name, std::string description)
-:
-NormalAsyncTest(name, description, 6),
-_resultsFileName(webrtc::test::OutputPath() + "benchmark.txt"),
-_codecName("Default")
-{
-}
-
-Benchmark::Benchmark(std::string name, std::string description, std::string resultsFileName, std::string codecName)
-:
-NormalAsyncTest(name, description, 6),
-_resultsFileName(resultsFileName),
-_codecName(codecName)
-{
-}
-
-void
-Benchmark::Perform()
-{
- std::vector<const VideoSource*> sources;
- std::vector<const VideoSource*>::iterator it;
-
- // Configuration --------------------------
- sources.push_back(new const VideoSource(webrtc::test::ProjectRootPath() +
- "resources/foreman_cif.yuv", kCIF));
-// sources.push_back(new const VideoSource(webrtc::test::ProjectRootPath() +
-// "resources/akiyo_cif.yuv", kCIF));
-
- const VideoSize size[] = {kQCIF, kCIF};
- const int frameRate[] = {10, 15, 30};
- // Specifies the framerates for which to perform a speed test.
- const bool speedTestMask[] = {false, false, false};
- const int bitRate[] = {50, 100, 200, 300, 400, 500, 600, 1000};
- // Determines the number of iterations to perform to arrive at the speed result.
- enum { kSpeedTestIterations = 10 };
- // ----------------------------------------
-
- const int nFrameRates = sizeof(frameRate)/sizeof(*frameRate);
- assert(sizeof(speedTestMask)/sizeof(*speedTestMask) == nFrameRates);
- const int nBitrates = sizeof(bitRate)/sizeof(*bitRate);
- int testIterations = 10;
-
- webrtc::test::QualityMetricsResult psnr[nBitrates];
- webrtc::test::QualityMetricsResult ssim[nBitrates];
- double fps[nBitrates];
- double totalEncodeTime[nBitrates];
- double totalDecodeTime[nBitrates];
-
- _results.open(_resultsFileName.c_str(), std::fstream::out);
- _results << GetMagicStr() << std::endl;
- _results << _codecName << std::endl;
-
- for (it = sources.begin() ; it < sources.end(); it++)
- {
- for (int i = 0; i < static_cast<int>(sizeof(size)/sizeof(*size)); i++)
- {
- for (int j = 0; j < nFrameRates; j++)
- {
- std::stringstream ss;
- std::string strFrameRate;
- std::string outFileName;
- ss << frameRate[j];
- ss >> strFrameRate;
- outFileName = (*it)->GetFilePath() + "/" + (*it)->GetName() + "_" +
- VideoSource::GetSizeString(size[i]) + "_" + strFrameRate + ".yuv";
-
- _target = new const VideoSource(outFileName, size[i], frameRate[j]);
- (*it)->Convert(*_target);
- if (VideoSource::FileExists(outFileName.c_str()))
- {
- _inname = outFileName;
- }
- else
- {
- _inname = (*it)->GetFileName();
- }
-
- std::cout << (*it)->GetName() << ", " << VideoSource::GetSizeString(size[i])
- << ", " << frameRate[j] << " fps" << std::endl << "Bitrate [kbps]:";
- _results << (*it)->GetName() << "," << VideoSource::GetSizeString(size[i])
- << "," << frameRate[j] << " fps" << std::endl << "Bitrate [kbps]";
-
- if (speedTestMask[j])
- {
- testIterations = kSpeedTestIterations;
- }
- else
- {
- testIterations = 1;
- }
-
- for (int k = 0; k < nBitrates; k++)
- {
- _bitRate = (bitRate[k]);
- double avgFps = 0.0;
- totalEncodeTime[k] = 0;
- totalDecodeTime[k] = 0;
-
- for (int l = 0; l < testIterations; l++)
- {
- PerformNormalTest();
- _appendNext = false;
-
- avgFps += _framecnt / (_totalEncodeTime + _totalDecodeTime);
- totalEncodeTime[k] += _totalEncodeTime;
- totalDecodeTime[k] += _totalDecodeTime;
-
- }
- avgFps /= testIterations;
- totalEncodeTime[k] /= testIterations;
- totalDecodeTime[k] /= testIterations;
-
- double actualBitRate = ActualBitRate(_framecnt) / 1000.0;
- std::cout << " " << actualBitRate;
- _results << "," << actualBitRate;
- webrtc::test::QualityMetricsResult psnr_result;
- I420PSNRFromFiles(_inname.c_str(), _outname.c_str(),
- _inst.width, _inst.height, &psnr[k]);
- if (SSIM_CALC)
- {
- webrtc::test::QualityMetricsResult ssim_result;
- I420SSIMFromFiles(_inname.c_str(), _outname.c_str(),
- _inst.width, _inst.height, &ssim[k]);
-
- }
- fps[k] = avgFps;
- }
- std::cout << std::endl << "Y-PSNR [dB]:";
- _results << std::endl << "Y-PSNR [dB]";
- for (int k = 0; k < nBitrates; k++)
- {
- std::cout << " " << psnr[k].average;
- _results << "," << psnr[k].average;
-
- }
- if (SSIM_CALC)
- {
- std::cout << std::endl << "SSIM: ";
- _results << std::endl << "SSIM ";
- for (int k = 0; k < nBitrates; k++)
- {
- std::cout << " " << ssim[k].average;
- _results << "," << ssim[k].average;
- }
-
- }
-
- std::cout << std::endl << "Encode Time[ms]:";
- _results << std::endl << "Encode Time[ms]";
- for (int k = 0; k < nBitrates; k++)
- {
- std::cout << " " << totalEncodeTime[k];
- _results << "," << totalEncodeTime[k];
-
- }
-
- std::cout << std::endl << "Decode Time[ms]:";
- _results << std::endl << "Decode Time[ms]";
- for (int k = 0; k < nBitrates; k++)
- {
- std::cout << " " << totalDecodeTime[k];
- _results << "," << totalDecodeTime[k];
-
- }
-
- if (speedTestMask[j])
- {
- std::cout << std::endl << "Speed [fps]:";
- _results << std::endl << "Speed [fps]";
- for (int k = 0; k < nBitrates; k++)
- {
- std::cout << " " << static_cast<int>(fps[k] + 0.5);
- _results << "," << static_cast<int>(fps[k] + 0.5);
- }
- }
- std::cout << std::endl << std::endl;
- _results << std::endl << std::endl;
-
- delete _target;
- }
- }
- delete *it;
- }
- _results.close();
-}
-
-void
-Benchmark::PerformNormalTest()
-{
- _encoder = GetNewEncoder();
- _decoder = GetNewDecoder();
- CodecSettings(_target->GetWidth(), _target->GetHeight(), _target->GetFrameRate(), _bitRate);
- Setup();
- EventWrapper* waitEvent = EventWrapper::Create();
-
- _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
- _decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
- _encoder->InitEncode(&_inst, 4, 1440);
- CodecSpecific_InitBitrate();
- _decoder->InitDecode(&_inst,1);
-
- FrameQueue frameQueue;
- VideoEncodeCompleteCallback encCallback(_encodedFile, &frameQueue, *this);
- VideoDecodeCompleteCallback decCallback(_decodedFile, *this);
- _encoder->RegisterEncodeCompleteCallback(&encCallback);
- _decoder->RegisterDecodeCompleteCallback(&decCallback);
-
- SetCodecSpecificParameters();
-
- _totalEncodeTime = _totalDecodeTime = 0;
- _totalEncodePipeTime = _totalDecodePipeTime = 0;
- bool complete = false;
- _framecnt = 0;
- _encFrameCnt = 0;
- _sumEncBytes = 0;
- _lengthEncFrame = 0;
- while (!complete)
- {
- complete = Encode();
- if (!frameQueue.Empty() || complete)
- {
- while (!frameQueue.Empty())
- {
- _frameToDecode = static_cast<FrameQueueTuple *>(frameQueue.PopFrame());
- DoPacketLoss();
- int ret = Decode();
- delete _frameToDecode;
- _frameToDecode = NULL;
- if (ret < 0)
- {
- fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
- exit(EXIT_FAILURE);
- }
- else if (ret == 0)
- {
- _framecnt++;
- }
- else
- {
- fprintf(stderr, "\n\nPositive return value from decode!\n\n");
- }
- }
- }
- waitEvent->Wait(5);
- }
-
- _inputVideoBuffer.Free();
- //_encodedVideoBuffer.Reset(); ?
- _encodedVideoBuffer.Free();
- _decodedVideoBuffer.Free();
-
- _encoder->Release();
- _decoder->Release();
- delete waitEvent;
- delete _encoder;
- delete _decoder;
- Teardown();
-}
-
-void
-Benchmark::CodecSpecific_InitBitrate()
-{
- if (_bitRate == 0)
- {
- _encoder->SetRates(600, _inst.maxFramerate);
- }
- else
- {
- _encoder->SetRates(_bitRate, _inst.maxFramerate);
- }
-}
-
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/benchmark.h b/trunk/src/modules/video_coding/codecs/test_framework/benchmark.h
deleted file mode 100644
index 57806e5..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/benchmark.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_BENCHMARK_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_BENCHMARK_H_
-
-#include "normal_async_test.h"
-
-class VideoSource;
-
-class Benchmark : public NormalAsyncTest
-{
-public:
- Benchmark();
- virtual void Perform();
-
-protected:
- Benchmark(std::string name, std::string description);
- Benchmark(std::string name, std::string description, std::string resultsFileName, std::string codecName);
- virtual webrtc::VideoEncoder* GetNewEncoder() = 0;
- virtual webrtc::VideoDecoder* GetNewDecoder() = 0;
- virtual void PerformNormalTest();
- virtual void CodecSpecific_InitBitrate();
- static const char* GetMagicStr() { return "#!benchmark1.0"; }
-
- const VideoSource* _target;
- std::string _resultsFileName;
- std::ofstream _results;
- std::string _codecName;
-};
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_BENCHMARK_H_
-
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/exportfig.m b/trunk/src/modules/video_coding/codecs/test_framework/exportfig.m
deleted file mode 100644
index d0d5ed9..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/exportfig.m
+++ /dev/null
@@ -1,500 +0,0 @@
-function exportfig(varargin)
-%EXPORTFIG Export a figure to Encapsulated Postscript.
-% EXPORTFIG(H, FILENAME) writes the figure H to FILENAME. H is
-% a figure handle and FILENAME is a string that specifies the
-% name of the output file.
-%
-% EXPORTFIG(...,PARAM1,VAL1,PARAM2,VAL2,...) specifies
-% parameters that control various characteristics of the output
-% file.
-%
-% Format Paramter:
-% 'Format' one of the strings 'eps','eps2','jpeg','png','preview'
-% specifies the output format. Defaults to 'eps'.
-% The output format 'preview' does not generate an output
-% file but instead creates a new figure window with a
-% preview of the exported figure. In this case the
-% FILENAME parameter is ignored.
-%
-% 'Preview' one of the strings 'none', 'tiff'
-% specifies a preview for EPS files. Defaults to 'none'.
-%
-% Size Parameters:
-% 'Width' a positive scalar
-% specifies the width in the figure's PaperUnits
-% 'Height' a positive scalar
-% specifies the height in the figure's PaperUnits
-%
-% Specifying only one dimension sets the other dimension
-% so that the exported aspect ratio is the same as the
-% figure's current aspect ratio.
-% If neither dimension is specified the size defaults to
-% the width and height from the figure's PaperPosition.
-%
-% Rendering Parameters:
-% 'Color' one of the strings 'bw', 'gray', 'cmyk'
-% 'bw' specifies that lines and text are exported in
-% black and all other objects in grayscale
-% 'gray' specifies that all objects are exported in grayscale
-% 'cmyk' specifies that all objects are exported in color
-% using the CMYK color space
-% 'Renderer' one of the strings 'painters', 'zbuffer', 'opengl'
-% specifies the renderer to use
-% 'Resolution' a positive scalar
-% specifies the resolution in dots-per-inch.
-%
-% The default color setting is 'bw'.
-%
-% Font Parameters:
-% 'FontMode' one of the strings 'scaled', 'fixed'
-% 'FontSize' a positive scalar
-% in 'scaled' mode multiplies with the font size of each
-% text object to obtain the exported font size
-% in 'fixed' mode specifies the font size of all text
-% objects in points
-% 'FontEncoding' one of the strings 'latin1', 'adobe'
-% specifies the character encoding of the font
-%
-% If FontMode is 'scaled' but FontSize is not specified then a
-% scaling factor is computed from the ratio of the size of the
-% exported figure to the size of the actual figure. The minimum
-% font size allowed after scaling is 5 points.
-% If FontMode is 'fixed' but FontSize is not specified then the
-% exported font sizes of all text objects is 7 points.
-%
-% The default 'FontMode' setting is 'scaled'.
-%
-% Line Width Parameters:
-% 'LineMode' one of the strings 'scaled', 'fixed'
-% 'LineWidth' a positive scalar
-% the semantics of LineMode and LineWidth are exactly the
-% same as FontMode and FontSize, except that they apply
-% to line widths instead of font sizes. The minumum line
-% width allowed after scaling is 0.5 points.
-% If LineMode is 'fixed' but LineWidth is not specified
-% then the exported line width of all line objects is 1
-% point.
-%
-% Examples:
-% exportfig(gcf,'fig1.eps','height',3);
-% Exports the current figure to the file named 'fig1.eps' with
-% a height of 3 inches (assuming the figure's PaperUnits is
-% inches) and an aspect ratio the same as the figure's aspect
-% ratio on screen.
-%
-% exportfig(gcf, 'fig2.eps', 'FontMode', 'fixed',...
-% 'FontSize', 10, 'color', 'cmyk' );
-% Exports the current figure to 'fig2.eps' in color with all
-% text in 10 point fonts. The size of the exported figure is
-% the figure's PaperPostion width and height.
-
-
-if (nargin < 2)
- error('Too few input arguments');
-end
-
-% exportfig(H, filename, ...)
-H = varargin{1};
-if ~ishandle(H) | ~strcmp(get(H,'type'), 'figure')
- error('First argument must be a handle to a figure.');
-end
-filename = varargin{2};
-if ~ischar(filename)
- error('Second argument must be a string.');
-end
-paramPairs = varargin(3:end);
-
-% Do some validity checking on param-value pairs
-if (rem(length(paramPairs),2) ~= 0)
- error(['Invalid input syntax. Optional parameters and values' ...
- ' must be in pairs.']);
-end
-
-format = 'eps';
-preview = 'none';
-width = -1;
-height = -1;
-color = 'bw';
-fontsize = -1;
-fontmode='scaled';
-linewidth = -1;
-linemode=[];
-fontencoding = 'latin1';
-renderer = [];
-resolution = [];
-
-% Process param-value pairs
-args = {};
-for k = 1:2:length(paramPairs)
- param = lower(paramPairs{k});
- if (~ischar(param))
- error('Optional parameter names must be strings');
- end
- value = paramPairs{k+1};
-
- switch (param)
- case 'format'
- format = value;
- if (~strcmp(format,{'eps','eps2','jpeg','png','preview'}))
- error(['Format must be ''eps'', ''eps2'', ''jpeg'', ''png'' or' ...
- ' ''preview''.']);
- end
- case 'preview'
- preview = value;
- if (~strcmp(preview,{'none','tiff'}))
- error('Preview must be ''none'' or ''tiff''.');
- end
- case 'width'
- width = LocalToNum(value);
- if(~LocalIsPositiveScalar(width))
- error('Width must be a numeric scalar > 0');
- end
- case 'height'
- height = LocalToNum(value);
- if(~LocalIsPositiveScalar(height))
- error('Height must be a numeric scalar > 0');
- end
- case 'color'
- color = lower(value);
- if (~strcmp(color,{'bw','gray','cmyk'}))
- error('Color must be ''bw'', ''gray'' or ''cmyk''.');
- end
- case 'fontmode'
- fontmode = lower(value);
- if (~strcmp(fontmode,{'scaled','fixed'}))
- error('FontMode must be ''scaled'' or ''fixed''.');
- end
- case 'fontsize'
- fontsize = LocalToNum(value);
- if(~LocalIsPositiveScalar(fontsize))
- error('FontSize must be a numeric scalar > 0');
- end
- case 'fontencoding'
- fontencoding = lower(value);
- if (~strcmp(fontencoding,{'latin1','adobe'}))
- error('FontEncoding must be ''latin1'' or ''adobe''.');
- end
- case 'linemode'
- linemode = lower(value);
- if (~strcmp(linemode,{'scaled','fixed'}))
- error('LineMode must be ''scaled'' or ''fixed''.');
- end
- case 'linewidth'
- linewidth = LocalToNum(value);
- if(~LocalIsPositiveScalar(linewidth))
- error('LineWidth must be a numeric scalar > 0');
- end
- case 'renderer'
- renderer = lower(value);
- if (~strcmp(renderer,{'painters','zbuffer','opengl'}))
- error('Renderer must be ''painters'', ''zbuffer'' or ''opengl''.');
- end
- case 'resolution'
- resolution = LocalToNum(value);
- if ~(isnumeric(value) & (prod(size(value)) == 1) & (value >= 0));
- error('Resolution must be a numeric scalar >= 0');
- end
- otherwise
- error(['Unrecognized option ' param '.']);
- end
-end
-
-allLines = findall(H, 'type', 'line');
-allText = findall(H, 'type', 'text');
-allAxes = findall(H, 'type', 'axes');
-allImages = findall(H, 'type', 'image');
-allLights = findall(H, 'type', 'light');
-allPatch = findall(H, 'type', 'patch');
-allSurf = findall(H, 'type', 'surface');
-allRect = findall(H, 'type', 'rectangle');
-allFont = [allText; allAxes];
-allColor = [allLines; allText; allAxes; allLights];
-allMarker = [allLines; allPatch; allSurf];
-allEdge = [allPatch; allSurf];
-allCData = [allImages; allPatch; allSurf];
-
-old.objs = {};
-old.prop = {};
-old.values = {};
-
-% Process format and preview parameter
-showPreview = strcmp(format,'preview');
-if showPreview
- format = 'png';
- filename = [tempName '.png'];
-end
-if strncmp(format,'eps',3) & ~strcmp(preview,'none')
- args = {args{:}, ['-' preview]};
-end
-
-hadError = 0;
-try
- % Process size parameters
- paperPos = get(H, 'PaperPosition');
- old = LocalPushOldData(old, H, 'PaperPosition', paperPos);
- figureUnits = get(H, 'Units');
- set(H, 'Units', get(H,'PaperUnits'));
- figurePos = get(H, 'Position');
- aspectRatio = figurePos(3)/figurePos(4);
- set(H, 'Units', figureUnits);
- if (width == -1) & (height == -1)
- width = paperPos(3);
- height = paperPos(4);
- elseif (width == -1)
- width = height * aspectRatio;
- elseif (height == -1)
- height = width / aspectRatio;
- end
- set(H, 'PaperPosition', [0 0 width height]);
- paperPosMode = get(H, 'PaperPositionMode');
- old = LocalPushOldData(old, H, 'PaperPositionMode', paperPosMode);
- set(H, 'PaperPositionMode', 'manual');
-
- % Process rendering parameters
- switch (color)
- case {'bw', 'gray'}
- if ~strcmp(color,'bw') & strncmp(format,'eps',3)
- format = [format 'c'];
- end
- args = {args{:}, ['-d' format]};
-
- %compute and set gray colormap
- oldcmap = get(H,'Colormap');
- newgrays = 0.30*oldcmap(:,1) + 0.59*oldcmap(:,2) + 0.11*oldcmap(:,3);
- newcmap = [newgrays newgrays newgrays];
- old = LocalPushOldData(old, H, 'Colormap', oldcmap);
- set(H, 'Colormap', newcmap);
-
- %compute and set ColorSpec and CData properties
- old = LocalUpdateColors(allColor, 'color', old);
- old = LocalUpdateColors(allAxes, 'xcolor', old);
- old = LocalUpdateColors(allAxes, 'ycolor', old);
- old = LocalUpdateColors(allAxes, 'zcolor', old);
- old = LocalUpdateColors(allMarker, 'MarkerEdgeColor', old);
- old = LocalUpdateColors(allMarker, 'MarkerFaceColor', old);
- old = LocalUpdateColors(allEdge, 'EdgeColor', old);
- old = LocalUpdateColors(allEdge, 'FaceColor', old);
- old = LocalUpdateColors(allCData, 'CData', old);
-
- case 'cmyk'
- if strncmp(format,'eps',3)
- format = [format 'c'];
- args = {args{:}, ['-d' format], '-cmyk'};
- else
- args = {args{:}, ['-d' format]};
- end
- otherwise
- error('Invalid Color parameter');
- end
- if (~isempty(renderer))
- args = {args{:}, ['-' renderer]};
- end
- if (~isempty(resolution)) | ~strncmp(format,'eps',3)
- if isempty(resolution)
- resolution = 0;
- end
- args = {args{:}, ['-r' int2str(resolution)]};
- end
-
- % Process font parameters
- if (~isempty(fontmode))
- oldfonts = LocalGetAsCell(allFont,'FontSize');
- switch (fontmode)
- case 'fixed'
- oldfontunits = LocalGetAsCell(allFont,'FontUnits');
- old = LocalPushOldData(old, allFont, {'FontUnits'}, oldfontunits);
- set(allFont,'FontUnits','points');
- if (fontsize == -1)
- set(allFont,'FontSize',7);
- else
- set(allFont,'FontSize',fontsize);
- end
- case 'scaled'
- if (fontsize == -1)
- wscale = width/figurePos(3);
- hscale = height/figurePos(4);
- scale = min(wscale, hscale);
- else
- scale = fontsize;
- end
- newfonts = LocalScale(oldfonts,scale,5);
- set(allFont,{'FontSize'},newfonts);
- otherwise
- error('Invalid FontMode parameter');
- end
- % make sure we push the size after the units
- old = LocalPushOldData(old, allFont, {'FontSize'}, oldfonts);
- end
- if strcmp(fontencoding,'adobe') & strncmp(format,'eps',3)
- args = {args{:}, '-adobecset'};
- end
-
- % Process linewidth parameters
- if (~isempty(linemode))
- oldlines = LocalGetAsCell(allMarker,'LineWidth');
- old = LocalPushOldData(old, allMarker, {'LineWidth'}, oldlines);
- switch (linemode)
- case 'fixed'
- if (linewidth == -1)
- set(allMarker,'LineWidth',1);
- else
- set(allMarker,'LineWidth',linewidth);
- end
- case 'scaled'
- if (linewidth == -1)
- wscale = width/figurePos(3);
- hscale = height/figurePos(4);
- scale = min(wscale, hscale);
- else
- scale = linewidth;
- end
- newlines = LocalScale(oldlines, scale, 0.5);
- set(allMarker,{'LineWidth'},newlines);
- otherwise
- error('Invalid LineMode parameter');
- end
- end
-
- % Export
- print(H, filename, args{:});
-
-catch
- hadError = 1;
-end
-
-% Restore figure settings
-for n=1:length(old.objs)
- set(old.objs{n}, old.prop{n}, old.values{n});
-end
-
-if hadError
- error(deblank(lasterr));
-end
-
-% Show preview if requested
-if showPreview
- X = imread(filename,'png');
- delete(filename);
- f = figure( 'Name', 'Preview', ...
- 'Menubar', 'none', ...
- 'NumberTitle', 'off', ...
- 'Visible', 'off');
- image(X);
- axis image;
- ax = findobj(f, 'type', 'axes');
- set(ax, 'Units', get(H,'PaperUnits'), ...
- 'Position', [0 0 width height], ...
- 'Visible', 'off');
- set(ax, 'Units', 'pixels');
- axesPos = get(ax,'Position');
- figPos = get(f,'Position');
- rootSize = get(0,'ScreenSize');
- figPos(3:4) = axesPos(3:4);
- if figPos(1) + figPos(3) > rootSize(3)
- figPos(1) = rootSize(3) - figPos(3) - 50;
- end
- if figPos(2) + figPos(4) > rootSize(4)
- figPos(2) = rootSize(4) - figPos(4) - 50;
- end
- set(f, 'Position',figPos, ...
- 'Visible', 'on');
-end
-
-%
-% Local Functions
-%
-
-function outData = LocalPushOldData(inData, objs, prop, values)
-outData.objs = {inData.objs{:}, objs};
-outData.prop = {inData.prop{:}, prop};
-outData.values = {inData.values{:}, values};
-
-function cellArray = LocalGetAsCell(fig,prop);
-cellArray = get(fig,prop);
-if (~isempty(cellArray)) & (~iscell(cellArray))
- cellArray = {cellArray};
-end
-
-function newArray = LocalScale(inArray, scale, minValue)
-n = length(inArray);
-newArray = cell(n,1);
-for k=1:n
- newArray{k} = max(minValue,scale*inArray{k}(1));
-end
-
-function newArray = LocalMapToGray(inArray);
-n = length(inArray);
-newArray = cell(n,1);
-for k=1:n
- color = inArray{k};
- if (~isempty(color))
- if ischar(color)
- switch color(1)
- case 'y'
- color = [1 1 0];
- case 'm'
- color = [1 0 1];
- case 'c'
- color = [0 1 1];
- case 'r'
- color = [1 0 0];
- case 'g'
- color = [0 1 0];
- case 'b'
- color = [0 0 1];
- case 'w'
- color = [1 1 1];
- case 'k'
- color = [0 0 0];
- otherwise
- newArray{k} = color;
- end
- end
- if ~ischar(color)
- color = 0.30*color(1) + 0.59*color(2) + 0.11*color(3);
- end
- end
- if isempty(color) | ischar(color)
- newArray{k} = color;
- else
- newArray{k} = [color color color];
- end
-end
-
-function newArray = LocalMapCData(inArray);
-n = length(inArray);
-newArray = cell(n,1);
-for k=1:n
- color = inArray{k};
- if (ndims(color) == 3) & isa(color,'double')
- gray = 0.30*color(:,:,1) + 0.59*color(:,:,2) + 0.11*color(:,:,3);
- color(:,:,1) = gray;
- color(:,:,2) = gray;
- color(:,:,3) = gray;
- end
- newArray{k} = color;
-end
-
-function outData = LocalUpdateColors(inArray, prop, inData)
-value = LocalGetAsCell(inArray,prop);
-outData.objs = {inData.objs{:}, inArray};
-outData.prop = {inData.prop{:}, {prop}};
-outData.values = {inData.values{:}, value};
-if (~isempty(value))
- if strcmp(prop,'CData')
- value = LocalMapCData(value);
- else
- value = LocalMapToGray(value);
- end
- set(inArray,{prop},value);
-end
-
-function bool = LocalIsPositiveScalar(value)
-bool = isnumeric(value) & ...
- prod(size(value)) == 1 & ...
- value > 0;
-
-function value = LocalToNum(value)
-if ischar(value)
- value = str2num(value);
-end
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/normal_async_test.cc b/trunk/src/modules/video_coding/codecs/test_framework/normal_async_test.cc
deleted file mode 100644
index e379758..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/normal_async_test.cc
+++ /dev/null
@@ -1,593 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "normal_async_test.h"
-
-#include <assert.h>
-#include <string.h>
-#include <sstream>
-#include <queue>
-
-#include "gtest/gtest.h"
-#include "tick_util.h"
-#include "testsupport/fileutils.h"
-#include "typedefs.h"
-
-using namespace webrtc;
-
-NormalAsyncTest::NormalAsyncTest()
-:
-NormalTest("Async Normal Test 1", "A test of normal execution of the codec",
- _testNo),
-_decodeCompleteTime(0),
-_encodeCompleteTime(0),
-_encFrameCnt(0),
-_decFrameCnt(0),
-_requestKeyFrame(false),
-_testNo(1),
-_appendNext(false),
-_missingFrames(false),
-_rttFrames(0),
-_hasReceivedSLI(false),
-_hasReceivedRPSI(false),
-_hasReceivedPLI(false),
-_waitForKey(false)
-{
-}
-
-NormalAsyncTest::NormalAsyncTest(WebRtc_UWord32 bitRate)
-:
-NormalTest("Async Normal Test 1", "A test of normal execution of the codec",
- bitRate, _testNo),
-_decodeCompleteTime(0),
-_encodeCompleteTime(0),
-_encFrameCnt(0),
-_decFrameCnt(0),
-_requestKeyFrame(false),
-_testNo(1),
-_appendNext(false),
-_missingFrames(false),
-_rttFrames(0),
-_hasReceivedSLI(false),
-_hasReceivedRPSI(false),
-_hasReceivedPLI(false),
-_waitForKey(false)
-{
-}
-
-NormalAsyncTest::NormalAsyncTest(std::string name, std::string description,
- unsigned int testNo)
-:
-NormalTest(name, description, _testNo),
-_decodeCompleteTime(0),
-_encodeCompleteTime(0),
-_encFrameCnt(0),
-_decFrameCnt(0),
-_requestKeyFrame(false),
-_testNo(testNo),
-_lengthEncFrame(0),
-_appendNext(false),
-_missingFrames(false),
-_rttFrames(0),
-_hasReceivedSLI(false),
-_hasReceivedRPSI(false),
-_hasReceivedPLI(false),
-_waitForKey(false)
-{
-}
-
-NormalAsyncTest::NormalAsyncTest(std::string name, std::string description,
- WebRtc_UWord32 bitRate, unsigned int testNo)
-:
-NormalTest(name, description, bitRate, _testNo),
-_decodeCompleteTime(0),
-_encodeCompleteTime(0),
-_encFrameCnt(0),
-_decFrameCnt(0),
-_requestKeyFrame(false),
-_testNo(testNo),
-_lengthEncFrame(0),
-_appendNext(false),
-_missingFrames(false),
-_rttFrames(0),
-_hasReceivedSLI(false),
-_hasReceivedRPSI(false),
-_hasReceivedPLI(false),
-_waitForKey(false)
-{
-}
-
-NormalAsyncTest::NormalAsyncTest(std::string name, std::string description,
- WebRtc_UWord32 bitRate, unsigned int testNo,
- unsigned int rttFrames)
-:
-NormalTest(name, description, bitRate, _testNo),
-_decodeCompleteTime(0),
-_encodeCompleteTime(0),
-_encFrameCnt(0),
-_decFrameCnt(0),
-_requestKeyFrame(false),
-_testNo(testNo),
-_lengthEncFrame(0),
-_appendNext(false),
-_missingFrames(false),
-_rttFrames(rttFrames),
-_hasReceivedSLI(false),
-_hasReceivedRPSI(false),
-_hasReceivedPLI(false),
-_waitForKey(false)
-{
-}
-
-void
-NormalAsyncTest::Setup()
-{
- Test::Setup();
- std::stringstream ss;
- std::string strTestNo;
- ss << _testNo;
- ss >> strTestNo;
-
- // Check if settings exist. Otherwise use defaults.
- if (_outname == "")
- {
- _outname = webrtc::test::OutputPath() + "out_normaltest" + strTestNo +
- ".yuv";
- }
-
- if (_encodedName == "")
- {
- _encodedName = webrtc::test::OutputPath() + "encoded_normaltest" +
- strTestNo + ".yuv";
- }
-
- if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
- {
- printf("Cannot read file %s.\n", _inname.c_str());
- exit(1);
- }
-
- if ((_encodedFile = fopen(_encodedName.c_str(), "wb")) == NULL)
- {
- printf("Cannot write encoded file.\n");
- exit(1);
- }
-
- char mode[3] = "wb";
- if (_appendNext)
- {
- strncpy(mode, "ab", 3);
- }
-
- if ((_decodedFile = fopen(_outname.c_str(), mode)) == NULL)
- {
- printf("Cannot write file %s.\n", _outname.c_str());
- exit(1);
- }
-
- _appendNext = true;
-}
-
-void
-NormalAsyncTest::Teardown()
-{
- Test::Teardown();
- fclose(_sourceFile);
- fclose(_encodedFile);
- fclose(_decodedFile);
-}
-
-FrameQueueTuple::~FrameQueueTuple()
-{
- if (_codecSpecificInfo != NULL)
- {
- delete _codecSpecificInfo;
- }
- if (_frame != NULL)
- {
- delete _frame;
- }
-}
-
-void FrameQueue::PushFrame(TestVideoEncodedBuffer *frame,
- webrtc::CodecSpecificInfo* codecSpecificInfo)
-{
- WriteLockScoped cs(_queueRWLock);
- _frameBufferQueue.push(new FrameQueueTuple(frame, codecSpecificInfo));
-}
-
-FrameQueueTuple* FrameQueue::PopFrame()
-{
- WriteLockScoped cs(_queueRWLock);
- if (_frameBufferQueue.empty())
- {
- return NULL;
- }
- FrameQueueTuple* tuple = _frameBufferQueue.front();
- _frameBufferQueue.pop();
- return tuple;
-}
-
-bool FrameQueue::Empty()
-{
- ReadLockScoped cs(_queueRWLock);
- return _frameBufferQueue.empty();
-}
-
-WebRtc_UWord32 VideoEncodeCompleteCallback::EncodedBytes()
-{
- return _encodedBytes;
-}
-
-WebRtc_Word32
-VideoEncodeCompleteCallback::Encoded(EncodedImage& encodedImage,
- const webrtc::CodecSpecificInfo* codecSpecificInfo,
- const webrtc::RTPFragmentationHeader*
- fragmentation)
-{
- _test.Encoded(encodedImage);
- TestVideoEncodedBuffer *newBuffer = new TestVideoEncodedBuffer();
- //newBuffer->VerifyAndAllocate(encodedImage._length);
- newBuffer->VerifyAndAllocate(encodedImage._size);
- _encodedBytes += encodedImage._length;
- // If _frameQueue would have been a fixed sized buffer we could have asked
- // it for an empty frame and then just do:
- // emptyFrame->SwapBuffers(encodedBuffer);
- // This is how it should be done in Video Engine to save in on memcpys
- webrtc::CodecSpecificInfo* codecSpecificInfoCopy =
- _test.CopyCodecSpecificInfo(codecSpecificInfo);
- _test.CopyEncodedImage(*newBuffer, encodedImage, codecSpecificInfoCopy);
- if (_encodedFile != NULL)
- {
- fwrite(newBuffer->GetBuffer(), 1, newBuffer->GetLength(), _encodedFile);
- }
- _frameQueue->PushFrame(newBuffer, codecSpecificInfoCopy);
- return 0;
-}
-
-WebRtc_UWord32 VideoDecodeCompleteCallback::DecodedBytes()
-{
- return _decodedBytes;
-}
-
-WebRtc_Word32
-VideoDecodeCompleteCallback::Decoded(RawImage& image)
-{
- _test.Decoded(image);
- _decodedBytes += image._length;
- if (_decodedFile != NULL)
- {
- fwrite(image._buffer, 1, image._length, _decodedFile);
- }
- return 0;
-}
-
-WebRtc_Word32
-VideoDecodeCompleteCallback::ReceivedDecodedReferenceFrame(
- const WebRtc_UWord64 pictureId)
-{
- return _test.ReceivedDecodedReferenceFrame(pictureId);
-}
-
-WebRtc_Word32
-VideoDecodeCompleteCallback::ReceivedDecodedFrame(
- const WebRtc_UWord64 pictureId)
-{
- return _test.ReceivedDecodedFrame(pictureId);
-}
-
-void
-NormalAsyncTest::Encoded(const EncodedImage& encodedImage)
-{
- _encodeCompleteTime = tGetTime();
- _encFrameCnt++;
- _totalEncodePipeTime += _encodeCompleteTime -
- _encodeTimes[encodedImage._timeStamp];
-}
-
-void
-NormalAsyncTest::Decoded(const RawImage& decodedImage)
-{
- _decodeCompleteTime = tGetTime();
- _decFrameCnt++;
- _totalDecodePipeTime += _decodeCompleteTime -
- _decodeTimes[decodedImage._timeStamp];
- _decodedWidth = decodedImage._width;
- _decodedHeight = decodedImage._height;
-}
-
-void
-NormalAsyncTest::Perform()
-{
- _inname = webrtc::test::ProjectRootPath() + "resources/foreman_cif.yuv";
- CodecSettings(352, 288, 30, _bitRate);
- Setup();
- _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
- _decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
- if(_encoder->InitEncode(&_inst, 1, 1440) < 0)
- {
- exit(EXIT_FAILURE);
- }
- _decoder->InitDecode(&_inst, 1);
- FrameQueue frameQueue;
- VideoEncodeCompleteCallback encCallback(_encodedFile, &frameQueue, *this);
- VideoDecodeCompleteCallback decCallback(_decodedFile, *this);
- _encoder->RegisterEncodeCompleteCallback(&encCallback);
- _decoder->RegisterDecodeCompleteCallback(&decCallback);
- if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
- {
- exit(EXIT_FAILURE);
- }
- _totalEncodeTime = _totalDecodeTime = 0;
- _totalEncodePipeTime = _totalDecodePipeTime = 0;
- bool complete = false;
- _framecnt = 0;
- _encFrameCnt = 0;
- _decFrameCnt = 0;
- _sumEncBytes = 0;
- _lengthEncFrame = 0;
- double starttime = tGetTime();
- while (!complete)
- {
- CodecSpecific_InitBitrate();
- complete = Encode();
- if (!frameQueue.Empty() || complete)
- {
- while (!frameQueue.Empty())
- {
- _frameToDecode =
- static_cast<FrameQueueTuple *>(frameQueue.PopFrame());
- int lost = DoPacketLoss();
- if (lost == 2)
- {
- // Lost the whole frame, continue
- _missingFrames = true;
- delete _frameToDecode;
- _frameToDecode = NULL;
- continue;
- }
- int ret = Decode(lost);
- delete _frameToDecode;
- _frameToDecode = NULL;
- if (ret < 0)
- {
- fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
- exit(EXIT_FAILURE);
- }
- else if (ret == 0)
- {
- _framecnt++;
- }
- else
- {
- fprintf(stderr,
- "\n\nPositive return value from decode!\n\n");
- }
- }
- }
- }
- double endtime = tGetTime();
- double totalExecutionTime = endtime - starttime;
- printf("Total execution time: %.1f s\n", totalExecutionTime);
- _sumEncBytes = encCallback.EncodedBytes();
- double actualBitRate = ActualBitRate(_encFrameCnt) / 1000.0;
- double avgEncTime = _totalEncodeTime / _encFrameCnt;
- double avgDecTime = _totalDecodeTime / _decFrameCnt;
- printf("Actual bitrate: %f kbps\n", actualBitRate);
- printf("Average encode time: %.1f ms\n", 1000 * avgEncTime);
- printf("Average decode time: %.1f ms\n", 1000 * avgDecTime);
- printf("Average encode pipeline time: %.1f ms\n",
- 1000 * _totalEncodePipeTime / _encFrameCnt);
- printf("Average decode pipeline time: %.1f ms\n",
- 1000 * _totalDecodePipeTime / _decFrameCnt);
- printf("Number of encoded frames: %u\n", _encFrameCnt);
- printf("Number of decoded frames: %u\n", _decFrameCnt);
- (*_log) << "Actual bitrate: " << actualBitRate << " kbps\tTarget: " <<
- _bitRate << " kbps" << std::endl;
- (*_log) << "Average encode time: " << avgEncTime << " s" << std::endl;
- (*_log) << "Average decode time: " << avgDecTime << " s" << std::endl;
- _encoder->Release();
- _decoder->Release();
- Teardown();
-}
-
-bool
-NormalAsyncTest::Encode()
-{
- _lengthEncFrame = 0;
- EXPECT_GT(fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile), 0u);
- _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
- _inputVideoBuffer.SetTimeStamp((unsigned int)
- (_encFrameCnt * 9e4 / _inst.maxFramerate));
- _inputVideoBuffer.SetWidth(_inst.width);
- _inputVideoBuffer.SetHeight(_inst.height);
- RawImage rawImage;
- VideoBufferToRawImage(_inputVideoBuffer, rawImage);
- if (feof(_sourceFile) != 0)
- {
- return true;
- }
- _encodeCompleteTime = 0;
- _encodeTimes[rawImage._timeStamp] = tGetTime();
- VideoFrameType frameType = kDeltaFrame;
-
- // check SLI queue
- _hasReceivedSLI = false;
- while (!_signalSLI.empty() && _signalSLI.front().delay == 0)
- {
- // SLI message has arrived at sender side
- _hasReceivedSLI = true;
- _pictureIdSLI = _signalSLI.front().id;
- _signalSLI.pop_front();
- }
- // decrement SLI queue times
- for (std::list<fbSignal>::iterator it = _signalSLI.begin();
- it !=_signalSLI.end(); it++)
- {
- (*it).delay--;
- }
-
- // check PLI queue
- _hasReceivedPLI = false;
- while (!_signalPLI.empty() && _signalPLI.front().delay == 0)
- {
- // PLI message has arrived at sender side
- _hasReceivedPLI = true;
- _signalPLI.pop_front();
- }
- // decrement PLI queue times
- for (std::list<fbSignal>::iterator it = _signalPLI.begin();
- it != _signalPLI.end(); it++)
- {
- (*it).delay--;
- }
-
- if (_hasReceivedPLI)
- {
- // respond to PLI by encoding a key frame
- frameType = kKeyFrame;
- _hasReceivedPLI = false;
- _hasReceivedSLI = false; // don't trigger both at once
- }
-
- webrtc::CodecSpecificInfo* codecSpecificInfo = CreateEncoderSpecificInfo();
- int ret = _encoder->Encode(rawImage, codecSpecificInfo, &frameType);
- EXPECT_EQ(ret, WEBRTC_VIDEO_CODEC_OK);
- if (codecSpecificInfo != NULL)
- {
- delete codecSpecificInfo;
- codecSpecificInfo = NULL;
- }
- if (_encodeCompleteTime > 0)
- {
- _totalEncodeTime += _encodeCompleteTime -
- _encodeTimes[rawImage._timeStamp];
- }
- else
- {
- _totalEncodeTime += tGetTime() - _encodeTimes[rawImage._timeStamp];
- }
- assert(ret >= 0);
- return false;
-}
-
-int
-NormalAsyncTest::Decode(int lossValue)
-{
- _sumEncBytes += _frameToDecode->_frame->GetLength();
- EncodedImage encodedImage;
- VideoEncodedBufferToEncodedImage(*(_frameToDecode->_frame), encodedImage);
- encodedImage._completeFrame = !lossValue;
- _decodeCompleteTime = 0;
- _decodeTimes[encodedImage._timeStamp] = tGetTime();
- int ret = WEBRTC_VIDEO_CODEC_OK;
- if (!_waitForKey || encodedImage._frameType == kKeyFrame)
- {
- _waitForKey = false;
- ret = _decoder->Decode(encodedImage, _missingFrames, NULL,
- _frameToDecode->_codecSpecificInfo);
-
- if (ret >= 0)
- {
- _missingFrames = false;
- }
- }
-
- // check for SLI
- if (ret == WEBRTC_VIDEO_CODEC_REQUEST_SLI)
- {
- // add an SLI feedback to the feedback "queue"
- // to be delivered to encoder with _rttFrames delay
- _signalSLI.push_back(fbSignal(_rttFrames,
- static_cast<WebRtc_UWord8>((_lastDecPictureId) & 0x3f))); // 6 lsb
-
- ret = WEBRTC_VIDEO_CODEC_OK;
- }
- else if (ret == WEBRTC_VIDEO_CODEC_ERR_REQUEST_SLI)
- {
- // add an SLI feedback to the feedback "queue"
- // to be delivered to encoder with _rttFrames delay
- _signalSLI.push_back(fbSignal(_rttFrames,
- static_cast<WebRtc_UWord8>((_lastDecPictureId + 1) & 0x3f)));//6 lsb
-
- ret = WEBRTC_VIDEO_CODEC_OK;
- }
- else if (ret == WEBRTC_VIDEO_CODEC_ERROR)
- {
- // wait for new key frame
- // add an PLI feedback to the feedback "queue"
- // to be delivered to encoder with _rttFrames delay
- _signalPLI.push_back(fbSignal(_rttFrames, 0 /* picId not used*/));
- _waitForKey = true;
-
- ret = WEBRTC_VIDEO_CODEC_OK;
- }
-
- if (_decodeCompleteTime > 0)
- {
- _totalDecodeTime += _decodeCompleteTime -
- _decodeTimes[encodedImage._timeStamp];
- }
- else
- {
- _totalDecodeTime += tGetTime() - _decodeTimes[encodedImage._timeStamp];
- }
- return ret;
-}
-
-webrtc::CodecSpecificInfo*
-NormalAsyncTest::CopyCodecSpecificInfo(
- const webrtc::CodecSpecificInfo* codecSpecificInfo) const
-{
- webrtc::CodecSpecificInfo* info = new webrtc::CodecSpecificInfo;
- *info = *codecSpecificInfo;
- return info;
-}
-
-void NormalAsyncTest::CodecSpecific_InitBitrate()
-{
- if (_bitRate == 0)
- {
- _encoder->SetRates(600, _inst.maxFramerate);
- }
- else
- {
- _encoder->SetRates(_bitRate, _inst.maxFramerate);
- }
-}
-
-void NormalAsyncTest::CopyEncodedImage(TestVideoEncodedBuffer& dest,
- EncodedImage& src,
- void* /*codecSpecificInfo*/) const
-{
- dest.CopyBuffer(src._length, src._buffer);
- dest.SetFrameType(src._frameType);
- dest.SetCaptureWidth((WebRtc_UWord16)src._encodedWidth);
- dest.SetCaptureHeight((WebRtc_UWord16)src._encodedHeight);
- dest.SetTimeStamp(src._timeStamp);
-}
-
-WebRtc_Word32 NormalAsyncTest::ReceivedDecodedReferenceFrame(
- const WebRtc_UWord64 pictureId) {
- _lastDecRefPictureId = pictureId;
- return 0;
-}
-
-WebRtc_Word32 NormalAsyncTest::ReceivedDecodedFrame(
- const WebRtc_UWord64 pictureId) {
- _lastDecPictureId = pictureId;
- return 0;
-}
-
-double
-NormalAsyncTest::tGetTime()
-{// return time in sec
- return ((double) (TickTime::MillisecondTimestamp())/1000);
- }
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/normal_async_test.h b/trunk/src/modules/video_coding/codecs/test_framework/normal_async_test.h
deleted file mode 100644
index e520601..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/normal_async_test.h
+++ /dev/null
@@ -1,189 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_ASYNC_TEST_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_ASYNC_TEST_H_
-
-#include "common_types.h"
-
-#include "normal_test.h"
-#include "rw_lock_wrapper.h"
-#include <list>
-#include <map>
-#include <queue>
-
-class FrameQueueTuple
-{
-public:
- FrameQueueTuple(TestVideoEncodedBuffer *frame,
- const webrtc::CodecSpecificInfo* codecSpecificInfo = NULL)
- :
- _frame(frame),
- _codecSpecificInfo(codecSpecificInfo)
- {};
- ~FrameQueueTuple();
- TestVideoEncodedBuffer* _frame;
- const webrtc::CodecSpecificInfo* _codecSpecificInfo;
-};
-
-class FrameQueue
-{
-public:
- FrameQueue()
- :
- _queueRWLock(*webrtc::RWLockWrapper::CreateRWLock()),
- _prevTS(-1)
- {
- }
-
- ~FrameQueue()
- {
- delete &_queueRWLock;
- }
-
- void PushFrame(TestVideoEncodedBuffer *frame,
- webrtc::CodecSpecificInfo* codecSpecificInfo = NULL);
- FrameQueueTuple* PopFrame();
- bool Empty();
-
-private:
- webrtc::RWLockWrapper& _queueRWLock;
- std::queue<FrameQueueTuple *> _frameBufferQueue;
- WebRtc_Word64 _prevTS;
-};
-
-// feedback signal to encoder
-struct fbSignal
-{
- fbSignal(int d, WebRtc_UWord8 pid) : delay(d), id(pid) {};
- int delay;
- WebRtc_UWord8 id;
-};
-
-class NormalAsyncTest : public NormalTest
-{
-public:
- NormalAsyncTest();
- NormalAsyncTest(WebRtc_UWord32 bitRate);
- NormalAsyncTest(std::string name, std::string description,
- unsigned int testNo);
- NormalAsyncTest(std::string name, std::string description,
- WebRtc_UWord32 bitRate, unsigned int testNo);
- NormalAsyncTest(std::string name, std::string description,
- WebRtc_UWord32 bitRate, unsigned int testNo,
- unsigned int rttFrames);
- virtual ~NormalAsyncTest() {};
- virtual void Perform();
- virtual void Encoded(const webrtc::EncodedImage& encodedImage);
- virtual void Decoded(const webrtc::RawImage& decodedImage);
- virtual webrtc::CodecSpecificInfo*
- CopyCodecSpecificInfo(
- const webrtc::CodecSpecificInfo* codecSpecificInfo) const;
- virtual void CopyEncodedImage(TestVideoEncodedBuffer& dest,
- webrtc::EncodedImage& src,
- void* /*codecSpecificInfo*/) const;
- virtual webrtc::CodecSpecificInfo* CreateEncoderSpecificInfo() const
- {
- return NULL;
- };
- virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(
- const WebRtc_UWord64 pictureId);
- virtual WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 pictureId);
-
-protected:
- virtual void Setup();
- virtual void Teardown();
- virtual bool Encode();
- virtual int Decode(int lossValue = 0);
- virtual void CodecSpecific_InitBitrate();
- virtual int SetCodecSpecificParameters() {return 0;};
- double tGetTime();// return time in sec
-
- FILE* _sourceFile;
- FILE* _decodedFile;
- WebRtc_UWord32 _decodedWidth;
- WebRtc_UWord32 _decodedHeight;
- double _totalEncodeTime;
- double _totalDecodeTime;
- double _decodeCompleteTime;
- double _encodeCompleteTime;
- double _totalEncodePipeTime;
- double _totalDecodePipeTime;
- int _framecnt;
- int _encFrameCnt;
- int _decFrameCnt;
- bool _requestKeyFrame;
- unsigned int _testNo;
- unsigned int _lengthEncFrame;
- FrameQueueTuple* _frameToDecode;
- bool _appendNext;
- std::map<WebRtc_UWord32, double> _encodeTimes;
- std::map<WebRtc_UWord32, double> _decodeTimes;
- bool _missingFrames;
- std::list<fbSignal> _signalSLI;
- int _rttFrames;
- mutable bool _hasReceivedSLI;
- mutable bool _hasReceivedRPSI;
- WebRtc_UWord8 _pictureIdSLI;
- WebRtc_UWord16 _pictureIdRPSI;
- WebRtc_UWord64 _lastDecRefPictureId;
- WebRtc_UWord64 _lastDecPictureId;
- std::list<fbSignal> _signalPLI;
- bool _hasReceivedPLI;
- bool _waitForKey;
-};
-
-class VideoEncodeCompleteCallback : public webrtc::EncodedImageCallback
-{
-public:
- VideoEncodeCompleteCallback(FILE* encodedFile, FrameQueue *frameQueue,
- NormalAsyncTest& test)
- :
- _encodedFile(encodedFile),
- _frameQueue(frameQueue),
- _test(test),
- _encodedBytes(0)
- {}
-
- WebRtc_Word32
- Encoded(webrtc::EncodedImage& encodedImage,
- const webrtc::CodecSpecificInfo* codecSpecificInfo = NULL,
- const webrtc::RTPFragmentationHeader* fragmentation = NULL);
- WebRtc_UWord32 EncodedBytes();
-private:
- FILE* _encodedFile;
- FrameQueue* _frameQueue;
- NormalAsyncTest& _test;
- WebRtc_UWord32 _encodedBytes;
-};
-
-class VideoDecodeCompleteCallback : public webrtc::DecodedImageCallback
-{
-public:
- VideoDecodeCompleteCallback(FILE* decodedFile, NormalAsyncTest& test)
- :
- _decodedFile(decodedFile),
- _test(test),
- _decodedBytes(0)
- {}
-
- virtual WebRtc_Word32 Decoded(webrtc::RawImage& decodedImage);
- virtual WebRtc_Word32
- ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId);
- virtual WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 pictureId);
-
- WebRtc_UWord32 DecodedBytes();
-private:
- FILE* _decodedFile;
- NormalAsyncTest& _test;
- WebRtc_UWord32 _decodedBytes;
-};
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_ASYNC_TEST_H_
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/normal_test.cc b/trunk/src/modules/video_coding/codecs/test_framework/normal_test.cc
deleted file mode 100644
index 2dd7bce..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/normal_test.cc
+++ /dev/null
@@ -1,253 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "normal_test.h"
-
-#include <time.h>
-#include <sstream>
-#include <string.h>
-
-#include "gtest/gtest.h"
-#include "testsupport/fileutils.h"
-
-NormalTest::NormalTest()
-:
-Test("Normal Test 1", "A test of normal execution of the codec"),
-_testNo(1),
-_lengthEncFrame(0),
-_appendNext(false)
-{
-}
-
-NormalTest::NormalTest(std::string name, std::string description,
- unsigned int testNo)
-:
-Test(name, description),
-_requestKeyFrame(false),
-_testNo(testNo),
-_lengthEncFrame(0),
-_appendNext(false)
-{
-}
-
-NormalTest::NormalTest(std::string name, std::string description,
- WebRtc_UWord32 bitRate, unsigned int testNo)
-:
-Test(name, description, bitRate),
-_requestKeyFrame(false),
-_testNo(testNo),
-_lengthEncFrame(0),
-_appendNext(false)
-{
-}
-
-void
-NormalTest::Setup()
-{
- Test::Setup();
- std::stringstream ss;
- std::string strTestNo;
- ss << _testNo;
- ss >> strTestNo;
-
- // Check if settings exist. Otherwise use defaults.
- if (_outname == "")
- {
- _outname = webrtc::test::OutputPath() + "out_normaltest" + strTestNo +
- ".yuv";
- }
-
- if (_encodedName == "")
- {
- _encodedName = webrtc::test::OutputPath() + "encoded_normaltest" +
- strTestNo + ".yuv";
- }
-
- if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
- {
- printf("Cannot read file %s.\n", _inname.c_str());
- exit(1);
- }
-
- if ((_encodedFile = fopen(_encodedName.c_str(), "wb")) == NULL)
- {
- printf("Cannot write encoded file.\n");
- exit(1);
- }
-
- char mode[3] = "wb";
- if (_appendNext)
- {
- strncpy(mode, "ab", 3);
- }
-
- if ((_decodedFile = fopen(_outname.c_str(), mode)) == NULL)
- {
- printf("Cannot write file %s.\n", _outname.c_str());
- exit(1);
- }
-
- _appendNext = true;
-}
-
-void
-NormalTest::Teardown()
-{
- Test::Teardown();
- fclose(_sourceFile);
- fclose(_decodedFile);
-}
-
-void
-NormalTest::Perform()
-{
- _inname = webrtc::test::ProjectRootPath() + "resources/foreman_cif.yuv";
- CodecSettings(352, 288, 30, _bitRate);
- Setup();
-
- _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
- _decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
- _encodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
-
- _encoder->InitEncode(&_inst, 1, 1460);
- CodecSpecific_InitBitrate();
- _decoder->InitDecode(&_inst,1);
-
- _totalEncodeTime = _totalDecodeTime = 0;
- _framecnt = 0;
- _sumEncBytes = 0;
- _lengthEncFrame = 0;
- int decodeLength = 0;
- while (!Encode())
- {
- DoPacketLoss();
- _encodedVideoBuffer.UpdateLength(_encodedVideoBuffer.GetLength());
- fwrite(_encodedVideoBuffer.GetBuffer(), 1, _encodedVideoBuffer.GetLength(), _encodedFile);
- decodeLength = Decode();
- if (decodeLength < 0)
- {
- fprintf(stderr,"\n\nError in decoder: %d\n\n", decodeLength);
- exit(EXIT_FAILURE);
- }
- fwrite(_decodedVideoBuffer.GetBuffer(), 1, decodeLength, _decodedFile);
- CodecSpecific_InitBitrate();
- _framecnt++;
- }
-
- // Ensure we empty the decoding queue.
- while (decodeLength > 0)
- {
- decodeLength = Decode();
- if (decodeLength < 0)
- {
- fprintf(stderr,"\n\nError in decoder: %d\n\n", decodeLength);
- exit(EXIT_FAILURE);
- }
- fwrite(_decodedVideoBuffer.GetBuffer(), 1, decodeLength, _decodedFile);
- }
-
- double actualBitRate = ActualBitRate(_framecnt) / 1000.0;
- double avgEncTime = _totalEncodeTime / _framecnt;
- double avgDecTime = _totalDecodeTime / _framecnt;
- printf("Actual bitrate: %f kbps\n", actualBitRate);
- printf("Average encode time: %f s\n", avgEncTime);
- printf("Average decode time: %f s\n", avgDecTime);
- (*_log) << "Actual bitrate: " << actualBitRate << " kbps\tTarget: " << _bitRate << " kbps" << std::endl;
- (*_log) << "Average encode time: " << avgEncTime << " s" << std::endl;
- (*_log) << "Average decode time: " << avgDecTime << " s" << std::endl;
-
- _inputVideoBuffer.Free();
- _encodedVideoBuffer.Reset();
- _decodedVideoBuffer.Free();
-
- _encoder->Release();
- _decoder->Release();
-
- Teardown();
-}
-
-bool
-NormalTest::Encode()
-{
- _lengthEncFrame = 0;
- EXPECT_GT(fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile), 0u);
- if (feof(_sourceFile) != 0)
- {
- return true;
- }
- _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
- _inputVideoBuffer.SetTimeStamp(_framecnt);
-
- // This multiple attempt ridiculousness is to accomodate VP7:
- // 1. The wrapper can unilaterally reduce the framerate for low bitrates.
- // 2. The codec inexplicably likes to reject some frames. Perhaps there
- // is a good reason for this...
- int encodingAttempts = 0;
- double starttime = 0;
- double endtime = 0;
- while (_lengthEncFrame == 0)
- {
- starttime = clock()/(double)CLOCKS_PER_SEC;
-
- _inputVideoBuffer.SetWidth(_inst.width);
- _inputVideoBuffer.SetHeight(_inst.height);
- //_lengthEncFrame = _encoder->Encode(_inputVideoBuffer, _encodedVideoBuffer, _frameInfo,
- // _inst.frameRate, _requestKeyFrame && !(_framecnt%50));
-
- endtime = clock()/(double)CLOCKS_PER_SEC;
-
- _encodedVideoBuffer.SetCaptureHeight(_inst.height);
- _encodedVideoBuffer.SetCaptureWidth(_inst.width);
- if (_lengthEncFrame < 0)
- {
- (*_log) << "Error in encoder: " << _lengthEncFrame << std::endl;
- fprintf(stderr,"\n\nError in encoder: %d\n\n", _lengthEncFrame);
- exit(EXIT_FAILURE);
- }
- _sumEncBytes += _lengthEncFrame;
-
- encodingAttempts++;
- if (encodingAttempts > 50)
- {
- (*_log) << "Unable to encode frame: " << _framecnt << std::endl;
- fprintf(stderr,"\n\nUnable to encode frame: %d\n\n", _framecnt);
- exit(EXIT_FAILURE);
- }
- }
- _totalEncodeTime += endtime - starttime;
-
- if (encodingAttempts > 1)
- {
- (*_log) << encodingAttempts << " attempts required to encode frame: " <<
- _framecnt + 1 << std::endl;
- fprintf(stderr,"\n%d attempts required to encode frame: %d\n", encodingAttempts,
- _framecnt + 1);
- }
-
- return false;
-}
-
-int
-NormalTest::Decode(int lossValue)
-{
- _encodedVideoBuffer.SetWidth(_inst.width);
- _encodedVideoBuffer.SetHeight(_inst.height);
- int lengthDecFrame = 0;
- //int lengthDecFrame = _decoder->Decode(_encodedVideoBuffer, _decodedVideoBuffer);
- //_totalDecodeTime += (double)((clock()/(double)CLOCKS_PER_SEC) - starttime);
- if (lengthDecFrame < 0)
- {
- return lengthDecFrame;
- }
- _encodedVideoBuffer.Reset();
- _encodedVideoBuffer.UpdateLength(0);
- return lengthDecFrame;
-}
-
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/normal_test.h b/trunk/src/modules/video_coding/codecs/test_framework/normal_test.h
deleted file mode 100644
index 0dcd9c2..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/normal_test.h
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_TEST_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_TEST_H_
-
-#include "test.h"
-
-class NormalTest : public Test
-{
-public:
- NormalTest();
- NormalTest(std::string name, std::string description, unsigned int testNo);
- NormalTest(std::string name, std::string description, WebRtc_UWord32 bitRate, unsigned int testNo);
- virtual ~NormalTest() {};
- virtual void Perform();
-
-protected:
- virtual void Setup();
- virtual void Teardown();
- virtual bool Encode();
- virtual int Decode(int lossValue = 0);
- virtual void CodecSpecific_InitBitrate()=0;
- virtual int DoPacketLoss() {return 0;};
-
- FILE* _sourceFile;
- FILE* _decodedFile;
- FILE* _encodedFile;
- double _totalEncodeTime;
- double _totalDecodeTime;
- unsigned int _framecnt;
- bool _requestKeyFrame;
- unsigned int _testNo;
- int _lengthEncFrame;
- bool _appendNext;
-};
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_NORMAL_TEST_H_
-
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/packet_loss_test.cc b/trunk/src/modules/video_coding/codecs/test_framework/packet_loss_test.cc
deleted file mode 100644
index c94fc9a..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/packet_loss_test.cc
+++ /dev/null
@@ -1,248 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "packet_loss_test.h"
-#include "video_source.h"
-#include <sstream>
-#include <cassert>
-#include <string.h>
-
-using namespace webrtc;
-
-PacketLossTest::PacketLossTest()
-:
-NormalAsyncTest("PacketLossTest", "Encode, remove lost packets, decode", 300,
- 5),
-_lossRate(0.1),
-_lossProbability(0.1),
-_lastFrame(NULL),
-_lastFrameLength(0)
-{
-}
-
-PacketLossTest::PacketLossTest(std::string name, std::string description)
-:
-NormalAsyncTest(name, description, 300, 5),
-_lossRate(0.1),
-_lossProbability(0.1),
-_lastFrame(NULL),
-_lastFrameLength(0)
-{
-}
-
-PacketLossTest::PacketLossTest(std::string name, std::string description, double lossRate, bool useNack, unsigned int rttFrames /* = 0*/)
-:
-NormalAsyncTest(name, description, 300, 5, rttFrames),
-_lossRate(lossRate),
-_lastFrame(NULL),
-_lastFrameLength(0)
-{
- assert(lossRate >= 0 && lossRate <= 1);
- if (useNack)
- {
- _lossProbability = 0;
- }
- else
- {
- _lossProbability = lossRate;
- }
-}
-
-void
-PacketLossTest::Encoded(const EncodedImage& encodedImage)
-{
- // push timestamp to queue
- _frameQueue.push_back(encodedImage._timeStamp);
- NormalAsyncTest::Encoded(encodedImage);
-}
-
-void
-PacketLossTest::Decoded(const RawImage& decodedImage)
-{
- // check the frame queue if any frames have gone missing
- assert(!_frameQueue.empty()); // decoded frame is not in the queue
- while(_frameQueue.front() < decodedImage._timeStamp)
- {
- // this frame is missing
- // write previous decoded frame again (frame freeze)
- if (_decodedFile && _lastFrame)
- {
- fwrite(_lastFrame, 1, _lastFrameLength, _decodedFile);
- }
-
- // remove frame from queue
- _frameQueue.pop_front();
- }
- assert(_frameQueue.front() == decodedImage._timeStamp); // decoded frame is not in the queue
-
- // pop the current frame
- _frameQueue.pop_front();
-
- // save image for future freeze-frame
- if (_lastFrameLength < decodedImage._length)
- {
- if (_lastFrame) delete [] _lastFrame;
-
- _lastFrame = new WebRtc_UWord8[decodedImage._length];
- }
- memcpy(_lastFrame, decodedImage._buffer, decodedImage._length);
- _lastFrameLength = decodedImage._length;
-
- NormalAsyncTest::Decoded(decodedImage);
-}
-
-void
-PacketLossTest::Teardown()
-{
- if (_totalKept + _totalThrown > 0)
- {
- printf("Target packet loss rate: %.4f\n", _lossProbability);
- printf("Actual packet loss rate: %.4f\n", (_totalThrown * 1.0f) / (_totalKept + _totalThrown));
- printf("Channel rate: %.2f kbps\n",
- 0.001 * 8.0 * _sumChannelBytes / ((_framecnt * 1.0f) / _inst.maxFramerate));
- }
- else
- {
- printf("No packet losses inflicted\n");
- }
-
- NormalAsyncTest::Teardown();
-}
-
-void
-PacketLossTest::Setup()
-{
- const VideoSource source(_inname, _inst.width, _inst.height, _inst.maxFramerate);
-
- std::stringstream ss;
- std::string lossRateStr;
- ss << _lossRate;
- ss >> lossRateStr;
- _encodedName = source.GetName() + "-" + lossRateStr;
- _outname = "out-" + source.GetName() + "-" + lossRateStr;
-
- if (_lossProbability != _lossRate)
- {
- _encodedName += "-nack";
- _outname += "-nack";
- }
- _encodedName += ".vp8";
- _outname += ".yuv";
-
- _totalKept = 0;
- _totalThrown = 0;
- _sumChannelBytes = 0;
-
- NormalAsyncTest::Setup();
-}
-
-void
-PacketLossTest::CodecSpecific_InitBitrate()
-{
- assert(_bitRate > 0);
- WebRtc_UWord32 simulatedBitRate;
- if (_lossProbability != _lossRate)
- {
- // Simulating NACK
- simulatedBitRate = WebRtc_UWord32(_bitRate / (1 + _lossRate));
- }
- else
- {
- simulatedBitRate = _bitRate;
- }
- int rtt = 0;
- if (_inst.maxFramerate > 0)
- rtt = _rttFrames * (1000 / _inst.maxFramerate);
- _encoder->SetChannelParameters((WebRtc_UWord32)(_lossProbability * 255.0),
- rtt);
- _encoder->SetRates(simulatedBitRate, _inst.maxFramerate);
-}
-
-int PacketLossTest::DoPacketLoss()
-{
- // Only packet loss for delta frames
- if (_frameToDecode->_frame->GetLength() == 0 || _frameToDecode->_frame->GetFrameType() != kDeltaFrame)
- {
- _sumChannelBytes += _frameToDecode->_frame->GetLength();
- return 0;
- }
- unsigned char *packet = NULL;
- TestVideoEncodedBuffer newEncBuf;
- newEncBuf.VerifyAndAllocate(_lengthSourceFrame);
- _inBufIdx = 0;
- _outBufIdx = 0;
- int size = 1;
- int kept = 0;
- int thrown = 0;
- while ((size = NextPacket(1500, &packet)) > 0)
- {
- if (!PacketLoss(_lossProbability, thrown))
- {
- InsertPacket(&newEncBuf, packet, size);
- kept++;
- }
- else
- {
- // Use the ByteLoss function if you want to lose only
- // parts of a packet, and not the whole packet.
-
- //int size2 = ByteLoss(size, packet, 15);
- thrown++;
- //if (size2 != size)
- //{
- // InsertPacket(&newEncBuf, packet, size2);
- //}
- }
- }
- int lossResult = (thrown!=0); // 0 = no loss 1 = loss(es)
- if (lossResult)
- {
- lossResult += (kept==0); // 2 = all lost = full frame
- }
- _frameToDecode->_frame->CopyBuffer(newEncBuf.GetLength(), newEncBuf.GetBuffer());
- _sumChannelBytes += newEncBuf.GetLength();
- _totalKept += kept;
- _totalThrown += thrown;
-
- return lossResult;
- //printf("Threw away: %d out of %d packets\n", thrown, thrown + kept);
- //printf("Encoded left: %d bytes\n", _encodedVideoBuffer.GetLength());
-}
-
-int PacketLossTest::NextPacket(int mtu, unsigned char **pkg)
-{
- unsigned char *buf = _frameToDecode->_frame->GetBuffer();
- *pkg = buf + _inBufIdx;
- if (static_cast<long>(_frameToDecode->_frame->GetLength()) - _inBufIdx <= mtu)
- {
- int size = _frameToDecode->_frame->GetLength() - _inBufIdx;
- _inBufIdx = _frameToDecode->_frame->GetLength();
- return size;
- }
- _inBufIdx += mtu;
- return mtu;
-}
-
-int PacketLossTest::ByteLoss(int size, unsigned char *pkg, int bytesToLose)
-{
- return size;
-}
-
-void PacketLossTest::InsertPacket(TestVideoEncodedBuffer *buf, unsigned char *pkg, int size)
-{
- if (static_cast<long>(buf->GetSize()) - _outBufIdx < size)
- {
- printf("InsertPacket error!\n");
- return;
- }
- memcpy(buf->GetBuffer() + _outBufIdx, pkg, size);
- buf->UpdateLength(buf->GetLength() + size);
- _outBufIdx += size;
-}
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/packet_loss_test.h b/trunk/src/modules/video_coding/codecs/test_framework/packet_loss_test.h
deleted file mode 100644
index ea37681..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/packet_loss_test.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PACKET_LOSS_TEST_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PACKET_LOSS_TEST_H_
-
-#include <list>
-
-#include "normal_async_test.h"
-
-class PacketLossTest : public NormalAsyncTest
-{
-public:
- PacketLossTest();
- virtual ~PacketLossTest() {if(_lastFrame) {delete [] _lastFrame; _lastFrame = NULL;}}
- virtual void Encoded(const webrtc::EncodedImage& encodedImage);
- virtual void Decoded(const webrtc::RawImage& decodedImage);
-protected:
- PacketLossTest(std::string name, std::string description);
- PacketLossTest(std::string name,
- std::string description,
- double lossRate,
- bool useNack,
- unsigned int rttFrames = 0);
-
- virtual void Setup();
- virtual void Teardown();
- virtual void CodecSpecific_InitBitrate();
- virtual int DoPacketLoss();
- virtual int NextPacket(int size, unsigned char **pkg);
- virtual int ByteLoss(int size, unsigned char *pkg, int bytesToLose);
- virtual void InsertPacket(TestVideoEncodedBuffer *buf, unsigned char *pkg, int size);
- int _inBufIdx;
- int _outBufIdx;
-
- // When NACK is being simulated _lossProbabilty is zero,
- // otherwise it is set equal to _lossRate.
- // Desired channel loss rate.
- double _lossRate;
- // Probability used to simulate packet drops.
- double _lossProbability;
-
- int _totalKept;
- int _totalThrown;
- int _sumChannelBytes;
- std::list<WebRtc_UWord32> _frameQueue;
- WebRtc_UWord8* _lastFrame;
- WebRtc_UWord32 _lastFrameLength;
-};
-
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PACKET_LOSS_TEST_H_
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/performance_test.cc b/trunk/src/modules/video_coding/codecs/test_framework/performance_test.cc
deleted file mode 100644
index 250400c..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/performance_test.cc
+++ /dev/null
@@ -1,296 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "performance_test.h"
-
-#include <assert.h>
-
-#include "gtest/gtest.h"
-#include "testsupport/fileutils.h"
-#include "tick_util.h"
-
-using namespace webrtc;
-
-#define NUM_FRAMES 300
-
-PerformanceTest::PerformanceTest(WebRtc_UWord32 bitRate)
-:
-NormalAsyncTest(bitRate),
-_numCodecs(0),
-_tests(NULL),
-_encoders(NULL),
-_decoders(NULL),
-_threads(NULL),
-_rawImageLock(NULL),
-_encodeEvents(new EventWrapper*[1]),
-_stopped(true),
-_encodeCompleteCallback(NULL),
-_decodeCompleteCallback(NULL)
-{
-}
-
-PerformanceTest::PerformanceTest(WebRtc_UWord32 bitRate, WebRtc_UWord8 numCodecs)
-:
-NormalAsyncTest(bitRate),
-_numCodecs(numCodecs),
-_tests(new PerformanceTest*[_numCodecs]),
-_encoders(new VideoEncoder*[_numCodecs]),
-_decoders(new VideoDecoder*[_numCodecs]),
-_threads(new ThreadWrapper*[_numCodecs]),
-_rawImageLock(RWLockWrapper::CreateRWLock()),
-_encodeEvents(new EventWrapper*[_numCodecs]),
-_stopped(true),
-_encodeCompleteCallback(NULL),
-_decodeCompleteCallback(NULL)
-{
- for (int i=0; i < _numCodecs; i++)
- {
- _tests[i] = new PerformanceTest(bitRate);
- _encodeEvents[i] = EventWrapper::Create();
- }
-}
-
-PerformanceTest::~PerformanceTest()
-{
- if (_encoders != NULL)
- {
- delete [] _encoders;
- }
- if (_decoders != NULL)
- {
- delete [] _decoders;
- }
- if (_tests != NULL)
- {
- delete [] _tests;
- }
- if (_threads != NULL)
- {
- delete [] _threads;
- }
- if (_rawImageLock != NULL)
- {
- delete _rawImageLock;
- }
- if (_encodeEvents != NULL)
- {
- delete [] _encodeEvents;
- }
-}
-
-void
-PerformanceTest::Setup()
-{
- _inname = webrtc::test::ProjectRootPath() + "resources/foreman_cif.yuv";
- NormalAsyncTest::Setup(); // Setup input and output files
- CodecSettings(352, 288, 30, _bitRate); // common to all codecs
- for (int i=0; i < _numCodecs; i++)
- {
- _encoders[i] = CreateEncoder();
- _decoders[i] = CreateDecoder();
- if (_encoders[i] == NULL)
- {
- printf("Must create a codec specific test!\n");
- exit(EXIT_FAILURE);
- }
- if(_encoders[i]->InitEncode(&_inst, 4, 1440) < 0)
- {
- exit(EXIT_FAILURE);
- }
- if (_decoders[i]->InitDecode(&_inst, 1))
- {
- exit(EXIT_FAILURE);
- }
- _tests[i]->SetEncoder(_encoders[i]);
- _tests[i]->SetDecoder(_decoders[i]);
- _tests[i]->_rawImageLock = _rawImageLock;
- _encodeEvents[i]->Reset();
- _tests[i]->_encodeEvents[0] = _encodeEvents[i];
- _tests[i]->_inst = _inst;
- _threads[i] = ThreadWrapper::CreateThread(PerformanceTest::RunThread, _tests[i]);
- unsigned int id = 0;
- _tests[i]->_stopped = false;
- _threads[i]->Start(id);
- }
-}
-
-void
-PerformanceTest::Perform()
-{
- Setup();
- EventWrapper& sleepEvent = *EventWrapper::Create();
- const WebRtc_Word64 startTime = TickTime::MillisecondTimestamp();
- for (int i=0; i < NUM_FRAMES; i++)
- {
- {
- // Read a new frame from file
- WriteLockScoped imageLock(*_rawImageLock);
- _lengthEncFrame = 0;
- EXPECT_GT(fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile),
- 0u);
- if (feof(_sourceFile) != 0)
- {
- rewind(_sourceFile);
- }
- _inputVideoBuffer.VerifyAndAllocate(_inst.width*_inst.height*3/2);
- _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
- _inputVideoBuffer.SetTimeStamp((unsigned int) (_encFrameCnt * 9e4 / static_cast<float>(_inst.maxFramerate)));
- _inputVideoBuffer.SetWidth(_inst.width);
- _inputVideoBuffer.SetHeight(_inst.height);
- for (int i=0; i < _numCodecs; i++)
- {
- _tests[i]->_inputVideoBuffer.CopyPointer(_inputVideoBuffer);
- _encodeEvents[i]->Set();
- }
- }
- if (i < NUM_FRAMES - 1)
- {
- sleepEvent.Wait(33);
- }
- }
- for (int i=0; i < _numCodecs; i++)
- {
- _tests[i]->_stopped = true;
- _encodeEvents[i]->Set();
- _threads[i]->Stop();
- }
- const WebRtc_UWord32 totalTime =
- static_cast<WebRtc_UWord32>(TickTime::MillisecondTimestamp() - startTime);
- printf("Total time: %u\n", totalTime);
- delete &sleepEvent;
- Teardown();
-}
-
-void PerformanceTest::Teardown()
-{
- if (_encodeCompleteCallback != NULL)
- {
- delete _encodeCompleteCallback;
- }
- if (_decodeCompleteCallback != NULL)
- {
- delete _decodeCompleteCallback;
- }
- // main test only, all others have numCodecs = 0:
- if (_numCodecs > 0)
- {
- WriteLockScoped imageLock(*_rawImageLock);
- _inputVideoBuffer.Free();
- NormalAsyncTest::Teardown();
- }
- for (int i=0; i < _numCodecs; i++)
- {
- _encoders[i]->Release();
- delete _encoders[i];
- _decoders[i]->Release();
- delete _decoders[i];
- _tests[i]->_inputVideoBuffer.ClearPointer();
- _tests[i]->_rawImageLock = NULL;
- _tests[i]->Teardown();
- delete _tests[i];
- delete _encodeEvents[i];
- delete _threads[i];
- }
-}
-
-bool
-PerformanceTest::RunThread(void* obj)
-{
- PerformanceTest& test = *static_cast<PerformanceTest*>(obj);
- return test.PerformSingleTest();
-}
-
-bool
-PerformanceTest::PerformSingleTest()
-{
- if (_encodeCompleteCallback == NULL)
- {
- _encodeCompleteCallback = new VideoEncodeCompleteCallback(NULL, &_frameQueue, *this);
- _encoder->RegisterEncodeCompleteCallback(_encodeCompleteCallback);
- }
- if (_decodeCompleteCallback == NULL)
- {
- _decodeCompleteCallback = new VideoDecodeCompleteCallback(NULL, *this);
- _decoder->RegisterDecodeCompleteCallback(_decodeCompleteCallback);
- }
- (*_encodeEvents)->Wait(WEBRTC_EVENT_INFINITE); // The first event is used for every single test
- CodecSpecific_InitBitrate();
- bool complete = false;
- {
- ReadLockScoped imageLock(*_rawImageLock);
- complete = Encode();
- }
- if (!_frameQueue.Empty() || complete)
- {
- while (!_frameQueue.Empty())
- {
- _frameToDecode = static_cast<FrameQueueTuple *>(_frameQueue.PopFrame());
- int lost = DoPacketLoss();
- if (lost == 2)
- {
- // Lost the whole frame, continue
- _missingFrames = true;
- delete _frameToDecode;
- _frameToDecode = NULL;
- continue;
- }
- int ret = Decode(lost);
- delete _frameToDecode;
- _frameToDecode = NULL;
- if (ret < 0)
- {
- fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
- return false;
- }
- else if (ret < 0)
- {
- fprintf(stderr, "\n\nPositive return value from decode!\n\n");
- return false;
- }
- }
- }
- if (_stopped)
- {
- return false;
- }
- return true;
-}
-
-bool PerformanceTest::Encode()
-{
- RawImage rawImage;
- VideoBufferToRawImage(_inputVideoBuffer, rawImage);
- VideoFrameType frameType = kDeltaFrame;
- if (_requestKeyFrame && !(_encFrameCnt%50))
- {
- frameType = kKeyFrame;
- }
- webrtc::CodecSpecificInfo* codecSpecificInfo = CreateEncoderSpecificInfo();
- int ret = _encoder->Encode(rawImage, codecSpecificInfo, &frameType);
- EXPECT_EQ(ret, WEBRTC_VIDEO_CODEC_OK);
- if (codecSpecificInfo != NULL)
- {
- delete codecSpecificInfo;
- codecSpecificInfo = NULL;
- }
- assert(ret >= 0);
- return false;
-}
-
-int PerformanceTest::Decode(int lossValue)
-{
- EncodedImage encodedImage;
- VideoEncodedBufferToEncodedImage(*(_frameToDecode->_frame), encodedImage);
- encodedImage._completeFrame = !lossValue;
- int ret = _decoder->Decode(encodedImage, _missingFrames, NULL,
- _frameToDecode->_codecSpecificInfo);
- _missingFrames = false;
- return ret;
-}
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/performance_test.h b/trunk/src/modules/video_coding/codecs/test_framework/performance_test.h
deleted file mode 100644
index d060832..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/performance_test.h
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PERFORMANCE_TEST_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PERFORMANCE_TEST_H_
-
-#include "normal_async_test.h"
-#include "thread_wrapper.h"
-#include "rw_lock_wrapper.h"
-#include "event_wrapper.h"
-
-class PerformanceTest : public NormalAsyncTest
-{
-public:
- PerformanceTest(WebRtc_UWord32 bitRate, WebRtc_UWord8 numCodecs);
- virtual ~PerformanceTest();
-
- virtual void Perform();
- virtual void Print() {};
-
-protected:
- PerformanceTest(WebRtc_UWord32 bitRate);
- virtual void Setup();
- virtual bool Encode();
- virtual int Decode(int lossValue = 0);
- virtual void Teardown();
- static bool RunThread(void* obj);
- bool PerformSingleTest();
-
- virtual webrtc::VideoEncoder* CreateEncoder() const { return NULL; };
- virtual webrtc::VideoDecoder* CreateDecoder() const { return NULL; };
-
- WebRtc_UWord8 _numCodecs;
- PerformanceTest** _tests;
- webrtc::VideoEncoder** _encoders;
- webrtc::VideoDecoder** _decoders;
- webrtc::ThreadWrapper** _threads;
- webrtc::RWLockWrapper* _rawImageLock;
- webrtc::EventWrapper** _encodeEvents;
- FrameQueue _frameQueue;
- bool _stopped;
- webrtc::EncodedImageCallback* _encodeCompleteCallback;
- webrtc::DecodedImageCallback* _decodeCompleteCallback;
- FILE* _outFile;
-};
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_PERFORMANCE_TEST_H_
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/plotBenchmark.m b/trunk/src/modules/video_coding/codecs/test_framework/plotBenchmark.m
deleted file mode 100644
index 33c8eb6..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/plotBenchmark.m
+++ /dev/null
@@ -1,427 +0,0 @@
-function plotBenchmark(fileNames, export)
-%PLOTBENCHMARK Plots and exports video codec benchmarking results.
-% PLOTBENCHMARK(FILENAMES, EXPORT) parses the video codec benchmarking result
-% files given by the cell array of strings FILENAME. It plots the results and
-% optionally exports each plot to an appropriately named file.
-%
-% EXPORT parameter:
-% 'none' No file exports.
-% 'eps' Exports to eps files (default).
-% 'pdf' Exports to eps files and uses the command-line utility
-% epstopdf to obtain pdf files.
-%
-% Example:
-% plotBenchmark({'H264Benchmark.txt' 'LSVXBenchmark.txt'}, 'pdf')
-
-if (nargin < 1)
- error('Too few input arguments');
-elseif (nargin < 2)
- export = 'eps';
-end
-
-if ~iscell(fileNames)
- if ischar(fileNames)
- % one single file name as a string is ok
- if size(fileNames,1) > 1
- % this is a char matrix, not ok
- error('First argument must not be a char matrix');
- end
- % wrap in a cell array
- fileNames = {fileNames};
- else
- error('First argument must be a cell array of strings');
- end
-end
-
-if ~ischar(export)
- error('Second argument must be a string');
-end
-
-outpath = 'BenchmarkPlots';
-[status, errMsg] = mkdir(outpath);
-if status == 0
- error(errMsg);
-end
-
-nCases = 0;
-testCases = [];
-% Read each test result file
-for fileIdx = 1:length(fileNames)
- if ~isstr(fileNames{fileIdx})
- error('First argument must be a cell array of strings');
- end
-
- fid = fopen(fileNames{fileIdx}, 'rt');
- if fid == -1
- error(['Unable to open ' fileNames{fileIdx}]);
- end
-
- version = '1.0';
- if ~strcmp(fgetl(fid), ['#!benchmark' version])
- fclose(fid);
- error(['Requires benchmark file format version ' version]);
- end
-
- % Parse results file into testCases struct
- codec = fgetl(fid);
- tline = fgetl(fid);
- while(tline ~= -1)
- nCases = nCases + 1;
-
- delim = strfind(tline, ',');
- name = tline(1:delim(1)-1);
- % Drop underscored suffix from name
- underscore = strfind(name, '_');
- if ~isempty(underscore)
- name = name(1:underscore(1)-1);
- end
-
- resolution = tline(delim(1)+1:delim(2)-1);
- frameRate = tline(delim(2)+1:end);
-
- tline = fgetl(fid);
- delim = strfind(tline, ',');
- bitrateLabel = tline(1:delim(1)-1);
- bitrate = sscanf(tline(delim(1):end),',%f');
-
- tline = fgetl(fid);
- delim = strfind(tline, ',');
- psnrLabel = tline(1:delim(1)-1);
- psnr = sscanf(tline(delim(1):end),',%f');
-
-
- % Default data for the optional lines
- speedLabel = 'Default';
- speed = 0;
- ssimLabel = 'Default';
- ssim = 0;
-
- tline = fgetl(fid);
- delim = strfind(tline, ',');
-
- while ~isempty(delim)
- % More data
- % Check type of data
- if strncmp(lower(tline), 'speed', 5)
- % Speed data included
- speedLabel = tline(1:delim(1)-1);
- speed = sscanf(tline(delim(1):end), ',%f');
-
- tline = fgetl(fid);
-
- elseif strncmp(lower(tline), 'encode time', 11)
- % Encode and decode times included
- % TODO: take care of the data
-
- % pop two lines from file
- tline = fgetl(fid);
- tline = fgetl(fid);
-
- elseif strncmp(tline, 'SSIM', 4)
- % SSIM data included
- ssimLabel = tline(1:delim(1)-1);
- ssim = sscanf(tline(delim(1):end), ',%f');
-
- tline = fgetl(fid);
- end
- delim = strfind(tline, ',');
- end
-
- testCases = [testCases struct('codec', codec, 'name', name, 'resolution', ...
- resolution, 'frameRate', frameRate, 'bitrate', bitrate, 'psnr', psnr, ...
- 'speed', speed, 'bitrateLabel', bitrateLabel, 'psnrLabel', psnrLabel, ...
- 'speedLabel', speedLabel, ...
- 'ssim', ssim, 'ssimLabel', ssimLabel)];
-
- tline = fgetl(fid);
- end
-
- fclose(fid);
-end
-
-i = 0;
-casesPsnr = testCases;
-while ~isempty(casesPsnr)
- i = i + 1;
- casesPsnr = plotOnePsnr(casesPsnr, i, export, outpath);
-end
-
-casesSSIM = testCases;
-while ~isempty(casesSSIM)
- i = i + 1;
- casesSSIM = plotOneSSIM(casesSSIM, i, export, outpath);
-end
-
-casesSpeed = testCases;
-while ~isempty(casesSpeed)
- if casesSpeed(1).speed == 0
- casesSpeed = casesSpeed(2:end);
- else
- i = i + 1;
- casesSpeed = plotOneSpeed(casesSpeed, i, export, outpath);
- end
-end
-
-
-
-%%%%%%%%%%%%%%%%%%
-%% SUBFUNCTIONS %%
-%%%%%%%%%%%%%%%%%%
-
-function casesOut = plotOnePsnr(cases, num, export, outpath)
-% Find matching specs
-plotIdx = 1;
-for i = 2:length(cases)
- if strcmp(cases(1).resolution, cases(i).resolution) & ...
- strcmp(cases(1).frameRate, cases(i).frameRate)
- plotIdx = [plotIdx i];
- end
-end
-
-% Return unplotted cases
-casesOut = cases(setdiff(1:length(cases), plotIdx));
-cases = cases(plotIdx);
-
-% Prune similar results
-for i = 1:length(cases)
- simIndx = find(abs(cases(i).bitrate - [cases(i).bitrate(2:end) ; 0]) < 10);
- while ~isempty(simIndx)
- diffIndx = setdiff(1:length(cases(i).bitrate), simIndx);
- cases(i).psnr = cases(i).psnr(diffIndx);
- cases(i).bitrate = cases(i).bitrate(diffIndx);
- simIndx = find(abs(cases(i).bitrate - [cases(i).bitrate(2:end) ; 0]) < 10);
- end
-end
-
-% Prepare figure with axis labels and so on
-hFig = figure(num);
-clf;
-hold on;
-grid on;
-axis([0 1100 20 50]);
-set(gca, 'XTick', 0:200:1000);
-set(gca, 'YTick', 20:10:60);
-xlabel(cases(1).bitrateLabel);
-ylabel(cases(1).psnrLabel);
-res = cases(1).resolution;
-frRate = cases(1).frameRate;
-title([res ', ' frRate]);
-
-hLines = [];
-codecs = {};
-sequences = {};
-i = 0;
-while ~isempty(cases)
- i = i + 1;
- [cases, hLine, codec, sequences] = plotOneCodec(cases, 'bitrate', 'psnr', i, sequences, 1);
-
- % Stored to generate the legend
- hLines = [hLines ; hLine];
- codecs = {codecs{:} codec};
-end
-legend(hLines, codecs, 4);
-hold off;
-
-if ~strcmp(export, 'none')
- % Export figure to an eps file
- res = stripws(res);
- frRate = stripws(frRate);
- exportName = [outpath '/psnr-' res '-' frRate];
- exportfig(hFig, exportName, 'Format', 'eps2', 'Color', 'cmyk');
-end
-
-if strcmp(export, 'pdf')
- % Use the epstopdf utility to convert to pdf
- system(['epstopdf ' exportName '.eps']);
-end
-
-
-function casesOut = plotOneSSIM(cases, num, export, outpath)
-% Find matching specs
-plotIdx = 1;
-for i = 2:length(cases)
- if strcmp(cases(1).resolution, cases(i).resolution) & ...
- strcmp(cases(1).frameRate, cases(i).frameRate)
- plotIdx = [plotIdx i];
- end
-end
-
-% Return unplotted cases
-casesOut = cases(setdiff(1:length(cases), plotIdx));
-cases = cases(plotIdx);
-
-% Prune similar results
-for i = 1:length(cases)
- simIndx = find(abs(cases(i).bitrate - [cases(i).bitrate(2:end) ; 0]) < 10);
- while ~isempty(simIndx)
- diffIndx = setdiff(1:length(cases(i).bitrate), simIndx);
- cases(i).ssim = cases(i).ssim(diffIndx);
- cases(i).bitrate = cases(i).bitrate(diffIndx);
- simIndx = find(abs(cases(i).bitrate - [cases(i).bitrate(2:end) ; 0]) < 10);
- end
-end
-
-% Prepare figure with axis labels and so on
-hFig = figure(num);
-clf;
-hold on;
-grid on;
-axis([0 1100 0.5 1]); % y-limit are set to 'auto' below
-set(gca, 'XTick', 0:200:1000);
-%set(gca, 'YTick', 20:10:60);
-xlabel(cases(1).bitrateLabel);
-ylabel(cases(1).ssimLabel);
-res = cases(1).resolution;
-frRate = cases(1).frameRate;
-title([res ', ' frRate]);
-
-hLines = [];
-codecs = {};
-sequences = {};
-i = 0;
-while ~isempty(cases)
- i = i + 1;
- [cases, hLine, codec, sequences] = plotOneCodec(cases, 'bitrate', 'ssim', i, sequences, 1);
-
- % Stored to generate the legend
- hLines = [hLines ; hLine];
- codecs = {codecs{:} codec};
-end
-%set(gca,'YLimMode','auto')
-set(gca,'YLim',[0.5 1])
-set(gca,'YScale','log')
-legend(hLines, codecs, 4);
-hold off;
-
-if ~strcmp(export, 'none')
- % Export figure to an eps file
- res = stripws(res);
- frRate = stripws(frRate);
- exportName = [outpath '/psnr-' res '-' frRate];
- exportfig(hFig, exportName, 'Format', 'eps2', 'Color', 'cmyk');
-end
-
-if strcmp(export, 'pdf')
- % Use the epstopdf utility to convert to pdf
- system(['epstopdf ' exportName '.eps']);
-end
-
-
-function casesOut = plotOneSpeed(cases, num, export, outpath)
-% Find matching specs
-plotIdx = 1;
-for i = 2:length(cases)
- if strcmp(cases(1).resolution, cases(i).resolution) & ...
- strcmp(cases(1).frameRate, cases(i).frameRate) & ...
- strcmp(cases(1).name, cases(i).name)
- plotIdx = [plotIdx i];
- end
-end
-
-% Return unplotted cases
-casesOut = cases(setdiff(1:length(cases), plotIdx));
-cases = cases(plotIdx);
-
-% Prune similar results
-for i = 1:length(cases)
- simIndx = find(abs(cases(i).psnr - [cases(i).psnr(2:end) ; 0]) < 0.25);
- while ~isempty(simIndx)
- diffIndx = setdiff(1:length(cases(i).psnr), simIndx);
- cases(i).psnr = cases(i).psnr(diffIndx);
- cases(i).speed = cases(i).speed(diffIndx);
- simIndx = find(abs(cases(i).psnr - [cases(i).psnr(2:end) ; 0]) < 0.25);
- end
-end
-
-hFig = figure(num);
-clf;
-hold on;
-%grid on;
-xlabel(cases(1).psnrLabel);
-ylabel(cases(1).speedLabel);
-res = cases(1).resolution;
-name = cases(1).name;
-frRate = cases(1).frameRate;
-title([name ', ' res ', ' frRate]);
-
-hLines = [];
-codecs = {};
-sequences = {};
-i = 0;
-while ~isempty(cases)
- i = i + 1;
- [cases, hLine, codec, sequences] = plotOneCodec(cases, 'psnr', 'speed', i, sequences, 0);
-
- % Stored to generate the legend
- hLines = [hLines ; hLine];
- codecs = {codecs{:} codec};
-end
-legend(hLines, codecs, 1);
-hold off;
-
-if ~strcmp(export, 'none')
- % Export figure to an eps file
- res = stripws(res);
- frRate = stripws(frRate);
- exportName = [outpath '/speed-' name '-' res '-' frRate];
- exportfig(hFig, exportName, 'Format', 'eps2', 'Color', 'cmyk');
-end
-
-if strcmp(export, 'pdf')
- % Use the epstopdf utility to convert to pdf
- system(['epstopdf ' exportName '.eps']);
-end
-
-
-function [casesOut, hLine, codec, sequences] = plotOneCodec(cases, xfield, yfield, num, sequences, annotatePlot)
-plotStr = {'gx-', 'bo-', 'r^-', 'kd-', 'cx-', 'go--', 'b^--'};
-% Find matching codecs
-plotIdx = 1;
-for i = 2:length(cases)
- if strcmp(cases(1).codec, cases(i).codec)
- plotIdx = [plotIdx i];
- end
-end
-
-% Return unplotted cases
-casesOut = cases(setdiff(1:length(cases), plotIdx));
-cases = cases(plotIdx);
-
-for i = 1:length(cases)
- % Plot a single case
- hLine = plot(getfield(cases(i), xfield), getfield(cases(i), yfield), plotStr{num}, ...
- 'LineWidth', 1.1, 'MarkerSize', 6);
-end
-
-% hLine handle and codec are returned to construct the legend afterwards
-codec = cases(1).codec;
-
-if annotatePlot == 0
- return;
-end
-
-for i = 1:length(cases)
- % Print the codec name as a text label
- % Ensure each codec is only printed once
- sequencePlotted = 0;
- for j = 1:length(sequences)
- if strcmp(cases(i).name, sequences{j})
- sequencePlotted = 1;
- break;
- end
- end
-
- if sequencePlotted == 0
- text(getfield(cases(i), xfield, {1}), getfield(cases(i), yfield, {1}), ...
- [' ' cases(i).name]);
- sequences = {sequences{:} cases(i).name};
- end
-end
-
-
-% Strip whitespace from string
-function str = stripws(str)
-if ~isstr(str)
- error('String required');
-end
-str = str(setdiff(1:length(str), find(isspace(str) == 1)));
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/test.cc b/trunk/src/modules/video_coding/codecs/test_framework/test.cc
deleted file mode 100644
index 3eb6935..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/test.cc
+++ /dev/null
@@ -1,165 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "test.h"
-
-#include <cstring>
-#include <iostream>
-
-#include "testsupport/metrics/video_metrics.h"
-
-using namespace webrtc;
-
-long filesize(const char *filename); // local function defined at end of file
-
-Test::Test(std::string name, std::string description)
-:
-_bitRate(0),
-_inname(""),
-_outname(""),
-_encodedName(""),
-_name(name),
-_description(description)
-{
- memset(&_inst, 0, sizeof(_inst));
- unsigned int seed = static_cast<unsigned int>(0);
- std::srand(seed);
-}
-
-Test::Test(std::string name, std::string description, WebRtc_UWord32 bitRate)
-:
-_bitRate(bitRate),
-_inname(""),
-_outname(""),
-_encodedName(""),
-_name(name),
-_description(description)
-{
- memset(&_inst, 0, sizeof(_inst));
- unsigned int seed = static_cast<unsigned int>(0);
- std::srand(seed);
-}
-
-void
-Test::Print()
-{
- std::cout << _name << " completed!" << std::endl;
- (*_log) << _name << std::endl;
- (*_log) << _description << std::endl;
- (*_log) << "Input file: " << _inname << std::endl;
- (*_log) << "Output file: " << _outname << std::endl;
- webrtc::test::QualityMetricsResult psnr;
- webrtc::test::QualityMetricsResult ssim;
- I420PSNRFromFiles(_inname.c_str(), _outname.c_str(), _inst.width,
- _inst.height, &psnr);
- I420SSIMFromFiles(_inname.c_str(), _outname.c_str(), _inst.width,
- _inst.height, &ssim);
-
- (*_log) << "PSNR: " << psnr.average << std::endl;
- std::cout << "PSNR: " << psnr.average << std::endl << std::endl;
- (*_log) << "SSIM: " << ssim.average << std::endl;
- std::cout << "SSIM: " << ssim.average << std::endl << std::endl;
- (*_log) << std::endl;
-}
-
-void
-Test::Setup()
-{
- int widhei = _inst.width*_inst.height;
- _lengthSourceFrame = 3*widhei/2;
- _sourceBuffer = new unsigned char[_lengthSourceFrame];
-}
-
-void
-Test::CodecSettings(int width, int height, WebRtc_UWord32 frameRate /*=30*/, WebRtc_UWord32 bitRate /*=0*/)
-{
- if (bitRate > 0)
- {
- _bitRate = bitRate;
- }
- else if (_bitRate == 0)
- {
- _bitRate = 600;
- }
- _inst.codecType = kVideoCodecVP8;
- _inst.codecSpecific.VP8.feedbackModeOn = true;
- _inst.maxFramerate = (unsigned char)frameRate;
- _inst.startBitrate = (int)_bitRate;
- _inst.maxBitrate = 8000;
- _inst.width = width;
- _inst.height = height;
-}
-
-void
-Test::Teardown()
-{
- delete [] _sourceBuffer;
-}
-
-void
-Test::SetEncoder(webrtc::VideoEncoder*encoder)
-{
- _encoder = encoder;
-}
-
-void
-Test::SetDecoder(VideoDecoder*decoder)
-{
- _decoder = decoder;
-}
-
-void
-Test::SetLog(std::fstream* log)
-{
- _log = log;
-}
-
-double Test::ActualBitRate(int nFrames)
-{
- return 8.0 * _sumEncBytes / (nFrames / _inst.maxFramerate);
-}
-
-bool Test::PacketLoss(double lossRate, int /*thrown*/)
-{
- return RandUniform() < lossRate;
-}
-
-void
-Test::VideoBufferToRawImage(TestVideoBuffer& videoBuffer, RawImage &image)
-{
- image._buffer = videoBuffer.GetBuffer();
- image._size = videoBuffer.GetSize();
- image._length = videoBuffer.GetLength();
- image._width = videoBuffer.GetWidth();
- image._height = videoBuffer.GetHeight();
- image._timeStamp = videoBuffer.GetTimeStamp();
-}
-void
-Test::VideoEncodedBufferToEncodedImage(TestVideoEncodedBuffer& videoBuffer, EncodedImage &image)
-{
- image._buffer = videoBuffer.GetBuffer();
- image._length = videoBuffer.GetLength();
- image._size = videoBuffer.GetSize();
- image._frameType = static_cast<VideoFrameType>(videoBuffer.GetFrameType());
- image._timeStamp = videoBuffer.GetTimeStamp();
- image._encodedWidth = videoBuffer.GetCaptureWidth();
- image._encodedHeight = videoBuffer.GetCaptureHeight();
- image._completeFrame = true;
-}
-
-long filesize(const char *filename)
-{
- FILE *f = fopen(filename,"rb"); /* open the file in read only */
- long size = 0;
- if (fseek(f,0,SEEK_END)==0) /* seek was successful */
- size = ftell(f);
- fclose(f);
- return size;
-}
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/test.h b/trunk/src/modules/video_coding/codecs/test_framework/test.h
deleted file mode 100644
index a31868b..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/test.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_TEST_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_TEST_H_
-
-#include "video_codec_interface.h"
-#include "video_buffer.h"
-#include <string>
-#include <fstream>
-#include <cstdlib>
-
-class Test
-{
-public:
- Test(std::string name, std::string description);
- Test(std::string name, std::string description, WebRtc_UWord32 bitRate);
- virtual ~Test() {};
- virtual void Perform()=0;
- virtual void Print();
- void SetEncoder(webrtc::VideoEncoder *encoder);
- void SetDecoder(webrtc::VideoDecoder *decoder);
- void SetLog(std::fstream* log);
-
-protected:
- virtual void Setup();
- virtual void CodecSettings(int width,
- int height,
- WebRtc_UWord32 frameRate=30,
- WebRtc_UWord32 bitRate=0);
- virtual void Teardown();
- double ActualBitRate(int nFrames);
- virtual bool PacketLoss(double lossRate, int /*thrown*/);
- static double RandUniform() { return (std::rand() + 1.0)/(RAND_MAX + 1.0); }
- static void VideoBufferToRawImage(TestVideoBuffer& videoBuffer,
- webrtc::RawImage &image);
- static void VideoEncodedBufferToEncodedImage(TestVideoEncodedBuffer& videoBuffer,
- webrtc::EncodedImage &image);
-
- webrtc::VideoEncoder* _encoder;
- webrtc::VideoDecoder* _decoder;
- WebRtc_UWord32 _bitRate;
- unsigned int _lengthSourceFrame;
- unsigned char* _sourceBuffer;
- TestVideoBuffer _inputVideoBuffer;
- TestVideoEncodedBuffer _encodedVideoBuffer;
- TestVideoBuffer _decodedVideoBuffer;
- webrtc::VideoCodec _inst;
- std::fstream* _log;
- std::string _inname;
- std::string _outname;
- std::string _encodedName;
- int _sumEncBytes;
-
-private:
- std::string _name;
- std::string _description;
-
-};
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAWEWORK_TEST_H_
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/test_framework.gypi b/trunk/src/modules/video_coding/codecs/test_framework/test_framework.gypi
deleted file mode 100644
index 3639e9a..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/test_framework.gypi
+++ /dev/null
@@ -1,71 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- # Exclude the test target when building with chromium.
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'test_framework',
- 'type': '<(library)',
-
- 'dependencies': [
- '<(webrtc_root)/../test/metrics.gyp:metrics',
- '<(webrtc_root)/../test/test.gyp:test_support',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
- ],
-
- 'include_dirs': [
- '../interface',
- '<(webrtc_root)/../testing/gtest/include',
- '../../../../common_video/interface',
- ],
-
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- ],
- },
-
- 'sources': [
- # header files
- 'benchmark.h',
- 'normal_async_test.h',
- 'normal_test.h',
- 'packet_loss_test.h',
- 'performance_test.h',
- 'test.h',
- 'unit_test.h',
- 'video_buffer.h',
- 'video_source.h',
-
- # source files
- 'benchmark.cc',
- 'normal_async_test.cc',
- 'normal_test.cc',
- 'packet_loss_test.cc',
- 'performance_test.cc',
- 'test.cc',
- 'unit_test.cc',
- 'video_buffer.cc',
- 'video_source.cc',
- ],
- },
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/unit_test.cc b/trunk/src/modules/video_coding/codecs/test_framework/unit_test.cc
deleted file mode 100644
index 99bd7fb..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/unit_test.cc
+++ /dev/null
@@ -1,817 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <math.h>
-#include <stdlib.h>
-#include <string.h>
-
-#include <cassert>
-
-#include "gtest/gtest.h"
-#include "testsupport/fileutils.h"
-#include "tick_util.h"
-#include "unit_test.h"
-#include "video_source.h"
-
-using namespace webrtc;
-
-UnitTest::UnitTest()
-:
-Test("UnitTest", "Unit test"),
-_tests(0),
-_errors(0),
-_source(NULL),
-_refFrame(NULL),
-_refEncFrame(NULL),
-_refDecFrame(NULL),
-_refEncFrameLength(0),
-_sourceFile(NULL),
-_encodeCompleteCallback(NULL),
-_decodeCompleteCallback(NULL)
-{
-}
-
-UnitTest::UnitTest(std::string name, std::string description)
-:
-Test(name, description),
-_tests(0),
-_errors(0),
-_source(NULL),
-_refFrame(NULL),
-_refEncFrame(NULL),
-_refDecFrame(NULL),
-_refEncFrameLength(0),
-_sourceFile(NULL),
-_encodeCompleteCallback(NULL),
-_decodeCompleteCallback(NULL)
-{
-}
-
-UnitTest::~UnitTest()
-{
- if (_encodeCompleteCallback) {
- delete _encodeCompleteCallback;
- }
-
- if (_decodeCompleteCallback) {
- delete _decodeCompleteCallback;
- }
-
- if (_source) {
- delete _source;
- }
-
- if (_refFrame) {
- delete [] _refFrame;
- }
-
- if (_refDecFrame) {
- delete [] _refDecFrame;
- }
-
- if (_sourceBuffer) {
- delete [] _sourceBuffer;
- }
-
- if (_sourceFile) {
- fclose(_sourceFile);
- }
-
- if (_refEncFrame) {
- delete [] _refEncFrame;
- }
-}
-
-WebRtc_Word32
-UnitTestEncodeCompleteCallback::Encoded(EncodedImage& encodedImage,
- const webrtc::CodecSpecificInfo* codecSpecificInfo,
- const webrtc::RTPFragmentationHeader*
- fragmentation)
-{
- _encodedVideoBuffer->VerifyAndAllocate(encodedImage._size);
- _encodedVideoBuffer->CopyBuffer(encodedImage._size, encodedImage._buffer);
- _encodedVideoBuffer->UpdateLength(encodedImage._length);
- _encodedVideoBuffer->SetFrameType(encodedImage._frameType);
- _encodedVideoBuffer->SetCaptureWidth(
- (WebRtc_UWord16)encodedImage._encodedWidth);
- _encodedVideoBuffer->SetCaptureHeight(
- (WebRtc_UWord16)encodedImage._encodedHeight);
- _encodedVideoBuffer->SetTimeStamp(encodedImage._timeStamp);
- _encodeComplete = true;
- _encodedFrameType = encodedImage._frameType;
- return 0;
-}
-
-WebRtc_Word32 UnitTestDecodeCompleteCallback::Decoded(RawImage& image)
-{
- _decodedVideoBuffer->VerifyAndAllocate(image._length);
- _decodedVideoBuffer->CopyBuffer(image._length, image._buffer);
- _decodedVideoBuffer->SetWidth(image._width);
- _decodedVideoBuffer->SetHeight(image._height);
- _decodedVideoBuffer->SetTimeStamp(image._timeStamp);
- _decodeComplete = true;
- return 0;
-}
-
-bool
-UnitTestEncodeCompleteCallback::EncodeComplete()
-{
- if (_encodeComplete)
- {
- _encodeComplete = false;
- return true;
- }
- return false;
-}
-
-VideoFrameType
-UnitTestEncodeCompleteCallback::EncodedFrameType() const
-{
- return _encodedFrameType;
-}
-
-bool
-UnitTestDecodeCompleteCallback::DecodeComplete()
-{
- if (_decodeComplete)
- {
- _decodeComplete = false;
- return true;
- }
- return false;
-}
-
-WebRtc_UWord32
-UnitTest::WaitForEncodedFrame() const
-{
- WebRtc_Word64 startTime = TickTime::MillisecondTimestamp();
- while (TickTime::MillisecondTimestamp() - startTime < kMaxWaitEncTimeMs)
- {
- if (_encodeCompleteCallback->EncodeComplete())
- {
- return _encodedVideoBuffer.GetLength();
- }
- }
- return 0;
-}
-
-WebRtc_UWord32
-UnitTest::WaitForDecodedFrame() const
-{
- WebRtc_Word64 startTime = TickTime::MillisecondTimestamp();
- while (TickTime::MillisecondTimestamp() - startTime < kMaxWaitDecTimeMs)
- {
- if (_decodeCompleteCallback->DecodeComplete())
- {
- return _decodedVideoBuffer.GetLength();
- }
- }
- return 0;
-}
-
-WebRtc_UWord32
-UnitTest::CodecSpecific_SetBitrate(WebRtc_UWord32 bitRate,
- WebRtc_UWord32 /* frameRate */)
-{
- return _encoder->SetRates(bitRate, _inst.maxFramerate);
-}
-
-void
-UnitTest::Setup()
-{
- // Use _sourceFile as a check to prevent multiple Setup() calls.
- if (_sourceFile != NULL)
- {
- return;
- }
-
- if (_encodeCompleteCallback == NULL)
- {
- _encodeCompleteCallback =
- new UnitTestEncodeCompleteCallback(&_encodedVideoBuffer);
- }
- if (_decodeCompleteCallback == NULL)
- {
- _decodeCompleteCallback =
- new UnitTestDecodeCompleteCallback(&_decodedVideoBuffer);
- }
-
- _encoder->RegisterEncodeCompleteCallback(_encodeCompleteCallback);
- _decoder->RegisterDecodeCompleteCallback(_decodeCompleteCallback);
-
- _source = new VideoSource(webrtc::test::ProjectRootPath() +
- "resources/foreman_cif.yuv", kCIF);
-
- _lengthSourceFrame = _source->GetFrameLength();
- _refFrame = new unsigned char[_lengthSourceFrame];
- _refDecFrame = new unsigned char[_lengthSourceFrame];
- _sourceBuffer = new unsigned char [_lengthSourceFrame];
- _sourceFile = fopen(_source->GetFileName().c_str(), "rb");
- ASSERT_TRUE(_sourceFile != NULL);
-
- _inst.maxFramerate = _source->GetFrameRate();
- _bitRate = 300;
- _inst.startBitrate = 300;
- _inst.maxBitrate = 4000;
- _inst.width = _source->GetWidth();
- _inst.height = _source->GetHeight();
-
- // Get input frame.
- _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
- ASSERT_TRUE(fread(_refFrame, 1, _lengthSourceFrame, _sourceFile)
- == _lengthSourceFrame);
- _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _refFrame);
- rewind(_sourceFile);
-
- // Get a reference encoded frame.
- _encodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
-
- RawImage image;
- VideoBufferToRawImage(_inputVideoBuffer, image);
-
- // Ensures our initial parameters are valid.
- EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
- VideoFrameType videoFrameType = kDeltaFrame;
- _encoder->Encode(image, NULL, &videoFrameType);
- _refEncFrameLength = WaitForEncodedFrame();
- ASSERT_TRUE(_refEncFrameLength > 0);
- _refEncFrame = new unsigned char[_refEncFrameLength];
- memcpy(_refEncFrame, _encodedVideoBuffer.GetBuffer(), _refEncFrameLength);
-
- // Get a reference decoded frame.
- _decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
- EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
-
- if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
- {
- exit(EXIT_FAILURE);
- }
-
- unsigned int frameLength = 0;
- int i=0;
- while (frameLength == 0)
- {
- if (i > 0)
- {
- // Insert yet another frame
- _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
- ASSERT_TRUE(fread(_refFrame, 1, _lengthSourceFrame,
- _sourceFile) == _lengthSourceFrame);
- _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _refFrame);
- _inputVideoBuffer.SetWidth(_source->GetWidth());
- _inputVideoBuffer.SetHeight(_source->GetHeight());
- VideoBufferToRawImage(_inputVideoBuffer, image);
- _encoder->Encode(image, NULL, &videoFrameType);
- ASSERT_TRUE(WaitForEncodedFrame() > 0);
- }
- EncodedImage encodedImage;
- VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
- ASSERT_TRUE(_decoder->Decode(encodedImage, 0, NULL)
- == WEBRTC_VIDEO_CODEC_OK);
- frameLength = WaitForDecodedFrame();
- _encodedVideoBuffer.Reset();
- _encodedVideoBuffer.UpdateLength(0);
- i++;
- }
- rewind(_sourceFile);
- EXPECT_TRUE(frameLength == _lengthSourceFrame);
- memcpy(_refDecFrame, _decodedVideoBuffer.GetBuffer(), _lengthSourceFrame);
-}
-
-void
-UnitTest::Teardown()
-{
- // Use _sourceFile as a check to prevent multiple Teardown() calls.
- if (_sourceFile == NULL)
- {
- return;
- }
-
- _encoder->Release();
- _decoder->Release();
-
- fclose(_sourceFile);
- _sourceFile = NULL;
- delete [] _refFrame;
- _refFrame = NULL;
- delete [] _refEncFrame;
- _refEncFrame = NULL;
- delete [] _refDecFrame;
- _refDecFrame = NULL;
- delete [] _sourceBuffer;
- _sourceBuffer = NULL;
-}
-
-void
-UnitTest::Print()
-{
- printf("Unit Test\n\n%i tests completed\n", _tests);
- if (_errors > 0)
- {
- printf("%i FAILED\n\n", _errors);
- }
- else
- {
- printf("ALL PASSED\n\n");
- }
-}
-
-int
-UnitTest::DecodeWithoutAssert()
-{
- EncodedImage encodedImage;
- VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
- int ret = _decoder->Decode(encodedImage, 0, NULL);
- int frameLength = WaitForDecodedFrame();
- _encodedVideoBuffer.Reset();
- _encodedVideoBuffer.UpdateLength(0);
- return ret == WEBRTC_VIDEO_CODEC_OK ? frameLength : ret;
-}
-
-int
-UnitTest::Decode()
-{
- EncodedImage encodedImage;
- VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
- if (encodedImage._length == 0)
- {
- return WEBRTC_VIDEO_CODEC_OK;
- }
- int ret = _decoder->Decode(encodedImage, 0, NULL);
- unsigned int frameLength = WaitForDecodedFrame();
- assert(ret == WEBRTC_VIDEO_CODEC_OK && (frameLength == 0 || frameLength
- == _lengthSourceFrame));
- EXPECT_TRUE(ret == WEBRTC_VIDEO_CODEC_OK && (frameLength == 0 || frameLength
- == _lengthSourceFrame));
- _encodedVideoBuffer.Reset();
- _encodedVideoBuffer.UpdateLength(0);
- return ret == WEBRTC_VIDEO_CODEC_OK ? frameLength : ret;
-}
-
-// Test pure virtual VideoEncoder and VideoDecoder APIs.
-void
-UnitTest::Perform()
-{
- UnitTest::Setup();
- int frameLength;
- RawImage inputImage;
- EncodedImage encodedImage;
- VideoFrameType videoFrameType = kDeltaFrame;
-
- //----- Encoder parameter tests -----
-
- //-- Calls before InitEncode() --
- // We want to revert the initialization done in Setup().
- EXPECT_TRUE(_encoder->Release() == WEBRTC_VIDEO_CODEC_OK);
- VideoBufferToRawImage(_inputVideoBuffer, inputImage);
- EXPECT_TRUE(_encoder->Encode(inputImage, NULL, &videoFrameType )
- == WEBRTC_VIDEO_CODEC_UNINITIALIZED);
-
- //-- InitEncode() errors --
- // Null pointer.
- EXPECT_TRUE(_encoder->InitEncode(NULL, 1, 1440) ==
- WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
- // bit rate exceeds max bit rate
- WebRtc_Word32 tmpBitRate = _inst.startBitrate;
- WebRtc_Word32 tmpMaxBitRate = _inst.maxBitrate;
- _inst.startBitrate = 4000;
- _inst.maxBitrate = 3000;
- EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
- WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
- _inst.startBitrate = tmpBitRate;
- _inst.maxBitrate = tmpMaxBitRate; //unspecified value
-
- // Bad framerate.
- _inst.maxFramerate = 0;
- EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
- WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
- // Seems like we should allow any framerate in range [0, 255].
- //_inst.frameRate = 100;
- //EXPECT_TRUE(_encoder->InitEncode(&_inst, 1) == -1); // FAILS
- _inst.maxFramerate = 30;
-
- // Bad bitrate.
- _inst.startBitrate = -1;
- EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
- WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
- _inst.maxBitrate = _inst.startBitrate - 1;
- EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
- WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
- _inst.maxBitrate = 0;
- _inst.startBitrate = 300;
-
- // Bad maxBitRate.
- _inst.maxBitrate = 200;
- EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
- WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
- _inst.maxBitrate = 4000;
-
- // Bad width.
- _inst.width = 0;
- EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) < 0);
- // Should there be a width and height cap?
- //_inst.width = 10000;
- //EXPECT_TRUE(_encoder->InitEncode(&_inst, 1) == -1);
- _inst.width = _source->GetWidth();
-
- // Bad height.
- _inst.height = 0;
- EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) < 0);
- _inst.height = _source->GetHeight();
-
- // Bad number of cores.
- EXPECT_TRUE(_encoder->InitEncode(&_inst, -1, 1440) ==
- WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
-
- EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
-
- //-- Encode() errors --
-
- // inputVideoBuffer unallocated.
- _inputVideoBuffer.Free();
- VideoBufferToRawImage(_inputVideoBuffer, inputImage);
- EXPECT_TRUE(_encoder->Encode(inputImage, NULL, &videoFrameType) ==
- WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
- _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
- _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _refFrame);
-
- //----- Encoder stress tests -----
-
- // Vary frame rate and I-frame request.
- VideoBufferToRawImage(_inputVideoBuffer, inputImage);
- for (int i = 1; i <= 60; i++)
- {
- VideoFrameType frameType = !(i % 2) ? kKeyFrame : kDeltaFrame;
- EXPECT_TRUE(_encoder->Encode(inputImage, NULL, &frameType) ==
- WEBRTC_VIDEO_CODEC_OK);
- EXPECT_TRUE(WaitForEncodedFrame() > 0);
- }
-
- // Init then encode.
- _encodedVideoBuffer.UpdateLength(0);
- _encodedVideoBuffer.Reset();
- EXPECT_TRUE(_encoder->Encode(inputImage, NULL, &videoFrameType) ==
- WEBRTC_VIDEO_CODEC_OK);
- EXPECT_TRUE(WaitForEncodedFrame() > 0);
-
- EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
- _encoder->Encode(inputImage, NULL, &videoFrameType);
- frameLength = WaitForEncodedFrame();
- EXPECT_TRUE(frameLength > 0);
- EXPECT_TRUE(CheckIfBitExact(_refEncFrame, _refEncFrameLength,
- _encodedVideoBuffer.GetBuffer(), frameLength) == true);
-
- // Reset then encode.
- _encodedVideoBuffer.UpdateLength(0);
- _encodedVideoBuffer.Reset();
- EXPECT_TRUE(_encoder->Encode(inputImage, NULL, &videoFrameType) ==
- WEBRTC_VIDEO_CODEC_OK);
- WaitForEncodedFrame();
- EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
- _encoder->Encode(inputImage, NULL, &videoFrameType);
- frameLength = WaitForEncodedFrame();
- EXPECT_TRUE(frameLength > 0);
- EXPECT_TRUE(CheckIfBitExact(_refEncFrame, _refEncFrameLength,
- _encodedVideoBuffer.GetBuffer(), frameLength) == true);
-
- // Release then encode.
- _encodedVideoBuffer.UpdateLength(0);
- _encodedVideoBuffer.Reset();
- EXPECT_TRUE(_encoder->Encode(inputImage, NULL, &videoFrameType) ==
- WEBRTC_VIDEO_CODEC_OK);
- WaitForEncodedFrame();
- EXPECT_TRUE(_encoder->Release() == WEBRTC_VIDEO_CODEC_OK);
- EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
- _encoder->Encode(inputImage, NULL, &videoFrameType);
- frameLength = WaitForEncodedFrame();
- EXPECT_TRUE(frameLength > 0);
- EXPECT_TRUE(CheckIfBitExact(_refEncFrame, _refEncFrameLength,
- _encodedVideoBuffer.GetBuffer(), frameLength) == true);
-
- //----- Decoder parameter tests -----
-
- //-- Calls before InitDecode() --
- // We want to revert the initialization done in Setup().
- EXPECT_TRUE(_decoder->Release() == WEBRTC_VIDEO_CODEC_OK);
- VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
- EXPECT_TRUE(_decoder->Decode(encodedImage, false, NULL) ==
- WEBRTC_VIDEO_CODEC_UNINITIALIZED);
- WaitForDecodedFrame();
- EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_UNINITIALIZED);
- EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
-
- if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
- {
- exit(EXIT_FAILURE);
- }
-
- //-- Decode() errors --
- // Unallocated encodedVideoBuffer.
- _encodedVideoBuffer.Free();
- VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
- encodedImage._length = 10; // Buffer NULL but length > 0
- EXPECT_EQ(_decoder->Decode(encodedImage, false, NULL),
- WEBRTC_VIDEO_CODEC_ERR_PARAMETER);
- _encodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
-
- //----- Decoder stress tests -----
- unsigned char* tmpBuf = new unsigned char[_lengthSourceFrame];
-
- // "Random" and zero data.
- // We either expect an error, or at the least, no output.
- // This relies on the codec's ability to detect an erroneous bitstream.
- /*
- EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
- EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
- if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
- {
- exit(EXIT_FAILURE);
- }
- for (int i = 0; i < 100; i++)
- {
- ASSERT_TRUE(fread(tmpBuf, 1, _refEncFrameLength, _sourceFile)
- == _refEncFrameLength);
- _encodedVideoBuffer.CopyBuffer(_refEncFrameLength, tmpBuf);
- VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
- FillDecoderSpecificInfo(encodedImage);
- int ret = _decoder->Decode(encodedImage, false, _decoderSpecificInfo);
- EXPECT_TRUE(ret <= 0);
- if (ret == 0)
- {
- EXPECT_TRUE(WaitForDecodedFrame() == 0);
- }
-
- memset(tmpBuf, 0, _refEncFrameLength);
- _encodedVideoBuffer.CopyBuffer(_refEncFrameLength, tmpBuf);
- VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
- FillDecoderSpecificInfo(encodedImage);
- ret = _decoder->Decode(encodedImage, false, _decoderSpecificInfo);
- EXPECT_TRUE(ret <= 0);
- if (ret == 0)
- {
- EXPECT_TRUE(WaitForDecodedFrame() == 0);
- }
- }
- */
- rewind(_sourceFile);
-
- _encodedVideoBuffer.UpdateLength(_refEncFrameLength);
- _encodedVideoBuffer.CopyBuffer(_refEncFrameLength, _refEncFrame);
-
- // Init then decode.
- EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
- if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
- {
- exit(EXIT_FAILURE);
- }
- frameLength = 0;
- VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
- while (frameLength == 0)
- {
- _decoder->Decode(encodedImage, false, NULL);
- frameLength = WaitForDecodedFrame();
- }
- EXPECT_TRUE(CheckIfBitExact(_decodedVideoBuffer.GetBuffer(), frameLength,
- _refDecFrame, _lengthSourceFrame) == true);
-
- // Reset then decode.
- EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
- frameLength = 0;
- VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
- while (frameLength == 0)
- {
- _decoder->Decode(encodedImage, false, NULL);
- frameLength = WaitForDecodedFrame();
- }
- EXPECT_TRUE(CheckIfBitExact(_decodedVideoBuffer.GetBuffer(), frameLength,
- _refDecFrame, _lengthSourceFrame) == true);
-
- // Decode with other size, reset, then decode with original size again
- // to verify that decoder is reset to a "fresh" state upon Reset().
- {
- // assert that input frame size is a factor of two, so that we can use
- // quarter size below
- EXPECT_TRUE((_inst.width % 2 == 0) && (_inst.height % 2 == 0));
-
- VideoCodec tempInst;
- memcpy(&tempInst, &_inst, sizeof(VideoCodec));
- tempInst.width /= 2;
- tempInst.height /= 2;
-
- // Encode reduced (quarter) frame size
- EXPECT_TRUE(_encoder->Release() == WEBRTC_VIDEO_CODEC_OK);
- EXPECT_TRUE(_encoder->InitEncode(&tempInst, 1, 1440) ==
- WEBRTC_VIDEO_CODEC_OK);
- RawImage tempInput(inputImage._buffer, inputImage._length/4,
- inputImage._size/4);
- VideoFrameType videoFrameType = kDeltaFrame;
- _encoder->Encode(tempInput, NULL, &videoFrameType);
- frameLength = WaitForEncodedFrame();
- EXPECT_TRUE(frameLength > 0);
-
- // Reset then decode.
- EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
- frameLength = 0;
- VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
- while (frameLength == 0)
- {
- _decoder->Decode(encodedImage, false, NULL);
- frameLength = WaitForDecodedFrame();
- }
-
- // Encode original frame again
- EXPECT_TRUE(_encoder->Release() == WEBRTC_VIDEO_CODEC_OK);
- EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) ==
- WEBRTC_VIDEO_CODEC_OK);
- _encoder->Encode(inputImage, NULL, &videoFrameType);
- frameLength = WaitForEncodedFrame();
- EXPECT_TRUE(frameLength > 0);
-
- // Reset then decode original frame again.
- EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
- frameLength = 0;
- VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
- while (frameLength == 0)
- {
- _decoder->Decode(encodedImage, false, NULL);
- frameLength = WaitForDecodedFrame();
- }
-
- // check that decoded frame matches with reference
- EXPECT_TRUE(CheckIfBitExact(_decodedVideoBuffer.GetBuffer(), frameLength,
- _refDecFrame, _lengthSourceFrame) == true);
-
- }
-
- // Release then decode.
- EXPECT_TRUE(_decoder->Release() == WEBRTC_VIDEO_CODEC_OK);
- EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
- if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
- {
- exit(EXIT_FAILURE);
- }
- frameLength = 0;
- VideoEncodedBufferToEncodedImage(_encodedVideoBuffer, encodedImage);
- while (frameLength == 0)
- {
- _decoder->Decode(encodedImage, false, NULL);
- frameLength = WaitForDecodedFrame();
- }
- EXPECT_TRUE(CheckIfBitExact(_decodedVideoBuffer.GetBuffer(), frameLength,
- _refDecFrame, _lengthSourceFrame) == true);
- _encodedVideoBuffer.UpdateLength(0);
- _encodedVideoBuffer.Reset();
-
- delete [] tmpBuf;
-
- //----- Function tests -----
- int frames = 0;
- // Do not specify maxBitRate (as in ViE).
- _inst.maxBitrate = 0;
-
- //-- Timestamp propagation --
- EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
- EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
- EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
- if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
- {
- exit(EXIT_FAILURE);
- }
-
- printf("\nTimestamp propagation test...\n");
- frames = 0;
- int frameDelay = 0;
- int encTimeStamp;
- _decodedVideoBuffer.SetTimeStamp(0);
- while (fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile) ==
- _lengthSourceFrame)
- {
- _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
- _inputVideoBuffer.SetTimeStamp(frames);
- VideoBufferToRawImage(_inputVideoBuffer, inputImage);
- VideoFrameType videoFrameType = kDeltaFrame;
- ASSERT_TRUE(_encoder->Encode(inputImage,
- NULL,
- &videoFrameType) ==
- WEBRTC_VIDEO_CODEC_OK);
- frameLength = WaitForEncodedFrame();
- //ASSERT_TRUE(frameLength);
- EXPECT_TRUE(frameLength > 0);
- encTimeStamp = _encodedVideoBuffer.GetTimeStamp();
- EXPECT_TRUE(_inputVideoBuffer.GetTimeStamp() ==
- static_cast<unsigned>(encTimeStamp));
-
- frameLength = Decode();
- if (frameLength == 0)
- {
- frameDelay++;
- }
-
- encTimeStamp -= frameDelay;
- if (encTimeStamp < 0)
- {
- encTimeStamp = 0;
- }
- EXPECT_TRUE(_decodedVideoBuffer.GetTimeStamp() ==
- static_cast<unsigned>(encTimeStamp));
- frames++;
- }
- ASSERT_TRUE(feof(_sourceFile) != 0);
- rewind(_sourceFile);
-
- RateControlTests();
-
- Teardown();
-}
-
-void
-UnitTest::RateControlTests()
-{
- std::string outFileName;
- int frames = 0;
- RawImage inputImage;
- WebRtc_UWord32 frameLength;
-
- // Do not specify maxBitRate (as in ViE).
- _inst.maxBitrate = 0;
- //-- Verify rate control --
- EXPECT_TRUE(_encoder->InitEncode(&_inst, 1, 1440) == WEBRTC_VIDEO_CODEC_OK);
- EXPECT_TRUE(_decoder->Reset() == WEBRTC_VIDEO_CODEC_OK);
- EXPECT_TRUE(_decoder->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
- // add: should also be 0, and 1
- const int bitRate[] =
- {100, 200, 300, 400, 500, 600, 800, 1000, 2000, 3000, 4000, 10000};
- const int nBitrates = sizeof(bitRate)/sizeof(*bitRate);
-
- printf("\nRate control test\n");
- for (int i = 0; i < nBitrates; i++)
- {
- _bitRate = bitRate[i];
- int totalBytes = 0;
- _inst.startBitrate = _bitRate;
- _encoder->InitEncode(&_inst, 4, 1440);
- _decoder->Reset();
- _decoder->InitDecode(&_inst, 1);
- frames = 0;
-
- if (_bitRate > _inst.maxBitrate)
- {
- CodecSpecific_SetBitrate(_bitRate, _inst.maxFramerate);
- }
- else
- {
- CodecSpecific_SetBitrate(_bitRate, _inst.maxFramerate);
- }
-
- while (fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile) ==
- _lengthSourceFrame)
- {
- _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
- _inputVideoBuffer.SetTimeStamp(_inputVideoBuffer.GetTimeStamp() +
- static_cast<WebRtc_UWord32>(9e4 /
- static_cast<float>(_inst.maxFramerate)));
- VideoBufferToRawImage(_inputVideoBuffer, inputImage);
- VideoFrameType videoFrameType = kDeltaFrame;
- ASSERT_EQ(_encoder->Encode(inputImage, NULL, &videoFrameType),
- WEBRTC_VIDEO_CODEC_OK);
- frameLength = WaitForEncodedFrame();
- ASSERT_GE(frameLength, 0u);
- totalBytes += frameLength;
- frames++;
-
- _encodedVideoBuffer.UpdateLength(0);
- _encodedVideoBuffer.Reset();
- }
- WebRtc_UWord32 actualBitrate =
- (totalBytes / frames * _inst.maxFramerate * 8)/1000;
- printf("Target bitrate: %d kbps, actual bitrate: %d kbps\n", _bitRate,
- actualBitrate);
- // Test for close match over reasonable range.
- if (_bitRate >= 100 && _bitRate <= 2500)
- {
- EXPECT_TRUE(abs(WebRtc_Word32(actualBitrate - _bitRate)) <
- 0.1 * _bitRate); // for VP8
- }
- ASSERT_TRUE(feof(_sourceFile) != 0);
- rewind(_sourceFile);
- }
-}
-
-bool
-UnitTest::CheckIfBitExact(const void* ptrA, unsigned int aLengthBytes,
- const void* ptrB, unsigned int bLengthBytes)
-{
- if (aLengthBytes != bLengthBytes)
- {
- return false;
- }
-
- return memcmp(ptrA, ptrB, aLengthBytes) == 0;
-}
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/unit_test.h b/trunk/src/modules/video_coding/codecs/test_framework/unit_test.h
deleted file mode 100644
index 01b348d..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/unit_test.h
+++ /dev/null
@@ -1,111 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_UNIT_TEST_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_UNIT_TEST_H_
-
-#include "test.h"
-#include "event_wrapper.h"
-
-// Disable "conditional expression is constant" warnings on the perfectly
-// acceptable
-// do { ... } while (0) constructions below.
-// Refer to http://stackoverflow.com/questions/1946445/
-// is-there-better-way-to-write-do-while0-construct-to-avoid-compiler-warnings
-// for some discussion of the issue.
-#ifdef _WIN32
-#pragma warning(disable : 4127)
-#endif
-
-class VideoSource;
-class UnitTestEncodeCompleteCallback;
-class UnitTestDecodeCompleteCallback;
-
-class UnitTest : public Test
-{
-public:
- UnitTest();
- virtual ~UnitTest();
- virtual void Perform();
- virtual void Print();
-
-protected:
- UnitTest(std::string name, std::string description);
- virtual WebRtc_UWord32 CodecSpecific_SetBitrate(
- WebRtc_UWord32 bitRate,
- WebRtc_UWord32 /* frameRate */);
- virtual void Setup();
- virtual void Teardown();
- virtual void RateControlTests();
- virtual int Decode();
- virtual int DecodeWithoutAssert();
- virtual int SetCodecSpecificParameters() {return 0;};
-
- virtual bool CheckIfBitExact(const void *ptrA, unsigned int aLengthBytes,
- const void *ptrB, unsigned int bLengthBytes);
-
- WebRtc_UWord32 WaitForEncodedFrame() const;
- WebRtc_UWord32 WaitForDecodedFrame() const;
-
- int _tests;
- int _errors;
-
- VideoSource* _source;
- unsigned char* _refFrame;
- unsigned char* _refEncFrame;
- unsigned char* _refDecFrame;
- int _refEncFrameLength;
- FILE* _sourceFile;
-
- UnitTestEncodeCompleteCallback* _encodeCompleteCallback;
- UnitTestDecodeCompleteCallback* _decodeCompleteCallback;
- enum { kMaxWaitEncTimeMs = 100 };
- enum { kMaxWaitDecTimeMs = 25 };
-};
-
-class UnitTestEncodeCompleteCallback : public webrtc::EncodedImageCallback
-{
-public:
- UnitTestEncodeCompleteCallback(TestVideoEncodedBuffer* buffer,
- WebRtc_UWord32 decoderSpecificSize = 0,
- void* decoderSpecificInfo = NULL) :
- _encodedVideoBuffer(buffer),
- _decoderSpecificInfo(decoderSpecificInfo),
- _decoderSpecificSize(decoderSpecificSize),
- _encodeComplete(false) {}
- WebRtc_Word32 Encoded(webrtc::EncodedImage& encodedImage,
- const webrtc::CodecSpecificInfo* codecSpecificInfo,
- const webrtc::RTPFragmentationHeader*
- fragmentation = NULL);
- bool EncodeComplete();
- // Note that this only makes sense if an encode has been completed
- webrtc::VideoFrameType EncodedFrameType() const;
-private:
- TestVideoEncodedBuffer* _encodedVideoBuffer;
- void* _decoderSpecificInfo;
- WebRtc_UWord32 _decoderSpecificSize;
- bool _encodeComplete;
- webrtc::VideoFrameType _encodedFrameType;
-};
-
-class UnitTestDecodeCompleteCallback : public webrtc::DecodedImageCallback
-{
-public:
- UnitTestDecodeCompleteCallback(TestVideoBuffer* buffer) :
- _decodedVideoBuffer(buffer), _decodeComplete(false) {}
- WebRtc_Word32 Decoded(webrtc::RawImage& image);
- bool DecodeComplete();
-private:
- TestVideoBuffer* _decodedVideoBuffer;
- bool _decodeComplete;
-};
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_UNIT_TEST_H_
-
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/video_buffer.cc b/trunk/src/modules/video_coding/codecs/test_framework/video_buffer.cc
deleted file mode 100644
index 3958e90..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/video_buffer.cc
+++ /dev/null
@@ -1,319 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <assert.h>
-#include <string.h>
-#include "video_buffer.h"
-
-using namespace webrtc;
-
-TestVideoBuffer::TestVideoBuffer():
-_buffer(0),
-_bufferSize(0),
-_bufferLength(0),
-_startOffset(0),
-_timeStamp(0),
-_width(0),
-_height(0)
-{
- //
-}
-
-
-TestVideoBuffer::~TestVideoBuffer()
-{
- _timeStamp = 0;
- _startOffset = 0;
- _bufferLength = 0;
- _bufferSize = 0;
-
- if(_buffer)
- {
- delete [] _buffer;
- _buffer = 0;
- }
-}
-
-TestVideoBuffer::TestVideoBuffer(const TestVideoBuffer& rhs)
-:
-_buffer(0),
-_bufferSize(rhs._bufferSize),
-_bufferLength(rhs._bufferLength),
-_startOffset(rhs._startOffset),
-_timeStamp(rhs._timeStamp),
-_width(rhs._width),
-_height(rhs._height)
-{
- // make sure that our buffer is big enough
- _buffer = new unsigned char[_bufferSize];
-
- // only copy required length
- memcpy(_buffer + _startOffset, rhs._buffer, _bufferLength); // GetBuffer() includes _startOffset
-}
-
-void TestVideoBuffer::SetTimeStamp(unsigned int timeStamp)
-{
- _timeStamp = timeStamp;
-}
-
-unsigned int
-TestVideoBuffer::GetWidth() const
-{
- return _width;
-}
-
-unsigned int
-TestVideoBuffer::GetHeight() const
-{
- return _height;
-}
-
-void
-TestVideoBuffer::SetWidth(unsigned int width)
-{
- _width = width;
-}
-
-void
-TestVideoBuffer::SetHeight(unsigned int height)
-{
- _height = height;
-}
-
-
-void TestVideoBuffer::Free()
-{
- _timeStamp = 0;
- _startOffset = 0;
- _bufferLength = 0;
- _bufferSize = 0;
- _height = 0;
- _width = 0;
-
- if(_buffer)
- {
- delete [] _buffer;
- _buffer = 0;
- }
-}
-
-void TestVideoBuffer::VerifyAndAllocate(unsigned int minimumSize)
-{
- if(minimumSize > _bufferSize)
- {
- // make sure that our buffer is big enough
- unsigned char * newBufferBuffer = new unsigned char[minimumSize];
- if(_buffer)
- {
- // copy the old data
- memcpy(newBufferBuffer, _buffer, _bufferSize);
- delete [] _buffer;
- }
- _buffer = newBufferBuffer;
- _bufferSize = minimumSize;
- }
-}
-
-int TestVideoBuffer::SetOffset(unsigned int length)
-{
- if (length > _bufferSize ||
- length > _bufferLength)
- {
- return -1;
- }
-
- unsigned int oldOffset = _startOffset;
-
- if(oldOffset > length)
- {
- unsigned int newLength = _bufferLength + (oldOffset-length);// increase by the diff
- assert(newLength <= _bufferSize);
- _bufferLength = newLength;
- }
- if(oldOffset < length)
- {
- if(_bufferLength > (length-oldOffset))
- {
- _bufferLength -= (length-oldOffset); // decrease by the diff
- }
- }
- _startOffset = length; // update
-
- return 0;
-}
-
-void TestVideoBuffer::UpdateLength(unsigned int newLength)
-{
- assert(newLength +_startOffset <= _bufferSize);
- _bufferLength = newLength;
-}
-
-void TestVideoBuffer::CopyBuffer(unsigned int length, const unsigned char* buffer)
-{
- assert(length+_startOffset <= _bufferSize);
- memcpy(_buffer+_startOffset, buffer, length);
- _bufferLength = length;
-}
-
-void TestVideoBuffer::CopyBuffer(TestVideoBuffer& fromVideoBuffer)
-{
- assert(fromVideoBuffer.GetLength() + fromVideoBuffer.GetStartOffset() <= _bufferSize);
- assert(fromVideoBuffer.GetSize() <= _bufferSize);
-
- _bufferLength = fromVideoBuffer.GetLength();
- _startOffset = fromVideoBuffer.GetStartOffset();
- _timeStamp = fromVideoBuffer.GetTimeStamp();
- _height = fromVideoBuffer.GetHeight();
- _width = fromVideoBuffer.GetWidth();
-
- // only copy required length
- memcpy(_buffer+_startOffset, fromVideoBuffer.GetBuffer(), fromVideoBuffer.GetLength()); // GetBuffer() includes _startOffset
-
-}
-
-void TestVideoBuffer::CopyPointer(const TestVideoBuffer& fromVideoBuffer)
-{
- _bufferSize = fromVideoBuffer.GetSize();
- _bufferLength = fromVideoBuffer.GetLength();
- _startOffset = fromVideoBuffer.GetStartOffset();
- _timeStamp = fromVideoBuffer.GetTimeStamp();
- _height = fromVideoBuffer.GetHeight();
- _width = fromVideoBuffer.GetWidth();
-
- _buffer = fromVideoBuffer.GetBuffer();
-}
-
-void TestVideoBuffer::ClearPointer()
-{
- _buffer = NULL;
-}
-
-void TestVideoBuffer::SwapBuffers(TestVideoBuffer& videoBuffer)
-{
- unsigned char* tempBuffer = _buffer;
- unsigned int tempSize = _bufferSize;
- unsigned int tempLength =_bufferLength;
- unsigned int tempOffset = _startOffset;
- unsigned int tempTime = _timeStamp;
- unsigned int tempHeight = _height;
- unsigned int tempWidth = _width;
-
- _buffer = videoBuffer.GetBuffer();
- _bufferSize = videoBuffer.GetSize();
- _bufferLength = videoBuffer.GetLength();
- _startOffset = videoBuffer.GetStartOffset();
- _timeStamp = videoBuffer.GetTimeStamp();
- _height = videoBuffer.GetHeight();
- _width = videoBuffer.GetWidth();
-
-
- videoBuffer.Set(tempBuffer, tempSize, tempLength, tempOffset, tempTime);
- videoBuffer.SetHeight(tempHeight);
- videoBuffer.SetWidth(tempWidth);
-}
-
-void TestVideoBuffer::Set(unsigned char* tempBuffer,unsigned int tempSize,unsigned int tempLength, unsigned int tempOffset,unsigned int timeStamp)
-{
- _buffer = tempBuffer;
- _bufferSize = tempSize;
- _bufferLength = tempLength;
- _startOffset = tempOffset;
- _timeStamp = timeStamp;
-}
-
-unsigned char* TestVideoBuffer::GetBuffer() const
-{
- return _buffer+_startOffset;
-}
-
-unsigned int TestVideoBuffer::GetStartOffset() const
-{
- return _startOffset;
-}
-
-unsigned int TestVideoBuffer::GetSize() const
-{
- return _bufferSize;
-}
-
-unsigned int TestVideoBuffer::GetLength() const
-{
- return _bufferLength;
-}
-
-unsigned int TestVideoBuffer::GetTimeStamp() const
-{
- return _timeStamp;
-}
-
-/**
-* TestVideoEncodedBuffer
-*
-*/
-
-TestVideoEncodedBuffer::TestVideoEncodedBuffer() :
- _captureWidth(0),
- _captureHeight(0),
- _frameRate(-1)
-{
- _frameType = kDeltaFrame;
-}
-
-TestVideoEncodedBuffer::~TestVideoEncodedBuffer()
-{
-}
-
-void TestVideoEncodedBuffer::SetCaptureWidth(unsigned short width)
-{
- _captureWidth = width;
-}
-
-void TestVideoEncodedBuffer::SetCaptureHeight(unsigned short height)
-{
- _captureHeight = height;
-}
-
-unsigned short TestVideoEncodedBuffer::GetCaptureWidth()
-{
- return _captureWidth;
-}
-
-unsigned short TestVideoEncodedBuffer::GetCaptureHeight()
-{
- return _captureHeight;
-}
-
-VideoFrameType TestVideoEncodedBuffer::GetFrameType()
-{
- return _frameType;
-}
-
-void TestVideoEncodedBuffer::SetFrameType(VideoFrameType frametype)
-{
- _frameType = frametype;
-}
-
-void TestVideoEncodedBuffer::Reset()
-{
- _captureWidth = 0;
- _captureHeight = 0;
- _frameRate = -1;
- _frameType = kDeltaFrame;
-}
-
-void TestVideoEncodedBuffer::SetFrameRate(float frameRate)
-{
- _frameRate = frameRate;
-}
-
-float TestVideoEncodedBuffer::GetFrameRate()
-{
- return _frameRate;
-}
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/video_buffer.h b/trunk/src/modules/video_coding/codecs/test_framework/video_buffer.h
deleted file mode 100644
index 824440e..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/video_buffer.h
+++ /dev/null
@@ -1,122 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_BUFFER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_BUFFER_H_
-
-#include "typedefs.h"
-#include "video_image.h"
-
-class TestVideoBuffer
-{
-public:
- TestVideoBuffer();
-
- virtual ~TestVideoBuffer();
-
- TestVideoBuffer(const TestVideoBuffer& rhs);
-
- /**
- * Verifies that current allocated buffer size is larger than or equal to the input size.
- * If the current buffer size is smaller, a new allocation is made and the old buffer data is copied to the new buffer.
- */
- void VerifyAndAllocate(unsigned int minimumSize);
-
- void UpdateLength(unsigned int newLength);
-
- void SwapBuffers(TestVideoBuffer& videoBuffer);
-
- void CopyBuffer(unsigned int length, const unsigned char* fromBuffer);
-
- void CopyBuffer(TestVideoBuffer& fromVideoBuffer);
-
- // Use with care, and remember to call ClearPointer() when done.
- void CopyPointer(const TestVideoBuffer& fromVideoBuffer);
-
- void ClearPointer();
-
- int SetOffset(unsigned int length); // Sets offset to beginning of frame in buffer
-
- void Free(); // Deletes frame buffer and resets members to zero
-
- void SetTimeStamp(unsigned int timeStamp); // Sets timestamp of frame (90kHz)
-
- /**
- * Gets pointer to frame buffer
- */
- unsigned char* GetBuffer() const;
-
- /**
- * Gets allocated buffer size
- */
- unsigned int GetSize() const;
-
- /**
- * Gets length of frame
- */
- unsigned int GetLength() const;
-
- /**
- * Gets timestamp of frame (90kHz)
- */
- unsigned int GetTimeStamp() const;
-
- unsigned int GetWidth() const;
- unsigned int GetHeight() const;
-
- void SetWidth(unsigned int width);
- void SetHeight(unsigned int height);
-
-private:
- TestVideoBuffer& operator=(const TestVideoBuffer& inBuffer);
-
-private:
- void Set(unsigned char* buffer,unsigned int size,unsigned int length,unsigned int offset, unsigned int timeStamp);
- unsigned int GetStartOffset() const;
-
- unsigned char* _buffer; // Pointer to frame buffer
- unsigned int _bufferSize; // Allocated buffer size
- unsigned int _bufferLength; // Length (in bytes) of frame
- unsigned int _startOffset; // Offset (in bytes) to beginning of frame in buffer
- unsigned int _timeStamp; // Timestamp of frame (90kHz)
- unsigned int _width;
- unsigned int _height;
-};
-
-class TestVideoEncodedBuffer: public TestVideoBuffer
-{
-public:
- TestVideoEncodedBuffer();
- ~TestVideoEncodedBuffer();
-
- void SetCaptureWidth(unsigned short width);
- void SetCaptureHeight(unsigned short height);
- unsigned short GetCaptureWidth();
- unsigned short GetCaptureHeight();
-
- webrtc::VideoFrameType GetFrameType();
- void SetFrameType(webrtc::VideoFrameType frametype);
-
- void Reset();
-
- void SetFrameRate(float frameRate);
- float GetFrameRate();
-
-private:
- TestVideoEncodedBuffer& operator=(const TestVideoEncodedBuffer& inBuffer);
-
-private:
- unsigned short _captureWidth;
- unsigned short _captureHeight;
- webrtc::VideoFrameType _frameType;
- float _frameRate;
-};
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_BUFFER_H_
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/video_source.cc b/trunk/src/modules/video_coding/codecs/test_framework/video_source.cc
deleted file mode 100644
index 8abad1b..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/video_source.cc
+++ /dev/null
@@ -1,419 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_source.h"
-
-#include <stdio.h>
-#include <cassert>
-
-#include "testsupport/fileutils.h"
-
-VideoSource::VideoSource()
-:
-_fileName(webrtc::test::ProjectRootPath() + "resources/foreman_cif.yuv"),
-_width(352),
-_height(288),
-_type(webrtc::kI420),
-_frameRate(30)
-{
-}
-
-VideoSource::VideoSource(std::string fileName, VideoSize size,
- int frameRate /*= 30*/, webrtc::VideoType type /*= webrtc::kI420*/)
-:
-_fileName(fileName),
-_type(type),
-_frameRate(frameRate)
-{
- assert(size != kUndefined && size != kNumberOfVideoSizes);
- assert(type != webrtc::kUnknown);
- assert(frameRate > 0);
- assert(GetWidthHeight(size, _width, _height) == 0);
-}
-
-VideoSource::VideoSource(std::string fileName, int width, int height,
- int frameRate /*= 30*/, webrtc::VideoType type /*= webrtc::kI420*/)
-:
-_fileName(fileName),
-_width(width),
-_height(height),
-_type(type),
-_frameRate(frameRate)
-{
- assert(width > 0);
- assert(height > 0);
- assert(type != webrtc::kUnknown);
- assert(frameRate > 0);
-}
-
-VideoSize
-VideoSource::GetSize() const
-{
- return GetSize(_width, _height);
-}
-
-VideoSize
-VideoSource::GetSize(WebRtc_UWord16 width, WebRtc_UWord16 height)
-{
- if(width == 128 && height == 96)
- {
- return kSQCIF;
- }else if(width == 160 && height == 120)
- {
- return kQQVGA;
- }else if(width == 176 && height == 144)
- {
- return kQCIF;
- }else if(width == 320 && height == 240)
- {
- return kQVGA;
- }else if(width == 352 && height == 288)
- {
- return kCIF;
- }else if(width == 640 && height == 480)
- {
- return kVGA;
- }else if(width == 720 && height == 480)
- {
- return kNTSC;
- }else if(width == 704 && height == 576)
- {
- return k4CIF;
- }else if(width == 800 && height == 600)
- {
- return kSVGA;
- }else if(width == 960 && height == 720)
- {
- return kHD;
- }else if(width == 1024 && height == 768)
- {
- return kXGA;
- }else if(width == 1440 && height == 1080)
- {
- return kFullHD;
- }else if(width == 400 && height == 240)
- {
- return kWQVGA;
- }else if(width == 800 && height == 480)
- {
- return kWVGA;
- }else if(width == 1280 && height == 720)
- {
- return kWHD;
- }else if(width == 1920 && height == 1080)
- {
- return kWFullHD;
- }
- return kUndefined;
-}
-
-unsigned int
-VideoSource::GetFrameLength() const
-{
- return webrtc::CalcBufferSize(_type, _width, _height);
-}
-
-const char*
-VideoSource::GetMySizeString() const
-{
- return VideoSource::GetSizeString(GetSize());
-}
-
-const char*
-VideoSource::GetSizeString(VideoSize size)
-{
- switch (size)
- {
- case kSQCIF:
- return "SQCIF";
- case kQQVGA:
- return "QQVGA";
- case kQCIF:
- return "QCIF";
- case kQVGA:
- return "QVGA";
- case kCIF:
- return "CIF";
- case kVGA:
- return "VGA";
- case kNTSC:
- return "NTSC";
- case k4CIF:
- return "4CIF";
- case kSVGA:
- return "SVGA";
- case kHD:
- return "HD";
- case kXGA:
- return "XGA";
- case kFullHD:
- return "Full_HD";
- case kWQVGA:
- return "WQVGA";
- case kWHD:
- return "WHD";
- case kWFullHD:
- return "WFull_HD";
- default:
- return "Undefined";
- }
-}
-
-std::string
-VideoSource::GetFilePath() const
-{
- size_t slashPos = _fileName.find_last_of("/\\");
- if (slashPos == std::string::npos)
- {
- return ".";
- }
-
- return _fileName.substr(0, slashPos);
-}
-
-std::string
-VideoSource::GetName() const
-{
- // Remove path.
- size_t slashPos = _fileName.find_last_of("/\\");
- if (slashPos == std::string::npos)
- {
- slashPos = 0;
- }
- else
- {
- slashPos++;
- }
-
- // Remove extension and underscored suffix if it exists.
- return _fileName.substr(slashPos, std::min(_fileName.find_last_of("_"),
- _fileName.find_last_of(".")) - slashPos);
-}
-
-void
-VideoSource::Convert(const VideoSource &target, bool force /* = false */) const
-{
- // Ensure target rate is less than or equal to source
- // (i.e. we are only temporally downsampling).
- assert(target.GetFrameRate() <= _frameRate);
- // Only supports YUV420 currently.
- assert(_type == webrtc::kI420 && target.GetType() == webrtc::kI420);
- if (!force && (FileExists(target.GetFileName().c_str()) ||
- (target.GetWidth() == _width && target.GetHeight() == _height && target.GetFrameRate() == _frameRate)))
- {
- // Assume that the filename uniquely defines the content.
- // If the file already exists, it is the correct file.
- return;
- }
- FILE *inFile = NULL;
- FILE *outFile = NULL;
-
- inFile = fopen(_fileName.c_str(), "rb");
- assert(inFile != NULL);
-
- outFile = fopen(target.GetFileName().c_str(), "wb");
- assert(outFile != NULL);
-
- FrameDropper fd;
- fd.SetFrameRate(target.GetFrameRate(), _frameRate);
-
- const size_t lengthOutFrame = webrtc::CalcBufferSize(target.GetType(),
- target.GetWidth(), target.GetHeight());
- assert(lengthOutFrame > 0);
- unsigned char *outFrame = new unsigned char[lengthOutFrame];
-
- const size_t lengthInFrame = webrtc::CalcBufferSize(_type, _width, _height);
- assert(lengthInFrame > 0);
- unsigned char *inFrame = new unsigned char[lengthInFrame];
-
- while (fread(inFrame, 1, lengthInFrame, inFile) == lengthInFrame)
- {
- if (!fd.DropFrame())
- {
- assert(target.GetWidth() == _width &&
- target.GetHeight() == _height); // Add video interpolator here!
- fwrite(outFrame, 1, lengthOutFrame, outFile);
- }
- }
-
- delete inFrame;
- delete outFrame;
- fclose(inFile);
- fclose(outFile);
-}
-
-bool VideoSource::FileExists(const char* fileName)
-{
- FILE* fp = NULL;
- fp = fopen(fileName, "rb");
- if(fp != NULL)
- {
- fclose(fp);
- return true;
- }
- return false;
-}
-
-
-int
-VideoSource::GetWidthHeight( VideoSize size, int & width, int& height)
-{
- switch(size)
- {
- case kSQCIF:
- width = 128;
- height = 96;
- return 0;
- case kQQVGA:
- width = 160;
- height = 120;
- return 0;
- case kQCIF:
- width = 176;
- height = 144;
- return 0;
- case kCGA:
- width = 320;
- height = 200;
- return 0;
- case kQVGA:
- width = 320;
- height = 240;
- return 0;
- case kSIF:
- width = 352;
- height = 240;
- return 0;
- case kWQVGA:
- width = 400;
- height = 240;
- return 0;
- case kCIF:
- width = 352;
- height = 288;
- return 0;
- case kW288p:
- width = 512;
- height = 288;
- return 0;
- case k448p:
- width = 576;
- height = 448;
- return 0;
- case kVGA:
- width = 640;
- height = 480;
- return 0;
- case k432p:
- width = 720;
- height = 432;
- return 0;
- case kW432p:
- width = 768;
- height = 432;
- return 0;
- case k4SIF:
- width = 704;
- height = 480;
- return 0;
- case kW448p:
- width = 768;
- height = 448;
- return 0;
- case kNTSC:
- width = 720;
- height = 480;
- return 0;
- case kFW448p:
- width = 800;
- height = 448;
- return 0;
- case kWVGA:
- width = 800;
- height = 480;
- return 0;
- case k4CIF:
- width = 704;
- height = 576;
- return 0;
- case kSVGA:
- width = 800;
- height = 600;
- return 0;
- case kW544p:
- width = 960;
- height = 544;
- return 0;
- case kW576p:
- width = 1024;
- height = 576;
- return 0;
- case kHD:
- width = 960;
- height = 720;
- return 0;
- case kXGA:
- width = 1024;
- height = 768;
- return 0;
- case kFullHD:
- width = 1440;
- height = 1080;
- return 0;
- case kWHD:
- width = 1280;
- height = 720;
- return 0;
- case kWFullHD:
- width = 1920;
- height = 1080;
- return 0;
- default:
- return -1;
- }
-}
-
-FrameDropper::FrameDropper()
-:
-_dropsBetweenRenders(0),
-_frameCounter(0)
-{
-}
-
-bool
-FrameDropper::DropFrame()
-{
- _frameCounter++;
- if (_frameCounter > _dropsBetweenRenders)
- {
- _frameCounter = 0;
- return false;
- }
- return true;
-}
-
-unsigned int
-FrameDropper::DropsBetweenRenders()
-{
- return _dropsBetweenRenders;
-}
-
-void
-FrameDropper::SetFrameRate(double frameRate, double maxFrameRate)
-{
- if (frameRate >= 1.0)
- {
- _dropsBetweenRenders = static_cast<unsigned int>(maxFrameRate / frameRate + 0.5) - 1;
- }
- else
- {
- _dropsBetweenRenders = 0;
- }
-}
diff --git a/trunk/src/modules/video_coding/codecs/test_framework/video_source.h b/trunk/src/modules/video_coding/codecs/test_framework/video_source.h
deleted file mode 100644
index 4f785ba..0000000
--- a/trunk/src/modules/video_coding/codecs/test_framework/video_source.h
+++ /dev/null
@@ -1,109 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_SOURCE_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_SOURCE_H_
-
-#include <string>
-#include "common_video/libyuv/include/libyuv.h"
-
-enum VideoSize
- {
- kUndefined,
- kSQCIF, // 128*96 = 12 288
- kQQVGA, // 160*120 = 19 200
- kQCIF, // 176*144 = 25 344
- kCGA, // 320*200 = 64 000
- kQVGA, // 320*240 = 76 800
- kSIF, // 352*240 = 84 480
- kWQVGA, // 400*240 = 96 000
- kCIF, // 352*288 = 101 376
- kW288p, // 512*288 = 147 456 (WCIF)
- k448p, // 576*448 = 281 088
- kVGA, // 640*480 = 307 200
- k432p, // 720*432 = 311 040
- kW432p, // 768*432 = 331 776
- k4SIF, // 704*480 = 337 920
- kW448p, // 768*448 = 344 064
- kNTSC, // 720*480 = 345 600
- kFW448p, // 800*448 = 358 400
- kWVGA, // 800*480 = 384 000
- k4CIF, // 704�576 = 405 504
- kSVGA, // 800*600 = 480 000
- kW544p, // 960*544 = 522 240
- kW576p, // 1024*576 = 589 824 (W4CIF)
- kHD, // 960*720 = 691 200
- kXGA, // 1024*768 = 786 432
- kWHD, // 1280*720 = 921 600
- kFullHD, // 1440*1080 = 1 555 200
- kWFullHD, // 1920*1080 = 2 073 600
-
- kNumberOfVideoSizes
- };
-
-class VideoSource
-{
-public:
- VideoSource();
- VideoSource(std::string fileName, VideoSize size, int frameRate = 30,
- webrtc::VideoType type = webrtc::kI420);
- VideoSource(std::string fileName, int width, int height, int frameRate = 30,
- webrtc::VideoType type = webrtc::kI420);
-
- std::string GetFileName() const { return _fileName; }
- int GetWidth() const { return _width; }
- int GetHeight() const { return _height; }
- webrtc::VideoType GetType() const { return _type; }
- int GetFrameRate() const { return _frameRate; }
-
- // Returns the file path without a trailing slash.
- std::string GetFilePath() const;
-
- // Returns the filename with the path (including the leading slash) removed.
- std::string GetName() const;
-
- VideoSize GetSize() const;
- static VideoSize GetSize(WebRtc_UWord16 width, WebRtc_UWord16 height);
- unsigned int GetFrameLength() const;
-
- // Returns a human-readable size string.
- static const char* GetSizeString(VideoSize size);
- const char* GetMySizeString() const;
-
- // Opens the video source, converting and writing to the specified target.
- // If force is true, the conversion will be done even if the target file
- // already exists.
- void Convert(const VideoSource& target, bool force = false) const;
- static bool FileExists(const char* fileName);
-private:
- static int GetWidthHeight( VideoSize size, int& width, int& height);
- std::string _fileName;
- int _width;
- int _height;
- webrtc::VideoType _type;
- int _frameRate;
-};
-
-class FrameDropper
-{
-public:
- FrameDropper();
- bool DropFrame();
- unsigned int DropsBetweenRenders();
- void SetFrameRate(double frameRate, double maxFrameRate);
-
-private:
- unsigned int _dropsBetweenRenders;
- unsigned int _frameCounter;
-};
-
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_TEST_FRAMEWORK_VIDEO_SOURCE_H_
-
diff --git a/trunk/src/modules/video_coding/codecs/tools/video_codecs_tools.gypi b/trunk/src/modules/video_coding/codecs/tools/video_codecs_tools.gypi
deleted file mode 100644
index f43f285..0000000
--- a/trunk/src/modules/video_coding/codecs/tools/video_codecs_tools.gypi
+++ /dev/null
@@ -1,37 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- # Exclude the test target when building with chromium.
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'video_quality_measurement',
- 'type': 'executable',
- 'dependencies': [
- 'video_codecs_test_framework',
- 'webrtc_video_coding',
- 'webrtc_vp8',
- '<(webrtc_root)/../test/metrics.gyp:metrics',
- '<(webrtc_root)/../third_party/google-gflags/google-gflags.gyp:google-gflags',
- ],
- 'sources': [
- 'video_quality_measurement.cc',
- ],
- },
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
\ No newline at end of file
diff --git a/trunk/src/modules/video_coding/codecs/tools/video_quality_measurement.cc b/trunk/src/modules/video_coding/codecs/tools/video_quality_measurement.cc
deleted file mode 100644
index 953e3b4..0000000
--- a/trunk/src/modules/video_coding/codecs/tools/video_quality_measurement.cc
+++ /dev/null
@@ -1,526 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdarg.h>
-#include <sys/stat.h> // To check for directory existence.
-
-#include <cassert>
-#include <cstdio>
-#include <ctime>
-
-#ifndef S_ISDIR // Not defined in stat.h on Windows.
-#define S_ISDIR(mode) (((mode) & S_IFMT) == S_IFDIR)
-#endif
-
-#include "common_types.h"
-#include "google/gflags.h"
-#include "modules/video_coding/codecs/test/packet_manipulator.h"
-#include "modules/video_coding/codecs/test/stats.h"
-#include "modules/video_coding/codecs/test/videoprocessor.h"
-#include "modules/video_coding/codecs/vp8/main/interface/vp8.h"
-#include "modules/video_coding/main/interface/video_coding.h"
-#include "system_wrappers/interface/trace.h"
-#include "testsupport/frame_reader.h"
-#include "testsupport/frame_writer.h"
-#include "testsupport/metrics/video_metrics.h"
-#include "testsupport/packet_reader.h"
-
-DEFINE_string(test_name, "Quality test", "The name of the test to run. ");
-DEFINE_string(test_description, "", "A more detailed description about what "
- "the current test is about.");
-DEFINE_string(input_filename, "", "Input file. "
- "The source video file to be encoded and decoded. Must be in "
- ".yuv format");
-DEFINE_int32(width, -1, "Width in pixels of the frames in the input file.");
-DEFINE_int32(height, -1, "Height in pixels of the frames in the input file.");
-DEFINE_int32(framerate, 30, "Frame rate of the input file, in FPS "
- "(frames-per-second). ");
-DEFINE_string(output_dir, ".", "Output directory. "
- "The directory where the output file will be put. Must already "
- "exist.");
-DEFINE_bool(use_single_core, false, "Force using a single core. If set to "
- "true, only one core will be used for processing. Using a single "
- "core is necessary to get a deterministic behavior for the"
- "encoded frames - using multiple cores will produce different "
- "encoded frames since multiple cores are competing to consume the "
- "byte budget for each frame in parallel. If set to false, "
- "the maximum detected number of cores will be used. ");
-DEFINE_bool(disable_fixed_random_seed , false, "Set this flag to disable the"
- "usage of a fixed random seed for the random generator used "
- "for packet loss. Disabling this will cause consecutive runs "
- "loose packets at different locations, which is bad for "
- "reproducibility.");
-DEFINE_string(output_filename, "", "Output file. "
- "The name of the output video file resulting of the processing "
- "of the source file. By default this is the same name as the "
- "input file with '_out' appended before the extension.");
-DEFINE_int32(bitrate, 500, "Bit rate in kilobits/second.");
-DEFINE_int32(keyframe_interval, 0, "Forces a keyframe every Nth frame. "
- "0 means the encoder decides when to insert keyframes. Note that "
- "the encoder may create a keyframe in other locations in addition "
- "to the interval that is set using this parameter.");
-DEFINE_int32(temporal_layers, 0, "The number of temporal layers to use "
- "(VP8 specific codec setting). Must be 0-4.");
-DEFINE_int32(packet_size, 1500, "Simulated network packet size in bytes (MTU). "
- "Used for packet loss simulation.");
-DEFINE_int32(max_payload_size, 1440, "Max payload size in bytes for the "
- "encoder.");
-DEFINE_string(packet_loss_mode, "uniform", "Packet loss mode. Two different "
- "packet loss models are supported: uniform or burst. This "
- "setting has no effect unless packet_loss_rate is >0. ");
-DEFINE_double(packet_loss_probability, 0.0, "Packet loss probability. A value "
- "between 0.0 and 1.0 that defines the probability of a packet "
- "being lost. 0.1 means 10% and so on.");
-DEFINE_int32(packet_loss_burst_length, 1, "Packet loss burst length. Defines "
- "how many packets will be lost in a burst when a packet has been "
- "decided to be lost. Must be >=1.");
-DEFINE_bool(csv, false, "CSV output. Enabling this will output all frame "
- "statistics at the end of execution. Recommended to run combined "
- "with --noverbose to avoid mixing output.");
-DEFINE_bool(python, false, "Python output. Enabling this will output all frame "
- "statistics as a Python script at the end of execution. "
- "Recommended to run combine with --noverbose to avoid mixing "
- "output.");
-DEFINE_bool(verbose, true, "Verbose mode. Prints a lot of debugging info. "
- "Suitable for tracking progress but not for capturing output. "
- "Disable with --noverbose flag.");
-
-// Custom log method that only prints if the verbose flag is given.
-// Supports all the standard printf parameters and formatting (just forwarded).
-int Log(const char *format, ...) {
- int result = 0;
- if (FLAGS_verbose) {
- va_list args;
- va_start(args, format);
- result = vprintf(format, args);
- va_end(args);
- }
- return result;
-}
-
-// Validates the arguments given as command line flags and fills in the
-// TestConfig struct with all configurations needed for video processing.
-// Returns 0 if everything is OK, otherwise an exit code.
-int HandleCommandLineFlags(webrtc::test::TestConfig* config) {
- // Validate the mandatory flags:
- if (FLAGS_input_filename == "" || FLAGS_width == -1 || FLAGS_height == -1) {
- printf("%s\n", google::ProgramUsage());
- return 1;
- }
- config->name = FLAGS_test_name;
- config->description = FLAGS_test_description;
-
- // Verify the input file exists and is readable.
- FILE* test_file;
- test_file = fopen(FLAGS_input_filename.c_str(), "rb");
- if (test_file == NULL) {
- fprintf(stderr, "Cannot read the specified input file: %s\n",
- FLAGS_input_filename.c_str());
- return 2;
- }
- fclose(test_file);
- config->input_filename = FLAGS_input_filename;
-
- // Verify the output dir exists.
- struct stat dir_info;
- if (!(stat(FLAGS_output_dir.c_str(), &dir_info) == 0 &&
- S_ISDIR(dir_info.st_mode))) {
- fprintf(stderr, "Cannot find output directory: %s\n",
- FLAGS_output_dir.c_str());
- return 3;
- }
- config->output_dir = FLAGS_output_dir;
-
- // Manufacture an output filename if none was given.
- if (FLAGS_output_filename == "") {
- // Cut out the filename without extension from the given input file
- // (which may include a path)
- int startIndex = FLAGS_input_filename.find_last_of("/") + 1;
- if (startIndex == 0) {
- startIndex = 0;
- }
- FLAGS_output_filename =
- FLAGS_input_filename.substr(startIndex,
- FLAGS_input_filename.find_last_of(".")
- - startIndex) + "_out.yuv";
- }
-
- // Verify output file can be written.
- if (FLAGS_output_dir == ".") {
- config->output_filename = FLAGS_output_filename;
- } else {
- config->output_filename = FLAGS_output_dir + "/"+ FLAGS_output_filename;
- }
- test_file = fopen(config->output_filename.c_str(), "wb");
- if (test_file == NULL) {
- fprintf(stderr, "Cannot write output file: %s\n",
- config->output_filename.c_str());
- return 4;
- }
- fclose(test_file);
-
- // Check single core flag.
- config->use_single_core = FLAGS_use_single_core;
-
- // Get codec specific configuration.
- webrtc::VideoCodingModule::Codec(webrtc::kVideoCodecVP8,
- config->codec_settings);
-
- // Check the temporal layers.
- if (FLAGS_temporal_layers < 0 ||
- FLAGS_temporal_layers > webrtc::kMaxTemporalStreams) {
- fprintf(stderr, "Temporal layers number must be 0-4, was: %d\n",
- FLAGS_temporal_layers);
- return 13;
- }
- config->codec_settings->codecSpecific.VP8.numberOfTemporalLayers =
- FLAGS_temporal_layers;
-
- // Check the bit rate.
- if (FLAGS_bitrate <= 0) {
- fprintf(stderr, "Bit rate must be >0 kbps, was: %d\n", FLAGS_bitrate);
- return 5;
- }
- config->codec_settings->startBitrate = FLAGS_bitrate;
-
- // Check the keyframe interval.
- if (FLAGS_keyframe_interval < 0) {
- fprintf(stderr, "Keyframe interval must be >=0, was: %d\n",
- FLAGS_keyframe_interval);
- return 6;
- }
- config->keyframe_interval = FLAGS_keyframe_interval;
-
- // Check packet size and max payload size.
- if (FLAGS_packet_size <= 0) {
- fprintf(stderr, "Packet size must be >0 bytes, was: %d\n",
- FLAGS_packet_size);
- return 7;
- }
- config->networking_config.packet_size_in_bytes = FLAGS_packet_size;
-
- if (FLAGS_max_payload_size <= 0) {
- fprintf(stderr, "Max payload size must be >0 bytes, was: %d\n",
- FLAGS_max_payload_size);
- return 8;
- }
- config->networking_config.max_payload_size_in_bytes =
- FLAGS_max_payload_size;
-
- // Check the width and height
- if (FLAGS_width <= 0 || FLAGS_height <= 0) {
- fprintf(stderr, "Width and height must be >0.");
- return 9;
- }
- config->codec_settings->width = FLAGS_width;
- config->codec_settings->height = FLAGS_height;
- config->codec_settings->maxFramerate = FLAGS_framerate;
-
- // Calculate the size of each frame to read (according to YUV spec).
- config->frame_length_in_bytes =
- 3 * config->codec_settings->width * config->codec_settings->height / 2;
-
- // Check packet loss settings
- if (FLAGS_packet_loss_mode != "uniform" &&
- FLAGS_packet_loss_mode != "burst") {
- fprintf(stderr, "Unsupported packet loss mode, must be 'uniform' or "
- "'burst'\n.");
- return 10;
- }
- config->networking_config.packet_loss_mode = webrtc::test::kUniform;
- if (FLAGS_packet_loss_mode == "burst") {
- config->networking_config.packet_loss_mode = webrtc::test::kBurst;
- }
-
- if (FLAGS_packet_loss_probability < 0.0 ||
- FLAGS_packet_loss_probability > 1.0) {
- fprintf(stderr, "Invalid packet loss probability. Must be 0.0 - 1.0, "
- "was: %f\n", FLAGS_packet_loss_probability);
- return 11;
- }
- config->networking_config.packet_loss_probability =
- FLAGS_packet_loss_probability;
-
- if (FLAGS_packet_loss_burst_length < 1) {
- fprintf(stderr, "Invalid packet loss burst length, must be >=1, "
- "was: %d\n", FLAGS_packet_loss_burst_length);
- return 12;
- }
- config->networking_config.packet_loss_burst_length =
- FLAGS_packet_loss_burst_length;
- config->verbose = FLAGS_verbose;
- return 0;
-}
-
-void CalculateSsimVideoMetrics(webrtc::test::TestConfig* config,
- webrtc::test::QualityMetricsResult* result) {
- Log("Calculating SSIM...\n");
- I420SSIMFromFiles(config->input_filename.c_str(),
- config->output_filename.c_str(),
- config->codec_settings->width,
- config->codec_settings->height, result);
- Log(" Average: %3.2f\n", result->average);
- Log(" Min : %3.2f (frame %d)\n", result->min, result->min_frame_number);
- Log(" Max : %3.2f (frame %d)\n", result->max, result->max_frame_number);
-}
-
-void CalculatePsnrVideoMetrics(webrtc::test::TestConfig* config,
- webrtc::test::QualityMetricsResult* result) {
- Log("Calculating PSNR...\n");
- I420PSNRFromFiles(config->input_filename.c_str(),
- config->output_filename.c_str(),
- config->codec_settings->width,
- config->codec_settings->height, result);
- Log(" Average: %3.2f\n", result->average);
- Log(" Min : %3.2f (frame %d)\n", result->min, result->min_frame_number);
- Log(" Max : %3.2f (frame %d)\n", result->max, result->max_frame_number);
-}
-
-void PrintConfigurationSummary(const webrtc::test::TestConfig& config) {
- Log("Quality test with parameters:\n");
- Log(" Test name : %s\n", config.name.c_str());
- Log(" Description : %s\n", config.description.c_str());
- Log(" Input filename : %s\n", config.input_filename.c_str());
- Log(" Output directory : %s\n", config.output_dir.c_str());
- Log(" Output filename : %s\n", config.output_filename.c_str());
- Log(" Frame length : %d bytes\n", config.frame_length_in_bytes);
- Log(" Packet size : %d bytes\n",
- config.networking_config.packet_size_in_bytes);
- Log(" Max payload size : %d bytes\n",
- config.networking_config.max_payload_size_in_bytes);
- Log(" Packet loss:\n");
- Log(" Mode : %s\n",
- PacketLossModeToStr(config.networking_config.packet_loss_mode));
- Log(" Probability : %2.1f\n",
- config.networking_config.packet_loss_probability);
- Log(" Burst length : %d packets\n",
- config.networking_config.packet_loss_burst_length);
-}
-
-void PrintCsvOutput(const webrtc::test::Stats& stats,
- const webrtc::test::QualityMetricsResult& ssim_result,
- const webrtc::test::QualityMetricsResult& psnr_result) {
- Log("\nCSV output (recommended to run with --noverbose to skip the "
- "above output)\n");
- printf("frame_number encoding_successful decoding_successful "
- "encode_return_code decode_return_code "
- "encode_time_in_us decode_time_in_us "
- "bit_rate_in_kbps encoded_frame_length_in_bytes frame_type "
- "packets_dropped total_packets "
- "ssim psnr\n");
-
- for (unsigned int i = 0; i < stats.stats_.size(); ++i) {
- const webrtc::test::FrameStatistic& f = stats.stats_[i];
- const webrtc::test::FrameResult& ssim = ssim_result.frames[i];
- const webrtc::test::FrameResult& psnr = psnr_result.frames[i];
- printf("%4d, %d, %d, %2d, %2d, %6d, %6d, %5d, %7d, %d, %2d, %2d, "
- "%5.3f, %5.2f\n",
- f.frame_number,
- f.encoding_successful,
- f.decoding_successful,
- f.encode_return_code,
- f.decode_return_code,
- f.encode_time_in_us,
- f.decode_time_in_us,
- f.bit_rate_in_kbps,
- f.encoded_frame_length_in_bytes,
- f.frame_type,
- f.packets_dropped,
- f.total_packets,
- ssim.value,
- psnr.value);
- }
-}
-
-void PrintPythonOutput(const webrtc::test::TestConfig& config,
- const webrtc::test::Stats& stats,
- const webrtc::test::QualityMetricsResult& ssim_result,
- const webrtc::test::QualityMetricsResult& psnr_result) {
- Log("\nPython output (recommended to run with --noverbose to skip the "
- "above output)\n");
- printf("test_configuration = ["
- "{'name': 'name', 'value': '%s'},\n"
- "{'name': 'description', 'value': '%s'},\n"
- "{'name': 'test_number', 'value': '%d'},\n"
- "{'name': 'input_filename', 'value': '%s'},\n"
- "{'name': 'output_filename', 'value': '%s'},\n"
- "{'name': 'output_dir', 'value': '%s'},\n"
- "{'name': 'packet_size_in_bytes', 'value': '%d'},\n"
- "{'name': 'max_payload_size_in_bytes', 'value': '%d'},\n"
- "{'name': 'packet_loss_mode', 'value': '%s'},\n"
- "{'name': 'packet_loss_probability', 'value': '%f'},\n"
- "{'name': 'packet_loss_burst_length', 'value': '%d'},\n"
- "{'name': 'exclude_frame_types', 'value': '%s'},\n"
- "{'name': 'frame_length_in_bytes', 'value': '%d'},\n"
- "{'name': 'use_single_core', 'value': '%s'},\n"
- "{'name': 'keyframe_interval;', 'value': '%d'},\n"
- "{'name': 'video_codec_type', 'value': '%s'},\n"
- "{'name': 'width', 'value': '%d'},\n"
- "{'name': 'height', 'value': '%d'},\n"
- "{'name': 'bit_rate_in_kbps', 'value': '%d'},\n"
- "]\n",
- config.name.c_str(),
- config.description.c_str(),
- config.test_number,
- config.input_filename.c_str(),
- config.output_filename.c_str(),
- config.output_dir.c_str(),
- config.networking_config.packet_size_in_bytes,
- config.networking_config.max_payload_size_in_bytes,
- PacketLossModeToStr(config.networking_config.packet_loss_mode),
- config.networking_config.packet_loss_probability,
- config.networking_config.packet_loss_burst_length,
- ExcludeFrameTypesToStr(config.exclude_frame_types),
- config.frame_length_in_bytes,
- config.use_single_core ? "True " : "False",
- config.keyframe_interval,
- webrtc::test::VideoCodecTypeToStr(config.codec_settings->codecType),
- config.codec_settings->width,
- config.codec_settings->height,
- config.codec_settings->startBitrate);
- printf("frame_data_types = {"
- "'frame_number': ('number', 'Frame number'),\n"
- "'encoding_successful': ('boolean', 'Encoding successful?'),\n"
- "'decoding_successful': ('boolean', 'Decoding successful?'),\n"
- "'encode_time': ('number', 'Encode time (us)'),\n"
- "'decode_time': ('number', 'Decode time (us)'),\n"
- "'encode_return_code': ('number', 'Encode return code'),\n"
- "'decode_return_code': ('number', 'Decode return code'),\n"
- "'bit_rate': ('number', 'Bit rate (kbps)'),\n"
- "'encoded_frame_length': "
- "('number', 'Encoded frame length (bytes)'),\n"
- "'frame_type': ('string', 'Frame type'),\n"
- "'packets_dropped': ('number', 'Packets dropped'),\n"
- "'total_packets': ('number', 'Total packets'),\n"
- "'ssim': ('number', 'SSIM'),\n"
- "'psnr': ('number', 'PSNR (dB)'),\n"
- "}\n");
- printf("frame_data = [");
- for (unsigned int i = 0; i < stats.stats_.size(); ++i) {
- const webrtc::test::FrameStatistic& f = stats.stats_[i];
- const webrtc::test::FrameResult& ssim = ssim_result.frames[i];
- const webrtc::test::FrameResult& psnr = psnr_result.frames[i];
- printf("{'frame_number': %d, "
- "'encoding_successful': %s, 'decoding_successful': %s, "
- "'encode_time': %d, 'decode_time': %d, "
- "'encode_return_code': %d, 'decode_return_code': %d, "
- "'bit_rate': %d, 'encoded_frame_length': %d, 'frame_type': %s, "
- "'packets_dropped': %d, 'total_packets': %d, "
- "'ssim': %f, 'psnr': %f},\n",
- f.frame_number,
- f.encoding_successful ? "True " : "False",
- f.decoding_successful ? "True " : "False",
- f.encode_time_in_us,
- f.decode_time_in_us,
- f.encode_return_code,
- f.decode_return_code,
- f.bit_rate_in_kbps,
- f.encoded_frame_length_in_bytes,
- f.frame_type == webrtc::kDeltaFrame ? "'Delta'" : "'Other'",
- f.packets_dropped,
- f.total_packets,
- ssim.value,
- psnr.value);
- }
- printf("]\n");
-}
-
-// Runs a quality measurement on the input file supplied to the program.
-// The input file must be in YUV format.
-int main(int argc, char* argv[]) {
- std::string program_name = argv[0];
- std::string usage = "Quality test application for video comparisons.\n"
- "Run " + program_name + " --helpshort for usage.\n"
- "Example usage:\n" + program_name +
- " --input_filename=filename.yuv --width=352 --height=288\n";
- google::SetUsageMessage(usage);
-
- google::ParseCommandLineFlags(&argc, &argv, true);
-
- // Create TestConfig and codec settings struct.
- webrtc::test::TestConfig config;
- webrtc::VideoCodec codec_settings;
- config.codec_settings = &codec_settings;
-
- int return_code = HandleCommandLineFlags(&config);
- // Exit if an invalid argument is supplied.
- if (return_code != 0) {
- return return_code;
- }
-
- PrintConfigurationSummary(config);
-
- webrtc::VP8Encoder* encoder = webrtc::VP8Encoder::Create();
- webrtc::VP8Decoder* decoder = webrtc::VP8Decoder::Create();
- webrtc::test::Stats stats;
- webrtc::test::FrameReaderImpl frame_reader(config.input_filename,
- config.frame_length_in_bytes);
- webrtc::test::FrameWriterImpl frame_writer(config.output_filename,
- config.frame_length_in_bytes);
- frame_reader.Init();
- frame_writer.Init();
- webrtc::test::PacketReader packet_reader;
-
- webrtc::test::PacketManipulatorImpl packet_manipulator(
- &packet_reader, config.networking_config, config.verbose);
- // By default the packet manipulator is seeded with a fixed random.
- // If disabled we must generate a new seed.
- if (FLAGS_disable_fixed_random_seed) {
- packet_manipulator.InitializeRandomSeed(time(NULL));
- }
- webrtc::test::VideoProcessor* processor =
- new webrtc::test::VideoProcessorImpl(encoder, decoder,
- &frame_reader,
- &frame_writer,
- &packet_manipulator,
- config, &stats);
- processor->Init();
-
- int frame_number = 0;
- while (processor->ProcessFrame(frame_number)) {
- if (frame_number % 80 == 0) {
- Log("\n"); // make the output a bit nicer.
- }
- Log(".");
- frame_number++;
- }
- Log("\n");
- Log("Processed %d frames\n", frame_number);
-
- // Release encoder and decoder to make sure they have finished processing.
- encoder->Release();
- decoder->Release();
-
- // Verify statistics are correct:
- assert(frame_number == static_cast<int>(stats.stats_.size()));
-
- // Close the files before we start using them for SSIM/PSNR calculations.
- frame_reader.Close();
- frame_writer.Close();
-
- stats.PrintSummary();
-
- webrtc::test::QualityMetricsResult ssim_result;
- CalculateSsimVideoMetrics(&config, &ssim_result);
- webrtc::test::QualityMetricsResult psnr_result;
- CalculatePsnrVideoMetrics(&config, &psnr_result);
-
- if (FLAGS_csv) {
- PrintCsvOutput(stats, ssim_result, psnr_result);
- }
- if (FLAGS_python) {
- PrintPythonOutput(config, stats, ssim_result, psnr_result);
- }
- delete processor;
- delete encoder;
- delete decoder;
- Log("Quality test finished!");
- return 0;
-}
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/interface/vp8.h b/trunk/src/modules/video_coding/codecs/vp8/main/interface/vp8.h
deleted file mode 100644
index 89cf067..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/interface/vp8.h
+++ /dev/null
@@ -1,236 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- *
- * WEBRTC VP8 wrapper interface
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_H_
-
-#include "video_codec_interface.h"
-
-// VPX forward declaration
-typedef struct vpx_codec_ctx vpx_codec_ctx_t;
-typedef struct vpx_codec_ctx vpx_dec_ctx_t;
-typedef struct vpx_codec_enc_cfg vpx_codec_enc_cfg_t;
-typedef struct vpx_image vpx_image_t;
-typedef struct vpx_ref_frame vpx_ref_frame_t;
-struct vpx_codec_cx_pkt;
-
-namespace webrtc
-{
-class TemporalLayers;
-class ReferencePictureSelection;
-
-class VP8Encoder : public VideoEncoder {
- public:
- static VP8Encoder* Create();
-
- virtual ~VP8Encoder();
-
- // Free encoder memory.
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual int Release();
-
- // Initialize the encoder with the information from the codecSettings
- //
- // Input:
- // - codec_settings : Codec settings
- // - number_of_cores : Number of cores available for the encoder
- // - max_payload_size : The maximum size each payload is allowed
- // to have. Usually MTU - overhead.
- //
- // Return value : Set bit rate if OK
- // <0 - Errors:
- // WEBRTC_VIDEO_CODEC_ERR_PARAMETER
- // WEBRTC_VIDEO_CODEC_ERR_SIZE
- // WEBRTC_VIDEO_CODEC_LEVEL_EXCEEDED
- // WEBRTC_VIDEO_CODEC_MEMORY
- // WEBRTC_VIDEO_CODEC_ERROR
- virtual int InitEncode(const VideoCodec* codec_settings,
- int number_of_cores,
- uint32_t max_payload_size);
-
- // Encode an I420 image (as a part of a video stream). The encoded image
- // will be returned to the user through the encode complete callback.
- //
- // Input:
- // - input_image : Image to be encoded
- // - frame_types : Frame type to be generated by the encoder.
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK
- // <0 - Errors:
- // WEBRTC_VIDEO_CODEC_ERR_PARAMETER
- // WEBRTC_VIDEO_CODEC_MEMORY
- // WEBRTC_VIDEO_CODEC_ERROR
- // WEBRTC_VIDEO_CODEC_TIMEOUT
-
- virtual int Encode(const RawImage& input_image,
- const CodecSpecificInfo* codec_specific_info,
- const VideoFrameType* frame_types);
-
- // Register an encode complete callback object.
- //
- // Input:
- // - callback : Callback object which handles encoded images.
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual int RegisterEncodeCompleteCallback(EncodedImageCallback* callback);
-
- // Inform the encoder of the new packet loss rate and the round-trip time of
- // the network.
- //
- // - packet_loss : Fraction lost
- // (loss rate in percent = 100 * packetLoss / 255)
- // - rtt : Round-trip time in milliseconds
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK
- // <0 - Errors: WEBRTC_VIDEO_CODEC_ERROR
- //
- virtual int SetChannelParameters(uint32_t packet_loss, int rtt);
-
- // Inform the encoder about the new target bit rate.
- //
- // - new_bitrate_kbit : New target bit rate
- // - frame_rate : The target frame rate
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual int SetRates(uint32_t new_bitrate_kbit, uint32_t frame_rate);
-
- private:
- VP8Encoder();
-
- // Call encoder initialize function and set control settings.
- int InitAndSetControlSettings();
-
- void PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
- const vpx_codec_cx_pkt& pkt);
-
- int GetEncodedFrame(const RawImage& input_image);
-
- int GetEncodedPartitions(const RawImage& input_image);
-
- // Determine maximum target for Intra frames
- //
- // Input:
- // - optimal_buffer_size : Optimal buffer size
- // Return Value : Max target size for Intra frames represented as
- // percentage of the per frame bandwidth
- uint32_t MaxIntraTarget(uint32_t optimal_buffer_size);
-
- EncodedImage encoded_image_;
- EncodedImageCallback* encoded_complete_callback_;
- VideoCodec codec_;
- bool inited_;
- uint32_t timestamp_;
- uint16_t picture_id_;
- bool feedback_mode_;
- int cpu_speed_;
- uint32_t rc_max_intra_target_;
- int token_partitions_;
- ReferencePictureSelection* rps_;
- TemporalLayers* temporal_layers_;
- vpx_codec_ctx_t* encoder_;
- vpx_codec_enc_cfg_t* config_;
- vpx_image_t* raw_;
-}; // end of VP8Encoder class
-
-
-class VP8Decoder : public VideoDecoder {
- public:
- static VP8Decoder* Create();
-
- virtual ~VP8Decoder();
-
- // Initialize the decoder.
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK.
- // <0 - Errors:
- // WEBRTC_VIDEO_CODEC_ERROR
- virtual int InitDecode(const VideoCodec* inst, int number_of_cores);
-
- // Decode encoded image (as a part of a video stream). The decoded image
- // will be returned to the user through the decode complete callback.
- //
- // Input:
- // - input_image : Encoded image to be decoded
- // - missing_frames : True if one or more frames have been lost
- // since the previous decode call.
- // - fragmentation : Specifies the start and length of each VP8
- // partition.
- // - codec_specific_info : pointer to specific codec data
- // - render_time_ms : Render time in Ms
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK
- // <0 - Errors:
- // WEBRTC_VIDEO_CODEC_ERROR
- // WEBRTC_VIDEO_CODEC_ERR_PARAMETER
- virtual int Decode(const EncodedImage& input_image,
- bool missing_frames,
- const RTPFragmentationHeader* fragmentation,
- const CodecSpecificInfo* codec_specific_info,
- int64_t /*render_time_ms*/);
-
- // Register a decode complete callback object.
- //
- // Input:
- // - callback : Callback object which handles decoded images.
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK, < 0 otherwise.
- virtual int RegisterDecodeCompleteCallback(DecodedImageCallback* callback);
-
- // Free decoder memory.
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK if OK
- // <0 - Errors:
- // WEBRTC_VIDEO_CODEC_ERROR
- virtual int Release();
-
- // Reset decoder state and prepare for a new call.
- //
- // Return value : WEBRTC_VIDEO_CODEC_OK.
- // <0 - Errors:
- // WEBRTC_VIDEO_CODEC_UNINITIALIZED
- // WEBRTC_VIDEO_CODEC_ERROR
- virtual int Reset();
-
- // Create a copy of the codec and its internal state.
- //
- // Return value : A copy of the instance if OK, NULL otherwise.
- virtual VideoDecoder* Copy();
-
- private:
- VP8Decoder();
-
- // Copy reference image from this _decoder to the _decoder in copyTo. Set
- // which frame type to copy in _refFrame->frame_type before the call to
- // this function.
- int CopyReference(VP8Decoder* copy);
-
- int DecodePartitions(const EncodedImage& input_image,
- const RTPFragmentationHeader* fragmentation);
-
- int ReturnFrame(const vpx_image_t* img, uint32_t timeStamp);
-
- RawImage decoded_image_;
- DecodedImageCallback* decode_complete_callback_;
- bool inited_;
- bool feedback_mode_;
- vpx_dec_ctx_t* decoder_;
- VideoCodec codec_;
- EncodedImage last_keyframe_;
- int image_format_;
- vpx_ref_frame_t* ref_frame_;
- int propagation_cnt_;
- bool latest_keyframe_complete_;
- bool mfqe_enabled_;
-}; // end of VP8Decoder class
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_H_
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/interface/vp8_common_types.h b/trunk/src/modules/video_coding/codecs/vp8/main/interface/vp8_common_types.h
deleted file mode 100644
index 6f347cd..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/interface/vp8_common_types.h
+++ /dev/null
@@ -1,29 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_COMMON_TYPES_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_COMMON_TYPES_H_
-
-#include "common_types.h"
-
-namespace webrtc {
-
-// Ratio allocation between temporal streams:
-// Values as required for the VP8 codec (accumulating).
-static const float
- kVp8LayerRateAlloction[kMaxTemporalStreams][kMaxTemporalStreams] = {
- {1.0f, 0, 0, 0}, // 1 layer
- {0.6f, 1.0f , 0 , 0}, // 2 layers {60%, 40%}
- {0.4f, 0.6f , 1.0f, 0}, // 3 layers {40%, 20%, 40%}
- {0.25f, 0.4f, 0.6f, 1.0f} // 4 layers {25%, 15%, 20%, 40%}
-};
-
-} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_COMMON_TYPES_H_
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/Android.mk b/trunk/src/modules/video_coding/codecs/vp8/main/source/Android.mk
deleted file mode 100644
index eb6c1af..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/source/Android.mk
+++ /dev/null
@@ -1,47 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../../../android-webrtc.mk
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_vp8
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := \
- reference_picture_selection.cc \
- vp8.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-# TODO(leozwang) Enable WEBRTC_LIBVPX_VERSION after libvpx is updateed
-# to a new version and also add temporal_layers.cc
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../../interface \
- $(LOCAL_PATH)/../../../../../.. \
- $(LOCAL_PATH)/../../../../../../common_video/interface \
- $(LOCAL_PATH)/../../../../../../common_video/vplib/main/interface \
- $(LOCAL_PATH)/../../../../../../modules/interface \
- $(LOCAL_PATH)/../../../../../../system_wrappers/interface \
- external/libvpx
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection.cc b/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection.cc
deleted file mode 100644
index 3ae6f19..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection.cc
+++ /dev/null
@@ -1,131 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "reference_picture_selection.h"
-
-#include "typedefs.h"
-#include "vpx/vpx_encoder.h"
-#include "vpx/vp8cx.h"
-
-namespace webrtc {
-
-ReferencePictureSelection::ReferencePictureSelection()
- : kRttConfidence(1.33),
- update_golden_next_(true),
- established_golden_(false),
- received_ack_(false),
- last_sent_ref_picture_id_(0),
- last_sent_ref_update_time_(0),
- established_ref_picture_id_(0),
- last_refresh_time_(0),
- rtt_(0) {
-}
-
-void ReferencePictureSelection::Init() {
- update_golden_next_ = true;
- established_golden_ = false;
- received_ack_ = false;
- last_sent_ref_picture_id_ = 0;
- last_sent_ref_update_time_ = 0;
- established_ref_picture_id_ = 0;
- last_refresh_time_ = 0;
- rtt_ = 0;
-}
-
-void ReferencePictureSelection::ReceivedRPSI(int rpsi_picture_id) {
- // Assume RPSI is signaled with 14 bits.
- if ((rpsi_picture_id & 0x3fff) == (last_sent_ref_picture_id_ & 0x3fff)) {
- // Remote peer has received our last reference frame, switch frame type.
- received_ack_ = true;
- established_golden_ = update_golden_next_;
- update_golden_next_ = !update_golden_next_;
- established_ref_picture_id_ = last_sent_ref_picture_id_;
- }
-}
-
-bool ReferencePictureSelection::ReceivedSLI(uint32_t now_ts) {
- bool send_refresh = false;
- // Don't send a refresh more than once per round-trip time.
- // This is to avoid too frequent refreshes, since the receiver
- // will signal an SLI for every corrupt frame.
- if (TimestampDiff(now_ts, last_refresh_time_) > rtt_) {
- send_refresh = true;
- last_refresh_time_ = now_ts;
- }
- return send_refresh;
-}
-
-int ReferencePictureSelection::EncodeFlags(int picture_id, bool send_refresh,
- uint32_t now_ts) {
- int flags = 0;
- // We can't refresh the decoder until we have established the key frame.
- if (send_refresh && received_ack_) {
- flags |= VP8_EFLAG_NO_REF_LAST; // Don't reference the last frame
- if (established_golden_)
- flags |= VP8_EFLAG_NO_REF_ARF; // Don't reference the alt-ref frame.
- else
- flags |= VP8_EFLAG_NO_REF_GF; // Don't reference the golden frame
- }
-
- // Make sure we don't update the reference frames too often. We must wait long
- // enough for an RPSI to arrive after the decoder decoded the reference frame.
- // Ideally that should happen after one round-trip time.
- // Add a margin defined by |kRttConfidence|.
- uint32_t update_interval = kRttConfidence * rtt_;
- if (update_interval < kMinUpdateInterval)
- update_interval = kMinUpdateInterval;
- // Don't send reference frame updates until we have an established reference.
- if (TimestampDiff(now_ts, last_sent_ref_update_time_) > update_interval &&
- received_ack_) {
- flags |= VP8_EFLAG_NO_REF_LAST; // Don't reference the last frame.
- if (update_golden_next_) {
- flags |= VP8_EFLAG_FORCE_GF; // Update the golden reference.
- flags |= VP8_EFLAG_NO_UPD_ARF; // Don't update alt-ref.
- flags |= VP8_EFLAG_NO_REF_GF; // Don't reference the golden frame.
- } else {
- flags |= VP8_EFLAG_FORCE_ARF; // Update the alt-ref reference.
- flags |= VP8_EFLAG_NO_UPD_GF; // Don't update the golden frame.
- flags |= VP8_EFLAG_NO_REF_ARF; // Don't reference the alt-ref frame.
- }
- last_sent_ref_picture_id_ = picture_id;
- last_sent_ref_update_time_ = now_ts;
- } else {
- // No update of golden or alt-ref. We can therefore freely reference the
- // established reference frame and the last frame.
- if (established_golden_)
- flags |= VP8_EFLAG_NO_REF_ARF; // Don't reference the alt-ref frame.
- else
- flags |= VP8_EFLAG_NO_REF_GF; // Don't reference the golden frame.
- flags |= VP8_EFLAG_NO_UPD_GF; // Don't update the golden frame.
- flags |= VP8_EFLAG_NO_UPD_ARF; // Don't update the alt-ref frame.
- }
- return flags;
-}
-
-void ReferencePictureSelection::EncodedKeyFrame(int picture_id) {
- last_sent_ref_picture_id_ = picture_id;
- received_ack_ = false;
-}
-
-void ReferencePictureSelection::SetRtt(int rtt) {
- // Convert from milliseconds to timestamp frequency.
- rtt_ = 90 * rtt;
-}
-
-uint32_t ReferencePictureSelection::TimestampDiff(uint32_t new_ts,
- uint32_t old_ts) {
- if (old_ts > new_ts) {
- // Assuming this is a wrap, doing a compensated subtraction.
- return (new_ts + (static_cast<int64_t>(1) << 32)) - old_ts;
- }
- return new_ts - old_ts;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection.h b/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection.h
deleted file mode 100644
index 59e5940..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection.h
+++ /dev/null
@@ -1,78 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file defines classes for doing reference picture selection, primarily
- * with VP8.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_REFERENCE_PICTURE_SELECTION_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_REFERENCE_PICTURE_SELECTION_H_
-
-#include "typedefs.h"
-
-namespace webrtc {
-
-class ReferencePictureSelection {
- public:
- ReferencePictureSelection();
- void Init();
-
- // Report a received reference picture selection indication. This will
- // introduce a new established reference if the received RPSI isn't too late.
- void ReceivedRPSI(int rpsi_picture_id);
-
- // Report a received slice loss indication. Returns true if a refresh frame
- // must be sent to the receiver, which is accomplished by only predicting
- // from the established reference.
- // |now_ts| is the RTP timestamp corresponding to the current time. Typically
- // the capture timestamp of the frame currently being processed.
- // Returns true if it's time to encode a decoder refresh, otherwise false.
- bool ReceivedSLI(uint32_t now_ts);
-
- // Returns the recommended VP8 encode flags needed. May refresh the decoder
- // and/or update the reference buffers.
- // |picture_id| picture id of the frame to be encoded.
- // |send_refresh| should be set to true if a decoder refresh should be
- // encoded, otherwise false.
- // |now_ts| is the RTP timestamp corresponding to the current time. Typically
- // the capture timestamp of the frame currently being processed.
- // Returns the flags to be given to the libvpx encoder when encoding the next
- // frame.
- int EncodeFlags(int picture_id, bool send_refresh, uint32_t now_ts);
-
- // Notify the RPS that the frame with picture id |picture_id| was encoded as
- // a key frame, effectively updating all reference buffers.
- void EncodedKeyFrame(int picture_id);
-
- // Set the round-trip time between the sender and the receiver to |rtt|
- // milliseconds.
- void SetRtt(int rtt);
-
- private:
- static uint32_t TimestampDiff(uint32_t new_ts, uint32_t old_ts);
-
- // The minimum time between reference frame updates.
- enum { kMinUpdateInterval = 90 * 10 }; // Timestamp frequency
- const double kRttConfidence;
-
- bool update_golden_next_;
- bool established_golden_;
- bool received_ack_;
- int last_sent_ref_picture_id_;
- uint32_t last_sent_ref_update_time_;
- int established_ref_picture_id_;
- uint32_t last_refresh_time_;
- uint32_t rtt_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_REFERENCE_PICTURE_SELECTION_H_
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection_unittest.cc b/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection_unittest.cc
deleted file mode 100644
index cdac76c..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/source/reference_picture_selection_unittest.cc
+++ /dev/null
@@ -1,107 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "gtest/gtest.h"
-#include "reference_picture_selection.h"
-#include "vpx/vpx_encoder.h"
-#include "vpx/vp8cx.h"
-
-using webrtc::ReferencePictureSelection;
-
-// The minimum time between reference frame updates. Should match the values
-// set in reference_picture_selection.h
-enum { kMinUpdateInterval = 10 };
-// The minimum time between decoder refreshes through restricted prediction.
-// Should match the values set in reference_picture_selection.h
-enum { kRtt = 10 };
-
-enum {
- kNoPropagationGolden = VP8_EFLAG_NO_REF_ARF |
- VP8_EFLAG_NO_UPD_GF |
- VP8_EFLAG_NO_UPD_ARF,
- kNoPropagationAltRef = VP8_EFLAG_NO_REF_GF |
- VP8_EFLAG_NO_UPD_GF |
- VP8_EFLAG_NO_UPD_ARF,
- kPropagateGolden = VP8_EFLAG_FORCE_GF |
- VP8_EFLAG_NO_UPD_ARF |
- VP8_EFLAG_NO_REF_GF |
- VP8_EFLAG_NO_REF_LAST,
- kPropagateAltRef = VP8_EFLAG_FORCE_ARF |
- VP8_EFLAG_NO_UPD_GF |
- VP8_EFLAG_NO_REF_ARF |
- VP8_EFLAG_NO_REF_LAST,
- kRefreshFromGolden = VP8_EFLAG_NO_REF_LAST |
- VP8_EFLAG_NO_REF_ARF,
- kRefreshFromAltRef = VP8_EFLAG_NO_REF_LAST |
- VP8_EFLAG_NO_REF_GF
-};
-
-class TestRPS : public ::testing::Test {
- protected:
- virtual void SetUp() {
- rps_.Init();
- // Initialize with sending a key frame and acknowledging it.
- rps_.EncodedKeyFrame(0);
- rps_.ReceivedRPSI(0);
- rps_.SetRtt(kRtt);
- }
-
- ReferencePictureSelection rps_;
-};
-
-TEST_F(TestRPS, TestPropagateReferenceFrames) {
- // Should propagate the alt-ref reference.
- uint32_t time = (4 * kMinUpdateInterval) / 3 + 1;
- EXPECT_EQ(rps_.EncodeFlags(1, false, 90 * time), kPropagateAltRef);
- rps_.ReceivedRPSI(1);
- time += (4 * (time + kMinUpdateInterval)) / 3 + 1;
- // Should propagate the golden reference.
- EXPECT_EQ(rps_.EncodeFlags(2, false, 90 * time), kPropagateGolden);
- rps_.ReceivedRPSI(2);
- // Should propagate the alt-ref reference.
- time = (4 * (time + kMinUpdateInterval)) / 3 + 1;
- EXPECT_EQ(rps_.EncodeFlags(3, false, 90 * time), kPropagateAltRef);
- rps_.ReceivedRPSI(3);
- // Shouldn't propagate any reference frames (except last), and the established
- // reference is alt-ref.
- time = time + kMinUpdateInterval;
- EXPECT_EQ(rps_.EncodeFlags(4, false, 90 * time), kNoPropagationAltRef);
-}
-
-TEST_F(TestRPS, TestDecoderRefresh) {
- uint32_t time = kRtt + 1;
- // No more than one refresh per RTT.
- EXPECT_EQ(rps_.ReceivedSLI(90 * time), true);
- time += 5;
- EXPECT_EQ(rps_.ReceivedSLI(90 * time), false);
- time += kRtt - 4;
- EXPECT_EQ(rps_.ReceivedSLI(90 * time), true);
- // Enough time have elapsed since the previous reference propagation, we will
- // therefore get both a refresh from golden and a propagation of alt-ref.
- EXPECT_EQ(rps_.EncodeFlags(5, true, 90 * time), kRefreshFromGolden |
- kPropagateAltRef);
- rps_.ReceivedRPSI(5);
- time += kRtt + 1;
- // Enough time for a new refresh, but not enough time for a reference
- // propagation.
- EXPECT_EQ(rps_.ReceivedSLI(90 * time), true);
- EXPECT_EQ(rps_.EncodeFlags(6, true, 90 * time), kRefreshFromAltRef |
- kNoPropagationAltRef);
-}
-
-TEST_F(TestRPS, TestWrap) {
- EXPECT_EQ(rps_.ReceivedSLI(0xffffffff), true);
- EXPECT_EQ(rps_.ReceivedSLI(1), false);
- EXPECT_EQ(rps_.ReceivedSLI(90 * 100), true);
-
- EXPECT_EQ(rps_.EncodeFlags(7, false, 0xffffffff), kPropagateAltRef);
- EXPECT_EQ(rps_.EncodeFlags(8, false, 1), kNoPropagationGolden);
- EXPECT_EQ(rps_.EncodeFlags(10, false, 90 * 100), kPropagateAltRef);
-}
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers.cc b/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers.cc
deleted file mode 100644
index 59ad9c2..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers.cc
+++ /dev/null
@@ -1,219 +0,0 @@
-/* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-*
-* Use of this source code is governed by a BSD-style license
-* that can be found in the LICENSE file in the root of the source
-* tree. An additional intellectual property rights grant can be found
-* in the file PATENTS. All contributing project authors may
-* be found in the AUTHORS file in the root of the source tree.
-*/
-
-#include "temporal_layers.h"
-
-#include <stdlib.h>
-#include <string.h>
-#include <cassert>
-
-#include "modules/interface/module_common_types.h"
-#include "modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "modules/video_coding/codecs/vp8/main/interface/vp8_common_types.h"
-
-#include "vpx/vpx_encoder.h"
-#include "vpx/vp8cx.h"
-
-namespace webrtc {
-
-TemporalLayers::TemporalLayers(int numberOfTemporalLayers)
- : number_of_temporal_layers_(numberOfTemporalLayers),
- temporal_ids_length_(0),
- temporal_pattern_length_(0),
- tl0_pic_idx_(rand()),
- pattern_idx_(255) {
- assert(kMaxTemporalStreams >= numberOfTemporalLayers);
- memset(temporal_ids_, 0, sizeof(temporal_ids_));
- memset(temporal_pattern_, 0, sizeof(temporal_pattern_));
-}
-
-bool TemporalLayers::ConfigureBitrates(int bitrateKbit,
- vpx_codec_enc_cfg_t* cfg) {
- switch (number_of_temporal_layers_) {
- case 0:
- case 1:
- // Do nothing.
- break;
- case 2:
- temporal_ids_length_ = 2;
- temporal_ids_[0] = 0;
- temporal_ids_[1] = 1;
- cfg->ts_number_layers = number_of_temporal_layers_;
- cfg->ts_periodicity = temporal_ids_length_;
- // Split stream 60% 40%.
- // Bitrate API for VP8 is the agregated bitrate for all lower layers.
- cfg->ts_target_bitrate[0] = bitrateKbit * kVp8LayerRateAlloction[1][0];
- cfg->ts_target_bitrate[1] = bitrateKbit;
- cfg->ts_rate_decimator[0] = 2;
- cfg->ts_rate_decimator[1] = 1;
- memcpy(cfg->ts_layer_id,
- temporal_ids_,
- sizeof(unsigned int) * temporal_ids_length_);
- temporal_pattern_length_ = 8;
- temporal_pattern_[0] = kTemporalUpdateLast;
- temporal_pattern_[1] = kTemporalUpdateGoldenWithoutDependency;
- temporal_pattern_[2] = kTemporalUpdateLast;
- temporal_pattern_[3] = kTemporalUpdateGolden;
- temporal_pattern_[4] = kTemporalUpdateLast;
- temporal_pattern_[5] = kTemporalUpdateGolden;
- temporal_pattern_[6] = kTemporalUpdateLast;
- temporal_pattern_[7] = kTemporalUpdateNoneNoRefAltref;
- break;
- case 3:
- temporal_ids_length_ = 4;
- temporal_ids_[0] = 0;
- temporal_ids_[1] = 2;
- temporal_ids_[2] = 1;
- temporal_ids_[3] = 2;
- cfg->ts_number_layers = number_of_temporal_layers_;
- cfg->ts_periodicity = temporal_ids_length_;
- // Split stream 40% 20% 40%.
- // Bitrate API for VP8 is the agregated bitrate for all lower layers.
- cfg->ts_target_bitrate[0] = bitrateKbit * kVp8LayerRateAlloction[2][0];
- cfg->ts_target_bitrate[1] = bitrateKbit * kVp8LayerRateAlloction[2][1];
- cfg->ts_target_bitrate[2] = bitrateKbit;
- cfg->ts_rate_decimator[0] = 4;
- cfg->ts_rate_decimator[1] = 2;
- cfg->ts_rate_decimator[2] = 1;
- memcpy(cfg->ts_layer_id,
- temporal_ids_,
- sizeof(unsigned int) * temporal_ids_length_);
- temporal_pattern_length_ = 8;
- temporal_pattern_[0] = kTemporalUpdateLast;
- temporal_pattern_[1] = kTemporalUpdateAltrefWithoutDependency;
- temporal_pattern_[2] = kTemporalUpdateGoldenWithoutDependency;
- temporal_pattern_[3] = kTemporalUpdateAltref;
- temporal_pattern_[4] = kTemporalUpdateLast;
- temporal_pattern_[5] = kTemporalUpdateAltref;
- temporal_pattern_[6] = kTemporalUpdateGolden;
- temporal_pattern_[7] = kTemporalUpdateNone;
- break;
- case 4:
- temporal_ids_length_ = 8;
- temporal_ids_[0] = 0;
- temporal_ids_[1] = 3;
- temporal_ids_[2] = 2;
- temporal_ids_[3] = 3;
- temporal_ids_[4] = 1;
- temporal_ids_[5] = 3;
- temporal_ids_[6] = 2;
- temporal_ids_[7] = 3;
- // Split stream 25% 15% 20% 40%.
- // Bitrate API for VP8 is the agregated bitrate for all lower layers.
- cfg->ts_number_layers = 4;
- cfg->ts_periodicity = temporal_ids_length_;
- cfg->ts_target_bitrate[0] = bitrateKbit * kVp8LayerRateAlloction[3][0];
- cfg->ts_target_bitrate[1] = bitrateKbit * kVp8LayerRateAlloction[3][1];
- cfg->ts_target_bitrate[2] = bitrateKbit * kVp8LayerRateAlloction[3][2];
- cfg->ts_target_bitrate[3] = bitrateKbit;
- cfg->ts_rate_decimator[0] = 8;
- cfg->ts_rate_decimator[1] = 4;
- cfg->ts_rate_decimator[2] = 2;
- cfg->ts_rate_decimator[3] = 1;
- memcpy(cfg->ts_layer_id,
- temporal_ids_,
- sizeof(unsigned int) * temporal_ids_length_);
- temporal_pattern_length_ = 16;
- temporal_pattern_[0] = kTemporalUpdateLast;
- temporal_pattern_[1] = kTemporalUpdateNone;
- temporal_pattern_[2] = kTemporalUpdateAltrefWithoutDependency;
- temporal_pattern_[3] = kTemporalUpdateNone;
- temporal_pattern_[4] = kTemporalUpdateGoldenWithoutDependency;
- temporal_pattern_[5] = kTemporalUpdateNone;
- temporal_pattern_[6] = kTemporalUpdateAltref;
- temporal_pattern_[7] = kTemporalUpdateNone;
- temporal_pattern_[8] = kTemporalUpdateLast;
- temporal_pattern_[9] = kTemporalUpdateNone;
- temporal_pattern_[10] = kTemporalUpdateAltref;
- temporal_pattern_[11] = kTemporalUpdateNone;
- temporal_pattern_[12] = kTemporalUpdateGolden;
- temporal_pattern_[13] = kTemporalUpdateNone;
- temporal_pattern_[14] = kTemporalUpdateAltref;
- temporal_pattern_[15] = kTemporalUpdateNone;
- break;
- default:
- assert(false);
- return false;
- }
- return true;
-}
-
-int TemporalLayers::EncodeFlags() {
- assert(number_of_temporal_layers_ > 1);
- assert(kMaxTemporalPattern >= temporal_pattern_length_);
- assert(0 < temporal_pattern_length_);
-
- int flags = 0;
- int patternIdx = ++pattern_idx_ % temporal_pattern_length_;
- assert(kMaxTemporalPattern >= patternIdx);
- switch (temporal_pattern_[patternIdx]) {
- case kTemporalUpdateLast:
- flags |= VP8_EFLAG_NO_UPD_GF;
- flags |= VP8_EFLAG_NO_UPD_ARF;
- flags |= VP8_EFLAG_NO_REF_GF;
- flags |= VP8_EFLAG_NO_REF_ARF;
- break;
- case kTemporalUpdateGoldenWithoutDependency:
- flags |= VP8_EFLAG_NO_REF_GF;
- // Deliberately no break here.
- case kTemporalUpdateGolden:
- flags |= VP8_EFLAG_NO_REF_ARF;
- flags |= VP8_EFLAG_NO_UPD_ARF;
- flags |= VP8_EFLAG_NO_UPD_LAST;
- break;
- case kTemporalUpdateAltrefWithoutDependency:
- flags |= VP8_EFLAG_NO_REF_ARF;
- flags |= VP8_EFLAG_NO_REF_GF;
- // Deliberately no break here.
- case kTemporalUpdateAltref:
- flags |= VP8_EFLAG_NO_UPD_GF;
- flags |= VP8_EFLAG_NO_UPD_LAST;
- break;
- case kTemporalUpdateNoneNoRefAltref:
- flags |= VP8_EFLAG_NO_REF_ARF;
- // Deliberately no break here.
- case kTemporalUpdateNone:
- flags |= VP8_EFLAG_NO_UPD_GF;
- flags |= VP8_EFLAG_NO_UPD_ARF;
- flags |= VP8_EFLAG_NO_UPD_LAST;
- flags |= VP8_EFLAG_NO_UPD_ENTROPY;
- break;
- }
- return flags;
-}
-
-void TemporalLayers::PopulateCodecSpecific(bool key_frame,
- CodecSpecificInfoVP8 *vp8_info) {
- assert(number_of_temporal_layers_ > 1);
- assert(0 < temporal_ids_length_);
-
- if (key_frame) {
- // Keyframe is always temporal layer 0
- vp8_info->temporalIdx = 0;
- } else {
- vp8_info->temporalIdx = temporal_ids_[pattern_idx_ % temporal_ids_length_];
- }
- TemporalReferences temporal_reference =
- temporal_pattern_[pattern_idx_ % temporal_pattern_length_];
-
- if (temporal_reference == kTemporalUpdateAltrefWithoutDependency ||
- temporal_reference == kTemporalUpdateGoldenWithoutDependency ||
- (temporal_reference == kTemporalUpdateNone &&
- number_of_temporal_layers_ == 4)) {
- vp8_info->layerSync = true;
- } else {
- vp8_info->layerSync = false;
- }
-
- if (vp8_info->temporalIdx == 0) {
- tl0_pic_idx_++;
- }
- vp8_info->tl0PicIdx = tl0_pic_idx_;
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers.h b/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers.h
deleted file mode 100644
index 2ca7229..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-*
-* Use of this source code is governed by a BSD-style license
-* that can be found in the LICENSE file in the root of the source
-* tree. An additional intellectual property rights grant can be found
-* in the file PATENTS. All contributing project authors may
-* be found in the AUTHORS file in the root of the source tree.
-*/
-/*
-* This file defines classes for doing temporal layers with VP8.
-*/
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
-
-#include <typedefs.h>
-
- // VPX forward declaration
-typedef struct vpx_codec_enc_cfg vpx_codec_enc_cfg_t;
-
-namespace webrtc {
-
-struct CodecSpecificInfoVP8;
-
-class TemporalLayers {
- public:
- TemporalLayers(int number_of_temporal_layers);
-
- // Returns the recommended VP8 encode flags needed. May refresh the decoder
- // and/or update the reference buffers.
- int EncodeFlags();
-
- bool ConfigureBitrates(int bitrate_kbit, vpx_codec_enc_cfg_t* cfg);
-
- void PopulateCodecSpecific(bool key_frame, CodecSpecificInfoVP8 *vp8_info);
-
- private:
- enum TemporalReferences {
- // Second layer and last frame in cycle, for 2 layers.
- kTemporalUpdateNoneNoRefAltref = 6,
- // Highest enhancement layer.
- kTemporalUpdateNone = 5,
- // Second enhancement layer.
- kTemporalUpdateAltref = 4,
- // Second enhancement layer without dependency on previous frames in
- // the second enhancement layer.
- kTemporalUpdateAltrefWithoutDependency = 3,
- // First enhancement layer.
- kTemporalUpdateGolden = 2,
- // First enhancement layer without dependency on previous frames in
- // the first enhancement layer.
- kTemporalUpdateGoldenWithoutDependency = 1,
- // Base layer.
- kTemporalUpdateLast = 0,
- };
- enum { kMaxTemporalPattern = 16 };
-
- int number_of_temporal_layers_;
- int temporal_ids_length_;
- int temporal_ids_[kMaxTemporalPattern];
- int temporal_pattern_length_;
- TemporalReferences temporal_pattern_[kMaxTemporalPattern];
- uint8_t tl0_pic_idx_;
- uint8_t pattern_idx_;
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_TEMPORAL_LAYERS_H_
-
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers_unittest.cc b/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers_unittest.cc
deleted file mode 100644
index 6525dda..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/source/temporal_layers_unittest.cc
+++ /dev/null
@@ -1,202 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-#include "gtest/gtest.h"
-#include "temporal_layers.h"
-#include "video_codec_interface.h"
-
-#include "vpx/vpx_encoder.h"
-#include "vpx/vp8cx.h"
-
-namespace webrtc {
-
-enum {
- kTemporalUpdateLast = VP8_EFLAG_NO_UPD_GF |
- VP8_EFLAG_NO_UPD_ARF |
- VP8_EFLAG_NO_REF_GF |
- VP8_EFLAG_NO_REF_ARF,
- kTemporalUpdateGoldenWithoutDependency = VP8_EFLAG_NO_REF_GF |
- VP8_EFLAG_NO_REF_ARF |
- VP8_EFLAG_NO_UPD_ARF |
- VP8_EFLAG_NO_UPD_LAST,
- kTemporalUpdateGolden = VP8_EFLAG_NO_REF_ARF |
- VP8_EFLAG_NO_UPD_ARF |
- VP8_EFLAG_NO_UPD_LAST,
- kTemporalUpdateAltrefWithoutDependency = VP8_EFLAG_NO_REF_ARF |
- VP8_EFLAG_NO_REF_GF |
- VP8_EFLAG_NO_UPD_GF |
- VP8_EFLAG_NO_UPD_LAST,
- kTemporalUpdateAltref = VP8_EFLAG_NO_UPD_GF |
- VP8_EFLAG_NO_UPD_LAST,
- kTemporalUpdateNone = VP8_EFLAG_NO_UPD_GF |
- VP8_EFLAG_NO_UPD_ARF |
- VP8_EFLAG_NO_UPD_LAST |
- VP8_EFLAG_NO_UPD_ENTROPY,
- kTemporalUpdateNoneNoRefAltRef = VP8_EFLAG_NO_REF_ARF |
- VP8_EFLAG_NO_UPD_GF |
- VP8_EFLAG_NO_UPD_ARF |
- VP8_EFLAG_NO_UPD_LAST |
- VP8_EFLAG_NO_UPD_ENTROPY,
-};
-
-TEST(TemporalLayersTest, 2Layers) {
- TemporalLayers tl(2);
- vpx_codec_enc_cfg_t cfg;
- CodecSpecificInfoVP8 vp8_info;
- tl.ConfigureBitrates(500, &cfg);
-
- int expected_flags[16] = { kTemporalUpdateLast,
- kTemporalUpdateGoldenWithoutDependency,
- kTemporalUpdateLast,
- kTemporalUpdateGolden,
- kTemporalUpdateLast,
- kTemporalUpdateGolden,
- kTemporalUpdateLast,
- kTemporalUpdateNoneNoRefAltRef,
- kTemporalUpdateLast,
- kTemporalUpdateGoldenWithoutDependency,
- kTemporalUpdateLast,
- kTemporalUpdateGolden,
- kTemporalUpdateLast,
- kTemporalUpdateGolden,
- kTemporalUpdateLast,
- kTemporalUpdateNoneNoRefAltRef
- };
- int expected_temporal_idx[16] =
- { 0, 1, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1, 0, 1 };
-
- bool expected_layer_sync[16] =
- { false, true, false, false, false, false, false, false,
- false, true, false, false, false, false, false, false };
-
- for (int i = 0; i < 16; ++i) {
- EXPECT_EQ(expected_flags[i], tl.EncodeFlags());
- tl.PopulateCodecSpecific(false, &vp8_info);
- EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
- bool expected_sync = expected_layer_sync[i];
- EXPECT_EQ(expected_sync, vp8_info.layerSync);
- }
-}
-
-TEST(TemporalLayersTest, 3Layers) {
- TemporalLayers tl(3);
- vpx_codec_enc_cfg_t cfg;
- CodecSpecificInfoVP8 vp8_info;
- tl.ConfigureBitrates(500, &cfg);
-
- int expected_flags[16] = { kTemporalUpdateLast,
- kTemporalUpdateAltrefWithoutDependency,
- kTemporalUpdateGoldenWithoutDependency,
- kTemporalUpdateAltref,
- kTemporalUpdateLast,
- kTemporalUpdateAltref,
- kTemporalUpdateGolden,
- kTemporalUpdateNone,
- kTemporalUpdateLast,
- kTemporalUpdateAltrefWithoutDependency,
- kTemporalUpdateGoldenWithoutDependency,
- kTemporalUpdateAltref,
- kTemporalUpdateLast,
- kTemporalUpdateAltref,
- kTemporalUpdateGolden,
- kTemporalUpdateNone,
- };
- int expected_temporal_idx[16] =
- { 0, 2, 1, 2, 0, 2, 1, 2, 0, 2, 1, 2, 0, 2, 1, 2 };
-
- bool expected_layer_sync[16] =
- { false, true, true, false, false, false, false, false,
- false, true, true, false, false, false, false, false };
-
- for (int i = 0; i < 16; ++i) {
- EXPECT_EQ(expected_flags[i], tl.EncodeFlags());
- tl.PopulateCodecSpecific(false, &vp8_info);
- EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
- bool expected_sync = expected_layer_sync[i];
- EXPECT_EQ(expected_sync, vp8_info.layerSync);
- }
-}
-
-TEST(TemporalLayersTest, 4Layers) {
- TemporalLayers tl(4);
- vpx_codec_enc_cfg_t cfg;
- CodecSpecificInfoVP8 vp8_info;
- tl.ConfigureBitrates(500, &cfg);
- int expected_flags[16] = {
- kTemporalUpdateLast,
- kTemporalUpdateNone,
- kTemporalUpdateAltrefWithoutDependency,
- kTemporalUpdateNone,
- kTemporalUpdateGoldenWithoutDependency,
- kTemporalUpdateNone,
- kTemporalUpdateAltref,
- kTemporalUpdateNone,
- kTemporalUpdateLast,
- kTemporalUpdateNone,
- kTemporalUpdateAltref,
- kTemporalUpdateNone,
- kTemporalUpdateGolden,
- kTemporalUpdateNone,
- kTemporalUpdateAltref,
- kTemporalUpdateNone,
- };
- int expected_temporal_idx[16] =
- { 0, 3, 2, 3, 1, 3, 2, 3, 0, 3, 2, 3, 1, 3, 2, 3 };
-
- bool expected_layer_sync[16] =
- { false, true, true, true, true, true, false, true,
- false, true, false, true, false, true, false, true };
-
- for (int i = 0; i < 16; ++i) {
- EXPECT_EQ(expected_flags[i], tl.EncodeFlags());
- tl.PopulateCodecSpecific(false, &vp8_info);
- EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
- bool expected_sync = expected_layer_sync[i];
- EXPECT_EQ(expected_sync, vp8_info.layerSync);
- }
-}
-
-TEST(TemporalLayersTest, KeyFrame) {
- TemporalLayers tl(3);
- vpx_codec_enc_cfg_t cfg;
- CodecSpecificInfoVP8 vp8_info;
- tl.ConfigureBitrates(500, &cfg);
-
- int expected_flags[8] = {
- kTemporalUpdateLast,
- kTemporalUpdateAltrefWithoutDependency,
- kTemporalUpdateGoldenWithoutDependency,
- kTemporalUpdateAltref,
- kTemporalUpdateLast,
- kTemporalUpdateAltref,
- kTemporalUpdateGolden,
- kTemporalUpdateNone,
- };
- int expected_temporal_idx[8] =
- { 0, 0, 0, 0, 0, 0, 0, 2};
-
- bool expected_layer_sync[8] =
- { false, true, true, false, false, false, false, false };
-
- for (int i = 0; i < 7; ++i) {
- EXPECT_EQ(expected_flags[i], tl.EncodeFlags());
- tl.PopulateCodecSpecific(true, &vp8_info);
- EXPECT_EQ(expected_temporal_idx[i], vp8_info.temporalIdx);
- bool expected_sync = expected_layer_sync[i];
- EXPECT_EQ(expected_sync, vp8_info.layerSync);
- }
- EXPECT_EQ(expected_flags[7], tl.EncodeFlags());
- tl.PopulateCodecSpecific(false, &vp8_info);
- EXPECT_EQ(expected_temporal_idx[7], vp8_info.temporalIdx);
- bool expected_sync = expected_layer_sync[7];
- EXPECT_EQ(expected_sync, vp8_info.layerSync);
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/vp8.cc b/trunk/src/modules/video_coding/codecs/vp8/main/source/vp8.cc
deleted file mode 100644
index a477237..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/source/vp8.cc
+++ /dev/null
@@ -1,977 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- *
- * This file contains the WEBRTC VP8 wrapper implementation
- *
- */
-#include "vp8.h"
-
-#include <stdlib.h>
-#include <string.h>
-#include <time.h>
-
-#include "module_common_types.h"
-#include "reference_picture_selection.h"
-#include "temporal_layers.h"
-#include "tick_util.h"
-#include "vpx/vpx_encoder.h"
-#include "vpx/vpx_decoder.h"
-#include "vpx/vp8cx.h"
-#include "vpx/vp8dx.h"
-
-enum { kVp8ErrorPropagationTh = 30 };
-
-namespace webrtc
-{
-
-VP8Encoder* VP8Encoder::Create() {
- return new VP8Encoder();
-}
-
-VP8Encoder::VP8Encoder()
- : encoded_image_(),
- encoded_complete_callback_(NULL),
- inited_(false),
- timestamp_(0),
- picture_id_(0),
- feedback_mode_(false),
- cpu_speed_(-6), // default value
- rc_max_intra_target_(0),
- token_partitions_(VP8_ONE_TOKENPARTITION),
- rps_(new ReferencePictureSelection),
-#if WEBRTC_LIBVPX_VERSION >= 971
- temporal_layers_(NULL),
-#endif
- encoder_(NULL),
- config_(NULL),
- raw_(NULL) {
- uint32_t seed = static_cast<uint32_t>(TickTime::MillisecondTimestamp());
- srand(seed);
-}
-
-VP8Encoder::~VP8Encoder() {
- Release();
- delete rps_;
-}
-
-int VP8Encoder::Release() {
- if (encoded_image_._buffer != NULL) {
- delete [] encoded_image_._buffer;
- encoded_image_._buffer = NULL;
- }
- if (encoder_ != NULL) {
- if (vpx_codec_destroy(encoder_)) {
- return WEBRTC_VIDEO_CODEC_MEMORY;
- }
- delete encoder_;
- encoder_ = NULL;
- }
- if (config_ != NULL) {
- delete config_;
- config_ = NULL;
- }
- if (raw_ != NULL) {
- vpx_img_free(raw_);
- delete raw_;
- raw_ = NULL;
- }
-#if WEBRTC_LIBVPX_VERSION >= 971
- if (temporal_layers_ != NULL) {
- delete temporal_layers_;
- temporal_layers_ = NULL;
- }
-#endif
- inited_ = false;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int VP8Encoder::SetRates(uint32_t new_bitrate_kbit, uint32_t new_framerate) {
- if (!inited_) {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
- if (encoder_->err) {
- return WEBRTC_VIDEO_CODEC_ERROR;
- }
- if (new_framerate < 1) {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- // update bit rate
- if (codec_.maxBitrate > 0 && new_bitrate_kbit > codec_.maxBitrate) {
- new_bitrate_kbit = codec_.maxBitrate;
- }
- config_->rc_target_bitrate = new_bitrate_kbit; // in kbit/s
-
-#if WEBRTC_LIBVPX_VERSION >= 971
- if (temporal_layers_) {
- temporal_layers_->ConfigureBitrates(new_bitrate_kbit, config_);
- }
-#endif
- codec_.maxFramerate = new_framerate;
-
- // update encoder context
- if (vpx_codec_enc_config_set(encoder_, config_)) {
- return WEBRTC_VIDEO_CODEC_ERROR;
- }
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int VP8Encoder::InitEncode(const VideoCodec* inst,
- int number_of_cores,
- uint32_t /*max_payload_size*/) {
- if (inst == NULL) {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- if (inst->maxFramerate < 1) {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- // allow zero to represent an unspecified maxBitRate
- if (inst->maxBitrate > 0 && inst->startBitrate > inst->maxBitrate) {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- if (inst->width < 1 || inst->height < 1) {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- if (number_of_cores < 1) {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- feedback_mode_ = inst->codecSpecific.VP8.feedbackModeOn;
-
- int retVal = Release();
- if (retVal < 0) {
- return retVal;
- }
- if (encoder_ == NULL) {
- encoder_ = new vpx_codec_ctx_t;
- }
- if (config_ == NULL) {
- config_ = new vpx_codec_enc_cfg_t;
- }
- if (raw_ == NULL) {
- raw_ = new vpx_image_t;
- }
- timestamp_ = 0;
-
- codec_ = *inst;
-
-#if WEBRTC_LIBVPX_VERSION >= 971
- if (inst->codecSpecific.VP8.numberOfTemporalLayers > 1) {
- assert(temporal_layers_ == NULL);
- temporal_layers_ =
- new TemporalLayers(inst->codecSpecific.VP8.numberOfTemporalLayers);
- }
-#endif
- // random start 16 bits is enough.
- picture_id_ = static_cast<uint16_t>(rand()) & 0x7FFF;
-
- // allocate memory for encoded image
- if (encoded_image_._buffer != NULL) {
- delete [] encoded_image_._buffer;
- }
- encoded_image_._size = (3 * codec_.width * codec_.height) >> 1;
- encoded_image_._buffer = new uint8_t[encoded_image_._size];
- encoded_image_._completeFrame = true;
-
- vpx_img_alloc(raw_, IMG_FMT_I420, codec_.width, codec_.height, 1);
- // populate encoder configuration with default values
- if (vpx_codec_enc_config_default(vpx_codec_vp8_cx(), config_, 0)) {
- return WEBRTC_VIDEO_CODEC_ERROR;
- }
- config_->g_w = codec_.width;
- config_->g_h = codec_.height;
- config_->rc_target_bitrate = inst->startBitrate; // in kbit/s
-
-#if WEBRTC_LIBVPX_VERSION >= 971
- if (temporal_layers_) {
- temporal_layers_->ConfigureBitrates(inst->startBitrate, config_);
- }
-#endif
- // setting the time base of the codec
- config_->g_timebase.num = 1;
- config_->g_timebase.den = 90000;
-
- // Set the error resilience mode according to user settings.
- switch (inst->codecSpecific.VP8.resilience) {
- case kResilienceOff:
- config_->g_error_resilient = 0;
-#if WEBRTC_LIBVPX_VERSION >= 971
- if (temporal_layers_) {
- // Must be on for temporal layers.
- config_->g_error_resilient = 1;
- }
-#endif
- break;
- case kResilientStream:
- config_->g_error_resilient = 1; // TODO(holmer): Replace with
- // VPX_ERROR_RESILIENT_DEFAULT when we
- // drop support for libvpx 9.6.0.
- break;
- case kResilientFrames:
-#ifdef INDEPENDENT_PARTITIONS
- config_->g_error_resilient = VPX_ERROR_RESILIENT_DEFAULT |
- VPX_ERROR_RESILIENT_PARTITIONS;
- break;
-#else
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER; // Not supported
-#endif
- }
- config_->g_lag_in_frames = 0; // 0- no frame lagging
-
- // Determining number of threads based on the image size
- if (codec_.width * codec_.height > 704 * 576 && number_of_cores > 1) {
- // 2 threads when larger than 4CIF
- config_->g_threads = 2;
- } else {
- config_->g_threads = 1;
- }
-
- // rate control settings
- config_->rc_dropframe_thresh = 30;
- config_->rc_end_usage = VPX_CBR;
- config_->g_pass = VPX_RC_ONE_PASS;
- config_->rc_resize_allowed = 0;
- config_->rc_min_quantizer = 8;
- config_->rc_max_quantizer = 56;
- config_->rc_undershoot_pct = 100;
- config_->rc_overshoot_pct = 15;
- config_->rc_buf_initial_sz = 500;
- config_->rc_buf_optimal_sz = 600;
- config_->rc_buf_sz = 1000;
- // set the maximum target size of any key-frame.
- rc_max_intra_target_ = MaxIntraTarget(config_->rc_buf_optimal_sz);
-
- if (feedback_mode_) {
- // Disable periodic key frames if we get feedback from the decoder
- // through SLI and RPSI.
- config_->kf_mode = VPX_KF_DISABLED;
- } else {
- config_->kf_mode = VPX_KF_AUTO;
- config_->kf_max_dist = 3000;
- }
- switch (inst->codecSpecific.VP8.complexity) {
- case kComplexityHigh:
- cpu_speed_ = -5;
- break;
- case kComplexityHigher:
- cpu_speed_ = -4;
- break;
- case kComplexityMax:
- cpu_speed_ = -3;
- break;
- default:
- cpu_speed_ = -6;
- break;
- }
- rps_->Init();
- return InitAndSetControlSettings();
-}
-
-int VP8Encoder::InitAndSetControlSettings() {
- vpx_codec_flags_t flags = 0;
- // TODO(holmer): We should make a smarter decision on the number of
- // partitions. Eight is probably not the optimal number for low resolution
- // video.
-
-#if WEBRTC_LIBVPX_VERSION >= 971
- flags |= VPX_CODEC_USE_OUTPUT_PARTITION;
-#endif
- if (vpx_codec_enc_init(encoder_, vpx_codec_vp8_cx(), config_, flags)) {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
- vpx_codec_control(encoder_, VP8E_SET_STATIC_THRESHOLD, 800);
- vpx_codec_control(encoder_, VP8E_SET_CPUUSED, cpu_speed_);
- vpx_codec_control(encoder_, VP8E_SET_TOKEN_PARTITIONS,
- static_cast<vp8e_token_partitions>(token_partitions_));
- vpx_codec_control(encoder_, VP8E_SET_NOISE_SENSITIVITY, 2);
-#if WEBRTC_LIBVPX_VERSION >= 971
- vpx_codec_control(encoder_, VP8E_SET_MAX_INTRA_BITRATE_PCT,
- rc_max_intra_target_);
-#endif
- inited_ = true;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-uint32_t VP8Encoder::MaxIntraTarget(uint32_t optimalBuffersize) {
- // Set max to the optimal buffer level (normalized by target BR),
- // and scaled by a scalePar.
- // Max target size = scalePar * optimalBufferSize * targetBR[Kbps].
- // This values is presented in percentage of perFrameBw:
- // perFrameBw = targetBR[Kbps] * 1000 / frameRate.
- // The target in % is as follows:
-
- float scalePar = 0.5;
- uint32_t targetPct = optimalBuffersize * scalePar * codec_.maxFramerate / 10;
-
- // Don't go below 3 times the per frame bandwidth.
- const uint32_t minIntraTh = 300;
- return (targetPct < minIntraTh) ? minIntraTh: targetPct;
-}
-
-int VP8Encoder::Encode(const RawImage& input_image,
- const CodecSpecificInfo* codec_specific_info,
- const VideoFrameType* frame_types) {
- if (!inited_) {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
- if (input_image._buffer == NULL) {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- if (encoded_complete_callback_ == NULL) {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
- // image in vpx_image_t format
- raw_->planes[PLANE_Y] = input_image._buffer;
- raw_->planes[PLANE_U] = &input_image._buffer[codec_.height * codec_.width];
- raw_->planes[PLANE_V] =
- &input_image._buffer[codec_.height * codec_.width * 5 >> 2];
-
- int flags = 0;
-#if WEBRTC_LIBVPX_VERSION >= 971
- if (temporal_layers_) {
- flags |= temporal_layers_->EncodeFlags();
- }
-#endif
- bool send_keyframe = frame_types && (*frame_types == kKeyFrame);
- if (send_keyframe) {
- // Key frame request from caller.
- // Will update both golden and alt-ref.
- flags = VPX_EFLAG_FORCE_KF;
- } else if (feedback_mode_ && codec_specific_info) {
- // Handle RPSI and SLI messages and set up the appropriate encode flags.
- bool sendRefresh = false;
- if (codec_specific_info->codecType == kVideoCodecVP8) {
- if (codec_specific_info->codecSpecific.VP8.hasReceivedRPSI) {
- rps_->ReceivedRPSI(
- codec_specific_info->codecSpecific.VP8.pictureIdRPSI);
- }
- if (codec_specific_info->codecSpecific.VP8.hasReceivedSLI) {
- sendRefresh = rps_->ReceivedSLI(input_image._timeStamp);
- }
- }
- flags = rps_->EncodeFlags(picture_id_, sendRefresh, input_image._timeStamp);
- }
-
- // TODO(holmer): Ideally the duration should be the timestamp diff of this
- // frame and the next frame to be encoded, which we don't have. Instead we
- // would like to use the duration of the previous frame. Unfortunately the
- // rate control seems to be off with that setup. Using the average input
- // frame rate to calculate an average duration for now.
- assert(codec_.maxFramerate > 0);
- uint32_t duration = 90000 / codec_.maxFramerate;
- if (vpx_codec_encode(encoder_, raw_, timestamp_, duration, flags,
- VPX_DL_REALTIME)) {
- return WEBRTC_VIDEO_CODEC_ERROR;
- }
- timestamp_ += duration;
-
-#if WEBRTC_LIBVPX_VERSION >= 971
- return GetEncodedPartitions(input_image);
-#else
- return GetEncodedFrame(input_image);
-#endif
-}
-
-void VP8Encoder::PopulateCodecSpecific(CodecSpecificInfo* codec_specific,
- const vpx_codec_cx_pkt& pkt) {
- assert(codec_specific != NULL);
- codec_specific->codecType = kVideoCodecVP8;
- CodecSpecificInfoVP8 *vp8Info = &(codec_specific->codecSpecific.VP8);
- vp8Info->pictureId = picture_id_;
- vp8Info->simulcastIdx = 0;
- vp8Info->keyIdx = kNoKeyIdx; // TODO(hlundin) populate this
- vp8Info->nonReference = (pkt.data.frame.flags & VPX_FRAME_IS_DROPPABLE) != 0;
-#if WEBRTC_LIBVPX_VERSION >= 971
- if (temporal_layers_) {
- temporal_layers_->PopulateCodecSpecific(
- (pkt.data.frame.flags & VPX_FRAME_IS_KEY) ? true : false, vp8Info);
- } else {
-#endif
- vp8Info->temporalIdx = kNoTemporalIdx;
- vp8Info->layerSync = false;
- vp8Info->tl0PicIdx = kNoTl0PicIdx;
-#if WEBRTC_LIBVPX_VERSION >= 971
- }
-#endif
- picture_id_ = (picture_id_ + 1) & 0x7FFF; // prepare next
-}
-
-int VP8Encoder::GetEncodedFrame(const RawImage& input_image) {
- vpx_codec_iter_t iter = NULL;
- encoded_image_._frameType = kDeltaFrame;
- const vpx_codec_cx_pkt_t *pkt= vpx_codec_get_cx_data(encoder_, &iter);
- if (pkt == NULL) {
- if (!encoder_->err) {
- // dropped frame
- return WEBRTC_VIDEO_CODEC_OK;
- } else {
- return WEBRTC_VIDEO_CODEC_ERROR;
- }
- } else if (pkt->kind == VPX_CODEC_CX_FRAME_PKT) {
- CodecSpecificInfo codecSpecific;
- PopulateCodecSpecific(&codecSpecific, *pkt);
-
- assert(pkt->data.frame.sz <= encoded_image_._size);
- memcpy(encoded_image_._buffer, pkt->data.frame.buf, pkt->data.frame.sz);
- encoded_image_._length = uint32_t(pkt->data.frame.sz);
- encoded_image_._encodedHeight = raw_->h;
- encoded_image_._encodedWidth = raw_->w;
-
- // check if encoded frame is a key frame
- if (pkt->data.frame.flags & VPX_FRAME_IS_KEY) {
- encoded_image_._frameType = kKeyFrame;
- rps_->EncodedKeyFrame(picture_id_);
- }
-
- if (encoded_image_._length > 0) {
- encoded_image_._timeStamp = input_image._timeStamp;
-
- // Figure out where partition boundaries are located.
- RTPFragmentationHeader fragInfo;
- fragInfo.VerifyAndAllocateFragmentationHeader(2);
- // two partitions: 1st and 2nd
-
- // First partition
- fragInfo.fragmentationOffset[0] = 0;
- uint8_t *firstByte = encoded_image_._buffer;
- uint32_t tmpSize = (firstByte[2] << 16) | (firstByte[1] << 8)
- | firstByte[0];
- fragInfo.fragmentationLength[0] = (tmpSize >> 5) & 0x7FFFF;
- fragInfo.fragmentationPlType[0] = 0; // not known here
- fragInfo.fragmentationTimeDiff[0] = 0;
-
- // Second partition
- fragInfo.fragmentationOffset[1] = fragInfo.fragmentationLength[0];
- fragInfo.fragmentationLength[1] = encoded_image_._length -
- fragInfo.fragmentationLength[0];
- fragInfo.fragmentationPlType[1] = 0; // not known here
- fragInfo.fragmentationTimeDiff[1] = 0;
-
- encoded_complete_callback_->Encoded(encoded_image_, &codecSpecific,
- &fragInfo);
- }
- return WEBRTC_VIDEO_CODEC_OK;
- }
- return WEBRTC_VIDEO_CODEC_ERROR;
-}
-
-#if WEBRTC_LIBVPX_VERSION >= 971
-int VP8Encoder::GetEncodedPartitions(const RawImage& input_image) {
- vpx_codec_iter_t iter = NULL;
- int part_idx = 0;
- encoded_image_._length = 0;
- encoded_image_._frameType = kDeltaFrame;
- RTPFragmentationHeader frag_info;
- frag_info.VerifyAndAllocateFragmentationHeader((1 << token_partitions_) + 1);
- CodecSpecificInfo codec_specific;
-
- const vpx_codec_cx_pkt_t *pkt = NULL;
- while ((pkt = vpx_codec_get_cx_data(encoder_, &iter)) != NULL) {
- switch(pkt->kind) {
- case VPX_CODEC_CX_FRAME_PKT: {
- memcpy(&encoded_image_._buffer[encoded_image_._length],
- pkt->data.frame.buf,
- pkt->data.frame.sz);
- frag_info.fragmentationOffset[part_idx] = encoded_image_._length;
- frag_info.fragmentationLength[part_idx] = pkt->data.frame.sz;
- frag_info.fragmentationPlType[part_idx] = 0; // not known here
- frag_info.fragmentationTimeDiff[part_idx] = 0;
- encoded_image_._length += pkt->data.frame.sz;
- assert(encoded_image_._length <= encoded_image_._size);
- ++part_idx;
- break;
- }
- default: {
- break;
- }
- }
- // End of frame
- if ((pkt->data.frame.flags & VPX_FRAME_IS_FRAGMENT) == 0) {
- // check if encoded frame is a key frame
- if (pkt->data.frame.flags & VPX_FRAME_IS_KEY) {
- encoded_image_._frameType = kKeyFrame;
- rps_->EncodedKeyFrame(picture_id_);
- }
- PopulateCodecSpecific(&codec_specific, *pkt);
- break;
- }
- }
- if (encoded_image_._length > 0) {
- encoded_image_._timeStamp = input_image._timeStamp;
- encoded_image_._encodedHeight = raw_->h;
- encoded_image_._encodedWidth = raw_->w;
- encoded_complete_callback_->Encoded(encoded_image_, &codec_specific,
- &frag_info);
- }
- return WEBRTC_VIDEO_CODEC_OK;
-}
-#endif
-
-int VP8Encoder::SetChannelParameters(uint32_t /*packet_loss*/, int rtt) {
- rps_->SetRtt(rtt);
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int VP8Encoder::RegisterEncodeCompleteCallback(
- EncodedImageCallback* callback) {
- encoded_complete_callback_ = callback;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-VP8Decoder* VP8Decoder::Create() {
- return new VP8Decoder();
-}
-
-VP8Decoder::VP8Decoder()
- : decode_complete_callback_(NULL),
- inited_(false),
- feedback_mode_(false),
- decoder_(NULL),
- last_keyframe_(),
- image_format_(VPX_IMG_FMT_NONE),
- ref_frame_(NULL),
- propagation_cnt_(-1),
- latest_keyframe_complete_(false),
- mfqe_enabled_(false) {
-}
-
-VP8Decoder::~VP8Decoder() {
- inited_ = true; // in order to do the actual release
- Release();
-}
-
-int VP8Decoder::Reset() {
- if (!inited_) {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
- InitDecode(&codec_, 1);
- propagation_cnt_ = -1;
- latest_keyframe_complete_ = false;
- mfqe_enabled_ = false;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int VP8Decoder::InitDecode(const VideoCodec* inst, int number_of_cores) {
- int ret_val = Release();
- if (ret_val < 0 ) {
- return ret_val;
- }
- if (decoder_ == NULL) {
- decoder_ = new vpx_dec_ctx_t;
- }
- if (inst && inst->codecType == kVideoCodecVP8) {
- feedback_mode_ = inst->codecSpecific.VP8.feedbackModeOn;
- }
- vpx_codec_dec_cfg_t cfg;
- // Setting number of threads to a constant value (1)
- cfg.threads = 1;
- cfg.h = cfg.w = 0; // set after decode
-
- vpx_codec_flags_t flags = 0;
-#if WEBRTC_LIBVPX_VERSION >= 971
- flags = VPX_CODEC_USE_ERROR_CONCEALMENT | VPX_CODEC_USE_POSTPROC;
-#ifdef INDEPENDENT_PARTITIONS
- flags |= VPX_CODEC_USE_INPUT_PARTITION;
-#endif
-#endif
-
- if (vpx_codec_dec_init(decoder_, vpx_codec_vp8_dx(), &cfg, flags)) {
- return WEBRTC_VIDEO_CODEC_MEMORY;
- }
-
-#if WEBRTC_LIBVPX_VERSION >= 971
- vp8_postproc_cfg_t ppcfg;
- // Disable deblocking for now due to uninitialized memory being returned.
- ppcfg.post_proc_flag = 0;
- // Strength of deblocking filter. Valid range:[0,16]
- //ppcfg.deblocking_level = 3;
- vpx_codec_control(decoder_, VP8_SET_POSTPROC, &ppcfg);
-#endif
-
- // Save VideoCodec instance for later; mainly for duplicating the decoder.
- codec_ = *inst;
- propagation_cnt_ = -1;
- latest_keyframe_complete_ = false;
-
- inited_ = true;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int VP8Decoder::Decode(const EncodedImage& input_image,
- bool missing_frames,
- const RTPFragmentationHeader* fragmentation,
- const CodecSpecificInfo* codec_specific_info,
- int64_t /*render_time_ms*/) {
- if (!inited_) {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
- if (decode_complete_callback_ == NULL) {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
- if (input_image._buffer == NULL && input_image._length > 0) {
- // Reset to avoid requesting key frames too often.
- if (propagation_cnt_ > 0)
- propagation_cnt_ = 0;
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
-
-#ifdef INDEPENDENT_PARTITIONS
- if (fragmentation == NULL) {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
-#endif
-
-#if WEBRTC_LIBVPX_VERSION >= 971
- if (!mfqe_enabled_ && codec_specific_info &&
- codec_specific_info->codecSpecific.VP8.temporalIdx > 0) {
- // Enable MFQE if we are receiving layers.
- // temporalIdx is set in the jitter buffer according to what the RTP
- // header says.
- mfqe_enabled_ = true;
- vp8_postproc_cfg_t ppcfg;
- ppcfg.post_proc_flag = VP8_MFQE;
- vpx_codec_control(decoder_, VP8_SET_POSTPROC, &ppcfg);
- }
-#endif
-
- // Restrict error propagation using key frame requests. Disabled when
- // the feedback mode is enabled (RPS).
- // Reset on a key frame refresh.
- if (!feedback_mode_) {
- if (input_image._frameType == kKeyFrame && input_image._completeFrame)
- propagation_cnt_ = -1;
- // Start count on first loss.
- else if ((!input_image._completeFrame || missing_frames) &&
- propagation_cnt_ == -1)
- propagation_cnt_ = 0;
- if (propagation_cnt_ >= 0)
- propagation_cnt_++;
- }
-
- vpx_dec_iter_t iter = NULL;
- vpx_image_t* img;
- int ret;
-
- // Check for missing frames.
- if (missing_frames) {
- // Call decoder with zero data length to signal missing frames.
- if (vpx_codec_decode(decoder_, NULL, 0, 0, VPX_DL_REALTIME)) {
- // Reset to avoid requesting key frames too often.
- if (propagation_cnt_ > 0)
- propagation_cnt_ = 0;
- return WEBRTC_VIDEO_CODEC_ERROR;
- }
- img = vpx_codec_get_frame(decoder_, &iter);
- iter = NULL;
- }
-
-#ifdef INDEPENDENT_PARTITIONS
- if (DecodePartitions(inputImage, fragmentation)) {
- // Reset to avoid requesting key frames too often.
- if (propagation_cnt_ > 0) {
- propagation_cnt_ = 0;
- }
- return WEBRTC_VIDEO_CODEC_ERROR;
- }
-#else
- uint8_t* buffer = input_image._buffer;
- if (input_image._length == 0) {
- buffer = NULL; // Triggers full frame concealment.
- }
- if (vpx_codec_decode(decoder_,
- buffer,
- input_image._length,
- 0,
- VPX_DL_REALTIME)) {
- // Reset to avoid requesting key frames too often.
- if (propagation_cnt_ > 0)
- propagation_cnt_ = 0;
- return WEBRTC_VIDEO_CODEC_ERROR;
- }
-#endif
-
- // Store encoded frame if key frame. (Used in Copy method.)
- if (input_image._frameType == kKeyFrame && input_image._buffer != NULL) {
- const uint32_t bytes_to_copy = input_image._length;
- if (last_keyframe_._size < bytes_to_copy) {
- delete [] last_keyframe_._buffer;
- last_keyframe_._buffer = NULL;
- last_keyframe_._size = 0;
- }
-
- uint8_t* temp_buffer = last_keyframe_._buffer; // Save buffer ptr.
- uint32_t temp_size = last_keyframe_._size; // Save size.
- last_keyframe_ = input_image; // Shallow copy.
- last_keyframe_._buffer = temp_buffer; // Restore buffer ptr.
- last_keyframe_._size = temp_size; // Restore buffer size.
- if (!last_keyframe_._buffer) {
- // Allocate memory.
- last_keyframe_._size = bytes_to_copy;
- last_keyframe_._buffer = new uint8_t[last_keyframe_._size];
- }
- // Copy encoded frame.
- memcpy(last_keyframe_._buffer, input_image._buffer, bytes_to_copy);
- last_keyframe_._length = bytes_to_copy;
- }
-
- img = vpx_codec_get_frame(decoder_, &iter);
- ret = ReturnFrame(img, input_image._timeStamp);
- if (ret != 0) {
- // Reset to avoid requesting key frames too often.
- if (ret < 0 && propagation_cnt_ > 0)
- propagation_cnt_ = 0;
- return ret;
- }
- if (feedback_mode_) {
- // Whenever we receive an incomplete key frame all reference buffers will
- // be corrupt. If that happens we must request new key frames until we
- // decode a complete.
- if (input_image._frameType == kKeyFrame)
- latest_keyframe_complete_ = input_image._completeFrame;
- if (!latest_keyframe_complete_)
- return WEBRTC_VIDEO_CODEC_ERROR;
-
- // Check for reference updates and last reference buffer corruption and
- // signal successful reference propagation or frame corruption to the
- // encoder.
- int reference_updates = 0;
- if (vpx_codec_control(decoder_, VP8D_GET_LAST_REF_UPDATES,
- &reference_updates)) {
- // Reset to avoid requesting key frames too often.
- if (propagation_cnt_ > 0)
- propagation_cnt_ = 0;
- return WEBRTC_VIDEO_CODEC_ERROR;
- }
- int corrupted = 0;
- if (vpx_codec_control(decoder_, VP8D_GET_FRAME_CORRUPTED, &corrupted)) {
- // Reset to avoid requesting key frames too often.
- if (propagation_cnt_ > 0)
- propagation_cnt_ = 0;
- return WEBRTC_VIDEO_CODEC_ERROR;
- }
- int16_t picture_id = -1;
- if (codec_specific_info) {
- picture_id = codec_specific_info->codecSpecific.VP8.pictureId;
- }
- if (picture_id > -1) {
- if (((reference_updates & VP8_GOLD_FRAME) ||
- (reference_updates & VP8_ALTR_FRAME)) && !corrupted) {
- decode_complete_callback_->ReceivedDecodedReferenceFrame(picture_id);
- }
- decode_complete_callback_->ReceivedDecodedFrame(picture_id);
- }
- if (corrupted) {
- // we can decode but with artifacts
- return WEBRTC_VIDEO_CODEC_REQUEST_SLI;
- }
- }
- // Check Vs. threshold
- if (propagation_cnt_ > kVp8ErrorPropagationTh) {
- // Reset to avoid requesting key frames too often.
- propagation_cnt_ = 0;
- return WEBRTC_VIDEO_CODEC_ERROR;
- }
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int VP8Decoder::DecodePartitions(
- const EncodedImage& input_image,
- const RTPFragmentationHeader* fragmentation) {
- for (int i = 0; i < fragmentation->fragmentationVectorSize; ++i) {
- const uint8_t* partition = input_image._buffer +
- fragmentation->fragmentationOffset[i];
- const uint32_t partition_length =
- fragmentation->fragmentationLength[i];
- if (vpx_codec_decode(decoder_,
- partition,
- partition_length,
- 0,
- VPX_DL_REALTIME)) {
- return WEBRTC_VIDEO_CODEC_ERROR;
- }
- }
- // Signal end of frame data. If there was no frame data this will trigger
- // a full frame concealment.
- if (vpx_codec_decode(decoder_, NULL, 0, 0, VPX_DL_REALTIME))
- return WEBRTC_VIDEO_CODEC_ERROR;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int VP8Decoder::ReturnFrame(const vpx_image_t* img, uint32_t timestamp) {
- if (img == NULL) {
- // Decoder OK and NULL image => No show frame
- return WEBRTC_VIDEO_CODEC_NO_OUTPUT;
- }
-
- // Allocate memory for decoded image
- uint32_t required_size = (3 * img->d_h * img->d_w) >> 1;
- if (required_size > decoded_image_._size) {
- delete [] decoded_image_._buffer;
- decoded_image_._buffer = NULL;
- }
- if (decoded_image_._buffer == NULL) {
- decoded_image_._size = required_size;
- decoded_image_._buffer = new uint8_t[decoded_image_._size];
- }
- uint8_t* buf;
- uint32_t pos = 0;
- uint32_t plane, y;
-
- for (plane = 0; plane < 3; plane++) {
- unsigned int width = (plane ? (img->d_w + 1) >> 1 : img->d_w);
- unsigned int height = (plane ? (img->d_h + 1) >> 1 : img->d_h);
- buf = img->planes[plane];
- for(y = 0; y < height; y++) {
- memcpy(&decoded_image_._buffer[pos], buf, width);
- pos += width;
- buf += img->stride[plane];
- }
- }
-
- // Set image parameters
- decoded_image_._height = img->d_h;
- decoded_image_._width = img->d_w;
- decoded_image_._length = (3 * img->d_h * img->d_w) >> 1;
- decoded_image_._timeStamp = timestamp;
- int ret = decode_complete_callback_->Decoded(decoded_image_);
- if (ret != 0)
- return ret;
-
- // Remember image format for later
- image_format_ = img->fmt;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int VP8Decoder::RegisterDecodeCompleteCallback(
- DecodedImageCallback* callback) {
- decode_complete_callback_ = callback;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-int VP8Decoder::Release() {
- if (decoded_image_._buffer != NULL) {
- delete [] decoded_image_._buffer;
- decoded_image_._buffer = NULL;
- }
- if (last_keyframe_._buffer != NULL) {
- delete [] last_keyframe_._buffer;
- last_keyframe_._buffer = NULL;
- }
- if (decoder_ != NULL) {
- if(vpx_codec_destroy(decoder_)) {
- return WEBRTC_VIDEO_CODEC_MEMORY;
- }
- delete decoder_;
- decoder_ = NULL;
- }
- if (ref_frame_ != NULL) {
- vpx_img_free(&ref_frame_->img);
- delete ref_frame_;
- ref_frame_ = NULL;
- }
- inited_ = false;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-VideoDecoder* VP8Decoder::Copy() {
- // Sanity checks.
- if (!inited_) {
- // Not initialized.
- assert(false);
- return NULL;
- }
- if (decoded_image_._buffer == NULL) {
- // Nothing has been decoded before; cannot clone.
- return NULL;
- }
- if (last_keyframe_._buffer == NULL) {
- // Cannot clone if we have no key frame to start with.
- return NULL;
- }
- // Create a new VideoDecoder object
- VP8Decoder *copy = new VP8Decoder;
-
- // Initialize the new decoder
- if (copy->InitDecode(&codec_, 1) != WEBRTC_VIDEO_CODEC_OK) {
- delete copy;
- return NULL;
- }
- // Inject last key frame into new decoder.
- if (vpx_codec_decode(copy->decoder_, last_keyframe_._buffer,
- last_keyframe_._length, NULL, VPX_DL_REALTIME)) {
- delete copy;
- return NULL;
- }
- // Allocate memory for reference image copy
- assert(decoded_image_._width > 0);
- assert(decoded_image_._height > 0);
- assert(image_format_ > VPX_IMG_FMT_NONE);
- // Check if frame format has changed.
- if (ref_frame_ &&
- (decoded_image_._width != ref_frame_->img.d_w ||
- decoded_image_._height != ref_frame_->img.d_h ||
- image_format_ != ref_frame_->img.fmt)) {
- vpx_img_free(&ref_frame_->img);
- delete ref_frame_;
- ref_frame_ = NULL;
- }
-
-
- if (!ref_frame_) {
- ref_frame_ = new vpx_ref_frame_t;
-
- if (!vpx_img_alloc(&ref_frame_->img,
- static_cast<vpx_img_fmt_t>(image_format_),
- decoded_image_._width, decoded_image_._height, 1)) {
- assert(false);
- delete copy;
- return NULL;
- }
- }
- const vpx_ref_frame_type_t type_vec[] = { VP8_LAST_FRAME, VP8_GOLD_FRAME,
- VP8_ALTR_FRAME };
- for (uint32_t ix = 0;
- ix < sizeof(type_vec) / sizeof(vpx_ref_frame_type_t); ++ix) {
- ref_frame_->frame_type = type_vec[ix];
- if (CopyReference(copy) < 0) {
- delete copy;
- return NULL;
- }
- }
- // Copy all member variables (that are not set in initialization).
- copy->feedback_mode_ = feedback_mode_;
- copy->image_format_ = image_format_;
- copy->last_keyframe_ = last_keyframe_; // Shallow copy.
- // Allocate memory. (Discard copied _buffer pointer.)
- copy->last_keyframe_._buffer = new uint8_t[last_keyframe_._size];
- memcpy(copy->last_keyframe_._buffer, last_keyframe_._buffer,
- last_keyframe_._length);
-
- // Initialize decoded_image_.
- copy->decoded_image_ = decoded_image_; // Shallow copy
- copy->decoded_image_._buffer = NULL;
- if (decoded_image_._size) {
- copy->decoded_image_._buffer = new uint8_t[decoded_image_._size];
- }
- return static_cast<VideoDecoder*>(copy);
-}
-
-int VP8Decoder::CopyReference(VP8Decoder* copyTo) {
- // The type of frame to copy should be set in ref_frame_->frame_type
- // before the call to this function.
- if (vpx_codec_control(decoder_, VP8_COPY_REFERENCE, ref_frame_)
- != VPX_CODEC_OK) {
- return -1;
- }
- if (vpx_codec_control(copyTo->decoder_, VP8_SET_REFERENCE, ref_frame_)
- != VPX_CODEC_OK) {
- return -1;
- }
- return 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/source/vp8.gypi b/trunk/src/modules/video_coding/codecs/vp8/main/source/vp8.gypi
deleted file mode 100644
index 89d8c78..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/source/vp8.gypi
+++ /dev/null
@@ -1,121 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'webrtc_vp8',
- 'type': '<(library)',
- 'dependencies': [
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
- ],
- 'include_dirs': [
- '../interface',
- '<(webrtc_root)/common_video/interface',
- '<(webrtc_root)/modules/video_coding/codecs/interface',
- '<(webrtc_root)/modules/interface',
- ],
- 'conditions': [
- ['build_with_chromium==1', {
- 'dependencies': [
- '<(webrtc_root)/../libvpx/libvpx.gyp:libvpx',
- ],
- 'defines': [
- 'WEBRTC_LIBVPX_VERSION=960' # Bali
- ],
- },{
- 'dependencies': [
- '<(webrtc_root)/../third_party/libvpx/libvpx.gyp:libvpx',
- ],
- 'defines': [
- 'WEBRTC_LIBVPX_VERSION=971' # Cayuga
- ],
- 'sources': [
- 'temporal_layers.h',
- 'temporal_layers.cc',
- ],
- }],
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- '<(webrtc_root)/common_video/interface',
- '<(webrtc_root)/modules/video_coding/codecs/interface',
- ],
- },
- 'sources': [
- 'reference_picture_selection.h',
- 'reference_picture_selection.cc',
- '../interface/vp8.h',
- '../interface/vp8_common_types.h',
- 'vp8.cc',
- ],
- },
- ], # targets
- # Exclude the test target when building with chromium.
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'vp8_test',
- 'type': 'executable',
- 'dependencies': [
- 'test_framework',
- 'webrtc_vp8',
- '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/../test/test.gyp:test_support',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'sources': [
- # header files
- '../test/benchmark.h',
- '../test/dual_decoder_test.h',
- '../test/normal_async_test.h',
- '../test/packet_loss_test.h',
- '../test/rps_test.h',
- '../test/unit_test.h',
-
- # source files
- '../test/benchmark.cc',
- '../test/dual_decoder_test.cc',
- '../test/normal_async_test.cc',
- '../test/packet_loss_test.cc',
- '../test/rps_test.cc',
- '../test/tester.cc',
- '../test/unit_test.cc',
- ],
- },
- {
- 'target_name': 'vp8_unittests',
- 'type': 'executable',
- 'dependencies': [
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../third_party/libvpx/libvpx.gyp:libvpx',
- 'webrtc_vp8',
- ],
- 'include_dirs': [
- '<(webrtc_root)/../third_party/libvpx/source/libvpx',
- ],
- 'sources': [
- 'reference_picture_selection_unittest.cc',
- 'temporal_layers_unittest.cc',
- ],
- },
- ], # targets
- }], # build_with_chromium
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/benchmark.cc b/trunk/src/modules/video_coding/codecs/vp8/main/test/benchmark.cc
deleted file mode 100644
index 4fc0e25..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/test/benchmark.cc
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "benchmark.h"
-#include "testsupport/fileutils.h"
-#include "vp8.h"
-
-using namespace webrtc;
-
-VP8Benchmark::VP8Benchmark()
- : Benchmark("VP8Benchmark", "VP8 benchmark over a range of test cases",
- webrtc::test::OutputPath() + "VP8Benchmark.txt", "VP8") {
-}
-
-VP8Benchmark::VP8Benchmark(std::string name, std::string description)
- : Benchmark(name, description,
- webrtc::test::OutputPath() + "VP8Benchmark.txt",
- "VP8") {
-}
-
-VP8Benchmark::VP8Benchmark(std::string name, std::string description,
- std::string resultsFileName)
- : Benchmark(name, description, resultsFileName, "VP8") {
-}
-
-VideoEncoder* VP8Benchmark::GetNewEncoder() {
- return VP8Encoder::Create();
-}
-
-VideoDecoder* VP8Benchmark::GetNewDecoder() {
- return VP8Decoder::Create();
-}
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/benchmark.h b/trunk/src/modules/video_coding/codecs/vp8/main/test/benchmark.h
deleted file mode 100644
index 5143de3..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/test/benchmark.h
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_BENCHMARK_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_BENCHMARK_H_
-
-#include "../../../test_framework/benchmark.h"
-
-class VP8Benchmark : public Benchmark
-{
-public:
- VP8Benchmark();
- VP8Benchmark(std::string name, std::string description);
- VP8Benchmark(std::string name, std::string description, std::string resultsFileName);
-
-protected:
- virtual webrtc::VideoEncoder* GetNewEncoder();
- virtual webrtc::VideoDecoder* GetNewDecoder();
-};
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_BENCHMARK_H_
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/dual_decoder_test.cc b/trunk/src/modules/video_coding/codecs/vp8/main/test/dual_decoder_test.cc
deleted file mode 100644
index 478b034..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/test/dual_decoder_test.cc
+++ /dev/null
@@ -1,221 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "dual_decoder_test.h"
-
-#include <assert.h>
-#include <string.h> // memcmp
-#include <time.h>
-
-#include "testsupport/fileutils.h"
-
-VP8DualDecoderTest::VP8DualDecoderTest(float bitRate)
-:
-VP8NormalAsyncTest(bitRate)
-{
- _decoder2 = NULL;
-}
-
-VP8DualDecoderTest::VP8DualDecoderTest()
-:
-VP8NormalAsyncTest("VP8 Dual Decoder Test", "Tests VP8 dual decoder", 1),
-_decoder2(NULL)
-{}
-
-VP8DualDecoderTest::~VP8DualDecoderTest()
-{
- if(_decoder2)
- {
- _decoder2->Release();
- delete _decoder2;
- }
-
- _decodedVideoBuffer2.Free();
-}
-
-void
-VP8DualDecoderTest::Perform()
-{
- _inname = webrtc::test::ProjectRootPath() + "resources/foreman_cif.yuv";
- CodecSettings(352, 288, 30, _bitRate);
- Setup();
- _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
- _decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
- _decodedVideoBuffer2.VerifyAndAllocate(_lengthSourceFrame);
- if(_encoder->InitEncode(&_inst, 4, 1460) < 0)
- {
- exit(EXIT_FAILURE);
- }
- _decoder->InitDecode(&_inst,1);
-
- FrameQueue frameQueue;
- VideoEncodeCompleteCallback encCallback(_encodedFile, &frameQueue, *this);
- DualDecoderCompleteCallback decCallback(&_decodedVideoBuffer);
- DualDecoderCompleteCallback decCallback2(&_decodedVideoBuffer2);
- _encoder->RegisterEncodeCompleteCallback(&encCallback);
- _decoder->RegisterDecodeCompleteCallback(&decCallback);
- if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
- {
- exit(EXIT_FAILURE);
- }
- _totalEncodeTime = _totalDecodeTime = 0;
- _totalEncodePipeTime = _totalDecodePipeTime = 0;
- bool complete = false;
- _framecnt = 0;
- _encFrameCnt = 0;
- _decFrameCnt = 0;
- _sumEncBytes = 0;
- _lengthEncFrame = 0;
- double starttime = clock()/(double)CLOCKS_PER_SEC;
- while (!complete)
- {
- if (_encFrameCnt == 10)
- {
- // initialize second decoder and copy state
- _decoder2 = static_cast<webrtc::VP8Decoder *>(_decoder->Copy());
- assert(_decoder2 != NULL);
- _decoder2->RegisterDecodeCompleteCallback(&decCallback2);
- }
- CodecSpecific_InitBitrate();
- complete = Encode();
- if (!frameQueue.Empty() || complete)
- {
- while (!frameQueue.Empty())
- {
- _frameToDecode =
- static_cast<FrameQueueTuple *>(frameQueue.PopFrame());
- int lost = DoPacketLoss();
- if (lost == 2)
- {
- // Lost the whole frame, continue
- _missingFrames = true;
- delete _frameToDecode;
- _frameToDecode = NULL;
- continue;
- }
- int ret = Decode(lost);
- delete _frameToDecode;
- _frameToDecode = NULL;
- if (ret < 0)
- {
- fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
- exit(EXIT_FAILURE);
- }
- else if (ret == 0)
- {
- _framecnt++;
- }
- else
- {
- fprintf(stderr,
- "\n\nPositive return value from decode!\n\n");
- }
- }
- }
- }
- double endtime = clock()/(double)CLOCKS_PER_SEC;
- double totalExecutionTime = endtime - starttime;
- printf("Total execution time: %.1f s\n", totalExecutionTime);
- _sumEncBytes = encCallback.EncodedBytes();
- double actualBitRate = ActualBitRate(_encFrameCnt) / 1000.0;
- double avgEncTime = _totalEncodeTime / _encFrameCnt;
- double avgDecTime = _totalDecodeTime / _decFrameCnt;
- printf("Actual bitrate: %f kbps\n", actualBitRate);
- printf("Average encode time: %.1f ms\n", 1000 * avgEncTime);
- printf("Average decode time: %.1f ms\n", 1000 * avgDecTime);
- printf("Average encode pipeline time: %.1f ms\n",
- 1000 * _totalEncodePipeTime / _encFrameCnt);
- printf("Average decode pipeline time: %.1f ms\n",
- 1000 * _totalDecodePipeTime / _decFrameCnt);
- printf("Number of encoded frames: %u\n", _encFrameCnt);
- printf("Number of decoded frames: %u\n", _decFrameCnt);
- (*_log) << "Actual bitrate: " << actualBitRate << " kbps\tTarget: " <<
- _bitRate << " kbps" << std::endl;
- (*_log) << "Average encode time: " << avgEncTime << " s" << std::endl;
- (*_log) << "Average decode time: " << avgDecTime << " s" << std::endl;
- _encoder->Release();
- _decoder->Release();
- Teardown();
-}
-
-
-int
-VP8DualDecoderTest::Decode(int lossValue)
-{
- _sumEncBytes += _frameToDecode->_frame->GetLength();
- webrtc::EncodedImage encodedImage;
- VideoEncodedBufferToEncodedImage(*(_frameToDecode->_frame), encodedImage);
- encodedImage._completeFrame = !lossValue;
- _decodeCompleteTime = 0;
- _decodeTimes[encodedImage._timeStamp] = clock()/(double)CLOCKS_PER_SEC;
- int ret = _decoder->Decode(encodedImage, _missingFrames, NULL,
- _frameToDecode->_codecSpecificInfo);
- // second decoder
- if (_decoder2)
- {
- int ret2 = _decoder2->Decode(encodedImage, _missingFrames, NULL,
- _frameToDecode->_codecSpecificInfo,
- 0 /* dummy */);
-
- // check return values
- if (ret < 0 || ret2 < 0 || ret2 != ret)
- {
- exit(EXIT_FAILURE);
- }
-
- // compare decoded images
- if (!CheckIfBitExact(_decodedVideoBuffer.GetBuffer(),
- _decodedVideoBuffer.GetLength(),
- _decodedVideoBuffer2.GetBuffer(), _decodedVideoBuffer.GetLength()))
- {
- fprintf(stderr,"\n\nClone output different from master.\n\n");
- exit(EXIT_FAILURE);
- }
-
- }
-
- _missingFrames = false;
- return ret;
-}
-
-
-bool
-VP8DualDecoderTest::CheckIfBitExact(const void* ptrA, unsigned int aLengthBytes,
- const void* ptrB, unsigned int bLengthBytes)
-{
- if (aLengthBytes != bLengthBytes)
- {
- return false;
- }
-
- return memcmp(ptrA, ptrB, aLengthBytes) == 0;
-}
-
-WebRtc_Word32 DualDecoderCompleteCallback::Decoded(webrtc::RawImage& image)
-{
- _decodedVideoBuffer->VerifyAndAllocate(image._length);
- _decodedVideoBuffer->CopyBuffer(image._length, image._buffer);
- _decodedVideoBuffer->SetWidth(image._width);
- _decodedVideoBuffer->SetHeight(image._height);
- _decodedVideoBuffer->SetTimeStamp(image._timeStamp);
- _decodeComplete = true;
- return 0;
-}
-
-bool DualDecoderCompleteCallback::DecodeComplete()
-{
- if (_decodeComplete)
- {
- _decodeComplete = false;
- return true;
- }
- return false;
-}
-
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/dual_decoder_test.h b/trunk/src/modules/video_coding/codecs/vp8/main/test/dual_decoder_test.h
deleted file mode 100644
index 4af4e3e..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/test/dual_decoder_test.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_DUAL_DECODER_TEST_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_DUAL_DECODER_TEST_H_
-
-#include "vp8.h"
-#include "normal_async_test.h"
-
-class DualDecoderCompleteCallback;
-
-class VP8DualDecoderTest : public VP8NormalAsyncTest
-{
-public:
- VP8DualDecoderTest(float bitRate);
- VP8DualDecoderTest();
- virtual ~VP8DualDecoderTest();
- virtual void Perform();
-protected:
- VP8DualDecoderTest(std::string name, std::string description,
- unsigned int testNo)
- : VP8NormalAsyncTest(name, description, testNo) {}
- virtual int Decode(int lossValue = 0);
-
- webrtc::VP8Decoder* _decoder2;
- TestVideoBuffer _decodedVideoBuffer2;
- static bool CheckIfBitExact(const void *ptrA, unsigned int aLengthBytes,
- const void *ptrB, unsigned int bLengthBytes);
-private:
-};
-
-class DualDecoderCompleteCallback : public webrtc::DecodedImageCallback
-{
-public:
- DualDecoderCompleteCallback(TestVideoBuffer* buffer)
- : _decodedVideoBuffer(buffer), _decodeComplete(false) {}
- WebRtc_Word32 Decoded(webrtc::RawImage& decodedImage);
- bool DecodeComplete();
-private:
- TestVideoBuffer* _decodedVideoBuffer;
- bool _decodeComplete;
-};
-
-
-#endif
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/normal_async_test.cc b/trunk/src/modules/video_coding/codecs/vp8/main/test/normal_async_test.cc
deleted file mode 100644
index 9ed508b..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/test/normal_async_test.cc
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "normal_async_test.h"
-
-using namespace webrtc;
-
-VP8NormalAsyncTest::VP8NormalAsyncTest(WebRtc_UWord32 bitRate) :
- NormalAsyncTest("VP8 Normal Test 1", "Tests VP8 normal execution", bitRate, 1),
- _hasReceivedRPSI(false)
-{
-}
-
-VP8NormalAsyncTest::VP8NormalAsyncTest(WebRtc_UWord32 bitRate, unsigned int testNo):
- NormalAsyncTest("VP8 Normal Test 1", "Tests VP8 normal execution", bitRate, testNo),
- _hasReceivedRPSI(false)
-{
-}
-
-void
-VP8NormalAsyncTest::CodecSettings(int width, int height, WebRtc_UWord32 frameRate /*=30*/, WebRtc_UWord32 bitRate /*=0*/)
-{
- if (bitRate > 0)
- {
- _bitRate = bitRate;
-
- }else if (_bitRate == 0)
- {
- _bitRate = 600;
- }
- _inst.codecType = kVideoCodecVP8;
- _inst.codecSpecific.VP8.feedbackModeOn = true;
- _inst.codecSpecific.VP8.pictureLossIndicationOn = true;
- _inst.codecSpecific.VP8.complexity = kComplexityNormal;
- _inst.maxFramerate = (unsigned char)frameRate;
- _inst.startBitrate = _bitRate;
- _inst.maxBitrate = 8000;
- _inst.width = width;
- _inst.height = height;
-}
-
-void
-VP8NormalAsyncTest::CodecSpecific_InitBitrate()
-{
- if (_bitRate == 0)
- {
- _encoder->SetRates(600, _inst.maxFramerate);
- }else
- {
- _encoder->SetRates(_bitRate, _inst.maxFramerate);
- }
-}
-
-WebRtc_Word32
-VP8NormalAsyncTest::ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId)
-{
- _pictureIdRPSI = pictureId;
- _hasReceivedRPSI = true;
- return 0;
-}
-
-CodecSpecificInfo*
-VP8NormalAsyncTest::CreateEncoderSpecificInfo() const
-{
- CodecSpecificInfo* vp8CodecSpecificInfo = new CodecSpecificInfo();
- vp8CodecSpecificInfo->codecType = kVideoCodecVP8;
- vp8CodecSpecificInfo->codecSpecific.VP8.hasReceivedRPSI = _hasReceivedRPSI;
- vp8CodecSpecificInfo->codecSpecific.VP8.pictureIdRPSI = _pictureIdRPSI;
- vp8CodecSpecificInfo->codecSpecific.VP8.hasReceivedSLI = _hasReceivedSLI;
- vp8CodecSpecificInfo->codecSpecific.VP8.pictureIdSLI = _pictureIdSLI;
-
- _hasReceivedSLI = false;
- _hasReceivedRPSI = false;
-
- return vp8CodecSpecificInfo;
-}
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/normal_async_test.h b/trunk/src/modules/video_coding/codecs/vp8/main/test/normal_async_test.h
deleted file mode 100644
index f49c812..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/test/normal_async_test.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_NORMAL_ASYNC_TEST_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_NORMAL_ASYNC_TEST_H_
-
-#include "../../../test_framework/normal_async_test.h"
-
-class VP8NormalAsyncTest : public NormalAsyncTest
-{
-public:
- VP8NormalAsyncTest(WebRtc_UWord32 bitRate);
- VP8NormalAsyncTest(WebRtc_UWord32 bitRate, unsigned int testNo);
- VP8NormalAsyncTest() : NormalAsyncTest("VP8 Normal Test 1", "Tests VP8 normal execution", 1) {}
-protected:
- VP8NormalAsyncTest(std::string name, std::string description, unsigned int testNo) : NormalAsyncTest(name, description, testNo) {}
- virtual void CodecSpecific_InitBitrate();
- virtual void CodecSettings(int width, int height, WebRtc_UWord32 frameRate=30, WebRtc_UWord32 bitRate=0);
- virtual webrtc::CodecSpecificInfo* CreateEncoderSpecificInfo() const;
- virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId);
-private:
- mutable bool _hasReceivedRPSI;
- WebRtc_UWord64 _pictureIdRPSI;
-};
-
-#endif
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/packet_loss_test.cc b/trunk/src/modules/video_coding/codecs/vp8/main/test/packet_loss_test.cc
deleted file mode 100644
index 69e028a..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/test/packet_loss_test.cc
+++ /dev/null
@@ -1,74 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "packet_loss_test.h"
-#include <cassert>
-
-VP8PacketLossTest::VP8PacketLossTest()
-:
-PacketLossTest("VP8PacketLossTest", "Encode, remove lost packets, decode")
-{
-}
-
-VP8PacketLossTest::VP8PacketLossTest(std::string name, std::string description)
-:
-PacketLossTest(name, description)
-{
-}
-
-VP8PacketLossTest::VP8PacketLossTest(double lossRate,
- bool useNack,
- int rttFrames)
-:
-PacketLossTest("VP8PacketLossTest", "Encode, remove lost packets, decode",
- lossRate, useNack, rttFrames)
-{
-}
-
-int VP8PacketLossTest::ByteLoss(int size, unsigned char* /* pkg */, int bytesToLose)
-{
- int retLength = size - bytesToLose;
- if (retLength < 4)
- {
- retLength = 4;
- }
- return retLength;
-}
-
-WebRtc_Word32
-VP8PacketLossTest::ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId)
-{
- _pictureIdRPSI = pictureId;
- _hasReceivedRPSI = true;
- return 0;
-}
-
-webrtc::CodecSpecificInfo*
-VP8PacketLossTest::CreateEncoderSpecificInfo() const
-{
- webrtc::CodecSpecificInfo* vp8CodecSpecificInfo =
- new webrtc::CodecSpecificInfo();
- vp8CodecSpecificInfo->codecType = webrtc::kVideoCodecVP8;
- vp8CodecSpecificInfo->codecSpecific.VP8.hasReceivedRPSI = _hasReceivedRPSI;
- vp8CodecSpecificInfo->codecSpecific.VP8.pictureIdRPSI = _pictureIdRPSI;
- vp8CodecSpecificInfo->codecSpecific.VP8.hasReceivedSLI = _hasReceivedSLI;
- vp8CodecSpecificInfo->codecSpecific.VP8.pictureIdSLI = _pictureIdSLI;
-
- _hasReceivedSLI = false;
- _hasReceivedRPSI = false;
-
- return vp8CodecSpecificInfo;
-}
-
-bool VP8PacketLossTest::PacketLoss(double lossRate, int numLosses) {
- if (numLosses)
- return true;
- return RandUniform() < lossRate;
-}
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/packet_loss_test.h b/trunk/src/modules/video_coding/codecs/vp8/main/test/packet_loss_test.h
deleted file mode 100644
index 262ee2f..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/test/packet_loss_test.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_PACKET_LOSS_TEST_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_PACKET_LOSS_TEST_H_
-
-#include "../../../test_framework/packet_loss_test.h"
-
-class VP8PacketLossTest : public PacketLossTest
-{
-public:
- VP8PacketLossTest();
- VP8PacketLossTest(double lossRate, bool useNack, int rttFrames);
-
-protected:
- VP8PacketLossTest(std::string name, std::string description);
- virtual int ByteLoss(int size, unsigned char *pkg, int bytesToLose);
- WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId);
- // |lossRate| is the probability of packet loss between 0 and 1.
- // |numLosses| is the number of packets already lost in the current frame.
- virtual bool PacketLoss(double lossRate, int numLosses);
-
- webrtc::CodecSpecificInfo* CreateEncoderSpecificInfo() const;
-
-};
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_PACKET_LOSS_TEST_H_
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/rps_test.cc b/trunk/src/modules/video_coding/codecs/vp8/main/test/rps_test.cc
deleted file mode 100644
index cd1c42e..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/test/rps_test.cc
+++ /dev/null
@@ -1,311 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rps_test.h"
-
-#include <assert.h>
-#include <string.h> // memcmp
-#include <time.h>
-
-#include "vp8.h"
-
-VP8RpsTest::VP8RpsTest(float bitRate)
- : VP8NormalAsyncTest(bitRate),
- decoder2_(webrtc::VP8Decoder::Create()),
- sli_(false) {
-}
-
-VP8RpsTest::VP8RpsTest()
- : VP8NormalAsyncTest("VP8 Reference Picture Selection Test",
- "VP8 Reference Picture Selection Test", 1),
- decoder2_(webrtc::VP8Decoder::Create()),
- sli_(false) {
-}
-
-VP8RpsTest::~VP8RpsTest() {
- if (decoder2_) {
- decoder2_->Release();
- delete decoder2_;
- }
- decoded_frame2_.Free();
-}
-
-void VP8RpsTest::Perform() {
- _inname = "test/testFiles/foreman_cif.yuv";
- CodecSettings(352, 288, 30, _bitRate);
- Setup();
- _inputVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
- _decodedVideoBuffer.VerifyAndAllocate(_lengthSourceFrame);
- decoded_frame2_.VerifyAndAllocate(_lengthSourceFrame);
-
- // Enable RPS functionality
- _inst.codecSpecific.VP8.pictureLossIndicationOn = true;
- _inst.codecSpecific.VP8.feedbackModeOn = true;
-
- if(_encoder->InitEncode(&_inst, 4, 1460) < 0)
- exit(EXIT_FAILURE);
-
- _decoder->InitDecode(&_inst,1);
- decoder2_->InitDecode(&_inst,1);
-
- FrameQueue frameQueue;
- VideoEncodeCompleteCallback encCallback(_encodedFile, &frameQueue, *this);
- RpsDecodeCompleteCallback decCallback(&_decodedVideoBuffer);
- RpsDecodeCompleteCallback decCallback2(&decoded_frame2_);
- _encoder->RegisterEncodeCompleteCallback(&encCallback);
- _decoder->RegisterDecodeCompleteCallback(&decCallback);
- decoder2_->RegisterDecodeCompleteCallback(&decCallback2);
-
- if (SetCodecSpecificParameters() != WEBRTC_VIDEO_CODEC_OK)
- exit(EXIT_FAILURE);
-
- _totalEncodeTime = _totalDecodeTime = 0;
- _totalEncodePipeTime = _totalDecodePipeTime = 0;
- bool complete = false;
- _framecnt = 0;
- _encFrameCnt = 0;
- _decFrameCnt = 0;
- _sumEncBytes = 0;
- _lengthEncFrame = 0;
- double starttime = clock()/(double)CLOCKS_PER_SEC;
- while (!complete) {
- CodecSpecific_InitBitrate();
- complete = EncodeRps(&decCallback2);
- if (!frameQueue.Empty() || complete) {
- while (!frameQueue.Empty()) {
- _frameToDecode =
- static_cast<FrameQueueTuple *>(frameQueue.PopFrame());
- int lost = DoPacketLoss();
- if (lost == 2) {
- // Lost the whole frame, continue
- _missingFrames = true;
- delete _frameToDecode;
- _frameToDecode = NULL;
- continue;
- }
- int ret = Decode(lost);
- delete _frameToDecode;
- _frameToDecode = NULL;
- if (ret < 0) {
- fprintf(stderr,"\n\nError in decoder: %d\n\n", ret);
- exit(EXIT_FAILURE);
- }
- else if (ret == 0) {
- _framecnt++;
- }
- else {
- fprintf(stderr,
- "\n\nPositive return value from decode!\n\n");
- }
- }
- }
- }
- double endtime = clock()/(double)CLOCKS_PER_SEC;
- double totalExecutionTime = endtime - starttime;
- printf("Total execution time: %.1f s\n", totalExecutionTime);
- _sumEncBytes = encCallback.EncodedBytes();
- double actualBitRate = ActualBitRate(_encFrameCnt) / 1000.0;
- double avgEncTime = _totalEncodeTime / _encFrameCnt;
- double avgDecTime = _totalDecodeTime / _decFrameCnt;
- printf("Actual bitrate: %f kbps\n", actualBitRate);
- printf("Average encode time: %.1f ms\n", 1000 * avgEncTime);
- printf("Average decode time: %.1f ms\n", 1000 * avgDecTime);
- printf("Average encode pipeline time: %.1f ms\n",
- 1000 * _totalEncodePipeTime / _encFrameCnt);
- printf("Average decode pipeline time: %.1f ms\n",
- 1000 * _totalDecodePipeTime / _decFrameCnt);
- printf("Number of encoded frames: %u\n", _encFrameCnt);
- printf("Number of decoded frames: %u\n", _decFrameCnt);
- (*_log) << "Actual bitrate: " << actualBitRate << " kbps\tTarget: " <<
- _bitRate << " kbps" << std::endl;
- (*_log) << "Average encode time: " << avgEncTime << " s" << std::endl;
- (*_log) << "Average decode time: " << avgDecTime << " s" << std::endl;
- _encoder->Release();
- _decoder->Release();
- Teardown();
-}
-
-bool VP8RpsTest::EncodeRps(RpsDecodeCompleteCallback* decodeCallback) {
- _lengthEncFrame = 0;
- size_t bytes_read = fread(_sourceBuffer, 1, _lengthSourceFrame, _sourceFile);
- if (bytes_read < _lengthSourceFrame)
- return true;
- _inputVideoBuffer.CopyBuffer(_lengthSourceFrame, _sourceBuffer);
- _inputVideoBuffer.SetTimeStamp((unsigned int)
- (_encFrameCnt * 9e4 / _inst.maxFramerate));
- _inputVideoBuffer.SetWidth(_inst.width);
- _inputVideoBuffer.SetHeight(_inst.height);
- webrtc::RawImage rawImage;
- VideoBufferToRawImage(_inputVideoBuffer, rawImage);
- if (feof(_sourceFile) != 0) {
- return true;
- }
- _encodeCompleteTime = 0;
- _encodeTimes[rawImage._timeStamp] = tGetTime();
- webrtc::VideoFrameType frameType = webrtc::kDeltaFrame;
-
- webrtc::CodecSpecificInfo* codecSpecificInfo = CreateEncoderSpecificInfo();
- codecSpecificInfo->codecSpecific.VP8.pictureIdRPSI =
- decodeCallback->LastDecodedRefPictureId(
- &codecSpecificInfo->codecSpecific.VP8.hasReceivedRPSI);
- if (sli_) {
- codecSpecificInfo->codecSpecific.VP8.pictureIdSLI =
- decodeCallback->LastDecodedPictureId();
- codecSpecificInfo->codecSpecific.VP8.hasReceivedSLI = true;
- sli_ = false;
- }
- printf("Encoding: %u\n", _framecnt);
- int ret = _encoder->Encode(rawImage, codecSpecificInfo, &frameType);
- if (ret < 0)
- printf("Failed to encode: %u\n", _framecnt);
-
- if (codecSpecificInfo != NULL) {
- delete codecSpecificInfo;
- codecSpecificInfo = NULL;
- }
- if (_encodeCompleteTime > 0) {
- _totalEncodeTime += _encodeCompleteTime -
- _encodeTimes[rawImage._timeStamp];
- }
- else {
- _totalEncodeTime += tGetTime() - _encodeTimes[rawImage._timeStamp];
- }
- return false;
-}
-
-//#define FRAME_LOSS 1
-
-int VP8RpsTest::Decode(int lossValue) {
- _sumEncBytes += _frameToDecode->_frame->GetLength();
- webrtc::EncodedImage encodedImage;
- VideoEncodedBufferToEncodedImage(*(_frameToDecode->_frame), encodedImage);
- encodedImage._completeFrame = !lossValue;
- _decodeCompleteTime = 0;
- _decodeTimes[encodedImage._timeStamp] = clock()/(double)CLOCKS_PER_SEC;
- int ret = _decoder->Decode(encodedImage, _missingFrames, NULL,
- _frameToDecode->_codecSpecificInfo);
- // Drop every 10th frame for the second decoder
-#if FRAME_LOSS
- if (_framecnt == 0 || _framecnt % 10 != 0) {
- printf("Decoding: %u\n", _framecnt);
- if (_framecnt > 1 && (_framecnt - 1) % 10 == 0)
- _missingFrames = true;
-#else
- if (true) {
- if (_framecnt > 0 && _framecnt % 10 == 0) {
- encodedImage._length = std::rand() % encodedImage._length;
- printf("Decoding with loss: %u\n", _framecnt);
- }
- else
- printf("Decoding: %u\n", _framecnt);
-#endif
- int ret2 = decoder2_->Decode(encodedImage, _missingFrames, NULL,
- _frameToDecode->_codecSpecificInfo,
- 0 /* dummy */);
-
- // check return values
- if (ret < 0 || ret2 < 0) {
- return -1;
- } else if (ret2 == WEBRTC_VIDEO_CODEC_ERR_REQUEST_SLI ||
- ret2 == WEBRTC_VIDEO_CODEC_REQUEST_SLI) {
- sli_ = true;
- }
-
- // compare decoded images
-#if FRAME_LOSS
- if (!_missingFrames) {
- if (!CheckIfBitExact(_decodedVideoBuffer.GetBuffer(),
- _decodedVideoBuffer.GetLength(),
- decoded_frame2_.GetBuffer(), _decodedVideoBuffer.GetLength())) {
- fprintf(stderr,"\n\nRPS decoder different from master: %u\n\n",
- _framecnt);
- return -1;
- }
- }
-#else
- if (_framecnt > 0 && _framecnt % 10 != 0) {
- if (!CheckIfBitExact(_decodedVideoBuffer.GetBuffer(),
- _decodedVideoBuffer.GetLength(),
- decoded_frame2_.GetBuffer(), _decodedVideoBuffer.GetLength())) {
- fprintf(stderr,"\n\nRPS decoder different from master: %u\n\n",
- _framecnt);
- return -1;
- }
- }
-#endif
- }
-#if FRAME_LOSS
- else
- printf("Dropping %u\n", _framecnt);
-#endif
- _missingFrames = false;
- return 0;
-}
-
-
-bool
-VP8RpsTest::CheckIfBitExact(const void* ptrA, unsigned int aLengthBytes,
- const void* ptrB, unsigned int bLengthBytes) {
- if (aLengthBytes != bLengthBytes)
- return false;
- return memcmp(ptrA, ptrB, aLengthBytes) == 0;
-}
-
-RpsDecodeCompleteCallback::RpsDecodeCompleteCallback(TestVideoBuffer* buffer)
- : decoded_frame_(buffer),
- decode_complete_(false),
- last_decoded_picture_id_(0),
- last_decoded_ref_picture_id_(0),
- updated_ref_picture_id_(false) {
-}
-
-WebRtc_Word32 RpsDecodeCompleteCallback::Decoded(webrtc::RawImage& image) {
- decoded_frame_->VerifyAndAllocate(image._length);
- decoded_frame_->CopyBuffer(image._length, image._buffer);
- decoded_frame_->SetWidth(image._width);
- decoded_frame_->SetHeight(image._height);
- decoded_frame_->SetTimeStamp(image._timeStamp);
- decode_complete_ = true;
- return 0;
-}
-
-bool RpsDecodeCompleteCallback::DecodeComplete() {
- if (decode_complete_)
- {
- decode_complete_ = false;
- return true;
- }
- return false;
-}
-
-WebRtc_Word32 RpsDecodeCompleteCallback::ReceivedDecodedReferenceFrame(
- const WebRtc_UWord64 picture_id) {
- last_decoded_ref_picture_id_ = picture_id & 0x7FFF;
- updated_ref_picture_id_ = true;
- return 0;
-}
-
-WebRtc_Word32 RpsDecodeCompleteCallback::ReceivedDecodedFrame(
- const WebRtc_UWord64 picture_id) {
- last_decoded_picture_id_ = picture_id & 0x3F;
- return 0;
-}
-
-WebRtc_UWord64 RpsDecodeCompleteCallback::LastDecodedPictureId() const {
- return last_decoded_picture_id_;
-}
-
-WebRtc_UWord64 RpsDecodeCompleteCallback::LastDecodedRefPictureId(
- bool *updated) {
- if (updated)
- *updated = updated_ref_picture_id_;
- updated_ref_picture_id_ = false;
- return last_decoded_ref_picture_id_;
-}
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/rps_test.h b/trunk/src/modules/video_coding/codecs/vp8/main/test/rps_test.h
deleted file mode 100644
index cbdbfa3..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/test/rps_test.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_RPS_TEST_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_RPS_TEST_H_
-
-#include "vp8.h"
-#include "normal_async_test.h"
-
-class RpsDecodeCompleteCallback;
-
-class VP8RpsTest : public VP8NormalAsyncTest {
- public:
- VP8RpsTest(float bitRate);
- VP8RpsTest();
- virtual ~VP8RpsTest();
- virtual void Perform();
- private:
- VP8RpsTest(std::string name, std::string description, unsigned int testNo)
- : VP8NormalAsyncTest(name, description, testNo) {}
- virtual bool EncodeRps(RpsDecodeCompleteCallback* decodeCallback);
- virtual int Decode(int lossValue = 0);
-
- static bool CheckIfBitExact(const void *ptrA, unsigned int aLengthBytes,
- const void *ptrB, unsigned int bLengthBytes);
-
- webrtc::VP8Decoder* decoder2_;
- TestVideoBuffer decoded_frame2_;
- bool sli_;
-};
-
-class RpsDecodeCompleteCallback : public webrtc::DecodedImageCallback {
- public:
- RpsDecodeCompleteCallback(TestVideoBuffer* buffer);
- WebRtc_Word32 Decoded(webrtc::RawImage& decodedImage);
- bool DecodeComplete();
- WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 picture_id);
- WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 picture_id);
- WebRtc_UWord64 LastDecodedPictureId() const;
- WebRtc_UWord64 LastDecodedRefPictureId(bool *updated);
-
- private:
- TestVideoBuffer* decoded_frame_;
- bool decode_complete_;
- WebRtc_UWord64 last_decoded_picture_id_;
- WebRtc_UWord64 last_decoded_ref_picture_id_;
- bool updated_ref_picture_id_;
-};
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_RPS_TEST_H_
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/tester.cc b/trunk/src/modules/video_coding/codecs/vp8/main/test/tester.cc
deleted file mode 100644
index 350d9a3..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/test/tester.cc
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <fstream>
-#include <iostream>
-#include <vector>
-
-#include "benchmark.h"
-#include "dual_decoder_test.h"
-#include "normal_async_test.h"
-#include "packet_loss_test.h"
-#include "unit_test.h"
-#include "rps_test.h"
-#include "testsupport/fileutils.h"
-#include "vp8.h"
-
-using namespace webrtc;
-
-void PopulateTests(std::vector<Test*>* tests)
-{
-// tests->push_back(new VP8RpsTest());
-// tests->push_back(new VP8UnitTest());
-// tests->push_back(new VP8DualDecoderTest());
-// tests->push_back(new VP8Benchmark());
-// tests->push_back(new VP8PacketLossTest(0.05, false, 5));
- tests->push_back(new VP8NormalAsyncTest());
-}
-
-int main()
-{
- VP8Encoder* enc;
- VP8Decoder* dec;
- std::vector<Test*> tests;
- PopulateTests(&tests);
- std::fstream log;
- std::string log_file = webrtc::test::OutputPath() + "VP8_test_log.txt";
- log.open(log_file.c_str(), std::fstream::out | std::fstream::app);
- std::vector<Test*>::iterator it;
- for (it = tests.begin() ; it < tests.end(); it++)
- {
- enc = VP8Encoder::Create();
- dec = VP8Decoder::Create();
- (*it)->SetEncoder(enc);
- (*it)->SetDecoder(dec);
- (*it)->SetLog(&log);
- (*it)->Perform();
- (*it)->Print();
- delete enc;
- delete dec;
- delete *it;
- }
- log.close();
- tests.pop_back();
- return 0;
-}
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/unit_test.cc b/trunk/src/modules/video_coding/codecs/vp8/main/test/unit_test.cc
deleted file mode 100644
index 5ba35ec..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/test/unit_test.cc
+++ /dev/null
@@ -1,161 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "unit_test.h"
-
-#include <string.h>
-
-#include "../../../test_framework/video_source.h"
-#include "gtest/gtest.h"
-#include "testsupport/fileutils.h"
-#include "vp8.h"
-
-using namespace webrtc;
-
-VP8UnitTest::VP8UnitTest()
-:
-UnitTest("VP8UnitTest", "Unit test")
-{
-}
-
-VP8UnitTest::VP8UnitTest(std::string name, std::string description)
-:
-UnitTest(name, description)
-{
-}
-
-WebRtc_UWord32
-VP8UnitTest::CodecSpecific_SetBitrate(WebRtc_UWord32 bitRate, WebRtc_UWord32 /*frameRate*/)
-{
- int rate = _encoder->SetRates(bitRate, _inst.maxFramerate);
- EXPECT_TRUE(rate >= 0);
- return rate;
-}
-
-bool
-VP8UnitTest::CheckIfBitExact(const void* ptrA, unsigned int aLengthBytes,
- const void* ptrB, unsigned int bLengthBytes)
-{
- const unsigned char* cPtrA = (const unsigned char*)ptrA;
- const unsigned char* cPtrB = (const unsigned char*)ptrB;
- // Skip picture ID before comparing
- int aSkip = PicIdLength(cPtrA);
- int bSkip = PicIdLength(cPtrB);
- return UnitTest::CheckIfBitExact(cPtrA + aSkip, aLengthBytes,
- cPtrB + bSkip, bLengthBytes);
-}
-
-int
-VP8UnitTest::PicIdLength(const unsigned char* ptr)
-{
- WebRtc_UWord8 numberOfBytes;
- WebRtc_UWord64 pictureID = 0;
- for (numberOfBytes = 0; (ptr[numberOfBytes] & 0x80) && numberOfBytes < 8; numberOfBytes++)
- {
- pictureID += ptr[numberOfBytes] & 0x7f;
- pictureID <<= 7;
- }
- pictureID += ptr[numberOfBytes] & 0x7f;
- numberOfBytes++;
- return numberOfBytes;
-}
-
-void
-VP8UnitTest::Perform()
-{
- Setup();
- FILE *outFile = NULL;
- std::string outFileName;
- VP8Encoder* enc = (VP8Encoder*)_encoder;
- VP8Decoder* dec = (VP8Decoder*)_decoder;
-
- //----- Encoder parameter tests -----
- //-- Calls before InitEncode() --
- EXPECT_EQ(enc->Release(), WEBRTC_VIDEO_CODEC_OK);
- EXPECT_EQ(enc->SetRates(_bitRate, _inst.maxFramerate),
- WEBRTC_VIDEO_CODEC_UNINITIALIZED);
-
- EXPECT_EQ(enc->SetRates(_bitRate, _inst.maxFramerate),
- WEBRTC_VIDEO_CODEC_UNINITIALIZED);
- // EXPECT_TRUE(enc->GetCodecConfigParameters(configParameters, sizeof(configParameters)) ==
- // WEBRTC_VIDEO_CODEC_UNINITIALIZED);
-
-
- VideoCodec codecInst;
- strncpy(codecInst.plName, "VP8", 31);
- codecInst.plType = 126;
- codecInst.maxBitrate = 0;
- codecInst.minBitrate = 0;
- codecInst.width = 1440;
- codecInst.height = 1080;
- codecInst.maxFramerate = 30;
- codecInst.startBitrate = 300;
- codecInst.codecSpecific.VP8.complexity = kComplexityNormal;
- EXPECT_EQ(enc->InitEncode(&codecInst, 1, 1440), WEBRTC_VIDEO_CODEC_OK);
-
-
- //-- Test two problematic level settings --
- strncpy(codecInst.plName, "VP8", 31);
- codecInst.plType = 126;
- codecInst.maxBitrate = 0;
- codecInst.minBitrate = 0;
- codecInst.width = 352;
- codecInst.height = 288;
- codecInst.maxFramerate = 30;
- codecInst.codecSpecific.VP8.complexity = kComplexityNormal;
- codecInst.startBitrate = 300;
- EXPECT_EQ(enc->InitEncode(&codecInst, 1, 1440), WEBRTC_VIDEO_CODEC_OK);
-
- // Settings not correct for this profile
- strncpy(codecInst.plName, "VP8", 31);
- codecInst.plType = 126;
- codecInst.maxBitrate = 0;
- codecInst.minBitrate = 0;
- codecInst.width = 176;
- codecInst.height = 144;
- codecInst.maxFramerate = 15;
- codecInst.codecSpecific.VP8.complexity = kComplexityNormal;
- codecInst.startBitrate = 300;
- //EXPECT_TRUE(enc->InitEncode(&codecInst, 1, 1440) == WEBRTC_VIDEO_CODEC_LEVEL_EXCEEDED);
-
- ASSERT_EQ(enc->InitEncode(&_inst, 1, 1440), WEBRTC_VIDEO_CODEC_OK);
-
-
- //-- ProcessNewBitrate() errors --
- // Bad bitrate.
- EXPECT_EQ(enc->SetRates(_inst.maxBitrate + 1, _inst.maxFramerate),
- WEBRTC_VIDEO_CODEC_OK);
-
- // Signaling not used.
-
- // Bad packetloss.
-// EXPECT_TRUE(enc->SetPacketLoss(300) < 0);
-
- //----- Decoder parameter tests -----
- //-- Calls before InitDecode() --
- EXPECT_TRUE(dec->Release() == 0);
- ASSERT_TRUE(dec->InitDecode(&_inst, 1) == WEBRTC_VIDEO_CODEC_OK);
-
- //-- SetCodecConfigParameters() errors --
- unsigned char tmpBuf[128];
- EXPECT_TRUE(dec->SetCodecConfigParameters(NULL, sizeof(tmpBuf)) == -1);
- EXPECT_TRUE(dec->SetCodecConfigParameters(tmpBuf, 1) == -1);
- // Garbage data.
- EXPECT_TRUE(dec->SetCodecConfigParameters(tmpBuf, sizeof(tmpBuf)) == -1);
-
- //----- Function tests -----
- outFileName = webrtc::test::OutputPath() + _source->GetName() + "-errResTest.yuv";
- outFile = fopen(outFileName.c_str(), "wb");
- ASSERT_TRUE(outFile != NULL);
-
- UnitTest::Perform();
- Teardown();
-
-}
diff --git a/trunk/src/modules/video_coding/codecs/vp8/main/test/unit_test.h b/trunk/src/modules/video_coding/codecs/vp8/main/test/unit_test.h
deleted file mode 100644
index e9287e6..0000000
--- a/trunk/src/modules/video_coding/codecs/vp8/main/test/unit_test.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_UNIT_TEST_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_UNIT_TEST_H_
-
-#include "../../../test_framework/unit_test.h"
-
-class VP8UnitTest : public UnitTest
-{
-public:
- VP8UnitTest();
- VP8UnitTest(std::string name, std::string description);
- virtual void Perform();
-
-protected:
- virtual WebRtc_UWord32 CodecSpecific_SetBitrate(WebRtc_UWord32 bitRate,
- WebRtc_UWord32 /*frameRate*/);
- virtual bool CheckIfBitExact(const void *ptrA, unsigned int aLengthBytes,
- const void *ptrB, unsigned int bLengthBytes);
- static int PicIdLength(const unsigned char* ptr);
-};
-
-////////////////////////////////////////////////////////////////
-// RESERVATIONS TO PASSING UNIT TEST ON VP8 CODEC //
-// Test that will not pass: //
-// 1. Check bit exact for decoded images. //
-// 2. Target bitrate - Allow a margin of 10% instead of 5% //
-// 3. Detecting errors in bit stream - NA. //
-////////////////////////////////////////////////////////////////
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODECS_VP8_UNIT_TEST_H_
diff --git a/trunk/src/modules/video_coding/main/OWNERS b/trunk/src/modules/video_coding/main/OWNERS
deleted file mode 100644
index 7183cf2..0000000
--- a/trunk/src/modules/video_coding/main/OWNERS
+++ /dev/null
@@ -1,4 +0,0 @@
-stefan@webrtc.org
-mikhal@webrtc.org
-marpan@webrtc.org
-henrik.lundin@webrtc.org
diff --git a/trunk/src/modules/video_coding/main/interface/mock/mock_vcm_callbacks.h b/trunk/src/modules/video_coding/main/interface/mock/mock_vcm_callbacks.h
deleted file mode 100644
index c84d5e7..0000000
--- a/trunk/src/modules/video_coding/main/interface/mock/mock_vcm_callbacks.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_INTERFACE_MOCK_MOCK_VCM_CALLBACKS_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_INTERFACE_MOCK_MOCK_VCM_CALLBACKS_H_
-
-#include "gmock/gmock.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-class MockVCMFrameTypeCallback : public VCMFrameTypeCallback {
- public:
- MOCK_METHOD0(RequestKeyFrame, int32_t());
- MOCK_METHOD1(SliceLossIndicationRequest,
- WebRtc_Word32(const WebRtc_UWord64 pictureId));
-};
-
-class MockPacketRequestCallback : public VCMPacketRequestCallback {
- public:
- MOCK_METHOD2(ResendPackets, int32_t(const uint16_t* sequenceNumbers,
- uint16_t length));
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_MAIN_INTERFACE_MOCK_MOCK_VCM_CALLBACKS_H_
diff --git a/trunk/src/modules/video_coding/main/interface/video_coding.h b/trunk/src/modules/video_coding/main/interface/video_coding.h
deleted file mode 100644
index 54e6742..0000000
--- a/trunk/src/modules/video_coding/main/interface/video_coding.h
+++ /dev/null
@@ -1,556 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_
-#define WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_
-
-#include "modules/interface/module.h"
-#include "modules/interface/module_common_types.h"
-#include "modules/video_coding/main/interface/video_coding_defines.h"
-
-namespace webrtc
-{
-
-class TickTimeBase;
-class VideoEncoder;
-class VideoDecoder;
-struct CodecSpecificInfo;
-
-class VideoCodingModule : public Module
-{
-public:
- enum SenderNackMode {
- kNackNone,
- kNackAll,
- kNackSelective
- };
-
- enum ReceiverRobustness {
- kNone,
- kHardNack,
- kSoftNack,
- kDualDecoder,
- kReferenceSelection
- };
-
- enum DecodeErrors {
- kNoDecodeErrors,
- kAllowDecodeErrors
- };
-
- static VideoCodingModule* Create(const WebRtc_Word32 id);
-
- static VideoCodingModule* Create(const WebRtc_Word32 id,
- TickTimeBase* clock);
-
- static void Destroy(VideoCodingModule* module);
-
- // Get number of supported codecs
- //
- // Return value : Number of supported codecs
- static WebRtc_UWord8 NumberOfCodecs();
-
- // Get supported codec settings with using id
- //
- // Input:
- // - listId : Id or index of the codec to look up
- // - codec : Memory where the codec settings will be stored
- //
- // Return value : VCM_OK, on success
- // VCM_PARAMETER_ERROR if codec not supported or id too high
- static WebRtc_Word32 Codec(const WebRtc_UWord8 listId, VideoCodec* codec);
-
- // Get supported codec settings using codec type
- //
- // Input:
- // - codecType : The codec type to get settings for
- // - codec : Memory where the codec settings will be stored
- //
- // Return value : VCM_OK, on success
- // VCM_PARAMETER_ERROR if codec not supported
- static WebRtc_Word32 Codec(VideoCodecType codecType, VideoCodec* codec);
-
- /*
- * Sender
- */
-
- // Any encoder-related state of VCM will be initialized to the
- // same state as when the VCM was created. This will not interrupt
- // or effect decoding functionality of VCM. VCM will lose all the
- // encoding-related settings by calling this function.
- // For instance, a send codec has to be registered again.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 InitializeSender() = 0;
-
- // Registers a codec to be used for encoding. Calling this
- // API multiple times overwrites any previously registered codecs.
- //
- // Input:
- // - sendCodec : Settings for the codec to be registered.
- // - numberOfCores : The number of cores the codec is allowed
- // to use.
- // - maxPayloadSize : The maximum size each payload is allowed
- // to have. Usually MTU - overhead.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 RegisterSendCodec(const VideoCodec* sendCodec,
- WebRtc_UWord32 numberOfCores,
- WebRtc_UWord32 maxPayloadSize) = 0;
-
- // API to get the current send codec in use.
- //
- // Input:
- // - currentSendCodec : Address where the sendCodec will be written.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 SendCodec(VideoCodec* currentSendCodec) const = 0;
-
- // API to get the current send codec type
- //
- // Return value : Codec type, on success.
- // kVideoCodecUnknown, on error or if no send codec is set
- virtual VideoCodecType SendCodec() const = 0;
-
- // Register an external encoder object. This can not be used together with
- // external decoder callbacks.
- //
- // Input:
- // - externalEncoder : Encoder object to be used for encoding frames inserted
- // with the AddVideoFrame API.
- // - payloadType : The payload type bound which this encoder is bound to.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 RegisterExternalEncoder(VideoEncoder* externalEncoder,
- WebRtc_UWord8 payloadType,
- bool internalSource = false) = 0;
-
- // API to get codec config parameters to be sent out-of-band to a receiver.
- //
- // Input:
- // - buffer : Memory where the codec config parameters should be written.
- // - size : Size of the memory available.
- //
- // Return value : Number of bytes written, on success.
- // < 0, on error.
- virtual WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* buffer, WebRtc_Word32 size) = 0;
-
- // API to get currently configured encoder target bitrate in kbit/s.
- //
- // Return value : 0, on success.
- // < 0, on error.
- virtual int Bitrate(unsigned int* bitrate) const = 0;
-
- // API to get currently configured encoder target frame rate.
- //
- // Return value : 0, on success.
- // < 0, on error.
- virtual int FrameRate(unsigned int* framerate) const = 0;
-
- // Sets the parameters describing the send channel. These parameters are inputs to the
- // Media Optimization inside the VCM and also specifies the target bit rate for the
- // encoder. Bit rate used by NACK should already be compensated for by the user.
- //
- // Input:
- // - availableBandWidth : Band width available for the VCM in kbit/s.
- // - lossRate : Fractions of lost packets the past second.
- // (loss rate in percent = 100 * packetLoss / 255)
- // - rtt : Current round-trip time in ms.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 SetChannelParameters(WebRtc_UWord32 availableBandWidth,
- WebRtc_UWord8 lossRate,
- WebRtc_UWord32 rtt) = 0;
-
- // Sets the parameters describing the receive channel. These parameters are inputs to the
- // Media Optimization inside the VCM.
- //
- // Input:
- // - rtt : Current round-trip time in ms.
- // with the most amount available bandwidth in a conference
- // scenario
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 SetReceiveChannelParameters(WebRtc_UWord32 rtt) = 0;
-
- // Register a transport callback which will be called to deliver the encoded data and
- // side information.
- //
- // Input:
- // - transport : The callback object to register.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 RegisterTransportCallback(VCMPacketizationCallback* transport) = 0;
-
- // Register video output information callback which will be called to deliver information
- // about the video stream produced by the encoder, for instance the average frame rate and
- // bit rate.
- //
- // Input:
- // - outputInformation : The callback object to register.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 RegisterSendStatisticsCallback(
- VCMSendStatisticsCallback* sendStats) = 0;
-
- // Register a video quality settings callback which will be called when
- // frame rate/dimensions need to be updated for video quality optimization
- //
- // Input:
- // - videoQMSettings : The callback object to register.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error
- virtual WebRtc_Word32 RegisterVideoQMCallback(VCMQMSettingsCallback* videoQMSettings) = 0;
-
- // Register a video protection callback which will be called to deliver
- // the requested FEC rate and NACK status (on/off).
- //
- // Input:
- // - protection : The callback object to register.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 RegisterProtectionCallback(VCMProtectionCallback* protection) = 0;
-
- // Enable or disable a video protection method.
- //
- // Input:
- // - videoProtection : The method to enable or disable.
- // - enable : True if the method should be enabled, false if
- // it should be disabled.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 SetVideoProtection(VCMVideoProtection videoProtection,
- bool enable) = 0;
-
- // Add one raw video frame to the encoder. This function does all the necessary
- // processing, then decides what frame type to encode, or if the frame should be
- // dropped. If the frame should be encoded it passes the frame to the encoder
- // before it returns.
- //
- // Input:
- // - videoFrame : Video frame to encode.
- // - codecSpecificInfo : Extra codec information, e.g., pre-parsed in-band signaling.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 AddVideoFrame(
- const VideoFrame& videoFrame,
- const VideoContentMetrics* contentMetrics = NULL,
- const CodecSpecificInfo* codecSpecificInfo = NULL) = 0;
-
- // Next frame encoded should be of the type frameType.
- //
- // Input:
- // - frameType : The frame type to encode next time a VideoFrame
- // is added to the module.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 FrameTypeRequest(FrameType frameType,
- WebRtc_UWord8 simulcastIdx) = 0;
-
- // Frame Dropper enable. Can be used to disable the frame dropping when the encoder
- // over-uses its bit rate. This API is designed to be used when the encoded frames
- // are supposed to be stored to an AVI file, or when the I420 codec is used and the
- // target bit rate shouldn't affect the frame rate.
- //
- // Input:
- // - enable : True to enable the setting, false to disable it.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 EnableFrameDropper(bool enable) = 0;
-
- // Sent frame counters
- virtual WebRtc_Word32 SentFrameCount(VCMFrameCount& frameCount) const = 0;
-
- /*
- * Receiver
- */
-
- // The receiver state of the VCM will be initialized to the
- // same state as when the VCM was created. This will not interrupt
- // or effect the send side functionality of VCM. VCM will lose all the
- // decoding-related settings by calling this function. All frames
- // inside the jitter buffer are flushed and the delay is reset.
- // For instance, a receive codec has to be registered again.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 InitializeReceiver() = 0;
-
- // Register possible receive codecs, can be called multiple times for different codecs.
- // The module will automatically switch between registered codecs depending on the
- // payload type of incoming frames. The actual decoder will be created when needed.
- //
- // Input:
- // - receiveCodec : Settings for the codec to be registered.
- // - numberOfCores : Number of CPU cores that the decoder is allowed to use.
- // - requireKeyFrame : Set this to true if you don't want any delta frames
- // to be decoded until the first key frame has been decoded.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 RegisterReceiveCodec(const VideoCodec* receiveCodec,
- WebRtc_Word32 numberOfCores,
- bool requireKeyFrame = false) = 0;
-
- // Register an externally defined decoder/renderer object. Can be a decoder only or a
- // decoder coupled with a renderer. Note that RegisterReceiveCodec must be called to
- // be used for decoding incoming streams.
- //
- // Input:
- // - externalDecoder : The external decoder/renderer object.
- // - payloadType : The payload type which this decoder should be
- // registered to.
- // - internalRenderTiming : True if the internal renderer (if any) of the decoder
- // object can make sure to render at a given time in ms.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 RegisterExternalDecoder(VideoDecoder* externalDecoder,
- WebRtc_UWord8 payloadType,
- bool internalRenderTiming) = 0;
-
- // Register a receive callback. Will be called whenever there is a new frame ready
- // for rendering.
- //
- // Input:
- // - receiveCallback : The callback object to be used by the module when a
- // frame is ready for rendering.
- // De-register with a NULL pointer.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 RegisterReceiveCallback(VCMReceiveCallback* receiveCallback) = 0;
-
- // Register a receive statistics callback which will be called to deliver information
- // about the video stream received by the receiving side of the VCM, for instance the
- // average frame rate and bit rate.
- //
- // Input:
- // - receiveStats : The callback object to register.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 RegisterReceiveStatisticsCallback(
- VCMReceiveStatisticsCallback* receiveStats) = 0;
-
- // Register a frame type request callback. This callback will be called when the
- // module needs to request specific frame types from the send side.
- //
- // Input:
- // - frameTypeCallback : The callback object to be used by the module when
- // requesting a specific type of frame from the send side.
- // De-register with a NULL pointer.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 RegisterFrameTypeCallback(
- VCMFrameTypeCallback* frameTypeCallback) = 0;
-
- // Register a frame storage callback. This callback will be called right before an
- // encoded frame is given to the decoder. Useful for recording the incoming video sequence.
- //
- // Input:
- // - frameStorageCallback : The callback object used by the module
- // to store a received encoded frame.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 RegisterFrameStorageCallback(
- VCMFrameStorageCallback* frameStorageCallback) = 0;
-
- // Registers a callback which is called whenever the receive side of the VCM
- // encounters holes in the packet sequence and needs packets to be retransmitted.
- //
- // Input:
- // - callback : The callback to be registered in the VCM.
- //
- // Return value : VCM_OK, on success.
- // <0, on error.
- virtual WebRtc_Word32 RegisterPacketRequestCallback(
- VCMPacketRequestCallback* callback) = 0;
-
- // Waits for the next frame in the jitter buffer to become complete
- // (waits no longer than maxWaitTimeMs), then passes it to the decoder for decoding.
- // Should be called as often as possible to get the most out of the decoder.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 Decode(WebRtc_UWord16 maxWaitTimeMs = 200) = 0;
-
- // Waits for the next frame in the dual jitter buffer to become complete
- // (waits no longer than maxWaitTimeMs), then passes it to the dual decoder
- // for decoding. This will never trigger a render callback. Should be
- // called frequently, and as long as it returns 1 it should be called again
- // as soon as possible.
- //
- // Return value : 1, if a frame was decoded
- // 0, if no frame was decoded
- // < 0, on error.
- virtual WebRtc_Word32 DecodeDualFrame(WebRtc_UWord16 maxWaitTimeMs = 200) = 0;
-
- // Decodes a frame and sets an appropriate render time in ms relative to the system time.
- // Should be used in conjunction with VCMFrameStorageCallback.
- //
- // Input:
- // - frameFromStorage : Encoded frame read from file or received through
- // the VCMFrameStorageCallback callback.
- //
- // Return value: : VCM_OK, on success
- // < 0, on error
- virtual WebRtc_Word32 DecodeFromStorage(const EncodedVideoData& frameFromStorage) = 0;
-
- // Reset the decoder state to the initial state.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 ResetDecoder() = 0;
-
- // API to get the codec which is currently used for decoding by the module.
- //
- // Input:
- // - currentReceiveCodec : Settings for the codec to be registered.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 ReceiveCodec(VideoCodec* currentReceiveCodec) const = 0;
-
- // API to get the codec type currently used for decoding by the module.
- //
- // Return value : codecy type, on success.
- // kVideoCodecUnknown, on error or if no receive codec is registered
- virtual VideoCodecType ReceiveCodec() const = 0;
-
- // Insert a parsed packet into the receiver side of the module. Will be placed in the
- // jitter buffer waiting for the frame to become complete. Returns as soon as the packet
- // has been placed in the jitter buffer.
- //
- // Input:
- // - incomingPayload : Payload of the packet.
- // - payloadLength : Length of the payload.
- // - rtpInfo : The parsed header.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incomingPayload,
- WebRtc_UWord32 payloadLength,
- const WebRtcRTPHeader& rtpInfo) = 0;
-
- // Minimum playout delay (Used for lip-sync). This is the minimum delay required
- // to sync with audio. Not included in VideoCodingModule::Delay()
- // Defaults to 0 ms.
- //
- // Input:
- // - minPlayoutDelayMs : Additional delay in ms.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 SetMinimumPlayoutDelay(WebRtc_UWord32 minPlayoutDelayMs) = 0;
-
- // Set the time required by the renderer to render a frame.
- //
- // Input:
- // - timeMS : The time in ms required by the renderer to render a frame.
- //
- // Return value : VCM_OK, on success.
- // < 0, on error.
- virtual WebRtc_Word32 SetRenderDelay(WebRtc_UWord32 timeMS) = 0;
-
- // The total delay desired by the VCM. Can be less than the minimum
- // delay set with SetMinimumPlayoutDelay.
- //
- // Return value : Total delay in ms, on success.
- // < 0, on error.
- virtual WebRtc_Word32 Delay() const = 0;
-
- // Get the received frame counters. Keeps track of the number of each frame type
- // received since the start of the call.
- //
- // Output:
- // - frameCount : Struct to be filled with the number of frames received.
- //
- // Return value : VCM_OK, on success.
- // <0, on error.
- virtual WebRtc_Word32 ReceivedFrameCount(VCMFrameCount& frameCount) const = 0;
-
- // Returns the number of packets discarded by the jitter buffer due to being
- // too late. This can include duplicated packets which arrived after the
- // frame was sent to the decoder. Therefore packets which were prematurely
- // NACKed will be counted.
- virtual WebRtc_UWord32 DiscardedPackets() const = 0;
-
-
- // Robustness APIs
-
- // Set the sender RTX/NACK mode.
- // Input:
- // - mode : the selected NACK mode.
- //
- // Return value : VCM_OK, on success;
- // < 0, on error.
- virtual int SetSenderNackMode(SenderNackMode mode) = 0;
-
- // Set the sender reference picture selection (RPS) mode.
- // Input:
- // - enable : true or false, for enable and disable, respectively.
- //
- // Return value : VCM_OK, on success;
- // < 0, on error.
- virtual int SetSenderReferenceSelection(bool enable) = 0;
-
- // Set the sender forward error correction (FEC) mode.
- // Input:
- // - enable : true or false, for enable and disable, respectively.
- //
- // Return value : VCM_OK, on success;
- // < 0, on error.
- virtual int SetSenderFEC(bool enable) = 0;
-
- // Set the key frame period, or disable periodic key frames (I-frames).
- // Input:
- // - periodMs : period in ms; <= 0 to disable periodic key frames.
- //
- // Return value : VCM_OK, on success;
- // < 0, on error.
- virtual int SetSenderKeyFramePeriod(int periodMs) = 0;
-
- // Set the receiver robustness mode. The mode decides how the receiver
- // responds to losses in the stream. The type of counter-measure (soft or
- // hard NACK, dual decoder, RPS, etc.) is selected through the
- // robustnessMode parameter. The errorMode parameter decides if it is
- // allowed to display frames corrupted by losses. Note that not all
- // combinations of the two parameters are feasible. An error will be
- // returned for invalid combinations.
- // Input:
- // - robustnessMode : selected robustness mode.
- // - errorMode : selected error mode.
- //
- // Return value : VCM_OK, on success;
- // < 0, on error.
- virtual int SetReceiverRobustnessMode(ReceiverRobustness robustnessMode,
- DecodeErrors errorMode) = 0;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_INTERFACE_VIDEO_CODING_H_
diff --git a/trunk/src/modules/video_coding/main/interface/video_coding_defines.h b/trunk/src/modules/video_coding/main/interface/video_coding_defines.h
deleted file mode 100644
index 3755f6d..0000000
--- a/trunk/src/modules/video_coding/main/interface/video_coding_defines.h
+++ /dev/null
@@ -1,195 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
-#define WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
-
-#include "typedefs.h"
-#include "modules/interface/module_common_types.h"
-
-namespace webrtc
-{
-
-// Error codes
-#define VCM_FRAME_NOT_READY 3
-#define VCM_REQUEST_SLI 2
-#define VCM_MISSING_CALLBACK 1
-#define VCM_OK 0
-#define VCM_GENERAL_ERROR -1
-#define VCM_LEVEL_EXCEEDED -2
-#define VCM_MEMORY -3
-#define VCM_PARAMETER_ERROR -4
-#define VCM_UNKNOWN_PAYLOAD -5
-#define VCM_CODEC_ERROR -6
-#define VCM_UNINITIALIZED -7
-#define VCM_NO_CODEC_REGISTERED -8
-#define VCM_JITTER_BUFFER_ERROR -9
-#define VCM_OLD_PACKET_ERROR -10
-#define VCM_NO_FRAME_DECODED -11
-#define VCM_ERROR_REQUEST_SLI -12
-#define VCM_NOT_IMPLEMENTED -20
-
-#define VCM_RED_PAYLOAD_TYPE 96
-#define VCM_ULPFEC_PAYLOAD_TYPE 97
-#define VCM_VP8_PAYLOAD_TYPE 120
-#define VCM_I420_PAYLOAD_TYPE 124
-
-enum VCMNackProperties
-{
- kNackHistoryLength = 450
-};
-
-enum VCMVideoProtection
-{
- kProtectionNack, // Both send-side and receive-side
- kProtectionNackSender, // Send-side only
- kProtectionNackReceiver, // Receive-side only
- kProtectionDualDecoder,
- kProtectionFEC,
- kProtectionNackFEC,
- kProtectionKeyOnLoss,
- kProtectionKeyOnKeyLoss,
- kProtectionPeriodicKeyFrames
-};
-
-enum VCMTemporalDecimation
-{
- kBitrateOverUseDecimation,
-};
-
-struct VCMFrameCount
-{
- WebRtc_UWord32 numKeyFrames;
- WebRtc_UWord32 numDeltaFrames;
-};
-
-
-// Callback class used for sending data ready to be packetized
-class VCMPacketizationCallback
-{
-public:
- virtual WebRtc_Word32 SendData(
- const FrameType frameType,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader& fragmentationHeader,
- const RTPVideoHeader* rtpVideoHdr) = 0;
-protected:
- virtual ~VCMPacketizationCallback() {}
-};
-
-// Callback class used for passing decoded frames which are ready to be rendered.
-class VCMFrameStorageCallback
-{
-public:
- virtual WebRtc_Word32 StoreReceivedFrame(const EncodedVideoData& frameToStore) = 0;
-
-protected:
- virtual ~VCMFrameStorageCallback() {}
-};
-
-// Callback class used for passing decoded frames which are ready to be rendered.
-class VCMReceiveCallback
-{
-public:
- virtual WebRtc_Word32 FrameToRender(VideoFrame& videoFrame) = 0;
- virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId) {return -1;}
-
-protected:
- virtual ~VCMReceiveCallback() {}
-};
-
-// Callback class used for informing the user of the bit rate and frame rate produced by the
-// encoder.
-class VCMSendStatisticsCallback
-{
-public:
- virtual WebRtc_Word32 SendStatistics(const WebRtc_UWord32 bitRate,
- const WebRtc_UWord32 frameRate) = 0;
-
-protected:
- virtual ~VCMSendStatisticsCallback() {}
-};
-
-// Callback class used for informing the user of the incoming bit rate and frame rate.
-class VCMReceiveStatisticsCallback
-{
-public:
- virtual WebRtc_Word32 ReceiveStatistics(const WebRtc_UWord32 bitRate,
- const WebRtc_UWord32 frameRate) = 0;
-
-protected:
- virtual ~VCMReceiveStatisticsCallback() {}
-};
-
-// Callback class used for telling the user about the requested amount of
-// bit stream protection: FEC rate for key and delta frame;
-// whether the FEC uses unequal protection (UEP) across packets,
-// for key and delta frame;
-// and whether NACK should be on or off.
-class VCMProtectionCallback
-{
-public:
- virtual int ProtectionRequest(
- uint8_t delta_fec_rate,
- uint8_t key_fec_rate,
- bool delta_use_uep_protection,
- bool key_use_uep_protection,
- bool nack_enabled,
- uint32_t* sent_video_rate_bps,
- uint32_t* sent_nack_rate_bps,
- uint32_t* sent_fec_rate_bps) = 0;
-
-protected:
- virtual ~VCMProtectionCallback() {}
-};
-
-// Callback class used for telling the user about what frame type needed to continue decoding.
-// Typically a key frame when the stream has been corrupted in some way.
-class VCMFrameTypeCallback
-{
-public:
- virtual WebRtc_Word32 RequestKeyFrame() = 0;
- virtual WebRtc_Word32 SliceLossIndicationRequest(const WebRtc_UWord64 pictureId) {return -1;}
-
-protected:
- virtual ~VCMFrameTypeCallback() {}
-};
-
-// Callback class used for telling the user about which packet sequence numbers are currently
-// missing and need to be resent.
-class VCMPacketRequestCallback
-{
-public:
- virtual WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequenceNumbers,
- WebRtc_UWord16 length) = 0;
-
-protected:
- virtual ~VCMPacketRequestCallback() {}
-};
-
-// Callback used to inform the user of the the desired resolution
-// as subscribed by Media Optimization (Quality Modes)
-class VCMQMSettingsCallback
-{
-public:
- virtual WebRtc_Word32 SetVideoQMSettings(const WebRtc_UWord32 frameRate,
- const WebRtc_UWord32 width,
- const WebRtc_UWord32 height) = 0;
-
-protected:
- virtual ~VCMQMSettingsCallback() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_INTERFACE_VIDEO_CODING_DEFINES_H_
diff --git a/trunk/src/modules/video_coding/main/source/Android.mk b/trunk/src/modules/video_coding/main/source/Android.mk
deleted file mode 100644
index 2ecf45d..0000000
--- a/trunk/src/modules/video_coding/main/source/Android.mk
+++ /dev/null
@@ -1,70 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_video_coding
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := \
- codec_database.cc \
- codec_timer.cc \
- content_metrics_processing.cc \
- decoding_state.cc \
- encoded_frame.cc \
- exp_filter.cc \
- frame_buffer.cc \
- frame_dropper.cc \
- generic_decoder.cc \
- generic_encoder.cc \
- inter_frame_delay.cc \
- jitter_buffer.cc \
- jitter_buffer_common.cc \
- jitter_estimator.cc \
- media_opt_util.cc \
- media_optimization.cc \
- packet.cc \
- qm_select.cc \
- receiver.cc \
- rtt_filter.cc \
- session_info.cc \
- timestamp_extrapolator.cc \
- timestamp_map.cc \
- timing.cc \
- video_coding_impl.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../codecs/interface \
- $(LOCAL_PATH)/../../codecs/i420/main/interface \
- $(LOCAL_PATH)/../../codecs/vp8/main/interface \
- $(LOCAL_PATH)/../../../interface \
- $(LOCAL_PATH)/../../../.. \
- $(LOCAL_PATH)/../../../../common_video/vplib/main/interface \
- $(LOCAL_PATH)/../../../../common_video/interface \
- $(LOCAL_PATH)/../../../../system_wrappers/interface
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/video_coding/main/source/codec_database.cc b/trunk/src/modules/video_coding/main/source/codec_database.cc
deleted file mode 100644
index 6f81895..0000000
--- a/trunk/src/modules/video_coding/main/source/codec_database.cc
+++ /dev/null
@@ -1,719 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "codec_database.h"
-
-#include <assert.h>
-
-#include "../../../../engine_configurations.h"
-#include "internal_defines.h"
-#include "trace.h"
-
-#if defined(_WIN32)
- // VS 2005: Don't warn for default initialized arrays. See help for more info.
- // Don't warn for strncpy being unsecure.
- // switch statement contains 'default' but no 'case' labels
-#pragma warning(disable:4351; disable:4996; disable:4065)
-#endif
-
-// Supported codecs
-#ifdef VIDEOCODEC_VP8
- #include "vp8.h"
-#endif
-#ifdef VIDEOCODEC_I420
- #include "i420.h"
-#endif
-
-namespace webrtc
-{
-
-VCMDecoderMapItem::VCMDecoderMapItem(VideoCodec* settings,
- WebRtc_UWord32 numberOfCores,
- bool requireKeyFrame)
-:
-_settings(settings),
-_numberOfCores(numberOfCores),
-_requireKeyFrame(requireKeyFrame)
-{
-}
-
-VCMExtDecoderMapItem::VCMExtDecoderMapItem(VideoDecoder* externalDecoderInstance,
- WebRtc_UWord8 payloadType,
- bool internalRenderTiming)
-:
-_payloadType(payloadType),
-_externalDecoderInstance(externalDecoderInstance),
-_internalRenderTiming(internalRenderTiming)
-{
-}
-
-VCMCodecDataBase::VCMCodecDataBase(WebRtc_Word32 id):
-_id(id),
-_numberOfCores(0),
-_maxPayloadSize(kDefaultPayloadSize),
-_periodicKeyFrames(false),
-_currentEncIsExternal(false),
-_sendCodec(),
-_receiveCodec(),
-_externalPayloadType(0),
-_externalEncoder(NULL),
-_internalSource(false),
-_ptrEncoder(NULL),
-_ptrDecoder(NULL),
-_currentDecIsExternal(false),
-_decMap(),
-_decExternalMap()
-{
- //
-}
-
-VCMCodecDataBase::~VCMCodecDataBase()
-{
- Reset();
-}
-
-WebRtc_Word32
-VCMCodecDataBase::Reset()
-{
- WebRtc_Word32 ret = ResetReceiver();
- if (ret < 0)
- {
- return ret;
- }
- ret = ResetSender();
- if (ret < 0)
- {
- return ret;
- }
- return VCM_OK;
-}
-
-WebRtc_Word32
-VCMCodecDataBase::ResetSender()
-{
- DeleteEncoder();
- _periodicKeyFrames = false;
- return VCM_OK;
-}
-
-VCMGenericEncoder* VCMCodecDataBase::CreateEncoder(
- const VideoCodecType type) const {
-
- switch(type)
- {
-#ifdef VIDEOCODEC_VP8
- case kVideoCodecVP8:
- return new VCMGenericEncoder(*(VP8Encoder::Create()));
-#endif
-#ifdef VIDEOCODEC_I420
- case kVideoCodecI420:
- return new VCMGenericEncoder(*(new I420Encoder));
-#endif
- default:
- return NULL;
- }
-}
-
-void
-VCMCodecDataBase::DeleteEncoder()
-{
- if (_ptrEncoder)
- {
- _ptrEncoder->Release();
- if (!_currentEncIsExternal)
- {
- delete &_ptrEncoder->_encoder;
- }
- delete _ptrEncoder;
- _ptrEncoder = NULL;
- }
-}
-
-WebRtc_UWord8
-VCMCodecDataBase::NumberOfCodecs()
-{
- return VCM_NUM_VIDEO_CODECS_AVAILABLE;
-}
-
-WebRtc_Word32
-VCMCodecDataBase::Codec(WebRtc_UWord8 listId, VideoCodec *settings)
-{
- if (settings == NULL)
- {
- return VCM_PARAMETER_ERROR;
- }
-
- if (listId >= VCM_NUM_VIDEO_CODECS_AVAILABLE)
- {
- return VCM_PARAMETER_ERROR;
- }
- memset(settings, 0, sizeof(VideoCodec));
- switch (listId)
- {
-#ifdef VIDEOCODEC_VP8
- case VCM_VP8_IDX:
- {
- strncpy(settings->plName, "VP8", 4);
- settings->codecType = kVideoCodecVP8;
- // 96 to 127 dynamic payload types for video codecs
- settings->plType = VCM_VP8_PAYLOAD_TYPE;
- settings->startBitrate = 100;
- settings->minBitrate = VCM_MIN_BITRATE;
- settings->maxBitrate = 0;
- settings->maxFramerate = VCM_DEFAULT_FRAME_RATE;
- settings->width = VCM_DEFAULT_CODEC_WIDTH;
- settings->height = VCM_DEFAULT_CODEC_HEIGHT;
- settings->numberOfSimulcastStreams = 0;
- settings->codecSpecific.VP8.resilience = kResilientStream;
- settings->codecSpecific.VP8.numberOfTemporalLayers = 1;
- break;
- }
-#endif
-#ifdef VIDEOCODEC_I420
- case VCM_I420_IDX:
- {
- strncpy(settings->plName, "I420", 5);
- settings->codecType = kVideoCodecI420;
- // 96 to 127 dynamic payload types for video codecs
- settings->plType = VCM_I420_PAYLOAD_TYPE;
- // Bitrate needed for this size and framerate
- settings->startBitrate = 3*VCM_DEFAULT_CODEC_WIDTH*
- VCM_DEFAULT_CODEC_HEIGHT*8*
- VCM_DEFAULT_FRAME_RATE/1000/2;
- settings->maxBitrate = settings->startBitrate;
- settings->maxFramerate = VCM_DEFAULT_FRAME_RATE;
- settings->width = VCM_DEFAULT_CODEC_WIDTH;
- settings->height = VCM_DEFAULT_CODEC_HEIGHT;
- settings->minBitrate = VCM_MIN_BITRATE;
- settings->numberOfSimulcastStreams = 0;
- break;
- }
-#endif
- default:
- {
- return VCM_PARAMETER_ERROR;
- }
- }
-
- return VCM_OK;
-}
-
-WebRtc_Word32
-VCMCodecDataBase::Codec(VideoCodecType codecType, VideoCodec* settings)
-{
- for (int i = 0; i < VCMCodecDataBase::NumberOfCodecs(); i++)
- {
- const WebRtc_Word32 ret = VCMCodecDataBase::Codec(i, settings);
- if (ret != VCM_OK)
- {
- return ret;
- }
- if (codecType == settings->codecType)
- {
- return VCM_OK;
- }
- }
- return VCM_PARAMETER_ERROR;
-}
-
-// assuming only one registered encoder - since only one used, no need for more
-WebRtc_Word32
-VCMCodecDataBase::RegisterSendCodec(const VideoCodec* sendCodec,
- WebRtc_UWord32 numberOfCores,
- WebRtc_UWord32 maxPayloadSize)
- {
- if (sendCodec == NULL)
- {
- return VCM_UNINITIALIZED;
- }
- if (maxPayloadSize == 0)
- {
- maxPayloadSize = kDefaultPayloadSize;
- }
- if (numberOfCores > 32)
- {
- return VCM_PARAMETER_ERROR;
- }
- if (sendCodec->plType <= 0)
- {
- return VCM_PARAMETER_ERROR;
- }
- // Make sure the start bit rate is sane...
- if (sendCodec->startBitrate > 1000000)
- {
- return VCM_PARAMETER_ERROR;
- }
- if (sendCodec->codecType == kVideoCodecUnknown)
- {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- _numberOfCores = numberOfCores;
- _maxPayloadSize = maxPayloadSize;
-
- memcpy(&_sendCodec, sendCodec, sizeof(VideoCodec));
-
- if (_sendCodec.maxBitrate == 0)
- {
- // max is one bit per pixel
- _sendCodec.maxBitrate = ((WebRtc_Word32)_sendCodec.height *
- (WebRtc_Word32)_sendCodec.width *
- (WebRtc_Word32)_sendCodec.maxFramerate) / 1000;
- if (_sendCodec.startBitrate > _sendCodec.maxBitrate)
- {
- // but if the customer tries to set a higher start bit rate we will increase
- // the max accordingly
- _sendCodec.maxBitrate = _sendCodec.startBitrate;
- }
- }
-
- return VCM_OK;
-}
-
-WebRtc_Word32
-VCMCodecDataBase::SendCodec(VideoCodec* currentSendCodec) const
-{
- WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideoCoding, VCMId(_id), "SendCodec");
-
- if(_ptrEncoder == NULL)
- {
- return VCM_UNINITIALIZED;
- }
- memcpy(currentSendCodec, &_sendCodec, sizeof(VideoCodec));
- return VCM_OK;
-}
-
-VideoCodecType
-VCMCodecDataBase::SendCodec() const
-{
- WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideoCoding, VCMId(_id),
- "SendCodec type");
- if (_ptrEncoder == NULL)
- {
- return kVideoCodecUnknown;
- }
- return _sendCodec.codecType;
-}
-
-WebRtc_Word32
-VCMCodecDataBase::DeRegisterExternalEncoder(WebRtc_UWord8 payloadType, bool& wasSendCodec)
-{
- wasSendCodec = false;
- if (_externalPayloadType != payloadType)
- {
- return VCM_PARAMETER_ERROR;
- }
- if (_sendCodec.plType == payloadType)
- {
- //De-register as send codec if needed
- DeleteEncoder();
- memset(&_sendCodec, 0, sizeof(VideoCodec));
- _currentEncIsExternal = false;
- wasSendCodec = true;
- }
- _externalPayloadType = 0;
- _externalEncoder = NULL;
- _internalSource = false;
- return VCM_OK;
-}
-
-WebRtc_Word32
-VCMCodecDataBase::RegisterExternalEncoder(VideoEncoder* externalEncoder,
- WebRtc_UWord8 payloadType,
- bool internalSource)
-{
- // since only one encoder can be used at a given time,
- // only one external encoder can be registered/used
- _externalEncoder = externalEncoder;
- _externalPayloadType = payloadType;
- _internalSource = internalSource;
-
- return VCM_OK;
-}
-
-VCMGenericEncoder*
-VCMCodecDataBase::SetEncoder(const VideoCodec* settings,
- VCMEncodedFrameCallback* VCMencodedFrameCallback)
-
-{
- // if encoder exists, will destroy it and create new one
- DeleteEncoder();
-
- if (settings->plType == _externalPayloadType)
- {
- // External encoder
- _ptrEncoder = new VCMGenericEncoder(*_externalEncoder, _internalSource);
- _currentEncIsExternal = true;
- }
- else
- {
- _ptrEncoder = CreateEncoder(settings->codecType);
- _currentEncIsExternal = false;
- }
- VCMencodedFrameCallback->SetPayloadType(settings->plType);
-
- if (_ptrEncoder == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceVideoCoding,
- VCMId(_id),
- "Failed to create encoder: %s.",
- settings->plName);
- return NULL;
- }
- if (_ptrEncoder->InitEncode(settings, _numberOfCores, _maxPayloadSize) < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceVideoCoding,
- VCMId(_id),
- "Failed to initialize encoder: %s.",
- settings->plName);
- DeleteEncoder();
- return NULL;
- }
- else if (_ptrEncoder->RegisterEncodeCallback(VCMencodedFrameCallback) < 0)
- {
- DeleteEncoder();
- return NULL;
- }
- // Intentionally don't check return value since the encoder registration
- // shouldn't fail because the codec doesn't support changing the
- // periodic key frame setting.
- _ptrEncoder->SetPeriodicKeyFrames(_periodicKeyFrames);
- return _ptrEncoder;
-}
-
-WebRtc_Word32
-VCMCodecDataBase::SetPeriodicKeyFrames(bool enable)
-{
- _periodicKeyFrames = enable;
- if (_ptrEncoder != NULL)
- {
- return _ptrEncoder->SetPeriodicKeyFrames(_periodicKeyFrames);
- }
- return VCM_OK;
-}
-
-WebRtc_Word32
-VCMCodecDataBase::RegisterReceiveCodec(const VideoCodec* receiveCodec,
- WebRtc_UWord32 numberOfCores,
- bool requireKeyFrame)
-{
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideoCoding, VCMId(_id),
- "Codec: %s, Payload type %d, Height %d, Width %d, Bitrate %d, Framerate %d.",
- receiveCodec->plName, receiveCodec->plType,
- receiveCodec->height, receiveCodec->width,
- receiveCodec->startBitrate, receiveCodec->maxFramerate);
-
- // check if payload value already exists, if so - erase old and insert new
- DeRegisterReceiveCodec(receiveCodec->plType);
- if (receiveCodec->codecType == kVideoCodecUnknown)
- {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- VideoCodec* newReceiveCodec = new VideoCodec(*receiveCodec);
- _decMap[receiveCodec->plType] =
- new VCMDecoderMapItem(newReceiveCodec, numberOfCores, requireKeyFrame);
-
- return VCM_OK;
-}
-
-WebRtc_Word32 VCMCodecDataBase::DeRegisterReceiveCodec(
- WebRtc_UWord8 payloadType)
-{
- DecoderMap::iterator it = _decMap.find(payloadType);
- if (it == _decMap.end())
- {
- return VCM_PARAMETER_ERROR;
- }
- VCMDecoderMapItem* decItem = (*it).second;
- delete decItem->_settings;
- delete decItem;
- _decMap.erase(it);
- if (_receiveCodec.plType == payloadType)
- {
- // This codec is currently in use.
- memset(&_receiveCodec, 0, sizeof(VideoCodec));
- _currentDecIsExternal = false;
- }
- return VCM_OK;
-}
-
-WebRtc_Word32
-VCMCodecDataBase::ResetReceiver()
-{
- ReleaseDecoder(_ptrDecoder);
- _ptrDecoder = NULL;
- memset(&_receiveCodec, 0, sizeof(VideoCodec));
- DecoderMap::iterator it = _decMap.begin();
- while (it != _decMap.end()) {
- if ((*it).second->_settings != NULL)
- {
- delete (*it).second->_settings;
- }
- delete (*it).second;
- _decMap.erase(it);
- it = _decMap.begin();
- }
- ExternalDecoderMap::iterator exterit = _decExternalMap.begin();
- while (exterit != _decExternalMap.end()) {
- delete (*exterit).second;
- _decExternalMap.erase(exterit);
- exterit = _decExternalMap.begin();
- }
-
- _currentDecIsExternal = false;
- return VCM_OK;
-}
-
-WebRtc_Word32
-VCMCodecDataBase::DeRegisterExternalDecoder(WebRtc_UWord8 payloadType)
-{
- ExternalDecoderMap::iterator it = _decExternalMap.find(payloadType);
- if (it == _decExternalMap.end())
- {
- // Not found
- return VCM_PARAMETER_ERROR;
- }
- if (_receiveCodec.plType == payloadType)
- {
- // Release it if it was registered and in use
- ReleaseDecoder(_ptrDecoder);
- _ptrDecoder = NULL;
- }
- DeRegisterReceiveCodec(payloadType);
- delete (*it).second;
- _decExternalMap.erase(it);
- return VCM_OK;
-}
-
-// Add the external encoder object to the list of external decoders.
-// Won't be registered as a receive codec until RegisterReceiveCodec is called.
-WebRtc_Word32
-VCMCodecDataBase::RegisterExternalDecoder(VideoDecoder* externalDecoder,
- WebRtc_UWord8 payloadType,
- bool internalRenderTiming)
-{
- // check if payload value already exists, if so - erase old and insert new
- VCMExtDecoderMapItem* extDecoder = new VCMExtDecoderMapItem(externalDecoder,
- payloadType,
- internalRenderTiming);
- if (extDecoder == NULL)
- {
- return VCM_MEMORY;
- }
- DeRegisterExternalDecoder(payloadType);
- _decExternalMap[payloadType] = extDecoder;
-
- return VCM_OK;
-}
-
-bool
-VCMCodecDataBase::DecoderRegistered() const
-{
- return !_decMap.empty();
-}
-
-WebRtc_Word32
-VCMCodecDataBase::ReceiveCodec(VideoCodec* currentReceiveCodec) const
-{
- if (_ptrDecoder == NULL)
- {
- return VCM_NO_FRAME_DECODED;
- }
- memcpy(currentReceiveCodec, &_receiveCodec, sizeof(VideoCodec));
- return VCM_OK;
-}
-
-VideoCodecType
-VCMCodecDataBase::ReceiveCodec() const
-{
- if (_ptrDecoder == NULL)
- {
- return kVideoCodecUnknown;
- }
- return _receiveCodec.codecType;
-}
-
-VCMGenericDecoder*
-VCMCodecDataBase::SetDecoder(WebRtc_UWord8 payloadType,
- VCMDecodedFrameCallback& callback)
-{
- if (payloadType == _receiveCodec.plType || payloadType == 0)
- {
- return _ptrDecoder;
- }
- // check for exisitng decoder, if exists - delete
- if (_ptrDecoder)
- {
- ReleaseDecoder(_ptrDecoder);
- _ptrDecoder = NULL;
- memset(&_receiveCodec, 0, sizeof(VideoCodec));
- }
- _ptrDecoder = CreateAndInitDecoder(payloadType, _receiveCodec,
- _currentDecIsExternal);
- if (_ptrDecoder == NULL)
- {
- return NULL;
- }
- if (_ptrDecoder->RegisterDecodeCompleteCallback(&callback) < 0)
- {
- ReleaseDecoder(_ptrDecoder);
- _ptrDecoder = NULL;
- memset(&_receiveCodec, 0, sizeof(VideoCodec));
- return NULL;
- }
- return _ptrDecoder;
-}
-
-VCMGenericDecoder*
-VCMCodecDataBase::CreateAndInitDecoder(WebRtc_UWord8 payloadType,
- VideoCodec& newCodec,
- bool &external) const
-{
- VCMDecoderMapItem* decoderItem = FindDecoderItem(payloadType);
- if (decoderItem == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(_id),
- "Unknown payload type: %u", payloadType);
- return NULL;
- }
- VCMGenericDecoder* ptrDecoder = NULL;
- VCMExtDecoderMapItem* externalDecItem = FindExternalDecoderItem(
- payloadType);
- if (externalDecItem != NULL)
- {
- // External codec
- ptrDecoder = new VCMGenericDecoder(
- *externalDecItem->_externalDecoderInstance,
- _id,
- true);
- external = true;
- }
- else
- {
- // create decoder
- ptrDecoder = CreateDecoder(decoderItem->_settings->codecType);
- external = false;
- }
- if (ptrDecoder == NULL)
- {
- return NULL;
- }
-
- if (ptrDecoder->InitDecode(decoderItem->_settings,
- decoderItem->_numberOfCores,
- decoderItem->_requireKeyFrame) < 0)
- {
- ReleaseDecoder(ptrDecoder);
- return NULL;
- }
- memcpy(&newCodec, decoderItem->_settings, sizeof(VideoCodec));
- return ptrDecoder;
-}
-
-VCMGenericDecoder*
-VCMCodecDataBase::CreateDecoderCopy() const
-{
- if (_ptrDecoder == NULL)
- {
- return NULL;
- }
- VideoDecoder* decoderCopy = _ptrDecoder->_decoder.Copy();
- if (decoderCopy == NULL)
- {
- return NULL;
- }
- return new VCMGenericDecoder(*decoderCopy, _id, _ptrDecoder->External());
-}
-
-void
-VCMCodecDataBase::CopyDecoder(const VCMGenericDecoder& decoder)
-{
- VideoDecoder* decoderCopy = decoder._decoder.Copy();
- if (decoderCopy != NULL)
- {
- VCMDecodedFrameCallback* cb = _ptrDecoder->_callback;
- ReleaseDecoder(_ptrDecoder);
- _ptrDecoder = new VCMGenericDecoder(*decoderCopy, _id,
- decoder.External());
- if (cb && _ptrDecoder->RegisterDecodeCompleteCallback(cb))
- {
- assert(false);
- }
- }
-}
-
-bool
-VCMCodecDataBase::RenderTiming() const
-{
- bool renderTiming = true;
- if (_currentDecIsExternal)
- {
- VCMExtDecoderMapItem* extItem = FindExternalDecoderItem(_receiveCodec.plType);
- renderTiming = extItem->_internalRenderTiming;
- }
- return renderTiming;
-}
-
-void
-VCMCodecDataBase::ReleaseDecoder(VCMGenericDecoder* decoder) const
-{
- if (decoder != NULL)
- {
- assert(&decoder->_decoder != NULL);
- decoder->Release();
- if (!decoder->External())
- {
- delete &decoder->_decoder;
- }
- delete decoder;
- }
-}
-
-VCMDecoderMapItem*
-VCMCodecDataBase::FindDecoderItem(WebRtc_UWord8 payloadType) const
-{
- DecoderMap::const_iterator it = _decMap.find(payloadType);
- if (it != _decMap.end())
- {
- return (*it).second;
- }
- return NULL;
-}
-
-VCMExtDecoderMapItem*
-VCMCodecDataBase::FindExternalDecoderItem(WebRtc_UWord8 payloadType) const
-{
- ExternalDecoderMap::const_iterator it = _decExternalMap.find(payloadType);
- if (it != _decExternalMap.end())
- {
- return (*it).second;
- }
- return NULL;
-}
-
-VCMGenericDecoder*
-VCMCodecDataBase::CreateDecoder(VideoCodecType type) const
-{
- switch(type)
- {
-#ifdef VIDEOCODEC_VP8
- case kVideoCodecVP8:
- return new VCMGenericDecoder(*(VP8Decoder::Create()), _id);
-#endif
-#ifdef VIDEOCODEC_I420
- case kVideoCodecI420:
- return new VCMGenericDecoder(*(new I420Decoder), _id);
-#endif
- default:
- return NULL;
- }
-}
-}
diff --git a/trunk/src/modules/video_coding/main/source/codec_database.h b/trunk/src/modules/video_coding/main/source/codec_database.h
deleted file mode 100644
index aab8229..0000000
--- a/trunk/src/modules/video_coding/main/source/codec_database.h
+++ /dev/null
@@ -1,210 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_DATABASE_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODEC_DATABASE_H_
-
-#include <map>
-
-#include "modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "modules/video_coding/main/interface/video_coding.h"
-#include "modules/video_coding/main/source/generic_decoder.h"
-#include "modules/video_coding/main/source/generic_encoder.h"
-#include "typedefs.h"
-
-namespace webrtc
-{
-
-enum VCMCodecDBProperties
-{
- kDefaultPayloadSize = 1440
-};
-
-class VCMDecoderMapItem {
-public:
- VCMDecoderMapItem(VideoCodec* settings,
- WebRtc_UWord32 numberOfCores,
- bool requireKeyFrame);
-
- VideoCodec* _settings;
- WebRtc_UWord32 _numberOfCores;
- bool _requireKeyFrame;
-};
-
-class VCMExtDecoderMapItem {
-public:
- VCMExtDecoderMapItem(VideoDecoder* externalDecoderInstance,
- WebRtc_UWord8 payloadType,
- bool internalRenderTiming);
-
- WebRtc_UWord8 _payloadType;
- VideoDecoder* _externalDecoderInstance;
- bool _internalRenderTiming;
-};
-
-/*******************************/
-/* VCMCodecDataBase class */
-/*******************************/
-class VCMCodecDataBase
-{
-public:
- VCMCodecDataBase(WebRtc_Word32 id);
- ~VCMCodecDataBase();
- /**
- * Release codecdatabase - release all memory for both send and receive side
- */
- WebRtc_Word32 Reset();
- /**
- * Sender Side
- */
- /**
- * Returns the number of supported codecs (or -1 in case of error).
- */
- static WebRtc_UWord8 NumberOfCodecs();
- /**
- * Get supported codecs with ID
- * Input Values:
- * listnr : Requested codec id number
- * codec_inst: Pointer to the struct in which the returned codec information is copied
- * Return Values: 0 if successful, otherwise
- */
- static WebRtc_Word32 Codec(WebRtc_UWord8 listId, VideoCodec* settings);
- static WebRtc_Word32 Codec(VideoCodecType codecType, VideoCodec* settings);
- /**
- * Reset Sender side
- */
- WebRtc_Word32 ResetSender();
- /**
- * Setting the sender side codec and initiaiting the desired codec given the VideoCodec
- * struct.
- * Return Value: 0 if the codec and the settings are supported, otherwise
- */
- WebRtc_Word32 RegisterSendCodec(const VideoCodec* sendCodec,
- WebRtc_UWord32 numberOfCores,
- WebRtc_UWord32 maxPayloadSize);
- /**
- * Get current send side codec. Relevant for internal codecs only.
- */
- WebRtc_Word32 SendCodec(VideoCodec* currentSendCodec) const;
- /**
- * Get current send side codec type. Relevant for internal codecs only.
- */
- VideoCodecType SendCodec() const;
- /**
- * Register external encoder - current assumption - if one is registered then it will also
- * be used, and therefore it is also initialized
- * Return value: A pointer to the encoder on success, or null, in case of an error.
- */
- WebRtc_Word32 DeRegisterExternalEncoder(WebRtc_UWord8 payloadType, bool& wasSendCodec);
- WebRtc_Word32 RegisterExternalEncoder(VideoEncoder* externalEncoder,
- WebRtc_UWord8 payloadType,
- bool internalSource);
- /**
- * Returns a encoder given a payloadname - to be used with internal encoders only.
- * Special cases:
- * Encoder exists - If payload matches, returns existing one, otherwise,
- * deletes existing one and creates new one.
- * No match found / Error - returns NULL.
- */
- VCMGenericEncoder* SetEncoder(const VideoCodec* settings,
- VCMEncodedFrameCallback* VCMencodedFrameCallback);
-
- WebRtc_Word32 SetPeriodicKeyFrames(bool enable);
-
- bool InternalSource() const;
-
- /*
- * Receiver Side
- */
- WebRtc_Word32 ResetReceiver();
- /**
- * Register external decoder/render object
- */
- WebRtc_Word32 DeRegisterExternalDecoder(WebRtc_UWord8 payloadType);
- WebRtc_Word32 RegisterExternalDecoder(VideoDecoder* externalDecoder,
- WebRtc_UWord8 payloadType,
- bool internalRenderTiming);
-
- bool DecoderRegistered() const;
- /**
- * Register recieve codec
- */
- WebRtc_Word32 RegisterReceiveCodec(const VideoCodec* receiveCodec,
- WebRtc_UWord32 numberOfCores,
- bool requireKeyFrame);
- WebRtc_Word32 DeRegisterReceiveCodec(WebRtc_UWord8 payloadType);
- /**
- * Get current receive side codec. Relevant for internal codecs only.
- */
- WebRtc_Word32 ReceiveCodec(VideoCodec* currentReceiveCodec) const;
- /**
- * Get current receive side codec type. Relevant for internal codecs only.
- */
- VideoCodecType ReceiveCodec() const;
- /**
- * Returns a decoder given which matches a payload type.
- * Special cases:
- * Decoder exists - If payload matches, returns existing one, otherwise, deletes
- * existing one, and creates new one.
- * No match found / Error - returns NULL.
- */
- VCMGenericDecoder* SetDecoder(WebRtc_UWord8 payloadType, VCMDecodedFrameCallback& callback);
-
- VCMGenericDecoder* CreateAndInitDecoder(WebRtc_UWord8 payloadType,
- VideoCodec& newCodec,
- bool &external) const;
-
- VCMGenericDecoder* CreateDecoderCopy() const;
-
- void ReleaseDecoder(VCMGenericDecoder* decoder) const;
-
- void CopyDecoder(const VCMGenericDecoder& decoder);
-
- bool RenderTiming() const;
-
-protected:
- /**
- * Create an internal Encoder given a codec type
- */
- VCMGenericEncoder* CreateEncoder(const VideoCodecType type) const;
-
- void DeleteEncoder();
- /*
- * Create an internal Decoder given a codec type
- */
- VCMGenericDecoder* CreateDecoder(VideoCodecType type) const;
-
- VCMDecoderMapItem* FindDecoderItem(WebRtc_UWord8 payloadType) const;
-
- VCMExtDecoderMapItem* FindExternalDecoderItem(WebRtc_UWord8 payloadType) const;
-
-private:
- typedef std::map<uint8_t, VCMDecoderMapItem*> DecoderMap;
- typedef std::map<uint8_t, VCMExtDecoderMapItem*> ExternalDecoderMap;
- WebRtc_Word32 _id;
- WebRtc_UWord32 _numberOfCores;
- WebRtc_UWord32 _maxPayloadSize;
- bool _periodicKeyFrames;
- bool _currentEncIsExternal;
- VideoCodec _sendCodec;
- VideoCodec _receiveCodec;
- WebRtc_UWord8 _externalPayloadType;
- VideoEncoder* _externalEncoder;
- bool _internalSource;
- VCMGenericEncoder* _ptrEncoder;
- VCMGenericDecoder* _ptrDecoder;
- bool _currentDecIsExternal;
- DecoderMap _decMap;
- ExternalDecoderMap _decExternalMap;
-}; // end of VCMCodecDataBase class definition
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODEC_DATABASE_H_
diff --git a/trunk/src/modules/video_coding/main/source/codec_timer.cc b/trunk/src/modules/video_coding/main/source/codec_timer.cc
deleted file mode 100644
index 1d112fa..0000000
--- a/trunk/src/modules/video_coding/main/source/codec_timer.cc
+++ /dev/null
@@ -1,133 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "codec_timer.h"
-
-#include <assert.h>
-
-namespace webrtc
-{
-
-VCMCodecTimer::VCMCodecTimer()
-:
-_filteredMax(0),
-_firstDecodeTime(true),
-_shortMax(0),
-_history()
-{
- Reset();
-}
-
-WebRtc_Word32 VCMCodecTimer::StopTimer(WebRtc_Word64 startTimeMs, WebRtc_Word64 nowMs)
-{
- const WebRtc_Word32 timeDiff = static_cast<WebRtc_Word32>(nowMs - startTimeMs);
- MaxFilter(timeDiff, nowMs);
- return timeDiff;
-}
-
-void VCMCodecTimer::Reset()
-{
- _filteredMax = 0;
- _firstDecodeTime = true;
- _shortMax = 0;
- for (int i=0; i < MAX_HISTORY_SIZE; i++)
- {
- _history[i].shortMax = 0;
- _history[i].timeMs = -1;
- }
-}
-
-// Update the max-value filter
-void VCMCodecTimer::MaxFilter(WebRtc_Word32 decodeTime, WebRtc_Word64 nowMs)
-{
- if (!_firstDecodeTime)
- {
- UpdateMaxHistory(decodeTime, nowMs);
- ProcessHistory(nowMs);
- }
- else
- {
- _firstDecodeTime = false;
- }
-}
-
-void
-VCMCodecTimer::UpdateMaxHistory(WebRtc_Word32 decodeTime, WebRtc_Word64 now)
-{
- if (_history[0].timeMs >= 0 &&
- now - _history[0].timeMs < SHORT_FILTER_MS)
- {
- if (decodeTime > _shortMax)
- {
- _shortMax = decodeTime;
- }
- }
- else
- {
- // Only add a new value to the history once a second
- if(_history[0].timeMs == -1)
- {
- // First, no shift
- _shortMax = decodeTime;
- }
- else
- {
- // Shift
- for(int i = (MAX_HISTORY_SIZE - 2); i >= 0 ; i--)
- {
- _history[i+1].shortMax = _history[i].shortMax;
- _history[i+1].timeMs = _history[i].timeMs;
- }
- }
- if (_shortMax == 0)
- {
- _shortMax = decodeTime;
- }
-
- _history[0].shortMax = _shortMax;
- _history[0].timeMs = now;
- _shortMax = 0;
- }
-}
-
-void
-VCMCodecTimer::ProcessHistory(WebRtc_Word64 nowMs)
-{
- _filteredMax = _shortMax;
- if (_history[0].timeMs == -1)
- {
- return;
- }
- for (int i=0; i < MAX_HISTORY_SIZE; i++)
- {
- if (_history[i].timeMs == -1)
- {
- break;
- }
- if (nowMs - _history[i].timeMs > MAX_HISTORY_SIZE * SHORT_FILTER_MS)
- {
- // This sample (and all samples after this) is too old
- break;
- }
- if (_history[i].shortMax > _filteredMax)
- {
- // This sample is the largest one this far into the history
- _filteredMax = _history[i].shortMax;
- }
- }
-}
-
-// Get the maximum observed time within a time window
-WebRtc_Word32 VCMCodecTimer::RequiredDecodeTimeMs(FrameType /*frameType*/) const
-{
- return _filteredMax;
-}
-
-}
diff --git a/trunk/src/modules/video_coding/main/source/codec_timer.h b/trunk/src/modules/video_coding/main/source/codec_timer.h
deleted file mode 100644
index e03c5bf..0000000
--- a/trunk/src/modules/video_coding/main/source/codec_timer.h
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
-
-#include "typedefs.h"
-#include "module_common_types.h"
-
-namespace webrtc
-{
-
-// MAX_HISTORY_SIZE * SHORT_FILTER_MS defines the window size in milliseconds
-#define MAX_HISTORY_SIZE 20
-#define SHORT_FILTER_MS 1000
-
-class VCMShortMaxSample
-{
-public:
- VCMShortMaxSample() : shortMax(0), timeMs(-1) {};
-
- WebRtc_Word32 shortMax;
- WebRtc_Word64 timeMs;
-};
-
-class VCMCodecTimer
-{
-public:
- VCMCodecTimer();
-
- // Updates and returns the max filtered decode time.
- WebRtc_Word32 StopTimer(WebRtc_Word64 startTimeMs, WebRtc_Word64 nowMs);
-
- // Empty the list of timers.
- void Reset();
-
- // Get the required decode time in ms.
- WebRtc_Word32 RequiredDecodeTimeMs(FrameType frameType) const;
-
-private:
- void UpdateMaxHistory(WebRtc_Word32 decodeTime, WebRtc_Word64 now);
- void MaxFilter(WebRtc_Word32 newTime, WebRtc_Word64 nowMs);
- void ProcessHistory(WebRtc_Word64 nowMs);
-
- WebRtc_Word32 _filteredMax;
- bool _firstDecodeTime;
- WebRtc_Word32 _shortMax;
- VCMShortMaxSample _history[MAX_HISTORY_SIZE];
-
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_CODEC_TIMER_H_
diff --git a/trunk/src/modules/video_coding/main/source/content_metrics_processing.cc b/trunk/src/modules/video_coding/main/source/content_metrics_processing.cc
deleted file mode 100644
index 99160c9..0000000
--- a/trunk/src/modules/video_coding/main/source/content_metrics_processing.cc
+++ /dev/null
@@ -1,125 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/video_coding/main/source/content_metrics_processing.h"
-
-#include <math.h>
-
-#include "modules/interface/module_common_types.h"
-#include "modules/video_coding/main/interface/video_coding_defines.h"
-
-namespace webrtc {
-//////////////////////////////////
-/// VCMContentMetricsProcessing //
-//////////////////////////////////
-
-VCMContentMetricsProcessing::VCMContentMetricsProcessing()
- : recursive_avg_factor_(1 / 150.0f), // matched to 30fps.
- frame_cnt_uniform_avg_(0),
- avg_motion_level_(0.0f),
- avg_spatial_level_(0.0f) {
- recursive_avg_ = new VideoContentMetrics();
- uniform_avg_ = new VideoContentMetrics();
-}
-
-VCMContentMetricsProcessing::~VCMContentMetricsProcessing() {
- delete recursive_avg_;
- delete uniform_avg_;
-}
-
-int VCMContentMetricsProcessing::Reset() {
- recursive_avg_->Reset();
- uniform_avg_->Reset();
- frame_cnt_uniform_avg_ = 0;
- avg_motion_level_ = 0.0f;
- avg_spatial_level_ = 0.0f;
- return VCM_OK;
-}
-
-void VCMContentMetricsProcessing::UpdateFrameRate(uint32_t frameRate) {
- // Update factor for recursive averaging.
- recursive_avg_factor_ = static_cast<float> (1000.0f) /
- static_cast<float>(frameRate * kQmMinIntervalMs);
-}
-
-VideoContentMetrics* VCMContentMetricsProcessing::LongTermAvgData() {
- return recursive_avg_;
-}
-
-VideoContentMetrics* VCMContentMetricsProcessing::ShortTermAvgData() {
- if (frame_cnt_uniform_avg_ == 0) {
- return NULL;
- }
- // Two metrics are used: motion and spatial level.
- uniform_avg_->motion_magnitude = avg_motion_level_ /
- static_cast<float>(frame_cnt_uniform_avg_);
- uniform_avg_->spatial_pred_err = avg_spatial_level_ /
- static_cast<float>(frame_cnt_uniform_avg_);
- return uniform_avg_;
-}
-
-void VCMContentMetricsProcessing::ResetShortTermAvgData() {
- // Reset.
- avg_motion_level_ = 0.0f;
- avg_spatial_level_ = 0.0f;
- frame_cnt_uniform_avg_ = 0;
-}
-
-int VCMContentMetricsProcessing::UpdateContentData(
- const VideoContentMetrics *contentMetrics) {
- if (contentMetrics == NULL) {
- return VCM_OK;
- }
- return ProcessContent(contentMetrics);
-}
-
-int VCMContentMetricsProcessing::ProcessContent(
- const VideoContentMetrics *contentMetrics) {
- // Update the recursive averaged metrics: average is over longer window
- // of time: over QmMinIntervalMs ms.
- UpdateRecursiveAvg(contentMetrics);
- // Update the uniform averaged metrics: average is over shorter window
- // of time: based on ~RTCP reports.
- UpdateUniformAvg(contentMetrics);
- return VCM_OK;
-}
-
-void VCMContentMetricsProcessing::UpdateUniformAvg(
- const VideoContentMetrics *contentMetrics) {
- // Update frame counter.
- frame_cnt_uniform_avg_ += 1;
- // Update averaged metrics: motion and spatial level are used.
- avg_motion_level_ += contentMetrics->motion_magnitude;
- avg_spatial_level_ += contentMetrics->spatial_pred_err;
- return;
-}
-
-void VCMContentMetricsProcessing::UpdateRecursiveAvg(
- const VideoContentMetrics *contentMetrics) {
-
- // Spatial metrics: 2x2, 1x2(H), 2x1(V).
- recursive_avg_->spatial_pred_err = (1 - recursive_avg_factor_) *
- recursive_avg_->spatial_pred_err +
- recursive_avg_factor_ * contentMetrics->spatial_pred_err;
-
- recursive_avg_->spatial_pred_err_h = (1 - recursive_avg_factor_) *
- recursive_avg_->spatial_pred_err_h +
- recursive_avg_factor_ * contentMetrics->spatial_pred_err_h;
-
- recursive_avg_->spatial_pred_err_v = (1 - recursive_avg_factor_) *
- recursive_avg_->spatial_pred_err_v +
- recursive_avg_factor_ * contentMetrics->spatial_pred_err_v;
-
- // Motion metric: Derived from NFD (normalized frame difference).
- recursive_avg_->motion_magnitude = (1 - recursive_avg_factor_) *
- recursive_avg_->motion_magnitude +
- recursive_avg_factor_ * contentMetrics->motion_magnitude;
-}
-} // end of namespace
diff --git a/trunk/src/modules/video_coding/main/source/content_metrics_processing.h b/trunk/src/modules/video_coding/main/source/content_metrics_processing.h
deleted file mode 100644
index 0317add..0000000
--- a/trunk/src/modules/video_coding/main/source/content_metrics_processing.h
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_CONTENT_METRICS_PROCESSING_H_
-#define WEBRTC_MODULES_VIDEO_CODING_CONTENT_METRICS_PROCESSING_H_
-
-#include "typedefs.h"
-
-namespace webrtc {
-
-struct VideoContentMetrics;
-
-// QM interval time (in ms)
-enum {
- kQmMinIntervalMs = 10000
-};
-
-// Flag for NFD metric vs motion metric
-enum {
- kNfdMetric = 1
-};
-
-/**********************************/
-/* Content Metrics Processing */
-/**********************************/
-class VCMContentMetricsProcessing {
- public:
- VCMContentMetricsProcessing();
- ~VCMContentMetricsProcessing();
-
- // Update class with latest metrics.
- int UpdateContentData(const VideoContentMetrics *contentMetrics);
-
- // Reset the short-term averaged content data.
- void ResetShortTermAvgData();
-
- // Initialize.
- int Reset();
-
- // Inform class of current frame rate.
- void UpdateFrameRate(uint32_t frameRate);
-
- // Returns the long-term averaged content data: recursive average over longer
- // time scale.
- VideoContentMetrics* LongTermAvgData();
-
- // Returns the short-term averaged content data: uniform average over
- // shorter time scalE.
- VideoContentMetrics* ShortTermAvgData();
-
- private:
- // Compute working average.
- int ProcessContent(const VideoContentMetrics *contentMetrics);
-
- // Update the recursive averaged metrics: longer time average (~5/10 secs).
- void UpdateRecursiveAvg(const VideoContentMetrics *contentMetrics);
-
- // Update the uniform averaged metrics: shorter time average (~RTCP report).
- void UpdateUniformAvg(const VideoContentMetrics *contentMetrics);
-
- VideoContentMetrics* recursive_avg_;
- VideoContentMetrics* uniform_avg_;
- float recursive_avg_factor_;
- uint32_t frame_cnt_uniform_avg_;
- float avg_motion_level_;
- float avg_spatial_level_;
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CODING_CONTENT_METRICS_PROCESSING_H_
diff --git a/trunk/src/modules/video_coding/main/source/decoding_state.cc b/trunk/src/modules/video_coding/main/source/decoding_state.cc
deleted file mode 100644
index eea0a2c..0000000
--- a/trunk/src/modules/video_coding/main/source/decoding_state.cc
+++ /dev/null
@@ -1,202 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/video_coding/main/source/decoding_state.h"
-
-#include "modules/video_coding/main/source/frame_buffer.h"
-#include "modules/video_coding/main/source/jitter_buffer_common.h"
-#include "modules/video_coding/main/source/packet.h"
-#include "modules/interface/module_common_types.h"
-
-namespace webrtc {
-
-VCMDecodingState::VCMDecodingState()
- : sequence_num_(0),
- time_stamp_(0),
- picture_id_(kNoPictureId),
- temporal_id_(kNoTemporalIdx),
- tl0_pic_id_(kNoTl0PicIdx),
- full_sync_(true),
- init_(true) {}
-
-VCMDecodingState::~VCMDecodingState() {}
-
-void VCMDecodingState::Reset() {
- // TODO(mikhal): Verify - not always would want to reset the sync
- sequence_num_ = 0;
- time_stamp_ = 0;
- picture_id_ = kNoPictureId;
- temporal_id_ = kNoTemporalIdx;
- tl0_pic_id_ = kNoTl0PicIdx;
- full_sync_ = true;
- init_ = true;
-}
-
-uint32_t VCMDecodingState::time_stamp() const {
- return time_stamp_;
-}
-
-uint16_t VCMDecodingState::sequence_num() const {
- return sequence_num_;
-}
-
-bool VCMDecodingState::IsOldFrame(const VCMFrameBuffer* frame) const {
- assert(frame != NULL);
- if (init_)
- return false;
- return (LatestTimestamp(time_stamp_, frame->TimeStamp(), NULL)
- == time_stamp_);
-}
-
-bool VCMDecodingState::IsOldPacket(const VCMPacket* packet) const {
- assert(packet != NULL);
- if (init_)
- return false;
- return (LatestTimestamp(time_stamp_, packet->timestamp, NULL)
- == time_stamp_);
-}
-
-void VCMDecodingState::SetState(const VCMFrameBuffer* frame) {
- assert(frame != NULL && frame->GetHighSeqNum() >= 0);
- UpdateSyncState(frame);
- sequence_num_ = static_cast<uint16_t>(frame->GetHighSeqNum());
- time_stamp_ = frame->TimeStamp();
- picture_id_ = frame->PictureId();
- temporal_id_ = frame->TemporalId();
- tl0_pic_id_ = frame->Tl0PicId();
- init_ = false;
-}
-
-void VCMDecodingState::SetStateOneBack(const VCMFrameBuffer* frame) {
- assert(frame != NULL && frame->GetHighSeqNum() >= 0);
- sequence_num_ = static_cast<uint16_t>(frame->GetHighSeqNum()) - 1u;
- time_stamp_ = frame->TimeStamp() - 1u;
- temporal_id_ = frame->TemporalId();
- if (frame->PictureId() != kNoPictureId) {
- if (frame->PictureId() == 0)
- picture_id_ = 0x7FFF;
- else
- picture_id_ = frame->PictureId() - 1;
- }
- if (frame->Tl0PicId() != kNoTl0PicIdx) {
- if (frame->Tl0PicId() == 0)
- tl0_pic_id_ = 0x00FF;
- else
- tl0_pic_id_ = frame->Tl0PicId() - 1;
- }
- init_ = false;
-}
-
-void VCMDecodingState::UpdateOldPacket(const VCMPacket* packet) {
- assert(packet != NULL);
- if (packet->timestamp == time_stamp_) {
- // Late packet belonging to the last decoded frame - make sure we update the
- // last decoded sequence number.
- sequence_num_ = LatestSequenceNumber(packet->seqNum, sequence_num_, NULL);
- }
-}
-
-void VCMDecodingState::SetSeqNum(uint16_t new_seq_num) {
- sequence_num_ = new_seq_num;
-}
-
-bool VCMDecodingState::init() const {
- return init_;
-}
-
-bool VCMDecodingState::full_sync() const {
- return full_sync_;
-}
-
-void VCMDecodingState::UpdateSyncState(const VCMFrameBuffer* frame) {
- if (init_)
- return;
- if (frame->TemporalId() == kNoTemporalIdx ||
- frame->Tl0PicId() == kNoTl0PicIdx) {
- full_sync_ = true;
- } else if (frame->FrameType() == kVideoFrameKey || frame->LayerSync()) {
- full_sync_ = true;
- } else if (full_sync_) {
- // Verify that we are still in sync.
- // Sync will be broken if continuity is true for layers but not for the
- // other methods (PictureId and SeqNum).
- if (!ContinuousPictureId(frame->PictureId()) &&
- !ContinuousSeqNum(static_cast<uint16_t>(frame->GetLowSeqNum()))) {
- // Non-layered methods have failed.
- full_sync_ = false;
- }
- }
-}
-
-bool VCMDecodingState::ContinuousFrame(const VCMFrameBuffer* frame) const {
- // Check continuity based on the following hierarchy:
- // - Temporal layers (stop here if out of sync).
- // - Picture Id when available.
- // - Sequence numbers.
- // Return true when in initial state.
- // Note that when a method is not applicable it will return false.
- assert(frame != NULL);
- if (init_)
- return true;
-
- if (!ContinuousLayer(frame->TemporalId(), frame->Tl0PicId())) {
- // Base layers are not continuous or temporal layers are inactive.
- // In the presence of temporal layers, check for Picture ID/sequence number
- // continuity if sync can be restored by this frame.
- if (!full_sync_ && !frame->LayerSync())
- return false;
- else if (!ContinuousPictureId(frame->PictureId()))
- return ContinuousSeqNum(static_cast<uint16_t>(frame->GetLowSeqNum()));
- }
- return true;
-}
-
-bool VCMDecodingState::ContinuousPictureId(int picture_id) const {
- // First, check if applicable.
- if (picture_id == kNoPictureId || picture_id_ == kNoPictureId)
- return false;
-
- int next_picture_id = picture_id_ + 1;
- if (picture_id < picture_id_) {
- // Wrap
- if (picture_id_ >= 0x80) {
- // 15 bits used for picture id
- return ((next_picture_id & 0x7FFF) == picture_id);
- } else {
- // 7 bits used for picture id
- return ((next_picture_id & 0x7F) == picture_id);
- }
- }
- // No wrap
- return (next_picture_id == picture_id);
-}
-
-bool VCMDecodingState::ContinuousSeqNum(uint16_t seq_num) const {
- return (seq_num == static_cast<uint16_t>(sequence_num_ + 1));
-}
-
-bool VCMDecodingState::ContinuousLayer(int temporal_id,
- int tl0_pic_id) const {
- // First, check if applicable.
- if (temporal_id == kNoTemporalIdx || tl0_pic_id == kNoTl0PicIdx)
- return false;
- // If this is the first frame to use temporal layers, make sure we start
- // from base.
- else if (tl0_pic_id_ == kNoTl0PicIdx && temporal_id_ == kNoTemporalIdx &&
- temporal_id == 0)
- return true;
-
- // Current implementation: Look for base layer continuity.
- if (temporal_id != 0)
- return false;
- return (static_cast<uint8_t>(tl0_pic_id_ + 1) == tl0_pic_id);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/decoding_state.h b/trunk/src/modules/video_coding/main/source/decoding_state.h
deleted file mode 100644
index afec3ba..0000000
--- a/trunk/src/modules/video_coding/main/source/decoding_state.h
+++ /dev/null
@@ -1,69 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_DECODING_STATE_H_
-#define WEBRTC_MODULES_VIDEO_CODING_DECODING_STATE_H_
-
-#include "typedefs.h"
-
-namespace webrtc {
-
-// Forward declarations
-class VCMFrameBuffer;
-class VCMPacket;
-
-class VCMDecodingState {
- public:
- VCMDecodingState();
- ~VCMDecodingState();
- // Check for old frame
- bool IsOldFrame(const VCMFrameBuffer* frame) const;
- // Check for old packet
- bool IsOldPacket(const VCMPacket* packet) const;
- // Check for frame continuity based on current decoded state. Use best method
- // possible, i.e. temporal info, picture ID or sequence number.
- bool ContinuousFrame(const VCMFrameBuffer* frame) const;
- void SetState(const VCMFrameBuffer* frame);
- // Set the decoding state one frame back.
- void SetStateOneBack(const VCMFrameBuffer* frame);
- // Update the sequence number if the timestamp matches current state and the
- // sequence number is higher than the current one. This accounts for packets
- // arriving late.
- void UpdateOldPacket(const VCMPacket* packet);
- void SetSeqNum(uint16_t new_seq_num);
- void Reset();
- uint32_t time_stamp() const;
- uint16_t sequence_num() const;
- // Return true if at initial state.
- bool init() const;
- // Return true when sync is on - decode all layers.
- bool full_sync() const;
-
- private:
- void UpdateSyncState(const VCMFrameBuffer* frame);
- // Designated continuity functions
- bool ContinuousPictureId(int picture_id) const;
- bool ContinuousSeqNum(uint16_t seq_num) const;
- bool ContinuousLayer(int temporal_id, int tl0_pic_id) const;
-
- // Keep state of last decoded frame.
- // TODO(mikhal/stefan): create designated classes to handle these types.
- uint16_t sequence_num_;
- uint32_t time_stamp_;
- int picture_id_;
- int temporal_id_;
- int tl0_pic_id_;
- bool full_sync_; // Sync flag when temporal layers are used.
- bool init_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_DECODING_STATE_H_
diff --git a/trunk/src/modules/video_coding/main/source/decoding_state_unittest.cc b/trunk/src/modules/video_coding/main/source/decoding_state_unittest.cc
deleted file mode 100644
index 853f42a..0000000
--- a/trunk/src/modules/video_coding/main/source/decoding_state_unittest.cc
+++ /dev/null
@@ -1,430 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string.h>
-
-#include "modules/video_coding/main/source/decoding_state.h"
-#include "modules/video_coding/main/source/frame_buffer.h"
-#include "gtest/gtest.h"
-#include "modules/video_coding/main/source/jitter_buffer_common.h"
-#include "modules/interface/module_common_types.h"
-#include "modules/video_coding/main/source/packet.h"
-
-namespace webrtc {
-
-
-TEST(TestDecodingState, Sanity) {
- VCMDecodingState dec_state;
- dec_state.Reset();
- EXPECT_TRUE(dec_state.init());
- EXPECT_TRUE(dec_state.full_sync());
-}
-
-TEST(TestDecodingState, FrameContinuity) {
- VCMDecodingState dec_state;
- // Check that makes decision based on correct method.
- VCMFrameBuffer frame;
- frame.SetState(kStateEmpty);
- VCMPacket* packet = new VCMPacket();
- packet->isFirstPacket = 1;
- packet->timestamp = 1;
- packet->seqNum = 0xffff;
- packet->frameType = kVideoFrameDelta;
- packet->codecSpecificHeader.codec = kRTPVideoVP8;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0x007F;
- frame.InsertPacket(*packet, 0, false, 0);
- // Should return true on init.
- dec_state.Reset();
- EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
- dec_state.SetState(&frame);
- frame.Reset();
- // Use pictureId
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0x0002;
- frame.InsertPacket(*packet, 0, false, 0);
- EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
- frame.Reset();
- frame.SetState(kStateEmpty);
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0;
- packet->seqNum = 10;
- frame.InsertPacket(*packet, 0, false, 0);
- EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
-
- // Use sequence numbers.
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = kNoPictureId;
- frame.Reset();
- frame.SetState(kStateEmpty);
- packet->seqNum = dec_state.sequence_num() - 1u;
- frame.InsertPacket(*packet, 0, false, 0);
- EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
- frame.Reset();
- frame.SetState(kStateEmpty);
- packet->seqNum = dec_state.sequence_num() + 1u;
- frame.InsertPacket(*packet, 0, false, 0);
- // Insert another packet to this frame
- packet->seqNum++;
- frame.InsertPacket(*packet, 0, false, 0);
- // Verify wrap.
- EXPECT_EQ(dec_state.sequence_num(), 0xffff);
- EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
- dec_state.SetState(&frame);
-
- // Insert packet with temporal info.
- dec_state.Reset();
- frame.Reset();
- frame.SetState(kStateEmpty);
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0;
- packet->seqNum = 1;
- packet->timestamp = 1;
- EXPECT_TRUE(dec_state.full_sync());
- frame.InsertPacket(*packet, 0, false, 0);
- dec_state.SetState(&frame);
- EXPECT_TRUE(dec_state.full_sync());
- frame.Reset();
- frame.SetState(kStateEmpty);
- // 1 layer up - still good.
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 1;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 1;
- packet->seqNum = 2;
- packet->timestamp = 2;
- frame.InsertPacket(*packet, 0, false, 0);
- EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
- dec_state.SetState(&frame);
- EXPECT_TRUE(dec_state.full_sync());
- frame.Reset();
- frame.SetState(kStateEmpty);
- // Lost non-base layer packet => should update sync parameter.
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 3;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 3;
- packet->seqNum = 4;
- packet->timestamp = 4;
- frame.InsertPacket(*packet, 0, false, 0);
- EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
- // Now insert the next non-base layer (belonging to a next tl0PicId).
- frame.Reset();
- frame.SetState(kStateEmpty);
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 1;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 2;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 4;
- packet->seqNum = 5;
- packet->timestamp = 5;
- frame.InsertPacket(*packet, 0, false, 0);
- // Checking continuity and not updating the state - this should not trigger
- // an update of sync state.
- EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
- EXPECT_TRUE(dec_state.full_sync());
- // Next base layer (dropped interim non-base layers) - should update sync.
- frame.Reset();
- frame.SetState(kStateEmpty);
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 1;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 5;
- packet->seqNum = 6;
- packet->timestamp = 6;
- frame.InsertPacket(*packet, 0, false, 0);
- EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
- dec_state.SetState(&frame);
- EXPECT_FALSE(dec_state.full_sync());
-
- // Check wrap for temporal layers.
- frame.Reset();
- frame.SetState(kStateEmpty);
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0x00FF;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 6;
- packet->seqNum = 7;
- packet->timestamp = 7;
- frame.InsertPacket(*packet, 0, false, 0);
- dec_state.SetState(&frame);
- EXPECT_FALSE(dec_state.full_sync());
- frame.Reset();
- frame.SetState(kStateEmpty);
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0x0000;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 7;
- packet->seqNum = 8;
- packet->timestamp = 8;
- frame.InsertPacket(*packet, 0, false, 0);
- EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
- // The current frame is not continuous
- dec_state.SetState(&frame);
- EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
- delete packet;
-}
-
-TEST(TestDecodingState, SetStateOneBack) {
- VCMDecodingState dec_state;
- VCMFrameBuffer frame;
- frame.SetState(kStateEmpty);
- VCMPacket* packet = new VCMPacket();
- // Based on PictureId.
- packet->frameType = kVideoFrameDelta;
- packet->codecSpecificHeader.codec = kRTPVideoVP8;
- packet->timestamp = 0;
- packet->seqNum = 0;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0;
- packet->frameType = kVideoFrameDelta;
- frame.InsertPacket(*packet, 0, false, 0);
- dec_state.SetStateOneBack(&frame);
- EXPECT_EQ(dec_state.sequence_num(), 0xFFFF);
- // Check continuity.
- EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
-
- // Based on Temporal layers.
- packet->timestamp = 0;
- packet->seqNum = 0;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = kNoPictureId;
- packet->frameType = kVideoFrameDelta;
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
- frame.InsertPacket(*packet, 0, false, 0);
- dec_state.SetStateOneBack(&frame);
- // Check continuity
- EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
- delete packet;
-}
-
-TEST(TestDecodingState, UpdateOldPacket) {
- VCMDecodingState dec_state;
- // Update only if zero size and newer than previous.
- // Should only update if the timeStamp match.
- VCMFrameBuffer frame;
- frame.SetState(kStateEmpty);
- VCMPacket* packet = new VCMPacket();
- packet->timestamp = 1;
- packet->seqNum = 1;
- packet->frameType = kVideoFrameDelta;
- frame.InsertPacket(*packet, 0, false, 0);
- dec_state.SetState(&frame);
- EXPECT_EQ(dec_state.sequence_num(), 1);
- // Insert an empty packet that does not belong to the same frame.
- // => Sequence num should be the same.
- packet->timestamp = 2;
- dec_state.UpdateOldPacket(packet);
- EXPECT_EQ(dec_state.sequence_num(), 1);
- // Now insert empty packet belonging to the same frame.
- packet->timestamp = 1;
- packet->seqNum = 2;
- packet->frameType = kFrameEmpty;
- packet->sizeBytes = 0;
- dec_state.UpdateOldPacket(packet);
- EXPECT_EQ(dec_state.sequence_num(), 2);
- // Now insert delta packet belonging to the same frame.
- packet->timestamp = 1;
- packet->seqNum = 3;
- packet->frameType = kVideoFrameDelta;
- packet->sizeBytes = 1400;
- dec_state.UpdateOldPacket(packet);
- EXPECT_EQ(dec_state.sequence_num(), 3);
- // Insert a packet belonging to an older timestamp - should not update the
- // sequence number.
- packet->timestamp = 0;
- packet->seqNum = 4;
- packet->frameType = kFrameEmpty;
- packet->sizeBytes = 0;
- dec_state.UpdateOldPacket(packet);
- EXPECT_EQ(dec_state.sequence_num(), 3);
-
- delete packet;
-}
-
-TEST(TestDecodingState, MultiLayerBehavior) {
- // Identify sync/non-sync when more than one layer.
- VCMDecodingState dec_state;
- // Identify packets belonging to old frames/packets.
- // Set state for current frames.
- // tl0PicIdx 0, temporal id 0.
- VCMFrameBuffer frame;
- VCMPacket* packet = new VCMPacket();
- packet->frameType = kVideoFrameDelta;
- packet->codecSpecificHeader.codec = kRTPVideoVP8;
- frame.SetState(kStateEmpty);
- packet->timestamp = 0;
- packet->seqNum = 0;
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0;
- frame.InsertPacket(*packet, 0, false, 0);
- dec_state.SetState(&frame);
- // tl0PicIdx 0, temporal id 1.
- frame.Reset();
- frame.SetState(kStateEmpty);
- packet->timestamp = 1;
- packet->seqNum = 1;
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 1;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 1;
- frame.InsertPacket(*packet, 0, false, 0);
- EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
- dec_state.SetState(&frame);
- EXPECT_TRUE(dec_state.full_sync());
- // Lost tl0PicIdx 0, temporal id 2.
- // Insert tl0PicIdx 0, temporal id 3.
- frame.Reset();
- frame.SetState(kStateEmpty);
- packet->timestamp = 3;
- packet->seqNum = 3;
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 3;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 3;
- frame.InsertPacket(*packet, 0, false, 0);
- EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
- dec_state.SetState(&frame);
- EXPECT_FALSE(dec_state.full_sync());
- // Insert next base layer
- frame.Reset();
- frame.SetState(kStateEmpty);
- packet->timestamp = 4;
- packet->seqNum = 4;
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 1;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 4;
- frame.InsertPacket(*packet, 0, false, 0);
- EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
- dec_state.SetState(&frame);
- EXPECT_FALSE(dec_state.full_sync());
- // Insert key frame - should update sync value.
- // A key frame is always a base layer.
- frame.Reset();
- frame.SetState(kStateEmpty);
- packet->frameType = kVideoFrameKey;
- packet->isFirstPacket = 1;
- packet->timestamp = 5;
- packet->seqNum = 5;
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 2;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 5;
- frame.InsertPacket(*packet, 0, false, 0);
- EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
- dec_state.SetState(&frame);
- EXPECT_TRUE(dec_state.full_sync());
- // After sync, a continuous PictureId is required
- // (continuous base layer is not enough )
- frame.Reset();
- frame.SetState(kStateEmpty);
- packet->frameType = kVideoFrameDelta;
- packet->timestamp = 6;
- packet->seqNum = 6;
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 3;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 6;
- frame.InsertPacket(*packet, 0, false, 0);
- EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
- EXPECT_TRUE(dec_state.full_sync());
- frame.Reset();
- frame.SetState(kStateEmpty);
- packet->frameType = kVideoFrameDelta;
- packet->isFirstPacket = 1;
- packet->timestamp = 8;
- packet->seqNum = 8;
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 4;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 8;
- frame.InsertPacket(*packet, 0, false, 0);
- EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
- EXPECT_TRUE(dec_state.full_sync());
- dec_state.SetState(&frame);
- EXPECT_FALSE(dec_state.full_sync());
-
- // Insert a non-ref frame - should update sync value.
- frame.Reset();
- frame.SetState(kStateEmpty);
- packet->frameType = kVideoFrameDelta;
- packet->isFirstPacket = 1;
- packet->timestamp = 9;
- packet->seqNum = 9;
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 4;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 2;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 9;
- packet->codecSpecificHeader.codecHeader.VP8.layerSync = true;
- frame.InsertPacket(*packet, 0, false, 0);
- dec_state.SetState(&frame);
- EXPECT_TRUE(dec_state.full_sync());
-
- // The following test will verify the sync flag behavior after a loss.
- // Create the following pattern:
- // Update base layer, lose packet 1 (sync flag on, layer 2), insert packet 3
- // (sync flag on, layer 2) check continuity and sync flag after inserting
- // packet 2 (sync flag on, layer 1).
- // Base layer.
- frame.Reset();
- dec_state.Reset();
- frame.SetState(kStateEmpty);
- packet->frameType = kVideoFrameDelta;
- packet->isFirstPacket = 1;
- packet->markerBit = 1;
- packet->timestamp = 0;
- packet->seqNum = 0;
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 0;
- packet->codecSpecificHeader.codecHeader.VP8.layerSync = false;
- frame.InsertPacket(*packet, 0, false, 0);
- dec_state.SetState(&frame);
- EXPECT_TRUE(dec_state.full_sync());
- // Layer 2 - 2 packets (insert one, lose one).
- frame.Reset();
- frame.SetState(kStateEmpty);
- packet->frameType = kVideoFrameDelta;
- packet->isFirstPacket = 1;
- packet->markerBit = 0;
- packet->timestamp = 1;
- packet->seqNum = 1;
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 2;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 1;
- packet->codecSpecificHeader.codecHeader.VP8.layerSync = true;
- frame.InsertPacket(*packet, 0, false, 0);
- EXPECT_TRUE(dec_state.ContinuousFrame(&frame));
- // Layer 1
- frame.Reset();
- frame.SetState(kStateEmpty);
- packet->frameType = kVideoFrameDelta;
- packet->isFirstPacket = 1;
- packet->markerBit = 1;
- packet->timestamp = 2;
- packet->seqNum = 3;
- packet->codecSpecificHeader.codecHeader.VP8.tl0PicIdx = 0;
- packet->codecSpecificHeader.codecHeader.VP8.temporalIdx = 1;
- packet->codecSpecificHeader.codecHeader.VP8.pictureId = 2;
- packet->codecSpecificHeader.codecHeader.VP8.layerSync = true;
- frame.InsertPacket(*packet, 0, false, 0);
- EXPECT_FALSE(dec_state.ContinuousFrame(&frame));
- EXPECT_TRUE(dec_state.full_sync());
-
- delete packet;
-}
-
-TEST(TestDecodingState, OldInput) {
- VCMDecodingState dec_state;
- // Identify packets belonging to old frames/packets.
- // Set state for current frames.
- VCMFrameBuffer frame;
- frame.SetState(kStateEmpty);
- VCMPacket* packet = new VCMPacket();
- packet->timestamp = 10;
- packet->seqNum = 1;
- frame.InsertPacket(*packet, 0, false, 0);
- dec_state.SetState(&frame);
- packet->timestamp = 9;
- EXPECT_TRUE(dec_state.IsOldPacket(packet));
- // Check for old frame
- frame.Reset();
- frame.InsertPacket(*packet, 0, false, 0);
- EXPECT_TRUE(dec_state.IsOldFrame(&frame));
-
-
- delete packet;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/encoded_frame.cc b/trunk/src/modules/video_coding/main/source/encoded_frame.cc
deleted file mode 100644
index 8e5d745..0000000
--- a/trunk/src/modules/video_coding/main/source/encoded_frame.cc
+++ /dev/null
@@ -1,253 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "encoded_frame.h"
-#include "generic_encoder.h"
-#include "jitter_buffer_common.h"
-#include "video_coding_defines.h"
-
-namespace webrtc {
-
-VCMEncodedFrame::VCMEncodedFrame()
-:
-webrtc::EncodedImage(),
-_renderTimeMs(-1),
-_payloadType(0),
-_missingFrame(false),
-_codec(kVideoCodecUnknown),
-_fragmentation()
-{
- _codecSpecificInfo.codecType = kVideoCodecUnknown;
-}
-
-VCMEncodedFrame::VCMEncodedFrame(const webrtc::EncodedImage& rhs)
-:
-webrtc::EncodedImage(rhs),
-_renderTimeMs(-1),
-_payloadType(0),
-_missingFrame(false),
-_codec(kVideoCodecUnknown),
-_fragmentation()
-{
- _codecSpecificInfo.codecType = kVideoCodecUnknown;
- _buffer = NULL;
- _size = 0;
- _length = 0;
- if (rhs._buffer != NULL)
- {
- VerifyAndAllocate(rhs._length);
- memcpy(_buffer, rhs._buffer, rhs._length);
- }
-}
-
-VCMEncodedFrame::VCMEncodedFrame(const VCMEncodedFrame& rhs)
- :
- webrtc::EncodedImage(rhs),
- _renderTimeMs(rhs._renderTimeMs),
- _payloadType(rhs._payloadType),
- _missingFrame(rhs._missingFrame),
- _codecSpecificInfo(rhs._codecSpecificInfo),
- _codec(rhs._codec),
- _fragmentation() {
- _buffer = NULL;
- _size = 0;
- _length = 0;
- if (rhs._buffer != NULL)
- {
- VerifyAndAllocate(rhs._length);
- memcpy(_buffer, rhs._buffer, rhs._length);
- _length = rhs._length;
- }
- // Deep operator=
- _fragmentation = rhs._fragmentation;
-}
-
-VCMEncodedFrame::~VCMEncodedFrame()
-{
- Free();
-}
-
-void VCMEncodedFrame::Free()
-{
- Reset();
- if (_buffer != NULL)
- {
- delete [] _buffer;
- _buffer = NULL;
- }
-}
-
-void VCMEncodedFrame::Reset()
-{
- _renderTimeMs = -1;
- _timeStamp = 0;
- _payloadType = 0;
- _frameType = kDeltaFrame;
- _encodedWidth = 0;
- _encodedHeight = 0;
- _completeFrame = false;
- _missingFrame = false;
- _length = 0;
- _codecSpecificInfo.codecType = kVideoCodecUnknown;
- _codec = kVideoCodecUnknown;
-}
-
-void VCMEncodedFrame::CopyCodecSpecific(const RTPVideoHeader* header)
-{
- if (header)
- {
- switch (header->codec)
- {
- case kRTPVideoVP8:
- {
- if (_codecSpecificInfo.codecType != kVideoCodecVP8)
- {
- // This is the first packet for this frame.
- _codecSpecificInfo.codecSpecific.VP8.pictureId = -1;
- _codecSpecificInfo.codecSpecific.VP8.temporalIdx = 0;
- _codecSpecificInfo.codecSpecific.VP8.layerSync = false;
- _codecSpecificInfo.codecSpecific.VP8.keyIdx = -1;
- _codecSpecificInfo.codecType = kVideoCodecVP8;
- }
- _codecSpecificInfo.codecSpecific.VP8.nonReference =
- header->codecHeader.VP8.nonReference;
- if (header->codecHeader.VP8.pictureId != kNoPictureId)
- {
- _codecSpecificInfo.codecSpecific.VP8.pictureId =
- header->codecHeader.VP8.pictureId;
- }
- if (header->codecHeader.VP8.temporalIdx != kNoTemporalIdx)
- {
- _codecSpecificInfo.codecSpecific.VP8.temporalIdx =
- header->codecHeader.VP8.temporalIdx;
- _codecSpecificInfo.codecSpecific.VP8.layerSync =
- header->codecHeader.VP8.layerSync;
- }
- if (header->codecHeader.VP8.keyIdx != kNoKeyIdx)
- {
- _codecSpecificInfo.codecSpecific.VP8.keyIdx =
- header->codecHeader.VP8.keyIdx;
- }
- break;
- }
- default:
- {
- _codecSpecificInfo.codecType = kVideoCodecUnknown;
- break;
- }
- }
- }
-}
-
-const RTPFragmentationHeader* VCMEncodedFrame::FragmentationHeader() const {
- return &_fragmentation;
-}
-
-WebRtc_Word32
-VCMEncodedFrame::Store(VCMFrameStorageCallback& storeCallback) const
-{
- EncodedVideoData frameToStore;
- frameToStore.codec = _codec;
- if (_buffer != NULL)
- {
- frameToStore.VerifyAndAllocate(_length);
- memcpy(frameToStore.payloadData, _buffer, _length);
- frameToStore.payloadSize = _length;
- }
- frameToStore.completeFrame = _completeFrame;
- frameToStore.encodedWidth = _encodedWidth;
- frameToStore.encodedHeight = _encodedHeight;
- frameToStore.frameType = ConvertFrameType(_frameType);
- frameToStore.missingFrame = _missingFrame;
- frameToStore.payloadType = _payloadType;
- frameToStore.renderTimeMs = _renderTimeMs;
- frameToStore.timeStamp = _timeStamp;
- storeCallback.StoreReceivedFrame(frameToStore);
- return VCM_OK;
-}
-
-WebRtc_Word32
-VCMEncodedFrame::VerifyAndAllocate(const WebRtc_UWord32 minimumSize)
-{
- if(minimumSize > _size)
- {
- // create buffer of sufficient size
- WebRtc_UWord8* newBuffer = new WebRtc_UWord8[minimumSize];
- if (newBuffer == NULL)
- {
- return -1;
- }
- if(_buffer)
- {
- // copy old data
- memcpy(newBuffer, _buffer, _size);
- delete [] _buffer;
- }
- _buffer = newBuffer;
- _size = minimumSize;
- }
- return 0;
-}
-
-webrtc::FrameType VCMEncodedFrame::ConvertFrameType(VideoFrameType frameType)
-{
- switch(frameType)
- {
- case kKeyFrame:
- {
- return kVideoFrameKey;
- }
- case kDeltaFrame:
- {
- return kVideoFrameDelta;
- }
- case kGoldenFrame:
- {
- return kVideoFrameGolden;
- }
- case kAltRefFrame:
- {
- return kVideoFrameAltRef;
- }
- default:
- {
- return kVideoFrameDelta;
- }
- }
-}
-
-VideoFrameType VCMEncodedFrame::ConvertFrameType(webrtc::FrameType frameType)
-{
- switch (frameType)
- {
- case kVideoFrameKey:
- {
- return kKeyFrame;
- }
- case kVideoFrameDelta:
- {
- return kDeltaFrame;
- }
- case kVideoFrameGolden:
- {
- return kGoldenFrame;
- }
- case kVideoFrameAltRef:
- {
- return kAltRefFrame;
- }
- default:
- {
- return kDeltaFrame;
- }
- }
-}
-
-}
diff --git a/trunk/src/modules/video_coding/main/source/encoded_frame.h b/trunk/src/modules/video_coding/main/source/encoded_frame.h
deleted file mode 100644
index 6289e9e..0000000
--- a/trunk/src/modules/video_coding/main/source/encoded_frame.h
+++ /dev/null
@@ -1,120 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
-#define WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
-
-#include "common_types.h"
-#include "common_video/interface/video_image.h"
-#include "modules/interface/module_common_types.h"
-#include "modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "modules/video_coding/main/interface/video_coding_defines.h"
-
-namespace webrtc
-{
-
-class VCMEncodedFrame : protected EncodedImage
-{
-public:
- VCMEncodedFrame();
- VCMEncodedFrame(const webrtc::EncodedImage& rhs);
- VCMEncodedFrame(const VCMEncodedFrame& rhs);
-
- ~VCMEncodedFrame();
- /**
- * Delete VideoFrame and resets members to zero
- */
- void Free();
- /**
- * Set render time in milliseconds
- */
- void SetRenderTime(const WebRtc_Word64 renderTimeMs) {_renderTimeMs = renderTimeMs;}
-
- /**
- * Set the encoded frame size
- */
- void SetEncodedSize(WebRtc_UWord32 width, WebRtc_UWord32 height)
- { _encodedWidth = width; _encodedHeight = height; }
- /**
- * Get the encoded image
- */
- const webrtc::EncodedImage& EncodedImage() const
- { return static_cast<const webrtc::EncodedImage&>(*this); }
- /**
- * Get pointer to frame buffer
- */
- const WebRtc_UWord8* Buffer() const {return _buffer;}
- /**
- * Get frame length
- */
- WebRtc_UWord32 Length() const {return _length;}
- /**
- * Get frame timestamp (90kHz)
- */
- WebRtc_UWord32 TimeStamp() const {return _timeStamp;}
- /**
- * Get render time in milliseconds
- */
- WebRtc_Word64 RenderTimeMs() const {return _renderTimeMs;}
- /**
- * Get frame type
- */
- webrtc::FrameType FrameType() const {return ConvertFrameType(_frameType);}
- /**
- * True if this frame is complete, false otherwise
- */
- bool Complete() const { return _completeFrame; }
- /**
- * True if there's a frame missing before this frame
- */
- bool MissingFrame() const { return _missingFrame; }
- /**
- * Payload type of the encoded payload
- */
- WebRtc_UWord8 PayloadType() const { return _payloadType; }
- /**
- * Get codec specific info.
- * The returned pointer is only valid as long as the VCMEncodedFrame
- * is valid. Also, VCMEncodedFrame owns the pointer and will delete
- * the object.
- */
- const CodecSpecificInfo* CodecSpecific() const {return &_codecSpecificInfo;}
-
- const RTPFragmentationHeader* FragmentationHeader() const;
-
- WebRtc_Word32 Store(VCMFrameStorageCallback& storeCallback) const;
-
- static webrtc::FrameType ConvertFrameType(VideoFrameType frameType);
- static VideoFrameType ConvertFrameType(webrtc::FrameType frameType);
-
-protected:
- /**
- * Verifies that current allocated buffer size is larger than or equal to the input size.
- * If the current buffer size is smaller, a new allocation is made and the old buffer data
- * is copied to the new buffer.
- * Buffer size is updated to minimumSize.
- */
- WebRtc_Word32 VerifyAndAllocate(const WebRtc_UWord32 minimumSize);
-
- void Reset();
-
- void CopyCodecSpecific(const RTPVideoHeader* header);
-
- WebRtc_Word64 _renderTimeMs;
- WebRtc_UWord8 _payloadType;
- bool _missingFrame;
- CodecSpecificInfo _codecSpecificInfo;
- webrtc::VideoCodecType _codec;
- RTPFragmentationHeader _fragmentation;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_ENCODED_FRAME_H_
diff --git a/trunk/src/modules/video_coding/main/source/er_tables_xor.h b/trunk/src/modules/video_coding/main/source/er_tables_xor.h
deleted file mode 100644
index 99163ba..0000000
--- a/trunk/src/modules/video_coding/main/source/er_tables_xor.h
+++ /dev/null
@@ -1,38742 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_ER_TABLES_XOR_H_
-#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_ER_TABLES_XOR_H_
-
-// This is a private header for media_opt_util.cc.
-// It should not be included by other files.
-
-namespace webrtc {
-
-// Table for average FEC recovery from packet loss, for XOR code.
-// From RPL model of random loss.
-// Input is the received packet loss (up to 50%), and FEC code parameters
-// (up to 24x24):
-// i.e., kAvgFECRecoveryXOR[k] where k = code_i*129 + loss_j;
-// code_i=1x1,2x1,2x2,..24x24, loss_j = 0,1,..128.
-
-// Maximum number of source packets in off-line model
-static const int kMaxNumPackets = 24;
-// Max value of loss rates in off-line model
-static const int kPacketLossMax = 129;
-
-// Table size for model is: kPacketLossMax * numberOfFecCodes = 38700
-// numberOfFecCodes is determined as:
-// {(1,1), (2,1), (2,2),...(n,1),..(n,n-1), (n,n)} = n*(n+1)/2
-// for n = kMaxNumPackets.
-static const int kSizeAvgFECRecoveryXOR = 38700;
-static const unsigned char kAvgFECRecoveryXOR[kSizeAvgFECRecoveryXOR] = {
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-11,
-12,
-13,
-14,
-15,
-16,
-17,
-18,
-20,
-21,
-22,
-23,
-25,
-26,
-27,
-29,
-30,
-31,
-32,
-33,
-35,
-36,
-37,
-38,
-39,
-40,
-41,
-42,
-43,
-43,
-44,
-45,
-45,
-46,
-46,
-47,
-47,
-47,
-47,
-48,
-48,
-48,
-48,
-48,
-47,
-47,
-47,
-47,
-46,
-46,
-45,
-45,
-44,
-44,
-43,
-42,
-42,
-41,
-40,
-39,
-39,
-38,
-37,
-36,
-35,
-34,
-33,
-32,
-31,
-30,
-29,
-28,
-27,
-27,
-26,
-25,
-24,
-23,
-22,
-21,
-20,
-19,
-18,
-18,
-17,
-16,
-
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_ER_TABLES_XOR_H_
diff --git a/trunk/src/modules/video_coding/main/source/event.h b/trunk/src/modules/video_coding/main/source/event.h
deleted file mode 100644
index 39fd494..0000000
--- a/trunk/src/modules/video_coding/main/source/event.h
+++ /dev/null
@@ -1,63 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_EVENT_H_
-#define WEBRTC_MODULES_VIDEO_CODING_EVENT_H_
-
-#include "event_wrapper.h"
-
-namespace webrtc
-{
-
-//#define EVENT_DEBUG
-
-class VCMEvent : public EventWrapper
-{
-public:
- VCMEvent() : _event(*EventWrapper::Create()) {};
-
- virtual ~VCMEvent() { delete &_event; };
-
- /**
- * Release waiting threads
- */
- bool Set() { return _event.Set(); };
-
- bool Reset() { return _event.Reset(); };
-
- /**
- * Wait for this event
- */
- EventTypeWrapper Wait(unsigned long maxTime)
- {
-#ifdef EVENT_DEBUG
- return kEventTimeout;
-#else
- return _event.Wait(maxTime);
-#endif
- };
-
- /**
- * Start a timer
- */
- bool StartTimer(bool periodic, unsigned long time)
- { return _event.StartTimer(periodic, time); };
- /**
- * Stop the timer
- */
- bool StopTimer() { return _event.StopTimer(); };
-
-private:
- EventWrapper& _event;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_EVENT_H_
diff --git a/trunk/src/modules/video_coding/main/source/exp_filter.cc b/trunk/src/modules/video_coding/main/source/exp_filter.cc
deleted file mode 100644
index 1d6f9a7..0000000
--- a/trunk/src/modules/video_coding/main/source/exp_filter.cc
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "exp_filter.h"
-
-#include <math.h>
-
-namespace webrtc {
-
-void
-VCMExpFilter::Reset(float alpha)
-{
- _alpha = alpha;
- _filtered = -1.0;
-}
-
-float
-VCMExpFilter::Apply(float exp, float sample)
-{
- if (_filtered == -1.0)
- {
- // Initialize filtered bit rates
- _filtered = sample;
- }
- else if (exp == 1.0)
- {
- _filtered = _alpha * _filtered + (1 - _alpha) * sample;
- }
- else
- {
- float alpha = pow(_alpha, exp);
- _filtered = alpha * _filtered + (1 - alpha) * sample;
- }
- if (_max != -1 && _filtered > _max)
- {
- _filtered = _max;
- }
- return _filtered;
-}
-
-void
-VCMExpFilter::UpdateBase(float alpha)
-{
- _alpha = alpha;
-}
-
-float
-VCMExpFilter::Value() const
-{
- return _filtered;
-}
-
-}
diff --git a/trunk/src/modules/video_coding/main/source/exp_filter.h b/trunk/src/modules/video_coding/main/source/exp_filter.h
deleted file mode 100644
index 46d206a..0000000
--- a/trunk/src/modules/video_coding/main/source/exp_filter.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_EXP_FILTER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_EXP_FILTER_H_
-
-namespace webrtc
-{
-
-/**********************/
-/* ExpFilter class */
-/**********************/
-
-class VCMExpFilter
-{
-public:
- VCMExpFilter(float alpha, float max = -1.0) : _alpha(alpha), _filtered(-1.0), _max(max) {}
-
- // Resets the filter to its initial state, and resets alpha to the given value
- //
- // Input:
- // - alpha : the new value of the filter factor base.
- void Reset(float alpha);
-
- // Applies the filter with the given exponent on the provided sample
- //
- // Input:
- // - exp : Exponent T in y(k) = alpha^T * y(k-1) + (1 - alpha^T) * x(k)
- // - sample : x(k) in the above filter equation
- float Apply(float exp, float sample);
-
- // Return current filtered value: y(k)
- //
- // Return value : The current filter output
- float Value() const;
-
- // Change the filter factor base
- //
- // Input:
- // - alpha : The new filter factor base.
- void UpdateBase(float alpha);
-
-private:
- float _alpha; // Filter factor base
- float _filtered; // Current filter output
- const float _max;
-}; // end of ExpFilter class
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_EXP_FILTER_H_
diff --git a/trunk/src/modules/video_coding/main/source/fec_tables_xor.h b/trunk/src/modules/video_coding/main/source/fec_tables_xor.h
deleted file mode 100644
index 27db9a4..0000000
--- a/trunk/src/modules/video_coding/main/source/fec_tables_xor.h
+++ /dev/null
@@ -1,6481 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_FEC_TABLES_XOR_H_
-#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_FEC_TABLES_XOR_H_
-
-// This is a private header for media_opt_util.cc.
-// It should not be included by other files.
-
-namespace webrtc {
-
-// Table for Protection factor (code rate) of delta frames, for the XOR FEC.
-// Input is the packet loss and an effective rate (bits/frame).
-// Output is array kCodeRateXORTable[k], where k = rate_i*129 + loss_j;
-// loss_j = 0,1,..128, and rate_i varies over some range.
-static const int kSizeCodeRateXORTable = 6450;
-static const unsigned char kCodeRateXORTable[kSizeCodeRateXORTable] = {
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-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-0,
-14,
-28,
-42,
-56,
-56,
-65,
-66,
-76,
-76,
-92,
-92,
-92,
-97,
-97,
-97,
-101,
-101,
-101,
-101,
-101,
-106,
-121,
-121,
-121,
-126,
-126,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-0,
-13,
-27,
-41,
-55,
-55,
-67,
-72,
-74,
-74,
-90,
-90,
-90,
-91,
-91,
-91,
-105,
-105,
-105,
-105,
-105,
-107,
-122,
-122,
-122,
-123,
-123,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-0,
-13,
-27,
-40,
-54,
-54,
-67,
-76,
-76,
-76,
-85,
-85,
-85,
-85,
-85,
-85,
-112,
-112,
-112,
-112,
-112,
-112,
-121,
-121,
-121,
-121,
-121,
-126,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-127,
-
-
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_FEC_TABLES_XOR_H_
diff --git a/trunk/src/modules/video_coding/main/source/frame_buffer.cc b/trunk/src/modules/video_coding/main/source/frame_buffer.cc
deleted file mode 100644
index 6339b05..0000000
--- a/trunk/src/modules/video_coding/main/source/frame_buffer.cc
+++ /dev/null
@@ -1,415 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "frame_buffer.h"
-#include "packet.h"
-
-#include <cassert>
-#include <string.h>
-
-#if defined(_WIN32)
- // VS 2005: Don't warn for default initialized arrays. See help for more info.
- #pragma warning(disable:4351)
-#endif
-
-namespace webrtc {
-
-VCMFrameBuffer::VCMFrameBuffer()
- :
- _state(kStateFree),
- _frameCounted(false),
- _nackCount(0),
- _latestPacketTimeMs(-1) {
-}
-
-VCMFrameBuffer::~VCMFrameBuffer() {
-}
-
-VCMFrameBuffer::VCMFrameBuffer(VCMFrameBuffer& rhs)
-:
-VCMEncodedFrame(rhs),
-_state(rhs._state),
-_frameCounted(rhs._frameCounted),
-_sessionInfo(),
-_nackCount(rhs._nackCount),
-_latestPacketTimeMs(rhs._latestPacketTimeMs)
-{
- _sessionInfo = rhs._sessionInfo;
- _sessionInfo.UpdateDataPointers(rhs._buffer, _buffer);
-}
-
-webrtc::FrameType
-VCMFrameBuffer::FrameType() const
-{
- return _sessionInfo.FrameType();
-}
-
-void
-VCMFrameBuffer::SetPreviousFrameLoss()
-{
- _sessionInfo.SetPreviousFrameLoss();
-}
-
-WebRtc_Word32
-VCMFrameBuffer::GetLowSeqNum() const
-{
- return _sessionInfo.LowSequenceNumber();
-}
-
-WebRtc_Word32
-VCMFrameBuffer::GetHighSeqNum() const
-{
- return _sessionInfo.HighSequenceNumber();
-}
-
-int VCMFrameBuffer::PictureId() const {
- return _sessionInfo.PictureId();
-}
-
-int VCMFrameBuffer::TemporalId() const {
- return _sessionInfo.TemporalId();
-}
-
-bool VCMFrameBuffer::LayerSync() const {
- return _sessionInfo.LayerSync();
-}
-
-int VCMFrameBuffer::Tl0PicId() const {
- return _sessionInfo.Tl0PicId();
-}
-
-bool VCMFrameBuffer::NonReference() const {
- return _sessionInfo.NonReference();
-}
-
-bool
-VCMFrameBuffer::IsSessionComplete() const
-{
- return _sessionInfo.complete();
-}
-
-// Insert packet
-VCMFrameBufferEnum
-VCMFrameBuffer::InsertPacket(const VCMPacket& packet, WebRtc_Word64 timeInMs,
- bool enableDecodableState, WebRtc_UWord32 rttMS)
-{
- if (_state == kStateDecoding)
- {
- // Do not insert packet
- return kNoError;
- }
-
- // Sanity to check if the frame has been freed. (Too old for example)
- if (_state == kStateFree)
- {
- return kStateError;
- }
-
- // is this packet part of this frame
- if (TimeStamp() && (TimeStamp() != packet.timestamp))
- {
- return kTimeStampError;
- }
-
- // sanity checks
- if (_size + packet.sizeBytes +
- (packet.insertStartCode ? kH264StartCodeLengthBytes : 0 )
- > kMaxJBFrameSizeBytes)
- {
- return kSizeError;
- }
- if (NULL == packet.dataPtr && packet.sizeBytes > 0)
- {
- return kSizeError;
- }
- if (packet.dataPtr != NULL)
- {
- _payloadType = packet.payloadType;
- }
-
- if (kStateEmpty == _state)
- {
- // First packet (empty and/or media) inserted into this frame.
- // store some info and set some initial values.
- _timeStamp = packet.timestamp;
- _codec = packet.codec;
- if (packet.frameType != kFrameEmpty)
- {
- // first media packet
- SetState(kStateIncomplete);
- }
- }
-
- WebRtc_UWord32 requiredSizeBytes = Length() + packet.sizeBytes +
- (packet.insertStartCode ? kH264StartCodeLengthBytes : 0);
- if (requiredSizeBytes >= _size)
- {
- const WebRtc_UWord8* prevBuffer = _buffer;
- const WebRtc_UWord32 increments = requiredSizeBytes /
- kBufferIncStepSizeBytes +
- (requiredSizeBytes %
- kBufferIncStepSizeBytes > 0);
- const WebRtc_UWord32 newSize = _size +
- increments * kBufferIncStepSizeBytes;
- if (newSize > kMaxJBFrameSizeBytes)
- {
- return kSizeError;
- }
- if (VerifyAndAllocate(newSize) == -1)
- {
- return kSizeError;
- }
- _sessionInfo.UpdateDataPointers(prevBuffer, _buffer);
- }
-
- CopyCodecSpecific(&packet.codecSpecificHeader);
-
- int retVal = _sessionInfo.InsertPacket(packet, _buffer,
- enableDecodableState,
- rttMS);
- if (retVal == -1)
- {
- return kSizeError;
- }
- else if (retVal == -2)
- {
- return kDuplicatePacket;
- }
- // update length
- _length = Length() + static_cast<WebRtc_UWord32>(retVal);
-
- _latestPacketTimeMs = timeInMs;
-
- if (_sessionInfo.complete()) {
- return kCompleteSession;
- } else if (_sessionInfo.decodable()) {
- SetState(kStateDecodable);
- return kDecodableSession;
- } else {
- // this layer is not complete
- if (_state == kStateComplete) {
- // we already have a complete layer
- // wait for all independent layers belonging to the same frame
- _state = kStateIncomplete;
- }
- }
- return kIncomplete;
-}
-
-WebRtc_Word64
-VCMFrameBuffer::LatestPacketTimeMs()
-{
- return _latestPacketTimeMs;
-}
-
-// Build hard NACK list:Zero out all entries in list up to and including the
-// (first) entry equal to _lowSeqNum.
-int VCMFrameBuffer::BuildHardNackList(int* list, int num) {
- if (_sessionInfo.BuildHardNackList(list, num) != 0) {
- return -1;
- }
- return 0;
-}
-
-// Build selective NACK list: Create a soft (selective) list of entries to zero
-// out up to and including the (first) entry equal to _lowSeqNum.
-int VCMFrameBuffer::BuildSoftNackList(int* list, int num, int rttMs) {
- return _sessionInfo.BuildSoftNackList(list, num, rttMs);
-}
-
-void
-VCMFrameBuffer::IncrementNackCount()
-{
- _nackCount++;
-}
-
-WebRtc_Word16
-VCMFrameBuffer::GetNackCount() const
-{
- return _nackCount;
-}
-
-bool
-VCMFrameBuffer::HaveLastPacket() const
-{
- return _sessionInfo.HaveLastPacket();
-}
-
-void
-VCMFrameBuffer::Reset()
-{
- _length = 0;
- _timeStamp = 0;
- _sessionInfo.Reset();
- _frameCounted = false;
- _payloadType = 0;
- _nackCount = 0;
- _latestPacketTimeMs = -1;
- _state = kStateFree;
- VCMEncodedFrame::Reset();
-}
-
-// Makes sure the session contains a decodable stream.
-void
-VCMFrameBuffer::MakeSessionDecodable()
-{
- WebRtc_UWord32 retVal;
-#ifdef INDEPENDENT_PARTITIONS
- if (_codec != kVideoCodecVP8) {
- retVal = _sessionInfo.MakeDecodable();
- _length -= retVal;
- }
-#else
- retVal = _sessionInfo.MakeDecodable();
- _length -= retVal;
-#endif
-}
-
-// Set state of frame
-void
-VCMFrameBuffer::SetState(VCMFrameBufferStateEnum state)
-{
- if (_state == state)
- {
- return;
- }
- switch (state)
- {
- case kStateFree:
- // Reset everything
- // We can go to this state from all other states.
- // The one setting the state to free must ensure
- // that the frame is removed from the timestamp
- // ordered frame list in the jb.
- Reset();
- break;
-
- case kStateIncomplete:
- // we can go to this state from state kStateEmpty
- assert(_state == kStateEmpty ||
- _state == kStateDecoding);
-
- // Do nothing, we received a packet
- break;
-
- case kStateComplete:
- assert(_state == kStateEmpty ||
- _state == kStateIncomplete ||
- _state == kStateDecodable);
-
- break;
-
- case kStateEmpty:
- assert(_state == kStateFree);
- // Do nothing
- break;
-
- case kStateDecoding:
- // A frame might have received empty packets, or media packets might
- // have been removed when making the frame decodable. The frame can
- // still be set to decodable since it can be used to inform the
- // decoder of a frame loss.
- assert(_state == kStateComplete || _state == kStateIncomplete ||
- _state == kStateDecodable || _state == kStateEmpty);
- // Transfer frame information to EncodedFrame and create any codec
- // specific information
- RestructureFrameInformation();
- break;
-
- case kStateDecodable:
- assert(_state == kStateEmpty ||
- _state == kStateIncomplete);
- break;
- }
- _state = state;
-}
-
-void
-VCMFrameBuffer::RestructureFrameInformation()
-{
- PrepareForDecode();
- _frameType = ConvertFrameType(_sessionInfo.FrameType());
- _completeFrame = _sessionInfo.complete();
- _missingFrame = _sessionInfo.PreviousFrameLoss();
-}
-
-WebRtc_Word32
-VCMFrameBuffer::ExtractFromStorage(const EncodedVideoData& frameFromStorage)
-{
- _frameType = ConvertFrameType(frameFromStorage.frameType);
- _timeStamp = frameFromStorage.timeStamp;
- _payloadType = frameFromStorage.payloadType;
- _encodedWidth = frameFromStorage.encodedWidth;
- _encodedHeight = frameFromStorage.encodedHeight;
- _missingFrame = frameFromStorage.missingFrame;
- _completeFrame = frameFromStorage.completeFrame;
- _renderTimeMs = frameFromStorage.renderTimeMs;
- _codec = frameFromStorage.codec;
- const WebRtc_UWord8 *prevBuffer = _buffer;
- if (VerifyAndAllocate(frameFromStorage.payloadSize) < 0)
- {
- return VCM_MEMORY;
- }
- _sessionInfo.UpdateDataPointers(prevBuffer, _buffer);
- memcpy(_buffer, frameFromStorage.payloadData, frameFromStorage.payloadSize);
- _length = frameFromStorage.payloadSize;
- return VCM_OK;
-}
-
-int VCMFrameBuffer::NotDecodablePackets() const {
- return _sessionInfo.packets_not_decodable();
-}
-
-// Set counted status (as counted by JB or not)
-void VCMFrameBuffer::SetCountedFrame(bool frameCounted)
-{
- _frameCounted = frameCounted;
-}
-
-bool VCMFrameBuffer::GetCountedFrame() const
-{
- return _frameCounted;
-}
-
-// Get current state of frame
-VCMFrameBufferStateEnum
-VCMFrameBuffer::GetState() const
-{
- return _state;
-}
-
-// Get current state of frame
-VCMFrameBufferStateEnum
-VCMFrameBuffer::GetState(WebRtc_UWord32& timeStamp) const
-{
- timeStamp = TimeStamp();
- return GetState();
-}
-
-bool
-VCMFrameBuffer::IsRetransmitted() const
-{
- return _sessionInfo.session_nack();
-}
-
-void
-VCMFrameBuffer::PrepareForDecode()
-{
-#ifdef INDEPENDENT_PARTITIONS
- if (_codec == kVideoCodecVP8)
- {
- _length =
- _sessionInfo.BuildVP8FragmentationHeader(_buffer, _length,
- &_fragmentation);
- }
-#endif
-}
-
-}
diff --git a/trunk/src/modules/video_coding/main/source/frame_buffer.h b/trunk/src/modules/video_coding/main/source/frame_buffer.h
deleted file mode 100644
index ea05754..0000000
--- a/trunk/src/modules/video_coding/main/source/frame_buffer.h
+++ /dev/null
@@ -1,102 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
-
-#include "modules/interface/module_common_types.h"
-#include "modules/video_coding/main/source/encoded_frame.h"
-#include "modules/video_coding/main/source/jitter_buffer_common.h"
-#include "modules/video_coding/main/source/session_info.h"
-#include "typedefs.h"
-
-namespace webrtc
-{
-
-class VCMFrameBuffer : public VCMEncodedFrame
-{
-public:
- VCMFrameBuffer();
- virtual ~VCMFrameBuffer();
-
- VCMFrameBuffer(VCMFrameBuffer& rhs);
-
- virtual void Reset();
-
- VCMFrameBufferEnum InsertPacket(const VCMPacket& packet,
- WebRtc_Word64 timeInMs,
- bool enableDecodableState,
- WebRtc_UWord32 rttMs);
-
- // State
- // Get current state of frame
- VCMFrameBufferStateEnum GetState() const;
- // Get current state and timestamp of frame
- VCMFrameBufferStateEnum GetState(WebRtc_UWord32& timeStamp) const;
- void SetState(VCMFrameBufferStateEnum state); // Set state of frame
-
- bool IsRetransmitted() const;
- bool IsSessionComplete() const;
- bool HaveLastPacket() const;
- // Makes sure the session contain a decodable stream.
- void MakeSessionDecodable();
-
- // Sequence numbers
- // Get lowest packet sequence number in frame
- WebRtc_Word32 GetLowSeqNum() const;
- // Get highest packet sequence number in frame
- WebRtc_Word32 GetHighSeqNum() const;
-
- int PictureId() const;
- int TemporalId() const;
- bool LayerSync() const;
- int Tl0PicId() const;
- bool NonReference() const;
-
- // Set counted status (as counted by JB or not)
- void SetCountedFrame(bool frameCounted);
- bool GetCountedFrame() const;
-
- // NACK - Building the NACK lists.
- // Build hard NACK list: Zero out all entries in list up to and including
- // _lowSeqNum.
- int BuildHardNackList(int* list, int num);
- // Build soft NACK list: Zero out only a subset of the packets, discard
- // empty packets.
- int BuildSoftNackList(int* list, int num, int rttMs);
- void IncrementNackCount();
- WebRtc_Word16 GetNackCount() const;
-
- WebRtc_Word64 LatestPacketTimeMs();
-
- webrtc::FrameType FrameType() const;
- void SetPreviousFrameLoss();
-
- WebRtc_Word32 ExtractFromStorage(const EncodedVideoData& frameFromStorage);
-
- // The number of packets discarded because the decoder can't make use of
- // them.
- int NotDecodablePackets() const;
-
-protected:
- void RestructureFrameInformation();
- void PrepareForDecode();
-
-private:
- VCMFrameBufferStateEnum _state; // Current state of the frame
- bool _frameCounted; // Was this frame counted by JB?
- VCMSessionInfo _sessionInfo;
- WebRtc_UWord16 _nackCount;
- WebRtc_Word64 _latestPacketTimeMs;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_FRAME_BUFFER_H_
diff --git a/trunk/src/modules/video_coding/main/source/frame_dropper.cc b/trunk/src/modules/video_coding/main/source/frame_dropper.cc
deleted file mode 100644
index 065e452..0000000
--- a/trunk/src/modules/video_coding/main/source/frame_dropper.cc
+++ /dev/null
@@ -1,331 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "frame_dropper.h"
-#include "internal_defines.h"
-#include "trace.h"
-
-namespace webrtc
-{
-
-VCMFrameDropper::VCMFrameDropper(WebRtc_Word32 vcmId)
-:
-_vcmId(vcmId),
-_keyFrameSizeAvgKbits(0.9f),
-_keyFrameRatio(0.99f),
-_dropRatio(0.9f, 0.96f)
-{
- Reset();
-}
-
-void
-VCMFrameDropper::Reset()
-{
- _keyFrameRatio.Reset(0.99f);
- _keyFrameRatio.Apply(1.0f, 1.0f/300.0f); // 1 key frame every 10th second in 30 fps
- _keyFrameSizeAvgKbits.Reset(0.9f);
- _keyFrameCount = 0;
- _accumulator = 0.0f;
- _accumulatorMax = 150.0f; // assume 300 kb/s and 0.5 s window
- _targetBitRate = 300.0f;
- _userFrameRate = 30;
- _keyFrameSpreadFrames = 0.5f * _userFrameRate;
- _dropNext = false;
- _dropRatio.Reset(0.9f);
- _dropRatio.Apply(0.0f, 0.0f); // Initialize to 0
- _dropCount = 0;
- _windowSize = 0.5f;
- _wasBelowMax = true;
- _enabled = true;
- _fastMode = false; // start with normal (non-aggressive) mode
-}
-
-void
-VCMFrameDropper::Enable(bool enable)
-{
- _enabled = enable;
-}
-
-void
-VCMFrameDropper::Fill(WebRtc_UWord32 frameSizeBytes, bool deltaFrame)
-{
- if (!_enabled)
- {
- return;
- }
- float frameSizeKbits = 8.0f * static_cast<float>(frameSizeBytes) / 1000.0f;
- if (!deltaFrame && !_fastMode) // fast mode does not treat key-frames any different
- {
- _keyFrameSizeAvgKbits.Apply(1, frameSizeKbits);
- _keyFrameRatio.Apply(1.0, 1.0);
- if (frameSizeKbits > _keyFrameSizeAvgKbits.Value())
- {
- // Remove the average key frame size since we
- // compensate for key frames when adding delta
- // frames.
- frameSizeKbits -= _keyFrameSizeAvgKbits.Value();
- }
- else
- {
- // Shouldn't be negative, so zero is the lower bound.
- frameSizeKbits = 0;
- }
- if (_keyFrameRatio.Value() > 1e-5 && 1 / _keyFrameRatio.Value() < _keyFrameSpreadFrames)
- {
- // We are sending key frames more often than our upper bound for
- // how much we allow the key frame compensation to be spread
- // out in time. Therefor we must use the key frame ratio rather
- // than keyFrameSpreadFrames.
- _keyFrameCount = static_cast<WebRtc_Word32>(1 / _keyFrameRatio.Value() + 0.5);
- }
- else
- {
- // Compensate for the key frame the following frames
- _keyFrameCount = static_cast<WebRtc_Word32>(_keyFrameSpreadFrames + 0.5);
- }
- }
- else
- {
- // Decrease the keyFrameRatio
- _keyFrameRatio.Apply(1.0, 0.0);
- }
- // Change the level of the accumulator (bucket)
- _accumulator += frameSizeKbits;
-}
-
-void
-VCMFrameDropper::Leak(WebRtc_UWord32 inputFrameRate)
-{
- if (!_enabled)
- {
- return;
- }
- if (inputFrameRate < 1)
- {
- return;
- }
- if (_targetBitRate < 0.0f)
- {
- return;
- }
- _keyFrameSpreadFrames = 0.5f * inputFrameRate;
- // T is the expected bits per frame (target). If all frames were the same size,
- // we would get T bits per frame. Notice that T is also weighted to be able to
- // force a lower frame rate if wanted.
- float T = _targetBitRate / inputFrameRate;
- if (_keyFrameCount > 0)
- {
- // Perform the key frame compensation
- if (_keyFrameRatio.Value() > 0 && 1 / _keyFrameRatio.Value() < _keyFrameSpreadFrames)
- {
- T -= _keyFrameSizeAvgKbits.Value() * _keyFrameRatio.Value();
- }
- else
- {
- T -= _keyFrameSizeAvgKbits.Value() / _keyFrameSpreadFrames;
- }
- _keyFrameCount--;
- }
- _accumulator -= T;
- UpdateRatio();
-
-}
-
-void
-VCMFrameDropper::UpdateNack(WebRtc_UWord32 nackBytes)
-{
- if (!_enabled)
- {
- return;
- }
- _accumulator += static_cast<float>(nackBytes) * 8.0f / 1000.0f;
-}
-
-void
-VCMFrameDropper::FillBucket(float inKbits, float outKbits)
-{
- _accumulator += (inKbits - outKbits);
-}
-
-void
-VCMFrameDropper::UpdateRatio()
-{
- if (_accumulator > 1.3f * _accumulatorMax)
- {
- // Too far above accumulator max, react faster
- _dropRatio.UpdateBase(0.8f);
- }
- else
- {
- // Go back to normal reaction
- _dropRatio.UpdateBase(0.9f);
- }
- if (_accumulator > _accumulatorMax)
- {
- // We are above accumulator max, and should ideally
- // drop a frame. Increase the dropRatio and drop
- // the frame later.
- if (_wasBelowMax)
- {
- _dropNext = true;
- }
- if (_fastMode)
- {
- // always drop in aggressive mode
- _dropNext = true;
- }
-
- _dropRatio.Apply(1.0f, 1.0f);
- _dropRatio.UpdateBase(0.9f);
- }
- else
- {
- _dropRatio.Apply(1.0f, 0.0f);
- }
- if (_accumulator < 0.0f)
- {
- _accumulator = 0.0f;
- }
- _wasBelowMax = _accumulator < _accumulatorMax;
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId), "FrameDropper: dropRatio = %f accumulator = %f, accumulatorMax = %f", _dropRatio.Value(), _accumulator, _accumulatorMax);
-}
-
-// This function signals when to drop frames to the caller. It makes use of the dropRatio
-// to smooth out the drops over time.
-bool
-VCMFrameDropper::DropFrame()
-{
- if (!_enabled)
- {
- return false;
- }
- if (_dropNext)
- {
- _dropNext = false;
- _dropCount = 0;
- }
-
- if (_dropRatio.Value() >= 0.5f) // Drops per keep
- {
- // limit is the number of frames we should drop between each kept frame
- // to keep our drop ratio. limit is positive in this case.
- float denom = 1.0f - _dropRatio.Value();
- if (denom < 1e-5)
- {
- denom = (float)1e-5;
- }
- WebRtc_Word32 limit = static_cast<WebRtc_Word32>(1.0f / denom - 1.0f + 0.5f);
- if (_dropCount < 0)
- {
- // Reset the _dropCount since it was negative and should be positive.
- if (_dropRatio.Value() > 0.4f)
- {
- _dropCount = -_dropCount;
- }
- else
- {
- _dropCount = 0;
- }
- }
- if (_dropCount < limit)
- {
- // As long we are below the limit we should drop frames.
- _dropCount++;
- return true;
- }
- else
- {
- // Only when we reset _dropCount a frame should be kept.
- _dropCount = 0;
- return false;
- }
- }
- else if (_dropRatio.Value() > 0.0f && _dropRatio.Value() < 0.5f) // Keeps per drop
- {
- // limit is the number of frames we should keep between each drop
- // in order to keep the drop ratio. limit is negative in this case,
- // and the _dropCount is also negative.
- float denom = _dropRatio.Value();
- if (denom < 1e-5)
- {
- denom = (float)1e-5;
- }
- WebRtc_Word32 limit = -static_cast<WebRtc_Word32>(1.0f / denom - 1.0f + 0.5f);
- if (_dropCount > 0)
- {
- // Reset the _dropCount since we have a positive
- // _dropCount, and it should be negative.
- if (_dropRatio.Value() < 0.6f)
- {
- _dropCount = -_dropCount;
- }
- else
- {
- _dropCount = 0;
- }
- }
- if (_dropCount > limit)
- {
- if (_dropCount == 0)
- {
- // Drop frames when we reset _dropCount.
- _dropCount--;
- return true;
- }
- else
- {
- // Keep frames as long as we haven't reached limit.
- _dropCount--;
- return false;
- }
- }
- else
- {
- _dropCount = 0;
- return false;
- }
- }
- _dropCount = 0;
- return false;
-
- // A simpler version, unfiltered and quicker
- //bool dropNext = _dropNext;
- //_dropNext = false;
- //return dropNext;
-}
-
-void
-VCMFrameDropper::SetRates(float bitRate, float userFrameRate)
-{
- // Bit rate of -1 means infinite bandwidth.
- _accumulatorMax = bitRate * _windowSize; // bitRate * windowSize (in seconds)
- if (_targetBitRate > 0.0f && bitRate < _targetBitRate && _accumulator > _accumulatorMax)
- {
- // Rescale the accumulator level if the accumulator max decreases
- _accumulator = bitRate / _targetBitRate * _accumulator;
- }
- _targetBitRate = bitRate;
- if (userFrameRate > 0.0f)
- {
- _userFrameRate = userFrameRate;
- }
-}
-
-float
-VCMFrameDropper::ActualFrameRate(WebRtc_UWord32 inputFrameRate) const
-{
- if (!_enabled)
- {
- return static_cast<float>(inputFrameRate);
- }
- return inputFrameRate * (1.0f - _dropRatio.Value());
-}
-
-}
diff --git a/trunk/src/modules/video_coding/main/source/frame_dropper.h b/trunk/src/modules/video_coding/main/source/frame_dropper.h
deleted file mode 100644
index 5e7e8a1..0000000
--- a/trunk/src/modules/video_coding/main/source/frame_dropper.h
+++ /dev/null
@@ -1,94 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_FRAME_DROPPER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_FRAME_DROPPER_H_
-
-#include "exp_filter.h"
-#include "typedefs.h"
-
-namespace webrtc
-{
-
-/******************************/
-/* VCMFrameDropper class */
-/****************************/
-// The Frame Dropper implements a variant of the leaky bucket algorithm
-// for keeping track of when to drop frames to avoid bit rate
-// over use when the encoder can't keep its bit rate.
-class VCMFrameDropper
-{
-public:
- VCMFrameDropper(WebRtc_Word32 vcmId = 0);
- // Resets the FrameDropper to its initial state.
- // This means that the frameRateWeight is set to its
- // default value as well.
- void Reset();
-
- void Enable(bool enable);
- // Answers the question if it's time to drop a frame
- // if we want to reach a given frame rate. Must be
- // called for every frame.
- //
- // Return value : True if we should drop the current frame
- bool DropFrame();
- // Updates the FrameDropper with the size of the latest encoded
- // frame. The FrameDropper calculates a new drop ratio (can be
- // seen as the probability to drop a frame) and updates its
- // internal statistics.
- //
- // Input:
- // - frameSizeBytes : The size of the latest frame
- // returned from the encoder.
- // - deltaFrame : True if the encoder returned
- // a key frame.
- void Fill(WebRtc_UWord32 frameSizeBytes, bool deltaFrame);
-
- void Leak(WebRtc_UWord32 inputFrameRate);
-
- void UpdateNack(WebRtc_UWord32 nackBytes);
-
- // Sets the target bit rate and the frame rate produced by
- // the camera.
- //
- // Input:
- // - bitRate : The target bit rate
- void SetRates(float bitRate, float userFrameRate);
-
- // Return value : The current average frame rate produced
- // if the DropFrame() function is used as
- // instruction of when to drop frames.
- float ActualFrameRate(WebRtc_UWord32 inputFrameRate) const;
-
-private:
- void FillBucket(float inKbits, float outKbits);
- void UpdateRatio();
-
- WebRtc_Word32 _vcmId;
- VCMExpFilter _keyFrameSizeAvgKbits;
- VCMExpFilter _keyFrameRatio;
- float _keyFrameSpreadFrames;
- WebRtc_Word32 _keyFrameCount;
- float _accumulator;
- float _accumulatorMax;
- float _targetBitRate;
- bool _dropNext;
- VCMExpFilter _dropRatio;
- WebRtc_Word32 _dropCount;
- float _windowSize;
- float _userFrameRate;
- bool _wasBelowMax;
- bool _enabled;
- bool _fastMode;
-}; // end of VCMFrameDropper class
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_FRAME_DROPPER_H_
diff --git a/trunk/src/modules/video_coding/main/source/generic_decoder.cc b/trunk/src/modules/video_coding/main/source/generic_decoder.cc
deleted file mode 100644
index 217c21d..0000000
--- a/trunk/src/modules/video_coding/main/source/generic_decoder.cc
+++ /dev/null
@@ -1,227 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_coding.h"
-#include "trace.h"
-#include "generic_decoder.h"
-#include "internal_defines.h"
-#include "tick_time_base.h"
-
-namespace webrtc {
-
-VCMDecodedFrameCallback::VCMDecodedFrameCallback(VCMTiming& timing,
- TickTimeBase* clock)
-:
-_critSect(CriticalSectionWrapper::CreateCriticalSection()),
-_clock(clock),
-_receiveCallback(NULL),
-_timing(timing),
-_timestampMap(kDecoderFrameMemoryLength),
-_lastReceivedPictureID(0)
-{
-}
-
-VCMDecodedFrameCallback::~VCMDecodedFrameCallback()
-{
- delete _critSect;
-}
-
-void VCMDecodedFrameCallback::SetUserReceiveCallback(
- VCMReceiveCallback* receiveCallback)
-{
- CriticalSectionScoped cs(_critSect);
- _receiveCallback = receiveCallback;
-}
-
-WebRtc_Word32 VCMDecodedFrameCallback::Decoded(RawImage& decodedImage)
-{
- // TODO(holmer): We should improve this so that we can handle multiple
- // callbacks from one call to Decode().
- CriticalSectionScoped cs(_critSect);
- VCMFrameInformation* frameInfo = static_cast<VCMFrameInformation*>(
- _timestampMap.Pop(decodedImage._timeStamp));
- if (frameInfo == NULL)
- {
- // The map should never be empty or full if this callback is called.
- return WEBRTC_VIDEO_CODEC_ERROR;
- }
-
- _timing.StopDecodeTimer(
- decodedImage._timeStamp,
- frameInfo->decodeStartTimeMs,
- _clock->MillisecondTimestamp());
-
- if (_receiveCallback != NULL)
- {
- _frame.Swap(decodedImage._buffer,
- decodedImage._length,
- decodedImage._size);
- _frame.SetWidth(decodedImage._width);
- _frame.SetHeight(decodedImage._height);
- _frame.SetTimeStamp(decodedImage._timeStamp);
- _frame.SetRenderTime(frameInfo->renderTimeMs);
- // Convert raw image to video frame
- WebRtc_Word32 callbackReturn = _receiveCallback->FrameToRender(_frame);
- if (callbackReturn < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug,
- webrtc::kTraceVideoCoding,
- -1,
- "Render callback returned error: %d", callbackReturn);
- }
- }
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-WebRtc_Word32
-VCMDecodedFrameCallback::ReceivedDecodedReferenceFrame(
- const WebRtc_UWord64 pictureId)
-{
- CriticalSectionScoped cs(_critSect);
- if (_receiveCallback != NULL)
- {
- return _receiveCallback->ReceivedDecodedReferenceFrame(pictureId);
- }
- return -1;
-}
-
-WebRtc_Word32
-VCMDecodedFrameCallback::ReceivedDecodedFrame(const WebRtc_UWord64 pictureId)
-{
- _lastReceivedPictureID = pictureId;
- return 0;
-}
-
-WebRtc_UWord64 VCMDecodedFrameCallback::LastReceivedPictureID() const
-{
- return _lastReceivedPictureID;
-}
-
-WebRtc_Word32 VCMDecodedFrameCallback::Map(WebRtc_UWord32 timestamp, VCMFrameInformation* frameInfo)
-{
- CriticalSectionScoped cs(_critSect);
- return _timestampMap.Add(timestamp, frameInfo);
-}
-
-WebRtc_Word32 VCMDecodedFrameCallback::Pop(WebRtc_UWord32 timestamp)
-{
- CriticalSectionScoped cs(_critSect);
- if (_timestampMap.Pop(timestamp) == NULL)
- {
- return VCM_GENERAL_ERROR;
- }
- return VCM_OK;
-}
-
-VCMGenericDecoder::VCMGenericDecoder(VideoDecoder& decoder, WebRtc_Word32 id, bool isExternal)
-:
-_id(id),
-_callback(NULL),
-_frameInfos(),
-_nextFrameInfoIdx(0),
-_decoder(decoder),
-_codecType(kVideoCodecUnknown),
-_isExternal(isExternal),
-_requireKeyFrame(false),
-_keyFrameDecoded(false)
-{
-}
-
-VCMGenericDecoder::~VCMGenericDecoder()
-{
-}
-
-WebRtc_Word32 VCMGenericDecoder::InitDecode(const VideoCodec* settings,
- WebRtc_Word32 numberOfCores,
- bool requireKeyFrame)
-{
- _requireKeyFrame = requireKeyFrame;
- _keyFrameDecoded = false;
- _codecType = settings->codecType;
-
- return _decoder.InitDecode(settings, numberOfCores);
-}
-
-WebRtc_Word32 VCMGenericDecoder::Decode(const VCMEncodedFrame& frame,
- int64_t nowMs)
-{
- if (_requireKeyFrame &&
- !_keyFrameDecoded &&
- frame.FrameType() != kVideoFrameKey &&
- frame.FrameType() != kVideoFrameGolden)
- {
- // Require key frame is enabled, meaning that one key frame must be decoded
- // before we can decode delta frames.
- return VCM_CODEC_ERROR;
- }
- _frameInfos[_nextFrameInfoIdx].decodeStartTimeMs = nowMs;
- _frameInfos[_nextFrameInfoIdx].renderTimeMs = frame.RenderTimeMs();
- _callback->Map(frame.TimeStamp(), &_frameInfos[_nextFrameInfoIdx]);
-
- WEBRTC_TRACE(webrtc::kTraceDebug,
- webrtc::kTraceVideoCoding,
- VCMId(_id),
- "Decoding timestamp %u", frame.TimeStamp());
-
- _nextFrameInfoIdx = (_nextFrameInfoIdx + 1) % kDecoderFrameMemoryLength;
-
- WebRtc_Word32 ret = _decoder.Decode(frame.EncodedImage(),
- frame.MissingFrame(),
- frame.FragmentationHeader(),
- frame.CodecSpecific(),
- frame.RenderTimeMs());
-
- if (ret < WEBRTC_VIDEO_CODEC_OK)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(_id), "Decoder error: %d\n", ret);
- _callback->Pop(frame.TimeStamp());
- return ret;
- }
- else if (ret == WEBRTC_VIDEO_CODEC_NO_OUTPUT ||
- ret == WEBRTC_VIDEO_CODEC_REQUEST_SLI)
- {
- // No output
- _callback->Pop(frame.TimeStamp());
- }
- // Update the key frame decoded variable so that we know whether or not we've decoded a key frame since reset.
- _keyFrameDecoded = (frame.FrameType() == kVideoFrameKey || frame.FrameType() == kVideoFrameGolden);
- return ret;
-}
-
-WebRtc_Word32
-VCMGenericDecoder::Release()
-{
- _keyFrameDecoded = false;
- return _decoder.Release();
-}
-
-WebRtc_Word32 VCMGenericDecoder::Reset()
-{
- _keyFrameDecoded = false;
- return _decoder.Reset();
-}
-
-WebRtc_Word32 VCMGenericDecoder::SetCodecConfigParameters(const WebRtc_UWord8* buffer, WebRtc_Word32 size)
-{
- return _decoder.SetCodecConfigParameters(buffer, size);
-}
-
-WebRtc_Word32 VCMGenericDecoder::RegisterDecodeCompleteCallback(VCMDecodedFrameCallback* callback)
-{
- _callback = callback;
- return _decoder.RegisterDecodeCompleteCallback(callback);
-}
-
-bool VCMGenericDecoder::External() const
-{
- return _isExternal;
-}
-
-} // namespace
diff --git a/trunk/src/modules/video_coding/main/source/generic_decoder.h b/trunk/src/modules/video_coding/main/source/generic_decoder.h
deleted file mode 100644
index 58dabc7..0000000
--- a/trunk/src/modules/video_coding/main/source/generic_decoder.h
+++ /dev/null
@@ -1,121 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
-
-#include "timing.h"
-#include "timestamp_map.h"
-#include "video_codec_interface.h"
-#include "encoded_frame.h"
-#include "module_common_types.h"
-
-namespace webrtc
-{
-
-class VCMReceiveCallback;
-
-enum { kDecoderFrameMemoryLength = 10 };
-
-struct VCMFrameInformation
-{
- WebRtc_Word64 renderTimeMs;
- WebRtc_Word64 decodeStartTimeMs;
- void* userData;
-};
-
-class VCMDecodedFrameCallback : public DecodedImageCallback
-{
-public:
- VCMDecodedFrameCallback(VCMTiming& timing, TickTimeBase* clock);
- virtual ~VCMDecodedFrameCallback();
- void SetUserReceiveCallback(VCMReceiveCallback* receiveCallback);
-
- virtual WebRtc_Word32 Decoded(RawImage& decodedImage);
- virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(const WebRtc_UWord64 pictureId);
- virtual WebRtc_Word32 ReceivedDecodedFrame(const WebRtc_UWord64 pictureId);
-
- WebRtc_UWord64 LastReceivedPictureID() const;
-
- WebRtc_Word32 Map(WebRtc_UWord32 timestamp, VCMFrameInformation* frameInfo);
- WebRtc_Word32 Pop(WebRtc_UWord32 timestamp);
-
-private:
- CriticalSectionWrapper* _critSect;
- TickTimeBase* _clock;
- VideoFrame _frame;
- VCMReceiveCallback* _receiveCallback;
- VCMTiming& _timing;
- VCMTimestampMap _timestampMap;
- WebRtc_UWord64 _lastReceivedPictureID;
-};
-
-
-class VCMGenericDecoder
-{
- friend class VCMCodecDataBase;
-public:
- VCMGenericDecoder(VideoDecoder& decoder, WebRtc_Word32 id = 0, bool isExternal = false);
- ~VCMGenericDecoder();
-
- /**
- * Initialize the decoder with the information from the VideoCodec
- */
- WebRtc_Word32 InitDecode(const VideoCodec* settings,
- WebRtc_Word32 numberOfCores,
- bool requireKeyFrame);
-
- /**
- * Decode to a raw I420 frame,
- *
- * inputVideoBuffer reference to encoded video frame
- */
- WebRtc_Word32 Decode(const VCMEncodedFrame& inputFrame, int64_t nowMs);
-
- /**
- * Free the decoder memory
- */
- WebRtc_Word32 Release();
-
- /**
- * Reset the decoder state, prepare for a new call
- */
- WebRtc_Word32 Reset();
-
- /**
- * Codec configuration data sent out-of-band, i.e. in SIP call setup
- *
- * buffer pointer to the configuration data
- * size the size of the configuration data in bytes
- */
- WebRtc_Word32 SetCodecConfigParameters(const WebRtc_UWord8* /*buffer*/,
- WebRtc_Word32 /*size*/);
-
- WebRtc_Word32 RegisterDecodeCompleteCallback(VCMDecodedFrameCallback* callback);
-
- bool External() const;
-
-protected:
-
- WebRtc_Word32 _id;
- VCMDecodedFrameCallback* _callback;
- VCMFrameInformation _frameInfos[kDecoderFrameMemoryLength];
- WebRtc_UWord32 _nextFrameInfoIdx;
- VideoDecoder& _decoder;
- VideoCodecType _codecType;
- bool _isExternal;
- bool _requireKeyFrame;
- bool _keyFrameDecoded;
-
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_GENERIC_DECODER_H_
diff --git a/trunk/src/modules/video_coding/main/source/generic_encoder.cc b/trunk/src/modules/video_coding/main/source/generic_encoder.cc
deleted file mode 100644
index 58cdd99..0000000
--- a/trunk/src/modules/video_coding/main/source/generic_encoder.cc
+++ /dev/null
@@ -1,280 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "encoded_frame.h"
-#include "generic_encoder.h"
-#include "media_optimization.h"
-#include "../../../../engine_configurations.h"
-
-namespace webrtc {
-
-//#define DEBUG_ENCODER_BIT_STREAM
-
-VCMGenericEncoder::VCMGenericEncoder(VideoEncoder& encoder, bool internalSource /*= false*/)
-:
-_encoder(encoder),
-_codecType(kVideoCodecUnknown),
-_VCMencodedFrameCallback(NULL),
-_bitRate(0),
-_frameRate(0),
-_internalSource(false)
-{
-}
-
-
-VCMGenericEncoder::~VCMGenericEncoder()
-{
-}
-
-WebRtc_Word32 VCMGenericEncoder::Release()
-{
- _bitRate = 0;
- _frameRate = 0;
- _VCMencodedFrameCallback = NULL;
- return _encoder.Release();
-}
-
-WebRtc_Word32
-VCMGenericEncoder::InitEncode(const VideoCodec* settings,
- WebRtc_Word32 numberOfCores,
- WebRtc_UWord32 maxPayloadSize)
-{
- _bitRate = settings->startBitrate;
- _frameRate = settings->maxFramerate;
- _codecType = settings->codecType;
- if (_VCMencodedFrameCallback != NULL)
- {
- _VCMencodedFrameCallback->SetCodecType(_codecType);
- }
- return _encoder.InitEncode(settings, numberOfCores, maxPayloadSize);
-}
-
-WebRtc_Word32
-VCMGenericEncoder::Encode(const VideoFrame& inputFrame,
- const CodecSpecificInfo* codecSpecificInfo,
- FrameType* frameType)
-{
- RawImage rawImage(inputFrame.Buffer(),
- inputFrame.Length(),
- inputFrame.Size());
- rawImage._width = inputFrame.Width();
- rawImage._height = inputFrame.Height();
- rawImage._timeStamp = inputFrame.TimeStamp();
-
- VideoFrameType videoFrameTypes[kMaxSimulcastStreams];
- for (int i = 0; i < kMaxSimulcastStreams; i++)
- {
- videoFrameTypes[i] = VCMEncodedFrame::ConvertFrameType(frameType[i]);
- }
- return _encoder.Encode(rawImage, codecSpecificInfo, videoFrameTypes);
-}
-
-WebRtc_Word32
-VCMGenericEncoder::SetChannelParameters(WebRtc_Word32 packetLoss, int rtt)
-{
- return _encoder.SetChannelParameters(packetLoss, rtt);
-}
-
-WebRtc_Word32
-VCMGenericEncoder::SetRates(WebRtc_UWord32 newBitRate, WebRtc_UWord32 frameRate)
-{
- WebRtc_Word32 ret = _encoder.SetRates(newBitRate, frameRate);
- if (ret < 0)
- {
- return ret;
- }
- _bitRate = newBitRate;
- _frameRate = frameRate;
- return VCM_OK;
-}
-
-WebRtc_Word32
-VCMGenericEncoder::CodecConfigParameters(WebRtc_UWord8* buffer, WebRtc_Word32 size)
-{
- WebRtc_Word32 ret = _encoder.CodecConfigParameters(buffer, size);
- if (ret < 0)
- {
- return ret;
- }
- return ret;
-}
-
-WebRtc_UWord32 VCMGenericEncoder::BitRate() const
-{
- return _bitRate;
-}
-
-WebRtc_UWord32 VCMGenericEncoder::FrameRate() const
-{
- return _frameRate;
-}
-
-WebRtc_Word32
-VCMGenericEncoder::SetPeriodicKeyFrames(bool enable)
-{
- return _encoder.SetPeriodicKeyFrames(enable);
-}
-
-WebRtc_Word32
-VCMGenericEncoder::RequestFrame(FrameType* frameTypes)
-{
- RawImage image;
- VideoFrameType videoFrameTypes[kMaxSimulcastStreams];
- for (int i = 0; i < kMaxSimulcastStreams; i++)
- {
- videoFrameTypes[i] = VCMEncodedFrame::ConvertFrameType(frameTypes[i]);
- }
- return _encoder.Encode(image, NULL, videoFrameTypes);
-}
-
-WebRtc_Word32
-VCMGenericEncoder::RegisterEncodeCallback(VCMEncodedFrameCallback* VCMencodedFrameCallback)
-{
- _VCMencodedFrameCallback = VCMencodedFrameCallback;
-
- _VCMencodedFrameCallback->SetCodecType(_codecType);
- _VCMencodedFrameCallback->SetInternalSource(_internalSource);
- return _encoder.RegisterEncodeCompleteCallback(_VCMencodedFrameCallback);
-}
-
-bool
-VCMGenericEncoder::InternalSource() const
-{
- return _internalSource;
-}
-
- /***************************
- * Callback Implementation
- ***************************/
-VCMEncodedFrameCallback::VCMEncodedFrameCallback():
-_sendCallback(),
-_mediaOpt(NULL),
-_encodedBytes(0),
-_payloadType(0),
-_bitStreamAfterEncoder(NULL)
-{
-#ifdef DEBUG_ENCODER_BIT_STREAM
- _bitStreamAfterEncoder = fopen("encoderBitStream.bit", "wb");
-#endif
-}
-
-VCMEncodedFrameCallback::~VCMEncodedFrameCallback()
-{
-#ifdef DEBUG_ENCODER_BIT_STREAM
- fclose(_bitStreamAfterEncoder);
-#endif
-}
-
-WebRtc_Word32
-VCMEncodedFrameCallback::SetTransportCallback(VCMPacketizationCallback* transport)
-{
- _sendCallback = transport;
- return VCM_OK;
-}
-
-WebRtc_Word32
-VCMEncodedFrameCallback::Encoded(
- EncodedImage &encodedImage,
- const CodecSpecificInfo* codecSpecificInfo,
- const RTPFragmentationHeader* fragmentationHeader)
-{
- FrameType frameType = VCMEncodedFrame::ConvertFrameType(encodedImage._frameType);
-
- WebRtc_UWord32 encodedBytes = 0;
- if (_sendCallback != NULL)
- {
- encodedBytes = encodedImage._length;
-
-#ifdef DEBUG_ENCODER_BIT_STREAM
- if (_bitStreamAfterEncoder != NULL)
- {
- fwrite(encodedImage._buffer, 1, encodedImage._length, _bitStreamAfterEncoder);
- }
-#endif
-
- RTPVideoHeader rtpVideoHeader;
- RTPVideoHeader* rtpVideoHeaderPtr = &rtpVideoHeader;
- if (codecSpecificInfo)
- {
- CopyCodecSpecific(*codecSpecificInfo, &rtpVideoHeaderPtr);
- }
- else
- {
- rtpVideoHeaderPtr = NULL;
- }
-
- WebRtc_Word32 callbackReturn = _sendCallback->SendData(
- frameType,
- _payloadType,
- encodedImage._timeStamp,
- encodedImage._buffer,
- encodedBytes,
- *fragmentationHeader,
- rtpVideoHeaderPtr);
- if (callbackReturn < 0)
- {
- return callbackReturn;
- }
- }
- else
- {
- return VCM_UNINITIALIZED;
- }
- _encodedBytes = encodedBytes;
- if (_mediaOpt != NULL) {
- _mediaOpt->UpdateWithEncodedData(_encodedBytes, frameType);
- if (_internalSource)
- {
- return _mediaOpt->DropFrame(); // Signal to encoder to drop next frame
- }
- }
- return VCM_OK;
-}
-
-WebRtc_UWord32
-VCMEncodedFrameCallback::EncodedBytes()
-{
- return _encodedBytes;
-}
-
-void
-VCMEncodedFrameCallback::SetMediaOpt(VCMMediaOptimization *mediaOpt)
-{
- _mediaOpt = mediaOpt;
-}
-
-void VCMEncodedFrameCallback::CopyCodecSpecific(const CodecSpecificInfo& info,
- RTPVideoHeader** rtp) {
- switch (info.codecType) {
- case kVideoCodecVP8: {
- (*rtp)->codecHeader.VP8.InitRTPVideoHeaderVP8();
- (*rtp)->codecHeader.VP8.pictureId =
- info.codecSpecific.VP8.pictureId;
- (*rtp)->codecHeader.VP8.nonReference =
- info.codecSpecific.VP8.nonReference;
- (*rtp)->codecHeader.VP8.temporalIdx =
- info.codecSpecific.VP8.temporalIdx;
- (*rtp)->codecHeader.VP8.layerSync =
- info.codecSpecific.VP8.layerSync;
- (*rtp)->codecHeader.VP8.tl0PicIdx =
- info.codecSpecific.VP8.tl0PicIdx;
- (*rtp)->codecHeader.VP8.keyIdx =
- info.codecSpecific.VP8.keyIdx;
- (*rtp)->simulcastIdx = info.codecSpecific.VP8.simulcastIdx;
- return;
- }
- default: {
- // No codec specific info. Change RTP header pointer to NULL.
- *rtp = NULL;
- return;
- }
- }
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/generic_encoder.h b/trunk/src/modules/video_coding/main/source/generic_encoder.h
deleted file mode 100644
index 99c2ce9..0000000
--- a/trunk/src/modules/video_coding/main/source/generic_encoder.h
+++ /dev/null
@@ -1,143 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
-
-#include "video_codec_interface.h"
-
-#include <stdio.h>
-
-namespace webrtc
-{
-
-class VCMMediaOptimization;
-
-/*************************************/
-/* VCMEncodeFrameCallback class */
-/***********************************/
-class VCMEncodedFrameCallback : public EncodedImageCallback
-{
-public:
- VCMEncodedFrameCallback();
- virtual ~VCMEncodedFrameCallback();
-
- /*
- * Callback implementation - codec encode complete
- */
- WebRtc_Word32 Encoded(
- EncodedImage& encodedImage,
- const CodecSpecificInfo* codecSpecificInfo = NULL,
- const RTPFragmentationHeader* fragmentationHeader = NULL);
- /*
- * Get number of encoded bytes
- */
- WebRtc_UWord32 EncodedBytes();
- /*
- * Callback implementation - generic encoder encode complete
- */
- WebRtc_Word32 SetTransportCallback(VCMPacketizationCallback* transport);
- /**
- * Set media Optimization
- */
- void SetMediaOpt (VCMMediaOptimization* mediaOpt);
-
- void SetPayloadType(WebRtc_UWord8 payloadType) { _payloadType = payloadType; };
- void SetCodecType(VideoCodecType codecType) {_codecType = codecType;};
- void SetInternalSource(bool internalSource) { _internalSource = internalSource; };
-
-private:
- /*
- * Map information from info into rtp. If no relevant information is found
- * in info, rtp is set to NULL.
- */
- static void CopyCodecSpecific(const CodecSpecificInfo& info,
- RTPVideoHeader** rtp);
-
- VCMPacketizationCallback* _sendCallback;
- VCMMediaOptimization* _mediaOpt;
- WebRtc_UWord32 _encodedBytes;
- WebRtc_UWord8 _payloadType;
- VideoCodecType _codecType;
- bool _internalSource;
- FILE* _bitStreamAfterEncoder;
-};// end of VCMEncodeFrameCallback class
-
-
-/******************************/
-/* VCMGenericEncoder class */
-/******************************/
-class VCMGenericEncoder
-{
- friend class VCMCodecDataBase;
-public:
- VCMGenericEncoder(VideoEncoder& encoder, bool internalSource = false);
- ~VCMGenericEncoder();
- /**
- * Free encoder memory
- */
- WebRtc_Word32 Release();
- /**
- * Initialize the encoder with the information from the VideoCodec
- */
- WebRtc_Word32 InitEncode(const VideoCodec* settings,
- WebRtc_Word32 numberOfCores,
- WebRtc_UWord32 maxPayloadSize);
- /**
- * Encode raw image
- * inputFrame : Frame containing raw image
- * codecSpecificInfo : Specific codec data
- * cameraFrameRate : request or information from the remote side
- * frameType : The requested frame type to encode
- */
- WebRtc_Word32 Encode(const VideoFrame& inputFrame,
- const CodecSpecificInfo* codecSpecificInfo,
- FrameType* frameType);
- /**
- * Set new target bit rate and frame rate
- * Return Value: new bit rate if OK, otherwise <0s
- */
- WebRtc_Word32 SetRates(WebRtc_UWord32 newBitRate, WebRtc_UWord32 frameRate);
- /**
- * Set a new packet loss rate and a new round-trip time in milliseconds.
- */
- WebRtc_Word32 SetChannelParameters(WebRtc_Word32 packetLoss, int rtt);
- WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* buffer, WebRtc_Word32 size);
- /**
- * Register a transport callback which will be called to deliver the encoded buffers
- */
- WebRtc_Word32 RegisterEncodeCallback(VCMEncodedFrameCallback* VCMencodedFrameCallback);
- /**
- * Get encoder bit rate
- */
- WebRtc_UWord32 BitRate() const;
- /**
- * Get encoder frame rate
- */
- WebRtc_UWord32 FrameRate() const;
-
- WebRtc_Word32 SetPeriodicKeyFrames(bool enable);
-
- WebRtc_Word32 RequestFrame(FrameType* frameTypes);
-
- bool InternalSource() const;
-
-private:
- VideoEncoder& _encoder;
- VideoCodecType _codecType;
- VCMEncodedFrameCallback* _VCMencodedFrameCallback;
- WebRtc_UWord32 _bitRate;
- WebRtc_UWord32 _frameRate;
- bool _internalSource;
-}; // end of VCMGenericEncoder class
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_GENERIC_ENCODER_H_
diff --git a/trunk/src/modules/video_coding/main/source/inter_frame_delay.cc b/trunk/src/modules/video_coding/main/source/inter_frame_delay.cc
deleted file mode 100644
index 3b520b3..0000000
--- a/trunk/src/modules/video_coding/main/source/inter_frame_delay.cc
+++ /dev/null
@@ -1,114 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "inter_frame_delay.h"
-
-namespace webrtc {
-
-VCMInterFrameDelay::VCMInterFrameDelay(int64_t currentWallClock)
-{
- Reset(currentWallClock);
-}
-
-// Resets the delay estimate
-void
-VCMInterFrameDelay::Reset(int64_t currentWallClock)
-{
- _zeroWallClock = currentWallClock;
- _wrapArounds = 0;
- _prevWallClock = 0;
- _prevTimestamp = 0;
- _dTS = 0;
-}
-
-// Calculates the delay of a frame with the given timestamp.
-// This method is called when the frame is complete.
-bool
-VCMInterFrameDelay::CalculateDelay(WebRtc_UWord32 timestamp,
- WebRtc_Word64 *delay,
- int64_t currentWallClock)
-{
- if (_prevWallClock == 0)
- {
- // First set of data, initialization, wait for next frame
- _prevWallClock = currentWallClock;
- _prevTimestamp = timestamp;
- *delay = 0;
- return true;
- }
-
- WebRtc_Word32 prevWrapArounds = _wrapArounds;
- CheckForWrapArounds(timestamp);
-
- // This will be -1 for backward wrap arounds and +1 for forward wrap arounds
- WebRtc_Word32 wrapAroundsSincePrev = _wrapArounds - prevWrapArounds;
-
- // Account for reordering in jitter variance estimate in the future?
- // Note that this also captures incomplete frames which are grabbed
- // for decoding after a later frame has been complete, i.e. real
- // packet losses.
- if ((wrapAroundsSincePrev == 0 && timestamp < _prevTimestamp) || wrapAroundsSincePrev < 0)
- {
- *delay = 0;
- return false;
- }
-
- // Compute the compensated timestamp difference and convert it to ms and
- // round it to closest integer.
- _dTS = static_cast<WebRtc_Word64>((timestamp + wrapAroundsSincePrev *
- (static_cast<WebRtc_Word64>(1)<<32) - _prevTimestamp) / 90.0 + 0.5);
-
- // frameDelay is the difference of dT and dTS -- i.e. the difference of
- // the wall clock time difference and the timestamp difference between
- // two following frames.
- *delay = static_cast<WebRtc_Word64>(currentWallClock - _prevWallClock - _dTS);
-
- _prevTimestamp = timestamp;
- _prevWallClock = currentWallClock;
-
- return true;
-}
-
-// Returns the current difference between incoming timestamps
-WebRtc_UWord32 VCMInterFrameDelay::CurrentTimeStampDiffMs() const
-{
- if (_dTS < 0)
- {
- return 0;
- }
- return static_cast<WebRtc_UWord32>(_dTS);
-}
-
-// Investigates if the timestamp clock has overflowed since the last timestamp and
-// keeps track of the number of wrap arounds since reset.
-void
-VCMInterFrameDelay::CheckForWrapArounds(WebRtc_UWord32 timestamp)
-{
- if (timestamp < _prevTimestamp)
- {
- // This difference will probably be less than -2^31 if we have had a wrap around
- // (e.g. timestamp = 1, _previousTimestamp = 2^32 - 1). Since it is cast to a Word32,
- // it should be positive.
- if (static_cast<WebRtc_Word32>(timestamp - _prevTimestamp) > 0)
- {
- // Forward wrap around
- _wrapArounds++;
- }
- }
- // This difference will probably be less than -2^31 if we have had a backward wrap around.
- // Since it is cast to a Word32, it should be positive.
- else if (static_cast<WebRtc_Word32>(_prevTimestamp - timestamp) > 0)
- {
- // Backward wrap around
- _wrapArounds--;
- }
-}
-
-}
diff --git a/trunk/src/modules/video_coding/main/source/inter_frame_delay.h b/trunk/src/modules/video_coding/main/source/inter_frame_delay.h
deleted file mode 100644
index 807c64b..0000000
--- a/trunk/src/modules/video_coding/main/source/inter_frame_delay.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
-#define WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
-
-#include "typedefs.h"
-
-namespace webrtc
-{
-
-class VCMInterFrameDelay
-{
-public:
- VCMInterFrameDelay(int64_t currentWallClock);
-
- // Resets the estimate. Zeros are given as parameters.
- void Reset(int64_t currentWallClock);
-
- // Calculates the delay of a frame with the given timestamp.
- // This method is called when the frame is complete.
- //
- // Input:
- // - timestamp : RTP timestamp of a received frame
- // - *delay : Pointer to memory where the result should be stored
- // - currentWallClock : The current time in milliseconds.
- // Should be -1 for normal operation, only used for testing.
- // Return value : true if OK, false when reordered timestamps
- bool CalculateDelay(WebRtc_UWord32 timestamp,
- WebRtc_Word64 *delay,
- int64_t currentWallClock);
-
- // Returns the current difference between incoming timestamps
- //
- // Return value : Wrap-around compensated difference between incoming
- // timestamps.
- WebRtc_UWord32 CurrentTimeStampDiffMs() const;
-
-private:
- // Controls if the RTP timestamp counter has had a wrap around
- // between the current and the previously received frame.
- //
- // Input:
- // - timestmap : RTP timestamp of the current frame.
- void CheckForWrapArounds(WebRtc_UWord32 timestamp);
-
- WebRtc_Word64 _zeroWallClock; // Local timestamp of the first video packet received
- WebRtc_Word32 _wrapArounds; // Number of wrapArounds detected
- // The previous timestamp passed to the delay estimate
- WebRtc_UWord32 _prevTimestamp;
- // The previous wall clock timestamp used by the delay estimate
- WebRtc_Word64 _prevWallClock;
- // Wrap-around compensated difference between incoming timestamps
- WebRtc_Word64 _dTS;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_INTER_FRAME_DELAY_H_
diff --git a/trunk/src/modules/video_coding/main/source/internal_defines.h b/trunk/src/modules/video_coding/main/source/internal_defines.h
deleted file mode 100644
index 0ba6385..0000000
--- a/trunk/src/modules/video_coding/main/source/internal_defines.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_
-#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_
-
-#include "typedefs.h"
-
-namespace webrtc
-{
-
-#define MASK_32_BITS(x) (0xFFFFFFFF & (x))
-
-inline WebRtc_UWord32 MaskWord64ToUWord32(WebRtc_Word64 w64)
-{
- return static_cast<WebRtc_UWord32>(MASK_32_BITS(w64));
-}
-
-#define VCM_MAX(a, b) (((a) > (b)) ? (a) : (b))
-#define VCM_MIN(a, b) (((a) < (b)) ? (a) : (b))
-
-#define VCM_DEFAULT_CODEC_WIDTH 352
-#define VCM_DEFAULT_CODEC_HEIGHT 288
-#define VCM_DEFAULT_FRAME_RATE 30
-#define VCM_MIN_BITRATE 30
-#define VCM_FLUSH_INDICATOR 4
-
-// Helper macros for creating the static codec list
-#define VCM_NO_CODEC_IDX -1
-#ifdef VIDEOCODEC_VP8
- #define VCM_VP8_IDX VCM_NO_CODEC_IDX + 1
-#else
- #define VCM_VP8_IDX VCM_NO_CODEC_IDX
-#endif
-#ifdef VIDEOCODEC_I420
- #define VCM_I420_IDX VCM_VP8_IDX + 1
-#else
- #define VCM_I420_IDX VCM_VP8_IDX
-#endif
-#define VCM_NUM_VIDEO_CODECS_AVAILABLE VCM_I420_IDX + 1
-
-#define VCM_NO_RECEIVER_ID 0
-
-inline WebRtc_Word32 VCMId(const WebRtc_Word32 vcmId, const WebRtc_Word32 receiverId = 0)
-{
- return static_cast<WebRtc_Word32>((vcmId << 16) + receiverId);
-}
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_INTERNAL_DEFINES_H_
diff --git a/trunk/src/modules/video_coding/main/source/jitter_buffer.cc b/trunk/src/modules/video_coding/main/source/jitter_buffer.cc
deleted file mode 100644
index 4b146b6..0000000
--- a/trunk/src/modules/video_coding/main/source/jitter_buffer.cc
+++ /dev/null
@@ -1,1775 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-#include "modules/video_coding/main/source/jitter_buffer.h"
-
-#include <algorithm>
-#include <cassert>
-
-#include "modules/video_coding/main/source/event.h"
-#include "modules/video_coding/main/source/frame_buffer.h"
-#include "modules/video_coding/main/source/inter_frame_delay.h"
-#include "modules/video_coding/main/source/internal_defines.h"
-#include "modules/video_coding/main/source/jitter_buffer_common.h"
-#include "modules/video_coding/main/source/jitter_estimator.h"
-#include "modules/video_coding/main/source/packet.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
-#include "system_wrappers/interface/critical_section_wrapper.h"
-#include "system_wrappers/interface/trace.h"
-
-#if defined(_WIN32)
- // VS 2005: Don't warn for default initialized arrays. See help for more info.
- #pragma warning(disable:4351)
-#endif
-
-namespace webrtc {
-
-// Predicates used when searching for frames in the frame buffer list
-class FrameSmallerTimestamp {
- public:
- FrameSmallerTimestamp(uint32_t timestamp) : timestamp_(timestamp) {}
- bool operator()(VCMFrameBuffer* frame) {
- return (LatestTimestamp(timestamp_, frame->TimeStamp(), NULL) ==
- timestamp_);
- }
-
- private:
- uint32_t timestamp_;
-};
-
-class FrameEqualTimestamp {
- public:
- FrameEqualTimestamp(uint32_t timestamp) : timestamp_(timestamp) {}
- bool operator()(VCMFrameBuffer* frame) {
- return (timestamp_ == frame->TimeStamp());
- }
-
- private:
- uint32_t timestamp_;
-};
-
-class CompleteDecodableKeyFrameCriteria {
- public:
- bool operator()(VCMFrameBuffer* frame) {
- return (frame->FrameType() == kVideoFrameKey) &&
- (frame->GetState() == kStateComplete ||
- frame->GetState() == kStateDecodable);
- }
-};
-
-// Constructor
-VCMJitterBuffer::VCMJitterBuffer(TickTimeBase* clock,
- WebRtc_Word32 vcmId,
- WebRtc_Word32 receiverId,
- bool master) :
- _vcmId(vcmId),
- _receiverId(receiverId),
- _clock(clock),
- _running(false),
- _critSect(CriticalSectionWrapper::CreateCriticalSection()),
- _master(master),
- _frameEvent(),
- _packetEvent(),
- _maxNumberOfFrames(kStartNumberOfFrames),
- _frameBuffers(),
- _frameList(),
- _lastDecodedState(),
- _packetsNotDecodable(0),
- _receiveStatistics(),
- _incomingFrameRate(0),
- _incomingFrameCount(0),
- _timeLastIncomingFrameCount(0),
- _incomingBitCount(0),
- _incomingBitRate(0),
- _dropCount(0),
- _numConsecutiveOldFrames(0),
- _numConsecutiveOldPackets(0),
- _discardedPackets(0),
- _jitterEstimate(vcmId, receiverId),
- _delayEstimate(_clock->MillisecondTimestamp()),
- _rttMs(0),
- _nackMode(kNoNack),
- _lowRttNackThresholdMs(-1),
- _highRttNackThresholdMs(-1),
- _NACKSeqNum(),
- _NACKSeqNumLength(0),
- _waitingForKeyFrame(false),
- _firstPacket(true)
-{
- memset(_frameBuffers, 0, sizeof(_frameBuffers));
- memset(_receiveStatistics, 0, sizeof(_receiveStatistics));
- memset(_NACKSeqNumInternal, -1, sizeof(_NACKSeqNumInternal));
-
- for (int i = 0; i< kStartNumberOfFrames; i++)
- {
- _frameBuffers[i] = new VCMFrameBuffer();
- }
-}
-
-// Destructor
-VCMJitterBuffer::~VCMJitterBuffer()
-{
- Stop();
- for (int i = 0; i< kMaxNumberOfFrames; i++)
- {
- if (_frameBuffers[i])
- {
- delete _frameBuffers[i];
- }
- }
- delete _critSect;
-}
-
-void
-VCMJitterBuffer::CopyFrom(const VCMJitterBuffer& rhs)
-{
- if (this != &rhs)
- {
- _critSect->Enter();
- rhs._critSect->Enter();
- _vcmId = rhs._vcmId;
- _receiverId = rhs._receiverId;
- _running = rhs._running;
- _master = !rhs._master;
- _maxNumberOfFrames = rhs._maxNumberOfFrames;
- _incomingFrameRate = rhs._incomingFrameRate;
- _incomingFrameCount = rhs._incomingFrameCount;
- _timeLastIncomingFrameCount = rhs._timeLastIncomingFrameCount;
- _incomingBitCount = rhs._incomingBitCount;
- _incomingBitRate = rhs._incomingBitRate;
- _dropCount = rhs._dropCount;
- _numConsecutiveOldFrames = rhs._numConsecutiveOldFrames;
- _numConsecutiveOldPackets = rhs._numConsecutiveOldPackets;
- _discardedPackets = rhs._discardedPackets;
- _jitterEstimate = rhs._jitterEstimate;
- _delayEstimate = rhs._delayEstimate;
- _waitingForCompletion = rhs._waitingForCompletion;
- _rttMs = rhs._rttMs;
- _NACKSeqNumLength = rhs._NACKSeqNumLength;
- _waitingForKeyFrame = rhs._waitingForKeyFrame;
- _firstPacket = rhs._firstPacket;
- _lastDecodedState = rhs._lastDecodedState;
- _packetsNotDecodable = rhs._packetsNotDecodable;
- memcpy(_receiveStatistics, rhs._receiveStatistics,
- sizeof(_receiveStatistics));
- memcpy(_NACKSeqNumInternal, rhs._NACKSeqNumInternal,
- sizeof(_NACKSeqNumInternal));
- memcpy(_NACKSeqNum, rhs._NACKSeqNum, sizeof(_NACKSeqNum));
- for (int i = 0; i < kMaxNumberOfFrames; i++)
- {
- if (_frameBuffers[i] != NULL)
- {
- delete _frameBuffers[i];
- _frameBuffers[i] = NULL;
- }
- }
- _frameList.clear();
- for (int i = 0; i < _maxNumberOfFrames; i++)
- {
- _frameBuffers[i] = new VCMFrameBuffer(*(rhs._frameBuffers[i]));
- if (_frameBuffers[i]->Length() > 0)
- {
- FrameList::reverse_iterator rit = std::find_if(
- _frameList.rbegin(), _frameList.rend(),
- FrameSmallerTimestamp(_frameBuffers[i]->TimeStamp()));
- _frameList.insert(rit.base(), _frameBuffers[i]);
- }
- }
- rhs._critSect->Leave();
- _critSect->Leave();
- }
-}
-
-// Start jitter buffer
-void
-VCMJitterBuffer::Start()
-{
- CriticalSectionScoped cs(_critSect);
- _running = true;
- _incomingFrameCount = 0;
- _incomingFrameRate = 0;
- _incomingBitCount = 0;
- _incomingBitRate = 0;
- _timeLastIncomingFrameCount = _clock->MillisecondTimestamp();
- memset(_receiveStatistics, 0, sizeof(_receiveStatistics));
-
- _numConsecutiveOldFrames = 0;
- _numConsecutiveOldPackets = 0;
- _discardedPackets = 0;
-
- _frameEvent.Reset(); // start in a non-signaled state
- _packetEvent.Reset(); // start in a non-signaled state
- _waitingForCompletion.frameSize = 0;
- _waitingForCompletion.timestamp = 0;
- _waitingForCompletion.latestPacketTime = -1;
- _firstPacket = true;
- _NACKSeqNumLength = 0;
- _waitingForKeyFrame = false;
- _rttMs = 0;
- _packetsNotDecodable = 0;
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId,
- _receiverId), "JB(0x%x): Jitter buffer: start", this);
-}
-
-
-// Stop jitter buffer
-void
-VCMJitterBuffer::Stop()
-{
- _critSect->Enter();
- _running = false;
- _lastDecodedState.Reset();
- _frameList.clear();
- for (int i = 0; i < kMaxNumberOfFrames; i++)
- {
- if (_frameBuffers[i] != NULL)
- {
- static_cast<VCMFrameBuffer*>(_frameBuffers[i])->SetState(kStateFree);
- }
- }
-
- _critSect->Leave();
- _frameEvent.Set(); // Make sure we exit from trying to get a frame to decoder
- _packetEvent.Set(); // Make sure we exit from trying to get a sequence number
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId,
- _receiverId), "JB(0x%x): Jitter buffer: stop", this);
-}
-
-bool
-VCMJitterBuffer::Running() const
-{
- CriticalSectionScoped cs(_critSect);
- return _running;
-}
-
-// Flush jitter buffer
-void
-VCMJitterBuffer::Flush()
-{
- CriticalSectionScoped cs(_critSect);
- FlushInternal();
-}
-
-// Must be called under the critical section _critSect
-void
-VCMJitterBuffer::FlushInternal()
-{
- // Erase all frames from the sorted list and set their state to free.
- _frameList.clear();
- for (WebRtc_Word32 i = 0; i < _maxNumberOfFrames; i++)
- {
- ReleaseFrameInternal(_frameBuffers[i]);
- }
- _lastDecodedState.Reset(); // TODO (mikhal): sync reset
- _packetsNotDecodable = 0;
-
- _frameEvent.Reset();
- _packetEvent.Reset();
-
- _numConsecutiveOldFrames = 0;
- _numConsecutiveOldPackets = 0;
-
- // Also reset the jitter and delay estimates
- _jitterEstimate.Reset();
- _delayEstimate.Reset(_clock->MillisecondTimestamp());
-
- _waitingForCompletion.frameSize = 0;
- _waitingForCompletion.timestamp = 0;
- _waitingForCompletion.latestPacketTime = -1;
-
- _firstPacket = true;
-
- _NACKSeqNumLength = 0;
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId,
- _receiverId), "JB(0x%x): Jitter buffer: flush", this);
-}
-
-// Set the frame state to free and remove it from the sorted
-// frame list. Must be called from inside the critical section _critSect.
-void
-VCMJitterBuffer::ReleaseFrameInternal(VCMFrameBuffer* frame)
-{
- if (frame != NULL && frame->GetState() != kStateDecoding)
- {
- frame->SetState(kStateFree);
- }
-}
-
-// Update frame state (set as complete if conditions are met)
-// Doing it here increases the degree of freedom for e.g. future
-// reconstructability of separate layers. Must be called under the
-// critical section _critSect.
-VCMFrameBufferEnum
-VCMJitterBuffer::UpdateFrameState(VCMFrameBuffer* frame)
-{
- if (frame == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
- VCMId(_vcmId, _receiverId), "JB(0x%x) FB(0x%x): "
- "UpdateFrameState NULL frame pointer", this, frame);
- return kNoError;
- }
-
- int length = frame->Length();
- if (_master)
- {
- // Only trace the primary jitter buffer to make it possible to parse
- // and plot the trace file.
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
- VCMId(_vcmId, _receiverId),
- "JB(0x%x) FB(0x%x): Complete frame added to jitter buffer,"
- " size:%d type %d",
- this, frame,length,frame->FrameType());
- }
-
- if (length != 0 && !frame->GetCountedFrame())
- {
- // ignore Ack frames
- _incomingFrameCount++;
- frame->SetCountedFrame(true);
- }
-
- // Check if we should drop frame
- // an old complete frame can arrive too late
- if (_lastDecodedState.IsOldFrame(frame))
- {
- // Frame is older than the latest decoded frame, drop it. Will be
- // released by CleanUpOldFrames later.
- frame->Reset();
- frame->SetState(kStateEmpty);
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
- VCMId(_vcmId, _receiverId),
- "JB(0x%x) FB(0x%x): Dropping old frame in Jitter buffer",
- this, frame);
- _dropCount++;
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
- VCMId(_vcmId, _receiverId),
- "Jitter buffer drop count: %d, consecutive drops: %u",
- _dropCount, _numConsecutiveOldFrames);
- // Flush() if this happens consistently.
- _numConsecutiveOldFrames++;
- if (_numConsecutiveOldFrames > kMaxConsecutiveOldFrames) {
- FlushInternal();
- return kFlushIndicator;
- }
- return kNoError;
- }
- _numConsecutiveOldFrames = 0;
- frame->SetState(kStateComplete);
-
-
- // Update receive statistics. We count all layers, thus when you use layers
- // adding all key and delta frames might differ from frame count
- if (frame->IsSessionComplete())
- {
- switch (frame->FrameType())
- {
- case kVideoFrameKey:
- {
- _receiveStatistics[0]++;
- break;
- }
- case kVideoFrameDelta:
- {
- _receiveStatistics[1]++;
- break;
- }
- case kVideoFrameGolden:
- {
- _receiveStatistics[2]++;
- break;
- }
- case kVideoFrameAltRef:
- {
- _receiveStatistics[3]++;
- break;
- }
- default:
- assert(false);
-
- }
- }
- const FrameList::iterator it = FindOldestCompleteContinuousFrame(false);
- VCMFrameBuffer* oldFrame = NULL;
- if (it != _frameList.end())
- {
- oldFrame = *it;
- }
-
- // Only signal if this is the oldest frame.
- // Not necessary the case due to packet reordering or NACK.
- if (!WaitForNack() || (oldFrame != NULL && oldFrame == frame))
- {
- _frameEvent.Set();
- }
- return kNoError;
-}
-
-// Get received key and delta frames
-WebRtc_Word32
-VCMJitterBuffer::GetFrameStatistics(WebRtc_UWord32& receivedDeltaFrames,
- WebRtc_UWord32& receivedKeyFrames) const
-{
- {
- CriticalSectionScoped cs(_critSect);
- receivedDeltaFrames = _receiveStatistics[1] + _receiveStatistics[3];
- receivedKeyFrames = _receiveStatistics[0] + _receiveStatistics[2];
- }
- return 0;
-}
-
-WebRtc_UWord32 VCMJitterBuffer::NumNotDecodablePackets() const {
- CriticalSectionScoped cs(_critSect);
- return _packetsNotDecodable;
-}
-
-WebRtc_UWord32 VCMJitterBuffer::DiscardedPackets() const {
- CriticalSectionScoped cs(_critSect);
- return _discardedPackets;
-}
-
-// Gets frame to use for this timestamp. If no match, get empty frame.
-WebRtc_Word32
-VCMJitterBuffer::GetFrame(const VCMPacket& packet, VCMEncodedFrame*& frame)
-{
- if (!_running) // don't accept incoming packets until we are started
- {
- return VCM_UNINITIALIZED;
- }
-
- _critSect->Enter();
- // Does this packet belong to an old frame?
- if (_lastDecodedState.IsOldPacket(&packet))
- {
- // Account only for media packets
- if (packet.sizeBytes > 0)
- {
- _discardedPackets++;
- _numConsecutiveOldPackets++;
- }
- // Update last decoded sequence number if the packet arrived late and
- // belongs to a frame with a timestamp equal to the last decoded
- // timestamp.
- _lastDecodedState.UpdateOldPacket(&packet);
-
- if (_numConsecutiveOldPackets > kMaxConsecutiveOldPackets)
- {
- FlushInternal();
- _critSect->Leave();
- return VCM_FLUSH_INDICATOR;
- }
- _critSect->Leave();
- return VCM_OLD_PACKET_ERROR;
- }
- _numConsecutiveOldPackets = 0;
-
- FrameList::iterator it = std::find_if(
- _frameList.begin(),
- _frameList.end(),
- FrameEqualTimestamp(packet.timestamp));
-
- if (it != _frameList.end()) {
- frame = *it;
- _critSect->Leave();
- return VCM_OK;
- }
-
- _critSect->Leave();
-
- // No match, return empty frame
- frame = GetEmptyFrame();
- if (frame != NULL)
- {
- return VCM_OK;
- }
- // No free frame! Try to reclaim some...
- _critSect->Enter();
- RecycleFramesUntilKeyFrame();
- _critSect->Leave();
-
- frame = GetEmptyFrame();
- if (frame != NULL)
- {
- return VCM_OK;
- }
- return VCM_JITTER_BUFFER_ERROR;
-}
-
-// Deprecated! Kept for testing purposes.
-VCMEncodedFrame*
-VCMJitterBuffer::GetFrame(const VCMPacket& packet)
-{
- VCMEncodedFrame* frame = NULL;
- if (GetFrame(packet, frame) < 0)
- {
- return NULL;
- }
- return frame;
-}
-
-// Get empty frame, creates new (i.e. increases JB size) if necessary
-VCMFrameBuffer*
-VCMJitterBuffer::GetEmptyFrame()
-{
- if (!_running) // don't accept incoming packets until we are started
- {
- return NULL;
- }
-
- _critSect->Enter();
-
- for (int i = 0; i <_maxNumberOfFrames; ++i)
- {
- if (kStateFree == _frameBuffers[i]->GetState())
- {
- // found a free buffer
- _frameBuffers[i]->SetState(kStateEmpty);
- _critSect->Leave();
- return _frameBuffers[i];
- }
- }
-
- // Check if we can increase JB size
- if (_maxNumberOfFrames < kMaxNumberOfFrames)
- {
- VCMFrameBuffer* ptrNewBuffer = new VCMFrameBuffer();
- ptrNewBuffer->SetState(kStateEmpty);
- _frameBuffers[_maxNumberOfFrames] = ptrNewBuffer;
- _maxNumberOfFrames++;
-
- _critSect->Leave();
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
- VCMId(_vcmId, _receiverId), "JB(0x%x) FB(0x%x): Jitter buffer "
- "increased to:%d frames", this, ptrNewBuffer, _maxNumberOfFrames);
- return ptrNewBuffer;
- }
- _critSect->Leave();
-
- // We have reached max size, cannot increase JB size
- return NULL;
-}
-
-
-// Find oldest complete frame used for getting next frame to decode
-// Must be called under critical section
-FrameList::iterator
-VCMJitterBuffer::FindOldestCompleteContinuousFrame(bool enable_decodable) {
- // If we have more than one frame done since last time, pick oldest.
- VCMFrameBuffer* oldest_frame = NULL;
- FrameList::iterator it = _frameList.begin();
-
- // When temporal layers are available, we search for a complete or decodable
- // frame until we hit one of the following:
- // 1. Continuous base or sync layer.
- // 2. The end of the list was reached.
- for (; it != _frameList.end(); ++it) {
- oldest_frame = *it;
- VCMFrameBufferStateEnum state = oldest_frame->GetState();
- // Is this frame complete or decodable and continuous?
- if ((state == kStateComplete ||
- (enable_decodable && state == kStateDecodable)) &&
- _lastDecodedState.ContinuousFrame(oldest_frame)) {
- break;
- } else {
- int temporal_id = oldest_frame->TemporalId();
- oldest_frame = NULL;
- if (temporal_id <= 0) {
- // When temporal layers are disabled or we have hit a base layer
- // we break (regardless of continuity and completeness).
- break;
- }
- }
- }
-
- if (oldest_frame == NULL) {
- // No complete frame no point to continue.
- return _frameList.end();
- } else if (_waitingForKeyFrame &&
- oldest_frame->FrameType() != kVideoFrameKey) {
- // We are waiting for a key frame.
- return _frameList.end();
- }
-
- // We have a complete continuous frame.
- return it;
-}
-
-// Call from inside the critical section _critSect
-void
-VCMJitterBuffer::RecycleFrame(VCMFrameBuffer* frame)
-{
- if (frame == NULL)
- {
- return;
- }
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
- VCMId(_vcmId, _receiverId),
- "JB(0x%x) FB(0x%x): RecycleFrame, size:%d",
- this, frame, frame->Length());
-
- ReleaseFrameInternal(frame);
-}
-
-// Calculate frame and bit rates
-WebRtc_Word32
-VCMJitterBuffer::GetUpdate(WebRtc_UWord32& frameRate, WebRtc_UWord32& bitRate)
-{
- CriticalSectionScoped cs(_critSect);
- const WebRtc_Word64 now = _clock->MillisecondTimestamp();
- WebRtc_Word64 diff = now - _timeLastIncomingFrameCount;
- if (diff < 1000 && _incomingFrameRate > 0 && _incomingBitRate > 0)
- {
- // Make sure we report something even though less than
- // 1 second has passed since last update.
- frameRate = _incomingFrameRate;
- bitRate = _incomingBitRate;
- }
- else if (_incomingFrameCount != 0)
- {
- // We have received frame(s) since last call to this function
-
- // Prepare calculations
- if (diff <= 0)
- {
- diff = 1;
- }
- // we add 0.5f for rounding
- float rate = 0.5f + ((_incomingFrameCount * 1000.0f) / diff);
- if (rate < 1.0f) // don't go below 1, can crash
- {
- rate = 1.0f;
- }
-
- // Calculate frame rate
- // Let r be rate.
- // r(0) = 1000*framecount/delta_time.
- // (I.e. frames per second since last calculation.)
- // frameRate = r(0)/2 + r(-1)/2
- // (I.e. fr/s average this and the previous calculation.)
- frameRate = (_incomingFrameRate + (WebRtc_Word32)rate) >> 1;
- _incomingFrameRate = (WebRtc_UWord8)rate;
-
- // Calculate bit rate
- if (_incomingBitCount == 0)
- {
- bitRate = 0;
- }
- else
- {
- bitRate = 10 * ((100 * _incomingBitCount) /
- static_cast<WebRtc_UWord32>(diff));
- }
- _incomingBitRate = bitRate;
-
- // Reset count
- _incomingFrameCount = 0;
- _incomingBitCount = 0;
- _timeLastIncomingFrameCount = now;
-
- }
- else
- {
- // No frames since last call
- _timeLastIncomingFrameCount = _clock->MillisecondTimestamp();
- frameRate = 0;
- bitRate = 0;
- _incomingBitRate = 0;
- }
-
- return 0;
-}
-
-// Returns immediately or a X ms event hang waiting for a complete frame,
-// X decided by caller
-VCMEncodedFrame*
-VCMJitterBuffer::GetCompleteFrameForDecoding(WebRtc_UWord32 maxWaitTimeMS)
-{
- if (!_running)
- {
- return NULL;
- }
-
- _critSect->Enter();
-
- CleanUpOldFrames();
-
- if (_lastDecodedState.init() && WaitForNack()) {
- _waitingForKeyFrame = true;
- }
-
- FrameList::iterator it = FindOldestCompleteContinuousFrame(false);
- if (it == _frameList.end())
- {
- if (maxWaitTimeMS == 0)
- {
- _critSect->Leave();
- return NULL;
- }
- const WebRtc_Word64 endWaitTimeMs = _clock->MillisecondTimestamp()
- + maxWaitTimeMS;
- WebRtc_Word64 waitTimeMs = maxWaitTimeMS;
- while (waitTimeMs > 0)
- {
- _critSect->Leave();
- const EventTypeWrapper ret =
- _frameEvent.Wait(static_cast<WebRtc_UWord32>(waitTimeMs));
- _critSect->Enter();
- if (ret == kEventSignaled)
- {
- // are we closing down the Jitter buffer
- if (!_running)
- {
- _critSect->Leave();
- return NULL;
- }
-
- // Finding oldest frame ready for decoder, but check
- // sequence number and size
- CleanUpOldFrames();
- it = FindOldestCompleteContinuousFrame(false);
- if (it == _frameList.end())
- {
- waitTimeMs = endWaitTimeMs -
- _clock->MillisecondTimestamp();
- }
- else
- {
- break;
- }
- }
- else
- {
- _critSect->Leave();
- return NULL;
- }
- }
- // Inside critSect
- }
- else
- {
- // we already have a frame reset the event
- _frameEvent.Reset();
- }
-
- if (it == _frameList.end())
- {
- // Even after signaling we're still missing a complete continuous frame
- _critSect->Leave();
- return NULL;
- }
-
- VCMFrameBuffer* oldestFrame = *it;
- it = _frameList.erase(it);
-
- // Update jitter estimate
- const bool retransmitted = (oldestFrame->GetNackCount() > 0);
- if (retransmitted)
- {
- _jitterEstimate.FrameNacked();
- }
- else if (oldestFrame->Length() > 0)
- {
- // Ignore retransmitted and empty frames.
- UpdateJitterAndDelayEstimates(*oldestFrame, false);
- }
-
- oldestFrame->SetState(kStateDecoding);
-
- CleanUpOldFrames();
-
- if (oldestFrame->FrameType() == kVideoFrameKey)
- {
- _waitingForKeyFrame = false;
- }
-
- _critSect->Leave();
-
- // We have a frame - update decoded state with frame info.
- _lastDecodedState.SetState(oldestFrame);
-
- return oldestFrame;
-}
-
-WebRtc_UWord32
-VCMJitterBuffer::GetEstimatedJitterMS()
-{
- CriticalSectionScoped cs(_critSect);
- return GetEstimatedJitterMsInternal();
-}
-
-WebRtc_UWord32
-VCMJitterBuffer::GetEstimatedJitterMsInternal()
-{
- WebRtc_UWord32 estimate = VCMJitterEstimator::OPERATING_SYSTEM_JITTER;
-
- // Compute RTT multiplier for estimation
- // _lowRttNackThresholdMs == -1 means no FEC.
- double rttMult = 1.0f;
- if (_nackMode == kNackHybrid && (_lowRttNackThresholdMs >= 0 &&
- static_cast<int>(_rttMs) > _lowRttNackThresholdMs))
- {
- // from here we count on FEC
- rttMult = 0.0f;
- }
- estimate += static_cast<WebRtc_UWord32>
- (_jitterEstimate.GetJitterEstimate(rttMult) + 0.5);
- return estimate;
-}
-
-void
-VCMJitterBuffer::UpdateRtt(WebRtc_UWord32 rttMs)
-{
- CriticalSectionScoped cs(_critSect);
- _rttMs = rttMs;
- _jitterEstimate.UpdateRtt(rttMs);
-}
-
-// wait for the first packet in the next frame to arrive
-WebRtc_Word64
-VCMJitterBuffer::GetNextTimeStamp(WebRtc_UWord32 maxWaitTimeMS,
- FrameType& incomingFrameType,
- WebRtc_Word64& renderTimeMs)
-{
- if (!_running)
- {
- return -1;
- }
-
- _critSect->Enter();
-
- // Finding oldest frame ready for decoder, check sequence number and size
- CleanUpOldFrames();
-
- FrameList::iterator it = _frameList.begin();
-
- if (it == _frameList.end())
- {
- _packetEvent.Reset();
- _critSect->Leave();
-
- if (_packetEvent.Wait(maxWaitTimeMS) == kEventSignaled)
- {
- // are we closing down the Jitter buffer
- if (!_running)
- {
- return -1;
- }
- _critSect->Enter();
-
- CleanUpOldFrames();
- it = _frameList.begin();
- }
- else
- {
- _critSect->Enter();
- }
- }
-
- if (it == _frameList.end())
- {
- _critSect->Leave();
- return -1;
- }
- // we have a frame
-
- // return frame type
- // All layers are assumed to have the same type
- incomingFrameType = (*it)->FrameType();
-
- renderTimeMs = (*it)->RenderTimeMs();
-
- const WebRtc_UWord32 timestamp = (*it)->TimeStamp();
-
- _critSect->Leave();
-
- // return current time
- return timestamp;
-}
-
-// Answers the question:
-// Will the packet sequence be complete if the next frame is grabbed for
-// decoding right now? That is, have we lost a frame between the last decoded
-// frame and the next, or is the next
-// frame missing one or more packets?
-bool
-VCMJitterBuffer::CompleteSequenceWithNextFrame()
-{
- CriticalSectionScoped cs(_critSect);
- // Finding oldest frame ready for decoder, check sequence number and size
- CleanUpOldFrames();
-
- if (_frameList.empty())
- return true;
-
- VCMFrameBuffer* oldestFrame = _frameList.front();
- if (_frameList.size() <= 1 &&
- oldestFrame->GetState() != kStateComplete)
- {
- // Frame not ready to be decoded.
- return true;
- }
- if (!oldestFrame->Complete())
- {
- return false;
- }
-
- // See if we have lost a frame before this one.
- if (_lastDecodedState.init())
- {
- // Following start, reset or flush -> check for key frame.
- if (oldestFrame->FrameType() != kVideoFrameKey)
- {
- return false;
- }
- }
- else if (oldestFrame->GetLowSeqNum() == -1)
- {
- return false;
- }
- else if (!_lastDecodedState.ContinuousFrame(oldestFrame))
- {
- return false;
- }
- return true;
-}
-
-// Returns immediately
-VCMEncodedFrame*
-VCMJitterBuffer::GetFrameForDecoding()
-{
- CriticalSectionScoped cs(_critSect);
- if (!_running)
- {
- return NULL;
- }
-
- if (WaitForNack())
- {
- return GetFrameForDecodingNACK();
- }
-
- CleanUpOldFrames();
-
- if (_frameList.empty()) {
- return NULL;
- }
-
- VCMFrameBuffer* oldestFrame = _frameList.front();
- if (_frameList.size() <= 1 &&
- oldestFrame->GetState() != kStateComplete) {
- return NULL;
- }
-
- // Incomplete frame pulled out from jitter buffer,
- // update the jitter estimate with what we currently know.
- // This frame shouldn't have been retransmitted, but if we recently
- // turned off NACK this might still happen.
- const bool retransmitted = (oldestFrame->GetNackCount() > 0);
- if (retransmitted)
- {
- _jitterEstimate.FrameNacked();
- }
- else if (oldestFrame->Length() > 0)
- {
- // Ignore retransmitted and empty frames.
- // Update with the previous incomplete frame first
- if (_waitingForCompletion.latestPacketTime >= 0)
- {
- UpdateJitterAndDelayEstimates(_waitingForCompletion, true);
- }
- // Then wait for this one to get complete
- _waitingForCompletion.frameSize = oldestFrame->Length();
- _waitingForCompletion.latestPacketTime =
- oldestFrame->LatestPacketTimeMs();
- _waitingForCompletion.timestamp = oldestFrame->TimeStamp();
- }
- _frameList.erase(_frameList.begin());
-
- // Look for previous frame loss
- VerifyAndSetPreviousFrameLost(*oldestFrame);
-
- // The state must be changed to decoding before cleaning up zero sized
- // frames to avoid empty frames being cleaned up and then given to the
- // decoder.
- // Set as decoding. Propagates the missingFrame bit.
- oldestFrame->SetState(kStateDecoding);
-
- CleanUpOldFrames();
-
- if (oldestFrame->FrameType() == kVideoFrameKey)
- {
- _waitingForKeyFrame = false;
- }
-
- _packetsNotDecodable += oldestFrame->NotDecodablePackets();
-
- // We have a frame - update decoded state with frame info.
- _lastDecodedState.SetState(oldestFrame);
-
- return oldestFrame;
-}
-
-VCMEncodedFrame*
-VCMJitterBuffer::GetFrameForDecodingNACK()
-{
- // when we use NACK we don't release non complete frames
- // unless we have a complete key frame.
- // In hybrid mode, we may release decodable frames (non-complete)
-
- // Clean up old frames and empty frames
- CleanUpOldFrames();
-
- // First look for a complete _continuous_ frame.
- // When waiting for nack, wait for a key frame, if a continuous frame cannot
- // be determined (i.e. initial decoding state).
- if (_lastDecodedState.init()) {
- _waitingForKeyFrame = true;
- }
-
- // Allow for a decodable frame when in Hybrid mode.
- bool enableDecodable = _nackMode == kNackHybrid ? true : false;
- FrameList::iterator it = FindOldestCompleteContinuousFrame(enableDecodable);
- if (it == _frameList.end())
- {
- // If we didn't find one we're good with a complete key/decodable frame.
- it = find_if(_frameList.begin(), _frameList.end(),
- CompleteDecodableKeyFrameCriteria());
- if (it == _frameList.end())
- {
- return NULL;
- }
- }
- VCMFrameBuffer* oldestFrame = *it;
- // Update jitter estimate
- const bool retransmitted = (oldestFrame->GetNackCount() > 0);
- if (retransmitted)
- {
- _jitterEstimate.FrameNacked();
- }
- else if (oldestFrame->Length() > 0)
- {
- // Ignore retransmitted and empty frames.
- UpdateJitterAndDelayEstimates(*oldestFrame, false);
- }
- it = _frameList.erase(it);
-
- // Look for previous frame loss
- VerifyAndSetPreviousFrameLost(*oldestFrame);
-
- // The state must be changed to decoding before cleaning up zero sized
- // frames to avoid empty frames being cleaned up and then given to the
- // decoder.
- oldestFrame->SetState(kStateDecoding);
-
- // Clean up old frames and empty frames
- CleanUpOldFrames();
-
- if (oldestFrame->FrameType() == kVideoFrameKey)
- {
- _waitingForKeyFrame = false;
- }
-
- // We have a frame - update decoded state with frame info.
- _lastDecodedState.SetState(oldestFrame);
-
- return oldestFrame;
-}
-
-// Must be called under the critical section _critSect. Should never be called
-// with retransmitted frames, they must be filtered out before this function is
-// called.
-void
-VCMJitterBuffer::UpdateJitterAndDelayEstimates(VCMJitterSample& sample,
- bool incompleteFrame)
-{
- if (sample.latestPacketTime == -1)
- {
- return;
- }
- if (incompleteFrame)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
- VCMId(_vcmId, _receiverId), "Received incomplete frame "
- "timestamp %u frame size %u at time %u",
- sample.timestamp, sample.frameSize,
- MaskWord64ToUWord32(sample.latestPacketTime));
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
- VCMId(_vcmId, _receiverId), "Received complete frame "
- "timestamp %u frame size %u at time %u",
- sample.timestamp, sample.frameSize,
- MaskWord64ToUWord32(sample.latestPacketTime));
- }
- UpdateJitterAndDelayEstimates(sample.latestPacketTime,
- sample.timestamp,
- sample.frameSize,
- incompleteFrame);
-}
-
-// Must be called under the critical section _critSect. Should never be
-// called with retransmitted frames, they must be filtered out before this
-// function is called.
-void
-VCMJitterBuffer::UpdateJitterAndDelayEstimates(VCMFrameBuffer& frame,
- bool incompleteFrame)
-{
- if (frame.LatestPacketTimeMs() == -1)
- {
- return;
- }
- // No retransmitted frames should be a part of the jitter
- // estimate.
- if (incompleteFrame)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
- VCMId(_vcmId, _receiverId),
- "Received incomplete frame timestamp %u frame type %d "
- "frame size %u at time %u, jitter estimate was %u",
- frame.TimeStamp(), frame.FrameType(), frame.Length(),
- MaskWord64ToUWord32(frame.LatestPacketTimeMs()),
- GetEstimatedJitterMsInternal());
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
- VCMId(_vcmId, _receiverId),"Received complete frame "
- "timestamp %u frame type %d frame size %u at time %u, "
- "jitter estimate was %u",
- frame.TimeStamp(), frame.FrameType(), frame.Length(),
- MaskWord64ToUWord32(frame.LatestPacketTimeMs()),
- GetEstimatedJitterMsInternal());
- }
- UpdateJitterAndDelayEstimates(frame.LatestPacketTimeMs(), frame.TimeStamp(),
- frame.Length(), incompleteFrame);
-}
-
-// Must be called under the critical section _critSect. Should never be called
-// with retransmitted frames, they must be filtered out before this function
-// is called.
-void
-VCMJitterBuffer::UpdateJitterAndDelayEstimates(WebRtc_Word64 latestPacketTimeMs,
- WebRtc_UWord32 timestamp,
- WebRtc_UWord32 frameSize,
- bool incompleteFrame)
-{
- if (latestPacketTimeMs == -1)
- {
- return;
- }
- WebRtc_Word64 frameDelay;
- // Calculate the delay estimate
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
- VCMId(_vcmId, _receiverId),
- "Packet received and sent to jitter estimate with: "
- "timestamp=%u wallClock=%u", timestamp,
- MaskWord64ToUWord32(latestPacketTimeMs));
- bool notReordered = _delayEstimate.CalculateDelay(timestamp,
- &frameDelay,
- latestPacketTimeMs);
- // Filter out frames which have been reordered in time by the network
- if (notReordered)
- {
- // Update the jitter estimate with the new samples
- _jitterEstimate.UpdateEstimate(frameDelay, frameSize, incompleteFrame);
- }
-}
-
-WebRtc_UWord16*
-VCMJitterBuffer::GetNackList(WebRtc_UWord16& nackSize,bool& listExtended)
-{
- return CreateNackList(nackSize,listExtended);
-}
-
-// Assume called internally with critsect
-WebRtc_Word32
-VCMJitterBuffer::GetLowHighSequenceNumbers(WebRtc_Word32& lowSeqNum,
- WebRtc_Word32& highSeqNum) const
-{
- // TODO (mikhal/stefan): refactor to use lastDecodedState
- WebRtc_Word32 i = 0;
- WebRtc_Word32 seqNum = -1;
-
- highSeqNum = -1;
- lowSeqNum = -1;
- if (!_lastDecodedState.init())
- lowSeqNum = _lastDecodedState.sequence_num();
-
- // find highest seq numbers
- for (i = 0; i < _maxNumberOfFrames; ++i)
- {
- seqNum = _frameBuffers[i]->GetHighSeqNum();
-
- // Ignore free / empty frames
- VCMFrameBufferStateEnum state = _frameBuffers[i]->GetState();
-
- if ((kStateFree != state) &&
- (kStateEmpty != state) &&
- (kStateDecoding != state) &&
- seqNum != -1)
- {
- bool wrap;
- highSeqNum = LatestSequenceNumber(seqNum, highSeqNum, &wrap);
- }
- } // for
- return 0;
-}
-
-
-WebRtc_UWord16*
-VCMJitterBuffer::CreateNackList(WebRtc_UWord16& nackSize, bool& listExtended)
-{
- // TODO (mikhal/stefan): Refactor to use lastDecodedState.
- CriticalSectionScoped cs(_critSect);
- int i = 0;
- WebRtc_Word32 lowSeqNum = -1;
- WebRtc_Word32 highSeqNum = -1;
- listExtended = false;
-
- // Don't create list, if we won't wait for it
- if (!WaitForNack())
- {
- nackSize = 0;
- return NULL;
- }
-
- // Find the lowest (last decoded) sequence number and
- // the highest (highest sequence number of the newest frame)
- // sequence number. The nack list is a subset of the range
- // between those two numbers.
- GetLowHighSequenceNumbers(lowSeqNum, highSeqNum);
-
- // write a list of all seq num we have
- if (lowSeqNum == -1 || highSeqNum == -1)
- {
- // This happens if we lose the first packet, nothing is popped
- if (highSeqNum == -1)
- {
- // we have not received any packets yet
- nackSize = 0;
- }
- else
- {
- // signal that we want a key frame request to be sent
- nackSize = 0xffff;
- }
- return NULL;
- }
-
- int numberOfSeqNum = 0;
- if (lowSeqNum > highSeqNum)
- {
- if (lowSeqNum - highSeqNum > 0x00ff)
- {
- // wrap
- numberOfSeqNum = (0xffff-lowSeqNum) + highSeqNum + 1;
- }
- }
- else
- {
- numberOfSeqNum = highSeqNum - lowSeqNum;
- }
-
- if (numberOfSeqNum > kNackHistoryLength)
- {
- // Nack list is too big, flush and try to restart.
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
- VCMId(_vcmId, _receiverId),
- "Nack list too large, try to find a key frame and restart "
- "from seq: %d. Lowest seq in jb %d", highSeqNum,lowSeqNum);
-
- // This nack size will trigger a key request...
- bool foundKeyFrame = false;
-
- while (numberOfSeqNum > kNackHistoryLength)
- {
- foundKeyFrame = RecycleFramesUntilKeyFrame();
-
- if (!foundKeyFrame)
- {
- break;
- }
-
- // Check if we still have too many packets in JB
- lowSeqNum = -1;
- highSeqNum = -1;
- GetLowHighSequenceNumbers(lowSeqNum, highSeqNum);
-
- if (highSeqNum == -1)
- {
- assert(lowSeqNum != -1); // This should never happen
- // We can't calculate the nack list length...
- return NULL;
- }
-
- numberOfSeqNum = 0;
- if (lowSeqNum > highSeqNum)
- {
- if (lowSeqNum - highSeqNum > 0x00ff)
- {
- // wrap
- numberOfSeqNum = (0xffff-lowSeqNum) + highSeqNum + 1;
- highSeqNum=lowSeqNum;
- }
- }
- else
- {
- numberOfSeqNum = highSeqNum - lowSeqNum;
- }
-
- } // end while
-
- if (!foundKeyFrame)
- {
- // No key frame in JB.
-
- // Set the last decoded sequence number to current high.
- // This is to not get a large nack list again right away
- _lastDecodedState.SetSeqNum(static_cast<uint16_t>(highSeqNum));
- // Set to trigger key frame signal
- nackSize = 0xffff;
- listExtended = true;
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, -1,
- "\tNo key frame found, request one. _lastDecodedSeqNum[0] "
- "%d", _lastDecodedState.sequence_num());
- }
- else
- {
- // We have cleaned up the jb and found a key frame
- // The function itself has set last decoded seq.
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, -1,
- "\tKey frame found. _lastDecodedSeqNum[0] %d",
- _lastDecodedState.sequence_num());
- nackSize = 0;
- }
-
- return NULL;
- }
-
- WebRtc_UWord16 seqNumberIterator = (WebRtc_UWord16)(lowSeqNum + 1);
- for (i = 0; i < numberOfSeqNum; i++)
- {
- _NACKSeqNumInternal[i] = seqNumberIterator;
- seqNumberIterator++;
- }
-
- // now we have a list of all sequence numbers that could have been sent
-
- // zero out the ones we have received
- for (i = 0; i < _maxNumberOfFrames; i++)
- {
- // loop all created frames
- // We don't need to check if frame is decoding since lowSeqNum is based
- // on _lastDecodedSeqNum
- // Ignore free frames
- VCMFrameBufferStateEnum state = _frameBuffers[i]->GetState();
-
- if ((kStateFree != state) &&
- (kStateEmpty != state) &&
- (kStateDecoding != state))
- {
- // Reaching thus far means we are going to update the nack list
- // When in hybrid mode, we use the soft NACKing feature.
- if (_nackMode == kNackHybrid)
- {
- _frameBuffers[i]->BuildSoftNackList(_NACKSeqNumInternal,
- numberOfSeqNum,
- _rttMs);
- }
- else
- {
- // Used when the frame is being processed by the decoding thread
- // don't need to use that info in this loop.
- _frameBuffers[i]->BuildHardNackList(_NACKSeqNumInternal,
- numberOfSeqNum);
- }
- }
- }
-
- // compress list
- int emptyIndex = -1;
- for (i = 0; i < numberOfSeqNum; i++)
- {
- if (_NACKSeqNumInternal[i] == -1 || _NACKSeqNumInternal[i] == -2 )
- {
- // this is empty
- if (emptyIndex == -1)
- {
- // no empty index before, remember this position
- emptyIndex = i;
- }
- }
- else
- {
- // this is not empty
- if (emptyIndex == -1)
- {
- // no empty index, continue
- }
- else
- {
- _NACKSeqNumInternal[emptyIndex] = _NACKSeqNumInternal[i];
- _NACKSeqNumInternal[i] = -1;
- emptyIndex++;
- }
- }
- } // for
-
- if (emptyIndex == -1)
- {
- // no empty
- nackSize = numberOfSeqNum;
- }
- else
- {
- nackSize = emptyIndex;
- }
-
- if (nackSize > _NACKSeqNumLength)
- {
- // Larger list: nack list was extended since the last call.
- listExtended = true;
- }
-
- for (WebRtc_UWord32 j = 0; j < nackSize; j++)
- {
- // Check if the list has been extended since it was last created. I.e,
- // new items have been added
- if (_NACKSeqNumLength > j && !listExtended)
- {
- WebRtc_UWord32 k = 0;
- for (k = j; k < _NACKSeqNumLength; k++)
- {
- // Found the item in the last list, i.e, no new items found yet.
- if (_NACKSeqNum[k] == (WebRtc_UWord16)_NACKSeqNumInternal[j])
- {
- break;
- }
- }
- if (k == _NACKSeqNumLength) // New item not found in last list.
- {
- listExtended = true;
- }
- }
- else
- {
- listExtended = true;
- }
- _NACKSeqNum[j] = (WebRtc_UWord16)_NACKSeqNumInternal[j];
- }
-
- _NACKSeqNumLength = nackSize;
-
- return _NACKSeqNum;
-}
-
-// Release frame when done with decoding. Should never be used to release
-// frames from within the jitter buffer.
-void
-VCMJitterBuffer::ReleaseFrame(VCMEncodedFrame* frame)
-{
- CriticalSectionScoped cs(_critSect);
- VCMFrameBuffer* frameBuffer = static_cast<VCMFrameBuffer*>(frame);
- if (frameBuffer != NULL)
- frameBuffer->SetState(kStateFree);
-}
-
-WebRtc_Word64
-VCMJitterBuffer::LastPacketTime(VCMEncodedFrame* frame,
- bool& retransmitted) const
-{
- CriticalSectionScoped cs(_critSect);
- retransmitted = (static_cast<VCMFrameBuffer*>(frame)->GetNackCount() > 0);
- return static_cast<VCMFrameBuffer*>(frame)->LatestPacketTimeMs();
-}
-
-WebRtc_Word64
-VCMJitterBuffer::LastDecodedTimestamp() const
-{
- CriticalSectionScoped cs(_critSect);
- return _lastDecodedState.time_stamp();
-}
-
-// Insert packet
-// Takes crit sect, and inserts packet in frame buffer, possibly does logging
-VCMFrameBufferEnum
-VCMJitterBuffer::InsertPacket(VCMEncodedFrame* buffer, const VCMPacket& packet)
-{
- CriticalSectionScoped cs(_critSect);
- WebRtc_Word64 nowMs = _clock->MillisecondTimestamp();
- VCMFrameBufferEnum bufferReturn = kSizeError;
- VCMFrameBufferEnum ret = kSizeError;
- VCMFrameBuffer* frame = static_cast<VCMFrameBuffer*>(buffer);
-
- // We are keeping track of the first seq num, the latest seq num and
- // the number of wraps to be able to calculate how many packets we expect.
- if (_firstPacket)
- {
- // Now it's time to start estimating jitter
- // reset the delay estimate.
- _delayEstimate.Reset(_clock->MillisecondTimestamp());
- _firstPacket = false;
- }
-
- // Empty packets may bias the jitter estimate (lacking size component),
- // therefore don't let empty packet trigger the following updates:
- if (packet.frameType != kFrameEmpty)
- {
- if (_waitingForCompletion.timestamp == packet.timestamp)
- {
- // This can get bad if we have a lot of duplicate packets,
- // we will then count some packet multiple times.
- _waitingForCompletion.frameSize += packet.sizeBytes;
- _waitingForCompletion.latestPacketTime = nowMs;
- }
- else if (_waitingForCompletion.latestPacketTime >= 0 &&
- _waitingForCompletion.latestPacketTime + 2000 <= nowMs)
- {
- // A packet should never be more than two seconds late
- UpdateJitterAndDelayEstimates(_waitingForCompletion, true);
- _waitingForCompletion.latestPacketTime = -1;
- _waitingForCompletion.frameSize = 0;
- _waitingForCompletion.timestamp = 0;
- }
- }
-
- if (frame != NULL)
- {
- VCMFrameBufferStateEnum state = frame->GetState();
- _lastDecodedState.UpdateOldPacket(&packet);
- // Insert packet
- // Check for first packet
- // High sequence number will be -1 if neither an empty packet nor
- // a media packet has been inserted.
- bool first = (frame->GetHighSeqNum() == -1);
- // When in Hybrid mode, we allow for a decodable state
- // Note: Under current version, a decodable frame will never be
- // triggered, as the body of the function is empty.
- // TODO (mikhal): Update when decodable is enabled.
- bufferReturn = frame->InsertPacket(packet, nowMs,
- _nackMode == kNackHybrid,
- _rttMs);
- ret = bufferReturn;
-
- if (bufferReturn > 0)
- {
- _incomingBitCount += packet.sizeBytes << 3;
-
- // Has this packet been nacked or is it about to be nacked?
- if (IsPacketRetransmitted(packet))
- {
- frame->IncrementNackCount();
- }
-
- // Insert each frame once on the arrival of the first packet
- // belonging to that frame (media or empty)
- if (state == kStateEmpty && first)
- {
- ret = kFirstPacket;
- FrameList::reverse_iterator rit = std::find_if(
- _frameList.rbegin(), _frameList.rend(),
- FrameSmallerTimestamp(frame->TimeStamp()));
- _frameList.insert(rit.base(), frame);
- }
- }
- }
- switch(bufferReturn)
- {
- case kStateError:
- case kTimeStampError:
- case kSizeError:
- {
- if (frame != NULL)
- {
- // Will be released when it gets old.
- frame->Reset();
- frame->SetState(kStateEmpty);
- }
- break;
- }
- case kCompleteSession:
- {
- // Only update return value for a JB flush indicator.
- if (UpdateFrameState(frame) == kFlushIndicator)
- ret = kFlushIndicator;
- // Signal that we have a received packet
- _packetEvent.Set();
- break;
- }
- case kDecodableSession:
- case kIncomplete:
- {
- // Signal that we have a received packet
- _packetEvent.Set();
- break;
- }
- case kNoError:
- case kDuplicatePacket:
- {
- break;
- }
- default:
- {
- assert(!"JitterBuffer::InsertPacket: Undefined value");
- }
- }
- return ret;
-}
-
-// Must be called from within _critSect
-void
-VCMJitterBuffer::UpdateOldJitterSample(const VCMPacket& packet)
-{
- if (_waitingForCompletion.timestamp != packet.timestamp &&
- LatestTimestamp(_waitingForCompletion.timestamp, packet.timestamp,
- NULL) == packet.timestamp)
- {
- // This is a newer frame than the one waiting for completion.
- _waitingForCompletion.frameSize = packet.sizeBytes;
- _waitingForCompletion.timestamp = packet.timestamp;
- }
- else
- {
- // This can get bad if we have a lot of duplicate packets,
- // we will then count some packet multiple times.
- _waitingForCompletion.frameSize += packet.sizeBytes;
- _jitterEstimate.UpdateMaxFrameSize(_waitingForCompletion.frameSize);
- }
-}
-
-// Must be called from within _critSect
-bool
-VCMJitterBuffer::IsPacketRetransmitted(const VCMPacket& packet) const
-{
- if (_NACKSeqNum && _NACKSeqNumLength > 0)
- {
- for (WebRtc_UWord16 i = 0; i < _NACKSeqNumLength; i++)
- {
- if (packet.seqNum == _NACKSeqNum[i])
- {
- return true;
- }
- }
- }
- return false;
-}
-
-// Get nack status (enabled/disabled)
-VCMNackMode
-VCMJitterBuffer::GetNackMode() const
-{
- CriticalSectionScoped cs(_critSect);
- return _nackMode;
-}
-
-// Set NACK mode
-void
-VCMJitterBuffer::SetNackMode(VCMNackMode mode,
- int lowRttNackThresholdMs,
- int highRttNackThresholdMs)
-{
- CriticalSectionScoped cs(_critSect);
- _nackMode = mode;
- assert(lowRttNackThresholdMs >= -1 && highRttNackThresholdMs >= -1);
- assert(highRttNackThresholdMs == -1 ||
- lowRttNackThresholdMs <= highRttNackThresholdMs);
- assert(lowRttNackThresholdMs > -1 || highRttNackThresholdMs == -1);
- _lowRttNackThresholdMs = lowRttNackThresholdMs;
- _highRttNackThresholdMs = highRttNackThresholdMs;
- if (_nackMode == kNoNack)
- {
- _jitterEstimate.ResetNackCount();
- }
-}
-
-
-// Recycle oldest frames up to a key frame, used if JB is completely full
-bool
-VCMJitterBuffer::RecycleFramesUntilKeyFrame()
-{
- // Remove up to oldest key frame
- while (_frameList.size() > 0)
- {
- // Throw at least one frame.
- _dropCount++;
- FrameList::iterator it = _frameList.begin();
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding,
- VCMId(_vcmId, _receiverId),
- "Jitter buffer drop count:%d, lowSeq %d", _dropCount,
- (*it)->GetLowSeqNum());
- RecycleFrame(*it);
- it = _frameList.erase(it);
- if (it != _frameList.end() && (*it)->FrameType() == kVideoFrameKey)
- {
- // Fake the lastDecodedState to match this key frame.
- _lastDecodedState.SetStateOneBack(*it);
- return true;
- }
- }
- _waitingForKeyFrame = true;
- _lastDecodedState.Reset(); // TODO (mikhal): no sync
- return false;
-}
-
-// Must be called under the critical section _critSect.
-void VCMJitterBuffer::CleanUpOldFrames() {
- while (_frameList.size() > 0) {
- VCMFrameBuffer* oldestFrame = _frameList.front();
- bool nextFrameEmpty = (_lastDecodedState.ContinuousFrame(oldestFrame) &&
- oldestFrame->GetState() == kStateEmpty);
- if (_lastDecodedState.IsOldFrame(oldestFrame) ||
- (nextFrameEmpty && _frameList.size() > 1)) {
- ReleaseFrameInternal(_frameList.front());
- _frameList.erase(_frameList.begin());
- } else {
- break;
- }
- }
-}
-
-// Used in GetFrameForDecoding
-void VCMJitterBuffer::VerifyAndSetPreviousFrameLost(VCMFrameBuffer& frame) {
- frame.MakeSessionDecodable(); // Make sure the session can be decoded.
- if (frame.FrameType() == kVideoFrameKey)
- return;
-
- if (!_lastDecodedState.ContinuousFrame(&frame))
- frame.SetPreviousFrameLoss();
-}
-
-bool
-VCMJitterBuffer::WaitForNack()
-{
- // NACK disabled -> can't wait
- if (_nackMode == kNoNack)
- {
- return false;
- }
- // NACK only -> always wait
- else if (_nackMode == kNackInfinite)
- {
- return true;
- }
- // else: hybrid mode, evaluate
- // RTT high, don't wait
- if (_highRttNackThresholdMs >= 0 &&
- _rttMs >= static_cast<unsigned int>(_highRttNackThresholdMs))
- {
- return false;
- }
- // Either NACK only or hybrid
- return true;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/jitter_buffer.h b/trunk/src/modules/video_coding/main/source/jitter_buffer.h
deleted file mode 100644
index d951187..0000000
--- a/trunk/src/modules/video_coding/main/source/jitter_buffer.h
+++ /dev/null
@@ -1,259 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_
-
-#include <list>
-
-#include "modules/interface/module_common_types.h"
-#include "modules/video_coding/main/interface/video_coding_defines.h"
-#include "modules/video_coding/main/source/decoding_state.h"
-#include "modules/video_coding/main/source/event.h"
-#include "modules/video_coding/main/source/inter_frame_delay.h"
-#include "modules/video_coding/main/source/jitter_buffer_common.h"
-#include "modules/video_coding/main/source/jitter_estimator.h"
-#include "system_wrappers/interface/constructor_magic.h"
-#include "system_wrappers/interface/critical_section_wrapper.h"
-#include "typedefs.h"
-
-namespace webrtc
-{
-
-enum VCMNackMode
-{
- kNackInfinite,
- kNackHybrid,
- kNoNack
-};
-
-typedef std::list<VCMFrameBuffer*> FrameList;
-
-// forward declarations
-class TickTimeBase;
-class VCMFrameBuffer;
-class VCMPacket;
-class VCMEncodedFrame;
-
-class VCMJitterSample
-{
-public:
- VCMJitterSample() : timestamp(0), frameSize(0), latestPacketTime(-1) {}
- WebRtc_UWord32 timestamp;
- WebRtc_UWord32 frameSize;
- WebRtc_Word64 latestPacketTime;
-};
-
-class VCMJitterBuffer
-{
-public:
- VCMJitterBuffer(TickTimeBase* clock,
- WebRtc_Word32 vcmId = -1,
- WebRtc_Word32 receiverId = -1,
- bool master = true);
- virtual ~VCMJitterBuffer();
-
- void CopyFrom(const VCMJitterBuffer& rhs);
-
- // We need a start and stop to break out of the wait event
- // used in GetCompleteFrameForDecoding
- void Start();
- void Stop();
- bool Running() const;
-
- // Empty the Jitter buffer of all its data
- void Flush();
-
- // Statistics, Get received key and delta frames
- WebRtc_Word32 GetFrameStatistics(WebRtc_UWord32& receivedDeltaFrames,
- WebRtc_UWord32& receivedKeyFrames) const;
-
- // The number of packets discarded by the jitter buffer because the decoder
- // won't be able to decode them.
- WebRtc_UWord32 NumNotDecodablePackets() const;
- // Get number of packets discarded by the jitter buffer
- WebRtc_UWord32 DiscardedPackets() const;
-
- // Statistics, Calculate frame and bit rates
- WebRtc_Word32 GetUpdate(WebRtc_UWord32& frameRate, WebRtc_UWord32& bitRate);
-
- // Wait for the first packet in the next frame to arrive, blocks
- // for <= maxWaitTimeMS ms
- WebRtc_Word64 GetNextTimeStamp(WebRtc_UWord32 maxWaitTimeMS,
- FrameType& incomingFrameType,
- WebRtc_Word64& renderTimeMs);
-
- // Will the packet sequence be complete if the next frame is grabbed
- // for decoding right now? That is, have we lost a frame between the
- // last decoded frame and the next, or is the next frame missing one
- // or more packets?
- bool CompleteSequenceWithNextFrame();
-
- // TODO (mikhal/stefan): Merge all GetFrameForDecoding into one.
- // Wait maxWaitTimeMS for a complete frame to arrive. After timeout NULL
- // is returned.
- VCMEncodedFrame* GetCompleteFrameForDecoding(WebRtc_UWord32 maxWaitTimeMS);
-
- // Get a frame for decoding (even an incomplete) without delay.
- VCMEncodedFrame* GetFrameForDecoding();
-
- VCMEncodedFrame* GetFrameForDecodingNACK();
-
- // Release frame (when done with decoding)
- void ReleaseFrame(VCMEncodedFrame* frame);
-
- // Get frame to use for this timestamp
- WebRtc_Word32 GetFrame(const VCMPacket& packet, VCMEncodedFrame*&);
- VCMEncodedFrame* GetFrame(const VCMPacket& packet); // deprecated
-
- // Returns the time in ms when the latest packet was inserted into the frame.
- // Retransmitted is set to true if any of the packets belonging to the frame
- // has been retransmitted.
- WebRtc_Word64 LastPacketTime(VCMEncodedFrame* frame,
- bool& retransmitted) const;
-
- // Insert a packet into a frame
- VCMFrameBufferEnum InsertPacket(VCMEncodedFrame* frame,
- const VCMPacket& packet);
-
- // Sync
- WebRtc_UWord32 GetEstimatedJitterMS();
- void UpdateRtt(WebRtc_UWord32 rttMs);
-
- // NACK
- // Set the NACK mode. "highRttNackThreshold" is an RTT threshold in ms above
- // which NACK will be disabled if the NACK mode is "kNackHybrid",
- // -1 meaning that NACK is always enabled in the Hybrid mode.
- // "lowRttNackThreshold" is an RTT threshold in ms below which we expect to
- // rely on NACK only, and therefore are using larger buffers to have time to
- // wait for retransmissions.
- void SetNackMode(VCMNackMode mode,
- int lowRttNackThresholdMs,
- int highRttNackThresholdMs);
- VCMNackMode GetNackMode() const; // Get nack mode
- // Get list of missing sequence numbers (size in number of elements)
- WebRtc_UWord16* GetNackList(WebRtc_UWord16& nackSize,
- bool& listExtended);
-
- WebRtc_Word64 LastDecodedTimestamp() const;
-
-private:
- // Misc help functions
- // Recycle (release) frame, used if we didn't receive whole frame
- void RecycleFrame(VCMFrameBuffer* frame);
- void ReleaseFrameInternal(VCMFrameBuffer* frame);
- // Flush and reset the jitter buffer. Call under critical section.
- void FlushInternal();
-
- // Help functions for insert packet
- // Get empty frame, creates new (i.e. increases JB size) if necessary
- VCMFrameBuffer* GetEmptyFrame();
- // Recycle oldest frames up to a key frame, used if JB is completely full
- bool RecycleFramesUntilKeyFrame();
- // Update frame state
- // (set as complete or reconstructable if conditions are met)
- VCMFrameBufferEnum UpdateFrameState(VCMFrameBuffer* frameListItem);
-
- // Help functions for getting a frame
- // Find oldest complete frame, used for getting next frame to decode
- // When enabled, will return a decodable frame
- FrameList::iterator FindOldestCompleteContinuousFrame(bool enableDecodable);
-
- void CleanUpOldFrames();
-
- void VerifyAndSetPreviousFrameLost(VCMFrameBuffer& frame);
- bool IsPacketRetransmitted(const VCMPacket& packet) const;
-
- void UpdateJitterAndDelayEstimates(VCMJitterSample& sample,
- bool incompleteFrame);
- void UpdateJitterAndDelayEstimates(VCMFrameBuffer& frame,
- bool incompleteFrame);
- void UpdateJitterAndDelayEstimates(WebRtc_Word64 latestPacketTimeMs,
- WebRtc_UWord32 timestamp,
- WebRtc_UWord32 frameSize,
- bool incompleteFrame);
- void UpdateOldJitterSample(const VCMPacket& packet);
- WebRtc_UWord32 GetEstimatedJitterMsInternal();
-
- // NACK help
- WebRtc_UWord16* CreateNackList(WebRtc_UWord16& nackSize,
- bool& listExtended);
- WebRtc_Word32 GetLowHighSequenceNumbers(WebRtc_Word32& lowSeqNum,
- WebRtc_Word32& highSeqNum) const;
-
- // Decide whether should wait for NACK (mainly relevant for hybrid mode)
- bool WaitForNack();
-
- WebRtc_Word32 _vcmId;
- WebRtc_Word32 _receiverId;
- TickTimeBase* _clock;
- // If we are running (have started) or not
- bool _running;
- CriticalSectionWrapper* _critSect;
- bool _master;
- // Event to signal when we have a frame ready for decoder
- VCMEvent _frameEvent;
- // Event to signal when we have received a packet
- VCMEvent _packetEvent;
- // Number of allocated frames
- WebRtc_Word32 _maxNumberOfFrames;
- // Array of pointers to the frames in JB
- VCMFrameBuffer* _frameBuffers[kMaxNumberOfFrames];
- FrameList _frameList;
-
- // timing
- VCMDecodingState _lastDecodedState;
- WebRtc_UWord32 _packetsNotDecodable;
-
- // Statistics
- // Frame counter for each type (key, delta, golden, key-delta)
- WebRtc_UWord8 _receiveStatistics[4];
- // Latest calculated frame rates of incoming stream
- WebRtc_UWord8 _incomingFrameRate;
- // Frame counter, reset in GetUpdate
- WebRtc_UWord32 _incomingFrameCount;
- // Real time for last _frameCount reset
- WebRtc_Word64 _timeLastIncomingFrameCount;
- // Received bits counter, reset in GetUpdate
- WebRtc_UWord32 _incomingBitCount;
- WebRtc_UWord32 _incomingBitRate;
- WebRtc_UWord32 _dropCount; // Frame drop counter
- // Number of frames in a row that have been too old
- WebRtc_UWord32 _numConsecutiveOldFrames;
- // Number of packets in a row that have been too old
- WebRtc_UWord32 _numConsecutiveOldPackets;
- // Number of packets discarded by the jitter buffer
- WebRtc_UWord32 _discardedPackets;
-
- // Filters for estimating jitter
- VCMJitterEstimator _jitterEstimate;
- // Calculates network delays used for jitter calculations
- VCMInterFrameDelay _delayEstimate;
- VCMJitterSample _waitingForCompletion;
- WebRtc_UWord32 _rttMs;
-
- // NACK
- VCMNackMode _nackMode;
- int _lowRttNackThresholdMs;
- int _highRttNackThresholdMs;
- // Holds the internal nack list (the missing sequence numbers)
- WebRtc_Word32 _NACKSeqNumInternal[kNackHistoryLength];
- WebRtc_UWord16 _NACKSeqNum[kNackHistoryLength];
- WebRtc_UWord32 _NACKSeqNumLength;
- bool _waitingForKeyFrame;
-
- bool _firstPacket;
-
- DISALLOW_COPY_AND_ASSIGN(VCMJitterBuffer);
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_H_
diff --git a/trunk/src/modules/video_coding/main/source/jitter_buffer_common.cc b/trunk/src/modules/video_coding/main/source/jitter_buffer_common.cc
deleted file mode 100644
index 79a21b4..0000000
--- a/trunk/src/modules/video_coding/main/source/jitter_buffer_common.cc
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "jitter_buffer_common.h"
-
-#include <cstdlib>
-
-namespace webrtc {
-
-WebRtc_UWord32 LatestTimestamp(WebRtc_UWord32 timestamp1,
- WebRtc_UWord32 timestamp2,
- bool* has_wrapped) {
- bool wrap = (timestamp2 < 0x0000ffff && timestamp1 > 0xffff0000) ||
- (timestamp2 > 0xffff0000 && timestamp1 < 0x0000ffff);
- if (has_wrapped != NULL)
- *has_wrapped = wrap;
- if (timestamp1 > timestamp2 && !wrap)
- return timestamp1;
- else if (timestamp1 <= timestamp2 && !wrap)
- return timestamp2;
- else if (timestamp1 < timestamp2 && wrap)
- return timestamp1;
- else
- return timestamp2;
-}
-
-WebRtc_Word32 LatestSequenceNumber(WebRtc_Word32 seq_num1,
- WebRtc_Word32 seq_num2,
- bool* has_wrapped) {
- if (seq_num1 < 0 && seq_num2 < 0)
- return -1;
- else if (seq_num1 < 0)
- return seq_num2;
- else if (seq_num2 < 0)
- return seq_num1;
-
- bool wrap = (seq_num1 < 0x00ff && seq_num2 > 0xff00) ||
- (seq_num1 > 0xff00 && seq_num2 < 0x00ff);
-
- if (has_wrapped != NULL)
- *has_wrapped = wrap;
-
- if (seq_num2 > seq_num1 && !wrap)
- return seq_num2;
- else if (seq_num2 <= seq_num1 && !wrap)
- return seq_num1;
- else if (seq_num2 < seq_num1 && wrap)
- return seq_num2;
- else
- return seq_num1;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/jitter_buffer_common.h b/trunk/src/modules/video_coding/main/source/jitter_buffer_common.h
deleted file mode 100644
index 38cea42..0000000
--- a/trunk/src/modules/video_coding/main/source/jitter_buffer_common.h
+++ /dev/null
@@ -1,84 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
-#define WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
-
-#include "typedefs.h"
-
-namespace webrtc
-{
-
-enum { kMaxNumberOfFrames = 100 };
-enum { kStartNumberOfFrames = 6 }; // in packets, 6 packets are approximately 198 ms,
- // we need at least one more for process
-enum { kMaxVideoDelayMs = 2000 }; // in ms
-
-enum VCMJitterBufferEnum
-{
- kMaxConsecutiveOldFrames = 60,
- kMaxConsecutiveOldPackets = 300,
- kMaxPacketsInSession = 800,
- kBufferIncStepSizeBytes = 30000, // >20 packets
- kMaxJBFrameSizeBytes = 4000000 // sanity don't go above 4Mbyte
-};
-
-enum VCMFrameBufferEnum
-{
- kStateError = -4,
- kFlushIndicator = -3, // Indicator that a flush has occurred.
- kTimeStampError = -2,
- kSizeError = -1,
- kNoError = 0,
- kIncomplete = 1, // Frame incomplete
- kFirstPacket = 2,
- kCompleteSession = 3, // at least one layer in the frame complete.
- kDecodableSession = 4, // Frame incomplete, but ready to be decoded
- kDuplicatePacket = 5 // We're receiving a duplicate packet.
-};
-
-enum VCMFrameBufferStateEnum
-{
- kStateFree, // Unused frame in the JB
- kStateEmpty, // frame popped by the RTP receiver
- kStateIncomplete, // frame that have one or more packet(s) stored
- kStateComplete, // frame that have all packets
- kStateDecoding, // frame popped by the decoding thread
- kStateDecodable // Hybrid mode - frame can be decoded
-};
-
-enum { kH264StartCodeLengthBytes = 4};
-
-// Used to indicate if a received packet contain a complete NALU (or equivalent)
-enum VCMNaluCompleteness
-{
- kNaluUnset = 0, //Packet has not been filled.
- kNaluComplete = 1, //Packet can be decoded as is.
- kNaluStart, // Packet contain beginning of NALU
- kNaluIncomplete, //Packet is not beginning or end of NALU
- kNaluEnd, // Packet is the end of a NALU
-};
-
-// Returns the latest of the two timestamps, compensating for wrap arounds.
-// This function assumes that the two timestamps are close in time.
-WebRtc_UWord32 LatestTimestamp(WebRtc_UWord32 timestamp1,
- WebRtc_UWord32 timestamp2,
- bool* has_wrapped);
-
-// Returns the latest of the two sequence numbers, compensating for wrap
-// arounds. This function assumes that the two sequence numbers are close in
-// time.
-WebRtc_Word32 LatestSequenceNumber(WebRtc_Word32 seq_num1,
- WebRtc_Word32 seq_num2,
- bool* has_wrapped);
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_JITTER_BUFFER_COMMON_H_
diff --git a/trunk/src/modules/video_coding/main/source/jitter_buffer_unittest.cc b/trunk/src/modules/video_coding/main/source/jitter_buffer_unittest.cc
deleted file mode 100644
index 6e48ea7..0000000
--- a/trunk/src/modules/video_coding/main/source/jitter_buffer_unittest.cc
+++ /dev/null
@@ -1,376 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string.h>
-
-#include <list>
-
-#include "gtest/gtest.h"
-#include "modules/video_coding/main/source/jitter_buffer.h"
-#include "modules/video_coding/main/source/media_opt_util.h"
-#include "modules/video_coding/main/source/mock/fake_tick_time.h"
-#include "modules/video_coding/main/source/packet.h"
-
-namespace webrtc {
-
-class StreamGenerator {
- public:
- StreamGenerator(uint16_t start_seq_num, uint32_t start_timestamp,
- int64_t current_time)
- : packets_(),
- sequence_number_(start_seq_num),
- timestamp_(start_timestamp),
- start_time_(current_time) {}
-
- void Init(uint16_t start_seq_num, uint32_t start_timestamp,
- int64_t current_time) {
- packets_.clear();
- sequence_number_ = start_seq_num;
- timestamp_ = start_timestamp;
- start_time_ = current_time;
- }
-
- void GenerateFrame(FrameType type, int num_media_packets,
- int num_empty_packets, int64_t current_time) {
- timestamp_ += 90 * (current_time - start_time_);
- // Move the sequence number counter if all packets from the previous frame
- // wasn't collected.
- sequence_number_ += packets_.size();
- packets_.clear();
- for (int i = 0; i < num_media_packets; ++i) {
- packets_.push_back(GeneratePacket(sequence_number_,
- timestamp_,
- (i == 0),
- (i == num_media_packets - 1),
- type));
- ++sequence_number_;
- }
- for (int i = 0; i < num_empty_packets; ++i) {
- packets_.push_back(GeneratePacket(sequence_number_,
- timestamp_,
- false,
- false,
- kFrameEmpty));
- ++sequence_number_;
- }
- }
-
- static VCMPacket GeneratePacket(uint16_t sequence_number,
- uint32_t timestamp,
- bool first_packet,
- bool marker_bit,
- FrameType type) {
- VCMPacket packet;
- packet.seqNum = sequence_number;
- packet.timestamp = timestamp;
- packet.frameType = type;
- packet.isFirstPacket = first_packet;
- packet.markerBit = marker_bit;
- if (packet.isFirstPacket)
- packet.completeNALU = kNaluStart;
- else if (packet.markerBit)
- packet.completeNALU = kNaluEnd;
- else
- packet.completeNALU = kNaluIncomplete;
- return packet;
- }
-
- bool PopPacket(VCMPacket* packet, int index) {
- std::list<VCMPacket>::iterator it = GetPacketIterator(index);
- if (it == packets_.end())
- return false;
- if (packet)
- *packet = (*it);
- packets_.erase(it);
- return true;
- }
-
- bool GetPacket(VCMPacket* packet, int index) {
- std::list<VCMPacket>::iterator it = GetPacketIterator(index);
- if (it == packets_.end())
- return false;
- if (packet)
- *packet = (*it);
- return true;
- }
-
- bool NextPacket(VCMPacket* packet) {
- if (packets_.empty())
- return false;
- if (packet != NULL)
- *packet = packets_.front();
- packets_.pop_front();
- return true;
- }
-
- uint16_t NextSequenceNumber() const {
- if (packets_.empty())
- return sequence_number_;
- return packets_.front().seqNum;
- }
-
- int PacketsRemaining() const {
- return packets_.size();
- }
-
- private:
- std::list<VCMPacket>::iterator GetPacketIterator(int index) {
- std::list<VCMPacket>::iterator it = packets_.begin();
- for (int i = 0; i < index; ++i) {
- ++it;
- if (it == packets_.end()) break;
- }
- return it;
- }
-
- std::list<VCMPacket> packets_;
- uint16_t sequence_number_;
- uint32_t timestamp_;
- int64_t start_time_;
-
- DISALLOW_COPY_AND_ASSIGN(StreamGenerator);
-};
-
-class TestRunningJitterBuffer : public ::testing::Test {
- protected:
- enum { kDataBufferSize = 10 };
- enum { kDefaultFrameRate = 25 };
- enum { kDefaultFramePeriodMs = 1000 / kDefaultFrameRate };
-
- virtual void SetUp() {
- clock_ = new FakeTickTime(0);
- jitter_buffer_ = new VCMJitterBuffer(clock_);
- stream_generator = new StreamGenerator(0, 0,
- clock_->MillisecondTimestamp());
- jitter_buffer_->Start();
- memset(data_buffer_, 0, kDataBufferSize);
- }
-
- virtual void TearDown() {
- jitter_buffer_->Stop();
- delete stream_generator;
- delete jitter_buffer_;
- delete clock_;
- }
-
- VCMFrameBufferEnum InsertPacketAndPop(int index) {
- VCMPacket packet;
- VCMEncodedFrame* frame;
-
- packet.dataPtr = data_buffer_;
- bool packet_available = stream_generator->PopPacket(&packet, index);
- EXPECT_TRUE(packet_available);
- if (!packet_available)
- return kStateError; // Return here to avoid crashes below.
- EXPECT_EQ(VCM_OK, jitter_buffer_->GetFrame(packet, frame));
- return jitter_buffer_->InsertPacket(frame, packet);
- }
-
- VCMFrameBufferEnum InsertPacket(int index) {
- VCMPacket packet;
- VCMEncodedFrame* frame;
-
- packet.dataPtr = data_buffer_;
- bool packet_available = stream_generator->GetPacket(&packet, index);
- EXPECT_TRUE(packet_available);
- if (!packet_available)
- return kStateError; // Return here to avoid crashes below.
- EXPECT_EQ(VCM_OK, jitter_buffer_->GetFrame(packet, frame));
- return jitter_buffer_->InsertPacket(frame, packet);
- }
-
- void InsertFrame(FrameType frame_type) {
- stream_generator->GenerateFrame(frame_type,
- (frame_type != kFrameEmpty) ? 1 : 0,
- (frame_type == kFrameEmpty) ? 1 : 0,
- clock_->MillisecondTimestamp());
- EXPECT_EQ(kFirstPacket, InsertPacketAndPop(0));
- clock_->IncrementDebugClock(kDefaultFramePeriodMs);
- }
-
- void InsertFrames(int num_frames, FrameType frame_type) {
- for (int i = 0; i < num_frames; ++i) {
- InsertFrame(frame_type);
- }
- }
-
- void DropFrame(int num_packets) {
- stream_generator->GenerateFrame(kVideoFrameDelta, num_packets, 0,
- clock_->MillisecondTimestamp());
- clock_->IncrementDebugClock(kDefaultFramePeriodMs);
- }
-
- bool DecodeCompleteFrame() {
- VCMEncodedFrame* frame = jitter_buffer_->GetCompleteFrameForDecoding(0);
- bool ret = (frame != NULL);
- jitter_buffer_->ReleaseFrame(frame);
- return ret;
- }
-
- bool DecodeFrame() {
- VCMEncodedFrame* frame = jitter_buffer_->GetFrameForDecoding();
- bool ret = (frame != NULL);
- jitter_buffer_->ReleaseFrame(frame);
- return ret;
- }
-
- VCMJitterBuffer* jitter_buffer_;
- StreamGenerator* stream_generator;
- FakeTickTime* clock_;
- uint8_t data_buffer_[kDataBufferSize];
-};
-
-class TestJitterBufferNack : public TestRunningJitterBuffer {
- protected:
- virtual void SetUp() {
- TestRunningJitterBuffer::SetUp();
- jitter_buffer_->SetNackMode(kNackInfinite, -1, -1);
- }
-
- virtual void TearDown() {
- TestRunningJitterBuffer::TearDown();
- }
-};
-
-TEST_F(TestRunningJitterBuffer, TestFull) {
- // Insert a key frame and decode it.
- InsertFrame(kVideoFrameKey);
- EXPECT_TRUE(DecodeCompleteFrame());
- DropFrame(1);
- // Fill the jitter buffer.
- InsertFrames(kMaxNumberOfFrames, kVideoFrameDelta);
- // Make sure we can't decode these frames.
- EXPECT_FALSE(DecodeCompleteFrame());
- // This frame will make the jitter buffer recycle frames until a key frame.
- // Since none is found it will have to wait until the next key frame before
- // decoding.
- InsertFrame(kVideoFrameDelta);
- EXPECT_FALSE(DecodeCompleteFrame());
-}
-
-TEST_F(TestRunningJitterBuffer, TestEmptyPackets) {
- // Make sure a frame can get complete even though empty packets are missing.
- stream_generator->GenerateFrame(kVideoFrameKey, 3, 3,
- clock_->MillisecondTimestamp());
- EXPECT_EQ(kFirstPacket, InsertPacketAndPop(4));
- EXPECT_EQ(kIncomplete, InsertPacketAndPop(4));
- EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
- EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
- EXPECT_EQ(kCompleteSession, InsertPacketAndPop(0));
-}
-
-TEST_F(TestJitterBufferNack, TestEmptyPackets) {
- // Make sure empty packets doesn't clog the jitter buffer.
- jitter_buffer_->SetNackMode(kNackHybrid, kLowRttNackMs, -1);
- InsertFrames(kMaxNumberOfFrames, kFrameEmpty);
- InsertFrame(kVideoFrameKey);
- EXPECT_TRUE(DecodeCompleteFrame());
-}
-
-TEST_F(TestJitterBufferNack, TestNackListFull) {
- // Insert a key frame and decode it.
- InsertFrame(kVideoFrameKey);
- EXPECT_TRUE(DecodeCompleteFrame());
-
- // Generate and drop |kNackHistoryLength| packets to fill the NACK list.
- DropFrame(kNackHistoryLength);
- // Insert a frame which should trigger a recycle until the next key frame.
- InsertFrame(kVideoFrameDelta);
- EXPECT_FALSE(DecodeCompleteFrame());
-
- uint16_t nack_list_length = kNackHistoryLength;
- bool extended;
- uint16_t* nack_list = jitter_buffer_->GetNackList(nack_list_length, extended);
- // Verify that the jitter buffer requests a key frame.
- EXPECT_TRUE(nack_list_length == 0xffff && nack_list == NULL);
-
- InsertFrame(kVideoFrameDelta);
- EXPECT_FALSE(DecodeCompleteFrame());
- EXPECT_FALSE(DecodeFrame());
-}
-
-TEST_F(TestJitterBufferNack, TestNackBeforeDecode) {
- DropFrame(10);
- // Insert a frame and try to generate a NACK list. Shouldn't get one.
- InsertFrame(kVideoFrameDelta);
- uint16_t nack_list_size = 0;
- bool extended = false;
- uint16_t* list = jitter_buffer_->GetNackList(nack_list_size, extended);
- // No list generated, and a key frame request is signaled.
- EXPECT_TRUE(list == NULL);
- EXPECT_EQ(0xFFFF, nack_list_size);
-}
-
-TEST_F(TestJitterBufferNack, TestNormalOperation) {
- EXPECT_EQ(kNackInfinite, jitter_buffer_->GetNackMode());
-
- InsertFrame(kVideoFrameKey);
- EXPECT_TRUE(DecodeFrame());
-
- // ----------------------------------------------------------------
- // | 1 | 2 | .. | 8 | 9 | x | 11 | 12 | .. | 19 | x | 21 | .. | 100 |
- // ----------------------------------------------------------------
- stream_generator->GenerateFrame(kVideoFrameKey, 100, 0,
- clock_->MillisecondTimestamp());
- clock_->IncrementDebugClock(kDefaultFramePeriodMs);
- EXPECT_EQ(kFirstPacket, InsertPacketAndPop(0));
- // Verify that the frame is incomplete.
- EXPECT_FALSE(DecodeCompleteFrame());
- while (stream_generator->PacketsRemaining() > 1) {
- if (stream_generator->NextSequenceNumber() % 10 != 0)
- EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
- else
- stream_generator->NextPacket(NULL); // Drop packet
- }
- EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
- EXPECT_EQ(0, stream_generator->PacketsRemaining());
- EXPECT_FALSE(DecodeCompleteFrame());
- EXPECT_FALSE(DecodeFrame());
- uint16_t nack_list_size = 0;
- bool extended = false;
- uint16_t* list = jitter_buffer_->GetNackList(nack_list_size, extended);
- // Verify the NACK list.
- const int kExpectedNackSize = 9;
- ASSERT_EQ(kExpectedNackSize, nack_list_size);
- for (int i = 0; i < nack_list_size; ++i)
- EXPECT_EQ((1 + i) * 10, list[i]);
-}
-
-TEST_F(TestJitterBufferNack, TestNormalOperationWrap) {
- // ------- ------------------------------------------------------------
- // | 65532 | | 65533 | 65534 | 65535 | x | 1 | .. | 9 | x | 11 |.....| 96 |
- // ------- ------------------------------------------------------------
- stream_generator->Init(65532, 0, clock_->MillisecondTimestamp());
- InsertFrame(kVideoFrameKey);
- EXPECT_TRUE(DecodeCompleteFrame());
- stream_generator->GenerateFrame(kVideoFrameDelta, 100, 0,
- clock_->MillisecondTimestamp());
- EXPECT_EQ(kFirstPacket, InsertPacketAndPop(0));
- while (stream_generator->PacketsRemaining() > 1) {
- if (stream_generator->NextSequenceNumber() % 10 != 0)
- EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
- else
- stream_generator->NextPacket(NULL); // Drop packet
- }
- EXPECT_EQ(kIncomplete, InsertPacketAndPop(0));
- EXPECT_EQ(0, stream_generator->PacketsRemaining());
- EXPECT_FALSE(DecodeCompleteFrame());
- EXPECT_FALSE(DecodeCompleteFrame());
- uint16_t nack_list_size = 0;
- bool extended = false;
- uint16_t* list = jitter_buffer_->GetNackList(nack_list_size, extended);
- // Verify the NACK list.
- const int kExpectedNackSize = 10;
- ASSERT_EQ(kExpectedNackSize, nack_list_size);
- for (int i = 0; i < nack_list_size; ++i)
- EXPECT_EQ(i * 10, list[i]);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/jitter_estimator.cc b/trunk/src/modules/video_coding/main/source/jitter_estimator.cc
deleted file mode 100644
index 93c6ccb..0000000
--- a/trunk/src/modules/video_coding/main/source/jitter_estimator.cc
+++ /dev/null
@@ -1,422 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "trace.h"
-#include "internal_defines.h"
-#include "jitter_estimator.h"
-#include "rtt_filter.h"
-
-#include <assert.h>
-#include <math.h>
-#include <stdlib.h>
-#include <string.h>
-
-namespace webrtc {
-
-VCMJitterEstimator::VCMJitterEstimator(WebRtc_Word32 vcmId, WebRtc_Word32 receiverId) :
-_vcmId(vcmId),
-_receiverId(receiverId),
-_phi(0.97),
-_psi(0.9999),
-_alphaCountMax(400),
-_beta(0.9994),
-_thetaLow(0.000001),
-_nackLimit(3),
-_numStdDevDelayOutlier(15),
-_numStdDevFrameSizeOutlier(3),
-_noiseStdDevs(2.33), // ~Less than 1% chance
- // (look up in normal distribution table)...
-_noiseStdDevOffset(30.0), // ...of getting 30 ms freezes
-_rttFilter(vcmId, receiverId)
-{
- Reset();
-}
-
-VCMJitterEstimator&
-VCMJitterEstimator::operator=(const VCMJitterEstimator& rhs)
-{
- if (this != &rhs)
- {
- memcpy(_thetaCov, rhs._thetaCov, sizeof(_thetaCov));
- memcpy(_Qcov, rhs._Qcov, sizeof(_Qcov));
-
- _vcmId = rhs._vcmId;
- _receiverId = rhs._receiverId;
- _avgFrameSize = rhs._avgFrameSize;
- _varFrameSize = rhs._varFrameSize;
- _maxFrameSize = rhs._maxFrameSize;
- _fsSum = rhs._fsSum;
- _fsCount = rhs._fsCount;
- _lastUpdateT = rhs._lastUpdateT;
- _prevEstimate = rhs._prevEstimate;
- _prevFrameSize = rhs._prevFrameSize;
- _avgNoise = rhs._avgNoise;
- _alphaCount = rhs._alphaCount;
- _filterJitterEstimate = rhs._filterJitterEstimate;
- _startupCount = rhs._startupCount;
- _latestNackTimestamp = rhs._latestNackTimestamp;
- _nackCount = rhs._nackCount;
- _rttFilter = rhs._rttFilter;
- }
- return *this;
-}
-
-// Resets the JitterEstimate
-void
-VCMJitterEstimator::Reset()
-{
- _theta[0] = 1/(512e3/8);
- _theta[1] = 0;
- _varNoise = 4.0;
-
- _thetaCov[0][0] = 1e-4;
- _thetaCov[1][1] = 1e2;
- _thetaCov[0][1] = _thetaCov[1][0] = 0;
- _Qcov[0][0] = 2.5e-10;
- _Qcov[1][1] = 1e-10;
- _Qcov[0][1] = _Qcov[1][0] = 0;
- _avgFrameSize = 500;
- _maxFrameSize = 500;
- _varFrameSize = 100;
- _lastUpdateT = -1;
- _prevEstimate = -1.0;
- _prevFrameSize = 0;
- _avgNoise = 0.0;
- _alphaCount = 1;
- _filterJitterEstimate = 0.0;
- _latestNackTimestamp = 0;
- _nackCount = 0;
- _fsSum = 0;
- _fsCount = 0;
- _startupCount = 0;
- _rttFilter.Reset();
-}
-
-void
-VCMJitterEstimator::ResetNackCount()
-{
- _nackCount = 0;
-}
-
-// Updates the estimates with the new measurements
-void
-VCMJitterEstimator::UpdateEstimate(WebRtc_Word64 frameDelayMS, WebRtc_UWord32 frameSizeBytes,
- bool incompleteFrame /* = false */)
-{
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
- VCMId(_vcmId, _receiverId),
- "Jitter estimate updated with: frameSize=%d frameDelayMS=%d",
- frameSizeBytes, frameDelayMS);
- if (frameSizeBytes == 0)
- {
- return;
- }
- int deltaFS = frameSizeBytes - _prevFrameSize;
- if (_fsCount < kFsAccuStartupSamples)
- {
- _fsSum += frameSizeBytes;
- _fsCount++;
- }
- else if (_fsCount == kFsAccuStartupSamples)
- {
- // Give the frame size filter
- _avgFrameSize = static_cast<double>(_fsSum) /
- static_cast<double>(_fsCount);
- _fsCount++;
- }
- if (!incompleteFrame || frameSizeBytes > _avgFrameSize)
- {
- double avgFrameSize = _phi * _avgFrameSize +
- (1 - _phi) * frameSizeBytes;
- if (frameSizeBytes < _avgFrameSize + 2 * sqrt(_varFrameSize))
- {
- // Only update the average frame size if this sample wasn't a
- // key frame
- _avgFrameSize = avgFrameSize;
- }
- // Update the variance anyway since we want to capture cases where we only get
- // key frames.
- _varFrameSize = VCM_MAX(_phi * _varFrameSize + (1 - _phi) *
- (frameSizeBytes - avgFrameSize) *
- (frameSizeBytes - avgFrameSize), 1.0);
- }
-
- // Update max frameSize estimate
- _maxFrameSize = VCM_MAX(_psi * _maxFrameSize, static_cast<double>(frameSizeBytes));
-
- if (_prevFrameSize == 0)
- {
- _prevFrameSize = frameSizeBytes;
- return;
- }
- _prevFrameSize = frameSizeBytes;
-
- // Only update the Kalman filter if the sample is not considered
- // an extreme outlier. Even if it is an extreme outlier from a
- // delay point of view, if the frame size also is large the
- // deviation is probably due to an incorrect line slope.
- double deviation = DeviationFromExpectedDelay(frameDelayMS, deltaFS);
-
- if (abs(deviation) < _numStdDevDelayOutlier * sqrt(_varNoise) ||
- frameSizeBytes > _avgFrameSize + _numStdDevFrameSizeOutlier * sqrt(_varFrameSize))
- {
- // Update the variance of the deviation from the
- // line given by the Kalman filter
- EstimateRandomJitter(deviation, incompleteFrame);
- // Prevent updating with frames which have been congested by a large
- // frame, and therefore arrives almost at the same time as that frame.
- // This can occur when we receive a large frame (key frame) which
- // has been delayed. The next frame is of normal size (delta frame),
- // and thus deltaFS will be << 0. This removes all frame samples
- // which arrives after a key frame.
- if ((!incompleteFrame || deviation >= 0.0) &&
- static_cast<double>(deltaFS) > - 0.25 * _maxFrameSize)
- {
- // Update the Kalman filter with the new data
- KalmanEstimateChannel(frameDelayMS, deltaFS);
- }
- }
- else
- {
- int nStdDev = (deviation >= 0) ? _numStdDevDelayOutlier : -_numStdDevDelayOutlier;
- EstimateRandomJitter(nStdDev * sqrt(_varNoise), incompleteFrame);
- }
- // Post process the total estimated jitter
- if (_startupCount >= kStartupDelaySamples)
- {
- PostProcessEstimate();
- }
- else
- {
- _startupCount++;
- }
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
- "Framesize statistics: max=%f average=%f", _maxFrameSize, _avgFrameSize);
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
- "The estimated slope is: theta=(%f, %f)", _theta[0], _theta[1]);
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
- "Random jitter: mean=%f variance=%f", _avgNoise, _varNoise);
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
- "Current jitter estimate: %f", _filterJitterEstimate);
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
- "Current max RTT: %u", _rttFilter.RttMs());
-}
-
-// Updates the nack/packet ratio
-void
-VCMJitterEstimator::FrameNacked()
-{
- // Wait until _nackLimit retransmissions has been received,
- // then always add ~1 RTT delay.
- // TODO(holmer): Should we ever remove the additional delay if the
- // the packet losses seem to have stopped? We could for instance scale
- // the number of RTTs to add with the amount of retransmissions in a given
- // time interval, or similar.
- if (_nackCount < _nackLimit)
- {
- _nackCount++;
- }
-}
-
-// Updates Kalman estimate of the channel
-// The caller is expected to sanity check the inputs.
-void
-VCMJitterEstimator::KalmanEstimateChannel(WebRtc_Word64 frameDelayMS,
- WebRtc_Word32 deltaFSBytes)
-{
- double Mh[2];
- double hMh_sigma;
- double kalmanGain[2];
- double measureRes;
- double t00, t01;
-
- // Kalman filtering
-
- // Prediction
- // M = M + Q
- _thetaCov[0][0] += _Qcov[0][0];
- _thetaCov[0][1] += _Qcov[0][1];
- _thetaCov[1][0] += _Qcov[1][0];
- _thetaCov[1][1] += _Qcov[1][1];
-
- // Kalman gain
- // K = M*h'/(sigma2n + h*M*h') = M*h'/(1 + h*M*h')
- // h = [dFS 1]
- // Mh = M*h'
- // hMh_sigma = h*M*h' + R
- Mh[0] = _thetaCov[0][0] * deltaFSBytes + _thetaCov[0][1];
- Mh[1] = _thetaCov[1][0] * deltaFSBytes + _thetaCov[1][1];
- // sigma weights measurements with a small deltaFS as noisy and
- // measurements with large deltaFS as good
- if (_maxFrameSize < 1.0)
- {
- return;
- }
- double sigma = (300.0 * exp(-abs(static_cast<double>(deltaFSBytes)) /
- (1e0 * _maxFrameSize)) + 1) * sqrt(_varNoise);
- if (sigma < 1.0)
- {
- sigma = 1.0;
- }
- hMh_sigma = deltaFSBytes * Mh[0] + Mh[1] + sigma;
- if ((hMh_sigma < 1e-9 && hMh_sigma >= 0) || (hMh_sigma > -1e-9 && hMh_sigma <= 0))
- {
- assert(false);
- return;
- }
- kalmanGain[0] = Mh[0] / hMh_sigma;
- kalmanGain[1] = Mh[1] / hMh_sigma;
-
- // Correction
- // theta = theta + K*(dT - h*theta)
- measureRes = frameDelayMS - (deltaFSBytes * _theta[0] + _theta[1]);
- _theta[0] += kalmanGain[0] * measureRes;
- _theta[1] += kalmanGain[1] * measureRes;
-
- if (_theta[0] < _thetaLow)
- {
- _theta[0] = _thetaLow;
- }
-
- // M = (I - K*h)*M
- t00 = _thetaCov[0][0];
- t01 = _thetaCov[0][1];
- _thetaCov[0][0] = (1 - kalmanGain[0] * deltaFSBytes) * t00 -
- kalmanGain[0] * _thetaCov[1][0];
- _thetaCov[0][1] = (1 - kalmanGain[0] * deltaFSBytes) * t01 -
- kalmanGain[0] * _thetaCov[1][1];
- _thetaCov[1][0] = _thetaCov[1][0] * (1 - kalmanGain[1]) -
- kalmanGain[1] * deltaFSBytes * t00;
- _thetaCov[1][1] = _thetaCov[1][1] * (1 - kalmanGain[1]) -
- kalmanGain[1] * deltaFSBytes * t01;
-
- // Covariance matrix, must be positive semi-definite
- assert(_thetaCov[0][0] + _thetaCov[1][1] >= 0 &&
- _thetaCov[0][0] * _thetaCov[1][1] - _thetaCov[0][1] * _thetaCov[1][0] >= 0 &&
- _thetaCov[0][0] >= 0);
-}
-
-// Calculate difference in delay between a sample and the
-// expected delay estimated by the Kalman filter
-double
-VCMJitterEstimator::DeviationFromExpectedDelay(WebRtc_Word64 frameDelayMS,
- WebRtc_Word32 deltaFSBytes) const
-{
- return frameDelayMS - (_theta[0] * deltaFSBytes + _theta[1]);
-}
-
-// Estimates the random jitter by calculating the variance of the
-// sample distance from the line given by theta.
-void
-VCMJitterEstimator::EstimateRandomJitter(double d_dT, bool incompleteFrame)
-{
- double alpha;
- if (_alphaCount == 0)
- {
- assert(_alphaCount > 0);
- return;
- }
- alpha = static_cast<double>(_alphaCount - 1) / static_cast<double>(_alphaCount);
- _alphaCount++;
- if (_alphaCount > _alphaCountMax)
- {
- _alphaCount = _alphaCountMax;
- }
- double avgNoise = alpha * _avgNoise + (1 - alpha) * d_dT;
- double varNoise = alpha * _varNoise +
- (1 - alpha) * (d_dT - _avgNoise) * (d_dT - _avgNoise);
- if (!incompleteFrame || varNoise > _varNoise)
- {
- _avgNoise = avgNoise;
- _varNoise = varNoise;
- }
- if (_varNoise < 1.0)
- {
- // The variance should never be zero, since we might get
- // stuck and consider all samples as outliers.
- _varNoise = 1.0;
- }
-}
-
-double
-VCMJitterEstimator::NoiseThreshold() const
-{
- double noiseThreshold = _noiseStdDevs * sqrt(_varNoise) - _noiseStdDevOffset;
- if (noiseThreshold < 1.0)
- {
- noiseThreshold = 1.0;
- }
- return noiseThreshold;
-}
-
-// Calculates the current jitter estimate from the filtered estimates
-double
-VCMJitterEstimator::CalculateEstimate()
-{
- double ret = _theta[0] * (_maxFrameSize - _avgFrameSize) + NoiseThreshold();
-
- // A very low estimate (or negative) is neglected
- if (ret < 1.0) {
- if (_prevEstimate <= 0.01)
- {
- ret = 1.0;
- }
- else
- {
- ret = _prevEstimate;
- }
- }
- if (ret > 10000.0) // Sanity
- {
- ret = 10000.0;
- }
- _prevEstimate = ret;
- return ret;
-}
-
-void
-VCMJitterEstimator::PostProcessEstimate()
-{
- _filterJitterEstimate = CalculateEstimate();
-}
-
-void
-VCMJitterEstimator::UpdateRtt(WebRtc_UWord32 rttMs)
-{
- _rttFilter.Update(rttMs);
-}
-
-void
-VCMJitterEstimator::UpdateMaxFrameSize(WebRtc_UWord32 frameSizeBytes)
-{
- if (_maxFrameSize < frameSizeBytes)
- {
- _maxFrameSize = frameSizeBytes;
- }
-}
-
-// Returns the current filtered estimate if available,
-// otherwise tries to calculate an estimate.
-double
-VCMJitterEstimator::GetJitterEstimate(double rttMultiplier)
-{
- double jitterMS = CalculateEstimate();
- if (_filterJitterEstimate > jitterMS)
- {
- jitterMS = _filterJitterEstimate;
- }
- if (_nackCount >= _nackLimit)
- {
- return jitterMS + _rttFilter.RttMs() * rttMultiplier;
- }
- return jitterMS;
-}
-
-}
diff --git a/trunk/src/modules/video_coding/main/source/jitter_estimator.h b/trunk/src/modules/video_coding/main/source/jitter_estimator.h
deleted file mode 100644
index 6fc4703..0000000
--- a/trunk/src/modules/video_coding/main/source/jitter_estimator.h
+++ /dev/null
@@ -1,155 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
-#define WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
-
-#include "typedefs.h"
-#include "rtt_filter.h"
-
-namespace webrtc
-{
-
-class VCMJitterEstimator
-{
-public:
- VCMJitterEstimator(WebRtc_Word32 vcmId = 0, WebRtc_Word32 receiverId = 0);
-
- VCMJitterEstimator& operator=(const VCMJitterEstimator& rhs);
-
- // Resets the estimate to the initial state
- void Reset();
- void ResetNackCount();
-
- // Updates the jitter estimate with the new data.
- //
- // Input:
- // - frameDelay : Delay-delta calculated by UTILDelayEstimate in milliseconds
- // - frameSize : Frame size of the current frame.
- // - incompleteFrame : Flags if the frame is used to update the estimate before it
- // was complete. Default is false.
- void UpdateEstimate(WebRtc_Word64 frameDelayMS,
- WebRtc_UWord32 frameSizeBytes,
- bool incompleteFrame = false);
-
- // Returns the current jitter estimate in milliseconds and adds
- // also adds an RTT dependent term in cases of retransmission.
- // Input:
- // - rttMultiplier : RTT param multiplier (when applicable).
- //
- // Return value : Jitter estimate in milliseconds
- double GetJitterEstimate(double rttMultiplier);
-
- // Updates the nack counter.
- void FrameNacked();
-
- // Updates the RTT filter.
- //
- // Input:
- // - rttMs : RTT in ms
- void UpdateRtt(WebRtc_UWord32 rttMs);
-
- void UpdateMaxFrameSize(WebRtc_UWord32 frameSizeBytes);
-
- // A constant describing the delay from the jitter buffer
- // to the delay on the receiving side which is not accounted
- // for by the jitter buffer nor the decoding delay estimate.
- static const WebRtc_UWord32 OPERATING_SYSTEM_JITTER = 10;
-
-protected:
- // These are protected for better testing possibilities
- double _theta[2]; // Estimated line parameters (slope, offset)
- double _varNoise; // Variance of the time-deviation from the line
-
-private:
- // Updates the Kalman filter for the line describing
- // the frame size dependent jitter.
- //
- // Input:
- // - frameDelayMS : Delay-delta calculated by UTILDelayEstimate in milliseconds
- // - deltaFSBytes : Frame size delta, i.e.
- // : frame size at time T minus frame size at time T-1
- void KalmanEstimateChannel(WebRtc_Word64 frameDelayMS, WebRtc_Word32 deltaFSBytes);
-
- // Updates the random jitter estimate, i.e. the variance
- // of the time deviations from the line given by the Kalman filter.
- //
- // Input:
- // - d_dT : The deviation from the kalman estimate
- // - incompleteFrame : True if the frame used to update the estimate
- // with was incomplete
- void EstimateRandomJitter(double d_dT, bool incompleteFrame);
-
- double NoiseThreshold() const;
-
- // Calculates the current jitter estimate.
- //
- // Return value : The current jitter estimate in milliseconds
- double CalculateEstimate();
-
- // Post process the calculated estimate
- void PostProcessEstimate();
-
- // Calculates the difference in delay between a sample and the
- // expected delay estimated by the Kalman filter.
- //
- // Input:
- // - frameDelayMS : Delay-delta calculated by UTILDelayEstimate in milliseconds
- // - deltaFS : Frame size delta, i.e. frame size at time
- // T minus frame size at time T-1
- //
- // Return value : The difference in milliseconds
- double DeviationFromExpectedDelay(WebRtc_Word64 frameDelayMS,
- WebRtc_Word32 deltaFSBytes) const;
-
- // Constants, filter parameters
- WebRtc_Word32 _vcmId;
- WebRtc_Word32 _receiverId;
- const double _phi;
- const double _psi;
- const WebRtc_UWord32 _alphaCountMax;
- const double _beta;
- const double _thetaLow;
- const WebRtc_UWord32 _nackLimit;
- const WebRtc_Word32 _numStdDevDelayOutlier;
- const WebRtc_Word32 _numStdDevFrameSizeOutlier;
- const double _noiseStdDevs;
- const double _noiseStdDevOffset;
-
- double _thetaCov[2][2]; // Estimate covariance
- double _Qcov[2][2]; // Process noise covariance
- double _avgFrameSize; // Average frame size
- double _varFrameSize; // Frame size variance
- double _maxFrameSize; // Largest frame size received (descending
- // with a factor _psi)
- WebRtc_UWord32 _fsSum;
- WebRtc_UWord32 _fsCount;
-
- WebRtc_Word64 _lastUpdateT;
- double _prevEstimate; // The previously returned jitter estimate
- WebRtc_UWord32 _prevFrameSize; // Frame size of the previous frame
- double _avgNoise; // Average of the random jitter
- WebRtc_UWord32 _alphaCount;
- double _filterJitterEstimate; // The filtered sum of jitter estimates
-
- WebRtc_UWord32 _startupCount;
-
- WebRtc_Word64 _latestNackTimestamp; // Timestamp in ms when the latest nack was seen
- WebRtc_UWord32 _nackCount; // Keeps track of the number of nacks received,
- // but never goes above _nackLimit
- VCMRttFilter _rttFilter;
-
- enum { kStartupDelaySamples = 30 };
- enum { kFsAccuStartupSamples = 5 };
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_JITTER_ESTIMATOR_H_
diff --git a/trunk/src/modules/video_coding/main/source/media_opt_util.cc b/trunk/src/modules/video_coding/main/source/media_opt_util.cc
deleted file mode 100644
index b520278..0000000
--- a/trunk/src/modules/video_coding/main/source/media_opt_util.cc
+++ /dev/null
@@ -1,893 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/video_coding/main/source/media_opt_util.h"
-
-#include <math.h>
-#include <float.h>
-#include <limits.h>
-
-#include "modules/interface/module_common_types.h"
-#include "modules/video_coding/codecs/vp8/main/interface/vp8_common_types.h"
-#include "modules/video_coding/main/interface/video_coding_defines.h"
-#include "modules/video_coding/main/source/er_tables_xor.h"
-#include "modules/video_coding/main/source/fec_tables_xor.h"
-#include "modules/video_coding/main/source/nack_fec_tables.h"
-
-namespace webrtc {
-
-VCMProtectionMethod::VCMProtectionMethod():
-_effectivePacketLoss(0),
-_protectionFactorK(0),
-_protectionFactorD(0),
-_residualPacketLossFec(0.0f),
-_scaleProtKey(2.0f),
-_maxPayloadSize(1460),
-_qmRobustness(new VCMQmRobustness()),
-_useUepProtectionK(false),
-_useUepProtectionD(true),
-_corrFecCost(1.0),
-_type(kNone),
-_efficiency(0)
-{
- //
-}
-
-VCMProtectionMethod::~VCMProtectionMethod()
-{
- delete _qmRobustness;
-}
-void
-VCMProtectionMethod::UpdateContentMetrics(const
- VideoContentMetrics* contentMetrics)
-{
- _qmRobustness->UpdateContent(contentMetrics);
-}
-
-VCMNackFecMethod::VCMNackFecMethod(int lowRttNackThresholdMs,
- int highRttNackThresholdMs)
- : VCMFecMethod(),
- _lowRttNackMs(lowRttNackThresholdMs),
- _highRttNackMs(highRttNackThresholdMs) {
- assert(lowRttNackThresholdMs >= -1 && highRttNackThresholdMs >= -1);
- assert(highRttNackThresholdMs == -1 ||
- lowRttNackThresholdMs <= highRttNackThresholdMs);
- assert(lowRttNackThresholdMs > -1 || highRttNackThresholdMs == -1);
- _type = kNackFec;
-}
-
-VCMNackFecMethod::~VCMNackFecMethod()
-{
- //
-}
-bool
-VCMNackFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters)
-{
- // Hybrid Nack FEC has three operational modes:
- // 1. Low RTT (below kLowRttNackMs) - Nack only: Set FEC rate
- // (_protectionFactorD) to zero. -1 means no FEC.
- // 2. High RTT (above _highRttNackMs) - FEC Only: Keep FEC factors.
- // -1 means always allow NACK.
- // 3. Medium RTT values - Hybrid mode: We will only nack the
- // residual following the decoding of the FEC (refer to JB logic). FEC
- // delta protection factor will be adjusted based on the RTT.
-
- // Otherwise: we count on FEC; if the RTT is below a threshold, then we
- // nack the residual, based on a decision made in the JB.
-
- // Compute the protection factors
- VCMFecMethod::ProtectionFactor(parameters);
- if (_lowRttNackMs == -1 || parameters->rtt < _lowRttNackMs)
- {
- _protectionFactorD = 0;
- VCMFecMethod::UpdateProtectionFactorD(_protectionFactorD);
- }
-
- // When in Hybrid mode (RTT range), adjust FEC rates based on the
- // RTT (NACK effectiveness) - adjustment factor is in the range [0,1].
- else if (_highRttNackMs == -1 || parameters->rtt < _highRttNackMs)
- {
- // TODO(mikhal): Disabling adjustment temporarily.
- // WebRtc_UWord16 rttIndex = (WebRtc_UWord16) parameters->rtt;
- float adjustRtt = 1.0f;// (float)VCMNackFecTable[rttIndex] / 100.0f;
-
- // Adjust FEC with NACK on (for delta frame only)
- // table depends on RTT relative to rttMax (NACK Threshold)
- _protectionFactorD = static_cast<WebRtc_UWord8>
- (adjustRtt *
- static_cast<float>(_protectionFactorD));
- // update FEC rates after applying adjustment
- VCMFecMethod::UpdateProtectionFactorD(_protectionFactorD);
- }
-
- return true;
-}
-
-bool
-VCMNackFecMethod::EffectivePacketLoss(const VCMProtectionParameters* parameters)
-{
- // Set the effective packet loss for encoder (based on FEC code).
- // Compute the effective packet loss and residual packet loss due to FEC.
- VCMFecMethod::EffectivePacketLoss(parameters);
- return true;
-}
-
-bool
-VCMNackFecMethod::UpdateParameters(const VCMProtectionParameters* parameters)
-{
- ProtectionFactor(parameters);
- EffectivePacketLoss(parameters);
-
- // Efficiency computation is based on FEC and NACK
-
- // Add FEC cost: ignore I frames for now
- float fecRate = static_cast<float> (_protectionFactorD) / 255.0f;
- _efficiency = parameters->bitRate * fecRate * _corrFecCost;
-
- // Add NACK cost, when applicable
- if (_highRttNackMs == -1 || parameters->rtt < _highRttNackMs)
- {
- // nackCost = (bitRate - nackCost) * (lossPr)
- _efficiency += parameters->bitRate * _residualPacketLossFec /
- (1.0f + _residualPacketLossFec);
- }
-
- // Protection/fec rates obtained above are defined relative to total number
- // of packets (total rate: source + fec) FEC in RTP module assumes
- // protection factor is defined relative to source number of packets so we
- // should convert the factor to reduce mismatch between mediaOpt's rate and
- // the actual one
- _protectionFactorK = VCMFecMethod::ConvertFECRate(_protectionFactorK);
- _protectionFactorD = VCMFecMethod::ConvertFECRate(_protectionFactorD);
-
- return true;
-}
-
-VCMNackMethod::VCMNackMethod():
-VCMProtectionMethod()
-{
- _type = kNack;
-}
-
-VCMNackMethod::~VCMNackMethod()
-{
- //
-}
-
-bool
-VCMNackMethod::EffectivePacketLoss(const VCMProtectionParameters* parameter)
-{
- // Effective Packet Loss, NA in current version.
- _effectivePacketLoss = 0;
- return true;
-}
-
-bool
-VCMNackMethod::UpdateParameters(const VCMProtectionParameters* parameters)
-{
- // Compute the effective packet loss
- EffectivePacketLoss(parameters);
-
- // nackCost = (bitRate - nackCost) * (lossPr)
- _efficiency = parameters->bitRate * parameters->lossPr /
- (1.0f + parameters->lossPr);
- return true;
-}
-
-VCMFecMethod::VCMFecMethod():
-VCMProtectionMethod()
-{
- _type = kFec;
-}
-VCMFecMethod::~VCMFecMethod()
-{
- //
-}
-
-WebRtc_UWord8
-VCMFecMethod::BoostCodeRateKey(WebRtc_UWord8 packetFrameDelta,
- WebRtc_UWord8 packetFrameKey) const
-{
- WebRtc_UWord8 boostRateKey = 2;
- // Default: ratio scales the FEC protection up for I frames
- WebRtc_UWord8 ratio = 1;
-
- if (packetFrameDelta > 0)
- {
- ratio = (WebRtc_Word8) (packetFrameKey / packetFrameDelta);
- }
- ratio = VCM_MAX(boostRateKey, ratio);
-
- return ratio;
-}
-
-WebRtc_UWord8
-VCMFecMethod::ConvertFECRate(WebRtc_UWord8 codeRateRTP) const
-{
- return static_cast<WebRtc_UWord8> (VCM_MIN(255,(0.5 + 255.0 * codeRateRTP /
- (float)(255 - codeRateRTP))));
-}
-
-// Update FEC with protectionFactorD
-void
-VCMFecMethod::UpdateProtectionFactorD(WebRtc_UWord8 protectionFactorD)
-{
- _protectionFactorD = protectionFactorD;
-}
-
-// Update FEC with protectionFactorK
-void
-VCMFecMethod::UpdateProtectionFactorK(WebRtc_UWord8 protectionFactorK)
-{
- _protectionFactorK = protectionFactorK;
-}
-
-// AvgRecoveryFEC: computes the residual packet loss (RPL) function.
-// This is the average recovery from the FEC, assuming random packet loss model.
-// Computed off-line for a range of FEC code parameters and loss rates.
-float
-VCMFecMethod::AvgRecoveryFEC(const VCMProtectionParameters* parameters) const
-{
- // Total (avg) bits available per frame: total rate over actual/sent frame
- // rate units are kbits/frame
- const WebRtc_UWord16 bitRatePerFrame = static_cast<WebRtc_UWord16>
- (parameters->bitRate / (parameters->frameRate));
-
- // Total (average) number of packets per frame (source and fec):
- const WebRtc_UWord8 avgTotPackets = 1 + static_cast<WebRtc_UWord8>
- (static_cast<float> (bitRatePerFrame * 1000.0) /
- static_cast<float> (8.0 * _maxPayloadSize) + 0.5);
-
- const float protectionFactor = static_cast<float>(_protectionFactorD) /
- 255.0;
-
- WebRtc_UWord8 fecPacketsPerFrame = static_cast<WebRtc_UWord8>
- (0.5 + protectionFactor * avgTotPackets);
-
- WebRtc_UWord8 sourcePacketsPerFrame = avgTotPackets - fecPacketsPerFrame;
-
- if ( (fecPacketsPerFrame == 0) || (sourcePacketsPerFrame == 0) )
- {
- // No protection, or rate too low: so average recovery from FEC == 0.
- return 0.0;
- }
-
- // Table defined up to kMaxNumPackets
- if (sourcePacketsPerFrame > kMaxNumPackets)
- {
- sourcePacketsPerFrame = kMaxNumPackets;
- }
-
- // Table defined up to kMaxNumPackets
- if (fecPacketsPerFrame > kMaxNumPackets)
- {
- fecPacketsPerFrame = kMaxNumPackets;
- }
-
- // Code index for tables: up to (kMaxNumPackets * kMaxNumPackets)
- WebRtc_UWord16 codeIndexTable[kMaxNumPackets * kMaxNumPackets];
- WebRtc_UWord16 k = 0;
- for (WebRtc_UWord8 i = 1; i <= kMaxNumPackets; i++)
- {
- for (WebRtc_UWord8 j = 1; j <= i; j++)
- {
- codeIndexTable[(j - 1) * kMaxNumPackets + i - 1] = k;
- k += 1;
- }
- }
-
- WebRtc_UWord8 lossRate = static_cast<WebRtc_UWord8> (255.0 *
- parameters->lossPr + 0.5f);
-
- // Constrain lossRate to 50%: tables defined up to 50%
- if (lossRate >= kPacketLossMax)
- {
- lossRate = kPacketLossMax - 1;
- }
-
- const WebRtc_UWord16 codeIndex = (fecPacketsPerFrame - 1) * kMaxNumPackets +
- (sourcePacketsPerFrame - 1);
-
- const WebRtc_UWord16 indexTable = codeIndexTable[codeIndex] * kPacketLossMax +
- lossRate;
-
- // Check on table index
- assert(indexTable < kSizeAvgFECRecoveryXOR);
- float avgFecRecov = static_cast<float>(kAvgFECRecoveryXOR[indexTable]);
-
- return avgFecRecov;
-}
-
-bool
-VCMFecMethod::ProtectionFactor(const VCMProtectionParameters* parameters)
-{
- // FEC PROTECTION SETTINGS: varies with packet loss and bitrate
-
- // No protection if (filtered) packetLoss is 0
- WebRtc_UWord8 packetLoss = (WebRtc_UWord8) (255 * parameters->lossPr);
- if (packetLoss == 0)
- {
- _protectionFactorK = 0;
- _protectionFactorD = 0;
- return true;
- }
-
- // Parameters for FEC setting:
- // first partition size, thresholds, table pars, spatial resoln fac.
-
- // First partition protection: ~ 20%
- WebRtc_UWord8 firstPartitionProt = (WebRtc_UWord8) (255 * 0.20);
-
- // Minimum protection level needed to generate one FEC packet for one
- // source packet/frame (in RTP sender)
- WebRtc_UWord8 minProtLevelFec = 85;
-
- // Threshold on packetLoss and bitRrate/frameRate (=average #packets),
- // above which we allocate protection to cover at least first partition.
- WebRtc_UWord8 lossThr = 0;
- WebRtc_UWord8 packetNumThr = 1;
-
- // Parameters for range of rate index of table.
- const WebRtc_UWord8 ratePar1 = 5;
- const WebRtc_UWord8 ratePar2 = 49;
-
- // Spatial resolution size, relative to a reference size.
- float spatialSizeToRef = static_cast<float>
- (parameters->codecWidth * parameters->codecHeight) /
- (static_cast<float>(704 * 576));
- // resolnFac: This parameter will generally increase/decrease the FEC rate
- // (for fixed bitRate and packetLoss) based on system size.
- // Use a smaller exponent (< 1) to control/soften system size effect.
- const float resolnFac = 1.0 / powf(spatialSizeToRef, 0.3f);
-
- const int bitRatePerFrame = BitsPerFrame(parameters);
-
-
- // Average number of packets per frame (source and fec):
- const WebRtc_UWord8 avgTotPackets = 1 + (WebRtc_UWord8)
- ((float) bitRatePerFrame * 1000.0
- / (float) (8.0 * _maxPayloadSize) + 0.5);
-
- // FEC rate parameters: for P and I frame
- WebRtc_UWord8 codeRateDelta = 0;
- WebRtc_UWord8 codeRateKey = 0;
-
- // Get index for table: the FEC protection depends on an effective rate.
- // The range on the rate index corresponds to rates (bps)
- // from ~200k to ~8000k, for 30fps
- const WebRtc_UWord16 effRateFecTable = static_cast<WebRtc_UWord16>
- (resolnFac * bitRatePerFrame);
- WebRtc_UWord8 rateIndexTable =
- (WebRtc_UWord8) VCM_MAX(VCM_MIN((effRateFecTable - ratePar1) /
- ratePar1, ratePar2), 0);
-
- // Restrict packet loss range to 50:
- // current tables defined only up to 50%
- if (packetLoss >= kPacketLossMax)
- {
- packetLoss = kPacketLossMax - 1;
- }
- WebRtc_UWord16 indexTable = rateIndexTable * kPacketLossMax + packetLoss;
-
- // Check on table index
- assert(indexTable < kSizeCodeRateXORTable);
-
- // Protection factor for P frame
- codeRateDelta = kCodeRateXORTable[indexTable];
-
- if (packetLoss > lossThr && avgTotPackets > packetNumThr)
- {
- // Set a minimum based on first partition size.
- if (codeRateDelta < firstPartitionProt)
- {
- codeRateDelta = firstPartitionProt;
- }
- }
-
- // Check limit on amount of protection for P frame; 50% is max.
- if (codeRateDelta >= kPacketLossMax)
- {
- codeRateDelta = kPacketLossMax - 1;
- }
-
- float adjustFec = 1.0f;
- // Avoid additional adjustments when layers are active.
- // TODO(mikhal/marco): Update adjusmtent based on layer info.
- if (parameters->numLayers == 1)
- {
- adjustFec = _qmRobustness->AdjustFecFactor(codeRateDelta,
- parameters->bitRate,
- parameters->frameRate,
- parameters->rtt,
- packetLoss);
- }
-
- codeRateDelta = static_cast<WebRtc_UWord8>(codeRateDelta * adjustFec);
-
- // For Key frame:
- // Effectively at a higher rate, so we scale/boost the rate
- // The boost factor may depend on several factors: ratio of packet
- // number of I to P frames, how much protection placed on P frames, etc.
- const WebRtc_UWord8 packetFrameDelta = (WebRtc_UWord8)
- (0.5 + parameters->packetsPerFrame);
- const WebRtc_UWord8 packetFrameKey = (WebRtc_UWord8)
- (0.5 + parameters->packetsPerFrameKey);
- const WebRtc_UWord8 boostKey = BoostCodeRateKey(packetFrameDelta,
- packetFrameKey);
-
- rateIndexTable = (WebRtc_UWord8) VCM_MAX(VCM_MIN(
- 1 + (boostKey * effRateFecTable - ratePar1) /
- ratePar1,ratePar2),0);
- WebRtc_UWord16 indexTableKey = rateIndexTable * kPacketLossMax + packetLoss;
-
- indexTableKey = VCM_MIN(indexTableKey, kSizeCodeRateXORTable);
-
- // Check on table index
- assert(indexTableKey < kSizeCodeRateXORTable);
-
- // Protection factor for I frame
- codeRateKey = kCodeRateXORTable[indexTableKey];
-
- // Boosting for Key frame.
- int boostKeyProt = _scaleProtKey * codeRateDelta;
- if (boostKeyProt >= kPacketLossMax)
- {
- boostKeyProt = kPacketLossMax - 1;
- }
-
- // Make sure I frame protection is at least larger than P frame protection,
- // and at least as high as filtered packet loss.
- codeRateKey = static_cast<WebRtc_UWord8> (VCM_MAX(packetLoss,
- VCM_MAX(boostKeyProt, codeRateKey)));
-
- // Check limit on amount of protection for I frame: 50% is max.
- if (codeRateKey >= kPacketLossMax)
- {
- codeRateKey = kPacketLossMax - 1;
- }
-
- _protectionFactorK = codeRateKey;
- _protectionFactorD = codeRateDelta;
-
- // Generally there is a rate mis-match between the FEC cost estimated
- // in mediaOpt and the actual FEC cost sent out in RTP module.
- // This is more significant at low rates (small # of source packets), where
- // the granularity of the FEC decreases. In this case, non-zero protection
- // in mediaOpt may generate 0 FEC packets in RTP sender (since actual #FEC
- // is based on rounding off protectionFactor on actual source packet number).
- // The correction factor (_corrFecCost) attempts to corrects this, at least
- // for cases of low rates (small #packets) and low protection levels.
-
- float numPacketsFl = 1.0f + ((float) bitRatePerFrame * 1000.0
- / (float) (8.0 * _maxPayloadSize) + 0.5);
-
- const float estNumFecGen = 0.5f + static_cast<float> (_protectionFactorD *
- numPacketsFl / 255.0f);
-
-
- // We reduce cost factor (which will reduce overhead for FEC and
- // hybrid method) and not the protectionFactor.
- _corrFecCost = 1.0f;
- if (estNumFecGen < 1.1f && _protectionFactorD < minProtLevelFec)
- {
- _corrFecCost = 0.5f;
- }
- if (estNumFecGen < 0.9f && _protectionFactorD < minProtLevelFec)
- {
- _corrFecCost = 0.0f;
- }
-
- // TODO (marpan): Set the UEP protection on/off for Key and Delta frames
- _useUepProtectionK = _qmRobustness->SetUepProtection(codeRateKey,
- parameters->bitRate,
- packetLoss,
- 0);
-
- _useUepProtectionD = _qmRobustness->SetUepProtection(codeRateDelta,
- parameters->bitRate,
- packetLoss,
- 1);
-
- // DONE WITH FEC PROTECTION SETTINGS
- return true;
-}
-
-int VCMFecMethod::BitsPerFrame(const VCMProtectionParameters* parameters) {
- // When temporal layers are available FEC will only be applied on the base
- // layer.
- const float bitRateRatio =
- kVp8LayerRateAlloction[parameters->numLayers - 1][0];
- float frameRateRatio = powf(1 / 2, parameters->numLayers - 1);
- float bitRate = parameters->bitRate * bitRateRatio;
- float frameRate = parameters->frameRate * frameRateRatio;
-
- // TODO(mikhal): Update factor following testing.
- float adjustmentFactor = 1;
-
- // Average bits per frame (units of kbits)
- return static_cast<int>(adjustmentFactor * bitRate / frameRate);
-}
-
-bool
-VCMFecMethod::EffectivePacketLoss(const VCMProtectionParameters* parameters)
-{
- // Effective packet loss to encoder is based on RPL (residual packet loss)
- // this is a soft setting based on degree of FEC protection
- // RPL = received/input packet loss - average_FEC_recovery
- // note: received/input packet loss may be filtered based on FilteredLoss
-
- // The packet loss:
- WebRtc_UWord8 packetLoss = (WebRtc_UWord8) (255 * parameters->lossPr);
-
- float avgFecRecov = AvgRecoveryFEC(parameters);
-
- // Residual Packet Loss:
- _residualPacketLossFec = (float) (packetLoss - avgFecRecov) / 255.0f;
-
- // Effective Packet Loss, NA in current version.
- _effectivePacketLoss = 0;
-
- return true;
-}
-
-bool
-VCMFecMethod::UpdateParameters(const VCMProtectionParameters* parameters)
-{
- // Compute the protection factor
- ProtectionFactor(parameters);
-
- // Compute the effective packet loss
- EffectivePacketLoss(parameters);
-
- // Compute the bit cost
- // Ignore key frames for now.
- float fecRate = static_cast<float> (_protectionFactorD) / 255.0f;
- if (fecRate >= 0.0f)
- {
- // use this formula if the fecRate (protection factor) is defined
- // relative to number of source packets
- // this is the case for the previous tables:
- // _efficiency = parameters->bitRate * ( 1.0 - 1.0 / (1.0 + fecRate));
-
- // in the new tables, the fecRate is defined relative to total number of
- // packets (total rate), so overhead cost is:
- _efficiency = parameters->bitRate * fecRate * _corrFecCost;
- }
- else
- {
- _efficiency = 0.0f;
- }
-
- // Protection/fec rates obtained above is defined relative to total number
- // of packets (total rate: source+fec) FEC in RTP module assumes protection
- // factor is defined relative to source number of packets so we should
- // convert the factor to reduce mismatch between mediaOpt suggested rate and
- // the actual rate
- _protectionFactorK = ConvertFECRate(_protectionFactorK);
- _protectionFactorD = ConvertFECRate(_protectionFactorD);
-
- return true;
-}
-VCMLossProtectionLogic::VCMLossProtectionLogic(int64_t nowMs):
-_selectedMethod(NULL),
-_currentParameters(),
-_rtt(0),
-_lossPr(0.0f),
-_bitRate(0.0f),
-_frameRate(0.0f),
-_keyFrameSize(0.0f),
-_fecRateKey(0),
-_fecRateDelta(0),
-_lastPrUpdateT(0),
-_lossPr255(0.9999f),
-_lossPrHistory(),
-_shortMaxLossPr255(0),
-_packetsPerFrame(0.9999f),
-_packetsPerFrameKey(0.9999f),
-_residualPacketLossFec(0),
-_boostRateKey(2),
-_codecWidth(0),
-_codecHeight(0),
-_numLayers(1)
-{
- Reset(nowMs);
-}
-
-VCMLossProtectionLogic::~VCMLossProtectionLogic()
-{
- Release();
-}
-
-bool
-VCMLossProtectionLogic::SetMethod(enum VCMProtectionMethodEnum newMethodType)
-{
- if (_selectedMethod != NULL)
- {
- if (_selectedMethod->Type() == newMethodType)
- {
- // Nothing to update
- return false;
- }
- // New method - delete existing one
- delete _selectedMethod;
- }
- VCMProtectionMethod *newMethod = NULL;
- switch (newMethodType)
- {
- case kNack:
- {
- newMethod = new VCMNackMethod();
- break;
- }
- case kFec:
- {
- newMethod = new VCMFecMethod();
- break;
- }
- case kNackFec:
- {
- // Default to always having NACK enabled for the hybrid mode.
- newMethod = new VCMNackFecMethod(kLowRttNackMs, -1);
- break;
- }
- default:
- {
- return false;
- break;
- }
-
- }
- _selectedMethod = newMethod;
- return true;
-}
-bool
-VCMLossProtectionLogic::RemoveMethod(enum VCMProtectionMethodEnum method)
-{
- if (_selectedMethod == NULL)
- {
- return false;
- }
- else if (_selectedMethod->Type() == method)
- {
- delete _selectedMethod;
- _selectedMethod = NULL;
- }
- return true;
-}
-
-float
-VCMLossProtectionLogic::RequiredBitRate() const
-{
- float RequiredBitRate = 0.0f;
- if (_selectedMethod != NULL)
- {
- RequiredBitRate = _selectedMethod->RequiredBitRate();
- }
- return RequiredBitRate;
-}
-
-void
-VCMLossProtectionLogic::UpdateRtt(WebRtc_UWord32 rtt)
-{
- _rtt = rtt;
-}
-
-void
-VCMLossProtectionLogic::UpdateResidualPacketLoss(float residualPacketLoss)
-{
- _residualPacketLossFec = residualPacketLoss;
-}
-
-void
-VCMLossProtectionLogic::UpdateMaxLossHistory(WebRtc_UWord8 lossPr255,
- WebRtc_Word64 now)
-{
- if (_lossPrHistory[0].timeMs >= 0 &&
- now - _lossPrHistory[0].timeMs < kLossPrShortFilterWinMs)
- {
- if (lossPr255 > _shortMaxLossPr255)
- {
- _shortMaxLossPr255 = lossPr255;
- }
- }
- else
- {
- // Only add a new value to the history once a second
- if (_lossPrHistory[0].timeMs == -1)
- {
- // First, no shift
- _shortMaxLossPr255 = lossPr255;
- }
- else
- {
- // Shift
- for (WebRtc_Word32 i = (kLossPrHistorySize - 2); i >= 0; i--)
- {
- _lossPrHistory[i + 1].lossPr255 = _lossPrHistory[i].lossPr255;
- _lossPrHistory[i + 1].timeMs = _lossPrHistory[i].timeMs;
- }
- }
- if (_shortMaxLossPr255 == 0)
- {
- _shortMaxLossPr255 = lossPr255;
- }
-
- _lossPrHistory[0].lossPr255 = _shortMaxLossPr255;
- _lossPrHistory[0].timeMs = now;
- _shortMaxLossPr255 = 0;
- }
-}
-
-WebRtc_UWord8
-VCMLossProtectionLogic::MaxFilteredLossPr(WebRtc_Word64 nowMs) const
-{
- WebRtc_UWord8 maxFound = _shortMaxLossPr255;
- if (_lossPrHistory[0].timeMs == -1)
- {
- return maxFound;
- }
- for (WebRtc_Word32 i = 0; i < kLossPrHistorySize; i++)
- {
- if (_lossPrHistory[i].timeMs == -1)
- {
- break;
- }
- if (nowMs - _lossPrHistory[i].timeMs >
- kLossPrHistorySize * kLossPrShortFilterWinMs)
- {
- // This sample (and all samples after this) is too old
- break;
- }
- if (_lossPrHistory[i].lossPr255 > maxFound)
- {
- // This sample is the largest one this far into the history
- maxFound = _lossPrHistory[i].lossPr255;
- }
- }
- return maxFound;
-}
-
-WebRtc_UWord8 VCMLossProtectionLogic::FilteredLoss(
- int64_t nowMs,
- FilterPacketLossMode filter_mode,
- WebRtc_UWord8 lossPr255) {
-
- // Update the max window filter.
- UpdateMaxLossHistory(lossPr255, nowMs);
-
- // Update the recursive average filter.
- _lossPr255.Apply(static_cast<float> (nowMs - _lastPrUpdateT),
- static_cast<float> (lossPr255));
- _lastPrUpdateT = nowMs;
-
- // Filtered loss: default is received loss (no filtering).
- WebRtc_UWord8 filtered_loss = lossPr255;
-
- switch (filter_mode) {
- case kNoFilter:
- break;
- case kAvgFilter:
- filtered_loss = static_cast<WebRtc_UWord8> (_lossPr255.Value() + 0.5);
- break;
- case kMaxFilter:
- filtered_loss = MaxFilteredLossPr(nowMs);
- break;
- }
-
- return filtered_loss;
-}
-
-void
-VCMLossProtectionLogic::UpdateFilteredLossPr(WebRtc_UWord8 packetLossEnc)
-{
- _lossPr = (float) packetLossEnc / (float) 255.0;
-}
-
-void
-VCMLossProtectionLogic::UpdateBitRate(float bitRate)
-{
- _bitRate = bitRate;
-}
-
-void
-VCMLossProtectionLogic::UpdatePacketsPerFrame(float nPackets, int64_t nowMs)
-{
- _packetsPerFrame.Apply(static_cast<float>(nowMs - _lastPacketPerFrameUpdateT),
- nPackets);
- _lastPacketPerFrameUpdateT = nowMs;
-}
-
-void
-VCMLossProtectionLogic::UpdatePacketsPerFrameKey(float nPackets, int64_t nowMs)
-{
- _packetsPerFrameKey.Apply(static_cast<float>(nowMs -
- _lastPacketPerFrameUpdateTKey), nPackets);
- _lastPacketPerFrameUpdateTKey = nowMs;
-}
-
-void
-VCMLossProtectionLogic::UpdateKeyFrameSize(float keyFrameSize)
-{
- _keyFrameSize = keyFrameSize;
-}
-
-void
-VCMLossProtectionLogic::UpdateFrameSize(WebRtc_UWord16 width,
- WebRtc_UWord16 height)
-{
- _codecWidth = width;
- _codecHeight = height;
-}
-
-void VCMLossProtectionLogic::UpdateNumLayers(int numLayers) {
- _numLayers = (numLayers == 0) ? 1 : numLayers;
-}
-
-bool
-VCMLossProtectionLogic::UpdateMethod()
-{
- if (_selectedMethod == NULL)
- {
- return false;
- }
- _currentParameters.rtt = _rtt;
- _currentParameters.lossPr = _lossPr;
- _currentParameters.bitRate = _bitRate;
- _currentParameters.frameRate = _frameRate; // rename actual frame rate?
- _currentParameters.keyFrameSize = _keyFrameSize;
- _currentParameters.fecRateDelta = _fecRateDelta;
- _currentParameters.fecRateKey = _fecRateKey;
- _currentParameters.packetsPerFrame = _packetsPerFrame.Value();
- _currentParameters.packetsPerFrameKey = _packetsPerFrameKey.Value();
- _currentParameters.residualPacketLossFec = _residualPacketLossFec;
- _currentParameters.codecWidth = _codecWidth;
- _currentParameters.codecHeight = _codecHeight;
- _currentParameters.numLayers = _numLayers;
- return _selectedMethod->UpdateParameters(&_currentParameters);
-}
-
-VCMProtectionMethod*
-VCMLossProtectionLogic::SelectedMethod() const
-{
- return _selectedMethod;
-}
-
-VCMProtectionMethodEnum
-VCMLossProtectionLogic::SelectedType() const
-{
- return _selectedMethod->Type();
-}
-
-void
-VCMLossProtectionLogic::Reset(int64_t nowMs)
-{
- _lastPrUpdateT = nowMs;
- _lastPacketPerFrameUpdateT = nowMs;
- _lastPacketPerFrameUpdateTKey = nowMs;
- _lossPr255.Reset(0.9999f);
- _packetsPerFrame.Reset(0.9999f);
- _fecRateDelta = _fecRateKey = 0;
- for (WebRtc_Word32 i = 0; i < kLossPrHistorySize; i++)
- {
- _lossPrHistory[i].lossPr255 = 0;
- _lossPrHistory[i].timeMs = -1;
- }
- _shortMaxLossPr255 = 0;
- Release();
-}
-
-void
-VCMLossProtectionLogic::Release()
-{
- delete _selectedMethod;
- _selectedMethod = NULL;
-}
-
-}
diff --git a/trunk/src/modules/video_coding/main/source/media_opt_util.h b/trunk/src/modules/video_coding/main/source/media_opt_util.h
deleted file mode 100644
index 1f808de..0000000
--- a/trunk/src/modules/video_coding/main/source/media_opt_util.h
+++ /dev/null
@@ -1,375 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
-
-#include "typedefs.h"
-#include "trace.h"
-#include "exp_filter.h"
-#include "internal_defines.h"
-#include "qm_select.h"
-
-#include <cmath>
-#include <cstdlib>
-
-
-namespace webrtc
-{
-// Number of time periods used for (max) window filter for packet loss
-// TODO (marpan): set reasonable window size for filtered packet loss,
-// adjustment should be based on logged/real data of loss stats/correlation.
-enum { kLossPrHistorySize = 10 };
-
-// 1000 ms, total filter length is (kLossPrHistorySize * 1000) ms
-enum { kLossPrShortFilterWinMs = 1000 };
-
-// The type of filter used on the received packet loss reports.
-enum FilterPacketLossMode {
- kNoFilter, // No filtering on received loss.
- kAvgFilter, // Recursive average filter.
- kMaxFilter // Max-window filter, over the time interval of:
- // (kLossPrHistorySize * kLossPrShortFilterWinMs) ms.
-};
-
-// Thresholds for hybrid NACK/FEC
-// common to media optimization and the jitter buffer.
-enum HybridNackTH {
- kHighRttNackMs = 100,
- kLowRttNackMs = 20
-};
-
-struct VCMProtectionParameters
-{
- VCMProtectionParameters() : rtt(0), lossPr(0.0f), bitRate(0.0f),
- packetsPerFrame(0.0f), packetsPerFrameKey(0.0f), frameRate(0.0f),
- keyFrameSize(0.0f), fecRateDelta(0), fecRateKey(0),
- residualPacketLossFec(0.0f), codecWidth(0), codecHeight(0),
- numLayers(1)
- {}
-
- int rtt;
- float lossPr;
- float bitRate;
- float packetsPerFrame;
- float packetsPerFrameKey;
- float frameRate;
- float keyFrameSize;
- WebRtc_UWord8 fecRateDelta;
- WebRtc_UWord8 fecRateKey;
- float residualPacketLossFec;
- WebRtc_UWord16 codecWidth;
- WebRtc_UWord16 codecHeight;
- int numLayers;
-};
-
-
-/******************************/
-/* VCMProtectionMethod class */
-/****************************/
-
-enum VCMProtectionMethodEnum
-{
- kNack,
- kFec,
- kNackFec,
- kNone
-};
-
-class VCMLossProbabilitySample
-{
-public:
- VCMLossProbabilitySample() : lossPr255(0), timeMs(-1) {};
-
- WebRtc_UWord8 lossPr255;
- WebRtc_Word64 timeMs;
-};
-
-
-class VCMProtectionMethod
-{
-public:
- VCMProtectionMethod();
- virtual ~VCMProtectionMethod();
-
- // Updates the efficiency of the method using the parameters provided
- //
- // Input:
- // - parameters : Parameters used to calculate efficiency
- //
- // Return value : True if this method is recommended in
- // the given conditions.
- virtual bool UpdateParameters(const VCMProtectionParameters* parameters) = 0;
-
- // Returns the protection type
- //
- // Return value : The protection type
- enum VCMProtectionMethodEnum Type() const { return _type; }
-
- // Returns the bit rate required by this protection method
- // during these conditions.
- //
- // Return value : Required bit rate
- virtual float RequiredBitRate() { return _efficiency; }
-
- // Returns the effective packet loss for ER, required by this protection method
- //
- // Return value : Required effective packet loss
- virtual WebRtc_UWord8 RequiredPacketLossER() { return _effectivePacketLoss; }
-
- // Extracts the FEC protection factor for Key frame, required by this protection method
- //
- // Return value : Required protectionFactor for Key frame
- virtual WebRtc_UWord8 RequiredProtectionFactorK() { return _protectionFactorK; }
-
- // Extracts the FEC protection factor for Delta frame, required by this protection method
- //
- // Return value : Required protectionFactor for delta frame
- virtual WebRtc_UWord8 RequiredProtectionFactorD() { return _protectionFactorD; }
-
- // Extracts whether the FEC Unequal protection (UEP) is used for Key frame.
- //
- // Return value : Required Unequal protection on/off state.
- virtual bool RequiredUepProtectionK() { return _useUepProtectionK; }
-
- // Extracts whether the the FEC Unequal protection (UEP) is used for Delta frame.
- //
- // Return value : Required Unequal protection on/off state.
- virtual bool RequiredUepProtectionD() { return _useUepProtectionD; }
-
- // Updates content metrics
- void UpdateContentMetrics(const VideoContentMetrics* contentMetrics);
-
-protected:
-
- WebRtc_UWord8 _effectivePacketLoss;
- WebRtc_UWord8 _protectionFactorK;
- WebRtc_UWord8 _protectionFactorD;
- // Estimation of residual loss after the FEC
- float _residualPacketLossFec;
- float _scaleProtKey;
- WebRtc_Word32 _maxPayloadSize;
-
- VCMQmRobustness* _qmRobustness;
- bool _useUepProtectionK;
- bool _useUepProtectionD;
- float _corrFecCost;
- enum VCMProtectionMethodEnum _type;
- float _efficiency;
-};
-
-class VCMNackMethod : public VCMProtectionMethod
-{
-public:
- VCMNackMethod();
- virtual ~VCMNackMethod();
- virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
- // Get the effective packet loss
- bool EffectivePacketLoss(const VCMProtectionParameters* parameter);
-};
-
-class VCMFecMethod : public VCMProtectionMethod
-{
-public:
- VCMFecMethod();
- virtual ~VCMFecMethod();
- virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
- // Get the effective packet loss for ER
- bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
- // Get the FEC protection factors
- bool ProtectionFactor(const VCMProtectionParameters* parameters);
- // Get the boost for key frame protection
- WebRtc_UWord8 BoostCodeRateKey(WebRtc_UWord8 packetFrameDelta,
- WebRtc_UWord8 packetFrameKey) const;
- // Convert the rates: defined relative to total# packets or source# packets
- WebRtc_UWord8 ConvertFECRate(WebRtc_UWord8 codeRate) const;
- // Get the average effective recovery from FEC: for random loss model
- float AvgRecoveryFEC(const VCMProtectionParameters* parameters) const;
- // Update FEC with protectionFactorD
- void UpdateProtectionFactorD(WebRtc_UWord8 protectionFactorD);
- // Update FEC with protectionFactorK
- void UpdateProtectionFactorK(WebRtc_UWord8 protectionFactorK);
- // Compute the bits per frame. Account for temporal layers when applicable.
- int BitsPerFrame(const VCMProtectionParameters* parameters);
-};
-
-
-class VCMNackFecMethod : public VCMFecMethod
-{
-public:
- VCMNackFecMethod(int lowRttNackThresholdMs,
- int highRttNackThresholdMs);
- virtual ~VCMNackFecMethod();
- virtual bool UpdateParameters(const VCMProtectionParameters* parameters);
- // Get the effective packet loss for ER
- bool EffectivePacketLoss(const VCMProtectionParameters* parameters);
- // Get the protection factors
- bool ProtectionFactor(const VCMProtectionParameters* parameters);
-
-private:
- int _lowRttNackMs;
- int _highRttNackMs;
-};
-
-class VCMLossProtectionLogic
-{
-public:
- VCMLossProtectionLogic(int64_t nowMs);
- ~VCMLossProtectionLogic();
-
- // Set the protection method to be used
- //
- // Input:
- // - newMethodType : New requested protection method type. If one
- // is already set, it will be deleted and replaced
- // Return value: Returns true on update
- bool SetMethod(enum VCMProtectionMethodEnum newMethodType);
-
- // Remove requested protection method
- // Input:
- // - method : method to be removed (if currently selected)
- //
- // Return value: Returns true on update
- bool RemoveMethod(enum VCMProtectionMethodEnum method);
-
- // Return required bit rate per selected protectin method
- float RequiredBitRate() const;
-
- // Update the round-trip time
- //
- // Input:
- // - rtt : Round-trip time in seconds.
- void UpdateRtt(WebRtc_UWord32 rtt);
-
- // Update residual packet loss
- //
- // Input:
- // - residualPacketLoss : residual packet loss:
- // effective loss after FEC recovery
- void UpdateResidualPacketLoss(float _residualPacketLoss);
-
- // Update the filtered packet loss.
- //
- // Input:
- // - packetLossEnc : The reported packet loss filtered
- // (max window or average)
- void UpdateFilteredLossPr(WebRtc_UWord8 packetLossEnc);
-
- // Update the current target bit rate.
- //
- // Input:
- // - bitRate : The current target bit rate in kbits/s
- void UpdateBitRate(float bitRate);
-
- // Update the number of packets per frame estimate, for delta frames
- //
- // Input:
- // - nPackets : Number of packets in the latest sent frame.
- void UpdatePacketsPerFrame(float nPackets, int64_t nowMs);
-
- // Update the number of packets per frame estimate, for key frames
- //
- // Input:
- // - nPackets : umber of packets in the latest sent frame.
- void UpdatePacketsPerFrameKey(float nPackets, int64_t nowMs);
-
- // Update the keyFrameSize estimate
- //
- // Input:
- // - keyFrameSize : The size of the latest sent key frame.
- void UpdateKeyFrameSize(float keyFrameSize);
-
- // Update the frame rate
- //
- // Input:
- // - frameRate : The current target frame rate.
- void UpdateFrameRate(float frameRate) { _frameRate = frameRate; }
-
- // Update the frame size
- //
- // Input:
- // - width : The codec frame width.
- // - height : The codec frame height.
- void UpdateFrameSize(WebRtc_UWord16 width, WebRtc_UWord16 height);
-
- // Update the number of active layers
- //
- // Input:
- // - numLayers : Number of layers used.
- void UpdateNumLayers(int numLayers);
-
- // The amount of packet loss to cover for with FEC.
- //
- // Input:
- // - fecRateKey : Packet loss to cover for with FEC when
- // sending key frames.
- // - fecRateDelta : Packet loss to cover for with FEC when
- // sending delta frames.
- void UpdateFECRates(WebRtc_UWord8 fecRateKey, WebRtc_UWord8 fecRateDelta)
- { _fecRateKey = fecRateKey;
- _fecRateDelta = fecRateDelta; }
-
- // Update the protection methods with the current VCMProtectionParameters
- // and set the requested protection settings.
- // Return value : Returns true on update
- bool UpdateMethod();
-
- // Returns the method currently selected.
- //
- // Return value : The protection method currently selected.
- VCMProtectionMethod* SelectedMethod() const;
-
- // Return the protection type of the currently selected method
- VCMProtectionMethodEnum SelectedType() const;
-
- // Updates the filtered loss for the average and max window packet loss,
- // and returns the filtered loss probability in the interval [0, 255].
- // The returned filtered loss value depends on the parameter |filter_mode|.
- // The input parameter |lossPr255| is the received packet loss.
-
- // Return value : The filtered loss probability
- WebRtc_UWord8 FilteredLoss(int64_t nowMs, FilterPacketLossMode filter_mode,
- WebRtc_UWord8 lossPr255);
-
- void Reset(int64_t nowMs);
-
- void Release();
-
-private:
- // Sets the available loss protection methods.
- void UpdateMaxLossHistory(WebRtc_UWord8 lossPr255, WebRtc_Word64 now);
- WebRtc_UWord8 MaxFilteredLossPr(WebRtc_Word64 nowMs) const;
- VCMProtectionMethod* _selectedMethod;
- VCMProtectionParameters _currentParameters;
- WebRtc_UWord32 _rtt;
- float _lossPr;
- float _bitRate;
- float _frameRate;
- float _keyFrameSize;
- WebRtc_UWord8 _fecRateKey;
- WebRtc_UWord8 _fecRateDelta;
- WebRtc_Word64 _lastPrUpdateT;
- WebRtc_Word64 _lastPacketPerFrameUpdateT;
- WebRtc_Word64 _lastPacketPerFrameUpdateTKey;
- VCMExpFilter _lossPr255;
- VCMLossProbabilitySample _lossPrHistory[kLossPrHistorySize];
- WebRtc_UWord8 _shortMaxLossPr255;
- VCMExpFilter _packetsPerFrame;
- VCMExpFilter _packetsPerFrameKey;
- float _residualPacketLossFec;
- WebRtc_UWord8 _boostRateKey;
- WebRtc_UWord16 _codecWidth;
- WebRtc_UWord16 _codecHeight;
- int _numLayers;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPT_UTIL_H_
diff --git a/trunk/src/modules/video_coding/main/source/media_optimization.cc b/trunk/src/modules/video_coding/main/source/media_optimization.cc
deleted file mode 100644
index cae4912..0000000
--- a/trunk/src/modules/video_coding/main/source/media_optimization.cc
+++ /dev/null
@@ -1,697 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "media_optimization.h"
-#include "content_metrics_processing.h"
-#include "frame_dropper.h"
-#include "qm_select.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
-
-namespace webrtc {
-
-VCMMediaOptimization::VCMMediaOptimization(WebRtc_Word32 id,
- TickTimeBase* clock):
-_id(id),
-_clock(clock),
-_maxBitRate(0),
-_sendCodecType(kVideoCodecUnknown),
-_codecWidth(0),
-_codecHeight(0),
-_initCodecWidth(0),
-_initCodecHeight(0),
-_userFrameRate(0),
-_packetLossEnc(0),
-_fractionLost(0),
-_sendStatisticsZeroEncode(0),
-_maxPayloadSize(1460),
-_targetBitRate(0),
-_incomingFrameRate(0),
-_enableQm(false),
-_videoProtectionCallback(NULL),
-_videoQMSettingsCallback(NULL),
-_encodedFrameSamples(),
-_avgSentBitRateBps(0.0f),
-_keyFrameCnt(0),
-_deltaFrameCnt(0),
-_lastQMUpdateTime(0),
-_lastChangeTime(0),
-_numLayers(0)
-{
- memset(_sendStatistics, 0, sizeof(_sendStatistics));
- memset(_incomingFrameTimes, -1, sizeof(_incomingFrameTimes));
-
- _frameDropper = new VCMFrameDropper(_id);
- _lossProtLogic = new VCMLossProtectionLogic(_clock->MillisecondTimestamp());
- _content = new VCMContentMetricsProcessing();
- _qmResolution = new VCMQmResolution();
-}
-
-VCMMediaOptimization::~VCMMediaOptimization(void)
-{
- _lossProtLogic->Release();
- delete _lossProtLogic;
- delete _frameDropper;
- delete _content;
- delete _qmResolution;
-}
-
-WebRtc_Word32
-VCMMediaOptimization::Reset()
-{
- memset(_incomingFrameTimes, -1, sizeof(_incomingFrameTimes));
- _incomingFrameRate = 0.0;
- _frameDropper->Reset();
- _lossProtLogic->Reset(_clock->MillisecondTimestamp());
- _frameDropper->SetRates(0, 0);
- _content->Reset();
- _qmResolution->Reset();
- _lossProtLogic->UpdateFrameRate(_incomingFrameRate);
- _lossProtLogic->Reset(_clock->MillisecondTimestamp());
- _sendStatisticsZeroEncode = 0;
- _targetBitRate = 0;
- _codecWidth = 0;
- _codecHeight = 0;
- _userFrameRate = 0;
- _keyFrameCnt = 0;
- _deltaFrameCnt = 0;
- _lastQMUpdateTime = 0;
- _lastChangeTime = 0;
- for (WebRtc_Word32 i = 0; i < kBitrateMaxFrameSamples; i++)
- {
- _encodedFrameSamples[i]._sizeBytes = -1;
- _encodedFrameSamples[i]._timeCompleteMs = -1;
- }
- _avgSentBitRateBps = 0.0f;
- _numLayers = 1;
- return VCM_OK;
-}
-
-WebRtc_UWord32
-VCMMediaOptimization::SetTargetRates(WebRtc_UWord32 bitRate,
- WebRtc_UWord8 &fractionLost,
- WebRtc_UWord32 roundTripTimeMs)
-{
- VCMProtectionMethod *selectedMethod = _lossProtLogic->SelectedMethod();
- _lossProtLogic->UpdateBitRate(static_cast<float>(bitRate));
- _lossProtLogic->UpdateRtt(roundTripTimeMs);
- _lossProtLogic->UpdateResidualPacketLoss(static_cast<float>(fractionLost));
-
- // Get frame rate for encoder: this is the actual/sent frame rate
- float actualFrameRate = SentFrameRate();
-
- // sanity
- if (actualFrameRate < 1.0)
- {
- actualFrameRate = 1.0;
- }
-
- // Update frame rate for the loss protection logic class: frame rate should
- // be the actual/sent rate
- _lossProtLogic->UpdateFrameRate(actualFrameRate);
-
- _fractionLost = fractionLost;
-
- // Returns the filtered packet loss, used for the protection setting.
- // The filtered loss may be the received loss (no filter), or some
- // filtered value (average or max window filter).
- // Use max window filter for now.
- FilterPacketLossMode filter_mode = kMaxFilter;
- WebRtc_UWord8 packetLossEnc = _lossProtLogic->FilteredLoss(
- _clock->MillisecondTimestamp(), filter_mode, fractionLost);
-
- // For now use the filtered loss for computing the robustness settings
- _lossProtLogic->UpdateFilteredLossPr(packetLossEnc);
-
- // Rate cost of the protection methods
- uint32_t protection_overhead_kbps = 0;
-
- // Update protection settings, when applicable
- float sent_video_rate = 0.0f;
- if (selectedMethod)
- {
- // Update protection method with content metrics
- selectedMethod->UpdateContentMetrics(_content->ShortTermAvgData());
-
- // Update method will compute the robustness settings for the given
- // protection method and the overhead cost
- // the protection method is set by the user via SetVideoProtection.
- _lossProtLogic->UpdateMethod();
-
- // Update protection callback with protection settings.
- uint32_t sent_video_rate_bps = 0;
- uint32_t sent_nack_rate_bps = 0;
- uint32_t sent_fec_rate_bps = 0;
- // Get the bit cost of protection method, based on the amount of
- // overhead data actually transmitted (including headers) the last
- // second.
- UpdateProtectionCallback(selectedMethod,
- &sent_video_rate_bps,
- &sent_nack_rate_bps,
- &sent_fec_rate_bps);
- uint32_t sent_total_rate_bps = sent_video_rate_bps +
- sent_nack_rate_bps + sent_fec_rate_bps;
- // Estimate the overhead costs of the next second as staying the same
- // wrt the source bitrate.
- if (sent_total_rate_bps > 0) {
- protection_overhead_kbps = static_cast<uint32_t>(bitRate *
- static_cast<double>(sent_nack_rate_bps + sent_fec_rate_bps) /
- sent_total_rate_bps + 0.5);
- }
- // Cap the overhead estimate to 50%.
- if (protection_overhead_kbps > bitRate / 2)
- protection_overhead_kbps = bitRate / 2;
-
- // Get the effective packet loss for encoder ER
- // when applicable, should be passed to encoder via fractionLost
- packetLossEnc = selectedMethod->RequiredPacketLossER();
- sent_video_rate = static_cast<float>(sent_video_rate_bps / 1000.0);
- }
-
- // Source coding rate: total rate - protection overhead
- _targetBitRate = bitRate - protection_overhead_kbps;
-
- // Update encoding rates following protection settings
- _frameDropper->SetRates(static_cast<float>(_targetBitRate), 0);
-
- if (_enableQm && _numLayers == 1)
- {
- // Update QM with rates
- _qmResolution->UpdateRates((float)_targetBitRate, sent_video_rate,
- _incomingFrameRate, _fractionLost);
- // Check for QM selection
- bool selectQM = checkStatusForQMchange();
- if (selectQM)
- {
- SelectQuality();
- }
- // Reset the short-term averaged content data.
- _content->ResetShortTermAvgData();
- }
-
- return _targetBitRate;
-}
-
-int VCMMediaOptimization::UpdateProtectionCallback(
- VCMProtectionMethod *selected_method,
- uint32_t* video_rate_bps,
- uint32_t* nack_overhead_rate_bps,
- uint32_t* fec_overhead_rate_bps)
-{
- if (!_videoProtectionCallback)
- {
- return VCM_OK;
- }
- // Get the FEC code rate for Key frames (set to 0 when NA)
- const WebRtc_UWord8
- codeRateKeyRTP = selected_method->RequiredProtectionFactorK();
-
- // Get the FEC code rate for Delta frames (set to 0 when NA)
- const WebRtc_UWord8
- codeRateDeltaRTP = selected_method->RequiredProtectionFactorD();
-
- // Get the FEC-UEP protection status for Key frames: UEP on/off
- const bool
- useUepProtectionKeyRTP = selected_method->RequiredUepProtectionK();
-
- // Get the FEC-UEP protection status for Delta frames: UEP on/off
- const bool
- useUepProtectionDeltaRTP = selected_method->RequiredUepProtectionD();
-
- // NACK is on for NACK and NackFec protection method: off for FEC method
- bool nackStatus = (selected_method->Type() == kNackFec ||
- selected_method->Type() == kNack);
-
- // TODO(Marco): Pass FEC protection values per layer.
-
- return _videoProtectionCallback->ProtectionRequest(codeRateDeltaRTP,
- codeRateKeyRTP,
- useUepProtectionDeltaRTP,
- useUepProtectionKeyRTP,
- nackStatus,
- video_rate_bps,
- nack_overhead_rate_bps,
- fec_overhead_rate_bps);
-}
-
-bool
-VCMMediaOptimization::DropFrame()
-{
- // leak appropriate number of bytes
- _frameDropper->Leak((WebRtc_UWord32)(InputFrameRate() + 0.5f));
- return _frameDropper->DropFrame();
-}
-
-WebRtc_Word32
-VCMMediaOptimization::SentFrameCount(VCMFrameCount &frameCount) const
-{
- frameCount.numDeltaFrames = _deltaFrameCnt;
- frameCount.numKeyFrames = _keyFrameCnt;
- return VCM_OK;
-}
-
-WebRtc_Word32
-VCMMediaOptimization::SetEncodingData(VideoCodecType sendCodecType,
- WebRtc_Word32 maxBitRate,
- WebRtc_UWord32 frameRate,
- WebRtc_UWord32 bitRate,
- WebRtc_UWord16 width,
- WebRtc_UWord16 height,
- int numLayers)
-{
- // Everything codec specific should be reset here since this means the codec
- // has changed. If native dimension values have changed, then either user
- // initiated change, or QM initiated change. Will be able to determine only
- // after the processing of the first frame.
- _lastChangeTime = _clock->MillisecondTimestamp();
- _content->Reset();
- _content->UpdateFrameRate(frameRate);
-
- _maxBitRate = maxBitRate;
- _sendCodecType = sendCodecType;
- _targetBitRate = bitRate;
- _lossProtLogic->UpdateBitRate(static_cast<float>(bitRate));
- _lossProtLogic->UpdateFrameRate(static_cast<float>(frameRate));
- _lossProtLogic->UpdateFrameSize(width, height);
- _lossProtLogic->UpdateNumLayers(numLayers);
- _frameDropper->Reset();
- _frameDropper->SetRates(static_cast<float>(bitRate),
- static_cast<float>(frameRate));
- _userFrameRate = static_cast<float>(frameRate);
- _codecWidth = width;
- _codecHeight = height;
- _initCodecWidth = width;
- _initCodecHeight = height;
- _numLayers = (numLayers <= 1) ? 1 : numLayers; // Can also be zero.
- WebRtc_Word32 ret = VCM_OK;
- ret = _qmResolution->Initialize((float)_targetBitRate, _userFrameRate,
- _codecWidth, _codecHeight);
- return ret;
-}
-
-WebRtc_Word32
-VCMMediaOptimization::RegisterProtectionCallback(VCMProtectionCallback*
- protectionCallback)
-{
- _videoProtectionCallback = protectionCallback;
- return VCM_OK;
-
-}
-
-void
-VCMMediaOptimization::EnableFrameDropper(bool enable)
-{
- _frameDropper->Enable(enable);
-}
-
-void
-VCMMediaOptimization::EnableProtectionMethod(bool enable,
- VCMProtectionMethodEnum method)
-{
- bool updated = false;
- if (enable)
- {
- updated = _lossProtLogic->SetMethod(method);
- }
- else
- {
- _lossProtLogic->RemoveMethod(method);
- }
- if (updated)
- {
- _lossProtLogic->UpdateMethod();
- }
-}
-
-bool
-VCMMediaOptimization::IsProtectionMethodEnabled(VCMProtectionMethodEnum method)
-{
- return (_lossProtLogic->SelectedType() == method);
-}
-
-void
-VCMMediaOptimization::SetMtu(WebRtc_Word32 mtu)
-{
- _maxPayloadSize = mtu;
-}
-
-float
-VCMMediaOptimization::SentFrameRate()
-{
- if (_frameDropper)
- {
- return _frameDropper->ActualFrameRate((WebRtc_UWord32)(InputFrameRate()
- + 0.5f));
- }
-
- return VCM_CODEC_ERROR;
-}
-
-float
-VCMMediaOptimization::SentBitRate()
-{
- UpdateBitRateEstimate(-1, _clock->MillisecondTimestamp());
- return _avgSentBitRateBps / 1000.0f;
-}
-
-WebRtc_Word32
-VCMMediaOptimization::MaxBitRate()
-{
- return _maxBitRate;
-}
-
-WebRtc_Word32
-VCMMediaOptimization::UpdateWithEncodedData(WebRtc_Word32 encodedLength,
- FrameType encodedFrameType)
-{
- // look into the ViE version - debug mode - needs also number of layers.
- UpdateBitRateEstimate(encodedLength, _clock->MillisecondTimestamp());
- if(encodedLength > 0)
- {
- const bool deltaFrame = (encodedFrameType != kVideoFrameKey &&
- encodedFrameType != kVideoFrameGolden);
-
- _frameDropper->Fill(encodedLength, deltaFrame);
- if (_maxPayloadSize > 0 && encodedLength > 0)
- {
- const float minPacketsPerFrame = encodedLength /
- static_cast<float>(_maxPayloadSize);
- if (deltaFrame)
- {
- _lossProtLogic->UpdatePacketsPerFrame(
- minPacketsPerFrame, _clock->MillisecondTimestamp());
- }
- else
- {
- _lossProtLogic->UpdatePacketsPerFrameKey(
- minPacketsPerFrame, _clock->MillisecondTimestamp());
- }
-
- if (_enableQm)
- {
- // update quality select with encoded length
- _qmResolution->UpdateEncodedSize(encodedLength,
- encodedFrameType);
- }
- }
- if (!deltaFrame && encodedLength > 0)
- {
- _lossProtLogic->UpdateKeyFrameSize(static_cast<float>(encodedLength));
- }
-
- // updating counters
- if (deltaFrame)
- {
- _deltaFrameCnt++;
- }
- else
- {
- _keyFrameCnt++;
- }
-
- }
-
- return VCM_OK;
-
-}
-
-void VCMMediaOptimization::UpdateBitRateEstimate(WebRtc_Word64 encodedLength,
- WebRtc_Word64 nowMs)
-{
- int i = kBitrateMaxFrameSamples - 1;
- WebRtc_UWord32 frameSizeSum = 0;
- WebRtc_Word64 timeOldest = -1;
- // Find an empty slot for storing the new sample and at the same time
- // accumulate the history.
- for (; i >= 0; i--)
- {
- if (_encodedFrameSamples[i]._sizeBytes == -1)
- {
- // Found empty slot
- break;
- }
- if (nowMs - _encodedFrameSamples[i]._timeCompleteMs <
- kBitrateAverageWinMs)
- {
- frameSizeSum += static_cast<WebRtc_UWord32>
- (_encodedFrameSamples[i]._sizeBytes);
- if (timeOldest == -1)
- {
- timeOldest = _encodedFrameSamples[i]._timeCompleteMs;
- }
- }
- }
- if (encodedLength > 0)
- {
- if (i < 0)
- {
- // No empty slot, shift
- for (i = kBitrateMaxFrameSamples - 2; i >= 0; i--)
- {
- _encodedFrameSamples[i + 1] = _encodedFrameSamples[i];
- }
- i++;
- }
- // Insert new sample
- _encodedFrameSamples[i]._sizeBytes = encodedLength;
- _encodedFrameSamples[i]._timeCompleteMs = nowMs;
- }
- if (timeOldest > -1)
- {
- // Update average bit rate
- float denom = static_cast<float>(nowMs - timeOldest);
- if (denom < 1.0)
- {
- denom = 1.0;
- }
- _avgSentBitRateBps = (frameSizeSum + encodedLength) * 8 * 1000 / denom;
- }
- else if (encodedLength > 0)
- {
- _avgSentBitRateBps = static_cast<float>(encodedLength * 8);
- }
- else
- {
- _avgSentBitRateBps = 0;
- }
-}
-
-
-WebRtc_Word32
-VCMMediaOptimization::RegisterVideoQMCallback(VCMQMSettingsCallback*
- videoQMSettings)
-{
- _videoQMSettingsCallback = videoQMSettings;
- // Callback setting controls QM
- if (_videoQMSettingsCallback != NULL)
- {
- _enableQm = true;
- }
- else
- {
- _enableQm = false;
- }
- return VCM_OK;
-}
-
-void
-VCMMediaOptimization::updateContentData(const VideoContentMetrics*
- contentMetrics)
-{
- // Updating content metrics
- if (contentMetrics == NULL)
- {
- // Disable QM if metrics are NULL
- _enableQm = false;
- _qmResolution->Reset();
- }
- else
- {
- _content->UpdateContentData(contentMetrics);
- }
-}
-
-WebRtc_Word32
-VCMMediaOptimization::SelectQuality()
-{
- // Reset quantities for QM select
- _qmResolution->ResetQM();
-
- // Update QM will long-term averaged content metrics.
- _qmResolution->UpdateContent(_content->LongTermAvgData());
-
- // Select quality mode
- VCMResolutionScale* qm = NULL;
- WebRtc_Word32 ret = _qmResolution->SelectResolution(&qm);
- if (ret < 0)
- {
- return ret;
- }
-
- // Check for updates to spatial/temporal modes
- QMUpdate(qm);
-
- // Reset all the rate and related frame counters quantities
- _qmResolution->ResetRates();
-
- // Reset counters
- _lastQMUpdateTime = _clock->MillisecondTimestamp();
-
- // Reset content metrics
- _content->Reset();
-
- return VCM_OK;
-}
-
-
-// Check timing constraints and look for significant change in:
-// (1) scene content
-// (2) target bit rate
-
-bool
-VCMMediaOptimization::checkStatusForQMchange()
-{
-
- bool status = true;
-
- // Check that we do not call QMSelect too often, and that we waited some time
- // (to sample the metrics) from the event lastChangeTime
- // lastChangeTime is the time where user changed the size/rate/frame rate
- // (via SetEncodingData)
- WebRtc_Word64 now = _clock->MillisecondTimestamp();
- if ((now - _lastQMUpdateTime) < kQmMinIntervalMs ||
- (now - _lastChangeTime) < kQmMinIntervalMs)
- {
- status = false;
- }
-
- return status;
-
-}
-
-bool
-VCMMediaOptimization::QMUpdate(VCMResolutionScale* qm)
-{
- // Check for no change
- if (qm->spatialHeightFact == 1 &&
- qm->spatialWidthFact == 1 &&
- qm->temporalFact == 1) {
- return false;
- }
-
- // Temporal
- WebRtc_UWord32 frameRate = static_cast<WebRtc_UWord32>
- (_incomingFrameRate + 0.5f);
-
- // Check if go back up in temporal resolution
- if (qm->temporalFact == 0) {
- // Currently only allow for 1/2 frame rate reduction per action.
- // TODO (marpan): allow for 2/3 reduction.
- frameRate = (WebRtc_UWord32) 2 * _incomingFrameRate;
- }
- // go down in temporal resolution
- else {
- frameRate = (WebRtc_UWord32)(_incomingFrameRate / qm->temporalFact + 1);
- }
- // Reset _incomingFrameRate if temporal action was selected.
- if (qm->temporalFact != 1) {
- memset(_incomingFrameTimes, -1, sizeof(_incomingFrameTimes));
- _incomingFrameRate = frameRate;
- }
-
- // Spatial
- WebRtc_UWord32 height = _codecHeight;
- WebRtc_UWord32 width = _codecWidth;
- // Check if go back up in spatial resolution, and update frame sizes.
- // Currently only allow for 2x2 spatial down-sampling.
- // TODO (marpan): allow for 1x2, 2x1, and 4/3x4/3 (or 3/2x3/2).
- if (qm->spatialHeightFact == 0 && qm->spatialWidthFact == 0) {
- width = _codecWidth * 2;
- height = _codecHeight * 2;
- } else {
- width = _codecWidth / qm->spatialWidthFact;
- height = _codecHeight / qm->spatialHeightFact;
- }
- _codecWidth = width;
- _codecHeight = height;
-
- // New frame sizes should never exceed the original sizes
- // from SetEncodingData().
- assert(_codecWidth <= _initCodecWidth);
- assert(_codecHeight <= _initCodecHeight);
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, _id,
- "Quality Mode Update: W = %d, H = %d, FR = %f",
- width, height, frameRate);
-
- // Update VPM with new target frame rate and size
- _videoQMSettingsCallback->SetVideoQMSettings(frameRate, width, height);
-
- _content->UpdateFrameRate(frameRate);
- _qmResolution->UpdateCodecFrameSize(width, height);
-
- return true;
-}
-
-void
-VCMMediaOptimization::UpdateIncomingFrameRate()
-{
- WebRtc_Word64 now = _clock->MillisecondTimestamp();
- if (_incomingFrameTimes[0] == 0)
- {
- // first no shift
- } else
- {
- // shift
- for(WebRtc_Word32 i = (kFrameCountHistorySize - 2); i >= 0 ; i--)
- {
- _incomingFrameTimes[i+1] = _incomingFrameTimes[i];
- }
- }
- _incomingFrameTimes[0] = now;
- ProcessIncomingFrameRate(now);
-}
-
-// allowing VCM to keep track of incoming frame rate
-void
-VCMMediaOptimization::ProcessIncomingFrameRate(WebRtc_Word64 now)
-{
- WebRtc_Word32 num = 0;
- WebRtc_Word32 nrOfFrames = 0;
- for (num = 1; num < (kFrameCountHistorySize - 1); num++)
- {
- if (_incomingFrameTimes[num] <= 0 ||
- // don't use data older than 2 s
- now - _incomingFrameTimes[num] > kFrameHistoryWinMs)
- {
- break;
- } else
- {
- nrOfFrames++;
- }
- }
- if (num > 1)
- {
- const WebRtc_Word64 diff = now - _incomingFrameTimes[num-1];
- _incomingFrameRate = 1.0;
- if(diff >0)
- {
- _incomingFrameRate = nrOfFrames * 1000.0f / static_cast<float>(diff);
- }
- }
-}
-
-WebRtc_UWord32
-VCMMediaOptimization::InputFrameRate()
-{
- ProcessIncomingFrameRate(_clock->MillisecondTimestamp());
- return WebRtc_UWord32 (_incomingFrameRate + 0.5f);
-}
-
-}
diff --git a/trunk/src/modules/video_coding/main/source/media_optimization.h b/trunk/src/modules/video_coding/main/source/media_optimization.h
deleted file mode 100644
index 14e5d1a..0000000
--- a/trunk/src/modules/video_coding/main/source/media_optimization.h
+++ /dev/null
@@ -1,212 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPTIMIZATION_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPTIMIZATION_H_
-
-#include "module_common_types.h"
-#include "video_coding.h"
-#include "trace.h"
-#include "media_opt_util.h"
-#include "qm_select.h"
-
-namespace webrtc
-{
-
-enum { kBitrateMaxFrameSamples = 60 };
-enum { kBitrateAverageWinMs = 1000 };
-
-class TickTimeBase;
-class VCMContentMetricsProcessing;
-class VCMFrameDropper;
-
-struct VCMEncodedFrameSample
-{
- VCMEncodedFrameSample() : _sizeBytes(-1), _timeCompleteMs(-1) {}
-
- WebRtc_Word64 _sizeBytes;
- WebRtc_Word64 _timeCompleteMs;
-};
-
-class VCMMediaOptimization
-{
-public:
- VCMMediaOptimization(WebRtc_Word32 id, TickTimeBase* clock);
- ~VCMMediaOptimization(void);
- /*
- * Reset the Media Optimization module
- */
- WebRtc_Word32 Reset();
- /**
- * Set target Rates for the encoder given the channel parameters
- * Inputs: bitRate - target bitRate, in the conference case this is the rate
- * between the sending client and the server
- * fractionLost - packet loss in % in the network
- * roundTripTimeMs - round trip time in miliseconds
- * minBitRate - the bit rate of the end-point with lowest rate
- * maxBitRate - the bit rate of the end-point with highest rate
- */
- WebRtc_UWord32 SetTargetRates(WebRtc_UWord32 bitRate,
- WebRtc_UWord8 &fractionLost,
- WebRtc_UWord32 roundTripTimeMs);
-
- /**
- * Inform media optimization of initial encoding state
- */
- WebRtc_Word32 SetEncodingData(VideoCodecType sendCodecType,
- WebRtc_Word32 maxBitRate,
- WebRtc_UWord32 frameRate,
- WebRtc_UWord32 bitRate,
- WebRtc_UWord16 width,
- WebRtc_UWord16 height,
- int numTemporalLayers);
- /**
- * Enable protection method
- */
- void EnableProtectionMethod(bool enable, VCMProtectionMethodEnum method);
- /**
- * Returns weather or not protection method is enabled
- */
- bool IsProtectionMethodEnabled(VCMProtectionMethodEnum method);
- /**
- * Updates the max pay load size
- */
- void SetMtu(WebRtc_Word32 mtu);
- /*
- * Get actual input frame rate
- */
- WebRtc_UWord32 InputFrameRate();
-
- /*
- * Get actual sent frame rate
- */
- float SentFrameRate();
- /*
- * Get actual sent bit rate
- */
- float SentBitRate();
- /*
- * Get maximum allowed bit rate
- */
- WebRtc_Word32 MaxBitRate();
- /*
- * Inform Media Optimization of encoding output: Length and frame type
- */
- WebRtc_Word32 UpdateWithEncodedData(WebRtc_Word32 encodedLength,
- FrameType encodedFrameType);
- /*
- * Register a protection callback to be used to inform the user about the
- * protection methods used
- */
- WebRtc_Word32 RegisterProtectionCallback(VCMProtectionCallback*
- protectionCallback);
- /*
- * Register a quality settings callback to be used to inform VPM/user about
- */
- WebRtc_Word32 RegisterVideoQMCallback(VCMQMSettingsCallback* videoQMSettings);
- void EnableFrameDropper(bool enable);
-
- bool DropFrame();
-
- /*
- * Get number of key/delta frames encoded
- */
- WebRtc_Word32 SentFrameCount(VCMFrameCount &frameCount) const;
-
- /*
- * update incoming frame rate value
- */
- void UpdateIncomingFrameRate();
-
- /**
- * Update content metric Data
- */
- void updateContentData(const VideoContentMetrics* contentMetrics);
-
- /**
- * Compute new Quality Mode
- */
- WebRtc_Word32 SelectQuality();
-
-private:
-
- /*
- * Update protection callback with protection settings
- */
- int UpdateProtectionCallback(VCMProtectionMethod *selected_method,
- uint32_t* total_video_rate_bps,
- uint32_t* nack_overhead_rate_bps,
- uint32_t* fec_overhead_rate_bps);
-
- void UpdateBitRateEstimate(WebRtc_Word64 encodedLength, WebRtc_Word64 nowMs);
- /*
- * verify if QM settings differ from default, i.e. if an update is required
- * Compute actual values, as will be sent to the encoder
- */
- bool QMUpdate(VCMResolutionScale* qm);
- /**
- * check if we should make a QM change
- * will return 1 if yes, 0 otherwise
- */
- bool checkStatusForQMchange();
-
- void ProcessIncomingFrameRate(WebRtc_Word64 now);
-
- enum { kFrameCountHistorySize = 90};
- enum { kFrameHistoryWinMs = 2000};
-
- WebRtc_Word32 _id;
- TickTimeBase* _clock;
- WebRtc_Word32 _maxBitRate;
- VideoCodecType _sendCodecType;
- WebRtc_UWord16 _codecWidth;
- WebRtc_UWord16 _codecHeight;
- WebRtc_UWord16 _initCodecWidth;
- WebRtc_UWord16 _initCodecHeight;
- float _userFrameRate;
-
- VCMFrameDropper* _frameDropper;
- VCMLossProtectionLogic* _lossProtLogic;
- WebRtc_UWord8 _packetLossEnc;
- WebRtc_UWord8 _fractionLost;
-
-
- WebRtc_UWord32 _sendStatistics[4];
- WebRtc_UWord32 _sendStatisticsZeroEncode;
- WebRtc_Word32 _maxPayloadSize;
- WebRtc_UWord32 _targetBitRate;
-
- float _incomingFrameRate;
- WebRtc_Word64 _incomingFrameTimes[kFrameCountHistorySize];
-
- bool _enableQm;
-
- VCMProtectionCallback* _videoProtectionCallback;
- VCMQMSettingsCallback* _videoQMSettingsCallback;
-
- VCMEncodedFrameSample _encodedFrameSamples[kBitrateMaxFrameSamples];
- float _avgSentBitRateBps;
-
- WebRtc_UWord32 _keyFrameCnt;
- WebRtc_UWord32 _deltaFrameCnt;
-
- VCMContentMetricsProcessing* _content;
- VCMQmResolution* _qmResolution;
-
- WebRtc_Word64 _lastQMUpdateTime;
- WebRtc_Word64 _lastChangeTime; // content/user triggered
- int _numLayers;
-
-
-}; // end of VCMMediaOptimization class definition
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_MEDIA_OPTIMIZATION_H_
diff --git a/trunk/src/modules/video_coding/main/source/mock/fake_tick_time.h b/trunk/src/modules/video_coding/main/source/mock/fake_tick_time.h
deleted file mode 100644
index c6da348..0000000
--- a/trunk/src/modules/video_coding/main/source/mock/fake_tick_time.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_MOCK_FAKE_TICK_TIME_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_MOCK_FAKE_TICK_TIME_H_
-
-#include <assert.h>
-
-#include <limits>
-
-#include "modules/video_coding/main/source/tick_time_base.h"
-
-namespace webrtc {
-
-// Provides a fake implementation of TickTimeBase, intended for offline
-// testing. This implementation does not query the system clock, but returns a
-// time value set by the user when creating the object, and incremented with
-// the method IncrementDebugClock.
-class FakeTickTime : public TickTimeBase {
- public:
- explicit FakeTickTime(int64_t start_time_ms) : fake_now_ms_(start_time_ms) {}
- virtual ~FakeTickTime() {}
- virtual int64_t MillisecondTimestamp() const {
- return fake_now_ms_;
- }
- virtual int64_t MicrosecondTimestamp() const {
- return 1000 * fake_now_ms_;
- }
- virtual void IncrementDebugClock(int64_t increase_ms) {
- assert(increase_ms <= std::numeric_limits<int64_t>::max() - fake_now_ms_);
- fake_now_ms_ += increase_ms;
- }
-
- private:
- int64_t fake_now_ms_;
-};
-
-} // namespace
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_MOCK_FAKE_TICK_TIME_H_
diff --git a/trunk/src/modules/video_coding/main/source/nack_fec_tables.h b/trunk/src/modules/video_coding/main/source/nack_fec_tables.h
deleted file mode 100644
index 88e225d..0000000
--- a/trunk/src/modules/video_coding/main/source/nack_fec_tables.h
+++ /dev/null
@@ -1,127 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_NACK_FEC_TABLES_H_
-#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_NACK_FEC_TABLES_H_
-
-namespace webrtc
-{
-
-// Table for adjusting FEC rate for NACK/FEC protection method
-// Table values are built as a sigmoid function, ranging from 0 to
-// kHighRttNackMs (100), based on the HybridNackTH values defined in
-// media_opt_util.h.
-const WebRtc_UWord16 VCMNackFecTable[100] = {
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-0,
-1,
-1,
-1,
-1,
-1,
-2,
-2,
-2,
-3,
-3,
-4,
-5,
-6,
-7,
-9,
-10,
-12,
-15,
-18,
-21,
-24,
-28,
-32,
-37,
-41,
-46,
-51,
-56,
-61,
-66,
-70,
-74,
-78,
-81,
-84,
-86,
-89,
-90,
-92,
-93,
-95,
-95,
-96,
-97,
-97,
-98,
-98,
-99,
-99,
-99,
-99,
-99,
-99,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-100,
-
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_NACK_FEC_TABLES_H_
diff --git a/trunk/src/modules/video_coding/main/source/packet.cc b/trunk/src/modules/video_coding/main/source/packet.cc
deleted file mode 100644
index e52cbdd..0000000
--- a/trunk/src/modules/video_coding/main/source/packet.cc
+++ /dev/null
@@ -1,119 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "packet.h"
-#include "module_common_types.h"
-
-#include <assert.h>
-
-namespace webrtc {
-
-VCMPacket::VCMPacket()
- :
- payloadType(0),
- timestamp(0),
- seqNum(0),
- dataPtr(NULL),
- sizeBytes(0),
- markerBit(false),
- frameType(kFrameEmpty),
- codec(kVideoCodecUnknown),
- isFirstPacket(false),
- completeNALU(kNaluUnset),
- insertStartCode(false),
- codecSpecificHeader() {
-}
-
-VCMPacket::VCMPacket(const WebRtc_UWord8* ptr,
- const WebRtc_UWord32 size,
- const WebRtcRTPHeader& rtpHeader) :
- payloadType(rtpHeader.header.payloadType),
- timestamp(rtpHeader.header.timestamp),
- seqNum(rtpHeader.header.sequenceNumber),
- dataPtr(ptr),
- sizeBytes(size),
- markerBit(rtpHeader.header.markerBit),
-
- frameType(rtpHeader.frameType),
- codec(kVideoCodecUnknown),
- isFirstPacket(rtpHeader.type.Video.isFirstPacket),
- completeNALU(kNaluComplete),
- insertStartCode(false),
- codecSpecificHeader(rtpHeader.type.Video)
-{
- CopyCodecSpecifics(rtpHeader.type.Video);
-}
-
-VCMPacket::VCMPacket(const WebRtc_UWord8* ptr, WebRtc_UWord32 size, WebRtc_UWord16 seq, WebRtc_UWord32 ts, bool mBit) :
- payloadType(0),
- timestamp(ts),
- seqNum(seq),
- dataPtr(ptr),
- sizeBytes(size),
- markerBit(mBit),
-
- frameType(kVideoFrameDelta),
- codec(kVideoCodecUnknown),
- isFirstPacket(false),
- completeNALU(kNaluComplete),
- insertStartCode(false),
- codecSpecificHeader()
-{}
-
-void VCMPacket::Reset() {
- payloadType = 0;
- timestamp = 0;
- seqNum = 0;
- dataPtr = NULL;
- sizeBytes = 0;
- markerBit = false;
- frameType = kFrameEmpty;
- codec = kVideoCodecUnknown;
- isFirstPacket = false;
- completeNALU = kNaluUnset;
- insertStartCode = false;
- memset(&codecSpecificHeader, 0, sizeof(RTPVideoHeader));
-}
-
-void VCMPacket::CopyCodecSpecifics(const RTPVideoHeader& videoHeader)
-{
- switch(videoHeader.codec)
- {
- case kRTPVideoVP8:
- {
- // Handle all packets within a frame as depending on the previous packet
- // TODO(holmer): This should be changed to make fragments independent
- // when the VP8 RTP receiver supports fragments.
- if (isFirstPacket && markerBit)
- completeNALU = kNaluComplete;
- else if (isFirstPacket)
- completeNALU = kNaluStart;
- else if (markerBit)
- completeNALU = kNaluEnd;
- else
- completeNALU = kNaluIncomplete;
-
- codec = kVideoCodecVP8;
- break;
- }
- case kRTPVideoI420:
- {
- codec = kVideoCodecI420;
- break;
- }
- default:
- {
- codec = kVideoCodecUnknown;
- break;
- }
- }
-}
-
-}
diff --git a/trunk/src/modules/video_coding/main/source/packet.h b/trunk/src/modules/video_coding/main/source/packet.h
deleted file mode 100644
index 2035653..0000000
--- a/trunk/src/modules/video_coding/main/source/packet.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
-#define WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
-
-#include "typedefs.h"
-#include "module_common_types.h"
-#include "jitter_buffer_common.h"
-
-namespace webrtc
-{
-
-class VCMPacket
-{
-public:
- VCMPacket();
- VCMPacket(const WebRtc_UWord8* ptr,
- const WebRtc_UWord32 size,
- const WebRtcRTPHeader& rtpHeader);
- VCMPacket(const WebRtc_UWord8* ptr,
- WebRtc_UWord32 size,
- WebRtc_UWord16 seqNum,
- WebRtc_UWord32 timestamp,
- bool markerBit);
-
- void Reset();
-
- WebRtc_UWord8 payloadType;
- WebRtc_UWord32 timestamp;
- WebRtc_UWord16 seqNum;
- const WebRtc_UWord8* dataPtr;
- WebRtc_UWord32 sizeBytes;
- bool markerBit;
-
- FrameType frameType;
- webrtc::VideoCodecType codec;
-
- bool isFirstPacket; // Is this first packet in a frame.
- VCMNaluCompleteness completeNALU; // Default is kNaluIncomplete.
- bool insertStartCode; // True if a start code should be inserted before this
- // packet.
- RTPVideoHeader codecSpecificHeader;
-
-protected:
- void CopyCodecSpecifics(const RTPVideoHeader& videoHeader);
-};
-
-} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CODING_PACKET_H_
diff --git a/trunk/src/modules/video_coding/main/source/qm_select.cc b/trunk/src/modules/video_coding/main/source/qm_select.cc
deleted file mode 100644
index c4bd707..0000000
--- a/trunk/src/modules/video_coding/main/source/qm_select.cc
+++ /dev/null
@@ -1,664 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/video_coding/main/source/qm_select.h"
-
-#include <math.h>
-
-#include "modules/interface/module_common_types.h"
-#include "modules/video_coding/main/source/internal_defines.h"
-#include "modules/video_coding/main/source/qm_select_data.h"
-#include "modules/video_coding/main/interface/video_coding_defines.h"
-#include "system_wrappers/interface/trace.h"
-
-namespace webrtc {
-
-// QM-METHOD class
-
-VCMQmMethod::VCMQmMethod()
- : _contentMetrics(NULL),
- _width(0),
- _height(0),
- _nativeWidth(0),
- _nativeHeight(0),
- _frameRateLevel(kDefault),
- _init(false) {
- ResetQM();
-}
-
-VCMQmMethod::~VCMQmMethod() {
-}
-
-void VCMQmMethod::ResetQM() {
- _aspectRatio = 1.0f;
- _imageType = 2;
- _motion.Reset();
- _spatial.Reset();
- _contentClass = 0;
-}
-
-uint8_t VCMQmMethod::ComputeContentClass() {
- ComputeMotionNFD();
- ComputeSpatial();
- return _contentClass = 3 * _motion.level + _spatial.level;
-}
-
-void VCMQmMethod::UpdateContent(const VideoContentMetrics* contentMetrics) {
- _contentMetrics = contentMetrics;
-}
-
-void VCMQmMethod::ComputeMotionNFD() {
- if (_contentMetrics) {
- _motion.value = _contentMetrics->motion_magnitude;
- }
- // Determine motion level.
- if (_motion.value < kLowMotionNfd) {
- _motion.level = kLow;
- } else if (_motion.value > kHighMotionNfd) {
- _motion.level = kHigh;
- } else {
- _motion.level = kDefault;
- }
-}
-
-void VCMQmMethod::ComputeSpatial() {
- float spatialErr = 0.0;
- float spatialErrH = 0.0;
- float spatialErrV = 0.0;
- if (_contentMetrics) {
- spatialErr = _contentMetrics->spatial_pred_err;
- spatialErrH = _contentMetrics->spatial_pred_err_h;
- spatialErrV = _contentMetrics->spatial_pred_err_v;
- }
- // Spatial measure: take average of 3 prediction errors.
- _spatial.value = (spatialErr + spatialErrH + spatialErrV) / 3.0f;
-
- // Reduce thresholds for large scenes/higher pixel correlation (~>=WHD).
- float scale2 = _imageType > 3 ? kScaleTexture : 1.0;
-
- if (_spatial.value > scale2 * kHighTexture) {
- _spatial.level = kHigh;
- } else if (_spatial.value < scale2 * kLowTexture) {
- _spatial.level = kLow;
- } else {
- _spatial.level = kDefault;
- }
-}
-
-uint8_t VCMQmMethod::GetImageType(uint16_t width,
- uint16_t height) {
- // Get the closest image type for encoder frame size.
- uint32_t imageSize = width * height;
- if (imageSize < kFrameSizeTh[0]) {
- return 0; // QCIF
- } else if (imageSize < kFrameSizeTh[1]) {
- return 1; // CIF
- } else if (imageSize < kFrameSizeTh[2]) {
- return 2; // VGA
- } else if (imageSize < kFrameSizeTh[3]) {
- return 3; // 4CIF
- } else if (imageSize < kFrameSizeTh[4]) {
- return 4; // 720,4:3
- } else if (imageSize < kFrameSizeTh[5]) {
- return 5; // WHD
- } else {
- return 6; // HD
- }
-}
-
-LevelClass VCMQmMethod::FrameRateLevel(float avgFrameRate) {
- if (avgFrameRate < kLowFrameRate) {
- return kLow;
- } else if (avgFrameRate > kHighFrameRate) {
- return kHigh;
- } else {
- return kDefault;
- }
-}
-
-// RESOLUTION CLASS
-
-VCMQmResolution::VCMQmResolution()
- : _qm(new VCMResolutionScale()) {
- Reset();
-}
-
-VCMQmResolution::~VCMQmResolution() {
- delete _qm;
-}
-
-void VCMQmResolution::ResetRates() {
- _sumTargetRate = 0.0f;
- _sumIncomingFrameRate = 0.0f;
- _sumRateMM = 0.0f;
- _sumRateMMSgn = 0;
- _sumPacketLoss = 0.0f;
- _frameCnt = 0;
- _frameCntDelta = 0;
- _lowBufferCnt = 0;
- _updateRateCnt = 0;
-}
-
-void VCMQmResolution::ResetDownSamplingState() {
- _stateDecFactorSpatial = 1;
- _stateDecFactorTemp = 1;
-}
-
-void VCMQmResolution::Reset() {
- _targetBitRate = 0.0f;
- _userFrameRate = 0.0f;
- _incomingFrameRate = 0.0f;
- _perFrameBandwidth =0.0f;
- _bufferLevel = 0.0f;
- _avgTargetRate = 0.0f;
- _avgIncomingFrameRate = 0.0f;
- _avgRatioBufferLow = 0.0f;
- _avgRateMisMatch = 0.0f;
- _avgRateMisMatchSgn = 0.0f;
- _avgPacketLoss = 0.0f;
- _encoderState = kStableEncoding;
- ResetRates();
- ResetDownSamplingState();
- ResetQM();
-}
-
-EncoderState VCMQmResolution::GetEncoderState() {
- return _encoderState;
-}
-
-// Initialize state after re-initializing the encoder,
-// i.e., after SetEncodingData() in mediaOpt.
-int VCMQmResolution::Initialize(float bitRate,
- float userFrameRate,
- uint16_t width,
- uint16_t height) {
- if (userFrameRate == 0.0f || width == 0 || height == 0) {
- return VCM_PARAMETER_ERROR;
- }
- Reset();
- _targetBitRate = bitRate;
- _userFrameRate = userFrameRate;
- _incomingFrameRate = userFrameRate;
- UpdateCodecFrameSize(width, height);
- _nativeWidth = width;
- _nativeHeight = height;
- // Initial buffer level.
- _bufferLevel = kInitBufferLevel * _targetBitRate;
- // Per-frame bandwidth.
- _perFrameBandwidth = _targetBitRate / _userFrameRate;
- _init = true;
- return VCM_OK;
-}
-
-void VCMQmResolution::UpdateCodecFrameSize(uint16_t width, uint16_t height) {
- _width = width;
- _height = height;
- // Set the imageType for the encoder width/height.
- _imageType = GetImageType(width, height);
-}
-
-// Update rate data after every encoded frame.
-void VCMQmResolution::UpdateEncodedSize(int encodedSize,
- FrameType encodedFrameType) {
- _frameCnt++;
- // Convert to Kbps.
- float encodedSizeKbits = static_cast<float>((encodedSize * 8.0) / 1000.0);
-
- // Update the buffer level:
- // Note this is not the actual encoder buffer level.
- // |_bufferLevel| is reset to 0 every time SelectResolution is called, and
- // does not account for frame dropping by encoder or VCM.
- _bufferLevel += _perFrameBandwidth - encodedSizeKbits;
- // Counter for occurrences of low buffer level:
- // low/negative values means encoder is likely dropping frames.
- if (_bufferLevel <= kPercBufferThr * kOptBufferLevel * _targetBitRate) {
- _lowBufferCnt++;
- }
-}
-
-// Update various quantities after SetTargetRates in MediaOpt.
-void VCMQmResolution::UpdateRates(float targetBitRate,
- float encoderSentRate,
- float incomingFrameRate,
- uint8_t packetLoss) {
- // Sum the target bitrate and incoming frame rate:
- // these values are the encoder rates (from previous update ~1sec),
- // i.e, before the update for next ~1sec.
- _sumTargetRate += _targetBitRate;
- _sumIncomingFrameRate += _incomingFrameRate;
- _updateRateCnt++;
-
- // Sum the received (from RTCP reports) packet loss rates.
- _sumPacketLoss += static_cast<float>(packetLoss / 255.0);
-
- // Sum the sequence rate mismatch:
- // Mismatch here is based on the difference between the target rate
- // used (in previous ~1sec) and the average actual encoding rate measured
- // at previous ~1sec.
- float diff = _targetBitRate - encoderSentRate;
- if (_targetBitRate > 0.0)
- _sumRateMM += fabs(diff) / _targetBitRate;
- int sgnDiff = diff > 0 ? 1 : (diff < 0 ? -1 : 0);
- // To check for consistent under(+)/over_shooting(-) of target rate.
- _sumRateMMSgn += sgnDiff;
-
- // Update with the current new target and frame rate:
- // these values are ones the encoder will use for the current/next ~1sec
- _targetBitRate = targetBitRate;
- _incomingFrameRate = incomingFrameRate;
-
- // Update the per_frame_bandwidth:
- // this is the per_frame_bw for the current/next ~1sec
- _perFrameBandwidth = 0.0f;
- if (_incomingFrameRate > 0.0f) {
- _perFrameBandwidth = _targetBitRate / _incomingFrameRate;
- }
-}
-
-// Select the resolution factors: frame size and frame rate change (qm scales).
-// Selection is for going down in resolution, or for going back up
-// (if a previous down-sampling action was taken).
-
-// In the current version the following constraints are imposed:
-// 1) we only allow for one action (either down or back up) at a given time.
-// 2) the possible down-sampling actions are: 2x2 spatial and 1/2 temporal.
-// 3) the total amount of down-sampling (spatial and/or temporal) from the
-// initial (native) resolution is limited by various factors.
-
-// TODO(marpan): extend to allow options for: 4/3x4/3, 1x2, 2x1 spatial,
-// and 2/3 temporal (i.e., skip every third frame).
-int VCMQmResolution::SelectResolution(VCMResolutionScale** qm) {
- if (!_init) {
- return VCM_UNINITIALIZED;
- }
- if (_contentMetrics == NULL) {
- Reset();
- *qm = _qm;
- return VCM_OK;
- }
-
- // Default settings: no action.
- _qm->spatialWidthFact = 1;
- _qm->spatialHeightFact = 1;
- _qm->temporalFact = 1;
- *qm = _qm;
-
- // Compute content class for selection.
- _contentClass = ComputeContentClass();
-
- // Compute various rate quantities for selection.
- ComputeRatesForSelection();
-
- // Get the encoder state.
- ComputeEncoderState();
-
- // Check for going back up in resolution, if we have had some down-sampling
- // relative to native state in Initialize (i.e., after SetEncodingData()
- // in mediaOpt.).
- if (_stateDecFactorSpatial > 1 || _stateDecFactorTemp > 1) {
- if (GoingUpResolution()) {
- *qm = _qm;
- return VCM_OK;
- }
- }
-
- // Check for going down in resolution, only if current total amount of
- // down-sampling state is below threshold.
- if (_stateDecFactorTemp * _stateDecFactorSpatial < kMaxDownSample) {
- if (GoingDownResolution()) {
- *qm = _qm;
- return VCM_OK;
- }
- }
- return VCM_OK;
-}
-
-void VCMQmResolution::ComputeRatesForSelection() {
- _avgTargetRate = 0.0f;
- _avgIncomingFrameRate = 0.0f;
- _avgRatioBufferLow = 0.0f;
- _avgRateMisMatch = 0.0f;
- _avgRateMisMatchSgn = 0.0f;
- _avgPacketLoss = 0.0f;
- if (_frameCnt > 0) {
- _avgRatioBufferLow = static_cast<float>(_lowBufferCnt) /
- static_cast<float>(_frameCnt);
- }
- if (_updateRateCnt > 0) {
- _avgRateMisMatch = static_cast<float>(_sumRateMM) /
- static_cast<float>(_updateRateCnt);
- _avgRateMisMatchSgn = static_cast<float>(_sumRateMMSgn) /
- static_cast<float>(_updateRateCnt);
- _avgTargetRate = static_cast<float>(_sumTargetRate) /
- static_cast<float>(_updateRateCnt);
- _avgIncomingFrameRate = static_cast<float>(_sumIncomingFrameRate) /
- static_cast<float>(_updateRateCnt);
- _avgPacketLoss = static_cast<float>(_sumPacketLoss) /
- static_cast<float>(_updateRateCnt);
- }
- // For selection we may want to weight some quantities more heavily
- // with the current (i.e., next ~1sec) rate values.
- float weight = 0.7f;
- _avgTargetRate = weight * _avgTargetRate + (1.0 - weight) * _targetBitRate;
- _avgIncomingFrameRate = weight * _avgIncomingFrameRate +
- (1.0 - weight) * _incomingFrameRate;
- _frameRateLevel = FrameRateLevel(_avgIncomingFrameRate);
-}
-
-void VCMQmResolution::ComputeEncoderState() {
- // Default.
- _encoderState = kStableEncoding;
-
- // Assign stressed state if:
- // 1) occurrences of low buffer levels is high, or
- // 2) rate mis-match is high, and consistent over-shooting by encoder.
- if ((_avgRatioBufferLow > kMaxBufferLow) ||
- ((_avgRateMisMatch > kMaxRateMisMatch) &&
- (_avgRateMisMatchSgn < -kRateOverShoot))) {
- _encoderState = kStressedEncoding;
- }
- // Assign easy state if:
- // 1) rate mis-match is high, and
- // 2) consistent under-shooting by encoder.
- if ((_avgRateMisMatch > kMaxRateMisMatch) &&
- (_avgRateMisMatchSgn > kRateUnderShoot)) {
- _encoderState = kEasyEncoding;
- }
-}
-
-bool VCMQmResolution::GoingUpResolution() {
- // Check if we should go up both spatially and temporally.
- if (_stateDecFactorSpatial > 1 && _stateDecFactorTemp > 1) {
- if (ConditionForGoingUp(2, 2, 2, kTransRateScaleUpSpatialTemp)) {
- _qm->spatialHeightFact = 0;
- _qm->spatialWidthFact = 0;
- _qm->temporalFact = 0;
- UpdateDownsamplingState(kUpResolution);
- return true;
- }
- } else {
- // Check if we should go up either spatially or temporally.
- bool selectedUpS = false;
- bool selectedUpT = false;
- if (_stateDecFactorSpatial > 1) {
- selectedUpS = ConditionForGoingUp(2, 2, 1, kTransRateScaleUpSpatial);
- }
- if (_stateDecFactorTemp > 1) {
- selectedUpT = ConditionForGoingUp(1, 1, 2, kTransRateScaleUpTemp);
- }
- if (selectedUpS && !selectedUpT) {
- _qm->spatialHeightFact = 0;
- _qm->spatialWidthFact = 0;
- UpdateDownsamplingState(kUpResolution);
- return true;
- } else if (!selectedUpS && selectedUpT) {
- _qm->temporalFact = 0;
- UpdateDownsamplingState(kUpResolution);
- return true;
- } else if (selectedUpS && selectedUpT) {
- // TODO(marpan): which one to pick?
- // pickSpatialOrTemporal()
- // For now take spatial over temporal.
- _qm->spatialHeightFact = 0;
- _qm->spatialWidthFact = 0;
- UpdateDownsamplingState(kUpResolution);
- return true;
- }
- }
- return false;
-}
-
-bool VCMQmResolution::ConditionForGoingUp(uint8_t facWidth,
- uint8_t facHeight,
- uint8_t facTemp,
- float scaleFac) {
- float estimatedTransitionRateUp = GetTransitionRate(facWidth, facHeight,
- facTemp, scaleFac);
- // Go back up if:
- // 1) target rate is above threshold and current encoder state is stable, or
- // 2) encoder state is easy (encoder is significantly under-shooting target).
- if (((_avgTargetRate > estimatedTransitionRateUp) &&
- (_encoderState == kStableEncoding)) ||
- (_encoderState == kEasyEncoding)) {
- return true;
- } else {
- return false;
- }
-}
-
-bool VCMQmResolution::GoingDownResolution() {
- float estimatedTransitionRateDown = GetTransitionRate(1, 1, 1, 1.0);
- float maxRate = kFrameRateFac[_frameRateLevel] * kMaxRateQm[_imageType];
-
- // TODO(marpan): Bias down-sampling based on packet loss conditions.
-
- // Resolution reduction if:
- // (1) target rate is below transition rate, or
- // (2) encoder is in stressed state and target rate below a max threshold.
- if ((_avgTargetRate < estimatedTransitionRateDown ) ||
- (_encoderState == kStressedEncoding && _avgTargetRate < maxRate)) {
- // Get the down-sampling action.
- uint8_t spatialFact = kSpatialAction[_contentClass];
- uint8_t tempFact = kTemporalAction[_contentClass];
-
- switch (spatialFact) {
- case 4: {
- _qm->spatialWidthFact = 2;
- _qm->spatialHeightFact = 2;
- break;
- }
- case 2: {
- assert(false); // Currently not used.
- // Select 1x2,2x1, or 4/3x4/3.
- // SelectSpatialDirectionMode((float) estimatedTransitionRateDown);
- break;
- }
- case 1: {
- _qm->spatialWidthFact = 1;
- _qm->spatialHeightFact = 1;
- break;
- }
- default: {
- assert(false);
- }
- }
- switch (tempFact) {
- case 2: {
- _qm->temporalFact = 2;
- break;
- }
- case 1: {
- _qm->temporalFact = 1;
- break;
- }
- default: {
- assert(false);
- }
- }
- // Adjust some cases based on frame rate.
- // TODO(marpan): will be modified when we add 1/2 spatial and 2/3 temporal.
- AdjustAction();
-
- // Sanity checks on down-sampling selection:
- // override the settings for too small image size and/or frame rate.
- // Also check the limit on current down-sampling states.
-
- // No spatial sampling if current frame size is too small (QCIF),
- // or if amount of spatial down-sampling is already too much.
- if ((_width * _height) <= kMinImageSize ||
- _stateDecFactorSpatial >= kMaxSpatialDown) {
- _qm->spatialWidthFact = 1;
- _qm->spatialHeightFact = 1;
- }
- // No frame rate reduction if average frame rate is below some point,
- // or if the amount of temporal down-sampling is already too much.
- if (_avgIncomingFrameRate <= kMinFrameRate ||
- _stateDecFactorTemp >= kMaxTempDown) {
- _qm->temporalFact = 1;
- }
-
- // Update down-sampling state.
- if (_qm->spatialWidthFact != 1 || _qm->spatialHeightFact != 1 ||
- _qm->temporalFact != 1) {
- UpdateDownsamplingState(kDownResolution);
- return true;
- }
- }
- return false;
-}
-
-float VCMQmResolution::GetTransitionRate(uint8_t facWidth,
- uint8_t facHeight,
- uint8_t facTemp,
- float scaleFac) {
- uint8_t imageType = GetImageType(facWidth * _width,
- facHeight * _height);
- LevelClass frameRateLevel = FrameRateLevel(facTemp * _avgIncomingFrameRate);
-
- // The maximum allowed rate below which down-sampling is allowed:
- // Nominal values based on image format (frame size and frame rate).
- float maxRate = kFrameRateFac[frameRateLevel] * kMaxRateQm[imageType];
-
- uint8_t imageClass = imageType > 3 ? 1: 0;
- uint8_t tableIndex = imageClass * 9 + _contentClass;
- // Scale factor for down-sampling transition threshold:
- // factor based on the content class and the image size.
- float scaleTransRate = kScaleTransRateQm[tableIndex];
-
- // Threshold bitrate for resolution action.
- return static_cast<float> (scaleFac * facTemp * _incomingFrameRate *
- scaleTransRate * maxRate / 30);
-}
-
-void VCMQmResolution::UpdateDownsamplingState(ResolutionAction action) {
- // Assumes for now only actions are 1/2 frame rate of 2x2 spatial.
- if (action == kUpResolution) {
- if (_qm->spatialHeightFact == 0 && _qm->spatialWidthFact == 0) {
- _stateDecFactorSpatial = _stateDecFactorSpatial / 4;
- assert(_stateDecFactorSpatial >= 1);
- }
- if (_qm->temporalFact == 0) {
- _stateDecFactorTemp = _stateDecFactorTemp / 2;
- assert(_stateDecFactorTemp >= 1);
- }
- } else if (action == kDownResolution) {
- _stateDecFactorSpatial = _stateDecFactorSpatial * _qm->spatialWidthFact
- * _qm->spatialHeightFact;
- _stateDecFactorTemp = _stateDecFactorTemp * _qm->temporalFact;
- assert(_stateDecFactorSpatial >= 1);
- assert(_stateDecFactorTemp >= 1);
- } else {
- assert(false);
- }
-}
-
-void VCMQmResolution::AdjustAction() {
- if (_spatial.level == kDefault && _motion.level != kHigh &&
- _frameRateLevel == kHigh) {
- _qm->temporalFact = 2;
- _qm->spatialWidthFact = 1;
- _qm->spatialHeightFact = 1;
- }
-}
-
-// TODO(marpan): Update this when we allow for 1/2 spatial down-sampling.
-void VCMQmResolution::SelectSpatialDirectionMode(float transRate) {
- // Default is 1x2 (H)
- // For bit rates well below transitional rate, we select 2x2.
- if (_targetBitRate < transRate * kRateRedSpatial2X2) {
- _qm->spatialWidthFact = 2;
- _qm->spatialHeightFact = 2;
- }
- // Otherwise check prediction errors and aspect ratio.
- float spatialErr = 0.0;
- float spatialErrH = 0.0;
- float spatialErrV = 0.0;
- if (_contentMetrics) {
- spatialErr = _contentMetrics->spatial_pred_err;
- spatialErrH = _contentMetrics->spatial_pred_err_h;
- spatialErrV = _contentMetrics->spatial_pred_err_v;
- }
-
- // Favor 1x2 if aspect_ratio is 16:9.
- if (_aspectRatio >= 16.0f / 9.0f) {
- // Check if 1x2 has lowest prediction error.
- if (spatialErrH < spatialErr && spatialErrH < spatialErrV) {
- _qm->spatialWidthFact = 2;
- _qm->spatialHeightFact = 1;
- }
- }
- // Check for 2x2 selection: favor 2x2 over 1x2 and 2x1.
- if (spatialErr < spatialErrH * (1.0f + kSpatialErr2x2VsHoriz) &&
- spatialErr < spatialErrV * (1.0f + kSpatialErr2X2VsVert)) {
- _qm->spatialWidthFact = 2;
- _qm->spatialHeightFact = 2;
- }
- // Check for 2x1 selection.
- if (spatialErrV < spatialErrH * (1.0f - kSpatialErrVertVsHoriz) &&
- spatialErrV < spatialErr * (1.0f - kSpatialErr2X2VsVert)) {
- _qm->spatialWidthFact = 1;
- _qm->spatialHeightFact = 2;
- }
-}
-
-// ROBUSTNESS CLASS
-
-VCMQmRobustness::VCMQmRobustness() {
- Reset();
-}
-
-VCMQmRobustness::~VCMQmRobustness() {
-}
-
-void VCMQmRobustness::Reset() {
- _prevTotalRate = 0.0f;
- _prevRttTime = 0;
- _prevPacketLoss = 0;
- _prevCodeRateDelta = 0;
- ResetQM();
-}
-
-// Adjust the FEC rate based on the content and the network state
-// (packet loss rate, total rate/bandwidth, round trip time).
-// Note that packetLoss here is the filtered loss value.
-float VCMQmRobustness::AdjustFecFactor(uint8_t codeRateDelta,
- float totalRate,
- float frameRate,
- uint32_t rttTime,
- uint8_t packetLoss) {
- // Default: no adjustment
- float adjustFec = 1.0f;
- if (_contentMetrics == NULL) {
- return adjustFec;
- }
- // Compute class state of the content.
- ComputeMotionNFD();
- ComputeSpatial();
-
- // TODO(marpan): Set FEC adjustment factor.
-
- // Keep track of previous values of network state:
- // adjustment may be also based on pattern of changes in network state.
- _prevTotalRate = totalRate;
- _prevRttTime = rttTime;
- _prevPacketLoss = packetLoss;
- _prevCodeRateDelta = codeRateDelta;
- return adjustFec;
-}
-
-// Set the UEP (unequal-protection across packets) on/off for the FEC.
-bool VCMQmRobustness::SetUepProtection(uint8_t codeRateDelta,
- float totalRate,
- uint8_t packetLoss,
- bool frameType) {
- // Default.
- return false;
-}
-} // end of namespace
diff --git a/trunk/src/modules/video_coding/main/source/qm_select.h b/trunk/src/modules/video_coding/main/source/qm_select.h
deleted file mode 100644
index 1859530..0000000
--- a/trunk/src/modules/video_coding/main/source/qm_select.h
+++ /dev/null
@@ -1,259 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_H_
-#define WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_H_
-
-#include "common_types.h"
-#include "typedefs.h"
-
-/******************************************************/
-/* Quality Modes: Resolution and Robustness settings */
-/******************************************************/
-
-namespace webrtc {
-struct VideoContentMetrics;
-
-struct VCMResolutionScale {
- VCMResolutionScale()
- : spatialWidthFact(1),
- spatialHeightFact(1),
- temporalFact(1) {
- }
- uint8_t spatialWidthFact;
- uint8_t spatialHeightFact;
- uint8_t temporalFact;
-};
-
-enum LevelClass {
- kLow,
- kHigh,
- kDefault
-};
-
-struct VCMContFeature {
- VCMContFeature()
- : value(0.0f),
- level(kDefault) {
- }
- void Reset() {
- value = 0.0f;
- level = kDefault;
- }
- float value;
- LevelClass level;
-};
-
-enum ResolutionAction {
- kDownResolution,
- kUpResolution,
- kNoChangeResolution
-};
-
-enum EncoderState {
- kStableEncoding, // Low rate mis-match, stable buffer levels.
- kStressedEncoding, // Significant over-shooting of target rate,
- // Buffer under-flow, etc.
- kEasyEncoding // Significant under-shooting of target rate.
-};
-
-// QmMethod class: main class for resolution and robustness settings
-
-class VCMQmMethod {
- public:
- VCMQmMethod();
- virtual ~VCMQmMethod();
-
- // Reset values
- void ResetQM();
- virtual void Reset() = 0;
-
- // Compute content class.
- uint8_t ComputeContentClass();
-
- // Update with the content metrics.
- void UpdateContent(const VideoContentMetrics* contentMetrics);
-
- // Compute spatial texture magnitude and level.
- // Spatial texture is a spatial prediction error measure.
- void ComputeSpatial();
-
- // Compute motion magnitude and level for NFD metric.
- // NFD is normalized frame difference (normalized by spatial variance).
- void ComputeMotionNFD();
-
- // Get the imageType (CIF, VGA, HD, etc) for the system width/height.
- uint8_t GetImageType(uint16_t width, uint16_t height);
-
- // Get the frame rate level.
- LevelClass FrameRateLevel(float frame_rate);
-
- protected:
- // Content Data.
- const VideoContentMetrics* _contentMetrics;
-
- // Encoder frame sizes and native frame sizes.
- uint16_t _width;
- uint16_t _height;
- uint16_t _nativeWidth;
- uint16_t _nativeHeight;
- float _aspectRatio;
- // Image type and frame rate leve, for the current encoder resolution.
- uint8_t _imageType;
- LevelClass _frameRateLevel;
- // Content class data.
- VCMContFeature _motion;
- VCMContFeature _spatial;
- uint8_t _contentClass;
- bool _init;
-};
-
-// Resolution settings class
-
-class VCMQmResolution : public VCMQmMethod {
- public:
- VCMQmResolution();
- virtual ~VCMQmResolution();
-
- // Reset all quantities.
- virtual void Reset();
-
- // Reset rate quantities and counters after every SelectResolution() call.
- void ResetRates();
-
- // Reset down-sampling state.
- void ResetDownSamplingState();
-
- // Get the encoder state.
- EncoderState GetEncoderState();
-
- // Initialize after SetEncodingData in media_opt.
- int Initialize(float bitRate, float userFrameRate,
- uint16_t width, uint16_t height);
-
- // Update the encoder frame size.
- void UpdateCodecFrameSize(uint16_t width, uint16_t height);
-
- // Update with actual bit rate (size of the latest encoded frame)
- // and frame type, after every encoded frame.
- void UpdateEncodedSize(int encodedSize,
- FrameType encodedFrameType);
-
- // Update with new target bitrate, actual encoder sent rate, frame_rate,
- // loss rate: every ~1 sec from SetTargetRates in media_opt.
- void UpdateRates(float targetBitRate, float encoderSentRate,
- float incomingFrameRate, uint8_t packetLoss);
-
- // Extract ST (spatio-temporal) resolution action.
- // Inputs: qm: Reference to the quality modes pointer.
- // Output: the spatial and/or temporal scale change.
- int SelectResolution(VCMResolutionScale** qm);
-
- // Compute rates for the selection of down-sampling action.
- void ComputeRatesForSelection();
-
- // Compute the encoder state.
- void ComputeEncoderState();
-
- // Return true if the action is to go back up in resolution.
- bool GoingUpResolution();
-
- // Return true if the action is to go down in resolution.
- bool GoingDownResolution();
-
- // Check the condition for going up in resolution by the scale factors:
- // |facWidth|, |facHeight|, |facTemp|.
- // |scaleFac| is a scale factor for the transition rate.
- bool ConditionForGoingUp(uint8_t facWidth, uint8_t facHeight,
- uint8_t facTemp,
- float scaleFac);
-
- // Get the bitrate threshold for the resolution action.
- // The case |facWidth|=|facHeight|=|facTemp|==1 is for down-sampling action.
- // |scaleFac| is a scale factor for the transition rate.
- float GetTransitionRate(uint8_t facWidth, uint8_t facHeight,
- uint8_t facTemp, float scaleFac);
-
- // Update the downsampling state.
- void UpdateDownsamplingState(ResolutionAction action);
-
- void AdjustAction();
-
- // Select the directional (1x2 or 2x1) spatial down-sampling action.
- void SelectSpatialDirectionMode(float transRate);
-
- private:
- VCMResolutionScale* _qm;
- // Encoder rate control parameters.
- float _targetBitRate;
- float _userFrameRate;
- float _incomingFrameRate;
- float _perFrameBandwidth;
- float _bufferLevel;
-
- // Data accumulated every ~1sec from MediaOpt.
- float _sumTargetRate;
- float _sumIncomingFrameRate;
- float _sumRateMM;
- float _sumRateMMSgn;
- float _sumPacketLoss;
- // Counters.
- uint32_t _frameCnt;
- uint32_t _frameCntDelta;
- uint32_t _updateRateCnt;
- uint32_t _lowBufferCnt;
-
- // Resolution state parameters.
- uint8_t _stateDecFactorSpatial;
- uint8_t _stateDecFactorTemp;
-
- // Quantities used for selection.
- float _avgTargetRate;
- float _avgIncomingFrameRate;
- float _avgRatioBufferLow;
- float _avgRateMisMatch;
- float _avgRateMisMatchSgn;
- float _avgPacketLoss;
- EncoderState _encoderState;
-};
-
-// Robustness settings class.
-
-class VCMQmRobustness : public VCMQmMethod {
- public:
- VCMQmRobustness();
- ~VCMQmRobustness();
-
- virtual void Reset();
-
- // Adjust FEC rate based on content: every ~1 sec from SetTargetRates.
- // Returns an adjustment factor.
- float AdjustFecFactor(uint8_t codeRateDelta,
- float totalRate,
- float frameRate,
- uint32_t rttTime,
- uint8_t packetLoss);
-
- // Set the UEP protection on/off.
- bool SetUepProtection(uint8_t codeRateDelta,
- float totalRate,
- uint8_t packetLoss,
- bool frameType);
-
- private:
- // Previous state of network parameters.
- float _prevTotalRate;
- uint32_t _prevRttTime;
- uint8_t _prevPacketLoss;
- uint8_t _prevCodeRateDelta;
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CODING_QM_SELECT_H_
-
diff --git a/trunk/src/modules/video_coding/main/source/qm_select_data.h b/trunk/src/modules/video_coding/main/source/qm_select_data.h
deleted file mode 100644
index d4af642..0000000
--- a/trunk/src/modules/video_coding/main/source/qm_select_data.h
+++ /dev/null
@@ -1,184 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_SOURCE_QM_SELECT_DATA_H_
-#define WEBRTC_MODULES_VIDEO_CODING_SOURCE_QM_SELECT_DATA_H_
-
-/***************************************************************
-*QMSelectData.h
-* This file includes parameters for content-aware media optimization
-****************************************************************/
-
-#include "typedefs.h"
-
-namespace webrtc {
-//
-// PARAMETERS FOR RESOLUTION ADAPTATION
-//
-
-// Initial level of buffer in secs: should corresponds to wrapper settings.
-const float kInitBufferLevel = 0.5f;
-
-// Optimal level of buffer in secs: should corresponds to wrapper settings.
-const float kOptBufferLevel = 0.6f;
-
-// Threshold of (max) buffer size below which we consider too low (underflow).
-const float kPercBufferThr = 0.10f;
-
-// Threshold on the occurrences of low buffer levels.
-const float kMaxBufferLow = 0.5f;
-
-// Threshold on rate mismatch
-const float kMaxRateMisMatch = 0.5f;
-
-// Threshold on amount of under/over encoder shooting.
-const float kRateOverShoot = 0.75f;
-const float kRateUnderShoot = 0.75f;
-
-// Factor for transitional rate for going back up in resolution.
-const float kTransRateScaleUpSpatial = 1.25f;
-const float kTransRateScaleUpTemp = 1.25f;
-const float kTransRateScaleUpSpatialTemp = 1.25f;
-
-// Threshold on packet loss rate, above which favor resolution reduction.
-const float kPacketLossThr = 0.1f;
-
-// Factor for reducing transitonal bitrate under packet loss.
-const float kPacketLossRateFac = 1.0f;
-
-// Maximum possible transitional rate for down-sampling:
-// (units in kbps), for 30fps.
-const uint16_t kMaxRateQm[7] = {
- 100, // QCIF
- 250, // CIF
- 500, // VGA
- 800, // 4CIF
- 1000, // 720 HD 4:3,
- 1500, // 720 HD 16:9
- 2000 // 1080HD
-};
-
-// Frame rate scale for maximum transition rate.
-const float kFrameRateFac[3] = {
- 0.7f, // L
- 1.0f, // H
- 0.8f // D
-};
-
-// Scale for transitional rate: based on content class
-// motion=L/H/D,spatial==L/H/D: for low, high, middle levels
-const float kScaleTransRateQm[18] = {
- // 4CIF and lower
- 0.50f, // L, L
- 0.50f, // L, H
- 0.50f, // L, D
- 0.50f, // H ,L
- 0.25f, // H, H
- 0.25f, // H, D
- 0.50f, // D, L
- 0.50f, // D, D
- 0.25f, // D, H
-
- // over 4CIF: WHD, HD
- 0.50f, // L, L
- 0.50f, // L, H
- 0.50f, // L, D
- 0.50f, // H ,L
- 0.25f, // H, H
- 0.25f, // H, D
- 0.50f, // D, L
- 0.50f, // D, D
- 0.25f, // D, H
-};
-
-// Action for down-sampling:
-// motion=L/H/D,spatial==L/H/D: for low, high, middle levels
-const uint8_t kSpatialAction[9] = {
- 1, // L, L
- 1, // L, H
- 1, // L, D
- 4, // H ,L
- 1, // H, H
- 4, // H, D
- 4, // D, L
- 1, // D, H
- 1, // D, D
-};
-
-const uint8_t kTemporalAction[9] = {
- 1, // L, L
- 2, // L, H
- 2, // L, D
- 1, // H ,L
- 2, // H, H
- 1, // H, D
- 1, // D, L
- 2, // D, H
- 1, // D, D
-};
-
-// Control the total amount of down-sampling allowed.
-const int kMaxSpatialDown = 16;
-const int kMaxTempDown = 4;
-const int kMaxDownSample = 16;
-
-// Minimum image size for a spatial down-sampling.
-const int kMinImageSize= 176 * 144;
-
-// Minimum frame rate for temporal down-sampling:
-// no frame rate reduction if incomingFrameRate <= MIN_FRAME_RATE
-const int kMinFrameRate = 8;
-
-// Boundaries for the closest standard frame size
-const uint32_t kFrameSizeTh[6] = {
- 63360, // between 176*144 and 352*288
- 204288, // between 352*288 and 640*480
- 356352, // between 640*480 and 704*576
- 548352, // between 704*576 and 960*720
- 806400, // between 960*720 and 1280*720
- 1497600, // between 1280*720 and 1920*1080
-};
-
-//
-// PARAMETERS FOR FEC ADJUSTMENT: TODO (marpan)
-//
-
-//
-// PARAMETETS FOR SETTING LOW/HIGH STATES OF CONTENT METRICS:
-//
-
-// Thresholds for frame rate:
-const int kLowFrameRate = 10;
-const int kHighFrameRate = 25;
-
-// Thresholds for motion: motion level is from NFD
-const float kHighMotionNfd = 0.075f;
-const float kLowMotionNfd = 0.04f;
-
-// Thresholds for spatial prediction error:
-// this is applied on the min(2x2,1x2,2x1)
-const float kHighTexture = 0.035f;
-const float kLowTexture = 0.025f;
-
-// Used to reduce thresholds for larger/HD scenes: correction factor since
-// higher correlation in HD scenes means lower spatial prediction error.
-const float kScaleTexture = 0.9f;
-
-// percentage reduction in transitional bitrate for 2x2 selected over 1x2/2x1
-const float kRateRedSpatial2X2 = 0.6f;
-
-const float kSpatialErr2x2VsHoriz = 0.1f; // percentage to favor 2x2 over H
-const float kSpatialErr2X2VsVert = 0.1f; // percentage to favor 2x2 over V
-const float kSpatialErrVertVsHoriz = 0.1f; // percentage to favor H over V
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_SOURCE_QM_SELECT_DATA_H_
-
diff --git a/trunk/src/modules/video_coding/main/source/qm_select_unittest.cc b/trunk/src/modules/video_coding/main/source/qm_select_unittest.cc
deleted file mode 100644
index e0ab7bf..0000000
--- a/trunk/src/modules/video_coding/main/source/qm_select_unittest.cc
+++ /dev/null
@@ -1,834 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file includes unit tests the QmResolution class
- * In particular, for the selection of spatial and/or temporal down-sampling.
- */
-
-#include <gtest/gtest.h>
-
-#include "modules/video_coding/main/source/qm_select.h"
-#include "modules/interface/module_common_types.h"
-
-namespace webrtc {
-
-class QmSelectTest : public ::testing::Test {
- protected:
- QmSelectTest()
- : qm_resolution_(new VCMQmResolution()),
- content_metrics_(new VideoContentMetrics()),
- qm_scale_(NULL) {
- }
- VCMQmResolution* qm_resolution_;
- VideoContentMetrics* content_metrics_;
- VCMResolutionScale* qm_scale_;
-
- void InitQmNativeData(float initial_bit_rate, int user_frame_rate,
- int native_width, int native_height);
-
- void UpdateQmEncodedFrame(int* encoded_size, int num_updates);
-
- void UpdateQmRateData(int* target_rate,
- int* encoder_sent_rate,
- int* incoming_frame_rate,
- uint8_t* fraction_lost,
- int num_updates);
-
- void UpdateQmContentData(float motion_metric,
- float spatial_metric,
- float spatial_metric_horiz,
- float spatial_metric_vert);
-
- bool IsSelectedActionCorrect(VCMResolutionScale* qm_scale,
- uint8_t fac_width,
- uint8_t fac_height,
- uint8_t fac_temp);
-
- void TearDown() {
- delete qm_resolution_;
- delete content_metrics_;
- }
-};
-
-TEST_F(QmSelectTest, HandleInputs) {
- // Expect parameter error. Initialize with invalid inputs.
- EXPECT_EQ(-4, qm_resolution_->Initialize(1000, 0, 640, 480));
- EXPECT_EQ(-4, qm_resolution_->Initialize(1000, 30, 640, 0));
- EXPECT_EQ(-4, qm_resolution_->Initialize(1000, 30, 0, 480));
-
- // Expect uninitialized error.: No valid initialization before selection.
- EXPECT_EQ(-7, qm_resolution_->SelectResolution(&qm_scale_));
-
- VideoContentMetrics* content_metrics = NULL;
- EXPECT_EQ(0, qm_resolution_->Initialize(1000, 30, 640, 480));
- qm_resolution_->UpdateContent(content_metrics);
- // Content metrics are NULL: Expect success and no down-sampling action.
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
-}
-
-// No down-sampling action at high rates.
-TEST_F(QmSelectTest, NoActionHighRate) {
- // Initialize with bitrate, frame rate, and native system width/height.
- InitQmNativeData(800, 30, 640, 480);
-
- // Update with encoder frame size.
- uint16_t codec_width = 640;
- uint16_t codec_height = 480;
- qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
- EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
-
- // Update rates for a sequence of intervals.
- int target_rate[] = {800, 800, 800};
- int encoder_sent_rate[] = {800, 800, 800};
- int incoming_frame_rate[] = {30, 30, 30};
- uint8_t fraction_lost[] = {10, 10, 10};
- UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
- fraction_lost, 3);
-
- // Update content: motion level, and 3 spatial prediction errors.
- UpdateQmContentData(0.01f, 0.01f, 0.01f, 0.01f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(0, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
-}
-
-// Rate is well below transition, down-sampling action is taken,
-// depending on the content state.
-TEST_F(QmSelectTest, DownActionLowRate) {
- // Initialize with bitrate, frame rate, and native system width/height.
- InitQmNativeData(100, 30, 640, 480);
-
- // Update with encoder frame size.
- uint16_t codec_width = 640;
- uint16_t codec_height = 480;
- qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
- EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
-
- // Update rates for a sequence of intervals.
- int target_rate[] = {100, 100, 100};
- int encoder_sent_rate[] = {100, 100, 100};
- int incoming_frame_rate[] = {30, 30, 30};
- uint8_t fraction_lost[] = {10, 10, 10};
- UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
- fraction_lost, 3);
-
- // Update content: motion level, and 3 spatial prediction errors.
- // High motion, low spatial: 2x2 spatial expected.
- UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
-
- qm_resolution_->ResetDownSamplingState();
- // Low motion, low spatial: no action expected: content is too low.
- UpdateQmContentData(0.01f, 0.01f, 0.01f, 0.01f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(0, qm_resolution_->ComputeContentClass());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
-
- qm_resolution_->ResetDownSamplingState();
- // Medium motion, low spatial: 2x2 spatial expected.
- UpdateQmContentData(0.06f, 0.01f, 0.01f, 0.01f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(6, qm_resolution_->ComputeContentClass());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
-
- qm_resolution_->ResetDownSamplingState();
- // High motion, high spatial: 1/2 temporal expected.
- UpdateQmContentData(0.1f, 0.1f, 0.1f, 0.1f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(4, qm_resolution_->ComputeContentClass());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
-
- qm_resolution_->ResetDownSamplingState();
- // Low motion, high spatial: 1/2 temporal expected.
- UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
-
- qm_resolution_->ResetDownSamplingState();
- // Medium motion, high spatial: 1/2 temporal expected.
- UpdateQmContentData(0.06f, 0.1f, 0.1f, 0.1f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(7, qm_resolution_->ComputeContentClass());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
-
- qm_resolution_->ResetDownSamplingState();
- // High motion, medium spatial: 2x2 spatial expected.
- UpdateQmContentData(0.1f, 0.03f, 0.03f, 0.03f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(5, qm_resolution_->ComputeContentClass());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
-
- qm_resolution_->ResetDownSamplingState();
- // Low motion, medium spatial: high frame rate, so 1/2 temporal expected.
- UpdateQmContentData(0.01f, 0.03f, 0.03f, 0.03f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(2, qm_resolution_->ComputeContentClass());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
-
- qm_resolution_->ResetDownSamplingState();
- // Medium motion, medium spatial: high frame rate, so 1/2 temporal expected.
- UpdateQmContentData(0.06f, 0.03f, 0.03f, 0.03f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(8, qm_resolution_->ComputeContentClass());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
-}
-
-// Rate mis-match is high, and we have over-shooting.
-// since target rate is below max for down-sampling, down-sampling is selected.
-TEST_F(QmSelectTest, DownActionHighRateMMOvershoot) {
- // Initialize with bitrate, frame rate, and native system width/height.
- InitQmNativeData(450, 30, 640, 480);
-
- // Update with encoder frame size.
- uint16_t codec_width = 640;
- uint16_t codec_height = 480;
- qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
- EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
-
- // Update rates for a sequence of intervals.
- int target_rate[] = {450, 450, 450};
- int encoder_sent_rate[] = {900, 900, 900};
- int incoming_frame_rate[] = {30, 30, 30};
- uint8_t fraction_lost[] = {10, 10, 10};
- UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
- fraction_lost, 3);
-
- // Update content: motion level, and 3 spatial prediction errors.
- // High motion, low spatial.
- UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
-
- qm_resolution_->ResetDownSamplingState();
- // Low motion, high spatial
- UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
-}
-
-// Rate mis-match is high, target rate is below max for down-sampling,
-// but since we have consistent under-shooting, no down-sampling action.
-TEST_F(QmSelectTest, NoActionHighRateMMUndershoot) {
- // Initialize with bitrate, frame rate, and native system width/height.
- InitQmNativeData(450, 30, 640, 480);
-
- // Update with encoder frame size.
- uint16_t codec_width = 640;
- uint16_t codec_height = 480;
- qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
- EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
-
- // Update rates for a sequence of intervals.
- int target_rate[] = {450, 450, 450};
- int encoder_sent_rate[] = {100, 100, 100};
- int incoming_frame_rate[] = {30, 30, 30};
- uint8_t fraction_lost[] = {10, 10, 10};
- UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
- fraction_lost, 3);
-
- // Update content: motion level, and 3 spatial prediction errors.
- // High motion, low spatial.
- UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kEasyEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
-
- qm_resolution_->ResetDownSamplingState();
- // Low motion, high spatial
- UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
-}
-
-// Buffer is underflowing, and target rate is below max for down-sampling,
-// so action is taken.
-TEST_F(QmSelectTest, DownActionBufferUnderflow) {
- // Initialize with bitrate, frame rate, and native system width/height.
- InitQmNativeData(450, 30, 640, 480);
-
- // Update with encoder frame size.
- uint16_t codec_width = 640;
- uint16_t codec_height = 480;
- qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
- EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
-
- // Update with encoded size over a number of frames.
- // per-frame bandwidth = 15 = 450/30: simulate (decoder) buffer underflow:
- int encoded_size[] = {200, 100, 50, 30, 60, 40, 20, 30, 20, 40};
- UpdateQmEncodedFrame(encoded_size, 10);
-
- // Update rates for a sequence of intervals.
- int target_rate[] = {450, 450, 450};
- int encoder_sent_rate[] = {450, 450, 450};
- int incoming_frame_rate[] = {30, 30, 30};
- uint8_t fraction_lost[] = {10, 10, 10};
- UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
- fraction_lost, 3);
-
- // Update content: motion level, and 3 spatial prediction errors.
- // High motion, low spatial.
- UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
-
- qm_resolution_->ResetDownSamplingState();
- // Low motion, high spatial
- UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
-}
-
-// Target rate is below max for down-sampling, but buffer level is stable,
-// so no action is taken.
-TEST_F(QmSelectTest, NoActionBufferStable) {
- // Initialize with bitrate, frame rate, and native system width/height.
- InitQmNativeData(450, 30, 640, 480);
-
- // Update with encoder frame size.
- uint16_t codec_width = 640;
- uint16_t codec_height = 480;
- qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
- EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
-
- // Update with encoded size over a number of frames.
- // per-frame bandwidth = 15 = 450/30: simulate stable (decoder) buffer levels.
- int32_t encoded_size[] = {40, 10, 10, 16, 18, 20, 17, 20, 16, 15};
- UpdateQmEncodedFrame(encoded_size, 10);
-
- // Update rates for a sequence of intervals.
- int target_rate[] = {450, 450, 450};
- int encoder_sent_rate[] = {450, 450, 450};
- int incoming_frame_rate[] = {30, 30, 30};
- uint8_t fraction_lost[] = {10, 10, 10};
- UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
- fraction_lost, 3);
-
- // Update content: motion level, and 3 spatial prediction errors.
- // High motion, low spatial.
- UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
-
- qm_resolution_->ResetDownSamplingState();
- // Low motion, high spatial
- UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
-}
-
-// Very low rate, but no spatial down-sampling below some size (QCIF).
-TEST_F(QmSelectTest, LimitDownSpatialAction) {
- // Initialize with bitrate, frame rate, and native system width/height.
- InitQmNativeData(10, 30, 176, 144);
-
- // Update with encoder frame size.
- uint16_t codec_width = 176;
- uint16_t codec_height = 144;
- qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
- EXPECT_EQ(0, qm_resolution_->GetImageType(codec_width, codec_height));
-
- // Update rates for a sequence of intervals.
- int target_rate[] = {10, 10, 10};
- int encoder_sent_rate[] = {10, 10, 10};
- int incoming_frame_rate[] = {30, 30, 30};
- uint8_t fraction_lost[] = {10, 10, 10};
- UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
- fraction_lost, 3);
-
- // Update content: motion level, and 3 spatial prediction errors.
- // High motion, low spatial.
- UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
-}
-
-// Very low rate, but no frame reduction below some frame_rate (8fps).
-TEST_F(QmSelectTest, LimitDownTemporalAction) {
- // Initialize with bitrate, frame rate, and native system width/height.
- InitQmNativeData(10, 8, 640, 480);
-
- // Update with encoder frame size.
- uint16_t codec_width = 640;
- uint16_t codec_height = 480;
- qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
- EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
-
- // Update rates for a sequence of intervals.
- int target_rate[] = {10, 10, 10};
- int encoder_sent_rate[] = {10, 10, 10};
- int incoming_frame_rate[] = {8, 8, 8};
- uint8_t fraction_lost[] = {10, 10, 10};
- UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
- fraction_lost, 3);
-
- // Update content: motion level, and 3 spatial prediction errors.
- // Low motion, medium spatial.
- UpdateQmContentData(0.01f, 0.03f, 0.03f, 0.03f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(2, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
-}
-
-// Two stages: spatial down-sample and then back up spatially,
-// as rate as increased.
-TEST_F(QmSelectTest, 2StageDownSpatialUpSpatial) {
- // Initialize with bitrate, frame rate, and native system width/height.
- InitQmNativeData(100, 30, 640, 480);
-
- // Update with encoder frame size.
- uint16_t codec_width = 640;
- uint16_t codec_height = 480;
- qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
- EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
-
- // Update rates for a sequence of intervals.
- int target_rate[] = {100, 100, 100};
- int encoder_sent_rate[] = {100, 100, 100};
- int incoming_frame_rate[] = {30, 30, 30};
- uint8_t fraction_lost[] = {10, 10, 10};
- UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
- fraction_lost, 3);
-
- // Update content: motion level, and 3 spatial prediction errors.
- // High motion, low spatial.
- UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
-
- // Reset and go up in rate: expected to go back up.
- qm_resolution_->ResetRates();
- qm_resolution_->UpdateCodecFrameSize(320, 240);
- EXPECT_EQ(1, qm_resolution_->GetImageType(320, 240));
- // Update rates for a sequence of intervals.
- int target_rate2[] = {400, 400, 400, 400, 400};
- int encoder_sent_rate2[] = {400, 400, 400, 400, 400};
- int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
- uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
- UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
- fraction_lost2, 5);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 0, 0, 1));
-}
-
-// Two stages: spatial down-sample and then back up spatially, since encoder
-// is under-shooting target even though rate has not increased much.
-TEST_F(QmSelectTest, 2StageDownSpatialUpSpatialUndershoot) {
- // Initialize with bitrate, frame rate, and native system width/height.
- InitQmNativeData(100, 30, 640, 480);
-
- // Update with encoder frame size.
- uint16_t codec_width = 640;
- uint16_t codec_height = 480;
- qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
- EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
-
- // Update rates for a sequence of intervals.
- int target_rate[] = {100, 100, 100};
- int encoder_sent_rate[] = {100, 100, 100};
- int incoming_frame_rate[] = {30, 30, 30};
- uint8_t fraction_lost[] = {10, 10, 10};
- UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
- fraction_lost, 3);
-
- // Update content: motion level, and 3 spatial prediction errors.
- // High motion, low spatial.
- UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
-
- // Reset rates and simulate under-shooting scenario.: expect to go back up.
- qm_resolution_->ResetRates();
- qm_resolution_->UpdateCodecFrameSize(320, 240);
- EXPECT_EQ(1, qm_resolution_->GetImageType(320, 240));
- // Update rates for a sequence of intervals.
- int target_rate2[] = {200, 200, 200, 200, 200};
- int encoder_sent_rate2[] = {50, 50, 50, 50, 50};
- int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
- uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
- UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
- fraction_lost2, 5);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(kEasyEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 0, 0, 1));
-}
-
-// Two stages: spatial down-sample and then no action to go up,
-// as encoding rate mis-match is too high.
-TEST_F(QmSelectTest, 2StageDownSpatialNoActionUp) {
- // Initialize with bitrate, frame rate, and native system width/height.
- InitQmNativeData(100, 30, 640, 480);
-
- // Update with encoder frame size.
- uint16_t codec_width = 640;
- uint16_t codec_height = 480;
- qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
- EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
-
- // Update rates for a sequence of intervals.
- int target_rate[] = {100, 100, 100};
- int encoder_sent_rate[] = {100, 100, 100};
- int incoming_frame_rate[] = {30, 30, 30};
- uint8_t fraction_lost[] = {10, 10, 10};
- UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
- fraction_lost, 3);
-
- // Update content: motion level, and 3 spatial prediction errors.
- // High motion, low spatial.
- UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
-
- // Reset and simulate large rate mis-match: expect no action to go back up.
- qm_resolution_->ResetRates();
- qm_resolution_->UpdateCodecFrameSize(320, 240);
- EXPECT_EQ(1, qm_resolution_->GetImageType(320, 240));
- // Update rates for a sequence of intervals.
- int target_rate2[] = {400, 400, 400, 400, 400};
- int encoder_sent_rate2[] = {1000, 1000, 1000, 1000, 1000};
- int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
- uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
- UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
- fraction_lost2, 5);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
-}
-// Two stages: temporally down-sample and then back up temporally,
-// as rate as increased.
-TEST_F(QmSelectTest, 2StatgeDownTemporalUpTemporal) {
- // Initialize with bitrate, frame rate, and native system width/height.
- InitQmNativeData(100, 30, 640, 480);
-
- // Update with encoder frame size.
- uint16_t codec_width = 640;
- uint16_t codec_height = 480;
- qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
- EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
-
- // Update rates for a sequence of intervals.
- int target_rate[] = {100, 100, 100};
- int encoder_sent_rate[] = {100, 100, 100};
- int incoming_frame_rate[] = {30, 30, 30};
- uint8_t fraction_lost[] = {10, 10, 10};
- UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
- fraction_lost, 3);
-
- // Update content: motion level, and 3 spatial prediction errors.
- // Low motion, high spatial.
- UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
-
- // Reset rates and go up in rate: expect to go back up.
- qm_resolution_->ResetRates();
- // Update rates for a sequence of intervals.
- int target_rate2[] = {400, 400, 400, 400, 400};
- int encoder_sent_rate2[] = {400, 400, 400, 400, 400};
- int incoming_frame_rate2[] = {15, 15, 15, 15, 15};
- uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
- UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
- fraction_lost2, 5);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 0));
-}
-
-// Two stages: temporal down-sample and then back up temporally, since encoder
-// is under-shooting target even though rate has not increased much.
-TEST_F(QmSelectTest, 2StatgeDownTemporalUpTemporalUndershoot) {
- // Initialize with bitrate, frame rate, and native system width/height.
- InitQmNativeData(100, 30, 640, 480);
-
- // Update with encoder frame size.
- uint16_t codec_width = 640;
- uint16_t codec_height = 480;
- qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
- EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
-
- // Update rates for a sequence of intervals.
- int target_rate[] = {100, 100, 100};
- int encoder_sent_rate[] = {100, 100, 100};
- int incoming_frame_rate[] = {30, 30, 30};
- uint8_t fraction_lost[] = {10, 10, 10};
- UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
- fraction_lost, 3);
-
- // Update content: motion level, and 3 spatial prediction errors.
- // Low motion, high spatial.
- UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
-
- // Reset rates and simulate under-shooting scenario.: expect to go back up.
- qm_resolution_->ResetRates();
- // Update rates for a sequence of intervals.
- int target_rate2[] = {200, 200, 200, 200, 200};
- int encoder_sent_rate2[] = {50, 50, 50, 50, 50};
- int incoming_frame_rate2[] = {15, 15, 15, 15, 15};
- uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
- UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
- fraction_lost2, 5);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(kEasyEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 0));
-}
-
-// Two stages: temporal down-sample and then no action to go up,
-// as encoding rate mis-match is too high.
-TEST_F(QmSelectTest, 2StageDownTemporalNoActionUp) {
- // Initialize with bitrate, frame rate, and native system width/height.
- InitQmNativeData(100, 30, 640, 480);
-
- // Update with encoder frame size.
- uint16_t codec_width = 640;
- uint16_t codec_height = 480;
- qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
- EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
-
- // Update rates for a sequence of intervals.
- int target_rate[] = {100, 100, 100};
- int encoder_sent_rate[] = {100, 100, 100};
- int incoming_frame_rate[] = {30, 30, 30};
- uint8_t fraction_lost[] = {10, 10, 10};
- UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
- fraction_lost, 3);
-
- // Update content: motion level, and 3 spatial prediction errors.
- // Low motion, high spatial.
- UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
-
- // Reset and simulate large rate mis-match: expect no action to go back up.
- qm_resolution_->ResetRates();
- // Update rates for a sequence of intervals.
- int target_rate2[] = {600, 600, 600, 600, 600};
- int encoder_sent_rate2[] = {1000, 1000, 1000, 1000, 1000};
- int incoming_frame_rate2[] = {15, 15, 15, 15, 15};
- uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
- UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
- fraction_lost2, 5);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(kStressedEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
-}
-// 3 stages: spatial down-sample, followed by temporal down-sample,
-// and then go up to full state, as encoding rate has increased.
-TEST_F(QmSelectTest, 3StageDownSpatialTemporlaUpSpatialTemporal) {
- // Initialize with bitrate, frame rate, and native system width/height.
- InitQmNativeData(100, 30, 640, 480);
-
- // Update with encoder frame size.
- uint16_t codec_width = 640;
- uint16_t codec_height = 480;
- qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
- EXPECT_EQ(2, qm_resolution_->GetImageType(codec_width, codec_height));
-
- // Update rates for a sequence of intervals.
- int target_rate[] = {100, 100, 100};
- int encoder_sent_rate[] = {100, 100, 100};
- int incoming_frame_rate[] = {30, 30, 30};
- uint8_t fraction_lost[] = {10, 10, 10};
- UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
- fraction_lost, 3);
-
- // Update content: motion level, and 3 spatial prediction errors.
- // High motion, low spatial.
- UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
-
- // Reset rate and change content data: expect temporal down-sample.
- qm_resolution_->ResetRates();
- qm_resolution_->UpdateCodecFrameSize(320, 240);
- EXPECT_EQ(1, qm_resolution_->GetImageType(320, 240));
-
- // Update content: motion level, and 3 spatial prediction errors.
- // Low motion, high spatial.
- UpdateQmContentData(0.01f, 0.1f, 0.1f, 0.1f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 2));
-
- // Reset rates and go high up in rate: expect to go back up both spatial
- // and temporally.
- qm_resolution_->ResetRates();
- // Update rates for a sequence of intervals.
- int target_rate2[] = {1000, 1000, 1000, 1000, 1000};
- int encoder_sent_rate2[] = {1000, 1000, 1000, 1000, 1000};
- int incoming_frame_rate2[] = {15, 15, 15, 15, 15};
- uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
- UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
- fraction_lost2, 5);
-
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(1, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 0, 0, 0));
-}
-
-// No down-sampling below some totol amount (factor of 16)
-TEST_F(QmSelectTest, NoActionTooMuchDownSampling) {
- // Initialize with bitrate, frame rate, and native system width/height.
- InitQmNativeData(400, 30, 1280, 720);
-
- // Update with encoder frame size.
- uint16_t codec_width = 1280;
- uint16_t codec_height = 720;
- qm_resolution_->UpdateCodecFrameSize(codec_width, codec_height);
- EXPECT_EQ(5, qm_resolution_->GetImageType(codec_width, codec_height));
-
- // Update rates for a sequence of intervals.
- int target_rate[] = {400, 400, 400};
- int encoder_sent_rate[] = {400, 400, 400};
- int incoming_frame_rate[] = {30, 30, 30};
- uint8_t fraction_lost[] = {10, 10, 10};
- UpdateQmRateData(target_rate, encoder_sent_rate, incoming_frame_rate,
- fraction_lost, 3);
-
- // Update content: motion level, and 3 spatial prediction errors.
- // High motion, low spatial: 2x2 spatial expected.
- UpdateQmContentData(0.1f, 0.01f, 0.01f, 0.01f);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
-
- // Reset and lower rates to get another spatial action.
- qm_resolution_->ResetRates();
- qm_resolution_->UpdateCodecFrameSize(640, 360);
- EXPECT_EQ(2, qm_resolution_->GetImageType(640, 360));
- // Update rates for a sequence of intervals.
- int target_rate2[] = {100, 100, 100, 100, 100};
- int encoder_sent_rate2[] = {100, 100, 100, 100, 100};
- int incoming_frame_rate2[] = {30, 30, 30, 30, 30};
- uint8_t fraction_lost2[] = {10, 10, 10, 10, 10};
- UpdateQmRateData(target_rate2, encoder_sent_rate2, incoming_frame_rate2,
- fraction_lost2, 5);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 2, 2, 1));
-
- // Reset and go to low rate: no action should be taken,
- // we went down too much already.
- qm_resolution_->ResetRates();
- qm_resolution_->UpdateCodecFrameSize(320, 180);
- EXPECT_EQ(0, qm_resolution_->GetImageType(320, 180));
- // Update rates for a sequence of intervals.
- int target_rate3[] = {10, 10, 10, 10, 10};
- int encoder_sent_rate3[] = {10, 10, 10, 10, 10};
- int incoming_frame_rate3[] = {30, 30, 30, 30, 30};
- uint8_t fraction_lost3[] = {10, 10, 10, 10, 10};
- UpdateQmRateData(target_rate3, encoder_sent_rate3, incoming_frame_rate3,
- fraction_lost3, 5);
- EXPECT_EQ(0, qm_resolution_->SelectResolution(&qm_scale_));
- EXPECT_EQ(3, qm_resolution_->ComputeContentClass());
- EXPECT_EQ(kStableEncoding, qm_resolution_->GetEncoderState());
- EXPECT_TRUE(IsSelectedActionCorrect(qm_scale_, 1, 1, 1));
-}
-
-void QmSelectTest::InitQmNativeData(float initial_bit_rate,
- int user_frame_rate,
- int native_width,
- int native_height) {
- EXPECT_EQ(0, qm_resolution_->Initialize(initial_bit_rate, user_frame_rate,
- native_width, native_height));
-}
-
-void QmSelectTest::UpdateQmContentData(float motion_metric,
- float spatial_metric,
- float spatial_metric_horiz,
- float spatial_metric_vert) {
- content_metrics_->motion_magnitude = motion_metric;
- content_metrics_->spatial_pred_err = spatial_metric;
- content_metrics_->spatial_pred_err_h = spatial_metric_horiz;
- content_metrics_->spatial_pred_err_v = spatial_metric_vert;
- qm_resolution_->UpdateContent(content_metrics_);
-}
-
-void QmSelectTest::UpdateQmEncodedFrame(int* encoded_size, int num_updates) {
- FrameType frame_type = kVideoFrameDelta;
- for (int i = 0; i < num_updates; i++) {
- // Convert to bytes.
- int32_t encoded_size_update = 1000 * encoded_size[i] / 8;
- qm_resolution_->UpdateEncodedSize(encoded_size_update, frame_type);
- }
-}
-
-void QmSelectTest::UpdateQmRateData(int* target_rate,
- int* encoder_sent_rate,
- int* incoming_frame_rate,
- uint8_t* fraction_lost,
- int num_updates) {
- for (int i = 0; i < num_updates; i++) {
- float target_rate_update = target_rate[i];
- float encoder_sent_rate_update = encoder_sent_rate[i];
- float incoming_frame_rate_update = incoming_frame_rate[i];
- uint8_t fraction_lost_update = fraction_lost[i];
- qm_resolution_->UpdateRates(target_rate_update,
- encoder_sent_rate_update,
- incoming_frame_rate_update,
- fraction_lost_update);
- }
-}
-
-// Check is the selected action from the QmResolution class is the same
-// as the expected scales from |fac_width|, |fac_height|, |fac_temp|.
-bool QmSelectTest::IsSelectedActionCorrect(VCMResolutionScale* qm_scale,
- uint8_t fac_width,
- uint8_t fac_height,
- uint8_t fac_temp) {
- if (qm_scale->spatialWidthFact == fac_width &&
- qm_scale->spatialHeightFact == fac_height &&
- qm_scale->temporalFact == fac_temp) {
- return true;
- } else {
- return false;
- }
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/receiver.cc b/trunk/src/modules/video_coding/main/source/receiver.cc
deleted file mode 100644
index 6be5336..0000000
--- a/trunk/src/modules/video_coding/main/source/receiver.cc
+++ /dev/null
@@ -1,493 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/video_coding/main/source/receiver.h"
-
-#include <assert.h>
-
-#include "modules/video_coding/main/interface/video_coding.h"
-#include "modules/video_coding/main/source/encoded_frame.h"
-#include "modules/video_coding/main/source/internal_defines.h"
-#include "modules/video_coding/main/source/media_opt_util.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
-#include "system_wrappers/interface/trace.h"
-
-namespace webrtc {
-
-VCMReceiver::VCMReceiver(VCMTiming& timing,
- TickTimeBase* clock,
- WebRtc_Word32 vcmId,
- WebRtc_Word32 receiverId,
- bool master)
- : _critSect(CriticalSectionWrapper::CreateCriticalSection()),
- _vcmId(vcmId),
- _clock(clock),
- _receiverId(receiverId),
- _master(master),
- _jitterBuffer(_clock, vcmId, receiverId, master),
- _timing(timing),
- _renderWaitEvent(*new VCMEvent()),
- _state(kPassive) {}
-
-VCMReceiver::~VCMReceiver()
-{
- _renderWaitEvent.Set();
- delete &_renderWaitEvent;
- delete _critSect;
-}
-
-void
-VCMReceiver::Reset()
-{
- CriticalSectionScoped cs(_critSect);
- if (!_jitterBuffer.Running())
- {
- _jitterBuffer.Start();
- }
- else
- {
- _jitterBuffer.Flush();
- }
- _renderWaitEvent.Reset();
- if (_master)
- {
- _state = kReceiving;
- }
- else
- {
- _state = kPassive;
- }
-}
-
-WebRtc_Word32
-VCMReceiver::Initialize()
-{
- CriticalSectionScoped cs(_critSect);
- Reset();
- if (!_master)
- {
- SetNackMode(kNoNack);
- }
- return VCM_OK;
-}
-
-void VCMReceiver::UpdateRtt(WebRtc_UWord32 rtt)
-{
- _jitterBuffer.UpdateRtt(rtt);
-}
-
-WebRtc_Word32
-VCMReceiver::InsertPacket(const VCMPacket& packet,
- WebRtc_UWord16 frameWidth,
- WebRtc_UWord16 frameHeight)
-{
- // Find an empty frame
- VCMEncodedFrame *buffer = NULL;
- const WebRtc_Word32 error = _jitterBuffer.GetFrame(packet, buffer);
- if (error == VCM_OLD_PACKET_ERROR)
- {
- return VCM_OK;
- }
- else if (error != VCM_OK)
- {
- return error;
- }
- assert(buffer);
- {
- CriticalSectionScoped cs(_critSect);
-
- if (frameWidth && frameHeight)
- {
- buffer->SetEncodedSize(static_cast<WebRtc_UWord32>(frameWidth),
- static_cast<WebRtc_UWord32>(frameHeight));
- }
-
- if (_master)
- {
- // Only trace the primary receiver to make it possible
- // to parse and plot the trace file.
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding,
- VCMId(_vcmId, _receiverId),
- "Packet seqNo %u of frame %u at %u",
- packet.seqNum, packet.timestamp,
- MaskWord64ToUWord32(_clock->MillisecondTimestamp()));
- }
-
- const WebRtc_Word64 nowMs = _clock->MillisecondTimestamp();
-
- WebRtc_Word64 renderTimeMs = _timing.RenderTimeMs(packet.timestamp, nowMs);
-
- if (renderTimeMs < 0)
- {
- // Render time error. Assume that this is due to some change in
- // the incoming video stream and reset the JB and the timing.
- _jitterBuffer.Flush();
- _timing.Reset(_clock->MillisecondTimestamp());
- return VCM_FLUSH_INDICATOR;
- }
- else if (renderTimeMs < nowMs - kMaxVideoDelayMs)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
- "This frame should have been rendered more than %u ms ago."
- "Flushing jitter buffer and resetting timing.", kMaxVideoDelayMs);
- _jitterBuffer.Flush();
- _timing.Reset(_clock->MillisecondTimestamp());
- return VCM_FLUSH_INDICATOR;
- }
- else if (_timing.TargetVideoDelay() > kMaxVideoDelayMs)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
- "More than %u ms target delay. Flushing jitter buffer and resetting timing.",
- kMaxVideoDelayMs);
- _jitterBuffer.Flush();
- _timing.Reset(_clock->MillisecondTimestamp());
- return VCM_FLUSH_INDICATOR;
- }
-
- // First packet received belonging to this frame.
- if (buffer->Length() == 0)
- {
- const WebRtc_Word64 nowMs = _clock->MillisecondTimestamp();
- if (_master)
- {
- // Only trace the primary receiver to make it possible to parse and plot the trace file.
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
- "First packet of frame %u at %u", packet.timestamp,
- MaskWord64ToUWord32(nowMs));
- }
- renderTimeMs = _timing.RenderTimeMs(packet.timestamp, nowMs);
- if (renderTimeMs >= 0)
- {
- buffer->SetRenderTime(renderTimeMs);
- }
- else
- {
- buffer->SetRenderTime(nowMs);
- }
- }
-
- // Insert packet into the jitter buffer
- // both media and empty packets
- const VCMFrameBufferEnum
- ret = _jitterBuffer.InsertPacket(buffer, packet);
- if (ret == kFlushIndicator) {
- return VCM_FLUSH_INDICATOR;
- } else if (ret < 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding,
- VCMId(_vcmId, _receiverId),
- "Error inserting packet seqNo=%u, timeStamp=%u",
- packet.seqNum, packet.timestamp);
- return VCM_JITTER_BUFFER_ERROR;
- }
- }
- return VCM_OK;
-}
-
-VCMEncodedFrame* VCMReceiver::FrameForDecoding(WebRtc_UWord16 maxWaitTimeMs,
- WebRtc_Word64& nextRenderTimeMs,
- bool renderTiming,
- VCMReceiver* dualReceiver)
-{
- // No need to enter the critical section here since the jitter buffer
- // is thread-safe.
- FrameType incomingFrameType = kVideoFrameDelta;
- nextRenderTimeMs = -1;
- const WebRtc_Word64 startTimeMs = _clock->MillisecondTimestamp();
- WebRtc_Word64 ret = _jitterBuffer.GetNextTimeStamp(maxWaitTimeMs,
- incomingFrameType,
- nextRenderTimeMs);
- if (ret < 0)
- {
- // No timestamp in jitter buffer at the moment
- return NULL;
- }
- const WebRtc_UWord32 timeStamp = static_cast<WebRtc_UWord32>(ret);
-
- // Update the timing
- _timing.SetRequiredDelay(_jitterBuffer.GetEstimatedJitterMS());
- _timing.UpdateCurrentDelay(timeStamp);
-
- const WebRtc_Word32 tempWaitTime = maxWaitTimeMs -
- static_cast<WebRtc_Word32>(_clock->MillisecondTimestamp() - startTimeMs);
- WebRtc_UWord16 newMaxWaitTime = static_cast<WebRtc_UWord16>(VCM_MAX(tempWaitTime, 0));
-
- VCMEncodedFrame* frame = NULL;
-
- if (renderTiming)
- {
- frame = FrameForDecoding(newMaxWaitTime, nextRenderTimeMs, dualReceiver);
- }
- else
- {
- frame = FrameForRendering(newMaxWaitTime, nextRenderTimeMs, dualReceiver);
- }
-
- if (frame != NULL)
- {
- bool retransmitted = false;
- const WebRtc_Word64 lastPacketTimeMs =
- _jitterBuffer.LastPacketTime(frame, retransmitted);
- if (lastPacketTimeMs >= 0 && !retransmitted)
- {
- // We don't want to include timestamps which have suffered from retransmission
- // here, since we compensate with extra retransmission delay within
- // the jitter estimate.
- _timing.IncomingTimestamp(timeStamp, lastPacketTimeMs);
- }
- if (dualReceiver != NULL)
- {
- dualReceiver->UpdateState(*frame);
- }
- }
- return frame;
-}
-
-VCMEncodedFrame*
-VCMReceiver::FrameForDecoding(WebRtc_UWord16 maxWaitTimeMs,
- WebRtc_Word64 nextRenderTimeMs,
- VCMReceiver* dualReceiver)
-{
- // How long can we wait until we must decode the next frame
- WebRtc_UWord32 waitTimeMs = _timing.MaxWaitingTime(nextRenderTimeMs,
- _clock->MillisecondTimestamp());
-
- // Try to get a complete frame from the jitter buffer
- VCMEncodedFrame* frame = _jitterBuffer.GetCompleteFrameForDecoding(0);
-
- if (frame == NULL && maxWaitTimeMs == 0 && waitTimeMs > 0)
- {
- // If we're not allowed to wait for frames to get complete we must
- // calculate if it's time to decode, and if it's not we will just return
- // for now.
- return NULL;
- }
-
- if (frame == NULL && VCM_MIN(waitTimeMs, maxWaitTimeMs) == 0)
- {
- // No time to wait for a complete frame,
- // check if we have an incomplete
- const bool dualReceiverEnabledAndPassive = (dualReceiver != NULL &&
- dualReceiver->State() == kPassive &&
- dualReceiver->NackMode() == kNackInfinite);
- if (dualReceiverEnabledAndPassive &&
- !_jitterBuffer.CompleteSequenceWithNextFrame())
- {
- // Jitter buffer state might get corrupt with this frame.
- dualReceiver->CopyJitterBufferStateFromReceiver(*this);
- frame = _jitterBuffer.GetFrameForDecoding();
- assert(frame);
- } else {
- frame = _jitterBuffer.GetFrameForDecoding();
- }
- }
- if (frame == NULL)
- {
- // Wait for a complete frame
- frame = _jitterBuffer.GetCompleteFrameForDecoding(maxWaitTimeMs);
- }
- if (frame == NULL)
- {
- // Get an incomplete frame
- if (_timing.MaxWaitingTime(nextRenderTimeMs,
- _clock->MillisecondTimestamp()) > 0)
- {
- // Still time to wait for a complete frame
- return NULL;
- }
-
- // No time left to wait, we must decode this frame now.
- const bool dualReceiverEnabledAndPassive = (dualReceiver != NULL &&
- dualReceiver->State() == kPassive &&
- dualReceiver->NackMode() == kNackInfinite);
- if (dualReceiverEnabledAndPassive &&
- !_jitterBuffer.CompleteSequenceWithNextFrame())
- {
- // Jitter buffer state might get corrupt with this frame.
- dualReceiver->CopyJitterBufferStateFromReceiver(*this);
- }
-
- frame = _jitterBuffer.GetFrameForDecoding();
- }
- return frame;
-}
-
-VCMEncodedFrame*
-VCMReceiver::FrameForRendering(WebRtc_UWord16 maxWaitTimeMs,
- WebRtc_Word64 nextRenderTimeMs,
- VCMReceiver* dualReceiver)
-{
- // How long MUST we wait until we must decode the next frame. This is different for the case
- // where we have a renderer which can render at a specified time. Here we must wait as long
- // as possible before giving the frame to the decoder, which will render the frame as soon
- // as it has been decoded.
- WebRtc_UWord32 waitTimeMs = _timing.MaxWaitingTime(nextRenderTimeMs,
- _clock->MillisecondTimestamp());
- if (maxWaitTimeMs < waitTimeMs)
- {
- // If we're not allowed to wait until the frame is supposed to be rendered
- // we will have to return NULL for now.
- return NULL;
- }
- // Wait until it's time to render
- _renderWaitEvent.Wait(waitTimeMs);
-
- // Get a complete frame if possible
- VCMEncodedFrame* frame = _jitterBuffer.GetCompleteFrameForDecoding(0);
-
- if (frame == NULL)
- {
- // Get an incomplete frame
- const bool dualReceiverEnabledAndPassive = dualReceiver != NULL &&
- dualReceiver->State() == kPassive &&
- dualReceiver->NackMode() == kNackInfinite;
- if (dualReceiverEnabledAndPassive && !_jitterBuffer.CompleteSequenceWithNextFrame())
- {
- // Jitter buffer state might get corrupt with this frame.
- dualReceiver->CopyJitterBufferStateFromReceiver(*this);
- }
-
- frame = _jitterBuffer.GetFrameForDecoding();
- }
- return frame;
-}
-
-void
-VCMReceiver::ReleaseFrame(VCMEncodedFrame* frame)
-{
- _jitterBuffer.ReleaseFrame(frame);
-}
-
-WebRtc_Word32
-VCMReceiver::ReceiveStatistics(WebRtc_UWord32& bitRate, WebRtc_UWord32& frameRate)
-{
- const WebRtc_Word32 ret = _jitterBuffer.GetUpdate(frameRate, bitRate);
- bitRate /= 1000; // Should be in kbps
- return ret;
-}
-
-WebRtc_Word32
-VCMReceiver::ReceivedFrameCount(VCMFrameCount& frameCount) const
-{
- return _jitterBuffer.GetFrameStatistics(frameCount.numDeltaFrames,
- frameCount.numKeyFrames);
-}
-
-WebRtc_UWord32 VCMReceiver::DiscardedPackets() const {
- return _jitterBuffer.DiscardedPackets();
-}
-
-void
-VCMReceiver::SetNackMode(VCMNackMode nackMode)
-{
- CriticalSectionScoped cs(_critSect);
- // Default to always having NACK enabled in hybrid mode.
- _jitterBuffer.SetNackMode(nackMode, kLowRttNackMs, -1);
- if (!_master)
- {
- _state = kPassive; // The dual decoder defaults to passive
- }
-}
-
-VCMNackMode
-VCMReceiver::NackMode() const
-{
- CriticalSectionScoped cs(_critSect);
- return _jitterBuffer.GetNackMode();
-}
-
-VCMNackStatus
-VCMReceiver::NackList(WebRtc_UWord16* nackList, WebRtc_UWord16& size)
-{
- bool extended = false;
- WebRtc_UWord16 nackListSize = 0;
- WebRtc_UWord16* internalNackList = _jitterBuffer.GetNackList(nackListSize, extended);
- if (internalNackList == NULL && nackListSize == 0xffff)
- {
- // This combination is used to trigger key frame requests.
- size = 0;
- return kNackKeyFrameRequest;
- }
- if (nackListSize > size)
- {
- size = nackListSize;
- return kNackNeedMoreMemory;
- }
- if (internalNackList != NULL && nackListSize > 0) {
- memcpy(nackList, internalNackList, nackListSize * sizeof(WebRtc_UWord16));
- }
- size = nackListSize;
- return kNackOk;
-}
-
-// Decide whether we should change decoder state. This should be done if the dual decoder
-// has caught up with the decoder decoding with packet losses.
-bool
-VCMReceiver::DualDecoderCaughtUp(VCMEncodedFrame* dualFrame, VCMReceiver& dualReceiver) const
-{
- if (dualFrame == NULL)
- {
- return false;
- }
- if (_jitterBuffer.LastDecodedTimestamp() == dualFrame->TimeStamp())
- {
- dualReceiver.UpdateState(kWaitForPrimaryDecode);
- return true;
- }
- return false;
-}
-
-void
-VCMReceiver::CopyJitterBufferStateFromReceiver(const VCMReceiver& receiver)
-{
- _jitterBuffer.CopyFrom(receiver._jitterBuffer);
-}
-
-VCMReceiverState
-VCMReceiver::State() const
-{
- CriticalSectionScoped cs(_critSect);
- return _state;
-}
-
-void
-VCMReceiver::UpdateState(VCMReceiverState newState)
-{
- CriticalSectionScoped cs(_critSect);
- assert(!(_state == kPassive && newState == kWaitForPrimaryDecode));
-// assert(!(_state == kReceiving && newState == kPassive));
- _state = newState;
-}
-
-void
-VCMReceiver::UpdateState(VCMEncodedFrame& frame)
-{
- if (_jitterBuffer.GetNackMode() == kNoNack)
- {
- // Dual decoder mode has not been enabled.
- return;
- }
- // Update the dual receiver state
- if (frame.Complete() && frame.FrameType() == kVideoFrameKey)
- {
- UpdateState(kPassive);
- }
- if (State() == kWaitForPrimaryDecode &&
- frame.Complete() && !frame.MissingFrame())
- {
- UpdateState(kPassive);
- }
- if (frame.MissingFrame() || !frame.Complete())
- {
- // State was corrupted, enable dual receiver.
- UpdateState(kReceiving);
- }
-}
-
-}
diff --git a/trunk/src/modules/video_coding/main/source/receiver.h b/trunk/src/modules/video_coding/main/source/receiver.h
deleted file mode 100644
index 0081ed1..0000000
--- a/trunk/src/modules/video_coding/main/source/receiver.h
+++ /dev/null
@@ -1,101 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_RECEIVER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_RECEIVER_H_
-
-#include "critical_section_wrapper.h"
-#include "jitter_buffer.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
-#include "timing.h"
-#include "packet.h"
-
-namespace webrtc
-{
-
-class VCMEncodedFrame;
-
-enum VCMNackStatus
-{
- kNackOk,
- kNackNeedMoreMemory,
- kNackKeyFrameRequest
-};
-
-
-enum VCMReceiverState
-{
- kReceiving,
- kPassive,
- kWaitForPrimaryDecode
-};
-
-class VCMReceiver
-{
-public:
- VCMReceiver(VCMTiming& timing,
- TickTimeBase* clock,
- WebRtc_Word32 vcmId = -1,
- WebRtc_Word32 receiverId = -1,
- bool master = true);
- ~VCMReceiver();
-
- void Reset();
- WebRtc_Word32 Initialize();
- void UpdateRtt(WebRtc_UWord32 rtt);
- WebRtc_Word32 InsertPacket(const VCMPacket& packet,
- WebRtc_UWord16 frameWidth,
- WebRtc_UWord16 frameHeight);
- VCMEncodedFrame* FrameForDecoding(WebRtc_UWord16 maxWaitTimeMs,
- WebRtc_Word64& nextRenderTimeMs,
- bool renderTiming = true,
- VCMReceiver* dualReceiver = NULL);
- void ReleaseFrame(VCMEncodedFrame* frame);
- WebRtc_Word32 ReceiveStatistics(WebRtc_UWord32& bitRate, WebRtc_UWord32& frameRate);
- WebRtc_Word32 ReceivedFrameCount(VCMFrameCount& frameCount) const;
- WebRtc_UWord32 DiscardedPackets() const;
-
- // NACK
- void SetNackMode(VCMNackMode nackMode);
- VCMNackMode NackMode() const;
- VCMNackStatus NackList(WebRtc_UWord16* nackList, WebRtc_UWord16& size);
-
- // Dual decoder
- bool DualDecoderCaughtUp(VCMEncodedFrame* dualFrame, VCMReceiver& dualReceiver) const;
- VCMReceiverState State() const;
-
-private:
- VCMEncodedFrame* FrameForDecoding(WebRtc_UWord16 maxWaitTimeMs,
- WebRtc_Word64 nextrenderTimeMs,
- VCMReceiver* dualReceiver);
- VCMEncodedFrame* FrameForRendering(WebRtc_UWord16 maxWaitTimeMs,
- WebRtc_Word64 nextrenderTimeMs,
- VCMReceiver* dualReceiver);
- void CopyJitterBufferStateFromReceiver(const VCMReceiver& receiver);
- void UpdateState(VCMReceiverState newState);
- void UpdateState(VCMEncodedFrame& frame);
- static WebRtc_Word32 GenerateReceiverId();
-
- CriticalSectionWrapper* _critSect;
- WebRtc_Word32 _vcmId;
- TickTimeBase* _clock;
- WebRtc_Word32 _receiverId;
- bool _master;
- VCMJitterBuffer _jitterBuffer;
- VCMTiming& _timing;
- VCMEvent& _renderWaitEvent;
- VCMReceiverState _state;
-
- static WebRtc_Word32 _receiverIdCounter;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_RECEIVER_H_
diff --git a/trunk/src/modules/video_coding/main/source/rtt_filter.cc b/trunk/src/modules/video_coding/main/source/rtt_filter.cc
deleted file mode 100644
index 36f7660..0000000
--- a/trunk/src/modules/video_coding/main/source/rtt_filter.cc
+++ /dev/null
@@ -1,214 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "trace.h"
-#include "internal_defines.h"
-#include "rtt_filter.h"
-
-#include <cmath>
-#include <stdlib.h>
-#include <string.h>
-
-namespace webrtc {
-
-VCMRttFilter::VCMRttFilter(WebRtc_Word32 vcmId, WebRtc_Word32 receiverId)
-:
-_vcmId(vcmId),
-_receiverId(receiverId),
-_filtFactMax(35),
-_jumpStdDevs(2.5),
-_driftStdDevs(3.5),
-_detectThreshold(kMaxDriftJumpCount)
-{
- Reset();
-}
-
-VCMRttFilter&
-VCMRttFilter::operator=(const VCMRttFilter& rhs)
-{
- if (this != &rhs)
- {
- _gotNonZeroUpdate = rhs._gotNonZeroUpdate;
- _avgRtt = rhs._avgRtt;
- _varRtt = rhs._varRtt;
- _maxRtt = rhs._maxRtt;
- _filtFactCount = rhs._filtFactCount;
- _jumpCount = rhs._jumpCount;
- _driftCount = rhs._driftCount;
- memcpy(_jumpBuf, rhs._jumpBuf, sizeof(_jumpBuf));
- memcpy(_driftBuf, rhs._driftBuf, sizeof(_driftBuf));
- }
- return *this;
-}
-
-void
-VCMRttFilter::Reset()
-{
- _gotNonZeroUpdate = false;
- _avgRtt = 0;
- _varRtt = 0;
- _maxRtt = 0;
- _filtFactCount = 1;
- _jumpCount = 0;
- _driftCount = 0;
- memset(_jumpBuf, 0, kMaxDriftJumpCount);
- memset(_driftBuf, 0, kMaxDriftJumpCount);
-}
-
-void
-VCMRttFilter::Update(WebRtc_UWord32 rttMs)
-{
- if (!_gotNonZeroUpdate)
- {
- if (rttMs == 0)
- {
- return;
- }
- _gotNonZeroUpdate = true;
- }
-
- // Sanity check
- if (rttMs > 3000)
- {
- rttMs = 3000;
- }
-
- double filtFactor = 0;
- if (_filtFactCount > 1)
- {
- filtFactor = static_cast<double>(_filtFactCount - 1) / _filtFactCount;
- }
- _filtFactCount++;
- if (_filtFactCount > _filtFactMax)
- {
- // This prevents filtFactor from going above
- // (_filtFactMax - 1) / _filtFactMax,
- // e.g., _filtFactMax = 50 => filtFactor = 49/50 = 0.98
- _filtFactCount = _filtFactMax;
- }
- double oldAvg = _avgRtt;
- double oldVar = _varRtt;
- _avgRtt = filtFactor * _avgRtt + (1 - filtFactor) * rttMs;
- _varRtt = filtFactor * _varRtt + (1 - filtFactor) *
- (rttMs - _avgRtt) * (rttMs - _avgRtt);
- _maxRtt = VCM_MAX(rttMs, _maxRtt);
- if (!JumpDetection(rttMs) || !DriftDetection(rttMs))
- {
- // In some cases we don't want to update the statistics
- _avgRtt = oldAvg;
- _varRtt = oldVar;
- }
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
- "RttFilter Update: sample=%u avgRtt=%f varRtt=%f maxRtt=%u",
- rttMs, _avgRtt, _varRtt, _maxRtt);
-}
-
-bool
-VCMRttFilter::JumpDetection(WebRtc_UWord32 rttMs)
-{
- double diffFromAvg = _avgRtt - rttMs;
- if (abs(diffFromAvg) > _jumpStdDevs * sqrt(_varRtt))
- {
- int diffSign = (diffFromAvg >= 0) ? 1 : -1;
- int jumpCountSign = (_jumpCount >= 0) ? 1 : -1;
- if (diffSign != jumpCountSign)
- {
- // Since the signs differ the samples currently
- // in the buffer is useless as they represent a
- // jump in a different direction.
- _jumpCount = 0;
- }
- if (abs(_jumpCount) < kMaxDriftJumpCount)
- {
- // Update the buffer used for the short time
- // statistics.
- // The sign of the diff is used for updating the counter since
- // we want to use the same buffer for keeping track of when
- // the RTT jumps down and up.
- _jumpBuf[abs(_jumpCount)] = rttMs;
- _jumpCount += diffSign;
- }
- if (abs(_jumpCount) >= _detectThreshold)
- {
- // Detected an RTT jump
- ShortRttFilter(_jumpBuf, abs(_jumpCount));
- _filtFactCount = _detectThreshold + 1;
- _jumpCount = 0;
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
- "Detected an RTT jump");
- }
- else
- {
- return false;
- }
- }
- else
- {
- _jumpCount = 0;
- }
- return true;
-}
-
-bool
-VCMRttFilter::DriftDetection(WebRtc_UWord32 rttMs)
-{
- if (_maxRtt - _avgRtt > _driftStdDevs * sqrt(_varRtt))
- {
- if (_driftCount < kMaxDriftJumpCount)
- {
- // Update the buffer used for the short time
- // statistics.
- _driftBuf[_driftCount] = rttMs;
- _driftCount++;
- }
- if (_driftCount >= _detectThreshold)
- {
- // Detected an RTT drift
- ShortRttFilter(_driftBuf, _driftCount);
- _filtFactCount = _detectThreshold + 1;
- _driftCount = 0;
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _receiverId),
- "Detected an RTT drift");
- }
- }
- else
- {
- _driftCount = 0;
- }
- return true;
-}
-
-void
-VCMRttFilter::ShortRttFilter(WebRtc_UWord32* buf, WebRtc_UWord32 length)
-{
- if (length == 0)
- {
- return;
- }
- _maxRtt = 0;
- _avgRtt = 0;
- for (WebRtc_UWord32 i=0; i < length; i++)
- {
- if (buf[i] > _maxRtt)
- {
- _maxRtt = buf[i];
- }
- _avgRtt += buf[i];
- }
- _avgRtt = _avgRtt / static_cast<double>(length);
-}
-
-WebRtc_UWord32
-VCMRttFilter::RttMs() const
-{
- return static_cast<WebRtc_UWord32>(_maxRtt + 0.5);
-}
-
-}
diff --git a/trunk/src/modules/video_coding/main/source/rtt_filter.h b/trunk/src/modules/video_coding/main/source/rtt_filter.h
deleted file mode 100644
index 5ec85fd..0000000
--- a/trunk/src/modules/video_coding/main/source/rtt_filter.h
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
-
-#include "typedefs.h"
-
-namespace webrtc
-{
-
-class VCMRttFilter
-{
-public:
- VCMRttFilter(WebRtc_Word32 vcmId = 0, WebRtc_Word32 receiverId = 0);
-
- VCMRttFilter& operator=(const VCMRttFilter& rhs);
-
- // Resets the filter.
- void Reset();
- // Updates the filter with a new sample.
- void Update(WebRtc_UWord32 rttMs);
- // A getter function for the current RTT level in ms.
- WebRtc_UWord32 RttMs() const;
-
-private:
- // The size of the drift and jump memory buffers
- // and thus also the detection threshold for these
- // detectors in number of samples.
- enum { kMaxDriftJumpCount = 5 };
- // Detects RTT jumps by comparing the difference between
- // samples and average to the standard deviation.
- // Returns true if the long time statistics should be updated
- // and false otherwise
- bool JumpDetection(WebRtc_UWord32 rttMs);
- // Detects RTT drifts by comparing the difference between
- // max and average to the standard deviation.
- // Returns true if the long time statistics should be updated
- // and false otherwise
- bool DriftDetection(WebRtc_UWord32 rttMs);
- // Computes the short time average and maximum of the vector buf.
- void ShortRttFilter(WebRtc_UWord32* buf, WebRtc_UWord32 length);
-
- WebRtc_Word32 _vcmId;
- WebRtc_Word32 _receiverId;
- bool _gotNonZeroUpdate;
- double _avgRtt;
- double _varRtt;
- WebRtc_UWord32 _maxRtt;
- WebRtc_UWord32 _filtFactCount;
- const WebRtc_UWord32 _filtFactMax;
- const double _jumpStdDevs;
- const double _driftStdDevs;
- WebRtc_Word32 _jumpCount;
- WebRtc_Word32 _driftCount;
- const WebRtc_Word32 _detectThreshold;
- WebRtc_UWord32 _jumpBuf[kMaxDriftJumpCount];
- WebRtc_UWord32 _driftBuf[kMaxDriftJumpCount];
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_RTT_FILTER_H_
diff --git a/trunk/src/modules/video_coding/main/source/session_info.cc b/trunk/src/modules/video_coding/main/source/session_info.cc
deleted file mode 100644
index 7261403..0000000
--- a/trunk/src/modules/video_coding/main/source/session_info.cc
+++ /dev/null
@@ -1,595 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/video_coding/main/source/session_info.h"
-
-#include "modules/video_coding/main/source/packet.h"
-
-namespace webrtc {
-
-VCMSessionInfo::VCMSessionInfo()
- : session_nack_(false),
- complete_(false),
- decodable_(false),
- frame_type_(kVideoFrameDelta),
- previous_frame_loss_(false),
- packets_(),
- empty_seq_num_low_(-1),
- empty_seq_num_high_(-1),
- packets_not_decodable_(0) {
-}
-
-void VCMSessionInfo::UpdateDataPointers(const uint8_t* old_base_ptr,
- const uint8_t* new_base_ptr) {
- for (PacketIterator it = packets_.begin(); it != packets_.end(); ++it)
- if ((*it).dataPtr != NULL) {
- assert(old_base_ptr != NULL && new_base_ptr != NULL);
- (*it).dataPtr = new_base_ptr + ((*it).dataPtr - old_base_ptr);
- }
-}
-
-int VCMSessionInfo::LowSequenceNumber() const {
- if (packets_.empty())
- return empty_seq_num_low_;
- return packets_.front().seqNum;
-}
-
-int VCMSessionInfo::HighSequenceNumber() const {
- if (packets_.empty())
- return empty_seq_num_high_;
- return LatestSequenceNumber(packets_.back().seqNum, empty_seq_num_high_,
- NULL);
-}
-
-int VCMSessionInfo::PictureId() const {
- if (packets_.empty() ||
- packets_.front().codecSpecificHeader.codec != kRTPVideoVP8)
- return kNoPictureId;
- return packets_.front().codecSpecificHeader.codecHeader.VP8.pictureId;
-}
-
-int VCMSessionInfo::TemporalId() const {
- if (packets_.empty() ||
- packets_.front().codecSpecificHeader.codec != kRTPVideoVP8)
- return kNoTemporalIdx;
- return packets_.front().codecSpecificHeader.codecHeader.VP8.temporalIdx;
-}
-
-bool VCMSessionInfo::LayerSync() const {
- if (packets_.empty() ||
- packets_.front().codecSpecificHeader.codec != kRTPVideoVP8)
- return false;
- return packets_.front().codecSpecificHeader.codecHeader.VP8.layerSync;
-}
-
-int VCMSessionInfo::Tl0PicId() const {
- if (packets_.empty() ||
- packets_.front().codecSpecificHeader.codec != kRTPVideoVP8)
- return kNoTl0PicIdx;
- return packets_.front().codecSpecificHeader.codecHeader.VP8.tl0PicIdx;
-}
-
-bool VCMSessionInfo::NonReference() const {
- if (packets_.empty() ||
- packets_.front().codecSpecificHeader.codec != kRTPVideoVP8)
- return false;
- return packets_.front().codecSpecificHeader.codecHeader.VP8.nonReference;
-}
-
-void VCMSessionInfo::Reset() {
- session_nack_ = false;
- complete_ = false;
- decodable_ = false;
- frame_type_ = kVideoFrameDelta;
- previous_frame_loss_ = false;
- packets_.clear();
- empty_seq_num_low_ = -1;
- empty_seq_num_high_ = -1;
- packets_not_decodable_ = 0;
-}
-
-int VCMSessionInfo::SessionLength() const {
- int length = 0;
- for (PacketIteratorConst it = packets_.begin(); it != packets_.end(); ++it)
- length += (*it).sizeBytes;
- return length;
-}
-
-int VCMSessionInfo::InsertBuffer(uint8_t* frame_buffer,
- PacketIterator packet_it) {
- VCMPacket& packet = *packet_it;
- PacketIterator it;
-
- int packet_size = packet.sizeBytes;
- packet_size += (packet.insertStartCode ? kH264StartCodeLengthBytes : 0);
-
- // Calculate the offset into the frame buffer for this packet.
- int offset = 0;
- for (it = packets_.begin(); it != packet_it; ++it)
- offset += (*it).sizeBytes;
-
- // Set the data pointer to pointing to the start of this packet in the
- // frame buffer.
- const uint8_t* data = packet.dataPtr;
- packet.dataPtr = frame_buffer + offset;
- packet.sizeBytes = packet_size;
-
- ShiftSubsequentPackets(packet_it, packet_size);
-
- const unsigned char startCode[] = {0, 0, 0, 1};
- if (packet.insertStartCode) {
- memcpy(const_cast<uint8_t*>(packet.dataPtr), startCode,
- kH264StartCodeLengthBytes);
- }
- memcpy(const_cast<uint8_t*>(packet.dataPtr
- + (packet.insertStartCode ? kH264StartCodeLengthBytes : 0)),
- data,
- packet.sizeBytes);
-
- return packet_size;
-}
-
-void VCMSessionInfo::ShiftSubsequentPackets(PacketIterator it,
- int steps_to_shift) {
- ++it;
- if (it == packets_.end())
- return;
- uint8_t* first_packet_ptr = const_cast<WebRtc_UWord8*>((*it).dataPtr);
- int shift_length = 0;
- // Calculate the total move length and move the data pointers in advance.
- for (; it != packets_.end(); ++it) {
- shift_length += (*it).sizeBytes;
- if ((*it).dataPtr != NULL)
- (*it).dataPtr += steps_to_shift;
- }
- memmove(first_packet_ptr + steps_to_shift, first_packet_ptr, shift_length);
-}
-
-void VCMSessionInfo::UpdateCompleteSession() {
- if (packets_.front().isFirstPacket && packets_.back().markerBit) {
- // Do we have all the packets in this session?
- bool complete_session = true;
- PacketIterator it = packets_.begin();
- PacketIterator prev_it = it;
- ++it;
- for (; it != packets_.end(); ++it) {
- if (!InSequence(it, prev_it)) {
- complete_session = false;
- break;
- }
- prev_it = it;
- }
- complete_ = complete_session;
- }
-}
-
-void VCMSessionInfo::UpdateDecodableSession(int rttMs) {
- // Irrelevant if session is already complete or decodable
- if (complete_ || decodable_)
- return;
- // First iteration - do nothing
-}
-
-bool VCMSessionInfo::complete() const {
- return complete_;
-}
-
-bool VCMSessionInfo::decodable() const {
- return decodable_;
-}
-
-// Find the end of the NAL unit which the packet pointed to by |packet_it|
-// belongs to. Returns an iterator to the last packet of the frame if the end
-// of the NAL unit wasn't found.
-VCMSessionInfo::PacketIterator VCMSessionInfo::FindNaluEnd(
- PacketIterator packet_it) const {
- if ((*packet_it).completeNALU == kNaluEnd ||
- (*packet_it).completeNALU == kNaluComplete) {
- return packet_it;
- }
- // Find the end of the NAL unit.
- for (; packet_it != packets_.end(); ++packet_it) {
- if (((*packet_it).completeNALU == kNaluComplete &&
- (*packet_it).sizeBytes > 0) ||
- // Found next NALU.
- (*packet_it).completeNALU == kNaluStart)
- return --packet_it;
- if ((*packet_it).completeNALU == kNaluEnd)
- return packet_it;
- }
- // The end wasn't found.
- return --packet_it;
-}
-
-int VCMSessionInfo::DeletePacketData(PacketIterator start,
- PacketIterator end) {
- int bytes_to_delete = 0; // The number of bytes to delete.
- PacketIterator packet_after_end = end;
- ++packet_after_end;
-
- // Get the number of bytes to delete.
- // Clear the size of these packets.
- for (PacketIterator it = start; it != packet_after_end; ++it) {
- bytes_to_delete += (*it).sizeBytes;
- (*it).sizeBytes = 0;
- (*it).dataPtr = NULL;
- ++packets_not_decodable_;
- }
- if (bytes_to_delete > 0)
- ShiftSubsequentPackets(end, -bytes_to_delete);
- return bytes_to_delete;
-}
-
-int VCMSessionInfo::BuildVP8FragmentationHeader(
- uint8_t* frame_buffer,
- int frame_buffer_length,
- RTPFragmentationHeader* fragmentation) {
- int new_length = 0;
- // Allocate space for max number of partitions
- fragmentation->VerifyAndAllocateFragmentationHeader(kMaxVP8Partitions);
- fragmentation->fragmentationVectorSize = 0;
- memset(fragmentation->fragmentationLength, 0,
- kMaxVP8Partitions * sizeof(WebRtc_UWord32));
- if (packets_.empty())
- return new_length;
- PacketIterator it = FindNextPartitionBeginning(packets_.begin(),
- &packets_not_decodable_);
- while (it != packets_.end()) {
- const int partition_id =
- (*it).codecSpecificHeader.codecHeader.VP8.partitionId;
- PacketIterator partition_end = FindPartitionEnd(it);
- fragmentation->fragmentationOffset[partition_id] =
- (*it).dataPtr - frame_buffer;
- assert(fragmentation->fragmentationOffset[partition_id] <
- static_cast<WebRtc_UWord32>(frame_buffer_length));
- fragmentation->fragmentationLength[partition_id] =
- (*partition_end).dataPtr + (*partition_end).sizeBytes - (*it).dataPtr;
- assert(fragmentation->fragmentationLength[partition_id] <=
- static_cast<WebRtc_UWord32>(frame_buffer_length));
- new_length += fragmentation->fragmentationLength[partition_id];
- ++partition_end;
- it = FindNextPartitionBeginning(partition_end, &packets_not_decodable_);
- if (partition_id + 1 > fragmentation->fragmentationVectorSize)
- fragmentation->fragmentationVectorSize = partition_id + 1;
- }
- // Set all empty fragments to start where the previous fragment ends,
- // and have zero length.
- if (fragmentation->fragmentationLength[0] == 0)
- fragmentation->fragmentationOffset[0] = 0;
- for (int i = 1; i < fragmentation->fragmentationVectorSize; ++i) {
- if (fragmentation->fragmentationLength[i] == 0)
- fragmentation->fragmentationOffset[i] =
- fragmentation->fragmentationOffset[i - 1] +
- fragmentation->fragmentationLength[i - 1];
- assert(i == 0 ||
- fragmentation->fragmentationOffset[i] >=
- fragmentation->fragmentationOffset[i - 1]);
- }
- assert(new_length <= frame_buffer_length);
- return new_length;
-}
-
-VCMSessionInfo::PacketIterator VCMSessionInfo::FindNextPartitionBeginning(
- PacketIterator it, int* packets_skipped) const {
- while (it != packets_.end()) {
- if ((*it).codecSpecificHeader.codecHeader.VP8.beginningOfPartition) {
- return it;
- } else if (packets_skipped != NULL) {
- // This packet belongs to a partition with a previous loss and can't
- // be decoded.
- ++(*packets_skipped);
- }
- ++it;
- }
- return it;
-}
-
-VCMSessionInfo::PacketIterator VCMSessionInfo::FindPartitionEnd(
- PacketIterator it) const {
- assert((*it).codec == kVideoCodecVP8);
- PacketIterator prev_it = it;
- const int partition_id =
- (*it).codecSpecificHeader.codecHeader.VP8.partitionId;
- while (it != packets_.end()) {
- bool beginning =
- (*it).codecSpecificHeader.codecHeader.VP8.beginningOfPartition;
- int current_partition_id =
- (*it).codecSpecificHeader.codecHeader.VP8.partitionId;
- bool packet_loss_found = (!beginning && !InSequence(it, prev_it));
- if (packet_loss_found ||
- (beginning && current_partition_id != partition_id)) {
- // Missing packet, the previous packet was the last in sequence.
- return prev_it;
- }
- prev_it = it;
- ++it;
- }
- return prev_it;
-}
-
-bool VCMSessionInfo::InSequence(const PacketIterator& packet_it,
- const PacketIterator& prev_packet_it) {
- // If the two iterators are pointing to the same packet they are considered
- // to be in sequence.
- return (packet_it == prev_packet_it ||
- (static_cast<WebRtc_UWord16>((*prev_packet_it).seqNum + 1) ==
- (*packet_it).seqNum));
-}
-
-int VCMSessionInfo::MakeDecodable() {
- int return_length = 0;
- if (packets_.empty()) {
- return 0;
- }
- PacketIterator it = packets_.begin();
- // Make sure we remove the first NAL unit if it's not decodable.
- if ((*it).completeNALU == kNaluIncomplete ||
- (*it).completeNALU == kNaluEnd) {
- PacketIterator nalu_end = FindNaluEnd(it);
- return_length += DeletePacketData(it, nalu_end);
- it = nalu_end;
- }
- PacketIterator prev_it = it;
- // Take care of the rest of the NAL units.
- for (; it != packets_.end(); ++it) {
- bool start_of_nalu = ((*it).completeNALU == kNaluStart ||
- (*it).completeNALU == kNaluComplete);
- if (!start_of_nalu && !InSequence(it, prev_it)) {
- // Found a sequence number gap due to packet loss.
- PacketIterator nalu_end = FindNaluEnd(it);
- return_length += DeletePacketData(it, nalu_end);
- it = nalu_end;
- }
- prev_it = it;
- }
- return return_length;
-}
-
-int VCMSessionInfo::BuildHardNackList(int* seq_num_list,
- int seq_num_list_length) {
- if (NULL == seq_num_list || seq_num_list_length < 1) {
- return -1;
- }
- if (packets_.empty()) {
- return 0;
- }
-
- // Find end point (index of entry equals the sequence number of the first
- // packet).
- int index = 0;
- for (; index < seq_num_list_length; ++index) {
- if (seq_num_list[index] == packets_.front().seqNum) {
- seq_num_list[index] = -1;
- ++index;
- break;
- }
- }
-
- // Zero out between the first entry and the end point.
- PacketIterator it = packets_.begin();
- PacketIterator prev_it = it;
- ++it;
- while (it != packets_.end() && index < seq_num_list_length) {
- if (!InSequence(it, prev_it)) {
- // Found a sequence number gap due to packet loss.
- index += PacketsMissing(it, prev_it);
- session_nack_ = true;
- }
- seq_num_list[index] = -1;
- ++index;
- prev_it = it;
- ++it;
- }
- if (!packets_.front().isFirstPacket)
- session_nack_ = true;
- return 0;
-}
-
-int VCMSessionInfo::BuildSoftNackList(int* seq_num_list,
- int seq_num_list_length,
- int rtt_ms) {
- if (NULL == seq_num_list || seq_num_list_length < 1) {
- return -1;
- }
- if (packets_.empty() && empty_seq_num_low_ == -1) {
- return 0;
- }
-
- int index = 0;
- int low_seq_num = (packets_.empty()) ? empty_seq_num_low_:
- packets_.front().seqNum;
- // Find entrance point (index of entry equals the sequence number of the
- // first packet).
- for (; index < seq_num_list_length; ++index) {
- if (seq_num_list[index] == low_seq_num) {
- seq_num_list[index] = -1;
- break;
- }
- }
-
- // TODO(mikhal): 1. Update score based on RTT value 2. Add partition data.
- // Use the previous available.
- bool base_available = false;
- if ((index > 0) && (seq_num_list[index] == -1)) {
- // Found first packet, for now let's go only one back.
- if ((seq_num_list[index - 1] == -1) || (seq_num_list[index - 1] == -2)) {
- // This is indeed the first packet, as previous packet was populated.
- base_available = true;
- }
- }
- bool allow_nack = ((packets_.size() > 0 && !packets_.front().isFirstPacket)
- || !base_available);
-
- // Zero out between first entry and end point.
-
- int media_high_seq_num;
- if (HaveLastPacket()) {
- media_high_seq_num = packets_.back().seqNum;
- } else {
- // Estimation.
- if (empty_seq_num_low_ >= 0) {
- // Assuming empty packets have later sequence numbers than media packets.
- media_high_seq_num = empty_seq_num_low_ - 1;
- } else {
- // Since this frame doesn't have the marker bit we can assume it should
- // contain at least one more packet.
- media_high_seq_num = static_cast<uint16_t>(packets_.back().seqNum + 1);
- }
- }
-
- // Compute session/packet scores and thresholds:
- // based on RTT and layer info (when available).
- float nack_score_threshold = 0.25f;
- float layer_score = TemporalId() > 0 ? 0.0f : 1.0f;
- float rtt_score = 1.0f;
- float score_multiplier = rtt_score * layer_score;
- // Zero out between first entry and end point.
- if (!packets_.empty()) {
- PacketIterator it = packets_.begin();
- PacketIterator prev_it = it;
- ++index;
- ++it;
- // TODO(holmer): Rewrite this in a way which better makes use of the list.
- while (it != packets_.end() && index < seq_num_list_length) {
- // Only process media packet sequence numbers.
- if (LatestSequenceNumber((*it).seqNum, media_high_seq_num, NULL) ==
- (*it).seqNum && (*it).seqNum != media_high_seq_num)
- break;
- if (!InSequence(it, prev_it)) {
- // Found a sequence number gap due to packet loss.
- int num_lost = PacketsMissing(it, prev_it);
- for (int i = 0 ; i < num_lost; ++i) {
- // Compute score of the packet.
- float score = 1.0f;
- // Multiply internal score (packet) by score multiplier.
- score *= score_multiplier;
- if (score > nack_score_threshold) {
- allow_nack = true;
- } else {
- seq_num_list[index] = -1;
- }
- ++index;
- }
- }
- seq_num_list[index] = -1;
- ++index;
- prev_it = it;
- ++it;
- }
- }
-
- // Empty packets follow the data packets, and therefore have a higher
- // sequence number. We do not want to NACK empty packets.
- if ((empty_seq_num_low_ != -1) && (empty_seq_num_high_ != -1) &&
- (index < seq_num_list_length)) {
- // First make sure that we are at least at the minimum value (if not we are
- // missing last packet(s)).
- while (seq_num_list[index] < empty_seq_num_low_ &&
- index < seq_num_list_length) {
- ++index;
- }
-
- // Mark empty packets.
- while (seq_num_list[index] <= empty_seq_num_high_ &&
- index < seq_num_list_length) {
- seq_num_list[index] = -2;
- ++index;
- }
- }
-
- session_nack_ = allow_nack;
- return 0;
-}
-
-int VCMSessionInfo::PacketsMissing(const PacketIterator& packet_it,
- const PacketIterator& prev_packet_it) {
- if (packet_it == prev_packet_it)
- return 0;
- if ((*prev_packet_it).seqNum > (*packet_it).seqNum) // Wrap.
- return static_cast<WebRtc_UWord16>(
- static_cast<WebRtc_UWord32>((*packet_it).seqNum + 0x10000) -
- (*prev_packet_it).seqNum) - 1;
- else
- return (*packet_it).seqNum - (*prev_packet_it).seqNum - 1;
-}
-
-bool
-VCMSessionInfo::HaveLastPacket() const {
- return (!packets_.empty() && packets_.back().markerBit);
-}
-
-bool
-VCMSessionInfo::session_nack() const {
- return session_nack_;
-}
-
-int VCMSessionInfo::InsertPacket(const VCMPacket& packet,
- uint8_t* frame_buffer,
- bool enable_decodable_state,
- int rtt_ms) {
- // Check if this is first packet (only valid for some codecs)
- if (packet.isFirstPacket) {
- // The first packet in a frame signals the frame type.
- frame_type_ = packet.frameType;
- } else if (frame_type_ == kFrameEmpty && packet.frameType != kFrameEmpty) {
- // Update the frame type with the first media packet.
- frame_type_ = packet.frameType;
- }
- if (packet.frameType == kFrameEmpty) {
- // Update sequence number of an empty packet.
- // Only media packets are inserted into the packet list.
- InformOfEmptyPacket(packet.seqNum);
- return 0;
- }
-
- if (packets_.size() == kMaxPacketsInSession)
- return -1;
-
- // Find the position of this packet in the packet list in sequence number
- // order and insert it. Loop over the list in reverse order.
- ReversePacketIterator rit = packets_.rbegin();
- for (; rit != packets_.rend(); ++rit)
- if (LatestSequenceNumber((*rit).seqNum, packet.seqNum, NULL) ==
- packet.seqNum)
- break;
-
- // Check for duplicate packets.
- if (rit != packets_.rend() &&
- (*rit).seqNum == packet.seqNum && (*rit).sizeBytes > 0)
- return -2;
-
- // The insert operation invalidates the iterator |rit|.
- PacketIterator packet_list_it = packets_.insert(rit.base(), packet);
-
- int returnLength = InsertBuffer(frame_buffer, packet_list_it);
- UpdateCompleteSession();
- if (enable_decodable_state)
- UpdateDecodableSession(rtt_ms);
- return returnLength;
-}
-
-void VCMSessionInfo::InformOfEmptyPacket(uint16_t seq_num) {
- // Empty packets may be FEC or filler packets. They are sequential and
- // follow the data packets, therefore, we should only keep track of the high
- // and low sequence numbers and may assume that the packets in between are
- // empty packets belonging to the same frame (timestamp).
- empty_seq_num_high_ = LatestSequenceNumber(seq_num, empty_seq_num_high_,
- NULL);
- if (empty_seq_num_low_ == -1 ||
- LatestSequenceNumber(seq_num, empty_seq_num_low_, NULL) ==
- empty_seq_num_low_)
- empty_seq_num_low_ = seq_num;
-}
-
-int VCMSessionInfo::packets_not_decodable() const {
- return packets_not_decodable_;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/session_info.h b/trunk/src/modules/video_coding/main/source/session_info.h
deleted file mode 100644
index 27533ce..0000000
--- a/trunk/src/modules/video_coding/main/source/session_info.h
+++ /dev/null
@@ -1,134 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_SESSION_INFO_H_
-#define WEBRTC_MODULES_VIDEO_CODING_SESSION_INFO_H_
-
-#include <list>
-
-#include "modules/interface/module_common_types.h"
-#include "modules/video_coding/main/source/packet.h"
-#include "typedefs.h" // NOLINT(build/include)
-
-namespace webrtc {
-
-class VCMSessionInfo {
- public:
- VCMSessionInfo();
-
- void UpdateDataPointers(const uint8_t* old_base_ptr,
- const uint8_t* new_base_ptr);
- // NACK - Building the NACK lists.
- // Build hard NACK list: Zero out all entries in list up to and including
- // _lowSeqNum.
- int BuildHardNackList(int* seq_num_list,
- int seq_num_list_length);
-
- // Build soft NACK list: Zero out only a subset of the packets, discard
- // empty packets.
- int BuildSoftNackList(int* seq_num_list,
- int seq_num_list_length,
- int rtt_ms);
- void Reset();
- int InsertPacket(const VCMPacket& packet,
- uint8_t* frame_buffer,
- bool enable_decodable_state,
- int rtt_ms);
- bool complete() const;
- bool decodable() const;
-
- // Builds fragmentation headers for VP8, each fragment being a decodable
- // VP8 partition. Returns the total number of bytes which are decodable. Is
- // used instead of MakeDecodable for VP8.
- int BuildVP8FragmentationHeader(uint8_t* frame_buffer,
- int frame_buffer_length,
- RTPFragmentationHeader* fragmentation);
-
- // Makes the frame decodable. I.e., only contain decodable NALUs. All
- // non-decodable NALUs will be deleted and packets will be moved to in
- // memory to remove any empty space.
- // Returns the number of bytes deleted from the session.
- int MakeDecodable();
- int SessionLength() const;
- bool HaveLastPacket() const;
- bool session_nack() const;
- webrtc::FrameType FrameType() const { return frame_type_; }
- int LowSequenceNumber() const;
-
- // Returns highest sequence number, media or empty.
- int HighSequenceNumber() const;
- int PictureId() const;
- int TemporalId() const;
- bool LayerSync() const;
- int Tl0PicId() const;
- bool NonReference() const;
- void SetPreviousFrameLoss() { previous_frame_loss_ = true; }
- bool PreviousFrameLoss() const { return previous_frame_loss_; }
-
- // The number of packets discarded because the decoder can't make use of
- // them.
- int packets_not_decodable() const;
-
- private:
- enum { kMaxVP8Partitions = 9 };
-
- typedef std::list<VCMPacket> PacketList;
- typedef PacketList::iterator PacketIterator;
- typedef PacketList::const_iterator PacketIteratorConst;
- typedef PacketList::reverse_iterator ReversePacketIterator;
-
- void InformOfEmptyPacket(uint16_t seq_num);
-
- // Finds the packet of the beginning of the next VP8 partition. If
- // none is found the returned iterator points to |packets_.end()|.
- // |it| is expected to point to the last packet of the previous partition,
- // or to the first packet of the frame. |packets_skipped| is incremented
- // for each packet found which doesn't have the beginning bit set.
- PacketIterator FindNextPartitionBeginning(PacketIterator it,
- int* packets_skipped) const;
-
- // Returns an iterator pointing to the last packet of the partition pointed to
- // by |it|.
- PacketIterator FindPartitionEnd(PacketIterator it) const;
- static bool InSequence(const PacketIterator& it,
- const PacketIterator& prev_it);
- static int PacketsMissing(const PacketIterator& packet_it,
- const PacketIterator& prev_packet_it);
- int InsertBuffer(uint8_t* frame_buffer,
- PacketIterator packetIterator);
- void ShiftSubsequentPackets(PacketIterator it, int steps_to_shift);
- PacketIterator FindNaluEnd(PacketIterator packet_iter) const;
- // Deletes the data of all packets between |start| and |end|, inclusively.
- // Note that this function doesn't delete the actual packets.
- int DeletePacketData(PacketIterator start,
- PacketIterator end);
- void UpdateCompleteSession();
-
- // When enabled, determine if session is decodable, i.e. incomplete but
- // would be sent to the decoder.
- void UpdateDecodableSession(int rtt_ms);
-
- // If this session has been NACKed by the jitter buffer.
- bool session_nack_;
- bool complete_;
- bool decodable_;
- webrtc::FrameType frame_type_;
- bool previous_frame_loss_;
- // Packets in this frame.
- PacketList packets_;
- int empty_seq_num_low_;
- int empty_seq_num_high_;
- // Number of packets discarded because the decoder can't use them.
- int packets_not_decodable_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_SESSION_INFO_H_
diff --git a/trunk/src/modules/video_coding/main/source/session_info_unittest.cc b/trunk/src/modules/video_coding/main/source/session_info_unittest.cc
deleted file mode 100644
index e017735..0000000
--- a/trunk/src/modules/video_coding/main/source/session_info_unittest.cc
+++ /dev/null
@@ -1,931 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string.h>
-
-#include "gtest/gtest.h"
-#include "modules/interface/module_common_types.h"
-#include "modules/video_coding/main/source/packet.h"
-#include "modules/video_coding/main/source/session_info.h"
-
-namespace webrtc {
-
-class TestSessionInfo : public ::testing::Test {
- protected:
- enum { kPacketBufferSize = 10 };
- enum { kFrameBufferSize = 10 * kPacketBufferSize };
-
- virtual void SetUp() {
- memset(packet_buffer_, 0, kPacketBufferSize);
- memset(frame_buffer_, 0, kFrameBufferSize);
- session_.Reset();
- packet_.Reset();
- packet_.frameType = kVideoFrameDelta;
- packet_.sizeBytes = kPacketBufferSize;
- packet_.dataPtr = packet_buffer_;
- packet_.seqNum = 0;
- packet_.timestamp = 0;
- }
-
- void FillPacket(uint8_t start_value) {
- for (int i = 0; i < kPacketBufferSize; ++i)
- packet_buffer_[i] = start_value + i;
- }
-
- void VerifyPacket(uint8_t* start_ptr, uint8_t start_value) {
- for (int j = 0; j < kPacketBufferSize; ++j) {
- ASSERT_EQ(start_value + j, start_ptr[j]);
- }
- }
-
- uint8_t packet_buffer_[kPacketBufferSize];
- uint8_t frame_buffer_[kFrameBufferSize];
- VCMSessionInfo session_;
- VCMPacket packet_;
-};
-
-class TestVP8Partitions : public TestSessionInfo {
- protected:
- enum { kMaxVP8Partitions = 9 };
-
- virtual void SetUp() {
- TestSessionInfo::SetUp();
- vp8_header_ = &packet_header_.type.Video.codecHeader.VP8;
- packet_header_.frameType = kVideoFrameDelta;
- packet_header_.type.Video.codec = kRTPVideoVP8;
- vp8_header_->InitRTPVideoHeaderVP8();
- fragmentation_.VerifyAndAllocateFragmentationHeader(kMaxVP8Partitions);
- }
-
- bool VerifyPartition(int partition_id,
- int packets_expected,
- int start_value) {
- EXPECT_EQ(static_cast<uint32_t>(packets_expected * kPacketBufferSize),
- fragmentation_.fragmentationLength[partition_id]);
- for (int i = 0; i < packets_expected; ++i) {
- int packet_index = fragmentation_.fragmentationOffset[partition_id] +
- i * kPacketBufferSize;
- if (packet_index + kPacketBufferSize > kFrameBufferSize)
- return false;
- VerifyPacket(frame_buffer_ + packet_index, start_value + i);
- }
- return true;
- }
-
- WebRtcRTPHeader packet_header_;
- RTPVideoHeaderVP8* vp8_header_;
- RTPFragmentationHeader fragmentation_;
-};
-
-class TestNalUnits : public TestSessionInfo {
- protected:
- virtual void SetUp() {
- TestSessionInfo::SetUp();
- packet_.codec = kVideoCodecVP8;
- }
-
- bool VerifyNalu(int offset, int packets_expected, int start_value) {
- EXPECT_GE(session_.SessionLength(),
- packets_expected * kPacketBufferSize);
- for (int i = 0; i < packets_expected; ++i) {
- int packet_index = offset * kPacketBufferSize + i * kPacketBufferSize;
- VerifyPacket(frame_buffer_ + packet_index, start_value + i);
- }
- return true;
- }
-};
-
-class TestNackList : public TestSessionInfo {
- protected:
- enum { kMaxSeqNumListLength = 30 };
-
- virtual void SetUp() {
- TestSessionInfo::SetUp();
- seq_num_list_length_ = 0;
- memset(seq_num_list_, 0, sizeof(seq_num_list_));
- }
-
- void BuildSeqNumList(uint16_t low,
- uint16_t high) {
- int i = 0;
- while (low != high + 1) {
- EXPECT_LT(i, kMaxSeqNumListLength);
- if (i >= kMaxSeqNumListLength) {
- seq_num_list_length_ = kMaxSeqNumListLength;
- return;
- }
- seq_num_list_[i] = low;
- low++;
- i++;
- }
- seq_num_list_length_ = i;
- }
-
- void VerifyAll(int value) {
- for (int i = 0; i < seq_num_list_length_; ++i)
- EXPECT_EQ(seq_num_list_[i], value);
- }
-
- int seq_num_list_[kMaxSeqNumListLength];
- int seq_num_list_length_;
-};
-
-TEST_F(TestSessionInfo, TestSimpleAPIs) {
- packet_.isFirstPacket = true;
- packet_.seqNum = 0xFFFE;
- packet_.sizeBytes = kPacketBufferSize;
- packet_.frameType = kVideoFrameKey;
- FillPacket(0);
- ASSERT_EQ(kPacketBufferSize,
- session_.InsertPacket(packet_, frame_buffer_, false, 0));
- EXPECT_FALSE(session_.HaveLastPacket());
- EXPECT_EQ(kVideoFrameKey, session_.FrameType());
-
- packet_.isFirstPacket = false;
- packet_.markerBit = true;
- packet_.seqNum += 1;
- ASSERT_EQ(kPacketBufferSize,
- session_.InsertPacket(packet_, frame_buffer_, false, 0));
- EXPECT_TRUE(session_.HaveLastPacket());
- EXPECT_EQ(packet_.seqNum, session_.HighSequenceNumber());
- EXPECT_EQ(0xFFFE, session_.LowSequenceNumber());
-
- // Insert empty packet which will be the new high sequence number.
- // To make things more difficult we will make sure to have a wrap here.
- packet_.isFirstPacket = false;
- packet_.markerBit = true;
- packet_.seqNum = 2;
- packet_.sizeBytes = 0;
- packet_.frameType = kFrameEmpty;
- ASSERT_EQ(0,
- session_.InsertPacket(packet_, frame_buffer_, false, 0));
- EXPECT_EQ(packet_.seqNum, session_.HighSequenceNumber());
-}
-
-TEST_F(TestSessionInfo, NormalOperation) {
- packet_.seqNum = 0xFFFF;
- packet_.isFirstPacket = true;
- packet_.markerBit = false;
- FillPacket(0);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- packet_.isFirstPacket = false;
- for (int i = 1; i < 9; ++i) {
- packet_.seqNum += 1;
- FillPacket(i);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
- }
-
- packet_.seqNum += 1;
- packet_.markerBit = true;
- FillPacket(9);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- EXPECT_EQ(0, session_.packets_not_decodable());
- EXPECT_EQ(10 * kPacketBufferSize, session_.SessionLength());
- for (int i = 0; i < 10; ++i) {
- SCOPED_TRACE("Calling VerifyPacket");
- VerifyPacket(frame_buffer_ + i * kPacketBufferSize, i);
- }
-}
-
-TEST_F(TestVP8Partitions, TwoPartitionsOneLoss) {
- // Partition 0 | Partition 1
- // [ 0 ] [ 2 ] | [ 3 ]
- packet_header_.type.Video.isFirstPacket = true;
- vp8_header_->beginningOfPartition = true;
- vp8_header_->partitionId = 0;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber = 0;
- FillPacket(0);
- VCMPacket* packet = new VCMPacket(packet_buffer_, kPacketBufferSize,
- packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 0;
- vp8_header_->beginningOfPartition = false;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber += 2;
- FillPacket(2);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 1;
- vp8_header_->beginningOfPartition = true;
- packet_header_.header.markerBit = true;
- packet_header_.header.sequenceNumber += 1;
- FillPacket(3);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- // One packet should be removed (end of partition 0).
- EXPECT_EQ(session_.BuildVP8FragmentationHeader(frame_buffer_,
- kFrameBufferSize,
- &fragmentation_),
- 2*kPacketBufferSize);
- SCOPED_TRACE("Calling VerifyPartition");
- EXPECT_TRUE(VerifyPartition(0, 1, 0));
- SCOPED_TRACE("Calling VerifyPartition");
- EXPECT_TRUE(VerifyPartition(1, 1, 3));
-}
-
-TEST_F(TestVP8Partitions, TwoPartitionsOneLoss2) {
- // Partition 0 | Partition 1
- // [ 1 ] [ 2 ] | [ 3 ] [ 5 ]
- packet_header_.type.Video.isFirstPacket = true;
- vp8_header_->beginningOfPartition = true;
- vp8_header_->partitionId = 0;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber = 1;
- FillPacket(1);
- VCMPacket* packet = new VCMPacket(packet_buffer_, kPacketBufferSize,
- packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0)
- , kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 0;
- vp8_header_->beginningOfPartition = false;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber += 1;
- FillPacket(2);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 1;
- vp8_header_->beginningOfPartition = true;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber += 1;
- FillPacket(3);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 1;
- vp8_header_->beginningOfPartition = false;
- packet_header_.header.markerBit = true;
- packet_header_.header.sequenceNumber += 2;
- FillPacket(5);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- // One packet should be removed (end of partition 2), 3 left.
- EXPECT_EQ(session_.BuildVP8FragmentationHeader(frame_buffer_,
- kFrameBufferSize,
- &fragmentation_),
- 3*kPacketBufferSize);
- SCOPED_TRACE("Calling VerifyPartition");
- EXPECT_TRUE(VerifyPartition(0, 2, 1));
- SCOPED_TRACE("Calling VerifyPartition");
- EXPECT_TRUE(VerifyPartition(1, 1, 3));
- EXPECT_EQ(1, session_.packets_not_decodable());
-}
-
-TEST_F(TestVP8Partitions, TwoPartitionsNoLossWrap) {
- // Partition 0 | Partition 1
- // [ fffd ] [ fffe ] | [ ffff ] [ 0 ]
- packet_header_.type.Video.isFirstPacket = true;
- vp8_header_->beginningOfPartition = true;
- vp8_header_->partitionId = 0;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber = 0xfffd;
- FillPacket(0);
- VCMPacket* packet = new VCMPacket(packet_buffer_, kPacketBufferSize,
- packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 0;
- vp8_header_->beginningOfPartition = false;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber += 1;
- FillPacket(1);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 1;
- vp8_header_->beginningOfPartition = true;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber += 1;
- FillPacket(2);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 1;
- vp8_header_->beginningOfPartition = false;
- packet_header_.header.markerBit = true;
- packet_header_.header.sequenceNumber += 1;
- FillPacket(3);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- // No packet should be removed.
- EXPECT_EQ(session_.BuildVP8FragmentationHeader(frame_buffer_,
- kFrameBufferSize,
- &fragmentation_),
- 4*kPacketBufferSize);
- SCOPED_TRACE("Calling VerifyPartition");
- EXPECT_TRUE(VerifyPartition(0, 2, 0));
- SCOPED_TRACE("Calling VerifyPartition");
- EXPECT_TRUE(VerifyPartition(1, 2, 2));
- EXPECT_EQ(0, session_.packets_not_decodable());
-}
-
-TEST_F(TestVP8Partitions, TwoPartitionsLossWrap) {
- // Partition 0 | Partition 1
- // [ fffd ] [ fffe ] | [ ffff ] [ 1 ]
- packet_header_.type.Video.isFirstPacket = true;
- vp8_header_->beginningOfPartition = true;
- vp8_header_->partitionId = 0;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber = 0xfffd;
- FillPacket(0);
- VCMPacket* packet = new VCMPacket(packet_buffer_, kPacketBufferSize,
- packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 0;
- vp8_header_->beginningOfPartition = false;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber += 1;
- FillPacket(1);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 1;
- vp8_header_->beginningOfPartition = true;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber += 1;
- FillPacket(2);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 1;
- vp8_header_->beginningOfPartition = false;
- packet_header_.header.markerBit = true;
- packet_header_.header.sequenceNumber += 2;
- FillPacket(3);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- // One packet should be removed from the last partition
- EXPECT_EQ(session_.BuildVP8FragmentationHeader(frame_buffer_,
- kFrameBufferSize,
- &fragmentation_),
- 3*kPacketBufferSize);
- SCOPED_TRACE("Calling VerifyPartition");
- EXPECT_TRUE(VerifyPartition(0, 2, 0));
- SCOPED_TRACE("Calling VerifyPartition");
- EXPECT_TRUE(VerifyPartition(1, 1, 2));
- EXPECT_EQ(1, session_.packets_not_decodable());
-}
-
-
-TEST_F(TestVP8Partitions, ThreePartitionsOneMissing) {
- // Partition 1 |Partition 2 | Partition 3
- // [ 1 ] [ 2 ] | | [ 5 ] | [ 6 ]
- packet_header_.type.Video.isFirstPacket = true;
- vp8_header_->beginningOfPartition = true;
- vp8_header_->partitionId = 0;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber = 1;
- FillPacket(1);
- VCMPacket* packet = new VCMPacket(packet_buffer_, kPacketBufferSize,
- packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 0;
- vp8_header_->beginningOfPartition = false;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber += 1;
- FillPacket(2);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 2;
- vp8_header_->beginningOfPartition = true;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber += 3;
- FillPacket(5);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 2;
- vp8_header_->beginningOfPartition = false;
- packet_header_.header.markerBit = true;
- packet_header_.header.sequenceNumber += 1;
- FillPacket(6);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- // No packet should be removed.
- EXPECT_EQ(session_.BuildVP8FragmentationHeader(frame_buffer_,
- kFrameBufferSize,
- &fragmentation_),
- 4*kPacketBufferSize);
- SCOPED_TRACE("Calling VerifyPartition");
- EXPECT_TRUE(VerifyPartition(0, 2, 1));
- SCOPED_TRACE("Calling VerifyPartition");
- EXPECT_TRUE(VerifyPartition(2, 2, 5));
- EXPECT_EQ(0, session_.packets_not_decodable());
-}
-
-TEST_F(TestVP8Partitions, ThreePartitionsLossInSecond) {
- // Partition 0 |Partition 1 | Partition 2
- // [ 1 ] [ 2 ] | [ 4 ] [ 5 ] | [ 6 ] [ 7 ]
- packet_header_.type.Video.isFirstPacket = true;
- vp8_header_->beginningOfPartition = true;
- vp8_header_->partitionId = 0;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber = 1;
- FillPacket(1);
- VCMPacket* packet = new VCMPacket(packet_buffer_, kPacketBufferSize,
- packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 0;
- vp8_header_->beginningOfPartition = false;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber += 1;
- FillPacket(2);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 1;
- vp8_header_->beginningOfPartition = false;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber += 2;
- FillPacket(4);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 1;
- vp8_header_->beginningOfPartition = false;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber += 1;
- FillPacket(5);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 2;
- vp8_header_->beginningOfPartition = true;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber += 1;
- FillPacket(6);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 2;
- vp8_header_->beginningOfPartition = false;
- packet_header_.header.markerBit = true;
- packet_header_.header.sequenceNumber += 1;
- FillPacket(7);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- // 2 partitions left. 2 packets removed from second partition
- EXPECT_EQ(session_.BuildVP8FragmentationHeader(frame_buffer_,
- kFrameBufferSize,
- &fragmentation_),
- 4*kPacketBufferSize);
- SCOPED_TRACE("Calling VerifyPartition");
- EXPECT_TRUE(VerifyPartition(0, 2, 1));
- SCOPED_TRACE("Calling VerifyPartition");
- EXPECT_TRUE(VerifyPartition(2, 2, 6));
- EXPECT_EQ(2, session_.packets_not_decodable());
-}
-
-TEST_F(TestVP8Partitions, AggregationOverTwoPackets) {
- // Partition 0 | Partition 1 | Partition 2
- // [ 0 | ] [ 1 ] | [ 2 ]
- packet_header_.type.Video.isFirstPacket = true;
- vp8_header_->beginningOfPartition = true;
- vp8_header_->partitionId = 0;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber = 0;
- FillPacket(0);
- VCMPacket* packet = new VCMPacket(packet_buffer_, kPacketBufferSize,
- packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 1;
- vp8_header_->beginningOfPartition = false;
- packet_header_.header.markerBit = false;
- packet_header_.header.sequenceNumber += 1;
- FillPacket(1);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- packet_header_.type.Video.isFirstPacket = false;
- vp8_header_->partitionId = 2;
- vp8_header_->beginningOfPartition = true;
- packet_header_.header.markerBit = true;
- packet_header_.header.sequenceNumber += 1;
- FillPacket(2);
- packet = new VCMPacket(packet_buffer_, kPacketBufferSize, packet_header_);
- ASSERT_EQ(session_.InsertPacket(*packet, frame_buffer_, false, 0),
- kPacketBufferSize);
- delete packet;
-
- // No packets removed.
- EXPECT_EQ(session_.BuildVP8FragmentationHeader(frame_buffer_,
- kFrameBufferSize,
- &fragmentation_),
- 3*kPacketBufferSize);
- EXPECT_EQ(0, session_.packets_not_decodable());
- SCOPED_TRACE("Calling VerifyPartition");
- EXPECT_TRUE(VerifyPartition(0, 2, 0));
- // This partition is aggregated in partition 0
- SCOPED_TRACE("Calling VerifyPartition");
- EXPECT_TRUE(VerifyPartition(1, 0, 0));
- SCOPED_TRACE("Calling VerifyPartition");
- EXPECT_TRUE(VerifyPartition(2, 1, 2));
-}
-
-TEST_F(TestNalUnits, OnlyReceivedEmptyPacket) {
- packet_.isFirstPacket = false;
- packet_.completeNALU = kNaluComplete;
- packet_.frameType = kFrameEmpty;
- packet_.sizeBytes = 0;
- packet_.seqNum = 0;
- packet_.markerBit = false;
- ASSERT_EQ(0, session_.InsertPacket(packet_, frame_buffer_, false, 0));
-
- EXPECT_EQ(0, session_.MakeDecodable());
- EXPECT_EQ(0, session_.SessionLength());
- EXPECT_EQ(0, session_.packets_not_decodable());
-}
-
-TEST_F(TestNalUnits, OneIsolatedNaluLoss) {
- packet_.isFirstPacket = true;
- packet_.completeNALU = kNaluComplete;
- packet_.seqNum = 0;
- packet_.markerBit = false;
- FillPacket(0);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- packet_.isFirstPacket = false;
- packet_.completeNALU = kNaluComplete;
- packet_.seqNum += 2;
- packet_.markerBit = true;
- FillPacket(2);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- EXPECT_EQ(0, session_.MakeDecodable());
- EXPECT_EQ(2 * kPacketBufferSize, session_.SessionLength());
- EXPECT_EQ(0, session_.packets_not_decodable());
- SCOPED_TRACE("Calling VerifyNalu");
- EXPECT_TRUE(VerifyNalu(0, 1, 0));
- SCOPED_TRACE("Calling VerifyNalu");
- EXPECT_TRUE(VerifyNalu(1, 1, 2));
-}
-
-TEST_F(TestNalUnits, LossInMiddleOfNalu) {
- packet_.isFirstPacket = true;
- packet_.completeNALU = kNaluComplete;
- packet_.seqNum = 0;
- packet_.markerBit = false;
- FillPacket(0);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- packet_.isFirstPacket = false;
- packet_.completeNALU = kNaluEnd;
- packet_.seqNum += 2;
- packet_.markerBit = true;
- FillPacket(2);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- EXPECT_EQ(kPacketBufferSize, session_.MakeDecodable());
- EXPECT_EQ(kPacketBufferSize, session_.SessionLength());
- EXPECT_EQ(1, session_.packets_not_decodable());
- SCOPED_TRACE("Calling VerifyNalu");
- EXPECT_TRUE(VerifyNalu(0, 1, 0));
-}
-
-TEST_F(TestNalUnits, StartAndEndOfLastNalUnitLost) {
- packet_.isFirstPacket = true;
- packet_.completeNALU = kNaluComplete;
- packet_.seqNum = 0;
- packet_.markerBit = false;
- FillPacket(0);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- packet_.isFirstPacket = false;
- packet_.completeNALU = kNaluIncomplete;
- packet_.seqNum += 2;
- packet_.markerBit = false;
- FillPacket(1);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- EXPECT_EQ(kPacketBufferSize, session_.MakeDecodable());
- EXPECT_EQ(kPacketBufferSize, session_.SessionLength());
- EXPECT_EQ(1, session_.packets_not_decodable());
- SCOPED_TRACE("Calling VerifyNalu");
- EXPECT_TRUE(VerifyNalu(0, 1, 0));
-}
-
-TEST_F(TestNalUnits, ReorderWrapNoLoss) {
- packet_.seqNum = 0xFFFF;
- packet_.isFirstPacket = false;
- packet_.completeNALU = kNaluIncomplete;
- packet_.seqNum += 1;
- packet_.markerBit = false;
- FillPacket(1);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- packet_.isFirstPacket = true;
- packet_.completeNALU = kNaluComplete;
- packet_.seqNum -= 1;
- packet_.markerBit = false;
- FillPacket(0);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- packet_.isFirstPacket = false;
- packet_.completeNALU = kNaluEnd;
- packet_.seqNum += 2;
- packet_.markerBit = true;
- FillPacket(2);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- EXPECT_EQ(0, session_.MakeDecodable());
- EXPECT_EQ(0, session_.packets_not_decodable());
- EXPECT_EQ(3*kPacketBufferSize, session_.SessionLength());
- SCOPED_TRACE("Calling VerifyNalu");
- EXPECT_TRUE(VerifyNalu(0, 1, 0));
-}
-
-TEST_F(TestNalUnits, WrapLosses) {
- packet_.seqNum = 0xFFFF;
- packet_.isFirstPacket = false;
- packet_.completeNALU = kNaluIncomplete;
- packet_.markerBit = false;
- FillPacket(1);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- packet_.isFirstPacket = false;
- packet_.completeNALU = kNaluEnd;
- packet_.seqNum += 2;
- packet_.markerBit = true;
- FillPacket(2);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- EXPECT_EQ(2 * kPacketBufferSize, session_.MakeDecodable());
- EXPECT_EQ(0, session_.SessionLength());
- EXPECT_EQ(2, session_.packets_not_decodable());
-}
-
-TEST_F(TestNalUnits, ReorderWrapLosses) {
- packet_.seqNum = 0xFFFF;
-
- packet_.isFirstPacket = false;
- packet_.completeNALU = kNaluEnd;
- packet_.seqNum += 2;
- packet_.markerBit = true;
- FillPacket(2);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- packet_.seqNum -= 2;
- packet_.isFirstPacket = false;
- packet_.completeNALU = kNaluIncomplete;
- packet_.markerBit = false;
- FillPacket(1);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- EXPECT_EQ(2 * kPacketBufferSize, session_.MakeDecodable());
- EXPECT_EQ(0, session_.SessionLength());
- EXPECT_EQ(2, session_.packets_not_decodable());
-}
-
-TEST_F(TestNackList, NoLosses) {
- uint16_t low = 0xFFFF - 5;
-
- packet_.seqNum = low;
- packet_.isFirstPacket = true;
- packet_.markerBit = false;
- FillPacket(0);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- for (int i = 1; i < 9; ++i) {
- packet_.seqNum += 1;
- packet_.isFirstPacket = false;
- packet_.markerBit = false;
- FillPacket(i + 1);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
- }
-
- packet_.seqNum += 1;
- packet_.isFirstPacket = false;
- packet_.markerBit = true;
- FillPacket(10);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- EXPECT_EQ(10 * kPacketBufferSize, session_.SessionLength());
- BuildSeqNumList(low, packet_.seqNum);
- EXPECT_EQ(0, session_.BuildHardNackList(seq_num_list_, seq_num_list_length_));
- EXPECT_FALSE(session_.session_nack());
- SCOPED_TRACE("Calling VerifyAll");
- VerifyAll(-1);
-
- BuildSeqNumList(low, packet_.seqNum);
- EXPECT_EQ(0, session_.BuildSoftNackList(seq_num_list_, seq_num_list_length_,
- 60));
- SCOPED_TRACE("Calling VerifyAll");
- VerifyAll(-1);
-}
-
-TEST_F(TestNackList, FiveLossesSpreadOut) {
- uint16_t low = 0xFFFF - 5;
-
- packet_.seqNum = low;
- packet_.isFirstPacket = false;
- packet_.markerBit = true;
- FillPacket(0);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- for (int i = 1; i < 9; ++i) {
- packet_.seqNum += 1;
- packet_.isFirstPacket = false;
- packet_.markerBit = false;
- FillPacket(i);
- if ((i + 1) % 2)
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
- }
-
- packet_.seqNum++; // Simulate loss of last packet.
-
- EXPECT_EQ(5 * kPacketBufferSize, session_.SessionLength());
- BuildSeqNumList(low, packet_.seqNum);
- EXPECT_EQ(0, session_.BuildHardNackList(seq_num_list_, seq_num_list_length_));
- for (int i = 0; i < seq_num_list_length_; ++i) {
- if (i % 2)
- EXPECT_EQ(static_cast<uint16_t>(low + i), seq_num_list_[i]);
- else
- EXPECT_EQ(-1, seq_num_list_[i]);
- }
-
- BuildSeqNumList(low, packet_.seqNum);
- EXPECT_EQ(0, session_.BuildSoftNackList(seq_num_list_, seq_num_list_length_,
- 60));
- EXPECT_EQ(true, session_.session_nack());
- for (int i = 0; i < seq_num_list_length_; ++i) {
- if (i % 2)
- EXPECT_EQ(static_cast<uint16_t>(low + i), seq_num_list_[i]);
- else
- EXPECT_EQ(-1, seq_num_list_[i]);
- }
-}
-
-TEST_F(TestNackList, FirstAndLastLost) {
- uint16_t low = 0xFFFF;
-
- packet_.seqNum = low + 1;
- packet_.isFirstPacket = false;
- packet_.markerBit = false;
- FillPacket(0);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0),
- kPacketBufferSize);
-
- EXPECT_EQ(kPacketBufferSize, session_.SessionLength());
- BuildSeqNumList(low, packet_.seqNum + 1);
- EXPECT_EQ(0, session_.BuildHardNackList(seq_num_list_, seq_num_list_length_));
- EXPECT_EQ(0xFFFF, seq_num_list_[0]);
- EXPECT_EQ(-1, seq_num_list_[1]);
- EXPECT_EQ(1, seq_num_list_[2]);
-
- BuildSeqNumList(low, packet_.seqNum + 1);
- EXPECT_EQ(0, session_.BuildSoftNackList(seq_num_list_,seq_num_list_length_,
- 60));
- EXPECT_EQ(true, session_.session_nack());
- EXPECT_EQ(0xFFFF, seq_num_list_[0]);
- EXPECT_EQ(-1, seq_num_list_[1]);
- EXPECT_EQ(1, seq_num_list_[2]);
-}
-
-TEST_F(TestNackList, LostAllButEmptyPackets) {
- uint16_t low = 0;
- packet_.seqNum = low + 1;
- packet_.isFirstPacket = false;
- packet_.markerBit = false;
- packet_.frameType = kFrameEmpty;
- packet_.sizeBytes = 0;
- FillPacket(0);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0), 0);
-
- packet_.seqNum = low + 3;
- packet_.isFirstPacket = false;
- packet_.markerBit = false;
- packet_.frameType = kFrameEmpty;
- packet_.sizeBytes = 0;
- FillPacket(0);
- ASSERT_EQ(session_.InsertPacket(packet_, frame_buffer_, false, 0), 0);
-
- EXPECT_EQ(0, session_.SessionLength());
- BuildSeqNumList(low, packet_.seqNum + 1);
- EXPECT_EQ(0, session_.BuildSoftNackList(seq_num_list_, seq_num_list_length_,
- 60));
- EXPECT_EQ(true, session_.session_nack());
- EXPECT_EQ(0, seq_num_list_[0]);
- EXPECT_EQ(-1, seq_num_list_[1]);
- EXPECT_EQ(-2, seq_num_list_[2]);
- EXPECT_EQ(-2, seq_num_list_[3]);
- EXPECT_EQ(4, seq_num_list_[4]);
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/tick_time_base.h b/trunk/src/modules/video_coding/main/source/tick_time_base.h
deleted file mode 100644
index a212591..0000000
--- a/trunk/src/modules/video_coding/main/source/tick_time_base.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TICK_TIME_BASE_H_
-#define WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TICK_TIME_BASE_H_
-
-#include "system_wrappers/interface/tick_util.h"
-
-namespace webrtc {
-
-// This class provides a mockable wrapper to TickTime.
-class TickTimeBase {
- public:
- virtual ~TickTimeBase() {}
-
- // "Now" in milliseconds.
- virtual int64_t MillisecondTimestamp() const {
- return TickTime::MillisecondTimestamp();
- }
-
- // "Now" in microseconds.
- virtual int64_t MicrosecondTimestamp() const {
- return TickTime::MicrosecondTimestamp();
- }
-};
-
-} // namespace
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_MAIN_SOURCE_TICK_TIME_BASE_H_
diff --git a/trunk/src/modules/video_coding/main/source/timestamp_extrapolator.cc b/trunk/src/modules/video_coding/main/source/timestamp_extrapolator.cc
deleted file mode 100644
index e272eb9..0000000
--- a/trunk/src/modules/video_coding/main/source/timestamp_extrapolator.cc
+++ /dev/null
@@ -1,262 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "internal_defines.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
-#include "timestamp_extrapolator.h"
-#include "trace.h"
-
-namespace webrtc {
-
-VCMTimestampExtrapolator::VCMTimestampExtrapolator(TickTimeBase* clock,
- WebRtc_Word32 vcmId,
- WebRtc_Word32 id)
-:
-_rwLock(RWLockWrapper::CreateRWLock()),
-_vcmId(vcmId),
-_id(id),
-_clock(clock),
-_startMs(0),
-_firstTimestamp(0),
-_wrapArounds(0),
-_prevTs90khz(0),
-_lambda(1),
-_firstAfterReset(true),
-_packetCount(0),
-_startUpFilterDelayInPackets(2),
-_detectorAccumulatorPos(0),
-_detectorAccumulatorNeg(0),
-_alarmThreshold(60e3),
-_accDrift(6600), // in timestamp ticks, i.e. 15 ms
-_accMaxError(7000),
-_P11(1e10)
-{
- Reset(_clock->MillisecondTimestamp());
-}
-
-VCMTimestampExtrapolator::~VCMTimestampExtrapolator()
-{
- delete _rwLock;
-}
-
-void
-VCMTimestampExtrapolator::Reset(const WebRtc_Word64 nowMs /* = -1 */)
-{
- WriteLockScoped wl(*_rwLock);
- if (nowMs > -1)
- {
- _startMs = nowMs;
- }
- else
- {
- _startMs = _clock->MillisecondTimestamp();
- }
- _prevMs = _startMs;
- _firstTimestamp = 0;
- _w[0] = 90.0;
- _w[1] = 0;
- _P[0][0] = 1;
- _P[1][1] = _P11;
- _P[0][1] = _P[1][0] = 0;
- _firstAfterReset = true;
- _prevTs90khz = 0;
- _wrapArounds = 0;
- _packetCount = 0;
- _detectorAccumulatorPos = 0;
- _detectorAccumulatorNeg = 0;
-}
-
-void
-VCMTimestampExtrapolator::Update(WebRtc_Word64 tMs, WebRtc_UWord32 ts90khz, bool trace)
-{
-
- _rwLock->AcquireLockExclusive();
- if (tMs - _prevMs > 10e3)
- {
- // Ten seconds without a complete frame.
- // Reset the extrapolator
- _rwLock->ReleaseLockExclusive();
- Reset();
- _rwLock->AcquireLockExclusive();
- }
- else
- {
- _prevMs = tMs;
- }
-
- // Remove offset to prevent badly scaled matrices
- tMs -= _startMs;
-
- WebRtc_Word32 prevWrapArounds = _wrapArounds;
- CheckForWrapArounds(ts90khz);
- WebRtc_Word32 wrapAroundsSincePrev = _wrapArounds - prevWrapArounds;
-
- if (wrapAroundsSincePrev == 0 && ts90khz < _prevTs90khz)
- {
- _rwLock->ReleaseLockExclusive();
- return;
- }
-
- if (_firstAfterReset)
- {
- // Make an initial guess of the offset,
- // should be almost correct since tMs - _startMs
- // should about zero at this time.
- _w[1] = -_w[0] * tMs;
- _firstTimestamp = ts90khz;
- _firstAfterReset = false;
- }
-
- // Compensate for wraparounds by changing the line offset
- _w[1] = _w[1] - wrapAroundsSincePrev * ((static_cast<WebRtc_Word64>(1)<<32) - 1);
-
- double residual = (static_cast<double>(ts90khz) - _firstTimestamp) - static_cast<double>(tMs) * _w[0] - _w[1];
- if (DelayChangeDetection(residual, trace) &&
- _packetCount >= _startUpFilterDelayInPackets)
- {
- // A sudden change of average network delay has been detected.
- // Force the filter to adjust its offset parameter by changing
- // the offset uncertainty. Don't do this during startup.
- _P[1][1] = _P11;
- }
- //T = [t(k) 1]';
- //that = T'*w;
- //K = P*T/(lambda + T'*P*T);
- double K[2];
- K[0] = _P[0][0] * tMs + _P[0][1];
- K[1] = _P[1][0] * tMs + _P[1][1];
- double TPT = _lambda + tMs * K[0] + K[1];
- K[0] /= TPT;
- K[1] /= TPT;
- //w = w + K*(ts(k) - that);
- _w[0] = _w[0] + K[0] * residual;
- _w[1] = _w[1] + K[1] * residual;
- //P = 1/lambda*(P - K*T'*P);
- double p00 = 1 / _lambda * (_P[0][0] - (K[0] * tMs * _P[0][0] + K[0] * _P[1][0]));
- double p01 = 1 / _lambda * (_P[0][1] - (K[0] * tMs * _P[0][1] + K[0] * _P[1][1]));
- _P[1][0] = 1 / _lambda * (_P[1][0] - (K[1] * tMs * _P[0][0] + K[1] * _P[1][0]));
- _P[1][1] = 1 / _lambda * (_P[1][1] - (K[1] * tMs * _P[0][1] + K[1] * _P[1][1]));
- _P[0][0] = p00;
- _P[0][1] = p01;
- if (_packetCount < _startUpFilterDelayInPackets)
- {
- _packetCount++;
- }
- if (trace)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _id), "w[0]=%f w[1]=%f ts=%u tMs=%u", _w[0], _w[1], ts90khz, tMs);
- }
- _rwLock->ReleaseLockExclusive();
-}
-
-WebRtc_UWord32
-VCMTimestampExtrapolator::ExtrapolateTimestamp(WebRtc_Word64 tMs) const
-{
- ReadLockScoped rl(*_rwLock);
- WebRtc_UWord32 timestamp = 0;
- if (_packetCount == 0)
- {
- timestamp = 0;
- }
- else if (_packetCount < _startUpFilterDelayInPackets)
- {
- timestamp = static_cast<WebRtc_UWord32>(90.0 * (tMs - _prevMs) + _prevTs90khz + 0.5);
- }
- else
- {
- timestamp = static_cast<WebRtc_UWord32>(_w[0] * (tMs - _startMs) + _w[1] + _firstTimestamp + 0.5);
- }
- return timestamp;
-}
-
-WebRtc_Word64
-VCMTimestampExtrapolator::ExtrapolateLocalTime(WebRtc_UWord32 timestamp90khz) const
-{
- ReadLockScoped rl(*_rwLock);
- WebRtc_Word64 localTimeMs = 0;
- if (_packetCount == 0)
- {
- localTimeMs = -1;
- }
- else if (_packetCount < _startUpFilterDelayInPackets)
- {
- localTimeMs = _prevMs + static_cast<WebRtc_Word64>(static_cast<double>(timestamp90khz - _prevTs90khz) / 90.0 + 0.5);
- }
- else
- {
- if (_w[0] < 1e-3)
- {
- localTimeMs = _startMs;
- }
- else
- {
- double timestampDiff = static_cast<double>(timestamp90khz) - static_cast<double>(_firstTimestamp);
- localTimeMs = static_cast<WebRtc_Word64>(static_cast<double>(_startMs) + (timestampDiff - _w[1]) / _w[0] + 0.5);
- }
- }
- return localTimeMs;
-}
-
-// Investigates if the timestamp clock has overflowed since the last timestamp and
-// keeps track of the number of wrap arounds since reset.
-void
-VCMTimestampExtrapolator::CheckForWrapArounds(WebRtc_UWord32 ts90khz)
-{
- if (_prevTs90khz == 0)
- {
- _prevTs90khz = ts90khz;
- return;
- }
- if (ts90khz < _prevTs90khz)
- {
- // This difference will probably be less than -2^31 if we have had a wrap around
- // (e.g. timestamp = 1, _previousTimestamp = 2^32 - 1). Since it is casted to a Word32,
- // it should be positive.
- if (static_cast<WebRtc_Word32>(ts90khz - _prevTs90khz) > 0)
- {
- // Forward wrap around
- _wrapArounds++;
- }
- }
- // This difference will probably be less than -2^31 if we have had a backward wrap around.
- // Since it is casted to a Word32, it should be positive.
- else if (static_cast<WebRtc_Word32>(_prevTs90khz - ts90khz) > 0)
- {
- // Backward wrap around
- _wrapArounds--;
- }
- _prevTs90khz = ts90khz;
-}
-
-bool
-VCMTimestampExtrapolator::DelayChangeDetection(double error, bool trace)
-{
- // CUSUM detection of sudden delay changes
- error = (error > 0) ? VCM_MIN(error, _accMaxError) : VCM_MAX(error, -_accMaxError);
- _detectorAccumulatorPos = VCM_MAX(_detectorAccumulatorPos + error - _accDrift, (double)0);
- _detectorAccumulatorNeg = VCM_MIN(_detectorAccumulatorNeg + error + _accDrift, (double)0);
- if (_detectorAccumulatorPos > _alarmThreshold || _detectorAccumulatorNeg < -_alarmThreshold)
- {
- // Alarm
- if (trace)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _id), "g1=%f g2=%f alarm=1", _detectorAccumulatorPos, _detectorAccumulatorNeg);
- }
- _detectorAccumulatorPos = _detectorAccumulatorNeg = 0;
- return true;
- }
- if (trace)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _id), "g1=%f g2=%f alarm=0", _detectorAccumulatorPos, _detectorAccumulatorNeg);
- }
- return false;
-}
-
-}
diff --git a/trunk/src/modules/video_coding/main/source/timestamp_extrapolator.h b/trunk/src/modules/video_coding/main/source/timestamp_extrapolator.h
deleted file mode 100644
index 901d8d4..0000000
--- a/trunk/src/modules/video_coding/main/source/timestamp_extrapolator.h
+++ /dev/null
@@ -1,63 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_
-
-#include "typedefs.h"
-#include "rw_lock_wrapper.h"
-
-namespace webrtc
-{
-
-class TickTimeBase;
-
-class VCMTimestampExtrapolator
-{
-public:
- VCMTimestampExtrapolator(TickTimeBase* clock,
- WebRtc_Word32 vcmId = 0,
- WebRtc_Word32 receiverId = 0);
- ~VCMTimestampExtrapolator();
- void Update(WebRtc_Word64 tMs, WebRtc_UWord32 ts90khz, bool trace = true);
- WebRtc_UWord32 ExtrapolateTimestamp(WebRtc_Word64 tMs) const;
- WebRtc_Word64 ExtrapolateLocalTime(WebRtc_UWord32 timestamp90khz) const;
- void Reset(WebRtc_Word64 nowMs = -1);
-
-private:
- void CheckForWrapArounds(WebRtc_UWord32 ts90khz);
- bool DelayChangeDetection(double error, bool trace = true);
- RWLockWrapper* _rwLock;
- WebRtc_Word32 _vcmId;
- WebRtc_Word32 _id;
- TickTimeBase* _clock;
- double _w[2];
- double _P[2][2];
- WebRtc_Word64 _startMs;
- WebRtc_Word64 _prevMs;
- WebRtc_UWord32 _firstTimestamp;
- WebRtc_Word32 _wrapArounds;
- WebRtc_UWord32 _prevTs90khz;
- const double _lambda;
- bool _firstAfterReset;
- WebRtc_UWord32 _packetCount;
- const WebRtc_UWord32 _startUpFilterDelayInPackets;
-
- double _detectorAccumulatorPos;
- double _detectorAccumulatorNeg;
- const double _alarmThreshold;
- const double _accDrift;
- const double _accMaxError;
- const double _P11;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_EXTRAPOLATOR_H_
diff --git a/trunk/src/modules/video_coding/main/source/timestamp_map.cc b/trunk/src/modules/video_coding/main/source/timestamp_map.cc
deleted file mode 100644
index f19819b..0000000
--- a/trunk/src/modules/video_coding/main/source/timestamp_map.cc
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "timestamp_map.h"
-#include <stdlib.h>
-#include <assert.h>
-
-namespace webrtc {
-
-// Constructor. Optional parameter specifies maximum number of
-// coexisting timers.
-VCMTimestampMap::VCMTimestampMap(WebRtc_Word32 length):
- _nextAddIx(0),
- _nextPopIx(0)
-{
- if (length <= 0)
- {
- // default
- length = 10;
- }
-
- _map = new VCMTimestampDataTuple[length];
- _length = length;
-}
-
-// Destructor.
-VCMTimestampMap::~VCMTimestampMap()
-{
- delete [] _map;
-}
-
-// Empty the list of timers.
-void
-VCMTimestampMap::Reset()
-{
- _nextAddIx = 0;
- _nextPopIx = 0;
-}
-
-WebRtc_Word32
-VCMTimestampMap::Add(WebRtc_UWord32 timestamp, void* data)
-{
- _map[_nextAddIx].timestamp = timestamp;
- _map[_nextAddIx].data = data;
- _nextAddIx = (_nextAddIx + 1) % _length;
-
- if (_nextAddIx == _nextPopIx)
- {
- // Circular list full; forget oldest entry
- _nextPopIx = (_nextPopIx + 1) % _length;
- return -1;
- }
- return 0;
-}
-
-void*
-VCMTimestampMap::Pop(WebRtc_UWord32 timestamp)
-{
- while (!IsEmpty())
- {
- if (_map[_nextPopIx].timestamp == timestamp)
- {
- // found start time for this timestamp
- void* data = _map[_nextPopIx].data;
- _map[_nextPopIx].data = NULL;
- _nextPopIx = (_nextPopIx + 1) % _length;
- return data;
- }
- else if (_map[_nextPopIx].timestamp > timestamp)
- {
- // the timestamp we are looking for is not in the list
- assert(_nextPopIx < _length && _nextPopIx >= 0);
- return NULL;
- }
-
- // not in this position, check next (and forget this position)
- _nextPopIx = (_nextPopIx + 1) % _length;
- }
-
- // could not find matching timestamp in list
- assert(_nextPopIx < _length && _nextPopIx >= 0);
- return NULL;
-}
-
-// Check if no timers are currently running
-bool
-VCMTimestampMap::IsEmpty() const
-{
- return (_nextAddIx == _nextPopIx);
-}
-
-}
diff --git a/trunk/src/modules/video_coding/main/source/timestamp_map.h b/trunk/src/modules/video_coding/main/source/timestamp_map.h
deleted file mode 100644
index fd532bc..0000000
--- a/trunk/src/modules/video_coding/main/source/timestamp_map.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
-
-#include "typedefs.h"
-
-namespace webrtc
-{
-
-struct VCMTimestampDataTuple
-{
- WebRtc_UWord32 timestamp;
- void* data;
-};
-
-class VCMTimestampMap
-{
-public:
- // Constructor. Optional parameter specifies maximum number of
- // timestamps in map.
- VCMTimestampMap(const WebRtc_Word32 length = 10);
-
- // Destructor.
- ~VCMTimestampMap();
-
- // Empty the map
- void Reset();
-
- WebRtc_Word32 Add(WebRtc_UWord32 timestamp, void* data);
- void* Pop(WebRtc_UWord32 timestamp);
-
-private:
- bool IsEmpty() const;
-
- VCMTimestampDataTuple* _map;
- WebRtc_Word32 _nextAddIx;
- WebRtc_Word32 _nextPopIx;
- WebRtc_Word32 _length;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TIMESTAMP_MAP_H_
diff --git a/trunk/src/modules/video_coding/main/source/timing.cc b/trunk/src/modules/video_coding/main/source/timing.cc
deleted file mode 100644
index aca05fa..0000000
--- a/trunk/src/modules/video_coding/main/source/timing.cc
+++ /dev/null
@@ -1,337 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "trace.h"
-#include "internal_defines.h"
-#include "jitter_buffer_common.h"
-#include "timing.h"
-#include "timestamp_extrapolator.h"
-
-namespace webrtc {
-
-VCMTiming::VCMTiming(TickTimeBase* clock,
- WebRtc_Word32 vcmId,
- WebRtc_Word32 timingId,
- VCMTiming* masterTiming)
-:
-_critSect(CriticalSectionWrapper::CreateCriticalSection()),
-_vcmId(vcmId),
-_clock(clock),
-_timingId(timingId),
-_master(false),
-_tsExtrapolator(),
-_codecTimer(),
-_renderDelayMs(kDefaultRenderDelayMs),
-_minTotalDelayMs(0),
-_requiredDelayMs(0),
-_currentDelayMs(0),
-_prevFrameTimestamp(0)
-{
- if (masterTiming == NULL)
- {
- _master = true;
- _tsExtrapolator = new VCMTimestampExtrapolator(_clock, vcmId, timingId);
- }
- else
- {
- _tsExtrapolator = masterTiming->_tsExtrapolator;
- }
-}
-
-VCMTiming::~VCMTiming()
-{
- if (_master)
- {
- delete _tsExtrapolator;
- }
- delete _critSect;
-}
-
-void
-VCMTiming::Reset(WebRtc_Word64 nowMs /* = -1 */)
-{
- CriticalSectionScoped cs(_critSect);
- if (nowMs > -1)
- {
- _tsExtrapolator->Reset(nowMs);
- }
- else
- {
- _tsExtrapolator->Reset();
- }
- _codecTimer.Reset();
- _renderDelayMs = kDefaultRenderDelayMs;
- _minTotalDelayMs = 0;
- _requiredDelayMs = 0;
- _currentDelayMs = 0;
- _prevFrameTimestamp = 0;
-}
-
-void VCMTiming::ResetDecodeTime()
-{
- _codecTimer.Reset();
-}
-
-void
-VCMTiming::SetRenderDelay(WebRtc_UWord32 renderDelayMs)
-{
- CriticalSectionScoped cs(_critSect);
- _renderDelayMs = renderDelayMs;
-}
-
-void
-VCMTiming::SetMinimumTotalDelay(WebRtc_UWord32 minTotalDelayMs)
-{
- CriticalSectionScoped cs(_critSect);
- _minTotalDelayMs = minTotalDelayMs;
-}
-
-void
-VCMTiming::SetRequiredDelay(WebRtc_UWord32 requiredDelayMs)
-{
- CriticalSectionScoped cs(_critSect);
- if (requiredDelayMs != _requiredDelayMs)
- {
- if (_master)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
- "Desired jitter buffer level: %u ms", requiredDelayMs);
- }
- _requiredDelayMs = requiredDelayMs;
- }
-}
-
-void VCMTiming::UpdateCurrentDelay(WebRtc_UWord32 frameTimestamp)
-{
- CriticalSectionScoped cs(_critSect);
- WebRtc_UWord32 targetDelayMs = TargetDelayInternal();
-
- // Make sure we try to sync with audio
- if (targetDelayMs < _minTotalDelayMs)
- {
- targetDelayMs = _minTotalDelayMs;
- }
-
- if (_currentDelayMs == 0)
- {
- // Not initialized, set current delay to target.
- _currentDelayMs = targetDelayMs;
- }
- else if (targetDelayMs != _currentDelayMs)
- {
- WebRtc_Word64 delayDiffMs = static_cast<WebRtc_Word64>(targetDelayMs) -
- _currentDelayMs;
- // Never change the delay with more than 100 ms every second. If we're changing the
- // delay in too large steps we will get noticable freezes. By limiting the change we
- // can increase the delay in smaller steps, which will be experienced as the video is
- // played in slow motion. When lowering the delay the video will be played at a faster
- // pace.
- WebRtc_Word64 maxChangeMs = 0;
- if (frameTimestamp < 0x0000ffff && _prevFrameTimestamp > 0xffff0000)
- {
- // wrap
- maxChangeMs = kDelayMaxChangeMsPerS * (frameTimestamp +
- (static_cast<WebRtc_Word64>(1)<<32) - _prevFrameTimestamp) / 90000;
- }
- else
- {
- maxChangeMs = kDelayMaxChangeMsPerS *
- (frameTimestamp - _prevFrameTimestamp) / 90000;
- }
- if (maxChangeMs <= 0)
- {
- // Any changes less than 1 ms are truncated and
- // will be postponed. Negative change will be due
- // to reordering and should be ignored.
- return;
- }
- else if (delayDiffMs < -maxChangeMs)
- {
- delayDiffMs = -maxChangeMs;
- }
- else if (delayDiffMs > maxChangeMs)
- {
- delayDiffMs = maxChangeMs;
- }
- _currentDelayMs = _currentDelayMs + static_cast<WebRtc_Word32>(delayDiffMs);
- }
- _prevFrameTimestamp = frameTimestamp;
-}
-
-void VCMTiming::UpdateCurrentDelay(WebRtc_Word64 renderTimeMs,
- WebRtc_Word64 actualDecodeTimeMs)
-{
- CriticalSectionScoped cs(_critSect);
- WebRtc_UWord32 targetDelayMs = TargetDelayInternal();
- // Make sure we try to sync with audio
- if (targetDelayMs < _minTotalDelayMs)
- {
- targetDelayMs = _minTotalDelayMs;
- }
- WebRtc_Word64 delayedMs = actualDecodeTimeMs -
- (renderTimeMs - MaxDecodeTimeMs() - _renderDelayMs);
- if (delayedMs < 0)
- {
- return;
- }
- else if (_currentDelayMs + delayedMs <= targetDelayMs)
- {
- _currentDelayMs += static_cast<WebRtc_UWord32>(delayedMs);
- }
- else
- {
- _currentDelayMs = targetDelayMs;
- }
-}
-
-WebRtc_Word32
-VCMTiming::StopDecodeTimer(WebRtc_UWord32 timeStamp,
- WebRtc_Word64 startTimeMs,
- WebRtc_Word64 nowMs)
-{
- CriticalSectionScoped cs(_critSect);
- const WebRtc_Word32 maxDecTime = MaxDecodeTimeMs();
- WebRtc_Word32 timeDiffMs = _codecTimer.StopTimer(startTimeMs, nowMs);
- if (timeDiffMs < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
- "Codec timer error: %d", timeDiffMs);
- assert(false);
- }
-
- if (_master)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
- "Frame decoded: timeStamp=%u decTime=%d maxDecTime=%u, at %u",
- timeStamp, timeDiffMs, maxDecTime, MaskWord64ToUWord32(nowMs));
- }
- return 0;
-}
-
-void
-VCMTiming::IncomingTimestamp(WebRtc_UWord32 timeStamp, WebRtc_Word64 nowMs)
-{
- CriticalSectionScoped cs(_critSect);
- _tsExtrapolator->Update(nowMs, timeStamp, _master);
-}
-
-WebRtc_Word64
-VCMTiming::RenderTimeMs(WebRtc_UWord32 frameTimestamp, WebRtc_Word64 nowMs) const
-{
- CriticalSectionScoped cs(_critSect);
- const WebRtc_Word64 renderTimeMs = RenderTimeMsInternal(frameTimestamp, nowMs);
- if (renderTimeMs < 0)
- {
- return renderTimeMs;
- }
- if (_master)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
- "Render frame %u at %u. Render delay %u, required delay %u,"
- " max decode time %u, min total delay %u",
- frameTimestamp, MaskWord64ToUWord32(renderTimeMs), _renderDelayMs,
- _requiredDelayMs, MaxDecodeTimeMs(),_minTotalDelayMs);
- }
- return renderTimeMs;
-}
-
-WebRtc_Word64
-VCMTiming::RenderTimeMsInternal(WebRtc_UWord32 frameTimestamp, WebRtc_Word64 nowMs) const
-{
- WebRtc_Word64 estimatedCompleteTimeMs =
- _tsExtrapolator->ExtrapolateLocalTime(frameTimestamp);
- if (estimatedCompleteTimeMs - nowMs > kMaxVideoDelayMs)
- {
- if (_master)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
- "Timestamp arrived 2 seconds early, reset statistics",
- frameTimestamp, estimatedCompleteTimeMs);
- }
- return -1;
- }
- if (_master)
- {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
- "ExtrapolateLocalTime(%u)=%u ms",
- frameTimestamp, MaskWord64ToUWord32(estimatedCompleteTimeMs));
- }
- if (estimatedCompleteTimeMs == -1)
- {
- estimatedCompleteTimeMs = nowMs;
- }
-
- return estimatedCompleteTimeMs + _currentDelayMs;
-}
-
-// Must be called from inside a critical section
-WebRtc_Word32
-VCMTiming::MaxDecodeTimeMs(FrameType frameType /*= kVideoFrameDelta*/) const
-{
- const WebRtc_Word32 decodeTimeMs = _codecTimer.RequiredDecodeTimeMs(frameType);
-
- if (decodeTimeMs < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoCoding, VCMId(_vcmId, _timingId),
- "Negative maximum decode time: %d", decodeTimeMs);
- return -1;
- }
- return decodeTimeMs;
-}
-
-WebRtc_UWord32
-VCMTiming::MaxWaitingTime(WebRtc_Word64 renderTimeMs, WebRtc_Word64 nowMs) const
-{
- CriticalSectionScoped cs(_critSect);
-
- const WebRtc_Word64 maxWaitTimeMs = renderTimeMs - nowMs -
- MaxDecodeTimeMs() - _renderDelayMs;
-
- if (maxWaitTimeMs < 0)
- {
- return 0;
- }
- return static_cast<WebRtc_UWord32>(maxWaitTimeMs);
-}
-
-bool
-VCMTiming::EnoughTimeToDecode(WebRtc_UWord32 availableProcessingTimeMs) const
-{
- CriticalSectionScoped cs(_critSect);
- WebRtc_Word32 maxDecodeTimeMs = MaxDecodeTimeMs();
- if (maxDecodeTimeMs < 0)
- {
- // Haven't decoded any frames yet, try decoding one to get an estimate
- // of the decode time.
- return true;
- }
- else if (maxDecodeTimeMs == 0)
- {
- // Decode time is less than 1, set to 1 for now since
- // we don't have any better precision. Count ticks later?
- maxDecodeTimeMs = 1;
- }
- return static_cast<WebRtc_Word32>(availableProcessingTimeMs) - maxDecodeTimeMs > 0;
-}
-
-WebRtc_UWord32
-VCMTiming::TargetVideoDelay() const
-{
- CriticalSectionScoped cs(_critSect);
- return TargetDelayInternal();
-}
-
-WebRtc_UWord32
-VCMTiming::TargetDelayInternal() const
-{
- return _requiredDelayMs + MaxDecodeTimeMs() + _renderDelayMs;
-}
-
-}
diff --git a/trunk/src/modules/video_coding/main/source/timing.h b/trunk/src/modules/video_coding/main/source/timing.h
deleted file mode 100644
index 41a4945..0000000
--- a/trunk/src/modules/video_coding/main/source/timing.h
+++ /dev/null
@@ -1,111 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TIMING_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TIMING_H_
-
-#include "typedefs.h"
-#include "critical_section_wrapper.h"
-#include "codec_timer.h"
-
-namespace webrtc
-{
-
-class TickTimeBase;
-class VCMTimestampExtrapolator;
-
-class VCMTiming
-{
-public:
- // The primary timing component should be passed
- // if this is the dual timing component.
- VCMTiming(TickTimeBase* clock,
- WebRtc_Word32 vcmId = 0,
- WebRtc_Word32 timingId = 0,
- VCMTiming* masterTiming = NULL);
- ~VCMTiming();
-
- // Resets the timing to the initial state.
- void Reset(WebRtc_Word64 nowMs = -1);
- void ResetDecodeTime();
-
- // The amount of time needed to render an image. Defaults to 10 ms.
- void SetRenderDelay(WebRtc_UWord32 renderDelayMs);
-
- // The minimum time the video must be delayed on the receiver to
- // get the desired jitter buffer level.
- void SetRequiredDelay(WebRtc_UWord32 requiredDelayMs);
-
- // Minimum total delay required to sync video with audio.
- void SetMinimumTotalDelay(WebRtc_UWord32 minTotalDelayMs);
-
- // Increases or decreases the current delay to get closer to the target delay.
- // Calculates how long it has been since the previous call to this function,
- // and increases/decreases the delay in proportion to the time difference.
- void UpdateCurrentDelay(WebRtc_UWord32 frameTimestamp);
-
- // Increases or decreases the current delay to get closer to the target delay.
- // Given the actual decode time in ms and the render time in ms for a frame, this
- // function calculates how late the frame is and increases the delay accordingly.
- void UpdateCurrentDelay(WebRtc_Word64 renderTimeMs, WebRtc_Word64 actualDecodeTimeMs);
-
- // Stops the decoder timer, should be called when the decoder returns a frame
- // or when the decoded frame callback is called.
- WebRtc_Word32 StopDecodeTimer(WebRtc_UWord32 timeStamp,
- WebRtc_Word64 startTimeMs,
- WebRtc_Word64 nowMs);
-
- // Used to report that a frame is passed to decoding. Updates the timestamp filter
- // which is used to map between timestamps and receiver system time.
- void IncomingTimestamp(WebRtc_UWord32 timeStamp, WebRtc_Word64 lastPacketTimeMs);
-
- // Returns the receiver system time when the frame with timestamp frameTimestamp
- // should be rendered, assuming that the system time currently is nowMs.
- WebRtc_Word64 RenderTimeMs(WebRtc_UWord32 frameTimestamp, WebRtc_Word64 nowMs) const;
-
- // Returns the maximum time in ms that we can wait for a frame to become complete
- // before we must pass it to the decoder.
- WebRtc_UWord32 MaxWaitingTime(WebRtc_Word64 renderTimeMs, WebRtc_Word64 nowMs) const;
-
- // Returns the current target delay which is required delay + decode time + render
- // delay.
- WebRtc_UWord32 TargetVideoDelay() const;
-
- // Calculates whether or not there is enough time to decode a frame given a
- // certain amount of processing time.
- bool EnoughTimeToDecode(WebRtc_UWord32 availableProcessingTimeMs) const;
-
- enum { kDefaultRenderDelayMs = 10 };
- enum { kDelayMaxChangeMsPerS = 100 };
-
-protected:
- WebRtc_Word32 MaxDecodeTimeMs(FrameType frameType = kVideoFrameDelta) const;
- WebRtc_Word64 RenderTimeMsInternal(WebRtc_UWord32 frameTimestamp,
- WebRtc_Word64 nowMs) const;
- WebRtc_UWord32 TargetDelayInternal() const;
-
-private:
- CriticalSectionWrapper* _critSect;
- WebRtc_Word32 _vcmId;
- TickTimeBase* _clock;
- WebRtc_Word32 _timingId;
- bool _master;
- VCMTimestampExtrapolator* _tsExtrapolator;
- VCMCodecTimer _codecTimer;
- WebRtc_UWord32 _renderDelayMs;
- WebRtc_UWord32 _minTotalDelayMs;
- WebRtc_UWord32 _requiredDelayMs;
- WebRtc_UWord32 _currentDelayMs;
- WebRtc_UWord32 _prevFrameTimestamp;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TIMING_H_
diff --git a/trunk/src/modules/video_coding/main/source/video_coding.gypi b/trunk/src/modules/video_coding/main/source/video_coding.gypi
deleted file mode 100644
index 3a99283..0000000
--- a/trunk/src/modules/video_coding/main/source/video_coding.gypi
+++ /dev/null
@@ -1,106 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'webrtc_video_coding',
- 'type': '<(library)',
- 'dependencies': [
- 'webrtc_i420',
- 'webrtc_vp8',
- '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '../interface',
- '../../../interface',
- '../../codecs/interface',
- '../../../../common_video/interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- '../../codecs/interface',
- ],
- },
- 'sources': [
- # interfaces
- '../interface/video_coding.h',
- '../interface/video_coding_defines.h',
-
- # headers
- 'codec_database.h',
- 'codec_timer.h',
- 'content_metrics_processing.h',
- 'decoding_state.h',
- 'encoded_frame.h',
- 'er_tables_xor.h',
- 'event.h',
- 'exp_filter.h',
- 'fec_tables_xor.h',
- 'frame_buffer.h',
- 'frame_dropper.h',
- 'generic_decoder.h',
- 'generic_encoder.h',
- 'inter_frame_delay.h',
- 'internal_defines.h',
- 'jitter_buffer.h',
- 'jitter_buffer_common.h',
- 'jitter_estimator.h',
- 'media_opt_util.h',
- 'media_optimization.h',
- 'nack_fec_tables.h',
- 'packet.h',
- 'qm_select_data.h',
- 'qm_select.h',
- 'receiver.h',
- 'rtt_filter.h',
- 'session_info.h',
- 'tick_time_base.h',
- 'timestamp_extrapolator.h',
- 'timestamp_map.h',
- 'timing.h',
- 'video_coding_impl.h',
-
- # sources
- 'codec_database.cc',
- 'codec_timer.cc',
- 'content_metrics_processing.cc',
- 'decoding_state.cc',
- 'encoded_frame.cc',
- 'exp_filter.cc',
- 'frame_buffer.cc',
- 'frame_dropper.cc',
- 'generic_decoder.cc',
- 'generic_encoder.cc',
- 'inter_frame_delay.cc',
- 'jitter_buffer.cc',
- 'jitter_buffer_common.cc',
- 'jitter_estimator.cc',
- 'media_opt_util.cc',
- 'media_optimization.cc',
- 'packet.cc',
- 'qm_select.cc',
- 'receiver.cc',
- 'rtt_filter.cc',
- 'session_info.cc',
- 'timestamp_extrapolator.cc',
- 'timestamp_map.cc',
- 'timing.cc',
- 'video_coding_impl.cc',
- ], # source
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/video_coding/main/source/video_coding_impl.cc b/trunk/src/modules/video_coding/main/source/video_coding_impl.cc
deleted file mode 100644
index 771f702..0000000
--- a/trunk/src/modules/video_coding/main/source/video_coding_impl.cc
+++ /dev/null
@@ -1,1385 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_coding_impl.h"
-#include "common_types.h"
-#include "encoded_frame.h"
-#include "jitter_buffer.h"
-#include "packet.h"
-#include "trace.h"
-#include "video_codec_interface.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
-
-namespace webrtc
-{
-
-//#define DEBUG_DECODER_BIT_STREAM
-//#define DEBUG_ENCODER_INPUT
-
-WebRtc_UWord32
-VCMProcessTimer::Period() const
-{
- return _periodMs;
-}
-
-WebRtc_UWord32
-VCMProcessTimer::TimeUntilProcess() const
-{
- return static_cast<WebRtc_UWord32>(
- VCM_MAX(static_cast<WebRtc_Word64>(_periodMs) -
- (_clock->MillisecondTimestamp() - _latestMs), 0));
-}
-
-void
-VCMProcessTimer::Processed()
-{
- _latestMs = _clock->MillisecondTimestamp();
-}
-
-VideoCodingModuleImpl::VideoCodingModuleImpl(const WebRtc_Word32 id,
- TickTimeBase* clock,
- bool delete_clock_on_destroy)
-:
-_id(id),
-clock_(clock),
-delete_clock_on_destroy_(delete_clock_on_destroy),
-_receiveCritSect(CriticalSectionWrapper::CreateCriticalSection()),
-_receiverInited(false),
-_timing(clock_, id, 1),
-_dualTiming(clock_, id, 2, &_timing),
-_receiver(_timing, clock_, id, 1),
-_dualReceiver(_dualTiming, clock_, id, 2, false),
-_decodedFrameCallback(_timing, clock_),
-_dualDecodedFrameCallback(_dualTiming, clock_),
-_frameTypeCallback(NULL),
-_frameStorageCallback(NULL),
-_receiveStatsCallback(NULL),
-_packetRequestCallback(NULL),
-_decoder(NULL),
-_dualDecoder(NULL),
-_bitStreamBeforeDecoder(NULL),
-_frameFromFile(),
-_keyRequestMode(kKeyOnError),
-_scheduleKeyRequest(false),
-
-_sendCritSect(CriticalSectionWrapper::CreateCriticalSection()),
-_encoder(),
-_encodedFrameCallback(),
-_mediaOpt(id, clock_),
-_sendCodecType(kVideoCodecUnknown),
-_sendStatsCallback(NULL),
-_encoderInputFile(NULL),
-
-_codecDataBase(id),
-_receiveStatsTimer(1000, clock_),
-_sendStatsTimer(1000, clock_),
-_retransmissionTimer(10, clock_),
-_keyRequestTimer(500, clock_)
-{
- assert(clock_);
- for (int i = 0; i < kMaxSimulcastStreams; i++)
- {
- _nextFrameType[i] = kVideoFrameDelta;
- }
-#ifdef DEBUG_DECODER_BIT_STREAM
- _bitStreamBeforeDecoder = fopen("decoderBitStream.bit", "wb");
-#endif
-#ifdef DEBUG_ENCODER_INPUT
- _encoderInputFile = fopen("encoderInput.yuv", "wb");
-#endif
-}
-
-VideoCodingModuleImpl::~VideoCodingModuleImpl()
-{
- if (_dualDecoder != NULL)
- {
- _codecDataBase.ReleaseDecoder(_dualDecoder);
- }
- delete _receiveCritSect;
- delete _sendCritSect;
- if (delete_clock_on_destroy_) delete clock_;
-#ifdef DEBUG_DECODER_BIT_STREAM
- fclose(_bitStreamBeforeDecoder);
-#endif
-#ifdef DEBUG_ENCODER_INPUT
- fclose(_encoderInputFile);
-#endif
-}
-
-VideoCodingModule*
-VideoCodingModule::Create(const WebRtc_Word32 id)
-{
- return new VideoCodingModuleImpl(id, new TickTimeBase(), true);
-}
-
-VideoCodingModule*
-VideoCodingModule::Create(const WebRtc_Word32 id, TickTimeBase* clock)
-{
- assert(clock);
- return new VideoCodingModuleImpl(id, clock, false);
-}
-
-void
-VideoCodingModule::Destroy(VideoCodingModule* module)
-{
- if (module != NULL)
- {
- delete static_cast<VideoCodingModuleImpl*>(module);
- }
-}
-
-WebRtc_Word32
-VideoCodingModuleImpl::Process()
-{
- WebRtc_Word32 returnValue = VCM_OK;
-
- // Receive-side statistics
- if (_receiveStatsTimer.TimeUntilProcess() == 0)
- {
- _receiveStatsTimer.Processed();
- if (_receiveStatsCallback != NULL)
- {
- WebRtc_UWord32 bitRate;
- WebRtc_UWord32 frameRate;
- const WebRtc_Word32 ret = _receiver.ReceiveStatistics(bitRate,
- frameRate);
- if (ret == 0)
- {
- _receiveStatsCallback->ReceiveStatistics(bitRate, frameRate);
- }
- else if (returnValue == VCM_OK)
- {
- returnValue = ret;
- }
- }
- }
-
- // Send-side statistics
- if (_sendStatsTimer.TimeUntilProcess() == 0)
- {
- _sendStatsTimer.Processed();
- if (_sendStatsCallback != NULL)
- {
- WebRtc_UWord32 bitRate;
- WebRtc_UWord32 frameRate;
- {
- CriticalSectionScoped cs(_sendCritSect);
- bitRate = static_cast<WebRtc_UWord32>(
- _mediaOpt.SentBitRate() + 0.5f);
- frameRate = static_cast<WebRtc_UWord32>(
- _mediaOpt.SentFrameRate() + 0.5f);
- }
- _sendStatsCallback->SendStatistics(bitRate, frameRate);
- }
- }
-
- // Packet retransmission requests
- if (_retransmissionTimer.TimeUntilProcess() == 0)
- {
- _retransmissionTimer.Processed();
- if (_packetRequestCallback != NULL)
- {
- WebRtc_UWord16 nackList[kNackHistoryLength];
- WebRtc_UWord16 length = kNackHistoryLength;
- const WebRtc_Word32 ret = NackList(nackList, length);
- if (ret != VCM_OK && returnValue == VCM_OK)
- {
- returnValue = ret;
- }
- if (length > 0)
- {
- _packetRequestCallback->ResendPackets(nackList, length);
- }
- }
- }
-
- // Key frame requests
- if (_keyRequestTimer.TimeUntilProcess() == 0)
- {
- _keyRequestTimer.Processed();
- if (_scheduleKeyRequest && _frameTypeCallback != NULL)
- {
- const WebRtc_Word32 ret = RequestKeyFrame();
- if (ret != VCM_OK && returnValue == VCM_OK)
- {
- returnValue = ret;
- }
- }
- }
-
- return returnValue;
-}
-
-WebRtc_Word32
-VideoCodingModuleImpl::Id() const
-{
- CriticalSectionScoped receiveCs(_receiveCritSect);
- {
- CriticalSectionScoped sendCs(_sendCritSect);
- return _id;
- }
-}
-
-// Change the unique identifier of this object
-WebRtc_Word32
-VideoCodingModuleImpl::ChangeUniqueId(const WebRtc_Word32 id)
-{
- CriticalSectionScoped receiveCs(_receiveCritSect);
- {
- CriticalSectionScoped sendCs(_sendCritSect);
- _id = id;
- return VCM_OK;
- }
-}
-
-// Returns the number of milliseconds until the module wants a worker thread to
-// call Process
-WebRtc_Word32
-VideoCodingModuleImpl::TimeUntilNextProcess()
-{
- WebRtc_UWord32 timeUntilNextProcess = VCM_MIN(
- _receiveStatsTimer.TimeUntilProcess(),
- _sendStatsTimer.TimeUntilProcess());
- if ((_receiver.NackMode() != kNoNack) ||
- (_dualReceiver.State() != kPassive))
- {
- // We need a Process call more often if we are relying on
- // retransmissions
- timeUntilNextProcess = VCM_MIN(timeUntilNextProcess,
- _retransmissionTimer.TimeUntilProcess());
- }
- timeUntilNextProcess = VCM_MIN(timeUntilNextProcess,
- _keyRequestTimer.TimeUntilProcess());
-
- return timeUntilNextProcess;
-}
-
-// Get number of supported codecs
-WebRtc_UWord8
-VideoCodingModule::NumberOfCodecs()
-{
- return VCMCodecDataBase::NumberOfCodecs();
-}
-
-// Get supported codec with id
-WebRtc_Word32
-VideoCodingModule::Codec(WebRtc_UWord8 listId, VideoCodec* codec)
-{
- if (codec == NULL)
- {
- return VCM_PARAMETER_ERROR;
- }
- return VCMCodecDataBase::Codec(listId, codec);
-}
-
-// Get supported codec with type
-WebRtc_Word32
-VideoCodingModule::Codec(VideoCodecType codecType, VideoCodec* codec)
-{
- if (codec == NULL)
- {
- return VCM_PARAMETER_ERROR;
- }
- return VCMCodecDataBase::Codec(codecType, codec);
-}
-
-/*
-* Sender
-*/
-
-// Reset send side to initial state - all components
-WebRtc_Word32
-VideoCodingModuleImpl::InitializeSender()
-{
- CriticalSectionScoped cs(_sendCritSect);
- _codecDataBase.ResetSender();
- _encoder = NULL;
- _encodedFrameCallback.SetTransportCallback(NULL);
- // setting default bitRate and frameRate to 0
- _mediaOpt.SetEncodingData(kVideoCodecUnknown, 0, 0, 0, 0, 0, 0);
- _mediaOpt.Reset(); // Resetting frame dropper
- return VCM_OK;
-}
-
-// Register the send codec to be used.
-WebRtc_Word32
-VideoCodingModuleImpl::RegisterSendCodec(const VideoCodec* sendCodec,
- WebRtc_UWord32 numberOfCores,
- WebRtc_UWord32 maxPayloadSize)
-{
- CriticalSectionScoped cs(_sendCritSect);
- if (sendCodec == NULL)
- {
- return VCM_PARAMETER_ERROR;
- }
- WebRtc_Word32 ret = _codecDataBase.RegisterSendCodec(sendCodec,
- numberOfCores,
- maxPayloadSize);
- if (ret < 0)
- {
- return ret;
- }
-
- _encoder = _codecDataBase.SetEncoder(sendCodec, &_encodedFrameCallback);
- if (_encoder == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceVideoCoding,
- VCMId(_id),
- "Failed to initialize encoder");
- return VCM_CODEC_ERROR;
- }
- _sendCodecType = sendCodec->codecType;
- int numLayers = (_sendCodecType != kVideoCodecVP8) ? 1 :
- sendCodec->codecSpecific.VP8.numberOfTemporalLayers;
-
- _mediaOpt.SetEncodingData(_sendCodecType,
- sendCodec->maxBitrate,
- sendCodec->maxFramerate,
- sendCodec->startBitrate,
- sendCodec->width,
- sendCodec->height,
- numLayers);
- _mediaOpt.SetMtu(maxPayloadSize);
-
- return VCM_OK;
-}
-
-// Get current send codec
-WebRtc_Word32
-VideoCodingModuleImpl::SendCodec(VideoCodec* currentSendCodec) const
-{
- CriticalSectionScoped cs(_sendCritSect);
-
- if (currentSendCodec == NULL)
- {
- return VCM_PARAMETER_ERROR;
- }
- return _codecDataBase.SendCodec(currentSendCodec);
-}
-
-// Get the current send codec type
-VideoCodecType
-VideoCodingModuleImpl::SendCodec() const
-{
- CriticalSectionScoped cs(_sendCritSect);
-
- return _codecDataBase.SendCodec();
-}
-
-// Register an external decoder object.
-// This can not be used together with external decoder callbacks.
-WebRtc_Word32
-VideoCodingModuleImpl::RegisterExternalEncoder(VideoEncoder* externalEncoder,
- WebRtc_UWord8 payloadType,
- bool internalSource /*= false*/)
-{
- CriticalSectionScoped cs(_sendCritSect);
-
- if (externalEncoder == NULL)
- {
- bool wasSendCodec = false;
- const WebRtc_Word32 ret = _codecDataBase.DeRegisterExternalEncoder(
- payloadType,
- wasSendCodec);
- if (wasSendCodec)
- {
- // Make sure the VCM doesn't use the de-registered codec
- _encoder = NULL;
- }
- return ret;
- }
- return _codecDataBase.RegisterExternalEncoder(externalEncoder,
- payloadType,
- internalSource);
-}
-
-// Get codec config parameters
-WebRtc_Word32
-VideoCodingModuleImpl::CodecConfigParameters(WebRtc_UWord8* buffer,
- WebRtc_Word32 size)
-{
- CriticalSectionScoped cs(_sendCritSect);
- if (_encoder != NULL)
- {
- return _encoder->CodecConfigParameters(buffer, size);
- }
- return VCM_UNINITIALIZED;
-}
-
-// Get encode bitrate
-int VideoCodingModuleImpl::Bitrate(unsigned int* bitrate) const
-{
- CriticalSectionScoped cs(_sendCritSect);
- // return the bit rate which the encoder is set to
- if (!_encoder) {
- return VCM_UNINITIALIZED;
- }
- *bitrate = _encoder->BitRate();
- return 0;
-}
-
-// Get encode frame rate
-int VideoCodingModuleImpl::FrameRate(unsigned int* framerate) const
-{
- CriticalSectionScoped cs(_sendCritSect);
- // input frame rate, not compensated
- if (!_encoder) {
- return VCM_UNINITIALIZED;
- }
- *framerate = _encoder->FrameRate();
- return 0;
-}
-
-// Set channel parameters
-WebRtc_Word32
-VideoCodingModuleImpl::SetChannelParameters(WebRtc_UWord32 availableBandWidth,
- WebRtc_UWord8 lossRate,
- WebRtc_UWord32 rtt)
-{
- WebRtc_Word32 ret = 0;
- {
- CriticalSectionScoped sendCs(_sendCritSect);
- WebRtc_UWord32 targetRate = _mediaOpt.SetTargetRates(availableBandWidth,
- lossRate,
- rtt);
- if (_encoder != NULL)
- {
- ret = _encoder->SetChannelParameters(lossRate, rtt);
- if (ret < 0 )
- {
- return ret;
- }
- ret = (WebRtc_Word32)_encoder->SetRates(targetRate,
- _mediaOpt.InputFrameRate());
- if (ret < 0)
- {
- return ret;
- }
- }
- else
- {
- return VCM_UNINITIALIZED;
- } // encoder
- }// send side
- return VCM_OK;
-}
-
-WebRtc_Word32
-VideoCodingModuleImpl::SetReceiveChannelParameters(WebRtc_UWord32 rtt)
-{
- CriticalSectionScoped receiveCs(_receiveCritSect);
- _receiver.UpdateRtt(rtt);
- return 0;
-}
-
-// Register a transport callback which will be called to deliver the encoded
-// buffers
-WebRtc_Word32
-VideoCodingModuleImpl::RegisterTransportCallback(
- VCMPacketizationCallback* transport)
-{
- CriticalSectionScoped cs(_sendCritSect);
- _encodedFrameCallback.SetMediaOpt(&_mediaOpt);
- _encodedFrameCallback.SetTransportCallback(transport);
- return VCM_OK;
-}
-
-// Register video output information callback which will be called to deliver
-// information about the video stream produced by the encoder, for instance the
-// average frame rate and bit rate.
-WebRtc_Word32
-VideoCodingModuleImpl::RegisterSendStatisticsCallback(
- VCMSendStatisticsCallback* sendStats)
-{
- CriticalSectionScoped cs(_sendCritSect);
- _sendStatsCallback = sendStats;
- return VCM_OK;
-}
-
-// Register a video quality settings callback which will be called when frame
-// rate/dimensions need to be updated for video quality optimization
-WebRtc_Word32
-VideoCodingModuleImpl::RegisterVideoQMCallback(
- VCMQMSettingsCallback* videoQMSettings)
-{
- CriticalSectionScoped cs(_sendCritSect);
- return _mediaOpt.RegisterVideoQMCallback(videoQMSettings);
-}
-
-
-// Register a video protection callback which will be called to deliver the
-// requested FEC rate and NACK status (on/off).
-WebRtc_Word32
-VideoCodingModuleImpl::RegisterProtectionCallback(
- VCMProtectionCallback* protection)
-{
- CriticalSectionScoped cs(_sendCritSect);
- _mediaOpt.RegisterProtectionCallback(protection);
- return VCM_OK;
-}
-
-// Enable or disable a video protection method.
-WebRtc_Word32
-VideoCodingModuleImpl::SetVideoProtection(VCMVideoProtection videoProtection,
- bool enable)
-{
-
- switch (videoProtection)
- {
-
- case kProtectionNack:
- {
- // Both send-side and receive-side
- SetVideoProtection(kProtectionNackSender, enable);
- SetVideoProtection(kProtectionNackReceiver, enable);
- break;
- }
-
- case kProtectionNackSender:
- {
- CriticalSectionScoped cs(_sendCritSect);
- _mediaOpt.EnableProtectionMethod(enable, kNack);
- break;
- }
-
- case kProtectionNackReceiver:
- {
- CriticalSectionScoped cs(_receiveCritSect);
- if (enable)
- {
- _receiver.SetNackMode(kNackInfinite);
- }
- else
- {
- _receiver.SetNackMode(kNoNack);
- }
- break;
- }
-
- case kProtectionDualDecoder:
- {
- CriticalSectionScoped cs(_receiveCritSect);
- if (enable)
- {
- _receiver.SetNackMode(kNoNack);
- _dualReceiver.SetNackMode(kNackInfinite);
- }
- else
- {
- _dualReceiver.SetNackMode(kNoNack);
- }
- break;
- }
-
- case kProtectionKeyOnLoss:
- {
- CriticalSectionScoped cs(_receiveCritSect);
- if (enable)
- {
- _keyRequestMode = kKeyOnLoss;
- }
- else if (_keyRequestMode == kKeyOnLoss)
- {
- _keyRequestMode = kKeyOnError; // default mode
- }
- else
- {
- return VCM_PARAMETER_ERROR;
- }
- break;
- }
-
- case kProtectionKeyOnKeyLoss:
- {
- CriticalSectionScoped cs(_receiveCritSect);
- if (enable)
- {
- _keyRequestMode = kKeyOnKeyLoss;
- }
- else if (_keyRequestMode == kKeyOnKeyLoss)
- {
- _keyRequestMode = kKeyOnError; // default mode
- }
- else
- {
- return VCM_PARAMETER_ERROR;
- }
- break;
- }
-
- case kProtectionNackFEC:
- {
- {
- // Receive side
- CriticalSectionScoped cs(_receiveCritSect);
- if (enable)
- {
- _receiver.SetNackMode(kNackHybrid);
- }
- else
- {
- _receiver.SetNackMode(kNoNack);
- }
- }
- // Send Side
- {
- CriticalSectionScoped cs(_sendCritSect);
- _mediaOpt.EnableProtectionMethod(enable, kNackFec);
- }
- break;
- }
-
- case kProtectionFEC:
- {
- CriticalSectionScoped cs(_sendCritSect);
- _mediaOpt.EnableProtectionMethod(enable, kFec);
- break;
- }
-
- case kProtectionPeriodicKeyFrames:
- {
- CriticalSectionScoped cs(_sendCritSect);
- return _codecDataBase.SetPeriodicKeyFrames(enable);
- break;
- }
- }
- return VCM_OK;
-}
-
-// Add one raw video frame to the encoder, blocking.
-WebRtc_Word32
-VideoCodingModuleImpl::AddVideoFrame(const VideoFrame& videoFrame,
- const VideoContentMetrics* contentMetrics,
- const CodecSpecificInfo* codecSpecificInfo)
-{
- CriticalSectionScoped cs(_sendCritSect);
-
- if (_encoder == NULL)
- {
- return VCM_UNINITIALIZED;
- }
- if (_nextFrameType[0] == kFrameEmpty)
- {
- return VCM_OK;
- }
- _mediaOpt.UpdateIncomingFrameRate();
-
- if (_mediaOpt.DropFrame())
- {
- WEBRTC_TRACE(webrtc::kTraceStream,
- webrtc::kTraceVideoCoding,
- VCMId(_id),
- "Drop frame due to bitrate");
- }
- else
- {
- _mediaOpt.updateContentData(contentMetrics);
- WebRtc_Word32 ret = _encoder->Encode(videoFrame,
- codecSpecificInfo,
- _nextFrameType);
-#ifdef DEBUG_ENCODER_INPUT
- if (_encoderInputFile != NULL)
- {
- fwrite(videoFrame.Buffer(), 1, videoFrame.Length(),
- _encoderInputFile);
- }
-#endif
- if (ret < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceVideoCoding,
- VCMId(_id),
- "Encode error: %d", ret);
- return ret;
- }
- for (int i = 0; i < kMaxSimulcastStreams; i++)
- {
- _nextFrameType[i] = kVideoFrameDelta; // default frame type
- }
- }
- return VCM_OK;
-}
-
-// Next frame encoded should be of the type frameType
-// Good for only one frame
-WebRtc_Word32
-VideoCodingModuleImpl::FrameTypeRequest(FrameType frameType,
- WebRtc_UWord8 simulcastIdx)
-{
- assert(simulcastIdx < kMaxSimulcastStreams);
-
-
- CriticalSectionScoped cs(_sendCritSect);
- _nextFrameType[simulcastIdx] = frameType;
- if (_encoder != NULL && _encoder->InternalSource())
- {
- // Try to request the frame if we have an external encoder with
- // internal source since AddVideoFrame never will be called.
- if (_encoder->RequestFrame(_nextFrameType) == WEBRTC_VIDEO_CODEC_OK)
- {
- _nextFrameType[simulcastIdx] = kVideoFrameDelta;
- }
- }
- return VCM_OK;
-}
-
-WebRtc_Word32
-VideoCodingModuleImpl::EnableFrameDropper(bool enable)
-{
- CriticalSectionScoped cs(_sendCritSect);
- _mediaOpt.EnableFrameDropper(enable);
- return VCM_OK;
-}
-
-
-WebRtc_Word32
-VideoCodingModuleImpl::SentFrameCount(VCMFrameCount &frameCount) const
-{
- CriticalSectionScoped cs(_sendCritSect);
- return _mediaOpt.SentFrameCount(frameCount);
-}
-
-// Initialize receiver, resets codec database etc
-WebRtc_Word32
-VideoCodingModuleImpl::InitializeReceiver()
-{
- CriticalSectionScoped cs(_receiveCritSect);
- WebRtc_Word32 ret = _receiver.Initialize();
- if (ret < 0)
- {
- return ret;
- }
-
- ret = _dualReceiver.Initialize();
- if (ret < 0)
- {
- return ret;
- }
- _codecDataBase.ResetReceiver();
- _timing.Reset();
-
- _decoder = NULL;
- _decodedFrameCallback.SetUserReceiveCallback(NULL);
- _receiverInited = true;
- _frameTypeCallback = NULL;
- _frameStorageCallback = NULL;
- _receiveStatsCallback = NULL;
- _packetRequestCallback = NULL;
- _keyRequestMode = kKeyOnError;
- _scheduleKeyRequest = false;
-
- return VCM_OK;
-}
-
-// Register a receive callback. Will be called whenever there is a new frame
-// ready for rendering.
-WebRtc_Word32
-VideoCodingModuleImpl::RegisterReceiveCallback(
- VCMReceiveCallback* receiveCallback)
-{
- CriticalSectionScoped cs(_receiveCritSect);
- _decodedFrameCallback.SetUserReceiveCallback(receiveCallback);
- return VCM_OK;
-}
-
-WebRtc_Word32
-VideoCodingModuleImpl::RegisterReceiveStatisticsCallback(
- VCMReceiveStatisticsCallback* receiveStats)
-{
- CriticalSectionScoped cs(_receiveCritSect);
- _receiveStatsCallback = receiveStats;
- return VCM_OK;
-}
-
-// Register an externally defined decoder/render object.
-// Can be a decoder only or a decoder coupled with a renderer.
-WebRtc_Word32
-VideoCodingModuleImpl::RegisterExternalDecoder(VideoDecoder* externalDecoder,
- WebRtc_UWord8 payloadType,
- bool internalRenderTiming)
-{
- CriticalSectionScoped cs(_receiveCritSect);
- if (externalDecoder == NULL)
- {
- // Make sure the VCM updates the decoder next time it decodes.
- _decoder = NULL;
- return _codecDataBase.DeRegisterExternalDecoder(payloadType);
- }
- else
- {
- return _codecDataBase.RegisterExternalDecoder(externalDecoder,
- payloadType,
- internalRenderTiming);
- }
-}
-
-// Register a frame type request callback.
-WebRtc_Word32
-VideoCodingModuleImpl::RegisterFrameTypeCallback(
- VCMFrameTypeCallback* frameTypeCallback)
-{
- CriticalSectionScoped cs(_receiveCritSect);
- _frameTypeCallback = frameTypeCallback;
- return VCM_OK;
-}
-
-WebRtc_Word32
-VideoCodingModuleImpl::RegisterFrameStorageCallback(
- VCMFrameStorageCallback* frameStorageCallback)
-{
- CriticalSectionScoped cs(_receiveCritSect);
- _frameStorageCallback = frameStorageCallback;
- return VCM_OK;
-}
-
-WebRtc_Word32
-VideoCodingModuleImpl::RegisterPacketRequestCallback(
- VCMPacketRequestCallback* callback)
-{
- CriticalSectionScoped cs(_receiveCritSect);
- _packetRequestCallback = callback;
- return VCM_OK;
-}
-
-// Decode next frame, blocking.
-// Should be called as often as possible to get the most out of the decoder.
-WebRtc_Word32
-VideoCodingModuleImpl::Decode(WebRtc_UWord16 maxWaitTimeMs)
-{
- WebRtc_Word64 nextRenderTimeMs;
- {
- CriticalSectionScoped cs(_receiveCritSect);
- if (!_receiverInited)
- {
- return VCM_UNINITIALIZED;
- }
- if (!_codecDataBase.DecoderRegistered())
- {
- return VCM_NO_CODEC_REGISTERED;
- }
- }
-
- const bool dualReceiverEnabledNotReceiving =
- (_dualReceiver.State() != kReceiving &&
- _dualReceiver.NackMode() == kNackInfinite);
-
- VCMEncodedFrame* frame = _receiver.FrameForDecoding(
- maxWaitTimeMs,
- nextRenderTimeMs,
- _codecDataBase.RenderTiming(),
- &_dualReceiver);
-
- if (dualReceiverEnabledNotReceiving && _dualReceiver.State() == kReceiving)
- {
- // Dual receiver is enabled (kNACK enabled), but was not receiving
- // before the call to FrameForDecoding(). After the call the state
- // changed to receiving, and therefore we must copy the primary decoder
- // state to the dual decoder to make it possible for the dual decoder to
- // start decoding retransmitted frames and recover.
- CriticalSectionScoped cs(_receiveCritSect);
- if (_dualDecoder != NULL)
- {
- _codecDataBase.ReleaseDecoder(_dualDecoder);
- }
- _dualDecoder = _codecDataBase.CreateDecoderCopy();
- if (_dualDecoder != NULL)
- {
- _dualDecoder->RegisterDecodeCompleteCallback(
- &_dualDecodedFrameCallback);
- }
- else
- {
- _dualReceiver.Reset();
- }
- }
-
- if (frame == NULL)
- return VCM_FRAME_NOT_READY;
- else
- {
- CriticalSectionScoped cs(_receiveCritSect);
-
- // If this frame was too late, we should adjust the delay accordingly
- _timing.UpdateCurrentDelay(frame->RenderTimeMs(),
- clock_->MillisecondTimestamp());
-
-#ifdef DEBUG_DECODER_BIT_STREAM
- if (_bitStreamBeforeDecoder != NULL)
- {
- // Write bit stream to file for debugging purposes
- fwrite(frame->Buffer(), 1, frame->Length(),
- _bitStreamBeforeDecoder);
- }
-#endif
- if (_frameStorageCallback != NULL)
- {
- WebRtc_Word32 ret = frame->Store(*_frameStorageCallback);
- if (ret < 0)
- {
- return ret;
- }
- }
-
- const WebRtc_Word32 ret = Decode(*frame);
- _receiver.ReleaseFrame(frame);
- frame = NULL;
- if (ret != VCM_OK)
- {
- return ret;
- }
- }
- return VCM_OK;
-}
-
-WebRtc_Word32
-VideoCodingModuleImpl::RequestSliceLossIndication(
- const WebRtc_UWord64 pictureID) const
-{
- if (_frameTypeCallback != NULL)
- {
- const WebRtc_Word32 ret =
- _frameTypeCallback->SliceLossIndicationRequest(pictureID);
- if (ret < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceVideoCoding,
- VCMId(_id),
- "Failed to request key frame");
- return ret;
- }
- } else
- {
- WEBRTC_TRACE(webrtc::kTraceWarning,
- webrtc::kTraceVideoCoding,
- VCMId(_id),
- "No frame type request callback registered");
- return VCM_MISSING_CALLBACK;
- }
- return VCM_OK;
-}
-
-WebRtc_Word32
-VideoCodingModuleImpl::RequestKeyFrame()
-{
- if (_frameTypeCallback != NULL)
- {
- const WebRtc_Word32 ret = _frameTypeCallback->RequestKeyFrame();
- if (ret < 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceVideoCoding,
- VCMId(_id),
- "Failed to request key frame");
- return ret;
- }
- _scheduleKeyRequest = false;
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceWarning,
- webrtc::kTraceVideoCoding,
- VCMId(_id),
- "No frame type request callback registered");
- return VCM_MISSING_CALLBACK;
- }
- return VCM_OK;
-}
-
-WebRtc_Word32
-VideoCodingModuleImpl::DecodeDualFrame(WebRtc_UWord16 maxWaitTimeMs)
-{
- CriticalSectionScoped cs(_receiveCritSect);
- if (_dualReceiver.State() != kReceiving ||
- _dualReceiver.NackMode() != kNackInfinite)
- {
- // The dual receiver is currently not receiving or
- // dual decoder mode is disabled.
- return VCM_OK;
- }
- WebRtc_Word64 dummyRenderTime;
- WebRtc_Word32 decodeCount = 0;
- VCMEncodedFrame* dualFrame = _dualReceiver.FrameForDecoding(
- maxWaitTimeMs,
- dummyRenderTime);
- if (dualFrame != NULL && _dualDecoder != NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceStream,
- webrtc::kTraceVideoCoding,
- VCMId(_id),
- "Decoding frame %u with dual decoder",
- dualFrame->TimeStamp());
- // Decode dualFrame and try to catch up
- WebRtc_Word32 ret = _dualDecoder->Decode(*dualFrame,
- clock_->MillisecondTimestamp());
- if (ret != WEBRTC_VIDEO_CODEC_OK)
- {
- WEBRTC_TRACE(webrtc::kTraceWarning,
- webrtc::kTraceVideoCoding,
- VCMId(_id),
- "Failed to decode frame with dual decoder");
- _dualReceiver.ReleaseFrame(dualFrame);
- return VCM_CODEC_ERROR;
- }
- if (_receiver.DualDecoderCaughtUp(dualFrame, _dualReceiver))
- {
- // Copy the complete decoder state of the dual decoder
- // to the primary decoder.
- WEBRTC_TRACE(webrtc::kTraceStream,
- webrtc::kTraceVideoCoding,
- VCMId(_id),
- "Dual decoder caught up");
- _codecDataBase.CopyDecoder(*_dualDecoder);
- _codecDataBase.ReleaseDecoder(_dualDecoder);
- _dualDecoder = NULL;
- }
- decodeCount++;
- }
- _dualReceiver.ReleaseFrame(dualFrame);
- return decodeCount;
-}
-
-
-// Must be called from inside the receive side critical section.
-WebRtc_Word32
-VideoCodingModuleImpl::Decode(const VCMEncodedFrame& frame)
-{
- // Change decoder if payload type has changed
- const bool renderTimingBefore = _codecDataBase.RenderTiming();
- _decoder = _codecDataBase.SetDecoder(frame.PayloadType(),
- _decodedFrameCallback);
- if (renderTimingBefore != _codecDataBase.RenderTiming())
- {
- // Make sure we reset the decode time estimate since it will
- // be zero for codecs without render timing.
- _timing.ResetDecodeTime();
- }
- if (_decoder == NULL)
- {
- return VCM_NO_CODEC_REGISTERED;
- }
- // Decode a frame
- WebRtc_Word32 ret = _decoder->Decode(frame, clock_->MillisecondTimestamp());
-
- // Check for failed decoding, run frame type request callback if needed.
- if (ret < 0)
- {
- if (ret == VCM_ERROR_REQUEST_SLI)
- {
- return RequestSliceLossIndication(
- _decodedFrameCallback.LastReceivedPictureID() + 1);
- }
- else
- {
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceVideoCoding,
- VCMId(_id),
- "Failed to decode frame %u, requesting key frame",
- frame.TimeStamp());
- ret = RequestKeyFrame();
- }
- }
- else if (ret == VCM_REQUEST_SLI)
- {
- ret = RequestSliceLossIndication(
- _decodedFrameCallback.LastReceivedPictureID() + 1);
- }
- if (!frame.Complete() || frame.MissingFrame())
- {
- switch (_keyRequestMode)
- {
- case kKeyOnKeyLoss:
- {
- if (frame.FrameType() == kVideoFrameKey)
- {
- _scheduleKeyRequest = true;
- return VCM_OK;
- }
- break;
- }
- case kKeyOnLoss:
- {
- _scheduleKeyRequest = true;
- return VCM_OK;
- }
- default:
- break;
- }
- }
- return ret;
-}
-
-WebRtc_Word32
-VideoCodingModuleImpl::DecodeFromStorage(
- const EncodedVideoData& frameFromStorage)
-{
- CriticalSectionScoped cs(_receiveCritSect);
- WebRtc_Word32 ret = _frameFromFile.ExtractFromStorage(frameFromStorage);
- if (ret < 0)
- {
- return ret;
- }
- return Decode(_frameFromFile);
-}
-
-// Reset the decoder state
-WebRtc_Word32
-VideoCodingModuleImpl::ResetDecoder()
-{
- CriticalSectionScoped cs(_receiveCritSect);
- if (_decoder != NULL)
- {
- _receiver.Initialize();
- _timing.Reset();
- _scheduleKeyRequest = false;
- _decoder->Reset();
- }
- if (_dualReceiver.State() != kPassive)
- {
- _dualReceiver.Initialize();
- }
- if (_dualDecoder != NULL)
- {
- _codecDataBase.ReleaseDecoder(_dualDecoder);
- _dualDecoder = NULL;
- }
- return VCM_OK;
-}
-
-// Register possible receive codecs, can be called multiple times
-WebRtc_Word32
-VideoCodingModuleImpl::RegisterReceiveCodec(const VideoCodec* receiveCodec,
- WebRtc_Word32 numberOfCores,
- bool requireKeyFrame)
-{
- CriticalSectionScoped cs(_receiveCritSect);
- if (receiveCodec == NULL)
- {
- return VCM_PARAMETER_ERROR;
- }
- return _codecDataBase.RegisterReceiveCodec(receiveCodec, numberOfCores,
- requireKeyFrame);
-}
-
-// Get current received codec
-WebRtc_Word32
-VideoCodingModuleImpl::ReceiveCodec(VideoCodec* currentReceiveCodec) const
-{
- CriticalSectionScoped cs(_receiveCritSect);
- if (currentReceiveCodec == NULL)
- {
- return VCM_PARAMETER_ERROR;
- }
- return _codecDataBase.ReceiveCodec(currentReceiveCodec);
-}
-
-// Get current received codec
-VideoCodecType
-VideoCodingModuleImpl::ReceiveCodec() const
-{
- CriticalSectionScoped cs(_receiveCritSect);
- return _codecDataBase.ReceiveCodec();
-}
-
-// Incoming packet from network parsed and ready for decode, non blocking.
-WebRtc_Word32
-VideoCodingModuleImpl::IncomingPacket(const WebRtc_UWord8* incomingPayload,
- WebRtc_UWord32 payloadLength,
- const WebRtcRTPHeader& rtpInfo)
-{
- const VCMPacket packet(incomingPayload, payloadLength, rtpInfo);
- WebRtc_Word32 ret;
- if (_dualReceiver.State() != kPassive)
- {
- ret = _dualReceiver.InsertPacket(packet,
- rtpInfo.type.Video.width,
- rtpInfo.type.Video.height);
- if (ret == VCM_FLUSH_INDICATOR) {
- RequestKeyFrame();
- ResetDecoder();
- } else if (ret < 0) {
- return ret;
- }
- }
- ret = _receiver.InsertPacket(packet, rtpInfo.type.Video.width,
- rtpInfo.type.Video.height);
- if (ret == VCM_FLUSH_INDICATOR) {
- RequestKeyFrame();
- ResetDecoder();
- } else if (ret < 0) {
- return ret;
- }
- return VCM_OK;
-}
-
-// Minimum playout delay (used for lip-sync). This is the minimum delay required
-// to sync with audio. Not included in VideoCodingModule::Delay()
-// Defaults to 0 ms.
-WebRtc_Word32
-VideoCodingModuleImpl::SetMinimumPlayoutDelay(WebRtc_UWord32 minPlayoutDelayMs)
-{
- _timing.SetMinimumTotalDelay(minPlayoutDelayMs);
- return VCM_OK;
-}
-
-// The estimated delay caused by rendering, defaults to
-// kDefaultRenderDelayMs = 10 ms
-WebRtc_Word32
-VideoCodingModuleImpl::SetRenderDelay(WebRtc_UWord32 timeMS)
-{
- _timing.SetRenderDelay(timeMS);
- return VCM_OK;
-}
-
-// Current video delay
-WebRtc_Word32
-VideoCodingModuleImpl::Delay() const
-{
- return _timing.TargetVideoDelay();
-}
-
-// Nack list
-WebRtc_Word32
-VideoCodingModuleImpl::NackList(WebRtc_UWord16* nackList, WebRtc_UWord16& size)
-{
- VCMNackStatus nackStatus = kNackOk;
- // Collect sequence numbers from the default receiver
- // if in normal nack mode. Otherwise collect them from
- // the dual receiver if the dual receiver is receiving.
- if (_receiver.NackMode() != kNoNack)
- {
- nackStatus = _receiver.NackList(nackList, size);
- }
- else if (_dualReceiver.State() != kPassive)
- {
- nackStatus = _dualReceiver.NackList(nackList, size);
- }
- else
- {
- size = 0;
- }
-
- switch (nackStatus)
- {
- case kNackNeedMoreMemory:
- {
- WEBRTC_TRACE(webrtc::kTraceError,
- webrtc::kTraceVideoCoding,
- VCMId(_id),
- "Out of memory");
- return VCM_MEMORY;
- }
- case kNackKeyFrameRequest:
- {
- CriticalSectionScoped cs(_receiveCritSect);
- WEBRTC_TRACE(webrtc::kTraceWarning,
- webrtc::kTraceVideoCoding,
- VCMId(_id),
- "Failed to get NACK list, requesting key frame");
- return RequestKeyFrame();
- }
- default:
- break;
- }
- return VCM_OK;
-}
-
-WebRtc_Word32
-VideoCodingModuleImpl::ReceivedFrameCount(VCMFrameCount& frameCount) const
-{
- return _receiver.ReceivedFrameCount(frameCount);
-}
-
-WebRtc_UWord32 VideoCodingModuleImpl::DiscardedPackets() const {
- return _receiver.DiscardedPackets();
-}
-
-int VideoCodingModuleImpl::SetSenderNackMode(SenderNackMode mode) {
- CriticalSectionScoped cs(_sendCritSect);
-
- switch (mode) {
- case kNackNone:
- _mediaOpt.EnableProtectionMethod(false, kNack);
- break;
- case kNackAll:
- _mediaOpt.EnableProtectionMethod(true, kNack);
- break;
- case kNackSelective:
- return VCM_NOT_IMPLEMENTED;
- break;
- }
- return VCM_OK;
-}
-
-int VideoCodingModuleImpl::SetSenderReferenceSelection(bool enable) {
- return VCM_NOT_IMPLEMENTED;
-}
-
-int VideoCodingModuleImpl::SetSenderFEC(bool enable) {
- CriticalSectionScoped cs(_sendCritSect);
- _mediaOpt.EnableProtectionMethod(enable, kFec);
- return VCM_OK;
-}
-
-int VideoCodingModuleImpl::SetSenderKeyFramePeriod(int periodMs) {
- return VCM_NOT_IMPLEMENTED;
-}
-
-int VideoCodingModuleImpl::SetReceiverRobustnessMode(
- ReceiverRobustness robustnessMode,
- DecodeErrors errorMode) {
- CriticalSectionScoped cs(_receiveCritSect);
- switch (robustnessMode) {
- case kNone:
- _receiver.SetNackMode(kNoNack);
- _dualReceiver.SetNackMode(kNoNack);
- if (errorMode == kNoDecodeErrors) {
- _keyRequestMode = kKeyOnLoss;
- } else {
- _keyRequestMode = kKeyOnError;
- }
- break;
- case kHardNack:
- if (errorMode == kAllowDecodeErrors) {
- return VCM_PARAMETER_ERROR;
- }
- _receiver.SetNackMode(kNackInfinite);
- _dualReceiver.SetNackMode(kNoNack);
- _keyRequestMode = kKeyOnError; // TODO(hlundin): On long NACK list?
- break;
- case kSoftNack:
- assert(false); // TODO(hlundin): Not completed.
- return VCM_NOT_IMPLEMENTED;
- _receiver.SetNackMode(kNackHybrid);
- _dualReceiver.SetNackMode(kNoNack);
- _keyRequestMode = kKeyOnError;
- break;
- case kDualDecoder:
- if (errorMode == kNoDecodeErrors) {
- return VCM_PARAMETER_ERROR;
- }
- _receiver.SetNackMode(kNoNack);
- _dualReceiver.SetNackMode(kNackInfinite);
- _keyRequestMode = kKeyOnError;
- break;
- case kReferenceSelection:
- assert(false); // TODO(hlundin): Not completed.
- return VCM_NOT_IMPLEMENTED;
- if (errorMode == kNoDecodeErrors) {
- return VCM_PARAMETER_ERROR;
- }
- _receiver.SetNackMode(kNoNack);
- _dualReceiver.SetNackMode(kNoNack);
- break;
- }
- return VCM_OK;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/video_coding_impl.h b/trunk/src/modules/video_coding/main/source/video_coding_impl.h
deleted file mode 100644
index 490c847..0000000
--- a/trunk/src/modules/video_coding/main/source/video_coding_impl.h
+++ /dev/null
@@ -1,311 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_VIDEO_CODING_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_CODING_VIDEO_CODING_IMPL_H_
-
-#include "video_coding.h"
-#include "critical_section_wrapper.h"
-#include "frame_buffer.h"
-#include "receiver.h"
-#include "timing.h"
-#include "jitter_buffer.h"
-#include "codec_database.h"
-#include "generic_decoder.h"
-#include "generic_encoder.h"
-#include "media_optimization.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
-
-#include <stdio.h>
-
-namespace webrtc
-{
-
-class VCMProcessTimer
-{
-public:
- VCMProcessTimer(WebRtc_UWord32 periodMs, TickTimeBase* clock)
- : _clock(clock),
- _periodMs(periodMs),
- _latestMs(_clock->MillisecondTimestamp()) {}
- WebRtc_UWord32 Period() const;
- WebRtc_UWord32 TimeUntilProcess() const;
- void Processed();
-
-private:
- TickTimeBase* _clock;
- WebRtc_UWord32 _periodMs;
- WebRtc_Word64 _latestMs;
-};
-
-enum VCMKeyRequestMode
-{
- kKeyOnError, // Normal mode, request key frames on decoder error
- kKeyOnKeyLoss, // Request key frames on decoder error and on packet loss
- // in key frames.
- kKeyOnLoss, // Request key frames on decoder error and on packet loss
- // in any frame
-};
-
-class VideoCodingModuleImpl : public VideoCodingModule
-{
-public:
- VideoCodingModuleImpl(const WebRtc_Word32 id,
- TickTimeBase* clock,
- bool delete_clock_on_destroy);
-
- virtual ~VideoCodingModuleImpl();
-
- WebRtc_Word32 Id() const;
-
- // Change the unique identifier of this object
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- // Returns the number of milliseconds until the module want a worker thread
- // to call Process
- virtual WebRtc_Word32 TimeUntilNextProcess();
-
- virtual WebRtc_Word32 Process();
-
- /*
- * Sender
- */
-
- // Initialize send codec
- virtual WebRtc_Word32 InitializeSender();
-
- // Register the send codec to be used.
- virtual WebRtc_Word32 RegisterSendCodec(const VideoCodec* sendCodec,
- WebRtc_UWord32 numberOfCores,
- WebRtc_UWord32 maxPayloadSize);
-
- // Get current send codec
- virtual WebRtc_Word32 SendCodec(VideoCodec* currentSendCodec) const;
-
- // Get current send codec type
- virtual VideoCodecType SendCodec() const;
-
- // Register an external encoder object.
- virtual WebRtc_Word32 RegisterExternalEncoder(VideoEncoder* externalEncoder,
- WebRtc_UWord8 payloadType,
- bool internalSource = false);
-
- // Get codec config parameters
- virtual WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* buffer,
- WebRtc_Word32 size);
-
- // Get encode bitrate
- virtual int Bitrate(unsigned int* bitrate) const;
-
- // Get encode frame rate
- virtual int FrameRate(unsigned int* framerate) const;
-
- // Set channel parameters
- virtual WebRtc_Word32 SetChannelParameters(
- WebRtc_UWord32 availableBandWidth,
- WebRtc_UWord8 lossRate,
- WebRtc_UWord32 rtt);
-
- // Set recieve channel parameters
- virtual WebRtc_Word32 SetReceiveChannelParameters(WebRtc_UWord32 rtt);
-
- // Register a transport callback which will be called to deliver the
- // encoded buffers
- virtual WebRtc_Word32 RegisterTransportCallback(
- VCMPacketizationCallback* transport);
-
- // Register a send statistics callback which will be called to deliver
- // information about the video stream produced by the encoder,
- // for instance the average frame rate and bit rate.
- virtual WebRtc_Word32 RegisterSendStatisticsCallback(
- VCMSendStatisticsCallback* sendStats);
-
- // Register a video quality settings callback which will be called when
- // frame rate/dimensions need to be updated for video quality optimization
- virtual WebRtc_Word32 RegisterVideoQMCallback(
- VCMQMSettingsCallback* videoQMSettings);
-
- // Register a video protection callback which will be called to deliver
- // the requested FEC rate and NACK status (on/off).
- virtual WebRtc_Word32 RegisterProtectionCallback(
- VCMProtectionCallback* protection);
-
- // Enable or disable a video protection method.
- virtual WebRtc_Word32 SetVideoProtection(VCMVideoProtection videoProtection,
- bool enable);
-
- // Add one raw video frame to the encoder, blocking.
- virtual WebRtc_Word32 AddVideoFrame(
- const VideoFrame& videoFrame,
- const VideoContentMetrics* _contentMetrics = NULL,
- const CodecSpecificInfo* codecSpecificInfo = NULL);
-
- // Next frame encoded should be of the type frameType.
- virtual WebRtc_Word32 FrameTypeRequest(FrameType frameType,
- WebRtc_UWord8 simulcastIdx);
-
- //Enable frame dropper
- virtual WebRtc_Word32 EnableFrameDropper(bool enable);
-
- // Sent frame counters
- virtual WebRtc_Word32 SentFrameCount(VCMFrameCount& frameCount) const;
-
- /*
- * Receiver
- */
-
- // Initialize receiver, resets codec database etc
- virtual WebRtc_Word32 InitializeReceiver();
-
- // Register possible reveive codecs, can be called multiple times
- virtual WebRtc_Word32 RegisterReceiveCodec(const VideoCodec* receiveCodec,
- WebRtc_Word32 numberOfCores,
- bool requireKeyFrame = false);
-
- // Register an externally defined decoder/render object.
- // Can be a decoder only or a decoder coupled with a renderer.
- virtual WebRtc_Word32 RegisterExternalDecoder(VideoDecoder* externalDecoder,
- WebRtc_UWord8 payloadType,
- bool internalRenderTiming);
-
- // Register a receive callback. Will be called whenever there are a new
- // frame ready for rendering.
- virtual WebRtc_Word32 RegisterReceiveCallback(
- VCMReceiveCallback* receiveCallback);
-
- // Register a receive statistics callback which will be called to deliver
- // information about the video stream received by the receiving side of the
- // VCM, for instance the average frame rate and bit rate.
- virtual WebRtc_Word32 RegisterReceiveStatisticsCallback(
- VCMReceiveStatisticsCallback* receiveStats);
-
- // Register a frame type request callback.
- virtual WebRtc_Word32 RegisterFrameTypeCallback(
- VCMFrameTypeCallback* frameTypeCallback);
-
- // Register a frame storage callback.
- virtual WebRtc_Word32 RegisterFrameStorageCallback(
- VCMFrameStorageCallback* frameStorageCallback);
-
- // Nack callback
- virtual WebRtc_Word32 RegisterPacketRequestCallback(
- VCMPacketRequestCallback* callback);
-
- // Decode next frame, blocks for a maximum of maxWaitTimeMs milliseconds.
- // Should be called as often as possible to get the most out of the decoder.
- virtual WebRtc_Word32 Decode(WebRtc_UWord16 maxWaitTimeMs = 200);
-
- // Decode next dual frame, blocks for a maximum of maxWaitTimeMs
- // milliseconds.
- virtual WebRtc_Word32 DecodeDualFrame(WebRtc_UWord16 maxWaitTimeMs = 200);
-
- // Reset the decoder state
- virtual WebRtc_Word32 ResetDecoder();
-
- // Get current received codec
- virtual WebRtc_Word32 ReceiveCodec(VideoCodec* currentReceiveCodec) const;
-
- // Get current received codec type
- virtual VideoCodecType ReceiveCodec() const;
-
- // Incoming packet from network parsed and ready for decode, non blocking.
- virtual WebRtc_Word32 IncomingPacket(const WebRtc_UWord8* incomingPayload,
- WebRtc_UWord32 payloadLength,
- const WebRtcRTPHeader& rtpInfo);
-
- // A part of an encoded frame to be decoded.
- // Used in conjunction with VCMFrameStorageCallback.
- virtual WebRtc_Word32 DecodeFromStorage(
- const EncodedVideoData& frameFromStorage);
-
- // Minimum playout delay (Used for lip-sync). This is the minimum delay
- // required to sync with audio. Not included in VideoCodingModule::Delay()
- // Defaults to 0 ms.
- virtual WebRtc_Word32 SetMinimumPlayoutDelay(
- WebRtc_UWord32 minPlayoutDelayMs);
-
- // The estimated delay caused by rendering
- virtual WebRtc_Word32 SetRenderDelay(WebRtc_UWord32 timeMS);
-
- // Current delay
- virtual WebRtc_Word32 Delay() const;
-
- // Received frame counters
- virtual WebRtc_Word32 ReceivedFrameCount(VCMFrameCount& frameCount) const;
-
- // Returns the number of packets discarded by the jitter buffer.
- virtual WebRtc_UWord32 DiscardedPackets() const;
-
-
- // Robustness APIs
-
- // Set the sender RTX/NACK mode.
- virtual int SetSenderNackMode(SenderNackMode mode);
-
- // Set the sender reference picture selection (RPS) mode.
- virtual int SetSenderReferenceSelection(bool enable);
-
- // Set the sender forward error correction (FEC) mode.
- virtual int SetSenderFEC(bool enable);
-
- // Set the key frame period, or disable periodic key frames (I-frames).
- virtual int SetSenderKeyFramePeriod(int periodMs);
-
- // Set the receiver robustness mode.
- virtual int SetReceiverRobustnessMode(ReceiverRobustness robustnessMode,
- DecodeErrors errorMode);
-
-protected:
- WebRtc_Word32 Decode(const webrtc::VCMEncodedFrame& frame);
- WebRtc_Word32 RequestKeyFrame();
- WebRtc_Word32 RequestSliceLossIndication(
- const WebRtc_UWord64 pictureID) const;
- WebRtc_Word32 NackList(WebRtc_UWord16* nackList, WebRtc_UWord16& size);
-
-private:
- WebRtc_Word32 _id;
- TickTimeBase* clock_;
- bool delete_clock_on_destroy_;
- CriticalSectionWrapper* _receiveCritSect;
- bool _receiverInited;
- VCMTiming _timing;
- VCMTiming _dualTiming;
- VCMReceiver _receiver;
- VCMReceiver _dualReceiver;
- VCMDecodedFrameCallback _decodedFrameCallback;
- VCMDecodedFrameCallback _dualDecodedFrameCallback;
- VCMFrameTypeCallback* _frameTypeCallback;
- VCMFrameStorageCallback* _frameStorageCallback;
- VCMReceiveStatisticsCallback* _receiveStatsCallback;
- VCMPacketRequestCallback* _packetRequestCallback;
- VCMGenericDecoder* _decoder;
- VCMGenericDecoder* _dualDecoder;
- FILE* _bitStreamBeforeDecoder;
- VCMFrameBuffer _frameFromFile;
- VCMKeyRequestMode _keyRequestMode;
- bool _scheduleKeyRequest;
-
- CriticalSectionWrapper* _sendCritSect; // Critical section for send side
- VCMGenericEncoder* _encoder;
- VCMEncodedFrameCallback _encodedFrameCallback;
- FrameType _nextFrameType[kMaxSimulcastStreams];
- VCMMediaOptimization _mediaOpt;
- VideoCodecType _sendCodecType;
- VCMSendStatisticsCallback* _sendStatsCallback;
- FILE* _encoderInputFile;
-
- VCMCodecDataBase _codecDataBase;
- VCMProcessTimer _receiveStatsTimer;
- VCMProcessTimer _sendStatsTimer;
- VCMProcessTimer _retransmissionTimer;
- VCMProcessTimer _keyRequestTimer;
-};
-} // namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_CODING_VIDEO_CODING_IMPL_H_
diff --git a/trunk/src/modules/video_coding/main/source/video_coding_robustness_unittest.cc b/trunk/src/modules/video_coding/main/source/video_coding_robustness_unittest.cc
deleted file mode 100644
index 0ee9657..0000000
--- a/trunk/src/modules/video_coding/main/source/video_coding_robustness_unittest.cc
+++ /dev/null
@@ -1,396 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "gmock/gmock.h"
-#include "gtest/gtest.h"
-#include "modules/video_coding/codecs/interface/mock/mock_video_codec_interface.h"
-#include "modules/video_coding/main/interface/video_coding.h"
-#include "modules/video_coding/main/interface/mock/mock_vcm_callbacks.h"
-#include "modules/video_coding/main/source/mock/fake_tick_time.h"
-
-namespace webrtc {
-
-using ::testing::Return;
-using ::testing::_;
-using ::testing::ElementsAre;
-using ::testing::AllOf;
-using ::testing::Args;
-using ::testing::Field;
-using ::testing::Pointee;
-using ::testing::NiceMock;
-using ::testing::Sequence;
-
-class VCMRobustnessTest : public ::testing::Test {
- protected:
- static const size_t kPayloadLen = 10;
-
- virtual void SetUp() {
- clock_ = new FakeTickTime(0);
- ASSERT_TRUE(clock_ != NULL);
- vcm_ = VideoCodingModule::Create(0, clock_);
- ASSERT_TRUE(vcm_ != NULL);
- ASSERT_EQ(0, vcm_->InitializeReceiver());
- ASSERT_EQ(0, vcm_->RegisterFrameTypeCallback(&frame_type_callback_));
- ASSERT_EQ(0, vcm_->RegisterPacketRequestCallback(&request_callback_));
- ASSERT_EQ(VCM_OK, vcm_->Codec(kVideoCodecVP8, &video_codec_));
- ASSERT_EQ(VCM_OK, vcm_->RegisterReceiveCodec(&video_codec_, 1));
- ASSERT_EQ(VCM_OK, vcm_->RegisterExternalDecoder(&decoder_,
- video_codec_.plType,
- true));
- }
-
- virtual void TearDown() {
- VideoCodingModule::Destroy(vcm_);
- delete clock_;
- }
-
- void InsertPacket(uint32_t timestamp,
- uint16_t seq_no,
- bool first,
- bool marker_bit,
- FrameType frame_type) {
- const uint8_t payload[kPayloadLen] = {0};
- WebRtcRTPHeader rtp_info;
- memset(&rtp_info, 0, sizeof(rtp_info));
- rtp_info.frameType = frame_type;
- rtp_info.header.timestamp = timestamp;
- rtp_info.header.sequenceNumber = seq_no;
- rtp_info.header.markerBit = marker_bit;
- rtp_info.header.payloadType = video_codec_.plType;
- rtp_info.type.Video.codec = kRTPVideoVP8;
- rtp_info.type.Video.codecHeader.VP8.InitRTPVideoHeaderVP8();
- rtp_info.type.Video.isFirstPacket = first;
-
- ASSERT_EQ(VCM_OK, vcm_->IncomingPacket(payload, kPayloadLen, rtp_info));
- }
-
- VideoCodingModule* vcm_;
- VideoCodec video_codec_;
- MockVCMFrameTypeCallback frame_type_callback_;
- MockPacketRequestCallback request_callback_;
- NiceMock<MockVideoDecoder> decoder_;
- NiceMock<MockVideoDecoder> decoderCopy_;
- FakeTickTime* clock_;
-};
-
-TEST_F(VCMRobustnessTest, TestHardNack) {
- Sequence s;
- EXPECT_CALL(request_callback_, ResendPackets(_, 2))
- .With(Args<0, 1>(ElementsAre(6, 7)))
- .Times(1);
- for (int ts = 0; ts <= 6000; ts += 3000) {
- EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, ts),
- Field(&EncodedImage::_length,
- kPayloadLen * 3),
- Field(&EncodedImage::_completeFrame,
- true)),
- false, _, _, _))
- .Times(1)
- .InSequence(s);
- }
-
- ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
- VideoCodingModule::kHardNack,
- VideoCodingModule::kNoDecodeErrors));
-
- InsertPacket(0, 0, true, false, kVideoFrameKey);
- InsertPacket(0, 1, false, false, kVideoFrameKey);
- InsertPacket(0, 2, false, true, kVideoFrameKey);
-
- InsertPacket(3000, 3, true, false, kVideoFrameDelta);
- InsertPacket(3000, 4, false, false, kVideoFrameDelta);
- InsertPacket(3000, 5, false, true, kVideoFrameDelta);
-
- ASSERT_EQ(VCM_OK, vcm_->Decode(0));
- ASSERT_EQ(VCM_OK, vcm_->Decode(0));
- ASSERT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
-
- clock_->IncrementDebugClock(10);
-
- ASSERT_EQ(VCM_OK, vcm_->Process());
-
- ASSERT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
-
- InsertPacket(6000, 8, false, true, kVideoFrameDelta);
- clock_->IncrementDebugClock(10);
- ASSERT_EQ(VCM_OK, vcm_->Process());
-
- ASSERT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
-
- InsertPacket(6000, 6, true, false, kVideoFrameDelta);
- InsertPacket(6000, 7, false, false, kVideoFrameDelta);
- clock_->IncrementDebugClock(10);
- ASSERT_EQ(VCM_OK, vcm_->Process());
-
- ASSERT_EQ(VCM_OK, vcm_->Decode(0));
-}
-
-TEST_F(VCMRobustnessTest, TestHardNackNoneDecoded) {
- EXPECT_CALL(request_callback_, ResendPackets(_, _))
- .Times(0);
- EXPECT_CALL(frame_type_callback_, RequestKeyFrame())
- .Times(1);
-
- ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
- VideoCodingModule::kHardNack,
- VideoCodingModule::kNoDecodeErrors));
-
- InsertPacket(3000, 3, true, false, kVideoFrameDelta);
- InsertPacket(3000, 4, false, false, kVideoFrameDelta);
- InsertPacket(3000, 5, false, true, kVideoFrameDelta);
-
- EXPECT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
- ASSERT_EQ(VCM_OK, vcm_->Process());
-
- clock_->IncrementDebugClock(10);
-
- EXPECT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
- ASSERT_EQ(VCM_OK, vcm_->Process());
-}
-
-TEST_F(VCMRobustnessTest, TestDualDecoder) {
- Sequence s1, s2;
- EXPECT_CALL(request_callback_, ResendPackets(_, 1))
- .With(Args<0, 1>(ElementsAre(4)))
- .Times(1);
-
- EXPECT_CALL(decoder_, Copy())
- .Times(1)
- .WillOnce(Return(&decoderCopy_));
- EXPECT_CALL(decoderCopy_, Copy())
- .Times(1)
- .WillOnce(Return(&decoder_));
-
- // Decode operations
- EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 0),
- Field(&EncodedImage::_completeFrame,
- true)),
- false, _, _, _))
- .Times(1)
- .InSequence(s1);
- EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 3000),
- Field(&EncodedImage::_completeFrame,
- false)),
- false, _, _, _))
- .Times(1)
- .InSequence(s1);
- EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 6000),
- Field(&EncodedImage::_completeFrame,
- true)),
- false, _, _, _))
- .Times(1)
- .InSequence(s1);
- EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 9000),
- Field(&EncodedImage::_completeFrame,
- true)),
- false, _, _, _))
- .Times(1)
- .InSequence(s1);
-
- EXPECT_CALL(decoderCopy_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 3000),
- Field(&EncodedImage::_completeFrame,
- true)),
- false, _, _, _))
- .Times(1)
- .InSequence(s2);
- EXPECT_CALL(decoderCopy_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 6000),
- Field(&EncodedImage::_completeFrame,
- true)),
- false, _, _, _))
- .Times(1)
- .InSequence(s2);
-
-
- ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
- VideoCodingModule::kDualDecoder,
- VideoCodingModule::kAllowDecodeErrors));
-
- InsertPacket(0, 0, true, false, kVideoFrameKey);
- InsertPacket(0, 1, false, false, kVideoFrameKey);
- InsertPacket(0, 2, false, true, kVideoFrameKey);
- EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 0.
-
- clock_->IncrementDebugClock(33);
- InsertPacket(3000, 3, true, false, kVideoFrameDelta);
- // Packet 4 missing
- InsertPacket(3000, 5, false, true, kVideoFrameDelta);
- EXPECT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
-
- clock_->IncrementDebugClock(33);
- InsertPacket(6000, 6, true, false, kVideoFrameDelta);
- InsertPacket(6000, 7, false, false, kVideoFrameDelta);
- InsertPacket(6000, 8, false, true, kVideoFrameDelta);
-
- EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 3000 incomplete.
- // Spawn a decoder copy.
- EXPECT_EQ(0, vcm_->DecodeDualFrame(0)); // Expect no dual decoder action.
-
- clock_->IncrementDebugClock(10);
- EXPECT_EQ(VCM_OK, vcm_->Process()); // Generate NACK list.
-
- EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 6000 complete.
- EXPECT_EQ(0, vcm_->DecodeDualFrame(0)); // Expect no dual decoder action.
-
- InsertPacket(3000, 4, false, false, kVideoFrameDelta);
- EXPECT_EQ(1, vcm_->DecodeDualFrame(0)); // Dual decode of timestamp 3000.
- EXPECT_EQ(1, vcm_->DecodeDualFrame(0)); // Dual decode of timestamp 6000.
- EXPECT_EQ(0, vcm_->DecodeDualFrame(0)); // No more frames.
-
- InsertPacket(9000, 9, true, false, kVideoFrameDelta);
- InsertPacket(9000, 10, false, false, kVideoFrameDelta);
- InsertPacket(9000, 11, false, true, kVideoFrameDelta);
- EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 9000 complete.
- EXPECT_EQ(0, vcm_->DecodeDualFrame(0)); // Expect no dual decoder action.
-}
-
-TEST_F(VCMRobustnessTest, TestModeNoneWithErrors) {
- EXPECT_CALL(decoder_, InitDecode(_, _)).Times(1);
- EXPECT_CALL(decoder_, Release()).Times(1);
- Sequence s1;
- EXPECT_CALL(request_callback_, ResendPackets(_, 1))
- .With(Args<0, 1>(ElementsAre(4)))
- .Times(0);
-
- EXPECT_CALL(decoder_, Copy())
- .Times(0);
- EXPECT_CALL(decoderCopy_, Copy())
- .Times(0);
-
- // Decode operations
- EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 0),
- Field(&EncodedImage::_completeFrame,
- true)),
- false, _, _, _))
- .Times(1)
- .InSequence(s1);
- EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 3000),
- Field(&EncodedImage::_completeFrame,
- false)),
- false, _, _, _))
- .Times(1)
- .InSequence(s1);
- EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 6000),
- Field(&EncodedImage::_completeFrame,
- true)),
- false, _, _, _))
- .Times(1)
- .InSequence(s1);
- EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 9000),
- Field(&EncodedImage::_completeFrame,
- true)),
- false, _, _, _))
- .Times(1)
- .InSequence(s1);
-
- ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
- VideoCodingModule::kNone,
- VideoCodingModule::kAllowDecodeErrors));
-
- InsertPacket(0, 0, true, false, kVideoFrameKey);
- InsertPacket(0, 1, false, false, kVideoFrameKey);
- InsertPacket(0, 2, false, true, kVideoFrameKey);
- EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 0.
- EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
-
- clock_->IncrementDebugClock(33);
- InsertPacket(3000, 3, true, false, kVideoFrameDelta);
- // Packet 4 missing
- InsertPacket(3000, 5, false, true, kVideoFrameDelta);
- EXPECT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
- EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
-
- clock_->IncrementDebugClock(33);
- InsertPacket(6000, 6, true, false, kVideoFrameDelta);
- InsertPacket(6000, 7, false, false, kVideoFrameDelta);
- InsertPacket(6000, 8, false, true, kVideoFrameDelta);
- EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 3000 incomplete.
- EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
-
- clock_->IncrementDebugClock(10);
- EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 6000 complete.
- EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
-
- clock_->IncrementDebugClock(23);
- InsertPacket(3000, 4, false, false, kVideoFrameDelta);
-
- InsertPacket(9000, 9, true, false, kVideoFrameDelta);
- InsertPacket(9000, 10, false, false, kVideoFrameDelta);
- InsertPacket(9000, 11, false, true, kVideoFrameDelta);
- EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 9000 complete.
-}
-
-TEST_F(VCMRobustnessTest, TestModeNoneWithoutErrors) {
- Sequence s1;
- EXPECT_CALL(decoder_, InitDecode(_, _)).Times(1);
- EXPECT_CALL(decoder_, Release()).Times(1);
- EXPECT_CALL(request_callback_, ResendPackets(_, 1))
- .With(Args<0, 1>(ElementsAre(4)))
- .Times(0);
-
- EXPECT_CALL(decoder_, Copy())
- .Times(0);
- EXPECT_CALL(decoderCopy_, Copy())
- .Times(0);
-
- // Decode operations
- EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 0),
- Field(&EncodedImage::_completeFrame,
- true)),
- false, _, _, _))
- .Times(1)
- .InSequence(s1);
- EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 3000),
- Field(&EncodedImage::_completeFrame,
- false)),
- false, _, _, _))
- .Times(1)
- .InSequence(s1);
- EXPECT_CALL(decoder_, Decode(AllOf(Field(&EncodedImage::_timeStamp, 6000),
- Field(&EncodedImage::_completeFrame,
- true)),
- false, _, _, _))
- .Times(1)
- .InSequence(s1);
- EXPECT_CALL(frame_type_callback_, RequestKeyFrame())
- .Times(1);
-
- ASSERT_EQ(VCM_OK, vcm_->SetReceiverRobustnessMode(
- VideoCodingModule::kNone,
- VideoCodingModule::kNoDecodeErrors));
-
- InsertPacket(0, 0, true, false, kVideoFrameKey);
- InsertPacket(0, 1, false, false, kVideoFrameKey);
- InsertPacket(0, 2, false, true, kVideoFrameKey);
- EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 0.
- EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
-
- clock_->IncrementDebugClock(33);
- InsertPacket(3000, 3, true, false, kVideoFrameDelta);
- // Packet 4 missing
- InsertPacket(3000, 5, false, true, kVideoFrameDelta);
- EXPECT_EQ(VCM_FRAME_NOT_READY, vcm_->Decode(0));
- EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
-
- clock_->IncrementDebugClock(33);
- InsertPacket(6000, 6, true, false, kVideoFrameDelta);
- InsertPacket(6000, 7, false, false, kVideoFrameDelta);
- InsertPacket(6000, 8, false, true, kVideoFrameDelta);
- EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 3000 incomplete.
- // Schedule key frame request.
- EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
-
- clock_->IncrementDebugClock(10);
- EXPECT_EQ(VCM_OK, vcm_->Decode(0)); // Decode timestamp 6000 complete.
- EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect no NACK list.
-
- clock_->IncrementDebugClock(500); // Wait for the key request timer to set.
- EXPECT_EQ(VCM_OK, vcm_->Process()); // Expect key frame request.
-}
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/source/video_coding_test.gypi b/trunk/src/modules/video_coding/main/source/video_coding_test.gypi
deleted file mode 100644
index 7ab265c..0000000
--- a/trunk/src/modules/video_coding/main/source/video_coding_test.gypi
+++ /dev/null
@@ -1,96 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [{
- 'target_name': 'video_coding_test',
- 'type': 'executable',
- 'dependencies': [
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../test/test.gyp:test_support',
- '<(webrtc_root)/../test/metrics.gyp:metrics',
- 'webrtc_video_coding',
- 'rtp_rtcp',
- 'webrtc_utility',
- 'video_processing',
- '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
- ],
- 'include_dirs': [
- '../../../interface',
- '../../codecs/vp8/main/interface',
- '../../../../system_wrappers/interface',
- '../../../../common_video/interface',
- '../source',
- ],
- 'sources': [
- # headers
- '../test/codec_database_test.h',
- '../test/generic_codec_test.h',
- '../test/jitter_estimate_test.h',
- '../test/media_opt_test.h',
- '../test/mt_test_common.h',
- '../test/normal_test.h',
- '../test/quality_modes_test.h',
- '../test/receiver_tests.h',
- '../test/release_test.h',
- '../test/rtp_player.h',
- '../test/test_callbacks.h',
- '../test/test_util.h',
- '../test/video_source.h',
-
- # sources
- '../test/codec_database_test.cc',
- '../test/decode_from_storage_test.cc',
- '../test/generic_codec_test.cc',
- '../test/jitter_buffer_test.cc',
- '../test/media_opt_test.cc',
- '../test/mt_test_common.cc',
- '../test/mt_rx_tx_test.cc',
- '../test/normal_test.cc',
- '../test/quality_modes_test.cc',
- '../test/receiver_timing_tests.cc',
- '../test/rtp_player.cc',
- '../test/test_callbacks.cc',
- '../test/test_util.cc',
- '../test/tester_main.cc',
- '../test/video_rtp_play_mt.cc',
- '../test/video_rtp_play.cc',
- '../test/video_source.cc',
- ], # source
- },
- {
- 'target_name': 'video_coding_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'webrtc_video_coding',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../testing/gmock.gyp:gmock',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '../../../interface',
- '../../codecs/interface',
- ],
- 'sources': [
- '../interface/mock/mock_vcm_callbacks.h',
- 'decoding_state_unittest.cc',
- 'jitter_buffer_unittest.cc',
- 'session_info_unittest.cc',
- 'video_coding_robustness_unittest.cc',
- 'qm_select_unittest.cc',
- ],
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/video_coding/main/test/codec_database_test.cc b/trunk/src/modules/video_coding/main/test/codec_database_test.cc
deleted file mode 100644
index f9ea3e5..0000000
--- a/trunk/src/modules/video_coding/main/test/codec_database_test.cc
+++ /dev/null
@@ -1,408 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Implementation of codec data base test
-// testing is done via the VCM module, no specific CodecDataBase module functionality.
-
-#include "codec_database_test.h"
-
-#include <assert.h>
-#include <stdio.h>
-
-#include "../../../../engine_configurations.h"
-#include "../source/event.h"
-#include "test_callbacks.h"
-#include "test_macros.h"
-#include "test_util.h"
-#include "testsupport/fileutils.h"
-#include "testsupport/metrics/video_metrics.h"
-#include "vp8.h" // for external codecs test
-
-
-using namespace webrtc;
-
-int CodecDataBaseTest::RunTest(CmdArgs& args)
-{
- VideoCodingModule* vcm = VideoCodingModule::Create(1);
- CodecDataBaseTest* cdbt = new CodecDataBaseTest(vcm);
- cdbt->Perform(args);
- VideoCodingModule::Destroy(vcm);
- delete cdbt;
- return 0;
-
-}
-
-CodecDataBaseTest::CodecDataBaseTest(VideoCodingModule* vcm):
-_vcm(vcm),
-_width(0),
-_height(0),
-_lengthSourceFrame(0),
-_timeStamp(0)
-{
- //
-}
-CodecDataBaseTest::~CodecDataBaseTest()
-{
- //
-}
-void
-CodecDataBaseTest::Setup(CmdArgs& args)
-{
- _inname= args.inputFile;
- _width = args.width;
- _height = args.height;
- _frameRate = args.frameRate;
- _lengthSourceFrame = 3*_width*_height/2;
- if (args.outputFile.compare(""))
- _outname = test::OutputPath() + "CDBtest_decoded.yuv";
- else
- _outname = args.outputFile;
- _outname = args.outputFile;
- _encodedName = test::OutputPath() + "CDBtest_encoded.vp8";
-
- if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
- {
- printf("Cannot read file %s.\n", _inname.c_str());
- exit(1);
- }
-
- if ((_encodedFile = fopen(_encodedName.c_str(), "wb")) == NULL)
- {
- printf("Cannot write encoded file.\n");
- exit(1);
- }
-
- if ((_decodedFile = fopen(_outname.c_str(), "wb")) == NULL)
- {
- printf("Cannot write file %s.\n", _outname.c_str());
- exit(1);
- }
-
- return;
-}
-
-
-
-WebRtc_Word32
-CodecDataBaseTest::Perform(CmdArgs& args)
-{
-#ifndef VIDEOCODEC_VP8
- assert(false);
-#endif
- Setup(args);
- EventWrapper* waitEvent = EventWrapper::Create();
-
- /**************************/
- /* General Sanity Checks */
- /************************/
- VideoCodec sendCodec, receiveCodec;
- TEST(VideoCodingModule::NumberOfCodecs() > 0);
- _vcm->InitializeReceiver();
- _vcm->InitializeSender();
- VCMDecodeCompleteCallback *_decodeCallback = new VCMDecodeCompleteCallback(_decodedFile);
- VCMEncodeCompleteCallback *_encodeCompleteCallback = new VCMEncodeCompleteCallback(_encodedFile);
- _vcm->RegisterReceiveCallback(_decodeCallback);
- _vcm->RegisterTransportCallback(_encodeCompleteCallback);
- _encodeCompleteCallback->SetFrameDimensions(_width, _height);
- // registering the callback - encode and decode with the same vcm (could be later changed)
- _encodeCompleteCallback->RegisterReceiverVCM(_vcm);
- // preparing a frame to be encoded
- VideoFrame sourceFrame;
- sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
- WebRtc_UWord8* tmpBuffer = new WebRtc_UWord8[_lengthSourceFrame];
- TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
- sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
- sourceFrame.SetHeight(_height);
- sourceFrame.SetWidth(_width);
- _timeStamp += (WebRtc_UWord32)(9e4 / _frameRate);
- sourceFrame.SetTimeStamp(_timeStamp);
- // Encoder registration
- TEST (VideoCodingModule::NumberOfCodecs() > 0);
- TEST(VideoCodingModule::Codec(-1, &sendCodec) == VCM_PARAMETER_ERROR);
- TEST(VideoCodingModule::Codec(VideoCodingModule::NumberOfCodecs() + 1, &sendCodec) == VCM_PARAMETER_ERROR);
- VideoCodingModule::Codec(1, &sendCodec);
- sendCodec.plType = 0; // random value
- TEST(_vcm->RegisterSendCodec(&sendCodec, 1, 1440) < 0);
- _vcm->InitializeReceiver();
- _vcm->InitializeSender();
- _vcm->RegisterReceiveCallback(_decodeCallback);
- _vcm->RegisterTransportCallback(_encodeCompleteCallback);
- printf(" \nNumber of Registered Codecs: %d \n\n", VideoCodingModule::NumberOfCodecs());
- printf("Registered codec names: ");
- for (int i=0; i < VideoCodingModule::NumberOfCodecs(); i++)
- {
- VideoCodingModule::Codec(i, &sendCodec);
- printf("%s ", sendCodec.plName);
- }
- printf("\n\nVerify that all requested codecs are used\n \n \n");
-
- // Testing with VP8.
- VideoCodingModule::Codec(kVideoCodecVP8, &sendCodec);
- _vcm->RegisterSendCodec(&sendCodec, 1, 1440);
- _encodeCompleteCallback->SetCodecType(kRTPVideoVP8);
- _vcm->InitializeReceiver();
- TEST (_vcm->AddVideoFrame(sourceFrame) == VCM_OK );
- _vcm->InitializeSender();
- TEST (_vcm->AddVideoFrame(sourceFrame) < 0 );
-
- // Test changing frame size while keeping the same payload type
- VideoCodingModule::Codec(0, &sendCodec);
- sendCodec.width = 352;
- sendCodec.height = 288;
- VideoCodec currentSendCodec;
- _vcm->RegisterSendCodec(&sendCodec, 1, 1440);
- _vcm->SendCodec(¤tSendCodec);
- TEST(currentSendCodec.width == sendCodec.width &&
- currentSendCodec.height == sendCodec.height);
- sendCodec.width = 352/2;
- sendCodec.height = 288/2;
- _vcm->RegisterSendCodec(&sendCodec, 1, 1440);
- _vcm->SendCodec(¤tSendCodec);
- TEST(currentSendCodec.width == sendCodec.width &&
- currentSendCodec.height == sendCodec.height);
-
- delete _decodeCallback;
- _decodeCallback = NULL;
- delete _encodeCompleteCallback;
- _encodeCompleteCallback = NULL;
-
- VCMEncodeCompleteCallback *_encodeCallback = new VCMEncodeCompleteCallback(_encodedFile);
-
- /*************************/
- /* External codecs */
- /*************************/
-
-
- _vcm->InitializeReceiver();
- VP8Decoder* decoder = VP8Decoder::Create();
- VideoCodec vp8DecSettings;
- VideoCodingModule::Codec(kVideoCodecVP8, &vp8DecSettings);
- TEST(_vcm->RegisterExternalDecoder(decoder, vp8DecSettings.plType, false) == VCM_OK);
- TEST(_vcm->RegisterReceiveCodec(&vp8DecSettings, 1, false) == VCM_OK);
- VP8Encoder* encoder = VP8Encoder::Create();
- VideoCodec vp8EncSettings;
- VideoCodingModule::Codec(kVideoCodecVP8, &vp8EncSettings);
- _vcm->RegisterTransportCallback(_encodeCallback); // encode returns error if callback uninitialized
- _encodeCallback->RegisterReceiverVCM(_vcm);
- _encodeCallback->SetCodecType(kRTPVideoVP8);
- TEST(_vcm->RegisterExternalEncoder(encoder, vp8EncSettings.plType) == VCM_OK);
- TEST(_vcm->RegisterSendCodec(&vp8EncSettings, 4, 1440) == VCM_OK);
- TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
- TEST(_vcm->Decode() == VCM_OK);
- waitEvent->Wait(33);
- _timeStamp += (WebRtc_UWord32)(9e4 / _frameRate);
- sourceFrame.SetTimeStamp(_timeStamp);
- TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
- TEST(_vcm->Decode() == VCM_OK);
-
- // De-register and try again.
- TEST(_vcm->RegisterExternalDecoder(NULL, vp8DecSettings.plType, false) == VCM_OK);
- TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
- TEST(_vcm->Decode() < 0); // Expect an error since we have de-registered the decoder
- TEST(_vcm->RegisterExternalEncoder(NULL, vp8DecSettings.plType) == VCM_OK);
- TEST(_vcm->AddVideoFrame(sourceFrame) < 0); // No send codec registered
-
- delete decoder;
- decoder = NULL;
- delete encoder;
- encoder = NULL;
-
- /***************************************
- * Test the "require key frame" setting*
- ***************************************/
-
- TEST(_vcm->InitializeSender() == VCM_OK);
- TEST(_vcm->InitializeReceiver() == VCM_OK);
- VideoCodingModule::Codec(kVideoCodecVP8, &receiveCodec);
- receiveCodec.height = _height;
- receiveCodec.width = _width;
- TEST(_vcm->RegisterSendCodec(&receiveCodec, 4, 1440) == VCM_OK);
- TEST(_vcm->RegisterReceiveCodec(&receiveCodec, 1, true) == VCM_OK); // Require key frame
- _vcm->RegisterTransportCallback(_encodeCallback); // encode returns error if callback uninitialized
- _encodeCallback->RegisterReceiverVCM(_vcm);
- _encodeCallback->SetCodecType(kRTPVideoVP8);
- TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
- TEST(_vcm->Decode() == VCM_OK);
- TEST(_vcm->ResetDecoder() == VCM_OK);
- waitEvent->Wait(33);
- _timeStamp += (WebRtc_UWord32)(9e4 / _frameRate);
- sourceFrame.SetTimeStamp(_timeStamp);
- TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
- // Try to decode a delta frame. Should get a warning since we have enabled the "require key frame" setting
- // and because no frame type request callback has been registered.
- TEST(_vcm->Decode() == VCM_MISSING_CALLBACK);
- TEST(_vcm->FrameTypeRequest(kVideoFrameKey, 0) == VCM_OK);
- _timeStamp += (WebRtc_UWord32)(9e4 / _frameRate);
- sourceFrame.SetTimeStamp(_timeStamp);
- TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
- TEST(_vcm->Decode() == VCM_OK);
-
- // Make sure we can register another codec with the same
- // payload type without crash.
- _vcm->InitializeReceiver();
- sendCodec.width = _width;
- sendCodec.height = _height;
- TEST(_vcm->RegisterReceiveCodec(&sendCodec, 1) == VCM_OK);
- TEST(_vcm->FrameTypeRequest(kVideoFrameKey, 0) == VCM_OK);
- waitEvent->Wait(33);
- _timeStamp += (WebRtc_UWord32)(9e4 / _frameRate);
- sourceFrame.SetTimeStamp(_timeStamp);
- TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
- TEST(_vcm->Decode() == VCM_OK);
- TEST(_vcm->RegisterReceiveCodec(&sendCodec, 1) == VCM_OK);
- waitEvent->Wait(33);
- _timeStamp += (WebRtc_UWord32)(9e4 / _frameRate);
- sourceFrame.SetTimeStamp(_timeStamp);
- TEST(_vcm->FrameTypeRequest(kVideoFrameKey, 0) == VCM_OK);
- TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
- TEST(_vcm->Decode() == VCM_OK);
- TEST(_vcm->ResetDecoder() == VCM_OK);
-
- delete _encodeCallback;
-
- /*************************/
- /* Send/Receive Control */
- /***********************/
- /*
- 1. check available codecs (N)
- 2. register all corresponding decoders
- 3. encode 300/N frames with each encoder, and hope to properly decode
- 4. encode without a matching decoder - expect an error
- */
- rewind(_sourceFile);
- _vcm->InitializeReceiver();
- _vcm->InitializeSender();
- sourceFrame.Free();
- VCMDecodeCompleteCallback* decodeCallCDT = new VCMDecodeCompleteCallback(_decodedFile);
- VCMEncodeCompleteCallback* encodeCallCDT = new VCMEncodeCompleteCallback(_encodedFile);
- _vcm->RegisterReceiveCallback(decodeCallCDT);
- _vcm->RegisterTransportCallback(encodeCallCDT);
- encodeCallCDT->RegisterReceiverVCM(_vcm);
- if (VideoCodingModule::NumberOfCodecs() > 0)
- {
- // Register all available decoders.
- int i, j;
- //double psnr;
- sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
- _vcm->RegisterReceiveCallback(decodeCallCDT);
- for (i=0; i < VideoCodingModule::NumberOfCodecs(); i++)
- {
- VideoCodingModule::Codec(i, &receiveCodec);
- if (strcmp(receiveCodec.plName, "I420") == 0)
- {
- receiveCodec.height = _height;
- receiveCodec.width = _width;
- }
- _vcm->RegisterReceiveCodec(&receiveCodec, 1);
- }
- // start encoding - iterating over available encoders
- _vcm->RegisterTransportCallback(encodeCallCDT);
- encodeCallCDT->RegisterReceiverVCM(_vcm);
- encodeCallCDT->Initialize();
- int frameCnt = 0;
- for (i=0; i < VideoCodingModule::NumberOfCodecs(); i++)
- {
- encodeCallCDT->ResetByteCount();
- VideoCodingModule::Codec(i, &sendCodec);
- sendCodec.height = _height;
- sendCodec.width = _width;
- sendCodec.startBitrate = 1000;
- sendCodec.maxBitrate = 8000;
- encodeCallCDT->SetFrameDimensions(_width, _height);
- encodeCallCDT->SetCodecType(ConvertCodecType(sendCodec.plName));
- TEST(_vcm->RegisterSendCodec(&sendCodec, 1, 1440) == VCM_OK);
-
- // We disable the frame dropper to avoid dropping frames due to
- // bad rate control. This isn't a codec performance test, and the
- // I420 codec is expected to produce too many bits.
- _vcm->EnableFrameDropper(false);
-
- printf("Encoding with %s \n\n", sendCodec.plName);
- for (j=0; j < int(300/VideoCodingModule::NumberOfCodecs()); j++)// assuming 300 frames, NumberOfCodecs <= 10
- {
- frameCnt++;
- TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
- // building source frame
- sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
- sourceFrame.SetHeight(_height);
- sourceFrame.SetWidth(_width);
- sourceFrame.SetLength(_lengthSourceFrame);
- _timeStamp += (WebRtc_UWord32)(9e4 / _frameRate);
- sourceFrame.SetTimeStamp(_timeStamp);
- // send frame to the encoder
- TEST (_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
- waitEvent->Wait(33); // was 100
-
- int ret =_vcm->Decode();
- TEST(ret == 0);
- if (ret < 0)
- {
- printf("Error #%d in frame number %d \n",ret, frameCnt);
- }
- // verifying matching payload types:
- _vcm->SendCodec(&sendCodec);
- _vcm->ReceiveCodec(&receiveCodec);
- TEST(sendCodec.plType == receiveCodec.plType);
- if (sendCodec.plType != receiveCodec.plType)
- {
- printf("frame number:%d\n",frameCnt);
- }
- } // end for:encode-decode
- // byte count for codec specific
-
- printf("Total bytes encoded: %f \n\n",(8.0/1000)*(encodeCallCDT->EncodedBytes()/((int)10/VideoCodingModule::NumberOfCodecs())));
- // decode what's left in the buffer....
- _vcm->Decode();
- _vcm->Decode();
- // Don't measure PSNR for I420 since it will be perfect.
- if (sendCodec.codecType != kVideoCodecI420) {
- webrtc::test::QualityMetricsResult psnr;
- I420PSNRFromFiles(_inname.c_str(), _outname.c_str(), _width,
- _height, &psnr);
- printf("\n @ %d KBPS: ", sendCodec.startBitrate);
- printf("PSNR from encoder-decoder send-receive control test"
- "is %f\n\n", psnr.average);
- }
- } // end: iterate codecs
- rewind(_sourceFile);
- sourceFrame.Free();
- delete [] tmpBuffer;
- delete decodeCallCDT;
- delete encodeCallCDT;
- // closing and calculating PSNR for prior encoder-decoder test
- TearDown(); // closing open files
- } // end of #codecs >1
-
- delete waitEvent;
- Print();
- return 0;
-}
-void
-CodecDataBaseTest::Print()
-{
- printf("\nVCM Codec DataBase Test: \n\n%i tests completed\n", vcmMacrosTests);
- if (vcmMacrosErrors > 0)
- {
- printf("%i FAILED\n\n", vcmMacrosErrors);
- }
- else
- {
- printf("ALL PASSED\n\n");
- }
-}
-
-void
-CodecDataBaseTest::TearDown()
-{
- fclose(_sourceFile);
- fclose(_decodedFile);
- fclose(_encodedFile);
- return;
-}
diff --git a/trunk/src/modules/video_coding/main/test/codec_database_test.h b/trunk/src/modules/video_coding/main/test/codec_database_test.h
deleted file mode 100644
index cc33e05..0000000
--- a/trunk/src/modules/video_coding/main/test/codec_database_test.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_CODEC_DATABASE_TEST_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TEST_CODEC_DATABASE_TEST_H_
-
-#include "video_coding.h"
-#include "test_util.h"
-
-#include <string.h>
-
-/*
-Test consists of:
-1. Sanity chacks: Send and Receive side (bad input, etc. )
-2. Send-side control (encoder registration etc.)
-3. Decoder-side control - encode with various encoders, and verify correct decoding
-*/
-
-class CodecDataBaseTest
-{
-public:
- CodecDataBaseTest(webrtc::VideoCodingModule* vcm);
- ~CodecDataBaseTest();
- static int RunTest(CmdArgs& args);
- WebRtc_Word32 Perform(CmdArgs& args);
-private:
- void TearDown();
- void Setup(CmdArgs& args);
- void Print();
- webrtc::VideoCodingModule* _vcm;
- std::string _inname;
- std::string _outname;
- std::string _encodedName;
- FILE* _sourceFile;
- FILE* _decodedFile;
- FILE* _encodedFile;
- WebRtc_UWord16 _width;
- WebRtc_UWord16 _height;
- WebRtc_UWord32 _lengthSourceFrame;
- WebRtc_UWord32 _timeStamp;
- float _frameRate;
-}; // end of codecDBTest class definition
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_CODEC_DATABASE_TEST_H_
diff --git a/trunk/src/modules/video_coding/main/test/decode_from_storage_test.cc b/trunk/src/modules/video_coding/main/test/decode_from_storage_test.cc
deleted file mode 100644
index 628d509..0000000
--- a/trunk/src/modules/video_coding/main/test/decode_from_storage_test.cc
+++ /dev/null
@@ -1,177 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "receiver_tests.h"
-#include "video_coding.h"
-#include "rtp_rtcp.h"
-#include "trace.h"
-#include "../source/event.h"
-#include "rtp_player.h"
-#include "modules/video_coding/main/source/mock/fake_tick_time.h"
-
-using namespace webrtc;
-
-class FrameStorageCallback : public VCMFrameStorageCallback
-{
-public:
- FrameStorageCallback(VideoCodingModule* vcm) : _vcm(vcm) {}
-
- WebRtc_Word32 StoreReceivedFrame(const EncodedVideoData& frameToStore)
- {
- _vcm->DecodeFromStorage(frameToStore);
- return VCM_OK;
- }
-
-private:
- VideoCodingModule* _vcm;
-};
-
-int DecodeFromStorageTest(CmdArgs& args)
-{
- // Make sure this test isn't executed without simulated events.
-#if !defined(EVENT_DEBUG)
- return -1;
-#endif
- // BEGIN Settings
-
- bool protectionEnabled = false;
- VCMVideoProtection protectionMethod = kProtectionNack;
- WebRtc_UWord32 rttMS = 100;
- float lossRate = 0.00f;
- bool reordering = false;
- WebRtc_UWord32 renderDelayMs = 0;
- WebRtc_UWord32 minPlayoutDelayMs = 0;
- const WebRtc_Word64 MAX_RUNTIME_MS = -1;
- std::string rtpFilename = args.inputFile;
- std::string outFilename = args.outputFile;
- if (outFilename == "")
- outFilename = test::OutputPath() + "DecodeFromStorage.yuv";
-
- FrameReceiveCallback receiveCallback(outFilename.c_str());
-
- // END Settings
-
- Trace::CreateTrace();
- Trace::SetTraceFile(
- (test::OutputPath() + "decodeFromStorageTestTrace.txt").c_str());
- Trace::SetLevelFilter(webrtc::kTraceAll);
-
-
- FakeTickTime clock(0);
- // TODO(hlundin): This test was not verified after changing to FakeTickTime.
- VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
- VideoCodingModule* vcmPlayback = VideoCodingModule::Create(2, &clock);
- FrameStorageCallback storageCallback(vcmPlayback);
- RtpDataCallback dataCallback(vcm);
- WebRtc_Word32 ret = vcm->InitializeReceiver();
- if (ret < 0)
- {
- return -1;
- }
- ret = vcmPlayback->InitializeReceiver();
- if (ret < 0)
- {
- return -1;
- }
- vcm->RegisterFrameStorageCallback(&storageCallback);
- vcmPlayback->RegisterReceiveCallback(&receiveCallback);
- RTPPlayer rtpStream(rtpFilename.c_str(), &dataCallback, &clock);
- PayloadTypeList payloadTypes;
- payloadTypes.push_front(new PayloadCodecTuple(VCM_VP8_PAYLOAD_TYPE, "VP8",
- kVideoCodecVP8));
-
- // Register receive codecs in VCM
- for (PayloadTypeList::iterator it = payloadTypes.begin();
- it != payloadTypes.end(); ++it) {
- PayloadCodecTuple* payloadType = *it;
- if (payloadType != NULL)
- {
- VideoCodec codec;
- memset(&codec, 0, sizeof(codec));
- strncpy(codec.plName, payloadType->name.c_str(), payloadType->name.length());
- codec.plName[payloadType->name.length()] = '\0';
- codec.plType = payloadType->payloadType;
- codec.codecType = payloadType->codecType;
- if (vcm->RegisterReceiveCodec(&codec, 1) < 0)
- {
- return -1;
- }
- if (vcmPlayback->RegisterReceiveCodec(&codec, 1) < 0)
- {
- return -1;
- }
- }
- }
- if (rtpStream.Initialize(&payloadTypes) < 0)
- {
- return -1;
- }
- bool nackEnabled = protectionEnabled && (protectionMethod == kProtectionNack ||
- protectionMethod == kProtectionDualDecoder);
- rtpStream.SimulatePacketLoss(lossRate, nackEnabled, rttMS);
- rtpStream.SetReordering(reordering);
- vcm->SetChannelParameters(0, 0, rttMS);
- vcm->SetVideoProtection(protectionMethod, protectionEnabled);
- vcm->SetRenderDelay(renderDelayMs);
- vcm->SetMinimumPlayoutDelay(minPlayoutDelayMs);
-
- ret = 0;
-
- // RTP stream main loop
- while ((ret = rtpStream.NextPacket(clock.MillisecondTimestamp())) == 0)
- {
- if (clock.MillisecondTimestamp() % 5 == 0)
- {
- ret = vcm->Decode();
- if (ret < 0)
- {
- return -1;
- }
- }
- if (vcm->TimeUntilNextProcess() <= 0)
- {
- vcm->Process();
- }
- if (MAX_RUNTIME_MS > -1 && clock.MillisecondTimestamp() >= MAX_RUNTIME_MS)
- {
- break;
- }
- clock.IncrementDebugClock(1);
- }
-
- switch (ret)
- {
- case 1:
- printf("Success\n");
- break;
- case -1:
- printf("Failed\n");
- break;
- case 0:
- printf("Timeout\n");
- break;
- }
-
- rtpStream.Print();
-
- // Tear down
- while (!payloadTypes.empty())
- {
- delete payloadTypes.front();
- payloadTypes.pop_front();
- }
- VideoCodingModule::Destroy(vcm);
- vcm = NULL;
- VideoCodingModule::Destroy(vcmPlayback);
- vcmPlayback = NULL;
- Trace::ReturnTrace();
-
- return 0;
-}
diff --git a/trunk/src/modules/video_coding/main/test/generic_codec_test.cc b/trunk/src/modules/video_coding/main/test/generic_codec_test.cc
deleted file mode 100644
index 56b3c86..0000000
--- a/trunk/src/modules/video_coding/main/test/generic_codec_test.cc
+++ /dev/null
@@ -1,587 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "generic_codec_test.h"
-#include <cmath>
-#include <stdio.h>
-#include "../source/event.h"
-#include "rtp_rtcp.h"
-#include "module_common_types.h"
-#include "test_macros.h"
-#include "modules/video_coding/main/source/mock/fake_tick_time.h"
-
-using namespace webrtc;
-
-enum { kMaxWaitEncTimeMs = 100 };
-
-int GenericCodecTest::RunTest(CmdArgs& args)
-{
-#if !defined(EVENT_DEBUG)
- printf("\n\nEnable debug events to run this test!\n\n");
- return -1;
-#endif
- FakeTickTime clock(0);
- VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
- GenericCodecTest* get = new GenericCodecTest(vcm, &clock);
- Trace::CreateTrace();
- Trace::SetTraceFile(
- (test::OutputPath() + "genericCodecTestTrace.txt").c_str());
- Trace::SetLevelFilter(webrtc::kTraceAll);
- get->Perform(args);
- Trace::ReturnTrace();
- delete get;
- VideoCodingModule::Destroy(vcm);
- return 0;
-}
-
-GenericCodecTest::GenericCodecTest(VideoCodingModule* vcm, FakeTickTime* clock):
-_clock(clock),
-_vcm(vcm),
-_width(0),
-_height(0),
-_frameRate(0),
-_lengthSourceFrame(0),
-_timeStamp(0)
-{
-}
-
-GenericCodecTest::~GenericCodecTest()
-{
-}
-
-void
-GenericCodecTest::Setup(CmdArgs& args)
-{
- _timeStamp = 0;
-
- /* Test Sequence parameters */
-
- _inname= args.inputFile;
- if (args.outputFile.compare(""))
- _outname = test::OutputPath() + "GCTest_decoded.yuv";
- else
- _outname = args.outputFile;
- _encodedName = test::OutputPath() + "GCTest_encoded.vp8";
- _width = args.width;
- _height = args.height;
- _frameRate = args.frameRate;
- _lengthSourceFrame = 3*_width*_height/2;
-
- /* File settings */
-
- if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
- {
- printf("Cannot read file %s.\n", _inname.c_str());
- exit(1);
- }
- if ((_encodedFile = fopen(_encodedName.c_str(), "wb")) == NULL)
- {
- printf("Cannot write encoded file.\n");
- exit(1);
- }
- if ((_decodedFile = fopen(_outname.c_str(), "wb")) == NULL)
- {
- printf("Cannot write file %s.\n", _outname.c_str());
- exit(1);
- }
-
- return;
-}
-WebRtc_Word32
-GenericCodecTest::Perform(CmdArgs& args)
-{
- WebRtc_Word32 ret;
- Setup(args);
- /*
- 1. sanity checks
- 2. encode/decoder individuality
- 3. API testing
- 4. Target bitrate (within a specific timespan)
- 5. Pipeline Delay
- */
-
- /*******************************/
- /* sanity checks on inputs */
- /*****************************/
- VideoCodec sendCodec, receiveCodec;
- sendCodec.maxBitrate = 8000;
- TEST(_vcm->NumberOfCodecs() > 0); // This works since we now initialize the list in the constructor
- TEST(_vcm->Codec(0, &sendCodec) == VCM_OK);
- _vcm->InitializeSender();
- _vcm->InitializeReceiver();
- WebRtc_Word32 NumberOfCodecs = _vcm->NumberOfCodecs();
- // registration of first codec in the list
- int i = 0;
- _vcm->Codec(0, &_sendCodec);
- TEST(_vcm->RegisterSendCodec(&_sendCodec, 4, 1440) == VCM_OK);
- // sanity on encoder registration
- VideoFrame sourceFrame;
- sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
- _vcm->InitializeSender();
- TEST(_vcm->Codec(kVideoCodecVP8, &sendCodec) == 0);
- TEST(_vcm->RegisterSendCodec(&sendCodec, -1, 1440) < 0); // bad number of cores
- sendCodec.maxBitrate = 8000;
- _vcm->RegisterSendCodec(&sendCodec, 1, 1440);
- _vcm->InitializeSender();
- _vcm->Codec(kVideoCodecVP8, &sendCodec);
- sendCodec.height = 0;
- TEST(_vcm->RegisterSendCodec(&sendCodec, 1, 1440) < 0); // bad height
- _vcm->Codec(kVideoCodecVP8, &sendCodec);
- sendCodec.startBitrate = -2;
- TEST(_vcm->RegisterSendCodec(&sendCodec, 1, 1440) < 0); // bad bit rate
- _vcm->Codec(kVideoCodecVP8, &sendCodec);
- _vcm->InitializeSender();
- TEST(_vcm->SetChannelParameters(100, 0, 0) < 0);// setting rate when encoder uninitialized
- // register all availbale decoders -- need to have more for this test
- for (i=0; i< NumberOfCodecs; i++)
- {
- _vcm->Codec(i, &receiveCodec);
- _vcm->RegisterReceiveCodec(&receiveCodec, 1);
- }
- WebRtc_UWord8* tmpBuffer = new WebRtc_UWord8[_lengthSourceFrame];
- TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
- // building source frame
- sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
- sourceFrame.SetHeight(_height);
- sourceFrame.SetWidth(_width);
- sourceFrame.SetTimeStamp(_timeStamp++);
- // encode/decode
- TEST(_vcm->AddVideoFrame(sourceFrame) < 0 ); // encoder uninitialized
- _vcm->InitializeReceiver();
- TEST(_vcm->SetChannelParameters(100, 0, 0) < 0);// setting rtt when receiver uninitialized
-
- /**************************************/
- /* encoder/decoder individuality test */
- /**************************************/
- //Register both encoder and decoder, reset decoder - encode, set up decoder, reset encoder - decode.
- rewind(_sourceFile);
- sourceFrame.Free();
- _vcm->InitializeReceiver();
- _vcm->InitializeSender();
- NumberOfCodecs = _vcm->NumberOfCodecs();
- // Register VP8
- _vcm->Codec(kVideoCodecVP8, &_sendCodec);
- _vcm->RegisterSendCodec(&_sendCodec, 4, 1440);
- _vcm->SendCodec(&sendCodec);
- sendCodec.startBitrate = 2000;
-
- // Set target frame rate to half of the incoming frame rate
- // to test the frame rate control in the VCM
- sendCodec.maxFramerate = (WebRtc_UWord8)(_frameRate / 2);
- sendCodec.width = _width;
- sendCodec.height = _height;
- TEST(strncmp(_sendCodec.plName, "VP8", 3) == 0); // was VP8
-
- _decodeCallback = new VCMDecodeCompleteCallback(_decodedFile);
- _encodeCompleteCallback = new VCMEncodeCompleteCallback(_encodedFile);
- _vcm->RegisterReceiveCallback(_decodeCallback);
- _vcm->RegisterTransportCallback(_encodeCompleteCallback);
- _encodeCompleteCallback->RegisterReceiverVCM(_vcm);
-
- _vcm->RegisterSendCodec(&sendCodec, 4, 1440);
- _encodeCompleteCallback->SetCodecType(ConvertCodecType(sendCodec.plName));
-
- _vcm->InitializeReceiver();
- _vcm->Process();
-
- //encoding 1 second of video
- for (i = 0; i < _frameRate; i++)
- {
- TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
- sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
- sourceFrame.SetHeight(_height);
- sourceFrame.SetWidth(_width);
- _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(_frameRate));
- sourceFrame.SetTimeStamp(_timeStamp);
- TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
- IncrementDebugClock(_frameRate);
- _vcm->Process();
- }
- sendCodec.maxFramerate = (WebRtc_UWord8)_frameRate;
- _vcm->InitializeSender();
- TEST(_vcm->RegisterReceiveCodec(&sendCodec, 1) == VCM_OK); // same codec for encode and decode
- ret = 0;
- i = 0;
- while ((i < 25) && (ret == 0) )
- {
- ret = _vcm->Decode();
- TEST(ret == VCM_OK);
- if (ret < 0)
- {
- printf("error in frame # %d \n", i);
- }
- IncrementDebugClock(_frameRate);
- i++;
- }
- //TEST((ret == 0) && (i = 50));
- if (ret == 0)
- {
- printf("Encoder/Decoder individuality test complete - View output files \n");
- }
- // last frame - not decoded
- _vcm->InitializeReceiver();
- TEST(_vcm->Decode() < 0); // frame to be encoded exists, decoder uninitialized
-
-
- // Test key frame request on packet loss mode.
- // This a frame as a key frame and fooling the receiver
- // that the last packet was lost. The decoding will succeed,
- // but the VCM will see a packet loss and request a new key frame.
- VCMEncComplete_KeyReqTest keyReqTest_EncCompleteCallback(*_vcm);
- KeyFrameReqTest frameTypeCallback;
- _vcm->RegisterTransportCallback(&keyReqTest_EncCompleteCallback);
- _encodeCompleteCallback->RegisterReceiverVCM(_vcm);
- _vcm->RegisterSendCodec(&sendCodec, 4, 1440);
- _encodeCompleteCallback->SetCodecType(ConvertCodecType(sendCodec.plName));
- TEST(_vcm->SetVideoProtection(kProtectionKeyOnKeyLoss, true) == VCM_OK);
- TEST(_vcm->RegisterFrameTypeCallback(&frameTypeCallback) == VCM_OK);
- TEST(_vcm->RegisterReceiveCodec(&sendCodec, 1) == VCM_OK);
- TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
- _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(_frameRate));
- sourceFrame.SetTimeStamp(_timeStamp);
- // First packet of a subsequent frame required before the jitter buffer
- // will allow decoding an incomplete frame.
- TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
- TEST(_vcm->Decode() == VCM_OK);
-
- printf("API tests complete \n");
-
- /*******************/
- /* Bit Rate Tests */
- /*****************/
- /* Requirements:
- * 1. OneSecReq = 15 % above/below target over a time period of 1s (_frameRate number of frames)
- * 3. FullReq = 10% for total seq. (for 300 frames/seq. coincides with #1)
- * 4. Test will go over all registered codecs
- //NOTE: time requirements are not part of the release tests
- */
- double FullReq = 0.1;
- //double OneSecReq = 0.15;
- printf("\n RATE CONTROL TEST\n");
- // initializing....
- _vcm->InitializeSender();
- _vcm->InitializeReceiver();
- rewind(_sourceFile);
- sourceFrame.Free();
- sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
- const float bitRate[] = {100, 400, 600, 1000, 2000};
- const float nBitrates = sizeof(bitRate)/sizeof(*bitRate);
- float _bitRate = 0;
- int _frameCnt = 0;
- float totalBytesOneSec;//, totalBytesTenSec;
- float totalBytes, actualBitrate;
- VCMFrameCount frameCount; // testing frame type counters
- // start test
- NumberOfCodecs = _vcm->NumberOfCodecs();
- // going over all available codecs
- _encodeCompleteCallback->SetFrameDimensions(_width, _height);
- SendStatsTest sendStats;
- for (int k = 0; k < NumberOfCodecs; k++)
- //for (int k = NumberOfCodecs - 1; k >=0; k--)
- {// static list starts from 0
- //just checking
- _vcm->InitializeSender();
- _sendCodec.maxBitrate = 8000;
- TEST(_vcm->Codec(k, &_sendCodec)== VCM_OK);
- _vcm->RegisterSendCodec(&_sendCodec, 1, 1440);
- _vcm->RegisterTransportCallback(_encodeCompleteCallback);
- _encodeCompleteCallback->SetCodecType(ConvertCodecType(_sendCodec.plName));
- printf (" \n\n Codec type = %s \n\n",_sendCodec.plName);
- for (i = 0; i < nBitrates; i++)
- {
- _bitRate = static_cast<float>(bitRate[i]);
- // just testing
- _vcm->InitializeSender();
- _sendCodec.startBitrate = (int)_bitRate;
- _sendCodec.maxBitrate = 8000;
- _sendCodec.maxFramerate = _frameRate;
- _vcm->RegisterSendCodec(&_sendCodec, 1, 1440);
- _vcm->RegisterTransportCallback(_encodeCompleteCallback);
- // up to here
- _vcm->SetChannelParameters((WebRtc_UWord32)_bitRate, 0, 20);
- _frameCnt = 0;
- totalBytes = 0;
- _encodeCompleteCallback->Initialize();
- sendStats.SetTargetFrameRate(static_cast<WebRtc_UWord32>(_frameRate));
- _vcm->RegisterSendStatisticsCallback(&sendStats);
- while (fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) ==
- _lengthSourceFrame)
- {
- _frameCnt++;
- sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
- sourceFrame.SetHeight(_height);
- sourceFrame.SetWidth(_width);
- _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(_frameRate));
- sourceFrame.SetTimeStamp(_timeStamp);
-
- ret = _vcm->AddVideoFrame(sourceFrame);
- IncrementDebugClock(_frameRate);
- // The following should be uncommneted for timing tests. Release tests only include
- // compliance with full sequence bit rate.
-
-
- //totalBytes = WaitForEncodedFrame();
- //currentTime = VCMTickTime::MillisecondTimestamp();//clock()/(double)CLOCKS_PER_SEC;
- if (_frameCnt == _frameRate)// @ 1sec
- {
- totalBytesOneSec = _encodeCompleteCallback->EncodedBytes();//totalBytes;
- }
- TEST(_vcm->TimeUntilNextProcess() >= 0);
- } // video seq. encode done
- TEST(_vcm->TimeUntilNextProcess() == 0);
- _vcm->Process(); // Let the module calculate its send bit rate estimate
- // estimating rates
- // complete sequence
- // bit rate assumes input frame rate is as specified
- totalBytes = _encodeCompleteCallback->EncodedBytes();
- actualBitrate = (float)(8.0/1000)*(totalBytes / (_frameCnt / _frameRate));
-
- printf("Complete Seq.: target bitrate: %.0f kbps, actual bitrate: %.1f kbps\n", _bitRate, actualBitrate);
- TEST((fabs(actualBitrate - _bitRate) < FullReq * _bitRate) ||
- (strncmp(_sendCodec.plName, "I420", 4) == 0));
-
- // 1 Sec.
- actualBitrate = (float)(8.0/1000)*(totalBytesOneSec);
- //actualBitrate = (float)(8.0*totalBytesOneSec)/(oneSecTime - startTime);
- //printf("First 1Sec: target bitrate: %.0f kbps, actual bitrate: %.1f kbps\n", _bitRate, actualBitrate);
- //TEST(fabs(actualBitrate - _bitRate) < OneSecReq * _bitRate);
- rewind(_sourceFile);
-
- //checking key/delta frame count
- _vcm->SentFrameCount(frameCount);
- printf("frame count: %d delta, %d key\n", frameCount.numDeltaFrames, frameCount.numKeyFrames);
- }// end per codec
-
- } // end rate control test
- /********************************/
- /* Encoder Pipeline Delay Test */
- /******************************/
- _vcm->InitializeSender();
- sourceFrame.Free();
- sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
- NumberOfCodecs = _vcm->NumberOfCodecs();
- bool encodeComplete = false;
- // going over all available codecs
- for (int k = 0; k < NumberOfCodecs; k++)
- {
- _vcm->Codec(k, &_sendCodec);
- _vcm->InitializeSender();
- _sendCodec.maxBitrate = 8000;
- _vcm->RegisterSendCodec(&_sendCodec, 4, 1440);
- _vcm->RegisterTransportCallback(_encodeCompleteCallback);
-
- _frameCnt = 0;
- encodeComplete = false;
- while (encodeComplete == false)
- {
- TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
- _frameCnt++;
- sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
- sourceFrame.SetHeight(_height);
- sourceFrame.SetWidth(_width);
- _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(_frameRate));
- sourceFrame.SetTimeStamp(_timeStamp);
- _vcm->AddVideoFrame(sourceFrame);
- encodeComplete = _encodeCompleteCallback->EncodeComplete();
- } // first frame encoded
- printf ("\n Codec type = %s \n", _sendCodec.plName);
- printf(" Encoder pipeline delay = %d frames\n", _frameCnt - 1);
- } // end for all codecs
-
- /********************************/
- /* Encoder Packet Size Test */
- /********************************/
- RtpRtcp& rtpModule = *RtpRtcp::CreateRtpRtcp(1, false);
- TEST(rtpModule.InitSender() == 0);
- RTPSendCallback_SizeTest sendCallback;
- rtpModule.RegisterSendTransport(&sendCallback);
-
- VCMRTPEncodeCompleteCallback encCompleteCallback(&rtpModule);
- _vcm->InitializeSender();
-
- // TEST DISABLED FOR NOW SINCE VP8 DOESN'T HAVE THIS FEATURE
-
-// sourceFrame.Free();
-// sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
-// NumberOfCodecs = _vcm->NumberOfCodecs();
-// WebRtc_UWord32 targetPayloadSize = 500;
-// rtpModule.SetMaxTransferUnit(targetPayloadSize);
-// // going over all available codecs
-// for (int k = 0; k < NumberOfCodecs; k++)
-// {
-// _vcm->Codec(k, &_sendCodec);
-// if (strncmp(_sendCodec.plName, "VP8", 3) == 0)
-// {
-// // Only test with VP8
-// continue;
-// }
-// rtpModule.RegisterSendPayload(_sendCodec.plName, _sendCodec.plType);
-// // Make sure we only get one NAL unit per packet
-// _vcm->InitializeSender();
-// _vcm->RegisterSendCodec(&_sendCodec, 4, targetPayloadSize);
-// sendCallback.SetMaxPayloadSize(targetPayloadSize);
-// _vcm->RegisterTransportCallback(&encCompleteCallback);
-// sendCallback.Reset();
-// _frameCnt = 0;
-// rewind(_sourceFile);
-// while (!feof(_sourceFile))
-// {
-// fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile);
-// _frameCnt++;
-// sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
-// sourceFrame.SetHeight(_height);
-// sourceFrame.SetWidth(_width);
-// _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(_frameRate));
-// sourceFrame.SetTimeStamp(_timeStamp);
-// ret = _vcm->AddVideoFrame(sourceFrame);
-// } // first frame encoded
-// printf ("\n Codec type = %s \n",_sendCodec.plName);
-// printf(" Average payload size = %f bytes, target = %u bytes\n", sendCallback.AveragePayloadSize(), targetPayloadSize);
-// } // end for all codecs
-
-
- // Test temporal decimation settings
- for (int k = 0; k < NumberOfCodecs; k++)
- {
- _vcm->Codec(k, &_sendCodec);
- if (strncmp(_sendCodec.plName, "I420", 4) == 0)
- {
- // Only test with I420
- break;
- }
- }
- TEST(strncmp(_sendCodec.plName, "I420", 4) == 0);
- _vcm->InitializeSender();
- _sendCodec.maxFramerate = static_cast<WebRtc_UWord8>(_frameRate / 2.0 + 0.5f);
- _vcm->RegisterSendCodec(&_sendCodec, 4, 1440);
- _vcm->SetChannelParameters(2000, 0, 0);
- _vcm->RegisterTransportCallback(_encodeCompleteCallback);
- // up to here
- _vcm->SetChannelParameters((WebRtc_UWord32)_bitRate, 0, 20);
- _encodeCompleteCallback->Initialize();
- sendStats.SetTargetFrameRate(static_cast<WebRtc_UWord32>(_frameRate));
- _vcm->RegisterSendStatisticsCallback(&sendStats);
- rewind(_sourceFile);
- while (fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) ==
- _lengthSourceFrame)
- {
- sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
- sourceFrame.SetHeight(_height);
- sourceFrame.SetWidth(_width);
- _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(_frameRate));
- sourceFrame.SetTimeStamp(_timeStamp);
- ret = _vcm->AddVideoFrame(sourceFrame);
- if (_vcm->TimeUntilNextProcess() <= 0)
- {
- _vcm->Process();
- }
- IncrementDebugClock(_frameRate);
- } // first frame encoded
-
- RtpRtcp::DestroyRtpRtcp(&rtpModule);
- Print();
- delete tmpBuffer;
- delete _decodeCallback;
- delete _encodeCompleteCallback;
- return 0;
-}
-
-
-void
-GenericCodecTest::Print()
-{
- printf(" \n\n VCM Generic Encoder Test: \n\n%i tests completed\n", vcmMacrosTests);
- if (vcmMacrosErrors > 0)
- {
- printf("%i FAILED\n\n", vcmMacrosErrors);
- }
- else
- {
- printf("ALL PASSED\n\n");
- }
-}
-
-float
-GenericCodecTest::WaitForEncodedFrame() const
-{
- WebRtc_Word64 startTime = _clock->MillisecondTimestamp();
- while (_clock->MillisecondTimestamp() - startTime < kMaxWaitEncTimeMs*10)
- {
- if (_encodeCompleteCallback->EncodeComplete())
- {
- return _encodeCompleteCallback->EncodedBytes();
- }
- }
- return 0;
-}
-
-void
-GenericCodecTest::IncrementDebugClock(float frameRate)
-{
- _clock->IncrementDebugClock(1000/frameRate);
-}
-
-int
-RTPSendCallback_SizeTest::SendPacket(int channel, const void *data, int len)
-{
- _nPackets++;
- _payloadSizeSum += len;
- // Make sure no payloads (len - header size) are larger than maxPayloadSize
- TEST(len > 0 && static_cast<WebRtc_UWord32>(len - 12) <= _maxPayloadSize);
- return 0;
-}
-
-void
-RTPSendCallback_SizeTest::SetMaxPayloadSize(WebRtc_UWord32 maxPayloadSize)
-{
- _maxPayloadSize = maxPayloadSize;
-}
-
-void
-RTPSendCallback_SizeTest::Reset()
-{
- _nPackets = 0;
- _payloadSizeSum = 0;
-}
-
-float
-RTPSendCallback_SizeTest::AveragePayloadSize() const
-{
- if (_nPackets > 0)
- {
- return _payloadSizeSum / static_cast<float>(_nPackets);
- }
- return 0;
-}
-
-WebRtc_Word32
-VCMEncComplete_KeyReqTest::SendData(
- const FrameType frameType,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader& /*fragmentationHeader*/,
- const webrtc::RTPVideoHeader* /*videoHdr*/)
-{
- WebRtcRTPHeader rtpInfo;
- rtpInfo.header.markerBit = true; // end of frame
- rtpInfo.type.Video.codecHeader.VP8.InitRTPVideoHeaderVP8();
- rtpInfo.type.Video.codec = kRTPVideoVP8;
- rtpInfo.header.payloadType = payloadType;
- rtpInfo.header.sequenceNumber = _seqNo;
- _seqNo += 2;
- rtpInfo.header.ssrc = 0;
- rtpInfo.header.timestamp = _timeStamp;
- _timeStamp += 3000;
- rtpInfo.type.Video.isFirstPacket = false;
- rtpInfo.frameType = kVideoFrameKey;
- return _vcm.IncomingPacket(payloadData, payloadSize, rtpInfo);
-}
diff --git a/trunk/src/modules/video_coding/main/test/generic_codec_test.h b/trunk/src/modules/video_coding/main/test/generic_codec_test.h
deleted file mode 100644
index c88280f..0000000
--- a/trunk/src/modules/video_coding/main/test/generic_codec_test.h
+++ /dev/null
@@ -1,109 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_GENERIC_CODEC_TEST_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TEST_GENERIC_CODEC_TEST_H_
-
-#include "video_coding.h"
-
-#include <string.h>
-#include <fstream>
-
-#include "test_callbacks.h"
-#include "test_util.h"
-/*
-Test consists of:
-1. Sanity checks
-2. Bit rate validation
-3. Encoder control test / General API functionality
-4. Decoder control test / General API functionality
-
-*/
-
-namespace webrtc {
-
-int VCMGenericCodecTest(CmdArgs& args);
-
-class FakeTickTime;
-
-class GenericCodecTest
-{
-public:
- GenericCodecTest(webrtc::VideoCodingModule* vcm,
- webrtc::FakeTickTime* clock);
- ~GenericCodecTest();
- static int RunTest(CmdArgs& args);
- WebRtc_Word32 Perform(CmdArgs& args);
- float WaitForEncodedFrame() const;
-
-private:
- void Setup(CmdArgs& args);
- void Print();
- WebRtc_Word32 TearDown();
- void IncrementDebugClock(float frameRate);
-
- webrtc::FakeTickTime* _clock;
- webrtc::VideoCodingModule* _vcm;
- webrtc::VideoCodec _sendCodec;
- webrtc::VideoCodec _receiveCodec;
- std::string _inname;
- std::string _outname;
- std::string _encodedName;
- WebRtc_Word32 _sumEncBytes;
- FILE* _sourceFile;
- FILE* _decodedFile;
- FILE* _encodedFile;
- WebRtc_UWord16 _width;
- WebRtc_UWord16 _height;
- float _frameRate;
- WebRtc_UWord32 _lengthSourceFrame;
- WebRtc_UWord32 _timeStamp;
- VCMDecodeCompleteCallback* _decodeCallback;
- VCMEncodeCompleteCallback* _encodeCompleteCallback;
-
-}; // end of GenericCodecTest class definition
-
-class RTPSendCallback_SizeTest : public webrtc::Transport
-{
-public:
- // constructor input: (receive side) rtp module to send encoded data to
- RTPSendCallback_SizeTest() : _maxPayloadSize(0), _payloadSizeSum(0), _nPackets(0) {}
- virtual int SendPacket(int channel, const void *data, int len);
- virtual int SendRTCPPacket(int channel, const void *data, int len) {return 0;}
- void SetMaxPayloadSize(WebRtc_UWord32 maxPayloadSize);
- void Reset();
- float AveragePayloadSize() const;
-private:
- WebRtc_UWord32 _maxPayloadSize;
- WebRtc_UWord32 _payloadSizeSum;
- WebRtc_UWord32 _nPackets;
-};
-
-class VCMEncComplete_KeyReqTest : public webrtc::VCMPacketizationCallback
-{
-public:
- VCMEncComplete_KeyReqTest(webrtc::VideoCodingModule &vcm) : _vcm(vcm), _seqNo(0), _timeStamp(0) {}
- WebRtc_Word32 SendData(
- const webrtc::FrameType frameType,
- const WebRtc_UWord8 payloadType,
- WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const webrtc::RTPFragmentationHeader& fragmentationHeader,
- const webrtc::RTPVideoHeader* videoHdr);
-private:
- webrtc::VideoCodingModule& _vcm;
- WebRtc_UWord16 _seqNo;
- WebRtc_UWord32 _timeStamp;
-}; // end of VCMEncodeCompleteCallback
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_GENERIC_CODEC_TEST_H_
diff --git a/trunk/src/modules/video_coding/main/test/jitter_buffer_test.cc b/trunk/src/modules/video_coding/main/test/jitter_buffer_test.cc
deleted file mode 100644
index 2066983..0000000
--- a/trunk/src/modules/video_coding/main/test/jitter_buffer_test.cc
+++ /dev/null
@@ -1,1936 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <math.h>
-#include <stdio.h>
-
-#include "common_types.h"
-#include "../source/event.h"
-#include "frame_buffer.h"
-#include "inter_frame_delay.h"
-#include "jitter_buffer.h"
-#include "jitter_estimate_test.h"
-#include "jitter_estimator.h"
-#include "media_opt_util.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
-#include "packet.h"
-#include "test_util.h"
-#include "test_macros.h"
-
-// TODO(holmer): Get rid of this to conform with style guide.
-using namespace webrtc;
-
-// TODO (Mikhal/Stefan): Update as gtest and separate to specific tests.
-
-int CheckOutFrame(VCMEncodedFrame* frameOut, unsigned int size, bool startCode)
-{
- if (frameOut == 0)
- {
- return -1;
- }
-
- const WebRtc_UWord8* outData = frameOut->Buffer();
-
- unsigned int i = 0;
-
- if(startCode)
- {
- if (outData[0] != 0 || outData[1] != 0 || outData[2] != 0 ||
- outData[3] != 1)
- {
- return -2;
- }
- i+= 4;
- }
-
- // check the frame data
- int count = 3;
-
- // check the frame length
- if (frameOut->Length() != size)
- {
- return -3;
- }
-
- for(; i < size; i++)
- {
- if (outData[i] == 0 && outData[i + 1] == 0 && outData[i + 2] == 0x80)
- {
- i += 2;
- }
- else if(startCode && outData[i] == 0 && outData[i + 1] == 0)
- {
- if (outData[i] != 0 || outData[i + 1] != 0 ||
- outData[i + 2] != 0 || outData[i + 3] != 1)
- {
- return -3;
- }
- i += 3;
- }
- else
- {
- if (outData[i] != count)
- {
- return -4;
- }
- count++;
- if(count == 10)
- {
- count = 3;
- }
- }
- }
- return 0;
-}
-
-
-int JitterBufferTest(CmdArgs& args)
-{
- // Don't run these tests with debug event.
-#if defined(EVENT_DEBUG)
- return -1;
-#endif
- TickTimeBase clock;
-
- // Start test
- WebRtc_UWord16 seqNum = 1234;
- WebRtc_UWord32 timeStamp = 0;
- int size = 1400;
- WebRtc_UWord8 data[1500];
- VCMPacket packet(data, size, seqNum, timeStamp, true);
-
- VCMJitterBuffer jb(&clock);
-
- seqNum = 1234;
- timeStamp = 123*90;
- FrameType incomingFrameType(kVideoFrameKey);
- VCMEncodedFrame* frameOut=NULL;
- WebRtc_Word64 renderTimeMs = 0;
- packet.timestamp = timeStamp;
- packet.seqNum = seqNum;
-
- // build a data vector with 0, 0, 0x80, 3, 4, 5, 6, 7, 8, 9, 0, 0, 0x80, 3....
- data[0] = 0;
- data[1] = 0;
- data[2] = 0x80;
- int count = 3;
- for (unsigned int i = 3; i < sizeof(data) - 3; ++i)
- {
- data[i] = count;
- count++;
- if(count == 10)
- {
- data[i+1] = 0;
- data[i+2] = 0;
- data[i+3] = 0x80;
- count = 3;
- i += 3;
- }
- }
-
- // Test out of range inputs
- TEST(kSizeError == jb.InsertPacket(0, packet));
- jb.ReleaseFrame(0);
-
- // Not started
- TEST(0 == jb.GetFrame(packet));
- TEST(-1 == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
- TEST(0 == jb.GetCompleteFrameForDecoding(10));
- TEST(0 == jb.GetFrameForDecoding());
-
- // Start
- jb.Start();
-
- // Get frame to use for this timestamp
- VCMEncodedFrame* frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // No packets inserted
- TEST(0 == jb.GetCompleteFrameForDecoding(10));
-
-
- //
- // TEST single packet frame
- //
- // --------
- // | 1234 |
- // --------
-
- // packet.frameType;
- // packet.dataPtr;
- // packet.sizeBytes;
- // packet.timestamp;
- // packet.seqNum;
- // packet.isFirstPacket;
- // packet.markerBit;
- //
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = true;
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- TEST(CheckOutFrame(frameOut, size, false) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameDelta);
-
- // Release frame (when done with decoding)
- jb.ReleaseFrame(frameOut);
-
- //printf("DONE delta frame 1 packet\n");
-
- //
- // TEST dual packet frame
- //
- // -----------------
- // | 1235 | 1236 |
- // -----------------
- //
-
- seqNum++;
- timeStamp += 33*90;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = false;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- // it should not be complete
- TEST(frameOut == 0);
-
- seqNum++;
- packet.isFirstPacket = false;
- packet.markerBit = true;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- TEST(CheckOutFrame(frameOut, size*2, false) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameDelta);
-
- // Release frame (when done with decoding)
- jb.ReleaseFrame(frameOut);
-
- //printf("DONE delta frame 2 packets\n");
-
-
- //
- // TEST 100 packets frame Key frame
- //
- // ----------------------------------
- // | 1237 | 1238 | .... | 1336 |
- // ----------------------------------
-
- // insert first packet
- timeStamp += 33*90;
- seqNum++;
- packet.frameType = kVideoFrameKey;
- packet.isFirstPacket = true;
- packet.markerBit = false;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameKey);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- // it should not be complete
- TEST(frameOut == 0);
-
- // insert 98 frames
- int loop = 0;
- do
- {
- seqNum++;
- packet.isFirstPacket = false;
- packet.markerBit = false;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
- loop++;
- } while (loop < 98);
-
- // insert last packet
- seqNum++;
- packet.isFirstPacket = false;
- packet.markerBit = true;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- TEST(CheckOutFrame(frameOut, size*100, false) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameKey);
-
- // Release frame (when done with decoding)
- jb.ReleaseFrame(frameOut);
-
- //printf("DONE key frame 100 packets\n");
-
- //
- // TEST 100 packets frame Delta frame
- //
- // ----------------------------------
- // | 1337 | 1238 | .... | 1436 |
- // ----------------------------------
-
- // insert first packet
- timeStamp += 33*90;
- seqNum++;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = false;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- // it should not be complete
- TEST(frameOut == 0);
-
- // insert 98 frames
- loop = 0;
- do
- {
- seqNum++;
- packet.isFirstPacket = false;
- packet.markerBit = false;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
- loop++;
- } while (loop < 98);
-
- // insert last packet
- seqNum++;
- packet.isFirstPacket = false;
- packet.markerBit = true;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- TEST(CheckOutFrame(frameOut, size*100, false) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameDelta);
-
- // Release frame (when done with decoding)
- jb.ReleaseFrame(frameOut);
-
- //printf("DONE delta frame 100 packets\n");
-
- //
- // TEST packet re-ordering reverse order
- //
- // ----------------------------------
- // | 1437 | 1238 | .... | 1536 |
- // ----------------------------------
- // <----------
-
- // insert "first" packet last seqnum
- timeStamp += 33*90;
- seqNum += 100;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = false;
- packet.markerBit = true;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- // it should not be complete
- TEST(frameOut == 0);
-
- // insert 98 packets
- loop = 0;
- do
- {
- seqNum--;
- packet.isFirstPacket = false;
- packet.markerBit = false;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
- loop++;
- } while (loop < 98);
-
- // insert last packet
- seqNum--;
- packet.isFirstPacket = true;
- packet.markerBit = false;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- TEST(CheckOutFrame(frameOut, size*100, false) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameDelta);
-
- // Release frame (when done with decoding)
- jb.ReleaseFrame(frameOut);
-
- //printf("DONE delta frame 100 packets reverse order\n");
-
- seqNum+= 100;
-
- //
- // TEST frame re-ordering 2 frames 2 packets each
- //
- // ----------------- -----------------
- // | 1539 | 1540 | | 1537 | 1538 |
- // ----------------- -----------------
-
- seqNum += 2;
- timeStamp += 2* 33 * 90;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = false;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- // it should not be complete
- TEST(frameOut == 0);
-
- seqNum++;
- packet.isFirstPacket = false;
- packet.markerBit = true;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
-
- // check that we fail to get frame since seqnum is not continuous
- frameOut = jb.GetCompleteFrameForDecoding(10);
- TEST(frameOut == 0);
-
- seqNum -= 3;
- timeStamp -= 33*90;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = false;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- // it should not be complete
- TEST(frameOut == 0);
-
- seqNum++;
- packet.isFirstPacket = false;
- packet.markerBit = true;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- TEST(CheckOutFrame(frameOut, size*2, false) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameDelta);
-
- // Release frame (when done with decoding)
- jb.ReleaseFrame(frameOut);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- TEST(CheckOutFrame(frameOut, size*2, false) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameDelta);
-
- // Release frame (when done with decoding)
- jb.ReleaseFrame(frameOut);
-
- seqNum += 2;
- //printf("DONE frame re-ordering 2 frames 2 packets\n");
-
- // restore
- packet.dataPtr = data;
- packet.codec = kVideoCodecUnknown;
-
- //
- // TEST duplicate packets
- //
- // -----------------
- // | 1543 | 1543 |
- // -----------------
- //
-
- seqNum++;
- timeStamp += 2*33*90;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = false;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- // it should not be complete
- TEST(frameOut == 0);
-
- packet.isFirstPacket = false;
- packet.markerBit = true;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kDuplicatePacket == jb.InsertPacket(frameIn, packet));
-
- seqNum++;
- packet.seqNum = seqNum;
-
- TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- TEST(CheckOutFrame(frameOut, size*2, false) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameDelta);
-
- // Release frame (when done with decoding)
- jb.ReleaseFrame(frameOut);
-
- //printf("DONE test duplicate packets\n");
-
- //
- // TEST H.264 insert start code
- //
- // -----------------
- // | 1544 | 1545 |
- // -----------------
- // insert start code, both packets
-
- seqNum++;
- timeStamp += 33 * 90;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = false;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
- packet.insertStartCode = true;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- // it should not be complete
- TEST(frameOut == 0);
-
- seqNum++;
- packet.isFirstPacket = false;
- packet.markerBit = true;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- TEST(CheckOutFrame(frameOut, size * 2 + 4 * 2, true) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameDelta);
-
- // Release frame (when done with decoding)
- jb.ReleaseFrame(frameOut);
-
- // reset
- packet.insertStartCode = false;
- //printf("DONE H.264 insert start code test 2 packets\n");
-
- //
- // TEST statistics
- //
- WebRtc_UWord32 numDeltaFrames = 0;
- WebRtc_UWord32 numKeyFrames = 0;
- TEST(jb.GetFrameStatistics(numDeltaFrames, numKeyFrames) == 0);
-
- TEST(numDeltaFrames == 8);
- TEST(numKeyFrames == 1);
-
- WebRtc_UWord32 frameRate;
- WebRtc_UWord32 bitRate;
- TEST(jb.GetUpdate(frameRate, bitRate) == 0);
-
- // these depend on CPU speed works on a T61
- TEST(frameRate > 30);
- TEST(bitRate > 10000000);
-
-
- jb.Flush();
-
- //
- // TEST packet loss. Verify missing packets statistics and not decodable
- // packets statistics.
- // Insert 10 frames consisting of 4 packets and remove one from all of them.
- // The last packet is an empty (non-media) packet
- //
-
- // Select a start seqNum which triggers a difficult wrap situation
- // The JB will only output (incomplete)frames if the next one has started
- // to arrive. Start by inserting one frame (key).
- seqNum = 0xffff - 4;
- seqNum++;
- timeStamp += 33*90;
- packet.frameType = kVideoFrameKey;
- packet.isFirstPacket = true;
- packet.markerBit = false;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
- packet.completeNALU = kNaluStart;
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- for (int i = 0; i < 11; ++i) {
- webrtc::FrameType frametype = kVideoFrameDelta;
- seqNum++;
- timeStamp += 33*90;
- packet.frameType = frametype;
- packet.isFirstPacket = true;
- packet.markerBit = false;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
- packet.completeNALU = kNaluStart;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // Get packet notification
- TEST(timeStamp - 33 * 90 == jb.GetNextTimeStamp(10, incomingFrameType,
- renderTimeMs));
-
- // Check incoming frame type
- if (i == 0)
- {
- TEST(incomingFrameType == kVideoFrameKey);
- }
- else
- {
- TEST(incomingFrameType == frametype);
- }
-
- // Get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- // Should not be complete
- TEST(frameOut == 0);
-
- seqNum += 2;
- packet.isFirstPacket = false;
- packet.markerBit = true;
- packet.seqNum = seqNum;
- packet.completeNALU = kNaluEnd;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
-
-
- // Insert an empty (non-media) packet
- seqNum++;
- packet.isFirstPacket = false;
- packet.markerBit = false;
- packet.seqNum = seqNum;
- packet.completeNALU = kNaluEnd;
- packet.frameType = kFrameEmpty;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
-
- // Get the frame
- frameOut = jb.GetFrameForDecoding();
-
- // One of the packets has been discarded by the jitter buffer.
- // Last frame can't be extracted yet.
- if (i < 10)
- {
- TEST(CheckOutFrame(frameOut, size, false) == 0);
-
- // check the frame type
- if (i == 0)
- {
- TEST(frameOut->FrameType() == kVideoFrameKey);
- }
- else
- {
- TEST(frameOut->FrameType() == frametype);
- }
- TEST(frameOut->Complete() == false);
- TEST(frameOut->MissingFrame() == false);
- }
-
- // Release frame (when done with decoding)
- jb.ReleaseFrame(frameOut);
- }
-
- TEST(jb.NumNotDecodablePackets() == 10);
-
- // Insert 3 old packets and verify that we have 3 discarded packets
- // Match value to actual latest timestamp decoded
- timeStamp -= 33 * 90;
- packet.timestamp = timeStamp - 1000;
- frameIn = jb.GetFrame(packet);
- TEST(frameIn == NULL);
-
- packet.timestamp = timeStamp - 500;
- frameIn = jb.GetFrame(packet);
- TEST(frameIn == NULL);
-
- packet.timestamp = timeStamp - 100;
- frameIn = jb.GetFrame(packet);
- TEST(frameIn == NULL);
-
- TEST(jb.DiscardedPackets() == 3);
-
- jb.Flush();
-
- // This statistic shouldn't be reset by a flush.
- TEST(jb.DiscardedPackets() == 3);
-
- //printf("DONE Statistics\n");
-
-
- // Temporarily do this to make the rest of the test work:
- timeStamp += 33*90;
- seqNum += 4;
-
-
- //
- // TEST delta frame 100 packets with seqNum wrap
- //
- // ---------------------------------------
- // | 65520 | 65521 | ... | 82 | 83 |
- // ---------------------------------------
- //
-
- jb.Flush();
-
- // insert first packet
- timeStamp += 33*90;
- seqNum = 0xfff0;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = false;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- // it should not be complete
- TEST(frameOut == 0);
-
- // insert 98 packets
- loop = 0;
- do
- {
- seqNum++;
- packet.isFirstPacket = false;
- packet.markerBit = false;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(2, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(2);
-
- // it should not be complete
- TEST(frameOut == 0);
-
- loop++;
- } while (loop < 98);
-
- // insert last packet
- seqNum++;
- packet.isFirstPacket = false;
- packet.markerBit = true;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- TEST(CheckOutFrame(frameOut, size*100, false) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameDelta);
-
- // Release frame (when done with decoding)
- jb.ReleaseFrame(frameOut);
-
- //printf("DONE delta frame 100 packets with wrap in seqNum\n");
-
- //
- // TEST packet re-ordering reverse order with neg seqNum warp
- //
- // ----------------------------------------
- // | 65447 | 65448 | ... | 9 | 10 |
- // ----------------------------------------
- // <-------------
-
- // test flush
- jb.Flush();
-
- // insert "first" packet last seqnum
- timeStamp += 33*90;
- seqNum = 10;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = false;
- packet.markerBit = true;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- // it should not be complete
- TEST(frameOut == 0);
-
- // insert 98 frames
- loop = 0;
- do
- {
- seqNum--;
- packet.isFirstPacket = false;
- packet.markerBit = false;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(2, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(2);
-
- // it should not be complete
- TEST(frameOut == 0);
-
- loop++;
- } while (loop < 98);
-
- // insert last packet
- seqNum--;
- packet.isFirstPacket = true;
- packet.markerBit = false;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- TEST(CheckOutFrame(frameOut, size*100, false) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameDelta);
-
- // Release frame (when done with decoding)
- jb.ReleaseFrame(frameOut);
-
- //printf("DONE delta frame 100 packets reverse order with wrap in seqNum \n");
-
- // test flush
- jb.Flush();
-
- //
- // TEST packet re-ordering with seqNum wrap
- //
- // -----------------------
- // | 1 | 65535 | 0 |
- // -----------------------
-
- // insert "first" packet last seqnum
- timeStamp += 33*90;
- seqNum = 1;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = false;
- packet.markerBit = true;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- // it should not be complete
- TEST(frameOut == 0);
-
- // insert last packet
- seqNum -= 2;
- packet.isFirstPacket = true;
- packet.markerBit = false;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- // it should not be complete
- TEST(frameOut == 0);
-
- seqNum++;
- packet.isFirstPacket = false;
- packet.markerBit = false;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- TEST(CheckOutFrame(frameOut, size*3, false) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameDelta);
-
- // Release frame (when done with decoding)
- jb.ReleaseFrame(frameOut);
-
- //printf("DONE delta frame 3 packets re-ordering with wrap in seqNum \n");
-
- // test flush
- jb.Flush();
-
- //
- // TEST insert old frame
- //
- // ------- -------
- // | 2 | | 1 |
- // ------- -------
- // t = 3000 t = 2000
-
- seqNum = 2;
- timeStamp = 3000;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = true;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(3000 == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
- TEST(kVideoFrameDelta == incomingFrameType);
-
- // Get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
- TEST(3000 == frameOut->TimeStamp());
-
- TEST(CheckOutFrame(frameOut, size, false) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameDelta);
-
- jb.ReleaseFrame(frameOut);
-
- seqNum--;
- timeStamp = 2000;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = true;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- // Changed behavior, never insert packets into frames older than the
- // last decoded frame.
- TEST(frameIn == 0);
-
- //printf("DONE insert old frame\n");
-
- jb.Flush();
-
- //
- // TEST insert old frame with wrap in timestamp
- //
- // ------- -------
- // | 2 | | 1 |
- // ------- -------
- // t = 3000 t = 0xffffff00
-
- seqNum = 2;
- timeStamp = 3000;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = true;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
- TEST(kVideoFrameDelta == incomingFrameType);
-
- // Get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
- TEST(timeStamp == frameOut->TimeStamp());
-
- TEST(CheckOutFrame(frameOut, size, false) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameDelta);
-
- jb.ReleaseFrame(frameOut);
-
- seqNum--;
- timeStamp = 0xffffff00;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = true;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- // This timestamp is old
- TEST(frameIn == 0);
-
- jb.Flush();
-
- //
- // TEST wrap in timeStamp
- //
- // --------------- ---------------
- // | 1 | 2 | | 3 | 4 |
- // --------------- ---------------
- // t = 0xffffff00 t = 33*90
-
- seqNum = 1;
- timeStamp = 0xffffff00;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = false;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- // it should not be complete
- TEST(frameOut == 0);
-
- seqNum++;
- packet.isFirstPacket = false;
- packet.markerBit = true;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
-
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- TEST(CheckOutFrame(frameOut, size*2, false) == 0);
-
- jb.ReleaseFrame(frameOut);
-
- seqNum++;
- timeStamp += 33*90;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = false;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- // it should not be complete
- TEST(frameOut == 0);
-
- seqNum++;
- packet.isFirstPacket = false;
- packet.markerBit = true;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kCompleteSession == jb.InsertPacket(frameIn, packet));
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
-
- TEST(CheckOutFrame(frameOut, size*2, false) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameDelta);
-
- // Release frame (when done with decoding)
- jb.ReleaseFrame(frameOut);
-
- //printf("DONE time stamp wrap 2 frames 2 packets\n");
-
- jb.Flush();
-
- //
- // TEST insert 2 frames with wrap in timeStamp
- //
- // ------- -------
- // | 1 | | 2 |
- // ------- -------
- // t = 0xffffff00 t = 2700
-
- seqNum = 1;
- timeStamp = 0xffffff00;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = true;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert first frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // Get packet notification
- TEST(0xffffff00 == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
- TEST(kVideoFrameDelta == incomingFrameType);
-
- // Insert next frame
- seqNum++;
- timeStamp = 2700;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = true;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // Get packet notification
- TEST(0xffffff00 == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
- TEST(kVideoFrameDelta == incomingFrameType);
-
- // Get frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
- TEST(0xffffff00 == frameOut->TimeStamp());
-
- TEST(CheckOutFrame(frameOut, size, false) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameDelta);
-
- // Get packet notification
- TEST(2700 == jb.GetNextTimeStamp(0, incomingFrameType, renderTimeMs));
- TEST(kVideoFrameDelta == incomingFrameType);
-
- // Get frame
- VCMEncodedFrame* frameOut2 = jb.GetCompleteFrameForDecoding(10);
- TEST(2700 == frameOut2->TimeStamp());
-
- TEST(CheckOutFrame(frameOut2, size, false) == 0);
-
- // check the frame type
- TEST(frameOut2->FrameType() == kVideoFrameDelta);
-
- // Release frame (when done with decoding)
- jb.ReleaseFrame(frameOut);
- jb.ReleaseFrame(frameOut2);
-
- //printf("DONE insert 2 frames (1 packet) with wrap in timestamp\n");
-
- jb.Flush();
-
- //
- // TEST insert 2 frames re-ordered with wrap in timeStamp
- //
- // ------- -------
- // | 2 | | 1 |
- // ------- -------
- // t = 2700 t = 0xffffff00
-
- seqNum = 2;
- timeStamp = 2700;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = true;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert first frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // Get packet notification
- TEST(2700 == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
- TEST(kVideoFrameDelta == incomingFrameType);
-
- // Insert second frame
- seqNum--;
- timeStamp = 0xffffff00;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = true;
- packet.markerBit = true;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // Get packet notification
- TEST(0xffffff00 == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
- TEST(kVideoFrameDelta == incomingFrameType);
-
- // Get frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
- TEST(0xffffff00 == frameOut->TimeStamp());
-
- TEST(CheckOutFrame(frameOut, size, false) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameDelta);
-
- // get packet notification
- TEST(2700 == jb.GetNextTimeStamp(0, incomingFrameType, renderTimeMs));
- TEST(kVideoFrameDelta == incomingFrameType);
-
- // Get frame
- frameOut2 = jb.GetCompleteFrameForDecoding(10);
- TEST(2700 == frameOut2->TimeStamp());
-
- TEST(CheckOutFrame(frameOut2, size, false) == 0);
-
- // check the frame type
- TEST(frameOut2->FrameType() == kVideoFrameDelta);
-
- // Release frame (when done with decoding)
- jb.ReleaseFrame(frameOut);
- jb.ReleaseFrame(frameOut2);
-
- //printf("DONE insert 2 frames (1 packet) re-ordered with wrap in timestamp\n");
-
- //
- // TEST delta frame with more than max number of packets
- //
-
- jb.Start();
-
- loop = 0;
- packet.timestamp += 33*90;
- bool firstPacket = true;
- // insert kMaxPacketsInJitterBuffer into frame
- do
- {
- seqNum++;
- packet.isFirstPacket = false;
- packet.markerBit = false;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert frame
- if (firstPacket)
- {
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
- firstPacket = false;
- }
- else
- {
- TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
- }
-
- // get packet notification
- TEST(packet.timestamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- loop++;
- } while (loop < kMaxPacketsInSession);
-
- // Max number of packets inserted
-
- // Insert one more packet
- seqNum++;
- packet.isFirstPacket = false;
- packet.markerBit = true;
- packet.seqNum = seqNum;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- // Insert the packet -> frame recycled
- TEST(kSizeError == jb.InsertPacket(frameIn, packet));
-
- TEST(0 == jb.GetCompleteFrameForDecoding(10));
-
- //printf("DONE fill frame - packets > max number of packets\n");
-
- //
- // TEST fill JB with more than max number of frame (50 delta frames +
- // 51 key frames) with wrap in seqNum
- //
- // --------------------------------------------------------------
- // | 65485 | 65486 | 65487 | .... | 65535 | 0 | 1 | 2 | .....| 50 |
- // --------------------------------------------------------------
- // |<-----------delta frames------------->|<------key frames----->|
-
- jb.Flush();
-
- loop = 0;
- seqNum = 65485;
- WebRtc_UWord32 timeStampStart = timeStamp + 33*90;
- WebRtc_UWord32 timeStampFirstKey = 0;
- VCMEncodedFrame* ptrLastDeltaFrame = NULL;
- VCMEncodedFrame* ptrFirstKeyFrame = NULL;
- // insert MAX_NUMBER_OF_FRAMES frames
- do
- {
- timeStamp += 33*90;
- seqNum++;
- packet.isFirstPacket = true;
- packet.markerBit = true;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- frameIn = jb.GetFrame(packet);
- TEST(frameIn != 0);
-
- if (loop == 49) // last delta
- {
- ptrLastDeltaFrame = frameIn;
- }
- if (loop == 50) // first key
- {
- ptrFirstKeyFrame = frameIn;
- packet.frameType = kVideoFrameKey;
- timeStampFirstKey = packet.timestamp;
- }
-
- // Insert frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // Get packet notification, should be first inserted frame
- TEST(timeStampStart == jb.GetNextTimeStamp(10, incomingFrameType,
- renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameDelta);
-
- loop++;
- } while (loop < kMaxNumberOfFrames);
-
- // Max number of frames inserted
-
- // Insert one more frame
- timeStamp += 33*90;
- seqNum++;
- packet.isFirstPacket = true;
- packet.markerBit = true;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
-
- // Now, no free frame - frames will be recycled until first key frame
- frameIn = jb.GetFrame(packet);
- // ptr to last inserted delta frame should be returned
- TEST(frameIn != 0 && frameIn && ptrLastDeltaFrame);
-
- // Insert frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // First inserted key frame should be oldest in buffer
- TEST(timeStampFirstKey == jb.GetNextTimeStamp(10, incomingFrameType,
- renderTimeMs));
-
- // check incoming frame type
- TEST(incomingFrameType == kVideoFrameKey);
-
- // get the first key frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
- TEST(ptrFirstKeyFrame == frameOut);
-
- TEST(CheckOutFrame(frameOut, size, false) == 0);
-
- // check the frame type
- TEST(frameOut->FrameType() == kVideoFrameKey);
-
- // Release frame (when done with decoding)
- jb.ReleaseFrame(frameOut);
-
- jb.Flush();
-
- // printf("DONE fill JB - nr of delta + key frames (w/ wrap in seqNum) >
- // max nr of frames\n");
-
- // Testing that 1 empty packet inserted last will not be set for decoding
- seqNum = 3;
- // Insert one empty packet per frame, should never return the last timestamp
- // inserted. Only return empty frames in the presence of subsequent frames.
- int maxSize = 1000;
- for (int i = 0; i < maxSize + 10; i++)
- {
- timeStamp += 33 * 90;
- seqNum++;
- packet.isFirstPacket = false;
- packet.markerBit = false;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
- packet.frameType = kFrameEmpty;
- VCMEncodedFrame* testFrame = jb.GetFrameForDecoding();
- // timestamp should bever be the last TS inserted
- if (testFrame != NULL)
- {
- TEST(testFrame->TimeStamp() < timeStamp);
- printf("Not null TS = %d\n",testFrame->TimeStamp());
- }
- }
-
- jb.Flush();
-
-
- // printf(DONE testing inserting empty packets to the JB)
-
-
- // H.264 tests
- // Test incomplete NALU frames
-
- jb.Flush();
- jb.SetNackMode(kNoNack, -1, -1);
- seqNum ++;
- timeStamp += 33 * 90;
- int insertedLength = 0;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
- packet.frameType = kVideoFrameKey;
- packet.isFirstPacket = true;
- packet.completeNALU = kNaluStart;
- packet.markerBit = false;
-
- frameIn = jb.GetFrame(packet);
-
- // Insert a packet into a frame
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- seqNum += 2; // Skip one packet
- packet.seqNum = seqNum;
- packet.frameType = kVideoFrameKey;
- packet.isFirstPacket = false;
- packet.completeNALU = kNaluIncomplete;
- packet.markerBit = false;
-
- // Insert a packet into a frame
- TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
-
- seqNum++;
- packet.seqNum = seqNum;
- packet.frameType = kVideoFrameKey;
- packet.isFirstPacket = false;
- packet.completeNALU = kNaluEnd;
- packet.markerBit = false;
-
- // Insert a packet into a frame
- TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
-
- seqNum++;
- packet.seqNum = seqNum;
- packet.completeNALU = kNaluComplete;
- packet.markerBit = true; // Last packet
- TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
-
-
- // The JB will only output (incomplete) frames if a packet belonging to a
- // subsequent frame was already inserted. Insert one packet of a subsequent
- // frame. place high timestamp so the JB would always have a next frame
- // (otherwise, for every inserted frame we need to take care of the next
- // frame as well).
- packet.seqNum = 1;
- packet.timestamp = timeStamp + 33 * 90 * 10;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = false;
- packet.completeNALU = kNaluStart;
- packet.markerBit = false;
- frameIn = jb.GetFrame(packet);
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- // Get packet notification
- TEST(timeStamp == jb.GetNextTimeStamp(10, incomingFrameType, renderTimeMs));
- frameOut = jb.GetFrameForDecoding();
-
- // We can decode everything from a NALU until a packet has been lost.
- // Thus we can decode the first packet of the first NALU and the second NALU
- // which consists of one packet.
- TEST(CheckOutFrame(frameOut, packet.sizeBytes * 2, false) == 0);
- jb.ReleaseFrame(frameOut);
-
- // Test reordered start frame + 1 lost
- seqNum += 2; // Reoreder 1 frame
- timeStamp += 33*90;
- insertedLength = 0;
-
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
- packet.frameType = kVideoFrameKey;
- packet.isFirstPacket = false;
- packet.completeNALU = kNaluEnd;
- packet.markerBit = false;
-
- TEST(frameIn = jb.GetFrame(packet));
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
- insertedLength += packet.sizeBytes; // This packet should be decoded
-
- seqNum--;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
- packet.frameType = kVideoFrameKey;
- packet.isFirstPacket = true;
- packet.completeNALU = kNaluStart;
- packet.markerBit = false;
- TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
- insertedLength += packet.sizeBytes; // This packet should be decoded
-
- seqNum += 3; // One packet drop
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
- packet.frameType = kVideoFrameKey;
- packet.isFirstPacket = false;
- packet.completeNALU = kNaluComplete;
- packet.markerBit = false;
- TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
- insertedLength += packet.sizeBytes; // This packet should be decoded
-
- seqNum += 1;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
- packet.frameType = kVideoFrameKey;
- packet.isFirstPacket = false;
- packet.completeNALU = kNaluStart;
- packet.markerBit = false;
- TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
- // This packet should be decoded since it's the beginning of a NAL
- insertedLength += packet.sizeBytes;
-
- seqNum += 2;
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
- packet.frameType = kVideoFrameKey;
- packet.isFirstPacket = false;
- packet.completeNALU = kNaluEnd;
- packet.markerBit = true;
- TEST(kIncomplete == jb.InsertPacket(frameIn, packet));
- // This packet should not be decoded because it is an incomplete NAL if it
- // is the last
-
- frameOut = jb.GetFrameForDecoding();
- // Only last NALU is complete
- TEST(CheckOutFrame(frameOut, insertedLength, false) == 0);
- jb.ReleaseFrame(frameOut);
-
-
- // Test to insert empty packet
- seqNum += 1;
- timeStamp += 33 * 90;
- VCMPacket emptypacket(data, 0, seqNum, timeStamp, true);
- emptypacket.seqNum = seqNum;
- emptypacket.timestamp = timeStamp;
- emptypacket.frameType = kVideoFrameKey;
- emptypacket.isFirstPacket = true;
- emptypacket.completeNALU = kNaluComplete;
- emptypacket.markerBit = true;
- TEST(frameIn = jb.GetFrame(emptypacket));
- TEST(kFirstPacket == jb.InsertPacket(frameIn, emptypacket));
- // This packet should not be decoded because it is an incomplete NAL if it
- // is the last
- insertedLength += 0;
-
- // Will be sent to the decoder, as a packet belonging to a subsequent frame
- // has arrived.
- frameOut = jb.GetFrameForDecoding();
-
-
- // Test that a frame can include an empty packet.
- seqNum += 1;
- timeStamp += 33 * 90;
-
- packet.seqNum = seqNum;
- packet.timestamp = timeStamp;
- packet.frameType = kVideoFrameKey;
- packet.isFirstPacket = true;
- packet.completeNALU = kNaluComplete;
- packet.markerBit = false;
- TEST(frameIn = jb.GetFrame(packet));
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- seqNum += 1;
- emptypacket.seqNum = seqNum;
- emptypacket.timestamp = timeStamp;
- emptypacket.frameType = kVideoFrameKey;
- emptypacket.isFirstPacket = true;
- emptypacket.completeNALU = kNaluComplete;
- emptypacket.markerBit = true;
- TEST(kCompleteSession == jb.InsertPacket(frameIn, emptypacket));
-
- // get the frame
- frameOut = jb.GetCompleteFrameForDecoding(10);
- // Only last NALU is complete
- TEST(CheckOutFrame(frameOut, packet.sizeBytes, false) == 0);
-
- jb.ReleaseFrame(frameOut);
-
- jb.Flush();
-
- // Test that a we cannot get incomplete frames from the JB if we haven't
- // received the marker bit, unless we have received a packet from a later
- // timestamp.
-
- packet.seqNum += 2;
- packet.frameType = kVideoFrameDelta;
- packet.isFirstPacket = false;
- packet.markerBit = false;
-
- TEST(frameIn = jb.GetFrame(packet));
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- frameOut = jb.GetFrameForDecoding();
- TEST(frameOut == NULL);
-
- packet.seqNum += 2;
- packet.timestamp += 33 * 90;
-
- TEST(frameIn = jb.GetFrame(packet));
- TEST(kFirstPacket == jb.InsertPacket(frameIn, packet));
-
- frameOut = jb.GetFrameForDecoding();
-
- TEST(frameOut != NULL);
- TEST(CheckOutFrame(frameOut, packet.sizeBytes, false) == 0);
- jb.ReleaseFrame(frameOut);
-
- jb.Stop();
-
- printf("DONE !!!\n");
-
- printf("\nVCM Jitter Buffer Test: \n\n%i tests completed\n",
- vcmMacrosTests);
- if (vcmMacrosErrors > 0)
- {
- printf("%i FAILED\n\n", vcmMacrosErrors);
- }
- else
- {
- printf("ALL PASSED\n\n");
- }
-
- return 0;
-
-}
diff --git a/trunk/src/modules/video_coding/main/test/jitter_estimate_test.cc b/trunk/src/modules/video_coding/main/test/jitter_estimate_test.cc
deleted file mode 100644
index ac74a8a..0000000
--- a/trunk/src/modules/video_coding/main/test/jitter_estimate_test.cc
+++ /dev/null
@@ -1,109 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <ctime>
-#include "JitterEstimateTest.h"
-
-using namespace webrtc;
-
-JitterEstimateTest::JitterEstimateTest(unsigned int frameRate) :
-_frameRate(frameRate),
-_capacity(2000),
-_rate(500),
-_jitter(5, 0),
-_keyFrameRate(1.0),
-_deltaFrameSize(10000, 1e6),
-_counter(0),
-_lossrate(0.0)
-{
- // Assign to random value between 0 and max of unsigned int
- _seed = static_cast<unsigned>(std::time(0));
- std::srand(_seed);
- _prevTimestamp = static_cast<unsigned int>((std::rand() + 1.0)/(RAND_MAX + 1.0)*(pow((float) 2, (long) sizeof(unsigned int)*8)-1));
- _prevWallClock = VCMTickTime::MillisecondTimestamp();
-}
-
-FrameSample
-JitterEstimateTest::GenerateFrameSample()
-{
- double increment = 1.0/_frameRate;
- unsigned int frameSize = static_cast<unsigned int>(_deltaFrameSize.RandValue());
- bool keyFrame = false;
- bool resent = false;
- _prevTimestamp += static_cast<unsigned int>(90000*increment + 0.5);
- double deltaFrameRate = _frameRate - _keyFrameRate;
- double ratio = deltaFrameRate/static_cast<double>(_keyFrameRate);
- if (ratio < 1.0)
- {
- ratio = 1.0/ratio;
- if (_counter >= ratio)
- _counter = 0;
- else
- {
- _counter++;
- frameSize += static_cast<unsigned int>(3*_deltaFrameSize.GetAverage());
- keyFrame = true;
- }
- }
- else
- {
- if (_counter >= ratio)
- {
- frameSize += static_cast<unsigned int>(3*_deltaFrameSize.GetAverage());
- _counter = 0;
- keyFrame = true;
- }
- else
- _counter++;
- }
- WebRtc_Word64 jitter = static_cast<WebRtc_Word64>(_jitter.RandValue() + 1.0/_capacity * frameSize + 0.5);
- _prevWallClock += static_cast<WebRtc_Word64>(1000*increment + 0.5);
- double rndValue = RandUniform();
- resent = (rndValue < _lossrate);
- //printf("rndValue = %f\n", rndValue);
- return FrameSample(_prevTimestamp, _prevWallClock + jitter, frameSize, keyFrame, resent);
-}
-
-void
-JitterEstimateTest::SetCapacity(unsigned int c)
-{
- _capacity = c;
-}
-
-void
-JitterEstimateTest::SetRate(unsigned int r)
-{
- _rate = r;
-}
-
-void
-JitterEstimateTest::SetJitter(double m, double v)
-{
- _jitter.SetParams(m, v);
-}
-
-void
-JitterEstimateTest::SetFrameSizeStats(double m, double v)
-{
- _deltaFrameSize.SetParams(m, v);
-}
-
-void
-JitterEstimateTest::SetKeyFrameRate(int rate)
-{
- _keyFrameRate = rate;
-}
-
-void
-JitterEstimateTest::SetLossRate(double rate)
-{
- _lossrate = rate;
-}
diff --git a/trunk/src/modules/video_coding/main/test/jitter_estimate_test.h b/trunk/src/modules/video_coding/main/test/jitter_estimate_test.h
deleted file mode 100644
index cd7338a..0000000
--- a/trunk/src/modules/video_coding/main/test/jitter_estimate_test.h
+++ /dev/null
@@ -1,105 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_JITTER_ESTIMATE_TEST_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TEST_JITTER_ESTIMATE_TEST_H_
-
-#include "typedefs.h"
-#include "jitter_buffer.h"
-#include "jitter_estimator.h"
-#include <cstdlib>
-#include <cmath>
-
-double const pi = 4*std::atan(1.0);
-
-class GaussDist
-{
-public:
- GaussDist(double m, double v): _mu(m), _sigma(sqrt(v)) {}
-
- double RandValue() // returns a single normally distributed number
- {
- double r1 = (std::rand() + 1.0)/(RAND_MAX + 1.0); // gives equal distribution in (0, 1]
- double r2 = (std::rand() + 1.0)/(RAND_MAX + 1.0);
- return _mu + _sigma * std::sqrt(-2*std::log(r1))*std::cos(2*pi*r2);
- }
-
- double GetAverage()
- {
- return _mu;
- }
-
- double GetVariance()
- {
- return _sigma*_sigma;
- }
-
- void SetParams(double m, double v)
- {
- _mu = m;
- _sigma = sqrt(v);
- }
-
-private:
- double _mu, _sigma;
-};
-
-class JitterEstimateTestWrapper : public webrtc::VCMJitterEstimator
-{
-public:
- JitterEstimateTestWrapper() : VCMJitterEstimator() {}
- double GetTheta() { return _theta[0]; }
- double GetVarNoise() { return _varNoise; }
-};
-
-class FrameSample
-{
-public:
- FrameSample() {FrameSample(0, 0, 0, false, false);}
- FrameSample(unsigned int ts, WebRtc_Word64 wallClk, unsigned int fs, bool _keyFrame, bool _resent):
- timestamp90Khz(ts), wallClockMs(wallClk), frameSize(fs), keyFrame(_keyFrame), resent(_resent) {}
-
- unsigned int timestamp90Khz;
- WebRtc_Word64 wallClockMs;
- unsigned int frameSize;
- bool keyFrame;
- bool resent;
-};
-
-class JitterEstimateTest
-{
-public:
- JitterEstimateTest(unsigned int frameRate);
- FrameSample GenerateFrameSample();
- void SetCapacity(unsigned int c);
- void SetRate(unsigned int r);
- void SetJitter(double m, double v);
- void SetFrameSizeStats(double m, double v);
- void SetKeyFrameRate(int rate);
- void SetLossRate(double rate);
-
-private:
- double RandUniform() { return (std::rand() + 1.0)/(RAND_MAX + 1.0); }
- unsigned int _frameRate;
- unsigned int _capacity;
- unsigned int _rate;
- GaussDist _jitter;
- //GaussDist _noResend;
- GaussDist _deltaFrameSize;
- unsigned int _prevTimestamp;
- WebRtc_Word64 _prevWallClock;
- unsigned int _nextDelay;
- double _keyFrameRate;
- unsigned int _counter;
- unsigned int _seed;
- double _lossrate;
-};
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_JITTER_ESTIMATE_TEST_H_
diff --git a/trunk/src/modules/video_coding/main/test/media_opt_test.cc b/trunk/src/modules/video_coding/main/test/media_opt_test.cc
deleted file mode 100644
index f767919..0000000
--- a/trunk/src/modules/video_coding/main/test/media_opt_test.cc
+++ /dev/null
@@ -1,560 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Implementation of Media Optimization Test
-// testing is done via the VCM module, no specific Media opt functionality.
-
-#include "media_opt_test.h"
-
-#include <string.h>
-#include <stdio.h>
-#include <time.h>
-#include <vector>
-
-#include "../source/event.h"
-#include "receiver_tests.h" // receive side callbacks
-#include "test_callbacks.h"
-#include "test_macros.h"
-#include "test_util.h" // send side callback
-#include "testsupport/metrics/video_metrics.h"
-#include "video_coding.h"
-
-
-using namespace webrtc;
-
-int MediaOptTest::RunTest(int testNum, CmdArgs& args)
-{
- Trace::CreateTrace();
- Trace::SetTraceFile((test::OutputPath() + "mediaOptTestTrace.txt").c_str());
- Trace::SetLevelFilter(webrtc::kTraceAll);
- TickTimeBase clock;
- VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
- MediaOptTest* mot = new MediaOptTest(vcm, &clock);
- if (testNum == 0)
- { // regular
- mot->Setup(0, args);
- mot->GeneralSetup();
- mot->Perform();
- mot->Print(1);// print to screen
- mot->TearDown();
- }
- if (testNum == 1)
- { // release test
- mot->Setup(0, args);
- mot->RTTest();
- }
- if (testNum == 2)
- { // release test, running from script
- mot->Setup(1, args);
- mot->GeneralSetup();
- mot->Perform();
- mot->Print(1);// print to screen
- mot->TearDown();
- }
-
- VideoCodingModule::Destroy(vcm);
- delete mot;
- Trace::ReturnTrace();
- return 0;
-
-}
-
-
-MediaOptTest::MediaOptTest(VideoCodingModule* vcm, TickTimeBase* clock):
-_vcm(vcm),
-_clock(clock),
-_width(0),
-_height(0),
-_lengthSourceFrame(0),
-_timeStamp(0),
-_frameRate(30.0f),
-_nackEnabled(false),
-_fecEnabled(false),
-_rttMS(0),
-_bitRate(300.0f),
-_lossRate(0.0f),
-_renderDelayMs(0),
-_frameCnt(0),
-_sumEncBytes(0),
-_numFramesDropped(0),
-_numberOfCores(4)
-{
- _rtp = RtpRtcp::CreateRtpRtcp(1, false);
-}
-
-MediaOptTest::~MediaOptTest()
-{
- RtpRtcp::DestroyRtpRtcp(_rtp);
-}
-void
-MediaOptTest::Setup(int testType, CmdArgs& args)
-{
- /*TEST USER SETTINGS*/
- // test parameters
- _inname = args.inputFile;
- if (args.outputFile == "")
- _outname = test::OutputPath() + "MOTest_out.vp8";
- else
- _outname = args.outputFile;
- // actual source after frame dropping
- _actualSourcename = test::OutputPath() + "MOTestSource.yuv";
- _codecName = args.codecName;
- _sendCodecType = args.codecType;
- _width = args.width;
- _height = args.height;
- _frameRate = args.frameRate;
- _bitRate = args.bitRate;
- _numberOfCores = 4;
-
- // error resilience
- _nackEnabled = false;
- _fecEnabled = true;
- _nackFecEnabled = false;
-
- _rttMS = 100;
- _lossRate = 0.00*255; // no packet loss
-
- _testType = testType;
-
- //For multiple runs with script
- if (_testType == 1)
- {
- float rateTest,lossTest;
- int numRuns;
- _fpinp = fopen("dat_inp","rb");
- _fpout = fopen("test_runs/dat_out","ab");
- _fpout2 = fopen("test_runs/dat_out2","ab");
- TEST(fscanf(_fpinp,"%f %f %d \n",&rateTest,&lossTest,&numRuns) > 0);
- _bitRate = rateTest;
- _lossRate = lossTest;
- _testNum = 0;
-
- // for bit rates: 500, 1000, 2000, 3000,4000
- // for loss rates: 0, 1, 3, 5, 10%
- _numParRuns = 25;
-
- _testNum = numRuns + 1;
- if (rateTest == 0.0) _lossRate = 0.0;
- else
- {
- if (rateTest == 4000) //final bit rate
- {
- if (lossTest == 0.1*255) _lossRate = 0.0; //start at 1%
- else
- if (lossTest == 0.05*255) _lossRate = 0.1*255; //final loss rate
- else
- if (lossTest == 0.0) _lossRate = 0.01*255;
- else _lossRate = lossTest + 0.02*255;
- }
- }
-
- if (rateTest == 0.0 || rateTest == 4000) _bitRate = 500; //starting bit rate
- else
- if (rateTest == 500) _bitRate = 1000;
- else _bitRate = rateTest + 1000;
- }
- //
-
- _renderDelayMs = 0;
- /* test settings end*/
-
- _lengthSourceFrame = 3*_width*_height/2;
- _log.open((test::OutputPath() + "VCM_MediaOptLog.txt").c_str(),
- std::fstream::out | std::fstream::app);
- return;
-}
-
-void
-MediaOptTest::GeneralSetup()
-{
- WebRtc_UWord32 minPlayoutDelayMs = 0;
-
- if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
- {
- printf("Cannot read file %s.\n", _inname.c_str());
- exit(1);
- }
-
- if ((_decodedFile = fopen(_outname.c_str(), "wb")) == NULL)
- {
- printf("Cannot read file %s.\n", _outname.c_str());
- exit(1);
- }
-
- if ((_actualSourceFile = fopen(_actualSourcename.c_str(), "wb")) == NULL)
- {
- printf("Cannot read file %s.\n", _actualSourcename.c_str());
- exit(1);
- }
-
- if (_rtp->InitReceiver() < 0)
- {
- exit(1);
- }
- if (_rtp->InitSender() < 0)
- {
- exit(1);
- }
- if (_vcm->InitializeReceiver() < 0)
- {
- exit(1);
- }
- if (_vcm->InitializeSender())
- {
- exit(1);
- }
-
- // Registering codecs for the RTP module
-
- // Register receive and send payload
- VideoCodec videoCodec;
- strncpy(videoCodec.plName, "VP8", 32);
- videoCodec.plType = VCM_VP8_PAYLOAD_TYPE;
- _rtp->RegisterReceivePayload(videoCodec);
- _rtp->RegisterSendPayload(videoCodec);
-
- strncpy(videoCodec.plName, "ULPFEC", 32);
- videoCodec.plType = VCM_ULPFEC_PAYLOAD_TYPE;
- _rtp->RegisterReceivePayload(videoCodec);
- _rtp->RegisterSendPayload(videoCodec);
-
- strncpy(videoCodec.plName, "RED", 32);
- videoCodec.plType = VCM_RED_PAYLOAD_TYPE;
- _rtp->RegisterReceivePayload(videoCodec);
- _rtp->RegisterSendPayload(videoCodec);
-
- if (_nackFecEnabled == 1)
- _rtp->SetGenericFECStatus(_nackFecEnabled, VCM_RED_PAYLOAD_TYPE,
- VCM_ULPFEC_PAYLOAD_TYPE);
- else
- _rtp->SetGenericFECStatus(_fecEnabled, VCM_RED_PAYLOAD_TYPE,
- VCM_ULPFEC_PAYLOAD_TYPE);
-
- // VCM: Registering codecs
- VideoCodec sendCodec;
- _vcm->InitializeSender();
- _vcm->InitializeReceiver();
- WebRtc_Word32 numberOfCodecs = _vcm->NumberOfCodecs();
- if (numberOfCodecs < 1)
- {
- exit(1);
- }
-
- if (_vcm->Codec(_sendCodecType, &sendCodec) != 0)
- {
- printf("Unknown codec\n");
- exit(1);
- }
- // register codec
- sendCodec.startBitrate = (int) _bitRate;
- sendCodec.height = _height;
- sendCodec.width = _width;
- sendCodec.maxFramerate = (WebRtc_UWord8)_frameRate;
- _vcm->RegisterSendCodec(&sendCodec, _numberOfCores, 1440);
- _vcm->RegisterReceiveCodec(&sendCodec, _numberOfCores); // same settings for encode and decode
-
- _vcm->SetRenderDelay(_renderDelayMs);
- _vcm->SetMinimumPlayoutDelay(minPlayoutDelayMs);
-
- return;
-}
-// The following test shall be conducted under release tests
-
-
-
-WebRtc_Word32
-MediaOptTest::Perform()
-{
- //Setup();
- EventWrapper* waitEvent = EventWrapper::Create();
-
- // callback settings
- VCMRTPEncodeCompleteCallback* encodeCompleteCallback = new VCMRTPEncodeCompleteCallback(_rtp);
- _vcm->RegisterTransportCallback(encodeCompleteCallback);
- encodeCompleteCallback->SetCodecType(ConvertCodecType(_codecName.c_str()));
- encodeCompleteCallback->SetFrameDimensions(_width, _height);
- // frame ready to be sent to network
- RTPSendCompleteCallback* outgoingTransport =
- new RTPSendCompleteCallback(_rtp, _clock);
- _rtp->RegisterSendTransport(outgoingTransport);
- //FrameReceiveCallback
- VCMDecodeCompleteCallback receiveCallback(_decodedFile);
- RtpDataCallback dataCallback(_vcm);
- _rtp->RegisterIncomingDataCallback(&dataCallback);
-
- VideoProtectionCallback protectionCallback;
- protectionCallback.RegisterRtpModule(_rtp);
- _vcm->RegisterProtectionCallback(&protectionCallback);
-
- // set error resilience / test parameters:
- outgoingTransport->SetLossPct(_lossRate);
- if (_nackFecEnabled == 1)
- _vcm->SetVideoProtection(kProtectionNackFEC, _nackFecEnabled);
- else
- {
- _vcm->SetVideoProtection(kProtectionNack, _nackEnabled);
- _vcm->SetVideoProtection(kProtectionFEC, _fecEnabled);
- }
-
- // START TEST
- VideoFrame sourceFrame;
- sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
- WebRtc_UWord8* tmpBuffer = new WebRtc_UWord8[_lengthSourceFrame];
- _vcm->SetChannelParameters((WebRtc_UWord32)_bitRate, (WebRtc_UWord8)_lossRate, _rttMS);
- _vcm->RegisterReceiveCallback(&receiveCallback);
-
- _frameCnt = 0;
- _sumEncBytes = 0.0;
- _numFramesDropped = 0;
-
- while (feof(_sourceFile)== 0)
- {
- TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
- _frameCnt++;
-
- sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
- sourceFrame.SetHeight(_height);
- sourceFrame.SetWidth(_width);
- _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(_frameRate));
- sourceFrame.SetTimeStamp(_timeStamp);
- TEST(_vcm->AddVideoFrame(sourceFrame) == VCM_OK);
- // inform RTP Module of error resilience features
- //_rtp->SetFECCodeRate(protectionCallback.FECKeyRate(),protectionCallback.FECDeltaRate());
- //_rtp->SetNACKStatus(protectionCallback.NACKMethod());
-
- WebRtc_Word32 ret = _vcm->Decode();
- if (ret < 0 )
- {
- TEST(ret == 0);
- printf ("Decode error in frame # %d",_frameCnt);
- }
-
- float encBytes = encodeCompleteCallback->EncodedBytes();
- if (encBytes == 0)
- {
- _numFramesDropped += 1;
- //printf("frame #%d dropped \n", _frameCnt );
- }
- else
- {
- // write frame to file
- fwrite(sourceFrame.Buffer(), 1, sourceFrame.Length(), _actualSourceFile);
- }
-
- _sumEncBytes += encBytes;
- //waitEvent->Wait(33);
- }
-
- //END TEST
- delete waitEvent;
- delete encodeCompleteCallback;
- delete outgoingTransport;
- delete tmpBuffer;
-
-return 0;
-
-}
-
-void
-MediaOptTest::RTTest()
-{
- // will only calculate PSNR - not create output files for all
- // SET UP
- // Set bit rates
- const float bitRateVec[] = {500, 1000, 2000,3000, 4000};
- //const float bitRateVec[] = {1000};
- // Set Packet loss values ([0,255])
- const double lossPctVec[] = {0.0*255, 0.0*255, 0.01*255, 0.01*255, 0.03*255, 0.03*255, 0.05*255, 0.05*255, 0.1*255, 0.1*255};
- const bool nackEnabledVec[] = {false , false, false, false, false, false, false, false , false, false};
- const bool fecEnabledVec[] = {false , true, false, true , false, true , false, true , false, true};
- // fec and nack are set according to the packet loss values
-
- const float nBitrates = sizeof(bitRateVec)/sizeof(*bitRateVec);
- const float nlossPct = sizeof(lossPctVec)/sizeof(*lossPctVec);
-
- std::vector<const VideoSource*> sources;
- std::vector<const VideoSource*>::iterator it;
-
- sources.push_back(new const VideoSource(_inname, _width, _height));
- int numOfSrc = 1;
-
- // constant settings (valid for entire run time)
- _rttMS = 20;
- _renderDelayMs = 0;
-
- // same out name for all
- _outname = test::OutputPath() + "RTMOTest_out.yuv";
- // actual source after frame dropping
- _actualSourcename = test::OutputPath() + "RTMOTestSource.yuv";
-
- _codecName = "VP8"; // for now just this one - later iterate over all codec types
- _log.open((test::OutputPath() + "/VCM_RTMediaOptLog.txt").c_str(),
- std::fstream::out | std::fstream::app);
- _outputRes=fopen((test::OutputPath() + "VCM_MediaOptResults.txt").c_str(),
- "ab");
-
- //char filename[128];
- /* test settings end*/
-
- // START TEST
- // iterate over test sequences
- printf("\n****START TEST OVER ALL RUNS ****\n");
- int runCnt = 0;
- for (it = sources.begin() ; it < sources.end(); it++)
- {
-
- // test set up
- _inname = (*it)->GetFileName();
- _width = (*it)->GetWidth();
- _height = (*it)->GetHeight();
- _lengthSourceFrame = 3*_width*_height/2;
- _frameRate = (*it)->GetFrameRate();
- //GeneralSetup();
-
-
- // iterate over all bit rates
- for (int i = 0; i < nBitrates; i++)
- {
- _bitRate = static_cast<float>(bitRateVec[i]);
- // iterate over all packet loss values
- for (int j = 0; j < nlossPct; j++)
- {
- _lossRate = static_cast<float>(lossPctVec[j]);
- _nackEnabled = static_cast<bool>(nackEnabledVec[j]);
- _fecEnabled = static_cast<bool>(fecEnabledVec[j]);
-
- runCnt++;
- printf("run #%d out of %d \n", runCnt,(int)(nlossPct*nBitrates*numOfSrc));
-
- //printf("**FOR RUN: **%d %d %d %d \n",_nackEnabled,_fecEnabled,int(lossPctVec[j]),int(_bitRate));
-
- /*
- int ch = sprintf(filename,"../test_mediaOpt/RTMOTest_%d_%d_%d_%d.yuv",_nackEnabled,_fecEnabled,int(lossPctVec[j]),int(_bitRate));
- _outname = filename;
-
- printf("**FOR RUN: **%d %d %d %d \n",_nackEnabled,_fecEnabled,int(lossPctVec[j]),int(_bitRate));
- */
- if (_rtp != NULL)
- {
- RtpRtcp::DestroyRtpRtcp(_rtp);
- }
- _rtp = RtpRtcp::CreateRtpRtcp(1, false);
- GeneralSetup();
- Perform();
- Print(1);
- TearDown();
- RtpRtcp::DestroyRtpRtcp(_rtp);
- _rtp = NULL;
-
- printf("\n");
- //printf("**DONE WITH RUN: **%d %d %f %d \n",_nackEnabled,_fecEnabled,lossPctVec[j],int(_bitRate));
- //
-
- }// end of packet loss loop
- }// end of bit rate loop
- delete *it;
- }// end of video sequence loop
- // at end of sequence
- fclose(_outputRes);
- printf("\nVCM Media Optimization Test: \n\n%i tests completed\n", vcmMacrosTests);
- if (vcmMacrosErrors > 0)
- {
- printf("%i FAILED\n\n", vcmMacrosErrors);
- }
- else
- {
- printf("ALL PASSED\n\n");
- }
-}
-
-
-void
-MediaOptTest::Print(int mode)
-{
- double ActualBitRate = 8.0 *( _sumEncBytes / (_frameCnt / _frameRate));
- double actualBitRate = ActualBitRate / 1000.0;
- webrtc::test::QualityMetricsResult psnr;
- I420PSNRFromFiles(_actualSourcename.c_str(), _outname.c_str(), _width,
- _height, &psnr);
-
- (_log) << "VCM: Media Optimization Test Cycle Completed!" << std::endl;
- (_log) << "Input file: " << _inname << std::endl;
- (_log) << "Output file:" << _outname << std::endl;
- ( _log) << "Actual bitrate: " << actualBitRate<< " kbps\tTarget: " << _bitRate << " kbps" << std::endl;
- (_log) << "Error Reslience: NACK:" << _nackEnabled << "; FEC: " << _fecEnabled << std::endl;
- (_log) << "Packet Loss applied= %f " << _lossRate << std::endl;
- (_log) << _numFramesDropped << " FRames were dropped" << std::endl;
- ( _log) << "PSNR: " << psnr.average << std::endl;
- (_log) << std::endl;
-
- if (_testType == 2)
- {
- fprintf(_outputRes,"************\n");
- fprintf(_outputRes,"\n\n\n");
- fprintf(_outputRes,"Actual bitrate: %f kbps\n", actualBitRate);
- fprintf(_outputRes,"Target bitrate: %f kbps\n", _bitRate);
- fprintf(_outputRes,"NACK: %s ",(_nackEnabled)?"true":"false");
- fprintf(_outputRes,"FEC: %s \n ",(_fecEnabled)?"true":"false");
- fprintf(_outputRes,"Packet loss applied = %f\n", _lossRate);
- fprintf(_outputRes,"%d frames were dropped, and total number of frames processed %d \n",_numFramesDropped,_frameCnt);
- fprintf(_outputRes,"PSNR: %f \n", psnr.average);
- fprintf(_outputRes,"************\n");
- }
-
-
- //
- if (_testType == 1)
- {
- fprintf(_fpout,"************\n");
- fprintf(_fpout,"\n\n\n");
- fprintf(_fpout,"Actual bitrate: %f kbps\n", actualBitRate);
- fprintf(_fpout,"Target bitrate: %f kbps\n", _bitRate);
- fprintf(_fpout,"NACK: %s ",(_nackEnabled)?"true":"false");
- fprintf(_fpout,"FEC: %s \n ",(_fecEnabled)?"true":"false");
- fprintf(_fpout,"Packet loss applied = %f\n", _lossRate);
- fprintf(_fpout,"%d frames were dropped, and total number of frames processed %d \n",_numFramesDropped,_frameCnt);
- fprintf(_fpout,"PSNR: %f \n", psnr.average);
- fprintf(_fpout,"************\n");
-
- int testNum1 = _testNum/(_numParRuns +1) + 1;
- int testNum2 = _testNum%_numParRuns;
- if (testNum2 == 0) testNum2 = _numParRuns;
- fprintf(_fpout2,"%d %d %f %f %f %f \n",testNum1,testNum2,_bitRate,actualBitRate,_lossRate,psnr.average);
- fclose(_fpinp);
- _fpinp = fopen("dat_inp","wb");
- fprintf(_fpinp,"%f %f %d \n",_bitRate,_lossRate,_testNum);
- }
- //
-
-
- if (mode == 1)
- {
- // print to screen
- printf("\n\n\n");
- printf("Actual bitrate: %f kbps\n", actualBitRate);
- printf("Target bitrate: %f kbps\n", _bitRate);
- printf("NACK: %s ",(_nackEnabled)?"true":"false");
- printf("FEC: %s \n",(_fecEnabled)?"true":"false");
- printf("Packet loss applied = %f\n", _lossRate);
- printf("%d frames were dropped, and total number of frames processed %d \n",_numFramesDropped,_frameCnt);
- printf("PSNR: %f \n", psnr.average);
- }
- TEST(psnr.average > 10); // low becuase of possible frame dropping (need to verify that OK for all packet loss values/ rates)
-}
-
-void
-MediaOptTest::TearDown()
-{
- _log.close();
- fclose(_sourceFile);
- fclose(_decodedFile);
- fclose(_actualSourceFile);
- return;
-}
diff --git a/trunk/src/modules/video_coding/main/test/media_opt_test.h b/trunk/src/modules/video_coding/main/test/media_opt_test.h
deleted file mode 100644
index 7d4e226..0000000
--- a/trunk/src/modules/video_coding/main/test/media_opt_test.h
+++ /dev/null
@@ -1,93 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// VCM Media Optimization Test
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_MEDIA_OPT_TEST_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TEST_MEDIA_OPT_TEST_H_
-
-
-#include <string>
-
-#include "rtp_rtcp.h"
-#include "test_util.h"
-#include "video_coding.h"
-#include "video_source.h"
-
-// media optimization test
-// This test simulates a complete encode-decode cycle via the RTP module.
-// allows error resilience tests, packet loss tests, etc.
-// Does not test the media optimization deirectly, but via the VCM API only.
-// The test allows two modes:
-// 1 - Standard, basic settings, one run
-// 2 - Release test - iterates over a number of video sequences, bit rates, packet loss values ,etc.
-
-class MediaOptTest
-{
-public:
- MediaOptTest(webrtc::VideoCodingModule* vcm,
- webrtc::TickTimeBase* clock);
- ~MediaOptTest();
-
- static int RunTest(int testNum, CmdArgs& args);
- // perform encode-decode of an entire sequence
- WebRtc_Word32 Perform();
- // Set up for a single mode test
- void Setup(int testType, CmdArgs& args);
- // General set up - applicable for both modes
- void GeneralSetup();
- // Run release testing
- void RTTest();
- void TearDown();
- // mode = 1; will print to screen, otherwise only to log file
- void Print(int mode);
-
-private:
-
- webrtc::VideoCodingModule* _vcm;
- webrtc::RtpRtcp* _rtp;
- webrtc::TickTimeBase* _clock;
- std::string _inname;
- std::string _outname;
- std::string _actualSourcename;
- std::fstream _log;
- FILE* _sourceFile;
- FILE* _decodedFile;
- FILE* _actualSourceFile;
- FILE* _outputRes;
- WebRtc_UWord16 _width;
- WebRtc_UWord16 _height;
- WebRtc_UWord32 _lengthSourceFrame;
- WebRtc_UWord32 _timeStamp;
- float _frameRate;
- bool _nackEnabled;
- bool _fecEnabled;
- bool _nackFecEnabled;
- WebRtc_UWord8 _rttMS;
- float _bitRate;
- double _lossRate;
- WebRtc_UWord32 _renderDelayMs;
- WebRtc_Word32 _frameCnt;
- float _sumEncBytes;
- WebRtc_Word32 _numFramesDropped;
- std::string _codecName;
- webrtc::VideoCodecType _sendCodecType;
- WebRtc_Word32 _numberOfCores;
-
- //for release test#2
- FILE* _fpinp;
- FILE* _fpout;
- FILE* _fpout2;
- int _testType;
- int _testNum;
- int _numParRuns;
-
-}; // end of MediaOptTest class definition
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_MEDIA_OPT_TEST_H_
diff --git a/trunk/src/modules/video_coding/main/test/mt_rx_tx_test.cc b/trunk/src/modules/video_coding/main/test/mt_rx_tx_test.cc
deleted file mode 100644
index d0a39ad..0000000
--- a/trunk/src/modules/video_coding/main/test/mt_rx_tx_test.cc
+++ /dev/null
@@ -1,362 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*************************************************
- *
- * Testing multi thread - receive and send sides
- *
- **************************************************/
-
-#include <string.h>
-
-#include "../source/event.h"
-#include "media_opt_test.h"
-#include "mt_test_common.h"
-#include "receiver_tests.h" // shared RTP state and receive side threads
-#include "rtp_rtcp.h"
-#include "test_macros.h"
-#include "test_util.h" // send side callback
-#include "thread_wrapper.h"
-#include "video_coding.h"
-
-using namespace webrtc;
-
-bool
-MainSenderThread(void* obj)
-{
- SendSharedState* state = static_cast<SendSharedState*>(obj);
- EventWrapper& waitEvent = *EventWrapper::Create();
- // preparing a frame for encoding
- VideoFrame sourceFrame;
- WebRtc_Word32 width = state->_args.width;
- WebRtc_Word32 height = state->_args.height;
- float frameRate = state->_args.frameRate;
- WebRtc_Word32 lengthSourceFrame = 3*width*height/2;
- sourceFrame.VerifyAndAllocate(lengthSourceFrame);
- WebRtc_UWord8* tmpBuffer = new WebRtc_UWord8[lengthSourceFrame];
-
- if (state->_sourceFile == NULL)
- {
- state->_sourceFile = fopen(state->_args.inputFile.c_str(), "rb");
- if (state->_sourceFile == NULL)
- {
- printf ("Error when opening file \n");
- delete &waitEvent;
- delete [] tmpBuffer;
- return false;
- }
- }
- if (feof(state->_sourceFile) == 0)
- {
- TEST(fread(tmpBuffer, 1, lengthSourceFrame,state->_sourceFile) > 0 ||
- feof(state->_sourceFile));
- state->_frameCnt++;
- sourceFrame.CopyFrame(lengthSourceFrame, tmpBuffer);
- sourceFrame.SetHeight(height);
- sourceFrame.SetWidth(width);
- state->_timestamp += (WebRtc_UWord32)(9e4 / frameRate);
- sourceFrame.SetTimeStamp(state->_timestamp);
-
- WebRtc_Word32 ret = state->_vcm.AddVideoFrame(sourceFrame);
- if (ret < 0)
- {
- printf("Add Frame error: %d\n", ret);
- delete &waitEvent;
- delete [] tmpBuffer;
- return false;
- }
- waitEvent.Wait(33);
- }
-
- delete &waitEvent;
- delete [] tmpBuffer;
-
- return true;
-}
-
-bool
-IntSenderThread(void* obj)
-{
- SendSharedState* state = static_cast<SendSharedState*>(obj);
- state->_vcm.SetChannelParameters(1000,30,0);
-
- return true;
-}
-
-
-int MTRxTxTest(CmdArgs& args)
-{
- /* TEST SETTINGS */
- std::string inname = args.inputFile;
- std::string outname;
- if (args.outputFile == "")
- outname = test::OutputPath() + "MTRxTxTest_decoded.yuv";
- else
- outname = args.outputFile;
-
- WebRtc_UWord16 width = args.width;
- WebRtc_UWord16 height = args.height;
-
- float frameRate = args.frameRate;
- float bitRate = args.bitRate;
- WebRtc_Word32 numberOfCores = 1;
-
- // error resilience/network
- // Nack support is currently not implemented in this test.
- bool nackEnabled = false;
- bool fecEnabled = false;
- WebRtc_UWord8 rttMS = 20;
- float lossRate = 0.0*255; // no packet loss
- WebRtc_UWord32 renderDelayMs = 0;
- WebRtc_UWord32 minPlayoutDelayMs = 0;
-
- /* TEST SET-UP */
-
- // Set up trace
- Trace::CreateTrace();
- Trace::SetTraceFile((test::OutputPath() + "MTRxTxTestTrace.txt").c_str());
- Trace::SetLevelFilter(webrtc::kTraceAll);
-
- FILE* sourceFile;
- FILE* decodedFile;
-
- if ((sourceFile = fopen(inname.c_str(), "rb")) == NULL)
- {
- printf("Cannot read file %s.\n", inname.c_str());
- return -1;
- }
-
- if ((decodedFile = fopen(outname.c_str(), "wb")) == NULL)
- {
- printf("Cannot read file %s.\n", outname.c_str());
- return -1;
- }
-
- //RTP
- RtpRtcp* rtp = RtpRtcp::CreateRtpRtcp(1, false);
- if (rtp->InitReceiver() < 0)
- {
- return -1;
- }
- if (rtp->InitSender() < 0)
- {
- return -1;
- }
- // registering codecs for the RTP module
- VideoCodec videoCodec;
- strncpy(videoCodec.plName, "ULPFEC", 32);
- videoCodec.plType = VCM_ULPFEC_PAYLOAD_TYPE;
- TEST(rtp->RegisterReceivePayload(videoCodec) == 0);
-
- strncpy(videoCodec.plName, "RED", 32);
- videoCodec.plType = VCM_RED_PAYLOAD_TYPE;
- TEST(rtp->RegisterReceivePayload(videoCodec) == 0);
-
- strncpy(videoCodec.plName, args.codecName.c_str(), 32);
- videoCodec.plType = VCM_VP8_PAYLOAD_TYPE;
- videoCodec.maxBitrate = 10000;
- videoCodec.codecType = args.codecType;
- TEST(rtp->RegisterReceivePayload(videoCodec) == 0);
- TEST(rtp->RegisterSendPayload(videoCodec) == 0);
-
- // inform RTP Module of error resilience features
- TEST(rtp->SetGenericFECStatus(fecEnabled, VCM_RED_PAYLOAD_TYPE, VCM_ULPFEC_PAYLOAD_TYPE) == 0);
-
- //VCM
- TickTimeBase clock;
- VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
- if (vcm->InitializeReceiver() < 0)
- {
- return -1;
- }
- if (vcm->InitializeSender())
- {
- return -1;
- }
- // registering codecs for the VCM module
- VideoCodec sendCodec;
- vcm->InitializeSender();
- WebRtc_Word32 numberOfCodecs = vcm->NumberOfCodecs();
- if (numberOfCodecs < 1)
- {
- return -1;
- }
-
- if (vcm->Codec(args.codecType, &sendCodec) != 0)
- {
- // desired codec unavailable
- printf("Codec not registered\n");
- return -1;
- }
- // register codec
- sendCodec.startBitrate = (int) bitRate;
- sendCodec.height = height;
- sendCodec.width = width;
- sendCodec.maxFramerate = (WebRtc_UWord8)frameRate;
- vcm->RegisterSendCodec(&sendCodec, numberOfCores, 1440);
- vcm->RegisterReceiveCodec(&sendCodec, numberOfCores); // same settings for encode and decode
-
- vcm->SetRenderDelay(renderDelayMs);
- vcm->SetMinimumPlayoutDelay(minPlayoutDelayMs);
-
- // Callback Settings
-
- PacketRequester packetRequester(*rtp);
- vcm->RegisterPacketRequestCallback(&packetRequester);
-
- VCMRTPEncodeCompleteCallback* encodeCompleteCallback = new VCMRTPEncodeCompleteCallback(rtp);
- vcm->RegisterTransportCallback(encodeCompleteCallback);
- encodeCompleteCallback->SetCodecType(ConvertCodecType(args.codecName.c_str()));
- encodeCompleteCallback->SetFrameDimensions(width, height);
- // frame ready to be sent to network
- RTPSendCompleteCallback* outgoingTransport =
- new RTPSendCompleteCallback(rtp, &clock, "dump.rtp");
- rtp->RegisterSendTransport(outgoingTransport);
- // FrameReceiveCallback
- VCMDecodeCompleteCallback receiveCallback(decodedFile);
- RtpDataCallback dataCallback(vcm);
- rtp->RegisterIncomingDataCallback(&dataCallback);
- vcm->RegisterReceiveCallback(&receiveCallback);
-
- VideoProtectionCallback protectionCallback;
- vcm->RegisterProtectionCallback(&protectionCallback);
-
- outgoingTransport->SetLossPct(lossRate);
- // Nack support is currently not implemented in this test
- assert(nackEnabled == false);
- vcm->SetVideoProtection(kProtectionNack, nackEnabled);
- vcm->SetVideoProtection(kProtectionFEC, fecEnabled);
-
- // inform RTP Module of error resilience features
- rtp->SetFECCodeRate(protectionCallback.FECKeyRate(),
- protectionCallback.FECDeltaRate());
- rtp->SetNACKStatus(protectionCallback.NACKMethod());
-
- vcm->SetChannelParameters((WebRtc_UWord32) bitRate,
- (WebRtc_UWord8) lossRate, rttMS);
-
- SharedRTPState mtState(*vcm, *rtp); // receive side
- SendSharedState mtSendState(*vcm, *rtp, args); // send side
-
- /*START TEST*/
-
- // Create and start all threads
- // send side threads
- ThreadWrapper* mainSenderThread = ThreadWrapper::CreateThread(MainSenderThread,
- &mtSendState, kNormalPriority, "MainSenderThread");
- ThreadWrapper* intSenderThread = ThreadWrapper::CreateThread(IntSenderThread,
- &mtSendState, kNormalPriority, "IntThread");
-
- if (mainSenderThread != NULL)
- {
- unsigned int tid;
- mainSenderThread->Start(tid);
- }
- else
- {
- printf("Unable to start main sender thread\n");
- return -1;
- }
-
- if (intSenderThread != NULL)
- {
- unsigned int tid;
- intSenderThread->Start(tid);
- }
- else
- {
- printf("Unable to start sender interference thread\n");
- return -1;
- }
-
- // Receive side threads
- ThreadWrapper* processingThread = ThreadWrapper::CreateThread(ProcessingThread,
- &mtState, kNormalPriority, "ProcessingThread");
- ThreadWrapper* decodeThread = ThreadWrapper::CreateThread(DecodeThread,
- &mtState, kNormalPriority, "DecodeThread");
-
- if (processingThread != NULL)
- {
- unsigned int tid;
- processingThread->Start(tid);
- }
- else
- {
- printf("Unable to start processing thread\n");
- return -1;
- }
-
- if (decodeThread != NULL)
- {
- unsigned int tid;
- decodeThread->Start(tid);
- }
- else
- {
- printf("Unable to start decode thread\n");
- return -1;
- }
-
- EventWrapper& waitEvent = *EventWrapper::Create();
-
- // Decode for 10 seconds and then tear down and exit.
- waitEvent.Wait(30000);
-
- // Tear down
-
- while (!mainSenderThread->Stop())
- {
- ;
- }
-
- while (!intSenderThread->Stop())
- {
- ;
- }
-
-
- while (!processingThread->Stop())
- {
- ;
- }
-
- while (!decodeThread->Stop())
- {
- ;
- }
-
- printf("\nVCM MT RX/TX Test: \n\n%i tests completed\n", vcmMacrosTests);
- if (vcmMacrosErrors > 0)
- {
- printf("%i FAILED\n\n", vcmMacrosErrors);
- }
- else
- {
- printf("ALL PASSED\n\n");
- }
-
- delete &waitEvent;
- delete mainSenderThread;
- delete intSenderThread;
- delete processingThread;
- delete decodeThread;
- delete encodeCompleteCallback;
- delete outgoingTransport;
- VideoCodingModule::Destroy(vcm);
- RtpRtcp::DestroyRtpRtcp(rtp);
- rtp = NULL;
- vcm = NULL;
- Trace::ReturnTrace();
- fclose(decodedFile);
- printf("Multi-Thread test Done: View output file \n");
- return 0;
-
-}
-
diff --git a/trunk/src/modules/video_coding/main/test/mt_test_common.cc b/trunk/src/modules/video_coding/main/test/mt_test_common.cc
deleted file mode 100644
index e3ebd97..0000000
--- a/trunk/src/modules/video_coding/main/test/mt_test_common.cc
+++ /dev/null
@@ -1,139 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "mt_test_common.h"
-
-#include <cmath>
-
-#include "modules/video_coding/main/source/tick_time_base.h"
-#include "rtp_dump.h"
-
-namespace webrtc {
-
-TransportCallback::TransportCallback(webrtc::RtpRtcp* rtp,
- TickTimeBase* clock,
- const char* filename):
-RTPSendCompleteCallback(rtp, clock, filename)
-{
- //
-}
-
-TransportCallback::~TransportCallback()
-{
- //
-}
-
-int
-TransportCallback::SendPacket(int channel, const void *data, int len)
-{
- _sendCount++;
- _totalSentLength += len;
-
- if (_rtpDump != NULL)
- {
- if (_rtpDump->DumpPacket((const WebRtc_UWord8*)data, len) != 0)
- {
- return -1;
- }
- }
-
- bool transmitPacket = true;
- // Off-line tests, don't drop first Key frame (approx.)
- if (_sendCount > 20)
- {
- transmitPacket = PacketLoss();
- }
-
- TickTimeBase clock;
- int64_t now = clock.MillisecondTimestamp();
- // Insert outgoing packet into list
- if (transmitPacket)
- {
- RtpPacket* newPacket = new RtpPacket();
- memcpy(newPacket->data, data, len);
- newPacket->length = len;
- // Simulate receive time = network delay + packet jitter
- // simulated as a Normal distribution random variable with
- // mean = networkDelay and variance = jitterVar
- WebRtc_Word32
- simulatedDelay = (WebRtc_Word32)NormalDist(_networkDelayMs,
- sqrt(_jitterVar));
- newPacket->receiveTime = now + simulatedDelay;
- _rtpPackets.push_back(newPacket);
- }
- return 0;
-}
-
-int
-TransportCallback::TransportPackets()
-{
- // Are we ready to send packets to the receiver?
- RtpPacket* packet = NULL;
- TickTimeBase clock;
- int64_t now = clock.MillisecondTimestamp();
-
- while (!_rtpPackets.empty())
- {
- // Take first packet in list
- packet = _rtpPackets.front();
- WebRtc_Word64 timeToReceive = packet->receiveTime - now;
- if (timeToReceive > 0)
- {
- // No available packets to send
- break;
- }
-
- _rtpPackets.pop_front();
- // Send to receive side
- if (_rtp->IncomingPacket((const WebRtc_UWord8*)packet->data,
- packet->length) < 0)
- {
- delete packet;
- packet = NULL;
- // Will return an error after the first packet that goes wrong
- return -1;
- }
- delete packet;
- packet = NULL;
- }
- return 0; // OK
-}
-
-
-
-bool VCMProcessingThread(void* obj)
-{
- SharedRTPState* state = static_cast<SharedRTPState*>(obj);
- if (state->_vcm.TimeUntilNextProcess() <= 0)
- {
- if (state->_vcm.Process() < 0)
- {
- return false;
- }
- }
- return true;
-}
-
-
-bool VCMDecodeThread(void* obj)
-{
- SharedRTPState* state = static_cast<SharedRTPState*>(obj);
- state->_vcm.Decode();
- return true;
-}
-
-bool TransportThread(void *obj)
-{
- SharedTransportState* state = static_cast<SharedTransportState*>(obj);
- state->_transport.TransportPackets();
- return true;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/test/mt_test_common.h b/trunk/src/modules/video_coding/main/test/mt_test_common.h
deleted file mode 100644
index 438f0be..0000000
--- a/trunk/src/modules/video_coding/main/test/mt_test_common.h
+++ /dev/null
@@ -1,88 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Common multi-thread functionality across video coding module tests
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_MT_TEST_COMMON_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TEST_MT_TEST_COMMON_H_
-
-#include "rtp_rtcp.h"
-#include "test_callbacks.h"
-#include "test_util.h"
-#include "video_coding.h"
-
-namespace webrtc {
-
-class SendSharedState
-{
-public:
- SendSharedState(webrtc::VideoCodingModule& vcm, webrtc::RtpRtcp& rtp,
- CmdArgs args) :
- _vcm(vcm),
- _rtp(rtp),
- _args(args),
- _sourceFile(NULL),
- _frameCnt(0),
- _timestamp(0) {}
-
- webrtc::VideoCodingModule& _vcm;
- webrtc::RtpRtcp& _rtp;
- CmdArgs _args;
- FILE* _sourceFile;
- WebRtc_Word32 _frameCnt;
- WebRtc_Word32 _timestamp;
-};
-
-// MT implementation of the RTPSendCompleteCallback (Transport)
-class TransportCallback:public RTPSendCompleteCallback
-{
- public:
- // constructor input: (receive side) rtp module to send encoded data to
- TransportCallback(webrtc::RtpRtcp* rtp, TickTimeBase* clock,
- const char* filename = NULL);
- virtual ~TransportCallback();
- // Add packets to list
- // Incorporate network conditions - delay and packet loss
- // Actual transmission will occur on a separate thread
- int SendPacket(int channel, const void *data, int len);
- // Send to the receiver packets which are ready to be submitted
- int TransportPackets();
-};
-
-class SharedRTPState
-{
-public:
- SharedRTPState(webrtc::VideoCodingModule& vcm, webrtc::RtpRtcp& rtp) :
- _vcm(vcm),
- _rtp(rtp) {}
- webrtc::VideoCodingModule& _vcm;
- webrtc::RtpRtcp& _rtp;
-};
-
-
-class SharedTransportState
-{
-public:
- SharedTransportState(webrtc::RtpRtcp& rtp, TransportCallback& transport):
- _rtp(rtp),
- _transport(transport) {}
- webrtc::RtpRtcp& _rtp;
- TransportCallback& _transport;
-};
-
-bool VCMProcessingThread(void* obj);
-bool VCMDecodeThread(void* obj);
-bool TransportThread(void *obj);
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_MT_TEST_COMMON_H_
diff --git a/trunk/src/modules/video_coding/main/test/normal_test.cc b/trunk/src/modules/video_coding/main/test/normal_test.cc
deleted file mode 100644
index 923d649..0000000
--- a/trunk/src/modules/video_coding/main/test/normal_test.cc
+++ /dev/null
@@ -1,402 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "normal_test.h"
-
-#include <assert.h>
-#include <iostream>
-#include <sstream>
-#include <time.h>
-
-#include "../source/event.h"
-#include "common_types.h"
-#include "modules/video_coding/main/source/mock/fake_tick_time.h"
-#include "test_callbacks.h"
-#include "test_macros.h"
-#include "test_util.h"
-#include "trace.h"
-#include "testsupport/metrics/video_metrics.h"
-
-using namespace webrtc;
-
-int NormalTest::RunTest(CmdArgs& args)
-{
-#if defined(EVENT_DEBUG)
- printf("SIMULATION TIME\n");
- FakeTickTime clock(0);
-#else
- printf("REAL-TIME\n");
- TickTimeBase clock;
-#endif
- Trace::CreateTrace();
- Trace::SetTraceFile(
- (test::OutputPath() + "VCMNormalTestTrace.txt").c_str());
- Trace::SetLevelFilter(webrtc::kTraceAll);
- VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
- NormalTest VCMNTest(vcm, &clock);
- VCMNTest.Perform(args);
- VideoCodingModule::Destroy(vcm);
- Trace::ReturnTrace();
- return 0;
-}
-
-////////////////
-// Callback Implementation
-//////////////
-
-VCMNTEncodeCompleteCallback::VCMNTEncodeCompleteCallback(FILE* encodedFile,
- NormalTest& test):
- _encodedFile(encodedFile),
- _encodedBytes(0),
- _skipCnt(0),
- _VCMReceiver(NULL),
- _seqNo(0),
- _test(test)
-{
- //
-}
-VCMNTEncodeCompleteCallback::~VCMNTEncodeCompleteCallback()
-{
-}
-
-void
-VCMNTEncodeCompleteCallback::RegisterTransportCallback(VCMPacketizationCallback* transport)
-{
-}
-
-WebRtc_Word32
-VCMNTEncodeCompleteCallback::SendData(
- const FrameType frameType,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader& /*fragmentationHeader*/,
- const webrtc::RTPVideoHeader* videoHdr)
-
-{
- // will call the VCMReceiver input packet
- _frameType = frameType;
- // writing encodedData into file
- fwrite(payloadData, 1, payloadSize, _encodedFile);
- WebRtcRTPHeader rtpInfo;
- rtpInfo.header.markerBit = true;
- rtpInfo.type.Video.width = 0;
- rtpInfo.type.Video.height = 0;
- switch (_test.VideoType())
- {
- case kVideoCodecVP8:
- rtpInfo.type.Video.codec = kRTPVideoVP8;
- rtpInfo.type.Video.codecHeader.VP8.InitRTPVideoHeaderVP8();
- rtpInfo.type.Video.codecHeader.VP8.nonReference =
- videoHdr->codecHeader.VP8.nonReference;
- rtpInfo.type.Video.codecHeader.VP8.pictureId =
- videoHdr->codecHeader.VP8.pictureId;
- break;
- case kVideoCodecI420:
- rtpInfo.type.Video.codec = kRTPVideoI420;
- break;
- default:
- assert(false);
- return -1;
- }
- rtpInfo.header.payloadType = payloadType;
- rtpInfo.header.sequenceNumber = _seqNo++;
- rtpInfo.header.ssrc = 0;
- rtpInfo.header.timestamp = timeStamp;
- rtpInfo.frameType = frameType;
- rtpInfo.type.Video.isFirstPacket = true;
- // Size should also be received from that table, since the payload type
- // defines the size.
-
- _encodedBytes += payloadSize;
- if (payloadSize < 20)
- {
- _skipCnt++;
- }
- _VCMReceiver->IncomingPacket(payloadData, payloadSize, rtpInfo);
- return 0;
-}
-void
-VCMNTEncodeCompleteCallback::RegisterReceiverVCM(VideoCodingModule *vcm)
-{
- _VCMReceiver = vcm;
- return;
-}
- WebRtc_Word32
-VCMNTEncodeCompleteCallback::EncodedBytes()
-{
- return _encodedBytes;
-}
-
-WebRtc_UWord32
-VCMNTEncodeCompleteCallback::SkipCnt()
-{
- return _skipCnt;
-}
-
-// Decoded Frame Callback Implementation
-VCMNTDecodeCompleCallback::~VCMNTDecodeCompleCallback()
-{
- if (_decodedFile)
- fclose(_decodedFile);
-}
- WebRtc_Word32
-VCMNTDecodeCompleCallback::FrameToRender(webrtc::VideoFrame& videoFrame)
-{
- if (videoFrame.Width() != _currentWidth ||
- videoFrame.Height() != _currentHeight)
- {
- _currentWidth = videoFrame.Width();
- _currentHeight = videoFrame.Height();
- if (_decodedFile != NULL)
- {
- fclose(_decodedFile);
- _decodedFile = NULL;
- }
- _decodedFile = fopen(_outname.c_str(), "wb");
- }
- fwrite(videoFrame.Buffer(), 1, videoFrame.Length(), _decodedFile);
- _decodedBytes+= videoFrame.Length();
- return VCM_OK;
-}
-
- WebRtc_Word32
-VCMNTDecodeCompleCallback::DecodedBytes()
-{
- return _decodedBytes;
-}
-
- //VCM Normal Test Class implementation
-
-NormalTest::NormalTest(VideoCodingModule* vcm, TickTimeBase* clock)
-:
-_clock(clock),
-_vcm(vcm),
-_sumEncBytes(0),
-_timeStamp(0),
-_totalEncodeTime(0),
-_totalDecodeTime(0),
-_decodeCompleteTime(0),
-_encodeCompleteTime(0),
-_totalEncodePipeTime(0),
-_totalDecodePipeTime(0),
-_frameCnt(0),
-_encFrameCnt(0),
-_decFrameCnt(0)
-{
- //
-}
-
-NormalTest::~NormalTest()
-{
- //
-}
-void
-NormalTest::Setup(CmdArgs& args)
-{
- _inname = args.inputFile;
- _encodedName = test::OutputPath() + "encoded_normaltest.yuv";
- _width = args.width;
- _height = args.height;
- _frameRate = args.frameRate;
- _bitRate = args.bitRate;
- if (args.outputFile == "")
- {
- std::ostringstream filename;
- filename << test::OutputPath() << "NormalTest_" <<
- _width << "x" << _height << "_" << _frameRate << "Hz_P420.yuv";
- _outname = filename.str();
- }
- else
- {
- _outname = args.outputFile;
- }
- _lengthSourceFrame = 3*_width*_height/2;
- _videoType = args.codecType;
-
- if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
- {
- printf("Cannot read file %s.\n", _inname.c_str());
- exit(1);
- }
- if ((_encodedFile = fopen(_encodedName.c_str(), "wb")) == NULL)
- {
- printf("Cannot write encoded file.\n");
- exit(1);
- }
-
- _log.open((test::OutputPath() + "TestLog.txt").c_str(),
- std::fstream::out | std::fstream::app);
- return;
-}
-
-WebRtc_Word32
-NormalTest::Perform(CmdArgs& args)
-{
- Setup(args);
- EventWrapper* waitEvent = EventWrapper::Create();
- VideoCodec _sendCodec;//, _receiveCodec; // tmp - sendCodecd used as receive codec
- _vcm->InitializeReceiver();
- _vcm->InitializeSender();
- TEST(VideoCodingModule::Codec(_videoType, &_sendCodec) == VCM_OK);
- _sendCodec.startBitrate = (int)_bitRate; // should be later on changed via the API
- _sendCodec.width = static_cast<WebRtc_UWord16>(_width);
- _sendCodec.height = static_cast<WebRtc_UWord16>(_height);
- _sendCodec.maxFramerate = _frameRate;
- TEST(_vcm->RegisterSendCodec(&_sendCodec, 4, 1400) == VCM_OK);// will also set and init the desired codec
- // register a decoder (same codec for decoder and encoder )
- TEST(_vcm->RegisterReceiveCodec(&_sendCodec, 1) == VCM_OK);
- /* Callback Settings */
- VCMNTDecodeCompleCallback _decodeCallback(_outname);
- _vcm->RegisterReceiveCallback(&_decodeCallback);
- VCMNTEncodeCompleteCallback _encodeCompleteCallback(_encodedFile, *this);
- _vcm->RegisterTransportCallback(&_encodeCompleteCallback);
- // encode and decode with the same vcm
- _encodeCompleteCallback.RegisterReceiverVCM(_vcm);
- ///////////////////////
- /// Start Test
- ///////////////////////
- VideoFrame sourceFrame;
- sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
- WebRtc_UWord8* tmpBuffer = new WebRtc_UWord8[_lengthSourceFrame];
- double startTime = clock()/(double)CLOCKS_PER_SEC;
- _vcm->SetChannelParameters((WebRtc_UWord32)_bitRate, 0, 0);
-
- SendStatsTest sendStats;
- sendStats.SetTargetFrameRate(static_cast<WebRtc_UWord32>(_frameRate));
- _vcm->RegisterSendStatisticsCallback(&sendStats);
-
- while (feof(_sourceFile) == 0)
- {
-#if !defined(EVENT_DEBUG)
- WebRtc_Word64 processStartTime = _clock->MillisecondTimestamp();
-#endif
- TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0 ||
- feof(_sourceFile));
- _frameCnt++;
- sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
- sourceFrame.SetHeight(_height);
- sourceFrame.SetWidth(_width);
- _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(_sendCodec.maxFramerate));
- sourceFrame.SetTimeStamp(_timeStamp);
- _encodeTimes[int(sourceFrame.TimeStamp())] = clock()/(double)CLOCKS_PER_SEC;
- WebRtc_Word32 ret = _vcm->AddVideoFrame(sourceFrame);
- double encodeTime = clock()/(double)CLOCKS_PER_SEC - _encodeTimes[int(sourceFrame.TimeStamp())];
- _totalEncodeTime += encodeTime;
- if (ret < 0)
- {
- printf("Error in AddFrame: %d\n", ret);
- //exit(1);
- }
- _decodeTimes[int(sourceFrame.TimeStamp())] = clock()/(double)CLOCKS_PER_SEC; // same timestamp value for encode and decode
- ret = _vcm->Decode();
- _totalDecodeTime += clock()/(double)CLOCKS_PER_SEC - _decodeTimes[int(sourceFrame.TimeStamp())];
- if (ret < 0)
- {
- printf("Error in Decode: %d\n", ret);
- //exit(1);
- }
- if (_vcm->TimeUntilNextProcess() <= 0)
- {
- _vcm->Process();
- }
- WebRtc_UWord32 framePeriod = static_cast<WebRtc_UWord32>(1000.0f/static_cast<float>(_sendCodec.maxFramerate) + 0.5f);
-#if defined(EVENT_DEBUG)
- static_cast<FakeTickTime*>(_clock)->IncrementDebugClock(framePeriod);
-#else
- WebRtc_Word64 timeSpent = _clock->MillisecondTimestamp() - processStartTime;
- if (timeSpent < framePeriod)
- {
- waitEvent->Wait(framePeriod - timeSpent);
- }
-#endif
- }
- double endTime = clock()/(double)CLOCKS_PER_SEC;
- _testTotalTime = endTime - startTime;
- _sumEncBytes = _encodeCompleteCallback.EncodedBytes();
-
- delete [] tmpBuffer;
- delete waitEvent;
- Teardown();
- Print();
- return 0;
-}
-
-void
-NormalTest::FrameEncoded(WebRtc_UWord32 timeStamp)
-{
- _encodeCompleteTime = clock()/(double)CLOCKS_PER_SEC;
- _encFrameCnt++;
- _totalEncodePipeTime += _encodeCompleteTime - _encodeTimes[int(timeStamp)];
-
-}
-
-void
-NormalTest::FrameDecoded(WebRtc_UWord32 timeStamp)
-{
- _decodeCompleteTime = clock()/(double)CLOCKS_PER_SEC;
- _decFrameCnt++;
- _totalDecodePipeTime += _decodeCompleteTime - _decodeTimes[timeStamp];
-}
-
-void
-NormalTest::Print()
-{
- std::cout << "Normal Test Completed!" << std::endl;
- (_log) << "Normal Test Completed!" << std::endl;
- (_log) << "Input file: " << _inname << std::endl;
- (_log) << "Output file: " << _outname << std::endl;
- (_log) << "Total run time: " << _testTotalTime << std::endl;
- printf("Total run time: %f s \n", _testTotalTime);
- double ActualBitRate = 8.0 *( _sumEncBytes / (_frameCnt / _frameRate));
- double actualBitRate = ActualBitRate / 1000.0;
- double avgEncTime = _totalEncodeTime / _frameCnt;
- double avgDecTime = _totalDecodeTime / _frameCnt;
- webrtc::test::QualityMetricsResult psnr, ssim;
- I420PSNRFromFiles(_inname.c_str(), _outname.c_str(), _width, _height,
- &psnr);
- I420SSIMFromFiles(_inname.c_str(), _outname.c_str(), _width, _height,
- &ssim);
- printf("Actual bitrate: %f kbps\n", actualBitRate);
- printf("Target bitrate: %f kbps\n", _bitRate);
- ( _log) << "Actual bitrate: " << actualBitRate<< " kbps\tTarget: " << _bitRate << " kbps" << std::endl;
- printf("Average encode time: %f s\n", avgEncTime);
- ( _log) << "Average encode time: " << avgEncTime << " s" << std::endl;
- printf("Average decode time: %f s\n", avgDecTime);
- ( _log) << "Average decode time: " << avgDecTime << " s" << std::endl;
- printf("PSNR: %f \n", psnr.average);
- ( _log) << "PSNR: " << psnr.average << std::endl;
- printf("SSIM: %f \n", ssim.average);
- ( _log) << "SSIM: " << ssim.average << std::endl;
- (_log) << std::endl;
-
- printf("\nVCM Normal Test: \n\n%i tests completed\n", vcmMacrosTests);
- if (vcmMacrosErrors > 0)
- {
- printf("%i FAILED\n\n", vcmMacrosErrors);
- }
- else
- {
- printf("ALL PASSED\n\n");
- }
-}
-void
-NormalTest::Teardown()
-{
- //_log.close();
- fclose(_sourceFile);
- fclose(_encodedFile);
- return;
-}
-
-
-
-
diff --git a/trunk/src/modules/video_coding/main/test/normal_test.h b/trunk/src/modules/video_coding/main/test/normal_test.h
deleted file mode 100644
index 6f75dfb..0000000
--- a/trunk/src/modules/video_coding/main/test/normal_test.h
+++ /dev/null
@@ -1,143 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_NORMAL_TEST_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TEST_NORMAL_TEST_H_
-
-#include "video_coding.h"
-#include "test_util.h"
-
-#include <map>
-
-class NormalTest;
-
-//Send Side - Packetization callback - will create and send a packet to the VCMReceiver
-class VCMNTEncodeCompleteCallback : public webrtc::VCMPacketizationCallback
-{
-public:
- // constructor input: file in which encoded data will be written
- VCMNTEncodeCompleteCallback(FILE* encodedFile, NormalTest& test);
- virtual ~VCMNTEncodeCompleteCallback();
- // Register transport callback
- void RegisterTransportCallback(webrtc::VCMPacketizationCallback* transport);
- // process encoded data received from the encoder, pass stream to the VCMReceiver module
- WebRtc_Word32 SendData(const webrtc::FrameType frameType,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const webrtc::RTPFragmentationHeader& fragmentationHeader,
- const webrtc::RTPVideoHeader* videoHdr);
-
- // Register exisitng VCM. Currently - encode and decode with the same vcm module.
- void RegisterReceiverVCM(webrtc::VideoCodingModule *vcm);
- // Return sum of encoded data (all frames in the sequence)
- WebRtc_Word32 EncodedBytes();
- // return number of encoder-skipped frames
- WebRtc_UWord32 SkipCnt();;
- // conversion function for payload type (needed for the callback function)
-// RTPVideoVideoCodecTypes ConvertPayloadType(WebRtc_UWord8 payloadType);
-
-private:
- FILE* _encodedFile;
- WebRtc_UWord32 _encodedBytes;
- WebRtc_UWord32 _skipCnt;
- webrtc::VideoCodingModule* _VCMReceiver;
- webrtc::FrameType _frameType;
- WebRtc_UWord8* _payloadData; // max payload size??
- WebRtc_UWord16 _seqNo;
- NormalTest& _test;
-}; // end of VCMEncodeCompleteCallback
-
-class VCMNTDecodeCompleCallback: public webrtc::VCMReceiveCallback
-{
-public:
- VCMNTDecodeCompleCallback(std::string outname): // or should it get a name?
- _decodedFile(NULL),
- _outname(outname),
- _decodedBytes(0),
- _currentWidth(0),
- _currentHeight(0) {}
- virtual ~VCMNTDecodeCompleCallback();
- void SetUserReceiveCallback(webrtc::VCMReceiveCallback* receiveCallback);
- // will write decoded frame into file
- WebRtc_Word32 FrameToRender(webrtc::VideoFrame& videoFrame);
- WebRtc_Word32 DecodedBytes();
-private:
- FILE* _decodedFile;
- std::string _outname;
- WebRtc_UWord32 _decodedBytes;
- WebRtc_UWord32 _currentWidth;
- WebRtc_UWord32 _currentHeight;
-
-}; // end of VCMDecodeCompleCallback class
-
-
-class NormalTest
-{
-public:
- NormalTest(webrtc::VideoCodingModule* vcm,
- webrtc::TickTimeBase* clock);
- ~NormalTest();
- static int RunTest(CmdArgs& args);
- WebRtc_Word32 Perform(CmdArgs& args);
- // option:: turn into private and call from perform
- WebRtc_UWord32 Width() const { return _width; };
- WebRtc_UWord32 Height() const { return _height; };
- webrtc::VideoCodecType VideoType() const { return _videoType; };
-
-
-protected:
- // test setup - open files, general initializations
- void Setup(CmdArgs& args);
- // close open files, delete used memory
- void Teardown();
- // print results to std output and to log file
- void Print();
- // calculating pipeline delay, and encoding time
- void FrameEncoded(WebRtc_UWord32 timeStamp);
- // calculating pipeline delay, and decoding time
- void FrameDecoded(WebRtc_UWord32 timeStamp);
-
- webrtc::TickTimeBase* _clock;
- webrtc::VideoCodingModule* _vcm;
- webrtc::VideoCodec _sendCodec;
- webrtc::VideoCodec _receiveCodec;
- std::string _inname;
- std::string _outname;
- std::string _encodedName;
- WebRtc_Word32 _sumEncBytes;
- FILE* _sourceFile;
- FILE* _decodedFile;
- FILE* _encodedFile;
- std::fstream _log;
- WebRtc_UWord32 _width;
- WebRtc_UWord32 _height;
- float _frameRate;
- float _bitRate;
- WebRtc_UWord32 _lengthSourceFrame;
- WebRtc_UWord32 _timeStamp;
- webrtc::VideoCodecType _videoType;
- double _totalEncodeTime;
- double _totalDecodeTime;
- double _decodeCompleteTime;
- double _encodeCompleteTime;
- double _totalEncodePipeTime;
- double _totalDecodePipeTime;
- double _testTotalTime;
- std::map<int, double> _encodeTimes;
- std::map<int, double> _decodeTimes;
- WebRtc_Word32 _frameCnt;
- WebRtc_Word32 _encFrameCnt;
- WebRtc_Word32 _decFrameCnt;
-
-}; // end of VCMNormalTestClass
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_NORMAL_TEST_H_
diff --git a/trunk/src/modules/video_coding/main/test/plotJitterEstimate.m b/trunk/src/modules/video_coding/main/test/plotJitterEstimate.m
deleted file mode 100644
index d6185f5..0000000
--- a/trunk/src/modules/video_coding/main/test/plotJitterEstimate.m
+++ /dev/null
@@ -1,52 +0,0 @@
-function plotJitterEstimate(filename)
-
-[timestamps, framedata, slopes, randJitters, framestats, timetable, filtjitter, rtt, rttStatsVec] = jitterBufferTraceParser(filename);
-
-x = 1:size(framestats, 1);
-%figure(2);
-subfigure(3, 2, 1);
-hold on;
-plot(x, slopes(x, 1).*(framestats(x, 1) - framestats(x, 2)) + 3*sqrt(randJitters(x,2)), 'b'); title('Estimate ms');
-plot(x, filtjitter, 'r');
-plot(x, slopes(x, 1).*(framestats(x, 1) - framestats(x, 2)), 'g');
-subfigure(3, 2, 2);
-%subplot(211);
-plot(x, slopes(x, 1)); title('Line slope');
-%subplot(212);
-%plot(x, slopes(x, 2)); title('Line offset');
-subfigure(3, 2, 3); hold on;
-plot(x, framestats); plot(x, framedata(x, 1)); title('frame size and average frame size');
-subfigure(3, 2, 4);
-plot(x, framedata(x, 2)); title('Delay');
-subfigure(3, 2, 5);
-hold on;
-plot(x, randJitters(x,1),'r');
-plot(x, randJitters(x,2)); title('Random jitter');
-
-subfigure(3, 2, 6);
-delays = framedata(:,2);
-dL = [0; framedata(2:end, 1) - framedata(1:end-1, 1)];
-hold on;
-plot(dL, delays, '.');
-s = [min(dL) max(dL)];
-plot(s, slopes(end, 1)*s + slopes(end, 2), 'g');
-plot(s, slopes(end, 1)*s + slopes(end, 2) + 3*sqrt(randJitters(end,2)), 'r');
-plot(s, slopes(end, 1)*s + slopes(end, 2) - 3*sqrt(randJitters(end,2)), 'r');
-title('theta(1)*x+theta(2), (dT-dTS)/dL');
-if sum(size(rttStatsVec)) > 0
- figure; hold on;
- rttNstdDevsDrift = 3.5;
- rttNstdDevsJump = 2.5;
- rttSamples = rttStatsVec(:, 1);
- rttAvgs = rttStatsVec(:, 2);
- rttStdDevs = sqrt(rttStatsVec(:, 3));
- rttMax = rttStatsVec(:, 4);
- plot(rttSamples, 'ko-');
- plot(rttAvgs, 'g');
- plot(rttAvgs + rttNstdDevsDrift*rttStdDevs, 'b--');
- plot(rttAvgs + rttNstdDevsJump*rttStdDevs, 'b');
- plot(rttAvgs - rttNstdDevsJump*rttStdDevs, 'b');
- plot(rttMax, 'r');
- %plot(driftRestarts*max(maxRtts), '.');
- %plot(jumpRestarts*max(maxRtts), '.');
-end
\ No newline at end of file
diff --git a/trunk/src/modules/video_coding/main/test/plotReceiveTrace.m b/trunk/src/modules/video_coding/main/test/plotReceiveTrace.m
deleted file mode 100644
index 4d262aa..0000000
--- a/trunk/src/modules/video_coding/main/test/plotReceiveTrace.m
+++ /dev/null
@@ -1,213 +0,0 @@
-function [t, TS] = plotReceiveTrace(filename, flat)
-fid=fopen(filename);
-%DEBUG ; ( 8:32:33:375 | 0) VIDEO:1 ; 5260; First packet of frame 1869537938
-%DEBUG ; ( 8:32:33:375 | 0) VIDEO CODING:1 ; 5260; Decoding timestamp 1869534934
-%DEBUG ; ( 8:32:33:375 | 0) VIDEO:1 ; 5260; Render frame 1869534934 at 20772610
-%DEBUG ; ( 8:32:33:375 | 0) VIDEO CODING:-1 ; 5260; Frame decoded: timeStamp=1870511259 decTime=0 maxDecTime=0, at 19965
-%DEBUG ; ( 7:59:42:500 | 0) VIDEO:-1 ; 2500; Received complete frame timestamp 1870514263 frame type 1 frame size 7862 at time 19965, jitter estimate was 130
-%DEBUG ; ( 8: 5:51:774 | 0) VIDEO:-1 ; 3968; ExtrapolateLocalTime(1870967878)=24971 ms
-
-if nargin == 1
- flat = 0;
-end
-line = fgetl(fid);
-estimatedArrivalTime = [];
-packetTime = [];
-firstPacketTime = [];
-decodeTime = [];
-decodeCompleteTime = [];
-renderTime = [];
-completeTime = [];
-while ischar(line)%line ~= -1
- if length(line) == 0
- line = fgetl(fid);
- continue;
- end
- % Parse the trace line header
- [tempres, count] = sscanf(line, 'DEBUG ; (%u:%u:%u:%u |%*lu)%13c:');
- if count < 5
- line = fgetl(fid);
- continue;
- end
- hr=tempres(1);
- mn=tempres(2);
- sec=tempres(3);
- ms=tempres(4);
- timeInMs=hr*60*60*1000 + mn*60*1000 + sec*1000 + ms;
- label = tempres(5:end);
- I = find(label ~= 32);
- label = label(I(1):end); % remove white spaces
- if ~strncmp(char(label), 'VIDEO', 5) & ~strncmp(char(label), 'VIDEO CODING', 12)
- line = fgetl(fid);
- continue;
- end
- message = line(72:end);
-
- % Parse message
- [p, count] = sscanf(message, 'ExtrapolateLocalTime(%lu)=%lu ms');
- if count == 2
- estimatedArrivalTime = [estimatedArrivalTime; p'];
- line = fgetl(fid);
- continue;
- end
-
- [p, count] = sscanf(message, 'Packet seqNo %u of frame %lu at %lu');
- if count == 3
- packetTime = [packetTime; p'];
- line = fgetl(fid);
- continue;
- end
-
- [p, count] = sscanf(message, 'First packet of frame %lu at %lu');
- if count == 2
- firstPacketTime = [firstPacketTime; p'];
- line = fgetl(fid);
- continue;
- end
-
- [p, count] = sscanf(message, 'Decoding timestamp %lu at %lu');
- if count == 2
- decodeTime = [decodeTime; p'];
- line = fgetl(fid);
- continue;
- end
-
- [p, count] = sscanf(message, 'Render frame %lu at %lu. Render delay %lu, required delay %lu, max decode time %lu, min total delay %lu');
- if count == 6
- renderTime = [renderTime; p'];
- line = fgetl(fid);
- continue;
- end
-
- [p, count] = sscanf(message, 'Frame decoded: timeStamp=%lu decTime=%d maxDecTime=%lu, at %lu');
- if count == 4
- decodeCompleteTime = [decodeCompleteTime; p'];
- line = fgetl(fid);
- continue;
- end
-
- [p, count] = sscanf(message, 'Received complete frame timestamp %lu frame type %u frame size %*u at time %lu, jitter estimate was %lu');
- if count == 4
- completeTime = [completeTime; p'];
- line = fgetl(fid);
- continue;
- end
-
- line = fgetl(fid);
-end
-fclose(fid);
-
-t = completeTime(:,3);
-TS = completeTime(:,1);
-
-figure;
-subplot(211);
-hold on;
-slope = 0;
-
-if sum(size(packetTime)) > 0
- % Plot the time when each packet arrives
- firstTimeStamp = packetTime(1,2);
- x = (packetTime(:,2) - firstTimeStamp)/90;
- if flat
- slope = x;
- end
- firstTime = packetTime(1,3);
- plot(x, packetTime(:,3) - firstTime - slope, 'b.');
-else
- % Plot the time when the first packet of a frame arrives
- firstTimeStamp = firstPacketTime(1,1);
- x = (firstPacketTime(:,1) - firstTimeStamp)/90;
- if flat
- slope = x;
- end
- firstTime = firstPacketTime(1,2);
- plot(x, firstPacketTime(:,2) - firstTime - slope, 'b.');
-end
-
-% Plot the frame complete time
-if prod(size(completeTime)) > 0
- x = (completeTime(:,1) - firstTimeStamp)/90;
- if flat
- slope = x;
- end
- plot(x, completeTime(:,3) - firstTime - slope, 'ks');
-end
-
-% Plot the time the decode starts
-if prod(size(decodeTime)) > 0
- x = (decodeTime(:,1) - firstTimeStamp)/90;
- if flat
- slope = x;
- end
- plot(x, decodeTime(:,2) - firstTime - slope, 'r.');
-end
-
-% Plot the decode complete time
-if prod(size(decodeCompleteTime)) > 0
- x = (decodeCompleteTime(:,1) - firstTimeStamp)/90;
- if flat
- slope = x;
- end
- plot(x, decodeCompleteTime(:,4) - firstTime - slope, 'g.');
-end
-
-if prod(size(renderTime)) > 0
- % Plot the wanted render time in ms
- x = (renderTime(:,1) - firstTimeStamp)/90;
- if flat
- slope = x;
- end
- plot(x, renderTime(:,2) - firstTime - slope, 'c-');
-
- % Plot the render time if there were no render delay or decoding delay.
- x = (renderTime(:,1) - firstTimeStamp)/90;
- if flat
- slope = x;
- end
- plot(x, renderTime(:,2) - firstTime - slope - renderTime(:, 3) - renderTime(:, 5), 'c--');
-
- % Plot the render time if there were no render delay.
- x = (renderTime(:,1) - firstTimeStamp)/90;
- if flat
- slope = x;
- end
- plot(x, renderTime(:,2) - firstTime - slope - renderTime(:, 3) - renderTime(:, 5), 'b-');
-end
-
-%plot(x, 90*x, 'r-');
-
-xlabel('RTP timestamp (in ms)');
-ylabel('Time (ms)');
-legend('Packet arrives', 'Frame complete', 'Decode', 'Decode complete', 'Time to render', 'Only jitter', 'Must decode');
-
-% subplot(312);
-% hold on;
-% completeTs = completeTime(:, 1);
-% arrivalTs = estimatedArrivalTime(:, 1);
-% [c, completeIdx, arrivalIdx] = intersect(completeTs, arrivalTs);
-% %plot(completeTs(completeIdx), completeTime(completeIdx, 3) - estimatedArrivalTime(arrivalIdx, 2));
-% timeUntilComplete = completeTime(completeIdx, 3) - estimatedArrivalTime(arrivalIdx, 2);
-% devFromAvgCompleteTime = timeUntilComplete - mean(timeUntilComplete);
-% plot(completeTs(completeIdx) - completeTs(completeIdx(1)), devFromAvgCompleteTime);
-% plot(completeTime(:, 1) - completeTime(1, 1), completeTime(:, 4), 'r');
-% plot(decodeCompleteTime(:, 1) - decodeCompleteTime(1, 1), decodeCompleteTime(:, 2), 'g');
-% plot(decodeCompleteTime(:, 1) - decodeCompleteTime(1, 1), decodeCompleteTime(:, 3), 'k');
-% xlabel('RTP timestamp');
-% ylabel('Time (ms)');
-% legend('Complete time - Estimated arrival time', 'Desired jitter buffer level', 'Actual decode time', 'Max decode time', 0);
-
-if prod(size(renderTime)) > 0
- subplot(212);
- hold on;
- firstTime = renderTime(1, 1);
- targetDelay = max(renderTime(:, 3) + renderTime(:, 4) + renderTime(:, 5), renderTime(:, 6));
- plot(renderTime(:, 1) - firstTime, renderTime(:, 3), 'r-');
- plot(renderTime(:, 1) - firstTime, renderTime(:, 4), 'b-');
- plot(renderTime(:, 1) - firstTime, renderTime(:, 5), 'g-');
- plot(renderTime(:, 1) - firstTime, renderTime(:, 6), 'k-');
- plot(renderTime(:, 1) - firstTime, targetDelay, 'c-');
- xlabel('RTP timestamp');
- ylabel('Time (ms)');
- legend('Render delay', 'Jitter delay', 'Decode delay', 'Extra delay', 'Min total delay');
-end
\ No newline at end of file
diff --git a/trunk/src/modules/video_coding/main/test/plotTimingTest.m b/trunk/src/modules/video_coding/main/test/plotTimingTest.m
deleted file mode 100644
index 52a6f30..0000000
--- a/trunk/src/modules/video_coding/main/test/plotTimingTest.m
+++ /dev/null
@@ -1,62 +0,0 @@
-function plotTimingTest(filename)
-fid=fopen(filename);
-
-%DEBUG ; ( 9:53:33:859 | 0) VIDEO:-1 ; 7132; Stochastic test 1
-%DEBUG ; ( 9:53:33:859 | 0) VIDEO CODING:-1 ; 7132; Frame decoded: timeStamp=3000 decTime=10 at 10012
-%DEBUG ; ( 9:53:33:859 | 0) VIDEO:-1 ; 7132; timeStamp=3000 clock=10037 maxWaitTime=0
-%DEBUG ; ( 9:53:33:859 | 0) VIDEO:-1 ; 7132; timeStampMs=33 renderTime=54
-line = fgetl(fid);
-decTime = [];
-waitTime = [];
-renderTime = [];
-foundStart = 0;
-testName = 'Stochastic test 1';
-while ischar(line)
- if length(line) == 0
- line = fgetl(fid);
- continue;
- end
- lineOrig = line;
- line = line(72:end);
- if ~foundStart
- if strncmp(line, testName, length(testName))
- foundStart = 1;
- end
- line = fgetl(fid);
- continue;
- end
- [p, count] = sscanf(line, 'Frame decoded: timeStamp=%lu decTime=%d maxDecTime=%d, at %lu');
- if count == 4
- decTime = [decTime; p'];
- line = fgetl(fid);
- continue;
- end
- [p, count] = sscanf(line, 'timeStamp=%u clock=%u maxWaitTime=%u');
- if count == 3
- waitTime = [waitTime; p'];
- line = fgetl(fid);
- continue;
- end
- [p, count] = sscanf(line, 'timeStamp=%u renderTime=%u');
- if count == 2
- renderTime = [renderTime; p'];
- line = fgetl(fid);
- continue;
- end
- line = fgetl(fid);
-end
-fclose(fid);
-
-% Compensate for wrap arounds and start counting from zero.
-timeStamps = waitTime(:, 1);
-tsDiff = diff(timeStamps);
-wrapIdx = find(tsDiff < 0);
-timeStamps(wrapIdx+1:end) = hex2dec('ffffffff') + timeStamps(wrapIdx+1:end);
-timeStamps = timeStamps - timeStamps(1);
-
-figure;
-hold on;
-plot(timeStamps, decTime(:, 2), 'r');
-plot(timeStamps, waitTime(:, 3), 'g');
-plot(timeStamps(2:end), diff(renderTime(:, 2)), 'b');
-legend('Decode time', 'Max wait time', 'Render time diff');
\ No newline at end of file
diff --git a/trunk/src/modules/video_coding/main/test/quality_modes_test.cc b/trunk/src/modules/video_coding/main/test/quality_modes_test.cc
deleted file mode 100644
index e7da691..0000000
--- a/trunk/src/modules/video_coding/main/test/quality_modes_test.cc
+++ /dev/null
@@ -1,485 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "quality_modes_test.h"
-
-#include <iostream>
-#include <string>
-#include <time.h>
-
-#include "../source/event.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
-#include "test_callbacks.h"
-#include "test_macros.h"
-#include "testsupport/metrics/video_metrics.h"
-#include "common_video/libyuv/include/libyuv.h"
-
-using namespace webrtc;
-
-int qualityModeTest()
-{
- // Don't run this test with debug events.
-#if defined(EVENT_DEBUG)
- return -1;
-#endif
- TickTimeBase clock;
- VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
- QualityModesTest QMTest(vcm, &clock);
- QMTest.Perform();
- VideoCodingModule::Destroy(vcm);
- return 0;
-}
-
-
-QualityModesTest::QualityModesTest(VideoCodingModule* vcm,
- TickTimeBase* clock):
-NormalTest(vcm, clock),
-_vpm()
-{
- //
-}
-
-
-QualityModesTest::~QualityModesTest()
-{
- //
-}
-
-void
-QualityModesTest::Setup()
-{
-
-
- _inname= test::ProjectRootPath() + "resources/crew_30f_4CIF.yuv";
- _outname = test::OutputPath() + "out_qmtest.yuv";
- _encodedName = test::OutputPath() + "encoded_qmtest.yuv";
-
- //NATIVE/SOURCE VALUES
- _nativeWidth = 2*352;
- _nativeHeight = 2*288;
- _nativeFrameRate = 30;
-
-
- //TARGET/ENCODER VALUES
- _width = 2*352;
- _height = 2*288;
- _frameRate = 30;
- //
- _bitRate = 400;
-
- _flagSSIM = false;
-
- _lengthSourceFrame = 3*_nativeWidth*_nativeHeight/2;
-
- if ((_sourceFile = fopen(_inname.c_str(), "rb")) == NULL)
- {
- printf("Cannot read file %s.\n", _inname.c_str());
- exit(1);
- }
- if ((_encodedFile = fopen(_encodedName.c_str(), "wb")) == NULL)
- {
- printf("Cannot write encoded file.\n");
- exit(1);
- }
- if ((_decodedFile = fopen(_outname.c_str(), "wb")) == NULL)
- {
- printf("Cannot write file %s.\n", _outname.c_str());
- exit(1);
- }
-
- _log.open((test::OutputPath() + "TestLog.txt").c_str(),
- std::fstream::out | std::fstream::app);
- return;
-}
-
-
-void
-QualityModesTest::Print()
-{
- std::cout << "Quality Modes Test Completed!" << std::endl;
- (_log) << "Quality Modes Test Completed!" << std::endl;
- (_log) << "Input file: " << _inname << std::endl;
- (_log) << "Output file: " << _outname << std::endl;
- (_log) << "Total run time: " << _testTotalTime << std::endl;
- printf("Total run time: %f s \n", _testTotalTime);
- double ActualBitRate = 8.0 *( _sumEncBytes / (_frameCnt / _nativeFrameRate));
- double actualBitRate = ActualBitRate / 1000.0;
- double avgEncTime = _totalEncodeTime / _frameCnt;
- double avgDecTime = _totalDecodeTime / _frameCnt;
- webrtc::test::QualityMetricsResult psnr,ssim;
- I420PSNRFromFiles(_inname.c_str(), _outname.c_str(), _nativeWidth,
- _nativeHeight, &psnr);
- printf("Actual bitrate: %f kbps\n", actualBitRate);
- printf("Target bitrate: %f kbps\n", _bitRate);
- ( _log) << "Actual bitrate: " << actualBitRate<< " kbps\tTarget: " << _bitRate << " kbps" << std::endl;
- printf("Average encode time: %f s\n", avgEncTime);
- ( _log) << "Average encode time: " << avgEncTime << " s" << std::endl;
- printf("Average decode time: %f s\n", avgDecTime);
- ( _log) << "Average decode time: " << avgDecTime << " s" << std::endl;
- printf("PSNR: %f \n", psnr.average);
- printf("**Number of frames dropped in VPM***%d \n",_numFramesDroppedVPM);
- ( _log) << "PSNR: " << psnr.average << std::endl;
- if (_flagSSIM == 1)
- {
- printf("***computing SSIM***\n");
- I420SSIMFromFiles(_inname.c_str(), _outname.c_str(), _nativeWidth,
- _nativeHeight, &ssim);
- printf("SSIM: %f \n", ssim.average);
- }
- (_log) << std::endl;
-
- printf("\nVCM Qualit Modes Test: \n\n%i tests completed\n", vcmMacrosTests);
- if (vcmMacrosErrors > 0)
- {
- printf("%i FAILED\n\n", vcmMacrosErrors);
- }
- else
- {
- printf("ALL PASSED\n\n");
- }
-}
-void
-QualityModesTest::Teardown()
-{
- _log.close();
- fclose(_sourceFile);
- fclose(_decodedFile);
- fclose(_encodedFile);
- return;
-}
-
-
-WebRtc_Word32
-QualityModesTest::Perform()
-{
- Setup();
- // changing bit/frame rate during the test
- const float bitRateUpdate[] = {1000};
- const float frameRateUpdate[] = {30};
- const int updateFrameNum[] = {10000}; // frame numbers at which an update will occur
-
- WebRtc_UWord32 numChanges = sizeof(updateFrameNum)/sizeof(*updateFrameNum);
- WebRtc_UWord8 change = 0;// change counter
-
- _vpm = VideoProcessingModule::Create(1);
-
- EventWrapper* waitEvent = EventWrapper::Create();
- VideoCodec codec;//both send and receive
- _vcm->InitializeReceiver();
- _vcm->InitializeSender();
- WebRtc_Word32 NumberOfCodecs = _vcm->NumberOfCodecs();
- for (int i = 0; i < NumberOfCodecs; i++)
- {
- _vcm->Codec(i, &codec);
- if(strncmp(codec.plName,"VP8" , 5) == 0)
- {
- codec.startBitrate = (int)_bitRate;
- codec.maxFramerate = (WebRtc_UWord8) _frameRate;
- codec.width = (WebRtc_UWord16)_width;
- codec.height = (WebRtc_UWord16)_height;
- TEST(_vcm->RegisterSendCodec(&codec, 2, 1440) == VCM_OK);// will also set and init the desired codec
- i = NumberOfCodecs;
- }
- }
-
- // register a decoder (same codec for decoder and encoder )
- TEST(_vcm->RegisterReceiveCodec(&codec, 2) == VCM_OK);
- /* Callback Settings */
- VCMQMDecodeCompleCallback _decodeCallback(_decodedFile);
- _vcm->RegisterReceiveCallback(&_decodeCallback);
- VCMNTEncodeCompleteCallback _encodeCompleteCallback(_encodedFile, *this);
- _vcm->RegisterTransportCallback(&_encodeCompleteCallback);
- // encode and decode with the same vcm
- _encodeCompleteCallback.RegisterReceiverVCM(_vcm);
-
- //quality modes callback
- QMTestVideoSettingsCallback QMCallback;
- QMCallback.RegisterVCM(_vcm);
- QMCallback.RegisterVPM(_vpm);
- _vcm->RegisterVideoQMCallback(&QMCallback);
-
- ///////////////////////
- /// Start Test
- ///////////////////////
- _vpm->EnableTemporalDecimation(true);
- _vpm->EnableContentAnalysis(true);
- _vpm->SetInputFrameResampleMode(kFastRescaling);
-
- // disabling internal VCM frame dropper
- _vcm->EnableFrameDropper(false);
-
- VideoFrame sourceFrame;
- VideoFrame *decimatedFrame = NULL;
- sourceFrame.VerifyAndAllocate(_lengthSourceFrame);
- WebRtc_UWord8* tmpBuffer = new WebRtc_UWord8[_lengthSourceFrame];
- double startTime = clock()/(double)CLOCKS_PER_SEC;
- _vcm->SetChannelParameters((WebRtc_UWord32)_bitRate, 0, 0);
-
- SendStatsTest sendStats;
- sendStats.SetTargetFrameRate(static_cast<WebRtc_UWord32>(_frameRate));
- _vcm->RegisterSendStatisticsCallback(&sendStats);
-
- VideoContentMetrics* contentMetrics = NULL;
- // setting user frame rate
- _vpm->SetMaxFrameRate((WebRtc_UWord32)(_nativeFrameRate+ 0.5f));
- // for starters: keeping native values:
- _vpm->SetTargetResolution(_width, _height, (WebRtc_UWord32)(_frameRate+ 0.5f));
- _decodeCallback.SetOriginalFrameDimensions(_nativeWidth, _nativeHeight);
-
- //tmp - disabling VPM frame dropping
- _vpm->EnableTemporalDecimation(false);
-
-
- WebRtc_Word32 ret = 0;
- _numFramesDroppedVPM = 0;
-
- while (feof(_sourceFile)== 0)
- {
- TEST(fread(tmpBuffer, 1, _lengthSourceFrame, _sourceFile) > 0);
- _frameCnt++;
- sourceFrame.CopyFrame(_lengthSourceFrame, tmpBuffer);
- sourceFrame.SetHeight(_nativeHeight);
- sourceFrame.SetWidth(_nativeWidth);
-
- _timeStamp += (WebRtc_UWord32)(9e4 / static_cast<float>(codec.maxFramerate));
- sourceFrame.SetTimeStamp(_timeStamp);
-
- ret = _vpm->PreprocessFrame(&sourceFrame, &decimatedFrame);
- if (ret == 1)
- {
- printf("VD: frame drop %d \n",_frameCnt);
- _numFramesDroppedVPM += 1;
- continue; // frame drop
- }
- else if (ret < 0)
- {
- printf("Error in PreprocessFrame: %d\n", ret);
- //exit(1);
- }
- contentMetrics = _vpm->ContentMetrics();
- if (contentMetrics == NULL)
- {
- printf("error: contentMetrics = NULL\n");
- }
-
- // counting only encoding time
- _encodeTimes[int(sourceFrame.TimeStamp())] = clock()/(double)CLOCKS_PER_SEC;
-
- WebRtc_Word32 ret = _vcm->AddVideoFrame(*decimatedFrame, contentMetrics);
-
- _totalEncodeTime += clock()/(double)CLOCKS_PER_SEC - _encodeTimes[int(sourceFrame.TimeStamp())];
-
- if (ret < 0)
- {
- printf("Error in AddFrame: %d\n", ret);
- //exit(1);
- }
- _decodeTimes[int(sourceFrame.TimeStamp())] = clock()/(double)CLOCKS_PER_SEC; // same timestamp value for encode and decode
- ret = _vcm->Decode();
- _totalDecodeTime += clock()/(double)CLOCKS_PER_SEC - _decodeTimes[int(sourceFrame.TimeStamp())];
- if (ret < 0)
- {
- printf("Error in Decode: %d\n", ret);
- //exit(1);
- }
- if (_vcm->TimeUntilNextProcess() <= 0)
- {
- _vcm->Process();
- }
- // mimicking setTargetRates - update every 1 sec
- // this will trigger QMSelect
- if (_frameCnt%((int)_frameRate) == 0)
- {
- _vcm->SetChannelParameters((WebRtc_UWord32)_bitRate, 0, 1);
- waitEvent->Wait(33);
- }
- waitEvent->Wait(33);
- // check for bit rate update
- if (change < numChanges && _frameCnt == updateFrameNum[change])
- {
- _bitRate = bitRateUpdate[change];
- _frameRate = frameRateUpdate[change];
- codec.startBitrate = (int)_bitRate;
- codec.maxFramerate = (WebRtc_UWord8) _frameRate;
- TEST(_vcm->RegisterSendCodec(&codec, 2, 1440) == VCM_OK);// will also set and init the desired codec
- change++;
- }
- }
-
- double endTime = clock()/(double)CLOCKS_PER_SEC;
- _testTotalTime = endTime - startTime;
- _sumEncBytes = _encodeCompleteCallback.EncodedBytes();
-
- delete tmpBuffer;
- delete waitEvent;
- _vpm->Reset();
- Teardown();
- Print();
- VideoProcessingModule::Destroy(_vpm);
- return 0;
-}
-
-
-// implementing callback to be called from VCM to update VPM of frame rate and size
-QMTestVideoSettingsCallback::QMTestVideoSettingsCallback():
-_vpm(NULL),
-_vcm(NULL)
-{
- //
-}
-
-void
-QMTestVideoSettingsCallback::RegisterVPM(VideoProcessingModule *vpm)
-{
- _vpm = vpm;
-}
-void
-QMTestVideoSettingsCallback::RegisterVCM(VideoCodingModule *vcm)
-{
- _vcm = vcm;
-}
-
-bool
-QMTestVideoSettingsCallback::Updated()
-{
- if (_updated)
- {
- _updated = false;
- return true;
- }
- return false;
-}
-
-WebRtc_Word32
-QMTestVideoSettingsCallback::SetVideoQMSettings(const WebRtc_UWord32 frameRate,
- const WebRtc_UWord32 width,
- const WebRtc_UWord32 height)
-{
- WebRtc_Word32 retVal = 0;
- printf("QM updates: W = %d, H = %d, FR = %d, \n", width, height, frameRate);
- retVal = _vpm->SetTargetResolution(width, height, frameRate);
- //Initialize codec with new values - is this the best place to do it?
- if (!retVal)
- {
- // first get current settings
- VideoCodec currentCodec;
- _vcm->SendCodec(¤tCodec);
- // now set new values:
- currentCodec.height = (WebRtc_UWord16)height;
- currentCodec.width = (WebRtc_UWord16)width;
- currentCodec.maxFramerate = (WebRtc_UWord8)frameRate;
-
- // re-register encoder
- retVal = _vcm->RegisterSendCodec(¤tCodec, 2, 1440);
- _updated = true;
- }
-
- return retVal;
-}
-
-
-// Decoded Frame Callback Implmentation
-VCMQMDecodeCompleCallback::VCMQMDecodeCompleCallback(FILE* decodedFile):
-_decodedFile(decodedFile),
-_decodedBytes(0),
-//_test(test),
-_origWidth(0),
-_origHeight(0),
-_decWidth(0),
-_decHeight(0),
-//_interpolator(NULL),
-_decBuffer(NULL),
-_frameCnt(0)
-{
- //
-}
-
-VCMQMDecodeCompleCallback::~VCMQMDecodeCompleCallback()
- {
-// if (_interpolator != NULL)
-// {
-// deleteInterpolator(_interpolator);
-// _interpolator = NULL;
-// }
- if (_decBuffer != NULL)
- {
- delete [] _decBuffer;
- _decBuffer = NULL;
- }
- }
-WebRtc_Word32
-VCMQMDecodeCompleCallback::FrameToRender(VideoFrame& videoFrame)
-{
- if ((_origWidth == videoFrame.Width()) && (_origHeight == videoFrame.Height()))
- {
- fwrite(videoFrame.Buffer(), 1, videoFrame.Length(), _decodedFile);
- _frameCnt++;
- //printf("frame dec # %d", _frameCnt);
- // no need for interpolator and decBuffer
- if (_decBuffer != NULL)
- {
- delete [] _decBuffer;
- _decBuffer = NULL;
- }
-// if (_interpolator != NULL)
-// {
-// deleteInterpolator(_interpolator);
-// _interpolator = NULL;
-// }
- _decWidth = 0;
- _decHeight = 0;
- }
- else
- {
- if ((_decWidth != videoFrame.Width()) || (_decHeight != videoFrame.Height()))
- {
- _decWidth = videoFrame.Width();
- _decHeight = videoFrame.Height();
- buildInterpolator();
- }
-
-// interpolateFrame(_interpolator, videoFrame.Buffer(),_decBuffer);
- fwrite(_decBuffer, 1, _origWidth*_origHeight*3/2, _decodedFile);
- _frameCnt++;
- }
-
- _decodedBytes += videoFrame.Length();
- return VCM_OK;
-}
-
-WebRtc_Word32
-VCMQMDecodeCompleCallback::DecodedBytes()
-{
- return _decodedBytes;
-}
-
-void
-VCMQMDecodeCompleCallback::SetOriginalFrameDimensions(WebRtc_Word32 width, WebRtc_Word32 height)
-{
- _origWidth = width;
- _origHeight = height;
-}
-
-WebRtc_Word32
-VCMQMDecodeCompleCallback::buildInterpolator()
-{
- WebRtc_UWord32 decFrameLength = _origWidth*_origHeight*3 >> 1;
- if (_decBuffer != NULL)
- {
- delete [] _decBuffer;
- }
- _decBuffer = new WebRtc_UWord8[decFrameLength];
- if (_decBuffer == NULL)
- {
- return -1;
- }
-
- return 0;
-}
diff --git a/trunk/src/modules/video_coding/main/test/quality_modes_test.h b/trunk/src/modules/video_coding/main/test/quality_modes_test.h
deleted file mode 100644
index 87fa01f..0000000
--- a/trunk/src/modules/video_coding/main/test/quality_modes_test.h
+++ /dev/null
@@ -1,95 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_QUALITY_MODSE_TEST_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TEST_QUALITY_MODSE_TEST_H_
-
-#include "video_processing.h"
-#include "normal_test.h"
-#include "video_coding_defines.h"
-
-int qualityModeTest();
-
-class QualityModesTest : public NormalTest
-{
-public:
- QualityModesTest(webrtc::VideoCodingModule* vcm,
- webrtc::TickTimeBase* clock);
- virtual ~QualityModesTest();
- WebRtc_Word32 Perform();
-
-private:
-
- void Setup();
- void Print();
- void Teardown();
- void SsimComp();
-
- webrtc::VideoProcessingModule* _vpm;
-
- WebRtc_UWord32 _width;
- WebRtc_UWord32 _height;
- float _frameRate;
- WebRtc_UWord32 _nativeWidth;
- WebRtc_UWord32 _nativeHeight;
- float _nativeFrameRate;
-
- WebRtc_UWord32 _numFramesDroppedVPM;
- bool _flagSSIM;
-
-}; // end of QualityModesTest class
-
-
-class VCMQMDecodeCompleCallback: public webrtc::VCMReceiveCallback
-{
-public:
- VCMQMDecodeCompleCallback(FILE* decodedFile);
- virtual ~VCMQMDecodeCompleCallback();
- void SetUserReceiveCallback(webrtc::VCMReceiveCallback* receiveCallback);
- // will write decoded frame into file
- WebRtc_Word32 FrameToRender(webrtc::VideoFrame& videoFrame);
- WebRtc_Word32 DecodedBytes();
- void SetOriginalFrameDimensions(WebRtc_Word32 width, WebRtc_Word32 height);
- WebRtc_Word32 buildInterpolator();
-private:
- FILE* _decodedFile;
- WebRtc_UWord32 _decodedBytes;
- // QualityModesTest& _test;
- WebRtc_UWord32 _origWidth;
- WebRtc_UWord32 _origHeight;
- WebRtc_UWord32 _decWidth;
- WebRtc_UWord32 _decHeight;
-// VideoInterpolator* _interpolator;
- WebRtc_UWord8* _decBuffer;
- WebRtc_UWord32 _frameCnt; // debug
-
-}; // end of VCMQMDecodeCompleCallback class
-
-class QMTestVideoSettingsCallback : public webrtc::VCMQMSettingsCallback
-{
-public:
- QMTestVideoSettingsCallback();
- // update VPM with QM settings
- WebRtc_Word32 SetVideoQMSettings(const WebRtc_UWord32 frameRate,
- const WebRtc_UWord32 width,
- const WebRtc_UWord32 height);
- // register VPM used by test
- void RegisterVPM(webrtc::VideoProcessingModule* vpm);
- void RegisterVCM(webrtc::VideoCodingModule* vcm);
- bool Updated();
-
-private:
- webrtc::VideoProcessingModule* _vpm;
- webrtc::VideoCodingModule* _vcm;
- bool _updated;
-};
-
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_QUALITY_MODSE_TEST_H_
diff --git a/trunk/src/modules/video_coding/main/test/receiver_tests.h b/trunk/src/modules/video_coding/main/test/receiver_tests.h
deleted file mode 100644
index 33d9f5f..0000000
--- a/trunk/src/modules/video_coding/main/test/receiver_tests.h
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
-
-#include "video_coding.h"
-#include "module_common_types.h"
-#include "common_types.h"
-#include "rtp_rtcp.h"
-#include "typedefs.h"
-#include "rtp_player.h"
-#include "test_util.h"
-
-#include <string>
-#include <stdio.h>
-
-class RtpDataCallback : public webrtc::RtpData
-{
-public:
- RtpDataCallback(webrtc::VideoCodingModule* vcm)
- : _vcm(vcm) {};
-
- virtual WebRtc_Word32 OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const webrtc::WebRtcRTPHeader* rtpHeader);
-private:
- webrtc::VideoCodingModule* _vcm;
-};
-
-class FrameReceiveCallback : public webrtc::VCMReceiveCallback
-{
-public:
- FrameReceiveCallback(std::string outFilename) :
- _outFilename(outFilename),
- _outFile(NULL),
- _timingFile(NULL) {}
-
- virtual ~FrameReceiveCallback();
-
- WebRtc_Word32 FrameToRender(webrtc::VideoFrame& videoFrame);
-
-private:
- std::string _outFilename;
- FILE* _outFile;
- FILE* _timingFile;
-};
-
-class SharedState
-{
-public:
- SharedState(webrtc::VideoCodingModule& vcm, RTPPlayer& rtpPlayer) :
- _vcm(vcm),
- _rtpPlayer(rtpPlayer) {}
- webrtc::VideoCodingModule& _vcm;
- RTPPlayer& _rtpPlayer;
-};
-
-
-int RtpPlay(CmdArgs& args);
-int RtpPlayMT(CmdArgs& args,
- int releaseTest = 0,
- webrtc::VideoCodecType releaseTestVideoType = webrtc::kVideoCodecVP8);
-int ReceiverTimingTests(CmdArgs& args);
-int JitterBufferTest(CmdArgs& args);
-int DecodeFromStorageTest(CmdArgs& args);
-
-// Thread functions:
-bool ProcessingThread(void* obj);
-bool RtpReaderThread(void* obj);
-bool DecodeThread(void* obj);
-bool NackThread(void* obj);
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RECEIVER_TESTS_H_
diff --git a/trunk/src/modules/video_coding/main/test/receiver_timing_tests.cc b/trunk/src/modules/video_coding/main/test/receiver_timing_tests.cc
deleted file mode 100644
index 0b09256..0000000
--- a/trunk/src/modules/video_coding/main/test/receiver_timing_tests.cc
+++ /dev/null
@@ -1,235 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "receiver_tests.h"
-#include "video_coding.h"
-#include "trace.h"
-#include "../source/event.h"
-#include "../source/internal_defines.h"
-#include "timing.h"
-#include "test_macros.h"
-#include "test_util.h"
-
-#include <cstdio>
-#include <cstdlib>
-#include <cmath>
-
-using namespace webrtc;
-
-float vcmFloatMax(float a, float b)
-{
- return a > b ? a : b;
-}
-
-float vcmFloatMin(float a, float b)
-{
- return a < b ? a : b;
-}
-
-double const pi = 4*std::atan(1.0);
-
-class GaussDist
-{
-public:
- static float RandValue(float m, float stdDev) // returns a single normally distributed number
- {
- float r1 = static_cast<float>((std::rand() + 1.0)/(RAND_MAX + 1.0)); // gives equal distribution in (0, 1]
- float r2 = static_cast<float>((std::rand() + 1.0)/(RAND_MAX + 1.0));
- return m + stdDev * static_cast<float>(std::sqrt(-2*std::log(r1))*std::cos(2*pi*r2));
- }
-};
-
-int ReceiverTimingTests(CmdArgs& args)
-{
- // Make sure this test is never executed with simulated events.
-#if defined(EVENT_DEBUG)
- return -1;
-#endif
-
- // Set up trace
- Trace::CreateTrace();
- Trace::SetTraceFile((test::OutputPath() + "receiverTestTrace.txt").c_str());
- Trace::SetLevelFilter(webrtc::kTraceAll);
-
- // A static random seed
- srand(0);
-
- TickTimeBase clock;
- VCMTiming timing(&clock);
- float clockInMs = 0.0;
- WebRtc_UWord32 waitTime = 0;
- WebRtc_UWord32 jitterDelayMs = 0;
- WebRtc_UWord32 maxDecodeTimeMs = 0;
- WebRtc_UWord32 timeStamp = 0;
-
- timing.Reset(static_cast<WebRtc_Word64>(clockInMs + 0.5));
-
- timing.UpdateCurrentDelay(timeStamp);
-
- timing.Reset(static_cast<WebRtc_Word64>(clockInMs + 0.5));
-
- timing.IncomingTimestamp(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5));
- jitterDelayMs = 20;
- timing.SetRequiredDelay(jitterDelayMs);
- timing.UpdateCurrentDelay(timeStamp);
- waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
- static_cast<WebRtc_Word64>(clockInMs + 0.5));
- // First update initializes the render time. Since we have no decode delay
- // we get waitTime = renderTime - now - renderDelay = jitter
- TEST(waitTime == jitterDelayMs);
-
- jitterDelayMs += VCMTiming::kDelayMaxChangeMsPerS + 10;
- timeStamp += 90000;
- clockInMs += 1000.0f;
- timing.SetRequiredDelay(jitterDelayMs);
- timing.UpdateCurrentDelay(timeStamp);
- waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
- static_cast<WebRtc_Word64>(clockInMs + 0.5));
- // Since we gradually increase the delay we only get
- // 100 ms every second.
- TEST(waitTime == jitterDelayMs - 10);
-
- timeStamp += 90000;
- clockInMs += 1000.0;
- timing.UpdateCurrentDelay(timeStamp);
- waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
- static_cast<WebRtc_Word64>(clockInMs + 0.5));
- TEST(waitTime == jitterDelayMs);
-
- // 300 incoming frames without jitter, verify that this gives the exact wait time
- for (int i=0; i < 300; i++)
- {
- clockInMs += 1000.0f/30.0f;
- timeStamp += 3000;
- timing.IncomingTimestamp(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5));
- }
- timing.UpdateCurrentDelay(timeStamp);
- waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
- static_cast<WebRtc_Word64>(clockInMs + 0.5));
- TEST(waitTime == jitterDelayMs);
-
- // Add decode time estimates
- for (int i=0; i < 10; i++)
- {
- WebRtc_Word64 startTimeMs = static_cast<WebRtc_Word64>(clockInMs + 0.5);
- clockInMs += 10.0f;
- timing.StopDecodeTimer(timeStamp, startTimeMs, static_cast<WebRtc_Word64>(clockInMs + 0.5));
- timeStamp += 3000;
- clockInMs += 1000.0f/30.0f - 10.0f;
- timing.IncomingTimestamp(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5));
- }
- maxDecodeTimeMs = 10;
- timing.SetRequiredDelay(jitterDelayMs);
- clockInMs += 1000.0f;
- timeStamp += 90000;
- timing.UpdateCurrentDelay(timeStamp);
- waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
- static_cast<WebRtc_Word64>(clockInMs + 0.5));
- TEST(waitTime == jitterDelayMs);
-
- WebRtc_UWord32 totalDelay1 = timing.TargetVideoDelay();
- WebRtc_UWord32 minTotalDelayMs = 200;
- timing.SetMinimumTotalDelay(minTotalDelayMs);
- clockInMs += 5000.0f;
- timeStamp += 5*90000;
- timing.UpdateCurrentDelay(timeStamp);
- waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
- static_cast<WebRtc_Word64>(clockInMs + 0.5));
- WebRtc_UWord32 totalDelay2 = timing.TargetVideoDelay();
- // We should at least have minTotalDelayMs - decodeTime (10) - renderTime (10) to wait
- TEST(waitTime == minTotalDelayMs - maxDecodeTimeMs - 10);
- // The total video delay should not increase with the extra delay,
- // the extra delay should be independent.
- TEST(totalDelay1 == totalDelay2);
-
- // Reset min total delay
- timing.SetMinimumTotalDelay(0);
- clockInMs += 5000.0f;
- timeStamp += 5*90000;
- timing.UpdateCurrentDelay(timeStamp);
-
- // A sudden increase in timestamp of 2.1 seconds
- clockInMs += 1000.0f/30.0f;
- timeStamp += static_cast<WebRtc_UWord32>(2.1*90000 + 0.5);
- WebRtc_Word64 ret = timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5));
- TEST(ret == -1);
- timing.Reset();
-
- // This test produces a trace which can be parsed with plotTimingTest.m. The plot
- // can be used to see that the timing is reasonable under noise, and that the
- // gradual transition between delays works as expected.
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, -1, "Stochastic test 1");
-
- jitterDelayMs = 60;
- maxDecodeTimeMs = 10;
-
- timeStamp = static_cast<WebRtc_UWord32>(-10000); // To produce a wrap
- clockInMs = 10000.0f;
- timing.Reset(static_cast<WebRtc_Word64>(clockInMs + 0.5));
-
- float noise = 0.0f;
- for (int i=0; i < 1400; i++)
- {
- if (i == 400)
- {
- jitterDelayMs = 30;
- }
- else if (i == 700)
- {
- jitterDelayMs = 100;
- }
- else if (i == 1000)
- {
- minTotalDelayMs = 200;
- timing.SetMinimumTotalDelay(minTotalDelayMs);
- }
- else if (i == 1200)
- {
- minTotalDelayMs = 0;
- timing.SetMinimumTotalDelay(minTotalDelayMs);
- }
- WebRtc_Word64 startTimeMs = static_cast<WebRtc_Word64>(clockInMs + 0.5);
- noise = vcmFloatMin(vcmFloatMax(GaussDist::RandValue(0, 2), -10.0f), 30.0f);
- clockInMs += 10.0f;
- timing.StopDecodeTimer(timeStamp, startTimeMs, static_cast<WebRtc_Word64>(clockInMs + noise + 0.5));
- timeStamp += 3000;
- clockInMs += 1000.0f/30.0f - 10.0f;
- noise = vcmFloatMin(vcmFloatMax(GaussDist::RandValue(0, 8), -15.0f), 15.0f);
- timing.IncomingTimestamp(timeStamp, static_cast<WebRtc_Word64>(clockInMs + noise + 0.5));
- timing.SetRequiredDelay(jitterDelayMs);
- timing.UpdateCurrentDelay(timeStamp);
- waitTime = timing.MaxWaitingTime(timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5)),
- static_cast<WebRtc_Word64>(clockInMs + 0.5));
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, -1, "timeStamp=%u clock=%u maxWaitTime=%u", timeStamp,
- static_cast<WebRtc_UWord32>(clockInMs + 0.5), waitTime);
-
- WebRtc_Word64 renderTimeMs = timing.RenderTimeMs(timeStamp, static_cast<WebRtc_Word64>(clockInMs + 0.5));
-
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, -1,
- "timeStamp=%u renderTime=%u",
- timeStamp,
- MaskWord64ToUWord32(renderTimeMs));
- }
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideoCoding, -1, "End Stochastic test 1");
-
- printf("\nVCM Timing Test: \n\n%i tests completed\n", vcmMacrosTests);
- if (vcmMacrosErrors > 0)
- {
- printf("%i FAILED\n\n", vcmMacrosErrors);
- }
- else
- {
- printf("ALL PASSED\n\n");
- }
-
- Trace::ReturnTrace();
- return 0;
-}
diff --git a/trunk/src/modules/video_coding/main/test/release_test.cc b/trunk/src/modules/video_coding/main/test/release_test.cc
deleted file mode 100644
index 8e3a073..0000000
--- a/trunk/src/modules/video_coding/main/test/release_test.cc
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "ReleaseTest.h"
-#include "ReceiverTests.h"
-#include "TestMacros.h"
-#include "MediaOptTest.h"
-#include "CodecDataBaseTest.h"
-#include "GenericCodecTest.h"
-
-
-
-
-int ReleaseTest()
-{
- printf("VCM RELEASE TESTS \n\n");
-
- // Automatic tests
-
- printf("Testing receive side timing...\n");
- TEST(ReceiverTimingTests() == 0);
-
- printf("Testing jitter buffer...\n");
- TEST(JitterBufferTest() == 0);
-
- printf("Testing Codec Data Base...\n");
- TEST(CodecDBTest() == 0);
-
- printf("Testing Media Optimization....\n");
- TEST(VCMMediaOptTest(1) == 0);
-
- // Tests requiring verification
-
- printf("Testing Multi thread send-receive....\n");
- TEST(MTRxTxTest() == 0);
- printf("Verify by viewing output file MTRxTx_out.yuv \n");
-
- return 0;
-}
\ No newline at end of file
diff --git a/trunk/src/modules/video_coding/main/test/release_test.h b/trunk/src/modules/video_coding/main/test/release_test.h
deleted file mode 100644
index 2578160..0000000
--- a/trunk/src/modules/video_coding/main/test/release_test.h
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef RELEASE_TEST_H
-#define RELEASE_TEST_H
-
-int ReleaseTest();
-int ReleaseTestPart2();
-
-#endif
\ No newline at end of file
diff --git a/trunk/src/modules/video_coding/main/test/release_test_pt2.cc b/trunk/src/modules/video_coding/main/test/release_test_pt2.cc
deleted file mode 100644
index 5ff48e5..0000000
--- a/trunk/src/modules/video_coding/main/test/release_test_pt2.cc
+++ /dev/null
@@ -1,31 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "ReleaseTest.h"
-#include "ReceiverTests.h"
-#include "TestMacros.h"
-#include "MediaOptTest.h"
-#include "CodecDataBaseTest.h"
-#include "GenericCodecTest.h"
-
-
-
-
-int ReleaseTestPart2()
-{
- printf("Verify that TICK_TIME_DEBUG and EVENT_DEBUG are uncommented");
- // Tests requiring verification
-
- printf("Testing Generic Codecs...\n");
- TEST(VCMGenericCodecTest() == 0);
- printf("Verify by viewing output file GCTest_out.yuv \n");
-
- return 0;
-}
\ No newline at end of file
diff --git a/trunk/src/modules/video_coding/main/test/rtp_player.cc b/trunk/src/modules/video_coding/main/test/rtp_player.cc
deleted file mode 100644
index 5361e31..0000000
--- a/trunk/src/modules/video_coding/main/test/rtp_player.cc
+++ /dev/null
@@ -1,443 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rtp_player.h"
-
-#include <cstdlib>
-#ifdef WIN32
-#include <windows.h>
-#include <Winsock2.h>
-#else
-#include <arpa/inet.h>
-#endif
-
-#include "../source/internal_defines.h"
-#include "gtest/gtest.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
-#include "rtp_rtcp.h"
-
-using namespace webrtc;
-
-RawRtpPacket::RawRtpPacket(uint8_t* rtp_data, uint16_t rtp_length)
- : data(rtp_data),
- length(rtp_length),
- resend_time_ms(-1) {
- data = new uint8_t[length];
- memcpy(data, rtp_data, length);
-}
-
-RawRtpPacket::~RawRtpPacket() {
- delete [] data;
-}
-
-LostPackets::LostPackets()
- : crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
- loss_count_(0),
- debug_file_(NULL),
- packets_() {
- debug_file_ = fopen("PacketLossDebug.txt", "w");
-}
-
-LostPackets::~LostPackets() {
- if (debug_file_) {
- fclose(debug_file_);
- }
- while (!packets_.empty()) {
- delete packets_.front();
- packets_.pop_front();
- }
- delete crit_sect_;
-}
-
-void LostPackets::AddPacket(RawRtpPacket* packet) {
- CriticalSectionScoped cs(crit_sect_);
- packets_.push_back(packet);
- uint16_t seq_num = (packet->data[2] << 8) + packet->data[3];
- if (debug_file_ != NULL) {
- fprintf(debug_file_, "%u Lost packet: %u\n", loss_count_, seq_num);
- }
- ++loss_count_;
-}
-
-void LostPackets::SetResendTime(uint16_t resend_seq_num,
- int64_t resend_time_ms,
- int64_t now_ms) {
- CriticalSectionScoped cs(crit_sect_);
- for (RtpPacketIterator it = packets_.begin(); it != packets_.end(); ++it) {
- const uint16_t seq_num = ((*it)->data[2] << 8) +
- (*it)->data[3];
- if (resend_seq_num == seq_num) {
- if ((*it)->resend_time_ms + 10 < now_ms) {
- if (debug_file_ != NULL) {
- fprintf(debug_file_, "Resend %u at %u\n", seq_num,
- MaskWord64ToUWord32(resend_time_ms));
- }
- (*it)->resend_time_ms = resend_time_ms;
- }
- return;
- }
- }
- assert(false);
-}
-
-RawRtpPacket* LostPackets::NextPacketToResend(int64_t timeNow) {
- CriticalSectionScoped cs(crit_sect_);
- for (RtpPacketIterator it = packets_.begin(); it != packets_.end(); ++it) {
- if (timeNow >= (*it)->resend_time_ms && (*it)->resend_time_ms != -1) {
- RawRtpPacket* packet = *it;
- it = packets_.erase(it);
- return packet;
- }
- }
- return NULL;
-}
-
-int LostPackets::NumberOfPacketsToResend() const {
- CriticalSectionScoped cs(crit_sect_);
- int count = 0;
- for (ConstRtpPacketIterator it = packets_.begin(); it != packets_.end();
- ++it) {
- if ((*it)->resend_time_ms >= 0) {
- count++;
- }
- }
- return count;
-}
-
-void LostPackets::SetPacketResent(uint16_t seq_num, int64_t now_ms) {
- CriticalSectionScoped cs(crit_sect_);
- if (debug_file_ != NULL) {
- fprintf(debug_file_, "Resent %u at %u\n", seq_num,
- MaskWord64ToUWord32(now_ms));
- }
-}
-
-void LostPackets::Print() const {
- CriticalSectionScoped cs(crit_sect_);
- printf("Lost packets: %u\n", loss_count_);
- printf("Packets waiting to be resent: %u\n",
- NumberOfPacketsToResend());
- printf("Packets still lost: %u\n",
- static_cast<unsigned int>(packets_.size()));
- printf("Sequence numbers:\n");
- for (ConstRtpPacketIterator it = packets_.begin(); it != packets_.end();
- ++it) {
- uint16_t seq_num = ((*it)->data[2] << 8) + (*it)->data[3];
- printf("%u, ", seq_num);
- }
- printf("\n");
-}
-
-RTPPlayer::RTPPlayer(const char* filename,
- RtpData* callback,
- TickTimeBase* clock)
-:
-_clock(clock),
-_rtpModule(*RtpRtcp::CreateRtpRtcp(1, false)),
-_nextRtpTime(0),
-_dataCallback(callback),
-_firstPacket(true),
-_lossRate(0.0f),
-_nackEnabled(false),
-_resendPacketCount(0),
-_noLossStartup(100),
-_endOfFile(false),
-_rttMs(0),
-_firstPacketRtpTime(0),
-_firstPacketTimeMs(0),
-_reorderBuffer(NULL),
-_reordering(false),
-_nextPacket(),
-_nextPacketLength(0),
-_randVec(),
-_randVecPos(0)
-{
- _rtpFile = fopen(filename, "rb");
- memset(_nextPacket, 0, sizeof(_nextPacket));
-}
-
-RTPPlayer::~RTPPlayer()
-{
- RtpRtcp::DestroyRtpRtcp(&_rtpModule);
- if (_rtpFile != NULL)
- {
- fclose(_rtpFile);
- }
- if (_reorderBuffer != NULL)
- {
- delete _reorderBuffer;
- _reorderBuffer = NULL;
- }
-}
-
-WebRtc_Word32 RTPPlayer::Initialize(const PayloadTypeList* payloadList)
-{
- std::srand(321);
- for (int i=0; i < RAND_VEC_LENGTH; i++)
- {
- _randVec[i] = rand();
- }
- _randVecPos = 0;
- WebRtc_Word32 ret = _rtpModule.SetNACKStatus(kNackOff);
- if (ret < 0)
- {
- return -1;
- }
- ret = _rtpModule.InitReceiver();
- if (ret < 0)
- {
- return -1;
- }
-
- _rtpModule.InitSender();
- _rtpModule.SetRTCPStatus(kRtcpNonCompound);
- _rtpModule.SetTMMBRStatus(true);
-
- ret = _rtpModule.RegisterIncomingDataCallback(_dataCallback);
- if (ret < 0)
- {
- return -1;
- }
- // Register payload types
- for (PayloadTypeList::const_iterator it = payloadList->begin();
- it != payloadList->end(); ++it) {
- PayloadCodecTuple* payloadType = *it;
- if (payloadType != NULL)
- {
- VideoCodec videoCodec;
- strncpy(videoCodec.plName, payloadType->name.c_str(), 32);
- videoCodec.plType = payloadType->payloadType;
- if (_rtpModule.RegisterReceivePayload(videoCodec) < 0)
- {
- return -1;
- }
- }
- }
- if (ReadHeader() < 0)
- {
- return -1;
- }
- memset(_nextPacket, 0, sizeof(_nextPacket));
- _nextPacketLength = ReadPacket(_nextPacket, &_nextRtpTime);
- return 0;
-}
-
-WebRtc_Word32 RTPPlayer::ReadHeader()
-{
- char firstline[FIRSTLINELEN];
- if (_rtpFile == NULL)
- {
- return -1;
- }
- EXPECT_TRUE(fgets(firstline, FIRSTLINELEN, _rtpFile) != NULL);
- if(strncmp(firstline,"#!rtpplay",9) == 0) {
- if(strncmp(firstline,"#!rtpplay1.0",12) != 0){
- printf("ERROR: wrong rtpplay version, must be 1.0\n");
- return -1;
- }
- }
- else if (strncmp(firstline,"#!RTPencode",11) == 0) {
- if(strncmp(firstline,"#!RTPencode1.0",14) != 0){
- printf("ERROR: wrong RTPencode version, must be 1.0\n");
- return -1;
- }
- }
- else {
- printf("ERROR: wrong file format of input file\n");
- return -1;
- }
-
- WebRtc_UWord32 start_sec;
- WebRtc_UWord32 start_usec;
- WebRtc_UWord32 source;
- WebRtc_UWord16 port;
- WebRtc_UWord16 padding;
-
- EXPECT_GT(fread(&start_sec, 4, 1, _rtpFile), 0u);
- start_sec=ntohl(start_sec);
- EXPECT_GT(fread(&start_usec, 4, 1, _rtpFile), 0u);
- start_usec=ntohl(start_usec);
- EXPECT_GT(fread(&source, 4, 1, _rtpFile), 0u);
- source=ntohl(source);
- EXPECT_GT(fread(&port, 2, 1, _rtpFile), 0u);
- port=ntohs(port);
- EXPECT_GT(fread(&padding, 2, 1, _rtpFile), 0u);
- padding=ntohs(padding);
- return 0;
-}
-
-WebRtc_UWord32 RTPPlayer::TimeUntilNextPacket() const
-{
- WebRtc_Word64 timeLeft = (_nextRtpTime - _firstPacketRtpTime) - (_clock->MillisecondTimestamp() - _firstPacketTimeMs);
- if (timeLeft < 0)
- {
- return 0;
- }
- return static_cast<WebRtc_UWord32>(timeLeft);
-}
-
-WebRtc_Word32 RTPPlayer::NextPacket(const WebRtc_Word64 timeNow)
-{
- // Send any packets ready to be resent,
- RawRtpPacket* resend_packet = _lostPackets.NextPacketToResend(timeNow);
- while (resend_packet != NULL) {
- const uint16_t seqNo = (resend_packet->data[2] << 8) +
- resend_packet->data[3];
- printf("Resend: %u\n", seqNo);
- int ret = SendPacket(resend_packet->data, resend_packet->length);
- delete resend_packet;
- _resendPacketCount++;
- if (ret > 0) {
- _lostPackets.SetPacketResent(seqNo, _clock->MillisecondTimestamp());
- } else if (ret < 0) {
- return ret;
- }
- resend_packet = _lostPackets.NextPacketToResend(timeNow);
- }
-
- // Send any packets from rtp file
- if (!_endOfFile && (TimeUntilNextPacket() == 0 || _firstPacket))
- {
- _rtpModule.Process();
- if (_firstPacket)
- {
- _firstPacketRtpTime = static_cast<WebRtc_Word64>(_nextRtpTime);
- _firstPacketTimeMs = _clock->MillisecondTimestamp();
- }
- if (_reordering && _reorderBuffer == NULL)
- {
- _reorderBuffer = new RawRtpPacket(reinterpret_cast<WebRtc_UWord8*>(_nextPacket), static_cast<WebRtc_UWord16>(_nextPacketLength));
- return 0;
- }
- WebRtc_Word32 ret = SendPacket(reinterpret_cast<WebRtc_UWord8*>(_nextPacket), static_cast<WebRtc_UWord16>(_nextPacketLength));
- if (_reordering && _reorderBuffer != NULL)
- {
- RawRtpPacket* rtpPacket = _reorderBuffer;
- _reorderBuffer = NULL;
- SendPacket(rtpPacket->data, rtpPacket->length);
- delete rtpPacket;
- }
- _firstPacket = false;
- if (ret < 0)
- {
- return ret;
- }
- _nextPacketLength = ReadPacket(_nextPacket, &_nextRtpTime);
- if (_nextPacketLength < 0)
- {
- _endOfFile = true;
- return 0;
- }
- else if (_nextPacketLength == 0)
- {
- return 0;
- }
- }
- if (_endOfFile && _lostPackets.NumberOfPacketsToResend() == 0)
- {
- return 1;
- }
- return 0;
-}
-
-WebRtc_Word32 RTPPlayer::SendPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen)
-{
- if ((_randVec[(_randVecPos++) % RAND_VEC_LENGTH] + 1.0)/(RAND_MAX + 1.0) < _lossRate &&
- _noLossStartup < 0)
- {
- if (_nackEnabled)
- {
- const WebRtc_UWord16 seqNo = (rtpData[2] << 8) + rtpData[3];
- printf("Throw: %u\n", seqNo);
- _lostPackets.AddPacket(new RawRtpPacket(rtpData, rtpLen));
- return 0;
- }
- }
- else if (rtpLen > 0)
- {
- WebRtc_Word32 ret = _rtpModule.IncomingPacket(rtpData, rtpLen);
- if (ret < 0)
- {
- return -1;
- }
- }
- if (_noLossStartup >= 0)
- {
- _noLossStartup--;
- }
- return 1;
-}
-
-WebRtc_Word32 RTPPlayer::ReadPacket(WebRtc_Word16* rtpdata, WebRtc_UWord32* offset)
-{
- WebRtc_UWord16 length, plen;
-
- if (fread(&length,2,1,_rtpFile)==0)
- return(-1);
- length=ntohs(length);
-
- if (fread(&plen,2,1,_rtpFile)==0)
- return(-1);
- plen=ntohs(plen);
-
- if (fread(offset,4,1,_rtpFile)==0)
- return(-1);
- *offset=ntohl(*offset);
-
- // Use length here because a plen of 0 specifies rtcp
- length = (WebRtc_UWord16) (length - HDR_SIZE);
- if (fread((unsigned short *) rtpdata,1,length,_rtpFile) != length)
- return(-1);
-
-#ifdef JUNK_DATA
- // destroy the RTP payload with random data
- if (plen > 12) { // ensure that we have more than just a header
- for ( int ix = 12; ix < plen; ix=ix+2 ) {
- rtpdata[ix>>1] = (short) (rtpdata[ix>>1] + (short) rand());
- }
- }
-#endif
- return plen;
-}
-
-WebRtc_Word32 RTPPlayer::SimulatePacketLoss(float lossRate, bool enableNack, WebRtc_UWord32 rttMs)
-{
- _nackEnabled = enableNack;
- _lossRate = lossRate;
- _rttMs = rttMs;
- return 0;
-}
-
-WebRtc_Word32 RTPPlayer::SetReordering(bool enabled)
-{
- _reordering = enabled;
- return 0;
-}
-
-WebRtc_Word32 RTPPlayer::ResendPackets(const WebRtc_UWord16* sequenceNumbers, WebRtc_UWord16 length)
-{
- if (sequenceNumbers == NULL)
- {
- return 0;
- }
- for (int i=0; i < length; i++)
- {
- _lostPackets.SetResendTime(sequenceNumbers[i],
- _clock->MillisecondTimestamp() + _rttMs,
- _clock->MillisecondTimestamp());
- }
- return 0;
-}
-
-void RTPPlayer::Print() const
-{
- printf("Resent packets: %u\n", _resendPacketCount);
- _lostPackets.Print();
-}
diff --git a/trunk/src/modules/video_coding/main/test/rtp_player.h b/trunk/src/modules/video_coding/main/test/rtp_player.h
deleted file mode 100644
index eac6ba8..0000000
--- a/trunk/src/modules/video_coding/main/test/rtp_player.h
+++ /dev/null
@@ -1,119 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
-
-#include "typedefs.h"
-#include "rtp_rtcp.h"
-#include "critical_section_wrapper.h"
-#include "video_coding_defines.h"
-#include "modules/video_coding/main/source/tick_time_base.h"
-
-#include <stdio.h>
-#include <list>
-#include <string>
-
-#define HDR_SIZE 8 // rtpplay packet header size in bytes
-#define FIRSTLINELEN 40
-#define RAND_VEC_LENGTH 4096
-
-struct PayloadCodecTuple;
-
-struct RawRtpPacket
-{
-public:
- RawRtpPacket(WebRtc_UWord8* rtp_data, WebRtc_UWord16 rtp_length);
- ~RawRtpPacket();
-
- uint8_t* data;
- uint16_t length;
- int64_t resend_time_ms;
-};
-
-typedef std::list<PayloadCodecTuple*> PayloadTypeList;
-typedef std::list<RawRtpPacket*> RtpPacketList;
-typedef RtpPacketList::iterator RtpPacketIterator;
-typedef RtpPacketList::const_iterator ConstRtpPacketIterator;
-
-class LostPackets {
- public:
- LostPackets();
- ~LostPackets();
-
- void AddPacket(RawRtpPacket* packet);
- void SetResendTime(uint16_t sequenceNumber,
- int64_t resendTime,
- int64_t nowMs);
- RawRtpPacket* NextPacketToResend(int64_t timeNow);
- int NumberOfPacketsToResend() const;
- void SetPacketResent(uint16_t seqNo, int64_t nowMs);
- void Print() const;
-
- private:
- webrtc::CriticalSectionWrapper* crit_sect_;
- int loss_count_;
- FILE* debug_file_;
- RtpPacketList packets_;
-};
-
-struct PayloadCodecTuple
-{
- PayloadCodecTuple(WebRtc_UWord8 plType, std::string codecName, webrtc::VideoCodecType type) :
- name(codecName), payloadType(plType), codecType(type) {};
- const std::string name;
- const WebRtc_UWord8 payloadType;
- const webrtc::VideoCodecType codecType;
-};
-
-class RTPPlayer : public webrtc::VCMPacketRequestCallback
-{
-public:
- RTPPlayer(const char* filename,
- webrtc::RtpData* callback,
- webrtc::TickTimeBase* clock);
- virtual ~RTPPlayer();
-
- WebRtc_Word32 Initialize(const PayloadTypeList* payloadList);
- WebRtc_Word32 NextPacket(const WebRtc_Word64 timeNow);
- WebRtc_UWord32 TimeUntilNextPacket() const;
- WebRtc_Word32 SimulatePacketLoss(float lossRate, bool enableNack = false, WebRtc_UWord32 rttMs = 0);
- WebRtc_Word32 SetReordering(bool enabled);
- WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequenceNumbers, WebRtc_UWord16 length);
- void Print() const;
-
-private:
- WebRtc_Word32 SendPacket(WebRtc_UWord8* rtpData, WebRtc_UWord16 rtpLen);
- WebRtc_Word32 ReadPacket(WebRtc_Word16* rtpdata, WebRtc_UWord32* offset);
- WebRtc_Word32 ReadHeader();
- webrtc::TickTimeBase* _clock;
- FILE* _rtpFile;
- webrtc::RtpRtcp& _rtpModule;
- WebRtc_UWord32 _nextRtpTime;
- webrtc::RtpData* _dataCallback;
- bool _firstPacket;
- float _lossRate;
- bool _nackEnabled;
- LostPackets _lostPackets;
- WebRtc_UWord32 _resendPacketCount;
- WebRtc_Word32 _noLossStartup;
- bool _endOfFile;
- WebRtc_UWord32 _rttMs;
- WebRtc_Word64 _firstPacketRtpTime;
- WebRtc_Word64 _firstPacketTimeMs;
- RawRtpPacket* _reorderBuffer;
- bool _reordering;
- WebRtc_Word16 _nextPacket[8000];
- WebRtc_Word32 _nextPacketLength;
- int _randVec[RAND_VEC_LENGTH];
- int _randVecPos;
-};
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_RTP_PLAYER_H_
diff --git a/trunk/src/modules/video_coding/main/test/subfigure.m b/trunk/src/modules/video_coding/main/test/subfigure.m
deleted file mode 100644
index eadfcb6..0000000
--- a/trunk/src/modules/video_coding/main/test/subfigure.m
+++ /dev/null
@@ -1,30 +0,0 @@
-function H = subfigure(m, n, p)
-%
-% H = SUBFIGURE(m, n, p)
-%
-% Create a new figure window and adjust position and size such that it will
-% become the p-th tile in an m-by-n matrix of windows. (The interpretation of
-% m, n, and p is the same as for SUBPLOT.
-%
-% Henrik Lundin, 2009-01-19
-%
-
-
-h = figure;
-
-[j, i] = ind2sub([n m], p);
-scrsz = get(0,'ScreenSize'); % get screen size
-%scrsz = [1, 1, 1600, 1200];
-
-taskbarSize = 58;
-windowbarSize = 68;
-windowBorder = 4;
-
-scrsz(2) = scrsz(2) + taskbarSize;
-scrsz(4) = scrsz(4) - taskbarSize;
-
-set(h, 'position', [(j-1)/n * scrsz(3) + scrsz(1) + windowBorder,...
- (m-i)/m * scrsz(4) + scrsz(2) + windowBorder, ...
- scrsz(3)/n - (windowBorder + windowBorder),...
- scrsz(4)/m - (windowbarSize + windowBorder + windowBorder)]);
-
diff --git a/trunk/src/modules/video_coding/main/test/test_callbacks.cc b/trunk/src/modules/video_coding/main/test/test_callbacks.cc
deleted file mode 100644
index 1ec0e59..0000000
--- a/trunk/src/modules/video_coding/main/test/test_callbacks.cc
+++ /dev/null
@@ -1,509 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "test_callbacks.h"
-
-#include <cmath>
-
-#include "modules/video_coding/main/source/tick_time_base.h"
-#include "rtp_dump.h"
-#include "test_macros.h"
-
-namespace webrtc {
-
-/******************************
- * VCMEncodeCompleteCallback
- *****************************/
-// Basic callback implementation
-// passes the encoded frame directly to the encoder
-// Packetization callback implementation
-VCMEncodeCompleteCallback::VCMEncodeCompleteCallback(FILE* encodedFile):
- _encodedFile(encodedFile),
- _encodedBytes(0),
- _VCMReceiver(NULL),
- _seqNo(0),
- _encodeComplete(false),
- _width(0),
- _height(0),
- _codecType(kRTPVideoNoVideo)
-{
- //
-}
-VCMEncodeCompleteCallback::~VCMEncodeCompleteCallback()
-{
-}
-
-void
-VCMEncodeCompleteCallback::RegisterTransportCallback(
- VCMPacketizationCallback* transport)
-{
-}
-
-WebRtc_Word32
-VCMEncodeCompleteCallback::SendData(
- const FrameType frameType,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader& fragmentationHeader,
- const RTPVideoHeader* videoHdr)
-{
- // will call the VCMReceiver input packet
- _frameType = frameType;
- // writing encodedData into file
- fwrite(payloadData, 1, payloadSize, _encodedFile);
- WebRtcRTPHeader rtpInfo;
- rtpInfo.header.markerBit = true; // end of frame
- rtpInfo.type.Video.isFirstPacket = true;
- rtpInfo.type.Video.codec = _codecType;
- rtpInfo.type.Video.height = (WebRtc_UWord16)_height;
- rtpInfo.type.Video.width = (WebRtc_UWord16)_width;
- switch (_codecType)
- {
- case webrtc::kRTPVideoVP8:
- rtpInfo.type.Video.codecHeader.VP8.InitRTPVideoHeaderVP8();
- rtpInfo.type.Video.codecHeader.VP8.nonReference =
- videoHdr->codecHeader.VP8.nonReference;
- rtpInfo.type.Video.codecHeader.VP8.pictureId =
- videoHdr->codecHeader.VP8.pictureId;
- break;
- case webrtc::kRTPVideoI420:
- break;
- default:
- assert(false);
- return -1;
- }
-
- rtpInfo.header.payloadType = payloadType;
- rtpInfo.header.sequenceNumber = _seqNo++;
- rtpInfo.header.ssrc = 0;
- rtpInfo.header.timestamp = timeStamp;
- rtpInfo.frameType = frameType;
- // Size should also be received from that table, since the payload type
- // defines the size.
-
- _encodedBytes += payloadSize;
- // directly to receiver
- int ret = _VCMReceiver->IncomingPacket(payloadData, payloadSize, rtpInfo);
- _encodeComplete = true;
-
- return ret;
-}
-
-float
-VCMEncodeCompleteCallback::EncodedBytes()
-{
- return _encodedBytes;
-}
-
-bool
-VCMEncodeCompleteCallback::EncodeComplete()
-{
- if (_encodeComplete)
- {
- _encodeComplete = false;
- return true;
- }
- return false;
-}
-
-void
-VCMEncodeCompleteCallback::Initialize()
-{
- _encodeComplete = false;
- _encodedBytes = 0;
- _seqNo = 0;
- return;
-}
-
-void
-VCMEncodeCompleteCallback::ResetByteCount()
-{
- _encodedBytes = 0;
-}
-
-/***********************************/
-/* VCMRTPEncodeCompleteCallback */
-/***********************************/
-// Encode Complete callback implementation
-// passes the encoded frame via the RTP module to the decoder
-// Packetization callback implementation
-
-WebRtc_Word32
-VCMRTPEncodeCompleteCallback::SendData(
- const FrameType frameType,
- const WebRtc_UWord8 payloadType,
- const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader& fragmentationHeader,
- const RTPVideoHeader* videoHdr)
-{
- _frameType = frameType;
- _encodedBytes+= payloadSize;
- _encodeComplete = true;
- return _RTPModule->SendOutgoingData(frameType,
- payloadType,
- timeStamp,
- payloadData,
- payloadSize,
- &fragmentationHeader,
- videoHdr);
-}
-
-float
-VCMRTPEncodeCompleteCallback::EncodedBytes()
-{
- // only good for one call - after which will reset value;
- float tmp = _encodedBytes;
- _encodedBytes = 0;
- return tmp;
- }
-
-bool
-VCMRTPEncodeCompleteCallback::EncodeComplete()
-{
- if (_encodeComplete)
- {
- _encodeComplete = false;
- return true;
- }
- return false;
-}
-
-// Decoded Frame Callback Implementation
-
-WebRtc_Word32
-VCMDecodeCompleteCallback::FrameToRender(VideoFrame& videoFrame)
-{
- fwrite(videoFrame.Buffer(), 1, videoFrame.Length(), _decodedFile);
- _decodedBytes+= videoFrame.Length();
- return VCM_OK;
- }
-
-WebRtc_Word32
-VCMDecodeCompleteCallback::DecodedBytes()
-{
- return _decodedBytes;
-}
-
-RTPSendCompleteCallback::RTPSendCompleteCallback(RtpRtcp* rtp,
- TickTimeBase* clock,
- const char* filename):
- _clock(clock),
- _sendCount(0),
- _rtp(rtp),
- _lossPct(0),
- _burstLength(0),
- _networkDelayMs(0),
- _jitterVar(0),
- _prevLossState(0),
- _totalSentLength(0),
- _rtpPackets(),
- _rtpDump(NULL)
-{
- if (filename != NULL)
- {
- _rtpDump = RtpDump::CreateRtpDump();
- _rtpDump->Start(filename);
- }
-}
-
-RTPSendCompleteCallback::~RTPSendCompleteCallback()
-{
- if (_rtpDump != NULL)
- {
- _rtpDump->Stop();
- RtpDump::DestroyRtpDump(_rtpDump);
- }
- // Delete remaining packets
- while (!_rtpPackets.empty())
- {
- // Take first packet in list
- delete _rtpPackets.front();
- _rtpPackets.pop_front();
- }
-}
-
-int
-RTPSendCompleteCallback::SendPacket(int channel, const void *data, int len)
-{
- _sendCount++;
- _totalSentLength += len;
-
- if (_rtpDump != NULL)
- {
- if (_rtpDump->DumpPacket((const WebRtc_UWord8*)data, len) != 0)
- {
- return -1;
- }
- }
-
- bool transmitPacket = true;
- transmitPacket = PacketLoss();
-
- WebRtc_UWord64 now = _clock->MillisecondTimestamp();
- // Insert outgoing packet into list
- if (transmitPacket)
- {
- RtpPacket* newPacket = new RtpPacket();
- memcpy(newPacket->data, data, len);
- newPacket->length = len;
- // Simulate receive time = network delay + packet jitter
- // simulated as a Normal distribution random variable with
- // mean = networkDelay and variance = jitterVar
- WebRtc_Word32
- simulatedDelay = (WebRtc_Word32)NormalDist(_networkDelayMs,
- sqrt(_jitterVar));
- newPacket->receiveTime = now + simulatedDelay;
- _rtpPackets.push_back(newPacket);
- }
-
- // Are we ready to send packets to the receiver?
- RtpPacket* packet = NULL;
-
- while (!_rtpPackets.empty())
- {
- // Take first packet in list
- packet = _rtpPackets.front();
- WebRtc_Word64 timeToReceive = packet->receiveTime - now;
- if (timeToReceive > 0)
- {
- // No available packets to send
- break;
- }
-
- _rtpPackets.pop_front();
- // Send to receive side
- if (_rtp->IncomingPacket((const WebRtc_UWord8*)packet->data,
- packet->length) < 0)
- {
- delete packet;
- packet = NULL;
- // Will return an error after the first packet that goes wrong
- return -1;
- }
- delete packet;
- packet = NULL;
- }
- return len; // OK
-}
-
-int
-RTPSendCompleteCallback::SendRTCPPacket(int channel, const void *data, int len)
-{
- // Incorporate network conditions
- return SendPacket(channel, data, len);
-}
-
-void
-RTPSendCompleteCallback::SetLossPct(double lossPct)
-{
- _lossPct = lossPct;
- return;
-}
-
-void
-RTPSendCompleteCallback::SetBurstLength(double burstLength)
-{
- _burstLength = burstLength;
- return;
-}
-
-bool
-RTPSendCompleteCallback::PacketLoss()
-{
- bool transmitPacket = true;
- if (_burstLength <= 1.0)
- {
- // Random loss: if _burstLength parameter is not set, or <=1
- if (UnifomLoss(_lossPct))
- {
- // drop
- transmitPacket = false;
- }
- }
- else
- {
- // Simulate bursty channel (Gilbert model)
- // (1st order) Markov chain model with memory of the previous/last
- // packet state (loss or received)
-
- // 0 = received state
- // 1 = loss state
-
- // probTrans10: if previous packet is lost, prob. to -> received state
- // probTrans11: if previous packet is lost, prob. to -> loss state
-
- // probTrans01: if previous packet is received, prob. to -> loss state
- // probTrans00: if previous packet is received, prob. to -> received
-
- // Map the two channel parameters (average loss rate and burst length)
- // to the transition probabilities:
- double probTrans10 = 100 * (1.0 / _burstLength);
- double probTrans11 = (100.0 - probTrans10);
- double probTrans01 = (probTrans10 * ( _lossPct / (100.0 - _lossPct)));
-
- // Note: Random loss (Bernoulli) model is a special case where:
- // burstLength = 100.0 / (100.0 - _lossPct) (i.e., p10 + p01 = 100)
-
- if (_prevLossState == 0 )
- {
- // previous packet was received
- if (UnifomLoss(probTrans01))
- {
- // drop, update previous state to loss
- _prevLossState = 1;
- transmitPacket = false;
- }
- }
- else if (_prevLossState == 1)
- {
- _prevLossState = 0;
- // previous packet was lost
- if (UnifomLoss(probTrans11))
- {
- // drop, update previous state to loss
- _prevLossState = 1;
- transmitPacket = false;
- }
- }
- }
- return transmitPacket;
-}
-
-
-bool
-RTPSendCompleteCallback::UnifomLoss(double lossPct)
-{
- double randVal = (std::rand() + 1.0)/(RAND_MAX + 1.0);
- return randVal < lossPct/100;
-}
-
-WebRtc_Word32
-PacketRequester::ResendPackets(const WebRtc_UWord16* sequenceNumbers,
- WebRtc_UWord16 length)
-{
- return _rtp.SendNACK(sequenceNumbers, length);
-}
-
-WebRtc_Word32
-SendStatsTest::SendStatistics(const WebRtc_UWord32 bitRate,
- const WebRtc_UWord32 frameRate)
-{
- TEST(frameRate <= _frameRate);
- TEST(bitRate > 0 && bitRate < 100000);
- printf("VCM 1 sec: Bit rate: %u\tFrame rate: %u\n", bitRate, frameRate);
- return 0;
-}
-
-WebRtc_Word32 KeyFrameReqTest::RequestKeyFrame() {
- printf("Key frame requested\n");
- return 0;
-}
-
-
-VideoProtectionCallback::VideoProtectionCallback():
-_deltaFECRate(0),
-_keyFECRate(0),
-_deltaUseUepProtection(0),
-_keyUseUepProtection(0),
-_nack(kNackOff)
-{
- //
-}
-
-VideoProtectionCallback::~VideoProtectionCallback()
-{
- //
-}
-
-WebRtc_Word32
-VideoProtectionCallback::ProtectionRequest(WebRtc_UWord8 deltaFECRate,
- WebRtc_UWord8 keyFECRate,
- bool deltaUseUepProtection,
- bool keyUseUepProtection,
- bool nack_enabled,
- WebRtc_UWord32* sent_video_rate_bps,
- WebRtc_UWord32* sent_nack_rate_bps,
- WebRtc_UWord32* sent_fec_rate_bps)
-{
- _deltaFECRate = deltaFECRate;
- _keyFECRate = keyFECRate;
- _deltaUseUepProtection = deltaUseUepProtection;
- _keyUseUepProtection = keyUseUepProtection;
- if (nack_enabled)
- {
- _nack = kNackRtcp;
- }
- else
- {
- _nack = kNackOff;
- }
-
- // Update RTP
- if (_rtp->SetFECCodeRate(keyFECRate, deltaFECRate) != 0)
- {
- printf("Error in Setting FEC rate\n");
- return -1;
-
- }
- if (_rtp->SetFECUepProtection(keyUseUepProtection,
- deltaUseUepProtection) != 0)
- {
- printf("Error in Setting FEC UEP protection\n");
- return -1;
- }
- return 0;
-
-}
-NACKMethod
-VideoProtectionCallback::NACKMethod()
-{
- return _nack;
-}
-
-WebRtc_UWord8
-VideoProtectionCallback::FECDeltaRate()
-{
- return _deltaFECRate;
-}
-
-WebRtc_UWord8
-VideoProtectionCallback::FECKeyRate()
-{
- return _keyFECRate;
-}
-
-bool
-VideoProtectionCallback::FECDeltaUepProtection()
-{
- return _deltaUseUepProtection;
-}
-
-bool
-VideoProtectionCallback::FECKeyUepProtection()
-{
- return _keyUseUepProtection;
-}
-
-void
-RTPFeedbackCallback::OnNetworkChanged(const WebRtc_Word32 id,
- const WebRtc_UWord32 bitrateBps,
- const WebRtc_UWord8 fractionLost,
- const WebRtc_UWord16 roundTripTimeMs)
-{
-
- _vcm->SetChannelParameters(bitrateBps / 1000, fractionLost,
- (WebRtc_UWord8)roundTripTimeMs);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_coding/main/test/test_callbacks.h b/trunk/src/modules/video_coding/main/test/test_callbacks.h
deleted file mode 100644
index 07820bb..0000000
--- a/trunk/src/modules/video_coding/main/test/test_callbacks.h
+++ /dev/null
@@ -1,279 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_CALLBACKS_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_CALLBACKS_H_
-
-/*
- * Declaration of general callbacks that are used throughout VCM's offline tests
- */
-
-
-#include <string.h>
-#include <cstdlib>
-#include <fstream>
-#include <list>
-
-#include "module_common_types.h"
-#include "rtp_rtcp.h"
-#include "test_util.h"
-#include "trace.h"
-#include "video_coding.h"
-
-namespace webrtc
-{
-class RtpDump;
-
-// Send Side - Packetization callback - send an encoded frame to the VCMReceiver
-class VCMEncodeCompleteCallback: public VCMPacketizationCallback
-{
-public:
- // Constructor input: file in which encoded data will be written
- VCMEncodeCompleteCallback(FILE* encodedFile);
- virtual ~VCMEncodeCompleteCallback();
- // Register transport callback
- void RegisterTransportCallback(VCMPacketizationCallback* transport);
- // Process encoded data received from the encoder, pass stream to the
- // VCMReceiver module
- WebRtc_Word32 SendData(const FrameType frameType,
- const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData, const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader& fragmentationHeader,
- const RTPVideoHeader* videoHdr);
- // Register exisitng VCM. Currently - encode and decode under same module.
- void RegisterReceiverVCM(VideoCodingModule *vcm) {_VCMReceiver = vcm;}
- // Return size of last encoded frame data (all frames in the sequence)
- // Good for only one call - after which will reset value
- // (to allow detection of frame drop)
- float EncodedBytes();
- // Return encode complete (true/false)
- bool EncodeComplete();
- // Inform callback of codec used
- void SetCodecType(RTPVideoCodecTypes codecType)
- {_codecType = codecType;}
- // Inform callback of frame dimensions
- void SetFrameDimensions(WebRtc_Word32 width, WebRtc_Word32 height)
- {
- _width = width;
- _height = height;
- }
- // Initialize callback data
- void Initialize();
- void ResetByteCount();
-
- // Conversion function for payload type (needed for the callback function)
-
-private:
- FILE* _encodedFile;
- float _encodedBytes;
- VideoCodingModule* _VCMReceiver;
- FrameType _frameType;
- WebRtc_UWord8* _payloadData;
- WebRtc_UWord16 _seqNo;
- bool _encodeComplete;
- WebRtc_Word32 _width;
- WebRtc_Word32 _height;
- RTPVideoCodecTypes _codecType;
-
-}; // end of VCMEncodeCompleteCallback
-
-// Send Side - Packetization callback - packetize an encoded frame via the
-// RTP module
-class VCMRTPEncodeCompleteCallback: public VCMPacketizationCallback
-{
-public:
- VCMRTPEncodeCompleteCallback(RtpRtcp* rtp) :
- _encodedBytes(0),
- _seqNo(0),
- _encodeComplete(false),
- _RTPModule(rtp) {}
-
- virtual ~VCMRTPEncodeCompleteCallback() {}
- // Process encoded data received from the encoder, pass stream to the
- // RTP module
- WebRtc_Word32 SendData(const FrameType frameType,
- const WebRtc_UWord8 payloadType, const WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData, const WebRtc_UWord32 payloadSize,
- const RTPFragmentationHeader& fragmentationHeader,
- const RTPVideoHeader* videoHdr);
- // Return size of last encoded frame. Value good for one call
- // (resets to zero after call to inform test of frame drop)
- float EncodedBytes();
- // Return encode complete (true/false)
- bool EncodeComplete();
- // Inform callback of codec used
- void SetCodecType(RTPVideoCodecTypes codecType)
- {_codecType = codecType;}
-
- // Inform callback of frame dimensions
- void SetFrameDimensions(WebRtc_Word16 width, WebRtc_Word16 height)
- {
- _width = width;
- _height = height;
- }
-
-private:
- float _encodedBytes;
- FrameType _frameType;
- WebRtc_UWord8* _payloadData;
- WebRtc_UWord16 _seqNo;
- bool _encodeComplete;
- RtpRtcp* _RTPModule;
- WebRtc_Word16 _width;
- WebRtc_Word16 _height;
- RTPVideoCodecTypes _codecType;
-}; // end of VCMEncodeCompleteCallback
-
-// Decode Complete callback
-// Writes the decoded frames to a given file.
-class VCMDecodeCompleteCallback: public VCMReceiveCallback
-{
-public:
- VCMDecodeCompleteCallback(FILE* decodedFile) :
- _decodedFile(decodedFile), _decodedBytes(0) {}
- virtual ~VCMDecodeCompleteCallback() {}
- // Write decoded frame into file
- WebRtc_Word32 FrameToRender(webrtc::VideoFrame& videoFrame);
- WebRtc_Word32 DecodedBytes();
-private:
- FILE* _decodedFile;
- WebRtc_UWord32 _decodedBytes;
-}; // end of VCMDecodeCompleCallback class
-
-// Transport callback
-// Called by the RTP Sender - simulates sending packets through a network to the
-// RTP receiver. User can set network conditions as: RTT, packet loss,
-// burst length and jitter.
-class RTPSendCompleteCallback: public Transport
-{
-public:
- // Constructor input: (receive side) rtp module to send encoded data to
- RTPSendCompleteCallback(RtpRtcp* rtp, TickTimeBase* clock,
- const char* filename = NULL);
- virtual ~RTPSendCompleteCallback();
- // Send Packet to receive side RTP module
- virtual int SendPacket(int channel, const void *data, int len);
- // Send RTCP Packet to receive side RTP module
- virtual int SendRTCPPacket(int channel, const void *data, int len);
- // Set percentage of channel loss in the network
- void SetLossPct(double lossPct);
- // Set average size of burst loss
- void SetBurstLength(double burstLength);
- // Set network delay in the network
- void SetNetworkDelay(WebRtc_UWord32 networkDelayMs)
- {_networkDelayMs = networkDelayMs;};
- // Set Packet jitter delay
- void SetJitterVar(WebRtc_UWord32 jitterVar)
- {_jitterVar = jitterVar;};
- // Return send count
- int SendCount() {return _sendCount; }
- // Return accumulated length in bytes of transmitted packets
- WebRtc_UWord32 TotalSentLength() {return _totalSentLength;}
-protected:
- // Randomly decide whether to drop packets, based on the channel model
- bool PacketLoss();
- // Random uniform loss model
- bool UnifomLoss(double lossPct);
-
- TickTimeBase* _clock;
- WebRtc_UWord32 _sendCount;
- RtpRtcp* _rtp;
- double _lossPct;
- double _burstLength;
- WebRtc_UWord32 _networkDelayMs;
- double _jitterVar;
- bool _prevLossState;
- WebRtc_UWord32 _totalSentLength;
- std::list<RtpPacket*> _rtpPackets;
- RtpDump* _rtpDump;
-};
-
-// Request re-transmission of packets (NACK)
-class PacketRequester: public VCMPacketRequestCallback
-{
-public:
- PacketRequester(RtpRtcp& rtp) :
- _rtp(rtp) {}
- WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequenceNumbers,
- WebRtc_UWord16 length);
-private:
- webrtc::RtpRtcp& _rtp;
-};
-
-// Key frame request
-class KeyFrameReqTest: public VCMFrameTypeCallback
-{
-public:
- WebRtc_Word32 RequestKeyFrame();
-};
-
-
-// VCM statistics
-class SendStatsTest: public webrtc::VCMSendStatisticsCallback
-{
-public:
- SendStatsTest() : _frameRate(15) {}
- WebRtc_Word32 SendStatistics(const WebRtc_UWord32 bitRate,
- const WebRtc_UWord32 frameRate);
- void SetTargetFrameRate(WebRtc_UWord32 frameRate) {_frameRate = frameRate;}
-private:
- WebRtc_UWord32 _frameRate;
-};
-
-// Protection callback - allows the VCM (media optimization) to inform the RTP
-// module of the required protection(FEC rates/settings and NACK mode).
-class VideoProtectionCallback: public VCMProtectionCallback
-{
-public:
- VideoProtectionCallback();
- virtual ~VideoProtectionCallback();
- void RegisterRtpModule(RtpRtcp* rtp) {_rtp = rtp;}
- WebRtc_Word32 ProtectionRequest(WebRtc_UWord8 deltaFECRate,
- WebRtc_UWord8 keyFECRate,
- bool deltaUseUepProtection,
- bool keyUseUepProtection,
- bool nack_enabled,
- WebRtc_UWord32* sent_video_rate_bps,
- WebRtc_UWord32* sent_nack_rate_bps,
- WebRtc_UWord32* sent_fec_rate_bps);
- enum NACKMethod NACKMethod();
- WebRtc_UWord8 FECDeltaRate();
- WebRtc_UWord8 FECKeyRate();
- bool FECDeltaUepProtection();
- bool FECKeyUepProtection();
-private:
- RtpRtcp* _rtp;
- WebRtc_UWord8 _deltaFECRate;
- WebRtc_UWord8 _keyFECRate;
- bool _deltaUseUepProtection;
- bool _keyUseUepProtection;
- enum NACKMethod _nack;
-};
-
-// Feed back from the RTP Module callback
-class RTPFeedbackCallback : public RtpVideoFeedback {
- public:
- RTPFeedbackCallback(VideoCodingModule* vcm) {_vcm = vcm;};
- void OnReceivedIntraFrameRequest(const WebRtc_Word32 id,
- const FrameType type,
- const WebRtc_UWord8 streamIdx) {};
-
- void OnNetworkChanged(const WebRtc_Word32 id,
- const WebRtc_UWord32 bitrateBps,
- const WebRtc_UWord8 fractionLost,
- const WebRtc_UWord16 roundTripTimeMs);
-
- private:
- VideoCodingModule* _vcm;
-};
-
-} // namespace webrtc
-
-#endif
diff --git a/trunk/src/modules/video_coding/main/test/test_macros.h b/trunk/src/modules/video_coding/main/test/test_macros.h
deleted file mode 100644
index 31693b5..0000000
--- a/trunk/src/modules/video_coding/main/test/test_macros.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef VCM_TEST_MACROS_H
-#define VCM_TEST_MACROS_H
-
-#include <cstdio>
-#include <cstdlib>
-
-extern int vcmMacrosTests;
-extern int vcmMacrosErrors;
-
-#define PRINT_ERR_MSG(msg) \
- do { \
- fprintf(stderr, "Error at line %i of %s\n%s", \
- __LINE__, __FILE__, msg); \
- } while(0)
-
-#define TEST(expr) \
- do { \
- vcmMacrosTests++; \
- if (!(expr)) { \
- PRINT_ERR_MSG("Assertion failed: " #expr "\n\n"); \
- vcmMacrosErrors++; \
- } \
- } while(0)
-
-#define TEST_EXIT_ON_FAIL(expr) \
- do { \
- vcmMacrosTests++; \
- if (!(expr)) { \
- PRINT_ERR_MSG("Assertion failed: " #expr "\nExiting...\n\n"); \
- vcmMacrosErrors++; \
- exit(EXIT_FAILURE); \
- } \
- } while(0)
-
-#endif
diff --git a/trunk/src/modules/video_coding/main/test/test_util.cc b/trunk/src/modules/video_coding/main/test/test_util.cc
deleted file mode 100644
index 1f89168..0000000
--- a/trunk/src/modules/video_coding/main/test/test_util.cc
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "test_util.h"
-#include "test_macros.h"
-#include "rtp_dump.h"
-#include <cmath>
-
-using namespace webrtc;
-
-// Normal Distribution
-#define PI 3.14159265
-double
-NormalDist(double mean, double stdDev)
-{
- // Creating a Normal distribution variable from two independent uniform
- // variables based on the Box-Muller transform
- double uniform1 = (std::rand() + 1.0) / (RAND_MAX + 1.0);
- double uniform2 = (std::rand() + 1.0) / (RAND_MAX + 1.0);
- return (mean + stdDev * sqrt(-2 * log(uniform1)) * cos(2 * PI * uniform2));
-}
-
-RTPVideoCodecTypes
-ConvertCodecType(const char* plname)
-{
- if (strncmp(plname,"VP8" , 3) == 0)
- {
- return kRTPVideoVP8;
- }
- else if (strncmp(plname,"I420" , 5) == 0)
- {
- return kRTPVideoI420;
- }
- else
- {
- return kRTPVideoNoVideo; // Default value
- }
-}
-
diff --git a/trunk/src/modules/video_coding/main/test/test_util.h b/trunk/src/modules/video_coding/main/test/test_util.h
deleted file mode 100644
index d705434..0000000
--- a/trunk/src/modules/video_coding/main/test/test_util.h
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TEST_TEST_UTIL_H_
-
-/*
- * General declarations used through out VCM offline tests.
- */
-
-#include <string.h>
-#include <fstream>
-#include <cstdlib>
-
-#include "module_common_types.h"
-#include "testsupport/fileutils.h"
-
-// Class used for passing command line arguments to tests
-class CmdArgs
-{
- public:
- CmdArgs()
- : codecName("VP8"),
- codecType(webrtc::kVideoCodecVP8),
- width(352),
- height(288),
- bitRate(500),
- frameRate(30),
- packetLoss(0),
- rtt(0),
- protectionMode(0),
- camaEnable(0),
- inputFile(webrtc::test::ProjectRootPath() +
- "/resources/foreman_cif.yuv"),
- outputFile(webrtc::test::OutputPath() +
- "video_coding_test_output_352x288.yuv"),
- testNum(11) {}
- std::string codecName;
- webrtc::VideoCodecType codecType;
- int width;
- int height;
- int bitRate;
- int frameRate;
- int packetLoss;
- int rtt;
- int protectionMode;
- int camaEnable;
- std::string inputFile;
- std::string outputFile;
- int testNum;
-};
-
-// forward declaration
-int MTRxTxTest(CmdArgs& args);
-double NormalDist(double mean, double stdDev);
-
-struct RtpPacket {
- WebRtc_Word8 data[1650]; // max packet size
- WebRtc_Word32 length;
- WebRtc_Word64 receiveTime;
-};
-
-
-// Codec type conversion
-webrtc::RTPVideoCodecTypes
-ConvertCodecType(const char* plname);
-
-#endif
diff --git a/trunk/src/modules/video_coding/main/test/tester_main.cc b/trunk/src/modules/video_coding/main/test/tester_main.cc
deleted file mode 100644
index e5d7cd3..0000000
--- a/trunk/src/modules/video_coding/main/test/tester_main.cc
+++ /dev/null
@@ -1,208 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "receiver_tests.h"
-#include "normal_test.h"
-#include "codec_database_test.h"
-#include "generic_codec_test.h"
-#include "../source/event.h"
-#include "media_opt_test.h"
-#include "quality_modes_test.h"
-#include "test_util.h"
-
-#include <stdlib.h>
-#include <string.h>
-
-#ifdef _WIN32
-//#include "vld.h"
-#endif
-
-using namespace webrtc;
-
-/*
- * Build with EVENT_DEBUG defined
- * to build the tests with simulated events.
- */
-
-int vcmMacrosTests = 0;
-int vcmMacrosErrors = 0;
-
-int ParseArguments(int argc, char **argv, CmdArgs& args)
-{
- int i = 1;
- while (i < argc)
- {
- if (argv[i][0] != '-')
- {
- return -1;
- }
- switch (argv[i][1])
- {
- case 'w':
- {
- int w = atoi(argv[i+1]);
- if (w < 1)
- return -1;
- args.width = w;
- break;
- }
- case 'h':
- {
- int h = atoi(argv[i+1]);
- if (h < 1)
- return -1;
- args.height = h;
- break;
- }
- case 'b':
- {
- int b = atoi(argv[i+1]);
- if (b < 1)
- return -1;
- args.bitRate = b;
- break;
- }
- case 'f':
- {
- int f = atoi(argv[i+1]);
- if (f < 1)
- return -1;
- args.frameRate = f;
- break;
- }
- case 'c':
- {
- // TODO(holmer): This should be replaced with a map if more codecs
- // are added
- args.codecName = argv[i+1];
- if (strncmp(argv[i+1], "VP8", 3) == 0)
- {
- args.codecType = kVideoCodecVP8;
- }
- else if (strncmp(argv[i+1], "I420", 4) == 0)
- {
- args.codecType = kVideoCodecI420;
- }
- else
- return -1;
-
- break;
- }
- case 'i':
- {
- args.inputFile = argv[i+1];
- break;
- }
- case 'o':
- args.outputFile = argv[i+1];
- break;
- case 'n':
- {
- int n = atoi(argv[i+1]);
- if (n < 1)
- return -1;
- args.testNum = n;
- break;
- }
- case 'p':
- {
- args.packetLoss = atoi(argv[i+1]);
- break;
- }
- case 'r':
- {
- args.rtt = atoi(argv[i+1]);
- break;
- }
- case 'm':
- {
- args.protectionMode = atoi(argv[i+1]);
- break;
- }
- case 'e':
- {
- args.camaEnable = atoi(argv[i+1]);
- break;
- }
- default:
- return -1;
- }
- i += 2;
- }
- return 0;
-}
-
-int main(int argc, char **argv)
-{
- CmdArgs args;
-
- if (ParseArguments(argc, argv, args) != 0)
- {
- printf("Unable to parse input arguments\n");
- printf("args: -n <test #> -w <width> -h <height> -f <fps> -b <bps> "
- "-c <codec> -i <input file> -o <output file> -p <packet loss> "
- "-r <round-trip-time> -e <cama enable> -m <protection mode> \n");
- return -1;
- }
-
- int ret = 0;
- switch (args.testNum)
- {
- case 1:
- ret = NormalTest::RunTest(args);
- break;
- case 2:
- ret = MTRxTxTest(args);
- break;
- case 3:
- ret = GenericCodecTest::RunTest(args);
- break;
- case 4:
- ret = CodecDataBaseTest::RunTest(args);
- break;
- case 5:
- // 0- normal, 1-Release test(50 runs) 2- from file
- ret = MediaOptTest::RunTest(0, args);
- break;
- case 6:
- ret = ReceiverTimingTests(args);
- break;
- case 7:
- ret = RtpPlay(args);
- break;
- case 8:
- ret = RtpPlayMT(args);
- break;
- case 9:
- ret = JitterBufferTest(args);
- break;
- case 10:
- ret = DecodeFromStorageTest(args);
- break;
- case 11:
- ret = NormalTest::RunTest(args);
- ret |= CodecDataBaseTest::RunTest(args);
- ret |= ReceiverTimingTests(args);
- ret |= JitterBufferTest(args);
- break;
- default:
- ret = -1;
- break;
- }
- if (ret != 0)
- {
- printf("Test failed!\n");
- return -1;
- }
- return 0;
-}
-
-
-
diff --git a/trunk/src/modules/video_coding/main/test/video_rtp_play.cc b/trunk/src/modules/video_coding/main/test/video_rtp_play.cc
deleted file mode 100644
index 577408f..0000000
--- a/trunk/src/modules/video_coding/main/test/video_rtp_play.cc
+++ /dev/null
@@ -1,202 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "receiver_tests.h"
-#include "video_coding.h"
-#include "rtp_rtcp.h"
-#include "trace.h"
-#include "../source/event.h"
-#include "../source/internal_defines.h"
-#include "test_macros.h"
-#include "rtp_player.h"
-#include "modules/video_coding/main/source/mock/fake_tick_time.h"
-
-#include <stdio.h>
-#include <string.h>
-
-using namespace webrtc;
-
-WebRtc_Word32
-RtpDataCallback::OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const WebRtcRTPHeader* rtpHeader)
-{
- return _vcm->IncomingPacket(payloadData, payloadSize, *rtpHeader);
-}
-
-FrameReceiveCallback::~FrameReceiveCallback()
-{
- if (_timingFile != NULL)
- {
- fclose(_timingFile);
- }
- if (_outFile != NULL)
- {
- fclose(_outFile);
- }
-}
-
-WebRtc_Word32
-FrameReceiveCallback::FrameToRender(VideoFrame& videoFrame)
-{
- if (_timingFile == NULL)
- {
- _timingFile = fopen((test::OutputPath() + "renderTiming.txt").c_str(),
- "w");
- if (_timingFile == NULL)
- {
- return -1;
- }
- }
- if (_outFile == NULL)
- {
- _outFile = fopen(_outFilename.c_str(), "wb");
- if (_outFile == NULL)
- {
- return -1;
- }
- }
- fprintf(_timingFile, "%u, %u\n",
- videoFrame.TimeStamp(),
- MaskWord64ToUWord32(videoFrame.RenderTimeMs()));
- fwrite(videoFrame.Buffer(), 1, videoFrame.Length(), _outFile);
- return 0;
-}
-
-int RtpPlay(CmdArgs& args)
-{
- // Make sure this test isn't executed without simulated events.
-#if !defined(EVENT_DEBUG)
- return -1;
-#endif
- // BEGIN Settings
-
- bool protectionEnabled = false;
- VCMVideoProtection protectionMethod = kProtectionNack;
- WebRtc_UWord32 rttMS = 0;
- float lossRate = 0.0f;
- bool reordering = false;
- WebRtc_UWord32 renderDelayMs = 0;
- WebRtc_UWord32 minPlayoutDelayMs = 0;
- const WebRtc_Word64 MAX_RUNTIME_MS = -1;
- std::string outFile = args.outputFile;
- if (outFile == "")
- outFile = test::OutputPath() + "RtpPlay_decoded.yuv";
- FrameReceiveCallback receiveCallback(outFile);
- FakeTickTime clock(0);
- VideoCodingModule* vcm = VideoCodingModule::Create(1, &clock);
- RtpDataCallback dataCallback(vcm);
- RTPPlayer rtpStream(args.inputFile.c_str(), &dataCallback, &clock);
-
-
- PayloadTypeList payloadTypes;
- payloadTypes.push_front(new PayloadCodecTuple(VCM_VP8_PAYLOAD_TYPE, "VP8",
- kVideoCodecVP8));
-
- Trace::CreateTrace();
- Trace::SetTraceFile((test::OutputPath() + "receiverTestTrace.txt").c_str());
- Trace::SetLevelFilter(webrtc::kTraceAll);
- // END Settings
-
- // Set up
-
- WebRtc_Word32 ret = vcm->InitializeReceiver();
- if (ret < 0)
- {
- return -1;
- }
- vcm->RegisterReceiveCallback(&receiveCallback);
- vcm->RegisterPacketRequestCallback(&rtpStream);
-
- // Register receive codecs in VCM
- for (PayloadTypeList::iterator it = payloadTypes.begin();
- it != payloadTypes.end(); ++it) {
- PayloadCodecTuple* payloadType = *it;
- if (payloadType != NULL)
- {
- VideoCodec codec;
- if (VideoCodingModule::Codec(payloadType->codecType, &codec) < 0)
- {
- return -1;
- }
- codec.plType = payloadType->payloadType;
- if (vcm->RegisterReceiveCodec(&codec, 1) < 0)
- {
- return -1;
- }
- }
- }
-
- if (rtpStream.Initialize(&payloadTypes) < 0)
- {
- return -1;
- }
- bool nackEnabled = protectionEnabled &&
- (protectionMethod == kProtectionNack ||
- protectionMethod == kProtectionDualDecoder);
- rtpStream.SimulatePacketLoss(lossRate, nackEnabled, rttMS);
- rtpStream.SetReordering(reordering);
- vcm->SetChannelParameters(0, 0, rttMS);
- vcm->SetVideoProtection(protectionMethod, protectionEnabled);
- vcm->SetRenderDelay(renderDelayMs);
- vcm->SetMinimumPlayoutDelay(minPlayoutDelayMs);
-
- ret = 0;
-
- // RTP stream main loop
- while ((ret = rtpStream.NextPacket(clock.MillisecondTimestamp())) == 0)
- {
- if (clock.MillisecondTimestamp() % 5 == 0)
- {
- ret = vcm->Decode();
- if (ret < 0)
- {
- return -1;
- }
- }
- while (vcm->DecodeDualFrame(0) == 1);
- if (vcm->TimeUntilNextProcess() <= 0)
- {
- vcm->Process();
- }
- if (MAX_RUNTIME_MS > -1 && clock.MillisecondTimestamp() >=
- MAX_RUNTIME_MS)
- {
- break;
- }
- clock.IncrementDebugClock(1);
- }
-
- switch (ret)
- {
- case 1:
- printf("Success\n");
- break;
- case -1:
- printf("Failed\n");
- break;
- case 0:
- printf("Timeout\n");
- break;
- }
-
- rtpStream.Print();
-
- // Tear down
- while (!payloadTypes.empty())
- {
- delete payloadTypes.front();
- payloadTypes.pop_front();
- }
- delete vcm;
- vcm = NULL;
- Trace::ReturnTrace();
- return 0;
-}
diff --git a/trunk/src/modules/video_coding/main/test/video_rtp_play_mt.cc b/trunk/src/modules/video_coding/main/test/video_rtp_play_mt.cc
deleted file mode 100644
index 21b9f73..0000000
--- a/trunk/src/modules/video_coding/main/test/video_rtp_play_mt.cc
+++ /dev/null
@@ -1,261 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "receiver_tests.h"
-#include "video_coding.h"
-#include "rtp_rtcp.h"
-#include "trace.h"
-#include "thread_wrapper.h"
-#include "../source/event.h"
-#include "test_macros.h"
-#include "rtp_player.h"
-
-#include <string.h>
-
-using namespace webrtc;
-
-bool ProcessingThread(void* obj)
-{
- SharedState* state = static_cast<SharedState*>(obj);
- if (state->_vcm.TimeUntilNextProcess() <= 0)
- {
- if (state->_vcm.Process() < 0)
- {
- return false;
- }
- }
- return true;
-}
-
-bool RtpReaderThread(void* obj)
-{
- SharedState* state = static_cast<SharedState*>(obj);
- EventWrapper& waitEvent = *EventWrapper::Create();
- // RTP stream main loop
- TickTimeBase clock;
- if (state->_rtpPlayer.NextPacket(clock.MillisecondTimestamp()) < 0)
- {
- return false;
- }
- waitEvent.Wait(state->_rtpPlayer.TimeUntilNextPacket());
- delete &waitEvent;
- return true;
-}
-
-bool DecodeThread(void* obj)
-{
- SharedState* state = static_cast<SharedState*>(obj);
- state->_vcm.Decode(10000);
- while (state->_vcm.DecodeDualFrame(0) == 1);
- return true;
-}
-
-int RtpPlayMT(CmdArgs& args, int releaseTestNo, webrtc::VideoCodecType releaseTestVideoType)
-{
- // Don't run these tests with debug events.
-#if defined(EVENT_DEBUG)
- return -1;
-#endif
-
- // BEGIN Settings
-
- bool protectionEnabled = true;
- VCMVideoProtection protection = kProtectionDualDecoder;
- WebRtc_UWord8 rttMS = 50;
- float lossRate = 0.05f;
- WebRtc_UWord32 renderDelayMs = 0;
- WebRtc_UWord32 minPlayoutDelayMs = 0;
- const WebRtc_Word64 MAX_RUNTIME_MS = 10000;
- std::string outFilename = args.outputFile;
- if (outFilename == "")
- outFilename = test::OutputPath() + "RtpPlayMT_decoded.yuv";
-
- bool nackEnabled = (protectionEnabled &&
- (protection == kProtectionDualDecoder ||
- protection == kProtectionNack ||
- kProtectionNackFEC));
- TickTimeBase clock;
- VideoCodingModule* vcm =
- VideoCodingModule::Create(1, &clock);
- RtpDataCallback dataCallback(vcm);
- std::string rtpFilename;
- rtpFilename = args.inputFile;
- if (releaseTestNo > 0)
- {
- // Setup a release test
- switch (releaseTestVideoType)
- {
- case webrtc::kVideoCodecVP8:
- rtpFilename = args.inputFile;
- outFilename = test::OutputPath() + "MTReceiveTest_VP8";
- break;
- default:
- return -1;
- }
- switch (releaseTestNo)
- {
- case 1:
- // Normal execution
- protectionEnabled = false;
- nackEnabled = false;
- rttMS = 0;
- lossRate = 0.0f;
- outFilename += "_Normal.yuv";
- break;
- case 2:
- // Packet loss
- protectionEnabled = false;
- nackEnabled = false;
- rttMS = 0;
- lossRate = 0.05f;
- outFilename += "_0.05.yuv";
- break;
- case 3:
- // Packet loss and NACK
- protection = kProtectionNack;
- nackEnabled = true;
- protectionEnabled = true;
- rttMS = 100;
- lossRate = 0.05f;
- outFilename += "_0.05_NACK_100ms.yuv";
- break;
- case 4:
- // Packet loss and dual decoder
- // Not implemented
- return 0;
- break;
- default:
- return -1;
- }
- printf("Watch %s to verify that the output is reasonable\n", outFilename.c_str());
- }
- RTPPlayer rtpStream(rtpFilename.c_str(), &dataCallback, &clock);
- PayloadTypeList payloadTypes;
- payloadTypes.push_front(new PayloadCodecTuple(VCM_VP8_PAYLOAD_TYPE, "VP8",
- kVideoCodecVP8));
- Trace::CreateTrace();
- Trace::SetTraceFile("receiverTestTrace.txt");
- Trace::SetLevelFilter(webrtc::kTraceAll);
-
- // END Settings
-
- // Set up
-
- SharedState mtState(*vcm, rtpStream);
-
- if (rtpStream.Initialize(&payloadTypes) < 0)
- {
- return -1;
- }
- rtpStream.SimulatePacketLoss(lossRate, nackEnabled, rttMS);
-
- WebRtc_Word32 ret = vcm->InitializeReceiver();
- if (ret < 0)
- {
- return -1;
- }
-
- // Create and start all threads
- ThreadWrapper* processingThread = ThreadWrapper::CreateThread(ProcessingThread,
- &mtState, kNormalPriority, "ProcessingThread");
- ThreadWrapper* rtpReaderThread = ThreadWrapper::CreateThread(RtpReaderThread,
- &mtState, kNormalPriority, "RtpReaderThread");
- ThreadWrapper* decodeThread = ThreadWrapper::CreateThread(DecodeThread,
- &mtState, kNormalPriority, "DecodeThread");
-
- // Register receive codecs in VCM
- for (PayloadTypeList::iterator it = payloadTypes.begin();
- it != payloadTypes.end(); ++it) {
- PayloadCodecTuple* payloadType = *it;
- if (payloadType != NULL)
- {
- VideoCodec codec;
- VideoCodingModule::Codec(payloadType->codecType, &codec);
- codec.plType = payloadType->payloadType;
- if (vcm->RegisterReceiveCodec(&codec, 1) < 0)
- {
- return -1;
- }
- }
- }
-
- if (processingThread != NULL)
- {
- unsigned int tid;
- processingThread->Start(tid);
- }
- else
- {
- printf("Unable to start processing thread\n");
- return -1;
- }
- if (rtpReaderThread != NULL)
- {
- unsigned int tid;
- rtpReaderThread->Start(tid);
- }
- else
- {
- printf("Unable to start RTP reader thread\n");
- return -1;
- }
- if (decodeThread != NULL)
- {
- unsigned int tid;
- decodeThread->Start(tid);
- }
- else
- {
- printf("Unable to start decode thread\n");
- return -1;
- }
-
- FrameReceiveCallback receiveCallback(outFilename);
- vcm->RegisterReceiveCallback(&receiveCallback);
- vcm->RegisterPacketRequestCallback(&rtpStream);
-
- vcm->SetChannelParameters(0, 0, rttMS);
- vcm->SetVideoProtection(protection, protectionEnabled);
- vcm->SetRenderDelay(renderDelayMs);
- vcm->SetMinimumPlayoutDelay(minPlayoutDelayMs);
-
- EventWrapper& waitEvent = *EventWrapper::Create();
-
- // Decode for 10 seconds and then tear down and exit.
- waitEvent.Wait(MAX_RUNTIME_MS);
-
- // Tear down
- while (!payloadTypes.empty())
- {
- delete payloadTypes.front();
- payloadTypes.pop_front();
- }
- while (!processingThread->Stop())
- {
- ;
- }
- while (!rtpReaderThread->Stop())
- {
- ;
- }
- while (!decodeThread->Stop())
- {
- ;
- }
- VideoCodingModule::Destroy(vcm);
- vcm = NULL;
- delete &waitEvent;
- delete processingThread;
- delete decodeThread;
- delete rtpReaderThread;
- rtpStream.Print();
- Trace::ReturnTrace();
- return 0;
-}
diff --git a/trunk/src/modules/video_coding/main/test/video_source.cc b/trunk/src/modules/video_coding/main/test/video_source.cc
deleted file mode 100644
index d7ba0b9..0000000
--- a/trunk/src/modules/video_coding/main/test/video_source.cc
+++ /dev/null
@@ -1,202 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_source.h"
-
-#include <cassert>
-
-#include "testsupport/fileutils.h"
-
-VideoSource::VideoSource()
-:
-_fileName(webrtc::test::ProjectRootPath() + "resources/foreman_cif.yuv"),
-_width(352),
-_height(288),
-_type(webrtc::kI420),
-_frameRate(30)
-{
- //
-}
-
-VideoSource::VideoSource(std::string fileName, VideoSize size,
- float frameRate, webrtc::VideoType type /*= webrtc::kI420*/)
-:
-_fileName(fileName),
-_width(0),
-_height(0),
-_type(type),
-_frameRate(frameRate)
-{
- assert(size != kUndefined && size != kNumberOfVideoSizes);
- assert(type != webrtc::kUnknown);
- assert(frameRate > 0);
- GetWidthHeight(size);
-}
-
-VideoSource::VideoSource(std::string fileName, WebRtc_UWord16 width, WebRtc_UWord16 height,
- float frameRate /*= 30*/, webrtc::VideoType type /*= webrtc::kI420*/)
-:
-_fileName(fileName),
-_width(width),
-_height(height),
-_type(type),
-_frameRate(frameRate)
-{
- assert(width > 0);
- assert(height > 0);
- assert(type != webrtc::kUnknown);
- assert(frameRate > 0);
-}
-
-WebRtc_Word32
-VideoSource::GetFrameLength() const
-{
- return webrtc::CalcBufferSize(_type, _width, _height);
-}
-
-std::string
-VideoSource::GetName() const
-{
- // Remove path.
- size_t slashPos = _fileName.find_last_of("/\\");
- if (slashPos == std::string::npos)
- {
- slashPos = 0;
- }
- else
- {
- slashPos++;
- }
-
- // Remove extension and underscored suffix if it exists.
- //return _fileName.substr(slashPos, std::min(_fileName.find_last_of("_"),
- // _fileName.find_last_of(".")) - slashPos);
- // MS: Removing suffix, not underscore....keeping full file name
- return _fileName.substr(slashPos, _fileName.find_last_of(".") - slashPos);
-
-}
-
-int
-VideoSource::GetWidthHeight( VideoSize size)
-{
- switch(size)
- {
- case kSQCIF:
- _width = 128;
- _height = 96;
- return 0;
- case kQQVGA:
- _width = 160;
- _height = 120;
- return 0;
- case kQCIF:
- _width = 176;
- _height = 144;
- return 0;
- case kCGA:
- _width = 320;
- _height = 200;
- return 0;
- case kQVGA:
- _width = 320;
- _height = 240;
- return 0;
- case kSIF:
- _width = 352;
- _height = 240;
- return 0;
- case kWQVGA:
- _width = 400;
- _height = 240;
- return 0;
- case kCIF:
- _width = 352;
- _height = 288;
- return 0;
- case kW288p:
- _width = 512;
- _height = 288;
- return 0;
- case k448p:
- _width = 576;
- _height = 448;
- return 0;
- case kVGA:
- _width = 640;
- _height = 480;
- return 0;
- case k432p:
- _width = 720;
- _height = 432;
- return 0;
- case kW432p:
- _width = 768;
- _height = 432;
- return 0;
- case k4SIF:
- _width = 704;
- _height = 480;
- return 0;
- case kW448p:
- _width = 768;
- _height = 448;
- return 0;
- case kNTSC:
- _width = 720;
- _height = 480;
- return 0;
- case kFW448p:
- _width = 800;
- _height = 448;
- return 0;
- case kWVGA:
- _width = 800;
- _height = 480;
- return 0;
- case k4CIF:
- _width = 704;
- _height = 576;
- return 0;
- case kSVGA:
- _width = 800;
- _height = 600;
- return 0;
- case kW544p:
- _width = 960;
- _height = 544;
- return 0;
- case kW576p:
- _width = 1024;
- _height = 576;
- return 0;
- case kHD:
- _width = 960;
- _height = 720;
- return 0;
- case kXGA:
- _width = 1024;
- _height = 768;
- return 0;
- case kFullHD:
- _width = 1440;
- _height = 1080;
- return 0;
- case kWHD:
- _width = 1280;
- _height = 720;
- return 0;
- case kWFullHD:
- _width = 1920;
- _height = 1080;
- return 0;
- default:
- return -1;
- }
-}
diff --git a/trunk/src/modules/video_coding/main/test/video_source.h b/trunk/src/modules/video_coding/main/test/video_source.h
deleted file mode 100644
index d9ad3d8..0000000
--- a/trunk/src/modules/video_coding/main/test/video_source.h
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
-#define WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
-
-#include "common_video/libyuv/include/libyuv.h"
-#include "typedefs.h"
-
-#include <string>
-
-enum VideoSize
- {
- kUndefined,
- kSQCIF, // 128*96 = 12 288
- kQQVGA, // 160*120 = 19 200
- kQCIF, // 176*144 = 25 344
- kCGA, // 320*200 = 64 000
- kQVGA, // 320*240 = 76 800
- kSIF, // 352*240 = 84 480
- kWQVGA, // 400*240 = 96 000
- kCIF, // 352*288 = 101 376
- kW288p, // 512*288 = 147 456 (WCIF)
- k448p, // 576*448 = 281 088
- kVGA, // 640*480 = 307 200
- k432p, // 720*432 = 311 040
- kW432p, // 768*432 = 331 776
- k4SIF, // 704*480 = 337 920
- kW448p, // 768*448 = 344 064
- kNTSC, // 720*480 = 345 600
- kFW448p, // 800*448 = 358 400
- kWVGA, // 800*480 = 384 000
- k4CIF, // 704*576 = 405 504
- kSVGA, // 800*600 = 480 000
- kW544p, // 960*544 = 522 240
- kW576p, // 1024*576 = 589 824 (W4CIF)
- kHD, // 960*720 = 691 200
- kXGA, // 1024*768 = 786 432
- kWHD, // 1280*720 = 921 600
- kFullHD, // 1440*1080 = 1 555 200
- kWFullHD, // 1920*1080 = 2 073 600
-
- kNumberOfVideoSizes
- };
-
-
-class VideoSource
-{
-public:
- VideoSource();
- VideoSource(std::string fileName, VideoSize size, float frameRate, webrtc::VideoType type = webrtc::kI420);
- VideoSource(std::string fileName, WebRtc_UWord16 width, WebRtc_UWord16 height,
- float frameRate = 30, webrtc::VideoType type = webrtc::kI420);
-
- std::string GetFileName() const { return _fileName; }
- WebRtc_UWord16 GetWidth() const { return _width; }
- WebRtc_UWord16 GetHeight() const { return _height; }
- webrtc::VideoType GetType() const { return _type; }
- float GetFrameRate() const { return _frameRate; }
- int GetWidthHeight( VideoSize size);
-
- // Returns the filename with the path (including the leading slash) removed.
- std::string GetName() const;
-
- WebRtc_Word32 GetFrameLength() const;
-
-private:
- std::string _fileName;
- WebRtc_UWord16 _width;
- WebRtc_UWord16 _height;
- webrtc::VideoType _type;
- float _frameRate;
-};
-
-#endif // WEBRTC_MODULES_VIDEO_CODING_TEST_VIDEO_SOURCE_H_
-
diff --git a/trunk/src/modules/video_processing/main/OWNERS b/trunk/src/modules/video_processing/main/OWNERS
deleted file mode 100644
index 7183cf2..0000000
--- a/trunk/src/modules/video_processing/main/OWNERS
+++ /dev/null
@@ -1,4 +0,0 @@
-stefan@webrtc.org
-mikhal@webrtc.org
-marpan@webrtc.org
-henrik.lundin@webrtc.org
diff --git a/trunk/src/modules/video_processing/main/interface/video_processing.h b/trunk/src/modules/video_processing/main/interface/video_processing.h
deleted file mode 100644
index 512cace..0000000
--- a/trunk/src/modules/video_processing/main/interface/video_processing.h
+++ /dev/null
@@ -1,382 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * video_processing.h
- * This header file contains the API required for the video
- * processing module class.
- */
-
-
-#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_H
-#define WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_H
-
-#include "module.h"
-#include "module_common_types.h"
-#include "video_processing_defines.h"
-
-/**
- The module is largely intended to process video streams, except functionality
- provided by static functions which operate independent of previous frames. It
- is recommended, but not required that a unique instance be used for each
- concurrently processed stream. Similarly, it is recommended to call Reset()
- before switching to a new stream, but this is not absolutely required.
-
- The module provides basic thread safety by permitting only a single function to
- execute concurrently.
-*/
-
-namespace webrtc {
-
-class VideoProcessingModule : public Module
-{
-public:
- /**
- Structure to hold frame statistics. Populate it with GetFrameStats().
- */
- struct FrameStats
- {
- FrameStats() :
- mean(0),
- sum(0),
- numPixels(0),
- subSamplWidth(0),
- subSamplHeight(0)
- {
- memset(hist, 0, sizeof(hist));
- }
-
- WebRtc_UWord32 hist[256]; /**< Histogram of frame */
- WebRtc_UWord32 mean; /**< Mean value of frame */
- WebRtc_UWord32 sum; /**< Sum of frame */
- WebRtc_UWord32 numPixels; /**< Number of pixels */
- WebRtc_UWord8 subSamplWidth; /**< Subsampling rate of width in powers of 2 */
- WebRtc_UWord8 subSamplHeight; /**< Subsampling rate of height in powers of 2 */
- };
-
- /**
- Specifies the warning types returned by BrightnessDetection().
- */
- enum BrightnessWarning
- {
- kNoWarning, /**< Frame has acceptable brightness */
- kDarkWarning, /**< Frame is too dark */
- kBrightWarning /**< Frame is too bright */
- };
-
- /*
- Creates a VPM object.
-
- \param[in] id
- Unique identifier of this object.
-
- \return Pointer to a VPM object.
- */
- static VideoProcessingModule* Create(WebRtc_Word32 id);
-
- /**
- Destroys a VPM object.
-
- \param[in] module
- Pointer to the VPM object to destroy.
- */
- static void Destroy(VideoProcessingModule* module);
-
- /**
- Not supported.
- */
- virtual WebRtc_Word32 TimeUntilNextProcess() { return -1; }
-
- /**
- Not supported.
- */
- virtual WebRtc_Word32 Process() { return -1; }
-
- /**
- Resets all processing components to their initial states. This should be
- called whenever a new video stream is started.
- */
- virtual void Reset() = 0;
-
- /**
- Retrieves statistics for the input frame. This function must be used to
- prepare a FrameStats struct for use in certain VPM functions.
-
- \param[out] stats
- The frame statistics will be stored here on return.
-
- \param[in] frame
- Pointer to the video frame.
-
- \param[in] width
- Frame width in pixels.
-
- \param[in] height
- Frame height in pixels.
-
- \return 0 on success, -1 on failure.
- */
- static WebRtc_Word32 GetFrameStats(FrameStats& stats,
- const WebRtc_UWord8* frame,
- WebRtc_UWord32 width,
- WebRtc_UWord32 height);
-
- /**
- \overload
- */
- static WebRtc_Word32 GetFrameStats(FrameStats& stats,
- const VideoFrame& frame);
-
- /**
- Checks the validity of a FrameStats struct. Currently, valid implies only
- that is had changed from its initialized state.
-
- \param[in] stats
- Frame statistics.
-
- \return True on valid stats, false on invalid stats.
- */
- static bool ValidFrameStats(const FrameStats& stats);
-
- /**
- Returns a FrameStats struct to its intialized state.
-
- \param[in,out] stats
- Frame statistics.
- */
- static void ClearFrameStats(FrameStats& stats);
-
- /**
- Enhances the color of an image through a constant mapping. Only the
- chrominance is altered. Has a fixed-point implementation.
-
- \param[in,out] frame
- Pointer to the video frame.
-
- \param[in] width
- Frame width in pixels.
-
- \param[in] height
- Frame height in pixels.
-
- \return 0 on success, -1 on failure.
- */
- static WebRtc_Word32 ColorEnhancement(WebRtc_UWord8* frame,
- WebRtc_UWord32 width,
- WebRtc_UWord32 height);
-
- /**
- \overload
- */
- static WebRtc_Word32 ColorEnhancement(VideoFrame& frame);
-
- /**
- Increases/decreases the luminance value.
-
- \param[in,out] frame
- Pointer to the video frame buffer.
-
- \param[in] width
- Frame width in pixels.
-
- \param[in] height
- Frame height in pixels.
-
- \param[in] delta
- The amount to change the chrominance value of every single pixel.
- Can be < 0 also.
-
- \return 0 on success, -1 on failure.
- */
- static WebRtc_Word32 Brighten(WebRtc_UWord8* frame,
- int width, int height, int delta);
- /**
- \overload
- */
- static WebRtc_Word32 Brighten(VideoFrame& frame, int delta);
-
- /**
- Detects and removes camera flicker from a video stream. Every frame from the
- stream must be passed in. A frame will only be altered if flicker has been
- detected. Has a fixed-point implementation.
-
- \param[in,out] frame
- Pointer to the video frame.
-
- \param[in] width
- Frame width in pixels.
-
- \param[in] height
- Frame height in pixels.
-
- \param[in] timestamp
- Frame timestamp in 90 kHz format.
-
- \param[in,out] stats
- Frame statistics provided by GetFrameStats(). On return the stats will
- be reset to zero if the frame was altered. Call GetFrameStats() again
- if the statistics for the altered frame are required.
-
- \return 0 on success, -1 on failure.
- */
- virtual WebRtc_Word32 Deflickering(WebRtc_UWord8* frame,
- WebRtc_UWord32 width,
- WebRtc_UWord32 height,
- WebRtc_UWord32 timestamp,
- FrameStats& stats) = 0;
-
- /**
- \overload
- */
- virtual WebRtc_Word32 Deflickering(VideoFrame& frame,
- FrameStats& stats) = 0;
-
- /**
- Denoises a video frame. Every frame from the stream should be passed in.
- Has a fixed-point implementation.
-
- \param[in,out] frame
- Pointer to the video frame.
-
- \param[in] width
- Frame width in pixels.
-
- \param[in] height
- Frame height in pixels.
-
- \return The number of modified pixels on success, -1 on failure.
- */
- virtual WebRtc_Word32 Denoising(WebRtc_UWord8* frame,
- WebRtc_UWord32 width,
- WebRtc_UWord32 height) = 0;
-
- /**
- \overload
- */
- virtual WebRtc_Word32 Denoising(VideoFrame& frame) = 0;
-
- /**
- Detects if a video frame is excessively bright or dark. Returns a warning if
- this is the case. Multiple frames should be passed in before expecting a
- warning. Has a floating-point implementation.
-
- \param[in] frame
- Pointer to the video frame.
-
- \param[in] width
- Frame width in pixels.
-
- \param[in] height
- Frame height in pixels.
-
- \param[in] stats
- Frame statistics provided by GetFrameStats().
-
- \return A member of BrightnessWarning on success, -1 on error
- */
- virtual WebRtc_Word32 BrightnessDetection(const WebRtc_UWord8* frame,
- WebRtc_UWord32 width,
- WebRtc_UWord32 height,
- const FrameStats& stats) = 0;
-
- /**
- \overload
- */
- virtual WebRtc_Word32 BrightnessDetection(const VideoFrame& frame,
- const FrameStats& stats) = 0;
-
-
- /**
- The following functions refer to the pre-processor unit within VPM. The pre-processor
- perfoms spatial/temporal decimation and content analysis on the frames prior to encoding.
- */
-
- /**
- Enable/disable temporal decimation
-
- \param[in] enable when true, temporal decimation is enabled
- */
- virtual void EnableTemporalDecimation(bool enable) = 0;
-
- /**
- Set target resolution
-
- \param[in] width
- Target width
-
- \param[in] height
- Target height
-
- \param[in] frameRate
- Target frameRate
-
- \return VPM_OK on success, a negative value on error (see error codes)
-
- */
- virtual WebRtc_Word32 SetTargetResolution(WebRtc_UWord32 width, WebRtc_UWord32 height, WebRtc_UWord32 frameRate) = 0;
-
- /**
- Set max frame rate
- \param[in] maxFrameRate: maximum frame rate (limited to native frame rate)
-
- \return VPM_OK on success, a negative value on error (see error codes)
- */
- virtual WebRtc_Word32 SetMaxFrameRate(WebRtc_UWord32 maxFrameRate) = 0;
-
- /**
- Get decimated(target) frame rate
- */
- virtual WebRtc_UWord32 DecimatedFrameRate() = 0;
-
- /**
- Get decimated(target) frame width
- */
- virtual WebRtc_UWord32 DecimatedWidth() const = 0;
-
- /**
- Get decimated(target) frame height
- */
- virtual WebRtc_UWord32 DecimatedHeight() const = 0 ;
-
- /**
- Set the spatial resampling settings of the VPM: The resampler may either be disabled or one of the following:
- scaling to a close to target dimension followed by crop/pad
-
- \param[in] resamplingMode
- Set resampling mode (a member of VideoFrameResampling)
- */
- virtual void SetInputFrameResampleMode(VideoFrameResampling resamplingMode) = 0;
-
- /**
- Get Processed (decimated) frame
-
- \param[in] frame pointer to the video frame.
-
- \param[in] processedFrame pointer (double) to the processed frame
-
- \return VPM_OK on success, a negative value on error (see error codes)
- */
- virtual WebRtc_Word32 PreprocessFrame(const VideoFrame* frame, VideoFrame** processedFrame) = 0;
-
- /**
- Return content metrics for the last processed frame
- */
- virtual VideoContentMetrics* ContentMetrics() const = 0 ;
-
- /**
- Enable content analysis
- */
- virtual void EnableContentAnalysis(bool enable) = 0;
-
-};
-
-} //namespace
-
-#endif
diff --git a/trunk/src/modules/video_processing/main/interface/video_processing_defines.h b/trunk/src/modules/video_processing/main/interface/video_processing_defines.h
deleted file mode 100644
index d9bebd4..0000000
--- a/trunk/src/modules/video_processing/main/interface/video_processing_defines.h
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * video_processing_defines.h
- * This header file includes the definitions used in the video processor module
- */
-
-#ifndef WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_DEFINES_H
-#define WEBRTC_MODULES_INTERFACE_VIDEO_PROCESSING_DEFINES_H
-
-#include "typedefs.h"
-
-namespace webrtc {
-
-// Error codes
-#define VPM_OK 0
-#define VPM_GENERAL_ERROR -1
-#define VPM_MEMORY -2
-#define VPM_PARAMETER_ERROR -3
-#define VPM_SCALE_ERROR -4
-#define VPM_UNINITIALIZED -5
-#define VPM_UNIMPLEMENTED -6
-
-enum VideoFrameResampling
-{
- // TODO: Do we still need crop/pad?
- kNoRescaling, // disables rescaling
- kFastRescaling, // point
- kBiLinear, // bi-linear interpolation
- kBox, // Box inteprolation
-};
-
-} //namespace
-
-#endif
diff --git a/trunk/src/modules/video_processing/main/source/Android.mk b/trunk/src/modules/video_processing/main/source/Android.mk
deleted file mode 100644
index 829fa96..0000000
--- a/trunk/src/modules/video_processing/main/source/Android.mk
+++ /dev/null
@@ -1,59 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_video_processing
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := \
- brighten.cc \
- brightness_detection.cc \
- color_enhancement.cc \
- content_analysis.cc \
- deflickering.cc \
- denoising.cc \
- frame_preprocessor.cc \
- spatial_resampler.cc \
- video_decimator.cc \
- video_processing_impl.cc
-
-ifeq ($(TARGET_ARCH),x86)
-LOCAL_SRC_FILES += \
- content_analysis_sse2.cc
-endif
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../../.. \
- $(LOCAL_PATH)/../../../interface \
- $(LOCAL_PATH)/../../../audio_coding/main/interface \
- $(LOCAL_PATH)/../../../utility/interface \
- $(LOCAL_PATH)/../../../../common_audio/signal_processing/include \
- $(LOCAL_PATH)/../../../../common_video/vplib/main/interface \
- $(LOCAL_PATH)/../../../../system_wrappers/interface
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/video_processing/main/source/brighten.cc b/trunk/src/modules/video_processing/main/source/brighten.cc
deleted file mode 100644
index 51e4b6b..0000000
--- a/trunk/src/modules/video_processing/main/source/brighten.cc
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/video_processing/main/source/brighten.h"
-
-#include <cstdlib>
-
-#include "system_wrappers/interface/trace.h"
-
-namespace webrtc {
-namespace VideoProcessing {
-
-WebRtc_Word32 Brighten(WebRtc_UWord8* frame,
- int width, int height, int delta) {
- if (frame == NULL) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, -1,
- "Null frame pointer");
- return VPM_PARAMETER_ERROR;
- }
-
- if (width <= 0 || height <= 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, -1,
- "Invalid frame size");
- return VPM_PARAMETER_ERROR;
- }
-
- int numPixels = width * height;
-
- int lookUp[256];
- for (int i = 0; i < 256; i++) {
- int val = i + delta;
- lookUp[i] = ((((val < 0) ? 0 : val) > 255) ? 255 : val);
- }
-
- WebRtc_UWord8* tempPtr = frame;
-
- for (int i = 0; i < numPixels; i++) {
- *tempPtr = static_cast<WebRtc_UWord8>(lookUp[*tempPtr]);
- tempPtr++;
- }
- return VPM_OK;
-}
-
-} // namespace VideoProcessing
-} // namespace webrtc
diff --git a/trunk/src/modules/video_processing/main/source/brighten.h b/trunk/src/modules/video_processing/main/source/brighten.h
deleted file mode 100644
index b7e6fb7..0000000
--- a/trunk/src/modules/video_processing/main/source/brighten.h
+++ /dev/null
@@ -1,26 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTEN_H_
-#define MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTEN_H_
-
-#include "typedefs.h"
-#include "modules/video_processing/main/interface/video_processing.h"
-
-namespace webrtc {
-namespace VideoProcessing {
-
-WebRtc_Word32 Brighten(WebRtc_UWord8* frame,
- int width, int height, int delta);
-
-} // namespace VideoProcessing
-} // namespace webrtc
-
-#endif // MODULES_VIDEO_PROCESSING_MAIN_SOURCE_BRIGHTEN_H_
diff --git a/trunk/src/modules/video_processing/main/source/brightness_detection.cc b/trunk/src/modules/video_processing/main/source/brightness_detection.cc
deleted file mode 100644
index 6840df2..0000000
--- a/trunk/src/modules/video_processing/main/source/brightness_detection.cc
+++ /dev/null
@@ -1,195 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_processing.h"
-#include "brightness_detection.h"
-#include "trace.h"
-
-#include <math.h>
-
-namespace webrtc {
-
-VPMBrightnessDetection::VPMBrightnessDetection() :
- _id(0)
-{
- Reset();
-}
-
-VPMBrightnessDetection::~VPMBrightnessDetection()
-{
-}
-
-WebRtc_Word32
-VPMBrightnessDetection::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
- return VPM_OK;
-}
-
-void
-VPMBrightnessDetection::Reset()
-{
- _frameCntBright = 0;
- _frameCntDark = 0;
-}
-
-WebRtc_Word32
-VPMBrightnessDetection::ProcessFrame(const WebRtc_UWord8* frame,
- const WebRtc_UWord32 width,
- const WebRtc_UWord32 height,
- const VideoProcessingModule::FrameStats& stats)
-{
- if (frame == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id, "Null frame pointer");
- return VPM_PARAMETER_ERROR;
- }
-
- if (width == 0 || height == 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id, "Invalid frame size");
- return VPM_PARAMETER_ERROR;
- }
-
- if (!VideoProcessingModule::ValidFrameStats(stats))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id, "Invalid frame stats");
- return VPM_PARAMETER_ERROR;
- }
-
- const WebRtc_UWord8 frameCntAlarm = 2;
-
- // Get proportion in lowest bins
- WebRtc_UWord8 lowTh = 20;
- float propLow = 0;
- for (WebRtc_UWord32 i = 0; i < lowTh; i++)
- {
- propLow += stats.hist[i];
- }
- propLow /= stats.numPixels;
-
- // Get proportion in highest bins
- unsigned char highTh = 230;
- float propHigh = 0;
- for (WebRtc_UWord32 i = highTh; i < 256; i++)
- {
- propHigh += stats.hist[i];
- }
- propHigh /= stats.numPixels;
-
- if(propHigh < 0.4)
- {
- if (stats.mean < 90 || stats.mean > 170)
- {
- // Standard deviation of Y
- float stdY = 0;
- for (WebRtc_UWord32 h = 0; h < height; h += (1 << stats.subSamplHeight))
- {
- WebRtc_UWord32 row = h*width;
- for (WebRtc_UWord32 w = 0; w < width; w += (1 << stats.subSamplWidth))
- {
- stdY += (frame[w + row] - stats.mean) * (frame[w + row] - stats.mean);
- }
- }
- stdY = sqrt(stdY / stats.numPixels);
-
- // Get percentiles
- WebRtc_UWord32 sum = 0;
- WebRtc_UWord32 medianY = 140;
- WebRtc_UWord32 perc05 = 0;
- WebRtc_UWord32 perc95 = 255;
- float posPerc05 = stats.numPixels * 0.05f;
- float posMedian = stats.numPixels * 0.5f;
- float posPerc95 = stats.numPixels * 0.95f;
- for (WebRtc_UWord32 i = 0; i < 256; i++)
- {
- sum += stats.hist[i];
-
- if (sum < posPerc05)
- {
- perc05 = i; // 5th perc
- }
- if (sum < posMedian)
- {
- medianY = i; // 50th perc
- }
- if (sum < posPerc95)
- {
- perc95 = i; // 95th perc
- }
- else
- {
- break;
- }
- }
-
- // Check if image is too dark
- if ((stdY < 55) && (perc05 < 50))
- {
- if (medianY < 60 || stats.mean < 80 || perc95 < 130 || propLow > 0.20)
- {
- _frameCntDark++;
- }
- else
- {
- _frameCntDark = 0;
- }
- }
- else
- {
- _frameCntDark = 0;
- }
-
- // Check if image is too bright
- if ((stdY < 52) && (perc95 > 200) && (medianY > 160))
- {
- if (medianY > 185 || stats.mean > 185 || perc05 > 140 || propHigh > 0.25)
- {
- _frameCntBright++;
- }
- else
- {
- _frameCntBright = 0;
- }
- }
- else
- {
- _frameCntBright = 0;
- }
-
- }
- else
- {
- _frameCntDark = 0;
- _frameCntBright = 0;
- }
-
- }
- else
- {
- _frameCntBright++;
- _frameCntDark = 0;
- }
-
- if (_frameCntDark > frameCntAlarm)
- {
- return VideoProcessingModule::kDarkWarning;
- }
- else if (_frameCntBright > frameCntAlarm)
- {
- return VideoProcessingModule::kBrightWarning;
- }
- else
- {
- return VideoProcessingModule::kNoWarning;
- }
-}
-
-} //namespace
diff --git a/trunk/src/modules/video_processing/main/source/brightness_detection.h b/trunk/src/modules/video_processing/main/source/brightness_detection.h
deleted file mode 100644
index 7bed556..0000000
--- a/trunk/src/modules/video_processing/main/source/brightness_detection.h
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * brightness_detection.h
- */
-#ifndef VPM_BRIGHTNESS_DETECTION_H
-#define VPM_BRIGHTNESS_DETECTION_H
-
-#include "typedefs.h"
-#include "video_processing.h"
-
-namespace webrtc {
-
-class VPMBrightnessDetection
-{
-public:
- VPMBrightnessDetection();
- ~VPMBrightnessDetection();
-
- WebRtc_Word32 ChangeUniqueId(WebRtc_Word32 id);
-
- void Reset();
-
- WebRtc_Word32 ProcessFrame(const WebRtc_UWord8* frame,
- WebRtc_UWord32 width,
- WebRtc_UWord32 height,
- const VideoProcessingModule::FrameStats& stats);
-
-private:
- WebRtc_Word32 _id;
-
- WebRtc_UWord32 _frameCntBright;
- WebRtc_UWord32 _frameCntDark;
-};
-
-} //namespace
-
-#endif // VPM_BRIGHTNESS_DETECTION_H
diff --git a/trunk/src/modules/video_processing/main/source/color_enhancement.cc b/trunk/src/modules/video_processing/main/source/color_enhancement.cc
deleted file mode 100644
index 426596f..0000000
--- a/trunk/src/modules/video_processing/main/source/color_enhancement.cc
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "color_enhancement.h"
-#include "color_enhancement_private.h"
-#include "trace.h"
-#include <cstdlib> // NULL
-
-namespace webrtc {
-
-namespace VideoProcessing
-{
- WebRtc_Word32
- ColorEnhancement(WebRtc_UWord8* frame,
- const WebRtc_UWord32 width,
- const WebRtc_UWord32 height)
- {
- // pointers to U and V color pixels
- WebRtc_UWord8* ptrU;
- WebRtc_UWord8* ptrV;
- WebRtc_UWord8 tempChroma;
- const WebRtc_UWord32 numPixels = width * height;
-
-
- if (frame == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, -1, "Null frame pointer");
- return VPM_GENERAL_ERROR;
- }
-
- if (width == 0 || height == 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, -1, "Invalid frame size");
- return VPM_GENERAL_ERROR;
- }
-
- // set pointers to first U and V pixels
-
- // stream format:
- // | numPixels bytes luminance | numPixels/4 bytes chroma U | numPixels/4 chroma V |
-
- ptrU = frame + numPixels; // skip luminance
- ptrV = ptrU + (numPixels>>2);
-
- // loop through all chrominance pixels and modify color
- for (WebRtc_UWord32 ix = 0; ix < (numPixels>>2); ix++)
- {
- tempChroma = colorTable[*ptrU][*ptrV];
- *ptrV = colorTable[*ptrV][*ptrU];
- *ptrU = tempChroma;
-
- // increment pointers
- ptrU++;
- ptrV++;
- }
- return VPM_OK;
- }
-
-} //namespace
-
-} //namespace webrtc
diff --git a/trunk/src/modules/video_processing/main/source/color_enhancement.h b/trunk/src/modules/video_processing/main/source/color_enhancement.h
deleted file mode 100644
index 87fabc3..0000000
--- a/trunk/src/modules/video_processing/main/source/color_enhancement.h
+++ /dev/null
@@ -1,31 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * color_enhancement.h
- */
-#ifndef VPM_COLOR_ENHANCEMENT_H
-#define VPM_COLOR_ENHANCEMENT_H
-
-#include "typedefs.h"
-#include "video_processing.h"
-
-namespace webrtc {
-
-namespace VideoProcessing
-{
- WebRtc_Word32 ColorEnhancement(WebRtc_UWord8* frame,
- WebRtc_UWord32 width,
- WebRtc_UWord32 height);
-}
-
-} //namespace
-
-#endif // VPM_COLOR_ENHANCEMENT_H
diff --git a/trunk/src/modules/video_processing/main/source/color_enhancement_private.h b/trunk/src/modules/video_processing/main/source/color_enhancement_private.h
deleted file mode 100644
index b88fc1a..0000000
--- a/trunk/src/modules/video_processing/main/source/color_enhancement_private.h
+++ /dev/null
@@ -1,273 +0,0 @@
-#ifndef VPM_COLOR_ENHANCEMENT_PRIVATE_H
-#define VPM_COLOR_ENHANCEMENT_PRIVATE_H
-
-#include "typedefs.h"
-
-namespace webrtc {
-
-//Table created with Matlab script createTable.m
-//Usage:
-// Umod=colorTable[U][V]
-// Vmod=colorTable[V][U]
-static const WebRtc_UWord8 colorTable[256][256] = {
- {0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0},
- {1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1, 1},
- {2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2, 2},
- {3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3, 3},
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- {249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249, 249},
- {250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250, 250},
- {251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251, 251},
- {252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252, 252},
- {253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253, 253},
- {254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254, 254},
- {255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255, 255}
-};
-
-} //namespace
-
-#endif // VPM_COLOR_ENHANCEMENT_PRIVATE_H
diff --git a/trunk/src/modules/video_processing/main/source/content_analysis.cc b/trunk/src/modules/video_processing/main/source/content_analysis.cc
deleted file mode 100644
index 8ea319c..0000000
--- a/trunk/src/modules/video_processing/main/source/content_analysis.cc
+++ /dev/null
@@ -1,338 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-#include "content_analysis.h"
-#include "tick_util.h"
-#include "system_wrappers/interface/cpu_features_wrapper.h"
-
-#include <math.h>
-#include <stdlib.h>
-
-namespace webrtc {
-
-VPMContentAnalysis::VPMContentAnalysis(bool runtime_cpu_detection):
-_origFrame(NULL),
-_prevFrame(NULL),
-_width(0),
-_height(0),
-_skipNum(1),
-_border(8),
-_motionMagnitude(0.0f),
-_spatialPredErr(0.0f),
-_spatialPredErrH(0.0f),
-_spatialPredErrV(0.0f),
-_firstFrame(true),
-_CAInit(false),
-_cMetrics(NULL)
-{
- ComputeSpatialMetrics = &VPMContentAnalysis::ComputeSpatialMetrics_C;
- TemporalDiffMetric = &VPMContentAnalysis::TemporalDiffMetric_C;
-
- if (runtime_cpu_detection)
- {
-#if defined(WEBRTC_ARCH_X86_FAMILY)
- if (WebRtc_GetCPUInfo(kSSE2))
- {
- ComputeSpatialMetrics =
- &VPMContentAnalysis::ComputeSpatialMetrics_SSE2;
- TemporalDiffMetric = &VPMContentAnalysis::TemporalDiffMetric_SSE2;
- }
-#endif
- }
-
- Release();
-}
-
-VPMContentAnalysis::~VPMContentAnalysis()
-{
- Release();
-}
-
-
-VideoContentMetrics*
-VPMContentAnalysis::ComputeContentMetrics(const VideoFrame* inputFrame)
-{
- if (inputFrame == NULL)
- {
- return NULL;
- }
-
- // Init if needed (native dimension change)
- if (_width != inputFrame->Width() || _height != inputFrame->Height())
- {
- if (VPM_OK != Initialize((WebRtc_UWord16)inputFrame->Width(),
- (WebRtc_UWord16)inputFrame->Height()))
- {
- return NULL;
- }
- }
-
- _origFrame = inputFrame->Buffer();
-
- // compute spatial metrics: 3 spatial prediction errors
- (this->*ComputeSpatialMetrics)();
-
- // compute motion metrics
- if (_firstFrame == false)
- ComputeMotionMetrics();
-
- // saving current frame as previous one: Y only
- memcpy(_prevFrame, _origFrame, _width * _height);
-
- _firstFrame = false;
- _CAInit = true;
-
- return ContentMetrics();
-}
-
-WebRtc_Word32
-VPMContentAnalysis::Release()
-{
- if (_cMetrics != NULL)
- {
- delete _cMetrics;
- _cMetrics = NULL;
- }
-
- if (_prevFrame != NULL)
- {
- delete [] _prevFrame;
- _prevFrame = NULL;
- }
-
- _width = 0;
- _height = 0;
- _firstFrame = true;
-
- return VPM_OK;
-}
-
-WebRtc_Word32
-VPMContentAnalysis::Initialize(WebRtc_UWord16 width, WebRtc_UWord16 height)
-{
- _width = width;
- _height = height;
- _firstFrame = true;
-
- // skip parameter: # of skipped rows: for complexity reduction
- // temporal also currently uses it for column reduction.
- _skipNum = 1;
-
- // use skipNum = 2 for 4CIF, WHD
- if ( (_height >= 576) && (_width >= 704) )
- {
- _skipNum = 2;
- }
- // use skipNum = 4 for FULLL_HD images
- if ( (_height >= 1080) && (_width >= 1920) )
- {
- _skipNum = 4;
- }
-
- if (_cMetrics != NULL)
- {
- delete _cMetrics;
- }
-
- if (_prevFrame != NULL)
- {
- delete [] _prevFrame;
- }
-
- // Spatial Metrics don't work on a border of 8. Minimum processing
- // block size is 16 pixels. So make sure the width and height support this.
- if (_width <= 32 || _height <= 32)
- {
- _CAInit = false;
- return VPM_PARAMETER_ERROR;
- }
-
- _cMetrics = new VideoContentMetrics();
- if (_cMetrics == NULL)
- {
- return VPM_MEMORY;
- }
-
- _prevFrame = new WebRtc_UWord8[_width * _height] ; // Y only
- if (_prevFrame == NULL)
- {
- return VPM_MEMORY;
- }
-
- return VPM_OK;
-}
-
-
-// Compute motion metrics: magnitude over non-zero motion vectors,
-// and size of zero cluster
-WebRtc_Word32
-VPMContentAnalysis::ComputeMotionMetrics()
-{
-
- // Motion metrics: only one is derived from normalized
- // (MAD) temporal difference
- (this->*TemporalDiffMetric)();
-
- return VPM_OK;
-}
-
-// Normalized temporal difference (MAD): used as a motion level metric
-// Normalize MAD by spatial contrast: images with more contrast
-// (pixel variance) likely have larger temporal difference
-// To reduce complexity, we compute the metric for a reduced set of points.
-WebRtc_Word32
-VPMContentAnalysis::TemporalDiffMetric_C()
-{
- // size of original frame
- WebRtc_UWord16 sizei = _height;
- WebRtc_UWord16 sizej = _width;
-
- WebRtc_UWord32 tempDiffSum = 0;
- WebRtc_UWord32 pixelSum = 0;
- WebRtc_UWord64 pixelSqSum = 0;
-
- WebRtc_UWord32 numPixels = 0; // counter for # of pixels
- WebRtc_UWord32 ssn;
-
- const WebRtc_Word32 width_end = ((_width - 2*_border) & -16) + _border;
-
- for(WebRtc_UWord16 i = _border; i < sizei - _border; i += _skipNum)
- {
- for(WebRtc_UWord16 j = _border; j < width_end; j++)
- {
- numPixels += 1;
- ssn = i * sizej + j;
-
- WebRtc_UWord8 currPixel = _origFrame[ssn];
- WebRtc_UWord8 prevPixel = _prevFrame[ssn];
-
- tempDiffSum += (WebRtc_UWord32)
- abs((WebRtc_Word16)(currPixel - prevPixel));
- pixelSum += (WebRtc_UWord32) currPixel;
- pixelSqSum += (WebRtc_UWord64) (currPixel * currPixel);
- }
- }
-
- // default
- _motionMagnitude = 0.0f;
-
- if (tempDiffSum == 0)
- {
- return VPM_OK;
- }
-
- // normalize over all pixels
- float const tempDiffAvg = (float)tempDiffSum / (float)(numPixels);
- float const pixelSumAvg = (float)pixelSum / (float)(numPixels);
- float const pixelSqSumAvg = (float)pixelSqSum / (float)(numPixels);
- float contrast = pixelSqSumAvg - (pixelSumAvg * pixelSumAvg);
-
- if (contrast > 0.0)
- {
- contrast = sqrt(contrast);
- _motionMagnitude = tempDiffAvg/contrast;
- }
-
- return VPM_OK;
-
-}
-
-// Compute spatial metrics:
-// To reduce complexity, we compute the metric for a reduced set of points.
-// The spatial metrics are rough estimates of the prediction error cost for
-// each QM spatial mode: 2x2,1x2,2x1
-// The metrics are a simple estimate of the up-sampling prediction error,
-// estimated assuming sub-sampling for decimation (no filtering),
-// and up-sampling back up with simple bilinear interpolation.
-WebRtc_Word32
-VPMContentAnalysis::ComputeSpatialMetrics_C()
-{
- //size of original frame
- const WebRtc_UWord16 sizei = _height;
- const WebRtc_UWord16 sizej = _width;
-
- // pixel mean square average: used to normalize the spatial metrics
- WebRtc_UWord32 pixelMSA = 0;
-
- WebRtc_UWord32 spatialErrSum = 0;
- WebRtc_UWord32 spatialErrVSum = 0;
- WebRtc_UWord32 spatialErrHSum = 0;
-
- // make sure work section is a multiple of 16
- const WebRtc_UWord32 width_end = ((sizej - 2*_border) & -16) + _border;
-
- for(WebRtc_UWord16 i = _border; i < sizei - _border; i += _skipNum)
- {
- for(WebRtc_UWord16 j = _border; j < width_end; j++)
- {
- WebRtc_UWord32 ssn1,ssn2,ssn3,ssn4,ssn5;
-
- ssn1= i * sizej + j;
- ssn2 = (i + 1) * sizej + j; // bottom
- ssn3 = (i - 1) * sizej + j; // top
- ssn4 = i * sizej + j + 1; // right
- ssn5 = i * sizej + j - 1; // left
-
- WebRtc_UWord16 refPixel1 = _origFrame[ssn1] << 1;
- WebRtc_UWord16 refPixel2 = _origFrame[ssn1] << 2;
-
- WebRtc_UWord8 bottPixel = _origFrame[ssn2];
- WebRtc_UWord8 topPixel = _origFrame[ssn3];
- WebRtc_UWord8 rightPixel = _origFrame[ssn4];
- WebRtc_UWord8 leftPixel = _origFrame[ssn5];
-
- spatialErrSum += (WebRtc_UWord32) abs((WebRtc_Word16)(refPixel2
- - (WebRtc_UWord16)(bottPixel + topPixel
- + leftPixel + rightPixel)));
- spatialErrVSum += (WebRtc_UWord32) abs((WebRtc_Word16)(refPixel1
- - (WebRtc_UWord16)(bottPixel + topPixel)));
- spatialErrHSum += (WebRtc_UWord32) abs((WebRtc_Word16)(refPixel1
- - (WebRtc_UWord16)(leftPixel + rightPixel)));
-
- pixelMSA += _origFrame[ssn1];
- }
- }
-
- // normalize over all pixels
- const float spatialErr = (float)(spatialErrSum >> 2);
- const float spatialErrH = (float)(spatialErrHSum >> 1);
- const float spatialErrV = (float)(spatialErrVSum >> 1);
- const float norm = (float)pixelMSA;
-
- // 2X2:
- _spatialPredErr = spatialErr / norm;
-
- // 1X2:
- _spatialPredErrH = spatialErrH / norm;
-
- // 2X1:
- _spatialPredErrV = spatialErrV / norm;
-
- return VPM_OK;
-}
-
-VideoContentMetrics*
-VPMContentAnalysis::ContentMetrics()
-{
- if (_CAInit == false)
- {
- return NULL;
- }
-
- _cMetrics->spatial_pred_err = _spatialPredErr;
- _cMetrics->spatial_pred_err_h = _spatialPredErrH;
- _cMetrics->spatial_pred_err_v = _spatialPredErrV;
- // Motion metric: normalized temporal difference (MAD)
- _cMetrics->motion_magnitude = _motionMagnitude;
-
- return _cMetrics;
-
-}
-
-} // namespace
diff --git a/trunk/src/modules/video_processing/main/source/content_analysis.h b/trunk/src/modules/video_processing/main/source/content_analysis.h
deleted file mode 100644
index 588712a..0000000
--- a/trunk/src/modules/video_processing/main/source/content_analysis.h
+++ /dev/null
@@ -1,90 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef VPM_CONTENT_ANALYSIS_H
-#define VPM_CONTENT_ANALYSIS_H
-
-#include "typedefs.h"
-#include "module_common_types.h"
-#include "video_processing_defines.h"
-
-namespace webrtc {
-
-class VPMContentAnalysis
-{
-public:
- // When |runtime_cpu_detection| is true, runtime selection of an optimized
- // code path is allowed.
- VPMContentAnalysis(bool runtime_cpu_detection);
- ~VPMContentAnalysis();
-
- // Initialize ContentAnalysis - should be called prior to
- // extractContentFeature
- // Inputs: width, height
- // Return value: 0 if OK, negative value upon error
- WebRtc_Word32 Initialize(WebRtc_UWord16 width, WebRtc_UWord16 height);
-
- // Extract content Feature - main function of ContentAnalysis
- // Input: new frame
- // Return value: pointer to structure containing content Analysis
- // metrics or NULL value upon error
- VideoContentMetrics* ComputeContentMetrics(const VideoFrame* inputFrame);
-
- // Release all allocated memory
- // Output: 0 if OK, negative value upon error
- WebRtc_Word32 Release();
-
-private:
-
- // return motion metrics
- VideoContentMetrics* ContentMetrics();
-
- // Normalized temporal difference metric: for motion magnitude
- typedef WebRtc_Word32 (VPMContentAnalysis::*TemporalDiffMetricFunc)();
- TemporalDiffMetricFunc TemporalDiffMetric;
- WebRtc_Word32 TemporalDiffMetric_C();
-
- // Motion metric method: call 2 metrics (magnitude and size)
- WebRtc_Word32 ComputeMotionMetrics();
-
- // Spatial metric method: computes the 3 frame-average spatial
- // prediction errors (1x2,2x1,2x2)
- typedef WebRtc_Word32 (VPMContentAnalysis::*ComputeSpatialMetricsFunc)();
- ComputeSpatialMetricsFunc ComputeSpatialMetrics;
- WebRtc_Word32 ComputeSpatialMetrics_C();
-
-#if defined(WEBRTC_ARCH_X86_FAMILY)
- WebRtc_Word32 ComputeSpatialMetrics_SSE2();
- WebRtc_Word32 TemporalDiffMetric_SSE2();
-#endif
-
- const WebRtc_UWord8* _origFrame;
- WebRtc_UWord8* _prevFrame;
- WebRtc_UWord16 _width;
- WebRtc_UWord16 _height;
- WebRtc_UWord32 _skipNum;
- WebRtc_Word32 _border;
-
- // Content Metrics:
- // stores the local average of the metrics
- float _motionMagnitude; // motion class
- float _spatialPredErr; // spatial class
- float _spatialPredErrH; // spatial class
- float _spatialPredErrV; // spatial class
- bool _firstFrame;
- bool _CAInit;
-
- VideoContentMetrics* _cMetrics;
-
-}; // end of VPMContentAnalysis class definition
-
-} // namespace
-
-#endif
diff --git a/trunk/src/modules/video_processing/main/source/content_analysis_sse2.cc b/trunk/src/modules/video_processing/main/source/content_analysis_sse2.cc
deleted file mode 100644
index f505850..0000000
--- a/trunk/src/modules/video_processing/main/source/content_analysis_sse2.cc
+++ /dev/null
@@ -1,300 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "content_analysis.h"
-
-#include <emmintrin.h>
-#include <math.h>
-
-namespace webrtc {
-
-WebRtc_Word32
-VPMContentAnalysis::TemporalDiffMetric_SSE2()
-{
- WebRtc_UWord32 numPixels = 0; // counter for # of pixels
-
- const WebRtc_UWord8* imgBufO = _origFrame + _border*_width + _border;
- const WebRtc_UWord8* imgBufP = _prevFrame + _border*_width + _border;
-
- const WebRtc_Word32 width_end = ((_width - 2*_border) & -16) + _border;
-
- __m128i sad_64 = _mm_setzero_si128();
- __m128i sum_64 = _mm_setzero_si128();
- __m128i sqsum_64 = _mm_setzero_si128();
- const __m128i z = _mm_setzero_si128();
-
- for(WebRtc_UWord16 i = 0; i < (_height - 2*_border); i += _skipNum)
- {
- __m128i sqsum_32 = _mm_setzero_si128();
-
- const WebRtc_UWord8 *lineO = imgBufO;
- const WebRtc_UWord8 *lineP = imgBufP;
-
- // Work on 16 pixels at a time. For HD content with a width of 1920
- // this loop will run ~67 times (depending on border). Maximum for
- // abs(o-p) and sum(o) will be 255. _mm_sad_epu8 produces 2 64 bit
- // results which are then accumulated. There is no chance of
- // rollover for these two accumulators.
- // o*o will have a maximum of 255*255 = 65025. This will roll over
- // a 16 bit accumulator as 67*65025 > 65535, but will fit in a
- // 32 bit accumulator.
- for(WebRtc_UWord16 j = 0; j < width_end - _border; j += 16)
- {
- const __m128i o = _mm_loadu_si128((__m128i*)(lineO));
- const __m128i p = _mm_loadu_si128((__m128i*)(lineP));
-
- lineO += 16;
- lineP += 16;
-
- // abs pixel difference between frames
- sad_64 = _mm_add_epi64 (sad_64, _mm_sad_epu8(o, p));
-
- // sum of all pixels in frame
- sum_64 = _mm_add_epi64 (sum_64, _mm_sad_epu8(o, z));
-
- // squared sum of all pixels in frame
- const __m128i olo = _mm_unpacklo_epi8(o,z);
- const __m128i ohi = _mm_unpackhi_epi8(o,z);
-
- const __m128i sqsum_32_lo = _mm_madd_epi16(olo, olo);
- const __m128i sqsum_32_hi = _mm_madd_epi16(ohi, ohi);
-
- sqsum_32 = _mm_add_epi32(sqsum_32, sqsum_32_lo);
- sqsum_32 = _mm_add_epi32(sqsum_32, sqsum_32_hi);
- }
-
- // Add to 64 bit running sum as to not roll over.
- sqsum_64 = _mm_add_epi64(sqsum_64,
- _mm_add_epi64(_mm_unpackhi_epi32(sqsum_32,z),
- _mm_unpacklo_epi32(sqsum_32,z)));
-
- imgBufO += _width * _skipNum;
- imgBufP += _width * _skipNum;
- numPixels += (width_end - _border);
- }
-
- __m128i sad_final_128;
- __m128i sum_final_128;
- __m128i sqsum_final_128;
-
- // bring sums out of vector registers and into integer register
- // domain, summing them along the way
- _mm_store_si128 (&sad_final_128, sad_64);
- _mm_store_si128 (&sum_final_128, sum_64);
- _mm_store_si128 (&sqsum_final_128, sqsum_64);
-
- WebRtc_UWord64 *sad_final_64 =
- reinterpret_cast<WebRtc_UWord64*>(&sad_final_128);
- WebRtc_UWord64 *sum_final_64 =
- reinterpret_cast<WebRtc_UWord64*>(&sum_final_128);
- WebRtc_UWord64 *sqsum_final_64 =
- reinterpret_cast<WebRtc_UWord64*>(&sqsum_final_128);
-
- const WebRtc_UWord32 pixelSum = sum_final_64[0] + sum_final_64[1];
- const WebRtc_UWord64 pixelSqSum = sqsum_final_64[0] + sqsum_final_64[1];
- const WebRtc_UWord32 tempDiffSum = sad_final_64[0] + sad_final_64[1];
-
- // default
- _motionMagnitude = 0.0f;
-
- if (tempDiffSum == 0)
- {
- return VPM_OK;
- }
-
- // normalize over all pixels
- const float tempDiffAvg = (float)tempDiffSum / (float)(numPixels);
- const float pixelSumAvg = (float)pixelSum / (float)(numPixels);
- const float pixelSqSumAvg = (float)pixelSqSum / (float)(numPixels);
- float contrast = pixelSqSumAvg - (pixelSumAvg * pixelSumAvg);
-
- if (contrast > 0.0)
- {
- contrast = sqrt(contrast);
- _motionMagnitude = tempDiffAvg/contrast;
- }
-
- return VPM_OK;
-}
-
-WebRtc_Word32
-VPMContentAnalysis::ComputeSpatialMetrics_SSE2()
-{
- const WebRtc_UWord8* imgBuf = _origFrame + _border*_width;
- const WebRtc_Word32 width_end = ((_width - 2*_border) & -16) + _border;
-
- __m128i se_32 = _mm_setzero_si128();
- __m128i sev_32 = _mm_setzero_si128();
- __m128i seh_32 = _mm_setzero_si128();
- __m128i msa_32 = _mm_setzero_si128();
- const __m128i z = _mm_setzero_si128();
-
- // Error is accumulated as a 32 bit value. Looking at HD content with a
- // height of 1080 lines, or about 67 macro blocks. If the 16 bit row
- // value is maxed out at 65529 for every row, 65529*1080 = 70777800, which
- // will not roll over a 32 bit accumulator.
- // _skipNum is also used to reduce the number of rows
- for(WebRtc_Word32 i = 0; i < (_height - 2*_border); i += _skipNum)
- {
- __m128i se_16 = _mm_setzero_si128();
- __m128i sev_16 = _mm_setzero_si128();
- __m128i seh_16 = _mm_setzero_si128();
- __m128i msa_16 = _mm_setzero_si128();
-
- // Row error is accumulated as a 16 bit value. There are 8
- // accumulators. Max value of a 16 bit number is 65529. Looking
- // at HD content, 1080p, has a width of 1920, 120 macro blocks.
- // A mb at a time is processed at a time. Absolute max error at
- // a point would be abs(0-255+255+255+255) which equals 1020.
- // 120*1020 = 122400. The probability of hitting this is quite low
- // on well behaved content. A specially crafted image could roll over.
- // _border could also be adjusted to concentrate on just the center of
- // the images for an HD capture in order to reduce the possiblity of
- // rollover.
- const WebRtc_UWord8 *lineTop = imgBuf - _width + _border;
- const WebRtc_UWord8 *lineCen = imgBuf + _border;
- const WebRtc_UWord8 *lineBot = imgBuf + _width + _border;
-
- for(WebRtc_Word32 j = 0; j < width_end - _border; j += 16)
- {
- const __m128i t = _mm_loadu_si128((__m128i*)(lineTop));
- const __m128i l = _mm_loadu_si128((__m128i*)(lineCen - 1));
- const __m128i c = _mm_loadu_si128((__m128i*)(lineCen));
- const __m128i r = _mm_loadu_si128((__m128i*)(lineCen + 1));
- const __m128i b = _mm_loadu_si128((__m128i*)(lineBot));
-
- lineTop += 16;
- lineCen += 16;
- lineBot += 16;
-
- // center pixel unpacked
- __m128i clo = _mm_unpacklo_epi8(c,z);
- __m128i chi = _mm_unpackhi_epi8(c,z);
-
- // left right pixels unpacked and added together
- const __m128i lrlo = _mm_add_epi16(_mm_unpacklo_epi8(l,z),
- _mm_unpacklo_epi8(r,z));
- const __m128i lrhi = _mm_add_epi16(_mm_unpackhi_epi8(l,z),
- _mm_unpackhi_epi8(r,z));
-
- // top & bottom pixels unpacked and added together
- const __m128i tblo = _mm_add_epi16(_mm_unpacklo_epi8(t,z),
- _mm_unpacklo_epi8(b,z));
- const __m128i tbhi = _mm_add_epi16(_mm_unpackhi_epi8(t,z),
- _mm_unpackhi_epi8(b,z));
-
- // running sum of all pixels
- msa_16 = _mm_add_epi16(msa_16, _mm_add_epi16(chi, clo));
-
- clo = _mm_slli_epi16(clo, 1);
- chi = _mm_slli_epi16(chi, 1);
- const __m128i sevtlo = _mm_subs_epi16(clo, tblo);
- const __m128i sevthi = _mm_subs_epi16(chi, tbhi);
- const __m128i sehtlo = _mm_subs_epi16(clo, lrlo);
- const __m128i sehthi = _mm_subs_epi16(chi, lrhi);
-
- clo = _mm_slli_epi16(clo, 1);
- chi = _mm_slli_epi16(chi, 1);
- const __m128i setlo = _mm_subs_epi16(clo,
- _mm_add_epi16(lrlo, tblo));
- const __m128i sethi = _mm_subs_epi16(chi,
- _mm_add_epi16(lrhi, tbhi));
-
- // Add to 16 bit running sum
- se_16 = _mm_add_epi16(se_16,
- _mm_max_epi16(setlo,
- _mm_subs_epi16(z, setlo)));
- se_16 = _mm_add_epi16(se_16,
- _mm_max_epi16(sethi,
- _mm_subs_epi16(z, sethi)));
- sev_16 = _mm_add_epi16(sev_16,
- _mm_max_epi16(sevtlo,
- _mm_subs_epi16(z, sevtlo)));
- sev_16 = _mm_add_epi16(sev_16,
- _mm_max_epi16(sevthi,
- _mm_subs_epi16(z, sevthi)));
- seh_16 = _mm_add_epi16(seh_16,
- _mm_max_epi16(sehtlo,
- _mm_subs_epi16(z, sehtlo)));
- seh_16 = _mm_add_epi16(seh_16,
- _mm_max_epi16(sehthi,
- _mm_subs_epi16(z, sehthi)));
- }
-
- // Add to 32 bit running sum as to not roll over.
- se_32 = _mm_add_epi32(se_32,
- _mm_add_epi32(_mm_unpackhi_epi16(se_16,z),
- _mm_unpacklo_epi16(se_16,z)));
- sev_32 = _mm_add_epi32(sev_32,
- _mm_add_epi32(_mm_unpackhi_epi16(sev_16,z),
- _mm_unpacklo_epi16(sev_16,z)));
- seh_32 = _mm_add_epi32(seh_32,
- _mm_add_epi32(_mm_unpackhi_epi16(seh_16,z),
- _mm_unpacklo_epi16(seh_16,z)));
- msa_32 = _mm_add_epi32(msa_32,
- _mm_add_epi32(_mm_unpackhi_epi16(msa_16,z),
- _mm_unpacklo_epi16(msa_16,z)));
-
- imgBuf += _width * _skipNum;
- }
-
- __m128i se_128;
- __m128i sev_128;
- __m128i seh_128;
- __m128i msa_128;
-
- // bring sums out of vector registers and into integer register
- // domain, summing them along the way
- _mm_store_si128 (&se_128,
- _mm_add_epi64(_mm_unpackhi_epi32(se_32,z),
- _mm_unpacklo_epi32(se_32,z)));
- _mm_store_si128 (&sev_128,
- _mm_add_epi64(_mm_unpackhi_epi32(sev_32,z),
- _mm_unpacklo_epi32(sev_32,z)));
- _mm_store_si128 (&seh_128,
- _mm_add_epi64(_mm_unpackhi_epi32(seh_32,z),
- _mm_unpacklo_epi32(seh_32,z)));
- _mm_store_si128 (&msa_128,
- _mm_add_epi64(_mm_unpackhi_epi32(msa_32,z),
- _mm_unpacklo_epi32(msa_32,z)));
-
- WebRtc_UWord64 *se_64 =
- reinterpret_cast<WebRtc_UWord64*>(&se_128);
- WebRtc_UWord64 *sev_64 =
- reinterpret_cast<WebRtc_UWord64*>(&sev_128);
- WebRtc_UWord64 *seh_64 =
- reinterpret_cast<WebRtc_UWord64*>(&seh_128);
- WebRtc_UWord64 *msa_64 =
- reinterpret_cast<WebRtc_UWord64*>(&msa_128);
-
- const WebRtc_UWord32 spatialErrSum = se_64[0] + se_64[1];
- const WebRtc_UWord32 spatialErrVSum = sev_64[0] + sev_64[1];
- const WebRtc_UWord32 spatialErrHSum = seh_64[0] + seh_64[1];
- const WebRtc_UWord32 pixelMSA = msa_64[0] + msa_64[1];
-
- // normalize over all pixels
- const float spatialErr = (float)(spatialErrSum >> 2);
- const float spatialErrH = (float)(spatialErrHSum >> 1);
- const float spatialErrV = (float)(spatialErrVSum >> 1);
- const float norm = (float)pixelMSA;
-
- // 2X2:
- _spatialPredErr = spatialErr / norm;
-
- // 1X2:
- _spatialPredErrH = spatialErrH / norm;
-
- // 2X1:
- _spatialPredErrV = spatialErrV / norm;
-
- return VPM_OK;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_processing/main/source/deflickering.cc b/trunk/src/modules/video_processing/main/source/deflickering.cc
deleted file mode 100644
index d0b8d3b..0000000
--- a/trunk/src/modules/video_processing/main/source/deflickering.cc
+++ /dev/null
@@ -1,445 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-#include <math.h>
-#include <stdlib.h>
-
-#include "deflickering.h"
-#include "trace.h"
-#include "signal_processing_library.h"
-#include "sort.h"
-
-namespace webrtc {
-
-// Detection constants
-enum { kFrequencyDeviation = 39 }; // (Q4) Maximum allowed deviation for detection
-enum { kMinFrequencyToDetect = 32 }; // (Q4) Minimum frequency that can be detected
-enum { kNumFlickerBeforeDetect = 2 }; // Number of flickers before we accept detection
-enum { kMeanValueScaling = 4 }; // (Q4) In power of 2
-enum { kZeroCrossingDeadzone = 10 }; // Deadzone region in terms of pixel values
-
-// Deflickering constants
-// Compute the quantiles over 1 / DownsamplingFactor of the image.
-enum { kDownsamplingFactor = 8 };
-enum { kLog2OfDownsamplingFactor = 3 };
-
-// To generate in Matlab:
-// >> probUW16 = round(2^11 * [0.05,0.1,0.2,0.3,0.4,0.5,0.6,0.7,0.8,0.9,0.95,0.97]);
-// >> fprintf('%d, ', probUW16)
-// Resolution reduced to avoid overflow when multiplying with the (potentially) large
-// number of pixels.
-const WebRtc_UWord16 VPMDeflickering::_probUW16[kNumProbs] =
- {102, 205, 410, 614, 819, 1024, 1229, 1434, 1638, 1843, 1946, 1987}; // <Q11>
-
-// To generate in Matlab:
-// >> numQuants = 14; maxOnlyLength = 5;
-// >> weightUW16 = round(2^15 * [linspace(0.5, 1.0, numQuants - maxOnlyLength)]);
-// >> fprintf('%d, %d,\n ', weightUW16);
-const WebRtc_UWord16 VPMDeflickering::_weightUW16[kNumQuants - kMaxOnlyLength] =
- {16384, 18432, 20480, 22528, 24576, 26624, 28672, 30720, 32768}; // <Q15>
-
-VPMDeflickering::VPMDeflickering() :
- _id(0)
-{
- Reset();
-}
-
-VPMDeflickering::~VPMDeflickering()
-{
-}
-
-WebRtc_Word32
-VPMDeflickering::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
- return 0;
-}
-
-void
-VPMDeflickering::Reset()
-{
- _meanBufferLength = 0;
- _detectionState = 0;
- _frameRate = 0;
-
- memset(_meanBuffer, 0, sizeof(WebRtc_Word32) * kMeanBufferLength);
- memset(_timestampBuffer, 0, sizeof(WebRtc_Word32) * kMeanBufferLength);
-
- // Initialize the history with a uniformly distributed histogram
- _quantHistUW8[0][0] = 0;
- _quantHistUW8[0][kNumQuants - 1] = 255;
- for (WebRtc_Word32 i = 0; i < kNumProbs; i++)
- {
- _quantHistUW8[0][i + 1] = static_cast<WebRtc_UWord8>((WEBRTC_SPL_UMUL_16_16(
- _probUW16[i], 255) + (1 << 10)) >> 11); // Unsigned round. <Q0>
- }
-
- for (WebRtc_Word32 i = 1; i < kFrameHistorySize; i++)
- {
- memcpy(_quantHistUW8[i], _quantHistUW8[0], sizeof(WebRtc_UWord8) * kNumQuants);
- }
-}
-
-WebRtc_Word32
-VPMDeflickering::ProcessFrame(WebRtc_UWord8* frame,
- const WebRtc_UWord32 width,
- const WebRtc_UWord32 height,
- const WebRtc_UWord32 timestamp,
- VideoProcessingModule::FrameStats& stats)
-{
- WebRtc_UWord32 frameMemory;
- WebRtc_UWord8 quantUW8[kNumQuants];
- WebRtc_UWord8 maxQuantUW8[kNumQuants];
- WebRtc_UWord8 minQuantUW8[kNumQuants];
- WebRtc_UWord16 targetQuantUW16[kNumQuants];
- WebRtc_UWord16 incrementUW16;
- WebRtc_UWord8 mapUW8[256];
-
- WebRtc_UWord16 tmpUW16;
- WebRtc_UWord32 tmpUW32;
-
- if (frame == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id, "Null frame pointer");
- return VPM_GENERAL_ERROR;
- }
-
- // Stricter height check due to subsampling size calculation below.
- if (width == 0 || height < 2)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id, "Invalid frame size");
- return VPM_GENERAL_ERROR;
- }
-
- if (!VideoProcessingModule::ValidFrameStats(stats))
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id, "Invalid frame stats");
- return VPM_GENERAL_ERROR;
- }
-
- if (PreDetection(timestamp, stats) == -1)
- {
- return VPM_GENERAL_ERROR;
- }
-
- // Flicker detection
- WebRtc_Word32 detFlicker = DetectFlicker();
- if (detFlicker < 0)
- { // Error
- return VPM_GENERAL_ERROR;
- }
- else if (detFlicker != 1)
- {
- return 0;
- }
-
- // Size of luminance component
- const WebRtc_UWord32 ySize = height * width;
-
- const WebRtc_UWord32 ySubSize = width * (((height - 1) >>
- kLog2OfDownsamplingFactor) + 1);
- WebRtc_UWord8* ySorted = new WebRtc_UWord8[ySubSize];
- WebRtc_UWord32 sortRowIdx = 0;
- for (WebRtc_UWord32 i = 0; i < height; i += kDownsamplingFactor)
- {
- memcpy(ySorted + sortRowIdx * width, frame + i * width, width);
- sortRowIdx++;
- }
-
- webrtc::Sort(ySorted, ySubSize, webrtc::TYPE_UWord8);
-
- WebRtc_UWord32 probIdxUW32 = 0;
- quantUW8[0] = 0;
- quantUW8[kNumQuants - 1] = 255;
-
- // Ensure we won't get an overflow below.
- // In practice, the number of subsampled pixels will not become this large.
- if (ySubSize > (1 << 21) - 1)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id,
- "Subsampled number of pixels too large");
- return -1;
- }
-
- for (WebRtc_Word32 i = 0; i < kNumProbs; i++)
- {
- probIdxUW32 = WEBRTC_SPL_UMUL_32_16(ySubSize, _probUW16[i]) >> 11; // <Q0>
- quantUW8[i + 1] = ySorted[probIdxUW32];
- }
-
- delete [] ySorted;
- ySorted = NULL;
-
- // Shift history for new frame.
- memmove(_quantHistUW8[1], _quantHistUW8[0], (kFrameHistorySize - 1) * kNumQuants *
- sizeof(WebRtc_UWord8));
- // Store current frame in history.
- memcpy(_quantHistUW8[0], quantUW8, kNumQuants * sizeof(WebRtc_UWord8));
-
- // We use a frame memory equal to the ceiling of half the frame rate to ensure we
- // capture an entire period of flicker.
- frameMemory = (_frameRate + (1 << 5)) >> 5; // Unsigned ceiling. <Q0>
- // _frameRate in Q4.
- if (frameMemory > kFrameHistorySize)
- {
- frameMemory = kFrameHistorySize;
- }
-
- // Get maximum and minimum.
- for (WebRtc_Word32 i = 0; i < kNumQuants; i++)
- {
- maxQuantUW8[i] = 0;
- minQuantUW8[i] = 255;
- for (WebRtc_UWord32 j = 0; j < frameMemory; j++)
- {
- if (_quantHistUW8[j][i] > maxQuantUW8[i])
- {
- maxQuantUW8[i] = _quantHistUW8[j][i];
- }
-
- if (_quantHistUW8[j][i] < minQuantUW8[i])
- {
- minQuantUW8[i] = _quantHistUW8[j][i];
- }
- }
- }
-
- // Get target quantiles.
- for (WebRtc_Word32 i = 0; i < kNumQuants - kMaxOnlyLength; i++)
- {
- targetQuantUW16[i] = static_cast<WebRtc_UWord16>((WEBRTC_SPL_UMUL_16_16(
- _weightUW16[i], maxQuantUW8[i]) + WEBRTC_SPL_UMUL_16_16((1 << 15) -
- _weightUW16[i], minQuantUW8[i])) >> 8); // <Q7>
- }
-
- for (WebRtc_Word32 i = kNumQuants - kMaxOnlyLength; i < kNumQuants; i++)
- {
- targetQuantUW16[i] = ((WebRtc_UWord16)maxQuantUW8[i]) << 7;
- }
-
- // Compute the map from input to output pixels.
- WebRtc_UWord16 mapUW16; // <Q7>
- for (WebRtc_Word32 i = 1; i < kNumQuants; i++)
- {
- // As quant and targetQuant are limited to UWord8, we're safe to use Q7 here.
- tmpUW32 = static_cast<WebRtc_UWord32>(targetQuantUW16[i] -
- targetQuantUW16[i - 1]); // <Q7>
- tmpUW16 = static_cast<WebRtc_UWord16>(quantUW8[i] - quantUW8[i - 1]); // <Q0>
-
- if (tmpUW16 > 0)
- {
- incrementUW16 = static_cast<WebRtc_UWord16>(WebRtcSpl_DivU32U16(tmpUW32,
- tmpUW16)); // <Q7>
- }
- else
- {
- // The value is irrelevant; the loop below will only iterate once.
- incrementUW16 = 0;
- }
-
- mapUW16 = targetQuantUW16[i - 1];
- for (WebRtc_UWord32 j = quantUW8[i - 1]; j < (WebRtc_UWord32)(quantUW8[i] + 1); j++)
- {
- mapUW8[j] = (WebRtc_UWord8)((mapUW16 + (1 << 6)) >> 7); // Unsigned round. <Q0>
- mapUW16 += incrementUW16;
- }
- }
-
- // Map to the output frame.
- for (WebRtc_UWord32 i = 0; i < ySize; i++)
- {
- frame[i] = mapUW8[frame[i]];
- }
-
- // Frame was altered, so reset stats.
- VideoProcessingModule::ClearFrameStats(stats);
-
- return 0;
-}
-
-/**
- Performs some pre-detection operations. Must be called before
- DetectFlicker().
-
- \param[in] timestamp Timestamp of the current frame.
- \param[in] stats Statistics of the current frame.
-
- \return 0: Success\n
- 2: Detection not possible due to flickering frequency too close to
- zero.\n
- -1: Error
-*/
-WebRtc_Word32
-VPMDeflickering::PreDetection(const WebRtc_UWord32 timestamp,
- const VideoProcessingModule::FrameStats& stats)
-{
- WebRtc_Word32 meanVal; // Mean value of frame (Q4)
- WebRtc_UWord32 frameRate = 0;
- WebRtc_Word32 meanBufferLength; // Temp variable
-
- meanVal = ((stats.sum << kMeanValueScaling) / stats.numPixels);
- /* Update mean value buffer.
- * This should be done even though we might end up in an unreliable detection.
- */
- memmove(_meanBuffer + 1, _meanBuffer, (kMeanBufferLength - 1) * sizeof(WebRtc_Word32));
- _meanBuffer[0] = meanVal;
-
- /* Update timestamp buffer.
- * This should be done even though we might end up in an unreliable detection.
- */
- memmove(_timestampBuffer + 1, _timestampBuffer, (kMeanBufferLength - 1) *
- sizeof(WebRtc_UWord32));
- _timestampBuffer[0] = timestamp;
-
- /* Compute current frame rate (Q4) */
- if (_timestampBuffer[kMeanBufferLength - 1] != 0)
- {
- frameRate = ((90000 << 4) * (kMeanBufferLength - 1));
- frameRate /= (_timestampBuffer[0] - _timestampBuffer[kMeanBufferLength - 1]);
- }else if (_timestampBuffer[1] != 0)
- {
- frameRate = (90000 << 4) / (_timestampBuffer[0] - _timestampBuffer[1]);
- }
-
- /* Determine required size of mean value buffer (_meanBufferLength) */
- if (frameRate == 0) {
- meanBufferLength = 1;
- }
- else {
- meanBufferLength = (kNumFlickerBeforeDetect * frameRate) / kMinFrequencyToDetect;
- }
- /* Sanity check of buffer length */
- if (meanBufferLength >= kMeanBufferLength)
- {
- /* Too long buffer. The flickering frequency is too close to zero, which
- * makes the estimation unreliable.
- */
- _meanBufferLength = 0;
- return 2;
- }
- _meanBufferLength = meanBufferLength;
-
- if ((_timestampBuffer[_meanBufferLength - 1] != 0) && (_meanBufferLength != 1))
- {
- frameRate = ((90000 << 4) * (_meanBufferLength - 1));
- frameRate /= (_timestampBuffer[0] - _timestampBuffer[_meanBufferLength - 1]);
- }else if (_timestampBuffer[1] != 0)
- {
- frameRate = (90000 << 4) / (_timestampBuffer[0] - _timestampBuffer[1]);
- }
- _frameRate = frameRate;
-
- return 0;
-}
-
-/**
- This function detects flicker in the video stream. As a side effect the mean value
- buffer is updated with the new mean value.
-
- \return 0: No flickering detected\n
- 1: Flickering detected\n
- 2: Detection not possible due to unreliable frequency interval
- -1: Error
-*/
-WebRtc_Word32 VPMDeflickering::DetectFlicker()
-{
- /* Local variables */
- WebRtc_UWord32 i;
- WebRtc_Word32 freqEst; // (Q4) Frequency estimate to base detection upon
- WebRtc_Word32 retVal = -1;
-
- /* Sanity check for _meanBufferLength */
- if (_meanBufferLength < 2)
- {
- /* Not possible to estimate frequency */
- return(2);
- }
- /* Count zero crossings with a dead zone to be robust against noise.
- * If the noise std is 2 pixel this corresponds to about 95% confidence interval.
- */
- WebRtc_Word32 deadzone = (kZeroCrossingDeadzone << kMeanValueScaling); // Q4
- WebRtc_Word32 meanOfBuffer = 0; // Mean value of mean value buffer
- WebRtc_Word32 numZeros = 0; // Number of zeros that cross the deadzone
- WebRtc_Word32 cntState = 0; // State variable for zero crossing regions
- WebRtc_Word32 cntStateOld = 0; // Previous state variable for zero crossing regions
-
- for (i = 0; i < _meanBufferLength; i++)
- {
- meanOfBuffer += _meanBuffer[i];
- }
- meanOfBuffer += (_meanBufferLength >> 1); // Rounding, not truncation
- meanOfBuffer /= _meanBufferLength;
-
- /* Count zero crossings */
- cntStateOld = (_meanBuffer[0] >= (meanOfBuffer + deadzone));
- cntStateOld -= (_meanBuffer[0] <= (meanOfBuffer - deadzone));
- for (i = 1; i < _meanBufferLength; i++)
- {
- cntState = (_meanBuffer[i] >= (meanOfBuffer + deadzone));
- cntState -= (_meanBuffer[i] <= (meanOfBuffer - deadzone));
- if (cntStateOld == 0)
- {
- cntStateOld = -cntState;
- }
- if (((cntState + cntStateOld) == 0) && (cntState != 0))
- {
- numZeros++;
- cntStateOld = cntState;
- }
- }
- /* END count zero crossings */
-
- /* Frequency estimation according to:
- * freqEst = numZeros * frameRate / 2 / _meanBufferLength;
- *
- * Resolution is set to Q4
- */
- freqEst = ((numZeros * 90000) << 3);
- freqEst /= (_timestampBuffer[0] - _timestampBuffer[_meanBufferLength - 1]);
-
- /* Translate frequency estimate to regions close to 100 and 120 Hz */
- WebRtc_UWord8 freqState = 0; // Current translation state;
- // (0) Not in interval,
- // (1) Within valid interval,
- // (2) Out of range
- WebRtc_Word32 freqAlias = freqEst;
- if (freqEst > kMinFrequencyToDetect)
- {
- WebRtc_UWord8 aliasState = 1;
- while(freqState == 0)
- {
- /* Increase frequency */
- freqAlias += (aliasState * _frameRate);
- freqAlias += ((freqEst << 1) * (1 - (aliasState << 1)));
- /* Compute state */
- freqState = (abs(freqAlias - (100 << 4)) <= kFrequencyDeviation);
- freqState += (abs(freqAlias - (120 << 4)) <= kFrequencyDeviation);
- freqState += 2 * (freqAlias > ((120 << 4) + kFrequencyDeviation));
- /* Switch alias state */
- aliasState++;
- aliasState &= 0x01;
- }
- }
- /* Is frequency estimate within detection region? */
- if (freqState == 1)
- {
- retVal = 1;
- }else if (freqState == 0)
- {
- retVal = 2;
- }else
- {
- retVal = 0;
- }
- return retVal;
-}
-
-} //namespace
diff --git a/trunk/src/modules/video_processing/main/source/deflickering.h b/trunk/src/modules/video_processing/main/source/deflickering.h
deleted file mode 100644
index ee5f90d..0000000
--- a/trunk/src/modules/video_processing/main/source/deflickering.h
+++ /dev/null
@@ -1,69 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * deflickering.h
- */
-
-#ifndef VPM_DEFLICKERING_H
-#define VPM_DEFLICKERING_H
-
-#include "typedefs.h"
-#include "video_processing.h"
-
-#include <cstring> // NULL
-
-namespace webrtc {
-
-class VPMDeflickering
-{
-public:
- VPMDeflickering();
- ~VPMDeflickering();
-
- WebRtc_Word32 ChangeUniqueId(WebRtc_Word32 id);
-
- void Reset();
-
- WebRtc_Word32 ProcessFrame(WebRtc_UWord8* frame,
- WebRtc_UWord32 width,
- WebRtc_UWord32 height,
- WebRtc_UWord32 timestamp,
- VideoProcessingModule::FrameStats& stats);
-private:
- WebRtc_Word32 PreDetection(WebRtc_UWord32 timestamp,
- const VideoProcessingModule::FrameStats& stats);
-
- WebRtc_Word32 DetectFlicker();
-
- enum { kMeanBufferLength = 32 };
- enum { kFrameHistorySize = 15 };
- enum { kNumProbs = 12 };
- enum { kNumQuants = kNumProbs + 2 };
- enum { kMaxOnlyLength = 5 };
-
- WebRtc_Word32 _id;
-
- WebRtc_UWord32 _meanBufferLength;
- WebRtc_UWord8 _detectionState; // 0: No flickering
- // 1: Flickering detected
- // 2: In flickering
- WebRtc_Word32 _meanBuffer[kMeanBufferLength];
- WebRtc_UWord32 _timestampBuffer[kMeanBufferLength];
- WebRtc_UWord32 _frameRate;
- static const WebRtc_UWord16 _probUW16[kNumProbs];
- static const WebRtc_UWord16 _weightUW16[kNumQuants - kMaxOnlyLength];
- WebRtc_UWord8 _quantHistUW8[kFrameHistorySize][kNumQuants];
-};
-
-} //namespace
-
-#endif // VPM_DEFLICKERING_H
-
diff --git a/trunk/src/modules/video_processing/main/source/denoising.cc b/trunk/src/modules/video_processing/main/source/denoising.cc
deleted file mode 100644
index d8931c9..0000000
--- a/trunk/src/modules/video_processing/main/source/denoising.cc
+++ /dev/null
@@ -1,180 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "denoising.h"
-#include "trace.h"
-
-#include <cstring>
-
-namespace webrtc {
-
-enum { kSubsamplingTime = 0 }; // Down-sampling in time (unit: number of frames)
-enum { kSubsamplingWidth = 0 }; // Sub-sampling in width (unit: power of 2)
-enum { kSubsamplingHeight = 0 }; // Sub-sampling in height (unit: power of 2)
-enum { kDenoiseFiltParam = 179 }; // (Q8) De-noising filter parameter
-enum { kDenoiseFiltParamRec = 77 }; // (Q8) 1 - filter parameter
-enum { kDenoiseThreshold = 19200 }; // (Q8) De-noising threshold level
-
-VPMDenoising::VPMDenoising() :
- _id(0),
- _moment1(NULL),
- _moment2(NULL)
-{
- Reset();
-}
-
-VPMDenoising::~VPMDenoising()
-{
- if (_moment1)
- {
- delete [] _moment1;
- _moment1 = NULL;
- }
-
- if (_moment2)
- {
- delete [] _moment2;
- _moment2 = NULL;
- }
-}
-
-WebRtc_Word32
-VPMDenoising::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
- return VPM_OK;
-}
-
-void
-VPMDenoising::Reset()
-{
- _frameSize = 0;
- _denoiseFrameCnt = 0;
-
- if (_moment1)
- {
- delete [] _moment1;
- _moment1 = NULL;
- }
-
- if (_moment2)
- {
- delete [] _moment2;
- _moment2 = NULL;
- }
-}
-
-WebRtc_Word32
-VPMDenoising::ProcessFrame(WebRtc_UWord8* frame,
- const WebRtc_UWord32 width,
- const WebRtc_UWord32 height)
-{
- WebRtc_Word32 thevar;
- WebRtc_UWord32 k;
- WebRtc_UWord32 jsub, ksub;
- WebRtc_Word32 diff0;
- WebRtc_UWord32 tmpMoment1;
- WebRtc_UWord32 tmpMoment2;
- WebRtc_UWord32 tmp;
- WebRtc_Word32 numPixelsChanged = 0;
-
- if (frame == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id, "Null frame pointer");
- return VPM_GENERAL_ERROR;
- }
-
- if (width == 0 || height == 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, _id, "Invalid frame size");
- return VPM_GENERAL_ERROR;
- }
-
- /* Size of luminance component */
- const WebRtc_UWord32 ysize = height * width;
-
- /* Initialization */
- if (ysize != _frameSize)
- {
- delete [] _moment1;
- _moment1 = NULL;
-
- delete [] _moment2;
- _moment2 = NULL;
- }
- _frameSize = ysize;
-
- if (!_moment1)
- {
- _moment1 = new WebRtc_UWord32[ysize];
- memset(_moment1, 0, sizeof(WebRtc_UWord32)*ysize);
- }
-
- if (!_moment2)
- {
- _moment2 = new WebRtc_UWord32[ysize];
- memset(_moment2, 0, sizeof(WebRtc_UWord32)*ysize);
- }
-
- /* Apply de-noising on each pixel, but update variance sub-sampled */
- for (WebRtc_UWord32 i = 0; i < height; i++)
- { // Collect over height
- k = i * width;
- ksub = ((i >> kSubsamplingHeight) << kSubsamplingHeight) * width;
- for (WebRtc_UWord32 j = 0; j < width; j++)
- { // Collect over width
- jsub = ((j >> kSubsamplingWidth) << kSubsamplingWidth);
- /* Update mean value for every pixel and every frame */
- tmpMoment1 = _moment1[k + j];
- tmpMoment1 *= kDenoiseFiltParam; // Q16
- tmpMoment1 += ((kDenoiseFiltParamRec * ((WebRtc_UWord32)frame[k + j])) << 8);
- tmpMoment1 >>= 8; // Q8
- _moment1[k + j] = tmpMoment1;
-
- tmpMoment2 = _moment2[ksub + jsub];
- if ((ksub == k) && (jsub == j) && (_denoiseFrameCnt == 0))
- {
- tmp = ((WebRtc_UWord32)frame[k + j] * (WebRtc_UWord32)frame[k + j]);
- tmpMoment2 *= kDenoiseFiltParam; // Q16
- tmpMoment2 += ((kDenoiseFiltParamRec * tmp)<<8);
- tmpMoment2 >>= 8; // Q8
- }
- _moment2[k + j] = tmpMoment2;
- /* Current event = deviation from mean value */
- diff0 = ((WebRtc_Word32)frame[k + j] << 8) - _moment1[k + j];
- /* Recent events = variance (variations over time) */
- thevar = _moment2[k + j];
- thevar -= ((_moment1[k + j] * _moment1[k + j]) >> 8);
- /***************************************************************************
- * De-noising criteria, i.e., when should we replace a pixel by its mean
- *
- * 1) recent events are minor
- * 2) current events are minor
- ***************************************************************************/
- if ((thevar < kDenoiseThreshold)
- && ((diff0 * diff0 >> 8) < kDenoiseThreshold))
- { // Replace with mean
- frame[k + j] = (WebRtc_UWord8)(_moment1[k + j] >> 8);
- numPixelsChanged++;
- }
- }
- }
-
- /* Update frame counter */
- _denoiseFrameCnt++;
- if (_denoiseFrameCnt > kSubsamplingTime)
- {
- _denoiseFrameCnt = 0;
- }
-
- return numPixelsChanged;
-}
-
-} //namespace
diff --git a/trunk/src/modules/video_processing/main/source/denoising.h b/trunk/src/modules/video_processing/main/source/denoising.h
deleted file mode 100644
index f53157c..0000000
--- a/trunk/src/modules/video_processing/main/source/denoising.h
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * denoising.h
- */
-#ifndef VPM_DENOISING_H
-#define VPM_DENOISING_H
-
-#include "typedefs.h"
-#include "video_processing.h"
-
-namespace webrtc {
-
-class VPMDenoising
-{
-public:
- VPMDenoising();
- ~VPMDenoising();
-
- WebRtc_Word32 ChangeUniqueId(WebRtc_Word32 id);
-
- void Reset();
-
- WebRtc_Word32 ProcessFrame(WebRtc_UWord8* frame,
- WebRtc_UWord32 width,
- WebRtc_UWord32 height);
-
-private:
- WebRtc_Word32 _id;
-
- WebRtc_UWord32* _moment1; // (Q8) First order moment (mean)
- WebRtc_UWord32* _moment2; // (Q8) Second order moment
- WebRtc_UWord32 _frameSize; // Size (# of pixels) of frame
- WebRtc_Word32 _denoiseFrameCnt; // Counter for subsampling in time
-};
-
-} //namespace
-
-#endif // VPM_DENOISING_H
-
diff --git a/trunk/src/modules/video_processing/main/source/frame_preprocessor.cc b/trunk/src/modules/video_processing/main/source/frame_preprocessor.cc
deleted file mode 100644
index 76fdac8..0000000
--- a/trunk/src/modules/video_processing/main/source/frame_preprocessor.cc
+++ /dev/null
@@ -1,182 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "frame_preprocessor.h"
-#include "trace.h"
-
-namespace webrtc {
-
-VPMFramePreprocessor::VPMFramePreprocessor():
-_id(0),
-_contentMetrics(NULL),
-_maxFrameRate(0),
-_resampledFrame(),
-_enableCA(false)
-{
- _spatialResampler = new VPMSimpleSpatialResampler();
- _ca = new VPMContentAnalysis(true);
- _vd = new VPMVideoDecimator();
-}
-
-VPMFramePreprocessor::~VPMFramePreprocessor()
-{
- Reset();
- delete _spatialResampler;
- delete _ca;
- delete _vd;
- _resampledFrame.Free(); // is this needed?
-}
-
-WebRtc_Word32
-VPMFramePreprocessor::ChangeUniqueId(const WebRtc_Word32 id)
-{
- _id = id;
- return VPM_OK;
-}
-
-void
-VPMFramePreprocessor::Reset()
-{
- _ca->Release();
- _vd->Reset();
- _contentMetrics = NULL;
- _spatialResampler->Reset();
- _enableCA = false;
-}
-
-
-void
-VPMFramePreprocessor::EnableTemporalDecimation(bool enable)
-{
- _vd->EnableTemporalDecimation(enable);
-}
-void
-VPMFramePreprocessor::EnableContentAnalysis(bool enable)
-{
- _enableCA = enable;
-}
-
-void
-VPMFramePreprocessor::SetInputFrameResampleMode(VideoFrameResampling resamplingMode)
-{
- _spatialResampler->SetInputFrameResampleMode(resamplingMode);
-}
-
-
-WebRtc_Word32
-VPMFramePreprocessor::SetMaxFrameRate(WebRtc_UWord32 maxFrameRate)
-{
- if (maxFrameRate == 0)
- {
- return VPM_PARAMETER_ERROR;
- }
- //Max allowed frame rate
- _maxFrameRate = maxFrameRate;
-
- return _vd->SetMaxFrameRate(maxFrameRate);
-}
-
-
-WebRtc_Word32
-VPMFramePreprocessor::SetTargetResolution(WebRtc_UWord32 width, WebRtc_UWord32 height, WebRtc_UWord32 frameRate)
-{
- if ( (width == 0) || (height == 0) || (frameRate == 0))
- {
- return VPM_PARAMETER_ERROR;
- }
- WebRtc_Word32 retVal = 0;
- retVal = _spatialResampler->SetTargetFrameSize(width, height);
- if (retVal < 0)
- {
- return retVal;
- }
- retVal = _vd->SetTargetFrameRate(frameRate);
- if (retVal < 0)
- {
- return retVal;
- }
-
- return VPM_OK;
-}
-
-void
-VPMFramePreprocessor::UpdateIncomingFrameRate()
-{
- _vd->UpdateIncomingFrameRate();
-}
-
-WebRtc_UWord32
-VPMFramePreprocessor::DecimatedFrameRate()
-{
- return _vd->DecimatedFrameRate();
-}
-
-
-WebRtc_UWord32
-VPMFramePreprocessor::DecimatedWidth() const
-{
- return _spatialResampler->TargetWidth();
-}
-
-
-WebRtc_UWord32
-VPMFramePreprocessor::DecimatedHeight() const
-{
- return _spatialResampler->TargetHeight();
-}
-
-
-WebRtc_Word32
-VPMFramePreprocessor::PreprocessFrame(const VideoFrame* frame, VideoFrame** processedFrame)
-{
- if (frame == NULL || frame->Height() == 0 || frame->Width() == 0)
- {
- return VPM_PARAMETER_ERROR;
- }
-
- _vd->UpdateIncomingFrameRate();
-
- if (_vd->DropFrame())
- {
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo, _id, "Drop frame due to frame rate");
- return 1; // drop 1 frame
- }
-
- // Resizing incoming frame if needed.
- // Note that we must make a copy of it.
- // We are not allowed to resample the input frame.
- *processedFrame = NULL;
- if (_spatialResampler->ApplyResample(frame->Width(), frame->Height())) {
- WebRtc_Word32 ret = _spatialResampler->ResampleFrame(*frame, _resampledFrame);
- if (ret != VPM_OK)
- return ret;
- *processedFrame = &_resampledFrame;
- }
-
- // Perform content analysis on the frame to be encoded
- if (_enableCA)
- {
- if (*processedFrame == NULL) {
- _contentMetrics = _ca->ComputeContentMetrics(frame);
- } else {
- _contentMetrics = _ca->ComputeContentMetrics(&_resampledFrame);
- }
- }
- return VPM_OK;
-}
-
-
-VideoContentMetrics*
-VPMFramePreprocessor::ContentMetrics() const
-{
- return _contentMetrics;
-}
-
-} //namespace
diff --git a/trunk/src/modules/video_processing/main/source/frame_preprocessor.h b/trunk/src/modules/video_processing/main/source/frame_preprocessor.h
deleted file mode 100644
index 3c07a47..0000000
--- a/trunk/src/modules/video_processing/main/source/frame_preprocessor.h
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * frame_preprocessor.h
- */
-#ifndef VPM_FRAME_PREPROCESSOR_H
-#define VPM_FRAME_PREPROCESSOR_H
-
-#include "typedefs.h"
-#include "video_processing.h"
-#include "content_analysis.h"
-#include "spatial_resampler.h"
-#include "video_decimator.h"
-
-namespace webrtc {
-
-
-class VPMFramePreprocessor
-{
-public:
-
- VPMFramePreprocessor();
- ~VPMFramePreprocessor();
-
- WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- void Reset();
-
- // Enable temporal decimation
- void EnableTemporalDecimation(bool enable);
-
- void SetInputFrameResampleMode(VideoFrameResampling resamplingMode);
-
- //Enable content analysis
- void EnableContentAnalysis(bool enable);
-
- //Set max frame rate
- WebRtc_Word32 SetMaxFrameRate(WebRtc_UWord32 maxFrameRate);
-
- //Set target resolution: frame rate and dimension
- WebRtc_Word32 SetTargetResolution(WebRtc_UWord32 width, WebRtc_UWord32 height, WebRtc_UWord32 frameRate);
-
- //Update incoming frame rate/dimension
- void UpdateIncomingFrameRate();
-
- WebRtc_Word32 updateIncomingFrameSize(WebRtc_UWord32 width, WebRtc_UWord32 height);
-
- //Set decimated values: frame rate/dimension
- WebRtc_UWord32 DecimatedFrameRate();
- WebRtc_UWord32 DecimatedWidth() const;
- WebRtc_UWord32 DecimatedHeight() const;
-
- //Preprocess output:
- WebRtc_Word32 PreprocessFrame(const VideoFrame* frame, VideoFrame** processedFrame);
- VideoContentMetrics* ContentMetrics() const;
-
-private:
-
- WebRtc_Word32 _id;
- VideoContentMetrics* _contentMetrics;
- WebRtc_UWord32 _maxFrameRate;
- VideoFrame _resampledFrame;
- VPMSpatialResampler* _spatialResampler;
- VPMContentAnalysis* _ca;
- VPMVideoDecimator* _vd;
- bool _enableCA;
-
-}; // end of VPMFramePreprocessor class definition
-
-} //namespace
-
-#endif // VPM_FRAME_PREPROCESS_H
diff --git a/trunk/src/modules/video_processing/main/source/spatial_resampler.cc b/trunk/src/modules/video_processing/main/source/spatial_resampler.cc
deleted file mode 100644
index b48c322..0000000
--- a/trunk/src/modules/video_processing/main/source/spatial_resampler.cc
+++ /dev/null
@@ -1,123 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "spatial_resampler.h"
-
-
-namespace webrtc {
-
-VPMSimpleSpatialResampler::VPMSimpleSpatialResampler()
-:
-_resamplingMode(kFastRescaling),
-_targetWidth(0),
-_targetHeight(0),
-_scaler()
-{
-}
-
-VPMSimpleSpatialResampler::~VPMSimpleSpatialResampler()
-{
- //
-}
-
-
-WebRtc_Word32
-VPMSimpleSpatialResampler::SetTargetFrameSize(WebRtc_Word32 width,
- WebRtc_Word32 height)
-{
- if (_resamplingMode == kNoRescaling) {
- return VPM_OK;
- }
-
- if (width < 1 || height < 1) {
- return VPM_PARAMETER_ERROR;
- }
-
- _targetWidth = width;
- _targetHeight = height;
-
- return VPM_OK;
-}
-
-void
-VPMSimpleSpatialResampler::SetInputFrameResampleMode(VideoFrameResampling
- resamplingMode)
-{
- _resamplingMode = resamplingMode;
-}
-
-void
-VPMSimpleSpatialResampler::Reset()
-{
- _resamplingMode = kFastRescaling;
- _targetWidth = 0;
- _targetHeight = 0;
-}
-
-WebRtc_Word32
-VPMSimpleSpatialResampler::ResampleFrame(const VideoFrame& inFrame,
- VideoFrame& outFrame)
-{
- if (_resamplingMode == kNoRescaling)
- return outFrame.CopyFrame(inFrame);
- // Check if re-sampling is needed
- if ((inFrame.Width() == (WebRtc_UWord32)_targetWidth) &&
- (inFrame.Height() == (WebRtc_UWord32)_targetHeight)) {
- return outFrame.CopyFrame(inFrame);
- }
-
- // Setting scaler
- //TODO: Modify scaler types
- int retVal = 0;
- retVal = _scaler.Set(inFrame.Width(), inFrame.Height(),
- _targetWidth, _targetHeight, kI420, kI420, kScaleBox);
- if (retVal < 0)
- return retVal;
-
-
- // Disabling cut/pad for now - only scaling.
- int requiredSize = (WebRtc_UWord32)(_targetWidth * _targetHeight * 3 >> 1);
- outFrame.VerifyAndAllocate(requiredSize);
- outFrame.SetTimeStamp(inFrame.TimeStamp());
- outFrame.SetWidth(_targetWidth);
- outFrame.SetHeight(_targetHeight);
-
- retVal = _scaler.Scale(inFrame.Buffer(), outFrame.Buffer(), requiredSize);
- outFrame.SetLength(requiredSize);
- if (retVal == 0)
- return VPM_OK;
- else
- return VPM_SCALE_ERROR;
-}
-
-WebRtc_Word32
-VPMSimpleSpatialResampler::TargetHeight()
-{
- return _targetHeight;
-}
-
-WebRtc_Word32
-VPMSimpleSpatialResampler::TargetWidth()
-{
- return _targetWidth;
-}
-
-bool
-VPMSimpleSpatialResampler::ApplyResample(WebRtc_Word32 width,
- WebRtc_Word32 height)
-{
- if ((width == _targetWidth && height == _targetHeight) ||
- _resamplingMode == kNoRescaling)
- return false;
- else
- return true;
-}
-
-} //namespace
diff --git a/trunk/src/modules/video_processing/main/source/spatial_resampler.h b/trunk/src/modules/video_processing/main/source/spatial_resampler.h
deleted file mode 100644
index 76a63fb..0000000
--- a/trunk/src/modules/video_processing/main/source/spatial_resampler.h
+++ /dev/null
@@ -1,69 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * spatial_resampler.h
- */
-
-#ifndef VPM_SPATIAL_RESAMPLER_H
-#define VPM_SPATIAL_RESAMPLER_H
-
-#include "typedefs.h"
-
-#include "module_common_types.h"
-#include "video_processing_defines.h"
-
-#include "common_video/libyuv/include/libyuv.h"
-#include "common_video/libyuv/include/scaler.h"
-
-namespace webrtc {
-
-class VPMSpatialResampler
-{
-public:
- virtual ~VPMSpatialResampler() {};
- virtual WebRtc_Word32 SetTargetFrameSize(WebRtc_Word32 width,
- WebRtc_Word32 height) = 0;
- virtual void SetInputFrameResampleMode(VideoFrameResampling
- resamplingMode) = 0;
- virtual void Reset() = 0;
- virtual WebRtc_Word32 ResampleFrame(const VideoFrame& inFrame,
- VideoFrame& outFrame) = 0;
- virtual WebRtc_Word32 TargetWidth() = 0;
- virtual WebRtc_Word32 TargetHeight() = 0;
- virtual bool ApplyResample(WebRtc_Word32 width, WebRtc_Word32 height) = 0;
-};
-
-class VPMSimpleSpatialResampler : public VPMSpatialResampler
-{
-public:
- VPMSimpleSpatialResampler();
- ~VPMSimpleSpatialResampler();
- virtual WebRtc_Word32 SetTargetFrameSize(WebRtc_Word32 width,
- WebRtc_Word32 height);
- virtual void SetInputFrameResampleMode(VideoFrameResampling resamplingMode);
- virtual void Reset();
- virtual WebRtc_Word32 ResampleFrame(const VideoFrame& inFrame,
- VideoFrame& outFrame);
- virtual WebRtc_Word32 TargetWidth();
- virtual WebRtc_Word32 TargetHeight();
- virtual bool ApplyResample(WebRtc_Word32 width, WebRtc_Word32 height);
-
-private:
-
- VideoFrameResampling _resamplingMode;
- WebRtc_Word32 _targetWidth;
- WebRtc_Word32 _targetHeight;
- Scaler _scaler;
-};
-
-} //namespace
-
-#endif
diff --git a/trunk/src/modules/video_processing/main/source/video_decimator.cc b/trunk/src/modules/video_processing/main/source/video_decimator.cc
deleted file mode 100644
index 43bda08..0000000
--- a/trunk/src/modules/video_processing/main/source/video_decimator.cc
+++ /dev/null
@@ -1,235 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_decimator.h"
-#include "tick_util.h"
-#include "video_processing.h"
-
-#define VD_MIN(a, b) ((a) < (b)) ? (a) : (b)
-
-namespace webrtc {
-
-VPMVideoDecimator::VPMVideoDecimator()
-:
-_overShootModifier(0),
-_dropCount(0),
-_keepCount(0),
-_targetFrameRate(30),
-_incomingFrameRate(0.0f),
-_maxFrameRate(30),
-_incomingFrameTimes(),
-_enableTemporalDecimation(true)
-{
- Reset();
-}
-
-VPMVideoDecimator::~VPMVideoDecimator()
-{
- //
-}
-
-void
-VPMVideoDecimator::Reset()
-{
- _overShootModifier = 0;
- _dropCount = 0;
- _keepCount = 0;
- _targetFrameRate = 30;
- _incomingFrameRate = 0.0f;
- _maxFrameRate = 30;
- memset(_incomingFrameTimes, 0, sizeof(_incomingFrameTimes));
- _enableTemporalDecimation = true;
-}
-
-void
-VPMVideoDecimator::EnableTemporalDecimation(bool enable)
-{
- _enableTemporalDecimation = enable;
-}
-WebRtc_Word32
-VPMVideoDecimator::SetMaxFrameRate(WebRtc_UWord32 maxFrameRate)
-{
- if (maxFrameRate == 0)
- {
- return VPM_PARAMETER_ERROR;
- }
-
- _maxFrameRate = maxFrameRate;
-
- if (_targetFrameRate > _maxFrameRate)
- {
- _targetFrameRate = _maxFrameRate;
-
- }
- return VPM_OK;
-}
-
-WebRtc_Word32
-VPMVideoDecimator::SetTargetFrameRate(WebRtc_UWord32 frameRate)
-{
- if (frameRate == 0)
- {
- return VPM_PARAMETER_ERROR;
- }
- if (frameRate > _maxFrameRate)
- {
- //override
- _targetFrameRate = _maxFrameRate;
- }
- else
- {
- _targetFrameRate = frameRate;
- }
- return VPM_OK;
-}
-
-bool
-VPMVideoDecimator::DropFrame()
-{
- if (!_enableTemporalDecimation)
- {
- return false;
- }
-
- if (_incomingFrameRate <= 0)
- {
- return false;
- }
-
- const WebRtc_UWord32 incomingFrameRate = static_cast<WebRtc_UWord32>(_incomingFrameRate + 0.5f);
-
- if (_targetFrameRate == 0)
- {
- return true;
- }
-
- bool drop = false;
- if (incomingFrameRate > _targetFrameRate)
- {
- WebRtc_Word32 overshoot = _overShootModifier + (incomingFrameRate - _targetFrameRate);
- if(overshoot < 0)
- {
- overshoot = 0;
- _overShootModifier = 0;
- }
-
- if (overshoot && 2 * overshoot < (WebRtc_Word32) incomingFrameRate)
- {
-
- if (_dropCount) // Just got here so drop to be sure.
- {
- _dropCount = 0;
- return true;
- }
- const WebRtc_UWord32 dropVar = incomingFrameRate / overshoot;
-
- if (_keepCount >= dropVar)
- {
- drop = true;
- _overShootModifier = -((WebRtc_Word32) incomingFrameRate % overshoot) / 3;
- _keepCount = 1;
- }
- else
- {
-
- _keepCount++;
- }
- }
- else
- {
- _keepCount = 0;
- const WebRtc_UWord32 dropVar = overshoot / _targetFrameRate;
- if (_dropCount < dropVar)
- {
- drop = true;
- _dropCount++;
- }
- else
- {
- _overShootModifier = overshoot % _targetFrameRate;
- drop = false;
- _dropCount = 0;
- }
- }
- }
-
- return drop;
-}
-
-
-WebRtc_UWord32
-VPMVideoDecimator::DecimatedFrameRate()
-{
- ProcessIncomingFrameRate(TickTime::MillisecondTimestamp());
- if (!_enableTemporalDecimation)
- {
- return static_cast<WebRtc_UWord32>(_incomingFrameRate + 0.5f);
- }
- return VD_MIN(_targetFrameRate, static_cast<WebRtc_UWord32>(_incomingFrameRate + 0.5f));
-}
-
-WebRtc_UWord32
-VPMVideoDecimator::InputFrameRate()
-{
- ProcessIncomingFrameRate(TickTime::MillisecondTimestamp());
- return static_cast<WebRtc_UWord32>(_incomingFrameRate + 0.5f);
-}
-
-void
-VPMVideoDecimator::UpdateIncomingFrameRate()
-{
- WebRtc_Word64 now = TickTime::MillisecondTimestamp();
- if(_incomingFrameTimes[0] == 0)
- {
- // first no shift
- } else
- {
- // shift
- for(int i = (kFrameCountHistorySize - 2); i >= 0 ; i--)
- {
- _incomingFrameTimes[i+1] = _incomingFrameTimes[i];
- }
- }
- _incomingFrameTimes[0] = now;
- ProcessIncomingFrameRate(now);
-}
-
-void
-VPMVideoDecimator::ProcessIncomingFrameRate(WebRtc_Word64 now)
-{
- WebRtc_Word32 num = 0;
- WebRtc_Word32 nrOfFrames = 0;
- for(num = 1; num < (kFrameCountHistorySize - 1); num++)
- {
- if (_incomingFrameTimes[num] <= 0 ||
- now - _incomingFrameTimes[num] > kFrameHistoryWindowMs) // don't use data older than 2sec
- {
- break;
- } else
- {
- nrOfFrames++;
- }
- }
- if (num > 1)
- {
- WebRtc_Word64 diff = now - _incomingFrameTimes[num-1];
- _incomingFrameRate = 1.0;
- if(diff >0)
- {
- _incomingFrameRate = nrOfFrames * 1000.0f / static_cast<float>(diff);
- }
- }
- else
- {
- _incomingFrameRate = static_cast<float>(nrOfFrames);
- }
-}
-
-} //namespace
diff --git a/trunk/src/modules/video_processing/main/source/video_decimator.h b/trunk/src/modules/video_processing/main/source/video_decimator.h
deleted file mode 100644
index e152bb9..0000000
--- a/trunk/src/modules/video_processing/main/source/video_decimator.h
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * video_decimator.h
- */
-#ifndef VPM_VIDEO_DECIMATOR_H
-#define VPM_VIDEO_DECIMATOR_H
-
-#include "typedefs.h"
-#include "module_common_types.h"
-
-namespace webrtc {
-
-class VPMVideoDecimator
-{
-public:
- VPMVideoDecimator();
- ~VPMVideoDecimator();
-
- void Reset();
-
- void EnableTemporalDecimation(bool enable);
-
- WebRtc_Word32 SetMaxFrameRate(WebRtc_UWord32 maxFrameRate);
- WebRtc_Word32 SetTargetFrameRate(WebRtc_UWord32 frameRate);
-
- bool DropFrame();
-
- void UpdateIncomingFrameRate();
-
- // Get Decimated Frame Rate/Dimensions
- WebRtc_UWord32 DecimatedFrameRate();
-
- //Get input frame rate
- WebRtc_UWord32 InputFrameRate();
-
-private:
- void ProcessIncomingFrameRate(WebRtc_Word64 now);
-
- enum { kFrameCountHistorySize = 90};
- enum { kFrameHistoryWindowMs = 2000};
-
- // Temporal decimation
- WebRtc_Word32 _overShootModifier;
- WebRtc_UWord32 _dropCount;
- WebRtc_UWord32 _keepCount;
- WebRtc_UWord32 _targetFrameRate;
- float _incomingFrameRate;
- WebRtc_UWord32 _maxFrameRate;
- WebRtc_Word64 _incomingFrameTimes[kFrameCountHistorySize];
- bool _enableTemporalDecimation;
-
-};
-
-} //namespace
-
-#endif
diff --git a/trunk/src/modules/video_processing/main/source/video_processing.gypi b/trunk/src/modules/video_processing/main/source/video_processing.gypi
deleted file mode 100644
index 3bc03bc..0000000
--- a/trunk/src/modules/video_processing/main/source/video_processing.gypi
+++ /dev/null
@@ -1,88 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'video_processing',
- 'type': '<(library)',
- 'dependencies': [
- 'webrtc_utility',
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '../interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- ],
- },
- 'sources': [
- '../interface/video_processing.h',
- '../interface/video_processing_defines.h',
- 'brighten.cc',
- 'brighten.h',
- 'brightness_detection.cc',
- 'brightness_detection.h',
- 'color_enhancement.cc',
- 'color_enhancement.h',
- 'color_enhancement_private.h',
- 'content_analysis.cc',
- 'content_analysis.h',
- 'deflickering.cc',
- 'deflickering.h',
- 'denoising.cc',
- 'denoising.h',
- 'frame_preprocessor.cc',
- 'frame_preprocessor.h',
- 'spatial_resampler.cc',
- 'spatial_resampler.h',
- 'video_decimator.cc',
- 'video_decimator.h',
- 'video_processing_impl.cc',
- 'video_processing_impl.h',
- ],
- 'conditions': [
- ['target_arch=="ia32" or target_arch=="x64"', {
- 'dependencies': [ 'video_processing_sse2', ],
- }],
- ],
- },
- ],
- 'conditions': [
- ['target_arch=="ia32" or target_arch=="x64"', {
- 'targets': [
- {
- 'target_name': 'video_processing_sse2',
- 'type': '<(library)',
- 'sources': [
- 'content_analysis_sse2.cc',
- ],
- 'include_dirs': [
- '../interface',
- '../../../interface',
- ],
- 'conditions': [
- ['os_posix==1 and OS!="mac"', {
- 'cflags': [ '-msse2', ],
- }],
- ['OS=="mac"', {
- 'xcode_settings': {
- 'OTHER_CFLAGS': [ '-msse2', ],
- },
- }],
- ],
- },
- ],
- }],
- ],
-}
-
diff --git a/trunk/src/modules/video_processing/main/source/video_processing_impl.cc b/trunk/src/modules/video_processing/main/source/video_processing_impl.cc
deleted file mode 100644
index 3619996..0000000
--- a/trunk/src/modules/video_processing/main/source/video_processing_impl.cc
+++ /dev/null
@@ -1,340 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_processing_impl.h"
-#include "critical_section_wrapper.h"
-#include "trace.h"
-
-#include <cassert>
-
-namespace webrtc {
-
-namespace
-{
- void
- SetSubSampling(VideoProcessingModule::FrameStats& stats,
- const WebRtc_Word32 width,
- const WebRtc_Word32 height)
- {
- if (width * height >= 640 * 480)
- {
- stats.subSamplWidth = 3;
- stats.subSamplHeight = 3;
- }
- else if (width * height >= 352 * 288)
- {
- stats.subSamplWidth = 2;
- stats.subSamplHeight = 2;
- }
- else if (width * height >= 176 * 144)
- {
- stats.subSamplWidth = 1;
- stats.subSamplHeight = 1;
- }
- else
- {
- stats.subSamplWidth = 0;
- stats.subSamplHeight = 0;
- }
- }
-}
-
-VideoProcessingModule*
-VideoProcessingModule::Create(const WebRtc_Word32 id)
-{
-
- return new VideoProcessingModuleImpl(id);
-}
-
-void
-VideoProcessingModule::Destroy(VideoProcessingModule* module)
-{
- if (module)
- {
- delete static_cast<VideoProcessingModuleImpl*>(module);
- }
-}
-
-WebRtc_Word32
-VideoProcessingModuleImpl::ChangeUniqueId(const WebRtc_Word32 id)
-{
- CriticalSectionScoped mutex(_mutex);
- _id = id;
- _brightnessDetection.ChangeUniqueId(id);
- _deflickering.ChangeUniqueId(id);
- _denoising.ChangeUniqueId(id);
- _framePreProcessor.ChangeUniqueId(id);
- return VPM_OK;
-}
-
-WebRtc_Word32
-VideoProcessingModuleImpl::Id() const
-{
- CriticalSectionScoped mutex(_mutex);
- return _id;
-}
-
-VideoProcessingModuleImpl::VideoProcessingModuleImpl(const WebRtc_Word32 id) :
- _id(id),
- _mutex(*CriticalSectionWrapper::CreateCriticalSection())
-{
- _brightnessDetection.ChangeUniqueId(id);
- _deflickering.ChangeUniqueId(id);
- _denoising.ChangeUniqueId(id);
- _framePreProcessor.ChangeUniqueId(id);
- WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideoPreocessing, _id, "Created");
-}
-
-
-VideoProcessingModuleImpl::~VideoProcessingModuleImpl()
-{
- WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideoPreocessing, _id, "Destroyed");
-
- delete &_mutex;
-}
-
-void
-VideoProcessingModuleImpl::Reset()
-{
- CriticalSectionScoped mutex(_mutex);
- _deflickering.Reset();
- _denoising.Reset();
- _brightnessDetection.Reset();
- _framePreProcessor.Reset();
-
-}
-
-WebRtc_Word32
-VideoProcessingModule::GetFrameStats(FrameStats& stats,
- const VideoFrame& frame)
-{
- return GetFrameStats(stats, frame.Buffer(), frame.Width(), frame.Height());
-}
-
-WebRtc_Word32
-VideoProcessingModule::GetFrameStats(FrameStats& stats,
- const WebRtc_UWord8* frame,
- const WebRtc_UWord32 width,
- const WebRtc_UWord32 height)
-{
- if (frame == NULL)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, -1, "Null frame pointer");
- return VPM_PARAMETER_ERROR;
- }
-
- if (width == 0 || height == 0)
- {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideoPreocessing, -1, "Invalid frame size");
- return VPM_PARAMETER_ERROR;
- }
-
- ClearFrameStats(stats); // The histogram needs to be zeroed out.
- SetSubSampling(stats, width, height);
-
- // Compute histogram and sum of frame
- for (WebRtc_UWord32 i = 0; i < height; i += (1 << stats.subSamplHeight))
- {
- WebRtc_Word32 k = i * width;
- for (WebRtc_UWord32 j = 0; j < width; j += (1 << stats.subSamplWidth))
- {
- stats.hist[frame[k + j]]++;
- stats.sum += frame[k + j];
- }
- }
-
- stats.numPixels = (width * height) / ((1 << stats.subSamplWidth) * (1 << stats.subSamplHeight));
- assert(stats.numPixels > 0);
-
- // Compute mean value of frame
- stats.mean = stats.sum / stats.numPixels;
-
- return VPM_OK;
-}
-
-bool
-VideoProcessingModule::ValidFrameStats(const FrameStats& stats)
-{
- if (stats.numPixels == 0)
- {
- return false;
- }
-
- return true;
-}
-
-void
-VideoProcessingModule::ClearFrameStats(FrameStats& stats)
-{
- stats.mean = 0;
- stats.sum = 0;
- stats.numPixels = 0;
- stats.subSamplWidth = 0;
- stats.subSamplHeight = 0;
- memset(stats.hist, 0, sizeof(stats.hist));
-}
-
-WebRtc_Word32
-VideoProcessingModule::ColorEnhancement(VideoFrame& frame)
-{
- return ColorEnhancement(frame.Buffer(), frame.Width(), frame.Height());
-}
-
-WebRtc_Word32
-VideoProcessingModule::ColorEnhancement(WebRtc_UWord8* frame,
- const WebRtc_UWord32 width,
- const WebRtc_UWord32 height)
-{
- return VideoProcessing::ColorEnhancement(frame, width, height);
-}
-
-WebRtc_Word32
-VideoProcessingModule::Brighten(VideoFrame& frame, int delta)
-{
- return Brighten(frame.Buffer(), frame.Width(), frame.Height(), delta);
-}
-
-WebRtc_Word32
-VideoProcessingModule::Brighten(WebRtc_UWord8* frame,
- int width,
- int height,
- int delta)
-{
- return VideoProcessing::Brighten(frame, width, height, delta);
-}
-
-WebRtc_Word32
-VideoProcessingModuleImpl::Deflickering(VideoFrame& frame,
- FrameStats& stats)
-{
- return Deflickering(frame.Buffer(), frame.Width(), frame.Height(),
- frame.TimeStamp(), stats);
-}
-
-WebRtc_Word32
-VideoProcessingModuleImpl::Deflickering(WebRtc_UWord8* frame,
- const WebRtc_UWord32 width,
- const WebRtc_UWord32 height,
- const WebRtc_UWord32 timestamp,
- FrameStats& stats)
-{
- CriticalSectionScoped mutex(_mutex);
- return _deflickering.ProcessFrame(frame, width, height, timestamp, stats);
-}
-
-WebRtc_Word32
-VideoProcessingModuleImpl::Denoising(VideoFrame& frame)
-{
- return Denoising(frame.Buffer(), frame.Width(), frame.Height());
-}
-
-WebRtc_Word32
-VideoProcessingModuleImpl::Denoising(WebRtc_UWord8* frame,
- const WebRtc_UWord32 width,
- const WebRtc_UWord32 height)
-{
- CriticalSectionScoped mutex(_mutex);
- return _denoising.ProcessFrame(frame, width, height);
-}
-
-WebRtc_Word32
-VideoProcessingModuleImpl::BrightnessDetection(const VideoFrame& frame,
- const FrameStats& stats)
-{
- return BrightnessDetection(frame.Buffer(), frame.Width(), frame.Height(), stats);
-}
-
-WebRtc_Word32
-VideoProcessingModuleImpl::BrightnessDetection(const WebRtc_UWord8* frame,
- const WebRtc_UWord32 width,
- const WebRtc_UWord32 height,
- const FrameStats& stats)
-{
- CriticalSectionScoped mutex(_mutex);
- return _brightnessDetection.ProcessFrame(frame, width, height, stats);
-}
-
-
-void
-VideoProcessingModuleImpl::EnableTemporalDecimation(bool enable)
-{
- CriticalSectionScoped mutex(_mutex);
- _framePreProcessor.EnableTemporalDecimation(enable);
-}
-
-
-void
-VideoProcessingModuleImpl::SetInputFrameResampleMode(VideoFrameResampling resamplingMode)
-{
- CriticalSectionScoped cs(_mutex);
- _framePreProcessor.SetInputFrameResampleMode(resamplingMode);
-}
-
-WebRtc_Word32
-VideoProcessingModuleImpl::SetMaxFrameRate(WebRtc_UWord32 maxFrameRate)
-{
- CriticalSectionScoped cs(_mutex);
- return _framePreProcessor.SetMaxFrameRate(maxFrameRate);
-
-}
-
-WebRtc_Word32
-VideoProcessingModuleImpl::SetTargetResolution(WebRtc_UWord32 width, WebRtc_UWord32 height, WebRtc_UWord32 frameRate)
-{
- CriticalSectionScoped cs(_mutex);
- return _framePreProcessor.SetTargetResolution(width, height, frameRate);
-}
-
-
-WebRtc_UWord32
-VideoProcessingModuleImpl::DecimatedFrameRate()
-{
- CriticalSectionScoped cs(_mutex);
- return _framePreProcessor.DecimatedFrameRate();
-}
-
-
-WebRtc_UWord32
-VideoProcessingModuleImpl::DecimatedWidth() const
-{
- CriticalSectionScoped cs(_mutex);
- return _framePreProcessor.DecimatedWidth();
-}
-
-WebRtc_UWord32
-VideoProcessingModuleImpl::DecimatedHeight() const
-{
- CriticalSectionScoped cs(_mutex);
- return _framePreProcessor.DecimatedHeight();
-}
-
-WebRtc_Word32
-VideoProcessingModuleImpl::PreprocessFrame(const VideoFrame *frame, VideoFrame **processedFrame)
-{
- CriticalSectionScoped mutex(_mutex);
- return _framePreProcessor.PreprocessFrame(frame, processedFrame);
-}
-
-VideoContentMetrics*
-VideoProcessingModuleImpl::ContentMetrics() const
-{
- CriticalSectionScoped mutex(_mutex);
- return _framePreProcessor.ContentMetrics();
-}
-
-
-void
-VideoProcessingModuleImpl::EnableContentAnalysis(bool enable)
-{
- CriticalSectionScoped mutex(_mutex);
- _framePreProcessor.EnableContentAnalysis(enable);
-}
-
-} //namespace
diff --git a/trunk/src/modules/video_processing/main/source/video_processing_impl.h b/trunk/src/modules/video_processing/main/source/video_processing_impl.h
deleted file mode 100644
index 3170ab1..0000000
--- a/trunk/src/modules/video_processing/main/source/video_processing_impl.h
+++ /dev/null
@@ -1,107 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULE_VIDEO_PROCESSING_IMPL_H
-#define WEBRTC_MODULE_VIDEO_PROCESSING_IMPL_H
-
-#include "video_processing.h"
-#include "brighten.h"
-#include "brightness_detection.h"
-#include "color_enhancement.h"
-#include "deflickering.h"
-#include "denoising.h"
-#include "frame_preprocessor.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class VideoProcessingModuleImpl : public VideoProcessingModule
-{
-public:
-
- VideoProcessingModuleImpl(WebRtc_Word32 id);
-
- virtual ~VideoProcessingModuleImpl();
-
- WebRtc_Word32 Id() const;
-
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- virtual void Reset();
-
- virtual WebRtc_Word32 Deflickering(WebRtc_UWord8* frame,
- WebRtc_UWord32 width,
- WebRtc_UWord32 height,
- WebRtc_UWord32 timestamp,
- FrameStats& stats);
-
- virtual WebRtc_Word32 Deflickering(VideoFrame& frame,
- FrameStats& stats);
-
- virtual WebRtc_Word32 Denoising(WebRtc_UWord8* frame,
- WebRtc_UWord32 width,
- WebRtc_UWord32 height);
-
- virtual WebRtc_Word32 Denoising(VideoFrame& frame);
-
- virtual WebRtc_Word32 BrightnessDetection(const WebRtc_UWord8* frame,
- WebRtc_UWord32 width,
- WebRtc_UWord32 height,
- const FrameStats& stats);
-
- virtual WebRtc_Word32 BrightnessDetection(const VideoFrame& frame,
- const FrameStats& stats);
-
-
- //Frame pre-processor functions
-
- //Enable temporal decimation
- virtual void EnableTemporalDecimation(bool enable);
-
- virtual void SetInputFrameResampleMode(VideoFrameResampling resamplingMode);
-
- //Enable content analysis
- virtual void EnableContentAnalysis(bool enable);
-
- //Set max frame rate
- virtual WebRtc_Word32 SetMaxFrameRate(WebRtc_UWord32 maxFrameRate);
-
- // Set Target Resolution: frame rate and dimension
- virtual WebRtc_Word32 SetTargetResolution(WebRtc_UWord32 width,
- WebRtc_UWord32 height,
- WebRtc_UWord32 frameRate);
-
-
- // Get decimated values: frame rate/dimension
- virtual WebRtc_UWord32 DecimatedFrameRate();
- virtual WebRtc_UWord32 DecimatedWidth() const;
- virtual WebRtc_UWord32 DecimatedHeight() const;
-
- // Preprocess:
- // Pre-process incoming frame: Sample when needed and compute content
- // metrics when enabled.
- // If no resampling takes place - processedFrame is set to NULL.
- virtual WebRtc_Word32 PreprocessFrame(const VideoFrame* frame,
- VideoFrame** processedFrame);
- virtual VideoContentMetrics* ContentMetrics() const;
-
-private:
- WebRtc_Word32 _id;
- CriticalSectionWrapper& _mutex;
-
- VPMDeflickering _deflickering;
- VPMDenoising _denoising;
- VPMBrightnessDetection _brightnessDetection;
- VPMFramePreprocessor _framePreProcessor;
-};
-
-} // namespace
-
-#endif
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/brightness_detection_test.cc b/trunk/src/modules/video_processing/main/test/unit_test/brightness_detection_test.cc
deleted file mode 100644
index 6510a5c..0000000
--- a/trunk/src/modules/video_processing/main/test/unit_test/brightness_detection_test.cc
+++ /dev/null
@@ -1,107 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "unit_test.h"
-#include "video_processing.h"
-
-using namespace webrtc;
-
-TEST_F(VideoProcessingModuleTest, BrightnessDetection)
-{
- WebRtc_UWord32 frameNum = 0;
- WebRtc_Word32 brightnessWarning = 0;
- WebRtc_UWord32 warningCount = 0;
- while (fread(_videoFrame.Buffer(), 1, _frameLength, _sourceFile) == _frameLength)
- {
- frameNum++;
- VideoProcessingModule::FrameStats stats;
- ASSERT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
- ASSERT_GE(brightnessWarning = _vpm->BrightnessDetection(_videoFrame, stats), 0);
- if (brightnessWarning != VideoProcessingModule::kNoWarning)
- {
- warningCount++;
- }
- }
- ASSERT_NE(0, feof(_sourceFile)) << "Error reading source file";
-
- // Expect few warnings
- float warningProportion = static_cast<float>(warningCount) / frameNum * 100;
- printf("\nWarning proportions:\n");
- printf("Stock foreman: %.1f %%\n", warningProportion);
- EXPECT_LT(warningProportion, 10);
-
- rewind(_sourceFile);
- frameNum = 0;
- warningCount = 0;
- while (fread(_videoFrame.Buffer(), 1, _frameLength, _sourceFile) == _frameLength &&
- frameNum < 300)
- {
- frameNum++;
-
- WebRtc_UWord8* frame = _videoFrame.Buffer();
- WebRtc_UWord32 yTmp = 0;
- for (WebRtc_UWord32 yIdx = 0; yIdx < _width * _height; yIdx++)
- {
- yTmp = frame[yIdx] << 1;
- if (yTmp > 255)
- {
- yTmp = 255;
- }
- frame[yIdx] = static_cast<WebRtc_UWord8>(yTmp);
- }
-
- VideoProcessingModule::FrameStats stats;
- ASSERT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
- ASSERT_GE(brightnessWarning = _vpm->BrightnessDetection(_videoFrame, stats), 0);
- EXPECT_NE(VideoProcessingModule::kDarkWarning, brightnessWarning);
- if (brightnessWarning == VideoProcessingModule::kBrightWarning)
- {
- warningCount++;
- }
- }
- ASSERT_NE(0, feof(_sourceFile)) << "Error reading source file";
-
- // Expect many brightness warnings
- warningProportion = static_cast<float>(warningCount) / frameNum * 100;
- printf("Bright foreman: %.1f %%\n", warningProportion);
- EXPECT_GT(warningProportion, 95);
-
- rewind(_sourceFile);
- frameNum = 0;
- warningCount = 0;
- while (fread(_videoFrame.Buffer(), 1, _frameLength, _sourceFile) == _frameLength &&
- frameNum < 300)
- {
- frameNum++;
-
- WebRtc_UWord8* frame = _videoFrame.Buffer();
- WebRtc_Word32 yTmp = 0;
- for (WebRtc_UWord32 yIdx = 0; yIdx < _width * _height; yIdx++)
- {
- yTmp = frame[yIdx] >> 1;
- frame[yIdx] = static_cast<WebRtc_UWord8>(yTmp);
- }
-
- VideoProcessingModule::FrameStats stats;
- ASSERT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
- ASSERT_GE(brightnessWarning = _vpm->BrightnessDetection(_videoFrame, stats), 0);
- EXPECT_NE(VideoProcessingModule::kBrightWarning, brightnessWarning);
- if (brightnessWarning == VideoProcessingModule::kDarkWarning)
- {
- warningCount++;
- }
- }
- ASSERT_NE(0, feof(_sourceFile)) << "Error reading source file";
-
- // Expect many darkness warnings
- warningProportion = static_cast<float>(warningCount) / frameNum * 100;
- printf("Dark foreman: %.1f %%\n\n", warningProportion);
- EXPECT_GT(warningProportion, 90);
-}
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/color_enhancement_test.cc b/trunk/src/modules/video_processing/main/test/unit_test/color_enhancement_test.cc
deleted file mode 100644
index c494c85..0000000
--- a/trunk/src/modules/video_processing/main/test/unit_test/color_enhancement_test.cc
+++ /dev/null
@@ -1,125 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cstdio>
-#include <cstdlib>
-
-#include "modules/video_processing/main/interface/video_processing.h"
-#include "modules/video_processing/main/test/unit_test/unit_test.h"
-#include "system_wrappers/interface/tick_util.h"
-#include "testsupport/fileutils.h"
-
-namespace webrtc {
-
-TEST_F(VideoProcessingModuleTest, ColorEnhancement)
-{
- TickTime t0;
- TickTime t1;
- TickInterval accTicks;
-
- // Use a shorter version of the Foreman clip for this test.
- fclose(_sourceFile);
- const std::string video_file =
- webrtc::test::ResourcePath("foreman_cif_short", "yuv");
- _sourceFile = fopen(video_file.c_str(), "rb");
- ASSERT_TRUE(_sourceFile != NULL) <<
- "Cannot read source file: " + video_file + "\n";
-
- std::string output_file = webrtc::test::OutputPath() +
- "foremanColorEnhancedVPM_cif_short.yuv";
- FILE* modFile = fopen(output_file.c_str(), "w+b");
- ASSERT_TRUE(modFile != NULL) << "Could not open output file.\n";
-
- WebRtc_UWord32 frameNum = 0;
- while (fread(_videoFrame.Buffer(), 1, _frameLength, _sourceFile) == _frameLength)
- {
- frameNum++;
- t0 = TickTime::Now();
- ASSERT_EQ(0, VideoProcessingModule::ColorEnhancement(_videoFrame));
- t1 = TickTime::Now();
- accTicks += t1 - t0;
- fwrite(_videoFrame.Buffer(), 1, _frameLength, modFile);
- }
- ASSERT_NE(0, feof(_sourceFile)) << "Error reading source file";
-
- printf("\nTime per frame: %d us \n",
- static_cast<int>(accTicks.Microseconds() / frameNum));
- rewind(modFile);
-
- printf("Comparing files...\n\n");
- std::string reference_filename =
- webrtc::test::ResourcePath("foremanColorEnhanced_cif_short", "yuv");
- FILE* refFile = fopen(reference_filename.c_str(), "rb");
- ASSERT_TRUE(refFile != NULL) << "Cannot open reference file: " <<
- reference_filename << "\n"
- "Create the reference by running Matlab script createTable.m.";
-
- // get file lenghts
- ASSERT_EQ(0, fseek(refFile, 0L, SEEK_END));
- long refLen = ftell(refFile);
- ASSERT_NE(-1L, refLen);
- rewind(refFile);
- ASSERT_EQ(0, fseek(modFile, 0L, SEEK_END));
- long testLen = ftell(modFile);
- ASSERT_NE(-1L, testLen);
- rewind(modFile);
- ASSERT_EQ(refLen, testLen) << "File lengths differ.";
-
- VideoFrame refVideoFrame;
- refVideoFrame.VerifyAndAllocate(_frameLength);
- refVideoFrame.SetWidth(_width);
- refVideoFrame.SetHeight(_height);
-
- // Compare frame-by-frame.
- while (fread(_videoFrame.Buffer(), 1, _frameLength, modFile) == _frameLength)
- {
- ASSERT_EQ(_frameLength, fread(refVideoFrame.Buffer(), 1, _frameLength, refFile));
- EXPECT_EQ(0, memcmp(_videoFrame.Buffer(), refVideoFrame.Buffer(), _frameLength));
- }
- ASSERT_NE(0, feof(_sourceFile)) << "Error reading source file";
-
- // Verify that all color pixels are enhanced, that no luminance values are altered,
- // and that the function does not write outside the vector.
- WebRtc_UWord32 safeGuard = 1000;
- WebRtc_UWord32 numPixels = 352*288; // CIF size
- WebRtc_UWord8 *testFrame = new WebRtc_UWord8[numPixels + (numPixels / 2) + (2 * safeGuard)];
- WebRtc_UWord8 *refFrame = new WebRtc_UWord8[numPixels + (numPixels / 2) + (2 * safeGuard)];
-
- // use value 128 as probe value, since we know that this will be changed in the enhancement
- memset(testFrame, 128, safeGuard);
- memset(&testFrame[safeGuard], 128, numPixels);
- memset(&testFrame[safeGuard + numPixels], 128, numPixels / 2);
- memset(&testFrame[safeGuard + numPixels + (numPixels / 2)], 128, safeGuard);
-
- memcpy(refFrame, testFrame, numPixels + (numPixels / 2) + (2 * safeGuard));
-
- ASSERT_EQ(0, VideoProcessingModule::ColorEnhancement(&testFrame[safeGuard], 352, 288));
-
- EXPECT_EQ(0, memcmp(testFrame, refFrame, safeGuard)) <<
- "Function is writing outside the frame memory.";
-
- EXPECT_EQ(0, memcmp(&testFrame[safeGuard + numPixels + (numPixels / 2)],
- &refFrame[safeGuard + numPixels + (numPixels / 2)], safeGuard)) <<
- "Function is writing outside the frame memory.";
-
- EXPECT_EQ(0, memcmp(&testFrame[safeGuard], &refFrame[safeGuard], numPixels)) <<
- "Function is modifying the luminance.";
-
- EXPECT_NE(0, memcmp(&testFrame[safeGuard + numPixels],
- &refFrame[safeGuard + numPixels], numPixels / 2)) <<
- "Function is not modifying all chrominance pixels";
-
- ASSERT_EQ(0, fclose(refFile));
- ASSERT_EQ(0, fclose(modFile));
- delete [] testFrame;
- delete [] refFrame;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/content_metrics_test.cc b/trunk/src/modules/video_processing/main/test/unit_test/content_metrics_test.cc
deleted file mode 100644
index 54a1390..0000000
--- a/trunk/src/modules/video_processing/main/test/unit_test/content_metrics_test.cc
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/video_processing/main/interface/video_processing.h"
-#include "modules/video_processing/main/source/content_analysis.h"
-#include "modules/video_processing/main/test/unit_test/unit_test.h"
-
-namespace webrtc {
-
-TEST_F(VideoProcessingModuleTest, ContentAnalysis)
-{
- VPMContentAnalysis _ca_c(false);
- VPMContentAnalysis _ca_sse(true);
- VideoContentMetrics *_cM_c, *_cM_SSE;
-
- _ca_c.Initialize(_width,_height);
- _ca_sse.Initialize(_width,_height);
-
- while (fread(_videoFrame.Buffer(), 1, _frameLength, _sourceFile)
- == _frameLength)
- {
- _cM_c = _ca_c.ComputeContentMetrics(&_videoFrame);
- _cM_SSE = _ca_sse.ComputeContentMetrics(&_videoFrame);
-
- ASSERT_EQ(_cM_c->spatial_pred_err, _cM_SSE->spatial_pred_err);
- ASSERT_EQ(_cM_c->spatial_pred_err_v, _cM_SSE->spatial_pred_err_v);
- ASSERT_EQ(_cM_c->spatial_pred_err_h, _cM_SSE->spatial_pred_err_h);
- ASSERT_EQ(_cM_c->motion_magnitude, _cM_SSE->motion_magnitude);
- }
- ASSERT_NE(0, feof(_sourceFile)) << "Error reading source file";
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/createTable.m b/trunk/src/modules/video_processing/main/test/unit_test/createTable.m
deleted file mode 100644
index 2c7fb52..0000000
--- a/trunk/src/modules/video_processing/main/test/unit_test/createTable.m
+++ /dev/null
@@ -1,179 +0,0 @@
-% Create the color enhancement look-up table and write it to
-% file colorEnhancementTable.cpp. Copy contents of that file into
-% the source file for the color enhancement function.
-
-clear
-close all
-
-
-% First, define the color enhancement in a normalized domain
-
-% Compander function is defined in three radial zones.
-% 1. From 0 to radius r0, the compander function
-% is a second-order polynomial intersecting the points (0,0)
-% and (r0, r0), and with a slope B in (0,0).
-% 2. From r0 to r1, the compander is a third-order polynomial
-% intersecting the points (r0, r0) and (r1, r1), and with the
-% same slope as the first part in the point (r0, r0) and slope
-% equal to 1 in (r1, r1).
-% 3. For radii larger than r1, the compander function is the
-% unity scale function (no scaling at all).
-
-r0=0.07; % Dead zone radius (must be > 0)
-r1=0.6; % Enhancement zone radius (must be > r0 and < 1)
-B=0.2; % initial slope of compander function (between 0 and 1)
-
-x0=linspace(0,r0).'; % zone 1
-x1=linspace(r0,r1).'; % zone 2
-x2=linspace(r1,1).'; % zone 3
-
-A=(1-B)/r0;
-f0=A*x0.^2+B*x0; % compander function in zone 1
-
-% equation system for finding second zone parameters
-M=[r0^3 r0^2 r0 1;
- 3*r0^2 2*r0 1 0;
- 3*r1^2 2*r1 1 0;
- r1^3 r1^2 r1 1];
-m=[A*r0^2+B*r0; 2*A*r0+B; 1; r1];
-% solve equations
-theta=M\m;
-
-% compander function in zone 1
-f1=[x1.^3 x1.^2 x1 ones(size(x1))]*theta;
-
-x=[x0; x1; x2];
-f=[f0; f1; x2];
-
-% plot it
-figure(1)
-plot(x,f,x,x,':')
-xlabel('Normalized radius')
-ylabel('Modified radius')
-
-
-% Now, create the look-up table in the integer color space
-[U,V]=meshgrid(0:255, 0:255); % U-V space
-U0=U;
-V0=V;
-
-% Conversion matrix from normalized YUV to RGB
-T=[1 0 1.13983; 1 -0.39465 -0.58060; 1 2.03211 0];
-Ylum=0.5;
-
-figure(2)
-Z(:,:,1)=Ylum + (U-127)/256*T(1,2) + (V-127)/256*T(1,3);
-Z(:,:,2)=Ylum + (U-127)/256*T(2,2) + (V-127)/256*T(2,3);
-Z(:,:,3)=Ylum + (U-127)/256*T(3,2) + (V-127)/256*T(3,3);
-Z=max(Z,0);
-Z=min(Z,1);
-subplot(121)
-image(Z);
-axis square
-axis off
-set(gcf,'color','k')
-
-R = sqrt((U-127).^2 + (V-127).^2);
-Rnorm = R/127;
-RnormMod = Rnorm;
-RnormMod(RnormMod==0)=1; % avoid division with zero
-
-% find indices to pixels in dead-zone (zone 1)
-ix=find(Rnorm<=r0);
-scaleMatrix = (A*Rnorm(ix).^2 + B*Rnorm(ix))./RnormMod(ix);
-U(ix)=(U(ix)-127).*scaleMatrix+127;
-V(ix)=(V(ix)-127).*scaleMatrix+127;
-
-% find indices to pixels in zone 2
-ix=find(Rnorm>r0 & Rnorm<=r1);
-scaleMatrix = (theta(1)*Rnorm(ix).^3 + theta(2)*Rnorm(ix).^2 + ...
- theta(3)*Rnorm(ix) + theta(4)) ./ RnormMod(ix);
-U(ix)=(U(ix)-127).*scaleMatrix + 127;
-V(ix)=(V(ix)-127).*scaleMatrix + 127;
-
-% round to integer values and saturate
-U=round(U);
-V=round(V);
-U=max(min(U,255),0);
-V=max(min(V,255),0);
-
-Z(:,:,1)=Ylum + (U-127)/256*T(1,2) + (V-127)/256*T(1,3);
-Z(:,:,2)=Ylum + (U-127)/256*T(2,2) + (V-127)/256*T(2,3);
-Z(:,:,3)=Ylum + (U-127)/256*T(3,2) + (V-127)/256*T(3,3);
-Z=max(Z,0);
-Z=min(Z,1);
-subplot(122)
-image(Z);
-axis square
-axis off
-
-figure(3)
-subplot(121)
-mesh(U-U0)
-subplot(122)
-mesh(V-V0)
-
-
-
-% Last, write to file
-% Write only one matrix, since U=V'
-
-fid = fopen('../out/Debug/colorEnhancementTable.h','wt');
-if fid==-1
- error('Cannot open file colorEnhancementTable.cpp');
-end
-
-fprintf(fid,'//colorEnhancementTable.h\n\n');
-fprintf(fid,'//Copy the constant table to the appropriate header file.\n\n');
-
-fprintf(fid,'//Table created with Matlab script createTable.m\n\n');
-fprintf(fid,'//Usage:\n');
-fprintf(fid,'// Umod=colorTable[U][V]\n');
-fprintf(fid,'// Vmod=colorTable[V][U]\n');
-
-fprintf(fid,'static unsigned char colorTable[%i][%i] = {\n', size(U,1), size(U,2));
-
-for u=1:size(U,2)
- fprintf(fid,' {%i', U(1,u));
- for v=2:size(U,1)
- fprintf(fid,', %i', U(v,u));
- end
- fprintf(fid,'}');
- if u<size(U,2)
- fprintf(fid,',');
- end
- fprintf(fid,'\n');
-end
-fprintf(fid,'};\n\n');
-fclose(fid);
-fprintf('done');
-
-
-answ=input('Create test vector (takes some time...)? y/n : ','s');
-if answ ~= 'y'
- return
-end
-
-% Also, create test vectors
-
-% Read test file foreman.yuv
-fprintf('Reading test file...')
-[y,u,v]=readYUV420file('../out/Debug/testFiles/foreman_cif.yuv',352,288);
-fprintf(' done\n');
-unew=uint8(zeros(size(u)));
-vnew=uint8(zeros(size(v)));
-
-% traverse all frames
-for k=1:size(y,3)
- fprintf('Frame %i\n', k);
- for r=1:size(u,1)
- for c=1:size(u,2)
- unew(r,c,k) = uint8(U(double(v(r,c,k))+1, double(u(r,c,k))+1));
- vnew(r,c,k) = uint8(V(double(v(r,c,k))+1, double(u(r,c,k))+1));
- end
- end
-end
-
-fprintf('\nWriting modified test file...')
-writeYUV420file('../out/Debug/foremanColorEnhanced.yuv',y,unew,vnew);
-fprintf(' done\n');
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/deflickering_test.cc b/trunk/src/modules/video_processing/main/test/unit_test/deflickering_test.cc
deleted file mode 100644
index f91dc13..0000000
--- a/trunk/src/modules/video_processing/main/test/unit_test/deflickering_test.cc
+++ /dev/null
@@ -1,91 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cstdio>
-#include <cstdlib>
-
-#include "modules/video_processing/main/interface/video_processing.h"
-#include "modules/video_processing/main/test/unit_test/unit_test.h"
-#include "system_wrappers/interface/tick_util.h"
-#include "testsupport/fileutils.h"
-
-namespace webrtc {
-
-TEST_F(VideoProcessingModuleTest, Deflickering)
-{
- enum { NumRuns = 30 };
- WebRtc_UWord32 frameNum = 0;
- const WebRtc_UWord32 frameRate = 15;
-
- WebRtc_Word64 minRuntime = 0;
- WebRtc_Word64 avgRuntime = 0;
-
- // Close automatically opened Foreman.
- fclose(_sourceFile);
- const std::string input_file =
- webrtc::test::ResourcePath("deflicker_before_cif_short", "yuv");
- _sourceFile = fopen(input_file.c_str(), "rb");
- ASSERT_TRUE(_sourceFile != NULL) <<
- "Cannot read input file: " << input_file << "\n";
-
- const std::string output_file =
- webrtc::test::OutputPath() + "deflicker_output_cif_short.yuv";
- FILE* deflickerFile = fopen(output_file.c_str(), "wb");
- ASSERT_TRUE(deflickerFile != NULL) <<
- "Could not open output file: " << output_file << "\n";
-
- printf("\nRun time [us / frame]:\n");
- for (WebRtc_UWord32 runIdx = 0; runIdx < NumRuns; runIdx++)
- {
- TickTime t0;
- TickTime t1;
- TickInterval accTicks;
- WebRtc_UWord32 timeStamp = 1;
-
- frameNum = 0;
- while (fread(_videoFrame.Buffer(), 1, _frameLength, _sourceFile) == _frameLength)
- {
- frameNum++;
- _videoFrame.SetTimeStamp(timeStamp);
-
- t0 = TickTime::Now();
- VideoProcessingModule::FrameStats stats;
- ASSERT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
- ASSERT_EQ(0, _vpm->Deflickering(_videoFrame, stats));
- t1 = TickTime::Now();
- accTicks += t1 - t0;
-
- if (runIdx == 0)
- {
- fwrite(_videoFrame.Buffer(), 1, _frameLength, deflickerFile);
- }
- timeStamp += (90000 / frameRate);
- }
- ASSERT_NE(0, feof(_sourceFile)) << "Error reading source file";
-
- printf("%u\n", static_cast<int>(accTicks.Microseconds() / frameNum));
- if (accTicks.Microseconds() < minRuntime || runIdx == 0)
- {
- minRuntime = accTicks.Microseconds();
- }
- avgRuntime += accTicks.Microseconds();
-
- rewind(_sourceFile);
- }
- ASSERT_EQ(0, fclose(deflickerFile));
- // TODO(kjellander): Add verification of deflicker output file.
-
- printf("\nAverage run time = %d us / frame\n",
- static_cast<int>(avgRuntime / frameNum / NumRuns));
- printf("Min run time = %d us / frame\n\n",
- static_cast<int>(minRuntime / frameNum));
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/denoising_test.cc b/trunk/src/modules/video_processing/main/test/unit_test/denoising_test.cc
deleted file mode 100644
index 173ceb8..0000000
--- a/trunk/src/modules/video_processing/main/test/unit_test/denoising_test.cc
+++ /dev/null
@@ -1,127 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cstdio>
-#include <cstdlib>
-
-#include "modules/video_processing/main/interface/video_processing.h"
-#include "modules/video_processing/main/test/unit_test/unit_test.h"
-#include "system_wrappers/interface/tick_util.h"
-#include "testsupport/fileutils.h"
-
-namespace webrtc {
-
-TEST_F(VideoProcessingModuleTest, Denoising)
-{
- enum { NumRuns = 10 };
- WebRtc_UWord32 frameNum = 0;
-
- WebRtc_Word64 minRuntime = 0;
- WebRtc_Word64 avgRuntime = 0;
-
- const std::string denoise_filename =
- webrtc::test::OutputPath() + "denoise_testfile.yuv";
- FILE* denoiseFile = fopen(denoise_filename.c_str(), "wb");
- ASSERT_TRUE(denoiseFile != NULL) <<
- "Could not open output file: " << denoise_filename << "\n";
-
- const std::string noise_filename =
- webrtc::test::OutputPath() + "noise_testfile.yuv";
- FILE* noiseFile = fopen(noise_filename.c_str(), "wb");
- ASSERT_TRUE(noiseFile != NULL) <<
- "Could not open noisy file: " << noise_filename << "\n";
-
- printf("\nRun time [us / frame]:\n");
- for (WebRtc_UWord32 runIdx = 0; runIdx < NumRuns; runIdx++)
- {
- TickTime t0;
- TickTime t1;
- TickInterval accTicks;
- WebRtc_Word32 modifiedPixels = 0;
-
- frameNum = 0;
- while (fread(_videoFrame.Buffer(), 1, _frameLength, _sourceFile) == _frameLength)
- {
- frameNum++;
- WebRtc_UWord8* sourceBuffer = _videoFrame.Buffer();
-
- // Add noise to a part in video stream
- // Random noise
- // TODO: investigate the effectiveness of this test.
-
- //for(WebRtc_UWord32 ir = 0; ir < _frameLength; ir++)
- // sourceBuffer[ir] = 128
- for (WebRtc_UWord32 ir = 0; ir < _height; ir++)
- {
- WebRtc_UWord32 ik = ir * _width;
- for (WebRtc_UWord32 ic = 0; ic < _width; ic++)
- {
- WebRtc_UWord8 r = rand() % 16;
- r -= 8;
- if (ir < _height / 4)
- r = 0;
- if (ir >= 3 * _height / 4)
- r = 0;
- if (ic < _width / 4)
- r = 0;
- if (ic >= 3 * _width / 4)
- r = 0;
-
- /*WebRtc_UWord8 pixelValue = 0;
- if (ir >= _height / 2)
- { // Region 3 or 4
- pixelValue = 170;
- }
- if (ic >= _width / 2)
- { // Region 2 or 4
- pixelValue += 85;
- }
- pixelValue += r;
- sourceBuffer[ik + ic] = pixelValue;
- */
- sourceBuffer[ik + ic] += r;
- }
- }
-
- if (runIdx == 0)
- {
- fwrite(_videoFrame.Buffer(), 1, _frameLength, noiseFile);
- }
-
- t0 = TickTime::Now();
- ASSERT_GE(modifiedPixels = _vpm->Denoising(_videoFrame), 0);
- t1 = TickTime::Now();
- accTicks += t1 - t0;
-
- if (runIdx == 0)
- {
- fwrite(_videoFrame.Buffer(), 1, _frameLength, denoiseFile);
- }
- }
- ASSERT_NE(0, feof(_sourceFile)) << "Error reading source file";
-
- printf("%u\n", static_cast<int>(accTicks.Microseconds() / frameNum));
- if (accTicks.Microseconds() < minRuntime || runIdx == 0)
- {
- minRuntime = accTicks.Microseconds();
- }
- avgRuntime += accTicks.Microseconds();
-
- rewind(_sourceFile);
- }
- ASSERT_EQ(0, fclose(denoiseFile));
- ASSERT_EQ(0, fclose(noiseFile));
- printf("\nAverage run time = %d us / frame\n",
- static_cast<int>(avgRuntime / frameNum / NumRuns));
- printf("Min run time = %d us / frame\n\n",
- static_cast<int>(minRuntime / frameNum));
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/readYUV420file.m b/trunk/src/modules/video_processing/main/test/unit_test/readYUV420file.m
deleted file mode 100644
index 03013ef..0000000
--- a/trunk/src/modules/video_processing/main/test/unit_test/readYUV420file.m
+++ /dev/null
@@ -1,45 +0,0 @@
-function [Y,U,V] = readYUV420file(filename, width, height)
-% [Y,U,V] = readYUVfile(filename, width, height)
-
-fid = fopen(filename,'rb');
-if fid==-1
- error(['Cannot open file ' filename]);
-end
-
-% Number of pixels per image
-nPx=width*height;
-
-% nPx bytes luminance, nPx/4 bytes U, nPx/4 bytes V
-frameSizeBytes = nPx*1.5;
-
-% calculate number of frames
-fseek(fid,0,'eof'); % move to end of file
-fileLen=ftell(fid); % number of bytes
-fseek(fid,0,'bof'); % rewind to start
-
-% calculate number of frames
-numFrames = floor(fileLen/frameSizeBytes);
-
-Y=uint8(zeros(height,width,numFrames));
-U=uint8(zeros(height/2,width/2,numFrames));
-V=uint8(zeros(height/2,width/2,numFrames));
-
-[X,nBytes]=fread(fid, frameSizeBytes, 'uchar');
-
-for k=1:numFrames
-
- % Store luminance
- Y(:,:,k)=uint8(reshape(X(1:nPx), width, height).');
-
- % Store U channel
- U(:,:,k)=uint8(reshape(X(nPx + (1:nPx/4)), width/2, height/2).');
-
- % Store V channel
- V(:,:,k)=uint8(reshape(X(nPx + nPx/4 + (1:nPx/4)), width/2, height/2).');
-
- % Read next frame
- [X,nBytes]=fread(fid, frameSizeBytes, 'uchar');
-end
-
-
-fclose(fid);
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/unit_test.cc b/trunk/src/modules/video_processing/main/test/unit_test/unit_test.cc
deleted file mode 100644
index 1dbe070..0000000
--- a/trunk/src/modules/video_processing/main/test/unit_test/unit_test.cc
+++ /dev/null
@@ -1,343 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "modules/video_processing/main/test/unit_test/unit_test.h"
-
-#include <string>
-
-#include "common_video/libyuv/include/libyuv.h"
-#include "system_wrappers/interface/tick_util.h"
-#include "testsupport/fileutils.h"
-
-namespace webrtc {
-
-void TestSize(VideoFrame& sourceFrame,
- WebRtc_UWord32 targetWidth, WebRtc_UWord32 targetHeight,
- WebRtc_UWord32 mode, VideoProcessingModule *vpm);
-
-VideoProcessingModuleTest::VideoProcessingModuleTest() :
- _vpm(NULL),
- _sourceFile(NULL),
- _width(352),
- _height(288),
- _frameLength(CalcBufferSize(kI420, 352, 288))
-{
-}
-
-void VideoProcessingModuleTest::SetUp()
-{
- _vpm = VideoProcessingModule::Create(0);
- ASSERT_TRUE(_vpm != NULL);
-
- ASSERT_EQ(0, _videoFrame.VerifyAndAllocate(_frameLength));
- _videoFrame.SetWidth(_width);
- _videoFrame.SetHeight(_height);
-
- const std::string video_file =
- webrtc::test::ResourcePath("foreman_cif", "yuv");
- _sourceFile = fopen(video_file.c_str(),"rb");
- ASSERT_TRUE(_sourceFile != NULL) <<
- "Cannot read source file: " + video_file + "\n";
-}
-
-void VideoProcessingModuleTest::TearDown()
-{
- if (_sourceFile != NULL) {
- ASSERT_EQ(0, fclose(_sourceFile));
- }
- _sourceFile = NULL;
-
- if (_vpm != NULL) {
- VideoProcessingModule::Destroy(_vpm);
- }
- _vpm = NULL;
-}
-
-TEST_F(VideoProcessingModuleTest, HandleNullBuffer)
-{
- VideoProcessingModule::FrameStats stats;
- ASSERT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
- // Video frame with unallocated buffer.
- VideoFrame videoFrame;
- videoFrame.SetWidth(_width);
- videoFrame.SetHeight(_height);
-
- EXPECT_EQ(-3, _vpm->GetFrameStats(stats, NULL, _width, _height));
- EXPECT_EQ(-3, _vpm->GetFrameStats(stats, videoFrame));
-
- EXPECT_EQ(-1, _vpm->ColorEnhancement(NULL, _width, _height));
- EXPECT_EQ(-1, _vpm->ColorEnhancement(videoFrame));
-
- EXPECT_EQ(-1, _vpm->Deflickering(NULL, _width, _height, 0, stats));
- EXPECT_EQ(-1, _vpm->Deflickering(videoFrame, stats));
-
- EXPECT_EQ(-1, _vpm->Denoising(NULL, _width, _height));
- EXPECT_EQ(-1, _vpm->Denoising(videoFrame));
-
- EXPECT_EQ(-3, _vpm->BrightnessDetection(NULL, _width, _height, stats));
- EXPECT_EQ(-3, _vpm->BrightnessDetection(videoFrame, stats));
-
- EXPECT_EQ(VPM_PARAMETER_ERROR, _vpm->PreprocessFrame(NULL, NULL));
-}
-
-TEST_F(VideoProcessingModuleTest, HandleBadStats)
-{
- VideoProcessingModule::FrameStats stats;
-
- ASSERT_EQ(_frameLength, fread(_videoFrame.Buffer(), 1, _frameLength,
- _sourceFile));
-
- EXPECT_EQ(-1, _vpm->Deflickering(_videoFrame.Buffer(), _width, _height, 0,
- stats));
- EXPECT_EQ(-1, _vpm->Deflickering(_videoFrame, stats));
-
- EXPECT_EQ(-3, _vpm->BrightnessDetection(_videoFrame.Buffer(), _width,
- _height, stats));
- EXPECT_EQ(-3, _vpm->BrightnessDetection(_videoFrame, stats));
-}
-
-TEST_F(VideoProcessingModuleTest, HandleBadSize)
-{
- VideoProcessingModule::FrameStats stats;
- ASSERT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
-
- // Bad width
- _videoFrame.SetWidth(0);
- EXPECT_EQ(-3, _vpm->GetFrameStats(stats, _videoFrame.Buffer(), 0, _height));
- EXPECT_EQ(-3, _vpm->GetFrameStats(stats, _videoFrame));
-
- EXPECT_EQ(-1, _vpm->ColorEnhancement(_videoFrame.Buffer(), 0, _height));
- EXPECT_EQ(-1, _vpm->ColorEnhancement(_videoFrame));
-
- EXPECT_EQ(-1, _vpm->Deflickering(_videoFrame.Buffer(), 0, _height, 0,
- stats));
- EXPECT_EQ(-1, _vpm->Deflickering(_videoFrame, stats));
-
- EXPECT_EQ(-1, _vpm->Denoising(_videoFrame.Buffer(), 0, _height));
- EXPECT_EQ(-1, _vpm->Denoising(_videoFrame));
-
- EXPECT_EQ(-3, _vpm->BrightnessDetection(_videoFrame.Buffer(), 0, _height,
- stats));
- EXPECT_EQ(-3, _vpm->BrightnessDetection(_videoFrame, stats));
-
-
- // Bad height
- _videoFrame.SetWidth(_width);
- _videoFrame.SetHeight(0);
- EXPECT_EQ(-3, _vpm->GetFrameStats(stats, _videoFrame.Buffer(), _width, 0));
- EXPECT_EQ(-3, _vpm->GetFrameStats(stats, _videoFrame));
-
- EXPECT_EQ(-1, _vpm->ColorEnhancement(_videoFrame.Buffer(), _width, 0));
- EXPECT_EQ(-1, _vpm->ColorEnhancement(_videoFrame));
-
- EXPECT_EQ(-1, _vpm->Deflickering(_videoFrame.Buffer(), _width, 0, 0,
- stats));
- EXPECT_EQ(-1, _vpm->Deflickering(_videoFrame, stats));
-
- EXPECT_EQ(-1, _vpm->Denoising(_videoFrame.Buffer(), _width, 0));
- EXPECT_EQ(-1, _vpm->Denoising(_videoFrame));
-
- EXPECT_EQ(-3, _vpm->BrightnessDetection(_videoFrame.Buffer(), _width, 0,
- stats));
- EXPECT_EQ(-3, _vpm->BrightnessDetection(_videoFrame, stats));
-
- EXPECT_EQ(VPM_PARAMETER_ERROR, _vpm->SetTargetResolution(0,0,0));
- EXPECT_EQ(VPM_PARAMETER_ERROR, _vpm->SetMaxFrameRate(0));
-
- VideoFrame *outFrame = NULL;
- EXPECT_EQ(VPM_PARAMETER_ERROR, _vpm->PreprocessFrame(&_videoFrame,
- &outFrame));
-}
-
-TEST_F(VideoProcessingModuleTest, IdenticalResultsAfterReset)
-{
- VideoFrame videoFrame2;
- VideoProcessingModule::FrameStats stats;
-
- ASSERT_EQ(0, videoFrame2.VerifyAndAllocate(_frameLength));
- videoFrame2.SetWidth(_width);
- videoFrame2.SetHeight(_height);
-
- // Only testing non-static functions here.
- ASSERT_EQ(_frameLength, fread(_videoFrame.Buffer(), 1, _frameLength,
- _sourceFile));
- ASSERT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
- memcpy(videoFrame2.Buffer(), _videoFrame.Buffer(), _frameLength);
- ASSERT_EQ(0, _vpm->Deflickering(_videoFrame, stats));
- _vpm->Reset();
- // Retrieve frame stats again in case Deflickering() has zeroed them.
- ASSERT_EQ(0, _vpm->GetFrameStats(stats, videoFrame2));
- ASSERT_EQ(0, _vpm->Deflickering(videoFrame2, stats));
- EXPECT_EQ(0, memcmp(_videoFrame.Buffer(), videoFrame2.Buffer(),
- _frameLength));
-
- ASSERT_EQ(_frameLength, fread(_videoFrame.Buffer(), 1, _frameLength,
- _sourceFile));
- memcpy(videoFrame2.Buffer(), _videoFrame.Buffer(), _frameLength);
- ASSERT_GE(_vpm->Denoising(_videoFrame), 0);
- _vpm->Reset();
- ASSERT_GE(_vpm->Denoising(videoFrame2), 0);
- EXPECT_EQ(0, memcmp(_videoFrame.Buffer(), videoFrame2.Buffer(),
- _frameLength));
-
- ASSERT_EQ(_frameLength, fread(_videoFrame.Buffer(), 1, _frameLength,
- _sourceFile));
- ASSERT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
- memcpy(videoFrame2.Buffer(), _videoFrame.Buffer(), _frameLength);
- ASSERT_EQ(0, _vpm->BrightnessDetection(_videoFrame, stats));
- _vpm->Reset();
- ASSERT_EQ(0, _vpm->BrightnessDetection(videoFrame2, stats));
- EXPECT_EQ(0, memcmp(_videoFrame.Buffer(), videoFrame2.Buffer(),
- _frameLength));
-}
-
-TEST_F(VideoProcessingModuleTest, FrameStats)
-{
- VideoProcessingModule::FrameStats stats;
- ASSERT_EQ(_frameLength, fread(_videoFrame.Buffer(), 1, _frameLength,
- _sourceFile));
-
- EXPECT_FALSE(_vpm->ValidFrameStats(stats));
- EXPECT_EQ(0, _vpm->GetFrameStats(stats, _videoFrame));
- EXPECT_TRUE(_vpm->ValidFrameStats(stats));
-
- printf("\nFrameStats\n");
- printf("mean: %u\nnumPixels: %u\nsubSamplWidth: "
- "%u\nsumSamplHeight: %u\nsum: %u\n\n",
- static_cast<unsigned int>(stats.mean),
- static_cast<unsigned int>(stats.numPixels),
- static_cast<unsigned int>(stats.subSamplHeight),
- static_cast<unsigned int>(stats.subSamplWidth),
- static_cast<unsigned int>(stats.sum));
-
- _vpm->ClearFrameStats(stats);
- EXPECT_FALSE(_vpm->ValidFrameStats(stats));
-}
-
-TEST_F(VideoProcessingModuleTest, PreprocessorLogic)
-{
- // Disable temporal sampling
- _vpm->EnableTemporalDecimation(false);
- ASSERT_EQ(VPM_OK, _vpm->SetMaxFrameRate(30));
- ASSERT_EQ(VPM_OK, _vpm->SetTargetResolution(100, 100, 15));
- // Revert
- _vpm->EnableTemporalDecimation(true);
- ASSERT_EQ(VPM_OK, _vpm->SetTargetResolution(100, 100, 30));
- // Disable spatial sampling
- _vpm->SetInputFrameResampleMode(kNoRescaling);
- ASSERT_EQ(VPM_OK, _vpm->SetTargetResolution(100, 100, 30));
- VideoFrame *outFrame = NULL;
- ASSERT_EQ(VPM_OK, _vpm->PreprocessFrame(&_videoFrame, &outFrame));
- // No rescaling=> output frame = NULL
- ASSERT_TRUE(outFrame == NULL);
-}
-
-TEST_F(VideoProcessingModuleTest, Resampler)
-{
- enum { NumRuns = 1 };
-
- WebRtc_Word64 minRuntime = 0;
- WebRtc_Word64 avgRuntime = 0;
-
- TickTime t0;
- TickTime t1;
- TickInterval accTicks;
- WebRtc_Word32 height = 288;
- WebRtc_Word32 width = 352;
- WebRtc_Word32 lengthSourceFrame = width*height*3/2;
-
- rewind(_sourceFile);
- ASSERT_TRUE(_sourceFile != NULL) <<
- "Cannot read input file \n";
-
- // CA not needed here
- _vpm->EnableContentAnalysis(false);
- // no temporal decimation
- _vpm->EnableTemporalDecimation(false);
-
- // Reading test frame
- VideoFrame sourceFrame;
- ASSERT_EQ(0, sourceFrame.VerifyAndAllocate(lengthSourceFrame));
- EXPECT_GT(fread(sourceFrame.Buffer(), 1, lengthSourceFrame, _sourceFile), 0u);
- ASSERT_EQ(0, sourceFrame.SetLength(lengthSourceFrame));
- sourceFrame.SetHeight(height);
- sourceFrame.SetWidth(width);
-
- for (WebRtc_UWord32 runIdx = 0; runIdx < NumRuns; runIdx++)
- {
- // initiate test timer
- t0 = TickTime::Now();
-
- // kFastRescaling
- _vpm->SetInputFrameResampleMode(kFastRescaling);
- // TESTING DIFFERENT SIZES
- TestSize(sourceFrame, 100, 50, 1, _vpm); // Cut, decimation 1x, interpolate
- TestSize(sourceFrame, 352/2, 288/2, 1, _vpm); // Even decimation
- TestSize(sourceFrame, 352, 288, 1, _vpm); // No resampling
- TestSize(sourceFrame, 2*352, 2*288,1, _vpm); // Even upsampling
- TestSize(sourceFrame, 400, 256, 1, _vpm); // Upsampling 1.5x and cut
- TestSize(sourceFrame, 960, 720, 1, _vpm); // Upsampling 3.5x and cut
- TestSize(sourceFrame, 1280, 720, 1, _vpm); // Upsampling 4x and cut
-
- // kBiLinear
- _vpm->SetInputFrameResampleMode(kBiLinear);
- // TESTING DIFFERENT SIZES
- TestSize(sourceFrame, 352/4, 288/4, 2, _vpm);
- TestSize(sourceFrame, 352/2, 288/2, 2, _vpm);
- TestSize(sourceFrame, 2*352, 2*288,2, _vpm);
- TestSize(sourceFrame, 480, 640, 2, _vpm);
- TestSize(sourceFrame, 960, 720, 2, _vpm);
- TestSize(sourceFrame, 1280, 720, 2, _vpm);
- // stop timer
- t1 = TickTime::Now();
- accTicks += t1 - t0;
-
- if (accTicks.Microseconds() < minRuntime || runIdx == 0) {
- minRuntime = accTicks.Microseconds();
- }
- avgRuntime += accTicks.Microseconds();
- }
-
- sourceFrame.Free();
-
- printf("\nAverage run time = %d us / frame\n",
- //static_cast<int>(avgRuntime / frameNum / NumRuns));
- static_cast<int>(avgRuntime));
- printf("Min run time = %d us / frame\n\n",
- //static_cast<int>(minRuntime / frameNum));
- static_cast<int>(minRuntime));
-}
-
-void TestSize(VideoFrame& sourceFrame, WebRtc_UWord32 targetWidth,
- WebRtc_UWord32 targetHeight,
- WebRtc_UWord32 mode, VideoProcessingModule *vpm)
-{
- VideoFrame *outFrame = NULL;
- std::ostringstream filename;
- filename << webrtc::test::OutputPath() << "Resampler_"<< mode << "_" <<
- targetWidth << "x" << targetHeight << "_30Hz_P420.yuv";
- // TODO(kjellander): Add automatic verification of these output files:
- std::cout << "Watch " << filename.str() << " and verify that it is okay."
- << std::endl;
- FILE* standAloneFile = fopen(filename.str().c_str(), "wb");
- ASSERT_EQ(VPM_OK, vpm->SetTargetResolution(targetWidth, targetHeight, 30));
- ASSERT_EQ(VPM_OK, vpm->PreprocessFrame(&sourceFrame, &outFrame));
- // Length should be updated only if frame was resampled
- if (targetWidth != sourceFrame.Width() ||
- targetHeight != sourceFrame.Height()) {
- ASSERT_EQ((targetWidth * targetHeight * 3 / 2), outFrame->Length());
- // Write to file for visual inspection
- fwrite(outFrame->Buffer(), 1, outFrame->Length(), standAloneFile);
- outFrame->Free();
- }
- fclose(standAloneFile);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/unit_test.h b/trunk/src/modules/video_processing/main/test/unit_test/unit_test.h
deleted file mode 100644
index 2363e1a..0000000
--- a/trunk/src/modules/video_processing/main/test/unit_test/unit_test.h
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_TEST_UNIT_TEST_VPM_UNIT_TEST_H
-#define WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_TEST_UNIT_TEST_VPM_UNIT_TEST_H
-
-#include "gtest/gtest.h"
-#include "modules/video_processing/main/interface/video_processing.h"
-#include "system_wrappers/interface/trace.h"
-#include "testsupport/fileutils.h"
-
-namespace webrtc {
-
-class VideoProcessingModuleTest : public ::testing::Test
-{
-protected:
- VideoProcessingModuleTest();
- virtual void SetUp();
- virtual void TearDown();
- static void SetUpTestCase()
- {
- Trace::CreateTrace();
- std::string trace_file = webrtc::test::OutputPath() + "VPMTrace.txt";
- ASSERT_EQ(0, Trace::SetTraceFile(trace_file.c_str()));
- }
- static void TearDownTestCase()
- {
- Trace::ReturnTrace();
- }
- VideoProcessingModule* _vpm;
- FILE* _sourceFile;
- VideoFrame _videoFrame;
- const WebRtc_UWord32 _width;
- const WebRtc_UWord32 _height;
- const WebRtc_UWord32 _frameLength;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_PROCESSING_MAIN_TEST_UNIT_TEST_VPM_UNIT_TEST_H
diff --git a/trunk/src/modules/video_processing/main/test/unit_test/writeYUV420file.m b/trunk/src/modules/video_processing/main/test/unit_test/writeYUV420file.m
deleted file mode 100644
index 69a8808..0000000
--- a/trunk/src/modules/video_processing/main/test/unit_test/writeYUV420file.m
+++ /dev/null
@@ -1,22 +0,0 @@
-function writeYUV420file(filename, Y, U, V)
-% writeYUV420file(filename, Y, U, V)
-
-fid = fopen(filename,'wb');
-if fid==-1
- error(['Cannot open file ' filename]);
-end
-
-numFrames=size(Y,3);
-
-for k=1:numFrames
- % Write luminance
- fwrite(fid,uint8(Y(:,:,k).'), 'uchar');
-
- % Write U channel
- fwrite(fid,uint8(U(:,:,k).'), 'uchar');
-
- % Write V channel
- fwrite(fid,uint8(V(:,:,k).'), 'uchar');
-end
-
-fclose(fid);
diff --git a/trunk/src/modules/video_processing/main/test/vpm_tests.gypi b/trunk/src/modules/video_processing/main/test/vpm_tests.gypi
deleted file mode 100644
index e53ac07..0000000
--- a/trunk/src/modules/video_processing/main/test/vpm_tests.gypi
+++ /dev/null
@@ -1,39 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'video_processing_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'video_processing',
- 'webrtc_utility',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'sources': [
- # headers
- 'unit_test/unit_test.h',
- # sources
- 'unit_test/brightness_detection_test.cc',
- 'unit_test/color_enhancement_test.cc',
- 'unit_test/content_metrics_test.cc',
- 'unit_test/deflickering_test.cc',
- 'unit_test/denoising_test.cc',
- 'unit_test/unit_test.cc',
- ], # sources
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/video_render/OWNERS b/trunk/src/modules/video_render/OWNERS
deleted file mode 100644
index ac607bd..0000000
--- a/trunk/src/modules/video_render/OWNERS
+++ /dev/null
@@ -1,4 +0,0 @@
-mflodman@webrtc.org
-perkj@webrtc.org
-wu@webrtc.org
-mallinath@webrtc.org
diff --git a/trunk/src/modules/video_render/main/interface/video_render.h b/trunk/src/modules/video_render/main/interface/video_render.h
deleted file mode 100644
index 42fce70..0000000
--- a/trunk/src/modules/video_render/main/interface/video_render.h
+++ /dev/null
@@ -1,280 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_H_
-
-/*
- * video_render.h
- *
- * This header file together with module.h and module_common_types.h
- * contains all of the APIs that are needed for using the video render
- * module class.
- *
- */
-
-#include "module.h"
-#include "video_render_defines.h"
-
-namespace webrtc {
-// Class definitions
-class VideoRender: public Module
-{
-public:
- /*
- * Create a video render module object
- *
- * id - unique identifier of this video render module object
- * window - pointer to the window to render to
- * fullscreen - true if this is a fullscreen renderer
- * videoRenderType - type of renderer to create
- */
- static VideoRender
- * CreateVideoRender(
- const WebRtc_Word32 id,
- void* window,
- const bool fullscreen,
- const VideoRenderType videoRenderType =
- kRenderDefault);
-
- /*
- * Destroy a video render module object
- *
- * module - object to destroy
- */
- static void DestroyVideoRender(VideoRender* module);
-
- /*
- * Change the unique identifier of this object
- *
- * id - new unique identifier of this video render module object
- */
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0;
-
- virtual WebRtc_Word32 TimeUntilNextProcess() = 0;
- virtual WebRtc_Word32 Process() = 0;
-
- /**************************************************************************
- *
- * Window functions
- *
- ***************************************************************************/
-
- /*
- * Get window for this renderer
- */
- virtual void* Window() = 0;
-
- /*
- * Change render window
- *
- * window - the new render window, assuming same type as originally created.
- */
- virtual WebRtc_Word32 ChangeWindow(void* window) = 0;
-
- /**************************************************************************
- *
- * Incoming Streams
- *
- ***************************************************************************/
-
- /*
- * Add incoming render stream
- *
- * streamID - id of the stream to add
- * zOrder - relative render order for the streams, 0 = on top
- * left - position of the stream in the window, [0.0f, 1.0f]
- * top - position of the stream in the window, [0.0f, 1.0f]
- * right - position of the stream in the window, [0.0f, 1.0f]
- * bottom - position of the stream in the window, [0.0f, 1.0f]
- *
- * Return - callback class to use for delivering new frames to render.
- */
- virtual VideoRenderCallback
- * AddIncomingRenderStream(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left, const float top,
- const float right, const float bottom) = 0;
- /*
- * Delete incoming render stream
- *
- * streamID - id of the stream to add
- */
- virtual WebRtc_Word32
- DeleteIncomingRenderStream(const WebRtc_UWord32 streamId) = 0;
-
- /*
- * Add incoming render callback, used for external rendering
- *
- * streamID - id of the stream the callback is used for
- * renderObject - the VideoRenderCallback to use for this stream, NULL to remove
- *
- * Return - callback class to use for delivering new frames to render.
- */
- virtual WebRtc_Word32
- AddExternalRenderCallback(const WebRtc_UWord32 streamId,
- VideoRenderCallback* renderObject) = 0;
-
- /*
- * Get the porperties for an incoming render stream
- *
- * streamID - [in] id of the stream to get properties for
- * zOrder - [out] relative render order for the streams, 0 = on top
- * left - [out] position of the stream in the window, [0.0f, 1.0f]
- * top - [out] position of the stream in the window, [0.0f, 1.0f]
- * right - [out] position of the stream in the window, [0.0f, 1.0f]
- * bottom - [out] position of the stream in the window, [0.0f, 1.0f]
- */
- virtual WebRtc_Word32
- GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
- WebRtc_UWord32& zOrder,
- float& left, float& top,
- float& right, float& bottom) const = 0;
- /*
- * The incoming frame rate to the module, not the rate rendered in the window.
- */
- virtual WebRtc_UWord32
- GetIncomingFrameRate(const WebRtc_UWord32 streamId) = 0;
-
- /*
- * Returns the number of incoming streams added to this render module
- */
- virtual WebRtc_UWord32 GetNumIncomingRenderStreams() const = 0;
-
- /*
- * Returns true if this render module has the streamId added, false otherwise.
- */
- virtual bool
- HasIncomingRenderStream(const WebRtc_UWord32 streamId) const = 0;
-
- /*
- * Registers a callback to get raw images in the same time as sent
- * to the renderer. To be used for external rendering.
- */
- virtual WebRtc_Word32
- RegisterRawFrameCallback(const WebRtc_UWord32 streamId,
- VideoRenderCallback* callbackObj) = 0;
-
- /*
- * This method is usefull to get last rendered frame for the stream specified
- */
- virtual WebRtc_Word32
- GetLastRenderedFrame(const WebRtc_UWord32 streamId,
- VideoFrame &frame) const = 0;
-
- /**************************************************************************
- *
- * Start/Stop
- *
- ***************************************************************************/
-
- /*
- * Starts rendering the specified stream
- */
- virtual WebRtc_Word32 StartRender(const WebRtc_UWord32 streamId) = 0;
-
- /*
- * Stops the renderer
- */
- virtual WebRtc_Word32 StopRender(const WebRtc_UWord32 streamId) = 0;
-
- /*
- * Resets the renderer
- * No streams are removed. The state should be as after AddStream was called.
- */
- virtual WebRtc_Word32 ResetRender() = 0;
-
- /**************************************************************************
- *
- * Properties
- *
- ***************************************************************************/
-
- /*
- * Returns the preferred render video type
- */
- virtual RawVideoType PreferredVideoType() const = 0;
-
- /*
- * Returns true if the renderer is in fullscreen mode, otherwise false.
- */
- virtual bool IsFullScreen() = 0;
-
- /*
- * Gets screen resolution in pixels
- */
- virtual WebRtc_Word32
- GetScreenResolution(WebRtc_UWord32& screenWidth,
- WebRtc_UWord32& screenHeight) const = 0;
-
- /*
- * Get the actual render rate for this stream. I.e rendered frame rate,
- * not frames delivered to the renderer.
- */
- virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId) = 0;
-
- /*
- * Set cropping of incoming stream
- */
- virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
- const float left, const float top,
- const float right,
- const float bottom) = 0;
-
- /*
- * re-configure renderer
- */
- virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
- const unsigned int zOrder,
- const float left, const float top,
- const float right,
- const float bottom) = 0;
-
- virtual WebRtc_Word32 SetTransparentBackground(const bool enable) = 0;
-
- virtual WebRtc_Word32 FullScreenRender(void* window, const bool enable) = 0;
-
- virtual WebRtc_Word32 SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey, const float left,
- const float top, const float right,
- const float bottom) = 0;
-
- virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 textColorRef,
- const WebRtc_UWord32 backgroundColorRef,
- const float left, const float top,
- const float right, const float bottom) = 0;
-
- /*
- * Set a start image. The image is rendered before the first image has been delivered
- */
- virtual WebRtc_Word32
- SetStartImage(const WebRtc_UWord32 streamId,
- const VideoFrame& videoFrame) = 0;
-
- /*
- * Set a timout image. The image is rendered if no videoframe has been delivered
- */
- virtual WebRtc_Word32 SetTimeoutImage(const WebRtc_UWord32 streamId,
- const VideoFrame& videoFrame,
- const WebRtc_UWord32 timeout)= 0;
-
- virtual WebRtc_Word32 MirrorRenderStream(const int renderId,
- const bool enable,
- const bool mirrorXAxis,
- const bool mirrorYAxis) = 0;
-
- static WebRtc_Word32 SetAndroidObjects(void* javaVM);
-};
-} //namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_H_
diff --git a/trunk/src/modules/video_render/main/interface/video_render_defines.h b/trunk/src/modules/video_render/main/interface/video_render_defines.h
deleted file mode 100644
index 2a87f3f..0000000
--- a/trunk/src/modules/video_render/main/interface/video_render_defines.h
+++ /dev/null
@@ -1,92 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_DEFINES_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_DEFINES_H_
-
-// Includes
-#include "common_types.h"
-#include "module_common_types.h"
-
-namespace webrtc
-{
-// Defines
-#ifndef NULL
-#define NULL 0
-#endif
-
-// Enums
-enum VideoRenderType
-{
- kRenderExternal = 0, // External
- kRenderWindows = 1, // Windows
- kRenderCocoa = 2, // Mac
- kRenderCarbon = 3,
- kRenderiPhone = 4, // iPhone
- kRenderAndroid = 5, // Android
- kRenderX11 = 6, // Linux
- kRenderDefault
-};
-
-// Runtime errors
-enum VideoRenderError
-{
- kRenderShutDown = 0,
- kRenderPerformanceAlarm = 1
-};
-
-// The object a module user uses to send new frames to the renderer
-// One object is used for each incoming stream
-class VideoRenderCallback
-{
-public:
- virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId,
- VideoFrame& videoFrame) = 0;
-
-protected:
- virtual ~VideoRenderCallback()
- {
- }
-};
-
-// Feedback class to be implemented by module user
-class VideoRenderFeedback
-{
-public:
- virtual void OnRenderError(const WebRtc_Word32 streamId,
- const VideoRenderError error) = 0;
-
-protected:
- virtual ~VideoRenderFeedback()
- {
- }
-};
-
-// Mobile enums
-enum StretchMode
-{
- kStretchToInsideEdge = 1,
- kStretchToOutsideEdge = 2,
- kStretchMatchWidth = 3,
- kStretchMatchHeight = 4,
- kStretchNone = 5
-};
-
-enum Rotation
-{
- kRotation0 = 0,
- kRotation90 = 1,
- kRotation180 = 2,
- kRotation270 = 3
-};
-
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_INTERFACE_VIDEO_RENDER_DEFINES_H_
diff --git a/trunk/src/modules/video_render/main/source/Android.mk b/trunk/src/modules/video_render/main/source/Android.mk
deleted file mode 100644
index 73eec93..0000000
--- a/trunk/src/modules/video_render/main/source/Android.mk
+++ /dev/null
@@ -1,53 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../android-webrtc.mk
-
-LOCAL_MODULE_CLASS := STATIC_LIBRARIES
-LOCAL_MODULE := libwebrtc_video_render
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := \
- incoming_video_stream.cc \
- video_render_frames.cc \
- video_render_impl.cc \
- external/video_render_external_impl.cc \
- Android/video_render_android_impl.cc \
- Android/video_render_android_native_opengl2.cc \
- Android/video_render_android_surface_view.cc \
- Android/video_render_opengles20.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS) \
- '-DWEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER'
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH) \
- $(LOCAL_PATH)/Android \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../../.. \
- $(LOCAL_PATH)/../../../audio_coding/main/interface \
- $(LOCAL_PATH)/../../../interface \
- $(LOCAL_PATH)/../../../utility/interface \
- $(LOCAL_PATH)/../../../../common_video/vplib/main/interface \
- $(LOCAL_PATH)/../../../../system_wrappers/interface
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViEAndroidGLES20.java b/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViEAndroidGLES20.java
deleted file mode 100644
index fc30607..0000000
--- a/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViEAndroidGLES20.java
+++ /dev/null
@@ -1,263 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-package org.webrtc.videoengine;
-
-import java.util.concurrent.locks.ReentrantLock;
-
-import javax.microedition.khronos.egl.EGL10;
-import javax.microedition.khronos.egl.EGLConfig;
-import javax.microedition.khronos.egl.EGLContext;
-import javax.microedition.khronos.egl.EGLDisplay;
-import javax.microedition.khronos.opengles.GL10;
-
-import android.app.ActivityManager;
-import android.content.Context;
-import android.content.pm.ConfigurationInfo;
-import android.opengl.GLSurfaceView;
-import android.util.Log;
-
-public class ViEAndroidGLES20 extends GLSurfaceView
- implements GLSurfaceView.Renderer {
- // True if onSurfaceCreated has been called.
- private boolean surfaceCreated = false;
- private boolean openGLCreated = false;
- // True if NativeFunctionsRegistered has been called.
- private boolean nativeFunctionsRegisted = false;
- private ReentrantLock nativeFunctionLock = new ReentrantLock();
- // Address of Native object that will do the drawing.
- private long nativeObject = 0;
- private int viewWidth = 0;
- private int viewHeight = 0;
-
- public static boolean UseOpenGL2(Object renderWindow) {
- return ViEAndroidGLES20.class.isInstance(renderWindow);
- }
-
- public ViEAndroidGLES20(Context context) {
- super(context);
-
- // Setup the context factory for 2.0 rendering.
- // See ContextFactory class definition below
- setEGLContextFactory(new ContextFactory());
-
- // We need to choose an EGLConfig that matches the format of
- // our surface exactly. This is going to be done in our
- // custom config chooser. See ConfigChooser class definition below
- // Use RGB 565 without an alpha channel.
- setEGLConfigChooser( new ConfigChooser(5, 6, 5, 0, 0, 0) );
-
- this.setRenderer(this);
- this.setRenderMode(GLSurfaceView.RENDERMODE_WHEN_DIRTY);
- }
-
- // IsSupported
- // Return true if this device support Open GL ES 2.0 rendering.
- public static boolean IsSupported(Context context) {
- ActivityManager am =
- (ActivityManager) context.getSystemService(Context.ACTIVITY_SERVICE);
- ConfigurationInfo info = am.getDeviceConfigurationInfo();
- if(info.reqGlEsVersion >= 0x20000) {
- // Open GL ES 2.0 is supported.
- return true;
- }
- return false;
- }
-
- public void onDrawFrame(GL10 gl) {
- nativeFunctionLock.lock();
- if(!nativeFunctionsRegisted || !surfaceCreated) {
- nativeFunctionLock.unlock();
- return;
- }
-
- if(!openGLCreated) {
- if(0 != CreateOpenGLNative(nativeObject, viewWidth, viewHeight)) {
- return; // Failed to create OpenGL
- }
- openGLCreated = true; // Created OpenGL successfully
- }
- DrawNative(nativeObject); // Draw the new frame
- nativeFunctionLock.unlock();
- }
-
- public void onSurfaceChanged(GL10 gl, int width, int height) {
- surfaceCreated = true;
- viewWidth = width;
- viewHeight = height;
-
- nativeFunctionLock.lock();
- if(nativeFunctionsRegisted) {
- if(CreateOpenGLNative(nativeObject,width,height) == 0)
- openGLCreated = true;
- }
- nativeFunctionLock.unlock();
- }
-
- public void onSurfaceCreated(GL10 gl, EGLConfig config) {
- }
-
- public void RegisterNativeObject(long nativeObject) {
- nativeFunctionLock.lock();
- nativeObject = nativeObject;
- nativeFunctionsRegisted = true;
- nativeFunctionLock.unlock();
- }
-
- public void DeRegisterNativeObject() {
- nativeFunctionLock.lock();
- nativeFunctionsRegisted = false;
- openGLCreated = false;
- nativeObject = 0;
- nativeFunctionLock.unlock();
- }
-
- public void ReDraw() {
- if(surfaceCreated) {
- // Request the renderer to redraw using the render thread context.
- this.requestRender();
- }
- }
-
- // EGL Context factory used for creating EGL 2.0 context
- // on Android 2.1(and later,
- // though there are simpler ways in 2.2)
- // Code is from the NDK samples\hello-gl2\src\com\android\gl2jni.
- private static class ContextFactory
- implements GLSurfaceView.EGLContextFactory {
- private static int EGL_CONTEXT_CLIENT_VERSION = 0x3098;
- public EGLContext createContext(EGL10 egl,
- EGLDisplay display,
- EGLConfig eglConfig) {
- //checkEglError("Before eglCreateContext", egl);
- int[] attrib_list = {EGL_CONTEXT_CLIENT_VERSION, 2, EGL10.EGL_NONE };
- // Create an Open GL ES 2.0 context
- EGLContext context = egl.eglCreateContext(display,
- eglConfig,
- EGL10.EGL_NO_CONTEXT,
- attrib_list);
- checkEglError("ContextFactory eglCreateContext", egl);
- return context;
- }
-
- public void destroyContext(EGL10 egl, EGLDisplay display,
- EGLContext context) {
- egl.eglDestroyContext(display, context);
- }
- }
-
- private static void checkEglError(String prompt, EGL10 egl) {
- int error;
- while ((error = egl.eglGetError()) != EGL10.EGL_SUCCESS) {
- Log.e("*WEBRTC*", String.format("%s: EGL error: 0x%x", prompt, error));
- }
- }
-
- // Code is from the NDK samples\hello-gl2\src\com\android\gl2jni
- private static class ConfigChooser
- implements GLSurfaceView.EGLConfigChooser {
-
- public ConfigChooser(int r, int g, int b, int a, int depth, int stencil) {
- mRedSize = r;
- mGreenSize = g;
- mBlueSize = b;
- mAlphaSize = a;
- mDepthSize = depth;
- mStencilSize = stencil;
- }
-
- // This EGL config specification is used to specify 2.0 rendering.
- // We use a minimum size of 4 bits for red/green/blue, but will
- // perform actual matching in chooseConfig() below.
- private static int EGL_OPENGL_ES2_BIT = 4;
- private static int[] s_configAttribs2 =
- {
- EGL10.EGL_RED_SIZE, 4,
- EGL10.EGL_GREEN_SIZE, 4,
- EGL10.EGL_BLUE_SIZE, 4,
- EGL10.EGL_RENDERABLE_TYPE, EGL_OPENGL_ES2_BIT,
- EGL10.EGL_NONE
- };
-
- public EGLConfig chooseConfig(EGL10 egl, EGLDisplay display) {
-
- // Get the number of minimally matching EGL configurations
- int[] num_config = new int[1];
- egl.eglChooseConfig(display, s_configAttribs2, null, 0, num_config);
-
- int numConfigs = num_config[0];
-
- if (numConfigs <= 0) {
- throw new IllegalArgumentException("No configs match configSpec");
- }
-
- // Allocate then read the array of minimally matching EGL configs
- EGLConfig[] configs = new EGLConfig[numConfigs];
- egl.eglChooseConfig(display, s_configAttribs2, configs,
- numConfigs, num_config);
-
- // Now return the "best" one
- return chooseConfig(egl, display, configs);
- }
-
- public EGLConfig chooseConfig(EGL10 egl, EGLDisplay display,
- EGLConfig[] configs) {
- for(EGLConfig config : configs) {
- int d = findConfigAttrib(egl, display, config,
- EGL10.EGL_DEPTH_SIZE, 0);
- int s = findConfigAttrib(egl, display, config,
- EGL10.EGL_STENCIL_SIZE, 0);
-
- // We need at least mDepthSize and mStencilSize bits
- if (d < mDepthSize || s < mStencilSize)
- continue;
-
- // We want an *exact* match for red/green/blue/alpha
- int r = findConfigAttrib(egl, display, config,
- EGL10.EGL_RED_SIZE, 0);
- int g = findConfigAttrib(egl, display, config,
- EGL10.EGL_GREEN_SIZE, 0);
- int b = findConfigAttrib(egl, display, config,
- EGL10.EGL_BLUE_SIZE, 0);
- int a = findConfigAttrib(egl, display, config,
- EGL10.EGL_ALPHA_SIZE, 0);
-
- if (r == mRedSize && g == mGreenSize &&
- b == mBlueSize && a == mAlphaSize)
- return config;
- }
- return null;
- }
-
- private int findConfigAttrib(EGL10 egl, EGLDisplay display,
- EGLConfig config, int attribute,
- int defaultValue) {
-
- if (egl.eglGetConfigAttrib(display, config, attribute, mValue)) {
- return mValue[0];
- }
- return defaultValue;
- }
-
- // Subclasses can adjust these values:
- protected int mRedSize;
- protected int mGreenSize;
- protected int mBlueSize;
- protected int mAlphaSize;
- protected int mDepthSize;
- protected int mStencilSize;
- private int[] mValue = new int[1];
- }
-
- private native int CreateOpenGLNative(long nativeObject,
- int width, int height);
- private native void DrawNative(long nativeObject);
-
-}
diff --git a/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViERenderer.java b/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViERenderer.java
deleted file mode 100644
index 56d5261..0000000
--- a/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViERenderer.java
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-package org.webrtc.videoengine;
-
-import android.content.Context;
-import android.view.SurfaceHolder;
-import android.view.SurfaceView;
-
-public class ViERenderer {
-
- // View used for local rendering that Cameras can use for Video Overlay.
- private static SurfaceHolder g_localRenderer;
-
- public static SurfaceView CreateRenderer(Context context) {
- return CreateRenderer(context,false);
- }
-
- public static SurfaceView CreateRenderer(Context context,
- boolean useOpenGLES2) {
- if(useOpenGLES2 == true && ViEAndroidGLES20.IsSupported(context))
- return new ViEAndroidGLES20(context);
- else
- return new SurfaceView(context);
- }
-
- // Creates a SurfaceView to be used by Android Camera
- // service to display a local preview.
- // This needs to be used on Android prior to version 2.1
- // in order to run the camera.
- // Call this function before ViECapture::StartCapture.
- // The created view needs to be added to a visible layout
- // after a camera has been allocated
- // (with the call ViECapture::AllocateCaptureDevice).
- // IE.
- // CreateLocalRenderer
- // ViECapture::AllocateCaptureDevice
- // LinearLayout.addview
- // ViECapture::StartCapture
- public static SurfaceView CreateLocalRenderer(Context context) {
- SurfaceView localRender = new SurfaceView(context);
- g_localRenderer = localRender.getHolder();
- g_localRenderer.setType(SurfaceHolder.SURFACE_TYPE_PUSH_BUFFERS);
- return localRender;
- }
-
- public static SurfaceHolder GetLocalRenderer() {
- return g_localRenderer;
- }
-
-}
diff --git a/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViESurfaceRenderer.java b/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViESurfaceRenderer.java
deleted file mode 100644
index 3412582..0000000
--- a/trunk/src/modules/video_render/main/source/Android/java/org/webrtc/videoengine/ViESurfaceRenderer.java
+++ /dev/null
@@ -1,155 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-package org.webrtc.videoengine;
-
-import java.nio.ByteBuffer;
-
-import android.graphics.Bitmap;
-import android.graphics.Canvas;
-import android.graphics.Rect;
-import android.util.Log;
-import android.view.SurfaceHolder;
-import android.view.SurfaceView;
-import android.view.SurfaceHolder.Callback;
-
-public class ViESurfaceRenderer implements Callback {
-
- // the bitmap used for drawing.
- private Bitmap bitmap = null;
- private ByteBuffer byteBuffer;
- private SurfaceHolder surfaceHolder;
- // Rect of the source bitmap to draw
- private Rect srcRect = new Rect();
- // Rect of the destination canvas to draw to
- private Rect dstRect = new Rect();
- private int dstHeight = 0;
- private int dstWidth = 0;
- private float dstTopScale = 0;
- private float dstBottomScale = 1;
- private float dstLeftScale = 0;
- private float dstRightScale = 1;
-
- public ViESurfaceRenderer(SurfaceView view) {
- surfaceHolder = view.getHolder();
- if(surfaceHolder == null)
- return;
-
- Canvas canvas = surfaceHolder.lockCanvas();
- if(canvas != null) {
- Rect dst =surfaceHolder.getSurfaceFrame();
- if(dst != null) {
- dstRect = dst;
- dstHeight =dstRect.bottom-dstRect.top;
- dstWidth = dstRect.right-dstRect.left;
- }
- surfaceHolder.unlockCanvasAndPost(canvas);
- }
-
- surfaceHolder.addCallback(this);
- }
-
- public void surfaceChanged(SurfaceHolder holder, int format,
- int in_width, int in_height) {
-
- dstHeight = in_height;
- dstWidth = in_width;
- dstRect.left = (int)(dstLeftScale*dstWidth);
- dstRect.top = (int)(dstTopScale*dstHeight);
- dstRect.bottom = (int)(dstBottomScale*dstHeight);
- dstRect.right = (int) (dstRightScale*dstWidth);
- }
-
- public void surfaceCreated(SurfaceHolder holder) {
- // TODO(leozwang) Auto-generated method stub
- }
-
- public void surfaceDestroyed(SurfaceHolder holder) {
- // TODO(leozwang) Auto-generated method stub
- }
-
- public Bitmap CreateBitmap(int width, int height) {
- if (bitmap == null) {
- try {
- android.os.Process.setThreadPriority(
- android.os.Process.THREAD_PRIORITY_DISPLAY);
- }
- catch (Exception e) {
- }
- }
- bitmap = Bitmap.createBitmap(width, height, Bitmap.Config.RGB_565);
- srcRect.left = 0;
- srcRect.top = 0;
- srcRect.bottom = height;
- srcRect.right = width;
-
- return bitmap;
- }
-
- public ByteBuffer CreateByteBuffer(int width, int height) {
- if (bitmap == null) {
- try {
- android.os.Process
- .setThreadPriority(android.os.Process.THREAD_PRIORITY_DISPLAY);
- }
- catch (Exception e) {
- }
- }
-
- try {
- bitmap = Bitmap.createBitmap(width, height, Bitmap.Config.RGB_565);
- byteBuffer = ByteBuffer.allocateDirect(width*height*2);
- srcRect.left = 0;
- srcRect.top = 0;
- srcRect.bottom = height;
- srcRect.right = width;
- }
- catch (Exception ex) {
- Log.e("*WEBRTC*", "Failed to CreateByteBuffer");
- bitmap = null;
- byteBuffer = null;
- }
-
- return byteBuffer;
- }
-
- public void SetCoordinates(float left, float top,
- float right, float bottom) {
- dstLeftScale = left;
- dstTopScale = top;
- dstRightScale = right;
- dstBottomScale = bottom;
-
- dstRect.left = (int)(dstLeftScale*dstWidth);
- dstRect.top = (int)(dstTopScale*dstHeight);
- dstRect.bottom = (int)(dstBottomScale*dstHeight);
- dstRect.right = (int) (dstRightScale*dstWidth);
- }
-
- public void DrawByteBuffer() {
- if(byteBuffer == null)
- return;
- byteBuffer.rewind();
- bitmap.copyPixelsFromBuffer(byteBuffer);
- DrawBitmap();
- }
-
- public void DrawBitmap() {
- if(bitmap == null)
- return;
-
- Canvas canvas = surfaceHolder.lockCanvas();
- if(canvas != null) {
- canvas.drawBitmap(bitmap, srcRect, dstRect, null);
- surfaceHolder.unlockCanvasAndPost(canvas);
- }
- }
-
-}
diff --git a/trunk/src/modules/video_render/main/source/Android/video_render_android_impl.cc b/trunk/src/modules/video_render/main/source/Android/video_render_android_impl.cc
deleted file mode 100644
index 22fa659..0000000
--- a/trunk/src/modules/video_render/main/source/Android/video_render_android_impl.cc
+++ /dev/null
@@ -1,392 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_render_android_impl.h"
-
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "tick_util.h"
-
-#ifdef ANDROID_LOG
-#include <stdio.h>
-#include <android/log.h>
-
-#undef WEBRTC_TRACE
-#define WEBRTC_TRACE(a,b,c,...) __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTCN*", __VA_ARGS__)
-#else
-#include "trace.h"
-#endif
-
-namespace webrtc {
-JavaVM* VideoRenderAndroid::g_jvm = NULL;
-
-WebRtc_Word32 VideoRenderAndroid::SetAndroidEnvVariables(void* javaVM)
-{
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, -1, "%s", __FUNCTION__);
-
- g_jvm = (JavaVM*) javaVM;
-
- return 0;
-
-}
-
-VideoRenderAndroid::VideoRenderAndroid(
- const WebRtc_Word32 id,
- const VideoRenderType videoRenderType,
- void* window,
- const bool /*fullscreen*/):
- _id(id),
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _renderType(videoRenderType),
- _ptrWindow((jobject)(window)),
- _streamsMap(),
- _javaShutDownFlag(false),
- _javaShutdownEvent(*EventWrapper::Create()),
- _javaRenderEvent(*EventWrapper::Create()),
- _lastJavaRenderEvent(0),
- _javaRenderJniEnv(NULL),
- _javaRenderThread(NULL)
-{
-}
-
-VideoRenderAndroid::~VideoRenderAndroid()
-{
-
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
- "VideoRenderAndroid dtor");
-
- if (_javaRenderThread)
- StopRender();
-
- for (MapItem* item = _streamsMap.First(); item != NULL; item
- = _streamsMap.Next(item))
- { // Delete streams
- delete static_cast<AndroidStream*> (item->GetItem());
- }
- delete &_javaShutdownEvent;
- delete &_javaRenderEvent;
- delete &_critSect;
-}
-
-WebRtc_Word32 VideoRenderAndroid::ChangeUniqueId(const WebRtc_Word32 id)
-{
- CriticalSectionScoped cs(_critSect);
- _id = id;
-
- return 0;
-}
-
-WebRtc_Word32 VideoRenderAndroid::ChangeWindow(void* /*window*/)
-{
- return -1;
-}
-
-VideoRenderCallback*
-VideoRenderAndroid::AddIncomingRenderStream(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left, const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_critSect);
-
- AndroidStream* renderStream = NULL;
- MapItem* item = _streamsMap.Find(streamId);
- if (item)
- {
- renderStream = (AndroidStream*) (item->GetItem());
- if (NULL != renderStream)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, -1,
- "%s: Render stream already exists", __FUNCTION__);
- return renderStream;
- }
- }
-
- renderStream = CreateAndroidRenderChannel(streamId, zOrder, left, top,
- right, bottom, *this);
- if (renderStream)
- {
- _streamsMap.Insert(streamId, renderStream);
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "(%s:%d): renderStream is NULL", __FUNCTION__, __LINE__);
- return NULL;
- }
- return renderStream;
-}
-
-WebRtc_Word32 VideoRenderAndroid::DeleteIncomingRenderStream(
- const WebRtc_UWord32 streamId)
-{
- CriticalSectionScoped cs(_critSect);
-
- MapItem* item = _streamsMap.Find(streamId);
- if (item)
- {
- delete (AndroidStream*) item->GetItem();
- _streamsMap.Erase(streamId);
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "(%s:%d): renderStream is NULL", __FUNCTION__, __LINE__);
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 VideoRenderAndroid::GetIncomingRenderStreamProperties(
- const WebRtc_UWord32 streamId,
- WebRtc_UWord32& zOrder,
- float& left,
- float& top,
- float& right,
- float& bottom) const
-{
-
- return -1;
-}
-
-WebRtc_Word32 VideoRenderAndroid::StartRender()
-{
- CriticalSectionScoped cs(_critSect);
-
- if (_javaRenderThread)
- {
- // StartRender is called when this stream should start render.
- // However StopRender is not called when the streams stop rendering. Thus the the thread is only deleted when the renderer is removed.
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
- "%s, Render thread already exist", __FUNCTION__);
- return 0;
- }
-
- _javaRenderThread = ThreadWrapper::CreateThread(JavaRenderThreadFun, this,
- kRealtimePriority,
- "AndroidRenderThread");
- if (!_javaRenderThread)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No thread", __FUNCTION__);
- return -1;
- }
-
- unsigned int tId = 0;
- if (_javaRenderThread->Start(tId))
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
- "%s: thread started: %u", __FUNCTION__, tId);
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: Could not start send thread", __FUNCTION__);
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 VideoRenderAndroid::StopRender()
-{
-
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:", __FUNCTION__);
- {
- CriticalSectionScoped cs(_critSect);
- if (!_javaRenderThread)
- {
- return -1;
- }
- _javaShutDownFlag = true;
- _javaRenderEvent.Set();
- }
-
- _javaShutdownEvent.Wait(3000);
- CriticalSectionScoped cs(_critSect);
- _javaRenderThread->SetNotAlive();
- if (_javaRenderThread->Stop())
- {
- delete _javaRenderThread;
- _javaRenderThread = NULL;
- }
- else
- {
- assert(false);
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
- "%s: Not able to stop thread, leaking", __FUNCTION__);
- _javaRenderThread = NULL;
- }
- return 0;
-}
-
-void VideoRenderAndroid::ReDraw()
-{
- CriticalSectionScoped cs(_critSect);
- if (_lastJavaRenderEvent < TickTime::MillisecondTimestamp() - 20) // Allow redraw if it was more than 20ms since last.
- {
- _lastJavaRenderEvent = TickTime::MillisecondTimestamp();
- _javaRenderEvent.Set();
- }
-}
-
-bool VideoRenderAndroid::JavaRenderThreadFun(void* obj)
-{
- return static_cast<VideoRenderAndroid*> (obj)->JavaRenderThreadProcess();
-}
-
-bool VideoRenderAndroid::JavaRenderThreadProcess()
-{
- _javaRenderEvent.Wait(1000);
-
- CriticalSectionScoped cs(_critSect);
- if (!_javaRenderJniEnv)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = g_jvm->AttachCurrentThread(&_javaRenderJniEnv, NULL);
-
- // Get the JNI env for this thread
- if ((res < 0) || !_javaRenderJniEnv)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: Could not attach thread to JVM (%d, %p)",
- __FUNCTION__, res, _javaRenderJniEnv);
- return false;
- }
- }
-
- for (MapItem* item = _streamsMap.First(); item != NULL; item
- = _streamsMap.Next(item))
- {
- static_cast<AndroidStream*> (item->GetItem())->DeliverFrame(
- _javaRenderJniEnv);
- }
-
- if (_javaShutDownFlag)
- {
- if (g_jvm->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
- "%s: Could not detach thread from JVM", __FUNCTION__);
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
- "%s: Java thread detached", __FUNCTION__);
- }
- _javaRenderJniEnv = false;
- _javaShutDownFlag = false;
- _javaShutdownEvent.Set();
- return false; // Do not run this thread again.
- }
- return true;
-}
-
-VideoRenderType VideoRenderAndroid::RenderType()
-{
- return _renderType;
-}
-
-RawVideoType VideoRenderAndroid::PerferedVideoType()
-{
- return kVideoI420;
-}
-
-bool VideoRenderAndroid::FullScreen()
-{
- return false;
-}
-
-WebRtc_Word32 VideoRenderAndroid::GetGraphicsMemory(
- WebRtc_UWord64& /*totalGraphicsMemory*/,
- WebRtc_UWord64& /*availableGraphicsMemory*/) const
-{
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s - not supported on Android", __FUNCTION__);
- return -1;
-}
-
-WebRtc_Word32 VideoRenderAndroid::GetScreenResolution(
- WebRtc_UWord32& /*screenWidth*/,
- WebRtc_UWord32& /*screenHeight*/) const
-{
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s - not supported on Android", __FUNCTION__);
- return -1;
-}
-
-WebRtc_UWord32 VideoRenderAndroid::RenderFrameRate(const WebRtc_UWord32 /*streamId*/)
-{
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s - not supported on Android", __FUNCTION__);
- return -1;
-}
-
-WebRtc_Word32 VideoRenderAndroid::SetStreamCropping(
- const WebRtc_UWord32 /*streamId*/,
- const float /*left*/,
- const float /*top*/,
- const float /*right*/,
- const float /*bottom*/)
-{
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s - not supported on Android", __FUNCTION__);
- return -1;
-}
-
-WebRtc_Word32 VideoRenderAndroid::SetTransparentBackground(const bool enable)
-{
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s - not supported on Android", __FUNCTION__);
- return -1;
-}
-
-WebRtc_Word32 VideoRenderAndroid::ConfigureRenderer(
- const WebRtc_UWord32 streamId,
- const unsigned int zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s - not supported on Android", __FUNCTION__);
- return -1;
-}
-
-WebRtc_Word32 VideoRenderAndroid::SetText(
- const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 textColorRef,
- const WebRtc_UWord32 backgroundColorRef,
- const float left, const float top,
- const float rigth, const float bottom)
-{
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s - not supported on Android", __FUNCTION__);
- return -1;
-}
-
-WebRtc_Word32 VideoRenderAndroid::SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey,
- const float left, const float top,
- const float right,
- const float bottom)
-{
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s - not supported on Android", __FUNCTION__);
- return -1;
-}
-} //namespace webrtc
-
diff --git a/trunk/src/modules/video_render/main/source/Android/video_render_android_impl.h b/trunk/src/modules/video_render/main/source/Android/video_render_android_impl.h
deleted file mode 100644
index 7058871..0000000
--- a/trunk/src/modules/video_render/main/source/Android/video_render_android_impl.h
+++ /dev/null
@@ -1,162 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_IMPL_H_
-
-#include <jni.h>
-#include "i_video_render.h"
-#include "map_wrapper.h"
-
-
-namespace webrtc {
-
-//#define ANDROID_LOG
-
-
-class CriticalSectionWrapper;
-class EventWrapper;
-class ThreadWrapper;
-
-
-// The object a module user uses to send new frames to the java renderer
-// Base class for android render streams.
-
-class AndroidStream: public VideoRenderCallback
-{
-public:
- /*
- * DeliverFrame is called from a thread connected to the Java VM.
- * Used for Delivering frame for rendering.
- */
- virtual void DeliverFrame(JNIEnv* jniEnv)=0;
-
- virtual ~AndroidStream()
- {
- };
-};
-
-class VideoRenderAndroid: IVideoRender
-{
-public:
- static WebRtc_Word32 SetAndroidEnvVariables(void* javaVM);
-
- VideoRenderAndroid(const WebRtc_Word32 id,
- const VideoRenderType videoRenderType, void* window,
- const bool fullscreen);
-
- virtual ~VideoRenderAndroid();
-
- virtual WebRtc_Word32 Init()=0;
-
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- virtual WebRtc_Word32 ChangeWindow(void* window);
-
- virtual VideoRenderCallback
- * AddIncomingRenderStream(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left, const float top,
- const float right, const float bottom);
-
- virtual WebRtc_Word32
- DeleteIncomingRenderStream(const WebRtc_UWord32 streamId);
-
- virtual WebRtc_Word32
- GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
- WebRtc_UWord32& zOrder,
- float& left, float& top,
- float& right, float& bottom) const;
- virtual WebRtc_Word32 StartRender();
-
- virtual WebRtc_Word32 StopRender();
-
- virtual void ReDraw();
-
- /**************************************************************************
- *
- * Properties
- *
- ***************************************************************************/
-
- virtual VideoRenderType RenderType();
-
- virtual RawVideoType PerferedVideoType();
-
- virtual bool FullScreen();
-
- virtual WebRtc_Word32
- GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
- WebRtc_UWord64& availableGraphicsMemory) const;
-
- virtual WebRtc_Word32
- GetScreenResolution(WebRtc_UWord32& screenWidth,
- WebRtc_UWord32& screenHeight) const;
-
- virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId);
-
- virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
- const float left, const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
-
- virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
- const unsigned int zOrder,
- const float left, const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 textColorRef,
- const WebRtc_UWord32 backgroundColorRef,
- const float left, const float top,
- const float rigth, const float bottom);
-
- virtual WebRtc_Word32 SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey, const float left,
- const float top, const float right,
- const float bottom);
-
-protected:
- virtual AndroidStream
- * CreateAndroidRenderChannel(WebRtc_Word32 streamId,
- WebRtc_Word32 zOrder,
- const float left, const float top,
- const float right, const float bottom,
- VideoRenderAndroid& renderer) = 0;
-
- WebRtc_Word32 _id;
- CriticalSectionWrapper& _critSect;
- VideoRenderType _renderType;
- jobject _ptrWindow;
-
- static JavaVM* g_jvm;
-
-private:
- static bool JavaRenderThreadFun(void* obj);
- bool JavaRenderThreadProcess();
-
- MapWrapper _streamsMap; // Map with streams to render.
- bool _javaShutDownFlag; // True if the _javaRenderThread thread shall be detached from the JVM.
- EventWrapper& _javaShutdownEvent;
- EventWrapper& _javaRenderEvent;
- WebRtc_Word64 _lastJavaRenderEvent;
- JNIEnv* _javaRenderJniEnv; // JNIEnv for the java render thread.
- ThreadWrapper* _javaRenderThread;
-};
-
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_IMPL_H_
diff --git a/trunk/src/modules/video_render/main/source/Android/video_render_android_native_opengl2.cc b/trunk/src/modules/video_render/main/source/Android/video_render_android_native_opengl2.cc
deleted file mode 100644
index a6d7133..0000000
--- a/trunk/src/modules/video_render/main/source/Android/video_render_android_native_opengl2.cc
+++ /dev/null
@@ -1,495 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_render_android_native_opengl2.h"
-#include "critical_section_wrapper.h"
-#include "tick_util.h"
-
-#ifdef ANDROID_LOG
-#include <stdio.h>
-#include <android/log.h>
-
-#undef WEBRTC_TRACE
-#define WEBRTC_TRACE(a,b,c,...) __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTC*", __VA_ARGS__)
-#else
-#include "trace.h"
-#endif
-
-namespace webrtc {
-
-AndroidNativeOpenGl2Renderer::AndroidNativeOpenGl2Renderer(
- const WebRtc_Word32 id,
- const VideoRenderType videoRenderType,
- void* window,
- const bool fullscreen) :
- VideoRenderAndroid(id, videoRenderType, window, fullscreen),
- _javaRenderObj(NULL),
- _javaRenderClass(NULL)
-{
-}
-
-bool AndroidNativeOpenGl2Renderer::UseOpenGL2(void* window)
-{
- if (!g_jvm)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
- "RendererAndroid():UseOpenGL No JVM set.");
- return false;
- }
- bool isAttached = false;
- JNIEnv* env = NULL;
- if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = g_jvm->AttachCurrentThread(&env, NULL);
-
- // Get the JNI env for this thread
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideoRenderer,
- -1,
- "RendererAndroid(): Could not attach thread to JVM (%d, %p)",
- res, env);
- return false;
- }
- isAttached = true;
- }
-
- // get the renderer class
- jclass javaRenderClassLocal =
- env->FindClass("org/webrtc/videoengine/ViEAndroidGLES20");
- if (!javaRenderClassLocal)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
- "%s: could not find ViEAndroidRenderer class",
- __FUNCTION__);
- return false;
- }
-
- // get the method ID for UseOpenGL
- jmethodID cidUseOpenGL = env->GetStaticMethodID(javaRenderClassLocal,
- "UseOpenGL2",
- "(Ljava/lang/Object;)Z");
- if (cidUseOpenGL == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
- "%s: could not get UseOpenGL ID", __FUNCTION__);
- return false;
- }
- jboolean res = env->CallStaticBooleanMethod(javaRenderClassLocal,
- cidUseOpenGL, (jobject) window);
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (g_jvm->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, -1,
- "%s: Could not detach thread from JVM", __FUNCTION__);
- }
- }
- return res;
-}
-
-AndroidNativeOpenGl2Renderer::~AndroidNativeOpenGl2Renderer()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
- "AndroidNativeOpenGl2Renderer dtor");
- if (g_jvm)
- {
- // get the JNI env for this thread
- bool isAttached = false;
- JNIEnv* env = NULL;
- if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = g_jvm->AttachCurrentThread(&env, NULL);
-
- // Get the JNI env for this thread
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: Could not attach thread to JVM (%d, %p)",
- __FUNCTION__, res, env);
- env = NULL;
- }
- else
- {
- isAttached = true;
- }
- }
-
- env->DeleteGlobalRef(_javaRenderObj);
- env->DeleteGlobalRef(_javaRenderClass);
-
- if (isAttached)
- {
- if (g_jvm->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
- "%s: Could not detach thread from JVM",
- __FUNCTION__);
- }
- }
- }
-}
-
-WebRtc_Word32 AndroidNativeOpenGl2Renderer::Init()
-{
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s", __FUNCTION__);
- if (!g_jvm)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "(%s): Not a valid Java VM pointer.", __FUNCTION__);
- return -1;
- }
- if (!_ptrWindow)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
- "(%s): No window have been provided.", __FUNCTION__);
- return -1;
- }
-
- // get the JNI env for this thread
- bool isAttached = false;
- JNIEnv* env = NULL;
- if (g_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = g_jvm->AttachCurrentThread(&env, NULL);
-
- // Get the JNI env for this thread
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: Could not attach thread to JVM (%d, %p)",
- __FUNCTION__, res, env);
- return -1;
- }
- isAttached = true;
- }
-
- // get the ViEAndroidGLES20 class
- jclass javaRenderClassLocal =
- env->FindClass("org/webrtc/videoengine/ViEAndroidGLES20");
- if (!javaRenderClassLocal)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: could not find ViEAndroidGLES20", __FUNCTION__);
- return -1;
- }
-
- // create a global reference to the class (to tell JNI that we are referencing it after this function has returned)
- _javaRenderClass
- = reinterpret_cast<jclass> (env->NewGlobalRef(javaRenderClassLocal));
- if (!_javaRenderClass)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: could not create Java SurfaceHolder class reference",
- __FUNCTION__);
- return -1;
- }
-
- // Delete local class ref, we only use the global ref
- env->DeleteLocalRef(javaRenderClassLocal);
-
- // create a reference to the object (to tell JNI that we are referencing it
- // after this function has returned)
- _javaRenderObj = env->NewGlobalRef(_ptrWindow);
- if (!_javaRenderObj)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideoRenderer,
- _id,
- "%s: could not create Java SurfaceRender object reference",
- __FUNCTION__);
- return -1;
- }
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (g_jvm->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
- "%s: Could not detach thread from JVM", __FUNCTION__);
- }
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s done",
- __FUNCTION__);
- return 0;
-
-}
-AndroidStream*
-AndroidNativeOpenGl2Renderer::CreateAndroidRenderChannel(
- WebRtc_Word32 streamId,
- WebRtc_Word32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom,
- VideoRenderAndroid& renderer)
-{
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s: Id %d",
- __FUNCTION__, streamId);
- AndroidNativeOpenGl2Channel* stream =
- new AndroidNativeOpenGl2Channel(streamId, g_jvm, renderer,
- _javaRenderObj);
- if (stream && stream->Init(zOrder, left, top, right, bottom) == 0)
- return stream;
- else
- {
- delete stream;
- }
- return NULL;
-}
-
-AndroidNativeOpenGl2Channel::AndroidNativeOpenGl2Channel(WebRtc_UWord32 streamId,
- JavaVM* jvm,
- VideoRenderAndroid& renderer,jobject javaRenderObj):
- _id(streamId),
- _renderCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _renderer(renderer), _jvm(jvm), _javaRenderObj(javaRenderObj),
- _registerNativeCID(NULL), _deRegisterNativeCID(NULL),
- _openGLRenderer(streamId)
-{
-
-}
-AndroidNativeOpenGl2Channel::~AndroidNativeOpenGl2Channel()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id,
- "AndroidNativeOpenGl2Channel dtor");
- delete &_renderCritSect;
- if (_jvm)
- {
- // get the JNI env for this thread
- bool isAttached = false;
- JNIEnv* env = NULL;
- if (_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = _jvm->AttachCurrentThread(&env, NULL);
-
- // Get the JNI env for this thread
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: Could not attach thread to JVM (%d, %p)",
- __FUNCTION__, res, env);
- env = NULL;
- }
- else
- {
- isAttached = true;
- }
- }
- if (env && _deRegisterNativeCID)
- {
- env->CallVoidMethod(_javaRenderObj, _deRegisterNativeCID);
- }
-
- if (isAttached)
- {
- if (_jvm->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
- "%s: Could not detach thread from JVM",
- __FUNCTION__);
- }
- }
- }
-}
-
-WebRtc_Word32 AndroidNativeOpenGl2Channel::Init(WebRtc_Word32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
- "%s: AndroidNativeOpenGl2Channel", __FUNCTION__);
- if (!_jvm)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: Not a valid Java VM pointer", __FUNCTION__);
- return -1;
- }
-
- // get the JNI env for this thread
- bool isAttached = false;
- JNIEnv* env = NULL;
- if (_jvm->GetEnv((void**) &env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = _jvm->AttachCurrentThread(&env, NULL);
-
- // Get the JNI env for this thread
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: Could not attach thread to JVM (%d, %p)",
- __FUNCTION__, res, env);
- return -1;
- }
- isAttached = true;
- }
-
- jclass javaRenderClass =
- env->FindClass("org/webrtc/videoengine/ViEAndroidGLES20");
- if (!javaRenderClass)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: could not find ViESurfaceRenderer", __FUNCTION__);
- return -1;
- }
-
- // get the method ID for the ReDraw function
- _redrawCid = env->GetMethodID(javaRenderClass, "ReDraw", "()V");
- if (_redrawCid == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: could not get ReDraw ID", __FUNCTION__);
- return -1;
- }
-
- _registerNativeCID = env->GetMethodID(javaRenderClass,
- "RegisterNativeObject", "(J)V");
- if (_registerNativeCID == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: could not get RegisterNativeObject ID", __FUNCTION__);
- return -1;
- }
-
- _deRegisterNativeCID = env->GetMethodID(javaRenderClass,
- "DeRegisterNativeObject", "()V");
- if (_deRegisterNativeCID == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: could not get DeRegisterNativeObject ID",
- __FUNCTION__);
- return -1;
- }
-
- JNINativeMethod
- nativeFunctions[2] = {
- "DrawNative",
- "(J)V",
- (void*) &AndroidNativeOpenGl2Channel::DrawNativeStatic,
- "CreateOpenGLNative",
- "(JII)I",
- (void*) &AndroidNativeOpenGl2Channel::CreateOpenGLNativeStatic };
- if (env->RegisterNatives(javaRenderClass, nativeFunctions, 2) == 0)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, -1,
- "%s: Registered native functions", __FUNCTION__);
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
- "%s: Failed to register native functions", __FUNCTION__);
- return -1;
- }
-
- env->CallVoidMethod(_javaRenderObj, _registerNativeCID, (jlong) this);
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (_jvm->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id,
- "%s: Could not detach thread from JVM", __FUNCTION__);
- }
- }
-
- if (_openGLRenderer.SetCoordinates(zOrder, left, top, right, bottom) != 0)
- {
- return -1;
- }
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
- "%s: AndroidNativeOpenGl2Channel done", __FUNCTION__);
- return 0;
-}
-
-WebRtc_Word32 AndroidNativeOpenGl2Channel::RenderFrame(const WebRtc_UWord32 /*streamId*/,
- VideoFrame& videoFrame)
-{
- // WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer,_id, "%s:" ,__FUNCTION__);
- _renderCritSect.Enter();
- _bufferToRender.SwapFrame(videoFrame);
- _renderCritSect.Leave();
- _renderer.ReDraw();
- return 0;
-}
-
-/*Implements AndroidStream
- * Calls the Java object and render the buffer in _bufferToRender
- */
-void AndroidNativeOpenGl2Channel::DeliverFrame(JNIEnv* jniEnv)
-{
- //TickTime timeNow=TickTime::Now();
-
- //Draw the Surface
- jniEnv->CallVoidMethod(_javaRenderObj, _redrawCid);
-
- //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer,_id, "%s: time to deliver %lld" ,__FUNCTION__,(TickTime::Now()-timeNow).Milliseconds());
-}
-
-/*
- * JNI callback from Java class. Called when the render want to render a frame. Called from the GLRenderThread
- * Method: DrawNative
- * Signature: (J)V
- */
-void JNICALL AndroidNativeOpenGl2Channel::DrawNativeStatic
-(JNIEnv * env, jobject, jlong context)
-{
- AndroidNativeOpenGl2Channel* renderChannel=reinterpret_cast<AndroidNativeOpenGl2Channel*>(context);
- renderChannel->DrawNative();
-}
-
-void AndroidNativeOpenGl2Channel::DrawNative()
-{
- _openGLRenderer.Render(_bufferToRender);
-}
-/*
- * JNI callback from Java class. Called when the GLSurfaceview have created a surface. Called from the GLRenderThread
- * Method: CreateOpenGLNativeStatic
- * Signature: (JII)I
- */
-jint JNICALL AndroidNativeOpenGl2Channel::CreateOpenGLNativeStatic(JNIEnv * env,
- jobject,
- jlong context,
- jint width,
- jint height)
-{
- AndroidNativeOpenGl2Channel* renderChannel =
- reinterpret_cast<AndroidNativeOpenGl2Channel*> (context);
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, -1, "%s:", __FUNCTION__);
- return renderChannel->CreateOpenGLNative(width, height);
-}
-
-jint AndroidNativeOpenGl2Channel::CreateOpenGLNative(int width, int height)
-{
-
- return _openGLRenderer.Setup(width, height);
-}
-
-} //namespace webrtc
-
diff --git a/trunk/src/modules/video_render/main/source/Android/video_render_android_native_opengl2.h b/trunk/src/modules/video_render/main/source/Android/video_render_android_native_opengl2.h
deleted file mode 100644
index 54532a6..0000000
--- a/trunk/src/modules/video_render/main/source/Android/video_render_android_native_opengl2.h
+++ /dev/null
@@ -1,93 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_NATIVE_OPENGL2_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_NATIVE_OPENGL2_H_
-
-#include <jni.h>
-
-#include "video_render_defines.h"
-
-#include "video_render_android_impl.h"
-#include "video_render_opengles20.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class AndroidNativeOpenGl2Channel: public AndroidStream
-{
-
-public:
- AndroidNativeOpenGl2Channel(WebRtc_UWord32 streamId,JavaVM* jvm,VideoRenderAndroid& renderer,jobject javaRenderObj);
- ~AndroidNativeOpenGl2Channel();
-
- WebRtc_Word32 Init(WebRtc_Word32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- //Implement VideoRenderCallback
- virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId, VideoFrame& videoFrame);
-
- //Implements AndroidStream
- virtual void DeliverFrame(JNIEnv* jniEnv);
-
-private:
- static jint CreateOpenGLNativeStatic(JNIEnv * env,jobject, jlong context, jint width, jint height);
- jint CreateOpenGLNative(int width, int height);
-
- static void DrawNativeStatic(JNIEnv * env,jobject, jlong context);
- void DrawNative();
- WebRtc_UWord32 _id;
- CriticalSectionWrapper& _renderCritSect;
-
- VideoFrame _bufferToRender;
- VideoRenderAndroid& _renderer;
- JavaVM* _jvm;
- jobject _javaRenderObj;
-
- jmethodID _redrawCid;
- jmethodID _registerNativeCID;
- jmethodID _deRegisterNativeCID;
- VideoRenderOpenGles20 _openGLRenderer;
-
-};
-
-
-class AndroidNativeOpenGl2Renderer: private VideoRenderAndroid
-{
-public:
- AndroidNativeOpenGl2Renderer(const WebRtc_Word32 id,
- const VideoRenderType videoRenderType,
- void* window,
- const bool fullscreen);
-
- ~AndroidNativeOpenGl2Renderer();
- static bool UseOpenGL2(void* window);
-
- WebRtc_Word32 Init();
- virtual AndroidStream* CreateAndroidRenderChannel(WebRtc_Word32 streamId,
- WebRtc_Word32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom,
- VideoRenderAndroid& renderer);
-
-private:
- jobject _javaRenderObj;
- jclass _javaRenderClass;
-
-};
-
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_NATIVE_OPENGL2_H_
diff --git a/trunk/src/modules/video_render/main/source/Android/video_render_android_surface_view.cc b/trunk/src/modules/video_render/main/source/Android/video_render_android_surface_view.cc
deleted file mode 100644
index 253d831..0000000
--- a/trunk/src/modules/video_render/main/source/Android/video_render_android_surface_view.cc
+++ /dev/null
@@ -1,470 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_render_android_surface_view.h"
-#include "critical_section_wrapper.h"
-#include "common_video/libyuv/include/libyuv.h"
-#include "tick_util.h"
-#ifdef ANDROID_NDK_8_OR_ABOVE
- #include <android/bitmap.h>
-#endif
-
-
-#ifdef ANDROID_LOG
-#include <stdio.h>
-#include <android/log.h>
-
-#undef WEBRTC_TRACE
-#define WEBRTC_TRACE(a,b,c,...) __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTC*", __VA_ARGS__)
-#else
-#include "trace.h"
-#endif
-
-namespace webrtc {
-
-AndroidSurfaceViewRenderer::AndroidSurfaceViewRenderer(const WebRtc_Word32 id,
- const VideoRenderType videoRenderType,
- void* window,
- const bool fullscreen)
-:
- VideoRenderAndroid(id,videoRenderType,window,fullscreen),
- _javaRenderObj(NULL),
- _javaRenderClass(NULL)
-{
-}
-
-AndroidSurfaceViewRenderer::~AndroidSurfaceViewRenderer()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "AndroidSurfaceViewRenderer dtor");
- if(g_jvm)
- {
- // get the JNI env for this thread
- bool isAttached = false;
- JNIEnv* env = NULL;
- if (g_jvm->GetEnv((void**)&env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = g_jvm->AttachCurrentThread(&env, NULL);
-
- // Get the JNI env for this thread
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: Could not attach thread to JVM (%d, %p)", __FUNCTION__, res, env);
- env=NULL;
- }
- else
- {
- isAttached = true;
- }
- }
- env->DeleteGlobalRef(_javaRenderObj);
- env->DeleteGlobalRef(_javaRenderClass);
-
- if (isAttached)
- {
- if (g_jvm->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "%s: Could not detach thread from JVM", __FUNCTION__);
- }
- }
- }
-}
-
-
-WebRtc_Word32
-AndroidSurfaceViewRenderer::Init()
-{
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s", __FUNCTION__);
- if (!g_jvm)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "(%s): Not a valid Java VM pointer.", __FUNCTION__);
- return -1;
- }
- if(!_ptrWindow)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "(%s): No window have been provided.", __FUNCTION__);
- return -1;
- }
-
- // get the JNI env for this thread
- bool isAttached = false;
- JNIEnv* env = NULL;
- if (g_jvm->GetEnv((void**)&env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = g_jvm->AttachCurrentThread(&env, NULL);
-
- // Get the JNI env for this thread
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: Could not attach thread to JVM (%d, %p)", __FUNCTION__, res, env);
- return -1;
- }
- isAttached = true;
- }
-
- // get the ViESurfaceRender class
- jclass javaRenderClassLocal = env->FindClass("org/webrtc/videoengine/ViESurfaceRenderer");
- if (!javaRenderClassLocal)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not find ViESurfaceRenderer", __FUNCTION__);
- return -1;
- }
-
- // create a global reference to the class (to tell JNI that we are referencing it after this function has returned)
- _javaRenderClass = reinterpret_cast<jclass>(env->NewGlobalRef(javaRenderClassLocal));
- if (!_javaRenderClass)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not create Java ViESurfaceRenderer class reference", __FUNCTION__);
- return -1;
- }
-
- // Delete local class ref, we only use the global ref
- env->DeleteLocalRef(javaRenderClassLocal);
-
- // get the method ID for the constructor
- jmethodID cid = env->GetMethodID(_javaRenderClass, "<init>", "(Landroid/view/SurfaceView;)V");
- if (cid == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not get constructor ID", __FUNCTION__);
- return -1; /* exception thrown */
- }
-
- // construct the object
- jobject javaRenderObjLocal = env->NewObject(_javaRenderClass, cid, _ptrWindow);
- if (!javaRenderObjLocal)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not create Java Render", __FUNCTION__);
- return -1;
- }
-
- // create a reference to the object (to tell JNI that we are referencing it
- // after this function has returned)
- _javaRenderObj = env->NewGlobalRef(javaRenderObjLocal);
- if (!_javaRenderObj)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not create Java SurfaceRender object reference", __FUNCTION__);
- return -1;
- }
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (g_jvm->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "%s: Could not detach thread from JVM", __FUNCTION__);
- }
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s done", __FUNCTION__);
- return 0;
-
-}
-AndroidStream*
-AndroidSurfaceViewRenderer::CreateAndroidRenderChannel(WebRtc_Word32 streamId,
- WebRtc_Word32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom,
- VideoRenderAndroid& renderer)
-{
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s: Id %d", __FUNCTION__,streamId);
- AndroidSurfaceViewChannel* stream=new AndroidSurfaceViewChannel(streamId,g_jvm,renderer,_javaRenderObj);
- if(stream && stream->Init(zOrder,left,top,right,bottom)==0)
- return stream;
- else
- delete stream;
- return NULL;
-}
-
-
-
-
-
-
-AndroidSurfaceViewChannel::AndroidSurfaceViewChannel(WebRtc_UWord32 streamId,JavaVM* jvm,VideoRenderAndroid& renderer,jobject javaRenderObj)
-:
-_id(streamId),
-_renderCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
-_renderer(renderer),
-_jvm(jvm),
-_javaRenderObj(javaRenderObj),
-_bitmapWidth(0),
-_bitmapHeight(0)
-{
-
-}
-AndroidSurfaceViewChannel::~AndroidSurfaceViewChannel()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "AndroidSurfaceViewChannel dtor");
- delete &_renderCritSect;
- if(_jvm)
- {
- // get the JNI env for this thread
- bool isAttached = false;
- JNIEnv* env = NULL;
- if (_jvm->GetEnv((void**)&env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = _jvm->AttachCurrentThread(&env, NULL);
-
- // Get the JNI env for this thread
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: Could not attach thread to JVM (%d, %p)", __FUNCTION__, res, env);
- env=NULL;
- }
- else
- {
- isAttached = true;
- }
- }
-
-#ifdef ANDROID_NDK_8_OR_ABOVE
- env->DeleteGlobalRef(_javaBitmapObj);
-#else
- env->DeleteGlobalRef(_javaByteBufferObj);
-#endif
- if (isAttached)
- {
- if (_jvm->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "%s: Could not detach thread from JVM", __FUNCTION__);
- }
- }
- }
-}
-
-WebRtc_Word32
-AndroidSurfaceViewChannel::Init(WebRtc_Word32 /*zOrder*/,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
-
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s: AndroidSurfaceViewChannel", __FUNCTION__);
- if (!_jvm)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer,_id, "%s: Not a valid Java VM pointer", __FUNCTION__);
- return -1;
- }
-
- if((top>1 || top<0) || (right>1 || right<0) || (bottom>1 || bottom<0) || (left>1 || left<0))
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: Wrong coordinates",
- __FUNCTION__);
- return -1;
- }
-
-
- // get the JNI env for this thread
- bool isAttached = false;
- JNIEnv* env = NULL;
- if (_jvm->GetEnv((void**)&env, JNI_VERSION_1_4) != JNI_OK)
- {
- // try to attach the thread and get the env
- // Attach this thread to JVM
- jint res = _jvm->AttachCurrentThread(&env, NULL);
-
- // Get the JNI env for this thread
- if ((res < 0) || !env)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: Could not attach thread to JVM (%d, %p)", __FUNCTION__, res, env);
- return -1;
- }
- isAttached = true;
- }
-
- jclass javaRenderClass = env->FindClass("org/webrtc/videoengine/ViESurfaceRenderer");
- if (!javaRenderClass)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not find ViESurfaceRenderer", __FUNCTION__);
- return -1;
- }
-#ifdef ANDROID_NDK_8_OR_ABOVE
- // get the method ID for the CreateBitmap
- _createBitmapCid = env->GetMethodID(_javaRenderClass, "CreateBitmap", "(II)Landroid/graphics/Bitmap;");
- if (_createBitmapCid == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not get CreateBitmap ID", __FUNCTION__);
- return -1; /* exception thrown */
- }
- // get the method ID for the DrawBitmap function
- _drawBitmapCid = env->GetMethodID(_javaRenderClass, "DrawBitmap", "()V");
- if (_drawBitmapCid == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not get DrawBitmap ID", __FUNCTION__);
- return -1; /* exception thrown */
- }
-#else
- // get the method ID for the CreateIntArray
- _createByteBufferCid = env->GetMethodID(javaRenderClass, "CreateByteBuffer", "(II)Ljava/nio/ByteBuffer;");
- if (_createByteBufferCid == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not get CreateByteBuffer ID", __FUNCTION__);
- return -1; /* exception thrown */
- }
-
- // get the method ID for the DrawByteBuffer function
- _drawByteBufferCid = env->GetMethodID(javaRenderClass, "DrawByteBuffer", "()V");
- if (_drawByteBufferCid == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not get DrawByteBuffer ID", __FUNCTION__);
- return -1; /* exception thrown */
- }
-#endif
-
- // get the method ID for the SetCoordinates function
- _setCoordinatesCid = env->GetMethodID(javaRenderClass, "SetCoordinates", "(FFFF)V");
- if (_setCoordinatesCid == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not get SetCoordinates ID", __FUNCTION__);
- return -1; /* exception thrown */
- }
-
- env->CallVoidMethod(_javaRenderObj,_setCoordinatesCid,left,top,right,bottom);
-
- // Detach this thread if it was attached
- if (isAttached)
- {
- if (_jvm->DetachCurrentThread() < 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "%s: Could not detach thread from JVM", __FUNCTION__);
- }
- }
-
-
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s: AndroidSurfaceViewChannel done", __FUNCTION__);
- return 0;
-}
-
-
-WebRtc_Word32 AndroidSurfaceViewChannel::RenderFrame(const WebRtc_UWord32 /*streamId*/, VideoFrame& videoFrame)
-{
- // WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer,_id, "%s:" ,__FUNCTION__);
- _renderCritSect.Enter();
- _bufferToRender.SwapFrame(videoFrame);
- _renderCritSect.Leave();
- _renderer.ReDraw();
- return 0;
-}
-
-
-/*Implements AndroidStream
- * Calls the Java object and render the buffer in _bufferToRender
- */
-void AndroidSurfaceViewChannel::DeliverFrame(JNIEnv* jniEnv) {
- _renderCritSect.Enter();
-
-#ifdef ANDROID_NDK_8_OR_ABOVE
- if (_bitmapWidth != _bufferToRender.Width() ||
- _bitmapHeight != _bufferToRender.Height()) {
- // Create the bitmap to write to
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s: Creating bitmap %u "
- "%u", __FUNCTION__, _bufferToRender.Width(),
- _bufferToRender.Height());
- if (_javaBitmapObj) {
- jniEnv->DeleteGlobalRef(_javaBitmapObj);
- _javaBitmapObj = NULL;
- }
- jobject javaBitmap = jniEnv->CallObjectMethod(_javaRenderObj,
- _createBitmapCid,
- videoFrame.Width(),
- videoFrame.Height());
- _javaBitmapObj = jniEnv->NewGlobalRef(javaBitmap);
- if (!_javaBitmapObj) {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not "
- "create Java Bitmap object reference", __FUNCTION__);
- _renderCritSect.Leave();
- return;
- } else {
- _bitmapWidth=_bufferToRender.Width();
- _bitmapHeight=_bufferToRender.Height();
- }
- }
- void* pixels;
- if (_javaBitmapObj &&
- AndroidBitmap_lockPixels(jniEnv, _javaBitmapObj, &pixels) >= 0) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s: Locked bitmap",
- __FUNCTION__);
- // Convert I420 straight into the Java bitmap.
- int ret = ConvertI420ToRGB565((unsigned char* )_bufferToRender.Buffer(),
- (unsigned char* ) pixels,
- _bitmapWidth, _bitmapHeight);
- if (ret < 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: Color conversion "
- "failed.", __FUNCTION__);
- }
-
- AndroidBitmap_unlockPixels(jniEnv, _javaBitmapObj);
- // Draw the Surface.
- jniEnv->CallVoidMethod(_javaRenderObj,_drawCid);
-
- } else {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: Could not lock "
- "bitmap", __FUNCTION__);
- }
- _renderCritSect.Leave();
-
-#else
- if (_bitmapWidth != _bufferToRender.Width() ||
- _bitmapHeight != _bufferToRender.Height()) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s: New render size %d "
- "%d",__FUNCTION__,
- _bufferToRender.Width(), _bufferToRender.Height());
- if (_javaByteBufferObj) {
- jniEnv->DeleteGlobalRef(_javaByteBufferObj);
- _javaByteBufferObj = NULL;
- _directBuffer = NULL;
- }
- jobject javaByteBufferObj =
- jniEnv->CallObjectMethod(_javaRenderObj, _createByteBufferCid,
- _bufferToRender.Width(),
- _bufferToRender.Height());
- _javaByteBufferObj = jniEnv->NewGlobalRef(javaByteBufferObj);
- if (!_javaByteBufferObj) {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: could not "
- "create Java ByteBuffer object reference", __FUNCTION__);
- _renderCritSect.Leave();
- return;
- } else {
- _directBuffer = static_cast<unsigned char*>
- (jniEnv->GetDirectBufferAddress(_javaByteBufferObj));
- _bitmapWidth = _bufferToRender.Width();
- _bitmapHeight = _bufferToRender.Height();
- }
- }
-
- if(_javaByteBufferObj && _bitmapWidth && _bitmapHeight) {
- // Android requires a vertically flipped image compared to std convert.
- // This is done by giving a negative height input.
- const int conversionResult =
- ConvertI420ToRGB565((unsigned char* )_bufferToRender.Buffer(),
- _directBuffer, _bitmapWidth, -_bitmapHeight);
- if (conversionResult < 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s: Color conversion"
- " failed.", __FUNCTION__);
- _renderCritSect.Leave();
- return;
- }
- }
- _renderCritSect.Leave();
- // Draw the Surface
- jniEnv->CallVoidMethod(_javaRenderObj, _drawByteBufferCid);
-#endif
-}
-
-} // namespace webrtc
-
diff --git a/trunk/src/modules/video_render/main/source/Android/video_render_android_surface_view.h b/trunk/src/modules/video_render/main/source/Android/video_render_android_surface_view.h
deleted file mode 100644
index f55e60b..0000000
--- a/trunk/src/modules/video_render/main/source/Android/video_render_android_surface_view.h
+++ /dev/null
@@ -1,96 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_SURFACE_VIEW_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_SURFACE_VIEW_H_
-
-#include <jni.h>
-
-#include "video_render_defines.h"
-
-#include "video_render_android_impl.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-
-class AndroidSurfaceViewChannel: public AndroidStream
-{
-
-public:
- AndroidSurfaceViewChannel(WebRtc_UWord32 streamId,
- JavaVM* jvm,
- VideoRenderAndroid& renderer,
- jobject javaRenderObj);
- ~AndroidSurfaceViewChannel();
-
- WebRtc_Word32 Init(WebRtc_Word32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- //Implement VideoRenderCallback
- virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId,
- VideoFrame& videoFrame);
-
- //Implements AndroidStream
- virtual void DeliverFrame(JNIEnv* jniEnv);
-
-private:
- WebRtc_UWord32 _id;
- CriticalSectionWrapper& _renderCritSect;
-
- VideoFrame _bufferToRender;
- VideoRenderAndroid& _renderer;
- JavaVM* _jvm;
- jobject _javaRenderObj;
-
-#ifdef ANDROID_NDK_8_OR_ABOVE
- jclass _javaBitmapClass;
- jmethodID _createBitmapCid;
- jobject _javaBitmapObj;
- jmethodID _drawBitmapCid;
-#else
- jobject _javaByteBufferObj;
- unsigned char* _directBuffer;
- jmethodID _createByteBufferCid;
- jmethodID _drawByteBufferCid;
-#endif
- jmethodID _setCoordinatesCid;
- unsigned int _bitmapWidth;
- unsigned int _bitmapHeight;
-};
-
-class AndroidSurfaceViewRenderer: private VideoRenderAndroid
-{
-public:
- AndroidSurfaceViewRenderer(const WebRtc_Word32 id,
- const VideoRenderType videoRenderType,
- void* window,
- const bool fullscreen);
- ~AndroidSurfaceViewRenderer();
- WebRtc_Word32 Init();
- virtual AndroidStream* CreateAndroidRenderChannel(WebRtc_Word32 streamId,
- WebRtc_Word32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom,
- VideoRenderAndroid& renderer);
-private:
- jobject _javaRenderObj;
- jclass _javaRenderClass;
-
-};
-
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_ANDROID_SURFACE_VIEW_H_
diff --git a/trunk/src/modules/video_render/main/source/Android/video_render_opengles20.cc b/trunk/src/modules/video_render/main/source/Android/video_render_opengles20.cc
deleted file mode 100644
index 8f4e5c5..0000000
--- a/trunk/src/modules/video_render/main/source/Android/video_render_opengles20.cc
+++ /dev/null
@@ -1,446 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <GLES2/gl2.h>
-#include <GLES2/gl2ext.h>
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "video_render_opengles20.h"
-
-//#define ANDROID_LOG
-
-#ifdef ANDROID_LOG
-#include <stdio.h>
-#include <android/log.h>
-
-#undef WEBRTC_TRACE
-#define WEBRTC_TRACE(a,b,c,...) __android_log_print(ANDROID_LOG_DEBUG, "*WEBRTCN*", __VA_ARGS__)
-#else
-#include "trace.h"
-#endif
-
-namespace webrtc {
-
-const char VideoRenderOpenGles20::g_indices[] = { 0, 3, 2, 0, 2, 1 };
-
-const char VideoRenderOpenGles20::g_vertextShader[] = {
- "attribute vec4 aPosition;\n"
- "attribute vec2 aTextureCoord;\n"
- "varying vec2 vTextureCoord;\n"
- "void main() {\n"
- " gl_Position = aPosition;\n"
- " vTextureCoord = aTextureCoord;\n"
- "}\n" };
-
-// The fragment shader.
-// Do YUV to RGB565 conversion.
-const char VideoRenderOpenGles20::g_fragmentShader[] = {
- "precision mediump float;\n"
- "uniform sampler2D Ytex;\n"
- "uniform sampler2D Utex,Vtex;\n"
- "varying vec2 vTextureCoord;\n"
- "void main(void) {\n"
- " float nx,ny,r,g,b,y,u,v;\n"
- " mediump vec4 txl,ux,vx;"
- " nx=vTextureCoord[0];\n"
- " ny=vTextureCoord[1];\n"
- " y=texture2D(Ytex,vec2(nx,ny)).r;\n"
- " u=texture2D(Utex,vec2(nx,ny)).r;\n"
- " v=texture2D(Vtex,vec2(nx,ny)).r;\n"
-
- //" y = v;\n"+
- " y=1.1643*(y-0.0625);\n"
- " u=u-0.5;\n"
- " v=v-0.5;\n"
-
- " r=y+1.5958*v;\n"
- " g=y-0.39173*u-0.81290*v;\n"
- " b=y+2.017*u;\n"
- " gl_FragColor=vec4(r,g,b,1.0);\n"
- "}\n" };
-
-VideoRenderOpenGles20::VideoRenderOpenGles20(WebRtc_Word32 id) :
- _id(id),
- _textureWidth(-1),
- _textureHeight(-1)
-
-{
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s: id %d",
- __FUNCTION__, (int) _id);
-
- const GLfloat vertices[20] = {
- // X, Y, Z, U, V
- -1, -1, 0, 0, 1, // Bottom Left
- 1, -1, 0, 1, 1, //Bottom Right
- 1, 1, 0, 1, 0, //Top Right
- -1, 1, 0, 0, 0 }; //Top Left
-
- memcpy(_vertices, vertices, sizeof(_vertices));
-}
-
-VideoRenderOpenGles20::~VideoRenderOpenGles20()
-{
-
-}
-
-WebRtc_Word32 VideoRenderOpenGles20::Setup(WebRtc_Word32 width,
- WebRtc_Word32 height)
-{
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
- "%s: width %d, height %d", __FUNCTION__, (int) width,
- (int) height);
-
- printGLString("Version", GL_VERSION);
- printGLString("Vendor", GL_VENDOR);
- printGLString("Renderer", GL_RENDERER);
- printGLString("Extensions", GL_EXTENSIONS);
-
- int maxTextureImageUnits[2];
- int maxTextureSize[2];
- glGetIntegerv(GL_MAX_TEXTURE_IMAGE_UNITS, maxTextureImageUnits);
- glGetIntegerv(GL_MAX_TEXTURE_SIZE, maxTextureSize);
-
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
- "%s: number of textures %d, size %d", __FUNCTION__,
- (int) maxTextureImageUnits[0], (int) maxTextureSize[0]);
-
- _program = createProgram(g_vertextShader, g_fragmentShader);
- if (!_program)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: Could not create program", __FUNCTION__);
- return -1;
- }
-
- int positionHandle = glGetAttribLocation(_program, "aPosition");
- checkGlError("glGetAttribLocation aPosition");
- if (positionHandle == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: Could not get aPosition handle", __FUNCTION__);
- return -1;
- }
- int textureHandle = glGetAttribLocation(_program, "aTextureCoord");
- checkGlError("glGetAttribLocation aTextureCoord");
- if (textureHandle == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: Could not get aTextureCoord handle", __FUNCTION__);
- return -1;
- }
-
- // set the vertices array in the shader
- // _vertices contains 4 vertices with 5 coordinates. 3 for (xyz) for the vertices and 2 for the texture
- glVertexAttribPointer(positionHandle, 3, GL_FLOAT, false, 5
- * sizeof(GLfloat), _vertices);
- checkGlError("glVertexAttribPointer aPosition");
-
- glEnableVertexAttribArray(positionHandle);
- checkGlError("glEnableVertexAttribArray positionHandle");
-
- // set the texture coordinate array in the shader
- // _vertices contains 4 vertices with 5 coordinates. 3 for (xyz) for the vertices and 2 for the texture
- glVertexAttribPointer(textureHandle, 2, GL_FLOAT, false, 5
- * sizeof(GLfloat), &_vertices[3]);
- checkGlError("glVertexAttribPointer maTextureHandle");
- glEnableVertexAttribArray(textureHandle);
- checkGlError("glEnableVertexAttribArray textureHandle");
-
- glUseProgram(_program);
- int i = glGetUniformLocation(_program, "Ytex");
- checkGlError("glGetUniformLocation");
- glUniform1i(i, 0); /* Bind Ytex to texture unit 0 */
- checkGlError("glUniform1i Ytex");
-
- i = glGetUniformLocation(_program, "Utex");
- checkGlError("glGetUniformLocation Utex");
- glUniform1i(i, 1); /* Bind Utex to texture unit 1 */
- checkGlError("glUniform1i Utex");
-
- i = glGetUniformLocation(_program, "Vtex");
- checkGlError("glGetUniformLocation");
- glUniform1i(i, 2); /* Bind Vtex to texture unit 2 */
- checkGlError("glUniform1i");
-
- glViewport(0, 0, width, height);
- checkGlError("glViewport");
- return 0;
-
-}
-/*
- * SetCoordinates
- * Sets the coordinates where the stream shall be rendered. Values must be between 0 and 1.
- */
-WebRtc_Word32 VideoRenderOpenGles20::SetCoordinates(WebRtc_Word32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- if ((top > 1 || top < 0) || (right > 1 || right < 0) || (bottom > 1
- || bottom < 0) || (left > 1 || left < 0))
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: Wrong coordinates", __FUNCTION__);
- return -1;
- }
- /*
- // X, Y, Z, U, V
- -1, -1, 0, 0, 1, // Bottom Left
- 1, -1, 0, 1, 1, //Bottom Right
- 1, 1, 0, 1, 0, //Top Right
- -1, 1, 0, 0, 0 }; //Top Left
- */
- // Bottom Left
- _vertices[0] = (left * 2) - 1;
- _vertices[1] = -1 * (2 * bottom) + 1;
- _vertices[2] = zOrder;
-
- //Bottom Right
- _vertices[5] = (right * 2) - 1;
- _vertices[6] = -1 * (2 * bottom) + 1;
- _vertices[7] = zOrder;
-
- //Top Right
- _vertices[10] = (right * 2) - 1;
- _vertices[11] = -1 * (2 * top) + 1;
- _vertices[12] = zOrder;
-
- //Top Left
- _vertices[15] = (left * 2) - 1;
- _vertices[16] = -1 * (2 * top) + 1;
- _vertices[17] = zOrder;
-
- return 0;
-
-}
-WebRtc_Word32 VideoRenderOpenGles20::Render(const VideoFrame& frameToRender)
-{
-
- if (frameToRender.Length() == 0)
- {
- return -1;
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s: id %d",
- __FUNCTION__, (int) _id);
-
- //glClearColor(0.0f, 0.0f, 1.0f, 1.0f);
- //glClear(GL_DEPTH_BUFFER_BIT | GL_COLOR_BUFFER_BIT);
-
- glUseProgram(_program);
- checkGlError("glUseProgram");
-
- if (_textureWidth != (GLsizei) frameToRender.Width() || _textureHeight
- != (GLsizei) frameToRender.Height())
- {
- SetupTextures(frameToRender);
- }
- else
- {
- UpdateTextures(frameToRender);
- }
-
- glDrawElements(GL_TRIANGLES, 6, GL_UNSIGNED_BYTE, g_indices);
- checkGlError("glDrawArrays");
-
- return 0;
-}
-
-GLuint VideoRenderOpenGles20::loadShader(GLenum shaderType,
- const char* pSource)
-{
- GLuint shader = glCreateShader(shaderType);
- if (shader)
- {
- glShaderSource(shader, 1, &pSource, NULL);
- glCompileShader(shader);
- GLint compiled = 0;
- glGetShaderiv(shader, GL_COMPILE_STATUS, &compiled);
- if (!compiled)
- {
- GLint infoLen = 0;
- glGetShaderiv(shader, GL_INFO_LOG_LENGTH, &infoLen);
- if (infoLen)
- {
- char* buf = (char*) malloc(infoLen);
- if (buf)
- {
- glGetShaderInfoLog(shader, infoLen, NULL, buf);
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: Could not compile shader %d: %s",
- __FUNCTION__, shaderType, buf);
- free(buf);
- }
- glDeleteShader(shader);
- shader = 0;
- }
- }
- }
- return shader;
-}
-
-GLuint VideoRenderOpenGles20::createProgram(const char* pVertexSource,
- const char* pFragmentSource)
-{
- GLuint vertexShader = loadShader(GL_VERTEX_SHADER, pVertexSource);
- if (!vertexShader)
- {
- return 0;
- }
-
- GLuint pixelShader = loadShader(GL_FRAGMENT_SHADER, pFragmentSource);
- if (!pixelShader)
- {
- return 0;
- }
-
- GLuint program = glCreateProgram();
- if (program)
- {
- glAttachShader(program, vertexShader);
- checkGlError("glAttachShader");
- glAttachShader(program, pixelShader);
- checkGlError("glAttachShader");
- glLinkProgram(program);
- GLint linkStatus = GL_FALSE;
- glGetProgramiv(program, GL_LINK_STATUS, &linkStatus);
- if (linkStatus != GL_TRUE)
- {
- GLint bufLength = 0;
- glGetProgramiv(program, GL_INFO_LOG_LENGTH, &bufLength);
- if (bufLength)
- {
- char* buf = (char*) malloc(bufLength);
- if (buf)
- {
- glGetProgramInfoLog(program, bufLength, NULL, buf);
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: Could not link program: %s",
- __FUNCTION__, buf);
- free(buf);
- }
- }
- glDeleteProgram(program);
- program = 0;
- }
- }
- return program;
-}
-
-void VideoRenderOpenGles20::printGLString(const char *name, GLenum s)
-{
- const char *v = (const char *) glGetString(s);
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "GL %s = %s\n",
- name, v);
-}
-
-void VideoRenderOpenGles20::checkGlError(const char* op)
-{
-#ifdef ANDROID_LOG
- for (GLint error = glGetError(); error; error
- = glGetError())
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "after %s() glError (0x%x)\n", op, error);
- }
-#else
- return;
-#endif
-}
-
-void VideoRenderOpenGles20::SetupTextures(const VideoFrame& frameToRender)
-{
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id,
- "%s: width %d, height %d length %u", __FUNCTION__,
- frameToRender.Width(), frameToRender.Height(),
- frameToRender.Length());
-
- const GLsizei width = frameToRender.Width();
- const GLsizei height = frameToRender.Height();
-
- glGenTextures(3, _textureIds); //Generate the Y, U and V texture
- GLuint currentTextureId = _textureIds[0]; // Y
- glActiveTexture( GL_TEXTURE0);
- glBindTexture(GL_TEXTURE_2D, currentTextureId);
-
- glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_NEAREST);
- glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
-
- glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
- glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
-
- glTexImage2D(GL_TEXTURE_2D, 0, GL_LUMINANCE, width, height, 0,
- GL_LUMINANCE, GL_UNSIGNED_BYTE,
- (const GLvoid*) frameToRender.Buffer());
-
- currentTextureId = _textureIds[1]; // U
- glActiveTexture( GL_TEXTURE1);
- glBindTexture(GL_TEXTURE_2D, currentTextureId);
-
- glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_NEAREST);
- glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
-
- glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
- glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
-
- const WebRtc_UWord8* uComponent = frameToRender.Buffer() + width * height;
- glTexImage2D(GL_TEXTURE_2D, 0, GL_LUMINANCE, width / 2, height / 2, 0,
- GL_LUMINANCE, GL_UNSIGNED_BYTE, (const GLvoid*) uComponent);
-
- currentTextureId = _textureIds[2]; // V
- glActiveTexture( GL_TEXTURE2);
- glBindTexture(GL_TEXTURE_2D, currentTextureId);
-
- glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_MIN_FILTER, GL_NEAREST);
- glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
- glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
- glTexParameterf(GL_TEXTURE_2D, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
-
- const WebRtc_UWord8* vComponent = uComponent + (width * height) / 4;
- glTexImage2D(GL_TEXTURE_2D, 0, GL_LUMINANCE, width / 2, height / 2, 0,
- GL_LUMINANCE, GL_UNSIGNED_BYTE, (const GLvoid*) vComponent);
- checkGlError("SetupTextures");
-
- _textureWidth = width;
- _textureHeight = height;
-}
-
-void VideoRenderOpenGles20::UpdateTextures(const VideoFrame& frameToRender)
-{
- const GLsizei width = frameToRender.Width();
- const GLsizei height = frameToRender.Height();
-
- GLuint currentTextureId = _textureIds[0]; // Y
- glActiveTexture( GL_TEXTURE0);
- glBindTexture(GL_TEXTURE_2D, currentTextureId);
- glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, width, height, GL_LUMINANCE,
- GL_UNSIGNED_BYTE, (const GLvoid*) frameToRender.Buffer());
-
- currentTextureId = _textureIds[1]; // U
- glActiveTexture( GL_TEXTURE1);
- glBindTexture(GL_TEXTURE_2D, currentTextureId);
- const WebRtc_UWord8* uComponent = frameToRender.Buffer() + width * height;
- glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, width / 2, height / 2,
- GL_LUMINANCE, GL_UNSIGNED_BYTE, (const GLvoid*) uComponent);
-
- currentTextureId = _textureIds[2]; // V
- glActiveTexture( GL_TEXTURE2);
- glBindTexture(GL_TEXTURE_2D, currentTextureId);
- const WebRtc_UWord8* vComponent = uComponent + (width * height) / 4;
- glTexSubImage2D(GL_TEXTURE_2D, 0, 0, 0, width / 2, height / 2,
- GL_LUMINANCE, GL_UNSIGNED_BYTE, (const GLvoid*) vComponent);
- checkGlError("UpdateTextures");
-
-}
-
-} //namespace webrtc
-
diff --git a/trunk/src/modules/video_render/main/source/Android/video_render_opengles20.h b/trunk/src/modules/video_render/main/source/Android/video_render_opengles20.h
deleted file mode 100644
index 379b1e7..0000000
--- a/trunk/src/modules/video_render/main/source/Android/video_render_opengles20.h
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_OPENGLES20_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_OPENGLES20_H_
-
-#include "video_render_defines.h"
-
-#include <GLES2/gl2.h>
-#include <GLES2/gl2ext.h>
-
-namespace webrtc
-{
-
-class VideoRenderOpenGles20
-{
-public:
- VideoRenderOpenGles20(WebRtc_Word32 id);
- ~VideoRenderOpenGles20();
-
- WebRtc_Word32 Setup(WebRtc_Word32 widht, WebRtc_Word32 height);
- WebRtc_Word32 Render(const VideoFrame& frameToRender);
- WebRtc_Word32 SetCoordinates(WebRtc_Word32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
-private:
- void printGLString(const char *name, GLenum s);
- void checkGlError(const char* op);
- GLuint loadShader(GLenum shaderType, const char* pSource);
- GLuint createProgram(const char* pVertexSource, const char* pFragmentSource);
- void SetupTextures(const VideoFrame& frameToRender);
- void UpdateTextures(const VideoFrame& frameToRender);
-
- WebRtc_Word32 _id;
- GLuint _textureIds[3]; // Texture id of Y,U and V texture.
- GLuint _program;
- GLuint _vPositionHandle;
- GLsizei _textureWidth;
- GLsizei _textureHeight;
-
- GLfloat _vertices[20];
- static const char g_indices[];
-
- static const char g_vertextShader[];
- static const char g_fragmentShader[];
-
-};
-
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_ANDROID_VIDEO_RENDER_OPENGLES20_H_
diff --git a/trunk/src/modules/video_render/main/source/external/video_render_external_impl.cc b/trunk/src/modules/video_render/main/source/external/video_render_external_impl.cc
deleted file mode 100644
index 7abb09a..0000000
--- a/trunk/src/modules/video_render/main/source/external/video_render_external_impl.cc
+++ /dev/null
@@ -1,205 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_render_external_impl.h"
-
-namespace webrtc {
-
-VideoRenderExternalImpl::VideoRenderExternalImpl(
- const WebRtc_Word32 id,
- const VideoRenderType videoRenderType,
- void* window,
- const bool fullscreen) :
- _id(id), _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _window(window), _fullscreen(fullscreen)
-{
-}
-
-VideoRenderExternalImpl::~VideoRenderExternalImpl()
-{
- delete &_critSect;
-}
-
-WebRtc_Word32 VideoRenderExternalImpl::Init()
-{
- return 0;
-}
-
-WebRtc_Word32 VideoRenderExternalImpl::ChangeUniqueId(const WebRtc_Word32 id)
-{
- CriticalSectionScoped cs(_critSect);
- _id = id;
- return 0;
-}
-
-WebRtc_Word32 VideoRenderExternalImpl::ChangeWindow(void* window)
-{
- CriticalSectionScoped cs(_critSect);
- return 0;
-}
-
-VideoRenderCallback*
-VideoRenderExternalImpl::AddIncomingRenderStream(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_critSect);
- return this;
-}
-
-WebRtc_Word32 VideoRenderExternalImpl::DeleteIncomingRenderStream(
- const WebRtc_UWord32 streamId)
-{
- CriticalSectionScoped cs(_critSect);
- return 0;
-}
-
-WebRtc_Word32 VideoRenderExternalImpl::GetIncomingRenderStreamProperties(
- const WebRtc_UWord32 streamId,
- WebRtc_UWord32& zOrder,
- float& left,
- float& top,
- float& right,
- float& bottom) const
-{
- CriticalSectionScoped cs(_critSect);
-
- zOrder = 0;
- left = 0;
- top = 0;
- right = 0;
- bottom = 0;
-
- return 0;
-}
-
-WebRtc_Word32 VideoRenderExternalImpl::StartRender()
-{
- CriticalSectionScoped cs(_critSect);
- return 0;
-}
-
-WebRtc_Word32 VideoRenderExternalImpl::StopRender()
-{
- CriticalSectionScoped cs(_critSect);
- return 0;
-}
-
-VideoRenderType VideoRenderExternalImpl::RenderType()
-{
- return kRenderExternal;
-}
-
-RawVideoType VideoRenderExternalImpl::PerferedVideoType()
-{
- return kVideoI420;
-}
-
-bool VideoRenderExternalImpl::FullScreen()
-{
- CriticalSectionScoped cs(_critSect);
- return _fullscreen;
-}
-
-WebRtc_Word32 VideoRenderExternalImpl::GetGraphicsMemory(
- WebRtc_UWord64& totalGraphicsMemory,
- WebRtc_UWord64& availableGraphicsMemory) const
-{
- totalGraphicsMemory = 0;
- availableGraphicsMemory = 0;
- return -1;
-}
-
-WebRtc_Word32 VideoRenderExternalImpl::GetScreenResolution(
- WebRtc_UWord32& screenWidth,
- WebRtc_UWord32& screenHeight) const
-{
- CriticalSectionScoped cs(_critSect);
- screenWidth = 0;
- screenHeight = 0;
- return 0;
-}
-
-WebRtc_UWord32 VideoRenderExternalImpl::RenderFrameRate(
- const WebRtc_UWord32 streamId)
-{
- CriticalSectionScoped cs(_critSect);
- return 0;
-}
-
-WebRtc_Word32 VideoRenderExternalImpl::SetStreamCropping(
- const WebRtc_UWord32 streamId,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_critSect);
- return 0;
-}
-
-WebRtc_Word32 VideoRenderExternalImpl::ConfigureRenderer(
- const WebRtc_UWord32 streamId,
- const unsigned int zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_critSect);
- return 0;
-}
-
-WebRtc_Word32 VideoRenderExternalImpl::SetTransparentBackground(
- const bool enable)
-{
- CriticalSectionScoped cs(_critSect);
- return 0;
-}
-
-WebRtc_Word32 VideoRenderExternalImpl::SetText(
- const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 textColorRef,
- const WebRtc_UWord32 backgroundColorRef,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_critSect);
- return 0;
-}
-
-WebRtc_Word32 VideoRenderExternalImpl::SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_critSect);
- return 0;
-}
-
-// VideoRenderCallback
-WebRtc_Word32 VideoRenderExternalImpl::RenderFrame(
- const WebRtc_UWord32 streamId,
- VideoFrame& videoFrame)
-{
- return 0;
-}
-} //namespace webrtc
-
diff --git a/trunk/src/modules/video_render/main/source/external/video_render_external_impl.h b/trunk/src/modules/video_render/main/source/external/video_render_external_impl.h
deleted file mode 100644
index e1374f4..0000000
--- a/trunk/src/modules/video_render/main/source/external/video_render_external_impl.h
+++ /dev/null
@@ -1,134 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_
-
-#include "i_video_render.h"
-#include "critical_section_wrapper.h"
-#include "module_common_types.h"
-
-namespace webrtc {
-
-// Class definitions
-class VideoRenderExternalImpl: IVideoRender, public VideoRenderCallback
-{
-public:
- /*
- * Constructor/destructor
- */
-
- VideoRenderExternalImpl(const WebRtc_Word32 id,
- const VideoRenderType videoRenderType,
- void* window, const bool fullscreen);
-
- virtual ~VideoRenderExternalImpl();
-
- virtual WebRtc_Word32 Init();
-
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- virtual WebRtc_Word32 ChangeWindow(void* window);
-
- /**************************************************************************
- *
- * Incoming Streams
- *
- ***************************************************************************/
-
- virtual VideoRenderCallback
- * AddIncomingRenderStream(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left, const float top,
- const float right, const float bottom);
-
- virtual WebRtc_Word32
- DeleteIncomingRenderStream(const WebRtc_UWord32 streamId);
-
- virtual WebRtc_Word32
- GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
- WebRtc_UWord32& zOrder,
- float& left, float& top,
- float& right, float& bottom) const;
-
- /**************************************************************************
- *
- * Start/Stop
- *
- ***************************************************************************/
-
- virtual WebRtc_Word32 StartRender();
-
- virtual WebRtc_Word32 StopRender();
-
- /**************************************************************************
- *
- * Properties
- *
- ***************************************************************************/
-
- virtual VideoRenderType RenderType();
-
- virtual RawVideoType PerferedVideoType();
-
- virtual bool FullScreen();
-
- virtual WebRtc_Word32
- GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
- WebRtc_UWord64& availableGraphicsMemory) const;
-
- virtual WebRtc_Word32
- GetScreenResolution(WebRtc_UWord32& screenWidth,
- WebRtc_UWord32& screenHeight) const;
-
- virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId);
-
- virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
- const float left, const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
- const unsigned int zOrder,
- const float left, const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
-
- virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 textColorRef,
- const WebRtc_UWord32 backgroundColorRef,
- const float left, const float top,
- const float right, const float bottom);
-
- virtual WebRtc_Word32 SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey, const float left,
- const float top, const float right,
- const float bottom);
-
- // VideoRenderCallback
- virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId,
- VideoFrame& videoFrame);
-
-private:
- WebRtc_Word32 _id;
- CriticalSectionWrapper& _critSect;
- void* _window;
- bool _fullscreen;
-};
-
-} //namespace webrtc
-
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_EXTERNAL_VIDEO_RENDER_EXTERNAL_IMPL_H_
diff --git a/trunk/src/modules/video_render/main/source/i_video_render.h b/trunk/src/modules/video_render/main/source/i_video_render.h
deleted file mode 100644
index 2799a79..0000000
--- a/trunk/src/modules/video_render/main/source/i_video_render.h
+++ /dev/null
@@ -1,133 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_I_VIDEO_RENDER_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_I_VIDEO_RENDER_H_
-
-#include "video_render.h"
-
-namespace webrtc {
-
-// Class definitions
-class IVideoRender
-{
-public:
- /*
- * Constructor/destructor
- */
-
- virtual ~IVideoRender()
- {
- };
-
- virtual WebRtc_Word32 Init() = 0;
-
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id) = 0;
-
- virtual WebRtc_Word32 ChangeWindow(void* window) = 0;
-
- /**************************************************************************
- *
- * Incoming Streams
- *
- ***************************************************************************/
-
- virtual VideoRenderCallback
- * AddIncomingRenderStream(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom) = 0;
-
- virtual WebRtc_Word32
- DeleteIncomingRenderStream(const WebRtc_UWord32 streamId) = 0;
-
- virtual WebRtc_Word32
- GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
- WebRtc_UWord32& zOrder,
- float& left,
- float& top,
- float& right,
- float& bottom) const = 0;
- // Implemented in common code?
- //virtual WebRtc_UWord32 GetNumIncomingRenderStreams() const = 0;
- //virtual bool HasIncomingRenderStream(const WebRtc_UWord16 stramId) const = 0;
-
-
- /**************************************************************************
- *
- * Start/Stop
- *
- ***************************************************************************/
-
- virtual WebRtc_Word32 StartRender() = 0;
-
- virtual WebRtc_Word32 StopRender() = 0;
-
- /**************************************************************************
- *
- * Properties
- *
- ***************************************************************************/
- virtual VideoRenderType RenderType() = 0;
-
- virtual RawVideoType PerferedVideoType() = 0;
-
- virtual bool FullScreen() = 0;
-
- // TODO: This should be treated in platform specific code only
- virtual WebRtc_Word32
- GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
- WebRtc_UWord64& availableGraphicsMemory) const = 0;
-
- virtual WebRtc_Word32
- GetScreenResolution(WebRtc_UWord32& screenWidth,
- WebRtc_UWord32& screenHeight) const = 0;
-
- virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId) = 0;
-
- virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
- const float left,
- const float top,
- const float right,
- const float bottom) = 0;
-
- virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
- const unsigned int zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom) = 0;
-
- virtual WebRtc_Word32 SetTransparentBackground(const bool enable) = 0;
-
- virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 textColorRef,
- const WebRtc_UWord32 backgroundColorRef,
- const float left,
- const float top,
- const float rigth,
- const float bottom) = 0;
-
- virtual WebRtc_Word32 SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey,
- const float left,
- const float top,
- const float right,
- const float bottom) = 0;
-
-};
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_I_VIDEO_RENDER_H_
diff --git a/trunk/src/modules/video_render/main/source/incoming_video_stream.cc b/trunk/src/modules/video_render/main/source/incoming_video_stream.cc
deleted file mode 100644
index 418a325..0000000
--- a/trunk/src/modules/video_render/main/source/incoming_video_stream.cc
+++ /dev/null
@@ -1,413 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "incoming_video_stream.h"
-
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "trace.h"
-#include "thread_wrapper.h"
-#include "video_render_frames.h"
-#include "tick_util.h"
-#include "map_wrapper.h"
-#include "common_video/libyuv/include/libyuv.h"
-
-#include <cassert>
-
-// Platform specifics
-#if defined(_WIN32)
-#include <windows.h>
-#elif defined(WEBRTC_LINUX)
-#include <ctime>
-#include <sys/time.h>
-#else
-#include <sys/time.h>
-#endif
-
-namespace webrtc {
-IncomingVideoStream::IncomingVideoStream(const WebRtc_Word32 moduleId,
- const WebRtc_UWord32 streamId) :
- _moduleId(moduleId),
- _streamId(streamId),
- _streamCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
- _threadCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
- _bufferCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
- _ptrIncomingRenderThread(),
- _deliverBufferEvent(*EventWrapper::Create()),
- _running(false),
- _ptrExternalCallback(NULL),
- _ptrRenderCallback(NULL),
- _renderBuffers(*(new VideoRenderFrames)),
- _callbackVideoType(kVideoI420),
- _callbackWidth(0),
- _callbackHeight(0),
- _incomingRate(0),
- _lastRateCalculationTimeMs(0),
- _numFramesSinceLastCalculation(0),
- _lastRenderedFrame(),
- _tempFrame(),
- _startImage(),
- _timeoutImage(),
- _timeoutTime(),
- _mirrorFramesEnabled(false),
- _mirroring(),
- _transformedVideoFrame()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVideoRenderer, _moduleId,
- "%s created for stream %d", __FUNCTION__, streamId);
-}
-
-IncomingVideoStream::~IncomingVideoStream()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVideoRenderer, _moduleId,
- "%s deleted for stream %d", __FUNCTION__, _streamId);
-
- Stop();
-
- // _ptrIncomingRenderThread - Delete in stop
- delete &_renderBuffers;
- delete &_streamCritsect;
- delete &_bufferCritsect;
- delete &_threadCritsect;
- delete &_deliverBufferEvent;
-
-}
-
-WebRtc_Word32 IncomingVideoStream::ChangeModuleId(const WebRtc_Word32 id)
-{
- CriticalSectionScoped cs(_streamCritsect);
-
- _moduleId = id;
- return 0;
-}
-
-VideoRenderCallback*
-IncomingVideoStream::ModuleCallback()
-{
- CriticalSectionScoped cs(_streamCritsect);
- return this;
-}
-
-WebRtc_Word32 IncomingVideoStream::RenderFrame(const WebRtc_UWord32 streamId,
- VideoFrame& videoFrame)
-{
-
- CriticalSectionScoped csS(_streamCritsect);
- WEBRTC_TRACE(kTraceStream, kTraceVideoRenderer, _moduleId,
- "%s for stream %d, render time: %u", __FUNCTION__, _streamId,
- videoFrame.RenderTimeMs());
-
- if (!_running)
- {
- WEBRTC_TRACE(kTraceStream, kTraceVideoRenderer, _moduleId,
- "%s: Not running", __FUNCTION__);
- return -1;
- }
-
- if (true == _mirrorFramesEnabled)
- {
- _transformedVideoFrame.VerifyAndAllocate(videoFrame.Length());
- if (_mirroring.mirrorXAxis)
- {
- MirrorI420UpDown(videoFrame.Buffer(),
- _transformedVideoFrame.Buffer(),
- videoFrame.Width(), videoFrame.Height());
- _transformedVideoFrame.SetLength(videoFrame.Length());
- _transformedVideoFrame.SetWidth(videoFrame.Width());
- _transformedVideoFrame.SetHeight(videoFrame.Height());
- videoFrame.SwapFrame(_transformedVideoFrame);
- }
- if (_mirroring.mirrorYAxis)
- {
- MirrorI420LeftRight(videoFrame.Buffer(),
- _transformedVideoFrame.Buffer(),
- videoFrame.Width(), videoFrame.Height());
- _transformedVideoFrame.SetLength(videoFrame.Length());
- _transformedVideoFrame.SetWidth(videoFrame.Width());
- _transformedVideoFrame.SetHeight(videoFrame.Height());
- videoFrame.SwapFrame(_transformedVideoFrame);
- }
- }
-
- // Rate statistics
- _numFramesSinceLastCalculation++;
- WebRtc_Word64 nowMs = TickTime::MillisecondTimestamp();
- if (nowMs >= _lastRateCalculationTimeMs + KFrameRatePeriodMs)
- {
- _incomingRate = (WebRtc_UWord32) (1000 * _numFramesSinceLastCalculation
- / (nowMs - _lastRateCalculationTimeMs));
- _numFramesSinceLastCalculation = 0;
- _lastRateCalculationTimeMs = nowMs;
- }
-
- // Insert frame
- CriticalSectionScoped csB(_bufferCritsect);
- if (_renderBuffers.AddFrame(&videoFrame) == 1)
- _deliverBufferEvent.Set();
-
- return 0;
-}
-
-WebRtc_Word32 IncomingVideoStream::SetStartImage(const VideoFrame& videoFrame)
-{
- CriticalSectionScoped csS(_threadCritsect);
- return _startImage.CopyFrame(videoFrame);
-}
-
-WebRtc_Word32 IncomingVideoStream::SetTimeoutImage(const VideoFrame& videoFrame,
- const WebRtc_UWord32 timeout)
-{
- CriticalSectionScoped csS(_threadCritsect);
- _timeoutTime = timeout;
- return _timeoutImage.CopyFrame(videoFrame);
-}
-
-WebRtc_Word32 IncomingVideoStream::SetRenderCallback(VideoRenderCallback* renderCallback)
-{
- CriticalSectionScoped cs(_streamCritsect);
-
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _moduleId,
- "%s(%x) for stream %d", __FUNCTION__, renderCallback,
- _streamId);
- _ptrRenderCallback = renderCallback;
- return 0;
-}
-
-WebRtc_Word32 IncomingVideoStream::EnableMirroring(const bool enable,
- const bool mirrorXAxis,
- const bool mirrorYAxis)
-{
- CriticalSectionScoped cs(_streamCritsect);
- _mirrorFramesEnabled = enable;
- _mirroring.mirrorXAxis = mirrorXAxis;
- _mirroring.mirrorYAxis = mirrorYAxis;
-
- return 0;
-}
-
-WebRtc_Word32 IncomingVideoStream::SetExternalCallback(VideoRenderCallback* externalCallback)
-{
- CriticalSectionScoped cs(_streamCritsect);
-
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _moduleId,
- "%s(%x) for stream %d", __FUNCTION__, externalCallback,
- _streamId);
- _ptrExternalCallback = externalCallback;
- _callbackVideoType = kVideoI420;
- _callbackWidth = 0;
- _callbackHeight = 0;
- return 0;
-}
-
-WebRtc_Word32 IncomingVideoStream::Start()
-{
- CriticalSectionScoped csS(_streamCritsect);
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _moduleId,
- "%s for stream %d", __FUNCTION__, _streamId);
- if (_running)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _moduleId,
- "%s: Already running", __FUNCTION__);
- return 0;
- }
-
- CriticalSectionScoped csT(_threadCritsect);
- assert(_ptrIncomingRenderThread == NULL);
-
- _ptrIncomingRenderThread
- = ThreadWrapper::CreateThread(IncomingVideoStreamThreadFun, this,
- kRealtimePriority,
- "IncomingVideoStreamThread");
- if (!_ptrIncomingRenderThread)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _moduleId,
- "%s: No thread", __FUNCTION__);
- return -1;
- }
-
- unsigned int tId = 0;
- if (_ptrIncomingRenderThread->Start(tId))
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _moduleId,
- "%s: thread started: %u", __FUNCTION__, tId);
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _moduleId,
- "%s: Could not start send thread", __FUNCTION__);
- return -1;
- }
- _deliverBufferEvent.StartTimer(false, KEventStartupTimeMS);
-
- _running = true;
- return 0;
-}
-
-WebRtc_Word32 IncomingVideoStream::Stop()
-{
- CriticalSectionScoped csStream(_streamCritsect);
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _moduleId,
- "%s for stream %d", __FUNCTION__, _streamId);
-
- if (!_running)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _moduleId,
- "%s: Not running", __FUNCTION__);
- return 0;
- }
-
- _threadCritsect.Enter();
- if (_ptrIncomingRenderThread)
- {
- ThreadWrapper* ptrThread = _ptrIncomingRenderThread;
- _ptrIncomingRenderThread = NULL;
- ptrThread->SetNotAlive();
-#ifndef _WIN32
- _deliverBufferEvent.StopTimer();
-#endif
- _threadCritsect.Leave();
- if (ptrThread->Stop())
- {
- delete ptrThread;
- }
- else
- {
- assert(false);
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _moduleId,
- "%s: Not able to stop thread, leaking", __FUNCTION__);
- }
- }
- else
- {
- _threadCritsect.Leave();
- }
- _running = false;
- return 0;
-}
-
-WebRtc_Word32 IncomingVideoStream::Reset()
-{
- CriticalSectionScoped csStream(_streamCritsect);
- CriticalSectionScoped csBuffer(_bufferCritsect);
-
- _renderBuffers.ReleaseAllFrames();
- return 0;
-}
-
-WebRtc_UWord32 IncomingVideoStream::StreamId() const
-{
- CriticalSectionScoped csStream(_streamCritsect);
- return _streamId;
-}
-
-WebRtc_UWord32 IncomingVideoStream::IncomingRate() const
-{
- CriticalSectionScoped cs(_streamCritsect);
- return _incomingRate;
-}
-
-bool IncomingVideoStream::IncomingVideoStreamThreadFun(void* obj)
-{
- return static_cast<IncomingVideoStream*> (obj)->IncomingVideoStreamProcess();
-}
-
-bool IncomingVideoStream::IncomingVideoStreamProcess()
-{
- if (kEventError != _deliverBufferEvent.Wait(KEventMaxWaitTimeMs))
- {
- if (_ptrIncomingRenderThread == NULL)
- {
- // Terminating
- return false;
- }
-
- _threadCritsect.Enter();
-
- VideoFrame* ptrFrameToRender = NULL;
-
- // Get a new frame to render and the time for the frame after this one.
- _bufferCritsect.Enter();
- ptrFrameToRender = _renderBuffers.FrameToRender();
- WebRtc_UWord32 waitTime = _renderBuffers.TimeToNextFrameRelease();
- _bufferCritsect.Leave();
-
- // Set timer for next frame to render
- if (waitTime > KEventMaxWaitTimeMs)
- {
- waitTime = KEventMaxWaitTimeMs;
- }
- _deliverBufferEvent.StartTimer(false, waitTime);
-
- if (!ptrFrameToRender)
- {
- if (_ptrRenderCallback)
- {
- if (_lastRenderedFrame.RenderTimeMs() == 0
- && _startImage.Size()) // And we have not rendered anything and have a start image
- {
- _tempFrame.CopyFrame(_startImage);// Copy the startimage if the renderer modifies the render buffer.
- _ptrRenderCallback->RenderFrame(_streamId, _tempFrame);
- }
- else if (_timeoutImage.Size()
- && _lastRenderedFrame.RenderTimeMs() + _timeoutTime
- < TickTime::MillisecondTimestamp()) // We have rendered something a long time ago and have a timeout image
- {
- _tempFrame.CopyFrame(_timeoutImage); // Copy the timeoutImage if the renderer modifies the render buffer.
- _ptrRenderCallback->RenderFrame(_streamId, _tempFrame);
- }
- }
-
- // No frame
- _threadCritsect.Leave();
- return true;
- }
-
- // Send frame for rendering
- if (_ptrExternalCallback)
- {
- WEBRTC_TRACE(kTraceStream,
- kTraceVideoRenderer,
- _moduleId,
- "%s: executing external renderer callback to deliver frame",
- __FUNCTION__, ptrFrameToRender->RenderTimeMs());
- _ptrExternalCallback->RenderFrame(_streamId, *ptrFrameToRender);
- }
- else
- {
- if (_ptrRenderCallback)
- {
- WEBRTC_TRACE(kTraceStream, kTraceVideoRenderer, _moduleId,
- "%s: Render frame, time: ", __FUNCTION__,
- ptrFrameToRender->RenderTimeMs());
- _ptrRenderCallback->RenderFrame(_streamId, *ptrFrameToRender);
- }
- }
-
- // Release critsect before calling the module user
- _threadCritsect.Leave();
-
- // We're done with this frame, delete it.
- if (ptrFrameToRender)
- {
- CriticalSectionScoped cs(_bufferCritsect);
- _lastRenderedFrame.SwapFrame(*ptrFrameToRender);
- _renderBuffers.ReturnFrame(ptrFrameToRender);
- }
- }
- return true;
-}
-WebRtc_Word32 IncomingVideoStream::GetLastRenderedFrame(VideoFrame& videoFrame) const
-{
- CriticalSectionScoped cs(_bufferCritsect);
- return videoFrame.CopyFrame(_lastRenderedFrame);
-}
-
-} //namespace webrtc
-
diff --git a/trunk/src/modules/video_render/main/source/incoming_video_stream.h b/trunk/src/modules/video_render/main/source/incoming_video_stream.h
deleted file mode 100644
index cd2785d..0000000
--- a/trunk/src/modules/video_render/main/source/incoming_video_stream.h
+++ /dev/null
@@ -1,140 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_INCOMING_VIDEO_STREAM_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_INCOMING_VIDEO_STREAM_H_
-
-#include "video_render.h"
-#include "map_wrapper.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-class ThreadWrapper;
-class VideoRenderCallback;
-class VideoRenderFrames;
-
-struct VideoMirroring
-{
- bool mirrorXAxis;
- bool mirrorYAxis;
- VideoMirroring() :
- mirrorXAxis(false), mirrorYAxis(false)
- {
- }
-};
-
-// Class definitions
-class IncomingVideoStream: public VideoRenderCallback
-{
-public:
- /*
- * VideoRenderer constructor/destructor
- */
- IncomingVideoStream(const WebRtc_Word32 moduleId,
- const WebRtc_UWord32 streamId);
- ~IncomingVideoStream();
-
- WebRtc_Word32 ChangeModuleId(const WebRtc_Word32 id);
-
- // Get callbck to deliver frames to the module
- VideoRenderCallback* ModuleCallback();
- virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId,
- VideoFrame& videoFrame);
-
- // Set callback to the platform dependant code
- WebRtc_Word32 SetRenderCallback(VideoRenderCallback* renderCallback);
-
- // Callback for file recording, snapshot, ...
- WebRtc_Word32 SetExternalCallback(VideoRenderCallback* renderObject);
-
- /*
- * Start/Stop
- */
- WebRtc_Word32 Start();
- WebRtc_Word32 Stop();
-
- // Clear all buffers
- WebRtc_Word32 Reset();
-
- /*
- * Properties
- */
- WebRtc_UWord32 StreamId() const;
- WebRtc_UWord32 IncomingRate() const;
-
- /*
- *
- */
- WebRtc_Word32 GetLastRenderedFrame(VideoFrame& videoFrame) const;
-
- WebRtc_Word32 SetStartImage(const VideoFrame& videoFrame);
-
- WebRtc_Word32 SetTimeoutImage(const VideoFrame& videoFrame,
- const WebRtc_UWord32 timeout);
-
- WebRtc_Word32 EnableMirroring(const bool enable,
- const bool mirrorXAxis,
- const bool mirrorYAxis);
-
-protected:
- static bool IncomingVideoStreamThreadFun(void* obj);
- bool IncomingVideoStreamProcess();
-
-private:
-
- // Enums
- enum
- {
- KEventStartupTimeMS = 10
- };
- enum
- {
- KEventMaxWaitTimeMs = 100
- };
- enum
- {
- KFrameRatePeriodMs = 1000
- };
-
- WebRtc_Word32 _moduleId;
- WebRtc_UWord32 _streamId;
- CriticalSectionWrapper& _streamCritsect; // Critsects in allowed to enter order
- CriticalSectionWrapper& _threadCritsect;
- CriticalSectionWrapper& _bufferCritsect;
- ThreadWrapper* _ptrIncomingRenderThread;
- EventWrapper& _deliverBufferEvent;
- bool _running;
-
- VideoRenderCallback* _ptrExternalCallback;
- VideoRenderCallback* _ptrRenderCallback;
- VideoRenderFrames& _renderBuffers;
-
- RawVideoType _callbackVideoType;
- WebRtc_UWord32 _callbackWidth;
- WebRtc_UWord32 _callbackHeight;
-
- WebRtc_UWord32 _incomingRate;
- WebRtc_Word64 _lastRateCalculationTimeMs;
- WebRtc_UWord16 _numFramesSinceLastCalculation;
- VideoFrame _lastRenderedFrame;
- VideoFrame _tempFrame;
- VideoFrame _startImage;
- VideoFrame _timeoutImage;
- WebRtc_UWord32 _timeoutTime;
-
- bool _mirrorFramesEnabled;
- VideoMirroring _mirroring;
- VideoFrame _transformedVideoFrame;
-};
-
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_INCOMING_VIDEO_STREAM_H_
diff --git a/trunk/src/modules/video_render/main/source/linux/video_render_linux_impl.cc b/trunk/src/modules/video_render/main/source/linux/video_render_linux_impl.cc
deleted file mode 100644
index a5e311a..0000000
--- a/trunk/src/modules/video_render/main/source/linux/video_render_linux_impl.cc
+++ /dev/null
@@ -1,271 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_render_linux_impl.h"
-
-#include "critical_section_wrapper.h"
-#include "trace.h"
-#include "video_x11_render.h"
-
-#include <X11/Xlib.h>
-
-namespace webrtc {
-
-VideoRenderLinuxImpl::VideoRenderLinuxImpl(
- const WebRtc_Word32 id,
- const VideoRenderType videoRenderType,
- void* window, const bool fullscreen) :
- _id(id),
- _renderLinuxCritsect(
- *CriticalSectionWrapper::CreateCriticalSection()),
- _ptrWindow(window), _fullscreen(fullscreen), _ptrX11Render(NULL),
- _renderType(videoRenderType)
-{
-}
-
-VideoRenderLinuxImpl::~VideoRenderLinuxImpl()
-{
- if (_ptrX11Render)
- delete _ptrX11Render;
-
- delete &_renderLinuxCritsect;
-}
-
-WebRtc_Word32 VideoRenderLinuxImpl::Init()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
- __FUNCTION__);
-
- CriticalSectionScoped cs(_renderLinuxCritsect);
- _ptrX11Render = new VideoX11Render((Window) _ptrWindow);
- if (!_ptrX11Render)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s",
- "Failed to create instance of VideoX11Render object");
- return -1;
- }
- int retVal = _ptrX11Render->Init();
- if (retVal == -1)
- {
- return -1;
- }
-
- return 0;
-
-}
-
-WebRtc_Word32 VideoRenderLinuxImpl::ChangeUniqueId(const WebRtc_Word32 id)
-{
- CriticalSectionScoped cs(_renderLinuxCritsect);
-
- _id = id;
- return 0;
-}
-
-WebRtc_Word32 VideoRenderLinuxImpl::ChangeWindow(void* window)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
- __FUNCTION__);
-
- CriticalSectionScoped cs(_renderLinuxCritsect);
- _ptrWindow = window;
-
- if (_ptrX11Render)
- {
- return _ptrX11Render->ChangeWindow((Window) window);
- }
-
- return -1;
-}
-
-VideoRenderCallback* VideoRenderLinuxImpl::AddIncomingRenderStream(
- const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
- __FUNCTION__);
- CriticalSectionScoped cs(_renderLinuxCritsect);
-
- VideoRenderCallback* renderCallback = NULL;
- if (_ptrX11Render)
- {
- VideoX11Channel* renderChannel =
- _ptrX11Render->CreateX11RenderChannel(streamId, zOrder, left,
- top, right, bottom);
- if (!renderChannel)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "Render channel creation failed for stream id: %d",
- streamId);
- return NULL;
- }
- renderCallback = (VideoRenderCallback *) renderChannel;
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "_ptrX11Render is NULL");
- return NULL;
- }
- return renderCallback;
-}
-
-WebRtc_Word32 VideoRenderLinuxImpl::DeleteIncomingRenderStream(
- const WebRtc_UWord32 streamId)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
- __FUNCTION__);
- CriticalSectionScoped cs(_renderLinuxCritsect);
-
- if (_ptrX11Render)
- {
- return _ptrX11Render->DeleteX11RenderChannel(streamId);
- }
- return -1;
-}
-
-WebRtc_Word32 VideoRenderLinuxImpl::GetIncomingRenderStreamProperties(
- const WebRtc_UWord32 streamId,
- WebRtc_UWord32& zOrder,
- float& left,
- float& top,
- float& right,
- float& bottom) const
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
- __FUNCTION__);
- CriticalSectionScoped cs(_renderLinuxCritsect);
-
- if (_ptrX11Render)
- {
- return _ptrX11Render->GetIncomingStreamProperties(streamId, zOrder,
- left, top, right,
- bottom);
- }
- return -1;
-}
-
-WebRtc_Word32 VideoRenderLinuxImpl::StartRender()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
- __FUNCTION__);
- return 0;
-}
-
-WebRtc_Word32 VideoRenderLinuxImpl::StopRender()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s",
- __FUNCTION__);
- return 0;
-}
-
-VideoRenderType VideoRenderLinuxImpl::RenderType()
-{
- return kRenderX11;
-}
-
-RawVideoType VideoRenderLinuxImpl::PerferedVideoType()
-{
- return kVideoI420;
-}
-
-bool VideoRenderLinuxImpl::FullScreen()
-{
- return false;
-}
-
-WebRtc_Word32 VideoRenderLinuxImpl::GetGraphicsMemory(
- WebRtc_UWord64& /*totalGraphicsMemory*/,
- WebRtc_UWord64& /*availableGraphicsMemory*/) const
-{
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s - not supported on Linux", __FUNCTION__);
- return -1;
-}
-
-WebRtc_Word32 VideoRenderLinuxImpl::GetScreenResolution(
- WebRtc_UWord32& /*screenWidth*/,
- WebRtc_UWord32& /*screenHeight*/) const
-{
- return -1;
-}
-
-WebRtc_UWord32 VideoRenderLinuxImpl::RenderFrameRate(const WebRtc_UWord32 /*streamId*/)
-{
- return -1;
-}
-
-WebRtc_Word32 VideoRenderLinuxImpl::SetStreamCropping(
- const WebRtc_UWord32 /*streamId*/,
- const float /*left*/,
- const float /*top*/,
- const float /*right*/,
- const float /*bottom*/)
-{
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s - not supported on Linux", __FUNCTION__);
- return -1;
-}
-
-WebRtc_Word32 VideoRenderLinuxImpl::SetTransparentBackground(const bool /*enable*/)
-{
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s - not supported on Linux", __FUNCTION__);
- return -1;
-}
-
-WebRtc_Word32 VideoRenderLinuxImpl::ConfigureRenderer(
- const WebRtc_UWord32 streamId,
- const unsigned int zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s - not supported on Linux", __FUNCTION__);
- return -1;
-}
-
-WebRtc_Word32 VideoRenderLinuxImpl::SetText(
- const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 textColorRef,
- const WebRtc_UWord32 backgroundColorRef,
- const float left, const float top,
- const float rigth,
- const float bottom)
-{
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s - not supported on Linux", __FUNCTION__);
- return -1;
-}
-
-WebRtc_Word32 VideoRenderLinuxImpl::SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s - not supported on Linux", __FUNCTION__);
- return -1;
-}
-
-} //namespace webrtc
-
diff --git a/trunk/src/modules/video_render/main/source/linux/video_render_linux_impl.h b/trunk/src/modules/video_render/main/source/linux/video_render_linux_impl.h
deleted file mode 100644
index 10460ec..0000000
--- a/trunk/src/modules/video_render/main/source/linux/video_render_linux_impl.h
+++ /dev/null
@@ -1,136 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_RENDER_LINUX_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_RENDER_LINUX_IMPL_H_
-
-#include "i_video_render.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class VideoX11Render;
-
-// Class definitions
-class VideoRenderLinuxImpl: IVideoRender
-{
-public:
- /*
- * Constructor/destructor
- */
-
- VideoRenderLinuxImpl(const WebRtc_Word32 id,
- const VideoRenderType videoRenderType,
- void* window, const bool fullscreen);
-
- virtual ~VideoRenderLinuxImpl();
-
- virtual WebRtc_Word32 Init();
-
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- virtual WebRtc_Word32 ChangeWindow(void* window);
-
- /**************************************************************************
- *
- * Incoming Streams
- *
- ***************************************************************************/
-
- virtual VideoRenderCallback
- * AddIncomingRenderStream(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left, const float top,
- const float right, const float bottom);
-
- virtual WebRtc_Word32
- DeleteIncomingRenderStream(const WebRtc_UWord32 streamId);
-
- virtual WebRtc_Word32
- GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
- WebRtc_UWord32& zOrder,
- float& left, float& top,
- float& right, float& bottom) const;
-
- /**************************************************************************
- *
- * Start/Stop
- *
- ***************************************************************************/
-
- virtual WebRtc_Word32 StartRender();
-
- virtual WebRtc_Word32 StopRender();
-
- /**************************************************************************
- *
- * Properties
- *
- ***************************************************************************/
-
- virtual VideoRenderType RenderType();
-
- virtual RawVideoType PerferedVideoType();
-
- virtual bool FullScreen();
-
- virtual WebRtc_Word32
- GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
- WebRtc_UWord64& availableGraphicsMemory) const;
-
- virtual WebRtc_Word32
- GetScreenResolution(WebRtc_UWord32& screenWidth,
- WebRtc_UWord32& screenHeight) const;
-
- virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId);
-
- virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
- const float left, const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
-
- virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
- const unsigned int zOrder,
- const float left, const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 textColorRef,
- const WebRtc_UWord32 backgroundColorRef,
- const float left, const float top,
- const float rigth, const float bottom);
-
- virtual WebRtc_Word32 SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey, const float left,
- const float top, const float right,
- const float bottom);
-
-private:
- WebRtc_Word32 _id;
- CriticalSectionWrapper& _renderLinuxCritsect;
-
- void* _ptrWindow;
- bool _fullscreen;
-
- // X11 Render
- VideoX11Render* _ptrX11Render;
-
- VideoRenderType _renderType;
-
-};
-
-} //namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_RENDER_LINUX_IMPL_H_
diff --git a/trunk/src/modules/video_render/main/source/linux/video_x11_channel.cc b/trunk/src/modules/video_render/main/source/linux/video_x11_channel.cc
deleted file mode 100644
index 5f8bfa1..0000000
--- a/trunk/src/modules/video_render/main/source/linux/video_x11_channel.cc
+++ /dev/null
@@ -1,332 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_x11_channel.h"
-
-#include "critical_section_wrapper.h"
-#include "trace.h"
-
-namespace webrtc {
-
-#define DISP_MAX 128
-
-static Display *dispArray[DISP_MAX];
-static int dispCount = 0;
-
-
-VideoX11Channel::VideoX11Channel(WebRtc_Word32 id) :
- _crit(*CriticalSectionWrapper::CreateCriticalSection()), _display(NULL),
- _shminfo(), _image(NULL), _window(0L), _gc(NULL),
- _width(DEFAULT_RENDER_FRAME_WIDTH),
- _height(DEFAULT_RENDER_FRAME_HEIGHT), _outWidth(0), _outHeight(0),
- _xPos(0), _yPos(0), _prepared(false), _dispCount(0), _buffer(NULL),
- _top(0.0), _left(0.0), _right(0.0), _bottom(0.0),
- _Id(id)
-{
-}
-
-VideoX11Channel::~VideoX11Channel()
-{
- if (_prepared)
- {
- _crit.Enter();
- ReleaseWindow();
- _crit.Leave();
- }
- delete &_crit;
-}
-
-WebRtc_Word32 VideoX11Channel::RenderFrame(const WebRtc_UWord32 streamId,
- VideoFrame& videoFrame)
-{
- CriticalSectionScoped cs(_crit);
- if (_width != (WebRtc_Word32) videoFrame.Width() || _height
- != (WebRtc_Word32) videoFrame.Height())
- {
- if (FrameSizeChange(videoFrame.Width(), videoFrame.Height(), 1) == -1)
- {
- return -1;
- }
- }
- return DeliverFrame(videoFrame.Buffer(), videoFrame.Length(),
- videoFrame.TimeStamp());
-}
-
-WebRtc_Word32 VideoX11Channel::FrameSizeChange(WebRtc_Word32 width,
- WebRtc_Word32 height,
- WebRtc_Word32 /*numberOfStreams */)
-{
- CriticalSectionScoped cs(_crit);
- if (_prepared)
- {
- RemoveRenderer();
- }
- if (CreateLocalRenderer(width, height) == -1)
- {
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 VideoX11Channel::DeliverFrame(unsigned char* buffer,
- WebRtc_Word32 bufferSize,
- unsigned WebRtc_Word32 /*timeStamp90kHz*/)
-{
- CriticalSectionScoped cs(_crit);
- if (!_prepared)
- {
- return 0;
- }
-
- if (!dispArray[_dispCount])
- {
- return -1;
- }
-
- unsigned char *pBuf = buffer;
- // convert to RGB32, setting stride = width.
- ConvertFromI420(pBuf, _width, kARGB, 0, _width, _height, _buffer);
-
- // put image in window
- XShmPutImage(_display, _window, _gc, _image, 0, 0, _xPos, _yPos, _width,
- _height, True);
-
- // very important for the image to update properly!
- XSync(_display, False);
- return 0;
-
-}
-
-WebRtc_Word32 VideoX11Channel::GetFrameSize(WebRtc_Word32& width,
- WebRtc_Word32& height)
-{
- width = _width;
- height = _height;
-
- return 0;
-}
-
-WebRtc_Word32 VideoX11Channel::Init(Window window, float left, float top,
- float right, float bottom)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
- __FUNCTION__);
- CriticalSectionScoped cs(_crit);
-
- _window = window;
- _left = left;
- _right = right;
- _top = top;
- _bottom = bottom;
-
- _display = XOpenDisplay(NULL); // Use default display
- if (!_window || !_display)
- {
- return -1;
- }
-
- if (dispCount < DISP_MAX)
- {
- dispArray[dispCount] = _display;
- _dispCount = dispCount;
- dispCount++;
- }
- else
- {
- return -1;
- }
-
- if ((1 < left || left < 0) || (1 < top || top < 0) || (1 < right || right
- < 0) || (1 < bottom || bottom < 0))
- {
- return -1;
- }
-
- // calculate position and size of rendered video
- int x, y;
- unsigned int winWidth, winHeight, borderwidth, depth;
- Window rootret;
- if (XGetGeometry(_display, _window, &rootret, &x, &y, &winWidth,
- &winHeight, &borderwidth, &depth) == 0)
- {
- return -1;
- }
-
- _xPos = (WebRtc_Word32) (winWidth * left);
- _yPos = (WebRtc_Word32) (winHeight * top);
- _outWidth = (WebRtc_Word32) (winWidth * (right - left));
- _outHeight = (WebRtc_Word32) (winHeight * (bottom - top));
- if (_outWidth % 2)
- _outWidth++; // the renderer want's sizes that are multiples of two
- if (_outHeight % 2)
- _outHeight++;
-
- _gc = XCreateGC(_display, _window, 0, 0);
- if (!_gc) {
- // Failed to create the graphics context.
- assert(false);
- return -1;
- }
-
- if (CreateLocalRenderer(winWidth, winHeight) == -1)
- {
- return -1;
- }
- return 0;
-
-}
-
-WebRtc_Word32 VideoX11Channel::ChangeWindow(Window window)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
- __FUNCTION__);
- CriticalSectionScoped cs(_crit);
-
- // Stop the rendering, if we are rendering...
- RemoveRenderer();
- _window = window;
-
- // calculate position and size of rendered video
- int x, y;
- unsigned int winWidth, winHeight, borderwidth, depth;
- Window rootret;
- if (XGetGeometry(_display, _window, &rootret, &x, &y, &winWidth,
- &winHeight, &borderwidth, &depth) == -1)
- {
- return -1;
- }
- _xPos = (int) (winWidth * _left);
- _yPos = (int) (winHeight * _top);
- _outWidth = (int) (winWidth * (_right - _left));
- _outHeight = (int) (winHeight * (_bottom - _top));
- if (_outWidth % 2)
- _outWidth++; // the renderer want's sizes that are multiples of two
- if (_outHeight % 2)
- _outHeight++;
-
- // Prepare rendering using the
- if (CreateLocalRenderer(_width, _height) == -1)
- {
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 VideoX11Channel::ReleaseWindow()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
- __FUNCTION__);
- CriticalSectionScoped cs(_crit);
-
- RemoveRenderer();
- if (_gc) {
- XFreeGC(_display, _gc);
- _gc = NULL;
- }
- if (_display)
- {
- XCloseDisplay(_display);
- _display = NULL;
- }
- return 0;
-}
-
-WebRtc_Word32 VideoX11Channel::CreateLocalRenderer(WebRtc_Word32 width,
- WebRtc_Word32 height)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
- __FUNCTION__);
- CriticalSectionScoped cs(_crit);
-
- if (!_window || !_display)
- {
- return -1;
- }
-
- if (_prepared)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _Id,
- "Renderer already prepared, exits.");
- return -1;
- }
-
- _width = width;
- _height = height;
-
- // create shared memory image
- _image = XShmCreateImage(_display, CopyFromParent, 24, ZPixmap, NULL,
- &_shminfo, _width, _height); // this parameter needs to be the same for some reason.
- _shminfo.shmid = shmget(IPC_PRIVATE, (_image->bytes_per_line
- * _image->height), IPC_CREAT | 0777);
- _shminfo.shmaddr = _image->data = (char*) shmat(_shminfo.shmid, 0, 0);
- if (_image->data == reinterpret_cast<char*>(-1))
- {
- return -1;
- }
- _buffer = (unsigned char*) _image->data;
- _shminfo.readOnly = False;
-
- // attach image to display
- if (!XShmAttach(_display, &_shminfo))
- {
- //printf("XShmAttach failed !\n");
- return -1;
- }
- XSync(_display, False);
-
- _prepared = true;
- return 0;
-}
-
-WebRtc_Word32 VideoX11Channel::RemoveRenderer()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
- __FUNCTION__);
-
- if (!_prepared)
- {
- return 0;
- }
- _prepared = false;
-
- // Free the memory.
- XShmDetach(_display, &_shminfo);
- XDestroyImage( _image );
- _image = NULL;
- shmdt(_shminfo.shmaddr);
- _shminfo.shmaddr = NULL;
- _buffer = NULL;
- shmctl(_shminfo.shmid, IPC_RMID, 0);
- _shminfo.shmid = 0;
- return 0;
-}
-
-WebRtc_Word32 VideoX11Channel::GetStreamProperties(WebRtc_UWord32& zOrder,
- float& left, float& top,
- float& right,
- float& bottom) const
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _Id, "%s",
- __FUNCTION__);
-
- zOrder = 0; // no z-order support yet
- left = _left;
- top = _top;
- right = _right;
- bottom = _bottom;
-
- return 0;
-}
-
-
-} //namespace webrtc
-
-
diff --git a/trunk/src/modules/video_render/main/source/linux/video_x11_channel.h b/trunk/src/modules/video_render/main/source/linux/video_x11_channel.h
deleted file mode 100644
index 1fb2dab..0000000
--- a/trunk/src/modules/video_render/main/source/linux/video_x11_channel.h
+++ /dev/null
@@ -1,97 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_CHANNEL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_CHANNEL_H_
-
-#include "video_render_defines.h"
-#include "common_video/libyuv/include/libyuv.h"
-#include <sys/shm.h>
-
-#include <X11/Xlib.h>
-#include <X11/Xutil.h>
-#include <X11/extensions/XShm.h>
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-#define DEFAULT_RENDER_FRAME_WIDTH 352
-#define DEFAULT_RENDER_FRAME_HEIGHT 288
-
-
-class VideoX11Channel: public VideoRenderCallback
-{
-public:
- VideoX11Channel(WebRtc_Word32 id);
-
- virtual ~VideoX11Channel();
-
- virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId,
- VideoFrame& videoFrame);
-
- WebRtc_Word32 FrameSizeChange(WebRtc_Word32 width, WebRtc_Word32 height,
- WebRtc_Word32 numberOfStreams);
- WebRtc_Word32 DeliverFrame(unsigned char* buffer, WebRtc_Word32 bufferSize,
- unsigned WebRtc_Word32 /*timeStamp90kHz*/);
- WebRtc_Word32 GetFrameSize(WebRtc_Word32& width, WebRtc_Word32& height);
- WebRtc_Word32 Init(Window window, float left, float top, float right,
- float bottom);
- WebRtc_Word32 ChangeWindow(Window window);
- WebRtc_Word32
- GetStreamProperties(WebRtc_UWord32& zOrder, float& left,
- float& top, float& right, float& bottom) const;
- WebRtc_Word32 ReleaseWindow();
-
- bool IsPrepared()
- {
- return _prepared;
- }
-
-private:
-
- WebRtc_Word32
- CreateLocalRenderer(WebRtc_Word32 width, WebRtc_Word32 height);
- WebRtc_Word32 RemoveRenderer();
-
- //FIXME a better place for this method? the GetWidthHeight no longer
- // supported by common_video.
- int GetWidthHeight(VideoType type, int bufferSize, int& width,
- int& height);
-
- CriticalSectionWrapper& _crit;
-
- Display* _display;
- XShmSegmentInfo _shminfo;
- XImage* _image;
- Window _window;
- GC _gc;
- WebRtc_Word32 _width; // incoming frame width
- WebRtc_Word32 _height; // incoming frame height
- WebRtc_Word32 _outWidth; // render frame width
- WebRtc_Word32 _outHeight; // render frame height
- WebRtc_Word32 _xPos; // position within window
- WebRtc_Word32 _yPos;
- bool _prepared; // true if ready to use
- WebRtc_Word32 _dispCount;
-
- unsigned char* _buffer;
- float _top;
- float _left;
- float _right;
- float _bottom;
-
- WebRtc_Word32 _Id;
-
-};
-
-
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_CHANNEL_H_
diff --git a/trunk/src/modules/video_render/main/source/linux/video_x11_render.cc b/trunk/src/modules/video_render/main/source/linux/video_x11_render.cc
deleted file mode 100644
index a3543e3..0000000
--- a/trunk/src/modules/video_render/main/source/linux/video_x11_render.cc
+++ /dev/null
@@ -1,154 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_x11_render.h"
-#include "video_x11_channel.h"
-
-#include "critical_section_wrapper.h"
-#include "trace.h"
-
-namespace webrtc {
-
-VideoX11Render::VideoX11Render(Window window) :
- _window(window),
- _critSect(*CriticalSectionWrapper::CreateCriticalSection())
-{
-}
-
-VideoX11Render::~VideoX11Render()
-{
- delete &_critSect;
-}
-
-WebRtc_Word32 VideoX11Render::Init()
-{
- CriticalSectionScoped cs(_critSect);
-
- _streamIdToX11ChannelMap.clear();
-
- return 0;
-}
-
-WebRtc_Word32 VideoX11Render::ChangeWindow(Window window)
-{
- CriticalSectionScoped cs(_critSect);
- VideoX11Channel* renderChannel = NULL;
-
- std::map<int, VideoX11Channel*>::iterator iter =
- _streamIdToX11ChannelMap.begin();
-
- while (iter != _streamIdToX11ChannelMap.end())
- {
- renderChannel = iter->second;
- if (renderChannel)
- {
- renderChannel->ChangeWindow(window);
- }
- iter++;
- }
-
- _window = window;
-
- return 0;
-}
-
-VideoX11Channel* VideoX11Render::CreateX11RenderChannel(
- WebRtc_Word32 streamId,
- WebRtc_Word32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_critSect);
- VideoX11Channel* renderChannel = NULL;
-
- std::map<int, VideoX11Channel*>::iterator iter =
- _streamIdToX11ChannelMap.find(streamId);
-
- if (iter == _streamIdToX11ChannelMap.end())
- {
- renderChannel = new VideoX11Channel(streamId);
- if (!renderChannel)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideoRenderer,
- -1,
- "Failed to create VideoX11Channel for streamId : %d",
- streamId);
- return NULL;
- }
- renderChannel->Init(_window, left, top, right, bottom);
- _streamIdToX11ChannelMap[streamId] = renderChannel;
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, -1,
- "Render Channel already exists for streamId: %d", streamId);
- renderChannel = iter->second;
- }
-
- return renderChannel;
-}
-
-WebRtc_Word32 VideoX11Render::DeleteX11RenderChannel(WebRtc_Word32 streamId)
-{
- CriticalSectionScoped cs(_critSect);
-
- std::map<int, VideoX11Channel*>::iterator iter =
- _streamIdToX11ChannelMap.find(streamId);
- if (iter != _streamIdToX11ChannelMap.end())
- {
- VideoX11Channel *renderChannel = iter->second;
- if (renderChannel)
- {
- renderChannel->ReleaseWindow();
- delete renderChannel;
- renderChannel = NULL;
- }
- _streamIdToX11ChannelMap.erase(iter);
- }
-
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
- "No VideoX11Channel object exists for stream id: %d",
- streamId);
- return -1;
-}
-
-WebRtc_Word32 VideoX11Render::GetIncomingStreamProperties(
- WebRtc_Word32 streamId,
- WebRtc_UWord32& zOrder,
- float& left,
- float& top,
- float& right,
- float& bottom)
-{
- CriticalSectionScoped cs(_critSect);
-
- std::map<int, VideoX11Channel*>::iterator iter =
- _streamIdToX11ChannelMap.find(streamId);
- if (iter != _streamIdToX11ChannelMap.end())
- {
- VideoX11Channel *renderChannel = iter->second;
- if (renderChannel)
- {
- renderChannel->GetStreamProperties(zOrder, left, top, right, bottom);
- }
- }
-
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
- "No VideoX11Channel object exists for stream id: %d",
- streamId);
- return -1;
-}
-
-} //namespace webrtc
-
diff --git a/trunk/src/modules/video_render/main/source/linux/video_x11_render.h b/trunk/src/modules/video_render/main/source/linux/video_x11_render.h
deleted file mode 100644
index 9b140ef..0000000
--- a/trunk/src/modules/video_render/main/source/linux/video_x11_render.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_RENDER_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_RENDER_H_
-
-#include "video_render_defines.h"
-
-#include <X11/Xlib.h>
-#include <map>
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class VideoX11Channel;
-
-class VideoX11Render
-{
-
-public:
- VideoX11Render(Window window);
- ~VideoX11Render();
-
- WebRtc_Word32 Init();
- WebRtc_Word32 ChangeWindow(Window window);
-
- VideoX11Channel* CreateX11RenderChannel(WebRtc_Word32 streamId,
- WebRtc_Word32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- WebRtc_Word32 DeleteX11RenderChannel(WebRtc_Word32 streamId);
-
- WebRtc_Word32 GetIncomingStreamProperties(WebRtc_Word32 streamId,
- WebRtc_UWord32& zOrder,
- float& left, float& top,
- float& right, float& bottom);
-
-private:
- Window _window;
- CriticalSectionWrapper& _critSect;
- std::map<int, VideoX11Channel*> _streamIdToX11ChannelMap;
-
-};
-
-
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_LINUX_VIDEO_X11_RENDER_H_
diff --git a/trunk/src/modules/video_render/main/source/mac/cocoa_full_screen_window.h b/trunk/src/modules/video_render/main/source/mac/cocoa_full_screen_window.h
deleted file mode 100644
index c8e98bb..0000000
--- a/trunk/src/modules/video_render/main/source/mac/cocoa_full_screen_window.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// cocoa_full_screen_window.h
-//
-//
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_FULL_SCREEN_WINDOW_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_FULL_SCREEN_WINDOW_H_
-
-#import <Cocoa/Cocoa.h>
-//#define GRAB_ALL_SCREENS 1
-
-@interface CocoaFullScreenWindow : NSObject {
- NSWindow* _window;
-}
-
--(id)init;
--(void)grabFullScreen;
--(void)releaseFullScreen;
--(NSWindow*)window;
-
-@end
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_FULL_SCREEN_WINDOW_H_
diff --git a/trunk/src/modules/video_render/main/source/mac/cocoa_full_screen_window.mm b/trunk/src/modules/video_render/main/source/mac/cocoa_full_screen_window.mm
deleted file mode 100644
index e86bab1..0000000
--- a/trunk/src/modules/video_render/main/source/mac/cocoa_full_screen_window.mm
+++ /dev/null
@@ -1,87 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#import "cocoa_full_screen_window.h"
-#include "trace.h"
-
-using namespace webrtc;
-
-@implementation CocoaFullScreenWindow
-
--(id)init{
-
- self = [super init];
- if(!self){
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, 0, "%s:%d COULD NOT CREATE INSTANCE", __FUNCTION__, __LINE__);
- return nil;
- }
-
-
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, 0, "%s:%d Created instance", __FUNCTION__, __LINE__);
- return self;
-}
-
--(void)grabFullScreen{
-
-#ifdef GRAB_ALL_SCREENS
- if(CGCaptureAllDisplays() != kCGErrorSuccess)
-#else
- if(CGDisplayCapture(kCGDirectMainDisplay) != kCGErrorSuccess)
-#endif
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, 0, "%s:%d Could not capture main level", __FUNCTION__, __LINE__);
- }
-
- // get the shielding window level
- int windowLevel = CGShieldingWindowLevel();
-
- // get the screen rect of main display
- NSRect screenRect = [[NSScreen mainScreen]frame];
-
- _window = [[NSWindow alloc]initWithContentRect:screenRect
- styleMask:NSBorderlessWindowMask
- backing:NSBackingStoreBuffered
- defer:NO
- screen:[NSScreen mainScreen]];
-
- [_window setLevel:windowLevel];
- [_window setBackgroundColor:[NSColor blackColor]];
- [_window makeKeyAndOrderFront:nil];
-
-}
-
--(void)releaseFullScreen
-{
- [_window orderOut:self];
-
-#ifdef GRAB_ALL_SCREENS
- if(CGReleaseAllDisplays() != kCGErrorSuccess)
-#else
- if(CGDisplayRelease(kCGDirectMainDisplay) != kCGErrorSuccess)
-#endif
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, 0, "%s:%d Could not release the displays", __FUNCTION__, __LINE__);
- }
-}
-
-- (NSWindow*)window
-{
- return _window;
-}
-
-- (void) dealloc
-{
- [self releaseFullScreen];
- [super dealloc];
-}
-
-
-
-@end
diff --git a/trunk/src/modules/video_render/main/source/mac/cocoa_render_view.h b/trunk/src/modules/video_render/main/source/mac/cocoa_render_view.h
deleted file mode 100644
index 15a8108..0000000
--- a/trunk/src/modules/video_render/main/source/mac/cocoa_render_view.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// cocoa_render_view.h
-//
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_RENDER_VIEW_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_RENDER_VIEW_H_
-
-#import <Cocoa/Cocoa.h>
-#import <OpenGL/gl.h>
-#import <OpenGL/glu.h>
-#import <OpenGL/OpenGL.h>
-
-@interface CocoaRenderView : NSOpenGLView {
- NSOpenGLContext* _nsOpenGLContext;
-}
-
--(void)initCocoaRenderView:(NSOpenGLPixelFormat*)fmt;
--(void)initCocoaRenderViewFullScreen:(NSOpenGLPixelFormat*)fmt;
--(NSOpenGLContext*)nsOpenGLContext;
-@end
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_COCOA_RENDER_VIEW_H_
diff --git a/trunk/src/modules/video_render/main/source/mac/cocoa_render_view.mm b/trunk/src/modules/video_render/main/source/mac/cocoa_render_view.mm
deleted file mode 100644
index 567d171..0000000
--- a/trunk/src/modules/video_render/main/source/mac/cocoa_render_view.mm
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#import <Cocoa/Cocoa.h>
-#import <AppKit/AppKit.h>
-#import "cocoa_render_view.h"
-#include "trace.h"
-
-using namespace webrtc;
-
-@implementation CocoaRenderView
-
--(void)initCocoaRenderView:(NSOpenGLPixelFormat*)fmt{
-
- self = [super initWithFrame:[self frame] pixelFormat:[fmt autorelease]];
- if (self == nil){
-
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, 0, "%s:%d Could not create instance", __FUNCTION__, __LINE__);
- }
-
-
- _nsOpenGLContext = [self openGLContext];
-
-}
-
--(NSOpenGLContext*)nsOpenGLContext {
- return _nsOpenGLContext;
-}
-
--(void)initCocoaRenderViewFullScreen:(NSOpenGLPixelFormat*)fmt{
-
- NSRect screenRect = [[NSScreen mainScreen]frame];
-// [_windowRef setFrame:screenRect];
-// [_windowRef setBounds:screenRect];
- self = [super initWithFrame:screenRect pixelFormat:[fmt autorelease]];
- if (self == nil){
-
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, 0, "%s:%d Could not create instance", __FUNCTION__, __LINE__);
- }
-
- _nsOpenGLContext = [self openGLContext];
-
-}
-
-@end
-
-
diff --git a/trunk/src/modules/video_render/main/source/mac/video_render_agl.cc b/trunk/src/modules/video_render/main/source/mac/video_render_agl.cc
deleted file mode 100644
index e968865..0000000
--- a/trunk/src/modules/video_render/main/source/mac/video_render_agl.cc
+++ /dev/null
@@ -1,2007 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "engine_configurations.h"
-
-#if defined(CARBON_RENDERING)
-
-#include "video_render_agl.h"
-
-// includes
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "trace.h"
-#include "thread_wrapper.h"
-#include "common_video/libyuv/include/libyuv.h"
-
-namespace webrtc {
-
-/*
- *
- * VideoChannelAGL
- *
- */
-
-#pragma mark VideoChannelAGL constructor
-
-VideoChannelAGL::VideoChannelAGL(AGLContext& aglContext, int iId, VideoRenderAGL* owner) :
- _aglContext( aglContext),
- _id( iId),
- _owner( owner),
- _width( 0),
- _height( 0),
- _stretchedWidth( 0),
- _stretchedHeight( 0),
- _startWidth( 0.0f),
- _startHeight( 0.0f),
- _stopWidth( 0.0f),
- _stopHeight( 0.0f),
- _xOldWidth( 0),
- _yOldHeight( 0),
- _oldStretchedHeight(0),
- _oldStretchedWidth( 0),
- _buffer( 0),
- _bufferSize( 0),
- _incommingBufferSize(0),
- _bufferIsUpdated( false),
- _sizeInitialized( false),
- _numberOfStreams( 0),
- _bVideoSizeStartedChanging(false),
- _pixelFormat( GL_RGBA),
- _pixelDataType( GL_UNSIGNED_INT_8_8_8_8),
- _texture( 0)
-
-{
- //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Constructor", __FUNCTION__, __LINE__);
-}
-
-VideoChannelAGL::~VideoChannelAGL()
-{
- //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Destructor", __FUNCTION__, __LINE__);
- if (_buffer)
- {
- delete [] _buffer;
- _buffer = NULL;
- }
-
- aglSetCurrentContext(_aglContext);
-
- if (_texture != 0)
- {
- glDeleteTextures(1, (const GLuint*) &_texture);
- _texture = 0;
- }
-}
-
-WebRtc_Word32 VideoChannelAGL::RenderFrame(const WebRtc_UWord32 streamId, VideoFrame& videoFrame)
-{
- _owner->LockAGLCntx();
- if(_width != videoFrame.Width() ||
- _height != videoFrame.Height())
- {
- if(FrameSizeChange(videoFrame.Width(), videoFrame.Height(), 1) == -1)
- { //WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d FrameSizeChange returned an error", __FUNCTION__, __LINE__);
- _owner->UnlockAGLCntx();
- return -1;
- }
- }
-
- _owner->UnlockAGLCntx();
- return DeliverFrame(videoFrame.Buffer(), videoFrame.Length(), videoFrame.TimeStamp());
-}
-
-int VideoChannelAGL::UpdateSize(int /*width*/, int /*height*/)
-{
- _owner->LockAGLCntx();
- _owner->UnlockAGLCntx();
- return 0;
-}
-
-int VideoChannelAGL::UpdateStretchSize(int stretchHeight, int stretchWidth)
-{
-
- _owner->LockAGLCntx();
- _stretchedHeight = stretchHeight;
- _stretchedWidth = stretchWidth;
- _owner->UnlockAGLCntx();
- return 0;
-}
-
-int VideoChannelAGL::FrameSizeChange(int width, int height, int numberOfStreams)
-{
- // We'll get a new frame size from VideoAPI, prepare the buffer
-
- _owner->LockAGLCntx();
-
- if (width == _width && _height == height)
- {
- // We already have a correct buffer size
- _numberOfStreams = numberOfStreams;
- _owner->UnlockAGLCntx();
- return 0;
- }
-
- _width = width;
- _height = height;
-
- // Delete the old buffer, create a new one with correct size.
- if (_buffer)
- {
- delete [] _buffer;
- _bufferSize = 0;
- }
-
- _incommingBufferSize = CalcBufferSize(kI420, _width, _height);
- _bufferSize = CalcBufferSize(kARGB, _width, _height);//_width * _height * bytesPerPixel;
- _buffer = new unsigned char [_bufferSize];
- memset(_buffer, 0, _bufferSize * sizeof(unsigned char));
-
- if (aglSetCurrentContext(_aglContext) == false)
- {
- _owner->UnlockAGLCntx();
- return -1;
- }
-
- // Delete a possible old texture
- if (_texture != 0)
- {
- glDeleteTextures(1, (const GLuint*) &_texture);
- _texture = 0;
- }
-
- // Create a new texture
- glGenTextures(1, (GLuint *) &_texture);
-
- GLenum glErr = glGetError();
-
- if (glErr != GL_NO_ERROR)
- {
- }
-
- // Do the setup for both textures
- // Note: we setup two textures even if we're not running full screen
- glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture);
-
- // Set texture parameters
- glTexParameterf(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_PRIORITY, 1.0);
-
- glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
- glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
-
- glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
- glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MIN_FILTER, GL_LINEAR);
- //glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MAG_FILTER, GL_NEAREST);
- //glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MIN_FILTER, GL_NEAREST);
-
- glTexEnvf(GL_TEXTURE_ENV, GL_TEXTURE_ENV_MODE, GL_MODULATE);
-
- glPixelStorei(GL_UNPACK_ALIGNMENT, 1);
-
- glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_STORAGE_HINT_APPLE, GL_STORAGE_SHARED_APPLE);
-
- // Maximum width/height for a texture
- GLint texSize;
- glGetIntegerv(GL_MAX_TEXTURE_SIZE, &texSize);
-
- if (texSize < _width || texSize < _height)
- {
- // Image too big for memory
- _owner->UnlockAGLCntx();
- return -1;
- }
-
- // Set up th texture type and size
- glTexImage2D(GL_TEXTURE_RECTANGLE_EXT, // target
- 0, // level
- GL_RGBA, // internal format
- _width, // width
- _height, // height
- 0, // border 0/1 = off/on
- _pixelFormat, // format, GL_BGRA
- _pixelDataType, // data type, GL_UNSIGNED_INT_8_8_8_8
- _buffer); // pixel data
-
- glErr = glGetError();
- if (glErr != GL_NO_ERROR)
- {
- _owner->UnlockAGLCntx();
- return -1;
- }
-
- _owner->UnlockAGLCntx();
- return 0;
-}
-
-// Called from video engine when a new frame should be rendered.
-int VideoChannelAGL::DeliverFrame(unsigned char* buffer, int bufferSize, unsigned int /*timeStamp90kHz*/)
-{
- _owner->LockAGLCntx();
-
- if (_texture == 0)
- {
- _owner->UnlockAGLCntx();
- return 0;
- }
-
- if (bufferSize != _incommingBufferSize)
- {
- _owner->UnlockAGLCntx();
- return -1;
- }
-
- // Setting stride = width.
- int rgbret = ConvertFromYV12(buffer, _width, kBGRA, 0, _width, _height,
- _buffer);
- if (rgbret < 0)
- {
- _owner->UnlockAGLCntx();
- return -1;
- }
-
- aglSetCurrentContext(_aglContext);
-
- // Put the new frame into the graphic card texture.
- glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture); // Make sure this texture is the active one
- GLenum glErr = glGetError();
- if (glErr != GL_NO_ERROR)
- {
- _owner->UnlockAGLCntx();
- return -1;
- }
-
- // Copy buffer to texture
- glTexSubImage2D(GL_TEXTURE_RECTANGLE_EXT,
- 0, // Level, not use
- 0, // start point x, (low left of pic)
- 0, // start point y,
- _width, // width
- _height, // height
- _pixelFormat, // pictue format for _buffer
- _pixelDataType, // data type of _buffer
- (const GLvoid*) _buffer); // the pixel data
-
- if (glGetError() != GL_NO_ERROR)
- {
- _owner->UnlockAGLCntx();
- return -1;
- }
-
- _bufferIsUpdated = true;
- _owner->UnlockAGLCntx();
-
- return 0;
-}
-
-int VideoChannelAGL::RenderOffScreenBuffer()
-{
-
- _owner->LockAGLCntx();
-
- if (_texture == 0)
- {
- _owner->UnlockAGLCntx();
- return 0;
- }
-
- GLfloat xStart = 2.0f * _startWidth - 1.0f;
- GLfloat xStop = 2.0f * _stopWidth - 1.0f;
- GLfloat yStart = 1.0f - 2.0f * _stopHeight;
- GLfloat yStop = 1.0f - 2.0f * _startHeight;
-
- aglSetCurrentContext(_aglContext);
- glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture);
-
- if(_stretchedWidth != _oldStretchedWidth || _stretchedHeight != _oldStretchedHeight)
- {
- glViewport(0, 0, _stretchedWidth, _stretchedHeight);
- }
- _oldStretchedHeight = _stretchedHeight;
- _oldStretchedWidth = _stretchedWidth;
-
- // Now really put the texture into the framebuffer
- glLoadIdentity();
-
- glEnable(GL_TEXTURE_RECTANGLE_EXT);
-
- glBegin(GL_POLYGON);
- {
- glTexCoord2f(0.0, 0.0); glVertex2f(xStart, yStop);
- glTexCoord2f(_width, 0.0); glVertex2f(xStop, yStop);
- glTexCoord2f(_width, _height); glVertex2f(xStop, yStart);
- glTexCoord2f(0.0, _height); glVertex2f(xStart, yStart);
- }
- glEnd();
-
- glDisable(GL_TEXTURE_RECTANGLE_EXT);
-
- _bufferIsUpdated = false;
-
- _owner->UnlockAGLCntx();
- return 0;
-}
-
-int VideoChannelAGL::IsUpdated(bool& isUpdated)
-{
- _owner->LockAGLCntx();
- isUpdated = _bufferIsUpdated;
- _owner->UnlockAGLCntx();
-
- return 0;
-}
-
-int VideoChannelAGL::SetStreamSettings(int /*streamId*/, float startWidth, float startHeight, float stopWidth, float stopHeight)
-{
-
- _owner->LockAGLCntx();
-
- _startWidth = startWidth;
- _stopWidth = stopWidth;
- _startHeight = startHeight;
- _stopHeight = stopHeight;
-
- int oldWidth = _width;
- int oldHeight = _height;
- int oldNumberOfStreams = _numberOfStreams;
-
- _width = 0;
- _height = 0;
-
- int retVal = FrameSizeChange(oldWidth, oldHeight, oldNumberOfStreams);
-
- _owner->UnlockAGLCntx();
-
- return retVal;
-}
-
-int VideoChannelAGL::SetStreamCropSettings(int /*streamId*/, float /*startWidth*/, float /*startHeight*/, float /*stopWidth*/, float /*stopHeight*/)
-{
- return -1;
-}
-
-#pragma mark VideoRenderAGL WindowRef constructor
-
-VideoRenderAGL::VideoRenderAGL(WindowRef windowRef, bool fullscreen, int iId) :
-_hiviewRef( 0),
-_windowRef( windowRef),
-_fullScreen( fullscreen),
-_id( iId),
-_renderCritSec(*CriticalSectionWrapper::CreateCriticalSection()),
-_screenUpdateThread( 0),
-_screenUpdateEvent( 0),
-_isHIViewRef( false),
-_aglContext( 0),
-_windowWidth( 0),
-_windowHeight( 0),
-_lastWindowWidth( -1),
-_lastWindowHeight( -1),
-_lastHiViewWidth( -1),
-_lastHiViewHeight( -1),
-_currentParentWindowHeight( 0),
-_currentParentWindowWidth( 0),
-_currentParentWindowBounds( ),
-_windowHasResized( false),
-_lastParentWindowBounds( ),
-_currentHIViewBounds( ),
-_lastHIViewBounds( ),
-_windowRect( ),
-_aglChannels( ),
-_zOrderToChannel( ),
-_hiviewEventHandlerRef( NULL),
-_windowEventHandlerRef( NULL),
-_currentViewBounds( ),
-_lastViewBounds( ),
-_renderingIsPaused( false),
-_threadID( )
-
-{
- //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s");
-
- _screenUpdateThread = ThreadWrapper::CreateThread(ScreenUpdateThreadProc, this, kRealtimePriority);
- _screenUpdateEvent = EventWrapper::Create();
-
- if(!IsValidWindowPtr(_windowRef))
- {
- //WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Invalid WindowRef:0x%x", __FUNCTION__, __LINE__, _windowRef);
- }
- else
- {
- //WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s:%d WindowRef 0x%x is valid", __FUNCTION__, __LINE__, _windowRef);
- }
-
- GetWindowRect(_windowRect);
-
- _lastViewBounds.origin.x = 0;
- _lastViewBounds.origin.y = 0;
- _lastViewBounds.size.width = 0;
- _lastViewBounds.size.height = 0;
-
-}
-
-// this is a static function. It has been registered (in class constructor) to be called on various window redrawing or resizing.
-// Since it is a static method, I have passed in "this" as the userData (one and only allowed) parameter, then calling member methods on it.
-#pragma mark WindowRef Event Handler
-pascal OSStatus VideoRenderAGL::sHandleWindowResized (EventHandlerCallRef /*nextHandler*/,
- EventRef theEvent,
- void* userData)
-{
- WindowRef windowRef = NULL;
-
- int eventType = GetEventKind(theEvent);
-
- // see https://dcs.sourcerepo.com/dcs/tox_view/trunk/tox/libraries/i686-win32/include/quicktime/CarbonEvents.h for a list of codes
- GetEventParameter (theEvent,
- kEventParamDirectObject,
- typeWindowRef,
- NULL,
- sizeof (WindowRef),
- NULL,
- &windowRef);
-
- VideoRenderAGL* obj = (VideoRenderAGL*)(userData);
-
- bool updateUI = true;
- if(kEventWindowBoundsChanged == eventType)
- {
- }
- else if(kEventWindowBoundsChanging == eventType)
- {
- }
- else if(kEventWindowZoomed == eventType)
- {
- }
- else if(kEventWindowExpanding == eventType)
- {
- }
- else if(kEventWindowExpanded == eventType)
- {
- }
- else if(kEventWindowClickResizeRgn == eventType)
- {
- }
- else if(kEventWindowClickDragRgn == eventType)
- {
- }
- else
- {
- updateUI = false;
- }
-
- if(true == updateUI)
- {
- obj->ParentWindowResized(windowRef);
- obj->UpdateClipping();
- obj->RenderOffScreenBuffers();
- }
-
- return noErr;
-}
-
-#pragma mark VideoRenderAGL HIViewRef constructor
-
-VideoRenderAGL::VideoRenderAGL(HIViewRef windowRef, bool fullscreen, int iId) :
-_hiviewRef( windowRef),
-_windowRef( 0),
-_fullScreen( fullscreen),
-_id( iId),
-_renderCritSec(*CriticalSectionWrapper::CreateCriticalSection()),
-_screenUpdateThread( 0),
-_screenUpdateEvent( 0),
-_isHIViewRef( false),
-_aglContext( 0),
-_windowWidth( 0),
-_windowHeight( 0),
-_lastWindowWidth( -1),
-_lastWindowHeight( -1),
-_lastHiViewWidth( -1),
-_lastHiViewHeight( -1),
-_currentParentWindowHeight( 0),
-_currentParentWindowWidth( 0),
-_currentParentWindowBounds( ),
-_windowHasResized( false),
-_lastParentWindowBounds( ),
-_currentHIViewBounds( ),
-_lastHIViewBounds( ),
-_windowRect( ),
-_aglChannels( ),
-_zOrderToChannel( ),
-_hiviewEventHandlerRef( NULL),
-_windowEventHandlerRef( NULL),
-_currentViewBounds( ),
-_lastViewBounds( ),
-_renderingIsPaused( false),
-_threadID( )
-{
- //WEBRTC_TRACE(kTraceDebug, "%s:%d Constructor", __FUNCTION__, __LINE__);
- // _renderCritSec = CriticalSectionWrapper::CreateCriticalSection();
-
- _screenUpdateThread = ThreadWrapper::CreateThread(ScreenUpdateThreadProc, this, kRealtimePriority);
- _screenUpdateEvent = EventWrapper::Create();
-
- GetWindowRect(_windowRect);
-
- _lastViewBounds.origin.x = 0;
- _lastViewBounds.origin.y = 0;
- _lastViewBounds.size.width = 0;
- _lastViewBounds.size.height = 0;
-
-#ifdef NEW_HIVIEW_PARENT_EVENT_HANDLER
- // This gets the parent window of the HIViewRef that's passed in and installs a WindowRef event handler on it
- // The event handler looks for window resize events and adjusts the offset of the controls.
-
- //WEBRTC_TRACE(kTraceDebug, "%s:%d Installing Eventhandler for hiviewRef's parent window", __FUNCTION__, __LINE__);
-
-
- static const EventTypeSpec windowEventTypes[] =
- {
- kEventClassWindow, kEventWindowBoundsChanged,
- kEventClassWindow, kEventWindowBoundsChanging,
- kEventClassWindow, kEventWindowZoomed,
- kEventClassWindow, kEventWindowExpanded,
- kEventClassWindow, kEventWindowClickResizeRgn,
- kEventClassWindow, kEventWindowClickDragRgn
- };
-
- WindowRef parentWindow = HIViewGetWindow(windowRef);
-
- InstallWindowEventHandler (parentWindow,
- NewEventHandlerUPP (sHandleWindowResized),
- GetEventTypeCount(windowEventTypes),
- windowEventTypes,
- (void *) this, // this is an arbitrary parameter that will be passed on to your event handler when it is called later
- &_windowEventHandlerRef);
-
-#endif
-
-#ifdef NEW_HIVIEW_EVENT_HANDLER
- //WEBRTC_TRACE(kTraceDebug, "%s:%d Installing Eventhandler for hiviewRef", __FUNCTION__, __LINE__);
-
- static const EventTypeSpec hiviewEventTypes[] =
- {
- kEventClassControl, kEventControlBoundsChanged,
- kEventClassControl, kEventControlDraw
- // kEventControlDragLeave
- // kEventControlDragReceive
- // kEventControlGetFocusPart
- // kEventControlApplyBackground
- // kEventControlDraw
- // kEventControlHit
-
- };
-
- HIViewInstallEventHandler(_hiviewRef,
- NewEventHandlerUPP(sHandleHiViewResized),
- GetEventTypeCount(hiviewEventTypes),
- hiviewEventTypes,
- (void *) this,
- &_hiviewEventHandlerRef);
-
-#endif
-}
-
-// this is a static function. It has been registered (in constructor) to be called on various window redrawing or resizing.
-// Since it is a static method, I have passed in "this" as the userData (one and only allowed) parameter, then calling member methods on it.
-#pragma mark HIViewRef Event Handler
-pascal OSStatus VideoRenderAGL::sHandleHiViewResized (EventHandlerCallRef nextHandler, EventRef theEvent, void* userData)
-{
- //static int callbackCounter = 1;
- HIViewRef hiviewRef = NULL;
-
- // see https://dcs.sourcerepo.com/dcs/tox_view/trunk/tox/libraries/i686-win32/include/quicktime/CarbonEvents.h for a list of codes
- int eventType = GetEventKind(theEvent);
- OSStatus status = noErr;
- status = GetEventParameter (theEvent,
- kEventParamDirectObject,
- typeControlRef,
- NULL,
- sizeof (ControlRef),
- NULL,
- &hiviewRef);
-
- VideoRenderAGL* obj = (VideoRenderAGL*)(userData);
- WindowRef parentWindow = HIViewGetWindow(hiviewRef);
- bool updateUI = true;
-
- if(kEventControlBoundsChanged == eventType)
- {
- }
- else if(kEventControlDraw == eventType)
- {
- }
- else
- {
- updateUI = false;
- }
-
- if(true == updateUI)
- {
- obj->ParentWindowResized(parentWindow);
- obj->UpdateClipping();
- obj->RenderOffScreenBuffers();
- }
-
- return status;
-}
-
-VideoRenderAGL::~VideoRenderAGL()
-{
-
- //WEBRTC_TRACE(kTraceDebug, "%s:%d Destructor", __FUNCTION__, __LINE__);
-
-
-#ifdef USE_EVENT_HANDLERS
- // remove event handlers
- OSStatus status;
- if(_isHIViewRef)
- {
- status = RemoveEventHandler(_hiviewEventHandlerRef);
- }
- else
- {
- status = RemoveEventHandler(_windowEventHandlerRef);
- }
- if(noErr != status)
- {
- if(_isHIViewRef)
- {
-
- //WEBRTC_TRACE(kTraceDebug, "%s:%d Failed to remove hiview event handler: %d", __FUNCTION__, __LINE__, (int)_hiviewEventHandlerRef);
- }
- else
- {
- //WEBRTC_TRACE(kTraceDebug, "%s:%d Failed to remove window event handler %d", __FUNCTION__, __LINE__, (int)_windowEventHandlerRef);
- }
- }
-
-#endif
-
- OSStatus status;
-#ifdef NEW_HIVIEW_PARENT_EVENT_HANDLER
- if(_windowEventHandlerRef)
- {
- status = RemoveEventHandler(_windowEventHandlerRef);
- if(status != noErr)
- {
- //WEBRTC_TRACE(kTraceDebug, "%s:%d failed to remove window event handler %d", __FUNCTION__, __LINE__, (int)_windowEventHandlerRef);
- }
- }
-#endif
-
-#ifdef NEW_HIVIEW_EVENT_HANDLER
- if(_hiviewEventHandlerRef)
- {
- status = RemoveEventHandler(_hiviewEventHandlerRef);
- if(status != noErr)
- {
- //WEBRTC_TRACE(kTraceDebug, "%s:%d Failed to remove hiview event handler: %d", __FUNCTION__, __LINE__, (int)_hiviewEventHandlerRef);
- }
- }
-#endif
-
- // Signal event to exit thread, then delete it
- ThreadWrapper* tmpPtr = _screenUpdateThread;
- _screenUpdateThread = NULL;
-
- if (tmpPtr)
- {
- tmpPtr->SetNotAlive();
- _screenUpdateEvent->Set();
- _screenUpdateEvent->StopTimer();
-
- if (tmpPtr->Stop())
- {
- delete tmpPtr;
- }
- delete _screenUpdateEvent;
- _screenUpdateEvent = NULL;
- }
-
- if (_aglContext != 0)
- {
- aglSetCurrentContext(_aglContext);
- aglDestroyContext(_aglContext);
- _aglContext = 0;
- }
-
- // Delete all channels
- std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.begin();
- while (it!= _aglChannels.end())
- {
- delete it->second;
- _aglChannels.erase(it);
- it = _aglChannels.begin();
- }
- _aglChannels.clear();
-
- // Clean the zOrder map
- std::multimap<int, int>::iterator zIt = _zOrderToChannel.begin();
- while(zIt != _zOrderToChannel.end())
- {
- _zOrderToChannel.erase(zIt);
- zIt = _zOrderToChannel.begin();
- }
- _zOrderToChannel.clear();
-
- //delete _renderCritSec;
-
-
-}
-
-int VideoRenderAGL::GetOpenGLVersion(int& aglMajor, int& aglMinor)
-{
- aglGetVersion((GLint *) &aglMajor, (GLint *) &aglMinor);
- return 0;
-}
-
-int VideoRenderAGL::Init()
-{
- LockAGLCntx();
-
- // Start rendering thread...
- if (!_screenUpdateThread)
- {
- UnlockAGLCntx();
- //WEBRTC_TRACE(kTraceError, "%s:%d Thread not created", __FUNCTION__, __LINE__);
- return -1;
- }
- unsigned int threadId;
- _screenUpdateThread->Start(threadId);
-
- // Start the event triggering the render process
- unsigned int monitorFreq = 60;
- _screenUpdateEvent->StartTimer(true, 1000/monitorFreq);
-
- // Create mixing textures
- if (CreateMixingContext() == -1)
- {
- //WEBRTC_TRACE(kTraceError, "%s:%d Could not create a mixing context", __FUNCTION__, __LINE__);
- UnlockAGLCntx();
- return -1;
- }
-
- UnlockAGLCntx();
- return 0;
-}
-
-VideoChannelAGL* VideoRenderAGL::CreateAGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight)
-{
-
- LockAGLCntx();
-
- //WEBRTC_TRACE(kTraceInfo, "%s:%d Creating AGL channel: %d", __FUNCTION__, __LINE__, channel);
-
- if (HasChannel(channel))
- {
- //WEBRTC_TRACE(kTraceError, "%s:%d Channel already exists", __FUNCTION__, __LINE__);
- UnlockAGLCntx();k
- return NULL;
- }
-
- if (_zOrderToChannel.find(zOrder) != _zOrderToChannel.end())
- {
- // There are already one channel using this zOrder
- // TODO: Allow multiple channels with same zOrder
- }
-
- VideoChannelAGL* newAGLChannel = new VideoChannelAGL(_aglContext, _id, this);
-
- if (newAGLChannel->SetStreamSettings(0, startWidth, startHeight, stopWidth, stopHeight) == -1)
- {
- if (newAGLChannel)
- {
- delete newAGLChannel;
- newAGLChannel = NULL;
- }
- //WEBRTC_LOG(kTraceError, "Could not create AGL channel");
- //WEBRTC_TRACE(kTraceError, "%s:%d Could not create AGL channel", __FUNCTION__, __LINE__);
- UnlockAGLCntx();
- return NULL;
- }
-k
- _aglChannels[channel] = newAGLChannel;
- _zOrderToChannel.insert(std::pair<int, int>(zOrder, channel));
-
- UnlockAGLCntx();
- return newAGLChannel;
-}
-
-int VideoRenderAGL::DeleteAllAGLChannels()
-{
- CriticalSectionScoped cs(_renderCritSec);
-
- //WEBRTC_TRACE(kTraceInfo, "%s:%d Deleting all AGL channels", __FUNCTION__, __LINE__);
- //int i = 0 ;
- std::map<int, VideoChannelAGL*>::iterator it;
- it = _aglChannels.begin();
-
- while (it != _aglChannels.end())
- {
- VideoChannelAGL* channel = it->second;
- if (channel)
- delete channel;
-
- _aglChannels.erase(it);
- it = _aglChannels.begin();
- }
- _aglChannels.clear();
- return 0;
-}
-
-int VideoRenderAGL::DeleteAGLChannel(int channel)
-{
- CriticalSectionScoped cs(_renderCritSec);
- //WEBRTC_TRACE(kTraceDebug, "%s:%d Deleting AGL channel %d", __FUNCTION__, __LINE__, channel);
-
- std::map<int, VideoChannelAGL*>::iterator it;
- it = _aglChannels.find(channel);
- if (it != _aglChannels.end())
- {
- delete it->second;
- _aglChannels.erase(it);
- }
- else
- {
- //WEBRTC_TRACE(kTraceWarning, "%s:%d Channel not found", __FUNCTION__, __LINE__);
- return -1;
- }
-
- std::multimap<int, int>::iterator zIt = _zOrderToChannel.begin();
- while( zIt != _zOrderToChannel.end())
- {
- if (zIt->second == channel)
- {
- _zOrderToChannel.erase(zIt);
- break;
- }
- zIt++;// = _zOrderToChannel.begin();
- }
-
- return 0;
-}
-
-int VideoRenderAGL::StopThread()
-{
- CriticalSectionScoped cs(_renderCritSec);
- ThreadWrapper* tmpPtr = _screenUpdateThread;
- //_screenUpdateThread = NULL;
-
- if (tmpPtr)
- {
- tmpPtr->SetNotAlive();
- _screenUpdateEvent->Set();
- if (tmpPtr->Stop())
- {
- delete tmpPtr;
- }
- }
-
- delete _screenUpdateEvent;
- _screenUpdateEvent = NULL;
-
- return 0;
-}
-
-bool VideoRenderAGL::IsFullScreen()
-{
- CriticalSectionScoped cs(_renderCritSec);
- return _fullScreen;
-}
-
-bool VideoRenderAGL::HasChannels()
-{
-
- CriticalSectionScoped cs(_renderCritSec);
-
- if (_aglChannels.begin() != _aglChannels.end())
- {
- return true;
- }
-
- return false;
-}
-
-bool VideoRenderAGL::HasChannel(int channel)
-{
- CriticalSectionScoped cs(_renderCritSec);
-
- std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.find(channel);
- if (it != _aglChannels.end())
- {
- return true;
- }
-
- return false;
-}
-
-int VideoRenderAGL::GetChannels(std::list<int>& channelList)
-{
-
- CriticalSectionScoped cs(_renderCritSec);
- std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.begin();
-
- while (it != _aglChannels.end())
- {
- channelList.push_back(it->first);
- it++;
- }
-
- return 0;
-}
-
-VideoChannelAGL* VideoRenderAGL::ConfigureAGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight)
-{
-
- CriticalSectionScoped cs(_renderCritSec);
-
- std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.find(channel);
-
- if (it != _aglChannels.end())
- {
- VideoChannelAGL* aglChannel = it->second;
- if (aglChannel->SetStreamSettings(0, startWidth, startHeight, stopWidth, stopHeight) == -1)
- {
- return NULL;
- }
-
- std::multimap<int, int>::iterator it = _zOrderToChannel.begin();
- while(it != _zOrderToChannel.end())
- {
- if (it->second == channel)
- {
- if (it->first != zOrder)
- {
- _zOrderToChannel.erase(it);
- _zOrderToChannel.insert(std::pair<int, int>(zOrder, channel));
- }
- break;
- }
- it++;
- }
- return aglChannel;
- }
-
- return NULL;
-}
-
-bool VideoRenderAGL::ScreenUpdateThreadProc(void* obj)
-{
- return static_cast<VideoRenderAGL*>(obj)->ScreenUpdateProcess();
-}
-
-bool VideoRenderAGL::ScreenUpdateProcess()
-{
- _screenUpdateEvent->Wait(100);
-
- LockAGLCntx();
-
- if (!_screenUpdateThread)
- {
- UnlockAGLCntx();
- return false;
- }
-
- if (aglSetCurrentContext(_aglContext) == GL_FALSE)
- {
- UnlockAGLCntx();
- return true;
- }
-
- if (GetWindowRect(_windowRect) == -1)
- {
- UnlockAGLCntx();
- return true;
- }
-
- if (_windowWidth != (_windowRect.right - _windowRect.left)
- || _windowHeight != (_windowRect.bottom - _windowRect.top))
- {
- // We have a new window size, update the context.
- if (aglUpdateContext(_aglContext) == GL_FALSE)
- {
- UnlockAGLCntx();
- return true;
- }
- _windowWidth = _windowRect.right - _windowRect.left;
- _windowHeight = _windowRect.bottom - _windowRect.top;
- }
-
- // this section will poll to see if the window size has changed
- // this is causing problem w/invalid windowRef
- // this code has been modified and exists now in the window event handler
-#ifndef NEW_HIVIEW_PARENT_EVENT_HANDLER
- if (_isHIViewRef)
- {
-
- if(FALSE == HIViewIsValid(_hiviewRef))
- {
-
- //WEBRTC_TRACE(kTraceDebug, "%s:%d Invalid windowRef", __FUNCTION__, __LINE__);
- UnlockAGLCntx();
- return true;
- }
- WindowRef window = HIViewGetWindow(_hiviewRef);
-
- if(FALSE == IsValidWindowPtr(window))
- {
- //WEBRTC_TRACE(kTraceDebug, "%s:%d Invalide hiviewRef", __FUNCTION__, __LINE__);
- UnlockAGLCntx();
- return true;
- }
- if (window == NULL)
- {
- //WEBRTC_TRACE(kTraceDebug, "%s:%d WindowRef = NULL", __FUNCTION__, __LINE__);
- UnlockAGLCntx();
- return true;
- }
-
- if(FALSE == MacIsWindowVisible(window))
- {
- //WEBRTC_TRACE(kTraceDebug, "%s:%d MacIsWindowVisible == FALSE. Returning early", __FUNCTION__, __LINE__);
- UnlockAGLCntx();
- return true;
- }
-
- HIRect viewBounds; // Placement and size for HIView
- int windowWidth = 0; // Parent window width
- int windowHeight = 0; // Parent window height
-
- // NOTE: Calling GetWindowBounds with kWindowStructureRgn will crash intermittentaly if the OS decides it needs to push it into the back for a moment.
- // To counter this, we get the titlebar height on class construction and then add it to the content region here. Content regions seems not to crash
- Rect contentBounds =
- { 0, 0, 0, 0}; // The bounds for the parent window
-
-#if defined(USE_CONTENT_RGN)
- GetWindowBounds(window, kWindowContentRgn, &contentBounds);
-#elif defined(USE_STRUCT_RGN)
- GetWindowBounds(window, kWindowStructureRgn, &contentBounds);
-#endif
-
- Rect globalBounds =
- { 0, 0, 0, 0}; // The bounds for the parent window
- globalBounds.top = contentBounds.top;
- globalBounds.right = contentBounds.right;
- globalBounds.bottom = contentBounds.bottom;
- globalBounds.left = contentBounds.left;
-
- windowHeight = globalBounds.bottom - globalBounds.top;
- windowWidth = globalBounds.right - globalBounds.left;
-
- // Get the size of the HIViewRef
- HIViewGetBounds(_hiviewRef, &viewBounds);
- HIViewConvertRect(&viewBounds, _hiviewRef, NULL);
-
- // Check if this is the first call..
- if (_lastWindowHeight == -1 &&
- _lastWindowWidth == -1)
- {
- _lastWindowWidth = windowWidth;
- _lastWindowHeight = windowHeight;
-
- _lastViewBounds.origin.x = viewBounds.origin.x;
- _lastViewBounds.origin.y = viewBounds.origin.y;
- _lastViewBounds.size.width = viewBounds.size.width;
- _lastViewBounds.size.height = viewBounds.size.height;
- }
- sfasdfasdf
-
- bool resized = false;
-
- // Check if parent window size has changed
- if (windowHeight != _lastWindowHeight ||
- windowWidth != _lastWindowWidth)
- {
- resized = true;
- }
-
- // Check if the HIView has new size or is moved in the parent window
- if (_lastViewBounds.origin.x != viewBounds.origin.x ||
- _lastViewBounds.origin.y != viewBounds.origin.y ||
- _lastViewBounds.size.width != viewBounds.size.width ||
- _lastViewBounds.size.height != viewBounds.size.height)
- {
- // The HiView is resized or has moved.
- resized = true;
- }
-
- if (resized)
- {
-
- //WEBRTC_TRACE(kTraceDebug, "%s:%d Window has resized", __FUNCTION__, __LINE__);
-
- // Calculate offset between the windows
- // {x, y, widht, height}, x,y = lower left corner
- const GLint offs[4] =
- { (int)(0.5f + viewBounds.origin.x),
- (int)(0.5f + windowHeight - (viewBounds.origin.y + viewBounds.size.height)),
- viewBounds.size.width, viewBounds.size.height};
-
- //WEBRTC_TRACE(kTraceDebug, "%s:%d contentBounds t:%d r:%d b:%d l:%d", __FUNCTION__, __LINE__,
- contentBounds.top, contentBounds.right, contentBounds.bottom, contentBounds.left);
- //WEBRTC_TRACE(kTraceDebug, "%s:%d windowHeight=%d", __FUNCTION__, __LINE__, windowHeight);
- //WEBRTC_TRACE(kTraceDebug, "%s:%d offs[4] = %d, %d, %d, %d", __FUNCTION__, __LINE__, offs[0], offs[1], offs[2], offs[3]);
-
- aglSetDrawable (_aglContext, GetWindowPort(window));
- aglSetInteger(_aglContext, AGL_BUFFER_RECT, offs);
- aglEnable(_aglContext, AGL_BUFFER_RECT);
-
- // We need to change the viewport too if the HIView size has changed
- glViewport(0.0f, 0.0f, (GLsizei) viewBounds.size.width, (GLsizei) viewBounds.size.height);
-
- }
- _lastWindowWidth = windowWidth;
- _lastWindowHeight = windowHeight;
-
- _lastViewBounds.origin.x = viewBounds.origin.x;
- _lastViewBounds.origin.y = viewBounds.origin.y;
- _lastViewBounds.size.width = viewBounds.size.width;
- _lastViewBounds.size.height = viewBounds.size.height;
-
- }
-#endif
- if (_fullScreen)
- {
- // TODO
- // We use double buffers, must always update
- //RenderOffScreenBuffersToBackBuffer();
- }
- else
- {
- // Check if there are any updated buffers
- bool updated = false;
-
- // TODO: check if window size is updated!
- // TODO Improvement: Walk through the zOrder Map to only render the ones in need of update
- std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.begin();
- while (it != _aglChannels.end())
- {
-
- VideoChannelAGL* aglChannel = it->second;
- aglChannel->UpdateStretchSize(_windowHeight, _windowWidth);
- aglChannel->IsUpdated(updated);
- if (updated)
- {
- break;
- }
- it++;
- }
-
- if (updated)
- {
- // At least on buffers is updated, we need to repaint the texture
- if (RenderOffScreenBuffers() != -1)
- {
- // MF
- //SwapAndDisplayBuffers();
- }
- else
- {
- // Error updating the mixing texture, don't swap.
- }
- }
- }
-
- UnlockAGLCntx();
-
- //WEBRTC_LOG(kTraceDebug, "Leaving ScreenUpdateProcess()");
- return true;
-}
-
-void VideoRenderAGL::ParentWindowResized(WindowRef window)
-{
- //WEBRTC_LOG(kTraceDebug, "%s HIViewRef:%d owner window has resized", __FUNCTION__, (int)_hiviewRef);
-
- LockAGLCntx();
-k
- // set flag
- _windowHasResized = false;
-
- if(FALSE == HIViewIsValid(_hiviewRef))
- {
- //WEBRTC_LOG(kTraceDebug, "invalid windowRef");
- UnlockAGLCntx();
- return;
- }
-
- if(FALSE == IsValidWindowPtr(window))
- {
- //WEBRTC_LOG(kTraceError, "invalid windowRef");
- UnlockAGLCntx();
- return;
- }
-
- if (window == NULL)
- {
- //WEBRTC_LOG(kTraceError, "windowRef = NULL");
- UnlockAGLCntx();
- return;
- }
-
- if(FALSE == MacIsWindowVisible(window))
- {
- //WEBRTC_LOG(kTraceDebug, "MacIsWindowVisible = FALSE. Returning early.");
- UnlockAGLCntx();
- return;
- }
-
- Rect contentBounds =
- { 0, 0, 0, 0};
-
-#if defined(USE_CONTENT_RGN)
- GetWindowBounds(window, kWindowContentRgn, &contentBounds);
-#elif defined(USE_STRUCT_RGN)
- GetWindowBounds(window, kWindowStructureRgn, &contentBounds);
-#endif
-
- //WEBRTC_LOG(kTraceDebug, "%s contentBounds t:%d r:%d b:%d l:%d", __FUNCTION__, contentBounds.top, contentBounds.right, contentBounds.bottom, contentBounds.left);
-
- // update global vars
- _currentParentWindowBounds.top = contentBounds.top;
- _currentParentWindowBounds.left = contentBounds.left;
- _currentParentWindowBounds.bottom = contentBounds.bottom;
- _currentParentWindowBounds.right = contentBounds.right;
-
- _currentParentWindowWidth = _currentParentWindowBounds.right - _currentParentWindowBounds.left;
- _currentParentWindowHeight = _currentParentWindowBounds.bottom - _currentParentWindowBounds.top;
-
- _windowHasResized = true;
-
- // ********* update AGL offsets
- HIRect viewBounds;
- HIViewGetBounds(_hiviewRef, &viewBounds);
- HIViewConvertRect(&viewBounds, _hiviewRef, NULL);
-
- const GLint offs[4] =
- { (int)(0.5f + viewBounds.origin.x),
- (int)(0.5f + _currentParentWindowHeight - (viewBounds.origin.y + viewBounds.size.height)),
- viewBounds.size.width, viewBounds.size.height};
- //WEBRTC_LOG(kTraceDebug, "%s _currentParentWindowHeight=%d", __FUNCTION__, _currentParentWindowHeight);
- //WEBRTC_LOG(kTraceDebug, "%s offs[4] = %d, %d, %d, %d", __FUNCTION__, offs[0], offs[1], offs[2], offs[3]);
-
- aglSetCurrentContext(_aglContext);
- aglSetDrawable (_aglContext, GetWindowPort(window));
- aglSetInteger(_aglContext, AGL_BUFFER_RECT, offs);
- aglEnable(_aglContext, AGL_BUFFER_RECT);
-
- // We need to change the viewport too if the HIView size has changed
- glViewport(0.0f, 0.0f, (GLsizei) viewBounds.size.width, (GLsizei) viewBounds.size.height);
-
- UnlockAGLCntx();
-
- return;
-}
-
-int VideoRenderAGL::CreateMixingContext()
-{
-
- LockAGLCntx();
-
- //WEBRTC_LOG(kTraceDebug, "Entering CreateMixingContext()");
-
- // Use both AGL_ACCELERATED and AGL_NO_RECOVERY to make sure
- // a hardware renderer is used and not a software renderer.
-
- GLint attributes[] =
- {
- AGL_DOUBLEBUFFER,
- AGL_WINDOW,
- AGL_RGBA,
- AGL_NO_RECOVERY,
- AGL_ACCELERATED,
- AGL_RED_SIZE, 8,
- AGL_GREEN_SIZE, 8,
- AGL_BLUE_SIZE, 8,
- AGL_ALPHA_SIZE, 8,
- AGL_DEPTH_SIZE, 24,
- AGL_NONE,
- };
-
- AGLPixelFormat aglPixelFormat;
-
- // ***** Set up the OpenGL Context *****
-
- // Get a pixel format for the attributes above
- aglPixelFormat = aglChoosePixelFormat(NULL, 0, attributes);
- if (NULL == aglPixelFormat)
- {
- //WEBRTC_LOG(kTraceError, "Could not create pixel format");
- UnlockAGLCntx();
- return -1;
- }
-
- // Create an AGL context
- _aglContext = aglCreateContext(aglPixelFormat, NULL);
- if (_aglContext == NULL)
- {
- //WEBRTC_LOG(kTraceError, "Could no create AGL context");
- UnlockAGLCntx();
- return -1;
- }
-
- // Release the pixel format memory
- aglDestroyPixelFormat(aglPixelFormat);
-
- // Set the current AGL context for the rest of the settings
- if (aglSetCurrentContext(_aglContext) == false)
- {
- //WEBRTC_LOG(kTraceError, "Could not set current context: %d", aglGetError());
- UnlockAGLCntx();
- return -1;
- }
-
- if (_isHIViewRef)
- {
- //---------------------------
- // BEGIN: new test code
-#if 0
- // Don't use this one!
- // There seems to be an OS X bug that can't handle
- // movements and resizing of the parent window
- // and or the HIView
- if (aglSetHIViewRef(_aglContext,_hiviewRef) == false)
- {
- //WEBRTC_LOG(kTraceError, "Could not set WindowRef: %d", aglGetError());
- UnlockAGLCntx();
- return -1;
- }
-#else
-
- // Get the parent window for this control
- WindowRef window = GetControlOwner(_hiviewRef);
-
- Rect globalBounds =
- { 0,0,0,0}; // The bounds for the parent window
- HIRect viewBounds; // Placemnt in the parent window and size.
- int windowHeight = 0;
-
- // Rect titleBounds = {0,0,0,0};
- // GetWindowBounds(window, kWindowTitleBarRgn, &titleBounds);
- // _titleBarHeight = titleBounds.top - titleBounds.bottom;
- // if(0 == _titleBarHeight)
- // {
- // //WEBRTC_LOG(kTraceError, "Titlebar height = 0");
- // //return -1;
- // }
-
-
- // Get the bounds for the parent window
-#if defined(USE_CONTENT_RGN)
- GetWindowBounds(window, kWindowContentRgn, &globalBounds);
-#elif defined(USE_STRUCT_RGN)
- GetWindowBounds(window, kWindowStructureRgn, &globalBounds);
-#endif
- windowHeight = globalBounds.bottom - globalBounds.top;
-
- // Get the bounds for the HIView
- HIViewGetBounds(_hiviewRef, &viewBounds);
-
- HIViewConvertRect(&viewBounds, _hiviewRef, NULL);
-
- const GLint offs[4] =
- { (int)(0.5f + viewBounds.origin.x),
- (int)(0.5f + windowHeight - (viewBounds.origin.y + viewBounds.size.height)),
- viewBounds.size.width, viewBounds.size.height};
-
- //WEBRTC_LOG(kTraceDebug, "%s offs[4] = %d, %d, %d, %d", __FUNCTION__, offs[0], offs[1], offs[2], offs[3]);
-
-
- aglSetDrawable (_aglContext, GetWindowPort(window));
- aglSetInteger(_aglContext, AGL_BUFFER_RECT, offs);
- aglEnable(_aglContext, AGL_BUFFER_RECT);
-
- GLint surfaceOrder = 1; // 1: above window, -1 below.
- //OSStatus status = aglSetInteger(_aglContext, AGL_SURFACE_ORDER, &surfaceOrder);
- aglSetInteger(_aglContext, AGL_SURFACE_ORDER, &surfaceOrder);
-
- glViewport(0.0f, 0.0f, (GLsizei) viewBounds.size.width, (GLsizei) viewBounds.size.height);
-#endif
-
- }
- else
- {
- if(GL_FALSE == aglSetDrawable (_aglContext, GetWindowPort(_windowRef)))
- {
- //WEBRTC_LOG(kTraceError, "Could not set WindowRef: %d", aglGetError());
- UnlockAGLCntx();
- return -1;
- }
- }
-
- _windowWidth = _windowRect.right - _windowRect.left;
- _windowHeight = _windowRect.bottom - _windowRect.top;
-
- // opaque surface
- int surfaceOpacity = 1;
- if (aglSetInteger(_aglContext, AGL_SURFACE_OPACITY, (const GLint *) &surfaceOpacity) == false)
- {
- //WEBRTC_LOG(kTraceError, "Could not set surface opacity: %d", aglGetError());
- UnlockAGLCntx();
- return -1;
- }
-
- // 1 -> sync to screen rat, slow...
- //int swapInterval = 0; // 0 don't sync with vertical trace
- int swapInterval = 0; // 1 sync with vertical trace
- if (aglSetInteger(_aglContext, AGL_SWAP_INTERVAL, (const GLint *) &swapInterval) == false)
- {
- //WEBRTC_LOG(kTraceError, "Could not set swap interval: %d", aglGetError());
- UnlockAGLCntx();
- return -1;
- }
-
- // Update the rect with the current size
- if (GetWindowRect(_windowRect) == -1)
- {
- //WEBRTC_LOG(kTraceError, "Could not get window size");
- UnlockAGLCntx();
- return -1;
- }
-
- // Disable not needed functionality to increase performance
- glDisable(GL_DITHER);
- glDisable(GL_ALPHA_TEST);
- glDisable(GL_STENCIL_TEST);
- glDisable(GL_FOG);
- glDisable(GL_TEXTURE_2D);
- glPixelZoom(1.0, 1.0);
-
- glDisable(GL_BLEND);
- glDisable(GL_DEPTH_TEST);
- glDepthMask(GL_FALSE);
- glDisable(GL_CULL_FACE);
-
- glClearColor(0.0f, 0.0f, 0.0f, 1.0f);
- glClear(GL_COLOR_BUFFER_BIT);
-
- GLenum glErr = glGetError();
-
- if (glErr)
- {
- }
-
- UpdateClipping();
-
- //WEBRTC_LOG(kTraceDebug, "Leaving CreateMixingContext()");
-
- UnlockAGLCntx();
- return 0;
-}
-
-int VideoRenderAGL::RenderOffScreenBuffers()
-{
- LockAGLCntx();
-
- // Get the current window size, it might have changed since last render.
- if (GetWindowRect(_windowRect) == -1)
- {
- //WEBRTC_LOG(kTraceError, "Could not get window rect");
- UnlockAGLCntx();
- return -1;
- }
-
- if (aglSetCurrentContext(_aglContext) == false)
- {
- //WEBRTC_LOG(kTraceError, "Could not set current context for rendering");
- UnlockAGLCntx();
- return -1;
- }
-
- // HERE - onl if updated!
- glClear(GL_COLOR_BUFFER_BIT);
-
- // Loop through all channels starting highest zOrder ending with lowest.
- for (std::multimap<int, int>::reverse_iterator rIt = _zOrderToChannel.rbegin();
- rIt != _zOrderToChannel.rend();
- rIt++)
- {
- int channelId = rIt->second;
- std::map<int, VideoChannelAGL*>::iterator it = _aglChannels.find(channelId);
-
- VideoChannelAGL* aglChannel = it->second;
-
- aglChannel->RenderOffScreenBuffer();
- }
-
- SwapAndDisplayBuffers();
-
- UnlockAGLCntx();
- return 0;
-}
-
-int VideoRenderAGL::SwapAndDisplayBuffers()
-{
-
- LockAGLCntx();
- if (_fullScreen)
- {
- // TODO:
- // Swap front and back buffers, rendering taking care of in the same call
- //aglSwapBuffers(_aglContext);
- // Update buffer index to the idx for the next rendering!
- //_textureIdx = (_textureIdx + 1) & 1;
- }
- else
- {
- // Single buffer rendering, only update context.
- glFlush();
- aglSwapBuffers(_aglContext);
- HIViewSetNeedsDisplay(_hiviewRef, true);
- }
-
- UnlockAGLCntx();
- return 0;
-}
-
-int VideoRenderAGL::GetWindowRect(Rect& rect)
-{
-
- LockAGLCntx();
-
- if (_isHIViewRef)
- {
- if (_hiviewRef)
- {
- HIRect HIViewRect1;
- if(FALSE == HIViewIsValid(_hiviewRef))
- {
- rect.top = 0;
- rect.left = 0;
- rect.right = 0;
- rect.bottom = 0;
- //WEBRTC_LOG(kTraceError,"GetWindowRect() HIViewIsValid() returned false");
- UnlockAGLCntx();
- }
- HIViewGetBounds(_hiviewRef,&HIViewRect1);
- HIRectConvert(&HIViewRect1, 1, NULL, 2, NULL);
- if(HIViewRect1.origin.x < 0)
- {
- rect.top = 0;
- //WEBRTC_LOG(kTraceDebug, "GetWindowRect() rect.top = 0");
- }
- else
- {
- rect.top = HIViewRect1.origin.x;
- }
-
- if(HIViewRect1.origin.y < 0)
- {
- rect.left = 0;
- //WEBRTC_LOG(kTraceDebug, "GetWindowRect() rect.left = 0");
- }
- else
- {
- rect.left = HIViewRect1.origin.y;
- }
-
- if(HIViewRect1.size.width < 0)
- {
- rect.right = 0;
- //WEBRTC_LOG(kTraceDebug, "GetWindowRect() rect.right = 0");
- }
- else
- {
- rect.right = HIViewRect1.size.width;
- }
-
- if(HIViewRect1.size.height < 0)
- {
- rect.bottom = 0;
- //WEBRTC_LOG(kTraceDebug, "GetWindowRect() rect.bottom = 0");
- }
- else
- {
- rect.bottom = HIViewRect1.size.height;
- }
-
- ////WEBRTC_LOG(kTraceDebug,"GetWindowRect() HIViewRef: rect.top = %d, rect.left = %d, rect.right = %d, rect.bottom =%d in GetWindowRect", rect.top,rect.left,rect.right,rect.bottom);
- UnlockAGLCntx();
- }
- else
- {
- //WEBRTC_LOG(kTraceError, "invalid HIViewRef");
- UnlockAGLCntx();
- }
- }
- else
- {
- if (_windowRef)
- {
- GetWindowBounds(_windowRef, kWindowContentRgn, &rect);
- UnlockAGLCntx();
- }
- else
- {
- //WEBRTC_LOG(kTraceError, "No WindowRef");
- UnlockAGLCntx();
- }
- }
-}
-
-int VideoRenderAGL::UpdateClipping()
-{
- //WEBRTC_LOG(kTraceDebug, "Entering UpdateClipping()");
- LockAGLCntx();
-
- if(_isHIViewRef)
- {
- if(FALSE == HIViewIsValid(_hiviewRef))
- {
- //WEBRTC_LOG(kTraceError, "UpdateClipping() _isHIViewRef is invalid. Returning -1");
- UnlockAGLCntx();
- return -1;
- }
-
- RgnHandle visibleRgn = NewRgn();
- SetEmptyRgn (visibleRgn);
-
- if(-1 == CalculateVisibleRegion((ControlRef)_hiviewRef, visibleRgn, true))
- {
- }
-
- if(GL_FALSE == aglSetCurrentContext(_aglContext))
- {
- GLenum glErr = aglGetError();
- //WEBRTC_LOG(kTraceError, "aglSetCurrentContext returned FALSE with error code %d at line %d", glErr, __LINE__);
- }
-
- if(GL_FALSE == aglEnable(_aglContext, AGL_CLIP_REGION))
- {
- GLenum glErr = aglGetError();
- //WEBRTC_LOG(kTraceError, "aglEnable returned FALSE with error code %d at line %d\n", glErr, __LINE__);
- }
-
- if(GL_FALSE == aglSetInteger(_aglContext, AGL_CLIP_REGION, (const GLint*)visibleRgn))
- {
- GLenum glErr = aglGetError();
- //WEBRTC_LOG(kTraceError, "aglSetInteger returned FALSE with error code %d at line %d\n", glErr, __LINE__);
- }
-
- DisposeRgn(visibleRgn);
- }
- else
- {
- //WEBRTC_LOG(kTraceDebug, "Not using a hiviewref!\n");
- }
-
- //WEBRTC_LOG(kTraceDebug, "Leaving UpdateClipping()");
- UnlockAGLCntx();
- return true;
-}
-
-int VideoRenderAGL::CalculateVisibleRegion(ControlRef control, RgnHandle &visibleRgn, bool clipChildren)
-{
-
- // LockAGLCntx();
-
- //WEBRTC_LOG(kTraceDebug, "Entering CalculateVisibleRegion()");
- OSStatus osStatus = 0;
- OSErr osErr = 0;
-
- RgnHandle tempRgn = NewRgn();
- if (IsControlVisible(control))
- {
- RgnHandle childRgn = NewRgn();
- WindowRef window = GetControlOwner(control);
- ControlRef rootControl;
- GetRootControl(window, &rootControl); // 'wvnc'
- ControlRef masterControl;
- osStatus = GetSuperControl(rootControl, &masterControl);
- // //WEBRTC_LOG(kTraceDebug, "IBM GetSuperControl=%d", osStatus);
-
- if (masterControl != NULL)
- {
- CheckValidRegion(visibleRgn);
- // init visibleRgn with region of 'wvnc'
- osStatus = GetControlRegion(rootControl, kControlStructureMetaPart, visibleRgn);
- // //WEBRTC_LOG(kTraceDebug, "IBM GetControlRegion=%d : %d", osStatus, __LINE__);
- //GetSuperControl(rootControl, &rootControl);
- ControlRef tempControl = control, lastControl = 0;
- while (tempControl != masterControl) // current control != master
-
- {
- CheckValidRegion(tempRgn);
-
- // //WEBRTC_LOG(kTraceDebug, "IBM tempControl=%d masterControl=%d", tempControl, masterControl);
- ControlRef subControl;
-
- osStatus = GetControlRegion(tempControl, kControlStructureMetaPart, tempRgn); // intersect the region of the current control with visibleRgn
- // //WEBRTC_LOG(kTraceDebug, "IBM GetControlRegion=%d : %d", osStatus, __LINE__);
- CheckValidRegion(tempRgn);
-
- osErr = HIViewConvertRegion(tempRgn, tempControl, rootControl);
- // //WEBRTC_LOG(kTraceDebug, "IBM HIViewConvertRegion=%d : %d", osErr, __LINE__);
- CheckValidRegion(tempRgn);
-
- SectRgn(tempRgn, visibleRgn, visibleRgn);
- CheckValidRegion(tempRgn);
- CheckValidRegion(visibleRgn);
- if (EmptyRgn(visibleRgn)) // if the region is empty, bail
- break;
-
- if (clipChildren || tempControl != control) // clip children if true, cut out the tempControl if it's not one passed to this function
-
- {
- UInt16 numChildren;
- osStatus = CountSubControls(tempControl, &numChildren); // count the subcontrols
- // //WEBRTC_LOG(kTraceDebug, "IBM CountSubControls=%d : %d", osStatus, __LINE__);
-
- // //WEBRTC_LOG(kTraceDebug, "IBM numChildren=%d", numChildren);
- for (int i = 0; i < numChildren; i++)
- {
- osErr = GetIndexedSubControl(tempControl, numChildren - i, &subControl); // retrieve the subcontrol in order by zorder
- // //WEBRTC_LOG(kTraceDebug, "IBM GetIndexedSubControls=%d : %d", osErr, __LINE__);
- if ( subControl == lastControl ) // break because of zorder
-
- {
- // //WEBRTC_LOG(kTraceDebug, "IBM breaking because of zorder %d", __LINE__);
- break;
- }
-
- if (!IsControlVisible(subControl)) // dont' clip invisible controls
-
- {
- // //WEBRTC_LOG(kTraceDebug, "IBM continue. Control is not visible %d", __LINE__);
- continue;
- }
-
- if(!subControl) continue;
-
- osStatus = GetControlRegion(subControl, kControlStructureMetaPart, tempRgn); //get the region of the current control and union to childrg
- // //WEBRTC_LOG(kTraceDebug, "IBM GetControlRegion=%d %d", osStatus, __LINE__);
- CheckValidRegion(tempRgn);
- if(osStatus != 0)
- {
- // //WEBRTC_LOG(kTraceDebug, "IBM ERROR! osStatus=%d. Continuing. %d", osStatus, __LINE__);
- continue;
- }
- if(!tempRgn)
- {
- // //WEBRTC_LOG(kTraceDebug, "IBM ERROR! !tempRgn %d", osStatus, __LINE__);
- continue;
- }
-
- osStatus = HIViewConvertRegion(tempRgn, subControl, rootControl);
- CheckValidRegion(tempRgn);
- // //WEBRTC_LOG(kTraceDebug, "IBM HIViewConvertRegion=%d %d", osStatus, __LINE__);
- if(osStatus != 0)
- {
- // //WEBRTC_LOG(kTraceDebug, "IBM ERROR! osStatus=%d. Continuing. %d", osStatus, __LINE__);
- continue;
- }
- if(!rootControl)
- {
- // //WEBRTC_LOG(kTraceDebug, "IBM ERROR! !rootControl %d", osStatus, __LINE__);
- continue;
- }
-
- UnionRgn(tempRgn, childRgn, childRgn);
- CheckValidRegion(tempRgn);
- CheckValidRegion(childRgn);
- CheckValidRegion(visibleRgn);
- if(!childRgn)
- {
- // //WEBRTC_LOG(kTraceDebug, "IBM ERROR! !childRgn %d", osStatus, __LINE__);
- continue;
- }
-
- } // next child control
- }
- lastControl = tempControl;
- GetSuperControl(tempControl, &subControl);
- tempControl = subControl;
- }
-
- DiffRgn(visibleRgn, childRgn, visibleRgn);
- CheckValidRegion(visibleRgn);
- CheckValidRegion(childRgn);
- DisposeRgn(childRgn);
- }
- else
- {
- CopyRgn(tempRgn, visibleRgn);
- CheckValidRegion(tempRgn);
- CheckValidRegion(visibleRgn);
- }
- DisposeRgn(tempRgn);
- }
-
- //WEBRTC_LOG(kTraceDebug, "Leaving CalculateVisibleRegion()");
- //_aglCritPtr->Leave();
- return 0;
-}
-
-bool VideoRenderAGL::CheckValidRegion(RgnHandle rHandle)
-{
-
- Handle hndSize = (Handle)rHandle;
- long size = GetHandleSize(hndSize);
- if(0 == size)
- {
-
- OSErr memErr = MemError();
- if(noErr != memErr)
- {
- // //WEBRTC_LOG(kTraceError, "IBM ERROR Could not get size of handle. MemError() returned %d", memErr);
- }
- else
- {
- // //WEBRTC_LOG(kTraceError, "IBM ERROR Could not get size of handle yet MemError() returned noErr");
- }
-
- }
- else
- {
- // //WEBRTC_LOG(kTraceDebug, "IBM handleSize = %d", size);
- }
-
- if(false == IsValidRgnHandle(rHandle))
- {
- // //WEBRTC_LOG(kTraceError, "IBM ERROR Invalid Region found : $%d", rHandle);
- assert(false);
- }
-
- int err = QDError();
- switch(err)
- {
- case 0:
- break;
- case -147:
- //WEBRTC_LOG(kTraceError, "ERROR region too big");
- assert(false);
- break;
-
- case -149:
- //WEBRTC_LOG(kTraceError, "ERROR not enough stack");
- assert(false);
- break;
-
- default:
- //WEBRTC_LOG(kTraceError, "ERROR Unknown QDError %d", err);
- assert(false);
- break;
- }
-
- return true;
-}
-
-int VideoRenderAGL::ChangeWindow(void* newWindowRef)
-{
-
- LockAGLCntx();
-
- UnlockAGLCntx();
- return -1;
-}
-WebRtc_Word32 VideoRenderAGL::ChangeUniqueID(WebRtc_Word32 id)
-{
- LockAGLCntx();
-
- UnlockAGLCntx();
- return -1;
-}
-
-WebRtc_Word32 VideoRenderAGL::StartRender()
-{
-
- LockAGLCntx();
- const unsigned int MONITOR_FREQ = 60;
- if(TRUE == _renderingIsPaused)
- {
- //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Rendering is paused. Restarting now", __FUNCTION__, __LINE__);
-
- // we already have the thread. Most likely StopRender() was called and they were paused
- if(FALSE == _screenUpdateThread->Start(_threadID))
- {
- //WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Failed to start screenUpdateThread", __FUNCTION__, __LINE__);
- UnlockAGLCntx();
- return -1;
- }
- if(FALSE == _screenUpdateEvent->StartTimer(true, 1000/MONITOR_FREQ))
- {
- //WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Failed to start screenUpdateEvent", __FUNCTION__, __LINE__);
- UnlockAGLCntx();
- return -1;
- }
-
- return 0;
- }
-
- _screenUpdateThread = ThreadWrapper::CreateThread(ScreenUpdateThreadProc, this, kRealtimePriority);
- _screenUpdateEvent = EventWrapper::Create();
-
- if (!_screenUpdateThread)
- {
- //WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Failed to start screenUpdateThread", __FUNCTION__, __LINE__);
- UnlockAGLCntx();
- return -1;
- }
-
- _screenUpdateThread->Start(_threadID);
- _screenUpdateEvent->StartTimer(true, 1000/MONITOR_FREQ);
-
- //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Started screenUpdateThread", __FUNCTION__, __LINE__);
-
- UnlockAGLCntx();
- return 0;
-
-}
-
-WebRtc_Word32 VideoRenderAGL::StopRender()
-{
- LockAGLCntx();
-
- if(!_screenUpdateThread || !_screenUpdateEvent)
- {
- _renderingIsPaused = TRUE;
- UnlockAGLCntx();
- return 0;
- }
-
- if(FALSE == _screenUpdateThread->Stop() || FALSE == _screenUpdateEvent->StopTimer())
- {
- _renderingIsPaused = FALSE;
- //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Could not stop either: screenUpdateThread or screenUpdateEvent", __FUNCTION__, __LINE__);
- UnlockAGLCntx();
- return -1;
- }
-
- _renderingIsPaused = TRUE;
-
- //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Stopped screenUpdateThread", __FUNCTION__, __LINE__);
- UnlockAGLCntx();
- return 0;
-}
-
-WebRtc_Word32 VideoRenderAGL::DeleteAGLChannel(const WebRtc_UWord32 streamID)
-{
-
- LockAGLCntx();
-
- std::map<int, VideoChannelAGL*>::iterator it;
- it = _aglChannels.begin();
-
- while (it != _aglChannels.end())
- {
- VideoChannelAGL* channel = it->second;
- //WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Deleting channel %d", __FUNCTION__, __LINE__, streamID);
- delete channel;
- it++;
- }
- _aglChannels.clear();
-
- UnlockAGLCntx();
- return 0;
-}
-
-WebRtc_Word32 VideoRenderAGL::GetChannelProperties(const WebRtc_UWord16 streamId,
-WebRtc_UWord32& zOrder,
-float& left,
-float& top,
-float& right,
-float& bottom)
-{
-
- LockAGLCntx();
- UnlockAGLCntx();
- return -1;
-
-}
-
-void VideoRenderAGL::LockAGLCntx()
-{
- _renderCritSec.Enter();
-}
-void VideoRenderAGL::UnlockAGLCntx()
-{
- _renderCritSec.Leave();
-}
-
-} //namespace webrtc
-
-#endif // CARBON_RENDERING
-
diff --git a/trunk/src/modules/video_render/main/source/mac/video_render_agl.h b/trunk/src/modules/video_render/main/source/mac/video_render_agl.h
deleted file mode 100644
index bdee619..0000000
--- a/trunk/src/modules/video_render/main/source/mac/video_render_agl.h
+++ /dev/null
@@ -1,185 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "engine_configurations.h"
-
-#if defined(CARBON_RENDERING)
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_AGL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_AGL_H_
-
-
-#include "video_render_defines.h"
-
-
-#define NEW_HIVIEW_PARENT_EVENT_HANDLER 1
-#define NEW_HIVIEW_EVENT_HANDLER 1
-#define USE_STRUCT_RGN
-
-#include <AGL/agl.h>
-#include <Carbon/Carbon.h>
-#include <OpenGL/OpenGL.h>
-#include <OpenGL/glu.h>
-#include <OpenGL/glext.h>
-#include <list>
-#include <map>
-
-class VideoRenderAGL;
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-class ThreadWrapper;
-
-class VideoChannelAGL : public VideoRenderCallback
-{
-public:
-
- VideoChannelAGL(AGLContext& aglContext, int iId, VideoRenderAGL* owner);
- virtual ~VideoChannelAGL();
- virtual int FrameSizeChange(int width, int height, int numberOfStreams);
- virtual int DeliverFrame(unsigned char* buffer, int bufferSize, unsigned int timeStame90kHz);
- virtual int UpdateSize(int width, int height);
- int SetStreamSettings(int streamId, float startWidth, float startHeight, float stopWidth, float stopHeight);
- int SetStreamCropSettings(int streamId, float startWidth, float startHeight, float stopWidth, float stopHeight);
- int RenderOffScreenBuffer();
- int IsUpdated(bool& isUpdated);
- virtual int UpdateStretchSize(int stretchHeight, int stretchWidth);
- virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId, VideoFrame& videoFrame);
-
-
-private:
-
- AGLContext _aglContext;
- int _id;
- VideoRenderAGL* _owner;
- int _width;
- int _height;
- int _stretchedWidth;
- int _stretchedHeight;
- float _startHeight;
- float _startWidth;
- float _stopWidth;
- float _stopHeight;
- int _xOldWidth;
- int _yOldHeight;
- int _oldStretchedHeight;
- int _oldStretchedWidth;
- unsigned char* _buffer;
- int _bufferSize;
- int _incommingBufferSize;
- bool _bufferIsUpdated;
- bool _sizeInitialized;
- int _numberOfStreams;
- bool _bVideoSizeStartedChanging;
- GLenum _pixelFormat;
- GLenum _pixelDataType;
- unsigned int _texture;
-};
-
-
-
-
-class VideoRenderAGL
-{
-public:
- VideoRenderAGL(WindowRef windowRef, bool fullscreen, int iId);
- VideoRenderAGL(HIViewRef windowRef, bool fullscreen, int iId);
- ~VideoRenderAGL();
-
- int Init();
- VideoChannelAGL* CreateAGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight);
- VideoChannelAGL* ConfigureAGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight);
- int DeleteAGLChannel(int channel);
- int DeleteAllAGLChannels();
- int StopThread();
- bool IsFullScreen();
- bool HasChannels();
- bool HasChannel(int channel);
- int GetChannels(std::list<int>& channelList);
- void LockAGLCntx();
- void UnlockAGLCntx();
-
- static int GetOpenGLVersion(int& aglMajor, int& aglMinor);
-
- // ********** new module functions ************ //
- int ChangeWindow(void* newWindowRef);
- WebRtc_Word32 ChangeUniqueID(WebRtc_Word32 id);
- WebRtc_Word32 StartRender();
- WebRtc_Word32 StopRender();
- WebRtc_Word32 DeleteAGLChannel(const WebRtc_UWord32 streamID);
- WebRtc_Word32 GetChannelProperties(const WebRtc_UWord16 streamId,
- WebRtc_UWord32& zOrder,
- float& left,
- float& top,
- float& right,
- float& bottom);
-
-protected:
- static bool ScreenUpdateThreadProc(void* obj);
- bool ScreenUpdateProcess();
- int GetWindowRect(Rect& rect);
-
-private:
- int CreateMixingContext();
- int RenderOffScreenBuffers();
- int SwapAndDisplayBuffers();
- int UpdateClipping();
- int CalculateVisibleRegion(ControlRef control, RgnHandle &visibleRgn, bool clipChildren);
- bool CheckValidRegion(RgnHandle rHandle);
- void ParentWindowResized(WindowRef window);
-
- // Carbon GUI event handlers
- static pascal OSStatus sHandleWindowResized (EventHandlerCallRef nextHandler, EventRef theEvent, void* userData);
- static pascal OSStatus sHandleHiViewResized (EventHandlerCallRef nextHandler, EventRef theEvent, void* userData);
-
- HIViewRef _hiviewRef;
- WindowRef _windowRef;
- bool _fullScreen;
- int _id;
- webrtc::CriticalSectionWrapper& _renderCritSec;
- webrtc::ThreadWrapper* _screenUpdateThread;
- webrtc::EventWrapper* _screenUpdateEvent;
- bool _isHIViewRef;
- AGLContext _aglContext;
- int _windowWidth;
- int _windowHeight;
- int _lastWindowWidth;
- int _lastWindowHeight;
- int _lastHiViewWidth;
- int _lastHiViewHeight;
- int _currentParentWindowHeight;
- int _currentParentWindowWidth;
- Rect _currentParentWindowBounds;
- bool _windowHasResized;
- Rect _lastParentWindowBounds;
- Rect _currentHIViewBounds;
- Rect _lastHIViewBounds;
- Rect _windowRect;
- std::map<int, VideoChannelAGL*> _aglChannels;
- std::multimap<int, int> _zOrderToChannel;
- EventHandlerRef _hiviewEventHandlerRef;
- EventHandlerRef _windowEventHandlerRef;
- HIRect _currentViewBounds;
- HIRect _lastViewBounds;
- bool _renderingIsPaused;
- unsigned int _threadID;
-
-
-
-
-};
-
-} //namespace webrtc
-
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_AGL_H_
-
-#endif // CARBON_RENDERING
diff --git a/trunk/src/modules/video_render/main/source/mac/video_render_mac_carbon_impl.cc b/trunk/src/modules/video_render/main/source/mac/video_render_mac_carbon_impl.cc
deleted file mode 100644
index 88dc1e6..0000000
--- a/trunk/src/modules/video_render/main/source/mac/video_render_mac_carbon_impl.cc
+++ /dev/null
@@ -1,297 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "engine_configurations.h"
-#if defined(CARBON_RENDERING)
-
-#include "video_render_mac_carbon_impl.h"
-#include "critical_section_wrapper.h"
-#include "video_render_agl.h"
-#include "trace.h"
-#include <AGL/agl.h>
-
-namespace webrtc {
-
-VideoRenderMacCarbonImpl::VideoRenderMacCarbonImpl(const WebRtc_Word32 id,
- const VideoRenderType videoRenderType,
- void* window,
- const bool fullscreen) :
-_id(id),
-_renderMacCarbonCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
-_fullScreen(fullscreen),
-_ptrWindow(window)
-{
-
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "Constructor %s:%d", __FUNCTION__, __LINE__);
-
-}
-
-VideoRenderMacCarbonImpl::~VideoRenderMacCarbonImpl()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "Destructor %s:%d", __FUNCTION__, __LINE__);
- delete &_renderMacCarbonCritsect;
-}
-
-WebRtc_Word32
-VideoRenderMacCarbonImpl::Init()
-{
- CriticalSectionScoped cs(_renderMacCarbonCritsect);
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d", __FUNCTION__, __LINE__);
-
- if (!_ptrWindow)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "Constructor %s:%d", __FUNCTION__, __LINE__);
- return -1;
- }
-
- // We don't know if the user passed us a WindowRef or a HIViewRef, so test.
- bool referenceIsValid = false;
-
- // Check if it's a valid WindowRef
- //WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s:%d _ptrWindowRef before WindowRef cast: %x", __FUNCTION__, __LINE__, _ptrWindowRef);
- WindowRef* windowRef = static_cast<WindowRef*>(_ptrWindow);
- //WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s:%d _ptrWindowRef after cast: %x", __FUNCTION__, __LINE__, _ptrWindowRef);
- if (IsValidWindowPtr(*windowRef))
- {
- _ptrCarbonRender = new VideoRenderAGL(*windowRef, _fullScreen, _id);
- referenceIsValid = true;
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Successfully initialized CarbonRenderer with WindowRef:%x", __FUNCTION__, __LINE__, *windowRef);
- }
- else
- {
- HIViewRef* hiviewRef = static_cast<HIViewRef*>(_ptrWindow);
- if (HIViewIsValid(*hiviewRef))
- {
- _ptrCarbonRender = new VideoRenderAGL(*hiviewRef, _fullScreen, _id);
- referenceIsValid = true;
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d Successfully initialized CarbonRenderer with HIViewRef:%x", __FUNCTION__, __LINE__, hiviewRef);
- }
- }
-
- if(!referenceIsValid)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Invalid WindowRef/HIViewRef Returning -1", __FUNCTION__, __LINE__);
- return -1;
- }
-
- if(!_ptrCarbonRender)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Failed to create an instance of VideoRenderAGL. Returning -1", __FUNCTION__, __LINE__);
- }
-
- int retVal = _ptrCarbonRender->Init();
- if (retVal == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s:%d Failed to init CarbonRenderer", __FUNCTION__, __LINE__);
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32
-VideoRenderMacCarbonImpl::ChangeUniqueId(const WebRtc_Word32 id)
-{
- return -1;
-
- CriticalSectionScoped cs(_renderMacCarbonCritsect);
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s", __FUNCTION__);
- _id = id;
-
- if(_ptrCarbonRender)
- {
- _ptrCarbonRender->ChangeUniqueID(_id);
- }
-
- return 0;
-}
-
-WebRtc_Word32
-VideoRenderMacCarbonImpl::ChangeWindow(void* window)
-{
- return -1;
- CriticalSectionScoped cs(_renderMacCarbonCritsect);
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s changing ID to ", __FUNCTION__, window);
-
- if (window == NULL)
- {
- return -1;
- }
- _ptrWindow = window;
-
-
- _ptrWindow = window;
-
- return 0;
-}
-
-VideoRenderCallback*
-VideoRenderMacCarbonImpl::AddIncomingRenderStream(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
-
- CriticalSectionScoped cs(_renderMacCarbonCritsect);
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s", __FUNCTION__);
- VideoChannelAGL* AGLChannel = NULL;
-
- if(!_ptrWindow)
- {
- }
-
- if(!AGLChannel)
- {
- AGLChannel = _ptrCocoaRender->CreateNSGLChannel(streamId, zOrder, left, top, right, bottom);
- }
-
- return AGLChannel;
-
-}
-
-WebRtc_Word32
-VideoRenderMacCarbonImpl::DeleteIncomingRenderStream(const WebRtc_UWord32 streamId)
-{
-
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s:%d", __FUNCTION__, __LINE__);
- CriticalSectionScoped cs(_renderMacCarbonCritsect);
- _ptrCarbonRender->DeleteAGLChannel(streamId);
-
- return 0;
-}
-
-WebRtc_Word32
-VideoRenderMacCarbonImpl::GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
- WebRtc_UWord32& zOrder,
- float& left,
- float& top,
- float& right,
- float& bottom) const
-{
- return -1;
- return _ptrCarbonRender->GetChannelProperties(streamId, zOrder, left, top, right, bottom);
-}
-
-WebRtc_Word32
-VideoRenderMacCarbonImpl::StartRender()
-{
- return _ptrCarbonRender->StartRender();
-}
-
-WebRtc_Word32
-VideoRenderMacCarbonImpl::StopRender()
-{
- return _ptrCarbonRender->StopRender();
-}
-
-VideoRenderType
-VideoRenderMacCarbonImpl::RenderType()
-{
- return kRenderCarbon;
-}
-
-RawVideoType
-VideoRenderMacCarbonImpl::PerferedVideoType()
-{
- return kVideoI420;
-}
-
-bool
-VideoRenderMacCarbonImpl::FullScreen()
-{
- return false;
-}
-
-WebRtc_Word32
-VideoRenderMacCarbonImpl::GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
- WebRtc_UWord64& availableGraphicsMemory) const
-{
- totalGraphicsMemory = 0;
- availableGraphicsMemory = 0;
- return 0;
-}
-
-WebRtc_Word32
-VideoRenderMacCarbonImpl::GetScreenResolution(WebRtc_UWord32& screenWidth,
- WebRtc_UWord32& screenHeight) const
-{
- CriticalSectionScoped cs(_renderMacCarbonCritsect);
- //NSScreen* mainScreen = [NSScreen mainScreen];
-
- //NSRect frame = [mainScreen frame];
-
- //screenWidth = frame.size.width;
- //screenHeight = frame.size.height;
- return 0;
-}
-
-WebRtc_UWord32
-VideoRenderMacCarbonImpl::RenderFrameRate(const WebRtc_UWord32 streamId)
-{
- CriticalSectionScoped cs(_renderMacCarbonCritsect);
- return 0;
-}
-
-WebRtc_Word32
-VideoRenderMacCarbonImpl::SetStreamCropping(const WebRtc_UWord32 streamId,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- return 0;
-}
-
-WebRtc_Word32 VideoRenderMacCarbonImpl::ConfigureRenderer(const WebRtc_UWord32 streamId,
- const unsigned int zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- return 0;
-}
-
-WebRtc_Word32
-VideoRenderMacCarbonImpl::SetTransparentBackground(const bool enable)
-{
- return 0;
-}
-
-WebRtc_Word32 VideoRenderMacCarbonImpl::SetText(const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 textColorRef,
- const WebRtc_UWord32 backgroundColorRef,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- return 0;
-}
-
-WebRtc_Word32 VideoRenderMacCarbonImpl::SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- return 0;
-}
-
-
-} //namespace webrtc
-
-#endif // CARBON_RENDERING
diff --git a/trunk/src/modules/video_render/main/source/mac/video_render_mac_carbon_impl.h b/trunk/src/modules/video_render/main/source/mac/video_render_mac_carbon_impl.h
deleted file mode 100644
index 3ff3d26..0000000
--- a/trunk/src/modules/video_render/main/source/mac/video_render_mac_carbon_impl.h
+++ /dev/null
@@ -1,148 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "engine_configurations.h"
-#if defined(CARBON_RENDERING)
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_CARBON_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_CARBON_IMPL_H_
-
-#include "i_video_render.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class VideoRenderAGL;
-
-// Class definitions
-class VideoRenderMacCarbonImpl : IVideoRender
-{
-public:
- /*
- * Constructor/destructor
- */
-
- VideoRenderMacCarbonImpl(const WebRtc_Word32 id,
- const VideoRenderType videoRenderType,
- void* window,
- const bool fullscreen);
-
- virtual ~VideoRenderMacCarbonImpl();
-
- virtual WebRtc_Word32 Init();
-
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- virtual WebRtc_Word32 ChangeWindow(void* window);
-
- /**************************************************************************
- *
- * Incoming Streams
- *
- ***************************************************************************/
- virtual VideoRenderCallback* AddIncomingRenderStream(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 DeleteIncomingRenderStream(const WebRtc_UWord32 streamId);
-
- virtual WebRtc_Word32 GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
- WebRtc_UWord32& zOrder,
- float& left,
- float& top,
- float& right,
- float& bottom) const;
-
- /**************************************************************************
- *
- * Start/Stop
- *
- ***************************************************************************/
-
- virtual WebRtc_Word32 StartRender();
-
- virtual WebRtc_Word32 StopRender();
-
- /**************************************************************************
- *
- * Properties
- *
- ***************************************************************************/
-
- virtual VideoRenderType RenderType();
-
- virtual RawVideoType PerferedVideoType();
-
- virtual bool FullScreen();
-
- virtual WebRtc_Word32 GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
- WebRtc_UWord64& availableGraphicsMemory) const;
-
- virtual WebRtc_Word32 GetScreenResolution(WebRtc_UWord32& screenWidth,
- WebRtc_UWord32& screenHeight) const;
-
- virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId);
-
- virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
- const unsigned int zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
-
- virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 textColorRef,
- const WebRtc_UWord32 backgroundColorRef,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 FullScreenRender(void* window, const bool enable)
- {
- // not supported in Carbon at this time
- return -1;
- }
-
-private:
- WebRtc_Word32 _id;
- CriticalSectionWrapper& _renderMacCarbonCritsect;
- bool _fullScreen;
- void* _ptrWindow;
- VideoRenderAGL* _ptrCarbonRender;
-
-};
-
-
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_CARBON_IMPL_H_
-#endif // CARBON_RENDERING
diff --git a/trunk/src/modules/video_render/main/source/mac/video_render_mac_cocoa_impl.h b/trunk/src/modules/video_render/main/source/mac/video_render_mac_cocoa_impl.h
deleted file mode 100644
index f9216bb..0000000
--- a/trunk/src/modules/video_render/main/source/mac/video_render_mac_cocoa_impl.h
+++ /dev/null
@@ -1,143 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "engine_configurations.h"
-
-#if defined(COCOA_RENDERING)
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_COCOA_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_COCOA_IMPL_H_
-
-#include "i_video_render.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class VideoRenderNSOpenGL;
-
-// Class definitions
-class VideoRenderMacCocoaImpl : IVideoRender
-{
-public:
- /*
- * Constructor/destructor
- */
-
- VideoRenderMacCocoaImpl(const WebRtc_Word32 id,
- const VideoRenderType videoRenderType,
- void* window,
- const bool fullscreen);
-
- virtual ~VideoRenderMacCocoaImpl();
-
- virtual WebRtc_Word32 Init();
-
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- virtual WebRtc_Word32 ChangeWindow(void* window);
-
- /**************************************************************************
- *
- * Incoming Streams
- *
- ***************************************************************************/
- virtual VideoRenderCallback* AddIncomingRenderStream(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 DeleteIncomingRenderStream(const WebRtc_UWord32 streamId);
-
- virtual WebRtc_Word32 GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
- WebRtc_UWord32& zOrder,
- float& left,
- float& top,
- float& right,
- float& bottom) const;
-
- /**************************************************************************
- *
- * Start/Stop
- *
- ***************************************************************************/
-
- virtual WebRtc_Word32 StartRender();
-
- virtual WebRtc_Word32 StopRender();
-
- /**************************************************************************
- *
- * Properties
- *
- ***************************************************************************/
-
- virtual VideoRenderType RenderType();
-
- virtual RawVideoType PerferedVideoType();
-
- virtual bool FullScreen();
-
- virtual WebRtc_Word32 GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
- WebRtc_UWord64& availableGraphicsMemory) const;
-
- virtual WebRtc_Word32 GetScreenResolution(WebRtc_UWord32& screenWidth,
- WebRtc_UWord32& screenHeight) const;
-
- virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId);
-
- virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
- const unsigned int zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
-
- virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 textColorRef,
- const WebRtc_UWord32 backgroundColorRef,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 FullScreenRender(void* window, const bool enable);
-
-private:
- WebRtc_Word32 _id;
- CriticalSectionWrapper& _renderMacCocoaCritsect;
- bool _fullScreen;
- void* _ptrWindow;
- VideoRenderNSOpenGL* _ptrCocoaRender;
-
-};
-
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_MAC_COCOA_IMPL_H_
-#endif // COCOA_RENDERING
diff --git a/trunk/src/modules/video_render/main/source/mac/video_render_mac_cocoa_impl.mm b/trunk/src/modules/video_render/main/source/mac/video_render_mac_cocoa_impl.mm
deleted file mode 100644
index 4698b6e..0000000
--- a/trunk/src/modules/video_render/main/source/mac/video_render_mac_cocoa_impl.mm
+++ /dev/null
@@ -1,269 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "engine_configurations.h"
-#if defined(COCOA_RENDERING)
-
-#import "cocoa_render_view.h"
-
-#include "video_render_mac_cocoa_impl.h"
-#include "critical_section_wrapper.h"
-#include "video_render_nsopengl.h"
-#include "trace.h"
-
-namespace webrtc {
-
-VideoRenderMacCocoaImpl::VideoRenderMacCocoaImpl(const WebRtc_Word32 id,
- const VideoRenderType videoRenderType,
- void* window,
- const bool fullscreen) :
-_id(id),
-_renderMacCocoaCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
-_fullScreen(fullscreen),
-_ptrWindow(window)
-{
-
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "Constructor %s:%d", __FUNCTION__, __LINE__);
-}
-
-VideoRenderMacCocoaImpl::~VideoRenderMacCocoaImpl()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "Destructor %s:%d", __FUNCTION__, __LINE__);
- delete &_renderMacCocoaCritsect;
- if (_ptrCocoaRender)
- {
- delete _ptrCocoaRender;
- _ptrCocoaRender = NULL;
- }
-}
-
-WebRtc_Word32
-VideoRenderMacCocoaImpl::Init()
-{
-
- CriticalSectionScoped cs(_renderMacCocoaCritsect);
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s:%d", __FUNCTION__, __LINE__);
-
- // cast ptrWindow from void* to CocoaRenderer. Void* was once NSOpenGLView, and CocoaRenderer is NSOpenGLView.
- _ptrCocoaRender = new VideoRenderNSOpenGL((CocoaRenderView*)_ptrWindow, _fullScreen, _id);
- if (!_ptrWindow)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "Constructor %s:%d", __FUNCTION__, __LINE__);
- return -1;
- }
- int retVal = _ptrCocoaRender->Init();
- if (retVal == -1)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "Failed to init %s:%d", __FUNCTION__, __LINE__);
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32
-VideoRenderMacCocoaImpl::ChangeUniqueId(const WebRtc_Word32 id)
-{
- CriticalSectionScoped cs(_renderMacCocoaCritsect);
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s", __FUNCTION__);
- _id = id;
-
- if(_ptrCocoaRender)
- {
- _ptrCocoaRender->ChangeUniqueID(_id);
- }
-
- return 0;
-}
-
-WebRtc_Word32
-VideoRenderMacCocoaImpl::ChangeWindow(void* window)
-{
-
- CriticalSectionScoped cs(_renderMacCocoaCritsect);
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s changing ID to ", __FUNCTION__, window);
-
- if (window == NULL)
- {
- return -1;
- }
- _ptrWindow = window;
-
-
- _ptrWindow = window;
- _ptrCocoaRender->ChangeWindow((CocoaRenderView*)_ptrWindow);
-
- return 0;
-}
-
-VideoRenderCallback*
-VideoRenderMacCocoaImpl::AddIncomingRenderStream(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_renderMacCocoaCritsect);
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s", __FUNCTION__);
- VideoChannelNSOpenGL* nsOpenGLChannel = NULL;
-
- if(!_ptrWindow)
- {
- }
-
- if(!nsOpenGLChannel)
- {
- nsOpenGLChannel = _ptrCocoaRender->CreateNSGLChannel(streamId, zOrder, left, top, right, bottom);
- }
-
- return nsOpenGLChannel;
-
-}
-
-WebRtc_Word32
-VideoRenderMacCocoaImpl::DeleteIncomingRenderStream(const WebRtc_UWord32 streamId)
-{
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "Constructor %s:%d", __FUNCTION__, __LINE__);
- CriticalSectionScoped cs(_renderMacCocoaCritsect);
- _ptrCocoaRender->DeleteNSGLChannel(streamId);
-
- return 0;
-}
-
-WebRtc_Word32
-VideoRenderMacCocoaImpl::GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
- WebRtc_UWord32& zOrder,
- float& left,
- float& top,
- float& right,
- float& bottom) const
-{
- return _ptrCocoaRender->GetChannelProperties(streamId, zOrder, left, top, right, bottom);
-}
-
-WebRtc_Word32
-VideoRenderMacCocoaImpl::StartRender()
-{
- return _ptrCocoaRender->StartRender();
-}
-
-WebRtc_Word32
-VideoRenderMacCocoaImpl::StopRender()
-{
- return _ptrCocoaRender->StopRender();
-}
-
-VideoRenderType
-VideoRenderMacCocoaImpl::RenderType()
-{
- return kRenderCocoa;
-}
-
-RawVideoType
-VideoRenderMacCocoaImpl::PerferedVideoType()
-{
- return kVideoI420;
-}
-
-bool
-VideoRenderMacCocoaImpl::FullScreen()
-{
- return false;
-}
-
-WebRtc_Word32
-VideoRenderMacCocoaImpl::GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
- WebRtc_UWord64& availableGraphicsMemory) const
-{
- totalGraphicsMemory = 0;
- availableGraphicsMemory = 0;
- return 0;
-}
-
-WebRtc_Word32
-VideoRenderMacCocoaImpl::GetScreenResolution(WebRtc_UWord32& screenWidth,
- WebRtc_UWord32& screenHeight) const
-{
- CriticalSectionScoped cs(_renderMacCocoaCritsect);
- NSScreen* mainScreen = [NSScreen mainScreen];
-
- NSRect frame = [mainScreen frame];
-
- screenWidth = frame.size.width;
- screenHeight = frame.size.height;
- return 0;
-}
-
-WebRtc_UWord32
-VideoRenderMacCocoaImpl::RenderFrameRate(const WebRtc_UWord32 streamId)
-{
- CriticalSectionScoped cs(_renderMacCocoaCritsect);
- return 0;
-}
-
-WebRtc_Word32
-VideoRenderMacCocoaImpl::SetStreamCropping(const WebRtc_UWord32 streamId,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- return 0;
-}
-
-WebRtc_Word32 VideoRenderMacCocoaImpl::ConfigureRenderer(const WebRtc_UWord32 streamId,
- const unsigned int zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- return 0;
-}
-
-WebRtc_Word32
-VideoRenderMacCocoaImpl::SetTransparentBackground(const bool enable)
-{
- return 0;
-}
-
-WebRtc_Word32 VideoRenderMacCocoaImpl::SetText(const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 textColorRef,
- const WebRtc_UWord32 backgroundColorRef,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- return _ptrCocoaRender->SetText(textId, text, textLength, textColorRef, backgroundColorRef, left, top, right, bottom);
-}
-
-WebRtc_Word32 VideoRenderMacCocoaImpl::SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- return 0;
-}
-
-WebRtc_Word32 VideoRenderMacCocoaImpl::FullScreenRender(void* window, const bool enable)
-{
- return -1;
-}
-
-} //namespace webrtc
-
-#endif // COCOA_RENDERING
diff --git a/trunk/src/modules/video_render/main/source/mac/video_render_nsopengl.h b/trunk/src/modules/video_render/main/source/mac/video_render_nsopengl.h
deleted file mode 100644
index cdd3be7..0000000
--- a/trunk/src/modules/video_render/main/source/mac/video_render_nsopengl.h
+++ /dev/null
@@ -1,193 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "engine_configurations.h"
-#if defined(COCOA_RENDERING)
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_NSOPENGL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_NSOPENGL_H_
-
-#import <Cocoa/Cocoa.h>
-#import <OpenGL/OpenGL.h>
-#import <OpenGL/glu.h>
-#import <OpenGL/glext.h>
-#include <QuickTime/QuickTime.h>
-#include <list>
-#include <map>
-
-#include "video_render_defines.h"
-
-#import "cocoa_render_view.h"
-#import "cocoa_full_screen_window.h"
-
-class Trace;
-
-namespace webrtc {
-class EventWrapper;
-class ThreadWrapper;
-class VideoRenderNSOpenGL;
-class CriticalSectionWrapper;
-
-class VideoChannelNSOpenGL : public VideoRenderCallback
-{
-
-public:
-
- VideoChannelNSOpenGL(NSOpenGLContext *nsglContext, int iId, VideoRenderNSOpenGL* owner);
- virtual ~VideoChannelNSOpenGL();
-
- // A new frame is delivered
- virtual int DeliverFrame(unsigned char* buffer, int bufferSize, unsigned int timeStame90kHz);
-
- // Called when the incomming frame size and/or number of streams in mix changes
- virtual int FrameSizeChange(int width, int height, int numberOfStreams);
-
- virtual int UpdateSize(int width, int height);
-
- // Setup
- int SetStreamSettings(int streamId, float startWidth, float startHeight, float stopWidth, float stopHeight);
- int SetStreamCropSettings(int streamId, float startWidth, float startHeight, float stopWidth, float stopHeight);
-
- // Called when it's time to render the last frame for the channel
- int RenderOffScreenBuffer();
-
- // Returns true if a new buffer has been delivered to the texture
- int IsUpdated(bool& isUpdated);
- virtual int UpdateStretchSize(int stretchHeight, int stretchWidth);
-
- // ********** new module functions ************ //
- virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId, VideoFrame& videoFrame);
-
- // ********** new module helper functions ***** //
- int ChangeContext(NSOpenGLContext *nsglContext);
- WebRtc_Word32 GetChannelProperties(float& left,
- float& top,
- float& right,
- float& bottom);
-
-private:
-
- NSOpenGLContext* _nsglContext;
- int _id;
- VideoRenderNSOpenGL* _owner;
- WebRtc_Word32 _width;
- WebRtc_Word32 _height;
- float _startWidth;
- float _startHeight;
- float _stopWidth;
- float _stopHeight;
- int _stretchedWidth;
- int _stretchedHeight;
- int _oldStretchedHeight;
- int _oldStretchedWidth;
- int _xOldWidth;
- int _yOldHeight;
- unsigned char* _buffer;
- int _bufferSize;
- int _incommingBufferSize;
- bool _bufferIsUpdated;
- int _numberOfStreams;
- GLenum _pixelFormat;
- GLenum _pixelDataType;
- unsigned int _texture;
- bool _bVideoSizeStartedChanging;
-};
-
-class VideoRenderNSOpenGL
-{
-
-public: // methods
- VideoRenderNSOpenGL(CocoaRenderView *windowRef, bool fullScreen, int iId);
- ~VideoRenderNSOpenGL();
-
- static int GetOpenGLVersion(int& nsglMajor, int& nsglMinor);
-
- // Allocates textures
- int Init();
- VideoChannelNSOpenGL* CreateNSGLChannel(int streamID, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight);
- VideoChannelNSOpenGL* ConfigureNSGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight);
- int DeleteNSGLChannel(int channel);
- int DeleteAllNSGLChannels();
- int StopThread();
- bool IsFullScreen();
- bool HasChannels();
- bool HasChannel(int channel);
- int GetChannels(std::list<int>& channelList);
- void LockAGLCntx();
- void UnlockAGLCntx();
-
- // ********** new module functions ************ //
- int ChangeWindow(CocoaRenderView* newWindowRef);
- WebRtc_Word32 ChangeUniqueID(WebRtc_Word32 id);
- WebRtc_Word32 StartRender();
- WebRtc_Word32 StopRender();
- WebRtc_Word32 DeleteNSGLChannel(const WebRtc_UWord32 streamID);
- WebRtc_Word32 GetChannelProperties(const WebRtc_UWord16 streamId,
- WebRtc_UWord32& zOrder,
- float& left,
- float& top,
- float& right,
- float& bottom);
-
- WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 textColorRef,
- const WebRtc_UWord32 backgroundColorRef,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- // ********** new module helper functions ***** //
- int configureNSOpenGLEngine();
- int configureNSOpenGLView();
- int setRenderTargetWindow();
- int setRenderTargetFullScreen();
-
-protected: // methods
- static bool ScreenUpdateThreadProc(void* obj);
- bool ScreenUpdateProcess();
- int GetWindowRect(Rect& rect);
-
-private: // methods
-
- int CreateMixingContext();
- int RenderOffScreenBuffers();
- int DisplayBuffers();
-
-private: // variables
-
-
- CocoaRenderView* _windowRef;
- bool _fullScreen;
- int _id;
- CriticalSectionWrapper& _nsglContextCritSec;
- ThreadWrapper* _screenUpdateThread;
- EventWrapper* _screenUpdateEvent;
- NSOpenGLContext* _nsglContext;
- NSOpenGLContext* _nsglFullScreenContext;
- CocoaFullScreenWindow* _fullScreenWindow;
- Rect _windowRect; // The size of the window
- int _windowWidth;
- int _windowHeight;
- std::map<int, VideoChannelNSOpenGL*> _nsglChannels;
- std::multimap<int, int> _zOrderToChannel;
- unsigned int _threadID;
- bool _renderingIsPaused;
- NSView* _windowRefSuperView;
- NSRect _windowRefSuperViewFrame;
-};
-
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_MAC_VIDEO_RENDER_NSOPENGL_H_
-#endif // COCOA_RENDERING
-
diff --git a/trunk/src/modules/video_render/main/source/mac/video_render_nsopengl.mm b/trunk/src/modules/video_render/main/source/mac/video_render_nsopengl.mm
deleted file mode 100644
index 4f1a6c5..0000000
--- a/trunk/src/modules/video_render/main/source/mac/video_render_nsopengl.mm
+++ /dev/null
@@ -1,1275 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "engine_configurations.h"
-#if defined(COCOA_RENDERING)
-
-#include "video_render_nsopengl.h"
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "trace.h"
-#include "thread_wrapper.h"
-#include "common_video/libyuv/include/libyuv.h"
-
-namespace webrtc {
-
-VideoChannelNSOpenGL::VideoChannelNSOpenGL(NSOpenGLContext *nsglContext, int iId, VideoRenderNSOpenGL* owner) :
-_nsglContext( nsglContext),
-_id( iId),
-_owner( owner),
-_width( 0),
-_height( 0),
-_startWidth( 0.0f),
-_startHeight( 0.0f),
-_stopWidth( 0.0f),
-_stopHeight( 0.0f),
-_stretchedWidth( 0),
-_stretchedHeight( 0),
-_oldStretchedHeight( 0),
-_oldStretchedWidth( 0),
-_xOldWidth( 0),
-_yOldHeight( 0),
-_buffer( 0),
-_bufferSize( 0),
-_incommingBufferSize( 0),
-_bufferIsUpdated( false),
-_numberOfStreams( 0),
-_pixelFormat( GL_RGBA),
-_pixelDataType( GL_UNSIGNED_INT_8_8_8_8),
-_texture( 0),
-_bVideoSizeStartedChanging(false)
-
-{
-
-}
-
-VideoChannelNSOpenGL::~VideoChannelNSOpenGL()
-{
- if (_buffer)
- {
- delete [] _buffer;
- _buffer = NULL;
- }
-
- if (_texture != 0)
- {
- [_nsglContext makeCurrentContext];
- glDeleteTextures(1, (const GLuint*) &_texture);
- _texture = 0;
- }
-}
-
-int VideoChannelNSOpenGL::ChangeContext(NSOpenGLContext *nsglContext)
-{
- _owner->UnlockAGLCntx();
-
- _nsglContext = nsglContext;
- [_nsglContext makeCurrentContext];
-
- _owner->UnlockAGLCntx();
- return 0;
-
-}
-
-WebRtc_Word32 VideoChannelNSOpenGL::GetChannelProperties(float& left,
- float& top,
- float& right,
- float& bottom)
-{
-
- _owner->LockAGLCntx();
-
- left = _startWidth;
- top = _startHeight;
- right = _stopWidth;
- bottom = _stopHeight;
-
- _owner->UnlockAGLCntx();
- return 0;
-}
-
-WebRtc_Word32 VideoChannelNSOpenGL::RenderFrame(const WebRtc_UWord32 /*streamId*/, VideoFrame& videoFrame)
-{
-
- _owner->LockAGLCntx();
-
- if(_width != (int)videoFrame.Width() ||
- _height != (int)videoFrame.Height())
- {
- if(FrameSizeChange(videoFrame.Width(), videoFrame.Height(), 1) == -1)
- {
- _owner->UnlockAGLCntx();
- return -1;
- }
- }
-
- int ret = DeliverFrame(videoFrame.Buffer(), videoFrame.Length(), videoFrame.TimeStamp());
-
- _owner->UnlockAGLCntx();
- return ret;
-}
-
-int VideoChannelNSOpenGL::UpdateSize(int width, int height)
-{
- _owner->LockAGLCntx();
- _width = width;
- _height = height;
- _owner->UnlockAGLCntx();
- return 0;
-}
-
-int VideoChannelNSOpenGL::UpdateStretchSize(int stretchHeight, int stretchWidth)
-{
-
- _owner->LockAGLCntx();
- _stretchedHeight = stretchHeight;
- _stretchedWidth = stretchWidth;
- _owner->UnlockAGLCntx();
- return 0;
-}
-
-int VideoChannelNSOpenGL::FrameSizeChange(int width, int height, int numberOfStreams)
-{
- // We got a new frame size from VideoAPI, prepare the buffer
-
- _owner->LockAGLCntx();
-
- if (width == _width && _height == height)
- {
- // We already have a correct buffer size
- _numberOfStreams = numberOfStreams;
- _owner->UnlockAGLCntx();
- return 0;
- }
-
- _width = width;
- _height = height;
-
- // Delete the old buffer, create a new one with correct size.
- if (_buffer)
- {
- delete [] _buffer;
- _bufferSize = 0;
- }
-
- _incommingBufferSize = CalcBufferSize(kI420, _width, _height);
- _bufferSize = CalcBufferSize(kARGB, _width, _height);//_width * _height * bytesPerPixel;
- _buffer = new unsigned char [_bufferSize];
- memset(_buffer, 0, _bufferSize * sizeof(unsigned char));
-
- [_nsglContext makeCurrentContext];
-
- if(glIsTexture(_texture))
- {
- glDeleteTextures(1, (const GLuint*) &_texture);
- _texture = 0;
- }
-
- // Create a new texture
- glGenTextures(1, (GLuint *) &_texture);
-
- GLenum glErr = glGetError();
-
- if (glErr != GL_NO_ERROR)
- {
-
- }
-
- glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture);
-
- GLint texSize;
- glGetIntegerv(GL_MAX_TEXTURE_SIZE, &texSize);
-
- if (texSize < _width || texSize < _height)
- {
- _owner->UnlockAGLCntx();
- return -1;
- }
-
- // Set up th texture type and size
- glTexImage2D(GL_TEXTURE_RECTANGLE_EXT, // target
- 0, // level
- GL_RGBA, // internal format
- _width, // width
- _height, // height
- 0, // border 0/1 = off/on
- _pixelFormat, // format, GL_RGBA
- _pixelDataType, // data type, GL_UNSIGNED_INT_8_8_8_8
- _buffer); // pixel data
-
- glErr = glGetError();
- if (glErr != GL_NO_ERROR)
- {
- _owner->UnlockAGLCntx();
- return -1;
- }
-
- _owner->UnlockAGLCntx();
- return 0;
-}
-
-int VideoChannelNSOpenGL::DeliverFrame(unsigned char* buffer, int bufferSize, unsigned int /*timeStamp90kHz*/)
-{
-
- _owner->LockAGLCntx();
-
- if (_texture == 0)
- {
- _owner->UnlockAGLCntx();
- return 0;
- }
-
- if (bufferSize != _incommingBufferSize)
- {
- _owner->UnlockAGLCntx();
- return -1;
- }
-
- int rgbRet = ConvertFromYV12(buffer, _width,
- kBGRA, 0, _width, _height,
- _buffer);
- if (rgbRet < 0)
- {
- _owner->UnlockAGLCntx();
- return -1;
- }
-
- [_nsglContext makeCurrentContext];
-
-
- glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture); // Make sure this texture is the active one
- GLenum glErr = glGetError();
- if (glErr != GL_NO_ERROR)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "ERROR %d while calling glBindTexture", glErr);
- _owner->UnlockAGLCntx();
- return -1;
- }
-
- glTexSubImage2D(GL_TEXTURE_RECTANGLE_EXT,
- 0, // Level, not use
- 0, // start point x, (low left of pic)
- 0, // start point y,
- _width, // width
- _height, // height
- _pixelFormat, // pictue format for _buffer
- _pixelDataType, // data type of _buffer
- (const GLvoid*) _buffer); // the pixel data
-
- glErr = glGetError();
- if (glErr != GL_NO_ERROR)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "ERROR %d while calling glTexSubImage2d", glErr);
- _owner->UnlockAGLCntx();
- return -1;
- }
-
- _bufferIsUpdated = true;
-
- _owner->UnlockAGLCntx();
- return 0;
-}
-
-int VideoChannelNSOpenGL::RenderOffScreenBuffer()
-{
-
- _owner->LockAGLCntx();
-
- if (_texture == 0)
- {
- _owner->UnlockAGLCntx();
- return 0;
- }
-
- // if(_fullscreen)
- // {
- // NSRect mainDisplayRect = [[NSScreen mainScreen] frame];
- // _width = mainDisplayRect.size.width;
- // _height = mainDisplayRect.size.height;
- // glViewport(0, 0, mainDisplayRect.size.width, mainDisplayRect.size.height);
- // float newX = mainDisplayRect.size.width/_width;
- // float newY = mainDisplayRect.size.height/_height;
-
- // convert from 0.0 <= size <= 1.0 to
- // open gl world -1.0 < size < 1.0
- GLfloat xStart = 2.0f * _startWidth - 1.0f;
- GLfloat xStop = 2.0f * _stopWidth - 1.0f;
- GLfloat yStart = 1.0f - 2.0f * _stopHeight;
- GLfloat yStop = 1.0f - 2.0f * _startHeight;
-
- [_nsglContext makeCurrentContext];
-
- glBindTexture(GL_TEXTURE_RECTANGLE_EXT, _texture);
- _oldStretchedHeight = _stretchedHeight;
- _oldStretchedWidth = _stretchedWidth;
-
- glLoadIdentity();
- glEnable(GL_TEXTURE_RECTANGLE_EXT);
- glBegin(GL_POLYGON);
- {
- glTexCoord2f(0.0, 0.0); glVertex2f(xStart, yStop);
- glTexCoord2f(_width, 0.0); glVertex2f(xStop, yStop);
- glTexCoord2f(_width, _height); glVertex2f(xStop, yStart);
- glTexCoord2f(0.0, _height); glVertex2f(xStart, yStart);
- }
- glEnd();
-
- glDisable(GL_TEXTURE_RECTANGLE_EXT);
-
- _bufferIsUpdated = false;
-
- _owner->UnlockAGLCntx();
- return 0;
-}
-
-int VideoChannelNSOpenGL::IsUpdated(bool& isUpdated)
-{
- _owner->LockAGLCntx();
-
- isUpdated = _bufferIsUpdated;
-
- _owner->UnlockAGLCntx();
- return 0;
-}
-
-int VideoChannelNSOpenGL::SetStreamSettings(int /*streamId*/, float startWidth, float startHeight, float stopWidth, float stopHeight)
-{
- _owner->LockAGLCntx();
-
- _startWidth = startWidth;
- _stopWidth = stopWidth;
- _startHeight = startHeight;
- _stopHeight = stopHeight;
-
- int oldWidth = _width;
- int oldHeight = _height;
- int oldNumberOfStreams = _numberOfStreams;
-
- _width = 0;
- _height = 0;
-
- int retVal = FrameSizeChange(oldWidth, oldHeight, oldNumberOfStreams);
-
- _owner->UnlockAGLCntx();
- return retVal;
-}
-
-int VideoChannelNSOpenGL::SetStreamCropSettings(int /*streamId*/, float /*startWidth*/, float /*startHeight*/, float /*stopWidth*/, float /*stopHeight*/)
-{
- return -1;
-}
-
-/*
- *
- * VideoRenderNSOpenGL
- *
- */
-
-VideoRenderNSOpenGL::VideoRenderNSOpenGL(CocoaRenderView *windowRef, bool fullScreen, int iId) :
-_windowRef( (CocoaRenderView*)windowRef),
-_fullScreen( fullScreen),
-_id( iId),
-_nsglContextCritSec( *CriticalSectionWrapper::CreateCriticalSection()),
-_screenUpdateThread( 0),
-_screenUpdateEvent( 0),
-_nsglContext( 0),
-_nsglFullScreenContext( 0),
-_fullScreenWindow( nil),
-_windowRect( ),
-_windowWidth( 0),
-_windowHeight( 0),
-_nsglChannels( ),
-_zOrderToChannel( ),
-_threadID (0),
-_renderingIsPaused (FALSE),
-_windowRefSuperView(NULL),
-_windowRefSuperViewFrame(NSMakeRect(0,0,0,0))
-{
- _screenUpdateThread = ThreadWrapper::CreateThread(ScreenUpdateThreadProc, this, kRealtimePriority);
- _screenUpdateEvent = EventWrapper::Create();
-}
-
-int VideoRenderNSOpenGL::ChangeWindow(CocoaRenderView* newWindowRef)
-{
-
- LockAGLCntx();
-
- _windowRef = newWindowRef;
-
- if(CreateMixingContext() == -1)
- {
- UnlockAGLCntx();
- return -1;
- }
-
- int error = 0;
- std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.begin();
- while (it!= _nsglChannels.end())
- {
- error |= (it->second)->ChangeContext(_nsglContext);
- it++;
- }
- if(error != 0)
- {
- UnlockAGLCntx();
- return -1;
- }
-
- UnlockAGLCntx();
- return 0;
-}
-
-/* Check if the thread and event already exist.
- * If so then they will simply be restarted
- * If not then create them and continue
- */
-WebRtc_Word32 VideoRenderNSOpenGL::StartRender()
-{
-
- LockAGLCntx();
-
- const unsigned int MONITOR_FREQ = 60;
- if(TRUE == _renderingIsPaused)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "Restarting screenUpdateThread");
-
- // we already have the thread. Most likely StopRender() was called and they were paused
- if(FALSE == _screenUpdateThread->Start(_threadID) ||
- FALSE == _screenUpdateEvent->StartTimer(true, 1000/MONITOR_FREQ))
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "Failed to restart screenUpdateThread or screenUpdateEvent");
- UnlockAGLCntx();
- return -1;
- }
-
- UnlockAGLCntx();
- return 0;
- }
-
-
- if (!_screenUpdateThread)
- {
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "failed start screenUpdateThread");
- UnlockAGLCntx();
- return -1;
- }
-
-
- UnlockAGLCntx();
- return 0;
-}
-WebRtc_Word32 VideoRenderNSOpenGL::StopRender()
-{
-
- LockAGLCntx();
-
- /* The code below is functional
- * but it pauses for several seconds
- */
-
- // pause the update thread and the event timer
- if(!_screenUpdateThread || !_screenUpdateEvent)
- {
- _renderingIsPaused = TRUE;
-
- UnlockAGLCntx();
- return 0;
- }
-
- if(FALSE == _screenUpdateThread->Stop() || FALSE == _screenUpdateEvent->StopTimer())
- {
- _renderingIsPaused = FALSE;
-
- UnlockAGLCntx();
- return -1;
- }
-
- _renderingIsPaused = TRUE;
-
- UnlockAGLCntx();
- return 0;
-}
-
-int VideoRenderNSOpenGL::configureNSOpenGLView()
-{
- return 0;
-
-}
-
-int VideoRenderNSOpenGL::configureNSOpenGLEngine()
-{
-
- LockAGLCntx();
-
- // Disable not needed functionality to increase performance
- glDisable(GL_DITHER);
- glDisable(GL_ALPHA_TEST);
- glDisable(GL_STENCIL_TEST);
- glDisable(GL_FOG);
- glDisable(GL_TEXTURE_2D);
- glPixelZoom(1.0, 1.0);
- glDisable(GL_BLEND);
- glDisable(GL_DEPTH_TEST);
- glDepthMask(GL_FALSE);
- glDisable(GL_CULL_FACE);
-
- // Set texture parameters
- glTexParameterf(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_PRIORITY, 1.0);
- glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_WRAP_S, GL_CLAMP_TO_EDGE);
- glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_WRAP_T, GL_CLAMP_TO_EDGE);
- glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MAG_FILTER, GL_LINEAR);
- glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_MIN_FILTER, GL_LINEAR);
- glTexEnvf(GL_TEXTURE_ENV, GL_TEXTURE_ENV_MODE, GL_MODULATE);
- glPixelStorei(GL_UNPACK_ALIGNMENT, 1);
- glTexParameteri(GL_TEXTURE_RECTANGLE_EXT, GL_TEXTURE_STORAGE_HINT_APPLE, GL_STORAGE_SHARED_APPLE);
-
- if (GetWindowRect(_windowRect) == -1)
- {
- UnlockAGLCntx();
- return true;
- }
-
- if (_windowWidth != (_windowRect.right - _windowRect.left)
- || _windowHeight != (_windowRect.bottom - _windowRect.top))
- {
- _windowWidth = _windowRect.right - _windowRect.left;
- _windowHeight = _windowRect.bottom - _windowRect.top;
- }
- glViewport(0, 0, _windowWidth, _windowHeight);
-
- // Synchronize buffer swaps with vertical refresh rate
- GLint swapInt = 1;
- [_nsglContext setValues:&swapInt forParameter:NSOpenGLCPSwapInterval];
-
- UnlockAGLCntx();
- return 0;
-}
-
-int VideoRenderNSOpenGL::setRenderTargetWindow()
-{
- LockAGLCntx();
-
-
- GLuint attribs[] =
- {
- NSOpenGLPFAColorSize, 24,
- NSOpenGLPFAAlphaSize, 8,
- NSOpenGLPFADepthSize, 16,
- NSOpenGLPFAAccelerated,
- 0
- };
-
- NSOpenGLPixelFormat* fmt = [[NSOpenGLPixelFormat alloc] initWithAttributes: (NSOpenGLPixelFormatAttribute*) attribs];
-
- if(_windowRef)
- {
- [_windowRef initCocoaRenderView:fmt];
- }
- else
- {
- UnlockAGLCntx();
- return -1;
- }
-
- [fmt release];
-
- _nsglContext = [_windowRef nsOpenGLContext];
- [_nsglContext makeCurrentContext];
-
- glClearColor(0.0f, 0.0f, 0.0f, 1.0f);
- glClear(GL_COLOR_BUFFER_BIT);
-
-
- DisplayBuffers();
-
- UnlockAGLCntx();
- return 0;
-}
-
-int VideoRenderNSOpenGL::setRenderTargetFullScreen()
-{
- LockAGLCntx();
-
-
- GLuint attribs[] =
- {
- NSOpenGLPFAColorSize, 24,
- NSOpenGLPFAAlphaSize, 8,
- NSOpenGLPFADepthSize, 16,
- NSOpenGLPFAAccelerated,
- 0
- };
-
- NSOpenGLPixelFormat* fmt = [[NSOpenGLPixelFormat alloc] initWithAttributes: (NSOpenGLPixelFormatAttribute*) attribs];
-
- // Store original superview and frame for use when exiting full screens
- _windowRefSuperViewFrame = [_windowRef frame];
- _windowRefSuperView = [_windowRef superview];
-
-
- // create new fullscreen window
- NSRect screenRect = [[NSScreen mainScreen]frame];
- [_windowRef setFrame:screenRect];
- [_windowRef setBounds:screenRect];
-
-
- _fullScreenWindow = [[CocoaFullScreenWindow alloc]init];
- [_fullScreenWindow grabFullScreen];
- [[[_fullScreenWindow window] contentView] addSubview:_windowRef];
-
- if(_windowRef)
- {
- [_windowRef initCocoaRenderViewFullScreen:fmt];
- }
- else
- {
- UnlockAGLCntx();
- return -1;
- }
-
- [fmt release];
-
- _nsglContext = [_windowRef nsOpenGLContext];
- [_nsglContext makeCurrentContext];
-
- glClearColor(0.0f, 0.0f, 0.0f, 1.0f);
- glClear(GL_COLOR_BUFFER_BIT);
-
- DisplayBuffers();
-
- UnlockAGLCntx();
- return 0;
-}
-
-VideoRenderNSOpenGL::~VideoRenderNSOpenGL()
-{
-
- if(_fullScreen)
- {
- if(_fullScreenWindow)
- {
- // Detach CocoaRenderView from full screen view back to
- // it's original parent.
- [_windowRef removeFromSuperview];
- if(_windowRefSuperView)
- {
- [_windowRefSuperView addSubview:_windowRef];
- [_windowRef setFrame:_windowRefSuperViewFrame];
- }
-
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, 0, "%s:%d Attempting to release fullscreen window", __FUNCTION__, __LINE__);
- [_fullScreenWindow releaseFullScreen];
-
- }
- }
-
- // Signal event to exit thread, then delete it
- ThreadWrapper* tmpPtr = _screenUpdateThread;
- _screenUpdateThread = NULL;
-
- if (tmpPtr)
- {
- tmpPtr->SetNotAlive();
- _screenUpdateEvent->Set();
- _screenUpdateEvent->StopTimer();
-
- if (tmpPtr->Stop())
- {
- delete tmpPtr;
- }
- delete _screenUpdateEvent;
- _screenUpdateEvent = NULL;
- }
-
- if (_nsglContext != 0)
- {
- [_nsglContext makeCurrentContext];
- _nsglContext = nil;
- }
-
- // Delete all channels
- std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.begin();
- while (it!= _nsglChannels.end())
- {
- delete it->second;
- _nsglChannels.erase(it);
- it = _nsglChannels.begin();
- }
- _nsglChannels.clear();
-
- // Clean the zOrder map
- std::multimap<int, int>::iterator zIt = _zOrderToChannel.begin();
- while(zIt != _zOrderToChannel.end())
- {
- _zOrderToChannel.erase(zIt);
- zIt = _zOrderToChannel.begin();
- }
- _zOrderToChannel.clear();
-
-}
-
-/* static */
-int VideoRenderNSOpenGL::GetOpenGLVersion(int& /*nsglMajor*/, int& /*nsglMinor*/)
-{
- return -1;
-}
-
-int VideoRenderNSOpenGL::Init()
-{
-
- LockAGLCntx();
- if (!_screenUpdateThread)
- {
- UnlockAGLCntx();
- return -1;
- }
-
- _screenUpdateThread->Start(_threadID);
-
- // Start the event triggering the render process
- unsigned int monitorFreq = 60;
- _screenUpdateEvent->StartTimer(true, 1000/monitorFreq);
-
- if (CreateMixingContext() == -1)
- {
- UnlockAGLCntx();
- return -1;
- }
-
- UnlockAGLCntx();
- return 0;
-}
-
-VideoChannelNSOpenGL* VideoRenderNSOpenGL::CreateNSGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight)
-{
- CriticalSectionScoped cs(_nsglContextCritSec);
-
- if (HasChannel(channel))
- {
- return NULL;
- }
-
- if (_zOrderToChannel.find(zOrder) != _zOrderToChannel.end())
- {
-
- }
-
- VideoChannelNSOpenGL* newAGLChannel = new VideoChannelNSOpenGL(_nsglContext, _id, this);
- if (newAGLChannel->SetStreamSettings(0, startWidth, startHeight, stopWidth, stopHeight) == -1)
- {
- if (newAGLChannel)
- {
- delete newAGLChannel;
- newAGLChannel = NULL;
- }
-
- return NULL;
- }
-
- _nsglChannels[channel] = newAGLChannel;
- _zOrderToChannel.insert(std::pair<int, int>(zOrder, channel));
-
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s successfully created NSGL channel number %d", __FUNCTION__, channel);
-
- return newAGLChannel;
-}
-
-int VideoRenderNSOpenGL::DeleteAllNSGLChannels()
-{
-
- CriticalSectionScoped cs(_nsglContextCritSec);
-
- std::map<int, VideoChannelNSOpenGL*>::iterator it;
- it = _nsglChannels.begin();
-
- while (it != _nsglChannels.end())
- {
- VideoChannelNSOpenGL* channel = it->second;
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s Deleting channel %d", __FUNCTION__, channel);
- delete channel;
- it++;
- }
- _nsglChannels.clear();
- return 0;
-}
-
-WebRtc_Word32 VideoRenderNSOpenGL::DeleteNSGLChannel(const WebRtc_UWord32 channel)
-{
-
- CriticalSectionScoped cs(_nsglContextCritSec);
-
- std::map<int, VideoChannelNSOpenGL*>::iterator it;
- it = _nsglChannels.find(channel);
- if (it != _nsglChannels.end())
- {
- delete it->second;
- _nsglChannels.erase(it);
- }
- else
- {
- return -1;
- }
-
- std::multimap<int, int>::iterator zIt = _zOrderToChannel.begin();
- while( zIt != _zOrderToChannel.end())
- {
- if (zIt->second == (int)channel)
- {
- _zOrderToChannel.erase(zIt);
- break;
- }
- zIt++;
- }
-
- return 0;
-}
-
-WebRtc_Word32 VideoRenderNSOpenGL::GetChannelProperties(const WebRtc_UWord16 streamId,
- WebRtc_UWord32& zOrder,
- float& left,
- float& top,
- float& right,
- float& bottom)
-{
-
- CriticalSectionScoped cs(_nsglContextCritSec);
-
- bool channelFound = false;
-
- // Loop through all channels until we find a match.
- // From that, get zorder.
- // From that, get T, L, R, B
- for (std::multimap<int, int>::reverse_iterator rIt = _zOrderToChannel.rbegin();
- rIt != _zOrderToChannel.rend();
- rIt++)
- {
- if(streamId == rIt->second)
- {
- channelFound = true;
-
- zOrder = rIt->second;
-
- std::map<int, VideoChannelNSOpenGL*>::iterator rIt = _nsglChannels.find(streamId);
- VideoChannelNSOpenGL* tempChannel = rIt->second;
-
- if(-1 == tempChannel->GetChannelProperties(left, top, right, bottom) )
- {
- return -1;
- }
- break;
- }
- }
-
- if(false == channelFound)
- {
-
- return -1;
- }
-
- return 0;
-}
-
-int VideoRenderNSOpenGL::StopThread()
-{
-
- ThreadWrapper* tmpPtr = _screenUpdateThread;
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s Stopping thread ", __FUNCTION__, _screenUpdateThread);
- _screenUpdateThread = NULL;
-
- if (tmpPtr)
- {
- tmpPtr->SetNotAlive();
- _screenUpdateEvent->Set();
- if (tmpPtr->Stop())
- {
- delete tmpPtr;
- }
- }
-
- delete _screenUpdateEvent;
- _screenUpdateEvent = NULL;
-
- return 0;
-}
-
-bool VideoRenderNSOpenGL::IsFullScreen()
-{
-
- CriticalSectionScoped cs(_nsglContextCritSec);
- return _fullScreen;
-}
-
-bool VideoRenderNSOpenGL::HasChannels()
-{
- CriticalSectionScoped cs(_nsglContextCritSec);
-
- if (_nsglChannels.begin() != _nsglChannels.end())
- {
- return true;
- }
- return false;
-}
-
-bool VideoRenderNSOpenGL::HasChannel(int channel)
-{
-
- CriticalSectionScoped cs(_nsglContextCritSec);
-
- std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.find(channel);
-
- if (it != _nsglChannels.end())
- {
- return true;
- }
- return false;
-}
-
-int VideoRenderNSOpenGL::GetChannels(std::list<int>& channelList)
-{
-
- CriticalSectionScoped cs(_nsglContextCritSec);
-
- std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.begin();
-
- while (it != _nsglChannels.end())
- {
- channelList.push_back(it->first);
- it++;
- }
-
- return 0;
-}
-
-VideoChannelNSOpenGL* VideoRenderNSOpenGL::ConfigureNSGLChannel(int channel, int zOrder, float startWidth, float startHeight, float stopWidth, float stopHeight)
-{
-
- CriticalSectionScoped cs(_nsglContextCritSec);
-
- std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.find(channel);
-
- if (it != _nsglChannels.end())
- {
- VideoChannelNSOpenGL* aglChannel = it->second;
- if (aglChannel->SetStreamSettings(0, startWidth, startHeight, stopWidth, stopHeight) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id, "%s failed to set stream settings: channel %d. channel=%d zOrder=%d startWidth=%d startHeight=%d stopWidth=%d stopHeight=%d",
- __FUNCTION__, channel, zOrder, startWidth, startHeight, stopWidth, stopHeight);
- return NULL;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, _id, "%s Configuring channel %d. channel=%d zOrder=%d startWidth=%d startHeight=%d stopWidth=%d stopHeight=%d",
- __FUNCTION__, channel, zOrder, startWidth, startHeight, stopWidth, stopHeight);
-
- std::multimap<int, int>::iterator it = _zOrderToChannel.begin();
- while(it != _zOrderToChannel.end())
- {
- if (it->second == channel)
- {
- if (it->first != zOrder)
- {
- _zOrderToChannel.erase(it);
- _zOrderToChannel.insert(std::pair<int, int>(zOrder, channel));
- }
- break;
- }
- it++;
- }
- return aglChannel;
- }
-
- return NULL;
-}
-
-/*
- *
- * Rendering process
- *
- */
-
-bool VideoRenderNSOpenGL::ScreenUpdateThreadProc(void* obj)
-{
- return static_cast<VideoRenderNSOpenGL*>(obj)->ScreenUpdateProcess();
-}
-
-bool VideoRenderNSOpenGL::ScreenUpdateProcess()
-{
-
- _screenUpdateEvent->Wait(10);
- LockAGLCntx();
-
- if (!_screenUpdateThread)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, _id, "%s no screen update thread", __FUNCTION__);
- UnlockAGLCntx();
- return false;
- }
-
- [_nsglContext makeCurrentContext];
-
- if (GetWindowRect(_windowRect) == -1)
- {
- UnlockAGLCntx();
- return true;
- }
-
- if (_windowWidth != (_windowRect.right - _windowRect.left)
- || _windowHeight != (_windowRect.bottom - _windowRect.top))
- {
- _windowWidth = _windowRect.right - _windowRect.left;
- _windowHeight = _windowRect.bottom - _windowRect.top;
- glViewport(0, 0, _windowWidth, _windowHeight);
- }
-
- // Check if there are any updated buffers
- bool updated = false;
- std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.begin();
- while (it != _nsglChannels.end())
- {
-
- VideoChannelNSOpenGL* aglChannel = it->second;
- aglChannel->UpdateStretchSize(_windowHeight, _windowWidth);
- aglChannel->IsUpdated(updated);
- if (updated)
- {
- break;
- }
- it++;
- }
-
- if (updated)
- {
-
- // At least on buffers is updated, we need to repaint the texture
- if (RenderOffScreenBuffers() != -1)
- {
- UnlockAGLCntx();
- return true;
- }
- }
- // }
- UnlockAGLCntx();
- return true;
-}
-
-/*
- *
- * Functions for creating mixing buffers and screen settings
- *
- */
-
-int VideoRenderNSOpenGL::CreateMixingContext()
-{
-
- CriticalSectionScoped cs(_nsglContextCritSec);
-
- if(_fullScreen)
- {
- if(-1 == setRenderTargetFullScreen())
- {
- return -1;
- }
- }
- else
- {
-
- if(-1 == setRenderTargetWindow())
- {
- return -1;
- }
- }
-
- configureNSOpenGLEngine();
-
- DisplayBuffers();
-
- GLenum glErr = glGetError();
- if (glErr)
- {
- }
-
- return 0;
-}
-
-/*
- *
- * Rendering functions
- *
- */
-
-int VideoRenderNSOpenGL::RenderOffScreenBuffers()
-{
- LockAGLCntx();
-
- // Get the current window size, it might have changed since last render.
- if (GetWindowRect(_windowRect) == -1)
- {
- UnlockAGLCntx();
- return -1;
- }
-
- [_nsglContext makeCurrentContext];
- glClear(GL_COLOR_BUFFER_BIT);
-
- // Loop through all channels starting highest zOrder ending with lowest.
- for (std::multimap<int, int>::reverse_iterator rIt = _zOrderToChannel.rbegin();
- rIt != _zOrderToChannel.rend();
- rIt++)
- {
- int channelId = rIt->second;
- std::map<int, VideoChannelNSOpenGL*>::iterator it = _nsglChannels.find(channelId);
-
- VideoChannelNSOpenGL* aglChannel = it->second;
-
- aglChannel->RenderOffScreenBuffer();
- }
-
- DisplayBuffers();
-
- UnlockAGLCntx();
- return 0;
-}
-
-/*
- *
- * Help functions
- *
- * All help functions assumes external protections
- *
- */
-
-int VideoRenderNSOpenGL::DisplayBuffers()
-{
-
- LockAGLCntx();
-
- glFinish();
- [_nsglContext flushBuffer];
-
- WEBRTC_TRACE(kTraceDebug, kTraceVideoRenderer, _id, "%s glFinish and [_nsglContext flushBuffer]", __FUNCTION__);
-
- UnlockAGLCntx();
- return 0;
-}
-
-int VideoRenderNSOpenGL::GetWindowRect(Rect& rect)
-{
-
- CriticalSectionScoped cs(_nsglContextCritSec);
-
- if (_windowRef)
- {
- if(_fullScreen)
- {
- NSRect mainDisplayRect = [[NSScreen mainScreen] frame];
- rect.bottom = 0;
- rect.left = 0;
- rect.right = mainDisplayRect.size.width;
- rect.top = mainDisplayRect.size.height;
- }
- else
- {
- rect.top = [_windowRef frame].origin.y;
- rect.left = [_windowRef frame].origin.x;
- rect.bottom = [_windowRef frame].origin.y + [_windowRef frame].size.height;
- rect.right = [_windowRef frame].origin.x + [_windowRef frame].size.width;
- }
-
- return 0;
- }
- else
- {
- return -1;
- }
-}
-
-WebRtc_Word32 VideoRenderNSOpenGL::ChangeUniqueID(WebRtc_Word32 id)
-{
-
- CriticalSectionScoped cs(_nsglContextCritSec);
- _id = id;
- return 0;
-}
-
-WebRtc_Word32 VideoRenderNSOpenGL::SetText(const WebRtc_UWord8 /*textId*/,
- const WebRtc_UWord8* /*text*/,
- const WebRtc_Word32 /*textLength*/,
- const WebRtc_UWord32 /*textColorRef*/,
- const WebRtc_UWord32 /*backgroundColorRef*/,
- const float /*left*/,
- const float /*top*/,
- const float /*right*/,
- const float /*bottom*/)
-{
-
- return 0;
-
-}
-
-void VideoRenderNSOpenGL::LockAGLCntx()
-{
- _nsglContextCritSec.Enter();
-}
-void VideoRenderNSOpenGL::UnlockAGLCntx()
-{
- _nsglContextCritSec.Leave();
-}
-
-/*
-
- bool VideoRenderNSOpenGL::SetFullScreen(bool fullscreen)
- {
- NSRect mainDisplayRect, viewRect;
-
- // Create a screen-sized window on the display you want to take over
- // Note, mainDisplayRect has a non-zero origin if the key window is on a secondary display
- mainDisplayRect = [[NSScreen mainScreen] frame];
- fullScreenWindow = [[NSWindow alloc] initWithContentRect:mainDisplayRect styleMask:NSBorderlessWindowMask
- backing:NSBackingStoreBuffered defer:YES];
-
- // Set the window level to be above the menu bar
- [fullScreenWindow setLevel:NSMainMenuWindowLevel+1];
-
- // Perform any other window configuration you desire
- [fullScreenWindow setOpaque:YES];
- [fullScreenWindow setHidesOnDeactivate:YES];
-
- // Create a view with a double-buffered OpenGL context and attach it to the window
- // By specifying the non-fullscreen context as the shareContext, we automatically inherit the OpenGL objects (textures, etc) it has defined
- viewRect = NSMakeRect(0.0, 0.0, mainDisplayRect.size.width, mainDisplayRect.size.height);
- fullScreenView = [[MyOpenGLView alloc] initWithFrame:viewRect shareContext:[openGLView openGLContext]];
- [fullScreenWindow setContentView:fullScreenView];
-
- // Show the window
- [fullScreenWindow makeKeyAndOrderFront:self];
-
- // Set the scene with the full-screen viewport and viewing transformation
- [scene setViewportRect:viewRect];
-
- // Assign the view's MainController to self
- [fullScreenView setMainController:self];
-
- if (!isAnimating) {
- // Mark the view as needing drawing to initalize its contents
- [fullScreenView setNeedsDisplay:YES];
- }
- else {
- // Start playing the animation
- [fullScreenView startAnimation];
- }
-
- }
-
-
-
- */
-
-
-} //namespace webrtc
-
-#endif // COCOA_RENDERING
diff --git a/trunk/src/modules/video_render/main/source/video_render.gypi b/trunk/src/modules/video_render/main/source/video_render.gypi
deleted file mode 100644
index 7a7545e..0000000
--- a/trunk/src/modules/video_render/main/source/video_render.gypi
+++ /dev/null
@@ -1,195 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'video_render_module',
- 'type': '<(library)',
- 'dependencies': [
- 'webrtc_utility',
- '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '.',
- '../interface',
- '../../../interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- '../../../interface',
- ],
- },
- 'sources': [
- # interfaces
- '../interface/video_render.h',
- '../interface/video_render_defines.h',
-
- # headers
- 'incoming_video_stream.h',
- 'video_render_frames.h',
- 'video_render_impl.h',
- 'i_video_render.h',
- # Linux
- 'linux/video_render_linux_impl.h',
- 'linux/video_x11_channel.h',
- 'linux/video_x11_render.h',
- # Mac
- 'mac/cocoa_full_screen_window.h',
- 'mac/cocoa_render_view.h',
- 'mac/video_render_agl.h',
- 'mac/video_render_mac_carbon_impl.h',
- 'mac/video_render_mac_cocoa_impl.h',
- 'mac/video_render_nsopengl.h',
- # Windows
- 'windows/i_video_render_win.h',
- 'windows/video_render_direct3d9.h',
- 'windows/video_render_directdraw.h',
- 'windows/video_render_windows_impl.h',
- # External
- 'external/video_render_external_impl.h',
-
- # PLATFORM INDEPENDENT SOURCE FILES
- 'incoming_video_stream.cc',
- 'video_render_frames.cc',
- 'video_render_impl.cc',
- # PLATFORM SPECIFIC SOURCE FILES - Will be filtered below
- # Linux
- 'linux/video_render_linux_impl.cc',
- 'linux/video_x11_channel.cc',
- 'linux/video_x11_render.cc',
- # Mac
- 'mac/video_render_nsopengl.mm',
- 'mac/video_render_mac_cocoa_impl.mm',
- 'mac/video_render_agl.cc',
- 'mac/video_render_mac_carbon_impl.cc',
- 'mac/cocoa_render_view.mm',
- 'mac/cocoa_full_screen_window.mm',
- # Windows
- 'windows/video_render_direct3d9.cc',
- 'windows/video_render_directdraw.cc',
- 'windows/video_render_windows_impl.cc',
- # External
- 'external/video_render_external_impl.cc',
- ],
- # TODO(andrew): with the proper suffix, these files will be excluded
- # automatically.
- 'conditions': [
- ['include_internal_video_render==1', {
- 'defines': [
- 'WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER',
- ],
- }],
- ['OS!="linux" or include_internal_video_render==0', {
- 'sources!': [
- 'linux/video_render_linux_impl.h',
- 'linux/video_x11_channel.h',
- 'linux/video_x11_render.h',
- 'linux/video_render_linux_impl.cc',
- 'linux/video_x11_channel.cc',
- 'linux/video_x11_render.cc',
- ],
- }],
- ['OS!="mac" or include_internal_video_render==0', {
- 'sources!': [
- 'mac/cocoa_full_screen_window.h',
- 'mac/cocoa_render_view.h',
- 'mac/video_render_agl.h',
- 'mac/video_render_mac_carbon_impl.h',
- 'mac/video_render_mac_cocoa_impl.h',
- 'mac/video_render_nsopengl.h',
- 'mac/video_render_nsopengl.mm',
- 'mac/video_render_mac_cocoa_impl.mm',
- 'mac/video_render_agl.cc',
- 'mac/video_render_mac_carbon_impl.cc',
- 'mac/cocoa_render_view.mm',
- 'mac/cocoa_full_screen_window.mm',
- ],
- }],
- ['OS=="mac"', {
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'mac',
- ],
- },
- }],
- ['OS!="win" or include_internal_video_render==0', {
- 'sources!': [
- 'windows/i_video_render_win.h',
- 'windows/video_render_direct3d9.h',
- 'windows/video_render_directdraw.h',
- 'windows/video_render_windows_impl.h',
- 'windows/video_render_direct3d9.cc',
- 'windows/video_render_directdraw.cc',
- 'windows/video_render_windows_impl.cc',
- ],
- }],
- ] # conditions
- }, # video_render_module
- ], # targets
-
- # Exclude the test target when building with chromium.
- 'conditions': [
- ['include_internal_video_render==1', {
- 'defines': [
- 'WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER',
- ],
- }],
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'video_render_module_test',
- 'type': 'executable',
- 'dependencies': [
- 'video_render_module',
- 'webrtc_utility',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
- ],
- 'sources': [
- '../test/testAPI/testAPI.cc',
- '../test/testAPI/testAPI.h',
- '../test/testAPI/testAPI_mac.mm',
- ],
- 'conditions': [
- ['OS=="mac" or OS=="linux"', {
- 'cflags': [
- '-Wno-write-strings',
- ],
- 'ldflags': [
- '-lpthread -lm',
- ],
- }],
- ['OS=="linux"', {
- 'libraries': [
- '-lrt',
- '-lXext',
- '-lX11',
- ],
- }],
- ['OS=="mac"', {
- 'xcode_settings': {
- 'OTHER_LDFLAGS': [
- '-framework Foundation -framework AppKit -framework Cocoa -framework OpenGL',
- ],
- },
- }],
- ] # conditions
- }, # video_render_module_test
- ], # targets
- }], # build_with_chromium==0
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/modules/video_render/main/source/video_render_frames.cc b/trunk/src/modules/video_render/main/source/video_render_frames.cc
deleted file mode 100644
index 28078e3..0000000
--- a/trunk/src/modules/video_render/main/source/video_render_frames.cc
+++ /dev/null
@@ -1,209 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_render_frames.h"
-#include "module_common_types.h"
-#include "tick_util.h"
-#include "trace.h"
-#include <cassert>
-
-namespace webrtc {
-
-VideoRenderFrames::VideoRenderFrames() :
- _incomingFrames(), _renderDelayMs(10)
-{
-}
-
-VideoRenderFrames::~VideoRenderFrames()
-{
- ReleaseAllFrames();
-}
-
-WebRtc_Word32 VideoRenderFrames::AddFrame(VideoFrame* ptrNewFrame)
-{
- const WebRtc_Word64 timeNow = TickTime::MillisecondTimestamp();
-
- if (ptrNewFrame->RenderTimeMs() + KOldRenderTimestampMS < timeNow)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, -1,
- "%s: too old frame.", __FUNCTION__);
- return -1;
- }
- if (ptrNewFrame->RenderTimeMs() > timeNow + KFutureRenderTimestampMS)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer, -1,
- "%s: frame too long into the future.", __FUNCTION__);
- return -1;
- }
-
- // Get an empty frame
- VideoFrame* ptrFrameToAdd = NULL;
- if (!_emptyFrames.Empty())
- {
- ListItem* item = _emptyFrames.First();
- if (item)
- {
- ptrFrameToAdd = static_cast<VideoFrame*> (item->GetItem());
- _emptyFrames.Erase(item);
- }
- }
- if (!ptrFrameToAdd)
- {
-
- if (_emptyFrames.GetSize() + _incomingFrames.GetSize()
- > KMaxNumberOfFrames)
- {
- // Already allocated toom many frames...
- WEBRTC_TRACE(kTraceWarning, kTraceVideoRenderer,
- -1, "%s: too many frames, limit: %d", __FUNCTION__,
- KMaxNumberOfFrames);
- return -1;
- }
-
- // Allocate new memory
- WEBRTC_TRACE(kTraceMemory, kTraceVideoRenderer, -1,
- "%s: allocating buffer %d", __FUNCTION__,
- _emptyFrames.GetSize() + _incomingFrames.GetSize());
-
- ptrFrameToAdd = new VideoFrame();
- if (!ptrFrameToAdd)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
- "%s: could not create new frame for", __FUNCTION__);
- return -1;
- }
- }
-
- ptrFrameToAdd->VerifyAndAllocate(ptrNewFrame->Length());
- ptrFrameToAdd->SwapFrame(const_cast<VideoFrame&> (*ptrNewFrame)); //remove const ness. Copying will be costly.
- _incomingFrames.PushBack(ptrFrameToAdd);
-
- return _incomingFrames.GetSize();
-}
-
-VideoFrame*
-VideoRenderFrames::FrameToRender()
-{
- VideoFrame* ptrRenderFrame = NULL;
- while (!_incomingFrames.Empty())
- {
- ListItem* item = _incomingFrames.First();
- if (item)
- {
- VideoFrame* ptrOldestFrameInList =
- static_cast<VideoFrame*> (item->GetItem());
- if (ptrOldestFrameInList->RenderTimeMs()
- <= TickTime::MillisecondTimestamp() + _renderDelayMs)
- {
- // This is the oldest one so far and it's ok to render
- if (ptrRenderFrame)
- {
- // This one is older than the newly found frame, remove this one.
- ptrRenderFrame->SetWidth(0);
- ptrRenderFrame->SetHeight(0);
- ptrRenderFrame->SetLength(0);
- ptrRenderFrame->SetRenderTime(0);
- ptrRenderFrame->SetTimeStamp(0);
- _emptyFrames.PushFront(ptrRenderFrame);
- }
- ptrRenderFrame = ptrOldestFrameInList;
- _incomingFrames.Erase(item);
- }
- else
- {
- // We can't release this one yet, we're done here.
- break;
- }
- }
- else
- {
- assert(false);
- }
- }
- return ptrRenderFrame;
-}
-
-WebRtc_Word32 VideoRenderFrames::ReturnFrame(VideoFrame* ptrOldFrame)
-{
- ptrOldFrame->SetWidth(0);
- ptrOldFrame->SetHeight(0);
- ptrOldFrame->SetRenderTime(0);
- ptrOldFrame->SetLength(0);
- _emptyFrames.PushBack(ptrOldFrame);
-
- return 0;
-}
-
-WebRtc_Word32 VideoRenderFrames::ReleaseAllFrames()
-{
- while (!_incomingFrames.Empty())
- {
- ListItem* item = _incomingFrames.First();
- if (item)
- {
- VideoFrame* ptrFrame =
- static_cast<VideoFrame*> (item->GetItem());
- assert(ptrFrame != NULL);
- ptrFrame->Free();
- delete ptrFrame;
- }
- _incomingFrames.Erase(item);
- }
- while (!_emptyFrames.Empty())
- {
- ListItem* item = _emptyFrames.First();
- if (item)
- {
- VideoFrame* ptrFrame =
- static_cast<VideoFrame*> (item->GetItem());
- assert(ptrFrame != NULL);
- ptrFrame->Free();
- delete ptrFrame;
- }
- _emptyFrames.Erase(item);
- }
- return 0;
-}
-
-WebRtc_Word32 KEventMaxWaitTimeMs = 200;
-
-WebRtc_UWord32 VideoRenderFrames::TimeToNextFrameRelease()
-{
- WebRtc_Word64 timeToRelease = 0;
- ListItem* item = _incomingFrames.First();
- if (item)
- {
- VideoFrame* oldestFrame =
- static_cast<VideoFrame*> (item->GetItem());
- timeToRelease = oldestFrame->RenderTimeMs() - _renderDelayMs
- - TickTime::MillisecondTimestamp();
- if (timeToRelease < 0)
- {
- timeToRelease = 0;
- }
- }
- else
- {
- timeToRelease = KEventMaxWaitTimeMs;
- }
-
- return (WebRtc_UWord32) timeToRelease;
-}
-
-//
-WebRtc_Word32 VideoRenderFrames::SetRenderDelay(
- const WebRtc_UWord32 renderDelay)
-{
- _renderDelayMs = renderDelay;
- return 0;
-}
-
-} //namespace webrtc
-
diff --git a/trunk/src/modules/video_render/main/source/video_render_frames.h b/trunk/src/modules/video_render/main/source/video_render_frames.h
deleted file mode 100644
index 84a2a1c..0000000
--- a/trunk/src/modules/video_render/main/source/video_render_frames.h
+++ /dev/null
@@ -1,78 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_FRAMES_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_FRAMES_H_
-
-#include "list_wrapper.h"
-#include "video_render.h"
-
-namespace webrtc {
-
-// Class definitions
-class VideoRenderFrames
-{
-public:
- VideoRenderFrames();
- ~VideoRenderFrames();
-
- /*
- * Add a frame to the render queue
- */
- WebRtc_Word32 AddFrame(VideoFrame* ptrNewFrame);
-
- /*
- * Get a frame for rendering, if it's time to render.
- */
- VideoFrame* FrameToRender();
-
- /*
- * Return an old frame
- */
- WebRtc_Word32 ReturnFrame(VideoFrame* ptrOldFrame);
-
- /*
- * Releases all frames
- */
- WebRtc_Word32 ReleaseAllFrames();
-
- /*
- * Returns the number of ms to next frame to render
- */
- WebRtc_UWord32 TimeToNextFrameRelease();
-
- /*
- * Sets estimates delay in renderer
- */
- WebRtc_Word32 SetRenderDelay(const WebRtc_UWord32 renderDelay);
-
-private:
- enum
- {
- KMaxNumberOfFrames = 300
- }; // 10 seconds for 30 fps.
- enum
- {
- KOldRenderTimestampMS = 500
- }; //Don't render frames with timestamp older than 500ms from now.
- enum
- {
- KFutureRenderTimestampMS = 10000
- }; //Don't render frames with timestamp more than 10s into the future.
-
- ListWrapper _incomingFrames; // Sorted oldest video frame first
- ListWrapper _emptyFrames; // Empty frames
-
- WebRtc_UWord32 _renderDelayMs; // Set render delay
-};
-
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_FRAMES_H_
diff --git a/trunk/src/modules/video_render/main/source/video_render_impl.cc b/trunk/src/modules/video_render/main/source/video_render_impl.cc
deleted file mode 100644
index 616d03b..0000000
--- a/trunk/src/modules/video_render/main/source/video_render_impl.cc
+++ /dev/null
@@ -1,981 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_render_impl.h"
-#include "engine_configurations.h"
-#include "critical_section_wrapper.h"
-#include "video_render_defines.h"
-#include "trace.h"
-#include "incoming_video_stream.h"
-#include "i_video_render.h"
-
-#include <cassert>
-
-#ifdef WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
-
-#if defined (_WIN32)
-#include "windows/video_render_windows_impl.h"
-#define STANDARD_RENDERING kRenderWindows
-
-#elif defined(MAC_IPHONE) // MAC_IPHONE should go before WEBRTC_MAC_INTEL because WEBRTC_MAC_INTEL gets defined if MAC_IPHONE is defined
-#if defined(IPHONE_GLES_RENDERING)
-#define STANDARD_RENDERING kRenderiPhone
-#include "iPhone/video_render_iphone_impl.h"
-#endif
-
-#elif defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL)
-#if defined(COCOA_RENDERING)
-#define STANDARD_RENDERING kRenderCocoa
-#include "mac/video_render_mac_cocoa_impl.h"
-#elif defined(CARBON_RENDERING)
-#define STANDARD_RENDERING kRenderCarbon
-#include "mac/video_render_mac_carbon_impl.h"
-#endif
-
-#elif defined(WEBRTC_ANDROID)
-#include "Android/video_render_android_impl.h"
-#include "Android/video_render_android_surface_view.h"
-#include "Android/video_render_android_native_opengl2.h"
-#define STANDARD_RENDERING kRenderAndroid
-
-#elif defined(WEBRTC_LINUX)
-#include "linux/video_render_linux_impl.h"
-#define STANDARD_RENDERING kRenderX11
-
-#else
-//Other platforms
-#endif
-
-#endif // WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
-
-// For external rendering
-#include "external/video_render_external_impl.h"
-#ifndef STANDARD_RENDERING
-#define STANDARD_RENDERING kRenderExternal
-#endif // STANDARD_RENDERING
-
-namespace webrtc {
-
-VideoRender*
-VideoRender::CreateVideoRender(const WebRtc_Word32 id,
- void* window,
- const bool fullscreen,
- const VideoRenderType videoRenderType/*=kRenderDefault*/)
-{
-
- VideoRenderType resultVideoRenderType = videoRenderType;
- if (videoRenderType == kRenderDefault)
- {
- resultVideoRenderType = STANDARD_RENDERING;
- }
- return new ModuleVideoRenderImpl(id, resultVideoRenderType, window,
- fullscreen);
-}
-
-void VideoRender::DestroyVideoRender(
- VideoRender* module)
-{
- if (module)
- {
- delete module;
- }
-}
-
-WebRtc_Word32 VideoRender::SetAndroidObjects(void *javaVM)
-{
-#ifdef WEBRTC_ANDROID
- return VideoRenderAndroid::SetAndroidEnvVariables(javaVM);
-#else
- return -1;
-#endif
-}
-
-ModuleVideoRenderImpl::ModuleVideoRenderImpl(
- const WebRtc_Word32 id,
- const VideoRenderType videoRenderType,
- void* window,
- const bool fullscreen) :
- _id(id), _moduleCrit(*CriticalSectionWrapper::CreateCriticalSection()),
- _ptrWindow(window), _renderType(videoRenderType),
- _fullScreen(fullscreen), _ptrRenderer(NULL),
- _streamRenderMap(*(new MapWrapper()))
-{
-
- // Create platform specific renderer
- switch (videoRenderType)
- {
-#ifdef WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
-
-#if defined(_WIN32)
- case kRenderWindows:
- {
- VideoRenderWindowsImpl* ptrRenderer;
- ptrRenderer = new VideoRenderWindowsImpl(_id, videoRenderType, window, _fullScreen);
- if (ptrRenderer)
- {
- _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
- }
- }
- break;
-
-#elif defined(MAC_IPHONE)
- case kRenderiPhone:
- {
- VideoRenderIPhoneImpl* ptrRenderer = new VideoRenderIPhoneImpl(_id, videoRenderType, window, _fullScreen);
- if(ptrRenderer)
- {
- _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
- }
- }
- break;
-
-#elif defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL)
-
-#if defined(COCOA_RENDERING)
- case kRenderCocoa:
- {
- VideoRenderMacCocoaImpl* ptrRenderer = new VideoRenderMacCocoaImpl(_id, videoRenderType, window, _fullScreen);
- if(ptrRenderer)
- {
- _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
- }
- }
-
- break;
-#elif defined(CARBON_RENDERING)
- case kRenderCarbon:
- {
- VideoRenderMacCarbonImpl* ptrRenderer = new VideoRenderMacCarbonImpl(_id, videoRenderType, window, _fullScreen);
- if(ptrRenderer)
- {
- _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
- }
- }
- break;
-#endif
-
-#elif defined(WEBRTC_ANDROID)
- case kRenderAndroid:
- {
- if(AndroidNativeOpenGl2Renderer::UseOpenGL2(window))
- {
- AndroidNativeOpenGl2Renderer* ptrRenderer = NULL;
- ptrRenderer = new AndroidNativeOpenGl2Renderer(_id, videoRenderType, window, _fullScreen);
- if (ptrRenderer)
- {
- _ptrRenderer = reinterpret_cast<IVideoRender*> (ptrRenderer);
- }
- }
- else
- {
- AndroidSurfaceViewRenderer* ptrRenderer = NULL;
- ptrRenderer = new AndroidSurfaceViewRenderer(_id, videoRenderType, window, _fullScreen);
- if (ptrRenderer)
- {
- _ptrRenderer = reinterpret_cast<IVideoRender*> (ptrRenderer);
- }
- }
-
- }
- break;
-#elif defined(WEBRTC_LINUX)
- case kRenderX11:
- {
- VideoRenderLinuxImpl* ptrRenderer = NULL;
- ptrRenderer = new VideoRenderLinuxImpl(_id, videoRenderType, window, _fullScreen);
- if ( ptrRenderer )
- {
- _ptrRenderer = reinterpret_cast<IVideoRender*> (ptrRenderer);
- }
- }
- break;
-
-#else
- // Other platforms
-#endif
-
-#endif // WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
- case kRenderExternal:
- {
- VideoRenderExternalImpl* ptrRenderer(NULL);
- ptrRenderer = new VideoRenderExternalImpl(_id, videoRenderType,
- window, _fullScreen);
- if (ptrRenderer)
- {
- _ptrRenderer = reinterpret_cast<IVideoRender*> (ptrRenderer);
- }
- }
- break;
- default:
- // Error...
- break;
- }
- if (_ptrRenderer)
- {
- if (_ptrRenderer->Init() == -1)
- {
- }
- }
-}
-
-ModuleVideoRenderImpl::~ModuleVideoRenderImpl()
-{
- delete &_moduleCrit;
-
- while (_streamRenderMap.Size() > 0)
- {
- MapItem* item = _streamRenderMap.First();
- IncomingVideoStream* ptrIncomingStream =
- static_cast<IncomingVideoStream*> (item->GetItem());
- assert(ptrIncomingStream != NULL);
- delete ptrIncomingStream;
- _streamRenderMap.Erase(item);
- }
- delete &_streamRenderMap;
-
- // Delete platform specific renderer
- if (_ptrRenderer)
- {
- VideoRenderType videoRenderType = _ptrRenderer->RenderType();
- switch (videoRenderType)
- {
- case kRenderExternal:
- {
- VideoRenderExternalImpl
- * ptrRenderer =
- reinterpret_cast<VideoRenderExternalImpl*> (_ptrRenderer);
- _ptrRenderer = NULL;
- delete ptrRenderer;
- }
- break;
-#ifdef WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
-
-#if defined(_WIN32)
- case kRenderWindows:
- {
- VideoRenderWindowsImpl* ptrRenderer = reinterpret_cast<VideoRenderWindowsImpl*>(_ptrRenderer);
- _ptrRenderer = NULL;
- delete ptrRenderer;
- }
- break;
-#elif defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL)
-
-#if defined(COCOA_RENDERING)
- case kRenderCocoa:
- {
- VideoRenderMacCocoaImpl* ptrRenderer = reinterpret_cast<VideoRenderMacCocoaImpl*> (_ptrRenderer);
- _ptrRenderer = NULL;
- delete ptrRenderer;
- }
- break;
-#elif defined(CARBON_RENDERING)
- case kRenderCarbon:
- {
- VideoRenderMacCarbonImpl* ptrRenderer = reinterpret_cast<VideoRenderMacCarbonImpl*> (_ptrRenderer);
- _ptrRenderer = NULL;
- delete ptrRenderer;
- }
- break;
-#endif
-
-#elif defined(MAC_IPHONE)
- case kRenderiPhone:
- break;
-
-#elif defined(WEBRTC_ANDROID)
- case kRenderAndroid:
- {
- VideoRenderAndroid* ptrRenderer = reinterpret_cast<VideoRenderAndroid*> (_ptrRenderer);
- _ptrRenderer = NULL;
- delete ptrRenderer;
- }
- break;
-
-#elif defined(WEBRTC_LINUX)
- case kRenderX11:
- {
- VideoRenderLinuxImpl* ptrRenderer = reinterpret_cast<VideoRenderLinuxImpl*> (_ptrRenderer);
- _ptrRenderer = NULL;
- delete ptrRenderer;
- }
- break;
-#else
- //other platforms
-#endif
-
-#endif // WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
-
- default:
- // Error...
- break;
- }
- }
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::ChangeUniqueId(const WebRtc_Word32 id)
-{
-
- CriticalSectionScoped cs(_moduleCrit);
-
- _id = id;
-
- if (_ptrRenderer)
- {
- _ptrRenderer->ChangeUniqueId(_id);
- }
-
- return 0;
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::TimeUntilNextProcess()
-{
- // Not used
- return 50;
-}
-WebRtc_Word32 ModuleVideoRenderImpl::Process()
-{
- // Not used
- return 0;
-}
-
-void*
-ModuleVideoRenderImpl::Window()
-{
- CriticalSectionScoped cs(_moduleCrit);
- return _ptrWindow;
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::ChangeWindow(void* window)
-{
-
- CriticalSectionScoped cs(_moduleCrit);
-
-#ifdef WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
-
-#if defined(MAC_IPHONE) // MAC_IPHONE must go before WEBRTC_MAC or WEBRTC_MAC_INTEL
- _ptrRenderer = NULL;
- delete _ptrRenderer;
-
- VideoRenderIPhoneImpl* ptrRenderer;
- ptrRenderer = new VideoRenderIPhoneImpl(_id, kRenderiPhone, window, _fullScreen);
- if (!ptrRenderer)
- {
- return -1;
- }
- _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
- return _ptrRenderer->ChangeWindow(window);
-
-#elif defined(WEBRTC_MAC) | defined(WEBRTC_MAC_INTEL)
-
- _ptrRenderer = NULL;
- delete _ptrRenderer;
-
-#if defined(COCOA_RENDERING)
- VideoRenderMacCocoaImpl* ptrRenderer;
- ptrRenderer = new VideoRenderMacCocoaImpl(_id, kRenderCocoa, window, _fullScreen);
-#elif defined(CARBON_RENDERING)
- VideoRenderMacCarbonImpl* ptrRenderer;
- ptrRenderer = new VideoRenderMacCarbonImpl(_id, kRenderCarbon, window, _fullScreen);
-#endif
- if (!ptrRenderer)
- {
- return -1;
- }
- _ptrRenderer = reinterpret_cast<IVideoRender*>(ptrRenderer);
- return _ptrRenderer->ChangeWindow(window);
-
-#else
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No renderer", __FUNCTION__);
- return -1;
- }
- return _ptrRenderer->ChangeWindow(window);
-
-#endif
-
-#else // WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
- return -1;
-#endif
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::Id()
-{
- CriticalSectionScoped cs(_moduleCrit);
- return _id;
-}
-
-WebRtc_UWord32 ModuleVideoRenderImpl::GetIncomingFrameRate(
- const WebRtc_UWord32 streamId)
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- MapItem* mapItem = _streamRenderMap.Find(streamId);
- if (mapItem == NULL)
- {
- // This stream doesn't exist
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: stream doesn't exist", __FUNCTION__);
- return 0;
- }
- IncomingVideoStream* incomingStream =
- static_cast<IncomingVideoStream*> (mapItem->GetItem());
- if (incomingStream == NULL)
- {
- // This should never happen
- assert(false);
- _streamRenderMap.Erase(mapItem);
- return 0;
- }
- return incomingStream->IncomingRate();
-}
-
-VideoRenderCallback*
-ModuleVideoRenderImpl::AddIncomingRenderStream(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No renderer", __FUNCTION__);
- return NULL;
- }
-
- if (_streamRenderMap.Find(streamId) != NULL)
- {
- // The stream already exists...
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: stream already exists", __FUNCTION__);
- return NULL;
- }
-
- VideoRenderCallback* ptrRenderCallback =
- _ptrRenderer->AddIncomingRenderStream(streamId, zOrder, left, top,
- right, bottom);
- if (ptrRenderCallback == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: Can't create incoming stream in renderer",
- __FUNCTION__);
- return NULL;
- }
-
- // Create platform independant code
- IncomingVideoStream* ptrIncomingStream = new IncomingVideoStream(_id,
- streamId);
- if (ptrIncomingStream == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: Can't create incoming stream", __FUNCTION__);
- return NULL;
- }
-
-
- if (ptrIncomingStream->SetRenderCallback(ptrRenderCallback) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: Can't set render callback", __FUNCTION__);
- delete ptrIncomingStream;
- _ptrRenderer->DeleteIncomingRenderStream(streamId);
- return NULL;
- }
-
- VideoRenderCallback* moduleCallback =
- ptrIncomingStream->ModuleCallback();
-
- // Store the stream
- _streamRenderMap.Insert(streamId, ptrIncomingStream);
-
- return moduleCallback;
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::DeleteIncomingRenderStream(
- const WebRtc_UWord32 streamId)
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No renderer", __FUNCTION__);
- return -1;
- }
-
- MapItem* mapItem = _streamRenderMap.Find(streamId);
- if (!mapItem)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: stream doesn't exist", __FUNCTION__);
- return -1;
- }
-
- IncomingVideoStream* ptrIncomingStream =
- static_cast<IncomingVideoStream*> (mapItem->GetItem());
- delete ptrIncomingStream;
- ptrIncomingStream = NULL;
- _ptrRenderer->DeleteIncomingRenderStream(streamId);
- _streamRenderMap.Erase(mapItem);
-
- return 0;
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::AddExternalRenderCallback(
- const WebRtc_UWord32 streamId,
- VideoRenderCallback* renderObject)
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- MapItem* mapItem = _streamRenderMap.Find(streamId);
- if (!mapItem)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: stream doesn't exist", __FUNCTION__);
- return -1;
- }
-
- IncomingVideoStream* ptrIncomingStream =
- static_cast<IncomingVideoStream*> (mapItem->GetItem());
- if (!ptrIncomingStream) {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: could not get stream", __FUNCTION__);
- return -1;
- }
- return ptrIncomingStream->SetExternalCallback(renderObject);
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::GetIncomingRenderStreamProperties(
- const WebRtc_UWord32 streamId,
- WebRtc_UWord32& zOrder,
- float& left,
- float& top,
- float& right,
- float& bottom) const
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No renderer", __FUNCTION__);
- return -1;
- }
-
- return _ptrRenderer->GetIncomingRenderStreamProperties(streamId, zOrder,
- left, top, right,
- bottom);
-}
-
-WebRtc_UWord32 ModuleVideoRenderImpl::GetNumIncomingRenderStreams() const
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- return (WebRtc_UWord32) _streamRenderMap.Size();
-}
-
-bool ModuleVideoRenderImpl::HasIncomingRenderStream(
- const WebRtc_UWord32 streamId) const
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- bool hasStream = false;
- if (_streamRenderMap.Find(streamId) != NULL)
- {
- hasStream = true;
- }
- return hasStream;
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::RegisterRawFrameCallback(
- const WebRtc_UWord32 streamId,
- VideoRenderCallback* callbackObj)
-{
- return -1;
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::StartRender(const WebRtc_UWord32 streamId)
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No renderer", __FUNCTION__);
- return -1;
- }
-
- // Start the stream
- MapItem* item = _streamRenderMap.Find(streamId);
- if (item == NULL)
- {
- return -1;
- }
-
- IncomingVideoStream* incomingStream =
- static_cast<IncomingVideoStream*> (item->GetItem());
- if (incomingStream->Start() == -1)
- {
- return -1;
- }
-
- // Start the HW renderer
- if (_ptrRenderer->StartRender() == -1)
- {
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::StopRender(const WebRtc_UWord32 streamId)
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s(%d): No renderer", __FUNCTION__, streamId);
- return -1;
- }
-
- // Stop the incoming stream
- MapItem* item = _streamRenderMap.Find(streamId);
- if (item == NULL)
- {
- return -1;
- }
-
- IncomingVideoStream* incomingStream =
- static_cast<IncomingVideoStream*> (item->GetItem());
- if (incomingStream->Stop() == -1)
- {
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::ResetRender()
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- WebRtc_Word32 error = 0;
-
- // Loop through all incoming streams and stop them
- MapItem* item = _streamRenderMap.First();
- while (item)
- {
- IncomingVideoStream* incomingStream =
- static_cast<IncomingVideoStream*> (item->GetItem());
- if (incomingStream->Reset() == -1)
- {
- error = -1;
- }
- item = _streamRenderMap.Next(item);
- }
- return error;
-}
-
-RawVideoType ModuleVideoRenderImpl::PreferredVideoType() const
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (_ptrRenderer == NULL)
- {
- return kVideoI420;
- }
-
- return _ptrRenderer->PerferedVideoType();
-}
-
-bool ModuleVideoRenderImpl::IsFullScreen()
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No renderer", __FUNCTION__);
- return false;
- }
- return _ptrRenderer->FullScreen();
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::GetScreenResolution(
- WebRtc_UWord32& screenWidth,
- WebRtc_UWord32& screenHeight) const
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No renderer", __FUNCTION__);
- return false;
- }
- return _ptrRenderer->GetScreenResolution(screenWidth, screenHeight);
-}
-
-WebRtc_UWord32 ModuleVideoRenderImpl::RenderFrameRate(
- const WebRtc_UWord32 streamId)
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No renderer", __FUNCTION__);
- return false;
- }
- return _ptrRenderer->RenderFrameRate(streamId);
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::SetStreamCropping(
- const WebRtc_UWord32 streamId,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No renderer", __FUNCTION__);
- return false;
- }
- return _ptrRenderer->SetStreamCropping(streamId, left, top, right, bottom);
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::SetTransparentBackground(const bool enable)
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No renderer", __FUNCTION__);
- return false;
- }
- return _ptrRenderer->SetTransparentBackground(enable);
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::FullScreenRender(void* window,
- const bool enable)
-{
- return -1;
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::SetText(
- const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 textColorRef,
- const WebRtc_UWord32 backgroundColorRef,
- const float left, const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No renderer", __FUNCTION__);
- return -1;
- }
- return _ptrRenderer->SetText(textId, text, textLength, textColorRef,
- backgroundColorRef, left, top, right, bottom);
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No renderer", __FUNCTION__);
- return -1;
- }
- return _ptrRenderer->SetBitmap(bitMap, pictureId, colorKey, left, top,
- right, bottom);
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::GetLastRenderedFrame(
- const WebRtc_UWord32 streamId,
- VideoFrame &frame) const
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No renderer", __FUNCTION__);
- return -1;
- }
-
- MapItem *item = _streamRenderMap.Find(streamId);
- if (item == NULL)
- {
- // This stream doesn't exist
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: stream doesn't exist", __FUNCTION__);
- return 0;
- }
- IncomingVideoStream* incomingStream =
- static_cast<IncomingVideoStream*> (item->GetItem());
- if (incomingStream == NULL)
- {
- // This should never happen
- assert(false);
- _streamRenderMap.Erase(item);
- return 0;
- }
- return incomingStream->GetLastRenderedFrame(frame);
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::ConfigureRenderer(
- const WebRtc_UWord32 streamId,
- const unsigned int zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No renderer", __FUNCTION__);
- return false;
- }
- return _ptrRenderer->ConfigureRenderer(streamId, zOrder, left, top, right,
- bottom);
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::SetStartImage(
- const WebRtc_UWord32 streamId,
- const VideoFrame& videoFrame)
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No renderer", __FUNCTION__);
- return -1;
- }
-
- MapItem *item = _streamRenderMap.Find(streamId);
- if (item == NULL)
- {
- // This stream doesn't exist
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: stream doesn't exist", __FUNCTION__);
- return -1;
- }
- IncomingVideoStream* incomingStream =
- static_cast<IncomingVideoStream*> (item->GetItem());
- if (incomingStream == NULL)
- {
- // This should never happen
- assert(false);
- _streamRenderMap.Erase(item);
- return 0;
- }
- return incomingStream->SetStartImage(videoFrame);
-
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::SetTimeoutImage(
- const WebRtc_UWord32 streamId,
- const VideoFrame& videoFrame,
- const WebRtc_UWord32 timeout)
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No renderer", __FUNCTION__);
- return -1;
- }
-
- MapItem *item = _streamRenderMap.Find(streamId);
- if (item == NULL)
- {
- // This stream doesn't exist
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: stream doesn't exist", __FUNCTION__);
- return -1;
- }
- IncomingVideoStream* incomingStream =
- static_cast<IncomingVideoStream*> (item->GetItem());
- if (incomingStream == NULL)
- {
- // This should never happen
- assert(false);
- _streamRenderMap.Erase(item);
- return 0;
- }
- return incomingStream->SetTimeoutImage(videoFrame, timeout);
-}
-
-WebRtc_Word32 ModuleVideoRenderImpl::MirrorRenderStream(const int renderId,
- const bool enable,
- const bool mirrorXAxis,
- const bool mirrorYAxis)
-{
- CriticalSectionScoped cs(_moduleCrit);
-
- if (!_ptrRenderer)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: No renderer", __FUNCTION__);
- return -1;
- }
-
- MapItem *item = _streamRenderMap.Find(renderId);
- if (item == NULL)
- {
- // This stream doesn't exist
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, _id,
- "%s: stream doesn't exist", __FUNCTION__);
- return 0;
- }
- IncomingVideoStream* incomingStream =
- static_cast<IncomingVideoStream*> (item->GetItem());
- if (incomingStream == NULL)
- {
- // This should never happen
- assert(false);
- _streamRenderMap.Erase(item);
- return 0;
- }
-
- return incomingStream->EnableMirroring(enable, mirrorXAxis, mirrorYAxis);
-}
-
-} //namespace webrtc
-
diff --git a/trunk/src/modules/video_render/main/source/video_render_impl.h b/trunk/src/modules/video_render/main/source/video_render_impl.h
deleted file mode 100644
index 16f934e..0000000
--- a/trunk/src/modules/video_render/main/source/video_render_impl.h
+++ /dev/null
@@ -1,229 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_IMPL_H_
-
-#include "engine_configurations.h"
-#include "video_render.h"
-#include "map_wrapper.h"
-
-//#include "video_render_defines.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class IncomingVideoStream;
-class IVideoRender;
-class MapWrapper;
-
-// Class definitions
-class ModuleVideoRenderImpl: public VideoRender
-{
-public:
- /*
- * VideoRenderer constructor/destructor
- */
- ModuleVideoRenderImpl(const WebRtc_Word32 id,
- const VideoRenderType videoRenderType,
- void* window, const bool fullscreen);
-
- virtual ~ModuleVideoRenderImpl();
-
- /*
- * Change the unique identifier of this object
- */
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- virtual WebRtc_Word32 TimeUntilNextProcess();
- virtual WebRtc_Word32 Process();
-
- /*
- * Returns the render window
- */
- virtual void* Window();
-
- /*
- * Change render window
- */
- virtual WebRtc_Word32 ChangeWindow(void* window);
-
- /*
- * Returns module id
- */
- WebRtc_Word32 Id();
-
- /**************************************************************************
- *
- * Incoming Streams
- *
- ***************************************************************************/
-
- /*
- * Add incoming render stream
- */
- virtual VideoRenderCallback
- * AddIncomingRenderStream(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left, const float top,
- const float right, const float bottom);
- /*
- * Delete incoming render stream
- */
- virtual WebRtc_Word32
- DeleteIncomingRenderStream(const WebRtc_UWord32 streamId);
-
- /*
- * Add incoming render callback, used for external rendering
- */
- virtual WebRtc_Word32
- AddExternalRenderCallback(const WebRtc_UWord32 streamId,
- VideoRenderCallback* renderObject);
-
- /*
- * Get the porperties for an incoming render stream
- */
- virtual WebRtc_Word32
- GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
- WebRtc_UWord32& zOrder,
- float& left, float& top,
- float& right, float& bottom) const;
- /*
- * Incoming frame rate for the specified stream.
- */
- virtual WebRtc_UWord32 GetIncomingFrameRate(const WebRtc_UWord32 streamId);
-
- /*
- * Returns the number of incoming streams added to this render module
- */
- virtual WebRtc_UWord32 GetNumIncomingRenderStreams() const;
-
- /*
- * Returns true if this render module has the streamId added, false otherwise.
- */
- virtual bool HasIncomingRenderStream(const WebRtc_UWord32 streamId) const;
-
- /*
- *
- */
- virtual WebRtc_Word32
- RegisterRawFrameCallback(const WebRtc_UWord32 streamId,
- VideoRenderCallback* callbackObj);
-
- virtual WebRtc_Word32 GetLastRenderedFrame(const WebRtc_UWord32 streamId,
- VideoFrame &frame) const;
-
- /**************************************************************************
- *
- * Start/Stop
- *
- ***************************************************************************/
-
- /*
- * Starts rendering the specified stream
- */
- virtual WebRtc_Word32 StartRender(const WebRtc_UWord32 streamId);
-
- /*
- * Stops the renderer
- */
- virtual WebRtc_Word32 StopRender(const WebRtc_UWord32 streamId);
-
- /*
- * Sets the renderer in start state, no streams removed.
- */
- virtual WebRtc_Word32 ResetRender();
-
- /**************************************************************************
- *
- * Properties
- *
- ***************************************************************************/
-
- /*
- * Returns the prefered render video type
- */
- virtual RawVideoType PreferredVideoType() const;
-
- /*
- * Returns true if the renderer is in fullscreen mode, otherwise false.
- */
- virtual bool IsFullScreen();
-
- /*
- * Gets screen resolution in pixels
- */
- virtual WebRtc_Word32
- GetScreenResolution(WebRtc_UWord32& screenWidth,
- WebRtc_UWord32& screenHeight) const;
-
- /*
- * Get the actual render rate for this stream. I.e rendered frame rate,
- * not frames delivered to the renderer.
- */
- virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId);
-
- /*
- * Set cropping of incoming stream
- */
- virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
- const float left, const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
- const unsigned int zOrder,
- const float left, const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
-
- virtual WebRtc_Word32 FullScreenRender(void* window, const bool enable);
-
- virtual WebRtc_Word32 SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey, const float left,
- const float top, const float right,
- const float bottom);
-
- virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 textColorRef,
- const WebRtc_UWord32 backgroundColorRef,
- const float left, const float top,
- const float right, const float bottom);
-
- virtual WebRtc_Word32 SetStartImage(const WebRtc_UWord32 streamId,
- const VideoFrame& videoFrame);
-
- virtual WebRtc_Word32 SetTimeoutImage(const WebRtc_UWord32 streamId,
- const VideoFrame& videoFrame,
- const WebRtc_UWord32 timeout);
-
- virtual WebRtc_Word32 MirrorRenderStream(const int renderId,
- const bool enable,
- const bool mirrorXAxis,
- const bool mirrorYAxis);
-
-private:
- WebRtc_Word32 _id;
- CriticalSectionWrapper& _moduleCrit;
- void* _ptrWindow;
- VideoRenderType _renderType;
- bool _fullScreen;
-
- IVideoRender* _ptrRenderer;
- MapWrapper& _streamRenderMap;
-};
-
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_VIDEO_RENDER_IMPL_H_
diff --git a/trunk/src/modules/video_render/main/source/windows/i_video_render_win.h b/trunk/src/modules/video_render/main/source/windows/i_video_render_win.h
deleted file mode 100644
index a765134..0000000
--- a/trunk/src/modules/video_render/main/source/windows/i_video_render_win.h
+++ /dev/null
@@ -1,118 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_I_VIDEO_RENDER_WIN_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_I_VIDEO_RENDER_WIN_H_
-
-#include "video_render.h"
-
-namespace webrtc {
-
-// Class definitions
-class IVideoRenderWin
-{
-public:
- /**************************************************************************
- *
- * Constructor/destructor
- *
- ***************************************************************************/
- virtual ~IVideoRenderWin()
- {
- };
-
- virtual WebRtc_Word32 Init() = 0;
-
- /**************************************************************************
- *
- * Incoming Streams
- *
- ***************************************************************************/
-
- virtual VideoRenderCallback
- * CreateChannel(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom) = 0;
-
- virtual WebRtc_Word32 DeleteChannel(const WebRtc_UWord32 streamId) = 0;
-
- virtual WebRtc_Word32 GetStreamSettings(const WebRtc_UWord32 channel,
- const WebRtc_UWord16 streamId,
- WebRtc_UWord32& zOrder,
- float& left,
- float& top,
- float& right,
- float& bottom) = 0;
-
- /**************************************************************************
- *
- * Start/Stop
- *
- ***************************************************************************/
-
- virtual WebRtc_Word32 StartRender() = 0;
-
- virtual WebRtc_Word32 StopRender() = 0;
-
- /**************************************************************************
- *
- * Properties
- *
- ***************************************************************************/
-
- virtual bool IsFullScreen() = 0;
-
- virtual WebRtc_Word32 SetCropping(const WebRtc_UWord32 channel,
- const WebRtc_UWord16 streamId,
- const float left,
- const float top,
- const float right,
- const float bottom) = 0;
-
- virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 channel,
- const WebRtc_UWord16 streamId,
- const unsigned int zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom) = 0;
-
- virtual WebRtc_Word32 SetTransparentBackground(const bool enable) = 0;
-
- virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 colorText,
- const WebRtc_UWord32 colorBg,
- const float left,
- const float top,
- const float rigth,
- const float bottom) = 0;
-
- virtual WebRtc_Word32 SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey,
- const float left,
- const float top,
- const float right,
- const float bottom) = 0;
-
- virtual WebRtc_Word32 ChangeWindow(void* window) = 0;
-
- virtual WebRtc_Word32 GetGraphicsMemory(WebRtc_UWord64& totalMemory,
- WebRtc_UWord64& availableMemory) = 0;
-
-};
-
-} //namespace webrtc
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_I_VIDEO_RENDER_WIN_H_
diff --git a/trunk/src/modules/video_render/main/source/windows/video_render_direct3d9.cc b/trunk/src/modules/video_render/main/source/windows/video_render_direct3d9.cc
deleted file mode 100644
index 00bc41b..0000000
--- a/trunk/src/modules/video_render/main/source/windows/video_render_direct3d9.cc
+++ /dev/null
@@ -1,1192 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Own include file
-#include "video_render_direct3d9.h"
-
-// System include files
-#include <windows.h>
-
-// WebRtc include files
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "trace.h"
-#include "thread_wrapper.h"
-#include "common_video/libyuv/include/libyuv.h"
-
-namespace webrtc {
-
-// A structure for our custom vertex type
-struct CUSTOMVERTEX
-{
- FLOAT x, y, z;
- DWORD color; // The vertex color
- FLOAT u, v;
-};
-
-// Our custom FVF, which describes our custom vertex structure
-#define D3DFVF_CUSTOMVERTEX (D3DFVF_XYZ|D3DFVF_DIFFUSE|D3DFVF_TEX1)
-
-/*
- *
- * D3D9Channel
- *
- */
-D3D9Channel::D3D9Channel(LPDIRECT3DDEVICE9 pd3DDevice,
- CriticalSectionWrapper* critSect,
- Trace* trace) :
- _width(0),
- _height(0),
- _pd3dDevice(pd3DDevice),
- _pTexture(NULL),
- _bufferIsUpdated(false),
- _critSect(critSect),
- _streamId(0),
- _zOrder(0),
- _startWidth(0),
- _startHeight(0),
- _stopWidth(0),
- _stopHeight(0)
-{
-
-}
-
-D3D9Channel::~D3D9Channel()
-{
- //release the texture
- if (_pTexture != NULL)
- {
- _pTexture->Release();
- _pTexture = NULL;
- }
-}
-
-void D3D9Channel::SetStreamSettings(WebRtc_UWord16 streamId,
- WebRtc_UWord32 zOrder,
- float startWidth,
- float startHeight,
- float stopWidth,
- float stopHeight)
-{
- _streamId = streamId;
- _zOrder = zOrder;
- _startWidth = startWidth;
- _startHeight = startHeight;
- _stopWidth = stopWidth;
- _stopHeight = stopHeight;
-}
-
-int D3D9Channel::GetStreamSettings(WebRtc_UWord16 streamId,
- WebRtc_UWord32& zOrder,
- float& startWidth,
- float& startHeight,
- float& stopWidth,
- float& stopHeight)
-{
- streamId = _streamId;
- zOrder = _zOrder;
- startWidth = _startWidth;
- startHeight = _startHeight;
- stopWidth = _stopWidth;
- stopHeight = _stopHeight;
- return 0;
-}
-
-int D3D9Channel::GetTextureWidth()
-{
- return _width;
-}
-
-int D3D9Channel::GetTextureHeight()
-{
- return _height;
-}
-
-// Called from video engine when a the frame size changed
-int D3D9Channel::FrameSizeChange(int width, int height, int numberOfStreams)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "FrameSizeChange, wifth: %d, height: %d, streams: %d", width,
- height, numberOfStreams);
-
- CriticalSectionScoped cs(*_critSect);
- _width = width;
- _height = height;
-
- //clean the previous texture
- if (_pTexture != NULL)
- {
- _pTexture->Release();
- _pTexture = NULL;
- }
-
- HRESULT ret = E_POINTER;
-
- if (_pd3dDevice)
- ret = _pd3dDevice->CreateTexture(_width, _height, 1, 0, D3DFMT_A8R8G8B8,
- D3DPOOL_MANAGED, &_pTexture, NULL);
-
- if (FAILED(ret))
- {
- _pTexture = NULL;
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 D3D9Channel::RenderFrame(const WebRtc_UWord32 streamId,
- VideoFrame& videoFrame)
-{
- CriticalSectionScoped cs(*_critSect);
- if (_width != videoFrame.Width() || _height != videoFrame.Height())
- {
- if (FrameSizeChange(videoFrame.Width(), videoFrame.Height(), 1) == -1)
- {
- return -1;
- }
- }
- return DeliverFrame(videoFrame.Buffer(), videoFrame.Length(),
- videoFrame.TimeStamp());
-}
-
-// Called from video engine when a new frame should be rendered.
-int D3D9Channel::DeliverFrame(unsigned char* buffer,
- int bufferSize,
- unsigned int timeStamp90kHz)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVideo, -1,
- "DeliverFrame to D3D9Channel");
-
- CriticalSectionScoped cs(*_critSect);
-
- //FIXME if _bufferIsUpdated is still true (not be renderred), do we what to update the texture?)
- //probably not
- if (_bufferIsUpdated)
- {
- WEBRTC_TRACE(kTraceStream, kTraceVideo, -1,
- "Last frame hasn't been rendered yet. Drop this frame.");
- return -1;
- }
-
- if (!_pd3dDevice)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "D3D for rendering not initialized.");
- return -1;
- }
-
- if (!_pTexture)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Texture for rendering not initialized.");
- return -1;
- }
-
- D3DLOCKED_RECT lr;
-
- if (FAILED(_pTexture->LockRect(0, &lr, NULL, 0)))
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Failed to lock a texture in D3D9 Channel.");
- return -1;
- }
- UCHAR* pRect = (UCHAR*) lr.pBits;
-
- ConvertFromI420(buffer, _width, kARGB, 0, _width, _height, pRect);
-
- if (FAILED(_pTexture->UnlockRect(0)))
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Failed to unlock a texture in D3D9 Channel.");
- return -1;
- }
-
- _bufferIsUpdated = true;
-
- return 0;
-}
-
-// Called by d3d channel owner to indicate the frame/texture has been rendered off
-int D3D9Channel::RenderOffFrame()
-{
- WEBRTC_TRACE(kTraceStream, kTraceVideo, -1,
- "Frame has been rendered to the screen.");
- CriticalSectionScoped cs(*_critSect);
- _bufferIsUpdated = false;
- return 0;
-}
-
-// Called by d3d channel owner to check if the texture is updated
-int D3D9Channel::IsUpdated(bool& isUpdated)
-{
- CriticalSectionScoped cs(*_critSect);
- isUpdated = _bufferIsUpdated;
- return 0;
-}
-
-// Called by d3d channel owner to get the texture
-LPDIRECT3DTEXTURE9 D3D9Channel::GetTexture()
-{
- CriticalSectionScoped cs(*_critSect);
- return _pTexture;
-}
-
-int D3D9Channel::ReleaseTexture()
-{
- CriticalSectionScoped cs(*_critSect);
-
- //release the texture
- if (_pTexture != NULL)
- {
- _pTexture->Release();
- _pTexture = NULL;
- }
- _pd3dDevice = NULL;
- return 0;
-}
-
-int D3D9Channel::RecreateTexture(LPDIRECT3DDEVICE9 pd3DDevice)
-{
- CriticalSectionScoped cs(*_critSect);
-
- _pd3dDevice = pd3DDevice;
-
- if (_pTexture != NULL)
- {
- _pTexture->Release();
- _pTexture = NULL;
- }
-
- HRESULT ret;
-
- ret = _pd3dDevice->CreateTexture(_width, _height, 1, 0, D3DFMT_A8R8G8B8,
- D3DPOOL_MANAGED, &_pTexture, NULL);
-
- if (FAILED(ret))
- {
- _pTexture = NULL;
- return -1;
- }
-
- return 0;
-}
-
-/*
- *
- * VideoRenderDirect3D9
- *
- */
-VideoRenderDirect3D9::VideoRenderDirect3D9(Trace* trace,
- HWND hWnd,
- bool fullScreen) :
- _refD3DCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
- _trace(trace),
- _hWnd(hWnd),
- _fullScreen(fullScreen),
- _pTextureLogo(NULL),
- _pVB(NULL),
- _pd3dDevice(NULL),
- _pD3D(NULL),
- _d3dChannels(),
- _d3dZorder(),
- _screenUpdateThread(NULL),
- _screenUpdateEvent(NULL),
- _logoLeft(0),
- _logoTop(0),
- _logoRight(0),
- _logoBottom(0),
- _pd3dSurface(NULL),
- _totalMemory(-1),
- _availableMemory(-1)
-{
- _screenUpdateThread = ThreadWrapper::CreateThread(ScreenUpdateThreadProc,
- this, kRealtimePriority);
- _screenUpdateEvent = EventWrapper::Create();
- SetRect(&_originalHwndRect, 0, 0, 0, 0);
-}
-
-VideoRenderDirect3D9::~VideoRenderDirect3D9()
-{
- //NOTE: we should not enter CriticalSection in here!
-
- // Signal event to exit thread, then delete it
- ThreadWrapper* tmpPtr = _screenUpdateThread;
- _screenUpdateThread = NULL;
- if (tmpPtr)
- {
- tmpPtr->SetNotAlive();
- _screenUpdateEvent->Set();
- _screenUpdateEvent->StopTimer();
-
- if (tmpPtr->Stop())
- {
- delete tmpPtr;
- }
- }
- delete _screenUpdateEvent;
-
- //close d3d device
- CloseDevice();
-
- // Delete all channels
- std::map<int, D3D9Channel*>::iterator it = _d3dChannels.begin();
- while (it != _d3dChannels.end())
- {
- delete it->second;
- it = _d3dChannels.erase(it);
- }
- // Clean the zOrder map
- _d3dZorder.clear();
-
- if (_fullScreen)
- {
- // restore hwnd to original size and position
- ::SetWindowPos(_hWnd, HWND_NOTOPMOST, _originalHwndRect.left,
- _originalHwndRect.top, _originalHwndRect.right
- - _originalHwndRect.left,
- _originalHwndRect.bottom - _originalHwndRect.top,
- SWP_FRAMECHANGED);
- ::RedrawWindow(_hWnd, NULL, NULL, RDW_INVALIDATE | RDW_UPDATENOW
- | RDW_ERASE);
- ::RedrawWindow(NULL, NULL, NULL, RDW_INVALIDATE | RDW_UPDATENOW
- | RDW_ERASE);
- }
-
- delete &_refD3DCritsect;
-}
-
-DWORD VideoRenderDirect3D9::GetVertexProcessingCaps()
-{
- D3DCAPS9 caps;
- DWORD dwVertexProcessing = D3DCREATE_SOFTWARE_VERTEXPROCESSING;
- if (SUCCEEDED(_pD3D->GetDeviceCaps(D3DADAPTER_DEFAULT, D3DDEVTYPE_HAL,
- &caps)))
- {
- if ((caps.DevCaps & D3DDEVCAPS_HWTRANSFORMANDLIGHT)
- == D3DDEVCAPS_HWTRANSFORMANDLIGHT)
- {
- dwVertexProcessing = D3DCREATE_HARDWARE_VERTEXPROCESSING;
- }
- }
- return dwVertexProcessing;
-}
-
-int VideoRenderDirect3D9::InitializeD3D(HWND hWnd,
- D3DPRESENT_PARAMETERS* pd3dpp)
-{
- // initialize Direct3D
- if (NULL == (_pD3D = Direct3DCreate9(D3D_SDK_VERSION)))
- {
- return -1;
- }
-
- // determine what type of vertex processing to use based on the device capabilities
- DWORD dwVertexProcessing = GetVertexProcessingCaps();
-
- // get the display mode
- D3DDISPLAYMODE d3ddm;
- _pD3D->GetAdapterDisplayMode(D3DADAPTER_DEFAULT, &d3ddm);
- pd3dpp->BackBufferFormat = d3ddm.Format;
-
- // create the D3D device
- if (FAILED(_pD3D->CreateDevice(D3DADAPTER_DEFAULT, D3DDEVTYPE_HAL, hWnd,
- dwVertexProcessing | D3DCREATE_MULTITHREADED
- | D3DCREATE_FPU_PRESERVE, pd3dpp,
- &_pd3dDevice)))
- {
- //try the ref device
- if (FAILED(_pD3D->CreateDevice(D3DADAPTER_DEFAULT, D3DDEVTYPE_REF,
- hWnd, dwVertexProcessing
- | D3DCREATE_MULTITHREADED
- | D3DCREATE_FPU_PRESERVE,
- pd3dpp, &_pd3dDevice)))
- {
- return -1;
- }
- }
-
- return 0;
-}
-
-int VideoRenderDirect3D9::ResetDevice()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "VideoRenderDirect3D9::ResetDevice");
-
- CriticalSectionScoped cs(_refD3DCritsect);
-
- //release the channel texture
- std::map<int, D3D9Channel*>::iterator it;
- it = _d3dChannels.begin();
- while (it != _d3dChannels.end())
- {
- if (it->second)
- {
- it->second->ReleaseTexture();
- }
- it++;
- }
-
- //close d3d device
- if (CloseDevice() != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "VideoRenderDirect3D9::ResetDevice failed to CloseDevice");
- return -1;
- }
-
- //reinit d3d device
- if (InitDevice() != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "VideoRenderDirect3D9::ResetDevice failed to InitDevice");
- return -1;
- }
-
- //recreate channel texture
- it = _d3dChannels.begin();
- while (it != _d3dChannels.end())
- {
- if (it->second)
- {
- it->second->RecreateTexture(_pd3dDevice);
- }
- it++;
- }
-
- return 0;
-}
-
-int VideoRenderDirect3D9::InitDevice()
-{
- // Set up the structure used to create the D3DDevice
- ZeroMemory(&_d3dpp, sizeof(_d3dpp));
- _d3dpp.SwapEffect = D3DSWAPEFFECT_DISCARD;
- _d3dpp.BackBufferFormat = D3DFMT_A8R8G8B8;
- if (GetWindowRect(_hWnd, &_originalHwndRect) == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "VideoRenderDirect3D9::InitDevice Could not get window size");
- return -1;
- }
- if (!_fullScreen)
- {
- _winWidth = _originalHwndRect.right - _originalHwndRect.left;
- _winHeight = _originalHwndRect.bottom - _originalHwndRect.top;
- _d3dpp.Windowed = TRUE;
- _d3dpp.BackBufferHeight = 0;
- _d3dpp.BackBufferWidth = 0;
- }
- else
- {
- _winWidth = (LONG) ::GetSystemMetrics(SM_CXSCREEN);
- _winHeight = (LONG) ::GetSystemMetrics(SM_CYSCREEN);
- _d3dpp.Windowed = FALSE;
- _d3dpp.BackBufferWidth = _winWidth;
- _d3dpp.BackBufferHeight = _winHeight;
- _d3dpp.PresentationInterval = D3DPRESENT_INTERVAL_IMMEDIATE;
- }
-
- if (InitializeD3D(_hWnd, &_d3dpp) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "VideoRenderDirect3D9::InitDevice failed in InitializeD3D");
- return -1;
- }
-
- // Turn off culling, so we see the front and back of the triangle
- _pd3dDevice->SetRenderState(D3DRS_CULLMODE, D3DCULL_NONE);
-
- // Turn off D3D lighting, since we are providing our own vertex colors
- _pd3dDevice->SetRenderState(D3DRS_LIGHTING, FALSE);
-
- // Settings for alpha blending
- _pd3dDevice->SetRenderState(D3DRS_ALPHABLENDENABLE, TRUE);
- _pd3dDevice->SetRenderState(D3DRS_SRCBLEND, D3DBLEND_SRCALPHA);
- _pd3dDevice->SetRenderState(D3DRS_DESTBLEND, D3DBLEND_INVSRCALPHA);
- //_pd3dDevice->SetTextureStageState(0,D3DTSS_ALPHAOP,D3DTOP_SELECTARG1);
- //_pd3dDevice->SetTextureStageState(0,D3DTSS_ALPHAARG1,D3DTA_TEXTURE);
-
- // Initialize Vertices
- CUSTOMVERTEX Vertices[] = {
- //front
- { -1.0f, -1.0f, 0.0f, 0xffffffff, 0, 1 }, { -1.0f, 1.0f, 0.0f,
- 0xffffffff, 0, 0 },
- { 1.0f, -1.0f, 0.0f, 0xffffffff, 1, 1 }, { 1.0f, 1.0f, 0.0f,
- 0xffffffff, 1, 0 } };
-
- // Create the vertex buffer.
- if (FAILED(_pd3dDevice->CreateVertexBuffer(sizeof(Vertices), 0,
- D3DFVF_CUSTOMVERTEX,
- D3DPOOL_DEFAULT, &_pVB, NULL )))
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Failed to create the vertex buffer.");
- return -1;
- }
-
- // Now we fill the vertex buffer.
- VOID* pVertices;
- if (FAILED(_pVB->Lock(0, sizeof(Vertices), (void**) &pVertices, 0)))
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Failed to lock the vertex buffer.");
- return -1;
- }
- memcpy(pVertices, Vertices, sizeof(Vertices));
- _pVB->Unlock();
-
- return 0;
-}
-
-WebRtc_Word32 VideoRenderDirect3D9::Init()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "VideoRenderDirect3D9::Init");
-
- CriticalSectionScoped cs(_refD3DCritsect);
-
- // Start rendering thread...
- if (!_screenUpdateThread)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Thread not created");
- return -1;
- }
- unsigned int threadId;
- _screenUpdateThread->Start(threadId);
-
- // Start the event triggering the render process
- unsigned int monitorFreq = 60;
- DEVMODE dm;
- // initialize the DEVMODE structure
- ZeroMemory(&dm, sizeof(dm));
- dm.dmSize = sizeof(dm);
- if (0 != EnumDisplaySettings(NULL, ENUM_CURRENT_SETTINGS, &dm))
- {
- monitorFreq = dm.dmDisplayFrequency;
- }
- _screenUpdateEvent->StartTimer(true, 1000 / monitorFreq);
-
- return InitDevice();
-}
-
-WebRtc_Word32 VideoRenderDirect3D9::ChangeWindow(void* window)
-{
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
- return -1;
-}
-
-int VideoRenderDirect3D9::UpdateRenderSurface()
-{
- CriticalSectionScoped cs(_refD3DCritsect);
-
- // Check if there are any updated buffers
- bool updated = false;
- std::map<int, D3D9Channel*>::iterator it;
- it = _d3dChannels.begin();
- while (it != _d3dChannels.end())
- {
-
- D3D9Channel* channel = it->second;
- channel->IsUpdated(updated);
- if (updated)
- {
- break;
- }
- it++;
- }
- //nothing is updated, continue
- if (!updated)
- return -1;
-
- // Clear the backbuffer to a black color
- _pd3dDevice->Clear(0, NULL, D3DCLEAR_TARGET, D3DCOLOR_XRGB(0, 0, 0), 1.0f,
- 0);
-
- // Begin the scene
- if (SUCCEEDED(_pd3dDevice->BeginScene()))
- {
- _pd3dDevice->SetStreamSource(0, _pVB, 0, sizeof(CUSTOMVERTEX));
- _pd3dDevice->SetFVF(D3DFVF_CUSTOMVERTEX);
-
- D3DXMATRIX matWorld;
- D3DXMATRIX matWorldTemp;
-
- //draw all the channels
- //get texture from the channels
- LPDIRECT3DTEXTURE9 textureFromChannel = NULL;
- DWORD textureWidth, textureHeight;
-
- std::multimap<int, unsigned int>::reverse_iterator it;
- it = _d3dZorder.rbegin();
- while (it != _d3dZorder.rend())
- {
- // loop through all channels and streams in Z order
- int channel = it->second & 0x0000ffff;
-
- std::map<int, D3D9Channel*>::iterator ddIt;
- ddIt = _d3dChannels.find(channel);
- if (ddIt != _d3dChannels.end())
- {
- // found the channel
- D3D9Channel* channelObj = ddIt->second;
- if (channelObj)
- {
- textureFromChannel = channelObj->GetTexture();
- textureWidth = channelObj->GetTextureWidth();
- textureHeight = channelObj->GetTextureHeight();
-
- WebRtc_UWord32 zOrder;
- float startWidth, startHeight, stopWidth, stopHeight;
- channelObj->GetStreamSettings(0, zOrder, startWidth,
- startHeight, stopWidth,
- stopHeight);
-
- //draw the video stream
- UpdateVerticeBuffer(_pVB, 0, startWidth, startHeight,
- stopWidth, stopHeight);
- _pd3dDevice->SetTexture(0, textureFromChannel);
- _pd3dDevice->DrawPrimitive(D3DPT_TRIANGLESTRIP, 0, 2);
-
- //Notice channel that this frame as been rendered
- channelObj->RenderOffFrame();
- }
- }
- it++;
- }
-
- //draw the logo
- if (_pTextureLogo)
- {
- UpdateVerticeBuffer(_pVB, 0, _logoLeft, _logoTop, _logoRight,
- _logoBottom);
- _pd3dDevice->SetTexture(0, _pTextureLogo);
- _pd3dDevice->DrawPrimitive(D3DPT_TRIANGLESTRIP, 0, 2);
- }
-
- // End the scene
- _pd3dDevice->EndScene();
- }
-
- // Present the backbuffer contents to the display
- _pd3dDevice->Present(NULL, NULL, NULL, NULL );
-
- return 0;
-}
-
-//set the alpha value of the pixal with a particular colorkey as 0
-int VideoRenderDirect3D9::SetTransparentColor(LPDIRECT3DTEXTURE9 pTexture,
- DDCOLORKEY* transparentColorKey,
- DWORD width,
- DWORD height)
-{
- D3DLOCKED_RECT lr;
- if (!pTexture)
- return -1;
-
- CriticalSectionScoped cs(_refD3DCritsect);
- if (SUCCEEDED(pTexture->LockRect(0, &lr, NULL, D3DLOCK_DISCARD)))
- {
- for (DWORD y = 0; y < height; y++)
- {
- DWORD dwOffset = y * width;
-
- for (DWORD x = 0; x < width; x)
- {
- DWORD a = (DWORD) 0;
-
- DWORD temp = ((DWORD*) lr.pBits)[dwOffset + x];
- if ((temp & 0x00FFFFFF)
- == transparentColorKey->dwColorSpaceLowValue)
- {
- temp &= 0x00FFFFFF;
- }
- else
- {
- temp |= 0xFF000000;
- }
- ((DWORD*) lr.pBits)[dwOffset + x] = temp;
- x++;
- }
- }
- pTexture->UnlockRect(0);
- return 0;
- }
- return -1;
-}
-
-/*
- *
- * Rendering process
- *
- */
-bool VideoRenderDirect3D9::ScreenUpdateThreadProc(void* obj)
-{
- return static_cast<VideoRenderDirect3D9*> (obj)->ScreenUpdateProcess();
-}
-
-bool VideoRenderDirect3D9::ScreenUpdateProcess()
-{
- _screenUpdateEvent->Wait(100);
-
- if (!_screenUpdateThread)
- {
- //stop the thread
- return false;
- }
- if (!_pd3dDevice)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "d3dDevice not created.");
- return true;
- }
-
- HRESULT hr = _pd3dDevice->TestCooperativeLevel();
-
- if (SUCCEEDED(hr))
- {
- UpdateRenderSurface();
- }
-
- if (hr == D3DERR_DEVICELOST)
- {
- //Device is lost and cannot be reset yet
-
- }
- else if (hr == D3DERR_DEVICENOTRESET)
- {
- //Lost but we can reset it now
- //Note: the standard way is to call Reset, however for some reason doesn't work here.
- //so we will release the device and create it again.
- ResetDevice();
- }
-
- return true;
-}
-
-int VideoRenderDirect3D9::CloseDevice()
-{
- CriticalSectionScoped cs(_refD3DCritsect);
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "VideoRenderDirect3D9::CloseDevice");
-
- if (_pTextureLogo != NULL)
- {
- _pTextureLogo->Release();
- _pTextureLogo = NULL;
- }
-
- if (_pVB != NULL)
- {
- _pVB->Release();
- _pVB = NULL;
- }
-
- if (_pd3dDevice != NULL)
- {
- _pd3dDevice->Release();
- _pd3dDevice = NULL;
- }
-
- if (_pD3D != NULL)
- {
- _pD3D->Release();
- _pD3D = NULL;
- }
-
- if (_pd3dSurface != NULL)
- _pd3dSurface->Release();
- return 0;
-}
-
-D3D9Channel* VideoRenderDirect3D9::GetD3DChannel(int channel)
-{
- std::map<int, D3D9Channel*>::iterator ddIt;
- ddIt = _d3dChannels.find(channel & 0x0000ffff);
- D3D9Channel* ddobj = NULL;
- if (ddIt != _d3dChannels.end())
- {
- ddobj = ddIt->second;
- }
- if (ddobj == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Direct3D render failed to find channel");
- return NULL;
- }
- return ddobj;
-}
-
-WebRtc_Word32 VideoRenderDirect3D9::DeleteChannel(const WebRtc_UWord32 streamId)
-{
- CriticalSectionScoped cs(_refD3DCritsect);
-
-
- std::multimap<int, unsigned int>::iterator it;
- it = _d3dZorder.begin();
- while (it != _d3dZorder.end())
- {
- if ((streamId & 0x0000ffff) == (it->second & 0x0000ffff))
- {
- it = _d3dZorder.erase(it);
- break;
- }
- it++;
- }
-
- std::map<int, D3D9Channel*>::iterator ddIt;
- ddIt = _d3dChannels.find(streamId & 0x0000ffff);
- D3D9Channel* ddobj = NULL;
- if (ddIt != _d3dChannels.end())
- {
- delete ddIt->second;
- _d3dChannels.erase(ddIt);
- return 0;
- }
- return -1;
-}
-
-VideoRenderCallback* VideoRenderDirect3D9::CreateChannel(const WebRtc_UWord32 channel,
- const WebRtc_UWord32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_refD3DCritsect);
-
- //FIXME this should be done in VideoAPIWindows? stop the frame deliver first
- //remove the old channel
- DeleteChannel(channel);
-
- D3D9Channel* d3dChannel = new D3D9Channel(_pd3dDevice,
- &_refD3DCritsect, _trace);
- d3dChannel->SetStreamSettings(0, zOrder, left, top, right, bottom);
-
- // store channel
- _d3dChannels[channel & 0x0000ffff] = d3dChannel;
-
- // store Z order
- // default streamID is 0
- _d3dZorder.insert(
- std::pair<int, unsigned int>(zOrder, channel & 0x0000ffff));
-
- return d3dChannel;
-}
-
-WebRtc_Word32 VideoRenderDirect3D9::GetStreamSettings(const WebRtc_UWord32 channel,
- const WebRtc_UWord16 streamId,
- WebRtc_UWord32& zOrder,
- float& left,
- float& top,
- float& right,
- float& bottom)
-{
- std::map<int, D3D9Channel*>::iterator ddIt;
- ddIt = _d3dChannels.find(channel & 0x0000ffff);
- D3D9Channel* ddobj = NULL;
- if (ddIt != _d3dChannels.end())
- {
- ddobj = ddIt->second;
- }
- if (ddobj == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Direct3D render failed to find channel");
- return -1;
- }
- // Only allow one stream per channel, demuxing is
- return ddobj->GetStreamSettings(0, zOrder, left, top, right, bottom);
- //return ddobj->GetStreamSettings(streamId, zOrder, left, top, right, bottom);
-}
-
-int VideoRenderDirect3D9::UpdateVerticeBuffer(LPDIRECT3DVERTEXBUFFER9 pVB,
- int offset,
- float startWidth,
- float startHeight,
- float stopWidth,
- float stopHeight)
-{
- if (pVB == NULL)
- return -1;
-
- float left, right, top, bottom;
-
- //update the vertice buffer
- //0,1 => -1,1
- left = startWidth * 2 - 1;
- right = stopWidth * 2 - 1;
-
- //0,1 => 1,-1
- top = 1 - startHeight * 2;
- bottom = 1 - stopHeight * 2;
-
- CUSTOMVERTEX newVertices[] = {
- //logo
- { left, bottom, 0.0f, 0xffffffff, 0, 1 }, { left, top, 0.0f,
- 0xffffffff, 0, 0 },
- { right, bottom, 0.0f, 0xffffffff, 1, 1 }, { right, top, 0.0f,
- 0xffffffff, 1, 0 }, };
- // Now we fill the vertex buffer.
- VOID* pVertices;
- if (FAILED(pVB->Lock(sizeof(CUSTOMVERTEX) * offset, sizeof(newVertices),
- (void**) &pVertices, 0)))
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Failed to lock the vertex buffer.");
- return -1;
- }
- memcpy(pVertices, newVertices, sizeof(newVertices));
- pVB->Unlock();
-
- return 0;
-}
-
-WebRtc_Word32 VideoRenderDirect3D9::StartRender()
-{
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
- return 0;
-}
-
-WebRtc_Word32 VideoRenderDirect3D9::StopRender()
-{
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
- return 0;
-}
-
-bool VideoRenderDirect3D9::IsFullScreen()
-{
- return _fullScreen;
-}
-
-WebRtc_Word32 VideoRenderDirect3D9::SetCropping(const WebRtc_UWord32 channel,
- const WebRtc_UWord16 streamId,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
- return 0;
-}
-
-WebRtc_Word32 VideoRenderDirect3D9::SetTransparentBackground(
- const bool enable)
-{
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
- return 0;
-}
-
-WebRtc_Word32 VideoRenderDirect3D9::SetText(const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 colorText,
- const WebRtc_UWord32 colorBg,
- const float left,
- const float top,
- const float rigth,
- const float bottom)
-{
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
- return 0;
-}
-
-WebRtc_Word32 VideoRenderDirect3D9::SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- if (!bitMap)
- {
- if (_pTextureLogo != NULL)
- {
- _pTextureLogo->Release();
- _pTextureLogo = NULL;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1, "Remove bitmap.");
- return 0;
- }
-
- // sanity
- if (left > 1.0f || left < 0.0f ||
- top > 1.0f || top < 0.0f ||
- right > 1.0f || right < 0.0f ||
- bottom > 1.0f || bottom < 0.0f)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Direct3D SetBitmap invalid parameter");
- return -1;
- }
-
- if ((bottom <= top) || (right <= left))
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Direct3D SetBitmap invalid parameter");
- return -1;
- }
-
- CriticalSectionScoped cs(_refD3DCritsect);
-
- unsigned char* srcPtr;
- HGDIOBJ oldhand;
- BITMAPINFO pbi;
- BITMAP bmap;
- HDC hdcNew;
- hdcNew = CreateCompatibleDC(0);
- // Fill out the BITMAP structure.
- GetObject((HBITMAP)bitMap, sizeof(bmap), &bmap);
- //Select the bitmap handle into the new device context.
- oldhand = SelectObject(hdcNew, (HGDIOBJ) bitMap);
- // we are done with this object
- DeleteObject(oldhand);
- pbi.bmiHeader.biSize = 40;
- pbi.bmiHeader.biWidth = bmap.bmWidth;
- pbi.bmiHeader.biHeight = bmap.bmHeight;
- pbi.bmiHeader.biPlanes = 1;
- pbi.bmiHeader.biBitCount = bmap.bmBitsPixel;
- pbi.bmiHeader.biCompression = BI_RGB;
- pbi.bmiHeader.biSizeImage = bmap.bmWidth * bmap.bmHeight * 3;
- srcPtr = new unsigned char[bmap.bmWidth * bmap.bmHeight * 4];
- // the original un-stretched image in RGB24
- int pixelHeight = GetDIBits(hdcNew, (HBITMAP)bitMap, 0, bmap.bmHeight, srcPtr, &pbi,
- DIB_RGB_COLORS);
- if (pixelHeight == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Direct3D failed to GetDIBits in SetBitmap");
- return -1;
- }
- DeleteDC(hdcNew);
- if (pbi.bmiHeader.biBitCount != 24 && pbi.bmiHeader.biBitCount != 32)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Direct3D failed to SetBitmap invalid bit depth");
- return -1;
- }
-
- HRESULT ret;
- //release the previous logo texture
- if (_pTextureLogo != NULL)
- {
- _pTextureLogo->Release();
- _pTextureLogo = NULL;
- }
- ret = _pd3dDevice->CreateTexture(bmap.bmWidth, bmap.bmHeight, 1, 0,
- D3DFMT_A8R8G8B8, D3DPOOL_MANAGED,
- &_pTextureLogo, NULL);
- if (FAILED(ret))
- {
- _pTextureLogo = NULL;
- return -1;
- }
- if (!_pTextureLogo)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Texture for rendering not initialized.");
- return -1;
- }
-
- D3DLOCKED_RECT lr;
- if (FAILED(_pTextureLogo->LockRect(0, &lr, NULL, 0)))
- {
- return -1;
- }
- unsigned char* dstPtr = (UCHAR*) lr.pBits;
- int pitch = bmap.bmWidth * 4;
-
- if (pbi.bmiHeader.biBitCount == 24)
- {
- ConvertRGB24ToARGB(srcPtr, dstPtr, bmap.bmWidth, bmap.bmHeight, 0);
- }
- else
- {
- unsigned char* srcTmp = srcPtr + (bmap.bmWidth * 4) * (bmap.bmHeight - 1);
- for (int i = 0; i < bmap.bmHeight; ++i)
- {
- memcpy(dstPtr, srcTmp, bmap.bmWidth * 4);
- srcTmp -= bmap.bmWidth * 4;
- dstPtr += pitch;
- }
- }
-
- delete srcPtr;
- if (FAILED(_pTextureLogo->UnlockRect(0)))
- {
- return -1;
- }
-
- if (colorKey)
- {
- DDCOLORKEY* ddColorKey =
- static_cast<DDCOLORKEY*> (const_cast<void*> (colorKey));
- SetTransparentColor(_pTextureLogo, ddColorKey, bmap.bmWidth,
- bmap.bmHeight);
- }
-
- //update the vertice buffer
- //0,1 => -1,1
- _logoLeft = left;
- _logoRight = right;
-
- //0,1 => 1,-1
- _logoTop = top;
- _logoBottom = bottom;
-
- return 0;
-
-}
-
-WebRtc_Word32 VideoRenderDirect3D9::GetGraphicsMemory(WebRtc_UWord64& totalMemory,
- WebRtc_UWord64& availableMemory)
-{
- if (_totalMemory == -1 || _availableMemory == -1)
- {
- totalMemory = 0;
- availableMemory = 0;
- return -1;
- }
- totalMemory = _totalMemory;
- availableMemory = _availableMemory;
- return 0;
-}
-
-WebRtc_Word32 VideoRenderDirect3D9::ConfigureRenderer(const WebRtc_UWord32 channel,
- const WebRtc_UWord16 streamId,
- const unsigned int zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- std::map<int, D3D9Channel*>::iterator ddIt;
- ddIt = _d3dChannels.find(channel & 0x0000ffff);
- D3D9Channel* ddobj = NULL;
- if (ddIt != _d3dChannels.end())
- {
- ddobj = ddIt->second;
- }
- if (ddobj == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Direct3D render failed to find channel");
- return -1;
- }
- // Only allow one stream per channel, demuxing is
- ddobj->SetStreamSettings(0, zOrder, left, top, right, bottom);
-
- return 0;
-}
-
-} //namespace webrtc
-
diff --git a/trunk/src/modules/video_render/main/source/windows/video_render_direct3d9.h b/trunk/src/modules/video_render/main/source/windows/video_render_direct3d9.h
deleted file mode 100644
index 6d6fef3..0000000
--- a/trunk/src/modules/video_render/main/source/windows/video_render_direct3d9.h
+++ /dev/null
@@ -1,267 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_DIRECT3D9_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_DIRECT3D9_H_
-
-// WebRtc includes
-#include "i_video_render_win.h"
-
-#include <d3d9.h>
-#include <d3dx9.h>
-#include "ddraw.h"
-
-#include <Map>
-
-// Added
-#include "video_render_defines.h"
-
-#pragma comment(lib, "d3d9.lib") // located in DirectX SDK
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-class Trace;
-class ThreadWrapper;
-
-class D3D9Channel: public VideoRenderCallback
-{
-public:
- D3D9Channel(LPDIRECT3DDEVICE9 pd3DDevice,
- CriticalSectionWrapper* critSect, Trace* trace);
-
- virtual ~D3D9Channel();
-
- // Inherited from VideoRencerCallback, called from VideoAPI class.
- // Called when the incomming frame size and/or number of streams in mix changes
- virtual int FrameSizeChange(int width, int height, int numberOfStreams);
-
- // A new frame is delivered
- virtual int DeliverFrame(unsigned char* buffer,
- int bufferSize,
- unsigned int timeStame90kHz);
- virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId,
- VideoFrame& videoFrame);
-
- // Called to check if the video frame is updated.
- int IsUpdated(bool& isUpdated);
- // Called after the video frame has been render to the screen
- int RenderOffFrame();
- // Called to get the texture that contains the video frame
- LPDIRECT3DTEXTURE9 GetTexture();
- // Called to get the texture(video frame) size
- int GetTextureWidth();
- int GetTextureHeight();
- //
- void SetStreamSettings(WebRtc_UWord16 streamId,
- WebRtc_UWord32 zOrder,
- float startWidth,
- float startHeight,
- float stopWidth,
- float stopHeight);
- int GetStreamSettings(WebRtc_UWord16 streamId,
- WebRtc_UWord32& zOrder,
- float& startWidth,
- float& startHeight,
- float& stopWidth,
- float& stopHeight);
-
- int ReleaseTexture();
- int RecreateTexture(LPDIRECT3DDEVICE9 pd3DDevice);
-
-protected:
-
-private:
- //critical section passed from the owner
- CriticalSectionWrapper* _critSect;
- LPDIRECT3DDEVICE9 _pd3dDevice;
- LPDIRECT3DTEXTURE9 _pTexture;
-
- bool _bufferIsUpdated;
- // the frame size
- int _width;
- int _height;
- //sream settings
- //TODO support multiple streams in one channel
- WebRtc_UWord16 _streamId;
- WebRtc_UWord32 _zOrder;
- float _startWidth;
- float _startHeight;
- float _stopWidth;
- float _stopHeight;
-};
-
-class VideoRenderDirect3D9: IVideoRenderWin
-{
-public:
- VideoRenderDirect3D9(Trace* trace, HWND hWnd, bool fullScreen);
- ~VideoRenderDirect3D9();
-
-public:
- //IVideoRenderWin
-
- /**************************************************************************
- *
- * Init
- *
- ***************************************************************************/
- virtual WebRtc_Word32 Init();
-
- /**************************************************************************
- *
- * Incoming Streams
- *
- ***************************************************************************/
- virtual VideoRenderCallback
- * CreateChannel(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 DeleteChannel(const WebRtc_UWord32 streamId);
-
- virtual WebRtc_Word32 GetStreamSettings(const WebRtc_UWord32 channel,
- const WebRtc_UWord16 streamId,
- WebRtc_UWord32& zOrder,
- float& left,
- float& top,
- float& right,
- float& bottom);
-
- /**************************************************************************
- *
- * Start/Stop
- *
- ***************************************************************************/
-
- virtual WebRtc_Word32 StartRender();
- virtual WebRtc_Word32 StopRender();
-
- /**************************************************************************
- *
- * Properties
- *
- ***************************************************************************/
-
- virtual bool IsFullScreen();
-
- virtual WebRtc_Word32 SetCropping(const WebRtc_UWord32 channel,
- const WebRtc_UWord16 streamId,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 channel,
- const WebRtc_UWord16 streamId,
- const unsigned int zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
-
- virtual WebRtc_Word32 ChangeWindow(void* window);
-
- virtual WebRtc_Word32 GetGraphicsMemory(WebRtc_UWord64& totalMemory,
- WebRtc_UWord64& availableMemory);
-
- virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 colorText,
- const WebRtc_UWord32 colorBg,
- const float left,
- const float top,
- const float rigth,
- const float bottom);
-
- virtual WebRtc_Word32 SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
-public:
- // Get a channel by channel id
- D3D9Channel* GetD3DChannel(int channel);
- int UpdateRenderSurface();
-
-protected:
- // The thread rendering the screen
- static bool ScreenUpdateThreadProc(void* obj);
- bool ScreenUpdateProcess();
-
-private:
- // Init/close the d3d device
- int InitDevice();
- int CloseDevice();
-
- // Transparent related functions
- int SetTransparentColor(LPDIRECT3DTEXTURE9 pTexture,
- DDCOLORKEY* transparentColorKey,
- DWORD width,
- DWORD height);
-
- CriticalSectionWrapper& _refD3DCritsect;
- Trace* _trace;
- ThreadWrapper* _screenUpdateThread;
- EventWrapper* _screenUpdateEvent;
-
- HWND _hWnd;
- bool _fullScreen;
- RECT _originalHwndRect;
- //FIXME we probably don't need this since all the information can be get from _d3dChannels
- int _channel;
- //Window size
- UINT _winWidth;
- UINT _winHeight;
-
- // Device
- LPDIRECT3D9 _pD3D; // Used to create the D3DDevice
- LPDIRECT3DDEVICE9 _pd3dDevice; // Our rendering device
- LPDIRECT3DVERTEXBUFFER9 _pVB; // Buffer to hold Vertices
- LPDIRECT3DTEXTURE9 _pTextureLogo;
-
- std::map<int, D3D9Channel*> _d3dChannels;
- std::multimap<int, unsigned int> _d3dZorder;
-
- // The position where the logo will be placed
- float _logoLeft;
- float _logoTop;
- float _logoRight;
- float _logoBottom;
-
- typedef HRESULT (WINAPI *DIRECT3DCREATE9EX)(UINT SDKVersion, IDirect3D9Ex**);
- LPDIRECT3DSURFACE9 _pd3dSurface;
-
- DWORD GetVertexProcessingCaps();
- int InitializeD3D(HWND hWnd, D3DPRESENT_PARAMETERS* pd3dpp);
-
- D3DPRESENT_PARAMETERS _d3dpp;
- int ResetDevice();
-
- int UpdateVerticeBuffer(LPDIRECT3DVERTEXBUFFER9 pVB, int offset,
- float startWidth, float startHeight,
- float stopWidth, float stopHeight);
-
- //code for providing graphics settings
- DWORD _totalMemory;
- DWORD _availableMemory;
-};
-
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_DIRECT3D9_H_
diff --git a/trunk/src/modules/video_render/main/source/windows/video_render_directdraw.cc b/trunk/src/modules/video_render/main/source/windows/video_render_directdraw.cc
deleted file mode 100644
index 034f710..0000000
--- a/trunk/src/modules/video_render/main/source/windows/video_render_directdraw.cc
+++ /dev/null
@@ -1,4012 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_render_directdraw.h"
-#include "video_render_windows_impl.h"
-#include "Windows.h"
-#include <ddraw.h>
-#include <assert.h>
-#include <initguid.h>
-#include <MMSystem.h> // timeGetTime
-DEFINE_GUID( IID_IDirectDraw7,0x15e65ec0,0x3b9c,0x11d2,0xb9,0x2f,0x00,0x60,0x97,0x97,0xea,0x5b );
-
-#include "thread_wrapper.h"
-#include "event_wrapper.h"
-#include "trace.h"
-#include "critical_section_wrapper.h"
-//#include "VideoErrors.h"
-
-// Added
-#include "module_common_types.h"
-
-#pragma warning(disable: 4355) // 'this' : used in base member initializer list
-// picture in picture do we need overlay? answer no we can blit directly
-// conference is easy since we can blt the quadrants seperatly
-
-// To determine if the driver supports DMA, retrieve the driver capabilities by calling the IDirectDraw::GetCaps method,
-// then look for DDBLTCAPS_READSYSMEM and/or DDBLTCAPS_WRITESYSMEM. If either of these flags is set, the device supports DMA.
-// Blt with SRCCOPY should do this can we use it?
-// investigate DDLOCK_NOSYSLOCK
-
-namespace webrtc {
-
-#define EXTRACT_BITS_RL(the_val, bits_start, bits_len) ((the_val >> (bits_start - 1)) & ((1 << bits_len) - 1))
-
-WindowsThreadCpuUsage::WindowsThreadCpuUsage() :
- _lastGetCpuUsageTime(0),
- _lastCpuUsageTime(0),
- _hThread(::GetCurrentThread()),
- _cores(0),
- _lastCpuUsage(0)
-{
-
- DWORD_PTR pmask, smask;
- DWORD access = PROCESS_QUERY_INFORMATION;
- if (GetProcessAffinityMask(
- OpenProcess(access, false, GetCurrentProcessId()),
- &pmask, &smask) != 0)
- {
-
- for (int i = 1; i < 33; i++)
- {
- if (EXTRACT_BITS_RL(pmask,i,1) == 0)
- {
- break;
- }
- _cores++;
- }
- //sanity
- if (_cores > 32)
- {
- _cores = 32;
- }
- if (_cores < 1)
- {
- _cores = 1;
- }
- }
- else
- {
- _cores = 1;
- }
- GetCpuUsage();
-}
-
-//in % since last call
-int WindowsThreadCpuUsage::GetCpuUsage()
-{
- DWORD now = timeGetTime();
-
- _int64 newTime = 0;
- FILETIME creationTime;
- FILETIME exitTime;
- _int64 kernelTime = 0;
- _int64 userTime = 0;
- if (GetThreadTimes(_hThread, (FILETIME*) &creationTime, &exitTime,
- (FILETIME*) &kernelTime, (FILETIME*) &userTime) != 0)
- {
- newTime = (kernelTime + userTime);
- }
- if (newTime == 0)
- {
- _lastGetCpuUsageTime = now;
- return _lastCpuUsage;
- }
-
- // calculate the time difference since last call
- const DWORD diffTime = (now - _lastGetCpuUsageTime);
- _lastGetCpuUsageTime = now;
-
- if (newTime < _lastCpuUsageTime)
- {
- _lastCpuUsageTime = newTime;
- return _lastCpuUsage;
- }
- const int cpuDiff = (int) (newTime - _lastCpuUsageTime) / 10000;
- _lastCpuUsageTime = newTime;
-
- // calculate the CPU usage
-
- _lastCpuUsage = (int) (float((cpuDiff * 100)) / (diffTime * _cores) + 0.5f);
-
- if (_lastCpuUsage > 100)
- {
- _lastCpuUsage = 100;
- }
- return _lastCpuUsage;
-
-}
-
-DirectDrawStreamSettings::DirectDrawStreamSettings() :
- _startWidth(0.0F),
- _stopWidth(1.0F),
- _startHeight(0.0F),
- _stopHeight(1.0F),
- _cropStartWidth(0.0F),
- _cropStopWidth(1.0F),
- _cropStartHeight(0.0F),
- _cropStopHeight(1.0F)
-{
-}
-;
-
-DirectDrawBitmapSettings::DirectDrawBitmapSettings() :
- _transparentBitMap(NULL),
- _transparentBitmapLeft(0.0f),
- _transparentBitmapRight(1.0f),
- _transparentBitmapTop(0.0f),
- _transparentBitmapBottom(1.0f),
- _transparentBitmapWidth(0),
- _transparentBitmapHeight(0),
- _transparentBitmapColorKey(NULL),
- _transparentBitmapSurface(NULL)
-{
-}
-;
-
-DirectDrawBitmapSettings::~DirectDrawBitmapSettings()
-{
- if (_transparentBitmapColorKey)
- {
- delete _transparentBitmapColorKey;
- }
- if (_transparentBitmapSurface)
- {
- _transparentBitmapSurface->Release();
- }
- _transparentBitmapColorKey = NULL;
- _transparentBitmapSurface = NULL;
-}
-;
-
-int DirectDrawBitmapSettings::SetBitmap(Trace* _trace,
- DirectDraw* directDraw)
-{
- VideoFrame tempVideoBuffer;
- HGDIOBJ oldhand;
- BITMAPINFO pbi;
- BITMAP bmap;
- HDC hdcNew;
-
- hdcNew = CreateCompatibleDC(0);
-
- // Fill out the BITMAP structure.
- GetObject(_transparentBitMap, sizeof(bmap), &bmap);
-
- //Select the bitmap handle into the new device context.
- oldhand = SelectObject(hdcNew, (HGDIOBJ) _transparentBitMap);
-
- // we are done with this object
- DeleteObject(oldhand);
-
- pbi.bmiHeader.biSize = 40;
- pbi.bmiHeader.biWidth = bmap.bmWidth;
- pbi.bmiHeader.biHeight = bmap.bmHeight;
- pbi.bmiHeader.biPlanes = 1;
- pbi.bmiHeader.biBitCount = bmap.bmBitsPixel;
- pbi.bmiHeader.biCompression = BI_RGB;
- pbi.bmiHeader.biSizeImage = bmap.bmWidth * bmap.bmHeight * 3;
-
- tempVideoBuffer.VerifyAndAllocate(bmap.bmWidth * bmap.bmHeight * 4);
-
- // the original un-stretched image in RGB24
- // todo is there another struct for pbi purify reports read of 24 bytes larger than size
- int pixelHeight = GetDIBits(hdcNew, _transparentBitMap, 0, bmap.bmHeight,
- tempVideoBuffer.Buffer(), &pbi, DIB_RGB_COLORS);
- if (pixelHeight == 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw failed to GetDIBits in SetBitmap.");
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
-
- DeleteDC(hdcNew);
-
- if (pbi.bmiHeader.biBitCount != 24 && pbi.bmiHeader.biBitCount != 32)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw failed to SetBitmap invalid bit depth");
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
-
- DirectDrawSurfaceDesc ddsd;
- memset(&ddsd, 0, sizeof(ddsd));
- ddsd.dwSize = sizeof(ddsd);
- ddsd.dwFlags = DDSD_CAPS | DDSD_HEIGHT | DDSD_WIDTH | DDSD_PIXELFORMAT;
- ddsd.ddsCaps.dwCaps = DDSCAPS_VIDEOMEMORY;
- ddsd.dwHeight = bmap.bmHeight;
- ddsd.dwWidth = bmap.bmWidth;
-
- ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
- ddsd.ddpfPixelFormat.dwFlags = DDPF_RGB;
-
- _transparentBitmapWidth = bmap.bmWidth;
- _transparentBitmapHeight = bmap.bmHeight;
-
- ddsd.ddpfPixelFormat.dwRGBBitCount = 32;
- ddsd.ddpfPixelFormat.dwRBitMask = 0xff0000;
- ddsd.ddpfPixelFormat.dwGBitMask = 0xff00;
- ddsd.ddpfPixelFormat.dwBBitMask = 0xff;
- ddsd.ddpfPixelFormat.dwRGBAlphaBitMask = 0;
-
- if (_transparentBitmapSurface)
- {
- _transparentBitmapSurface->Release();
- _transparentBitmapSurface = NULL;
- }
-
- HRESULT ddrval =
- directDraw->CreateSurface(&ddsd, &_transparentBitmapSurface, NULL);
- if (FAILED(ddrval))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "DirectDraw failed to CreateSurface _transparentBitmapSurface: 0x%x",
- ddrval);
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
-
- memset(&ddsd, 0, sizeof(DDSURFACEDESC));
- ddsd.dwSize = sizeof(DDSURFACEDESC);
- ddrval = _transparentBitmapSurface->Lock(NULL, &ddsd, DDLOCK_WAIT, NULL);
- if (ddrval == DDERR_SURFACELOST)
- {
- ddrval = _transparentBitmapSurface->Restore();
- if (ddrval != DD_OK)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
- "DirectDraw failed to restore lost _transparentBitmapSurface");
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw restored lost _transparentBitmapSurface");
-
- ddrval
- = _transparentBitmapSurface->Lock(NULL, &ddsd, DDLOCK_WAIT,
- NULL);
- if (ddrval != DD_OK)
- {
- WEBRTC_TRACE(
- kTraceInfo,
- kTraceVideo,
- -1,
- "DirectDraw lock error 0x%x _transparentBitmapSurface",
- ddrval);
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- }
- unsigned char* dstPtr = (unsigned char*) ddsd.lpSurface;
- unsigned char* srcPtr = (unsigned char*) tempVideoBuffer.Buffer();
-
- int pitch = bmap.bmWidth * 4;
- if (ddsd.dwFlags & DDSD_PITCH)
- {
- pitch = ddsd.lPitch;
- }
-
- if (pbi.bmiHeader.biBitCount == 24)
- {
- ConvertRGB24ToARGB(srcPtr, dstPtr, bmap.bmWidth, bmap.bmHeight,
- 0);
- }
- else
- {
- srcPtr += (bmap.bmWidth * 4) * (bmap.bmHeight - 1);
-
- for (int i = 0; i < bmap.bmHeight; ++i)
- {
- memcpy(dstPtr, srcPtr, bmap.bmWidth * 4);
- srcPtr -= bmap.bmWidth * 4;
- dstPtr += pitch;
- }
- }
-
- _transparentBitmapSurface->Unlock(NULL);
- return 0;
-}
-/**
- *
- * DirectDrawTextSettings
- *
- */
-DirectDrawTextSettings::DirectDrawTextSettings() :
- _ptrText(NULL),
- _textLength(0),
- _colorRefText(RGB(255, 255, 255)), // white
- _colorRefBackground(RGB(0, 0, 0)), // black
- _textLeft(0.0f),
- _textRight(0.0f),
- _textTop(0.0f),
- _textBottom(0.0f),
- _transparent(true)
-{
-}
-
-DirectDrawTextSettings::~DirectDrawTextSettings()
-{
- if (_ptrText)
- {
- delete[] _ptrText;
- }
-}
-
-int DirectDrawTextSettings::SetText(const char* text, int textLength,
- COLORREF colorText, COLORREF colorBg,
- float left, float top, float right,
- float bottom)
-{
- if (_ptrText)
- {
- delete[] _ptrText;
- }
- _ptrText = new char[textLength];
- memcpy(_ptrText, text, textLength);
- _textLength = textLength;
- _colorRefText = colorText;
- _colorRefBackground = colorBg;
- //_transparent = transparent;
- _textLeft = left;
- _textRight = right;
- _textTop = top;
- _textBottom = bottom;
- return 0;
-}
-
-/**
- *
- * DirectDrawChannel
- *
- *
- */
-
-// this need to have a refcount dueto multiple HWNDS demux
-DirectDrawChannel::DirectDrawChannel(DirectDraw* directDraw,
- VideoType blitVideoType,
- VideoType incomingVideoType,
- VideoType screenVideoType,
- VideoRenderDirectDraw* owner) :
-
- _critSect(CriticalSectionWrapper::CreateCriticalSection()), _refCount(1),
- _width(0), _height(0), _numberOfStreams(0), _doubleBuffer(false),
- _directDraw(directDraw), _offScreenSurface(NULL),
- _offScreenSurfaceNext(NULL), _incomingVideoType(incomingVideoType),
- _blitVideoType(blitVideoType),
- _originalBlitVideoType(blitVideoType),
- _screenVideoType(screenVideoType), _deliverInScreenType(false),
- _owner(owner)
-{
- _directDraw->AddRef();
-}
-
-DirectDrawChannel::~DirectDrawChannel()
-{
- if (_directDraw)
- {
- _directDraw->Release();
- }
- if (_offScreenSurface)
- {
- _offScreenSurface->Release();
- }
- if (_offScreenSurfaceNext)
- {
- _offScreenSurfaceNext->Release();
- }
- std::map<unsigned long long, DirectDrawStreamSettings*>::iterator it =
- _streamIdToSettings.begin();
- while (it != _streamIdToSettings.end())
- {
- DirectDrawStreamSettings* streamSettings = it->second;
- if (streamSettings)
- {
- delete streamSettings;
- }
- it = _streamIdToSettings.erase(it);
- }
- delete _critSect;
-}
-
-void DirectDrawChannel::AddRef()
-{
- CriticalSectionScoped cs(*_critSect);
- _refCount++;
-}
-
-void DirectDrawChannel::Release()
-{
- bool deleteObj = false;
- _critSect->Enter();
- _refCount--;
- if (_refCount == 0)
- {
- deleteObj = true;
- }
- _critSect->Leave();
-
- if (deleteObj)
- {
- delete this;
- }
-}
-
-void DirectDrawChannel::SetStreamSettings(VideoRenderDirectDraw* DDobj,
- short streamId, float startWidth,
- float startHeight,
- float stopWidth, float stopHeight)
-{
- // we can save 5 bits due to 16 byte alignment of the pointer
- unsigned long long lookupID = reinterpret_cast<unsigned long long> (DDobj);
- lookupID &= 0xffffffffffffffe0;
- lookupID <<= 11;
- lookupID += streamId;
-
- CriticalSectionScoped cs(*_critSect);
-
- DirectDrawStreamSettings* streamSettings = NULL;
-
- std::map<unsigned long long, DirectDrawStreamSettings*>::iterator it =
- _streamIdToSettings.find(lookupID);
- if (it == _streamIdToSettings.end())
- {
- streamSettings = new DirectDrawStreamSettings();
- _streamIdToSettings[lookupID] = streamSettings;
- }
- else
- {
- streamSettings = it->second;
- }
-
- streamSettings->_startHeight = startHeight;
- streamSettings->_startWidth = startWidth;
- streamSettings->_stopWidth = stopWidth;
- streamSettings->_stopHeight = stopHeight;
-
- _offScreenSurfaceUpdated = false;
-}
-
-void DirectDrawChannel::SetStreamCropSettings(VideoRenderDirectDraw* DDObj,
- short streamId,
- float startWidth,
- float startHeight,
- float stopWidth,
- float stopHeight)
-{
- unsigned long long lookupID = reinterpret_cast<unsigned long long> (DDObj);
- lookupID &= 0xffffffffffffffe0;
- lookupID <<= 11;
- lookupID += streamId;
-
- CriticalSectionScoped cs(*_critSect);
-
- DirectDrawStreamSettings* streamSettings = NULL;
- std::map<unsigned long long, DirectDrawStreamSettings*>::iterator it =
- _streamIdToSettings.find(lookupID);
- if (it == _streamIdToSettings.end())
- {
- streamSettings = new DirectDrawStreamSettings();
- _streamIdToSettings[streamId] = streamSettings;
- }
- else
- {
- streamSettings = it->second;
- }
- streamSettings->_cropStartWidth = startWidth;
- streamSettings->_cropStopWidth = stopWidth;
- streamSettings->_cropStartHeight = startHeight;
- streamSettings->_cropStopHeight = stopHeight;
-}
-
-int DirectDrawChannel::GetStreamSettings(VideoRenderDirectDraw* DDObj,
- short streamId, float& startWidth,
- float& startHeight,
- float& stopWidth,
- float& stopHeight)
-{
- CriticalSectionScoped cs(*_critSect);
-
- unsigned long long lookupID = reinterpret_cast<unsigned long long> (DDObj);
- lookupID &= 0xffffffffffffffe0;
- lookupID <<= 11;
- lookupID += streamId;
-
- DirectDrawStreamSettings* streamSettings = NULL;
- std::map<unsigned long long, DirectDrawStreamSettings*>::iterator it =
- _streamIdToSettings.find(lookupID);
- if (it == _streamIdToSettings.end())
- {
- // Didn't find this stream...
- return -1;
- }
- streamSettings = it->second;
- startWidth = streamSettings->_startWidth;
- startHeight = streamSettings->_startHeight;
- stopWidth = streamSettings->_stopWidth;
- stopHeight = streamSettings->_stopHeight;
-
- return 0;
-}
-
-bool DirectDrawChannel::IsOffScreenSurfaceUpdated(VideoRenderDirectDraw* DDobj)
-{
- CriticalSectionScoped cs(*_critSect);
- return _offScreenSurfaceUpdated;
-}
-
-void DirectDrawChannel::GetLargestSize(RECT* mixingRect)
-{
- CriticalSectionScoped cs(*_critSect);
- if (mixingRect)
- {
- if (mixingRect->bottom < _height)
- {
- mixingRect->bottom = _height;
- }
- if (mixingRect->right < _width)
- {
- mixingRect->right = _width;
- }
- }
-}
-
-int DirectDrawChannel::ChangeDeliverColorFormat(bool useScreenType)
-{
- _deliverInScreenType = useScreenType;
- return FrameSizeChange(0, 0, 0);
-}
-
-WebRtc_Word32 DirectDrawChannel::RenderFrame(const WebRtc_UWord32 streamId,
- VideoFrame& videoFrame)
-{
- CriticalSectionScoped cs(*_critSect);
- if (_width != videoFrame.Width() || _height != videoFrame.Height())
- {
- if (FrameSizeChange(videoFrame.Width(), videoFrame.Height(), 1) == -1)
- {
- return -1;
- }
- }
- return DeliverFrame(videoFrame.Buffer(), videoFrame.Length(),
- videoFrame.TimeStamp());
-}
-
-int DirectDrawChannel::FrameSizeChange(int width, int height,
- int numberOfStreams)
-{
- CriticalSectionScoped cs(*_critSect);
-
- if (_directDraw == NULL)
- {
- return -1; // signal that we are not ready for the change
- }
- if (_width == width && _height == height && _offScreenSurface
- && _offScreenSurfaceNext)
- {
- _numberOfStreams = numberOfStreams;
- return 0;
- }
- if (_offScreenSurface)
- {
- _offScreenSurface->Release();
- _offScreenSurface = NULL;
- }
- if (_offScreenSurfaceNext)
- {
- _offScreenSurfaceNext->Release();
- _offScreenSurfaceNext = NULL;
- }
- if (width && height)
- {
- _width = width;
- _height = height;
- _numberOfStreams = numberOfStreams;
- }
-
- // create this channels offscreen buffer
- DirectDrawSurfaceDesc ddsd;
- HRESULT ddrval = DD_OK;
- memset(&ddsd, 0, sizeof(ddsd));
- ddsd.dwSize = sizeof(ddsd);
- ddsd.dwFlags = DDSD_CAPS | DDSD_HEIGHT | DDSD_WIDTH | DDSD_PIXELFORMAT;
- ddsd.ddsCaps.dwCaps = DDSCAPS_VIDEOMEMORY;
- ddsd.dwHeight = _height;
- ddsd.dwWidth = _width;
- /*
- char logStr[256];
- _snprintf(logStr,256, "offscreen H:%d W:%d \n",_height, _width);
- OutputDebugString(logStr);
- */
- //Fix for bad video driver on HP Mini. If it takes to long time to deliver a frame - try to blit using the same pixel format as used by the screen.
- if (_deliverInScreenType && _screenVideoType != kUnknown)
- {
- //The HP mini netbook, which this fix for, uses the VIA processor.
- //The measuring shows that this fix will impact systems with Intel processor, including Atom.
- //So let's disable it here. If we really need this for VIA processor, we should have additional logic to detect
- //the processor model.
- //WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1, "DirectDrawChannel changing to screen video type");
- //_blitVideoType=_screenVideoType;
- }
- else
- {
- WEBRTC_TRACE(
- kTraceInfo,
- kTraceVideo,
- -1,
- "DirectDrawChannel changing to originial blit video type %d",
- _originalBlitVideoType);
- _blitVideoType = _originalBlitVideoType;
- }
-
- WEBRTC_TRACE(
- kTraceInfo,
- kTraceVideo,
- -1,
- "DirectDrawChannel::FrameSizeChange height %d, width %d, _blitVideoType %d",
- ddsd.dwHeight, ddsd.dwWidth, _blitVideoType);
- switch (_blitVideoType)
- {
- case kYV12:
- {
- ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
- ddsd.ddpfPixelFormat.dwFlags = DDPF_FOURCC;
- ddsd.ddpfPixelFormat.dwFourCC = MAKEFOURCC('Y', 'V', '1', '2');
- }
- break;
- case kYUY2:
- {
- ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
- ddsd.ddpfPixelFormat.dwFlags = DDPF_FOURCC;
- ddsd.ddpfPixelFormat.dwFourCC = MAKEFOURCC('Y', 'U', 'Y', '2');
- }
- break;
- case kUYVY:
- {
- ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
- ddsd.ddpfPixelFormat.dwFlags = DDPF_FOURCC;
- ddsd.ddpfPixelFormat.dwFourCC = MAKEFOURCC('U', 'Y', 'V', 'Y');
- }
- break;
- case kIYUV:
- {
- ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
- ddsd.ddpfPixelFormat.dwFlags = DDPF_FOURCC;
- ddsd.ddpfPixelFormat.dwFourCC = MAKEFOURCC('I', 'Y', 'U', 'V');
- }
- break;
- case kARGB:
- {
- ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
- ddsd.ddpfPixelFormat.dwFlags = DDPF_RGB;
- ddsd.ddpfPixelFormat.dwRGBBitCount = 32;
- ddsd.ddpfPixelFormat.dwRBitMask = 0xff0000;
- ddsd.ddpfPixelFormat.dwGBitMask = 0xff00;
- ddsd.ddpfPixelFormat.dwBBitMask = 0xff;
- ddsd.ddpfPixelFormat.dwRGBAlphaBitMask = 0;
- }
- break;
- case kRGB24:
- {
- ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
- ddsd.ddpfPixelFormat.dwFlags = DDPF_RGB;
- ddsd.ddpfPixelFormat.dwRGBBitCount = 24;
- ddsd.ddpfPixelFormat.dwRBitMask = 0xff0000;
- ddsd.ddpfPixelFormat.dwGBitMask = 0xff00;
- ddsd.ddpfPixelFormat.dwBBitMask = 0xff;
- ddsd.ddpfPixelFormat.dwRGBAlphaBitMask = 0;
- }
- break;
- case kRGB565:
- {
- ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
- ddsd.ddpfPixelFormat.dwFlags = DDPF_RGB;
- ddsd.ddpfPixelFormat.dwRGBBitCount = 16;
- ddsd.ddpfPixelFormat.dwRBitMask = 0x0000F800;
- ddsd.ddpfPixelFormat.dwGBitMask = 0x000007e0;
- ddsd.ddpfPixelFormat.dwBBitMask = 0x0000001F;
- ddsd.ddpfPixelFormat.dwRGBAlphaBitMask = 0;
- }
- break;
- case kARGB4444:
- {
- ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
- ddsd.ddpfPixelFormat.dwFlags = DDPF_RGB;
- ddsd.ddpfPixelFormat.dwRGBBitCount = 16;
- ddsd.ddpfPixelFormat.dwRBitMask = 0x00000f00;
- ddsd.ddpfPixelFormat.dwGBitMask = 0x000000f0;
- ddsd.ddpfPixelFormat.dwBBitMask = 0x0000000f;
- ddsd.ddpfPixelFormat.dwRGBAlphaBitMask = 0;
- break;
- }
- case kARGB1555:
- {
- ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
- ddsd.ddpfPixelFormat.dwFlags = DDPF_RGB;
- ddsd.ddpfPixelFormat.dwRGBBitCount = 16;
- ddsd.ddpfPixelFormat.dwRBitMask = 0x00007C00;
- ddsd.ddpfPixelFormat.dwGBitMask = 0x3E0;
- ddsd.ddpfPixelFormat.dwBBitMask = 0x1F;
- ddsd.ddpfPixelFormat.dwRGBAlphaBitMask = 0;
- break;
- }
- case kI420:
- {
- ddsd.ddpfPixelFormat.dwSize = sizeof(DDPIXELFORMAT);
- ddsd.ddpfPixelFormat.dwFlags = DDPF_FOURCC;
- ddsd.ddpfPixelFormat.dwFourCC = MAKEFOURCC('I', '4', '2', '0');
- }
- break;
- default:
- ddrval = S_FALSE;
- }
-
- if (ddrval == DD_OK)
- {
- if (!_owner->IsPrimaryOrMixingSurfaceOnSystem())
- {
- ddrval
- = _directDraw->CreateSurface(&ddsd, &_offScreenSurface,
- NULL);
- if (FAILED(ddrval))
- {
- WEBRTC_TRACE(
- kTraceInfo,
- kTraceVideo,
- -1,
- "CreateSurface failed for _offScreenSurface on VideoMemory, trying on System Memory");
-
- memset(&ddsd, 0, sizeof(ddsd));
- ddsd.dwSize = sizeof(ddsd);
- ddsd.dwFlags = DDSD_CAPS | DDSD_HEIGHT | DDSD_WIDTH;
-
- ddsd.dwHeight = _height;
- ddsd.dwWidth = _width;
-
- ddsd.ddsCaps.dwCaps = DDSCAPS_SYSTEMMEMORY;
- _blitVideoType = kARGB;
-
- ddrval = _directDraw->CreateSurface(&ddsd, &_offScreenSurface,
- NULL);
- if (FAILED(ddrval))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "DirectDraw failed to CreateSurface _offScreenSurface using SystemMemory: 0x%x",
- ddrval);
- }
- ddrval = _directDraw->CreateSurface(&ddsd,
- &_offScreenSurfaceNext,
- NULL);
- if (FAILED(ddrval))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "DirectDraw failed to CreateSurface _offScreenSurfaceNext using SystemMemory: 0x%x",
- ddrval);
- }
- }
- else
- {
- ddrval = _directDraw->CreateSurface(&ddsd,
- &_offScreenSurfaceNext,
- NULL);
- if (ddrval == DDERR_OUTOFVIDEOMEMORY)
- {
- WEBRTC_TRACE(
- kTraceInfo,
- kTraceVideo,
- -1,
- "CreateSurface failed for _offScreenSurfaceNext on VideoMemory, trying on System Memory");
-
- memset(&ddsd, 0, sizeof(ddsd));
- ddsd.dwSize = sizeof(ddsd);
- ddsd.dwFlags = DDSD_CAPS | DDSD_HEIGHT | DDSD_WIDTH;
-
- ddsd.dwHeight = _height;
- ddsd.dwWidth = _width;
-
- ddsd.ddsCaps.dwCaps = DDSCAPS_SYSTEMMEMORY;
- _blitVideoType = kARGB;
-
- ddrval = _directDraw->CreateSurface(&ddsd,
- &_offScreenSurfaceNext,
- NULL);
- if (FAILED(ddrval))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "DirectDraw failed to CreateSurface _offScreenSurfaceNext using SystemMemory: 0x%x",
- ddrval);
- }
- }
- }
- }
- else
- {
- memset(&ddsd, 0, sizeof(ddsd));
- ddsd.dwSize = sizeof(ddsd);
- ddsd.dwFlags = DDSD_CAPS | DDSD_HEIGHT | DDSD_WIDTH;
-
- ddsd.dwHeight = _height;
- ddsd.dwWidth = _width;
-
- ddsd.ddsCaps.dwCaps = DDSCAPS_SYSTEMMEMORY;
- if (_owner->CanBltFourCC())
- {
- _blitVideoType = kARGB;
- }
- else
- {
- _blitVideoType = _originalBlitVideoType;
- }
-
- ddrval
- = _directDraw->CreateSurface(&ddsd, &_offScreenSurface,
- NULL);
- if (FAILED(ddrval))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "DirectDraw failed to CreateSurface _offScreenSurface using SystemMemory: 0x%x",
- ddrval);
- }
-
- ddrval = _directDraw->CreateSurface(&ddsd, &_offScreenSurfaceNext,
- NULL);
- if (FAILED(ddrval))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "DirectDraw failed to CreateSurface _offScreenSurfaceNext using SystemMemory: 0x%x",
- ddrval);
- }
- }
- }
-
- if (FAILED(ddrval))
- {
- // failed to change size
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw failed to CreateSurface : 0x%x", ddrval);
- return -1;
- }
-
- return 0;
-}
-
-int DirectDrawChannel::DeliverFrame(unsigned char* buffer, int bufferSize,
- unsigned int /*timeStamp90KHz*/)
-{
- CriticalSectionScoped cs(*_critSect);
-
- if (CalcBufferSize(_incomingVideoType, _width, _height)
- != bufferSize)
- {
- // sanity
- return -1;
- }
- if (!_offScreenSurface || !_offScreenSurfaceNext)
- {
- if (_width && _height && _numberOfStreams)
- {
- // our surface was lost recreate it
- FrameSizeChange(_width, _height, _numberOfStreams);
- }
- return -1;
- }
- if (_offScreenSurface->IsLost() == DDERR_SURFACELOST)
- {
- HRESULT ddrval = _offScreenSurface->Restore();
- if (ddrval != DD_OK)
- {
- // failed to restore our surface remove it and it will be re-created in next frame
- _offScreenSurface->Release();
- _offScreenSurface = NULL;
- _offScreenSurfaceNext->Release();
- _offScreenSurfaceNext = NULL;
- return -1;
- }
- ddrval = _offScreenSurfaceNext->Restore();
- if (ddrval != DD_OK)
- {
- // failed to restore our surface remove it and it will be re-created in next frame
- _offScreenSurface->Release();
- _offScreenSurface = NULL;
- _offScreenSurfaceNext->Release();
- _offScreenSurfaceNext = NULL;
- return -1;
- }
- }
- _doubleBuffer = false;
-
- // check if _offScreenSurfaceUpdated is true
- DirectDrawSurface* offScreenSurface = _offScreenSurface;
- {
-
- if (_offScreenSurfaceUpdated)
- {
- // this frame is not yet rendered
- offScreenSurface = _offScreenSurfaceNext;
- _doubleBuffer = true;
- }
- }
-
- DirectDrawSurfaceDesc ddsd;
- memset(&ddsd, 0, sizeof(ddsd));
- ddsd.dwSize = sizeof(ddsd);
- HRESULT ddrval = offScreenSurface->Lock(NULL, &ddsd, DDLOCK_WAIT, NULL);
- if (ddrval == DDERR_SURFACELOST)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDrawChannel::DeliverFrame offScreenSurface lost");
- ddrval = offScreenSurface->Restore();
- if (ddrval != DD_OK)
- {
- // failed to restore our surface remove it and it will be re-created in next frame
- _offScreenSurface->Release();
- _offScreenSurface = NULL;
- _offScreenSurfaceNext->Release();
- _offScreenSurfaceNext = NULL;
- return -1;
- }
- return 0;
- }
- if (ddrval != DD_OK)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDrawChannel::DeliverFrame failed to lock");
- // failed to lock our surface remove it and it will be re-created in next frame
- _offScreenSurface->Release();
- _offScreenSurface = NULL;
- _offScreenSurfaceNext->Release();
- _offScreenSurfaceNext = NULL;
- return -1;
- }
-
- unsigned char* ptr = (unsigned char*) ddsd.lpSurface;
- // ddsd.lPitch; distance in bytes
-
-
- switch (_incomingVideoType)
- {
- case kI420:
- {
- switch (_blitVideoType)
- {
- case kYUY2:
- case kUYVY:
- case kIYUV: // same as kYV12
- case kYV12:
- ConvertFromI420(buffer, _width,
- _blitVideoType, 0,
- _width, _height,
- ptr);
- break;
- case kRGB24:
- {
- _tempRenderBuffer.VerifyAndAllocate(_width * _height * 3);
- unsigned char *ptrTempBuffer = _tempRenderBuffer.Buffer();
- ConvertFromI420(buffer, _width, kRGB24, 0, _width, _height,
- ptrTempBuffer);
- for (int i = 0; i < _height; i++)
- {
- memcpy(ptr, ptrTempBuffer, _width * 3);
- ptrTempBuffer += _width * 3;
- ptr += ddsd.lPitch;
- }
- break;
- }
- case kARGB:
- ConvertFromI420(buffer, ddsd.lPitch, kARGB, 0,
- _width, _height, ptr);
- break;
- case kARGB4444:
- ConvertI420ToARGB4444(buffer, ptr, _width, _height,
- (ddsd.lPitch >> 1) - _width);
- break;
- case kARGB1555:
- ConvertI420ToARGB1555(buffer, ptr, _width, _height,
- (ddsd.lPitch >> 1) - _width);
- break;
- case kRGB565:
- {
- _tempRenderBuffer.VerifyAndAllocate(_width * _height * 2);
- unsigned char *ptrTempBuffer = _tempRenderBuffer.Buffer();
- ConvertI420ToRGB565(buffer, ptrTempBuffer, _width, _height);
- ptr += ddsd.lPitch * (_height - 1);
- for (int i = 0; i < _height; i++)
- {
- memcpy(ptr, ptrTempBuffer, _width * 2);
- ptrTempBuffer += _width * 2;
- ptr -= ddsd.lPitch;
- }
- break;
- }
- default:
- assert(!"DirectDrawChannel::DeliverFrame unknown blitVideoType");
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "DirectDrawChannel::DeliverFrame unknown blitVideoType %d",
- _blitVideoType);
- }
- break;
- }
- default:
- assert(!"DirectDrawChannel::DeliverFrame wrong incomming video type");
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "DirectDrawChannel::DeliverFrame wrong incomming %d",
- _incomingVideoType);
- }
-
- _offScreenSurfaceUpdated = true;
- offScreenSurface->Unlock(NULL);
- return 0;
-}
-
-int DirectDrawChannel::BlitFromOffscreenBufferToMixingBuffer(
- VideoRenderDirectDraw* DDobj,
- short streamID,
- DirectDrawSurface* mixingSurface,
- RECT &hwndRect,
- bool demuxing)
-{
- HRESULT ddrval;
- RECT srcRect;
- RECT dstRect;
- DirectDrawStreamSettings* streamSettings = NULL;
- unsigned long long lookupID = reinterpret_cast<unsigned long long> (DDobj);
- lookupID &= 0xffffffffffffffe0;
- lookupID <<= 11;
- lookupID += streamID;
-
- CriticalSectionScoped cs(*_critSect);
-
- if (_offScreenSurface == NULL)
- {
- // The offscreen surface has been deleted but not restored yet
- return 0;
- }
- if (mixingSurface == NULL)
- {
- // Not a valid input argument
- return 0;
- }
-
- std::map<unsigned long long, DirectDrawStreamSettings*>::iterator it =
- _streamIdToSettings.find(lookupID);
- if (it == _streamIdToSettings.end())
- {
- // ignore this stream id
- return 0;
- }
- streamSettings = it->second;
-
- int numberOfStreams = _numberOfStreams;
- if (!demuxing)
- {
- numberOfStreams = 1; // treat as one stream if we only have one config
- }
-
- switch (numberOfStreams)
- {
- case 0:
- return 0;
- case 1:
- {
- // no demux
- if (streamID > 0)
- return 0;
-
- ::SetRect(&srcRect, int(_width * streamSettings->_cropStartWidth),
- int(_height * streamSettings->_cropStartHeight),
- int(_width * streamSettings->_cropStopWidth), int(_height
- * streamSettings->_cropStopHeight));
-
- ::SetRect(&dstRect, int(hwndRect.right
- * streamSettings->_startWidth), int(hwndRect.bottom
- * streamSettings->_startHeight), int(hwndRect.right
- * streamSettings->_stopWidth), int(hwndRect.bottom
- * streamSettings->_stopHeight));
- }
- break;
- case 2:
- case 3:
- case 4:
- // classic quadrant demux
- {
- int width = _width >> 1;
- int height = _height >> 1;
- ::SetRect(&srcRect, int(width * streamSettings->_cropStartWidth),
- int(height * streamSettings->_cropStartHeight), int(width
- * streamSettings->_cropStopWidth), int(height
- * streamSettings->_cropStopHeight));
-
- ::SetRect(&dstRect, int(hwndRect.right
- * streamSettings->_startWidth), int(hwndRect.bottom
- * streamSettings->_startHeight), int(hwndRect.right
- * streamSettings->_stopWidth), int(hwndRect.bottom
- * streamSettings->_stopHeight));
-
- // stream id to select quadrant
- if (streamID == 1)
- {
- ::OffsetRect(&srcRect, width, 0);
- }
- if (streamID == 2)
- {
- ::OffsetRect(&srcRect, 0, height);
- }
- if (streamID == 3)
- {
- ::OffsetRect(&srcRect, width, height);
- }
- }
- break;
- case 5:
- case 6:
- {
- const int width = (_width / (3 * 16)) * 16;
- const int widthMidCol = width + ((_width % (16 * 3)) / 16) * 16;
- const int height = _height / (2 * 16) * 16;
- if (streamID == 1 || streamID == 4)
- {
- ::SetRect(&srcRect, int(widthMidCol
- * streamSettings->_cropStartWidth), int(height
- * streamSettings->_cropStartHeight), int(widthMidCol
- * streamSettings->_cropStopWidth), int(height
- * streamSettings->_cropStopHeight));
- }
- else
- {
- ::SetRect(&srcRect,
- int(width * streamSettings->_cropStartWidth),
- int(height * streamSettings->_cropStartHeight),
- int(width * streamSettings->_cropStopWidth),
- int(height * streamSettings->_cropStopHeight));
- }
- ::SetRect(&dstRect, int(hwndRect.right
- * streamSettings->_startWidth), int(hwndRect.bottom
- * streamSettings->_startHeight), int(hwndRect.right
- * streamSettings->_stopWidth), int(hwndRect.bottom
- * streamSettings->_stopHeight));
-
- // stream id to select quadrant
- switch (streamID)
- {
- case 1:
- ::OffsetRect(&srcRect, width, 0);
- break;
- case 2:
- ::OffsetRect(&srcRect, width + widthMidCol, 0);
- break;
- case 3:
- ::OffsetRect(&srcRect, 0, height);
- break;
- case 4:
- ::OffsetRect(&srcRect, width, height);
- break;
- case 5:
- ::OffsetRect(&srcRect, width + widthMidCol, height);
- break;
- }
- }
- break;
- case 7:
- case 8:
- case 9:
-
- {
- const int width = (_width / (3 * 16)) * 16;
- const int widthMidCol = width + ((_width % (16 * 3)) / 16) * 16;
- const int height = _height / (3 * 16) * 16;
- const int heightMidRow = height + ((_height % (16 * 3)) / 16) * 16;
-
- ::SetRect(&dstRect, int(hwndRect.right
- * streamSettings->_startWidth), int(hwndRect.bottom
- * streamSettings->_startHeight), int(hwndRect.right
- * streamSettings->_stopWidth), int(hwndRect.bottom
- * streamSettings->_stopHeight));
-
- switch (streamID)
- {
- case 0:
- //Size
- ::SetRect(&srcRect, int(width
- * streamSettings->_cropStartWidth), int(height
- * streamSettings->_cropStartHeight), int(width
- * streamSettings->_cropStopWidth), int(height
- * streamSettings->_cropStopHeight));
- //Position
- ::OffsetRect(&srcRect, 0, 0);
- break;
- case 1:
- ::SetRect(
- &srcRect,
- int(widthMidCol * streamSettings->_cropStartWidth),
- int(height * streamSettings->_cropStartHeight),
- int(widthMidCol * streamSettings->_cropStopWidth),
- int(height * streamSettings->_cropStopHeight));
- ::OffsetRect(&srcRect, width, 0);
- break;
- case 2:
- ::SetRect(&srcRect, int(width
- * streamSettings->_cropStartWidth), int(height
- * streamSettings->_cropStartHeight), int(width
- * streamSettings->_cropStopWidth), int(height
- * streamSettings->_cropStopHeight));
- ::OffsetRect(&srcRect, width + widthMidCol, 0);
- break;
- case 3:
- ::SetRect(&srcRect, int(width
- * streamSettings->_cropStartWidth),
- int(heightMidRow
- * streamSettings->_cropStartHeight),
- int(width * streamSettings->_cropStopWidth),
- int(heightMidRow
- * streamSettings->_cropStopHeight));
- ::OffsetRect(&srcRect, 0, height);
- break;
- case 4:
- ::SetRect(
- &srcRect,
- int(widthMidCol * streamSettings->_cropStartWidth),
- int(heightMidRow
- * streamSettings->_cropStartHeight),
- int(widthMidCol * streamSettings->_cropStopWidth),
- int(heightMidRow
- * streamSettings->_cropStopHeight));
- ::OffsetRect(&srcRect, width, height);
-
- break;
- case 5:
- ::SetRect(&srcRect, int(width
- * streamSettings->_cropStartWidth),
- int(heightMidRow
- * streamSettings->_cropStartHeight),
- int(width * streamSettings->_cropStopWidth),
- int(heightMidRow
- * streamSettings->_cropStopHeight));
- ::OffsetRect(&srcRect, width + widthMidCol, height);
- break;
- case 6:
- ::SetRect(&srcRect, int(width
- * streamSettings->_cropStartWidth), int(height
- * streamSettings->_cropStartHeight), int(width
- * streamSettings->_cropStopWidth), int(height
- * streamSettings->_cropStopHeight));
- ::OffsetRect(&srcRect, 0, height + heightMidRow);
- break;
- case 7:
- ::SetRect(
- &srcRect,
- int(widthMidCol * streamSettings->_cropStartWidth),
- int(height * streamSettings->_cropStartHeight),
- int(widthMidCol * streamSettings->_cropStopWidth),
- int(height * streamSettings->_cropStopHeight));
- ::OffsetRect(&srcRect, width, height + heightMidRow);
- break;
- case 8:
- ::SetRect(&srcRect, int(width
- * streamSettings->_cropStartWidth), int(height
- * streamSettings->_cropStartHeight), int(width
- * streamSettings->_cropStopWidth), int(height
- * streamSettings->_cropStopHeight));
- ::OffsetRect(&srcRect, width + widthMidCol, height
- + heightMidRow);
- break;
- }
- }
- break;
- case 10:
- case 11:
- case 12:
- case 13:
- case 14:
- case 15:
- case 16:
- default:
- {
- ::SetRect(&srcRect, int(_width * streamSettings->_cropStartWidth),
- int(_height * streamSettings->_cropStartHeight),
- int(_width * streamSettings->_cropStopWidth), int(_height
- * streamSettings->_cropStopHeight));
-
- ::SetRect(&dstRect, int(hwndRect.right
- * streamSettings->_startWidth), int(hwndRect.bottom
- * streamSettings->_startHeight), int(hwndRect.right
- * streamSettings->_stopWidth), int(hwndRect.bottom
- * streamSettings->_stopHeight));
- }
- }
-
- if (dstRect.right > hwndRect.right)
- {
- srcRect.right -= (int) ((float) (srcRect.right - srcRect.left)
- * ((float) (dstRect.right - hwndRect.right)
- / (float) (dstRect.right - dstRect.left)));
- dstRect.right = hwndRect.right;
- }
- if (dstRect.left < hwndRect.left)
- {
- srcRect.left += (int) ((float) (srcRect.right - srcRect.left)
- * ((float) (hwndRect.left - dstRect.left)
- / (float) (dstRect.right - dstRect.left)));
- dstRect.left = hwndRect.left;
- }
- if (dstRect.bottom > hwndRect.bottom)
- {
- srcRect.bottom -= (int) ((float) (srcRect.bottom - srcRect.top)
- * ((float) (dstRect.bottom - hwndRect.bottom)
- / (float) (dstRect.bottom - dstRect.top)));
- dstRect.bottom = hwndRect.bottom;
- }
- if (dstRect.top < hwndRect.top)
- {
- srcRect.top += (int) ((float) (srcRect.bottom - srcRect.top)
- * ((float) (hwndRect.top - dstRect.top)
- / (float) (dstRect.bottom - dstRect.top)));
- dstRect.top = hwndRect.top;
- }
-
- DDBLTFX ddbltfx;
- ZeroMemory(&ddbltfx, sizeof(ddbltfx));
- ddbltfx.dwSize = sizeof(ddbltfx);
- ddbltfx.dwDDFX = DDBLTFX_NOTEARING;
-
- // wait for the _mixingSurface to be available
- ddrval = mixingSurface->Blt(&dstRect, _offScreenSurface, &srcRect,
- DDBLT_WAIT | DDBLT_DDFX, &ddbltfx);
- if (ddrval == DDERR_SURFACELOST)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "mixingSurface->Blt surface lost");
- ddrval = mixingSurface->Restore();
- if (ddrval != DD_OK)
- {
- // we dont own the surface just report the error
- return -1;
- }
- }
- else if (ddrval == DDERR_INVALIDRECT)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "mixingSurface->Blt DDERR_INVALIDRECT");
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "dstRect co-ordinates - top: %d left: %d bottom: %d right: %d",
- dstRect.top, dstRect.left, dstRect.bottom, dstRect.right);
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "srcRect co-ordinates - top: %d left: %d bottom: %d right: %d",
- srcRect.top, srcRect.left, srcRect.bottom, srcRect.right);
-
- // ignore
- }
- else if (ddrval != DD_OK)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "mixingSurface->Blt !DD_OK");
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "DirectDraw blt mixingSurface BlitFromOffscreenBufferToMixingBuffer error 0x%x ",
- ddrval);
-
- //logging the co-ordinates and hwnd
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "dstRect co-ordinates - top: %d left: %d bottom: %d right: %d",
- dstRect.top, dstRect.left, dstRect.bottom, dstRect.right);
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "srcRect co-ordinates - top: %d left: %d bottom: %d right: %d",
- srcRect.top, srcRect.left, srcRect.bottom, srcRect.right);
-
- /* char logStr[256];
- _snprintf(logStr,256, "srcRect T:%d L:%d B:%d R:%d\n",srcRect.top, srcRect.left, srcRect.bottom, srcRect.right);
- OutputDebugString(logStr);
- char logStr1[256];
- _snprintf(logStr1,256, "dstRect T:%d L:%d B:%d R:%d\n",dstRect.top, dstRect.left, dstRect.bottom, dstRect.right);
- OutputDebugString(logStr1);
- char logStr2[256];
- _snprintf(logStr2,256, "error 0x:%x \n",ddrval);
- OutputDebugString(logStr2);
- */
- // we dont own the surface just report the error
- return -1;
- }
- if (_doubleBuffer)
- {
- DirectDrawSurface* oldOffScreenSurface = _offScreenSurface;
- _offScreenSurface = _offScreenSurfaceNext;
- _offScreenSurfaceNext = oldOffScreenSurface;
- _doubleBuffer = false;
- }
- else
- {
- _offScreenSurfaceUpdated = false;
- }
- return 0;
-}
-
-/**
- *
- * VideoRenderDirectDraw
- *
- *
- */
-
-VideoRenderDirectDraw::VideoRenderDirectDraw(Trace* trace,
- HWND hWnd, bool fullscreen) :
- _trace(trace),
- _confCritSect(CriticalSectionWrapper::CreateCriticalSection()),
- _fullscreen(fullscreen),
- _demuxing(false),
- _transparentBackground(false),
- _supportTransparency(false),
- _canStretch(false),
- _canMirrorLeftRight(false),
- _clearMixingSurface(false),
- _deliverInScreenType(false),
- _renderModeWaitForCorrectScanLine(false),
- _deliverInHalfFrameRate(false),
- _deliverInQuarterFrameRate(false),
- _bCanBltFourcc(true),
- _frameChanged(false),
- _processCount(0),
- _hWnd(hWnd),
- _screenRect(),
- _mixingRect(),
-
- _incomingVideoType(kUnknown),
- _blitVideoType(kUnknown),
- _rgbVideoType(kUnknown),
-
- _directDraw(NULL),
- _primarySurface(NULL),
- _backSurface(NULL),
- _mixingSurface(NULL),
- _bitmapSettings(),
- _textSettings(),
- _directDrawChannels(),
- _directDrawZorder(),
-
- _fullScreenWaitEvent(EventWrapper::Create()),
- _screenEvent(EventWrapper::Create()),
- _screenRenderThread(
- ThreadWrapper::CreateThread(
- RemoteRenderingThreadProc,
- this,
- kRealtimePriority,
- "Video_directdraw_thread")),
- _blit(true), _lastRenderModeCpuUsage(-1), _totalMemory(-1),
- _availableMemory(-1), _systemCPUUsage(0), _maxAllowedRenderTime(0),
- _nrOfTooLongRenderTimes(0),
- _isPrimaryOrMixingSurfaceOnSystem(false)
-{
- SetRect(&_screenRect, 0, 0, 0, 0);
- SetRect(&_mixingRect, 0, 0, 0, 0);
- SetRect(&_originalHwndRect, 0, 0, 0, 0);
- ::GetClientRect(_hWnd, &_hwndRect);
-}
-
-VideoRenderDirectDraw::~VideoRenderDirectDraw()
-{
- ThreadWrapper* temp = _screenRenderThread;
- _screenRenderThread = NULL;
- if (temp)
- {
- temp->SetNotAlive();
- _screenEvent->Set();
- _screenEvent->StopTimer();
- _fullScreenWaitEvent->StopTimer();
-
- if (temp->Stop())
- {
- delete temp;
- }
- }
- delete _screenEvent;
- delete _fullScreenWaitEvent;
-
- std::map<int, DirectDrawChannel*>::iterator it;
- it = _directDrawChannels.begin();
- while (it != _directDrawChannels.end())
- {
- it->second->Release();
- it = _directDrawChannels.erase(it);
- }
- if (_primarySurface)
- {
- _primarySurface->Release();
- }
- if (_mixingSurface)
- {
- _mixingSurface->Release();
- }
-
- std::map<unsigned char, DirectDrawBitmapSettings*>::iterator bitIt;
-
- bitIt = _bitmapSettings.begin();
- while (_bitmapSettings.end() != bitIt)
- {
- delete bitIt->second;
- bitIt = _bitmapSettings.erase(bitIt);
- }
-
- std::map<unsigned char, DirectDrawTextSettings*>::iterator textIt;
- textIt = _textSettings.begin();
- while (_textSettings.end() != textIt)
- {
- delete textIt->second;
- textIt = _textSettings.erase(textIt);
- }
- if (_directDraw)
- {
- _directDraw->Release();
- if (_fullscreen)
- {
- // restore hwnd to original size and position
- ::SetWindowPos(_hWnd, HWND_NOTOPMOST, _originalHwndRect.left,
- _originalHwndRect.top, _originalHwndRect.right
- - _originalHwndRect.left,
- _originalHwndRect.bottom - _originalHwndRect.top,
- SWP_FRAMECHANGED);
- ::RedrawWindow(_hWnd, NULL, NULL, RDW_INVALIDATE | RDW_UPDATENOW
- | RDW_ERASE);
- ::RedrawWindow(NULL, NULL, NULL, RDW_INVALIDATE | RDW_UPDATENOW
- | RDW_ERASE);
- }
- }
- delete _confCritSect;
-}
-
-WebRtc_Word32 VideoRenderDirectDraw::Init()
-{
- int retVal = 0;
- HRESULT ddrval = DirectDrawCreateEx(NULL, (void**) &_directDraw,
- IID_IDirectDraw7, NULL);
- if (FAILED(ddrval) || NULL == _directDraw)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Failed to created DirectDraw7 object");
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- retVal = CheckCapabilities();
- if (retVal != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw CheckCapabilities failed");
- return retVal;
- }
- if (_hWnd)
- {
- retVal = CreatePrimarySurface();
- if (retVal != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw failed to CreatePrimarySurface");
- return retVal;
- }
- retVal = CreateMixingSurface();
- if (retVal != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw failed to CreateMixingSurface");
- return retVal;
- }
- if (_screenRenderThread)
- {
- unsigned int tid;
- _screenRenderThread->Start(tid);
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "Screen Render thread started, thread id: %d", tid);
- }
- DWORD freq = 0;
- _directDraw->GetMonitorFrequency(&freq);
- if (freq == 0)
- {
- freq = 60;
- }
- // Do this now to not do it in each render process loop
- _maxAllowedRenderTime = (int) (1000 / freq * 0.8F);
- _nrOfTooLongRenderTimes = 0;
-
- _screenEvent->StartTimer(true, 1000 / freq);
-
- _deliverInScreenType = false;
- _renderModeWaitForCorrectScanLine = false;
- _deliverInHalfFrameRate = false;
- _deliverInQuarterFrameRate = false;
-
- _lastRenderModeCpuUsage = -1;
- if (_fullscreen)
- {
- _fullScreenWaitEvent->StartTimer(true, 1);
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "Screen freq %d", freq);
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "Created DirectDraw object");
- return 0;
-}
-
-WebRtc_Word32 VideoRenderDirectDraw::GetGraphicsMemory(
- WebRtc_UWord64& totalMemory,
- WebRtc_UWord64& availableMemory)
-{
- CriticalSectionScoped cs(*_confCritSect);
-
- if (_totalMemory == -1 || _availableMemory == -1)
- {
- totalMemory = 0;
- availableMemory = 0;
- return -1;
- }
- totalMemory = _totalMemory;
- availableMemory = _availableMemory;
- return 0;
-}
-
-int VideoRenderDirectDraw::GetScreenResolution(int& screenWidth,
- int& screenHeight)
-{
- CriticalSectionScoped cs(*_confCritSect);
-
- screenWidth = _screenRect.right - _screenRect.left;
- screenHeight = _screenRect.bottom - _screenRect.top;
- return 0;
-}
-
-int VideoRenderDirectDraw::UpdateSystemCPUUsage(int systemCPU)
-{
- CriticalSectionScoped cs(*_confCritSect);
- if (systemCPU <= 100 && systemCPU >= 0)
- {
- _systemCPUUsage = systemCPU;
- }
- return 0;
-}
-
-int VideoRenderDirectDraw::CheckCapabilities()
-{
- HRESULT ddrval = DD_OK;
- DDCAPS ddcaps;
- DDCAPS ddcapsEmul;
- memset(&ddcaps, 0, sizeof(ddcaps));
- memset(&ddcapsEmul, 0, sizeof(ddcapsEmul));
- ddcaps.dwSize = sizeof(ddcaps);
- ddcapsEmul.dwSize = sizeof(ddcapsEmul);
- if (_directDraw == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw object not created");
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- if (IsRectEmpty(&_screenRect))
- {
- ::GetWindowRect(GetDesktopWindow(), &_screenRect);
- }
- // Log Screen resolution
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "ScreenRect. Top: %d, left: %d, bottom: %d, right: %d",
- _screenRect.top, _screenRect.left, _screenRect.bottom,
- _screenRect.right);
-
- bool fullAccelerationEnabled = false;
- bool badDriver = false;
- VideoRenderWindowsImpl::CheckHWDriver(badDriver, fullAccelerationEnabled);
- if (!fullAccelerationEnabled)
- {
-
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Direct draw Hardware acceleration is not enabled.");
- return -1;
- //return VIDEO_DIRECT_DRAW_HWACC_NOT_ENABLED;
-
- }
-
- // ddcaps supported by the HW
- // ddcapsEmul supported by the OS emulating the HW
- ddrval = _directDraw->GetCaps(&ddcaps, &ddcapsEmul);
- if (ddrval != DD_OK)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw HW: could not get capabilities: %x", ddrval);
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
-
- unsigned int minVideoMemory = 3 * 4 * (_screenRect.right
- * _screenRect.bottom); // assuming ARGB size (4 bytes)
-
- // Store the memory for possible calls to GetMemory()
- _totalMemory = ddcaps.dwVidMemTotal;
- _availableMemory = ddcaps.dwVidMemFree;
-
- if (ddcaps.dwVidMemFree < minVideoMemory)
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "DirectDraw HW does not have enough memory, freeMem:%d, requiredMem:%d",
- ddcaps.dwVidMemFree, minVideoMemory);
- // If memory is not available on the Video Card...allocate it on RAM
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw video memory, freeMem:%d, totalMem:%d",
- ddcaps.dwVidMemFree, ddcaps.dwVidMemTotal);
- }
-
- /*
- DirectDrawCaps ddsCaps ;
- ZeroMemory(&ddsCaps, sizeof(ddsCaps)) ;
- ddsCaps.dwCaps = DDSCAPS_VIDEOMEMORY;
- DWORD memTotal=0;
- DWORD memFree=0;
- ddrval = _directDraw->GetAvailableVidMem(&ddsCaps, &memTotal, &memFree);
- if(ddrval == DD_OK)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1, "DirectDraw video memory, freeMem:%d, totalMem:%d", memFree, memTotal);
- }
- */
- // Determine if the hardware supports overlay deinterlacing
- // bCanDeinterlace = (ddcaps.dwCaps2 & DDCAPS2_CANFLIPODDEVEN) ? 1 : 0;
-
- // this fail since we check before we set the mode
- // bool bCanFlip =(ddcaps.dwCaps & DDSCAPS_FLIP) ? 1 : 0;
-
- // Determine if the hardware supports colorkeying
- _supportTransparency = (ddcaps.dwCaps & DDCAPS_COLORKEY) ? 1 : 0;
- if (_supportTransparency)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw support colorkey");
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
- "DirectDraw don't support colorkey");
- }
-
- if (ddcaps.dwCaps2 & DDCAPS2_CANRENDERWINDOWED)
- {
- // required for _directDraw->FlipToGDISurface();
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw support CANRENDERWINDOWED");
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw don't support CANRENDERWINDOWED");
- }
-
- // Determine if the hardware supports scaling during a blit
- _canStretch = (ddcaps.dwCaps & DDCAPS_BLTSTRETCH) ? 1 : 0;
- if (_canStretch)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw blit can stretch");
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
- "DirectDraw blit can't stretch");
- }
-
- _canMirrorLeftRight = (ddcaps.dwFXAlphaCaps & DDBLTFX_MIRRORLEFTRIGHT) ? 1
- : 0;
- if (_canMirrorLeftRight)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw mirroring is supported");
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw mirroring is not supported");
- }
-
- // Determine if the hardware supports color conversion during a blit
- _bCanBltFourcc = (ddcaps.dwCaps & DDCAPS_BLTFOURCC) ? 1 : 0;
- if (_bCanBltFourcc)
- _bCanBltFourcc = (ddcaps.dwCKeyCaps & DDCKEYCAPS_DESTBLT) ? 1 : 0;
-
- if (_bCanBltFourcc)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw can blit Fourcc");
- DWORD i_codes;
- ddrval = _directDraw->GetFourCCCodes(&i_codes, NULL);
-
- if (i_codes > 0)
- {
- DWORD* pi_codes = new DWORD[i_codes];
-
- ddrval = _directDraw->GetFourCCCodes(&i_codes, pi_codes);
- for (unsigned int i = 0; i < i_codes && _blitVideoType
- != kI420; i++)
- {
- DWORD w = pi_codes[i];
- switch (w)
- {
- case MAKEFOURCC('I', '4', '2', '0'):
- // _blitVideoType = kI420;
- // not enabled since its not tested
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- -1, "DirectDraw support I420");
- break;
- case MAKEFOURCC('I', 'Y', 'U', 'V'): // same as YV12
- // _blitVideoType = kIYUV;
- // not enabled since its not tested
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- -1, "DirectDraw support IYUV");
- break;
- case MAKEFOURCC('U', 'Y', 'N', 'V'): // same shit different name
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- -1, "DirectDraw support UYNV");
- // not enabled since its not tested
- break;
- case MAKEFOURCC('Y', '4', '2', '2'): // same shit different name
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- -1, "DirectDraw support Y422");
- // not enabled since its not tested
- break;
- case MAKEFOURCC('Y', 'U', 'N', 'V'): // same shit different name
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- -1, "DirectDraw support YUNV");
- // not enabled since its not tested
- break;
- case MAKEFOURCC('Y', 'V', '1', '2'):
- _blitVideoType = kYV12;
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- -1, "DirectDraw support YV12");
- break;
- case MAKEFOURCC('Y', 'U', 'Y', '2'):
- if (_blitVideoType != kYV12)
- {
- _blitVideoType = kYUY2;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- -1, "DirectDraw support YUY2");
- break;
- case MAKEFOURCC('U', 'Y', 'V', 'Y'):
- if (_blitVideoType != kYV12)
- {
- _blitVideoType = kUYVY;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- -1, "DirectDraw support UYVY");
- break;
- default:
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- -1, "DirectDraw unknown blit type %x", w);
- break;
- }
- }
- delete[] pi_codes;
- }
- }
- return 0;
-}
-
-int VideoRenderDirectDraw::Stop()
-{
- _confCritSect->Enter();
-
- _blit = false;
-
- _confCritSect->Leave();
- return 0;
-}
-
-bool VideoRenderDirectDraw::IsPrimaryOrMixingSurfaceOnSystem()
-{
- return _isPrimaryOrMixingSurfaceOnSystem;
-}
-
-int VideoRenderDirectDraw::CreatePrimarySurface()
-{
- // Create the primary surface
- DirectDrawSurfaceDesc ddsd;
- ZeroMemory(&ddsd, sizeof(ddsd));
- ddsd.dwSize = sizeof(ddsd);
- HRESULT ddrval = DD_OK;
-
- if (_directDraw == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw object not created");
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- if (_primarySurface)
- {
- _primarySurface->Release();
- _primarySurface = NULL;
- }
-
- if (!_fullscreen)
- {
- // create a normal window
- ddrval = _directDraw->SetCooperativeLevel(_hWnd, DDSCL_NORMAL);
- if (FAILED(ddrval))
- {
- //******** Potential workaround for D#4608 *************** Ignore error.
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
- "DirectDraw failed to set SetCooperativeLevel %x, ddrval");
- }
- // we cant size the primary surface based on _hwndRect
- ddsd.dwFlags = DDSD_CAPS;
- ddsd.ddsCaps.dwCaps = DDSCAPS_PRIMARYSURFACE | DDSCAPS_VIDEOMEMORY;
-
-#ifndef NOGRAPHICSCARD_MEMORY
- ddrval = _directDraw->CreateSurface(&ddsd, &_primarySurface, NULL);
- if (FAILED(ddrval))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "DirectDraw failed to CreateSurface _primarySurface using VideoMemory: 0x%x",
- ddrval);
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "\t HWND: 0x%x, top: %d, left: %d, bottom: %d, right: %d, dwFlags: %d. Line : %d",
- _hWnd, _hwndRect.top, _hwndRect.left,
- _hwndRect.bottom, _hwndRect.right, ddsd.dwFlags,
- __LINE__);
-
-#endif
- //allocate using System memory
- ddsd.ddsCaps.dwCaps = DDSCAPS_PRIMARYSURFACE | DDSCAPS_SYSTEMMEMORY;
- ddrval = _directDraw->CreateSurface(&ddsd, &_primarySurface, NULL);
- if (FAILED(ddrval))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "DirectDraw failed to CreateSurface _primarySurface using SystemMemory: 0x%x",
- ddrval);
- if (ddrval != 0x887600E1)
- {
- _directDraw->Release();
- _directDraw = 0;
- }
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- _isPrimaryOrMixingSurfaceOnSystem = true;
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw _primarySurface on SystemMemory");
-
-#ifndef NOGRAPHICSCARD_MEMORY
- }
-#endif
-
- // Create a clipper to ensure that our drawing stays inside our window
- LPDIRECTDRAWCLIPPER directDrawClipper;
- ddrval = _directDraw->CreateClipper(0, &directDrawClipper, NULL );
- if (ddrval != DD_OK)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw failed to CreateClipper");
- _primarySurface->Release();
- _directDraw->Release();
- _primarySurface = 0;
- _directDraw = 0;
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- // setting it to our hwnd gives the clipper the coordinates from our window
- // when using cliplist we run into problem with transparent HWNDs (such as REX)
- ddrval = directDrawClipper->SetHWnd(0, _hWnd);
- if (ddrval != DD_OK)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw failed to SetHWnd");
- _primarySurface->Release();
- _directDraw->Release();
- _primarySurface = 0;
- _directDraw = 0;
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- // attach the clipper to the primary surface
- ddrval = _primarySurface->SetClipper(directDrawClipper);
- directDrawClipper->Release(); // no need to keep the clipper around
- if (ddrval != DD_OK)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw failed to SetClipper");
- _primarySurface->Release();
- _directDraw->Release();
- _primarySurface = 0;
- _directDraw = 0;
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- }
- else
- {
- /* The cooperative level determines how much control we have over the
- * screen. This must at least be either DDSCL_EXCLUSIVE or DDSCL_NORMAL
- *
- * DDSCL_EXCLUSIVE allows us to change video modes, and requires
- * the DDSCL_FULLSCREEN flag, which will cause the window to take over
- * the fullscreen. This is the preferred DirectDraw mode because it allows
- * us to have control of the whole screen without regard for GDI.
- *
- * DDSCL_NORMAL is used to allow the DirectDraw app to run windowed.
- */
-
- // Note: debuging in fullscreen mode does not work, thanks MS...
- ::GetWindowRect(_hWnd, &_originalHwndRect);
-
- // DDSCL_NOWINDOWCHANGES prevents DD to change the window but it give us trouble too, not using it
- ddrval = _directDraw->SetCooperativeLevel(_hWnd, DDSCL_EXCLUSIVE
- | DDSCL_FULLSCREEN | DDSCL_ALLOWREBOOT);
-
- if (FAILED(ddrval))
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw failed to SetCooperativeLevel DDSCL_EXCLUSIVE");
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "\t HWND: 0x%x, top: %d, left: %d, bottom: %d, right: %d, dwFlags: %d. Line : %d",
- _hWnd, _hwndRect.top, _hwndRect.left,
- _hwndRect.bottom, _hwndRect.right, ddsd.dwFlags,
- __LINE__);
-
- _directDraw->Release();
- _directDraw = 0;
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- ddsd.dwFlags = DDSD_CAPS | DDSD_BACKBUFFERCOUNT;
- ddsd.ddsCaps.dwCaps = DDSCAPS_PRIMARYSURFACE | DDSCAPS_FLIP
- | DDSCAPS_COMPLEX | DDSCAPS_VIDEOMEMORY;
- ddsd.dwBackBufferCount = 1;
-
- ddrval = _directDraw->CreateSurface(&ddsd, &_primarySurface, NULL);
- if (FAILED(ddrval))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "DirectDraw failed to CreateSurface _primarySurface, fullscreen mode: 0x%x",
- ddrval);
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "\t HWND: 0x%x, top: %d, left: %d, bottom: %d, right: %d, dwFlags: %d. Line : %d",
- _hWnd, _hwndRect.top, _hwndRect.left,
- _hwndRect.bottom, _hwndRect.right, ddsd.dwFlags,
- __LINE__);
-
- _directDraw->Release();
- _directDraw = 0;
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- // Get a pointer to the back buffer
- DirectDrawCaps ddsCaps;
- ZeroMemory(&ddsCaps, sizeof(ddsCaps));
- ddsCaps.dwCaps = DDSCAPS_BACKBUFFER | DDSCAPS_VIDEOMEMORY;
-
- ddrval = _primarySurface->GetAttachedSurface(&ddsCaps, &_backSurface);
- if (FAILED(ddrval))
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw failed to GetAttachedSurface, fullscreen mode ");
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "\t HWND: 0x%x, top: %d, left: %d, bottom: %d, right: %d, dwFlags: %d. Line : %d",
- _hWnd, _hwndRect.top, _hwndRect.left,
- _hwndRect.bottom, _hwndRect.right, ddsd.dwFlags,
- __LINE__);
-
- _primarySurface->Release();
- _directDraw->Release();
- _primarySurface = 0;
- _directDraw = 0;
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- // Get the screen size and save it as a rect
- ZeroMemory(&ddsd, sizeof(ddsd));
- ddsd.dwSize = sizeof(ddsd);
- }
-
- ZeroMemory(&ddsd, sizeof(ddsd));
- ddsd.dwSize = sizeof(ddsd);
-
- // get our prinmary surface description
- ddrval = _primarySurface->GetSurfaceDesc(&ddsd);
- if (!(SUCCEEDED(ddrval) && (ddsd.dwFlags & DDSD_WIDTH) && (ddsd.dwFlags
- & DDSD_HEIGHT)))
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw failed to GetSurfaceDesc _primarySurface");
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "\t HWND: 0x%x, top: %d, left: %d, bottom: %d, right: %d, dwFlags: %d. Line : %d",
- _hWnd, _hwndRect.top, _hwndRect.left, _hwndRect.bottom,
- _hwndRect.right, ddsd.dwFlags, __LINE__);
-
- _primarySurface->Release();
- _directDraw->Release();
- _primarySurface = 0;
- _directDraw = 0;
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- // first we need to figure out the size of the primary surface
-
- // store screen size
- ::SetRect(&_screenRect, 0, 0, ddsd.dwWidth, ddsd.dwHeight);
-
- // store RGB type
- if (ddsd.ddpfPixelFormat.dwFlags & DDPF_RGB)
- {
- // RGB surface
- switch (ddsd.ddpfPixelFormat.dwRGBBitCount)
- {
- case 16:
- switch (ddsd.ddpfPixelFormat.dwGBitMask)
- {
- case 0x00e0:
- _rgbVideoType = kARGB4444;
- break;
- case 0x03e0:
- _rgbVideoType = kARGB1555;
- break;
- case 0x07e0:
- _rgbVideoType = kRGB565;
- break;
- }
- break;
- case 24:
- _rgbVideoType = kRGB24;
- break;
- case 32:
- _rgbVideoType = kARGB;
- break;
- }
- }
- switch (_blitVideoType)
- {
- case kI420:
- case kIYUV:
- case kYUY2:
- case kYV12:
- case kUYVY:
- _incomingVideoType = kI420;
- break;
- case kUnknown:
- _blitVideoType = _rgbVideoType;
- _incomingVideoType = kI420;
- break;
- default:
- _blitVideoType = _rgbVideoType;
- _incomingVideoType = kI420;
- break;
- }
- WEBRTC_TRACE(
- kTraceInfo,
- kTraceVideo,
- -1,
- "DirectDraw created _primarySurface, _blitVideoType %d, _rgbvideoType %d",
- _blitVideoType, _rgbVideoType);
- return 0;
-}
-
-int VideoRenderDirectDraw::CreateMixingSurface()
-{
- if (_directDraw == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw object not created");
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
-
- if (_fullscreen)
- {
- ::CopyRect(&_hwndRect, &_screenRect);
- }
- else
- {
- // update our _hWnd size
- ::GetClientRect(_hWnd, &_hwndRect);
- }
-
- if (_mixingSurface)
- {
- _mixingSurface->Release();
- _mixingSurface = NULL;
- }
- // create mixing surface
- DirectDrawSurfaceDesc ddsd;
- memset(&ddsd, 0, sizeof(ddsd));
- ddsd.dwSize = sizeof(ddsd);
- ddsd.dwFlags = DDSD_CAPS | DDSD_HEIGHT | DDSD_WIDTH;
- ddsd.ddsCaps.dwCaps = DDSCAPS_VIDEOMEMORY;
- ddsd.dwHeight = _hwndRect.bottom;
- ddsd.dwWidth = _hwndRect.right;
-
- /* char logStr[256];
- _snprintf(logStr,256, "CreateMixingSurface H:%d W:%d \n",_hwndRect.bottom, _hwndRect.right);
- OutputDebugString(logStr);
- */
-
-#ifndef NOGRAPHICSCARD_MEMORY
- HRESULT ddrval = _directDraw->CreateSurface(&ddsd, &_mixingSurface, NULL);
- if (FAILED(ddrval))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "DirectDraw failed to CreateSurface _mixingSurface using VideoMemory: 0x%x",
- ddrval);
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "\t HWND: 0x%x, top: %d, left: %d, bottom: %d, right: %d, dwFlags: %d",
- _hWnd, _hwndRect.top, _hwndRect.left, _hwndRect.bottom,
- _hwndRect.right, ddsd.dwFlags);
-#endif
-
- ddsd.ddsCaps.dwCaps = DDSCAPS_SYSTEMMEMORY;
- HRESULT ddrval = _directDraw->CreateSurface(&ddsd, &_mixingSurface,
- NULL);
- if (FAILED(ddrval))
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "DirectDraw failed to CreateSurface _mixingSurface on System Memory: 0x%x",
- ddrval);
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- _isPrimaryOrMixingSurfaceOnSystem = true;
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw CreateSurface _mixingSurface on SystemMemory");
-
-#ifndef NOGRAPHICSCARD_MEMORY
- }
-#endif
-
- _clearMixingSurface = true;
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw _mixingSurface created");
- return 0;
-}
-
-VideoRenderCallback* VideoRenderDirectDraw::CreateChannel(WebRtc_UWord32 channel,
- WebRtc_UWord32 zOrder,
- float startWidth,
- float startHeight,
- float stopWidth,
- float stopHeight)
-{
- if (!_canStretch)
- {
- if (startWidth != 0.0f || startHeight != 0.0f || stopWidth != 1.0f
- || stopHeight != 1.0f)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw failed to CreateChannel HW don't support stretch");
- return NULL;
- }
- }
- DirectDrawChannel* ddobj =
- new DirectDrawChannel(_directDraw, _blitVideoType,
- _incomingVideoType, _rgbVideoType, this);
- ddobj->SetStreamSettings(this, 0, startWidth, startHeight, stopWidth,
- stopHeight);
-
- // store channel
- _directDrawChannels[channel & 0x0000ffff] = ddobj;
-
- // store Z order
- // default streamID is 0
- _directDrawZorder.insert(ZorderPair(zOrder, channel & 0x0000ffff));
- return ddobj;
-}
-
-int VideoRenderDirectDraw::AddDirectDrawChannel(int channel,
- unsigned char streamID,
- int zOrder,
- DirectDrawChannel* ddObj)
-{
- // Only allow one stream per channel, demuxing is done outside of DirectDraw...
- streamID = 0;
- unsigned int streamChannel = (streamID << 16) + (channel & 0x0000ffff);
-
- // store channel
- _directDrawChannels[channel & 0x0000ffff] = ddObj;
-
- _demuxing = true; // with this function it's always demux
-
- // store Z order
- _directDrawZorder.insert(ZorderPair(zOrder, streamChannel));
- return 0;
-}
-
-DirectDrawChannel* VideoRenderDirectDraw::ShareDirectDrawChannel(
- int channel)
-{
- CriticalSectionScoped cs(*_confCritSect);
-
- DirectDrawChannel* obj = NULL;
-
- std::map<int, DirectDrawChannel*>::iterator ddIt;
- ddIt = _directDrawChannels.find(channel & 0x0000ffff);
- if (ddIt != _directDrawChannels.end())
- {
- obj = ddIt->second;
- obj->AddRef();
- }
- return obj;
-}
-
-WebRtc_Word32 VideoRenderDirectDraw::DeleteChannel(const WebRtc_UWord32 channel)
-{
- CriticalSectionScoped cs(*_confCritSect);
-
- // Remove the old z order
-
- //unsigned int streamChannel = (streamID << 16) + (channel & 0x0000ffff);
- std::multimap<int, unsigned int>::iterator it;
- it = _directDrawZorder.begin();
- while (it != _directDrawZorder.end())
- {
- //if(streamChannel == it->second )
- if ((channel & 0x0000ffff) == (it->second & 0x0000ffff))
- {
- it = _directDrawZorder.erase(it);
- break;
- }
- it++;
- }
-
- std::map<int, DirectDrawChannel*>::iterator ddIt;
- ddIt = _directDrawChannels.find(channel & 0x0000ffff);
- if (ddIt != _directDrawChannels.end())
- {
- ddIt->second->Release();
- _directDrawChannels.erase(ddIt);
- _clearMixingSurface = true;
- }
-
- return 0;
-}
-
-WebRtc_Word32 VideoRenderDirectDraw::GetStreamSettings(const WebRtc_UWord32 channel,
- const WebRtc_UWord16 streamId,
- WebRtc_UWord32& zOrder,
- float& startWidth,
- float& startHeight,
- float& stopWidth,
- float& stopHeight)
-{
- CriticalSectionScoped cs(*_confCritSect);
-
- std::map<int, DirectDrawChannel*>::iterator ddIt;
- ddIt = _directDrawChannels.find(channel & 0x0000ffff);
- if (ddIt == _directDrawChannels.end())
- {
- // This channel doesn't exist.
- return -1;
- }
-
- DirectDrawChannel* ptrChannel = ddIt->second;
- // Only support one stream per channel, is demuxing done outside if DD.
- //if (ptrChannel->GetStreamSettings(this, streamId, startWidth, startHeight, stopWidth, stopHeight) == -1)
- if (ptrChannel->GetStreamSettings(this, 0, startWidth, startHeight,
- stopWidth, stopHeight) == -1)
- {
- // Error for this stream
- return -1;
- }
-
- // Get the zOrder
- std::multimap<int, unsigned int>::iterator it;
- it = _directDrawZorder.begin();
- while (it != _directDrawZorder.end())
- {
- if ((channel & 0x0000ffff) == (it->second & 0x0000ffff))
- {
- // We found our channel zOrder
- zOrder = (unsigned int) (it->first);
- break;
- }
- it++;
- }
-
- return 0;
-}
-
-int VideoRenderDirectDraw::GetChannels(std::list<int>& channelList)
-{
- CriticalSectionScoped cs(*_confCritSect);
-
- std::map<int, DirectDrawChannel*>::iterator ddIt;
- ddIt = _directDrawChannels.begin();
-
- while (ddIt != _directDrawChannels.end())
- {
- int channel = ddIt->first;
- if (channel == 0x0000ffff)
- {
- channel = -1;
- }
- channelList.push_back(channel);
- ddIt++;
- }
- return 0;
-}
-
-bool VideoRenderDirectDraw::HasChannel(int channel)
-{
- CriticalSectionScoped cs(*_confCritSect);
-
- std::map<int, DirectDrawChannel*>::iterator ddIt;
- ddIt = _directDrawChannels.find(channel & 0x0000ffff);
- if (ddIt != _directDrawChannels.end())
- {
- return true;
- }
- return false;
-}
-
-bool VideoRenderDirectDraw::HasChannels()
-{
- CriticalSectionScoped cs(*_confCritSect);
-
- if (_directDrawChannels.begin() != _directDrawChannels.end())
- {
- return true;
- }
- return false;
-}
-
-bool VideoRenderDirectDraw::IsFullScreen()
-{
- return _fullscreen;
-}
-
-VideoType VideoRenderDirectDraw::GetPerferedVideoFormat()
-{
- return _incomingVideoType;
-}
-
-// this can be called rutime from another thread
-DirectDrawChannel* VideoRenderDirectDraw::ConfigureDirectDrawChannel(int channel,
- unsigned char streamID,
- int zOrder,
- float left,
- float top,
- float right,
- float bottom)
-{
- // Only support one stream per channel, is demuxing done outside if DD.
- streamID = 0;
-
- CriticalSectionScoped cs(*_confCritSect);
-
- if (!_canStretch)
- {
- if (left != 0.0f || top != 0.0f || right != 1.0f || bottom != 1.0f)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw failed to ConfigureDirectDrawChannel HW don't support stretch");
- return NULL;
- }
- }
- std::map<int, DirectDrawChannel*>::iterator ddIt;
- ddIt = _directDrawChannels.find(channel & 0x0000ffff);
- DirectDrawChannel* ddobj = NULL;
- if (ddIt != _directDrawChannels.end())
- {
- ddobj = ddIt->second;
- }
- if (ddobj == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
- "DirectDraw failed to find channel");
- return NULL;
- }
- unsigned int streamChannel = (streamID << 16) + (channel & 0x0000ffff);
- // remove the old z order
- std::multimap<int, unsigned int>::iterator it;
- it = _directDrawZorder.begin();
- while (it != _directDrawZorder.end())
- {
- if (streamChannel == it->second)
- {
- it = _directDrawZorder.erase(it);
- break;
- }
- it++;
- }
- // if this channel already are in the zOrder map it's demux
- it = _directDrawZorder.begin();
- while (it != _directDrawZorder.end())
- {
- if (channel == (it->second & 0x0000ffff))
- {
- _demuxing = true;
- break;
- }
- it++;
- }
- if (it == _directDrawZorder.end())
- {
- _demuxing = false;
- }
-
- _clearMixingSurface = true;
-
- if (left == 0.0f && top == 0.0f && right == 0.0f && bottom == 0.0f)
- {
- // remove
- _directDrawChannels.erase(ddIt);
- ddobj->Release();
- return NULL;
- }
- ddobj->SetStreamSettings(this, streamID, left, top, right, bottom);
-
- _directDrawZorder.insert(ZorderPair(zOrder, streamChannel));
- return ddobj;
-}
-
-WebRtc_Word32 VideoRenderDirectDraw::SetCropping(const WebRtc_UWord32 channel,
- const WebRtc_UWord16 streamID,
- float left, float top,
- float right, float bottom)
-{
- CriticalSectionScoped cs(*_confCritSect);
- if (!_canStretch)
- {
- if (left != 0.0f || top != 0.0f || right != 1.0f || bottom != 1.0f)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideoRenderer, -1,
- "DirectDraw failed to SetCropping HW don't support stretch");
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- }
-
- std::map<int, DirectDrawChannel*>::iterator ddIt;
- ddIt = _directDrawChannels.find(channel & 0x0000ffff);
- if (ddIt != _directDrawChannels.end())
- {
- DirectDrawChannel* ddobj = ddIt->second;
- if (ddobj)
- {
- // Only support one stream per channel, is demuxing done outside if DD.
- ddobj->SetStreamCropSettings(this, 0, left, top, right, bottom);
- //ddobj->SetStreamCropSettings(this, streamID, left, top, right, bottom);
- }
- }
- return 0;
-}
-
-WebRtc_Word32 VideoRenderDirectDraw::ConfigureRenderer(const WebRtc_UWord32 channel,
- const WebRtc_UWord16 streamId,
- const unsigned int zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- if (ConfigureDirectDrawChannel(channel, (unsigned char) streamId, zOrder,
- left, top, right, bottom) == NULL)
- {
- if (left == 0.0f && top == 0.0f && right == 0.0f && bottom == 0.0f)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideoRenderer, -1,
- "ConfigureRender, removed channel:%d streamId:%d",
- channel, streamId);
- }
- else
- {
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideoRenderer,
- -1,
- "DirectDraw failed to ConfigureRenderer for channel: %d",
- channel);
- return -1;
- }
- }
- return 0;
-}
-
-// this can be called runtime from another thread
-WebRtc_Word32 VideoRenderDirectDraw::SetText(const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 colorText,
- const WebRtc_UWord32 colorBg,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- DirectDrawTextSettings* textSetting = NULL;
-
- CriticalSectionScoped cs(*_confCritSect);
-
- _frameChanged = true;
-
- std::map<unsigned char, DirectDrawTextSettings*>::iterator it;
- it = _textSettings.find(textId);
- if (it != _textSettings.end())
- {
- if (it->second)
- {
- textSetting = it->second;
- }
- }
- _clearMixingSurface = true;
-
- if (text == NULL || textLength == 0)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw remove text textId:%d", textId);
- if (textSetting)
- {
- delete textSetting;
- _textSettings.erase(it);
- }
- return 0;
- }
-
- // sanity
- if (left > 1.0f || left < 0.0f)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw SetText invalid parameter");
- return -1;
- //return VIDEO_DIRECT_DRAW_INVALID_ARG;
- }
- if (top > 1.0f || top < 0.0f)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw SetText invalid parameter");
- return -1;
- //return VIDEO_DIRECT_DRAW_INVALID_ARG;
- }
- if (right > 1.0f || right < 0.0f)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw SetText invalid parameter");
- return -1;
- //return VIDEO_DIRECT_DRAW_INVALID_ARG;
- }
- if (bottom > 1.0f || bottom < 0.0f)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw SetText invalid parameter");
- return -1;
- //return VIDEO_DIRECT_DRAW_INVALID_ARG;
- }
- if (textSetting == NULL)
- {
- textSetting = new DirectDrawTextSettings();
- }
- int retVal = textSetting->SetText((const char*) text, textLength,
- (COLORREF) colorText, (COLORREF) colorBg,
- left, top, right, bottom);
- if (retVal != 0)
- {
- delete textSetting;
- textSetting = NULL;
- _textSettings.erase(textId);
- return retVal;
- }
- if (textSetting)
- {
- _textSettings[textId] = textSetting;
- }
- return retVal;
-}
-
-// this can be called runtime from another thread
-WebRtc_Word32 VideoRenderDirectDraw::SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- DirectDrawBitmapSettings* bitmapSetting = NULL;
-
- CriticalSectionScoped cs(*_confCritSect);
-
- _frameChanged = true;
- std::map<unsigned char, DirectDrawBitmapSettings*>::iterator it;
- it = _bitmapSettings.find(pictureId);
- if (it != _bitmapSettings.end())
- {
- if (it->second)
- {
- bitmapSetting = it->second;
- }
- }
- _clearMixingSurface = true;
-
- if (bitMap == NULL)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw remove bitmap pictureId:%d", pictureId);
- if (bitmapSetting)
- {
- delete bitmapSetting;
- _bitmapSettings.erase(it);
- }
- return 0;
- }
-
- // sanity
- if (left > 1.0f || left < 0.0f)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw SetBitmap invalid parameter");
- return -1;
- //return VIDEO_DIRECT_DRAW_INVALID_ARG;
- }
- if (top > 1.0f || top < 0.0f)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw SetBitmap invalid parameter");
- return -1;
- //return VIDEO_DIRECT_DRAW_INVALID_ARG;
- }
- if (right > 1.0f || right < 0.0f)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw SetBitmap invalid parameter");
- return -1;
- //return VIDEO_DIRECT_DRAW_INVALID_ARG;
- }
- if (bottom > 1.0f || bottom < 0.0f)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw SetBitmap invalid parameter");
- return -1;
- //return VIDEO_DIRECT_DRAW_INVALID_ARG;
- }
- if (!_canStretch)
- {
- if (left != 0.0f || top != 0.0f || right != 1.0f || bottom != 1.0f)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw failed to SetBitmap HW don't support stretch");
- return -1;
- //return VIDEO_DIRECT_DRAW_INVALID_ARG;
- }
- }
- if (bitmapSetting == NULL)
- {
- bitmapSetting = new DirectDrawBitmapSettings();
- }
-
- bitmapSetting->_transparentBitMap = (HBITMAP) bitMap;
- bitmapSetting->_transparentBitmapLeft = left;
- bitmapSetting->_transparentBitmapRight = right;
- bitmapSetting->_transparentBitmapTop = top;
- bitmapSetting->_transparentBitmapBottom = bottom;
-
- // colorKey == NULL equals no transparency
- if (colorKey)
- {
- // first remove constness
- DDCOLORKEY* ddColorKey =
- static_cast<DDCOLORKEY*> (const_cast<void*> (colorKey));
- if (!_supportTransparency)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw failed to SetBitmap HW don't support transparency");
- return -1;
- //return VIDEO_DIRECT_DRAW_INVALID_ARG;
- }
- if (bitmapSetting->_transparentBitmapColorKey == NULL)
- {
- bitmapSetting->_transparentBitmapColorKey = new DDCOLORKEY();
- }
-
- if (ddColorKey)
- {
- bitmapSetting->_transparentBitmapColorKey->dwColorSpaceLowValue
- = ddColorKey->dwColorSpaceLowValue;
- bitmapSetting->_transparentBitmapColorKey->dwColorSpaceHighValue
- = ddColorKey->dwColorSpaceHighValue;
- }
- }
- int retval = bitmapSetting->SetBitmap(_trace, _directDraw);
- if (retval != 0)
- {
- delete bitmapSetting;
- bitmapSetting = NULL;
- _bitmapSettings.erase(pictureId);
- return retval;
- }
- if (bitmapSetting)
- {
- _bitmapSettings[pictureId] = bitmapSetting;
- }
- return retval;
-}
-
-// this can be called rutime from another thread
-WebRtc_Word32 VideoRenderDirectDraw::SetTransparentBackground(
- const bool enable)
-{
- CriticalSectionScoped cs(*_confCritSect);
-
- if (_supportTransparency)
- {
- _transparentBackground = enable;
- if (enable)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw enabled TransparentBackground");
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw disabled TransparentBackground");
- }
- return 0;
- }
- WEBRTC_TRACE(
- kTraceError,
- kTraceVideo,
- -1,
- "DirectDraw failed to EnableTransparentBackground HW don't support transparency");
- return -1;
- //return VIDEO_DIRECT_DRAW_INVALID_ARG;
-}
-
-int VideoRenderDirectDraw::FillSurface(DirectDrawSurface *pDDSurface,
- RECT* rect)
-{
- // sanity checks
- if (NULL == pDDSurface)
- {
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- if (NULL == rect)
- {
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
-
- // Repaint the whole specified surface
- HRESULT ddrval;
- DDBLTFX ddFX;
-
- ZeroMemory(&ddFX, sizeof(ddFX));
- ddFX.dwSize = sizeof(ddFX);
- ddFX.dwFillColor = RGB(0, 0, 0);
-
- // Draw color key on the video area of given surface
- ddrval = pDDSurface->Blt(rect, NULL, NULL, DDBLT_COLORFILL | DDBLT_WAIT,
- &ddFX);
- if (FAILED(ddrval))
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw failed to fill surface");
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- return 0;
-}
-
-// the real rendering thread
-bool VideoRenderDirectDraw::RemoteRenderingThreadProc(void *obj)
-{
- return static_cast<VideoRenderDirectDraw*> (obj)->RemoteRenderingProcess();
-}
-
-bool VideoRenderDirectDraw::RemoteRenderingProcess()
-{
- bool hwndChanged = false;
- int waitTime = 0;
-
- _screenEvent->Wait(100);
-
- _confCritSect->Enter();
-
- if (_blit == false)
- {
- _confCritSect->Leave();
- return true;
- }
-
- if (!::GetForegroundWindow())
- {
- //no window, i.e the user have clicked CTRL+ALT+DEL, return true and wait
- _confCritSect->Leave();
- return true;
- }
-
- // Skip to blit if last render to primare surface took too long time.
- _processCount++;
- if (_deliverInQuarterFrameRate)
- {
- if (_processCount % 4 != 0)
- {
- _confCritSect->Leave();
- return true;
- }
- }
- else if (_deliverInHalfFrameRate)
- {
- if (_processCount % 2 != 0)
- {
- _confCritSect->Leave();
- return true;
- }
- }
-
- // Calculate th erender process time
- unsigned int startProcessTime = timeGetTime();
-
- hwndChanged = HasHWNDChanged();
- if (hwndChanged)
- {
- _clearMixingSurface = true;
- }
-
- std::map<int, DirectDrawChannel*>::iterator it;
- it = _directDrawChannels.begin();
- while (it != _directDrawChannels.end() && !_frameChanged)
- {
- if (it->second)
- {
- int channel = it->first;
- _frameChanged = it->second->IsOffScreenSurfaceUpdated(this);
- }
- it++;
- }
- if (_backSurface)
- {
- if (hwndChanged || _frameChanged)
- {
- BlitFromOffscreenBuffersToMixingBuffer();
- BlitFromBitmapBuffersToMixingBuffer();
- BlitFromTextToMixingBuffer();
- }
- BlitFromMixingBufferToBackBuffer();
- WaitAndFlip(waitTime);
- }
- else
- {
- if (hwndChanged || _frameChanged)
- {
- BlitFromOffscreenBuffersToMixingBuffer();
- BlitFromBitmapBuffersToMixingBuffer();
- BlitFromTextToMixingBuffer();
- }
- BlitFromMixingBufferToFrontBuffer(hwndChanged, waitTime);
-
- }
- // Check the total time it took processing all rendering. Don't consider waitTime.
- //const int totalRenderTime=GET_TIME_IN_MS()- startProcessTime-waitTime;
- const int totalRenderTime = ::timeGetTime() - startProcessTime - waitTime;
- DecideBestRenderingMode(hwndChanged, totalRenderTime);
- _frameChanged = false;
- _confCritSect->Leave();
-
- return true;
-}
-void VideoRenderDirectDraw::DecideBestRenderingMode(bool hwndChanged,
- int totalRenderTime)
-{
- /* Apply variuos fixes for bad graphic drivers.
- 1. If cpu to high- test wait fix
- 2. If cpu still too high render in 1/2 display update period.
- 3. If RemoteRenderingProcess take to long time reduce the blit period to 1/2 display update period.
- 4. If RemoteRenderingProcess still take to long time try color conversion fix. It do color conversion in VieoRenderDirectDrawChannel::DeliverFrame
- 5. If RemoteRenderingProcess still take to long time reduce the blit period to 1/4 display update period and disable color conversion fix.
- 6 if RemoteRenderingProcess still take to long time reduce the blit period to 1/4 display update period and enable color conversion fix again.
- */
-
- const int timesSinceLastCPUCheck = timeGetTime()
- - _screenRenderCpuUsage.LastGetCpuTime();
- int cpu = 0;
-
- if (hwndChanged) // Render window changed.
- {
- cpu = _screenRenderCpuUsage.GetCpuUsage(); // Get CPU usage for this thread. (Called if hwndCanged just to reset the GET CPU Usage function)
- _nrOfTooLongRenderTimes = 0; // Reset count of too long render times.
- return; // Return - nothing more to do since the window has changed.
- }
- // Check total rendering times
- if (_maxAllowedRenderTime > 0 && totalRenderTime > _maxAllowedRenderTime)
- {
- if (!_deliverInHalfFrameRate || totalRenderTime > 2
- * _maxAllowedRenderTime)
- {
- _nrOfTooLongRenderTimes += totalRenderTime / _maxAllowedRenderTime; //Weighted with the number of to long render times
- }
- }
-
- // If we are not using back surface (ie full screen rendering) we might try to switch BlitFromMixingBufferToFrontBuffer mode.
- if (timesSinceLastCPUCheck > WindowsThreadCpuUsage::CPU_CHECK_INTERVAL)
- {
- cpu = _screenRenderCpuUsage.GetCpuUsage(); // Get CPU usage for this thread. (Called if hwndCanged just to reset the GET CPU Usage function)
- WEBRTC_TRACE(
- kTraceStream,
- kTraceVideo,
- -1,
- "Screen render thread cpu usage. (Tid %d), cpu usage %d processTime %d, no of too long render times %d",
- GetCurrentThreadId(), cpu, totalRenderTime,
- _nrOfTooLongRenderTimes);
-
- // If this screen render thread uses more than 5% of the total CPU time and the
- // 1. try waitFix
- if (cpu >= 5 && _renderModeWaitForCorrectScanLine == false
- && !_backSurface)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVideo,
- -1,
- "HIGH screen render thread cpu usage. (Tid %d), cpu usage %d, applying wait for scan line",
- GetCurrentThreadId(), cpu);
- _renderModeWaitForCorrectScanLine = true;
- _fullScreenWaitEvent->StartTimer(true, 1);
- }
- else if (cpu >= 10 && _deliverInHalfFrameRate == false)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVideo,
- -1,
- "HIGH screen render thread cpu usage. (Tid %d), cpu usage %d, Render half rate",
- GetCurrentThreadId(), cpu);
- _deliverInHalfFrameRate = true;
- }
- else
- {
- // Check if rendering takes too long time
- if (_nrOfTooLongRenderTimes > 15 || totalRenderTime
- >= WindowsThreadCpuUsage::CPU_CHECK_INTERVAL)
- {
-
- // The rendering is taking too long time
- if (_deliverInHalfFrameRate == false)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "Render half rate, tid: %d",
- GetCurrentThreadId());
- _deliverInHalfFrameRate = true;
- }
- else if (_deliverInScreenType == false
- && !_deliverInQuarterFrameRate)
- {
- WEBRTC_TRACE(
- kTraceInfo,
- kTraceVideo,
- -1,
- "Applying deliver in screen type format, tid: %d",
- GetCurrentThreadId());
- // 2. try RGB fix
- std::map<int, DirectDrawChannel*>::iterator it;
- it = _directDrawChannels.begin();
- while (it != _directDrawChannels.end())
- {
- it->second->ChangeDeliverColorFormat(true);
- it++;
- }
- _deliverInScreenType = true;
- }
- else if (_deliverInQuarterFrameRate == false)
- {
- WEBRTC_TRACE(
- kTraceInfo,
- kTraceVideo,
- -1,
- "Render quarter rate and disable deliver in screen type format, tid: %d",
- GetCurrentThreadId());
- _deliverInQuarterFrameRate = true;
- if (_deliverInScreenType)
- {
- //Disable RGB fix
- std::map<int, DirectDrawChannel*>::iterator it;
- it = _directDrawChannels.begin();
- while (it != _directDrawChannels.end())
- {
- it->second->ChangeDeliverColorFormat(false);
- it++;
- }
- _deliverInScreenType = false;
- }
- }
- else if (_deliverInQuarterFrameRate == true
- && !_deliverInScreenType)
- {
- WEBRTC_TRACE(
- kTraceInfo,
- kTraceVideo,
- -1,
- "Render quarter rate and enable RGB fix, tid: %d",
- GetCurrentThreadId());
- _deliverInQuarterFrameRate = true;
-
- //Enabe RGB fix
- std::map<int, DirectDrawChannel*>::iterator it;
- it = _directDrawChannels.begin();
- while (it != _directDrawChannels.end())
- {
- it->second->ChangeDeliverColorFormat(true);
- it++;
- }
- _deliverInScreenType = true;
- }
- }
- }
- _nrOfTooLongRenderTimes = 0; // Reset count of too long render times.
- }
-}
-
-/*
- * Internal help functions for blitting
- */
-
-bool VideoRenderDirectDraw::HasHWNDChanged()
-{
- // we check if the HWND has changed
- if (!_fullscreen)
- {
- RECT currentRect;
- ::GetClientRect(_hWnd, ¤tRect);
- if (!EqualRect(¤tRect, &_hwndRect))
- {
- int retVal = CreateMixingSurface(); // this will delete the old mixing surface
- if (retVal != 0)
- {
- return false;
- }
- return true;
- }
- }
- return false;
-}
-
-int VideoRenderDirectDraw::BlitFromOffscreenBuffersToMixingBuffer()
-{
- bool updateAll = false; // used to minimize the number of blt
-
- DDBLTFX ddbltfx;
- ZeroMemory(&ddbltfx, sizeof(ddbltfx));
- ddbltfx.dwSize = sizeof(ddbltfx);
- ddbltfx.dwDDFX = DDBLTFX_NOTEARING;
-
- if (_mixingSurface == NULL)
- {
- int retVal = CreateMixingSurface();
- if (retVal != 0)
- {
- // trace done
- return retVal;
- }
- }
- RECT mixingRect;
- ::SetRectEmpty(&mixingRect);
-
- if (_fullscreen)
- {
- ::CopyRect(&mixingRect, &_screenRect);
- }
- else
- {
- ::CopyRect(&mixingRect, &_hwndRect);
- // what if largest size is larger than screen
- if (mixingRect.right > _screenRect.right)
- {
- mixingRect.right = _screenRect.right;
- }
- if (mixingRect.bottom > _screenRect.bottom)
- {
- mixingRect.bottom = _screenRect.bottom;
- }
- }
- if (!EqualRect(&_mixingRect, &mixingRect))
- {
- // size changed
- CopyRect(&_mixingRect, &mixingRect);
- FillSurface(_mixingSurface, &mixingRect);
- updateAll = true;
- }
-
- if (_clearMixingSurface)
- {
- FillSurface(_mixingSurface, &_mixingRect);
- _clearMixingSurface = false;
- updateAll = true;
- }
-
- std::multimap<int, unsigned int>::reverse_iterator it;
- it = _directDrawZorder.rbegin();
- while (it != _directDrawZorder.rend())
- {
- // loop through all channels and streams in Z order
- short streamID = (it->second >> 16);
- int channel = it->second & 0x0000ffff;
-
- std::map<int, DirectDrawChannel*>::iterator ddIt;
- ddIt = _directDrawChannels.find(channel);
- if (ddIt != _directDrawChannels.end())
- {
- // found the channel
- DirectDrawChannel* channelObj = ddIt->second;
- if (channelObj && _mixingSurface)
- {
- if (updateAll || channelObj->IsOffScreenSurfaceUpdated(this))
- {
- updateAll = true;
- if (channelObj->BlitFromOffscreenBufferToMixingBuffer(
- this,
- streamID,
- _mixingSurface,
- _mixingRect,
- _demuxing)
- != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- -1,
- "DirectDraw error BlitFromOffscreenBufferToMixingBuffer ");
- _mixingSurface->Release();
- _mixingSurface = NULL;
- }
- }
- }
- }
- it++;
- }
- return 0;
-}
-
-int VideoRenderDirectDraw::BlitFromTextToMixingBuffer()
-{
- if (_directDraw == NULL)
- {
- return -1;
- }
- if (!_mixingSurface)
- {
- return -1;
- }
- if (_textSettings.empty())
- {
- return 0;
- }
-
- HDC hdcDDSurface;
- HRESULT res = _mixingSurface->GetDC(&hdcDDSurface);
- if (res != S_OK)
- {
- return -1;
- }
- //
- std::map<unsigned char, DirectDrawTextSettings*>::reverse_iterator it;
- it = _textSettings.rbegin();
-
- while (it != _textSettings.rend())
- {
- DirectDrawTextSettings* settings = it->second;
- it++;
- if (settings == NULL)
- {
- continue;
- }
- SetTextColor(hdcDDSurface, settings->_colorRefText);
- SetBkColor(hdcDDSurface, settings->_colorRefBackground);
-
- if (settings->_transparent)
- {
- SetBkMode(hdcDDSurface, TRANSPARENT); // do we need to call this all the time?
- }
- else
- {
- SetBkMode(hdcDDSurface, OPAQUE); // do we need to call this all the time?
- }
- RECT textRect;
- textRect.left = int(_mixingRect.right * settings->_textLeft);
- textRect.right = int(_mixingRect.right * settings->_textRight);
- textRect.top = int(_mixingRect.bottom * settings->_textTop);
- textRect.bottom = int(_mixingRect.bottom * settings->_textBottom);
-
- DrawTextA(hdcDDSurface, settings->_ptrText, settings->_textLength,
- &textRect, DT_LEFT);
- }
- _mixingSurface->ReleaseDC(hdcDDSurface);
- return 0;
-}
-
-int VideoRenderDirectDraw::BlitFromBitmapBuffersToMixingBuffer()
-{
- HRESULT ddrval;
- DDBLTFX ddbltfx;
- ZeroMemory(&ddbltfx, sizeof(ddbltfx));
- ddbltfx.dwSize = sizeof(ddbltfx);
- ddbltfx.dwDDFX = DDBLTFX_NOTEARING;
-
- if (_directDraw == NULL)
- {
- return -1; // signal that we are not ready for the change
- }
-
- std::map<unsigned char, DirectDrawBitmapSettings*>::reverse_iterator it;
- it = _bitmapSettings.rbegin();
-
- while (it != _bitmapSettings.rend())
- {
- DirectDrawBitmapSettings* settings = it->second;
- it++;
- if (settings == NULL)
- {
- continue;
- }
-
- // Color keying lets you set colors on a surface to be completely transparent.
- // always blit _transparentBitmapSurface last
- if (_mixingSurface && settings->_transparentBitmapSurface
- && settings->_transparentBitmapWidth
- && settings->_transparentBitmapHeight)
- {
- DWORD signal = DDBLT_WAIT | DDBLT_DDFX;
- // Set transparent color
- if (settings->_transparentBitmapColorKey)
- {
- signal |= DDBLT_KEYSRC;
- settings->_transparentBitmapSurface->SetColorKey(
- DDCKEY_SRCBLT,
- settings->_transparentBitmapColorKey);
- }
-
- // Now we can blt the transparent surface to another surface
- RECT srcRect;
- SetRect(&srcRect, 0, 0, settings->_transparentBitmapWidth,
- settings->_transparentBitmapHeight);
-
- RECT dstRect;
- if (settings->_transparentBitmapLeft
- != settings->_transparentBitmapRight
- && settings->_transparentBitmapTop
- != settings->_transparentBitmapBottom)
- {
- CopyRect(&dstRect, &_mixingRect);
- dstRect.left = (int) (dstRect.right
- * settings->_transparentBitmapLeft);
- dstRect.right = (int) (dstRect.right
- * settings->_transparentBitmapRight);
- dstRect.top = (int) (dstRect.bottom
- * settings->_transparentBitmapTop);
- dstRect.bottom = (int) (dstRect.bottom
- * settings->_transparentBitmapBottom);
- }
- else
- {
-
- // if left, right, top and bottom are describing one point use the original size
- CopyRect(&dstRect, &srcRect);
- POINT startp;
- startp.x = (int) (_mixingRect.right
- * settings->_transparentBitmapLeft);
- startp.y = (int) (_mixingRect.bottom
- * settings->_transparentBitmapTop);
- OffsetRect(&dstRect, startp.x, startp.y);
-
- // make sure that we blit inside our surface
- if (dstRect.bottom > _mixingRect.bottom)
- {
- srcRect.bottom -= dstRect.bottom - _mixingRect.bottom;
- // sanity
- if (srcRect.bottom < 0)
- {
- srcRect.bottom = 0;
- }
- dstRect.bottom = _mixingRect.bottom;
- }
- if (dstRect.right > _mixingRect.right)
- {
- srcRect.right -= dstRect.right - _mixingRect.right;
- // sanity
- if (srcRect.right < 0)
- {
- srcRect.right = 0;
- }
- dstRect.right = _mixingRect.right;
- }
- }
- // ddbltfx.dwDDFX |= DDBLTFX_MIRRORUPDOWN; //only for test requires hw support
-
- // wait for the _mixingSurface to be available
- ddrval = _mixingSurface->Blt(&dstRect,
- settings->_transparentBitmapSurface,
- &srcRect, signal, &ddbltfx);
- if (ddrval == DDERR_SURFACELOST)
- {
- if (!::GetForegroundWindow())
- {
- // no window, i.e the user have clicked CTRL+ALT+DEL
- return 0;
- }
- // always re-creted via the SetBitmap call
- settings->_transparentBitmapSurface->Release();
- settings->_transparentBitmapSurface = NULL;
-
- _clearMixingSurface = true;
-
- if (settings->_transparentBitMap)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw re-set transparent bitmap");
- settings->SetBitmap(_trace, _directDraw);
- }
- }
- else if (ddrval != DD_OK)
- {
- settings->_transparentBitmapSurface->Release();
- settings->_transparentBitmapSurface = NULL;
- WEBRTC_TRACE(
- kTraceInfo,
- kTraceVideo,
- -1,
- "DirectDraw blt error 0x%x _transparentBitmapSurface",
- ddrval);
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- }
- }
- return 0;
-}
-
-/**
- * normal blitting
- */
-int VideoRenderDirectDraw::BlitFromMixingBufferToFrontBuffer(
- bool hwndChanged,
- int& waitTime)
-{
- DDBLTFX ddbltfx;
- ZeroMemory(&ddbltfx, sizeof(ddbltfx));
- ddbltfx.dwSize = sizeof(ddbltfx);
- ddbltfx.dwDDFX = DDBLTFX_NOTEARING;
- RECT rcRectDest;
-
- // test for changing mode
- /* for(int i= 0; i< 6000000; i ++)
- {
- rcRectDest.left = i;
- }
- */
-
- if (IsRectEmpty(&_mixingRect))
- {
- // no error just nothing to blit
- return 0;
- }
- if (_mixingSurface == NULL)
- {
- // The mixing surface has probably been deleted
- // and we haven't had time to restore it yet. Wait...
- return 0;
- }
- if (_primarySurface == NULL)
- {
- int retVal = CreatePrimarySurface();
- if (retVal != 0)
- {
- // tracing done
- return retVal;
- }
- }
-
- // first we need to figure out where on the primary surface our window lives
- ::GetWindowRect(_hWnd, &rcRectDest);
-
- DWORD signal = DDBLT_WAIT | DDBLT_DDFX;
-
- // Set transparent color
- if (_transparentBackground)
- {
- signal |= DDBLT_KEYSRC;
- DDCOLORKEY ColorKey;
- ColorKey.dwColorSpaceLowValue = RGB(0, 0, 0);
- ColorKey.dwColorSpaceHighValue = RGB(0, 0, 0);
- _mixingSurface->SetColorKey(DDCKEY_SRCBLT, &ColorKey);
- }
-
- if (_renderModeWaitForCorrectScanLine)
- {
- // wait for previus draw to complete
- DWORD scanLines = 0;
- DWORD screenLines = _screenRect.bottom - 1; // scanlines start on 0
- DWORD screenLines90 = (screenLines * 9) / 10; // % of the screen is rendered
- //waitTime=GET_TIME_IN_MS();
- waitTime = ::timeGetTime();
- HRESULT hr = _directDraw->GetScanLine(&scanLines);
- while (screenLines90 > scanLines && hr == DD_OK)
- {
- _confCritSect->Leave();
- _fullScreenWaitEvent->Wait(3);
- _confCritSect->Enter();
- if (_directDraw == NULL)
- {
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- hr = _directDraw->GetScanLine(&scanLines);
- }
- //waitTime=GET_TIME_IN_MS()-waitTime;
- waitTime = ::timeGetTime() - waitTime;
- }
-
- HRESULT ddrval = _primarySurface->Blt(&rcRectDest, _mixingSurface,
- &_mixingRect, signal, &ddbltfx);
- if (ddrval == DDERR_SURFACELOST)
- {
- if (!::GetForegroundWindow())
- {
- // no window, i.e the user have clicked CTRL+ALT+DEL
- return 0;
- }
- ddrval = _primarySurface->Restore();
- if (ddrval == DD_OK) // Try again
- {
- ddrval = _primarySurface->Blt(&rcRectDest, _mixingSurface,
- &_mixingRect, signal, &ddbltfx);
- }
- if (ddrval != DD_OK) // If restore failed or second time blt failed. Delete the surface. It will be recreated next time.
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVideo,
- -1,
- "DirectDraw failed to restore lost _primarySurface 0x%x",
- ddrval);
- _primarySurface->Release();
- _primarySurface = NULL;
- if (_mixingSurface)
- {
- _mixingSurface->Release();
- _mixingSurface = NULL;
- }
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw restored lost _primarySurface");
- }
- else if (ddrval == DDERR_EXCEPTION)
- {
- _primarySurface->Release();
- _primarySurface = NULL;
- if (_mixingSurface)
- {
- _mixingSurface->Release();
- _mixingSurface = NULL;
- }
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw exception in _primarySurface");
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- if (ddrval != DD_OK)
- {
- if (ddrval != 0x80004005) // Undefined error. Ignore
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw blt error 0x%x _primarySurface", ddrval);
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- }
- return 0;
-}
-
-/**
- * fullscreen mode blitting
- */
-
-int VideoRenderDirectDraw::WaitAndFlip(int& waitTime)
-{
- if (_primarySurface == NULL)
- {
- // no trace, too much in the file
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- if (_directDraw == NULL)
- {
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- // wait for previus draw to complete
- DWORD scanLines = 0;
- DWORD screenLines = _screenRect.bottom - 1; // scanlines start on 0
- DWORD screenLines90 = (screenLines * 9) / 10; // % of the screen is rendered
-
- //waitTime=GET_TIME_IN_MS();
- waitTime = ::timeGetTime();
- HRESULT hr = _directDraw->GetScanLine(&scanLines);
- while (screenLines90 > scanLines && hr == DD_OK)
- {
- _confCritSect->Leave();
- _fullScreenWaitEvent->Wait(3);
- _confCritSect->Enter();
- if (_directDraw == NULL)
- {
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- hr = _directDraw->GetScanLine(&scanLines);
- }
- //waitTime=GET_TIME_IN_MS()-waitTime;
- waitTime = ::timeGetTime() - waitTime;
- if (screenLines > scanLines)
- {
- // this function sucks a lot of the CPU... but it's worth it
- _directDraw->WaitForVerticalBlank(DDWAITVB_BLOCKBEGIN, NULL);
- }
-
- // schedule a flip
- HRESULT ddrval = _primarySurface->Flip(NULL, DDFLIP_WAIT); // schedule flip DDFLIP_WAIT
- if (ddrval == DDERR_SURFACELOST)
- {
- if (!::GetForegroundWindow())
- {
- // no window, i.e the user have clicked CTRL+ALT+DEL
- return 0;
- }
- //if(::IsIconic(_hWnd))
- //{
- // need to do this before Restore
- //WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1, "DirectDraw our window is an icon maximize it ");
- // When the full screen window is switched out by ALT-TAB or ALT-CTRL-DEL-TASKMANAGER,
- // this call will hang the app. Remove it to fix the problem.
- // FIXME:
- // 1) Why we want to active and max the window when it was minimized?
- // 2) Why this is needed before restore? We didn't do that in non full screen mode.
- //::ShowWindow(_hWnd, SW_SHOWMAXIMIZED);
- //}
- ddrval = _primarySurface->Restore();
- if (ddrval != DD_OK)
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVideo,
- -1,
- "DirectDraw failed to restore _primarySurface, in flip, 0x%x",
- ddrval);
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw restore _primarySurface in flip");
-
- }
- else if (ddrval != DD_OK)
- {
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- return 0;
-}
-
-int VideoRenderDirectDraw::BlitFromMixingBufferToBackBuffer()
-{
- if (_backSurface == NULL)
- {
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- if (IsRectEmpty(&_mixingRect))
- {
- // nothing to blit
- return 0;
- }
- DDBLTFX ddbltfx;
- ZeroMemory(&ddbltfx, sizeof(ddbltfx));
- ddbltfx.dwSize = sizeof(ddbltfx);
- ddbltfx.dwDDFX = DDBLTFX_NOTEARING;
-
- // wait for the _backSurface to be available
- HRESULT ddrval = _backSurface->Blt(&_screenRect, _mixingSurface,
- &_mixingRect, DDBLT_WAIT | DDBLT_DDFX,
- &ddbltfx);
- if (ddrval == DDERR_SURFACELOST)
- {
- if (!::GetForegroundWindow())
- {
- // no window, i.e the user have clicked CTRL+ALT+DEL
- return 0;
- }
- //if(::IsIconic(_hWnd))
- //{
- // need to do this before Restore
- //WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1, "DirectDraw our window is an icon maximize it ");
- //WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1, "DirectDraw show our window is an icon maximize it ");
- // When the full screen window is switch out by ALT-TAB or ALT-CTRL-DEL-TASKMANAGER,
- // this call will hang the app. Remove it to fix the problem.
- // FIXME:
- // 1) Why we want to active and max the window when it was minimized?
- // 2) Why this is needed before restore? We didn't do that in non full screen mode.
- //::ShowWindow(_hWnd, SW_SHOWMAXIMIZED);
- //}
- ddrval = _primarySurface->Restore();
- if (ddrval != DD_OK)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
- "DirectDraw failed to restore _primarySurface");
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "DirectDraw restored _primarySurface");
-
- _clearMixingSurface = true;
-
- }
- else if (ddrval != DD_OK)
- {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "DirectDraw blt error 0x%x _backSurface", ddrval);
- return -1;
- //return VIDEO_DIRECT_DRAW_FAILURE;
- }
- return 0;
-}
-
-/*
- Saving the code for using a clip list instead of HWND, problem was that other transparent
- HWNDs caused us not to update an area or that we painted in other HWNDs area.
-
- RECT hWndRect;
- ::GetWindowRect(_hWnd, &hWndRect);
-
- LPRGNDATA lpClipList = (LPRGNDATA)malloc(sizeof(RGNDATAHEADER) + sizeof(RECT));
-
- // now fill out all the structure fields
- memcpy(lpClipList->Buffer, &hWndRect, sizeof(RECT));
-
- ::CopyRect(&(lpClipList->rdh.rcBound), &hWndRect);
- lpClipList->rdh.dwSize = sizeof(RGNDATAHEADER);
- lpClipList->rdh.iType = RDH_RECTANGLES;
- lpClipList->rdh.nCount = 1;
- lpClipList->rdh.nRgnSize = sizeof(RECT) * lpClipList->rdh.nCount;
- ddrval= _directDrawClipper->SetClipList(lpClipList, 0);
-
- void Visible(HWND hwnd, HRGN &hRgn)
- {
- if (!IsWindowVisible(hwnd)) // If the window is visible
- {
- if(CombineRgn(hRgn, hRgn, hRgn, RGN_XOR) == NULLREGION)
- {
- return;
- }
- }
- // Gets the topmost window
- HWND hWnd=GetTopWindow(NULL);
- while (hWnd != NULL && hWnd != hwnd) // If the window is above in Z-order
- {
- if (IsWindowVisible(hWnd)) // If the window is visible
- {
- RECT Rect;
- // Gets window dimension
- GetWindowRect(hWnd, &Rect);
- // Creates a region corresponding to the window
- if(Rect.left > 0) // test fo rnow
- {
- HRGN hrgnWnd = CreateRectRgn(Rect.left, Rect.top, Rect.right, Rect.bottom);
- // int err = GetUpdateRgn(hWnd, hrgnWnd, FALSE);
- // Creates a region corresponding to region not overlapped
- if(CombineRgn(hRgn, hRgn, hrgnWnd, RGN_DIFF) == COMPLEXREGION)
- {
- int a = 0;
- }
- DeleteObject(hrgnWnd);
- }
- }
- // Loops through all windows till the specified window
- hWnd = GetWindow(hWnd, GW_HWNDNEXT);
- }
-
- HRGN region;
- region = CreateRectRgn(0, 0, 500, 500);
-
- // Get the affected region
- // if (GetUpdateRgn(_hWnd, region, FALSE) != ERROR)
- HDC dc = GetDC(_hWnd);
- if(GetClipRgn(dc, region) > 0)
- {
- int buffsize;
- UINT x;
- RGNDATA *buff;
- POINT TopLeft;
-
- // Get the top-left point of the client area
- TopLeft.x = 0;
- TopLeft.y = 0;
- if (!ClientToScreen(_hWnd, &TopLeft))
- {
- int a = 0;
- }
-
-
- // Get the size of buffer required
- buffsize = GetRegionData(region, 0, 0);
- if (buffsize != 0)
- {
- buff = (RGNDATA *) new BYTE [buffsize];
- if (buff == NULL)
- {
- int a = 0;
- }
-
- // Now get the region data
- if(GetRegionData(region, buffsize, buff))
- {
- if(buff->rdh.nCount > 0)
- {
- ::OffsetRect(&(buff->rdh.rcBound), TopLeft.x, TopLeft.y);
- for (x=0; x<(buff->rdh.nCount); x++)
- {
- RECT *urect = (RECT *) (((BYTE *) buff) + sizeof(RGNDATAHEADER) + (x * sizeof(RECT)));
- ::OffsetRect(urect, TopLeft.x, TopLeft.y);
- char logStr[256];
- _snprintf(logStr,256, "rect T:%d L:%d B:%d R:%d\n",urect->top, urect->left, urect->bottom, urect->right);
- OutputDebugString(logStr);
-
- }
- OutputDebugString("\n");
- _directDrawClipper->SetClipList(buff, 0);
- }
- LPRGNDATA lpClipList = (LPRGNDATA)malloc(sizeof(RGNDATAHEADER) + sizeof(RECT) * buff->rdh.nCount);
- if(buff->rdh.nCount > 0)
- {
- _directDrawClipper->SetClipList(lpClipList, 0);
-
- lpClipList->
- DWORD size = sizeof(RGNDATAHEADER) + sizeof(RECT)* buff->rdh.nCount;
- lpClipList->rdh.dwSize = sizeof(RGNDATAHEADER);
- lpClipList->rdh.iType = RDH_RECTANGLES;
- lpClipList->rdh.nCount = 1;
-
- HRESULT ddrval1 = _directDrawClipper->GetClipList(NULL, lpClipList, &size);
- memcpy(lpClipList->Buffer, &rcRectDest, sizeof(RECT));
- ::CopyRect(&(lpClipList->rdh.rcBound), &rcRectDest);
- _directDrawClipper->SetClipList(lpClipList, 0);
- } }
-
- for (x=0; x<(buff->rdh.nCount); x++)
- {
- // Obtain the rectangles from the list
- RECT *urect = (RECT *) (((BYTE *) buff) + sizeof(RGNDATAHEADER) + (x * sizeof(RECT)));
- int a = 0;
-
- }
- delete lpClipList;
- }
- delete buff;
- }
- }
- */
-/*
- void VideoRenderDirectDraw::Wait()
- {
- // wait for previus draw to complete
- int count = 0;
- DWORD scanLines = 0;
- DWORD screenLines = _screenRect.bottom -1; // scanlines start on 0
- DWORD screenLines75 = (screenLines*3)/4; // % of the screen is rendered
- HRESULT hr = DD_OK;
- if(_directDraw == NULL)
- {
- return;
- }
- hr =_directDraw->GetScanLine(&scanLines);
- while ( screenLines75 > scanLines && hr == DD_OK)
- {
- // _confCritSect->Leave();
- _screenEvent->Wait(10);
- // _confCritSect->Enter();
- if(_directDraw == NULL)
- {
- return;
- }
- hr = _directDraw->GetScanLine(&scanLines);
- }
- }
- */
-
-WebRtc_Word32 VideoRenderDirectDraw::StartRender()
-{
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
- return 0;
-}
-
-WebRtc_Word32 VideoRenderDirectDraw::StopRender()
-{
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
- return 0;
-}
-
-WebRtc_Word32 VideoRenderDirectDraw::ChangeWindow(void* window)
-{
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Not supported.");
- return -1;
-}
-
-} //namespace webrtc
-
diff --git a/trunk/src/modules/video_render/main/source/windows/video_render_directdraw.h b/trunk/src/modules/video_render/main/source/windows/video_render_directdraw.h
deleted file mode 100644
index e790661..0000000
--- a/trunk/src/modules/video_render/main/source/windows/video_render_directdraw.h
+++ /dev/null
@@ -1,399 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_DIRECTDRAW_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_DIRECTDRAW_H_
-
-#include "typedefs.h"
-#include "i_video_render_win.h"
-#include "common_video/libyuv/include/libyuv.h"
-
-#include "ddraw.h"
-#include <Map>
-#include <List>
-
-// Added
-#include "video_render_defines.h"
-
-#pragma comment(lib, "ddraw.lib") // located in DirectX SDK
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-class ThreadWrapper;
-class Trace;
-
-class VideoRenderDirectDraw;
-
-// some typedefs to make it easy to test different versions
-typedef IDirectDraw7 DirectDraw;
-typedef IDirectDrawSurface7 DirectDrawSurface;
-typedef DDSURFACEDESC2 DirectDrawSurfaceDesc;
-typedef DDSCAPS2 DirectDrawCaps;
-typedef std::pair<int, unsigned int> ZorderPair;
-
-class WindowsThreadCpuUsage
-{
-public:
- WindowsThreadCpuUsage();
- int GetCpuUsage(); //in % since last call
- DWORD LastGetCpuTime()
- {
- return _lastGetCpuUsageTime;
- }
- const enum
- {
- CPU_CHECK_INTERVAL = 1000
- };
-private:
- _int64 _lastCpuUsageTime;
- DWORD _lastGetCpuUsageTime;
- int _lastCpuUsage;
- HANDLE _hThread;
- int _cores;
-};
-
-class DirectDrawStreamSettings
-{
-public:
- DirectDrawStreamSettings();
-
- float _startWidth;
- float _stopWidth;
- float _startHeight;
- float _stopHeight;
-
- float _cropStartWidth;
- float _cropStopWidth;
- float _cropStartHeight;
- float _cropStopHeight;
-};
-
-class DirectDrawBitmapSettings
-{
-public:
- DirectDrawBitmapSettings();
- ~DirectDrawBitmapSettings();
-
- int SetBitmap(Trace* trace, DirectDraw* directDraw);
-
- HBITMAP _transparentBitMap;
- float _transparentBitmapLeft;
- float _transparentBitmapRight;
- float _transparentBitmapTop;
- float _transparentBitmapBottom;
- int _transparentBitmapWidth;
- int _transparentBitmapHeight;
- DDCOLORKEY* _transparentBitmapColorKey;
- DirectDrawSurface* _transparentBitmapSurface; // size of bitmap image
-};
-
-class DirectDrawTextSettings
-{
-public:
- DirectDrawTextSettings();
- ~DirectDrawTextSettings();
-
- int SetText(const char* text, int textLength, COLORREF colorText,
- COLORREF colorBg, float left, float top, float right,
- float bottom);
-
- char* _ptrText;
- WebRtc_UWord32 _textLength;
- COLORREF _colorRefText;
- COLORREF _colorRefBackground;
- float _textLeft;
- float _textRight;
- float _textTop;
- float _textBottom;
- bool _transparent;
-};
-
-class DirectDrawChannel: public VideoRenderCallback
-{
-public:
- DirectDrawChannel(DirectDraw* directDraw,
- VideoType blitVideoType,
- VideoType incomingVideoType,
- VideoType screenVideoType,
- VideoRenderDirectDraw* owner);
-
- int FrameSizeChange(int width, int height, int numberOfStreams);
- int DeliverFrame(unsigned char* buffer, int buffeSize,
- unsigned int timeStamp90KHz);
- virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId,
- VideoFrame& videoFrame);
-
- int ChangeDeliverColorFormat(bool useScreenType);
-
- void AddRef();
- void Release();
-
- void SetStreamSettings(VideoRenderDirectDraw* DDObj, short streamId,
- float startWidth, float startHeight,
- float stopWidth, float stopHeight);
- void SetStreamCropSettings(VideoRenderDirectDraw* DDObj,
- short streamId, float startWidth,
- float startHeight, float stopWidth,
- float stopHeight);
-
- int GetStreamSettings(VideoRenderDirectDraw* DDObj, short streamId,
- float& startWidth, float& startHeight,
- float& stopWidth, float& stopHeight);
-
- void GetLargestSize(RECT* mixingRect);
- int
- BlitFromOffscreenBufferToMixingBuffer(
- VideoRenderDirectDraw* DDObj,
- short streamID,
- DirectDrawSurface* mixingSurface,
- RECT &dstRect, bool demuxing);
- bool IsOffScreenSurfaceUpdated(VideoRenderDirectDraw* DDobj);
-
-protected:
- virtual ~DirectDrawChannel();
-
-private:
- CriticalSectionWrapper* _critSect; // protect members from change while using them
- int _refCount;
- int _width;
- int _height;
- int _numberOfStreams;
- bool _deliverInScreenType;
- bool _doubleBuffer;
- DirectDraw* _directDraw;
- DirectDrawSurface* _offScreenSurface; // size of incoming stream
- DirectDrawSurface* _offScreenSurfaceNext; // size of incoming stream
- VideoType _blitVideoType;
- VideoType _originalBlitVideoType;
- VideoType _incomingVideoType;
- VideoType _screenVideoType;
- enum
- {
- MAX_FRAMEDELIVER_TIME = 20
- }; //Maximum time it might take to deliver a frame (process time in DeliverFrame)
- enum
- {
- MAX_NO_OF_LATE_FRAMEDELIVER_TIME = 10
- }; //No of times we allow DeliverFrame process time to exceed MAX_FRAMEDELIVER_TIME before we take action.
- VideoFrame _tempRenderBuffer;
-
- std::map<unsigned long long, DirectDrawStreamSettings*>
- _streamIdToSettings;
- bool _offScreenSurfaceUpdated;
- VideoRenderDirectDraw* _owner;
-};
-
-class VideoRenderDirectDraw: IVideoRenderWin
-{
-public:
- VideoRenderDirectDraw(Trace* trace, HWND hWnd, bool fullscreen);
- ~VideoRenderDirectDraw();
-public:
- //IVideoRenderWin
-
- /**************************************************************************
- *
- * Init
- *
- ***************************************************************************/
- virtual WebRtc_Word32 Init();
-
- /**************************************************************************
- *
- * Incoming Streams
- *
- ***************************************************************************/
- virtual VideoRenderCallback
- * CreateChannel(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder, const float left,
- const float top, const float right,
- const float bottom);
-
- virtual WebRtc_Word32 DeleteChannel(const WebRtc_UWord32 streamId);
-
- virtual WebRtc_Word32 GetStreamSettings(const WebRtc_UWord32 channel,
- const WebRtc_UWord16 streamId,
- WebRtc_UWord32& zOrder,
- float& left, float& top,
- float& right, float& bottom);
-
- /**************************************************************************
- *
- * Start/Stop
- *
- ***************************************************************************/
-
- virtual WebRtc_Word32 StartRender();
- virtual WebRtc_Word32 StopRender();
-
- /**************************************************************************
- *
- * Properties
- *
- ***************************************************************************/
-
- virtual bool IsFullScreen();
-
- virtual WebRtc_Word32 SetCropping(const WebRtc_UWord32 channel,
- const WebRtc_UWord16 streamId,
- const float left, const float top,
- const float right, const float bottom);
-
- virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
-
- virtual WebRtc_Word32 ChangeWindow(void* window);
-
- virtual WebRtc_Word32 GetGraphicsMemory(WebRtc_UWord64& totalMemory,
- WebRtc_UWord64& availableMemory);
-
- virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 colorText,
- const WebRtc_UWord32 colorBg,
- const float left, const float top,
- const float rigth, const float bottom);
-
- virtual WebRtc_Word32 SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey, const float left,
- const float top, const float right,
- const float bottom);
-
- virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 channel,
- const WebRtc_UWord16 streamId,
- const unsigned int zOrder,
- const float left, const float top,
- const float right,
- const float bottom);
-public:
-
- // Used for emergency stops...
- int Stop();
-
- DirectDrawChannel* ShareDirectDrawChannel(int channel);
- DirectDrawChannel* ConfigureDirectDrawChannel(int channel,
- unsigned char streamID,
- int zOrder, float left,
- float top, float right,
- float bottom);
-
- int AddDirectDrawChannel(int channel, unsigned char streamID, int zOrder,
- DirectDrawChannel*);
-
- VideoType GetPerferedVideoFormat();
- bool HasChannels();
- bool HasChannel(int channel);
- bool DeliverInScreenType();
- int GetChannels(std::list<int>& channelList);
-
- // code for getting graphics settings
- int GetScreenResolution(int& screenWidth, int& screenHeight);
- int UpdateSystemCPUUsage(int systemCPU);
-
- int SetBitmap(HBITMAP bitMap, unsigned char pictureId,
- DDCOLORKEY* colorKey, float left, float top, float rigth,
- float bottom);
-
- bool IsPrimaryOrMixingSurfaceOnSystem();
- bool CanBltFourCC()
- {
- return _bCanBltFourcc;
- }
-
-protected:
- static bool RemoteRenderingThreadProc(void* obj);
- bool RemoteRenderingProcess();
-
-private:
- int CheckCapabilities();
- int CreateMixingSurface();
- int CreatePrimarySurface();
-
- int FillSurface(DirectDrawSurface *pDDSurface, RECT* rect);
- int DrawOnSurface(unsigned char* buffer, int buffeSize);
- int BlitFromOffscreenBuffersToMixingBuffer();
- int BlitFromBitmapBuffersToMixingBuffer();
- int BlitFromTextToMixingBuffer();
-
- bool HasHWNDChanged();
- void DecideBestRenderingMode(bool hwndChanged, int totalRenderTime);
-
- // in fullscreen flip mode
- int WaitAndFlip(int& waitTime);
- int BlitFromMixingBufferToBackBuffer();
-
- // in normal window mode
- int BlitFromMixingBufferToFrontBuffer(bool hwndChanged, int& waitTime);
-
- // private members
- Trace* _trace;
- CriticalSectionWrapper* _confCritSect; // protect members from change while using them
-
- bool _fullscreen;
- bool _demuxing;
- bool _transparentBackground;
- bool _supportTransparency;
- bool _canStretch;
- bool _canMirrorLeftRight;
- bool _clearMixingSurface;
- bool _deliverInScreenType;
- bool _renderModeWaitForCorrectScanLine;
- bool _deliverInHalfFrameRate;
- bool _deliverInQuarterFrameRate;
- bool _bCanBltFourcc;
- bool _frameChanged; // True if a frame has changed or bitmap or text has changed.
- int _processCount;
- HWND _hWnd;
- RECT _screenRect; // whole screen as a rect
- RECT _mixingRect;
- RECT _originalHwndRect;
- RECT _hwndRect;
-
- VideoType _incomingVideoType;
- VideoType _blitVideoType;
- VideoType _rgbVideoType;
-
- DirectDraw* _directDraw;
- DirectDrawSurface* _primarySurface; // size of screen
- DirectDrawSurface* _backSurface; // size of screen
- DirectDrawSurface* _mixingSurface; // size of screen
-
- std::map<unsigned char, DirectDrawBitmapSettings*> _bitmapSettings;
- std::map<unsigned char, DirectDrawTextSettings*> _textSettings;
- std::map<int, DirectDrawChannel*> _directDrawChannels;
- std::multimap<int, unsigned int> _directDrawZorder;
-
- EventWrapper* _fullScreenWaitEvent;
- EventWrapper* _screenEvent;
- ThreadWrapper* _screenRenderThread;
- WindowsThreadCpuUsage _screenRenderCpuUsage;
-
- int _lastRenderModeCpuUsage;
-
- // Used for emergency stop caused by OnDisplayChange
- bool _blit;
-
- //code for providing graphics settings
- DWORD _totalMemory;
- DWORD _availableMemory;
- int _systemCPUUsage;
-
- // Variables used for checking render time
- int _maxAllowedRenderTime;
- int _nrOfTooLongRenderTimes;
- bool _isPrimaryOrMixingSurfaceOnSystem;
-};
-
-} //namespace webrtc
-
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_DIRECTDRAW_H_
diff --git a/trunk/src/modules/video_render/main/source/windows/video_render_windows_impl.cc b/trunk/src/modules/video_render/main/source/windows/video_render_windows_impl.cc
deleted file mode 100644
index 6f2237f..0000000
--- a/trunk/src/modules/video_render/main/source/windows/video_render_windows_impl.cc
+++ /dev/null
@@ -1,989 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "engine_configurations.h"
-#include "video_render_windows_impl.h"
-
-#include "critical_section_wrapper.h"
-#include "trace.h"
-#ifdef DIRECTDRAW_RENDERING
-#include "video_render_directdraw.h"
-#endif
-#ifdef DIRECT3D9_RENDERING
-#include "video_render_direct3d9.h"
-#endif
-
-#include <tchar.h>
-
-namespace webrtc {
-
-VideoRenderWindowsImpl::VideoRenderWindowsImpl(
- const WebRtc_Word32 id,
- const VideoRenderType videoRenderType,
- void* window,
- const bool fullscreen) :
- _id(id),
- _renderWindowsCritsect(
- *CriticalSectionWrapper::CreateCriticalSection()),
- _prtWindow(window), _fullscreen(fullscreen), _ptrRendererWin(NULL)
-{
-}
-
-VideoRenderWindowsImpl::~VideoRenderWindowsImpl()
-{
- delete &_renderWindowsCritsect;
- if (_ptrRendererWin)
- {
- delete _ptrRendererWin;
- _ptrRendererWin = NULL;
- }
-}
-
-WebRtc_Word32 VideoRenderWindowsImpl::Init()
-{
- //LogOSAndHardwareDetails();
- CheckHWAcceleration();
-
- _renderMethod = kVideoRenderWinD3D9;
-
- // Create the win renderer
- switch (_renderMethod)
- {
- case kVideoRenderWinDd:
- {
-#ifdef DIRECTDRAW_RENDERING
- VideoRenderDirectDraw* ptrRenderer;
- ptrRenderer = new VideoRenderDirectDraw(NULL, (HWND) _prtWindow, _fullscreen);
- if (ptrRenderer == NULL)
- {
- break;
- }
- _ptrRendererWin = reinterpret_cast<IVideoRenderWin*>(ptrRenderer);
-#else
- return NULL;
-#endif //DIRECTDRAW_RENDERING
- }
- break;
- case kVideoRenderWinD3D9:
- {
-#ifdef DIRECT3D9_RENDERING
- VideoRenderDirect3D9* ptrRenderer;
- ptrRenderer = new VideoRenderDirect3D9(NULL, (HWND) _prtWindow, _fullscreen);
- if (ptrRenderer == NULL)
- {
- break;
- }
- _ptrRendererWin = reinterpret_cast<IVideoRenderWin*>(ptrRenderer);
-#else
- return NULL;
-#endif //DIRECT3D9_RENDERING
- }
- break;
- default:
- break;
- }
-
- //Init renderer
- if (_ptrRendererWin)
- return _ptrRendererWin->Init();
- else
- return -1;
-}
-
-WebRtc_Word32 VideoRenderWindowsImpl::ChangeUniqueId(const WebRtc_Word32 id)
-{
- CriticalSectionScoped cs(_renderWindowsCritsect);
- _id = id;
- return 0;
-}
-
-WebRtc_Word32 VideoRenderWindowsImpl::ChangeWindow(void* window)
-{
- CriticalSectionScoped cs(_renderWindowsCritsect);
- if (!_ptrRendererWin)
- {
- return -1;
- }
- else
- {
- return _ptrRendererWin->ChangeWindow(window);
- }
-}
-
-VideoRenderCallback*
-VideoRenderWindowsImpl::AddIncomingRenderStream(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_renderWindowsCritsect);
- VideoRenderCallback* renderCallback = NULL;
-
- if (!_ptrRendererWin)
- {
- }
- else
- {
- renderCallback = _ptrRendererWin->CreateChannel(streamId, zOrder, left,
- top, right, bottom);
- }
-
- return renderCallback;
-}
-
-WebRtc_Word32 VideoRenderWindowsImpl::DeleteIncomingRenderStream(
- const WebRtc_UWord32 streamId)
-{
- CriticalSectionScoped cs(_renderWindowsCritsect);
- WebRtc_Word32 error = -1;
- if (!_ptrRendererWin)
- {
- }
- else
- {
- error = _ptrRendererWin->DeleteChannel(streamId);
- }
- return error;
-}
-
-WebRtc_Word32 VideoRenderWindowsImpl::GetIncomingRenderStreamProperties(
- const WebRtc_UWord32 streamId,
- WebRtc_UWord32& zOrder,
- float& left,
- float& top,
- float& right,
- float& bottom) const
-{
- CriticalSectionScoped cs(_renderWindowsCritsect);
- zOrder = 0;
- left = 0;
- top = 0;
- right = 0;
- bottom = 0;
-
- WebRtc_Word32 error = -1;
- if (!_ptrRendererWin)
- {
- }
- else
- {
- error = _ptrRendererWin->GetStreamSettings(streamId, 0, zOrder, left,
- top, right, bottom);
- }
- return error;
-}
-
-WebRtc_Word32 VideoRenderWindowsImpl::StartRender()
-{
- CriticalSectionScoped cs(_renderWindowsCritsect);
- WebRtc_Word32 error = -1;
- if (!_ptrRendererWin)
- {
- }
- else
- {
- error = _ptrRendererWin->StartRender();
- }
- return error;
-}
-
-WebRtc_Word32 VideoRenderWindowsImpl::StopRender()
-{
- CriticalSectionScoped cs(_renderWindowsCritsect);
- WebRtc_Word32 error = -1;
- if (!_ptrRendererWin)
- {
- }
- else
- {
- error = _ptrRendererWin->StopRender();
- }
- return error;
-}
-
-VideoRenderType VideoRenderWindowsImpl::RenderType()
-{
- return kRenderWindows;
-}
-
-RawVideoType VideoRenderWindowsImpl::PerferedVideoType()
-{
- return kVideoI420;
-}
-
-bool VideoRenderWindowsImpl::FullScreen()
-{
- CriticalSectionScoped cs(_renderWindowsCritsect);
- bool fullscreen = false;
- if (!_ptrRendererWin)
- {
- }
- else
- {
- fullscreen = _ptrRendererWin->IsFullScreen();
- }
- return fullscreen;
-}
-
-WebRtc_Word32 VideoRenderWindowsImpl::GetGraphicsMemory(
- WebRtc_UWord64& totalGraphicsMemory,
- WebRtc_UWord64& availableGraphicsMemory) const
-{
- if (_ptrRendererWin)
- {
- return _ptrRendererWin->GetGraphicsMemory(totalGraphicsMemory,
- availableGraphicsMemory);
- }
-
- totalGraphicsMemory = 0;
- availableGraphicsMemory = 0;
- return -1;
-}
-
-WebRtc_Word32 VideoRenderWindowsImpl::GetScreenResolution(
- WebRtc_UWord32& screenWidth,
- WebRtc_UWord32& screenHeight) const
-{
- CriticalSectionScoped cs(_renderWindowsCritsect);
- screenWidth = 0;
- screenHeight = 0;
- return 0;
-}
-
-WebRtc_UWord32 VideoRenderWindowsImpl::RenderFrameRate(
- const WebRtc_UWord32 streamId)
-{
- CriticalSectionScoped cs(_renderWindowsCritsect);
- return 0;
-}
-
-WebRtc_Word32 VideoRenderWindowsImpl::SetStreamCropping(
- const WebRtc_UWord32 streamId,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_renderWindowsCritsect);
- WebRtc_Word32 error = -1;
- if (!_ptrRendererWin)
- {
- }
- else
- {
- error = _ptrRendererWin->SetCropping(streamId, 0, left, top, right,
- bottom);
- }
- return error;
-}
-
-WebRtc_Word32 VideoRenderWindowsImpl::ConfigureRenderer(
- const WebRtc_UWord32 streamId,
- const unsigned int zOrder,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_renderWindowsCritsect);
- WebRtc_Word32 error = -1;
- if (!_ptrRendererWin)
- {
- }
- else
- {
- error = _ptrRendererWin->ConfigureRenderer(streamId, 0, zOrder, left,
- top, right, bottom);
- }
-
- return error;
-}
-
-WebRtc_Word32 VideoRenderWindowsImpl::SetTransparentBackground(
- const bool enable)
-{
- CriticalSectionScoped cs(_renderWindowsCritsect);
- WebRtc_Word32 error = -1;
- if (!_ptrRendererWin)
- {
- }
- else
- {
- error = _ptrRendererWin->SetTransparentBackground(enable);
- }
- return error;
-}
-
-WebRtc_Word32 VideoRenderWindowsImpl::SetText(
- const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 textColorRef,
- const WebRtc_UWord32 backgroundColorRef,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_renderWindowsCritsect);
- WebRtc_Word32 error = -1;
- if (!_ptrRendererWin)
- {
- }
- else
- {
- error = _ptrRendererWin->SetText(textId, text, textLength,
- textColorRef, backgroundColorRef,
- left, top, right, bottom);
- }
- return error;
-}
-
-WebRtc_Word32 VideoRenderWindowsImpl::SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey,
- const float left,
- const float top,
- const float right,
- const float bottom)
-{
- CriticalSectionScoped cs(_renderWindowsCritsect);
- WebRtc_Word32 error = -1;
- if (!_ptrRendererWin)
- {
- }
- else
- {
- error = _ptrRendererWin->SetBitmap(bitMap, pictureId, colorKey, left,
- top, right, bottom);
- }
- return error;
-}
-
-void VideoRenderWindowsImpl::LogOSAndHardwareDetails()
-{
- HRESULT hr;
- IDxDiagProvider* m_pDxDiagProvider = NULL;
- IDxDiagContainer* m_pDxDiagRoot = NULL;
-
- hr = CoInitializeEx(NULL, COINIT_MULTITHREADED);
- bool coUninitializeIsRequired = true;
- if (FAILED(hr))
- {
- // Avoid calling CoUninitialize() since CoInitializeEx() failed.
- coUninitializeIsRequired = false;
- if (hr == RPC_E_CHANGED_MODE)
- {
- // Calling thread has already initialized COM to be used in a single-threaded
- // apartment (STA). We are then prevented from using STA.
- // Details: hr = 0x80010106 <=> "Cannot change thread mode after it is set".
- //
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVideoRenderer,
- _id,
- "VideoRenderWindowsImpl::LogOSAndHardwareDetails() CoInitializeEx(NULL, COINIT_APARTMENTTHREADED) => RPC_E_CHANGED_MODE, error 0x%x",
- hr);
- }
- }
-
- hr = CoCreateInstance(CLSID_DxDiagProvider, NULL, CLSCTX_INPROC_SERVER,
- IID_IDxDiagProvider, (LPVOID*) &m_pDxDiagProvider);
-
- if (FAILED(hr) || m_pDxDiagProvider == NULL)
- {
- if (coUninitializeIsRequired)
- CoUninitialize();
- return;
- }
-
- // Fill out a DXDIAG_INIT_PARAMS struct and pass it to IDxDiagContainer::Initialize
- // Passing in TRUE for bAllowWHQLChecks, allows dxdiag to check if drivers are
- // digital signed as logo'd by WHQL which may connect via internet to update
- // WHQL certificates.
- DXDIAG_INIT_PARAMS dxDiagInitParam;
- ZeroMemory(&dxDiagInitParam, sizeof(DXDIAG_INIT_PARAMS));
-
- dxDiagInitParam.dwSize = sizeof(DXDIAG_INIT_PARAMS);
- dxDiagInitParam.dwDxDiagHeaderVersion = DXDIAG_DX9_SDK_VERSION;
- dxDiagInitParam.bAllowWHQLChecks = TRUE;
- dxDiagInitParam.pReserved = NULL;
-
- hr = m_pDxDiagProvider->Initialize(&dxDiagInitParam);
- if (FAILED(hr))
- {
- m_pDxDiagProvider->Release();
- if (coUninitializeIsRequired)
- CoUninitialize();
- return;
- }
-
- hr = m_pDxDiagProvider->GetRootContainer(&m_pDxDiagRoot);
- if (FAILED(hr) || m_pDxDiagRoot == NULL)
- {
- m_pDxDiagProvider->Release();
- if (coUninitializeIsRequired)
- CoUninitialize();
- return;
- }
-
- IDxDiagContainer* pObject = NULL;
-
- hr = m_pDxDiagRoot->GetChildContainer(L"DxDiag_SystemInfo", &pObject);
- if (FAILED(hr) || pObject == NULL)
- {
- m_pDxDiagRoot->Release();
- m_pDxDiagProvider->Release();
- if (coUninitializeIsRequired)
- CoUninitialize();
- return;
- }
-
- TCHAR m_szDirectXVersionLongEnglish[100];
- TCHAR m_szOSLocalized[100];
- TCHAR m_szProcessorEnglish[200];
- TCHAR m_szSystemManufacturerEnglish[200];
-
- ZeroMemory(m_szDirectXVersionLongEnglish, sizeof(TCHAR) * 100);
- ZeroMemory(m_szOSLocalized, sizeof(TCHAR) * 100);
- ZeroMemory(m_szProcessorEnglish, sizeof(TCHAR) * 200);
- ZeroMemory(m_szSystemManufacturerEnglish, sizeof(TCHAR) * 200);
-
- GetStringValue( pObject, L"szDirectXVersionLongEnglish",
- EXPAND(m_szDirectXVersionLongEnglish) );
- GetStringValue(pObject, L"szOSLocalized", EXPAND(m_szOSLocalized) );
- GetStringValue(pObject, L"szProcessorEnglish", EXPAND(m_szProcessorEnglish) );
- GetStringValue( pObject, L"szSystemManufacturerEnglish",
- EXPAND(m_szSystemManufacturerEnglish) );
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
- "System Manufacturer --- %s",
- m_szSystemManufacturerEnglish);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
- "Processor --- %s", m_szProcessorEnglish);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
- "Operating System --- %s", m_szOSLocalized);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
- "DirectX Version --- %s",
- m_szDirectXVersionLongEnglish);
-
- if (pObject)
- pObject->Release();
-
- struct DisplayInfo
- {
- TCHAR m_szDescription[200];
- TCHAR m_szManufacturer[200];
- TCHAR m_szChipType[100];
- TCHAR m_szDisplayMemoryEnglish[100];
- TCHAR m_szDisplayModeEnglish[100];
- TCHAR m_szDriverName[100];
- TCHAR m_szDriverVersion[100];
- TCHAR m_szDDStatusEnglish[100];
- TCHAR m_szD3DStatusEnglish[100];
- BOOL m_bDDAccelerationEnabled;
- BOOL m_bNoHardware;
- BOOL m_b3DAccelerationExists;
- BOOL m_b3DAccelerationEnabled;
- };
-
- WCHAR wszContainer[256];
- IDxDiagContainer* pContainer = NULL;
-
- DWORD nInstanceCount = 0;
- DWORD nItem = 0;
- DWORD nCurCount = 0;
-
- // Get the IDxDiagContainer object called "DxDiag_DisplayDevices".
- // This call may take some time while dxdiag gathers the info.
- if (FAILED(hr = m_pDxDiagRoot->GetChildContainer(L"DxDiag_DisplayDevices",
- &pContainer)))
- {
- m_pDxDiagRoot->Release();
- m_pDxDiagProvider->Release();
- if (coUninitializeIsRequired)
- CoUninitialize();
- return;
- }
-
- if (FAILED(hr = pContainer->GetNumberOfChildContainers(&nInstanceCount)))
- {
- pContainer->Release();
- m_pDxDiagRoot->Release();
- m_pDxDiagProvider->Release();
- if (coUninitializeIsRequired)
- CoUninitialize();
- return;
- }
-
- DisplayInfo *pDisplayInfo = new DisplayInfo;
- if (pDisplayInfo == NULL)
- return;
- ZeroMemory(pDisplayInfo, sizeof(DisplayInfo));
-
- hr = pContainer->EnumChildContainerNames(nItem, wszContainer, 256);
- if (FAILED(hr))
- {
- delete pDisplayInfo;
- pContainer->Release();
- m_pDxDiagRoot->Release();
- m_pDxDiagProvider->Release();
- if (coUninitializeIsRequired)
- CoUninitialize();
- return;
- }
-
- hr = pContainer->GetChildContainer(wszContainer, &pObject);
- if (FAILED(hr) || pObject == NULL)
- {
- delete pDisplayInfo;
- pContainer->Release();
- m_pDxDiagRoot->Release();
- m_pDxDiagProvider->Release();
- if (coUninitializeIsRequired)
- CoUninitialize();
- return;
- }
-
- GetStringValue( pObject, L"szDescription",
- EXPAND(pDisplayInfo->m_szDescription) );
- GetStringValue( pObject, L"szManufacturer",
- EXPAND(pDisplayInfo->m_szManufacturer) );
- GetStringValue(pObject, L"szChipType", EXPAND(pDisplayInfo->m_szChipType) );
- GetStringValue( pObject, L"szDisplayMemoryEnglish",
- EXPAND(pDisplayInfo->m_szDisplayMemoryEnglish) );
- GetStringValue( pObject, L"szDisplayModeEnglish",
- EXPAND(pDisplayInfo->m_szDisplayModeEnglish) );
- GetStringValue( pObject, L"szDriverName",
- EXPAND(pDisplayInfo->m_szDriverName) );
- GetStringValue( pObject, L"szDriverVersion",
- EXPAND(pDisplayInfo->m_szDriverVersion) );
- GetBoolValue(pObject, L"bDDAccelerationEnabled",
- &pDisplayInfo->m_bDDAccelerationEnabled);
- GetBoolValue(pObject, L"bNoHardware", &pDisplayInfo->m_bNoHardware);
- GetBoolValue(pObject, L"bDDAccelerationEnabled",
- &pDisplayInfo->m_bDDAccelerationEnabled);
- GetBoolValue(pObject, L"b3DAccelerationExists",
- &pDisplayInfo->m_b3DAccelerationExists);
- GetBoolValue(pObject, L"b3DAccelerationEnabled",
- &pDisplayInfo->m_b3DAccelerationEnabled);
- GetStringValue( pObject, L"szDDStatusEnglish",
- EXPAND(pDisplayInfo->m_szDDStatusEnglish));
- GetStringValue( pObject, L"szD3DStatusEnglish",
- EXPAND(pDisplayInfo->m_szD3DStatusEnglish));
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
- "Device Name --- %s",
- pDisplayInfo->m_szDescription);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
- "Device Manufacturer --- %s",
- pDisplayInfo->m_szManufacturer);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
- "Device ChipType --- %s",
- pDisplayInfo->m_szChipType);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
- "Approx. Total Device Memory --- %s",
- pDisplayInfo->m_szDisplayMemoryEnglish);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
- "Current Display Mode --- %s",
- pDisplayInfo->m_szDisplayModeEnglish);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
- "Device Driver Name --- %s",
- pDisplayInfo->m_szDriverName);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
- "Device Driver Version --- %s",
- pDisplayInfo->m_szDriverVersion);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
- "DirectDraw Acceleration Enabled --- %s",
- pDisplayInfo->m_szDescription ? "Enabled" : "Disabled");
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
- "bNoHardware --- %s",
- pDisplayInfo->m_bNoHardware ? "Enabled" : "Disabled");
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
- "b3DAccelerationExists Enabled --- %s",
- pDisplayInfo->m_b3DAccelerationExists ? "Enabled" : "Disabled");
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
- "b3DAccelerationEnabled Enabled --- %s",
- pDisplayInfo->m_b3DAccelerationEnabled ? "Enabled"
- : "Disabled");
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
- "DDraw Status --- %s",
- pDisplayInfo->m_szDDStatusEnglish);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, -1,
- "D3D Status --- %s",
- pDisplayInfo->m_szD3DStatusEnglish);
-
- // Get OS version
- OSVERSIONINFOEX osvie;
- osvie.dwOSVersionInfoSize = sizeof(OSVERSIONINFOEX);
- GetVersionEx((LPOSVERSIONINFO) & osvie);
- /*
- Operating system Version number dwMajorVersion dwMinorVersion
- Windows 7 6.1 6 1
- Windows Server 2008 R2 6.1 6 1
- Windows Server 2008 6.0 6 0
- Windows Vista 6.0 6 0
- Windows Server 2003 R2 5.2 5 2
- Windows Server 2003 5.2 5 2
- Windows XP 5.1 5 1
- Windows 2000 5.0 5 0
- */
- //RDP problem exists only when XP is involved
- if (osvie.dwMajorVersion < 6)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideoRenderer, _id,
- "Checking for RDP driver");
- if (_tcsncmp(pDisplayInfo->m_szDriverName, _T("RDPDD.dll"), 9) == 0)
- {
- //
- }
- }
-
- if (pObject)
- {
- pObject->Release();
- pObject = NULL;
- }
-
- if (pContainer)
- pContainer->Release();
-
- if (m_pDxDiagProvider)
- m_pDxDiagProvider->Release();
-
- if (m_pDxDiagRoot)
- m_pDxDiagRoot->Release();
-
- if (pDisplayInfo)
- delete pDisplayInfo;
-
- if (coUninitializeIsRequired)
- CoUninitialize();
-
- return;
-}
-
-//-----------------------------------------------------------------------------
-// Name: GetStringValue()
-// Desc: Get a string value from a IDxDiagContainer object
-//-----------------------------------------------------------------------------
-HRESULT VideoRenderWindowsImpl::GetStringValue(IDxDiagContainer* pObject,
- WCHAR* wstrName,
- TCHAR* strValue, int nStrLen)
-{
- HRESULT hr;
- VARIANT var;
- VariantInit(&var);
-
- if (FAILED(hr = pObject->GetProp(wstrName, &var)))
- return hr;
-
- if (var.vt != VT_BSTR)
- return E_INVALIDARG;
-
-#ifdef _UNICODE
- wcsncpy( strValue, var.bstrVal, nStrLen-1 );
-#else
- wcstombs(strValue, var.bstrVal, nStrLen);
-#endif
- strValue[nStrLen - 1] = TEXT('\0');
- VariantClear(&var);
-
- return S_OK;
-}
-
-//-----------------------------------------------------------------------------
-// Name: GetBoolValue()
-// Desc: Get a BOOL value from a IDxDiagContainer object
-//-----------------------------------------------------------------------------
-HRESULT VideoRenderWindowsImpl::GetBoolValue(IDxDiagContainer* pObject,
- WCHAR* wstrName, BOOL* pbValue)
-{
- HRESULT hr;
- VARIANT var;
- VariantInit(&var);
-
- if (FAILED(hr = pObject->GetProp(wstrName, &var)))
- return hr;
-
- if (var.vt != VT_BOOL)
- return E_INVALIDARG;
-
- *pbValue = (var.boolVal != 0);
- VariantClear(&var);
-
- return S_OK;
-}
-
-int VideoRenderWindowsImpl::CheckHWAcceleration()
-{
- // Read the registry to check if HW acceleration is enabled or not.
- HKEY regKey;
- DWORD value = 0;
- DWORD valueLength = 4;
-
- bool directDraw = true;
- bool direct3D = true;
- bool dci = true;
-
- // DirectDraw
- if (RegOpenKeyEx(HKEY_LOCAL_MACHINE, _T("SOFTWARE\\Microsoft\\DirectDraw"),
- 0, KEY_QUERY_VALUE, ®Key) == ERROR_SUCCESS)
- {
- // We have the registry key
- value = 0;
- if (RegQueryValueEx(regKey, _T("EmulationOnly"), NULL, NULL,
- (BYTE*) &value, &valueLength) == ERROR_SUCCESS)
- {
- if (value == 1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
- "DirectDraw acceleration is disabled");
- directDraw = false;
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
- "DirectDraw acceleration is enabled");
- }
- }
- else
- {
- // Could not get the value for this one.
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "Could not find EmulationOnly key, DirectDraw acceleration is probably enabled");
- }
- RegCloseKey(regKey);
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "Could not open DirectDraw settings");
- }
-
- // Direct3D
- if (RegOpenKeyEx(HKEY_LOCAL_MACHINE,
- _T("SOFTWARE\\Microsoft\\Direct3D\\Drivers"), 0,
- KEY_QUERY_VALUE, ®Key) == ERROR_SUCCESS)
- {
- // We have the registry key
- value = 0;
- if (RegQueryValueEx(regKey, _T("SoftwareOnly"), NULL, NULL,
- (BYTE*) &value, &valueLength) == ERROR_SUCCESS)
- {
- if (value == 1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
- "Direct3D acceleration is disabled");
- direct3D = false;
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
- "Direct3D acceleration is enabled");
- }
- }
- else
- {
- // Could not get the value for this one.
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "Could not find SoftwarOnly key, Direct3D acceleration is probably enabled");
- }
- RegCloseKey(regKey);
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "Could not open Direct3D settings");
- }
-
- // DCI
- if (RegOpenKeyEx(
- HKEY_LOCAL_MACHINE,
- _T(
- "SYSTEM\\CurrentControlSet\\Control\\GraphicsDrivers\\DCI"),
- 0, KEY_QUERY_VALUE, ®Key) == ERROR_SUCCESS)
- {
- // We have found the registry key
- value = 0;
- if (RegQueryValueEx(regKey, _T("Timeout"), NULL, NULL, (BYTE*) &value,
- &valueLength) == ERROR_SUCCESS)
- {
- if (value == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
- "DCI - DirectDraw acceleration is disabled");
- dci = false;
- }
- else if (value == 7)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, -1,
- "DCI is fully enabled");
- }
- else
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVideo,
- -1,
- "DCI - DirectDraw acceleration is enabled, but short timeout: %d",
- value);
- }
- }
- else
- {
- // Could not get the value for this one.
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "Could not find Timeout key");
- }
- RegCloseKey(regKey);
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "Could not open DCI settings");
- }
-
- // We don't care about Direct3D right now...
- if (dci == false || directDraw == false)
- {
- return -1;
- }
-
- return 0;
-}
-
-void VideoRenderWindowsImpl::CheckHWDriver(bool& badDriver,
- bool& fullAccelerationEnabled)
-{
- // Read the registry to check if HW acceleration is enabled or not.
- HKEY regKey;
- DWORD value = 0;
- DWORD valueLength = 4;
-
- //Assume the best
- badDriver = false;
- fullAccelerationEnabled = true;
-
- // Check the path to the currently used driver
- if (RegOpenKeyEx(HKEY_LOCAL_MACHINE, _T("HARDWARE\\DEVICEMAP\\VIDEO"), 0,
- KEY_QUERY_VALUE, ®Key) == ERROR_SUCCESS)
- {
- // We have found the registry key containing the driver location
- value = 0;
- DWORD driverPathLen = 512;
- TCHAR driverPath[512];
- memset(driverPath, 0, driverPathLen * sizeof(TCHAR));
- DWORD dwType = REG_SZ;
-
- long retVal = RegQueryValueEx(regKey, _T("\\Device\\Video0"), NULL,
- NULL, (BYTE*) driverPath, &driverPathLen);
-
- // Close the key...
- RegCloseKey(regKey);
-
- if (retVal == ERROR_SUCCESS)
- {
- // We have the path to the currently used video card
-
- // trueDriverPath = modified nameStr, from above, that works
- // for RegOpenKeyEx
- TCHAR trueDriverPath[512];
- memset(trueDriverPath, 0, 512 * sizeof(TCHAR));
-
- // Convert the path to correct format.
- // - Remove \Registry\Machine\
- // - Replace '\' with '\\'
- // Should be something like this: System\\CurrentControlSet\\Control\\Video\\{F6987E15-F12C-4B15-8C84-0F635F3F09EA}\\0000"
- int idx = 0;
- for (DWORD i = 18; i < (driverPathLen / sizeof(TCHAR)); i++)
- {
- trueDriverPath[idx++] = driverPath[i];
- if (driverPath[i] == _T('\\'))
- {
- trueDriverPath[idx++] = driverPath[i];
- }
- }
-
- // Open the driver key
- if (RegOpenKeyEx(HKEY_LOCAL_MACHINE, trueDriverPath, 0,
- KEY_QUERY_VALUE, ®Key) == ERROR_SUCCESS)
- {
- TCHAR driverName[64];
- memset(driverName, 0, 64 * sizeof(TCHAR));
- DWORD driverNameLength = 64;
- retVal = RegQueryValueEx(regKey, _T("drv"), NULL, NULL,
- (BYTE*) driverName, &driverNameLength);
- if (retVal == ERROR_SUCCESS)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, -1,
- "Graphics card driver name: %s", driverName);
- }
- DWORD accLevel = 0;
- DWORD accLevelS = sizeof(accLevel);
-
- RegQueryValueEx(regKey, _T("Acceleration.Level"), NULL, NULL,
- (LPBYTE) & accLevel, &accLevelS);
- //Don't care if the key is not found. It probably means that acceleration is enabled
- if (accLevel != 0)
- {
- // Close the key...
- RegCloseKey(regKey);
-
- if (RegOpenKeyEx(HKEY_LOCAL_MACHINE, trueDriverPath, 0,
- KEY_SET_VALUE, ®Key) == ERROR_SUCCESS)
- {
- // try setting it to full
- accLevel = 0;
- LONG retVal;
- retVal = RegSetValueEx(regKey,
- _T("Acceleration.Level"), NULL,
- REG_DWORD, (PBYTE) & accLevel,
- sizeof(DWORD));
- if (retVal != ERROR_SUCCESS)
- {
- fullAccelerationEnabled = false;
- }
- else
- {
- RegQueryValueEx(regKey, _T("Acceleration.Level"),
- NULL, NULL, (LPBYTE) & accLevel,
- &accLevelS);
- if (accLevel != 0)
- {
- fullAccelerationEnabled = false;
- }
- else
- {
- fullAccelerationEnabled = true;
- }
- }
- }
- else
- {
- fullAccelerationEnabled = false;
- }
- }
- else
- {
- fullAccelerationEnabled = true;
- }
-
- // Close the key...
- RegCloseKey(regKey);
- }
- }
- }
-}
-
-} //namespace webrtc
-
diff --git a/trunk/src/modules/video_render/main/source/windows/video_render_windows_impl.h b/trunk/src/modules/video_render/main/source/windows/video_render_windows_impl.h
deleted file mode 100644
index bdc0a0a..0000000
--- a/trunk/src/modules/video_render/main/source/windows/video_render_windows_impl.h
+++ /dev/null
@@ -1,155 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_WINDOWS_IMPL_H_
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_WINDOWS_IMPL_H_
-
-#include <Winerror.h>
-#include <dxdiag.h>
-
-#include "i_video_render.h"
-#include "i_video_render_win.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-#define EXPAND(x) x, sizeof(x)/sizeof(TCHAR)
-
-#pragma comment(lib, "dxguid.lib")
-
-enum VideoRenderWinMethod
-{
- kVideoRenderWinDd = 0, kVideoRenderWinD3D9 = 1
-};
-
-// Class definitions
-class VideoRenderWindowsImpl: IVideoRender
-{
-public:
- /*
- * Constructor/destructor
- */
-
- VideoRenderWindowsImpl(const WebRtc_Word32 id,
- const VideoRenderType videoRenderType,
- void* window, const bool fullscreen);
-
- virtual ~VideoRenderWindowsImpl();
-
- virtual WebRtc_Word32 Init();
-
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- virtual WebRtc_Word32 ChangeWindow(void* window);
-
- /**************************************************************************
- *
- * Incoming Streams
- *
- ***************************************************************************/
-
- virtual VideoRenderCallback
- * AddIncomingRenderStream(const WebRtc_UWord32 streamId,
- const WebRtc_UWord32 zOrder,
- const float left, const float top,
- const float right, const float bottom);
-
- virtual WebRtc_Word32
- DeleteIncomingRenderStream(const WebRtc_UWord32 streamId);
-
- virtual WebRtc_Word32
- GetIncomingRenderStreamProperties(const WebRtc_UWord32 streamId,
- WebRtc_UWord32& zOrder,
- float& left, float& top,
- float& right, float& bottom) const;
-
- /**************************************************************************
- *
- * Start/Stop
- *
- ***************************************************************************/
-
- virtual WebRtc_Word32 StartRender();
-
- virtual WebRtc_Word32 StopRender();
-
- /**************************************************************************
- *
- * Properties
- *
- ***************************************************************************/
-
- virtual VideoRenderType RenderType();
-
- virtual RawVideoType PerferedVideoType();
-
- virtual bool FullScreen();
-
- virtual WebRtc_Word32
- GetGraphicsMemory(WebRtc_UWord64& totalGraphicsMemory,
- WebRtc_UWord64& availableGraphicsMemory) const;
-
- virtual WebRtc_Word32
- GetScreenResolution(WebRtc_UWord32& screenWidth,
- WebRtc_UWord32& screenHeight) const;
-
- virtual WebRtc_UWord32 RenderFrameRate(const WebRtc_UWord32 streamId);
-
- virtual WebRtc_Word32 SetStreamCropping(const WebRtc_UWord32 streamId,
- const float left, const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 ConfigureRenderer(const WebRtc_UWord32 streamId,
- const unsigned int zOrder,
- const float left, const float top,
- const float right,
- const float bottom);
-
- virtual WebRtc_Word32 SetTransparentBackground(const bool enable);
-
- virtual WebRtc_Word32 SetText(const WebRtc_UWord8 textId,
- const WebRtc_UWord8* text,
- const WebRtc_Word32 textLength,
- const WebRtc_UWord32 textColorRef,
- const WebRtc_UWord32 backgroundColorRef,
- const float left, const float top,
- const float right, const float bottom);
-
- virtual WebRtc_Word32 SetBitmap(const void* bitMap,
- const WebRtc_UWord8 pictureId,
- const void* colorKey, const float left,
- const float top, const float right,
- const float bottom);
-
- static int CheckHWAcceleration();
- static void CheckHWDriver(bool& badDriver, bool& fullAccelerationEnabled);
-
-private:
-
- void LogOSAndHardwareDetails();
- HRESULT GetBoolValue(IDxDiagContainer* pObject, WCHAR* wstrName,
- BOOL* pbValue);
- HRESULT GetStringValue(IDxDiagContainer* pObject, WCHAR* wstrName,
- TCHAR* strValue, int nStrLen);
-
- WebRtc_Word32 _id;
- CriticalSectionWrapper& _renderWindowsCritsect;
-
- void* _prtWindow;
- bool _fullscreen;
-
- VideoRenderWinMethod _renderMethod;
- IVideoRenderWin* _ptrRendererWin;
-};
-
-} //namespace webrtc
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_SOURCE_WINDOWS_VIDEO_RENDER_WINDOWS_IMPL_H_
diff --git a/trunk/src/modules/video_render/main/test/testAPI/renderStartImage.bmp b/trunk/src/modules/video_render/main/test/testAPI/renderStartImage.bmp
deleted file mode 100644
index c443a58..0000000
--- a/trunk/src/modules/video_render/main/test/testAPI/renderStartImage.bmp
+++ /dev/null
Binary files differ
diff --git a/trunk/src/modules/video_render/main/test/testAPI/testAPI.cc b/trunk/src/modules/video_render/main/test/testAPI/testAPI.cc
deleted file mode 100644
index 607a866..0000000
--- a/trunk/src/modules/video_render/main/test/testAPI/testAPI.cc
+++ /dev/null
@@ -1,680 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testAPI.h"
-
-#include <stdio.h>
-
-#if defined(_WIN32)
-#include <tchar.h>
-#include <windows.h>
-#include <cassert>
-#include <fstream>
-#include <iostream>
-#include <string>
-#include <windows.h>
-#include <ddraw.h>
-
-#elif defined(WEBRTC_LINUX)
-
-#include <iostream>
-#include <X11/Xlib.h>
-#include <X11/Xutil.h>
-#include <sys/time.h>
-
-#endif
-
-#include "common_types.h"
-#include "process_thread.h"
-#include "module_common_types.h"
-#include "video_render_defines.h"
-#include "video_render.h"
-#include "tick_util.h"
-#include "trace.h"
-
-using namespace webrtc;
-
-void GetTestVideoFrame(WebRtc_UWord8* frame,
- WebRtc_Word32 width,
- WebRtc_Word32 height,
- WebRtc_UWord8 startColor);
-int TestSingleStream(VideoRender* renderModule);
-int TestFullscreenStream(VideoRender* &renderModule,
- void* window,
- const VideoRenderType videoRenderType);
-int TestBitmapText(VideoRender* renderModule);
-int TestMultipleStreams(VideoRender* renderModule);
-int TestExternalRender(VideoRender* renderModule);
-
-#define TEST_FRAME_RATE 30
-#define TEST_TIME_SECOND 5
-#define TEST_FRAME_NUM (TEST_FRAME_RATE*TEST_TIME_SECOND)
-#define TEST_STREAM0_START_COLOR 0
-#define TEST_STREAM1_START_COLOR 64
-#define TEST_STREAM2_START_COLOR 128
-#define TEST_STREAM3_START_COLOR 192
-
-#if defined(_WIN32) && defined(_DEBUG)
-// #include "vld.h"
-#define SLEEP(x) ::Sleep(x)
-#elif defined(WEBRTC_LINUX)
-
-#define GET_TIME_IN_MS timeGetTime()
-#define SLEEP(x) Sleep(x)
-
-void Sleep(unsigned long x)
-{
- timespec t;
- t.tv_sec = x/1000;
- t.tv_nsec = (x-(x/1000)*1000)*1000000;
- nanosleep(&t,NULL);
-}
-
-unsigned long timeGetTime()
-{
- struct timeval tv;
- struct timezone tz;
- unsigned long val;
-
- gettimeofday(&tv, &tz);
- val= tv.tv_sec*1000+ tv.tv_usec/1000;
- return(val);
-}
-
-#elif defined(WEBRTC_MAC_INTEL)
-
-#include <unistd.h>
-
-#define GET_TIME_IN_MS timeGetTime()
-#define SLEEP(x) usleep(x * 1000)
-
-unsigned long timeGetTime()
-{
- return 0;
-}
-
-#else
-
-#define GET_TIME_IN_MS ::timeGetTime()
-#define SLEEP(x) ::Sleep(x)
-
-#endif
-
-using namespace std;
-
-#if defined(_WIN32)
-LRESULT CALLBACK WebRtcWinProc( HWND hWnd,UINT uMsg,WPARAM wParam,LPARAM lParam)
-{
- switch(uMsg)
- {
- case WM_DESTROY:
- break;
- case WM_COMMAND:
- break;
- }
- return DefWindowProc(hWnd,uMsg,wParam,lParam);
-}
-
-int WebRtcCreateWindow(HWND &hwndMain,int winNum, int width, int height)
-{
- HINSTANCE hinst = GetModuleHandle(0);
- WNDCLASSEX wcx;
- wcx.hInstance = hinst;
- wcx.lpszClassName = TEXT("VideoRenderTest");
- wcx.lpfnWndProc = (WNDPROC)WebRtcWinProc;
- wcx.style = CS_DBLCLKS;
- wcx.hIcon = LoadIcon (NULL, IDI_APPLICATION);
- wcx.hIconSm = LoadIcon (NULL, IDI_APPLICATION);
- wcx.hCursor = LoadCursor (NULL, IDC_ARROW);
- wcx.lpszMenuName = NULL;
- wcx.cbSize = sizeof (WNDCLASSEX);
- wcx.cbClsExtra = 0;
- wcx.cbWndExtra = 0;
- wcx.hbrBackground = GetSysColorBrush(COLOR_3DFACE);
-
- // Register our window class with the operating system.
- // If there is an error, exit program.
- if ( !RegisterClassEx (&wcx) )
- {
- MessageBox( 0, TEXT("Failed to register window class!"),TEXT("Error!"), MB_OK|MB_ICONERROR );
- return 0;
- }
-
- // Create the main window.
- hwndMain = CreateWindowEx(
- 0, // no extended styles
- TEXT("VideoRenderTest"), // class name
- TEXT("VideoRenderTest Window"), // window name
- WS_OVERLAPPED |WS_THICKFRAME, // overlapped window
- 800, // horizontal position
- 0, // vertical position
- width, // width
- height, // height
- (HWND) NULL, // no parent or owner window
- (HMENU) NULL, // class menu used
- hinst, // instance handle
- NULL); // no window creation data
-
- if (!hwndMain)
- {
- int error = GetLastError();
- return -1;
- }
-
- // Show the window using the flag specified by the program
- // that started the application, and send the application
- // a WM_PAINT message.
-
- ShowWindow(hwndMain, SW_SHOWDEFAULT);
- UpdateWindow(hwndMain);
- return 0;
-}
-
-#elif defined(WEBRTC_LINUX)
-
-int WebRtcCreateWindow(Window *outWindow, Display **outDisplay, int winNum, int width, int height) // unsigned char* title, int titleLength)
-
-{
- int screen, xpos = 10, ypos = 10;
- XEvent evnt;
- XSetWindowAttributes xswa; // window attribute struct
- XVisualInfo vinfo; // screen visual info struct
- unsigned long mask; // attribute mask
-
- // get connection handle to xserver
- Display* _display = XOpenDisplay( NULL );
-
- // get screen number
- screen = DefaultScreen(_display);
-
- // put desired visual info for the screen in vinfo
- if( XMatchVisualInfo(_display, screen, 24, TrueColor, &vinfo) != 0 )
- {
- //printf( "Screen visual info match!\n" );
- }
-
- // set window attributes
- xswa.colormap = XCreateColormap(_display, DefaultRootWindow(_display), vinfo.visual, AllocNone);
- xswa.event_mask = StructureNotifyMask | ExposureMask;
- xswa.background_pixel = 0;
- xswa.border_pixel = 0;
-
- // value mask for attributes
- mask = CWBackPixel | CWBorderPixel | CWColormap | CWEventMask;
-
- switch( winNum )
- {
- case 0:
- xpos = 200;
- ypos = 200;
- break;
- case 1:
- xpos = 300;
- ypos = 200;
- break;
- default:
- break;
- }
-
- // create a subwindow for parent (defroot)
- Window _window = XCreateWindow(_display, DefaultRootWindow(_display),
- xpos, ypos,
- width,
- height,
- 0, vinfo.depth,
- InputOutput,
- vinfo.visual,
- mask, &xswa);
-
- // Set window name
- if( winNum == 0 )
- {
- XStoreName(_display, _window, "VE MM Local Window");
- XSetIconName(_display, _window, "VE MM Local Window");
- }
- else if( winNum == 1 )
- {
- XStoreName(_display, _window, "VE MM Remote Window");
- XSetIconName(_display, _window, "VE MM Remote Window");
- }
-
- // make x report events for mask
- XSelectInput(_display, _window, StructureNotifyMask);
-
- // map the window to the display
- XMapWindow(_display, _window);
-
- // wait for map event
- do
- {
- XNextEvent(_display, &evnt);
- }
- while (evnt.type != MapNotify || evnt.xmap.event != _window);
-
- *outWindow = _window;
- *outDisplay = _display;
-
- return 0;
-}
-#endif // LINUX
-
-// Note: Mac code is in testApi_mac.mm.
-
-class MyRenderCallback: public VideoRenderCallback
-{
-public:
- MyRenderCallback() :
- _cnt(0)
- {
- }
- ;
- ~MyRenderCallback()
- {
- }
- ;
- virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 streamId,
- VideoFrame& videoFrame)
- {
- _cnt++;
- if (_cnt % 100 == 0)
- {
- printf("Render callback %d \n",_cnt);
- }
- return 0;
- }
- WebRtc_Word32 _cnt;
-};
-
-void GetTestVideoFrame(WebRtc_UWord8* frame,
- WebRtc_Word32 width,
- WebRtc_Word32 height,
- WebRtc_UWord8 startColor) {
- // changing color
- static WebRtc_UWord8 color = startColor;
-
- WebRtc_UWord8* destY = frame;
- WebRtc_UWord8* destU = &frame[width*height];
- WebRtc_UWord8* destV = &frame[width*height*5/4];
- //Y
- for (WebRtc_Word32 y=0; y<(width*height); y++)
- {
- destY[y] = color;
- }
- //U
- for (WebRtc_Word32 u=0; u<(width*height/4); u++)
- {
- destU[u] = color;
- }
- //V
- for (WebRtc_Word32 v=0; v<(width*height/4); v++)
- {
- destV[v] = color;
- }
-
- color++;
-}
-
-int TestSingleStream(VideoRender* renderModule) {
- int error = 0;
- // Add settings for a stream to render
- printf("Add stream 0 to entire window\n");
- const int streamId0 = 0;
- VideoRenderCallback* renderCallback0 = renderModule->AddIncomingRenderStream(streamId0, 0, 0.0f, 0.0f, 1.0f, 1.0f);
- assert(renderCallback0 != NULL);
-
-#ifndef WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
- MyRenderCallback externalRender;
- renderModule->AddExternalRenderCallback(streamId0, &externalRender);
-#endif
-
- printf("Start render\n");
- error = renderModule->StartRender(streamId0);
- if (error != 0) {
- // TODO(phoglund): This test will not work if compiled in release mode.
- // This rather silly construct here is to avoid compilation errors when
- // compiling in release. Release => no asserts => unused 'error' variable.
- assert(false);
- }
-
- // Loop through an I420 file and render each frame
- const WebRtc_UWord32 width = 352;
- const WebRtc_UWord32 height = 288;
- const WebRtc_UWord32 numBytes = (WebRtc_UWord32)(1.5 * width * height);
-
- VideoFrame videoFrame0;
- videoFrame0.VerifyAndAllocate(numBytes);
-
- const WebRtc_UWord32 renderDelayMs = 500;
-
- for (int i=0; i<TEST_FRAME_NUM; i++) {
- GetTestVideoFrame(videoFrame0.Buffer(), width, height, TEST_STREAM0_START_COLOR);
- videoFrame0.SetRenderTime(TickTime::MillisecondTimestamp() + renderDelayMs); // Render this frame with the specified delay
- videoFrame0.SetWidth(width);
- videoFrame0.SetHeight(height);
- videoFrame0.SetLength(numBytes);
- renderCallback0->RenderFrame(streamId0, videoFrame0);
- SLEEP(1000/TEST_FRAME_RATE);
- }
-
- videoFrame0.Free();
-
- // Shut down
- printf("Closing...\n");
- error = renderModule->StopRender(streamId0);
- assert(error == 0);
-
- error = renderModule->DeleteIncomingRenderStream(streamId0);
- assert(error == 0);
-
- return 0;
-}
-
-int TestFullscreenStream(VideoRender* &renderModule,
- void* window,
- const VideoRenderType videoRenderType) {
- VideoRender::DestroyVideoRender(renderModule);
- renderModule = VideoRender::CreateVideoRender(12345, window, true, videoRenderType);
-
- TestSingleStream(renderModule);
-
- VideoRender::DestroyVideoRender(renderModule);
- renderModule = VideoRender::CreateVideoRender(12345, window, false, videoRenderType);
-
- return 0;
-}
-
-int TestBitmapText(VideoRender* renderModule) {
-#if defined(WIN32)
-
- int error = 0;
- // Add settings for a stream to render
- printf("Add stream 0 to entire window\n");
- const int streamId0 = 0;
- VideoRenderCallback* renderCallback0 = renderModule->AddIncomingRenderStream(streamId0, 0, 0.0f, 0.0f, 1.0f, 1.0f);
- assert(renderCallback0 != NULL);
-
- printf("Adding Bitmap\n");
- DDCOLORKEY ColorKey; // black
- ColorKey.dwColorSpaceHighValue = RGB(0, 0, 0);
- ColorKey.dwColorSpaceLowValue = RGB(0, 0, 0);
- HBITMAP hbm = (HBITMAP)LoadImage(NULL,
- (LPCTSTR)_T("renderStartImage.bmp"),
- IMAGE_BITMAP, 0, 0, LR_LOADFROMFILE);
- renderModule->SetBitmap(hbm, 0, &ColorKey, 0.0f, 0.0f, 0.3f,
- 0.3f);
-
- printf("Adding Text\n");
- renderModule->SetText(1, (WebRtc_UWord8*) "WebRtc Render Demo App", 20,
- RGB(255, 0, 0), RGB(0, 0, 0), 0.25f, 0.1f, 1.0f,
- 1.0f);
-
- printf("Start render\n");
- error = renderModule->StartRender(streamId0);
- assert(error == 0);
-
- // Loop through an I420 file and render each frame
- const WebRtc_UWord32 width = 352;
- const WebRtc_UWord32 height = 288;
- const WebRtc_UWord32 numBytes = (WebRtc_UWord32)(1.5 * width * height);
-
- VideoFrame videoFrame0;
- videoFrame0.VerifyAndAllocate(numBytes);
-
- const WebRtc_UWord32 renderDelayMs = 500;
-
- for (int i=0; i<TEST_FRAME_NUM; i++) {
- GetTestVideoFrame(videoFrame0.Buffer(), width, height, TEST_STREAM0_START_COLOR);
- videoFrame0.SetRenderTime(TickTime::MillisecondTimestamp() + renderDelayMs); // Render this frame with the specified delay
- videoFrame0.SetWidth(width);
- videoFrame0.SetHeight(height);
- videoFrame0.SetLength(numBytes);
- renderCallback0->RenderFrame(streamId0, videoFrame0);
- SLEEP(1000/TEST_FRAME_RATE);
- }
- videoFrame0.Free();
- // Sleep and let all frames be rendered before closing
- SLEEP(renderDelayMs*2);
-
-
- // Shut down
- printf("Closing...\n");
- ColorKey.dwColorSpaceHighValue = RGB(0,0,0);
- ColorKey.dwColorSpaceLowValue = RGB(0,0,0);
- renderModule->SetBitmap(NULL, 0, &ColorKey, 0.0f, 0.0f, 0.0f, 0.0f);
- renderModule->SetText(1, NULL, 20, RGB(255,255,255),
- RGB(0,0,0), 0.0f, 0.0f, 0.0f, 0.0f);
-
- error = renderModule->StopRender(streamId0);
- assert(error == 0);
-
- error = renderModule->DeleteIncomingRenderStream(streamId0);
- assert(error == 0);
-#endif
-
- return 0;
-}
-
-int TestMultipleStreams(VideoRender* renderModule) {
- // Add settings for a stream to render
- printf("Add stream 0\n");
- const int streamId0 = 0;
- VideoRenderCallback* renderCallback0 =
- renderModule->AddIncomingRenderStream(streamId0, 0, 0.0f, 0.0f, 0.45f, 0.45f);
- assert(renderCallback0 != NULL);
- printf("Add stream 1\n");
- const int streamId1 = 1;
- VideoRenderCallback* renderCallback1 =
- renderModule->AddIncomingRenderStream(streamId1, 0, 0.55f, 0.0f, 1.0f, 0.45f);
- assert(renderCallback1 != NULL);
- printf("Add stream 2\n");
- const int streamId2 = 2;
- VideoRenderCallback* renderCallback2 =
- renderModule->AddIncomingRenderStream(streamId2, 0, 0.0f, 0.55f, 0.45f, 1.0f);
- assert(renderCallback2 != NULL);
- printf("Add stream 3\n");
- const int streamId3 = 3;
- VideoRenderCallback* renderCallback3 =
- renderModule->AddIncomingRenderStream(streamId3, 0, 0.55f, 0.55f, 1.0f, 1.0f);
- assert(renderCallback3 != NULL);
- assert(renderModule->StartRender(streamId0) == 0);
- assert(renderModule->StartRender(streamId1) == 0);
- assert(renderModule->StartRender(streamId2) == 0);
- assert(renderModule->StartRender(streamId3) == 0);
-
- // Loop through an I420 file and render each frame
- const WebRtc_UWord32 width = 352;
- const WebRtc_UWord32 height = 288;
- const WebRtc_UWord32 numBytes = (WebRtc_UWord32)(1.5 * width * height);
-
- VideoFrame videoFrame0;
- videoFrame0.VerifyAndAllocate(numBytes);
- VideoFrame videoFrame1;
- videoFrame1.VerifyAndAllocate(numBytes);
- VideoFrame videoFrame2;
- videoFrame2.VerifyAndAllocate(numBytes);
- VideoFrame videoFrame3;
- videoFrame3.VerifyAndAllocate(numBytes);
-
- const WebRtc_UWord32 renderDelayMs = 500;
-
- for (int i=0; i<TEST_FRAME_NUM; i++) {
- GetTestVideoFrame(videoFrame0.Buffer(), width, height, TEST_STREAM0_START_COLOR);
- videoFrame0.SetRenderTime(TickTime::MillisecondTimestamp() + renderDelayMs); // Render this frame with the specified delay
- videoFrame0.SetWidth(width);
- videoFrame0.SetHeight(height);
- videoFrame0.SetLength(numBytes);
- renderCallback0->RenderFrame(streamId0, videoFrame0);
-
- GetTestVideoFrame(videoFrame1.Buffer(), width, height, TEST_STREAM1_START_COLOR);
- videoFrame1.SetRenderTime(TickTime::MillisecondTimestamp() + renderDelayMs); // Render this frame with the specified delay
- videoFrame1.SetWidth(width);
- videoFrame1.SetHeight(height);
- videoFrame1.SetLength(numBytes);
- renderCallback1->RenderFrame(streamId1, videoFrame1);
-
- GetTestVideoFrame(videoFrame2.Buffer(), width, height, TEST_STREAM2_START_COLOR);
- videoFrame2.SetRenderTime(TickTime::MillisecondTimestamp() + renderDelayMs); // Render this frame with the specified delay
- videoFrame2.SetWidth(width);
- videoFrame2.SetHeight(height);
- videoFrame2.SetLength(numBytes);
- renderCallback2->RenderFrame(streamId2, videoFrame2);
-
- GetTestVideoFrame(videoFrame3.Buffer(), width, height, TEST_STREAM3_START_COLOR);
- videoFrame3.SetRenderTime(TickTime::MillisecondTimestamp() + renderDelayMs); // Render this frame with the specified delay
- videoFrame3.SetWidth(width);
- videoFrame3.SetHeight(height);
- videoFrame3.SetLength(numBytes);
- renderCallback3->RenderFrame(streamId3, videoFrame3);
-
- SLEEP(1000/TEST_FRAME_RATE);
- }
-
- videoFrame0.Free();
- videoFrame1.Free();
- videoFrame2.Free();
- videoFrame3.Free();
-
- // Shut down
- printf("Closing...\n");
- assert(renderModule->StopRender(streamId0) == 0);
- assert(renderModule->DeleteIncomingRenderStream(streamId0) == 0);
- assert(renderModule->StopRender(streamId1) == 0);
- assert(renderModule->DeleteIncomingRenderStream(streamId1) == 0);
- assert(renderModule->StopRender(streamId2) == 0);
- assert(renderModule->DeleteIncomingRenderStream(streamId2) == 0);
- assert(renderModule->StopRender(streamId3) == 0);
- assert(renderModule->DeleteIncomingRenderStream(streamId3) == 0);
-
- return 0;
-}
-
-int TestExternalRender(VideoRender* renderModule) {
- MyRenderCallback *externalRender = new MyRenderCallback();
-
- const int streamId0 = 0;
- VideoRenderCallback* renderCallback0 =
- renderModule->AddIncomingRenderStream(streamId0, 0, 0.0f, 0.0f,
- 1.0f, 1.0f);
- assert(renderCallback0 != NULL);
- assert(renderModule->AddExternalRenderCallback(streamId0,
- externalRender) == 0);
-
- assert(renderModule->StartRender(streamId0) == 0);
-
- const WebRtc_UWord32 width = 352;
- const WebRtc_UWord32 height = 288;
- const WebRtc_UWord32 numBytes = (WebRtc_UWord32) (1.5 * width * height);
- VideoFrame videoFrame0;
- videoFrame0.VerifyAndAllocate(numBytes);
-
- const WebRtc_UWord32 renderDelayMs = 500;
- int frameCount = TEST_FRAME_NUM;
- for (int i=0; i<frameCount; i++) {
- videoFrame0.SetRenderTime(TickTime::MillisecondTimestamp() + renderDelayMs);
- videoFrame0.SetWidth(width);
- videoFrame0.SetHeight(height);
- renderCallback0->RenderFrame(streamId0, videoFrame0);
- SLEEP(33);
- }
-
- // Sleep and let all frames be rendered before closing
- SLEEP(2*renderDelayMs);
- videoFrame0.Free();
-
- assert(renderModule->StopRender(streamId0) == 0);
- assert(renderModule->DeleteIncomingRenderStream(streamId0) == 0);
- assert(frameCount == externalRender->_cnt);
-
- delete externalRender;
- externalRender = NULL;
-
- return 0;
-}
-
-void RunVideoRenderTests(void* window, VideoRenderType windowType) {
-#ifndef WEBRTC_INCLUDE_INTERNAL_VIDEO_RENDER
- windowType = kRenderExternal;
-#endif
-
- int myId = 12345;
-
- // Create the render module
- printf("Create render module\n");
- VideoRender* renderModule = NULL;
- renderModule = VideoRender::CreateVideoRender(myId,
- window,
- false,
- windowType);
- assert(renderModule != NULL);
-
-
- // ##### Test single stream rendering ####
- printf("#### TestSingleStream ####\n");
- if (TestSingleStream(renderModule) != 0) {
- printf ("TestSingleStream failed\n");
- }
-
- // ##### Test fullscreen rendering ####
- printf("#### TestFullscreenStream ####\n");
- if (TestFullscreenStream(renderModule, window, windowType) != 0) {
- printf ("TestFullscreenStream failed\n");
- }
-
- // ##### Test bitmap and text ####
- printf("#### TestBitmapText ####\n");
- if (TestBitmapText(renderModule) != 0) {
- printf ("TestBitmapText failed\n");
- }
-
- // ##### Test multiple streams ####
- printf("#### TestMultipleStreams ####\n");
- if (TestMultipleStreams(renderModule) != 0) {
- printf ("TestMultipleStreams failed\n");
- }
-
- // ##### Test multiple streams ####
- printf("#### TestExternalRender ####\n");
- if (TestExternalRender(renderModule) != 0) {
- printf ("TestExternalRender failed\n");
- }
-
- delete renderModule;
- renderModule = NULL;
-
- printf("VideoRender unit tests passed.\n");
-}
-
-// Note: The Mac main is implemented in testApi_mac.mm.
-#if defined(_WIN32)
-int _tmain(int argc, _TCHAR* argv[])
-#elif defined(WEBRTC_LINUX)
-int main(int argc, char* argv[])
-#endif
-#if !defined(WEBRTC_MAC)
-{
- // Create a window for testing.
- void* window = NULL;
-#if defined (_WIN32)
- HWND testHwnd;
- WebRtcCreateWindow(testHwnd, 0, 352, 288);
- window = (void*)testHwnd;
- VideoRenderType windowType = kRenderWindows;
-#elif defined(WEBRTC_LINUX)
- Window testWindow;
- Display* display;
- WebRtcCreateWindow(&testWindow, &display, 0, 352, 288);
- VideoRenderType windowType = kRenderX11;
- window = (void*)testWindow;
-#endif // WEBRTC_LINUX
-
- RunVideoRenderTests(window, windowType);
- return 0;
-}
-#endif // !WEBRTC_MAC
diff --git a/trunk/src/modules/video_render/main/test/testAPI/testAPI.h b/trunk/src/modules/video_render/main/test/testAPI/testAPI.h
deleted file mode 100644
index e0e0631..0000000
--- a/trunk/src/modules/video_render/main/test/testAPI/testAPI.h
+++ /dev/null
@@ -1,18 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_MODULES_VIDEO_RENDER_MAIN_TEST_TESTAPI_TESTAPI_H
-#define WEBRTC_MODULES_VIDEO_RENDER_MAIN_TEST_TESTAPI_TESTAPI_H
-
-#include "video_render_defines.h"
-
-void RunVideoRenderTests(void* window, webrtc::VideoRenderType windowType);
-
-#endif // WEBRTC_MODULES_VIDEO_RENDER_MAIN_TEST_TESTAPI_TESTAPI_H
diff --git a/trunk/src/modules/video_render/main/test/testAPI/testAPI_mac.mm b/trunk/src/modules/video_render/main/test/testAPI/testAPI_mac.mm
deleted file mode 100644
index 2f836ab..0000000
--- a/trunk/src/modules/video_render/main/test/testAPI/testAPI_mac.mm
+++ /dev/null
@@ -1,69 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testAPI.h"
-
-#include <iostream>
-
-#import <Foundation/Foundation.h>
-#import <Cocoa/Cocoa.h>
-#import <AppKit/AppKit.h>
-#import <QTKit/QTKit.h>
-#include <sys/time.h>
-
-#include "common_types.h"
-#import "modules/video_render/main/source/mac/cocoa_render_view.h"
-#include "module_common_types.h"
-#include "process_thread.h"
-#include "tick_util.h"
-#include "trace.h"
-#include "video_render_defines.h"
-#include "video_render.h"
-
-using namespace webrtc;
-
-int WebRtcCreateWindow(CocoaRenderView*& cocoaRenderer, int winNum, int width, int height)
-{
- // In Cocoa, rendering is not done directly to a window like in Windows and Linux.
- // It is rendererd to a Subclass of NSOpenGLView
-
- // create cocoa container window
- NSRect outWindowFrame = NSMakeRect(200, 800, width + 20, height + 20);
- NSWindow* outWindow = [[NSWindow alloc] initWithContentRect:outWindowFrame
- styleMask:NSTitledWindowMask
- backing:NSBackingStoreBuffered
- defer:NO];
- [outWindow orderOut:nil];
- [outWindow setTitle:@"Cocoa Renderer"];
- [outWindow setBackgroundColor:[NSColor blueColor]];
-
- // create renderer and attach to window
- NSRect cocoaRendererFrame = NSMakeRect(10, 10, width, height);
- cocoaRenderer = [[CocoaRenderView alloc] initWithFrame:cocoaRendererFrame];
- [[outWindow contentView] addSubview:(NSView*)cocoaRenderer];
-
- [outWindow makeKeyAndOrderFront:NSApp];
-
- return 0;
-}
-
-int main (int argc, const char * argv[]) {
- NSAutoreleasePool * pool = [[NSAutoreleasePool alloc] init];
- [NSApplication sharedApplication];
-
- CocoaRenderView* testWindow;
- WebRtcCreateWindow(testWindow, 0, 352, 288);
- VideoRenderType windowType = kRenderCocoa;
- void* window = (void*)testWindow;
-
- RunVideoRenderTests(window, windowType);
-
- [pool release];
-}
diff --git a/trunk/src/system_wrappers/OWNERS b/trunk/src/system_wrappers/OWNERS
deleted file mode 100644
index 4091a93..0000000
--- a/trunk/src/system_wrappers/OWNERS
+++ /dev/null
@@ -1,7 +0,0 @@
-henrike@webrtc.org
-pwestin@webrtc.org
-perkj@webrtc.org
-henrika@webrtc.org
-henrikg@webrtc.org
-mflodman@webrtc.org
-niklas.enbom@webrtc.org
\ No newline at end of file
diff --git a/trunk/src/system_wrappers/interface/aligned_malloc.h b/trunk/src/system_wrappers/interface/aligned_malloc.h
deleted file mode 100644
index c229435..0000000
--- a/trunk/src/system_wrappers/interface/aligned_malloc.h
+++ /dev/null
@@ -1,25 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ALIGNED_MALLOC_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ALIGNED_MALLOC_H_
-
-#include <stddef.h>
-
-namespace webrtc
-{
- void* AlignedMalloc(
- size_t size,
- size_t alignment);
- void AlignedFree(
- void* memBlock);
-}
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ALIGNED_MALLOC_H_
diff --git a/trunk/src/system_wrappers/interface/atomic32_wrapper.h b/trunk/src/system_wrappers/interface/atomic32_wrapper.h
deleted file mode 100644
index 40862fb..0000000
--- a/trunk/src/system_wrappers/interface/atomic32_wrapper.h
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Atomic system independant 32-bit integer.
-// Note: uses full memory barriers.
-// Note: assumes 32-bit (or higher) system
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ATOMIC32_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ATOMIC32_WRAPPER_H_
-
-#include "common_types.h"
-
-namespace webrtc {
-class Atomic32Impl;
-class Atomic32Wrapper
-{
-public:
- Atomic32Wrapper(WebRtc_Word32 initialValue = 0);
- ~Atomic32Wrapper();
-
- // Prefix operator!
- WebRtc_Word32 operator++();
- WebRtc_Word32 operator--();
-
- Atomic32Wrapper& operator=(const Atomic32Wrapper& rhs);
- Atomic32Wrapper& operator=(WebRtc_Word32 rhs);
-
- WebRtc_Word32 operator+=(WebRtc_Word32 rhs);
- WebRtc_Word32 operator-=(WebRtc_Word32 rhs);
-
- // Sets the value atomically to newValue if the value equals compare value.
- // The function returns true if the exchange happened.
- bool CompareExchange(WebRtc_Word32 newValue, WebRtc_Word32 compareValue);
- WebRtc_Word32 Value() const;
-private:
- // Disable the + and - operator since it's unclear what these operations
- // should do.
- Atomic32Wrapper operator+(const Atomic32Wrapper& rhs);
- Atomic32Wrapper operator-(const Atomic32Wrapper& rhs);
-
- WebRtc_Word32& operator++(int);
- WebRtc_Word32& operator--(int);
-
- // Cheshire cat to hide the implementation (faster than
- // using virtual functions)
- Atomic32Impl& _impl;
-};
-} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_ATOMIC32_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/condition_variable_wrapper.h b/trunk/src/system_wrappers/interface/condition_variable_wrapper.h
deleted file mode 100644
index c040fbf..0000000
--- a/trunk/src/system_wrappers/interface/condition_variable_wrapper.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONDITION_VARIABLE_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONDITION_VARIABLE_WRAPPER_H_
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class ConditionVariableWrapper
-{
-public:
- // Factory method, constructor disabled.
- static ConditionVariableWrapper* CreateConditionVariable();
-
- virtual ~ConditionVariableWrapper() {}
-
- // Calling thread will atomically release critSect and wait until next
- // some other thread calls Wake() or WakeAll().
- virtual void SleepCS(CriticalSectionWrapper& critSect) = 0;
-
- // Same as above but with a timeout.
- virtual bool SleepCS(CriticalSectionWrapper& critSect,
- unsigned long maxTimeInMS) = 0;
-
- // Wakes one thread calling SleepCS().
- virtual void Wake() = 0;
-
- // Wakes all threads calling SleepCS().
- virtual void WakeAll() = 0;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONDITION_VARIABLE_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/constructor_magic.h b/trunk/src/system_wrappers/interface/constructor_magic.h
deleted file mode 100644
index b2aabc5..0000000
--- a/trunk/src/system_wrappers/interface/constructor_magic.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * WebRtc
- * Copy from third_party/libjingle/source/talk/base/constructormagic.h
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONSTRUCTOR_MAGIC_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONSTRUCTOR_MAGIC_H_
-
-#ifndef DISALLOW_ASSIGN
-#define DISALLOW_ASSIGN(TypeName) \
- void operator=(const TypeName&)
-#endif
-
-#ifndef DISALLOW_COPY_AND_ASSIGN
-// A macro to disallow the evil copy constructor and operator= functions
-// This should be used in the private: declarations for a class
-#define DISALLOW_COPY_AND_ASSIGN(TypeName) \
- TypeName(const TypeName&); \
- DISALLOW_ASSIGN(TypeName)
-#endif
-
-#ifndef DISALLOW_EVIL_CONSTRUCTORS
-// Alternative, less-accurate legacy name.
-#define DISALLOW_EVIL_CONSTRUCTORS(TypeName) \
- DISALLOW_COPY_AND_ASSIGN(TypeName)
-#endif
-
-#ifndef DISALLOW_IMPLICIT_CONSTRUCTORS
-// A macro to disallow all the implicit constructors, namely the
-// default constructor, copy constructor and operator= functions.
-//
-// This should be used in the private: declarations for a class
-// that wants to prevent anyone from instantiating it. This is
-// especially useful for classes containing only static methods.
-#define DISALLOW_IMPLICIT_CONSTRUCTORS(TypeName) \
- TypeName(); \
- DISALLOW_EVIL_CONSTRUCTORS(TypeName)
-#endif
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CONSTRUCTOR_MAGIC_H_
diff --git a/trunk/src/system_wrappers/interface/cpu_features_wrapper.h b/trunk/src/system_wrappers/interface/cpu_features_wrapper.h
deleted file mode 100644
index d949592..0000000
--- a/trunk/src/system_wrappers/interface/cpu_features_wrapper.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_FEATURES_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_FEATURES_WRAPPER_H_
-
-#if defined(__cplusplus) || defined(c_plusplus)
-extern "C" {
-#endif
-
-#include <typedefs.h>
-
-// List of features in x86.
-typedef enum {
- kSSE2,
- kSSE3
-} CPUFeature;
-
-// List of features in ARM.
-enum {
- kCPUFeatureARMv7 = (1 << 0),
- kCPUFeatureVFPv3 = (1 << 1),
- kCPUFeatureNEON = (1 << 2),
- kCPUFeatureLDREXSTREX = (1 << 3)
-};
-
-typedef int (*WebRtc_CPUInfo)(CPUFeature feature);
-// returns true if the CPU supports the feature.
-extern WebRtc_CPUInfo WebRtc_GetCPUInfo;
-// No CPU feature is available => straight C path.
-extern WebRtc_CPUInfo WebRtc_GetCPUInfoNoASM;
-
-// Return the features in an ARM device.
-// It detects the features in the hardware platform, and returns supported
-// values in the above enum definition as a bitmask.
-extern uint64_t WebRtc_GetCPUFeaturesARM(void);
-
-#if defined(__cplusplus) || defined(c_plusplus)
-} // extern "C"
-#endif
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_FEATURES_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/cpu_info.h b/trunk/src/system_wrappers/interface/cpu_info.h
deleted file mode 100644
index a6da29f..0000000
--- a/trunk/src/system_wrappers/interface/cpu_info.h
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_INFO_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_INFO_H_
-
-#include "typedefs.h"
-
-namespace webrtc {
-class CpuInfo
-{
-public:
- static WebRtc_UWord32 DetectNumberOfCores();
-
-private:
- CpuInfo() {}
- static WebRtc_UWord32 _numberOfCores;
-};
-} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_INFO_H_
diff --git a/trunk/src/system_wrappers/interface/cpu_wrapper.h b/trunk/src/system_wrappers/interface/cpu_wrapper.h
deleted file mode 100644
index d938741..0000000
--- a/trunk/src/system_wrappers/interface/cpu_wrapper.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_WRAPPER_H_
-
-#include "typedefs.h"
-
-namespace webrtc {
-class CpuWrapper
-{
-public:
- static CpuWrapper* CreateCpu();
- virtual ~CpuWrapper() {}
-
- // Returns the average CPU usage for all processors. The CPU usage can be
- // between and including 0 to 100 (%)
- virtual WebRtc_Word32 CpuUsage() = 0;
- virtual WebRtc_Word32 CpuUsage(WebRtc_Word8* processName,
- WebRtc_UWord32 length) = 0;
- virtual WebRtc_Word32 CpuUsage(WebRtc_UWord32 dwProcessID) = 0;
-
- // The CPU usage per core is returned in cpu_usage. The CPU can be between
- // and including 0 to 100 (%)
- // Note that the pointer passed as cpu_usage is redirected to a local member
- // of the CPU wrapper.
- // numCores is the number of cores in the cpu_usage array.
- // The return value is -1 for failure or 0-100, indicating the average
- // CPU usage across all cores.
- // Note: on some OSs this class is initialized lazy. This means that it
- // might not yet be possible to retrieve any CPU metrics. When this happens
- // the return value will be zero (indicating that there is not a failure),
- // numCores will be 0 and cpu_usage will be set to NULL (indicating that
- // no metrics are available yet). Once the initialization is completed,
- // which can take in the order of seconds, CPU metrics can be retrieved.
- virtual WebRtc_Word32 CpuUsageMultiCore(WebRtc_UWord32& numCores,
- WebRtc_UWord32*& cpu_usage) = 0;
-
- virtual void Reset() = 0;
- virtual void Stop() = 0;
-
-protected:
- CpuWrapper() {}
-};
-} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CPU_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/critical_section_wrapper.h b/trunk/src/system_wrappers/interface/critical_section_wrapper.h
deleted file mode 100644
index cfec9ae..0000000
--- a/trunk/src/system_wrappers/interface/critical_section_wrapper.h
+++ /dev/null
@@ -1,74 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CRITICAL_SECTION_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CRITICAL_SECTION_WRAPPER_H_
-
-// If the critical section is heavily contended it may be beneficial to use
-// read/write locks instead.
-
-#include "common_types.h"
-
-namespace webrtc {
-class CriticalSectionWrapper
-{
-public:
- // Factory method, constructor disabled
- static CriticalSectionWrapper* CreateCriticalSection();
-
- virtual ~CriticalSectionWrapper() {}
-
- // Tries to grab lock, beginning of a critical section. Will wait for the
- // lock to become available if the grab failed.
- virtual void Enter() = 0;
-
- // Returns a grabbed lock, end of critical section.
- virtual void Leave() = 0;
-};
-
-// RAII extension of the critical section. Prevents Enter/Leave mismatches and
-// provides more compact critical section syntax.
-class CriticalSectionScoped
-{
-public:
- // Deprecated, don't add more users of this constructor.
- // TODO(mflodman) Remove this version of the constructor when no one is
- // using it any longer.
- explicit CriticalSectionScoped(CriticalSectionWrapper& critsec)
- : _ptrCritSec(&critsec)
- {
- _ptrCritSec->Enter();
- }
-
- explicit CriticalSectionScoped(CriticalSectionWrapper* critsec)
- : _ptrCritSec(critsec)
- {
- _ptrCritSec->Enter();
- }
-
- ~CriticalSectionScoped()
- {
- if (_ptrCritSec)
- {
- Leave();
- }
- }
-
-private:
- void Leave()
- {
- _ptrCritSec->Leave();
- _ptrCritSec = 0;
- }
-
- CriticalSectionWrapper* _ptrCritSec;
-};
-} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_CRITICAL_SECTION_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/data_log.h b/trunk/src/system_wrappers/interface/data_log.h
deleted file mode 100644
index 6fc1d64..0000000
--- a/trunk/src/system_wrappers/interface/data_log.h
+++ /dev/null
@@ -1,121 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This singleton can be used for logging data for offline processing. Data
- * logged with it can conveniently be parsed and processed with e.g. Matlab.
- *
- * Following is an example of the log file format, starting with the header
- * row at line 1, and the data rows following.
- * col1,col2,col3,multi-value-col4[3],,,col5
- * 123,10.2,-243,1,2,3,100
- * 241,12.3,233,1,2,3,200
- * 13,16.4,-13,1,2,3,300
- *
- * As can be seen in the example, a multi-value-column is specified with the
- * name followed the number of elements it contains. This followed by
- * number of elements - 1 empty columns.
- *
- * Without multi-value-columns this format can be natively by Matlab. With
- * multi-value-columns a small Matlab script is needed, available at
- * trunk/tools/matlab/parseLog.m.
- *
- * Table names and column names are case sensitive.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_H_
-
-#include <string>
-
-#include "data_log_impl.h"
-
-namespace webrtc {
-
-class DataLog {
- public:
- // Creates a log which uses a separate thread (referred to as the file
- // writer thread) for writing log rows to file.
- //
- // Calls to this function after the log object has been created will only
- // increment the reference counter.
- static int CreateLog();
-
- // Decrements the reference counter and deletes the log when the counter
- // reaches 0. Should be called equal number of times as successful calls to
- // CreateLog or memory leak will occur.
- static void ReturnLog();
-
- // Combines the string table_name and the integer table_id into a new string
- // table_name + _ + table_id. The new string will be lower-case.
- static std::string Combine(const std::string& table_name, int table_id);
-
- // Adds a new table, with the name table_name, and creates the file, with the
- // name table_name + ".txt", to which the table will be written.
- // table_name is treated in a case sensitive way.
- static int AddTable(const std::string& table_name);
-
- // Adds a new column to a table. The column will be a multi-value-column
- // if multi_value_length is greater than 1.
- // table_name and column_name are treated in a case sensitive way.
- static int AddColumn(const std::string& table_name,
- const std::string& column_name,
- int multi_value_length);
-
- // Inserts a single value into a table with name table_name at the column with
- // name column_name.
- // Note that the ValueContainer makes use of the copy constructor,
- // operator= and operator<< of the type T, and that the template type must
- // implement a deep copy copy constructor and operator=.
- // Copy constructor and operator= must not be disabled for the type T.
- // table_name and column_name are treated in a case sensitive way.
- template<class T>
- static int InsertCell(const std::string& table_name,
- const std::string& column_name,
- T value) {
- DataLogImpl* data_log = DataLogImpl::StaticInstance();
- if (data_log == NULL)
- return -1;
- return data_log->InsertCell(
- table_name,
- column_name,
- new ValueContainer<T>(value));
- }
-
- // Inserts an array of values into a table with name table_name at the
- // column specified by column_name, which must be a multi-value-column.
- // Note that the MultiValueContainer makes use of the copy constructor,
- // operator= and operator<< of the type T, and that the template type
- // must implement a deep copy copy constructor and operator=.
- // Copy constructor and operator= must not be disabled for the type T.
- // table_name and column_name are treated in a case sensitive way.
- template<class T>
- static int InsertCell(const std::string& table_name,
- const std::string& column_name,
- const T* array,
- int length) {
- DataLogImpl* data_log = DataLogImpl::StaticInstance();
- if (data_log == NULL)
- return -1;
- return data_log->InsertCell(
- table_name,
- column_name,
- new MultiValueContainer<T>(array, length));
- }
-
- // For the table with name table_name: Writes the current row to file.
- // Starts a new empty row.
- // table_name is treated in a case-sensitive way.
- static int NextRow(const std::string& table_name);
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_H_
diff --git a/trunk/src/system_wrappers/interface/data_log_c.h b/trunk/src/system_wrappers/interface/data_log_c.h
deleted file mode 100644
index fffbb4f..0000000
--- a/trunk/src/system_wrappers/interface/data_log_c.h
+++ /dev/null
@@ -1,89 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This is a pure C wrapper of the DataLog class. The functions are directly
- * mapped here except for InsertCell as C does not support templates.
- * See data_log.h for a description of the functions.
- */
-
-#ifndef SRC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_C_H_
-#define SRC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_C_H_
-
-#include <stddef.h> /* size_t */
-
-#include "typedefs.h" /* NOLINT(build/include) */
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-/*
- * All char* parameters in this file are expected to be null-terminated
- * character sequences.
- */
-int WebRtcDataLog_CreateLog();
-void WebRtcDataLog_ReturnLog();
-char* WebRtcDataLog_Combine(char* combined_name, size_t combined_len,
- const char* table_name, int table_id);
-int WebRtcDataLog_AddTable(const char* table_name);
-int WebRtcDataLog_AddColumn(const char* table_name, const char* column_name,
- int multi_value_length);
-
-int WebRtcDataLog_InsertCell_int(const char* table_name,
- const char* column_name,
- int value);
-int WebRtcDataLog_InsertArray_int(const char* table_name,
- const char* column_name,
- const int* values,
- int length);
-int WebRtcDataLog_InsertCell_float(const char* table_name,
- const char* column_name,
- float value);
-int WebRtcDataLog_InsertArray_float(const char* table_name,
- const char* column_name,
- const float* values,
- int length);
-int WebRtcDataLog_InsertCell_double(const char* table_name,
- const char* column_name,
- double value);
-int WebRtcDataLog_InsertArray_double(const char* table_name,
- const char* column_name,
- const double* values,
- int length);
-int WebRtcDataLog_InsertCell_int32(const char* table_name,
- const char* column_name,
- int32_t value);
-int WebRtcDataLog_InsertArray_int32(const char* table_name,
- const char* column_name,
- const int32_t* values,
- int length);
-int WebRtcDataLog_InsertCell_uint32(const char* table_name,
- const char* column_name,
- uint32_t value);
-int WebRtcDataLog_InsertArray_uint32(const char* table_name,
- const char* column_name,
- const uint32_t* values,
- int length);
-int WebRtcDataLog_InsertCell_int64(const char* table_name,
- const char* column_name,
- int64_t value);
-int WebRtcDataLog_InsertArray_int64(const char* table_name,
- const char* column_name,
- const int64_t* values,
- int length);
-
-int WebRtcDataLog_NextRow(const char* table_name);
-
-#ifdef __cplusplus
-} /* end of extern "C" */
-#endif
-
-#endif /* SRC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_C_H_ */ /* NOLINT */
diff --git a/trunk/src/system_wrappers/interface/data_log_impl.h b/trunk/src/system_wrappers/interface/data_log_impl.h
deleted file mode 100644
index cef4964..0000000
--- a/trunk/src/system_wrappers/interface/data_log_impl.h
+++ /dev/null
@@ -1,157 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the helper classes for the DataLog APIs. See data_log.h
- * for the APIs.
- *
- * These classes are helper classes used for logging data for offline
- * processing. Data logged with these classes can conveniently be parsed and
- * processed with e.g. Matlab.
- */
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_IMPL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_IMPL_H_
-
-#include <map>
-#include <sstream>
-#include <string>
-#include <vector>
-
-#include "scoped_ptr.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class EventWrapper;
-class LogTable;
-class RWLockWrapper;
-class ThreadWrapper;
-
-// All container classes need to implement a ToString-function to be
-// writable to file. Enforce this via the Container interface.
-class Container {
- public:
- virtual ~Container() {}
-
- virtual void ToString(std::string* container_string) const = 0;
-};
-
-template<class T>
-class ValueContainer : public Container {
- public:
- explicit ValueContainer(T data) : data_(data) {}
-
- virtual void ToString(std::string* container_string) const {
- *container_string = "";
- std::stringstream ss;
- ss << data_ << ",";
- ss >> *container_string;
- }
-
- private:
- T data_;
-};
-
-template<class T>
-class MultiValueContainer : public Container {
- public:
- MultiValueContainer(const T* data, int length)
- : data_(data, data + length) {
- }
-
- virtual void ToString(std::string* container_string) const {
- *container_string = "";
- std::stringstream ss;
- for (size_t i = 0; i < data_.size(); ++i)
- ss << data_[i] << ",";
- *container_string += ss.str();
- }
-
- private:
- std::vector<T> data_;
-};
-
-class DataLogImpl {
- public:
- ~DataLogImpl();
-
- // The implementation of the CreateLog() method declared in data_log.h.
- // See data_log.h for a description.
- static int CreateLog();
-
- // The implementation of the StaticInstance() method declared in data_log.h.
- // See data_log.h for a description.
- static DataLogImpl* StaticInstance();
-
- // The implementation of the ReturnLog() method declared in data_log.h. See
- // data_log.h for a description.
- static void ReturnLog();
-
- // The implementation of the AddTable() method declared in data_log.h. See
- // data_log.h for a description.
- int AddTable(const std::string& table_name);
-
- // The implementation of the AddColumn() method declared in data_log.h. See
- // data_log.h for a description.
- int AddColumn(const std::string& table_name,
- const std::string& column_name,
- int multi_value_length);
-
- // Inserts a Container into a table with name table_name at the column
- // with name column_name.
- // column_name is treated in a case sensitive way.
- int InsertCell(const std::string& table_name,
- const std::string& column_name,
- const Container* value_container);
-
- // The implementation of the NextRow() method declared in data_log.h. See
- // data_log.h for a description.
- int NextRow(const std::string& table_name);
-
- private:
- DataLogImpl();
-
- // Initializes the DataLogImpl object, allocates and starts the
- // thread file_writer_thread_.
- int Init();
-
- // Write all complete rows in every table to file.
- // This function should only be called by the file_writer_thread_ if that
- // thread is running to avoid race conditions.
- void Flush();
-
- // Run() is called by the thread file_writer_thread_.
- static bool Run(void* obj);
-
- // This function writes data to file. Note, it blocks if there is no data
- // that should be written to file availble. Flush is the non-blocking
- // version of this function.
- void Process();
-
- // Stops the continuous calling of Process().
- void StopThread();
-
- // Collection of tables indexed by the table name as std::string.
- typedef std::map<std::string, LogTable*> TableMap;
- typedef webrtc::scoped_ptr<CriticalSectionWrapper> CritSectScopedPtr;
-
- static CritSectScopedPtr crit_sect_;
- static DataLogImpl* instance_;
- int counter_;
- TableMap tables_;
- EventWrapper* flush_event_;
- ThreadWrapper* file_writer_thread_;
- RWLockWrapper* tables_lock_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_DATA_LOG_IMPL_H_
diff --git a/trunk/src/system_wrappers/interface/event_wrapper.h b/trunk/src/system_wrappers/interface/event_wrapper.h
deleted file mode 100644
index 0c9a908..0000000
--- a/trunk/src/system_wrappers/interface/event_wrapper.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_EVENT_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_EVENT_WRAPPER_H_
-
-namespace webrtc {
-enum EventTypeWrapper
-{
- kEventSignaled = 1,
- kEventError = 2,
- kEventTimeout = 3
-};
-
-#define WEBRTC_EVENT_10_SEC 10000
-#define WEBRTC_EVENT_INFINITE 0xffffffff
-
-class EventWrapper
-{
-public:
- // Factory method. Constructor disabled.
- static EventWrapper* Create();
- virtual ~EventWrapper() {}
-
- // Releases threads who are calling Wait() and has started waiting. Please
- // note that a thread calling Wait() will not start waiting immediately.
- // assumptions to the contrary is a very common source of issues in
- // multithreaded programming.
- // Set is sticky in the sense that it will release at least one thread
- // either immediately or some time in the future.
- virtual bool Set() = 0;
-
- // Prevents future Wait() calls from finishing without a new Set() call.
- virtual bool Reset() = 0;
-
- // Puts the calling thread into a wait state. The thread may be released
- // by a Set() call depending on if other threads are waiting and if so on
- // timing. The thread that was released will call Reset() before leaving
- // preventing more threads from being released. If multiple threads
- // are waiting for the same Set(), only one (random) thread is guaranteed to
- // be released. It is possible that multiple (random) threads are released
- // Depending on timing.
- virtual EventTypeWrapper Wait(unsigned long maxTime) = 0;
-
- // Starts a timer that will call a non-sticky version of Set() either once
- // or periodically. If the timer is periodic it ensures that there is no
- // drift over time relative to the system clock.
- virtual bool StartTimer(bool periodic, unsigned long time) = 0;
-
- virtual bool StopTimer() = 0;
-
- // Only implemented on Windows
- // Returns 1 if a key has been pressed since last call to this function.
- // -1 indicates failure
- // 0 indicates no key has been pressed since last call
- // TODO(hellner) this function does not seem to belong here
- static int KeyPressed();
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_EVENT_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/file_wrapper.h b/trunk/src/system_wrappers/interface/file_wrapper.h
deleted file mode 100644
index 4d17438..0000000
--- a/trunk/src/system_wrappers/interface/file_wrapper.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FILE_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FILE_WRAPPER_H_
-
-#include <stddef.h>
-
-#include "common_types.h"
-#include "typedefs.h"
-
-// Implementation of an InStream and OutStream that can read (exclusive) or
-// write from/to a file.
-
-namespace webrtc {
-
-class FileWrapper : public InStream, public OutStream
-{
-public:
- static const size_t kMaxFileNameSize = 1024;
-
- // Factory method. Constructor disabled.
- static FileWrapper* Create();
-
- // Returns true if a file has been opened.
- virtual bool Open() const = 0;
-
- // Opens a file in read or write mode, decided by the readOnly parameter.
- virtual int OpenFile(const char* fileNameUTF8,
- bool readOnly,
- bool loop = false,
- bool text = false) = 0;
-
- virtual int CloseFile() = 0;
-
- // Limits the file size to |bytes|. Writing will fail after the cap
- // is hit. Pass zero to use an unlimited size.
- virtual int SetMaxFileSize(size_t bytes) = 0;
-
- // Flush any pending writes.
- virtual int Flush() = 0;
-
- // Returns the opened file's name in |fileNameUTF8|. Provide the size of
- // the buffer in bytes in |size|. The name will be truncated if |size| is
- // too small.
- virtual int FileName(char* fileNameUTF8,
- size_t size) const = 0;
-
- // Write |format| to the opened file. Arguments are taken in the same manner
- // as printf. That is, supply a format string containing text and
- // specifiers. Returns the number of characters written or -1 on error.
- virtual int WriteText(const char* format, ...) = 0;
-
- // Inherited from Instream.
- // Reads |length| bytes from file to |buf|. Returns the number of bytes read
- // or -1 on error.
- virtual int Read(void* buf, int length) = 0;
-
- // Inherited from OutStream.
- // Writes |length| bytes from |buf| to file. The actual writing may happen
- // some time later. Call Flush() to force a write.
- virtual bool Write(const void *buf, int length) = 0;
-
- // Inherited from both Instream and OutStream.
- // Rewinds the file to the start. Only available when OpenFile() has been
- // called with |loop| == true or |readOnly| == true.
- virtual int Rewind() = 0;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_FILE_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/fix_interlocked_exchange_pointer_win.h b/trunk/src/system_wrappers/interface/fix_interlocked_exchange_pointer_win.h
deleted file mode 100644
index d85c724..0000000
--- a/trunk/src/system_wrappers/interface/fix_interlocked_exchange_pointer_win.h
+++ /dev/null
@@ -1,35 +0,0 @@
-// Copyright (c) 2006-2008 The Chromium Authors. All rights reserved.
-// Use of this source code is governed by a BSD-style license that can be
-// found in the LICENSE file under third_party_mods/chromium directory of
-// source tree or at
-// http://src.chromium.org/viewvc/chrome/trunk/src/LICENSE
-
-// Various inline functions and macros to fix compilation of 32 bit target
-// on MSVC with /Wp64 flag enabled.
-
-// The original code can be found here:
-// http://src.chromium.org/svn/trunk/src/base/fix_wp64.h
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_FIX_INTERLOCKED_EXCHANGE_POINTER_WINDOWS_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_FIX_INTERLOCKED_EXCHANGE_POINTER_WINDOWS_H_
-
-#include <windows.h>
-
-// Platform SDK fixes when building with /Wp64 for a 32 bits target.
-#if !defined(_WIN64) && defined(_Wp64)
-
-#ifdef InterlockedExchangePointer
-#undef InterlockedExchangePointer
-// The problem is that the macro provided for InterlockedExchangePointer() is
-// doing a (LONG) C-style cast that triggers invariably the warning C4312 when
-// building on 32 bits.
-inline void* InterlockedExchangePointer(void* volatile* target, void* value) {
- return reinterpret_cast<void*>(static_cast<LONG_PTR>(InterlockedExchange(
- reinterpret_cast<volatile LONG*>(target),
- static_cast<LONG>(reinterpret_cast<LONG_PTR>(value)))));
-}
-#endif // #ifdef InterlockedExchangePointer
-
-#endif // #if !defined(_WIN64) && defined(_Wp64)
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_FIX_INTERLOCKED_EXCHANGE_POINTER_WINDOWS_H_
diff --git a/trunk/src/system_wrappers/interface/list_wrapper.h b/trunk/src/system_wrappers/interface/list_wrapper.h
deleted file mode 100644
index 3608ada..0000000
--- a/trunk/src/system_wrappers/interface/list_wrapper.h
+++ /dev/null
@@ -1,107 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_LIST_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_LIST_WRAPPER_H_
-
-#include "constructor_magic.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class ListItem
-{
-friend class ListWrapper;
-
-public:
- ListItem(const void* ptr);
- ListItem(const unsigned int item);
- virtual ~ListItem();
- void* GetItem() const;
- unsigned int GetUnsignedItem() const;
-
-protected:
- ListItem* next_;
- ListItem* prev_;
-
-private:
- const void* item_ptr_;
- const unsigned int item_;
-};
-
-class ListWrapper
-{
-public:
- ListWrapper();
- virtual ~ListWrapper();
-
- // Returns the number of elements stored in the list.
- unsigned int GetSize() const;
-
- // Puts a pointer to anything last in the list.
- int PushBack(const void* ptr);
- // Puts a pointer to anything first in the list.
- int PushFront(const void* ptr);
-
- // Puts a copy of the specified integer last in the list.
- int PushBack(const unsigned int item_id);
- // Puts a copy of the specified integer first in the list.
- int PushFront(const unsigned int item_id);
-
- // Pops the first ListItem from the list
- int PopFront();
-
- // Pops the last ListItem from the list
- int PopBack();
-
- // Returns true if the list is empty
- bool Empty() const;
-
- // Returns a pointer to the first ListItem in the list.
- ListItem* First() const;
-
- // Returns a pointer to the last ListItem in the list.
- ListItem* Last() const;
-
- // Returns a pointer to the ListItem stored after item in the list.
- ListItem* Next(ListItem* item) const;
-
- // Returns a pointer to the ListItem stored before item in the list.
- ListItem* Previous(ListItem* item) const;
-
- // Removes item from the list.
- int Erase(ListItem* item);
-
- // Insert list item after existing_previous_item. Please note that new_item
- // must be created using new ListItem(). The map will take ownership of
- // new_item following a successfull insert. If insert fails new_item will
- // not be released by the List
- int Insert(ListItem* existing_previous_item,
- ListItem* new_item);
-
- // Insert list item before existing_next_item. Please note that new_item
- // must be created using new ListItem(). The map will take ownership of
- // new_item following a successfull insert. If insert fails new_item will
- // not be released by the List
- int InsertBefore(ListItem* existing_next_item,
- ListItem* new_item);
-
-private:
- void PushBackImpl(ListItem* item);
- void PushFrontImpl(ListItem* item);
-
- CriticalSectionWrapper* critical_section_;
- ListItem* first_;
- ListItem* last_;
- unsigned int size_;
-};
-} //namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_LIST_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/map_wrapper.h b/trunk/src/system_wrappers/interface/map_wrapper.h
deleted file mode 100644
index 7d4e733..0000000
--- a/trunk/src/system_wrappers/interface/map_wrapper.h
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_MAP_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_MAP_WRAPPER_H_
-
-#include <map>
-
-#include "constructor_magic.h"
-
-namespace webrtc {
-class MapItem
-{
-friend class MapWrapper;
-
-public:
- MapItem(int id, void* ptr);
- virtual ~MapItem();
- void* GetItem();
- int GetId();
- unsigned int GetUnsignedId();
- void SetItem(void* ptr);
-
-private:
- int item_id_;
- void* item_pointer_;
-};
-
-class MapWrapper
-{
-public:
- MapWrapper();
- ~MapWrapper();
-
- // Puts a pointer to anything in the map and associates it with id. Note, id
- // needs to be unique for all items in the map.
- int Insert(int id, void* ptr);
-
- // Removes item from map.
- int Erase(MapItem* item);
-
- // Finds item with associated with id and removes it from the map.
- int Erase(int id);
-
- // Returns the number of elements stored in the map.
- int Size() const;
-
- // Returns a pointer to the first MapItem in the map.
- MapItem* First() const;
-
- // Returns a pointer to the last MapItem in the map.
- MapItem* Last() const;
-
- // Returns a pointer to the MapItem stored after item in the map.
- MapItem* Next(MapItem* item) const;
-
- // Returns a pointer to the MapItem stored before item in the map.
- MapItem* Previous(MapItem* item) const;
-
- // Returns a pointer to the MapItem associated with id from the map.
- MapItem* Find(int id) const;
-
-private:
- std::map<int, MapItem*> map_;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_MAP_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/ref_count.h b/trunk/src/system_wrappers/interface/ref_count.h
deleted file mode 100644
index f90b0b3..0000000
--- a/trunk/src/system_wrappers/interface/ref_count.h
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SYSTEM_WRAPPERS_INTERFACE_REF_COUNT_H_
-#define SYSTEM_WRAPPERS_INTERFACE_REF_COUNT_H_
-
-#include "system_wrappers/interface/atomic32_wrapper.h"
-
-namespace webrtc {
-
-// This class can be used for instantiating
-// reference counted objects.
-// int32_t AddRef() and int32_t Release().
-// Usage:
-// RefCountImpl<T>* implementation = new RefCountImpl<T>(p);
-//
-// Example:
-// class MyInterface {
-// public:
-// virtual void DoSomething() = 0;
-// virtual int32_t AddRef() = 0;
-// virtual int32_t Release() = 0:
-// private:
-// virtual ~MyInterface(){};
-// }
-// class MyImplementation : public MyInterface {
-// public:
-// virtual DoSomething() { printf("hello"); };
-// };
-// MyImplementation* CreateMyImplementation() {
-// RefCountImpl<MyImplementation>* implementation =
-// new RefCountImpl<MyImplementation>();
-// return implementation;
-// }
-
-template <class T>
-class RefCountImpl : public T {
- public:
- RefCountImpl() : ref_count_(0) {}
-
- template<typename P>
- explicit RefCountImpl(P p) : T(p), ref_count_(0) {}
-
- template<typename P1, typename P2>
- RefCountImpl(P1 p1, P2 p2) : T(p1, p2), ref_count_(0) {}
-
- template<typename P1, typename P2, typename P3>
- RefCountImpl(P1 p1, P2 p2, P3 p3) : T(p1, p2, p3), ref_count_(0) {}
-
- template<typename P1, typename P2, typename P3, typename P4>
- RefCountImpl(P1 p1, P2 p2, P3 p3, P4 p4) : T(p1, p2, p3, p4), ref_count_(0) {}
-
- template<typename P1, typename P2, typename P3, typename P4, typename P5>
- RefCountImpl(P1 p1, P2 p2, P3 p3, P4 p4, P5 p5)
- : T(p1, p2, p3, p4, p5), ref_count_(0) {}
-
- virtual int32_t AddRef() {
- return ++ref_count_;
- }
-
- virtual int32_t Release() {
- int32_t ref_count;
- ref_count = --ref_count_;
- if (ref_count == 0)
- delete this;
- return ref_count;
- }
-
- protected:
- Atomic32Wrapper ref_count_;
-};
-
-} // namespace webrtc
-
-#endif // SYSTEM_WRAPPERS_INTERFACE_REF_COUNT_H_
diff --git a/trunk/src/system_wrappers/interface/rw_lock_wrapper.h b/trunk/src/system_wrappers/interface/rw_lock_wrapper.h
deleted file mode 100644
index f0842ac..0000000
--- a/trunk/src/system_wrappers/interface/rw_lock_wrapper.h
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_RW_LOCK_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_RW_LOCK_WRAPPER_H_
-
-// Note, Windows pre-Vista version of RW locks are not supported nativly. For
-// these OSs regular critical sections have been used to approximate RW lock
-// functionality and will therefore have worse performance.
-
-namespace webrtc {
-class RWLockWrapper
-{
-public:
- static RWLockWrapper* CreateRWLock();
- virtual ~RWLockWrapper();
-
- virtual void AcquireLockExclusive() = 0;
- virtual void ReleaseLockExclusive() = 0;
-
- virtual void AcquireLockShared() = 0;
- virtual void ReleaseLockShared() = 0;
-
-protected:
- virtual int Init() = 0;
-};
-
-// RAII extensions of the RW lock. Prevents Acquire/Release missmatches and
-// provides more compact locking syntax.
-class ReadLockScoped
-{
-public:
- ReadLockScoped(RWLockWrapper& rwLock)
- :
- _rwLock(rwLock)
- {
- _rwLock.AcquireLockShared();
- }
-
- ~ReadLockScoped()
- {
- _rwLock.ReleaseLockShared();
- }
-
-private:
- RWLockWrapper& _rwLock;
-};
-
-class WriteLockScoped
-{
-public:
- WriteLockScoped(RWLockWrapper& rwLock)
- :
- _rwLock(rwLock)
- {
- _rwLock.AcquireLockExclusive();
- }
-
- ~WriteLockScoped()
- {
- _rwLock.ReleaseLockExclusive();
- }
-
-private:
- RWLockWrapper& _rwLock;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_RW_LOCK_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/scoped_ptr.h b/trunk/src/system_wrappers/interface/scoped_ptr.h
deleted file mode 100644
index 74b6ad3..0000000
--- a/trunk/src/system_wrappers/interface/scoped_ptr.h
+++ /dev/null
@@ -1,258 +0,0 @@
-// (C) Copyright Greg Colvin and Beman Dawes 1998, 1999.
-// Copyright (c) 2001, 2002 Peter Dimov
-//
-// Permission to copy, use, modify, sell and distribute this software
-// is granted provided this copyright notice appears in all copies.
-// This software is provided "as is" without express or implied
-// warranty, and with no claim as to its suitability for any purpose.
-//
-// See http://www.boost.org/libs/smart_ptr/scoped_ptr.htm for documentation.
-//
-
-// scoped_ptr mimics a built-in pointer except that it guarantees deletion
-// of the object pointed to, either on destruction of the scoped_ptr or via
-// an explicit reset(). scoped_ptr is a simple solution for simple needs;
-// use shared_ptr or std::auto_ptr if your needs are more complex.
-
-// scoped_ptr_malloc added in by Google. When one of
-// these goes out of scope, instead of doing a delete or delete[], it
-// calls free(). scoped_ptr_malloc<char> is likely to see much more
-// use than any other specializations.
-
-// release() added in by Google. Use this to conditionally
-// transfer ownership of a heap-allocated object to the caller, usually on
-// method success.
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SCOPED_PTR_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SCOPED_PTR_H_
-
-#include <assert.h> // for assert
-#include <stdlib.h> // for free() decl
-
-#include <cstddef> // for std::ptrdiff_t
-
-#ifdef _WIN32
-namespace std { using ::ptrdiff_t; };
-#endif // _WIN32
-
-namespace webrtc {
-
-template <typename T>
-class scoped_ptr {
- private:
-
- T* ptr;
-
- scoped_ptr(scoped_ptr const &);
- scoped_ptr & operator=(scoped_ptr const &);
-
- public:
-
- typedef T element_type;
-
- explicit scoped_ptr(T* p = NULL): ptr(p) {}
-
- ~scoped_ptr() {
- typedef char type_must_be_complete[sizeof(T)];
- delete ptr;
- }
-
- void reset(T* p = NULL) {
- typedef char type_must_be_complete[sizeof(T)];
-
- if (ptr != p) {
- T* obj = ptr;
- ptr = p;
- // Delete last, in case obj destructor indirectly results in ~scoped_ptr
- delete obj;
- }
- }
-
- T& operator*() const {
- assert(ptr != NULL);
- return *ptr;
- }
-
- T* operator->() const {
- assert(ptr != NULL);
- return ptr;
- }
-
- T* get() const {
- return ptr;
- }
-
- void swap(scoped_ptr & b) {
- T* tmp = b.ptr;
- b.ptr = ptr;
- ptr = tmp;
- }
-
- T* release() {
- T* tmp = ptr;
- ptr = NULL;
- return tmp;
- }
-
- T** accept() {
- if (ptr) {
- delete ptr;
- ptr = NULL;
- }
- return &ptr;
- }
-
- T** use() {
- return &ptr;
- }
-};
-
-template<typename T> inline
-void swap(scoped_ptr<T>& a, scoped_ptr<T>& b) {
- a.swap(b);
-}
-
-
-
-
-// scoped_array extends scoped_ptr to arrays. Deletion of the array pointed to
-// is guaranteed, either on destruction of the scoped_array or via an explicit
-// reset(). Use shared_array or std::vector if your needs are more complex.
-
-template<typename T>
-class scoped_array {
- private:
-
- T* ptr;
-
- scoped_array(scoped_array const &);
- scoped_array & operator=(scoped_array const &);
-
- public:
-
- typedef T element_type;
-
- explicit scoped_array(T* p = NULL) : ptr(p) {}
-
- ~scoped_array() {
- typedef char type_must_be_complete[sizeof(T)];
- delete[] ptr;
- }
-
- void reset(T* p = NULL) {
- typedef char type_must_be_complete[sizeof(T)];
-
- if (ptr != p) {
- T* arr = ptr;
- ptr = p;
- // Delete last, in case arr destructor indirectly results in ~scoped_array
- delete [] arr;
- }
- }
-
- T& operator[](std::ptrdiff_t i) const {
- assert(ptr != NULL);
- assert(i >= 0);
- return ptr[i];
- }
-
- T* get() const {
- return ptr;
- }
-
- void swap(scoped_array & b) {
- T* tmp = b.ptr;
- b.ptr = ptr;
- ptr = tmp;
- }
-
- T* release() {
- T* tmp = ptr;
- ptr = NULL;
- return tmp;
- }
-
- T** accept() {
- if (ptr) {
- delete [] ptr;
- ptr = NULL;
- }
- return &ptr;
- }
-};
-
-template<class T> inline
-void swap(scoped_array<T>& a, scoped_array<T>& b) {
- a.swap(b);
-}
-
-// scoped_ptr_malloc<> is similar to scoped_ptr<>, but it accepts a
-// second template argument, the function used to free the object.
-
-template<typename T, void (*FF)(void*) = free> class scoped_ptr_malloc {
- private:
-
- T* ptr;
-
- scoped_ptr_malloc(scoped_ptr_malloc const &);
- scoped_ptr_malloc & operator=(scoped_ptr_malloc const &);
-
- public:
-
- typedef T element_type;
-
- explicit scoped_ptr_malloc(T* p = 0): ptr(p) {}
-
- ~scoped_ptr_malloc() {
- FF(static_cast<void*>(ptr));
- }
-
- void reset(T* p = 0) {
- if (ptr != p) {
- FF(static_cast<void*>(ptr));
- ptr = p;
- }
- }
-
- T& operator*() const {
- assert(ptr != 0);
- return *ptr;
- }
-
- T* operator->() const {
- assert(ptr != 0);
- return ptr;
- }
-
- T* get() const {
- return ptr;
- }
-
- void swap(scoped_ptr_malloc & b) {
- T* tmp = b.ptr;
- b.ptr = ptr;
- ptr = tmp;
- }
-
- T* release() {
- T* tmp = ptr;
- ptr = 0;
- return tmp;
- }
-
- T** accept() {
- if (ptr) {
- FF(static_cast<void*>(ptr));
- ptr = 0;
- }
- return &ptr;
- }
-};
-
-template<typename T, void (*FF)(void*)> inline
-void swap(scoped_ptr_malloc<T,FF>& a, scoped_ptr_malloc<T,FF>& b) {
- a.swap(b);
-}
-
-} // namespace webrtc
-
-#endif // #ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SCOPED_PTR_H_
diff --git a/trunk/src/system_wrappers/interface/scoped_refptr.h b/trunk/src/system_wrappers/interface/scoped_refptr.h
deleted file mode 100644
index 0df15be..0000000
--- a/trunk/src/system_wrappers/interface/scoped_refptr.h
+++ /dev/null
@@ -1,137 +0,0 @@
-// Copyright (c) 2011 The Chromium Authors. All rights reserved.
-// Use of this source code is governed by a BSD-style license that can be
-// found in the LICENSE file under third_party_mods/chromium or at:
-// http://src.chromium.org/svn/trunk/src/LICENSE
-
-#ifndef SYSTEM_WRAPPERS_INTERFACE_SCOPED_REFPTR_H_
-#define SYSTEM_WRAPPERS_INTERFACE_SCOPED_REFPTR_H_
-
-namespace webrtc {
-
-// Extracted from Chromium's src/base/memory/ref_counted.h.
-
-//
-// A smart pointer class for reference counted objects. Use this class instead
-// of calling AddRef and Release manually on a reference counted object to
-// avoid common memory leaks caused by forgetting to Release an object
-// reference. Sample usage:
-//
-// class MyFoo : public RefCounted<MyFoo> {
-// ...
-// };
-//
-// void some_function() {
-// scoped_refptr<MyFoo> foo = new MyFoo();
-// foo->Method(param);
-// // |foo| is released when this function returns
-// }
-//
-// void some_other_function() {
-// scoped_refptr<MyFoo> foo = new MyFoo();
-// ...
-// foo = NULL; // explicitly releases |foo|
-// ...
-// if (foo)
-// foo->Method(param);
-// }
-//
-// The above examples show how scoped_refptr<T> acts like a pointer to T.
-// Given two scoped_refptr<T> classes, it is also possible to exchange
-// references between the two objects, like so:
-//
-// {
-// scoped_refptr<MyFoo> a = new MyFoo();
-// scoped_refptr<MyFoo> b;
-//
-// b.swap(a);
-// // now, |b| references the MyFoo object, and |a| references NULL.
-// }
-//
-// To make both |a| and |b| in the above example reference the same MyFoo
-// object, simply use the assignment operator:
-//
-// {
-// scoped_refptr<MyFoo> a = new MyFoo();
-// scoped_refptr<MyFoo> b;
-//
-// b = a;
-// // now, |a| and |b| each own a reference to the same MyFoo object.
-// }
-//
-template <class T>
-class scoped_refptr {
- public:
- scoped_refptr() : ptr_(NULL) {
- }
-
- scoped_refptr(T* p) : ptr_(p) {
- if (ptr_)
- ptr_->AddRef();
- }
-
- scoped_refptr(const scoped_refptr<T>& r) : ptr_(r.ptr_) {
- if (ptr_)
- ptr_->AddRef();
- }
-
- template <typename U>
- scoped_refptr(const scoped_refptr<U>& r) : ptr_(r.get()) {
- if (ptr_)
- ptr_->AddRef();
- }
-
- ~scoped_refptr() {
- if (ptr_)
- ptr_->Release();
- }
-
- T* get() const { return ptr_; }
- operator T*() const { return ptr_; }
- T* operator->() const { return ptr_; }
-
- // Release a pointer.
- // The return value is the current pointer held by this object.
- // If this object holds a NULL pointer, the return value is NULL.
- // After this operation, this object will hold a NULL pointer,
- // and will not own the object any more.
- T* release() {
- T* retVal = ptr_;
- ptr_ = NULL;
- return retVal;
- }
-
- scoped_refptr<T>& operator=(T* p) {
- // AddRef first so that self assignment should work
- if (p)
- p->AddRef();
- if (ptr_ )
- ptr_->Release();
- ptr_ = p;
- return *this;
- }
-
- scoped_refptr<T>& operator=(const scoped_refptr<T>& r) {
- return *this = r.ptr_;
- }
-
- template <typename U>
- scoped_refptr<T>& operator=(const scoped_refptr<U>& r) {
- return *this = r.get();
- }
-
- void swap(T** pp) {
- T* p = ptr_;
- ptr_ = *pp;
- *pp = p;
- }
-
- void swap(scoped_refptr<T>& r) {
- swap(&r.ptr_);
- }
-
- protected:
- T* ptr_;
-};
-} // namespace webrtc
-
-#endif // SYSTEM_WRAPPERS_INTERFACE_SCOPED_REFPTR_H_
diff --git a/trunk/src/system_wrappers/interface/sort.h b/trunk/src/system_wrappers/interface/sort.h
deleted file mode 100644
index fb25ecf..0000000
--- a/trunk/src/system_wrappers/interface/sort.h
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Generic unstable sorting routines.
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SORT_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SORT_H_
-
-#include "typedefs.h"
-#include "common_types.h"
-
-namespace webrtc
-{
- enum Type
- {
- TYPE_Word8,
- TYPE_UWord8,
- TYPE_Word16,
- TYPE_UWord16,
- TYPE_Word32,
- TYPE_UWord32,
- TYPE_Word64,
- TYPE_UWord64,
- TYPE_Float32,
- TYPE_Float64
- };
- // Sorts intrinsic data types.
- //
- // data [in/out] A pointer to an array of intrinsic type.
- // Upon return it will be sorted in ascending order.
- // numOfElements The number of elements in the array.
- // dataType Enum corresponding to the type of the array.
- //
- // returns 0 on success, -1 on failure.
- WebRtc_Word32 Sort(void* data, WebRtc_UWord32 numOfElements, Type dataType);
-
- // Sorts arbitrary data types. This requires an array of intrinsically typed
- // key values which will be used to sort the data array. There must be a
- // one-to-one correspondence between data elements and key elements, with
- // corresponding elements sharing the same position in their respective
- // arrays.
- //
- // data [in/out] A pointer to an array of arbitrary type.
- // Upon return it will be sorted in ascending order.
- // key [in] A pointer to an array of keys used to sort the
- // data array.
- // numOfElements The number of elements in the arrays.
- // sizeOfElement The size, in bytes, of the data array.
- // keyType Enum corresponding to the type of the key array.
- //
- // returns 0 on success, -1 on failure.
- //
- WebRtc_Word32 KeySort(void* data, void* key, WebRtc_UWord32 numOfElements,
- WebRtc_UWord32 sizeOfElement, Type keyType);
-}
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_SORT_H_
diff --git a/trunk/src/system_wrappers/interface/static_instance.h b/trunk/src/system_wrappers/interface/static_instance.h
deleted file mode 100644
index 8fe91cc..0000000
--- a/trunk/src/system_wrappers/interface/static_instance.h
+++ /dev/null
@@ -1,155 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STATICINSTANCETEMPLATE_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STATICINSTANCETEMPLATE_H_
-
-#include <assert.h>
-
-#include "critical_section_wrapper.h"
-#ifdef _WIN32
-#include "fix_interlocked_exchange_pointer_win.h"
-#endif
-
-namespace webrtc {
-
-enum CountOperation {
- kRelease,
- kAddRef,
- kAddRefNoCreate
-};
-enum CreateOperation {
- kInstanceExists,
- kCreate,
- kDestroy
-};
-
-template <class T>
-// Construct On First Use idiom. Avoids
-// "static initialization order fiasco".
-static T* GetStaticInstance(CountOperation count_operation) {
- // TODO (hellner): use atomic wrapper instead.
- static volatile long instance_count = 0;
- static T* volatile instance = NULL;
- CreateOperation state = kInstanceExists;
-#ifndef _WIN32
- // This memory is staticly allocated once. The application does not try to
- // free this memory. This approach is taken to avoid issues with
- // destruction order for statically allocated memory. The memory will be
- // reclaimed by the OS and memory leak tools will not recognize memory
- // reachable from statics leaked so no noise is added by doing this.
- static CriticalSectionWrapper* crit_sect(
- CriticalSectionWrapper::CreateCriticalSection());
- CriticalSectionScoped lock(crit_sect);
-
- if (count_operation ==
- kAddRefNoCreate && instance_count == 0) {
- return NULL;
- }
- if (count_operation ==
- kAddRef ||
- count_operation == kAddRefNoCreate) {
- instance_count++;
- if (instance_count == 1) {
- state = kCreate;
- }
- } else {
- instance_count--;
- if (instance_count == 0) {
- state = kDestroy;
- }
- }
- if (state == kCreate) {
- instance = T::CreateInstance();
- } else if (state == kDestroy) {
- T* old_instance = instance;
- instance = NULL;
- // The state will not change past this point. Release the critical
- // section while deleting the object in case it would be blocking on
- // access back to this object. (This is the case for the tracing class
- // since the thread owned by the tracing class also traces).
- // TODO(hellner): this is a bit out of place but here goes, de-couple
- // thread implementation with trace implementation.
- crit_sect->Leave();
- if (old_instance) {
- delete old_instance;
- }
- // Re-acquire the lock since the scoped critical section will release
- // it.
- crit_sect->Enter();
- return NULL;
- }
-#else // _WIN32
- if (count_operation ==
- kAddRefNoCreate && instance_count == 0) {
- return NULL;
- }
- if (count_operation == kAddRefNoCreate) {
- if (1 == InterlockedIncrement(&instance_count)) {
- // The instance has been destroyed by some other thread. Rollback.
- InterlockedDecrement(&instance_count);
- assert(false);
- return NULL;
- }
- // Sanity to catch corrupt state.
- if (instance == NULL) {
- assert(false);
- InterlockedDecrement(&instance_count);
- return NULL;
- }
- } else if (count_operation == kAddRef) {
- if (instance_count == 0) {
- state = kCreate;
- } else {
- if (1 == InterlockedIncrement(&instance_count)) {
- // InterlockedDecrement because reference count should not be
- // updated just yet (that's done when the instance is created).
- InterlockedDecrement(&instance_count);
- state = kCreate;
- }
- }
- } else {
- int newValue = InterlockedDecrement(&instance_count);
- if (newValue == 0) {
- state = kDestroy;
- }
- }
-
- if (state == kCreate) {
- // Create instance and let whichever thread finishes first assign its
- // local copy to the global instance. All other threads reclaim their
- // local copy.
- T* new_instance = T::CreateInstance();
- if (1 == InterlockedIncrement(&instance_count)) {
- T* old_value = static_cast<T*> (InterlockedExchangePointer(
- reinterpret_cast<void* volatile*>(&instance), new_instance));
- assert(old_value == NULL);
- assert(instance);
- } else {
- InterlockedDecrement(&instance_count);
- if (new_instance) {
- delete static_cast<T*>(new_instance);
- }
- }
- } else if (state == kDestroy) {
- T* old_value = static_cast<T*> (InterlockedExchangePointer(
- reinterpret_cast<void* volatile*>(&instance), NULL));
- if (old_value) {
- delete static_cast<T*>(old_value);
- }
- return NULL;
- }
-#endif // #ifndef _WIN32
- return instance;
-}
-
-} // namspace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_STATICINSTANCETEMPLATE_H_
diff --git a/trunk/src/system_wrappers/interface/thread_wrapper.h b/trunk/src/system_wrappers/interface/thread_wrapper.h
deleted file mode 100644
index 030ac8a..0000000
--- a/trunk/src/system_wrappers/interface/thread_wrapper.h
+++ /dev/null
@@ -1,95 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// System independant wrapper for spawning threads
-// Note: the spawned thread will loop over the callback function until stopped.
-// Note: The callback function is expected to return every 2 seconds or more
-// often.
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_THREAD_WRAPPER_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_THREAD_WRAPPER_H_
-
-#include "common_types.h"
-#include "typedefs.h"
-
-namespace webrtc {
-// Object that will be passed by the spawned thread when it enters the callback
-// function.
-#define ThreadObj void*
-
-// Callback function that the spawned thread will enter once spawned.
-// A return value of false is interpreted as that the function has no
-// more work to do and that the thread can be released.
-typedef bool(*ThreadRunFunction)(ThreadObj);
-
-enum ThreadPriority
-{
- kLowPriority = 1,
- kNormalPriority = 2,
- kHighPriority = 3,
- kHighestPriority = 4,
- kRealtimePriority = 5
-};
-
-class ThreadWrapper
-{
-public:
- enum {kThreadMaxNameLength = 64};
-
- virtual ~ThreadWrapper() {};
-
- // Factory method. Constructor disabled.
- //
- // func Pointer to a, by user, specified callback function.
- // obj Object associated with the thread. Passed in the callback
- // function.
- // prio Thread priority. May require root/admin rights.
- // threadName NULL terminated thread name, will be visable in the Windows
- // debugger.
- static ThreadWrapper* CreateThread(ThreadRunFunction func = 0,
- ThreadObj obj= 0,
- ThreadPriority prio = kNormalPriority,
- const char* threadName = 0);
-
- // Get the current thread's kernel thread ID.
- static uint32_t GetThreadId();
-
- // Non blocking termination of the spawned thread. Note that it is not safe
- // to delete this class until the spawned thread has been reclaimed.
- virtual void SetNotAlive() = 0;
-
- // Spawns the thread. This will start the triggering of the callback
- // function.
- virtual bool Start(unsigned int& id) = 0;
-
- // Sets the threads CPU affinity. CPUs are listed 0 - (number of CPUs - 1).
- // The numbers in processorNumbers specify which CPUs are allowed to run the
- // thread. processorNumbers should not contain any duplicates and elements
- // should be lower than (number of CPUs - 1). amountOfProcessors should be
- // equal to the number of processors listed in processorNumbers
- virtual bool SetAffinity(const int* /*processorNumbers*/,
- const unsigned int /*amountOfProcessors*/) {
- return false;
- }
-
- // Stops the spawned thread and waits for it to be reclaimed with a timeout
- // of two seconds. Will return false if the thread was not reclaimed.
- // Multiple tries to Stop are allowed (e.g. to wait longer than 2 seconds).
- // It's ok to call Stop() even if the spawned thread has been reclaimed.
- virtual bool Stop() = 0;
-
- // Stops the spawned thread dead in its tracks. Will likely result in a
- // corrupt state. There should be an extremely good reason for even looking
- // at this function. Can cause many problems deadlock being one of them.
- virtual bool Shutdown() {return false;}
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_THREAD_WRAPPER_H_
diff --git a/trunk/src/system_wrappers/interface/tick_util.h b/trunk/src/system_wrappers/interface/tick_util.h
deleted file mode 100644
index e78e53d..0000000
--- a/trunk/src/system_wrappers/interface/tick_util.h
+++ /dev/null
@@ -1,325 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// System independant wrapper for polling elapsed time in ms and us.
-// The implementation works in the tick domain which can be mapped over to the
-// time domain.
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TICK_UTIL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TICK_UTIL_H_
-
-#if _WIN32
-#include <windows.h>
-#include <mmsystem.h>
-#elif WEBRTC_LINUX
-#include <ctime>
-#else
-#include <sys/time.h>
-#include <time.h>
-#endif
-
-#include "typedefs.h"
-
-namespace webrtc {
-class TickInterval;
-
-class TickTime
-{
-public:
- // Current time in the tick domain.
- static TickTime Now();
-
- // Now in the time domain in ms.
- static WebRtc_Word64 MillisecondTimestamp();
-
- // Now in the time domain in us.
- static WebRtc_Word64 MicrosecondTimestamp();
-
- WebRtc_Word64 Ticks() const;
-
- static WebRtc_Word64 MillisecondsToTicks(const WebRtc_Word64 ms);
-
- static WebRtc_Word64 TicksToMilliseconds(const WebRtc_Word64 ticks);
-
- // Returns a TickTime that is ticks later than the passed TickTime
- friend TickTime operator+(const TickTime lhs, const WebRtc_Word64 ticks);
- TickTime& operator+=(const WebRtc_Word64& rhs);
-
-
- // Returns a TickInterval that is the difference in ticks beween rhs and lhs
- friend TickInterval operator-(const TickTime& lhs, const TickTime& rhs);
-private:
- WebRtc_Word64 _ticks;
-};
-
-class TickInterval
-{
-public:
- TickInterval();
-
- WebRtc_Word64 Milliseconds() const;
- WebRtc_Word64 Microseconds() const;
-
- // Returns the sum of two TickIntervals as a TickInterval
- friend TickInterval operator+(const TickInterval& lhs,
- const TickInterval& rhs);
- TickInterval& operator-=(const TickInterval& rhs);
-
- // Returns a TickInterval corresponding to rhs - lhs
- friend TickInterval operator-(const TickInterval& lhs,
- const TickInterval& rhs);
- TickInterval& operator+=(const TickInterval& rhs);
-
- friend bool operator>(const TickInterval& lhs, const TickInterval& rhs);
- friend bool operator<=(const TickInterval& lhs, const TickInterval& rhs);
- friend bool operator<(const TickInterval& lhs, const TickInterval& rhs);
- friend bool operator>=(const TickInterval& lhs, const TickInterval& rhs);
-
-private:
- TickInterval(WebRtc_Word64 interval);
-
- friend class TickTime;
- friend TickInterval operator-(const TickTime& lhs, const TickTime& rhs);
-
-private:
- WebRtc_Word64 _interval;
-};
-
-inline TickInterval operator+(const TickInterval& lhs, const TickInterval& rhs)
-{
- return TickInterval(lhs._interval + rhs._interval);
-}
-
-inline TickInterval operator-(const TickInterval& lhs, const TickInterval& rhs)
-{
- return TickInterval(lhs._interval - rhs._interval);
-}
-
-inline TickInterval operator-(const TickTime& lhs,const TickTime& rhs)
-{
- return TickInterval(lhs._ticks - rhs._ticks);
-}
-
-inline TickTime operator+(const TickTime lhs, const WebRtc_Word64 ticks)
-{
- TickTime time = lhs;
- time._ticks += ticks;
- return time;
-}
-inline bool operator>(const TickInterval& lhs, const TickInterval& rhs)
-{
- return lhs._interval > rhs._interval;
-}
-inline bool operator<=(const TickInterval& lhs, const TickInterval& rhs)
-{
- return lhs._interval <= rhs._interval;
-}
-inline bool operator<(const TickInterval& lhs, const TickInterval& rhs)
-{
- return lhs._interval <= rhs._interval;
-}
-inline bool operator>=(const TickInterval& lhs, const TickInterval& rhs)
-{
- return lhs._interval >= rhs._interval;
-}
-
-inline TickTime TickTime::Now()
-{
- TickTime result;
-#if _WIN32
- #ifdef USE_QUERY_PERFORMANCE_COUNTER
- // QueryPerformanceCounter returns the value from the TSC which is
- // incremented at the CPU frequency. The algorithm used requires
- // the CPU frequency to be constant. Technology like speed stepping
- // which has variable CPU frequency will therefore yield unpredictable,
- // incorrect time estimations.
- LARGE_INTEGER qpcnt;
- QueryPerformanceCounter(&qpcnt);
- result._ticks = qpcnt.QuadPart;
- #else
- static volatile LONG lastTimeGetTime = 0;
- static volatile WebRtc_Word64 numWrapTimeGetTime = 0;
- volatile LONG* lastTimeGetTimePtr = &lastTimeGetTime;
- DWORD now = timeGetTime();
- // Atomically update the last gotten time
- DWORD old = InterlockedExchange(lastTimeGetTimePtr, now);
- if(now < old)
- {
- // If now is earlier than old, there may have been a race between
- // threads.
- // 0x0fffffff ~3.1 days, the code will not take that long to execute
- // so it must have been a wrap around.
- if(old > 0xf0000000 && now < 0x0fffffff)
- {
- numWrapTimeGetTime++;
- }
- }
- result._ticks = now + (numWrapTimeGetTime<<32);
- #endif
-#elif defined(WEBRTC_LINUX)
- struct timespec ts;
- #ifdef WEBRTC_CLOCK_TYPE_REALTIME
- clock_gettime(CLOCK_REALTIME, &ts);
- #else
- clock_gettime(CLOCK_MONOTONIC, &ts);
- #endif
- result._ticks = 1000000000LL * static_cast<WebRtc_Word64>(ts.tv_sec) + static_cast<WebRtc_Word64>(ts.tv_nsec);
-#else
- struct timeval tv;
- gettimeofday(&tv, NULL);
- result._ticks = 1000000LL * static_cast<WebRtc_Word64>(tv.tv_sec) + static_cast<WebRtc_Word64>(tv.tv_usec);
-#endif
- return result;
-}
-
-inline WebRtc_Word64 TickTime::MillisecondTimestamp()
-{
- TickTime now = TickTime::Now();
-#if _WIN32
- #ifdef USE_QUERY_PERFORMANCE_COUNTER
- LARGE_INTEGER qpfreq;
- QueryPerformanceFrequency(&qpfreq);
- return (now._ticks * 1000) / qpfreq.QuadPart;
- #else
- return now._ticks;
- #endif
-#elif WEBRTC_LINUX
- return now._ticks / 1000000LL;
-#else
- return now._ticks / 1000LL;
-#endif
-}
-
-inline WebRtc_Word64 TickTime::MicrosecondTimestamp()
-{
- TickTime now = TickTime::Now();
-
-#if _WIN32
- #ifdef USE_QUERY_PERFORMANCE_COUNTER
- LARGE_INTEGER qpfreq;
- QueryPerformanceFrequency(&qpfreq);
- return (now._ticks * 1000) / (qpfreq.QuadPart/1000);
- #else
- return now._ticks *1000LL;
- #endif
-#elif WEBRTC_LINUX
- return now._ticks / 1000LL;
-#else
- return now._ticks;
-#endif
-}
-
-inline WebRtc_Word64 TickTime::Ticks() const
-{
- return _ticks;
-}
-
-inline WebRtc_Word64 TickTime::MillisecondsToTicks(const WebRtc_Word64 ms)
-{
-#if _WIN32
- #ifdef USE_QUERY_PERFORMANCE_COUNTER
- LARGE_INTEGER qpfreq;
- QueryPerformanceFrequency(&qpfreq);
- return (qpfreq.QuadPart * ms) / 1000;
- #else
- return ms;
- #endif
-#elif WEBRTC_LINUX
- return ms * 1000000LL;
-#else
- return ms * 1000LL;
-#endif
-}
-
-inline WebRtc_Word64 TickTime::TicksToMilliseconds(const WebRtc_Word64 ticks)
-{
-#if _WIN32
- #ifdef USE_QUERY_PERFORMANCE_COUNTER
- LARGE_INTEGER qpfreq;
- QueryPerformanceFrequency(&qpfreq);
- return (ticks * 1000) / qpfreq.QuadPart;
- #else
- return ticks;
- #endif
-#elif WEBRTC_LINUX
- return ticks / 1000000LL;
-#else
- return ticks / 1000LL;
-#endif
-}
-
-inline TickTime& TickTime::operator+=(const WebRtc_Word64& ticks)
-{
- _ticks += ticks;
- return *this;
-}
-
-inline TickInterval::TickInterval() : _interval(0)
-{
-}
-
-inline TickInterval::TickInterval(const WebRtc_Word64 interval)
- : _interval(interval)
-{
-}
-
-inline WebRtc_Word64 TickInterval::Milliseconds() const
-{
-#if _WIN32
- #ifdef USE_QUERY_PERFORMANCE_COUNTER
- LARGE_INTEGER qpfreq;
- QueryPerformanceFrequency(&qpfreq);
- return (_interval * 1000) / qpfreq.QuadPart;
- #else
- // _interval is in ms
- return _interval;
- #endif
-#elif WEBRTC_LINUX
- // _interval is in ns
- return _interval / 1000000;
-#else
- // _interval is usecs
- return _interval / 1000;
-#endif
-}
-
-inline WebRtc_Word64 TickInterval::Microseconds() const
-{
-#if _WIN32
- #ifdef USE_QUERY_PERFORMANCE_COUNTER
- LARGE_INTEGER qpfreq;
- QueryPerformanceFrequency(&qpfreq);
- return (_interval * 1000000) / qpfreq.QuadPart;
- #else
- // _interval is in ms
- return _interval *1000LL;
- #endif
-#elif WEBRTC_LINUX
- // _interval is in ns
- return _interval / 1000;
-#else
- // _interval is usecs
- return _interval;
-#endif
-}
-
-inline TickInterval& TickInterval::operator+=(const TickInterval& rhs)
-{
- _interval += rhs._interval;
- return *this;
-}
-
-inline TickInterval& TickInterval::operator-=(const TickInterval& rhs)
-{
- _interval -= rhs._interval;
- return *this;
-}
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TICK_UTIL_H_
diff --git a/trunk/src/system_wrappers/interface/trace.h b/trunk/src/system_wrappers/interface/trace.h
deleted file mode 100644
index 0333e76..0000000
--- a/trunk/src/system_wrappers/interface/trace.h
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// System independant wrapper for logging runtime information to file.
-// Note: All log messages will be written to the same trace file.
-// Note: If to many messages are written to file there will be a build up of
-// messages. Apply filtering to avoid that.
-#ifndef WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TRACE_H_
-#define WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TRACE_H_
-
-#include "common_types.h"
-#include "typedefs.h"
-
-#define WEBRTC_TRACE Trace::Add
-
-namespace webrtc {
-class Trace
-{
-public:
-
- // Increments the reference count to the trace.
- static void CreateTrace();
- // Decrements the reference count to the trace.
- static void ReturnTrace();
- // Note: any instance that writes to the trace file should increment and
- // decrement the reference count on construction and destruction
- // respectively
-
- // Specifies what type of messages should be written to the trace file. The
- // filter parameter is a bitmask where each message type is enumerated by
- // the TraceLevel enumerator. TODO (hellner) why is the
- // TraceLevel enumerator not defined in this file?
- static WebRtc_Word32 SetLevelFilter(const WebRtc_UWord32 filter);
-
- // Returns what type of messages are written to the trace file.
- static WebRtc_Word32 LevelFilter(WebRtc_UWord32& filter);
-
- // Sets the file name. If addFileCounter is false the same file will be
- // reused when it fills up. If it's true a new file with incremented name
- // will be used.
- static WebRtc_Word32 SetTraceFile(const char* fileName,
- const bool addFileCounter = false);
-
- // Returns the name of the file that the trace is currently writing to.
- static WebRtc_Word32 TraceFile(char fileName[1024]);
-
- // Registers callback to receive trace messages. TODO (hellner)
- // why not use OutStream instead? Why is TraceCallback not defined in this
- // file
- static WebRtc_Word32 SetTraceCallback(TraceCallback* callback);
-
- // Adds a trace message for writing to file. The message is put in a queue
- // for writing to file whenever possible for performance reasons. I.e. there
- // is a crash it is possible that the last, vital logs are not logged yet.
- // level is the the type of message to log. If that type of messages is
- // filtered it will not be written to file. module is an identifier for what
- // part of the code the message is comming.
- // id is an identifier that should be unique for that set of classes that
- // are associated (e.g. all instances owned by an engine).
- // msg and the elipsis are the same as e.g. sprintf.
- // TODO (hellner) Why is TraceModule not defined in this file?
- static void Add(const TraceLevel level,
- const TraceModule module,
- const WebRtc_Word32 id,
- const char* msg, ...);
-
-};
-} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_INTERFACE_TRACE_H_
diff --git a/trunk/src/system_wrappers/source/Android.mk b/trunk/src/system_wrappers/source/Android.mk
deleted file mode 100644
index aea9b32..0000000
--- a/trunk/src/system_wrappers/source/Android.mk
+++ /dev/null
@@ -1,60 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE := libwebrtc_system_wrappers
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := \
- map.cc \
- sort.cc \
- aligned_malloc.cc \
- atomic32.cc \
- condition_variable.cc \
- cpu_no_op.cc \
- cpu_features.cc \
- cpu_features_arm.c \
- cpu_info.cc \
- critical_section.cc \
- event.cc \
- file_impl.cc \
- list_no_stl.cc \
- rw_lock.cc \
- thread.cc \
- trace_impl.cc \
- condition_variable_posix.cc \
- cpu_linux.cc \
- critical_section_posix.cc \
- event_posix.cc \
- thread_posix.cc \
- trace_posix.cc \
- rw_lock_posix.cc
-
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../.. \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/spreadsortlib
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/system_wrappers/source/aligned_malloc.cc b/trunk/src/system_wrappers/source/aligned_malloc.cc
deleted file mode 100644
index bb10c6b..0000000
--- a/trunk/src/system_wrappers/source/aligned_malloc.cc
+++ /dev/null
@@ -1,121 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "aligned_malloc.h"
-
-#include <assert.h>
-#include <memory.h>
-
-#ifdef WEBRTC_ANDROID
-#include <stdlib.h>
-#endif
-
-#if WEBRTC_MAC
- #include <malloc/malloc.h>
-#else
- #include <malloc.h>
-#endif
-
-#if _WIN32
- #include <windows.h>
-#else
- #include <stdint.h>
-#endif
-
-#include "typedefs.h"
-
-// Ok reference on memory alignment:
-// http://stackoverflow.com/questions/227897/solve-the-memory-alignment-in-c-interview-question-that-stumped-me
-
-namespace webrtc
-{
-// TODO (hellner) better to create just one memory block and
-// interpret the first sizeof(AlignedMemory) bytes as
-// an AlignedMemory struct.
-struct AlignedMemory
-{
- void* alignedBuffer;
- void* memoryPointer;
-};
-
-void* AlignedMalloc(size_t size, size_t alignment)
-{
- if(alignment == 0)
- {
- // Don't allow alignment 0 since it's undefined.
- return NULL;
- }
- // Make sure that the alignment is an integer power of two or fail.
- if(alignment & (alignment - 1))
- {
- return NULL;
- }
-
- AlignedMemory* returnValue = new AlignedMemory();
- if(returnValue == NULL)
- {
- return NULL;
- }
-
- // The memory is aligned towards the lowest address that so only
- // alignment - 1 bytes needs to be allocated.
- // A pointer to AlignedMemory must be stored so that it can be retreived for
- // deletion, ergo the sizeof(uintptr_t).
- returnValue->memoryPointer = malloc(size + sizeof(uintptr_t) +
- alignment - 1);
- if(returnValue->memoryPointer == NULL)
- {
- delete returnValue;
- return NULL;
- }
-
- // Alligning after the sizeof(header) bytes will leave room for the header
- // in the same memory block.
- uintptr_t alignStartPos = (uintptr_t)returnValue->memoryPointer;
- alignStartPos += sizeof(uintptr_t);
-
- // The buffer should be aligned with 'alignment' bytes. The - 1 guarantees
- // that we align towards the lowest address.
- uintptr_t alignedPos = (alignStartPos + alignment - 1) & ~(alignment - 1);
-
- // alignedPos is the address sought for.
- returnValue->alignedBuffer = (void*)alignedPos;
-
- // Store the address to the AlignedMemory struct in the header so that a
- // it's possible to reclaim all memory.
- uintptr_t headerPos = alignedPos;
- headerPos -= sizeof(uintptr_t);
- void* headerPtr = (void*) headerPos;
- uintptr_t headerValue = (uintptr_t)returnValue;
- memcpy(headerPtr,&headerValue,sizeof(uintptr_t));
-
- return returnValue->alignedBuffer;
-}
-
-void AlignedFree(void* memBlock)
-{
- if(memBlock == NULL)
- {
- return;
- }
- uintptr_t alignedPos = (uintptr_t)memBlock;
- uintptr_t headerPos = alignedPos - sizeof(uintptr_t);
-
- // Read out the address of the AlignedMemory struct from the header.
- uintptr_t* headerPtr = (uintptr_t*)headerPos;
- AlignedMemory* deleteMemory = (AlignedMemory*) *headerPtr;
-
- if(deleteMemory->memoryPointer != NULL)
- {
- free(deleteMemory->memoryPointer);
- }
- delete deleteMemory;
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/atomic32.cc b/trunk/src/system_wrappers/source/atomic32.cc
deleted file mode 100644
index 588dd3e..0000000
--- a/trunk/src/system_wrappers/source/atomic32.cc
+++ /dev/null
@@ -1,84 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "atomic32_wrapper.h"
-
-#if defined(_WIN32)
- #include "atomic32_win.h"
-#elif defined(WEBRTC_LINUX)
- #include "atomic32_linux.h"
-#elif defined(WEBRTC_MAC)
- #include "atomic32_mac.h"
-#else
- #error unsupported os!
-#endif
-
-namespace webrtc {
-Atomic32Wrapper::Atomic32Wrapper(WebRtc_Word32 initialValue)
- : _impl(*new Atomic32Impl(initialValue))
-{
-}
-
-Atomic32Wrapper::~Atomic32Wrapper()
-{
- delete &_impl;
-}
-
-WebRtc_Word32 Atomic32Wrapper::operator++()
-{
- return ++_impl;
-}
-
-WebRtc_Word32 Atomic32Wrapper::operator--()
-{
- return --_impl;
-}
-
-// Read and write to properly aligned variables are atomic operations.
-// Ex reference (for Windows): http://msdn.microsoft.com/en-us/library/ms684122(v=VS.85).aspx
-// TODO (hellner) operator= and Atomic32Wrapper::Value() can be fully
-// implemented here.
-Atomic32Wrapper& Atomic32Wrapper::operator=(const Atomic32Wrapper& rhs)
-{
- if(this == &rhs)
- {
- return *this;
- }
- _impl = rhs._impl;
- return *this;
-}
-
-Atomic32Wrapper& Atomic32Wrapper::operator=(WebRtc_Word32 rhs)
-{
- _impl = rhs;
- return *this;
-}
-
-WebRtc_Word32 Atomic32Wrapper::operator+=(WebRtc_Word32 rhs)
-{
- return _impl += rhs;
-}
-
-WebRtc_Word32 Atomic32Wrapper::operator-=(WebRtc_Word32 rhs)
-{
- return _impl -= rhs;
-}
-
-bool Atomic32Wrapper::CompareExchange(WebRtc_Word32 newValue,
- WebRtc_Word32 compareValue)
-{
- return _impl.CompareExchange(newValue,compareValue);
-}
-
-WebRtc_Word32 Atomic32Wrapper::Value() const
-{
- return _impl.Value();
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/atomic32_linux.h b/trunk/src/system_wrappers/source/atomic32_linux.h
deleted file mode 100644
index f9f5650..0000000
--- a/trunk/src/system_wrappers/source/atomic32_linux.h
+++ /dev/null
@@ -1,119 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Atomic system independant 32-bit signed integer.
-// Linux implementation.
-// Note: Requires gcc 4.1.2 or later.
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_LINUX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_LINUX_H_
-
-#include <inttypes.h>
-#include <malloc.h>
-
-#include "common_types.h"
-
-namespace webrtc {
-class Atomic32Impl
-{
-public:
- inline Atomic32Impl(WebRtc_Word32 initialValue);
- inline ~Atomic32Impl();
-
- inline WebRtc_Word32 operator++();
- inline WebRtc_Word32 operator--();
-
- inline Atomic32Impl& operator=(const Atomic32Impl& rhs);
- inline Atomic32Impl& operator=(WebRtc_Word32 rhs);
- inline WebRtc_Word32 operator+=(WebRtc_Word32 rhs);
- inline WebRtc_Word32 operator-=(WebRtc_Word32 rhs);
-
- inline bool CompareExchange(WebRtc_Word32 newValue,
- WebRtc_Word32 compareValue);
-
- inline WebRtc_Word32 Value() const;
-private:
- void* _ptrMemory;
- // Volatile ensures full memory barriers.
- volatile WebRtc_Word32* _value;
-};
-
-// TODO (hellner) use aligned_malloc instead of doing it manually.
-inline Atomic32Impl::Atomic32Impl(WebRtc_Word32 initialValue)
- : _ptrMemory(NULL),
- _value(NULL)
-{ // Align the memory associated with _value on a 32-bit boundary. This is a
- // requirement for the used Linux APIs to be atomic.
- // Keep _ptrMemory to be able to reclaim memory.
- _ptrMemory = malloc(sizeof(WebRtc_Word32)*2);
- _value = (WebRtc_Word32*) (((uintptr_t)_ptrMemory+3)&(~0x3));
- *_value = initialValue;
-}
-
-inline Atomic32Impl::~Atomic32Impl()
-{
- if(_ptrMemory != NULL)
- {
- free(_ptrMemory);
- }
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator++()
-{
- WebRtc_Word32 returnValue = __sync_fetch_and_add(_value,1);
- returnValue++;
- return returnValue;
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator--()
-{
- WebRtc_Word32 returnValue = __sync_fetch_and_sub(_value,1);
- returnValue--;
- return returnValue;
-}
-
-inline Atomic32Impl& Atomic32Impl::operator=(const Atomic32Impl& rhs)
-{
- *_value = *rhs._value;
- return *this;
-}
-
-inline Atomic32Impl& Atomic32Impl::operator=(WebRtc_Word32 rhs)
-{
- *_value = rhs;
- return *this;
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator+=(WebRtc_Word32 rhs)
-{
- WebRtc_Word32 returnValue = __sync_fetch_and_add(_value,rhs);
- returnValue += rhs;
- return returnValue;
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator-=(WebRtc_Word32 rhs)
-{
- WebRtc_Word32 returnValue = __sync_fetch_and_sub(_value,rhs);
- returnValue -= rhs;
- return returnValue;
-}
-
-inline bool Atomic32Impl::CompareExchange(WebRtc_Word32 newValue,
- WebRtc_Word32 compareValue)
-{
- return __sync_bool_compare_and_swap(_value,compareValue,newValue);
-}
-
-inline WebRtc_Word32 Atomic32Impl::Value() const
-{
- return *_value;
-}
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_LINUX_H_
diff --git a/trunk/src/system_wrappers/source/atomic32_mac.h b/trunk/src/system_wrappers/source/atomic32_mac.h
deleted file mode 100644
index bf8febc..0000000
--- a/trunk/src/system_wrappers/source/atomic32_mac.h
+++ /dev/null
@@ -1,117 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Atomic system independant 32-bit signed integer.
-// Mac implementation.
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_MAC_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_MAC_H_
-
-#include <stdlib.h>
-#include <libkern/OSAtomic.h>
-
-#include "common_types.h"
-
-namespace webrtc {
-class Atomic32Impl
-{
-public:
- inline Atomic32Impl(WebRtc_Word32 initialValue);
- inline ~Atomic32Impl();
-
- inline WebRtc_Word32 operator++();
- inline WebRtc_Word32 operator--();
-
- inline Atomic32Impl& operator=(const Atomic32Impl& rhs);
- inline Atomic32Impl& operator=(WebRtc_Word32 rhs);
- inline WebRtc_Word32 operator+=(WebRtc_Word32 rhs);
- inline WebRtc_Word32 operator-=(WebRtc_Word32 rhs);
-
- inline bool CompareExchange(WebRtc_Word32 newValue,
- WebRtc_Word32 compareValue);
-
- inline WebRtc_Word32 Value() const;
-private:
- void* _ptrMemory;
- // Volatile ensures full memory barriers.
- volatile WebRtc_Word32* _value;
-};
-
-// TODO (hellner) use aligned_malloc instead of doing it manually.
-inline Atomic32Impl::Atomic32Impl(WebRtc_Word32 initialValue)
- :
- _ptrMemory(NULL),
- _value(NULL)
-{ // Align the memory associated with _value on a 32-bit boundary. This is a
- // requirement for the used Mac APIs to be atomic.
- // Keep _ptrMemory to be able to reclaim memory.
- _ptrMemory = malloc(sizeof(WebRtc_Word32)*2);
- _value = (WebRtc_Word32*) (((uintptr_t)_ptrMemory+3)&(~0x3));
- *_value = initialValue;
-}
-
-inline Atomic32Impl::~Atomic32Impl()
-{
- if(_ptrMemory != NULL)
- {
- free(_ptrMemory);
- }
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator++()
-{
- return OSAtomicIncrement32Barrier(
- reinterpret_cast<volatile int32_t*>(_value));
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator--()
-{
- return OSAtomicDecrement32Barrier(
- reinterpret_cast<volatile int32_t*>(_value));
-}
-
-inline Atomic32Impl& Atomic32Impl::operator=(const Atomic32Impl& rhs)
-{
- *_value = *rhs._value;
- return *this;
-}
-
-inline Atomic32Impl& Atomic32Impl::operator=(WebRtc_Word32 rhs)
-{
- *_value = rhs;
- return *this;
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator+=(WebRtc_Word32 rhs)
-{
- return OSAtomicAdd32Barrier(rhs,
- reinterpret_cast<volatile int32_t*>(_value));
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator-=(WebRtc_Word32 rhs)
-{
- return OSAtomicAdd32Barrier(-rhs,
- reinterpret_cast<volatile int32_t*>(_value));
-}
-
-inline bool Atomic32Impl::CompareExchange(WebRtc_Word32 newValue,
- WebRtc_Word32 compareValue)
-{
- return OSAtomicCompareAndSwap32Barrier(
- compareValue,
- newValue,
- reinterpret_cast<volatile int32_t*>(_value));
-}
-
-inline WebRtc_Word32 Atomic32Impl::Value() const
-{
- return *_value;
-}
-} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_MAC_H_
diff --git a/trunk/src/system_wrappers/source/atomic32_win.h b/trunk/src/system_wrappers/source/atomic32_win.h
deleted file mode 100644
index c27e48e..0000000
--- a/trunk/src/system_wrappers/source/atomic32_win.h
+++ /dev/null
@@ -1,113 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Atomic system independant 32-bit signed integer.
-// Windows implementation.
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_WINDOWS_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_WINDOWS_H_
-
-#include <malloc.h>
-#include <windows.h>
-
-#include "common_types.h"
-
-namespace webrtc {
-class Atomic32Impl
-{
-public:
- inline Atomic32Impl(WebRtc_Word32 initialValue);
- inline ~Atomic32Impl();
-
- inline WebRtc_Word32 operator++();
- inline WebRtc_Word32 operator--();
-
- inline Atomic32Impl& operator=(const Atomic32Impl& rhs);
- inline Atomic32Impl& operator=(WebRtc_Word32 rhs);
- inline WebRtc_Word32 operator+=(WebRtc_Word32 rhs);
- inline WebRtc_Word32 operator-=(WebRtc_Word32 rhs);
-
- inline bool CompareExchange(WebRtc_Word32 newValue,
- WebRtc_Word32 compareValue);
-
- inline WebRtc_Word32 Value() const;
-private:
- void* _ptrMemory;
- // Volatile ensures full memory barriers.
- volatile LONG* _value;
-};
-
-// TODO (hellner) use aligned_malloc instead of doing it manually.
-inline Atomic32Impl::Atomic32Impl(WebRtc_Word32 initialValue)
- : _ptrMemory(NULL),
- _value(NULL)
-{ // Align the memory associated with _value on a 32-bit boundary. This is a
- // requirement for the used Windows APIs to be atomic.
- // Keep _ptrMemory to be able to reclaim memory.
- _ptrMemory = malloc(sizeof(WebRtc_Word32)*2);
- _value = reinterpret_cast<LONG*> (((uintptr_t)_ptrMemory+3)&(~0x3));
- *_value = initialValue;
-}
-
-inline Atomic32Impl::~Atomic32Impl()
-{
- if(_ptrMemory != NULL)
- {
- free(_ptrMemory);
- }
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator++()
-{
- return (WebRtc_Word32)InterlockedIncrement(_value);
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator--()
-{
- return (WebRtc_Word32)InterlockedDecrement(_value);
-}
-
-inline Atomic32Impl& Atomic32Impl::operator=(const Atomic32Impl& rhs)
-{
- *_value = *rhs._value;
- return *this;
-}
-
-inline Atomic32Impl& Atomic32Impl::operator=(WebRtc_Word32 rhs)
-{
- *_value = rhs;
- return *this;
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator+=(WebRtc_Word32 rhs)
-{
- return InterlockedExchangeAdd(_value,rhs);
-}
-
-inline WebRtc_Word32 Atomic32Impl::operator-=(WebRtc_Word32 rhs)
-{
- return InterlockedExchangeAdd(_value,-rhs);
-}
-
-inline bool Atomic32Impl::CompareExchange(WebRtc_Word32 newValue,
- WebRtc_Word32 compareValue)
-{
- const LONG oldValue = InterlockedCompareExchange(_value,newValue,
- compareValue);
- // If the old value and the compare value is the same an exchange happened.
- return (oldValue == compareValue);
-}
-
-inline WebRtc_Word32 Atomic32Impl::Value() const
-{
- return *_value;
-}
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_ATOMIC32_WINDOWS_H_
diff --git a/trunk/src/system_wrappers/source/condition_variable.cc b/trunk/src/system_wrappers/source/condition_variable.cc
deleted file mode 100644
index b37d037..0000000
--- a/trunk/src/system_wrappers/source/condition_variable.cc
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#if defined(_WIN32)
- #include <windows.h>
- #include "condition_variable_wrapper.h"
- #include "condition_variable_win.h"
-#elif defined(WEBRTC_LINUX)
- #include <pthread.h>
- #include "condition_variable_wrapper.h"
- #include "condition_variable_posix.h"
-#elif defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL)
- #include <pthread.h>
- #include "condition_variable_wrapper.h"
- #include "condition_variable_posix.h"
-#endif
-
-namespace webrtc {
-ConditionVariableWrapper*
-ConditionVariableWrapper::CreateConditionVariable()
-{
-#if defined(_WIN32)
- return new ConditionVariableWindows;
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL)
- return ConditionVariablePosix::Create();
-#else
- return NULL;
-#endif
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/condition_variable_posix.cc b/trunk/src/system_wrappers/source/condition_variable_posix.cc
deleted file mode 100644
index 48835ab..0000000
--- a/trunk/src/system_wrappers/source/condition_variable_posix.cc
+++ /dev/null
@@ -1,151 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "condition_variable_posix.h"
-
-#if defined(WEBRTC_LINUX)
-#include <ctime>
-#else
-#include <sys/time.h>
-#endif
-
-#include <errno.h>
-
-#include "critical_section_posix.h"
-
-namespace webrtc {
-ConditionVariableWrapper* ConditionVariablePosix::Create()
-{
- ConditionVariablePosix* ptr = new ConditionVariablePosix;
- if (!ptr)
- {
- return NULL;
- }
-
- const int error = ptr->Construct();
- if (error)
- {
- delete ptr;
- return NULL;
- }
-
- return ptr;
-}
-
-ConditionVariablePosix::ConditionVariablePosix()
-{
-}
-
-int ConditionVariablePosix::Construct()
-{
- int result = 0;
-#ifdef WEBRTC_CLOCK_TYPE_REALTIME
- result = pthread_cond_init(&_cond, NULL);
-#else
- pthread_condattr_t condAttr;
- result = pthread_condattr_init(&condAttr);
- if (result != 0)
- {
- return -1;
- }
- result = pthread_condattr_setclock(&condAttr, CLOCK_MONOTONIC);
- if (result != 0)
- {
- return -1;
- }
- result = pthread_cond_init(&_cond, &condAttr);
- if (result != 0)
- {
- return -1;
- }
- result = pthread_condattr_destroy(&condAttr);
- if (result != 0)
- {
- return -1;
- }
-#endif
- return 0;
-}
-
-ConditionVariablePosix::~ConditionVariablePosix()
-{
- pthread_cond_destroy(&_cond);
-}
-
-void ConditionVariablePosix::SleepCS(CriticalSectionWrapper& critSect)
-{
- CriticalSectionPosix* cs = reinterpret_cast<CriticalSectionPosix*>(
- &critSect);
- pthread_cond_wait(&_cond, &cs->_mutex);
-}
-
-
-bool
-ConditionVariablePosix::SleepCS(
- CriticalSectionWrapper& critSect,
- unsigned long maxTimeInMS)
-{
- const unsigned long INFINITE = 0xFFFFFFFF;
-
- const int MILLISECONDS_PER_SECOND = 1000;
-#ifndef WEBRTC_LINUX
- const int MICROSECONDS_PER_MILLISECOND = 1000;
-#endif
- const int NANOSECONDS_PER_SECOND = 1000000000;
- const int NANOSECONDS_PER_MILLISECOND = 1000000;
-
- CriticalSectionPosix* cs = reinterpret_cast<CriticalSectionPosix*>(
- &critSect);
-
- if (maxTimeInMS != INFINITE)
- {
- timespec ts;
-#ifndef WEBRTC_MAC
-#ifdef WEBRTC_CLOCK_TYPE_REALTIME
- clock_gettime(CLOCK_REALTIME, &ts);
-#else
- clock_gettime(CLOCK_MONOTONIC, &ts);
-#endif
-#else
- struct timeval tv;
- gettimeofday(&tv, 0);
- ts.tv_sec = tv.tv_sec;
- ts.tv_nsec = tv.tv_usec * MICROSECONDS_PER_MILLISECOND;
-#endif
-
- ts.tv_sec += maxTimeInMS / MILLISECONDS_PER_SECOND;
- ts.tv_nsec += (maxTimeInMS - ((maxTimeInMS / MILLISECONDS_PER_SECOND)*
- MILLISECONDS_PER_SECOND)) * NANOSECONDS_PER_MILLISECOND;
-
- if (ts.tv_nsec >= NANOSECONDS_PER_SECOND)
- {
- ts.tv_sec += ts.tv_nsec / NANOSECONDS_PER_SECOND;
- ts.tv_nsec %= NANOSECONDS_PER_SECOND;
- }
- const int res = pthread_cond_timedwait(&_cond, &cs->_mutex, &ts);
- return (res == ETIMEDOUT) ? false : true;
- }
- else
- {
- pthread_cond_wait(&_cond, &cs->_mutex);
- return true;
- }
-}
-
-void ConditionVariablePosix::Wake()
-{
- pthread_cond_signal(&_cond);
-}
-
-void ConditionVariablePosix::WakeAll()
-{
- pthread_cond_broadcast(&_cond);
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/condition_variable_posix.h b/trunk/src/system_wrappers/source/condition_variable_posix.h
deleted file mode 100644
index c239a47..0000000
--- a/trunk/src/system_wrappers/source/condition_variable_posix.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_POSIX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_POSIX_H_
-
-#include "condition_variable_wrapper.h"
-
-#include <pthread.h>
-
-namespace webrtc {
-class ConditionVariablePosix : public ConditionVariableWrapper
-{
-public:
- static ConditionVariableWrapper* Create();
- ~ConditionVariablePosix();
-
- void SleepCS(CriticalSectionWrapper& critSect);
- bool SleepCS(CriticalSectionWrapper& critSect, unsigned long maxTimeInMS);
- void Wake();
- void WakeAll();
-
-private:
- ConditionVariablePosix();
- int Construct();
-
-private:
- pthread_cond_t _cond;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_POSIX_H_
diff --git a/trunk/src/system_wrappers/source/condition_variable_win.cc b/trunk/src/system_wrappers/source/condition_variable_win.cc
deleted file mode 100644
index f4fae0b..0000000
--- a/trunk/src/system_wrappers/source/condition_variable_win.cc
+++ /dev/null
@@ -1,224 +0,0 @@
-/*
- * Use of this source code is governed by the ACE copyright license which
- * can be found in the LICENSE file in the third_party_mods/ace directory of
- * the source tree or at http://www1.cse.wustl.edu/~schmidt/ACE-copying.html.
- */
-/*
- * This source code contain modifications to the original source code
- * which can be found here:
- * http://www.cs.wustl.edu/~schmidt/win32-cv-1.html (section 3.2).
- * Modifications:
- * 1) Dynamic detection of native support for condition variables.
- * 2) Use of WebRTC defined types and classes. Renaming of some functions.
- * 3) Introduction of a second event for wake all functionality. This prevents
- * a thread from spinning on the same condition variable, preventing other
- * threads from waking up.
- */
-
-// TODO (hellner): probably nicer to split up native and generic
-// implementation into two different files
-
-#include "condition_variable_win.h"
-
-#include "critical_section_win.h"
-#include "trace.h"
-
-namespace webrtc {
-bool ConditionVariableWindows::_winSupportConditionVariablesPrimitive = false;
-static HMODULE library = NULL;
-
-PInitializeConditionVariable _PInitializeConditionVariable;
-PSleepConditionVariableCS _PSleepConditionVariableCS;
-PWakeConditionVariable _PWakeConditionVariable;
-PWakeAllConditionVariable _PWakeAllConditionVariable;
-
-typedef void (WINAPI *PInitializeConditionVariable)(PCONDITION_VARIABLE);
-typedef BOOL (WINAPI *PSleepConditionVariableCS)(PCONDITION_VARIABLE,
- PCRITICAL_SECTION, DWORD);
-typedef void (WINAPI *PWakeConditionVariable)(PCONDITION_VARIABLE);
-typedef void (WINAPI *PWakeAllConditionVariable)(PCONDITION_VARIABLE);
-
-ConditionVariableWindows::ConditionVariableWindows()
- : _eventID(WAKEALL_0)
-{
- if (!library)
- {
- // Use native implementation if supported (i.e Vista+)
- library = LoadLibrary(TEXT("Kernel32.dll"));
- if (library)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
- "Loaded Kernel.dll");
-
- _PInitializeConditionVariable =
- (PInitializeConditionVariable) GetProcAddress(
- library,
- "InitializeConditionVariable");
- _PSleepConditionVariableCS =
- (PSleepConditionVariableCS)GetProcAddress(
- library,
- "SleepConditionVariableCS");
- _PWakeConditionVariable =
- (PWakeConditionVariable)GetProcAddress(
- library,
- "WakeConditionVariable");
- _PWakeAllConditionVariable =
- (PWakeAllConditionVariable)GetProcAddress(
- library,
- "WakeAllConditionVariable");
-
- if(_PInitializeConditionVariable &&
- _PSleepConditionVariableCS &&
- _PWakeConditionVariable &&
- _PWakeAllConditionVariable)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
- "Loaded native condition variables");
- _winSupportConditionVariablesPrimitive = true;
- }
- }
- }
-
- if (_winSupportConditionVariablesPrimitive)
- {
- _PInitializeConditionVariable(&_conditionVariable);
-
- _events[WAKEALL_0] = NULL;
- _events[WAKEALL_1] = NULL;
- _events[WAKE] = NULL;
-
- } else {
- memset(&_numWaiters[0],0,sizeof(_numWaiters));
-
- InitializeCriticalSection(&_numWaitersCritSect);
-
- _events[WAKEALL_0] = CreateEvent(NULL, // no security attributes
- TRUE, // manual-reset, sticky event
- FALSE, // initial state non-signaled
- NULL); // no name for event
-
- _events[WAKEALL_1] = CreateEvent(NULL, // no security attributes
- TRUE, // manual-reset, sticky event
- FALSE, // initial state non-signaled
- NULL); // no name for event
-
- _events[WAKE] = CreateEvent(NULL, // no security attributes
- FALSE, // auto-reset, sticky event
- FALSE, // initial state non-signaled
- NULL); // no name for event
- }
-}
-
-ConditionVariableWindows::~ConditionVariableWindows()
-{
- if(!_winSupportConditionVariablesPrimitive)
- {
- CloseHandle(_events[WAKE]);
- CloseHandle(_events[WAKEALL_1]);
- CloseHandle(_events[WAKEALL_0]);
-
- DeleteCriticalSection(&_numWaitersCritSect);
- }
-}
-
-void ConditionVariableWindows::SleepCS(CriticalSectionWrapper& critSect)
-{
- SleepCS(critSect, INFINITE);
-}
-
-bool ConditionVariableWindows::SleepCS(CriticalSectionWrapper& critSect,
- unsigned long maxTimeInMS)
-{
- CriticalSectionWindows* cs = reinterpret_cast<CriticalSectionWindows*>(
- &critSect);
-
- if(_winSupportConditionVariablesPrimitive)
- {
- BOOL retVal = _PSleepConditionVariableCS(&_conditionVariable,
- &(cs->crit),maxTimeInMS);
- return (retVal == 0) ? false : true;
-
- }else
- {
- EnterCriticalSection(&_numWaitersCritSect);
- // Get the eventID for the event that will be triggered by next
- // WakeAll() call and start waiting for it.
- const EventWakeUpType eventID = (WAKEALL_0 == _eventID) ?
- WAKEALL_1 : WAKEALL_0;
- ++(_numWaiters[eventID]);
- LeaveCriticalSection(&_numWaitersCritSect);
-
- LeaveCriticalSection(&cs->crit);
- HANDLE events[2];
- events[0] = _events[WAKE];
- events[1] = _events[eventID];
- const DWORD result = WaitForMultipleObjects(2, // Wait on 2 events.
- events,
- FALSE, // Wait for either.
- maxTimeInMS);
-
- const bool retVal = (result != WAIT_TIMEOUT);
-
- EnterCriticalSection(&_numWaitersCritSect);
- --(_numWaiters[eventID]);
- // Last waiter should only be true for WakeAll(). WakeAll() correspond
- // to position 1 in events[] -> (result == WAIT_OBJECT_0 + 1)
- const bool lastWaiter = (result == WAIT_OBJECT_0 + 1) &&
- (_numWaiters[eventID] == 0);
- LeaveCriticalSection(&_numWaitersCritSect);
-
- if (lastWaiter)
- {
- // Reset/unset the WakeAll() event since all threads have been
- // released.
- ResetEvent(_events[eventID]);
- }
-
- EnterCriticalSection(&cs->crit);
- return retVal;
- }
-}
-
-void
-ConditionVariableWindows::Wake()
-{
- if(_winSupportConditionVariablesPrimitive)
- {
- _PWakeConditionVariable(&_conditionVariable);
- }else
- {
- EnterCriticalSection(&_numWaitersCritSect);
- const bool haveWaiters = (_numWaiters[WAKEALL_0] > 0) ||
- (_numWaiters[WAKEALL_1] > 0);
- LeaveCriticalSection(&_numWaitersCritSect);
-
- if (haveWaiters)
- {
- SetEvent(_events[WAKE]);
- }
- }
-}
-
-void
-ConditionVariableWindows::WakeAll()
-{
- if(_winSupportConditionVariablesPrimitive)
- {
- _PWakeAllConditionVariable(&_conditionVariable);
- }else
- {
- EnterCriticalSection(&_numWaitersCritSect);
- // Update current WakeAll() event
- _eventID = (WAKEALL_0 == _eventID) ? WAKEALL_1 : WAKEALL_0;
- // Trigger current event
- const EventWakeUpType eventID = _eventID;
- const bool haveWaiters = _numWaiters[eventID] > 0;
- LeaveCriticalSection(&_numWaitersCritSect);
-
- if (haveWaiters)
- {
- SetEvent(_events[eventID]);
- }
- }
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/condition_variable_win.h b/trunk/src/system_wrappers/source/condition_variable_win.h
deleted file mode 100644
index aab2564..0000000
--- a/trunk/src/system_wrappers/source/condition_variable_win.h
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_WINDOWS_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_WINDOWS_H_
-
-#include "condition_variable_wrapper.h"
-
-#include <windows.h>
-
-namespace webrtc {
-#if !defined CONDITION_VARIABLE_INIT
- typedef struct _RTL_CONDITION_VARIABLE
- {
- void* Ptr;
- } RTL_CONDITION_VARIABLE, *PRTL_CONDITION_VARIABLE;
-
- typedef RTL_CONDITION_VARIABLE CONDITION_VARIABLE, *PCONDITION_VARIABLE;
-#endif
-
-typedef void (WINAPI *PInitializeConditionVariable)(PCONDITION_VARIABLE);
-typedef BOOL (WINAPI *PSleepConditionVariableCS)(PCONDITION_VARIABLE,
- PCRITICAL_SECTION, DWORD);
-typedef void (WINAPI *PWakeConditionVariable)(PCONDITION_VARIABLE);
-typedef void (WINAPI *PWakeAllConditionVariable)(PCONDITION_VARIABLE);
-
-
-class ConditionVariableWindows : public ConditionVariableWrapper
-{
-public:
- ConditionVariableWindows();
- ~ConditionVariableWindows();
-
- void SleepCS(CriticalSectionWrapper& critSect);
- bool SleepCS(CriticalSectionWrapper& critSect, unsigned long maxTimeInMS);
- void Wake();
- void WakeAll();
-
-private:
- enum EventWakeUpType
- {
- WAKEALL_0 = 0,
- WAKEALL_1 = 1,
- WAKE = 2,
- EVENT_COUNT = 3
- };
-
-private:
- // Native support for Windows Vista+
- static bool _winSupportConditionVariablesPrimitive;
- CONDITION_VARIABLE _conditionVariable;
-
- unsigned int _numWaiters[2];
- EventWakeUpType _eventID;
- CRITICAL_SECTION _numWaitersCritSect;
- HANDLE _events[EVENT_COUNT];
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CONDITION_VARIABLE_WINDOWS_H_
diff --git a/trunk/src/system_wrappers/source/cpu.cc b/trunk/src/system_wrappers/source/cpu.cc
deleted file mode 100644
index 3df5d18..0000000
--- a/trunk/src/system_wrappers/source/cpu.cc
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "cpu_wrapper.h"
-
-#if defined(_WIN32)
- #include "cpu_win.h"
-#elif defined(WEBRTC_MAC)
- #include "cpu_mac.h"
-#elif defined(WEBRTC_MAC_INTEL)
- #include "cpu_mac.h"
-#elif defined(WEBRTC_ANDROID)
- // Not implemented yet, might be possible to use Linux implementation
-#else // defined(WEBRTC_LINUX)
- #include "cpu_linux.h"
-#endif
-
-namespace webrtc {
-CpuWrapper* CpuWrapper::CreateCpu()
-{
-#if defined(_WIN32)
- return new CpuWindows();
-#elif (defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL))
- return new CpuWrapperMac();
-#elif defined(WEBRTC_ANDROID)
- return 0;
-#else
- return new CpuLinux();
-#endif
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/cpu_features.cc b/trunk/src/system_wrappers/source/cpu_features.cc
deleted file mode 100644
index 41a86e3..0000000
--- a/trunk/src/system_wrappers/source/cpu_features.cc
+++ /dev/null
@@ -1,74 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Parts of this file derived from Chromium's base/cpu.cc.
-
-#include "cpu_features_wrapper.h"
-
-#include "typedefs.h"
-
-#if defined(WEBRTC_ARCH_X86_FAMILY)
-#if defined(_MSC_VER)
-#include <intrin.h>
-#endif
-#endif
-
-// No CPU feature is available => straight C path.
-int GetCPUInfoNoASM(CPUFeature feature) {
- (void)feature;
- return 0;
-}
-
-#if defined(WEBRTC_ARCH_X86_FAMILY)
-#ifndef _MSC_VER
-// Intrinsic for "cpuid".
-#if defined(__pic__) && defined(__i386__)
-static inline void __cpuid(int cpu_info[4], int info_type) {
- __asm__ volatile (
- "mov %%ebx, %%edi\n"
- "cpuid\n"
- "xchg %%edi, %%ebx\n"
- : "=a"(cpu_info[0]), "=D"(cpu_info[1]), "=c"(cpu_info[2]), "=d"(cpu_info[3])
- : "a"(info_type));
-}
-#else
-static inline void __cpuid(int cpu_info[4], int info_type) {
- __asm__ volatile (
- "cpuid\n"
- : "=a"(cpu_info[0]), "=b"(cpu_info[1]), "=c"(cpu_info[2]), "=d"(cpu_info[3])
- : "a"(info_type));
-}
-#endif
-#endif // _MSC_VER
-#endif // WEBRTC_ARCH_X86_FAMILY
-
-#if defined(WEBRTC_ARCH_X86_FAMILY)
-// Actual feature detection for x86.
-static int GetCPUInfo(CPUFeature feature) {
- int cpu_info[4];
- __cpuid(cpu_info, 1);
- if (feature == kSSE2) {
- return 0 != (cpu_info[3] & 0x04000000);
- }
- if (feature == kSSE3) {
- return 0 != (cpu_info[2] & 0x00000001);
- }
- return 0;
-}
-#else
-// Default to straight C for other platforms.
-static int GetCPUInfo(CPUFeature feature) {
- (void)feature;
- return 0;
-}
-#endif
-
-WebRtc_CPUInfo WebRtc_GetCPUInfo = GetCPUInfo;
-WebRtc_CPUInfo WebRtc_GetCPUInfoNoASM = GetCPUInfoNoASM;
diff --git a/trunk/src/system_wrappers/source/cpu_features_arm.c b/trunk/src/system_wrappers/source/cpu_features_arm.c
deleted file mode 100644
index 1065118..0000000
--- a/trunk/src/system_wrappers/source/cpu_features_arm.c
+++ /dev/null
@@ -1,333 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file is derived from Android's NDK package r7, located at
-// <ndk>/sources/android/cpufeatures/ (downloadable from
-// http://developer.android.com/sdk/ndk/index.html).
-
-#include "cpu_features_wrapper.h"
-
-#include <fcntl.h>
-#include <errno.h>
-#include <pthread.h>
-#include <stdio.h>
-#include <stdlib.h>
-
-// Define CPU family.
-typedef enum {
- CPU_FAMILY_UNKNOWN = 0,
- CPU_FAMILY_ARM,
- CPU_FAMILY_X86,
- CPU_FAMILY_MAX // Do not remove.
-} CpuFamily;
-
-static pthread_once_t g_once;
-static CpuFamily g_cpuFamily;
-static uint64_t g_cpuFeatures;
-static int g_cpuCount;
-
-static const int cpufeatures_debug = 0;
-
-#ifdef __arm__
-# define DEFAULT_CPU_FAMILY CPU_FAMILY_ARM
-#elif defined __i386__
-# define DEFAULT_CPU_FAMILY CPU_FAMILY_X86
-#else
-# define DEFAULT_CPU_FAMILY CPU_FAMILY_UNKNOWN
-#endif
-
-#define D(...) \
- do { \
- if (cpufeatures_debug) { \
- printf(__VA_ARGS__); fflush(stdout); \
- } \
- } while (0)
-
-/* Read the content of /proc/cpuinfo into a user-provided buffer.
- * Return the length of the data, or -1 on error. Does *not*
- * zero-terminate the content. Will not read more
- * than 'buffsize' bytes.
- */
-static int read_file(const char* pathname, char* buffer, size_t buffsize) {
- int fd, len;
-
- fd = open(pathname, O_RDONLY);
- if (fd < 0)
- return -1;
-
- do {
- len = read(fd, buffer, buffsize);
- } while (len < 0 && errno == EINTR);
-
- close(fd);
-
- return len;
-}
-
-/* Extract the content of a the first occurence of a given field in
- * the content of /proc/cpuinfo and return it as a heap-allocated
- * string that must be freed by the caller.
- *
- * Return NULL if not found
- */
-static char* extract_cpuinfo_field(char* buffer, int buflen, const char* field) {
- int fieldlen = strlen(field);
- char* bufend = buffer + buflen;
- char* result = NULL;
- int len, ignore;
- const char* p, *q;
-
- /* Look for first field occurence, and ensures it starts the line.
- */
- p = buffer;
- bufend = buffer + buflen;
- for (;;) {
- p = memmem(p, bufend - p, field, fieldlen);
- if (p == NULL)
- goto EXIT;
-
- if (p == buffer || p[-1] == '\n')
- break;
-
- p += fieldlen;
- }
-
- /* Skip to the first column followed by a space */
- p += fieldlen;
- p = memchr(p, ':', bufend - p);
- if (p == NULL || p[1] != ' ')
- goto EXIT;
-
- /* Find the end of the line */
- p += 2;
- q = memchr(p, '\n', bufend - p);
- if (q == NULL)
- q = bufend;
-
- /* Copy the line into a heap-allocated buffer */
- len = q - p;
- result = malloc(len + 1);
- if (result == NULL)
- goto EXIT;
-
- memcpy(result, p, len);
- result[len] = '\0';
-
-EXIT:
- return result;
-}
-
-/* Count the number of occurences of a given field prefix in /proc/cpuinfo.
- */
-static int count_cpuinfo_field(char* buffer, int buflen, const char* field) {
- int fieldlen = strlen(field);
- const char* p = buffer;
- const char* bufend = buffer + buflen;
- const char* q;
- int count = 0;
-
- for (;;) {
- const char* q;
-
- p = memmem(p, bufend - p, field, fieldlen);
- if (p == NULL)
- break;
-
- /* Ensure that the field is at the start of a line */
- if (p > buffer && p[-1] != '\n') {
- p += fieldlen;
- continue;
- }
-
-
- /* skip any whitespace */
- q = p + fieldlen;
- while (q < bufend && (*q == ' ' || *q == '\t'))
- q++;
-
- /* we must have a colon now */
- if (q < bufend && *q == ':') {
- count += 1;
- q ++;
- }
- p = q;
- }
-
- return count;
-}
-
-/* Like strlen(), but for constant string literals */
-#define STRLEN_CONST(x) ((sizeof(x)-1)
-
-
-/* Checks that a space-separated list of items contains one given 'item'.
- * Returns 1 if found, 0 otherwise.
- */
-static int has_list_item(const char* list, const char* item) {
- const char* p = list;
- int itemlen = strlen(item);
-
- if (list == NULL)
- return 0;
-
- while (*p) {
- const char* q;
-
- /* skip spaces */
- while (*p == ' ' || *p == '\t')
- p++;
-
- /* find end of current list item */
- q = p;
- while (*q && *q != ' ' && *q != '\t')
- q++;
-
- if (itemlen == q - p && !memcmp(p, item, itemlen))
- return 1;
-
- /* skip to next item */
- p = q;
- }
- return 0;
-}
-
-
-static void cpuInit(void) {
- char cpuinfo[4096];
- int cpuinfo_len;
-
- g_cpuFamily = DEFAULT_CPU_FAMILY;
- g_cpuFeatures = 0;
- g_cpuCount = 1;
-
- cpuinfo_len = read_file("/proc/cpuinfo", cpuinfo, sizeof cpuinfo);
- D("cpuinfo_len is (%d):\n%.*s\n", cpuinfo_len,
- cpuinfo_len >= 0 ? cpuinfo_len : 0, cpuinfo);
-
- if (cpuinfo_len < 0) { /* should not happen */
- return;
- }
-
- /* Count the CPU cores, the value may be 0 for single-core CPUs */
- g_cpuCount = count_cpuinfo_field(cpuinfo, cpuinfo_len, "processor");
- if (g_cpuCount == 0) {
- g_cpuCount = count_cpuinfo_field(cpuinfo, cpuinfo_len, "Processor");
- if (g_cpuCount == 0) {
- g_cpuCount = 1;
- }
- }
-
- D("found cpuCount = %d\n", g_cpuCount);
-
-#ifdef __arm__
- {
- char* features = NULL;
- char* architecture = NULL;
-
- /* Extract architecture from the "CPU Architecture" field.
- * The list is well-known, unlike the the output of
- * the 'Processor' field which can vary greatly.
- *
- * See the definition of the 'proc_arch' array in
- * $KERNEL/arch/arm/kernel/setup.c and the 'c_show' function in
- * same file.
- */
- char* cpuArch = extract_cpuinfo_field(cpuinfo, cpuinfo_len,
- "CPU architecture");
-
- if (cpuArch != NULL) {
- char* end;
- long archNumber;
- int hasARMv7 = 0;
-
- D("found cpuArch = '%s'\n", cpuArch);
-
- /* read the initial decimal number, ignore the rest */
- archNumber = strtol(cpuArch, &end, 10);
-
- /* Here we assume that ARMv8 will be upwards compatible with v7
- * in the future. Unfortunately, there is no 'Features' field to
- * indicate that Thumb-2 is supported.
- */
- if (end > cpuArch && archNumber >= 7) {
- hasARMv7 = 1;
- }
-
- /* Unfortunately, it seems that certain ARMv6-based CPUs
- * report an incorrect architecture number of 7!
- *
- * We try to correct this by looking at the 'elf_format'
- * field reported by the 'Processor' field, which is of the
- * form of "(v7l)" for an ARMv7-based CPU, and "(v6l)" for
- * an ARMv6-one.
- */
- if (hasARMv7) {
- char* cpuProc = extract_cpuinfo_field(cpuinfo, cpuinfo_len,
- "Processor");
- if (cpuProc != NULL) {
- D("found cpuProc = '%s'\n", cpuProc);
- if (has_list_item(cpuProc, "(v6l)")) {
- D("CPU processor and architecture mismatch!!\n");
- hasARMv7 = 0;
- }
- free(cpuProc);
- }
- }
-
- if (hasARMv7) {
- g_cpuFeatures |= kCPUFeatureARMv7;
- }
-
- /* The LDREX / STREX instructions are available from ARMv6 */
- if (archNumber >= 6) {
- g_cpuFeatures |= kCPUFeatureLDREXSTREX;
- }
-
- free(cpuArch);
- }
-
- /* Extract the list of CPU features from 'Features' field */
- char* cpuFeatures = extract_cpuinfo_field(cpuinfo, cpuinfo_len,
- "Features");
-
- if (cpuFeatures != NULL) {
-
- D("found cpuFeatures = '%s'\n", cpuFeatures);
-
- if (has_list_item(cpuFeatures, "vfpv3"))
- g_cpuFeatures |= kCPUFeatureVFPv3;
-
- else if (has_list_item(cpuFeatures, "vfpv3d16"))
- g_cpuFeatures |= kCPUFeatureVFPv3;
-
- if (has_list_item(cpuFeatures, "neon")) {
- /* Note: Certain kernels only report neon but not vfpv3
- * in their features list. However, ARM mandates
- * that if Neon is implemented, so must be VFPv3
- * so always set the flag.
- */
- g_cpuFeatures |= kCPUFeatureNEON |
- kCPUFeatureVFPv3;
- }
- free(cpuFeatures);
- }
- }
-#endif // __arm__
-
-#ifdef __i386__
- g_cpuFamily = CPU_FAMILY_X86;
-#endif
-}
-
-
-uint64_t WebRtc_GetCPUFeaturesARM(void) {
- pthread_once(&g_once, cpuInit);
- return g_cpuFeatures;
-}
diff --git a/trunk/src/system_wrappers/source/cpu_info.cc b/trunk/src/system_wrappers/source/cpu_info.cc
deleted file mode 100644
index e367abf..0000000
--- a/trunk/src/system_wrappers/source/cpu_info.cc
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "cpu_info.h"
-
-#if defined(_WIN32)
-#include <Windows.h>
-#elif defined(WEBRTC_MAC)
-#include <sys/types.h>
-#include <sys/sysctl.h>
-#elif defined(WEBRTC_MAC_INTEL)
-// Intentionally empty
-#elif defined(WEBRTC_ANDROID)
-// Not implemented yet, might be possible to use Linux implementation
-#else // defined(WEBRTC_LINUX)
-#include <sys/sysinfo.h>
-#endif
-
-#include "trace.h"
-
-namespace webrtc {
-
-WebRtc_UWord32 CpuInfo::_numberOfCores = 0;
-
-WebRtc_UWord32 CpuInfo::DetectNumberOfCores()
-{
- if (!_numberOfCores)
- {
-#if defined(_WIN32)
- SYSTEM_INFO si;
- GetSystemInfo(&si);
- _numberOfCores = static_cast<WebRtc_UWord32>(si.dwNumberOfProcessors);
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
- "Available number of cores:%d", _numberOfCores);
-
-#elif defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
- _numberOfCores = get_nprocs();
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
- "Available number of cores:%d", _numberOfCores);
-
-#elif (defined(WEBRTC_MAC) || defined(WEBRTC_MAC_INTEL))
- int name[] = {CTL_HW, HW_AVAILCPU};
- int ncpu;
- size_t size = sizeof(ncpu);
- if(0 == sysctl(name, 2, &ncpu, &size, NULL, 0))
- {
- _numberOfCores = static_cast<WebRtc_UWord32>(ncpu);
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
- "Available number of cores:%d", _numberOfCores);
- } else
- {
- WEBRTC_TRACE(kTraceError, kTraceUtility, -1,
- "Failed to get number of cores");
- _numberOfCores = 1;
- }
-#else
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, -1,
- "No function to get number of cores");
- _numberOfCores = 1;
-#endif
- }
- return _numberOfCores;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/cpu_linux.cc b/trunk/src/system_wrappers/source/cpu_linux.cc
deleted file mode 100644
index 8e8ecda..0000000
--- a/trunk/src/system_wrappers/source/cpu_linux.cc
+++ /dev/null
@@ -1,202 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "cpu_linux.h"
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <unistd.h>
-
-namespace webrtc {
-CpuLinux::CpuLinux()
- : m_oldBusyTime(0),
- m_oldIdleTime(0),
- m_oldBusyTimeMulti(NULL),
- m_oldIdleTimeMulti(NULL),
- m_idleArray(NULL),
- m_busyArray(NULL),
- m_resultArray(NULL),
- m_numCores(0) {
- const int result = GetNumCores();
- if (result != -1) {
- m_numCores = result;
- m_oldBusyTimeMulti = new long long[m_numCores];
- memset(m_oldBusyTimeMulti, 0, sizeof(long long) * m_numCores);
- m_oldIdleTimeMulti = new long long[m_numCores];
- memset(m_oldIdleTimeMulti, 0, sizeof(long long) * m_numCores);
- m_idleArray = new long long[m_numCores];
- memset(m_idleArray, 0, sizeof(long long) * m_numCores);
- m_busyArray = new long long[m_numCores];
- memset(m_busyArray, 0, sizeof(long long) * m_numCores);
- m_resultArray = new WebRtc_UWord32[m_numCores];
-
- GetData(m_oldBusyTime, m_oldIdleTime, m_busyArray, m_idleArray);
- }
-}
-
-CpuLinux::~CpuLinux()
-{
- delete [] m_oldBusyTimeMulti;
- delete [] m_oldIdleTimeMulti;
- delete [] m_idleArray;
- delete [] m_busyArray;
- delete [] m_resultArray;
-}
-
-WebRtc_Word32 CpuLinux::CpuUsage()
-{
- WebRtc_UWord32 dummy = 0;
- WebRtc_UWord32* dummyArray = NULL;
- return CpuUsageMultiCore(dummy, dummyArray);
-}
-
-WebRtc_Word32 CpuLinux::CpuUsageMultiCore(WebRtc_UWord32& numCores,
- WebRtc_UWord32*& coreArray)
-{
- coreArray = m_resultArray;
- numCores = m_numCores;
- long long busy = 0;
- long long idle = 0;
- if (GetData(busy, idle, m_busyArray, m_idleArray) != 0)
- return -1;
-
- long long deltaBusy = busy - m_oldBusyTime;
- long long deltaIdle = idle - m_oldIdleTime;
- m_oldBusyTime = busy;
- m_oldIdleTime = idle;
-
- int retVal = -1;
- if (deltaBusy + deltaIdle == 0)
- {
- retVal = 0;
- }
- else
- {
- retVal = (int)(100 * (deltaBusy) / (deltaBusy + deltaIdle));
- }
-
- if (coreArray == NULL)
- {
- return retVal;
- }
-
- for (WebRtc_UWord32 i = 0; i < m_numCores; i++)
- {
- deltaBusy = m_busyArray[i] - m_oldBusyTimeMulti[i];
- deltaIdle = m_idleArray[i] - m_oldIdleTimeMulti[i];
- m_oldBusyTimeMulti[i] = m_busyArray[i];
- m_oldIdleTimeMulti[i] = m_idleArray[i];
- if(deltaBusy + deltaIdle == 0)
- {
- coreArray[i] = 0;
- }
- else
- {
- coreArray[i] = (int)(100 * (deltaBusy) / (deltaBusy+deltaIdle));
- }
- }
- return retVal;
-}
-
-
-int CpuLinux::GetData(long long& busy, long long& idle, long long*& busyArray,
- long long*& idleArray)
-{
- FILE* fp = fopen("/proc/stat", "r");
- if (!fp)
- {
- return -1;
- }
-
- char line[100];
- if (fgets(line, 100, fp) == NULL) {
- fclose(fp);
- return -1;
- }
- char firstWord[100];
- if (sscanf(line, "%s ", firstWord) != 1) {
- fclose(fp);
- return -1;
- }
- if (strncmp(firstWord, "cpu", 3) != 0) {
- fclose(fp);
- return -1;
- }
- char sUser[100];
- char sNice[100];
- char sSystem[100];
- char sIdle[100];
- if (sscanf(line, "%s %s %s %s %s ",
- firstWord, sUser, sNice, sSystem, sIdle) != 5) {
- fclose(fp);
- return -1;
- }
- long long luser = atoll(sUser);
- long long lnice = atoll(sNice);
- long long lsystem = atoll(sSystem);
- long long lidle = atoll (sIdle);
-
- busy = luser + lnice + lsystem;
- idle = lidle;
- for (WebRtc_UWord32 i = 0; i < m_numCores; i++)
- {
- if (fgets(line, 100, fp) == NULL) {
- fclose(fp);
- return -1;
- }
- if (sscanf(line, "%s %s %s %s %s ", firstWord, sUser, sNice, sSystem,
- sIdle) != 5) {
- fclose(fp);
- return -1;
- }
- luser = atoll(sUser);
- lnice = atoll(sNice);
- lsystem = atoll(sSystem);
- lidle = atoll (sIdle);
- busyArray[i] = luser + lnice + lsystem;
- idleArray[i] = lidle;
- }
- fclose(fp);
- return 0;
-}
-
-int CpuLinux::GetNumCores()
-{
- FILE* fp = fopen("/proc/stat", "r");
- if (!fp)
- {
- return -1;
- }
- // Skip first line
- char line[100];
- if (!fgets(line, 100, fp))
- {
- fclose(fp);
- return -1;
- }
- int numCores = -1;
- char firstWord[100];
- do
- {
- numCores++;
- if (fgets(line, 100, fp))
- {
- if (sscanf(line, "%s ", firstWord) != 1) {
- firstWord[0] = '\0';
- }
- } else {
- break;
- }
- } while (strncmp(firstWord, "cpu", 3) == 0);
- fclose(fp);
- return numCores;
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/cpu_linux.h b/trunk/src/system_wrappers/source/cpu_linux.h
deleted file mode 100644
index 9b22e83..0000000
--- a/trunk/src/system_wrappers/source/cpu_linux.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_LINUX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_LINUX_H_
-
-#include "cpu_wrapper.h"
-
-namespace webrtc {
-class CpuLinux : public CpuWrapper
-{
-public:
- CpuLinux();
- virtual ~CpuLinux();
-
- virtual WebRtc_Word32 CpuUsage();
- virtual WebRtc_Word32 CpuUsage(WebRtc_Word8* /*pProcessName*/,
- WebRtc_UWord32 /*length*/) {return 0;}
- virtual WebRtc_Word32 CpuUsage(WebRtc_UWord32 /*dwProcessID*/) {return 0;}
-
- virtual WebRtc_Word32 CpuUsageMultiCore(WebRtc_UWord32& numCores,
- WebRtc_UWord32*& array);
-
- virtual void Reset() {return;}
- virtual void Stop() {return;}
-private:
- int GetData(long long& busy, long long& idle, long long*& busyArray,
- long long*& idleArray);
- int GetNumCores();
-
- long long m_oldBusyTime;
- long long m_oldIdleTime;
-
- long long* m_oldBusyTimeMulti;
- long long* m_oldIdleTimeMulti;
-
- long long* m_idleArray;
- long long* m_busyArray;
- WebRtc_UWord32* m_resultArray;
- WebRtc_UWord32 m_numCores;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_LINUX_H_
diff --git a/trunk/src/system_wrappers/source/cpu_mac.cc b/trunk/src/system_wrappers/source/cpu_mac.cc
deleted file mode 100644
index d82bf07..0000000
--- a/trunk/src/system_wrappers/source/cpu_mac.cc
+++ /dev/null
@@ -1,143 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "cpu_mac.h"
-
-#include <iostream>
-#include <mach/mach.h>
-#include <mach/mach_error.h>
-
-#include "tick_util.h"
-
-namespace webrtc {
-CpuWrapperMac::CpuWrapperMac()
- : _cpuCount(0),
- _cpuUsage(NULL),
- _totalCpuUsage(0),
- _lastTickCount(NULL),
- _lastTime(0)
-{
- natural_t cpuCount;
- processor_info_array_t infoArray;
- mach_msg_type_number_t infoCount;
-
- kern_return_t error = host_processor_info(mach_host_self(),
- PROCESSOR_CPU_LOAD_INFO,
- &cpuCount,
- &infoArray,
- &infoCount);
- if (error)
- {
- return;
- }
-
- _cpuCount = cpuCount;
- _cpuUsage = new WebRtc_UWord32[cpuCount];
- _lastTickCount = new WebRtc_Word64[cpuCount];
- _lastTime = TickTime::MillisecondTimestamp();
-
- processor_cpu_load_info_data_t* cpuLoadInfo =
- (processor_cpu_load_info_data_t*) infoArray;
- for (unsigned int cpu= 0; cpu < cpuCount; cpu++)
- {
- WebRtc_Word64 ticks = 0;
- for (int state = 0; state < 2; state++)
- {
- ticks += cpuLoadInfo[cpu].cpu_ticks[state];
- }
- _lastTickCount[cpu] = ticks;
- _cpuUsage[cpu] = 0;
- }
- vm_deallocate(mach_task_self(), (vm_address_t)infoArray, infoCount);
-}
-
-CpuWrapperMac::~CpuWrapperMac()
-{
- delete[] _cpuUsage;
- delete[] _lastTickCount;
-}
-
-WebRtc_Word32 CpuWrapperMac::CpuUsage()
-{
- WebRtc_UWord32 numCores;
- WebRtc_UWord32* array = NULL;
- return CpuUsageMultiCore(numCores, array);
-}
-
-WebRtc_Word32
-CpuWrapperMac::CpuUsageMultiCore(WebRtc_UWord32& numCores,
- WebRtc_UWord32*& array)
-{
- // sanity check
- if(_cpuUsage == NULL)
- {
- return -1;
- }
-
- WebRtc_Word64 now = TickTime::MillisecondTimestamp();
- WebRtc_Word64 timeDiffMS = now - _lastTime;
- if(timeDiffMS >= 500)
- {
- if(Update(timeDiffMS) != 0)
- {
- return -1;
- }
- _lastTime = now;
- }
-
- numCores = _cpuCount;
- array = _cpuUsage;
- return _totalCpuUsage / _cpuCount;
-}
-
-WebRtc_Word32 CpuWrapperMac::Update(WebRtc_Word64 timeDiffMS)
-{
- natural_t cpuCount;
- processor_info_array_t infoArray;
- mach_msg_type_number_t infoCount;
-
- kern_return_t error = host_processor_info(mach_host_self(),
- PROCESSOR_CPU_LOAD_INFO,
- &cpuCount,
- &infoArray,
- &infoCount);
- if (error)
- {
- return -1;
- }
-
- processor_cpu_load_info_data_t* cpuLoadInfo =
- (processor_cpu_load_info_data_t*) infoArray;
-
- _totalCpuUsage = 0;
- for (unsigned int cpu = 0; cpu < cpuCount; cpu++)
- {
- WebRtc_Word64 ticks = 0;
- for (int state = 0; state < 2; state++)
- {
- ticks += cpuLoadInfo[cpu].cpu_ticks[state];
- }
- if(timeDiffMS <= 0)
- {
- _cpuUsage[cpu] = 0;
- }else {
- _cpuUsage[cpu] = (WebRtc_UWord32)((1000 *
- (ticks - _lastTickCount[cpu])) /
- timeDiffMS);
- }
- _lastTickCount[cpu] = ticks;
- _totalCpuUsage += _cpuUsage[cpu];
- }
-
- vm_deallocate(mach_task_self(), (vm_address_t)infoArray, infoCount);
-
- return 0;
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/cpu_mac.h b/trunk/src/system_wrappers/source/cpu_mac.h
deleted file mode 100644
index f9f8207..0000000
--- a/trunk/src/system_wrappers/source/cpu_mac.h
+++ /dev/null
@@ -1,48 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_MAC_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_MAC_H_
-
-#include "cpu_wrapper.h"
-
-namespace webrtc {
-class CpuWrapperMac : public CpuWrapper
-{
-public:
- CpuWrapperMac();
- virtual ~CpuWrapperMac();
-
- virtual WebRtc_Word32 CpuUsage();
- virtual WebRtc_Word32 CpuUsage(WebRtc_Word8* /*pProcessName*/,
- WebRtc_UWord32 /*length*/) {return -1;}
- virtual WebRtc_Word32 CpuUsage(WebRtc_UWord32 /*dwProcessID*/) {return -1;}
-
- // Note: this class will block the call and sleep if called too fast
- // This function blocks the calling thread if the thread is calling it more
- // often than every 500 ms.
- virtual WebRtc_Word32 CpuUsageMultiCore(WebRtc_UWord32& numCores,
- WebRtc_UWord32*& array);
-
- virtual void Reset() {}
- virtual void Stop() {}
-
-private:
- WebRtc_Word32 Update(WebRtc_Word64 timeDiffMS);
-
- WebRtc_UWord32 _cpuCount;
- WebRtc_UWord32* _cpuUsage;
- WebRtc_Word32 _totalCpuUsage;
- WebRtc_Word64* _lastTickCount;
- WebRtc_Word64 _lastTime;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_MAC_H_
diff --git a/trunk/src/system_wrappers/source/cpu_measurement_harness.cc b/trunk/src/system_wrappers/source/cpu_measurement_harness.cc
deleted file mode 100644
index 237e776..0000000
--- a/trunk/src/system_wrappers/source/cpu_measurement_harness.cc
+++ /dev/null
@@ -1,127 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "system_wrappers/interface/cpu_wrapper.h"
-#include "system_wrappers/interface/event_wrapper.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-#include "system_wrappers/source/cpu_measurement_harness.h"
-
-const int kCpuCheckPeriodMs = 100;
-
-namespace webrtc {
-
-CpuMeasurementHarness* CpuMeasurementHarness::Create(
- CpuTarget* target,
- int work_period_ms,
- int work_iterations_per_period,
- int duration_ms) {
- if (target == NULL) {
- return NULL;
- }
- if (work_period_ms > duration_ms) {
- return NULL;
- }
- if (work_period_ms < 0) {
- return NULL;
- }
- if (duration_ms < 0) {
- return NULL;
- }
- if (work_iterations_per_period < 1) {
- return NULL;
- }
- return new CpuMeasurementHarness(target, work_period_ms,
- work_iterations_per_period, duration_ms);
-}
-
-CpuMeasurementHarness::CpuMeasurementHarness(CpuTarget* target,
- int work_period_ms,
- int work_iterations_per_period,
- int duration_ms)
- : cpu_target_(target),
- work_period_ms_(work_period_ms),
- work_iterations_per_period_(work_iterations_per_period),
- duration_ms_(duration_ms),
- cpu_sum_(0),
- cpu_iterations_(0),
- cpu_(CpuWrapper::CreateCpu()),
- event_(EventWrapper::Create()) {
-}
-
-CpuMeasurementHarness::~CpuMeasurementHarness() {
-}
-
-bool CpuMeasurementHarness::Run() {
- if (!WaitForCpuInit()) {
- return false;
- }
- // No need for precision. Run for approximately the asked for duration.
- // TODO(hellner): very low prio if at all, the actual duration of the test
- // will be longer if calling DoWork() is not negligable and/or called many
- // times. It may make sense to compensate for drift here. This will,
- // however, only add complexity with minimal gains. Perhaps renaming the
- // duration_ms_ to something more fuzzy is a better idea. However, the name
- // would be very convoluted if it is to be self documenting.
- int elapsed_time_ms = 0;
- int last_measured_time = 0;
- while (elapsed_time_ms < duration_ms_) {
- if (((elapsed_time_ms - last_measured_time) / kCpuCheckPeriodMs) >= 1) {
- last_measured_time = elapsed_time_ms;
- Measure();
- }
- if (!DoWork()) {
- return false;
- }
- event_->Wait(work_period_ms_);
- elapsed_time_ms += work_period_ms_;
- }
- return true;
-}
-
-int CpuMeasurementHarness::AverageCpu() {
- if (cpu_iterations_ == 0) {
- return 0;
- }
- assert(cpu_sum_ >= 0);
- assert(cpu_iterations_ >= 0);
- return cpu_sum_ / cpu_iterations_;
-}
-
-bool CpuMeasurementHarness::WaitForCpuInit() {
- bool cpu_usage_available = false;
- int num_iterations = 0;
- // Initializing the CPU measurements may take a couple of seconds on Windows.
- // Since the initialization is lazy we need to wait until it is completed.
- // Should not take more than 10000 ms.
- while (!cpu_usage_available && (++num_iterations < 10000)) {
- event_->Wait(1);
- cpu_usage_available = cpu_->CpuUsage() != -1;
- }
- return cpu_usage_available;
-}
-
-void CpuMeasurementHarness::Measure() {
- WebRtc_UWord32 num_cores = 0;
- WebRtc_UWord32* cores = NULL;
- // Return the average CPU for now.
- cpu_sum_ = cpu_->CpuUsageMultiCore(num_cores, cores);
- ++cpu_iterations_;
-}
-
-bool CpuMeasurementHarness::DoWork() {
- for (int i = 0; i < work_iterations_per_period_; ++i) {
- if (!cpu_target_->DoWork()) {
- return false;
- }
- }
- return true;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/cpu_measurement_harness.h b/trunk/src/system_wrappers/source/cpu_measurement_harness.h
deleted file mode 100644
index 3b87f27..0000000
--- a/trunk/src/system_wrappers/source/cpu_measurement_harness.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_SYSTEM_WRAPPERS_SOURCE_CPU_MEASUREMENT_HARNESS_H_
-#define SRC_SYSTEM_WRAPPERS_SOURCE_CPU_MEASUREMENT_HARNESS_H_
-
-#include "system_wrappers/interface/scoped_ptr.h"
-
-namespace webrtc {
-
-class CpuWrapper;
-class EventWrapper;
-class ThreadWrapper;
-
-// This abstract class provides an interface that should be passed to
-// CpuMeasurementHarness. CpuMeasurementHarness will call it with the
-// frequency requested and measure the CPU usage for all calls.
-class CpuTarget {
- public:
- // Callback function for which the CPU usage should be calculated.
- virtual bool DoWork() = 0;
-
- protected:
- CpuTarget() {}
- virtual ~CpuTarget() {}
-};
-
-class CpuMeasurementHarness {
- public:
- static CpuMeasurementHarness* Create(CpuTarget* target,
- int work_period_ms,
- int work_iterations_per_period,
- int duration_ms);
- ~CpuMeasurementHarness();
- bool Run();
- int AverageCpu();
-
- protected:
- CpuMeasurementHarness(CpuTarget* target, int work_period_ms,
- int work_iterations_per_period, int duration_ms);
-
- private:
- bool WaitForCpuInit();
- void Measure();
- bool DoWork();
-
- CpuTarget* cpu_target_;
- const int work_period_ms_;
- const int work_iterations_per_period_;
- const int duration_ms_;
- int cpu_sum_;
- int cpu_iterations_;
- scoped_ptr<CpuWrapper> cpu_;
- scoped_ptr<EventWrapper> event_;
-};
-
-} // namespace webrtc
-
-#endif // SRC_SYSTEM_WRAPPERS_SOURCE_CPU_MEASUREMENT_HARNESS_H_
diff --git a/trunk/src/system_wrappers/source/cpu_no_op.cc b/trunk/src/system_wrappers/source/cpu_no_op.cc
deleted file mode 100644
index e42ef91..0000000
--- a/trunk/src/system_wrappers/source/cpu_no_op.cc
+++ /dev/null
@@ -1,22 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stddef.h>
-
-#include "cpu_wrapper.h"
-
-namespace webrtc {
-
-CpuWrapper* CpuWrapper::CreateCpu()
-{
- return NULL;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/cpu_win.cc b/trunk/src/system_wrappers/source/cpu_win.cc
deleted file mode 100644
index 5792023..0000000
--- a/trunk/src/system_wrappers/source/cpu_win.cc
+++ /dev/null
@@ -1,534 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "cpu_win.h"
-
-#define _WIN32_DCOM
-
-#include <assert.h>
-#include <iostream>
-#include <Wbemidl.h>
-
-#pragma comment(lib, "wbemuuid.lib")
-
-#include "condition_variable_wrapper.h"
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-
-namespace webrtc {
-WebRtc_Word32 CpuWindows::CpuUsage()
-{
- if (!has_initialized_)
- {
- return -1;
- }
- // Last element is the average
- return cpu_usage_[number_of_objects_ - 1];
-}
-
-WebRtc_Word32 CpuWindows::CpuUsageMultiCore(WebRtc_UWord32& num_cores,
- WebRtc_UWord32*& cpu_usage)
-{
- if (has_terminated_) {
- num_cores = 0;
- cpu_usage = NULL;
- return -1;
- }
- if (!has_initialized_)
- {
- num_cores = 0;
- cpu_usage = NULL;
- return -1;
- }
- num_cores = number_of_objects_ - 1;
- cpu_usage = cpu_usage_;
- return cpu_usage_[number_of_objects_-1];
-}
-
-CpuWindows::CpuWindows()
- : cpu_polling_thread(NULL),
- initialize_(true),
- has_initialized_(false),
- terminate_(false),
- has_terminated_(false),
- cpu_usage_(NULL),
- wbem_enum_access_(NULL),
- number_of_objects_(0),
- cpu_usage_handle_(0),
- previous_processor_timestamp_(NULL),
- timestamp_sys_100_ns_handle_(0),
- previous_100ns_timestamp_(NULL),
- wbem_service_(NULL),
- wbem_service_proxy_(NULL),
- wbem_refresher_(NULL),
- wbem_enum_(NULL)
-{
- // All resources are allocated in PollingCpu().
- if (AllocateComplexDataTypes())
- {
- StartPollingCpu();
- }
- else
- {
- assert(false);
- }
-}
-
-CpuWindows::~CpuWindows()
-{
- // All resources are reclaimed in StopPollingCpu().
- const bool success = StopPollingCpu();
- assert(success);
- DeAllocateComplexDataTypes();
-}
-
-bool CpuWindows::AllocateComplexDataTypes()
-{
- cpu_polling_thread = ThreadWrapper::CreateThread(
- CpuWindows::Process,
- reinterpret_cast<void*>(this),
- kNormalPriority,
- "CpuWindows");
- init_crit_ = CriticalSectionWrapper::CreateCriticalSection();
- init_cond_ = ConditionVariableWrapper::CreateConditionVariable();
- terminate_crit_ = CriticalSectionWrapper::CreateCriticalSection();
- terminate_cond_ = ConditionVariableWrapper::CreateConditionVariable();
- sleep_event = EventWrapper::Create();
- return (cpu_polling_thread != NULL) && (init_crit_ != NULL) &&
- (init_cond_ != NULL) && (terminate_crit_ != NULL) &&
- (terminate_cond_ != NULL) && (sleep_event != NULL);
-}
-
-void CpuWindows::DeAllocateComplexDataTypes()
-{
- if (sleep_event != NULL)
- {
- delete sleep_event;
- sleep_event = NULL;
- }
- if (terminate_cond_ != NULL)
- {
- delete terminate_cond_;
- terminate_cond_ = NULL;
- }
- if (terminate_crit_ != NULL)
- {
- delete terminate_crit_;
- terminate_crit_ = NULL;
- }
- if (init_cond_ != NULL)
- {
- delete init_cond_;
- init_cond_ = NULL;
- }
- if (init_crit_ != NULL)
- {
- delete init_crit_;
- init_crit_ = NULL;
- }
- if (cpu_polling_thread != NULL)
- {
- delete cpu_polling_thread;
- cpu_polling_thread = NULL;
- }
-}
-
-void CpuWindows::StartPollingCpu()
-{
- unsigned int dummy_id = 0;
- if (!cpu_polling_thread->Start(dummy_id))
- {
- initialize_ = false;
- has_terminated_ = true;
- assert(false);
- }
-}
-
-bool CpuWindows::StopPollingCpu()
-{
- {
- // If StopPollingCpu is called immediately after StartPollingCpu() it is
- // possible that cpu_polling_thread is in the process of initializing.
- // Let initialization finish to avoid getting into a bad state.
- CriticalSectionScoped cs(init_crit_);
- while(initialize_)
- {
- init_cond_->SleepCS(*init_crit_);
- }
- }
-
- CriticalSectionScoped cs(terminate_crit_);
- terminate_ = true;
- sleep_event->Set();
- while (!has_terminated_)
- {
- terminate_cond_->SleepCS(*terminate_crit_);
- }
- cpu_polling_thread->Stop();
- delete cpu_polling_thread;
- cpu_polling_thread = NULL;
- return true;
-}
-
-bool CpuWindows::Process(void* thread_object)
-{
- return reinterpret_cast<CpuWindows*>(thread_object)->ProcessImpl();
-}
-
-bool CpuWindows::ProcessImpl()
-{
- {
- CriticalSectionScoped cs(terminate_crit_);
- if (terminate_)
- {
- const bool success = Terminate();
- assert(success);
- terminate_cond_->WakeAll();
- return false;
- }
- }
- // Initialize on first iteration
- if (initialize_)
- {
- CriticalSectionScoped cs(init_crit_);
- initialize_ = false;
- const bool success = Initialize();
- init_cond_->WakeAll();
- if (!success || !has_initialized_)
- {
- has_initialized_ = false;
- terminate_ = true;
- return true;
- }
- }
- // Approximately one seconds sleep for each CPU measurement. Precision is
- // not important. 1 second refresh rate is also used by Performance Monitor
- // (perfmon).
- if(kEventTimeout != sleep_event->Wait(1000))
- {
- // Terminating. No need to update CPU usage.
- assert(terminate_);
- return true;
- }
-
- // UpdateCpuUsage() returns false if a single (or more) CPU read(s) failed.
- // Not a major problem if it happens but make sure it doesnt trigger in
- // debug.
- const bool success = UpdateCpuUsage();
- assert(success);
- return true;
-}
-
-bool CpuWindows::CreateWmiConnection()
-{
- IWbemLocator* service_locator = NULL;
- HRESULT hr = CoCreateInstance(CLSID_WbemLocator, NULL,
- CLSCTX_INPROC_SERVER, IID_IWbemLocator,
- reinterpret_cast<void**> (&service_locator));
- if (FAILED(hr))
- {
- return false;
- }
- // To get the WMI service specify the WMI namespace.
- BSTR wmi_namespace = SysAllocString(L"\\\\.\\root\\cimv2");
- if (wmi_namespace == NULL)
- {
- // This type of failure signifies running out of memory.
- service_locator->Release();
- return false;
- }
- hr = service_locator->ConnectServer(wmi_namespace, NULL, NULL, NULL, 0L,
- NULL, NULL, &wbem_service_);
- SysFreeString(wmi_namespace);
- service_locator->Release();
- return !FAILED(hr);
-}
-
-// Sets up WMI refresher and enum
-bool CpuWindows::CreatePerfOsRefresher()
-{
- // Create refresher.
- HRESULT hr = CoCreateInstance(CLSID_WbemRefresher, NULL,
- CLSCTX_INPROC_SERVER, IID_IWbemRefresher,
- reinterpret_cast<void**> (&wbem_refresher_));
- if (FAILED(hr))
- {
- return false;
- }
- // Create PerfOS_Processor enum.
- IWbemConfigureRefresher* wbem_refresher_config = NULL;
- hr = wbem_refresher_->QueryInterface(
- IID_IWbemConfigureRefresher,
- reinterpret_cast<void**> (&wbem_refresher_config));
- if (FAILED(hr))
- {
- return false;
- }
-
- // Create a proxy to the IWbemServices so that a local authentication
- // can be set up (this is needed to be able to successfully call
- // IWbemConfigureRefresher::AddEnum). Setting authentication with
- // CoInitializeSecurity is process-wide (which is too intrusive).
- hr = CoCopyProxy(static_cast<IUnknown*> (wbem_service_),
- reinterpret_cast<IUnknown**> (&wbem_service_proxy_));
- if(FAILED(hr))
- {
- return false;
- }
- // Set local authentication.
- // RPC_C_AUTHN_WINNT means using NTLM instead of Kerberos which is default.
- hr = CoSetProxyBlanket(static_cast<IUnknown*> (wbem_service_proxy_),
- RPC_C_AUTHN_WINNT, RPC_C_AUTHZ_NONE, NULL,
- RPC_C_AUTHN_LEVEL_DEFAULT,
- RPC_C_IMP_LEVEL_IMPERSONATE, NULL, EOAC_NONE);
- if(FAILED(hr))
- {
- return false;
- }
-
- // Don't care about the particular id for the enum.
- long enum_id = 0;
- hr = wbem_refresher_config->AddEnum(wbem_service_proxy_,
- L"Win32_PerfRawData_PerfOS_Processor",
- 0, NULL, &wbem_enum_, &enum_id);
- wbem_refresher_config->Release();
- wbem_refresher_config = NULL;
- return !FAILED(hr);
-}
-
-// Have to pull the first round of data to be able set the handles.
-bool CpuWindows::CreatePerfOsCpuHandles()
-{
- // Update the refresher so that there is data available in wbem_enum_.
- wbem_refresher_->Refresh(0L);
-
- // The number of enumerators is the number of processor + 1 (the total).
- // This is unknown at this point.
- DWORD number_returned = 0;
- HRESULT hr = wbem_enum_->GetObjects(0L, number_of_objects_,
- wbem_enum_access_, &number_returned);
- // number_returned indicates the number of enumerators that are needed.
- if (hr == WBEM_E_BUFFER_TOO_SMALL &&
- number_returned > number_of_objects_)
- {
- // Allocate the number IWbemObjectAccess asked for by the
- // GetObjects(..) function.
- wbem_enum_access_ = new IWbemObjectAccess*[number_returned];
- cpu_usage_ = new WebRtc_UWord32[number_returned];
- previous_processor_timestamp_ = new unsigned __int64[number_returned];
- previous_100ns_timestamp_ = new unsigned __int64[number_returned];
- if ((wbem_enum_access_ == NULL) || (cpu_usage_ == NULL) ||
- (previous_processor_timestamp_ == NULL) ||
- (previous_100ns_timestamp_ == NULL))
- {
- // Out of memory.
- return false;
- }
-
- SecureZeroMemory(wbem_enum_access_, number_returned *
- sizeof(IWbemObjectAccess*));
- memset(cpu_usage_, 0, sizeof(int) * number_returned);
- memset(previous_processor_timestamp_, 0, sizeof(unsigned __int64) *
- number_returned);
- memset(previous_100ns_timestamp_, 0, sizeof(unsigned __int64) *
- number_returned);
-
- number_of_objects_ = number_returned;
- // Read should be successfull now that memory has been allocated.
- hr = wbem_enum_->GetObjects(0L, number_of_objects_, wbem_enum_access_,
- &number_returned);
- if (FAILED(hr))
- {
- return false;
- }
- }
- else
- {
- // 0 enumerators should not be enough. Something has gone wrong here.
- return false;
- }
-
- // Get the enumerator handles that are needed for calculating CPU usage.
- CIMTYPE cpu_usage_type;
- hr = wbem_enum_access_[0]->GetPropertyHandle(L"PercentProcessorTime",
- &cpu_usage_type,
- &cpu_usage_handle_);
- if (FAILED(hr))
- {
- return false;
- }
- CIMTYPE timestamp_sys_100_ns_type;
- hr = wbem_enum_access_[0]->GetPropertyHandle(L"TimeStamp_Sys100NS",
- ×tamp_sys_100_ns_type,
- ×tamp_sys_100_ns_handle_);
- return !FAILED(hr);
-}
-
-bool CpuWindows::Initialize()
-{
- if (terminate_)
- {
- return false;
- }
- // Initialize COM library.
- HRESULT hr = CoInitializeEx(NULL,COINIT_MULTITHREADED);
- if (FAILED(hr))
- {
- return false;
- }
- if (FAILED(hr))
- {
- return false;
- }
-
- if (!CreateWmiConnection())
- {
- return false;
- }
- if (!CreatePerfOsRefresher())
- {
- return false;
- }
- if (!CreatePerfOsCpuHandles())
- {
- return false;
- }
- has_initialized_ = true;
- return true;
-}
-
-bool CpuWindows::Terminate()
-{
- if (has_terminated_)
- {
- return false;
- }
- // Reverse order of Initialize().
- // Some compilers complain about deleting NULL though it's well defined
- if (previous_100ns_timestamp_ != NULL)
- {
- delete[] previous_100ns_timestamp_;
- previous_100ns_timestamp_ = NULL;
- }
- if (previous_processor_timestamp_ != NULL)
- {
- delete[] previous_processor_timestamp_;
- previous_processor_timestamp_ = NULL;
- }
- if (cpu_usage_ != NULL)
- {
- delete[] cpu_usage_;
- cpu_usage_ = NULL;
- }
- if (wbem_enum_access_ != NULL)
- {
- for (DWORD i = 0; i < number_of_objects_; i++)
- {
- if(wbem_enum_access_[i] != NULL)
- {
- wbem_enum_access_[i]->Release();
- }
- }
- delete[] wbem_enum_access_;
- wbem_enum_access_ = NULL;
- }
- if (wbem_enum_ != NULL)
- {
- wbem_enum_->Release();
- wbem_enum_ = NULL;
- }
- if (wbem_refresher_ != NULL)
- {
- wbem_refresher_->Release();
- wbem_refresher_ = NULL;
- }
- if (wbem_service_proxy_ != NULL)
- {
- wbem_service_proxy_->Release();
- wbem_service_proxy_ = NULL;
- }
- if (wbem_service_ != NULL)
- {
- wbem_service_->Release();
- wbem_service_ = NULL;
- }
- // CoUninitialized should be called once for every CoInitializeEx.
- // Regardless if it failed or not.
- CoUninitialize();
- has_terminated_ = true;
- return true;
-}
-
-bool CpuWindows::UpdateCpuUsage()
-{
- wbem_refresher_->Refresh(0L);
- DWORD number_returned = 0;
- HRESULT hr = wbem_enum_->GetObjects(0L, number_of_objects_,
- wbem_enum_access_,&number_returned);
- if (FAILED(hr))
- {
- // wbem_enum_access_ has already been allocated. Unless the number of
- // CPUs change runtime this should not happen.
- return false;
- }
- unsigned __int64 cpu_usage = 0;
- unsigned __int64 timestamp_100ns = 0;
- bool returnValue = true;
- for (DWORD i = 0; i < number_returned; i++)
- {
- hr = wbem_enum_access_[i]->ReadQWORD(cpu_usage_handle_,&cpu_usage);
- if (FAILED(hr))
- {
- returnValue = false;
- }
- hr = wbem_enum_access_[i]->ReadQWORD(timestamp_sys_100_ns_handle_,
- ×tamp_100ns);
- if (FAILED(hr))
- {
- returnValue = false;
- }
- wbem_enum_access_[i]->Release();
- wbem_enum_access_[i] = NULL;
-
- const bool wrapparound =
- (previous_processor_timestamp_[i] > cpu_usage) ||
- (previous_100ns_timestamp_[i] > timestamp_100ns);
- const bool first_time = (previous_processor_timestamp_[i] == 0) ||
- (previous_100ns_timestamp_[i] == 0);
- if (wrapparound || first_time)
- {
- previous_processor_timestamp_[i] = cpu_usage;
- previous_100ns_timestamp_[i] = timestamp_100ns;
- continue;
- }
- const unsigned __int64 processor_timestamp_delta =
- cpu_usage - previous_processor_timestamp_[i];
- const unsigned __int64 timestamp_100ns_delta =
- timestamp_100ns - previous_100ns_timestamp_[i];
-
- if (processor_timestamp_delta >= timestamp_100ns_delta)
- {
- cpu_usage_[i] = 0;
- } else {
- // Quotient must be float since the division is guaranteed to yield
- // a value between 0 and 1 which is 0 in integer division.
- const float delta_quotient =
- static_cast<float>(processor_timestamp_delta) /
- static_cast<float>(timestamp_100ns_delta);
- cpu_usage_[i] = 100 - static_cast<WebRtc_UWord32>(delta_quotient *
- 100);
- }
- previous_processor_timestamp_[i] = cpu_usage;
- previous_100ns_timestamp_[i] = timestamp_100ns;
- }
- return returnValue;
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/cpu_win.h b/trunk/src/system_wrappers/source/cpu_win.h
deleted file mode 100644
index d15073c..0000000
--- a/trunk/src/system_wrappers/source/cpu_win.h
+++ /dev/null
@@ -1,103 +0,0 @@
-// This file contains a Windows implementation of CpuWrapper.
-// Note: Windows XP, Windows Server 2003 are the minimum requirements.
-// The requirements are due to the implementation being based on
-// WMI.
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_WINDOWS_NO_CPOL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_WINDOWS_NO_CPOL_H_
-
-#include "cpu_wrapper.h"
-
-#include <Wbemidl.h>
-
-namespace webrtc {
-class ConditionVariableWrapper;
-class CriticalSectionWrapper;
-class EventWrapper;
-class ThreadWrapper;
-
-class CpuWindows : public CpuWrapper
-{
-public:
- virtual WebRtc_Word32 CpuUsage();
- virtual WebRtc_Word32 CpuUsage(WebRtc_Word8* /*pProcessName*/,
- WebRtc_UWord32 /*length*/) {return -1;}
- virtual WebRtc_Word32 CpuUsage(WebRtc_UWord32 /*dwProcessID*/) {return -1;}
-
- virtual WebRtc_Word32 CpuUsageMultiCore(WebRtc_UWord32& num_cores,
- WebRtc_UWord32*& cpu_usage);
-
- virtual void Reset() {}
- virtual void Stop() {}
-
- CpuWindows();
- virtual ~CpuWindows();
-private:
- bool AllocateComplexDataTypes();
- void DeAllocateComplexDataTypes();
-
- void StartPollingCpu();
- bool StopPollingCpu();
-
- static bool Process(void* thread_object);
- bool ProcessImpl();
-
- bool CreateWmiConnection();
- bool CreatePerfOsRefresher();
- bool CreatePerfOsCpuHandles();
- bool Initialize();
- bool Terminate();
-
- bool UpdateCpuUsage();
-
- ThreadWrapper* cpu_polling_thread;
-
- bool initialize_;
- bool has_initialized_;
- CriticalSectionWrapper* init_crit_;
- ConditionVariableWrapper* init_cond_;
-
- bool terminate_;
- bool has_terminated_;
- CriticalSectionWrapper* terminate_crit_;
- ConditionVariableWrapper* terminate_cond_;
-
- // For sleep with wake-up functionality.
- EventWrapper* sleep_event;
-
- // Will be an array. Just care about CPU 0 for now.
- WebRtc_UWord32* cpu_usage_;
-
- // One IWbemObjectAccess for each processor and one for the total.
- // 0-n-1 is the individual processors.
- // n is the total.
- IWbemObjectAccess** wbem_enum_access_;
- DWORD number_of_objects_;
-
- // Cpu timestamp
- long cpu_usage_handle_;
- unsigned __int64* previous_processor_timestamp_;
-
- // Timestamp
- long timestamp_sys_100_ns_handle_;
- unsigned __int64* previous_100ns_timestamp_;
-
- IWbemServices* wbem_service_;
- IWbemServices* wbem_service_proxy_;
-
- IWbemRefresher* wbem_refresher_;
-
- IWbemHiPerfEnum* wbem_enum_;
-
-};
-} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CPU_WINDOWS_NO_CPOL_H_
diff --git a/trunk/src/system_wrappers/source/cpu_wrapper_unittest.cc b/trunk/src/system_wrappers/source/cpu_wrapper_unittest.cc
deleted file mode 100644
index cd149dc..0000000
--- a/trunk/src/system_wrappers/source/cpu_wrapper_unittest.cc
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "system_wrappers/interface/cpu_wrapper.h"
-
-#include "gtest/gtest.h"
-#include "system_wrappers/interface/cpu_info.h"
-#include "system_wrappers/interface/event_wrapper.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-#include "system_wrappers/interface/trace.h"
-#include "testsupport/fileutils.h"
-
-using webrtc::CpuInfo;
-using webrtc::CpuWrapper;
-using webrtc::EventWrapper;
-using webrtc::scoped_ptr;
-using webrtc::Trace;
-
-// This test is flaky on Windows/Release.
-// http://code.google.com/p/webrtc/issues/detail?id=290
-#ifdef _WIN32
-#define MAYBE_Usage DISABLED_Usage
-#else
-#define MAYBE_Usage Usage
-#endif
-TEST(CpuWrapperTest, MAYBE_Usage) {
- Trace::CreateTrace();
- std::string trace_file = webrtc::test::OutputPath() +
- "cpu_wrapper_unittest.txt";
- Trace::SetTraceFile(trace_file.c_str());
- Trace::SetLevelFilter(webrtc::kTraceAll);
- printf("Number of cores detected:%u\n", CpuInfo::DetectNumberOfCores());
- scoped_ptr<CpuWrapper> cpu(CpuWrapper::CreateCpu());
- ASSERT_TRUE(cpu.get() != NULL);
- scoped_ptr<EventWrapper> sleep_event(EventWrapper::Create());
- ASSERT_TRUE(sleep_event.get() != NULL);
-
- int num_iterations = 0;
- WebRtc_UWord32 num_cores = 0;
- WebRtc_UWord32* cores = NULL;
- bool cpu_usage_available = cpu->CpuUsageMultiCore(num_cores, cores) != -1;
- // Initializing the CPU measurements may take a couple of seconds on Windows.
- // Since the initialization is lazy we need to wait until it is completed.
- // Should not take more than 10000 ms.
- while (!cpu_usage_available && (++num_iterations < 10000)) {
- if (cores != NULL) {
- ASSERT_GT(num_cores, 0u);
- break;
- }
- sleep_event->Wait(1);
- cpu_usage_available = cpu->CpuUsageMultiCore(num_cores, cores) != -1;
- }
- ASSERT_TRUE(cpu_usage_available);
-
- const WebRtc_Word32 average = cpu->CpuUsageMultiCore(num_cores, cores);
- ASSERT_TRUE(cores != NULL);
- EXPECT_GT(num_cores, 0u);
- EXPECT_GE(average, 0);
- EXPECT_LE(average, 100);
-
- printf("\nNumber of cores:%d\n", num_cores);
- printf("Average cpu:%d\n", average);
- for (WebRtc_UWord32 i = 0; i < num_cores; i++) {
- printf("Core:%u CPU:%u \n", i, cores[i]);
- EXPECT_GE(cores[i], 0u);
- EXPECT_LE(cores[i], 100u);
- }
-
- Trace::ReturnTrace();
-};
diff --git a/trunk/src/system_wrappers/source/critical_section.cc b/trunk/src/system_wrappers/source/critical_section.cc
deleted file mode 100644
index d3f3f01..0000000
--- a/trunk/src/system_wrappers/source/critical_section.cc
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#if defined(_WIN32)
- #include <windows.h>
- #include "critical_section_win.h"
-#else
- #include "critical_section_posix.h"
-#endif
-
-namespace webrtc {
-CriticalSectionWrapper* CriticalSectionWrapper::CreateCriticalSection()
-{
-#ifdef _WIN32
- return new CriticalSectionWindows();
-#else
- return new CriticalSectionPosix();
-#endif
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/critical_section_posix.cc b/trunk/src/system_wrappers/source/critical_section_posix.cc
deleted file mode 100644
index 70f85f9..0000000
--- a/trunk/src/system_wrappers/source/critical_section_posix.cc
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// General note: return values for the various pthread synchronization APIs
-// are explicitly ignored here. In Chromium, the same thing is done for release.
-// However, in debugging, failure in these APIs are logged. There is currently
-// no equivalent to DCHECK_EQ in WebRTC code so this is the best we can do here.
-// TODO(henrike): add logging when pthread synchronization APIs are failing.
-
-#include "critical_section_posix.h"
-
-namespace webrtc {
-
-CriticalSectionPosix::CriticalSectionPosix()
-{
- pthread_mutexattr_t attr;
- (void) pthread_mutexattr_init(&attr);
- (void) pthread_mutexattr_settype(&attr, PTHREAD_MUTEX_RECURSIVE);
- (void) pthread_mutex_init(&_mutex, &attr);
-}
-
-CriticalSectionPosix::~CriticalSectionPosix()
-{
- (void) pthread_mutex_destroy(&_mutex);
-}
-
-void
-CriticalSectionPosix::Enter()
-{
- (void) pthread_mutex_lock(&_mutex);
-}
-
-void
-CriticalSectionPosix::Leave()
-{
- (void) pthread_mutex_unlock(&_mutex);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/critical_section_posix.h b/trunk/src/system_wrappers/source/critical_section_posix.h
deleted file mode 100644
index 40b7dc9..0000000
--- a/trunk/src/system_wrappers/source/critical_section_posix.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_POSIX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_POSIX_H_
-
-#include "critical_section_wrapper.h"
-
-#include <pthread.h>
-
-namespace webrtc {
-class CriticalSectionPosix : public CriticalSectionWrapper
-{
-public:
- CriticalSectionPosix();
-
- virtual ~CriticalSectionPosix();
-
- virtual void Enter();
- virtual void Leave();
-
-private:
- pthread_mutex_t _mutex;
- friend class ConditionVariablePosix;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_POSIX_H_
diff --git a/trunk/src/system_wrappers/source/critical_section_win.cc b/trunk/src/system_wrappers/source/critical_section_win.cc
deleted file mode 100644
index bbc66e5..0000000
--- a/trunk/src/system_wrappers/source/critical_section_win.cc
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "critical_section_win.h"
-
-namespace webrtc {
-CriticalSectionWindows::CriticalSectionWindows()
-{
- InitializeCriticalSection(&crit);
-}
-
-CriticalSectionWindows::~CriticalSectionWindows()
-{
- DeleteCriticalSection(&crit);
-}
-
-void
-CriticalSectionWindows::Enter()
-{
- EnterCriticalSection(&crit);
-}
-
-void
-CriticalSectionWindows::Leave()
-{
- LeaveCriticalSection(&crit);
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/critical_section_win.h b/trunk/src/system_wrappers/source/critical_section_win.h
deleted file mode 100644
index 9556fa9..0000000
--- a/trunk/src/system_wrappers/source/critical_section_win.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_WINDOWS_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_WINDOWS_H_
-
-#include "typedefs.h"
-#include "critical_section_wrapper.h"
-#include <windows.h>
-
-namespace webrtc {
-class CriticalSectionWindows : public CriticalSectionWrapper
-{
-public:
- CriticalSectionWindows();
-
- virtual ~CriticalSectionWindows();
-
- virtual void Enter();
- virtual void Leave();
-
-private:
- CRITICAL_SECTION crit;
-
- friend class ConditionVariableWindows;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_CRITICAL_SECTION_WINDOWS_H_
diff --git a/trunk/src/system_wrappers/source/data_log.cc b/trunk/src/system_wrappers/source/data_log.cc
deleted file mode 100644
index f123896..0000000
--- a/trunk/src/system_wrappers/source/data_log.cc
+++ /dev/null
@@ -1,455 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "data_log.h"
-
-#include <assert.h>
-
-#include <algorithm>
-#include <list>
-
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "file_wrapper.h"
-#include "rw_lock_wrapper.h"
-#include "thread_wrapper.h"
-
-namespace webrtc {
-
-DataLogImpl::CritSectScopedPtr DataLogImpl::crit_sect_(
- CriticalSectionWrapper::CreateCriticalSection());
-
-DataLogImpl* DataLogImpl::instance_ = NULL;
-
-// A Row contains cells, which are indexed by the column names as std::string.
-// The string index is treated in a case sensitive way.
-class Row {
- public:
- Row();
- ~Row();
-
- // Inserts a Container into the cell of the column specified with
- // column_name.
- // column_name is treated in a case sensitive way.
- int InsertCell(const std::string& column_name,
- const Container* value_container);
-
- // Converts the value at the column specified by column_name to a string
- // stored in value_string.
- // column_name is treated in a case sensitive way.
- void ToString(const std::string& column_name, std::string* value_string);
-
- private:
- // Collection of containers indexed by column name as std::string
- typedef std::map<std::string, const Container*> CellMap;
-
- CellMap cells_;
- CriticalSectionWrapper* cells_lock_;
-};
-
-// A LogTable contains multiple rows, where only the latest row is active for
-// editing. The rows are defined by the ColumnMap, which contains the name of
-// each column and the length of the column (1 for one-value-columns and greater
-// than 1 for multi-value-columns).
-class LogTable {
- public:
- LogTable();
- ~LogTable();
-
- // Adds the column with name column_name to the table. The column will be a
- // multi-value-column if multi_value_length is greater than 1.
- // column_name is treated in a case sensitive way.
- int AddColumn(const std::string& column_name, int multi_value_length);
-
- // Buffers the current row while it is waiting to be written to file,
- // which is done by a call to Flush(). A new row is available when the
- // function returns
- void NextRow();
-
- // Inserts a Container into the cell of the column specified with
- // column_name.
- // column_name is treated in a case sensitive way.
- int InsertCell(const std::string& column_name,
- const Container* value_container);
-
- // Creates a log file, named as specified in the string file_name, to
- // where the table will be written when calling Flush().
- int CreateLogFile(const std::string& file_name);
-
- // Write all complete rows to file.
- // May not be called by two threads simultaneously (doing so may result in
- // a race condition). Will be called by the file_writer_thread_ when that
- // thread is running.
- void Flush();
-
- private:
- // Collection of multi_value_lengths indexed by column name as std::string
- typedef std::map<std::string, int> ColumnMap;
- typedef std::list<Row*> RowList;
-
- ColumnMap columns_;
- RowList rows_[2];
- RowList* rows_history_;
- RowList* rows_flush_;
- Row* current_row_;
- FileWrapper* file_;
- bool write_header_;
- CriticalSectionWrapper* table_lock_;
-};
-
-Row::Row()
- : cells_(),
- cells_lock_(CriticalSectionWrapper::CreateCriticalSection()) {
-}
-
-Row::~Row() {
- for (CellMap::iterator it = cells_.begin(); it != cells_.end();) {
- delete it->second;
- // For maps all iterators (except the erased) are valid after an erase
- cells_.erase(it++);
- }
- delete cells_lock_;
-}
-
-int Row::InsertCell(const std::string& column_name,
- const Container* value_container) {
- CriticalSectionScoped synchronize(cells_lock_);
- assert(cells_.count(column_name) == 0);
- if (cells_.count(column_name) > 0)
- return -1;
- cells_[column_name] = value_container;
- return 0;
-}
-
-void Row::ToString(const std::string& column_name,
- std::string* value_string) {
- CriticalSectionScoped synchronize(cells_lock_);
- const Container* container = cells_[column_name];
- if (container == NULL) {
- *value_string = "NaN,";
- return;
- }
- container->ToString(value_string);
-}
-
-LogTable::LogTable()
- : columns_(),
- rows_(),
- rows_history_(&rows_[0]),
- rows_flush_(&rows_[1]),
- current_row_(new Row),
- file_(FileWrapper::Create()),
- write_header_(true),
- table_lock_(CriticalSectionWrapper::CreateCriticalSection()) {
-}
-
-LogTable::~LogTable() {
- for (RowList::iterator row_it = rows_history_->begin();
- row_it != rows_history_->end();) {
- delete *row_it;
- row_it = rows_history_->erase(row_it);
- }
- for (ColumnMap::iterator col_it = columns_.begin();
- col_it != columns_.end();) {
- // For maps all iterators (except the erased) are valid after an erase
- columns_.erase(col_it++);
- }
- if (file_ != NULL) {
- file_->Flush();
- file_->CloseFile();
- delete file_;
- }
- delete current_row_;
- delete table_lock_;
-}
-
-int LogTable::AddColumn(const std::string& column_name,
- int multi_value_length) {
- assert(multi_value_length > 0);
- if (!write_header_) {
- // It's not allowed to add new columns after the header
- // has been written.
- assert(false);
- return -1;
- } else {
- CriticalSectionScoped synchronize(table_lock_);
- if (write_header_)
- columns_[column_name] = multi_value_length;
- else
- return -1;
- }
- return 0;
-}
-
-void LogTable::NextRow() {
- CriticalSectionScoped sync_rows(table_lock_);
- rows_history_->push_back(current_row_);
- current_row_ = new Row;
-}
-
-int LogTable::InsertCell(const std::string& column_name,
- const Container* value_container) {
- CriticalSectionScoped synchronize(table_lock_);
- assert(columns_.count(column_name) > 0);
- if (columns_.count(column_name) == 0)
- return -1;
- return current_row_->InsertCell(column_name, value_container);
-}
-
-int LogTable::CreateLogFile(const std::string& file_name) {
- if (file_name.length() == 0)
- return -1;
- if (file_->Open())
- return -1;
- file_->OpenFile(file_name.c_str(),
- false, // Open with read/write permissions
- false, // Don't wraparound and write at the beginning when
- // the file is full
- true); // Open as a text file
- if (file_ == NULL)
- return -1;
- return 0;
-}
-
-void LogTable::Flush() {
- ColumnMap::iterator column_it;
- bool commit_header = false;
- if (write_header_) {
- CriticalSectionScoped synchronize(table_lock_);
- if (write_header_) {
- commit_header = true;
- write_header_ = false;
- }
- }
- if (commit_header) {
- for (column_it = columns_.begin();
- column_it != columns_.end(); ++column_it) {
- if (column_it->second > 1) {
- file_->WriteText("%s[%u],", column_it->first.c_str(),
- column_it->second);
- for (int i = 1; i < column_it->second; ++i)
- file_->WriteText(",");
- } else {
- file_->WriteText("%s,", column_it->first.c_str());
- }
- }
- if (columns_.size() > 0)
- file_->WriteText("\n");
- }
-
- // Swap the list used for flushing with the list containing the row history
- // and clear the history. We also create a local pointer to the new
- // list used for flushing to avoid race conditions if another thread
- // calls this function while we are writing.
- // We don't want to block the list while we're writing to file.
- {
- CriticalSectionScoped synchronize(table_lock_);
- RowList* tmp = rows_flush_;
- rows_flush_ = rows_history_;
- rows_history_ = tmp;
- rows_history_->clear();
- }
-
- // Write all complete rows to file and delete them
- for (RowList::iterator row_it = rows_flush_->begin();
- row_it != rows_flush_->end();) {
- for (column_it = columns_.begin();
- column_it != columns_.end(); ++column_it) {
- std::string row_string;
- (*row_it)->ToString(column_it->first, &row_string);
- file_->WriteText("%s", row_string.c_str());
- }
- if (columns_.size() > 0)
- file_->WriteText("\n");
- delete *row_it;
- row_it = rows_flush_->erase(row_it);
- }
-}
-
-int DataLog::CreateLog() {
- return DataLogImpl::CreateLog();
-}
-
-void DataLog::ReturnLog() {
- return DataLogImpl::ReturnLog();
-}
-
-std::string DataLog::Combine(const std::string& table_name, int table_id) {
- std::stringstream ss;
- std::string combined_id = table_name;
- std::string number_suffix;
- ss << "_" << table_id;
- ss >> number_suffix;
- combined_id += number_suffix;
- std::transform(combined_id.begin(), combined_id.end(), combined_id.begin(),
- ::tolower);
- return combined_id;
-}
-
-int DataLog::AddTable(const std::string& table_name) {
- DataLogImpl* data_log = DataLogImpl::StaticInstance();
- if (data_log == NULL)
- return -1;
- return data_log->AddTable(table_name);
-}
-
-int DataLog::AddColumn(const std::string& table_name,
- const std::string& column_name,
- int multi_value_length) {
- DataLogImpl* data_log = DataLogImpl::StaticInstance();
- if (data_log == NULL)
- return -1;
- return data_log->DataLogImpl::StaticInstance()->AddColumn(table_name,
- column_name,
- multi_value_length);
-}
-
-int DataLog::NextRow(const std::string& table_name) {
- DataLogImpl* data_log = DataLogImpl::StaticInstance();
- if (data_log == NULL)
- return -1;
- return data_log->DataLogImpl::StaticInstance()->NextRow(table_name);
-}
-
-DataLogImpl::DataLogImpl()
- : counter_(1),
- tables_(),
- flush_event_(EventWrapper::Create()),
- file_writer_thread_(NULL),
- tables_lock_(RWLockWrapper::CreateRWLock()) {
-}
-
-DataLogImpl::~DataLogImpl() {
- StopThread();
- Flush(); // Write any remaining rows
- delete file_writer_thread_;
- delete flush_event_;
- for (TableMap::iterator it = tables_.begin(); it != tables_.end();) {
- delete static_cast<LogTable*>(it->second);
- // For maps all iterators (except the erased) are valid after an erase
- tables_.erase(it++);
- }
- delete tables_lock_;
-}
-
-int DataLogImpl::CreateLog() {
- CriticalSectionScoped synchronize(crit_sect_.get());
- if (instance_ == NULL) {
- instance_ = new DataLogImpl();
- return instance_->Init();
- } else {
- ++instance_->counter_;
- }
- return 0;
-}
-
-int DataLogImpl::Init() {
- file_writer_thread_ = ThreadWrapper::CreateThread(
- DataLogImpl::Run,
- instance_,
- kHighestPriority,
- "DataLog");
- if (file_writer_thread_ == NULL)
- return -1;
- unsigned int thread_id = 0;
- bool success = file_writer_thread_->Start(thread_id);
- if (!success)
- return -1;
- return 0;
-}
-
-DataLogImpl* DataLogImpl::StaticInstance() {
- return instance_;
-}
-
-void DataLogImpl::ReturnLog() {
- CriticalSectionScoped synchronize(crit_sect_.get());
- if (instance_ && instance_->counter_ > 1) {
- --instance_->counter_;
- return;
- }
- delete instance_;
- instance_ = NULL;
-}
-
-int DataLogImpl::AddTable(const std::string& table_name) {
- WriteLockScoped synchronize(*tables_lock_);
- // Make sure we don't add a table which already exists
- if (tables_.count(table_name) > 0)
- return -1;
- tables_[table_name] = new LogTable();
- if (tables_[table_name]->CreateLogFile(table_name + ".txt") == -1)
- return -1;
- return 0;
-}
-
-int DataLogImpl::AddColumn(const std::string& table_name,
- const std::string& column_name,
- int multi_value_length) {
- ReadLockScoped synchronize(*tables_lock_);
- if (tables_.count(table_name) == 0)
- return -1;
- return tables_[table_name]->AddColumn(column_name, multi_value_length);
-}
-
-int DataLogImpl::InsertCell(const std::string& table_name,
- const std::string& column_name,
- const Container* value_container) {
- ReadLockScoped synchronize(*tables_lock_);
- assert(tables_.count(table_name) > 0);
- if (tables_.count(table_name) == 0)
- return -1;
- return tables_[table_name]->InsertCell(column_name, value_container);
-}
-
-int DataLogImpl::NextRow(const std::string& table_name) {
- ReadLockScoped synchronize(*tables_lock_);
- if (tables_.count(table_name) == 0)
- return -1;
- tables_[table_name]->NextRow();
- if (file_writer_thread_ == NULL) {
- // Write every row to file as they get complete.
- tables_[table_name]->Flush();
- } else {
- // Signal a complete row
- flush_event_->Set();
- }
- return 0;
-}
-
-void DataLogImpl::Flush() {
- ReadLockScoped synchronize(*tables_lock_);
- for (TableMap::iterator it = tables_.begin(); it != tables_.end(); ++it) {
- it->second->Flush();
- }
-}
-
-bool DataLogImpl::Run(void* obj) {
- static_cast<DataLogImpl*>(obj)->Process();
- return true;
-}
-
-void DataLogImpl::Process() {
- // Wait for a row to be complete
- flush_event_->Wait(WEBRTC_EVENT_INFINITE);
- Flush();
-}
-
-void DataLogImpl::StopThread() {
- if (file_writer_thread_ != NULL) {
- file_writer_thread_->SetNotAlive();
- flush_event_->Set();
- // Call Stop() repeatedly, waiting for the Flush() call in Process() to
- // finish.
- while (!file_writer_thread_->Stop()) continue;
- }
-}
-
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/data_log_c.cc b/trunk/src/system_wrappers/source/data_log_c.cc
deleted file mode 100644
index f8d7efd..0000000
--- a/trunk/src/system_wrappers/source/data_log_c.cc
+++ /dev/null
@@ -1,145 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This is the pure C wrapper of the DataLog class.
- */
-
-#include "system_wrappers/interface/data_log_c.h"
-
-#include <string>
-
-#include "system_wrappers/interface/data_log.h"
-
-extern "C" int WebRtcDataLog_CreateLog() {
- return webrtc::DataLog::CreateLog();
-}
-
-extern "C" void WebRtcDataLog_ReturnLog() {
- return webrtc::DataLog::ReturnLog();
-}
-
-extern "C" char* WebRtcDataLog_Combine(char* combined_name, size_t combined_len,
- const char* table_name, int table_id) {
- if (!table_name) return NULL;
- std::string combined = webrtc::DataLog::Combine(table_name, table_id);
- if (combined.size() >= combined_len) return NULL;
- std::copy(combined.begin(), combined.end(), combined_name);
- combined_name[combined.size()] = '\0';
- return combined_name;
-}
-
-extern "C" int WebRtcDataLog_AddTable(const char* table_name) {
- if (!table_name) return -1;
- return webrtc::DataLog::AddTable(table_name);
-}
-
-extern "C" int WebRtcDataLog_AddColumn(const char* table_name,
- const char* column_name,
- int multi_value_length) {
- if (!table_name || !column_name) return -1;
- return webrtc::DataLog::AddColumn(table_name, column_name,
- multi_value_length);
-}
-
-extern "C" int WebRtcDataLog_InsertCell_int(const char* table_name,
- const char* column_name,
- int value) {
- if (!table_name || !column_name) return -1;
- return webrtc::DataLog::InsertCell(table_name, column_name, value);
-}
-
-extern "C" int WebRtcDataLog_InsertArray_int(const char* table_name,
- const char* column_name,
- const int* values,
- int length) {
- if (!table_name || !column_name) return -1;
- return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
-}
-
-extern "C" int WebRtcDataLog_InsertCell_float(const char* table_name,
- const char* column_name,
- float value) {
- if (!table_name || !column_name) return -1;
- return webrtc::DataLog::InsertCell(table_name, column_name, value);
-}
-
-extern "C" int WebRtcDataLog_InsertArray_float(const char* table_name,
- const char* column_name,
- const float* values,
- int length) {
- if (!table_name || !column_name) return -1;
- return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
-}
-
-extern "C" int WebRtcDataLog_InsertCell_double(const char* table_name,
- const char* column_name,
- double value) {
- if (!table_name || !column_name) return -1;
- return webrtc::DataLog::InsertCell(table_name, column_name, value);
-}
-
-extern "C" int WebRtcDataLog_InsertArray_double(const char* table_name,
- const char* column_name,
- const double* values,
- int length) {
- if (!table_name || !column_name) return -1;
- return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
-}
-
-extern "C" int WebRtcDataLog_InsertCell_int32(const char* table_name,
- const char* column_name,
- int32_t value) {
- if (!table_name || !column_name) return -1;
- return webrtc::DataLog::InsertCell(table_name, column_name, value);
-}
-
-extern "C" int WebRtcDataLog_InsertArray_int32(const char* table_name,
- const char* column_name,
- const int32_t* values,
- int length) {
- if (!table_name || !column_name) return -1;
- return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
-}
-
-extern "C" int WebRtcDataLog_InsertCell_uint32(const char* table_name,
- const char* column_name,
- uint32_t value) {
- if (!table_name || !column_name) return -1;
- return webrtc::DataLog::InsertCell(table_name, column_name, value);
-}
-
-extern "C" int WebRtcDataLog_InsertArray_uint32(const char* table_name,
- const char* column_name,
- const uint32_t* values,
- int length) {
- if (!table_name || !column_name) return -1;
- return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
-}
-
-extern "C" int WebRtcDataLog_InsertCell_int64(const char* table_name,
- const char* column_name,
- int64_t value) {
- if (!table_name || !column_name) return -1;
- return webrtc::DataLog::InsertCell(table_name, column_name, value);
-}
-
-extern "C" int WebRtcDataLog_InsertArray_int64(const char* table_name,
- const char* column_name,
- const int64_t* values,
- int length) {
- if (!table_name || !column_name) return -1;
- return webrtc::DataLog::InsertCell(table_name, column_name, values, length);
-}
-
-extern "C" int WebRtcDataLog_NextRow(const char* table_name) {
- if (!table_name) return -1;
- return webrtc::DataLog::NextRow(table_name);
-}
diff --git a/trunk/src/system_wrappers/source/data_log_c_helpers_unittest.c b/trunk/src/system_wrappers/source/data_log_c_helpers_unittest.c
deleted file mode 100644
index e78a0e3..0000000
--- a/trunk/src/system_wrappers/source/data_log_c_helpers_unittest.c
+++ /dev/null
@@ -1,124 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "system_wrappers/source/data_log_c_helpers_unittest.h"
-
-#include <assert.h>
-#include <string.h>
-#include <stdlib.h>
-
-#include "system_wrappers/interface/data_log_c.h"
-
-enum { kTestArrayLen = 4 };
-static const char kTableName[] = "c_wrapper_table";
-static const char kColumnName1[] = "Scalar";
-static const char kColumnName2[] = "Vector";
-
-int WebRtcDataLogCHelper_TestCreateLog() {
- return WebRtcDataLog_CreateLog();
-}
-
-int WebRtcDataLogCHelper_TestReturnLog() {
- WebRtcDataLog_ReturnLog();
- return 0;
-}
-
-int WebRtcDataLogCHelper_TestCombine() {
- const int kOutLen = strlen(kTableName) + 4; /* Room for "_17" + '\0' */
- char* combined_name = malloc(kOutLen * sizeof(char));
- char* out_ptr = WebRtcDataLog_Combine(combined_name, kOutLen, kTableName, 17);
- int return_code = 0;
- if (!out_ptr) {
- return_code = -1;
- }
- if (strcmp(combined_name, "c_wrapper_table_17") != 0) {
- return_code = -2;
- }
- free(combined_name);
- return return_code;
-}
-
-int WebRtcDataLogCHelper_TestAddTable() {
- return WebRtcDataLog_AddTable(kTableName);
-}
-
-int WebRtcDataLogCHelper_TestAddColumn() {
- if (WebRtcDataLog_AddColumn(kTableName, kColumnName1, 1) != 0) {
- return -1;
- }
- if (WebRtcDataLog_AddColumn(kTableName, kColumnName2, kTestArrayLen) != 0) {
- return -2;
- }
- return 0;
-}
-
-int WebRtcDataLogCHelper_TestNextRow() {
- return WebRtcDataLog_NextRow(kTableName);
-}
-
-int WebRtcDataLogCHelper_TestInsertCell_int() {
- return WebRtcDataLog_InsertCell_int(kTableName, kColumnName1, 17);
-}
-
-int WebRtcDataLogCHelper_TestInsertArray_int() {
- int values[kTestArrayLen] = {1, 2, 3, 4};
- return WebRtcDataLog_InsertArray_int(kTableName, kColumnName2, values,
- kTestArrayLen);
-}
-
-int WebRtcDataLogCHelper_TestInsertCell_float() {
- return WebRtcDataLog_InsertCell_float(kTableName, kColumnName1, 17.0f);
-}
-
-int WebRtcDataLogCHelper_TestInsertArray_float() {
- float values[kTestArrayLen] = {1.0f, 2.0f, 3.0f, 4.0f};
- return WebRtcDataLog_InsertArray_float(kTableName, kColumnName2, values,
- kTestArrayLen);
-}
-
-int WebRtcDataLogCHelper_TestInsertCell_double() {
- return WebRtcDataLog_InsertCell_int(kTableName, kColumnName1, 17.0);
-}
-
-int WebRtcDataLogCHelper_TestInsertArray_double() {
- double values[kTestArrayLen] = {1.0, 2.0, 3.0, 4.0};
- return WebRtcDataLog_InsertArray_double(kTableName, kColumnName2, values,
- kTestArrayLen);
-}
-
-int WebRtcDataLogCHelper_TestInsertCell_int32() {
- return WebRtcDataLog_InsertCell_int32(kTableName, kColumnName1, 17);
-}
-
-int WebRtcDataLogCHelper_TestInsertArray_int32() {
- int32_t values[kTestArrayLen] = {1, 2, 3, 4};
- return WebRtcDataLog_InsertArray_int32(kTableName, kColumnName2, values,
- kTestArrayLen);
-}
-
-int WebRtcDataLogCHelper_TestInsertCell_uint32() {
- return WebRtcDataLog_InsertCell_uint32(kTableName, kColumnName1, 17);
-}
-
-int WebRtcDataLogCHelper_TestInsertArray_uint32() {
- uint32_t values[kTestArrayLen] = {1, 2, 3, 4};
- return WebRtcDataLog_InsertArray_uint32(kTableName, kColumnName2, values,
- kTestArrayLen);
-}
-
-int WebRtcDataLogCHelper_TestInsertCell_int64() {
- return WebRtcDataLog_InsertCell_int64(kTableName, kColumnName1, 17);
-}
-
-int WebRtcDataLogCHelper_TestInsertArray_int64() {
- int64_t values[kTestArrayLen] = {1, 2, 3, 4};
- return WebRtcDataLog_InsertArray_int64(kTableName, kColumnName2, values,
- kTestArrayLen);
-}
diff --git a/trunk/src/system_wrappers/source/data_log_c_helpers_unittest.h b/trunk/src/system_wrappers/source/data_log_c_helpers_unittest.h
deleted file mode 100644
index ef86eae..0000000
--- a/trunk/src/system_wrappers/source/data_log_c_helpers_unittest.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_SYSTEM_WRAPPERS_SOURCE_DATA_LOG_C_HELPERS_UNITTEST_H_
-#define SRC_SYSTEM_WRAPPERS_SOURCE_DATA_LOG_C_HELPERS_UNITTEST_H_
-
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-int WebRtcDataLogCHelper_TestCreateLog();
-
-int WebRtcDataLogCHelper_TestReturnLog();
-
-int WebRtcDataLogCHelper_TestCombine();
-
-int WebRtcDataLogCHelper_TestAddTable();
-
-int WebRtcDataLogCHelper_TestAddColumn();
-
-int WebRtcDataLogCHelper_TestNextRow();
-
-int WebRtcDataLogCHelper_TestInsertCell_int();
-
-int WebRtcDataLogCHelper_TestInsertArray_int();
-
-int WebRtcDataLogCHelper_TestInsertCell_float();
-
-int WebRtcDataLogCHelper_TestInsertArray_float();
-
-int WebRtcDataLogCHelper_TestInsertCell_double();
-
-int WebRtcDataLogCHelper_TestInsertArray_double();
-
-int WebRtcDataLogCHelper_TestInsertCell_int32();
-
-int WebRtcDataLogCHelper_TestInsertArray_int32();
-
-int WebRtcDataLogCHelper_TestInsertCell_uint32();
-
-int WebRtcDataLogCHelper_TestInsertArray_uint32();
-
-int WebRtcDataLogCHelper_TestInsertCell_int64();
-
-int WebRtcDataLogCHelper_TestInsertArray_int64();
-
-#ifdef __cplusplus
-} // end of extern "C"
-#endif
-
-#endif // SRC_SYSTEM_WRAPPERS_SOURCE_DATA_LOG_C_HELPERS_UNITTEST_H_
diff --git a/trunk/src/system_wrappers/source/data_log_helpers_unittest.cc b/trunk/src/system_wrappers/source/data_log_helpers_unittest.cc
deleted file mode 100644
index 94b4d6e..0000000
--- a/trunk/src/system_wrappers/source/data_log_helpers_unittest.cc
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <string>
-
-#include "data_log.h"
-#include "gtest/gtest.h"
-
-using ::webrtc::DataLog;
-
-TEST(TestDataLog, IntContainers) {
- int c = 5;
- webrtc::ValueContainer<int> v1(c);
- c = 10;
- webrtc::ValueContainer<int> v2(c);
- std::string s1, s2;
- v1.ToString(&s1);
- v2.ToString(&s2);
- ASSERT_EQ(s1, "5,");
- ASSERT_EQ(s2, "10,");
- v1 = v2;
- v1.ToString(&s1);
- ASSERT_EQ(s1, s2);
-}
-
-TEST(TestDataLog, DoubleContainers) {
- double c = 3.5;
- webrtc::ValueContainer<double> v1(c);
- c = 10.3;
- webrtc::ValueContainer<double> v2(c);
- std::string s1, s2;
- v1.ToString(&s1);
- v2.ToString(&s2);
- ASSERT_EQ(s1, "3.5,");
- ASSERT_EQ(s2, "10.3,");
- v1 = v2;
- v1.ToString(&s1);
- ASSERT_EQ(s1, s2);
-}
-
-TEST(TestDataLog, MultiValueContainers) {
- int a[3] = {1, 2, 3};
- int b[3] = {4, 5, 6};
- webrtc::MultiValueContainer<int> m1(a, 3);
- webrtc::MultiValueContainer<int> m2(b, 3);
- webrtc::MultiValueContainer<int> m3(a, 3);
- std::string s1, s2, s3;
- m1.ToString(&s1);
- m2.ToString(&s2);
- ASSERT_EQ(s1, "1,2,3,");
- ASSERT_EQ(s2, "4,5,6,");
- m1 = m2;
- m1.ToString(&s1);
- ASSERT_EQ(s1, s2);
- m3.ToString(&s3);
- ASSERT_EQ(s3, "1,2,3,");
-}
diff --git a/trunk/src/system_wrappers/source/data_log_no_op.cc b/trunk/src/system_wrappers/source/data_log_no_op.cc
deleted file mode 100644
index bedc82a..0000000
--- a/trunk/src/system_wrappers/source/data_log_no_op.cc
+++ /dev/null
@@ -1,88 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "data_log.h"
-
-#include <string>
-
-namespace webrtc {
-
-int DataLog::CreateLog() {
- return 0;
-}
-
-void DataLog::ReturnLog() {
-}
-
-std::string DataLog::Combine(const std::string& table_name, int table_id) {
- return std::string();
-}
-
-int DataLog::AddTable(const std::string& /*table_name*/) {
- return 0;
-}
-
-int DataLog::AddColumn(const std::string& /*table_name*/,
- const std::string& /*column_name*/,
- int /*multi_value_length*/) {
- return 0;
-}
-
-int DataLog::NextRow(const std::string& /*table_name*/) {
- return 0;
-}
-
-DataLogImpl::DataLogImpl() {
-}
-
-DataLogImpl::~DataLogImpl() {
-}
-
-DataLogImpl* DataLogImpl::StaticInstance() {
- return NULL;
-}
-
-void DataLogImpl::ReturnLog() {
-}
-
-int DataLogImpl::AddTable(const std::string& /*table_name*/) {
- return 0;
-}
-
-int DataLogImpl::AddColumn(const std::string& /*table_name*/,
- const std::string& /*column_name*/,
- int /*multi_value_length*/) {
- return 0;
-}
-
-int DataLogImpl::InsertCell(const std::string& /*table_name*/,
- const std::string& /*column_name*/,
- const Container* /*value_container*/) {
- return 0;
-}
-
-int DataLogImpl::NextRow(const std::string& /*table_name*/) {
- return 0;
-}
-
-void DataLogImpl::Flush() {
-}
-
-bool DataLogImpl::Run(void* /*obj*/) {
- return true;
-}
-
-void DataLogImpl::Process() {
-}
-
-void DataLogImpl::StopThread() {
-}
-
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/data_log_unittest.cc b/trunk/src/system_wrappers/source/data_log_unittest.cc
deleted file mode 100644
index c64ed94..0000000
--- a/trunk/src/system_wrappers/source/data_log_unittest.cc
+++ /dev/null
@@ -1,310 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <map>
-#include <string>
-
-#include "system_wrappers/interface/data_log.h"
-#include "system_wrappers/interface/data_log_c.h"
-#include "system_wrappers/source/data_log_c_helpers_unittest.h"
-#include "gtest/gtest.h"
-
-using ::webrtc::DataLog;
-
-// A class for storing the values expected from a log table column when
-// verifying a log table file.
-struct ExpectedValues {
- public:
- ExpectedValues()
- : values(NULL),
- multi_value_length(1) {
- }
-
- ExpectedValues(std::vector<std::string> expected_values,
- int expected_multi_value_length)
- : values(expected_values),
- multi_value_length(expected_multi_value_length) {
- }
-
- std::vector<std::string> values;
- int multi_value_length;
-};
-
-typedef std::map<std::string, ExpectedValues> ExpectedValuesMap;
-
-// A static class used for parsing and verifying data log files.
-class DataLogParser {
- public:
- // Verifies that the log table stored in the file "log_file" corresponds to
- // the cells and columns specified in "columns".
- static int VerifyTable(FILE* log_file, const ExpectedValuesMap& columns) {
- int row = 0;
- char line_buffer[kMaxLineLength];
- char* ret = fgets(line_buffer, kMaxLineLength, log_file);
- EXPECT_FALSE(ret == NULL);
- if (ret == NULL)
- return -1;
-
- std::string line(line_buffer, kMaxLineLength);
- VerifyHeader(line, columns);
- while (fgets(line_buffer, kMaxLineLength, log_file) != NULL) {
- line = std::string(line_buffer, kMaxLineLength);
- size_t line_position = 0;
-
- for (ExpectedValuesMap::const_iterator it = columns.begin();
- it != columns.end(); ++it) {
- std::string str = ParseElement(line, &line_position,
- it->second.multi_value_length);
- EXPECT_EQ(str, it->second.values[row]);
- if (str != it->second.values[row])
- return -1;
- }
- ++row;
- }
- return 0;
- }
-
- // Verifies the table header stored in "line" to correspond with the header
- // specified in "columns".
- static int VerifyHeader(const std::string& line,
- const ExpectedValuesMap& columns) {
- size_t line_position = 0;
- for (ExpectedValuesMap::const_iterator it = columns.begin();
- it != columns.end(); ++it) {
- std::string str = ParseElement(line, &line_position,
- it->second.multi_value_length);
- EXPECT_EQ(str, it->first);
- if (str != it->first)
- return -1;
- }
- return 0;
- }
-
- // Parses out and returns one element from the string "line", which contains
- // one line read from a log table file. An element can either be a column
- // header or a cell of a row.
- static std::string ParseElement(const std::string& line,
- size_t* line_position,
- int multi_value_length) {
- std::string parsed_cell;
- parsed_cell = "";
- for (int i = 0; i < multi_value_length; ++i) {
- size_t next_separator = line.find(',', *line_position);
- EXPECT_NE(next_separator, std::string::npos);
- if (next_separator == std::string::npos)
- break;
- parsed_cell += line.substr(*line_position,
- next_separator - *line_position + 1);
- *line_position = next_separator + 1;
- }
- return parsed_cell;
- }
-
- // This constant defines the maximum line length the DataLogParser can
- // parse.
- enum { kMaxLineLength = 100 };
-};
-
-TEST(TestDataLog, CreateReturnTest) {
- for (int i = 0; i < 10; ++i)
- ASSERT_EQ(DataLog::CreateLog(), 0);
- ASSERT_EQ(DataLog::AddTable(DataLog::Combine("a proper table", 1)), 0);
- for (int i = 0; i < 10; ++i)
- DataLog::ReturnLog();
- ASSERT_LT(DataLog::AddTable(DataLog::Combine("table failure", 1)), 0);
-}
-
-TEST(TestDataLog, VerifyCombineMethod) {
- EXPECT_EQ(std::string("a proper table_1"),
- DataLog::Combine("a proper table", 1));
-}
-
-TEST(TestDataLog, VerifySingleTable) {
- DataLog::CreateLog();
- DataLog::AddTable(DataLog::Combine("table", 1));
- DataLog::AddColumn(DataLog::Combine("table", 1), "arrival", 1);
- DataLog::AddColumn(DataLog::Combine("table", 1), "timestamp", 1);
- DataLog::AddColumn(DataLog::Combine("table", 1), "size", 5);
- WebRtc_UWord32 sizes[5] = {1400, 1500, 1600, 1700, 1800};
- for (int i = 0; i < 10; ++i) {
- DataLog::InsertCell(DataLog::Combine("table", 1), "arrival",
- static_cast<double>(i));
- DataLog::InsertCell(DataLog::Combine("table", 1), "timestamp",
- static_cast<WebRtc_Word64>(4354 + i));
- DataLog::InsertCell(DataLog::Combine("table", 1), "size", sizes, 5);
- DataLog::NextRow(DataLog::Combine("table", 1));
- }
- DataLog::ReturnLog();
- // Verify file
- FILE* table = fopen("table_1.txt", "r");
- ASSERT_FALSE(table == NULL);
- // Read the column names and verify with the expected columns.
- // Note that the columns are written to file in alphabetical order.
- // Data expected from parsing the file
- const int kNumberOfRows = 10;
- std::string string_arrival[kNumberOfRows] = {
- "0,", "1,", "2,", "3,", "4,",
- "5,", "6,", "7,", "8,", "9,"
- };
- std::string string_timestamp[kNumberOfRows] = {
- "4354,", "4355,", "4356,", "4357,",
- "4358,", "4359,", "4360,", "4361,",
- "4362,", "4363,"
- };
- std::string string_sizes = "1400,1500,1600,1700,1800,";
- ExpectedValuesMap expected;
- expected["arrival,"] = ExpectedValues(
- std::vector<std::string>(string_arrival,
- string_arrival +
- kNumberOfRows),
- 1);
- expected["size[5],,,,,"] = ExpectedValues(
- std::vector<std::string>(10, string_sizes), 5);
- expected["timestamp,"] = ExpectedValues(
- std::vector<std::string>(string_timestamp,
- string_timestamp +
- kNumberOfRows),
- 1);
- ASSERT_EQ(DataLogParser::VerifyTable(table, expected), 0);
- fclose(table);
-}
-
-TEST(TestDataLog, VerifyMultipleTables) {
- DataLog::CreateLog();
- DataLog::AddTable(DataLog::Combine("table", 2));
- DataLog::AddTable(DataLog::Combine("table", 3));
- DataLog::AddColumn(DataLog::Combine("table", 2), "arrival", 1);
- DataLog::AddColumn(DataLog::Combine("table", 2), "timestamp", 1);
- DataLog::AddColumn(DataLog::Combine("table", 2), "size", 1);
- DataLog::AddTable(DataLog::Combine("table", 4));
- DataLog::AddColumn(DataLog::Combine("table", 3), "timestamp", 1);
- DataLog::AddColumn(DataLog::Combine("table", 3), "arrival", 1);
- DataLog::AddColumn(DataLog::Combine("table", 4), "size", 1);
- for (WebRtc_Word32 i = 0; i < 10; ++i) {
- DataLog::InsertCell(DataLog::Combine("table", 2), "arrival",
- static_cast<WebRtc_Word32>(i));
- DataLog::InsertCell(DataLog::Combine("table", 2), "timestamp",
- static_cast<WebRtc_Word32>(4354 + i));
- DataLog::InsertCell(DataLog::Combine("table", 2), "size",
- static_cast<WebRtc_Word32>(1200 + 10 * i));
- DataLog::InsertCell(DataLog::Combine("table", 3), "timestamp",
- static_cast<WebRtc_Word32>(4354 + i));
- DataLog::InsertCell(DataLog::Combine("table", 3), "arrival",
- static_cast<WebRtc_Word32>(i));
- DataLog::InsertCell(DataLog::Combine("table", 4), "size",
- static_cast<WebRtc_Word32>(1200 + 10 * i));
- DataLog::NextRow(DataLog::Combine("table", 4));
- DataLog::NextRow(DataLog::Combine("table", 2));
- DataLog::NextRow(DataLog::Combine("table", 3));
- }
- DataLog::ReturnLog();
-
- // Data expected from parsing the file
- const int kNumberOfRows = 10;
- std::string string_arrival[kNumberOfRows] = {
- "0,", "1,", "2,", "3,", "4,",
- "5,", "6,", "7,", "8,", "9,"
- };
- std::string string_timestamp[kNumberOfRows] = {
- "4354,", "4355,", "4356,", "4357,",
- "4358,", "4359,", "4360,", "4361,",
- "4362,", "4363,"
- };
- std::string string_size[kNumberOfRows] = {
- "1200,", "1210,", "1220,", "1230,",
- "1240,", "1250,", "1260,", "1270,",
- "1280,", "1290,"
- };
-
- // Verify table 2
- {
- FILE* table = fopen("table_2.txt", "r");
- ASSERT_FALSE(table == NULL);
- ExpectedValuesMap expected;
- expected["arrival,"] = ExpectedValues(
- std::vector<std::string>(string_arrival,
- string_arrival +
- kNumberOfRows),
- 1);
- expected["size,"] = ExpectedValues(
- std::vector<std::string>(string_size,
- string_size + kNumberOfRows),
- 1);
- expected["timestamp,"] = ExpectedValues(
- std::vector<std::string>(string_timestamp,
- string_timestamp +
- kNumberOfRows),
- 1);
- ASSERT_EQ(DataLogParser::VerifyTable(table, expected), 0);
- fclose(table);
- }
-
- // Verify table 3
- {
- FILE* table = fopen("table_3.txt", "r");
- ASSERT_FALSE(table == NULL);
- ExpectedValuesMap expected;
- expected["arrival,"] = ExpectedValues(
- std::vector<std::string>(string_arrival,
- string_arrival +
- kNumberOfRows),
- 1);
- expected["timestamp,"] = ExpectedValues(
- std::vector<std::string>(string_timestamp,
- string_timestamp +
- kNumberOfRows),
- 1);
- ASSERT_EQ(DataLogParser::VerifyTable(table, expected), 0);
- fclose(table);
- }
-
- // Verify table 4
- {
- FILE* table = fopen("table_4.txt", "r");
- ASSERT_FALSE(table == NULL);
- ExpectedValuesMap expected;
- expected["size,"] = ExpectedValues(
- std::vector<std::string>(string_size,
- string_size +
- kNumberOfRows),
- 1);
- ASSERT_EQ(DataLogParser::VerifyTable(table, expected), 0);
- fclose(table);
- }
-}
-
-TEST(TestDataLogCWrapper, VerifyCWrapper) {
- // Simply call all C wrapper log functions through the C helper unittests.
- // Main purpose is to make sure that the linkage is correct.
-
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestCreateLog());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestCombine());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestAddTable());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestAddColumn());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_int());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_int());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_float());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_float());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_double());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_double());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_int32());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_int32());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_uint32());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_uint32());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertCell_int64());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestInsertArray_int64());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestNextRow());
- EXPECT_EQ(0, WebRtcDataLogCHelper_TestReturnLog());
-}
diff --git a/trunk/src/system_wrappers/source/data_log_unittest_disabled.cc b/trunk/src/system_wrappers/source/data_log_unittest_disabled.cc
deleted file mode 100644
index 9d630b6..0000000
--- a/trunk/src/system_wrappers/source/data_log_unittest_disabled.cc
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "system_wrappers/interface/data_log.h"
-
-#include <cstdio>
-
-#include "gtest/gtest.h"
-
-using ::webrtc::DataLog;
-
-const char* kDataLogFileName = "table_1.txt";
-
-void PerformLogging(std::string table_name) {
- // Simulate normal DataTable logging behavior using this table name.
- ASSERT_EQ(0, DataLog::AddTable(table_name));
- ASSERT_EQ(0, DataLog::AddColumn(table_name, "test", 1));
- for (int i = 0; i < 10; ++i) {
- // TODO(kjellander): Check InsertCell result when the DataLog dummy is
- // fixed.
- DataLog::InsertCell(table_name, "test", static_cast<double>(i));
- ASSERT_EQ(0, DataLog::NextRow(table_name));
- }
-}
-
-// Simple test to verify DataLog is still working when the GYP variable
-// enable_data_logging==0 (the default case).
-TEST(TestDataLogDisabled, VerifyLoggingWorks) {
- ASSERT_EQ(0, DataLog::CreateLog());
- // Generate a table_name name and assure it's an empty string
- // (dummy behavior).
- std::string table_name = DataLog::Combine("table", 1);
- ASSERT_EQ("", table_name);
- PerformLogging(table_name);
- DataLog::ReturnLog();
-}
-
-TEST(TestDataLogDisabled, EnsureNoFileIsWritten) {
- // Remove any previous data files on disk:
- std::remove(kDataLogFileName);
- ASSERT_EQ(0, DataLog::CreateLog());
- // Don't use the table name we would get from Combine on a disabled DataLog.
- // Use "table_1" instead (which is what an enabled DataLog would give us).
- PerformLogging("table_1");
- DataLog::ReturnLog();
- // Verify no data log file have been written:
- ASSERT_EQ(NULL, fopen(kDataLogFileName, "r"));
-}
diff --git a/trunk/src/system_wrappers/source/event.cc b/trunk/src/system_wrappers/source/event.cc
deleted file mode 100644
index 608cd53..0000000
--- a/trunk/src/system_wrappers/source/event.cc
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "event_wrapper.h"
-
-#if defined(_WIN32)
- #include <windows.h>
- #include "event_win.h"
-#elif defined(WEBRTC_MAC_INTEL)
- #include <ApplicationServices/ApplicationServices.h>
- #include <pthread.h>
- #include "event_posix.h"
-#else
- #include <pthread.h>
- #include "event_posix.h"
-#endif
-
-namespace webrtc {
-EventWrapper* EventWrapper::Create()
-{
-#if defined(_WIN32)
- return new EventWindows();
-#else
- return EventPosix::Create();
-#endif
-}
-
-int EventWrapper::KeyPressed()
-{
-#if defined(_WIN32)
- int keyDown = 0;
- for(int key = 0x20; key < 0x90; key++)
- {
- short res = GetAsyncKeyState(key);
- keyDown |= res%2; // Get the LSB
- }
- if(keyDown)
- {
- return 1;
- }
- else
- {
- return 0;
- }
-#elif defined(WEBRTC_MAC_INTEL)
- bool keyDown = false;
- // loop through all Mac virtual key constant values
- for(int keyIndex = 0; keyIndex <= 0x5C; keyIndex++)
- {
- keyDown |= CGEventSourceKeyState(kCGEventSourceStateHIDSystemState, keyIndex);
- }
- if(keyDown)
- {
- return 1;
- }
- else
- {
- return 0;
- }
-#else
- return -1;
-#endif
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/event_posix.cc b/trunk/src/system_wrappers/source/event_posix.cc
deleted file mode 100644
index b77b902..0000000
--- a/trunk/src/system_wrappers/source/event_posix.cc
+++ /dev/null
@@ -1,324 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "event_posix.h"
-
-#include <errno.h>
-#include <pthread.h>
-#include <signal.h>
-#include <stdio.h>
-#include <string.h>
-#include <sys/time.h>
-#include <unistd.h>
-
-namespace webrtc {
-const long int E6 = 1000000;
-const long int E9 = 1000 * E6;
-
-EventWrapper* EventPosix::Create()
-{
- EventPosix* ptr = new EventPosix;
- if (!ptr)
- {
- return NULL;
- }
-
- const int error = ptr->Construct();
- if (error)
- {
- delete ptr;
- return NULL;
- }
- return ptr;
-}
-
-
-EventPosix::EventPosix()
- : _timerThread(0),
- _timerEvent(0),
- _periodic(false),
- _time(0),
- _count(0),
- _state(kDown)
-{
-}
-
-int EventPosix::Construct()
-{
- // Set start time to zero
- memset(&_tCreate, 0, sizeof(_tCreate));
-
- int result = pthread_mutex_init(&mutex, 0);
- if (result != 0)
- {
- return -1;
- }
-#ifdef WEBRTC_CLOCK_TYPE_REALTIME
- result = pthread_cond_init(&cond, 0);
- if (result != 0)
- {
- return -1;
- }
-#else
- pthread_condattr_t condAttr;
- result = pthread_condattr_init(&condAttr);
- if (result != 0)
- {
- return -1;
- }
- result = pthread_condattr_setclock(&condAttr, CLOCK_MONOTONIC);
- if (result != 0)
- {
- return -1;
- }
- result = pthread_cond_init(&cond, &condAttr);
- if (result != 0)
- {
- return -1;
- }
- result = pthread_condattr_destroy(&condAttr);
- if (result != 0)
- {
- return -1;
- }
-#endif
- return 0;
-}
-
-EventPosix::~EventPosix()
-{
- StopTimer();
- pthread_cond_destroy(&cond);
- pthread_mutex_destroy(&mutex);
-}
-
-bool EventPosix::Reset()
-{
- if (0 != pthread_mutex_lock(&mutex))
- {
- return false;
- }
- _state = kDown;
- pthread_mutex_unlock(&mutex);
- return true;
-}
-
-bool EventPosix::Set()
-{
- if (0 != pthread_mutex_lock(&mutex))
- {
- return false;
- }
- _state = kUp;
- // Release all waiting threads
- pthread_cond_broadcast(&cond);
- pthread_mutex_unlock(&mutex);
- return true;
-}
-
-EventTypeWrapper EventPosix::Wait(unsigned long timeout)
-{
- int retVal = 0;
- if (0 != pthread_mutex_lock(&mutex))
- {
- return kEventError;
- }
-
- if (kDown == _state)
- {
- if (WEBRTC_EVENT_INFINITE != timeout)
- {
- timespec tEnd;
-#ifndef WEBRTC_MAC
-#ifdef WEBRTC_CLOCK_TYPE_REALTIME
- clock_gettime(CLOCK_REALTIME, &tEnd);
-#else
- clock_gettime(CLOCK_MONOTONIC, &tEnd);
-#endif
-#else
- timeval tVal;
- struct timezone tZone;
- tZone.tz_minuteswest = 0;
- tZone.tz_dsttime = 0;
- gettimeofday(&tVal,&tZone);
- TIMEVAL_TO_TIMESPEC(&tVal,&tEnd);
-#endif
- tEnd.tv_sec += timeout / 1000;
- tEnd.tv_nsec += (timeout - (timeout / 1000) * 1000) * E6;
-
- if (tEnd.tv_nsec >= E9)
- {
- tEnd.tv_sec++;
- tEnd.tv_nsec -= E9;
- }
- retVal = pthread_cond_timedwait(&cond, &mutex, &tEnd);
- } else {
- retVal = pthread_cond_wait(&cond, &mutex);
- }
- }
-
- _state = kDown;
- pthread_mutex_unlock(&mutex);
-
- switch(retVal)
- {
- case 0:
- return kEventSignaled;
- case ETIMEDOUT:
- return kEventTimeout;
- default:
- return kEventError;
- }
-}
-
-EventTypeWrapper EventPosix::Wait(timespec& tPulse)
-{
- int retVal = 0;
- if (0 != pthread_mutex_lock(&mutex))
- {
- return kEventError;
- }
-
- if (kUp != _state)
- {
- retVal = pthread_cond_timedwait(&cond, &mutex, &tPulse);
- }
- _state = kDown;
-
- pthread_mutex_unlock(&mutex);
-
- switch(retVal)
- {
- case 0:
- return kEventSignaled;
- case ETIMEDOUT:
- return kEventTimeout;
- default:
- return kEventError;
- }
-}
-
-bool EventPosix::StartTimer(bool periodic, unsigned long time)
-{
- if (_timerThread)
- {
- if(_periodic)
- {
- // Timer already started.
- return false;
- } else {
- // New one shot timer
- _time = time;
- _tCreate.tv_sec = 0;
- _timerEvent->Set();
- return true;
- }
- }
-
- // Start the timer thread
- _timerEvent = static_cast<EventPosix*>(EventWrapper::Create());
- const char* threadName = "WebRtc_event_timer_thread";
- _timerThread = ThreadWrapper::CreateThread(Run, this, kRealtimePriority,
- threadName);
- _periodic = periodic;
- _time = time;
- unsigned int id = 0;
- if (_timerThread->Start(id))
- {
- return true;
- }
- return false;
-}
-
-bool EventPosix::Run(ThreadObj obj)
-{
- return static_cast<EventPosix*>(obj)->Process();
-}
-
-bool EventPosix::Process()
-{
- if (_tCreate.tv_sec == 0)
- {
-#ifndef WEBRTC_MAC
-#ifdef WEBRTC_CLOCK_TYPE_REALTIME
- clock_gettime(CLOCK_REALTIME, &_tCreate);
-#else
- clock_gettime(CLOCK_MONOTONIC, &_tCreate);
-#endif
-#else
- timeval tVal;
- struct timezone tZone;
- tZone.tz_minuteswest = 0;
- tZone.tz_dsttime = 0;
- gettimeofday(&tVal,&tZone);
- TIMEVAL_TO_TIMESPEC(&tVal,&_tCreate);
-#endif
- _count=0;
- }
-
- timespec tEnd;
- unsigned long long time = _time * ++_count;
- tEnd.tv_sec = _tCreate.tv_sec + time/1000;
- tEnd.tv_nsec = _tCreate.tv_nsec + (time - (time/1000)*1000)*E6;
-
- if ( tEnd.tv_nsec >= E9 )
- {
- tEnd.tv_sec++;
- tEnd.tv_nsec -= E9;
- }
-
- switch(_timerEvent->Wait(tEnd))
- {
- case kEventSignaled:
- return true;
- case kEventError:
- return false;
- case kEventTimeout:
- break;
- }
- if(_periodic || _count==1)
- {
- Set();
- }
- return true;
-}
-
-bool EventPosix::StopTimer()
-{
- if(_timerThread)
- {
- _timerThread->SetNotAlive();
- }
- if (_timerEvent)
- {
- _timerEvent->Set();
- }
- if (_timerThread)
- {
- if(!_timerThread->Stop())
- {
- return false;
- }
-
- delete _timerThread;
- _timerThread = 0;
- }
- if (_timerEvent)
- {
- delete _timerEvent;
- _timerEvent = 0;
- }
-
- // Set time to zero to force new reference time for the timer.
- memset(&_tCreate, 0, sizeof(_tCreate));
- _count=0;
- return true;
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/event_posix.h b/trunk/src/system_wrappers/source/event_posix.h
deleted file mode 100644
index 0e5893b..0000000
--- a/trunk/src/system_wrappers/source/event_posix.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_POSIX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_POSIX_H_
-
-#include "event_wrapper.h"
-
-#include <pthread.h>
-#include <time.h>
-
-#include "thread_wrapper.h"
-
-namespace webrtc {
-enum State
-{
- kUp = 1,
- kDown = 2
-};
-
-class EventPosix : public EventWrapper
-{
-public:
- static EventWrapper* Create();
-
- virtual ~EventPosix();
-
- virtual EventTypeWrapper Wait(unsigned long maxTime);
- virtual bool Set();
- virtual bool Reset();
-
- virtual bool StartTimer(bool periodic, unsigned long time);
- virtual bool StopTimer();
-
-private:
- EventPosix();
- int Construct();
-
- static bool Run(ThreadObj obj);
- bool Process();
- EventTypeWrapper Wait(timespec& tPulse);
-
-
-private:
- pthread_cond_t cond;
- pthread_mutex_t mutex;
-
- ThreadWrapper* _timerThread;
- EventPosix* _timerEvent;
- timespec _tCreate;
-
- bool _periodic;
- unsigned long _time; // In ms
- unsigned long _count;
- State _state;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_POSIX_H_
diff --git a/trunk/src/system_wrappers/source/event_win.cc b/trunk/src/system_wrappers/source/event_win.cc
deleted file mode 100644
index efcb5af..0000000
--- a/trunk/src/system_wrappers/source/event_win.cc
+++ /dev/null
@@ -1,84 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "event_win.h"
-
-#include "Mmsystem.h"
-
-namespace webrtc {
-EventWindows::EventWindows()
- : _event(::CreateEvent(NULL /* security attributes */,
- FALSE /* manual reset */,
- FALSE /* initial state */,
- NULL /* name of event */)),
- _timerID(NULL)
-{
-}
-
-EventWindows::~EventWindows()
-{
- CloseHandle(_event);
-}
-
-bool EventWindows::Set()
-{
- // Note: setting an event that is already set has no effect.
- return SetEvent(_event) == 1 ? true : false;
-}
-
-bool EventWindows::Reset()
-{
- return ResetEvent(_event) == 1 ? true : false;
-}
-
-EventTypeWrapper EventWindows::Wait(unsigned long maxTime)
-{
- unsigned long res = WaitForSingleObject(_event, maxTime);
- switch(res)
- {
- case WAIT_OBJECT_0:
- return kEventSignaled;
- case WAIT_TIMEOUT:
- return kEventTimeout;
- default:
- return kEventError;
- }
-}
-
-bool EventWindows::StartTimer(bool periodic, unsigned long time)
-{
- if (_timerID != NULL)
- {
- timeKillEvent(_timerID);
- _timerID=NULL;
- }
- if (periodic)
- {
- _timerID=timeSetEvent(time, 0,(LPTIMECALLBACK)HANDLE(_event),0,
- TIME_PERIODIC|TIME_CALLBACK_EVENT_PULSE);
- } else {
- _timerID=timeSetEvent(time, 0,(LPTIMECALLBACK)HANDLE(_event),0,
- TIME_ONESHOT|TIME_CALLBACK_EVENT_SET);
- }
-
- if (_timerID == NULL)
- {
- return false;
- }
- return true;
-}
-
-bool EventWindows::StopTimer()
-{
- timeKillEvent(_timerID);
- _timerID = NULL;
- return true;
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/event_win.h b/trunk/src/system_wrappers/source/event_win.h
deleted file mode 100644
index 8ca1360..0000000
--- a/trunk/src/system_wrappers/source/event_win.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_WINDOWS_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_WINDOWS_H_
-
-#include <windows.h>
-
-#include "event_wrapper.h"
-
-#include "typedefs.h"
-
-namespace webrtc {
-class EventWindows : public EventWrapper
-{
-public:
- EventWindows();
- virtual ~EventWindows();
-
- virtual EventTypeWrapper Wait(unsigned long maxTime);
- virtual bool Set();
- virtual bool Reset();
-
- virtual bool StartTimer(bool periodic, unsigned long time);
- virtual bool StopTimer();
-
-private:
- HANDLE _event;
- WebRtc_UWord32 _timerID;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_EVENT_WINDOWS_H_
diff --git a/trunk/src/system_wrappers/source/file_impl.cc b/trunk/src/system_wrappers/source/file_impl.cc
deleted file mode 100644
index 4d06c54..0000000
--- a/trunk/src/system_wrappers/source/file_impl.cc
+++ /dev/null
@@ -1,265 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "file_impl.h"
-
-#include <assert.h>
-
-#ifdef _WIN32
-#include <Windows.h>
-#else
-#include <stdarg.h>
-#include <string.h>
-#endif
-
-namespace webrtc {
-
-FileWrapper* FileWrapper::Create()
-{
- return new FileWrapperImpl();
-}
-
-FileWrapperImpl::FileWrapperImpl()
- : _id(NULL),
- _open(false),
- _looping(false),
- _readOnly(false),
- _maxSizeInBytes(0),
- _sizeInBytes(0)
-{
- memset(_fileNameUTF8, 0, kMaxFileNameSize);
-}
-
-FileWrapperImpl::~FileWrapperImpl()
-{
- if (_id != NULL)
- {
- fclose(_id);
- }
-}
-
-int FileWrapperImpl::CloseFile()
-{
- if (_id != NULL)
- {
- fclose(_id);
- _id = NULL;
- }
- memset(_fileNameUTF8, 0, kMaxFileNameSize);
- _open = false;
- return 0;
-}
-
-int FileWrapperImpl::Rewind()
-{
- if(_looping || !_readOnly)
- {
- if (_id != NULL)
- {
- _sizeInBytes = 0;
- return fseek(_id, 0, SEEK_SET);
- }
- }
- return -1;
-}
-
-int FileWrapperImpl::SetMaxFileSize(size_t bytes)
-{
- _maxSizeInBytes = bytes;
- return 0;
-}
-
-int FileWrapperImpl::Flush()
-{
- if (_id != NULL)
- {
- return fflush(_id);
- }
- return -1;
-}
-
-int FileWrapperImpl::FileName(char* fileNameUTF8,
- size_t size) const
-{
- size_t length = strlen(_fileNameUTF8);
- if(length > kMaxFileNameSize)
- {
- assert(false);
- return -1;
- }
- if(length < 1)
- {
- return -1;
- }
-
- // Make sure to NULL terminate
- if(size < length)
- {
- length = size - 1;
- }
- memcpy(fileNameUTF8, _fileNameUTF8, length);
- fileNameUTF8[length] = 0;
- return 0;
-}
-
-bool FileWrapperImpl::Open() const
-{
- return _open;
-}
-
-int FileWrapperImpl::OpenFile(const char *fileNameUTF8, bool readOnly,
- bool loop, bool text)
-{
- size_t length = strlen(fileNameUTF8);
- if (length > kMaxFileNameSize - 1)
- {
- return -1;
- }
-
- _readOnly = readOnly;
-
- FILE *tmpId = NULL;
-#if defined _WIN32
- wchar_t wideFileName[kMaxFileNameSize];
- wideFileName[0] = 0;
-
- MultiByteToWideChar(CP_UTF8,
- 0 /*UTF8 flag*/,
- fileNameUTF8,
- -1 /*Null terminated string*/,
- wideFileName,
- kMaxFileNameSize);
- if(text)
- {
- if(readOnly)
- {
- tmpId = _wfopen(wideFileName, L"rt");
- } else {
- tmpId = _wfopen(wideFileName, L"wt");
- }
- } else {
- if(readOnly)
- {
- tmpId = _wfopen(wideFileName, L"rb");
- } else {
- tmpId = _wfopen(wideFileName, L"wb");
- }
- }
-#else
- if(text)
- {
- if(readOnly)
- {
- tmpId = fopen(fileNameUTF8, "rt");
- } else {
- tmpId = fopen(fileNameUTF8, "wt");
- }
- } else {
- if(readOnly)
- {
- tmpId = fopen(fileNameUTF8, "rb");
- } else {
- tmpId = fopen(fileNameUTF8, "wb");
- }
- }
-#endif
-
- if (tmpId != NULL)
- {
- // +1 comes from copying the NULL termination character.
- memcpy(_fileNameUTF8, fileNameUTF8, length + 1);
- if (_id != NULL)
- {
- fclose(_id);
- }
- _id = tmpId;
- _looping = loop;
- _open = true;
- return 0;
- }
- return -1;
-}
-
-int FileWrapperImpl::Read(void* buf, int length)
-{
- if (length < 0)
- return -1;
-
- if (_id == NULL)
- return -1;
-
- int bytes_read = static_cast<int>(fread(buf, 1, length, _id));
- if (bytes_read != length && !_looping)
- {
- CloseFile();
- }
- return bytes_read;
-}
-
-int FileWrapperImpl::WriteText(const char* format, ...)
-{
- if (format == NULL)
- return -1;
-
- if (_readOnly)
- return -1;
-
- if (_id == NULL)
- return -1;
-
- va_list args;
- va_start(args, format);
- int num_chars = vfprintf(_id, format, args);
- va_end(args);
-
- if (num_chars >= 0)
- {
- return num_chars;
- }
- else
- {
- CloseFile();
- return -1;
- }
-}
-
-bool FileWrapperImpl::Write(const void* buf, int length)
-{
- if (buf == NULL)
- return false;
-
- if (length < 0)
- return false;
-
- if (_readOnly)
- return false;
-
- if (_id == NULL)
- return false;
-
- // Check if it's time to stop writing.
- if (_maxSizeInBytes > 0 && (_sizeInBytes + length) > _maxSizeInBytes)
- {
- Flush();
- return false;
- }
-
- size_t num_bytes = fwrite(buf, 1, length, _id);
- if (num_bytes > 0)
- {
- _sizeInBytes += num_bytes;
- return true;
- }
-
- CloseFile();
- return false;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/file_impl.h b/trunk/src/system_wrappers/source/file_impl.h
deleted file mode 100644
index 31ab31e..0000000
--- a/trunk/src/system_wrappers/source/file_impl.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_FILE_IMPL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_FILE_IMPL_H_
-
-#include "file_wrapper.h"
-
-#include <stdio.h>
-
-namespace webrtc {
-
-class FileWrapperImpl : public FileWrapper
-{
-public:
- FileWrapperImpl();
- virtual ~FileWrapperImpl();
-
- virtual int FileName(char* fileNameUTF8,
- size_t size) const;
-
- virtual bool Open() const;
-
- virtual int OpenFile(const char* fileNameUTF8,
- bool readOnly,
- bool loop = false,
- bool text = false);
-
- virtual int CloseFile();
- virtual int SetMaxFileSize(size_t bytes);
- virtual int Flush();
-
- virtual int Read(void* buf, int length);
- virtual bool Write(const void *buf, int length);
- virtual int WriteText(const char* format, ...);
- virtual int Rewind();
-
-private:
- FILE* _id;
- bool _open;
- bool _looping;
- bool _readOnly;
- size_t _maxSizeInBytes; // -1 indicates file size limitation is off
- size_t _sizeInBytes;
- char _fileNameUTF8[kMaxFileNameSize];
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_FILE_IMPL_H_
diff --git a/trunk/src/system_wrappers/source/list_no_stl.cc b/trunk/src/system_wrappers/source/list_no_stl.cc
deleted file mode 100644
index dbba571..0000000
--- a/trunk/src/system_wrappers/source/list_no_stl.cc
+++ /dev/null
@@ -1,289 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "list_wrapper.h"
-
-#include "critical_section_wrapper.h"
-#include "trace.h"
-
-namespace webrtc {
-ListItem::ListItem(const void* item)
- : next_(0),
- prev_(0),
- item_ptr_(item),
- item_(0)
-{
-}
-
-ListItem::ListItem(const unsigned int item)
- : next_(0),
- prev_(0),
- item_ptr_(0),
- item_(item)
-{
-}
-
-ListItem::~ListItem()
-{
-}
-
-void* ListItem::GetItem() const
-{
- return const_cast<void*>(item_ptr_);
-}
-
-unsigned int ListItem::GetUnsignedItem() const
-{
- return item_;
-}
-
-ListWrapper::ListWrapper()
- : critical_section_(CriticalSectionWrapper::CreateCriticalSection()),
- first_(0),
- last_(0),
- size_(0)
-{
-}
-
-ListWrapper::~ListWrapper()
-{
- if (!Empty())
- {
- // TODO (hellner) I'm not sure this loggin is useful.
- WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1,
- "Potential memory leak in ListWrapper");
- // Remove all remaining list items.
- while (Erase(First()) == 0)
- {}
- }
- delete critical_section_;
-}
-
-bool ListWrapper::Empty() const
-{
- return !first_ && !last_;
-}
-
-unsigned int ListWrapper::GetSize() const
-{
- return size_;
-}
-
-int ListWrapper::PushBack(const void* ptr)
-{
- ListItem* item = new ListItem(ptr);
- CriticalSectionScoped lock(critical_section_);
- PushBackImpl(item);
- return 0;
-}
-
-int ListWrapper::PushBack(const unsigned int item_id)
-{
- ListItem* item = new ListItem(item_id);
- CriticalSectionScoped lock(critical_section_);
- PushBackImpl(item);
- return 0;
-}
-
-int ListWrapper::PushFront(const unsigned int item_id)
-{
- ListItem* item = new ListItem(item_id);
- CriticalSectionScoped lock(critical_section_);
- PushFrontImpl(item);
- return 0;
-}
-
-int ListWrapper::PushFront(const void* ptr)
-{
- ListItem* item = new ListItem(ptr);
- CriticalSectionScoped lock(critical_section_);
- PushFrontImpl(item);
- return 0;
-}
-
-int ListWrapper::PopFront()
-{
- return Erase(first_);
-}
-
-int ListWrapper::PopBack()
-{
- return Erase(last_);
-}
-
-ListItem* ListWrapper::First() const
-{
- return first_;
-}
-
-ListItem* ListWrapper::Last() const
-{
- return last_;
-}
-
-ListItem* ListWrapper::Next(ListItem* item) const
-{
- if(!item)
- {
- return 0;
- }
- return item->next_;
-}
-
-ListItem* ListWrapper::Previous(ListItem* item) const
-{
- if (!item)
- {
- return 0;
- }
- return item->prev_;
-}
-
-int ListWrapper::Insert(ListItem* existing_previous_item, ListItem* new_item)
-{
- if (!new_item)
- {
- return -1;
- }
- // Allow existing_previous_item to be NULL if the list is empty.
- // TODO (hellner) why allow this? Keep it as is for now to avoid
- // breaking API contract.
- if (!existing_previous_item && !Empty())
- {
- return -1;
- }
- CriticalSectionScoped lock(critical_section_);
- if (!existing_previous_item)
- {
- PushBackImpl(new_item);
- return 0;
- }
- ListItem* next_item = existing_previous_item->next_;
- new_item->next_ = existing_previous_item->next_;
- new_item->prev_ = existing_previous_item;
- existing_previous_item->next_ = new_item;
- if (next_item)
- {
- next_item->prev_ = new_item;
- }
- else
- {
- last_ = new_item;
- }
- size_++;
- return 0;
-}
-
-int ListWrapper::InsertBefore(ListItem* existing_next_item,
- ListItem* new_item)
-{
- if (!new_item)
- {
- return -1;
- }
- // Allow existing_next_item to be NULL if the list is empty.
- // Todo: why allow this? Keep it as is for now to avoid breaking API
- // contract.
- if (!existing_next_item && !Empty())
- {
- return -1;
- }
- CriticalSectionScoped lock(critical_section_);
- if (!existing_next_item)
- {
- PushBackImpl(new_item);
- return 0;
- }
-
- ListItem* previous_item = existing_next_item->prev_;
- new_item->next_ = existing_next_item;
- new_item->prev_ = previous_item;
- existing_next_item->prev_ = new_item;
- if (previous_item)
- {
- previous_item->next_ = new_item;
- }
- else
- {
- first_ = new_item;
- }
- size_++;
- return 0;
-}
-
-int ListWrapper::Erase(ListItem* item)
-{
- if (!item)
- {
- return -1;
- }
- size_--;
- ListItem* previous_item = item->prev_;
- ListItem* next_item = item->next_;
- if (!previous_item)
- {
- if(next_item)
- {
- next_item->prev_ = 0;
- }
- first_ = next_item;
- }
- else
- {
- previous_item->next_ = next_item;
- }
- if (!next_item)
- {
- if(previous_item)
- {
- previous_item->next_ = 0;
- }
- last_ = previous_item;
- }
- else
- {
- next_item->prev_ = previous_item;
- }
- delete item;
- return 0;
-}
-
-void ListWrapper::PushBackImpl(ListItem* item)
-{
- if (Empty())
- {
- first_ = item;
- last_ = item;
- size_++;
- return;
- }
-
- item->prev_ = last_;
- last_->next_ = item;
- last_ = item;
- size_++;
-}
-
-void ListWrapper::PushFrontImpl(ListItem* item)
-{
- if (Empty())
- {
- first_ = item;
- last_ = item;
- size_++;
- return;
- }
-
- item->next_ = first_;
- first_->prev_ = item;
- first_ = item;
- size_++;
-}
-} //namespace webrtc
diff --git a/trunk/src/system_wrappers/source/list_no_stl.h b/trunk/src/system_wrappers/source/list_no_stl.h
deleted file mode 100644
index 26d844c..0000000
--- a/trunk/src/system_wrappers/source/list_no_stl.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_NO_STL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_NO_STL_H_
-
-#include "constructor_magic.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class ListNoStlItem
-{
-public:
- ListNoStlItem(const void* ptr);
- ListNoStlItem(const unsigned int item);
- virtual ~ListNoStlItem();
- void* GetItem() const;
- unsigned int GetUnsignedItem() const;
-
-protected:
- ListNoStlItem* next_;
- ListNoStlItem* prev_;
-
-private:
- friend class ListNoStl;
-
- const void* item_ptr_;
- const unsigned int item_;
- DISALLOW_COPY_AND_ASSIGN(ListNoStlItem);
-};
-
-
-class ListNoStl
-{
-public:
- ListNoStl();
- virtual ~ListNoStl();
-
- // ListWrapper functions
- unsigned int GetSize() const;
- int PushBack(const void* ptr);
- int PushBack(const unsigned int item_id);
- int PushFront(const void* ptr);
- int PushFront(const unsigned int item_id);
- int PopFront();
- int PopBack();
- bool Empty() const;
- ListNoStlItem* First() const;
- ListNoStlItem* Last() const;
- ListNoStlItem* Next(ListNoStlItem* item) const;
- ListNoStlItem* Previous(ListNoStlItem* item) const;
- int Erase(ListNoStlItem* item);
- int Insert(ListNoStlItem* existing_previous_item,
- ListNoStlItem* new_item);
-
- int InsertBefore(ListNoStlItem* existing_next_item,
- ListNoStlItem* new_item);
-
-private:
- void PushBack(ListNoStlItem* item);
- void PushFront(ListNoStlItem* item);
-
- CriticalSectionWrapper* critical_section_;
- ListNoStlItem* first_;
- ListNoStlItem* last_;
- unsigned int size_;
- DISALLOW_COPY_AND_ASSIGN(ListNoStl);
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_NO_STL_H_
diff --git a/trunk/src/system_wrappers/source/list_stl.cc b/trunk/src/system_wrappers/source/list_stl.cc
deleted file mode 100644
index dcc63c3..0000000
--- a/trunk/src/system_wrappers/source/list_stl.cc
+++ /dev/null
@@ -1,244 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "list_wrapper.h"
-
-#include "trace.h"
-
-namespace webrtc {
-ListItem::ListItem(const void* item)
- : this_iter_(),
- item_ptr_(item),
- item_(0)
-{
-}
-
-ListItem::ListItem(const unsigned int item)
- : this_iter_(),
- item_ptr_(0),
- item_(item)
-{
-}
-
-ListItem::~ListItem()
-{
-}
-
-void* ListItem::GetItem() const
-{
- return const_cast<void*>(item_ptr_);
-}
-
-unsigned int ListItem::GetUnsignedItem() const
-{
- return item_;
-}
-
-ListWrapper::ListWrapper() : list_()
-{
-}
-
-ListWrapper::~ListWrapper()
-{
- if (!Empty())
- {
- // TODO (hellner) I'm not sure this loggin is useful.
- WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1,
- "Potential memory leak in ListWrapper");
- // Remove all remaining list items.
- while (Erase(First()) == 0)
- {}
- }
-}
-
-bool ListWrapper::Empty() const
-{
- return list_.empty();
-}
-
-unsigned int ListWrapper::GetSize() const
-{
- return list_.size();
-}
-
-int ListWrapper::PushBack(const void* ptr)
-{
- ListItem* item = new ListItem(ptr);
- list_.push_back(item);
- return 0;
-}
-
-int ListWrapper::PushBack(const unsigned int item_id)
-{
- ListItem* item = new ListItem(item_id);
- list_.push_back(item);
- return 0;
-}
-
-int ListWrapper::PushFront(const unsigned int item_id)
-{
- ListItem* item = new ListItem(item_id);
- list_.push_front(item);
- return 0;
-}
-
-int ListWrapper::PushFront(const void* ptr)
-{
- ListItem* item = new ListItem(ptr);
- list_.push_front(item);
- return 0;
-}
-
-int ListWrapper::PopFront()
-{
- if(list_.empty())
- {
- return -1;
- }
- list_.pop_front();
- return 0;
-}
-
-int ListWrapper::PopBack()
-{
- if(list_.empty())
- {
- return -1;
- }
- list_.pop_back();
- return 0;
-}
-
-ListItem* ListWrapper::First() const
-{
- if(list_.empty())
- {
- return NULL;
- }
- std::list<ListItem*>::iterator item_iter = list_.begin();
- ListItem* return_item = (*item_iter);
- return_item->this_iter_ = item_iter;
- return return_item;
-}
-
-ListItem* ListWrapper::Last() const
-{
- if(list_.empty())
- {
- return NULL;
- }
- // std::list::end() addresses the last item + 1. Decrement so that the
- // actual last is accessed.
- std::list<ListItem*>::iterator item_iter = list_.end();
- --item_iter;
- ListItem* return_item = (*item_iter);
- return_item->this_iter_ = item_iter;
- return return_item;
-}
-
-ListItem* ListWrapper::Next(ListItem* item) const
-{
- if(item == NULL)
- {
- return NULL;
- }
- std::list<ListItem*>::iterator item_iter = item->this_iter_;
- ++item_iter;
- if (item_iter == list_.end())
- {
- return NULL;
- }
- ListItem* return_item = (*item_iter);
- return_item->this_iter_ = item_iter;
- return return_item;
-}
-
-ListItem* ListWrapper::Previous(ListItem* item) const
-{
- if(item == NULL)
- {
- return NULL;
- }
- std::list<ListItem*>::iterator item_iter = item->this_iter_;
- if (item_iter == list_.begin())
- {
- return NULL;
- }
- --item_iter;
- ListItem* return_item = (*item_iter);
- return_item->this_iter_ = item_iter;
- return return_item;
-}
-
-int ListWrapper::Insert(ListItem* existing_previous_item,
- ListItem* new_item)
-{
- // Allow existingPreviousItem to be NULL if the list is empty.
- // TODO (hellner) why allow this? Keep it as is for now to avoid
- // breaking API contract.
- if (!existing_previous_item && !Empty())
- {
- return -1;
- }
-
- if (!new_item)
- {
- return -1;
- }
-
- std::list<ListItem*>::iterator insert_location = list_.begin();
- if (!Empty())
- {
- insert_location = existing_previous_item->this_iter_;
- if(insert_location != list_.end())
- {
- ++insert_location;
- }
- }
-
- list_.insert(insert_location,new_item);
- return 0;
-}
-
-int ListWrapper::InsertBefore(ListItem* existing_next_item,
- ListItem* new_item)
-{
- // Allow existing_next_item to be NULL if the list is empty.
- // Todo: why allow this? Keep it as is for now to avoid breaking API
- // contract.
- if (!existing_next_item && !Empty())
- {
- return -1;
- }
- if (!new_item)
- {
- return -1;
- }
-
- std::list<ListItem*>::iterator insert_location = list_.begin();
- if (!Empty())
- {
- insert_location = existing_next_item->this_iter_;
- }
-
- list_.insert(insert_location,new_item);
- return 0;
-}
-
-int ListWrapper::Erase(ListItem* item)
-{
- if(item == NULL)
- {
- return -1;
- }
- list_.erase(item->this_iter_);
- return 0;
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/list_stl.h b/trunk/src/system_wrappers/source/list_stl.h
deleted file mode 100644
index b83a664..0000000
--- a/trunk/src/system_wrappers/source/list_stl.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_STL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_STL_H_
-
-#include <list>
-
-#include "constructor_magic.h"
-
-namespace webrtc {
-class ListItem
-{
-friend class ListWrapper;
-
-public:
- ListItem(const void* ptr);
- ListItem(const unsigned int item);
- virtual ~ListItem();
- void* GetItem() const;
- unsigned int GetUnsignedItem() const;
-
-private:
- mutable std::list<ListItem*>::iterator this_iter_;
- const void* item_ptr_;
- const unsigned int item_;
- DISALLOW_COPY_AND_ASSIGN(ListItem);
-};
-
-class ListWrapper
-{
-public:
- ListWrapper();
- ~ListWrapper();
-
- // ListWrapper functions
- unsigned int GetSize() const;
- int PushBack(const void* ptr);
- int PushBack(const unsigned int item_id);
- int PushFront(const void* ptr);
- int PushFront(const unsigned int item_id);
- int PopFront();
- int PopBack();
- bool Empty() const;
- ListItem* First() const;
- ListItem* Last() const;
- ListItem* Next(ListItem* item) const;
- ListItem* Previous(ListItem* item) const;
- int Erase(ListItem* item);
- int Insert(ListItem* existing_previous_item, ListItem* new_item);
- int InsertBefore(ListItem* existing_next_item, ListItem* new_item);
-
-private:
- mutable std::list<ListItem*> list_;
- DISALLOW_COPY_AND_ASSIGN(ListWrapper);
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_LIST_STL_H_
diff --git a/trunk/src/system_wrappers/source/list_unittest.cc b/trunk/src/system_wrappers/source/list_unittest.cc
deleted file mode 100644
index 4d32f59..0000000
--- a/trunk/src/system_wrappers/source/list_unittest.cc
+++ /dev/null
@@ -1,479 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "gtest/gtest.h"
-
-#include "system_wrappers/interface/list_wrapper.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-
-using ::webrtc::ListWrapper;
-using ::webrtc::ListItem;
-using ::webrtc::scoped_ptr;
-
-// Note: kNumberOfElements needs to be even.
-const unsigned int kNumberOfElements = 10;
-
-// An opaque implementation of dynamic or statically allocated unsigned ints.
-// This class makes it possible to use the exact same code for testing of both
-// the dynamic and static implementation of ListWrapper.
-// Clarification: ListWrapper has two versions of PushBack(..). It takes an
-// unsigned integer or a void pointer. The integer implementation takes care
-// of memory management. The void pointer version expect the caller to manage
-// the memory associated with the void pointer.
-// This class works like the integer version but can be implemented on top of
-// either the integer version or void pointer version of ListWrapper.
-// Note: the non-virtual fuctions behave the same for both versions.
-class ListWrapperSimple {
-public:
- static ListWrapperSimple* Create(bool static_allocation);
- virtual ~ListWrapperSimple() {}
-
- // These three functions should be used for manipulating ListItems so that
- // they are the type corresponding to the underlying implementation.
- virtual unsigned int GetUnsignedItem(
- const ListItem* item) const = 0;
- virtual ListItem* CreateListItem(unsigned int item_id) = 0;
- unsigned int GetSize() const {
- return list_.GetSize();
- }
- virtual int PushBack(const unsigned int item_id) = 0;
- virtual int PushFront(const unsigned int item_id) = 0;
- virtual int PopFront() = 0;
- virtual int PopBack() = 0;
- bool Empty() const {
- return list_.Empty();
- }
- ListItem* First() const {
- return list_.First();
- }
- ListItem* Last() const {
- return list_.Last();
- }
- ListItem* Next(ListItem* item) const {
- return list_.Next(item);
- }
- ListItem* Previous(ListItem* item) const {
- return list_.Previous(item);
- }
- virtual int Erase(ListItem* item) = 0;
- int Insert(ListItem* existing_previous_item,
- ListItem* new_item) {
- const int retval = list_.Insert(existing_previous_item, new_item);
- if (retval != 0) {
- EXPECT_TRUE(DestroyListItem(new_item));
- }
- return retval;
- }
-
- int InsertBefore(ListItem* existing_next_item,
- ListItem* new_item) {
- const int retval = list_.InsertBefore(existing_next_item, new_item);
- if (retval != 0) {
- EXPECT_TRUE(DestroyListItem(new_item));
- }
- return retval;
- }
-protected:
- ListWrapperSimple() {}
-
- virtual bool DestroyListItemContent(ListItem* item) = 0;
- bool DestroyListItem(ListItem* item) {
- const bool retval = DestroyListItemContent(item);
- delete item;
- return retval;
- }
-
- ListWrapper list_;
-};
-
-void ClearList(ListWrapperSimple* list_wrapper) {
- if (list_wrapper == NULL) {
- return;
- }
- ListItem* list_item = list_wrapper->First();
- while (list_item != NULL) {
- EXPECT_EQ(list_wrapper->Erase(list_item), 0);
- list_item = list_wrapper->First();
- }
-}
-
-class ListWrapperStatic : public ListWrapperSimple {
-public:
- ListWrapperStatic() {}
- virtual ~ListWrapperStatic() {
- ClearList(this);
- }
-
- virtual unsigned int GetUnsignedItem(const ListItem* item) const {
- return item->GetUnsignedItem();
- }
- virtual ListItem* CreateListItem(unsigned int item_id) {
- return new ListItem(item_id);
- }
- virtual bool DestroyListItemContent(ListItem* item) {
- return true;
- }
- virtual int PushBack(const unsigned int item_id) {
- return list_.PushBack(item_id);
- }
- virtual int PushFront(const unsigned int item_id) {
- return list_.PushFront(item_id);
- }
- virtual int PopFront() {
- return list_.PopFront();
- }
- virtual int PopBack() {
- return list_.PopBack();
- }
- virtual int Erase(ListItem* item) {
- return list_.Erase(item);
- }
-};
-
-class ListWrapperDynamic : public ListWrapperSimple {
-public:
- ListWrapperDynamic() {}
- virtual ~ListWrapperDynamic() {
- ClearList(this);
- }
-
- virtual unsigned int GetUnsignedItem(const ListItem* item) const {
- const unsigned int* return_value_pointer =
- reinterpret_cast<unsigned int*> (item->GetItem());
- if (return_value_pointer == NULL) {
- return -1;
- }
- return *return_value_pointer;
- }
- virtual ListItem* CreateListItem(unsigned int item_id) {
- unsigned int* item_id_pointer = new unsigned int;
- if (item_id_pointer == NULL) {
- return NULL;
- }
- *item_id_pointer = item_id;
- ListItem* return_value = new ListItem(
- reinterpret_cast<void*>(item_id_pointer));
- if (return_value == NULL) {
- delete item_id_pointer;
- return NULL;
- }
- return return_value;
- }
- virtual bool DestroyListItemContent(ListItem* item) {
- if (item == NULL) {
- return false;
- }
- bool return_value = false;
- unsigned int* item_id_ptr = reinterpret_cast<unsigned int*>(
- item->GetItem());
- if (item_id_ptr != NULL) {
- return_value = true;
- delete item_id_ptr;
- }
- return return_value;
- }
- virtual int PushBack(const unsigned int item_id) {
- unsigned int* item_id_ptr = new unsigned int;
- if (item_id_ptr == NULL) {
- return -1;
- }
- *item_id_ptr = item_id;
- const int return_value = list_.PushBack(
- reinterpret_cast<void*>(item_id_ptr));
- if (return_value != 0) {
- delete item_id_ptr;
- }
- return return_value;
- }
- virtual int PushFront(const unsigned int item_id) {
- unsigned int* item_id_ptr = new unsigned int;
- if (item_id_ptr == NULL) {
- return -1;
- }
- *item_id_ptr = item_id;
- const int return_value = list_.PushFront(
- reinterpret_cast<void*>(item_id_ptr));
- if (return_value != 0) {
- delete item_id_ptr;
- }
- return return_value;
- }
- virtual int PopFront() {
- return Erase(list_.First());
- }
- virtual int PopBack() {
- return Erase(list_.Last());
- }
- virtual int Erase(ListItem* item) {
- if (item == NULL) {
- return -1;
- }
- int retval = 0;
- if (!DestroyListItemContent(item)) {
- retval = -1;
- ADD_FAILURE();
- }
- if (list_.Erase(item) != 0) {
- retval = -1;
- }
- return retval;
- }
-};
-
-ListWrapperSimple* ListWrapperSimple::Create(bool static_allocation) {
- if(static_allocation)
- {
- return new ListWrapperStatic();
- }
- return new ListWrapperDynamic();
-}
-
-ListWrapperSimple* CreateAscendingList(bool static_allocation) {
- ListWrapperSimple* return_value = ListWrapperSimple::Create(
- static_allocation);
- if (return_value == NULL) {
- return NULL;
- }
- for (unsigned int i = 0; i < kNumberOfElements; ++i) {
- if (return_value->PushBack(i) == -1) {
- ClearList(return_value);
- delete return_value;
- return NULL;
- }
- }
- return return_value;
-}
-
-ListWrapperSimple* CreateDescendingList(bool static_allocation) {
- ListWrapperSimple* return_value = ListWrapperSimple::Create(
- static_allocation);
- if (return_value == NULL) {
- return NULL;
- }
- for (unsigned int i = 0; i < kNumberOfElements; ++i) {
- if (return_value->PushBack(kNumberOfElements - i - 1) == -1) {
- ClearList(return_value);
- delete return_value;
- return NULL;
- }
- }
- return return_value;
-}
-
-// [0,kNumberOfElements - 1,1,kNumberOfElements - 2,...] (this is why
-// kNumberOfElements need to be even)
-ListWrapperSimple* CreateInterleavedList(bool static_allocation) {
- ListWrapperSimple* return_value = ListWrapperSimple::Create(
- static_allocation);
- if (return_value == NULL) {
- return NULL;
- }
- unsigned int uneven_count = 0;
- unsigned int even_count = 0;
- for (unsigned int i = 0; i < kNumberOfElements; i++) {
- unsigned int push_value = 0;
- if ((i % 2) == 0) {
- push_value = even_count;
- even_count++;
- } else {
- push_value = kNumberOfElements - uneven_count - 1;
- uneven_count++;
- }
- if (return_value->PushBack(push_value) == -1) {
- ClearList(return_value);
- delete return_value;
- return NULL;
- }
- }
- return return_value;
-}
-
-void PrintList(const ListWrapperSimple* list) {
- ListItem* list_item = list->First();
- printf("[");
- while (list_item != NULL)
- {
- printf("%3u", list->GetUnsignedItem(list_item));
- list_item = list->Next(list_item);
- }
- printf("]\n");
-}
-
-bool CompareLists(const ListWrapperSimple* lhs, const ListWrapperSimple* rhs) {
- const unsigned int list_size = lhs->GetSize();
- if (lhs->GetSize() != rhs->GetSize()) {
- return false;
- }
- if (lhs->Empty()) {
- return rhs->Empty();
- }
- unsigned int i = 0;
- ListItem* lhs_item = lhs->First();
- ListItem* rhs_item = rhs->First();
- while (i < list_size) {
- if (lhs_item == NULL) {
- return false;
- }
- if (rhs_item == NULL) {
- return false;
- }
- if (lhs->GetUnsignedItem(lhs_item) != rhs->GetUnsignedItem(rhs_item)) {
- return false;
- }
- i++;
- lhs_item = lhs->Next(lhs_item);
- rhs_item = rhs->Next(rhs_item);
- }
- return true;
-}
-
-TEST(ListWrapperTest,ReverseNewIntList) {
- // Create a new temporary list with elements reversed those of
- // new_int_list_
- const scoped_ptr<ListWrapperSimple> descending_list(
- CreateDescendingList(rand()%2));
- ASSERT_FALSE(descending_list.get() == NULL);
- ASSERT_FALSE(descending_list->Empty());
- ASSERT_EQ(kNumberOfElements,descending_list->GetSize());
-
- const scoped_ptr<ListWrapperSimple> ascending_list(
- CreateAscendingList(rand()%2));
- ASSERT_FALSE(ascending_list.get() == NULL);
- ASSERT_FALSE(ascending_list->Empty());
- ASSERT_EQ(kNumberOfElements,ascending_list->GetSize());
-
- scoped_ptr<ListWrapperSimple> list_to_reverse(
- ListWrapperSimple::Create(rand()%2));
-
- // Reverse the list using PushBack and Previous.
- for (ListItem* item = ascending_list->Last(); item != NULL;
- item = ascending_list->Previous(item)) {
- list_to_reverse->PushBack(ascending_list->GetUnsignedItem(item));
- }
-
- ASSERT_TRUE(CompareLists(descending_list.get(), list_to_reverse.get()));
-
- scoped_ptr<ListWrapperSimple> list_to_un_reverse(
- ListWrapperSimple::Create(rand()%2));
- ASSERT_FALSE(list_to_un_reverse.get() == NULL);
- // Reverse the reversed list using PushFront and Next.
- for (ListItem* item = list_to_reverse->First(); item != NULL;
- item = list_to_reverse->Next(item)) {
- list_to_un_reverse->PushFront(list_to_reverse->GetUnsignedItem(item));
- }
- ASSERT_TRUE(CompareLists(ascending_list.get(), list_to_un_reverse.get()));
-}
-
-TEST(ListWrapperTest,PopTest) {
- scoped_ptr<ListWrapperSimple> ascending_list(CreateAscendingList(rand()%2));
- ASSERT_FALSE(ascending_list.get() == NULL);
- ASSERT_FALSE(ascending_list->Empty());
- EXPECT_EQ(0, ascending_list->PopFront());
- EXPECT_EQ(1U, ascending_list->GetUnsignedItem(ascending_list->First()));
-
- EXPECT_EQ(0, ascending_list->PopBack());
- EXPECT_EQ(kNumberOfElements - 2, ascending_list->GetUnsignedItem(
- ascending_list->Last()));
- EXPECT_EQ(kNumberOfElements - 2, ascending_list->GetSize());
-}
-
-// Use Insert to interleave two lists.
-TEST(ListWrapperTest,InterLeaveTest) {
- scoped_ptr<ListWrapperSimple> interleave_list(
- CreateAscendingList(rand()%2));
- ASSERT_FALSE(interleave_list.get() == NULL);
- ASSERT_FALSE(interleave_list->Empty());
-
- scoped_ptr<ListWrapperSimple> descending_list(
- CreateDescendingList(rand()%2));
- ASSERT_FALSE(descending_list.get() == NULL);
-
- for (unsigned int i = 0; i < kNumberOfElements/2; ++i) {
- ASSERT_EQ(0, interleave_list->PopBack());
- ASSERT_EQ(0, descending_list->PopBack());
- }
- ASSERT_EQ(kNumberOfElements/2, interleave_list->GetSize());
- ASSERT_EQ(kNumberOfElements/2, descending_list->GetSize());
-
- unsigned int insert_position = kNumberOfElements/2;
- ASSERT_EQ(insert_position * 2, kNumberOfElements);
- while (!descending_list->Empty())
- {
- ListItem* item = descending_list->Last();
- ASSERT_FALSE(item == NULL);
-
- const unsigned int item_id = descending_list->GetUnsignedItem(item);
- ASSERT_EQ(0, descending_list->Erase(item));
-
- ListItem* insert_item = interleave_list->CreateListItem(item_id);
- ASSERT_FALSE(insert_item == NULL);
- item = interleave_list->First();
- ASSERT_FALSE(item == NULL);
- for (unsigned int j = 0; j < insert_position - 1; ++j) {
- item = interleave_list->Next(item);
- ASSERT_FALSE(item == NULL);
- }
- EXPECT_EQ(0, interleave_list->Insert(item, insert_item));
- --insert_position;
- }
-
- scoped_ptr<ListWrapperSimple> interleaved_list(
- CreateInterleavedList(rand()%2));
- ASSERT_FALSE(interleaved_list.get() == NULL);
- ASSERT_FALSE(interleaved_list->Empty());
- ASSERT_TRUE(CompareLists(interleaved_list.get(), interleave_list.get()));
-}
-
-// Use InsertBefore to interleave two lists.
-TEST(ListWrapperTest,InterLeaveTestII) {
- scoped_ptr<ListWrapperSimple> interleave_list(
- CreateDescendingList(rand()%2));
- ASSERT_FALSE(interleave_list.get() == NULL);
- ASSERT_FALSE(interleave_list->Empty());
-
- scoped_ptr<ListWrapperSimple> ascending_list(CreateAscendingList(rand()%2));
- ASSERT_FALSE(ascending_list.get() == NULL);
-
- for (unsigned int i = 0; i < kNumberOfElements/2; ++i) {
- ASSERT_EQ(0, interleave_list->PopBack());
- ASSERT_EQ(0, ascending_list->PopBack());
- }
- ASSERT_EQ(kNumberOfElements/2, interleave_list->GetSize());
- ASSERT_EQ(kNumberOfElements/2, ascending_list->GetSize());
-
- unsigned int insert_position = kNumberOfElements/2;
- ASSERT_EQ(insert_position * 2, kNumberOfElements);
- while (!ascending_list->Empty())
- {
- ListItem* item = ascending_list->Last();
- ASSERT_FALSE(item == NULL);
-
- const unsigned int item_id = ascending_list->GetUnsignedItem(item);
- ASSERT_EQ(0,ascending_list->Erase(item));
-
- ListItem* insert_item = interleave_list->CreateListItem(item_id);
- ASSERT_FALSE(insert_item == NULL);
- item = interleave_list->First();
- ASSERT_FALSE(item == NULL);
- for (unsigned int j = 0; j < insert_position - 1; ++j) {
- item = interleave_list->Next(item);
- ASSERT_FALSE(item == NULL);
- }
- EXPECT_EQ(interleave_list->InsertBefore(item, insert_item), 0);
- --insert_position;
- }
-
- scoped_ptr<ListWrapperSimple> interleaved_list(
- CreateInterleavedList(rand()%2));
- ASSERT_FALSE(interleaved_list.get() == NULL);
- ASSERT_FALSE(interleaved_list->Empty());
-
- ASSERT_TRUE(CompareLists(interleaved_list.get(), interleave_list.get()));
-}
diff --git a/trunk/src/system_wrappers/source/map.cc b/trunk/src/system_wrappers/source/map.cc
deleted file mode 100644
index 331da32..0000000
--- a/trunk/src/system_wrappers/source/map.cc
+++ /dev/null
@@ -1,166 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "map_wrapper.h"
-
-#include "trace.h"
-
-namespace webrtc {
-MapItem::MapItem(int id, void* item) : item_id_(id), item_pointer_(item)
-{
-}
-
-MapItem::~MapItem()
-{
-}
-
-void* MapItem::GetItem()
-{
- return item_pointer_;
-}
-
-int MapItem::GetId()
-{
- return item_id_;
-}
-
-unsigned int MapItem::GetUnsignedId()
-{
- return static_cast<unsigned int>(item_id_);
-}
-
-void MapItem::SetItem(void* ptr)
-{
- item_pointer_ = ptr;
-}
-
-MapWrapper::MapWrapper() : map_()
-{
-}
-
-MapWrapper::~MapWrapper()
-{
- if (!map_.empty())
- {
- WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1,
- "Potential memory leak in MapWrapper");
- // Remove all map items. Please note that std::map::clear() can't be
- // used because each item has some dynamically allocated memory
- // associated with it (i.e. using std::map::clear would introduce a
- // memory leak).
- while (Erase(First()) == 0)
- {}
- }
-}
-
-int MapWrapper::Size() const
-{
- return (int)map_.size();
-}
-
-int MapWrapper::Insert(int id, void* ptr)
-{
- map_[id] = new MapItem(id,ptr);
- return 0;
-}
-
-MapItem* MapWrapper::First() const
-{
- std::map<int, MapItem*>::const_iterator it = map_.begin();
- if (it != map_.end())
- {
- return it->second;
- }
- return 0;
-}
-
-MapItem* MapWrapper::Last() const
-{
- std::map<int, MapItem*>::const_reverse_iterator it = map_.rbegin();
- if (it != map_.rend())
- {
- return it->second;
- }
- return 0;
-}
-
-MapItem* MapWrapper::Next(MapItem* item) const
-{
- if (item == 0)
- {
- return 0;
- }
- std::map<int, MapItem*>::const_iterator it = map_.find(item->item_id_);
- if (it != map_.end())
- {
- it++;
- if (it != map_.end())
- {
- return it->second;
- }
- }
- return 0;
-}
-
-MapItem* MapWrapper::Previous(MapItem* item) const
-{
- if (item == 0)
- {
- return 0;
- }
-
- std::map<int, MapItem*>::const_iterator it = map_.find(item->item_id_);
- if ((it != map_.end()) &&
- (it != map_.begin()))
- {
- --it;
- return it->second;
- }
- return 0;
-}
-
-MapItem* MapWrapper::Find(int id) const
-{
- std::map<int, MapItem*>::const_iterator it = map_.find(id);
- if (it != map_.end())
- {
- return it->second;
- }
- return 0;
-}
-
-int MapWrapper::Erase(MapItem* item)
-{
- if (item == 0)
- {
- return -1;
- }
- std::map<int, MapItem*>::iterator it = map_.find(item->item_id_);
- if (it != map_.end())
- {
- delete it->second;
- map_.erase(it);
- return 0;
- }
- return -1;
-}
-
-int MapWrapper::Erase(const int id)
-{
- std::map<int, MapItem*>::iterator it = map_.find(id);
- if (it != map_.end())
- {
- delete it->second;
- map_.erase(it);
- return 0;
- }
- return -1;
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/map_no_stl.cc b/trunk/src/system_wrappers/source/map_no_stl.cc
deleted file mode 100644
index ef93a1f..0000000
--- a/trunk/src/system_wrappers/source/map_no_stl.cc
+++ /dev/null
@@ -1,217 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "map_no_stl.h"
-
-#include "critical_section_wrapper.h"
-#include "trace.h"
-
-namespace webrtc {
-MapNoStlItem::MapNoStlItem(int id, void* item)
- : next_(0),
- prev_(0),
- item_id_(id),
- item_ptr_(item)
-{
-}
-
-MapNoStlItem::~MapNoStlItem()
-{
-}
-
-void* MapNoStlItem::GetItem()
-{
- return item_ptr_;
-}
-
-int MapNoStlItem::GetId()
-{
- return item_id_;
-}
-
-unsigned int MapNoStlItem::GetUnsignedId()
-{
- return static_cast<unsigned int>(item_id_);
-}
-
-void MapNoStlItem::SetItem(void* ptr)
-{
- item_ptr_ = ptr;
-}
-
-MapNoStl::MapNoStl()
- : critical_section_(CriticalSectionWrapper::CreateCriticalSection()),
- first_(0),
- last_(0),
- size_(0)
-{
-}
-
-MapNoStl::~MapNoStl()
-{
- if (First())
- {
- WEBRTC_TRACE(kTraceMemory, kTraceUtility, -1,
- "Potential memory leak in MapNoStl");
- while (Erase(First()) == 0)
- {}
- }
- delete critical_section_;
-}
-
-int MapNoStl::Size() const
-{
- return size_;
-}
-
-int MapNoStl::Insert(int id, void* ptr)
-{
- MapNoStlItem* new_item = new MapNoStlItem(id, ptr);
-
- CriticalSectionScoped lock(critical_section_);
- MapNoStlItem* item = first_;
- size_++;
- if (!item)
- {
- first_ = new_item;
- last_ = new_item;
- return 0;
- }
- while(item->next_)
- {
- // Three scenarios
- // 1. Item should be inserted first.
- // 2. Item should be inserted between two items
- // 3. Item should be inserted last
- if (item->GetId() > id)
- {
- new_item->next_ = item;
- item->prev_ = new_item;
- if (item == first_)
- {
- first_ = new_item;
- }
- else
- {
- new_item->prev_ = item->prev_;
- new_item->prev_->next_ = new_item;
- }
- return 0;
- }
- item = item->next_;
- }
- // 3
- item->next_ = new_item;
- new_item->prev_ = item;
- last_ = new_item;
- return 0;
-}
-
-MapNoStlItem* MapNoStl::First() const
-{
- return first_;
-}
-
-MapNoStlItem* MapNoStl::Last() const
-{
- return last_;
-}
-
-MapNoStlItem* MapNoStl::Next(MapNoStlItem* item) const
-{
- if (!item)
- {
- return 0;
- }
- return item->next_;
-}
-
-MapNoStlItem* MapNoStl::Previous(MapNoStlItem* item) const
-{
- if (!item)
- {
- return 0;
- }
- return item->prev_;
-}
-
-MapNoStlItem* MapNoStl::Find(int id) const
-{
- CriticalSectionScoped lock(critical_section_);
- MapNoStlItem* item = Locate(id);
- return item;
-}
-
-int MapNoStl::Erase(MapNoStlItem* item)
-{
- if(!item)
- {
- return -1;
- }
- CriticalSectionScoped lock(critical_section_);
- return Remove(item);
-}
-
-int MapNoStl::Erase(const int id)
-{
- CriticalSectionScoped lock(critical_section_);
- MapNoStlItem* item = Locate(id);
- if(!item)
- {
- return -1;
- }
- return Remove(item);
-}
-
-MapNoStlItem* MapNoStl::Locate(int id) const
-{
- MapNoStlItem* item = first_;
- while(item)
- {
- if (item->GetId() == id)
- {
- return item;
- }
- item = item->next_;
- }
- return 0;
-}
-
-int MapNoStl::Remove(MapNoStlItem* item)
-{
- if (!item)
- {
- return -1;
- }
- size_--;
- MapNoStlItem* previous_item = item->prev_;
- MapNoStlItem* next_item = item->next_;
- if (!previous_item)
- {
- next_item->prev_ = 0;
- first_ = next_item;
- }
- else
- {
- previous_item->next_ = next_item;
- }
- if (!next_item)
- {
- previous_item->next_ = 0;
- last_ = previous_item;
- }
- else
- {
- next_item->prev_ = previous_item;
- }
- delete item;
- return 0;
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/map_no_stl.h b/trunk/src/system_wrappers/source/map_no_stl.h
deleted file mode 100644
index 51bc011..0000000
--- a/trunk/src/system_wrappers/source/map_no_stl.h
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_MAP_NO_STL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_MAP_NO_STL_H_
-
-#include "constructor_magic.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class MapNoStlItem
-{
-friend class Map;
-
-public:
- MapNoStlItem(int id, void* ptr);
- virtual ~MapNoStlItem();
- void* GetItem();
- int GetId();
- unsigned int GetUnsignedId();
- void SetItem(void* ptr);
-
-protected:
- MapNoStlItem* next_;
- MapNoStlItem* prev_;
-
-private:
- int item_id_;
- void* item_ptr_;
- DISALLOW_COPY_AND_ASSIGN(MapNoStlItem);
-};
-
-class MapNoStl
-{
-public:
- MapNoStl();
- virtual ~MapNoStl();
-
- // MapWrapper functions.
- int Insert(int id, void* ptr);
- int Erase(MapNoStlItem* item);
- int Erase(int id);
- int Size() const;
- MapNoStlItem* First() const;
- MapNoStlItem* Last() const;
- MapNoStlItem* Next(MapNoStlItem* item) const;
- MapNoStlItem* Previous(MapNoStlItem* item) const;
- MapNoStlItem* Find(int id) const;
-
-private:
- MapNoStlItem* Locate(int id) const;
- int Remove(MapNoStlItem* item);
-
- CriticalSection* critical_section_;
- MapNoStlItem* first_;
- MapNoStlItem* last_;
- int size_;
- DISALLOW_COPY_AND_ASSIGN(MapNoStl);
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_MAP_NO_STL_H_
diff --git a/trunk/src/system_wrappers/source/map_unittest.cc b/trunk/src/system_wrappers/source/map_unittest.cc
deleted file mode 100644
index 1c85a92..0000000
--- a/trunk/src/system_wrappers/source/map_unittest.cc
+++ /dev/null
@@ -1,231 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "gtest/gtest.h"
-
-#include "map_wrapper.h"
-
-using ::webrtc::MapWrapper;
-using ::webrtc::MapItem;
-
-const int kNumberOfElements = 10;
-
-int* ItemPointer(MapItem* item) {
- if (item == NULL) {
- return NULL;
- }
- return reinterpret_cast<int*>(item->GetItem());
-}
-
-bool DeleteItemContent(MapItem* item) {
- if(item == NULL) {
- return false;
- }
- int* value_ptr = ItemPointer(item);
- delete value_ptr;
- return true;
-}
-
-int ItemValue(MapItem* item) {
- if (item == NULL) {
- return -1;
- }
- const int* value_ptr = ItemPointer(item);
- if (value_ptr == 0) {
- return -1;
- }
- return *value_ptr;
-}
-
-void PrintToConsole(const char* message, bool supress) {
- if (supress) {
- return;
- }
- printf("%s", message);
-}
-
-bool CreateAscendingMap(MapWrapper* ascending_map) {
- int* insert_value = NULL;
- for (int i = 0; i < kNumberOfElements; ++i) {
- insert_value = new int;
- if (insert_value == NULL) {
- return false;
- }
- *insert_value = i;
- if (0 != ascending_map->Insert(
- i,
- reinterpret_cast<void*>(insert_value))) {
- return false;
- }
- }
- return true;
-}
-
-bool ClearMap(MapWrapper* clear_map) {
- bool success = true;
- while (clear_map->Size() != 0) {
- MapItem* remove_item = clear_map->First();
- if (remove_item == NULL) {
- return false;
- }
- if (!DeleteItemContent(remove_item)) {
- success = false;
- }
- if (clear_map->Erase(remove_item) != 0) {
- return false;
- }
- }
- return success;
-}
-
-void PrintMapItem(MapItem* item, bool supress) {
- const int id = item->GetId();
- const int value = ItemValue(item);
- char print_buffer[16];
- sprintf(print_buffer, "(%3i,%3i) ", id, value);
- PrintToConsole(print_buffer, supress);
-}
-
-// Succeeds only if all the items were printed.
-bool PrintMap(const MapWrapper& print_map, bool supress) {
- const int elements_to_print = print_map.Size();
- int elements_printed = 0;
- MapItem* item = print_map.First();
- PrintToConsole("[", supress);
- while (item != NULL) {
- PrintMapItem(item, supress);
- ++elements_printed;
- item = print_map.Next(item);
- }
- PrintToConsole("]\n", supress);
- return elements_printed == elements_to_print;
-}
-
-// Succeeds only if all the items were printed.
-bool ReversePrintMap(const MapWrapper& print_map, bool supress) {
- const int elements_to_print = print_map.Size();
- int elements_printed = 0;
- MapItem* item = print_map.Last();
- PrintToConsole("[", supress);
- while (item != NULL) {
- PrintMapItem(item, supress);
- ++elements_printed;
- item = print_map.Previous(item);
- }
- PrintToConsole("]\n", supress);
- return elements_printed == elements_to_print;
-}
-
-// Returns true if the map items contain the same item.
-bool CompareItems(MapItem* lhs_item, MapItem* rhs_item) {
- if ((lhs_item == NULL) || (rhs_item == NULL)) {
- return false;
- }
- if (lhs_item->GetId() != rhs_item->GetId()) {
- return false;
- }
- return lhs_item->GetItem() == rhs_item->GetItem();
-}
-
-// Returns true if the map contains the same items.
-bool CompareMaps(const MapWrapper& lhs, const MapWrapper& rhs) {
- const int map_size = lhs.Size();
- if (map_size != rhs.Size()) {
- return false;
- }
- int item_count = 0;
- MapItem* lhs_item = lhs.First();
- while (lhs_item != NULL) {
- MapItem* rhs_item = rhs.Find(lhs_item->GetId());
- if (rhs_item == NULL) {
- return false;
- }
- if (!CompareItems(lhs_item, rhs_item)) {
- return false;
- }
- ++item_count;
- lhs_item = lhs.Next(lhs_item);
- }
- return item_count == map_size;
-}
-
-class MapWrapperTest : public ::testing::Test {
-protected:
- virtual void SetUp() {
- ASSERT_TRUE(CreateAscendingMap(&ascending_map_));
- }
-
- virtual void TearDown() {
- EXPECT_TRUE(ClearMap(&ascending_map_));
- }
- MapWrapper ascending_map_;
-};
-
-TEST_F(MapWrapperTest,RemoveTest) {
- // Erase using int id
- { // Create local scope to avoid accidental re-use
- MapItem* item_first = ascending_map_.First();
- ASSERT_FALSE(item_first == NULL);
- const int first_value_id = item_first->GetId();
- const int first_value = ItemValue(item_first);
- EXPECT_TRUE(first_value == 0);
- EXPECT_EQ(first_value_id,first_value);
- EXPECT_FALSE(NULL == ascending_map_.Find(first_value_id));
- EXPECT_TRUE(DeleteItemContent(item_first));
- ascending_map_.Erase(first_value_id);
- EXPECT_TRUE(NULL == ascending_map_.Find(first_value_id));
- EXPECT_EQ(kNumberOfElements-1,ascending_map_.Size());
- }
-
- // Erase using MapItem* item
- MapItem* item_last = ascending_map_.Last();
- ASSERT_FALSE(item_last == NULL);
- const int last_value_id = item_last->GetId();
- const int last_value = ItemValue(item_last);
- EXPECT_TRUE(last_value == kNumberOfElements - 1);
- EXPECT_EQ(last_value_id, last_value);
- EXPECT_FALSE(NULL == ascending_map_.Find(last_value_id));
- EXPECT_TRUE(DeleteItemContent(item_last));
- ascending_map_.Erase(last_value_id);
- EXPECT_TRUE(NULL == ascending_map_.Find(last_value_id));
- EXPECT_EQ(kNumberOfElements-2,ascending_map_.Size());
-}
-
-TEST_F(MapWrapperTest, PrintTest) {
- const bool supress = true; // Don't spam the console
-
- EXPECT_TRUE(PrintMap(ascending_map_, supress));
- EXPECT_TRUE(ReversePrintMap(ascending_map_, supress));
-}
-
-TEST_F(MapWrapperTest, CopyTest) {
- MapWrapper compare_map;
- ASSERT_TRUE(CreateAscendingMap(&compare_map));
- const int map_size = compare_map.Size();
- ASSERT_EQ(ascending_map_.Size(), map_size);
- // CompareMaps compare the pointers not value of the pointers.
- // (the values are the same since both are ascending maps).
- EXPECT_FALSE(CompareMaps(compare_map,ascending_map_));
-
- int copy_count = 0;
- MapItem* ascend_item = ascending_map_.First();
- while (ascend_item != NULL) {
- MapItem* compare_item = compare_map.Find(ascend_item->GetId());
- ASSERT_FALSE(compare_item == NULL);
- DeleteItemContent(compare_item);
- compare_item->SetItem(ascend_item->GetItem());
- ascend_item = ascending_map_.Next(ascend_item);
- ++copy_count;
- }
- EXPECT_TRUE(CompareMaps(compare_map,ascending_map_));
- while (compare_map.Erase(compare_map.First()) == 0) {
- }
- EXPECT_EQ(map_size, copy_count);
-}
diff --git a/trunk/src/system_wrappers/source/rw_lock.cc b/trunk/src/system_wrappers/source/rw_lock.cc
deleted file mode 100644
index 16da0e3..0000000
--- a/trunk/src/system_wrappers/source/rw_lock.cc
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rw_lock_wrapper.h"
-
-#include <assert.h>
-
-#if defined(_WIN32)
- #include "rw_lock_win.h"
-#else
- #include "rw_lock_posix.h"
-#endif
-
-namespace webrtc {
-RWLockWrapper* RWLockWrapper::CreateRWLock()
-{
-#ifdef _WIN32
- RWLockWrapper* lock = new RWLockWindows();
-#else
- RWLockWrapper* lock = new RWLockPosix();
-#endif
- if(lock->Init() != 0)
- {
- delete lock;
- assert(false);
- return NULL;
- }
- return lock;
-}
-
-RWLockWrapper::~RWLockWrapper()
-{
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/rw_lock_generic.cc b/trunk/src/system_wrappers/source/rw_lock_generic.cc
deleted file mode 100644
index a468ef3..0000000
--- a/trunk/src/system_wrappers/source/rw_lock_generic.cc
+++ /dev/null
@@ -1,106 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rw_lock_generic.h"
-
-#include "condition_variable_wrapper.h"
-#include "critical_section_wrapper.h"
-
-namespace webrtc {
-RWLockWrapperGeneric::RWLockWrapperGeneric()
- : _readersActive(0),
- _writerActive(false),
- _readersWaiting(0),
- _writersWaiting(0)
-{
- _critSectPtr = CriticalSectionWrapper::CreateCriticalSection();
- _readCondPtr = ConditionVariableWrapper::CreateConditionVariable();
- _writeCondPtr = ConditionVariableWrapper::CreateConditionVariable();
-}
-
-RWLockWrapperGeneric::~RWLockWrapperGeneric()
-{
- delete _writeCondPtr;
- delete _readCondPtr;
- delete _critSectPtr;
-}
-
-int RWLockWrapperGeneric::Init()
-{
- return 0;
-}
-
-void RWLockWrapperGeneric::AcquireLockExclusive()
-{
- _critSectPtr->Enter();
-
- if (_writerActive || _readersActive > 0)
- {
- ++_writersWaiting;
-
- while (_writerActive || _readersActive > 0)
- {
- _writeCondPtr->SleepCS(*_critSectPtr);
- }
-
- --_writersWaiting;
- }
- _writerActive = true;
- _critSectPtr->Leave();
-}
-
-void RWLockWrapperGeneric::ReleaseLockExclusive()
-{
- _critSectPtr->Enter();
-
- _writerActive = false;
-
- if (_writersWaiting > 0)
- {
- _writeCondPtr->Wake();
-
- }else if (_readersWaiting > 0)
- {
- _readCondPtr->WakeAll();
- }
- _critSectPtr->Leave();
-}
-
-void RWLockWrapperGeneric::AcquireLockShared()
-{
- _critSectPtr->Enter();
-
- if (_writerActive || _writersWaiting > 0)
- {
- ++_readersWaiting;
-
- while (_writerActive || _writersWaiting > 0)
- {
- _readCondPtr->SleepCS(*_critSectPtr);
- }
- --_readersWaiting;
- }
- ++_readersActive;
- _critSectPtr->Leave();
-}
-
-void RWLockWrapperGeneric::ReleaseLockShared()
-{
- _critSectPtr->Enter();
-
- --_readersActive;
-
- if (_readersActive == 0 && _writersWaiting > 0)
- {
- _writeCondPtr->Wake();
- }
- _critSectPtr->Leave();
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/rw_lock_generic.h b/trunk/src/system_wrappers/source/rw_lock_generic.h
deleted file mode 100644
index fff5e5d..0000000
--- a/trunk/src/system_wrappers/source/rw_lock_generic.h
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_GENERIC_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_GENERIC_H_
-
-#include "rw_lock_wrapper.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class ConditionVariableWrapper;
-
-class RWLockWrapperGeneric : public RWLockWrapper
-{
-public:
- RWLockWrapperGeneric();
- virtual ~RWLockWrapperGeneric();
-
- virtual void AcquireLockExclusive();
- virtual void ReleaseLockExclusive();
-
- virtual void AcquireLockShared();
- virtual void ReleaseLockShared();
-
-protected:
- virtual int Init();
-
-private:
- CriticalSectionWrapper* _critSectPtr;
- ConditionVariableWrapper* _readCondPtr;
- ConditionVariableWrapper* _writeCondPtr;
-
- int _readersActive;
- bool _writerActive;
- int _readersWaiting;
- int _writersWaiting;
-};
-} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_GENERIC_H_
diff --git a/trunk/src/system_wrappers/source/rw_lock_posix.cc b/trunk/src/system_wrappers/source/rw_lock_posix.cc
deleted file mode 100644
index 81a161e..0000000
--- a/trunk/src/system_wrappers/source/rw_lock_posix.cc
+++ /dev/null
@@ -1,47 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rw_lock_posix.h"
-
-namespace webrtc {
-RWLockPosix::RWLockPosix() : _lock()
-{
-}
-
-RWLockPosix::~RWLockPosix()
-{
- pthread_rwlock_destroy(&_lock);
-}
-
-int RWLockPosix::Init()
-{
- return pthread_rwlock_init(&_lock, 0);
-}
-
-void RWLockPosix::AcquireLockExclusive()
-{
- pthread_rwlock_wrlock(&_lock);
-}
-
-void RWLockPosix::ReleaseLockExclusive()
-{
- pthread_rwlock_unlock(&_lock);
-}
-
-void RWLockPosix::AcquireLockShared()
-{
- pthread_rwlock_rdlock(&_lock);
-}
-
-void RWLockPosix::ReleaseLockShared()
-{
- pthread_rwlock_unlock(&_lock);
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/rw_lock_posix.h b/trunk/src/system_wrappers/source/rw_lock_posix.h
deleted file mode 100644
index 929bbb8..0000000
--- a/trunk/src/system_wrappers/source/rw_lock_posix.h
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_POSIX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_POSIX_H_
-
-#include "rw_lock_wrapper.h"
-
-#include <pthread.h>
-
-namespace webrtc {
-class RWLockPosix : public RWLockWrapper
-{
-public:
- RWLockPosix();
- virtual ~RWLockPosix();
-
- virtual void AcquireLockExclusive();
- virtual void ReleaseLockExclusive();
-
- virtual void AcquireLockShared();
- virtual void ReleaseLockShared();
-
-protected:
- virtual int Init();
-
-private:
- pthread_rwlock_t _lock;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_POSIX_H_
diff --git a/trunk/src/system_wrappers/source/rw_lock_win.cc b/trunk/src/system_wrappers/source/rw_lock_win.cc
deleted file mode 100644
index 82cd0ac..0000000
--- a/trunk/src/system_wrappers/source/rw_lock_win.cc
+++ /dev/null
@@ -1,186 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "rw_lock_win.h"
-
-#include "critical_section_wrapper.h"
-#include "condition_variable_wrapper.h"
-#include "trace.h"
-
-// TODO (hellner) why not just use the rw_lock_generic.cc solution if
-// native is not supported? Unnecessary redundancy!
-
-namespace webrtc {
-bool RWLockWindows::_winSupportRWLockPrimitive = false;
-static HMODULE library = NULL;
-
-PInitializeSRWLock _PInitializeSRWLock;
-PAcquireSRWLockExclusive _PAcquireSRWLockExclusive;
-PAcquireSRWLockShared _PAcquireSRWLockShared;
-PReleaseSRWLockShared _PReleaseSRWLockShared;
-PReleaseSRWLockExclusive _PReleaseSRWLockExclusive;
-
-RWLockWindows::RWLockWindows()
- : _critSectPtr(NULL),
- _readCondPtr(NULL),
- _writeCondPtr(NULL),
- _readersActive(0),
- _writerActive(false),
- _readersWaiting(0),
- _writersWaiting(0)
-{
-}
-
-RWLockWindows::~RWLockWindows()
-{
- delete _writeCondPtr;
- delete _readCondPtr;
- delete _critSectPtr;
-}
-
-int RWLockWindows::Init()
-{
- if(!library)
- {
- // Use native implementation if supported (i.e Vista+)
- library = LoadLibrary(TEXT("Kernel32.dll"));
- if(library)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
- "Loaded Kernel.dll");
-
- _PInitializeSRWLock =
- (PInitializeSRWLock)GetProcAddress(
- library,
- "InitializeSRWLock");
-
- _PAcquireSRWLockExclusive =
- (PAcquireSRWLockExclusive)GetProcAddress(
- library,
- "AcquireSRWLockExclusive");
- _PReleaseSRWLockExclusive =
- (PReleaseSRWLockExclusive)GetProcAddress(
- library,
- "ReleaseSRWLockExclusive");
- _PAcquireSRWLockShared =
- (PAcquireSRWLockShared)GetProcAddress(
- library,
- "AcquireSRWLockShared");
- _PReleaseSRWLockShared =
- (PReleaseSRWLockShared)GetProcAddress(
- library,
- "ReleaseSRWLockShared");
-
- if( _PInitializeSRWLock &&
- _PAcquireSRWLockExclusive &&
- _PReleaseSRWLockExclusive &&
- _PAcquireSRWLockShared &&
- _PReleaseSRWLockShared )
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
- "Loaded Simple RW Lock");
- _winSupportRWLockPrimitive = true;
- }
- }
- }
- if(_winSupportRWLockPrimitive)
- {
- _PInitializeSRWLock(&_lock);
- } else {
- _critSectPtr = CriticalSectionWrapper::CreateCriticalSection();
- _readCondPtr = ConditionVariableWrapper::CreateConditionVariable();
- _writeCondPtr = ConditionVariableWrapper::CreateConditionVariable();
- }
- return 0;
-}
-
-void RWLockWindows::AcquireLockExclusive()
-{
- if (_winSupportRWLockPrimitive)
- {
- _PAcquireSRWLockExclusive(&_lock);
- } else {
- _critSectPtr->Enter();
-
- if (_writerActive || _readersActive > 0)
- {
- ++_writersWaiting;
- while (_writerActive || _readersActive > 0)
- {
- _writeCondPtr->SleepCS(*_critSectPtr);
- }
- --_writersWaiting;
- }
- _writerActive = true;
- _critSectPtr->Leave();
- }
-}
-
-void RWLockWindows::ReleaseLockExclusive()
-{
- if(_winSupportRWLockPrimitive)
- {
- _PReleaseSRWLockExclusive(&_lock);
- } else {
- _critSectPtr->Enter();
- _writerActive = false;
- if (_writersWaiting > 0)
- {
- _writeCondPtr->Wake();
-
- }else if (_readersWaiting > 0) {
- _readCondPtr->WakeAll();
- }
- _critSectPtr->Leave();
- }
-}
-
-void RWLockWindows::AcquireLockShared()
-{
- if(_winSupportRWLockPrimitive)
- {
- _PAcquireSRWLockShared(&_lock);
- } else
- {
- _critSectPtr->Enter();
- if (_writerActive || _writersWaiting > 0)
- {
- ++_readersWaiting;
-
- while (_writerActive || _writersWaiting > 0)
- {
- _readCondPtr->SleepCS(*_critSectPtr);
- }
- --_readersWaiting;
- }
- ++_readersActive;
- _critSectPtr->Leave();
- }
-}
-
-void RWLockWindows::ReleaseLockShared()
-{
- if(_winSupportRWLockPrimitive)
- {
- _PReleaseSRWLockShared(&_lock);
- } else
- {
- _critSectPtr->Enter();
-
- --_readersActive;
-
- if (_readersActive == 0 && _writersWaiting > 0)
- {
- _writeCondPtr->Wake();
- }
- _critSectPtr->Leave();
- }
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/rw_lock_win.h b/trunk/src/system_wrappers/source/rw_lock_win.h
deleted file mode 100644
index dc5355e..0000000
--- a/trunk/src/system_wrappers/source/rw_lock_win.h
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_WINDOWS_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_WINDOWS_H_
-
-#include "rw_lock_wrapper.h"
-
-#include <Windows.h>
-
-#if !defined(RTL_SRWLOCK_INIT)
- typedef struct _RTL_SRWLOCK
- {
- void* Ptr;
- } RTL_SRWLOCK, *PRTL_SRWLOCK;
- typedef RTL_SRWLOCK SRWLOCK, *PSRWLOCK;
-#endif
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class ConditionVariableWrapper;
-
-typedef void (WINAPI *PInitializeSRWLock)(PSRWLOCK);
-
-typedef void (WINAPI *PAcquireSRWLockExclusive)(PSRWLOCK);
-typedef void (WINAPI *PReleaseSRWLockExclusive)(PSRWLOCK);
-
-typedef void (WINAPI *PAcquireSRWLockShared)(PSRWLOCK);
-typedef void (WINAPI *PReleaseSRWLockShared)(PSRWLOCK);
-
-
-class RWLockWindows :public RWLockWrapper
-{
-public:
- RWLockWindows();
- virtual ~RWLockWindows();
-
- virtual void AcquireLockExclusive();
- virtual void ReleaseLockExclusive();
-
- virtual void AcquireLockShared();
- virtual void ReleaseLockShared();
-
-protected:
- virtual int Init();
-
-private:
- // For native implementation.
- static bool _winSupportRWLockPrimitive;
- SRWLOCK _lock;
-
- // Genric implementation, fallback if native is not supported.
- CriticalSectionWrapper* _critSectPtr;
- ConditionVariableWrapper* _readCondPtr;
- ConditionVariableWrapper* _writeCondPtr;
-
- int _readersActive;
- bool _writerActive;
- int _readersWaiting;
- int _writersWaiting;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_RW_LOCK_WINDOWS_H_
diff --git a/trunk/src/system_wrappers/source/set_thread_name_win.h b/trunk/src/system_wrappers/source/set_thread_name_win.h
deleted file mode 100644
index a46f4d6..0000000
--- a/trunk/src/system_wrappers/source/set_thread_name_win.h
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * Use of this source code is governed by the MICROSOFT LIMITED PUBLIC LICENSE
- * copyright license which can be found in the LICENSE file in the
- * third_party_mods/mslpl directory of the source tree or at
- * http://msdn.microsoft.com/en-us/cc300389.aspx#P.
- */
-/*
- * The original code can be found here:
- * http://msdn.microsoft.com/en-us/library/xcb2z8hs(VS.71).aspx
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_WINDOWS_SET_NAME_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_WINDOWS_SET_NAME_H_
-
-namespace webrtc {
-
-struct THREADNAME_INFO
-{
- DWORD dwType; // must be 0x1000
- LPCSTR szName; // pointer to name (in user addr space)
- DWORD dwThreadID; // thread ID (-1 = caller thread)
- DWORD dwFlags; // reserved for future use, must be zero
-};
-
-void SetThreadName(DWORD dwThreadID, LPCSTR szThreadName)
-{
- THREADNAME_INFO info;
- info.dwType = 0x1000;
- info.szName = szThreadName;
- info.dwThreadID = dwThreadID;
- info.dwFlags = 0;
-
- __try
- {
- RaiseException(0x406D1388, 0, sizeof(info) / sizeof(DWORD),
- (ULONG_PTR*)&info);
- }
- __except (EXCEPTION_CONTINUE_EXECUTION)
- {
- }
-}
-} // namespace webrtc
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_WINDOWS_SET_NAME_H_
diff --git a/trunk/src/system_wrappers/source/sort.cc b/trunk/src/system_wrappers/source/sort.cc
deleted file mode 100644
index 34aa437..0000000
--- a/trunk/src/system_wrappers/source/sort.cc
+++ /dev/null
@@ -1,549 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// When the platform supports STL, the functions are implemented using a
-// templated spreadsort algorithm (http://sourceforge.net/projects/spreadsort/),
-// part of the Boost C++ library collection. Otherwise, the C standard library's
-// qsort() will be used.
-
-#include "sort.h"
-
-#include <cassert>
-#include <cstring> // memcpy
-#include <new> // nothrow new
-
-#ifdef NO_STL
-#include <cstdlib> // qsort
-#else
-#include <algorithm> // std::sort
-#include <vector>
-#include "spreadsort.hpp" // TODO (ajm) upgrade to spreadsortv2.
-#endif
-
-#ifdef NO_STL
-#define COMPARE_DEREFERENCED(XT, YT) \
- do \
- { \
- if ((XT) > (YT)) \
- { \
- return 1; \
- } \
- else if ((XT) < (YT)) \
- { \
- return -1; \
- } \
- \
- return 0; \
- } \
- while(0)
-
-#define COMPARE_FOR_QSORT(X, Y, TYPE) \
- do \
- { \
- TYPE xT = static_cast<TYPE>(*static_cast<const TYPE*>(X)); \
- TYPE yT = static_cast<TYPE>(*static_cast<const TYPE*>(Y)); \
- COMPARE_DEREFERENCED(xT, yT); \
- } \
- while(0)
-
-#define COMPARE_KEY_FOR_QSORT(SORT_KEY_X, SORT_KEY_Y, TYPE) \
- do \
- { \
- TYPE xT = static_cast<TYPE>(*static_cast<TYPE*> \
- (static_cast<const SortKey*>(SORT_KEY_X)->key)); \
- TYPE yT = static_cast<TYPE>(*static_cast<TYPE*> \
- (static_cast<const SortKey*>(SORT_KEY_Y)->key)); \
- COMPARE_DEREFERENCED(xT, yT); \
- } \
- while(0)
-
-#define KEY_QSORT(SORT_KEY, KEY, NUM_OF_ELEMENTS, KEY_TYPE, COMPARE_FUNC) \
- do \
- { \
- KEY_TYPE* keyT = (KEY_TYPE*)(key); \
- for (WebRtc_UWord32 i = 0; i < (NUM_OF_ELEMENTS); i++) \
- { \
- ptrSortKey[i].key = &keyT[i]; \
- ptrSortKey[i].index = i; \
- } \
- \
- qsort((SORT_KEY), (NUM_OF_ELEMENTS), sizeof(SortKey), (COMPARE_FUNC));\
- } \
- while(0)
-#endif
-
-namespace webrtc
-{
-#ifdef NO_STL
- struct SortKey
- {
- void* key;
- WebRtc_UWord32 index;
- };
-#else
- template<typename KeyType>
- struct SortKey
- {
- KeyType key;
- WebRtc_UWord32 index;
- };
-#endif
-
- namespace // Unnamed namespace provides internal linkage.
- {
-#ifdef NO_STL
- int CompareWord8(const void* x, const void* y)
- {
- COMPARE_FOR_QSORT(x, y, WebRtc_Word8);
- }
-
- int CompareUWord8(const void* x, const void* y)
- {
- COMPARE_FOR_QSORT(x, y, WebRtc_UWord8);
- }
-
- int CompareWord16(const void* x, const void* y)
- {
- COMPARE_FOR_QSORT(x, y, WebRtc_Word16);
- }
-
- int CompareUWord16(const void* x, const void* y)
- {
- COMPARE_FOR_QSORT(x, y, WebRtc_UWord16);
- }
-
- int CompareWord32(const void* x, const void* y)
- {
- COMPARE_FOR_QSORT(x, y, WebRtc_Word32);
- }
-
- int CompareUWord32(const void* x, const void* y)
- {
- COMPARE_FOR_QSORT(x, y, WebRtc_UWord32);
- }
-
- int CompareWord64(const void* x, const void* y)
- {
- COMPARE_FOR_QSORT(x, y, WebRtc_Word64);
- }
-
- int CompareUWord64(const void* x, const void* y)
- {
- COMPARE_FOR_QSORT(x, y, WebRtc_UWord64);
- }
-
- int CompareFloat32(const void* x, const void* y)
- {
- COMPARE_FOR_QSORT(x, y, float);
- }
-
- int CompareFloat64(const void* x, const void* y)
- {
- COMPARE_FOR_QSORT(x, y, double);
- }
-
- int CompareKeyWord8(const void* sortKeyX, const void* sortKeyY)
- {
- COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_Word8);
- }
-
- int CompareKeyUWord8(const void* sortKeyX, const void* sortKeyY)
- {
- COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_UWord8);
- }
-
- int CompareKeyWord16(const void* sortKeyX, const void* sortKeyY)
- {
- COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_Word16);
- }
-
- int CompareKeyUWord16(const void* sortKeyX, const void* sortKeyY)
- {
- COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_UWord16);
- }
-
- int CompareKeyWord32(const void* sortKeyX, const void* sortKeyY)
- {
- COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_Word32);
- }
-
- int CompareKeyUWord32(const void* sortKeyX, const void* sortKeyY)
- {
- COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_UWord32);
- }
-
- int CompareKeyWord64(const void* sortKeyX, const void* sortKeyY)
- {
- COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_Word64);
- }
-
- int CompareKeyUWord64(const void* sortKeyX, const void* sortKeyY)
- {
- COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, WebRtc_UWord64);
- }
-
- int CompareKeyFloat32(const void* sortKeyX, const void* sortKeyY)
- {
- COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, float);
- }
-
- int CompareKeyFloat64(const void* sortKeyX, const void* sortKeyY)
- {
- COMPARE_KEY_FOR_QSORT(sortKeyX, sortKeyY, double);
- }
-#else
- template <typename KeyType>
- struct KeyLessThan
- {
- bool operator()(const SortKey<KeyType>& sortKeyX,
- const SortKey<KeyType>& sortKeyY) const
- {
- return sortKeyX.key < sortKeyY.key;
- }
- };
-
- template <typename KeyType>
- struct KeyRightShift
- {
- KeyType operator()(const SortKey<KeyType>& sortKey,
- const unsigned offset) const
- {
- return sortKey.key >> offset;
- }
- };
-
- template <typename DataType>
- inline void IntegerSort(void* data, WebRtc_UWord32 numOfElements)
- {
- DataType* dataT = static_cast<DataType*>(data);
- boost::integer_sort(dataT, dataT + numOfElements);
- }
-
- template <typename DataType, typename IntegerType>
- inline void FloatSort(void* data, WebRtc_UWord32 numOfElements)
- {
- DataType* dataT = static_cast<DataType*>(data);
- IntegerType cVal = 0;
- boost::float_sort_cast(dataT, dataT + numOfElements, cVal);
- }
-
- template <typename DataType>
- inline void StdSort(void* data, WebRtc_UWord32 numOfElements)
- {
- DataType* dataT = static_cast<DataType*>(data);
- std::sort(dataT, dataT + numOfElements);
- }
-
- template<typename KeyType>
- inline WebRtc_Word32 SetupKeySort(void* key,
- SortKey<KeyType>*& ptrSortKey,
- WebRtc_UWord32 numOfElements)
- {
- ptrSortKey = new(std::nothrow) SortKey<KeyType>[numOfElements];
- if (ptrSortKey == NULL)
- {
- return -1;
- }
-
- KeyType* keyT = static_cast<KeyType*>(key);
- for (WebRtc_UWord32 i = 0; i < numOfElements; i++)
- {
- ptrSortKey[i].key = keyT[i];
- ptrSortKey[i].index = i;
- }
-
- return 0;
- }
-
- template<typename KeyType>
- inline WebRtc_Word32 TeardownKeySort(void* data,
- SortKey<KeyType>* ptrSortKey,
- WebRtc_UWord32 numOfElements, WebRtc_UWord32 sizeOfElement)
- {
- WebRtc_UWord8* ptrData = static_cast<WebRtc_UWord8*>(data);
- WebRtc_UWord8* ptrDataSorted = new(std::nothrow) WebRtc_UWord8
- [numOfElements * sizeOfElement];
- if (ptrDataSorted == NULL)
- {
- return -1;
- }
-
- for (WebRtc_UWord32 i = 0; i < numOfElements; i++)
- {
- memcpy(ptrDataSorted + i * sizeOfElement, ptrData +
- ptrSortKey[i].index * sizeOfElement, sizeOfElement);
- }
- memcpy(ptrData, ptrDataSorted, numOfElements * sizeOfElement);
- delete[] ptrSortKey;
- delete[] ptrDataSorted;
- return 0;
- }
-
- template<typename KeyType>
- inline WebRtc_Word32 IntegerKeySort(void* data, void* key,
- WebRtc_UWord32 numOfElements,
- WebRtc_UWord32 sizeOfElement)
- {
- SortKey<KeyType>* ptrSortKey;
- if (SetupKeySort<KeyType>(key, ptrSortKey, numOfElements) != 0)
- {
- return -1;
- }
-
- boost::integer_sort(ptrSortKey, ptrSortKey + numOfElements,
- KeyRightShift<KeyType>(), KeyLessThan<KeyType>());
-
- if (TeardownKeySort<KeyType>(data, ptrSortKey, numOfElements,
- sizeOfElement) != 0)
- {
- return -1;
- }
-
- return 0;
- }
-
- template<typename KeyType>
- inline WebRtc_Word32 StdKeySort(void* data, void* key,
- WebRtc_UWord32 numOfElements,
- WebRtc_UWord32 sizeOfElement)
- {
- SortKey<KeyType>* ptrSortKey;
- if (SetupKeySort<KeyType>(key, ptrSortKey, numOfElements) != 0)
- {
- return -1;
- }
-
- std::sort(ptrSortKey, ptrSortKey + numOfElements,
- KeyLessThan<KeyType>());
-
- if (TeardownKeySort<KeyType>(data, ptrSortKey, numOfElements,
- sizeOfElement) != 0)
- {
- return -1;
- }
-
- return 0;
- }
-#endif
- }
-
- WebRtc_Word32 Sort(void* data, WebRtc_UWord32 numOfElements, Type type)
- {
- if (data == NULL)
- {
- return -1;
- }
-
-#ifdef NO_STL
- switch (type)
- {
- case TYPE_Word8:
- qsort(data, numOfElements, sizeof(WebRtc_Word8), CompareWord8);
- break;
- case TYPE_UWord8:
- qsort(data, numOfElements, sizeof(WebRtc_UWord8), CompareUWord8);
- break;
- case TYPE_Word16:
- qsort(data, numOfElements, sizeof(WebRtc_Word16), CompareWord16);
- break;
- case TYPE_UWord16:
- qsort(data, numOfElements, sizeof(WebRtc_UWord16), CompareUWord16);
- break;
- case TYPE_Word32:
- qsort(data, numOfElements, sizeof(WebRtc_Word32), CompareWord32);
- break;
- case TYPE_UWord32:
- qsort(data, numOfElements, sizeof(WebRtc_UWord32), CompareUWord32);
- break;
- case TYPE_Word64:
- qsort(data, numOfElements, sizeof(WebRtc_Word64), CompareWord64);
- break;
- case TYPE_UWord64:
- qsort(data, numOfElements, sizeof(WebRtc_UWord64), CompareUWord64);
- break;
- case TYPE_Float32:
- qsort(data, numOfElements, sizeof(float), CompareFloat32);
- break;
- case TYPE_Float64:
- qsort(data, numOfElements, sizeof(double), CompareFloat64);
- break;
- default:
- return -1;
- }
-#else
- // Fall back to std::sort for 64-bit types and floats due to compiler
- // warnings and VS 2003 build crashes respectively with spreadsort.
- switch (type)
- {
- case TYPE_Word8:
- IntegerSort<WebRtc_Word8>(data, numOfElements);
- break;
- case TYPE_UWord8:
- IntegerSort<WebRtc_UWord8>(data, numOfElements);
- break;
- case TYPE_Word16:
- IntegerSort<WebRtc_Word16>(data, numOfElements);
- break;
- case TYPE_UWord16:
- IntegerSort<WebRtc_UWord16>(data, numOfElements);
- break;
- case TYPE_Word32:
- IntegerSort<WebRtc_Word32>(data, numOfElements);
- break;
- case TYPE_UWord32:
- IntegerSort<WebRtc_UWord32>(data, numOfElements);
- break;
- case TYPE_Word64:
- StdSort<WebRtc_Word64>(data, numOfElements);
- break;
- case TYPE_UWord64:
- StdSort<WebRtc_UWord64>(data, numOfElements);
- break;
- case TYPE_Float32:
- StdSort<float>(data, numOfElements);
- break;
- case TYPE_Float64:
- StdSort<double>(data, numOfElements);
- break;
- }
-#endif
- return 0;
- }
-
- WebRtc_Word32 KeySort(void* data, void* key, WebRtc_UWord32 numOfElements,
- WebRtc_UWord32 sizeOfElement, Type keyType)
- {
- if (data == NULL)
- {
- return -1;
- }
-
- if (key == NULL)
- {
- return -1;
- }
-
- if ((WebRtc_UWord64)numOfElements * sizeOfElement > 0xffffffff)
- {
- return -1;
- }
-
-#ifdef NO_STL
- SortKey* ptrSortKey = new(std::nothrow) SortKey[numOfElements];
- if (ptrSortKey == NULL)
- {
- return -1;
- }
-
- switch (keyType)
- {
- case TYPE_Word8:
- KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_Word8,
- CompareKeyWord8);
- break;
- case TYPE_UWord8:
- KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_UWord8,
- CompareKeyUWord8);
- break;
- case TYPE_Word16:
- KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_Word16,
- CompareKeyWord16);
- break;
- case TYPE_UWord16:
- KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_UWord16,
- CompareKeyUWord16);
- break;
- case TYPE_Word32:
- KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_Word32,
- CompareKeyWord32);
- break;
- case TYPE_UWord32:
- KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_UWord32,
- CompareKeyUWord32);
- break;
- case TYPE_Word64:
- KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_Word64,
- CompareKeyWord64);
- break;
- case TYPE_UWord64:
- KEY_QSORT(ptrSortKey, key, numOfElements, WebRtc_UWord64,
- CompareKeyUWord64);
- break;
- case TYPE_Float32:
- KEY_QSORT(ptrSortKey, key, numOfElements, float,
- CompareKeyFloat32);
- break;
- case TYPE_Float64:
- KEY_QSORT(ptrSortKey, key, numOfElements, double,
- CompareKeyFloat64);
- break;
- default:
- return -1;
- }
-
- // Shuffle into sorted position based on index map.
- WebRtc_UWord8* ptrData = static_cast<WebRtc_UWord8*>(data);
- WebRtc_UWord8* ptrDataSorted = new(std::nothrow) WebRtc_UWord8
- [numOfElements * sizeOfElement];
- if (ptrDataSorted == NULL)
- {
- return -1;
- }
-
- for (WebRtc_UWord32 i = 0; i < numOfElements; i++)
- {
- memcpy(ptrDataSorted + i * sizeOfElement, ptrData +
- ptrSortKey[i].index * sizeOfElement, sizeOfElement);
- }
- memcpy(ptrData, ptrDataSorted, numOfElements * sizeOfElement);
-
- delete[] ptrSortKey;
- delete[] ptrDataSorted;
-
- return 0;
-#else
- // Fall back to std::sort for 64-bit types and floats due to compiler
- // warnings and errors respectively with spreadsort.
- switch (keyType)
- {
- case TYPE_Word8:
- return IntegerKeySort<WebRtc_Word8>(data, key, numOfElements,
- sizeOfElement);
- case TYPE_UWord8:
- return IntegerKeySort<WebRtc_UWord8>(data, key, numOfElements,
- sizeOfElement);
- case TYPE_Word16:
- return IntegerKeySort<WebRtc_Word16>(data, key, numOfElements,
- sizeOfElement);
- case TYPE_UWord16:
- return IntegerKeySort<WebRtc_UWord16>(data, key, numOfElements,
- sizeOfElement);
- case TYPE_Word32:
- return IntegerKeySort<WebRtc_Word32>(data, key, numOfElements,
- sizeOfElement);
- case TYPE_UWord32:
- return IntegerKeySort<WebRtc_UWord32>(data, key, numOfElements,
- sizeOfElement);
- case TYPE_Word64:
- return StdKeySort<WebRtc_Word64>(data, key, numOfElements,
- sizeOfElement);
- case TYPE_UWord64:
- return StdKeySort<WebRtc_UWord64>(data, key, numOfElements,
- sizeOfElement);
- case TYPE_Float32:
- return StdKeySort<float>(data, key, numOfElements, sizeOfElement);
- case TYPE_Float64:
- return StdKeySort<double>(data, key, numOfElements, sizeOfElement);
- }
- assert(false);
- return -1;
-#endif
- }
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/spreadsortlib/constants.hpp b/trunk/src/system_wrappers/source/spreadsortlib/constants.hpp
deleted file mode 100644
index fa81ece..0000000
--- a/trunk/src/system_wrappers/source/spreadsortlib/constants.hpp
+++ /dev/null
@@ -1,42 +0,0 @@
-/*Boost Software License - Version 1.0 - August 17th, 2003
-
-Permission is hereby granted, free of charge, to any person or organization
-obtaining a copy of the software and accompanying documentation covered by
-this license (the "Software") to use, reproduce, display, distribute,
-execute, and transmit the Software, and to prepare derivative works of the
-Software, and to permit third-parties to whom the Software is furnished to
-do so, all subject to the following:
-
-The copyright notices in the Software and this entire statement, including
-the above license grant, this restriction and the following disclaimer,
-must be included in all copies of the Software, in whole or in part, and
-all derivative works of the Software, unless such copies or derivative
-works are solely in the form of machine-executable object code generated by
-a source language processor.
-
-THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
-IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
-FITNESS FOR A PARTICULAR PURPOSE, TITLE AND NON-INFRINGEMENT. IN NO EVENT
-SHALL THE COPYRIGHT HOLDERS OR ANYONE DISTRIBUTING THE SOFTWARE BE LIABLE
-FOR ANY DAMAGES OR OTHER LIABILITY, WHETHER IN CONTRACT, TORT OR OTHERWISE,
-ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
-DEALINGS IN THE SOFTWARE.*/
-#ifndef BOOST_SPREADSORT_CONSTANTS
-#define BOOST_SPREADSORT_CONSTANTS
-namespace boost {
-namespace detail {
-//Tuning constants
-//Sets the minimum number of items per bin.
-static const unsigned LOG_MEAN_BIN_SIZE = 2;
-//This should be tuned to your processor cache; if you go too large you get cache misses on bins
-//The smaller this number, the less worst-case memory usage. If too small, too many recursions slow down spreadsort
-static const unsigned MAX_SPLITS = 10;
-//Used to force a comparison-based sorting for small bins, if it's faster. Minimum value 0
-static const unsigned LOG_MIN_SPLIT_COUNT = 5;
-//There is a minimum size below which it is not worth using spreadsort
-static const long MIN_SORT_SIZE = 1000;
-//This is the constant on the log base n of m calculation; make this larger the faster std::sort is relative to spreadsort
-static const unsigned LOG_CONST = 2;
-}
-}
-#endif
diff --git a/trunk/src/system_wrappers/source/spreadsortlib/spreadsort.hpp b/trunk/src/system_wrappers/source/spreadsortlib/spreadsort.hpp
deleted file mode 100644
index 2d1529a..0000000
--- a/trunk/src/system_wrappers/source/spreadsortlib/spreadsort.hpp
+++ /dev/null
@@ -1,1688 +0,0 @@
-//Templated spread_sort library
-
-// Copyright Steven J. Ross 2001 - 2009.
-// Distributed under the Boost Software License, Version 1.0.
-// (See accompanying file LICENSE_1_0.txt or copy at
-// http://www.boost.org/LICENSE_1_0.txt)
-
-// See http://www.boost.org/ for updates, documentation, and revision history.
-
-/*
-Some improvements suggested by:
-Phil Endecott and Frank Gennari
-Cygwin fix provided by:
-Scott McMurray
-*/
-
-#ifndef BOOST_SPREAD_SORT_H
-#define BOOST_SPREAD_SORT_H
-#include <algorithm>
-#include <vector>
-#include "constants.hpp"
-#include <cstring>
-
-namespace boost {
- namespace detail {
- //This only works on unsigned data types
- template <typename T>
- inline unsigned
- rough_log_2_size(const T& input)
- {
- unsigned result = 0;
- //The && is necessary on some compilers to avoid infinite loops; it doesn't significantly impair performance
- while((input >> result) && (result < (8*sizeof(T)))) ++result;
- return result;
- }
-
- //Gets the maximum size which we'll call spread_sort on to control worst-case performance
- //Maintains both a minimum size to recurse and a check of distribution size versus count
- //This is called for a set of bins, instead of bin-by-bin, to avoid performance overhead
- inline size_t
- get_max_count(unsigned log_range, size_t count)
- {
- unsigned divisor = rough_log_2_size(count);
- //Making sure the divisor is positive
- if(divisor > LOG_MEAN_BIN_SIZE)
- divisor -= LOG_MEAN_BIN_SIZE;
- else
- divisor = 1;
- unsigned relative_width = (LOG_CONST * log_range)/((divisor > MAX_SPLITS) ? MAX_SPLITS : divisor);
- //Don't try to bitshift more than the size of an element
- if((8*sizeof(size_t)) <= relative_width)
- relative_width = (8*sizeof(size_t)) - 1;
- return (size_t)1 << ((relative_width < (LOG_MEAN_BIN_SIZE + LOG_MIN_SPLIT_COUNT)) ?
- (LOG_MEAN_BIN_SIZE + LOG_MIN_SPLIT_COUNT) : relative_width);
- }
-
- //Find the minimum and maximum using <
- template <class RandomAccessIter>
- inline void
- find_extremes(RandomAccessIter current, RandomAccessIter last, RandomAccessIter & max, RandomAccessIter & min)
- {
- min = max = current;
- //Start from the second item, as max and min are initialized to the first
- while(++current < last) {
- if(*max < *current)
- max = current;
- else if(*current < *min)
- min = current;
- }
- }
-
- //Uses a user-defined comparison operator to find minimum and maximum
- template <class RandomAccessIter, class compare>
- inline void
- find_extremes(RandomAccessIter current, RandomAccessIter last, RandomAccessIter & max, RandomAccessIter & min, compare comp)
- {
- min = max = current;
- while(++current < last) {
- if(comp(*max, *current))
- max = current;
- else if(comp(*current, *min))
- min = current;
- }
- }
-
- //Gets a non-negative right bit shift to operate as a logarithmic divisor
- inline int
- get_log_divisor(size_t count, unsigned log_range)
- {
- int log_divisor;
- //If we can finish in one iteration without exceeding either (2 to the MAX_SPLITS) or n bins, do so
- if((log_divisor = log_range - rough_log_2_size(count)) <= 0 && log_range < MAX_SPLITS)
- log_divisor = 0;
- else {
- //otherwise divide the data into an optimized number of pieces
- log_divisor += LOG_MEAN_BIN_SIZE;
- if(log_divisor < 0)
- log_divisor = 0;
- //Cannot exceed MAX_SPLITS or cache misses slow down bin lookups dramatically
- if((log_range - log_divisor) > MAX_SPLITS)
- log_divisor = log_range - MAX_SPLITS;
- }
- return log_divisor;
- }
-
- template <class RandomAccessIter>
- inline RandomAccessIter *
- size_bins(std::vector<size_t> &bin_sizes, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset, unsigned &cache_end, unsigned bin_count)
- {
- //Assure space for the size of each bin, followed by initializing sizes
- if(bin_count > bin_sizes.size())
- bin_sizes.resize(bin_count);
- for(size_t u = 0; u < bin_count; u++)
- bin_sizes[u] = 0;
- //Make sure there is space for the bins
- cache_end = cache_offset + bin_count;
- if(cache_end > bin_cache.size())
- bin_cache.resize(cache_end);
- return &(bin_cache[cache_offset]);
- }
-
- //Implementation for recursive integer sorting
- template <class RandomAccessIter, class div_type, class data_type>
- inline void
- spread_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
- , std::vector<size_t> &bin_sizes)
- {
- //This step is roughly 10% of runtime, but it helps avoid worst-case behavior and improve behavior with real data
- //If you know the maximum and minimum ahead of time, you can pass those values in and skip this step for the first iteration
- RandomAccessIter max, min;
- find_extremes(first, last, max, min);
- //max and min will be the same (the first item) iff all values are equivalent
- if(max == min)
- return;
- RandomAccessIter * target_bin;
- unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(*max >> 0) - (*min >> 0)));
- div_type div_min = *min >> log_divisor;
- div_type div_max = *max >> log_divisor;
- unsigned bin_count = div_max - div_min + 1;
- unsigned cache_end;
- RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-
- //Calculating the size of each bin; this takes roughly 10% of runtime
- for (RandomAccessIter current = first; current != last;)
- bin_sizes[(*(current++) >> log_divisor) - div_min]++;
- //Assign the bin positions
- bins[0] = first;
- for(unsigned u = 0; u < bin_count - 1; u++)
- bins[u + 1] = bins[u] + bin_sizes[u];
-
- //Swap into place
- //This dominates runtime, mostly in the swap and bin lookups
- RandomAccessIter nextbinstart = first;
- for(unsigned u = 0; u < bin_count - 1; ++u) {
- RandomAccessIter * local_bin = bins + u;
- nextbinstart += bin_sizes[u];
- //Iterating over each element in this bin
- for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
- //Swapping elements in current into place until the correct element has been swapped in
- for(target_bin = (bins + ((*current >> log_divisor) - div_min)); target_bin != local_bin;
- target_bin = bins + ((*current >> log_divisor) - div_min)) {
- //3-way swap; this is about 1% faster than a 2-way swap with integers
- //The main advantage is less copies are involved per item put in the correct place
- data_type tmp;
- RandomAccessIter b = (*target_bin)++;
- RandomAccessIter * b_bin = bins + ((*b >> log_divisor) - div_min);
- if (b_bin != local_bin) {
- RandomAccessIter c = (*b_bin)++;
- tmp = *c;
- *c = *b;
- }
- else
- tmp = *b;
- *b = *current;
- *current = tmp;
- }
- }
- *local_bin = nextbinstart;
- }
- bins[bin_count - 1] = last;
-
- //If we've bucketsorted, the array is sorted and we should skip recursion
- if(!log_divisor)
- return;
-
- //Recursing; log_divisor is the remaining range
- size_t max_count = get_max_count(log_divisor, last - first);
- RandomAccessIter lastPos = first;
- for(unsigned u = cache_offset; u < cache_end; lastPos = bin_cache[u], ++u) {
- size_t count = bin_cache[u] - lastPos;
- //don't sort unless there are at least two items to compare
- if(count < 2)
- continue;
- //using std::sort if its worst-case is better
- if(count < max_count)
- std::sort(lastPos, bin_cache[u]);
- else
- spread_sort_rec<RandomAccessIter, div_type, data_type>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes);
- }
- }
-
- //Generic bitshift-based 3-way swapping code
- template <class RandomAccessIter, class div_type, class data_type, class right_shift>
- inline void inner_swap_loop(RandomAccessIter * bins, const RandomAccessIter & nextbinstart, unsigned ii, right_shift &shift
- , const unsigned log_divisor, const div_type div_min)
- {
- RandomAccessIter * local_bin = bins + ii;
- for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
- for(RandomAccessIter * target_bin = (bins + (shift(*current, log_divisor) - div_min)); target_bin != local_bin;
- target_bin = bins + (shift(*current, log_divisor) - div_min)) {
- data_type tmp;
- RandomAccessIter b = (*target_bin)++;
- RandomAccessIter * b_bin = bins + (shift(*b, log_divisor) - div_min);
- //Three-way swap; if the item to be swapped doesn't belong in the current bin, swap it to where it belongs
- if (b_bin != local_bin) {
- RandomAccessIter c = (*b_bin)++;
- tmp = *c;
- *c = *b;
- }
- //Note: we could increment current once the swap is done in this case, but that seems to impair performance
- else
- tmp = *b;
- *b = *current;
- *current = tmp;
- }
- }
- *local_bin = nextbinstart;
- }
-
- //Standard swapping wrapper for ascending values
- template <class RandomAccessIter, class div_type, class data_type, class right_shift>
- inline void swap_loop(RandomAccessIter * bins, RandomAccessIter & nextbinstart, unsigned ii, right_shift &shift
- , const std::vector<size_t> &bin_sizes, const unsigned log_divisor, const div_type div_min)
- {
- nextbinstart += bin_sizes[ii];
- inner_swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, ii, shift, log_divisor, div_min);
- }
-
- //Functor implementation for recursive sorting
- template <class RandomAccessIter, class div_type, class data_type, class right_shift, class compare>
- inline void
- spread_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
- , std::vector<size_t> &bin_sizes, right_shift shift, compare comp)
- {
- RandomAccessIter max, min;
- find_extremes(first, last, max, min, comp);
- if(max == min)
- return;
- unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(shift(*max, 0)) - (shift(*min, 0))));
- div_type div_min = shift(*min, log_divisor);
- div_type div_max = shift(*max, log_divisor);
- unsigned bin_count = div_max - div_min + 1;
- unsigned cache_end;
- RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-
- //Calculating the size of each bin
- for (RandomAccessIter current = first; current != last;)
- bin_sizes[shift(*(current++), log_divisor) - div_min]++;
- bins[0] = first;
- for(unsigned u = 0; u < bin_count - 1; u++)
- bins[u + 1] = bins[u] + bin_sizes[u];
-
- //Swap into place
- RandomAccessIter nextbinstart = first;
- for(unsigned u = 0; u < bin_count - 1; ++u)
- swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, u, shift, bin_sizes, log_divisor, div_min);
- bins[bin_count - 1] = last;
-
- //If we've bucketsorted, the array is sorted and we should skip recursion
- if(!log_divisor)
- return;
-
- //Recursing
- size_t max_count = get_max_count(log_divisor, last - first);
- RandomAccessIter lastPos = first;
- for(unsigned u = cache_offset; u < cache_end; lastPos = bin_cache[u], ++u) {
- size_t count = bin_cache[u] - lastPos;
- if(count < 2)
- continue;
- if(count < max_count)
- std::sort(lastPos, bin_cache[u], comp);
- else
- spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift, compare>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes, shift, comp);
- }
- }
-
- //Functor implementation for recursive sorting with only Shift overridden
- template <class RandomAccessIter, class div_type, class data_type, class right_shift>
- inline void
- spread_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
- , std::vector<size_t> &bin_sizes, right_shift shift)
- {
- RandomAccessIter max, min;
- find_extremes(first, last, max, min);
- if(max == min)
- return;
- unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(shift(*max, 0)) - (shift(*min, 0))));
- div_type div_min = shift(*min, log_divisor);
- div_type div_max = shift(*max, log_divisor);
- unsigned bin_count = div_max - div_min + 1;
- unsigned cache_end;
- RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-
- //Calculating the size of each bin
- for (RandomAccessIter current = first; current != last;)
- bin_sizes[shift(*(current++), log_divisor) - div_min]++;
- bins[0] = first;
- for(unsigned u = 0; u < bin_count - 1; u++)
- bins[u + 1] = bins[u] + bin_sizes[u];
-
- //Swap into place
- RandomAccessIter nextbinstart = first;
- for(unsigned ii = 0; ii < bin_count - 1; ++ii)
- swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, ii, shift, bin_sizes, log_divisor, div_min);
- bins[bin_count - 1] = last;
-
- //If we've bucketsorted, the array is sorted and we should skip recursion
- if(!log_divisor)
- return;
-
- //Recursing
- size_t max_count = get_max_count(log_divisor, last - first);
- RandomAccessIter lastPos = first;
- for(unsigned u = cache_offset; u < cache_end; lastPos = bin_cache[u], ++u) {
- size_t count = bin_cache[u] - lastPos;
- if(count < 2)
- continue;
- if(count < max_count)
- std::sort(lastPos, bin_cache[u]);
- else
- spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes, shift);
- }
- }
-
- //Holds the bin vector and makes the initial recursive call
- template <class RandomAccessIter, class div_type, class data_type>
- inline void
- spread_sort(RandomAccessIter first, RandomAccessIter last, div_type, data_type)
- {
- std::vector<size_t> bin_sizes;
- std::vector<RandomAccessIter> bin_cache;
- spread_sort_rec<RandomAccessIter, div_type, data_type>(first, last, bin_cache, 0, bin_sizes);
- }
-
- template <class RandomAccessIter, class div_type, class data_type, class right_shift, class compare>
- inline void
- spread_sort(RandomAccessIter first, RandomAccessIter last, div_type, data_type, right_shift shift, compare comp)
- {
- std::vector<size_t> bin_sizes;
- std::vector<RandomAccessIter> bin_cache;
- spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift, compare>(first, last, bin_cache, 0, bin_sizes, shift, comp);
- }
-
- template <class RandomAccessIter, class div_type, class data_type, class right_shift>
- inline void
- spread_sort(RandomAccessIter first, RandomAccessIter last, div_type, data_type, right_shift shift)
- {
- std::vector<size_t> bin_sizes;
- std::vector<RandomAccessIter> bin_cache;
- spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(first, last, bin_cache, 0, bin_sizes, shift);
- }
- }
-
- //Top-level sorting call for integers
- template <class RandomAccessIter>
- inline void integer_sort(RandomAccessIter first, RandomAccessIter last)
- {
- //Don't sort if it's too small to optimize
- if(last - first < detail::MIN_SORT_SIZE)
- std::sort(first, last);
- else
- detail::spread_sort(first, last, *first >> 0, *first);
- }
-
- //integer_sort with functors
- template <class RandomAccessIter, class right_shift, class compare>
- inline void integer_sort(RandomAccessIter first, RandomAccessIter last,
- right_shift shift, compare comp) {
- if(last - first < detail::MIN_SORT_SIZE)
- std::sort(first, last, comp);
- else
- detail::spread_sort(first, last, shift(*first, 0), *first, shift, comp);
- }
-
- //integer_sort with right_shift functor
- template <class RandomAccessIter, class right_shift>
- inline void integer_sort(RandomAccessIter first, RandomAccessIter last,
- right_shift shift) {
- if(last - first < detail::MIN_SORT_SIZE)
- std::sort(first, last);
- else
- detail::spread_sort(first, last, shift(*first, 0), *first, shift);
- }
-
- //------------------------------------------------------ float_sort source --------------------------------------
- //Casts a RandomAccessIter to the specified data type
- template<class cast_type, class RandomAccessIter>
- inline cast_type
- cast_float_iter(const RandomAccessIter & floatiter)
- {
- cast_type result;
- std::memcpy(&result, &(*floatiter), sizeof(cast_type));
- return result;
- }
-
- //Casts a data element to the specified datinner_float_a type
- template<class data_type, class cast_type>
- inline cast_type
- mem_cast(const data_type & data)
- {
- cast_type result;
- std::memcpy(&result, &data, sizeof(cast_type));
- return result;
- }
-
- namespace detail {
- template <class RandomAccessIter, class div_type, class right_shift>
- inline void
- find_extremes(RandomAccessIter current, RandomAccessIter last, div_type & max, div_type & min, right_shift shift)
- {
- min = max = shift(*current, 0);
- while(++current < last) {
- div_type value = shift(*current, 0);
- if(max < value)
- max = value;
- else if(value < min)
- min = value;
- }
- }
-
- //Specialized swap loops for floating-point casting
- template <class RandomAccessIter, class div_type, class data_type>
- inline void inner_float_swap_loop(RandomAccessIter * bins, const RandomAccessIter & nextbinstart, unsigned ii
- , const unsigned log_divisor, const div_type div_min)
- {
- RandomAccessIter * local_bin = bins + ii;
- for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
- for(RandomAccessIter * target_bin = (bins + ((cast_float_iter<div_type, RandomAccessIter>(current) >> log_divisor) - div_min)); target_bin != local_bin;
- target_bin = bins + ((cast_float_iter<div_type, RandomAccessIter>(current) >> log_divisor) - div_min)) {
- data_type tmp;
- RandomAccessIter b = (*target_bin)++;
- RandomAccessIter * b_bin = bins + ((cast_float_iter<div_type, RandomAccessIter>(b) >> log_divisor) - div_min);
- //Three-way swap; if the item to be swapped doesn't belong in the current bin, swap it to where it belongs
- if (b_bin != local_bin) {
- RandomAccessIter c = (*b_bin)++;
- tmp = *c;
- *c = *b;
- }
- else
- tmp = *b;
- *b = *current;
- *current = tmp;
- }
- }
- *local_bin = nextbinstart;
- }
-
- template <class RandomAccessIter, class div_type, class data_type>
- inline void float_swap_loop(RandomAccessIter * bins, RandomAccessIter & nextbinstart, unsigned ii
- , const std::vector<size_t> &bin_sizes, const unsigned log_divisor, const div_type div_min)
- {
- nextbinstart += bin_sizes[ii];
- inner_float_swap_loop<RandomAccessIter, div_type, data_type>(bins, nextbinstart, ii, log_divisor, div_min);
- }
-
- template <class RandomAccessIter, class cast_type>
- inline void
- find_extremes(RandomAccessIter current, RandomAccessIter last, cast_type & max, cast_type & min)
- {
- min = max = cast_float_iter<cast_type, RandomAccessIter>(current);
- while(++current < last) {
- cast_type value = cast_float_iter<cast_type, RandomAccessIter>(current);
- if(max < value)
- max = value;
- else if(value < min)
- min = value;
- }
- }
-
- //Special-case sorting of positive floats with casting instead of a right_shift
- template <class RandomAccessIter, class div_type, class data_type>
- inline void
- positive_float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
- , std::vector<size_t> &bin_sizes)
- {
- div_type max, min;
- find_extremes(first, last, max, min);
- if(max == min)
- return;
- unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
- div_type div_min = min >> log_divisor;
- div_type div_max = max >> log_divisor;
- unsigned bin_count = div_max - div_min + 1;
- unsigned cache_end;
- RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-
- //Calculating the size of each bin
- for (RandomAccessIter current = first; current != last;)
- bin_sizes[(cast_float_iter<div_type, RandomAccessIter>(current++) >> log_divisor) - div_min]++;
- bins[0] = first;
- for(unsigned u = 0; u < bin_count - 1; u++)
- bins[u + 1] = bins[u] + bin_sizes[u];
-
- //Swap into place
- RandomAccessIter nextbinstart = first;
- for(unsigned u = 0; u < bin_count - 1; ++u)
- float_swap_loop<RandomAccessIter, div_type, data_type>(bins, nextbinstart, u, bin_sizes, log_divisor, div_min);
- bins[bin_count - 1] = last;
-
- //Return if we've completed bucketsorting
- if(!log_divisor)
- return;
-
- //Recursing
- size_t max_count = get_max_count(log_divisor, last - first);
- RandomAccessIter lastPos = first;
- for(unsigned u = cache_offset; u < cache_end; lastPos = bin_cache[u], ++u) {
- size_t count = bin_cache[u] - lastPos;
- if(count < 2)
- continue;
- if(count < max_count)
- std::sort(lastPos, bin_cache[u]);
- else
- positive_float_sort_rec<RandomAccessIter, div_type, data_type>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes);
- }
- }
-
- //Sorting negative_ float_s
- //Note that bins are iterated in reverse order because max_neg_float = min_neg_int
- template <class RandomAccessIter, class div_type, class data_type>
- inline void
- negative_float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
- , std::vector<size_t> &bin_sizes)
- {
- div_type max, min;
- find_extremes(first, last, max, min);
- if(max == min)
- return;
- unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
- div_type div_min = min >> log_divisor;
- div_type div_max = max >> log_divisor;
- unsigned bin_count = div_max - div_min + 1;
- unsigned cache_end;
- RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-
- //Calculating the size of each bin
- for (RandomAccessIter current = first; current != last;)
- bin_sizes[(cast_float_iter<div_type, RandomAccessIter>(current++) >> log_divisor) - div_min]++;
- bins[bin_count - 1] = first;
- for(int ii = bin_count - 2; ii >= 0; --ii)
- bins[ii] = bins[ii + 1] + bin_sizes[ii + 1];
-
- //Swap into place
- RandomAccessIter nextbinstart = first;
- //The last bin will always have the correct elements in it
- for(int ii = bin_count - 1; ii > 0; --ii)
- float_swap_loop<RandomAccessIter, div_type, data_type>(bins, nextbinstart, ii, bin_sizes, log_divisor, div_min);
- //Since we don't process the last bin, we need to update its end position
- bin_cache[cache_offset] = last;
-
- //Return if we've completed bucketsorting
- if(!log_divisor)
- return;
-
- //Recursing
- size_t max_count = get_max_count(log_divisor, last - first);
- RandomAccessIter lastPos = first;
- for(int ii = cache_end - 1; ii >= (int)cache_offset; lastPos = bin_cache[ii], --ii) {
- size_t count = bin_cache[ii] - lastPos;
- if(count < 2)
- continue;
- if(count < max_count)
- std::sort(lastPos, bin_cache[ii]);
- else
- negative_float_sort_rec<RandomAccessIter, div_type, data_type>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes);
- }
- }
-
- //Sorting negative_ float_s
- //Note that bins are iterated in reverse order because max_neg_float = min_neg_int
- template <class RandomAccessIter, class div_type, class data_type, class right_shift>
- inline void
- negative_float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
- , std::vector<size_t> &bin_sizes, right_shift shift)
- {
- div_type max, min;
- find_extremes(first, last, max, min, shift);
- if(max == min)
- return;
- unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
- div_type div_min = min >> log_divisor;
- div_type div_max = max >> log_divisor;
- unsigned bin_count = div_max - div_min + 1;
- unsigned cache_end;
- RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-
- //Calculating the size of each bin
- for (RandomAccessIter current = first; current != last;)
- bin_sizes[shift(*(current++), log_divisor) - div_min]++;
- bins[bin_count - 1] = first;
- for(int ii = bin_count - 2; ii >= 0; --ii)
- bins[ii] = bins[ii + 1] + bin_sizes[ii + 1];
-
- //Swap into place
- RandomAccessIter nextbinstart = first;
- //The last bin will always have the correct elements in it
- for(int ii = bin_count - 1; ii > 0; --ii)
- swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, ii, shift, bin_sizes, log_divisor, div_min);
- //Since we don't process the last bin, we need to update its end position
- bin_cache[cache_offset] = last;
-
- //Return if we've completed bucketsorting
- if(!log_divisor)
- return;
-
- //Recursing
- size_t max_count = get_max_count(log_divisor, last - first);
- RandomAccessIter lastPos = first;
- for(int ii = cache_end - 1; ii >= (int)cache_offset; lastPos = bin_cache[ii], --ii) {
- size_t count = bin_cache[ii] - lastPos;
- if(count < 2)
- continue;
- if(count < max_count)
- std::sort(lastPos, bin_cache[ii]);
- else
- negative_float_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes, shift);
- }
- }
-
- template <class RandomAccessIter, class div_type, class data_type, class right_shift, class compare>
- inline void
- negative_float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
- , std::vector<size_t> &bin_sizes, right_shift shift, compare comp)
- {
- div_type max, min;
- find_extremes(first, last, max, min, shift);
- if(max == min)
- return;
- unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
- div_type div_min = min >> log_divisor;
- div_type div_max = max >> log_divisor;
- unsigned bin_count = div_max - div_min + 1;
- unsigned cache_end;
- RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-
- //Calculating the size of each bin
- for (RandomAccessIter current = first; current != last;)
- bin_sizes[shift(*(current++), log_divisor) - div_min]++;
- bins[bin_count - 1] = first;
- for(int ii = bin_count - 2; ii >= 0; --ii)
- bins[ii] = bins[ii + 1] + bin_sizes[ii + 1];
-
- //Swap into place
- RandomAccessIter nextbinstart = first;
- //The last bin will always have the correct elements in it
- for(int ii = bin_count - 1; ii > 0; --ii)
- swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, ii, shift, bin_sizes, log_divisor, div_min);
- //Since we don't process the last bin, we need to update its end position
- bin_cache[cache_offset] = last;
-
- //Return if we've completed bucketsorting
- if(!log_divisor)
- return;
-
- //Recursing
- size_t max_count = get_max_count(log_divisor, last - first);
- RandomAccessIter lastPos = first;
- for(int ii = cache_end - 1; ii >= (int)cache_offset; lastPos = bin_cache[ii], --ii) {
- size_t count = bin_cache[ii] - lastPos;
- if(count < 2)
- continue;
- if(count < max_count)
- std::sort(lastPos, bin_cache[ii], comp);
- else
- negative_float_sort_rec<RandomAccessIter, div_type, data_type, right_shift, compare>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes, shift, comp);
- }
- }
-
- //Casting special-case for floating-point sorting
- template <class RandomAccessIter, class div_type, class data_type>
- inline void
- float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
- , std::vector<size_t> &bin_sizes)
- {
- div_type max, min;
- find_extremes(first, last, max, min);
- if(max == min)
- return;
- unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
- div_type div_min = min >> log_divisor;
- div_type div_max = max >> log_divisor;
- unsigned bin_count = div_max - div_min + 1;
- unsigned cache_end;
- RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-
- //Calculating the size of each bin
- for (RandomAccessIter current = first; current != last;)
- bin_sizes[(cast_float_iter<div_type, RandomAccessIter>(current++) >> log_divisor) - div_min]++;
- //The index of the first positive bin
- div_type first_positive = (div_min < 0) ? -div_min : 0;
- //Resetting if all bins are negative
- if(cache_offset + first_positive > cache_end)
- first_positive = cache_end - cache_offset;
- //Reversing the order of the negative bins
- //Note that because of the negative/positive ordering direction flip
- //We can not depend upon bin order and positions matching up
- //so bin_sizes must be reused to contain the end of the bin
- if(first_positive > 0) {
- bins[first_positive - 1] = first;
- for(int ii = first_positive - 2; ii >= 0; --ii) {
- bins[ii] = first + bin_sizes[ii + 1];
- bin_sizes[ii] += bin_sizes[ii + 1];
- }
- //Handling positives following negatives
- if((unsigned)first_positive < bin_count) {
- bins[first_positive] = first + bin_sizes[0];
- bin_sizes[first_positive] += bin_sizes[0];
- }
- }
- else
- bins[0] = first;
- for(unsigned u = first_positive; u < bin_count - 1; u++) {
- bins[u + 1] = first + bin_sizes[u];
- bin_sizes[u + 1] += bin_sizes[u];
- }
-
- //Swap into place
- RandomAccessIter nextbinstart = first;
- for(unsigned u = 0; u < bin_count; ++u) {
- nextbinstart = first + bin_sizes[u];
- inner_float_swap_loop<RandomAccessIter, div_type, data_type>(bins, nextbinstart, u, log_divisor, div_min);
- }
-
- if(!log_divisor)
- return;
-
- //Handling negative values first
- size_t max_count = get_max_count(log_divisor, last - first);
- RandomAccessIter lastPos = first;
- for(int ii = cache_offset + first_positive - 1; ii >= (int)cache_offset ; lastPos = bin_cache[ii--]) {
- size_t count = bin_cache[ii] - lastPos;
- if(count < 2)
- continue;
- if(count < max_count)
- std::sort(lastPos, bin_cache[ii]);
- //sort negative values using reversed-bin spread_sort
- else
- negative_float_sort_rec<RandomAccessIter, div_type, data_type>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes);
- }
-
- for(unsigned u = cache_offset + first_positive; u < cache_end; lastPos = bin_cache[u], ++u) {
- size_t count = bin_cache[u] - lastPos;
- if(count < 2)
- continue;
- if(count < max_count)
- std::sort(lastPos, bin_cache[u]);
- //sort positive values using normal spread_sort
- else
- positive_float_sort_rec<RandomAccessIter, div_type, data_type>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes);
- }
- }
-
- //Functor implementation for recursive sorting
- template <class RandomAccessIter, class div_type, class data_type, class right_shift>
- inline void
- float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
- , std::vector<size_t> &bin_sizes, right_shift shift)
- {
- div_type max, min;
- find_extremes(first, last, max, min, shift);
- if(max == min)
- return;
- unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
- div_type div_min = min >> log_divisor;
- div_type div_max = max >> log_divisor;
- unsigned bin_count = div_max - div_min + 1;
- unsigned cache_end;
- RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-
- //Calculating the size of each bin
- for (RandomAccessIter current = first; current != last;)
- bin_sizes[shift(*(current++), log_divisor) - div_min]++;
- //The index of the first positive bin
- div_type first_positive = (div_min < 0) ? -div_min : 0;
- //Resetting if all bins are negative
- if(cache_offset + first_positive > cache_end)
- first_positive = cache_end - cache_offset;
- //Reversing the order of the negative bins
- //Note that because of the negative/positive ordering direction flip
- //We can not depend upon bin order and positions matching up
- //so bin_sizes must be reused to contain the end of the bin
- if(first_positive > 0) {
- bins[first_positive - 1] = first;
- for(int ii = first_positive - 2; ii >= 0; --ii) {
- bins[ii] = first + bin_sizes[ii + 1];
- bin_sizes[ii] += bin_sizes[ii + 1];
- }
- //Handling positives following negatives
- if((unsigned)first_positive < bin_count) {
- bins[first_positive] = first + bin_sizes[0];
- bin_sizes[first_positive] += bin_sizes[0];
- }
- }
- else
- bins[0] = first;
- for(unsigned u = first_positive; u < bin_count - 1; u++) {
- bins[u + 1] = first + bin_sizes[u];
- bin_sizes[u + 1] += bin_sizes[u];
- }
-
- //Swap into place
- RandomAccessIter nextbinstart = first;
- for(unsigned u = 0; u < bin_count; ++u) {
- nextbinstart = first + bin_sizes[u];
- inner_swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, u, shift, log_divisor, div_min);
- }
-
- //Return if we've completed bucketsorting
- if(!log_divisor)
- return;
-
- //Handling negative values first
- size_t max_count = get_max_count(log_divisor, last - first);
- RandomAccessIter lastPos = first;
- for(int ii = cache_offset + first_positive - 1; ii >= (int)cache_offset ; lastPos = bin_cache[ii--]) {
- size_t count = bin_cache[ii] - lastPos;
- if(count < 2)
- continue;
- if(count < max_count)
- std::sort(lastPos, bin_cache[ii]);
- //sort negative values using reversed-bin spread_sort
- else
- negative_float_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes, shift);
- }
-
- for(unsigned u = cache_offset + first_positive; u < cache_end; lastPos = bin_cache[u], ++u) {
- size_t count = bin_cache[u] - lastPos;
- if(count < 2)
- continue;
- if(count < max_count)
- std::sort(lastPos, bin_cache[u]);
- //sort positive values using normal spread_sort
- else
- spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes, shift);
- }
- }
-
- template <class RandomAccessIter, class div_type, class data_type, class right_shift, class compare>
- inline void
- float_sort_rec(RandomAccessIter first, RandomAccessIter last, std::vector<RandomAccessIter> &bin_cache, unsigned cache_offset
- , std::vector<size_t> &bin_sizes, right_shift shift, compare comp)
- {
- div_type max, min;
- find_extremes(first, last, max, min, shift);
- if(max == min)
- return;
- unsigned log_divisor = get_log_divisor(last - first, rough_log_2_size((size_t)(max) - min));
- div_type div_min = min >> log_divisor;
- div_type div_max = max >> log_divisor;
- unsigned bin_count = div_max - div_min + 1;
- unsigned cache_end;
- RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, bin_count);
-
- //Calculating the size of each bin
- for (RandomAccessIter current = first; current != last;)
- bin_sizes[shift(*(current++), log_divisor) - div_min]++;
- //The index of the first positive bin
- div_type first_positive = (div_min < 0) ? -div_min : 0;
- //Resetting if all bins are negative
- if(cache_offset + first_positive > cache_end)
- first_positive = cache_end - cache_offset;
- //Reversing the order of the negative bins
- //Note that because of the negative/positive ordering direction flip
- //We can not depend upon bin order and positions matching up
- //so bin_sizes must be reused to contain the end of the bin
- if(first_positive > 0) {
- bins[first_positive - 1] = first;
- for(int ii = first_positive - 2; ii >= 0; --ii) {
- bins[ii] = first + bin_sizes[ii + 1];
- bin_sizes[ii] += bin_sizes[ii + 1];
- }
- //Handling positives following negatives
- if((unsigned)first_positive < bin_count) {
- bins[first_positive] = first + bin_sizes[0];
- bin_sizes[first_positive] += bin_sizes[0];
- }
- }
- else
- bins[0] = first;
- for(unsigned u = first_positive; u < bin_count - 1; u++) {
- bins[u + 1] = first + bin_sizes[u];
- bin_sizes[u + 1] += bin_sizes[u];
- }
-
- //Swap into place
- RandomAccessIter nextbinstart = first;
- for(unsigned u = 0; u < bin_count; ++u) {
- nextbinstart = first + bin_sizes[u];
- inner_swap_loop<RandomAccessIter, div_type, data_type, right_shift>(bins, nextbinstart, u, shift, log_divisor, div_min);
- }
-
- //Return if we've completed bucketsorting
- if(!log_divisor)
- return;
-
- //Handling negative values first
- size_t max_count = get_max_count(log_divisor, last - first);
- RandomAccessIter lastPos = first;
- for(int ii = cache_offset + first_positive - 1; ii >= (int)cache_offset ; lastPos = bin_cache[ii--]) {
- size_t count = bin_cache[ii] - lastPos;
- if(count < 2)
- continue;
- if(count < max_count)
- std::sort(lastPos, bin_cache[ii]);
- //sort negative values using reversed-bin spread_sort
- else
- negative_float_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[ii], bin_cache, cache_end, bin_sizes, shift, comp);
- }
-
- for(unsigned u = cache_offset + first_positive; u < cache_end; lastPos = bin_cache[u], ++u) {
- size_t count = bin_cache[u] - lastPos;
- if(count < 2)
- continue;
- if(count < max_count)
- std::sort(lastPos, bin_cache[u]);
- //sort positive values using normal spread_sort
- else
- spread_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(lastPos, bin_cache[u], bin_cache, cache_end, bin_sizes, shift, comp);
- }
- }
-
- template <class RandomAccessIter, class cast_type, class data_type>
- inline void
- float_Sort(RandomAccessIter first, RandomAccessIter last, cast_type, data_type)
- {
- std::vector<size_t> bin_sizes;
- std::vector<RandomAccessIter> bin_cache;
- float_sort_rec<RandomAccessIter, cast_type, data_type>(first, last, bin_cache, 0, bin_sizes);
- }
-
- template <class RandomAccessIter, class div_type, class data_type, class right_shift>
- inline void
- float_Sort(RandomAccessIter first, RandomAccessIter last, div_type, data_type, right_shift shift)
- {
- std::vector<size_t> bin_sizes;
- std::vector<RandomAccessIter> bin_cache;
- float_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(first, last, bin_cache, 0, bin_sizes, shift);
- }
-
- template <class RandomAccessIter, class div_type, class data_type, class right_shift, class compare>
- inline void
- float_Sort(RandomAccessIter first, RandomAccessIter last, div_type, data_type, right_shift shift, compare comp)
- {
- std::vector<size_t> bin_sizes;
- std::vector<RandomAccessIter> bin_cache;
- float_sort_rec<RandomAccessIter, div_type, data_type, right_shift>(first, last, bin_cache, 0, bin_sizes, shift, comp);
- }
- }
-
- //float_sort with casting
- //The cast_type must be equal in size to the data type, and must be a signed integer
- template <class RandomAccessIter, class cast_type>
- inline void float_sort_cast(RandomAccessIter first, RandomAccessIter last, cast_type cVal)
- {
- if(last - first < detail::MIN_SORT_SIZE)
- std::sort(first, last);
- else
- detail::float_Sort(first, last, cVal, *first);
- }
-
- //float_sort with casting to an int
- //Only use this with IEEE floating-point numbers
- template <class RandomAccessIter>
- inline void float_sort_cast_to_int(RandomAccessIter first, RandomAccessIter last)
- {
- int cVal = 0;
- float_sort_cast(first, last, cVal);
- }
-
- //float_sort with functors
- template <class RandomAccessIter, class right_shift>
- inline void float_sort(RandomAccessIter first, RandomAccessIter last, right_shift shift)
- {
- if(last - first < detail::MIN_SORT_SIZE)
- std::sort(first, last);
- else
- detail::float_Sort(first, last, shift(*first, 0), *first, shift);
- }
-
- template <class RandomAccessIter, class right_shift, class compare>
- inline void float_sort(RandomAccessIter first, RandomAccessIter last, right_shift shift, compare comp)
- {
- if(last - first < detail::MIN_SORT_SIZE)
- std::sort(first, last, comp);
- else
- detail::float_Sort(first, last, shift(*first, 0), *first, shift, comp);
- }
-
- //------------------------------------------------- string_sort source ---------------------------------------------
- namespace detail {
- //Offsetting on identical characters. This function works a character at a time for optimal worst-case performance.
- template<class RandomAccessIter>
- inline void
- update_offset(RandomAccessIter first, RandomAccessIter finish, unsigned &char_offset)
- {
- unsigned nextOffset = char_offset;
- bool done = false;
- while(!done) {
- RandomAccessIter curr = first;
- do {
- //ignore empties, but if the nextOffset would exceed the length or not match, exit; we've found the last matching character
- if((*curr).size() > char_offset && ((*curr).size() <= (nextOffset + 1) || (*curr)[nextOffset] != (*first)[nextOffset])) {
- done = true;
- break;
- }
- } while(++curr != finish);
- if(!done)
- ++nextOffset;
- }
- char_offset = nextOffset;
- }
-
- //Offsetting on identical characters. This function works a character at a time for optimal worst-case performance.
- template<class RandomAccessIter, class get_char, class get_length>
- inline void
- update_offset(RandomAccessIter first, RandomAccessIter finish, unsigned &char_offset, get_char getchar, get_length length)
- {
- unsigned nextOffset = char_offset;
- bool done = false;
- while(!done) {
- RandomAccessIter curr = first;
- do {
- //ignore empties, but if the nextOffset would exceed the length or not match, exit; we've found the last matching character
- if(length(*curr) > char_offset && (length(*curr) <= (nextOffset + 1) || getchar((*curr), nextOffset) != getchar((*first), nextOffset))) {
- done = true;
- break;
- }
- } while(++curr != finish);
- if(!done)
- ++nextOffset;
- }
- char_offset = nextOffset;
- }
-
- //A comparison functor for strings that assumes they are identical up to char_offset
- template<class data_type, class unsignedchar_type>
- struct offset_lessthan {
- offset_lessthan(unsigned char_offset) : fchar_offset(char_offset){}
- inline bool operator()(const data_type &x, const data_type &y) const
- {
- unsigned minSize = std::min(x.size(), y.size());
- for(unsigned u = fchar_offset; u < minSize; ++u) {
- if(static_cast<unsignedchar_type>(x[u]) < static_cast<unsignedchar_type>(y[u]))
- return true;
- else if(static_cast<unsignedchar_type>(y[u]) < static_cast<unsignedchar_type>(x[u]))
- return false;
- }
- return x.size() < y.size();
- }
- unsigned fchar_offset;
- };
-
- //A comparison functor for strings that assumes they are identical up to char_offset
- template<class data_type, class unsignedchar_type>
- struct offset_greaterthan {
- offset_greaterthan(unsigned char_offset) : fchar_offset(char_offset){}
- inline bool operator()(const data_type &x, const data_type &y) const
- {
- unsigned minSize = std::min(x.size(), y.size());
- for(unsigned u = fchar_offset; u < minSize; ++u) {
- if(static_cast<unsignedchar_type>(x[u]) > static_cast<unsignedchar_type>(y[u]))
- return true;
- else if(static_cast<unsignedchar_type>(y[u]) > static_cast<unsignedchar_type>(x[u]))
- return false;
- }
- return x.size() > y.size();
- }
- unsigned fchar_offset;
- };
-
- //A comparison functor for strings that assumes they are identical up to char_offset
- template<class data_type, class get_char, class get_length>
- struct offset_char_lessthan {
- offset_char_lessthan(unsigned char_offset) : fchar_offset(char_offset){}
- inline bool operator()(const data_type &x, const data_type &y) const
- {
- unsigned minSize = std::min(length(x), length(y));
- for(unsigned u = fchar_offset; u < minSize; ++u) {
- if(getchar(x, u) < getchar(y, u))
- return true;
- else if(getchar(y, u) < getchar(x, u))
- return false;
- }
- return length(x) < length(y);
- }
- unsigned fchar_offset;
- get_char getchar;
- get_length length;
- };
-
- //String sorting recursive implementation
- template <class RandomAccessIter, class data_type, class unsignedchar_type>
- inline void
- string_sort_rec(RandomAccessIter first, RandomAccessIter last, unsigned char_offset, std::vector<RandomAccessIter> &bin_cache
- , unsigned cache_offset, std::vector<size_t> &bin_sizes)
- {
- //This section is not strictly necessary, but makes handling of long identical substrings much faster, with a mild average performance impact.
- //Iterate to the end of the empties. If all empty, return
- while((*first).size() <= char_offset) {
- if(++first == last)
- return;
- }
- RandomAccessIter finish = last - 1;
- //Getting the last non-empty
- for(;(*finish).size() <= char_offset; --finish) { }
- ++finish;
- //Offsetting on identical characters. This section works a character at a time for optimal worst-case performance.
- update_offset(first, finish, char_offset);
-
- const unsigned bin_count = (1 << (sizeof(unsignedchar_type)*8));
- //Equal worst-case between radix and comparison-based is when bin_count = n*log(n).
- const unsigned max_size = bin_count;
- const unsigned membin_count = bin_count + 1;
- unsigned cache_end;
- RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, membin_count) + 1;
-
- //Calculating the size of each bin; this takes roughly 10% of runtime
- for (RandomAccessIter current = first; current != last; ++current) {
- if((*current).size() <= char_offset) {
- bin_sizes[0]++;
- }
- else
- bin_sizes[static_cast<unsignedchar_type>((*current)[char_offset]) + 1]++;
- }
- //Assign the bin positions
- bin_cache[cache_offset] = first;
- for(unsigned u = 0; u < membin_count - 1; u++)
- bin_cache[cache_offset + u + 1] = bin_cache[cache_offset + u] + bin_sizes[u];
-
- //Swap into place
- RandomAccessIter nextbinstart = first;
- //handling empty bins
- RandomAccessIter * local_bin = &(bin_cache[cache_offset]);
- nextbinstart += bin_sizes[0];
- RandomAccessIter * target_bin;
- //Iterating over each element in the bin of empties
- for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
- //empties belong in this bin
- while((*current).size() > char_offset) {
- target_bin = bins + static_cast<unsignedchar_type>((*current)[char_offset]);
- iter_swap(current, (*target_bin)++);
- }
- }
- *local_bin = nextbinstart;
- //iterate backwards to find the last bin with elements in it; this saves iterations in multiple loops
- unsigned last_bin = bin_count - 1;
- for(; last_bin && !bin_sizes[last_bin + 1]; --last_bin) { }
- //This dominates runtime, mostly in the swap and bin lookups
- for(unsigned u = 0; u < last_bin; ++u) {
- local_bin = bins + u;
- nextbinstart += bin_sizes[u + 1];
- //Iterating over each element in this bin
- for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
- //Swapping elements in current into place until the correct element has been swapped in
- for(target_bin = bins + static_cast<unsignedchar_type>((*current)[char_offset]); target_bin != local_bin;
- target_bin = bins + static_cast<unsignedchar_type>((*current)[char_offset]))
- iter_swap(current, (*target_bin)++);
- }
- *local_bin = nextbinstart;
- }
- bins[last_bin] = last;
- //Recursing
- RandomAccessIter lastPos = bin_cache[cache_offset];
- //Skip this loop for empties
- for(unsigned u = cache_offset + 1; u < cache_offset + last_bin + 2; lastPos = bin_cache[u], ++u) {
- size_t count = bin_cache[u] - lastPos;
- //don't sort unless there are at least two items to compare
- if(count < 2)
- continue;
- //using std::sort if its worst-case is better
- if(count < max_size)
- std::sort(lastPos, bin_cache[u], offset_lessthan<data_type, unsignedchar_type>(char_offset + 1));
- else
- string_sort_rec<RandomAccessIter, data_type, unsignedchar_type>(lastPos, bin_cache[u], char_offset + 1, bin_cache, cache_end, bin_sizes);
- }
- }
-
- //Sorts strings in reverse order, with empties at the end
- template <class RandomAccessIter, class data_type, class unsignedchar_type>
- inline void
- reverse_string_sort_rec(RandomAccessIter first, RandomAccessIter last, unsigned char_offset, std::vector<RandomAccessIter> &bin_cache
- , unsigned cache_offset, std::vector<size_t> &bin_sizes)
- {
- //This section is not strictly necessary, but makes handling of long identical substrings much faster, with a mild average performance impact.
- RandomAccessIter curr = first;
- //Iterate to the end of the empties. If all empty, return
- while((*curr).size() <= char_offset) {
- if(++curr == last)
- return;
- }
- //Getting the last non-empty
- while((*(--last)).size() <= char_offset) { }
- ++last;
- //Offsetting on identical characters. This section works a character at a time for optimal worst-case performance.
- update_offset(curr, last, char_offset);
- RandomAccessIter * target_bin;
-
- const unsigned bin_count = (1 << (sizeof(unsignedchar_type)*8));
- //Equal worst-case between radix and comparison-based is when bin_count = n*log(n).
- const unsigned max_size = bin_count;
- const unsigned membin_count = bin_count + 1;
- const unsigned max_bin = bin_count - 1;
- unsigned cache_end;
- RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, membin_count);
- RandomAccessIter * end_bin = &(bin_cache[cache_offset + max_bin]);
-
- //Calculating the size of each bin; this takes roughly 10% of runtime
- for (RandomAccessIter current = first; current != last; ++current) {
- if((*current).size() <= char_offset) {
- bin_sizes[bin_count]++;
- }
- else
- bin_sizes[max_bin - static_cast<unsignedchar_type>((*current)[char_offset])]++;
- }
- //Assign the bin positions
- bin_cache[cache_offset] = first;
- for(unsigned u = 0; u < membin_count - 1; u++)
- bin_cache[cache_offset + u + 1] = bin_cache[cache_offset + u] + bin_sizes[u];
-
- //Swap into place
- RandomAccessIter nextbinstart = last;
- //handling empty bins
- RandomAccessIter * local_bin = &(bin_cache[cache_offset + bin_count]);
- RandomAccessIter lastFull = *local_bin;
- //Iterating over each element in the bin of empties
- for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
- //empties belong in this bin
- while((*current).size() > char_offset) {
- target_bin = end_bin - static_cast<unsignedchar_type>((*current)[char_offset]);
- iter_swap(current, (*target_bin)++);
- }
- }
- *local_bin = nextbinstart;
- nextbinstart = first;
- //iterate backwards to find the last bin with elements in it; this saves iterations in multiple loops
- unsigned last_bin = max_bin;
- for(; last_bin && !bin_sizes[last_bin]; --last_bin) { }
- //This dominates runtime, mostly in the swap and bin lookups
- for(unsigned u = 0; u < last_bin; ++u) {
- local_bin = bins + u;
- nextbinstart += bin_sizes[u];
- //Iterating over each element in this bin
- for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
- //Swapping elements in current into place until the correct element has been swapped in
- for(target_bin = end_bin - static_cast<unsignedchar_type>((*current)[char_offset]); target_bin != local_bin;
- target_bin = end_bin - static_cast<unsignedchar_type>((*current)[char_offset]))
- iter_swap(current, (*target_bin)++);
- }
- *local_bin = nextbinstart;
- }
- bins[last_bin] = lastFull;
- //Recursing
- RandomAccessIter lastPos = first;
- //Skip this loop for empties
- for(unsigned u = cache_offset; u <= cache_offset + last_bin; lastPos = bin_cache[u], ++u) {
- size_t count = bin_cache[u] - lastPos;
- //don't sort unless there are at least two items to compare
- if(count < 2)
- continue;
- //using std::sort if its worst-case is better
- if(count < max_size)
- std::sort(lastPos, bin_cache[u], offset_greaterthan<data_type, unsignedchar_type>(char_offset + 1));
- else
- reverse_string_sort_rec<RandomAccessIter, data_type, unsignedchar_type>(lastPos, bin_cache[u], char_offset + 1, bin_cache, cache_end, bin_sizes);
- }
- }
-
- //String sorting recursive implementation
- template <class RandomAccessIter, class data_type, class unsignedchar_type, class get_char, class get_length>
- inline void
- string_sort_rec(RandomAccessIter first, RandomAccessIter last, unsigned char_offset, std::vector<RandomAccessIter> &bin_cache
- , unsigned cache_offset, std::vector<size_t> &bin_sizes, get_char getchar, get_length length)
- {
- //This section is not strictly necessary, but makes handling of long identical substrings much faster, with a mild average performance impact.
- //Iterate to the end of the empties. If all empty, return
- while(length(*first) <= char_offset) {
- if(++first == last)
- return;
- }
- RandomAccessIter finish = last - 1;
- //Getting the last non-empty
- for(;length(*finish) <= char_offset; --finish) { }
- ++finish;
- update_offset(first, finish, char_offset, getchar, length);
-
- const unsigned bin_count = (1 << (sizeof(unsignedchar_type)*8));
- //Equal worst-case between radix and comparison-based is when bin_count = n*log(n).
- const unsigned max_size = bin_count;
- const unsigned membin_count = bin_count + 1;
- unsigned cache_end;
- RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, membin_count) + 1;
-
- //Calculating the size of each bin; this takes roughly 10% of runtime
- for (RandomAccessIter current = first; current != last; ++current) {
- if(length(*current) <= char_offset) {
- bin_sizes[0]++;
- }
- else
- bin_sizes[getchar((*current), char_offset) + 1]++;
- }
- //Assign the bin positions
- bin_cache[cache_offset] = first;
- for(unsigned u = 0; u < membin_count - 1; u++)
- bin_cache[cache_offset + u + 1] = bin_cache[cache_offset + u] + bin_sizes[u];
-
- //Swap into place
- RandomAccessIter nextbinstart = first;
- //handling empty bins
- RandomAccessIter * local_bin = &(bin_cache[cache_offset]);
- nextbinstart += bin_sizes[0];
- RandomAccessIter * target_bin;
- //Iterating over each element in the bin of empties
- for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
- //empties belong in this bin
- while(length(*current) > char_offset) {
- target_bin = bins + getchar((*current), char_offset);
- iter_swap(current, (*target_bin)++);
- }
- }
- *local_bin = nextbinstart;
- //iterate backwards to find the last bin with elements in it; this saves iterations in multiple loops
- unsigned last_bin = bin_count - 1;
- for(; last_bin && !bin_sizes[last_bin + 1]; --last_bin) { }
- //This dominates runtime, mostly in the swap and bin lookups
- for(unsigned ii = 0; ii < last_bin; ++ii) {
- local_bin = bins + ii;
- nextbinstart += bin_sizes[ii + 1];
- //Iterating over each element in this bin
- for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
- //Swapping elements in current into place until the correct element has been swapped in
- for(target_bin = bins + getchar((*current), char_offset); target_bin != local_bin;
- target_bin = bins + getchar((*current), char_offset))
- iter_swap(current, (*target_bin)++);
- }
- *local_bin = nextbinstart;
- }
- bins[last_bin] = last;
-
- //Recursing
- RandomAccessIter lastPos = bin_cache[cache_offset];
- //Skip this loop for empties
- for(unsigned u = cache_offset + 1; u < cache_offset + last_bin + 2; lastPos = bin_cache[u], ++u) {
- size_t count = bin_cache[u] - lastPos;
- //don't sort unless there are at least two items to compare
- if(count < 2)
- continue;
- //using std::sort if its worst-case is better
- if(count < max_size)
- std::sort(lastPos, bin_cache[u], offset_char_lessthan<data_type, get_char, get_length>(char_offset + 1));
- else
- string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length>(lastPos, bin_cache[u], char_offset + 1, bin_cache, cache_end, bin_sizes, getchar, length);
- }
- }
-
- //String sorting recursive implementation
- template <class RandomAccessIter, class data_type, class unsignedchar_type, class get_char, class get_length, class compare>
- inline void
- string_sort_rec(RandomAccessIter first, RandomAccessIter last, unsigned char_offset, std::vector<RandomAccessIter> &bin_cache
- , unsigned cache_offset, std::vector<size_t> &bin_sizes, get_char getchar, get_length length, compare comp)
- {
- //This section is not strictly necessary, but makes handling of long identical substrings much faster, with a mild average performance impact.
- //Iterate to the end of the empties. If all empty, return
- while(length(*first) <= char_offset) {
- if(++first == last)
- return;
- }
- RandomAccessIter finish = last - 1;
- //Getting the last non-empty
- for(;length(*finish) <= char_offset; --finish) { }
- ++finish;
- update_offset(first, finish, char_offset, getchar, length);
-
- const unsigned bin_count = (1 << (sizeof(unsignedchar_type)*8));
- //Equal worst-case between radix and comparison-based is when bin_count = n*log(n).
- const unsigned max_size = bin_count;
- const unsigned membin_count = bin_count + 1;
- unsigned cache_end;
- RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, membin_count) + 1;
-
- //Calculating the size of each bin; this takes roughly 10% of runtime
- for (RandomAccessIter current = first; current != last; ++current) {
- if(length(*current) <= char_offset) {
- bin_sizes[0]++;
- }
- else
- bin_sizes[getchar((*current), char_offset) + 1]++;
- }
- //Assign the bin positions
- bin_cache[cache_offset] = first;
- for(unsigned u = 0; u < membin_count - 1; u++)
- bin_cache[cache_offset + u + 1] = bin_cache[cache_offset + u] + bin_sizes[u];
-
- //Swap into place
- RandomAccessIter nextbinstart = first;
- //handling empty bins
- RandomAccessIter * local_bin = &(bin_cache[cache_offset]);
- nextbinstart += bin_sizes[0];
- RandomAccessIter * target_bin;
- //Iterating over each element in the bin of empties
- for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
- //empties belong in this bin
- while(length(*current) > char_offset) {
- target_bin = bins + getchar((*current), char_offset);
- iter_swap(current, (*target_bin)++);
- }
- }
- *local_bin = nextbinstart;
- //iterate backwards to find the last bin with elements in it; this saves iterations in multiple loops
- unsigned last_bin = bin_count - 1;
- for(; last_bin && !bin_sizes[last_bin + 1]; --last_bin) { }
- //This dominates runtime, mostly in the swap and bin lookups
- for(unsigned u = 0; u < last_bin; ++u) {
- local_bin = bins + u;
- nextbinstart += bin_sizes[u + 1];
- //Iterating over each element in this bin
- for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
- //Swapping elements in current into place until the correct element has been swapped in
- for(target_bin = bins + getchar((*current), char_offset); target_bin != local_bin;
- target_bin = bins + getchar((*current), char_offset))
- iter_swap(current, (*target_bin)++);
- }
- *local_bin = nextbinstart;
- }
- bins[last_bin] = last;
-
- //Recursing
- RandomAccessIter lastPos = bin_cache[cache_offset];
- //Skip this loop for empties
- for(unsigned u = cache_offset + 1; u < cache_offset + last_bin + 2; lastPos = bin_cache[u], ++u) {
- size_t count = bin_cache[u] - lastPos;
- //don't sort unless there are at least two items to compare
- if(count < 2)
- continue;
- //using std::sort if its worst-case is better
- if(count < max_size)
- std::sort(lastPos, bin_cache[u], comp);
- else
- string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length, compare>(lastPos
- , bin_cache[u], char_offset + 1, bin_cache, cache_end, bin_sizes, getchar, length, comp);
- }
- }
-
- //Sorts strings in reverse order, with empties at the end
- template <class RandomAccessIter, class data_type, class unsignedchar_type, class get_char, class get_length, class compare>
- inline void
- reverse_string_sort_rec(RandomAccessIter first, RandomAccessIter last, unsigned char_offset, std::vector<RandomAccessIter> &bin_cache
- , unsigned cache_offset, std::vector<size_t> &bin_sizes, get_char getchar, get_length length, compare comp)
- {
- //This section is not strictly necessary, but makes handling of long identical substrings much faster, with a mild average performance impact.
- RandomAccessIter curr = first;
- //Iterate to the end of the empties. If all empty, return
- while(length(*curr) <= char_offset) {
- if(++curr == last)
- return;
- }
- //Getting the last non-empty
- while(length(*(--last)) <= char_offset) { }
- ++last;
- //Offsetting on identical characters. This section works a character at a time for optimal worst-case performance.
- update_offset(first, last, char_offset, getchar, length);
-
- const unsigned bin_count = (1 << (sizeof(unsignedchar_type)*8));
- //Equal worst-case between radix and comparison-based is when bin_count = n*log(n).
- const unsigned max_size = bin_count;
- const unsigned membin_count = bin_count + 1;
- const unsigned max_bin = bin_count - 1;
- unsigned cache_end;
- RandomAccessIter * bins = size_bins(bin_sizes, bin_cache, cache_offset, cache_end, membin_count);
- RandomAccessIter *end_bin = &(bin_cache[cache_offset + max_bin]);
-
- //Calculating the size of each bin; this takes roughly 10% of runtime
- for (RandomAccessIter current = first; current != last; ++current) {
- if(length(*current) <= char_offset) {
- bin_sizes[bin_count]++;
- }
- else
- bin_sizes[max_bin - getchar((*current), char_offset)]++;
- }
- //Assign the bin positions
- bin_cache[cache_offset] = first;
- for(unsigned u = 0; u < membin_count - 1; u++)
- bin_cache[cache_offset + u + 1] = bin_cache[cache_offset + u] + bin_sizes[u];
-
- //Swap into place
- RandomAccessIter nextbinstart = last;
- //handling empty bins
- RandomAccessIter * local_bin = &(bin_cache[cache_offset + bin_count]);
- RandomAccessIter lastFull = *local_bin;
- RandomAccessIter * target_bin;
- //Iterating over each element in the bin of empties
- for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
- //empties belong in this bin
- while(length(*current) > char_offset) {
- target_bin = end_bin - getchar((*current), char_offset);
- iter_swap(current, (*target_bin)++);
- }
- }
- *local_bin = nextbinstart;
- nextbinstart = first;
- //iterate backwards to find the last bin with elements in it; this saves iterations in multiple loops
- unsigned last_bin = max_bin;
- for(; last_bin && !bin_sizes[last_bin]; --last_bin) { }
- //This dominates runtime, mostly in the swap and bin lookups
- for(unsigned u = 0; u < last_bin; ++u) {
- local_bin = bins + u;
- nextbinstart += bin_sizes[u];
- //Iterating over each element in this bin
- for(RandomAccessIter current = *local_bin; current < nextbinstart; ++current) {
- //Swapping elements in current into place until the correct element has been swapped in
- for(target_bin = end_bin - getchar((*current), char_offset); target_bin != local_bin;
- target_bin = end_bin - getchar((*current), char_offset))
- iter_swap(current, (*target_bin)++);
- }
- *local_bin = nextbinstart;
- }
- bins[last_bin] = lastFull;
- //Recursing
- RandomAccessIter lastPos = first;
- //Skip this loop for empties
- for(unsigned u = cache_offset; u <= cache_offset + last_bin; lastPos = bin_cache[u], ++u) {
- size_t count = bin_cache[u] - lastPos;
- //don't sort unless there are at least two items to compare
- if(count < 2)
- continue;
- //using std::sort if its worst-case is better
- if(count < max_size)
- std::sort(lastPos, bin_cache[u], comp);
- else
- reverse_string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length, compare>(lastPos
- , bin_cache[u], char_offset + 1, bin_cache, cache_end, bin_sizes, getchar, length, comp);
- }
- }
-
- //Holds the bin vector and makes the initial recursive call
- template <class RandomAccessIter, class data_type, class unsignedchar_type>
- inline void
- string_sort(RandomAccessIter first, RandomAccessIter last, data_type, unsignedchar_type)
- {
- std::vector<size_t> bin_sizes;
- std::vector<RandomAccessIter> bin_cache;
- string_sort_rec<RandomAccessIter, data_type, unsignedchar_type>(first, last, 0, bin_cache, 0, bin_sizes);
- }
-
- //Holds the bin vector and makes the initial recursive call
- template <class RandomAccessIter, class data_type, class unsignedchar_type>
- inline void
- reverse_string_sort(RandomAccessIter first, RandomAccessIter last, data_type, unsignedchar_type)
- {
- std::vector<size_t> bin_sizes;
- std::vector<RandomAccessIter> bin_cache;
- reverse_string_sort_rec<RandomAccessIter, data_type, unsignedchar_type>(first, last, 0, bin_cache, 0, bin_sizes);
- }
-
- //Holds the bin vector and makes the initial recursive call
- template <class RandomAccessIter, class get_char, class get_length, class data_type, class unsignedchar_type>
- inline void
- string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length, data_type, unsignedchar_type)
- {
- std::vector<size_t> bin_sizes;
- std::vector<RandomAccessIter> bin_cache;
- string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length>(first, last, 0, bin_cache, 0, bin_sizes, getchar, length);
- }
-
- //Holds the bin vector and makes the initial recursive call
- template <class RandomAccessIter, class get_char, class get_length, class compare, class data_type, class unsignedchar_type>
- inline void
- string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length, compare comp, data_type, unsignedchar_type)
- {
- std::vector<size_t> bin_sizes;
- std::vector<RandomAccessIter> bin_cache;
- string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length, compare>(first, last, 0, bin_cache, 0, bin_sizes, getchar, length, comp);
- }
-
- //Holds the bin vector and makes the initial recursive call
- template <class RandomAccessIter, class get_char, class get_length, class compare, class data_type, class unsignedchar_type>
- inline void
- reverse_string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length, compare comp, data_type, unsignedchar_type)
- {
- std::vector<size_t> bin_sizes;
- std::vector<RandomAccessIter> bin_cache;
- reverse_string_sort_rec<RandomAccessIter, data_type, unsignedchar_type, get_char, get_length, compare>(first, last, 0, bin_cache, 0, bin_sizes, getchar, length, comp);
- }
- }
-
- //Allows character-type overloads
- template <class RandomAccessIter, class unsignedchar_type>
- inline void string_sort(RandomAccessIter first, RandomAccessIter last, unsignedchar_type unused)
- {
- //Don't sort if it's too small to optimize
- if(last - first < detail::MIN_SORT_SIZE)
- std::sort(first, last);
- else
- detail::string_sort(first, last, *first, unused);
- }
-
- //Top-level sorting call; wraps using default of unsigned char
- template <class RandomAccessIter>
- inline void string_sort(RandomAccessIter first, RandomAccessIter last)
- {
- unsigned char unused = '\0';
- string_sort(first, last, unused);
- }
-
- //Allows character-type overloads
- template <class RandomAccessIter, class compare, class unsignedchar_type>
- inline void reverse_string_sort(RandomAccessIter first, RandomAccessIter last, compare comp, unsignedchar_type unused)
- {
- //Don't sort if it's too small to optimize
- if(last - first < detail::MIN_SORT_SIZE)
- std::sort(first, last, comp);
- else
- detail::reverse_string_sort(first, last, *first, unused);
- }
-
- //Top-level sorting call; wraps using default of unsigned char
- template <class RandomAccessIter, class compare>
- inline void reverse_string_sort(RandomAccessIter first, RandomAccessIter last, compare comp)
- {
- unsigned char unused = '\0';
- reverse_string_sort(first, last, comp, unused);
- }
-
- template <class RandomAccessIter, class get_char, class get_length>
- inline void string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length)
- {
- //Don't sort if it's too small to optimize
- if(last - first < detail::MIN_SORT_SIZE)
- std::sort(first, last);
- else {
- //skipping past empties at the beginning, which allows us to get the character type
- //.empty() is not used so as not to require a user declaration of it
- while(!length(*first)) {
- if(++first == last)
- return;
- }
- detail::string_sort(first, last, getchar, length, *first, getchar((*first), 0));
- }
- }
-
- template <class RandomAccessIter, class get_char, class get_length, class compare>
- inline void string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length, compare comp)
- {
- //Don't sort if it's too small to optimize
- if(last - first < detail::MIN_SORT_SIZE)
- std::sort(first, last, comp);
- else {
- //skipping past empties at the beginning, which allows us to get the character type
- //.empty() is not used so as not to require a user declaration of it
- while(!length(*first)) {
- if(++first == last)
- return;
- }
- detail::string_sort(first, last, getchar, length, comp, *first, getchar((*first), 0));
- }
- }
-
- template <class RandomAccessIter, class get_char, class get_length, class compare>
- inline void reverse_string_sort(RandomAccessIter first, RandomAccessIter last, get_char getchar, get_length length, compare comp)
- {
- //Don't sort if it's too small to optimize
- if(last - first < detail::MIN_SORT_SIZE)
- std::sort(first, last, comp);
- else {
- //skipping past empties at the beginning, which allows us to get the character type
- //.empty() is not used so as not to require a user declaration of it
- while(!length(*(--last))) {
- //Note: if there is just one non-empty, and it's at the beginning, then it's already in sorted order
- if(first == last)
- return;
- }
- //making last just after the end of the non-empty part of the array
- ++last;
- detail::reverse_string_sort(first, last, getchar, length, comp, *first, getchar((*first), 0));
- }
- }
-}
-
-#endif
diff --git a/trunk/src/system_wrappers/source/system_wrappers.gyp b/trunk/src/system_wrappers/source/system_wrappers.gyp
deleted file mode 100644
index 79820e9..0000000
--- a/trunk/src/system_wrappers/source/system_wrappers.gyp
+++ /dev/null
@@ -1,190 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'includes': [ '../../build/common.gypi', ],
- 'targets': [
- {
- 'target_name': 'system_wrappers',
- 'type': '<(library)',
- 'include_dirs': [
- 'spreadsortlib',
- '../interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../interface',
- ],
- },
- 'sources': [
- '../interface/aligned_malloc.h',
- '../interface/atomic32_wrapper.h',
- '../interface/condition_variable_wrapper.h',
- '../interface/cpu_info.h',
- '../interface/cpu_wrapper.h',
- '../interface/cpu_features_wrapper.h',
- '../interface/critical_section_wrapper.h',
- '../interface/data_log.h',
- '../interface/data_log_c.h',
- '../interface/data_log_impl.h',
- '../interface/event_wrapper.h',
- '../interface/file_wrapper.h',
- '../interface/fix_interlocked_exchange_pointer_win.h',
- '../interface/list_wrapper.h',
- '../interface/map_wrapper.h',
- '../interface/ref_count.h',
- '../interface/rw_lock_wrapper.h',
- '../interface/scoped_ptr.h',
- '../interface/scoped_refptr.h',
- '../interface/sort.h',
- '../interface/static_instance.h',
- '../interface/thread_wrapper.h',
- '../interface/tick_util.h',
- '../interface/trace.h',
- 'aligned_malloc.cc',
- 'atomic32.cc',
- 'atomic32_linux.h',
- 'atomic32_mac.h',
- 'atomic32_win.h',
- 'condition_variable.cc',
- 'condition_variable_posix.cc',
- 'condition_variable_posix.h',
- 'condition_variable_win.cc',
- 'condition_variable_win.h',
- 'cpu.cc',
- 'cpu_no_op.cc',
- 'cpu_info.cc',
- 'cpu_linux.cc',
- 'cpu_linux.h',
- 'cpu_mac.cc',
- 'cpu_mac.h',
- 'cpu_win.cc',
- 'cpu_win.h',
- 'cpu_features.cc',
- 'critical_section.cc',
- 'critical_section_posix.cc',
- 'critical_section_posix.h',
- 'critical_section_win.cc',
- 'critical_section_win.h',
- 'data_log.cc',
- 'data_log_c.cc',
- 'data_log_no_op.cc',
- 'event.cc',
- 'event_posix.cc',
- 'event_posix.h',
- 'event_win.cc',
- 'event_win.h',
- 'file_impl.cc',
- 'file_impl.h',
- 'list_no_stl.cc',
- 'map.cc',
- 'rw_lock.cc',
- 'rw_lock_posix.cc',
- 'rw_lock_posix.h',
- 'rw_lock_win.cc',
- 'rw_lock_win.h',
- 'sort.cc',
- 'thread.cc',
- 'thread_posix.cc',
- 'thread_posix.h',
- 'thread_win.cc',
- 'thread_win.h',
- 'set_thread_name_win.h',
- 'trace_impl.cc',
- 'trace_impl.h',
- 'trace_impl_no_op.cc',
- 'trace_posix.cc',
- 'trace_posix.h',
- 'trace_win.cc',
- 'trace_win.h',
- ],
- 'conditions': [
- ['enable_data_logging==1', {
- 'sources!': [ 'data_log_no_op.cc', ],
- },{
- 'sources!': [ 'data_log.cc', ],
- },],
- ['OS=="linux"', {
- 'link_settings': {
- 'libraries': [ '-lrt', ],
- },
- }],
- ['OS=="mac"', {
- 'link_settings': {
- 'libraries': [ '$(SDKROOT)/System/Library/Frameworks/ApplicationServices.framework', ],
- },
- }],
- ['OS=="win"', {
- 'link_settings': {
- 'libraries': [ '-lwinmm.lib', ],
- },
- }],
- ['build_with_chromium==1', {
- 'sources!': [
- 'cpu.cc',
- 'cpu_linux.h',
- 'cpu_mac.h',
- 'cpu_win.h',
- 'trace_impl.cc',
- 'trace_impl.h',
- 'trace_posix.cc',
- 'trace_posix.h',
- 'trace_win.cc',
- 'trace_win.h',
- ],
- }, {
- 'sources!': [
- 'cpu_no_op.cc',
- 'trace_impl_no_op.cc',
- ],
- }]
- ] # conditions
- },
- ], # targets
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'system_wrappers_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'system_wrappers',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- ],
- 'sources': [
- 'cpu_wrapper_unittest.cc',
- 'cpu_measurement_harness.h',
- 'cpu_measurement_harness.cc',
- 'list_unittest.cc',
- 'map_unittest.cc',
- 'data_log_unittest.cc',
- 'data_log_unittest_disabled.cc',
- 'data_log_helpers_unittest.cc',
- 'data_log_c_helpers_unittest.c',
- 'data_log_c_helpers_unittest.h',
- 'trace_unittest.cc',
- ],
- 'conditions': [
- ['enable_data_logging==1', {
- 'sources!': [ 'data_log_unittest_disabled.cc', ],
- }, {
- 'sources!': [ 'data_log_unittest.cc', ],
- }],
- ],
- },
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/system_wrappers/source/thread.cc b/trunk/src/system_wrappers/source/thread.cc
deleted file mode 100644
index 32dcc63..0000000
--- a/trunk/src/system_wrappers/source/thread.cc
+++ /dev/null
@@ -1,30 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "thread_wrapper.h"
-
-#if defined(_WIN32)
- #include "thread_win.h"
-#else
- #include "thread_posix.h"
-#endif
-
-namespace webrtc {
-ThreadWrapper* ThreadWrapper::CreateThread(ThreadRunFunction func,
- ThreadObj obj, ThreadPriority prio,
- const char* threadName)
-{
-#if defined(_WIN32)
- return new ThreadWindows(func, obj, prio, threadName);
-#else
- return ThreadPosix::Create(func, obj, prio, threadName);
-#endif
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/thread_posix.cc b/trunk/src/system_wrappers/source/thread_posix.cc
deleted file mode 100644
index 05e7943..0000000
--- a/trunk/src/system_wrappers/source/thread_posix.cc
+++ /dev/null
@@ -1,340 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "thread_posix.h"
-
-#include <errno.h>
-#include <string.h> // strncpy
-#include <time.h> // nanosleep
-#include <unistd.h>
-#ifdef WEBRTC_LINUX
-#include <sys/types.h>
-#include <sched.h>
-#include <sys/syscall.h>
-#include <linux/unistd.h>
-#include <sys/prctl.h>
-#endif
-
-#if defined(WEBRTC_MAC)
-#include <mach/mach.h>
-#endif
-
-#include "event_wrapper.h"
-#include "trace.h"
-
-namespace webrtc {
-extern "C"
-{
- static void* StartThread(void* lpParameter)
- {
- static_cast<ThreadPosix*>(lpParameter)->Run();
- return 0;
- }
-}
-
-ThreadWrapper* ThreadPosix::Create(ThreadRunFunction func, ThreadObj obj,
- ThreadPriority prio, const char* threadName)
-{
- ThreadPosix* ptr = new ThreadPosix(func, obj, prio, threadName);
- if (!ptr)
- {
- return NULL;
- }
- const int error = ptr->Construct();
- if (error)
- {
- delete ptr;
- return NULL;
- }
- return ptr;
-}
-
-ThreadPosix::ThreadPosix(ThreadRunFunction func, ThreadObj obj,
- ThreadPriority prio, const char* threadName)
- : _runFunction(func),
- _obj(obj),
- _alive(false),
- _dead(true),
- _prio(prio),
- _event(EventWrapper::Create()),
- _name(),
- _setThreadName(false),
-#if (defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID))
- _pid(-1),
-#endif
- _attr(),
- _thread(0)
-{
- if (threadName != NULL)
- {
- _setThreadName = true;
- strncpy(_name, threadName, kThreadMaxNameLength);
- _name[kThreadMaxNameLength - 1] = '\0';
- }
-}
-
-uint32_t ThreadWrapper::GetThreadId() {
-#if defined(WEBRTC_ANDROID) || defined(WEBRTC_LINUX)
- return static_cast<uint32_t>(syscall(__NR_gettid));
-#elif defined(WEBRTC_MAC)
- return static_cast<uint32_t>(mach_thread_self());
-#else
- return reinterpret_cast<uint32_t>(pthread_self());
-#endif
-}
-
-int ThreadPosix::Construct()
-{
- int result = 0;
-#if !defined(WEBRTC_ANDROID)
- // Enable immediate cancellation if requested, see Shutdown()
- result = pthread_setcancelstate(PTHREAD_CANCEL_ENABLE, NULL);
- if (result != 0)
- {
- return -1;
- }
- result = pthread_setcanceltype(PTHREAD_CANCEL_ASYNCHRONOUS, NULL);
- if (result != 0)
- {
- return -1;
- }
-#endif
- result = pthread_attr_init(&_attr);
- if (result != 0)
- {
- return -1;
- }
- return 0;
-}
-
-ThreadPosix::~ThreadPosix()
-{
- pthread_attr_destroy(&_attr);
- delete _event;
-}
-
-#define HAS_THREAD_ID !defined(MAC_IPHONE) && !defined(MAC_IPHONE_SIM) && \
- !defined(WEBRTC_MAC) && !defined(WEBRTC_MAC_INTEL) && \
- !defined(MAC_DYLIB) && !defined(MAC_INTEL_DYLIB)
-#if HAS_THREAD_ID
-bool ThreadPosix::Start(unsigned int& threadID)
-#else
-bool ThreadPosix::Start(unsigned int& /*threadID*/)
-#endif
-{
- if (!_runFunction)
- {
- return false;
- }
- int result = pthread_attr_setdetachstate(&_attr, PTHREAD_CREATE_DETACHED);
- // Set the stack stack size to 1M.
- result |= pthread_attr_setstacksize(&_attr, 1024*1024);
-#ifdef WEBRTC_THREAD_RR
- const int policy = SCHED_RR;
-#else
- const int policy = SCHED_FIFO;
-#endif
- _event->Reset();
- result |= pthread_create(&_thread, &_attr, &StartThread, this);
- if (result != 0)
- {
- return false;
- }
-
- // Wait up to 10 seconds for the OS to call the callback function. Prevents
- // race condition if Stop() is called too quickly after start.
- if (kEventSignaled != _event->Wait(WEBRTC_EVENT_10_SEC))
- {
- // Timed out. Something went wrong.
- _runFunction = NULL;
- return false;
- }
-
-#if HAS_THREAD_ID
- threadID = static_cast<unsigned int>(_thread);
-#endif
- sched_param param;
-
- const int minPrio = sched_get_priority_min(policy);
- const int maxPrio = sched_get_priority_max(policy);
- if ((minPrio == EINVAL) || (maxPrio == EINVAL))
- {
- return false;
- }
-
- switch (_prio)
- {
- case kLowPriority:
- param.sched_priority = minPrio + 1;
- break;
- case kNormalPriority:
- param.sched_priority = (minPrio + maxPrio) / 2;
- break;
- case kHighPriority:
- param.sched_priority = maxPrio - 3;
- break;
- case kHighestPriority:
- param.sched_priority = maxPrio - 2;
- break;
- case kRealtimePriority:
- param.sched_priority = maxPrio - 1;
- break;
- }
- result = pthread_setschedparam(_thread, policy, ¶m);
- if (result == EINVAL)
- {
- return false;
- }
- return true;
-}
-
-// CPU_ZERO and CPU_SET are not available in NDK r7, so disable
-// SetAffinity on Android for now.
-#if (defined(WEBRTC_LINUX) && (!defined(WEBRTC_ANDROID)))
-bool ThreadPosix::SetAffinity(const int* processorNumbers,
- const unsigned int amountOfProcessors) {
- if (!processorNumbers || (amountOfProcessors == 0)) {
- return false;
- }
- cpu_set_t mask;
- CPU_ZERO(&mask);
-
- for (unsigned int processor = 0;
- processor < amountOfProcessors;
- processor++) {
- CPU_SET(processorNumbers[processor], &mask);
- }
-#if defined(WEBRTC_ANDROID)
- // Android.
- const int result = syscall(__NR_sched_setaffinity,
- _pid,
- sizeof(mask),
- &mask);
-#else
- // "Normal" Linux.
- const int result = sched_setaffinity(_pid,
- sizeof(mask),
- &mask);
-#endif
- if (result != 0) {
- return false;
- }
- return true;
-}
-
-#else
-// NOTE: On Mac OS X, use the Thread affinity API in
-// /usr/include/mach/thread_policy.h: thread_policy_set and mach_thread_self()
-// instead of Linux gettid() syscall.
-bool ThreadPosix::SetAffinity(const int* , const unsigned int)
-{
- return false;
-}
-#endif
-
-void ThreadPosix::SetNotAlive()
-{
- _alive = false;
-}
-
-bool ThreadPosix::Shutdown()
-{
-#if !defined(WEBRTC_ANDROID)
- if (_thread && (0 != pthread_cancel(_thread)))
- {
- return false;
- }
-
- return true;
-#else
- return false;
-#endif
-}
-
-bool ThreadPosix::Stop()
-{
- _alive = false;
-
- // TODO (hellner) why not use an event here?
- // Wait up to 10 seconds for the thread to terminate
- for (int i = 0; i < 1000 && !_dead; i++)
- {
- timespec t;
- t.tv_sec = 0;
- t.tv_nsec = 10*1000*1000;
- nanosleep(&t, NULL);
- }
- if (_dead)
- {
- return true;
- }
- else
- {
- return false;
- }
-}
-
-void ThreadPosix::Run()
-{
- _alive = true;
- _dead = false;
-#if (defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID))
- _pid = GetThreadId();
-#endif
- // The event the Start() is waiting for.
- _event->Set();
-
- if (_setThreadName)
- {
-#ifdef WEBRTC_LINUX
- prctl(PR_SET_NAME, (unsigned long)_name, 0, 0, 0);
-#endif
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility,-1,
- "Thread with name:%s started ", _name);
- } else
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, -1,
- "Thread without name started");
- }
- do
- {
- if (_runFunction)
- {
- if (!_runFunction(_obj))
- {
- _alive = false;
- }
- }
- else
- {
- _alive = false;
- }
- }
- while (_alive);
-
- if (_setThreadName)
- {
- // Don't set the name for the trace thread because it may cause a
- // deadlock. TODO (hellner) there should be a better solution than
- // coupling the thread and the trace class like this.
- if (strcmp(_name, "Trace"))
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility,-1,
- "Thread with name:%s stopped", _name);
- }
- }
- else
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility,-1,
- "Thread without name stopped");
- }
- _dead = true;
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/thread_posix.h b/trunk/src/system_wrappers/source/thread_posix.h
deleted file mode 100644
index fc5757d..0000000
--- a/trunk/src/system_wrappers/source/thread_posix.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_POSIX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_POSIX_H_
-
-#include "thread_wrapper.h"
-#include <pthread.h>
-
-namespace webrtc {
-class EventWrapper;
-
-class ThreadPosix : public ThreadWrapper
-{
-public:
- static ThreadWrapper* Create(ThreadRunFunction func, ThreadObj obj,
- ThreadPriority prio, const char* threadName);
-
- ThreadPosix(ThreadRunFunction func, ThreadObj obj, ThreadPriority prio,
- const char* threadName);
- ~ThreadPosix();
-
- // From ThreadWrapper
- virtual void SetNotAlive();
- virtual bool Start(unsigned int& id);
- // Not implemented on Mac
- virtual bool SetAffinity(const int* processorNumbers,
- unsigned int amountOfProcessors);
- virtual bool Stop();
- virtual bool Shutdown();
-
- void Run();
-
-private:
- int Construct();
-
-private:
- // processing function
- ThreadRunFunction _runFunction;
- ThreadObj _obj;
-
- // internal state
- bool _alive;
- bool _dead;
- ThreadPriority _prio;
- EventWrapper* _event;
-
- // zero-terminated thread name string
- char _name[kThreadMaxNameLength];
- bool _setThreadName;
-
- // handle to thread
-#if (defined(WEBRTC_LINUX) || defined(WEBRTC_ANDROID))
- pid_t _pid;
-#endif
- pthread_attr_t _attr;
- pthread_t _thread;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_POSIX_H_
diff --git a/trunk/src/system_wrappers/source/thread_win.cc b/trunk/src/system_wrappers/source/thread_win.cc
deleted file mode 100644
index 4599bed..0000000
--- a/trunk/src/system_wrappers/source/thread_win.cc
+++ /dev/null
@@ -1,238 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "thread_win.h"
-
-#include <assert.h>
-#include <process.h>
-#include <stdio.h>
-#include <windows.h>
-
-#include "set_thread_name_win.h"
-#include "trace.h"
-
-#if defined(_WIN32)
-// VS 2005: Disable warnings for default initialized arrays.
-#pragma warning(disable:4351)
-#endif
-
-namespace webrtc {
-ThreadWindows::ThreadWindows(ThreadRunFunction func, ThreadObj obj,
- ThreadPriority prio, const char* threadName)
- : ThreadWrapper(),
- _runFunction(func),
- _obj(obj),
- _alive(false),
- _dead(true),
- _doNotCloseHandle(false),
- _prio(prio),
- _event(NULL),
- _thread(NULL),
- _id(0),
- _name(),
- _setThreadName(false)
-{
- _event = EventWrapper::Create();
- _critsectStop = CriticalSectionWrapper::CreateCriticalSection();
- if (threadName != NULL)
- {
- // Set the thread name to appear in the VS debugger.
- _setThreadName = true;
- strncpy(_name, threadName, kThreadMaxNameLength);
- }
-}
-
-ThreadWindows::~ThreadWindows()
-{
-#ifdef _DEBUG
- assert(!_alive);
-#endif
- if (_thread)
- {
- CloseHandle(_thread);
- }
- if(_event)
- {
- delete _event;
- }
- if(_critsectStop)
- {
- delete _critsectStop;
- }
-}
-
-uint32_t ThreadWrapper::GetThreadId() {
- return GetCurrentThreadId();
-}
-
-unsigned int WINAPI ThreadWindows::StartThread(LPVOID lpParameter)
-{
- static_cast<ThreadWindows*>(lpParameter)->Run();
- return 0;
-}
-
-bool ThreadWindows::Start(unsigned int& threadID)
-{
- _doNotCloseHandle = false;
-
- // Set stack size to 1M
- _thread=(HANDLE)_beginthreadex(NULL, 1024*1024, StartThread, (void*)this, 0,
- &threadID);
- if(_thread == NULL)
- {
- return false;
- }
- _id = threadID;
- _event->Wait(INFINITE);
-
- switch(_prio)
- {
- case kLowPriority:
- SetThreadPriority(_thread, THREAD_PRIORITY_BELOW_NORMAL);
- break;
- case kNormalPriority:
- SetThreadPriority(_thread, THREAD_PRIORITY_NORMAL);
- break;
- case kHighPriority:
- SetThreadPriority(_thread, THREAD_PRIORITY_ABOVE_NORMAL);
- break;
- case kHighestPriority:
- SetThreadPriority(_thread, THREAD_PRIORITY_HIGHEST);
- break;
- case kRealtimePriority:
- SetThreadPriority(_thread, THREAD_PRIORITY_TIME_CRITICAL);
- break;
- };
- return true;
-}
-
-bool ThreadWindows::SetAffinity(const int* processorNumbers,
- const unsigned int amountOfProcessors)
-{
- DWORD_PTR processorBitMask = 0;
- for(unsigned int processorIndex = 0;
- processorIndex < amountOfProcessors;
- processorIndex++)
- {
- // Convert from an array with processor numbers to a bitmask
- // Processor numbers start at zero.
- // TODO (hellner): this looks like a bug. Shouldn't the '=' be a '+='?
- // Or even better |=
- processorBitMask = 1 << processorNumbers[processorIndex];
- }
- return SetThreadAffinityMask(_thread,processorBitMask) != 0;
-}
-
-void ThreadWindows::SetNotAlive()
-{
- _alive = false;
-}
-
-bool ThreadWindows::Shutdown()
-{
- DWORD exitCode = 0;
- BOOL ret = TRUE;
- if (_thread)
- {
- ret = TerminateThread(_thread, exitCode);
- _alive = false;
- _dead = true;
- _thread = NULL;
- }
- return ret == TRUE;
-}
-
-bool ThreadWindows::Stop()
-{
- _critsectStop->Enter();
- // Prevents the handle from being closed in ThreadWindows::Run()
- _doNotCloseHandle = true;
- _alive = false;
- bool signaled = false;
- if (_thread && !_dead)
- {
- _critsectStop->Leave();
- // Wait up to 2 seconds for the thread to complete.
- if( WAIT_OBJECT_0 == WaitForSingleObject(_thread, 2000))
- {
- signaled = true;
- }
- _critsectStop->Enter();
- }
- if (_thread)
- {
- CloseHandle(_thread);
- _thread = NULL;
- }
- _critsectStop->Leave();
-
- if (_dead || signaled)
- {
- return true;
- }
- else
- {
- return false;
- }
-}
-
-void ThreadWindows::Run()
-{
- _alive = true;
- _dead = false;
- _event->Set();
-
- // All tracing must be after _event->Set to avoid deadlock in Trace.
- if (_setThreadName)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, _id,
- "Thread with name:%s started ", _name);
- SetThreadName(-1, _name); // -1, set thread name for the calling thread.
- }else
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, _id,
- "Thread without name started");
- }
-
- do
- {
- if (_runFunction)
- {
- if (!_runFunction(_obj))
- {
- _alive = false;
- }
- } else {
- _alive = false;
- }
- } while(_alive);
-
- if (_setThreadName)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility, _id,
- "Thread with name:%s stopped", _name);
- } else {
- WEBRTC_TRACE(kTraceStateInfo, kTraceUtility,_id,
- "Thread without name stopped");
- }
-
- _critsectStop->Enter();
-
- if (_thread && !_doNotCloseHandle)
- {
- HANDLE thread = _thread;
- _thread = NULL;
- CloseHandle(thread);
- }
- _dead = true;
-
- _critsectStop->Leave();
-};
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/thread_win.h b/trunk/src/system_wrappers/source/thread_win.h
deleted file mode 100644
index 4fd7523..0000000
--- a/trunk/src/system_wrappers/source/thread_win.h
+++ /dev/null
@@ -1,66 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_WINDOWS_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_WINDOWS_H_
-
-#include "thread_wrapper.h"
-#include "event_wrapper.h"
-#include "critical_section_wrapper.h"
-
-#include <windows.h>
-
-namespace webrtc {
-
-class ThreadWindows : public ThreadWrapper
-{
-public:
- ThreadWindows(ThreadRunFunction func, ThreadObj obj, ThreadPriority prio,
- const char* threadName);
- virtual ~ThreadWindows();
-
- virtual bool Start(unsigned int& id);
- bool SetAffinity(const int* processorNumbers,
- const unsigned int amountOfProcessors);
- virtual bool Stop();
- virtual void SetNotAlive();
-
- static unsigned int WINAPI StartThread(LPVOID lpParameter);
-
- virtual bool Shutdown();
-
-protected:
- virtual void Run();
-
-private:
- ThreadRunFunction _runFunction;
- ThreadObj _obj;
-
- bool _alive;
- bool _dead;
-
- // TODO (hellner)
- // _doNotCloseHandle member seem pretty redundant. Should be able to remove
- // it. Basically it should be fine to reclaim the handle when calling stop
- // and in the destructor.
- bool _doNotCloseHandle;
- ThreadPriority _prio;
- EventWrapper* _event;
- CriticalSectionWrapper* _critsectStop;
-
- HANDLE _thread;
- unsigned int _id;
- char _name[kThreadMaxNameLength];
- bool _setThreadName;
-
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_THREAD_WINDOWS_H_
diff --git a/trunk/src/system_wrappers/source/trace_impl.cc b/trunk/src/system_wrappers/source/trace_impl.cc
deleted file mode 100644
index ad908a3..0000000
--- a/trunk/src/system_wrappers/source/trace_impl.cc
+++ /dev/null
@@ -1,829 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "trace_impl.h"
-
-#include <cassert>
-#include <string.h> // memset
-
-#ifdef _WIN32
-#include "trace_win.h"
-#else
-#include <stdio.h>
-#include <time.h>
-#include <stdarg.h>
-#include "trace_posix.h"
-#endif // _WIN32
-
-#define KEY_LEN_CHARS 31
-
-#ifdef _WIN32
- #pragma warning(disable:4355)
-// VS 2005: Disable warnings for default initialized arrays.
- #pragma warning(disable:4351)
-#endif // _WIN32
-
-namespace webrtc {
-static WebRtc_UWord32 levelFilter = kTraceDefault;
-
-// Construct On First Use idiom. Avoids "static initialization order fiasco".
-TraceImpl* TraceImpl::StaticInstance(CountOperation count_operation,
- const TraceLevel level)
-{
- // Sanities to avoid taking lock unless absolutely necessary (for
- // performance reasons).
- // count_operation == kAddRefNoCreate implies that a message will be
- // written to file.
- if((level != kTraceAll) && (count_operation == kAddRefNoCreate))
- {
- if(!(level & levelFilter))
- {
- return NULL;
- }
- }
- TraceImpl* impl =
- GetStaticInstance<TraceImpl>(count_operation);
- return impl;
-}
-
-TraceImpl* TraceImpl::GetTrace(const TraceLevel level)
-{
- return StaticInstance(kAddRefNoCreate, level);
-}
-
-TraceImpl* TraceImpl::CreateInstance()
-{
-#if defined(_WIN32)
- return new TraceWindows();
-#else
- return new TracePosix();
-#endif
-}
-
-TraceImpl::TraceImpl()
- : _critsectInterface(CriticalSectionWrapper::CreateCriticalSection()),
- _callback(NULL),
- _rowCountText(0),
- _fileCountText(0),
- _traceFile(*FileWrapper::Create()),
- _thread(*ThreadWrapper::CreateThread(TraceImpl::Run, this,
- kHighestPriority, "Trace")),
- _event(*EventWrapper::Create()),
- _critsectArray(CriticalSectionWrapper::CreateCriticalSection()),
- _nextFreeIdx(),
- _level(),
- _length(),
- _messageQueue(),
- _activeQueue(0)
-{
- _nextFreeIdx[0] = 0;
- _nextFreeIdx[1] = 0;
-
- unsigned int tid = 0;
- _thread.Start(tid);
-
- for(int m = 0; m < WEBRTC_TRACE_NUM_ARRAY; m++)
- {
- for(int n = 0; n < WEBRTC_TRACE_MAX_QUEUE; n++)
- {
- _messageQueue[m][n] = new
- char[WEBRTC_TRACE_MAX_MESSAGE_SIZE];
- }
- }
-}
-
-bool TraceImpl::StopThread()
-{
- // Release the worker thread so that it can flush any lingering messages.
- _event.Set();
-
- // Allow 10 ms for pending messages to be flushed out.
- // TODO (hellner): why not use condition variables to do this? Or let the
- // worker thread die and let this thread flush remaining
- // messages?
-#ifdef _WIN32
- Sleep(10);
-#else
- timespec t;
- t.tv_sec = 0;
- t.tv_nsec = 10*1000000;
- nanosleep(&t,NULL);
-#endif
-
- _thread.SetNotAlive();
- // Make sure the thread finishes as quickly as possible (instead of having
- // to wait for the timeout).
- _event.Set();
- bool stopped = _thread.Stop();
-
- CriticalSectionScoped lock(_critsectInterface);
- _traceFile.Flush();
- _traceFile.CloseFile();
- return stopped;
-}
-
-TraceImpl::~TraceImpl()
-{
- StopThread();
- delete &_event;
- delete &_traceFile;
- delete &_thread;
- delete _critsectInterface;
- delete _critsectArray;
-
- for(int m = 0; m < WEBRTC_TRACE_NUM_ARRAY; m++)
- {
- for(int n = 0; n < WEBRTC_TRACE_MAX_QUEUE; n++)
- {
- delete [] _messageQueue[m][n];
- }
- }
-}
-
-WebRtc_Word32 TraceImpl::AddThreadId(char* traceMessage) const {
- WebRtc_UWord32 threadId = ThreadWrapper::GetThreadId();
- // Messages is 12 characters.
- return sprintf(traceMessage, "%10u; ", threadId);
-}
-
-WebRtc_Word32 TraceImpl::AddLevel(char* szMessage, const TraceLevel level) const
-{
- switch (level)
- {
- case kTraceStateInfo:
- sprintf (szMessage, "STATEINFO ; ");
- break;
- case kTraceWarning:
- sprintf (szMessage, "WARNING ; ");
- break;
- case kTraceError:
- sprintf (szMessage, "ERROR ; ");
- break;
- case kTraceCritical:
- sprintf (szMessage, "CRITICAL ; ");
- break;
- case kTraceInfo:
- sprintf (szMessage, "DEBUGINFO ; ");
- break;
- case kTraceModuleCall:
- sprintf (szMessage, "MODULECALL; ");
- break;
- case kTraceMemory:
- sprintf (szMessage, "MEMORY ; ");
- break;
- case kTraceTimer:
- sprintf (szMessage, "TIMER ; ");
- break;
- case kTraceStream:
- sprintf (szMessage, "STREAM ; ");
- break;
- case kTraceApiCall:
- sprintf (szMessage, "APICALL ; ");
- break;
- case kTraceDebug:
- sprintf (szMessage, "DEBUG ; ");
- break;
- default:
- assert(false);
- return 0;
- }
- // All messages are 12 characters.
- return 12;
-}
-
-WebRtc_Word32 TraceImpl::AddModuleAndId(char* traceMessage,
- const TraceModule module,
- const WebRtc_Word32 id) const
-{
- // Use long int to prevent problems with different definitions of
- // WebRtc_Word32.
- // TODO (hellner): is this actually a problem? If so, it should be better to
- // clean up WebRtc_Word32
- const long int idl = id;
- if(idl != -1)
- {
- const unsigned long int idEngine = id>>16;
- const unsigned long int idChannel = id & 0xffff;
-
- switch (module)
- {
- case kTraceVoice:
- sprintf(traceMessage, " VOICE:%5ld %5ld;", idEngine,
- idChannel);
- break;
- case kTraceVideo:
- sprintf(traceMessage, " VIDEO:%5ld %5ld;", idEngine,
- idChannel);
- break;
- case kTraceUtility:
- sprintf(traceMessage, " UTILITY:%5ld %5ld;", idEngine,
- idChannel);
- break;
- case kTraceRtpRtcp:
- sprintf(traceMessage, " RTP/RTCP:%5ld %5ld;", idEngine,
- idChannel);
- break;
- case kTraceTransport:
- sprintf(traceMessage, " TRANSPORT:%5ld %5ld;", idEngine,
- idChannel);
- break;
- case kTraceAudioCoding:
- sprintf(traceMessage, "AUDIO CODING:%5ld %5ld;", idEngine,
- idChannel);
- break;
- case kTraceSrtp:
- sprintf(traceMessage, " SRTP:%5ld %5ld;", idEngine,
- idChannel);
- break;
- case kTraceAudioMixerServer:
- sprintf(traceMessage, " AUDIO MIX/S:%5ld %5ld;", idEngine,
- idChannel);
- break;
- case kTraceAudioMixerClient:
- sprintf(traceMessage, " AUDIO MIX/C:%5ld %5ld;", idEngine,
- idChannel);
- break;
- case kTraceVideoCoding:
- sprintf(traceMessage, "VIDEO CODING:%5ld %5ld;", idEngine,
- idChannel);
- break;
- case kTraceVideoMixer:
- // Print sleep time and API call
- sprintf(traceMessage, " VIDEO MIX:%5ld %5ld;", idEngine,
- idChannel);
- break;
- case kTraceFile:
- sprintf(traceMessage, " FILE:%5ld %5ld;", idEngine,
- idChannel);
- break;
- case kTraceAudioProcessing:
- sprintf(traceMessage, " AUDIO PROC:%5ld %5ld;", idEngine,
- idChannel);
- break;
- case kTraceAudioDevice:
- sprintf(traceMessage, "AUDIO DEVICE:%5ld %5ld;", idEngine,
- idChannel);
- break;
- case kTraceVideoRenderer:
- sprintf(traceMessage, "VIDEO RENDER:%5ld %5ld;", idEngine,
- idChannel);
- break;
- case kTraceVideoCapture:
- sprintf(traceMessage, "VIDEO CAPTUR:%5ld %5ld;", idEngine,
- idChannel);
- break;
- case kTraceVideoPreocessing:
- sprintf(traceMessage, " VIDEO PROC:%5ld %5ld;", idEngine,
- idChannel);
- break;
- }
- } else {
- switch (module)
- {
- case kTraceVoice:
- sprintf (traceMessage, " VOICE:%11ld;", idl);
- break;
- case kTraceVideo:
- sprintf (traceMessage, " VIDEO:%11ld;", idl);
- break;
- case kTraceUtility:
- sprintf (traceMessage, " UTILITY:%11ld;", idl);
- break;
- case kTraceRtpRtcp:
- sprintf (traceMessage, " RTP/RTCP:%11ld;", idl);
- break;
- case kTraceTransport:
- sprintf (traceMessage, " TRANSPORT:%11ld;", idl);
- break;
- case kTraceAudioCoding:
- sprintf (traceMessage, "AUDIO CODING:%11ld;", idl);
- break;
- case kTraceSrtp:
- sprintf (traceMessage, " SRTP:%11ld;", idl);
- break;
- case kTraceAudioMixerServer:
- sprintf (traceMessage, " AUDIO MIX/S:%11ld;", idl);
- break;
- case kTraceAudioMixerClient:
- sprintf (traceMessage, " AUDIO MIX/C:%11ld;", idl);
- break;
- case kTraceVideoCoding:
- sprintf (traceMessage, "VIDEO CODING:%11ld;", idl);
- break;
- case kTraceVideoMixer:
- sprintf (traceMessage, " VIDEO MIX:%11ld;", idl);
- break;
- case kTraceFile:
- sprintf (traceMessage, " FILE:%11ld;", idl);
- break;
- case kTraceAudioProcessing:
- sprintf (traceMessage, " AUDIO PROC:%11ld;", idl);
- break;
- case kTraceAudioDevice:
- sprintf (traceMessage, "AUDIO DEVICE:%11ld;", idl);
- break;
- case kTraceVideoRenderer:
- sprintf (traceMessage, "VIDEO RENDER:%11ld;", idl);
- break;
- case kTraceVideoCapture:
- sprintf (traceMessage, "VIDEO CAPTUR:%11ld;", idl);
- break;
- case kTraceVideoPreocessing:
- sprintf (traceMessage, " VIDEO PROC:%11ld;", idl);
- break;
- }
- }
- // All messages are 25 characters.
- return 25;
-}
-
-WebRtc_Word32 TraceImpl::SetTraceFileImpl(const char* fileNameUTF8,
- const bool addFileCounter)
-{
- CriticalSectionScoped lock(_critsectInterface);
-
- _traceFile.Flush();
- _traceFile.CloseFile();
-
- if(fileNameUTF8)
- {
- if(addFileCounter)
- {
- _fileCountText = 1;
-
- char fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize];
- CreateFileName(fileNameUTF8, fileNameWithCounterUTF8,
- _fileCountText);
- if(_traceFile.OpenFile(fileNameWithCounterUTF8, false, false,
- true) == -1)
- {
- return -1;
- }
- }else {
- _fileCountText = 0;
- if(_traceFile.OpenFile(fileNameUTF8, false, false, true) == -1)
- {
- return -1;
- }
- }
- }
- _rowCountText = 0;
- return 0;
-}
-
-WebRtc_Word32 TraceImpl::TraceFileImpl(
- char fileNameUTF8[FileWrapper::kMaxFileNameSize])
-{
- CriticalSectionScoped lock(_critsectInterface);
- return _traceFile.FileName(fileNameUTF8, FileWrapper::kMaxFileNameSize);
-}
-
-WebRtc_Word32 TraceImpl::SetTraceCallbackImpl(TraceCallback* callback)
-{
- CriticalSectionScoped lock(_critsectInterface);
- _callback = callback;
- return 0;
-}
-
-WebRtc_Word32 TraceImpl::AddMessage(
- char* traceMessage,
- const char msg[WEBRTC_TRACE_MAX_MESSAGE_SIZE],
- const WebRtc_UWord16 writtenSoFar) const
-
-{
- int length = 0;
- if(writtenSoFar >= WEBRTC_TRACE_MAX_MESSAGE_SIZE)
- {
- return -1;
- }
- // - 2 to leave room for newline and NULL termination
-#ifdef _WIN32
- length = _snprintf(traceMessage,
- WEBRTC_TRACE_MAX_MESSAGE_SIZE - writtenSoFar - 2,
- "%s",msg);
- if(length < 0)
- {
- length = WEBRTC_TRACE_MAX_MESSAGE_SIZE - writtenSoFar - 2;
- traceMessage[length] = 0;
- }
-#else
- length = snprintf(traceMessage,
- WEBRTC_TRACE_MAX_MESSAGE_SIZE-writtenSoFar-2, "%s",msg);
- if(length < 0 || length > WEBRTC_TRACE_MAX_MESSAGE_SIZE-writtenSoFar - 2)
- {
- length = WEBRTC_TRACE_MAX_MESSAGE_SIZE - writtenSoFar - 2;
- traceMessage[length] = 0;
- }
-#endif
- // Length with NULL termination.
- return length+1;
-}
-
-void TraceImpl::AddMessageToList(
- const char traceMessage[WEBRTC_TRACE_MAX_MESSAGE_SIZE],
- const WebRtc_UWord16 length,
- const TraceLevel level)
-{
- CriticalSectionScoped lock(_critsectArray);
-
- if(_nextFreeIdx[_activeQueue] >= WEBRTC_TRACE_MAX_QUEUE)
- {
- if( ! _traceFile.Open() &&
- !_callback)
- {
- // Keep at least the last 1/4 of old messages when not logging.
- // TODO (hellner): isn't this redundant. The user will make it known
- // when to start logging. Why keep messages before
- // that?
- for(int n = 0; n < WEBRTC_TRACE_MAX_QUEUE/4; n++)
- {
- const int lastQuarterOffset = (3*WEBRTC_TRACE_MAX_QUEUE/4);
- memcpy(_messageQueue[_activeQueue][n],
- _messageQueue[_activeQueue][n + lastQuarterOffset],
- WEBRTC_TRACE_MAX_MESSAGE_SIZE);
- }
- _nextFreeIdx[_activeQueue] = WEBRTC_TRACE_MAX_QUEUE/4;
- } else {
- // More messages are being written than there is room for in the
- // buffer. Drop any new messages.
- // TODO (hellner): its probably better to drop old messages instead
- // of new ones. One step further: if this happens
- // it's due to writing faster than what can be
- // processed. Maybe modify the filter at this point.
- // E.g. turn of STREAM.
- return;
- }
- }
-
- WebRtc_UWord16 idx = _nextFreeIdx[_activeQueue];
- _nextFreeIdx[_activeQueue]++;
-
- _level[_activeQueue][idx] = level;
- _length[_activeQueue][idx] = length;
- memcpy(_messageQueue[_activeQueue][idx], traceMessage, length);
-
- if(_nextFreeIdx[_activeQueue] == WEBRTC_TRACE_MAX_QUEUE-1)
- {
- // Logging more messages than can be worked off. Log a warning.
- const char warning_msg[] = "WARNING MISSING TRACE MESSAGES\n";
- _level[_activeQueue][_nextFreeIdx[_activeQueue]] = kTraceWarning;
- _length[_activeQueue][_nextFreeIdx[_activeQueue]] = strlen(warning_msg);
- memcpy(_messageQueue[_activeQueue][_nextFreeIdx[_activeQueue]],
- warning_msg, _length[_activeQueue][idx]);
- _nextFreeIdx[_activeQueue]++;
- }
-}
-
-bool TraceImpl::Run(void* obj)
-{
- return static_cast<TraceImpl*>(obj)->Process();
-}
-
-bool TraceImpl::Process()
-{
- if(_event.Wait(1000) == kEventSignaled)
- {
- if(_traceFile.Open() || _callback)
- {
- // File mode (not calback mode).
- WriteToFile();
- }
- } else {
- _traceFile.Flush();
- }
- return true;
-}
-
-void TraceImpl::WriteToFile()
-{
- WebRtc_UWord8 localQueueActive = 0;
- WebRtc_UWord16 localNextFreeIdx = 0;
-
- // There are two buffer. One for reading (for writing to file) and one for
- // writing (for storing new messages). Let new messages be posted to the
- // unused buffer so that the current buffer can be flushed safely.
- {
- CriticalSectionScoped lock(_critsectArray);
- localNextFreeIdx = _nextFreeIdx[_activeQueue];
- _nextFreeIdx[_activeQueue] = 0;
- localQueueActive = _activeQueue;
- if(_activeQueue == 0)
- {
- _activeQueue = 1;
- } else
- {
- _activeQueue = 0;
- }
- }
- if(localNextFreeIdx == 0)
- {
- return;
- }
-
- CriticalSectionScoped lock(_critsectInterface);
-
- for(WebRtc_UWord16 idx = 0; idx <localNextFreeIdx; idx++)
- {
- TraceLevel localLevel = _level[localQueueActive][idx];
- if(_callback)
- {
- _callback->Print(localLevel, _messageQueue[localQueueActive][idx],
- _length[localQueueActive][idx]);
- }
- if(_traceFile.Open())
- {
- if(_rowCountText > WEBRTC_TRACE_MAX_FILE_SIZE)
- {
- // wrap file
- _rowCountText = 0;
- _traceFile.Flush();
-
- if(_fileCountText == 0)
- {
- _traceFile.Rewind();
- } else
- {
- char oldFileName[FileWrapper::kMaxFileNameSize];
- char newFileName[FileWrapper::kMaxFileNameSize];
-
- // get current name
- _traceFile.FileName(oldFileName,
- FileWrapper::kMaxFileNameSize);
- _traceFile.CloseFile();
-
- _fileCountText++;
-
- UpdateFileName(oldFileName, newFileName, _fileCountText);
-
- if(_traceFile.OpenFile(newFileName, false, false,
- true) == -1)
- {
- return;
- }
- }
- }
- if(_rowCountText == 0)
- {
- char message[WEBRTC_TRACE_MAX_MESSAGE_SIZE + 1];
- WebRtc_Word32 length = AddDateTimeInfo(message);
- if(length != -1)
- {
- message[length] = 0;
- message[length-1] = '\n';
- _traceFile.Write(message, length);
- _rowCountText++;
- }
- length = AddBuildInfo(message);
- if(length != -1)
- {
- message[length+1] = 0;
- message[length] = '\n';
- message[length-1] = '\n';
- _traceFile.Write(message, length+1);
- _rowCountText++;
- _rowCountText++;
- }
- }
- WebRtc_UWord16 length = _length[localQueueActive][idx];
- _messageQueue[localQueueActive][idx][length] = 0;
- _messageQueue[localQueueActive][idx][length-1] = '\n';
- _traceFile.Write(_messageQueue[localQueueActive][idx], length);
- _rowCountText++;
- }
- }
-}
-
-void TraceImpl::AddImpl(const TraceLevel level, const TraceModule module,
- const WebRtc_Word32 id,
- const char msg[WEBRTC_TRACE_MAX_MESSAGE_SIZE])
-{
- if (TraceCheck(level))
- {
- char traceMessage[WEBRTC_TRACE_MAX_MESSAGE_SIZE];
- char* meassagePtr = traceMessage;
-
- WebRtc_Word32 len = 0;
- WebRtc_Word32 ackLen = 0;
-
- len = AddLevel(meassagePtr, level);
- if(len == -1)
- {
- return;
- }
- meassagePtr += len;
- ackLen += len;
-
- len = AddTime(meassagePtr, level);
- if(len == -1)
- {
- return;
- }
- meassagePtr += len;
- ackLen += len;
-
- len = AddModuleAndId(meassagePtr, module, id);
- if(len == -1)
- {
- return;
- }
- meassagePtr += len;
- ackLen += len;
-
- len = AddThreadId(meassagePtr);
- if(len < 0)
- {
- return;
- }
- meassagePtr += len;
- ackLen += len;
-
- len = AddMessage(meassagePtr, msg, (WebRtc_UWord16)ackLen);
- if(len == -1)
- {
- return;
- }
- ackLen += len;
- AddMessageToList(traceMessage,(WebRtc_UWord16)ackLen, level);
-
- // Make sure that messages are written as soon as possible.
- _event.Set();
- }
-}
-
-bool TraceImpl::TraceCheck(const TraceLevel level) const
-{
- return (level & levelFilter)? true:false;
-}
-
-bool TraceImpl::UpdateFileName(
- const char fileNameUTF8[FileWrapper::kMaxFileNameSize],
- char fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
- const WebRtc_UWord32 newCount) const
-{
- WebRtc_Word32 length = (WebRtc_Word32)strlen(fileNameUTF8);
- if(length < 0)
- {
- return false;
- }
-
- WebRtc_Word32 lengthWithoutFileEnding = length-1;
- while(lengthWithoutFileEnding > 0)
- {
- if(fileNameUTF8[lengthWithoutFileEnding] == '.')
- {
- break;
- } else {
- lengthWithoutFileEnding--;
- }
- }
- if(lengthWithoutFileEnding == 0)
- {
- lengthWithoutFileEnding = length;
- }
- WebRtc_Word32 lengthTo_ = lengthWithoutFileEnding - 1;
- while(lengthTo_ > 0)
- {
- if(fileNameUTF8[lengthTo_] == '_')
- {
- break;
- } else {
- lengthTo_--;
- }
- }
-
- memcpy(fileNameWithCounterUTF8, fileNameUTF8, lengthTo_);
- sprintf(fileNameWithCounterUTF8+lengthTo_, "_%lu%s",
- static_cast<long unsigned int> (newCount),
- fileNameUTF8+lengthWithoutFileEnding);
- return true;
-}
-
-bool TraceImpl::CreateFileName(
- const char fileNameUTF8[FileWrapper::kMaxFileNameSize],
- char fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
- const WebRtc_UWord32 newCount) const
-{
- WebRtc_Word32 length = (WebRtc_Word32)strlen(fileNameUTF8);
- if(length < 0)
- {
- return false;
- }
-
- WebRtc_Word32 lengthWithoutFileEnding = length-1;
- while(lengthWithoutFileEnding > 0)
- {
- if(fileNameUTF8[lengthWithoutFileEnding] == '.')
- {
- break;
- }else
- {
- lengthWithoutFileEnding--;
- }
- }
- if(lengthWithoutFileEnding == 0)
- {
- lengthWithoutFileEnding = length;
- }
- memcpy(fileNameWithCounterUTF8, fileNameUTF8, lengthWithoutFileEnding);
- sprintf(fileNameWithCounterUTF8+lengthWithoutFileEnding, "_%lu%s",
- static_cast<long unsigned int> (newCount),
- fileNameUTF8+lengthWithoutFileEnding);
- return true;
-}
-
-void Trace::CreateTrace()
-{
- TraceImpl::StaticInstance(kAddRef);
-}
-
-void Trace::ReturnTrace()
-{
- TraceImpl::StaticInstance(kRelease);
-}
-
-WebRtc_Word32 Trace::SetLevelFilter(WebRtc_UWord32 filter)
-{
- levelFilter = filter;
- return 0;
-}
-
-WebRtc_Word32 Trace::LevelFilter(WebRtc_UWord32& filter)
-{
- filter = levelFilter;
- return 0;
-}
-
-WebRtc_Word32 Trace::TraceFile(char fileName[FileWrapper::kMaxFileNameSize])
-{
- TraceImpl* trace = TraceImpl::GetTrace();
- if(trace)
- {
- int retVal = trace->TraceFileImpl(fileName);
- ReturnTrace();
- return retVal;
- }
- return -1;
-}
-
-WebRtc_Word32 Trace::SetTraceFile(const char* fileName,
- const bool addFileCounter)
-{
- TraceImpl* trace = TraceImpl::GetTrace();
- if(trace)
- {
- int retVal = trace->SetTraceFileImpl(fileName, addFileCounter);
- ReturnTrace();
- return retVal;
- }
- return -1;
-}
-
-WebRtc_Word32 Trace::SetTraceCallback(TraceCallback* callback)
-{
- TraceImpl* trace = TraceImpl::GetTrace();
- if(trace)
- {
- int retVal = trace->SetTraceCallbackImpl(callback);
- ReturnTrace();
- return retVal;
- }
- return -1;
-}
-
-void Trace::Add(const TraceLevel level, const TraceModule module,
- const WebRtc_Word32 id, const char* msg, ...)
-
-{
- TraceImpl* trace = TraceImpl::GetTrace(level);
- if(trace)
- {
- if(trace->TraceCheck(level))
- {
- char tempBuff[WEBRTC_TRACE_MAX_MESSAGE_SIZE];
- char* buff = 0;
- if(msg)
- {
- va_list args;
- va_start(args, msg);
-#ifdef _WIN32
- _vsnprintf(tempBuff,WEBRTC_TRACE_MAX_MESSAGE_SIZE-1,msg,args);
-#else
- vsnprintf(tempBuff,WEBRTC_TRACE_MAX_MESSAGE_SIZE-1,msg,args);
-#endif
- va_end(args);
- buff = tempBuff;
- }
- trace->AddImpl(level, module, id, buff);
- }
- ReturnTrace();
- }
-}
-
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/trace_impl.h b/trunk/src/system_wrappers/source/trace_impl.h
deleted file mode 100644
index 2b85813..0000000
--- a/trunk/src/system_wrappers/source/trace_impl.h
+++ /dev/null
@@ -1,132 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_IMPL_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_IMPL_H_
-
-#include "system_wrappers/interface/critical_section_wrapper.h"
-#include "system_wrappers/interface/event_wrapper.h"
-#include "system_wrappers/interface/file_wrapper.h"
-#include "system_wrappers/interface/static_instance.h"
-#include "system_wrappers/interface/trace.h"
-#include "system_wrappers/interface/thread_wrapper.h"
-
-namespace webrtc {
-
-// TODO (pwestin) WEBRTC_TRACE_MAX_QUEUE needs to be tweaked
-// TODO (hellner) the buffer should be close to how much the system can write to
-// file. Increasing the buffer will not solve anything. Sooner or
-// later the buffer is going to fill up anyways.
-#if defined(MAC_IPHONE)
- #define WEBRTC_TRACE_MAX_QUEUE 2000
-#else
- #define WEBRTC_TRACE_MAX_QUEUE 8000
-#endif
-#define WEBRTC_TRACE_NUM_ARRAY 2
-#define WEBRTC_TRACE_MAX_MESSAGE_SIZE 256
-// Total buffer size is WEBRTC_TRACE_NUM_ARRAY (number of buffer partitions) *
-// WEBRTC_TRACE_MAX_QUEUE (number of lines per buffer partition) *
-// WEBRTC_TRACE_MAX_MESSAGE_SIZE (number of 1 byte charachters per line) =
-// 1 or 4 Mbyte
-
-#define WEBRTC_TRACE_MAX_FILE_SIZE 100*1000
-// Number of rows that may be written to file. On average 110 bytes per row (max
-// 256 bytes per row). So on average 110*100*1000 = 11 Mbyte, max 256*100*1000 =
-// 25.6 Mbyte
-
-class TraceImpl : public Trace
-{
-public:
- virtual ~TraceImpl();
-
- static TraceImpl* CreateInstance();
- static TraceImpl* GetTrace(const TraceLevel level = kTraceAll);
-
- WebRtc_Word32 SetTraceFileImpl(const char* fileName,
- const bool addFileCounter);
- WebRtc_Word32 TraceFileImpl(
- char fileName[FileWrapper::kMaxFileNameSize]);
-
- WebRtc_Word32 SetTraceCallbackImpl(TraceCallback* callback);
-
- void AddImpl(const TraceLevel level, const TraceModule module,
- const WebRtc_Word32 id, const char* msg);
-
- bool StopThread();
-
- bool TraceCheck(const TraceLevel level) const;
-
-protected:
- TraceImpl();
-
- static TraceImpl* StaticInstance(CountOperation count_operation,
- const TraceLevel level = kTraceAll);
-
- WebRtc_Word32 AddThreadId(char* traceMessage) const;
-
- // OS specific implementations
- virtual WebRtc_Word32 AddTime(char* traceMessage,
- const TraceLevel level) const = 0;
-
- virtual WebRtc_Word32 AddBuildInfo(char* traceMessage) const = 0;
- virtual WebRtc_Word32 AddDateTimeInfo(char* traceMessage) const = 0;
-
- static bool Run(void* obj);
- bool Process();
-
-private:
- friend class Trace;
-
- WebRtc_Word32 AddLevel(char* szMessage, const TraceLevel level) const;
-
- WebRtc_Word32 AddModuleAndId(char* traceMessage, const TraceModule module,
- const WebRtc_Word32 id) const;
-
- WebRtc_Word32 AddMessage(char* traceMessage,
- const char msg[WEBRTC_TRACE_MAX_MESSAGE_SIZE],
- const WebRtc_UWord16 writtenSoFar) const;
-
- void AddMessageToList(
- const char traceMessage[WEBRTC_TRACE_MAX_MESSAGE_SIZE],
- const WebRtc_UWord16 length,
- const TraceLevel level);
-
- bool UpdateFileName(
- const char fileNameUTF8[FileWrapper::kMaxFileNameSize],
- char fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
- const WebRtc_UWord32 newCount) const;
-
- bool CreateFileName(
- const char fileNameUTF8[FileWrapper::kMaxFileNameSize],
- char fileNameWithCounterUTF8[FileWrapper::kMaxFileNameSize],
- const WebRtc_UWord32 newCount) const;
-
- void WriteToFile();
-
- CriticalSectionWrapper* _critsectInterface;
- TraceCallback* _callback;
- WebRtc_UWord32 _rowCountText;
- WebRtc_UWord32 _fileCountText;
-
- FileWrapper& _traceFile;
- ThreadWrapper& _thread;
- EventWrapper& _event;
-
- // _critsectArray protects _activeQueue
- CriticalSectionWrapper* _critsectArray;
- WebRtc_UWord16 _nextFreeIdx[WEBRTC_TRACE_NUM_ARRAY];
- TraceLevel _level[WEBRTC_TRACE_NUM_ARRAY][WEBRTC_TRACE_MAX_QUEUE];
- WebRtc_UWord16 _length[WEBRTC_TRACE_NUM_ARRAY][WEBRTC_TRACE_MAX_QUEUE];
- char* _messageQueue[WEBRTC_TRACE_NUM_ARRAY][WEBRTC_TRACE_MAX_QUEUE];
- WebRtc_UWord8 _activeQueue;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_IMPL_H_
diff --git a/trunk/src/system_wrappers/source/trace_impl_no_op.cc b/trunk/src/system_wrappers/source/trace_impl_no_op.cc
deleted file mode 100644
index 1752871..0000000
--- a/trunk/src/system_wrappers/source/trace_impl_no_op.cc
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "trace.h"
-
-namespace webrtc {
-
-void Trace::CreateTrace()
-{
-}
-
-void Trace::ReturnTrace()
-{
-}
-
-WebRtc_Word32 Trace::SetLevelFilter(WebRtc_UWord32 /*filter*/)
-{
- return 0;
-}
-
-WebRtc_Word32 Trace::LevelFilter(WebRtc_UWord32& /*filter*/)
-{
- return 0;
-}
-
-WebRtc_Word32 Trace::TraceFile(
- WebRtc_Word8 /*fileName*/[1024])
-{
- return -1;
-}
-
-WebRtc_Word32 Trace::SetTraceFile(const WebRtc_Word8* /*fileName*/,
- const bool /*addFileCounter*/)
-{
- return -1;
-}
-
-WebRtc_Word32 Trace::SetTraceCallback(TraceCallback* /*callback*/)
-{
- return -1;
-}
-
-void Trace::Add(const TraceLevel /*level*/, const TraceModule /*module*/,
- const WebRtc_Word32 /*id*/, const char* /*msg*/, ...)
-
-{
-}
-
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/trace_posix.cc b/trunk/src/system_wrappers/source/trace_posix.cc
deleted file mode 100644
index caf1128..0000000
--- a/trunk/src/system_wrappers/source/trace_posix.cc
+++ /dev/null
@@ -1,118 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "trace_posix.h"
-
-#include <cassert>
-#include <stdarg.h>
-#include <stdio.h>
-#include <string.h>
-#include <sys/time.h>
-#include <time.h>
-#ifdef WEBRTC_ANDROID
- #include <pthread.h>
-#else
- #include <iostream>
-#endif
-
-#if defined(_DEBUG)
- #define BUILDMODE "d"
-#elif defined(DEBUG)
- #define BUILDMODE "d"
-#elif defined(NDEBUG)
- #define BUILDMODE "r"
-#else
- #define BUILDMODE "?"
-#endif
-#define BUILDTIME __TIME__
-#define BUILDDATE __DATE__
-// example: "Oct 10 2002 12:05:30 r"
-#define BUILDINFO BUILDDATE " " BUILDTIME " " BUILDMODE
-
-namespace webrtc {
-TracePosix::TracePosix()
-{
- struct timeval systemTimeHighRes;
- gettimeofday(&systemTimeHighRes, 0);
- _prevAPITickCount = _prevTickCount = systemTimeHighRes.tv_sec;
-}
-
-TracePosix::~TracePosix()
-{
- StopThread();
-}
-
-WebRtc_Word32 TracePosix::AddTime(char* traceMessage,
- const TraceLevel level) const
-{
- struct timeval systemTimeHighRes;
- if (gettimeofday(&systemTimeHighRes, 0) == -1)
- {
- return -1;
- }
- const struct tm* systemTime =
- localtime(&systemTimeHighRes.tv_sec);
-
- const WebRtc_UWord32 ms_time = systemTimeHighRes.tv_usec / 1000;
- WebRtc_UWord32 prevTickCount = 0;
- if (level == kTraceApiCall)
- {
- prevTickCount = _prevTickCount;
- _prevTickCount = ms_time;
- } else {
- prevTickCount = _prevAPITickCount;
- _prevAPITickCount = ms_time;
- }
- WebRtc_UWord32 dwDeltaTime = ms_time - prevTickCount;
- if (prevTickCount == 0)
- {
- dwDeltaTime = 0;
- }
- if (dwDeltaTime > 0x0fffffff)
- {
- // Either wraparound or data race.
- dwDeltaTime = 0;
- }
- if(dwDeltaTime > 99999)
- {
- dwDeltaTime = 99999;
- }
-
- sprintf(traceMessage, "(%2u:%2u:%2u:%3u |%5lu) ", systemTime->tm_hour,
- systemTime->tm_min, systemTime->tm_sec, ms_time,
- static_cast<unsigned long>(dwDeltaTime));
- // Messages are 22 characters.
- return 22;
-}
-
-WebRtc_Word32 TracePosix::AddBuildInfo(char* traceMessage) const
-{
- sprintf(traceMessage, "Build info: %s", BUILDINFO);
- // Include NULL termination (hence + 1).
- return strlen(traceMessage) + 1;
-}
-
-WebRtc_Word32 TracePosix::AddDateTimeInfo(char* traceMessage) const
-{
- time_t t;
- time(&t);
- sprintf(traceMessage, "Local Date: %s", ctime(&t));
- WebRtc_Word32 len = static_cast<WebRtc_Word32>(strlen(traceMessage));
-
- if ('\n' == traceMessage[len - 1])
- {
- traceMessage[len - 1] = '\0';
- --len;
- }
-
- // Messages is 12 characters.
- return len + 1;
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/trace_posix.h b/trunk/src/system_wrappers/source/trace_posix.h
deleted file mode 100644
index 8c37cd2..0000000
--- a/trunk/src/system_wrappers/source/trace_posix.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_POSIX_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_POSIX_H_
-
-#include "critical_section_wrapper.h"
-#include "trace_impl.h"
-
-namespace webrtc {
-class TracePosix : public TraceImpl
-{
-public:
- TracePosix();
- virtual ~TracePosix();
-
- virtual WebRtc_Word32 AddTime(char* traceMessage,
- const TraceLevel level) const;
-
- virtual WebRtc_Word32 AddBuildInfo(char* traceMessage) const;
- virtual WebRtc_Word32 AddDateTimeInfo(char* traceMessage) const;
-
-private:
- volatile mutable WebRtc_UWord32 _prevAPITickCount;
- volatile mutable WebRtc_UWord32 _prevTickCount;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_POSIX_H_
diff --git a/trunk/src/system_wrappers/source/trace_unittest.cc b/trunk/src/system_wrappers/source/trace_unittest.cc
deleted file mode 100644
index 982e715..0000000
--- a/trunk/src/system_wrappers/source/trace_unittest.cc
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "system_wrappers/interface/trace.h"
-
-#include "gtest/gtest.h"
-#include "system_wrappers/source/cpu_measurement_harness.h"
-#include "testsupport/fileutils.h"
-
-using webrtc::CpuMeasurementHarness;
-using webrtc::Trace;
-using webrtc::kTraceWarning;
-using webrtc::kTraceUtility;
-
-class Logger : public webrtc::CpuTarget {
- public:
- Logger() {
- Trace::CreateTrace();
- std::string trace_file = webrtc::test::OutputPath() +
- "trace_unittest.txt";
- Trace::SetTraceFile(trace_file.c_str());
- Trace::SetLevelFilter(webrtc::kTraceAll);
- }
- virtual ~Logger() {
- Trace::ReturnTrace();
- }
-
- virtual bool DoWork() {
- // Use input paremeters to WEBRTC_TRACE that are not likely to be removed
- // in future code. E.g. warnings will likely be kept and this file is in
- // utility so it should use kTraceUtility.
- WEBRTC_TRACE(kTraceWarning, kTraceUtility, 0, "Log line");
- return true;
- }
-};
-
-// This test is disabled because it measures CPU usage. This is flaky because
-// the CPU usage for a machine may spike due to OS or other application.
-TEST(TraceTest, DISABLED_CpuUsage) {
- Logger logger;
- const int periodicity_ms = 1;
- const int iterations_per_period = 10;
- const int duration_ms = 1000;
- CpuMeasurementHarness* cpu_harness =
- CpuMeasurementHarness::Create(&logger, periodicity_ms,
- iterations_per_period, duration_ms);
- cpu_harness->Run();
- const int average_cpu = cpu_harness->AverageCpu();
- EXPECT_GE(5, average_cpu);
-}
diff --git a/trunk/src/system_wrappers/source/trace_win.cc b/trunk/src/system_wrappers/source/trace_win.cc
deleted file mode 100644
index db3da76..0000000
--- a/trunk/src/system_wrappers/source/trace_win.cc
+++ /dev/null
@@ -1,133 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "trace_win.h"
-
-#include <cassert>
-#include <stdarg.h>
-
-#include "Mmsystem.h"
-
-#if defined(_DEBUG)
- #define BUILDMODE "d"
-#elif defined(DEBUG)
- #define BUILDMODE "d"
-#elif defined(NDEBUG)
- #define BUILDMODE "r"
-#else
- #define BUILDMODE "?"
-#endif
-#define BUILDTIME __TIME__
-#define BUILDDATE __DATE__
-// Example: "Oct 10 2002 12:05:30 r"
-#define BUILDINFO BUILDDATE " " BUILDTIME " " BUILDMODE
-
-namespace webrtc {
-TraceWindows::TraceWindows()
- : _prevAPITickCount(0),
- _prevTickCount(0)
-{
-}
-
-TraceWindows::~TraceWindows()
-{
- StopThread();
-}
-
-WebRtc_Word32 TraceWindows::AddTime(char* traceMessage,
- const TraceLevel level) const
-{
- WebRtc_UWord32 dwCurrentTime = timeGetTime();
- SYSTEMTIME systemTime;
- GetSystemTime(&systemTime);
-
- if(level == kTraceApiCall)
- {
- WebRtc_UWord32 dwDeltaTime = dwCurrentTime- _prevTickCount;
- _prevTickCount = dwCurrentTime;
-
- if(_prevTickCount == 0)
- {
- dwDeltaTime = 0;
- }
- if(dwDeltaTime > 0x0fffffff)
- {
- // Either wraparound or data race.
- dwDeltaTime = 0;
- }
- if(dwDeltaTime > 99999)
- {
- dwDeltaTime = 99999;
- }
-
- sprintf (traceMessage, "(%2u:%2u:%2u:%3u |%5lu) ", systemTime.wHour,
- systemTime.wMinute, systemTime.wSecond,
- systemTime.wMilliseconds, dwDeltaTime);
- } else {
- WebRtc_UWord32 dwDeltaTime = dwCurrentTime - _prevAPITickCount;
- _prevAPITickCount = dwCurrentTime;
-
- if(_prevAPITickCount == 0)
- {
- dwDeltaTime = 0;
- }
- if(dwDeltaTime > 0x0fffffff)
- {
- // Either wraparound or data race.
- dwDeltaTime = 0;
- }
- if(dwDeltaTime > 99999)
- {
- dwDeltaTime = 99999;
- }
- sprintf (traceMessage, "(%2u:%2u:%2u:%3u |%5lu) ", systemTime.wHour,
- systemTime.wMinute, systemTime.wSecond,
- systemTime.wMilliseconds, dwDeltaTime);
- }
- // Messages is 12 characters.
- return 22;
-}
-
-WebRtc_Word32 TraceWindows::AddBuildInfo(char* traceMessage) const
-{
- // write data and time to text file
- sprintf(traceMessage, "Build info: %s", BUILDINFO);
- // Include NULL termination (hence + 1).
- return static_cast<WebRtc_Word32>(strlen(traceMessage)+1);
-}
-
-WebRtc_Word32 TraceWindows::AddDateTimeInfo(char* traceMessage) const
-{
- _prevAPITickCount = timeGetTime();
- _prevTickCount = _prevAPITickCount;
-
- SYSTEMTIME sysTime;
- GetLocalTime (&sysTime);
-
- TCHAR szDateStr[20];
- TCHAR szTimeStr[20];
- TCHAR *pSzDateStr = szDateStr;
- TCHAR *pSzTimeStr = szTimeStr;
-
- // Create date string (e.g. Apr 04 2002)
- GetDateFormat(LOCALE_SYSTEM_DEFAULT, 0, &sysTime, TEXT("MMM dd yyyy"),
- szDateStr, 20);
-
- // Create time string (e.g. 15:32:08)
- GetTimeFormat(LOCALE_SYSTEM_DEFAULT, 0, &sysTime, TEXT("HH':'mm':'ss"),
- szTimeStr, 20);
-
- sprintf(traceMessage, "Local Date: %s Local Time: %s", szDateStr,
- szTimeStr);
-
- // Include NULL termination (hence + 1).
- return static_cast<WebRtc_Word32>(strlen(traceMessage)+ 1);
-}
-} // namespace webrtc
diff --git a/trunk/src/system_wrappers/source/trace_win.h b/trunk/src/system_wrappers/source/trace_win.h
deleted file mode 100644
index 803198e..0000000
--- a/trunk/src/system_wrappers/source/trace_win.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_WINDOWS_H_
-#define WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_WINDOWS_H_
-
-#include "trace_impl.h"
-#include <stdio.h>
-#include <windows.h>
-
-namespace webrtc {
-class TraceWindows : public TraceImpl
-{
-public:
- TraceWindows();
- virtual ~TraceWindows();
-
- virtual WebRtc_Word32 AddTime(char* traceMessage,
- const TraceLevel level) const;
-
- virtual WebRtc_Word32 AddBuildInfo(char* traceMessage) const;
- virtual WebRtc_Word32 AddDateTimeInfo(char* traceMessage) const;
-private:
- volatile mutable WebRtc_UWord32 _prevAPITickCount;
- volatile mutable WebRtc_UWord32 _prevTickCount;
-};
-} // namespace webrtc
-
-#endif // WEBRTC_SYSTEM_WRAPPERS_SOURCE_TRACE_WINDOWS_H_
diff --git a/trunk/src/system_wrappers/test/TestSort/TestSort.cc b/trunk/src/system_wrappers/test/TestSort/TestSort.cc
deleted file mode 100644
index 6846a71..0000000
--- a/trunk/src/system_wrappers/test/TestSort/TestSort.cc
+++ /dev/null
@@ -1,265 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cstdio>
-#include <algorithm>
-#include <cstring>
-
-#include "sort.h"
-#include "tick_util.h"
-
-// Excellent work polluting the global namespace Visual Studio...
-#undef max
-#undef min
-#include <limits>
-
-template<typename KeyType>
-struct LotsOfData
-{
- KeyType key;
- char data[64];
-};
-
-template<typename DataType>
-int Compare(const void* dataX, const void* dataY)
-{
- DataType dataX = (DataType)*(const DataType*)dataX;
- DataType dataY = (DataType)*(const DataType*)dataY;
- if (dataX > dataY)
- {
- return 1;
- }
- else if (dataX < dataY)
- {
- return -1;
- }
-
- return 0;
-};
-
-template<typename DataType, typename KeyType>
-int CompareKey(const void* dataX, const void* dataY)
-{
- KeyType keyX = ((const DataType*)dataX)->key;
- KeyType keyY = ((const DataType*)dataY)->key;
- if (keyX > keyY)
- {
- return 1;
- }
- else if (keyX < keyY)
- {
- return -1;
- }
-
- return 0;
-}
-
-template<typename DataType>
-struct KeyLessThan
-{
- bool operator()(const DataType &dataX, const DataType &dataY) const
- {
- return dataX.key < dataY.key;
- }
-};
-
-const char* TypeEnumToString(webrtc::Type type)
-{
- switch (type)
- {
- using namespace webrtc;
- case TYPE_Word8:
- return "Word8";
- case TYPE_UWord8:
- return "UWord8";
- case TYPE_Word16:
- return "Word16";
- case TYPE_UWord16:
- return "UWord16";
- case TYPE_Word32:
- return "Word32";
- case TYPE_UWord32:
- return "UWord32";
- case TYPE_Word64:
- return "Word64";
- case TYPE_UWord64:
- return "UWord64";
- case TYPE_Float32:
- return "Float32";
- case TYPE_Float64:
- return "Float64";
- default:
- return "Unrecognized";
- }
-}
-
-template<typename Type>
-Type TypedRand()
-{
- if (std::numeric_limits<Type>::is_integer)
- {
- double floatRand = static_cast<double>(rand()) / RAND_MAX;
- if (std::numeric_limits<Type>::is_signed)
- {
- floatRand -= 0.5;
- }
-
- // Uniform [-max()/2, max()/2] for signed
- // [0, max()] for unsigned
- return static_cast<Type>(floatRand * std::numeric_limits<Type>::max());
- }
- else // Floating point
- {
- // Uniform [-0.5, 0.5]
- // The outer cast is to remove template warnings.
- return static_cast<Type>((static_cast<Type>(rand()) / RAND_MAX) - 0.5);
- }
-}
-
-template<typename KeyType>
-void RunSortTest(webrtc::Type sortType, bool keySort)
-{
- enum { DataLength = 1000 };
- enum { NumOfTests = 10000 };
- KeyType key[DataLength];
- KeyType keyRef[DataLength];
- LotsOfData<KeyType> data[DataLength];
- LotsOfData<KeyType> dataRef[DataLength];
- WebRtc_Word32 retVal = 0;
-
- if (keySort)
- {
- printf("Running %s KeySort() tests...\n", TypeEnumToString(sortType));
- }
- else
- {
- printf("Running %s Sort() tests...\n", TypeEnumToString(sortType));
- }
-
- TickInterval accTicks;
- for (int i = 0; i < NumOfTests; i++)
- {
- for (int j = 0; j < DataLength; j++)
- {
- key[j] = TypedRand<KeyType>();
- data[j].key = key[j];
- // Write index to payload. We use this later for verification.
- sprintf(data[j].data, "%d", j);
- }
-
- memcpy(dataRef, data, sizeof(data));
- memcpy(keyRef, key, sizeof(key));
-
- retVal = 0;
- TickTime t0 = TickTime::Now();
- if (keySort)
- {
- retVal = webrtc::KeySort(data, key, DataLength, sizeof(LotsOfData<KeyType>),
- sortType);
-
- //std::sort(data, data + DataLength, KeyLessThan<KeyType>());
- //qsort(data, DataLength, sizeof(LotsOfData<KeyType>),
- // CompareKey<LotsOfData<KeyType>, KeyType>);
- }
- else
- {
- retVal = webrtc::Sort(key, DataLength, sortType);
-
- //std::sort(key, key + DataLength);
- //qsort(key, DataLength, sizeof(KeyType), Compare<KeyType>);
- }
- TickTime t1 = TickTime::Now();
- accTicks += (t1 - t0);
-
- if (retVal != 0)
- {
- printf("Test failed at iteration %d:\n", i);
- printf("Sort returned an error. ");
- printf("It likely does not support the requested type\nExiting...\n");
- exit(0);
- }
-
- // Reference sort.
- if (!keySort)
- {
- std::sort(keyRef, keyRef + DataLength);
- }
-
- if (keySort)
- {
- for (int j = 0; j < DataLength - 1; j++)
- {
- if (data[j].key > data[j + 1].key)
- {
- printf("Test failed at iteration %d:\n", i);
- printf("Keys are not monotonically increasing\nExiting...\n");
- exit(0);
- }
-
- int index = atoi(data[j].data);
- if (index < 0 || index >= DataLength || data[j].key != dataRef[index].key)
- {
- printf("Test failed at iteration %d:\n", i);
- printf("Payload data is corrupt\nExiting...\n");
- exit(0);
- }
- }
- }
- else
- {
- for (int j = 0; j < DataLength - 1; j++)
- {
- if (key[j] > key[j + 1])
- {
- printf("Test failed at iteration %d:\n", i);
- printf("Data is not monotonically increasing\nExiting...\n");
- exit(0);
- }
- }
-
- if (memcmp(key, keyRef, sizeof(key)) != 0)
- {
- printf("Test failed at iteration %d:\n", i);
- printf("Sort data differs from std::sort reference\nExiting...\n");
- exit(0);
- }
- }
- }
-
- printf("Compliance test passed over %d iterations\n", NumOfTests);
-
- WebRtc_Word64 executeTime = accTicks.Milliseconds();
- printf("Execute time: %.2f s\n\n", (float)executeTime / 1000);
-}
-
-int main()
-{
- // Seed rand().
- srand(42);
- bool keySort = false;
- for (int i = 0; i < 2; i++) {
- RunSortTest<WebRtc_Word8>(webrtc::TYPE_Word8, keySort);
- RunSortTest<WebRtc_UWord8>(webrtc::TYPE_UWord8, keySort);
- RunSortTest<WebRtc_Word16>(webrtc::TYPE_Word16, keySort);
- RunSortTest<WebRtc_UWord16>(webrtc::TYPE_UWord16, keySort);
- RunSortTest<WebRtc_Word32>(webrtc::TYPE_Word32, keySort);
- RunSortTest<WebRtc_UWord32>(webrtc::TYPE_UWord32, keySort);
- RunSortTest<WebRtc_Word64>(webrtc::TYPE_Word64, keySort);
- RunSortTest<WebRtc_UWord64>(webrtc::TYPE_UWord64, keySort);
- RunSortTest<float>(webrtc::TYPE_Float32, keySort);
- RunSortTest<double>(webrtc::TYPE_Float64, keySort);
-
- keySort = !keySort;
- }
-
- printf("All tests passed\n");
-
- return 0;
-}
diff --git a/trunk/src/system_wrappers/test/list/list.cc b/trunk/src/system_wrappers/test/list/list.cc
deleted file mode 100644
index 5c4f0b9..0000000
--- a/trunk/src/system_wrappers/test/list/list.cc
+++ /dev/null
@@ -1,174 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "list_wrapper.h"
-
-const int kNumberOfElements = 10;
-
-void FailTest(bool failed)
-{
- if (failed)
- {
- printf("Test failed!\n");
- printf("Press enter to continue:");
- getchar();
- exit(0);
- }
-}
-
-int GetStoredIntegerValue(ListItem* list_item)
-{
- void* list_item_pointer = list_item->GetItem();
- if (list_item_pointer != NULL)
- {
- return *(reinterpret_cast<int*>(list_item_pointer));
- }
- return static_cast<int>(list_item->GetUnsignedItem());
-}
-
-void PrintList(ListWrapper& list)
-{
- ListItem* list_item = list.First();
- printf("List: ");
- while (list_item != NULL)
- {
- int item_value = GetStoredIntegerValue(list_item);
- FailTest(item_value < 0);
- printf(" %d",item_value);
- list_item = list.Next(list_item);
- }
- printf("\n");
-}
-
-// The list should always be in ascending order
-void ListSanity(ListWrapper& list)
-{
- if(list.Empty())
- {
- return;
- }
- ListItem* item_iter = list.First();
- // Fake a previous value for the first iteration
- int previous_value = GetStoredIntegerValue(item_iter) - 1;
- while (item_iter != NULL)
- {
- const int value = GetStoredIntegerValue(item_iter);
- FailTest(value != previous_value + 1);
- previous_value = value;
- item_iter = list.Next(item_iter);
- }
-}
-
-int main(int /*argc*/, char* /*argv*/[])
-{
- printf("List Test:\n");
- int element_array[kNumberOfElements];
- for (int i = 0; i < kNumberOfElements; i++)
- {
- element_array[i] = i;
- }
- // Test PushBack 1
- ListWrapper test_list;
- for (int i = 2; i < kNumberOfElements - 2; i++)
- {
- FailTest(test_list.PushBack((void*)&element_array[i]) != 0);
- }
- // Test PushBack 2
- FailTest(test_list.PushBack(element_array[kNumberOfElements - 2]) != 0);
- FailTest(test_list.PushBack(element_array[kNumberOfElements - 1]) != 0);
- // Test PushFront 2
- FailTest(test_list.PushFront(element_array[1]) != 0);
- // Test PushFront 1
- FailTest(test_list.PushFront((void*)&element_array[0]) != 0);
- // Test GetSize
- FailTest(test_list.GetSize() != kNumberOfElements);
- PrintList(test_list);
- //Test PopFront
- FailTest(test_list.PopFront() != 0);
- //Test PopBack
- FailTest(test_list.PopBack() != 0);
- // Test GetSize
- FailTest(test_list.GetSize() != kNumberOfElements - 2);
- // Test Empty
- FailTest(test_list.Empty());
- // Test First
- ListItem* first_item = test_list.First();
- FailTest(first_item == NULL);
- // Test Last
- ListItem* last_item = test_list.Last();
- FailTest(last_item == NULL);
- // Test Next
- ListItem* second_item = test_list.Next(first_item);
- FailTest(second_item == NULL);
- FailTest(test_list.Next(last_item) != NULL);
- FailTest(test_list.Next(NULL) != NULL);
- // Test Previous
- ListItem* second_to_last_item = test_list.Previous(last_item);
- FailTest(second_to_last_item == NULL);
- FailTest(test_list.Previous(first_item) != NULL);
- FailTest(test_list.Previous(NULL) != NULL);
- // Test GetUnsignedItem
- FailTest(last_item->GetUnsignedItem() !=
- kNumberOfElements - 2);
- FailTest(last_item->GetItem() !=
- NULL);
- // Test GetItem
- FailTest(GetStoredIntegerValue(second_to_last_item) !=
- kNumberOfElements - 3);
- FailTest(second_to_last_item->GetUnsignedItem() != 0);
- // Pop last and first since they are pushed as unsigned items.
- FailTest(test_list.PopFront() != 0);
- FailTest(test_list.PopBack() != 0);
- // Test Insert. Please note that old iterators are no longer valid at
- // this point.
- ListItem* insert_item_last = new ListItem(reinterpret_cast<void*>(&element_array[kNumberOfElements - 2]));
- FailTest(test_list.Insert(test_list.Last(),insert_item_last) != 0);
- FailTest(test_list.Insert(NULL,insert_item_last) == 0);
- ListItem* insert_item_last2 = new ListItem(reinterpret_cast<void*>(&element_array[kNumberOfElements - 2]));
- FailTest(test_list.Insert(insert_item_last2,NULL) == 0);
- // test InsertBefore
- ListItem* insert_item_first = new ListItem(reinterpret_cast<void*>(&element_array[1]));
- FailTest(test_list.InsertBefore(test_list.First(),insert_item_first) != 0);
- FailTest(test_list.InsertBefore(NULL,insert_item_first) == 0);
- ListItem* insert_item_first2 = new ListItem(reinterpret_cast<void*>(&element_array[1]));
- FailTest(test_list.InsertBefore(insert_item_first2,NULL) == 0);
- PrintList(test_list);
- ListSanity(test_list);
- // Erase the whole list
- int counter = 0;
- while (test_list.PopFront() == 0)
- {
- FailTest(counter++ > kNumberOfElements);
- }
- PrintList(test_list);
- // Test APIs when list is empty
- FailTest(test_list.GetSize() != 0);
- FailTest(test_list.PopFront() != -1);
- FailTest(test_list.PopBack() != -1);
- FailTest(!test_list.Empty());
- FailTest(test_list.First() != NULL);
- FailTest(test_list.Last() != NULL);
- FailTest(test_list.Next(NULL) != NULL);
- FailTest(test_list.Previous(NULL) != NULL);
- FailTest(test_list.Erase(NULL) != -1);
- // Test Insert APIs when list is empty
- ListItem* new_item = new ListItem(reinterpret_cast<void*>(&element_array[0]));
- FailTest(test_list.Insert(NULL,new_item) != 0);
- FailTest(test_list.Empty());
- FailTest(test_list.PopFront() != 0);
- ListItem* new_item2 = new ListItem(reinterpret_cast<void*>(&element_array[0]));
- FailTest(test_list.InsertBefore(NULL,new_item2) != 0);
- FailTest(test_list.Empty());
-
- printf("Tests passed successfully!\n");
-}
diff --git a/trunk/src/system_wrappers/test/map/map.cc b/trunk/src/system_wrappers/test/map/map.cc
deleted file mode 100644
index 8a0d3e3..0000000
--- a/trunk/src/system_wrappers/test/map/map.cc
+++ /dev/null
@@ -1,112 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-
-#include "map_wrapper.h"
-
-const int kNumberOfElements = 10;
-
-void FailTest(bool failed)
-{
- if (failed)
- {
- printf("Test failed!\n");
- printf("Press enter to continue:");
- getchar();
- exit(0);
- }
-}
-
-int GetStoredIntegerValue(MapItem* map_item)
-{
- void* map_item_pointer = map_item->GetItem();
- if (map_item_pointer != NULL)
- {
- return *(reinterpret_cast<int*>(map_item_pointer));
- }
- return static_cast<int>(map_item->GetUnsignedId());
-}
-
-void PrintMap(MapWrapper& map)
-{
- MapItem* map_item = map.First();
- printf("Map: ");
- while (map_item != NULL)
- {
- int item_value = GetStoredIntegerValue(map_item);
- FailTest(item_value < 0);
- printf(" %d",item_value);
- map_item = map.Next(map_item);
- }
- printf("\n");
-}
-
-int main(int /*argc*/, char* /*argv*/[])
-{
- int element_array[kNumberOfElements];
- for (int i = 0; i < kNumberOfElements; i++)
- {
- element_array[i] = i;
- }
- // Test insert
- MapWrapper test_map;
- for (int i = 0; i < kNumberOfElements; i++)
- {
- test_map.Insert(i,(void*)&element_array[i]);
- }
- // Test Erase1
- MapItem* remove_item = test_map.Find(2);
- FailTest(remove_item == NULL);
- FailTest(test_map.Erase(remove_item) != 0);
- FailTest(test_map.Find(2) != NULL);
- remove_item = NULL;
- FailTest(test_map.Erase(remove_item) != -1);
- // Test Erase2
- FailTest(test_map.Erase(1) != 0);
- FailTest(test_map.Find(1) != NULL);
- FailTest(test_map.Erase(1) != -1);
- // Test Size
- FailTest(test_map.Size() != kNumberOfElements - 2);
- PrintMap(test_map);
- // Test First
- MapItem* first_item = test_map.First();
- FailTest(first_item == NULL);
- FailTest(GetStoredIntegerValue(first_item) != 0);
- // Test Last
- MapItem* last_item = test_map.Last();
- FailTest(last_item == NULL);
- FailTest(GetStoredIntegerValue(last_item) != 9);
- // Test Next
- MapItem* second_item = test_map.Next(first_item);
- FailTest(second_item == NULL);
- FailTest(GetStoredIntegerValue(second_item) != 3);
- FailTest(test_map.Next(last_item) != NULL);
- // Test Previous
- MapItem* second_to_last_item = test_map.Previous(last_item);
- FailTest(second_to_last_item == NULL);
- FailTest(GetStoredIntegerValue(second_to_last_item) != 8);
- FailTest(test_map.Previous(first_item) != NULL);
- // Test Find (only improper usage untested)
- FailTest(test_map.Find(kNumberOfElements + 2) != NULL);
- // Test GetId
- FailTest(*(reinterpret_cast<int*>(second_to_last_item->GetItem())) !=
- second_to_last_item->GetId());
- FailTest(second_to_last_item->GetUnsignedId() !=
- static_cast<unsigned int>(second_to_last_item->GetId()));
- // Test SetItem
- int swapped_item = kNumberOfElements;
- last_item->SetItem(reinterpret_cast<void*>(&swapped_item));
- FailTest(GetStoredIntegerValue(last_item) !=
- swapped_item);
-
- printf("Tests passed successfully!\n");
-}
diff --git a/trunk/src/typedefs.h b/trunk/src/typedefs.h
deleted file mode 100644
index ba87309..0000000
--- a/trunk/src/typedefs.h
+++ /dev/null
@@ -1,142 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This file contains platform-specific typedefs and defines.
-
-#ifndef WEBRTC_TYPEDEFS_H_
-#define WEBRTC_TYPEDEFS_H_
-
-// Reserved words definitions
-// TODO(andrew): Look at removing these.
-#define WEBRTC_EXTERN extern
-#define G_CONST const
-#define WEBRTC_INLINE extern __inline
-
-// Define WebRTC preprocessor identifiers based on the current build platform.
-// TODO(andrew): Clean these up. We can probably remove everything in this
-// block.
-// - TARGET_MAC_INTEL and TARGET_MAC aren't used anywhere.
-// - In the few places where TARGET_PC is used, it should be replaced by
-// something more specific.
-// - Do we really support PowerPC? Probably not. Remove WEBRTC_MAC_INTEL
-// from build/common.gypi as well.
-#if defined(WIN32)
- // Windows & Windows Mobile.
- #if !defined(WEBRTC_TARGET_PC)
- #define WEBRTC_TARGET_PC
- #endif
-#elif defined(__APPLE__)
- // Mac OS X.
- #if defined(__LITTLE_ENDIAN__ )
- #if !defined(WEBRTC_TARGET_MAC_INTEL)
- #define WEBRTC_TARGET_MAC_INTEL
- #endif
- #else
- #if !defined(WEBRTC_TARGET_MAC)
- #define WEBRTC_TARGET_MAC
- #endif
- #endif
-#else
- // Linux etc.
- #if !defined(WEBRTC_TARGET_PC)
- #define WEBRTC_TARGET_PC
- #endif
-#endif
-
-// Derived from Chromium's build/build_config.h
-// Processor architecture detection. For more info on what's defined, see:
-// http://msdn.microsoft.com/en-us/library/b0084kay.aspx
-// http://www.agner.org/optimize/calling_conventions.pdf
-// or with gcc, run: "echo | gcc -E -dM -"
-// TODO(andrew): replace WEBRTC_LITTLE_ENDIAN with WEBRTC_ARCH_LITTLE_ENDIAN?
-#if defined(_M_X64) || defined(__x86_64__)
-#define WEBRTC_ARCH_X86_FAMILY
-#define WEBRTC_ARCH_X86_64
-#define WEBRTC_ARCH_64_BITS
-#define WEBRTC_ARCH_LITTLE_ENDIAN
-#elif defined(_M_IX86) || defined(__i386__)
-#define WEBRTC_ARCH_X86_FAMILY
-#define WEBRTC_ARCH_X86
-#define WEBRTC_ARCH_32_BITS
-#define WEBRTC_ARCH_LITTLE_ENDIAN
-#elif defined(__ARMEL__)
-// TODO(andrew): We'd prefer to control platform defines here, but this is
-// currently provided by the Android makefiles. Commented to avoid duplicate
-// definition warnings.
-//#define WEBRTC_ARCH_ARM
-// TODO(andrew): Chromium uses the following two defines. Should we switch?
-//#define WEBRTC_ARCH_ARM_FAMILY
-//#define WEBRTC_ARCH_ARMEL
-#define WEBRTC_ARCH_32_BITS
-#define WEBRTC_ARCH_LITTLE_ENDIAN
-#else
-#error Please add support for your architecture in typedefs.h
-#endif
-
-#if defined(__SSE2__) || defined(_MSC_VER)
-#define WEBRTC_USE_SSE2
-#endif
-
-#if defined(WEBRTC_TARGET_PC)
-
-#if !defined(_MSC_VER)
- #include <stdint.h>
-#else
- // Define C99 equivalent types.
- // Since MSVC doesn't include these headers, we have to write our own
- // version to provide a compatibility layer between MSVC and the WebRTC
- // headers.
- typedef signed char int8_t;
- typedef signed short int16_t;
- typedef signed int int32_t;
- typedef signed long long int64_t;
- typedef unsigned char uint8_t;
- typedef unsigned short uint16_t;
- typedef unsigned int uint32_t;
- typedef unsigned long long uint64_t;
-#endif
-
-#if defined(WIN32)
- typedef __int64 WebRtc_Word64;
- typedef unsigned __int64 WebRtc_UWord64;
-#else
- typedef int64_t WebRtc_Word64;
- typedef uint64_t WebRtc_UWord64;
-#endif
- typedef int32_t WebRtc_Word32;
- typedef uint32_t WebRtc_UWord32;
- typedef int16_t WebRtc_Word16;
- typedef uint16_t WebRtc_UWord16;
- typedef char WebRtc_Word8;
- typedef uint8_t WebRtc_UWord8;
-
- // Define endian for the platform
- #define WEBRTC_LITTLE_ENDIAN
-
-#elif defined(WEBRTC_TARGET_MAC_INTEL)
- #include <stdint.h>
-
- typedef int64_t WebRtc_Word64;
- typedef uint64_t WebRtc_UWord64;
- typedef int32_t WebRtc_Word32;
- typedef uint32_t WebRtc_UWord32;
- typedef int16_t WebRtc_Word16;
- typedef char WebRtc_Word8;
- typedef uint16_t WebRtc_UWord16;
- typedef uint8_t WebRtc_UWord8;
-
- // Define endian for the platform
- #define WEBRTC_LITTLE_ENDIAN
-
-#else
- #error "No platform defined for WebRTC type definitions (typedefs.h)"
-#endif
-
-#endif // WEBRTC_TYPEDEFS_H_
diff --git a/trunk/src/video_engine/Android.mk b/trunk/src/video_engine/Android.mk
deleted file mode 100644
index 164c289..0000000
--- a/trunk/src/video_engine/Android.mk
+++ /dev/null
@@ -1,94 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE := libwebrtc_vie_core
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := \
- vie_base_impl.cc \
- vie_capture_impl.cc \
- vie_codec_impl.cc \
- vie_encryption_impl.cc \
- vie_external_codec_impl.cc \
- vie_file_impl.cc \
- vie_image_process_impl.cc \
- vie_impl.cc \
- vie_network_impl.cc \
- vie_ref_count.cc \
- vie_render_impl.cc \
- vie_rtp_rtcp_impl.cc \
- vie_shared_data.cc \
- vie_capturer.cc \
- vie_channel.cc \
- vie_channel_manager.cc \
- vie_encoder.cc \
- vie_file_image.cc \
- vie_file_player.cc \
- vie_file_recorder.cc \
- vie_frame_provider_base.cc \
- vie_input_manager.cc \
- vie_manager_base.cc \
- vie_performance_monitor.cc \
- vie_receiver.cc \
- vie_remb.cc \
- vie_renderer.cc \
- vie_render_manager.cc \
- vie_sender.cc \
- vie_sync_module.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS)
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/include\
- $(LOCAL_PATH)/.. \
- $(LOCAL_PATH)/../common_video/interface \
- $(LOCAL_PATH)/../common_video/jpeg/main/interface \
- $(LOCAL_PATH)/../common_video/vplib/main/interface \
- $(LOCAL_PATH)/../modules/interface \
- $(LOCAL_PATH)/../modules/audio_coding/main/interface \
- $(LOCAL_PATH)/../modules/media_file/interface \
- $(LOCAL_PATH)/../modules/rtp_rtcp/interface \
- $(LOCAL_PATH)/../modules/udp_transport/interface \
- $(LOCAL_PATH)/../modules/utility/interface \
- $(LOCAL_PATH)/../modules/video_capture/main/interface \
- $(LOCAL_PATH)/../modules/video_capture/main/source \
- $(LOCAL_PATH)/../modules/video_capture/main/source/Android \
- $(LOCAL_PATH)/../modules/video_coding/codecs/interface \
- $(LOCAL_PATH)/../modules/video_coding/main/interface \
- $(LOCAL_PATH)/../modules/video_mixer/main/interface \
- $(LOCAL_PATH)/../modules/video_processing/main/interface \
- $(LOCAL_PATH)/../modules/video_render/main/interface \
- $(LOCAL_PATH)/../system_wrappers/interface \
- $(LOCAL_PATH)/../voice_engine/main/interface
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
-LOCAL_LDLIBS += -ldl -lpthread
-endif
-
-ifneq ($(TARGET_SIMULATOR),true)
-LOCAL_SHARED_LIBRARIES += libdl
-endif
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/video_engine/OWNERS b/trunk/src/video_engine/OWNERS
deleted file mode 100644
index ac607bd..0000000
--- a/trunk/src/video_engine/OWNERS
+++ /dev/null
@@ -1,4 +0,0 @@
-mflodman@webrtc.org
-perkj@webrtc.org
-wu@webrtc.org
-mallinath@webrtc.org
diff --git a/trunk/src/video_engine/include/vie_base.h b/trunk/src/video_engine/include/vie_base.h
deleted file mode 100644
index 2757a78..0000000
--- a/trunk/src/video_engine/include/vie_base.h
+++ /dev/null
@@ -1,137 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-// - Creating and deleting VideoEngine instances.
-// - Creating and deleting channels.
-// - Connect a video channel with a corresponding voice channel for audio/video
-// synchronization.
-// - Start and stop sending and receiving.
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_BASE_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_BASE_H_
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VoiceEngine;
-
-// Class used for all callbacks from ViEBase.
-class WEBRTC_DLLEXPORT ViEBaseObserver {
- public:
- // This method will be called periodically if the average system CPU usage
- // exceeds 75%.
- virtual void PerformanceAlarm(const unsigned int cpu_load) = 0;
-
- protected:
- virtual ~ViEBaseObserver() {}
-};
-
-class WEBRTC_DLLEXPORT VideoEngine {
- public:
- // Creates a VideoEngine object, which can then be used to acquire sub‐APIs.
- static VideoEngine* Create();
-
- // Deletes a VideoEngine instance.
- static bool Delete(VideoEngine*& video_engine);
-
- // Specifies the amount and type of trace information, which will be created
- // by the VideoEngine.
- static int SetTraceFilter(const unsigned int filter);
-
- // Sets the name of the trace file and enables non‐encrypted trace messages.
- static int SetTraceFile(const char* file_nameUTF8,
- const bool add_file_counter = false);
-
- // Installs the TraceCallback implementation to ensure that the VideoEngine
- // user receives callbacks for generated trace messages.
- static int SetTraceCallback(TraceCallback* callback);
-
- // Android specific.
- // Provides VideoEngine with pointers to objects supplied by the Java
- // applications JNI interface.
- static int SetAndroidObjects(void* java_vm, void* java_context);
-
- protected:
- VideoEngine() {}
- virtual ~VideoEngine() {}
-};
-
-class WEBRTC_DLLEXPORT ViEBase {
- public:
- // Factory for the ViEBase sub‐API and increases an internal reference
- // counter if successful. Returns NULL if the API is not supported or if
- // construction fails.
- static ViEBase* GetInterface(VideoEngine* video_engine);
-
- // Releases the ViEBase sub-API and decreases an internal reference counter.
- // Returns the new reference count. This value should be zero
- // for all sub-API:s before the VideoEngine object can be safely deleted.
- virtual int Release() = 0;
-
- // Initiates all common parts of the VideoEngine.
- virtual int Init() = 0;
-
- // Connects a VideoEngine instance to a VoiceEngine instance for audio video
- // synchronization.
- virtual int SetVoiceEngine(VoiceEngine* voice_engine) = 0;
-
- // Creates a new channel, either with a new encoder instance or by sharing
- // encoder instance with an already created channel.
- virtual int CreateChannel(int& video_channel) = 0;
- virtual int CreateChannel(int& video_channel, int original_channel) = 0;
-
- // Deletes an existing channel and releases the utilized resources.
- virtual int DeleteChannel(const int video_channel) = 0;
-
- // Specifies the VoiceEngine and VideoEngine channel pair to use for
- // audio/video synchronization.
- virtual int ConnectAudioChannel(const int video_channel,
- const int audio_channel) = 0;
-
- // Disconnects a previously paired VideoEngine and VoiceEngine channel pair.
- virtual int DisconnectAudioChannel(const int video_channel) = 0;
-
- // Starts sending packets to an already specified IP address and port number
- // for a specified channel.
- virtual int StartSend(const int video_channel) = 0;
-
- // Stops packets from being sent for a specified channel.
- virtual int StopSend(const int video_channel) = 0;
-
- // Prepares VideoEngine for receiving packets on the specified channel.
- virtual int StartReceive(const int video_channel) = 0;
-
- // Stops receiving incoming RTP and RTCP packets on the specified channel.
- virtual int StopReceive(const int video_channel) = 0;
-
- // Registers an instance of a user implementation of the ViEBase
- // observer.
- virtual int RegisterObserver(ViEBaseObserver& observer) = 0;
-
- // Removes an already registered instance of ViEBaseObserver.
- virtual int DeregisterObserver() = 0;
-
- // Retrieves the version information for VideoEngine and its components.
- virtual int GetVersion(char version[1024]) = 0;
-
- // Returns the last VideoEngine error code.
- virtual int LastError() = 0;
-
- protected:
- ViEBase() {}
- virtual ~ViEBase() {}
-};
-
-} // namespace webrtc
-
-#endif // #define WEBRTC_VIDEO_ENGINE_MAIN_INTERFACE_VIE_BASE_H_
diff --git a/trunk/src/video_engine/include/vie_capture.h b/trunk/src/video_engine/include/vie_capture.h
deleted file mode 100644
index 45d62e5..0000000
--- a/trunk/src/video_engine/include/vie_capture.h
+++ /dev/null
@@ -1,255 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-// - Allocating capture devices.
-// - Connect a capture device with one or more channels.
-// - Start and stop capture devices.
-// - Getting capture device capabilities.
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CAPTURE_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CAPTURE_H_
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VideoEngine;
-class VideoCaptureModule;
-
-// This structure describes one set of the supported capabilities for a capture
-// device.
-struct CaptureCapability {
- unsigned int width;
- unsigned int height;
- unsigned int maxFPS;
- RawVideoType rawType;
- VideoCodecType codecType;
- unsigned int expectedCaptureDelay;
- bool interlaced;
- CaptureCapability() {
- width = 0;
- height = 0;
- maxFPS = 0;
- rawType = kVideoI420;
- codecType = kVideoCodecUnknown;
- expectedCaptureDelay = 0;
- interlaced = false;
- }
-};
-
-// This enumerator tells the current brightness alarm mode.
-enum Brightness {
- Normal = 0,
- Bright = 1,
- Dark = 2
-};
-
-// This enumerator describes the capture alarm mode.
-enum CaptureAlarm {
- AlarmRaised = 0,
- AlarmCleared = 1
-};
-
-enum RotateCapturedFrame {
- RotateCapturedFrame_0 = 0,
- RotateCapturedFrame_90 = 90,
- RotateCapturedFrame_180 = 180,
- RotateCapturedFrame_270 = 270
-};
-
-struct ViEVideoFrameI420 {
- ViEVideoFrameI420() {
- y_plane = NULL;
- u_plane = NULL;
- v_plane = NULL;
- y_pitch = 0;
- u_pitch = 0;
- v_pitch = 0;
- width = 0;
- height = 0;
- }
-
- unsigned char* y_plane;
- unsigned char* u_plane;
- unsigned char* v_plane;
-
- int y_pitch;
- int u_pitch;
- int v_pitch;
-
- unsigned short width;
- unsigned short height;
-};
-
-// This class declares an abstract interface to be used when implementing
-// a user-defined capture device. This interface is not meant to be
-// implemented by the user. Instead, the user should call AllocateCaptureDevice
-// in the ViECapture interface, which will create a suitable implementation.
-// The user should then call IncomingFrame in this interface to deliver
-// captured frames to the system.
-class WEBRTC_DLLEXPORT ViEExternalCapture {
- public:
- ViEExternalCapture() {}
- virtual ~ViEExternalCapture() {}
-
- // This method is called by the user to deliver a new captured frame to
- // VideoEngine.
- virtual int IncomingFrame(unsigned char* video_frame,
- unsigned int video_frame_length,
- unsigned short width,
- unsigned short height,
- RawVideoType video_type,
- unsigned long long capture_time = 0) = 0;
-
- // This method is specifically for delivering a new captured I420 frame to
- // VideoEngine.
- virtual int IncomingFrameI420(
- const ViEVideoFrameI420& video_frame,
- unsigned long long capture_time = 0) = 0;
-};
-
-// This class declares an abstract interface for a user defined observer. It is
-// up to the VideoEngine user to implement a derived class which implements the
-// observer class. The observer is registered using RegisterObserver() and
-// deregistered using DeregisterObserver().
-class WEBRTC_DLLEXPORT ViECaptureObserver {
- public:
- // This method is called if a bright or dark captured image is detected.
- virtual void BrightnessAlarm(const int capture_id,
- const Brightness brightness) = 0;
-
- // This method is called periodically telling the capture device frame rate.
- virtual void CapturedFrameRate(const int capture_id,
- const unsigned char frame_rate) = 0;
-
- // This method is called if the capture device stops delivering images to
- // VideoEngine.
- virtual void NoPictureAlarm(const int capture_id,
- const CaptureAlarm alarm) = 0;
-
- protected:
- virtual ~ViECaptureObserver() {}
-};
-
-class WEBRTC_DLLEXPORT ViECapture {
- public:
- // Factory for the ViECapture sub‐API and increases an internal reference
- // counter if successful. Returns NULL if the API is not supported or if
- // construction fails.
- static ViECapture* GetInterface(VideoEngine* video_engine);
-
- // Releases the ViECapture sub-API and decreases an internal reference
- // counter.
- // Returns the new reference count. This value should be zero
- // for all sub-API:s before the VideoEngine object can be safely deleted.
- virtual int Release() = 0;
-
- // Gets the number of available capture devices.
- virtual int NumberOfCaptureDevices() = 0;
-
- // Gets the name and unique id of a capture device.
- virtual int GetCaptureDevice(unsigned int list_number,
- char* device_nameUTF8,
- const unsigned int device_nameUTF8Length,
- char* unique_idUTF8,
- const unsigned int unique_idUTF8Length) = 0;
-
- // Allocates a capture device to be used in VideoEngine.
- virtual int AllocateCaptureDevice(const char* unique_idUTF8,
- const unsigned int unique_idUTF8Length,
- int& capture_id) = 0;
-
- // Registers an external capture device to be used in VideoEngine
- virtual int AllocateExternalCaptureDevice(
- int& capture_id,
- ViEExternalCapture *&external_capture) = 0;
-
- // Use capture device using external capture module.
- virtual int AllocateCaptureDevice(VideoCaptureModule& capture_module,
- int& capture_id) = 0;
-
- // Releases a capture device and makes it available for other applications.
- virtual int ReleaseCaptureDevice(const int capture_id) = 0;
-
- // This function connects a capture device with a channel. Multiple channels
- // can be connected to the same capture device.
- virtual int ConnectCaptureDevice(const int capture_id,
- const int video_channel) = 0;
-
- // Disconnects a capture device as input for a specified channel.
- virtual int DisconnectCaptureDevice(const int video_channel) = 0;
-
- // Makes a capture device start capturing video frames.
- virtual int StartCapture(
- const int capture_id,
- const CaptureCapability& capture_capability = CaptureCapability()) = 0;
-
- // Stops a started capture device from capturing video frames.
- virtual int StopCapture(const int capture_id) = 0;
-
- // Rotates captured frames before encoding and sending.
- // Used on mobile devices with rotates cameras.
- virtual int SetRotateCapturedFrames(const int capture_id,
- const RotateCapturedFrame rotation) = 0;
-
- // This function sets the expected delay from when a video frame is captured
- // to when that frame is delivered to VideoEngine.
- virtual int SetCaptureDelay(const int capture_id,
- const unsigned int capture_delay_ms) = 0;
-
- // Returns the number of sets of capture capabilities the capture device
- // supports.
- virtual int NumberOfCapabilities(
- const char* unique_id_utf8,
- const unsigned int unique_id_utf8_length) = 0;
-
- // Gets a set of capture capabilities for a specified capture device.
- virtual int GetCaptureCapability(const char* unique_id_utf8,
- const unsigned int unique_id_utf8_length,
- const unsigned int capability_number,
- CaptureCapability& capability) = 0;
-
- // Displays the capture device property dialog box for the specified capture
- // device. Windows only.
- virtual int ShowCaptureSettingsDialogBox(
- const char* unique_idUTF8,
- const unsigned int unique_id_utf8_length,
- const char* dialog_title,
- void* parent_window = NULL,
- const unsigned int x = 200,
- const unsigned int y = 200) = 0;
-
- // Gets the clockwise angle the frames from the camera must be rotated in
- // order to display the frames correctly if the display is rotated in its
- // natural orientation.
- virtual int GetOrientation(const char* unique_id_utf8,
- RotateCapturedFrame& orientation) = 0;
-
- // Enables brightness alarm detection and the brightness alarm callback.
- virtual int EnableBrightnessAlarm(const int capture_id,
- const bool enable) = 0;
-
- // Registers an instance of a user implementation of the ViECaptureObserver.
- virtual int RegisterObserver(const int capture_id,
- ViECaptureObserver& observer) = 0;
-
- // Removes an already registered instance of ViECaptureObserver.
- virtual int DeregisterObserver(const int capture_id) = 0;
-
- protected:
- ViECapture() {}
- virtual ~ViECapture() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CAPTURE_H_
diff --git a/trunk/src/video_engine/include/vie_codec.h b/trunk/src/video_engine/include/vie_codec.h
deleted file mode 100644
index 0535038..0000000
--- a/trunk/src/video_engine/include/vie_codec.h
+++ /dev/null
@@ -1,176 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-// - Setting send and receive codecs.
-// - Codec specific settings.
-// - Key frame signaling.
-// - Stream management settings.
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CODEC_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CODEC_H_
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VideoEngine;
-struct VideoCodec;
-
-// This class declares an abstract interface for a user defined observer. It is
-// up to the VideoEngine user to implement a derived class which implements the
-// observer class. The observer is registered using RegisterEncoderObserver()
-// and deregistered using DeregisterEncoderObserver().
-class WEBRTC_DLLEXPORT ViEEncoderObserver {
- public:
- // This method is called once per second with the current encoded frame rate
- // and bit rate.
- virtual void OutgoingRate(const int video_channel,
- const unsigned int framerate,
- const unsigned int bitrate) = 0;
- protected:
- virtual ~ViEEncoderObserver() {}
-};
-
-// This class declares an abstract interface for a user defined observer. It is
-// up to the VideoEngine user to implement a derived class which implements the
-// observer class. The observer is registered using RegisterDecoderObserver()
-// and deregistered using DeregisterDecoderObserver().
-class WEBRTC_DLLEXPORT ViEDecoderObserver {
- public:
- // This method is called when a new incoming stream is detected, normally
- // triggered by a new incoming SSRC or payload type.
- virtual void IncomingCodecChanged(const int video_channel,
- const VideoCodec& video_codec) = 0;
-
- // This method is called once per second containing the frame rate and bit
- // rate for the incoming stream
- virtual void IncomingRate(const int video_channel,
- const unsigned int framerate,
- const unsigned int bitrate) = 0;
-
- // This method is called when the decoder needs a new key frame from encoder
- // on the sender.
- virtual void RequestNewKeyFrame(const int video_channel) = 0;
-
- protected:
- virtual ~ViEDecoderObserver() {}
-};
-
-class WEBRTC_DLLEXPORT ViECodec {
- public:
- // Factory for the ViECodec sub‐API and increases an internal reference
- // counter if successful. Returns NULL if the API is not supported or if
- // construction fails.
- static ViECodec* GetInterface(VideoEngine* video_engine);
-
- // Releases the ViECodec sub-API and decreases an internal reference
- // counter.
- // Returns the new reference count. This value should be zero
- // for all sub-API:s before the VideoEngine object can be safely deleted.
- virtual int Release() = 0;
-
- // Gets the number of available codecs for the VideoEngine build.
- virtual int NumberOfCodecs() const = 0;
-
- // Gets a VideoCodec struct for a codec containing the default configuration
- // for that codec type.
- virtual int GetCodec(const unsigned char list_number,
- VideoCodec& video_codec) const = 0;
-
- // Sets the send codec to use for a specified channel.
- virtual int SetSendCodec(const int video_channel,
- const VideoCodec& video_codec) = 0;
-
- // Gets the current send codec settings.
- virtual int GetSendCodec(const int video_channel,
- VideoCodec& video_codec) const = 0;
-
- // Prepares VideoEngine to receive a certain codec type and setting for a
- // specified payload type.
- virtual int SetReceiveCodec(const int video_channel,
- const VideoCodec& video_codec) = 0;
-
- // Gets the current receive codec.
- virtual int GetReceiveCodec(const int video_channel,
- VideoCodec& video_codec) const = 0;
-
- // This function is used to get codec configuration parameters to be
- // signaled from the encoder to the decoder in the call setup.
- virtual int GetCodecConfigParameters(
- const int video_channel,
- unsigned char config_parameters[kConfigParameterSize],
- unsigned char& config_parameters_size) const = 0;
-
- // Enables advanced scaling of the captured video stream if the stream
- // differs from the send codec settings.
- virtual int SetImageScaleStatus(const int video_channel,
- const bool enable) = 0;
-
- // Gets the number of sent key frames and number of sent delta frames.
- virtual int GetSendCodecStastistics(const int video_channel,
- unsigned int& key_frames,
- unsigned int& delta_frames) const = 0;
-
- // Gets the number of decoded key frames and number of decoded delta frames.
- virtual int GetReceiveCodecStastistics(const int video_channel,
- unsigned int& key_frames,
- unsigned int& delta_frames) const = 0;
-
- // Gets the bitrate targeted by the video codec rate control in kbit/s.
- virtual int GetCodecTargetBitrate(const int video_channel,
- unsigned int* bitrate) const = 0;
-
- // Gets the number of packets discarded by the jitter buffer because they
- // arrived too late.
- virtual unsigned int GetDiscardedPackets(const int video_channel) const = 0;
-
- // Enables key frame request callback in ViEDecoderObserver.
- virtual int SetKeyFrameRequestCallbackStatus(const int video_channel,
- const bool enable) = 0;
-
- // Enables key frame requests for detected lost packets.
- virtual int SetSignalKeyPacketLossStatus(
- const int video_channel,
- const bool enable,
- const bool only_key_frames = false) = 0;
-
- // Registers an instance of a user implementation of the ViEEncoderObserver.
- virtual int RegisterEncoderObserver(const int video_channel,
- ViEEncoderObserver& observer) = 0;
-
- // Removes an already registered instance of ViEEncoderObserver.
- virtual int DeregisterEncoderObserver(const int video_channel) = 0;
-
- // Registers an instance of a user implementation of the ViEDecoderObserver.
- virtual int RegisterDecoderObserver(const int video_channel,
- ViEDecoderObserver& observer) = 0;
-
- // Removes an already registered instance of ViEDecoderObserver.
- virtual int DeregisterDecoderObserver(const int video_channel) = 0;
-
- // This function forces the next encoded frame to be a key frame. This is
- // normally used when the remote endpoint only supports out‐band key frame
- // request.
- virtual int SendKeyFrame(const int video_channel) = 0;
-
- // This function makes the decoder wait for a key frame before starting to
- // decode the incoming video stream.
- virtual int WaitForFirstKeyFrame(const int video_channel,
- const bool wait) = 0;
-
- protected:
- ViECodec() {}
- virtual ~ViECodec() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_CODEC_H_
diff --git a/trunk/src/video_engine/include/vie_encryption.h b/trunk/src/video_engine/include/vie_encryption.h
deleted file mode 100644
index 8e7c955..0000000
--- a/trunk/src/video_engine/include/vie_encryption.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-// - External encryption and decryption.
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_ENCRYPTION_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_ENCRYPTION_H_
-
-#include "common_types.h"
-
-namespace webrtc {
-class VideoEngine;
-
-class WEBRTC_DLLEXPORT ViEEncryption {
- public:
- // Factory for the ViEEncryption sub‐API and increases an internal reference
- // counter if successful. Returns NULL if the API is not supported or if
- // construction fails.
- static ViEEncryption* GetInterface(VideoEngine* video_engine);
-
- // Releases the ViEEncryption sub-API and decreases an internal reference
- // counter.
- // Returns the new reference count. This value should be zero
- // for all sub-API:s before the VideoEngine object can be safely deleted.
- virtual int Release() = 0;
-
- // This function registers a encryption derived instance and enables
- // external encryption for the specified channel.
- virtual int RegisterExternalEncryption(const int video_channel,
- Encryption& encryption) = 0;
-
- // This function deregisters a registered encryption derived instance
- // and disables external encryption.
- virtual int DeregisterExternalEncryption(const int video_channel) = 0;
-
- protected:
- ViEEncryption() {}
- virtual ~ViEEncryption() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_ENCRYPTION_H_
diff --git a/trunk/src/video_engine/include/vie_errors.h b/trunk/src/video_engine/include/vie_errors.h
deleted file mode 100644
index 63100b5..0000000
--- a/trunk/src/video_engine/include/vie_errors.h
+++ /dev/null
@@ -1,119 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_ERRORS_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_ERRORS_H_
-
-enum ViEErrors {
- // ViEBase.
- kViENotInitialized = 12000, // Init has not been called successfully.
- kViEBaseVoEFailure, // SetVoiceEngine. ViE failed to use VE instance. Check VE instance pointer.ConnectAudioChannel failed to set voice channel. Have SetVoiceEngine been called? Is the voice channel correct.
- kViEBaseChannelCreationFailed, // CreateChannel.
- kViEBaseInvalidChannelId, // The channel does not exist.
- kViEAPIDoesNotExist, // Release called on Interface that has not been created.
- kViEBaseInvalidArgument,
- kViEBaseAlreadySending, // StartSend called on channel that is already sending.
- kViEBaseNotSending, // StopSend called on channel that is not sending.
- kViEBaseAlreadyReceiving, // StartReceive called on channel that is already receiving.
- kViEBaseObserverAlreadyRegistered, // RegisterObserver- an observer has already been set.
- kViEBaseObserverNotRegistered, // DeregisterObserver - no observer has been registered.
- kViEBaseUnknownError, // An unknown error has occurred. Check the log file.
-
- // ViECodec.
- kViECodecInvalidArgument = 12100, // Wrong input parameter to function.
- kViECodecObserverAlreadyRegistered, // RegisterEncoderObserver, RegisterDecoderObserver.
- kViECodecObserverNotRegistered, // DeregisterEncoderObserver, DeregisterDecoderObserver.
- kViECodecInvalidCodec, // SetSendCodec,SetReceiveCodec- The codec structure is invalid.
- kViECodecInvalidChannelId, // The channel does not exist.
- kViECodecInUse, // SetSendCodec- Can't change codec size or type when multiple channels use the same encoder.
- kViECodecUnknownError, // An unknown error has occurred. Check the log file.
-
- // ViERender.
- kViERenderInvalidRenderId = 12200, // No renderer with the ID exist. In AddRenderer - The render ID is invalid. No capture device, channel or file is allocated with that id.
- kViERenderAlreadyExists, // AddRenderer: the renderer already exist.
- kViERenderInvalidFrameFormat, // AddRender (external renderer). The user has requested a frame format that we don't support.
- kViERenderUnknownError, // An unknown error has occurred. Check the log file.
-
- // ViECapture.
- kViECaptureDeviceAlreadyConnected = 12300, // ConnectCaptureDevice - A capture device has already been connected to this video channel.
- kViECaptureDeviceDoesNotExist, // No capture device exist with the provided capture id or unique name.
- kViECaptureDeviceInvalidChannelId, // ConnectCaptureDevice, DisconnectCaptureDevice- No Channel exist with the provided channel id.
- kViECaptureDeviceNotConnected, // DisconnectCaptureDevice- No capture device is connected to the channel.
- kViECaptureDeviceNotStarted, // Stop- The capture device is not started.
- kViECaptureDeviceAlreadyStarted, // Start- The capture device is already started.
- kViECaptureDeviceAlreadyAllocated, // AllocateCaptureDevice The device is already allocated.
- kViECaptureDeviceMaxNoDevicesAllocated, // AllocateCaptureDevice Max number of devices already allocated.
- kViECaptureObserverAlreadyRegistered, // RegisterObserver- An observer is already registered. Need to deregister first.
- kViECaptureDeviceObserverNotRegistered, // DeregisterObserver- No observer is registered.
- kViECaptureDeviceUnknownError, // An unknown error has occurred. Check the log file.
- kViECaptureDeviceMacQtkitNotSupported, // QTKit handles the capture devices automatically. Thus querying capture capabilities is not supported.
-
- // ViEFile.
- kViEFileInvalidChannelId = 12400, // No Channel exist with the provided channel id.
- kViEFileInvalidArgument, // Incorrect input argument
- kViEFileAlreadyRecording, // StartRecordOutgoingVideo - already recording channel
- kViEFileVoENotSet, // StartRecordOutgoingVideo. Failed to access voice engine. Has SetVoiceEngine been called?
- kViEFileNotRecording, // StopRecordOutgoingVideo
- kViEFileMaxNoOfFilesOpened, // StartPlayFile
- kViEFileNotPlaying, // StopPlayFile. The file with the provided id is not playing.
- kViEFileObserverAlreadyRegistered, // RegisterObserver
- kViEFileObserverNotRegistered, // DeregisterObserver
- kViEFileInputAlreadyConnected, // SendFileOnChannel- the video channel already have a connected input.
- kViEFileNotConnected, // StopSendFileOnChannel- No file is being sent on the channel.
- kViEFileVoEFailure, // SendFileOnChannel,StartPlayAudioLocally - failed to play audio stream
- kViEFileInvalidRenderId, // SetRenderTimeoutImage and SetRenderStartImage: Renderer with the provided render id does not exist.
- kViEFileInvalidFile, // Can't open the file with provided filename. Is the path and file format correct?
- kViEFileInvalidCapture, // Can't use ViEPicture. Is the object correct?
- kViEFileSetRenderTimeoutError, // SetRenderTimeoutImage- Please see log file.
- kViEFileInvalidCaptureId, // SetCaptureDeviceImage capture id does not exist.
- kViEFileSetCaptureImageError, // SetCaptureDeviceImage error. Please see log file.
- kViEFileSetStartImageError, // SetRenderStartImage error. Please see log file.
- kViEFileUnknownError, // An unknown error has occurred. Check the log file.
-
- // ViENetwork.
- kViENetworkInvalidChannelId = 12500, // No Channel exist with the provided channel id.
- kViENetworkAlreadyReceiving, // SetLocalReceiver: Can not change ports while receiving.
- kViENetworkLocalReceiverNotSet, // GetLocalReceiver: SetLocalReceiver not called.
- kViENetworkAlreadySending, // SetSendDestination
- kViENetworkDestinationNotSet, // GetSendDestination
- kViENetworkInvalidArgument, // GetLocalIP- Check function arguments.
- kViENetworkSendCodecNotSet, // SetSendGQoS- Need to set the send codec first.
- kViENetworkServiceTypeNotSupported, // SetSendGQoS
- kViENetworkNotSupported, // SetSendGQoS Not supported on this OS.
- kViENetworkObserverAlreadyRegistered, // RegisterObserver
- kViENetworkObserverNotRegistered, // SetPeriodicDeadOrAliveStatus - Need to call RegisterObserver first, DeregisterObserver if no observer is registered.
- kViENetworkUnknownError, // An unknown error has occurred. Check the log file.
-
- // ViERTP_RTCP.
- kViERtpRtcpInvalidChannelId = 12600, // No Channel exist with the provided channel id.
- kViERtpRtcpAlreadySending, // The channel is already sending. Need to stop send before calling this API.
- kViERtpRtcpNotSending, // The channel needs to be sending in order for this function to work.
- kViERtpRtcpRtcpDisabled, // Functions failed because RTCP is disabled.
- kViERtpRtcpObserverAlreadyRegistered, // An observer is already registered. Need to deregister the old first.
- kViERtpRtcpObserverNotRegistered, // No observer registered.
- kViERtpRtcpUnknownError, // An unknown error has occurred. Check the log file.
-
- // ViEEncryption.
- kViEEncryptionInvalidChannelId = 12700, // Channel id does not exist.
- kViEEncryptionInvalidSrtpParameter, // EnableSRTPSend, EnableSRTPReceive- Check the SRTP parameters.
- kViEEncryptionSrtpNotSupported, // This build does not support SRTP.
- kViEEncryptionUnknownError, // An unknown error has occurred. Check the log file.
-
- // ViEImageProcess.
- kViEImageProcessInvalidChannelId = 12800, // No Channel exist with the provided channel id.
- kViEImageProcessInvalidCaptureId, // No capture device exist with the provided capture id.
- kViEImageProcessFilterExists, // RegisterCaptureEffectFilter,RegisterSendEffectFilter,RegisterRenderEffectFilter - Effect filter already registered.
- kViEImageProcessFilterDoesNotExist, // DeRegisterCaptureEffectFilter,DeRegisterSendEffectFilter,DeRegisterRenderEffectFilter - Effect filter not registered.
- kViEImageProcessAlreadyEnabled, // EnableDeflickering,EnableDenoising,EnableColorEnhancement- Function already enabled.
- kViEImageProcessAlreadyDisabled, // EnableDeflickering,EnableDenoising,EnableColorEnhancement- Function already disabled.
- kViEImageProcessUnknownError // An unknown error has occurred. Check the log file.
-};
-
-#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_ERRORS_H_
diff --git a/trunk/src/video_engine/include/vie_external_codec.h b/trunk/src/video_engine/include/vie_external_codec.h
deleted file mode 100644
index a2e686a..0000000
--- a/trunk/src/video_engine/include/vie_external_codec.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_EXTERNAL_CODEC_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_EXTERNAL_CODEC_H_
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VideoDecoder;
-class VideoEncoder;
-class VideoEngine;
-
-class WEBRTC_DLLEXPORT ViEExternalCodec {
- public:
- static ViEExternalCodec* GetInterface(VideoEngine* video_engine);
-
- virtual int Release() = 0;
-
- virtual int RegisterExternalSendCodec(const int video_channel,
- const unsigned char pl_type,
- VideoEncoder* encoder) = 0;
-
- virtual int DeRegisterExternalSendCodec(const int video_channel,
- const unsigned char pl_type) = 0;
-
- virtual int RegisterExternalReceiveCodec(const int video_channel,
- const unsigned int pl_type,
- VideoDecoder* decoder,
- bool decoder_render = false,
- int render_delay = 0) = 0;
-
- virtual int DeRegisterExternalReceiveCodec(const int video_channel,
- const unsigned char pl_type) = 0;
-
- protected:
- ViEExternalCodec() {}
- virtual ~ViEExternalCodec() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_EXTERNAL_CODEC_H_
diff --git a/trunk/src/video_engine/include/vie_file.h b/trunk/src/video_engine/include/vie_file.h
deleted file mode 100644
index fbc3d2b..0000000
--- a/trunk/src/video_engine/include/vie_file.h
+++ /dev/null
@@ -1,221 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-// - File recording and playing.
-// - Snapshots.
-// - Background images.
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_FILE_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_FILE_H_
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VideoEngine;
-struct VideoCodec;
-
-// This structure contains picture data and describes the picture type.
-struct ViEPicture {
- unsigned char* data;
- unsigned int size;
- unsigned int width;
- unsigned int height;
- RawVideoType type;
-
- ViEPicture() {
- data = NULL;
- size = 0;
- width = 0;
- height = 0;
- type = kVideoI420;
- }
-
- // Call FreePicture to free data.
- ~ViEPicture() {
- data = NULL;
- size = 0;
- width = 0;
- height = 0;
- type = kVideoUnknown;
- }
-};
-
-// This enumerator tells which audio source to use for media files.
-enum AudioSource {
- NO_AUDIO,
- MICROPHONE,
- PLAYOUT,
- VOICECALL
-};
-
-// This class declares an abstract interface for a user defined observer. It is
-// up to the VideoEngine user to implement a derived class which implements the
-// observer class. The observer is registered using RegisterObserver() and
-// deregistered using DeregisterObserver().
-class WEBRTC_DLLEXPORT ViEFileObserver {
- public:
- // This method is called when the end is reached of a played file.
- virtual void PlayFileEnded(const WebRtc_Word32 file_id) = 0;
-
- protected:
- virtual ~ViEFileObserver() {}
-};
-
-class WEBRTC_DLLEXPORT ViEFile {
- public:
- // Factory for the ViEFile sub‐API and increases an internal reference
- // counter if successful. Returns NULL if the API is not supported or if
- // construction fails.
- static ViEFile* GetInterface(VideoEngine* video_engine);
-
- // Releases the ViEFile sub-API and decreases an internal reference counter.
- // Returns the new reference count. This value should be zero
- // for all sub-API:s before the VideoEngine object can be safely deleted.
- virtual int Release() = 0;
-
- // Starts playing a video file.
- virtual int StartPlayFile(
- const char* file_name_utf8,
- int& file_id,
- const bool loop = false,
- const FileFormats file_format = kFileFormatAviFile) = 0;
-
- // Stops a file from being played.
- virtual int StopPlayFile(const int file_id) = 0;
-
- // Registers an instance of a user implementation of the ViEFileObserver.
- virtual int RegisterObserver(int file_id, ViEFileObserver& observer) = 0;
-
- // Removes an already registered instance of ViEFileObserver.
- virtual int DeregisterObserver(int file_id, ViEFileObserver& observer) = 0;
-
- // This function tells which channel, if any, the file should be sent on.
- virtual int SendFileOnChannel(const int file_id, const int video_channel) = 0;
-
- // Stops a file from being sent on a a channel.
- virtual int StopSendFileOnChannel(const int video_channel) = 0;
-
- // Starts playing the file audio as microphone input for the specified voice
- // channel.
- virtual int StartPlayFileAsMicrophone(const int file_id,
- const int audio_channel,
- bool mix_microphone = false,
- float volume_scaling = 1) = 0;
-
- // The function stop the audio from being played on a VoiceEngine channel.
- virtual int StopPlayFileAsMicrophone(const int file_id,
- const int audio_channel) = 0;
-
- // The function plays and mixes the file audio with the local speaker signal
- // for playout.
- virtual int StartPlayAudioLocally(const int file_id, const int audio_channel,
- float volume_scaling = 1) = 0;
-
- // Stops the audio from a file from being played locally.
- virtual int StopPlayAudioLocally(const int file_id,
- const int audio_channel) = 0;
-
- // This function starts recording the video transmitted to another endpoint.
- virtual int StartRecordOutgoingVideo(
- const int video_channel,
- const char* file_name_utf8,
- AudioSource audio_source,
- const CodecInst& audio_codec,
- const VideoCodec& video_codec,
- const FileFormats file_format = kFileFormatAviFile) = 0;
-
- // This function starts recording the incoming video stream on a channel.
- virtual int StartRecordIncomingVideo(
- const int video_channel,
- const char* file_name_utf8,
- AudioSource audio_source,
- const CodecInst& audio_codec,
- const VideoCodec& video_codec,
- const FileFormats file_format = kFileFormatAviFile) = 0;
-
- // Stops the file recording of the outgoing stream.
- virtual int StopRecordOutgoingVideo(const int video_channel) = 0;
-
- // Stops the file recording of the incoming stream.
- virtual int StopRecordIncomingVideo(const int video_channel) = 0;
-
- // Gets the audio codec, video codec and file format of a recorded file.
- virtual int GetFileInformation(
- const char* file_name,
- VideoCodec& video_codec,
- CodecInst& audio_codec,
- const FileFormats file_format = kFileFormatAviFile) = 0;
-
- // The function takes a snapshot of the last rendered image for a video
- // channel.
- virtual int GetRenderSnapshot(const int video_channel,
- const char* file_name_utf8) = 0;
-
- // The function takes a snapshot of the last rendered image for a video
- // channel
- virtual int GetRenderSnapshot(const int video_channel,
- ViEPicture& picture) = 0;
-
- // The function takes a snapshot of the last captured image by a specified
- // capture device.
- virtual int GetCaptureDeviceSnapshot(const int capture_id,
- const char* file_name_utf8) = 0;
-
- // The function takes a snapshot of the last captured image by a specified
- // capture device.
- virtual int GetCaptureDeviceSnapshot(const int capture_id,
- ViEPicture& picture) = 0;
-
- // This function sets a jpg image to show before the first frame is captured
- // by the capture device. This frame will be encoded and transmitted to a
- // possible receiver
- virtual int SetCaptureDeviceImage(const int capture_id,
- const char* file_name_utf8) = 0;
-
- // This function sets an image to show before the first frame is captured by
- // the capture device. This frame will be encoded and transmitted to a
- // possible receiver
- virtual int SetCaptureDeviceImage(const int capture_id,
- const ViEPicture& picture) = 0;
-
- virtual int FreePicture(ViEPicture& picture) = 0;
-
- // This function sets a jpg image to render before the first received video
- // frame is decoded for a specified channel.
- virtual int SetRenderStartImage(const int video_channel,
- const char* file_name_utf8) = 0;
-
- // This function sets an image to render before the first received video
- // frame is decoded for a specified channel.
- virtual int SetRenderStartImage(const int video_channel,
- const ViEPicture& picture) = 0;
-
- // This function sets a jpg image to render if no frame is decoded for a
- // specified time interval.
- virtual int SetRenderTimeoutImage(const int video_channel,
- const char* file_name_utf8,
- const unsigned int timeout_ms = 1000) = 0;
-
- // This function sets an image to render if no frame is decoded for a
- // specified time interval.
- virtual int SetRenderTimeoutImage(const int video_channel,
- const ViEPicture& picture,
- const unsigned int timeout_ms) = 0;
-
- protected:
- ViEFile() {}
- virtual ~ViEFile() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_FILE_H_
diff --git a/trunk/src/video_engine/include/vie_image_process.h b/trunk/src/video_engine/include/vie_image_process.h
deleted file mode 100644
index dfad08d..0000000
--- a/trunk/src/video_engine/include/vie_image_process.h
+++ /dev/null
@@ -1,101 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-// - Effect filters
-// - Deflickering
-// - Denoising
-// - Color enhancement
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_IMAGE_PROCESS_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_IMAGE_PROCESS_H_
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VideoEngine;
-
-// This class declares an abstract interface for a user defined effect filter.
-// The effect filter is registered using RegisterCaptureEffectFilter(),
-// RegisterSendEffectFilter() or RegisterRenderEffectFilter() and deregistered
-// with the corresponding deregister function.
-class WEBRTC_DLLEXPORT ViEEffectFilter {
- public:
- // This method is called with an I420 video frame allowing the user to
- // modify the video frame.
- virtual int Transform(int size,
- unsigned char* frame_buffer,
- unsigned int time_stamp90KHz,
- unsigned int width,
- unsigned int height) = 0;
- protected:
- ViEEffectFilter() {}
- virtual ~ViEEffectFilter() {}
-};
-
-class WEBRTC_DLLEXPORT ViEImageProcess {
- public:
- // Factory for the ViEImageProcess sub‐API and increases an internal
- // reference counter if successful. Returns NULL if the API is not supported
- // or if construction fails.
- static ViEImageProcess* GetInterface(VideoEngine* video_engine);
-
- // Releases the ViEImageProcess sub-API and decreases an internal reference
- // counter. Returns the new reference count. This value should be zero
- // for all sub-API:s before the VideoEngine object can be safely deleted.
- virtual int Release() = 0;
-
- // This function registers a EffectFilter to use for a specified capture
- // device.
- virtual int RegisterCaptureEffectFilter(const int capture_id,
- ViEEffectFilter& capture_filter) = 0;
-
- // This function deregisters a EffectFilter for a specified capture device.
- virtual int DeregisterCaptureEffectFilter(const int capture_id) = 0;
-
- // This function registers an EffectFilter to use for a specified channel.
- virtual int RegisterSendEffectFilter(const int video_channel,
- ViEEffectFilter& send_filter) = 0;
-
- // This function deregisters a send effect filter for a specified channel.
- virtual int DeregisterSendEffectFilter(const int video_channel) = 0;
-
- // This function registers a EffectFilter to use for the rendered video
- // stream on an incoming channel.
- virtual int RegisterRenderEffectFilter(const int video_channel,
- ViEEffectFilter& render_filter) = 0;
-
- // This function deregisters a render effect filter for a specified channel.
- virtual int DeregisterRenderEffectFilter(const int video_channel) = 0;
-
- // All cameras run the risk of getting in almost perfect sync with
- // florescent lamps, which will result in a very annoying flickering of the
- // image. Most cameras have some type of filter to protect against this but
- // not all of them succeed. Enabling this function will remove the flicker.
- virtual int EnableDeflickering(const int capture_id, const bool enable) = 0;
-
- // Some cameras produce very noisy captured images, especially in low‐light
- // conditions. This functionality will reduce the camera noise.
- virtual int EnableDenoising(const int capture_id, const bool enable) = 0;
-
- // This function enhances the colors on the decoded video stream, enabled by
- // default.
- virtual int EnableColorEnhancement(const int video_channel,
- const bool enable) = 0;
-
- protected:
- ViEImageProcess() {}
- virtual ~ViEImageProcess() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_IMAGE_PROCESS_H_
diff --git a/trunk/src/video_engine/include/vie_network.h b/trunk/src/video_engine/include/vie_network.h
deleted file mode 100644
index 9752008..0000000
--- a/trunk/src/video_engine/include/vie_network.h
+++ /dev/null
@@ -1,215 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_
-
-// This sub-API supports the following functionalities:
-// - Configuring send and receive addresses.
-// - External transport support.
-// - Port and address filters.
-// - Windows GQoS functions and ToS functions.
-// - Packet timeout notification.
-// - Dead‐or‐Alive connection observations.
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class Transport;
-class VideoEngine;
-
-// This enumerator describes VideoEngine packet timeout states.
-enum ViEPacketTimeout {
- NoPacket = 0,
- PacketReceived = 1
-};
-
-// This class declares an abstract interface for a user defined observer. It is
-// up to the VideoEngine user to implement a derived class which implements the
-// observer class. The observer is registered using RegisterObserver() and
-// deregistered using DeregisterObserver().
-class WEBRTC_DLLEXPORT ViENetworkObserver {
- public:
- // This method will be called periodically delivering a dead‐or‐alive
- // decision for a specified channel.
- virtual void OnPeriodicDeadOrAlive(const int video_channel,
- const bool alive) = 0;
-
- // This method is called once if a packet timeout occurred.
- virtual void PacketTimeout(const int video_channel,
- const ViEPacketTimeout timeout) = 0;
- protected:
- virtual ~ViENetworkObserver() {}
-};
-
-class WEBRTC_DLLEXPORT ViENetwork {
- public:
- // Default values.
- enum { KDefaultSampleTimeSeconds = 2 };
-
- // Factory for the ViENetwork sub‐API and increases an internal reference
- // counter if successful. Returns NULL if the API is not supported or if
- // construction fails.
- static ViENetwork* GetInterface(VideoEngine* video_engine);
-
- // Releases the ViENetwork sub-API and decreases an internal reference
- // counter.Returns the new reference count. This value should be zero
- // for all sub-API:s before the VideoEngine object can be safely deleted.
- virtual int Release() = 0;
-
- // Specifies the ports to receive RTP packets on. It is also possible to set
- // port for RTCP and local IP address.
- virtual int SetLocalReceiver(const int video_channel,
- const unsigned short rtp_port,
- const unsigned short rtcp_port = 0,
- const char* ip_address = NULL) = 0;
-
- // Gets the local receiver ports and address for a specified channel.
- virtual int GetLocalReceiver(const int video_channel,
- unsigned short& rtp_port,
- unsigned short& rtcp_port, char* ip_address) = 0;
-
- // Specifies the destination port and IP address for a specified channel.
- virtual int SetSendDestination(const int video_channel,
- const char* ip_address,
- const unsigned short rtp_port,
- const unsigned short rtcp_port = 0,
- const unsigned short source_rtp_port = 0,
- const unsigned short source_rtcp_port = 0) = 0;
-
- // Get the destination port and address for a specified channel.
- virtual int GetSendDestination(const int video_channel,
- char* ip_address,
- unsigned short& rtp_port,
- unsigned short& rtcp_port,
- unsigned short& source_rtp_port,
- unsigned short& source_rtcp_port) = 0;
-
- // This function registers a user implementation of Transport to use for
- // sending RTP and RTCP packets on this channel.
- virtual int RegisterSendTransport(const int video_channel,
- Transport& transport) = 0;
-
- // This function deregisters a used Transport for a specified channel.
- virtual int DeregisterSendTransport(const int video_channel) = 0;
-
- // When using external transport for a channel, received RTP packets should
- // be passed to VideoEngine using this function. The input should contain
- // the RTP header and payload.
- virtual int ReceivedRTPPacket(const int video_channel,
- const void* data,
- const int length) = 0;
-
- // When using external transport for a channel, received RTCP packets should
- // be passed to VideoEngine using this function.
- virtual int ReceivedRTCPPacket(const int video_channel,
- const void* data,
- const int length) = 0;
-
- // Gets the source ports and IP address of the incoming stream for a
- // specified channel.
- virtual int GetSourceInfo(const int video_channel,
- unsigned short& rtp_port,
- unsigned short& rtcp_port,
- char* ip_address,
- unsigned int ip_address_length) = 0;
-
- // Gets the local IP address, in string format.
- virtual int GetLocalIP(char ip_address[64], bool ipv6 = false) = 0;
-
- // Enables IPv6, instead of IPv4, for a specified channel.
- virtual int EnableIPv6(int video_channel) = 0;
-
- // The function returns true if IPv6 is enabled, false otherwise.
- virtual bool IsIPv6Enabled(int video_channel) = 0;
-
- // Enables a port and IP address filtering for incoming packets on a
- // specific channel.
- virtual int SetSourceFilter(const int video_channel,
- const unsigned short rtp_port,
- const unsigned short rtcp_port = 0,
- const char* ip_address = NULL) = 0;
-
- // Gets current port and IP address filter for a specified channel.
- virtual int GetSourceFilter(const int video_channel,
- unsigned short& rtp_port,
- unsigned short& rtcp_port,
- char* ip_address) = 0;
-
- // This function sets the six‐bit Differentiated Services Code Point (DSCP)
- // in the IP header of the outgoing stream for a specific channel.
- // Windows and Linux only.
- virtual int SetSendToS(const int video_channel,
- const int DSCP,
- const bool use_set_sockOpt = false) = 0;
-
- // Retrieves the six‐bit Differentiated Services Code Point (DSCP) in the IP
- // header of the outgoing stream for a specific channel.
- virtual int GetSendToS(const int video_channel,
- int& DSCP,
- bool& use_set_sockOpt) = 0;
-
- // This function sets the Generic Quality of Service (GQoS) service level.
- // The Windows operating system then maps to a Differentiated Services Code
- // Point (DSCP) and to an 802.1p setting. Windows only.
- virtual int SetSendGQoS(const int video_channel, const bool enable,
- const int service_type,
- const int overrideDSCP = 0) = 0;
-
- // This function retrieves the currently set GQoS service level for a
- // specific channel.
- virtual int GetSendGQoS(const int video_channel,
- bool& enabled,
- int& service_type,
- int& overrideDSCP) = 0;
-
- // This function sets the Maximum Transition Unit (MTU) for a channel. The
- // RTP packet will be packetized based on this MTU to optimize performance
- // over the network.
- virtual int SetMTU(int video_channel, unsigned int mtu) = 0;
-
- // This function enables or disables warning reports if packets have not
- // been received for a specified time interval.
- virtual int SetPacketTimeoutNotification(const int video_channel,
- bool enable,
- int timeout_seconds) = 0;
-
- // Registers an instance of a user implementation of the ViENetwork
- // observer.
- virtual int RegisterObserver(const int video_channel,
- ViENetworkObserver& observer) = 0;
-
- // Removes a registered instance of ViENetworkObserver.
- virtual int DeregisterObserver(const int video_channel) = 0;
-
- // This function enables or disables the periodic dead‐or‐alive callback
- // functionality for a specified channel.
- virtual int SetPeriodicDeadOrAliveStatus(
- const int video_channel,
- const bool enable,
- const unsigned int sample_time_seconds = KDefaultSampleTimeSeconds) = 0;
-
- // This function handles sending a raw UDP data packet over an existing RTP
- // or RTCP socket.
- virtual int SendUDPPacket(const int video_channel,
- const void* data,
- const unsigned int length,
- int& transmitted_bytes,
- bool use_rtcp_socket = false) = 0;
-
- protected:
- ViENetwork() {}
- virtual ~ViENetwork() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_NETWORK_H_
diff --git a/trunk/src/video_engine/include/vie_render.h b/trunk/src/video_engine/include/vie_render.h
deleted file mode 100644
index 0b8328e..0000000
--- a/trunk/src/video_engine/include/vie_render.h
+++ /dev/null
@@ -1,110 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-// - Specify render destinations for incoming video streams, capture devices
-// and files.
-// - Configuring render streams.
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RENDER_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RENDER_H_
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VideoEngine;
-class VideoRender;
-
-// This class declares an abstract interface to be used for external renderers.
-// The user implemented derived class is registered using AddRenderer().
-class WEBRTC_DLLEXPORT ExternalRenderer {
- public:
- // This method will be called when the stream to be rendered changes in
- // resolution or number of streams mixed in the image.
- virtual int FrameSizeChange(unsigned int width,
- unsigned int height,
- unsigned int number_of_streams) = 0;
-
- // This method is called when a new frame should be rendered.
- virtual int DeliverFrame(unsigned char* buffer,
- int buffer_size,
- // RTP timestamp in 90kHz.
- uint32_t time_stamp,
- // Wallclock render time in miliseconds
- int64_t render_time) = 0;
-
- protected:
- virtual ~ExternalRenderer() {}
-};
-
-class WEBRTC_DLLEXPORT ViERender {
- public:
- // Factory for the ViERender sub‐API and increases an internal reference
- // counter if successful. Returns NULL if the API is not supported or if
- // construction fails.
- static ViERender* GetInterface(VideoEngine* video_engine);
-
- // Releases the ViERender sub-API and decreases an internal reference
- // counter. Returns the new reference count. This value should be zero
- // for all sub-API:s before the VideoEngine object can be safely deleted.
- virtual int Release() = 0;
-
- // Registers render module.
- virtual int RegisterVideoRenderModule(VideoRender& render_module) = 0;
-
- // Deregisters render module.
- virtual int DeRegisterVideoRenderModule(VideoRender& render_module) = 0;
-
- // Sets the render destination for a given render ID.
- virtual int AddRenderer(const int render_id,
- void* window,
- const unsigned int z_order,
- const float left,
- const float top,
- const float right,
- const float bottom) = 0;
-
- // Removes the renderer for a stream.
- virtual int RemoveRenderer(const int render_id) = 0;
-
- // Starts rendering a render stream.
- virtual int StartRender(const int render_id) = 0;
-
- // Stops rendering a render stream.
- virtual int StopRender(const int render_id) = 0;
-
- // Configures an already added render stream.
- virtual int ConfigureRender(int render_id,
- const unsigned int z_order,
- const float left,
- const float top,
- const float right,
- const float bottom) = 0;
-
- // This function mirrors the rendered stream left and right or up and down.
- virtual int MirrorRenderStream(const int render_id,
- const bool enable,
- const bool mirror_xaxis,
- const bool mirror_yaxis) = 0;
-
- // External render.
- virtual int AddRenderer(const int render_id,
- RawVideoType video_input_format,
- ExternalRenderer* renderer) = 0;
-
- protected:
- ViERender() {}
- virtual ~ViERender() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RENDER_H_
diff --git a/trunk/src/video_engine/include/vie_rtp_rtcp.h b/trunk/src/video_engine/include/vie_rtp_rtcp.h
deleted file mode 100644
index 1397222..0000000
--- a/trunk/src/video_engine/include/vie_rtp_rtcp.h
+++ /dev/null
@@ -1,311 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-// - Callbacks for RTP and RTCP events such as modified SSRC or CSRC.
-// - SSRC handling.
-// - Transmission of RTCP reports.
-// - Obtaining RTCP data from incoming RTCP sender reports.
-// - RTP and RTCP statistics (jitter, packet loss, RTT etc.).
-// - Forward Error Correction (FEC).
-// - RTP Keep‐alive for maintaining the NAT mappings associated to RTP flows.
-// - Writing RTP and RTCP packets to binary files for off‐line analysis of the
-// call quality.
-// - Inserting extra RTP packets into active audio stream.
-
-#ifndef WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RTP_RTCP_H_
-#define WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RTP_RTCP_H_
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VideoEngine;
-
-// This enumerator sets the RTCP mode.
-enum ViERTCPMode {
- kRtcpNone = 0,
- kRtcpCompound_RFC4585 = 1,
- kRtcpNonCompound_RFC5506 = 2
-};
-
-// This enumerator describes the key frame request mode.
-enum ViEKeyFrameRequestMethod {
- kViEKeyFrameRequestNone = 0,
- kViEKeyFrameRequestPliRtcp = 1,
- kViEKeyFrameRequestFirRtp = 2,
- kViEKeyFrameRequestFirRtcp = 3
-};
-
-enum StreamType {
- kViEStreamTypeNormal = 0, // Normal media stream
- kViEStreamTypeRtx = 1 // Retransmission media stream
-};
-
-// This class declares an abstract interface for a user defined observer. It is
-// up to the VideoEngine user to implement a derived class which implements the
-// observer class. The observer is registered using RegisterRTPObserver() and
-// deregistered using DeregisterRTPObserver().
-class WEBRTC_DLLEXPORT ViERTPObserver {
- public:
- // This method is called if SSRC of the incoming stream is changed.
- virtual void IncomingSSRCChanged(const int video_channel,
- const unsigned int SSRC) = 0;
-
- // This method is called if a field in CSRC changes or if the number of
- // CSRCs changes.
- virtual void IncomingCSRCChanged(const int video_channel,
- const unsigned int CSRC,
- const bool added) = 0;
- protected:
- virtual ~ViERTPObserver() {}
-};
-
-// This class declares an abstract interface for a user defined observer. It is
-// up to the VideoEngine user to implement a derived class which implements the
-// observer class. The observer is registered using RegisterRTCPObserver() and
-// deregistered using DeregisterRTCPObserver().
-
-class WEBRTC_DLLEXPORT ViERTCPObserver {
- public:
- // This method is called if a application-defined RTCP packet has been
- // received.
- virtual void OnApplicationDataReceived(
- const int video_channel,
- const unsigned char sub_type,
- const unsigned int name,
- const char* data,
- const unsigned short data_length_in_bytes) = 0;
- protected:
- virtual ~ViERTCPObserver() {}
-};
-
-class WEBRTC_DLLEXPORT ViERTP_RTCP {
- public:
- enum { KDefaultDeltaTransmitTimeSeconds = 15 };
- enum { KMaxRTCPCNameLength = 256 };
-
- // Factory for the ViERTP_RTCP sub‐API and increases an internal reference
- // counter if successful. Returns NULL if the API is not supported or if
- // construction fails.
- static ViERTP_RTCP* GetInterface(VideoEngine* video_engine);
-
- // Releases the ViERTP_RTCP sub-API and decreases an internal reference
- // counter. Returns the new reference count. This value should be zero
- // for all sub-API:s before the VideoEngine object can be safely deleted.
- virtual int Release() = 0;
-
- // This function enables you to specify the RTP synchronization source
- // identifier (SSRC) explicitly.
- virtual int SetLocalSSRC(const int video_channel,
- const unsigned int SSRC,
- const StreamType usage = kViEStreamTypeNormal,
- const unsigned char simulcast_idx = 0) = 0;
-
- // This function gets the SSRC for the outgoing RTP stream for the specified
- // channel.
- virtual int GetLocalSSRC(const int video_channel,
- unsigned int& SSRC) const = 0;
-
- // This function map a incoming SSRC to a StreamType so that the engine
- // can know which is the normal stream and which is the RTX
- virtual int SetRemoteSSRCType(const int video_channel,
- const StreamType usage,
- const unsigned int SSRC) const = 0;
-
- // This function gets the SSRC for the incoming RTP stream for the specified
- // channel.
- virtual int GetRemoteSSRC(const int video_channel,
- unsigned int& SSRC) const = 0;
-
- // This function returns the CSRCs of the incoming RTP packets.
- virtual int GetRemoteCSRCs(const int video_channel,
- unsigned int CSRCs[kRtpCsrcSize]) const = 0;
-
- // This function enables manual initialization of the sequence number. The
- // start sequence number is normally a random number.
- virtual int SetStartSequenceNumber(const int video_channel,
- unsigned short sequence_number) = 0;
-
- // This function sets the RTCP status for the specified channel.
- // Default mode is kRtcpCompound_RFC4585.
- virtual int SetRTCPStatus(const int video_channel,
- const ViERTCPMode rtcp_mode) = 0;
-
- // This function gets the RTCP status for the specified channel.
- virtual int GetRTCPStatus(const int video_channel,
- ViERTCPMode& rtcp_mode) const = 0;
-
- // This function sets the RTCP canonical name (CNAME) for the RTCP reports
- // on a specific channel.
- virtual int SetRTCPCName(const int video_channel,
- const char rtcp_cname[KMaxRTCPCNameLength]) = 0;
-
- // This function gets the RTCP canonical name (CNAME) for the RTCP reports
- // sent the specified channel.
- virtual int GetRTCPCName(const int video_channel,
- char rtcp_cname[KMaxRTCPCNameLength]) const = 0;
-
- // This function gets the RTCP canonical name (CNAME) for the RTCP reports
- // received on the specified channel.
- virtual int GetRemoteRTCPCName(
- const int video_channel,
- char rtcp_cname[KMaxRTCPCNameLength]) const = 0;
-
- // This function sends an RTCP APP packet on a specific channel.
- virtual int SendApplicationDefinedRTCPPacket(
- const int video_channel,
- const unsigned char sub_type,
- unsigned int name,
- const char* data,
- unsigned short data_length_in_bytes) = 0;
-
- // This function enables Negative Acknowledgment (NACK) using RTCP,
- // implemented based on RFC 4585. NACK retransmits RTP packets if lost on
- // the network. This creates a lossless transport at the expense of delay.
- // If using NACK, NACK should be enabled on both endpoints in a call.
- virtual int SetNACKStatus(const int video_channel, const bool enable) = 0;
-
- // This function enables Forward Error Correction (FEC) using RTCP,
- // implemented based on RFC 5109, to improve packet loss robustness. Extra
- // FEC packets are sent together with the usual media packets, hence
- // part of the bitrate will be used for FEC packets.
- virtual int SetFECStatus(const int video_channel,
- const bool enable,
- const unsigned char payload_typeRED,
- const unsigned char payload_typeFEC) = 0;
-
- // This function enables hybrid Negative Acknowledgment using RTCP
- // and Forward Error Correction (FEC) implemented based on RFC 5109,
- // to improve packet loss robustness. Extra
- // FEC packets are sent together with the usual media packets, hence will
- // part of the bitrate be used for FEC packets.
- // The hybrid mode will choose between nack only, fec only and both based on
- // network conditions. When both are applied, only packets that were not
- // recovered by the FEC will be nacked.
- virtual int SetHybridNACKFECStatus(const int video_channel,
- const bool enable,
- const unsigned char payload_typeRED,
- const unsigned char payload_typeFEC) = 0;
-
- // This function enables RTCP key frame requests.
- virtual int SetKeyFrameRequestMethod(
- const int video_channel, const ViEKeyFrameRequestMethod method) = 0;
-
- // This function enables signaling of temporary bitrate constraints using
- // RTCP, implemented based on RFC4585.
- virtual int SetTMMBRStatus(const int video_channel, const bool enable) = 0;
-
- // Enables and disables REMB packets for this channel. |sender| indicates
- // this channel is encoding, |receiver| tells the bitrate estimate for
- // this channel should be included in the REMB packet.
- virtual int SetRembStatus(int video_channel,
- bool sender,
- bool receiver) = 0;
-
- // This function returns our locally created statistics of the received RTP
- // stream.
- virtual int GetReceivedRTCPStatistics(
- const int video_channel,
- unsigned short& fraction_lost,
- unsigned int& cumulative_lost,
- unsigned int& extended_max,
- unsigned int& jitter,
- int& rtt_ms) const = 0;
-
- // This function returns statistics reported by the remote client in a RTCP
- // packet.
- virtual int GetSentRTCPStatistics(const int video_channel,
- unsigned short& fraction_lost,
- unsigned int& cumulative_lost,
- unsigned int& extended_max,
- unsigned int& jitter,
- int& rtt_ms) const = 0;
-
- // The function gets statistics from the sent and received RTP streams.
- virtual int GetRTPStatistics(const int video_channel,
- unsigned int& bytes_sent,
- unsigned int& packets_sent,
- unsigned int& bytes_received,
- unsigned int& packets_received) const = 0;
-
- // The function gets bandwidth usage statistics from the sent RTP streams in
- // bits/s.
- virtual int GetBandwidthUsage(const int video_channel,
- unsigned int& total_bitrate_sent,
- unsigned int& video_bitrate_sent,
- unsigned int& fec_bitrate_sent,
- unsigned int& nackBitrateSent) const = 0;
-
- // This function gets the send-side estimated bandwidth available for video,
- // including overhead, in bits/s.
- virtual int GetEstimatedSendBandwidth(
- const int video_channel,
- unsigned int* estimated_bandwidth) const = 0;
-
- // This function gets the receive-side estimated bandwidth available for
- // video, including overhead, in bits/s.
- // Returns -1 when no valid estimate is available.
- virtual int GetEstimatedReceiveBandwidth(
- const int video_channel,
- unsigned int* estimated_bandwidth) const = 0;
-
- // This function enables or disables an RTP keep-alive mechanism which can
- // be used to maintain an existing Network Address Translator (NAT) mapping
- // while regular RTP is no longer transmitted.
- virtual int SetRTPKeepAliveStatus(
- const int video_channel,
- bool enable,
- const char unknown_payload_type,
- const unsigned int delta_transmit_time_seconds =
- KDefaultDeltaTransmitTimeSeconds) = 0;
-
- // This function gets the RTP keep-alive status.
- virtual int GetRTPKeepAliveStatus(
- const int video_channel,
- bool& enabled,
- char& unkown_payload_type,
- unsigned int& delta_transmit_time_seconds) const = 0;
-
- // This function enables capturing of RTP packets to a binary file on a
- // specific channel and for a given direction. The file can later be
- // replayed using e.g. RTP Tools rtpplay since the binary file format is
- // compatible with the rtpdump format.
- virtual int StartRTPDump(const int video_channel,
- const char file_nameUTF8[1024],
- RTPDirections direction) = 0;
-
- // This function disables capturing of RTP packets to a binary file on a
- // specific channel and for a given direction.
- virtual int StopRTPDump(const int video_channel,
- RTPDirections direction) = 0;
-
- // Registers an instance of a user implementation of the ViERTPObserver.
- virtual int RegisterRTPObserver(const int video_channel,
- ViERTPObserver& observer) = 0;
-
- // Removes a registered instance of ViERTPObserver.
- virtual int DeregisterRTPObserver(const int video_channel) = 0;
-
- // Registers an instance of a user implementation of the ViERTCPObserver.
- virtual int RegisterRTCPObserver(const int video_channel,
- ViERTCPObserver& observer) = 0;
-
- // Removes a registered instance of ViERTCPObserver.
- virtual int DeregisterRTCPObserver(const int video_channel) = 0;
-
- protected:
- ViERTP_RTCP() {}
- virtual ~ViERTP_RTCP() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_INCLUDE_VIE_RTP_RTCP_H_
diff --git a/trunk/src/video_engine/main/test/SimpleCocoaGUI/GUI_Defines.h b/trunk/src/video_engine/main/test/SimpleCocoaGUI/GUI_Defines.h
deleted file mode 100644
index 8382844..0000000
--- a/trunk/src/video_engine/main/test/SimpleCocoaGUI/GUI_Defines.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * GUI_Defines.h
- *
- */
-
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_SIMPLECOCOAGUI_GUI_DEFINES_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_SIMPLECOCOAGUI_GUI_DEFINES_H_
-
-#define ViE_TEST(x) if(-1 == x){ \
-int errNum = _ptrViEBase->LastError(); \
-NSLog(@"ERROR: %d at %s:%d", errNum, __FUNCTION__, __LINE__); \
-}
-
-
-// Video Engine Related
-#define V_CAPTURE_DEVICE_INDEX 0
-#define V_VIE_CAPTURE_ID 747
-#define V_DEVICE_NAME_LENGTH 256
-#define V_CODEC_INDEX 2
-#define V_IP_ADDRESS "127.0.0.1"
-#define V_RTP_PORT 8000
-
-
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_SIMPLECOCOAGUI_GUI_DEFINES_H_
diff --git a/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUI-Info.plist b/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUI-Info.plist
deleted file mode 100644
index d0d3a18..0000000
--- a/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUI-Info.plist
+++ /dev/null
@@ -1,32 +0,0 @@
-<?xml version="1.0" encoding="UTF-8"?>
-<!DOCTYPE plist PUBLIC "-//Apple//DTD PLIST 1.0//EN" "http://www.apple.com/DTDs/PropertyList-1.0.dtd">
-<plist version="1.0">
-<dict>
- <key>CFBundleDevelopmentRegion</key>
- <string>English</string>
- <key>CFBundleExecutable</key>
- <string>${EXECUTABLE_NAME}</string>
- <key>CFBundleIconFile</key>
- <string></string>
- <key>CFBundleIdentifier</key>
- <string>com.yourcompany.${PRODUCT_NAME:rfc1034identifier}</string>
- <key>CFBundleInfoDictionaryVersion</key>
- <string>6.0</string>
- <key>CFBundleName</key>
- <string>${PRODUCT_NAME}</string>
- <key>CFBundlePackageType</key>
- <string>APPL</string>
- <key>CFBundleShortVersionString</key>
- <string>1.0</string>
- <key>CFBundleSignature</key>
- <string>????</string>
- <key>CFBundleVersion</key>
- <string>1</string>
- <key>LSMinimumSystemVersion</key>
- <string>${MACOSX_DEPLOYMENT_TARGET}</string>
- <key>NSMainNibFile</key>
- <string>SimpleCocoaGUI</string>
- <key>NSPrincipalClass</key>
- <string>NSApplication</string>
-</dict>
-</plist>
diff --git a/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUIAppDelegate.h b/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUIAppDelegate.h
deleted file mode 100644
index 10d52fc..0000000
--- a/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUIAppDelegate.h
+++ /dev/null
@@ -1,77 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// SimpleCocoaGUIAppDelegate.h
-//
-
-#import <Cocoa/Cocoa.h>
-#include <iostream>
-using namespace std;
-
-@class ViECocoaRenderView;
-
-#include "GUI_Defines.h"
-
-#include "common_types.h"
-#include "voe_base.h"
-
-#include "vie_base.h"
-#include "vie_capture.h"
-#include "vie_codec.h"
-#include "vie_file.h"
-#include "vie_network.h"
-#include "vie_render.h"
-#include "vie_rtp_rtcp.h"
-#include "vie_errors.h"
-
-
-
-@interface SimpleCocoaGUIAppDelegate : NSObject <NSApplicationDelegate> {
- NSWindow* _window;
- IBOutlet NSOpenGLView* _vieCocoaRenderView1;
- IBOutlet NSOpenGLView* _vieCocoaRenderView2;
- IBOutlet NSButton* _butRestartLoopback;
- VideoEngine* _ptrViE;
- ViEBase* _ptrViEBase;
- ViECapture* _ptrViECapture;
- ViERender* _ptrViERender;
- ViECodec* _ptrViECodec;
- ViENetwork* _ptrViENetwork;
-
- bool _fullScreen;
- int _videoChannel;
-
- int _captureId;
-
- VideoEngine* ptrViE;
- ViEBase* ptrViEBase;
- ViECapture* ptrViECapture;
- ViERTP_RTCP* ptrViERtpRtcp;
- ViERender* ptrViERender;
- ViECodec* ptrViECodec;
- ViENetwork* ptrViENetwork;
-}
-
-@property (assign) IBOutlet NSWindow* window;
--(void)createUI:(bool)fullScreen;
--(void)initViECocoaTest;
--(void)initializeVariables;
--(void)NSLogVideoCodecs;
--(void)startViECocoaTest;
--(int)initLoopback;
--(int)ioLooback;
--(int)startLoopback;
--(int)stopLooback;
-
--(IBAction)handleRestart:(id)sender;
-
-
-@end
diff --git a/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUIAppDelegate.mm b/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUIAppDelegate.mm
deleted file mode 100644
index d594cfe..0000000
--- a/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUIAppDelegate.mm
+++ /dev/null
@@ -1,1075 +0,0 @@
-//
-// SimpleCocoaGUIAppDelegate.m
-//
-
-#import "SimpleCocoaGUIAppDelegate.h"
-
-@implementation SimpleCocoaGUIAppDelegate
-
-@synthesize window = _window;
-
-- (void)applicationDidFinishLaunching:(NSNotification *)aNotification {
-
-// [self initializeVariables];
- [self createUI];
-// [self initViECocoaTest];
-// [self NSLogVideoCodecs];
-// [self startViECocoaTest];
-
-// [self startLoopback];
-
- [self ioLooback];
-}
-
--(void)createUI{
-
- NSRect outWindow1Frame = NSMakeRect(200, 200, 200, 200);
- NSWindow* outWindow1 = [[NSWindow alloc] initWithContentRect:outWindow1Frame styleMask:NSTitledWindowMask backing:NSBackingStoreBuffered defer:NO];
- [outWindow1 orderOut:nil];
- NSRect vieAutotestCocoaRenderView1Frame = NSMakeRect(0, 0, 200, 200);
- _vieCocoaRenderView1 = [[ViECocoaRenderView alloc] initWithFrame:vieAutotestCocoaRenderView1Frame];
- [[outWindow1 contentView] addSubview:_vieCocoaRenderView1];
- [outWindow1 setTitle:[NSString stringWithFormat:@"window1"]];
- [outWindow1 makeKeyAndOrderFront:NSApp];
-
-
- NSRect outWindow2Frame = NSMakeRect(400, 200, 200, 200);
- NSWindow* outWindow2 = [[NSWindow alloc] initWithContentRect:outWindow2Frame styleMask:NSTitledWindowMask backing:NSBackingStoreBuffered defer:NO];
- [outWindow2 orderOut:nil];
- NSRect vieAutotestCocoaRenderView2Frame = NSMakeRect(0, 0, 200, 200);
- _vieCocoaRenderView2 = [[ViECocoaRenderView alloc] initWithFrame:vieAutotestCocoaRenderView2Frame];
- [[outWindow2 contentView] addSubview:_vieCocoaRenderView2];
- [outWindow2 setTitle:[NSString stringWithFormat:@"window2"]];
- [outWindow2 makeKeyAndOrderFront:NSApp];
-
-
-
-
-
-
-
-}
-
--(void)initViECocoaTest{
-
- int _error = 0;
- _ptrViE = VideoEngine::Create();
- _ptrViEBase = ViEBase::GetInterface(_ptrViE);
- _error = _ptrViEBase->Init();
-
- _ptrViECapture = ViECapture::GetInterface(_ptrViE);
- _ptrViERender = ViERender::GetInterface(_ptrViE);
- _ptrViECodec = ViECodec::GetInterface(_ptrViE);
- _ptrViENetwork = ViENetwork::GetInterface(_ptrViE);
-
-
- _error = _ptrViE->SetTraceFile("ViEBaseStandardTest.txt");
- _error = _ptrViE->SetEncryptedTraceFile("ViEBaseStandardTestEncrypted.txt");
-
-
-}
-
-
--(void)initializeVariables{
- _fullScreen = YES;
-}
-
--(void)NSLogVideoCodecs{
- NSLog(@"Searching for video codecs.....");
-
- VideoCodec videoCodec;
- memset(&videoCodec, 0, sizeof(VideoCodec));
- for(int index = 0; index < _ptrViECodec->NumberOfCodecs(); index++)
- {
- ViE_TEST(_ptrViECodec->GetCodec(index, videoCodec));
- NSLog(@"Video codec found: %s", videoCodec.plName);
- }
-
-}
--(void)startViECocoaTest{
-
-
-
-
- int error=0;
-
- char deviceName[128];
- char deviceUniqueName[512];
- int captureId = 0;
- int dummy = 0;
-
- //ViE_TEST(_ptrViEBase->CreateChannel(_videoChannel));
- //ViE_TEST(_ptrViECapture->GetCaptureDevice(0,deviceName,sizeof(deviceName),deviceUniqueName,sizeof(deviceUniqueName)));
- //ViE_TEST(_ptrViECapture->AllocateCaptureDevice(deviceUniqueName,sizeof(deviceUniqueName),captureId));
- //ViE_TEST(_ptrViECapture->AllocateCaptureDevice("dummydevicethatdoesnotexist",sizeof(deviceUniqueName),dummy));
-
- char captureDeviceName[V_DEVICE_NAME_LENGTH] = "";
- char captureDeviceUniqueId[V_DEVICE_NAME_LENGTH] = "";
- int captureDeviceId = 0;
-
-
-
- ViE_TEST(_ptrViE->SetTraceFilter(webrtc::TR_ALL));
- ViE_TEST(_ptrViE->SetTraceFile("ViECocoaTrace.txt"));
- ViE_TEST(_ptrViE->SetEncryptedTraceFile("ViECocoaEncryptedTrace.txt"));
-
-
-
-
- // base
- ViE_TEST(_ptrViEBase->CreateChannel(_videoChannel));
-
- // capture device
- ViE_TEST(_ptrViECapture->GetCaptureDevice(V_CAPTURE_DEVICE_INDEX, captureDeviceName, V_DEVICE_NAME_LENGTH, captureDeviceUniqueId, V_DEVICE_NAME_LENGTH));
- ViE_TEST(_ptrViECapture->AllocateCaptureDevice(captureDeviceUniqueId, V_DEVICE_NAME_LENGTH, captureDeviceId));
- ViE_TEST(_ptrViECapture->ConnectCaptureDevice(captureDeviceId, _videoChannel));
- ViE_TEST(_ptrViECapture->StartCapture(captureDeviceId));
-
- // renderer
- ViE_TEST(_ptrViERender->AddRenderer(captureDeviceId, (void*)_vieCocoaRenderView1, 0, 0.0, 0.0, 1.0, 1.0));
- ViE_TEST(_ptrViERender->StartRender(captureDeviceId));
-// usleep(3 * 1000);
-// ViE_TEST(_ptrViERender->RemoveRenderer(captureDeviceId));
- //exit(0);
-
-
-// // codec
-// [self NSLogVideoCodecs];
-// VideoCodec videoCodec;
-// memset(&videoCodec, 0, sizeof(VideoCodec));
-// ViE_TEST(_ptrViECodec->GetCodec(V_CODEC_INDEX, videoCodec));
-// ViE_TEST(_ptrViECodec->SetReceiveCodec(_videoChannel, videoCodec));
-// ViE_TEST(_ptrViECodec->SetSendCodec(_videoChannel, videoCodec));
-//
-// // network + base
-// ViE_TEST(_ptrViENetwork->SetLocalReceiver(_videoChannel, V_RTP_PORT));
-// ViE_TEST(_ptrViEBase->StartReceive(_videoChannel));
-// ViE_TEST(_ptrViENetwork->SetSendDestination(_videoChannel, V_IP_ADDRESS, V_RTP_PORT));
-// ViE_TEST(_ptrViEBase->StartSend(_videoChannel));
-// ViE_TEST(_ptrViERender->MirrorRenderStream(captureDeviceId, true, false, true));
-
-
-}
-
--(int)initLoopback
-{
-
-}
--(int)startLoopback
-{
- //********************************************************
- // Begin create/initialize Video Engine for testing
- //********************************************************
-
- int error = 0;
- bool succeeded = true;
- int numberOfErrors = 0;
- std::string str;
-
- //
- // Create a VideoEngine instance
- //
-// VideoEngine* ptrViE = NULL;
- ptrViE = VideoEngine::Create();
- if (ptrViE == NULL)
- {
- printf("ERROR in VideoEngine::Create\n");
- return -1;
- }
-
- error = ptrViE->SetTraceFilter(webrtc::TR_ALL);
- if (error == -1)
- {
- printf("ERROR in VideoEngine::SetTraceLevel\n");
- return -1;
- }
-
-
- error = ptrViE->SetTraceFile("ViETrace.txt");
- if (error == -1)
- {
- printf("ERROR in VideoEngine::SetTraceFile\n");
- return -1;
- }
-
- error = ptrViE->SetEncryptedTraceFile("ViEEncryptedTrace.txt");
- if (error == -1)
- {
- printf("ERROR in VideoEngine::SetEncryptedTraceFile\n");
- return -1;
- }
-
- //
- // Init VideoEngine and create a channel
- //
- ptrViEBase = ViEBase::GetInterface(ptrViE);
- if (ptrViEBase == NULL)
- {
- printf("ERROR in ViEBase::GetInterface\n");
- return -1;
- }
-
- error = ptrViEBase->Init();
- if (error == -1)
- {
- printf("ERROR in ViEBase::Init\n");
- return -1;
- }
-
- int videoChannel = -1;
- error = ptrViEBase->CreateChannel(_videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViEBase::CreateChannel\n");
- return -1;
- }
-
- //
- // List available capture devices, allocate and connect.
- //
- ptrViECapture = ViECapture::GetInterface(ptrViE);
- if (ptrViEBase == NULL)
- {
- printf("ERROR in ViECapture::GetInterface\n");
- return -1;
- }
-
- const unsigned int KMaxDeviceNameLength = 128;
- const unsigned int KMaxUniqueIdLength = 256;
- char deviceName[KMaxDeviceNameLength];
- memset(deviceName, 0, KMaxDeviceNameLength);
- char uniqueId[KMaxUniqueIdLength];
- memset(uniqueId, 0, KMaxUniqueIdLength);
-
- std::cout << std::endl;
- std::cout << "Available capture devices:" << std::endl;
- unsigned int captureIdx = 0;
- for (captureIdx = 0;
- captureIdx < ptrViECapture->NumberOfCaptureDevices();
- captureIdx++)
- {
- memset(deviceName, 0, KMaxDeviceNameLength);
- memset(uniqueId, 0, KMaxUniqueIdLength);
-
- error = ptrViECapture->GetCaptureDevice(captureIdx,
- deviceName, KMaxDeviceNameLength, uniqueId, KMaxUniqueIdLength);
- if (error == -1)
- {
- printf("ERROR in ViECapture::GetCaptureDevice\n");
- return -1;
- }
- std::cout << " " << captureIdx+1 << ". " << deviceName
- << std::endl;
- }
- std::cout << std::endl;
- std::cout << "Choose capture devices: ";
-// std::getline(std::cin, str);
-// captureIdx = atoi(str.c_str()) - 1;
- captureIdx = 0;
- error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
- KMaxDeviceNameLength, uniqueId, KMaxUniqueIdLength);
- if (error == -1)
- {
- printf("ERROR in ViECapture::GetCaptureDevice\n");
- return -1;
- }
-
- _captureId = 0;
- error = ptrViECapture->AllocateCaptureDevice(uniqueId,
- KMaxUniqueIdLength, _captureId);
- if (error == -1)
- {
- printf("ERROR in ViECapture::AllocateCaptureDevice\n");
- return -1;
- }
-
- error = ptrViECapture->ConnectCaptureDevice(_captureId,
- _videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViECapture::ConnectCaptureDevice\n");
- return -1;
- }
-
- error = ptrViECapture->StartCapture(_captureId);
- if (error == -1)
- {
- printf("ERROR in ViECapture::StartCapture\n");
- return -1;
- }
-
- //
- // RTP/RTCP settings
- //
- ptrViERtpRtcp = ViERTP_RTCP::GetInterface(ptrViE);
- if (ptrViERtpRtcp == NULL)
- {
- printf("ERROR in ViERTP_RTCP::GetInterface\n");
- return -1;
- }
-
- error = ptrViERtpRtcp->SetRTCPStatus(_videoChannel,
- kRtcpCompound_RFC4585);
- if (error == -1)
- {
- printf("ERROR in ViERTP_RTCP::SetRTCPStatus\n");
- return -1;
- }
-
- error = ptrViERtpRtcp->SetKeyFrameRequestMethod(_videoChannel,
- kViEKeyFrameRequestPliRtcp);
- if (error == -1)
- {
- printf("ERROR in ViERTP_RTCP::SetKeyFrameRequestMethod\n");
- return -1;
- }
-
- error = ptrViERtpRtcp->SetTMMBRStatus(_videoChannel, true);
- if (error == -1)
- {
- printf("ERROR in ViERTP_RTCP::SetTMMBRStatus\n");
- return -1;
- }
-
- //
- // Set up rendering
- //
- ptrViERender = ViERender::GetInterface(ptrViE);
- if (ptrViERender == NULL)
- {
- printf("ERROR in ViERender::GetInterface\n");
- return -1;
- }
-
- error = ptrViERender->AddRenderer(_captureId, _vieCocoaRenderView1,
- 0, 0.0, 0.0, 1.0, 1.0);
- if (error == -1)
- {
- printf("ERROR in ViERender::AddRenderer\n");
- return -1;
- }
-
- error = ptrViERender->StartRender(_captureId);
- if (error == -1)
- {
- printf("ERROR in ViERender::StartRender\n");
- return -1;
- }
-
- error = ptrViERender->AddRenderer(_videoChannel, _vieCocoaRenderView2,
- 1, 0.0, 0.0, 1.0, 1.0);
- if (error == -1)
- {
- printf("ERROR in ViERender::AddRenderer\n");
- return -1;
- }
-
- error = ptrViERender->StartRender(_videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViERender::StartRender\n");
- return -1;
- }
-
- //
- // Setup codecs
- //
- ptrViECodec = ViECodec::GetInterface(ptrViE);
- if (ptrViECodec == NULL)
- {
- printf("ERROR in ViECodec::GetInterface\n");
- return -1;
- }
-
- std::cout << std::endl;
- std::cout << "Available codecs:" << std::endl;
-
- // Check available codecs and prepare receive codecs
- VideoCodec videoCodec;
- memset(&videoCodec, 0, sizeof(VideoCodec));
- unsigned int codecIdx = 0;
- for (codecIdx = 0;
- codecIdx < ptrViECodec->NumberOfCodecs();
- codecIdx++)
- {
- error = ptrViECodec->GetCodec(codecIdx, videoCodec);
- if (error == -1)
- {
- printf("ERROR in ViECodec::GetCodec\n");
- return -1;
- }
-
- error = ptrViECodec->SetReceiveCodec(_videoChannel,
- videoCodec);
- if (error == -1)
- {
- printf("ERROR in ViECodec::SetReceiveCodec\n");
- return -1;
- }
- if (videoCodec.codecType != kVideoCodecRED &&
- videoCodec.codecType != kVideoCodecULPFEC)
- {
- std::cout << " " << codecIdx+1 << ". " << videoCodec.plName
- << std::endl;
- }
- }
-// std::cout << std::endl;
-// std::cout << "Choose codec: ";
-// std::getline(std::cin, str);
-// codecIdx = atoi(str.c_str()) - 1;
- codecIdx = 0;
-
- error = ptrViECodec->GetCodec(codecIdx, videoCodec);
- if (error == -1)
- {
- printf("ERROR in ViECodec::GetCodec\n");
- return -1;
- }
-
- error = ptrViECodec->SetSendCodec(_videoChannel, videoCodec);
- if (error == -1)
- {
- printf("ERROR in ViECodec::SetSendCodec\n");
- return -1;
- }
-
- //
- // Address settings
- //
- ptrViENetwork = ViENetwork::GetInterface(ptrViE);
- if (ptrViENetwork == NULL)
- {
- printf("ERROR in ViENetwork::GetInterface\n");
- return -1;
- }
-
- const char* ipAddress = "127.0.0.1";
- const unsigned short rtpPort = 6000;
- error = ptrViENetwork->SetLocalReceiver(_videoChannel, rtpPort);
- if (error == -1)
- {
- printf("ERROR in ViENetwork::SetLocalReceiver\n");
- return -1;
- }
-
- error = ptrViEBase->StartReceive(_videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViENetwork::StartReceive\n");
- return -1;
- }
-
- error = ptrViENetwork->SetSendDestination(_videoChannel,
- ipAddress, rtpPort);
- if (error == -1)
- {
- printf("ERROR in ViENetwork::SetSendDestination\n");
- return -1;
- }
-
- error = ptrViEBase->StartSend(_videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViENetwork::StartSend\n");
- return -1;
- }
-
-
- //********************************************************
- // Engine started
- //********************************************************
-
-
- // Call started
- std::cout << std::endl;
- std::cout << "Loopback call started" << std::endl;
-// std::cout << std::endl << std::endl;
-// std::cout << "Press enter to stop...";
-// std::getline(std::cin, str);
-}
-
--(int)stopLooback
-{
- int error = 0;
-
-
-
- //********************************************************
- // Testing finished. Tear down Video Engine
- //********************************************************
-
- error = ptrViEBase->StopReceive(_videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViEBase::StopReceive\n");
- return -1;
- }
-
- error = ptrViEBase->StopSend(_videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViEBase::StopSend\n");
- return -1;
- }
-
- error = ptrViERender->StopRender(_captureId);
- if (error == -1)
- {
- printf("ERROR in ViERender::StopRender\n");
- return -1;
- }
-
- error = ptrViERender->RemoveRenderer(_captureId);
- if (error == -1)
- {
- printf("ERROR in ViERender::RemoveRenderer\n");
- return -1;
- }
-
- error = ptrViERender->StopRender(_videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViERender::StopRender\n");
- return -1;
- }
-
- error = ptrViERender->RemoveRenderer(_videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViERender::RemoveRenderer\n");
- return -1;
- }
-
- error = ptrViECapture->StopCapture(_captureId);
- if (error == -1)
- {
- printf("ERROR in ViECapture::StopCapture\n");
- return -1;
- }
-
- error = ptrViECapture->DisconnectCaptureDevice(_videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViECapture::DisconnectCaptureDevice\n");
- return -1;
- }
-
- error = ptrViECapture->ReleaseCaptureDevice(_captureId);
- if (error == -1)
- {
- printf("ERROR in ViECapture::ReleaseCaptureDevice\n");
- return -1;
- }
-
- error = ptrViEBase->DeleteChannel(_videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViEBase::DeleteChannel\n");
- return -1;
- }
-
- int remainingInterfaces = 0;
- remainingInterfaces = ptrViECodec->Release();
- remainingInterfaces += ptrViECapture->Release();
- remainingInterfaces += ptrViERtpRtcp->Release();
- remainingInterfaces += ptrViERender->Release();
- remainingInterfaces += ptrViENetwork->Release();
- remainingInterfaces += ptrViEBase->Release();
- if (remainingInterfaces > 0)
- {
- printf("ERROR: Could not release all interfaces\n");
- return -1;
- }
-
- bool deleted = VideoEngine::Delete(ptrViE);
- if (deleted == false)
- {
- printf("ERROR in VideoEngine::Delete\n");
- return -1;
- }
-
- return 0;
-
- // ===================================================================
- //
- // END: VideoEngine 3.0 Sample Code
- //
- // ===================================================================
-
-
-}
-
--(int)ioLooback
-{
- //********************************************************
- // Begin create/initialize Video Engine for testing
- //********************************************************
-
- int error = 0;
- bool succeeded = true;
- int numberOfErrors = 0;
- std::string str;
-
- //
- // Create a VideoEngine instance
- //
- VideoEngine* ptrViE = NULL;
- ptrViE = VideoEngine::Create();
- if (ptrViE == NULL)
- {
- printf("ERROR in VideoEngine::Create\n");
- return -1;
- }
-
- error = ptrViE->SetTraceFilter(webrtc::TR_ALL);
- if (error == -1)
- {
- printf("ERROR in VideoEngine::SetTraceLevel\n");
- return -1;
- }
-
-
- error = ptrViE->SetTraceFile("ViETrace.txt");
- if (error == -1)
- {
- printf("ERROR in VideoEngine::SetTraceFile\n");
- return -1;
- }
-
- error = ptrViE->SetEncryptedTraceFile("ViEEncryptedTrace.txt");
- if (error == -1)
- {
- printf("ERROR in VideoEngine::SetEncryptedTraceFile\n");
- return -1;
- }
-
- //
- // Init VideoEngine and create a channel
- //
- ViEBase* ptrViEBase = ViEBase::GetInterface(ptrViE);
- if (ptrViEBase == NULL)
- {
- printf("ERROR in ViEBase::GetInterface\n");
- return -1;
- }
-
- error = ptrViEBase->Init();
- if (error == -1)
- {
- printf("ERROR in ViEBase::Init\n");
- return -1;
- }
-
- int videoChannel = -1;
- error = ptrViEBase->CreateChannel(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViEBase::CreateChannel\n");
- return -1;
- }
-
- //
- // List available capture devices, allocate and connect.
- //
- ViECapture* ptrViECapture =
- ViECapture::GetInterface(ptrViE);
- if (ptrViEBase == NULL)
- {
- printf("ERROR in ViECapture::GetInterface\n");
- return -1;
- }
-
- const unsigned int KMaxDeviceNameLength = 128;
- const unsigned int KMaxUniqueIdLength = 256;
- char deviceName[KMaxDeviceNameLength];
- memset(deviceName, 0, KMaxDeviceNameLength);
- char uniqueId[KMaxUniqueIdLength];
- memset(uniqueId, 0, KMaxUniqueIdLength);
-
- std::cout << std::endl;
- std::cout << "Available capture devices:" << std::endl;
- unsigned int captureIdx = 0;
- for (captureIdx = 0;
- captureIdx < ptrViECapture->NumberOfCaptureDevices();
- captureIdx++)
- {
- memset(deviceName, 0, KMaxDeviceNameLength);
- memset(uniqueId, 0, KMaxUniqueIdLength);
-
- error = ptrViECapture->GetCaptureDevice(captureIdx,
- deviceName, KMaxDeviceNameLength, uniqueId, KMaxUniqueIdLength);
- if (error == -1)
- {
- printf("ERROR in ViECapture::GetCaptureDevice\n");
- return -1;
- }
- std::cout << " " << captureIdx+1 << ". " << deviceName
- << std::endl;
- }
- std::cout << std::endl;
- std::cout << "Choose capture devices: ";
-// std::getline(std::cin, str);
-// captureIdx = atoi(str.c_str()) - 1;
- captureIdx = 0;
- error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
- KMaxDeviceNameLength, uniqueId, KMaxUniqueIdLength);
- if (error == -1)
- {
- printf("ERROR in ViECapture::GetCaptureDevice\n");
- return -1;
- }
-
- int captureId = 0;
- error = ptrViECapture->AllocateCaptureDevice(uniqueId,
- KMaxUniqueIdLength, captureId);
- if (error == -1)
- {
- printf("ERROR in ViECapture::AllocateCaptureDevice\n");
- return -1;
- }
-
- error = ptrViECapture->ConnectCaptureDevice(captureId,
- videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViECapture::ConnectCaptureDevice\n");
- return -1;
- }
-
- error = ptrViECapture->StartCapture(captureId);
- if (error == -1)
- {
- printf("ERROR in ViECapture::StartCapture\n");
- return -1;
- }
-
- //
- // RTP/RTCP settings
- //
- ViERTP_RTCP* ptrViERtpRtcp =
- ViERTP_RTCP::GetInterface(ptrViE);
- if (ptrViERtpRtcp == NULL)
- {
- printf("ERROR in ViERTP_RTCP::GetInterface\n");
- return -1;
- }
-
- error = ptrViERtpRtcp->SetRTCPStatus(videoChannel,
- kRtcpCompound_RFC4585);
- if (error == -1)
- {
- printf("ERROR in ViERTP_RTCP::SetRTCPStatus\n");
- return -1;
- }
-
- error = ptrViERtpRtcp->SetKeyFrameRequestMethod(videoChannel,
- kViEKeyFrameRequestPliRtcp);
- if (error == -1)
- {
- printf("ERROR in ViERTP_RTCP::SetKeyFrameRequestMethod\n");
- return -1;
- }
-
- error = ptrViERtpRtcp->SetTMMBRStatus(videoChannel, true);
- if (error == -1)
- {
- printf("ERROR in ViERTP_RTCP::SetTMMBRStatus\n");
- return -1;
- }
-
- //
- // Set up rendering
- //
- ViERender* ptrViERender =
- ViERender::GetInterface(ptrViE);
- if (ptrViERender == NULL)
- {
- printf("ERROR in ViERender::GetInterface\n");
- return -1;
- }
-
-// error = ptrViERender->EnableFullScreenRender(_vieCocoaRenderView1);
-// if (error == -1)
-// {
-// printf("ERROR in ViERender::AddRenderer\n");
-// return -1;
-// }
-
-
- error = ptrViERender->AddRenderer(captureId, _vieCocoaRenderView1,
- 0, 0.5, 0.5, 1.0, 1.0);
- if (error == -1)
- {
- printf("ERROR in ViERender::AddRenderer\n");
- return -1;
- }
-
- error = ptrViERender->StartRender(captureId);
- if (error == -1)
- {
- printf("ERROR in ViERender::StartRender\n");
- return -1;
- }
-
- error = ptrViERender->AddRenderer(videoChannel, _vieCocoaRenderView2,
- 1, 0.0, 0.0, 1.0, 1.0);
- if (error == -1)
- {
- printf("ERROR in ViERender::AddRenderer\n");
- return -1;
- }
-
- error = ptrViERender->StartRender(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViERender::StartRender\n");
- return -1;
- }
-
- //
- // Setup codecs
- //
- ViECodec* ptrViECodec = ViECodec::GetInterface(ptrViE);
- if (ptrViECodec == NULL)
- {
- printf("ERROR in ViECodec::GetInterface\n");
- return -1;
- }
-
- std::cout << std::endl;
- std::cout << "Available codecs:" << std::endl;
-
- // Check available codecs and prepare receive codecs
- VideoCodec videoCodec;
- memset(&videoCodec, 0, sizeof(VideoCodec));
- unsigned int codecIdx = 0;
- for (codecIdx = 0;
- codecIdx < ptrViECodec->NumberOfCodecs();
- codecIdx++)
- {
- error = ptrViECodec->GetCodec(codecIdx, videoCodec);
- if (error == -1)
- {
- printf("ERROR in ViECodec::GetCodec\n");
- return -1;
- }
-
- error = ptrViECodec->SetReceiveCodec(videoChannel,
- videoCodec);
- if (error == -1)
- {
- printf("ERROR in ViECodec::SetReceiveCodec\n");
- return -1;
- }
- if (videoCodec.codecType != kVideoCodecRED &&
- videoCodec.codecType != kVideoCodecULPFEC)
- {
- std::cout << " " << codecIdx+1 << ". " << videoCodec.plName
- << std::endl;
- }
- }
- std::cout << std::endl;
- std::cout << "Choose codec: ";
-// std::getline(std::cin, str);
-// codecIdx = atoi(str.c_str()) - 1;
-
-
- error = ptrViECapture->ShowCaptureSettingsDialogBox("unique",10, "mytitle");
- codecIdx = 0;
- error = ptrViECodec->GetCodec(codecIdx, videoCodec);
- if (error == -1)
- {
- printf("ERROR in ViECodec::GetCodec\n");
- return -1;
- }
-
- error = ptrViECodec->SetSendCodec(videoChannel, videoCodec);
- if (error == -1)
- {
- printf("ERROR in ViECodec::SetSendCodec\n");
- return -1;
- }
-
- //
- // Address settings
- //
- ViENetwork* ptrViENetwork =
- ViENetwork::GetInterface(ptrViE);
- if (ptrViENetwork == NULL)
- {
- printf("ERROR in ViENetwork::GetInterface\n");
- return -1;
- }
-
- const char* ipAddress = "127.0.0.1";
- const unsigned short rtpPort = 6000;
- error = ptrViENetwork->SetLocalReceiver(videoChannel, rtpPort);
- if (error == -1)
- {
- printf("ERROR in ViENetwork::SetLocalReceiver\n");
- return -1;
- }
-
- error = ptrViEBase->StartReceive(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViENetwork::StartReceive\n");
- return -1;
- }
-
- error = ptrViENetwork->SetSendDestination(videoChannel,
- ipAddress, rtpPort);
- if (error == -1)
- {
- printf("ERROR in ViENetwork::SetSendDestination\n");
- return -1;
- }
-
- error = ptrViEBase->StartSend(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViENetwork::StartSend\n");
- return -1;
- }
-
-
- //********************************************************
- // Engine started
- //********************************************************
-
-
- // Call started
- std::cout << std::endl;
- std::cout << "Loopback call started" << std::endl;
- std::cout << std::endl << std::endl;
- std::cout << "Press enter to stop...";
-// [[NSRunLoop currentRunLoop] runUntilDate:[NSDate dateWithTimeIntervalSinceNow:1]];
-// std::getline(std::cin, str);
- usleep(5 * 1000 * 1000);
-
- //int i = 0;
-// while(1)
-// {
-// NSLog(@"app iteration %d", i);
-// i++;
-// [[NSRunLoop currentRunLoop] runUntilDate:[NSDate dateWithTimeIntervalSinceNow:1]];
-// std::getline(std::cin, str);
-// if(i > 3)
-// {
-// break;
-// }
-// }
-
- //********************************************************
- // Testing finished. Tear down Video Engine
- //********************************************************
-
- error = ptrViEBase->StopReceive(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViEBase::StopReceive\n");
- return -1;
- }
-
- error = ptrViEBase->StopSend(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViEBase::StopSend\n");
- return -1;
- }
-
- error = ptrViERender->StopRender(captureId);
- if (error == -1)
- {
- printf("ERROR in ViERender::StopRender\n");
- return -1;
- }
-
- error = ptrViERender->RemoveRenderer(captureId);
- if (error == -1)
- {
- printf("ERROR in ViERender::RemoveRenderer\n");
- return -1;
- }
-
- error = ptrViERender->StopRender(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViERender::StopRender\n");
- return -1;
- }
-
- error = ptrViERender->RemoveRenderer(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViERender::RemoveRenderer\n");
- return -1;
- }
-
- error = ptrViECapture->StopCapture(captureId);
- if (error == -1)
- {
- printf("ERROR in ViECapture::StopCapture\n");
- return -1;
- }
-
- error = ptrViECapture->DisconnectCaptureDevice(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViECapture::DisconnectCaptureDevice\n");
- return -1;
- }
-
- error = ptrViECapture->ReleaseCaptureDevice(captureId);
- if (error == -1)
- {
- printf("ERROR in ViECapture::ReleaseCaptureDevice\n");
- return -1;
- }
-
- error = ptrViEBase->DeleteChannel(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViEBase::DeleteChannel\n");
- return -1;
- }
-
- int remainingInterfaces = 0;
- remainingInterfaces = ptrViECodec->Release();
- remainingInterfaces += ptrViECapture->Release();
- remainingInterfaces += ptrViERtpRtcp->Release();
- remainingInterfaces += ptrViERender->Release();
- remainingInterfaces += ptrViENetwork->Release();
- remainingInterfaces += ptrViEBase->Release();
- if (remainingInterfaces > 0)
- {
- printf("ERROR: Could not release all interfaces\n");
- return -1;
- }
-
- bool deleted = VideoEngine::Delete(ptrViE);
- if (deleted == false)
- {
- printf("ERROR in VideoEngine::Delete\n");
- return -1;
- }
-
- NSLog(@"Finished function");
- return 0;
-
- //
- // END: VideoEngine 3.0 Sample Code
- //
- // ===================================================================
-}
-
-
-
-
--(IBAction)handleRestart:(id)sender
-{
-// [self stopLooback];
-// [self startLoopback];
- [self ioLooback];
-}
-@end
diff --git a/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUI_Prefix.pch b/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUI_Prefix.pch
deleted file mode 100644
index 72b5870..0000000
--- a/trunk/src/video_engine/main/test/SimpleCocoaGUI/SimpleCocoaGUI_Prefix.pch
+++ /dev/null
@@ -1,7 +0,0 @@
-//
-// Prefix header for all source files of the 'SimpleCocoaGUI' target in the 'SimpleCocoaGUI' project
-//
-
-#ifdef __OBJC__
- #import <Cocoa/Cocoa.h>
-#endif
diff --git a/trunk/src/video_engine/main/test/SimpleCocoaGUI/main.m b/trunk/src/video_engine/main/test/SimpleCocoaGUI/main.m
deleted file mode 100644
index 9d52a1c..0000000
--- a/trunk/src/video_engine/main/test/SimpleCocoaGUI/main.m
+++ /dev/null
@@ -1,12 +0,0 @@
-//
-// main.m
-// SimpleCocoaGUI
-//
-//
-
-#import <Cocoa/Cocoa.h>
-
-int main(int argc, char *argv[])
-{
- return NSApplicationMain(argc, (const char **) argv);
-}
diff --git a/trunk/src/video_engine/main/test/WindowsTest/Capture.rc b/trunk/src/video_engine/main/test/WindowsTest/Capture.rc
deleted file mode 100644
index 962256c..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/Capture.rc
+++ /dev/null
@@ -1,255 +0,0 @@
-// Microsoft Visual C++ generated resource script.
-//
-#include "resource.h"
-
-#define APSTUDIO_READONLY_SYMBOLS
-/////////////////////////////////////////////////////////////////////////////
-//
-// Generated from the TEXTINCLUDE 2 resource.
-//
-#include "afxres.h"
-
-/////////////////////////////////////////////////////////////////////////////
-#undef APSTUDIO_READONLY_SYMBOLS
-
-/////////////////////////////////////////////////////////////////////////////
-// Korean resources
-
-#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_KOR)
-#ifdef _WIN32
-LANGUAGE LANG_KOREAN, SUBLANG_DEFAULT
-#pragma code_page(949)
-#endif //_WIN32
-
-#ifdef APSTUDIO_INVOKED
-/////////////////////////////////////////////////////////////////////////////
-//
-// TEXTINCLUDE
-//
-
-1 TEXTINCLUDE
-BEGIN
- "resource.h\0"
-END
-
-2 TEXTINCLUDE
-BEGIN
- "#include ""afxres.h""\r\n"
- "\0"
-END
-
-3 TEXTINCLUDE
-BEGIN
- "#define _AFX_NO_SPLITTER_RESOURCES\r\n"
- "#define _AFX_NO_OLE_RESOURCES\r\n"
- "#define _AFX_NO_TRACKER_RESOURCES\r\n"
- "#define _AFX_NO_PROPERTY_RESOURCES\r\n"
- "\r\n"
- "#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_KOR)\r\n"
- "#ifdef _WIN32\r\n"
- "LANGUAGE 18, 1\r\n"
- "#pragma code_page(949)\r\n"
- "#endif //_WIN32\r\n"
- "#include ""res\\Capture.rc2"" // non-Microsoft Visual C++ edited resources\r\n"
- "#include ""l.kor\\afxres.rc"" // Standard components\r\n"
- "#endif\r\n"
- "\0"
-END
-
-#endif // APSTUDIO_INVOKED
-
-
-/////////////////////////////////////////////////////////////////////////////
-//
-// Version
-//
-
-VS_VERSION_INFO VERSIONINFO
- FILEVERSION 1,0,0,1
- PRODUCTVERSION 1,0,0,1
- FILEFLAGSMASK 0x3fL
-#ifdef _DEBUG
- FILEFLAGS 0x1L
-#else
- FILEFLAGS 0x0L
-#endif
- FILEOS 0x4L
- FILETYPE 0x1L
- FILESUBTYPE 0x0L
-BEGIN
-END
-
-#endif // Korean resources
-/////////////////////////////////////////////////////////////////////////////
-
-
-/////////////////////////////////////////////////////////////////////////////
-// Swedish resources
-
-#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_SVE)
-#ifdef _WIN32
-LANGUAGE LANG_SWEDISH, SUBLANG_DEFAULT
-#pragma code_page(1252)
-#endif //_WIN32
-
-/////////////////////////////////////////////////////////////////////////////
-//
-// Dialog
-//
-
-IDD_SLAVE_CHANNEL DIALOGEX 0, 0, 677, 358
-STYLE DS_SETFONT | DS_MODALFRAME | DS_FIXEDSYS | WS_POPUP | WS_CAPTION | WS_SYSMENU
-CAPTION "Slave channel"
-FONT 8, "MS Shell Dlg", 400, 0, 0x1
-BEGIN
- CONTROL "",IDC_IPADDRESS1,"SysIPAddress32",WS_TABSTOP,485,18,105,15
- EDITTEXT IDC_LOCAL_PORT1,631,18,36,16,ES_AUTOHSCROLL
- LTEXT "IP-address",IDC_STATIC,495,7,42,9
- LTEXT "Local Port",IDC_STATIC,633,7,37,9
- EDITTEXT IDC_REMOTE_PORT1,593,18,36,16,ES_AUTOHSCROLL
- LTEXT "Port",IDC_STATIC,595,7,17,9
- CONTROL "Ext",IDC_EXTTRANSPORT,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,489,41,29,12
- LTEXT "delay",IDC_STATIC,589,41,21,9
- COMBOBOX IDC_PACKETLOSS,535,40,45,82,CBS_DROPDOWN | WS_DISABLED | WS_VSCROLL | WS_TABSTOP
- COMBOBOX IDC_DELAY,611,40,45,82,CBS_DROPDOWN | WS_DISABLED | WS_VSCROLL | WS_TABSTOP
-END
-
-
-/////////////////////////////////////////////////////////////////////////////
-//
-// DESIGNINFO
-//
-
-#ifdef APSTUDIO_INVOKED
-GUIDELINES DESIGNINFO
-BEGIN
- IDD_SLAVE_CHANNEL, DIALOG
- BEGIN
- LEFTMARGIN, 7
- RIGHTMARGIN, 670
- TOPMARGIN, 7
- BOTTOMMARGIN, 351
- END
-END
-#endif // APSTUDIO_INVOKED
-
-#endif // Swedish resources
-/////////////////////////////////////////////////////////////////////////////
-
-
-/////////////////////////////////////////////////////////////////////////////
-// Neutral (Sys. Default) resources
-
-#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_NEUSD)
-#ifdef _WIN32
-LANGUAGE LANG_NEUTRAL, SUBLANG_SYS_DEFAULT
-#pragma code_page(1252)
-#endif //_WIN32
-
-/////////////////////////////////////////////////////////////////////////////
-//
-// Dialog
-//
-
-IDD_DXQUALITY_DIALOG DIALOGEX 0, 0, 699, 385
-STYLE DS_ABSALIGN | DS_SETFONT | DS_MODALFRAME | DS_3DLOOK | WS_MINIMIZEBOX | WS_POPUP | WS_VISIBLE | WS_CAPTION | WS_SYSMENU
-EXSTYLE WS_EX_WINDOWEDGE | WS_EX_STATICEDGE | WS_EX_APPWINDOW | WS_EX_NOINHERITLAYOUT
-CAPTION "webrtc ViE test program"
-FONT 9, "Arial", 400, 0, 0x0
-BEGIN
- PUSHBUTTON "Start Send",IDC_STARTSEND,589,270,50,19
- PUSHBUTTON "Stop Send",IDC_STOPSend,639,270,50,19
- PUSHBUTTON "Start Listen",IDC_STARTLISTEN,589,291,50,19
- PUSHBUTTON "StopListen",IDC_STOPLISTEN,639,291,50,19
- CONTROL "",IDC_LIVEVIDEO,"Static",SS_BITMAP | SS_CENTERIMAGE | SS_SUNKEN,450,179,139,101
- COMBOBOX IDC_DEVICE,487,14,185,30,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
- CTEXT "Select Capture Device",IDC_STATIC,485,7,78,8
- COMBOBOX IDC_CODEC_LIST,490,90,58,30,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
- LTEXT "Codec",IDC_STATIC,490,82,21,8
- COMBOBOX IDC_CODEC_SIZE,627,90,62,30,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
- LTEXT "Codec Size",IDC_STATIC,611,82,36,8
- CONTROL "",IDC_IPADDRESS1,"SysIPAddress32",WS_TABSTOP,490,46,90,13
- EDITTEXT IDC_LOCAL_PORT1,615,46,31,14,ES_AUTOHSCROLL
- LTEXT "IP-address",IDC_STATIC,498,37,36,8
- LTEXT "Local Port",IDC_STATIC,616,36,32,8
- LTEXT "Start Bitrate",IDC_STATIC,553,80,37,8
- COMBOBOX IDC_BITRATE,558,90,49,30,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
- EDITTEXT IDC_REMOTE_PORT1,582,46,31,14,ES_AUTOHSCROLL
- LTEXT "Port",IDC_STATIC,584,37,14,8
- GROUPBOX "Remote client 1",IDC_STATIC,487,27,203,50
- LTEXT "Max FrameRate",IDC_STATIC,488,106,50,8
- COMBOBOX IDC_MIN_FRAME_RATE,488,115,48,82,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
- CONTROL "",IDC_CAPTURE,"Static",SS_BITMAP | SS_CENTERIMAGE | SS_REALSIZEIMAGE,7,7,418,276
- CONTROL "TMMBR",IDC_TMMBR,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,486,138,40,10
- GROUPBOX "Standard Protection",IDC_STATIC,607,138,72,55
- CONTROL "None",IDC_PROT_NONE,"Button",BS_AUTORADIOBUTTON | WS_GROUP,615,146,33,10
- CONTROL "NACK",IDC_PROT_NACK,"Button",BS_AUTORADIOBUTTON,615,165,35,10
- CONTROL "FEC",IDC_PROT_FEC,"Button",BS_AUTORADIOBUTTON,615,155,30,10
- CONTROL "NACK & FEC",IDC_PROT_NACKFEC,"Button",BS_AUTORADIOBUTTON,615,174,52,10
- COMBOBOX IDC_RTCPMODE,571,119,80,57,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
- LTEXT "RTCP Mode",IDC_STATIC,571,110,39,8
- LISTBOX IDC_INFORMATION,476,309,214,63,LBS_SORT | LBS_NOINTEGRALHEIGHT | WS_VSCROLL | WS_TABSTOP
- COMBOBOX IDC_PACKETBURST,653,118,36,57,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
- LTEXT "Packet Burst",IDC_STATIC,649,109,40,8
- CONTROL "Stop Log",IDC_FREEZELOG,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,465,292,44,10
- PUSHBUTTON "Version",IDC_VERSION,530,291,55,16
- CONTROL "Ext",IDC_EXTTRANSPORT,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,493,66,25,10
- LTEXT "loss",IDC_STATIC,519,66,15,8
- LTEXT "delay",IDC_STATIC,578,66,18,8
- COMBOBOX IDC_PACKETLOSS,533,65,38,82,CBS_DROPDOWN | WS_DISABLED | WS_VSCROLL | WS_TABSTOP
- COMBOBOX IDC_DELAY,598,65,38,82,CBS_DROPDOWN | WS_DISABLED | WS_VSCROLL | WS_TABSTOP
- PUSHBUTTON "Record Incoming",IDC_BTN_RECORD_INCOMING,587,198,69,14
- PUSHBUTTON "Record outgoing",IDC_BTN_RECORD_OUTGOING,587,212,69,14
- PUSHBUTTON "Create Slave",IDC_BTN_CREATE_SLAVE,586,231,50,14
-END
-
-
-/////////////////////////////////////////////////////////////////////////////
-//
-// DESIGNINFO
-//
-
-#ifdef APSTUDIO_INVOKED
-GUIDELINES DESIGNINFO
-BEGIN
- IDD_DXQUALITY_DIALOG, DIALOG
- BEGIN
- LEFTMARGIN, 7
- RIGHTMARGIN, 690
- VERTGUIDE, 321
- VERTGUIDE, 372
- VERTGUIDE, 425
- VERTGUIDE, 465
- TOPMARGIN, 7
- BOTTOMMARGIN, 372
- END
-END
-#endif // APSTUDIO_INVOKED
-
-#endif // Neutral (Sys. Default) resources
-/////////////////////////////////////////////////////////////////////////////
-
-
-
-#ifndef APSTUDIO_INVOKED
-/////////////////////////////////////////////////////////////////////////////
-//
-// Generated from the TEXTINCLUDE 3 resource.
-//
-#define _AFX_NO_SPLITTER_RESOURCES
-#define _AFX_NO_OLE_RESOURCES
-#define _AFX_NO_TRACKER_RESOURCES
-#define _AFX_NO_PROPERTY_RESOURCES
-
-#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_KOR)
-#ifdef _WIN32
-LANGUAGE 18, 1
-#pragma code_page(949)
-#endif //_WIN32
-#include "res\Capture.rc2" // non-Microsoft Visual C++ edited resources
-#include "l.kor\afxres.rc" // Standard components
-#endif
-
-/////////////////////////////////////////////////////////////////////////////
-#endif // not APSTUDIO_INVOKED
-
diff --git a/trunk/src/video_engine/main/test/WindowsTest/CaptureDevicePool.cc b/trunk/src/video_engine/main/test/WindowsTest/CaptureDevicePool.cc
deleted file mode 100644
index 8f7c9a2..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/CaptureDevicePool.cc
+++ /dev/null
@@ -1,98 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "CaptureDevicePool.h"
-#include "map_wrapper.h"
-#include <string.h>
-#include <assert.h>
-#include "critical_section_wrapper.h"
-#include "vie_file.h"
-
-CaptureDevicePool::CaptureDevicePool(VideoEngine* videoEngine):
-_critSect(*CriticalSectionWrapper::CreateCriticalSection()),
-_vieCapture(ViECapture::GetInterface(videoEngine)),
-_vieFile(ViEFile::GetInterface(videoEngine))
-{
-}
-
-CaptureDevicePool::~CaptureDevicePool(void)
-{
- assert(_deviceMap.Size()==0);
- _vieCapture->Release();
- _vieFile->Release();
- delete &_critSect;
-}
-
-WebRtc_Word32 CaptureDevicePool::GetCaptureDevice(int& captureId, const char* uniqeDeviceName)
-{
- CriticalSectionScoped cs(_critSect);
- DeviceItem* device=NULL;
-
- for(MapItem* item=_deviceMap.First();
- item!=NULL;
- item=_deviceMap.Next(item))
- {
- //Found the device?
- if(strcmp(uniqeDeviceName,(static_cast<DeviceItem*>( item->GetItem()))->uniqeDeviceName)==0)
- {
- device=static_cast<DeviceItem*>( item->GetItem());
- device->refCount++;
- captureId=device->captureId;
- return 0;
- }
- }
- device = new DeviceItem;
- strncpy(device->uniqeDeviceName,uniqeDeviceName,255);
-
-
- // Device does not exist. Create it.
- WebRtc_Word32 result=_vieCapture->AllocateCaptureDevice(device->uniqeDeviceName,strlen(device->uniqeDeviceName),device->captureId);
- if(result==0)
- {
- //CaptureCapability cap;
- /*cap.height=1080;
- cap.width=1920;
- cap.maxFPS=25;
- cap.interlaced=true;*/
- // result=_vieCapture->StartCapture(device->captureId,cap);
- result=_vieFile->SetCaptureDeviceImage(device->captureId,"captureDeviceImage.jpg");
- }
- captureId=device->captureId;
- _deviceMap.Insert(captureId,device);
- device->refCount++;
-
- return result;
-
-
-}
-WebRtc_Word32 CaptureDevicePool::ReturnCaptureDevice(int captureId)
-{
- CriticalSectionScoped cs(_critSect);
-
- MapItem* mapItem=_deviceMap.Find(captureId);
- if(!mapItem)
- return -1;
-
- DeviceItem* item=static_cast<DeviceItem*> (mapItem->GetItem());
- if(!item)
- return 0;
- item->refCount--;
- WebRtc_Word32 result=0;
-
- if(item->refCount==0)
- {
- result=_vieCapture->ReleaseCaptureDevice(captureId);
-
- _deviceMap.Erase(mapItem);
- delete item;
-
- }
- return result;
-}
diff --git a/trunk/src/video_engine/main/test/WindowsTest/CaptureDevicePool.h b/trunk/src/video_engine/main/test/WindowsTest/CaptureDevicePool.h
deleted file mode 100644
index 104b84f..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/CaptureDevicePool.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#pragma once
-
-#include "common_types.h"
-
-#include "vie_base.h"
-#include "vie_capture.h"
-#include "vie_file.h"
-#include "map_wrapper.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-}
-using namespace webrtc;
-class CaptureDevicePool
-{
-public:
- CaptureDevicePool(VideoEngine* videoEngine);
- ~CaptureDevicePool(void);
- WebRtc_Word32 GetCaptureDevice(int& captureId, const char uniqeDeviceName[256]);
- WebRtc_Word32 ReturnCaptureDevice(int captureId);
-
- private:
- struct DeviceItem
- {
- int captureId;
- WebRtc_Word32 refCount;
- char uniqeDeviceName[256];
- DeviceItem()
- {
- captureId=-1;
- refCount=0;
- }
- };
- CriticalSectionWrapper& _critSect;
- ViECapture* _vieCapture;
- ViEFile* _vieFile;
- MapWrapper _deviceMap;
-
-};
diff --git a/trunk/src/video_engine/main/test/WindowsTest/ChannelDlg.cc b/trunk/src/video_engine/main/test/WindowsTest/ChannelDlg.cc
deleted file mode 100644
index cccfa8e..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/ChannelDlg.cc
+++ /dev/null
@@ -1,1297 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "ChannelDlg.h"
-#include "VideoSize.h"
-#include "CaptureDevicePool.h"
-#include "ChannelPool.h"
-
-#include <Mmsystem.h>
-#include <dbt.h>
-
-
-#include "assert.h"
-
-
-#include <process.h> // threads.
-
-#if defined _WIN32
- #define SLEEP_10_SEC ::Sleep(10000)
- #define GET_TIME_IN_MS timeGetTime
-#endif
-
-// Hack to convert char to TCHAR, using two buffers to be able to
-// call twice in the same statement
-TCHAR convertTemp1[256] = {0};
-TCHAR convertTemp2[256] = {0};
-bool convertBufferSwitch(false);
-TCHAR* CharToTchar(const char* str, int len)
-{
-#ifdef _UNICODE
- TCHAR* temp = convertBufferSwitch ? convertTemp1 : convertTemp2;
- convertBufferSwitch = !convertBufferSwitch;
- memset(temp, 0, sizeof(convertTemp1));
- MultiByteToWideChar(CP_UTF8, 0, str, len, temp, 256);
- return temp;
-#else
- return str;
-#endif
-}
-
-// Hack to convert TCHAR to char
-char convertTemp3[256] = {0};
-char* TcharToChar(TCHAR* str, int len)
-{
-#ifdef _UNICODE
- memset(convertTemp3, 0, sizeof(convertTemp3));
- WideCharToMultiByte(CP_UTF8, 0, str, len, convertTemp3, 256, 0, 0);
- return convertTemp3;
-#else
- return str;
-#endif
-}
-
-/////////////////////////////////////////////////////////////////////////////
-// CDXChannelDlg dialog
-
-CDXChannelDlg::CDXChannelDlg(VideoEngine* videoEngine,
- CaptureDevicePool& captureDevicePool,
- ChannelPool& channelPool,
- void* voiceEngine
- ,CWnd* pParent,CDXChannelDlgObserver* observer,
- int parentChannel/*=-1*/)
- : CDialog(CDXChannelDlg::IDD, pParent),
- _canAddLog(true),
- _dialogObserver(observer),
- _videoEngine(videoEngine),
- _captureDevicePool(captureDevicePool),
- _channelPool(channelPool),
- _parentChannel(parentChannel),
-#ifndef NO_VOICE_ENGINE
- _voiceEngine((VoiceEngine*) voiceEngine),
-#endif
- _callbackEvent(::CreateEvent( NULL, FALSE, FALSE, NULL)),
- _externalTransport(NULL)
-{
- strcpy(_logMsg,"");
- _channelId = -1;
- _audioChannel=-1;
- _captureId=-1;
-
- //_transport = NULL;
-
-
- //{{AFX_DATA_INIT(CDXChannelDlg)
- //}}AFX_DATA_INIT
- // Note that LoadIcon does not require a subsequent DestroyIcon in Win32
-
- InitializeCriticalSection(&_critCallback);
- unsigned int threadID;
- _callbackThread=(HANDLE)_beginthreadex(NULL,1024*1024,CallbackThread,(void*)this,0, &threadID);
-
-
-}
-
-void CDXChannelDlg::DoDataExchange(CDataExchange* pDX)
-{
- CDialog::DoDataExchange(pDX);
- //{{AFX_DATA_MAP(CDXChannelDlg)
- DDX_Control(pDX, IDC_DEVICE, m_ctrlDevice);
- DDX_Control(pDX, IDC_CODEC_LIST, m_ctrlCodec);
- DDX_Control(pDX, IDC_CAPTURE, m_ctrlLiveRemoteVideo);
- DDX_Control(pDX, IDC_LIVEVIDEO, m_ctrlLiveVideo);
- DDX_Control(pDX, IDC_LOCAL_PORT1, m_localPort1);
- DDX_Control(pDX, IDC_REMOTE_PORT1, m_remotePort1);
- DDX_Control(pDX, IDC_IPADDRESS1, m_remoteIp1);
- DDX_Control(pDX, IDC_CODEC_SIZE, m_ctrlCodecSize);
- DDX_Control(pDX, IDC_RTCPMODE, m_ctrlRtcpMode);
- DDX_Control(pDX, IDC_PACKETBURST, m_ctrlPacketBurst);
- DDX_Control(pDX, IDC_BITRATE, m_ctrlBitrate);
- DDX_Control(pDX, IDC_MIN_FRAME_RATE, m_ctrlMinFrameRate);
- DDX_Control(pDX, IDC_TMMBR,m_cbTmmbr);
- DDX_Control(pDX, IDC_EXTTRANSPORT,m_cbExternalTransport);
- DDX_Control(pDX, IDC_PACKETLOSS,m_ctrlPacketLoss);
- DDX_Control(pDX, IDC_DELAY,m_ctrlDelay);
- DDX_Control(pDX, IDC_FREEZELOG,m_cbFreezeLog);
- DDX_Control(pDX,IDC_INFORMATION,m_ctrlInfo);
- //}}AFX_DATA_MAP
-}
-
-// ON_WM_SYSKEYDOWN ALT+key
-
-BEGIN_MESSAGE_MAP(CDXChannelDlg, CDialog)
- //{{AFX_MSG_MAP(CDXChannelDlg)
- ON_WM_SYSCOMMAND()
- ON_WM_RBUTTONUP()
- //ON_WM_DEVICECHANGE()
- ON_WM_PAINT()
- ON_WM_QUERYDRAGICON()
- ON_BN_CLICKED(IDC_STARTSEND, OnStartSend)
- ON_BN_CLICKED(IDC_STOPSend, OnStopSend)
- //ON_WM_TIMER()
- ON_WM_DESTROY()
- //}}AFX_MSG_MAP
- ON_CBN_SELCHANGE(IDC_CODEC_LIST, OnCbnSelchangeCodecList)
- ON_CBN_SELCHANGE(IDC_DEVICE, OnCbnSelchangeDevice)
- ON_CBN_SELCHANGE(IDC_CODEC_SIZE, OnCbnSelchangeSize)
- ON_CBN_SELCHANGE(IDC_BITRATE, OnCbnSelchangeBitrate)
- //ON_MESSAGE(WM_DISPLAYCHANGE, OnDisplayChange)
- ON_CBN_SELCHANGE(IDC_MIN_FRAME_RATE, OnCbnSelchangeMinFrameRate)
- ON_BN_CLICKED(IDC_STARTLISTEN, OnBnClickedStartlisten)
- ON_BN_CLICKED(IDC_STOPLISTEN, OnBnClickedStoplisten)
- ON_BN_CLICKED(IDC_TMMBR, &CDXChannelDlg::OnBnClickedTmmbr)
- ON_CBN_SELCHANGE(IDC_RTCPMODE, &CDXChannelDlg::OnCbnSelchangeRtcpmode)
- ON_BN_CLICKED(IDC_PROT_NACK, &CDXChannelDlg::OnBnClickedProtNack)
- ON_BN_CLICKED(IDC_PROT_NONE, &CDXChannelDlg::OnBnClickedProtNone)
- ON_BN_CLICKED(IDC_PROT_FEC, &CDXChannelDlg::OnBnClickedProtFec)
- ON_BN_CLICKED(IDC_FREEZELOG, &CDXChannelDlg::OnBnClickedFreezelog)
- ON_BN_CLICKED(IDC_VERSION, &CDXChannelDlg::OnBnClickedVersion)
- ON_BN_CLICKED(IDC_EXTTRANSPORT, &CDXChannelDlg::OnBnClickedExttransport)
- ON_CBN_SELCHANGE(IDC_PACKETLOSS, &CDXChannelDlg::OnCbnSelchangePacketloss)
- ON_CBN_SELCHANGE(IDC_DELAY, &CDXChannelDlg::OnCbnSelchangeDelay)
- ON_BN_CLICKED(IDC_BTN_RECORD_INCOMING, &CDXChannelDlg::OnBnClickedBtnRecordIncoming)
- ON_BN_CLICKED(IDC_BTN_RECORD_OUTGOING, &CDXChannelDlg::OnBnClickedBtnRecordOutgoing)
- ON_BN_CLICKED(IDC_BTN_CREATE_SLAVE, &CDXChannelDlg::OnBnClickedBtnCreateSlave)
- ON_BN_CLICKED(IDC_PROT_NACKFEC, &CDXChannelDlg::OnBnClickedProtNackFec)
-END_MESSAGE_MAP()
-
-/////////////////////////////////////////////////////////////////////////////
-// CDXChannelDlg message handlers
-
-
-BOOL CDXChannelDlg::OnInitDialog()
-{
- CDialog::OnInitDialog();
-
- // Set the icon for this dialog. The framework does this automatically
- // when the application's main window is not a dialog
- SetIcon(m_hIcon, TRUE); // Set big icon
- SetIcon(m_hIcon, FALSE); // Set small icon
-
-
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("5"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("6"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("7"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("8"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("9"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("10"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("11"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("12"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("13"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("14"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("15"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("16"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("17"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("18"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("19"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("20"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("21"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("22"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("23"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("24"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("25"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("26"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("27"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("28"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("29"));
- ::SendMessage(m_ctrlMinFrameRate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("30"));
- m_ctrlMinFrameRate.SetCurSel(25);
-
- // Codec sizes
- for(VideoSize i=UNDEFINED;i<NUMBER_OF_VIDEO_SIZE;i=VideoSize(i+1))
- {
- char sizeStr[64];
- int width=0;
- int height=0;
- GetWidthHeight(i,width,height);
- sprintf(sizeStr,"%d x %d",width,height);
- ::SendMessage(m_ctrlCodecSize.m_hWnd, CB_ADDSTRING, 0,(LPARAM) CharToTchar(sizeStr,-1));
- }
- m_ctrlCodecSize.SetCurSel(8);
-
- // RTCP mode
- /*
- kRtcpNone = 0,
- kRtcpCompound_RFC4585 = 1,
- kRtcpNonCompound_RFC5506 = 2 */
- ::SendMessage(m_ctrlRtcpMode.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("RTCP_NONE"));
- ::SendMessage(m_ctrlRtcpMode.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("RTCP_COMPOUND_RFC4585"));
- ::SendMessage(m_ctrlRtcpMode.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("RTCP_NON_COMPOUND_RFC5506"));
- m_ctrlRtcpMode.SetCurSel(2);
-
-
- //Packet Burst
- ::SendMessage(m_ctrlPacketBurst.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("0"));
- ::SendMessage(m_ctrlPacketBurst.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("10"));
- ::SendMessage(m_ctrlPacketBurst.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("20"));
- ::SendMessage(m_ctrlPacketBurst.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("30"));
- m_ctrlPacketBurst.SetCurSel(0);
-
-
- //Send Bitrate
- ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("50"));
- ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("100"));
- ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("200"));
- ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("300"));
- ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("500"));
- ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("1000"));
- ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("2000"));
- ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("3000"));
- ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("4000"));
- ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("5000"));
- ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("6000"));
- ::SendMessage(m_ctrlBitrate.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("7000"));
-
- m_ctrlBitrate.SetCurSel(3);
-
- // External transport packet loss
- ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("0"));
- ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("2"));
- ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("4"));
- ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("6"));
- ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("8"));
- ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("10"));
- ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("12"));
- ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("14"));
- ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("16"));
- ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("18"));
- ::SendMessage(m_ctrlPacketLoss.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("20"));
- m_ctrlPacketLoss.SetCurSel(0);
-
- // External transport delay
- ::SendMessage(m_ctrlDelay.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("0"));
- ::SendMessage(m_ctrlDelay.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("30"));
- ::SendMessage(m_ctrlDelay.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("60"));
- ::SendMessage(m_ctrlDelay.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("90"));
- ::SendMessage(m_ctrlDelay.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("120"));
- ::SendMessage(m_ctrlDelay.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("150"));
- ::SendMessage(m_ctrlDelay.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("180"));
- ::SendMessage(m_ctrlDelay.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("210"));
- m_ctrlDelay.SetCurSel(0);
-
-
- _vieBase=ViEBase::GetInterface(_videoEngine);
- TEST_MUSTPASS(_vieBase==0,-5);
-
- _vieCapture=ViECapture::GetInterface(_videoEngine);
- TEST_MUSTPASS(_vieCapture==0,-5);
-
- _vieRTPRTCP=ViERTP_RTCP::GetInterface(_videoEngine);
- TEST_MUSTPASS(_vieRTPRTCP==0,-5);
-
- _vieRender=ViERender::GetInterface(_videoEngine);
- TEST_MUSTPASS(_vieRender==0,-5);
-
- _vieCodec=ViECodec::GetInterface(_videoEngine);
- TEST_MUSTPASS(_vieCodec==0,-5);
- _vieNetwork=ViENetwork::GetInterface(_videoEngine);
- TEST_MUSTPASS(_vieNetwork==0,-5);
-
- _vieFile=ViEFile::GetInterface(_videoEngine);
- TEST_MUSTPASS(_vieFile==0,-5);
-
-#ifndef NO_VOICE_ENGINE
-
- _veBase = VoEBase::GetInterface(_voiceEngine);
- _veNetwork = VoENetwork::GetInterface(_voiceEngine);
- _veCodec = VoECodec::GetInterface(_voiceEngine);
- _veRTCP = VoERTP_RTCP::GetInterface(_voiceEngine);
- TEST_MUSTPASS(_vieBase->SetVoiceEngine(_voiceEngine),-5);
-#endif
-
- int err = 0;
-
- char str[64];
- bool found = false;
-
-
- int captureIdx = 0;
- while (-1 !=_vieCapture->GetCaptureDevice(captureIdx,str,sizeof(str),NULL,0))
- {
- char* tmp = strstr(str,"(VFW)");
- if (!tmp)
- {
- ::SendMessage(m_ctrlDevice.m_hWnd, CB_ADDSTRING, 0,(LPARAM)CharToTchar(str,-1));
- found = true;
- }
- captureIdx++;
- memset(str, 0, 64);
- }
- WIN32_FIND_DATA FindFileData;
- HANDLE hFind;
- //char fileSearch[256];
- //strcpy(fileSearch,_T("*.avi"));
- hFind = FindFirstFile(_T("*.avi"), &FindFileData);
- if (hFind != INVALID_HANDLE_VALUE)
- {
-
- ::SendMessage(m_ctrlDevice.m_hWnd, CB_ADDSTRING, 0,(LPARAM)(FindFileData.cFileName));
- while(FindNextFile(hFind,&FindFileData))
- {
- ::SendMessage(m_ctrlDevice.m_hWnd, CB_ADDSTRING, 0,(LPARAM)(FindFileData.cFileName));
- }
- FindClose(hFind);
- }
-
-
-
- ::SendMessage(m_ctrlDevice.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("Conference"));
- ::SendMessage(m_ctrlDevice.m_hWnd, CB_ADDSTRING, 0,(LPARAM)_T("None"));
-
- if (!found)
- {
- strncpy(str,"N/A",64);
- ::SendMessage(m_ctrlDevice.m_hWnd, CB_ADDSTRING, 0,(LPARAM)CharToTchar(str,-1));
- }
- m_ctrlDevice.SetCurSel(0);
-
- //Codecs
- int numOfCodecs = _vieCodec->NumberOfCodecs();
- for(int i=0; i<numOfCodecs;++i)
- {
- VideoCodec codec;
- if(-1 !=_vieCodec->GetCodec(i,codec))
- {
- ::SendMessage(m_ctrlCodec.m_hWnd, CB_ADDSTRING, 0,(LPARAM)CharToTchar(codec.plName,-1));
- }
- }
- m_ctrlCodec.SetCurSel(0);
-
-#ifndef NO_VOICE_ENGINE
- CodecInst voiceCodec;
- int numOfVeCodecs = _veCodec->NumOfCodecs();
- for(int i=0; i<numOfVeCodecs;++i)
- {
- if(_veCodec->GetCodec(i,voiceCodec)!=-1)
- {
- if(strncmp(voiceCodec.plname,"ISAC",4)==0)
- break;
-
-
- }
- }
-
- _audioChannel = _veBase->CreateChannel();
-
-
- TEST_MUSTPASS(_veRTCP->SetRTCPStatus(_audioChannel, true),-5);
- TEST_MUSTPASS(_veCodec->SetSendCodec(_audioChannel, voiceCodec),-5);
- TEST_MUSTPASS(_veBase->StartPlayout(_audioChannel),-5);
-
-#endif //NO_VOICE_ENGINE
-
- if(_parentChannel==-1)
- {
- TEST_MUSTPASS(_vieBase->CreateChannel(_channelId),-5);
- }
- else // This is a slave channel
- {
- TEST_MUSTPASS(_vieBase->CreateChannel(_channelId,_parentChannel),-5);
- }
-#ifndef NO_VOICE_ENGINE
- TEST_MUSTPASS(_vieBase->ConnectAudioChannel(_channelId,_audioChannel),-5);
-#endif
-
- _channelPool.AddChannel(_channelId);
-
- //Set Receive codec
- {
- VideoCodec codec;
- int numOfCodecs = _vieCodec->NumberOfCodecs();;
- for(int i=0; i<numOfCodecs;++i)
- {
- if(-1 !=_vieCodec->GetCodec(i,codec))
- {
- if(codec.codecType == webrtc::kVideoCodecVP8)
- {
- codec.codecSpecific.VP8.feedbackModeOn = true;
- codec.codecSpecific.VP8.pictureLossIndicationOn = true;
- }
- TEST_MUSTPASS(_vieCodec->SetReceiveCodec(_channelId,codec),-5);
- }
- }
- }
-
- //TMMBR
- m_cbTmmbr.SetCheck(BST_CHECKED);
- OnBnClickedTmmbr();
-
- //Packet Burst
- m_ctrlPacketBurst.SetCurSel(0);
-
-
- //Protection method none
- CButton *opProtection = (CButton *) GetDlgItem(IDC_PROT_NONE);
- opProtection->SetCheck(BST_CHECKED);
- OnBnClickedProtNone();
-
-
- // Configure the renderer
- ConfigureRender();
-
- TEST_MUSTPASS(_vieCodec->RegisterEncoderObserver(_channelId,*this),kViECodecObserverAlreadyRegistered);
- TEST_MUSTPASS(_vieCodec->RegisterDecoderObserver(_channelId,*this),-5);
-
- TEST_MUSTPASS(_vieBase->RegisterObserver(*this),kViEBaseObserverAlreadyRegistered);
-
-
-
-
- //Set captions based on channel id
- m_remoteIp1.SetAddress(127,0,0,1);
- CString port;
- port.AppendFormat(_T("%d"),11111+_channelId*4);
- m_remotePort1.SetWindowText(port);
- m_localPort1.SetWindowText(port);
-
- CString title;
- this->GetWindowText(title);
- if(_parentChannel==-1)
- {
- title.AppendFormat(_T("%s - channel %d"),title,_channelId);
- }
- else
- {
- title.AppendFormat(_T("%s - slave channel %d - parent %d"),title,_channelId,_parentChannel);
- }
- this->SetWindowText(title);
-
- if(_parentChannel!=-1)
- m_ctrlDevice.EnableWindow(FALSE); //Prevent from changing capture device
-
- return TRUE; // return TRUE unless you set the focus to a control
-}
-
-
-
-void CDXChannelDlg::OnTimer(UINT nIDEvent)
-{
- CDialog::OnTimer(nIDEvent);
-}
-
-void CDXChannelDlg::SetSendCodec()
-{
- // Get the codec stucture
- int codecSel= m_ctrlCodec.GetCurSel();
- VideoCodec codec;
- TEST_MUSTPASS(_vieCodec->GetCodec(codecSel,codec),-5);
-
-
- // Set Codec Size
- VideoSize sizeSel=VideoSize(m_ctrlCodecSize.GetCurSel());
- int width, height;
- GetWidthHeight(sizeSel, width, height);
- codec.width=width;
- codec.height=height;
-
- //Set the codec bitrate
- CString bitrateStr;
- m_ctrlBitrate.GetLBText(m_ctrlBitrate.GetCurSel(), bitrateStr);
- int bitrate = _ttoi(bitrateStr.GetBuffer(0));
- if(codec.codecType!=kVideoCodecI420)
- {
- codec.startBitrate=bitrate;
- codec.maxBitrate=bitrate*4;
- }
-
-
- //Set the codec frame rate
- codec.maxFramerate = m_ctrlMinFrameRate.GetCurSel() +5;
-
- if(strncmp(codec.plName, "VP8", 5) == 0)
- {
- codec.codecSpecific.VP8.feedbackModeOn = true;
- codec.codecSpecific.VP8.pictureLossIndicationOn = true;
- TEST_MUSTPASS(_vieRTPRTCP->SetKeyFrameRequestMethod(_channelId, kViEKeyFrameRequestPliRtcp),-5);
- }else
- {
- TEST_MUSTPASS(_vieRTPRTCP->SetKeyFrameRequestMethod(_channelId, kViEKeyFrameRequestPliRtcp),-5);
- }
- TEST_MUSTPASS(_vieCodec->SetSendCodec(_channelId, codec),-5);
-
- if (codec.codecType == webrtc::kVideoCodecI420)
- { // Need to set the receive codec size
- _vieCodec->SetReceiveCodec(_channelId, codec);
- }
-}
-
-void CDXChannelDlg::SetSendDestination()
-{
- if(_externalTransport)
- return;
-
- BYTE part1, part2, part3, part4;
- char sendIP1[16];
- m_remoteIp1.GetAddress(part1, part2, part3, part4);
- sprintf(sendIP1,"%d.%d.%d.%d",part1,part2,part3,part4);
-
- CString strPort;
- m_remotePort1.GetWindowText(strPort);
- int remotePort1 = _ttoi(strPort.GetString());
-
-#ifdef IPV6
- char* recIP = "::0";
-#else
- char* recIP = "0.0.0.0";
-#endif //IPV6
-
- TEST_MUSTPASS(_vieNetwork->SetSendDestination(_channelId,sendIP1,remotePort1),kViENetworkAlreadySending);
-
- #ifndef NO_VOICE_ENGINE
- m_localPort1.GetWindowText(strPort);
- int localPort1 = _ttoi(strPort.GetString());
- int res=_veBase->SetLocalReceiver(_audioChannel,localPort1+2);
-
- TEST_MUSTPASS(_veBase->SetSendDestination(_audioChannel, remotePort1+2, sendIP1),-5)
- #endif
-
-}
-
-void CDXChannelDlg::SetLocalReceiver()
-{
- if(_externalTransport)
- return;
-
- CString strPort;
- m_localPort1.GetWindowText(strPort);
- int localPort1 = _ttoi(strPort.GetString());
-
-
-
- // May fail because we are sending
- TEST_MUSTPASS(_vieNetwork->SetLocalReceiver(_channelId, localPort1),-5);
-
- #ifndef NO_VOICE_ENGINE
- int res=_veBase->SetLocalReceiver(_audioChannel,localPort1+2);
- #endif
-
-
-
-}
-
-void CDXChannelDlg::SetCaptureDevice()
-{
- if(_parentChannel!=-1) // don't accept changing input on slave channels.
- return;
-
- int camSel=-1;
- camSel=m_ctrlDevice.GetCurSel();
-
- CString captureStr;
- //captureStr.Compare
- m_ctrlDevice.GetLBText(camSel, captureStr);
- if(captureStr!=_T("N/A") != 0)
- {
-
- TEST_MUSTPASS(_vieFile->StopPlayFile(_captureId),kViEFileNotPlaying);
- TEST_MUSTPASS(_vieCapture->DisconnectCaptureDevice(_channelId),kViECaptureDeviceNotConnected);
- TEST_MUSTPASS(_vieRender->RemoveRenderer(_captureId),kViERenderInvalidRenderId);
-
- if(_captureId>=0x1001 && _captureId<0x10FF)// ID is a capture device
- {
- TEST_MUSTPASS(_captureDevicePool.ReturnCaptureDevice(_captureId),-5);
- }
-
- if(captureStr!=_T("None")==0)
- {
- _captureId=-1;
- }
- else if(_tcsstr(captureStr,_T(".avi"))!=NULL ) // Selected an AVI file
- {
- TEST_MUSTPASS(_vieFile->StartPlayFile(TcharToChar(captureStr.GetBuffer(),-1),_captureId,false,webrtc::kFileFormatAviFile),-5);
- TEST_MUSTPASS(_vieRender->AddRenderer(_captureId,m_ctrlLiveVideo.m_hWnd, 0, 0.0f, 0.0f,1.0f,1.0f),-5);
- TEST_MUSTPASS(_vieRender->StartRender(_captureId),-5);
- TEST_MUSTPASS(_vieFile->SendFileOnChannel(_captureId,_channelId),-5);
- TEST_MUSTPASS(_vieFile->StartPlayFileAsMicrophone(_captureId,_channelId,true),-5);
- //TEST_MUSTPASS(_vieFile->StartPlayAudioLocally(_captureId,_channelId),-5);
- }
- else
- {
-
- char captureName[256];
- char uniqueCaptureName[256];
-
- TEST_MUSTPASS(_vieCapture->GetCaptureDevice(camSel,captureName,256,uniqueCaptureName,256),-5);
-
- TEST_MUSTPASS(_captureDevicePool.GetCaptureDevice(_captureId,uniqueCaptureName),-5);
- TEST_MUSTPASS(_vieCapture->StartCapture(_captureId),kViECaptureDeviceAlreadyStarted);
- TEST_MUSTPASS(_vieCapture->RegisterObserver(_captureId,*this),kViECaptureObserverAlreadyRegistered);
-
- TEST_MUSTPASS(_vieRender->AddRenderer(_captureId,m_ctrlLiveVideo.m_hWnd, 0, 0.0f, 0.0f,1.0f,1.0f),-5);
- TEST_MUSTPASS(_vieCapture->ConnectCaptureDevice(_captureId,_channelId),-5);
- TEST_MUSTPASS(_vieRender->StartRender(_captureId),-5);
- }
- }
-
-}
-
-
-
-void CDXChannelDlg::OnBnClickedStartlisten()
-{
-
-
- // Configure the local ports
- SetLocalReceiver();
-
- //Configure the remote destination- needed in order to be able to respond to RTCP messages
- SetSendDestination();
-
-
- #ifndef NO_VOICE_ENGINE
- TEST_MUSTPASS(_veBase->StartReceive(_audioChannel),-5);
- #endif
- TEST_MUSTPASS(_vieBase->StartReceive(_channelId),-5);
-
-
-}
-
-void CDXChannelDlg::OnStartSend()
-{
-
- // Set the send destination
- SetSendDestination();
-
- // Configure the local ports (Needed to be able to receive RTCP
- //SetLocalReceiver();
-
-
- // Set the send codec
- SetSendCodec();
-
- if(_captureId==-1) // If no capture device has been set.
- SetCaptureDevice(); //Set the capture device
-
-
-
- //TEST_MUSTPASS(_vieRTPRTCP->SetStartSequenceNumber(_channelId,1),-5);
-
- // Start sending
- TEST_MUSTPASS(_vieBase->StartSend(_channelId),-5);
-
-
- #ifndef NO_VOICE_ENGINE
- TEST_MUSTPASS(_veBase->StartSend(_audioChannel),-5);
- #endif
-
-
-}
-
-void CDXChannelDlg::ConfigureRender()
-{
- TEST_MUSTPASS(_vieRender->AddRenderer(_channelId,m_ctrlLiveRemoteVideo.m_hWnd, 0, 0.0f, 0.0f,1.0f,1.0f),-5);
-
- TEST_MUSTPASS(_vieFile->SetRenderStartImage(_channelId,"renderStartImage.jpg"),-5);
- TEST_MUSTPASS(_vieRender->StartRender(_channelId),-5);
- TEST_MUSTPASS(_vieFile->SetRenderTimeoutImage(_channelId,"renderTimeoutImage.jpg"),-5);
-
-
-}
-
-
-void CDXChannelDlg::OnStopSend()
-{
-
- #ifndef NO_VOICE_ENGINE
- TEST_MUSTPASS(_veBase->StopSend(_audioChannel),-5);
- #endif
-
-
- TEST_MUSTPASS(_vieBase->StopSend(_channelId),kViEBaseNotSending); // Accept error Not sending
-
-
-}
-void CDXChannelDlg::OnBnClickedStoplisten()
-{
-
-
- #ifndef NO_VOICE_ENGINE
- TEST_MUSTPASS(_veBase->StopReceive(_audioChannel),-5);
- #endif
- TEST_MUSTPASS(_vieBase->StopReceive(_channelId),-5);
-}
-
-
-void CDXChannelDlg::OnDestroy()
-{
-
- OnStopSend();
- OnBnClickedStoplisten();
-
- if(_vieCapture && _parentChannel==-1)
- {
- _vieCapture->DisconnectCaptureDevice(_channelId);
- _captureDevicePool.ReturnCaptureDevice(_captureId);
- }
- if(_vieFile && _parentChannel!=-1)
- {
- TEST_MUSTPASS(_vieFile->StopPlayFile(_captureId),kViEFileNotPlaying);
- }
-
-
-
-
- if(_videoEngine)
- {
- if(_parentChannel==-1)
- {
- _vieCodec->DeregisterEncoderObserver(_channelId);
- }
- _vieBase->DeleteChannel(_channelId);
- _channelPool.RemoveChannel(_channelId);
- }
-
- _videoEngine = NULL;
-#ifndef NO_VOICE_ENGINE
-
- if (_voiceEngine)
- {
- _veBase->DeleteChannel(_audioChannel);
- _veBase->Release();
- _veNetwork->Release();
- _veCodec->Release();
- _veRTCP->Release();
- }
-#endif
-
-
- strcpy(_logMsg,"");
- SetEvent(_callbackEvent);
- MSG msg; // Wait until the callback thread exits. Need to handle messages since the callback thread can call SendMessage when updating UI
- while(WaitForSingleObject(_callbackThread,10)==WAIT_TIMEOUT)
- {
- DWORD ret = PeekMessage( &msg, NULL, 0, 0,PM_REMOVE );
- if (ret >0)
- {
- TranslateMessage(&msg);
- DispatchMessage(&msg);
- }
- }
-
- CloseHandle(_callbackThread);
- CloseHandle(_callbackEvent);
- DeleteCriticalSection(&_critCallback);
-
- TEST_MUSTPASS(_vieCapture->Release()<0,-5);
- TEST_MUSTPASS(_vieRTPRTCP->Release()<0,-5);
- TEST_MUSTPASS(_vieRender->Release()<0,-5);
- TEST_MUSTPASS(_vieCodec->Release()<0,-5);
- TEST_MUSTPASS(_vieNetwork->Release()<0,-5);
- TEST_MUSTPASS(_vieFile->Release()<0,-5);
- TEST_MUSTPASS(_vieBase->Release()<0,-5);
-
-
-
-#ifdef TEST_EXTERNAL_TRANSPORT
- if(_transport)
- delete _transport;
- _transport = NULL;
-#endif
-
- delete _externalTransport;
-
- CDialog::OnDestroy();
- if(_dialogObserver)
- {
- _dialogObserver->ChannelDialogEnded(this);
- }
-}
-
-void CDXChannelDlg::OnCancel()
-{
- DestroyWindow();
-}
-// If you add a minimize button to your dialog, you will need the code below
-// to draw the icon. For MFC applications using the document/view model,
-// this is automatically done for you by the framework.
-
-void CDXChannelDlg::OnPaint()
-{
- if (IsIconic())
- {
- CPaintDC dc(this); // device context for painting
-
- SendMessage(WM_ICONERASEBKGND, (WPARAM) dc.GetSafeHdc(), 0);
-
- // Center icon in client rectangle
- int cxIcon = GetSystemMetrics(SM_CXICON);
- int cyIcon = GetSystemMetrics(SM_CYICON);
- CRect rect;
- GetClientRect(&rect);
- int x = (rect.Width() - cxIcon + 1) / 2;
- int y = (rect.Height() - cyIcon + 1) / 2;
-
- // Draw the icon
- dc.DrawIcon(x, y, m_hIcon);
- }
- else
- {
- CDialog::OnPaint();
- }
-}
-
-BOOL CDXChannelDlg::OnDeviceChange( UINT nID, DWORD lParam)
-{
- if(nID == DBT_DEVNODES_CHANGED)
- {
- // SetCaptureDevice();
- }
- return CDialog::OnDeviceChange(nID, lParam);
-}
-
-
-void CDXChannelDlg::OnSysCommand(UINT nID, LPARAM lParam)
-{
- if(SC_MAXIMIZE == nID)
- {}
- CDialog::OnSysCommand(nID, lParam);
-}
-
-
-static bool fullScreen = false;
-void CDXChannelDlg::OnRButtonUp( UINT nFlags, CPoint point)
-{
- CDialog::OnRButtonUp( nFlags, point);
-}
-
-// The system calls this to obtain the cursor to display while the user drags
-// the minimized window.
-HCURSOR CDXChannelDlg::OnQueryDragIcon()
-{
- return (HCURSOR) m_hIcon;
-}
-
-void CDXChannelDlg::OnCbnSelchangeCodecList()
-{
- SetSendCodec();
-}
-
-
-void CDXChannelDlg::OnCbnSelchangeSize()
-{
- SetSendCodec();
-}
-
-void CDXChannelDlg::OnCbnSelchangeDevice()
-{
-
-
- SetCaptureDevice();
-
-}
-
-
-void CDXChannelDlg::OnCbnSelchangeBitrate()
-{
-
- SetSendCodec();
-
-}
-
-void CDXChannelDlg::OnCbnSelchangeMinFrameRate()
-{
-
- SetSendCodec();
-
-}
-
-
-void CDXChannelDlg::OnBnClickedTmmbr()
-{
-
- TEST_MUSTPASS(_vieRTPRTCP->SetTMMBRStatus(_channelId,m_cbTmmbr.GetCheck()==BST_CHECKED),-5);
-
-}
-
-void CDXChannelDlg::OnCbnSelchangeRtcpmode()
-{
-
- /*
- kRtcpNone = 0,
- kRtcpCompound_RFC4585 = 1,
- kRtcpNonCompound_RFC5506 = 2 */
- ViERTCPMode mode=ViERTCPMode(m_ctrlRtcpMode.GetCurSel());
- TEST_MUSTPASS(_vieRTPRTCP->SetRTCPStatus(_channelId,mode),-5);
-
-}
-
-void CDXChannelDlg::OnBnClickedFreezelog()
-{
- _canAddLog=m_cbFreezeLog.GetCheck()!=BST_CHECKED;
-}
-
-void CDXChannelDlg::OnBnClickedProtNack()
-{
-
- TEST_MUSTPASS(_vieRTPRTCP->SetNACKStatus(_channelId,true),-5);
-
-}
-
-void CDXChannelDlg::OnBnClickedProtNone()
-{
-
- TEST_MUSTPASS(_vieRTPRTCP->SetNACKStatus(_channelId,false),-5);
- TEST_MUSTPASS(_vieRTPRTCP->SetFECStatus(_channelId,false,0,0),-5);
- TEST_MUSTPASS(_vieRTPRTCP->SetHybridNACKFECStatus(_channelId,false,0,0),-5);
-}
-
-void CDXChannelDlg::OnBnClickedProtFec()
-{
- int noCodec=_vieCodec->NumberOfCodecs();
- int redPayloadType=0;
- int fecPayloadType=0;
- for(unsigned char i=0;i<noCodec;++i)
- {
- VideoCodec codec;
- _vieCodec->GetCodec(i,codec);
- if(codec.codecType==webrtc::kVideoCodecRED)
- {
- redPayloadType=codec.plType;
- }
- if(codec.codecType==webrtc::kVideoCodecULPFEC)
- {
- fecPayloadType=codec.plType;
- }
- }
- TEST_MUSTPASS(_vieRTPRTCP->SetFECStatus(_channelId,true,redPayloadType,fecPayloadType),-5);
-}
-
-void CDXChannelDlg::OnBnClickedProtNackFec()
-{
- int noCodec=_vieCodec->NumberOfCodecs();
- int redPayloadType=0;
- int fecPayloadType=0;
- for(unsigned char i=0;i<noCodec;++i)
- {
- VideoCodec codec;
- _vieCodec->GetCodec(i,codec);
- if(codec.codecType==webrtc::kVideoCodecRED)
- {
- redPayloadType=codec.plType;
- }
- if(codec.codecType==webrtc::kVideoCodecULPFEC)
- {
- fecPayloadType=codec.plType;
- }
- }
- TEST_MUSTPASS(_vieRTPRTCP->SetHybridNACKFECStatus(_channelId,true,
- redPayloadType,
- fecPayloadType),-5);
-
-}
-
-void CDXChannelDlg::OnBnClickedVersion()
-{
- char version[1024];
- _vieBase->GetVersion(version);
- MessageBox(CharToTchar(version,-1));
- int p=strlen(version);
-#ifndef NO_VOICE_ENGINE
- _veBase->GetVersion(version);
- MessageBox(CharToTchar(version,-1));
-#endif
-}
-
-unsigned int WINAPI CDXChannelDlg::CallbackThread(LPVOID lpParameter)
-{
- static_cast<CDXChannelDlg*>(lpParameter)->CallbackThreadProcess();
- return 0;
-}
-
-void CDXChannelDlg::CallbackThreadProcess()
-{
- while(1)
- {
- if(WAIT_OBJECT_0==WaitForSingleObject(_callbackEvent,INFINITE))
- {
- char smsg[512];
- EnterCriticalSection(&_critCallback);
- strncpy(smsg,_logMsg,strlen(_logMsg)+1);
- strcpy(_logMsg,"");
-
-
- LeaveCriticalSection(&_critCallback);
- if(strstr(smsg,"Send")!=NULL)
- {
- unsigned short fractionLost=0;
- unsigned int cumulativeLost=0;
- unsigned int extendedMax=0;
- unsigned int jitter=0;
- int rttMs=0;
-
-
-
- _vieRTPRTCP->GetReceivedRTCPStatistics(_channelId,
- fractionLost,
- cumulativeLost,
- extendedMax,
- jitter,
- rttMs);
-
- //int bw=0;
- //if(_vieCodec->GetAvailableBandwidth(_channelId,bw)==0)
- //{
- // sprintf(smsg,"%s, rtt %d, loss %d,bw %d", smsg,rttMs,fractionLost,bw);
- //}
- //else
- //{
- // _vieBase->LastError(); // Reset last error.
- //}
-
-
-
- }
- if(strlen(smsg))
- {
- m_ctrlInfo.InsertString(0,(LPCTSTR) CharToTchar(smsg,-1));
- while(m_ctrlInfo.GetCount()==151)
- m_ctrlInfo.DeleteString(150);
- }
- else
- {
- break; // End the callback thread
- }
- }
- }
-
-}
-void CDXChannelDlg::AddToInfo(const char* msg)
-{
- if(!_canAddLog)
- return;
- EnterCriticalSection(&_critCallback);
-
- SYSTEMTIME systemTime;
- GetSystemTime(&systemTime);
-
- if(strlen(_logMsg)==0)
- {
- SetEvent(_callbackEvent); // Notify of new
- }
-
- sprintf (_logMsg, "(%2u:%2u:%2u:%3u) %s", systemTime.wHour,
- systemTime.wMinute,
- systemTime.wSecond,
- systemTime.wMilliseconds,
- msg
- );
-
-
-
- LeaveCriticalSection(&_critCallback);
-
-
-}
-
-void CDXChannelDlg::IncomingRate(const int videoChannel,
- unsigned int framerate,
- unsigned int bitrate)
-{
- char str[64];
- sprintf(str,"Incoming Fr:%d br %d\n", framerate, bitrate);
- AddToInfo(str);
-}
-
-void CDXChannelDlg::RequestNewKeyFrame(int channel)
-{
- assert(!"(RequestNewKeyFrame why is it called");
-}
-void CDXChannelDlg::PerformanceAlarm(unsigned int cpuLoad)
-{
- char str[64];
- sprintf(str,"Performance alarm %d",cpuLoad);
- AddToInfo(str);
-}
-void CDXChannelDlg::OutgoingRate(const int videoChannel,
- unsigned int framerate,
- unsigned int bitrate)
- {
- char str[64];
- sprintf(str,"Send Fr:%d br %d", framerate, bitrate);
- AddToInfo(str);
- }
-void CDXChannelDlg::IncomingCodecChanged(const int videoChannel,
- const VideoCodec& videoCodec)
- {
- char str[128];
- sprintf(str,"Incoming codec channel:%d pltype:%d width:%d height:%d\n", videoChannel, videoCodec.plType, videoCodec.width,videoCodec.height);
- AddToInfo(str);
- }
-void CDXChannelDlg::BrightnessAlarm(const int captureId,
- const Brightness brightness)
-{
-
- switch(brightness)
- {
- case Normal:
- AddToInfo("BrightnessAlarm - image ok.\n");
- break;
- case Bright:
- AddToInfo("BrightnessAlarm - light image.\n");
- break;
- case Dark:
- AddToInfo("BrightnessAlarm - dark image.\n");
- break;
- }
-}
-
-void CDXChannelDlg::CapturedFrameRate(const int captureId,
- const unsigned char frameRate)
-{
- char str[64];
- sprintf(str,"Local Camera Frame rate:%d \n", frameRate);
- AddToInfo(str);
-}
-
-void CDXChannelDlg::NoPictureAlarm(const int captureId,
- const CaptureAlarm alarm)
-{
- char str[64];
- sprintf(str,"No Picture alarm\n");
- AddToInfo(str);
-
-}
-
-
-void CDXChannelDlg::OnBnClickedExttransport()
-{
- if(m_cbExternalTransport.GetCheck()==BST_CHECKED)
- {
- m_localPort1.EnableWindow(FALSE);
- m_remotePort1.EnableWindow(FALSE);
- m_remoteIp1.EnableWindow(FALSE);
- m_ctrlPacketLoss.EnableWindow(TRUE);
- m_ctrlDelay.EnableWindow(TRUE);
- _externalTransport= new TbExternalTransport(*_vieNetwork);
- _vieNetwork->RegisterSendTransport(_channelId,*_externalTransport);
- }
- else
- {
- _vieNetwork->DeregisterSendTransport(_channelId);
-
- delete _externalTransport;
- _externalTransport=NULL;
- m_localPort1.EnableWindow(TRUE);
- m_remotePort1.EnableWindow(TRUE);
- m_remoteIp1.EnableWindow(TRUE);
- m_ctrlPacketLoss.EnableWindow(FALSE);
- m_ctrlDelay.EnableWindow(FALSE);
- }
-}
-
-
-void CDXChannelDlg::OnCbnSelchangePacketloss()
-{
- if(_externalTransport)
- {
- _externalTransport->SetPacketLoss(m_ctrlPacketLoss.GetCurSel()*2);
- }
-}
-
-
-void CDXChannelDlg::OnCbnSelchangeDelay()
-{
- if(_externalTransport)
- {
- _externalTransport->SetNetworkDelay(m_ctrlDelay.GetCurSel()*30);
- }
-
-}
-
-void CDXChannelDlg::OnBnClickedBtnRecordIncoming()
-{
-
- CButton *recordBtn = (CButton *) GetDlgItem(IDC_BTN_RECORD_INCOMING);
-
- CString text;
- recordBtn->GetWindowText(text);
- if(text!=_T("Stop Rec Inc")!=0)
- {
- recordBtn->SetWindowText(_T("Stop Rec Inc"));
- SYSTEMTIME time;
- GetSystemTime(&time);
- sprintf(_fileName,"IncomingChannel%d_%4d%2d%2d%2d%2d.avi",_channelId,time.wYear,time.wMonth,time.wDay,time.wHour,time.wMinute);
-
- AudioSource audioSource=PLAYOUT;
- webrtc::CodecInst audioCodec;
- strcpy(audioCodec.plname,"L16");
- audioCodec.rate = 256000;
- audioCodec.plfreq = 16000;
- audioCodec.pacsize = 160;
-
- webrtc::VideoCodec videoCodec;
- memset(&videoCodec,0,sizeof(videoCodec));
-
- strcpy(videoCodec.plName,"VP8");
- videoCodec.maxBitrate=1000;
- videoCodec.startBitrate=1000;
- videoCodec.width=352;
- videoCodec.height=288;
- videoCodec.codecType=webrtc::kVideoCodecVP8;
- videoCodec.maxFramerate=30;
- TEST_MUSTPASS(_vieFile->StartRecordIncomingVideo(_channelId,_fileName,audioSource,audioCodec, videoCodec),-5);
- }
- else
- {
- recordBtn->SetWindowText(_T("Record Incoming"));
- TEST_MUSTPASS(_vieFile->StopRecordIncomingVideo(_channelId),-5);
- CString msg;
- msg.AppendFormat(_T("Recorded file %s"),_fileName);
- MessageBox(msg);
- }
-}
-
-void CDXChannelDlg::OnBnClickedBtnRecordOutgoing()
-{
-
- CButton *recordBtn = (CButton *) GetDlgItem(IDC_BTN_RECORD_OUTGOING);
- CString text;
- recordBtn->GetWindowText(text);
- if(text!=_T("Stop Rec Out"))
- {
- recordBtn->SetWindowText(_T("Stop Rec Out"));
- SYSTEMTIME time;
- GetSystemTime(&time);
- sprintf(_fileName,"OutgoingChannel%d_%4d%2d%2d%2d%2d.avi",_channelId,time.wYear,time.wMonth,time.wDay,time.wHour,time.wMinute);
-
- AudioSource audioSource=MICROPHONE;
- webrtc::CodecInst audioCodec;
- strcpy(audioCodec.plname,"L16");
- audioCodec.rate = 256000;
- audioCodec.plfreq = 16000;
- audioCodec.pacsize = 160;
-
- webrtc::VideoCodec videoCodec;
- memset(&videoCodec,0,sizeof(videoCodec));
-
- strcpy(videoCodec.plName,"VP8");
- videoCodec.maxBitrate=1000;
- videoCodec.startBitrate=1000;
- videoCodec.width=352;
- videoCodec.height=288;
- videoCodec.codecType=webrtc::kVideoCodecVP8;
- videoCodec.maxFramerate=30;
- TEST_MUSTPASS(_vieFile->StartRecordOutgoingVideo(_channelId,_fileName,audioSource,audioCodec,videoCodec),-5);
- }
- else
- {
- recordBtn->SetWindowText(_T("Record Outgoing"));
- TEST_MUSTPASS(_vieFile->StopRecordOutgoingVideo(_channelId),-5);
- CString msg;
- msg.AppendFormat(_T("Recorded file %s"),_fileName);
- MessageBox(msg);
- }
-}
-
-void CDXChannelDlg::OnBnClickedBtnCreateSlave()
-{
- CDXChannelDlg* newSlave =new CDXChannelDlg(_videoEngine,_captureDevicePool,_channelPool,_voiceEngine,NULL,_dialogObserver,_channelId);
- newSlave->Create(CDXChannelDlg::IDD,NULL);
-}
diff --git a/trunk/src/video_engine/main/test/WindowsTest/ChannelDlg.h b/trunk/src/video_engine/main/test/WindowsTest/ChannelDlg.h
deleted file mode 100644
index 43aeb09..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/ChannelDlg.h
+++ /dev/null
@@ -1,273 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_CHANNELDLG_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_CHANNELDLG_H_
-
-#include "StdAfx.h"
-//#define NO_VOICE_ENGINE
-
-/////////////////////////////////////////////////////////////////////////////
-// CDXChannelDlg dialog
-// Include ViE headers
-
-#include "common_types.h"
-
-#include "vie_base.h"
-#include "vie_capture.h"
-#include "vie_codec.h"
-#include "vie_network.h"
-#include "vie_render.h"
-#include "vie_rtp_rtcp.h"
-#include "vie_errors.h"
-#include "vie_file.h"
-#include "tbExternalTransport.h"
-
-#include "resource.h" // main symbols
-
-
-#ifndef NO_VOICE_ENGINE
-
-#include "voe_base.h"
-#include "voe_errors.h"
-#include "voe_base.h"
-#include "voe_network.h"
-#include "voe_codec.h"
-#include "voe_rtp_rtcp.h"
-#endif
-
-using namespace webrtc;
-class CDXChannelDlg;
-class CaptureDevicePool;
-class ChannelPool;
-
-#define TEST_MUSTPASS(expr,oklasterror) \
- { \
- if ((expr)) \
- { \
- CString r_msg; \
- int r_lastError=_vieBase->LastError(); \
- CString exp; \
- exp=#expr;\
- r_msg.Format(_T("\nError at line:%i, %s \nError code: %i\n"),__LINE__, exp,r_lastError); \
- if(r_lastError!=oklasterror) \
- ::MessageBox (NULL, (LPCTSTR)r_msg, TEXT("Error Message"), MB_OK | MB_ICONINFORMATION); \
- } \
- }
-
-class CDXChannelDlgObserver
-{
-public:
- virtual void ChannelDialogEnded(CDXChannelDlg* context)=0;
-
-protected:
- virtual ~CDXChannelDlgObserver(){};
-
-};
-
-class CDXChannelDlg : public CDialog , public ViEEncoderObserver, public ViEDecoderObserver, public ViEBaseObserver, public ViECaptureObserver
-{
-// Construction
-public:
- CDXChannelDlg(VideoEngine* videoEngine,
- CaptureDevicePool& captureDevicePool,
- ChannelPool& channelPool,
- void* voiceEngine=NULL
- ,CWnd* pParent = NULL,CDXChannelDlgObserver* observer=NULL,int parentChannel=-1); // standard constructor
-
-// Dialog Data
- //{{AFX_DATA(CDXChannelDlg)
- enum { IDD = IDD_DXQUALITY_DIALOG };
- CComboBox m_ctrlDevice;
- CComboBox m_ctrlCodec;
- CComboBox m_ctrlBitrate;
- CComboBox m_ctrlCodecSize;
- CComboBox m_ctrlRtcpMode;
- CComboBox m_ctrlPacketBurst;
- CComboBox m_ctrlMinFrameRate;
-
- CListBox m_ctrlInfo;
-
- CStatic m_ctrlLiveRemoteVideo;
- CStatic m_ctrlLiveVideo;
- CEdit m_localPort1;
- CEdit m_remotePort1;
- CIPAddressCtrl m_remoteIp1;
- CButton m_cbTmmbr;
- CButton m_cbExternalTransport;
- CButton m_cbFreezeLog;
- CButton m_cbDefaultSendChannel;
- CComboBox m_ctrlPacketLoss;
- CComboBox m_ctrlDelay;
-
-
- //}}AFX_DATA
-
- // ClassWizard generated virtual function overrides
- //{{AFX_VIRTUAL(CDXChannelDlg)
- protected:
- virtual void DoDataExchange(CDataExchange* pDX); // DDX/DDV support
- //}}AFX_VIRTUAL
-
-
-
-public :
- // Callback
-
- //Capture observer
- virtual void BrightnessAlarm(const int captureId,
- const Brightness brightness);
-
- virtual void CapturedFrameRate(const int captureId,
- const unsigned char frameRate);
-
- virtual void NoPictureAlarm(const int captureId,
- const CaptureAlarm alarm);
-
-
- // same callback method is being used to raise also to clear.
- // true - raise, false - clear
- // virtual void NoPictureAlarm(bool active = true);
-
- // Encoder observer
- virtual void OutgoingRate(const int videoChannel,
- const unsigned int framerate,
- const unsigned int bitrate) ;
-
- //Decoder observer
- virtual void IncomingCodecChanged(const int videoChannel,
- const VideoCodec& videoCodec);
-
- virtual void IncomingRate(const int videoChannel,
- const unsigned int framerate,
- const unsigned int bitrate);
-
- virtual void RequestNewKeyFrame(const int videoChannel);
-
- // Base observer
- virtual void PerformanceAlarm(const unsigned int cpuLoad);
-
-
- //virtual void IncomingCSRCChanged(int channel, unsigned int csrc, bool added);
-
-
-
-// Implementation
-protected:
- HICON m_hIcon;
- int _channelId;
- int _parentChannel;
- int _audioChannel;
- bool _canAddLog;
-
- // Thread and function for callbacks
- CRITICAL_SECTION _critCallback;
- HANDLE _callbackThread;
- HANDLE _callbackEvent;
- char _logMsg[512];
- static unsigned int WINAPI CallbackThread(LPVOID lpParameter);
- void CallbackThreadProcess();
-
-
-
- //void GetSize(int sizeSel, int &width, int &height);
- virtual void ConfigureRender();
-
- virtual void SetCaptureDevice();
- virtual void SetLocalReceiver();
- virtual void SetSendDestination();
- virtual void SetSendCodec();
-
-
- void AddToInfo(const char* msg);
-
- // afx_msg void Command(UINT nID, LPARAM lParam);
-
- // Generated message map functions
- //{{AFX_MSG(CDXChannelDlg)
- virtual BOOL OnInitDialog();
- afx_msg void OnSysCommand(UINT nID, LPARAM lParam);
- afx_msg void OnRButtonUp( UINT nFlags, CPoint point);
- afx_msg BOOL OnDeviceChange( UINT, DWORD );
- afx_msg void OnPaint();
- //afx_msg LRESULT OnDisplayChange(WPARAM, LPARAM);
- afx_msg HCURSOR OnQueryDragIcon();
- virtual afx_msg void OnStartSend();
- virtual afx_msg void OnDestroy();
- virtual afx_msg void OnStopSend();
- virtual afx_msg void OnCancel();
- afx_msg void OnTimer(UINT nIDEvent);
-
- //}}AFX_MSG
- DECLARE_MESSAGE_MAP()
-
-private:
- CDXChannelDlgObserver* _dialogObserver;
-
- VideoEngine* _videoEngine;
- ViEBase* _vieBase;
- ViECapture* _vieCapture;
- ViERTP_RTCP* _vieRTPRTCP;
- ViERender* _vieRender;
- ViECodec* _vieCodec;
- ViENetwork* _vieNetwork;
- ViEFile* _vieFile;
- TbExternalTransport* _externalTransport;
- char _fileName[256];
-
-
-#ifndef NO_VOICE_ENGINE
- VoiceEngine* _voiceEngine;
- VoEBase* _veBase;
- VoENetwork* _veNetwork;
- VoECodec* _veCodec;
- VoERTP_RTCP* _veRTCP;
-#else
- void* _voiceEngine;
-
-#endif
-
- VideoCodec _sendCodec;
- int _captureId;
- CaptureDevicePool& _captureDevicePool;
- ChannelPool& _channelPool;
-
-
- afx_msg void OnCbnSelchangeCodecList();
- afx_msg void OnCbnSelchangeDevice();
- afx_msg void OnCbnSelchangeSize();
- afx_msg void OnCbnSelchangeBitrate();
- afx_msg void OnCbnSelchangeWindowSize();
- afx_msg void OnBnClickedversion();
- afx_msg void OnCbnSelchangeMinFrameRate();
- afx_msg void OnBnClickedStartlisten();
- afx_msg void OnBnClickedStoplisten();
- afx_msg void OnBnClickedStopsend();
- afx_msg void OnBnClickedTmmbr();
- afx_msg void OnCbnSelchangeRtcpmode();
- afx_msg void OnBnClickedProtNack();
- afx_msg void OnBnClickedProtNone();
- afx_msg void OnBnClickedProtFec();
- afx_msg void OnBnClickedProtNackFec();
- afx_msg void OnBnClickedFreezelog();
-public:
- afx_msg void OnBnClickedExttransport();
- afx_msg void OnCbnSelchangePacketloss();
- afx_msg void OnCbnSelchangeDelay();
- afx_msg void OnBnClickedBtnRecordIncoming();
- afx_msg void OnBnClickedBtnRecordOutgoing();
- afx_msg void OnBnClickedBtnCreateSlave();
- afx_msg void OnBnClickedVersion();
-};
-
-//{{AFX_INSERT_LOCATION}}
-// Microsoft Visual C++ will insert additional declarations immediately before the previous line.
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_CHANNELDLG_H_
diff --git a/trunk/src/video_engine/main/test/WindowsTest/ChannelPool.cc b/trunk/src/video_engine/main/test/WindowsTest/ChannelPool.cc
deleted file mode 100644
index dbd1644..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/ChannelPool.cc
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "ChannelPool.h"
-#include "map_wrapper.h"
-#include <string.h>
-#include <assert.h>
-#include "critical_section_wrapper.h"
-
-ChannelPool::ChannelPool():
-_critSect(*webrtc::CriticalSectionWrapper::CreateCriticalSection())
-{
-}
-
-ChannelPool::~ChannelPool(void)
-{
- assert(_channelMap.Size()==0);
- delete &_critSect;
-}
-
-WebRtc_Word32 ChannelPool::AddChannel(int channel)
-{
- return _channelMap.Insert(channel,(void*) channel);
-}
-WebRtc_Word32 ChannelPool::RemoveChannel(int channel)
-{
- return _channelMap.Erase(channel);
-}
-
-webrtc::MapWrapper& ChannelPool::ChannelMap()
-{
- return _channelMap;
-}
diff --git a/trunk/src/video_engine/main/test/WindowsTest/ChannelPool.h b/trunk/src/video_engine/main/test/WindowsTest/ChannelPool.h
deleted file mode 100644
index 374c676..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/ChannelPool.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#pragma once
-#include "StdAfx.h"
-#include "common_types.h"
-
-#include "vie_base.h"
-#include "map_wrapper.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-}
-
-class ChannelPool
-{
-public:
- ChannelPool();
- ~ChannelPool(void);
- WebRtc_Word32 AddChannel(int channel);
- WebRtc_Word32 RemoveChannel(int channel);
-
- webrtc::MapWrapper& ChannelMap();
-
- private:
- webrtc::CriticalSectionWrapper& _critSect;
- webrtc::MapWrapper _channelMap;
-
-};
diff --git a/trunk/src/video_engine/main/test/WindowsTest/StdAfx.h b/trunk/src/video_engine/main/test/WindowsTest/StdAfx.h
deleted file mode 100644
index 78b1fbd..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/StdAfx.h
+++ /dev/null
@@ -1,37 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// stdafx.h : include file for standard system include files,
-// or project specific include files that are used frequently, but
-// are changed infrequently
-//
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_STDAFX_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_STDAFX_H_
-
-#if _MSC_VER > 1000
-#pragma once
-#endif // _MSC_VER > 1000
-
-#define VC_EXTRALEAN // Exclude rarely-used stuff from Windows headers
-
-#include <afxwin.h> // MFC core and standard components
-#include <afxext.h> // MFC extensions
-#include <afxdisp.h> // MFC Automation classes
-#include <afxdtctl.h> // MFC support for Internet Explorer 4 Common Controls
-#ifndef _AFX_NO_AFXCMN_SUPPORT
-#include <afxcmn.h> // MFC support for Windows Common Controls
-#endif // _AFX_NO_AFXCMN_SUPPORT
-
-
-//{{AFX_INSERT_LOCATION}}
-// Microsoft Visual C++ will insert additional declarations immediately before the previous line.
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_STDAFX_H_
diff --git a/trunk/src/video_engine/main/test/WindowsTest/VideoSize.h b/trunk/src/video_engine/main/test/WindowsTest/VideoSize.h
deleted file mode 100644
index 60e2bdd..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/VideoSize.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_VIDEOSIZE_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_VIDEOSIZE_H_
-#include "StdAfx.h"
-enum VideoSize
- {
- UNDEFINED,
- SQCIF, // 128*96 = 12 288
- QQVGA, // 160*120 = 19 200
- QCIF, // 176*144 = 25 344
- CGA, // 320*200 = 64 000
- QVGA, // 320*240 = 76 800
- SIF, // 352*240 = 84 480
- WQVGA, // 400*240 = 96 000
- CIF, // 352*288 = 101 376
- W288P, // 512*288 = 147 456 (WCIF)
- W368P, // 640*368 = 235 520
- S_448P, // 576*448 = 281 088
- VGA, // 640*480 = 307 200
- S_432P, // 720*432 = 311 040
- W432P, // 768*432 = 331 776 (a.k.a WVGA 16:9)
- S_4SIF, // 704*480 = 337 920
- W448P, // 768*448 = 344 064
- NTSC, // 720*480 = 345 600
- FW448P, // 800*448 = 358 400
- S_768x480P, // 768*480 = 368 640 (a.k.a WVGA 16:10)
- WVGA, // 800*480 = 384 000
- S_4CIF, // 704576 = 405 504
- SVGA, // 800*600 = 480 000
- W544P, // 960*544 = 522 240
- W576P, // 1024*576 = 589 824 (W4CIF)
- HD, // 960*720 = 691 200
- XGA, // 1024*768 = 786 432
- WHD, // 1280*720 = 921 600
- FULL_HD, // 1440*1080 = 1 555 200
- UXGA, // 1600*1200 = 1 920 000
- WFULL_HD, // 1920*1080 = 2 073 600
- NUMBER_OF_VIDEO_SIZE
- };
-
-int GetWidthHeight(VideoSize size, int& width, int& height);
-
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_VIDEOSIZE_H_
diff --git a/trunk/src/video_engine/main/test/WindowsTest/WindowsTest.cc b/trunk/src/video_engine/main/test/WindowsTest/WindowsTest.cc
deleted file mode 100644
index ceda8a9..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/WindowsTest.cc
+++ /dev/null
@@ -1,114 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "WindowsTest.h"
-#include "ChannelDlg.h"
-#include "WindowsTestMainDlg.h"
-#include "engine_configurations.h"
-
-#ifdef _DEBUG
-#define new DEBUG_NEW
-#undef THIS_FILE
-static char THIS_FILE[] = __FILE__;
-#endif
-
-// Check memory leaks id running debug
-#if (defined(_DEBUG) && defined(_WIN32))
-// #include "vld.h"
-#endif
-/////////////////////////////////////////////////////////////////////////////
-// CDXWindowsTestApp
-
-BEGIN_MESSAGE_MAP(CDXWindowsTestApp, CWinApp)
- //{{AFX_MSG_MAP(CDXWindowsTestApp)
- // NOTE - the ClassWizard will add and remove mapping macros here.
- // DO NOT EDIT what you see in these blocks of generated code!
- //}}AFX_MSG
- ON_COMMAND(ID_HELP, CWinApp::OnHelp)
-END_MESSAGE_MAP()
-
-/////////////////////////////////////////////////////////////////////////////
-// CDXWindowsTestApp construction
-
-CDXWindowsTestApp::CDXWindowsTestApp()
-{
-
-}
-
-/////////////////////////////////////////////////////////////////////////////
-// The one and only object
-
-CDXWindowsTestApp theApp;
-
-/////////////////////////////////////////////////////////////////////////////
-// CDXWindowsTestApp initialization
-
-BOOL CDXWindowsTestApp::InitInstance()
-{
- int result=0;
- #ifndef NO_VOICE_ENGINE
- _voiceEngine = VoiceEngine::Create();
- _veBase = VoEBase::GetInterface(_voiceEngine);
- result+=_veBase->Init();
- #else
- _voiceEngine=NULL;
- #endif
-
- _videoEngine = VideoEngine::Create();
-
- _videoEngine->SetTraceFilter(webrtc::kTraceDefault);//webrtc::kTraceDebug | webrtc::kTraceError | webrtc::kTraceApiCall | webrtc::kTraceWarning | webrtc::kTraceCritical | webrtc::kTraceStateInfo | webrtc::kTraceInfo | webrtc::kTraceStream);
- _videoEngine->SetTraceFile("trace.txt");
-
- ViEBase* vieBase=ViEBase::GetInterface(_videoEngine);
- result+=vieBase->Init();
- if(result!=0)
- {
- ::MessageBox (NULL, (LPCTSTR)("failed to init VideoEngine"), TEXT("Error Message"), MB_OK | MB_ICONINFORMATION);
- }
-
- {
- WindowsTestMainDlg dlg(_videoEngine,_voiceEngine);
-
- m_pMainWnd = &dlg;
- int nResponse = dlg.DoModal();
- }
-
- vieBase->Release();
-
- if(!VideoEngine::Delete(_videoEngine))
- {
- char errorMsg[255];
- sprintf(errorMsg,"All VideoEngine interfaces are not released properly!");
- ::MessageBox (NULL, (LPCTSTR)errorMsg, TEXT("Error Message"), MB_OK | MB_ICONINFORMATION);
- }
-
- #ifndef NO_VOICE_ENGINE
-
- _veBase->Terminate();
- if(_veBase->Release()!=0)
- {
- // ensure that no interface is still referenced
- char errorMsg[256];
- sprintf(errorMsg,"All VoiceEngine interfaces are not released properly!");
- ::MessageBox (NULL, (LPCTSTR)errorMsg, TEXT("Error Message"), MB_OK | MB_ICONINFORMATION);
- }
-
- if (false == VoiceEngine::Delete(_voiceEngine))
- {
- char errorMsg[256];
- sprintf(errorMsg,"VoiceEngine::Delete() failed!");
- ::MessageBox (NULL, (LPCTSTR)errorMsg, TEXT("Error Message"), MB_OK | MB_ICONINFORMATION);
- }
- #endif
-
- // Since the dialog has been closed, return FALSE so that we exit the
- // application, rather than start the application's message pump.
- return FALSE;
-}
diff --git a/trunk/src/video_engine/main/test/WindowsTest/WindowsTest.h b/trunk/src/video_engine/main/test/WindowsTest/WindowsTest.h
deleted file mode 100644
index dc3ee9d..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/WindowsTest.h
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_WINDOWSTEST_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_WINDOWSTEST_H_
-
-
-#include "StdAfx.h"
-#include "resource.h" // main symbols
-
-
-
-/////////////////////////////////////////////////////////////////////////////
-
-//Forward declarations
-namespace webrtc {
- class VoiceEngine;
- class VoEBase;
- class VideoEngine;
-}
-using namespace webrtc;
-
-class CDXWindowsTestApp : public CWinApp
-{
-public:
- CDXWindowsTestApp();
-
-// Overrides
- // ClassWizard generated virtual function overrides
- //{{AFX_VIRTUAL(CDXWindowsTestApp)
- public:
- virtual BOOL InitInstance();
- //}}AFX_VIRTUAL
-
-// Implementation
-
- //{{AFX_MSG(CDXWindowsTestApp)
- // NOTE - the ClassWizard will add and remove member functions here.
- // DO NOT EDIT what you see in these blocks of generated code !
- //}}AFX_MSG
- DECLARE_MESSAGE_MAP()
-
- VideoEngine* _videoEngine;
- VoiceEngine* _voiceEngine;
- VoEBase* _veBase;
-};
-
-
-/////////////////////////////////////////////////////////////////////////////
-
-//{{AFX_INSERT_LOCATION}}
-// Microsoft Visual C++ will insert additional declarations immediately before the previous line.
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_WINDOWSTEST_H_
diff --git a/trunk/src/video_engine/main/test/WindowsTest/WindowsTestMainDlg.cc b/trunk/src/video_engine/main/test/WindowsTest/WindowsTestMainDlg.cc
deleted file mode 100644
index fcc490d..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/WindowsTestMainDlg.cc
+++ /dev/null
@@ -1,142 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// WindowsTestMainDlg.cpp : implementation file
-//
-#include "WindowsTestMainDlg.h"
-#include "WindowsTest.h"
-#include "ChannelDlg.h"
-
-#include "voe_base.h"
-
-// WindowsTestMainDlg dialog
-
-IMPLEMENT_DYNAMIC(WindowsTestMainDlg, CDialog)
-
-WindowsTestMainDlg::WindowsTestMainDlg(VideoEngine* videoEngine,void* voiceEngine,CWnd* pParent /*=NULL*/)
- : CDialog(WindowsTestMainDlg::IDD, pParent),
- _videoEngine(videoEngine),
- _voiceEngine((VoiceEngine*) voiceEngine),
- _testDlg1(NULL),
- _testDlg2(NULL),
- _testDlg3(NULL),
- _testDlg4(NULL),
- _externalInWidth(0),
- _externalInHeight(0),
- _externalInVideoType(0),
- _captureDevicePool(videoEngine)
-{
-
-}
-
-WindowsTestMainDlg::~WindowsTestMainDlg()
-{
-}
-
-void WindowsTestMainDlg::DoDataExchange(CDataExchange* pDX)
-{
- CDialog::DoDataExchange(pDX);
-}
-
-
-BEGIN_MESSAGE_MAP(WindowsTestMainDlg, CDialog)
- ON_BN_CLICKED(IDC_CHANNEL1, &WindowsTestMainDlg::OnBnClickedChannel1)
- ON_BN_CLICKED(IDC_CHANNEL2, &WindowsTestMainDlg::OnBnClickedChannel2)
- ON_BN_CLICKED(IDC_CHANNEL3, &WindowsTestMainDlg::OnBnClickedChannel3)
- ON_BN_CLICKED(IDC_CHANNEL4, &WindowsTestMainDlg::OnBnClickedChannel4)
-END_MESSAGE_MAP()
-
-
-
-void WindowsTestMainDlg::OnBnClickedChannel1()
-{
- if(!_testDlg1)
- {
- _testDlg1=new CDXChannelDlg(_videoEngine,_captureDevicePool,_channelPool,_voiceEngine,NULL,this);
- _testDlg1->Create(CDXChannelDlg::IDD,this);
- }
- else
- {
- _testDlg1->SetActiveWindow();
- }
-}
-
-void WindowsTestMainDlg::OnBnClickedChannel2()
-{
- if(!_testDlg2)
- {
- _testDlg2=new CDXChannelDlg(_videoEngine,_captureDevicePool,_channelPool,_voiceEngine,NULL,this);
- _testDlg2->Create(CDXChannelDlg::IDD,this);
-
- }
- else
- {
- _testDlg2->SetActiveWindow();
- }
-}
-
-void WindowsTestMainDlg::ChannelDialogEnded(CDXChannelDlg* context)
-{
- if(context==_testDlg4)
- {
- delete _testDlg4;
- _testDlg4=NULL;
- }
- else if(context==_testDlg3)
- {
- delete _testDlg3;
- _testDlg3=NULL;
- }
- else if(context==_testDlg2)
- {
- delete _testDlg2;
- _testDlg2=NULL;
- }
- else if(context==_testDlg1)
- {
- delete _testDlg1;
- _testDlg1=NULL;
- }
- else // Slave channel
- {
- delete context;
- }
-
-}
-
-
-
-void WindowsTestMainDlg::OnBnClickedChannel3()
-{
- if(!_testDlg3)
- {
- _testDlg3=new CDXChannelDlg(_videoEngine,_captureDevicePool,_channelPool,_voiceEngine,NULL,this);
- _testDlg3->Create(CDXChannelDlg::IDD,this);
-
- }
- else
- {
- _testDlg3->SetActiveWindow();
- }
-}
-
-void WindowsTestMainDlg::OnBnClickedChannel4()
-{
- if(!_testDlg4)
- {
- _testDlg4=new CDXChannelDlg(_videoEngine,_captureDevicePool,_channelPool,_voiceEngine,NULL,this);
- _testDlg4->Create(CDXChannelDlg::IDD,this);
-
- }
- else
- {
- _testDlg4->SetActiveWindow();
- }
-}
diff --git a/trunk/src/video_engine/main/test/WindowsTest/WindowsTestMainDlg.h b/trunk/src/video_engine/main/test/WindowsTest/WindowsTestMainDlg.h
deleted file mode 100644
index 8aae99a..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/WindowsTestMainDlg.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#pragma once
-#include "StdAfx.h"
-#include "WindowsTestResource.h"
-
-#include "ChannelDlg.h"
-#include "CaptureDevicePool.h"
-#include "ChannelPool.h"
-
-//Forward declarations
-namespace webrtc {
- class VideoEngine;
- class VoiceEngine;
-}
-using namespace webrtc;
-class CDXCaptureDlg;
-
-
-class WindowsTestMainDlg : public CDialog, private CDXChannelDlgObserver
-{
- DECLARE_DYNAMIC(WindowsTestMainDlg)
-
-public:
- WindowsTestMainDlg(VideoEngine* videoEngine,void* voiceEngine=NULL,CWnd* pParent = NULL); // standard constructor
- virtual ~WindowsTestMainDlg();
-
-// Dialog Data
- enum { IDD = IDD_WINDOWSTEST_MAIN };
-
-protected:
- virtual void DoDataExchange(CDataExchange* pDX); // DDX/DDV support
-
- DECLARE_MESSAGE_MAP()
-public:
- afx_msg void OnBnClickedChannel1();
- afx_msg void OnBnClickedChannel2();
- afx_msg void OnBnClickedChannel3();
- afx_msg void OnBnClickedChannel4();
-
-
- VideoEngine* _videoEngine;
- VoiceEngine* _voiceEngine;
- VoEBase* _veBase;
-
- CDXChannelDlg* _testDlg1;
- CDXChannelDlg* _testDlg2;
- CDXChannelDlg* _testDlg3;
- CDXChannelDlg* _testDlg4;
-
- int _externalInWidth;
- int _externalInHeight;
- int _externalInVideoType;
-
- CaptureDevicePool _captureDevicePool;
- ChannelPool _channelPool;
-
-
-private:
- virtual void ChannelDialogEnded(CDXChannelDlg* context);
-
-public:
-
-};
diff --git a/trunk/src/video_engine/main/test/WindowsTest/WindowsTestResouce.rc b/trunk/src/video_engine/main/test/WindowsTest/WindowsTestResouce.rc
deleted file mode 100644
index 5e866ad..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/WindowsTestResouce.rc
+++ /dev/null
@@ -1,101 +0,0 @@
-// Microsoft Visual C++ generated resource script.
-//
-#include "WindowsTestResource.h"
-
-#define APSTUDIO_READONLY_SYMBOLS
-/////////////////////////////////////////////////////////////////////////////
-//
-// Generated from the TEXTINCLUDE 2 resource.
-//
-#include "afxres.h"
-
-/////////////////////////////////////////////////////////////////////////////
-#undef APSTUDIO_READONLY_SYMBOLS
-
-/////////////////////////////////////////////////////////////////////////////
-// Swedish resources
-
-#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_SVE)
-#ifdef _WIN32
-LANGUAGE LANG_SWEDISH, SUBLANG_DEFAULT
-#pragma code_page(1252)
-#endif //_WIN32
-
-#ifdef APSTUDIO_INVOKED
-/////////////////////////////////////////////////////////////////////////////
-//
-// TEXTINCLUDE
-//
-
-1 TEXTINCLUDE
-BEGIN
- "WindowsTestResource.h\0"
-END
-
-2 TEXTINCLUDE
-BEGIN
- "#include ""afxres.h""\r\n"
- "\0"
-END
-
-3 TEXTINCLUDE
-BEGIN
- "\r\n"
- "\0"
-END
-
-#endif // APSTUDIO_INVOKED
-
-
-/////////////////////////////////////////////////////////////////////////////
-//
-// Dialog
-//
-
-IDD_WINDOWSTEST_MAIN DIALOGEX 0, 0, 186, 156
-STYLE DS_SETFONT | DS_MODALFRAME | DS_FIXEDSYS | WS_POPUP | WS_CAPTION | WS_SYSMENU
-CAPTION "Windows ViE Test"
-FONT 8, "MS Shell Dlg", 400, 0, 0x1
-BEGIN
- DEFPUSHBUTTON "OK",IDOK,129,7,50,14
- PUSHBUTTON "Cancel",IDCANCEL,129,24,50,14
- PUSHBUTTON "Channel 1",IDC_CHANNEL1,129,45,50,14
- PUSHBUTTON "Channel 2",IDC_CHANNEL2,129,62,50,14
- PUSHBUTTON "Channel 3",IDC_CHANNEL3,129,79,50,14
- PUSHBUTTON "Channel 4",IDC_CHANNEL4,129,96,50,14
-END
-
-
-/////////////////////////////////////////////////////////////////////////////
-//
-// DESIGNINFO
-//
-
-#ifdef APSTUDIO_INVOKED
-GUIDELINES DESIGNINFO
-BEGIN
- IDD_WINDOWSTEST_MAIN, DIALOG
- BEGIN
- LEFTMARGIN, 7
- RIGHTMARGIN, 179
- TOPMARGIN, 7
- BOTTOMMARGIN, 149
- END
-END
-#endif // APSTUDIO_INVOKED
-
-#endif // Swedish resources
-/////////////////////////////////////////////////////////////////////////////
-
-
-
-#ifndef APSTUDIO_INVOKED
-/////////////////////////////////////////////////////////////////////////////
-//
-// Generated from the TEXTINCLUDE 3 resource.
-//
-
-
-/////////////////////////////////////////////////////////////////////////////
-#endif // not APSTUDIO_INVOKED
-
diff --git a/trunk/src/video_engine/main/test/WindowsTest/WindowsTestResource.h b/trunk/src/video_engine/main/test/WindowsTest/WindowsTestResource.h
deleted file mode 100644
index 2d49c28..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/WindowsTestResource.h
+++ /dev/null
@@ -1,28 +0,0 @@
-//{{NO_DEPENDENCIES}}
-// Microsoft Visual C++ generated include file.
-// Used by WindowsTestResouce.rc
-//
-#define IDD_WINDOWSTEST_MAIN 101
-#define IDC_CHANNEL1 1001
-#define IDC_CHANNEL2 1002
-#define IDC_CHANNEL3 1004
-#define IDC_CHANNEL4 1005
-#define IDC_POSITION 1009
-#define IDC_INFORMATION 1050
-#define IDC_CHECK_CHANNEL1 1070
-#define IDC_CHECK_CHANNEL2 1071
-#define IDC_CHECK_CHANNEL3 1072
-#define IDC_CHECK_CHANNEL4 1073
-#define IDC_COMBO1 1074
-#define IDC_BTN_CREATE2 1076
-
-// Next default values for new objects
-//
-#ifdef APSTUDIO_INVOKED
-#ifndef APSTUDIO_READONLY_SYMBOLS
-#define _APS_NEXT_RESOURCE_VALUE 106
-#define _APS_NEXT_COMMAND_VALUE 40001
-#define _APS_NEXT_CONTROL_VALUE 1076
-#define _APS_NEXT_SYMED_VALUE 107
-#endif
-#endif
diff --git a/trunk/src/video_engine/main/test/WindowsTest/captureDeviceImage.jpg b/trunk/src/video_engine/main/test/WindowsTest/captureDeviceImage.jpg
deleted file mode 100644
index 3bb3ba4..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/captureDeviceImage.jpg
+++ /dev/null
Binary files differ
diff --git a/trunk/src/video_engine/main/test/WindowsTest/renderStartImage.jpg b/trunk/src/video_engine/main/test/WindowsTest/renderStartImage.jpg
deleted file mode 100644
index b10a842..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/renderStartImage.jpg
+++ /dev/null
Binary files differ
diff --git a/trunk/src/video_engine/main/test/WindowsTest/renderTimeoutImage.jpg b/trunk/src/video_engine/main/test/WindowsTest/renderTimeoutImage.jpg
deleted file mode 100644
index cb34d67..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/renderTimeoutImage.jpg
+++ /dev/null
Binary files differ
diff --git a/trunk/src/video_engine/main/test/WindowsTest/res/Capture.rc2 b/trunk/src/video_engine/main/test/WindowsTest/res/Capture.rc2
deleted file mode 100644
index d9acfd2..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/res/Capture.rc2
+++ /dev/null
@@ -1,13 +0,0 @@
-//
-// DXCAPTURE.RC2 - resources Microsoft Visual C++ does not edit directly
-//
-
-#ifdef APSTUDIO_INVOKED
- #error this file is not editable by Microsoft Visual C++
-#endif //APSTUDIO_INVOKED
-
-
-/////////////////////////////////////////////////////////////////////////////
-// Add manually edited resources here...
-
-/////////////////////////////////////////////////////////////////////////////
diff --git a/trunk/src/video_engine/main/test/WindowsTest/resource.h b/trunk/src/video_engine/main/test/WindowsTest/resource.h
deleted file mode 100644
index 34e90eb..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/resource.h
+++ /dev/null
@@ -1,58 +0,0 @@
-//{{NO_DEPENDENCIES}}
-// Microsoft Visual C++ generated include file.
-// Used by Capture.rc
-//
-#define IDM_ABOUTBOX 0x0010
-#define IDD_DXQUALITY_DIALOG 102
-#define IDI_ICON1 130
-#define IDD_SLAVE_CHANNEL 132
-#define IDC_LIVEVIDEO 1000
-#define IDC_CAPTURE 1001
-#define IDC_DEVICE 1003
-#define IDC_STARTSEND 1004
-#define IDC_STARTLISTEN 1006
-#define IDC_STOPLISTEN 1007
-#define IDC_STOPSend 1008
-#define IDC_CODEC_LIST 1010
-#define IDC_CODEC_SIZE 1011
-#define IDC_IPADDRESS1 1012
-#define IDC_LOCAL_PORT1 1013
-#define IDC_CHANGE_SIZE 1017
-#define IDC_QUALITY 1018
-#define IDC_BITRATE 1019
-#define IDC_WINDOW_SIZE 1022
-#define IDC_REMOTE_PORT1 1025
-#define IDC_START_REC 1030
-#define IDC_STOP_REC 1031
-#define IDC_CAM_PROPERTY 1033
-#define IDC_ONMODE 1034
-#define IDC_CAPTURECAP 1038
-#define IDC_RADIO1 1039
-#define IDC_MIN_FRAME_RATE 1040
-#define IDC_RTCPMODE 1042
-#define IDC_TMMBR 1043
-#define IDC_PACKETBURST 1044
-#define IDC_PROT_NONE 1045
-#define IDC_PROT_NACK 1046
-#define IDC_PROT_FEC 1047
-#define IDC_PROT_NACKFEC 1048
-#define IDC_INFORMATION 1050
-#define IDC_PACKETLOSS 1051
-#define IDC_FREEZELOG 1052
-#define IDC_VERSION 1053
-#define IDC_EXTTRANSPORT 1054
-#define IDC_DELAY 1055
-#define IDC_BTN_RECORD_INCOMING 1056
-#define IDC_BTN_RECORD_OUTGOING 1057
-#define IDC_BTN_CREATE_SLAVE 1058
-
-// Next default values for new objects
-//
-#ifdef APSTUDIO_INVOKED
-#ifndef APSTUDIO_READONLY_SYMBOLS
-#define _APS_NEXT_RESOURCE_VALUE 133
-#define _APS_NEXT_COMMAND_VALUE 32771
-#define _APS_NEXT_CONTROL_VALUE 1059
-#define _APS_NEXT_SYMED_VALUE 101
-#endif
-#endif
diff --git a/trunk/src/video_engine/main/test/WindowsTest/tbExternalTransport.cc b/trunk/src/video_engine/main/test/WindowsTest/tbExternalTransport.cc
deleted file mode 100644
index f0910ee..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/tbExternalTransport.cc
+++ /dev/null
@@ -1,317 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// tbExternalTransport.cpp
-//
-
-#include "tbExternalTransport.h"
-
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "tick_util.h"
-#include "vie_network.h"
-#include "tick_util.h"
-
-using namespace webrtc;
-
-TbExternalTransport::TbExternalTransport(ViENetwork& vieNetwork)
- :
- _vieNetwork(vieNetwork),
- _thread(*ThreadWrapper::CreateThread(ViEExternalTransportRun, this, kHighPriority, "AutotestTransport")),
- _event(*EventWrapper::Create()),
- _crit(*CriticalSectionWrapper::CreateCriticalSection()),
- _statCrit(*CriticalSectionWrapper::CreateCriticalSection()),
- _lossRate(0),
- _networkDelayMs(0),
- _rtpCount(0),
- _dropCount(0),
- _rtcpCount(0),
- _rtpPackets(),
- _rtcpPackets(),
- _checkSSRC(false),
- _lastSSRC(0),
- _checkSequenceNumber(0),
- _firstSequenceNumber(0),
- _lastSeq(0)
-{
- srand((int)TickTime::MicrosecondTimestamp());
- unsigned int tId = 0;
- _thread.Start(tId);
-}
-
-
-TbExternalTransport::~TbExternalTransport()
-{
- // TODO: stop thread
- _thread.SetNotAlive();
- _event.Set();
- if (_thread.Stop())
- {
- delete &_thread;
- delete &_event;
- }
- delete &_crit;
- delete &_statCrit;
-}
-
-
-
-
-
-int TbExternalTransport::SendPacket(int channel, const void *data, int len)
-{
- _statCrit.Enter();
- _rtpCount++;
- _statCrit.Leave();
-
-
- unsigned short sequenceNumber = (((unsigned char*) data)[2]) << 8;
- sequenceNumber += (((unsigned char*) data)[3]);
-
-
- int marker=((unsigned char*)data)[1] & 0x80;
- unsigned int timestamp=((((unsigned char*)data)[4]) << 24) + ((((unsigned char*)data)[5])<<16) +((((unsigned char*)data)[6])<<8)+(((unsigned char*)data)[7]);
-
-
- // Packet loss
- int dropThis = rand() % 100;
- bool nacked=false;
- if(sequenceNumber<_lastSeq)
- {
- nacked=true;
- }
- else
- {
- _lastSeq=sequenceNumber;
- }
-
- if (dropThis < _lossRate)
- {
- _statCrit.Enter();
- _dropCount++;
- _statCrit.Leave();
-
-
- /* char str[256];
- sprintf(str,"Dropping seq %d length %d m %d, ts %u\n", sequenceNumber,len,marker,timestamp) ;
- OutputDebugString(str);*/
-
- return len;
- }
- else
- {
- if(nacked)
- {
- /*char str[256];
- sprintf(str,"Resending seq %d length %d m %d, ts %u\n", sequenceNumber,len,marker,timestamp) ;
- OutputDebugString(str);*/
- }
- else
- {
- /*char str[256];
- sprintf(str,"Sending seq %d length %d m %d, ts %u\n", sequenceNumber,len,marker,timestamp) ;
- OutputDebugString(str);*/
-
- }
- }
-
-
- VideoPacket* newPacket = new VideoPacket();
- memcpy(newPacket->packetBuffer, data, len);
- newPacket->length = len;
- newPacket->channel = channel;
-
- _crit.Enter();
- newPacket->receiveTime = NowMs() + _networkDelayMs;
- _rtpPackets.push(newPacket);
- _event.Set();
- _crit.Leave();
- return len;
-}
-
-int TbExternalTransport::SendRTCPPacket(int channel, const void *data, int len)
-{
- _statCrit.Enter();
- _rtcpCount++;
- _statCrit.Leave();
-
- VideoPacket* newPacket = new VideoPacket();
- memcpy(newPacket->packetBuffer, data, len);
- newPacket->length = len;
- newPacket->channel = channel;
-
- _crit.Enter();
- newPacket->receiveTime = NowMs() + _networkDelayMs;
- _rtcpPackets.push(newPacket);
- _event.Set();
- _crit.Leave();
- return len;
-}
-
-WebRtc_Word32 TbExternalTransport::SetPacketLoss(WebRtc_Word32 lossRate)
-{
- CriticalSectionScoped cs(_statCrit);
- _lossRate = lossRate;
- return 0;
-}
-
-void TbExternalTransport::SetNetworkDelay(WebRtc_Word64 delayMs)
-{
- CriticalSectionScoped cs(_crit);
- _networkDelayMs = delayMs;
- return;
-}
-
-void TbExternalTransport::ClearStats()
-{
- CriticalSectionScoped cs(_statCrit);
- _rtpCount = 0;
- _dropCount = 0;
- _rtcpCount = 0;
- return;
-}
-
-void TbExternalTransport::GetStats(WebRtc_Word32& numRtpPackets, WebRtc_Word32& numDroppedPackets, WebRtc_Word32& numRtcpPackets)
-{
- CriticalSectionScoped cs(_statCrit);
- numRtpPackets = _rtpCount;
- numDroppedPackets = _dropCount;
- numRtcpPackets = _rtcpCount;
- return;
-}
-
-void TbExternalTransport::EnableSSRCCheck()
-{
- CriticalSectionScoped cs(_statCrit);
- _checkSSRC = true;
-}
-unsigned int TbExternalTransport::ReceivedSSRC()
-{
- CriticalSectionScoped cs(_statCrit);
- return _lastSSRC;
-}
-
-void TbExternalTransport::EnableSequenceNumberCheck()
-{
- CriticalSectionScoped cs(_statCrit);
- _checkSequenceNumber = true;
-}
-
-unsigned short TbExternalTransport::GetFirstSequenceNumber()
-{
- CriticalSectionScoped cs(_statCrit);
- return _firstSequenceNumber;
-}
-
-
-bool TbExternalTransport::ViEExternalTransportRun(void* object)
-{
- return static_cast<TbExternalTransport*>(object)->ViEExternalTransportProcess();
-}
-bool TbExternalTransport::ViEExternalTransportProcess()
-{
- unsigned int waitTime = KMaxWaitTimeMs;
-
- VideoPacket* packet = NULL;
-
- while (!_rtpPackets.empty())
- {
- // Take first packet in queue
- _crit.Enter();
- packet = _rtpPackets.front();
- WebRtc_Word64 timeToReceive = packet->receiveTime - NowMs();
- if (timeToReceive > 0)
- {
- // No packets to receive yet
- if (timeToReceive < waitTime &&
- timeToReceive > 0)
- {
- waitTime = (unsigned int) timeToReceive;
- }
- _crit.Leave();
- break;
- }
- _rtpPackets.pop();
- _crit.Leave();
-
- // Send to ViE
- if (packet)
- {
- {
- CriticalSectionScoped cs(_statCrit);
- if (_checkSSRC)
- {
- _lastSSRC = ((packet->packetBuffer[8]) << 24);
- _lastSSRC += (packet->packetBuffer[9] << 16);
- _lastSSRC += (packet->packetBuffer[10] << 8);
- _lastSSRC += packet->packetBuffer[11];
- _checkSSRC = false;
- }
- if (_checkSequenceNumber)
- {
- _firstSequenceNumber = (unsigned char)packet->packetBuffer[2] << 8;
- _firstSequenceNumber += (unsigned char)packet->packetBuffer[3];
- _checkSequenceNumber = false;
- }
- }
- /*
- unsigned short sequenceNumber = (unsigned char)packet->packetBuffer[2] << 8;
- sequenceNumber += (unsigned char)packet->packetBuffer[3];
-
- int marker=packet->packetBuffer[1] & 0x80;
- unsigned int timestamp=((((unsigned char*)packet->packetBuffer)[4]) << 24) + ((((unsigned char*)packet->packetBuffer)[5])<<16) +((((unsigned char*)packet->packetBuffer)[6])<<8)+(((unsigned char*)packet->packetBuffer)[7]);
- char str[256];
- sprintf(str,"Receiving seq %u length %d m %d, ts %u\n", sequenceNumber,packet->length,marker,timestamp) ;
- OutputDebugString(str);*/
-
- _vieNetwork.ReceivedRTPPacket(packet->channel, packet->packetBuffer, packet->length);
- delete packet;
- packet = NULL;
- }
- }
- while (!_rtcpPackets.empty())
- {
- // Take first packet in queue
- _crit.Enter();
- packet = _rtcpPackets.front();
- WebRtc_Word64 timeToReceive = packet->receiveTime - NowMs();
- if (timeToReceive > 0)
- {
- // No packets to receive yet
- if (timeToReceive < waitTime &&
- timeToReceive > 0)
- {
- waitTime = (unsigned int) timeToReceive;
- }
- _crit.Leave();
- break;
- }
- packet = _rtcpPackets.front();
- _rtcpPackets.pop();
- _crit.Leave();
-
- // Send to ViE
- if (packet)
- {
- _vieNetwork.ReceivedRTCPPacket(packet->channel, packet->packetBuffer, packet->length);
- delete packet;
- packet = NULL;
- }
- }
- _event.Wait(waitTime + 1); // Add 1 ms to not call to early...
- return true;
-}
-
-WebRtc_Word64 TbExternalTransport::NowMs()
-{
- return TickTime::MillisecondTimestamp();
-}
diff --git a/trunk/src/video_engine/main/test/WindowsTest/tbExternalTransport.h b/trunk/src/video_engine/main/test/WindowsTest/tbExternalTransport.h
deleted file mode 100644
index 53226c6..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/tbExternalTransport.h
+++ /dev/null
@@ -1,106 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// tbExternalTransport.h
-//
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_TBEXTERNALTRANSPORT_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_TBEXTERNALTRANSPORT_H_
-
-#include "common_types.h"
-#include <queue>
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-class EventWrapper;
-class ThreadWrapper;
-class ViENetwork;
-}
-
-class TbExternalTransport : public webrtc::Transport
-{
-public:
- TbExternalTransport(webrtc::ViENetwork& vieNetwork);
- ~TbExternalTransport(void);
-
- virtual int SendPacket(int channel, const void *data, int len);
- virtual int SendRTCPPacket(int channel, const void *data, int len);
-
- WebRtc_Word32 SetPacketLoss(WebRtc_Word32 lossRate); // Rate in %
- void SetNetworkDelay(WebRtc_Word64 delayMs);
-
- void ClearStats();
- void GetStats(WebRtc_Word32& numRtpPackets, WebRtc_Word32& numDroppedPackets, WebRtc_Word32& numRtcpPackets);
-
- void EnableSSRCCheck();
- unsigned int ReceivedSSRC();
-
- void EnableSequenceNumberCheck();
- unsigned short GetFirstSequenceNumber();
-
-
-protected:
- static bool ViEExternalTransportRun(void* object);
- bool ViEExternalTransportProcess();
-private:
- WebRtc_Word64 NowMs();
-
- enum { KMaxPacketSize = 1650};
- enum { KMaxWaitTimeMs = 100};
- typedef struct
- {
- WebRtc_Word8 packetBuffer[KMaxPacketSize];
- WebRtc_Word32 length;
- WebRtc_Word32 channel;
- WebRtc_Word64 receiveTime;
- } VideoPacket;
-
- typedef std::queue<VideoPacket*> VideoPacketQueue;
-
-
- webrtc::ViENetwork& _vieNetwork;
- webrtc::ThreadWrapper& _thread;
- webrtc::EventWrapper& _event;
- webrtc::CriticalSectionWrapper& _crit;
- webrtc::CriticalSectionWrapper& _statCrit;
-
- WebRtc_Word32 _lossRate;
- WebRtc_Word64 _networkDelayMs;
- WebRtc_Word32 _rtpCount;
- WebRtc_Word32 _rtcpCount;
- WebRtc_Word32 _dropCount;
-
- VideoPacketQueue _rtpPackets;
- VideoPacketQueue _rtcpPackets;
-
- bool _checkSSRC;
- WebRtc_UWord32 _lastSSRC;
- bool _checkSequenceNumber;
- WebRtc_UWord16 _firstSequenceNumber;
- WebRtc_Word32 _lastSeq;
-
- //int& numberOfErrors;
-
- //int _bits;
- //int _lastTicks;
- //int _dropCnt;
- //int _sentCount;
- //int _frameCount;
- //int _packetLoss;
-
- //VideoEngine* _video;
-
- //ReceiveBufferQueue _videoBufferQueue;
- //ReceiveBufferQueue _rtcpBufferQueue;
-};
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_WINDOWSTEST_TBEXTERNALTRANSPORT_H_
diff --git a/trunk/src/video_engine/main/test/WindowsTest/videosize.cc b/trunk/src/video_engine/main/test/WindowsTest/videosize.cc
deleted file mode 100644
index a675ec5..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/videosize.cc
+++ /dev/null
@@ -1,140 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "VideoSize.h"
-int GetWidthHeight( VideoSize size, int& width, int& height)
-{
- switch(size)
- {
- case SQCIF:
- width = 128;
- height = 96;
- return 0;
- case QQVGA:
- width = 160;
- height = 120;
- return 0;
- case QCIF:
- width = 176;
- height = 144;
- return 0;
- case CGA:
- width = 320;
- height = 200;
- return 0;
- case QVGA:
- width = 320;
- height = 240;
- return 0;
- case SIF:
- width = 352;
- height = 240;
- return 0;
- case WQVGA:
- width = 400;
- height = 240;
- return 0;
- case CIF:
- width = 352;
- height = 288;
- return 0;
- case W288P:
- width = 512;
- height = 288;
- return 0;
- case W368P:
- width = 640;
- height = 368;
- return 0;
- case S_448P:
- width = 576;
- height = 448;
- return 0;
- case VGA:
- width = 640;
- height = 480;
- return 0;
- case S_432P:
- width = 720;
- height = 432;
- return 0;
- case W432P:
- width = 768;
- height = 432;
- return 0;
- case S_4SIF:
- width = 704;
- height = 480;
- return 0;
- case W448P:
- width = 768;
- height = 448;
- return 0;
- case NTSC:
- width = 720;
- height = 480;
- return 0;
- case FW448P:
- width = 800;
- height = 448;
- return 0;
- case S_768x480P:
- width = 768;
- height = 480;
- return 0;
- case WVGA:
- width = 800;
- height = 480;
- return 0;
- case S_4CIF:
- width = 704;
- height = 576;
- return 0;
- case SVGA:
- width = 800;
- height = 600;
- return 0;
- case W544P:
- width = 960;
- height = 544;
- return 0;
- case W576P:
- width = 1024;
- height = 576;
- return 0;
- case HD:
- width = 960;
- height = 720;
- return 0;
- case XGA:
- width = 1024;
- height = 768;
- return 0;
- case FULL_HD:
- width = 1440;
- height = 1080;
- return 0;
- case WHD:
- width = 1280;
- height = 720;
- return 0;
- case UXGA:
- width = 1600;
- height = 1200;
- return 0;
- case WFULL_HD:
- width = 1920;
- height = 1080;
- return 0;
- default:
- return -1;
- }
- return -1;
-}
\ No newline at end of file
diff --git a/trunk/src/video_engine/main/test/WindowsTest/windowstest.gypi b/trunk/src/video_engine/main/test/WindowsTest/windowstest.gypi
deleted file mode 100644
index 0bebd98..0000000
--- a/trunk/src/video_engine/main/test/WindowsTest/windowstest.gypi
+++ /dev/null
@@ -1,84 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'conditions': [
- ['OS=="win"', {
- 'targets': [
- # WinTest - GUI test for Windows
- {
- 'target_name': 'vie_win_test',
- 'type': 'executable',
- 'dependencies': [
- '<(webrtc_root)/modules/modules.gyp:video_render_module',
- '<(webrtc_root)/modules/modules.gyp:video_capture_module',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ## VoiceEngine
- '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine_core',
- ## VideoEngine
- 'video_engine_core',
- ],
- 'include_dirs': [
- './interface',
- '../../../../', # common_types.h and typedefs.h
- '../commonTestClasses/'
- ],
- 'sources': [
- 'Capture.rc',
- 'captureDeviceImage.jpg',
- 'ChannelDlg.cc',
- 'ChannelDlg.h',
- 'ChannelPool.cc',
- 'ChannelPool.h',
- 'renderStartImage.jpg',
- 'renderTimeoutImage.jpg',
- 'res\Capture.rc2',
- 'resource.h',
- 'StdAfx.h',
- 'videosize.cc',
- 'VideoSize.h',
- 'WindowsTest.cc',
- 'WindowsTest.h',
- 'WindowsTestMainDlg.cc',
- 'WindowsTestMainDlg.h',
- 'WindowsTestResouce.rc',
- 'WindowsTestResource.h',
- 'tbExternalTransport.cc',
- 'CaptureDevicePool.cc',
- 'tbExternalTransport.h',
- 'CaptureDevicePool.h',
-
- ],
- 'configurations': {
- 'Common_Base': {
- 'msvs_configuration_attributes': {
- 'conditions': [
- ['component=="shared_library"', {
- 'UseOfMFC': '2', # Shared DLL
- },{
- 'UseOfMFC': '1', # Static
- }],
- ],
- },
- },
- },
- 'msvs_settings': {
- 'VCLinkerTool': {
- 'SubSystem': '2', # Windows
- },
- },
- },
- ],
- }],
- ],
-}
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/video_engine/main/test/android_test/.classpath b/trunk/src/video_engine/main/test/android_test/.classpath
deleted file mode 100644
index f2adf55..0000000
--- a/trunk/src/video_engine/main/test/android_test/.classpath
+++ /dev/null
@@ -1,9 +0,0 @@
-<?xml version="1.0" encoding="UTF-8"?>
-<classpath>
- <classpathentry kind="src" path="src"/>
- <classpathentry kind="src" path="gen"/>
- <classpathentry kind="con" path="com.android.ide.eclipse.adt.ANDROID_FRAMEWORK"/>
- <classpathentry kind="lib" path="libs/VideoEngine_android_java.jar"/>
- <classpathentry kind="lib" path="libs/VoiceEngine_android_java.jar"/>
- <classpathentry kind="output" path="bin"/>
-</classpath>
diff --git a/trunk/src/video_engine/main/test/android_test/Android.mk b/trunk/src/video_engine/main/test/android_test/Android.mk
deleted file mode 100644
index 3ee8627..0000000
--- a/trunk/src/video_engine/main/test/android_test/Android.mk
+++ /dev/null
@@ -1,41 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH:= $(call my-dir)
-
-include $(CLEAR_VARS)
-
-MY_CAPTURE_FOLDER := ../../../../modules/video_capture/main/source
-MY_CAPTURE_JAVA_FOLDER := Android/java/org/webrtc/videoengine
-MY_CAPTURE_PATH := $(MY_CAPTURE_FOLDER)/$(MY_CAPTURE_JAVA_FOLDER)
-
-MY_RENDER_FOLDER := ../../../../modules/video_render/main/source
-MY_RENDER_JAVA_FOLDER := Android/java/org/webrtc/videoengine
-MY_RENDER_PATH := $(MY_RENDER_FOLDER)/$(MY_RENDER_JAVA_FOLDER)
-
-LOCAL_MODULE_TAGS := tests
-
-LOCAL_SRC_FILES := \
- src/org/webrtc/videoengineapp/ViEAndroidDemo.java \
- src/org/webrtc/videoengineapp/ViEAndroidJavaAPI.java \
- src/org/webrtc/videoengineapp/IViEAndroidCallback.java \
- $(MY_CAPTURE_PATH)/CaptureCapabilityAndroid.java \
- $(MY_CAPTURE_PATH)/VideoCaptureAndroid.java \
- $(MY_CAPTURE_PATH)/VideoCaptureDeviceInfoAndroid.java \
- $(MY_RENDER_PATH)/ViEAndroidGLES20.java \
- $(MY_RENDER_PATH)/ViERenderer.java \
- $(MY_RENDER_PATH)/ViESurfaceRenderer.java
-
-LOCAL_PACKAGE_NAME := webrtc-video-demo
-LOCAL_CERTIFICATE := platform
-
-LOCAL_JNI_SHARED_LIBRARIES := libwebrtc-video-demo-jni
-
-include $(BUILD_PACKAGE)
-
-include $(call all-makefiles-under,$(LOCAL_PATH))
diff --git a/trunk/src/video_engine/main/test/android_test/AndroidManifest.xml b/trunk/src/video_engine/main/test/android_test/AndroidManifest.xml
deleted file mode 100644
index 0714aed..0000000
--- a/trunk/src/video_engine/main/test/android_test/AndroidManifest.xml
+++ /dev/null
@@ -1,28 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<manifest xmlns:android="http://schemas.android.com/apk/res/android"
- android:versionCode="1" package="org.webrtc.videoengineapp" android:versionName="1.07">
- <application android:icon="@drawable/logo"
- android:label="@string/app_name"
- android:debuggable="true">
- <activity android:name=".ViEAndroidDemo"
- android:label="@string/app_name"
- android:configChanges="keyboardHidden|orientation"
- >
- <!--android:configChanges="keyboardHidden|orientation" -->
- <!-- android:screenOrientation="landscape" -->
- <intent-filter>
- <action android:name="android.intent.action.MAIN" />
- <category android:name="android.intent.category.LAUNCHER" />
- </intent-filter>
- </activity>
- </application>
- <uses-sdk android:minSdkVersion="7" />
- <uses-permission android:name="android.permission.CAMERA"></uses-permission>
- <uses-feature android:name="android.hardware.camera" />
- <uses-feature android:name="android.hardware.camera.autofocus" />
- <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
- <uses-permission android:name="android.permission.RECORD_AUDIO" />
- <uses-permission android:name="android.permission.INTERNET" />
- <uses-permission android:name="android.permission.WAKE_LOCK" />
- <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/>
-</manifest>
diff --git a/trunk/src/video_engine/main/test/android_test/default.properties b/trunk/src/video_engine/main/test/android_test/default.properties
deleted file mode 100644
index c206f58..0000000
--- a/trunk/src/video_engine/main/test/android_test/default.properties
+++ /dev/null
@@ -1,13 +0,0 @@
-# This file is automatically generated by Android Tools.
-# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
-#
-# This file must be checked in Version Control Systems.
-#
-# To customize properties used by the Ant build system use,
-# "build.properties", and override values to adapt the script to your
-# project structure.
-
-# Indicates whether an apk should be generated for each density.
-split.density=false
-# Project target.
-target=android-9
diff --git a/trunk/src/video_engine/main/test/android_test/gen/org/webrtc/androidapp/R.java b/trunk/src/video_engine/main/test/android_test/gen/org/webrtc/androidapp/R.java
deleted file mode 100644
index c35047b..0000000
--- a/trunk/src/video_engine/main/test/android_test/gen/org/webrtc/androidapp/R.java
+++ /dev/null
@@ -1,68 +0,0 @@
-/* AUTO-GENERATED FILE. DO NOT MODIFY.
- *
- * This class was automatically generated by the
- * aapt tool from the resource data it found. It
- * should not be modified by hand.
- */
-
-package org.webrtc.videoengineapp;
-
-public final class R {
- public static final class array {
- public static final int codecSize=0x7f040001;
- public static final int codectype=0x7f040000;
- public static final int voiceCodecType=0x7f040002;
- }
- public static final class attr {
- }
- public static final class drawable {
- public static final int bar=0x7f020000;
- public static final int bg=0x7f020001;
- public static final int logo=0x7f020002;
- public static final int robot=0x7f020003;
- public static final int video=0x7f020004;
- }
- public static final class id {
- public static final int LinearLayout01=0x7f060010;
- public static final int LinearLayout02=0x7f060006;
- public static final int TextView01=0x7f060005;
- public static final int TextView02=0x7f06000b;
- public static final int TextView03=0x7f060004;
- public static final int btStartBoth=0x7f060013;
- public static final int btStartListen=0x7f060011;
- public static final int btStartSend=0x7f060012;
- public static final int cbLoopback=0x7f06000e;
- public static final int cbVoice=0x7f06000d;
- public static final int etRemoteIp=0x7f06000c;
- public static final int ivPreview=0x7f060014;
- public static final int ivTopBar=0x7f060002;
- public static final int rlSurfaces=0x7f060000;
- public static final int spCodecSize=0x7f06000a;
- public static final int spCodecType=0x7f060007;
- public static final int spVoiceCodecType=0x7f060008;
- public static final int svLocal=0x7f060001;
- public static final int tvCodecSize=0x7f060009;
- public static final int tvLocalIp=0x7f06000f;
- public static final int tvTitle=0x7f060003;
- }
- public static final class layout {
- public static final int both=0x7f030000;
- public static final int main=0x7f030001;
- public static final int send=0x7f030002;
- }
- public static final class string {
- public static final int app_name=0x7f050001;
- public static final int codecSize=0x7f050007;
- public static final int codecType=0x7f050006;
- public static final int codectype_prompt=0x7f050004;
- public static final int demoTitle=0x7f050005;
- public static final int enableVoice=0x7f05000d;
- public static final int error=0x7f050002;
- public static final int errorCamera=0x7f050003;
- public static final int loopback=0x7f050009;
- public static final int remoteIp=0x7f050008;
- public static final int startBoth=0x7f05000c;
- public static final int startListen=0x7f05000a;
- public static final int startSend=0x7f05000b;
- }
-}
diff --git a/trunk/src/video_engine/main/test/android_test/jni/Android.mk b/trunk/src/video_engine/main/test/android_test/jni/Android.mk
deleted file mode 100644
index 4121cc6..0000000
--- a/trunk/src/video_engine/main/test/android_test/jni/Android.mk
+++ /dev/null
@@ -1,40 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-INTERFACES_PATH := $(LOCAL_PATH)/../../../../../../../build/interface
-LIBS_PATH := $(LOCAL_PATH)/../../../../../../../build/libraries
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_TAGS := tests
-LOCAL_MODULE := libwebrtc-video-demo-jni
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := vie_android_java_api.cc
-LOCAL_CFLAGS := \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_ANDROID'
-
-LOCAL_C_INCLUDES := \
- external/gtest/include \
- $(LOCAL_PATH)/../../../../.. \
- $(LOCAL_PATH)/../../../../include \
- $(LOCAL_PATH)/../../../../../voice_engine/main/interface
-
-LOCAL_PRELINK_MODULE := false
-
-LOCAL_SHARED_LIBRARIES := \
- libutils \
- libstlport \
- libandroid \
- libwebrtc \
- libGLESv2
-LOCAL_LDLIBS := $(LIBS_PATH)/VideoEngine_android_gcc.a -llog -lgcc
-
-include $(BUILD_SHARED_LIBRARY)
diff --git a/trunk/src/video_engine/main/test/android_test/jni/org_webrtc_videoengineapp_vie_android_java_api.h b/trunk/src/video_engine/main/test/android_test/jni/org_webrtc_videoengineapp_vie_android_java_api.h
deleted file mode 100644
index 37a9c43..0000000
--- a/trunk/src/video_engine/main/test/android_test/jni/org_webrtc_videoengineapp_vie_android_java_api.h
+++ /dev/null
@@ -1,494 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/* DO NOT EDIT THIS FILE - it is machine generated */
-#include <jni.h>
-/* Header for class org_webrtc_videoengineapp_ViEAndroidJavaAPI */
-
-#ifndef _Included_org_webrtc_videoengineapp_ViEAndroidJavaAPI
-#define _Included_org_webrtc_videoengineapp_ViEAndroidJavaAPI
-#ifdef __cplusplus
-extern "C" {
-#endif
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: NativeInit
- * Signature: (Landroid/content/Context;)Z
- */
-JNIEXPORT jboolean JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_NativeInit
-(JNIEnv *, jobject, jobject);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: GetVideoEngine
- * Signature: ()I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_GetVideoEngine
-(JNIEnv *, jobject);
-
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: Init
- * Signature: (IIIZ)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_Init
-(JNIEnv *, jobject, jboolean);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: Terminate
- * Signature: ()I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_Terminate
-(JNIEnv *, jobject);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: StartSend
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartSend
-(JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: StopRender
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopRender
-(JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: StopSend
- * Signature: ()I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopSend
-(JNIEnv *, jobject,jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: StartReceive
- * Signature: ()I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartReceive
-(JNIEnv *, jobject,jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: StopReceive
- * Signature: ()I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopReceive
-(JNIEnv *, jobject,jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: CreateChannel
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_CreateChannel
-(JNIEnv *, jobject,jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: SetLocalReceiver
- * Signature: (II)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetLocalReceiver
-(JNIEnv *, jobject, jint, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: SetSendDestination
- * Signature: (II[B)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetSendDestination
-(JNIEnv *, jobject, jint, jint, jbyteArray);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: SetReceiveCodec
- * Signature: (IIIIIIZ)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetReceiveCodec
-(JNIEnv *, jobject, jint, jint, jint, jint, jint, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: SetSendCodec
- * Signature: (IIIIIIZ)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetSendCodec
-(JNIEnv *, jobject, jint, jint, jint, jint, jint, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: AddRemoteRenderer
- * Signature: (ILandroid/view/SurfaceView;)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_AddRemoteRenderer
-(JNIEnv *, jobject, jint, jobject);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: RemoveRemoteRenderer
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_RemoveRemoteRenderer
-(JNIEnv *, jobject,jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: StartRender
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartRender
-(JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: StartCamera
- * Signature: (II)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartCamera
-(JNIEnv *, jobject,jint channel,jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: StopCamera
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopCamera
-(JNIEnv *, jobject,jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: GetCameraOrientation
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_GetCameraOrientation
-(JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: SetRotation
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetRotation
-(JNIEnv *, jobject, jint, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: EnableNACK
- * Signature: (IZ)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_EnableNACK
-(JNIEnv *, jobject, jint, jboolean);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: EnablePLI
- * Signature: (IZ)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_EnablePLI
-(JNIEnv *, jobject, jint, jboolean);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: StartSendNative
- * Signature: (III[BIIIIIILandroid/view/SurfaceView;III)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartSendNative
-(JNIEnv *, jobject, jint, jint, jint, jbyteArray, jint, jint,
-jint, jint, jint, jint, jobject, jint, jint, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: StartListenNative
- * Signature: (III[BIIIIIIILandroid/view/SurfaceView;)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartListenNative
-(JNIEnv *, jobject, jint, jint, jint, jbyteArray, jint, jint,
-jint, jint, jint, jint, jint, jobject);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: StopAllNative
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopAllNative
-(JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: SetCallback
- * Signature: (ILorg/webrtc/videoengineapp/IViEAndroidCallback;)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetCallback
-(JNIEnv *, jobject, jint, jobject);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_Create
- * Signature: (Landroid/app/Activity;)Z
- */
-JNIEXPORT jboolean JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_Create
-(JNIEnv *, jobject, jobject);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_Delete
- * Signature: ()Z
- */
-JNIEXPORT jboolean JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_Delete
-(JNIEnv *, jobject);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_Init
- * Signature: (IIIZZ)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_Init
-(JNIEnv *, jobject, jint, jint, jint, jboolean, jboolean);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_Terminate
- * Signature: ()I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_Terminate
-(JNIEnv *, jobject);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_CreateChannel
- * Signature: ()I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_CreateChannel
-(JNIEnv *, jobject);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_DeleteChannel
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_DeleteChannel
-(JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_SetLocalReceiver
- * Signature: (II)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetLocalReceiver
-(JNIEnv *, jobject, jint, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_SetSendDestination
- * Signature: (IILjava/lang/String;)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetSendDestination
-(JNIEnv *, jobject, jint, jint, jstring);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_StartListen
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartListen
-(JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_StartPlayout
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartPlayout
-(JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_StartSend
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartSend
-(JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_StopListen
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopListen
-(JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_StopPlayout
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopPlayout
-(JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_StopSend
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopSend
-(JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_SetSpeakerVolume
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetSpeakerVolume
-(JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_SetLoudspeakerStatus
- * Signature: (Z)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetLoudspeakerStatus
-(JNIEnv *, jobject, jboolean);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_StartPlayingFileLocally
- * Signature: (ILjava/lang/String;Z)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartPlayingFileLocally
-(JNIEnv *, jobject, jint, jstring, jboolean);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_StopPlayingFileLocally
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopPlayingFileLocally
-(JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_StartPlayingFileAsMicrophone
- * Signature: (ILjava/lang/String;Z)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartPlayingFileAsMicrophone
-(JNIEnv *, jobject, jint, jstring, jboolean);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_StopPlayingFileAsMicrophone
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopPlayingFileAsMicrophone
-(JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_NumOfCodecs
- * Signature: ()I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_NumOfCodecs
-(JNIEnv *, jobject);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_SetSendCodec
- * Signature: (II)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetSendCodec
-(JNIEnv *, jobject, jint, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_SetECStatus
- * Signature: (ZI)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetECStatus
-(JNIEnv *, jobject, jboolean, jint, jint, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_SetNSStatus
- * Signature: (ZI)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetNSStatus
-(JNIEnv *, jobject, jboolean, jint);
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: VE_SetAGCStatus
- * Signature: (ZI)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetAGCStatus
-(JNIEnv *, jobject, jboolean, jint);
-
-#ifdef __cplusplus
-}
-#endif
-#endif
diff --git a/trunk/src/video_engine/main/test/android_test/jni/vie_android_java_api.cc b/trunk/src/video_engine/main/test/android_test/jni/vie_android_java_api.cc
deleted file mode 100644
index 112667f..0000000
--- a/trunk/src/video_engine/main/test/android_test/jni/vie_android_java_api.cc
+++ /dev/null
@@ -1,1563 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <string.h>
-#include <android/log.h>
-
-#include "org_webrtc_videoengineapp_vie_android_java_api.h"
-
-#include "voe_base.h"
-#include "voe_codec.h"
-#include "voe_file.h"
-#include "voe_network.h"
-#include "voe_audio_processing.h"
-#include "voe_volume_control.h"
-#include "voe_hardware.h"
-
-#include "vie_base.h"
-#include "vie_codec.h"
-#include "vie_capture.h"
-#include "vie_network.h"
-#include "vie_render.h"
-#include "vie_rtp_rtcp.h"
-
-#include "common_types.h"
-
-#define WEBRTC_LOG_TAG "*WEBRTCN*"
-#define VALIDATE_BASE_POINTER \
- if (!veData.base) \
- { \
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
- "Base pointer doesn't exist"); \
- return -1; \
- }
-#define VALIDATE_CODEC_POINTER \
- if (!veData.codec) \
- { \
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
- "Codec pointer doesn't exist"); \
- return -1; \
- }
-#define VALIDATE_FILE_POINTER \
- if (!veData.file) \
- { \
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
- "File pointer doesn't exist"); \
- return -1; \
- }
-#define VALIDATE_APM_POINTER \
- if (!veData.codec) \
- { \
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
- "Apm pointer doesn't exist"); \
- return -1; \
- }
-#define VALIDATE_HARDWARE_POINTER \
- if (!veData.hardware) \
- { \
- __android_log_write( \
- ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
- "Hardware pointer doesn't exist"); \
- return -1; \
- }
-#define VALIDATE_VOLUME_POINTER \
- if (!veData.volume) \
- { \
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
- "Volume pointer doesn't exist"); \
- return -1; \
- }
-
-using namespace webrtc;
-
-//Forward declaration.
-class VideoCallbackAndroid;
-
-// VoiceEngine data struct
-typedef struct
-{
- // VoiceEngine
- VoiceEngine* ve;
- // Sub-APIs
- VoEBase* base;
- VoECodec* codec;
- VoEFile* file;
- VoENetwork* netw;
- VoEAudioProcessing* apm;
- VoEVolumeControl* volume;
- VoEHardware* hardware;
- JavaVM* jvm;
-} VoiceEngineData;
-
-class AndroidVideoRenderCallback;
-// VideoEngine data struct
-typedef struct
-{
- VideoEngine* vie;
- ViEBase* base;
- ViECodec* codec;
- ViENetwork* netw;
- ViERTP_RTCP* rtp;
- ViERender* render;
- ViECapture* capture;
- VideoCallbackAndroid* callback;
-
-} VideoEngineData;
-
-// Global variables
-JavaVM* webrtcGlobalVM;
-
-// Global variables visible in this file
-static VoiceEngineData veData;
-static VideoEngineData vieData;
-
-// "Local" functions (i.e. not Java accessible)
-#define WEBRTC_TRACE_MAX_MESSAGE_SIZE 1024
-static bool VE_GetSubApis();
-static bool VE_ReleaseSubApis();
-
-#define CHECK_API_RETURN(ret) \
- if (ret!=0) \
- { \
- __android_log_print(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
- "Return error %d",ret); \
- break; \
- }
-
-class VideoCallbackAndroid: public ViEDecoderObserver,
- public ViEEncoderObserver
-{
-
- // Implements ViEDecoderObserver
- virtual void IncomingRate(const int videoChannel,
- const unsigned int framerate,
- const unsigned int bitrate)
- {
- // Let's print out the network statistics from this call back as well
- unsigned short fraction_lost;
- unsigned int dummy;
- int intdummy;
- _vieData.rtp->GetReceivedRTCPStatistics(videoChannel, fraction_lost,
- dummy, dummy, dummy, intdummy);
- unsigned short packetLossRate = 0;
- if (fraction_lost > 0)
- {
- // Change from frac to %
- packetLossRate = (fraction_lost * 100) >> 8;
- }
-
- JNIEnv* threadEnv = NULL;
- int ret = webrtcGlobalVM->AttachCurrentThread(&threadEnv, NULL);
- // Get the JNI env for this thread
- if ((ret < 0) || !threadEnv)
- {
- __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "Could not attach thread to JVM (%d, %p)", ret,
- threadEnv);
- return;
- }
- threadEnv->CallIntMethod(_callbackObj, _callbackId, framerate, bitrate,
- packetLossRate, _frameRateO, _bitRateO);
- webrtcGlobalVM->DetachCurrentThread();
- }
- ;
-
- virtual void IncomingCodecChanged(const int videoChannel,
- const webrtc::VideoCodec& videoCodec)
- {
- }
- ;
-
- virtual void RequestNewKeyFrame(const int videoChannel)
- {
- }
- ;
-
- virtual void OutgoingRate(const int videoChannel,
- const unsigned int framerate,
- const unsigned int bitrate)
- {
- _frameRateO = framerate;
- _bitRateO = bitrate;
- //__android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- // "SendRate frameRate %d bitrate %d\n",frameRate,bitrate);
- }
- ;
-
-public:
- VideoEngineData& _vieData;
- JNIEnv * _env;
- jobject _callbackObj;
- jclass _callbackCls;
- jmethodID _callbackId;
- int _frameRateO, _bitRateO;
- VideoCallbackAndroid(VideoEngineData& vieData, JNIEnv * env,
- jobject callback) :
- _vieData(vieData), _env(env), _callbackObj(callback),
- _frameRateO(0), _bitRateO(0) {
- _callbackCls = _env->GetObjectClass(_callbackObj);
- _callbackId
- = _env->GetMethodID(_callbackCls, "UpdateStats", "(IIIII)I");
- if (_callbackId == NULL) {
- __android_log_print(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to get jid");
- }
- _callbackObj = _env->NewGlobalRef(_callbackObj);
- }
-};
-
-// JNI_OnLoad
-jint JNI_OnLoad(JavaVM* vm, void* /*reserved*/) {
- webrtcGlobalVM = vm;
- if (!webrtcGlobalVM)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "JNI_OnLoad did not receive a valid VM pointer");
- return -1;
- }
-
- // Get JNI
- JNIEnv* env;
- if (JNI_OK != vm->GetEnv(reinterpret_cast<void**> (&env),
- JNI_VERSION_1_4)) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "JNI_OnLoad could not get JNI env");
- return -1;
- }
-
- // Init VoiceEngine data
- memset(&veData, 0, sizeof(veData));
- // Store the JVM
- veData.jvm = vm;
-
- // Init VideoEngineData data
- memset(&vieData, 0, sizeof(vieData));
-
- return JNI_VERSION_1_4;
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: NativeInit
-// Signature: (Landroid/content/Context;)Z
-JNIEXPORT jboolean JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_NativeInit(
- JNIEnv * env,
- jobject,
- jobject context)
-{
- return true;
-}
-
-// VideoEngine API wrapper functions
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: GetVideoEngine
-// Signature: ()I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_GetVideoEngine(
- JNIEnv *,
- jobject context) {
-
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "GetVideoEngine");
-
- VideoEngine::SetAndroidObjects(webrtcGlobalVM, context);
-
- // Check if already got
- if (vieData.vie) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "ViE already got");
- return -1;
- }
-
- // Create
- vieData.vie = VideoEngine::Create();
- if (!vieData.vie) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, "Get ViE failed");
- return -1;
- }
- vieData.base = ViEBase::GetInterface(vieData.vie);
- if (!vieData.base) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get base sub-API failed");
- return -1;
- }
-
- vieData.codec = ViECodec::GetInterface(vieData.vie);
- if (!vieData.codec) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get codec sub-API failed");
- return -1;
- }
-
- vieData.netw = ViENetwork::GetInterface(vieData.vie);
- if (!vieData.netw) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get network sub-API failed");
- return -1;
- }
-
- vieData.rtp = ViERTP_RTCP::GetInterface(vieData.vie);
- if (!vieData.rtp) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get RTP sub-API failed");
- return -1;
- }
-
- vieData.render = ViERender::GetInterface(vieData.vie);
- if (!vieData.render) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get Render sub-API failed");
- return -1;
- }
-
- vieData.capture = ViECapture::GetInterface(vieData.vie);
- if (!vieData.capture) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get Capture sub-API failed");
- return -1;
- }
-
- return 0;
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: Init
-// Signature: (IIIZ)I
-JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_Init(
- JNIEnv *,
- jobject,
- jboolean enableTrace)
-{
- if (vieData.vie) {
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "Init");
-
- int ret = vieData.base->Init();
- __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "Init return %d", ret);
- if (enableTrace)
- {
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "SetTraceFile");
- if (0 != vieData.vie->SetTraceFile(("/sdcard/trace.txt"), false))
- {
- __android_log_print(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Video Engine could not enable trace");
- }
-
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "SetTraceFilter");
- if (0 != vieData.vie->SetTraceFilter(webrtc::kTraceDefault))
- {
- __android_log_write(ANDROID_LOG_WARN, WEBRTC_LOG_TAG,
- "Could not set trace filter");
- }
- }
- else
- {
- if (0 != vieData.vie->SetTraceFilter(webrtc::kTraceNone))
- {
- __android_log_write(ANDROID_LOG_WARN, WEBRTC_LOG_TAG,
- "Could not set trace filter");
- }
- }
- if (veData.ve) // VoiceEngine is enabled
- {
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "SetVoiceEngine");
- if (0 != vieData.base->SetVoiceEngine(veData.ve))
- {
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "SetVoiceEngine failed");
- }
- }
- return ret;
- }
- else
- {
- return -1;
- }
-}
-
-/*
- * Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
- * Method: Terminate
- * Signature: ()I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_Terminate(
- JNIEnv *,
- jobject)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "Terminate");
-
- if (vieData.vie) {
- if (!vieData.rtp || vieData.rtp->Release() != 0) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to release RTP sub-API");
- }
-
- if (!vieData.netw || vieData.netw->Release() != 0) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to release Network sub-API");
- }
-
- if (!vieData.codec || vieData.codec->Release() != 0) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to release Codec sub-API");
- }
-
- if (!vieData.render || vieData.render->Release()) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to release Render sub-API");
- }
-
- if (!vieData.capture || vieData.capture->Release()) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to release Capture sub-API");
- }
-
- if (!vieData.base || vieData.base->Release() != 0) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to release Base sub-API");
- }
-
- // Delete Vie
- if (!VideoEngine::Delete(vieData.vie)) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to delete ViE ");
- return -1;
- }
- memset(&vieData, 0, sizeof(vieData));
- VideoEngine::SetAndroidObjects(NULL, NULL);
- return 0;
- }
- else {
- return -1;
- }
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: StartSend
-// Signature: (I)I
-JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartSend(
- JNIEnv *,
- jobject,
- jint channel)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StartSend");
-
- if (vieData.base) {
- int ret = vieData.base->StartSend(channel);
- return ret;
- }
- else {
- return -1;
- }
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: StopRender
-// Signature: (I)I
-JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopRender(
- JNIEnv *,
- jobject,
- jint channel)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StopRender");
-
- if (vieData.render) {
- return vieData.render->StopRender(channel);
- }
- else {
- return -1;
- }
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: Stop
-// Signature: ()I
-JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopSend(
- JNIEnv *,
- jobject,
- jint channel)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StopSend");
-
- if (vieData.base) {
- return vieData.base->StopSend(channel);
- }
- else {
- return -1;
- }
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: StartReceive
-// Signature: ()I
-JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartReceive(
- JNIEnv *,
- jobject,
- jint channel)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StartReceive");
-
- if (vieData.base) {
- return vieData.base->StartReceive(channel);
- }
- else {
- return -1;
- }
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: StopReceive
-// Signature: ()I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopReceive(
- JNIEnv *,
- jobject,
- jint channel)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StopReceive");
- if (vieData.base) {
- return vieData.base->StopReceive(channel);
- }
- else {
- return -1;
- }
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: CreateChannel
-// Signature: ()I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_CreateChannel(
- JNIEnv *,
- jobject,
- jint voiceChannel)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "CreateChannel");
-
- if (vieData.vie) {
- int channel = 0;
- if (vieData.base->CreateChannel(channel) != 0) {
- return -1;
- }
- if (voiceChannel >= 0) {
- vieData.base->ConnectAudioChannel(channel, voiceChannel);
- }
-
- return channel;
- }
- else {
- return -1;
- }
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: SetLocalReceiver
-// Signature: (II)I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetLocalReceiver(
- JNIEnv *,
- jobject,
- jint channel,
- jint port)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "SetLocalReceiver");
-
- if (vieData.vie) {
- int ret = vieData.netw->SetLocalReceiver(channel, port);
- return ret;
- }
- else {
- return -1;
- }
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: SetSendDestination
-// Signature: (II[B)I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetSendDestination(
- JNIEnv * env,
- jobject,
- jint channel,
- jint port,
- jbyteArray ipadr)
-{
-
- if (NULL == vieData.vie)
- return -1;
-
- char ip[64];
- jsize len = env->GetArrayLength(ipadr);
- if ((len >= 64) || (len == 0))
- return -1;
- env->GetByteArrayRegion(ipadr, 0, len, (jbyte*) ip);
- ip[len] = '\0';
-
- __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "SetSendDestination: channel=%d, port=%d, ip=%s\n",
- channel, port, ip);
-
- return vieData.netw->SetSendDestination(channel, ip, port);
-}
-
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: SetReceiveCodec
-// Signature: (IIIIII)I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetReceiveCodec(
- JNIEnv *,
- jobject,
- jint channel,
- jint codecNum,
- jint intbitRate,
- jint width,
- jint height,
- jint frameRate)
-{
- if (NULL == vieData.codec)
- return -1;
-
- //Create codec
- webrtc::VideoCodec codec;
- vieData.codec->GetCodec(codecNum, codec);
-
- __android_log_print(
- ANDROID_LOG_DEBUG,
- WEBRTC_LOG_TAG,
- "SetReceiveCodec %s, pltype=%d, bitRate=%d, maxBitRate=%d,"
- " width=%d, height=%d, frameRate=%d, codecSpecific=%d \n",
- codec.plName, codec.plType, codec.startBitrate,
- codec.maxBitrate, codec.width, codec.height,
- codec.maxFramerate, codec.codecSpecific);
- int ret = vieData.codec->SetReceiveCodec(channel, codec);
- __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "SetReceiveCodec return %d", ret);
- return ret;
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: SetSendCodec
-// Signature: (IIIIII)I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetSendCodec(
- JNIEnv *,
- jobject,
- jint channel,
- jint codecNum,
- jint intbitRate,
- jint width,
- jint height,
- jint frameRate)
-{
- if (NULL == vieData.codec)
- return -1;
-
- //Create codec
- webrtc::VideoCodec codec;
- vieData.codec->GetCodec(codecNum, codec);
- codec.startBitrate = intbitRate;
- codec.maxBitrate = 600;
- codec.width = width;
- codec.height = height;
- codec.maxFramerate = frameRate;
-
- for (int i = 0; i < vieData.codec->NumberOfCodecs(); ++i) {
- webrtc::VideoCodec codecToList;
- vieData.codec->GetCodec(i, codecToList);
- __android_log_print(
- ANDROID_LOG_DEBUG,
- WEBRTC_LOG_TAG,
- "Codec list %s, pltype=%d, bitRate=%d, maxBitRate=%d,"
- " width=%d, height=%d, frameRate=%d\n",
- codecToList.plName, codecToList.plType,
- codecToList.startBitrate, codecToList.maxBitrate,
- codecToList.width, codecToList.height,
- codecToList.maxFramerate);
- }
- __android_log_print(
- ANDROID_LOG_DEBUG,
- WEBRTC_LOG_TAG,
- "SetSendCodec %s, pltype=%d, bitRate=%d, maxBitRate=%d, "
- "width=%d, height=%d, frameRate=%d\n",
- codec.plName, codec.plType, codec.startBitrate,
- codec.maxBitrate, codec.width, codec.height,
- codec.maxFramerate);
-
- return vieData.codec->SetSendCodec(channel, codec);
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: AddRemoteRenderer
-// Signature: (ILandroid/view/SurfaceView;)I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_AddRemoteRenderer(
- JNIEnv *,
- jobject,
- jint channel,
- jobject glSurface)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "AddRemoteRenderer");
- if (vieData.vie) {
- return vieData.render->AddRenderer(channel, glSurface, 0, 0, 0, 1, 1);
- }
- else {
- return -1;
- }
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: RemoveRemoteRenderer
-// Signature: (I)I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_RemoveRemoteRenderer(
- JNIEnv *,
- jobject,
- jint channel)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "RemoveRemoteRenderer");
-
- if (vieData.vie) {
- return vieData.render->RemoveRenderer(channel);
- }
- else {
- return -1;
- }
- return 0;
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: StartRender
-// Signature: (I)I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartRender(
- JNIEnv *,
- jobject,
- jint channel)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StartRender");
-
- if (vieData.render) {
- return vieData.render->StartRender(channel);
- }
- else {
- return -1;
- }
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: StarteCamera
-// Signature: (II)I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StartCamera(
- JNIEnv * env,
- jobject,
- jint channel,
- jint cameraNum)
-{
- if (NULL == vieData.vie)
- return -1;
-
- int i = 0;
- char deviceName[64];
- char deviceUniqueName[64];
- int re;
- do {
- re = vieData.capture->GetCaptureDevice(i, deviceName,
- sizeof(deviceName),
- deviceUniqueName,
- sizeof(deviceUniqueName));
- __android_log_print(
- ANDROID_LOG_DEBUG,
- WEBRTC_LOG_TAG,
- "GetCaptureDevice ret %d devicenum %d deviceUniqueName %s",
- re, i, deviceUniqueName);
- i++;
- } while (re == 0);
-
- int ret;
- int cameraId;
- vieData.capture->GetCaptureDevice(cameraNum, deviceName,
- sizeof(deviceName), deviceUniqueName,
- sizeof(deviceUniqueName));
- vieData.capture->AllocateCaptureDevice(deviceUniqueName,
- sizeof(deviceUniqueName), cameraId);
-
- if (cameraId >= 0) { //Connect the
- ret = vieData.capture->ConnectCaptureDevice(cameraId, channel);
- __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "ConnectCaptureDevice ret %d ", ret);
-
- ret = vieData.capture->StartCapture(cameraId);
- __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "StartCapture ret %d ", ret);
- }
-
- return cameraId;
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: StopCamera
-// Signature: ()I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_StopCamera(
- JNIEnv *,
- jobject,
- jint cameraId)
-{
- if (NULL == vieData.capture)
- return -1;
-
- int ret = vieData.capture->StopCapture(cameraId);
- __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "StopCapture ret %d ", ret);
- ret = vieData.capture->ReleaseCaptureDevice(cameraId);
- __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "ReleaseCaptureDevice ret %d ", ret);
-
- return ret;
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: GetCameraOrientation
-// Signature: (I)I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_GetCameraOrientation(
- JNIEnv *,
- jobject,
- jint cameraNum)
-{
- char deviceName[64];
- char deviceUniqueName[64];
- int ret;
-
- ret = vieData.capture->GetCaptureDevice(cameraNum, deviceName,
- sizeof(deviceName),
- deviceUniqueName,
- sizeof(deviceUniqueName));
- if (ret != 0) {
- return -1;
- }
-
- RotateCapturedFrame orientation;
- ret = vieData.capture->GetOrientation(deviceUniqueName, orientation);
- __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "GetOrientation ret %d orientation %d", ret,
- orientation);
-
- return (jint) orientation;
-
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: SetRotation
-// Signature: (II)I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetRotation(
- JNIEnv *,
- jobject,
- jint captureId,
- jint degrees)
-{
-
- if (NULL == vieData.capture)
- return -1;
- RotateCapturedFrame rotation = RotateCapturedFrame_0;
- if (degrees == 90)
- rotation = RotateCapturedFrame_90;
- else if (degrees == 180)
- rotation = RotateCapturedFrame_180;
- else if (degrees == 270)
- rotation = RotateCapturedFrame_270;
-
- int ret = vieData.capture->SetRotateCapturedFrames(captureId, rotation);
- return ret;
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: EnableNACK
-// Signature: (IZ)I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_EnableNACK(
- JNIEnv *,
- jobject,
- jint channel,
- jboolean enable)
-{
- if (NULL == vieData.rtp)
- return -1;
-
- if (enable)
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "EnableNACK enable");
- else
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "EnableNACK disable");
-
- int ret = vieData.rtp->SetNACKStatus(channel, enable);
- return ret;
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: EnablePLI
-// Signature: (IZ)I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_EnablePLI(
- JNIEnv *,
- jobject,
- jint channel,
- jboolean enable)
-{
- if (NULL == vieData.rtp)
- return -1;
-
- if (enable)
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "EnablePLI enable");
- else
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "EnablePLI disable");
-
- int ret = vieData.rtp->SetKeyFrameRequestMethod(channel,
- kViEKeyFrameRequestPliRtcp);
- return ret;
-}
-
-// Class: org_webrtc_videoengineapp_ViEAndroidJavaAPI
-// Method: SetCallback
-// Signature: (ILorg/webrtc/videoengineapp/IViEAndroidCallback;)I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_SetCallback(
- JNIEnv * env,
- jobject,
- jint channel,
- jobject callback)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "SetCallback");
-
- if (NULL == vieData.codec)
- return -1;
- if (vieData.callback == NULL) {
- vieData.callback = new VideoCallbackAndroid(vieData, env, callback);
- }
- else if (vieData.codec) {
- vieData.codec->DeregisterDecoderObserver(channel); // Wrong channel?
- vieData.codec->DeregisterEncoderObserver(channel);
- }
-
- vieData.codec->RegisterDecoderObserver(channel, *vieData.callback);
- vieData.codec->RegisterEncoderObserver(channel, *vieData.callback);
-
- return 0;
-}
-
-//
-// VoiceEngine API wrapper functions
-//
-
-// Create VoiceEngine instance
-JNIEXPORT jboolean JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_Create(
- JNIEnv *env,
- jobject,
- jobject context)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "Create");
-
- // Check if already created
- if (veData.ve) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "VoE already created");
- return false;
- }
-
- // Init Android Object
- VoiceEngine::SetAndroidObjects(veData.jvm, env, context);
- // Create
- veData.ve = VoiceEngine::Create();
- if (!veData.ve) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Create VoE failed");
- return false;
- }
-
- // Get sub-APIs
- if (!VE_GetSubApis()) {
- // If not OK, release all sub-APIs and delete VoE
- VE_ReleaseSubApis();
- if (!VoiceEngine::Delete(veData.ve)) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Delete VoE failed");
- }
- return false;
- }
-
- return true;
-}
-
-// Delete VoiceEngine instance
-JNIEXPORT jboolean JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_Delete(
- JNIEnv *,
- jobject)
-{
- // Check if exists
- if (!veData.ve) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "VoE does not exist");
- return false;
- }
-
- // Release sub-APIs
- VE_ReleaseSubApis();
-
- // Delete
- if (!VoiceEngine::Delete(veData.ve)) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Delete VoE failed");
- return false;
- }
-
- veData.ve = NULL;
-
- // Clear instance independent Java objects
- VoiceEngine::SetAndroidObjects(NULL, NULL, NULL);
-
- return true;
-}
-
-// [Base] Initialize VoiceEngine
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_Init(
- JNIEnv *,
- jobject,
- jboolean enableTrace,
- jboolean useExtTrans)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "VE_Init");
-
- VALIDATE_BASE_POINTER;
-
- if (useExtTrans) {
- // Not implemented
- return -1;
- }
-
- return veData.base->Init();
-}
-
-// [Base] Terminate VoiceEngine
-JNIEXPORT jint
-JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_Terminate(
- JNIEnv *,
- jobject)
-{
- VALIDATE_BASE_POINTER;
-
- jint retVal = veData.base->Terminate();
- return retVal;
-}
-
-// [Base] Create channel
-JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_CreateChannel(
- JNIEnv *,
- jobject)
-{
- VALIDATE_BASE_POINTER;
-
- webrtc::CodecInst voiceCodec;
- int numOfVeCodecs = veData.codec->NumOfCodecs();
-
- //enum all the supported codec
- __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "Supported Voice Codec:\n");
- for (int i = 0; i < numOfVeCodecs; ++i) {
- if (veData.codec->GetCodec(i, voiceCodec) != -1) {
- __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "num: %d name: %s\n", i, voiceCodec.plname);
- }
- }
-
- jint channel = veData.base->CreateChannel();
-
- return channel;
-}
-
-// [Base] Delete channel
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_DeleteChannel(
- JNIEnv *,
- jobject,
- jint channel)
-{
- VALIDATE_BASE_POINTER;
- return veData.base->DeleteChannel(channel);
-}
-
-// [Base] SetLocalReceiver
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetLocalReceiver(
- JNIEnv *,
- jobject,
- jint channel,
- jint port)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "SetLocalReceiver");
- VALIDATE_BASE_POINTER;
- return veData.base->SetLocalReceiver(channel, port);
-}
-
-// [Base] SetSendDestination
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetSendDestination(
- JNIEnv *env,
- jobject,
- jint channel,
- jint port,
- jstring ipaddr)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "SetSendDestination");
- VALIDATE_BASE_POINTER;
-
- const char* ipaddrNative = env->GetStringUTFChars(ipaddr, NULL);
- if (!ipaddrNative) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Could not get UTF string");
- return -1;
- }
- jint retVal = veData.base->SetSendDestination(channel, port, ipaddrNative);
- env->ReleaseStringUTFChars(ipaddr, ipaddrNative);
- return retVal;
-}
-
-// [Base] StartListen
-JNIEXPORT jint JNICALL Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartListen(
- JNIEnv *,
- jobject,
- jint channel)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StartListen");
- VALIDATE_BASE_POINTER;
- return veData.base->StartReceive(channel);
-}
-
-// [Base] Start playout
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartPlayout(
- JNIEnv *,
- jobject,
- jint channel)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StartPlayout");
- VALIDATE_BASE_POINTER;
- return veData.base->StartPlayout(channel);
-}
-
-// [Base] Start send
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartSend(
- JNIEnv *,
- jobject,
- jint channel)
-{
- __android_log_write(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "StartSend");
- VALIDATE_BASE_POINTER;
- return veData.base->StartSend(channel);
-}
-
-// [Base] Stop listen
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopListen(
- JNIEnv *,
- jobject,
- jint channel)
-{
- VALIDATE_BASE_POINTER;
- return veData.base->StartReceive(channel);
-}
-
-// [Base] Stop playout
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopPlayout(
- JNIEnv *,
- jobject,
- jint channel)
-{
- VALIDATE_BASE_POINTER;
- return veData.base->StopPlayout(channel);
-}
-
-// [Base] Stop send
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopSend(
- JNIEnv *,
- jobject,
- jint channel)
-{
- VALIDATE_BASE_POINTER;
- return veData.base->StopSend(channel);
-}
-
-// [codec] Number of codecs
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_NumOfCodecs(
- JNIEnv *,
- jobject)
-{
- VALIDATE_CODEC_POINTER;
- return veData.codec->NumOfCodecs();
-}
-
-// [codec] Set send codec
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetSendCodec(
- JNIEnv *,
- jobject,
- jint channel,
- jint index)
-{
- VALIDATE_CODEC_POINTER;
-
- webrtc::CodecInst codec;
-
- for (int i = 0; i < veData.codec->NumOfCodecs(); ++i) {
- webrtc::CodecInst codecToList;
- veData.codec->GetCodec(i, codecToList);
- __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "VE Codec list %s, pltype=%d\n",
- codecToList.plname, codecToList.pltype);
- }
-
- if (veData.codec->GetCodec(index, codec) != 0) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to get codec");
- return -1;
- }
- __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG, "SetSendCodec %s\n",
- codec.plname);
-
- return veData.codec->SetSendCodec(channel, codec);
-}
-
-// [audioprocessing] SetNSStatus
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetNSStatus(
- JNIEnv *,
- jobject,
- jboolean enable,
- jint mode)
-{
- //TODO implement
- return -1;
-}
-
-// [audioprocessing] SetAGCStatus
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetAGCStatus(
- JNIEnv *,
- jobject,
- jboolean enable,
- jint mode)
-{
- //TODO implement
- return -1;
-}
-
-// [audioprocessing] SetECStatus
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetECStatus(
- JNIEnv *,
- jobject,
- jboolean enable,
- jint mode,
- jint AESmode,
- jint AESattenuation)
-{
- //TODO implement
- return -1;
-}
-
-// [File] Start play file locally
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartPlayingFileLocally(
- JNIEnv * env,
- jobject,
- jint channel,
- jstring fileName,
- jboolean loop)
-{
- VALIDATE_FILE_POINTER;
-
- const char* fileNameNative = env->GetStringUTFChars(fileName, NULL);
- if (!fileNameNative) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Could not get UTF string");
- return -1;
- }
-
- jint retVal = veData.file->StartPlayingFileLocally(channel,
- fileNameNative,
- loop);
-
- env->ReleaseStringUTFChars(fileName, fileNameNative);
-
- return retVal;
-}
-
-// [File] Stop play file locally
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopPlayingFileLocally(
- JNIEnv *,
- jobject,
- jint channel)
-{
- VALIDATE_FILE_POINTER;
- return veData.file->StopPlayingFileLocally(channel);
-}
-
-// [File] Start playing file as microphone
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StartPlayingFileAsMicrophone(
- JNIEnv *env,
- jobject,
- jint channel,
- jstring fileName,
- jboolean loop)
-{
- VALIDATE_FILE_POINTER;
-
- const char* fileNameNative = env->GetStringUTFChars(fileName, NULL);
- if (!fileNameNative) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Could not get UTF string");
- return -1;
- }
-
- jint retVal = veData.file->StartPlayingFileAsMicrophone(channel,
- fileNameNative,
- loop);
-
- env->ReleaseStringUTFChars(fileName, fileNameNative);
-
- return retVal;
-}
-
-// [File] Stop playing file as microphone
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_StopPlayingFileAsMicrophone(
- JNIEnv *,
- jobject,
- jint channel)
-{
- VALIDATE_FILE_POINTER;
- return veData.file->StopPlayingFileAsMicrophone(channel);
-}
-
-// [Volume] Set speaker volume
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetSpeakerVolume(
- JNIEnv *,
- jobject,
- jint level)
-{
- VALIDATE_VOLUME_POINTER;
-
- if (veData.volume->SetSpeakerVolume(level) != 0) {
- return -1;
- }
-
- unsigned int storedVolume = 0;
- if (veData.volume->GetSpeakerVolume(storedVolume) != 0) {
- return -1;
- }
-
- if (storedVolume != level) {
- return -1;
- }
-
- return 0;
-}
-
-// [Hardware] Set speaker volume
-JNIEXPORT jint JNICALL
-Java_org_webrtc_videoengineapp_ViEAndroidJavaAPI_VE_SetLoudspeakerStatus(
- JNIEnv *,
- jobject,
- jboolean enable)
-{
- VALIDATE_HARDWARE_POINTER;
-
- if (veData.hardware->SetLoudspeakerStatus(enable) != 0) {
- return -1;
- }
-
- return 0;
-}
-
-//
-// local function
-//
-
-// Get all sub-APIs
-bool VE_GetSubApis() {
- bool getOK = true;
-
- // Base
- veData.base = VoEBase::GetInterface(veData.ve);
- if (!veData.base) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get base sub-API failed");
- getOK = false;
- }
-
- // Codec
- veData.codec = VoECodec::GetInterface(veData.ve);
- if (!veData.codec) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get codec sub-API failed");
- getOK = false;
- }
-
- // File
- veData.file = VoEFile::GetInterface(veData.ve);
- if (!veData.file) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get file sub-API failed");
- getOK = false;
- }
-
- // Network
- veData.netw = VoENetwork::GetInterface(veData.ve);
- if (!veData.netw) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get network sub-API failed");
- getOK = false;
- }
-
- // audioprocessing
- veData.apm = VoEAudioProcessing::GetInterface(veData.ve);
- if (!veData.apm) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get VoEAudioProcessing sub-API failed");
- getOK = false;
- }
-
- // Volume
- veData.volume = VoEVolumeControl::GetInterface(veData.ve);
- if (!veData.volume) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get volume sub-API failed");
- getOK = false;
- }
-
- // Hardware
- veData.hardware = VoEHardware::GetInterface(veData.ve);
- if (!veData.hardware) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get hardware sub-API failed");
- getOK = false;
- }
-
- return getOK;
-}
-
-// Release all sub-APIs
-bool VE_ReleaseSubApis() {
- bool releaseOK = true;
-
- // Base
- if (veData.base) {
- if (0 != veData.base->Release()) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Release base sub-API failed");
- releaseOK = false;
- }
- else {
- veData.base = NULL;
- }
- }
-
- // Codec
- if (veData.codec) {
- if (0 != veData.codec->Release()) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Release codec sub-API failed");
- releaseOK = false;
- }
- else {
- veData.codec = NULL;
- }
- }
-
- // File
- if (veData.file) {
- if (0 != veData.file->Release()) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Release file sub-API failed");
- releaseOK = false;
- }
- else {
- veData.file = NULL;
- }
- }
-
- // Network
- if (veData.netw) {
- if (0 != veData.netw->Release()) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Release network sub-API failed");
- releaseOK = false;
- }
- else {
- veData.netw = NULL;
- }
- }
-
- // apm
- if (veData.apm) {
- if (0 != veData.apm->Release()) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Release apm sub-API failed");
- releaseOK = false;
- }
- else {
- veData.apm = NULL;
- }
- }
-
- // Volume
- if (veData.volume) {
- if (0 != veData.volume->Release()) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Release volume sub-API failed");
- releaseOK = false;
- }
- else {
- veData.volume = NULL;
- }
- }
-
- // Hardware
- if (veData.hardware) {
- if (0 != veData.hardware->Release()) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Release hardware sub-API failed");
- releaseOK = false;
- }
- else {
- veData.hardware = NULL;
- }
- }
-
- return releaseOK;
-}
diff --git a/trunk/src/video_engine/main/test/android_test/res/drawable/logo.png b/trunk/src/video_engine/main/test/android_test/res/drawable/logo.png
deleted file mode 100644
index a07c69f..0000000
--- a/trunk/src/video_engine/main/test/android_test/res/drawable/logo.png
+++ /dev/null
Binary files differ
diff --git a/trunk/src/video_engine/main/test/android_test/res/layout/aconfig.xml b/trunk/src/video_engine/main/test/android_test/res/layout/aconfig.xml
deleted file mode 100644
index e88b7c6..0000000
--- a/trunk/src/video_engine/main/test/android_test/res/layout/aconfig.xml
+++ /dev/null
@@ -1,81 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<LinearLayout android:layout_width="fill_parent"
- android:layout_height="fill_parent" android:orientation="vertical"
- xmlns:android="http://schemas.android.com/apk/res/android">
- <TextView android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:textStyle="bold"
- android:textSize="24dip"
- android:text="Audio Settings"></TextView>
- <TextView android:id="@+id/TextView03"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </TextView>
- <TextView android:id="@+id/TextView01"
- android:layout_height="wrap_content"
- android:layout_gravity="bottom"
- android:layout_width="wrap_content"
- android:text="@string/codecType">
- </TextView>
- <Spinner android:layout_height="wrap_content"
- android:layout_width="fill_parent"
- android:id="@+id/spVoiceCodecType">
- </Spinner>
-
- <LinearLayout android:id="@+id/LinearLayout02"
- android:layout_height="wrap_content"
- android:layout_width="fill_parent">
- <TextView android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:text="@string/aTxPort">
- </TextView>
- <EditText android:layout_height="wrap_content"
- android:layout_width="wrap_content"
- android:id="@+id/etATxPort">
- </EditText>
- <TextView android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:text="@string/aRxPort">
- </TextView>
- <EditText android:layout_height="wrap_content"
- android:layout_width="wrap_content"
- android:id="@+id/etARxPort" >
- </EditText>
- </LinearLayout>
-
- <LinearLayout android:id="@+id/LinearLayout02"
- android:layout_height="wrap_content"
- android:layout_width="fill_parent">
- <CheckBox android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:id="@+id/cbSpeaker"
- android:text="@string/speaker">
- </CheckBox>
- <CheckBox android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:id="@+id/cbAutoGainControl"
- android:text="@string/AutoGainControl">
- </CheckBox>
- </LinearLayout>
-
- <LinearLayout android:id="@+id/LinearLayout02"
- android:layout_height="wrap_content"
- android:layout_width="fill_parent">
- <CheckBox android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:id="@+id/cbAECM"
- android:text="@string/AECM">
- </CheckBox>
- </LinearLayout>
-
- <LinearLayout android:id="@+id/LinearLayout03"
- android:layout_height="wrap_content"
- android:layout_width="fill_parent">
- <CheckBox android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:id="@+id/cbNoiseSuppression"
- android:text="@string/NoiseSuppression">
- </CheckBox>
- </LinearLayout>
-
-</LinearLayout>
diff --git a/trunk/src/video_engine/main/test/android_test/res/layout/both.xml b/trunk/src/video_engine/main/test/android_test/res/layout/both.xml
deleted file mode 100644
index d29d906..0000000
--- a/trunk/src/video_engine/main/test/android_test/res/layout/both.xml
+++ /dev/null
@@ -1,43 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<LinearLayout
- xmlns:android="http://schemas.android.com/apk/res/android"
- android:orientation="horizontal"
- android:layout_width="fill_parent"
- android:layout_height="fill_parent">
- <LinearLayout
- android:orientation="vertical"
- android:layout_width="120dip"
- android:layout_height="fill_parent">
- <LinearLayout android:id="@+id/llLocalView"
- android:layout_width="fill_parent"
- android:layout_height="80dip">
- </LinearLayout>
- <TextView
- android:layout_width="fill_parent"
- android:layout_height="fill_parent"
- android:layout_weight="1"
- android:text="" />
- <Button android:id="@+id/btSwitchCamera"
- android:layout_width="fill_parent"
- android:layout_height="wrap_content"
- android:text="@string/frontCamera"
- android:layout_gravity="bottom"/>
- <Button android:id="@+id/btStartStopCall"
- android:layout_width="fill_parent"
- android:layout_height="wrap_content"
- android:text="@string/startCall"
- android:layout_gravity="bottom"/>
- <Button android:id="@+id/btExit"
- android:layout_width="fill_parent"
- android:layout_height="wrap_content"
- android:layout_gravity="bottom"
- android:text="@string/exit"/>
- </LinearLayout>
- <LinearLayout
- android:id="@+id/llRemoteView"
- android:layout_width="fill_parent"
- android:layout_height="fill_parent"
- android:layout_weight="1">
- </LinearLayout>
- </LinearLayout >
-
diff --git a/trunk/src/video_engine/main/test/android_test/res/layout/main.xml b/trunk/src/video_engine/main/test/android_test/res/layout/main.xml
deleted file mode 100644
index 3b16cb7..0000000
--- a/trunk/src/video_engine/main/test/android_test/res/layout/main.xml
+++ /dev/null
@@ -1,54 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<LinearLayout android:layout_width="fill_parent"
- android:layout_height="fill_parent" android:layout_gravity="right"
- android:orientation="vertical"
- xmlns:android="http://schemas.android.com/apk/res/android">
-
- <TextView android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:id="@+id/tvTitle"
- android:textStyle="bold"
- android:textSize="24dip"
- android:text="@string/gSettings">
- </TextView>
- <TextView android:id="@+id/TextView03"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </TextView>
- <LinearLayout android:id="@+id/LinearLayout02"
- android:layout_height="wrap_content"
- android:layout_width="fill_parent">
- <CheckBox android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:id="@+id/cbVideoReceive"
- android:text="@string/enableVideoReceive">
- </CheckBox>
- <CheckBox android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:id="@+id/cbVideoSend"
- android:text="@string/enableVideoSend">
- </CheckBox>
- </LinearLayout>
-
- <CheckBox android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:id="@+id/cbVoice"
- android:text="@string/enableVoice">
- </CheckBox>
-
- <TextView android:id="@+id/TextView02"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:text="@string/remoteIp">
- </TextView>
- <EditText android:layout_height="wrap_content"
- android:layout_width="fill_parent"
- android:id="@+id/etRemoteIp" >
- </EditText>
-
- <CheckBox android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:id="@+id/cbLoopback"
- android:text="@string/loopback">
- </CheckBox>
-</LinearLayout>
diff --git a/trunk/src/video_engine/main/test/android_test/res/layout/send.xml b/trunk/src/video_engine/main/test/android_test/res/layout/send.xml
deleted file mode 100644
index ee230f5..0000000
--- a/trunk/src/video_engine/main/test/android_test/res/layout/send.xml
+++ /dev/null
@@ -1,17 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android"
- android:orientation="vertical"
- android:layout_width="fill_parent"
- android:layout_height="fill_parent">
-
-
- <SurfaceView android:id="@+id/svLocal"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </SurfaceView>
- <ImageView android:id="@+id/ivPreview"
- android:layout_height="fill_parent"
- android:layout_width="fill_parent">
- </ImageView>
-
-</LinearLayout>
diff --git a/trunk/src/video_engine/main/test/android_test/res/layout/stats.xml b/trunk/src/video_engine/main/test/android_test/res/layout/stats.xml
deleted file mode 100644
index 1612ec3..0000000
--- a/trunk/src/video_engine/main/test/android_test/res/layout/stats.xml
+++ /dev/null
@@ -1,59 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<LinearLayout android:layout_width="fill_parent"
- android:layout_height="fill_parent" android:orientation="vertical"
- xmlns:android="http://schemas.android.com/apk/res/android">
- <TextView android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:textStyle="bold"
- android:textSize="24dip"
- android:text="Stats">
- </TextView>
-
- <TextView android:id="@+id/TextView03"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </TextView>
-
- <TextView android:layout_height="wrap_content"
- android:layout_width="fill_parent"
- android:id="@+id/tvLocalIp"
- android:textStyle="bold">
- </TextView>
-
- <TextView android:id="@+id/TextView03"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </TextView>
-
- <TextView android:layout_height="wrap_content"
- android:layout_width="fill_parent"
- android:id="@+id/tvFrameRateI"
- android:textStyle="bold">
- </TextView>
- <TextView android:layout_height="wrap_content"
- android:layout_width="fill_parent"
- android:id="@+id/tvBitRateI"
- android:textStyle="bold">
- </TextView>
- <TextView android:layout_height="wrap_content"
- android:layout_width="fill_parent"
- android:id="@+id/tvPacketLoss"
- android:textStyle="bold">
- </TextView>
-
- <TextView android:id="@+id/TextView03"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </TextView>
-
- <TextView android:layout_height="wrap_content"
- android:layout_width="fill_parent"
- android:id="@+id/tvFrameRateO"
- android:textStyle="bold">
- </TextView>
- <TextView android:layout_height="wrap_content"
- android:layout_width="fill_parent"
- android:id="@+id/tvBitRateO"
- android:textStyle="bold">
- </TextView>
-</LinearLayout>
diff --git a/trunk/src/video_engine/main/test/android_test/res/layout/tabhost.xml b/trunk/src/video_engine/main/test/android_test/res/layout/tabhost.xml
deleted file mode 100644
index 9997113..0000000
--- a/trunk/src/video_engine/main/test/android_test/res/layout/tabhost.xml
+++ /dev/null
@@ -1,25 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<TabHost xmlns:android="http://schemas.android.com/apk/res/android"
- android:id="@android:id/tabhost"
- android:layout_width="fill_parent"
- android:layout_height="fill_parent">
- <LinearLayout
- android:orientation="vertical"
- android:layout_width="fill_parent"
- android:layout_height="fill_parent">
- <TabWidget
- android:id="@android:id/tabs"
- android:layout_width="fill_parent"
- android:layout_height="wrap_content" />
- <FrameLayout
- android:id="@android:id/tabcontent"
- android:layout_width="fill_parent"
- android:layout_height="fill_parent">
- <include android:id="@+id/tab_video" layout="@layout/both" />
- <include android:id="@+id/tab_config" layout="@layout/main" />
- <include android:id="@+id/tab_vconfig" layout="@layout/vconfig" />
- <include android:id="@+id/tab_aconfig" layout="@layout/aconfig" />
- <include android:id="@+id/tab_stats" layout="@layout/stats" />
- </FrameLayout>
- </LinearLayout>
-</TabHost>
diff --git a/trunk/src/video_engine/main/test/android_test/res/layout/vconfig.xml b/trunk/src/video_engine/main/test/android_test/res/layout/vconfig.xml
deleted file mode 100644
index 5270c10..0000000
--- a/trunk/src/video_engine/main/test/android_test/res/layout/vconfig.xml
+++ /dev/null
@@ -1,67 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<LinearLayout android:layout_width="fill_parent"
- android:layout_height="fill_parent" android:orientation="vertical"
- xmlns:android="http://schemas.android.com/apk/res/android">
- <TextView android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:textStyle="bold"
- android:textSize="24dip"
- android:text="@string/vSettings">
- </TextView>
-
- <TextView android:id="@+id/TextView03"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </TextView>
-
- <TextView android:id="@+id/TextView01"
- android:layout_height="wrap_content"
- android:layout_width="wrap_content"
- android:text="@string/codecType">
- </TextView>
- <Spinner android:layout_height="wrap_content"
- android:layout_width="fill_parent"
- android:id="@+id/spCodecType">
- </Spinner>
-
- <TextView android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:id="@+id/tvCodecSize"
- android:text="@string/codecSize">
- </TextView>
- <Spinner android:layout_height="wrap_content"
- android:layout_width="fill_parent"
- android:id="@+id/spCodecSize">
- </Spinner>
-
- <LinearLayout android:id="@+id/LinearLayout02"
- android:layout_height="wrap_content"
- android:layout_width="fill_parent">
- <TextView android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:text="@string/vTxPort">
- </TextView>
- <EditText android:layout_height="wrap_content"
- android:layout_width="wrap_content"
- android:id="@+id/etVTxPort" >
- </EditText>
- <TextView android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:text="@string/vRxPort">
- </TextView>
- <EditText android:layout_height="wrap_content"
- android:layout_width="wrap_content"
- android:id="@+id/etVRxPort" >
- </EditText>
- </LinearLayout>
-
- <LinearLayout android:id="@+id/LinearLayout03"
- android:layout_height="wrap_content"
- android:layout_width="fill_parent">
- <CheckBox android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:id="@+id/cbNack"
- android:text="@string/nack">
- </CheckBox>
- </LinearLayout>
-</LinearLayout>
diff --git a/trunk/src/video_engine/main/test/android_test/res/values/arrays.xml b/trunk/src/video_engine/main/test/android_test/res/values/arrays.xml
deleted file mode 100644
index 77dd017..0000000
--- a/trunk/src/video_engine/main/test/android_test/res/values/arrays.xml
+++ /dev/null
@@ -1,22 +0,0 @@
-<resources>
-
- <string-array name="codectype">
- <item>vp8</item>
- </string-array>
-
-<string-array name="codecSize"><item>176x144</item><item>320x240</item>
-<item>352x288</item><item>640x480</item>
-
-</string-array>
-
-<string-array name="voiceCodecType">
-<item>iPCM-wb</item>
-<item>iSAC</item>
-<item>iSAC-LC</item>
-<item>EG711U</item>
-<item>EG711A</item>
-<item>PCMU</item>
-<item>PCMA</item>
-<item>iLBC</item>
-</string-array>
-</resources>
diff --git a/trunk/src/video_engine/main/test/android_test/res/values/strings.xml b/trunk/src/video_engine/main/test/android_test/res/values/strings.xml
deleted file mode 100644
index 6597bec..0000000
--- a/trunk/src/video_engine/main/test/android_test/res/values/strings.xml
+++ /dev/null
@@ -1,36 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<resources>
- <string name="app_name">ViEDemo</string>
- <string name="error">Error</string>
- <string name="errorCamera">Camera Error</string>
-<string name="codectype_prompt">Choose a codec type</string>
-<string name="demoTitle">Video Engine Android Demo</string>
-<string name="codecType">Codec Type</string>
-<string name="codecSize">Codec Size</string>
-<string name="remoteIp">Remote IP address</string>
-<string name="loopback">Loopback</string>
-<string name="startListen">Start Listen</string>
-<string name="startSend">Start Send</string>
-<string name="startBoth">Start Both</string>
-<string name="enableVoice">Voice</string>
-<string name="enableVideoReceive">Video Receive</string>
-<string name="enableVideoSend">Video Send</string>
-<string name="gSettings">Global Settings</string>
-<string name="vSettings">Video Settings</string>
-<string name="vTxPort">Video Tx Port</string>
-<string name="vRxPort">Video Rx Port</string>
-<string name="aTxPort">Audio Tx Port</string>
-<string name="aRxPort">Audio Rx Port</string>
-<string name="AutoGainControl">Auto Gain Control</string>
-<string name="VoiceActivityDetection">Voice Activity Detection</string>
-<string name="AECM">Acoustic Echo Control</string>
-<string name="NoiseSuppression">Noise Suppression</string>
-<string name="nack">NACK</string>
-<string name="frontCamera">SwitchToFront</string>
-<string name="backCamera">SwitchToBack</string>
-<string name="startCall">StartCall</string>
-<string name="stopCall">StopCall</string>
-<string name="exit">Exit</string>
-<string name="speaker">Speaker</string>
-<string name="h263DecSw">H263 SW Decoder</string>
-</resources>
diff --git a/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/IViEAndroidCallback.java b/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/IViEAndroidCallback.java
deleted file mode 100644
index 7a00d14..0000000
--- a/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/IViEAndroidCallback.java
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-package org.webrtc.videoengineapp;
-
-public interface IViEAndroidCallback {
- public int UpdateStats(int frameRateI, int bitRateI,
- int packetLoss, int frameRateO,
- int bitRateO);
-}
diff --git a/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/ViEAndroidDemo.java b/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/ViEAndroidDemo.java
deleted file mode 100644
index 9301948..0000000
--- a/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/ViEAndroidDemo.java
+++ /dev/null
@@ -1,803 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-package org.webrtc.videoengineapp;
-
-import java.net.InetAddress;
-import java.net.NetworkInterface;
-import java.net.SocketException;
-import java.util.Enumeration;
-
-import org.webrtc.videoengine.ViERenderer;
-
-
-import android.app.TabActivity;
-import android.content.Context;
-import android.content.res.Configuration;
-import android.hardware.SensorManager;
-import android.media.AudioManager;
-import android.os.Bundle;
-import android.os.PowerManager;
-import android.os.PowerManager.WakeLock;
-
-import android.util.Log;
-import android.view.KeyEvent;
-import android.view.Surface;
-import android.view.SurfaceView;
-import android.view.View;
-import android.view.Display;
-import android.view.Window;
-import android.view.WindowManager;
-
-import android.widget.AdapterView;
-import android.widget.ArrayAdapter;
-import android.widget.Button;
-import android.widget.CheckBox;
-
-import android.widget.EditText;
-import android.widget.LinearLayout;
-import android.widget.Spinner;
-import android.widget.TabHost;
-import android.widget.TextView;
-import android.widget.AdapterView.OnItemSelectedListener;
-import android.widget.TabHost.TabSpec;
-import android.view.OrientationEventListener;
-
-public class ViEAndroidDemo extends TabActivity implements IViEAndroidCallback,
- View.OnClickListener,
- OnItemSelectedListener {
- private ViEAndroidJavaAPI ViEAndroidAPI = null;
-
- // remote renderer
- private SurfaceView remoteSurfaceView = null;
-
- // local renderer and camera
- private SurfaceView svLocal = null;
-
- // channel number
- private int channel;
- private int cameraId;
- private int voiceChannel = -1;
-
- // flags
- private boolean viERunning = false;
- private boolean voERunning = false;
-
- // debug
- private boolean enableTrace = false;
-
- // Constant
- private static final String LOG_TAG = "*WEBRTCJ*";
- private static final int RECEIVE_CODEC_FRAMERATE = 30;
- private static final int SEND_CODEC_FRAMERATE = 15;
- private static final int INIT_BITRATE = 400;
-
- private static final int EXPIRARY_YEAR = 2010;
- private static final int EXPIRARY_MONTH = 10;
- private static final int EXPIRARY_DAY = 22;
-
- private int volumeLevel = 204;
-
- private TabHost mTabHost = null;
-
- private TabSpec mTabSpecConfig;
- private TabSpec mTabSpecVideo;
-
- private LinearLayout mLlRemoteSurface = null;
- private LinearLayout mLlLocalSurface = null;
-
- private Button btStartStopCall;
- private Button btSwitchCamera;
-
- //Global Settings
- private CheckBox cbVideoSend;
- private boolean enableVideoSend = true;
- private CheckBox cbVideoReceive;
- private boolean enableVideoReceive = true;
- private boolean enableVideo = true;
- private CheckBox cbVoice;
- private boolean enableVoice = false;
- private EditText etRemoteIp;
- private String remoteIp = "10.1.100.68";
- private CheckBox cbLoopback;
- private boolean loopbackMode = true;
-
- //Video settings
- private Spinner spCodecType;
- private int codecType = 0;
- private Spinner spCodecSize;
- private int codecSizeWidth = 352;
- private int codecSizeHeight = 288;
- private TextView etVRxPort;
- private int receivePortVideo = 11111;
- private TextView etVTxPort;
- private int destinationPortVideo = 11111;
- private CheckBox cbEnableNack;
- private boolean enableNack = false;
-
- //Audio settings
- private Spinner spVoiceCodecType;
- private int voiceCodecType = 5; //PCMU = 5
- private TextView etARxPort;
- private int receivePortVoice = 11113;
- private TextView etATxPort;
- private int destinationPortVoice = 11113;
- private CheckBox cbEnableSpeaker;
- private boolean enableSpeaker = false;
- private CheckBox cbEnableAGC;
- private boolean enableAGC = false;
- private CheckBox cbEnableAECM;
- private boolean enableAECM = false;
- private CheckBox cbEnableNS;
- private boolean enableNS = false;
-
- //Stats
- private TextView tvFrameRateI;
- private TextView tvBitRateI;
- private TextView tvPacketLoss;
- private TextView tvFrameRateO;
- private TextView tvBitRateO;
- private int frameRateI;
- private int bitRateI;
- private int packetLoss;
- private int frameRateO;
- private int bitRateO;
-
- private WakeLock wakeLock;
-
- private boolean usingFrontCamera = false;
-
- private OrientationEventListener orientationListener;
- int currentOrientation = OrientationEventListener.ORIENTATION_UNKNOWN;
- int currentCameraOrientation = 0;
-
- //Convert current display orientation to how much the camera should be rotated.
- public int GetCameraOrientation(int cameraOrientation) {
- Display display = this.getWindowManager().getDefaultDisplay();
- int displatyRotation = display.getRotation();
- int degrees = 0;
- switch (displatyRotation) {
- case Surface.ROTATION_0: degrees = 0; break;
- case Surface.ROTATION_90: degrees = 90; break;
- case Surface.ROTATION_180: degrees = 180; break;
- case Surface.ROTATION_270: degrees = 270; break;
- }
- int result=0;
- if(cameraOrientation>180) {
- result=(cameraOrientation + degrees) % 360;
- }
- else {
- result=(cameraOrientation - degrees+360) % 360;
- }
-
- return result;
- }
-
- public void onConfigurationChanged(Configuration newConfig) {
- super.onConfigurationChanged(newConfig);
- int newRotation = GetCameraOrientation(currentCameraOrientation);
- if (viERunning){
- ViEAndroidAPI.SetRotation(cameraId,newRotation);
- }
- }
-
- // Called when the activity is first created.
- @Override
- public void onCreate(Bundle savedInstanceState) {
- super.onCreate(savedInstanceState);
- requestWindowFeature(Window.FEATURE_NO_TITLE);
- getWindow().setFlags(WindowManager.LayoutParams.FLAG_FULLSCREEN,
- WindowManager.LayoutParams.FLAG_FULLSCREEN);
-
- PowerManager pm = (PowerManager)this.getSystemService(
- Context.POWER_SERVICE);
- wakeLock = pm.newWakeLock(
- PowerManager.SCREEN_DIM_WAKE_LOCK, LOG_TAG);
-
- setContentView(R.layout.tabhost);
- mTabHost = getTabHost();
-
- //Video tab
- mTabSpecVideo = mTabHost.newTabSpec("tab_video");
- mTabSpecVideo.setIndicator("Video");
- mTabSpecVideo.setContent(R.id.tab_video);
- mTabHost.addTab(mTabSpecVideo);
-
- //Shared config tab
- mTabHost = getTabHost();
- mTabSpecConfig = mTabHost.newTabSpec("tab_config");
- mTabSpecConfig.setIndicator("Config");
- mTabSpecConfig.setContent(R.id.tab_config);
- mTabHost.addTab(mTabSpecConfig);
-
- TabSpec mTabv;
- mTabv = mTabHost.newTabSpec("tab_vconfig");
- mTabv.setIndicator("V. Config");
- mTabv.setContent(R.id.tab_vconfig);
- mTabHost.addTab(mTabv);
- TabSpec mTaba;
- mTaba = mTabHost.newTabSpec("tab_aconfig");
- mTaba.setIndicator("A. Config");
- mTaba.setContent(R.id.tab_aconfig);
- mTabHost.addTab(mTaba);
- TabSpec mTabs;
- mTabs = mTabHost.newTabSpec("tab_stats");
- mTabs.setIndicator("Stats");
- mTabs.setContent(R.id.tab_stats);
- mTabHost.addTab(mTabs);
-
- int childCount = mTabHost.getTabWidget().getChildCount();
- for (int i=0; i<childCount; i++)
- mTabHost.getTabWidget().getChildAt(i).getLayoutParams().height = 50;
-
- orientationListener =
- new OrientationEventListener(this,SensorManager.SENSOR_DELAY_UI) {
- public void onOrientationChanged (int orientation) {
- if (orientation != ORIENTATION_UNKNOWN) {
- currentOrientation = orientation;
- }
- }
- };
- orientationListener.enable ();
-
- StartMain();
- return;
- }
-
- private String GetLocalIpAddress() {
- String localIPs = "";
- try {
- for (Enumeration<NetworkInterface> en = NetworkInterface
- .getNetworkInterfaces(); en.hasMoreElements();) {
- NetworkInterface intf = en.nextElement();
- for (Enumeration<InetAddress> enumIpAddr = intf
- .getInetAddresses(); enumIpAddr.hasMoreElements();) {
- InetAddress inetAddress = enumIpAddr.nextElement();
- if (!inetAddress.isLoopbackAddress()) {
- localIPs += inetAddress.getHostAddress().toString() + " ";
- //set the remote ip address the same as
- // the local ip address of the last netif
- remoteIp = inetAddress.getHostAddress().toString();
- }
- }
- }
- } catch (SocketException ex) {
- Log.e(LOG_TAG, ex.toString());
- }
- return localIPs;
- }
-
- @Override
- public boolean onKeyDown(int keyCode, KeyEvent event) {
- if (keyCode == KeyEvent.KEYCODE_BACK) {
- if (viERunning) {
- StopAll();
- StartMain();
- }
- finish();
- return true;
- }
- return super.onKeyDown(keyCode, event);
- }
-
- private void StopAll() {
- if (ViEAndroidAPI != null) {
- if (voERunning) {
- voERunning = false;
- StopVoiceEngine();
- }
-
- if (viERunning) {
- viERunning = false;
- ViEAndroidAPI.StopRender(channel);
- ViEAndroidAPI.StopReceive(channel);
- ViEAndroidAPI.StopSend(channel);
- ViEAndroidAPI.RemoveRemoteRenderer(channel);
- // stop the camera
- ViEAndroidAPI.StopCamera(cameraId);
- ViEAndroidAPI.Terminate();
- mLlRemoteSurface.removeView(remoteSurfaceView);
- mLlLocalSurface.removeView(svLocal);
- remoteSurfaceView = null;
-
- svLocal = null;
- }
- }
- }
-
- private void StartMain() {
- mTabHost.setCurrentTab(0);
-
- mLlRemoteSurface = (LinearLayout) findViewById(R.id.llRemoteView);
- mLlLocalSurface = (LinearLayout) findViewById(R.id.llLocalView);
-
- if (null == ViEAndroidAPI)
- ViEAndroidAPI = new ViEAndroidJavaAPI(this);
-
- //setContentView(R.layout.main);
- btSwitchCamera = (Button)findViewById(R.id.btSwitchCamera);
- btSwitchCamera.setOnClickListener(this);
- btStartStopCall = (Button)findViewById(R.id.btStartStopCall);
- btStartStopCall.setOnClickListener(this);
- findViewById(R.id.btExit).setOnClickListener(this);
-
- // cleaning
- remoteSurfaceView = null;
- svLocal = null;
-
- // init UI
- ArrayAdapter<?> adapter;
-
- int resource = android.R.layout.simple_spinner_item;
- int dropdownRes = android.R.layout.simple_spinner_dropdown_item;
-
- // video codec
- spCodecType = (Spinner) findViewById(R.id.spCodecType);
- adapter = ArrayAdapter.createFromResource(this,
- R.array.codectype,
- resource);
- adapter.setDropDownViewResource(dropdownRes);
- spCodecType.setAdapter(adapter);
- spCodecType.setSelection(codecType);
- spCodecType.setOnItemSelectedListener(this);
-
- // voice codec
- spVoiceCodecType = (Spinner) findViewById(R.id.spVoiceCodecType);
- adapter = ArrayAdapter.createFromResource(this, R.array.voiceCodecType,
- resource);
- adapter.setDropDownViewResource(dropdownRes);
- spVoiceCodecType.setAdapter(adapter);
- spVoiceCodecType.setSelection(voiceCodecType);
- spVoiceCodecType.setOnItemSelectedListener(this);
-
- spCodecSize = (Spinner) findViewById(R.id.spCodecSize);
- adapter = ArrayAdapter.createFromResource(this, R.array.codecSize,
- resource);
- adapter.setDropDownViewResource(dropdownRes);
- spCodecSize.setAdapter(adapter);
- spCodecSize.setOnItemSelectedListener(this);
-
- String ip = GetLocalIpAddress();
- TextView tvLocalIp = (TextView) findViewById(R.id.tvLocalIp);
- tvLocalIp.setText("Local IP address - " + ip);
-
- etRemoteIp = (EditText) findViewById(R.id.etRemoteIp);
- etRemoteIp.setText(remoteIp);
-
- cbLoopback = (CheckBox) findViewById(R.id.cbLoopback);
- cbLoopback.setChecked(loopbackMode);
-
- cbVoice = (CheckBox) findViewById(R.id.cbVoice);
- cbVoice.setChecked(enableVoice);
-
- cbVideoSend = (CheckBox) findViewById(R.id.cbVideoSend);
- cbVideoSend.setChecked(enableVideoSend);
- cbVideoReceive = (CheckBox) findViewById(R.id.cbVideoReceive);
- cbVideoReceive.setChecked(enableVideoReceive);
-
- etVTxPort = (EditText) findViewById(R.id.etVTxPort);
- etVTxPort.setText(Integer.toString(destinationPortVideo));
-
- etVRxPort = (EditText) findViewById(R.id.etVRxPort);
- etVRxPort.setText(Integer.toString(receivePortVideo));
-
- etATxPort = (EditText) findViewById(R.id.etATxPort);
- etATxPort.setText(Integer.toString(destinationPortVoice));
-
- etARxPort = (EditText) findViewById(R.id.etARxPort);
- etARxPort.setText(Integer.toString(receivePortVoice));
-
- cbEnableNack = (CheckBox) findViewById(R.id.cbNack);
- cbEnableNack.setChecked(enableNack);
-
- cbEnableSpeaker = (CheckBox) findViewById(R.id.cbSpeaker);
- cbEnableSpeaker.setChecked(enableSpeaker);
- cbEnableAGC = (CheckBox) findViewById(R.id.cbAutoGainControl);
- cbEnableAGC.setChecked(enableAGC);
- cbEnableAECM = (CheckBox) findViewById(R.id.cbAECM);
- cbEnableAECM.setChecked(enableAECM);
- cbEnableNS = (CheckBox) findViewById(R.id.cbNoiseSuppression);
- cbEnableNS.setChecked(enableNS);
-
- cbEnableNack.setOnClickListener(this);
- cbEnableSpeaker.setOnClickListener(this);
- cbEnableAECM.setOnClickListener(this);
-
- cbEnableAGC.setOnClickListener(this);
- cbEnableNS.setOnClickListener(this);
-
- tvFrameRateI = (TextView) findViewById(R.id.tvFrameRateI);
- tvBitRateI = (TextView) findViewById(R.id.tvBitRateI);
- tvPacketLoss = (TextView) findViewById(R.id.tvPacketLoss);
- tvFrameRateO = (TextView) findViewById(R.id.tvFrameRateO);
- tvBitRateO = (TextView) findViewById(R.id.tvBitRateO);
-
- }
-
- @Override
- protected void onPause() {
- super.onPause();
- // if (remoteSurfaceView != null)
- // glSurfaceView.onPause();
- }
-
- @Override
- protected void onResume() {
- super.onResume();
- // if (glSurfaceView != null)
- // glSurfaceView.onResume();
- }
-
- private void StartCall() {
- int ret = 0;
-
- if (enableVoice) {
- SetupVoE();
- StartVoiceEngine();
- }
-
- if (enableVideo) {
- if (enableVideoSend) {
- // camera and preview surface
- svLocal = ViERenderer.CreateLocalRenderer(this);
- }
-
- ret = ViEAndroidAPI.GetVideoEngine();
- ret = ViEAndroidAPI.Init(enableTrace);
- channel = ViEAndroidAPI.CreateChannel(voiceChannel);
- ret = ViEAndroidAPI.SetLocalReceiver(channel,
- receivePortVideo);
- ret = ViEAndroidAPI.SetSendDestination(channel,
- destinationPortVideo,
- remoteIp.getBytes());
-
- if (enableVideoReceive) {
- if(android.os.Build.MANUFACTURER.equals("samsung")) {
- // Create an Open GL renderer
- remoteSurfaceView = ViERenderer.CreateRenderer(this, true);
- ret = ViEAndroidAPI.AddRemoteRenderer(channel, remoteSurfaceView);
- }
- else {
- remoteSurfaceView = ViERenderer.CreateRenderer(this, false);
- ret = ViEAndroidAPI.AddRemoteRenderer(channel, remoteSurfaceView);
- }
-
- ret = ViEAndroidAPI.SetReceiveCodec(channel,
- codecType,
- INIT_BITRATE,
- codecSizeWidth,
- codecSizeHeight,
- RECEIVE_CODEC_FRAMERATE);
- ret = ViEAndroidAPI.StartRender(channel);
- ret = ViEAndroidAPI.StartReceive(channel);
- }
-
- if (enableVideoSend) {
- currentCameraOrientation =
- ViEAndroidAPI.GetCameraOrientation(usingFrontCamera?1:0);
- ret = ViEAndroidAPI.SetSendCodec(channel,
- codecType,
- INIT_BITRATE,
- codecSizeWidth,
- codecSizeHeight,
- SEND_CODEC_FRAMERATE);
- int cameraId = ViEAndroidAPI.StartCamera(channel, usingFrontCamera?1:0);
-
- if(cameraId>0) {
- cameraId = cameraId;
- int neededRotation = GetCameraOrientation(currentCameraOrientation);
- ViEAndroidAPI.SetRotation(cameraId,neededRotation);
- }
- else {
- ret=cameraId;
- }
- ret = ViEAndroidAPI.StartSend(channel);
- }
-
- ret = ViEAndroidAPI.SetCallback(channel, this);
-
- if (enableVideoSend) {
- if (mLlLocalSurface != null)
- mLlLocalSurface.addView(svLocal);
- }
-
- if (enableVideoReceive) {
- if (mLlRemoteSurface != null)
- mLlRemoteSurface.addView(remoteSurfaceView);
- }
-
- viERunning = true;
- }
-
- }
-
- private void DemoLog(String msg) {
- Log.d("*WEBRTC*", msg);
- }
-
- private void StopVoiceEngine() {
- // Stop send
- if (0 != ViEAndroidAPI.VoE_StopSend(voiceChannel)) {
- DemoLog("VoE stop send failed");
- }
-
- // Stop listen
- if (0 != ViEAndroidAPI.VoE_StopListen(voiceChannel)) {
- DemoLog("VoE stop listen failed");
- }
-
- // Stop playout
- if (0 != ViEAndroidAPI.VoE_StopPlayout(voiceChannel)) {
- DemoLog("VoE stop playout failed");
- }
-
- if (0 != ViEAndroidAPI.VoE_DeleteChannel(voiceChannel)) {
- DemoLog("VoE delete channel failed");
- }
- voiceChannel=-1;
-
- // Terminate
- if (0 != ViEAndroidAPI.VoE_Terminate()) {
- DemoLog("VoE terminate failed");
- }
- }
-
- private void SetupVoE() {
- // Create VoiceEngine
- // Error logging is done in native API wrapper
- ViEAndroidAPI.VoE_Create(this);
-
- // Initialize
- if (0 != ViEAndroidAPI.VoE_Init(enableTrace)) {
- DemoLog("VoE init failed");
- }
-
- // Create channel
- voiceChannel = ViEAndroidAPI.VoE_CreateChannel();
- if (0 != voiceChannel) {
- DemoLog("VoE create channel failed");
- }
-
- // Suggest to use the voice call audio stream for hardware volume controls
- setVolumeControlStream(AudioManager.STREAM_VOICE_CALL);
- }
-
- private int StartVoiceEngine() {
- // Set local receiver
- if (0 != ViEAndroidAPI.VoE_SetLocalReceiver(voiceChannel,
- receivePortVoice)) {
- DemoLog("VoE set local receiver failed");
- }
-
- if (0 != ViEAndroidAPI.VoE_StartListen(voiceChannel)) {
- DemoLog("VoE start listen failed");
- }
-
- // Route audio
- RouteAudio(enableSpeaker);
-
- // set volume to default value
- if (0 != ViEAndroidAPI.VoE_SetSpeakerVolume(volumeLevel)) {
- DemoLog("VoE set speaker volume failed");
- }
-
- // Start playout
- if (0 != ViEAndroidAPI.VoE_StartPlayout(voiceChannel)) {
- DemoLog("VoE start playout failed");
- }
-
- if (0 != ViEAndroidAPI.VoE_SetSendDestination(voiceChannel,
- destinationPortVoice,
- remoteIp)) {
- DemoLog("VoE set send destination failed");
- }
-
- // 0 = iPCM-wb, 5 = PCMU
- if (0 != ViEAndroidAPI.VoE_SetSendCodec(voiceChannel, voiceCodecType)) {
- DemoLog("VoE set send codec failed");
- }
-
- if (0 != ViEAndroidAPI.VoE_SetECStatus(enableAECM, 5, 0, 28)){
- DemoLog("VoE set EC Status failed");
- }
-
- if (0 != ViEAndroidAPI.VoE_StartSend(voiceChannel)) {
- DemoLog("VoE start send failed");
- }
-
- voERunning = true;
- return 0;
- }
-
- private void RouteAudio(boolean enableSpeaker) {
- int sdkVersion = Integer.parseInt(android.os.Build.VERSION.SDK);
- if (sdkVersion >= 5) {
- AudioManager am =
- (AudioManager) this.getSystemService(Context.AUDIO_SERVICE);
- am.setSpeakerphoneOn(enableSpeaker);
- }
- else {
- if (0 != ViEAndroidAPI.VoE_SetLoudspeakerStatus(enableSpeaker)) {
- DemoLog("VoE set louspeaker status failed");
- }
- }
- }
-
- public void onClick(View arg0) {
- switch (arg0.getId()) {
- case R.id.btSwitchCamera:
- if (usingFrontCamera ){
- btSwitchCamera.setText(R.string.frontCamera);
- }
- else {
- btSwitchCamera.setText(R.string.backCamera);
- }
- usingFrontCamera = !usingFrontCamera;
-
- if (viERunning) {
- currentCameraOrientation =
- ViEAndroidAPI.GetCameraOrientation(usingFrontCamera?1:0);
- ViEAndroidAPI.StopCamera(cameraId);
- mLlLocalSurface.removeView(svLocal);
-
- ViEAndroidAPI.StartCamera(channel,usingFrontCamera?1:0);
- mLlLocalSurface.addView(svLocal);
- int neededRotation = GetCameraOrientation(currentCameraOrientation);
- ViEAndroidAPI.SetRotation(cameraId, neededRotation);
- }
- break;
- case R.id.btStartStopCall:
- ReadSettings();
- if (viERunning || voERunning) {
- StopAll();
- wakeLock.release();//release the wake lock
- btStartStopCall.setText(R.string.startCall);
- }
- else if (enableVoice || enableVideo){
- StartCall();
- wakeLock.acquire();//screen stay on during the call
- btStartStopCall.setText(R.string.stopCall);
- }
- break;
- case R.id.btExit:
- StopAll();
- finish();
- break;
- case R.id.cbNack:
- enableNack = cbEnableNack.isChecked();
- if (viERunning) {
- ViEAndroidAPI.EnableNACK(channel, enableNack);
- }
- break;
- case R.id.cbSpeaker:
- enableSpeaker = cbEnableSpeaker.isChecked();
- if (voERunning){
- RouteAudio(enableSpeaker);
- }
- break;
- case R.id.cbAutoGainControl:
- enableAGC=cbEnableAGC.isChecked();
- if(voERunning) {
- //Enable AGC default mode.
- ViEAndroidAPI.VoE_SetAGCStatus(enableAGC,1);
- }
- break;
- case R.id.cbNoiseSuppression:
- enableNS=cbEnableNS.isChecked();
- if(voERunning) {
- //Enable NS default mode.
- ViEAndroidAPI.VoE_SetNSStatus(enableNS, 1);
- }
- break;
- case R.id.cbAECM:
- enableAECM = cbEnableAECM.isChecked();
- if (voERunning) {
- //EC_AECM=5
- //AECM_DEFAULT=0
- ViEAndroidAPI.VoE_SetECStatus(enableAECM, 5, 0, 28);
- }
- break;
- }
- }
-
- private void ReadSettings() {
- codecType = spCodecType.getSelectedItemPosition();
- voiceCodecType = spVoiceCodecType.getSelectedItemPosition();
-
- String sCodecSize = spCodecSize.getSelectedItem().toString();
- String[] aCodecSize = sCodecSize.split("x");
- codecSizeWidth = Integer.parseInt(aCodecSize[0]);
- codecSizeHeight = Integer.parseInt(aCodecSize[1]);
-
- loopbackMode = cbLoopback.isChecked();
- enableVoice = cbVoice.isChecked();
- enableVideoSend = cbVideoSend.isChecked();
- enableVideoReceive = cbVideoReceive.isChecked();
- enableVideo = enableVideoSend || enableVideoReceive;
-
- destinationPortVideo =
- Integer.parseInt(etVTxPort.getText().toString());
- receivePortVideo =
- Integer.parseInt(etVRxPort.getText().toString());
- destinationPortVoice =
- Integer.parseInt(etATxPort.getText().toString());
- receivePortVoice =
- Integer.parseInt(etARxPort.getText().toString());
-
- enableNack = cbEnableNack.isChecked();
- enableSpeaker = cbEnableSpeaker.isChecked();
- enableAGC = cbEnableAGC.isChecked();
- enableAECM = cbEnableAECM.isChecked();
- enableNS = cbEnableNS.isChecked();
-
- if (loopbackMode)
- remoteIp = "127.0.0.1";
- else
- remoteIp = etRemoteIp.getText().toString();
- }
-
- public void onItemSelected(AdapterView<?> adapterView, View view,
- int position, long id) {
- if ((adapterView==spCodecType || adapterView==spCodecSize) &&
- viERunning) {
- ReadSettings();
- //change the codectype
- if (enableVideoReceive) {
- if (0 !=ViEAndroidAPI.SetReceiveCodec(channel, codecType,
- INIT_BITRATE, codecSizeWidth,
- codecSizeHeight,
- RECEIVE_CODEC_FRAMERATE))
- DemoLog("ViE set receive codec failed");
- }
- if (enableVideoSend) {
- if (0!=ViEAndroidAPI.SetSendCodec(channel, codecType, INIT_BITRATE,
- codecSizeWidth,
- codecSizeHeight,
- SEND_CODEC_FRAMERATE))
- DemoLog("ViE set send codec failed");
- }
- }
- else if ((adapterView==spVoiceCodecType) && voERunning) {
- //change voice engine codec
- ReadSettings();
- if (0 != ViEAndroidAPI.VoE_SetSendCodec(voiceChannel, voiceCodecType)) {
- DemoLog("VoE set send codec failed");
- }
- }
- }
-
- public void onNothingSelected(AdapterView<?> arg0) {
- DemoLog("No setting selected");
- }
-
- public int UpdateStats(int in_frameRateI, int in_bitRateI, int in_packetLoss,
- int in_frameRateO, int in_bitRateO) {
- frameRateI = in_frameRateI;
- bitRateI = in_bitRateI;
- packetLoss = in_packetLoss;
- frameRateO = in_frameRateO;
- bitRateO = in_bitRateO;
- runOnUiThread(new Runnable() {
- public void run() {
- tvFrameRateI.setText("Incoming FrameRate - " +
- Integer.toString(frameRateI));
- tvBitRateI.setText("Incoming BitRate - " +
- Integer.toString(bitRateI));
- tvPacketLoss.setText("Incoming Packet Loss - " +
- Integer.toString(packetLoss));
- tvFrameRateO.setText("Send FrameRate - " +
- Integer.toString(frameRateO));
- tvBitRateO.setText("Send BitRate - " +
- Integer.toString(bitRateO));
- }
- });
- return 0;
- }
-}
diff --git a/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/ViEAndroidJavaAPI.java b/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/ViEAndroidJavaAPI.java
deleted file mode 100644
index 701f8bd..0000000
--- a/trunk/src/video_engine/main/test/android_test/src/org/webrtc/videoengineapp/ViEAndroidJavaAPI.java
+++ /dev/null
@@ -1,138 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-package org.webrtc.videoengineapp;
-
-import android.app.Activity;
-import android.content.Context;
-import android.util.Log;
-import android.view.SurfaceHolder;
-import android.view.SurfaceView;
-
-public class ViEAndroidJavaAPI {
-
- public ViEAndroidJavaAPI(Context context) {
- Log.d("*WEBRTCJ*", "Loading ViEAndroidJavaAPI...");
- System.loadLibrary("webrtc-video-demo-jni");
-
- Log.d("*WEBRTCJ*", "Calling native init...");
- if (!NativeInit(context)) {
- Log.e("*WEBRTCJ*", "Native init failed");
- throw new RuntimeException("Native init failed");
- }
- else {
- Log.d("*WEBRTCJ*", "Native init successful");
- }
- String a = "";
- a.getBytes();
- }
-
- // API Native
- private native boolean NativeInit(Context context);
-
- // Video Engine API
- // Initialization and Termination functions
- public native int GetVideoEngine();
- public native int Init(boolean enableTrace);
- public native int Terminate();
-
-
- public native int StartSend(int channel);
- public native int StopRender(int channel);
- public native int StopSend(int channel);
- public native int StartReceive(int channel);
- public native int StopReceive(int channel);
- // Channel functions
- public native int CreateChannel(int voiceChannel);
- // Receiver & Destination functions
- public native int SetLocalReceiver(int channel, int port);
- public native int SetSendDestination(int channel, int port, byte ipadr[]);
- // Codec
- public native int SetReceiveCodec(int channel, int codecNum,
- int intbitRate, int width,
- int height, int frameRate);
- public native int SetSendCodec(int channel, int codecNum,
- int intbitRate, int width,
- int height, int frameRate);
- // Rendering
- public native int AddRemoteRenderer(int channel, Object glSurface);
- public native int RemoveRemoteRenderer(int channel);
- public native int StartRender(int channel);
-
- // Capture
- public native int StartCamera(int channel, int cameraNum);
- public native int StopCamera(int cameraId);
- public native int GetCameraOrientation(int cameraNum);
- public native int SetRotation(int cameraId,int degrees);
-
- // NACK
- public native int EnableNACK(int channel, boolean enable);
-
- //PLI for H.264
- public native int EnablePLI(int channel, boolean enable);
-
- // Enable stats callback
- public native int SetCallback(int channel, IViEAndroidCallback callback);
-
- // Voice Engine API
- // Create and Delete functions
- public native boolean VoE_Create(Activity context);
- public native boolean VoE_Delete();
-
- // Initialization and Termination functions
- public native int VoE_Authenticate(String key);
- public native int VoE_Init(boolean enableTrace);
- public native int VoE_Terminate();
-
- // Channel functions
- public native int VoE_CreateChannel();
- public native int VoE_DeleteChannel(int channel);
-
- // Receiver & Destination functions
- public native int VoE_SetLocalReceiver(int channel, int port);
- public native int VoE_SetSendDestination(int channel, int port,
- String ipaddr);
-
- // Media functions
- public native int VoE_StartListen(int channel);
- public native int VoE_StartPlayout(int channel);
- public native int VoE_StartSend(int channel);
- public native int VoE_StopListen(int channel);
- public native int VoE_StopPlayout(int channel);
- public native int VoE_StopSend(int channel);
-
- // Volume
- public native int VoE_SetSpeakerVolume(int volume);
-
- // Hardware
- public native int VoE_SetLoudspeakerStatus(boolean enable);
-
- // Playout file locally
- public native int VoE_StartPlayingFileLocally(int channel,
- String fileName,
- boolean loop);
- public native int VoE_StopPlayingFileLocally(int channel);
-
- // Play file as microphone
- public native int VoE_StartPlayingFileAsMicrophone(int channel,
- String fileName,
- boolean loop);
- public native int VoE_StopPlayingFileAsMicrophone(int channel);
-
- // Codec-setting functions
- public native int VoE_NumOfCodecs();
- public native int VoE_SetSendCodec(int channel, int index);
-
- //VE funtions
- public native int VoE_SetECStatus(boolean enable, int mode,
- int AESmode, int AESattenuation);
- public native int VoE_SetAGCStatus(boolean enable, int mode);
- public native int VoE_SetNSStatus(boolean enable, int mode);
-}
diff --git a/trunk/src/video_engine/test/auto_test/android/.classpath b/trunk/src/video_engine/test/auto_test/android/.classpath
deleted file mode 100644
index f2adf55..0000000
--- a/trunk/src/video_engine/test/auto_test/android/.classpath
+++ /dev/null
@@ -1,9 +0,0 @@
-<?xml version="1.0" encoding="UTF-8"?>
-<classpath>
- <classpathentry kind="src" path="src"/>
- <classpathentry kind="src" path="gen"/>
- <classpathentry kind="con" path="com.android.ide.eclipse.adt.ANDROID_FRAMEWORK"/>
- <classpathentry kind="lib" path="libs/VideoEngine_android_java.jar"/>
- <classpathentry kind="lib" path="libs/VoiceEngine_android_java.jar"/>
- <classpathentry kind="output" path="bin"/>
-</classpath>
diff --git a/trunk/src/video_engine/test/auto_test/android/Android.mk b/trunk/src/video_engine/test/auto_test/android/Android.mk
deleted file mode 100644
index 2b20e80..0000000
--- a/trunk/src/video_engine/test/auto_test/android/Android.mk
+++ /dev/null
@@ -1,39 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH:= $(call my-dir)
-
-include $(CLEAR_VARS)
-
-MY_CAPTURE_FOLDER := ../../../../modules/video_capture/main/source
-MY_CAPTURE_JAVA_FOLDER := Android/java/org/webrtc/videoengine
-MY_CAPTURE_PATH := $(MY_CAPTURE_FOLDER)/$(MY_CAPTURE_JAVA_FOLDER)
-
-MY_RENDER_FOLDER := ../../../../modules/video_render/main/source
-MY_RENDER_JAVA_FOLDER := Android/java/org/webrtc/videoengine
-MY_RENDER_PATH := $(MY_RENDER_FOLDER)/$(MY_RENDER_JAVA_FOLDER)
-
-LOCAL_MODULE_TAGS := tests
-
-LOCAL_SRC_FILES := \
- src/org/webrtc/vieautotest/ViEAutotest.java \
- $(MY_CAPTURE_PATH)/CaptureCapabilityAndroid.java \
- $(MY_CAPTURE_PATH)/VideoCaptureAndroid.java \
- $(MY_CAPTURE_PATH)/VideoCaptureDeviceInfoAndroid.java \
- $(MY_RENDER_PATH)/ViEAndroidGLES20.java \
- $(MY_RENDER_PATH)/ViERenderer.java \
- $(MY_RENDER_PATH)/ViESurfaceRenderer.java
-
-LOCAL_PACKAGE_NAME := webrtc-video-autotest
-LOCAL_CERTIFICATE := platform
-
-LOCAL_JNI_SHARED_LIBRARIES := libwebrtc-video-autotest-jni
-
-include $(BUILD_PACKAGE)
-
-include $(call all-makefiles-under,$(LOCAL_PATH))
diff --git a/trunk/src/video_engine/test/auto_test/android/AndroidManifest.xml b/trunk/src/video_engine/test/auto_test/android/AndroidManifest.xml
deleted file mode 100644
index 11b3e27..0000000
--- a/trunk/src/video_engine/test/auto_test/android/AndroidManifest.xml
+++ /dev/null
@@ -1,26 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<manifest xmlns:android="http://schemas.android.com/apk/res/android"
- android:versionCode="1"
- android:versionName="1.0" package="org.webrtc.vieautotest">
- <application android:label="@string/app_name"
- android:debuggable="true"
- android:icon="@drawable/logo">
- <activity android:label="@string/app_name"
- android:name="ViEAutotest">
- <intent-filter>
- <action android:name="android.intent.action.MAIN" />
- <category android:name="android.intent.category.LAUNCHER" />
- </intent-filter>
- </activity>
-
- </application>
- <uses-sdk android:minSdkVersion="3" android:targetSdkVersion="8" />
- <uses-permission android:name="android.permission.CAMERA"></uses-permission>
- <uses-feature android:name="android.hardware.camera" />
- <uses-feature android:name="android.hardware.camera.autofocus" />
- <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
- <uses-permission android:name="android.permission.RECORD_AUDIO" />
- <uses-permission android:name="android.permission.INTERNET" />
- <uses-permission android:name="android.permission.WAKE_LOCK" />
- <uses-permission android:name="android.permission.WRITE_EXTERNAL_STORAGE"/>
-</manifest>
diff --git a/trunk/src/video_engine/test/auto_test/android/default.properties b/trunk/src/video_engine/test/auto_test/android/default.properties
deleted file mode 100644
index 2ad44a4..0000000
--- a/trunk/src/video_engine/test/auto_test/android/default.properties
+++ /dev/null
@@ -1,11 +0,0 @@
-# This file is automatically generated by Android Tools.
-# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
-#
-# This file must be checked in Version Control Systems.
-#
-# To customize properties used by the Ant build system use,
-# "build.properties", and override values to adapt the script to your
-# project structure.
-
-# Project target.
-target=android-9
diff --git a/trunk/src/video_engine/test/auto_test/android/gen/org/webrtc/vieautotest/R.java b/trunk/src/video_engine/test/auto_test/android/gen/org/webrtc/vieautotest/R.java
deleted file mode 100644
index 4b46020..0000000
--- a/trunk/src/video_engine/test/auto_test/android/gen/org/webrtc/vieautotest/R.java
+++ /dev/null
@@ -1,37 +0,0 @@
-/* AUTO-GENERATED FILE. DO NOT MODIFY.
- *
- * This class was automatically generated by the
- * aapt tool from the resource data it found. It
- * should not be modified by hand.
- */
-
-package org.webrtc.vieautotest;
-
-public final class R {
- public static final class array {
- public static final int subtest_array=0x7f050001;
- public static final int test_array=0x7f050000;
- }
- public static final class attr {
- }
- public static final class drawable {
- public static final int logo=0x7f020000;
- }
- public static final class id {
- public static final int Button01=0x7f060004;
- public static final int LocalView=0x7f060001;
- public static final int RemoteView=0x7f060000;
- public static final int subtestSpinner=0x7f060003;
- public static final int testSpinner=0x7f060002;
- }
- public static final class layout {
- public static final int main=0x7f030000;
- }
- public static final class string {
- public static final int SpinnerSubtest=0x7f040004;
- public static final int SpinnerTitle=0x7f040003;
- public static final int TitleName=0x7f040001;
- public static final int app_name=0x7f040000;
- public static final int run_button=0x7f040002;
- }
-}
diff --git a/trunk/src/video_engine/test/auto_test/android/jni/Android.mk b/trunk/src/video_engine/test/auto_test/android/jni/Android.mk
deleted file mode 100644
index ac43975..0000000
--- a/trunk/src/video_engine/test/auto_test/android/jni/Android.mk
+++ /dev/null
@@ -1,77 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-# the follow two lines are for NDK build
-INTERFACES_PATH := $(LOCAL_PATH)/../../../../../../build/interface
-LIBS_PATH := $(LOCAL_PATH)/../../../../../../build/libraries
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_TAGS := tests
-LOCAL_MODULE := libwebrtc-video-autotest-jni
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := \
- vie_autotest_jni.cc \
- ../../source/vie_autotest_android.cc \
- ../../source/vie_autotest.cc \
- ../../source/vie_autotest_base.cc \
- ../../source/vie_autotest_capture.cc \
- ../../source/vie_autotest_codec.cc \
- ../../source/vie_autotest_encryption.cc \
- ../../source/vie_autotest_file.cc \
- ../../source/vie_autotest_image_process.cc \
- ../../source/vie_autotest_loopback.cc \
- ../../source/vie_autotest_network.cc \
- ../../source/vie_autotest_render.cc \
- ../../source/vie_autotest_rtp_rtcp.cc \
- ../../source/tb_I420_codec.cc \
- ../../source/tb_capture_device.cc \
- ../../source/tb_external_transport.cc \
- ../../source/tb_interfaces.cc \
- ../../source/tb_video_channel.cc
-
-LOCAL_CFLAGS := \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES'
-
-LOCAL_C_INCLUDES := \
- external/gtest/include \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../interface \
- $(LOCAL_PATH)/../../../interface \
- $(LOCAL_PATH)/../../.. \
- $(LOCAL_PATH)/../../../../.. \
- $(LOCAL_PATH)/../../../../../common_video/interface \
- $(LOCAL_PATH)/../../../../../common_video/vplib/main/interface \
- $(LOCAL_PATH)/../../../../../modules/interface \
- $(LOCAL_PATH)/../../../../../modules/video_capture/main/interface \
- $(LOCAL_PATH)/../../../../../modules/video_capture/main/source \
- $(LOCAL_PATH)/../../../../../modules/video_coding/codecs/interface \
- $(LOCAL_PATH)/../../../../../modules/video_render/main/interface \
- $(LOCAL_PATH)/../../../../../voice_engine/main/interface \
- $(LOCAL_PATH)/../../../../../system_wrappers/interface
-
-LOCAL_PRELINK_MODULE := false
-
-LOCAL_SHARED_LIBRARIES := \
- libutils \
- libstlport \
- libandroid \
- libwebrtc \
- libGLESv2
-
-# the following line is for NDK build
-LOCAL_LDLIBS := $(LIBS_PATH)/VideoEngine_android_gcc.a -llog -lgcc
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_SHARED_LIBRARY)
diff --git a/trunk/src/video_engine/test/auto_test/android/jni/org_webrtc_vieautotest_vie_autotest.h b/trunk/src/video_engine/test/auto_test/android/jni/org_webrtc_vieautotest_vie_autotest.h
deleted file mode 100644
index 68ec601..0000000
--- a/trunk/src/video_engine/test/auto_test/android/jni/org_webrtc_vieautotest_vie_autotest.h
+++ /dev/null
@@ -1,34 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/* DO NOT EDIT THIS FILE - it is machine generated */
-#include <jni.h>
-/* Header for class org_webrtc_vieautotest_ViEAutotest */
-
-#ifndef _Included_org_webrtc_vieautotest_ViEAutotest
-#define _Included_org_webrtc_vieautotest_ViEAutotest
-#ifdef __cplusplus
-extern "C" {
-#endif
-
-
-/*
- * Class: org_webrtc_vieautotest_ViEAutotest
- * Method: RunTest
- * Signature: (IILandroid/view/SurfaceView;Landroid/view/SurfaceView;)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_vieautotest_ViEAutotest_RunTest__IILandroid_view_SurfaceView_2Landroid_view_SurfaceView_2
-(JNIEnv *, jobject, jint, jint, jobject, jobject);
-
-#ifdef __cplusplus
-}
-#endif
-#endif
diff --git a/trunk/src/video_engine/test/auto_test/android/jni/vie_autotest_jni.cc b/trunk/src/video_engine/test/auto_test/android/jni/vie_autotest_jni.cc
deleted file mode 100644
index 2e55283..0000000
--- a/trunk/src/video_engine/test/auto_test/android/jni/vie_autotest_jni.cc
+++ /dev/null
@@ -1,137 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <string.h>
-#include <android/log.h>
-#include <pthread.h>
-#include <unistd.h>
-
-#include "org_webrtc_vieautotest_vie_autotest.h"
-
-#include "vie_autotest_android.h"
-
-#define WEBRTC_LOG_TAG "*WEBRTCN*"
-
-// VideoEngine data struct
-typedef struct
-{
- JavaVM* jvm;
-} VideoEngineData;
-
-// Global variables
-JavaVM* webrtcGlobalVM;
-
-// Global variables visible in this file
-static VideoEngineData vieData;
-
-// "Local" functions (i.e. not Java accessible)
-#define WEBRTC_TRACE_MAX_MESSAGE_SIZE 1024
-
-static bool GetSubAPIs(VideoEngineData& vieData);
-static bool ReleaseSubAPIs(VideoEngineData& vieData);
-
-//
-// General functions
-//
-
-// JNI_OnLoad
-jint JNI_OnLoad(JavaVM* vm, void* /*reserved*/) {
- if (!vm) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "JNI_OnLoad did not receive a valid VM pointer");
- return -1;
- }
-
- JNIEnv* env;
- if (JNI_OK != vm->GetEnv(reinterpret_cast<void**> (&env),
- JNI_VERSION_1_4)) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "JNI_OnLoad could not get JNI env");
- return -1;
- }
-
- // Init ViE data
- vieData.jvm = vm;
-
- return JNI_VERSION_1_4;
-}
-
-// Class: org_webrtc_vieautotest_ViEAutotest
-// Method: RunTest
-// Signature: (IILandroid/opengl/GLSurfaceView;Landroid/opengl/GLSurfaceView;)I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_vieautotest_ViEAutotest_RunTest__IILandroid_opengl_GLSurfaceView_2Landroid_opengl_GLSurfaceView_2(
- JNIEnv* env,
- jobject context,
- jint testType,
- jint subtestType,
- jobject glView1,
- jobject glView2)
-{
- int numErrors = -1;
- numErrors = ViEAutoTestAndroid::RunAutotest(testType, subtestType, glView1,
- glView2, vieData.jvm, env,
- context);
- return numErrors;
-}
-
-// Class: org_webrtc_vieautotest_ViEAutotest
-// Method: RunTest
-// Signature: (IILandroid/view/SurfaceView;Landroid/view/SurfaceView;)I
-JNIEXPORT jint JNICALL
-Java_org_webrtc_vieautotest_ViEAutotest_RunTest__IILandroid_view_SurfaceView_2Landroid_view_SurfaceView_2(
- JNIEnv* env,
- jobject context,
- jint testType,
- jint subtestType,
- jobject surfaceHolder1,
- jobject surfaceHolder2)
-{
- int numErrors = -1;
- numErrors = ViEAutoTestAndroid::RunAutotest(testType, subtestType,
- surfaceHolder1, surfaceHolder2,
- vieData.jvm, env, context);
- return numErrors;
-}
-
-//
-//local function
-//
-
-bool GetSubAPIs(VideoEngineData& vieData) {
- bool retVal = true;
- //vieData.base = ViEBase::GetInterface(vieData.vie);
- //if (vieData.base == NULL)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Could not get Base API");
- retVal = false;
- }
- return retVal;
-}
-
-bool ReleaseSubAPIs(VideoEngineData& vieData) {
- bool releaseOk = true;
- //if (vieData.base)
- {
- //if (vieData.base->Release() != 0)
- if (false) {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Release base sub-API failed");
- releaseOk = false;
- }
- else {
- //vieData.base = NULL;
- }
- }
-
- return releaseOk;
-}
diff --git a/trunk/src/video_engine/test/auto_test/android/res/drawable/logo.png b/trunk/src/video_engine/test/auto_test/android/res/drawable/logo.png
deleted file mode 100644
index c3e0a12..0000000
--- a/trunk/src/video_engine/test/auto_test/android/res/drawable/logo.png
+++ /dev/null
Binary files differ
diff --git a/trunk/src/video_engine/test/auto_test/android/res/layout/main.xml b/trunk/src/video_engine/test/auto_test/android/res/layout/main.xml
deleted file mode 100644
index 1f2aaf9..0000000
--- a/trunk/src/video_engine/test/auto_test/android/res/layout/main.xml
+++ /dev/null
@@ -1,64 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<FrameLayout xmlns:android="http://schemas.android.com/apk/res/android"
- android:layout_width="fill_parent"
- android:layout_height="fill_parent">
- <RelativeLayout
- android:layout_width="fill_parent"
- android:layout_height="fill_parent">
- <LinearLayout
- android:id="@+id/RemoteView"
- android:layout_width="fill_parent"
- android:layout_height="fill_parent"
- android:layout_weight="1">
- <!-- log instead of video
- <ImageView
- android:layout_width="fill_parent"
- android:layout_height="fill_parent"
- android:scaleType="fitXY"
- android:src="@drawable/logo" /> -->
-
- </LinearLayout>
- <LinearLayout
- android:id="@+id/LocalView"
- android:layout_width="120dip"
- android:layout_height="120dip"
- android:layout_weight="1">
- <!-- <ImageView
- android:layout_width="fill_parent"
- android:layout_height="fill_parent"
- android:scaleType="fitXY"
- android:src="@drawable/logo" /> -->
- </LinearLayout>
- <LinearLayout
- android:orientation="horizontal"
- android:layout_width="fill_parent"
- android:layout_height="wrap_content"
- android:layout_alignParentBottom="true">
- <LinearLayout
- android:orientation="vertical"
- android:layout_width="fill_parent"
- android:layout_height="wrap_content"
- android:layout_alignParentBottom="true">
- <Spinner
- android:id="@+id/testSpinner"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:prompt="@string/SpinnerTitle"
- />
- <Spinner
- android:id="@+id/subtestSpinner"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content"
- android:prompt="@string/SpinnerSubtest"
- />
- <Button
- android:text="@string/run_button"
- android:id="@+id/Button01"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </Button>
- </LinearLayout>
-
- </LinearLayout>
- </RelativeLayout>
-</FrameLayout>
diff --git a/trunk/src/video_engine/test/auto_test/android/res/values/strings.xml b/trunk/src/video_engine/test/auto_test/android/res/values/strings.xml
deleted file mode 100644
index ba59c5e..0000000
--- a/trunk/src/video_engine/test/auto_test/android/res/values/strings.xml
+++ /dev/null
@@ -1,31 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<resources>
-
-<string name="app_name">ViEAutotest</string>
-<string name="TitleName">ViEAutotest</string>
-<string name="run_button">Run Test</string>
-<string name="SpinnerTitle">Test type...</string>
-<string-array name="test_array">
- <item>Standard</item>
- <item>API</item>
- <item>Extended</item>
- <item>Loopback</item>
- <item>Custom</item>
-</string-array>
-<string name="SpinnerSubtest">Run...</string>
-<string-array name="subtest_array">
- <item>All</item>
- <item>Base</item>
- <item>Capture</item>
- <item>Codec</item>
- <item>Mix</item>
- <item>Encryption</item>
- <item>External Codec</item>
- <item>File</item>
- <item>Image Process</item>
- <item>Network</item>
- <item>Render</item>
- <item>RTP/RTCP</item>
-</string-array>
-
-</resources>
diff --git a/trunk/src/video_engine/test/auto_test/android/src/org/webrtc/vieautotest/ViEAutotest.java b/trunk/src/video_engine/test/auto_test/android/src/org/webrtc/vieautotest/ViEAutotest.java
deleted file mode 100644
index de228a8..0000000
--- a/trunk/src/video_engine/test/auto_test/android/src/org/webrtc/vieautotest/ViEAutotest.java
+++ /dev/null
@@ -1,162 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-package org.webrtc.vieautotest;
-
-import org.webrtc.vieautotest.R;
-
-import android.app.Activity;
-import android.os.Bundle;
-import android.util.Log;
-import android.widget.Button;
-import android.view.SurfaceView;
-import android.view.View;
-import android.view.SurfaceHolder;
-import android.widget.LinearLayout;
-import android.opengl.GLSurfaceView;
-import android.widget.Spinner;
-import android.widget.ArrayAdapter;
-import android.widget.AdapterView;
-
-public class ViEAutotest extends Activity
- implements
- AdapterView.OnItemSelectedListener,
- View.OnClickListener {
-
- private Thread testThread;
- private Spinner testSpinner;
- private Spinner subtestSpinner;
- private int testSelection;
- private int subTestSelection;
-
- // View for remote video
- private LinearLayout remoteSurface = null;
- private GLSurfaceView glSurfaceView = null;
- private SurfaceView surfaceView = null;
-
- private LinearLayout localSurface = null;
- private GLSurfaceView glLocalSurfaceView = null;
- private SurfaceView localSurfaceView = null;
-
- /** Called when the activity is first created. */
- @Override
- public void onCreate(Bundle savedInstanceState) {
-
- Log.d("*WEBRTC*", "onCreate called");
-
- super.onCreate(savedInstanceState);
- setContentView(R.layout.main);
-
- // Set the Start button action
- final Button buttonStart = (Button) findViewById(R.id.Button01);
- buttonStart.setOnClickListener(this);
-
- // Set test spinner
- testSpinner = (Spinner) findViewById(R.id.testSpinner);
- ArrayAdapter<CharSequence> adapter =
- ArrayAdapter.createFromResource(this, R.array.test_array,
- android.R.layout.simple_spinner_item);
-
- int resource = android.R.layout.simple_spinner_dropdown_item;
- adapter.setDropDownViewResource(resource);
- testSpinner.setAdapter(adapter);
- testSpinner.setOnItemSelectedListener(this);
-
- // Set sub test spinner
- subtestSpinner = (Spinner) findViewById(R.id.subtestSpinner);
- ArrayAdapter<CharSequence> subtestAdapter =
- ArrayAdapter.createFromResource(this, R.array.subtest_array,
- android.R.layout.simple_spinner_item);
-
- subtestAdapter.setDropDownViewResource(resource);
- subtestSpinner.setAdapter(subtestAdapter);
- subtestSpinner.setOnItemSelectedListener(this);
-
- remoteSurface = (LinearLayout) findViewById(R.id.RemoteView);
- surfaceView = new SurfaceView(this);
- remoteSurface.addView(surfaceView);
-
- localSurface = (LinearLayout) findViewById(R.id.LocalView);
- localSurfaceView = new SurfaceView(this);
- localSurfaceView.setZOrderMediaOverlay(true);
- localSurface.addView(localSurfaceView);
-
- // Set members
- testSelection = 0;
- subTestSelection = 0;
- }
-
- public void onClick(View v) {
- Log.d("*WEBRTC*", "Button clicked...");
- switch (v.getId()) {
- case R.id.Button01:
- new Thread(new Runnable() {
- public void run() {
- Log.d("*WEBRTC*", "Calling RunTest...");
- RunTest(testSelection, subTestSelection,
- localSurfaceView, surfaceView);
- Log.d("*WEBRTC*", "RunTest done");
- }
- }).start();
- }
- }
-
- public void onItemSelected(AdapterView<?> parent, View v,
- int position, long id) {
-
- if (parent == (Spinner) findViewById(R.id.testSpinner)) {
- testSelection = position;
- } else {
- subTestSelection = position;
- }
- }
-
- public void onNothingSelected(AdapterView<?> parent) {
- }
-
- @Override
- protected void onStart() {
- super.onStart();
- }
-
- @Override
- protected void onResume() {
- super.onResume();
- }
-
- @Override
- protected void onPause() {
- super.onPause();
- }
-
- @Override
- protected void onStop() {
- super.onStop();
- }
-
- @Override
- protected void onDestroy() {
-
- super.onDestroy();
- }
-
- // C++ function performing the chosen test
- // private native int RunTest(int testSelection, int subtestSelection,
- // GLSurfaceView window1, GLSurfaceView window2);
- private native int RunTest(int testSelection, int subtestSelection,
- SurfaceView window1, SurfaceView window2);
-
- // this is used to load the 'ViEAutotestJNIAPI' library on application
- // startup.
- static {
- Log.d("*WEBRTC*", "Loading ViEAutotest...");
- System.loadLibrary("webrtc-video-autotest-jni");
- }
-}
diff --git a/trunk/src/video_engine/test/auto_test/automated/legacy_fixture.cc b/trunk/src/video_engine/test/auto_test/automated/legacy_fixture.cc
deleted file mode 100644
index 591a567..0000000
--- a/trunk/src/video_engine/test/auto_test/automated/legacy_fixture.cc
+++ /dev/null
@@ -1,28 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/test/auto_test/automated/legacy_fixture.h"
-
-#include "video_engine/test/auto_test/interface/vie_autotest.h"
-
-void LegacyFixture::SetUpTestCase() {
- TwoWindowsFixture::SetUpTestCase();
-
- // Create the test cases
- tests_ = new ViEAutoTest(window_1_, window_2_);
-}
-
-void LegacyFixture::TearDownTestCase() {
- delete tests_;
-
- TwoWindowsFixture::TearDownTestCase();
-}
-
-ViEAutoTest* LegacyFixture::tests_ = NULL;
diff --git a/trunk/src/video_engine/test/auto_test/automated/legacy_fixture.h b/trunk/src/video_engine/test/auto_test/automated/legacy_fixture.h
deleted file mode 100644
index b452766..0000000
--- a/trunk/src/video_engine/test/auto_test/automated/legacy_fixture.h
+++ /dev/null
@@ -1,29 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VIDEO_ENGINE_TEST_AUTO_TEST_AUTOMATED_VIE_LEGACY_FIXTURE_H_
-#define SRC_VIDEO_ENGINE_TEST_AUTO_TEST_AUTOMATED_VIE_LEGACY_FIXTURE_H_
-
-#include "video_engine/test/auto_test/automated/two_windows_fixture.h"
-
-// Inherited by old-style standard integration tests based on ViEAutoTest.
-class LegacyFixture : public TwoWindowsFixture {
- public:
- // Initializes ViEAutoTest in addition to the work done by ViEIntegrationTest.
- static void SetUpTestCase();
-
- // Releases anything allocated by SetupTestCase.
- static void TearDownTestCase();
-
- protected:
- static ViEAutoTest* tests_;
-};
-
-#endif // SRC_VIDEO_ENGINE_TEST_AUTO_TEST_AUTOMATED_VIE_LEGACY_FIXTURE_H_
diff --git a/trunk/src/video_engine/test/auto_test/automated/two_windows_fixture.cc b/trunk/src/video_engine/test/auto_test/automated/two_windows_fixture.cc
deleted file mode 100644
index d181aa5..0000000
--- a/trunk/src/video_engine/test/auto_test/automated/two_windows_fixture.cc
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/test/auto_test/automated/two_windows_fixture.h"
-
-#include "video_engine/test/auto_test/helpers/vie_window_creator.h"
-#include "video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h"
-
-void TwoWindowsFixture::SetUpTestCase() {
- window_creator_ = new ViEWindowCreator();
-
- ViEAutoTestWindowManagerInterface* window_manager =
- window_creator_->CreateTwoWindows();
-
- window_1_ = window_manager->GetWindow1();
- window_2_ = window_manager->GetWindow2();
-}
-
-void TwoWindowsFixture::TearDownTestCase() {
- window_creator_->TerminateWindows();
- delete window_creator_;
-}
-
-ViEWindowCreator* TwoWindowsFixture::window_creator_ = NULL;
-void* TwoWindowsFixture::window_1_ = NULL;
-void* TwoWindowsFixture::window_2_ = NULL;
diff --git a/trunk/src/video_engine/test/auto_test/automated/two_windows_fixture.h b/trunk/src/video_engine/test/auto_test/automated/two_windows_fixture.h
deleted file mode 100644
index 175a42d..0000000
--- a/trunk/src/video_engine/test/auto_test/automated/two_windows_fixture.h
+++ /dev/null
@@ -1,35 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_AUTOMATED_TWO_WINDOWS_FIXTURE_H_
-#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_AUTOMATED_TWO_WINDOWS_FIXTURE_H_
-
-#include "gtest/gtest.h"
-
-class ViEWindowCreator;
-class ViEAutoTest;
-
-// Meant to be inherited by all standard test who require two windows.
-class TwoWindowsFixture : public testing::Test {
- public:
- // Launches two windows in a platform-dependent manner and stores the handles
- // in the window_1_ and window_2_ fields.
- static void SetUpTestCase();
-
- // Releases anything allocated by SetupTestCase.
- static void TearDownTestCase();
-
- protected:
- static void* window_1_;
- static void* window_2_;
- static ViEWindowCreator* window_creator_;
-};
-
-#endif // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_AUTOMATED_TWO_WINDOWS_FIXTURE_H_
diff --git a/trunk/src/video_engine/test/auto_test/automated/vie_api_integration_test.cc b/trunk/src/video_engine/test/auto_test/automated/vie_api_integration_test.cc
deleted file mode 100644
index a2c65a2..0000000
--- a/trunk/src/video_engine/test/auto_test/automated/vie_api_integration_test.cc
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/**
- * Runs "extended" integration tests.
- */
-
-#include "gtest/gtest.h"
-#include "legacy_fixture.h"
-#include "vie_autotest.h"
-
-namespace {
-
-class ViEApiIntegrationTest : public LegacyFixture {
-};
-
-TEST_F(ViEApiIntegrationTest, RunsBaseTestWithoutErrors) {
- tests_->ViEBaseAPITest();
-}
-
-TEST_F(ViEApiIntegrationTest, RunsCaptureTestWithoutErrors) {
- tests_->ViECaptureAPITest();
-}
-
-TEST_F(ViEApiIntegrationTest, RunsCodecTestWithoutErrors) {
- tests_->ViECodecAPITest();
-}
-
-TEST_F(ViEApiIntegrationTest, RunsEncryptionTestWithoutErrors) {
- tests_->ViEEncryptionAPITest();
-}
-
-TEST_F(ViEApiIntegrationTest, RunsFileTestWithoutErrors) {
- tests_->ViEFileAPITest();
-}
-
-TEST_F(ViEApiIntegrationTest, RunsImageProcessTestWithoutErrors) {
- tests_->ViEImageProcessAPITest();
-}
-
-TEST_F(ViEApiIntegrationTest, RunsNetworkTestWithoutErrors) {
- tests_->ViENetworkAPITest();
-}
-
-TEST_F(ViEApiIntegrationTest, RunsRenderTestWithoutErrors) {
- tests_->ViERenderAPITest();
-}
-
-TEST_F(ViEApiIntegrationTest, RunsRtpRtcpTestWithoutErrors) {
- tests_->ViERtpRtcpAPITest();
-}
-
-} // namespace
diff --git a/trunk/src/video_engine/test/auto_test/automated/vie_extended_integration_test.cc b/trunk/src/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
deleted file mode 100644
index facb659..0000000
--- a/trunk/src/video_engine/test/auto_test/automated/vie_extended_integration_test.cc
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/**
- * Runs "extended" integration tests.
- */
-
-#include "gtest/gtest.h"
-#include "legacy_fixture.h"
-#include "vie_autotest.h"
-
-namespace {
-
-class ViEExtendedIntegrationTest : public LegacyFixture {
-};
-
-TEST_F(ViEExtendedIntegrationTest, RunsBaseTestWithoutErrors) {
- tests_->ViEBaseExtendedTest();
-}
-
-TEST_F(ViEExtendedIntegrationTest, RunsCaptureTestWithoutErrors) {
- tests_->ViECaptureExtendedTest();
-}
-
-TEST_F(ViEExtendedIntegrationTest, RunsCodecTestWithoutErrors) {
- tests_->ViECodecExtendedTest();
-}
-
-TEST_F(ViEExtendedIntegrationTest, RunsEncryptionTestWithoutErrors) {
- tests_->ViEEncryptionExtendedTest();
-}
-
-TEST_F(ViEExtendedIntegrationTest, RunsFileTestWithoutErrors) {
- tests_->ViEFileExtendedTest();
-}
-
-TEST_F(ViEExtendedIntegrationTest, RunsImageProcessTestWithoutErrors) {
- tests_->ViEImageProcessExtendedTest();
-}
-
-TEST_F(ViEExtendedIntegrationTest, RunsNetworkTestWithoutErrors) {
- tests_->ViENetworkExtendedTest();
-}
-
-TEST_F(ViEExtendedIntegrationTest, RunsRenderTestWithoutErrors) {
- tests_->ViERenderExtendedTest();
-}
-
-TEST_F(ViEExtendedIntegrationTest, RunsRtpRtcpTestWithoutErrors) {
- tests_->ViERtpRtcpExtendedTest();
-}
-
-} // namespace
diff --git a/trunk/src/video_engine/test/auto_test/automated/vie_rtp_fuzz_test.cc b/trunk/src/video_engine/test/auto_test/automated/vie_rtp_fuzz_test.cc
deleted file mode 100644
index f8740be..0000000
--- a/trunk/src/video_engine/test/auto_test/automated/vie_rtp_fuzz_test.cc
+++ /dev/null
@@ -1,150 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <ctime>
-
-#include "gtest/gtest.h"
-#include "gflags/gflags.h"
-#include "video_engine/test/auto_test/automated/two_windows_fixture.h"
-#include "video_engine/test/auto_test/helpers/bit_flip_encryption.h"
-#include "video_engine/test/auto_test/helpers/random_encryption.h"
-#include "video_engine/test/auto_test/helpers/vie_window_creator.h"
-#include "video_engine/test/auto_test/interface/tb_capture_device.h"
-#include "video_engine/test/auto_test/interface/tb_interfaces.h"
-#include "video_engine/test/auto_test/interface/tb_video_channel.h"
-#include "video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h"
-#include "video_engine/test/auto_test/primitives/general_primitives.h"
-
-namespace {
-
-DEFINE_int32(rtp_fuzz_test_rand_seed, 0, "The rand seed to use for "
- "the RTP fuzz test. Defaults to time(). 0 cannot be specified.");
-
-class ViERtpFuzzTest : public TwoWindowsFixture {
- protected:
- TbVideoChannel* video_channel_;
- TbInterfaces* video_engine_;
- TbCaptureDevice* capture_device_;
-
- void SetUp() {
- video_engine_ = new TbInterfaces(
- "ViERtpTryInjectingRandomPacketsIntoRtpStream");
- video_channel_ = new TbVideoChannel(
- *video_engine_, webrtc::kVideoCodecVP8);
- capture_device_ = new TbCaptureDevice(*video_engine_);
-
- capture_device_->ConnectTo(video_channel_->videoChannel);
-
- // Enable PLI RTCP, which will allow the video engine to recover better.
- video_engine_->rtp_rtcp->SetKeyFrameRequestMethod(
- video_channel_->videoChannel, webrtc::kViEKeyFrameRequestPliRtcp);
-
- video_channel_->StartReceive();
- video_channel_->StartSend();
-
- RenderInWindow(
- video_engine_->render, capture_device_->captureId, window_1_, 0);
- RenderInWindow(
- video_engine_->render, video_channel_->videoChannel, window_2_, 1);
- }
-
- void TearDown() {
- delete capture_device_;
- delete video_channel_;
- delete video_engine_;
- }
-
- unsigned int FetchRandSeed() {
- if (FLAGS_rtp_fuzz_test_rand_seed != 0) {
- return FLAGS_rtp_fuzz_test_rand_seed;
- }
- return std::time(NULL);
- }
-
- // Pass in a number [0, 1] which will be the bit flip probability per byte.
- void BitFlipFuzzTest(float flip_probability) {
- unsigned int rand_seed = FetchRandSeed();
- ViETest::Log("Running test with rand seed %d.", rand_seed);
-
- ViETest::Log("Running as usual. You should see video output.");
- AutoTestSleep(2000);
-
- ViETest::Log("Starting to flip bits in packets (%f%% chance per byte).",
- flip_probability * 100);
- BitFlipEncryption bit_flip_encryption(rand_seed, flip_probability);
- video_engine_->encryption->RegisterExternalEncryption(
- video_channel_->videoChannel, bit_flip_encryption);
-
- AutoTestSleep(5000);
-
- ViETest::Log("Back to normal. Flipped %d bits.",
- bit_flip_encryption.flip_count());
- video_engine_->encryption->DeregisterExternalEncryption(
- video_channel_->videoChannel);
-
- AutoTestSleep(5000);
- }
-};
-
-TEST_F(ViERtpFuzzTest, VideoEngineDealsWithASmallNumberOfTamperedPackets) {
- // Try 0.005% bit flip chance per byte.
- BitFlipFuzzTest(0.00005f);
-}
-
-TEST_F(ViERtpFuzzTest, VideoEngineDealsWithAMediumNumberOfTamperedPackets) {
- // Try 0.05% bit flip chance per byte.
- BitFlipFuzzTest(0.0005f);
-}
-
-TEST_F(ViERtpFuzzTest, VideoEngineDealsWithALargeNumberOfTamperedPackets) {
- // Try 0.5% bit flip chance per byte.
- BitFlipFuzzTest(0.005f);
-}
-
-TEST_F(ViERtpFuzzTest, VideoEngineDealsWithAVeryLargeNumberOfTamperedPackets) {
- // Try 5% bit flip chance per byte.
- BitFlipFuzzTest(0.05f);
-}
-
-TEST_F(ViERtpFuzzTest,
- VideoEngineDealsWithAExtremelyLargeNumberOfTamperedPackets) {
- // Try 25% bit flip chance per byte (madness!)
- BitFlipFuzzTest(0.25f);
-}
-
-TEST_F(ViERtpFuzzTest, VideoEngineDealsWithSeveralPeriodsOfTamperedPackets) {
- // Try 0.05% bit flip chance per byte.
- BitFlipFuzzTest(0.0005f);
- BitFlipFuzzTest(0.0005f);
- BitFlipFuzzTest(0.0005f);
-}
-
-TEST_F(ViERtpFuzzTest, VideoEngineRecoversAfterSomeCompletelyRandomPackets) {
- unsigned int rand_seed = FetchRandSeed();
- ViETest::Log("Running test with rand seed %d.", rand_seed);
-
- ViETest::Log("Running as usual. You should see video output.");
- AutoTestSleep(2000);
-
- ViETest::Log("Injecting completely random packets...");
- RandomEncryption random_encryption(rand_seed);
- video_engine_->encryption->RegisterExternalEncryption(
- video_channel_->videoChannel, random_encryption);
-
- AutoTestSleep(5000);
-
- ViETest::Log("Back to normal.");
- video_engine_->encryption->DeregisterExternalEncryption(
- video_channel_->videoChannel);
-
- AutoTestSleep(5000);
-}
-
-}
diff --git a/trunk/src/video_engine/test/auto_test/automated/vie_standard_integration_test.cc b/trunk/src/video_engine/test/auto_test/automated/vie_standard_integration_test.cc
deleted file mode 100644
index e85b6e4..0000000
--- a/trunk/src/video_engine/test/auto_test/automated/vie_standard_integration_test.cc
+++ /dev/null
@@ -1,71 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains the "standard" suite of integration tests, implemented
- * as a GUnit test. This file is a part of the effort to try to automate all
- * tests in this section of the code. Currently, this code makes no attempt
- * to verify any video output - it only checks for direct errors.
- */
-
-#include <cstdio>
-
-#include "gflags/gflags.h"
-#include "gtest/gtest.h"
-#include "legacy_fixture.h"
-#include "testsupport/metrics/video_metrics.h"
-#include "vie_autotest.h"
-#include "vie_autotest_window_manager_interface.h"
-#include "vie_to_file_renderer.h"
-#include "vie_window_creator.h"
-#include "testsupport/metrics/video_metrics.h"
-
-namespace {
-
-class ViEStandardIntegrationTest : public LegacyFixture {
-};
-
-TEST_F(ViEStandardIntegrationTest, RunsBaseTestWithoutErrors) {
- tests_->ViEBaseStandardTest();
-}
-
-TEST_F(ViEStandardIntegrationTest, RunsCodecTestWithoutErrors) {
- tests_->ViECodecStandardTest();
-}
-
-TEST_F(ViEStandardIntegrationTest, RunsCaptureTestWithoutErrors) {
- tests_->ViECaptureStandardTest();
-}
-
-TEST_F(ViEStandardIntegrationTest, RunsEncryptionTestWithoutErrors) {
- tests_->ViEEncryptionStandardTest();
-}
-
-TEST_F(ViEStandardIntegrationTest, RunsFileTestWithoutErrors) {
- tests_->ViEFileStandardTest();
-}
-
-TEST_F(ViEStandardIntegrationTest, RunsImageProcessTestWithoutErrors) {
- tests_->ViEImageProcessStandardTest();
-}
-
-TEST_F(ViEStandardIntegrationTest, RunsNetworkTestWithoutErrors) {
- tests_->ViENetworkStandardTest();
-}
-
-TEST_F(ViEStandardIntegrationTest, RunsRenderTestWithoutErrors) {
- tests_->ViERenderStandardTest();
-}
-
-TEST_F(ViEStandardIntegrationTest, RunsRtpRtcpTestWithoutErrors) {
- tests_->ViERtpRtcpStandardTest();
-}
-
-} // namespace
diff --git a/trunk/src/video_engine/test/auto_test/automated/vie_video_verification_test.cc b/trunk/src/video_engine/test/auto_test/automated/vie_video_verification_test.cc
deleted file mode 100644
index 861af8a..0000000
--- a/trunk/src/video_engine/test/auto_test/automated/vie_video_verification_test.cc
+++ /dev/null
@@ -1,253 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <vector>
-
-#include "gflags/gflags.h"
-#include "gtest/gtest.h"
-#include "testsupport/fileutils.h"
-#include "testsupport/metrics/video_metrics.h"
-#include "video_engine/test/auto_test/helpers/vie_to_file_renderer.h"
-#include "video_engine/test/auto_test/interface/vie_autotest.h"
-#include "video_engine/test/auto_test/interface/vie_file_based_comparison_tests.h"
-#include "video_engine/test/auto_test/primitives/framedrop_primitives.h"
-
-namespace {
-
-// The input file must be QCIF since I420 gets scaled to that in the tests
-// (it is so bandwidth-heavy we have no choice). Our comparison algorithms
-// wouldn't like scaling, so this will work when we compare with the original.
-const int kInputWidth = 176;
-const int kInputHeight = 144;
-const int kVerifyingTestMaxNumAttempts = 3;
-
-class ViEVideoVerificationTest : public testing::Test {
- protected:
- void SetUp() {
- input_file_ = webrtc::test::ResourcePath("paris_qcif", "yuv");
- }
-
- void TearDown() {
- TearDownFileRenderer(local_file_renderer_);
- TearDownFileRenderer(remote_file_renderer_);
- }
-
- void InitializeFileRenderers() {
- local_file_renderer_ = new ViEToFileRenderer();
- remote_file_renderer_ = new ViEToFileRenderer();
- SetUpLocalFileRenderer(local_file_renderer_);
- SetUpRemoteFileRenderer(remote_file_renderer_);
- }
-
- void SetUpLocalFileRenderer(ViEToFileRenderer* file_renderer) {
- SetUpFileRenderer(file_renderer, "-local-preview.yuv");
- }
-
- void SetUpRemoteFileRenderer(ViEToFileRenderer* file_renderer) {
- SetUpFileRenderer(file_renderer, "-remote.yuv");
- }
-
- // Must be called manually inside the tests.
- void StopRenderers() {
- local_file_renderer_->StopRendering();
- remote_file_renderer_->StopRendering();
- }
-
- void TearDownFileRenderer(ViEToFileRenderer* file_renderer) {
- assert(file_renderer);
- bool test_failed = ::testing::UnitTest::GetInstance()->
- current_test_info()->result()->Failed();
- if (test_failed) {
- // Leave the files for analysis if the test failed.
- file_renderer->SaveOutputFile("failed-");
- } else {
- // No reason to keep the files if we succeeded.
- file_renderer->DeleteOutputFile();
- }
- delete file_renderer;
- }
-
- void CompareFiles(const std::string& reference_file,
- const std::string& test_file,
- double* psnr_result, double *ssim_result) {
- webrtc::test::QualityMetricsResult psnr;
- int error = I420PSNRFromFiles(reference_file.c_str(), test_file.c_str(),
- kInputWidth, kInputHeight, &psnr);
-
- EXPECT_EQ(0, error) << "PSNR routine failed - output files missing?";
- *psnr_result = psnr.average;
-
- webrtc::test::QualityMetricsResult ssim;
- error = I420SSIMFromFiles(reference_file.c_str(), test_file.c_str(),
- kInputWidth, kInputHeight, &ssim);
- EXPECT_EQ(0, error) << "SSIM routine failed - output files missing?";
- *ssim_result = ssim.average;
-
- ViETest::Log("Results: PSNR is %f (dB), SSIM is %f (1 is perfect)",
- psnr.average, ssim.average);
- }
-
- std::string input_file_;
- ViEToFileRenderer* local_file_renderer_;
- ViEToFileRenderer* remote_file_renderer_;
- ViEFileBasedComparisonTests tests_;
-
- private:
- void SetUpFileRenderer(ViEToFileRenderer* file_renderer,
- const std::string& suffix) {
- std::string test_case_name =
- ::testing::UnitTest::GetInstance()->current_test_info()->name();
-
- std::string output_path = ViETest::GetResultOutputPath();
- std::string filename = test_case_name + suffix;
-
- if (!file_renderer->PrepareForRendering(output_path, filename)) {
- FAIL() << "Could not open output file " << filename <<
- " for writing.";
- }
- }
-};
-
-TEST_F(ViEVideoVerificationTest, RunsBaseStandardTestWithoutErrors) {
- // The I420 test should give pretty good values since it's a lossless codec
- // running on the default bitrate. It should average about 30 dB but there
- // may be cases where it dips as low as 26 under adverse conditions. That's
- // why we have a retrying mechanism in place for this test.
- const double kExpectedMinimumPSNR = 30;
- const double kExpectedMinimumSSIM = 0.95;
-
- for (int attempt = 0; attempt < kVerifyingTestMaxNumAttempts; attempt++) {
- InitializeFileRenderers();
- ASSERT_TRUE(tests_.TestCallSetup(input_file_, kInputWidth, kInputHeight,
- local_file_renderer_,
- remote_file_renderer_));
- std::string output_file = remote_file_renderer_->GetFullOutputPath();
- StopRenderers();
-
- double actual_psnr = 0;
- double actual_ssim = 0;
- CompareFiles(input_file_, output_file, &actual_psnr, &actual_ssim);
-
- if (actual_psnr >= kExpectedMinimumPSNR &&
- actual_ssim >= kExpectedMinimumSSIM) {
- // Test succeeded!
- return;
- }
- }
-
- ADD_FAILURE() << "Failed to achieve PSNR " << kExpectedMinimumPSNR <<
- " and SSIM " << kExpectedMinimumSSIM << " after " <<
- kVerifyingTestMaxNumAttempts << " attempts.";
-}
-
-TEST_F(ViEVideoVerificationTest, RunsCodecTestWithoutErrors) {
- // We compare the local and remote here instead of with the original.
- // The reason is that it is hard to say when the three consecutive tests
- // switch over into each other, at which point we would have to restart the
- // original to get a fair comparison.
- //
- // The PSNR and SSIM values are quite low here, and they have to be since
- // the codec switches will lead to lag in the output. This is considered
- // acceptable, but it probably shouldn't get worse than this.
- const double kExpectedMinimumPSNR = 20;
- const double kExpectedMinimumSSIM = 0.7;
-
- for (int attempt = 0; attempt < kVerifyingTestMaxNumAttempts; attempt++) {
- InitializeFileRenderers();
- ASSERT_TRUE(tests_.TestCodecs(input_file_, kInputWidth, kInputHeight,
- local_file_renderer_,
- remote_file_renderer_));
- std::string reference_file = local_file_renderer_->GetFullOutputPath();
- std::string output_file = remote_file_renderer_->GetFullOutputPath();
- StopRenderers();
-
- double actual_psnr = 0;
- double actual_ssim = 0;
- CompareFiles(reference_file, output_file, &actual_psnr, &actual_ssim);
-
- if (actual_psnr >= kExpectedMinimumPSNR &&
- actual_ssim >= kExpectedMinimumSSIM) {
- // Test succeeded!
- return;
- }
- }
-}
-
-// Runs a whole stack processing with tracking of which frames are dropped
-// in the encoder. The local and remote file will not be of equal size because
-// of unknown reasons. Tests show that they start at the same frame, which is
-// the important thing when doing frame-to-frame comparison with PSNR/SSIM.
-TEST_F(ViEVideoVerificationTest, RunsFullStackWithoutErrors) {
- FrameDropDetector detector;
- local_file_renderer_ = new ViEToFileRenderer();
- remote_file_renderer_ = new FrameDropMonitoringRemoteFileRenderer(&detector);
- SetUpLocalFileRenderer(local_file_renderer_);
- SetUpRemoteFileRenderer(remote_file_renderer_);
-
- // Set a low bit rate so the encoder budget will be tight, causing it to drop
- // frames every now and then.
- const int kBitRateKbps = 50;
- const int kPacketLossPercent = 10;
- const int kNetworkDelayMs = 100;
- ViETest::Log("Bit rate : %5d kbps", kBitRateKbps);
- ViETest::Log("Packet loss : %5d %%", kPacketLossPercent);
- ViETest::Log("Network delay: %5d ms", kNetworkDelayMs);
- tests_.TestFullStack(input_file_, kInputWidth, kInputHeight, kBitRateKbps,
- kPacketLossPercent, kNetworkDelayMs,
- local_file_renderer_, remote_file_renderer_, &detector);
- const std::string reference_file = local_file_renderer_->GetFullOutputPath();
- const std::string output_file = remote_file_renderer_->GetFullOutputPath();
- StopRenderers();
-
- detector.CalculateResults();
- detector.PrintReport();
-
- if (detector.GetNumberOfFramesDroppedAt(FrameDropDetector::kRendered) !=
- detector.GetNumberOfFramesDroppedAt(FrameDropDetector::kDecoded)) {
- detector.PrintDebugDump();
- }
-
- ASSERT_EQ(detector.GetNumberOfFramesDroppedAt(FrameDropDetector::kRendered),
- detector.GetNumberOfFramesDroppedAt(FrameDropDetector::kDecoded))
- << "The number of dropped frames on the decode and render steps are not "
- "equal. This may be because we have a major problem in the buffers of "
- "the ViEToFileRenderer?";
-
- // We may have dropped frames during the processing, which means the output
- // file does not contain all the frames that are present in the input file.
- // To make the quality measurement correct, we must adjust the output file to
- // that by copying the last successful frame into the place where the dropped
- // frame would be, for all dropped frames.
- const int frame_length_in_bytes = 3 * kInputHeight * kInputWidth / 2;
- ViETest::Log("Frame length: %d bytes", frame_length_in_bytes);
- std::vector<Frame*> all_frames = detector.GetAllFrames();
- FixOutputFileForComparison(output_file, frame_length_in_bytes, all_frames);
-
- // Verify all sent frames are present in the output file.
- size_t output_file_size = webrtc::test::GetFileSize(output_file);
- EXPECT_EQ(all_frames.size(), output_file_size / frame_length_in_bytes)
- << "The output file size is incorrect. It should be equal to the number "
- "of frames multiplied by the frame size. This will likely affect "
- "PSNR/SSIM calculations in a bad way.";
-
- // We are running on a lower bitrate here so we need to settle for somewhat
- // lower PSNR and SSIM values.
- double actual_psnr = 0;
- double actual_ssim = 0;
- CompareFiles(reference_file, output_file, &actual_psnr, &actual_ssim);
-
- const double kExpectedMinimumPSNR = 24;
- const double kExpectedMinimumSSIM = 0.7;
-
- EXPECT_GE(actual_psnr, kExpectedMinimumPSNR);
- EXPECT_GE(actual_ssim, kExpectedMinimumSSIM);
-}
-
-} // namespace
diff --git a/trunk/src/video_engine/test/auto_test/helpers/bit_flip_encryption.cc b/trunk/src/video_engine/test/auto_test/helpers/bit_flip_encryption.cc
deleted file mode 100644
index 423ad5a..0000000
--- a/trunk/src/video_engine/test/auto_test/helpers/bit_flip_encryption.cc
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/test/auto_test/helpers/bit_flip_encryption.h"
-
-#include <cstdlib>
-
-#include "video_engine/test/auto_test/interface/vie_autotest_defines.h"
-
-float NormalizedRand() {
- return static_cast<float>(rand()) /
- static_cast<float>(RAND_MAX);
-}
-
-BitFlipEncryption::BitFlipEncryption(unsigned int rand_seed,
- float flip_probability)
- : flip_probability_(flip_probability),
- flip_count_(0) {
- srand(rand_seed);
-}
-
-void BitFlipEncryption::FlipSomeBitsInData(const unsigned char* in_data,
- unsigned char* out_data,
- int bytes_in, int* bytes_out) {
- for (int i = 0; i < bytes_in; i++) {
- out_data[i] = in_data[i];
-
- if (NormalizedRand() < flip_probability_) {
- int bit_to_flip = rand() % 8;
- out_data[i] ^= 1 << bit_to_flip;
- flip_count_++;
- }
- }
- *bytes_out = bytes_in;
-}
diff --git a/trunk/src/video_engine/test/auto_test/helpers/bit_flip_encryption.h b/trunk/src/video_engine/test/auto_test/helpers/bit_flip_encryption.h
deleted file mode 100644
index ccc6974..0000000
--- a/trunk/src/video_engine/test/auto_test/helpers/bit_flip_encryption.h
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_BIT_FLIP_ENCRYPTION_H_
-#define SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_BIT_FLIP_ENCRYPTION_H_
-
-#include "video_engine/include/vie_encryption.h"
-
-// This encryption scheme will randomly flip bits every now and then in the
-// input data.
-class BitFlipEncryption : public webrtc::Encryption {
- public:
- // Args:
- // rand_seed: the seed to initialize the test's random generator with.
- // flip_probability: A number [0, 1] which is the percentage chance a bit
- // gets flipped in a particular byte.
- BitFlipEncryption(unsigned int rand_seed, float flip_probability);
-
- virtual void encrypt(int channel_no, unsigned char* in_data,
- unsigned char* out_data, int bytes_in, int* bytes_out) {
- FlipSomeBitsInData(in_data, out_data, bytes_in, bytes_out);
- }
-
- virtual void decrypt(int channel_no, unsigned char* in_data,
- unsigned char* out_data, int bytes_in, int* bytes_out) {
- FlipSomeBitsInData(in_data, out_data, bytes_in, bytes_out);
- }
-
- virtual void encrypt_rtcp(int channel_no, unsigned char* in_data,
- unsigned char* out_data, int bytes_in,
- int* bytes_out) {
- FlipSomeBitsInData(in_data, out_data, bytes_in, bytes_out);
- }
-
- virtual void decrypt_rtcp(int channel_no, unsigned char* in_data,
- unsigned char* out_data, int bytes_in,
- int* bytes_out) {
- FlipSomeBitsInData(in_data, out_data, bytes_in, bytes_out);
- }
-
- int64_t flip_count() const { return flip_count_; }
-
- private:
- // The flip probability ([0, 1]).
- float flip_probability_;
- // The number of bits we've flipped so far.
- int64_t flip_count_;
-
- // Flips some bits in the data at random.
- void FlipSomeBitsInData(const unsigned char *in_data, unsigned char* out_data,
- int bytes_in, int* bytes_out);
-};
-
-#endif // SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_BIT_FLIP_ENCRYPTION_H_
diff --git a/trunk/src/video_engine/test/auto_test/helpers/random_encryption.cc b/trunk/src/video_engine/test/auto_test/helpers/random_encryption.cc
deleted file mode 100644
index 04904b5..0000000
--- a/trunk/src/video_engine/test/auto_test/helpers/random_encryption.cc
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/test/auto_test/helpers/random_encryption.h"
-
-#include <algorithm>
-#include <cstdlib>
-#include <cmath>
-
-#include "video_engine/vie_defines.h"
-
-static int Saturate(int value, int min, int max) {
- return std::min(std::max(value, min), max);
-}
-
-RandomEncryption::RandomEncryption(unsigned int rand_seed) {
- srand(rand_seed);
-}
-
-// Generates some completely random data with roughly the right length.
-void RandomEncryption::GenerateRandomData(unsigned char* out_data, int bytes_in,
- int* bytes_out) {
- int out_length = MakeUpSimilarLength(bytes_in);
- for (int i = 0; i < out_length; i++) {
- // The modulo will skew the random distribution a bit, but I think it
- // will be random enough.
- out_data[i] = static_cast<unsigned char>(rand() % 256);
- }
- *bytes_out = out_length;
-}
-
-// Makes up a length within +- 50 of the original length, without
-// overstepping the contract for encrypt / decrypt.
-int RandomEncryption::MakeUpSimilarLength(int original_length) {
- int sign = rand() - RAND_MAX / 2;
- int length = original_length + sign * rand() % 50;
-
- return Saturate(length, 0, static_cast<int>(webrtc::kViEMaxMtu));
-}
diff --git a/trunk/src/video_engine/test/auto_test/helpers/random_encryption.h b/trunk/src/video_engine/test/auto_test/helpers/random_encryption.h
deleted file mode 100644
index cbca844..0000000
--- a/trunk/src/video_engine/test/auto_test/helpers/random_encryption.h
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_RANDOM_ENCRYPTION_H_
-#define SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_RANDOM_ENCRYPTION_H_
-
-#include "video_engine/include/vie_encryption.h"
-
-// These algorithms attempt to create an uncrackable encryption
-// scheme by completely disregarding the input data.
-class RandomEncryption : public webrtc::Encryption {
- public:
- explicit RandomEncryption(unsigned int rand_seed);
-
- virtual void encrypt(int channel_no, unsigned char* in_data,
- unsigned char* out_data, int bytes_in, int* bytes_out) {
- GenerateRandomData(out_data, bytes_in, bytes_out);
- }
-
- virtual void decrypt(int channel_no, unsigned char* in_data,
- unsigned char* out_data, int bytes_in, int* bytes_out) {
- GenerateRandomData(out_data, bytes_in, bytes_out);
- }
-
- virtual void encrypt_rtcp(int channel_no, unsigned char* in_data,
- unsigned char* out_data, int bytes_in,
- int* bytes_out) {
- GenerateRandomData(out_data, bytes_in, bytes_out);
- }
-
- virtual void decrypt_rtcp(int channel_no, unsigned char* in_data,
- unsigned char* out_data, int bytes_in,
- int* bytes_out) {
- GenerateRandomData(out_data, bytes_in, bytes_out);
- }
-
- private:
- // Generates some completely random data with roughly the right length.
- void GenerateRandomData(unsigned char* out_data, int bytes_in,
- int* bytes_out);
-
- // Makes up a length within +- 50 of the original length, without
- // overstepping the contract for encrypt / decrypt.
- int MakeUpSimilarLength(int original_length);
-};
-
-#endif // SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_RANDOM_ENCRYPTION_H_
diff --git a/trunk/src/video_engine/test/auto_test/helpers/vie_fake_camera.cc b/trunk/src/video_engine/test/auto_test/helpers/vie_fake_camera.cc
deleted file mode 100644
index f7dae2f..0000000
--- a/trunk/src/video_engine/test/auto_test/helpers/vie_fake_camera.cc
+++ /dev/null
@@ -1,87 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-#include "vie_fake_camera.h"
-
-#include <assert.h>
-
-#include "vie_capture.h"
-#include "vie_file_capture_device.h"
-#include "thread_wrapper.h"
-
-// This callback runs the camera thread:
-bool StreamVideoFileRepeatedlyIntoCaptureDevice(void* data) {
- ViEFileCaptureDevice* file_capture_device =
- reinterpret_cast<ViEFileCaptureDevice*>(data);
-
- // We want to interrupt the camera feeding thread every now and then in order
- // to follow the contract for the system_wrappers thread library. 1.5 seconds
- // seems about right here.
- uint64_t time_slice_ms = 1500;
- uint32_t max_fps = 30;
-
- file_capture_device->ReadFileFor(time_slice_ms, max_fps);
-
- return true;
-}
-
-ViEFakeCamera::ViEFakeCamera(webrtc::ViECapture* capture_interface)
- : capture_interface_(capture_interface),
- capture_id_(-1),
- camera_thread_(NULL),
- file_capture_device_(NULL) {
-}
-
-ViEFakeCamera::~ViEFakeCamera() {
-}
-
-bool ViEFakeCamera::StartCameraInNewThread(
- const std::string& i420_test_video_path, int width, int height) {
-
- assert(file_capture_device_ == NULL && camera_thread_ == NULL);
-
- webrtc::ViEExternalCapture* externalCapture;
- int result = capture_interface_->
- AllocateExternalCaptureDevice(capture_id_, externalCapture);
- if (result != 0) {
- return false;
- }
-
- file_capture_device_ = new ViEFileCaptureDevice(externalCapture);
- if (!file_capture_device_->OpenI420File(i420_test_video_path,
- width,
- height)) {
- return false;
- }
-
- // Set up a thread which runs the fake camera. The capturer object is
- // thread-safe.
- camera_thread_ = webrtc::ThreadWrapper::CreateThread(
- StreamVideoFileRepeatedlyIntoCaptureDevice, file_capture_device_);
- unsigned int id;
- camera_thread_->Start(id);
-
- return true;
-}
-
-bool ViEFakeCamera::StopCamera() {
- assert(file_capture_device_ != NULL && camera_thread_ != NULL);
-
- camera_thread_->Stop();
- file_capture_device_->CloseFile();
-
- int result = capture_interface_->ReleaseCaptureDevice(capture_id_);
-
- delete camera_thread_;
- delete file_capture_device_;
- camera_thread_ = NULL;
- file_capture_device_ = NULL;
-
- return result == 0;
-}
diff --git a/trunk/src/video_engine/test/auto_test/helpers/vie_fake_camera.h b/trunk/src/video_engine/test/auto_test/helpers/vie_fake_camera.h
deleted file mode 100644
index 0b9c2e7..0000000
--- a/trunk/src/video_engine/test/auto_test/helpers/vie_fake_camera.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FAKE_CAMERA_H_
-#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FAKE_CAMERA_H_
-
-#include <string>
-
-namespace webrtc {
-class ViECapture;
-class ThreadWrapper;
-}
-
-class ViEFileCaptureDevice;
-
-// Registers an external capture device with the provided capture interface
-// and starts running a fake camera by reading frames from a file. The frame-
-// reading code runs in a separate thread which makes it possible to run tests
-// while the fake camera feeds data into the system. This class is not thread-
-// safe in itself (but handles its own thread in a safe manner).
-class ViEFakeCamera {
- public:
- // The argument is the capture interface to register with.
- explicit ViEFakeCamera(webrtc::ViECapture* capture_interface);
- virtual ~ViEFakeCamera();
-
- // Runs the scenario in the class comments.
- bool StartCameraInNewThread(const std::string& i420_test_video_path,
- int width,
- int height);
- // Stops the camera and cleans up everything allocated by the start method.
- bool StopCamera();
-
- int capture_id() const { return capture_id_; }
-
- private:
- webrtc::ViECapture* capture_interface_;
-
- int capture_id_;
- webrtc::ThreadWrapper* camera_thread_;
- ViEFileCaptureDevice* file_capture_device_;
-};
-
-#endif // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FAKE_CAMERA_H_
diff --git a/trunk/src/video_engine/test/auto_test/helpers/vie_file_capture_device.cc b/trunk/src/video_engine/test/auto_test/helpers/vie_file_capture_device.cc
deleted file mode 100644
index e4caa2a..0000000
--- a/trunk/src/video_engine/test/auto_test/helpers/vie_file_capture_device.cc
+++ /dev/null
@@ -1,106 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-#include "vie_file_capture_device.h"
-
-#include <assert.h>
-
-#include "common_types.h"
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "module_common_types.h"
-#include "vie_autotest_defines.h"
-#include "vie_capture.h"
-#include "tick_util.h"
-
-// This class ensures we are not exceeding the max FPS.
-class FramePacemaker {
- public:
- explicit FramePacemaker(uint32_t max_fps)
- : time_per_frame_ms_(1000 / max_fps) {
- frame_start_ = webrtc::TickTime::MillisecondTimestamp();
- }
-
- void SleepIfNecessary(webrtc::EventWrapper* sleeper) {
- uint64_t now = webrtc::TickTime::MillisecondTimestamp();
- if (now - frame_start_ < time_per_frame_ms_) {
- sleeper->Wait(time_per_frame_ms_ - (now - frame_start_));
- }
- }
-
- private:
- uint64_t frame_start_;
- uint64_t time_per_frame_ms_;
-};
-
-ViEFileCaptureDevice::ViEFileCaptureDevice(
- webrtc::ViEExternalCapture* input_sink)
- : input_sink_(input_sink),
- input_file_(NULL) {
- mutex_ = webrtc::CriticalSectionWrapper::CreateCriticalSection();
-}
-
-ViEFileCaptureDevice::~ViEFileCaptureDevice() {
- delete mutex_;
-}
-
-bool ViEFileCaptureDevice::OpenI420File(const std::string& path,
- int width,
- int height) {
- webrtc::CriticalSectionScoped cs(*mutex_);
- assert(input_file_ == NULL);
-
- input_file_ = std::fopen(path.c_str(), "rb");
- if (input_file_ == NULL) {
- return false;
- }
-
- frame_length_ = 3 * width * height / 2;
- width_ = width;
- height_ = height;
- return true;
-}
-
-void ViEFileCaptureDevice::ReadFileFor(uint64_t time_slice_ms,
- uint32_t max_fps) {
- webrtc::CriticalSectionScoped cs(*mutex_);
- assert(input_file_ != NULL);
-
- unsigned char* frame_buffer = new unsigned char[frame_length_];
-
- webrtc::EventWrapper* sleeper = webrtc::EventWrapper::Create();
-
- uint64_t start_time_ms = webrtc::TickTime::MillisecondTimestamp();
- uint64_t elapsed_ms = 0;
-
- while (elapsed_ms < time_slice_ms) {
- FramePacemaker pacemaker(max_fps);
- int read = std::fread(frame_buffer, 1, frame_length_, input_file_);
-
- if (std::feof(input_file_)) {
- std::rewind(input_file_);
- }
- input_sink_->IncomingFrame(frame_buffer, read, width_, height_,
- webrtc::kVideoI420,
- webrtc::TickTime::MillisecondTimestamp());
-
- pacemaker.SleepIfNecessary(sleeper);
- elapsed_ms = webrtc::TickTime::MillisecondTimestamp() - start_time_ms;
- }
-
- delete sleeper;
- delete[] frame_buffer;
-}
-
-void ViEFileCaptureDevice::CloseFile() {
- webrtc::CriticalSectionScoped cs(*mutex_);
- assert(input_file_ != NULL);
-
- std::fclose(input_file_);
-}
diff --git a/trunk/src/video_engine/test/auto_test/helpers/vie_file_capture_device.h b/trunk/src/video_engine/test/auto_test/helpers/vie_file_capture_device.h
deleted file mode 100644
index 5e62c48..0000000
--- a/trunk/src/video_engine/test/auto_test/helpers/vie_file_capture_device.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FILE_CAPTURE_DEVICE_H_
-#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FILE_CAPTURE_DEVICE_H_
-
-#include <cstdio>
-
-#include <string>
-
-#include "typedefs.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-class ViEExternalCapture;
-}
-
-// This class opens a i420 file and feeds it into a ExternalCapture instance,
-// thereby acting as a faked capture device with deterministic input.
-class ViEFileCaptureDevice {
- public:
- // The input sink is where to send the I420 video frames.
- explicit ViEFileCaptureDevice(webrtc::ViEExternalCapture* input_sink);
- virtual ~ViEFileCaptureDevice();
-
- // Opens the provided I420 file and interprets it according to the provided
- // width and height. Returns false if the file doesn't exist.
- bool OpenI420File(const std::string& path, int width, int height);
-
- // Reads the previously opened file for at most time_slice_ms milliseconds,
- // after which it will return. It will make sure to sleep accordingly so we
- // do not send more than max_fps cap (we may send less, though).
- void ReadFileFor(uint64_t time_slice_ms, uint32_t max_fps);
-
- // Closes the opened input file.
- void CloseFile();
-
- private:
- webrtc::ViEExternalCapture* input_sink_;
-
- std::FILE* input_file_;
- webrtc::CriticalSectionWrapper* mutex_;
-
- WebRtc_UWord32 frame_length_;
- WebRtc_UWord8* frame_buffer_;
- WebRtc_UWord32 width_;
- WebRtc_UWord32 height_;
-};
-
-#endif // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_FILE_CAPTURE_DEVICE_H_
diff --git a/trunk/src/video_engine/test/auto_test/helpers/vie_to_file_renderer.cc b/trunk/src/video_engine/test/auto_test/helpers/vie_to_file_renderer.cc
deleted file mode 100644
index d9c8da3..0000000
--- a/trunk/src/video_engine/test/auto_test/helpers/vie_to_file_renderer.cc
+++ /dev/null
@@ -1,94 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vie_to_file_renderer.h"
-
-#include <assert.h>
-
-ViEToFileRenderer::ViEToFileRenderer()
- : output_file_(NULL) {
-}
-
-ViEToFileRenderer::~ViEToFileRenderer() {
-}
-
-bool ViEToFileRenderer::PrepareForRendering(
- const std::string& output_path,
- const std::string& output_filename) {
-
- assert(output_file_ == NULL);
-
- output_file_ = std::fopen((output_path + output_filename).c_str(), "wb");
- if (output_file_ == NULL) {
- return false;
- }
-
- output_filename_ = output_filename;
- output_path_ = output_path;
- return true;
-}
-
-void ViEToFileRenderer::StopRendering() {
- assert(output_file_ != NULL);
- std::fclose(output_file_);
- output_file_ = NULL;
-}
-
-bool ViEToFileRenderer::SaveOutputFile(const std::string& prefix) {
- assert(output_file_ == NULL && output_filename_ != "");
- if (std::rename((output_path_ + output_filename_).c_str(),
- (output_path_ + prefix + output_filename_).c_str()) != 0) {
- std::perror("Failed to rename output file");
- return false;
- }
- ForgetOutputFile();
- return true;
-}
-
-bool ViEToFileRenderer::DeleteOutputFile() {
- assert(output_file_ == NULL && output_filename_ != "");
- if (std::remove((output_path_ + output_filename_).c_str()) != 0) {
- std::perror("Failed to delete output file");
- return false;
- }
- ForgetOutputFile();
- return true;
-}
-
-const std::string ViEToFileRenderer::GetFullOutputPath() const {
- return output_path_ + output_filename_;
-}
-
-void ViEToFileRenderer::ForgetOutputFile() {
- output_filename_ = "";
- output_path_ = "";
-}
-
-int ViEToFileRenderer::DeliverFrame(unsigned char *buffer,
- int buffer_size,
- uint32_t time_stamp,
- int64_t render_time) {
- assert(output_file_ != NULL);
-
- int written = std::fwrite(buffer, sizeof(unsigned char),
- buffer_size, output_file_);
-
- if (written == buffer_size) {
- return 0;
- } else {
- return -1;
- }
-}
-
-int ViEToFileRenderer::FrameSizeChange(unsigned int width,
- unsigned int height,
- unsigned int number_of_streams) {
- return 0;
-}
diff --git a/trunk/src/video_engine/test/auto_test/helpers/vie_to_file_renderer.h b/trunk/src/video_engine/test/auto_test/helpers/vie_to_file_renderer.h
deleted file mode 100644
index 08559af..0000000
--- a/trunk/src/video_engine/test/auto_test/helpers/vie_to_file_renderer.h
+++ /dev/null
@@ -1,59 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_VIE_TO_FILE_RENDERER_H_
-#define SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_VIE_TO_FILE_RENDERER_H_
-
-#include <cstdio>
-#include <string>
-
-#include "video_engine/include/vie_render.h"
-
-class ViEToFileRenderer: public webrtc::ExternalRenderer {
- public:
- ViEToFileRenderer();
- virtual ~ViEToFileRenderer();
-
- // Returns false if we fail opening the output filename for writing.
- bool PrepareForRendering(const std::string& output_path,
- const std::string& output_filename);
-
- // Closes the output file.
- void StopRendering();
-
- // Deletes the closed output file from the file system. This is one option
- // after calling StopRendering, the other being KeepOutputFile. This file
- // renderer will forget about the file after this call and can be used again.
- bool DeleteOutputFile();
-
- // Renames the closed output file to its previous name with the provided
- // prefix prepended. This file renderer will forget about the file after this
- // call and can be used again.
- bool SaveOutputFile(const std::string& prefix);
-
- // Implementation of ExternalRenderer:
- int FrameSizeChange(unsigned int width, unsigned int height,
- unsigned int number_of_streams);
-
- int DeliverFrame(unsigned char* buffer, int buffer_size,
- uint32_t time_stamp,
- int64_t render_time);
-
- const std::string GetFullOutputPath() const;
-
- private:
- void ForgetOutputFile();
-
- std::FILE* output_file_;
- std::string output_path_;
- std::string output_filename_;
-};
-
-#endif // SRC_VIDEO_ENGINE_TEST_AUTO_TEST_HELPERS_VIE_TO_FILE_RENDERER_H_
diff --git a/trunk/src/video_engine/test/auto_test/helpers/vie_window_creator.cc b/trunk/src/video_engine/test/auto_test/helpers/vie_window_creator.cc
deleted file mode 100644
index 40ec0bb..0000000
--- a/trunk/src/video_engine/test/auto_test/helpers/vie_window_creator.cc
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vie_window_creator.h"
-
-#include "vie_autotest_main.h"
-#include "vie_codec.h"
-#include "voe_codec.h"
-#include "vie_window_manager_factory.h"
-#include "vie_autotest_window_manager_interface.h"
-
-#if defined(WIN32)
-#include <tchar.h>
-#endif
-
-ViEWindowCreator::ViEWindowCreator() {
- window_manager_ =
- ViEWindowManagerFactory::CreateWindowManagerForCurrentPlatform();
-}
-
-ViEWindowCreator::~ViEWindowCreator() {
- delete window_manager_;
-}
-
-ViEAutoTestWindowManagerInterface*
- ViEWindowCreator::CreateTwoWindows() {
-#if defined(WIN32)
- TCHAR window1Title[1024] = _T("ViE Autotest Window 1");
- TCHAR window2Title[1024] = _T("ViE Autotest Window 2");
-#else
- char window1Title[1024] = "ViE Autotest Window 1";
- char window2Title[1024] = "ViE Autotest Window 2";
-#endif
-
- AutoTestRect window1Size(352, 288, 600, 100);
- AutoTestRect window2Size(352, 288, 1000, 100);
- window_manager_->CreateWindows(window1Size, window2Size, window1Title,
- window2Title);
- window_manager_->SetTopmostWindow();
-
- return window_manager_;
-}
-
-void ViEWindowCreator::TerminateWindows() {
- window_manager_->TerminateWindows();
-}
-
diff --git a/trunk/src/video_engine/test/auto_test/helpers/vie_window_creator.h b/trunk/src/video_engine/test/auto_test/helpers/vie_window_creator.h
deleted file mode 100644
index 25c23a3..0000000
--- a/trunk/src/video_engine/test/auto_test/helpers/vie_window_creator.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_WINDOW_CREATOR_H_
-#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_WINDOW_CREATOR_H_
-
-class ViEAutoTestWindowManagerInterface;
-
-class ViEWindowCreator {
- public:
- ViEWindowCreator();
- virtual ~ViEWindowCreator();
-
- // The pointer returned here will still be owned by this object.
- // Only use it to retrieve the created windows.
- ViEAutoTestWindowManagerInterface* CreateTwoWindows();
-
- // Terminates windows opened by CreateTwoWindows, which must
- // have been called before this method.
- void TerminateWindows();
- private:
- ViEAutoTestWindowManagerInterface* window_manager_;
-};
-
-#endif // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_HELPERS_VIE_WINDOW_CREATOR_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/tb_I420_codec.h b/trunk/src/video_engine/test/auto_test/interface/tb_I420_codec.h
deleted file mode 100644
index d6be4b2..0000000
--- a/trunk/src/video_engine/test/auto_test/interface/tb_I420_codec.h
+++ /dev/null
@@ -1,134 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * tb_I420_codec.h
- *
- * This file contains the interface to I420 "codec"
- * This is a dummy wrapper to allow VCM deal with raw I420 sequences
- *
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_I420_CODEC_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_I420_CODEC_H_
-
-#include "video_codec_interface.h"
-
-class TbI420Encoder: public webrtc::VideoEncoder
-{
-public:
- TbI420Encoder();
- virtual ~TbI420Encoder();
-
- static WebRtc_Word32 VersionStatic(WebRtc_Word8* version,
- WebRtc_Word32 length);
- virtual WebRtc_Word32 Version(WebRtc_Word8 *version,
- WebRtc_Word32 length) const;
-
- virtual WebRtc_Word32 InitEncode(const webrtc::VideoCodec* codecSettings,
- WebRtc_Word32 numberOfCores,
- WebRtc_UWord32 maxPayloadSize);
-
- virtual WebRtc_Word32 Encode(
- const webrtc::RawImage& inputImage,
- const webrtc::CodecSpecificInfo* codecSpecificInfo,
- const webrtc::VideoFrameType* frameType);
-
- virtual WebRtc_Word32 RegisterEncodeCompleteCallback(
- webrtc::EncodedImageCallback* callback);
-
- virtual WebRtc_Word32 Release();
-
- virtual WebRtc_Word32 Reset();
-
- virtual WebRtc_Word32 SetChannelParameters(WebRtc_UWord32 packetLoss,
- int rtt);
-
- virtual WebRtc_Word32 SetPacketLoss(WebRtc_UWord32 packetLoss);
-
- virtual WebRtc_Word32 SetRates(WebRtc_UWord32 newBitRate,
- WebRtc_UWord32 frameRate);
-
- virtual WebRtc_Word32 SetPeriodicKeyFrames(bool enable);
-
- virtual WebRtc_Word32 CodecConfigParameters(WebRtc_UWord8* /*buffer*/,
- WebRtc_Word32 /*size*/);
-
- struct FunctionCalls
- {
- WebRtc_Word32 InitEncode;
- WebRtc_Word32 Encode;
- WebRtc_Word32 RegisterEncodeCompleteCallback;
- WebRtc_Word32 Release;
- WebRtc_Word32 Reset;
- WebRtc_Word32 SetRates;
- WebRtc_Word32 SetPacketLoss;
- WebRtc_Word32 SetPeriodicKeyFrames;
- WebRtc_Word32 CodecConfigParameters;
-
- };
-
- FunctionCalls GetFunctionCalls();
-private:
- bool _inited;
- webrtc::EncodedImage _encodedImage;
- FunctionCalls _functionCalls;
- webrtc::EncodedImageCallback* _encodedCompleteCallback;
-
-}; // end of tbI420Encoder class
-
-
-/***************************/
-/* tbI420Decoder class */
-/***************************/
-
-class TbI420Decoder: public webrtc::VideoDecoder
-{
-public:
- TbI420Decoder();
- virtual ~TbI420Decoder();
-
- virtual WebRtc_Word32 InitDecode(const webrtc::VideoCodec* inst,
- WebRtc_Word32 numberOfCores);
- virtual WebRtc_Word32 Decode(
- const webrtc::EncodedImage& inputImage,
- bool missingFrames,
- const webrtc::RTPFragmentationHeader* fragmentation,
- const webrtc::CodecSpecificInfo* codecSpecificInfo = NULL,
- WebRtc_Word64 renderTimeMs = -1);
-
- virtual WebRtc_Word32
- RegisterDecodeCompleteCallback(webrtc::DecodedImageCallback* callback);
- virtual WebRtc_Word32 Release();
- virtual WebRtc_Word32 Reset();
-
- struct FunctionCalls
- {
- WebRtc_Word32 InitDecode;
- WebRtc_Word32 Decode;
- WebRtc_Word32 RegisterDecodeCompleteCallback;
- WebRtc_Word32 Release;
- WebRtc_Word32 Reset;
- };
-
- FunctionCalls GetFunctionCalls();
-
-private:
-
- webrtc::RawImage _decodedImage;
- WebRtc_Word32 _width;
- WebRtc_Word32 _height;
- bool _inited;
- FunctionCalls _functionCalls;
- webrtc::DecodedImageCallback* _decodeCompleteCallback;
-
-}; // end of tbI420Decoder class
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_I420_CODEC_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/tb_capture_device.h b/trunk/src/video_engine/test/auto_test/interface/tb_capture_device.h
deleted file mode 100644
index 3ade1e0..0000000
--- a/trunk/src/video_engine/test/auto_test/interface/tb_capture_device.h
+++ /dev/null
@@ -1,31 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_CAPTURE_DEVICE_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_CAPTURE_DEVICE_H_
-
-#include "tb_interfaces.h"
-#include "video_capture_factory.h"
-
-class TbCaptureDevice
-{
-public:
- TbCaptureDevice(TbInterfaces& Engine);
- ~TbCaptureDevice(void);
-
- int captureId;
- void ConnectTo(int videoChannel);
- void Disconnect(int videoChannel);
-private:
- TbInterfaces& ViE;
- webrtc::VideoCaptureModule* vcpm_;
-};
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_CAPTURE_DEVICE_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/tb_external_transport.h b/trunk/src/video_engine/test/auto_test/interface/tb_external_transport.h
deleted file mode 100644
index 6578811..0000000
--- a/trunk/src/video_engine/test/auto_test/interface/tb_external_transport.h
+++ /dev/null
@@ -1,159 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// tb_external_transport.h
-//
-
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_AUTOTEST_INTERFACE_TB_EXTERNAL_TRANSPORT_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_AUTOTEST_INTERFACE_TB_EXTERNAL_TRANSPORT_H_
-
-#include <list>
-
-#include "common_types.h"
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-class EventWrapper;
-class ThreadWrapper;
-class ViENetwork;
-}
-
-// Allows to subscribe for callback when a frame is started being sent.
-class SendFrameCallback
-{
-public:
- // Called once per frame (when a new RTP timestamp is detected) when the
- // first data packet of the frame is being sent using the
- // TbExternalTransport.SendPacket method.
- virtual void FrameSent(unsigned int rtp_timestamp) = 0;
-protected:
- SendFrameCallback() {}
- virtual ~SendFrameCallback() {}
-};
-
-// Allows to subscribe for callback when the first packet of a frame is
-// received.
-class ReceiveFrameCallback
-{
-public:
- // Called once per frame (when a new RTP timestamp is detected)
- // during the processing of the RTP packet queue in
- // TbExternalTransport::ViEExternalTransportProcess.
- virtual void FrameReceived(unsigned int rtp_timestamp) = 0;
-protected:
- ReceiveFrameCallback() {}
- virtual ~ReceiveFrameCallback() {}
-};
-
-// External transport implementation for testing purposes.
-// A packet loss probability must be set in order to drop packets from the data
-// being sent to this class.
-// Will never drop packets from the first frame of a video sequence.
-class TbExternalTransport : public webrtc::Transport
-{
-public:
- TbExternalTransport(webrtc::ViENetwork& vieNetwork);
- ~TbExternalTransport(void);
-
- virtual int SendPacket(int channel, const void *data, int len);
- virtual int SendRTCPPacket(int channel, const void *data, int len);
-
- // Should only be called before/after traffic is being processed.
- // Only one observer can be set (multiple calls will overwrite each other).
- virtual void RegisterSendFrameCallback(SendFrameCallback* callback);
-
- // Should only be called before/after traffic is being processed.
- // Only one observer can be set (multiple calls will overwrite each other).
- virtual void RegisterReceiveFrameCallback(ReceiveFrameCallback* callback);
-
- // The probability of a packet of being dropped. Packets belonging to the
- // first packet (same RTP timestamp) will never be dropped.
- WebRtc_Word32 SetPacketLoss(WebRtc_Word32 lossRate); // Rate in %
- void SetNetworkDelay(WebRtc_Word64 delayMs);
- void SetSSRCFilter(WebRtc_UWord32 SSRC);
-
- void ClearStats();
- void GetStats(WebRtc_Word32& numRtpPackets,
- WebRtc_Word32& numDroppedPackets,
- WebRtc_Word32& numRtcpPackets);
-
- void SetTemporalToggle(unsigned char layers);
- void EnableSSRCCheck();
- unsigned int ReceivedSSRC();
-
- void EnableSequenceNumberCheck();
- unsigned short GetFirstSequenceNumber();
-
-protected:
- static bool ViEExternalTransportRun(void* object);
- bool ViEExternalTransportProcess();
-private:
- WebRtc_Word64 NowMs();
-
- enum
- {
- KMaxPacketSize = 1650
- };
- enum
- {
- KMaxWaitTimeMs = 100
- };
- typedef struct
- {
- WebRtc_Word8 packetBuffer[KMaxPacketSize];
- WebRtc_Word32 length;
- WebRtc_Word32 channel;
- WebRtc_Word64 receiveTime;
- } VideoPacket;
-
- webrtc::ViENetwork& _vieNetwork;
- webrtc::ThreadWrapper& _thread;
- webrtc::EventWrapper& _event;
- webrtc::CriticalSectionWrapper& _crit;
- webrtc::CriticalSectionWrapper& _statCrit;
-
- WebRtc_Word32 _lossRate;
- WebRtc_Word64 _networkDelayMs;
- WebRtc_Word32 _rtpCount;
- WebRtc_Word32 _rtcpCount;
- WebRtc_Word32 _dropCount;
-
- std::list<VideoPacket*> _rtpPackets;
- std::list<VideoPacket*> _rtcpPackets;
-
- SendFrameCallback* _send_frame_callback;
- ReceiveFrameCallback* _receive_frame_callback;
-
- unsigned char _temporalLayers;
- unsigned short _seqNum;
- unsigned short _sendPID;
- unsigned char _receivedPID;
- bool _switchLayer;
- unsigned char _currentRelayLayer;
- unsigned int _lastTimeMs;
-
- bool _checkSSRC;
- WebRtc_UWord32 _lastSSRC;
- bool _filterSSRC;
- WebRtc_UWord32 _SSRC;
- bool _checkSequenceNumber;
- WebRtc_UWord16 _firstSequenceNumber;
-
- // Keep track of the first RTP timestamp so we don't do packet loss on
- // the first frame.
- WebRtc_UWord32 _firstRTPTimestamp;
- // Track RTP timestamps so we invoke callbacks properly (if registered).
- WebRtc_UWord32 _lastSendRTPTimestamp;
- WebRtc_UWord32 _lastReceiveRTPTimestamp;
-};
-
-#endif // WEBRTC_VIDEO_ENGINE_TEST_AUTOTEST_INTERFACE_TB_EXTERNAL_TRANSPORT_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/tb_interfaces.h b/trunk/src/video_engine/test/auto_test/interface/tb_interfaces.h
deleted file mode 100644
index 63e78a3..0000000
--- a/trunk/src/video_engine/test/auto_test/interface/tb_interfaces.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_INTERFACES_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_INTERFACES_H_
-
-#include "vie_autotest_defines.h"
-
-#include "common_types.h"
-#include "vie_base.h"
-#include "vie_capture.h"
-#include "vie_codec.h"
-#include "vie_image_process.h"
-#include "vie_network.h"
-#include "vie_render.h"
-#include "vie_rtp_rtcp.h"
-#include "vie_encryption.h"
-#include "vie_defines.h"
-
-// This class deals with all the tedium of setting up video engine interfaces.
-// It does its work in constructor and destructor, so keeping it in scope is
-// enough.
-class TbInterfaces
-{
-public:
- TbInterfaces(const char* test_name);
- ~TbInterfaces(void);
-
- webrtc::VideoEngine* video_engine;
- webrtc::ViEBase* base;
- webrtc::ViECapture* capture;
- webrtc::ViERender* render;
- webrtc::ViERTP_RTCP* rtp_rtcp;
- webrtc::ViECodec* codec;
- webrtc::ViENetwork* network;
- webrtc::ViEImageProcess* image_process;
- webrtc::ViEEncryption* encryption;
-
- int LastError() {
- return base->LastError();
- }
-};
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_INTERFACES_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/tb_video_channel.h b/trunk/src/video_engine/test/auto_test/interface/tb_video_channel.h
deleted file mode 100644
index 5961e62..0000000
--- a/trunk/src/video_engine/test/auto_test/interface/tb_video_channel.h
+++ /dev/null
@@ -1,41 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_VIDEO_CHANNEL_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_VIDEO_CHANNEL_H_
-
-#include "tb_interfaces.h"
-class TbVideoChannel
-{
-public:
- TbVideoChannel(TbInterfaces& Engine,
- webrtc::VideoCodecType sendCodec = webrtc::kVideoCodecVP8,
- int width = 352, int height = 288, int frameRate = 30,
- int startBitrate = 300);
-
- ~TbVideoChannel(void);
-
- void SetFrameSettings(int width, int height, int frameRate);
-
- void StartSend(const unsigned short rtpPort = 11000,
- const char* ipAddress = "127.0.0.1");
-
- void StopSend();
-
- void StartReceive(const unsigned short rtpPort = 11000);
-
- void StopReceive();
-
- int videoChannel;
-private:
- TbInterfaces& ViE;
-};
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_TB_VIDEO_CHANNEL_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest.h
deleted file mode 100644
index a508cd4..0000000
--- a/trunk/src/video_engine/test/auto_test/interface/vie_autotest.h
+++ /dev/null
@@ -1,137 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest.h
-//
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_H_
-
-#include "common_types.h"
-
-#include "voe_base.h"
-#include "voe_codec.h"
-#include "voe_hardware.h"
-#include "voe_audio_processing.h"
-
-#include "vie_base.h"
-#include "vie_capture.h"
-#include "vie_codec.h"
-#include "vie_file.h"
-#include "vie_network.h"
-#include "vie_render.h"
-#include "vie_rtp_rtcp.h"
-#include "vie_defines.h"
-#include "vie_errors.h"
-#include "video_render_defines.h"
-
-#include "vie_autotest_defines.h"
-
-#ifndef WEBRTC_ANDROID
-#include <string>
-#endif
-
-class TbCaptureDevice;
-class TbInterfaces;
-class TbVideoChannel;
-class ViEToFileRenderer;
-
-// This class provides a bunch of methods, implemented across several .cc
-// files, which runs tests on the video engine. All methods will report
-// errors using standard googletest macros, except when marked otherwise.
-class ViEAutoTest
-{
-public:
- ViEAutoTest(void* window1, void* window2);
- ~ViEAutoTest();
-
- // These three are special and should not be run in a googletest harness.
- // They keep track of their errors by themselves and return the number
- // of errors.
- int ViELoopbackCall();
- int ViESimulcastCall();
- int ViECustomCall();
-
- // All functions except the three above are meant to run in a
- // googletest harness.
- void ViEStandardTest();
- void ViEExtendedTest();
- void ViEAPITest();
-
- // vie_autotest_base.cc
- void ViEBaseStandardTest();
- void ViEBaseExtendedTest();
- void ViEBaseAPITest();
-
- // vie_autotest_capture.cc
- void ViECaptureStandardTest();
- void ViECaptureExtendedTest();
- void ViECaptureAPITest();
- void ViECaptureExternalCaptureTest();
-
- // vie_autotest_codec.cc
- void ViECodecStandardTest();
- void ViECodecExtendedTest();
- void ViECodecExternalCodecTest();
- void ViECodecAPITest();
-
- // vie_autotest_encryption.cc
- void ViEEncryptionStandardTest();
- void ViEEncryptionExtendedTest();
- void ViEEncryptionAPITest();
-
- // vie_autotest_file.ccs
- void ViEFileStandardTest();
- void ViEFileExtendedTest();
- void ViEFileAPITest();
-
- // vie_autotest_image_process.cc
- void ViEImageProcessStandardTest();
- void ViEImageProcessExtendedTest();
- void ViEImageProcessAPITest();
-
- // vie_autotest_network.cc
- void ViENetworkStandardTest();
- void ViENetworkExtendedTest();
- void ViENetworkAPITest();
-
- // vie_autotest_render.cc
- void ViERenderStandardTest();
- void ViERenderExtendedTest();
- void ViERenderAPITest();
-
- // vie_autotest_rtp_rtcp.cc
- void ViERtpRtcpStandardTest();
- void ViERtpRtcpExtendedTest();
- void ViERtpRtcpAPITest();
-
- // vie_autotest_rtp_fuzz.cc
- void ViERtpTryInjectingRandomPacketsIntoRtpStream(long rand_seed);
-
-private:
- void PrintAudioCodec(const webrtc::CodecInst audioCodec);
- void PrintVideoCodec(const webrtc::VideoCodec videoCodec);
-
- // Sets up rendering so the capture device output goes to window 1 and
- // the video engine output goes to window 2.
- void RenderCaptureDeviceAndOutputStream(TbInterfaces* video_engine,
- TbVideoChannel* video_channel,
- TbCaptureDevice* capture_device);
-
- void* _window1;
- void* _window2;
-
- webrtc::VideoRenderType _renderType;
- webrtc::VideoRender* _vrm1;
- webrtc::VideoRender* _vrm2;
-};
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_android.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_android.h
deleted file mode 100644
index 53b8cc3..0000000
--- a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_android.h
+++ /dev/null
@@ -1,26 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_ANDROID_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_ANDROID_H_
-
-class ViEAutoTestAndroid
-{
-public:
- static int RunAutotest(int testSelection,
- int subTestSelection,
- void* window1,
- void* window2,
- void* javaVM,
- void* env,
- void* context);
-};
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_ANDROID_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_defines.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_defines.h
deleted file mode 100644
index 34d489b..0000000
--- a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_defines.h
+++ /dev/null
@@ -1,222 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest_defines.h
-//
-
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_DEFINES_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_DEFINES_H_
-
-#include <cassert>
-#include <stdarg.h>
-#include <stdio.h>
-#include <string>
-
-#include "engine_configurations.h"
-#include "gtest/gtest.h"
-
-#if defined(_WIN32)
-#include <windows.h>
-#elif defined (WEBRTC_ANDROID)
-#include <android/log.h>
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
-#include <string.h>
-#include <time.h>
-#include <sys/time.h>
-#include <stdlib.h>
-#endif
-
-// Choose how to log
-//#define VIE_LOG_TO_FILE
-#define VIE_LOG_TO_STDOUT
-
-// Choose one way to test error
-#define VIE_ASSERT_ERROR
-
-#define VIE_LOG_FILE_NAME "ViEAutotestLog.txt"
-
-#undef RGB
-#define RGB(r,g,b) r|g<<8|b<<16
-
-enum {
- KAutoTestSleepTimeMs = 5000
-};
-
-struct AutoTestSize {
- unsigned int width;
- unsigned int height;
- AutoTestSize() :
- width(0), height(0) {
- }
- AutoTestSize(unsigned int iWidth, unsigned int iHeight) :
- width(iWidth), height(iHeight) {
- }
-};
-
-struct AutoTestOrigin {
- unsigned int x;
- unsigned int y;
- AutoTestOrigin() :
- x(0), y(0) {
- }
- AutoTestOrigin(unsigned int iX, unsigned int iY) :
- x(iX), y(iY) {
- }
-};
-
-struct AutoTestRect {
- AutoTestSize size;
- AutoTestOrigin origin;
- AutoTestRect() :
- size(), origin() {
- }
-
- AutoTestRect(unsigned int iX, unsigned int iY, unsigned int iWidth, unsigned int iHeight) :
- size(iX, iY), origin(iWidth, iHeight) {
- }
-
- void Copy(AutoTestRect iRect) {
- origin.x = iRect.origin.x;
- origin.y = iRect.origin.y;
- size.width = iRect.size.width;
- size.height = iRect.size.height;
- }
-};
-
-// ============================================
-
-class ViETest {
- public:
- static int Init() {
-#ifdef VIE_LOG_TO_FILE
- log_file_ = fopen(VIE_LOG_FILE_NAME, "w+t");
-#else
- log_file_ = NULL;
-#endif
- log_str_ = new char[kMaxLogSize];
- memset(log_str_, 0, kMaxLogSize);
- return 0;
- }
-
- static int Terminate() {
- if (log_file_) {
- fclose(log_file_);
- log_file_ = NULL;
- }
- if (log_str_) {
- delete[] log_str_;
- log_str_ = NULL;
- }
- return 0;
- }
-
- static void Log(const char* fmt, ...) {
- va_list va;
- va_start(va, fmt);
- memset(log_str_, 0, kMaxLogSize);
- vsprintf(log_str_, fmt, va);
- va_end(va);
-
- WriteToSuitableOutput(log_str_);
- }
-
- // Writes to a suitable output, depending on platform and log mode.
- static void WriteToSuitableOutput(const char* message) {
-#ifdef VIE_LOG_TO_FILE
- if (log_file_)
- {
- fwrite(log_str_, 1, strlen(log_str_), log_file_);
- fwrite("\n", 1, 1, log_file_);
- fflush(log_file_);
- }
-#endif
-#ifdef VIE_LOG_TO_STDOUT
-#if WEBRTC_ANDROID
- __android_log_write(ANDROID_LOG_DEBUG, "*WebRTCN*", log_str_);
-#else
- printf("%s\n", log_str_);
-#endif
-#endif
- }
-
- // Deprecated(phoglund): Prefer to use googletest macros in all cases
- // except the custom call case.
- static int TestError(bool expr, const char* fmt, ...) {
- if (!expr) {
- va_list va;
- va_start(va, fmt);
- memset(log_str_, 0, kMaxLogSize);
- vsprintf(log_str_, fmt, va);
-#ifdef WEBRTC_ANDROID
- __android_log_write(ANDROID_LOG_ERROR, "*WebRTCN*", log_str_);
-#endif
- WriteToSuitableOutput(log_str_);
- va_end(va);
-
- AssertError(log_str_);
- return 1;
- }
- return 0;
- }
-
- // Returns a suitable path to write trace and result files to.
- // You should always use this when you want to write output files.
- // The returned path is guaranteed to end with a path separator.
- // This function may be run at any time during the program's execution.
- // Implemented in vie_autotest.cc
- static std::string GetResultOutputPath();
-
-private:
- static void AssertError(const char* message) {
-#ifdef VIE_ASSERT_ERROR
- assert(false);
-#endif
- }
-
- static FILE* log_file_;
- enum {
- kMaxLogSize = 512
- };
- static char* log_str_;
-};
-
-// milliseconds
-#if defined(_WIN32)
-#define AutoTestSleep ::Sleep
-#elif defined(WEBRTC_MAC_INTEL)
-#define AutoTestSleep(x) usleep(x * 1000)
-#elif defined(WEBRTC_LINUX)
-namespace {
- void Sleep(unsigned long x) {
- timespec t;
- t.tv_sec = x/1000;
- t.tv_nsec = (x-(x/1000)*1000)*1000000;
- nanosleep(&t,NULL);
- }
-}
-#define AutoTestSleep ::Sleep
-#endif
-
-#ifdef WEBRTC_ANDROID
-namespace {
- void Sleep(unsigned long x) {
- timespec t;
- t.tv_sec = x/1000;
- t.tv_nsec = (x-(x/1000)*1000)*1000000;
- nanosleep(&t,NULL);
- }
-}
-
-#define AutoTestSleep ::Sleep
-#endif
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_DEFINES_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_linux.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_linux.h
deleted file mode 100644
index fb1d319..0000000
--- a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_linux.h
+++ /dev/null
@@ -1,44 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_LINUX_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_LINUX_H_
-
-#include "vie_autotest_window_manager_interface.h"
-#include <X11/Xlib.h>
-#include <X11/Xutil.h>
-
-// Forward declaration
-
-class ViEAutoTestWindowManager: public ViEAutoTestWindowManagerInterface
-{
-public:
- ViEAutoTestWindowManager();
- virtual ~ViEAutoTestWindowManager();
- virtual void* GetWindow1();
- virtual void* GetWindow2();
- virtual int TerminateWindows();
- virtual int CreateWindows(AutoTestRect window1Size,
- AutoTestRect window2Size, void* window1Title,
- void* window2Title);
- virtual bool SetTopmostWindow();
-
-private:
- int ViECreateWindow(Window *outWindow, Display **outDisplay, int xpos,
- int ypos, int width, int height, char* title);
- int ViEDestroyWindow(Window *window, Display *display);
-
- Window _hwnd1;
- Window _hwnd2;
- Display* _hdsp1;
- Display* _hdsp2;
-};
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_LINUX_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_mac_carbon.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_mac_carbon.h
deleted file mode 100644
index 6c35baa..0000000
--- a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_mac_carbon.h
+++ /dev/null
@@ -1,63 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "EngineConfigurations.h"
-
-#if defined(CARBON_RENDERING)
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAC_CARBON_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAC_CARBON_H_
-
-#include "vie_autotest_window_manager_interface.h"
-
-// #define HIVIEWREF_MODE 1
-
-#include <Carbon/Carbon.h>
-#import <Cocoa/Cocoa.h>
-
-class ViEAutoTestWindowManager: public ViEAutoTestWindowManagerInterface
-{
-public:
- ViEAutoTestWindowManager();
- virtual ~ViEAutoTestWindowManager();
- virtual void* GetWindow1();
- virtual void* GetWindow2();
- virtual int CreateWindows(AutoTestRect window1Size,
- AutoTestRect window2Size, char* window1Title,
- char* window2Title);
- virtual int TerminateWindows();
- virtual bool SetTopmostWindow();
-
- // event handler static methods
-static pascal OSStatus HandleWindowEvent (EventHandlerCallRef nextHandler,
- EventRef theEvent, void* userData);
-static pascal OSStatus HandleHIViewEvent (EventHandlerCallRef nextHandler,
- EventRef theEvent, void* userData);
-private:
- WindowRef* _carbonWindow1;
- WindowRef* _carbonWindow2;
- HIViewRef* _hiView1;
- HIViewRef* _hiView2;
-
- EventHandlerRef _carbonWindow1EventHandlerRef;
- EventHandlerRef _carbonWindow2EventHandlerRef;
- EventHandlerRef _carbonHIView1EventHandlerRef;
- EventHandlerRef _carbonHIView2EventHandlerRef;
-
-};
-
-@interface AutoTestClass : NSObject
-{
-}
-
--(void)autoTestWithArg:(NSString*)answerFile;
-@end
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAC_CARBON_H_
-#endif CARBON_RENDERING
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_mac_cocoa.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_mac_cocoa.h
deleted file mode 100644
index 745431d..0000000
--- a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_mac_cocoa.h
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "engine_configurations.h"
-
-#if defined(COCOA_RENDERING)
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAC_COCOA_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAC_COCOA_H_
-
-#include "vie_autotest_window_manager_interface.h"
-#define MAC_COCOA_USE_NSRUNLOOP 1
-
-@class CocoaRenderView;
-
-#import <Cocoa/Cocoa.h>
-
-class ViEAutoTestWindowManager: public ViEAutoTestWindowManagerInterface {
- public:
- ViEAutoTestWindowManager();
- virtual ~ViEAutoTestWindowManager() {}
- virtual void* GetWindow1();
- virtual void* GetWindow2();
- virtual int CreateWindows(AutoTestRect window1Size,
- AutoTestRect window2Size,
- void* window1Title,
- void* window2Title);
- virtual int TerminateWindows();
- virtual bool SetTopmostWindow();
-
- private:
- CocoaRenderView* _cocoaRenderView1;
- CocoaRenderView* _cocoaRenderView2;
- NSWindow* outWindow1_;
- NSWindow* outWindow2_;
-};
-
-@interface AutoTestClass : NSObject {
- int argc_;
- char** argv_;
- int result_;
-}
-
--(void)setArgc:(int)argc argv:(char**)argv;
--(int) result;
--(void)autoTestWithArg:(NSObject*)ignored;
-
-@end
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAC_COCOA_H_
-#endif // COCOA_RENDERING
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_main.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_main.h
deleted file mode 100644
index d24de5c..0000000
--- a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_main.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAIN_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAIN_H_
-
-#include <string>
-#include <map>
-
-class ViEAutoTestMain {
- public:
- ViEAutoTestMain();
-
- // Runs the test according to the specified arguments.
- // Pass in --automated to run in automated mode; interactive
- // mode is default. All usual googletest flags also apply.
- int RunTests(int argc, char** argv);
-
- private:
- std::map<int, std::string> index_to_test_method_map_;
-
- static const int kInvalidChoice = -1;
-
- // Starts interactive mode.
- bool RunInteractiveMode();
- // Prompts the user for a specific test method in the provided test case.
- // Returns 0 on success, nonzero otherwise.
- int RunSpecificTestCaseIn(const std::string test_case_name);
- // Asks the user for a particular test case to run.
- int AskUserForTestCase();
- // Retrieves a number from the user in the interval
- // [min_allowed, max_allowed]. Returns kInvalidChoice on failure.
- int AskUserForNumber(int min_allowed, int max_allowed);
- // Runs all tests matching the provided filter. * are wildcards.
- // Returns the test runner result (0 == OK).
- int RunTestMatching(const std::string test_case,
- const std::string test_method);
- // Runs a non-gtest test case. Choice must be [7,9]. Returns 0 on success.
- int RunSpecialTestCase(int choice);
-};
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_MAIN_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h
deleted file mode 100644
index 6dd043e..0000000
--- a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_window_manager_interface.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * vie_autotest_window_manager_interface.h
- */
-
-#include "vie_autotest_defines.h"
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOW_MANAGER_INTERFACE_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOW_MANAGER_INTERFACE_H_
-
-class ViEAutoTestWindowManagerInterface
-{
-public:
- virtual int CreateWindows(AutoTestRect window1Size,
- AutoTestRect window2Size, void* window1Title,
- void* window2Title) = 0;
- virtual int TerminateWindows() = 0;
- virtual void* GetWindow1() = 0;
- virtual void* GetWindow2() = 0;
- virtual bool SetTopmostWindow() = 0;
- virtual ~ViEAutoTestWindowManagerInterface() {}
-};
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOW_MANAGER_INTERFACE_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_windows.h b/trunk/src/video_engine/test/auto_test/interface/vie_autotest_windows.h
deleted file mode 100644
index 314a121..0000000
--- a/trunk/src/video_engine/test/auto_test/interface/vie_autotest_windows.h
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOWS_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOWS_H_
-
-#include "vie_autotest_window_manager_interface.h"
-#include "engine_configurations.h"
-
-#include <windows.h>
-#define TITLE_LENGTH 1024
-
-// Forward declaration
-namespace webrtc {
-class ThreadWrapper;
-class CriticalSectionWrapper;
-}
-
-class ViEAutoTestWindowManager: public ViEAutoTestWindowManagerInterface
-{
-public:
- ViEAutoTestWindowManager();
- virtual ~ViEAutoTestWindowManager();
- virtual void* GetWindow1();
- virtual void* GetWindow2();
- virtual int CreateWindows(AutoTestRect window1Size,
- AutoTestRect window2Size, void* window1Title,
- void* window2Title);
- virtual int TerminateWindows();
- virtual bool SetTopmostWindow();
-protected:
- static bool EventProcess(void* obj);
- bool EventLoop();
-
-private:
- int ViECreateWindow(HWND &hwndMain, int xPos, int yPos, int width,
- int height, TCHAR* className);
- int ViEDestroyWindow(HWND& hwnd);
-
- void* _window1;
- void* _window2;
-
- bool _terminate;
- webrtc::ThreadWrapper& _eventThread;
- webrtc::CriticalSectionWrapper& _crit;
- HWND _hwndMain;
- HWND _hwnd1;
- HWND _hwnd2;
-
- AutoTestRect _hwnd1Size;
- AutoTestRect _hwnd2Size;
- TCHAR _hwnd1Title[TITLE_LENGTH];
- TCHAR _hwnd2Title[TITLE_LENGTH];
-
-};
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_AUTOTEST_WINDOWS_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_file_based_comparison_tests.h b/trunk/src/video_engine/test/auto_test/interface/vie_file_based_comparison_tests.h
deleted file mode 100644
index a9560bc..0000000
--- a/trunk/src/video_engine/test/auto_test/interface/vie_file_based_comparison_tests.h
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VIDEO_ENGINE_TEST_AUTO_TEST_INTERFACE_VIE_COMPARISON_TESTS_H_
-#define SRC_VIDEO_ENGINE_TEST_AUTO_TEST_INTERFACE_VIE_COMPARISON_TESTS_H_
-
-#include <string>
-
-class FrameDropDetector;
-class ViEToFileRenderer;
-
-// This class contains comparison tests, which will exercise video engine
-// functionality and then run comparison tests on the result using PSNR and
-// SSIM algorithms. These tests are intended mostly as sanity checks so that
-// we know we are outputting roughly the right thing and not random noise or
-// black screens.
-//
-// We will set up a fake ExternalCapture device which will pose as a webcam
-// and read the input from the provided raw YUV file. Output will be written
-// as a local preview in the local file renderer; the remote side output gets
-// written to the provided remote file renderer.
-//
-// The local preview is a straight, unaltered copy of the input. This can be
-// useful for comparisons if the test method contains several stages where the
-// input is restarted between stages.
-class ViEFileBasedComparisonTests {
- public:
- // Test a typical simple call setup. Returns false if the input file
- // could not be opened; reports errors using googletest macros otherwise.
- bool TestCallSetup(
- const std::string& i420_test_video_path,
- int width,
- int height,
- ViEToFileRenderer* local_file_renderer,
- ViEToFileRenderer* remote_file_renderer);
-
- // Tries testing the I420 and VP8 codecs in turn. Returns false if the
- // input file could not be opened; reports errors using googletest macros
- // otherwise.
- bool TestCodecs(
- const std::string& i420_video_file,
- int width,
- int height,
- ViEToFileRenderer* local_file_renderer,
- ViEToFileRenderer* remote_file_renderer);
-
- // Runs a full stack test using the VP8 codec. Tests the full stack and uses
- // RTP timestamps to sync frames between the endpoints.
- void TestFullStack(
- const std::string& i420_video_file,
- int width,
- int height,
- int bit_rate_kbps,
- int packet_loss_percent,
- int network_delay_ms,
- ViEToFileRenderer* local_file_renderer,
- ViEToFileRenderer* remote_file_renderer,
- FrameDropDetector* frame_drop_detector);
-};
-
-#endif // SRC_VIDEO_ENGINE_TEST_AUTO_TEST_INTERFACE_VIE_COMPARISON_TESTS_H_
diff --git a/trunk/src/video_engine/test/auto_test/interface/vie_window_manager_factory.h b/trunk/src/video_engine/test/auto_test/interface/vie_window_manager_factory.h
deleted file mode 100644
index a85280d..0000000
--- a/trunk/src/video_engine/test/auto_test/interface/vie_window_manager_factory.h
+++ /dev/null
@@ -1,25 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_WINDOW_MANAGER_FACTORY_H_
-#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_WINDOW_MANAGER_FACTORY_H_
-
-class ViEAutoTestWindowManagerInterface;
-
-class ViEWindowManagerFactory {
- public:
- // This method is implemented in different files depending on platform.
- // The caller is responsible for freeing the resulting object using
- // the delete operator.
- static ViEAutoTestWindowManagerInterface*
- CreateWindowManagerForCurrentPlatform();
-};
-
-#endif // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_INTERFACE_VIE_WINDOW_MANAGER_FACTORY_H_
diff --git a/trunk/src/video_engine/test/auto_test/media/captureDeviceImage.bmp b/trunk/src/video_engine/test/auto_test/media/captureDeviceImage.bmp
deleted file mode 100644
index 6cd34ba..0000000
--- a/trunk/src/video_engine/test/auto_test/media/captureDeviceImage.bmp
+++ /dev/null
Binary files differ
diff --git a/trunk/src/video_engine/test/auto_test/media/captureDeviceImage.jpg b/trunk/src/video_engine/test/auto_test/media/captureDeviceImage.jpg
deleted file mode 100644
index 3bb3ba4..0000000
--- a/trunk/src/video_engine/test/auto_test/media/captureDeviceImage.jpg
+++ /dev/null
Binary files differ
diff --git a/trunk/src/video_engine/test/auto_test/media/renderStartImage.bmp b/trunk/src/video_engine/test/auto_test/media/renderStartImage.bmp
deleted file mode 100644
index c443a58..0000000
--- a/trunk/src/video_engine/test/auto_test/media/renderStartImage.bmp
+++ /dev/null
Binary files differ
diff --git a/trunk/src/video_engine/test/auto_test/media/renderStartImage.jpg b/trunk/src/video_engine/test/auto_test/media/renderStartImage.jpg
deleted file mode 100644
index b10a842..0000000
--- a/trunk/src/video_engine/test/auto_test/media/renderStartImage.jpg
+++ /dev/null
Binary files differ
diff --git a/trunk/src/video_engine/test/auto_test/media/renderTimeoutImage.bmp b/trunk/src/video_engine/test/auto_test/media/renderTimeoutImage.bmp
deleted file mode 100644
index 8159bad..0000000
--- a/trunk/src/video_engine/test/auto_test/media/renderTimeoutImage.bmp
+++ /dev/null
Binary files differ
diff --git a/trunk/src/video_engine/test/auto_test/media/renderTimeoutImage.jpg b/trunk/src/video_engine/test/auto_test/media/renderTimeoutImage.jpg
deleted file mode 100644
index cb34d67..0000000
--- a/trunk/src/video_engine/test/auto_test/media/renderTimeoutImage.jpg
+++ /dev/null
Binary files differ
diff --git a/trunk/src/video_engine/test/auto_test/primitives/base_primitives.cc b/trunk/src/video_engine/test/auto_test/primitives/base_primitives.cc
deleted file mode 100644
index b8c827e..0000000
--- a/trunk/src/video_engine/test/auto_test/primitives/base_primitives.cc
+++ /dev/null
@@ -1,67 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "base_primitives.h"
-
-#include "vie_autotest.h"
-#include "vie_autotest_defines.h"
-#include "video_capture_factory.h"
-
-void TestI420CallSetup(webrtc::ViECodec* codec_interface,
- webrtc::VideoEngine* video_engine,
- webrtc::ViEBase* base_interface,
- webrtc::ViENetwork* network_interface,
- int video_channel,
- const unsigned char *device_name) {
- webrtc::VideoCodec video_codec;
- memset(&video_codec, 0, sizeof(webrtc::VideoCodec));
-
- // Set up the codec interface with all known receive codecs and with
- // I420 as the send codec.
- for (int i = 0; i < codec_interface->NumberOfCodecs(); i++) {
- EXPECT_EQ(0, codec_interface->GetCodec(i, video_codec));
-
- // Try to keep the test frame size small when I420.
- if (video_codec.codecType == webrtc::kVideoCodecI420) {
- video_codec.width = 176;
- video_codec.height = 144;
- EXPECT_EQ(0, codec_interface->SetSendCodec(video_channel, video_codec));
- }
-
- EXPECT_EQ(0, codec_interface->SetReceiveCodec(video_channel, video_codec));
- }
-
- // Verify that we really found the I420 codec.
- EXPECT_EQ(0, codec_interface->GetSendCodec(video_channel, video_codec));
- EXPECT_EQ(webrtc::kVideoCodecI420, video_codec.codecType);
-
- // Set up senders and receivers.
- char version[1024] = "";
- EXPECT_EQ(0, base_interface->GetVersion(version));
- ViETest::Log("\nUsing WebRTC Video Engine version: %s", version);
-
- const char *ipAddress = "127.0.0.1";
- WebRtc_UWord16 rtpPortListen = 6100;
- WebRtc_UWord16 rtpPortSend = 6100;
- EXPECT_EQ(0, network_interface->SetLocalReceiver(video_channel,
- rtpPortListen));
- EXPECT_EQ(0, base_interface->StartReceive(video_channel));
- EXPECT_EQ(0, network_interface->SetSendDestination(video_channel, ipAddress,
- rtpPortSend));
- EXPECT_EQ(0, base_interface->StartSend(video_channel));
-
- // Call started.
- ViETest::Log("Call started");
-
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- // Done.
- EXPECT_EQ(0, base_interface->StopSend(video_channel));
-}
diff --git a/trunk/src/video_engine/test/auto_test/primitives/base_primitives.h b/trunk/src/video_engine/test/auto_test/primitives/base_primitives.h
deleted file mode 100644
index 19b6f23..0000000
--- a/trunk/src/video_engine/test/auto_test/primitives/base_primitives.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_SOURCE_BASE_PRIMITIVES_H_
-#define SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_SOURCE_BASE_PRIMITIVES_H_
-
-namespace webrtc {
-class VideoEngine;
-class ViEBase;
-class ViECodec;
-class ViENetwork;
-}
-
-// Tests a I420-to-I420 call. This test exercises the most basic WebRTC
-// functionality by training the codec interface to recognize the most common
-// codecs, and the initiating a I420 call. A video channel with a capture device
-// must be set up prior to this call.
-void TestI420CallSetup(webrtc::ViECodec* codec_interface,
- webrtc::VideoEngine* video_engine,
- webrtc::ViEBase* base_interface,
- webrtc::ViENetwork* network_interface,
- int video_channel,
- const unsigned char *device_name);
-
-#endif // SRC_VIDEO_ENGINE_MAIN_TEST_AUTOTEST_SOURCE_BASE_PRIMITIVES_H_
diff --git a/trunk/src/video_engine/test/auto_test/primitives/codec_primitives.cc b/trunk/src/video_engine/test/auto_test/primitives/codec_primitives.cc
deleted file mode 100644
index 2d43440..0000000
--- a/trunk/src/video_engine/test/auto_test/primitives/codec_primitives.cc
+++ /dev/null
@@ -1,184 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "codec_primitives.h"
-
-#include "general_primitives.h"
-#include "vie_autotest.h"
-#include "vie_autotest_defines.h"
-#include "vie_to_file_renderer.h"
-#include "video_capture_factory.h"
-#include "tb_interfaces.h"
-
-// Helper functions.
-
-void TestCodecImageProcess(webrtc::VideoCodec video_codec,
- webrtc::ViECodec* codec_interface,
- int video_channel,
- webrtc::ViEImageProcess* image_process) {
-
- EXPECT_EQ(0, codec_interface->SetSendCodec(video_channel, video_codec));
- FrameCounterEffectFilter frame_counter;
- EXPECT_EQ(0, image_process->RegisterRenderEffectFilter(video_channel,
- frame_counter));
- AutoTestSleep (KAutoTestSleepTimeMs);
-
- int max_number_of_rendered_frames = video_codec.maxFramerate *
- KAutoTestSleepTimeMs / 1000;
-
- if (video_codec.codecType == webrtc::kVideoCodecI420) {
- // Due to that I420 needs a huge bandwidth, rate control can set
- // frame rate very low. This happen since we use the same channel
- // as we just tested with vp8.
- EXPECT_GT(frame_counter.numFrames, 0);
- } else {
-#ifdef WEBRTC_ANDROID
- // Special case to get the autotest to pass on some slow devices
- EXPECT_GT(frame_counter.numFrames, max_number_of_rendered_frames / 6);
-#else
- EXPECT_GT(frame_counter.numFrames, max_number_of_rendered_frames / 4);
-#endif
- }
- EXPECT_EQ(0, image_process->DeregisterRenderEffectFilter(video_channel));
-}
-
-// Test switching from i420 to VP8 as send codec and make sure that
-// the codec observer gets called after the switch.
-void TestCodecCallbacks(webrtc::ViEBase *& base_interface,
- webrtc::ViECodec *codec_interface,
- int video_channel,
- int forced_codec_width,
- int forced_codec_height) {
-
- // Set I420 as send codec so we don't make any assumptions about what
- // we currently have as send codec:
- SetSendCodec(webrtc::kVideoCodecI420, codec_interface, video_channel,
- forced_codec_width, forced_codec_height);
-
- // Register the observer:
- ViEAutotestCodecObserver codec_observer;
- EXPECT_EQ(0, codec_interface->RegisterEncoderObserver(video_channel,
- codec_observer));
- EXPECT_EQ(0, codec_interface->RegisterDecoderObserver(video_channel,
- codec_observer));
-
- // Make the switch.
- ViETest::Log("Testing codec callbacks...");
-
- SetSendCodec(webrtc::kVideoCodecVP8, codec_interface, video_channel,
- forced_codec_width, forced_codec_height);
-
- AutoTestSleep (KAutoTestSleepTimeMs);
-
- // Verify that we got the right codec.
- EXPECT_EQ(webrtc::kVideoCodecVP8, codec_observer.incomingCodec.codecType);
-
- // Clean up.
- EXPECT_EQ(0, codec_interface->DeregisterEncoderObserver(video_channel));
- EXPECT_EQ(0, codec_interface->DeregisterDecoderObserver(video_channel));
-
- // Verify results.
- EXPECT_GT(codec_observer.incomingCodecCalled, 0);
- EXPECT_GT(codec_observer.incomingRatecalled, 0);
- EXPECT_GT(codec_observer.outgoingRatecalled, 0);
-}
-
-void TestCodecs(const TbInterfaces& interfaces,
- int capture_id,
- int video_channel,
- int forced_codec_width,
- int forced_codec_height) {
- webrtc::VideoEngine *video_engine_interface = interfaces.video_engine;
- webrtc::ViEBase *base_interface = interfaces.base;
- webrtc::ViECapture *capture_interface = interfaces.capture;
- webrtc::ViERender *render_interface = interfaces.render;
- webrtc::ViECodec *codec_interface = interfaces.codec;
- webrtc::ViENetwork *network_interface = interfaces.network;
-
- // ***************************************************************
- // Engine ready. Begin testing class
- // ***************************************************************
- webrtc::VideoCodec video_codec;
- memset(&video_codec, 0, sizeof (webrtc::VideoCodec));
-
- // Set up all receive codecs. This basically trains the codec interface
- // to be able to recognize all receive codecs based on payload type.
- for (int idx = 0; idx < codec_interface->NumberOfCodecs(); idx++) {
- EXPECT_EQ(0, codec_interface->GetCodec(idx, video_codec));
- SetSuitableResolution(&video_codec,
- forced_codec_width,
- forced_codec_height);
-
- EXPECT_EQ(0, codec_interface->SetReceiveCodec(video_channel, video_codec));
- }
- const char *ip_address = "127.0.0.1";
- const unsigned short rtp_port = 6000;
- EXPECT_EQ(0, network_interface->SetLocalReceiver(video_channel, rtp_port));
- EXPECT_EQ(0, base_interface->StartReceive(video_channel));
- EXPECT_EQ(0, network_interface->SetSendDestination(video_channel, ip_address,
- rtp_port));
- EXPECT_EQ(0, base_interface->StartSend(video_channel));
-
- // Run all found codecs
- webrtc::ViEImageProcess *image_process =
- webrtc::ViEImageProcess::GetInterface(video_engine_interface);
- EXPECT_TRUE(image_process != NULL);
-
- ViETest::Log("Loop through all codecs for %d seconds",
- KAutoTestSleepTimeMs / 1000);
- for (int i = 0; i < codec_interface->NumberOfCodecs(); i++) {
- EXPECT_EQ(0, codec_interface->GetCodec(i, video_codec));
-
- if (video_codec.codecType == webrtc::kVideoCodecRED ||
- video_codec.codecType == webrtc::kVideoCodecULPFEC) {
- ViETest::Log("\t %d. %s not tested", i, video_codec.plName);
- } else {
- ViETest::Log("\t %d. %s", i, video_codec.plName);
- SetSuitableResolution(&video_codec, forced_codec_width,
- forced_codec_height);
- TestCodecImageProcess(video_codec, codec_interface, video_channel,
- image_process);
- }
- }
- image_process->Release();
-
- TestCodecCallbacks(base_interface, codec_interface, video_channel,
- forced_codec_width, forced_codec_height);
-
- ViETest::Log("Done!");
-
- // ***************************************************************
- // Testing finished. Tear down Video Engine
- // ***************************************************************
- EXPECT_EQ(0, base_interface->StopSend(video_channel));
- EXPECT_EQ(0, base_interface->StopReceive(video_channel));
- EXPECT_EQ(0, render_interface->StopRender(capture_id));
- EXPECT_EQ(0, render_interface->StopRender(video_channel));
- EXPECT_EQ(0, render_interface->RemoveRenderer(capture_id));
- EXPECT_EQ(0, render_interface->RemoveRenderer(video_channel));
- EXPECT_EQ(0, capture_interface->DisconnectCaptureDevice(video_channel));
- EXPECT_EQ(0, base_interface->DeleteChannel(video_channel));
-}
-
-void SetSendCodec(webrtc::VideoCodecType of_type,
- webrtc::ViECodec* codec_interface,
- int video_channel,
- int forced_codec_width,
- int forced_codec_height) {
- webrtc::VideoCodec codec;
- bool ok;
- EXPECT_TRUE(ok = FindSpecificCodec(of_type, codec_interface, &codec));
- if (!ok) {
- return;
- }
-
- SetSuitableResolution(&codec, forced_codec_width, forced_codec_height);
- EXPECT_EQ(0, codec_interface->SetSendCodec(video_channel, codec));
-}
diff --git a/trunk/src/video_engine/test/auto_test/primitives/codec_primitives.h b/trunk/src/video_engine/test/auto_test/primitives/codec_primitives.h
deleted file mode 100644
index e778044..0000000
--- a/trunk/src/video_engine/test/auto_test/primitives/codec_primitives.h
+++ /dev/null
@@ -1,126 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_CODEC_PRIMITIVES_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_CODEC_PRIMITIVES_H_
-
-#include "video_engine/include/vie_codec.h"
-#include "video_engine/include/vie_image_process.h"
-#include "video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "video_engine/test/auto_test/primitives/general_primitives.h"
-
-class TbInterfaces;
-
-// Tests that a codec actually renders frames by registering a basic
-// render effect filter on the codec and then running it. This test is
-// quite lenient on the number of frames that get rendered, so it should not
-// be seen as a end-user-visible quality measure - it is more a sanity check
-// that the codec at least gets some frames through.
-
-// The codec resolution can be forced by specifying the forced* variables
-// (pass in kDoNotForceResolution if you don't care).
-void TestCodecs(const TbInterfaces& interfaces,
- int capture_id,
- int video_channel,
- int forced_codec_width,
- int forced_codec_height);
-
-// This helper function will set the send codec in the codec interface to a
-// codec of the specified type. It will generate a test failure if we do not
-// support the provided codec type.
-
-// The codec resolution can be forced by specifying the forced* variables
-// (pass in kDoNotForceResolution if you don't care).
-void SetSendCodec(webrtc::VideoCodecType of_type,
- webrtc::ViECodec* codec_interface,
- int video_channel,
- int forced_codec_width,
- int forced_codec_height);
-
-class ViEAutotestCodecObserver: public webrtc::ViEEncoderObserver,
- public webrtc::ViEDecoderObserver {
- public:
- int incomingCodecCalled;
- int incomingRatecalled;
- int outgoingRatecalled;
-
- unsigned char lastPayloadType;
- unsigned short lastWidth;
- unsigned short lastHeight;
-
- unsigned int lastOutgoingFramerate;
- unsigned int lastOutgoingBitrate;
- unsigned int lastIncomingFramerate;
- unsigned int lastIncomingBitrate;
-
- webrtc::VideoCodec incomingCodec;
-
- ViEAutotestCodecObserver() {
- incomingCodecCalled = 0;
- incomingRatecalled = 0;
- outgoingRatecalled = 0;
- lastPayloadType = 0;
- lastWidth = 0;
- lastHeight = 0;
- lastOutgoingFramerate = 0;
- lastOutgoingBitrate = 0;
- lastIncomingFramerate = 0;
- lastIncomingBitrate = 0;
- memset(&incomingCodec, 0, sizeof(incomingCodec));
- }
- virtual void IncomingCodecChanged(const int videoChannel,
- const webrtc::VideoCodec& videoCodec) {
- incomingCodecCalled++;
- lastPayloadType = videoCodec.plType;
- lastWidth = videoCodec.width;
- lastHeight = videoCodec.height;
-
- memcpy(&incomingCodec, &videoCodec, sizeof(videoCodec));
- }
-
- virtual void IncomingRate(const int videoChannel,
- const unsigned int framerate,
- const unsigned int bitrate) {
- incomingRatecalled++;
- lastIncomingFramerate += framerate;
- lastIncomingBitrate += bitrate;
- }
-
- virtual void OutgoingRate(const int videoChannel,
- const unsigned int framerate,
- const unsigned int bitrate) {
- outgoingRatecalled++;
- lastOutgoingFramerate += framerate;
- lastOutgoingBitrate += bitrate;
- }
-
- virtual void RequestNewKeyFrame(const int videoChannel) {
- }
-};
-
-class FrameCounterEffectFilter : public webrtc::ViEEffectFilter
-{
- public:
- int numFrames;
- FrameCounterEffectFilter() {
- numFrames = 0;
- }
- virtual ~FrameCounterEffectFilter() {
- }
-
- virtual int Transform(int size, unsigned char* frameBuffer,
- unsigned int timeStamp90KHz, unsigned int width,
- unsigned int height) {
- numFrames++;
- return 0;
- }
-};
-
-#endif // WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_CODEC_PRIMITIVES_H_
diff --git a/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives.cc b/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives.cc
deleted file mode 100644
index 6767cc6..0000000
--- a/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives.cc
+++ /dev/null
@@ -1,513 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cassert>
-#include <string>
-
-#include "modules/video_capture/main/interface/video_capture_factory.h"
-#include "system_wrappers/interface/tick_util.h"
-#include "testsupport/fileutils.h"
-#include "testsupport/frame_reader.h"
-#include "testsupport/frame_writer.h"
-#include "video_engine/test/auto_test/helpers/vie_to_file_renderer.h"
-#include "video_engine/test/auto_test/interface/tb_interfaces.h"
-#include "video_engine/test/auto_test/interface/tb_external_transport.h"
-#include "video_engine/test/auto_test/interface/vie_autotest.h"
-#include "video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "video_engine/test/auto_test/primitives/framedrop_primitives.h"
-#include "video_engine/test/auto_test/primitives/general_primitives.h"
-
-// Tracks which frames are created on the local side and reports them to the
-// FrameDropDetector class.
-class CreatedTimestampEffectFilter : public webrtc::ViEEffectFilter {
- public:
- explicit CreatedTimestampEffectFilter(FrameDropDetector* frame_drop_detector)
- : frame_drop_detector_(frame_drop_detector) {}
- virtual ~CreatedTimestampEffectFilter() {}
- virtual int Transform(int size, unsigned char* frameBuffer,
- unsigned int timeStamp90KHz, unsigned int width,
- unsigned int height) {
- frame_drop_detector_->ReportFrameState(FrameDropDetector::kCreated,
- timeStamp90KHz);
- return 0;
- }
-
- private:
- FrameDropDetector* frame_drop_detector_;
-};
-
-// Tracks which frames are sent in external transport on the local side
-// and reports them to the FrameDropDetector class.
-class FrameSentCallback : public SendFrameCallback {
- public:
- explicit FrameSentCallback(FrameDropDetector* frame_drop_detector)
- : frame_drop_detector_(frame_drop_detector) {}
- virtual ~FrameSentCallback() {}
- virtual void FrameSent(unsigned int rtp_timestamp) {
- frame_drop_detector_->ReportFrameState(FrameDropDetector::kSent,
- rtp_timestamp);
- }
-
- private:
- FrameDropDetector* frame_drop_detector_;
-};
-
-// Tracks which frames are received in external transport on the remote side
-// and reports them to the FrameDropDetector class.
-class FrameReceivedCallback : public ReceiveFrameCallback {
- public:
- explicit FrameReceivedCallback(FrameDropDetector* frame_drop_detector)
- : frame_drop_detector_(frame_drop_detector) {}
- virtual ~FrameReceivedCallback() {}
- virtual void FrameReceived(unsigned int rtp_timestamp) {
- frame_drop_detector_->ReportFrameState(FrameDropDetector::kReceived,
- rtp_timestamp);
- }
-
- private:
- FrameDropDetector* frame_drop_detector_;
-};
-
-// Tracks when frames are decoded on the remote side (received from the
-// jitter buffer) and reports them to the FrameDropDetector class.
-class DecodedTimestampEffectFilter : public webrtc::ViEEffectFilter {
- public:
- explicit DecodedTimestampEffectFilter(FrameDropDetector* frame_drop_detector)
- : frame_drop_detector_(frame_drop_detector) {}
- virtual ~DecodedTimestampEffectFilter() {}
- virtual int Transform(int size, unsigned char* frameBuffer,
- unsigned int timeStamp90KHz, unsigned int width,
- unsigned int height) {
- frame_drop_detector_->ReportFrameState(FrameDropDetector::kDecoded,
- timeStamp90KHz);
- return 0;
- }
-
- private:
- FrameDropDetector* frame_drop_detector_;
-};
-
-void TestFullStack(const TbInterfaces& interfaces,
- int capture_id,
- int video_channel,
- int width,
- int height,
- int bit_rate_kbps,
- int packet_loss_percent,
- int network_delay_ms,
- FrameDropDetector* frame_drop_detector) {
- webrtc::VideoEngine *video_engine_interface = interfaces.video_engine;
- webrtc::ViEBase *base_interface = interfaces.base;
- webrtc::ViECapture *capture_interface = interfaces.capture;
- webrtc::ViERender *render_interface = interfaces.render;
- webrtc::ViECodec *codec_interface = interfaces.codec;
- webrtc::ViENetwork *network_interface = interfaces.network;
-
- // ***************************************************************
- // Engine ready. Begin testing class
- // ***************************************************************
- webrtc::VideoCodec video_codec;
- memset(&video_codec, 0, sizeof (webrtc::VideoCodec));
-
- // Set up all receive codecs. This basically setup the codec interface
- // to be able to recognize all receive codecs based on payload type.
- for (int idx = 0; idx < codec_interface->NumberOfCodecs(); idx++) {
- EXPECT_EQ(0, codec_interface->GetCodec(idx, video_codec));
- SetSuitableResolution(&video_codec, width, height);
-
- EXPECT_EQ(0, codec_interface->SetReceiveCodec(video_channel, video_codec));
- }
-
- // Configure External transport to simulate network interference:
- TbExternalTransport external_transport(*interfaces.network);
- external_transport.SetPacketLoss(packet_loss_percent);
- external_transport.SetNetworkDelay(network_delay_ms);
-
- FrameSentCallback frame_sent_callback(frame_drop_detector);
- FrameReceivedCallback frame_received_callback(frame_drop_detector);
- external_transport.RegisterSendFrameCallback(&frame_sent_callback);
- external_transport.RegisterReceiveFrameCallback(&frame_received_callback);
- EXPECT_EQ(0, network_interface->RegisterSendTransport(video_channel,
- external_transport));
- EXPECT_EQ(0, base_interface->StartReceive(video_channel));
-
- // Setup only the VP8 codec, which is what we'll use.
- webrtc::VideoCodec codec;
- EXPECT_TRUE(FindSpecificCodec(webrtc::kVideoCodecVP8, codec_interface,
- &codec));
- codec.startBitrate = bit_rate_kbps;
- codec.maxBitrate = bit_rate_kbps;
- codec.width = width;
- codec.height = height;
- EXPECT_EQ(0, codec_interface->SetSendCodec(video_channel, codec));
-
- webrtc::ViEImageProcess *image_process =
- webrtc::ViEImageProcess::GetInterface(video_engine_interface);
- EXPECT_TRUE(image_process);
-
- // Setup the effect filters
- CreatedTimestampEffectFilter create_filter(frame_drop_detector);
- EXPECT_EQ(0, image_process->RegisterSendEffectFilter(video_channel,
- create_filter));
- DecodedTimestampEffectFilter decode_filter(frame_drop_detector);
- EXPECT_EQ(0, image_process->RegisterRenderEffectFilter(video_channel,
- decode_filter));
- // Send video.
- EXPECT_EQ(0, base_interface->StartSend(video_channel));
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- // Cleanup.
- EXPECT_EQ(0, image_process->DeregisterSendEffectFilter(video_channel));
- EXPECT_EQ(0, image_process->DeregisterRenderEffectFilter(video_channel));
- image_process->Release();
- ViETest::Log("Done!");
-
- WebRtc_Word32 num_rtp_packets = 0;
- WebRtc_Word32 num_dropped_packets = 0;
- WebRtc_Word32 num_rtcp_packets = 0;
- external_transport.GetStats(num_rtp_packets, num_dropped_packets,
- num_rtcp_packets);
- ViETest::Log("RTP packets : %5d", num_rtp_packets);
- ViETest::Log("Dropped packets: %5d", num_dropped_packets);
- ViETest::Log("RTCP packets : %5d", num_rtcp_packets);
-
- // ***************************************************************
- // Testing finished. Tear down Video Engine
- // ***************************************************************
- EXPECT_EQ(0, base_interface->StopSend(video_channel));
- EXPECT_EQ(0, base_interface->StopReceive(video_channel));
- EXPECT_EQ(0, network_interface->DeregisterSendTransport(video_channel));
- EXPECT_EQ(0, render_interface->StopRender(capture_id));
- EXPECT_EQ(0, render_interface->StopRender(video_channel));
- EXPECT_EQ(0, render_interface->RemoveRenderer(capture_id));
- EXPECT_EQ(0, render_interface->RemoveRenderer(video_channel));
- EXPECT_EQ(0, capture_interface->DisconnectCaptureDevice(video_channel));
- EXPECT_EQ(0, base_interface->DeleteChannel(video_channel));
-}
-
-void FixOutputFileForComparison(const std::string& output_file,
- int frame_length_in_bytes,
- const std::vector<Frame*>& frames) {
- webrtc::test::FrameReaderImpl frame_reader(output_file,
- frame_length_in_bytes);
- const std::string temp_file = output_file + ".fixed";
- webrtc::test::FrameWriterImpl frame_writer(temp_file, frame_length_in_bytes);
- frame_reader.Init();
- frame_writer.Init();
-
- ASSERT_FALSE(frames.front()->dropped_at_render) << "It should not be "
- "possible to drop the first frame. Both because we don't have anything "
- "useful to fill that gap with and it is impossible to detect it without "
- "any previous timestamps to compare with.";
-
- WebRtc_UWord8* last_frame_data = new WebRtc_UWord8[frame_length_in_bytes];
-
- // Process the file and write frame duplicates for all dropped frames.
- for (std::vector<Frame*>::const_iterator it = frames.begin();
- it != frames.end(); ++it) {
- if ((*it)->dropped_at_render) {
- // Write the previous frame to the output file:
- EXPECT_TRUE(frame_writer.WriteFrame(last_frame_data));
- } else {
- EXPECT_TRUE(frame_reader.ReadFrame(last_frame_data));
- EXPECT_TRUE(frame_writer.WriteFrame(last_frame_data));
- }
- }
- delete[] last_frame_data;
- frame_reader.Close();
- frame_writer.Close();
- ASSERT_EQ(0, std::remove(output_file.c_str()));
- ASSERT_EQ(0, std::rename(temp_file.c_str(), output_file.c_str()));
-}
-
-void FrameDropDetector::ReportFrameState(State state, unsigned int timestamp) {
- dirty_ = true;
- switch (state) {
- case kCreated: {
- int number = created_frames_vector_.size();
- Frame* frame = new Frame(number, timestamp);
- frame->created_timestamp_in_us_ =
- webrtc::TickTime::MicrosecondTimestamp();
- created_frames_vector_.push_back(frame);
- created_frames_[timestamp] = frame;
- num_created_frames_++;
- break;
- }
- case kSent:
- sent_frames_[timestamp] = webrtc::TickTime::MicrosecondTimestamp();
- if (timestamp_diff_ == 0) {
- // When the first created frame arrives we calculate the fixed
- // difference between the timestamps of the frames entering and leaving
- // the encoder. This diff is used to identify the frames from the
- // created_frames_ map.
- timestamp_diff_ =
- timestamp - created_frames_vector_.front()->frame_timestamp_;
- }
- num_sent_frames_++;
- break;
- case kReceived:
- received_frames_[timestamp] = webrtc::TickTime::MicrosecondTimestamp();
- num_received_frames_++;
- break;
- case kDecoded:
- decoded_frames_[timestamp] = webrtc::TickTime::MicrosecondTimestamp();
- num_decoded_frames_++;
- break;
- case kRendered:
- rendered_frames_[timestamp] = webrtc::TickTime::MicrosecondTimestamp();
- num_rendered_frames_++;
- break;
- }
-}
-
-void FrameDropDetector::CalculateResults() {
- // Fill in all fields of the Frame objects in the created_frames_ map.
- // Iterate over the maps from converted timestamps to the arrival timestamps.
- std::map<unsigned int, int64_t>::const_iterator it;
- for (it = sent_frames_.begin(); it != sent_frames_.end(); ++it) {
- int created_timestamp = it->first - timestamp_diff_;
- created_frames_[created_timestamp]->sent_timestamp_in_us_ = it->second;
- }
- for (it = received_frames_.begin(); it != received_frames_.end(); ++it) {
- int created_timestamp = it->first - timestamp_diff_;
- created_frames_[created_timestamp]->received_timestamp_in_us_ = it->second;
- }
- for (it = decoded_frames_.begin(); it != decoded_frames_.end(); ++it) {
- int created_timestamp = it->first - timestamp_diff_;
- created_frames_[created_timestamp]->decoded_timestamp_in_us_ =it->second;
- }
- for (it = rendered_frames_.begin(); it != rendered_frames_.end(); ++it) {
- int created_timestamp = it->first - timestamp_diff_;
- created_frames_[created_timestamp]->rendered_timestamp_in_us_ = it->second;
- }
- // Find out where the frames were not present in the different states.
- dropped_frames_at_send_ = 0;
- dropped_frames_at_receive_ = 0;
- dropped_frames_at_decode_ = 0;
- dropped_frames_at_render_ = 0;
- for (std::vector<Frame*>::const_iterator it = created_frames_vector_.begin();
- it != created_frames_vector_.end(); ++it) {
- int encoded_timestamp = (*it)->frame_timestamp_ + timestamp_diff_;
- if (sent_frames_.find(encoded_timestamp) == sent_frames_.end()) {
- (*it)->dropped_at_send = true;
- dropped_frames_at_send_++;
- }
- if (received_frames_.find(encoded_timestamp) == received_frames_.end()) {
- (*it)->dropped_at_receive = true;
- dropped_frames_at_receive_++;
- }
- if (decoded_frames_.find(encoded_timestamp) == decoded_frames_.end()) {
- (*it)->dropped_at_decode = true;
- dropped_frames_at_decode_++;
- }
- if (rendered_frames_.find(encoded_timestamp) == rendered_frames_.end()) {
- (*it)->dropped_at_render = true;
- dropped_frames_at_render_++;
- }
- }
- dirty_ = false;
-}
-
-void FrameDropDetector::PrintReport() {
- assert(!dirty_);
- ViETest::Log("Frame Drop Detector report:");
- ViETest::Log(" Created frames: %ld", created_frames_.size());
- ViETest::Log(" Sent frames: %ld", sent_frames_.size());
- ViETest::Log(" Received frames: %ld", received_frames_.size());
- ViETest::Log(" Decoded frames: %ld", decoded_frames_.size());
- ViETest::Log(" Rendered frames: %ld", rendered_frames_.size());
-
- // Display all frames and stats for them:
- long last_created = 0;
- long last_sent = 0;
- long last_received = 0;
- long last_decoded = 0;
- long last_rendered = 0;
- ViETest::Log("\nDeltas between sent frames and drop status:");
- ViETest::Log("Unit: Microseconds");
- ViETest::Log("Frame Created Sent Received Decoded Rendered "
- "Dropped at Dropped at Dropped at Dropped at");
- ViETest::Log(" nbr delta delta delta delta delta "
- " Send? Receive? Decode? Render?");
- for (std::vector<Frame*>::const_iterator it = created_frames_vector_.begin();
- it != created_frames_vector_.end(); ++it) {
- int created_delta =
- static_cast<int>((*it)->created_timestamp_in_us_ - last_created);
- int sent_delta = (*it)->dropped_at_send ? -1 :
- static_cast<int>((*it)->sent_timestamp_in_us_ - last_sent);
- int received_delta = (*it)->dropped_at_receive ? -1 :
- static_cast<int>((*it)->received_timestamp_in_us_ - last_received);
- int decoded_delta = (*it)->dropped_at_decode ? -1 :
- static_cast<int>((*it)->decoded_timestamp_in_us_ - last_decoded);
- int rendered_delta = (*it)->dropped_at_render ? -1 :
- static_cast<int>((*it)->rendered_timestamp_in_us_ - last_rendered);
-
- // Set values to -1 for the first frame:
- if ((*it)->number_ == 0) {
- created_delta = -1;
- sent_delta = -1;
- received_delta = -1;
- decoded_delta = -1;
- rendered_delta = -1;
- }
- ViETest::Log("%5d %8d %8d %8d %8d %8d %10s %10s %10s %10s",
- (*it)->number_,
- created_delta,
- sent_delta,
- received_delta,
- decoded_delta,
- rendered_delta,
- (*it)->dropped_at_send ? "DROPPED" : " ",
- (*it)->dropped_at_receive ? "DROPPED" : " ",
- (*it)->dropped_at_decode ? "DROPPED" : " ",
- (*it)->dropped_at_render ? "DROPPED" : " ");
- last_created = (*it)->created_timestamp_in_us_;
- if (!(*it)->dropped_at_send) {
- last_sent = (*it)->sent_timestamp_in_us_;
- }
- if (!(*it)->dropped_at_receive) {
- last_received = (*it)->received_timestamp_in_us_;
- }
- if (!(*it)->dropped_at_decode) {
- last_decoded = (*it)->decoded_timestamp_in_us_;
- }
- if (!(*it)->dropped_at_render) {
- last_rendered = (*it)->rendered_timestamp_in_us_;
- }
- }
- ViETest::Log("\nLatency between states (-1 means N/A because of drop):");
- ViETest::Log("Unit: Microseconds");
- ViETest::Log("Frame Created Sent Received Decoded Total "
- " Total");
- ViETest::Log(" nbr ->Sent ->Received ->Decoded ->Rendered latency "
- " latency");
- ViETest::Log(" (incl network)"
- "(excl network)");
- for (std::vector<Frame*>::const_iterator it = created_frames_vector_.begin();
- it != created_frames_vector_.end(); ++it) {
- int created_to_sent = (*it)->dropped_at_send ? -1 :
- static_cast<int>((*it)->sent_timestamp_in_us_ -
- (*it)->created_timestamp_in_us_);
- int sent_to_received = (*it)->dropped_at_receive ? -1 :
- static_cast<int>((*it)->received_timestamp_in_us_ -
- (*it)->sent_timestamp_in_us_);
- int received_to_decoded = (*it)->dropped_at_decode ? -1 :
- static_cast<int>((*it)->decoded_timestamp_in_us_ -
- (*it)->received_timestamp_in_us_);
- int decoded_to_render = (*it)->dropped_at_render ? -1 :
- static_cast<int>((*it)->rendered_timestamp_in_us_ -
- (*it)->decoded_timestamp_in_us_);
- int total_latency_incl_network = (*it)->dropped_at_render ? -1 :
- static_cast<int>((*it)->rendered_timestamp_in_us_ -
- (*it)->created_timestamp_in_us_);
- int total_latency_excl_network = (*it)->dropped_at_render ? -1 :
- static_cast<int>((*it)->rendered_timestamp_in_us_ -
- (*it)->created_timestamp_in_us_ - sent_to_received);
- ViETest::Log("%5d %9d %9d %9d %9d %12d %12d",
- (*it)->number_,
- created_to_sent,
- sent_to_received,
- received_to_decoded,
- decoded_to_render,
- total_latency_incl_network,
- total_latency_excl_network);
- }
- // Find and print the dropped frames.
- ViETest::Log("\nTotal # dropped frames at:");
- ViETest::Log(" Send : %d", dropped_frames_at_send_);
- ViETest::Log(" Receive: %d", dropped_frames_at_receive_);
- ViETest::Log(" Decode : %d", dropped_frames_at_decode_);
- ViETest::Log(" Render : %d", dropped_frames_at_render_);
-}
-
-void FrameDropDetector::PrintDebugDump() {
- assert(!dirty_);
- ViETest::Log("\nPrintDebugDump: Frame objects:");
- ViETest::Log("Frame FrTimeStamp Created Sent Received Decoded"
- " Rendered ");
- for (std::vector<Frame*>::const_iterator it = created_frames_vector_.begin();
- it != created_frames_vector_.end(); ++it) {
- ViETest::Log("%5d %11d %11d %11d %11d %11d %11d",
- (*it)->number_,
- (*it)->frame_timestamp_,
- (*it)->created_timestamp_in_us_,
- (*it)->sent_timestamp_in_us_,
- (*it)->received_timestamp_in_us_,
- (*it)->decoded_timestamp_in_us_,
- (*it)->rendered_timestamp_in_us_);
- }
- std::vector<int> mismatch_frame_num_list;
- for (std::vector<Frame*>::const_iterator it = created_frames_vector_.begin();
- it != created_frames_vector_.end(); ++it) {
- if ((*it)->dropped_at_render != (*it)->dropped_at_decode) {
- mismatch_frame_num_list.push_back((*it)->number_);
- }
- }
- if (mismatch_frame_num_list.size() > 0) {
- ViETest::Log("\nDecoded/Rendered mismatches:");
- ViETest::Log("Frame FrTimeStamp Created Sent Received "
- "Decoded Rendered ");
- for (std::vector<int>::const_iterator it = mismatch_frame_num_list.begin();
- it != mismatch_frame_num_list.end(); ++it) {
- Frame* frame = created_frames_vector_[*it];
- ViETest::Log("%5d %11d %11d %11d %11d %11d %11d",
- frame->number_,
- frame->frame_timestamp_,
- frame->created_timestamp_in_us_,
- frame->sent_timestamp_in_us_,
- frame->received_timestamp_in_us_,
- frame->decoded_timestamp_in_us_,
- frame->rendered_timestamp_in_us_);
- }
- }
-
- ViETest::Log("\nReportFrameState method invocations:");
- ViETest::Log(" Created : %d", num_created_frames_);
- ViETest::Log(" Send : %d", num_sent_frames_);
- ViETest::Log(" Received: %d", num_received_frames_);
- ViETest::Log(" Decoded : %d", num_decoded_frames_);
- ViETest::Log(" Rendered: %d", num_rendered_frames_);
-}
-
-const std::vector<Frame*>& FrameDropDetector::GetAllFrames() {
- assert(!dirty_);
- return created_frames_vector_;
-}
-
-int FrameDropDetector::GetNumberOfFramesDroppedAt(State state) {
- assert(!dirty_);
- switch (state) {
- case kSent:
- return dropped_frames_at_send_;
- case kReceived:
- return dropped_frames_at_receive_;
- case kDecoded:
- return dropped_frames_at_decode_;
- case kRendered:
- return dropped_frames_at_render_;
- default:
- return 0;
- }
-}
-
-int FrameDropMonitoringRemoteFileRenderer::DeliverFrame(
- unsigned char *buffer, int buffer_size, uint32_t time_stamp,
- int64_t render_time) {
- // Register that this frame has been rendered:
- frame_drop_detector_->ReportFrameState(FrameDropDetector::kRendered,
- time_stamp);
- return ViEToFileRenderer::DeliverFrame(buffer, buffer_size,
- time_stamp, render_time);
-}
-
-int FrameDropMonitoringRemoteFileRenderer::FrameSizeChange(
- unsigned int width, unsigned int height, unsigned int number_of_streams) {
- return ViEToFileRenderer::FrameSizeChange(width, height, number_of_streams);
-}
diff --git a/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives.h b/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives.h
deleted file mode 100644
index 80575c3..0000000
--- a/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives.h
+++ /dev/null
@@ -1,229 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_SOURCE_FRAMEDROP_PRIMITIVES_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_SOURCE_FRAMEDROP_PRIMITIVES_H_
-
-#include <map>
-#include <vector>
-
-#include "video_engine/include/vie_codec.h"
-#include "video_engine/include/vie_image_process.h"
-#include "video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "video_engine/test/auto_test/helpers/vie_to_file_renderer.h"
-
-class FrameDropDetector;
-class TbInterfaces;
-
-// Initializes the Video engine and its components, runs video playback using
-// for KAutoTestSleepTimeMs milliseconds, then shuts down everything.
-// The bit rate and packet loss parameters should be configured so that
-// frames are dropped, in order to test the frame drop detection that is
-// performed by the FrameDropDetector class.
-void TestFullStack(const TbInterfaces& interfaces,
- int capture_id,
- int video_channel,
- int width,
- int height,
- int bit_rate_kbps,
- int packet_loss_percent,
- int network_delay_ms,
- FrameDropDetector* frame_drop_detector);
-
-// A frame in a video file. The four different points in the stack when
-// register the frame state are (in time order): created, transmitted, decoded,
-// rendered.
-class Frame {
- public:
- Frame(int number, unsigned int timestamp)
- : number_(number),
- frame_timestamp_(timestamp),
- created_timestamp_in_us_(-1),
- sent_timestamp_in_us_(-1),
- received_timestamp_in_us_(-1),
- decoded_timestamp_in_us_(-1),
- rendered_timestamp_in_us_(-1),
- dropped_at_send(false),
- dropped_at_receive(false),
- dropped_at_decode(false),
- dropped_at_render(false) {}
-
- // Frame number, starting at 0.
- int number_;
-
- // Frame timestamp, that is used by Video Engine and RTP headers and set when
- // the frame is sent into the stack.
- unsigned int frame_timestamp_;
-
- // Timestamps for our measurements of when the frame is in different states.
- int64_t created_timestamp_in_us_;
- int64_t sent_timestamp_in_us_;
- int64_t received_timestamp_in_us_;
- int64_t decoded_timestamp_in_us_;
- int64_t rendered_timestamp_in_us_;
-
- // Where the frame was dropped (more than one may be true).
- bool dropped_at_send;
- bool dropped_at_receive;
- bool dropped_at_decode;
- bool dropped_at_render;
-};
-
-// Fixes the output file by copying the last successful frame into the place
-// where the dropped frame would be, for all dropped frames (if any).
-// This method will not be able to fix data for the first frame if that is
-// dropped, since there'll be no previous frame to copy. This case should never
-// happen because of encoder frame dropping at least.
-// Parameters:
-// output_file The output file to modify (pad with frame copies
-// for all dropped frames)
-// frame_length_in_bytes Byte length of each frame.
-// frames A vector of all Frame objects. Must be sorted by
-// frame number. If empty this method will do nothing.
-void FixOutputFileForComparison(const std::string& output_file,
- int frame_length_in_bytes,
- const std::vector<Frame*>& frames);
-
-// Handles statistics about dropped frames. Frames travel through the stack
-// with different timestamps. The frames created and sent to the encoder have
-// one timestamp on the sending side while the decoded/rendered frames have
-// another timestamp on the receiving side. The difference between these
-// timestamps is fixed, which we can use to identify the frames when they
-// arrive, since the FrameDropDetector class gets data reported from both sides.
-// The four different points in the stack when this class examines the frame
-// states are (in time order): created, sent, received, decoded, rendered.
-//
-// The flow can be visualized like this:
-//
-// Created Sent Received Decoded Rendered
-// +-------+ | +-------+ | +---------+ | +------+ +-------+ | +--------+
-// |Capture| | |Encoder| | | Ext. | | |Jitter| |Decoder| | | Ext. |
-// | device|---->| |-->|transport|-->|buffer|->| |---->|renderer|
-// +-------+ +-------+ +---------+ +------+ +-------+ +--------+
-//
-// This class has no intention of being thread-safe.
-class FrameDropDetector {
- public:
- enum State {
- // A frame being created, i.e. sent to the encoder; the first step of
- // a frame's life cycle. This timestamp becomes the frame timestamp in the
- // Frame objects.
- kCreated,
- // A frame being sent in external transport (to the simulated network). This
- // timestamp differs from the one in the Created state by a constant diff.
- kSent,
- // A frame being received in external transport (from the simulated
- // network). This timestamp differs from the one in the Created state by a
- // constant diff.
- kReceived,
- // A frame that has been decoded in the decoder. This timestamp differs
- // from the one in the Created state by a constant diff.
- kDecoded,
- // A frame that has been rendered; the last step of a frame's life cycle.
- // This timestamp differs from the one in the Created state by a constant
- // diff.
- kRendered
- };
-
- FrameDropDetector()
- : dirty_(true),
- dropped_frames_at_send_(0),
- dropped_frames_at_receive_(0),
- dropped_frames_at_decode_(0),
- dropped_frames_at_render_(0),
- num_created_frames_(0),
- num_sent_frames_(0),
- num_received_frames_(0),
- num_decoded_frames_(0),
- num_rendered_frames_(0),
- timestamp_diff_(0) {}
-
- // Reports a frame has reached a state in the frame life cycle.
- void ReportFrameState(State state, unsigned int timestamp);
-
- // Uses all the gathered timestamp information to calculate which frames have
- // been dropped during the test and where they were dropped. Not until
- // this method has been executed, the Frame objects will have all fields
- // filled with the proper timestamp information.
- void CalculateResults();
-
- // Calculates the number of frames have been registered as dropped at the
- // specified state of the frame life cycle.
- // CalculateResults() must be called before calling this method.
- int GetNumberOfFramesDroppedAt(State state);
-
- // Gets a vector of all the created frames.
- // CalculateResults() must be called before calling this method to have all
- // fields of the Frame objects to represent the current state.
- const std::vector<Frame*>& GetAllFrames();
-
- // Prints a detailed report about all the different frame states and which
- // ones are detected as dropped, using ViETest::Log.
- // CalculateResults() must be called before calling this method.
- void PrintReport();
-
- // Prints all the timestamp maps. Mainly used for debugging purposes to find
- // missing timestamps.
- void PrintDebugDump();
- private:
- // Will be false until CalculateResults() is called. Switches to true
- // as soon as new timestamps are reported using ReportFrameState().
- bool dirty_;
-
- // Map of frame creation timestamps to all Frame objects.
- std::map<unsigned int, Frame*> created_frames_;
-
- // Maps converted frame timestamps (differ from creation timestamp) to the
- // time they arrived in the different states of the frame's life cycle.
- std::map<unsigned int, int64_t> sent_frames_;
- std::map<unsigned int, int64_t> received_frames_;
- std::map<unsigned int, int64_t> decoded_frames_;
- std::map<unsigned int, int64_t> rendered_frames_;
-
- // A vector with the frames sorted in their created order.
- std::vector<Frame*> created_frames_vector_;
-
- // Statistics.
- int dropped_frames_at_send_;
- int dropped_frames_at_receive_;
- int dropped_frames_at_decode_;
- int dropped_frames_at_render_;
-
- int num_created_frames_;
- int num_sent_frames_;
- int num_received_frames_;
- int num_decoded_frames_;
- int num_rendered_frames_;
-
- // The constant diff between the created and transmitted frames, since their
- // timestamps are converted.
- unsigned int timestamp_diff_;
-};
-
-// Tracks which frames are received on the remote side and reports back to the
-// FrameDropDetector class when they are rendered.
-class FrameDropMonitoringRemoteFileRenderer : public ViEToFileRenderer {
- public:
- explicit FrameDropMonitoringRemoteFileRenderer(
- FrameDropDetector* frame_drop_detector)
- : frame_drop_detector_(frame_drop_detector) {}
- virtual ~FrameDropMonitoringRemoteFileRenderer() {}
-
- // Implementation of ExternalRenderer:
- int FrameSizeChange(unsigned int width, unsigned int height,
- unsigned int number_of_streams);
- int DeliverFrame(unsigned char* buffer, int buffer_size,
- uint32_t time_stamp,
- int64_t render_time);
- private:
- FrameDropDetector* frame_drop_detector_;
-};
-
-#endif // WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_SOURCE_FRAMEDROP_PRIMITIVES_H_
diff --git a/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives_unittest.cc b/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives_unittest.cc
deleted file mode 100644
index f4cc390..0000000
--- a/trunk/src/video_engine/test/auto_test/primitives/framedrop_primitives_unittest.cc
+++ /dev/null
@@ -1,96 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "framedrop_primitives.h"
-
-#include <cstdio>
-#include <vector>
-
-#include "gtest/gtest.h"
-#include "testsupport/fileutils.h"
-#include "testsupport/frame_reader.h"
-#include "testsupport/frame_writer.h"
-
-namespace webrtc {
-
-const std::string kOutputFilename = "temp_outputfile.tmp";
-const int kFrameLength = 1000;
-
-class FrameDropPrimitivesTest: public testing::Test {
- protected:
- FrameDropPrimitivesTest() {}
- virtual ~FrameDropPrimitivesTest() {}
- void SetUp() {
- // Cleanup any previous output file.
- std::remove(kOutputFilename.c_str());
- }
- void TearDown() {
- // Cleanup the temporary file.
- std::remove(kOutputFilename.c_str());
- }
-};
-
-TEST_F(FrameDropPrimitivesTest, FixOutputFileForComparison) {
- // Create test frame objects, where the second and fourth frame is marked
- // as dropped at rendering.
- std::vector<Frame*> frames;
- Frame first_frame(0, kFrameLength);
- Frame second_frame(0, kFrameLength);
- Frame third_frame(0, kFrameLength);
- Frame fourth_frame(0, kFrameLength);
-
- second_frame.dropped_at_render = true;
- fourth_frame.dropped_at_render = true;
-
- frames.push_back(&first_frame);
- frames.push_back(&second_frame);
- frames.push_back(&third_frame);
- frames.push_back(&fourth_frame);
-
- // Prepare data for the first and third frames:
- WebRtc_UWord8 first_frame_data[kFrameLength];
- memset(first_frame_data, 5, kFrameLength); // Fill it with 5's to identify.
- WebRtc_UWord8 third_frame_data[kFrameLength];
- memset(third_frame_data, 7, kFrameLength); // Fill it with 7's to identify.
-
- // Write the first and third frames to the temporary file. This means the fix
- // method should add two frames of data by filling the file with data from
- // the first and third frames after executing.
- webrtc::test::FrameWriterImpl frame_writer(kOutputFilename, kFrameLength);
- EXPECT_TRUE(frame_writer.Init());
- EXPECT_TRUE(frame_writer.WriteFrame(first_frame_data));
- EXPECT_TRUE(frame_writer.WriteFrame(third_frame_data));
- frame_writer.Close();
- EXPECT_EQ(2 * kFrameLength,
- static_cast<int>(webrtc::test::GetFileSize(kOutputFilename)));
-
- FixOutputFileForComparison(kOutputFilename, kFrameLength, frames);
-
- // Verify that the output file has correct size.
- EXPECT_EQ(4 * kFrameLength,
- static_cast<int>(webrtc::test::GetFileSize(kOutputFilename)));
-
- webrtc::test::FrameReaderImpl frame_reader(kOutputFilename, kFrameLength);
- frame_reader.Init();
- WebRtc_UWord8 read_buffer[kFrameLength];
- EXPECT_TRUE(frame_reader.ReadFrame(read_buffer));
- EXPECT_EQ(0, memcmp(read_buffer, first_frame_data, kFrameLength));
- EXPECT_TRUE(frame_reader.ReadFrame(read_buffer));
- EXPECT_EQ(0, memcmp(read_buffer, first_frame_data, kFrameLength));
-
- EXPECT_TRUE(frame_reader.ReadFrame(read_buffer));
- EXPECT_EQ(0, memcmp(read_buffer, third_frame_data, kFrameLength));
- EXPECT_TRUE(frame_reader.ReadFrame(read_buffer));
- EXPECT_EQ(0, memcmp(read_buffer, third_frame_data, kFrameLength));
-
- frame_reader.Close();
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/test/auto_test/primitives/general_primitives.cc b/trunk/src/video_engine/test/auto_test/primitives/general_primitives.cc
deleted file mode 100644
index 6198a64..0000000
--- a/trunk/src/video_engine/test/auto_test/primitives/general_primitives.cc
+++ /dev/null
@@ -1,130 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "general_primitives.h"
-
-#include "video_capture_factory.h"
-#include "vie_autotest.h"
-#include "vie_autotest_defines.h"
-#include "vie_to_file_renderer.h"
-
-void FindCaptureDeviceOnSystem(webrtc::ViECapture* capture,
- unsigned char* device_name,
- unsigned int device_name_length,
- int* device_id,
- webrtc::VideoCaptureModule** device_video) {
-
- bool capture_device_set = false;
- webrtc::VideoCaptureModule::DeviceInfo *dev_info =
- webrtc::VideoCaptureFactory::CreateDeviceInfo(0);
-
- const unsigned int kMaxUniqueIdLength = 256;
- WebRtc_UWord8 unique_id[kMaxUniqueIdLength];
- memset(unique_id, 0, kMaxUniqueIdLength);
-
- for (unsigned int i = 0; i < dev_info->NumberOfDevices(); i++) {
- EXPECT_EQ(0, dev_info->GetDeviceName(i, device_name, device_name_length,
- unique_id, kMaxUniqueIdLength));
-
- *device_video = webrtc::VideoCaptureFactory::Create(4571, unique_id);
- EXPECT_TRUE(*device_video != NULL);
-
- (*device_video)->AddRef();
-
- int error = capture->AllocateCaptureDevice(**device_video, *device_id);
- if (error == 0) {
- ViETest::Log("Using capture device: %s, captureId: %d.",
- device_name, *device_id);
- capture_device_set = true;
- break;
- } else {
- (*device_video)->Release();
- (*device_video) = NULL;
- }
- }
- delete dev_info;
- EXPECT_TRUE(capture_device_set) << "Found no suitable camera on your system.";
-}
-
-void RenderInWindow(webrtc::ViERender* video_render_interface,
- int frame_provider_id,
- void* os_window,
- float z_index) {
- EXPECT_EQ(0,
- video_render_interface->AddRenderer(frame_provider_id, os_window,
- z_index, 0.0, 0.0, 1.0, 1.0));
- EXPECT_EQ(0, video_render_interface->StartRender(frame_provider_id));
-}
-
-void RenderToFile(webrtc::ViERender* renderer_interface,
- int frame_provider_id,
- ViEToFileRenderer *to_file_renderer) {
- EXPECT_EQ(0, renderer_interface->AddRenderer(
- frame_provider_id, webrtc::kVideoI420, to_file_renderer));
- EXPECT_EQ(0, renderer_interface->StartRender(frame_provider_id));
-}
-
-void StopAndRemoveRenderers(webrtc::ViEBase* base_interface,
- webrtc::ViERender* render_interface,
- int channel_id,
- int capture_id) {
- EXPECT_EQ(0, render_interface->StopRender(channel_id));
- EXPECT_EQ(0, render_interface->RemoveRenderer(channel_id));
- EXPECT_EQ(0, render_interface->RemoveRenderer(capture_id));
-}
-
-void ConfigureRtpRtcp(webrtc::ViERTP_RTCP* rtcp_interface,
- int video_channel) {
- EXPECT_EQ(0, rtcp_interface->SetRTCPStatus(video_channel,
- webrtc::kRtcpCompound_RFC4585));
- EXPECT_EQ(0, rtcp_interface->SetKeyFrameRequestMethod(
- video_channel, webrtc::kViEKeyFrameRequestPliRtcp));
- EXPECT_EQ(0, rtcp_interface->SetTMMBRStatus(video_channel, true));
-}
-
-bool FindSpecificCodec(webrtc::VideoCodecType of_type,
- webrtc::ViECodec* codec_interface,
- webrtc::VideoCodec* result) {
-
- memset(result, 0, sizeof(webrtc::VideoCodec));
-
- for (int i = 0; i < codec_interface->NumberOfCodecs(); i++) {
- webrtc::VideoCodec codec;
- memset(&codec, 0, sizeof(webrtc::VideoCodec));
- if (codec_interface->GetCodec(i, codec) != 0) {
- return false;
- }
- if (codec.codecType == of_type) {
- // Done
- *result = codec;
- return true;
- }
- }
- // Didn't find it
- return false;
-}
-
-void SetSuitableResolution(webrtc::VideoCodec* video_codec,
- int forced_codec_width,
- int forced_codec_height) {
- if (forced_codec_width != kDoNotForceResolution &&
- forced_codec_height != kDoNotForceResolution) {
- video_codec->width = forced_codec_width;
- video_codec->height = forced_codec_height;
- } else if (video_codec->codecType == webrtc::kVideoCodecI420) {
- // I420 is very bandwidth heavy, so limit it here.
- video_codec->width = 176;
- video_codec->height = 144;
- } else {
- // Otherwise go with 640x480.
- video_codec->width = 640;
- video_codec->height = 480;
- }
-}
diff --git a/trunk/src/video_engine/test/auto_test/primitives/general_primitives.h b/trunk/src/video_engine/test/auto_test/primitives/general_primitives.h
deleted file mode 100644
index 07eca0d..0000000
--- a/trunk/src/video_engine/test/auto_test/primitives/general_primitives.h
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_GENERAL_PRIMITIVES_H_
-#define WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_GENERAL_PRIMITIVES_H_
-
-class ViEToFileRenderer;
-
-#include "common_types.h"
-
-namespace webrtc {
-class VideoCaptureModule;
-class ViEBase;
-class ViECapture;
-class ViECodec;
-class ViERender;
-class ViERTP_RTCP;
-struct VideoCodec;
-}
-
-// This constant can be used as input to various functions to not force the
-// codec resolution.
-const int kDoNotForceResolution = 0;
-
-// Finds a suitable capture device (e.g. camera) on the current system
-// and allocates it. Details about the found device are filled into the out
-// parameters. If this operation fails, device_id is assigned a negative value
-// and number_of_errors is incremented.
-void FindCaptureDeviceOnSystem(webrtc::ViECapture* capture,
- unsigned char* device_name,
- const unsigned int kDeviceNameLength,
- int* device_id,
- webrtc::VideoCaptureModule** device_video);
-
-// Sets up rendering in a window previously created using a Window Manager
-// (See vie_window_manager_factory.h for more details on how to make one of
-// those). The frame provider id is a source of video frames, for instance
-// a capture device or a video channel.
-void RenderInWindow(webrtc::ViERender* video_render_interface,
- int frame_provider_id,
- void* os_window,
- float z_index);
-
-// Similar in function to RenderInWindow, this function instead renders to
-// a file using a to-file-renderer. The frame provider id is a source of
-// video frames, for instance a capture device or a video channel.
-void RenderToFile(webrtc::ViERender* renderer_interface,
- int frame_provider_id,
- ViEToFileRenderer* to_file_renderer);
-
-// Stops all rendering given the normal case that we have a capture device
-// and a video channel set up for rendering.
-void StopAndRemoveRenderers(webrtc::ViEBase* base_interface,
- webrtc::ViERender* render_interface,
- int channel_id,
- int capture_id);
-
-// Configures RTP-RTCP.
-void ConfigureRtpRtcp(webrtc::ViERTP_RTCP* rtcp_interface,
- int video_channel);
-
-// Finds a codec in the codec list. Returns true on success, false otherwise.
-// The resulting codec is filled into result on success but is zeroed out
-// on failure.
-bool FindSpecificCodec(webrtc::VideoCodecType of_type,
- webrtc::ViECodec* codec_interface,
- webrtc::VideoCodec* result);
-
-// Sets up the provided codec with a resolution that takes individual codec
-// quirks into account (except if the forced* variables are
-// != kDoNotForceResolution)
-void SetSuitableResolution(webrtc::VideoCodec* video_codec,
- int forced_codec_width,
- int forced_codec_height);
-
-#endif // WEBRTC_VIDEO_ENGINE_TEST_AUTO_TEST_PRIMITIVES_GENERAL_PRIMITIVES_H_
diff --git a/trunk/src/video_engine/test/auto_test/source/Android.mk b/trunk/src/video_engine/test/auto_test/source/Android.mk
deleted file mode 100644
index 9e19418..0000000
--- a/trunk/src/video_engine/test/auto_test/source/Android.mk
+++ /dev/null
@@ -1,65 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH:= $(call my-dir)
-
-# voice engine test app
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../android-webrtc.mk
-
-LOCAL_MODULE_TAGS := tests
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES:= \
- vie_autotest.cc \
- vie_autotest_android.cc \
- vie_autotest_base.cc \
- vie_autotest_capture.cc \
- vie_autotest_codec.cc \
- vie_autotest_encryption.cc \
- vie_autotest_file.cc \
- vie_autotest_image_process.cc \
- vie_autotest_loopback.cc \
- vie_autotest_network.cc \
- vie_autotest_render.cc \
- vie_autotest_rtp_rtcp.cc \
- vie_comparison_tests.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_ANDROID' \
- '-DWEBRTC_ANDROID_OPENSLES'
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../helpers \
- $(LOCAL_PATH)/../primitives \
- $(LOCAL_PATH)/../../../.. \
- $(LOCAL_PATH)/../../../include \
- $(LOCAL_PATH)/../../.. \
- $(LOCAL_PATH)/../../../../modules/video_coding/codecs/interface \
- $(LOCAL_PATH)/../../../../system_wrappers/interface \
- $(LOCAL_PATH)/../../../../modules/video_render/main/interface \
- $(LOCAL_PATH)/../../../../modules/interface \
- $(LOCAL_PATH)/../../../../modules/video_capture/main/interface \
- $(LOCAL_PATH)/../../../../common_video/vplib/main/interface \
- $(LOCAL_PATH)/../../../../voice_engine/main/interface
-
-LOCAL_SHARED_LIBRARIES := \
- libutils \
- libstlport \
- libwebrtc
-
-LOCAL_MODULE:= webrtc_video_test
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_EXECUTABLE)
diff --git a/trunk/src/video_engine/test/auto_test/source/tb_I420_codec.cc b/trunk/src/video_engine/test/auto_test/source/tb_I420_codec.cc
deleted file mode 100644
index df837c7..0000000
--- a/trunk/src/video_engine/test/auto_test/source/tb_I420_codec.cc
+++ /dev/null
@@ -1,324 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * tb_I420_codec.cc
- *
- */
-
-#include "tb_I420_codec.h"
-#include <string.h>
-#include <stdio.h>
-#include <assert.h>
-
-TbI420Encoder::TbI420Encoder() :
- _inited(false), _encodedImage(), _encodedCompleteCallback(NULL)
-{
- //
- memset(&_functionCalls, 0, sizeof(_functionCalls));
-}
-
-TbI420Encoder::~TbI420Encoder()
-{
- _inited = false;
- if (_encodedImage._buffer != NULL)
- {
- delete[] _encodedImage._buffer;
- _encodedImage._buffer = NULL;
- }
-}
-
-WebRtc_Word32 TbI420Encoder::VersionStatic(WebRtc_Word8* version,
- WebRtc_Word32 length)
-{
- const WebRtc_Word8* str = "I420 version 1.0.0\n";
- WebRtc_Word32 verLen = (WebRtc_Word32) strlen(str);
- if (verLen > length)
- {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- strncpy(version, str, length);
- return verLen;
-}
-
-WebRtc_Word32 TbI420Encoder::Version(WebRtc_Word8 *version,
- WebRtc_Word32 length) const
-{
- return VersionStatic(version, length);
-}
-
-WebRtc_Word32 TbI420Encoder::Release()
-{
- _functionCalls.Release++;
- // should allocate an encoded frame and then release it here, for that we
- // actaully need an init flag
- if (_encodedImage._buffer != NULL)
- {
- delete[] _encodedImage._buffer;
- _encodedImage._buffer = NULL;
- }
- _inited = false;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-WebRtc_Word32 TbI420Encoder::Reset()
-{
- _functionCalls.Reset++;
- if (!_inited)
- {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
- return WEBRTC_VIDEO_CODEC_OK;
-
-}
-
-WebRtc_Word32 TbI420Encoder::SetChannelParameters(WebRtc_UWord32 packetLoss,
- int rtt) {
- return 0;
-}
-
-WebRtc_Word32 TbI420Encoder::InitEncode(const webrtc::VideoCodec* inst,
- WebRtc_Word32 /*numberOfCores*/,
- WebRtc_UWord32 /*maxPayloadSize */)
-{
- _functionCalls.InitEncode++;
- if (inst == NULL)
- {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- if (inst->width < 1 || inst->height < 1)
- {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
-
- // allocating encoded memory
- if (_encodedImage._buffer != NULL)
- {
- delete[] _encodedImage._buffer;
- _encodedImage._buffer = NULL;
- _encodedImage._size = 0;
- }
- const WebRtc_UWord32 newSize = (3 * inst->width * inst->height) >> 1;
- WebRtc_UWord8* newBuffer = new WebRtc_UWord8[newSize];
- if (newBuffer == NULL)
- {
- return WEBRTC_VIDEO_CODEC_MEMORY;
- }
- _encodedImage._size = newSize;
- _encodedImage._buffer = newBuffer;
-
- // if no memeory allocation, no point to init
- _inited = true;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-WebRtc_Word32 TbI420Encoder::Encode(
- const webrtc::RawImage& inputImage,
- const webrtc::CodecSpecificInfo* /*codecSpecificInfo*/,
- const webrtc::VideoFrameType* /*frameType*/)
-{
- _functionCalls.Encode++;
- if (!_inited)
- {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
- if (_encodedCompleteCallback == NULL)
- {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
-
- _encodedImage._frameType = webrtc::kKeyFrame; // no coding
- _encodedImage._timeStamp = inputImage._timeStamp;
- _encodedImage._encodedHeight = inputImage._height;
- _encodedImage._encodedWidth = inputImage._width;
- if (inputImage._length > _encodedImage._size)
- {
-
- // allocating encoded memory
- if (_encodedImage._buffer != NULL)
- {
- delete[] _encodedImage._buffer;
- _encodedImage._buffer = NULL;
- _encodedImage._size = 0;
- }
- const WebRtc_UWord32 newSize = (3 * _encodedImage._encodedWidth
- * _encodedImage._encodedHeight) >> 1;
- WebRtc_UWord8* newBuffer = new WebRtc_UWord8[newSize];
- if (newBuffer == NULL)
- {
- return WEBRTC_VIDEO_CODEC_MEMORY;
- }
- _encodedImage._size = newSize;
- _encodedImage._buffer = newBuffer;
- }
- assert(_encodedImage._size >= inputImage._length);
- memcpy(_encodedImage._buffer, inputImage._buffer, inputImage._length);
- _encodedImage._length = inputImage._length;
- _encodedCompleteCallback->Encoded(_encodedImage);
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-WebRtc_Word32 TbI420Encoder::RegisterEncodeCompleteCallback(
- webrtc::EncodedImageCallback* callback)
-{
- _functionCalls.RegisterEncodeCompleteCallback++;
- _encodedCompleteCallback = callback;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-WebRtc_Word32 TbI420Encoder::SetPacketLoss(WebRtc_UWord32 packetLoss)
-{
- _functionCalls.SetPacketLoss++;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-WebRtc_Word32 TbI420Encoder::SetRates(WebRtc_UWord32 newBitRate,
- WebRtc_UWord32 frameRate)
-{
- _functionCalls.SetRates++;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-WebRtc_Word32 TbI420Encoder::SetPeriodicKeyFrames(bool enable)
-{
- _functionCalls.SetPeriodicKeyFrames++;
- return WEBRTC_VIDEO_CODEC_ERROR;
-}
-
-WebRtc_Word32 TbI420Encoder::CodecConfigParameters(WebRtc_UWord8* /*buffer*/,
- WebRtc_Word32 /*size*/)
-{
- _functionCalls.CodecConfigParameters++;
- return WEBRTC_VIDEO_CODEC_ERROR;
-}
-TbI420Encoder::FunctionCalls TbI420Encoder::GetFunctionCalls()
-{
- return _functionCalls;
-}
-
-TbI420Decoder::TbI420Decoder():
- _decodedImage(), _width(0), _height(0), _inited(false),
- _decodeCompleteCallback(NULL)
-{
- memset(&_functionCalls, 0, sizeof(_functionCalls));
-}
-
-TbI420Decoder::~TbI420Decoder()
-{
- Release();
-}
-
-WebRtc_Word32 TbI420Decoder::Reset()
-{
- _functionCalls.Reset++;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-WebRtc_Word32 TbI420Decoder::InitDecode(const webrtc::VideoCodec* inst,
- WebRtc_Word32 /*numberOfCores */)
-{
- _functionCalls.InitDecode++;
- if (inst == NULL)
- {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- else if (inst->width < 1 || inst->height < 1)
- {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- _width = inst->width;
- _height = inst->height;
- _inited = true;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-WebRtc_Word32 TbI420Decoder::Decode(
- const webrtc::EncodedImage& inputImage,
- bool /*missingFrames*/,
- const webrtc::RTPFragmentationHeader* /*fragmentation*/,
- const webrtc::CodecSpecificInfo* /*codecSpecificInfo*/,
- WebRtc_Word64 /*renderTimeMs*/)
-{
- _functionCalls.Decode++;
- if (inputImage._buffer == NULL)
- {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- if (_decodeCompleteCallback == NULL)
- {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
- if (inputImage._length <= 0)
- {
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
- }
- if (!_inited)
- {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
-
- // Allocate memory for decoded image
-
- if (_decodedImage._buffer != NULL)
- {
- delete[] _decodedImage._buffer;
- _decodedImage._buffer = NULL;
- _decodedImage._size = 0;
- }
- if (_decodedImage._buffer == NULL)
- {
- const WebRtc_UWord32 newSize = (3 * _width * _height) >> 1;
- WebRtc_UWord8* newBuffer = new WebRtc_UWord8[newSize];
- if (newBuffer == NULL)
- {
- return WEBRTC_VIDEO_CODEC_MEMORY;
- }
- _decodedImage._size = newSize;
- _decodedImage._buffer = newBuffer;
- }
-
- // Set decoded image parameters
- _decodedImage._height = _height;
- _decodedImage._width = _width;
- _decodedImage._timeStamp = inputImage._timeStamp;
- assert(_decodedImage._size >= inputImage._length);
- memcpy(_decodedImage._buffer, inputImage._buffer, inputImage._length);
- _decodedImage._length = inputImage._length;
- //_decodedImage._buffer = inputImage._buffer;
-
- _decodeCompleteCallback->Decoded(_decodedImage);
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-WebRtc_Word32 TbI420Decoder::RegisterDecodeCompleteCallback(
- webrtc::DecodedImageCallback* callback)
-{
- _functionCalls.RegisterDecodeCompleteCallback++;
- _decodeCompleteCallback = callback;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-WebRtc_Word32 TbI420Decoder::Release()
-{
- _functionCalls.Release++;
- if (_decodedImage._buffer != NULL)
- {
- delete[] _decodedImage._buffer;
- _decodedImage._buffer = NULL;
- }
- _inited = false;
- return WEBRTC_VIDEO_CODEC_OK;
-}
-
-TbI420Decoder::FunctionCalls TbI420Decoder::GetFunctionCalls()
-{
- return _functionCalls;
-}
-
diff --git a/trunk/src/video_engine/test/auto_test/source/tb_capture_device.cc b/trunk/src/video_engine/test/auto_test/source/tb_capture_device.cc
deleted file mode 100644
index 0d4e46a..0000000
--- a/trunk/src/video_engine/test/auto_test/source/tb_capture_device.cc
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "tb_capture_device.h"
-
-TbCaptureDevice::TbCaptureDevice(TbInterfaces& Engine) :
- captureId(-1),
- ViE(Engine),
- vcpm_(NULL)
-{
- const unsigned int KMaxDeviceNameLength = 128;
- const unsigned int KMaxUniqueIdLength = 256;
- WebRtc_UWord8 deviceName[KMaxDeviceNameLength];
- memset(deviceName, 0, KMaxDeviceNameLength);
- WebRtc_UWord8 uniqueId[KMaxUniqueIdLength];
- memset(uniqueId, 0, KMaxUniqueIdLength);
-
- bool captureDeviceSet = false;
-
- webrtc::VideoCaptureModule::DeviceInfo* devInfo =
- webrtc::VideoCaptureFactory::CreateDeviceInfo(0);
- for (size_t captureIdx = 0;
- captureIdx < devInfo->NumberOfDevices();
- captureIdx++)
- {
- EXPECT_EQ(0, devInfo->GetDeviceName(captureIdx, deviceName,
- KMaxDeviceNameLength, uniqueId,
- KMaxUniqueIdLength));
-
- vcpm_ = webrtc::VideoCaptureFactory::Create(
- captureIdx, uniqueId);
- if (vcpm_ == NULL) // Failed to open this device. Try next.
- {
- continue;
- }
- vcpm_->AddRef();
-
- int error = ViE.capture->AllocateCaptureDevice(*vcpm_, captureId);
- if (error == 0)
- {
- ViETest::Log("Using capture device: %s, captureId: %d", deviceName,
- captureId);
- captureDeviceSet = true;
- break;
- }
- }
- delete devInfo;
- EXPECT_TRUE(captureDeviceSet);
- if (!captureDeviceSet) {
- return;
- }
-
- ViETest::Log("Starting capture device %s with captureId %d\n", deviceName,
- captureId);
- EXPECT_EQ(0, ViE.capture->StartCapture(captureId));
-}
-
-TbCaptureDevice::~TbCaptureDevice(void)
-{
- ViETest::Log("Stopping capture device with id %d\n", captureId);
- EXPECT_EQ(0, ViE.capture->StopCapture(captureId));
- EXPECT_EQ(0, ViE.capture->ReleaseCaptureDevice(captureId));
- vcpm_->Release();
-}
-
-void TbCaptureDevice::ConnectTo(int videoChannel)
-{
- EXPECT_EQ(0, ViE.capture->ConnectCaptureDevice(captureId, videoChannel));
-}
-
-void TbCaptureDevice::Disconnect(int videoChannel)
-{
- EXPECT_EQ(0, ViE.capture->DisconnectCaptureDevice(videoChannel));
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/tb_external_transport.cc b/trunk/src/video_engine/test/auto_test/source/tb_external_transport.cc
deleted file mode 100644
index 8670c40..0000000
--- a/trunk/src/video_engine/test/auto_test/source/tb_external_transport.cc
+++ /dev/null
@@ -1,445 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "tb_external_transport.h"
-
-#include <stdio.h> // printf
-#include <stdlib.h> // rand
-#include <cassert>
-
-#if defined(WEBRTC_LINUX) || defined(__linux__)
-#include <string.h>
-#endif
-#if defined(WEBRTC_MAC)
-#include <cstring>
-#endif
-
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "tick_util.h"
-#include "vie_network.h"
-
-#if defined(_WIN32)
-#pragma warning(disable: 4355) // 'this' : used in base member initializer list
-#endif
-
-TbExternalTransport::TbExternalTransport(webrtc::ViENetwork& vieNetwork) :
- _vieNetwork(vieNetwork),
- _thread(*webrtc::ThreadWrapper::CreateThread(
- ViEExternalTransportRun, this, webrtc::kHighPriority,
- "AutotestTransport")),
- _event(*webrtc::EventWrapper::Create()),
- _crit(*webrtc::CriticalSectionWrapper::CreateCriticalSection()),
- _statCrit(*webrtc::CriticalSectionWrapper::CreateCriticalSection()),
- _lossRate(0),
- _networkDelayMs(0),
- _rtpCount(0),
- _rtcpCount(0),
- _dropCount(0),
- _rtpPackets(),
- _rtcpPackets(),
- _send_frame_callback(NULL),
- _receive_frame_callback(NULL),
- _temporalLayers(0),
- _seqNum(0),
- _sendPID(0),
- _receivedPID(0),
- _switchLayer(false),
- _currentRelayLayer(0),
- _lastTimeMs(webrtc::TickTime::MillisecondTimestamp()),
- _checkSSRC(false),
- _lastSSRC(0),
- _filterSSRC(false),
- _SSRC(0),
- _checkSequenceNumber(0),
- _firstSequenceNumber(0),
- _firstRTPTimestamp(0),
- _lastSendRTPTimestamp(0),
- _lastReceiveRTPTimestamp(0)
-{
- srand((int) webrtc::TickTime::MicrosecondTimestamp());
- unsigned int tId = 0;
- _thread.Start(tId);
-}
-
-TbExternalTransport::~TbExternalTransport()
-{
- _thread.SetNotAlive();
- _event.Set();
- if (_thread.Stop())
- {
- delete &_thread;
- delete &_event;
- }
- delete &_crit;
- delete &_statCrit;
-}
-
-int TbExternalTransport::SendPacket(int channel, const void *data, int len)
-{
- // Parse timestamp from RTP header according to RFC 3550, section 5.1.
- WebRtc_UWord8* ptr = (WebRtc_UWord8*)data;
- WebRtc_UWord32 rtp_timestamp = ptr[4] << 24;
- rtp_timestamp += ptr[5] << 16;
- rtp_timestamp += ptr[6] << 8;
- rtp_timestamp += ptr[7];
- _crit.Enter();
- if (_firstRTPTimestamp == 0) {
- _firstRTPTimestamp = rtp_timestamp;
- }
- _crit.Leave();
- if (_send_frame_callback != NULL &&
- _lastSendRTPTimestamp != rtp_timestamp) {
- _send_frame_callback->FrameSent(rtp_timestamp);
- }
- _lastSendRTPTimestamp = rtp_timestamp;
-
- if (_filterSSRC)
- {
- WebRtc_UWord8* ptr = (WebRtc_UWord8*)data;
- WebRtc_UWord32 ssrc = ptr[8] << 24;
- ssrc += ptr[9] << 16;
- ssrc += ptr[10] << 8;
- ssrc += ptr[11];
- if (ssrc != _SSRC)
- {
- return len; // return len to avoid error in trace file
- }
- }
- if (_temporalLayers) {
- // parse out vp8 temporal layers
- // 12 bytes RTP
- WebRtc_UWord8* ptr = (WebRtc_UWord8*)data;
-
- if (ptr[12] & 0x80 && // X-bit
- ptr[13] & 0x20) // T-bit
- {
- int offset = 1;
- if (ptr[13] & 0x80) // PID-bit
- {
- offset++;
- if (ptr[14] & 0x80) // 2 byte PID
- {
- offset++;
- }
- }
- if (ptr[13] & 0x40)
- {
- offset++;
- }
- unsigned char TID = (ptr[13 + offset] >> 5);
- unsigned int timeMs = NowMs();
-
- // Every 5 second switch layer
- if (_lastTimeMs + 5000 < timeMs)
- {
- _lastTimeMs = timeMs;
- _switchLayer = true;
- }
- // Switch at the non ref frame
- if (_switchLayer && (ptr[12] & 0x20))
- { // N-bit
- _currentRelayLayer++;
- if (_currentRelayLayer >= _temporalLayers)
- _currentRelayLayer = 0;
-
- _switchLayer = false;
- printf("\t Switching to layer:%d\n", _currentRelayLayer);
- }
- if (_currentRelayLayer < TID)
- {
- return len; // return len to avoid error in trace file
- }
- if (ptr[14] & 0x80) // 2 byte PID
- {
- if(_receivedPID != ptr[15])
- {
- _sendPID++;
- _receivedPID = ptr[15];
- }
- } else
- {
- if(_receivedPID != ptr[14])
- {
- _sendPID++;
- _receivedPID = ptr[14];
- }
- }
- }
- }
- _statCrit.Enter();
- _rtpCount++;
- _statCrit.Leave();
-
- // Packet loss. Never drop packets from the first RTP timestamp, i.e. the
- // first frame being transmitted.
- int dropThis = rand() % 100;
- if (dropThis < _lossRate && _firstRTPTimestamp != rtp_timestamp)
- {
- _statCrit.Enter();
- _dropCount++;
- _statCrit.Leave();
- return 0;
- }
-
- VideoPacket* newPacket = new VideoPacket();
- memcpy(newPacket->packetBuffer, data, len);
-
- if (_temporalLayers)
- {
- // rewrite seqNum
- newPacket->packetBuffer[2] = _seqNum >> 8;
- newPacket->packetBuffer[3] = _seqNum;
- _seqNum++;
-
- // rewrite PID
- if (newPacket->packetBuffer[14] & 0x80) // 2 byte PID
- {
- newPacket->packetBuffer[14] = (_sendPID >> 8) | 0x80;
- newPacket->packetBuffer[15] = _sendPID;
- } else
- {
- newPacket->packetBuffer[14] = (_sendPID & 0x7f);
- }
- }
- newPacket->length = len;
- newPacket->channel = channel;
-
- _crit.Enter();
- newPacket->receiveTime = NowMs() + _networkDelayMs;
- _rtpPackets.push_back(newPacket);
- _event.Set();
- _crit.Leave();
- return len;
-}
-
-void TbExternalTransport::RegisterSendFrameCallback(
- SendFrameCallback* callback) {
- _send_frame_callback = callback;
-}
-
-void TbExternalTransport::RegisterReceiveFrameCallback(
- ReceiveFrameCallback* callback) {
- _receive_frame_callback = callback;
-}
-
-// Set to 0 to disable.
-void TbExternalTransport::SetTemporalToggle(unsigned char layers)
-{
- _temporalLayers = layers;
-}
-
-int TbExternalTransport::SendRTCPPacket(int channel, const void *data, int len)
-{
- _statCrit.Enter();
- _rtcpCount++;
- _statCrit.Leave();
-
- VideoPacket* newPacket = new VideoPacket();
- memcpy(newPacket->packetBuffer, data, len);
- newPacket->length = len;
- newPacket->channel = channel;
-
- _crit.Enter();
- newPacket->receiveTime = NowMs() + _networkDelayMs;
- _rtcpPackets.push_back(newPacket);
- _event.Set();
- _crit.Leave();
- return len;
-}
-
-WebRtc_Word32 TbExternalTransport::SetPacketLoss(WebRtc_Word32 lossRate)
-{
- webrtc::CriticalSectionScoped cs(_statCrit);
- _lossRate = lossRate;
- return 0;
-}
-
-void TbExternalTransport::SetNetworkDelay(WebRtc_Word64 delayMs)
-{
- webrtc::CriticalSectionScoped cs(_crit);
- _networkDelayMs = delayMs;
-}
-
-void TbExternalTransport::SetSSRCFilter(WebRtc_UWord32 ssrc)
-{
- webrtc::CriticalSectionScoped cs(_crit);
- _filterSSRC = true;
- _SSRC = ssrc;
-}
-
-void TbExternalTransport::ClearStats()
-{
- webrtc::CriticalSectionScoped cs(_statCrit);
- _rtpCount = 0;
- _dropCount = 0;
- _rtcpCount = 0;
-}
-
-void TbExternalTransport::GetStats(WebRtc_Word32& numRtpPackets,
- WebRtc_Word32& numDroppedPackets,
- WebRtc_Word32& numRtcpPackets)
-{
- webrtc::CriticalSectionScoped cs(_statCrit);
- numRtpPackets = _rtpCount;
- numDroppedPackets = _dropCount;
- numRtcpPackets = _rtcpCount;
-}
-
-void TbExternalTransport::EnableSSRCCheck()
-{
- webrtc::CriticalSectionScoped cs(_statCrit);
- _checkSSRC = true;
-}
-
-unsigned int TbExternalTransport::ReceivedSSRC()
-{
- webrtc::CriticalSectionScoped cs(_statCrit);
- return _lastSSRC;
-}
-
-void TbExternalTransport::EnableSequenceNumberCheck()
-{
- webrtc::CriticalSectionScoped cs(_statCrit);
- _checkSequenceNumber = true;
-}
-
-unsigned short TbExternalTransport::GetFirstSequenceNumber()
-{
- webrtc::CriticalSectionScoped cs(_statCrit);
- return _firstSequenceNumber;
-}
-
-bool TbExternalTransport::ViEExternalTransportRun(void* object)
-{
- return static_cast<TbExternalTransport*>
- (object)->ViEExternalTransportProcess();
-}
-bool TbExternalTransport::ViEExternalTransportProcess()
-{
- unsigned int waitTime = KMaxWaitTimeMs;
-
- VideoPacket* packet = NULL;
-
- while (!_rtpPackets.empty())
- {
- // Take first packet in queue
- _crit.Enter();
- packet = _rtpPackets.front();
- WebRtc_Word64 timeToReceive = 0;
- if (packet)
- {
- timeToReceive = packet->receiveTime - NowMs();
- }
- else
- {
- // There should never be any empty packets in the list.
- assert(false);
- }
- if (timeToReceive > 0)
- {
- // No packets to receive yet
- if (timeToReceive < waitTime && timeToReceive > 0)
- {
- waitTime = (unsigned int) timeToReceive;
- }
- _crit.Leave();
- break;
- }
- _rtpPackets.pop_front();
- _crit.Leave();
-
- // Send to ViE
- if (packet)
- {
- {
- webrtc::CriticalSectionScoped cs(_statCrit);
- if (_checkSSRC)
- {
- _lastSSRC = ((packet->packetBuffer[8]) << 24);
- _lastSSRC += (packet->packetBuffer[9] << 16);
- _lastSSRC += (packet->packetBuffer[10] << 8);
- _lastSSRC += packet->packetBuffer[11];
- _checkSSRC = false;
- }
- if (_checkSequenceNumber)
- {
- _firstSequenceNumber
- = (unsigned char) packet->packetBuffer[2] << 8;
- _firstSequenceNumber
- += (unsigned char) packet->packetBuffer[3];
- _checkSequenceNumber = false;
- }
- }
- // Signal received packet of frame
- WebRtc_UWord8* ptr = (WebRtc_UWord8*)packet->packetBuffer;
- WebRtc_UWord32 rtp_timestamp = ptr[4] << 24;
- rtp_timestamp += ptr[5] << 16;
- rtp_timestamp += ptr[6] << 8;
- rtp_timestamp += ptr[7];
- if (_receive_frame_callback != NULL &&
- _lastReceiveRTPTimestamp != rtp_timestamp) {
- _receive_frame_callback->FrameReceived(rtp_timestamp);
- }
- _lastReceiveRTPTimestamp = rtp_timestamp;
-
- _vieNetwork.ReceivedRTPPacket(packet->channel,
- packet->packetBuffer, packet->length);
- delete packet;
- packet = NULL;
- }
- }
- while (!_rtcpPackets.empty())
- {
- // Take first packet in queue
- _crit.Enter();
- packet = _rtcpPackets.front();
- WebRtc_Word64 timeToReceive = 0;
- if (packet)
- {
- timeToReceive = packet->receiveTime - NowMs();
- }
- else
- {
- // There should never be any empty packets in the list.
- assert(false);
- }
- if (timeToReceive > 0)
- {
- // No packets to receive yet
- if (timeToReceive < waitTime && timeToReceive > 0)
- {
- waitTime = (unsigned int) timeToReceive;
- }
- _crit.Leave();
- break;
- }
- _rtcpPackets.pop_front();
- _crit.Leave();
-
- // Send to ViE
- if (packet)
- {
- _vieNetwork.ReceivedRTCPPacket(
- packet->channel,
- packet->packetBuffer, packet->length);
- delete packet;
- packet = NULL;
- }
- }
- _event.Wait(waitTime + 1); // Add 1 ms to not call to early...
- return true;
-}
-
-WebRtc_Word64 TbExternalTransport::NowMs()
-{
- return webrtc::TickTime::MillisecondTimestamp();
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/tb_interfaces.cc b/trunk/src/video_engine/test/auto_test/source/tb_interfaces.cc
deleted file mode 100644
index b6b13a2..0000000
--- a/trunk/src/video_engine/test/auto_test/source/tb_interfaces.cc
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "tb_interfaces.h"
-
-#include "gtest/gtest.h"
-
-TbInterfaces::TbInterfaces(const char* test_name) {
- std::string trace_file_path =
- (ViETest::GetResultOutputPath() + test_name) + "_trace.txt";
-
- ViETest::Log("Creating ViE Interfaces for test %s\n", test_name);
-
- video_engine = webrtc::VideoEngine::Create();
- EXPECT_TRUE(video_engine != NULL);
-
- EXPECT_EQ(0, video_engine->SetTraceFile(trace_file_path.c_str()));
- EXPECT_EQ(0, video_engine->SetTraceFilter(webrtc::kTraceAll));
-
- base = webrtc::ViEBase::GetInterface(video_engine);
- EXPECT_TRUE(base != NULL);
-
- EXPECT_EQ(0, base->Init());
-
- capture = webrtc::ViECapture::GetInterface(video_engine);
- EXPECT_TRUE(capture != NULL);
-
- rtp_rtcp = webrtc::ViERTP_RTCP::GetInterface(video_engine);
- EXPECT_TRUE(rtp_rtcp != NULL);
-
- render = webrtc::ViERender::GetInterface(video_engine);
- EXPECT_TRUE(render != NULL);
-
- codec = webrtc::ViECodec::GetInterface(video_engine);
- EXPECT_TRUE(codec != NULL);
-
- network = webrtc::ViENetwork::GetInterface(video_engine);
- EXPECT_TRUE(network != NULL);
-
- image_process = webrtc::ViEImageProcess::GetInterface(video_engine);
- EXPECT_TRUE(image_process != NULL);
-
- encryption = webrtc::ViEEncryption::GetInterface(video_engine);
- EXPECT_TRUE(encryption != NULL);
-}
-
-TbInterfaces::~TbInterfaces(void)
-{
- EXPECT_EQ(0, encryption->Release());
- EXPECT_EQ(0, image_process->Release());
- EXPECT_EQ(0, codec->Release());
- EXPECT_EQ(0, capture->Release());
- EXPECT_EQ(0, render->Release());
- EXPECT_EQ(0, rtp_rtcp->Release());
- EXPECT_EQ(0, network->Release());
- EXPECT_EQ(0, base->Release());
- EXPECT_TRUE(webrtc::VideoEngine::Delete(video_engine)) <<
- "Since we have released all interfaces at this point, deletion "
- "should be successful.";
-
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/tb_video_channel.cc b/trunk/src/video_engine/test/auto_test/source/tb_video_channel.cc
deleted file mode 100644
index 1e62a6b..0000000
--- a/trunk/src/video_engine/test/auto_test/source/tb_video_channel.cc
+++ /dev/null
@@ -1,90 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "tb_video_channel.h"
-
-TbVideoChannel::TbVideoChannel(TbInterfaces& Engine,
- webrtc::VideoCodecType sendCodec, int width,
- int height, int frameRate, int startBitrate) :
- videoChannel(-1), ViE(Engine)
-{
- EXPECT_EQ(0, ViE.base->CreateChannel(videoChannel));
-
- webrtc::VideoCodec videoCodec;
- memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
- bool sendCodecSet = false;
- for (int idx = 0; idx < ViE.codec->NumberOfCodecs(); idx++)
- {
- EXPECT_EQ(0, ViE.codec->GetCodec(idx, videoCodec));
- videoCodec.width = width;
- videoCodec.height = height;
- videoCodec.maxFramerate = frameRate;
-
- if (videoCodec.codecType == sendCodec && sendCodecSet == false)
- {
- if(videoCodec.codecType != webrtc::kVideoCodecI420 )
- {
- videoCodec.startBitrate = startBitrate;
- videoCodec.maxBitrate = startBitrate * 3;
- }
- EXPECT_EQ(0, ViE.codec->SetSendCodec(videoChannel, videoCodec));
- sendCodecSet = true;
- }
- if (videoCodec.codecType == webrtc::kVideoCodecVP8)
- {
- videoCodec.width = 352;
- videoCodec.height = 288;
- }
- EXPECT_EQ(0, ViE.codec->SetReceiveCodec(videoChannel, videoCodec));
- }
- EXPECT_TRUE(sendCodecSet);
-}
-
-TbVideoChannel::~TbVideoChannel(void)
-{
- EXPECT_EQ(0, ViE.base->DeleteChannel(videoChannel));
-}
-
-void TbVideoChannel::StartSend(const unsigned short rtpPort /*= 11000*/,
- const char* ipAddress /*= "127.0.0.1"*/)
-{
- EXPECT_EQ(0, ViE.network->SetSendDestination(videoChannel, ipAddress,
- rtpPort));
-
- EXPECT_EQ(0, ViE.base->StartSend(videoChannel));
-}
-
-void TbVideoChannel::SetFrameSettings(int width, int height, int frameRate)
-{
- webrtc::VideoCodec videoCodec;
- EXPECT_EQ(0, ViE.codec->GetSendCodec(videoChannel, videoCodec));
- videoCodec.width = width;
- videoCodec.height = height;
- videoCodec.maxFramerate = frameRate;
-
- EXPECT_EQ(0, ViE.codec->SetSendCodec(videoChannel, videoCodec));
- EXPECT_EQ(0, ViE.codec->SetReceiveCodec(videoChannel, videoCodec));
-}
-
-void TbVideoChannel::StopSend()
-{
- EXPECT_EQ(0, ViE.base->StopSend(videoChannel));
-}
-
-void TbVideoChannel::StartReceive(const unsigned short rtpPort /*= 11000*/)
-{
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(videoChannel, rtpPort));
- EXPECT_EQ(0, ViE.base->StartReceive(videoChannel));
-}
-
-void TbVideoChannel::StopReceive()
-{
- EXPECT_EQ(0, ViE.base->StopReceive(videoChannel));
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest.cc
deleted file mode 100644
index 8281ea2..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest.cc
+++ /dev/null
@@ -1,159 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest.cc
-//
-
-#include "vie_autotest.h"
-
-#include <stdio.h>
-
-#include "engine_configurations.h"
-#include "general_primitives.h"
-#include "tb_interfaces.h"
-#include "tb_video_channel.h"
-#include "tb_capture_device.h"
-#include "testsupport/fileutils.h"
-#include "video_render.h"
-#include "vie_autotest_defines.h"
-
-// ViETest implementation
-FILE* ViETest::log_file_ = NULL;
-char* ViETest::log_str_ = NULL;
-
-std::string ViETest::GetResultOutputPath() {
-#ifdef WEBRTC_ANDROID
- return "/sdcard/";
-#else
- return webrtc::test::OutputPath();
-#endif
-}
-
-// ViEAutoTest implementation
-ViEAutoTest::ViEAutoTest(void* window1, void* window2) :
- _window1(window1),
- _window2(window2),
- _renderType(webrtc::kRenderDefault),
- _vrm1(webrtc::VideoRender::CreateVideoRender(
- 4561, window1, false, _renderType)),
- _vrm2(webrtc::VideoRender::CreateVideoRender(
- 4562, window2, false, _renderType))
-{
- assert(_vrm1);
- assert(_vrm2);
-}
-
-ViEAutoTest::~ViEAutoTest()
-{
- webrtc::VideoRender::DestroyVideoRender(_vrm1);
- _vrm1 = NULL;
- webrtc::VideoRender::DestroyVideoRender(_vrm2);
- _vrm2 = NULL;
-}
-
-void ViEAutoTest::ViEStandardTest()
-{
- ViEBaseStandardTest();
- ViECaptureStandardTest();
- ViECodecStandardTest();
- ViEEncryptionStandardTest();
- ViEFileStandardTest();
- ViEImageProcessStandardTest();
- ViENetworkStandardTest();
- ViERenderStandardTest();
- ViERtpRtcpStandardTest();
-}
-
-void ViEAutoTest::ViEExtendedTest()
-{
- ViEBaseExtendedTest();
- ViECaptureExtendedTest();
- ViECodecExtendedTest();
- ViEEncryptionExtendedTest();
- ViEFileExtendedTest();
- ViEImageProcessExtendedTest();
- ViENetworkExtendedTest();
- ViERenderExtendedTest();
- ViERtpRtcpExtendedTest();
-}
-
-void ViEAutoTest::ViEAPITest()
-{
- ViEBaseAPITest();
- ViECaptureAPITest();
- ViECodecAPITest();
- ViEEncryptionAPITest();
- ViEFileAPITest();
- ViEImageProcessAPITest();
- ViENetworkAPITest();
- ViERenderAPITest();
- ViERtpRtcpAPITest();
-}
-
-void ViEAutoTest::PrintVideoCodec(const webrtc::VideoCodec videoCodec)
-{
- ViETest::Log("Video Codec Information:");
-
- switch (videoCodec.codecType)
- {
- case webrtc::kVideoCodecVP8:
- ViETest::Log("\tcodecType: VP8");
- break;
- // TODO(sh): keep or remove MPEG4?
- // case webrtc::kVideoCodecMPEG4:
- // ViETest::Log("\tcodecType: MPEG4");
- // break;
- case webrtc::kVideoCodecI420:
- ViETest::Log("\tcodecType: I420");
- break;
- case webrtc::kVideoCodecRED:
- ViETest::Log("\tcodecType: RED");
- break;
- case webrtc::kVideoCodecULPFEC:
- ViETest::Log("\tcodecType: ULPFEC");
- break;
- case webrtc::kVideoCodecUnknown:
- ViETest::Log("\tcodecType: ????");
- break;
- }
-
- ViETest::Log("\theight: %u", videoCodec.height);
- ViETest::Log("\tmaxBitrate: %u", videoCodec.maxBitrate);
- ViETest::Log("\tmaxFramerate: %u", videoCodec.maxFramerate);
- ViETest::Log("\tminBitrate: %u", videoCodec.minBitrate);
- ViETest::Log("\tplName: %s", videoCodec.plName);
- ViETest::Log("\tplType: %u", videoCodec.plType);
- ViETest::Log("\tstartBitrate: %u", videoCodec.startBitrate);
- ViETest::Log("\twidth: %u", videoCodec.width);
- ViETest::Log("");
-}
-
-void ViEAutoTest::PrintAudioCodec(const webrtc::CodecInst audioCodec)
-{
- ViETest::Log("Audio Codec Information:");
- ViETest::Log("\tchannels: %u", audioCodec.channels);
- ViETest::Log("\t: %u", audioCodec.pacsize);
- ViETest::Log("\t: %u", audioCodec.plfreq);
- ViETest::Log("\t: %s", audioCodec.plname);
- ViETest::Log("\t: %u", audioCodec.pltype);
- ViETest::Log("\t: %u", audioCodec.rate);
- ViETest::Log("");
-}
-
-void ViEAutoTest::RenderCaptureDeviceAndOutputStream(
- TbInterfaces* video_engine,
- TbVideoChannel* video_channel,
- TbCaptureDevice* capture_device) {
- RenderInWindow(
- video_engine->render, capture_device->captureId, _window1, 0);
- RenderInWindow(
- video_engine->render, video_channel->videoChannel, _window1, 1);
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_android.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_android.cc
deleted file mode 100644
index 81af69b..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_android.cc
+++ /dev/null
@@ -1,209 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "../interface/vie_autotest_android.h"
-
-#include <android/log.h>
-#include <stdio.h>
-
-#include "vie_autotest.h"
-#include "vie_autotest_defines.h"
-
-int ViEAutoTestAndroid::RunAutotest(int testSelection, int subTestSelection,
- void* window1, void* window2,
- void* javaVM, void* env, void* context) {
- ViEAutoTest vieAutoTest(window1, window2);
- ViETest::Log("RunAutoTest(%d, %d)", testSelection, subTestSelection);
- webrtc::VideoEngine::SetAndroidObjects(javaVM, context);
-#ifndef WEBRTC_ANDROID_OPENSLES
- // voice engine calls into ADM directly
- webrtc::VoiceEngine::SetAndroidAudioDeviceObjects(javaVM, env, context);
-#endif
- int testErrors = 0;
-
- if (subTestSelection == 0) {
- // Run all selected test
- switch (testSelection) {
- case 0:
- testErrors += vieAutoTest.ViEStandardTest();
- if (testErrors == 0) {
- // No errors found in delivery test, create delivery
- ViETest::Log("Standard/delivery passed. ");
- }
- else {
- // Didn't pass
- ViETest::Log("\nStandard/delivery test failed.");
- }
- break;
- case 1:
- testErrors += vieAutoTest.ViEAPITest();
- break;
- case 2:
- testErrors += vieAutoTest.ViEExtendedTest();
- break;
- case 3:
- testErrors += vieAutoTest.ViELoopbackCall();
- break;
- default:
- break;
- }
- }
-
- switch (testSelection) {
- case 0: // Specific standard test
- switch (subTestSelection) {
- case 1: // base
- testErrors += vieAutoTest.ViEBaseStandardTest();
- break;
-
- case 2: // capture
- testErrors += vieAutoTest.ViECaptureStandardTest();
- break;
-
- case 3: // codec
- testErrors += vieAutoTest.ViECodecStandardTest();
- break;
-
- case 5: //encryption
- testErrors += vieAutoTest.ViEEncryptionStandardTest();
- break;
-
- case 6: // file
- testErrors += vieAutoTest.ViEFileStandardTest();
- break;
-
- case 7: // image process
- testErrors += vieAutoTest.ViEImageProcessStandardTest();
- break;
-
- case 8: // network
- testErrors += vieAutoTest.ViENetworkStandardTest();
- break;
-
- case 9: // Render
- testErrors += vieAutoTest.ViERenderStandardTest();
- break;
-
- case 10: // RTP/RTCP
- testErrors += vieAutoTest.ViERtpRtcpStandardTest();
- break;
-
- default:
- break;
- }
- break;
-
- case 1:// specific API
- switch (subTestSelection) {
- case 1: // base
- testErrors += vieAutoTest.ViEBaseAPITest();
- break;
-
- case 2: // capture
- testErrors += vieAutoTest.ViECaptureAPITest();
- break;
-
- case 3: // codec
- testErrors += vieAutoTest.ViECodecAPITest();
- break;
-
- case 5: //encryption
- testErrors += vieAutoTest.ViEEncryptionAPITest();
- break;
-
- case 6: // file
- testErrors += vieAutoTest.ViEFileAPITest();
- break;
-
- case 7: // image process
- testErrors += vieAutoTest.ViEImageProcessAPITest();
- break;
-
- case 8: // network
- testErrors += vieAutoTest.ViENetworkAPITest();
- break;
-
- case 9: // Render
- testErrors += vieAutoTest.ViERenderAPITest();
- break;
-
- case 10: // RTP/RTCP
- testErrors += vieAutoTest.ViERtpRtcpAPITest();
- break;
- case 11:
- break;
-
- default:
- break;
- }
- break;
-
- case 2:// specific extended
- switch (subTestSelection) {
- case 1: // base
- testErrors += vieAutoTest.ViEBaseExtendedTest();
- break;
-
- case 2: // capture
- testErrors += vieAutoTest.ViECaptureExtendedTest();
- break;
-
- case 3: // codec
- testErrors += vieAutoTest.ViECodecExtendedTest();
- break;
-
- case 5: //encryption
- testErrors += vieAutoTest.ViEEncryptionExtendedTest();
- break;
-
- case 6: // file
- testErrors += vieAutoTest.ViEFileExtendedTest();
- break;
-
- case 7: // image process
- testErrors += vieAutoTest.ViEImageProcessExtendedTest();
- break;
-
- case 8: // network
- testErrors += vieAutoTest.ViENetworkExtendedTest();
- break;
-
- case 9: // Render
- testErrors += vieAutoTest.ViERenderExtendedTest();
- break;
-
- case 10: // RTP/RTCP
- testErrors += vieAutoTest.ViERtpRtcpExtendedTest();
- break;
-
- case 11:
- break;
-
- default:
- break;
- }
- break;
-
- case 3:
- testErrors += vieAutoTest.ViELoopbackCall();
- break;
-
- default:
- break;
- }
-
- if (testErrors) {
- ViETest::Log("Test done with %d errors!\n", testErrors);
- }
- else {
- ViETest::Log("Test passed!\n");
- }
- return testErrors;
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_base.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_base.cc
deleted file mode 100644
index d2a29b2..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_base.cc
+++ /dev/null
@@ -1,208 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vie_autotest.h"
-
-#include "base_primitives.h"
-#include "general_primitives.h"
-#include "tb_interfaces.h"
-#include "vie_autotest_defines.h"
-#include "video_capture_factory.h"
-
-class BaseObserver : public webrtc::ViEBaseObserver {
- public:
- BaseObserver()
- : cpu_load_(0) {}
-
- virtual void PerformanceAlarm(const unsigned int cpu_load) {
- cpu_load_ = cpu_load;
- }
- unsigned int cpu_load_;
-};
-
-void ViEAutoTest::ViEBaseStandardTest() {
- // ***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- // ***************************************************************
-
- TbInterfaces interfaces("ViEBaseStandardTest");
-
- // ***************************************************************
- // Engine ready. Set up the test case:
- // ***************************************************************
- int video_channel = -1;
- EXPECT_EQ(0, interfaces.base->CreateChannel(video_channel));
-
- webrtc::VideoCaptureModule* video_capture_module(NULL);
- const unsigned int kMaxDeviceNameLength = 128;
- WebRtc_UWord8 device_name[kMaxDeviceNameLength];
- memset(device_name, 0, kMaxDeviceNameLength);
- int capture_id;
-
- webrtc::ViEBase *base_interface = interfaces.base;
- webrtc::ViERender *render_interface = interfaces.render;
- webrtc::ViECapture *capture_interface = interfaces.capture;
-
- FindCaptureDeviceOnSystem(capture_interface,
- device_name,
- kMaxDeviceNameLength,
- &capture_id,
- &video_capture_module);
-
- EXPECT_EQ(0, capture_interface->ConnectCaptureDevice(capture_id,
- video_channel));
- EXPECT_EQ(0, capture_interface->StartCapture(capture_id));
-
- ConfigureRtpRtcp(interfaces.rtp_rtcp, video_channel);
-
- EXPECT_EQ(0, render_interface->RegisterVideoRenderModule(*_vrm1));
- EXPECT_EQ(0, render_interface->RegisterVideoRenderModule(*_vrm2));
-
- RenderInWindow(render_interface, capture_id, _window1, 0);
- RenderInWindow(render_interface, video_channel, _window2, 1);
-
- // ***************************************************************
- // Run the actual test:
- // ***************************************************************
- ViETest::Log("You should shortly see a local preview from camera %s"
- " in window 1 and the remote video in window 2.", device_name);
- ::TestI420CallSetup(interfaces.codec, interfaces.video_engine,
- base_interface, interfaces.network, video_channel,
- device_name);
-
- // ***************************************************************
- // Testing finished. Tear down Video Engine
- // ***************************************************************
- EXPECT_EQ(0, capture_interface->StopCapture(capture_id));
- EXPECT_EQ(0, base_interface->StopReceive(video_channel));
-
- StopAndRemoveRenderers(base_interface, render_interface, video_channel,
- capture_id);
-
- EXPECT_EQ(0, render_interface->DeRegisterVideoRenderModule(*_vrm1));
- EXPECT_EQ(0, render_interface->DeRegisterVideoRenderModule(*_vrm2));
-
- EXPECT_EQ(0, capture_interface->ReleaseCaptureDevice(capture_id));
-
- video_capture_module->Release();
- video_capture_module = NULL;
-
- EXPECT_EQ(0, base_interface->DeleteChannel(video_channel));
-}
-
-void ViEAutoTest::ViEBaseExtendedTest() {
- // Start with standard test
- ViEBaseAPITest();
- ViEBaseStandardTest();
-
- // ***************************************************************
- // Test BaseObserver
- // ***************************************************************
- // TODO(mflodman) Add test for base observer. Cpu load must be over 75%.
-// BaseObserver base_observer;
-// EXPECT_EQ(ptrViEBase->RegisterObserver(base_observer), 0);
-//
-// AutoTestSleep(KAutoTestSleepTimeMs);
-//
-// EXPECT_EQ(ptrViEBase->DeregisterObserver(), 0);
-// EXPECT_GT(base_observer.cpu_load, 0);
-}
-
-void ViEAutoTest::ViEBaseAPITest() {
- // ***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- // ***************************************************************
- // Get the ViEBase API
- webrtc::ViEBase* ptrViEBase = webrtc::ViEBase::GetInterface(NULL);
- EXPECT_EQ(NULL, ptrViEBase) << "Should return null for a bad ViE pointer";
-
- webrtc::VideoEngine* ptrViE = webrtc::VideoEngine::Create();
- EXPECT_TRUE(NULL != ptrViE);
-
- std::string trace_file_path =
- ViETest::GetResultOutputPath() + "ViEBaseAPI_trace.txt";
- EXPECT_EQ(0, ptrViE->SetTraceFile(trace_file_path.c_str()));
-
- ptrViEBase = webrtc::ViEBase::GetInterface(ptrViE);
- EXPECT_TRUE(NULL != ptrViEBase);
-
- // ***************************************************************
- // Engine ready. Begin testing class
- // ***************************************************************
- char version[1024] = "";
- EXPECT_EQ(0, ptrViEBase->GetVersion(version));
- EXPECT_EQ(0, ptrViEBase->LastError());
-
- // Create without init
- int videoChannel = -1;
- EXPECT_NE(0, ptrViEBase->CreateChannel(videoChannel)) <<
- "Should fail since Init has not been called yet";
- EXPECT_EQ(0, ptrViEBase->Init());
- EXPECT_EQ(0, ptrViEBase->CreateChannel(videoChannel));
-
- int videoChannel2 = -1;
- EXPECT_EQ(0, ptrViEBase->CreateChannel(videoChannel2));
- EXPECT_NE(videoChannel, videoChannel2) <<
- "Should allocate new number for independent channel";
-
- EXPECT_EQ(0, ptrViEBase->DeleteChannel(videoChannel2));
-
- EXPECT_EQ(-1, ptrViEBase->CreateChannel(videoChannel2, videoChannel + 1)) <<
- "Should fail since neither channel exists (the second must)";
-
- EXPECT_EQ(0, ptrViEBase->CreateChannel(videoChannel2, videoChannel));
-
- // Test Voice Engine integration with Video Engine.
- webrtc::VoiceEngine* ptrVoE = NULL;
- webrtc::VoEBase* ptrVoEBase = NULL;
- int audioChannel = -1;
-
- ptrVoE = webrtc::VoiceEngine::Create();
- EXPECT_TRUE(NULL != ptrVoE);
-
- ptrVoEBase = webrtc::VoEBase::GetInterface(ptrVoE);
- EXPECT_TRUE(NULL != ptrVoEBase);
- EXPECT_EQ(0, ptrVoEBase->Init());
-
- audioChannel = ptrVoEBase->CreateChannel();
- EXPECT_NE(-1, audioChannel);
-
- // Connect before setting VoE.
- EXPECT_NE(0, ptrViEBase->ConnectAudioChannel(videoChannel, audioChannel)) <<
- "Should fail since Voice Engine is not set yet.";
-
- // Then do it right.
- EXPECT_EQ(0, ptrViEBase->SetVoiceEngine(ptrVoE));
- EXPECT_EQ(0, ptrViEBase->ConnectAudioChannel(videoChannel, audioChannel));
-
- // ***************************************************************
- // Testing finished. Tear down Video Engine
- // ***************************************************************
- EXPECT_NE(0, ptrViEBase->DisconnectAudioChannel(videoChannel + 5)) <<
- "Should fail: disconnecting bogus channel";
-
- EXPECT_EQ(0, ptrViEBase->DisconnectAudioChannel(videoChannel));
-
- // Clean up voice engine
- EXPECT_EQ(0, ptrViEBase->SetVoiceEngine(NULL));
- EXPECT_EQ(0, ptrVoEBase->Release());
- EXPECT_TRUE(webrtc::VoiceEngine::Delete(ptrVoE));
-
- webrtc::ViEBase* ptrViEBase2 = webrtc::ViEBase::GetInterface(ptrViE);
- EXPECT_TRUE(NULL != ptrViEBase2);
-
- EXPECT_EQ(1, ptrViEBase->Release()) << "There should be one interface left.";
-
- EXPECT_FALSE(webrtc::VideoEngine::Delete(ptrViE)) <<
- "Should fail since there are interfaces left.";
-
- EXPECT_EQ(0, ptrViEBase->Release());
- EXPECT_TRUE(webrtc::VideoEngine::Delete(ptrViE));
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_capture.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_capture.cc
deleted file mode 100644
index cbb8b03..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_capture.cc
+++ /dev/null
@@ -1,559 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vie_autotest.h"
-
-#include "common_types.h"
-#include "engine_configurations.h"
-#include "gflags/gflags.h"
-#include "tb_interfaces.h"
-#include "tb_video_channel.h"
-#include "tick_util.h"
-#include "vie_autotest_defines.h"
-#include "video_capture_factory.h"
-#include "vie_base.h"
-#include "vie_capture.h"
-#include "vie_codec.h"
-#include "vie_network.h"
-#include "vie_render.h"
-#include "vie_rtp_rtcp.h"
-#include "voe_base.h"
-
-DEFINE_bool(capture_test_ensure_resolution_alignment_in_capture_device, true,
- "If true, we will give resolutions slightly below a reasonable "
- "value to test the camera's ability to choose a good resolution. "
- "If false, we will provide reasonable resolutions instead.");
-
-class CaptureObserver: public webrtc::ViECaptureObserver
-{
-public:
- CaptureObserver() :
- _brightness(webrtc::Normal),
- _alarm(webrtc::AlarmCleared),
- _frameRate(0) {}
-
- virtual void BrightnessAlarm(const int captureId,
- const webrtc::Brightness brightness)
- {
- _brightness = brightness;
- switch (brightness)
- {
- case webrtc::Normal:
- ViETest::Log(" BrightnessAlarm Normal");
- break;
- case webrtc::Bright:
- ViETest::Log(" BrightnessAlarm Bright");
- break;
- case webrtc::Dark:
- ViETest::Log(" BrightnessAlarm Dark");
- break;
- }
- }
-
- virtual void CapturedFrameRate(const int captureId,
- const unsigned char frameRate)
- {
- ViETest::Log(" CapturedFrameRate %u", frameRate);
- _frameRate = frameRate;
- }
-
- virtual void NoPictureAlarm(const int captureId,
- const webrtc::CaptureAlarm alarm)
- {
- _alarm = alarm;
- if (alarm == webrtc::AlarmRaised)
- {
- ViETest::Log("NoPictureAlarm CARaised.");
- }
- else
- {
- ViETest::Log("NoPictureAlarm CACleared.");
- }
- }
-
- webrtc::Brightness _brightness;
- webrtc::CaptureAlarm _alarm;
- unsigned char _frameRate;
-};
-
-class CaptureEffectFilter: public webrtc::ViEEffectFilter {
- public:
- CaptureEffectFilter(unsigned int expected_width, unsigned int expected_height)
- : number_of_captured_frames_(0),
- expected_width_(expected_width),
- expected_height_(expected_height) {
- }
-
- // Implements ViEEffectFilter
- virtual int Transform(int size, unsigned char* frameBuffer,
- unsigned int timeStamp90KHz, unsigned int width,
- unsigned int height) {
- EXPECT_TRUE(frameBuffer != NULL);
- EXPECT_EQ(expected_width_, width);
- EXPECT_EQ(expected_height_, height);
- ++number_of_captured_frames_;
- return 0;
- }
-
- int number_of_captured_frames_;
-
- protected:
- unsigned int expected_width_;
- unsigned int expected_height_;
- };
-
-void ViEAutoTest::ViECaptureStandardTest()
-{
- //***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- //***************************************************************
-
- //***************************************************************
- // Engine ready. Begin testing class
- //***************************************************************
-
- TbInterfaces ViE("ViECaptureStandardTest");
-
- webrtc::VideoCaptureModule::DeviceInfo* devInfo =
- webrtc::VideoCaptureFactory::CreateDeviceInfo(0);
-
- int numberOfCaptureDevices = devInfo->NumberOfDevices();
- ViETest::Log("Number of capture devices %d", numberOfCaptureDevices);
- EXPECT_GT(numberOfCaptureDevices, 0);
-
- int captureDeviceId[10];
- memset(captureDeviceId, 0, sizeof(captureDeviceId));
- webrtc::VideoCaptureModule* vcpms[10];
- memset(vcpms, 0, sizeof(vcpms));
-
- // Check capabilities
- for (int deviceIndex = 0;
- deviceIndex < numberOfCaptureDevices;
- ++deviceIndex)
- {
- WebRtc_UWord8 deviceName[128];
- WebRtc_UWord8 deviceUniqueName[512];
-
- EXPECT_EQ(0, devInfo->GetDeviceName(deviceIndex,
- deviceName,
- sizeof(deviceName),
- deviceUniqueName,
- sizeof(deviceUniqueName)));
- ViETest::Log("Found capture device %s\nUnique name %s", deviceName,
- deviceUniqueName);
-
-#if !defined(WEBRTC_MAC_INTEL) // these functions will return -1
- int numberOfCapabilities =
- devInfo->NumberOfCapabilities(deviceUniqueName);
- EXPECT_GT(numberOfCapabilities, 0);
-
- for (int capIndex = 0; capIndex < numberOfCapabilities; ++capIndex)
- {
- webrtc::VideoCaptureCapability capability;
- EXPECT_EQ(0, devInfo->GetCapability(deviceUniqueName, capIndex,
- capability));
- ViETest::Log("Capture capability %d (of %u)", capIndex + 1,
- numberOfCapabilities);
- ViETest::Log("witdh %d, height %d, frame rate %d",
- capability.width, capability.height, capability.maxFPS);
- ViETest::Log("expected delay %d, color type %d, encoding %d",
- capability.expectedCaptureDelay, capability.rawType,
- capability.codecType);
-
- EXPECT_GT(capability.width, 0);
- EXPECT_GT(capability.height, 0);
- EXPECT_GT(capability.maxFPS, -1); // >= 0
- EXPECT_GT(capability.expectedCaptureDelay, 0);
- }
-#endif
- }
- // Capture Capability Functions are not supported on WEBRTC_MAC_INTEL.
-#if !defined(WEBRTC_MAC_INTEL)
-
- // Check allocation. Try to allocate them all after each other.
- for (int deviceIndex = 0;
- deviceIndex < numberOfCaptureDevices;
- ++deviceIndex)
- {
- WebRtc_UWord8 deviceName[128];
- WebRtc_UWord8 deviceUniqueName[512];
-
- EXPECT_EQ(0, devInfo->GetDeviceName(deviceIndex,
- deviceName,
- sizeof(deviceName),
- deviceUniqueName,
- sizeof(deviceUniqueName)));
-
- webrtc::VideoCaptureModule* vcpm =
- webrtc::VideoCaptureFactory::Create(
- deviceIndex, deviceUniqueName);
- EXPECT_TRUE(vcpm != NULL);
- vcpm->AddRef();
- vcpms[deviceIndex] = vcpm;
-
- EXPECT_EQ(0, ViE.capture->AllocateCaptureDevice(
- *vcpm, captureDeviceId[deviceIndex]));
-
- webrtc::VideoCaptureCapability capability;
- EXPECT_EQ(0, devInfo->GetCapability(deviceUniqueName, 0, capability));
-
- // Test that the camera select the closest capability to the selected
- // width and height.
- CaptureEffectFilter filter(capability.width, capability.height);
- EXPECT_EQ(0, ViE.image_process->RegisterCaptureEffectFilter(
- captureDeviceId[deviceIndex], filter));
-
- ViETest::Log("Testing Device %s capability width %d height %d",
- deviceUniqueName, capability.width, capability.height);
-
- if (FLAGS_capture_test_ensure_resolution_alignment_in_capture_device) {
- // This tests that the capture device properly aligns to a
- // multiple of 16 (or at least 8).
- capability.height = capability.height - 2;
- capability.width = capability.width - 2;
- }
-
- webrtc::CaptureCapability vieCapability;
- vieCapability.width = capability.width;
- vieCapability.height = capability.height;
- vieCapability.codecType = capability.codecType;
- vieCapability.maxFPS = capability.maxFPS;
- vieCapability.rawType = capability.rawType;
-
- EXPECT_EQ(0, ViE.capture->StartCapture(captureDeviceId[deviceIndex],
- vieCapability));
- webrtc::TickTime startTime = webrtc::TickTime::Now();
-
- while (filter.number_of_captured_frames_ < 10
- && (webrtc::TickTime::Now() - startTime).Milliseconds() < 10000)
- {
- AutoTestSleep(100);
- }
-
- EXPECT_GT(filter.number_of_captured_frames_, 9) <<
- "Should capture at least some frames";
-
- EXPECT_EQ(0, ViE.image_process->DeregisterCaptureEffectFilter(
- captureDeviceId[deviceIndex]));
-
-#ifdef WEBRTC_ANDROID // Can only allocate one camera at the time on Android.
- EXPECT_EQ(0, ViE.capture->StopCapture(captureDeviceId[deviceIndex]));
- EXPECT_EQ(0, ViE.capture->ReleaseCaptureDevice(
- captureDeviceId[deviceIndex]));
-#endif
- }
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
-
-
- // Stop all started capture devices.
- for (int deviceIndex = 0;
- deviceIndex < numberOfCaptureDevices;
- ++deviceIndex) {
-#if !defined(WEBRTC_ANDROID)
- // Don't stop on Android since we can only allocate one camera.
- EXPECT_EQ(0, ViE.capture->StopCapture(
- captureDeviceId[deviceIndex]));
- EXPECT_EQ(0, ViE.capture->ReleaseCaptureDevice(
- captureDeviceId[deviceIndex]));
-#endif // !WEBRTC_ANDROID
- vcpms[deviceIndex]->Release();
- }
-#endif // !WEBRTC_MAC_INTEL
-}
-
-void ViEAutoTest::ViECaptureExtendedTest() {
- ViECaptureStandardTest();
- ViECaptureAPITest();
- ViECaptureExternalCaptureTest();
-}
-
-void ViEAutoTest::ViECaptureAPITest()
-{
- //***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- //***************************************************************
-
- //***************************************************************
- // Engine ready. Begin testing class
- //***************************************************************
- TbInterfaces ViE("ViECaptureAPITest");
-
- ViE.capture->NumberOfCaptureDevices();
-
- WebRtc_UWord8 deviceName[128];
- WebRtc_UWord8 deviceUniqueName[512];
- int captureId = 0;
-
- webrtc::VideoCaptureModule::DeviceInfo* devInfo =
- webrtc::VideoCaptureFactory::CreateDeviceInfo(0);
- EXPECT_TRUE(devInfo != NULL);
-
- // Get the first capture device
- EXPECT_EQ(0, devInfo->GetDeviceName(0, deviceName,
- sizeof(deviceName),
- deviceUniqueName,
- sizeof(deviceUniqueName)));
-
- webrtc::VideoCaptureModule* vcpm =
- webrtc::VideoCaptureFactory::Create(0, deviceUniqueName);
- vcpm->AddRef();
- EXPECT_TRUE(vcpm != NULL);
-
- // Allocate capture device.
- EXPECT_EQ(0, ViE.capture->AllocateCaptureDevice(*vcpm, captureId));
-
- // Start the capture device.
- EXPECT_EQ(0, ViE.capture->StartCapture(captureId));
-
- // Start again. Should fail.
- EXPECT_NE(0, ViE.capture->StartCapture(captureId));
- EXPECT_EQ(kViECaptureDeviceAlreadyStarted, ViE.LastError());
-
- // Start invalid capture device.
- EXPECT_NE(0, ViE.capture->StartCapture(captureId + 1));
- EXPECT_EQ(kViECaptureDeviceDoesNotExist, ViE.LastError());
-
- // Stop invalid capture device.
- EXPECT_NE(0, ViE.capture->StopCapture(captureId + 1));
- EXPECT_EQ(kViECaptureDeviceDoesNotExist, ViE.LastError());
-
- // Stop the capture device.
- EXPECT_EQ(0, ViE.capture->StopCapture(captureId));
-
- // Stop the capture device again.
- EXPECT_NE(0, ViE.capture->StopCapture(captureId));
- EXPECT_EQ(kViECaptureDeviceNotStarted, ViE.LastError());
-
- // Connect to invalid channel.
- EXPECT_NE(0, ViE.capture->ConnectCaptureDevice(captureId, 0));
- EXPECT_EQ(kViECaptureDeviceInvalidChannelId, ViE.LastError());
-
- TbVideoChannel channel(ViE);
-
- // Connect invalid captureId.
- EXPECT_NE(0, ViE.capture->ConnectCaptureDevice(captureId + 1,
- channel.videoChannel));
- EXPECT_EQ(kViECaptureDeviceDoesNotExist, ViE.LastError());
-
- // Connect the capture device to the channel.
- EXPECT_EQ(0, ViE.capture->ConnectCaptureDevice(captureId,
- channel.videoChannel));
-
- // Connect the channel again.
- EXPECT_NE(0, ViE.capture->ConnectCaptureDevice(captureId,
- channel.videoChannel));
- EXPECT_EQ(kViECaptureDeviceAlreadyConnected, ViE.LastError());
-
- // Start the capture device.
- EXPECT_EQ(0, ViE.capture->StartCapture(captureId));
-
- // Release invalid capture device.
- EXPECT_NE(0, ViE.capture->ReleaseCaptureDevice(captureId + 1));
- EXPECT_EQ(kViECaptureDeviceDoesNotExist, ViE.LastError());
-
- // Release the capture device.
- EXPECT_EQ(0, ViE.capture->ReleaseCaptureDevice(captureId));
-
- // Release the capture device again.
- EXPECT_NE(0, ViE.capture->ReleaseCaptureDevice(captureId));
- EXPECT_EQ(kViECaptureDeviceDoesNotExist, ViE.LastError());
-
- // Test GetOrientation.
- webrtc::VideoCaptureRotation orientation;
- WebRtc_UWord8 dummy_name[5];
- EXPECT_NE(0, devInfo->GetOrientation(dummy_name, orientation));
-
- // Test SetRotation.
- EXPECT_NE(0, ViE.capture->SetRotateCapturedFrames(
- captureId, webrtc::RotateCapturedFrame_90));
- EXPECT_EQ(kViECaptureDeviceDoesNotExist, ViE.LastError());
-
- // Allocate capture device.
- EXPECT_EQ(0, ViE.capture->AllocateCaptureDevice(*vcpm, captureId));
-
- EXPECT_EQ(0, ViE.capture->SetRotateCapturedFrames(
- captureId, webrtc::RotateCapturedFrame_0));
- EXPECT_EQ(0, ViE.capture->SetRotateCapturedFrames(
- captureId, webrtc::RotateCapturedFrame_90));
- EXPECT_EQ(0, ViE.capture->SetRotateCapturedFrames(
- captureId, webrtc::RotateCapturedFrame_180));
- EXPECT_EQ(0, ViE.capture->SetRotateCapturedFrames(
- captureId, webrtc::RotateCapturedFrame_270));
-
- // Release the capture device
- EXPECT_EQ(0, ViE.capture->ReleaseCaptureDevice(captureId));
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
- delete devInfo;
- vcpm->Release();
-}
-
-void ViEAutoTest::ViECaptureExternalCaptureTest()
-{
- //***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- //***************************************************************
-
- TbInterfaces ViE("ViECaptureExternalCaptureTest");
- TbVideoChannel channel(ViE);
- channel.StartReceive();
- channel.StartSend();
-
- webrtc::VideoCaptureExternal* externalCapture;
- int captureId = 0;
-
- // Allocate the external capture device.
- webrtc::VideoCaptureModule* vcpm = webrtc::VideoCaptureFactory::Create(
- 0, externalCapture);
- EXPECT_TRUE(vcpm != NULL);
- EXPECT_TRUE(externalCapture != NULL);
- vcpm->AddRef();
-
- EXPECT_EQ(0, ViE.capture->AllocateCaptureDevice(*vcpm, captureId));
-
- // Connect the capture device to the channel.
- EXPECT_EQ(0, ViE.capture->ConnectCaptureDevice(captureId,
- channel.videoChannel));
-
- // Render the local capture.
- EXPECT_EQ(0, ViE.render->AddRenderer(captureId, _window1, 1, 0.0, 0.0,
- 1.0, 1.0));
-
- // Render the remote capture.
- EXPECT_EQ(0, ViE.render->AddRenderer(channel.videoChannel, _window2, 1,
- 0.0, 0.0, 1.0, 1.0));
- EXPECT_EQ(0, ViE.render->StartRender(captureId));
- EXPECT_EQ(0, ViE.render->StartRender(channel.videoChannel));
-
- // Register observer.
- CaptureObserver observer;
- EXPECT_EQ(0, ViE.capture->RegisterObserver(captureId, observer));
-
- // Enable brightness alarm.
- EXPECT_EQ(0, ViE.capture->EnableBrightnessAlarm(captureId, true));
-
- CaptureEffectFilter effectFilter(176, 144);
- EXPECT_EQ(0, ViE.image_process->RegisterCaptureEffectFilter(captureId,
- effectFilter));
-
- // Call started.
- ViETest::Log("You should see local preview from external capture\n"
- "in window 1 and the remote video in window 2.\n");
-
- //***************************************************************
- // Engine ready. Begin testing class
- //***************************************************************
- const unsigned int videoFrameLength = (176 * 144 * 3) / 2;
- unsigned char* videoFrame = new unsigned char[videoFrameLength];
- memset(videoFrame, 128, 176 * 144);
-
- int frameCount = 0;
- webrtc::VideoCaptureCapability capability;
- capability.width = 176;
- capability.height = 144;
- capability.rawType = webrtc::kVideoI420;
-
- ViETest::Log("Testing external capturing and frame rate callbacks.");
- // TODO: Change when using a real file!
- // while (fread(videoFrame, videoFrameLength, 1, foreman) == 1)
- while (frameCount < 120)
- {
- externalCapture->IncomingFrame(
- videoFrame, videoFrameLength, capability,
- webrtc::TickTime::Now().MillisecondTimestamp());
- AutoTestSleep(33);
-
- if (effectFilter.number_of_captured_frames_ > 2)
- {
- EXPECT_EQ(webrtc::Normal, observer._brightness) <<
- "Brightness or picture alarm should not have been called yet.";
- EXPECT_EQ(webrtc::AlarmCleared, observer._alarm) <<
- "Brightness or picture alarm should not have been called yet.";
- }
- frameCount++;
- }
-
- // Test brightness alarm
- // Test bright image
- for (int i = 0; i < 176 * 144; ++i)
- {
- if (videoFrame[i] <= 155)
- videoFrame[i] = videoFrame[i] + 100;
- else
- videoFrame[i] = 255;
- }
- ViETest::Log("Testing Brighness alarm");
- for (int frame = 0; frame < 30; ++frame)
- {
- externalCapture->IncomingFrame(
- videoFrame, videoFrameLength, capability,
- webrtc::TickTime::Now().MillisecondTimestamp());
- AutoTestSleep(33);
- }
- EXPECT_EQ(webrtc::Bright, observer._brightness) <<
- "Should be bright at this point since we are using a bright image.";
-
- // Test Dark image
- for (int i = 0; i < 176 * 144; ++i)
- {
- videoFrame[i] = videoFrame[i] > 200 ? videoFrame[i] - 200 : 0;
- }
- for (int frame = 0; frame < 30; ++frame)
- {
- externalCapture->IncomingFrame(
- videoFrame, videoFrameLength, capability,
- webrtc::TickTime::Now().MillisecondTimestamp());
- AutoTestSleep(33);
- }
- EXPECT_EQ(webrtc::Dark, observer._brightness) <<
- "Should be dark at this point since we are using a dark image.";
- EXPECT_GT(effectFilter.number_of_captured_frames_, 150) <<
- "Frames should have been played.";
-
- EXPECT_GE(observer._frameRate, 29) <<
- "Frame rate callback should be approximately correct.";
- EXPECT_LE(observer._frameRate, 30) <<
- "Frame rate callback should be approximately correct.";
-
- // Test no picture alarm
- ViETest::Log("Testing NoPictureAlarm.");
- AutoTestSleep(1050);
-
- EXPECT_EQ(webrtc::AlarmRaised, observer._alarm) <<
- "No picture alarm should be raised.";
- for (int frame = 0; frame < 10; ++frame)
- {
- externalCapture->IncomingFrame(
- videoFrame, videoFrameLength, capability,
- webrtc::TickTime::Now().MillisecondTimestamp());
- AutoTestSleep(33);
- }
- EXPECT_EQ(webrtc::AlarmCleared, observer._alarm) <<
- "Alarm should be cleared since ge just got some data.";
-
- delete videoFrame;
-
- // Release the capture device
- EXPECT_EQ(0, ViE.capture->ReleaseCaptureDevice(captureId));
-
- // Release the capture device again
- EXPECT_NE(0, ViE.capture->ReleaseCaptureDevice(captureId));
- EXPECT_EQ(kViECaptureDeviceDoesNotExist, ViE.LastError());
- vcpm->Release();
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_carbon_mac.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_carbon_mac.cc
deleted file mode 100644
index 38d9c4e..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_carbon_mac.cc
+++ /dev/null
@@ -1,355 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "engine_configurations.h"
-
-#if defined(CARBON_RENDERING)
-#include "vie_autotest_mac_carbon.h"
-#include "vie_autotest_defines.h"
-#include "vie_autotest.h"
-#include "vie_autotest_main.h"
-
-ViEAutoTestWindowManager::ViEAutoTestWindowManager() :
- _carbonWindow1(new WindowRef()),
- _carbonWindow2(new WindowRef()),
- _hiView1(new HIViewRef()),
- _hiView2(new HIViewRef())
-{
-}
-
-ViEAutoTestWindowManager::~ViEAutoTestWindowManager()
-{
- if (_carbonWindow1EventHandlerRef)
- RemoveEventHandler(_carbonWindow1EventHandlerRef);
-
- if (_carbonWindow2EventHandlerRef)
- RemoveEventHandler(_carbonWindow2EventHandlerRef);
-
- if (_carbonHIView1EventHandlerRef)
- RemoveEventHandler(_carbonHIView1EventHandlerRef);
-
- if (_carbonHIView2EventHandlerRef)
- RemoveEventHandler(_carbonHIView2EventHandlerRef);
-
- delete _carbonWindow1;
- delete _carbonWindow2;
- delete _hiView1;
- delete _hiView2;
-}
-
-int ViEAutoTestWindowManager::CreateWindows(AutoTestRect window1Size,
- AutoTestRect window2Size,
- char* window1Title,
- char* window2Title)
-{
-
- WindowAttributes windowAttributes = kWindowStandardDocumentAttributes
- | kWindowStandardHandlerAttribute | kWindowCompositingAttribute;
- Rect windowContentRect;
- static const EventTypeSpec
- windowEventTypes[] = { kEventClassWindow, kEventWindowBoundsChanged,
- kEventClassWindow, kEventWindowBoundsChanging, kEventClassWindow,
- kEventWindowZoomed, kEventClassWindow, kEventWindowExpanded,
- kEventClassWindow, kEventWindowClickResizeRgn, kEventClassWindow,
- kEventWindowClickDragRgn };
-
- // ************* Window 1 and Event Handler ***********************
-
- SetRect(&windowContentRect, window1Size.origin.x, window1Size.origin.y,
- window1Size.origin.x + window1Size.size.width, window1Size.origin.y
- + window1Size.size.height);
-
- CreateNewWindow(kDocumentWindowClass, windowAttributes, &windowContentRect,
- _carbonWindow1);
- SetWindowTitleWithCFString(*_carbonWindow1, CFSTR("Carbon Window 1"));
- ShowWindow(*_carbonWindow1);
- InitCursor();
- InstallWindowEventHandler(*_carbonWindow1,
- NewEventHandlerUPP(HandleWindowEvent),
- GetEventTypeCount(windowEventTypes),
- windowEventTypes, (void*) this,
- &_carbonWindow1EventHandlerRef);
-
- // ************* Window 2 and Event Handler ***********************
-
- SetRect(&windowContentRect, window2Size.origin.x, window2Size.origin.y,
- window2Size.origin.x + window2Size.size.width, window2Size.origin.y
- + window2Size.size.height);
-
- CreateNewWindow(kDocumentWindowClass, windowAttributes, &windowContentRect,
- _carbonWindow2);
- SetWindowTitleWithCFString(*_carbonWindow2, CFSTR("Carbon Window 2"));
- ShowWindow(*_carbonWindow2);
- InitCursor();
- InstallWindowEventHandler(*_carbonWindow2,
- NewEventHandlerUPP(HandleWindowEvent),
- GetEventTypeCount(windowEventTypes),
- windowEventTypes, (void*) this,
- &_carbonWindow2EventHandlerRef);
-
-#if defined(HIVIEWREF_MODE)
- OSStatus status;
- static const EventTypeSpec hiviewEventTypes[] = { kEventClassControl,
- kEventControlBoundsChanged, kEventClassControl, kEventControlDraw };
-
- HIRect hiView1Rect = { 10, 10, 200, 200 };
- status = HICreateCustomView(&hiView1Rect, &_hiView1);
- status = HIViewAddSubview(&_carbonWindow1, _hiView1);
- HIViewSetZOrder(_hiView1, kHIViewZOrderAbove, NULL);
- HIViewSetVisible(_hiView1, true);
-
- HIViewInstallEventHandler(_hiView1, NewEventHandlerUPP(HandleHIViewEvent),
- GetEventTypeCount(hiviewEventTypes),
- hiviewEventTypes, (void *) this,
- &_carbonHIView1EventHandlerRef);
-
- HIRect hiView2Rect = { 10, 10, 200, 200 };
- status = HICreateCustomView(&hiView2Rect, &_hiView2);
- status = HIViewAddSubview(&_carbonWindow2, _hiView2);
- HIViewSetZOrder(_hiView2, kHIViewZOrderAbove, NULL);
- HIViewSetVisible(_hiView2, true);
-
- HIViewInstallEventHandler(_hiView2, NewEventHandlerUPP(HandleHIViewEvent),
- GetEventTypeCount(hiviewEventTypes),
- hiviewEventTypes, (void *) this,
- &_carbonHIView2EventHandlerRef);
-#endif
-
- return 0;
-}
-
-pascal OSStatus ViEAutoTestWindowManager::HandleWindowEvent(
- EventHandlerCallRef nextHandler, EventRef theEvent, void* userData)
-{
-
- WindowRef windowRef = NULL;
-
- int eventType = GetEventKind(theEvent);
-
- // see https://dcs.sourcerepo.com/dcs/tox_view/trunk/tox/libraries/
- // i686-win32/include/quicktime/CarbonEvents.h for a list of codes
- GetEventParameter(theEvent, kEventParamDirectObject, typeWindowRef, NULL,
- sizeof(WindowRef), NULL, &windowRef);
-
- ViEAutoTestWindowManager* obj = (ViEAutoTestWindowManager*) (userData);
-
- if (windowRef == obj->GetWindow1())
- {
- // event was triggered on window 1
- }
- else if (windowRef == obj->GetWindow2())
- {
- // event was triggered on window 2
- }
-
- if (kEventWindowBoundsChanged == eventType)
- {
- }
- else if (kEventWindowBoundsChanging == eventType)
- {
- }
- else if (kEventWindowZoomed == eventType)
- {
- }
- else if (kEventWindowExpanding == eventType)
- {
- }
- else if (kEventWindowExpanded == eventType)
- {
- }
- else if (kEventWindowClickResizeRgn == eventType)
- {
- }
- else if (kEventWindowClickDragRgn == eventType)
- {
- }
- else
- {
- }
-
- return noErr;
-}
-
-pascal OSStatus ViEAutoTestWindowManager::HandleHIViewEvent(
- EventHandlerCallRef nextHandler, EventRef theEvent, void* userData)
-{
- HIViewRef hiviewRef = NULL;
-
- // see https://dcs.sourcerepo.com/dcs/tox_view/trunk/tox/libraries/
- // i686-win32/include/quicktime/CarbonEvents.h for a list of codes
- int eventType = GetEventKind(theEvent);
- OSStatus status = noErr;
- status = GetEventParameter(theEvent, kEventParamDirectObject,
- typeControlRef, NULL, sizeof(ControlRef), NULL,
- &hiviewRef);
-
- if (GetEventClass(theEvent) == kEventClassControl)
- {
- if (GetEventKind(theEvent) == kEventControlDraw)
- {
- ViEAutoTestWindowManager* obj =
- (ViEAutoTestWindowManager*) (userData);
-
- CGContextRef context;
- status = GetEventParameter(theEvent, kEventParamCGContextRef,
- typeCGContextRef, NULL, sizeof(context),
- NULL, &context);
- HIRect viewBounds;
-
- HIViewRef* ptrHIViewRef =
- static_cast<HIViewRef*> (obj->GetWindow1());
- if (hiviewRef == *ptrHIViewRef)
- {
- // color hiview1
- CGContextSetRGBFillColor(context, 1, 0, 0, 1);
- HIViewGetBounds(hiviewRef, &viewBounds);
- CGContextFillRect(context, viewBounds);
- }
-
- ptrHIViewRef = static_cast<HIViewRef*> (obj->GetWindow1());
- if (hiviewRef == *ptrHIViewRef)
- {
- CGContextSetRGBFillColor(context, 0, 1, 0, 1);
- HIViewGetBounds(hiviewRef, &viewBounds);
- CGContextFillRect(context, viewBounds);
- }
-
- }
- }
-
- /*
-
-
- VideoRenderAGL* obj = (VideoRenderAGL*)(userData);
- WindowRef parentWindow = HIViewGetWindow(hiviewRef);
- bool updateUI = true;
-
- if(kEventControlBoundsChanged == eventType){
- }
- else if(kEventControlDraw == eventType){
- }
- else{
- updateUI = false;
- }
-
- if(true == updateUI){
- obj->ParentWindowResized(parentWindow);
- obj->UpdateClipping();
- obj->RenderOffScreenBuffers();
- }
- */
-
- return status;
-}
-
-int ViEAutoTestWindowManager::TerminateWindows()
-{
- return 0;
-}
-
-void* ViEAutoTestWindowManager::GetWindow1()
-{
-#if defined(HIVIEWREF_MODE)
- return (void*)_hiView1;
-#else
- return (void*) _carbonWindow1;
-#endif
-
-}
-void* ViEAutoTestWindowManager::GetWindow2()
-{
-#if defined(HIVIEWREF_MODE)
- return (void*)_hiView2;
-#else
- return (void*) _carbonWindow2;
-#endif
-
-}
-
-bool ViEAutoTestWindowManager::SetTopmostWindow()
-{
- return true;
-}
-
-/*
-
- int main (int argc, const char * argv[])
- {
- ViEAutoTestMain autoTest;
-
- if(argc > 1){
- autoTest.UseAnswerFile(argv[1]);
- }
-
- int success = autoTest.BeginOSIndependentTesting();
-
- }
-
- */
-
-int main(int argc, const char * argv[])
-{
- NSAutoreleasePool* pool = [[NSAutoreleasePool alloc] init];
-
- [NSApplication sharedApplication];
-
- // We have to run the test in a secondary thread because we need to run a
- // runloop, which blocks.
- if (argc > 1)
- {
-AutoTestClass * autoTestClass = [[AutoTestClass alloc]init];
- [NSThread detachNewThreadSelector:@selector(autoTestWithArg:)
- toTarget:autoTestClass withObject:[NSString stringWithFormat:@"%s",
- argv[1]]];
-}
-else
-{
- AutoTestClass* autoTestClass = [[AutoTestClass alloc]init];
- [NSThread detachNewThreadSelector:@selector(autoTestWithArg:)
- toTarget:autoTestClass withObject:nil];
-}
-
-// process OS events. Blocking call
-[[NSRunLoop currentRunLoop]run];
-[pool release];
-}
-
-@implementation AutoTestClass
-
--(void)autoTestWithArg:(NSString*)answerFile;
-{
- // TODO(phoglund): Rewrite this file to work with the new way of running
- // vie_auto_test. The file doesn't seem to be used at the moment though.
- NSAutoreleasePool* pool = [[NSAutoreleasePool alloc] init];
-
- ViEAutoTestMain autoTest;
-
- if(NSOrderedSame != [answerFile compare:@""])
- {
- char answerFileUTF8[1024] = "";
- strcpy(answerFileUTF8, (char*)[answerFileUTF8 UTF8]);
- autoTest.UseAnswerFile(answerFileUTF8);
- }
-
- int success = autoTest.BeginOSIndependentTesting();
-
- [pool release];
- return;
-}
-// TODO: move window creation to Obj-c class so GUI commands can be run on the
-// main NSThread
-// -(void)createWindow1:(AutoTestRect)window1Size
-// AndWindow2:(AutoTestRect)window2Size WithTitle1:(char*)window1Title
-// AndTitle2:(char*)window2Title{
-
-@end
-
-#endif
-
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_cocoa_mac.mm b/trunk/src/video_engine/test/auto_test/source/vie_autotest_cocoa_mac.mm
deleted file mode 100644
index 3f9ef6b..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_cocoa_mac.mm
+++ /dev/null
@@ -1,132 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "engine_configurations.h"
-
-#if defined(COCOA_RENDERING)
-#import "cocoa_render_view.h"
-#include "vie_autotest_mac_cocoa.h"
-#include "vie_autotest_defines.h"
-#include "vie_autotest.h"
-#include "vie_autotest_main.h"
-
-ViEAutoTestWindowManager::ViEAutoTestWindowManager()
- : _cocoaRenderView1(nil), _cocoaRenderView2(nil) {
-}
-
-int ViEAutoTestWindowManager::CreateWindows(AutoTestRect window1Size,
- AutoTestRect window2Size,
- void* window1Title,
- void* window2Title) {
- NSRect outWindow1Frame = NSMakeRect(window1Size.origin.x,
- window1Size.origin.y,
- window1Size.size.width,
- window1Size.size.height);
- outWindow1_ = [[NSWindow alloc] initWithContentRect:outWindow1Frame
- styleMask:NSTitledWindowMask
- backing:NSBackingStoreBuffered defer:NO];
- [outWindow1_ orderOut:nil];
- NSRect cocoaRenderView1Frame = NSMakeRect(0, 0, window1Size.size.width,
- window1Size.size.height);
- _cocoaRenderView1 = [[CocoaRenderView alloc]
- initWithFrame:cocoaRenderView1Frame];
- [[outWindow1_ contentView] addSubview:(NSView*)_cocoaRenderView1];
- [outWindow1_ setTitle:[NSString stringWithFormat:@"%s", window1Title]];
- [outWindow1_ makeKeyAndOrderFront:NSApp];
-
- NSRect outWindow2Frame = NSMakeRect(window2Size.origin.x,
- window2Size.origin.y,
- window2Size.size.width,
- window2Size.size.height);
- outWindow2_ = [[NSWindow alloc] initWithContentRect:outWindow2Frame
- styleMask:NSTitledWindowMask
- backing:NSBackingStoreBuffered defer:NO];
- [outWindow2_ orderOut:nil];
- NSRect cocoaRenderView2Frame = NSMakeRect(0, 0, window2Size.size.width,
- window2Size.size.height);
- _cocoaRenderView2 = [[CocoaRenderView alloc]
- initWithFrame:cocoaRenderView2Frame];
- [[outWindow2_ contentView] addSubview:(NSView*)_cocoaRenderView2];
- [outWindow2_ setTitle:[NSString stringWithFormat:@"%s", window2Title]];
- [outWindow2_ makeKeyAndOrderFront:NSApp];
-
- return 0;
-}
-
-int ViEAutoTestWindowManager::TerminateWindows() {
- [outWindow1_ close];
- [outWindow2_ close];
- return 0;
-}
-
-void* ViEAutoTestWindowManager::GetWindow1() {
- return _cocoaRenderView1;
-}
-
-void* ViEAutoTestWindowManager::GetWindow2() {
- return _cocoaRenderView2;
-}
-
-bool ViEAutoTestWindowManager::SetTopmostWindow() {
- return true;
-}
-
-int main(int argc, char * argv[]) {
- NSAutoreleasePool* pool = [[NSAutoreleasePool alloc] init];
-
- [NSApplication sharedApplication];
-
- int result = 0;
-#if defined(MAC_COCOA_USE_NSRUNLOOP)
- AutoTestClass* tests = [[AutoTestClass alloc] init];
-
- [tests setArgc:argc argv:argv];
- [NSThread detachNewThreadSelector:@selector(autoTestWithArg:)
- toTarget:tests withObject:nil];
- // Process OS events. Blocking call.
- [[NSRunLoop mainRunLoop]run];
-
- result = [tests result];
-
-#else
- ViEAutoTestMain autoTest;
- result = autoTest.RunTests(argc, argv);
-
-#endif
- [pool release];
- return result;
-}
-
-@implementation AutoTestClass
-
-- (void)setArgc:(int)argc argv:(char**)argv {
- argc_ = argc;
- argv_ = argv;
-}
-
-- (void)autoTestWithArg:(NSObject*)ignored {
- NSAutoreleasePool* pool = [[NSAutoreleasePool alloc] init];
-
- ViEAutoTestMain auto_test;
-
- result_ = auto_test.RunTests(argc_, argv_);
-
- [pool release];
- return;
-}
-
-- (int)result {
- return result_;
-}
-
-@end
-
-#endif
-
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_codec.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_codec.cc
deleted file mode 100644
index ed8f20b..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_codec.cc
+++ /dev/null
@@ -1,455 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vie_autotest.h"
-
-#include "codec_primitives.h"
-#include "common_types.h"
-#include "general_primitives.h"
-#include "tb_capture_device.h"
-#include "tb_I420_codec.h"
-#include "tb_interfaces.h"
-#include "tb_video_channel.h"
-#include "vie_autotest_defines.h"
-
-class RenderFilter : public webrtc::ViEEffectFilter {
- public:
- RenderFilter()
- : last_render_width_(0),
- last_render_height_(0) {}
-
- ~RenderFilter() {}
-
- virtual int Transform(int size, unsigned char* frame_buffer,
- unsigned int time_stamp, unsigned int width,
- unsigned int height) {
- last_render_width_ = width;
- last_render_height_ = height;
- return 0;
- }
- unsigned int last_render_width_;
- unsigned int last_render_height_;
-};
-
-void ViEAutoTest::ViECodecStandardTest()
-{
- TbInterfaces interfaces = TbInterfaces("ViECodecStandardTest");
- TbCaptureDevice capture_device = TbCaptureDevice(interfaces);
-
- int video_channel = -1;
-
- EXPECT_EQ(0, interfaces.base->CreateChannel(video_channel));
- EXPECT_EQ(0, interfaces.capture->ConnectCaptureDevice(
- capture_device.captureId, video_channel));
-
- ConfigureRtpRtcp(interfaces.rtp_rtcp,
- video_channel);
-
- RenderInWindow(interfaces.render, capture_device.captureId, _window1, 0);
- RenderInWindow(interfaces.render, video_channel, _window2, 1);
-
- TestCodecs(interfaces, capture_device.captureId, video_channel,
- kDoNotForceResolution, kDoNotForceResolution);
-}
-
-void ViEAutoTest::ViECodecExtendedTest()
-{
- {
- ViECodecAPITest();
- ViECodecStandardTest();
- ViECodecExternalCodecTest();
-
- TbInterfaces interfaces = TbInterfaces("ViECodecExtendedTest");
- webrtc::ViEBase* ptrViEBase = interfaces.base;
- webrtc::ViECapture* ptrViECapture = interfaces.capture;
- webrtc::ViERender* ptrViERender = interfaces.render;
- webrtc::ViECodec* ptrViECodec = interfaces.codec;
- webrtc::ViERTP_RTCP* ptrViERtpRtcp = interfaces.rtp_rtcp;
- webrtc::ViENetwork* ptrViENetwork = interfaces.network;
-
- TbCaptureDevice captureDevice = TbCaptureDevice(interfaces);
- int captureId = captureDevice.captureId;
-
- int videoChannel = -1;
- EXPECT_EQ(0, ptrViEBase->CreateChannel(videoChannel));
- EXPECT_EQ(0, ptrViECapture->ConnectCaptureDevice(
- captureId, videoChannel));
-
- EXPECT_EQ(0, ptrViERtpRtcp->SetRTCPStatus(
- videoChannel, webrtc::kRtcpCompound_RFC4585));
- EXPECT_EQ(0, ptrViERtpRtcp->SetKeyFrameRequestMethod(
- videoChannel, webrtc::kViEKeyFrameRequestPliRtcp));
- EXPECT_EQ(0, ptrViERtpRtcp->SetTMMBRStatus(videoChannel, true));
- EXPECT_EQ(0, ptrViERender->AddRenderer(
- captureId, _window1, 0, 0.0, 0.0, 1.0, 1.0));
- EXPECT_EQ(0, ptrViERender->AddRenderer(
- videoChannel, _window2, 1, 0.0, 0.0, 1.0, 1.0));
- EXPECT_EQ(0, ptrViERender->StartRender(captureId));
- EXPECT_EQ(0, ptrViERender->StartRender(videoChannel));
-
- webrtc::VideoCodec videoCodec;
- memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
- for (int idx = 0; idx < ptrViECodec->NumberOfCodecs(); idx++)
- {
- EXPECT_EQ(0, ptrViECodec->GetCodec(idx, videoCodec));
-
- if (videoCodec.codecType != webrtc::kVideoCodecI420)
- {
- videoCodec.width = 640;
- videoCodec.height = 480;
- }
- EXPECT_EQ(0, ptrViECodec->SetReceiveCodec(
- videoChannel, videoCodec));
- }
-
- const char* ipAddress = "127.0.0.1";
- const unsigned short rtpPort = 6000;
- EXPECT_EQ(0, ptrViENetwork->SetLocalReceiver(videoChannel, rtpPort));
- EXPECT_EQ(0, ptrViEBase->StartReceive(videoChannel));
- EXPECT_EQ(0, ptrViENetwork->SetSendDestination(
- videoChannel, ipAddress, rtpPort));
- EXPECT_EQ(0, ptrViEBase->StartSend(videoChannel));
-
- //
- // Codec specific tests
- //
- memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
- EXPECT_EQ(0, ptrViEBase->StopSend(videoChannel));
- ViEAutotestCodecObserver codecObserver;
- EXPECT_EQ(0, ptrViECodec->RegisterEncoderObserver(
- videoChannel, codecObserver));
- EXPECT_EQ(0, ptrViECodec->RegisterDecoderObserver(
- videoChannel, codecObserver));
-
- EXPECT_EQ(0, ptrViEBase->StopReceive(videoChannel));
- EXPECT_NE(0, ptrViEBase->StopSend(videoChannel)); // Already stopped
-
- EXPECT_EQ(0, ptrViERender->StopRender(videoChannel));
- EXPECT_EQ(0, ptrViERender->RemoveRenderer(captureId));
- EXPECT_EQ(0, ptrViERender->RemoveRenderer(videoChannel));
- EXPECT_EQ(0, ptrViECapture->DisconnectCaptureDevice(videoChannel));
- EXPECT_EQ(0, ptrViEBase->DeleteChannel(videoChannel));
- }
-
- //
- // Default channel
- //
- {
- // Create VIE
- TbInterfaces ViE("ViECodecExtendedTest2");
- // Create a capture device
- TbCaptureDevice tbCapture(ViE);
-
- // Create channel 1
- int videoChannel1 = -1;
- EXPECT_EQ(0, ViE.base->CreateChannel(videoChannel1));
-
- unsigned short rtpPort1 = 12000;
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(
- videoChannel1, rtpPort1));
- EXPECT_EQ(0, ViE.network->SetSendDestination(
- videoChannel1, "127.0.0.1", rtpPort1));
- tbCapture.ConnectTo(videoChannel1);
-
- EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
- videoChannel1, webrtc::kViEKeyFrameRequestPliRtcp));
- EXPECT_EQ(0, ViE.render->AddRenderer(
- videoChannel1, _window1, 0, 0.0, 0.0, 1.0, 1.0));
- EXPECT_EQ(0, ViE.render->StartRender(videoChannel1));
-
- ViEAutotestCodecObserver codecObserver1;
- EXPECT_EQ(0, ViE.codec->RegisterEncoderObserver(
- videoChannel1, codecObserver1));
- EXPECT_EQ(0, ViE.codec->RegisterDecoderObserver(
- videoChannel1, codecObserver1));
-
- // Set Send codec
- unsigned short codecWidth = 176;
- unsigned short codecHeight = 144;
- bool codecSet = false;
- webrtc::VideoCodec videoCodec;
- for (int idx = 0; idx < ViE.codec->NumberOfCodecs(); idx++)
- {
- EXPECT_EQ(0, ViE.codec->GetCodec(idx, videoCodec));
- EXPECT_EQ(0, ViE.codec->SetReceiveCodec(videoChannel1, videoCodec));
- if (videoCodec.codecType == webrtc::kVideoCodecVP8)
- {
- videoCodec.width = codecWidth;
- videoCodec.height = codecHeight;
- videoCodec.startBitrate = 200;
- videoCodec.maxBitrate = 300;
- EXPECT_EQ(0, ViE.codec->SetSendCodec(
- videoChannel1, videoCodec));
- codecSet = true;
- break;
- }
- }
- EXPECT_TRUE(codecSet);
- webrtc::VideoCodec send_codec;
- memcpy(&send_codec, &videoCodec, sizeof(videoCodec));
-
- EXPECT_EQ(0, ViE.base->StartSend(videoChannel1));
- EXPECT_EQ(0, ViE.base->StartReceive(videoChannel1));
-
- // Create channel 2, based on channel 1
- int videoChannel2 = -1;
- EXPECT_EQ(0, ViE.base->CreateChannel(videoChannel2, videoChannel1));
- EXPECT_NE(videoChannel1, videoChannel2) <<
- "Channel 2 should be seop";
-
- EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
- videoChannel2, webrtc::kViEKeyFrameRequestPliRtcp));
-
- // Prepare receive codecs
- for (int idx = 0; idx < ViE.codec->NumberOfCodecs(); idx++)
- {
- EXPECT_EQ(0, ViE.codec->GetCodec(idx, videoCodec));
- EXPECT_EQ(0, ViE.codec->SetReceiveCodec(videoChannel2, videoCodec));
- }
-
- ViEAutotestCodecObserver codecObserver2;
- EXPECT_EQ(0, ViE.codec->RegisterDecoderObserver(
- videoChannel2, codecObserver2));
- EXPECT_EQ(0, ViE.render->AddRenderer(
- videoChannel2, _window2, 0, 0.0, 0.0, 1.0, 1.0));
- EXPECT_EQ(0, ViE.render->StartRender(videoChannel2));
-
- unsigned short rtpPort2 = 13000;
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(videoChannel2, rtpPort2));
- EXPECT_EQ(0, ViE.network->SetSendDestination(
- videoChannel2, "127.0.0.1", rtpPort2));
-
- EXPECT_EQ(0, ViE.base->StartReceive(videoChannel2));
- EXPECT_EQ(-1, ViE.base->StartSend(videoChannel2));
-
- EXPECT_EQ(0, ViE.base->DeleteChannel(videoChannel1));
- EXPECT_EQ(0, ViE.base->DeleteChannel(videoChannel2));
-
- }
-}
-
-void ViEAutoTest::ViECodecAPITest()
-{
- // ***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- // ***************************************************************
- webrtc::VideoEngine* ptrViE = NULL;
- ptrViE = webrtc::VideoEngine::Create();
- EXPECT_TRUE(ptrViE != NULL);
-
- webrtc::ViEBase* ptrViEBase = webrtc::ViEBase::GetInterface(ptrViE);
- EXPECT_TRUE(ptrViEBase != NULL);
-
- EXPECT_EQ(0, ptrViEBase->Init());
-
- int videoChannel = -1;
- EXPECT_EQ(0, ptrViEBase->CreateChannel(videoChannel));
-
- webrtc::ViECodec* ptrViECodec = webrtc::ViECodec::GetInterface(ptrViE);
- EXPECT_TRUE(ptrViECodec != NULL);
-
- //***************************************************************
- // Engine ready. Begin testing class
- //***************************************************************
-
- //
- // SendCodec
- //
- webrtc::VideoCodec videoCodec;
- memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
-
- const int numberOfCodecs = ptrViECodec->NumberOfCodecs();
- EXPECT_GT(numberOfCodecs, 0);
-
- SetSendCodec(webrtc::kVideoCodecVP8, ptrViECodec, videoChannel,
- kDoNotForceResolution, kDoNotForceResolution);
-
- memset(&videoCodec, 0, sizeof(videoCodec));
- EXPECT_EQ(0, ptrViECodec->GetSendCodec(videoChannel, videoCodec));
- EXPECT_EQ(webrtc::kVideoCodecVP8, videoCodec.codecType);
- // Verify that the target bit rate is equal to the start bitrate.
- unsigned int target_bitrate = 0;
- EXPECT_EQ(0, ptrViECodec->GetCodecTargetBitrate(videoChannel,
- &target_bitrate));
- EXPECT_EQ(videoCodec.startBitrate, target_bitrate);
-
- SetSendCodec(webrtc::kVideoCodecI420, ptrViECodec, videoChannel,
- kDoNotForceResolution, kDoNotForceResolution);
- memset(&videoCodec, 0, sizeof(videoCodec));
- EXPECT_EQ(0, ptrViECodec->GetSendCodec(videoChannel, videoCodec));
- EXPECT_EQ(webrtc::kVideoCodecI420, videoCodec.codecType);
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
-
- EXPECT_EQ(0, ptrViEBase->DeleteChannel(videoChannel));
-
- EXPECT_EQ(0, ptrViECodec->Release());
- EXPECT_EQ(0, ptrViEBase->Release());
- EXPECT_TRUE(webrtc::VideoEngine::Delete(ptrViE));
-}
-
-#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
-#include "vie_external_codec.h"
-#endif
-void ViEAutoTest::ViECodecExternalCodecTest()
-{
- // ***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- // ***************************************************************
-
-
- // ***************************************************************
- // Engine ready. Begin testing class
- // ***************************************************************
-
-#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
- {
- TbInterfaces ViE("ViEExternalCodec");
- TbCaptureDevice captureDevice(ViE);
- TbVideoChannel channel(
- ViE, webrtc::kVideoCodecI420, 352,288,30,(352*288*3*8*30)/(2*1000));
-
- captureDevice.ConnectTo(channel.videoChannel);
-
- EXPECT_EQ(0, ViE.render->AddRenderer(
- channel.videoChannel, _window1, 0, 0.0, 0.0, 1.0, 1.0));
- EXPECT_EQ(0, ViE.render->StartRender(channel.videoChannel));
-
- channel.StartReceive();
- channel.StartSend();
-
- ViETest::Log("Using internal I420 codec");
- AutoTestSleep(KAutoTestSleepTimeMs/2);
-
- webrtc::ViEExternalCodec* ptrViEExtCodec =
- webrtc::ViEExternalCodec::GetInterface(ViE.video_engine);
- EXPECT_TRUE(ptrViEExtCodec != NULL);
-
- webrtc::VideoCodec codecStruct;
-
- EXPECT_EQ(0, ViE.codec->GetSendCodec(
- channel.videoChannel, codecStruct));
-
- // Use external encoder instead
- {
- TbI420Encoder extEncoder;
-
- // Test to register on wrong channel
- EXPECT_NE(0, ptrViEExtCodec->RegisterExternalSendCodec(
- channel.videoChannel+5,codecStruct.plType,&extEncoder));
- EXPECT_EQ(kViECodecInvalidArgument, ViE.LastError());
-
- EXPECT_EQ(0, ptrViEExtCodec->RegisterExternalSendCodec(
- channel.videoChannel,codecStruct.plType,&extEncoder));
-
- // Use new external encoder
- EXPECT_EQ(0, ViE.codec->SetSendCodec(
- channel.videoChannel, codecStruct));
-
- TbI420Decoder extDecoder;
- EXPECT_EQ(0, ptrViEExtCodec->RegisterExternalReceiveCodec(
- channel.videoChannel,codecStruct.plType,&extDecoder));
-
- EXPECT_EQ(0, ViE.codec->SetReceiveCodec(
- channel.videoChannel, codecStruct));
-
- ViETest::Log("Using external I420 codec");
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- // Test to deregister on wrong channel
- EXPECT_NE(0, ptrViEExtCodec->DeRegisterExternalSendCodec(
- channel.videoChannel+5,codecStruct.plType));
- EXPECT_EQ(kViECodecInvalidArgument, ViE.LastError());
-
- // Test to deregister wrong payload type.
- EXPECT_NE(0, ptrViEExtCodec->DeRegisterExternalSendCodec(
- channel.videoChannel,codecStruct.plType-1));
-
- // Deregister external send codec
- EXPECT_EQ(0, ptrViEExtCodec->DeRegisterExternalSendCodec(
- channel.videoChannel,codecStruct.plType));
-
- EXPECT_EQ(0, ptrViEExtCodec->DeRegisterExternalReceiveCodec(
- channel.videoChannel,codecStruct.plType));
-
- // Verify that the encoder and decoder has been used
- TbI420Encoder::FunctionCalls encodeCalls =
- extEncoder.GetFunctionCalls();
- EXPECT_EQ(1, encodeCalls.InitEncode);
- EXPECT_EQ(1, encodeCalls.Release);
- EXPECT_EQ(1, encodeCalls.RegisterEncodeCompleteCallback);
- EXPECT_GT(encodeCalls.Encode, 30);
- EXPECT_GT(encodeCalls.SetRates, 1);
- EXPECT_GT(encodeCalls.SetPacketLoss, 1);
-
- TbI420Decoder::FunctionCalls decodeCalls =
- extDecoder.GetFunctionCalls();
- EXPECT_EQ(1, decodeCalls.InitDecode);
- EXPECT_EQ(1, decodeCalls.Release);
- EXPECT_EQ(1, decodeCalls.RegisterDecodeCompleteCallback);
- EXPECT_GT(decodeCalls.Decode, 30);
-
- ViETest::Log("Changing payload type Using external I420 codec");
-
- codecStruct.plType = codecStruct.plType - 1;
- EXPECT_EQ(0, ptrViEExtCodec->RegisterExternalReceiveCodec(
- channel.videoChannel, codecStruct.plType, &extDecoder));
-
- EXPECT_EQ(0, ViE.codec->SetReceiveCodec(
- channel.videoChannel, codecStruct));
-
- EXPECT_EQ(0, ptrViEExtCodec->RegisterExternalSendCodec(
- channel.videoChannel, codecStruct.plType, &extEncoder));
-
- // Use new external encoder
- EXPECT_EQ(0, ViE.codec->SetSendCodec(
- channel.videoChannel, codecStruct));
-
- AutoTestSleep(KAutoTestSleepTimeMs/2);
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
-
- EXPECT_EQ(0, ptrViEExtCodec->DeRegisterExternalSendCodec(
- channel.videoChannel,codecStruct.plType));
- EXPECT_EQ(0, ptrViEExtCodec->DeRegisterExternalReceiveCodec(
- channel.videoChannel,codecStruct.plType));
-
- // Verify that the encoder and decoder has been used
- encodeCalls = extEncoder.GetFunctionCalls();
- EXPECT_EQ(2, encodeCalls.InitEncode);
- EXPECT_EQ(2, encodeCalls.Release);
- EXPECT_EQ(2, encodeCalls.RegisterEncodeCompleteCallback);
- EXPECT_GT(encodeCalls.Encode, 30);
- EXPECT_GT(encodeCalls.SetRates, 1);
- EXPECT_GT(encodeCalls.SetPacketLoss, 1);
-
- decodeCalls = extDecoder.GetFunctionCalls();
-
- EXPECT_EQ(2, decodeCalls.InitDecode);
- EXPECT_EQ(2, decodeCalls.Release);
- EXPECT_EQ(2, decodeCalls.RegisterDecodeCompleteCallback);
- EXPECT_GT(decodeCalls.Decode, 30);
-
- EXPECT_EQ(0, ptrViEExtCodec->Release());
- } // tbI420Encoder and extDecoder goes out of scope
-
- ViETest::Log("Using internal I420 codec");
- AutoTestSleep(KAutoTestSleepTimeMs/2);
-
- }
-
-#else
- ViETest::Log(" ViEExternalCodec not enabled\n");
-#endif
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_custom_call.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_custom_call.cc
deleted file mode 100644
index fa8bf76..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_custom_call.cc
+++ /dev/null
@@ -1,2058 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * vie_autotest_custom_call.cc
- *
- */
-
-#include <iostream>
-
-#include "vie_autotest.h"
-#include "vie_autotest_defines.h"
-
-#define VCM_RED_PAYLOAD_TYPE 96
-#define VCM_ULPFEC_PAYLOAD_TYPE 97
-#define DEFAULT_SEND_IP "127.0.0.1"
-#define DEFAULT_VIDEO_PORT 11111
-#define DEFAULT_VIDEO_CODEC "vp8"
-#define DEFAULT_VIDEO_CODEC_WIDTH 640
-#define DEFAULT_VIDEO_CODEC_HEIGHT 480
-#define DEFAULT_VIDEO_CODEC_BITRATE 300
-#define DEFAULT_VIDEO_CODEC_MIN_BITRATE 100
-#define DEFAULT_VIDEO_CODEC_MAX_BITRATE 1000
-#define DEFAULT_AUDIO_PORT 11113
-#define DEFAULT_AUDIO_CODEC "ISAC"
-#define DEFAULT_INCOMING_FILE_NAME "IncomingFile.avi"
-#define DEFAULT_OUTGOING_FILE_NAME "OutgoingFile.avi"
-#define DEFAULT_VIDEO_CODEC_MAX_FRAMERATE 30
-#define DEFAULT_VIDEO_PROTECTION_METHOD 0
-#define DEFAULT_TEMPORAL_LAYER 0
-
-enum StatisticsType {
- kSendStatistic,
- kReceivedStatistic
-};
-
-class ViEAutotestFileObserver: public webrtc::ViEFileObserver {
- public:
- ViEAutotestFileObserver() {};
- ~ViEAutotestFileObserver() {};
-
- void PlayFileEnded(const WebRtc_Word32 fileId) {
- ViETest::Log("PlayFile ended");
- }
-};
-
-class ViEAutotestEncoderObserver: public webrtc::ViEEncoderObserver {
- public:
- ViEAutotestEncoderObserver() {};
- ~ViEAutotestEncoderObserver() {};
-
- void OutgoingRate(const int videoChannel,
- const unsigned int framerate,
- const unsigned int bitrate) {
- std::cout << "Send FR: " << framerate
- << " BR: " << bitrate << std::endl;
- }
-};
-
-class ViEAutotestDecoderObserver: public webrtc::ViEDecoderObserver {
- public:
- ViEAutotestDecoderObserver() {};
- ~ViEAutotestDecoderObserver() {};
-
- void IncomingRate(const int videoChannel,
- const unsigned int framerate,
- const unsigned int bitrate) {
- std::cout << "Received FR: " << framerate
- << " BR: " << bitrate << std::endl;
- }
- void IncomingCodecChanged(const int videoChannel,
- const webrtc::VideoCodec& codec) {}
- void RequestNewKeyFrame(const int videoChannel) {
- std::cout << "Decoder requesting a new key frame." << std::endl;
- }
-};
-
-// The following are general helper functions.
-bool GetVideoDevice(webrtc::ViEBase* ptrViEBase,
- webrtc::ViECapture* ptrViECapture,
- char* captureDeviceName, char* captureDeviceUniqueId);
-bool GetIPAddress(char* IP);
-#ifndef WEBRTC_ANDROID
-bool ValidateIP(std::string iStr);
-#endif
-// The following are Print to stdout functions.
-void PrintCallInformation(char* IP, char* videoCaptureDeviceName,
- char* videoCaptureUniqueId,
- webrtc::VideoCodec videoCodec, int videoTxPort,
- int videoRxPort, char* audioCaptureDeviceName,
- char* audioPlaybackDeviceName,
- webrtc::CodecInst audioCodec, int audioTxPort,
- int audioRxPort, int protectionMethod);
-void PrintRTCCPStatistics(webrtc::ViERTP_RTCP* ptrViERtpRtcp,
- int videoChannel, StatisticsType statType);
-void PrintRTPStatistics(webrtc::ViERTP_RTCP* ptrViERtpRtcp,
- int videoChannel);
-void PrintBandwidthUsage(webrtc::ViERTP_RTCP* ptrViERtpRtcp,
- int videoChannel);
-void PrintCodecStatistics(webrtc::ViECodec* ptrViECodec, int videoChannel,
- StatisticsType statType);
-void PrintGetDiscardedPackets(webrtc::ViECodec* ptrViECodec, int videoChannel);
-void PrintVideoStreamInformation(webrtc::ViECodec* ptrViECodec,
- int videoChannel);
-void PrintVideoCodec(webrtc::VideoCodec videoCodec);
-
-// The following are video functions.
-// TODO(amyfong): change to pointers as input arguments
-// instead of references
-bool SetVideoPorts(int* txPort, int* rxPort);
-bool SetVideoCodecType(webrtc::ViECodec* ptrViECodec,
- webrtc::VideoCodec& videoCodec);
-bool SetVideoCodecResolution(webrtc::ViECodec* ptrViECodec,
- webrtc::VideoCodec& videoCodec);
-bool SetVideoCodecSize(webrtc::ViECodec* ptrViECodec,
- webrtc::VideoCodec& videoCodec);
-bool SetVideoCodecBitrate(webrtc::ViECodec* ptrViECodec,
- webrtc::VideoCodec& videoCodec);
-bool SetVideoCodecMinBitrate(webrtc::ViECodec* ptrViECodec,
- webrtc::VideoCodec& videoCodec);
-bool SetVideoCodecMaxBitrate(webrtc::ViECodec* ptrViECodec,
- webrtc::VideoCodec& videoCodec);
-bool SetVideoCodecMaxFramerate(webrtc::ViECodec* ptrViECodec,
- webrtc::VideoCodec& videoCodec);
-bool SetVideoCodecTemporalLayer(webrtc::VideoCodec& videoCodec);
-int GetVideoProtection();
-bool SetVideoProtection(webrtc::ViECodec* ptrViECodec,
- webrtc::ViERTP_RTCP* ptrViERtpRtcp,
- int videoChannel, int protectionMethod);
-bool GetBitrateSignaling();
-
-// The following are audio helper functions.
-bool GetAudioDevices(webrtc::VoEBase* ptrVEBase,
- webrtc::VoEHardware* ptrVEHardware,
- char* recordingDeviceName, int& recordingDeviceIndex,
- char* playbackDeviceName, int& playbackDeviceIndex);
-bool GetAudioDevices(webrtc::VoEBase* ptrVEBase,
- webrtc::VoEHardware* ptrVEHardware,
- int& recordingDeviceIndex, int& playbackDeviceIndex);
-bool GetAudioPorts(int* txPort, int* rxPort);
-bool GetAudioCodec(webrtc::VoECodec* ptrVeCodec,
- webrtc::CodecInst& audioCodec);
-
-int ViEAutoTest::ViECustomCall()
-{
- ViETest::Log(" ");
- ViETest::Log("========================================");
- ViETest::Log(" Enter values to use custom settings\n");
-
- int error = 0;
- int numberOfErrors = 0;
- std::string str;
-
- // Create the VoE and get the VoE interfaces.
- webrtc::VoiceEngine* ptrVE = webrtc::VoiceEngine::Create();
- numberOfErrors += ViETest::TestError(ptrVE != NULL, "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- webrtc::VoEBase* ptrVEBase = webrtc::VoEBase::GetInterface(ptrVE);
- numberOfErrors += ViETest::TestError(ptrVEBase != NULL,
- "ERROR: %s at line %d", __FUNCTION__,
- __LINE__);
-
- error = ptrVEBase->Init();
- numberOfErrors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- webrtc::VoECodec* ptrVECodec = webrtc::VoECodec::GetInterface(ptrVE);
- numberOfErrors += ViETest::TestError(ptrVECodec != NULL,
- "ERROR: %s at line %d", __FUNCTION__,
- __LINE__);
-
- webrtc::VoEHardware* ptrVEHardware =
- webrtc::VoEHardware::GetInterface(ptrVE);
- numberOfErrors += ViETest::TestError(ptrVEHardware != NULL,
- "ERROR: %s at line %d", __FUNCTION__,
- __LINE__);
-
- webrtc::VoEAudioProcessing* ptrVEAPM =
- webrtc::VoEAudioProcessing::GetInterface(ptrVE);
- numberOfErrors += ViETest::TestError(ptrVEAPM != NULL,
- "ERROR: %s at line %d", __FUNCTION__,
- __LINE__);
-
- // Create the ViE and get the ViE Interfaces.
- webrtc::VideoEngine* ptrViE = webrtc::VideoEngine::Create();
- numberOfErrors += ViETest::TestError(ptrViE != NULL,
- "ERROR: %s at line %d", __FUNCTION__,
- __LINE__);
-
- webrtc::ViEBase* ptrViEBase = webrtc::ViEBase::GetInterface(ptrViE);
- numberOfErrors += ViETest::TestError(ptrViEBase != NULL,
- "ERROR: %s at line %d", __FUNCTION__,
- __LINE__);
-
- error = ptrViEBase->Init();
- numberOfErrors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- webrtc::ViECapture* ptrViECapture =
- webrtc::ViECapture::GetInterface(ptrViE);
- numberOfErrors += ViETest::TestError(ptrViECapture != NULL,
- "ERROR: %s at line %d", __FUNCTION__,
- __LINE__);
-
- webrtc::ViERender* ptrViERender = webrtc::ViERender::GetInterface(ptrViE);
- numberOfErrors += ViETest::TestError(ptrViERender != NULL,
- "ERROR: %s at line %d", __FUNCTION__,
- __LINE__);
-
- webrtc::ViECodec* ptrViECodec = webrtc::ViECodec::GetInterface(ptrViE);
- numberOfErrors += ViETest::TestError(ptrViECodec != NULL,
- "ERROR: %s at line %d", __FUNCTION__,
- __LINE__);
-
- webrtc::ViENetwork* ptrViENetwork =
- webrtc::ViENetwork::GetInterface(ptrViE);
- numberOfErrors += ViETest::TestError(ptrViENetwork != NULL,
- "ERROR: %s at line %d", __FUNCTION__,
- __LINE__);
-
- webrtc::ViEFile* ptrViEFile = webrtc::ViEFile::GetInterface(ptrViE);
- numberOfErrors += ViETest::TestError(ptrViEFile != NULL,
- "ERROR: %s at line %d", __FUNCTION__,
- __LINE__);
-
- bool startCall = false;
- const unsigned int kMaxIPLength = 16;
- char ipAddress[kMaxIPLength] = "";
- const unsigned int KMaxUniqueIdLength = 256;
- char uniqueId[KMaxUniqueIdLength] = "";
- char deviceName[KMaxUniqueIdLength] = "";
- int videoTxPort = 0;
- int videoRxPort = 0;
- int videoChannel = -1;
- webrtc::VideoCodec videoSendCodec;
- char audioCaptureDeviceName[KMaxUniqueIdLength] = "";
- char audioPlaybackDeviceName[KMaxUniqueIdLength] = "";
- int audioCaptureDeviceIndex = -1;
- int audioPlaybackDeviceIndex = -1;
- int audioTxPort = 0;
- int audioRxPort = 0;
- webrtc::CodecInst audioCodec;
- int audioChannel = -1;
- bool isImageScaleEnabled = false;
- int protectionMethod = DEFAULT_VIDEO_PROTECTION_METHOD;
- bool remb = true;
-
- while (!startCall) {
- // Get the IP address to use from call.
- memset(ipAddress, 0, kMaxIPLength);
- GetIPAddress(ipAddress);
-
- // Get the video device to use for call.
- memset(deviceName, 0, KMaxUniqueIdLength);
- memset(uniqueId, 0, KMaxUniqueIdLength);
- GetVideoDevice(ptrViEBase, ptrViECapture, deviceName, uniqueId);
-
- // Get and set the video ports for the call.
- videoTxPort = 0;
- videoRxPort = 0;
- SetVideoPorts(&videoTxPort, &videoRxPort);
-
- // Get and set the video codec parameters for the call.
- memset((void*)&videoSendCodec, 0, sizeof(webrtc::VideoCodec));
- SetVideoCodecType(ptrViECodec, videoSendCodec);
- SetVideoCodecSize(ptrViECodec, videoSendCodec);
- SetVideoCodecBitrate(ptrViECodec, videoSendCodec);
- SetVideoCodecMinBitrate(ptrViECodec, videoSendCodec);
- SetVideoCodecMaxBitrate(ptrViECodec, videoSendCodec);
- SetVideoCodecMaxFramerate(ptrViECodec, videoSendCodec);
- SetVideoCodecTemporalLayer(videoSendCodec);
- remb = GetBitrateSignaling();
-
- // Get the video protection method for the call.
- protectionMethod = GetVideoProtection();
-
- // Get the audio device for the call.
- memset(audioCaptureDeviceName, 0, KMaxUniqueIdLength);
- memset(audioPlaybackDeviceName, 0, KMaxUniqueIdLength);
- GetAudioDevices(ptrVEBase, ptrVEHardware, audioCaptureDeviceName,
- audioCaptureDeviceIndex, audioPlaybackDeviceName,
- audioPlaybackDeviceIndex);
-
- // Get the audio port for the call.
- audioTxPort = 0;
- audioRxPort = 0;
- GetAudioPorts(&audioTxPort, &audioRxPort);
-
- // Get the audio codec for the call.
- memset((void*)&audioCodec, 0, sizeof(audioCodec));
- GetAudioCodec(ptrVECodec, audioCodec);
-
- // Now ready to start the call. Check user wants to continue.
- PrintCallInformation(ipAddress, deviceName, uniqueId, videoSendCodec,
- videoTxPort, videoRxPort, audioCaptureDeviceName,
- audioPlaybackDeviceName, audioCodec, audioTxPort,
- audioRxPort, protectionMethod);
-
- std::cout << std::endl;
- std::cout << "1. Start the call" << std::endl;
- std::cout << "2. Reconfigure call settings" << std::endl;
- std::cout << "What do you want to do? Press enter for default "
- << "(Start the call): ";
-
- std::getline(std::cin, str);
- int selection = 0;
- selection = atoi(str.c_str());
-
- switch (selection) {
- case 0:
- startCall = true;
- break;
- case 1:
- startCall = true;
- break;
- case 2:
- startCall = false;
- break;
- default:
- // Invalid selection gets error mesage.
- std::cout << "ERROR: Code=" << error
- << " Invalid selection" << std::endl;
- continue;
- }
- }
- //***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing.
- //***************************************************************
- if (startCall == true) {
- // Configure audio channel first.
- audioChannel = ptrVEBase->CreateChannel();
- error = ptrVEBase->SetSendDestination(audioChannel, audioTxPort,
- ipAddress);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrVEBase->SetLocalReceiver(audioChannel, audioRxPort);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrVEHardware->SetRecordingDevice(audioCaptureDeviceIndex);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrVEHardware->SetPlayoutDevice(audioPlaybackDeviceIndex);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrVECodec->SetSendCodec(audioChannel, audioCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrVEAPM->SetAgcStatus(true, webrtc::kAgcDefault);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrVEAPM->SetNsStatus(true, webrtc::kNsHighSuppression);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- // Now configure the video channel.
- error = ptrViE->SetTraceFilter(webrtc::kTraceAll);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- std::string trace_file =
- ViETest::GetResultOutputPath() + "ViECustomCall_trace.txt";
- error = ptrViE->SetTraceFile(trace_file.c_str());
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViEBase->SetVoiceEngine(ptrVE);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViEBase->CreateChannel(videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViEBase->ConnectAudioChannel(videoChannel, audioChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- int captureId = 0;
- error = ptrViECapture->AllocateCaptureDevice(uniqueId,
- KMaxUniqueIdLength,
- captureId);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViECapture->ConnectCaptureDevice(captureId, videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViECapture->StartCapture(captureId);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- webrtc::ViERTP_RTCP* ptrViERtpRtcp =
- webrtc::ViERTP_RTCP::GetInterface(ptrViE);
- numberOfErrors += ViETest::TestError(ptrViE != NULL,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViERtpRtcp->SetRTCPStatus(videoChannel,
- webrtc::kRtcpCompound_RFC4585);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViERtpRtcp->SetKeyFrameRequestMethod(videoChannel,
- webrtc::kViEKeyFrameRequestPliRtcp);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- if (remb) {
- error = ptrViERtpRtcp->SetRembStatus(videoChannel, true, true);
- numberOfErrors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- } else {
- error = ptrViERtpRtcp->SetTMMBRStatus(videoChannel, true);
- numberOfErrors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- }
-
- error = ptrViERender->AddRenderer(captureId, _window1, 0, 0.0, 0.0,
- 1.0, 1.0);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViERender->AddRenderer(videoChannel, _window2, 1, 0.0, 0.0,
- 1.0, 1.0);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- error = ptrViENetwork->SetSendDestination(videoChannel, ipAddress,
- videoTxPort);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViENetwork->SetLocalReceiver(videoChannel, videoRxPort);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViECodec->SetSendCodec(videoChannel, videoSendCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViECodec->SetReceiveCodec(videoChannel, videoSendCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- // Set the Video Protection before start send and receive.
- SetVideoProtection(ptrViECodec, ptrViERtpRtcp,
- videoChannel, protectionMethod);
-
- // Start Voice Playout and Receive.
- error = ptrVEBase->StartReceive(audioChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrVEBase->StartPlayout(audioChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrVEBase->StartSend(audioChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- // Now start the Video Send & Receive.
- error = ptrViEBase->StartSend(videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViEBase->StartReceive(videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViERender->StartRender(captureId);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViERender->StartRender(videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- ViEAutotestFileObserver fileObserver;
- int fileId;
-
- ViEAutotestEncoderObserver* codecEncoderObserver = NULL;
- ViEAutotestDecoderObserver* codecDecoderObserver = NULL;
-
- //***************************************************************
- // Engine ready. Wait for input
- //***************************************************************
-
- // Call started.
- std::cout << std::endl;
- std::cout << "Custom call started" << std::endl;
- std::cout << std::endl << std::endl;
-
- // Modify call or stop call.
-
- std::cout << "Custom call in progress, would you like do?" << std::endl;
- std::cout << " 0. Stop the call" << std::endl;
- std::cout << " 1. Modify the call" << std::endl;
- std::cout << "What do you want to do? "
- << "Press enter for default (Stop the call): ";
-
- std::getline(std::cin, str);
- int selection = 0;
- selection = atoi(str.c_str());
-
- // Keep on modifying the call until user selects finish modify call.
- bool modify_call = false;
-
- while (selection == 1) {
- std::cout << "Modify Custom Call" << std::endl;
- std::cout << " 0. Finished modifying custom call" << std::endl;
- std::cout << " 1. Change Video Send Codec" << std::endl;
- std::cout << " 2. Change Video Send Size by Common Resolutions"
- << std::endl;
- std::cout << " 3. Change Video Send Size by Width & Height" << std::endl;
- std::cout << " 4. Change Video Capture Device" << std::endl;
- std::cout << " 5. Record Incoming Call" << std::endl;
- std::cout << " 6. Record Outgoing Call" << std::endl;
- std::cout << " 7. Play File on Video Channel"
- << "(Assumes you recorded incoming & outgoing call)"
- << std::endl;
- std::cout << " 8. Change Video Protection Method" << std::endl;
- std::cout << " 9. Toggle Encoder Observer" << std::endl;
- std::cout << " 10. Toggle Decoder Observer" << std::endl;
- std::cout << " 11. Print Call Information" << std::endl;
- std::cout << " 12. Print Call Statistics" << std::endl;
- std::cout << " 13. Toggle Image Scaling "
- << "(Warning high CPU usage when enabled)"
- << std::endl;
- std::cout << "What do you want to do? ";
- std::cout << "Press enter for default "
- << "(Finished modifying custom call): ";
-
- std::getline(std::cin, str);
- int modify_selection = 0;
- int file_selection = 0;
-
- modify_selection = atoi(str.c_str());
-
- switch (modify_selection) {
- case 0:
- std::cout << "Finished modifying custom call." << std::endl;
- modify_call = false;
- break;
- case 1:
- // Change video Codec
- SetVideoCodecType(ptrViECodec, videoSendCodec);
- SetVideoCodecSize(ptrViECodec, videoSendCodec);
- SetVideoCodecBitrate(ptrViECodec, videoSendCodec);
- SetVideoCodecMinBitrate(ptrViECodec, videoSendCodec);
- SetVideoCodecMaxBitrate(ptrViECodec, videoSendCodec);
- SetVideoCodecMaxFramerate(ptrViECodec, videoSendCodec);
- SetVideoCodecTemporalLayer(videoSendCodec);
- PrintCallInformation(ipAddress, deviceName,
- uniqueId, videoSendCodec,
- videoTxPort, videoRxPort,
- audioCaptureDeviceName,
- audioPlaybackDeviceName, audioCodec,
- audioTxPort, audioRxPort, protectionMethod);
- error = ptrViECodec->SetSendCodec(videoChannel, videoSendCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- error = ptrViECodec->SetReceiveCodec(videoChannel, videoSendCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- modify_call = true;
- break;
- case 2:
- // Change Video codec size by common resolution
- SetVideoCodecResolution(ptrViECodec, videoSendCodec);
- PrintCallInformation(ipAddress, deviceName,
- uniqueId, videoSendCodec,
- videoTxPort, videoRxPort,
- audioCaptureDeviceName,
- audioPlaybackDeviceName, audioCodec,
- audioTxPort, audioRxPort, protectionMethod);
- error = ptrViECodec->SetSendCodec(videoChannel, videoSendCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- error = ptrViECodec->SetReceiveCodec(videoChannel, videoSendCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- modify_call = true;
- break;
- case 3:
- // Change Video codec by size height and width
- SetVideoCodecSize(ptrViECodec, videoSendCodec);
- PrintCallInformation(ipAddress, deviceName,
- uniqueId, videoSendCodec,
- videoTxPort, videoRxPort,
- audioCaptureDeviceName,
- audioPlaybackDeviceName, audioCodec,
- audioTxPort, audioRxPort, protectionMethod);
- error = ptrViECodec->SetSendCodec(videoChannel, videoSendCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- error = ptrViECodec->SetReceiveCodec(videoChannel, videoSendCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- modify_call = true;
- break;
- case 4:
- error = ptrViERender->StopRender(captureId);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- error = ptrViERender->RemoveRenderer(captureId);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- error = ptrViECapture->StopCapture(captureId);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- error = ptrViECapture->DisconnectCaptureDevice(videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- error = ptrViECapture->ReleaseCaptureDevice(captureId);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- memset(deviceName, 0, KMaxUniqueIdLength);
- memset(uniqueId, 0, KMaxUniqueIdLength);
- GetVideoDevice(ptrViEBase, ptrViECapture, deviceName, uniqueId);
- captureId = 0;
- error = ptrViECapture->AllocateCaptureDevice(uniqueId,
- KMaxUniqueIdLength,
- captureId);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- error = ptrViECapture->ConnectCaptureDevice(captureId,
- videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- error = ptrViECapture->StartCapture(captureId);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- error = ptrViERender->AddRenderer(
- captureId, _window1, 0, 0.0, 0.0, 1.0, 1.0);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- error = ptrViERender->StartRender(captureId);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- modify_call = true;
- break;
- case 5:
- // Record the incoming call
- std::cout << "Start Recording Incoming Video "
- << DEFAULT_INCOMING_FILE_NAME << std::endl;
- error = ptrViEFile->StartRecordIncomingVideo(
- videoChannel, DEFAULT_INCOMING_FILE_NAME,
- webrtc::NO_AUDIO, audioCodec, videoSendCodec);
- std::cout << "Press enter to stop...";
- std::getline(std::cin, str);
- error = ptrViEFile->StopRecordIncomingVideo(videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR:%d %s at line %d",
- ptrViEBase->LastError(),
- __FUNCTION__, __LINE__);
- modify_call = true;
- break;
- case 6:
- // Record the outgoing call
- std::cout << "Start Recording Outgoing Video "
- << DEFAULT_OUTGOING_FILE_NAME << std::endl;
- error = ptrViEFile->StartRecordOutgoingVideo(
- videoChannel, DEFAULT_OUTGOING_FILE_NAME,
- webrtc::NO_AUDIO, audioCodec, videoSendCodec);
- std::cout << "Press enter to stop...";
- std::getline(std::cin, str);
- error = ptrViEFile->StopRecordOutgoingVideo(videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR:%d %s at line %d",
- ptrViEBase->LastError(),
- __FUNCTION__, __LINE__);
- modify_call = true;
- break;
- case 7:
- // Send the file on the videoChannel
- file_selection = 0;
- std::cout << "Available files to play" << std::endl;
- std::cout << " 0. " << DEFAULT_INCOMING_FILE_NAME << std::endl;
- std::cout << " 1. " << DEFAULT_OUTGOING_FILE_NAME << std::endl;
- std::cout << "Press enter for default ("
- << DEFAULT_INCOMING_FILE_NAME << "): ";
- std::getline(std::cin, str);
- file_selection = atoi(str.c_str());
- // Disconnect the camera first
- error = ptrViECapture->DisconnectCaptureDevice(videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR:%d %s at line %d",
- ptrViEBase->LastError(),
- __FUNCTION__, __LINE__);
- if (file_selection == 1)
- error = ptrViEFile->StartPlayFile(DEFAULT_OUTGOING_FILE_NAME,
- fileId, true);
- else
- error = ptrViEFile->StartPlayFile(DEFAULT_INCOMING_FILE_NAME,
- fileId, true);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR:%d %s at line %d",
- ptrViEBase->LastError(),
- __FUNCTION__, __LINE__);
- ViETest::Log("Registering file observer");
- error = ptrViEFile->RegisterObserver(fileId, fileObserver);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR:%d %s at line %d",
- ptrViEBase->LastError(),
- __FUNCTION__, __LINE__);
- std::cout << std::endl;
- std::cout << "Start sending the file that is played in a loop "
- << std::endl;
- error = ptrViEFile->SendFileOnChannel(fileId, videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR:%d %s at line %d",
- ptrViEBase->LastError(),
- __FUNCTION__, __LINE__);
- std::cout << "Press enter to stop...";
- std::getline(std::cin, str);
- ViETest::Log("Stopped sending video on channel");
- error = ptrViEFile->StopSendFileOnChannel(videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR:%d %s at line %d",
- ptrViEBase->LastError(),
- __FUNCTION__, __LINE__);
- ViETest::Log("Stop playing the file.");
- error = ptrViEFile->StopPlayFile(fileId);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR:%d %s at line %d",
- ptrViEBase->LastError(),
- __FUNCTION__, __LINE__);
- error = ptrViECapture->ConnectCaptureDevice(captureId,
- videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR:%d %s at line %d",
- ptrViEBase->LastError(),
- __FUNCTION__, __LINE__);
- error = ptrViEFile->DeregisterObserver(fileId, fileObserver);
- numberOfErrors += ViETest::TestError(error == -1,
- "ERROR:%d %s at line %d",
- ptrViEBase->LastError(),
- __FUNCTION__, __LINE__);
- modify_call = true;
- break;
- case 8:
- // Change the Video Protection
- protectionMethod = GetVideoProtection();
- SetVideoProtection(ptrViECodec, ptrViERtpRtcp,
- videoChannel, protectionMethod);
- modify_call = true;
- break;
- case 9:
- // Toggle Encoder Observer
- if (!codecEncoderObserver) {
- std::cout << "Registering Encoder Observer" << std::endl;
- codecEncoderObserver = new ViEAutotestEncoderObserver();
- error = ptrViECodec->RegisterEncoderObserver(videoChannel,
- *codecEncoderObserver);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- } else {
- std::cout << "Deregistering Encoder Observer" << std::endl;
- error = ptrViECodec->DeregisterEncoderObserver(videoChannel);
- delete codecEncoderObserver;
- codecEncoderObserver = NULL;
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- }
- modify_call = true;
- break;
- case 10:
- // Toggle Decoder Observer
- if (!codecDecoderObserver) {
- std::cout << "Registering Decoder Observer" << std::endl;
- codecDecoderObserver = new ViEAutotestDecoderObserver();
- error = ptrViECodec->RegisterDecoderObserver(videoChannel,
- *codecDecoderObserver);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- } else {
- std::cout << "Deregistering Decoder Observer" << std::endl;
- error = ptrViECodec->DeregisterDecoderObserver(videoChannel);
- delete codecDecoderObserver;
- codecDecoderObserver = NULL;
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- }
- modify_call = true;
- break;
- case 11:
- // Print Call information
- PrintCallInformation(ipAddress, deviceName,
- uniqueId, videoSendCodec,
- videoTxPort, videoRxPort,
- audioCaptureDeviceName,
- audioPlaybackDeviceName,
- audioCodec, audioTxPort,
- audioRxPort, protectionMethod);
- PrintVideoStreamInformation(ptrViECodec,
- videoChannel);
- modify_call = true;
- break;
- case 12:
- // Print Call statistics
- PrintRTCCPStatistics(ptrViERtpRtcp, videoChannel,
- kSendStatistic);
- PrintRTCCPStatistics(ptrViERtpRtcp, videoChannel,
- kReceivedStatistic);
- PrintRTPStatistics(ptrViERtpRtcp, videoChannel);
- PrintBandwidthUsage(ptrViERtpRtcp, videoChannel);
- PrintCodecStatistics(ptrViECodec, videoChannel,
- kSendStatistic);
- PrintCodecStatistics(ptrViECodec, videoChannel,
- kReceivedStatistic);
- PrintGetDiscardedPackets(ptrViECodec, videoChannel);
- modify_call = true;
- break;
- case 13:
- isImageScaleEnabled = !isImageScaleEnabled;
- ptrViECodec->SetImageScaleStatus(videoChannel, isImageScaleEnabled);
- if (isImageScaleEnabled) {
- std::cout << "Image Scale is now enabled" << std::endl;
- } else {
- std::cout << "Image Scale is now disabled" << std::endl;
- }
- modify_call = true;
- break;
- default:
- // invalid selection, shows options menu again
- std::cout << "Invalid selection. Select Again." << std::endl;
- break;
- }
- // modify_call is false if user does not select one of the
- // modify options
- if (modify_call == false) {
- selection = 0;
- }
- }
- // Stop the Call
- std::cout << "Press enter to stop...";
- std::getline(std::cin, str);
- //***************************************************************
- // Testing finished. Tear down Voice and Video Engine.
- //***************************************************************
- // Tear down the VE first.
- error = ptrVEBase->StopReceive(audioChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrVEBase->StopPlayout(audioChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrVEBase->DeleteChannel(audioChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- // Now tear down the ViE engine.
- error = ptrViEBase->DisconnectAudioChannel(videoChannel);
-
- // If Encoder/Decoder Observer is running, delete them.
- if (codecEncoderObserver) {
- error = ptrViECodec->DeregisterEncoderObserver(videoChannel);
- delete codecEncoderObserver;
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- }
- if (codecDecoderObserver) {
- error = ptrViECodec->DeregisterDecoderObserver(videoChannel);
- delete codecDecoderObserver;
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- }
-
- error = ptrViEBase->StopReceive(videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViEBase->StopSend(videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViERender->StopRender(captureId);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViERender->StopRender(videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViERender->RemoveRenderer(captureId);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViERender->RemoveRenderer(videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViECapture->StopCapture(captureId);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViECapture->DisconnectCaptureDevice(videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViECapture->ReleaseCaptureDevice(captureId);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- error = ptrViEBase->DeleteChannel(videoChannel);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- int remainingInterfaces = 0;
- remainingInterfaces = ptrViEFile->Release();
- numberOfErrors += ViETest::TestError(remainingInterfaces == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- remainingInterfaces = ptrViECodec->Release();
- numberOfErrors += ViETest::TestError(remainingInterfaces == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- remainingInterfaces = ptrViECapture->Release();
- numberOfErrors += ViETest::TestError(remainingInterfaces == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- remainingInterfaces = ptrViERtpRtcp->Release();
- numberOfErrors += ViETest::TestError(remainingInterfaces == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- remainingInterfaces = ptrViERender->Release();
- numberOfErrors += ViETest::TestError(remainingInterfaces == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- remainingInterfaces = ptrViENetwork->Release();
- numberOfErrors += ViETest::TestError(remainingInterfaces == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- remainingInterfaces = ptrViEBase->Release();
- numberOfErrors += ViETest::TestError(remainingInterfaces == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- bool deleted = webrtc::VideoEngine::Delete(ptrViE);
- numberOfErrors += ViETest::TestError(deleted == true,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- ViETest::Log(" ");
- ViETest::Log(" ViE Autotest Custom Call Started");
- ViETest::Log("========================================");
- ViETest::Log(" ");
- }
- return numberOfErrors;
-}
-
-bool GetVideoDevice(webrtc::ViEBase* ptrViEBase,
- webrtc::ViECapture* ptrViECapture,
- char* captureDeviceName,
- char* captureDeviceUniqueId) {
- int error = 0;
- int numberOfErrors = 0;
- int captureDeviceIndex = 0;
- std::string str;
-
- const unsigned int KMaxDeviceNameLength = 128;
- const unsigned int KMaxUniqueIdLength = 256;
- char deviceName[KMaxDeviceNameLength];
- char uniqueId[KMaxUniqueIdLength];
-
- while (1) {
- memset(deviceName, 0, KMaxDeviceNameLength);
- memset(uniqueId, 0, KMaxUniqueIdLength);
-
- std::cout << std::endl;
- std::cout << "Available video capture devices:" << std::endl;
- int captureIdx = 0;
- for (captureIdx = 0;
- captureIdx < ptrViECapture->NumberOfCaptureDevices();
- captureIdx++) {
- memset(deviceName, 0, KMaxDeviceNameLength);
- memset(uniqueId, 0, KMaxUniqueIdLength);
-
- error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
- KMaxDeviceNameLength,
- uniqueId,
- KMaxUniqueIdLength);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- std::cout << " " << captureIdx+1 << ". " << deviceName
- << "/" << uniqueId
- << std::endl;
- }
- // Get the devName of the default (or first) camera for display.
- error = ptrViECapture->GetCaptureDevice(0, deviceName,
- KMaxDeviceNameLength,
- uniqueId,
- KMaxUniqueIdLength);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- std::cout << "Choose a video capture device. Press enter for default ("
- << deviceName << "/" << uniqueId << "): ";
- std::getline(std::cin, str);
- captureDeviceIndex = atoi(str.c_str());
-
- if (captureDeviceIndex == 0) {
- // Use the default (or first) camera.
- error = ptrViECapture->GetCaptureDevice(0, deviceName,
- KMaxDeviceNameLength,
- uniqueId,
- KMaxUniqueIdLength);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- strcpy(captureDeviceUniqueId, uniqueId);
- strcpy(captureDeviceName, deviceName);
- return true;
- } else if (captureDeviceIndex < 0
- || (captureDeviceIndex >
- (int)ptrViECapture->NumberOfCaptureDevices())) {
- // invalid selection
- continue;
- } else {
- error = ptrViECapture->GetCaptureDevice(captureDeviceIndex - 1,
- deviceName,
- KMaxDeviceNameLength,
- uniqueId,
- KMaxUniqueIdLength);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- strcpy(captureDeviceUniqueId, uniqueId);
- strcpy(captureDeviceName, deviceName);
- return true;
- }
- }
-}
-
-bool GetAudioDevices(webrtc::VoEBase* ptrVEBase,
- webrtc::VoEHardware* ptrVEHardware,
- char* recordingDeviceName,
- int& recordingDeviceIndex,
- char* playbackDeviceName,
- int& playbackDeviceIndex) {
- int error = 0;
- int numberOfErrors = 0;
- std::string str;
-
- const unsigned int KMaxDeviceNameLength = 128;
- const unsigned int KMaxUniqueIdLength = 128;
- char recordingDeviceUniqueName[KMaxDeviceNameLength];
- char playbackDeviceUniqueName[KMaxUniqueIdLength];
-
- int numberOfRecordingDevices = -1;
- error = ptrVEHardware->GetNumOfRecordingDevices(numberOfRecordingDevices);
- numberOfErrors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- while (1) {
- recordingDeviceIndex = -1;
- std::cout << std::endl;
- std::cout << "Available audio capture devices:" << std::endl;
- int captureIdx = 0;
-
- for (captureIdx = 0; captureIdx < numberOfRecordingDevices;
- captureIdx++) {
- memset(recordingDeviceName, 0, KMaxDeviceNameLength);
- memset(recordingDeviceUniqueName, 0, KMaxDeviceNameLength);
- error = ptrVEHardware->GetRecordingDeviceName(
- captureIdx, recordingDeviceName, recordingDeviceUniqueName);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- std::cout << " " << captureIdx+1 << ". " << recordingDeviceName
- << std::endl;
- }
-
- std::cout << "Choose an audio capture device. Press enter for default("
- << recordingDeviceName << "): ";
- std::getline(std::cin, str);
- int captureDeviceIndex = atoi(str.c_str());
-
- if (captureDeviceIndex == 0) {
- // Use the default (or first) recording device.
- recordingDeviceIndex = 0;
- error = ptrVEHardware->GetRecordingDeviceName(
- recordingDeviceIndex, recordingDeviceName,
- recordingDeviceUniqueName);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- break;
- } else if (captureDeviceIndex < 0
- || captureDeviceIndex > numberOfRecordingDevices) {
- // invalid selection
- continue;
- } else {
- recordingDeviceIndex = captureDeviceIndex - 1;
- error = ptrVEHardware->GetRecordingDeviceName(
- recordingDeviceIndex, recordingDeviceName,
- recordingDeviceUniqueName);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- break;
- }
- }
-
- int numberOfPlaybackDevices = -1;
- error = ptrVEHardware->GetNumOfPlayoutDevices(numberOfPlaybackDevices);
- numberOfErrors += ViETest::TestError(error == 0, "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- while (1) {
- playbackDeviceIndex = -1;
- std::cout << std::endl;
- std::cout << "Available audio playout devices:" << std::endl;
- int captureIdx = 0;
-
- for (captureIdx = 0; captureIdx < numberOfPlaybackDevices;
- captureIdx++) {
- memset(playbackDeviceName, 0, KMaxDeviceNameLength);
- memset(playbackDeviceUniqueName, 0, KMaxDeviceNameLength);
- error = ptrVEHardware->GetPlayoutDeviceName(
- captureIdx, playbackDeviceName, playbackDeviceUniqueName);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- std::cout << " " << captureIdx+1 << ". " << playbackDeviceName
- << std::endl;
- }
-
- std::cout <<
- "Choose an audio playback device. Press enter for default ("
- << playbackDeviceName << "): ";
- std::getline(std::cin, str);
- int captureDeviceIndex = atoi(str.c_str());
-
- if (captureDeviceIndex == 0) {
- // Use the default (or first) playout device.
- playbackDeviceIndex = 0;
- error = ptrVEHardware->GetPlayoutDeviceName(
- playbackDeviceIndex, playbackDeviceName,
- playbackDeviceUniqueName);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- return true;
- } else if (captureDeviceIndex < 0
- || captureDeviceIndex > numberOfPlaybackDevices) {
- // invalid selection
- continue;
- } else {
- playbackDeviceIndex = captureDeviceIndex - 1;
- error = ptrVEHardware->GetPlayoutDeviceName(
- playbackDeviceIndex, playbackDeviceName,
- playbackDeviceUniqueName);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- return true;
- }
- }
-}
-
-// General helper functions.
-
-bool GetIPAddress(char* iIP) {
- char oIP[16] = DEFAULT_SEND_IP;
- std::string str;
-
- while (1) {
- std::cout << std::endl;
- std::cout << "Enter destination IP. Press enter for default ("
- << oIP << "): ";
- std::getline(std::cin, str);
-
- if (str.compare("") == 0) {
- // use default value;
- strcpy(iIP, oIP);
- return true;
- }
- if (ValidateIP(str) == false) {
- std::cout << "Invalid entry. Try again." << std::endl;
- continue;
- }
- // done. Copy std::string to c_string and return
- strcpy(iIP, str.c_str());
- return true;
- }
- assert(false);
- return false;
-}
-
-bool ValidateIP(std::string iStr) {
- if (0 == iStr.compare("")) {
- return false;
- }
- return true;
-}
-
-// Video settings functions.
-
-bool SetVideoPorts(int* txPort, int* rxPort) {
- std::string str;
- int port = 0;
-
- // set to default values
- *txPort = DEFAULT_VIDEO_PORT;
- *rxPort = DEFAULT_VIDEO_PORT;
-
- while (1) {
- std::cout << "Enter video send port. Press enter for default ("
- << *txPort << "): ";
- std::getline(std::cin, str);
- port = atoi(str.c_str());
-
- if (port == 0) {
- // default value
- break;
- } else {
- // user selection
- if (port <= 0 || port > 63556) {
- // invalid selection
- continue;
- } else {
- *txPort = port;
- break; // move on to rxport
- }
- }
- }
-
- while (1) {
- std::cout << "Enter video receive port. Press enter for default ("
- << *rxPort << "): ";
- std::getline(std::cin, str);
- port = atoi(str.c_str());
-
- if (port == 0) {
- // default value
- return true;
- } else {
- // user selection
- if (port <= 0 || port > 63556) {
- // invalid selection
- continue;
- } else {
- *rxPort = port;
- return true;
- }
- }
- }
- assert(false);
- return false;
-}
-
-// Audio settings functions.
-
-bool GetAudioPorts(int* txPort, int* rxPort) {
- int port = 0;
- std::string str;
-
- // set to default values
- *txPort = DEFAULT_AUDIO_PORT;
- *rxPort = DEFAULT_AUDIO_PORT;
-
- while (1) {
- std::cout << "Enter audio send port. Press enter for default ("
- << *txPort << "): ";
- std::getline(std::cin, str);
- port = atoi(str.c_str());
-
- if (port == 0) {
- // default value
- break;
- } else {
- // user selection
- if (port <= 0 || port > 63556) {
- // invalid selection
- continue;
- } else {
- *txPort = port;
- break; // move on to rxport
- }
- }
- }
-
- while (1) {
- std::cout << "Enter audio receive port. Press enter for default ("
- << *rxPort << "): ";
- std::getline(std::cin, str);
- port = atoi(str.c_str());
-
- if (port == 0) {
- // default value
- return true;
- } else {
- // user selection
- if (port <= 0 || port > 63556) {
- // invalid selection
- continue;
- } else {
- *rxPort = port;
- return true;
- }
- }
- }
- assert(false);
- return false;
-}
-
-bool GetAudioCodec(webrtc::VoECodec* ptrVeCodec,
- webrtc::CodecInst& audioCodec) {
- int error = 0;
- int numberOfErrors = 0;
- int codecSelection = 0;
- std::string str;
- memset(&audioCodec, 0, sizeof(webrtc::CodecInst));
-
- while (1) {
- std::cout << std::endl;
- std::cout << "Available audio codecs:" << std::endl;
- int codecIdx = 0;
- int defaultCodecIdx = 0;
- for (codecIdx = 0; codecIdx < ptrVeCodec->NumOfCodecs(); codecIdx++) {
- error = ptrVeCodec->GetCodec(codecIdx, audioCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
-
- // test for default codec index
- if (strcmp(audioCodec.plname, DEFAULT_AUDIO_CODEC) == 0) {
- defaultCodecIdx = codecIdx;
- }
- std::cout << " " << codecIdx+1 << ". " << audioCodec.plname
- << std::endl;
- }
- std::cout << std::endl;
- std::cout << "Choose audio codec. Press enter for default ("
- << DEFAULT_AUDIO_CODEC << "): ";
- std::getline(std::cin, str);
- codecSelection = atoi(str.c_str());
-
- if (codecSelection == 0) {
- // use default
- error = ptrVeCodec->GetCodec(defaultCodecIdx, audioCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- return true;
- } else {
- // user selection
- codecSelection = atoi(str.c_str())-1;
- error = ptrVeCodec->GetCodec(codecSelection, audioCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- if (error != 0) {
- std::cout << "ERROR: Code = " << error << " Invalid selection"
- << std::endl;
- continue;
- }
- return true;
- }
- }
- assert(false);
- return false;
-}
-
-void PrintCallInformation(char* IP, char* videoCaptureDeviceName,
- char* videoCaptureUniqueId,
- webrtc::VideoCodec videoCodec,
- int videoTxPort, int videoRxPort,
- char* audioCaptureDeviceName,
- char* audioPlaybackDeviceName,
- webrtc::CodecInst audioCodec,
- int audioTxPort, int audioRxPort,
- int protectionMethod) {
- std::string str;
-
- std::cout << "************************************************"
- << std::endl;
- std::cout << "The call has the following settings: " << std::endl;
- std::cout << "\tIP: " << IP << std::endl;
- std::cout << "\tVideo Capture Device: " << videoCaptureDeviceName
- << std::endl;
- std::cout << "\t\tName: " << videoCaptureDeviceName << std::endl;
- std::cout << "\t\tUniqueId: " << videoCaptureUniqueId << std::endl;
- PrintVideoCodec(videoCodec);
- std::cout << "\t Video Tx Port: " << videoTxPort << std::endl;
- std::cout << "\t Video Rx Port: " << videoRxPort << std::endl;
- std::cout << "\t Video Protection Method: " << protectionMethod
- << std::endl;
- std::cout << "\tAudio Capture Device: " << audioCaptureDeviceName
- << std::endl;
- std::cout << "\tAudio Playback Device: " << audioPlaybackDeviceName
- << std::endl;
- std::cout << "\tAudio Codec: " << std::endl;
- std::cout << "\t\tplname: " << audioCodec.plname << std::endl;
- std::cout << "\t\tpltype: " << (int)audioCodec.pltype << std::endl;
- std::cout << "\t Audio Tx Port: " << audioTxPort << std::endl;
- std::cout << "\t Audio Rx Port: " << audioRxPort << std::endl;
- std::cout << "************************************************"
- << std::endl;
-}
-
-bool SetVideoCodecType(webrtc::ViECodec* ptrViECodec,
- webrtc::VideoCodec& videoCodec) {
- int error = 0;
- int numberOfErrors = 0;
- int codecSelection = 0;
- std::string str;
- memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
-
- bool exitLoop=false;
- while (!exitLoop) {
- std::cout << std::endl;
- std::cout << "Available video codecs:" << std::endl;
- int codecIdx = 0;
- int defaultCodecIdx = 0;
- // Print out all the codecs available to set Codec to.
- for (codecIdx = 0; codecIdx < ptrViECodec->NumberOfCodecs(); codecIdx++) {
- error = ptrViECodec->GetCodec(codecIdx, videoCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- // test for default codec index
- if (strcmp(videoCodec.plName, DEFAULT_VIDEO_CODEC) == 0) {
- defaultCodecIdx = codecIdx;
- }
- std::cout << " " << codecIdx+1 << ". " << videoCodec.plName
- << std::endl;
- }
- std::cout << std::endl;
- std::cout << "Choose video codec. Press enter for default ("
- << DEFAULT_VIDEO_CODEC << "): ";
- std::getline(std::cin, str);
- codecSelection = atoi(str.c_str());
- if (codecSelection == 0) {
- // use default
- error = ptrViECodec->GetCodec(defaultCodecIdx, videoCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- exitLoop=true;
- } else {
- // user selection
- codecSelection = atoi(str.c_str())-1;
- error = ptrViECodec->GetCodec(codecSelection, videoCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- if (error != 0) {
- std::cout << "ERROR: Code=" << error << " Invalid selection"
- << std::endl;
- continue;
- }
- exitLoop=true;
- }
- }
- if (videoCodec.codecType == webrtc::kVideoCodecI420) {
- videoCodec.width = 176;
- videoCodec.height = 144;
- }
- return true;
-}
-
-bool SetVideoCodecResolution(webrtc::ViECodec* ptrViECodec,
- webrtc::VideoCodec& videoCodec) {
- std::string str;
- int sizeOption = 5;
-
- if (videoCodec.codecType == webrtc::kVideoCodecVP8) {
- std::cout << std::endl;
- std::cout << "Available Common Resolutions : " << std::endl;
- std::cout << " 1. SQCIF (128X96) " << std::endl;
- std::cout << " 2. QQVGA (160X120) " << std::endl;
- std::cout << " 3. QCIF (176X144) " << std::endl;
- std::cout << " 4. CIF (352X288) " << std::endl;
- std::cout << " 5. VGA (640X480) " << std::endl;
- std::cout << " 6. WVGA (800x480) " << std::endl;
- std::cout << " 7. 4CIF (704X576) " << std::endl;
- std::cout << " 8. SVGA (800X600) " << std::endl;
- std::cout << " 9. HD (1280X720) " << std::endl;
- std::cout << " 10. XGA (1024x768) " << std::endl;
- std::cout << "Enter frame size option: " << std::endl;
-
- std::getline(std::cin, str);
- sizeOption = atoi(str.c_str());
-
- switch (sizeOption) {
- case 1:
- videoCodec.width = 128;
- videoCodec.height = 96;
- break;
- case 2:
- videoCodec.width = 160;
- videoCodec.height = 120;
- break;
- case 3:
- videoCodec.width = 176;
- videoCodec.height = 144;
- break;
- case 4:
- videoCodec.width = 352;
- videoCodec.height = 288;
- break;
- case 5:
- videoCodec.width = 640;
- videoCodec.height = 480;
- break;
- case 6:
- videoCodec.width = 800;
- videoCodec.height = 480;
- break;
- case 7:
- videoCodec.width = 704;
- videoCodec.height = 576;
- break;
- case 8:
- videoCodec.width = 800;
- videoCodec.height = 600;
- break;
- case 9:
- videoCodec.width = 1280;
- videoCodec.height = 720;
- break;
- case 10:
- videoCodec.width = 1024;
- videoCodec.height = 768;
- break;
- }
- } else {
- std::cout << "Can Only change codec size if it's VP8" << std::endl;
- }
- return true;
-}
-
-bool SetVideoCodecSize(webrtc::ViECodec* ptrViECodec,
- webrtc::VideoCodec& videoCodec) {
- if (videoCodec.codecType == webrtc::kVideoCodecVP8) {
- std::string str;
- videoCodec.width = DEFAULT_VIDEO_CODEC_WIDTH;
- videoCodec.height = DEFAULT_VIDEO_CODEC_HEIGHT;
- std::cout << "Choose video width. Press enter for default ("
- << DEFAULT_VIDEO_CODEC_WIDTH << "): ";
- std::getline(std::cin, str);
- int sizeSelection = atoi(str.c_str());
- if (sizeSelection!=0) {
- videoCodec.width=sizeSelection;
- }
- std::cout << "Choose video height. Press enter for default ("
- << DEFAULT_VIDEO_CODEC_HEIGHT << "): ";
- std::getline(std::cin, str);
- sizeSelection = atoi(str.c_str());
- if (sizeSelection!=0) {
- videoCodec.height=sizeSelection;
- }
- } else {
- std::cout << "Can Only change codec size if it's VP8" << std::endl;
- }
- return true;
-}
-
-bool SetVideoCodecBitrate(webrtc::ViECodec* ptrViECodec,
- webrtc::VideoCodec& videoCodec) {
- std::string str;
- std::cout << std::endl;
- std::cout << "Choose start rate (in kbps). Press enter for default ("
- << DEFAULT_VIDEO_CODEC_BITRATE << "): ";
- std::getline(std::cin, str);
- int startRate = atoi(str.c_str());
- videoCodec.startBitrate = DEFAULT_VIDEO_CODEC_BITRATE;
- if (startRate != 0) {
- videoCodec.startBitrate = startRate;
- }
- return true;
-}
-
-bool SetVideoCodecMaxBitrate(webrtc::ViECodec* ptrViECodec,
- webrtc::VideoCodec& videoCodec) {
- std::string str;
- std::cout << std::endl;
- std::cout << "Choose max bitrate (in kbps). Press enter for default ("
- << DEFAULT_VIDEO_CODEC_MAX_BITRATE << "): ";
- std::getline(std::cin, str);
- int maxRate = atoi(str.c_str());
- videoCodec.maxBitrate = DEFAULT_VIDEO_CODEC_MAX_BITRATE;
- if (maxRate != 0) {
- videoCodec.maxBitrate = maxRate;
- }
- return true;
-}
-
-bool SetVideoCodecMinBitrate(webrtc::ViECodec* ptrViECodec,
- webrtc::VideoCodec& videoCodec) {
- std::string str;
- std::cout << std::endl;
- std::cout << "Choose min bitrate (in fps). Press enter for default ("
- << DEFAULT_VIDEO_CODEC_MIN_BITRATE << "): ";
- std::getline(std::cin, str);
- char minBitRate = atoi(str.c_str());
- videoCodec.minBitrate = DEFAULT_VIDEO_CODEC_MIN_BITRATE;
- if (minBitRate != 0) {
- videoCodec.minBitrate = minBitRate;
- }
- return true;
-}
-
-bool SetVideoCodecMaxFramerate(webrtc::ViECodec* ptrViECodec,
- webrtc::VideoCodec& videoCodec) {
- std::string str;
- std::cout << std::endl;
- std::cout << "Choose max framerate (in fps). Press enter for default ("
- << DEFAULT_VIDEO_CODEC_MAX_FRAMERATE << "): ";
- std::getline(std::cin, str);
- char maxFrameRate = atoi(str.c_str());
- videoCodec.maxFramerate = DEFAULT_VIDEO_CODEC_MAX_FRAMERATE;
- if (maxFrameRate != 0) {
- videoCodec.maxFramerate = maxFrameRate;
- }
- return true;
-}
-
-bool SetVideoCodecTemporalLayer(webrtc::VideoCodec& videoCodec) {
- if (videoCodec.codecType == webrtc::kVideoCodecVP8) {
- std::string str;
- std::cout << std::endl;
- std::cout << "Choose number of temporal layers (1 to 4). "
- << "Press enter for default ("
- << DEFAULT_TEMPORAL_LAYER << "): ";
- std::getline(std::cin, str);
- char numTemporalLayers = atoi(str.c_str());
- videoCodec.codecSpecific.VP8.numberOfTemporalLayers
- = DEFAULT_TEMPORAL_LAYER;
- if(numTemporalLayers != 0) {
- videoCodec.codecSpecific.VP8.numberOfTemporalLayers
- = numTemporalLayers;
- }
- }
- return true;
-}
-// GetVideoProtection only prints the prompt to get a number
-// that SetVideoProtection method uses
-// 0 = None
-// 1 = FEC
-// 2 = NACK
-// 3 = NACK + FEC (aka Hybrid)
-// Default = DEFAULT_VIDEO_PROTECTION METHOD
-int GetVideoProtection() {
- int protectionMethod = DEFAULT_VIDEO_PROTECTION_METHOD;
-
- std::cout << "Available Video Protection Method." << std::endl;
- std::cout << " 0. None" << std::endl;
- std::cout << " 1. FEC" << std::endl;
- std::cout << " 2. NACK" << std::endl;
- std::cout << " 3. NACK+FEC" << std::endl;
- std::cout << "Enter Video Protection Method. "
- << "Press enter for default (" << protectionMethod << "):"
- << std::endl;
- std::string method;
- std::getline(std::cin, method);
- protectionMethod = atoi(method.c_str());
-
- return protectionMethod;
-}
-
-bool SetVideoProtection(webrtc::ViECodec* ptrViECodec,
- webrtc::ViERTP_RTCP* ptrViERtpRtcp,
- int videoChannel, int protectionMethod) {
- int error = 0;
- int numberOfErrors = 0;
- webrtc::VideoCodec videoCodec;
-
- memset((void*)&videoCodec, 0, sizeof(webrtc::VideoCodec));
-
- // Set all video protection to false initially
- error = ptrViERtpRtcp->SetHybridNACKFECStatus(videoChannel, false,
- VCM_RED_PAYLOAD_TYPE,
- VCM_ULPFEC_PAYLOAD_TYPE);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- error = ptrViERtpRtcp->SetFECStatus(videoChannel, false,
- VCM_RED_PAYLOAD_TYPE,
- VCM_ULPFEC_PAYLOAD_TYPE);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- error = ptrViERtpRtcp->SetNACKStatus(videoChannel, false);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- // Set video protection for FEC, NACK or Hybrid
- switch (protectionMethod) {
- case 0: // None
- // No protection selected, all protection already at false
- std::cout << "Call using None protection Method"
- << std::endl;
- break;
- case 1: // FEC only
- std::cout << "Call using FEC protection Method"
- << std::endl;
- error = ptrViERtpRtcp->SetFECStatus(videoChannel, true,
- VCM_RED_PAYLOAD_TYPE,
- VCM_ULPFEC_PAYLOAD_TYPE);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- break;
- case 2: // NACK only
- std::cout << "Call using NACK protection Method"
- << std::endl;
- error = ptrViERtpRtcp->SetNACKStatus(videoChannel, true);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- break;
- case 3: // Hybrid NACK and FEC
- std::cout << "Call using Hybrid NACK and FEC protection Method"
- << std::endl;
- error = ptrViERtpRtcp->SetHybridNACKFECStatus(videoChannel, true,
- VCM_RED_PAYLOAD_TYPE,
- VCM_ULPFEC_PAYLOAD_TYPE);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- break;
- }
-
- // Set receive codecs for FEC and hybrid NACK/FEC
- if (protectionMethod == 1 || protectionMethod == 3) {
- // RED
- error = ptrViECodec->GetCodec(ptrViECodec->NumberOfCodecs() - 2,
- videoCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- videoCodec.plType = VCM_RED_PAYLOAD_TYPE;
- error = ptrViECodec->SetReceiveCodec(videoChannel, videoCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- std::cout << "RED Codec Information:" << std::endl;
- PrintVideoCodec(videoCodec);
- // ULPFEC
- error = ptrViECodec->GetCodec(ptrViECodec->NumberOfCodecs() - 1,
- videoCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- videoCodec.plType = VCM_ULPFEC_PAYLOAD_TYPE;
- error = ptrViECodec->SetReceiveCodec(videoChannel, videoCodec);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- std::cout << "ULPFEC Codec Information:" << std::endl;
- PrintVideoCodec(videoCodec);
- }
-
-
-
- return true;
-}
-
-bool GetBitrateSignaling() {
- std::cout << std::endl;
- std::cout << "Available bitrate signaling methods." << std::endl;
- std::cout << " 0. REMB" << std::endl;
- std::cout << " 1. TMMBR" << std::endl;
- std::cout << "Enter bitrate signaling methods. "
- << "Press enter for default (REMB): " << std::endl;
- std::string method;
- std::getline(std::cin, method);
- if (atoi(method.c_str()) == 1) {
- return false;
- }
- return true;
-}
-
-void PrintRTCCPStatistics(webrtc::ViERTP_RTCP* ptrViERtpRtcp,
- int videoChannel, StatisticsType statType) {
- int error = 0;
- int numberOfErrors =0;
- unsigned short fractionLost = 0;
- unsigned int cumulativeLost = 0;
- unsigned int extendedMax = 0;
- unsigned int jitter = 0;
- int rttMS = 0;
-
- switch (statType) {
- case kReceivedStatistic:
- std::cout << "RTCP Received statistics"
- << std::endl;
- // Get and print the Received RTCP Statistics
- error = ptrViERtpRtcp->GetReceivedRTCPStatistics(videoChannel,
- fractionLost,
- cumulativeLost,
- extendedMax,
- jitter, rttMS);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- break;
- case kSendStatistic:
- std::cout << "RTCP Sent statistics"
- << std::endl;
- // Get and print the Sent RTCP Statistics
- error = ptrViERtpRtcp->GetSentRTCPStatistics(videoChannel, fractionLost,
- cumulativeLost, extendedMax,
- jitter, rttMS);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- break;
- }
- std::cout << "\tRTCP fraction of lost packets: "
- << fractionLost << std::endl;
- std::cout << "\tRTCP cumulative number of lost packets: "
- << cumulativeLost << std::endl;
- std::cout << "\tRTCP max received sequence number "
- << extendedMax << std::endl;
- std::cout << "\tRTCP jitter: "
- << jitter << std::endl;
- std::cout << "\tRTCP round trip (ms): "
- << rttMS<< std::endl;
-}
-
-void PrintRTPStatistics(webrtc::ViERTP_RTCP* ptrViERtpRtcp,
- int videoChannel) {
- int error = 0;
- int numberOfErrors =0;
- unsigned int bytesSent = 0;
- unsigned int packetsSent= 0;
- unsigned int bytesReceived = 0;
- unsigned int packetsReceived = 0;
-
- std::cout << "RTP statistics"
- << std::endl;
-
- // Get and print the RTP Statistics
- error = ptrViERtpRtcp->GetRTPStatistics(videoChannel, bytesSent, packetsSent,
- bytesReceived, packetsReceived);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- std::cout << "\tRTP bytes sent: "
- << bytesSent << std::endl;
- std::cout << "\tRTP packets sent: "
- << packetsSent << std::endl;
- std::cout << "\tRTP bytes received: "
- << bytesReceived << std::endl;
- std::cout << "\tRTP packets received: "
- << packetsReceived << std::endl;
-}
-
-void PrintBandwidthUsage(webrtc::ViERTP_RTCP* ptrViERtpRtcp,
- int videoChannel) {
- int error = 0;
- int numberOfErrors = 0;
- unsigned int totalBitrateSent = 0;
- unsigned int videoBitrateSent = 0;
- unsigned int fecBitrateSent = 0;
- unsigned int nackBitrateSent = 0;
- double percentageFEC = 0;
- double percentageNACK = 0;
-
- std::cout << "Bandwidth Usage"
- << std::endl;
-
- // Get and print Bandwidth usage
- error = ptrViERtpRtcp->GetBandwidthUsage(videoChannel, totalBitrateSent,
- videoBitrateSent, fecBitrateSent,
- nackBitrateSent);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- std::cout << "\tTotal bitrate sent (Kbit/s): "
- << totalBitrateSent << std::endl;
- std::cout << "\tVideo bitrate sent (Kbit/s): "
- << videoBitrateSent << std::endl;
- std::cout << "\tFEC bitrate sent (Kbit/s): "
- << fecBitrateSent << std::endl;
- percentageFEC = ((double)fecBitrateSent/(double)totalBitrateSent) * 100;
- std::cout << "\tPercentage FEC bitrate sent from total bitrate: "
- << percentageFEC << std::endl;
- std::cout << "\tNACK bitrate sent (Kbit/s): "
- << nackBitrateSent << std::endl;
- percentageNACK = ((double)nackBitrateSent/(double)totalBitrateSent) * 100;
- std::cout << "\tPercentage NACK bitrate sent from total bitrate: "
- << percentageNACK << std::endl;
-}
-
-void PrintCodecStatistics(webrtc::ViECodec* ptrViECodec, int videoChannel,
- StatisticsType statType) {
- int error = 0;
- int numberOfErrors = 0;
- unsigned int keyFrames = 0;
- unsigned int deltaFrames = 0;
- switch(statType) {
- case kReceivedStatistic:
- std::cout << "Codec Receive statistics"
- << std::endl;
- // Get and print the Receive Codec Statistics
- error = ptrViECodec->GetReceiveCodecStastistics(videoChannel, keyFrames,
- deltaFrames);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- break;
- case kSendStatistic:
- std::cout << "Codec Send statistics"
- << std::endl;
- // Get and print the Send Codec Statistics
- error = ptrViECodec->GetSendCodecStastistics(videoChannel, keyFrames,
- deltaFrames);
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);
- break;
- }
- std::cout << "\tNumber of encoded key frames: "
- << keyFrames << std::endl;
- std::cout << "\tNumber of encoded delta frames: "
- << deltaFrames << std::endl;
-}
-
-void PrintGetDiscardedPackets(webrtc::ViECodec* ptrViECodec, int videoChannel) {
- std::cout << "Discarded Packets"
- << std::endl;
- int discardedPackets = 0;
- discardedPackets = ptrViECodec->GetDiscardedPackets(videoChannel);
- std::cout << "\tNumber of discarded packets: "
- << discardedPackets << std::endl;
-}
-
-void PrintVideoStreamInformation(webrtc::ViECodec* ptrViECodec,
- int videoChannel) {
- webrtc::VideoCodec outgoingCodec;
- webrtc::VideoCodec incomingCodec;
-
- memset((void*)&outgoingCodec, 0, sizeof(webrtc::VideoCodec));
- memset((void*)&incomingCodec, 0, sizeof(webrtc::VideoCodec));
-
- ptrViECodec->GetSendCodec(videoChannel, outgoingCodec);
- ptrViECodec->GetReceiveCodec(videoChannel, incomingCodec);
-
- std::cout << "************************************************"
- << std::endl;
- std::cout << "ChannelId: " << videoChannel << std::endl;
- std::cout << "Outgoing Stream information:" <<std::endl;
- PrintVideoCodec(outgoingCodec);
- std::cout << "Incoming Stream information:" <<std::endl;
- PrintVideoCodec(incomingCodec);
- std::cout << "************************************************"
- << std::endl;
-}
-
-void PrintVideoCodec(webrtc::VideoCodec videoCodec) {
- std::cout << "\t\tplName: " << videoCodec.plName << std::endl;
- std::cout << "\t\tplType: " << (int)videoCodec.plType << std::endl;
- std::cout << "\t\twidth: " << videoCodec.width << std::endl;
- std::cout << "\t\theight: " << videoCodec.height << std::endl;
- std::cout << "\t\tstartBitrate: " << videoCodec.startBitrate
- << std::endl;
- std::cout << "\t\tminBitrate: " << videoCodec.minBitrate
- << std::endl;
- std::cout << "\t\tmaxBitrate: " << videoCodec.maxBitrate
- << std::endl;
- std::cout << "\t\tmaxFramerate: " << (int)videoCodec.maxFramerate
- << std::endl;
- if (videoCodec.codecType == webrtc::kVideoCodecVP8) {
- std::cout << "\t\tVP8 Temporal Layer: "
- << (int)videoCodec.codecSpecific.VP8.numberOfTemporalLayers
- << std::endl;
- }
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_encryption.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_encryption.cc
deleted file mode 100644
index 0c06f32..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_encryption.cc
+++ /dev/null
@@ -1,567 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest_encryption.cc
-//
-
-#include "vie_autotest_defines.h"
-#include "vie_autotest.h"
-#include "engine_configurations.h"
-
-#include "tb_capture_device.h"
-#include "tb_external_transport.h"
-#include "tb_interfaces.h"
-#include "tb_video_channel.h"
-
-class ViEAutotestEncryption: public webrtc::Encryption
-{
-public:
- ViEAutotestEncryption()
- {
- }
- ~ViEAutotestEncryption()
- {
- }
-
- virtual void encrypt(int channel_no, unsigned char * in_data,
- unsigned char * out_data, int bytes_in, int* bytes_out)
- {
- for (int i = 0; i < bytes_in; i++)
- {
- out_data[i] = ~in_data[i];
- }
- *bytes_out = bytes_in + 2;
- }
-
- virtual void decrypt(int channel_no, unsigned char * in_data,
- unsigned char * out_data, int bytes_in, int* bytes_out)
- {
- for (int i = 0; i < bytes_in - 2; i++)
- {
- out_data[i] = ~in_data[i];
- }
- *bytes_out = bytes_in - 2;
- }
-
- virtual void encrypt_rtcp(int channel_no, unsigned char * in_data,
- unsigned char * out_data, int bytes_in,
- int* bytes_out)
- {
- for (int i = 0; i < bytes_in; i++)
- {
- out_data[i] = ~in_data[i];
- }
- *bytes_out = bytes_in + 2;
- }
-
- virtual void decrypt_rtcp(int channel_no, unsigned char * in_data,
- unsigned char * out_data, int bytes_in,
- int* bytes_out)
- {
- for (int i = 0; i < bytes_in - 2; i++)
- {
- out_data[i] = ~in_data[i];
- }
- *bytes_out = bytes_in - 2;
- }
-};
-
-void ViEAutoTest::ViEEncryptionStandardTest()
-{
- //***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- //***************************************************************
-
- // Create VIE
- TbInterfaces ViE("ViEEncryptionStandardTest");
- // Create a video channel
- TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
-
- // Create a capture device
- TbCaptureDevice tbCapture(ViE);
- tbCapture.ConnectTo(tbChannel.videoChannel);
-
- tbChannel.StartReceive();
-
- tbChannel.StartSend();
-
- RenderCaptureDeviceAndOutputStream(&ViE, &tbChannel, &tbCapture);
-
-#ifdef WEBRTC_SRTP
- //***************************************************************
- // Engine ready. Begin testing class
- //***************************************************************
-
- //
- // SRTP
- //
- unsigned char srtpKey1[30] =
- { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3,
- 4, 5, 6, 7, 8, 9};
-
- // Encryption only
- EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthNull, 0, 0, webrtc::kEncryption, srtpKey1));
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthNull, 0, 0, webrtc::kEncryption, srtpKey1));
- ViETest::Log("SRTP encryption only");
- AutoTestSleep(KAutoTestSleepTimeMs);
- EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
-
- // Authentication only
- EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
- 20, 4, webrtc::kAuthentication, srtpKey1));
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
- 20, 4, webrtc::kAuthentication, srtpKey1));
-
- ViETest::Log("SRTP authentication only");
- AutoTestSleep(KAutoTestSleepTimeMs);
- EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
-
- // Full protection
- EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey1));
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey1));
-
- ViETest::Log("SRTP full protection");
- AutoTestSleep(KAutoTestSleepTimeMs);
- EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
-#endif // WEBRTC_SRTP
-
- //
- // External encryption
- //
- ViEAutotestEncryption testEncryption;
- // Note(qhogpat): StartSend fails, not sure if this is intentional.
- EXPECT_NE(0, ViE.base->StartSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->RegisterExternalEncryption(
- tbChannel.videoChannel, testEncryption));
- ViETest::Log(
- "External encryption/decryption added, you should still see video");
- AutoTestSleep(KAutoTestSleepTimeMs);
- EXPECT_EQ(0, ViE.encryption->DeregisterExternalEncryption(
- tbChannel.videoChannel));
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
-}
-
-void ViEAutoTest::ViEEncryptionExtendedTest()
-{
- //***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- //***************************************************************
-
- // Create VIE
- TbInterfaces ViE("ViEEncryptionExtendedTest");
- // Create a video channel
- TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
-
- // Create a capture device
- TbCaptureDevice tbCapture(ViE);
- tbCapture.ConnectTo(tbChannel.videoChannel);
-
- tbChannel.StartReceive();
- tbChannel.StartSend();
-
- RenderCaptureDeviceAndOutputStream(&ViE, &tbChannel, &tbCapture);
-
- //***************************************************************
- // Engine ready. Begin testing class
- //***************************************************************
-
-#ifdef WEBRTC_SRTP
-
- //
- // SRTP
- //
- unsigned char srtpKey1[30] =
- { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3,
- 4, 5, 6, 7, 8, 9};
- unsigned char srtpKey2[30] =
- { 9, 8, 7, 6, 5, 4, 3, 2, 1, 0, 9, 8, 7, 6, 5, 4, 3, 2, 1, 0, 9, 8, 7, 6,
- 5, 4, 3, 2, 1, 0};
- // NULL
- EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthNull, 0, 0,
- webrtc::kNoProtection, srtpKey1));
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthNull, 0, 0,
- webrtc::kNoProtection, srtpKey1));
-
- ViETest::Log("SRTP NULL encryption/authentication");
- AutoTestSleep(KAutoTestSleepTimeMs);
- EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
-
- // Encryption only
- EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthNull, 0, 0, webrtc::kEncryption, srtpKey1));
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthNull, 0, 0, webrtc::kEncryption, srtpKey1));
-
- ViETest::Log("SRTP encryption only");
- AutoTestSleep(KAutoTestSleepTimeMs);
- EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
-
- // Authentication only
- EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
- 20, 4, webrtc::kAuthentication, srtpKey1));
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
- 20, 4, webrtc::kAuthentication, srtpKey1));
-
- ViETest::Log("SRTP authentication only");
- AutoTestSleep(KAutoTestSleepTimeMs);
- EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
-
- // Full protection
- EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey1));
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey1));
-
- ViETest::Log("SRTP full protection");
- AutoTestSleep(KAutoTestSleepTimeMs);
- EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
-
- // Change receive key, but not send key...
- EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey2));
-
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey1));
-
- ViETest::Log(
- "\nSRTP receive key changed, you should not see any remote images");
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- // Change send key too
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey2));
-
- ViETest::Log("\nSRTP send key changed too, you should see remote video "
- "again with some decoding artefacts at start");
- AutoTestSleep(KAutoTestSleepTimeMs);
- EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
-
- // Disable receive, keep send
- ViETest::Log("SRTP receive disabled , you shouldn't see any video");
- AutoTestSleep(KAutoTestSleepTimeMs);
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
-
-#endif //WEBRTC_SRTP
- //
- // External encryption
- //
- ViEAutotestEncryption testEncryption;
- EXPECT_EQ(0, ViE.encryption->RegisterExternalEncryption(
- tbChannel.videoChannel, testEncryption));
- ViETest::Log(
- "External encryption/decryption added, you should still see video");
- AutoTestSleep(KAutoTestSleepTimeMs);
- EXPECT_EQ(0, ViE.encryption->DeregisterExternalEncryption(
- tbChannel.videoChannel));
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
-}
-
-void ViEAutoTest::ViEEncryptionAPITest()
-{
- //***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- //***************************************************************
-
- //***************************************************************
- // Engine ready. Begin testing class
- //***************************************************************
-
- // Create VIE
- TbInterfaces ViE("ViEEncryptionAPITest");
- // Create a video channel
- TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
-
- // Create a capture device
- TbCaptureDevice tbCapture(ViE);
- // Connect to channel
- tbCapture.ConnectTo(tbChannel.videoChannel);
-
-#ifdef WEBRTC_SRTP
- unsigned char srtpKey[30] =
- { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3,
- 4, 5, 6, 7, 8, 9};
-
- //
- // EnableSRTPSend and DisableSRTPSend
- //
-
- // Incorrect input argument, complete protection not enabled
- EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kNoProtection, srtpKey));
- EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryption, srtpKey));
- EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kAuthentication, srtpKey));
-
- // Incorrect cipher key length
- EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 15,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey));
- EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 257,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey));
- EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherNull, 15, webrtc::kAuthHmacSha1,
- 20, 4, webrtc::kEncryptionAndAuthentication, srtpKey));
-
- EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherNull, 257, webrtc::kAuthHmacSha1,
- 20, 4, webrtc::kEncryptionAndAuthentication, srtpKey));
-
- // Incorrect auth key length
- EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode,
- 30, webrtc::kAuthHmacSha1, 21, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey));
- EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthNull, 257, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey));
- EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 21, webrtc::kEncryptionAndAuthentication,
- srtpKey));
- EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthNull, 20, 13, webrtc::kEncryptionAndAuthentication,
- srtpKey));
-
- // NULL input
- EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- NULL));
-
- // Double enable and disable
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey));
- EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey));
-
- // Note(qhogpat): the second check is likely incorrect.
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
-
- // No protection
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthNull, 0, 0,
- webrtc::kNoProtection, srtpKey));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
-
- // Authentication only
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
- 20, 4, webrtc::kAuthentication, srtpKey));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
- 1, 4, webrtc::kAuthentication, srtpKey));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
- 20, 20, webrtc::kAuthentication, srtpKey));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
- 1, 1, webrtc::kAuthentication, srtpKey));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
-
- // Encryption only
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthNull, 0, 0, webrtc::kEncryption, srtpKey));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 16,
- webrtc::kAuthNull, 0, 0, webrtc::kEncryption, srtpKey));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
-
- // Full protection
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
-
- //
- // EnableSRTPReceive and DisableSRTPReceive
- //
-
- // Incorrect input argument, complete protection not enabled
- EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kNoProtection, srtpKey));
- EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryption, srtpKey));
- EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kAuthentication, srtpKey));
-
- // Incorrect cipher key length
- EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 15,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey));
- EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 257,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey));
- EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherNull, 15, webrtc::kAuthHmacSha1,
- 20, 4, webrtc::kEncryptionAndAuthentication, srtpKey));
- EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherNull, 257, webrtc::kAuthHmacSha1,
- 20, 4, webrtc::kEncryptionAndAuthentication, srtpKey));
-
- // Incorrect auth key length
- EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 21, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey));
- EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthNull, 257, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey));
- EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 21, webrtc::kEncryptionAndAuthentication,
- srtpKey));
- EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthNull, 20, 13, webrtc::kEncryptionAndAuthentication,
- srtpKey));
-
- // NULL input
- EXPECT_NE(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- NULL));
-
- // Double enable and disable
- EXPECT_EQ(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey));
- EXPECT_NE(0, ViE.encryption->EnableSRTPSend(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPSend(tbChannel.videoChannel));
-
- // No protection
- EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthNull, 0, 0,
- webrtc::kNoProtection, srtpKey));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
-
- // Authentication only
- EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
- 1, 4, webrtc::kAuthentication, srtpKey));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherNull, 0,
- webrtc::kAuthHmacSha1, 20, 20, webrtc::kAuthentication, srtpKey));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherNull, 0, webrtc::kAuthHmacSha1,
- 1, 1, webrtc::kAuthentication, srtpKey));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
-
- // Encryption only
- EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthNull, 0, 0, webrtc::kEncryption, srtpKey));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 16,
- webrtc::kAuthNull, 0, 0, webrtc::kEncryption, srtpKey));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
-
- // Full protection
- EXPECT_EQ(0, ViE.encryption->EnableSRTPReceive(
- tbChannel.videoChannel, webrtc::kCipherAes128CounterMode, 30,
- webrtc::kAuthHmacSha1, 20, 4, webrtc::kEncryptionAndAuthentication,
- srtpKey));
- EXPECT_EQ(0, ViE.encryption->DisableSRTPReceive(tbChannel.videoChannel));
-#endif //WEBRTC_SRTP
- //
- // External encryption
- //
-
- ViEAutotestEncryption testEncryption;
- EXPECT_EQ(0, ViE.encryption->RegisterExternalEncryption(
- tbChannel.videoChannel, testEncryption));
- EXPECT_NE(0, ViE.encryption->RegisterExternalEncryption(
- tbChannel.videoChannel, testEncryption));
- EXPECT_EQ(0, ViE.encryption->DeregisterExternalEncryption(
- tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.encryption->DeregisterExternalEncryption(
- tbChannel.videoChannel));
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_file.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_file.cc
deleted file mode 100644
index 83ace52..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_file.cc
+++ /dev/null
@@ -1,493 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vie_autotest_defines.h"
-#include "vie_autotest.h"
-#include "engine_configurations.h"
-
-#include "testsupport/fileutils.h"
-#include "tb_interfaces.h"
-#include "tb_capture_device.h"
-
-#include "voe_codec.h"
-
-class ViEAutotestFileObserver: public webrtc::ViEFileObserver
-{
-public:
- ViEAutotestFileObserver() {};
- ~ViEAutotestFileObserver() {};
-
- void PlayFileEnded(const WebRtc_Word32 fileId)
- {
- ViETest::Log("PlayFile ended");
- }
-};
-
-void ViEAutoTest::ViEFileStandardTest()
-{
-#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
- //***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- //***************************************************************
- {
- ViETest::Log("Starting a loopback call...");
-
- TbInterfaces interfaces = TbInterfaces("ViEFileStandardTest");
-
- webrtc::VideoEngine* ptrViE = interfaces.video_engine;
- webrtc::ViEBase* ptrViEBase = interfaces.base;
- webrtc::ViECapture* ptrViECapture = interfaces.capture;
- webrtc::ViERender* ptrViERender = interfaces.render;
- webrtc::ViECodec* ptrViECodec = interfaces.codec;
- webrtc::ViERTP_RTCP* ptrViERtpRtcp = interfaces.rtp_rtcp;
- webrtc::ViENetwork* ptrViENetwork = interfaces.network;
-
- TbCaptureDevice captureDevice = TbCaptureDevice(interfaces);
- int captureId = captureDevice.captureId;
-
- int videoChannel = -1;
- EXPECT_EQ(0, ptrViEBase->CreateChannel(videoChannel));
- EXPECT_EQ(0, ptrViECapture->ConnectCaptureDevice(
- captureId, videoChannel));
-
- EXPECT_EQ(0, ptrViERtpRtcp->SetRTCPStatus(
- videoChannel, webrtc::kRtcpCompound_RFC4585));
- EXPECT_EQ(0, ptrViERtpRtcp->SetKeyFrameRequestMethod(
- videoChannel, webrtc::kViEKeyFrameRequestPliRtcp));
- EXPECT_EQ(0, ptrViERtpRtcp->SetTMMBRStatus(videoChannel, true));
-
- EXPECT_EQ(0, ptrViERender->AddRenderer(
- captureId, _window1, 0, 0.0, 0.0, 1.0, 1.0));
- EXPECT_EQ(0, ptrViERender->AddRenderer(
- videoChannel, _window2, 1, 0.0, 0.0, 1.0, 1.0));
- EXPECT_EQ(0, ptrViERender->StartRender(captureId));
- EXPECT_EQ(0, ptrViERender->StartRender(videoChannel));
-
- webrtc::VideoCodec videoCodec;
- memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
- for (int idx = 0; idx < ptrViECodec->NumberOfCodecs(); idx++)
- {
- EXPECT_EQ(0, ptrViECodec->GetCodec(idx, videoCodec));
- EXPECT_EQ(0, ptrViECodec->SetReceiveCodec(videoChannel,
- videoCodec));
- }
-
- // Find the codec used for encoding the channel
- for (int idx = 0; idx < ptrViECodec->NumberOfCodecs(); idx++)
- {
- EXPECT_EQ(0, ptrViECodec->GetCodec(idx, videoCodec));
- if (videoCodec.codecType == webrtc::kVideoCodecVP8)
- {
- EXPECT_EQ(0, ptrViECodec->SetSendCodec(videoChannel, videoCodec));
- break;
- }
- }
- // Find the codec used for recording.
- for (int idx = 0; idx < ptrViECodec->NumberOfCodecs(); idx++)
- {
- EXPECT_EQ(0, ptrViECodec->GetCodec(idx, videoCodec));
- if (videoCodec.codecType == webrtc::kVideoCodecI420)
- {
- break;
- }
- }
-
-
- const char* ipAddress = "127.0.0.1";
- const unsigned short rtpPort = 6000;
- EXPECT_EQ(0, ptrViENetwork->SetLocalReceiver(videoChannel, rtpPort));
- EXPECT_EQ(0, ptrViEBase->StartReceive(videoChannel));
- EXPECT_EQ(0, ptrViENetwork->SetSendDestination(
- videoChannel, ipAddress, rtpPort));
- EXPECT_EQ(0, ptrViEBase->StartSend(videoChannel));
- webrtc::ViEFile* ptrViEFile = webrtc::ViEFile::GetInterface(ptrViE);
- EXPECT_TRUE(ptrViEFile != NULL);
-
- webrtc::VoiceEngine* ptrVEEngine = webrtc::VoiceEngine::Create();
- webrtc::VoEBase* ptrVEBase = webrtc::VoEBase::GetInterface(ptrVEEngine);
- ptrVEBase->Init();
-
- int audioChannel = ptrVEBase->CreateChannel();
- ptrViEBase->SetVoiceEngine(ptrVEEngine);
- ptrViEBase->ConnectAudioChannel(videoChannel, audioChannel);
-
- webrtc::CodecInst audioCodec;
- webrtc::VoECodec* ptrVECodec =
- webrtc::VoECodec::GetInterface(ptrVEEngine);
- for (int index = 0; index < ptrVECodec->NumOfCodecs(); index++)
- {
- ptrVECodec->GetCodec(index, audioCodec);
- if (0 == strcmp(audioCodec.plname, "PCMU") || 0
- == strcmp(audioCodec.plname, "PCMA"))
- {
- break; // these two types are allowed as avi recording formats
- }
- }
-
- webrtc::CodecInst audioCodec2;
-
- //***************************************************************
- // Engine ready. Begin testing class
- //***************************************************************
-
- // Call started
- ViETest::Log("Call started\nYou should see local preview from camera\n"
- "in window 1 and the remote video in window 2.");
- AutoTestSleep(2000);
-
- const int RENDER_TIMEOUT = 1000;
- const int TEST_SPACING = 1000;
- const int VIDEO_LENGTH = 5000;
-
- const std::string root = webrtc::test::ProjectRootPath() +
- "src/video_engine/test/auto_test/media/";
- const std::string renderStartImage = root + "renderStartImage.jpg";
- const std::string captureDeviceImage = root + "captureDeviceImage.jpg";
- const std::string renderTimeoutFile = root + "renderTimeoutImage.jpg";
-
- const std::string output = webrtc::test::OutputPath();
- const std::string snapshotCaptureDeviceFileName =
- output + "snapshotCaptureDevice.jpg";
- const std::string incomingVideo = output + "incomingVideo.avi";
- const std::string outgoingVideo = output + "outgoingVideo.avi";
- const std::string snapshotRenderFileName =
- output + "snapshotRenderer.jpg";
-
- webrtc::ViEPicture capturePicture;
- webrtc::ViEPicture renderPicture;
- webrtc::ViEPicture renderTimeoutPicture; // TODO: init with and image
-
- ViEAutotestFileObserver fileObserver;
- int fileId;
-
- AutoTestSleep(TEST_SPACING);
-
- // testing StartRecordIncomingVideo and StopRecordIncomingVideo
- {
- ViETest::Log("Recording incoming video (currently no audio) for %d "
- "seconds", VIDEO_LENGTH);
-
- EXPECT_EQ(0, ptrViEFile->StartRecordIncomingVideo(
- videoChannel, incomingVideo.c_str(), webrtc::NO_AUDIO,
- audioCodec2, videoCodec));
-
- AutoTestSleep(VIDEO_LENGTH);
- ViETest::Log("Stop recording incoming video");
-
- EXPECT_EQ(0, ptrViEFile->StopRecordIncomingVideo(videoChannel));
- ViETest::Log("Done\n");
- }
-
- AutoTestSleep(TEST_SPACING);
-
- // testing GetFileInformation
- {
- webrtc::VideoCodec fileVideoCodec;
- webrtc::CodecInst fileAudioCodec;
- ViETest::Log("Reading video file information");
-
- EXPECT_EQ(0, ptrViEFile->GetFileInformation(
- incomingVideo.c_str(), fileVideoCodec, fileAudioCodec));
- PrintAudioCodec(fileAudioCodec);
- PrintVideoCodec(fileVideoCodec);
- }
-
- // testing StartPlayFile and RegisterObserver
- {
- ViETest::Log("Start playing file: %s with observer",
- incomingVideo.c_str());
- EXPECT_EQ(0, ptrViEFile->StartPlayFile(incomingVideo.c_str(),
- fileId));
-
- ViETest::Log("Registering file observer");
- EXPECT_EQ(0, ptrViEFile->RegisterObserver(fileId, fileObserver));
- ViETest::Log("Done\n");
- }
-
- // testing SendFileOnChannel and StopSendFileOnChannel
- {
- ViETest::Log("Sending video on channel");
- // should fail since we are sending the capture device.
- EXPECT_NE(0, ptrViEFile->SendFileOnChannel(fileId, videoChannel));
-
- // Disconnect the camera
- EXPECT_EQ(0, ptrViECapture->DisconnectCaptureDevice(videoChannel));
-
- // And try playing the file again.
- EXPECT_EQ(0, ptrViEFile->SendFileOnChannel(fileId, videoChannel));
-
- AutoTestSleep(VIDEO_LENGTH);
- ViETest::Log("Stopped sending video on channel");
- EXPECT_EQ(0, ptrViEFile->StopSendFileOnChannel(videoChannel));
- ViETest::Log("Done\n");
- }
-
- AutoTestSleep(TEST_SPACING);
-
- // stop playing the file
- {
- ViETest::Log("Stop playing the file.");
- EXPECT_EQ(0, ptrViEFile->StopPlayFile(fileId));
- ViETest::Log("Done\n");
- }
-
- // testing StartRecordOutgoingVideo and StopRecordOutgoingVideo
- {
- // connect the camera to the output.
- EXPECT_EQ(0, ptrViECapture->ConnectCaptureDevice(
- captureId, videoChannel));
-
- ViETest::Log("Recording outgoing video (currently no audio) for %d "
- "seconds", VIDEO_LENGTH);
- EXPECT_EQ(0, ptrViEFile->StartRecordOutgoingVideo(
- videoChannel, outgoingVideo.c_str(), webrtc::NO_AUDIO,
- audioCodec2, videoCodec));
-
- AutoTestSleep(VIDEO_LENGTH);
- ViETest::Log("Stop recording outgoing video");
- EXPECT_EQ(0, ptrViEFile->StopRecordOutgoingVideo(videoChannel));
- ViETest::Log("Done\n");
- }
-
- // again testing GetFileInformation
- {
- EXPECT_EQ(0, ptrViEFile->GetFileInformation(
- incomingVideo.c_str(), videoCodec, audioCodec2));
- PrintAudioCodec(audioCodec2);
- PrintVideoCodec(videoCodec);
- }
-
- AutoTestSleep(TEST_SPACING);
-
- // GetCaptureDeviceSnapshot
- {
- ViETest::Log("Testing GetCaptureDeviceSnapshot(int, ViEPicture)");
- ViETest::Log("Taking a picture to use for displaying ViEPictures "
- "for the rest of file test");
- ViETest::Log("Hold an object to the camera. Ready?...");
- AutoTestSleep(1000);
- ViETest::Log("3");
- AutoTestSleep(1000);
- ViETest::Log("...2");
- AutoTestSleep(1000);
- ViETest::Log("...1");
- AutoTestSleep(1000);
- ViETest::Log("...Taking picture!");
- EXPECT_EQ(0, ptrViEFile->GetCaptureDeviceSnapshot(
- captureId, capturePicture));
- ViETest::Log("Picture has been taken.");
- AutoTestSleep(TEST_SPACING);
-
- ViETest::Log("Done\n");
- }
-
- AutoTestSleep(TEST_SPACING);
-
- // GetRenderSnapshot
- {
- ViETest::Log("Testing GetRenderSnapshot(int, char*)");
-
- ViETest::Log("Taking snapshot of videoChannel %d", captureId);
- EXPECT_EQ(0, ptrViEFile->GetRenderSnapshot(
- captureId, snapshotRenderFileName.c_str()));
- ViETest::Log("Wrote image to file %s",
- snapshotRenderFileName.c_str());
- ViETest::Log("Done\n");
- AutoTestSleep(TEST_SPACING);
- }
-
- // GetRenderSnapshot
- {
- ViETest::Log("Testing GetRenderSnapshot(int, ViEPicture)");
- EXPECT_EQ(0, ptrViEFile->GetRenderSnapshot(
- captureId, renderPicture));
- ViETest::Log("Done\n");
- }
-
- AutoTestSleep(TEST_SPACING);
-
- // GetCaptureDeviceSnapshot
- {
- ViETest::Log("Testing GetCaptureDeviceSnapshot(int, char*)");
- ViETest::Log("Taking snapshot from capture device %d", captureId);
- EXPECT_EQ(0, ptrViEFile->GetCaptureDeviceSnapshot(
- captureId, snapshotCaptureDeviceFileName.c_str()));
- ViETest::Log("Wrote image to file %s",
- snapshotCaptureDeviceFileName.c_str());
- ViETest::Log("Done\n");
- }
-
- AutoTestSleep(TEST_SPACING);
-
- // Testing: SetCaptureDeviceImage
- {
- ViETest::Log("Testing SetCaptureDeviceImage(int, char*)");
- EXPECT_EQ(0, ptrViECapture->StopCapture(captureId));
- EXPECT_EQ(0, ptrViEFile->SetCaptureDeviceImage(
- captureId, captureDeviceImage.c_str()));
-
- ViETest::Log("you should see the capture device image now");
- AutoTestSleep(2 * RENDER_TIMEOUT);
- EXPECT_EQ(0, ptrViECapture->StartCapture(captureId));
- ViETest::Log("Done\n");
- }
-
- AutoTestSleep(TEST_SPACING);
-
- // Testing: SetCaptureDeviceImage
- {
- ViETest::Log("Testing SetCaptureDeviceImage(int, ViEPicture)");
- EXPECT_EQ(0, ptrViECapture->StopCapture(captureId));
- EXPECT_EQ(0, ptrViEFile->SetCaptureDeviceImage(
- captureId, capturePicture));
-
- ViETest::Log("you should see the capture device image now");
- AutoTestSleep(2 * RENDER_TIMEOUT);
- EXPECT_EQ(0, ptrViECapture->StartCapture(captureId));
- ViETest::Log("Done\n");
- }
-
- AutoTestSleep(TEST_SPACING);
-
- // testing SetRenderStartImage(videoChannel, renderStartImage);
- {
- ViETest::Log("Testing SetRenderStartImage(int, char*)");
- // set render image, then stop capture and stop render to display it
- ViETest::Log("Stoping renderer, setting start image, then "
- "restarting");
- EXPECT_EQ(0, ptrViEFile->SetRenderStartImage(
- videoChannel, renderStartImage.c_str()));
- EXPECT_EQ(0, ptrViECapture->StopCapture(captureId));
- EXPECT_EQ(0, ptrViERender->StopRender(videoChannel));
-
- ViETest::Log("Render start image should be displayed.");
- AutoTestSleep(RENDER_TIMEOUT);
-
- // restarting capture and render
- EXPECT_EQ(0, ptrViECapture->StartCapture(captureId));
- EXPECT_EQ(0, ptrViERender->StartRender(videoChannel));
- ViETest::Log("Done\n");
- }
-
- AutoTestSleep(TEST_SPACING);
-
- // testing SetRenderStartImage(videoChannel, renderStartImage);
- {
- ViETest::Log("Testing SetRenderStartImage(int, ViEPicture)");
- // set render image, then stop capture and stop render to display it
- ViETest::Log("Stoping renderer, setting start image, then "
- "restarting");
- EXPECT_EQ(0, ptrViEFile->SetRenderStartImage(
- videoChannel, capturePicture));
- EXPECT_EQ(0, ptrViECapture->StopCapture(captureId));
- EXPECT_EQ(0, ptrViERender->StopRender(videoChannel));
-
- ViETest::Log("Render start image should be displayed.");
- AutoTestSleep(RENDER_TIMEOUT);
-
- // restarting capture and render
- EXPECT_EQ(0, ptrViECapture->StartCapture(captureId));
- EXPECT_EQ(0, ptrViERender->StartRender(videoChannel));
- ViETest::Log("Done\n");
- }
-
- AutoTestSleep(TEST_SPACING);
-
- // testing SetRenderTimeoutImage(videoChannel, renderTimeoutFile,
- // RENDER_TIMEOUT);
- {
- ViETest::Log("Testing SetRenderTimeoutImage(int, char*)");
- ViETest::Log("Stopping capture device to induce timeout of %d ms",
- RENDER_TIMEOUT);
- EXPECT_EQ(0, ptrViEFile->SetRenderTimeoutImage(
- videoChannel, renderTimeoutFile.c_str(), RENDER_TIMEOUT));
-
- // now stop sending frames to the remote renderer and wait for
- // timeout
- EXPECT_EQ(0, ptrViECapture->StopCapture(captureId));
- AutoTestSleep(RENDER_TIMEOUT);
- ViETest::Log("Timeout image should be displayed now for %d ms",
- RENDER_TIMEOUT * 2);
- AutoTestSleep(RENDER_TIMEOUT * 2);
-
- // restart the capture device to undo the timeout
- EXPECT_EQ(0, ptrViECapture->StartCapture(captureId));
- ViETest::Log("Restarting capture device");
- AutoTestSleep(RENDER_TIMEOUT);
- ViETest::Log("Done\n");
- }
-
- AutoTestSleep(TEST_SPACING);
-
- // Need to create a ViEPicture object to pass into this function.
- // SetRenderTimeoutImage(videoChannel, renderTimeoutFile,
- // RENDER_TIMEOUT);
- {
- ViETest::Log("Testing SetRenderTimeoutImage(int, ViEPicture)");
- ViETest::Log("Stopping capture device to induce timeout of %d",
- RENDER_TIMEOUT);
- EXPECT_EQ(0, ptrViEFile->SetRenderTimeoutImage(
- videoChannel, capturePicture, RENDER_TIMEOUT));
-
- // now stop sending frames to the remote renderer and wait for
- // timeout
- EXPECT_EQ(0, ptrViECapture->StopCapture(captureId));
- AutoTestSleep(RENDER_TIMEOUT);
- ViETest::Log("Timeout image should be displayed now for %d",
- RENDER_TIMEOUT * 2);
- AutoTestSleep(RENDER_TIMEOUT * 2);
-
- // restart the capture device to undo the timeout
- EXPECT_EQ(0, ptrViECapture->StartCapture(captureId));
- ViETest::Log("Restarting capture device");
- ViETest::Log("Done\n");
- }
-
- // testing DeregisterObserver
- {
- ViETest::Log("Deregistering file observer");
- // Should fail since we don't observe this file.
- EXPECT_NE(0, ptrViEFile->DeregisterObserver(fileId, fileObserver));
- }
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
-
- EXPECT_EQ(0, ptrViEBase->DisconnectAudioChannel(videoChannel));
- EXPECT_EQ(0, ptrViEBase->SetVoiceEngine(NULL));
- EXPECT_EQ(0, ptrVEBase->DeleteChannel(audioChannel));
- EXPECT_EQ(0, ptrVEBase->Release());
- EXPECT_EQ(0, ptrVECodec->Release());
- EXPECT_TRUE(webrtc::VoiceEngine::Delete(ptrVEEngine));
-
- EXPECT_EQ(0, ptrViEBase->StopReceive(videoChannel));
- EXPECT_EQ(0, ptrViEBase->StopSend(videoChannel));
- EXPECT_EQ(0, ptrViERender->StopRender(videoChannel));
- EXPECT_EQ(0, ptrViERender->RemoveRenderer(captureId));
- EXPECT_EQ(0, ptrViERender->RemoveRenderer(videoChannel));
- EXPECT_EQ(0, ptrViECapture->DisconnectCaptureDevice(videoChannel));
- EXPECT_EQ(0, ptrViEFile->FreePicture(capturePicture));
- EXPECT_EQ(0, ptrViEFile->FreePicture(renderPicture));
- EXPECT_EQ(0, ptrViEFile->FreePicture(renderTimeoutPicture));
- EXPECT_EQ(0, ptrViEBase->DeleteChannel(videoChannel));
-
- EXPECT_EQ(0, ptrViEFile->Release());
- }
-#endif
-}
-
-void ViEAutoTest::ViEFileExtendedTest()
-{
-}
-
-void ViEAutoTest::ViEFileAPITest()
-{
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_image_process.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_image_process.cc
deleted file mode 100644
index 887fe6e..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_image_process.cc
+++ /dev/null
@@ -1,240 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest_image_process.cc
-//
-
-// Settings
-#include "vie_autotest_defines.h"
-#include "vie_autotest.h"
-#include "engine_configurations.h"
-
-#include "tb_interfaces.h"
-#include "tb_video_channel.h"
-#include "tb_capture_device.h"
-
-class MyEffectFilter: public webrtc::ViEEffectFilter
-{
-public:
- MyEffectFilter() {}
-
- ~MyEffectFilter() {}
-
- virtual int Transform(int size, unsigned char* frameBuffer,
- unsigned int timeStamp90KHz, unsigned int width,
- unsigned int height)
- {
- // Black and white
- memset(frameBuffer + (2 * size) / 3, 0x7f, size / 3);
- return 0;
- }
-};
-
-void ViEAutoTest::ViEImageProcessStandardTest()
-{
- //***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- //***************************************************************
- int rtpPort = 6000;
- // Create VIE
- TbInterfaces ViE("ViEImageProcessAPITest");
- // Create a video channel
- TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
- // Create a capture device
- TbCaptureDevice tbCapture(ViE);
-
- tbCapture.ConnectTo(tbChannel.videoChannel);
- tbChannel.StartReceive(rtpPort);
- tbChannel.StartSend(rtpPort);
-
- MyEffectFilter effectFilter;
-
- RenderCaptureDeviceAndOutputStream(&ViE, &tbChannel, &tbCapture);
-
- ViETest::Log("Capture device is renderered in Window 1");
- ViETest::Log("Remote stream is renderered in Window 2");
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- //***************************************************************
- // Engine ready. Begin testing class
- //***************************************************************
-
-
- EXPECT_EQ(0, ViE.image_process->RegisterCaptureEffectFilter(
- tbCapture.captureId, effectFilter));
-
- ViETest::Log("Black and white filter registered for capture device, "
- "affects both windows");
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- EXPECT_EQ(0, ViE.image_process->DeregisterCaptureEffectFilter(
- tbCapture.captureId));
-
- EXPECT_EQ(0, ViE.image_process->RegisterRenderEffectFilter(
- tbChannel.videoChannel, effectFilter));
-
- ViETest::Log("Remove capture effect filter, adding filter for incoming "
- "stream");
- ViETest::Log("Only Window 2 should be black and white");
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- EXPECT_EQ(0, ViE.render->StopRender(tbCapture.captureId));
- EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
-
- int rtpPort2 = rtpPort + 100;
- // Create a video channel
- TbVideoChannel tbChannel2(ViE, webrtc::kVideoCodecVP8);
-
- tbCapture.ConnectTo(tbChannel2.videoChannel);
- tbChannel2.StartReceive(rtpPort2);
- tbChannel2.StartSend(rtpPort2);
-
- EXPECT_EQ(0, ViE.render->AddRenderer(
- tbChannel2.videoChannel, _window1, 1, 0.0, 0.0, 1.0, 1.0));
- EXPECT_EQ(0, ViE.render->StartRender(tbChannel2.videoChannel));
- EXPECT_EQ(0, ViE.image_process->DeregisterRenderEffectFilter(
- tbChannel.videoChannel));
-
- ViETest::Log("Local renderer removed, added new channel and rendering in "
- "Window1.");
-
- EXPECT_EQ(0, ViE.image_process->RegisterCaptureEffectFilter(
- tbCapture.captureId, effectFilter));
-
- ViETest::Log("Black and white filter registered for capture device, "
- "affects both windows");
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- EXPECT_EQ(0, ViE.image_process->DeregisterCaptureEffectFilter(
- tbCapture.captureId));
-
- EXPECT_EQ(0, ViE.image_process->RegisterSendEffectFilter(
- tbChannel.videoChannel, effectFilter));
-
- ViETest::Log("Capture filter removed.");
- ViETest::Log("Black and white filter registered for one channel, Window2 "
- "should be black and white");
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- EXPECT_EQ(0, ViE.image_process->DeregisterSendEffectFilter(
- tbChannel.videoChannel));
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
-}
-
-void ViEAutoTest::ViEImageProcessExtendedTest()
-{
- ViEImageProcessStandardTest();
-}
-
-void ViEAutoTest::ViEImageProcessAPITest()
-{
- TbInterfaces ViE("ViEImageProcessAPITest");
- TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
- TbCaptureDevice tbCapture(ViE);
-
- tbCapture.ConnectTo(tbChannel.videoChannel);
-
- MyEffectFilter effectFilter;
-
- //
- // Capture effect filter
- //
- // Add effect filter
- EXPECT_EQ(0, ViE.image_process->RegisterCaptureEffectFilter(
- tbCapture.captureId, effectFilter));
- // Add again -> error
- EXPECT_NE(0, ViE.image_process->RegisterCaptureEffectFilter(
- tbCapture.captureId, effectFilter));
- EXPECT_EQ(0, ViE.image_process->DeregisterCaptureEffectFilter(
- tbCapture.captureId));
-
- // Double deregister
- EXPECT_NE(0, ViE.image_process->DeregisterCaptureEffectFilter(
- tbCapture.captureId));
- // Non-existing capture device
- EXPECT_NE(0, ViE.image_process->RegisterCaptureEffectFilter(
- tbChannel.videoChannel, effectFilter));
-
- //
- // Render effect filter
- //
- EXPECT_EQ(0, ViE.image_process->RegisterRenderEffectFilter(
- tbChannel.videoChannel, effectFilter));
- EXPECT_NE(0, ViE.image_process->RegisterRenderEffectFilter(
- tbChannel.videoChannel, effectFilter));
- EXPECT_EQ(0, ViE.image_process->DeregisterRenderEffectFilter(
- tbChannel.videoChannel));
- EXPECT_NE(0, ViE.image_process->DeregisterRenderEffectFilter(
- tbChannel.videoChannel));
-
- // Non-existing channel id
- EXPECT_NE(0, ViE.image_process->RegisterRenderEffectFilter(
- tbCapture.captureId, effectFilter));
-
- //
- // Send effect filter
- //
- EXPECT_EQ(0, ViE.image_process->RegisterSendEffectFilter(
- tbChannel.videoChannel, effectFilter));
- EXPECT_NE(0, ViE.image_process->RegisterSendEffectFilter(
- tbChannel.videoChannel, effectFilter));
- EXPECT_EQ(0, ViE.image_process->DeregisterSendEffectFilter(
- tbChannel.videoChannel));
- EXPECT_NE(0, ViE.image_process->DeregisterSendEffectFilter(
- tbChannel.videoChannel));
- EXPECT_NE(0, ViE.image_process->RegisterSendEffectFilter(
- tbCapture.captureId, effectFilter));
-
- //
- // Denoising
- //
- EXPECT_EQ(0, ViE.image_process->EnableDenoising(tbCapture.captureId, true));
- EXPECT_NE(0, ViE.image_process->EnableDenoising(tbCapture.captureId, true));
- EXPECT_EQ(0, ViE.image_process->EnableDenoising(
- tbCapture.captureId, false));
- EXPECT_NE(0, ViE.image_process->EnableDenoising(
- tbCapture.captureId, false));
- EXPECT_NE(0, ViE.image_process->EnableDenoising(
- tbChannel.videoChannel, true));
-
- //
- // Deflickering
- //
- EXPECT_EQ(0, ViE.image_process->EnableDeflickering(
- tbCapture.captureId, true));
- EXPECT_NE(0, ViE.image_process->EnableDeflickering(
- tbCapture.captureId, true));
- EXPECT_EQ(0, ViE.image_process->EnableDeflickering(
- tbCapture.captureId, false));
- EXPECT_NE(0, ViE.image_process->EnableDeflickering(
- tbCapture.captureId, false));
- EXPECT_NE(0, ViE.image_process->EnableDeflickering(
- tbChannel.videoChannel, true));
-
- //
- // Color enhancement
- //
- EXPECT_EQ(0, ViE.image_process->EnableColorEnhancement(
- tbChannel.videoChannel, false));
- EXPECT_EQ(0, ViE.image_process->EnableColorEnhancement(
- tbChannel.videoChannel, true));
- EXPECT_NE(0, ViE.image_process->EnableColorEnhancement(
- tbChannel.videoChannel, true));
- EXPECT_EQ(0, ViE.image_process->EnableColorEnhancement(
- tbChannel.videoChannel, false));
- EXPECT_NE(0, ViE.image_process->EnableColorEnhancement(
- tbChannel.videoChannel, false));
- EXPECT_NE(0, ViE.image_process->EnableColorEnhancement(
- tbCapture.captureId, true));
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_linux.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_linux.cc
deleted file mode 100644
index aca96bd..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_linux.cc
+++ /dev/null
@@ -1,142 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest_linux.cc
-//
-#include "vie_autotest_linux.h"
-
-#include <string>
-
-#include "vie_autotest_defines.h"
-#include "vie_autotest_main.h"
-#include "engine_configurations.h"
-#include "critical_section_wrapper.h"
-#include "thread_wrapper.h"
-
-ViEAutoTestWindowManager::ViEAutoTestWindowManager()
- : _hdsp1(NULL),
- _hdsp2(NULL) {
-}
-
-ViEAutoTestWindowManager::~ViEAutoTestWindowManager() {
- TerminateWindows();
-}
-
-void* ViEAutoTestWindowManager::GetWindow1() {
- return reinterpret_cast<void*>(_hwnd1);
-}
-
-void* ViEAutoTestWindowManager::GetWindow2() {
- return reinterpret_cast<void*>(_hwnd2);
-}
-
-int ViEAutoTestWindowManager::TerminateWindows() {
- if (_hdsp1) {
- ViEDestroyWindow(&_hwnd1, _hdsp1);
- _hdsp1 = NULL;
- }
- if (_hdsp2) {
- ViEDestroyWindow(&_hwnd2, _hdsp2);
- _hdsp2 = NULL;
- }
- return 0;
-}
-
-int ViEAutoTestWindowManager::CreateWindows(AutoTestRect window1Size,
- AutoTestRect window2Size,
- void* window1Title,
- void* window2Title) {
- ViECreateWindow(&_hwnd1, &_hdsp1, window1Size.origin.x,
- window1Size.origin.y, window1Size.size.width,
- window1Size.size.height,
- reinterpret_cast<char*>(window1Title));
- ViECreateWindow(&_hwnd2, &_hdsp2, window2Size.origin.x,
- window2Size.origin.y, window2Size.size.width,
- window2Size.size.height,
- reinterpret_cast<char*>(window2Title));
-
- return 0;
-}
-
-int ViEAutoTestWindowManager::ViECreateWindow(Window *out_window,
- Display **out_display, int x_pos,
- int y_pos, int width, int height,
- char* title) {
- Display* display = XOpenDisplay(NULL);
- if (display == NULL) {
- // There's no point to continue if this happens: nothing will work anyway.
- printf("Failed to connect to X server: X environment likely broken\n");
- exit(-1);
- }
-
- int screen = DefaultScreen(display);
-
- // Try to establish a 24-bit TrueColor display
- // (our environment must allow this).
- XVisualInfo visual_info;
- if (XMatchVisualInfo(display, screen, 24, TrueColor, &visual_info) == 0) {
- printf("Failed to establish 24-bit TrueColor in X environment.\n");
- exit(-1);
- }
-
- // Create suitable window attributes.
- XSetWindowAttributes window_attributes;
- window_attributes.colormap = XCreateColormap(
- display, DefaultRootWindow(display), visual_info.visual, AllocNone);
- window_attributes.event_mask = StructureNotifyMask | ExposureMask;
- window_attributes.background_pixel = 0;
- window_attributes.border_pixel = 0;
-
- unsigned long attribute_mask = CWBackPixel | CWBorderPixel | CWColormap |
- CWEventMask;
-
- Window _window = XCreateWindow(display, DefaultRootWindow(display), x_pos,
- y_pos, width, height, 0, visual_info.depth,
- InputOutput, visual_info.visual,
- attribute_mask, &window_attributes);
-
- // Set window name.
- XStoreName(display, _window, title);
- XSetIconName(display, _window, title);
-
- // Make x report events for mask.
- XSelectInput(display, _window, StructureNotifyMask);
-
- // Map the window to the display.
- XMapWindow(display, _window);
-
- // Wait for map event.
- XEvent event;
- do {
- XNextEvent(display, &event);
- } while (event.type != MapNotify || event.xmap.event != _window);
-
- *out_window = _window;
- *out_display = display;
- return 0;
-}
-
-int ViEAutoTestWindowManager::ViEDestroyWindow(Window *window,
- Display *display) {
- XUnmapWindow(display, *window);
- XDestroyWindow(display, *window);
- XSync(display, false);
- return 0;
-}
-
-bool ViEAutoTestWindowManager::SetTopmostWindow() {
- return 0;
-}
-
-int main(int argc, char** argv) {
- ViEAutoTestMain auto_test;
- return auto_test.RunTests(argc, argv);
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_loopback.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_loopback.cc
deleted file mode 100644
index d660745..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_loopback.cc
+++ /dev/null
@@ -1,663 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest_loopback.cc
-//
-// This code is also used as sample code for ViE 3.0
-//
-
-// ===================================================================
-//
-// BEGIN: VideoEngine 3.0 Sample Code
-//
-
-#include <iostream>
-
-#include "common_types.h"
-#include "tb_external_transport.h"
-#include "voe_base.h"
-#include "vie_autotest_defines.h"
-#include "vie_autotest.h"
-#include "vie_base.h"
-#include "vie_capture.h"
-#include "vie_codec.h"
-#include "vie_network.h"
-#include "vie_render.h"
-#include "vie_rtp_rtcp.h"
-
-#define VCM_RED_PAYLOAD_TYPE 96
-#define VCM_ULPFEC_PAYLOAD_TYPE 97
-
-int VideoEngineSampleCode(void* window1, void* window2)
-{
- //********************************************************
- // Begin create/initialize Video Engine for testing
- //********************************************************
-
- int error = 0;
-
- //
- // Create a VideoEngine instance
- //
- webrtc::VideoEngine* ptrViE = NULL;
- ptrViE = webrtc::VideoEngine::Create();
- if (ptrViE == NULL)
- {
- printf("ERROR in VideoEngine::Create\n");
- return -1;
- }
-
- error = ptrViE->SetTraceFilter(webrtc::kTraceAll);
- if (error == -1)
- {
- printf("ERROR in VideoEngine::SetTraceLevel\n");
- return -1;
- }
-
- std::string trace_file =
- ViETest::GetResultOutputPath() + "ViELoopbackCall_trace.txt";
- error = ptrViE->SetTraceFile(trace_file.c_str());
- if (error == -1)
- {
- printf("ERROR in VideoEngine::SetTraceFile\n");
- return -1;
- }
-
- //
- // Init VideoEngine and create a channel
- //
- webrtc::ViEBase* ptrViEBase = webrtc::ViEBase::GetInterface(ptrViE);
- if (ptrViEBase == NULL)
- {
- printf("ERROR in ViEBase::GetInterface\n");
- return -1;
- }
-
- error = ptrViEBase->Init();
- if (error == -1)
- {
- printf("ERROR in ViEBase::Init\n");
- return -1;
- }
-
- int videoChannel = -1;
- error = ptrViEBase->CreateChannel(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViEBase::CreateChannel\n");
- return -1;
- }
-
- //
- // List available capture devices, allocate and connect.
- //
- webrtc::ViECapture* ptrViECapture =
- webrtc::ViECapture::GetInterface(ptrViE);
- if (ptrViEBase == NULL)
- {
- printf("ERROR in ViECapture::GetInterface\n");
- return -1;
- }
-
- const unsigned int KMaxDeviceNameLength = 128;
- const unsigned int KMaxUniqueIdLength = 256;
- char deviceName[KMaxDeviceNameLength];
- memset(deviceName, 0, KMaxDeviceNameLength);
- char uniqueId[KMaxUniqueIdLength];
- memset(uniqueId, 0, KMaxUniqueIdLength);
-
- printf("Available capture devices:\n");
- int captureIdx = 0;
- for (captureIdx = 0;
- captureIdx < ptrViECapture->NumberOfCaptureDevices();
- captureIdx++)
- {
- memset(deviceName, 0, KMaxDeviceNameLength);
- memset(uniqueId, 0, KMaxUniqueIdLength);
-
- error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
- KMaxDeviceNameLength, uniqueId,
- KMaxUniqueIdLength);
- if (error == -1)
- {
- printf("ERROR in ViECapture::GetCaptureDevice\n");
- return -1;
- }
- printf("\t %d. %s\n", captureIdx + 1, deviceName);
- }
- printf("\nChoose capture device: ");
-#ifdef WEBRTC_ANDROID
- captureIdx = 0;
- printf("0\n");
-#else
- if (scanf("%d", &captureIdx) != 1)
- {
- printf("Error in scanf()\n");
- return -1;
- }
- getchar();
- captureIdx = captureIdx - 1; // Compensate for idx start at 1.
-#endif
- error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
- KMaxDeviceNameLength, uniqueId,
- KMaxUniqueIdLength);
- if (error == -1)
- {
- printf("ERROR in ViECapture::GetCaptureDevice\n");
- return -1;
- }
-
- int captureId = 0;
- error = ptrViECapture->AllocateCaptureDevice(uniqueId, KMaxUniqueIdLength,
- captureId);
- if (error == -1)
- {
- printf("ERROR in ViECapture::AllocateCaptureDevice\n");
- return -1;
- }
-
- error = ptrViECapture->ConnectCaptureDevice(captureId, videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViECapture::ConnectCaptureDevice\n");
- return -1;
- }
-
- error = ptrViECapture->StartCapture(captureId);
- if (error == -1)
- {
- printf("ERROR in ViECapture::StartCapture\n");
- return -1;
- }
-
- //
- // RTP/RTCP settings
- //
- webrtc::ViERTP_RTCP* ptrViERtpRtcp =
- webrtc::ViERTP_RTCP::GetInterface(ptrViE);
- if (ptrViERtpRtcp == NULL)
- {
- printf("ERROR in ViERTP_RTCP::GetInterface\n");
- return -1;
- }
-
- error = ptrViERtpRtcp->SetRTCPStatus(videoChannel,
- webrtc::kRtcpCompound_RFC4585);
- if (error == -1)
- {
- printf("ERROR in ViERTP_RTCP::SetRTCPStatus\n");
- return -1;
- }
-
- error = ptrViERtpRtcp->SetKeyFrameRequestMethod(
- videoChannel, webrtc::kViEKeyFrameRequestPliRtcp);
- if (error == -1)
- {
- printf("ERROR in ViERTP_RTCP::SetKeyFrameRequestMethod\n");
- return -1;
- }
-
- error = ptrViERtpRtcp->SetRembStatus(videoChannel, true, true);
- if (error == -1)
- {
- printf("ERROR in ViERTP_RTCP::SetTMMBRStatus\n");
- return -1;
- }
-
- //
- // Set up rendering
- //
- webrtc::ViERender* ptrViERender = webrtc::ViERender::GetInterface(ptrViE);
- if (ptrViERender == NULL)
- {
- printf("ERROR in ViERender::GetInterface\n");
- return -1;
- }
-
- error
- = ptrViERender->AddRenderer(captureId, window1, 0, 0.0, 0.0, 1.0, 1.0);
- if (error == -1)
- {
- printf("ERROR in ViERender::AddRenderer\n");
- return -1;
- }
-
- error = ptrViERender->StartRender(captureId);
- if (error == -1)
- {
- printf("ERROR in ViERender::StartRender\n");
- return -1;
- }
-
- error = ptrViERender->AddRenderer(videoChannel, window2, 1, 0.0, 0.0, 1.0,
- 1.0);
- if (error == -1)
- {
- printf("ERROR in ViERender::AddRenderer\n");
- return -1;
- }
-
- error = ptrViERender->StartRender(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViERender::StartRender\n");
- return -1;
- }
-
- //
- // Setup codecs
- //
- webrtc::ViECodec* ptrViECodec = webrtc::ViECodec::GetInterface(ptrViE);
- if (ptrViECodec == NULL)
- {
- printf("ERROR in ViECodec::GetInterface\n");
- return -1;
- }
-
- // Check available codecs and prepare receive codecs
- printf("\nAvailable codecs:\n");
- webrtc::VideoCodec videoCodec;
- memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
- int codecIdx = 0;
- for (codecIdx = 0; codecIdx < ptrViECodec->NumberOfCodecs(); codecIdx++)
- {
- error = ptrViECodec->GetCodec(codecIdx, videoCodec);
- if (error == -1)
- {
- printf("ERROR in ViECodec::GetCodec\n");
- return -1;
- }
-
- // try to keep the test frame size small when I420
- if (videoCodec.codecType == webrtc::kVideoCodecI420)
- {
- videoCodec.width = 176;
- videoCodec.height = 144;
- }
-
- error = ptrViECodec->SetReceiveCodec(videoChannel, videoCodec);
- if (error == -1)
- {
- printf("ERROR in ViECodec::SetReceiveCodec\n");
- return -1;
- }
- if (videoCodec.codecType != webrtc::kVideoCodecRED
- && videoCodec.codecType != webrtc::kVideoCodecULPFEC)
- {
- printf("\t %d. %s\n", codecIdx + 1, videoCodec.plName);
- }
- }
- printf("Choose codec: ");
-#ifdef WEBRTC_ANDROID
- codecIdx = 0;
- printf("0\n");
-#else
- if (scanf("%d", &codecIdx) != 1)
- {
- printf("Error in scanf()\n");
- return -1;
- }
- getchar();
- codecIdx = codecIdx - 1; // Compensate for idx start at 1.
-#endif
-
- error = ptrViECodec->GetCodec(codecIdx, videoCodec);
- if (error == -1)
- {
- printf("ERROR in ViECodec::GetCodec\n");
- return -1;
- }
-
- // Set spatial resolution option
- std::string str;
- std::cout << std::endl;
- std::cout << "Enter frame size option (default is CIF):" << std::endl;
- std::cout << "1. QCIF (176X144) " << std::endl;
- std::cout << "2. CIF (352X288) " << std::endl;
- std::cout << "3. VGA (640X480) " << std::endl;
- std::cout << "4. 4CIF (704X576) " << std::endl;
- std::cout << "5. WHD (1280X720) " << std::endl;
- std::getline(std::cin, str);
- int resolnOption = atoi(str.c_str());
- // Try to keep the test frame size small when I420
- if (videoCodec.codecType == webrtc::kVideoCodecI420)
- {
- resolnOption = 1;
- }
- switch (resolnOption)
- {
- case 1:
- videoCodec.width = 176;
- videoCodec.height = 144;
- break;
- case 2:
- videoCodec.width = 352;
- videoCodec.height = 288;
- break;
- case 3:
- videoCodec.width = 640;
- videoCodec.height = 480;
- break;
- case 4:
- videoCodec.width = 704;
- videoCodec.height = 576;
- break;
- case 5:
- videoCodec.width = 1280;
- videoCodec.height = 720;
- break;
- }
-
- // Set number of temporal layers.
- std::cout << std::endl;
- std::cout << "Choose number of temporal layers (1 to 4).";
- std::cout << "Press enter for default: \n";
- std::getline(std::cin, str);
- int numTemporalLayers = atoi(str.c_str());
- if(numTemporalLayers != 0)
- {
- videoCodec.codecSpecific.VP8.numberOfTemporalLayers = numTemporalLayers;
- }
-
- // Set start bit rate
- std::cout << std::endl;
- std::cout << "Choose start rate (in kbps). Press enter for default: ";
- std::getline(std::cin, str);
- int startRate = atoi(str.c_str());
- if(startRate != 0)
- {
- videoCodec.startBitrate=startRate;
- }
-
- error = ptrViECodec->SetSendCodec(videoChannel, videoCodec);
- if (error == -1)
- {
- printf("ERROR in ViECodec::SetSendCodec\n");
- return -1;
- }
-
- //
- // Choose Protection Mode
- //
- std::cout << std::endl;
- std::cout << "Enter Protection Method:" << std::endl;
- std::cout << "0. None" << std::endl;
- std::cout << "1. FEC" << std::endl;
- std::cout << "2. NACK" << std::endl;
- std::cout << "3. NACK+FEC" << std::endl;
- std::getline(std::cin, str);
- int protectionMethod = atoi(str.c_str());
- error = 0;
- bool temporalToggling = true;
- switch (protectionMethod)
- {
- case 0: // None: default is no protection
- break;
-
- case 1: // FEC only
- error = ptrViERtpRtcp->SetFECStatus(videoChannel,
- true,
- VCM_RED_PAYLOAD_TYPE,
- VCM_ULPFEC_PAYLOAD_TYPE);
- temporalToggling = false;
- break;
-
- case 2: // Nack only
- error = ptrViERtpRtcp->SetNACKStatus(videoChannel, true);
-
- break;
-
- case 3: // Hybrid NAck and FEC
- error = ptrViERtpRtcp->SetHybridNACKFECStatus(
- videoChannel,
- true,
- VCM_RED_PAYLOAD_TYPE,
- VCM_ULPFEC_PAYLOAD_TYPE);
- temporalToggling = false;
- break;
- }
-
- if (error < 0)
- {
- printf("ERROR in ViERTP_RTCP::SetProtectionStatus\n");
- }
-
-
- //
- // Address settings
- //
- webrtc::ViENetwork* ptrViENetwork =
- webrtc::ViENetwork::GetInterface(ptrViE);
- if (ptrViENetwork == NULL)
- {
- printf("ERROR in ViENetwork::GetInterface\n");
- return -1;
- }
-
- // Setting External transport
- TbExternalTransport extTransport(*(ptrViENetwork));
-
- int testMode = 0;
- std::cout << std::endl;
- std::cout << "Enter 1 for testing packet loss and delay with "
- "external transport: ";
- std::string test_str;
- std::getline(std::cin, test_str);
- testMode = atoi(test_str.c_str());
- if (testMode == 1)
- {
- // Avoid changing SSRC due to collision.
- error = ptrViERtpRtcp->SetLocalSSRC(videoChannel, 1);
-
- error = ptrViENetwork->RegisterSendTransport(videoChannel,
- extTransport);
- if (error == -1)
- {
- printf("ERROR in ViECodec::RegisterSendTransport \n");
- return -1;
- }
-
- // Set up packet loss value
- std::cout << "Enter Packet Loss Percentage" << std::endl;
- std::string rate_str;
- std::getline(std::cin, rate_str);
- int rate = atoi(rate_str.c_str());
- extTransport.SetPacketLoss(rate);
- if (rate) {
- temporalToggling = false;
- }
-
- // Set network delay value
- std::cout << "Enter network delay value [mS]" << std::endl;
- std::string delay_str;
- std::getline(std::cin, delay_str);
- int delayMs = atoi(delay_str.c_str());
- extTransport.SetNetworkDelay(delayMs);
-
- if (numTemporalLayers > 1 && temporalToggling) {
- extTransport.SetTemporalToggle(numTemporalLayers);
- } else {
- // Disabled
- extTransport.SetTemporalToggle(0);
- }
- }
- else
- {
- const char* ipAddress = "127.0.0.1";
- const unsigned short rtpPort = 6000;
- std::cout << std::endl;
- std::cout << "Using rtp port: " << rtpPort << std::endl;
- std::cout << std::endl;
- error = ptrViENetwork->SetLocalReceiver(videoChannel, rtpPort);
- if (error == -1)
- {
- printf("ERROR in ViENetwork::SetLocalReceiver\n");
- return -1;
- }
- error = ptrViENetwork->SetSendDestination(videoChannel,
- ipAddress, rtpPort);
- if (error == -1)
- {
- printf("ERROR in ViENetwork::SetSendDestination\n");
- return -1;
- }
- }
-
- error = ptrViEBase->StartReceive(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViENetwork::StartReceive\n");
- return -1;
- }
-
- error = ptrViEBase->StartSend(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViENetwork::StartSend\n");
- return -1;
- }
-
- //********************************************************
- // Engine started
- //********************************************************
-
-
- // Call started
- printf("\nLoopback call started\n\n");
- printf("Press enter to stop...");
- while ((getchar()) != '\n')
- ;
-
- //********************************************************
- // Testing finished. Tear down Video Engine
- //********************************************************
-
- error = ptrViEBase->StopReceive(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViEBase::StopReceive\n");
- return -1;
- }
-
- error = ptrViEBase->StopSend(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViEBase::StopSend\n");
- return -1;
- }
-
- error = ptrViERender->StopRender(captureId);
- if (error == -1)
- {
- printf("ERROR in ViERender::StopRender\n");
- return -1;
- }
-
- error = ptrViERender->RemoveRenderer(captureId);
- if (error == -1)
- {
- printf("ERROR in ViERender::RemoveRenderer\n");
- return -1;
- }
-
- error = ptrViERender->StopRender(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViERender::StopRender\n");
- return -1;
- }
-
- error = ptrViERender->RemoveRenderer(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViERender::RemoveRenderer\n");
- return -1;
- }
-
- error = ptrViECapture->StopCapture(captureId);
- if (error == -1)
- {
- printf("ERROR in ViECapture::StopCapture\n");
- return -1;
- }
-
- error = ptrViECapture->DisconnectCaptureDevice(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViECapture::DisconnectCaptureDevice\n");
- return -1;
- }
-
- error = ptrViECapture->ReleaseCaptureDevice(captureId);
- if (error == -1)
- {
- printf("ERROR in ViECapture::ReleaseCaptureDevice\n");
- return -1;
- }
-
- error = ptrViEBase->DeleteChannel(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViEBase::DeleteChannel\n");
- return -1;
- }
-
- int remainingInterfaces = 0;
- remainingInterfaces = ptrViECodec->Release();
- remainingInterfaces += ptrViECapture->Release();
- remainingInterfaces += ptrViERtpRtcp->Release();
- remainingInterfaces += ptrViERender->Release();
- remainingInterfaces += ptrViENetwork->Release();
- remainingInterfaces += ptrViEBase->Release();
- if (remainingInterfaces > 0)
- {
- printf("ERROR: Could not release all interfaces\n");
- return -1;
- }
-
- bool deleted = webrtc::VideoEngine::Delete(ptrViE);
- if (deleted == false)
- {
- printf("ERROR in VideoEngine::Delete\n");
- return -1;
- }
-
- return 0;
-
- //
- // END: VideoEngine 3.0 Sample Code
- //
- // ===================================================================
-}
-
-int ViEAutoTest::ViELoopbackCall()
-{
- ViETest::Log(" ");
- ViETest::Log("========================================");
- ViETest::Log(" ViE Autotest Loopback Call\n");
-
- if (VideoEngineSampleCode(_window1, _window2) == 0)
- {
- ViETest::Log(" ");
- ViETest::Log(" ViE Autotest Loopback Call Done");
- ViETest::Log("========================================");
- ViETest::Log(" ");
-
- return 0;
- }
-
- ViETest::Log(" ");
- ViETest::Log(" ViE Autotest Loopback Call Failed");
- ViETest::Log("========================================");
- ViETest::Log(" ");
- return 1;
-
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_main.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_main.cc
deleted file mode 100644
index b9f6958..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_main.cc
+++ /dev/null
@@ -1,188 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vie_autotest_main.h"
-
-#include "gflags/gflags.h"
-#include "gtest/gtest.h"
-#include "vie_autotest.h"
-#include "vie_autotest_window_manager_interface.h"
-#include "vie_window_creator.h"
-
-DEFINE_bool(automated, false, "Run Video engine tests in noninteractive mode.");
-
-static const std::string kStandardTest = "ViEStandardIntegrationTest";
-static const std::string kExtendedTest = "ViEExtendedIntegrationTest";
-static const std::string kApiTest = "ViEApiIntegrationTest";
-
-ViEAutoTestMain::ViEAutoTestMain() {
- index_to_test_method_map_[1] = "RunsBaseTestWithoutErrors";
- index_to_test_method_map_[2] = "RunsCaptureTestWithoutErrors";
- index_to_test_method_map_[3] = "RunsCodecTestWithoutErrors";
- index_to_test_method_map_[4] = "RunsEncryptionTestWithoutErrors";
- index_to_test_method_map_[5] = "RunsFileTestWithoutErrors";
- index_to_test_method_map_[6] = "RunsImageProcessTestWithoutErrors";
- index_to_test_method_map_[7] = "RunsNetworkTestWithoutErrors";
- index_to_test_method_map_[8] = "RunsRenderTestWithoutErrors";
- index_to_test_method_map_[9] = "RunsRtpRtcpTestWithoutErrors";
-}
-
-int ViEAutoTestMain::RunTests(int argc, char** argv) {
- // Initialize logging.
- ViETest::Init();
- // Initialize the testing framework.
- testing::InitGoogleTest(&argc, argv);
- // Parse remaining flags:
- google::ParseCommandLineFlags(&argc, &argv, true);
-
- int result;
- if (FLAGS_automated) {
- // Run in automated mode.
- result = RUN_ALL_TESTS();
- } else {
- // Run in interactive mode.
- result = RunInteractiveMode();
- }
-
- ViETest::Terminate();
- return result;
-}
-
-int ViEAutoTestMain::AskUserForTestCase() {
- int choice;
- std::string answer;
-
- do {
- ViETest::Log("\nSpecific tests:");
- ViETest::Log("\t 0. Go back to previous menu.");
-
- // Print all test method choices. Assumes that map sorts on its key.
- int last_valid_choice = 0;
- std::map<int, std::string>::const_iterator iterator;
- for (iterator = index_to_test_method_map_.begin();
- iterator != index_to_test_method_map_.end();
- ++iterator) {
- ViETest::Log("\t %d. %s", iterator->first, iterator->second.c_str());
- last_valid_choice = iterator->first;
- }
-
- ViETest::Log("Choose specific test:");
- choice = AskUserForNumber(0, last_valid_choice);
- } while (choice == kInvalidChoice);
-
- return choice;
-}
-
-int ViEAutoTestMain::AskUserForNumber(int min_allowed, int max_allowed) {
- int result;
- if (scanf("%d", &result) <= 0) {
- ViETest::Log("\nPlease enter a number instead, then hit enter.");
- getchar();
- return kInvalidChoice;
- }
- getchar(); // Consume enter key.
-
- if (result < min_allowed || result > max_allowed) {
- ViETest::Log("%d-%d are valid choices. Please try again.", min_allowed,
- max_allowed);
- return kInvalidChoice;
- }
-
- return result;
-}
-
-int ViEAutoTestMain::RunTestMatching(const std::string test_case,
- const std::string test_method) {
- testing::FLAGS_gtest_filter = test_case + "." + test_method;
- return RUN_ALL_TESTS();
-}
-
-int ViEAutoTestMain::RunSpecificTestCaseIn(const std::string test_case_name)
-{
- // If user says 0, it means don't run anything.
- int specific_choice = AskUserForTestCase();
- if (specific_choice != 0){
- return RunTestMatching(test_case_name,
- index_to_test_method_map_[specific_choice]);
- }
- return 0;
-}
-
-int ViEAutoTestMain::RunSpecialTestCase(int choice) {
- // 7-9 don't run in GTest and need to initialize by themselves.
- assert(choice >= 7 && choice <= 9);
-
- // Create the windows
- ViEWindowCreator windowCreator;
- ViEAutoTestWindowManagerInterface* windowManager =
- windowCreator.CreateTwoWindows();
-
- // Create the test cases
- ViEAutoTest vieAutoTest(windowManager->GetWindow1(),
- windowManager->GetWindow2());
-
- int errors = 0;
- switch (choice) {
- case 7: errors = vieAutoTest.ViELoopbackCall(); break;
- case 8: errors = vieAutoTest.ViECustomCall(); break;
- case 9: errors = vieAutoTest.ViESimulcastCall(); break;
- }
-
- windowCreator.TerminateWindows();
- return errors;
-}
-
-bool ViEAutoTestMain::RunInteractiveMode() {
- ViETest::Log(" ============================== ");
- ViETest::Log(" WebRTC ViE 3.x Autotest ");
- ViETest::Log(" ============================== \n");
-
- int choice = 0;
- int errors = 0;
- do {
- ViETest::Log("Test types: ");
- ViETest::Log("\t 0. Quit");
- ViETest::Log("\t 1. All standard tests (delivery test)");
- ViETest::Log("\t 2. All API tests");
- ViETest::Log("\t 3. All extended test");
- ViETest::Log("\t 4. Specific standard test");
- ViETest::Log("\t 5. Specific API test");
- ViETest::Log("\t 6. Specific extended test");
- ViETest::Log("\t 7. Simple loopback call");
- ViETest::Log("\t 8. Custom configure a call");
- ViETest::Log("\t 9. Simulcast in loopback");
- ViETest::Log("Select type of test:");
-
- choice = AskUserForNumber(0, 9);
- if (choice == kInvalidChoice) {
- continue;
- }
- switch (choice) {
- case 0: break;
- case 1: errors = RunTestMatching(kStandardTest, "*"); break;
- case 2: errors = RunTestMatching(kApiTest, "*"); break;
- case 3: errors = RunTestMatching(kExtendedTest, "*"); break;
- case 4: errors = RunSpecificTestCaseIn(kStandardTest); break;
- case 5: errors = RunSpecificTestCaseIn(kApiTest); break;
- case 6: errors = RunSpecificTestCaseIn(kExtendedTest); break;
- default: errors = RunSpecialTestCase(choice); break;
- }
- } while (choice != 0);
-
- if (errors) {
- ViETest::Log("Test done with errors, see ViEAutotestLog.txt for test "
- "result.\n");
- } else {
- ViETest::Log("Test done without errors, see ViEAutotestLog.txt for "
- "test result.\n");
- }
- return true;
-}
-
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_network.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_network.cc
deleted file mode 100644
index 2a84644..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_network.cc
+++ /dev/null
@@ -1,563 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest_network.cc
-//
-
-#include "vie_autotest_defines.h"
-#include "vie_autotest.h"
-#include "engine_configurations.h"
-
-#include "tb_capture_device.h"
-#include "tb_external_transport.h"
-#include "tb_interfaces.h"
-#include "tb_video_channel.h"
-
-#if defined(_WIN32)
-#include <qos.h>
-#elif defined(WEBRTC_MAC_INTEL)
-
-#endif
-
-class ViEAutoTestNetworkObserver: public webrtc::ViENetworkObserver
-{
-public:
- ViEAutoTestNetworkObserver()
- {
- }
- virtual ~ViEAutoTestNetworkObserver()
- {
- }
- virtual void OnPeriodicDeadOrAlive(const int videoChannel, const bool alive)
- {
- }
- virtual void PacketTimeout(const int videoChannel,
- const webrtc::ViEPacketTimeout timeout)
- {
- }
-};
-
-void ViEAutoTest::ViENetworkStandardTest()
-{
- TbInterfaces ViE("ViENetworkStandardTest"); // Create VIE
- TbCaptureDevice tbCapture(ViE);
- {
- // Create a video channel
- TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
- tbCapture.ConnectTo(tbChannel.videoChannel);
-
- RenderCaptureDeviceAndOutputStream(&ViE, &tbChannel, &tbCapture);
-
- // ***************************************************************
- // Engine ready. Begin testing class
- // ***************************************************************
-
- //
- // Transport
- //
- TbExternalTransport testTransport(*ViE.network);
- EXPECT_EQ(0, ViE.network->RegisterSendTransport(
- tbChannel.videoChannel, testTransport));
- EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
- tbChannel.videoChannel, webrtc::kViEKeyFrameRequestPliRtcp));
-
- ViETest::Log("Call started using external transport, video should "
- "see video in both windows\n");
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.network->DeregisterSendTransport(
- tbChannel.videoChannel));
-
- char myIpAddress[64];
- memset(myIpAddress, 0, 64);
- unsigned short rtpPort = 1234;
- memcpy(myIpAddress, "127.0.0.1", sizeof("127.0.0.1"));
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(
- tbChannel.videoChannel, rtpPort, rtpPort + 1, myIpAddress));
- EXPECT_EQ(0, ViE.network->SetSendDestination(
- tbChannel.videoChannel, myIpAddress, rtpPort,
- rtpPort + 1, rtpPort));
- EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
- ViETest::Log("Changed to WebRTC SocketTransport, you should still see "
- "video in both windows\n");
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- EXPECT_EQ(0, ViE.network->SetSourceFilter(
- tbChannel.videoChannel, rtpPort + 10, rtpPort + 11, myIpAddress));
- ViETest::Log("Added UDP port filter for incorrect ports, you should "
- "not see video in Window2");
- AutoTestSleep(2000);
- EXPECT_EQ(0, ViE.network->SetSourceFilter(
- tbChannel.videoChannel, rtpPort, rtpPort + 1, "123.1.1.0"));
- ViETest::Log("Added IP filter for incorrect IP address, you should not "
- "see video in Window2");
- AutoTestSleep(2000);
- EXPECT_EQ(0, ViE.network->SetSourceFilter(
- tbChannel.videoChannel, rtpPort, rtpPort + 1, myIpAddress));
- ViETest::Log("Added IP filter for this computer, you should see video "
- "in Window2 again\n");
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- tbCapture.Disconnect(tbChannel.videoChannel);
- }
-}
-
-void ViEAutoTest::ViENetworkExtendedTest()
-{
- //***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- //***************************************************************
-
- TbInterfaces ViE("ViENetworkExtendedTest"); // Create VIE
- TbCaptureDevice tbCapture(ViE);
- EXPECT_EQ(0, ViE.render->AddRenderer(
- tbCapture.captureId, _window1, 0, 0.0, 0.0, 1.0, 1.0));
- EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
-
- {
- //
- // ToS
- //
- // Create a video channel
- TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
- tbCapture.ConnectTo(tbChannel.videoChannel);
- const char* remoteIp = "192.168.200.1";
- int DSCP = 0;
- bool useSetSockOpt = false;
-
- webrtc::VideoCodec videoCodec;
- EXPECT_EQ(0, ViE.codec->GetSendCodec(
- tbChannel.videoChannel, videoCodec));
- videoCodec.maxFramerate = 5;
- EXPECT_EQ(0, ViE.codec->SetSendCodec(
- tbChannel.videoChannel, videoCodec));
-
- //***************************************************************
- // Engine ready. Begin testing class
- //***************************************************************
-
- char myIpAddress[64];
- memset(myIpAddress, 0, 64);
- unsigned short rtpPort = 9000;
- EXPECT_EQ(0, ViE.network->GetLocalIP(myIpAddress, false));
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(
- tbChannel.videoChannel, rtpPort, rtpPort + 1, myIpAddress));
- EXPECT_EQ(0, ViE.network->SetSendDestination(
- tbChannel.videoChannel, remoteIp, rtpPort, rtpPort + 1, rtpPort));
-
- // ToS
- int tos_result = ViE.network->SetSendToS(tbChannel.videoChannel, 2);
- EXPECT_EQ(0, tos_result);
- if (tos_result != 0)
- {
- ViETest::Log("ViESetSendToS error!.");
- ViETest::Log("You must be admin to run these tests.");
- ViETest::Log("On Win7 and late Vista, you need to right click the "
- "exe and choose");
- ViETest::Log("\"Run as administrator\"\n");
- getchar();
- }
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
-
- EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
- ViETest::Log("Use Wireshark to capture the outgoing video stream and "
- "verify ToS settings\n");
- ViETest::Log(" DSCP set to 0x%x\n", DSCP);
- AutoTestSleep(1000);
-
- EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 63));
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
- ViETest::Log(" DSCP set to 0x%x\n", DSCP);
- AutoTestSleep(1000);
-
- EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 0));
- EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 2, true));
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
- ViETest::Log(" DSCP set to 0x%x\n", DSCP);
- AutoTestSleep(1000);
-
- EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 63, true));
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
- ViETest::Log(" DSCP set to 0x%x\n", DSCP);
- AutoTestSleep(1000);
-
- tbCapture.Disconnect(tbChannel.videoChannel);
- }
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
-}
-
-void ViEAutoTest::ViENetworkAPITest()
-{
- //***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- //***************************************************************
-
- TbInterfaces ViE("ViENetworkAPITest"); // Create VIE
- {
- // Create a video channel
- TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
-
- //***************************************************************
- // Engine ready. Begin testing class
- //***************************************************************
-
- //
- // External transport
- //
- TbExternalTransport testTransport(*ViE.network);
- EXPECT_EQ(0, ViE.network->RegisterSendTransport(
- tbChannel.videoChannel, testTransport));
- EXPECT_NE(0, ViE.network->RegisterSendTransport(
- tbChannel.videoChannel, testTransport));
-
- unsigned char packet[1500];
- packet[0] = 0x80; // V=2, P=0, X=0, CC=0
- packet[1] = 0x78; // M=0, PT = 120 (VP8)
- EXPECT_NE(0, ViE.network->ReceivedRTPPacket(
- tbChannel.videoChannel, packet, 1500));
- EXPECT_NE(0, ViE.network->ReceivedRTCPPacket(
- tbChannel.videoChannel, packet, 1500));
- EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.network->ReceivedRTPPacket(
- tbChannel.videoChannel, packet, 1500));
- EXPECT_EQ(0, ViE.network->ReceivedRTCPPacket(
- tbChannel.videoChannel, packet, 1500));
- EXPECT_NE(0, ViE.network->ReceivedRTPPacket(
- tbChannel.videoChannel, packet, 11));
- EXPECT_NE(0, ViE.network->ReceivedRTPPacket(
- tbChannel.videoChannel, packet, 11));
- EXPECT_EQ(0, ViE.network->ReceivedRTPPacket(
- tbChannel.videoChannel, packet, 3000));
- EXPECT_EQ(0, ViE.network->ReceivedRTPPacket(
- tbChannel.videoChannel, packet, 3000));
- EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
- EXPECT_NE(0, ViE.network->DeregisterSendTransport(
- tbChannel.videoChannel)); // Sending
- EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.network->DeregisterSendTransport(
- tbChannel.videoChannel));
- EXPECT_NE(0, ViE.network->DeregisterSendTransport(
- tbChannel.videoChannel)); // Already deregistered
-
- //
- // Local receiver
- //
- // TODO (perkj) change when B 4239431 is fixed.
- /*error = ViE.ptrViENetwork->SetLocalReceiver(tbChannel.videoChannel,
- 1234, 1234, "127.0.0.1");
- numberOfErrors += ViETest::TestError(error == 0,
- "ERROR: %s at line %d",
- __FUNCTION__, __LINE__);*/
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(
- tbChannel.videoChannel, 1234, 1235, "127.0.0.1"));
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(
- tbChannel.videoChannel, 1234, 1235, "127.0.0.1"));
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(
- tbChannel.videoChannel, 1236, 1237, "127.0.0.1"));
-
- unsigned short rtpPort = 0;
- unsigned short rtcpPort = 0;
- char ipAddress[64];
- memset(ipAddress, 0, 64);
- EXPECT_EQ(0, ViE.network->GetLocalReceiver(
- tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
- EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
- EXPECT_NE(0, ViE.network->SetLocalReceiver(
- tbChannel.videoChannel, 1234, 1235, "127.0.0.1"));
- EXPECT_EQ(0, ViE.network->GetLocalReceiver(
- tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
- EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-
- //
- // Send destination
- //
- EXPECT_EQ(0, ViE.network->SetSendDestination(
- tbChannel.videoChannel, "127.0.0.1", 1234, 1235, 1234, 1235));
- EXPECT_EQ(0, ViE.network->SetSendDestination(
- tbChannel.videoChannel, "127.0.0.1", 1236, 1237, 1234, 1235));
-
- unsigned short sourceRtpPort = 0;
- unsigned short sourceRtcpPort = 0;
- EXPECT_EQ(0, ViE.network->GetSendDestination(
- tbChannel.videoChannel, ipAddress, rtpPort, rtcpPort,
- sourceRtpPort, sourceRtcpPort));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
- // Not allowed while sending
- EXPECT_NE(0, ViE.network->SetSendDestination(
- tbChannel.videoChannel, "127.0.0.1", 1234, 1235, 1234, 1235));
- EXPECT_EQ(kViENetworkAlreadySending, ViE.base->LastError());
-
- EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.network->SetSendDestination(
- tbChannel.videoChannel, "127.0.0.1", 1234, 1235, 1234, 1235));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.network->GetSendDestination(
- tbChannel.videoChannel, ipAddress, rtpPort, rtcpPort,
- sourceRtpPort, sourceRtcpPort));
- EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-
- //
- // Address information
- //
-
- // GetSourceInfo: Tested in functional test
- EXPECT_EQ(0, ViE.network->GetLocalIP(ipAddress, false));
-
- // TODO(unknown): IPv6
-
- //
- // Filter
- //
- EXPECT_NE(0, ViE.network->GetSourceFilter(
- tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
- EXPECT_EQ(0, ViE.network->SetSourceFilter(
- tbChannel.videoChannel, 1234, 1235, "10.10.10.10"));
- EXPECT_EQ(0, ViE.network->SetSourceFilter(
- tbChannel.videoChannel, 1236, 1237, "127.0.0.1"));
- EXPECT_EQ(0, ViE.network->GetSourceFilter(
- tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
- EXPECT_EQ(0, ViE.network->SetSourceFilter(
- tbChannel.videoChannel, 0, 0, NULL));
- EXPECT_NE(0, ViE.network->GetSourceFilter(
- tbChannel.videoChannel, rtpPort, rtcpPort, ipAddress));
- }
- {
- TbVideoChannel tbChannel(ViE); // Create a video channel
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(
- tbChannel.videoChannel, 1234));
-
- int DSCP = 0;
- bool useSetSockOpt = false;
- // SetSockOpt should work without a locally bind socket
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
- EXPECT_EQ(0, DSCP);
-
- // Invalid input
- EXPECT_NE(0, ViE.network->SetSendToS(tbChannel.videoChannel, -1, true));
-
- // Invalid input
- EXPECT_NE(0, ViE.network->SetSendToS(tbChannel.videoChannel, 64, true));
-
- // Valid
- EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 20, true));
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt));
-
- EXPECT_EQ(20, DSCP);
- EXPECT_TRUE(useSetSockOpt);
-
- // Disable
- EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 0, true));
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt));
- EXPECT_EQ(0, DSCP);
-
- char myIpAddress[64];
- memset(myIpAddress, 0, 64);
- // Get local ip to be able to set ToS withtou setSockOpt
- EXPECT_EQ(0, ViE.network->GetLocalIP(myIpAddress, false));
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(
- tbChannel.videoChannel, 1234, 1235, myIpAddress));
-
- // Invalid input
- EXPECT_NE(0, ViE.network->SetSendToS(
- tbChannel.videoChannel, -1, false));
- EXPECT_NE(0, ViE.network->SetSendToS(
- tbChannel.videoChannel, 64, false)); // Invalid input
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt)); // No ToS set
- EXPECT_EQ(0, DSCP);
- int tos_result = ViE.network->SetSendToS(
- tbChannel.videoChannel, 20, false); // Valid
- EXPECT_EQ(0, tos_result);
- if (tos_result != 0)
- {
- ViETest::Log("ViESetSendToS error!.");
- ViETest::Log("You must be admin to run these tests.");
- ViETest::Log("On Win7 and late Vista, you need to right click the "
- "exe and choose");
- ViETest::Log("\"Run as administrator\"\n");
- getchar();
- }
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt));
- EXPECT_EQ(20, DSCP);
-#ifdef _WIN32
- EXPECT_FALSE(useSetSockOpt);
-#else // useSetSockOpt is true on Linux and Mac
- EXPECT_TRUE(useSetSockOpt);
-#endif
- EXPECT_EQ(0, ViE.network->SetSendToS(tbChannel.videoChannel, 0, false));
- EXPECT_EQ(0, ViE.network->GetSendToS(
- tbChannel.videoChannel, DSCP, useSetSockOpt));
- EXPECT_EQ(0, DSCP);
- }
- {
- // From qos.h. (*) -> supported by ViE
- //
- // #define SERVICETYPE_NOTRAFFIC 0x00000000
- // #define SERVICETYPE_BESTEFFORT 0x00000001 (*)
- // #define SERVICETYPE_CONTROLLEDLOAD 0x00000002 (*)
- // #define SERVICETYPE_GUARANTEED 0x00000003 (*)
- // #define SERVICETYPE_NETWORK_UNAVAILABLE 0x00000004
- // #define SERVICETYPE_GENERAL_INFORMATION 0x00000005
- // #define SERVICETYPE_NOCHANGE 0x00000006
- // #define SERVICETYPE_NONCONFORMING 0x00000009
- // #define SERVICETYPE_NETWORK_CONTROL 0x0000000A
- // #define SERVICETYPE_QUALITATIVE 0x0000000D (*)
- //
- // #define SERVICE_BESTEFFORT 0x80010000
- // #define SERVICE_CONTROLLEDLOAD 0x80020000
- // #define SERVICE_GUARANTEED 0x80040000
- // #define SERVICE_QUALITATIVE 0x80200000
-
- TbVideoChannel tbChannel(ViE); // Create a video channel
-
-
-#if defined(_WIN32)
- // No socket
- EXPECT_NE(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, true, SERVICETYPE_BESTEFFORT));
-
- EXPECT_EQ(0, ViE.network->SetLocalReceiver(
- tbChannel.videoChannel, 1234));
-
- // Sender not initialized
- EXPECT_NE(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, true, SERVICETYPE_BESTEFFORT));
- EXPECT_EQ(0, ViE.network->SetSendDestination(
- tbChannel.videoChannel, "127.0.0.1", 12345));
-
- // Try to set all non-supported service types
- EXPECT_NE(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, true, SERVICETYPE_NOTRAFFIC));
- EXPECT_NE(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, true, SERVICETYPE_NETWORK_UNAVAILABLE));
- EXPECT_NE(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, true, SERVICETYPE_GENERAL_INFORMATION));
- EXPECT_NE(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, true, SERVICETYPE_NOCHANGE));
- EXPECT_NE(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, true, SERVICETYPE_NONCONFORMING));
- EXPECT_NE(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, true, SERVICETYPE_NOTRAFFIC));
- EXPECT_NE(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, true, SERVICETYPE_NETWORK_CONTROL));
- EXPECT_NE(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, true, SERVICE_BESTEFFORT));
- EXPECT_NE(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, true, SERVICE_CONTROLLEDLOAD));
- EXPECT_NE(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, true, SERVICE_GUARANTEED));
- EXPECT_NE(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, true, SERVICE_QUALITATIVE));
-
- // Loop through valid service settings
- bool enabled = false;
- int serviceType = 0;
- int overrideDSCP = 0;
-
- EXPECT_EQ(0, ViE.network->GetSendGQoS(
- tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
- EXPECT_FALSE(enabled);
- EXPECT_EQ(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, true, SERVICETYPE_BESTEFFORT));
- EXPECT_EQ(0, ViE.network->GetSendGQoS(
- tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
- EXPECT_TRUE(enabled);
- EXPECT_EQ(SERVICETYPE_BESTEFFORT, serviceType);
- EXPECT_FALSE(overrideDSCP);
-
- EXPECT_EQ(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, true, SERVICETYPE_CONTROLLEDLOAD));
- EXPECT_EQ(0, ViE.network->GetSendGQoS(
- tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
- EXPECT_TRUE(enabled);
- EXPECT_EQ(SERVICETYPE_CONTROLLEDLOAD, serviceType);
- EXPECT_FALSE(overrideDSCP);
-
- EXPECT_EQ(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, true, SERVICETYPE_GUARANTEED));
- EXPECT_EQ(0, ViE.network->GetSendGQoS(
- tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
- EXPECT_TRUE(enabled);
- EXPECT_EQ(SERVICETYPE_GUARANTEED, serviceType);
- EXPECT_FALSE(overrideDSCP);
-
- EXPECT_EQ(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, true, SERVICETYPE_QUALITATIVE));
- EXPECT_EQ(0, ViE.network->GetSendGQoS(
- tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
- EXPECT_TRUE(enabled);
- EXPECT_EQ(SERVICETYPE_QUALITATIVE, serviceType);
- EXPECT_FALSE(overrideDSCP);
-
- EXPECT_EQ(0, ViE.network->SetSendGQoS(
- tbChannel.videoChannel, false, SERVICETYPE_QUALITATIVE));
- EXPECT_EQ(0, ViE.network->GetSendGQoS(
- tbChannel.videoChannel, enabled, serviceType, overrideDSCP));
- EXPECT_FALSE(enabled);
-#endif
- }
- {
- //
- // MTU and packet burst
- //
- // Create a video channel
- TbVideoChannel tbChannel(ViE);
- // Invalid input
- EXPECT_NE(0, ViE.network->SetMTU(tbChannel.videoChannel, 1600));
- // Valid input
- EXPECT_EQ(0, ViE.network->SetMTU(tbChannel.videoChannel, 800));
-
- //
- // Observer and timeout
- //
- ViEAutoTestNetworkObserver vieTestObserver;
- EXPECT_EQ(0, ViE.network->RegisterObserver(
- tbChannel.videoChannel, vieTestObserver));
- EXPECT_NE(0, ViE.network->RegisterObserver(
- tbChannel.videoChannel, vieTestObserver));
- EXPECT_EQ(0, ViE.network->SetPeriodicDeadOrAliveStatus(
- tbChannel.videoChannel, true)); // No observer
- EXPECT_EQ(0, ViE.network->DeregisterObserver(tbChannel.videoChannel));
-
- EXPECT_NE(0, ViE.network->DeregisterObserver(tbChannel.videoChannel));
- EXPECT_NE(0, ViE.network->SetPeriodicDeadOrAliveStatus(
- tbChannel.videoChannel, true)); // No observer
-
- // Packet timout notification
- EXPECT_EQ(0, ViE.network->SetPacketTimeoutNotification(
- tbChannel.videoChannel, true, 10));
- }
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_render.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_render.cc
deleted file mode 100644
index 2a73a6c..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_render.cc
+++ /dev/null
@@ -1,296 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest_render.cc
-//
-
-#include "vie_autotest_defines.h"
-#include "vie_autotest.h"
-#include "engine_configurations.h"
-
-#include "video_render.h"
-
-#include "tb_interfaces.h"
-#include "tb_video_channel.h"
-#include "tb_capture_device.h"
-
-#if defined(WIN32)
-#include <windows.h>
-#include <ddraw.h>
-#include <tchar.h>
-#elif defined(WEBRTC_LINUX)
- //From windgi.h
- #undef RGB
- #define RGB(r,g,b) ((unsigned long)(((unsigned char)(r)|((unsigned short)((unsigned char)(g))<<8))|(((unsigned long)(unsigned char)(b))<<16)))
- //From ddraw.h
-/* typedef struct _DDCOLORKEY
- {
- DWORD dwColorSpaceLowValue; // low boundary of color space that is to
- DWORD dwColorSpaceHighValue; // high boundary of color space that is
- } DDCOLORKEY;*/
-#elif defined(WEBRTC_MAC)
-#endif
-
-class ViEAutoTestExternalRenderer: public webrtc::ExternalRenderer
-{
-public:
- ViEAutoTestExternalRenderer() :
- _width(0),
- _height(0)
- {
- }
- virtual int FrameSizeChange(unsigned int width, unsigned int height,
- unsigned int numberOfStreams)
- {
- _width = width;
- _height = height;
- return 0;
- }
-
- virtual int DeliverFrame(unsigned char* buffer, int bufferSize,
- uint32_t time_stamp,
- int64_t render_time)
- {
- if (bufferSize != _width * _height * 3 / 2)
- {
- ViETest::Log("incorrect render buffer received, of length = %d\n",
- bufferSize);
- return 0;
- }
- ViETest::Log("callback DeliverFrame is good\n");
- return 0;
- }
-
-public:
- virtual ~ViEAutoTestExternalRenderer()
- {
- }
-private:
- int _width, _height;
-};
-
-void ViEAutoTest::ViERenderStandardTest()
-{
- //***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- //***************************************************************
- int rtpPort = 6000;
-
- TbInterfaces ViE("ViERenderStandardTest");
-
- // Create a video channel
- TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
- TbCaptureDevice tbCapture(ViE); // Create a capture device
- tbCapture.ConnectTo(tbChannel.videoChannel);
- tbChannel.StartReceive(rtpPort);
- tbChannel.StartSend(rtpPort);
-
- EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm1));
- EXPECT_EQ(0, ViE.render->AddRenderer(
- tbCapture.captureId, _window1, 0, 0.0, 0.0, 1.0, 1.0));
- EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
- EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm2));
- EXPECT_EQ(0, ViE.render->AddRenderer(
- tbChannel.videoChannel, _window2, 1, 0.0, 0.0, 1.0, 1.0));
- EXPECT_EQ(0, ViE.render->StartRender(tbChannel.videoChannel));
-
- ViETest::Log("\nCapture device is renderered in Window 1");
- ViETest::Log("Remote stream is renderered in Window 2");
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- EXPECT_EQ(0, ViE.render->StopRender(tbCapture.captureId));
- EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
-
- // PIP and full screen rendering is not supported on Android
-#ifndef WEBRTC_ANDROID
- EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm1));
- EXPECT_EQ(0, ViE.render->AddRenderer(
- tbCapture.captureId, _window2, 0, 0.75, 0.75, 1.0, 1.0));
- EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
-
- ViETest::Log("\nCapture device is now rendered in Window 2, PiP.");
- ViETest::Log("Switching to full screen rendering in %d seconds.\n",
- KAutoTestSleepTimeMs / 1000);
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
- EXPECT_EQ(0, ViE.render->RemoveRenderer(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm2));
-
- // Destroy render module and create new in full screen mode
- webrtc::VideoRender::DestroyVideoRender(_vrm1);
- _vrm1 = NULL;
- _vrm1 = webrtc::VideoRender::CreateVideoRender(
- 4563, _window1, true, _renderType);
- EXPECT_TRUE(_vrm1 != NULL);
-
- EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm1));
- EXPECT_EQ(0, ViE.render->AddRenderer(
- tbCapture.captureId, _window1, 0, 0.75f, 0.75f, 1.0f, 1.0f));
- EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
- EXPECT_EQ(0, ViE.render->AddRenderer(
- tbChannel.videoChannel, _window1, 1, 0.0, 0.0, 1.0, 1.0));
- EXPECT_EQ(0, ViE.render->StartRender(tbChannel.videoChannel));
-
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
-
- EXPECT_EQ(0, ViE.render->RemoveRenderer(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm1));
-
- // Destroy full screen render module and create new in normal mode
- webrtc::VideoRender::DestroyVideoRender(_vrm1);
- _vrm1 = NULL;
- _vrm1 = webrtc::VideoRender::CreateVideoRender(
- 4561, _window1, false, _renderType);
- EXPECT_TRUE(_vrm1 != NULL);
-#endif
-
- //***************************************************************
- // Engine ready. Begin testing class
- //***************************************************************
-
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
- tbCapture.Disconnect(tbChannel.videoChannel);
-}
-
-void ViEAutoTest::ViERenderExtendedTest()
-{
- int rtpPort = 6000;
-
- TbInterfaces ViE("ViERenderExtendedTest");
-
- // Create a video channel
- TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
- TbCaptureDevice tbCapture(ViE); // Create a capture device
- tbCapture.ConnectTo(tbChannel.videoChannel);
- tbChannel.StartReceive(rtpPort);
- tbChannel.StartSend(rtpPort);
-
- EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm1));
- EXPECT_EQ(0, ViE.render->AddRenderer(
- tbCapture.captureId, _window1, 0, 0.0, 0.0, 1.0, 1.0));
- EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
- EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm2));
- EXPECT_EQ(0, ViE.render->AddRenderer(
- tbChannel.videoChannel, _window2, 1, 0.0, 0.0, 1.0, 1.0));
- EXPECT_EQ(0, ViE.render->StartRender(tbChannel.videoChannel));
-
- ViETest::Log("\nCapture device is renderered in Window 1");
- ViETest::Log("Remote stream is renderered in Window 2");
- AutoTestSleep(KAutoTestSleepTimeMs);
-
-#ifdef _WIN32
- ViETest::Log("\nConfiguring Window2");
- ViETest::Log("you will see video only in first quadrant");
- EXPECT_EQ(0, ViE.render->ConfigureRender(
- tbChannel.videoChannel, 0, 0.0f, 0.0f, 0.5f, 0.5f));
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- ViETest::Log("you will see video only in fourth quadrant");
- EXPECT_EQ(0, ViE.render->ConfigureRender(
- tbChannel.videoChannel, 0, 0.5f, 0.5f, 1.0f, 1.0f));
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- ViETest::Log("normal video on Window2");
- EXPECT_EQ(0, ViE.render->ConfigureRender(
- tbChannel.videoChannel, 0, 0.0f, 0.0f, 1.0f, 1.0f));
- AutoTestSleep(KAutoTestSleepTimeMs);
-#endif
-
- ViETest::Log("Mirroring Local Preview (Window1) Left-Right");
- EXPECT_EQ(0, ViE.render->MirrorRenderStream(
- tbCapture.captureId, true, false, true));
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- ViETest::Log("\nMirroring Local Preview (Window1) Left-Right and Up-Down");
- EXPECT_EQ(0, ViE.render->MirrorRenderStream(
- tbCapture.captureId, true, true, true));
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- ViETest::Log("\nMirroring Remote Window(Window2) Up-Down");
- EXPECT_EQ(0, ViE.render->MirrorRenderStream(
- tbChannel.videoChannel, true, true, false));
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- ViETest::Log("Disabling Mirroing on Window1 and Window2");
- EXPECT_EQ(0, ViE.render->MirrorRenderStream(
- tbCapture.captureId, false, false, false));
- AutoTestSleep(KAutoTestSleepTimeMs);
- EXPECT_EQ(0, ViE.render->MirrorRenderStream(
- tbChannel.videoChannel, false, false, false));
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- ViETest::Log("\nEnabling Full Screen render in 5 sec");
-
- EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
- EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm1));
- EXPECT_EQ(0, ViE.render->RemoveRenderer(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm2));
-
- // Destroy render module and create new in full screen mode
- webrtc::VideoRender::DestroyVideoRender(_vrm1);
- _vrm1 = NULL;
- _vrm1 = webrtc::VideoRender::CreateVideoRender(
- 4563, _window1, true, _renderType);
- EXPECT_TRUE(_vrm1 != NULL);
-
- EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm1));
- EXPECT_EQ(0, ViE.render->AddRenderer(
- tbCapture.captureId, _window1, 0, 0.0f, 0.0f, 1.0f, 1.0f));
- EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- ViETest::Log("\nStop renderer");
- EXPECT_EQ(0, ViE.render->StopRender(tbCapture.captureId));
- ViETest::Log("\nRemove renderer");
- EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
-
- EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm1));
-
- // Destroy full screen render module and create new for external rendering
- webrtc::VideoRender::DestroyVideoRender(_vrm1);
- _vrm1 = NULL;
- _vrm1 = webrtc::VideoRender::CreateVideoRender(4564, NULL, false,
- _renderType);
- EXPECT_TRUE(_vrm1 != NULL);
-
- EXPECT_EQ(0, ViE.render->RegisterVideoRenderModule(*_vrm1));
-
- ViETest::Log("\nExternal Render Test");
- ViEAutoTestExternalRenderer externalRenderObj;
- EXPECT_EQ(0, ViE.render->AddRenderer(
- tbCapture.captureId, webrtc::kVideoI420, &externalRenderObj));
- EXPECT_EQ(0, ViE.render->StartRender(tbCapture.captureId));
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- EXPECT_EQ(0, ViE.render->StopRender(tbCapture.captureId));
- EXPECT_EQ(0, ViE.render->RemoveRenderer(tbCapture.captureId));
- EXPECT_EQ(0, ViE.render->DeRegisterVideoRenderModule(*_vrm1));
-
- // Destroy render module for external rendering and create new in normal
- // mode
- webrtc::VideoRender::DestroyVideoRender(_vrm1);
- _vrm1 = NULL;
- _vrm1 = webrtc::VideoRender::CreateVideoRender(
- 4561, _window1, false, _renderType);
- EXPECT_TRUE(_vrm1 != NULL);
- tbCapture.Disconnect(tbChannel.videoChannel);
-}
-
-void ViEAutoTest::ViERenderAPITest()
-{
- // TODO(unknown): add the real tests cases
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
deleted file mode 100644
index 4ea8383..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_rtp_rtcp.cc
+++ /dev/null
@@ -1,682 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest_rtp_rtcp.cc
-//
-#include <iostream>
-
-#include "engine_configurations.h"
-#include "tb_capture_device.h"
-#include "tb_external_transport.h"
-#include "tb_interfaces.h"
-#include "tb_video_channel.h"
-#include "testsupport/fileutils.h"
-#include "vie_autotest.h"
-#include "vie_autotest_defines.h"
-
-class ViERtpObserver: public webrtc::ViERTPObserver
-{
-public:
- ViERtpObserver()
- {
- }
- virtual ~ViERtpObserver()
- {
- }
-
- virtual void IncomingSSRCChanged(const int videoChannel,
- const unsigned int SSRC)
- {
- }
- virtual void IncomingCSRCChanged(const int videoChannel,
- const unsigned int CSRC, const bool added)
- {
- }
-};
-
-class ViERtcpObserver: public webrtc::ViERTCPObserver
-{
-public:
- int _channel;
- unsigned char _subType;
- unsigned int _name;
- char* _data;
- unsigned short _dataLength;
-
- ViERtcpObserver() :
- _channel(-1),
- _subType(0),
- _name(-1),
- _data(NULL),
- _dataLength(0)
- {
- }
- ~ViERtcpObserver()
- {
- if (_data)
- {
- delete[] _data;
- }
- }
- virtual void OnApplicationDataReceived(
- const int videoChannel, const unsigned char subType,
- const unsigned int name, const char* data,
- const unsigned short dataLengthInBytes)
- {
- _channel = videoChannel;
- _subType = subType;
- _name = name;
- if (dataLengthInBytes > _dataLength)
- {
- delete[] _data;
- _data = NULL;
- }
- if (_data == NULL)
- {
- _data = new char[dataLengthInBytes];
- }
- memcpy(_data, data, dataLengthInBytes);
- _dataLength = dataLengthInBytes;
- }
-};
-
-void ViEAutoTest::ViERtpRtcpStandardTest()
-{
- // ***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- // ***************************************************************
-
- // Create VIE
- TbInterfaces ViE("ViERtpRtcpStandardTest");
- // Create a video channel
- TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
-
- // Create a capture device
- TbCaptureDevice tbCapture(ViE);
- tbCapture.ConnectTo(tbChannel.videoChannel);
-
- ViETest::Log("\n");
- TbExternalTransport myTransport(*(ViE.network));
-
- EXPECT_EQ(0, ViE.network->RegisterSendTransport(
- tbChannel.videoChannel, myTransport));
-
- // ***************************************************************
- // Engine ready. Begin testing class
- // ***************************************************************
-
- unsigned short startSequenceNumber = 12345;
- ViETest::Log("Set start sequence number: %u", startSequenceNumber);
- EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
- tbChannel.videoChannel, startSequenceNumber));
-
- myTransport.EnableSequenceNumberCheck();
-
- EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
- AutoTestSleep(2000);
-
- unsigned short receivedSequenceNumber =
- myTransport.GetFirstSequenceNumber();
- ViETest::Log("First received sequence number: %u\n",
- receivedSequenceNumber);
- EXPECT_EQ(startSequenceNumber, receivedSequenceNumber);
-
- EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-
- //
- // RTCP CName
- //
- ViETest::Log("Testing CName\n");
- const char* sendCName = "ViEAutoTestCName\0";
- EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPCName(tbChannel.videoChannel, sendCName));
-
- char returnCName[webrtc::ViERTP_RTCP::KMaxRTCPCNameLength];
- memset(returnCName, 0, webrtc::ViERTP_RTCP::KMaxRTCPCNameLength);
- EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPCName(
- tbChannel.videoChannel, returnCName));
- EXPECT_STRCASEEQ(sendCName, returnCName);
-
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
- AutoTestSleep(1000);
-
- char remoteCName[webrtc::ViERTP_RTCP::KMaxRTCPCNameLength];
- memset(remoteCName, 0, webrtc::ViERTP_RTCP::KMaxRTCPCNameLength);
- EXPECT_EQ(0, ViE.rtp_rtcp->GetRemoteRTCPCName(
- tbChannel.videoChannel, remoteCName));
- EXPECT_STRCASEEQ(sendCName, remoteCName);
-
- //
- // Statistics
- //
- // Stop and restart to clear stats
- ViETest::Log("Testing statistics\n");
- EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-
- myTransport.ClearStats();
- int rate = 20;
- myTransport.SetPacketLoss(rate);
-
- // Start send to verify sending stats
-
- EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
- tbChannel.videoChannel, startSequenceNumber));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- unsigned short sentFractionsLost = 0;
- unsigned int sentCumulativeLost = 0;
- unsigned int sentExtendedMax = 0;
- unsigned int sentJitter = 0;
- int sentRttMs = 0;
- unsigned short recFractionsLost = 0;
- unsigned int recCumulativeLost = 0;
- unsigned int recExtendedMax = 0;
- unsigned int recJitter = 0;
- int recRttMs = 0;
-
- unsigned int sentTotalBitrate = 0;
- unsigned int sentVideoBitrate = 0;
- unsigned int sentFecBitrate = 0;
- unsigned int sentNackBitrate = 0;
-
- EXPECT_EQ(0, ViE.rtp_rtcp->GetBandwidthUsage(
- tbChannel.videoChannel, sentTotalBitrate, sentVideoBitrate,
- sentFecBitrate, sentNackBitrate));
-
- EXPECT_GT(sentTotalBitrate, 0u);
- EXPECT_EQ(sentFecBitrate, 0u);
- EXPECT_EQ(sentNackBitrate, 0u);
-
- EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
-
- AutoTestSleep(2000);
-
- EXPECT_EQ(0, ViE.rtp_rtcp->GetSentRTCPStatistics(
- tbChannel.videoChannel, sentFractionsLost, sentCumulativeLost,
- sentExtendedMax, sentJitter, sentRttMs));
- EXPECT_GT(sentCumulativeLost, 0u);
- EXPECT_GT(sentExtendedMax, startSequenceNumber);
- EXPECT_GT(sentJitter, 0u);
- EXPECT_GT(sentRttMs, 0);
-
- EXPECT_EQ(0, ViE.rtp_rtcp->GetReceivedRTCPStatistics(
- tbChannel.videoChannel, recFractionsLost, recCumulativeLost,
- recExtendedMax, recJitter, recRttMs));
-
- EXPECT_GT(recCumulativeLost, 0u);
- EXPECT_GT(recExtendedMax, startSequenceNumber);
- EXPECT_GT(recJitter, 0u);
- EXPECT_GT(recRttMs, 0);
-
- unsigned int estimated_bandwidth = 0;
- EXPECT_EQ(0, ViE.rtp_rtcp->GetEstimatedSendBandwidth(
- tbChannel.videoChannel,
- &estimated_bandwidth));
- EXPECT_GT(estimated_bandwidth, 0u);
-
- EXPECT_EQ(0, ViE.rtp_rtcp->GetEstimatedReceiveBandwidth(
- tbChannel.videoChannel,
- &estimated_bandwidth));
- EXPECT_GT(estimated_bandwidth, 0u);
-
- // Check that rec stats extended max is greater than what we've sent.
- EXPECT_GE(recExtendedMax, sentExtendedMax);
- EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-
- //
- // Test bandwidth statistics with NACK and FEC separately
- //
-
- myTransport.ClearStats();
- myTransport.SetPacketLoss(rate);
-
- EXPECT_EQ(0, ViE.rtp_rtcp->SetFECStatus(
- tbChannel.videoChannel, true, 96, 97));
- EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- EXPECT_EQ(0, ViE.rtp_rtcp->GetBandwidthUsage(
- tbChannel.videoChannel, sentTotalBitrate, sentVideoBitrate,
- sentFecBitrate, sentNackBitrate));
-
- EXPECT_GT(sentTotalBitrate, 0u);
- EXPECT_GE(sentFecBitrate, 10u);
- EXPECT_EQ(sentNackBitrate, 0u);
-
- EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.rtp_rtcp->SetFECStatus(
- tbChannel.videoChannel, false, 96, 97));
- EXPECT_EQ(0, ViE.rtp_rtcp->SetNACKStatus(tbChannel.videoChannel, true));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- EXPECT_EQ(0, ViE.rtp_rtcp->GetBandwidthUsage(
- tbChannel.videoChannel, sentTotalBitrate, sentVideoBitrate,
- sentFecBitrate, sentNackBitrate));
-
- // TODO(holmer): Write a non-flaky verification of this API.
- // numberOfErrors += ViETest::TestError(sentTotalBitrate > 0 &&
- // sentFecBitrate == 0 &&
- // sentNackBitrate > 0,
- // "ERROR: %s at line %d",
- // __FUNCTION__, __LINE__);
-
- EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.rtp_rtcp->SetNACKStatus(tbChannel.videoChannel, false));
-
- //
- // Keepalive
- //
- ViETest::Log("Testing RTP keep alive...\n");
- EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
-
- myTransport.SetPacketLoss(0);
- myTransport.ClearStats();
-
- const char keepAlivePT = 109;
- unsigned int deltaTimeSeconds = 2;
- EXPECT_EQ(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
- tbChannel.videoChannel, true, keepAlivePT, deltaTimeSeconds));
-
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- EXPECT_EQ(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
- tbChannel.videoChannel, false, keepAlivePT, deltaTimeSeconds));
-
- WebRtc_Word32 numRtpPackets = 0;
- WebRtc_Word32 numDroppedPackets = 0;
- WebRtc_Word32 numRtcpPackets = 0;
- myTransport.GetStats(numRtpPackets, numDroppedPackets, numRtcpPackets);
- WebRtc_Word32 expectedPackets = KAutoTestSleepTimeMs / (1000 *
- static_cast<WebRtc_Word32>(deltaTimeSeconds));
- EXPECT_EQ(expectedPackets, numRtpPackets);
-
- // Test to set SSRC
- unsigned int setSSRC = 0x01234567;
- ViETest::Log("Set SSRC %u", setSSRC);
- EXPECT_EQ(0, ViE.rtp_rtcp->SetLocalSSRC(tbChannel.videoChannel, setSSRC));
- EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
- myTransport.EnableSSRCCheck();
-
- AutoTestSleep(2000);
- unsigned int receivedSSRC = myTransport.ReceivedSSRC();
- ViETest::Log("Received SSRC %u\n", receivedSSRC);
- EXPECT_EQ(setSSRC, receivedSSRC);
-
- unsigned int localSSRC = 0;
- EXPECT_EQ(0, ViE.rtp_rtcp->GetLocalSSRC(tbChannel.videoChannel, localSSRC));
- EXPECT_EQ(setSSRC, localSSRC);
-
- unsigned int remoteSSRC = 0;
- EXPECT_EQ(0, ViE.rtp_rtcp->GetRemoteSSRC(
- tbChannel.videoChannel, remoteSSRC));
- EXPECT_EQ(setSSRC, remoteSSRC);
-
- EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-
- ViETest::Log("Testing RTP dump...\n");
-
- std::string inDumpName =
- ViETest::GetResultOutputPath() + "IncomingRTPDump.rtp";
- std::string outDumpName =
- ViETest::GetResultOutputPath() + "OutgoingRTPDump.rtp";
- EXPECT_EQ(0, ViE.rtp_rtcp->StartRTPDump(
- tbChannel.videoChannel, inDumpName.c_str(), webrtc::kRtpIncoming));
- EXPECT_EQ(0, ViE.rtp_rtcp->StartRTPDump(
- tbChannel.videoChannel, outDumpName.c_str(), webrtc::kRtpOutgoing));
-
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-
- AutoTestSleep(1000);
-
- EXPECT_EQ(0, ViE.rtp_rtcp->StopRTPDump(
- tbChannel.videoChannel, webrtc::kRtpIncoming));
- EXPECT_EQ(0, ViE.rtp_rtcp->StopRTPDump(
- tbChannel.videoChannel, webrtc::kRtpOutgoing));
-
- // Make sure data was actually saved to the file and we stored the same
- // amount of data in both files
- FILE* inDump = fopen(inDumpName.c_str(), "r");
- fseek(inDump, 0L, SEEK_END);
- long inEndPos = ftell(inDump);
- fclose(inDump);
- FILE* outDump = fopen(outDumpName.c_str(), "r");
- fseek(outDump, 0L, SEEK_END);
- long outEndPos = ftell(outDump);
- fclose(outDump);
-
- EXPECT_GT(inEndPos, 0);
- EXPECT_LT(inEndPos, outEndPos + 100);
-
- // Deregister external transport
- EXPECT_EQ(0, ViE.network->DeregisterSendTransport(tbChannel.videoChannel));
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
-}
-
-void ViEAutoTest::ViERtpRtcpExtendedTest()
-{
- //***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- //***************************************************************
- ViERtpRtcpStandardTest();
-
- // Create VIE
- TbInterfaces ViE("ViERtpRtcpExtendedTest");
- // Create a video channel
- TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
- // Create a capture device
- TbCaptureDevice tbCapture(ViE);
- tbCapture.ConnectTo(tbChannel.videoChannel);
-
- //tbChannel.StartReceive(rtpPort);
- //tbChannel.StartSend(rtpPort);
- TbExternalTransport myTransport(*(ViE.network));
-
- EXPECT_EQ(0, ViE.network->RegisterSendTransport(
- tbChannel.videoChannel, myTransport));
- EXPECT_EQ(0, ViE.base->StartReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
- //***************************************************************
- // Engine ready. Begin testing class
- //***************************************************************
-
- //
- // Application specific RTCP
- //
- //
-
- ViERtcpObserver rtcpObserver;
- EXPECT_EQ(0, ViE.rtp_rtcp->RegisterRTCPObserver(
- tbChannel.videoChannel, rtcpObserver));
-
- unsigned char subType = 3;
- unsigned int name = static_cast<unsigned int> (0x41424344); // 'ABCD';
- const char* data = "ViEAutoTest Data of length 32 -\0";
- const unsigned short numBytes = 32;
-
- EXPECT_EQ(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
- tbChannel.videoChannel, subType, name, data, numBytes));
-
- ViETest::Log("Sending RTCP application data...\n");
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- EXPECT_EQ(subType, rtcpObserver._subType);
- EXPECT_STRCASEEQ(data, rtcpObserver._data);
- EXPECT_EQ(name, rtcpObserver._name);
- EXPECT_EQ(numBytes, rtcpObserver._dataLength);
-
- ViETest::Log("\t RTCP application data received\n");
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
- EXPECT_EQ(0, ViE.base->StopReceive(tbChannel.videoChannel));
- EXPECT_EQ(0, ViE.base->StopSend(tbChannel.videoChannel));
-
- EXPECT_EQ(0, ViE.network->DeregisterSendTransport(tbChannel.videoChannel));
-}
-
-void ViEAutoTest::ViERtpRtcpAPITest()
-{
- //***************************************************************
- // Begin create/initialize WebRTC Video Engine for testing
- //***************************************************************
- // Create VIE
- TbInterfaces ViE("ViERtpRtcpAPITest");
- // Create a video channel
- TbVideoChannel tbChannel(ViE, webrtc::kVideoCodecVP8);
- // Create a capture device
- TbCaptureDevice tbCapture(ViE);
- tbCapture.ConnectTo(tbChannel.videoChannel);
-
- //***************************************************************
- // Engine ready. Begin testing class
- //***************************************************************
-
- //
- // Check different RTCP modes
- //
- webrtc::ViERTCPMode rtcpMode = webrtc::kRtcpNone;
- EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPStatus(
- tbChannel.videoChannel, rtcpMode));
- EXPECT_EQ(webrtc::kRtcpCompound_RFC4585, rtcpMode);
- EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPStatus(
- tbChannel.videoChannel, webrtc::kRtcpCompound_RFC4585));
- EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPStatus(
- tbChannel.videoChannel, rtcpMode));
- EXPECT_EQ(webrtc::kRtcpCompound_RFC4585, rtcpMode);
- EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPStatus(
- tbChannel.videoChannel, webrtc::kRtcpNonCompound_RFC5506));
- EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPStatus(
- tbChannel.videoChannel, rtcpMode));
- EXPECT_EQ(webrtc::kRtcpNonCompound_RFC5506, rtcpMode);
- EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPStatus(
- tbChannel.videoChannel, webrtc::kRtcpNone));
- EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPStatus(
- tbChannel.videoChannel, rtcpMode));
- EXPECT_EQ(webrtc::kRtcpNone, rtcpMode);
- EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPStatus(
- tbChannel.videoChannel, webrtc::kRtcpCompound_RFC4585));
-
- //
- // CName is testedn in SimpleTest
- // Start sequence number is tested in SimplTEst
- //
- const char* testCName = "ViEAutotestCName";
- EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPCName(
- tbChannel.videoChannel, testCName));
-
- char returnCName[256];
- memset(returnCName, 0, 256);
- EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPCName(
- tbChannel.videoChannel, returnCName));
- EXPECT_STRCASEEQ(testCName, returnCName);
-
- //
- // SSRC
- //
- EXPECT_EQ(0, ViE.rtp_rtcp->SetLocalSSRC(
- tbChannel.videoChannel, 0x01234567));
- EXPECT_EQ(0, ViE.rtp_rtcp->SetLocalSSRC(
- tbChannel.videoChannel, 0x76543210));
-
- unsigned int ssrc = 0;
- EXPECT_EQ(0, ViE.rtp_rtcp->GetLocalSSRC(tbChannel.videoChannel, ssrc));
- EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
- tbChannel.videoChannel, 1000));
- tbChannel.StartSend();
- EXPECT_NE(0, ViE.rtp_rtcp->SetStartSequenceNumber(
- tbChannel.videoChannel, 12345));
- tbChannel.StopSend();
-
- //
- // Start sequence number
- //
- EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
- tbChannel.videoChannel, 12345));
- EXPECT_EQ(0, ViE.rtp_rtcp->SetStartSequenceNumber(
- tbChannel.videoChannel, 1000));
- tbChannel.StartSend();
- EXPECT_NE(0, ViE.rtp_rtcp->SetStartSequenceNumber(
- tbChannel.videoChannel, 12345));
- tbChannel.StopSend();
-
- //
- // Application specific RTCP
- //
- {
- unsigned char subType = 3;
- unsigned int name = static_cast<unsigned int> (0x41424344); // 'ABCD';
- const char* data = "ViEAutoTest Data of length 32 --";
- const unsigned short numBytes = 32;
-
- tbChannel.StartSend();
- EXPECT_EQ(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
- tbChannel.videoChannel, subType, name, data, numBytes));
- EXPECT_NE(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
- tbChannel.videoChannel, subType, name, NULL, numBytes)) <<
- "Should fail on NULL input.";
- EXPECT_NE(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
- tbChannel.videoChannel, subType, name, data, numBytes - 1)) <<
- "Should fail on incorrect length.";
-
- EXPECT_EQ(0, ViE.rtp_rtcp->GetRTCPStatus(
- tbChannel.videoChannel, rtcpMode));
- EXPECT_EQ(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
- tbChannel.videoChannel, subType, name, data, numBytes));
- EXPECT_EQ(0, ViE.rtp_rtcp->SetRTCPStatus(
- tbChannel.videoChannel, webrtc::kRtcpCompound_RFC4585));
- tbChannel.StopSend();
- EXPECT_NE(0, ViE.rtp_rtcp->SendApplicationDefinedRTCPPacket(
- tbChannel.videoChannel, subType, name, data, numBytes));
- }
-
- //
- // Statistics
- //
- // Tested in SimpleTest(), we'll get errors if we haven't received a RTCP
- // packet.
-
- //
- // RTP Keepalive
- //
- {
- char setPT = 123;
- unsigned int setDeltaTime = 10;
- bool enabled = false;
- char getPT = 0;
- unsigned int getDeltaTime = 0;
- EXPECT_EQ(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
- tbChannel.videoChannel, true, 119));
- EXPECT_NE(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
- tbChannel.videoChannel, true, setPT, setDeltaTime));
- EXPECT_EQ(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
- tbChannel.videoChannel, false, setPT, setDeltaTime));
- EXPECT_EQ(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
- tbChannel.videoChannel, true, setPT, setDeltaTime));
- EXPECT_EQ(0, ViE.rtp_rtcp->GetRTPKeepAliveStatus(
- tbChannel.videoChannel, enabled, getPT, getDeltaTime));
-
- EXPECT_TRUE(enabled);
- EXPECT_EQ(setPT, getPT);
- EXPECT_EQ(setDeltaTime, getDeltaTime);
-
- EXPECT_EQ(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
- tbChannel.videoChannel, false, setPT, setDeltaTime));
-
- EXPECT_EQ(0, ViE.base->StartSend(tbChannel.videoChannel));
-
- EXPECT_EQ(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
- tbChannel.videoChannel, true, setPT, setDeltaTime));
-
- EXPECT_NE(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
- tbChannel.videoChannel, true, setPT, setDeltaTime));
-
- tbChannel.StopSend();
- EXPECT_NE(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
- tbChannel.videoChannel, enabled, getPT, 0));
- EXPECT_NE(0, ViE.rtp_rtcp->SetRTPKeepAliveStatus(
- tbChannel.videoChannel, enabled, getPT, 61));
- }
- //
- // RTP Dump
- //
- {
-#ifdef WEBRTC_ANDROID
- const char* dumpName = "/sdcard/DumpFileName.rtp";
-#else
- std::string output_file = webrtc::test::OutputPath() +
- "DumpFileName.rtp";
- const char* dumpName = output_file.c_str();
-#endif
- EXPECT_EQ(0, ViE.rtp_rtcp->StartRTPDump(
- tbChannel.videoChannel, dumpName, webrtc::kRtpIncoming));
- EXPECT_EQ(0, ViE.rtp_rtcp->StopRTPDump(
- tbChannel.videoChannel, webrtc::kRtpIncoming));
- EXPECT_NE(0, ViE.rtp_rtcp->StopRTPDump(
- tbChannel.videoChannel, webrtc::kRtpIncoming));
- EXPECT_EQ(0, ViE.rtp_rtcp->StartRTPDump(
- tbChannel.videoChannel, dumpName, webrtc::kRtpOutgoing));
- EXPECT_EQ(0, ViE.rtp_rtcp->StopRTPDump(
- tbChannel.videoChannel, webrtc::kRtpOutgoing));
- EXPECT_NE(0, ViE.rtp_rtcp->StopRTPDump(
- tbChannel.videoChannel, webrtc::kRtpOutgoing));
- EXPECT_NE(0, ViE.rtp_rtcp->StartRTPDump(
- tbChannel.videoChannel, dumpName, (webrtc::RTPDirections) 3));
- }
- //
- // RTP/RTCP Observers
- //
- {
- ViERtpObserver rtpObserver;
- EXPECT_EQ(0, ViE.rtp_rtcp->RegisterRTPObserver(
- tbChannel.videoChannel, rtpObserver));
- EXPECT_NE(0, ViE.rtp_rtcp->RegisterRTPObserver(
- tbChannel.videoChannel, rtpObserver));
- EXPECT_EQ(0, ViE.rtp_rtcp->DeregisterRTPObserver(
- tbChannel.videoChannel));
- EXPECT_NE(0, ViE.rtp_rtcp->DeregisterRTPObserver(
- tbChannel.videoChannel));
-
- ViERtcpObserver rtcpObserver;
- EXPECT_EQ(0, ViE.rtp_rtcp->RegisterRTCPObserver(
- tbChannel.videoChannel, rtcpObserver));
- EXPECT_NE(0, ViE.rtp_rtcp->RegisterRTCPObserver(
- tbChannel.videoChannel, rtcpObserver));
- EXPECT_EQ(0, ViE.rtp_rtcp->DeregisterRTCPObserver(
- tbChannel.videoChannel));
- EXPECT_NE(0, ViE.rtp_rtcp->DeregisterRTCPObserver(
- tbChannel.videoChannel));
- }
- //
- // PLI
- //
- {
- EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
- tbChannel.videoChannel, webrtc::kViEKeyFrameRequestPliRtcp));
- EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
- tbChannel.videoChannel, webrtc::kViEKeyFrameRequestPliRtcp));
- EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
- tbChannel.videoChannel, webrtc::kViEKeyFrameRequestNone));
- EXPECT_EQ(0, ViE.rtp_rtcp->SetKeyFrameRequestMethod(
- tbChannel.videoChannel, webrtc::kViEKeyFrameRequestNone));
- }
- //
- // NACK
- //
- {
- EXPECT_EQ(0, ViE.rtp_rtcp->SetNACKStatus(tbChannel.videoChannel, true));
- }
-
- //***************************************************************
- // Testing finished. Tear down Video Engine
- //***************************************************************
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_simulcast.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_simulcast.cc
deleted file mode 100644
index 6d77115..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_simulcast.cc
+++ /dev/null
@@ -1,531 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <iostream>
-
-#include "common_types.h"
-#include "tb_external_transport.h"
-#include "voe_base.h"
-#include "vie_autotest_defines.h"
-#include "vie_autotest.h"
-#include "vie_base.h"
-#include "vie_capture.h"
-#include "vie_codec.h"
-#include "vie_network.h"
-#include "vie_render.h"
-#include "vie_rtp_rtcp.h"
-
-#define VCM_RED_PAYLOAD_TYPE 96
-#define VCM_ULPFEC_PAYLOAD_TYPE 97
-
-int VideoEngineSimulcastTest(void* window1, void* window2)
-{
- //********************************************************
- // Begin create/initialize Video Engine for testing
- //********************************************************
-
- int error = 0;
-
- //
- // Create a VideoEngine instance
- //
- webrtc::VideoEngine* ptrViE = NULL;
- ptrViE = webrtc::VideoEngine::Create();
- if (ptrViE == NULL)
- {
- printf("ERROR in VideoEngine::Create\n");
- return -1;
- }
-
- error = ptrViE->SetTraceFilter(webrtc::kTraceAll);
- if (error == -1)
- {
- printf("ERROR in VideoEngine::SetTraceLevel\n");
- return -1;
- }
-
-
- std::string trace_file =
- ViETest::GetResultOutputPath() + "ViESimulcast_trace.txt";
- error = ptrViE->SetTraceFile(trace_file.c_str());
- if (error == -1)
- {
- printf("ERROR in VideoEngine::SetTraceFile\n");
- return -1;
- }
-
- //
- // Init VideoEngine and create a channel
- //
- webrtc::ViEBase* ptrViEBase = webrtc::ViEBase::GetInterface(ptrViE);
- if (ptrViEBase == NULL)
- {
- printf("ERROR in ViEBase::GetInterface\n");
- return -1;
- }
-
- error = ptrViEBase->Init();
- if (error == -1)
- {
- printf("ERROR in ViEBase::Init\n");
- return -1;
- }
-
- int videoChannel = -1;
- error = ptrViEBase->CreateChannel(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViEBase::CreateChannel\n");
- return -1;
- }
-
- //
- // List available capture devices, allocate and connect.
- //
- webrtc::ViECapture* ptrViECapture =
- webrtc::ViECapture::GetInterface(ptrViE);
- if (ptrViEBase == NULL)
- {
- printf("ERROR in ViECapture::GetInterface\n");
- return -1;
- }
-
- const unsigned int KMaxDeviceNameLength = 128;
- const unsigned int KMaxUniqueIdLength = 256;
- char deviceName[KMaxDeviceNameLength];
- memset(deviceName, 0, KMaxDeviceNameLength);
- char uniqueId[KMaxUniqueIdLength];
- memset(uniqueId, 0, KMaxUniqueIdLength);
-
- printf("Available capture devices:\n");
- int captureIdx = 0;
- for (captureIdx = 0;
- captureIdx < ptrViECapture->NumberOfCaptureDevices();
- captureIdx++)
- {
- memset(deviceName, 0, KMaxDeviceNameLength);
- memset(uniqueId, 0, KMaxUniqueIdLength);
-
- error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
- KMaxDeviceNameLength, uniqueId,
- KMaxUniqueIdLength);
- if (error == -1)
- {
- printf("ERROR in ViECapture::GetCaptureDevice\n");
- return -1;
- }
- printf("\t %d. %s\n", captureIdx + 1, deviceName);
- }
- printf("\nChoose capture device: ");
-#ifdef WEBRTC_ANDROID
- captureIdx = 0;
- printf("0\n");
-#else
- if (scanf("%d", &captureIdx) != 1)
- {
- printf("Error in scanf()\n");
- return -1;
- }
- getchar();
- captureIdx = captureIdx - 1; // Compensate for idx start at 1.
-#endif
- error = ptrViECapture->GetCaptureDevice(captureIdx, deviceName,
- KMaxDeviceNameLength, uniqueId,
- KMaxUniqueIdLength);
- if (error == -1)
- {
- printf("ERROR in ViECapture::GetCaptureDevice\n");
- return -1;
- }
-
- int captureId = 0;
- error = ptrViECapture->AllocateCaptureDevice(uniqueId, KMaxUniqueIdLength,
- captureId);
- if (error == -1)
- {
- printf("ERROR in ViECapture::AllocateCaptureDevice\n");
- return -1;
- }
-
- error = ptrViECapture->ConnectCaptureDevice(captureId, videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViECapture::ConnectCaptureDevice\n");
- return -1;
- }
-
- error = ptrViECapture->StartCapture(captureId);
- if (error == -1)
- {
- printf("ERROR in ViECapture::StartCapture\n");
- return -1;
- }
-
- //
- // RTP/RTCP settings
- //
- webrtc::ViERTP_RTCP* ptrViERtpRtcp =
- webrtc::ViERTP_RTCP::GetInterface(ptrViE);
- if (ptrViERtpRtcp == NULL)
- {
- printf("ERROR in ViERTP_RTCP::GetInterface\n");
- return -1;
- }
-
- error = ptrViERtpRtcp->SetRTCPStatus(videoChannel,
- webrtc::kRtcpCompound_RFC4585);
- if (error == -1)
- {
- printf("ERROR in ViERTP_RTCP::SetRTCPStatus\n");
- return -1;
- }
-
- error = ptrViERtpRtcp->SetKeyFrameRequestMethod(
- videoChannel, webrtc::kViEKeyFrameRequestPliRtcp);
- if (error == -1)
- {
- printf("ERROR in ViERTP_RTCP::SetKeyFrameRequestMethod\n");
- return -1;
- }
-
- //
- // Set up rendering
- //
- webrtc::ViERender* ptrViERender = webrtc::ViERender::GetInterface(ptrViE);
- if (ptrViERender == NULL)
- {
- printf("ERROR in ViERender::GetInterface\n");
- return -1;
- }
-
- error
- = ptrViERender->AddRenderer(captureId, window1, 0, 0.0, 0.0, 1.0, 1.0);
- if (error == -1)
- {
- printf("ERROR in ViERender::AddRenderer\n");
- return -1;
- }
-
- error = ptrViERender->StartRender(captureId);
- if (error == -1)
- {
- printf("ERROR in ViERender::StartRender\n");
- return -1;
- }
-
- error = ptrViERender->AddRenderer(videoChannel, window2, 1, 0.0, 0.0, 1.0,
- 1.0);
- if (error == -1)
- {
- printf("ERROR in ViERender::AddRenderer\n");
- return -1;
- }
-
- error = ptrViERender->StartRender(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViERender::StartRender\n");
- return -1;
- }
-
- //
- // Setup codecs
- //
- webrtc::ViECodec* ptrViECodec = webrtc::ViECodec::GetInterface(ptrViE);
- if (ptrViECodec == NULL)
- {
- printf("ERROR in ViECodec::GetInterface\n");
- return -1;
- }
-
- // Check available codecs and prepare receive codecs
- printf("\nAvailable codecs:\n");
- webrtc::VideoCodec videoCodec;
- memset(&videoCodec, 0, sizeof(webrtc::VideoCodec));
- int codecIdx = 0;
- for (codecIdx = 0; codecIdx < ptrViECodec->NumberOfCodecs(); codecIdx++)
- {
- error = ptrViECodec->GetCodec(codecIdx, videoCodec);
- if (error == -1)
- {
- printf("ERROR in ViECodec::GetCodec\n");
- return -1;
- }
- // try to keep the test frame size small when I420
- if (videoCodec.codecType != webrtc::kVideoCodecVP8)
- {
- continue;
- }
- error = ptrViECodec->SetReceiveCodec(videoChannel, videoCodec);
- if (error == -1)
- {
- printf("ERROR in ViECodec::SetReceiveCodec\n");
- return -1;
- }
- if (videoCodec.codecType != webrtc::kVideoCodecRED
- && videoCodec.codecType != webrtc::kVideoCodecULPFEC)
- {
- printf("\t %d. %s\n", codecIdx + 1, videoCodec.plName);
- }
- break;
- }
- error = ptrViECodec->GetCodec(codecIdx, videoCodec);
- if (error == -1)
- {
- printf("ERROR in ViECodec::GetCodec\n");
- return -1;
- }
-
- // Set spatial resolution option
- videoCodec.width = 1280;
- videoCodec.height = 720;
-
- // simulcast settings
- videoCodec.numberOfSimulcastStreams = 3;
- videoCodec.simulcastStream[0].width = 320;
- videoCodec.simulcastStream[0].height = 180;
- videoCodec.simulcastStream[0].numberOfTemporalLayers = 0;
- videoCodec.simulcastStream[0].maxBitrate = 100;
- videoCodec.simulcastStream[0].qpMax = videoCodec.qpMax;
-
- videoCodec.simulcastStream[1].width = 640;
- videoCodec.simulcastStream[1].height = 360;
- videoCodec.simulcastStream[1].numberOfTemporalLayers = 0;
- videoCodec.simulcastStream[1].maxBitrate = 500;
- videoCodec.simulcastStream[1].qpMax = videoCodec.qpMax;
-
- videoCodec.simulcastStream[2].width = 1280;
- videoCodec.simulcastStream[2].height = 720;
- videoCodec.simulcastStream[2].numberOfTemporalLayers = 0;
- videoCodec.simulcastStream[2].maxBitrate = 1200;
- videoCodec.simulcastStream[2].qpMax = videoCodec.qpMax;
-
- // Set start bit rate
- std::string str;
- std::cout << std::endl;
- std::cout << "Choose start rate (in kbps). Press enter for default: ";
- std::getline(std::cin, str);
- int startRate = atoi(str.c_str());
- if(startRate != 0)
- {
- videoCodec.startBitrate=startRate;
- }
-
- error = ptrViECodec->SetSendCodec(videoChannel, videoCodec);
- if (error == -1)
- {
- printf("ERROR in ViECodec::SetSendCodec\n");
- return -1;
- }
- //
- // Address settings
- //
- webrtc::ViENetwork* ptrViENetwork =
- webrtc::ViENetwork::GetInterface(ptrViE);
- if (ptrViENetwork == NULL)
- {
- printf("ERROR in ViENetwork::GetInterface\n");
- return -1;
- }
-
- // Setting External transport
- TbExternalTransport extTransport(*(ptrViENetwork));
-
- error = ptrViENetwork->RegisterSendTransport(videoChannel,
- extTransport);
- if (error == -1)
- {
- printf("ERROR in ViECodec::RegisterSendTransport \n");
- return -1;
- }
-
- extTransport.SetPacketLoss(0);
-
- // Set network delay value
- extTransport.SetNetworkDelay(10);
-
- extTransport.SetSSRCFilter(3);
-
- for (int idx = 0; idx < 3; idx++)
- {
- error = ptrViERtpRtcp->SetLocalSSRC(videoChannel,
- idx+1, // SSRC
- webrtc::kViEStreamTypeNormal,
- idx);
- if (error == -1)
- {
- printf("ERROR in ViERTP_RTCP::SetLocalSSRC(idx:%d)\n", idx);
- return -1;
- }
- }
-
- error = ptrViEBase->StartReceive(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViENetwork::StartReceive\n");
- return -1;
- }
-
- error = ptrViEBase->StartSend(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViENetwork::StartSend\n");
- return -1;
- }
-
- //********************************************************
- // Engine started
- //********************************************************
-
- printf("\nSimulcast call started\n\n");
- do
- {
- printf("Enter new SSRC filter 1,2 or 3\n");
- printf("Press enter to stop...");
- str.clear();
- std::getline(std::cin, str);
- if (!str.empty())
- {
- int ssrc = atoi(str.c_str());
- if (ssrc > 0 && ssrc < 4)
- {
- extTransport.SetSSRCFilter(ssrc);
- } else
- {
- printf("Invalid SSRC\n");
- }
- } else
- {
- break;
- }
- } while (true);
-
- //********************************************************
- // Testing finished. Tear down Video Engine
- //********************************************************
-
- error = ptrViEBase->StopReceive(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViEBase::StopReceive\n");
- return -1;
- }
-
- error = ptrViEBase->StopSend(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViEBase::StopSend\n");
- return -1;
- }
-
- error = ptrViERender->StopRender(captureId);
- if (error == -1)
- {
- printf("ERROR in ViERender::StopRender\n");
- return -1;
- }
-
- error = ptrViERender->RemoveRenderer(captureId);
- if (error == -1)
- {
- printf("ERROR in ViERender::RemoveRenderer\n");
- return -1;
- }
-
- error = ptrViERender->StopRender(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViERender::StopRender\n");
- return -1;
- }
-
- error = ptrViERender->RemoveRenderer(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViERender::RemoveRenderer\n");
- return -1;
- }
-
- error = ptrViECapture->StopCapture(captureId);
- if (error == -1)
- {
- printf("ERROR in ViECapture::StopCapture\n");
- return -1;
- }
-
- error = ptrViECapture->DisconnectCaptureDevice(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViECapture::DisconnectCaptureDevice\n");
- return -1;
- }
-
- error = ptrViECapture->ReleaseCaptureDevice(captureId);
- if (error == -1)
- {
- printf("ERROR in ViECapture::ReleaseCaptureDevice\n");
- return -1;
- }
-
- error = ptrViEBase->DeleteChannel(videoChannel);
- if (error == -1)
- {
- printf("ERROR in ViEBase::DeleteChannel\n");
- return -1;
- }
-
- int remainingInterfaces = 0;
- remainingInterfaces = ptrViECodec->Release();
- remainingInterfaces += ptrViECapture->Release();
- remainingInterfaces += ptrViERtpRtcp->Release();
- remainingInterfaces += ptrViERender->Release();
- remainingInterfaces += ptrViENetwork->Release();
- remainingInterfaces += ptrViEBase->Release();
- if (remainingInterfaces > 0)
- {
- printf("ERROR: Could not release all interfaces\n");
- return -1;
- }
-
- bool deleted = webrtc::VideoEngine::Delete(ptrViE);
- if (deleted == false)
- {
- printf("ERROR in VideoEngine::Delete\n");
- return -1;
- }
- return 0;
-
- //
- // END: VideoEngine 3.0 Sample Code
- //
- // ===================================================================
-}
-
-int ViEAutoTest::ViESimulcastCall()
-{
- ViETest::Log(" ");
- ViETest::Log("========================================");
- ViETest::Log(" ViE Autotest Simulcast Call\n");
-
- if (VideoEngineSimulcastTest(_window1, _window2) == 0)
- {
- ViETest::Log(" ");
- ViETest::Log(" ViE Autotest Simulcast Call Done");
- ViETest::Log("========================================");
- ViETest::Log(" ");
-
- return 0;
- }
- ViETest::Log(" ");
- ViETest::Log(" ViE Autotest Simulcast Call Failed");
- ViETest::Log("========================================");
- ViETest::Log(" ");
- return 1;
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_autotest_win.cc b/trunk/src/video_engine/test/auto_test/source/vie_autotest_win.cc
deleted file mode 100755
index 7887767..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_autotest_win.cc
+++ /dev/null
@@ -1,214 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-//
-// vie_autotest_windows.cc
-//
-
-#include "vie_autotest_windows.h"
-
-#include "vie_autotest_defines.h"
-#include "vie_autotest_main.h"
-
-#include "engine_configurations.h"
-#include "critical_section_wrapper.h"
-#include "thread_wrapper.h"
-
-#include <windows.h>
-
-#ifdef _DEBUG
-//#include "vld.h"
-#endif
-
-// Disable Visual studio warnings
-// 'this' : used in base member initializer list
-#pragma warning(disable: 4355)
-// new behavior: elements of array 'XXX' will be default initialized
-#pragma warning(disable: 4351)
-
-LRESULT CALLBACK ViEAutoTestWinProc(HWND hWnd, UINT uMsg, WPARAM wParam,
- LPARAM lParam) {
- switch (uMsg) {
- case WM_DESTROY:
- PostQuitMessage( WM_QUIT);
- break;
- case WM_COMMAND:
- break;
- }
- return DefWindowProc(hWnd, uMsg, wParam, lParam);
-}
-
-ViEAutoTestWindowManager::ViEAutoTestWindowManager()
- : _window1(NULL),
- _window2(NULL),
- _terminate(false),
- _eventThread(*webrtc::ThreadWrapper::CreateThread(
- EventProcess, this, webrtc::kNormalPriority,
- "ViEAutotestEventThread")),
- _crit(*webrtc::CriticalSectionWrapper::CreateCriticalSection()),
- _hwnd1(NULL),
- _hwnd2(NULL),
- _hwnd1Size(),
- _hwnd2Size(),
- _hwnd1Title(),
- _hwnd2Title() {
-}
-
-ViEAutoTestWindowManager::~ViEAutoTestWindowManager() {
- if (_hwnd1) {
- ViEDestroyWindow(_hwnd1);
- }
- if (_hwnd2) {
- ViEDestroyWindow(_hwnd1);
- }
- delete &_crit;
-}
-
-void* ViEAutoTestWindowManager::GetWindow1() {
- return _window1;
-}
-
-void* ViEAutoTestWindowManager::GetWindow2() {
- return _window2;
-}
-
-int ViEAutoTestWindowManager::CreateWindows(AutoTestRect window1Size,
- AutoTestRect window2Size,
- void* window1Title,
- void* window2Title) {
- _hwnd1Size.Copy(window1Size);
- _hwnd2Size.Copy(window2Size);
- memcpy(_hwnd1Title, window1Title, TITLE_LENGTH);
- memcpy(_hwnd2Title, window2Title, TITLE_LENGTH);
-
- unsigned int tId = 0;
- _eventThread.Start(tId);
-
- do {
- _crit.Enter();
- if (_window1 != NULL) {
- break;
- }
- _crit.Leave();
- AutoTestSleep(10);
- } while (true);
- _crit.Leave();
- return 0;
-}
-
-int ViEAutoTestWindowManager::TerminateWindows() {
- _eventThread.SetNotAlive();
-
- _terminate = true;
- if (_eventThread.Stop()) {
- _crit.Enter();
- delete &_eventThread;
- _crit.Leave();
- }
-
- return 0;
-}
-
-bool ViEAutoTestWindowManager::EventProcess(void* obj) {
- return static_cast<ViEAutoTestWindowManager*> (obj)->EventLoop();
-}
-
-bool ViEAutoTestWindowManager::EventLoop() {
- _crit.Enter();
-
- ViECreateWindow(_hwnd1, _hwnd1Size.origin.x, _hwnd1Size.origin.y,
- _hwnd1Size.size.width, _hwnd1Size.size.height, _hwnd1Title);
- ViECreateWindow(_hwnd2, _hwnd2Size.origin.x, _hwnd2Size.origin.y,
- _hwnd2Size.size.width, _hwnd2Size.size.height, _hwnd2Title);
-
- _window1 = (void*) _hwnd1;
- _window2 = (void*) _hwnd2;
- MSG msg;
- while (!_terminate) {
- if (PeekMessage(&msg, NULL, 0, 0, PM_REMOVE)) {
- TranslateMessage(&msg);
- DispatchMessage(&msg);
- }
- _crit.Leave();
- AutoTestSleep(10);
- _crit.Enter();
- }
- ViEDestroyWindow(_hwnd1);
- ViEDestroyWindow(_hwnd2);
- _crit.Leave();
-
- return false;
-}
-
-int ViEAutoTestWindowManager::ViECreateWindow(HWND &hwndMain, int xPos,
- int yPos, int width, int height,
- TCHAR* className) {
- HINSTANCE hinst = GetModuleHandle(0);
- WNDCLASSEX wcx;
- wcx.hInstance = hinst;
- wcx.lpszClassName = className;
- wcx.lpfnWndProc = (WNDPROC) ViEAutoTestWinProc;
- wcx.style = CS_DBLCLKS;
- wcx.hIcon = LoadIcon(NULL, IDI_APPLICATION);
- wcx.hIconSm = LoadIcon(NULL, IDI_APPLICATION);
- wcx.hCursor = LoadCursor(NULL, IDC_ARROW);
- wcx.lpszMenuName = NULL;
- wcx.cbSize = sizeof(WNDCLASSEX);
- wcx.cbClsExtra = 0;
- wcx.cbWndExtra = 0;
- wcx.hbrBackground = GetSysColorBrush(COLOR_3DFACE);
-
- RegisterClassEx(&wcx);
-
- // Create the main window.
- hwndMain = CreateWindowEx(0, // no extended styles
- className, // class name
- className, // window name
- WS_OVERLAPPED | WS_THICKFRAME, // overlapped window
- xPos, // horizontal position
- yPos, // vertical position
- width, // width
- height, // height
- (HWND) NULL, // no parent or owner window
- (HMENU) NULL, // class menu used
- hinst, // instance handle
- NULL); // no window creation data
-
- if (!hwndMain) {
- int error = GetLastError();
- return -1;
- }
-
- // Show the window using the flag specified by the program
- // that started the application, and send the application
- // a WM_PAINT message.
- ShowWindow(hwndMain, SW_SHOWDEFAULT);
- UpdateWindow(hwndMain);
-
- ::SetWindowPos(hwndMain, HWND_TOP, xPos, yPos, width, height,
- SWP_FRAMECHANGED);
-
- return 0;
-}
-
-int ViEAutoTestWindowManager::ViEDestroyWindow(HWND& hwnd) {
- ::DestroyWindow(hwnd);
- return 0;
-}
-
-bool ViEAutoTestWindowManager::SetTopmostWindow() {
- // Meant to put terminal window on top
- return true;
-}
-
-int main(int argc, char* argv[]) {
- ViEAutoTestMain auto_test;
- return auto_test.RunTests(argc, argv);
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_file_based_comparison_tests.cc b/trunk/src/video_engine/test/auto_test/source/vie_file_based_comparison_tests.cc
deleted file mode 100644
index aff5118..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_file_based_comparison_tests.cc
+++ /dev/null
@@ -1,160 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/test/auto_test/interface/vie_file_based_comparison_tests.h"
-
-#include "video_engine/test/auto_test/interface/tb_interfaces.h"
-#include "video_engine/test/auto_test/interface/vie_autotest_defines.h"
-#include "video_engine/test/auto_test/helpers/vie_fake_camera.h"
-#include "video_engine/test/auto_test/helpers/vie_to_file_renderer.h"
-#include "video_engine/test/auto_test/primitives/base_primitives.h"
-#include "video_engine/test/auto_test/primitives/codec_primitives.h"
-#include "video_engine/test/auto_test/primitives/framedrop_primitives.h"
-#include "video_engine/test/auto_test/primitives/general_primitives.h"
-
-bool ViEFileBasedComparisonTests::TestCallSetup(
- const std::string& i420_video_file,
- int width,
- int height,
- ViEToFileRenderer* local_file_renderer,
- ViEToFileRenderer* remote_file_renderer) {
-
- TbInterfaces interfaces("TestCallSetup");
-
- int video_channel = -1;
- EXPECT_EQ(0, interfaces.base->CreateChannel(video_channel));
-
- ViEFakeCamera fake_camera(interfaces.capture);
- if (!fake_camera.StartCameraInNewThread(i420_video_file,
- width,
- height)) {
- // No point in continuing if we have no proper video source
- ADD_FAILURE() << "Could not open input video " << i420_video_file <<
- ": aborting test...";
- return false;
- }
- int capture_id = fake_camera.capture_id();
-
- // Apparently, we need to connect external capture devices, but we should
- // not start them since the external device is not a proper device.
- EXPECT_EQ(0, interfaces.capture->ConnectCaptureDevice(
- capture_id, video_channel));
-
- ConfigureRtpRtcp(interfaces.rtp_rtcp, video_channel);
-
- webrtc::ViERender *render_interface = interfaces.render;
-
- RenderToFile(render_interface, capture_id, local_file_renderer);
- RenderToFile(render_interface, video_channel, remote_file_renderer);
-
- // Run the test itself:
- const WebRtc_UWord8* device_name =
- reinterpret_cast<const WebRtc_UWord8*>("Fake Capture Device");
-
- ::TestI420CallSetup(interfaces.codec, interfaces.video_engine,
- interfaces.base, interfaces.network, video_channel,
- device_name);
-
- AutoTestSleep(KAutoTestSleepTimeMs);
-
- EXPECT_EQ(0, interfaces.base->StopReceive(video_channel));
-
- StopAndRemoveRenderers(interfaces.base, render_interface, video_channel,
- capture_id);
-
- interfaces.capture->DisconnectCaptureDevice(video_channel);
-
- // Stop sending data, clean up the camera thread and release the capture
- // device. Note that this all happens after StopEverything, so this
- // tests that the system doesn't mind that the external capture device sends
- // data after rendering has been stopped.
- fake_camera.StopCamera();
-
- EXPECT_EQ(0, interfaces.base->DeleteChannel(video_channel));
- return true;
-}
-
-bool ViEFileBasedComparisonTests::TestCodecs(
- const std::string& i420_video_file,
- int width,
- int height,
- ViEToFileRenderer* local_file_renderer,
- ViEToFileRenderer* remote_file_renderer) {
-
- TbInterfaces interfaces = TbInterfaces("TestCodecs");
-
- ViEFakeCamera fake_camera(interfaces.capture);
- if (!fake_camera.StartCameraInNewThread(i420_video_file, width, height)) {
- // No point in continuing if we have no proper video source
- ADD_FAILURE() << "Could not open input video " << i420_video_file <<
- ": aborting test...";
- return false;
- }
-
- int video_channel = -1;
- int capture_id = fake_camera.capture_id();
-
- EXPECT_EQ(0, interfaces.base->CreateChannel(video_channel));
- EXPECT_EQ(0, interfaces.capture->ConnectCaptureDevice(
- capture_id, video_channel));
-
- ConfigureRtpRtcp(interfaces.rtp_rtcp, video_channel);
-
- RenderToFile(interfaces.render, capture_id, local_file_renderer);
- RenderToFile(interfaces.render, video_channel, remote_file_renderer);
-
- // Force the codec resolution to what our input video is so we can make
- // comparisons later. Our comparison algorithms wouldn't like scaling.
- ::TestCodecs(interfaces, capture_id, video_channel, width, height);
-
- fake_camera.StopCamera();
- return true;
-}
-
-void ViEFileBasedComparisonTests::TestFullStack(
- const std::string& i420_video_file,
- int width,
- int height,
- int bit_rate_kbps,
- int packet_loss_percent,
- int network_delay_ms,
- ViEToFileRenderer* local_file_renderer,
- ViEToFileRenderer* remote_file_renderer,
- FrameDropDetector* frame_drop_detector) {
- TbInterfaces interfaces = TbInterfaces("TestFullStack");
-
- // Setup camera capturing from file.
- ViEFakeCamera fake_camera(interfaces.capture);
- if (!fake_camera.StartCameraInNewThread(i420_video_file, width, height)) {
- // No point in continuing if we have no proper video source
- ADD_FAILURE() << "Could not open input video " << i420_video_file <<
- ": aborting test...";
- return;
- }
- int video_channel = -1;
- int capture_id = fake_camera.capture_id();
- EXPECT_EQ(0, interfaces.base->CreateChannel(video_channel));
-
- // Must set SSRC to avoid SSRC collision detection since we're sending and
- // receiving from the same machine (that would cause frames being discarded
- // and decoder reset).
- EXPECT_EQ(0, interfaces.rtp_rtcp->SetLocalSSRC(video_channel, 12345));
-
- EXPECT_EQ(0, interfaces.capture->ConnectCaptureDevice(
- capture_id, video_channel));
- ConfigureRtpRtcp(interfaces.rtp_rtcp, video_channel);
- RenderToFile(interfaces.render, capture_id, local_file_renderer);
- RenderToFile(interfaces.render, video_channel, remote_file_renderer);
-
- ::TestFullStack(interfaces, capture_id, video_channel, width, height,
- bit_rate_kbps, packet_loss_percent, network_delay_ms,
- frame_drop_detector);
- EXPECT_TRUE(fake_camera.StopCamera());
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_linux.cc b/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_linux.cc
deleted file mode 100644
index 1b16878..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_linux.cc
+++ /dev/null
@@ -1,18 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vie_window_manager_factory.h"
-
-#include "vie_autotest_linux.h"
-
-ViEAutoTestWindowManagerInterface*
-ViEWindowManagerFactory::CreateWindowManagerForCurrentPlatform() {
- return new ViEAutoTestWindowManager();
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_mac.mm b/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_mac.mm
deleted file mode 100644
index 806d10f..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_mac.mm
+++ /dev/null
@@ -1,23 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vie_window_manager_factory.h"
-
-#include "engine_configurations.h"
-#if defined(COCOA_RENDERING)
-#include "vie_autotest_mac_cocoa.h"
-#elif defined(CARBON_RENDERING)
-#include "vie_autotest_mac_carbon.h"
-#endif
-
-ViEAutoTestWindowManagerInterface*
-ViEWindowManagerFactory::CreateWindowManagerForCurrentPlatform() {
- return new ViEAutoTestWindowManager();
-}
diff --git a/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_win.cc b/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_win.cc
deleted file mode 100644
index 11114fd..0000000
--- a/trunk/src/video_engine/test/auto_test/source/vie_window_manager_factory_win.cc
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-#include "vie_window_manager_factory.h"
-
-#include "vie_autotest_windows.h"
-
-ViEAutoTestWindowManagerInterface*
-ViEWindowManagerFactory::CreateWindowManagerForCurrentPlatform() {
- return new ViEAutoTestWindowManager();
-}
diff --git a/trunk/src/video_engine/test/auto_test/vie_auto_test.gypi b/trunk/src/video_engine/test/auto_test/vie_auto_test.gypi
deleted file mode 100644
index cf403b7..0000000
--- a/trunk/src/video_engine/test/auto_test/vie_auto_test.gypi
+++ /dev/null
@@ -1,151 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'vie_auto_test',
- 'type': 'executable',
- 'dependencies': [
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/modules/modules.gyp:video_render_module',
- '<(webrtc_root)/modules/modules.gyp:video_capture_module',
- '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine_core',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../third_party/google-gflags/google-gflags.gyp:google-gflags',
- '<(webrtc_root)/../test/metrics.gyp:metrics',
- '<(webrtc_root)/../test/test.gyp:test_support',
- 'video_engine_core',
- ],
- 'include_dirs': [
- 'interface/',
- 'helpers/',
- 'primitives',
- '../../include',
- '../..',
- '../../../modules/video_coding/codecs/interface',
- '../../../common_video/interface',
- ],
- 'sources': [
- 'interface/tb_capture_device.h',
- 'interface/tb_external_transport.h',
- 'interface/tb_I420_codec.h',
- 'interface/tb_interfaces.h',
- 'interface/tb_video_channel.h',
- 'interface/vie_autotest.h',
- 'interface/vie_autotest_defines.h',
- 'interface/vie_autotest_linux.h',
- 'interface/vie_autotest_mac_carbon.h',
- 'interface/vie_autotest_mac_cocoa.h',
- 'interface/vie_autotest_main.h',
- 'interface/vie_autotest_window_manager_interface.h',
- 'interface/vie_autotest_windows.h',
- 'interface/vie_file_based_comparison_tests.h',
- 'interface/vie_window_manager_factory.h',
-
- # Helper classes
- 'helpers/bit_flip_encryption.cc',
- 'helpers/bit_flip_encryption.h',
- 'helpers/random_encryption.cc',
- 'helpers/random_encryption.h',
- 'helpers/vie_fake_camera.cc',
- 'helpers/vie_fake_camera.h',
- 'helpers/vie_file_capture_device.cc',
- 'helpers/vie_file_capture_device.h',
- 'helpers/vie_to_file_renderer.cc',
- 'helpers/vie_to_file_renderer.h',
- 'helpers/vie_window_creator.cc',
- 'helpers/vie_window_creator.h',
-
- # New, fully automated tests
- 'automated/legacy_fixture.cc',
- 'automated/two_windows_fixture.cc',
- 'automated/vie_api_integration_test.cc',
- 'automated/vie_extended_integration_test.cc',
- 'automated/vie_rtp_fuzz_test.cc',
- 'automated/vie_standard_integration_test.cc',
- 'automated/vie_video_verification_test.cc',
-
- # Test primitives
- 'primitives/base_primitives.cc',
- 'primitives/base_primitives.h',
- 'primitives/codec_primitives.cc',
- 'primitives/codec_primitives.h',
- 'primitives/framedrop_primitives.h',
- 'primitives/framedrop_primitives.cc',
- 'primitives/framedrop_primitives_unittest.cc',
- 'primitives/general_primitives.cc',
- 'primitives/general_primitives.h',
-
- # Platform independent
- 'source/tb_capture_device.cc',
- 'source/tb_external_transport.cc',
- 'source/tb_I420_codec.cc',
- 'source/tb_interfaces.cc',
- 'source/tb_video_channel.cc',
- 'source/vie_autotest.cc',
- 'source/vie_autotest_base.cc',
- 'source/vie_autotest_capture.cc',
- 'source/vie_autotest_codec.cc',
- 'source/vie_autotest_encryption.cc',
- 'source/vie_autotest_file.cc',
- 'source/vie_autotest_image_process.cc',
- 'source/vie_autotest_loopback.cc',
- 'source/vie_autotest_main.cc',
- 'source/vie_autotest_network.cc',
- 'source/vie_autotest_render.cc',
- 'source/vie_autotest_rtp_rtcp.cc',
- 'source/vie_autotest_custom_call.cc',
- 'source/vie_autotest_simulcast.cc',
- 'source/vie_file_based_comparison_tests.cc',
-
- # Platform dependent
- # Linux
- 'source/vie_autotest_linux.cc',
- 'source/vie_window_manager_factory_linux.cc',
- # Mac
- 'source/vie_autotest_cocoa_mac.mm',
- 'source/vie_autotest_carbon_mac.cc',
- 'source/vie_window_manager_factory_mac.mm',
- # Windows
- 'source/vie_autotest_win.cc',
- 'source/vie_window_manager_factory_win.cc',
- ],
- 'conditions': [
- # TODO(andrew): this likely isn't an actual dependency. It should be
- # included in webrtc.gyp or video_engine.gyp instead.
- ['OS=="win"', {
- 'dependencies': [
- 'vie_win_test',
- ],
- }],
- ['OS=="linux"', {
- # TODO(andrew): these should be provided directly by the projects
- # # which require them instead.
- 'libraries': [
- '-lXext',
- '-lX11',
- ],
- }],
- ['OS=="mac"', {
- 'xcode_settings': {
- 'OTHER_LDFLAGS': [
- '-framework Foundation -framework AppKit -framework Cocoa -framework OpenGL -framework CoreVideo -framework CoreAudio -framework AudioToolbox',
- ],
- },
- }],
- ], # conditions
- },
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/video_engine/video_engine.gyp b/trunk/src/video_engine/video_engine.gyp
deleted file mode 100644
index 261292c..0000000
--- a/trunk/src/video_engine/video_engine.gyp
+++ /dev/null
@@ -1,25 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'includes': [
- '../build/common.gypi',
- './video_engine_core.gypi',
- ],
-
- # Test targets, excluded when building with Chromium.
- 'conditions': [
- ['build_with_chromium==0', {
- 'includes': [
- 'test/auto_test/vie_auto_test.gypi',
- 'main/test/WindowsTest/windowstest.gypi',
- ],
- }],
- ],
-}
-
diff --git a/trunk/src/video_engine/video_engine_core.gypi b/trunk/src/video_engine/video_engine_core.gypi
deleted file mode 100644
index 38015af..0000000
--- a/trunk/src/video_engine/video_engine_core.gypi
+++ /dev/null
@@ -1,160 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'video_engine_core',
- 'type': '<(library)',
- 'dependencies': [
-
- # common_video
- '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
- '<(webrtc_root)/common_video/common_video.gyp:webrtc_jpeg',
-
- # ModulesShared
- '<(webrtc_root)/modules/modules.gyp:media_file',
- '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
- '<(webrtc_root)/modules/modules.gyp:udp_transport',
- '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
-
- # ModulesVideo
- '<(webrtc_root)/modules/modules.gyp:video_capture_module',
- '<(webrtc_root)/modules/modules.gyp:webrtc_video_coding',
- '<(webrtc_root)/modules/modules.gyp:video_processing',
- '<(webrtc_root)/modules/modules.gyp:video_render_module',
-
- # VoiceEngine
- '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine_core',
-
- # system_wrappers
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- 'include',
- '../common_video/interface',
- '../modules/video_capture/main/interface',
- '../modules/video_render/main/interface',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- 'include',
- ],
- },
- 'sources': [
- # interface
- 'include/vie_base.h',
- 'include/vie_capture.h',
- 'include/vie_codec.h',
- 'include/vie_encryption.h',
- 'include/vie_errors.h',
- 'include/vie_external_codec.h',
- 'include/vie_file.h',
- 'include/vie_image_process.h',
- 'include/vie_network.h',
- 'include/vie_render.h',
- 'include/vie_rtp_rtcp.h',
-
- # headers
- 'vie_base_impl.h',
- 'vie_capture_impl.h',
- 'vie_codec_impl.h',
- 'vie_defines.h',
- 'vie_encryption_impl.h',
- 'vie_external_codec_impl.h',
- 'vie_file_impl.h',
- 'vie_image_process_impl.h',
- 'vie_impl.h',
- 'vie_network_impl.h',
- 'vie_ref_count.h',
- 'vie_remb.h',
- 'vie_render_impl.h',
- 'vie_rtp_rtcp_impl.h',
- 'vie_shared_data.h',
- 'vie_capturer.h',
- 'vie_channel.h',
- 'vie_channel_manager.h',
- 'vie_encoder.h',
- 'vie_file_image.h',
- 'vie_file_player.h',
- 'vie_file_recorder.h',
- 'vie_frame_provider_base.h',
- 'vie_input_manager.h',
- 'vie_manager_base.h',
- 'vie_performance_monitor.h',
- 'vie_receiver.h',
- 'vie_renderer.h',
- 'vie_render_manager.h',
- 'vie_sender.h',
- 'vie_sync_module.h',
-
- # ViE
- 'vie_base_impl.cc',
- 'vie_capture_impl.cc',
- 'vie_codec_impl.cc',
- 'vie_encryption_impl.cc',
- 'vie_external_codec_impl.cc',
- 'vie_file_impl.cc',
- 'vie_image_process_impl.cc',
- 'vie_impl.cc',
- 'vie_network_impl.cc',
- 'vie_ref_count.cc',
- 'vie_render_impl.cc',
- 'vie_rtp_rtcp_impl.cc',
- 'vie_shared_data.cc',
- 'vie_capturer.cc',
- 'vie_channel.cc',
- 'vie_channel_manager.cc',
- 'vie_encoder.cc',
- 'vie_file_image.cc',
- 'vie_file_player.cc',
- 'vie_file_recorder.cc',
- 'vie_frame_provider_base.cc',
- 'vie_input_manager.cc',
- 'vie_manager_base.cc',
- 'vie_performance_monitor.cc',
- 'vie_receiver.cc',
- 'vie_remb.cc',
- 'vie_renderer.cc',
- 'vie_render_manager.cc',
- 'vie_sender.cc',
- 'vie_sync_module.cc',
- ], # source
- },
- ], # targets
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'video_engine_core_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'video_engine_core',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../testing/gmock.gyp:gmock',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- ],
- 'include_dirs': [
- '..',
- '../modules/interface',
- '../modules/rtp_rtcp/interface',
- ],
- 'sources': [
- 'vie_remb_unittest.cc',
- ],
- },
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/video_engine/vie_base_impl.cc b/trunk/src/video_engine/vie_base_impl.cc
deleted file mode 100644
index 4f0eae2..0000000
--- a/trunk/src/video_engine/vie_base_impl.cc
+++ /dev/null
@@ -1,494 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-
-#include "critical_section_wrapper.h"
-#include "engine_configurations.h"
-#include "rtp_rtcp.h"
-#include "trace.h"
-#include "video_coding.h"
-#include "video_processing.h"
-#include "video_render.h"
-#include "vie_base_impl.h"
-#include "vie_channel.h"
-#include "vie_channel_manager.h"
-#include "vie_defines.h"
-#include "vie_encoder.h"
-#include "vie_errors.h"
-#include "vie_impl.h"
-#include "vie_input_manager.h"
-#include "vie_performance_monitor.h"
-#include "vie_shared_data.h"
-
-namespace webrtc {
-
-ViEBase* ViEBase::GetInterface(VideoEngine* video_engine) {
- if (!video_engine) {
- return NULL;
- }
- VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
- ViEBaseImpl* vie_base_impl = vie_impl;
- (*vie_base_impl)++; // Increase ref count.
-
- return vie_base_impl;
-}
-
-int ViEBaseImpl::Release() {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_.instance_id(),
- "ViEBase::Release()");
- (*this)--; // Decrease ref count.
-
- WebRtc_Word32 ref_count = GetCount();
- if (ref_count < 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_.instance_id(),
- "ViEBase release too many times");
- shared_data_.SetLastError(kViEAPIDoesNotExist);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_.instance_id(),
- "ViEBase reference count: %d", ref_count);
- return ref_count;
-}
-
-ViEBaseImpl::ViEBaseImpl() {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_.instance_id(),
- "ViEBaseImpl::ViEBaseImpl() Ctor");
-}
-
-ViEBaseImpl::~ViEBaseImpl() {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_.instance_id(),
- "ViEBaseImpl::ViEBaseImpl() Dtor");
-}
-
-int ViEBaseImpl::Init() {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_.instance_id(),
- "Init");
- if (shared_data_.Initialized()) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_.instance_id(),
- "Init called twice");
- return 0;
- }
-
- shared_data_.SetInitialized();
- return 0;
-}
-
-int ViEBaseImpl::SetVoiceEngine(VoiceEngine* voice_engine) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s", __FUNCTION__);
- if (!(shared_data_.Initialized())) {
- shared_data_.SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_.instance_id());
- return -1;
- }
-
- if (shared_data_.channel_manager()->SetVoiceEngine(voice_engine) != 0) {
- shared_data_.SetLastError(kViEBaseVoEFailure);
- return -1;
- }
- return 0;
-}
-
-int ViEBaseImpl::CreateChannel(int& video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s", __FUNCTION__);
-
- if (!(shared_data_.Initialized())) {
- shared_data_.SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_.instance_id());
- return -1;
- }
-
- if (shared_data_.channel_manager()->CreateChannel(video_channel) == -1) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s: Could not create channel", __FUNCTION__);
- video_channel = -1;
- shared_data_.SetLastError(kViEBaseChannelCreationFailed);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s: channel created: %d", __FUNCTION__, video_channel);
- return 0;
-}
-
-int ViEBaseImpl::CreateChannel(int& video_channel, int original_channel) {
- if (!(shared_data_.Initialized())) {
- shared_data_.SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_.instance_id());
- return -1;
- }
-
- ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
- if (!cs.Channel(original_channel)) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s - original_channel does not exist.", __FUNCTION__,
- shared_data_.instance_id());
- shared_data_.SetLastError(kViEBaseInvalidChannelId);
- return -1;
- }
-
- if (shared_data_.channel_manager()->CreateChannel(video_channel,
- original_channel) == -1) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s: Could not create channel", __FUNCTION__);
- video_channel = -1;
- shared_data_.SetLastError(kViEBaseChannelCreationFailed);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s: channel created: %d", __FUNCTION__, video_channel);
- return 0;
-}
-
-int ViEBaseImpl::DeleteChannel(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s(%d)", __FUNCTION__, video_channel);
-
- if (!(shared_data_.Initialized())) {
- shared_data_.SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_.instance_id());
- return -1;
- }
-
- {
- ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_.instance_id()),
- "%s: channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_.SetLastError(kViEBaseInvalidChannelId);
- return -1;
- }
-
- // Deregister the ViEEncoder if no other channel is using it.
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (cs.ChannelUsingViEEncoder(video_channel) == false) {
- ViEInputManagerScoped is(*(shared_data_.input_manager()));
- ViEFrameProviderBase* provider = is.FrameProvider(vie_encoder);
- if (provider) {
- provider->DeregisterFrameCallback(vie_encoder);
- }
- }
- }
-
- if (shared_data_.channel_manager()->DeleteChannel(video_channel) == -1) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s: Could not delete channel %d", __FUNCTION__,
- video_channel);
- shared_data_.SetLastError(kViEBaseUnknownError);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s: channel deleted: %d", __FUNCTION__, video_channel);
- return 0;
-}
-
-int ViEBaseImpl::ConnectAudioChannel(const int video_channel,
- const int audio_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s(%d)", __FUNCTION__, video_channel);
-
- if (!(shared_data_.Initialized())) {
- shared_data_.SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_.instance_id());
- return -1;
- }
-
- ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
- if (!cs.Channel(video_channel)) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s: channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_.SetLastError(kViEBaseInvalidChannelId);
- return -1;
- }
-
- if (shared_data_.channel_manager()->ConnectVoiceChannel(video_channel,
- audio_channel) != 0) {
- shared_data_.SetLastError(kViEBaseVoEFailure);
- return -1;
- }
- return 0;
-}
-
-int ViEBaseImpl::DisconnectAudioChannel(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s(%d)", __FUNCTION__, video_channel);
- if (!(shared_data_.Initialized())) {
- shared_data_.SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_.instance_id());
- return -1;
- }
- ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
- if (!cs.Channel(video_channel)) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s: channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_.SetLastError(kViEBaseInvalidChannelId);
- return -1;
- }
-
- if (shared_data_.channel_manager()->DisconnectVoiceChannel(
- video_channel) != 0) {
- shared_data_.SetLastError(kViEBaseVoEFailure);
- return -1;
- }
- return 0;
-}
-
-int ViEBaseImpl::StartSend(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_.instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_.instance_id(), video_channel),
- "%s: Channel %d does not exist", __FUNCTION__, video_channel);
- shared_data_.SetLastError(kViEBaseInvalidChannelId);
- return -1;
- }
-
- // Verify no other channel using the same encoder is sending.
- ChannelList channels;
- cs.ChannelsUsingViEEncoder(video_channel, &channels);
- for (ChannelList::iterator it = channels.begin(); it != channels.end();
- ++it) {
- if ((*it)->Sending()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_.instance_id(), video_channel),
- "A channel using this encoder is already synding");
- shared_data_.SetLastError(kViEBaseAlreadySending);
- return -1;
- }
- }
-
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- assert(false);
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_.instance_id(), video_channel),
- "%s: Could not find encoder for channel %d", __FUNCTION__,
- video_channel);
- return -1;
- }
-
- // Pause and trigger a key frame.
- vie_encoder->Pause();
- WebRtc_Word32 error = vie_channel->StartSend();
- if (error != 0) {
- vie_encoder->Restart();
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_.instance_id(), video_channel),
- "%s: Could not start sending on channel %d", __FUNCTION__,
- video_channel);
- if (error == kViEBaseAlreadySending) {
- shared_data_.SetLastError(kViEBaseAlreadySending);
- }
- shared_data_.SetLastError(kViEBaseUnknownError);
- return -1;
- }
- vie_encoder->SendKeyFrame();
- vie_encoder->Restart();
- return 0;
-}
-
-int ViEBaseImpl::StopSend(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_.instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_.instance_id(), video_channel),
- "%s: Channel %d does not exist", __FUNCTION__, video_channel);
- shared_data_.SetLastError(kViEBaseInvalidChannelId);
- return -1;
- }
-
- WebRtc_Word32 error = vie_channel->StopSend();
- if (error != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_.instance_id(), video_channel),
- "%s: Could not stop sending on channel %d", __FUNCTION__,
- video_channel);
- if (error == kViEBaseNotSending) {
- shared_data_.SetLastError(kViEBaseNotSending);
- } else {
- shared_data_.SetLastError(kViEBaseUnknownError);
- }
- return -1;
- }
- return 0;
-}
-
-int ViEBaseImpl::StartReceive(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_.instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_.instance_id(), video_channel),
- "%s: Channel %d does not exist", __FUNCTION__, video_channel);
- shared_data_.SetLastError(kViEBaseInvalidChannelId);
- return -1;
- }
- if (vie_channel->Receiving()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_.instance_id(), video_channel),
- "%s: Channel %d already receive.", __FUNCTION__,
- video_channel);
- shared_data_.SetLastError(kViEBaseAlreadyReceiving);
- return -1;
- }
- if (vie_channel->StartReceive() != 0) {
- shared_data_.SetLastError(kViEBaseUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViEBaseImpl::StopReceive(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_.instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_.channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_.instance_id(), video_channel),
- "%s: Channel %d does not exist", __FUNCTION__, video_channel);
- shared_data_.SetLastError(kViEBaseInvalidChannelId);
- return -1;
- }
- if (vie_channel->StopReceive() != 0) {
- shared_data_.SetLastError(kViEBaseUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViEBaseImpl::RegisterObserver(ViEBaseObserver& observer) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s", __FUNCTION__);
- if (shared_data_.vie_performance_monitor()->ViEBaseObserverRegistered()) {
- shared_data_.SetLastError(kViEBaseObserverAlreadyRegistered);
- return -1;
- }
- return shared_data_.vie_performance_monitor()->Init(&observer);
-}
-
-int ViEBaseImpl::DeregisterObserver() {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_.instance_id()),
- "%s", __FUNCTION__);
-
- if (!shared_data_.vie_performance_monitor()->ViEBaseObserverRegistered()) {
- shared_data_.SetLastError(kViEBaseObserverNotRegistered);
- WEBRTC_TRACE(kTraceError, kTraceVideo, shared_data_.instance_id(),
- "%s No observer registered.", __FUNCTION__);
- return -1;
- }
- shared_data_.vie_performance_monitor()->Terminate();
- return 0;
-}
-
-int ViEBaseImpl::GetVersion(char version[1024]) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_.instance_id()),
- "GetVersion(version=?)");
- assert(kViEVersionMaxMessageSize == 1024);
-
- if (!version) {
- shared_data_.SetLastError(kViEBaseInvalidArgument);
- return -1;
- }
-
- char version_buf[kViEVersionMaxMessageSize];
- char* version_ptr = version_buf;
-
- WebRtc_Word32 len = 0; // Does not include NULL termination.
- WebRtc_Word32 acc_len = 0;
-
- len = AddViEVersion(version_ptr);
- if (len == -1) {
- shared_data_.SetLastError(kViEBaseUnknownError);
- return -1;
- }
- version_ptr += len;
- acc_len += len;
- assert(acc_len < kViEVersionMaxMessageSize);
-
- len = AddBuildInfo(version_ptr);
- if (len == -1) {
- shared_data_.SetLastError(kViEBaseUnknownError);
- return -1;
- }
- version_ptr += len;
- acc_len += len;
- assert(acc_len < kViEVersionMaxMessageSize);
-
- len = AddExternalTransportBuild(version_ptr);
- if (len == -1) {
- shared_data_.SetLastError(kViEBaseUnknownError);
- return -1;
- }
- version_ptr += len;
- acc_len += len;
- assert(acc_len < kViEVersionMaxMessageSize);
-
- memcpy(version, version_buf, acc_len);
- version[acc_len] = '\0';
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo,
- ViEId(shared_data_.instance_id()), "GetVersion() => %s",
- version);
- return 0;
-}
-
-int ViEBaseImpl::LastError() {
- return shared_data_.LastErrorInternal();
-}
-
-WebRtc_Word32 ViEBaseImpl::AddBuildInfo(char* str) const {
- return sprintf(str, "Build: %s\n", BUILDINFO);
-}
-
-WebRtc_Word32 ViEBaseImpl::AddViEVersion(char* str) const {
- return sprintf(str, "VideoEngine 3.2.0\n");
-}
-
-WebRtc_Word32 ViEBaseImpl::AddExternalTransportBuild(char* str) const {
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
- return sprintf(str, "External transport build\n");
-#else
- return 0;
-#endif
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_base_impl.h b/trunk/src/video_engine/vie_base_impl.h
deleted file mode 100644
index a950300..0000000
--- a/trunk/src/video_engine/vie_base_impl.h
+++ /dev/null
@@ -1,96 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_BASE_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_BASE_IMPL_H_
-
-#include "vie_base.h"
-#include "vie_defines.h"
-#include "vie_ref_count.h"
-#include "vie_shared_data.h"
-
-namespace webrtc {
-
-class Module;
-class VoiceEngine;
-
-class ViEBaseImpl
- : public ViEBase,
- public ViERefCount {
- public:
- virtual int Release();
-
- // Initializes VideoEngine and must be called before any other API is called.
- virtual int Init();
-
- // Connects ViE to a VoE instance. Pass in NULL to forget about a previously
- // set voice engine and release all resources we allocated from it.
- virtual int SetVoiceEngine(VoiceEngine* voice_engine);
-
- // Creates a new ViE channel.
- virtual int CreateChannel(int& video_channel);
-
- // Creates a new ViE channel that will use the same capture device and encoder
- // as |original_channel|.
- virtual int CreateChannel(int& video_channel, int original_channel);
-
- // Deletes a ViE channel.
- virtual int DeleteChannel(const int video_channel);
-
- // Connects a ViE channel with a VoE channel.
- virtual int ConnectAudioChannel(const int video_channel,
- const int audio_channel);
-
- // Disconnects a video/voice channel pair.
- virtual int DisconnectAudioChannel(const int video_channel);
-
- // Starts sending on video_channel and also starts the encoder.
- virtual int StartSend(const int video_channel);
-
- // Stops sending on the specified channel.
- virtual int StopSend(const int video_channel);
-
- // Starts receiving on the channel and also start decoding.
- virtual int StartReceive(const int video_channel);
-
- // Stops receiving on the specified channel.
- virtual int StopReceive(const int video_channel);
-
- // Registers a customer implemented observer.
- virtual int RegisterObserver(ViEBaseObserver& observer);
-
- // Deregisters the observer.
- virtual int DeregisterObserver();
-
- // Prints version information into |version|.
- virtual int GetVersion(char version[1024]);
-
- // Returns the error code for the last registered error.
- virtual int LastError();
-
- protected:
- ViEBaseImpl();
- virtual ~ViEBaseImpl();
-
- ViESharedData* shared_data() { return &shared_data_; }
-
- private:
- // Version functions.
- WebRtc_Word32 AddViEVersion(char* str) const;
- WebRtc_Word32 AddBuildInfo(char* str) const;
- WebRtc_Word32 AddExternalTransportBuild(char* str) const;
-
- // ViEBaseImpl owns ViESharedData used by all interface implementations.
- ViESharedData shared_data_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_BASE_IMPL_H_
diff --git a/trunk/src/video_engine/vie_capture_impl.cc b/trunk/src/video_engine/vie_capture_impl.cc
deleted file mode 100644
index a94c29c..0000000
--- a/trunk/src/video_engine/vie_capture_impl.cc
+++ /dev/null
@@ -1,590 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_capture_impl.h"
-
-#include "system_wrappers/interface/trace.h"
-#include "video_engine/include/vie_errors.h"
-#include "video_engine/vie_capturer.h"
-#include "video_engine/vie_channel.h"
-#include "video_engine/vie_channel_manager.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_encoder.h"
-#include "video_engine/vie_impl.h"
-#include "video_engine/vie_input_manager.h"
-#include "video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-ViECapture* ViECapture::GetInterface(VideoEngine* video_engine) {
-#ifdef WEBRTC_VIDEO_ENGINE_CAPTURE_API
- if (!video_engine) {
- return NULL;
- }
- VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
- ViECaptureImpl* vie_capture_impl = vie_impl;
- // Increase ref count.
- (*vie_capture_impl)++;
- return vie_capture_impl;
-#else
- return NULL;
-#endif
-}
-
-int ViECaptureImpl::Release() {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
- "ViECapture::Release()");
- // Decrease ref count
- (*this)--;
-
- WebRtc_Word32 ref_count = GetCount();
- if (ref_count < 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
- "ViECapture release too many times");
- shared_data_->SetLastError(kViEAPIDoesNotExist);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
- "ViECapture reference count: %d", ref_count);
- return ref_count;
-}
-
-ViECaptureImpl::ViECaptureImpl(ViESharedData* shared_data)
- : shared_data_(shared_data) {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViECaptureImpl::ViECaptureImpl() Ctor");
-}
-
-ViECaptureImpl::~ViECaptureImpl() {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViECaptureImpl::~ViECaptureImpl() Dtor");
-}
-
-int ViECaptureImpl::NumberOfCaptureDevices() {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s", __FUNCTION__);
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
- return shared_data_->input_manager()->NumberOfCaptureDevices();
-}
-
-
-int ViECaptureImpl::GetCaptureDevice(unsigned int list_number,
- char* device_nameUTF8,
- unsigned int device_nameUTF8Length,
- char* unique_idUTF8,
- unsigned int unique_idUTF8Length) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(list_number: %d)", __FUNCTION__, list_number);
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
- return shared_data_->input_manager()->GetDeviceName(
- list_number,
- reinterpret_cast<WebRtc_UWord8*>(device_nameUTF8), device_nameUTF8Length,
- reinterpret_cast<WebRtc_UWord8*>(unique_idUTF8), unique_idUTF8Length);
-}
-
-int ViECaptureImpl::AllocateCaptureDevice(
- const char* unique_idUTF8,
- const unsigned int unique_idUTF8Length,
- int& capture_id) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(unique_idUTF8: %s)", __FUNCTION__, unique_idUTF8);
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
- const WebRtc_Word32 result =
- shared_data_->input_manager()->CreateCaptureDevice(
- reinterpret_cast<const WebRtc_UWord8*>(unique_idUTF8),
- static_cast<const WebRtc_UWord32>(unique_idUTF8Length), capture_id);
- if (result != 0) {
- shared_data_->SetLastError(result);
- return -1;
- }
- return 0;
-}
-
-int ViECaptureImpl::AllocateExternalCaptureDevice(
- int& capture_id, ViEExternalCapture*& external_capture) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s", __FUNCTION__);
-
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
- const WebRtc_Word32 result =
- shared_data_->input_manager()->CreateExternalCaptureDevice(
- external_capture, capture_id);
-
- if (result != 0) {
- shared_data_->SetLastError(result);
- return -1;
- }
- return 0;
-}
-
-int ViECaptureImpl::AllocateCaptureDevice(VideoCaptureModule& capture_module,
- int& capture_id) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s", __FUNCTION__);
-
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
- const WebRtc_Word32 result =
- shared_data_->input_manager()->CreateCaptureDevice(capture_module,
- capture_id);
- if (result != 0) {
- shared_data_->SetLastError(result);
- return -1;
- }
- return 0;
-}
-
-
-int ViECaptureImpl::ReleaseCaptureDevice(const int capture_id) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(capture_id: %d)", __FUNCTION__, capture_id);
- {
- ViEInputManagerScoped is((*(shared_data_->input_manager())));
- ViECapturer* vie_capture = is.Capture(capture_id);
- if (!vie_capture) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Capture device %d doesn't exist", __FUNCTION__,
- capture_id);
- shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
- return -1;
- }
- }
-
- // Destroy the capture device.
- return shared_data_->input_manager()->DestroyCaptureDevice(capture_id);
-}
-
-int ViECaptureImpl::ConnectCaptureDevice(const int capture_id,
- const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(capture_id: %d, video_channel: %d)", __FUNCTION__,
- capture_id, video_channel);
-
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViECapturer* vie_capture = is.Capture(capture_id);
- if (!vie_capture) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Capture device %d doesn't exist", __FUNCTION__,
- capture_id);
- shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
- return -1;
- }
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViECaptureDeviceInvalidChannelId);
- return -1;
- }
- // Check if the encoder already has a connected frame provider
- if (is.FrameProvider(vie_encoder) != NULL) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d already connected to a capture device.",
- __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECaptureDeviceAlreadyConnected);
- return -1;
- }
- VideoCodec codec;
- bool use_hardware_encoder = false;
- if (vie_encoder->GetEncoder(codec) == 0) {
- // Try to provide the encoder with pre-encoded frames if possible.
- if (vie_capture->PreEncodeToViEEncoder(codec, *vie_encoder,
- video_channel) == 0) {
- use_hardware_encoder = true;
- }
- }
- // If we don't use the camera as hardware encoder, we register the vie_encoder
- // for callbacks.
- if (!use_hardware_encoder &&
- vie_capture->RegisterFrameCallback(video_channel, vie_encoder) != 0) {
- shared_data_->SetLastError(kViECaptureDeviceUnknownError);
- return -1;
- }
- return 0;
-}
-
-
-int ViECaptureImpl::DisconnectCaptureDevice(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id()),
- "%s: Channel %d doesn't exist", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViECaptureDeviceInvalidChannelId);
- return -1;
- }
-
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViEFrameProviderBase* frame_provider = is.FrameProvider(vie_encoder);
- if (!frame_provider) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo,
- ViEId(shared_data_->instance_id()),
- "%s: No capture device connected to channel %d",
- __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECaptureDeviceNotConnected);
- return -1;
- }
- if (frame_provider->Id() < kViECaptureIdBase ||
- frame_provider->Id() > kViECaptureIdMax) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo,
- ViEId(shared_data_->instance_id()),
- "%s: No capture device connected to channel %d",
- __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECaptureDeviceNotConnected);
- return -1;
- }
-
- if (frame_provider->DeregisterFrameCallback(vie_encoder) != 0) {
- shared_data_->SetLastError(kViECaptureDeviceUnknownError);
- return -1;
- }
-
- return 0;
-}
-
-int ViECaptureImpl::StartCapture(const int capture_id,
- const CaptureCapability& capture_capability) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(capture_id: %d)", __FUNCTION__, capture_id);
-
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViECapturer* vie_capture = is.Capture(capture_id);
- if (!vie_capture) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), capture_id),
- "%s: Capture device %d doesn't exist", __FUNCTION__,
- capture_id);
- shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
- return -1;
- }
- if (vie_capture->Started()) {
- shared_data_->SetLastError(kViECaptureDeviceAlreadyStarted);
- return -1;
- }
- if (vie_capture->Start(capture_capability) != 0) {
- shared_data_->SetLastError(kViECaptureDeviceUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViECaptureImpl::StopCapture(const int capture_id) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(capture_id: %d)", __FUNCTION__, capture_id);
-
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViECapturer* vie_capture = is.Capture(capture_id);
- if (!vie_capture) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), capture_id),
- "%s: Capture device %d doesn't exist", __FUNCTION__,
- capture_id);
- shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
- return -1;
- }
- if (!vie_capture->Started()) {
- shared_data_->SetLastError(kViECaptureDeviceNotStarted);
- return -1;
- }
- if (vie_capture->Stop() != 0) {
- shared_data_->SetLastError(kViECaptureDeviceUnknownError);
- return -1;
- }
-
- return 0;
-}
-
-int ViECaptureImpl::SetRotateCapturedFrames(
- const int capture_id,
- const RotateCapturedFrame rotation) {
- int i_rotation = -1;
- switch (rotation) {
- case RotateCapturedFrame_0:
- i_rotation = 0;
- break;
- case RotateCapturedFrame_90:
- i_rotation = 90;
- break;
- case RotateCapturedFrame_180:
- i_rotation = 180;
- break;
- case RotateCapturedFrame_270:
- i_rotation = 270;
- break;
- }
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(rotation: %d)", __FUNCTION__, i_rotation);
-
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViECapturer* vie_capture = is.Capture(capture_id);
- if (!vie_capture) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), capture_id),
- "%s: Capture device %d doesn't exist", __FUNCTION__,
- capture_id);
- shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
- return -1;
- }
- if (vie_capture->SetRotateCapturedFrames(rotation) != 0) {
- shared_data_->SetLastError(kViECaptureDeviceUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViECaptureImpl::SetCaptureDelay(const int capture_id,
- const unsigned int capture_delay_ms) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(capture_id: %d, capture_delay_ms %u)", __FUNCTION__,
- capture_id, capture_delay_ms);
-
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViECapturer* vie_capture = is.Capture(capture_id);
- if (!vie_capture) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), capture_id),
- "%s: Capture device %d doesn't exist", __FUNCTION__,
- capture_id);
- shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
- return -1;
- }
-
- if (vie_capture->SetCaptureDelay(capture_delay_ms) != 0) {
- shared_data_->SetLastError(kViECaptureDeviceUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViECaptureImpl::NumberOfCapabilities(
- const char* unique_idUTF8,
- const unsigned int unique_idUTF8Length) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(capture_device_name: %s)", __FUNCTION__, unique_idUTF8);
-
-#if defined(WEBRTC_MAC_INTEL)
- // TODO(mflodman) Move to capture module!
- // QTKit framework handles all capabilities and capture settings
- // automatically (mandatory).
- // Thus this function cannot be supported on the Mac platform.
- shared_data_->SetLastError(kViECaptureDeviceMacQtkitNotSupported);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s This API is not supported on Mac OS", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
-#endif
-
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
- return shared_data_->input_manager()->NumberOfCaptureCapabilities(
- reinterpret_cast<const WebRtc_UWord8*>(unique_idUTF8));
-}
-
-
-int ViECaptureImpl::GetCaptureCapability(const char* unique_idUTF8,
- const unsigned int unique_idUTF8Length,
- const unsigned int capability_number,
- CaptureCapability& capability) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(capture_device_name: %s)", __FUNCTION__, unique_idUTF8);
-
-#if defined(WEBRTC_MAC_INTEL)
- // TODO(mflodman) Move to capture module!
- // QTKit framework handles all capabilities and capture settings
- // automatically (mandatory).
- // Thus this function cannot be supported on the Mac platform.
- shared_data_->SetLastError(kViECaptureDeviceMacQtkitNotSupported);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s This API is not supported on Mac OS", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
-#endif
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
- if (shared_data_->input_manager()->GetCaptureCapability(
- reinterpret_cast<const WebRtc_UWord8*>(unique_idUTF8),
- capability_number, capability) != 0) {
- shared_data_->SetLastError(kViECaptureDeviceUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViECaptureImpl::ShowCaptureSettingsDialogBox(
- const char* unique_idUTF8,
- const unsigned int unique_idUTF8Length,
- const char* dialog_title,
- void* parent_window,
- const unsigned int x,
- const unsigned int y) {
-#if defined(WEBRTC_MAC_INTEL)
- // TODO(mflodman) Move to capture module
- // QTKit framework handles all capabilities and capture settings
- // automatically (mandatory).
- // Thus this function cannot be supported on the Mac platform.
- shared_data_->SetLastError(kViECaptureDeviceMacQtkitNotSupported);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s This API is not supported on Mac OS", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
-#endif
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s capture_id (capture_device_name: %s)", __FUNCTION__,
- unique_idUTF8);
-
- return shared_data_->input_manager()->DisplayCaptureSettingsDialogBox(
- reinterpret_cast<const WebRtc_UWord8*>(unique_idUTF8),
- reinterpret_cast<const WebRtc_UWord8*>(dialog_title),
- parent_window, x, y);
-}
-
-int ViECaptureImpl::GetOrientation(const char* unique_idUTF8,
- RotateCapturedFrame& orientation) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s (capture_device_name: %s)", __FUNCTION__, unique_idUTF8);
-
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
- if (shared_data_->input_manager()->GetOrientation(
- reinterpret_cast<const WebRtc_UWord8*>(unique_idUTF8),
- orientation) != 0) {
- shared_data_->SetLastError(kViECaptureDeviceUnknownError);
- return -1;
- }
- return 0;
-}
-
-
-int ViECaptureImpl::EnableBrightnessAlarm(const int capture_id,
- const bool enable) {
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViECapturer* vie_capture = is.Capture(capture_id);
- if (!vie_capture) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), capture_id),
- "%s: Capture device %d doesn't exist", __FUNCTION__,
- capture_id);
- shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
- return -1;
- }
- if (vie_capture->EnableBrightnessAlarm(enable) != 0) {
- shared_data_->SetLastError(kViECaptureDeviceUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViECaptureImpl::RegisterObserver(const int capture_id,
- ViECaptureObserver& observer) {
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViECapturer* vie_capture = is.Capture(capture_id);
- if (!vie_capture) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), capture_id),
- "%s: Capture device %d doesn't exist", __FUNCTION__,
- capture_id);
- shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
- return -1;
- }
- if (vie_capture->IsObserverRegistered()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), capture_id),
- "%s: Observer already registered", __FUNCTION__);
- shared_data_->SetLastError(kViECaptureObserverAlreadyRegistered);
- return -1;
- }
- if (vie_capture->RegisterObserver(observer) != 0) {
- shared_data_->SetLastError(kViECaptureDeviceUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViECaptureImpl::DeregisterObserver(const int capture_id) {
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViECapturer* vie_capture = is.Capture(capture_id);
- if (!vie_capture) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), capture_id),
- "%s: Capture device %d doesn't exist", __FUNCTION__,
- capture_id);
- shared_data_->SetLastError(kViECaptureDeviceDoesNotExist);
- return -1;
- }
- if (!vie_capture->IsObserverRegistered()) {
- shared_data_->SetLastError(kViECaptureDeviceObserverNotRegistered);
- return -1;
- }
-
- if (vie_capture->DeRegisterObserver() != 0) {
- shared_data_->SetLastError(kViECaptureDeviceUnknownError);
- return -1;
- }
- return 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_capture_impl.h b/trunk/src/video_engine/vie_capture_impl.h
deleted file mode 100644
index ca88a06..0000000
--- a/trunk/src/video_engine/vie_capture_impl.h
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_CAPTURE_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_CAPTURE_IMPL_H_
-
-#include "typedefs.h"
-#include "video_engine/include/vie_capture.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_ref_count.h"
-
-namespace webrtc {
-
-class ViESharedData;
-
-class ViECaptureImpl
- : public ViECapture,
- public ViERefCount {
- public:
- // Implements ViECapture.
- virtual int Release();
- virtual int NumberOfCaptureDevices();
- virtual int GetCaptureDevice(unsigned int list_number, char* device_nameUTF8,
- const unsigned int device_nameUTF8Length,
- char* unique_idUTF8,
- const unsigned int unique_idUTF8Length);
- virtual int AllocateCaptureDevice(const char* unique_idUTF8,
- const unsigned int unique_idUTF8Length,
- int& capture_id);
- virtual int AllocateCaptureDevice(VideoCaptureModule& capture_module,
- int& capture_id);
- virtual int AllocateExternalCaptureDevice(
- int& capture_id, ViEExternalCapture *&external_capture);
- virtual int ReleaseCaptureDevice(const int capture_id);
-
- virtual int ConnectCaptureDevice(const int capture_id,
- const int video_channel);
- virtual int DisconnectCaptureDevice(const int video_channel);
- virtual int StartCapture(
- const int capture_id,
- const CaptureCapability& capture_capability = CaptureCapability());
- virtual int StopCapture(const int capture_id);
- virtual int SetRotateCapturedFrames(const int capture_id,
- const RotateCapturedFrame rotation);
- virtual int SetCaptureDelay(const int capture_id,
- const unsigned int capture_delay_ms);
- virtual int NumberOfCapabilities(const char* unique_idUTF8,
- const unsigned int unique_idUTF8Length);
- virtual int GetCaptureCapability(const char* unique_idUTF8,
- const unsigned int unique_idUTF8Length,
- const unsigned int capability_number,
- CaptureCapability& capability);
- virtual int ShowCaptureSettingsDialogBox(
- const char* unique_idUTF8, const unsigned int unique_idUTF8Length,
- const char* dialog_title, void* parent_window = NULL,
- const unsigned int x = 200, const unsigned int y = 200);
- virtual int GetOrientation(const char* unique_idUTF8,
- RotateCapturedFrame& orientation);
- virtual int EnableBrightnessAlarm(const int capture_id, const bool enable);
- virtual int RegisterObserver(const int capture_id,
- ViECaptureObserver& observer);
- virtual int DeregisterObserver(const int capture_id);
-
- protected:
- ViECaptureImpl(ViESharedData* shared_data);
- virtual ~ViECaptureImpl();
-
- private:
- ViESharedData* shared_data_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_CAPTURE_IMPL_H_
diff --git a/trunk/src/video_engine/vie_capturer.cc b/trunk/src/video_engine/vie_capturer.cc
deleted file mode 100644
index cd9bd0c..0000000
--- a/trunk/src/video_engine/vie_capturer.cc
+++ /dev/null
@@ -1,900 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_capturer.h"
-
-#include "modules/interface/module_common_types.h"
-#include "modules/utility/interface/process_thread.h"
-#include "modules/video_capture/main/interface/video_capture_factory.h"
-#include "modules/video_processing/main/interface/video_processing.h"
-#include "modules/video_render/main/interface/video_render_defines.h"
-#include "system_wrappers/interface/critical_section_wrapper.h"
-#include "system_wrappers/interface/event_wrapper.h"
-#include "system_wrappers/interface/thread_wrapper.h"
-#include "system_wrappers/interface/trace.h"
-#include "video_engine/include/vie_image_process.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_encoder.h"
-
-namespace webrtc {
-
-const int kThreadWaitTimeMs = 100;
-const int kMaxDeliverWaitTime = 500;
-
-ViECapturer::ViECapturer(int capture_id,
- int engine_id,
- ProcessThread& module_process_thread)
- : ViEFrameProviderBase(capture_id, engine_id),
- capture_cs_(CriticalSectionWrapper::CreateCriticalSection()),
- deliver_cs_(CriticalSectionWrapper::CreateCriticalSection()),
- capture_module_(NULL),
- external_capture_module_(NULL),
- module_process_thread_(module_process_thread),
- capture_id_(capture_id),
- capture_thread_(*ThreadWrapper::CreateThread(ViECaptureThreadFunction,
- this, kHighPriority,
- "ViECaptureThread")),
- capture_event_(*EventWrapper::Create()),
- deliver_event_(*EventWrapper::Create()),
- effect_filter_(NULL),
- image_proc_module_(NULL),
- image_proc_module_ref_counter_(0),
- deflicker_frame_stats_(NULL),
- brightness_frame_stats_(NULL),
- current_brightness_level_(Normal),
- reported_brightness_level_(Normal),
- denoising_enabled_(false),
- observer_cs_(CriticalSectionWrapper::CreateCriticalSection()),
- observer_(NULL),
- encoding_cs_(CriticalSectionWrapper::CreateCriticalSection()),
- capture_encoder_(NULL),
- encode_complete_callback_(NULL),
- vie_encoder_(NULL),
- vcm_(NULL),
- decoder_initialized_(false) {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, ViEId(engine_id, capture_id),
- "ViECapturer::ViECapturer(capture_id: %d, engine_id: %d)",
- capture_id, engine_id);
- unsigned int t_id = 0;
- if (capture_thread_.Start(t_id)) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id, capture_id),
- "%s: thread started: %u", __FUNCTION__, t_id);
- } else {
- assert(false);
- }
-}
-
-ViECapturer::~ViECapturer() {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, ViEId(engine_id_, capture_id_),
- "ViECapturer::~ViECapturer() - capture_id: %d, engine_id: %d",
- capture_id_, engine_id_);
-
- // Stop the thread.
- deliver_cs_->Enter();
- capture_cs_->Enter();
- capture_thread_.SetNotAlive();
- capture_event_.Set();
- capture_cs_->Leave();
- deliver_cs_->Leave();
-
- provider_cs_->Enter();
- if (vie_encoder_) {
- vie_encoder_->DeRegisterExternalEncoder(codec_.plType);
- }
- provider_cs_->Leave();
-
- // Stop the camera input.
- if (capture_module_) {
- module_process_thread_.DeRegisterModule(capture_module_);
- capture_module_->DeRegisterCaptureDataCallback();
- capture_module_->Release();
- capture_module_ = NULL;
- }
- if (capture_thread_.Stop()) {
- // Thread stopped.
- delete &capture_thread_;
- delete &capture_event_;
- delete &deliver_event_;
- } else {
- assert(false);
- WEBRTC_TRACE(kTraceMemory, kTraceVideoRenderer,
- ViEId(engine_id_, capture_id_),
- "%s: Not able to stop capture thread for device %d, leaking",
- __FUNCTION__, capture_id_);
- }
-
- if (image_proc_module_) {
- VideoProcessingModule::Destroy(image_proc_module_);
- }
- if (deflicker_frame_stats_) {
- delete deflicker_frame_stats_;
- deflicker_frame_stats_ = NULL;
- }
- delete brightness_frame_stats_;
- if (vcm_) {
- delete vcm_;
- }
-}
-
-ViECapturer* ViECapturer::CreateViECapture(
- int capture_id,
- int engine_id,
- VideoCaptureModule& capture_module,
- ProcessThread& module_process_thread) {
- ViECapturer* capture = new ViECapturer(capture_id, engine_id,
- module_process_thread);
- if (!capture || capture->Init(capture_module) != 0) {
- delete capture;
- capture = NULL;
- }
- return capture;
-}
-
-WebRtc_Word32 ViECapturer::Init(VideoCaptureModule& capture_module) {
- assert(capture_module_ == NULL);
- capture_module_ = &capture_module;
- capture_module_->RegisterCaptureDataCallback(*this);
- capture_module_->AddRef();
- if (module_process_thread_.RegisterModule(capture_module_) != 0) {
- return -1;
- }
-
- return 0;
-}
-
-ViECapturer* ViECapturer::CreateViECapture(
- int capture_id,
- int engine_id,
- const WebRtc_UWord8* device_unique_idUTF8,
- const WebRtc_UWord32 device_unique_idUTF8Length,
- ProcessThread& module_process_thread) {
- ViECapturer* capture = new ViECapturer(capture_id, engine_id,
- module_process_thread);
- if (!capture ||
- capture->Init(device_unique_idUTF8, device_unique_idUTF8Length) != 0) {
- delete capture;
- capture = NULL;
- }
- return capture;
-}
-
-WebRtc_Word32 ViECapturer::Init(
- const WebRtc_UWord8* device_unique_idUTF8,
- const WebRtc_UWord32 device_unique_idUTF8Length) {
- assert(capture_module_ == NULL);
- if (device_unique_idUTF8 == NULL) {
- capture_module_ = VideoCaptureFactory::Create(
- ViEModuleId(engine_id_, capture_id_), external_capture_module_);
- } else {
- capture_module_ = VideoCaptureFactory::Create(
- ViEModuleId(engine_id_, capture_id_), device_unique_idUTF8);
- }
- if (!capture_module_) {
- return -1;
- }
- capture_module_->AddRef();
- capture_module_->RegisterCaptureDataCallback(*this);
- if (module_process_thread_.RegisterModule(capture_module_) != 0) {
- return -1;
- }
-
- return 0;
-}
-
-int ViECapturer::FrameCallbackChanged() {
- if (Started() && !EncoderActive() && !CaptureCapabilityFixed()) {
- // Reconfigure the camera if a new size is required and the capture device
- // does not provide encoded frames.
- int best_width;
- int best_height;
- int best_frame_rate;
- VideoCaptureCapability capture_settings;
- capture_module_->CaptureSettings(capture_settings);
- GetBestFormat(best_width, best_height, best_frame_rate);
- if (best_width != 0 && best_height != 0 && best_frame_rate != 0) {
- if (best_width != capture_settings.width ||
- best_height != capture_settings.height ||
- best_frame_rate != capture_settings.maxFPS ||
- capture_settings.codecType != kVideoCodecUnknown) {
- Stop();
- Start(requested_capability_);
- }
- }
- }
- return 0;
-}
-
-WebRtc_Word32 ViECapturer::Start(const CaptureCapability& capture_capability) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_), "%s",
- __FUNCTION__);
- int width;
- int height;
- int frame_rate;
- VideoCaptureCapability capability;
- requested_capability_ = capture_capability;
- if (EncoderActive()) {
- CriticalSectionScoped cs(encoding_cs_.get());
- capability.width = codec_.width;
- capability.height = codec_.height;
- capability.maxFPS = codec_.maxFramerate;
- capability.codecType = codec_.codecType;
- capability.rawType = kVideoI420;
-
- } else if (!CaptureCapabilityFixed()) {
- // Ask the observers for best size.
- GetBestFormat(width, height, frame_rate);
- if (width == 0) {
- width = kViECaptureDefaultWidth;
- }
- if (height == 0) {
- height = kViECaptureDefaultHeight;
- }
- if (frame_rate == 0) {
- frame_rate = kViECaptureDefaultFramerate;
- }
- capability.height = height;
- capability.width = width;
- capability.maxFPS = frame_rate;
- capability.rawType = kVideoI420;
- capability.codecType = kVideoCodecUnknown;
- } else {
- // Width, height and type specified with call to Start, not set by
- // observers.
- capability.width = requested_capability_.width;
- capability.height = requested_capability_.height;
- capability.maxFPS = requested_capability_.maxFPS;
- capability.rawType = requested_capability_.rawType;
- capability.interlaced = requested_capability_.interlaced;
- }
- return capture_module_->StartCapture(capability);
-}
-
-WebRtc_Word32 ViECapturer::Stop() {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_), "%s",
- __FUNCTION__);
- requested_capability_ = CaptureCapability();
- return capture_module_->StopCapture();
-}
-
-bool ViECapturer::Started() {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_), "%s",
- __FUNCTION__);
- return capture_module_->CaptureStarted();
-}
-
-const WebRtc_UWord8* ViECapturer::CurrentDeviceName() const {
- return capture_module_->CurrentDeviceName();
-}
-
-WebRtc_Word32 ViECapturer::SetCaptureDelay(WebRtc_Word32 delay_ms) {
- return capture_module_->SetCaptureDelay(delay_ms);
-}
-
-WebRtc_Word32 ViECapturer::SetRotateCapturedFrames(
- const RotateCapturedFrame rotation) {
- VideoCaptureRotation converted_rotation = kCameraRotate0;
- switch (rotation) {
- case RotateCapturedFrame_0:
- converted_rotation = kCameraRotate0;
- break;
- case RotateCapturedFrame_90:
- converted_rotation = kCameraRotate90;
- break;
- case RotateCapturedFrame_180:
- converted_rotation = kCameraRotate180;
- break;
- case RotateCapturedFrame_270:
- converted_rotation = kCameraRotate270;
- break;
- }
- return capture_module_->SetCaptureRotation(converted_rotation);
-}
-
-int ViECapturer::IncomingFrame(unsigned char* video_frame,
- unsigned int video_frame_length,
- unsigned short width,
- unsigned short height,
- RawVideoType video_type,
- unsigned long long capture_time) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
- "ExternalCapture::IncomingFrame width %d, height %d, "
- "capture_time %u", width, height, capture_time);
-
- if (!external_capture_module_) {
- return -1;
- }
- VideoCaptureCapability capability;
- capability.width = width;
- capability.height = height;
- capability.rawType = video_type;
- return external_capture_module_->IncomingFrame(video_frame,
- video_frame_length,
- capability, capture_time);
-}
-
-int ViECapturer::IncomingFrameI420(const ViEVideoFrameI420& video_frame,
- unsigned long long capture_time) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
- "ExternalCapture::IncomingFrame width %d, height %d, "
- " capture_time %u", video_frame.width, video_frame.height,
- capture_time);
-
- if (!external_capture_module_) {
- return -1;
- }
-
- VideoFrameI420 frame;
- frame.width = video_frame.width;
- frame.height = video_frame.height;
- frame.y_plane = video_frame.y_plane;
- frame.u_plane = video_frame.u_plane;
- frame.v_plane = video_frame.v_plane;
- frame.y_pitch = video_frame.y_pitch;
- frame.u_pitch = video_frame.u_pitch;
- frame.v_pitch = video_frame.v_pitch;
-
- return external_capture_module_->IncomingFrameI420(frame, capture_time);
-}
-
-void ViECapturer::OnIncomingCapturedFrame(const WebRtc_Word32 capture_id,
- VideoFrame& video_frame,
- VideoCodecType codec_type) {
- WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s(capture_id: %d)", __FUNCTION__, capture_id);
-
- CriticalSectionScoped cs(capture_cs_.get());
- if (codec_type != kVideoCodecUnknown) {
- if (encoded_frame_.Length() != 0) {
- // The last encoded frame has not been sent yet. Need to wait.
- deliver_event_.Reset();
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s(capture_id: %d) Last encoded frame not yet delivered.",
- __FUNCTION__, capture_id);
- capture_cs_->Leave();
- // Wait for the coded frame to be sent before unblocking this.
- deliver_event_.Wait(kMaxDeliverWaitTime);
- assert(encoded_frame_.Length() == 0);
- capture_cs_->Enter();
- }
- encoded_frame_.SwapFrame(video_frame);
- } else {
- captured_frame_.SwapFrame(video_frame);
- }
- capture_event_.Set();
- return;
-}
-
-void ViECapturer::OnCaptureDelayChanged(const WebRtc_Word32 id,
- const WebRtc_Word32 delay) {
- WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s(capture_id: %d) delay %d", __FUNCTION__, capture_id_,
- delay);
-
- // Deliver the network delay to all registered callbacks.
- ViEFrameProviderBase::SetFrameDelay(delay);
- CriticalSectionScoped cs(encoding_cs_.get());
- if (vie_encoder_) {
- vie_encoder_->DelayChanged(id, delay);
- }
-}
-
-WebRtc_Word32 ViECapturer::RegisterEffectFilter(
- ViEEffectFilter* effect_filter) {
- CriticalSectionScoped cs(deliver_cs_.get());
-
- if (!effect_filter) {
- if (!effect_filter_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s: no effect filter added for capture device %d",
- __FUNCTION__, capture_id_);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s: deregister effect filter for device %d", __FUNCTION__,
- capture_id_);
- } else {
- if (effect_filter_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s: effect filter already added for capture device %d",
- __FUNCTION__, capture_id_);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s: register effect filter for device %d", __FUNCTION__,
- capture_id_);
- }
- effect_filter_ = effect_filter;
- return 0;
-}
-
-WebRtc_Word32 ViECapturer::IncImageProcRefCount() {
- if (!image_proc_module_) {
- assert(image_proc_module_ref_counter_ == 0);
- image_proc_module_ = VideoProcessingModule::Create(
- ViEModuleId(engine_id_, capture_id_));
- if (!image_proc_module_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s: could not create video processing module",
- __FUNCTION__);
- return -1;
- }
- }
- image_proc_module_ref_counter_++;
- return 0;
-}
-
-WebRtc_Word32 ViECapturer::DecImageProcRefCount() {
- image_proc_module_ref_counter_--;
- if (image_proc_module_ref_counter_ == 0) {
- // Destroy module.
- VideoProcessingModule::Destroy(image_proc_module_);
- image_proc_module_ = NULL;
- }
- return 0;
-}
-
-WebRtc_Word32 ViECapturer::EnableDenoising(bool enable) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s(capture_device_id: %d, enable: %d)", __FUNCTION__,
- capture_id_, enable);
-
- CriticalSectionScoped cs(deliver_cs_.get());
- if (enable) {
- if (denoising_enabled_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s: denoising already enabled", __FUNCTION__);
- return -1;
- }
- denoising_enabled_ = true;
- if (IncImageProcRefCount() != 0) {
- return -1;
- }
- } else {
- if (denoising_enabled_ == false) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s: denoising not enabled", __FUNCTION__);
- return -1;
- }
- denoising_enabled_ = false;
- DecImageProcRefCount();
- }
-
- return 0;
-}
-
-WebRtc_Word32 ViECapturer::EnableDeflickering(bool enable) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s(capture_device_id: %d, enable: %d)", __FUNCTION__,
- capture_id_, enable);
-
- CriticalSectionScoped cs(deliver_cs_.get());
- if (enable) {
- if (deflicker_frame_stats_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s: deflickering already enabled", __FUNCTION__);
- return -1;
- }
- if (IncImageProcRefCount() != 0) {
- return -1;
- }
- deflicker_frame_stats_ = new VideoProcessingModule::FrameStats();
- } else {
- if (deflicker_frame_stats_ == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s: deflickering not enabled", __FUNCTION__);
- return -1;
- }
- DecImageProcRefCount();
- delete deflicker_frame_stats_;
- deflicker_frame_stats_ = NULL;
- }
- return 0;
-}
-
-WebRtc_Word32 ViECapturer::EnableBrightnessAlarm(bool enable) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s(capture_device_id: %d, enable: %d)", __FUNCTION__,
- capture_id_, enable);
-
- CriticalSectionScoped cs(deliver_cs_.get());
- if (enable) {
- if (brightness_frame_stats_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s: BrightnessAlarm already enabled", __FUNCTION__);
- return -1;
- }
- if (IncImageProcRefCount() != 0) {
- return -1;
- }
- brightness_frame_stats_ = new VideoProcessingModule::FrameStats();
- } else {
- DecImageProcRefCount();
- if (brightness_frame_stats_ == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s: deflickering not enabled", __FUNCTION__);
- return -1;
- }
- delete brightness_frame_stats_;
- brightness_frame_stats_ = NULL;
- }
- return 0;
-}
-
-bool ViECapturer::ViECaptureThreadFunction(void* obj) {
- return static_cast<ViECapturer*>(obj)->ViECaptureProcess();
-}
-
-bool ViECapturer::ViECaptureProcess() {
- if (capture_event_.Wait(kThreadWaitTimeMs) == kEventSignaled) {
- deliver_cs_->Enter();
- if (captured_frame_.Length() > 0) {
- // New I420 frame.
- capture_cs_->Enter();
- deliver_frame_.SwapFrame(captured_frame_);
- captured_frame_.SetLength(0);
- capture_cs_->Leave();
- DeliverI420Frame(deliver_frame_);
- }
- if (encoded_frame_.Length() > 0) {
- capture_cs_->Enter();
- deliver_frame_.SwapFrame(encoded_frame_);
- encoded_frame_.SetLength(0);
- deliver_event_.Set();
- capture_cs_->Leave();
- DeliverCodedFrame(deliver_frame_);
- }
- deliver_cs_->Leave();
- if (current_brightness_level_ != reported_brightness_level_) {
- CriticalSectionScoped cs(observer_cs_.get());
- if (observer_) {
- observer_->BrightnessAlarm(id_, current_brightness_level_);
- reported_brightness_level_ = current_brightness_level_;
- }
- }
- }
- // We're done!
- return true;
-}
-
-void ViECapturer::DeliverI420Frame(VideoFrame& video_frame) {
- // Apply image enhancement and effect filter.
- if (deflicker_frame_stats_) {
- if (image_proc_module_->GetFrameStats(*deflicker_frame_stats_,
- video_frame) == 0) {
- image_proc_module_->Deflickering(video_frame, *deflicker_frame_stats_);
- } else {
- WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s: could not get frame stats for captured frame",
- __FUNCTION__);
- }
- }
- if (denoising_enabled_) {
- image_proc_module_->Denoising(video_frame);
- }
- if (brightness_frame_stats_) {
- if (image_proc_module_->GetFrameStats(*brightness_frame_stats_,
- video_frame) == 0) {
- WebRtc_Word32 brightness = image_proc_module_->BrightnessDetection(
- video_frame, *brightness_frame_stats_);
-
- switch (brightness) {
- case VideoProcessingModule::kNoWarning:
- current_brightness_level_ = Normal;
- break;
- case VideoProcessingModule::kDarkWarning:
- current_brightness_level_ = Dark;
- break;
- case VideoProcessingModule::kBrightWarning:
- current_brightness_level_ = Bright;
- break;
- default:
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s: Brightness detection failed", __FUNCTION__);
- }
- }
- }
- if (effect_filter_) {
- effect_filter_->Transform(video_frame.Length(), video_frame.Buffer(),
- video_frame.TimeStamp(), video_frame.Width(),
- video_frame.Height());
- }
- // Deliver the captured frame to all observers (channels, renderer or file).
- ViEFrameProviderBase::DeliverFrame(video_frame);
-}
-
-void ViECapturer::DeliverCodedFrame(VideoFrame& video_frame) {
- if (encode_complete_callback_) {
- EncodedImage encoded_image(video_frame.Buffer(), video_frame.Length(),
- video_frame.Size());
- encoded_image._timeStamp = 90 * (WebRtc_UWord32) video_frame.RenderTimeMs();
- encode_complete_callback_->Encoded(encoded_image);
- }
-
- if (NumberOfRegisteredFrameCallbacks() > 0 && decoder_initialized_) {
- video_frame.Swap(decode_buffer_.payloadData, decode_buffer_.bufferSize,
- decode_buffer_.payloadSize);
- decode_buffer_.encodedHeight = video_frame.Height();
- decode_buffer_.encodedWidth = video_frame.Width();
- decode_buffer_.renderTimeMs = video_frame.RenderTimeMs();
- decode_buffer_.timeStamp = 90 * (WebRtc_UWord32) video_frame.RenderTimeMs();
- decode_buffer_.payloadType = codec_.plType;
- vcm_->DecodeFromStorage(decode_buffer_);
- }
-}
-
-int ViECapturer::DeregisterFrameCallback(
- const ViEFrameCallback* callbackObject) {
- provider_cs_->Enter();
- if (callbackObject == vie_encoder_) {
- // Don't use this camera as encoder anymore. Need to tell the ViEEncoder.
- ViEEncoder* vie_encoder = NULL;
- vie_encoder = vie_encoder_;
- vie_encoder_ = NULL;
- provider_cs_->Leave();
-
- // Need to take this here in order to avoid deadlock with VCM. The reason is
- // that VCM will call ::Release and a deadlock can occur.
- deliver_cs_->Enter();
- vie_encoder->DeRegisterExternalEncoder(codec_.plType);
- deliver_cs_->Leave();
- return 0;
- }
- provider_cs_->Leave();
- return ViEFrameProviderBase::DeregisterFrameCallback(callbackObject);
-}
-
-bool ViECapturer::IsFrameCallbackRegistered(
- const ViEFrameCallback* callbackObject) {
- CriticalSectionScoped cs(provider_cs_.get());
- if (callbackObject == vie_encoder_) {
- return true;
- }
- return ViEFrameProviderBase::IsFrameCallbackRegistered(callbackObject);
-}
-
-WebRtc_Word32 ViECapturer::PreEncodeToViEEncoder(const VideoCodec& codec,
- ViEEncoder& vie_encoder,
- WebRtc_Word32 vie_encoder_id) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s(capture_device_id: %d)", __FUNCTION__, capture_id_);
- if (vie_encoder_ && &vie_encoder != vie_encoder_) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s(capture_device_id: %d Capture device already encoding)",
- __FUNCTION__, capture_id_);
- return -1;
- }
-
- CriticalSectionScoped cs(encoding_cs_.get());
- VideoCaptureModule::VideoCaptureEncodeInterface* capture_encoder =
- capture_module_->GetEncodeInterface(codec);
- if (!capture_encoder) {
- // Encoding not supported?
- return -1;
- }
- capture_encoder_ = capture_encoder;
-
- // Create VCM module used for decoding frames if needed.
- if (!vcm_) {
- vcm_ = VideoCodingModule::Create(capture_id_);
- }
-
- if (vie_encoder.RegisterExternalEncoder(this, codec.plType) != 0) {
- return -1;
- }
- if (vie_encoder.SetEncoder(codec) != 0) {
- vie_encoder.DeRegisterExternalEncoder(codec.plType);
- return -1;
- }
-
- // Make sure the encoder is not an I420 observer.
- ViEFrameProviderBase::DeregisterFrameCallback(&vie_encoder);
- // Store the vie_encoder using this capture device.
- vie_encoder_ = &vie_encoder;
- vie_encoder_id_ = vie_encoder_id;
- memcpy(&codec_, &codec, sizeof(VideoCodec));
- return 0;
-}
-
-bool ViECapturer::EncoderActive() {
- return vie_encoder_ != NULL;
-}
-
-bool ViECapturer::CaptureCapabilityFixed() {
- return requested_capability_.width != 0 &&
- requested_capability_.height != 0 &&
- requested_capability_.maxFPS != 0;
-}
-
-WebRtc_Word32 ViECapturer::Version(WebRtc_Word8* version,
- WebRtc_Word32 length) const {
- return 0;
-}
-
-WebRtc_Word32 ViECapturer::InitEncode(const VideoCodec* codec_settings,
- WebRtc_Word32 number_of_cores,
- WebRtc_UWord32 max_payload_size) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s(capture_device_id: %d)", __FUNCTION__, capture_id_);
-
- CriticalSectionScoped cs(encoding_cs_.get());
- if (!capture_encoder_ || !codec_settings) {
- return WEBRTC_VIDEO_CODEC_ERROR;
- }
-
- if (vcm_) {
- // Initialize VCM to be able to decode frames if needed.
- if (vcm_->InitializeReceiver() == 0) {
- if (vcm_->RegisterReceiveCallback(this) == 0) {
- if (vcm_->RegisterReceiveCodec(codec_settings, number_of_cores,
- false) == 0) {
- decoder_initialized_ = true;
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s(capture_device_id: %d) VCM Decoder initialized",
- __FUNCTION__, capture_id_);
- }
- }
- }
- }
- return capture_encoder_->ConfigureEncoder(*codec_settings, max_payload_size);
-}
-
-WebRtc_Word32 ViECapturer::Encode(const RawImage& input_image,
- const CodecSpecificInfo* codec_specific_info,
- const VideoFrameType* frame_types) {
- CriticalSectionScoped cs(encoding_cs_.get());
- if (!capture_encoder_) {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
- if (*frame_types == kKeyFrame) {
- return capture_encoder_->EncodeFrameType(kVideoFrameKey);
- }
- if (*frame_types == kSkipFrame) {
- return capture_encoder_->EncodeFrameType(kFrameEmpty);
- }
- return WEBRTC_VIDEO_CODEC_ERR_PARAMETER;
-}
-
-WebRtc_Word32 ViECapturer::RegisterEncodeCompleteCallback(
- EncodedImageCallback* callback) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s(capture_device_id: %d)", __FUNCTION__, capture_id_);
-
- CriticalSectionScoped cs(deliver_cs_.get());
- if (!capture_encoder_) {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
- encode_complete_callback_ = callback;
- return 0;
-}
-
-WebRtc_Word32 ViECapturer::Release() {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s(capture_device_id: %d)", __FUNCTION__, capture_id_);
- {
- CriticalSectionScoped cs(deliver_cs_.get());
- encode_complete_callback_ = NULL;
- }
-
- {
- CriticalSectionScoped cs(encoding_cs_.get());
-
- decoder_initialized_ = false;
- codec_.codecType = kVideoCodecUnknown;
- // Reset the camera to output I420.
- capture_encoder_->ConfigureEncoder(codec_, 0);
-
- if (vie_encoder_) {
- // Need to add the encoder as an observer of I420.
- ViEFrameProviderBase::RegisterFrameCallback(vie_encoder_id_,
- vie_encoder_);
- }
- vie_encoder_ = NULL;
- }
- return 0;
-}
-
-// Should reset the capture device to the state it was in after the InitEncode
-// function. Current implementation do nothing.
-WebRtc_Word32 ViECapturer::Reset() {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s(capture_device_id: %d)", __FUNCTION__, capture_id_);
- return 0;
-}
-
-WebRtc_Word32 ViECapturer::SetChannelParameters(WebRtc_UWord32 packet_loss,
- int rtt) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s(capture_device_id: %d)", __FUNCTION__, capture_id_);
-
- CriticalSectionScoped cs(encoding_cs_.get());
- if (!capture_encoder_) {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
- return capture_encoder_->SetChannelParameters(packet_loss, rtt);
-}
-
-WebRtc_Word32 ViECapturer::SetRates(WebRtc_UWord32 new_bit_rate,
- WebRtc_UWord32 frame_rate) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s(capture_device_id: %d)", __FUNCTION__, capture_id_);
-
- CriticalSectionScoped cs(encoding_cs_.get());
- if (!capture_encoder_) {
- return WEBRTC_VIDEO_CODEC_UNINITIALIZED;
- }
- return capture_encoder_->SetRates(new_bit_rate, frame_rate);
-}
-
-WebRtc_Word32 ViECapturer::FrameToRender(VideoFrame& video_frame) {
- deliver_cs_->Enter();
- DeliverI420Frame(video_frame);
- deliver_cs_->Leave();
- return 0;
-}
-
-WebRtc_Word32 ViECapturer::RegisterObserver(ViECaptureObserver& observer) {
- if (observer_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s Observer already registered", __FUNCTION__, capture_id_);
- return -1;
- }
- if (capture_module_->RegisterCaptureCallback(*this) != 0) {
- return -1;
- }
- capture_module_->EnableFrameRateCallback(true);
- capture_module_->EnableNoPictureAlarm(true);
- observer_ = &observer;
- return 0;
-}
-
-WebRtc_Word32 ViECapturer::DeRegisterObserver() {
- CriticalSectionScoped cs(observer_cs_.get());
- if (!observer_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, capture_id_),
- "%s No observer registered", __FUNCTION__, capture_id_);
- return -1;
- }
- capture_module_->EnableFrameRateCallback(false);
- capture_module_->EnableNoPictureAlarm(false);
- capture_module_->DeRegisterCaptureCallback();
- observer_ = NULL;
- return 0;
-}
-
-bool ViECapturer::IsObserverRegistered() {
- CriticalSectionScoped cs(observer_cs_.get());
- return observer_ != NULL;
-}
-
-void ViECapturer::OnCaptureFrameRate(const WebRtc_Word32 id,
- const WebRtc_UWord32 frame_rate) {
- WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, capture_id_),
- "OnCaptureFrameRate %d", frame_rate);
-
- CriticalSectionScoped cs(observer_cs_.get());
- observer_->CapturedFrameRate(id_, (WebRtc_UWord8) frame_rate);
-}
-
-void ViECapturer::OnNoPictureAlarm(const WebRtc_Word32 id,
- const VideoCaptureAlarm alarm) {
- WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, capture_id_),
- "OnNoPictureAlarm %d", alarm);
-
- CriticalSectionScoped cs(observer_cs_.get());
- CaptureAlarm vie_alarm = (alarm == Raised) ? AlarmRaised : AlarmCleared;
- observer_->NoPictureAlarm(id, vie_alarm);
-}
-
-WebRtc_Word32 ViECapturer::SetCaptureDeviceImage(
- const VideoFrame& capture_device_image) {
- return capture_module_->StartSendImage(capture_device_image, 10);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_capturer.h b/trunk/src/video_engine/vie_capturer.h
deleted file mode 100644
index da68278..0000000
--- a/trunk/src/video_engine/vie_capturer.h
+++ /dev/null
@@ -1,222 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_CAPTURER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_CAPTURER_H_
-
-#include "common_types.h"
-#include "engine_configurations.h"
-#include "modules/video_capture/main/interface/video_capture.h"
-#include "modules/video_coding/codecs/interface/video_codec_interface.h"
-#include "modules/video_coding/main/interface/video_coding.h"
-#include "modules/video_processing/main/interface/video_processing.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-#include "typedefs.h"
-#include "video_engine/include/vie_capture.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_frame_provider_base.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class EventWrapper;
-class ProcessThread;
-class ThreadWrapper;
-class ViEEffectFilter;
-class ViEEncoder;
-struct ViEPicture;
-
-class ViECapturer
- : public ViEFrameProviderBase,
- public ViEExternalCapture,
- protected VCMReceiveCallback,
- protected VideoCaptureDataCallback,
- protected VideoCaptureFeedBack,
- protected VideoEncoder {
- public:
- static ViECapturer* CreateViECapture(int capture_id,
- int engine_id,
- VideoCaptureModule& capture_module,
- ProcessThread& module_process_thread);
-
- static ViECapturer* CreateViECapture(
- int capture_id,
- int engine_id,
- const WebRtc_UWord8* device_unique_idUTF8,
- WebRtc_UWord32 device_unique_idUTF8Length,
- ProcessThread& module_process_thread);
-
- ~ViECapturer();
-
- // Implements ViEFrameProviderBase.
- int FrameCallbackChanged();
- virtual int DeregisterFrameCallback(const ViEFrameCallback* callbackObject);
- bool IsFrameCallbackRegistered(const ViEFrameCallback* callbackObject);
-
- // Implements ExternalCapture.
- virtual int IncomingFrame(unsigned char* video_frame,
- unsigned int video_frame_length,
- unsigned short width, unsigned short height,
- RawVideoType video_type,
- unsigned long long capture_time = 0);
-
- virtual int IncomingFrameI420(const ViEVideoFrameI420& video_frame,
- unsigned long long capture_time = 0);
-
- // Use this capture device as encoder.
- // Returns 0 if the codec is supported by this capture device.
- virtual WebRtc_Word32 PreEncodeToViEEncoder(const VideoCodec& codec,
- ViEEncoder& vie_encoder,
- WebRtc_Word32 vie_encoder_id);
-
- // Start/Stop.
- WebRtc_Word32 Start(
- const CaptureCapability& capture_capability = CaptureCapability());
- WebRtc_Word32 Stop();
- bool Started();
-
- // Overrides the capture delay.
- WebRtc_Word32 SetCaptureDelay(WebRtc_Word32 delay_ms);
-
- // Sets rotation of the incoming captured frame.
- WebRtc_Word32 SetRotateCapturedFrames(const RotateCapturedFrame rotation);
-
- // Effect filter.
- WebRtc_Word32 RegisterEffectFilter(ViEEffectFilter* effect_filter);
- WebRtc_Word32 EnableDenoising(bool enable);
- WebRtc_Word32 EnableDeflickering(bool enable);
- WebRtc_Word32 EnableBrightnessAlarm(bool enable);
-
- // Statistics observer.
- WebRtc_Word32 RegisterObserver(ViECaptureObserver& observer);
- WebRtc_Word32 DeRegisterObserver();
- bool IsObserverRegistered();
-
- // Information.
- const WebRtc_UWord8* CurrentDeviceName() const;
-
- // Set device image.
- WebRtc_Word32 SetCaptureDeviceImage(const VideoFrame& capture_device_image);
-
- protected:
- ViECapturer(int capture_id,
- int engine_id,
- ProcessThread& module_process_thread);
-
- WebRtc_Word32 Init(VideoCaptureModule& capture_module);
- WebRtc_Word32 Init(const WebRtc_UWord8* device_unique_idUTF8,
- const WebRtc_UWord32 device_unique_idUTF8Length);
-
- // Implements VideoCaptureDataCallback.
- virtual void OnIncomingCapturedFrame(const WebRtc_Word32 id,
- VideoFrame& video_frame,
- VideoCodecType codec_type);
- virtual void OnCaptureDelayChanged(const WebRtc_Word32 id,
- const WebRtc_Word32 delay);
-
- bool EncoderActive();
-
- // Returns true if the capture capability has been set in |StartCapture|
- // function and may not be changed.
- bool CaptureCapabilityFixed();
-
- // Help function used for keeping track of VideoImageProcesingModule.
- // Creates the module if it is needed, returns 0 on success and guarantees
- // that the image proc module exist.
- WebRtc_Word32 IncImageProcRefCount();
- WebRtc_Word32 DecImageProcRefCount();
-
- // Implements VideoEncoder.
- virtual WebRtc_Word32 Version(WebRtc_Word8* version,
- WebRtc_Word32 length) const;
- virtual WebRtc_Word32 InitEncode(const VideoCodec* codec_settings,
- WebRtc_Word32 number_of_cores,
- WebRtc_UWord32 max_payload_size);
- virtual WebRtc_Word32 Encode(const RawImage& input_image,
- const CodecSpecificInfo* codec_specific_info,
- const VideoFrameType* frame_types);
- virtual WebRtc_Word32 RegisterEncodeCompleteCallback(
- EncodedImageCallback* callback);
- virtual WebRtc_Word32 Release();
- virtual WebRtc_Word32 Reset();
- virtual WebRtc_Word32 SetChannelParameters(WebRtc_UWord32 packet_loss,
- int rtt);
- virtual WebRtc_Word32 SetRates(WebRtc_UWord32 new_bit_rate,
- WebRtc_UWord32 frame_rate);
-
- // Implements VCMReceiveCallback.
- virtual WebRtc_Word32 FrameToRender(VideoFrame& video_frame);
-
- // Implements VideoCaptureFeedBack
- virtual void OnCaptureFrameRate(const WebRtc_Word32 id,
- const WebRtc_UWord32 frame_rate);
- virtual void OnNoPictureAlarm(const WebRtc_Word32 id,
- const VideoCaptureAlarm alarm);
-
- // Thread functions for deliver captured frames to receivers.
- static bool ViECaptureThreadFunction(void* obj);
- bool ViECaptureProcess();
-
- void DeliverI420Frame(VideoFrame& video_frame);
- void DeliverCodedFrame(VideoFrame& video_frame);
-
- private:
- // Never take capture_cs_ before deliver_cs_!
- scoped_ptr<CriticalSectionWrapper> capture_cs_;
- scoped_ptr<CriticalSectionWrapper> deliver_cs_;
- VideoCaptureModule* capture_module_;
- VideoCaptureExternal* external_capture_module_;
- ProcessThread& module_process_thread_;
- const int capture_id_;
-
- // Capture thread.
- ThreadWrapper& capture_thread_;
- EventWrapper& capture_event_;
- EventWrapper& deliver_event_;
-
- VideoFrame captured_frame_;
- VideoFrame deliver_frame_;
- VideoFrame encoded_frame_;
-
- // Image processing.
- ViEEffectFilter* effect_filter_;
- VideoProcessingModule* image_proc_module_;
- int image_proc_module_ref_counter_;
- VideoProcessingModule::FrameStats* deflicker_frame_stats_;
- VideoProcessingModule::FrameStats* brightness_frame_stats_;
- Brightness current_brightness_level_;
- Brightness reported_brightness_level_;
- bool denoising_enabled_;
-
- // Statistics observer.
- scoped_ptr<CriticalSectionWrapper> observer_cs_;
- ViECaptureObserver* observer_;
-
- // Encoding using encoding capable cameras.
- scoped_ptr<CriticalSectionWrapper> encoding_cs_;
- VideoCaptureModule::VideoCaptureEncodeInterface* capture_encoder_;
- EncodedImageCallback* encode_complete_callback_;
- VideoCodec codec_;
- // The ViEEncoder we are encoding for.
- ViEEncoder* vie_encoder_;
- // ViEEncoder id we are encoding for.
- WebRtc_Word32 vie_encoder_id_;
- // Used for decoding preencoded frames.
- VideoCodingModule* vcm_;
- EncodedVideoData decode_buffer_;
- bool decoder_initialized_;
- CaptureCapability requested_capability_;
-
- VideoFrame capture_device_image_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_CAPTURER_H_
diff --git a/trunk/src/video_engine/vie_channel.cc b/trunk/src/video_engine/vie_channel.cc
deleted file mode 100644
index e6f7a0a..0000000
--- a/trunk/src/video_engine/vie_channel.cc
+++ /dev/null
@@ -1,2529 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_channel.h"
-
-#include <algorithm>
-#include <vector>
-
-#include "modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "modules/udp_transport/interface/udp_transport.h"
-#include "modules/utility/interface/process_thread.h"
-#include "modules/video_coding/main/interface/video_coding.h"
-#include "modules/video_processing/main/interface/video_processing.h"
-#include "modules/video_render/main/interface/video_render_defines.h"
-#include "system_wrappers/interface/critical_section_wrapper.h"
-#include "system_wrappers/interface/thread_wrapper.h"
-#include "system_wrappers/interface/trace.h"
-#include "video_engine/include/vie_codec.h"
-#include "video_engine/include/vie_errors.h"
-#include "video_engine/include/vie_image_process.h"
-#include "video_engine/include/vie_rtp_rtcp.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_receiver.h"
-#include "video_engine/vie_sender.h"
-#include "video_engine/vie_sync_module.h"
-
-namespace webrtc {
-
-const int kMaxDecodeWaitTimeMs = 50;
-
-ViEChannel::ViEChannel(WebRtc_Word32 channel_id,
- WebRtc_Word32 engine_id,
- WebRtc_UWord32 number_of_cores,
- ProcessThread& module_process_thread)
- : ViEFrameProviderBase(channel_id, engine_id),
- channel_id_(channel_id),
- engine_id_(engine_id),
- number_of_cores_(number_of_cores),
- num_socket_threads_(kViESocketThreads),
- callback_cs_(CriticalSectionWrapper::CreateCriticalSection()),
- rtp_rtcp_(*RtpRtcp::CreateRtpRtcp(ViEModuleId(engine_id, channel_id),
- false)),
- default_rtp_rtcp_(NULL),
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- socket_transport_(*UdpTransport::Create(
- ViEModuleId(engine_id, channel_id), num_socket_threads_)),
-#endif
- vcm_(*VideoCodingModule::Create(ViEModuleId(engine_id, channel_id))),
- vie_receiver_(*(new ViEReceiver(engine_id, channel_id, rtp_rtcp_, vcm_))),
- vie_sender_(*(new ViESender(engine_id, channel_id))),
- vie_sync_(*(new ViESyncModule(ViEId(engine_id, channel_id), vcm_,
- rtp_rtcp_))),
- module_process_thread_(module_process_thread),
- codec_observer_(NULL),
- do_key_frame_callbackRequest_(false),
- rtp_observer_(NULL),
- rtcp_observer_(NULL),
- networkObserver_(NULL),
- rtp_packet_timeout_(false),
- using_packet_spread_(false),
- external_transport_(NULL),
- decoder_reset_(true),
- wait_for_key_frame_(false),
- decode_thread_(NULL),
- external_encryption_(NULL),
- effect_filter_(NULL),
- color_enhancement_(true),
- vcm_rttreported_(TickTime::Now()),
- file_recorder_(channel_id) {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, ViEId(engine_id, channel_id),
- "ViEChannel::ViEChannel(channel_id: %d, engine_id: %d)",
- channel_id, engine_id);
-}
-
-WebRtc_Word32 ViEChannel::Init() {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: channel_id: %d, engine_id: %d)", __FUNCTION__, channel_id_,
- engine_id_);
- // RTP/RTCP initialization.
- if (rtp_rtcp_.InitSender() != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: RTP::InitSender failure", __FUNCTION__);
- return -1;
- }
- if (rtp_rtcp_.SetSendingMediaStatus(false) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: RTP::SetSendingMediaStatus failure", __FUNCTION__);
- return -1;
- }
- if (rtp_rtcp_.InitReceiver() != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: RTP::InitReceiver failure", __FUNCTION__);
- return -1;
- }
- if (rtp_rtcp_.RegisterIncomingDataCallback(
- static_cast<RtpData*>(&vie_receiver_)) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: RTP::RegisterIncomingDataCallback failure", __FUNCTION__);
- return -1;
- }
- if (rtp_rtcp_.RegisterSendTransport(
- static_cast<Transport*>(&vie_sender_)) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: RTP::RegisterSendTransport failure", __FUNCTION__);
- return -1;
- }
- if (module_process_thread_.RegisterModule(&rtp_rtcp_) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: RTP::RegisterModule failure", __FUNCTION__);
- return -1;
- }
- if (rtp_rtcp_.SetKeyFrameRequestMethod(kKeyFrameReqFirRtp) != 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: RTP::SetKeyFrameRequestMethod failure", __FUNCTION__);
- }
- if (rtp_rtcp_.SetRTCPStatus(kRtcpCompound) != 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: RTP::SetRTCPStatus failure", __FUNCTION__);
- }
- if (rtp_rtcp_.RegisterIncomingRTPCallback(this) != 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: RTP::RegisterIncomingRTPCallback failure", __FUNCTION__);
- return -1;
- }
- if (rtp_rtcp_.RegisterIncomingRTCPCallback(this) != 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: RTP::RegisterIncomingRTCPCallback failure", __FUNCTION__);
- return -1;
- }
-
- // VCM initialization
- if (vcm_.InitializeReceiver() != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: VCM::InitializeReceiver failure", __FUNCTION__);
- return -1;
- }
- if (vcm_.RegisterReceiveCallback(this) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: VCM::RegisterReceiveCallback failure", __FUNCTION__);
- return -1;
- }
- if (vcm_.RegisterFrameTypeCallback(this) != 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: VCM::RegisterFrameTypeCallback failure", __FUNCTION__);
- }
- if (vcm_.RegisterReceiveStatisticsCallback(this) != 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: VCM::RegisterReceiveStatisticsCallback failure",
- __FUNCTION__);
- }
- if (vcm_.SetRenderDelay(kViEDefaultRenderDelayMs) != 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: VCM::SetRenderDelay failure", __FUNCTION__);
- }
- if (module_process_thread_.RegisterModule(&vcm_) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: VCM::RegisterModule(vcm) failure", __FUNCTION__);
- return -1;
- }
-#ifdef VIDEOCODEC_VP8
- VideoCodec video_codec;
- if (vcm_.Codec(kVideoCodecVP8, &video_codec) == VCM_OK) {
- rtp_rtcp_.RegisterSendPayload(video_codec);
- rtp_rtcp_.RegisterReceivePayload(video_codec);
- vcm_.RegisterReceiveCodec(&video_codec, number_of_cores_);
- vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
- rtp_rtcp_.MaxDataPayloadLength());
- } else {
- assert(false);
- }
-#endif
-
- return 0;
-}
-
-ViEChannel::~ViEChannel() {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, ViEId(engine_id_, channel_id_),
- "ViEChannel Destructor, channel_id: %d, engine_id: %d",
- channel_id_, engine_id_);
-
- // Make sure we don't get more callbacks from the RTP module.
- rtp_rtcp_.RegisterIncomingRTPCallback(NULL);
- rtp_rtcp_.RegisterSendTransport(NULL);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- socket_transport_.StopReceiving();
-#endif
- module_process_thread_.DeRegisterModule(&rtp_rtcp_);
- module_process_thread_.DeRegisterModule(&vcm_);
- module_process_thread_.DeRegisterModule(&vie_sync_);
- while (simulcast_rtp_rtcp_.size() > 0) {
- std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
- RtpRtcp* rtp_rtcp = *it;
- rtp_rtcp->RegisterIncomingRTCPCallback(NULL);
- rtp_rtcp->RegisterSendTransport(NULL);
- module_process_thread_.DeRegisterModule(rtp_rtcp);
- RtpRtcp::DestroyRtpRtcp(rtp_rtcp);
- simulcast_rtp_rtcp_.erase(it);
- }
- if (decode_thread_) {
- StopDecodeThread();
- }
-
- delete &vie_receiver_;
- delete &vie_sender_;
- delete &vie_sync_;
-
- // Release modules.
- RtpRtcp::DestroyRtpRtcp(&rtp_rtcp_);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- UdpTransport::Destroy(&socket_transport_);
-#endif
- VideoCodingModule::Destroy(&vcm_);
-}
-
-WebRtc_Word32 ViEChannel::SetSendCodec(const VideoCodec& video_codec,
- bool new_stream) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: codec_type: %d", __FUNCTION__, video_codec.codecType);
-
- if (video_codec.codecType == kVideoCodecRED ||
- video_codec.codecType == kVideoCodecULPFEC) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: codec_type: %d is not a valid send codec.", __FUNCTION__,
- video_codec.codecType);
- return -1;
- }
- if (kMaxSimulcastStreams < video_codec.numberOfSimulcastStreams) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Too many simulcast streams", __FUNCTION__);
- return -1;
- }
- // Update the RTP module with the settings.
- // Stop and Start the RTP module -> trigger new SSRC, if an SSRC hasn't been
- // set explicitly.
- bool restart_rtp = false;
- if (rtp_rtcp_.Sending() && new_stream) {
- restart_rtp = true;
- rtp_rtcp_.SetSendingStatus(false);
- }
- if (video_codec.numberOfSimulcastStreams > 0) {
- WebRtc_UWord32 start_bitrate = video_codec.startBitrate * 1000;
- WebRtc_UWord32 stream_bitrate =
- std::min(start_bitrate, video_codec.simulcastStream[0].maxBitrate);
- start_bitrate -= stream_bitrate;
- // Set correct bitrate to base layer.
- rtp_rtcp_.SetSendBitrate(stream_bitrate, video_codec.minBitrate,
- video_codec.simulcastStream[0].maxBitrate);
- // Create our simulcast RTP modules.
- for (int i = simulcast_rtp_rtcp_.size();
- i < video_codec.numberOfSimulcastStreams - 1;
- i++) {
- RtpRtcp* rtp_rtcp = RtpRtcp::CreateRtpRtcp(
- ViEModuleId(engine_id_, channel_id_), false);
- if (rtp_rtcp->RegisterDefaultModule(default_rtp_rtcp_)) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not register default module", __FUNCTION__);
- return -1;
- }
- simulcast_rtp_rtcp_.push_back(rtp_rtcp);
- }
- // Remove last in list if we have too many.
- for (int j = simulcast_rtp_rtcp_.size();
- j > (video_codec.numberOfSimulcastStreams - 1);
- j--) {
- RtpRtcp* rtp_rtcp = simulcast_rtp_rtcp_.back();
- rtp_rtcp->RegisterIncomingRTCPCallback(NULL);
- rtp_rtcp->RegisterSendTransport(NULL);
- module_process_thread_.DeRegisterModule(rtp_rtcp);
- RtpRtcp::DestroyRtpRtcp(rtp_rtcp);
- simulcast_rtp_rtcp_.pop_back();
- }
- VideoCodec video_codec;
- if (vcm_.Codec(kVideoCodecVP8, &video_codec) != VCM_OK) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: VCM: failure geting default VP8 pl_type", __FUNCTION__);
- return -1;
- }
- WebRtc_UWord8 idx = 0;
- // Configure all simulcast modules.
- for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
- it != simulcast_rtp_rtcp_.end();
- it++) {
- idx++;
- RtpRtcp* rtp_rtcp = *it;
- if (rtp_rtcp->InitSender() != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: RTP::InitSender failure", __FUNCTION__);
- return -1;
- }
- if (rtp_rtcp->InitReceiver() != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: RTP::InitReceiver failure", __FUNCTION__);
- return -1;
- }
- if (rtp_rtcp->RegisterSendTransport(
- static_cast<Transport*>(&vie_sender_)) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: RTP::RegisterSendTransport failure", __FUNCTION__);
- return -1;
- }
- // Silently ignore error.
- module_process_thread_.RegisterModule(rtp_rtcp);
- if (rtp_rtcp->SetRTCPStatus(rtp_rtcp_.RTCP()) != 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: RTP::SetRTCPStatus failure", __FUNCTION__);
- }
- rtp_rtcp->DeRegisterSendPayload(video_codec.plType);
- if (rtp_rtcp->RegisterSendPayload(video_codec) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not register payload type", __FUNCTION__);
- return -1;
- }
- if (restart_rtp) {
- rtp_rtcp->SetSendingStatus(true);
- }
- // Configure all simulcast streams min and max bitrates
- const WebRtc_UWord32 stream_bitrate =
- std::min(start_bitrate, video_codec.simulcastStream[idx].maxBitrate);
- start_bitrate -= stream_bitrate;
- rtp_rtcp->SetSendBitrate(stream_bitrate, video_codec.minBitrate,
- video_codec.simulcastStream[idx].maxBitrate);
- }
- vie_receiver_.RegisterSimulcastRtpRtcpModules(simulcast_rtp_rtcp_);
- } else {
- if (!simulcast_rtp_rtcp_.empty()) {
- // Delete all simulcast rtp modules.
- while (!simulcast_rtp_rtcp_.empty()) {
- RtpRtcp* rtp_rtcp = simulcast_rtp_rtcp_.back();
- rtp_rtcp->RegisterIncomingRTCPCallback(NULL);
- rtp_rtcp->RegisterSendTransport(NULL);
- module_process_thread_.DeRegisterModule(rtp_rtcp);
- RtpRtcp::DestroyRtpRtcp(rtp_rtcp);
- simulcast_rtp_rtcp_.pop_back();
- }
- }
- // Clear any previous modules.
- vie_receiver_.RegisterSimulcastRtpRtcpModules(simulcast_rtp_rtcp_);
-
- rtp_rtcp_.SetSendBitrate(video_codec.startBitrate * 1000,
- video_codec.minBitrate,
- video_codec.maxBitrate);
- }
- // Enable this if H264 is available.
- // This sets the wanted packetization mode.
- // if (video_codec.plType == kVideoCodecH264) {
- // if (video_codec.codecSpecific.H264.packetization == kH264SingleMode) {
- // rtp_rtcp_.SetH264PacketizationMode(H264_SINGLE_NAL_MODE);
- // } else {
- // rtp_rtcp_.SetH264PacketizationMode(H264_NON_INTERLEAVED_MODE);
- // }
- // if (video_codec.codecSpecific.H264.configParametersSize > 0) {
- // rtp_rtcp_.SetH264SendModeNALU_PPS_SPS(true);
- // }
- // }
-
- // Don't log this error, no way to check in advance if this pl_type is
- // registered or not...
- rtp_rtcp_.DeRegisterSendPayload(video_codec.plType);
- if (rtp_rtcp_.RegisterSendPayload(video_codec) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not register payload type", __FUNCTION__);
- return -1;
- }
- if (restart_rtp) {
- rtp_rtcp_.SetSendingStatus(true);
- }
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::SetReceiveCodec(const VideoCodec& video_codec) {
- // We will not receive simulcast streams, so no need to handle that use case.
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
-
- WebRtc_Word8 old_pltype = -1;
- if (rtp_rtcp_.ReceivePayloadType(video_codec, &old_pltype) != -1) {
- rtp_rtcp_.DeRegisterReceivePayload(old_pltype);
- }
-
- if (rtp_rtcp_.RegisterReceivePayload(video_codec) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not register receive payload type", __FUNCTION__);
- return -1;
- }
-
- if (video_codec.codecType != kVideoCodecRED &&
- video_codec.codecType != kVideoCodecULPFEC) {
- // Register codec type with VCM, but do not register RED or ULPFEC.
- if (vcm_.RegisterReceiveCodec(&video_codec, number_of_cores_,
- wait_for_key_frame_) != VCM_OK) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not register decoder", __FUNCTION__);
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::GetReceiveCodec(VideoCodec& video_codec) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
-
- if (vcm_.ReceiveCodec(&video_codec) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not get receive codec", __FUNCTION__);
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::RegisterCodecObserver(ViEDecoderObserver* observer) {
- CriticalSectionScoped cs(callback_cs_.get());
- if (observer) {
- if (codec_observer_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: already added", __FUNCTION__);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: observer added", __FUNCTION__);
- codec_observer_ = observer;
- } else {
- if (!codec_observer_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: no observer added", __FUNCTION__);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: observer removed", __FUNCTION__);
- codec_observer_ = NULL;
- }
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::RegisterExternalDecoder(const WebRtc_UWord8 pl_type,
- VideoDecoder* decoder,
- bool decoder_render,
- WebRtc_Word32 render_delay) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
-
- WebRtc_Word32 result = 0;
- result = vcm_.RegisterExternalDecoder(decoder, pl_type, decoder_render);
- if (decoder_render && result == 0) {
- // Let VCM know how long before the actual render time the decoder needs
- // to get a frame for decoding.
- result = vcm_.SetRenderDelay(render_delay);
- }
- return result;
-}
-
-WebRtc_Word32 ViEChannel::DeRegisterExternalDecoder(
- const WebRtc_UWord8 pl_type) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s pl_type", __FUNCTION__, pl_type);
-
- VideoCodec current_receive_codec;
- WebRtc_Word32 result = 0;
- result = vcm_.ReceiveCodec(¤t_receive_codec);
- if (vcm_.RegisterExternalDecoder(NULL, pl_type, false) != VCM_OK) {
- return -1;
- }
-
- if (result == 0 && current_receive_codec.plType == pl_type) {
- result = vcm_.RegisterReceiveCodec(¤t_receive_codec, number_of_cores_,
- wait_for_key_frame_);
- }
- return result;
-}
-
-WebRtc_Word32 ViEChannel::ReceiveCodecStatistics(
- WebRtc_UWord32& num_key_frames, WebRtc_UWord32& num_delta_frames) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
-
- VCMFrameCount received_frames;
- if (vcm_.ReceivedFrameCount(received_frames) != VCM_OK) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not get received frame information", __FUNCTION__);
- return -1;
- }
- num_key_frames = received_frames.numKeyFrames;
- num_delta_frames = received_frames.numDeltaFrames;
- return 0;
-}
-
-WebRtc_UWord32 ViEChannel::DiscardedPackets() const {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- return vcm_.DiscardedPackets();
-}
-
-WebRtc_Word32 ViEChannel::WaitForKeyFrame(bool wait) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s(wait: %d)", __FUNCTION__, wait);
- wait_for_key_frame_ = wait;
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::SetSignalPacketLossStatus(bool enable,
- bool only_key_frames) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s(enable: %d)", __FUNCTION__, enable);
- if (enable) {
- if (only_key_frames) {
- vcm_.SetVideoProtection(kProtectionKeyOnLoss, false);
- if (vcm_.SetVideoProtection(kProtectionKeyOnKeyLoss, true) != VCM_OK) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s failed %d", __FUNCTION__, enable);
- return -1;
- }
- } else {
- vcm_.SetVideoProtection(kProtectionKeyOnKeyLoss, false);
- if (vcm_.SetVideoProtection(kProtectionKeyOnLoss, true) != VCM_OK) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s failed %d", __FUNCTION__, enable);
- return -1;
- }
- }
- } else {
- vcm_.SetVideoProtection(kProtectionKeyOnLoss, false);
- vcm_.SetVideoProtection(kProtectionKeyOnKeyLoss, false);
- }
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::SetRTCPMode(const RTCPMethod rtcp_mode) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: %d", __FUNCTION__, rtcp_mode);
-
- for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
- it != simulcast_rtp_rtcp_.end();
- it++) {
- RtpRtcp* rtp_rtcp = *it;
- rtp_rtcp->SetRTCPStatus(rtcp_mode);
- }
- return rtp_rtcp_.SetRTCPStatus(rtcp_mode);
-}
-
-WebRtc_Word32 ViEChannel::GetRTCPMode(RTCPMethod& rtcp_mode) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
- rtcp_mode = rtp_rtcp_.RTCP();
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::SetNACKStatus(const bool enable) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s(enable: %d)", __FUNCTION__, enable);
-
- // Update the decoding VCM.
- if (vcm_.SetVideoProtection(kProtectionNack, enable) != VCM_OK) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not set VCM NACK protection: %d", __FUNCTION__,
- enable);
- return -1;
- }
- if (enable) {
- // Disable possible FEC.
- SetFECStatus(false, 0, 0);
- }
- // Update the decoding VCM.
- if (vcm_.SetVideoProtection(kProtectionNack, enable) != VCM_OK) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not set VCM NACK protection: %d", __FUNCTION__,
- enable);
- return -1;
- }
- return ProcessNACKRequest(enable);
-}
-
-WebRtc_Word32 ViEChannel::ProcessNACKRequest(const bool enable) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s(enable: %d)", __FUNCTION__, enable);
-
- if (enable) {
- // Turn on NACK.
- NACKMethod nackMethod = kNackRtcp;
- if (rtp_rtcp_.RTCP() == kRtcpOff) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not enable NACK, RTPC not on ", __FUNCTION__);
- return -1;
- }
- if (rtp_rtcp_.SetNACKStatus(nackMethod) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not set NACK method %d", __FUNCTION__,
- nackMethod);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Using NACK method %d", __FUNCTION__, nackMethod);
- rtp_rtcp_.SetStorePacketsStatus(true, kNackHistorySize);
-
- vcm_.RegisterPacketRequestCallback(this);
-
- for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
- it != simulcast_rtp_rtcp_.end();
- it++) {
- RtpRtcp* rtp_rtcp = *it;
- rtp_rtcp->SetStorePacketsStatus(true, kNackHistorySize);
- }
- } else {
- for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
- it != simulcast_rtp_rtcp_.end();
- it++) {
- RtpRtcp* rtp_rtcp = *it;
- rtp_rtcp->SetStorePacketsStatus(false);
- }
- rtp_rtcp_.SetStorePacketsStatus(false);
- vcm_.RegisterPacketRequestCallback(NULL);
- if (rtp_rtcp_.SetNACKStatus(kNackOff) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not turn off NACK", __FUNCTION__);
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::SetFECStatus(const bool enable,
- const unsigned char payload_typeRED,
- const unsigned char payload_typeFEC) {
- // Disable possible NACK.
- if (enable) {
- SetNACKStatus(false);
- }
-
- return ProcessFECRequest(enable, payload_typeRED, payload_typeFEC);
-}
-
-WebRtc_Word32 ViEChannel::ProcessFECRequest(
- const bool enable,
- const unsigned char payload_typeRED,
- const unsigned char payload_typeFEC) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s(enable: %d, payload_typeRED: %u, payload_typeFEC: %u)",
- __FUNCTION__, enable, payload_typeRED, payload_typeFEC);
-
- if (rtp_rtcp_.SetGenericFECStatus(enable, payload_typeRED,
- payload_typeFEC) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not change FEC status to %d", __FUNCTION__,
- enable);
- return -1;
- }
- for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
- it != simulcast_rtp_rtcp_.end();
- it++) {
- RtpRtcp* rtp_rtcp = *it;
- rtp_rtcp->SetGenericFECStatus(enable, payload_typeRED, payload_typeFEC);
- }
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::SetHybridNACKFECStatus(
- const bool enable,
- const unsigned char payload_typeRED,
- const unsigned char payload_typeFEC) {
- // Update the decoding VCM with hybrid mode.
- if (vcm_.SetVideoProtection(kProtectionNackFEC, enable) != VCM_OK) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not set VCM NACK protection: %d", __FUNCTION__,
- enable);
- return -1;
- }
-
- WebRtc_Word32 ret_val = 0;
- ret_val = ProcessNACKRequest(enable);
- if (ret_val < 0) {
- return ret_val;
- }
- return ProcessFECRequest(enable, payload_typeRED, payload_typeFEC);
-}
-
-WebRtc_Word32 ViEChannel::SetKeyFrameRequestMethod(
- const KeyFrameRequestMethod method) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: %d", __FUNCTION__, method);
- return rtp_rtcp_.SetKeyFrameRequestMethod(method);
-}
-
-bool ViEChannel::EnableRemb(bool enable) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "ViEChannel::EnableRemb: %d", enable);
- if (rtp_rtcp_.SetREMBStatus(enable) != 0)
- return false;
- return true;
-}
-
-WebRtc_Word32 ViEChannel::EnableTMMBR(const bool enable) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: %d", __FUNCTION__, enable);
- return rtp_rtcp_.SetTMMBRStatus(enable);
-}
-
-WebRtc_Word32 ViEChannel::EnableKeyFrameRequestCallback(const bool enable) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: %d", __FUNCTION__, enable);
-
- CriticalSectionScoped cs(callback_cs_.get());
- if (enable && !codec_observer_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: No ViECodecObserver set", __FUNCTION__, enable);
- return -1;
- }
- do_key_frame_callbackRequest_ = enable;
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::SetSSRC(const WebRtc_UWord32 SSRC,
- const StreamType usage,
- const uint8_t simulcast_idx) {
- WEBRTC_TRACE(webrtc::kTraceInfo,
- webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s(usage:%d, SSRC: 0x%x, idx:%u)",
- __FUNCTION__, usage, SSRC, simulcast_idx);
- if (simulcast_idx == 0) {
- return rtp_rtcp_.SetSSRC(SSRC);
- }
- std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
- for (int i = 1; i < simulcast_idx; ++i, ++it) {
- if (it == simulcast_rtp_rtcp_.end()) {
- return -1;
- }
- }
- RtpRtcp* rtp_rtcp = *it;
- if (usage == kViEStreamTypeRtx) {
- return rtp_rtcp->SetRTXSendStatus(true, true, SSRC);
- }
- return rtp_rtcp->SetSSRC(SSRC);
-}
-
-WebRtc_Word32 ViEChannel::SetRemoteSSRCType(const StreamType usage,
- const uint32_t SSRC) const {
- WEBRTC_TRACE(webrtc::kTraceInfo,
- webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s(usage:%d, SSRC: 0x%x)",
- __FUNCTION__, usage, SSRC);
-
- return rtp_rtcp_.SetRTXReceiveStatus(true, SSRC);
-}
-
-WebRtc_Word32 ViEChannel::GetLocalSSRC(WebRtc_UWord32& SSRC) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
- SSRC = rtp_rtcp_.SSRC();
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::GetRemoteSSRC(WebRtc_UWord32& SSRC) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- SSRC = rtp_rtcp_.RemoteSSRC();
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::GetRemoteCSRC(unsigned int CSRCs[kRtpCsrcSize]) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- WebRtc_UWord32 arrayCSRC[kRtpCsrcSize];
- memset(arrayCSRC, 0, sizeof(arrayCSRC));
-
- WebRtc_Word32 num_csrcs = rtp_rtcp_.RemoteCSRCs(arrayCSRC);
- if (num_csrcs > 0) {
- memcpy(CSRCs, arrayCSRC, num_csrcs * sizeof(WebRtc_UWord32));
- for (int idx = 0; idx < num_csrcs; idx++) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "\tCSRC[%d] = %lu", idx, CSRCs[idx]);
- }
- } else {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: CSRC list is empty", __FUNCTION__);
- }
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::SetStartSequenceNumber(
- WebRtc_UWord16 sequence_number) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- if (rtp_rtcp_.Sending()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: already sending", __FUNCTION__);
- return -1;
- }
- return rtp_rtcp_.SetSequenceNumber(sequence_number);
-}
-
-WebRtc_Word32 ViEChannel::SetRTCPCName(const WebRtc_Word8 rtcp_cname[]) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
- if (rtp_rtcp_.Sending()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: already sending", __FUNCTION__);
- return -1;
- }
- return rtp_rtcp_.SetCNAME(rtcp_cname);
-}
-
-WebRtc_Word32 ViEChannel::GetRTCPCName(WebRtc_Word8 rtcp_cname[]) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
- return rtp_rtcp_.CNAME(rtcp_cname);
-}
-
-WebRtc_Word32 ViEChannel::GetRemoteRTCPCName(WebRtc_Word8 rtcp_cname[]) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- WebRtc_UWord32 remoteSSRC = rtp_rtcp_.RemoteSSRC();
- return rtp_rtcp_.RemoteCNAME(remoteSSRC, rtcp_cname);
-}
-
-WebRtc_Word32 ViEChannel::RegisterRtpObserver(ViERTPObserver* observer) {
- CriticalSectionScoped cs(callback_cs_.get());
- if (observer) {
- if (rtp_observer_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: observer alread added", __FUNCTION__);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: observer added", __FUNCTION__);
- rtp_observer_ = observer;
- } else {
- if (!rtp_observer_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: no observer added", __FUNCTION__);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: observer removed", __FUNCTION__);
- rtp_observer_ = NULL;
- }
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::RegisterRtcpObserver(ViERTCPObserver* observer) {
- CriticalSectionScoped cs(callback_cs_.get());
- if (observer) {
- if (rtcp_observer_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: observer alread added", __FUNCTION__);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: observer added", __FUNCTION__);
- rtcp_observer_ = observer;
- } else {
- if (!rtcp_observer_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: no observer added", __FUNCTION__);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: observer removed", __FUNCTION__);
- rtcp_observer_ = NULL;
- }
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::SendApplicationDefinedRTCPPacket(
- const WebRtc_UWord8 sub_type,
- WebRtc_UWord32 name,
- const WebRtc_UWord8* data,
- WebRtc_UWord16 data_length_in_bytes) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- if (!rtp_rtcp_.Sending()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not sending", __FUNCTION__);
- return -1;
- }
- if (!data) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: no input argument", __FUNCTION__);
- return -1;
- }
- if (data_length_in_bytes % 4 != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: input length error", __FUNCTION__);
- return -1;
- }
- RTCPMethod rtcp_method = rtp_rtcp_.RTCP();
- if (rtcp_method == kRtcpOff) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: RTCP not enabled", __FUNCTION__);
- return -1;
- }
- // Create and send packet.
- if (rtp_rtcp_.SetRTCPApplicationSpecificData(sub_type, name, data,
- data_length_in_bytes) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not send RTCP application data", __FUNCTION__);
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::GetSendRtcpStatistics(WebRtc_UWord16& fraction_lost,
- WebRtc_UWord32& cumulative_lost,
- WebRtc_UWord32& extended_max,
- WebRtc_UWord32& jitter_samples,
- WebRtc_Word32& rtt_ms) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- // TODO(pwestin) how do we do this for simulcast ? average for all
- // except cumulative_lost that is the sum ?
- // for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
- // it != simulcast_rtp_rtcp_.end();
- // it++) {
- // RtpRtcp* rtp_rtcp = *it;
- // }
- WebRtc_UWord32 remoteSSRC = rtp_rtcp_.RemoteSSRC();
-
- // Get all RTCP receiver report blocks that have been received on this
- // channel. If we receive RTP packets from a remote source we know the
- // remote SSRC and use the report block from him.
- // Otherwise use the first report block.
- std::vector<RTCPReportBlock> remote_stats;
- if (rtp_rtcp_.RemoteRTCPStat(&remote_stats) != 0 || remote_stats.empty()) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not get remote stats", __FUNCTION__);
- return -1;
- }
- std::vector<RTCPReportBlock>::const_iterator statistics =
- remote_stats.begin();
- for (; statistics != remote_stats.end(); ++statistics) {
- if (statistics->remoteSSRC == remoteSSRC)
- break;
- }
-
- if (statistics == remote_stats.end()) {
- // If we have not received any RTCP packets from this SSRC it probably means
- // we have not received any RTP packets.
- // Use the first received report block instead.
- statistics = remote_stats.begin();
- remoteSSRC = statistics->remoteSSRC;
- }
-
- fraction_lost = statistics->fractionLost;
- cumulative_lost = statistics->cumulativeLost;
- extended_max = statistics->extendedHighSeqNum;
- jitter_samples = statistics->jitter;
-
- WebRtc_UWord16 dummy;
- WebRtc_UWord16 rtt = 0;
- if (rtp_rtcp_.RTT(remoteSSRC, &rtt, &dummy, &dummy, &dummy) != 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not get RTT", __FUNCTION__);
- return -1;
- }
- rtt_ms = rtt;
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::GetReceivedRtcpStatistics(
- WebRtc_UWord16& fraction_lost,
- WebRtc_UWord32& cumulative_lost,
- WebRtc_UWord32& extended_max,
- WebRtc_UWord32& jitter_samples,
- WebRtc_Word32& rtt_ms) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
-
- WebRtc_UWord8 frac_lost = 0;
- if (rtp_rtcp_.StatisticsRTP(&frac_lost, &cumulative_lost, &extended_max,
- &jitter_samples) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not get received RTP statistics", __FUNCTION__);
- return -1;
- }
- fraction_lost = frac_lost;
-
- WebRtc_UWord32 remoteSSRC = rtp_rtcp_.RemoteSSRC();
- WebRtc_UWord16 dummy = 0;
- WebRtc_UWord16 rtt = 0;
- if (rtp_rtcp_.RTT(remoteSSRC, &rtt, &dummy, &dummy, &dummy) != 0) {
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not get RTT", __FUNCTION__);
- }
- rtt_ms = rtt;
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::GetRtpStatistics(
- WebRtc_UWord32& bytes_sent,
- WebRtc_UWord32& packets_sent,
- WebRtc_UWord32& bytes_received,
- WebRtc_UWord32& packets_received) const {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- if (rtp_rtcp_.DataCountersRTP(&bytes_sent,
- &packets_sent,
- &bytes_received,
- &packets_received) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not get counters", __FUNCTION__);
- return -1;
- }
- for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
- it != simulcast_rtp_rtcp_.end();
- it++) {
- WebRtc_UWord32 bytes_sent_temp = 0;
- WebRtc_UWord32 packets_sent_temp = 0;
- RtpRtcp* rtp_rtcp = *it;
- rtp_rtcp->DataCountersRTP(&bytes_sent_temp, &packets_sent_temp, NULL, NULL);
- bytes_sent += bytes_sent_temp;
- packets_sent += packets_sent_temp;
- }
- return 0;
-}
-
-void ViEChannel::GetBandwidthUsage(WebRtc_UWord32& total_bitrate_sent,
- WebRtc_UWord32& video_bitrate_sent,
- WebRtc_UWord32& fec_bitrate_sent,
- WebRtc_UWord32& nackBitrateSent) const {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- rtp_rtcp_.BitrateSent(&total_bitrate_sent,
- &video_bitrate_sent,
- &fec_bitrate_sent,
- &nackBitrateSent);
- for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
- it != simulcast_rtp_rtcp_.end(); it++) {
- WebRtc_UWord32 stream_rate = 0;
- WebRtc_UWord32 video_rate = 0;
- WebRtc_UWord32 fec_rate = 0;
- WebRtc_UWord32 nackRate = 0;
- RtpRtcp* rtp_rtcp = *it;
- rtp_rtcp->BitrateSent(&stream_rate, &video_rate, &fec_rate, &nackRate);
- total_bitrate_sent += stream_rate;
- fec_bitrate_sent += fec_rate;
- nackBitrateSent += nackRate;
- }
-}
-
-int ViEChannel::GetEstimatedReceiveBandwidth(
- WebRtc_UWord32* estimated_bandwidth) const {
- return rtp_rtcp_.EstimatedReceiveBandwidth(estimated_bandwidth);
-}
-
-WebRtc_Word32 ViEChannel::SetKeepAliveStatus(
- const bool enable,
- const WebRtc_Word8 unknown_payload_type,
- const WebRtc_UWord16 delta_transmit_timeMS) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
-
- if (enable && rtp_rtcp_.RTPKeepalive()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: RTP keepalive already enabled", __FUNCTION__);
- return -1;
- } else if (!enable && !rtp_rtcp_.RTPKeepalive()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: RTP keepalive already disabled", __FUNCTION__);
- return -1;
- }
-
- if (rtp_rtcp_.SetRTPKeepaliveStatus(enable, unknown_payload_type,
- delta_transmit_timeMS) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not set RTP keepalive status %d", __FUNCTION__,
- enable);
- if (enable == false && !rtp_rtcp_.DefaultModuleRegistered()) {
- // Not sending media and we try to disable keep alive
- rtp_rtcp_.ResetSendDataCountersRTP();
- rtp_rtcp_.SetSendingStatus(false);
- }
- return -1;
- }
-
- if (enable && !rtp_rtcp_.Sending()) {
- // Enable sending to start sending Sender reports instead of receive
- // reports.
- if (rtp_rtcp_.SetSendingStatus(true) != 0) {
- rtp_rtcp_.SetRTPKeepaliveStatus(false, 0, 0);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not start sending", __FUNCTION__);
- return -1;
- }
- } else if (!enable && !rtp_rtcp_.SendingMedia()) {
- // Not sending media and we're disabling keep alive.
- rtp_rtcp_.ResetSendDataCountersRTP();
- if (rtp_rtcp_.SetSendingStatus(false) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not stop sending", __FUNCTION__);
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::GetKeepAliveStatus(
- bool& enabled,
- WebRtc_Word8& unknown_payload_type,
- WebRtc_UWord16& delta_transmit_time_ms) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- if (rtp_rtcp_.RTPKeepaliveStatus(&enabled, &unknown_payload_type,
- &delta_transmit_time_ms) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not get RTP keepalive status", __FUNCTION__);
- return -1;
- }
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: enabled = %d, unknown_payload_type = %d, "
- "delta_transmit_time_ms = %ul",
- __FUNCTION__, enabled, (WebRtc_Word32) unknown_payload_type,
- delta_transmit_time_ms);
-
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::StartRTPDump(const char file_nameUTF8[1024],
- RTPDirections direction) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- if (direction != kRtpIncoming && direction != kRtpOutgoing) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: invalid input", __FUNCTION__);
- return -1;
- }
-
- if (direction == kRtpIncoming) {
- return vie_receiver_.StartRTPDump(file_nameUTF8);
- } else {
- return vie_sender_.StartRTPDump(file_nameUTF8);
- }
-}
-
-WebRtc_Word32 ViEChannel::StopRTPDump(RTPDirections direction) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
-
- if (direction != kRtpIncoming && direction != kRtpOutgoing) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: invalid input", __FUNCTION__);
- return -1;
- }
-
- if (direction == kRtpIncoming) {
- return vie_receiver_.StopRTPDump();
- } else {
- return vie_sender_.StopRTPDump();
- }
-}
-
-WebRtc_Word32 ViEChannel::SetLocalReceiver(const WebRtc_UWord16 rtp_port,
- const WebRtc_UWord16 rtcp_port,
- const WebRtc_Word8* ip_address) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- callback_cs_->Enter();
- if (external_transport_) {
- callback_cs_->Leave();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: external transport registered", __FUNCTION__);
- return -1;
- }
- callback_cs_->Leave();
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.Receiving()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: already receiving", __FUNCTION__);
- return -1;
- }
-
- const WebRtc_Word8* multicast_ip_address = NULL;
- if (socket_transport_.InitializeReceiveSockets(&vie_receiver_, rtp_port,
- ip_address,
- multicast_ip_address,
- rtcp_port) != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not initialize receive sockets. Socket error: %d",
- __FUNCTION__, socket_error);
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::GetLocalReceiver(WebRtc_UWord16& rtp_port,
- WebRtc_UWord16& rtcp_port,
- WebRtc_Word8* ip_address) const {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- callback_cs_->Enter();
- if (external_transport_) {
- callback_cs_->Leave();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: external transport registered", __FUNCTION__);
- return -1;
- }
- callback_cs_->Leave();
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.ReceiveSocketsInitialized() == false) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: receive sockets not initialized", __FUNCTION__);
- return -1;
- }
-
- WebRtc_Word8 multicast_ip_address[UdpTransport::kIpAddressVersion6Length];
- if (socket_transport_.ReceiveSocketInformation(ip_address, rtp_port,
- rtcp_port,
- multicast_ip_address) != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not get receive socket information. Socket error: %d",
- __FUNCTION__, socket_error);
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::SetSendDestination(
- const WebRtc_Word8* ip_address,
- const WebRtc_UWord16 rtp_port,
- const WebRtc_UWord16 rtcp_port,
- const WebRtc_UWord16 source_rtp_port,
- const WebRtc_UWord16 source_rtcp_port) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- callback_cs_->Enter();
- if (external_transport_) {
- callback_cs_->Leave();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: external transport registered", __FUNCTION__);
- return -1;
- }
- callback_cs_->Leave();
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- const bool is_ipv6 = socket_transport_.IpV6Enabled();
- if (UdpTransport::IsIpAddressValid(ip_address, is_ipv6) == false) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Not a valid IP address: %s", __FUNCTION__, ip_address);
- return -1;
- }
- if (socket_transport_.InitializeSendSockets(ip_address, rtp_port,
- rtcp_port)!= 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not initialize send socket. Socket error: %d",
- __FUNCTION__, socket_error);
- return -1;
- }
-
- if (source_rtp_port != 0) {
- WebRtc_UWord16 receive_rtp_port = 0;
- WebRtc_UWord16 receive_rtcp_port = 0;
- if (socket_transport_.ReceiveSocketInformation(NULL, receive_rtp_port,
- receive_rtcp_port,
- NULL) != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not get receive port information. Socket error: %d",
- __FUNCTION__, socket_error);
- return -1;
- }
- // Initialize an extra socket only if send port differs from receive
- // port.
- if (source_rtp_port != receive_rtp_port) {
- if (socket_transport_.InitializeSourcePorts(source_rtp_port,
- source_rtcp_port) != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not set source ports. Socket error: %d",
- __FUNCTION__, socket_error);
- return -1;
- }
- }
- }
- vie_sender_.RegisterSendTransport(&socket_transport_);
-
- // Workaround to avoid SSRC colision detection in loppback tests.
- if (!is_ipv6) {
- WebRtc_UWord32 local_host_address = 0;
- const WebRtc_UWord32 current_ip_address =
- UdpTransport::InetAddrIPV4(ip_address);
-
- if ((UdpTransport::LocalHostAddress(local_host_address) == 0 &&
- local_host_address == current_ip_address) ||
- strncmp("127.0.0.1", ip_address, 9) == 0) {
- rtp_rtcp_.SetSSRC(0xFFFFFFFF);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "Running in loopback. Forcing fixed SSRC");
- }
- } else {
- char local_host_address[16];
- char current_ip_address[16];
-
- WebRtc_Word32 conv_result =
- UdpTransport::LocalHostAddressIPV6(local_host_address);
- conv_result += socket_transport_.InetPresentationToNumeric(
- 23, ip_address, current_ip_address);
- if (conv_result == 0) {
- bool local_host = true;
- for (WebRtc_Word32 i = 0; i < 16; i++) {
- if (local_host_address[i] != current_ip_address[i]) {
- local_host = false;
- break;
- }
- }
- if (!local_host) {
- local_host = true;
- for (WebRtc_Word32 i = 0; i < 15; i++) {
- if (current_ip_address[i] != 0) {
- local_host = false;
- break;
- }
- }
- if (local_host == true && current_ip_address[15] != 1) {
- local_host = false;
- }
- }
- if (local_host) {
- rtp_rtcp_.SetSSRC(0xFFFFFFFF);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "Running in loopback. Forcing fixed SSRC");
- }
- }
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::GetSendDestination(
- WebRtc_Word8* ip_address,
- WebRtc_UWord16& rtp_port,
- WebRtc_UWord16& rtcp_port,
- WebRtc_UWord16& source_rtp_port,
- WebRtc_UWord16& source_rtcp_port) const {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- callback_cs_->Enter();
- if (external_transport_) {
- callback_cs_->Leave();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: external transport registered", __FUNCTION__);
- return -1;
- }
- callback_cs_->Leave();
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.SendSocketsInitialized() == false) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: send sockets not initialized", __FUNCTION__);
- return -1;
- }
- if (socket_transport_.SendSocketInformation(ip_address, rtp_port, rtcp_port)
- != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not get send socket information. Socket error: %d",
- __FUNCTION__, socket_error);
- return -1;
- }
- source_rtp_port = 0;
- source_rtcp_port = 0;
- if (socket_transport_.SourcePortsInitialized()) {
- socket_transport_.SourcePorts(source_rtp_port, source_rtcp_port);
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::StartSend() {
- CriticalSectionScoped cs(callback_cs_.get());
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!external_transport_) {
- if (socket_transport_.SendSocketsInitialized() == false) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: send sockets not initialized", __FUNCTION__);
- return -1;
- }
- }
-#endif
- rtp_rtcp_.SetSendingMediaStatus(true);
-
- if (rtp_rtcp_.Sending() && !rtp_rtcp_.RTPKeepalive()) {
- if (rtp_rtcp_.RTPKeepalive()) {
- // Sending Keep alive, don't trigger an error.
- return 0;
- }
- // Already sending.
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Already sending", __FUNCTION__);
- return kViEBaseAlreadySending;
- }
- if (rtp_rtcp_.SetSendingStatus(true) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not start sending RTP", __FUNCTION__);
- return -1;
- }
- for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
- it != simulcast_rtp_rtcp_.end();
- it++) {
- RtpRtcp* rtp_rtcp = *it;
- rtp_rtcp->SetSendingMediaStatus(true);
- rtp_rtcp->SetSendingStatus(true);
- }
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::StopSend() {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- rtp_rtcp_.SetSendingMediaStatus(false);
- for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
- it != simulcast_rtp_rtcp_.end();
- it++) {
- RtpRtcp* rtp_rtcp = *it;
- rtp_rtcp->SetSendingMediaStatus(false);
- }
- if (rtp_rtcp_.RTPKeepalive()) {
- // Don't turn off sending since we'll send keep alive packets.
- return 0;
- }
- if (!rtp_rtcp_.Sending()) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Not sending", __FUNCTION__);
- return kViEBaseNotSending;
- }
-
- // Reset.
- rtp_rtcp_.ResetSendDataCountersRTP();
- if (rtp_rtcp_.SetSendingStatus(false) != 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not stop RTP sending", __FUNCTION__);
- return -1;
- }
- for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
- it != simulcast_rtp_rtcp_.end();
- it++) {
- RtpRtcp* rtp_rtcp = *it;
- rtp_rtcp->ResetSendDataCountersRTP();
- rtp_rtcp->SetSendingStatus(false);
- }
- return 0;
-}
-
-bool ViEChannel::Sending() {
- return rtp_rtcp_.Sending();
-}
-
-WebRtc_Word32 ViEChannel::StartReceive() {
- CriticalSectionScoped cs(callback_cs_.get());
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!external_transport_) {
- if (socket_transport_.Receiving()) {
- // Warning, don't return error.
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: already receiving", __FUNCTION__);
- return 0;
- }
- if (socket_transport_.ReceiveSocketsInitialized() == false) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: receive sockets not initialized", __FUNCTION__);
- return -1;
- }
- if (socket_transport_.StartReceiving(kViENumReceiveSocketBuffers) != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not get receive socket information. Socket error:%d",
- __FUNCTION__, socket_error);
- return -1;
- }
- }
-#endif
- if (StartDecodeThread() != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not start decoder thread", __FUNCTION__);
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- socket_transport_.StopReceiving();
-#endif
- vie_receiver_.StopReceive();
- return -1;
- }
- vie_receiver_.StartReceive();
-
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::StopReceive() {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- vie_receiver_.StopReceive();
- StopDecodeThread();
- vcm_.ResetDecoder();
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (external_transport_) {
- return 0;
- }
- }
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.Receiving() == false) {
- // Warning, don't return error
- WEBRTC_TRACE(kTraceWarning, kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s: not receiving",
- __FUNCTION__);
- return 0;
- }
- if (socket_transport_.StopReceiving() != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Socket error: %d", __FUNCTION__, socket_error);
- return -1;
- }
-#endif
-
- return 0;
-}
-
-bool ViEChannel::Receiving() {
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- return socket_transport_.Receiving();
-#else
- return false;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::GetSourceInfo(WebRtc_UWord16& rtp_port,
- WebRtc_UWord16& rtcp_port,
- WebRtc_Word8* ip_address,
- WebRtc_UWord32 ip_address_length) {
- {
- CriticalSectionScoped cs(callback_cs_.get());
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- if (external_transport_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: external transport registered", __FUNCTION__);
- return -1;
- }
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.IpV6Enabled() &&
- ip_address_length < UdpTransport::kIpAddressVersion6Length) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: IP address length is too small for IPv6", __FUNCTION__);
- return -1;
- } else if (ip_address_length < UdpTransport::kIpAddressVersion4Length) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: IP address length is too small for IPv4", __FUNCTION__);
- return -1;
- }
-
- if (socket_transport_.RemoteSocketInformation(ip_address, rtp_port, rtcp_port)
- != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Error getting source ports. Socket error: %d",
- __FUNCTION__, socket_error);
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-WebRtc_Word32 ViEChannel::RegisterSendTransport(Transport& transport) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.SendSocketsInitialized() ||
- socket_transport_.ReceiveSocketsInitialized()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: socket transport already initialized", __FUNCTION__);
- return -1;
- }
-#endif
- if (rtp_rtcp_.Sending()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Sending", __FUNCTION__);
- return -1;
- }
-
- CriticalSectionScoped cs(callback_cs_.get());
- if (external_transport_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: transport already registered", __FUNCTION__);
- return -1;
- }
- external_transport_ = &transport;
- vie_sender_.RegisterSendTransport(&transport);
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Transport registered: 0x%p", __FUNCTION__,
- &external_transport_);
-
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::DeregisterSendTransport() {
- CriticalSectionScoped cs(callback_cs_.get());
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- if (!external_transport_) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: no transport registered", __FUNCTION__);
- return -1;
- }
- if (rtp_rtcp_.Sending()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Sending", __FUNCTION__);
- return -1;
- }
- external_transport_ = NULL;
- vie_sender_.DeregisterSendTransport();
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::ReceivedRTPPacket(
- const void* rtp_packet, const WebRtc_Word32 rtp_packet_length) {
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (!external_transport_) {
- return -1;
- }
- }
- return vie_receiver_.ReceivedRTPPacket(rtp_packet, rtp_packet_length);
-}
-
-WebRtc_Word32 ViEChannel::ReceivedRTCPPacket(
- const void* rtcp_packet, const WebRtc_Word32 rtcp_packet_length) {
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (!external_transport_) {
- return -1;
- }
- }
- return vie_receiver_.ReceivedRTCPPacket(rtcp_packet, rtcp_packet_length);
-}
-
-WebRtc_Word32 ViEChannel::EnableIPv6() {
- callback_cs_->Enter();
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
-
- if (external_transport_) {
- callback_cs_->Leave();
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return -1;
- }
- callback_cs_->Leave();
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.IpV6Enabled()) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: IPv6 already enabled", __FUNCTION__);
- return -1;
- }
-
- if (socket_transport_.EnableIpV6() != 0) {
- WebRtc_Word32 socket_error = socket_transport_.LastError();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not enable IPv6. Socket error: %d", __FUNCTION__,
- socket_error);
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-bool ViEChannel::IsIPv6Enabled() {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (external_transport_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return false;
- }
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- return socket_transport_.IpV6Enabled();
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return false;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::SetSourceFilter(const WebRtc_UWord16 rtp_port,
- const WebRtc_UWord16 rtcp_port,
- const WebRtc_Word8* ip_address) {
- callback_cs_->Enter();
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- if (external_transport_) {
- callback_cs_->Leave();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return -1;
- }
- callback_cs_->Leave();
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.SetFilterIP(ip_address) != 0) {
- // Logging done in module.
- return -1;
- }
- if (socket_transport_.SetFilterPorts(rtp_port, rtcp_port) != 0) {
- // Logging done.
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::GetSourceFilter(WebRtc_UWord16& rtp_port,
- WebRtc_UWord16& rtcp_port,
- WebRtc_Word8* ip_address) const {
- callback_cs_->Enter();
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- if (external_transport_) {
- callback_cs_->Leave();
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return -1;
- }
- callback_cs_->Leave();
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.FilterIP(ip_address) != 0) {
- // Logging done in module.
- return -1;
- }
- if (socket_transport_.FilterPorts(rtp_port, rtcp_port) != 0) {
- // Logging done in module.
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::SetToS(const WebRtc_Word32 DSCP,
- const bool use_set_sockOpt) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (external_transport_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return -1;
- }
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.SetToS(DSCP, use_set_sockOpt) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Socket error: %d", __FUNCTION__,
- socket_transport_.LastError());
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::GetToS(WebRtc_Word32& DSCP,
- bool& use_set_sockOpt) const {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (external_transport_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return -1;
- }
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.ToS(DSCP, use_set_sockOpt) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Socket error: %d", __FUNCTION__,
- socket_transport_.LastError());
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::SetSendGQoS(const bool enable,
- const WebRtc_Word32 service_type,
- const WebRtc_UWord32 max_bitrate,
- const WebRtc_Word32 overrideDSCP) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (external_transport_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return -1;
- }
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.SetQoS(enable, service_type, max_bitrate, overrideDSCP,
- false) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Socket error: %d", __FUNCTION__,
- socket_transport_.LastError());
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::GetSendGQoS(bool& enabled,
- WebRtc_Word32& service_type,
- WebRtc_Word32& overrideDSCP) const {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (external_transport_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return -1;
- }
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.QoS(enabled, service_type, overrideDSCP) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Socket error: %d", __FUNCTION__,
- socket_transport_.LastError());
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::SetMTU(WebRtc_UWord16 mtu) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- if (rtp_rtcp_.SetMaxTransferUnit(mtu) != 0) {
- // Logging done.
- return -1;
- }
- for (std::list<RtpRtcp*>::iterator it = simulcast_rtp_rtcp_.begin();
- it != simulcast_rtp_rtcp_.end();
- it++) {
- RtpRtcp* rtp_rtcp = *it;
- rtp_rtcp->SetMaxTransferUnit(mtu);
- }
- return 0;
-}
-
-WebRtc_UWord16 ViEChannel::MaxDataPayloadLength() const {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
- return rtp_rtcp_.MaxDataPayloadLength();
-}
-
-WebRtc_Word32 ViEChannel::SetPacketTimeoutNotification(
- bool enable, WebRtc_UWord32 timeout_seconds) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- if (enable) {
- WebRtc_UWord32 timeout_ms = 1000 * timeout_seconds;
- if (rtp_rtcp_.SetPacketTimeout(timeout_ms, 0) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
- return -1;
- }
- } else {
- if (rtp_rtcp_.SetPacketTimeout(0, 0) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::RegisterNetworkObserver(
- ViENetworkObserver* observer) {
- CriticalSectionScoped cs(callback_cs_.get());
- if (observer) {
- if (networkObserver_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: observer alread added", __FUNCTION__);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: observer added", __FUNCTION__);
- networkObserver_ = observer;
- } else {
- if (!networkObserver_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: no observer added", __FUNCTION__);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: observer removed", __FUNCTION__);
- networkObserver_ = NULL;
- }
- return 0;
-}
-
-bool ViEChannel::NetworkObserverRegistered() {
- CriticalSectionScoped cs(callback_cs_.get());
- return networkObserver_ != NULL;
-}
-
-WebRtc_Word32 ViEChannel::SetPeriodicDeadOrAliveStatus(
- const bool enable, const WebRtc_UWord32 sample_time_seconds) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- CriticalSectionScoped cs(callback_cs_.get());
- if (!networkObserver_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: no observer added", __FUNCTION__);
- return -1;
- }
-
- bool enabled = false;
- WebRtc_UWord8 current_sampletime_seconds = 0;
-
- // Get old settings.
- rtp_rtcp_.PeriodicDeadOrAliveStatus(enabled, current_sampletime_seconds);
- // Set new settings.
- if (rtp_rtcp_.SetPeriodicDeadOrAliveStatus(
- enable, static_cast<WebRtc_UWord8>(sample_time_seconds)) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not set periodic dead-or-alive status",
- __FUNCTION__);
- return -1;
- }
- if (!enable) {
- // Restore last utilized sample time.
- // Without this trick, the sample time would always be reset to default
- // (2 sec), each time dead-or-alive was disabled without sample-time
- // parameter.
- rtp_rtcp_.SetPeriodicDeadOrAliveStatus(enable, current_sampletime_seconds);
- }
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::SendUDPPacket(const WebRtc_Word8* data,
- const WebRtc_UWord32 length,
- WebRtc_Word32& transmitted_bytes,
- bool use_rtcp_socket) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (external_transport_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: External transport registered", __FUNCTION__);
- return -1;
- }
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- transmitted_bytes = socket_transport_.SendRaw(data, length, use_rtcp_socket);
- if (transmitted_bytes == -1) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not available for external transport", __FUNCTION__);
- return -1;
-#endif
-}
-
-WebRtc_Word32 ViEChannel::EnableColorEnhancement(bool enable) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s(enable: %d)", __FUNCTION__, enable);
-
- CriticalSectionScoped cs(callback_cs_.get());
- if (enable && color_enhancement_) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Already enabled", __FUNCTION__);
- return -1;
- } else if (!enable && !color_enhancement_) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: not enabled", __FUNCTION__);
- return -1;
- }
- color_enhancement_ = enable;
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::RegisterSendRtpRtcpModule(
- RtpRtcp& send_rtp_rtcp_module) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- WebRtc_Word32 ret_val = rtp_rtcp_.RegisterDefaultModule(
- &send_rtp_rtcp_module);
- if (ret_val == 0) {
- // We need to store this for the SetSendCodec call.
- default_rtp_rtcp_ = &send_rtp_rtcp_module;
- }
- return ret_val;
-}
-
-WebRtc_Word32 ViEChannel::DeregisterSendRtpRtcpModule() {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- default_rtp_rtcp_ = NULL;
-
- for (std::list<RtpRtcp*>::const_iterator it = simulcast_rtp_rtcp_.begin();
- it != simulcast_rtp_rtcp_.end();
- it++) {
- RtpRtcp* rtp_rtcp = *it;
- rtp_rtcp->DeRegisterDefaultModule();
- }
- return rtp_rtcp_.DeRegisterDefaultModule();
-}
-
-RtpRtcp* ViEChannel::rtp_rtcp() {
- return &rtp_rtcp_;
-}
-
-
-WebRtc_Word32 ViEChannel::FrameToRender(VideoFrame& video_frame) {
- CriticalSectionScoped cs(callback_cs_.get());
-
- if (decoder_reset_) {
- // Trigger a callback to the user if the incoming codec has changed.
- if (codec_observer_) {
- VideoCodec decoder;
- memset(&decoder, 0, sizeof(decoder));
- if (vcm_.ReceiveCodec(&decoder) == VCM_OK) {
- // VCM::ReceiveCodec returns the codec set by
- // RegisterReceiveCodec, which might not be the size we're
- // actually decoding.
- decoder.width = static_cast<unsigned short>(video_frame.Width());
- decoder.height = static_cast<unsigned short>(video_frame.Height());
- codec_observer_->IncomingCodecChanged(channel_id_, decoder);
- } else {
- assert(false);
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: Could not get receive codec", __FUNCTION__);
- }
- }
- decoder_reset_ = false;
- }
- if (effect_filter_) {
- effect_filter_->Transform(video_frame.Length(), video_frame.Buffer(),
- video_frame.TimeStamp(), video_frame.Width(),
- video_frame.Height());
- }
- if (color_enhancement_) {
- VideoProcessingModule::ColorEnhancement(video_frame);
- }
-
- // Record videoframe.
- file_recorder_.RecordVideoFrame(video_frame);
-
- WebRtc_UWord32 arr_ofCSRC[kRtpCsrcSize];
- WebRtc_Word32 no_of_csrcs = rtp_rtcp_.RemoteCSRCs(arr_ofCSRC);
- if (no_of_csrcs <= 0) {
- arr_ofCSRC[0] = rtp_rtcp_.RemoteSSRC();
- no_of_csrcs = 1;
- }
- DeliverFrame(video_frame, no_of_csrcs, arr_ofCSRC);
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::ReceivedDecodedReferenceFrame(
- const WebRtc_UWord64 picture_id) {
- return rtp_rtcp_.SendRTCPReferencePictureSelection(picture_id);
-}
-
-WebRtc_Word32 ViEChannel::StoreReceivedFrame(
- const EncodedVideoData& frame_to_store) {
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::ReceiveStatistics(const WebRtc_UWord32 bit_rate,
- const WebRtc_UWord32 frame_rate) {
- CriticalSectionScoped cs(callback_cs_.get());
- if (codec_observer_) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: bitrate %u, framerate %u", __FUNCTION__, bit_rate,
- frame_rate);
- codec_observer_->IncomingRate(channel_id_, frame_rate, bit_rate);
- }
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::RequestKeyFrame() {
- WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (codec_observer_ && do_key_frame_callbackRequest_) {
- codec_observer_->RequestNewKeyFrame(channel_id_);
- }
- }
- return rtp_rtcp_.RequestKeyFrame();
-}
-
-WebRtc_Word32 ViEChannel::SliceLossIndicationRequest(
- const WebRtc_UWord64 picture_id) {
- return rtp_rtcp_.SendRTCPSliceLossIndication((WebRtc_UWord8) picture_id);
-}
-
-WebRtc_Word32 ViEChannel::ResendPackets(const WebRtc_UWord16* sequence_numbers,
- WebRtc_UWord16 length) {
- WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s(length: %d)", __FUNCTION__, length);
- return rtp_rtcp_.SendNACK(sequence_numbers, length);
-}
-
-bool ViEChannel::ChannelDecodeThreadFunction(void* obj) {
- return static_cast<ViEChannel*>(obj)->ChannelDecodeProcess();
-}
-
-bool ViEChannel::ChannelDecodeProcess() {
- // Decode is blocking, but sleep some time anyway to not get a spin.
- vcm_.Decode(kMaxDecodeWaitTimeMs);
-
- if ((TickTime::Now() - vcm_rttreported_).Milliseconds() > 1000) {
- WebRtc_UWord16 RTT;
- WebRtc_UWord16 avgRTT;
- WebRtc_UWord16 minRTT;
- WebRtc_UWord16 maxRTT;
-
- if (rtp_rtcp_.RTT(rtp_rtcp_.RemoteSSRC(), &RTT, &avgRTT, &minRTT, &maxRTT)
- == 0) {
- vcm_.SetReceiveChannelParameters(RTT);
- vcm_rttreported_ = TickTime::Now();
- } else if (!rtp_rtcp_.Sending() &&
- (TickTime::Now() - vcm_rttreported_).Milliseconds() > 5000) {
- // Wait at least 5 seconds before faking a 200 ms RTT. This is to
- // make sure we have a chance to start sending before we decide to fake.
- vcm_.SetReceiveChannelParameters(200);
- vcm_rttreported_ = TickTime::Now();
- }
- }
- return true;
-}
-
-WebRtc_Word32 ViEChannel::StartDecodeThread() {
- // Start the decode thread
- if (decode_thread_) {
- // Already started.
- return 0;
- }
- decode_thread_ = ThreadWrapper::CreateThread(ChannelDecodeThreadFunction,
- this, kHighestPriority,
- "DecodingThread");
- if (!decode_thread_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not create decode thread", __FUNCTION__);
- return -1;
- }
-
- unsigned int thread_id;
- if (decode_thread_->Start(thread_id) == false) {
- delete decode_thread_;
- decode_thread_ = NULL;
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not start decode thread", __FUNCTION__);
- return -1;
- }
-
- // Used to make sure that we don't give the VCM a faked RTT
- // too early.
- vcm_rttreported_ = TickTime::Now();
-
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: decode thread with id %u started", __FUNCTION__);
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::StopDecodeThread() {
- if (!decode_thread_) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: decode thread not running", __FUNCTION__);
- return 0;
- }
-
- decode_thread_->SetNotAlive();
- if (decode_thread_->Stop()) {
- delete decode_thread_;
- } else {
- // Couldn't stop the thread, leak instead of crash.
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: could not stop decode thread", __FUNCTION__);
- assert(!"could not stop decode thread");
- }
- decode_thread_ = NULL;
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::RegisterExternalEncryption(Encryption* encryption) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- CriticalSectionScoped cs(callback_cs_.get());
- if (external_encryption_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: external encryption already registered", __FUNCTION__);
- return -1;
- }
-
- external_encryption_ = encryption;
-
- vie_receiver_.RegisterExternalDecryption(encryption);
- vie_sender_.RegisterExternalEncryption(encryption);
-
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s", "external encryption object registerd with channel=%d",
- channel_id_);
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::DeRegisterExternalEncryption() {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- CriticalSectionScoped cs(callback_cs_.get());
- if (!external_encryption_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: external encryption is not registered", __FUNCTION__);
- return -1;
- }
-
- external_transport_ = NULL;
- vie_receiver_.DeregisterExternalDecryption();
- vie_sender_.DeregisterExternalEncryption();
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s external encryption object de-registerd with channel=%d",
- __FUNCTION__, channel_id_);
- return 0;
-}
-
-WebRtc_Word32 ViEChannel::SetVoiceChannel(WebRtc_Word32 ve_channel_id,
- VoEVideoSync* ve_sync_interface) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s, audio channel %d, video channel %d", __FUNCTION__,
- ve_channel_id, channel_id_);
-
- if (ve_sync_interface) {
- // Register lip sync
- module_process_thread_.RegisterModule(&vie_sync_);
- } else {
- module_process_thread_.DeRegisterModule(&vie_sync_);
- }
- return vie_sync_.SetVoiceChannel(ve_channel_id, ve_sync_interface);
-}
-
-WebRtc_Word32 ViEChannel::VoiceChannel() {
- return vie_sync_.VoiceChannel();
-}
-
-WebRtc_Word32 ViEChannel::RegisterEffectFilter(ViEEffectFilter* effect_filter) {
- CriticalSectionScoped cs(callback_cs_.get());
- if (!effect_filter) {
- if (!effect_filter_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: no effect filter added for channel %d",
- __FUNCTION__, channel_id_);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: deregister effect filter for device %d", __FUNCTION__,
- channel_id_);
- } else {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: register effect filter for device %d", __FUNCTION__,
- channel_id_);
- if (effect_filter_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: effect filter already added for channel %d",
- __FUNCTION__, channel_id_);
- return -1;
- }
- }
- effect_filter_ = effect_filter;
- return 0;
-}
-
-ViEFileRecorder& ViEChannel::GetIncomingFileRecorder() {
- // Start getting callback of all frames before they are decoded.
- vcm_.RegisterFrameStorageCallback(this);
- return file_recorder_;
-}
-
-void ViEChannel::ReleaseIncomingFileRecorder() {
- // Stop getting callback of all frames before they are decoded.
- vcm_.RegisterFrameStorageCallback(NULL);
-}
-
-void ViEChannel::OnLipSyncUpdate(const WebRtc_Word32 id,
- const WebRtc_Word32 audio_video_offset) {
- if (channel_id_ != ChannelId(id)) {
- WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s, incorrect id", __FUNCTION__, id);
- return;
- }
- vie_sync_.SetNetworkDelay(audio_video_offset);
-}
-
-void ViEChannel::OnApplicationDataReceived(const WebRtc_Word32 id,
- const WebRtc_UWord8 sub_type,
- const WebRtc_UWord32 name,
- const WebRtc_UWord16 length,
- const WebRtc_UWord8* data) {
- if (channel_id_ != ChannelId(id)) {
- WEBRTC_TRACE(kTraceStream, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s, incorrect id", __FUNCTION__, id);
- return;
- }
- CriticalSectionScoped cs(callback_cs_.get());
- {
- if (rtcp_observer_) {
- rtcp_observer_->OnApplicationDataReceived(
- channel_id_, sub_type, name, reinterpret_cast<const char*>(data),
- length);
- }
- }
-}
-
-WebRtc_Word32 ViEChannel::OnInitializeDecoder(
- const WebRtc_Word32 id,
- const WebRtc_Word8 payload_type,
- const WebRtc_Word8 payload_name[RTP_PAYLOAD_NAME_SIZE],
- const int frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: payload_type %d, payload_name %s", __FUNCTION__,
- payload_type, payload_name);
- vcm_.ResetDecoder();
-
- callback_cs_->Enter();
- decoder_reset_ = true;
- callback_cs_->Leave();
- return 0;
-}
-
-void ViEChannel::OnPacketTimeout(const WebRtc_Word32 id) {
- assert(ChannelId(id) == channel_id_);
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
-
- CriticalSectionScoped cs(callback_cs_.get());
- if (networkObserver_) {
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (socket_transport_.Receiving() || external_transport_) {
-#else
- if (external_transport_) {
-#endif
- networkObserver_->PacketTimeout(channel_id_, NoPacket);
- rtp_packet_timeout_ = true;
- }
- }
-}
-
-void ViEChannel::OnReceivedPacket(const WebRtc_Word32 id,
- const RtpRtcpPacketType packet_type) {
- assert(ChannelId(id) == channel_id_);
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- if (rtp_packet_timeout_ && packet_type == kPacketRtp) {
- CriticalSectionScoped cs(callback_cs_.get());
- if (networkObserver_) {
- networkObserver_->PacketTimeout(channel_id_, PacketReceived);
- }
-
- // Reset even if no observer set, might have been removed during timeout.
- rtp_packet_timeout_ = false;
- }
-}
-
-void ViEChannel::OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
- const RTPAliveType alive) {
- assert(ChannelId(id) == channel_id_);
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s(id=%d, alive=%d)", __FUNCTION__, id, alive);
-
- CriticalSectionScoped cs(callback_cs_.get());
- if (!networkObserver_) {
- return;
- }
- bool is_alive = true;
- if (alive == kRtpDead) {
- is_alive = false;
- }
- networkObserver_->OnPeriodicDeadOrAlive(channel_id_, is_alive);
- return;
-}
-
-void ViEChannel::OnIncomingSSRCChanged(const WebRtc_Word32 id,
- const WebRtc_UWord32 SSRC) {
- if (channel_id_ != ChannelId(id)) {
- assert(false);
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s, incorrect id", __FUNCTION__, id);
- return;
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: %u", __FUNCTION__, SSRC);
-
- CriticalSectionScoped cs(callback_cs_.get());
- {
- if (rtp_observer_) {
- rtp_observer_->IncomingSSRCChanged(channel_id_, SSRC);
- }
- }
-}
-
-void ViEChannel::OnIncomingCSRCChanged(const WebRtc_Word32 id,
- const WebRtc_UWord32 CSRC,
- const bool added) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: %u added: %d", __FUNCTION__, CSRC, added);
-
- if (channel_id_ != ChannelId(id)) {
- assert(false);
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s, incorrect id", __FUNCTION__, id);
- return;
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_),
- "%s: %u", __FUNCTION__, CSRC);
-
- CriticalSectionScoped cs(callback_cs_.get());
- {
- if (rtp_observer_) {
- rtp_observer_->IncomingCSRCChanged(channel_id_, CSRC, added);
- }
- }
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_channel.h b/trunk/src/video_engine/vie_channel.h
deleted file mode 100644
index dac268f..0000000
--- a/trunk/src/video_engine/vie_channel.h
+++ /dev/null
@@ -1,404 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// vie_channel.h
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
-
-#include <list>
-
-#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "modules/udp_transport/interface/udp_transport.h"
-#include "modules/video_coding/main/interface/video_coding_defines.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-#include "system_wrappers/interface/tick_util.h"
-#include "typedefs.h"
-#include "video_engine/include/vie_network.h"
-#include "video_engine/include/vie_rtp_rtcp.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_file_recorder.h"
-#include "video_engine/vie_frame_provider_base.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class Encryption;
-class ProcessThread;
-class RtpRtcp;
-class ThreadWrapper;
-class VideoCodingModule;
-class VideoDecoder;
-class VideoRenderCallback;
-class ViEDecoderObserver;
-class ViEEffectFilter;
-class ViENetworkObserver;
-class ViEReceiver;
-class ViERTCPObserver;
-class ViERTPObserver;
-class ViESender;
-class ViESyncModule;
-class VoEVideoSync;
-
-class ViEChannel
- : public VCMFrameTypeCallback,
- public VCMReceiveCallback,
- public VCMReceiveStatisticsCallback,
- public VCMPacketRequestCallback,
- public VCMFrameStorageCallback,
- public RtcpFeedback,
- public RtpFeedback,
- public ViEFrameProviderBase {
- public:
- ViEChannel(WebRtc_Word32 channel_id,
- WebRtc_Word32 engine_id,
- WebRtc_UWord32 number_of_cores,
- ProcessThread& module_process_thread);
- ~ViEChannel();
-
- WebRtc_Word32 Init();
-
- // Sets the encoder to use for the channel. |new_stream| indicates the encoder
- // type has changed and we should start a new RTP stream.
- WebRtc_Word32 SetSendCodec(const VideoCodec& video_codec,
- bool new_stream = true);
- WebRtc_Word32 SetReceiveCodec(const VideoCodec& video_codec);
- WebRtc_Word32 GetReceiveCodec(VideoCodec& video_codec);
- WebRtc_Word32 RegisterCodecObserver(ViEDecoderObserver* observer);
- // Registers an external decoder. |decoder_render| is set to true if the
- // decoder will do the rendering. If |decoder_render| is set,|render_delay|
- // indicates the time needed to decode and render a frame.
- WebRtc_Word32 RegisterExternalDecoder(const WebRtc_UWord8 pl_type,
- VideoDecoder* decoder,
- bool decoder_render,
- WebRtc_Word32 render_delay);
- WebRtc_Word32 DeRegisterExternalDecoder(const WebRtc_UWord8 pl_type);
- WebRtc_Word32 ReceiveCodecStatistics(WebRtc_UWord32& num_key_frames,
- WebRtc_UWord32& num_delta_frames);
- WebRtc_UWord32 DiscardedPackets() const;
-
- // Only affects calls to SetReceiveCodec done after this call.
- WebRtc_Word32 WaitForKeyFrame(bool wait);
-
- // If enabled, a key frame request will be sent as soon as there are lost
- // packets. If |only_key_frames| are set, requests are only sent for loss in
- // key frames.
- WebRtc_Word32 SetSignalPacketLossStatus(bool enable, bool only_key_frames);
-
- WebRtc_Word32 SetRTCPMode(const RTCPMethod rtcp_mode);
- WebRtc_Word32 GetRTCPMode(RTCPMethod& rtcp_mode);
- WebRtc_Word32 SetNACKStatus(const bool enable);
- WebRtc_Word32 SetFECStatus(const bool enable,
- const unsigned char payload_typeRED,
- const unsigned char payload_typeFEC);
- WebRtc_Word32 SetHybridNACKFECStatus(const bool enable,
- const unsigned char payload_typeRED,
- const unsigned char payload_typeFEC);
- WebRtc_Word32 SetKeyFrameRequestMethod(const KeyFrameRequestMethod method);
- bool EnableRemb(bool enable);
- WebRtc_Word32 EnableTMMBR(const bool enable);
- WebRtc_Word32 EnableKeyFrameRequestCallback(const bool enable);
-
- // Sets SSRC for outgoing stream.
- WebRtc_Word32 SetSSRC(const WebRtc_UWord32 SSRC,
- const StreamType usage,
- const unsigned char simulcast_idx);
-
- // Gets SSRC for outgoing stream.
- WebRtc_Word32 GetLocalSSRC(WebRtc_UWord32& SSRC);
-
- // Gets SSRC for the incoming stream.
- WebRtc_Word32 GetRemoteSSRC(WebRtc_UWord32& SSRC);
-
- // Gets the CSRC for the incoming stream.
- WebRtc_Word32 GetRemoteCSRC(unsigned int CSRCs[kRtpCsrcSize]);
-
- // Sets the starting sequence number, must be called before StartSend.
- WebRtc_Word32 SetStartSequenceNumber(WebRtc_UWord16 sequence_number);
-
- // Sets the CName for the outgoing stream on the channel.
- WebRtc_Word32 SetRTCPCName(const WebRtc_Word8 rtcp_cname[]);
-
- // Gets the CName for the outgoing stream on the channel.
- WebRtc_Word32 GetRTCPCName(WebRtc_Word8 rtcp_cname[]);
-
- // Gets the CName of the incoming stream.
- WebRtc_Word32 GetRemoteRTCPCName(WebRtc_Word8 rtcp_cname[]);
- WebRtc_Word32 RegisterRtpObserver(ViERTPObserver* observer);
- WebRtc_Word32 RegisterRtcpObserver(ViERTCPObserver* observer);
- WebRtc_Word32 SendApplicationDefinedRTCPPacket(
- const WebRtc_UWord8 sub_type,
- WebRtc_UWord32 name,
- const WebRtc_UWord8* data,
- WebRtc_UWord16 data_length_in_bytes);
-
- // Returns statistics reported by the remote client in an RTCP packet.
- WebRtc_Word32 GetSendRtcpStatistics(WebRtc_UWord16& fraction_lost,
- WebRtc_UWord32& cumulative_lost,
- WebRtc_UWord32& extended_max,
- WebRtc_UWord32& jitter_samples,
- WebRtc_Word32& rtt_ms);
-
- // Returns our localy created statistics of the received RTP stream.
- WebRtc_Word32 GetReceivedRtcpStatistics(WebRtc_UWord16& fraction_lost,
- WebRtc_UWord32& cumulative_lost,
- WebRtc_UWord32& extended_max,
- WebRtc_UWord32& jitter_samples,
- WebRtc_Word32& rtt_ms);
-
- // Gets sent/received packets statistics.
- WebRtc_Word32 GetRtpStatistics(WebRtc_UWord32& bytes_sent,
- WebRtc_UWord32& packets_sent,
- WebRtc_UWord32& bytes_received,
- WebRtc_UWord32& packets_received) const;
- void GetBandwidthUsage(WebRtc_UWord32& total_bitrate_sent,
- WebRtc_UWord32& video_bitrate_sent,
- WebRtc_UWord32& fec_bitrate_sent,
- WebRtc_UWord32& nackBitrateSent) const;
- int GetEstimatedReceiveBandwidth(WebRtc_UWord32* estimated_bandwidth) const;
- WebRtc_Word32 SetKeepAliveStatus(const bool enable,
- const WebRtc_Word8 unknown_payload_type,
- const WebRtc_UWord16 delta_transmit_timeMS);
- WebRtc_Word32 GetKeepAliveStatus(bool& enable,
- WebRtc_Word8& unknown_payload_type,
- WebRtc_UWord16& delta_transmit_timeMS);
- WebRtc_Word32 StartRTPDump(const char file_nameUTF8[1024],
- RTPDirections direction);
- WebRtc_Word32 StopRTPDump(RTPDirections direction);
-
- // Implements RtcpFeedback.
- virtual void OnLipSyncUpdate(const WebRtc_Word32 id,
- const WebRtc_Word32 audio_video_offset);
- virtual void OnApplicationDataReceived(const WebRtc_Word32 id,
- const WebRtc_UWord8 sub_type,
- const WebRtc_UWord32 name,
- const WebRtc_UWord16 length,
- const WebRtc_UWord8* data);
-
- // Implements RtpFeedback.
- virtual WebRtc_Word32 OnInitializeDecoder(
- const WebRtc_Word32 id,
- const WebRtc_Word8 payload_type,
- const WebRtc_Word8 payload_name[RTP_PAYLOAD_NAME_SIZE],
- const int frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate);
- virtual void OnPacketTimeout(const WebRtc_Word32 id);
- virtual void OnReceivedPacket(const WebRtc_Word32 id,
- const RtpRtcpPacketType packet_type);
- virtual void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
- const RTPAliveType alive);
- virtual void OnIncomingSSRCChanged(const WebRtc_Word32 id,
- const WebRtc_UWord32 SSRC);
- virtual void OnIncomingCSRCChanged(const WebRtc_Word32 id,
- const WebRtc_UWord32 CSRC,
- const bool added);
-
- WebRtc_Word32 SetLocalReceiver(const WebRtc_UWord16 rtp_port,
- const WebRtc_UWord16 rtcp_port,
- const WebRtc_Word8* ip_address);
- WebRtc_Word32 GetLocalReceiver(WebRtc_UWord16& rtp_port,
- WebRtc_UWord16& rtcp_port,
- WebRtc_Word8* ip_address) const;
- WebRtc_Word32 SetSendDestination(const WebRtc_Word8* ip_address,
- const WebRtc_UWord16 rtp_port,
- const WebRtc_UWord16 rtcp_port,
- const WebRtc_UWord16 source_rtp_port,
- const WebRtc_UWord16 source_rtcp_port);
- WebRtc_Word32 GetSendDestination(WebRtc_Word8* ip_address,
- WebRtc_UWord16& rtp_port,
- WebRtc_UWord16& rtcp_port,
- WebRtc_UWord16& source_rtp_port,
- WebRtc_UWord16& source_rtcp_port) const;
- WebRtc_Word32 GetSourceInfo(WebRtc_UWord16& rtp_port,
- WebRtc_UWord16& rtcp_port,
- WebRtc_Word8* ip_address,
- WebRtc_UWord32 ip_address_length);
-
- WebRtc_Word32 SetRemoteSSRCType(const StreamType usage,
- const uint32_t SSRC) const;
-
- WebRtc_Word32 StartSend();
- WebRtc_Word32 StopSend();
- bool Sending();
- WebRtc_Word32 StartReceive();
- WebRtc_Word32 StopReceive();
- bool Receiving();
-
- WebRtc_Word32 RegisterSendTransport(Transport& transport);
- WebRtc_Word32 DeregisterSendTransport();
-
- // Incoming packet from external transport.
- WebRtc_Word32 ReceivedRTPPacket(const void* rtp_packet,
- const WebRtc_Word32 rtp_packet_length);
-
- // Incoming packet from external transport.
- WebRtc_Word32 ReceivedRTCPPacket(const void* rtcp_packet,
- const WebRtc_Word32 rtcp_packet_length);
-
- WebRtc_Word32 EnableIPv6();
- bool IsIPv6Enabled();
- WebRtc_Word32 SetSourceFilter(const WebRtc_UWord16 rtp_port,
- const WebRtc_UWord16 rtcp_port,
- const WebRtc_Word8* ip_address);
- WebRtc_Word32 GetSourceFilter(WebRtc_UWord16& rtp_port,
- WebRtc_UWord16& rtcp_port,
- WebRtc_Word8* ip_address) const;
-
- WebRtc_Word32 SetToS(const WebRtc_Word32 DSCP, const bool use_set_sockOpt);
- WebRtc_Word32 GetToS(WebRtc_Word32& DSCP, bool& use_set_sockOpt) const;
- WebRtc_Word32 SetSendGQoS(const bool enable,
- const WebRtc_Word32 service_type,
- const WebRtc_UWord32 max_bitrate,
- const WebRtc_Word32 overrideDSCP);
- WebRtc_Word32 GetSendGQoS(bool& enabled, WebRtc_Word32& service_type,
- WebRtc_Word32& overrideDSCP) const;
-
- // Sets the maximum transfer unit size for the network link, i.e. including
- // IP, UDP and RTP headers.
- WebRtc_Word32 SetMTU(WebRtc_UWord16 mtu);
-
- // Returns maximum allowed payload size, i.e. the maximum allowed size of
- // encoded data in each packet.
- WebRtc_UWord16 MaxDataPayloadLength() const;
- WebRtc_Word32 SetMaxPacketBurstSize(WebRtc_UWord16 max_number_of_packets);
- WebRtc_Word32 SetPacketBurstSpreadState(bool enable,
- const WebRtc_UWord16 frame_periodMS);
-
- WebRtc_Word32 SetPacketTimeoutNotification(bool enable,
- WebRtc_UWord32 timeout_seconds);
- WebRtc_Word32 RegisterNetworkObserver(ViENetworkObserver* observer);
- bool NetworkObserverRegistered();
- WebRtc_Word32 SetPeriodicDeadOrAliveStatus(
- const bool enable, const WebRtc_UWord32 sample_time_seconds);
-
- WebRtc_Word32 SendUDPPacket(const WebRtc_Word8* data,
- const WebRtc_UWord32 length,
- WebRtc_Word32& transmitted_bytes,
- bool use_rtcp_socket);
-
- WebRtc_Word32 EnableColorEnhancement(bool enable);
-
- // Register send RTP RTCP module, which will deliver encoded frames to the
- // to the channel RTP module.
- WebRtc_Word32 RegisterSendRtpRtcpModule(RtpRtcp& send_rtp_rtcp_module);
-
- // Deregisters the send RTP RTCP module, which will stop the encoder input to
- // the channel.
- WebRtc_Word32 DeregisterSendRtpRtcpModule();
-
- // Gets the modules used by the channel.
- RtpRtcp* rtp_rtcp();
-
- // Implements VCMReceiveCallback.
- virtual WebRtc_Word32 FrameToRender(VideoFrame& video_frame);
-
- // Implements VCMReceiveCallback.
- virtual WebRtc_Word32 ReceivedDecodedReferenceFrame(
- const WebRtc_UWord64 picture_id);
-
- // Implements VCM.
- virtual WebRtc_Word32 StoreReceivedFrame(
- const EncodedVideoData& frame_to_store);
-
- // Implements VideoReceiveStatisticsCallback.
- virtual WebRtc_Word32 ReceiveStatistics(const WebRtc_UWord32 bit_rate,
- const WebRtc_UWord32 frame_rate);
-
- // Implements VideoFrameTypeCallback.
- virtual WebRtc_Word32 RequestKeyFrame();
-
- // Implements VideoFrameTypeCallback.
- virtual WebRtc_Word32 SliceLossIndicationRequest(
- const WebRtc_UWord64 picture_id);
-
- // Implements VideoPacketRequestCallback.
- virtual WebRtc_Word32 ResendPackets(const WebRtc_UWord16* sequence_numbers,
- WebRtc_UWord16 length);
-
- WebRtc_Word32 RegisterExternalEncryption(Encryption* encryption);
- WebRtc_Word32 DeRegisterExternalEncryption();
-
- WebRtc_Word32 SetVoiceChannel(WebRtc_Word32 ve_channel_id,
- VoEVideoSync* ve_sync_interface);
- WebRtc_Word32 VoiceChannel();
-
- // Implements ViEFrameProviderBase.
- virtual int FrameCallbackChanged() {return -1;}
-
- WebRtc_Word32 RegisterEffectFilter(ViEEffectFilter* effect_filter);
-
- ViEFileRecorder& GetIncomingFileRecorder();
- void ReleaseIncomingFileRecorder();
-
- protected:
- static bool ChannelDecodeThreadFunction(void* obj);
- bool ChannelDecodeProcess();
-
- private:
- // Assumed to be protected.
- WebRtc_Word32 StartDecodeThread();
- WebRtc_Word32 StopDecodeThread();
-
- WebRtc_Word32 ProcessNACKRequest(const bool enable);
- WebRtc_Word32 ProcessFECRequest(const bool enable,
- const unsigned char payload_typeRED,
- const unsigned char payload_typeFEC);
-
- WebRtc_Word32 channel_id_;
- WebRtc_Word32 engine_id_;
- WebRtc_UWord32 number_of_cores_;
- WebRtc_UWord8 num_socket_threads_;
-
- // Used for all registered callbacks except rendering.
- scoped_ptr<CriticalSectionWrapper> callback_cs_;
-
- // Owned modules/classes.
- RtpRtcp& rtp_rtcp_;
- RtpRtcp* default_rtp_rtcp_;
- std::list<RtpRtcp*> simulcast_rtp_rtcp_;
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- UdpTransport& socket_transport_;
-#endif
- VideoCodingModule& vcm_;
- ViEReceiver& vie_receiver_;
- ViESender& vie_sender_;
- ViESyncModule& vie_sync_;
-
- // Not owned.
- ProcessThread& module_process_thread_;
- ViEDecoderObserver* codec_observer_;
- bool do_key_frame_callbackRequest_;
- ViERTPObserver* rtp_observer_;
- ViERTCPObserver* rtcp_observer_;
- ViENetworkObserver* networkObserver_;
- bool rtp_packet_timeout_;
- bool using_packet_spread_;
-
- Transport* external_transport_;
-
- bool decoder_reset_;
- bool wait_for_key_frame_;
- ThreadWrapper* decode_thread_;
-
- Encryption* external_encryption_;
-
- ViEEffectFilter* effect_filter_;
- bool color_enhancement_;
-
- // Time when RTT time was last reported to VCM JB.
- TickTime vcm_rttreported_;
-
- ViEFileRecorder file_recorder_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_H_
diff --git a/trunk/src/video_engine/vie_channel_manager.cc b/trunk/src/video_engine/vie_channel_manager.cc
deleted file mode 100644
index 268bc18..0000000
--- a/trunk/src/video_engine/vie_channel_manager.cc
+++ /dev/null
@@ -1,484 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_channel_manager.h"
-
-#include "engine_configurations.h"
-#include "modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "modules/utility/interface/process_thread.h"
-#include "system_wrappers/interface/critical_section_wrapper.h"
-#include "system_wrappers/interface/map_wrapper.h"
-#include "system_wrappers/interface/trace.h"
-#include "video_engine/vie_channel.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_encoder.h"
-#include "video_engine/vie_remb.h"
-#include "voice_engine/main/interface/voe_video_sync.h"
-
-namespace webrtc {
-
-ViEChannelManager::ViEChannelManager(
- int engine_id,
- int number_of_cores,
- ViEPerformanceMonitor& vie_performance_monitor)
- : channel_id_critsect_(CriticalSectionWrapper::CreateCriticalSection()),
- engine_id_(engine_id),
- number_of_cores_(number_of_cores),
- vie_performance_monitor_(vie_performance_monitor),
- free_channel_ids_(new bool[kViEMaxNumberOfChannels]),
- free_channel_ids_size_(kViEMaxNumberOfChannels),
- voice_sync_interface_(NULL),
- remb_(new VieRemb(engine_id)),
- voice_engine_(NULL),
- module_process_thread_(NULL) {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, ViEId(engine_id),
- "ViEChannelManager::ViEChannelManager(engine_id: %d)",
- engine_id);
- for (int idx = 0; idx < free_channel_ids_size_; idx++) {
- free_channel_ids_[idx] = true;
- }
-}
-
-ViEChannelManager::~ViEChannelManager() {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, ViEId(engine_id_),
- "ViEChannelManager Destructor, engine_id: %d", engine_id_);
-
- module_process_thread_->DeRegisterModule(remb_.get());
- while (channel_map_.size() > 0) {
- ChannelMap::iterator it = channel_map_.begin();
- // DeleteChannel will erase this channel from the map and invalidate |it|.
- DeleteChannel(it->first);
- }
-
- if (voice_sync_interface_) {
- voice_sync_interface_->Release();
- }
- if (channel_id_critsect_) {
- delete channel_id_critsect_;
- channel_id_critsect_ = NULL;
- }
- if (free_channel_ids_) {
- delete[] free_channel_ids_;
- free_channel_ids_ = NULL;
- free_channel_ids_size_ = 0;
- }
-}
-
-void ViEChannelManager::SetModuleProcessThread(
- ProcessThread& module_process_thread) {
- assert(!module_process_thread_);
- module_process_thread_ = &module_process_thread;
- module_process_thread_->RegisterModule(remb_.get());
-}
-
-int ViEChannelManager::CreateChannel(int& channel_id) {
- CriticalSectionScoped cs(*channel_id_critsect_);
-
- // Get a free id for the new channel.
- if (!GetFreeChannelId(channel_id)) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
- "Max number of channels reached: %d", channel_map_.size());
- return -1;
- }
-
- ViEChannel* vie_channel = new ViEChannel(channel_id, engine_id_,
- number_of_cores_,
- *module_process_thread_);
- if (!vie_channel) {
- ReturnChannelId(channel_id);
- return -1;
- }
- if (vie_channel->Init() != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
- "%s could not init channel", __FUNCTION__, channel_id);
- ReturnChannelId(channel_id);
- delete vie_channel;
- vie_channel = NULL;
- return -1;
- }
-
- // There is no ViEEncoder for this channel, create one with default settings.
- ViEEncoder* vie_encoder = new ViEEncoder(engine_id_, channel_id,
- number_of_cores_,
- *module_process_thread_);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
- "%s(video_channel_id: %d) - Could not create a new encoder",
- __FUNCTION__, channel_id);
- delete vie_channel;
- return -1;
- }
-
- vie_encoder_map_[channel_id] = vie_encoder;
- channel_map_[channel_id] = vie_channel;
-
- // Register the channel at the encoder.
- RtpRtcp* send_rtp_rtcp_module = vie_encoder->SendRtpRtcpModule();
- if (vie_channel->RegisterSendRtpRtcpModule(*send_rtp_rtcp_module) != 0) {
- assert(false);
- vie_encoder_map_.erase(channel_id);
- channel_map_.erase(channel_id);
- ReturnChannelId(channel_id);
- delete vie_channel;
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id),
- "%s: Could not register rtp module %d", __FUNCTION__,
- channel_id);
- return -1;
- }
- return 0;
-}
-
-int ViEChannelManager::CreateChannel(int& channel_id, int original_channel) {
- CriticalSectionScoped cs(*channel_id_critsect_);
-
- // Check that original_channel already exists.
- ViEEncoder* vie_encoder = ViEEncoderPtr(original_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
- "%s: Original channel doesn't exist", __FUNCTION__,
- original_channel);
- return -1;
- }
- VideoCodec video_codec;
- vie_encoder->GetEncoder(video_codec);
-
- // Get a free id for the new channel.
- if (GetFreeChannelId(channel_id) == false) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
- "Max number of channels reached: %d", channel_map_.size());
- return -1;
- }
- ViEChannel* vie_channel = new ViEChannel(channel_id, engine_id_,
- number_of_cores_,
- *module_process_thread_);
- if (!vie_channel) {
- ReturnChannelId(channel_id);
- return -1;
- }
- if (vie_channel->Init() != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
- "%s could not init channel", __FUNCTION__, channel_id);
- ReturnChannelId(channel_id);
- delete vie_channel;
- vie_channel = NULL;
- return -1;
- }
- vie_encoder_map_[channel_id] = vie_encoder;
-
- // Set the same encoder settings for the channel as used by the master
- // channel. Do this before attaching rtp module to ensure all rtp children has
- // the same codec type.
- VideoCodec encoder;
- if (vie_encoder->GetEncoder(encoder) == 0) {
- vie_channel->SetSendCodec(encoder);
- }
- channel_map_[channel_id] = vie_channel;
-
- // Register the channel at the encoder.
- RtpRtcp* send_rtp_rtcp_module = vie_encoder->SendRtpRtcpModule();
- if (vie_channel->RegisterSendRtpRtcpModule(*send_rtp_rtcp_module) != 0) {
- assert(false);
- vie_encoder_map_.erase(channel_id);
- channel_map_.erase(channel_id);
- ReturnChannelId(channel_id);
- delete vie_channel;
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id),
- "%s: Could not register rtp module %d", __FUNCTION__,
- channel_id);
- return -1;
- }
- return 0;
-}
-
-int ViEChannelManager::DeleteChannel(int channel_id) {
- ViEChannel* vie_channel = NULL;
- ViEEncoder* vie_encoder = NULL;
- {
- // Write lock to make sure no one is using the channel.
- ViEManagerWriteScoped wl(*this);
-
- // Protect the map.
- CriticalSectionScoped cs(*channel_id_critsect_);
- ChannelMap::iterator c_it = channel_map_.find(channel_id);
- if (c_it == channel_map_.end()) {
- // No such channel.
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
- "%s Channel doesn't exist: %d", __FUNCTION__, channel_id);
- return -1;
- }
- vie_channel = c_it->second;
- channel_map_.erase(c_it);
-
- // Deregister possible remb modules.
- RtpRtcp* rtp_module = vie_channel->rtp_rtcp();
- remb_->RemoveRembSender(rtp_module);
- remb_->RemoveReceiveChannel(rtp_module);
-
- // Deregister the channel from the ViEEncoder to stop the media flow.
- vie_channel->DeregisterSendRtpRtcpModule();
- ReturnChannelId(channel_id);
-
- // Find the encoder object.
- EncoderMap::iterator e_it = vie_encoder_map_.find(channel_id);
- assert(e_it != vie_encoder_map_.end());
- vie_encoder = e_it->second;
-
- remb_->RemoveSendChannel(vie_encoder->SendRtpRtcpModule());
-
- // Check if other channels are using the same encoder.
- if (ChannelUsingViEEncoder(channel_id)) {
- // Don't delete the ViEEncoder, at least one other channel is using it.
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_),
- "%s ViEEncoder removed from map for channel %d, not deleted",
- __FUNCTION__, channel_id);
- vie_encoder = NULL;
- } else {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_),
- "%s ViEEncoder deleted for channel %d", __FUNCTION__,
- channel_id);
- // Delete later when we've released the critsect.
- }
-
- // We can't erase the item before we've checked for other channels using
- // same ViEEncoder.
- vie_encoder_map_.erase(e_it);
- }
-
- // Leave the write critsect before deleting the objects.
- // Deleting a channel can cause other objects, such as renderers, to be
- // deleted, which might take time.
- if (vie_encoder) {
- delete vie_encoder;
- }
- delete vie_channel;
-
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_),
- "%s Channel %d deleted", __FUNCTION__, channel_id);
- return 0;
-}
-
-int ViEChannelManager::SetVoiceEngine(VoiceEngine* voice_engine) {
- // Write lock to make sure no one is using the channel.
- ViEManagerWriteScoped wl(*this);
-
- CriticalSectionScoped cs(*channel_id_critsect_);
-
- VoEVideoSync* sync_interface = NULL;
- if (voice_engine) {
- // Get new sync interface.
- sync_interface = VoEVideoSync::GetInterface(voice_engine);
- if (!sync_interface) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
- "%s Can't get audio sync interface from VoiceEngine.",
- __FUNCTION__);
- return -1;
- }
- }
-
- for (ChannelMap::iterator it = channel_map_.begin(); it != channel_map_.end();
- ++it) {
- it->second->SetVoiceChannel(-1, sync_interface);
- }
- if (voice_sync_interface_) {
- voice_sync_interface_->Release();
- }
- voice_engine_ = voice_engine;
- voice_sync_interface_ = sync_interface;
- return 0;
-}
-
-int ViEChannelManager::ConnectVoiceChannel(int channel_id,
- int audio_channel_id) {
- CriticalSectionScoped cs(*channel_id_critsect_);
- if (!voice_sync_interface_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, channel_id),
- "No VoE set");
- return -1;
- }
- ViEChannel* channel = ViEChannelPtr(channel_id);
- if (!channel) {
- return -1;
- }
- return channel->SetVoiceChannel(audio_channel_id, voice_sync_interface_);
-}
-
-int ViEChannelManager::DisconnectVoiceChannel(int channel_id) {
- CriticalSectionScoped cs(*channel_id_critsect_);
- ViEChannel* channel = ViEChannelPtr(channel_id);
- if (channel) {
- channel->SetVoiceChannel(-1, NULL);
- return 0;
- }
- return -1;
-}
-
-VoiceEngine* ViEChannelManager::GetVoiceEngine() {
- CriticalSectionScoped cs(*channel_id_critsect_);
- return voice_engine_;
-}
-
-bool ViEChannelManager::SetRembStatus(int channel_id, bool sender,
- bool receiver) {
- CriticalSectionScoped cs(*channel_id_critsect_);
- ViEChannel* channel = ViEChannelPtr(channel_id);
- if (!channel) {
- return false;
- }
-
- ViEEncoder* encoder = ViEEncoderPtr(channel_id);
-
- if (sender || receiver) {
- if (!channel->EnableRemb(true)) {
- return false;
- }
- } else {
- channel->EnableRemb(false);
- }
- RtpRtcp* rtp_module = channel->rtp_rtcp();
- if (sender) {
- remb_->AddRembSender(rtp_module);
- remb_->AddSendChannel(encoder->SendRtpRtcpModule());
- } else {
- remb_->RemoveRembSender(rtp_module);
- remb_->RemoveSendChannel(encoder->SendRtpRtcpModule());
- }
- if (receiver) {
- remb_->AddReceiveChannel(rtp_module);
- } else {
- remb_->RemoveReceiveChannel(rtp_module);
- }
- if (sender || receiver) {
- rtp_module->SetRemoteBitrateObserver(remb_.get());
- } else {
- rtp_module->SetRemoteBitrateObserver(NULL);
- }
- return true;
-}
-
-ViEChannel* ViEChannelManager::ViEChannelPtr(int channel_id) const {
- CriticalSectionScoped cs(*channel_id_critsect_);
- ChannelMap::const_iterator it = channel_map_.find(channel_id);
- if (it == channel_map_.end()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_),
- "%s Channel doesn't exist: %d", __FUNCTION__, channel_id);
- return NULL;
- }
- return it->second;
-}
-
-void ViEChannelManager::GetViEChannels(MapWrapper& channel_map) {
- CriticalSectionScoped cs(*channel_id_critsect_);
- if (channel_map.Size() == 0) {
- return;
- }
-
- // Add all items to 'channelMap'.
- for (ChannelMap::iterator it = channel_map_.begin(); it != channel_map_.end();
- ++it) {
- channel_map.Insert(it->first, it->second);
- }
-}
-
-ViEEncoder* ViEChannelManager::ViEEncoderPtr(int video_channel_id) const {
- CriticalSectionScoped cs(*channel_id_critsect_);
- EncoderMap::const_iterator it = vie_encoder_map_.find(video_channel_id);
- if (it == vie_encoder_map_.end()) {
- return NULL;
- }
- return it->second;
-}
-
-bool ViEChannelManager::GetFreeChannelId(int& free_channel_id) {
- CriticalSectionScoped cs(*channel_id_critsect_);
- int idx = 0;
- while (idx < free_channel_ids_size_) {
- if (free_channel_ids_[idx] == true) {
- // We've found a free id, allocate it and return.
- free_channel_ids_[idx] = false;
- free_channel_id = idx + kViEChannelIdBase;
- return true;
- }
- idx++;
- }
- // No free channel id.
- free_channel_id = -1;
- return false;
-}
-
-void ViEChannelManager::ReturnChannelId(int channel_id) {
- CriticalSectionScoped cs(*channel_id_critsect_);
- assert(channel_id < kViEMaxNumberOfChannels + kViEChannelIdBase &&
- channel_id >= kViEChannelIdBase);
- free_channel_ids_[channel_id - kViEChannelIdBase] = true;
-}
-
-bool ViEChannelManager::ChannelUsingViEEncoder(int channel_id) const {
- CriticalSectionScoped cs(*channel_id_critsect_);
- EncoderMap::const_iterator orig_it = vie_encoder_map_.find(channel_id);
- if(orig_it == vie_encoder_map_.end()) {
- // No ViEEncoder for this channel.
- return false;
- }
-
- // Loop through all other channels to see if anyone points at the same
- // ViEEncoder.
- for (EncoderMap::const_iterator comp_it = vie_encoder_map_.begin();
- comp_it != vie_encoder_map_.end(); ++comp_it) {
- // Make sure we're not comparing the same channel with itself.
- if (comp_it->first != channel_id) {
- if (comp_it->second == orig_it->second) {
- return true;
- }
- }
- }
- return false;
-}
-
-void ViEChannelManager::ChannelsUsingViEEncoder(int channel_id,
- ChannelList* channels) const {
- CriticalSectionScoped cs(*channel_id_critsect_);
- EncoderMap::const_iterator orig_it = vie_encoder_map_.find(channel_id);
-
- for (ChannelMap::const_iterator c_it = channel_map_.begin();
- c_it != channel_map_.end(); ++c_it) {
- EncoderMap::const_iterator comp_it = vie_encoder_map_.find(c_it->first);
- assert(comp_it != vie_encoder_map_.end());
- if (comp_it->second == orig_it->second) {
- channels->push_back(c_it->second);
- }
- }
-}
-
-ViEChannelManagerScoped::ViEChannelManagerScoped(
- const ViEChannelManager& vie_channel_manager)
- : ViEManagerScopedBase(vie_channel_manager) {
-}
-
-ViEChannel* ViEChannelManagerScoped::Channel(int vie_channel_id) const {
- return static_cast<const ViEChannelManager*>(vie_manager_)->ViEChannelPtr(
- vie_channel_id);
-}
-ViEEncoder* ViEChannelManagerScoped::Encoder(int vie_channel_id) const {
- return static_cast<const ViEChannelManager*>(vie_manager_)->ViEEncoderPtr(
- vie_channel_id);
-}
-
-bool ViEChannelManagerScoped::ChannelUsingViEEncoder(int channel_id) const {
- return (static_cast<const ViEChannelManager*>(vie_manager_))->
- ChannelUsingViEEncoder(channel_id);
-}
-
-void ViEChannelManagerScoped::ChannelsUsingViEEncoder(
- int channel_id, ChannelList* channels) const {
- (static_cast<const ViEChannelManager*>(vie_manager_))->
- ChannelsUsingViEEncoder(channel_id, channels);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_channel_manager.h b/trunk/src/video_engine/vie_channel_manager.h
deleted file mode 100644
index f0fa348..0000000
--- a/trunk/src/video_engine/vie_channel_manager.h
+++ /dev/null
@@ -1,131 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_MANAGER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_MANAGER_H_
-
-#include <list>
-#include <map>
-
-#include "engine_configurations.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-#include "typedefs.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_manager_base.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class MapWrapper;
-class ProcessThread;
-class ViEChannel;
-class ViEEncoder;
-class ViEPerformanceMonitor;
-class VieRemb;
-class VoEVideoSync;
-class VoiceEngine;
-
-typedef std::list<ViEChannel*> ChannelList;
-typedef std::map<int, ViEChannel*> ChannelMap;
-typedef std::map<int, ViEEncoder*> EncoderMap;
-
-class ViEChannelManager: private ViEManagerBase {
- friend class ViEChannelManagerScoped;
- public:
- ViEChannelManager(int engine_id,
- int number_of_cores,
- ViEPerformanceMonitor& vie_performance_monitor);
- ~ViEChannelManager();
-
- void SetModuleProcessThread(ProcessThread& module_process_thread);
-
- // Creates a new channel. 'channelId' will be the id of the created channel.
- int CreateChannel(int& channel_id);
-
- // Creates a channel and attaches to an already existing ViEEncoder.
- int CreateChannel(int& channel_id, int original_channel);
-
- // Deletes a channel.
- int DeleteChannel(int channel_id);
-
- // Set the voice engine instance to be used by all video channels.
- int SetVoiceEngine(VoiceEngine* voice_engine);
-
- // Enables lip sync of the channel.
- int ConnectVoiceChannel(int channel_id, int audio_channel_id);
-
- // Disables lip sync of the channel.
- int DisconnectVoiceChannel(int channel_id);
-
- VoiceEngine* GetVoiceEngine();
-
- // Adds a channel to include when sending REMB.
- bool SetRembStatus(int channel_id, bool sender, bool receiver);
-
- private:
- // Used by ViEChannelScoped, forcing a manager user to use scoped.
- // Returns a pointer to the channel with id 'channelId'.
- ViEChannel* ViEChannelPtr(int channel_id) const;
-
- // Adds all channels to channel_map.
- void GetViEChannels(MapWrapper& channel_map);
-
- // Methods used by ViECaptureScoped and ViEEncoderScoped.
- // Gets the ViEEncoder used as input for video_channel_id
- ViEEncoder* ViEEncoderPtr(int video_channel_id) const;
-
- // Returns true if we found a new channel id, free_channel_id, false
- // otherwise.
- bool GetFreeChannelId(int& free_channel_id);
-
- // Returns a previously allocated channel id.
- void ReturnChannelId(int channel_id);
-
- // Returns true if at least one other channels uses the same ViEEncoder as
- // channel_id.
- bool ChannelUsingViEEncoder(int channel_id) const;
- void ChannelsUsingViEEncoder(int channel_id, ChannelList* channels) const;
-
- // Protects channel_map_ and free_channel_ids_.
- CriticalSectionWrapper* channel_id_critsect_;
- int engine_id_;
- int number_of_cores_;
- ViEPerformanceMonitor& vie_performance_monitor_;
- ChannelMap channel_map_;
- bool* free_channel_ids_;
- int free_channel_ids_size_;
-
- // Maps Channel id -> ViEEncoder.
- EncoderMap vie_encoder_map_;
- VoEVideoSync* voice_sync_interface_;
- scoped_ptr<VieRemb> remb_;
- VoiceEngine* voice_engine_;
- ProcessThread* module_process_thread_;
-};
-
-class ViEChannelManagerScoped: private ViEManagerScopedBase {
- public:
- explicit ViEChannelManagerScoped(
- const ViEChannelManager& vie_channel_manager);
- ViEChannel* Channel(int vie_channel_id) const;
- ViEEncoder* Encoder(int vie_channel_id) const;
-
- // Returns true if at least one other channels uses the same ViEEncoder as
- // channel_id.
- bool ChannelUsingViEEncoder(int channel_id) const;
-
- // Returns a list with pointers to all channels using the same encoder as the
- // channel with |channel_id|, including the one with the specified id.
- void ChannelsUsingViEEncoder(int channel_id, ChannelList* channels) const;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_CHANNEL_MANAGER_H_
diff --git a/trunk/src/video_engine/vie_codec_impl.cc b/trunk/src/video_engine/vie_codec_impl.cc
deleted file mode 100644
index f995117..0000000
--- a/trunk/src/video_engine/vie_codec_impl.cc
+++ /dev/null
@@ -1,748 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_codec_impl.h"
-
-#include "engine_configurations.h"
-#include "modules/video_coding/main/interface/video_coding.h"
-#include "system_wrappers/interface/trace.h"
-#include "video_engine/include/vie_errors.h"
-#include "video_engine/vie_capturer.h"
-#include "video_engine/vie_channel.h"
-#include "video_engine/vie_channel_manager.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_encoder.h"
-#include "video_engine/vie_impl.h"
-#include "video_engine/vie_input_manager.h"
-#include "video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-ViECodec* ViECodec::GetInterface(VideoEngine* video_engine) {
-#ifdef WEBRTC_VIDEO_ENGINE_CODEC_API
- if (!video_engine) {
- return NULL;
- }
- VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
- ViECodecImpl* vie_codec_impl = vie_impl;
- // Increase ref count.
- (*vie_codec_impl)++;
- return vie_codec_impl;
-#else
- return NULL;
-#endif
-}
-
-int ViECodecImpl::Release() {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
- "ViECodecImpl::Release()");
- // Decrease ref count.
- (*this)--;
-
- WebRtc_Word32 ref_count = GetCount();
- if (ref_count < 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
- "ViECodec released too many times");
- shared_data_->SetLastError(kViEAPIDoesNotExist);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
- "ViECodec reference count: %d", ref_count);
- return ref_count;
-}
-
-ViECodecImpl::ViECodecImpl(ViESharedData* shared_data)
- : shared_data_(shared_data) {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViECodecImpl::ViECodecImpl() Ctor");
-}
-
-ViECodecImpl::~ViECodecImpl() {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViECodecImpl::~ViECodecImpl() Dtor");
-}
-
-int ViECodecImpl::NumberOfCodecs() const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s", __FUNCTION__);
- // +2 because of FEC(RED and ULPFEC)
- return static_cast<int>((VideoCodingModule::NumberOfCodecs() + 2));
-}
-
-int ViECodecImpl::GetCodec(const unsigned char list_number,
- VideoCodec& video_codec) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(list_number: %d, codec_type: %d)", __FUNCTION__,
- list_number, video_codec.codecType);
- if (list_number == VideoCodingModule::NumberOfCodecs()) {
- memset(&video_codec, 0, sizeof(VideoCodec));
- strcpy(video_codec.plName, "red");
- video_codec.codecType = kVideoCodecRED;
- video_codec.plType = VCM_RED_PAYLOAD_TYPE;
- } else if (list_number == VideoCodingModule::NumberOfCodecs() + 1) {
- memset(&video_codec, 0, sizeof(VideoCodec));
- strcpy(video_codec.plName, "ulpfec");
- video_codec.codecType = kVideoCodecULPFEC;
- video_codec.plType = VCM_ULPFEC_PAYLOAD_TYPE;
- } else if (VideoCodingModule::Codec(list_number, &video_codec) != VCM_OK) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Could not get codec for list_number: %u", __FUNCTION__,
- list_number);
- shared_data_->SetLastError(kViECodecInvalidArgument);
- return -1;
- }
- return 0;
-}
-
-int ViECodecImpl::SetSendCodec(const int video_channel,
- const VideoCodec& video_codec) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(),video_channel),
- "%s(video_channel: %d, codec_type: %d)", __FUNCTION__,
- video_channel, video_codec.codecType);
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: codec: %d, pl_type: %d, width: %d, height: %d, bitrate: %d"
- "maxBr: %d, min_br: %d, frame_rate: %d, qpMax: %u,"
- "numberOfSimulcastStreams: %u )", __FUNCTION__,
- video_codec.codecType, video_codec.plType, video_codec.width,
- video_codec.height, video_codec.startBitrate,
- video_codec.maxBitrate, video_codec.minBitrate,
- video_codec.maxFramerate, video_codec.qpMax,
- video_codec.numberOfSimulcastStreams);
- if (video_codec.codecType == kVideoCodecVP8) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "pictureLossIndicationOn: %d, feedbackModeOn: %d, "
- "complexity: %d, resilience: %d, numberOfTemporalLayers: %u",
- video_codec.codecSpecific.VP8.pictureLossIndicationOn,
- video_codec.codecSpecific.VP8.feedbackModeOn,
- video_codec.codecSpecific.VP8.complexity,
- video_codec.codecSpecific.VP8.resilience,
- video_codec.codecSpecific.VP8.numberOfTemporalLayers);
- }
- if (!CodecValid(video_codec)) {
- // Error logged.
- shared_data_->SetLastError(kViECodecInvalidCodec);
- return -1;
- }
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
-
- // Set a max_bitrate if the user hasn't set one.
- VideoCodec video_codec_internal;
- memcpy(&video_codec_internal, &video_codec, sizeof(VideoCodec));
- if (video_codec_internal.maxBitrate == 0) {
- // Max is one bit per pixel.
- video_codec_internal.maxBitrate = (video_codec_internal.width *
- video_codec_internal.height *
- video_codec_internal.maxFramerate)
- / 1000;
- if (video_codec_internal.startBitrate > video_codec_internal.maxBitrate) {
- // Don't limit the set start bitrate.
- video_codec_internal.maxBitrate = video_codec_internal.startBitrate;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: New max bitrate set to %d kbps", __FUNCTION__,
- video_codec_internal.maxBitrate);
- }
-
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- assert(false);
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No encoder found for channel %d", __FUNCTION__);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
-
- VideoCodec encoder;
- vie_encoder->GetEncoder(encoder);
- if (encoder.codecType != video_codec_internal.codecType &&
- cs.ChannelUsingViEEncoder(video_channel)) {
- // We don't allow changing codec type when several channels share encoder.
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Settings differs from other channels using encoder",
- __FUNCTION__);
- shared_data_->SetLastError(kViECodecInUse);
- return -1;
- }
- // Make sure to generate a new SSRC if the codec type and/or resolution has
- // changed. This won't have any effect if the user has set an SSRC.
- bool new_rtp_stream = false;
- if (encoder.codecType != video_codec_internal.codecType ||
- encoder.width != video_codec_internal.width ||
- encoder.height != video_codec_internal.height) {
- new_rtp_stream = true;
- }
- if (video_codec_internal.numberOfSimulcastStreams > 1) {
- if (cs.ChannelUsingViEEncoder(video_channel)) {
- // We don't allow simulcast channels to share encoder.
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Can't share simulcast encoder",
- __FUNCTION__);
- shared_data_->SetLastError(kViECodecInUse);
- return -1;
- }
- }
-
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViEFrameProviderBase* frame_provider = NULL;
-
- // Stop the media flow while reconfiguring.
- vie_encoder->Pause();
-
- // Check if we have a frame provider that is a camera and can provide this
- // codec for us.
- bool use_capture_device_as_encoder = false;
- frame_provider = is.FrameProvider(vie_encoder);
- if (frame_provider) {
- if (frame_provider->Id() >= kViECaptureIdBase &&
- frame_provider->Id() <= kViECaptureIdMax) {
- ViECapturer* vie_capture = static_cast<ViECapturer*>(frame_provider);
- // Try to get preencoded. Nothing to do if it is not supported.
- if (vie_capture && vie_capture->PreEncodeToViEEncoder(
- video_codec_internal,
- *vie_encoder,
- video_channel) == 0) {
- use_capture_device_as_encoder = true;
- }
- }
- }
-
- // Update the encoder settings if we are not using a capture device capable
- // of this codec.
- if (!use_capture_device_as_encoder &&
- vie_encoder->SetEncoder(video_codec_internal) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Could not change encoder for channel %d", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViECodecUnknownError);
- return -1;
- }
-
- // Give the channel(s) the new information.
- ChannelList channels;
- cs.ChannelsUsingViEEncoder(video_channel, &channels);
- for (ChannelList::iterator it = channels.begin(); it != channels.end();
- ++it) {
- bool ret = true;
- if ((*it)->SetSendCodec(video_codec_internal, new_rtp_stream) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Could not set send codec for channel %d", __FUNCTION__,
- video_channel);
- ret = false;
- }
- if (!ret) {
- shared_data_->SetLastError(kViECodecUnknownError);
- return -1;
- }
- }
-
- // Update the protection mode, we might be switching NACK/FEC.
- vie_encoder->UpdateProtectionMethod();
-
- // Get new best format for frame provider.
- if (frame_provider) {
- frame_provider->FrameCallbackChanged();
- }
- // Restart the media flow
- if (new_rtp_stream) {
- // Stream settings changed, make sure we get a key frame.
- vie_encoder->SendKeyFrame();
- }
- vie_encoder->Restart();
- return 0;
-}
-
-int ViECodecImpl::GetSendCodec(const int video_channel,
- VideoCodec& video_codec) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No encoder for channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
- return vie_encoder->GetEncoder(video_codec);
-}
-
-int ViECodecImpl::SetReceiveCodec(const int video_channel,
- const VideoCodec& video_codec) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d, codec_type: %d)", __FUNCTION__,
- video_channel, video_codec.codecType);
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: codec: %d, pl_type: %d, width: %d, height: %d, bitrate: %d,"
- "maxBr: %d, min_br: %d, frame_rate: %d", __FUNCTION__,
- video_codec.codecType, video_codec.plType, video_codec.width,
- video_codec.height, video_codec.startBitrate,
- video_codec.maxBitrate, video_codec.minBitrate,
- video_codec.maxFramerate);
-
- if (CodecValid(video_codec) == false) {
- // Error logged.
- shared_data_->SetLastError(kViECodecInvalidCodec);
- return -1;
- }
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
-
- if (vie_channel->SetReceiveCodec(video_codec) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Could not set receive codec for channel %d",
- __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViECodecImpl::GetReceiveCodec(const int video_channel,
- VideoCodec& video_codec) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d, codec_type: %d)", __FUNCTION__,
- video_channel, video_codec.codecType);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
-
- if (vie_channel->GetReceiveCodec(video_codec) != 0) {
- shared_data_->SetLastError(kViECodecUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViECodecImpl::GetCodecConfigParameters(
- const int video_channel,
- unsigned char config_parameters[kConfigParameterSize],
- unsigned char& config_parameters_size) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No encoder for channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
-
- if (vie_encoder->GetCodecConfigParameters(config_parameters,
- config_parameters_size) != 0) {
- shared_data_->SetLastError(kViECodecUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViECodecImpl::SetImageScaleStatus(const int video_channel,
- const bool enable) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d, enable: %d)", __FUNCTION__, video_channel,
- enable);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
-
- if (vie_encoder->ScaleInputImage(enable) != 0) {
- shared_data_->SetLastError(kViECodecUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViECodecImpl::GetSendCodecStastistics(const int video_channel,
- unsigned int& key_frames,
- unsigned int& delta_frames) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No send codec for channel %d", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
-
- if (vie_encoder->SendCodecStatistics(key_frames, delta_frames) != 0) {
- shared_data_->SetLastError(kViECodecUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViECodecImpl::GetReceiveCodecStastistics(const int video_channel,
- unsigned int& key_frames,
- unsigned int& delta_frames) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d, codec_type: %d)", __FUNCTION__,
- video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
- if (vie_channel->ReceiveCodecStatistics(key_frames, delta_frames) != 0) {
- shared_data_->SetLastError(kViECodecUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViECodecImpl::GetCodecTargetBitrate(const int video_channel,
- unsigned int* bitrate) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d, codec_type: %d)", __FUNCTION__,
- video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No send codec for channel %d", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
- return vie_encoder->CodecTargetBitrate(static_cast<WebRtc_UWord32*>(bitrate));
-}
-
-unsigned int ViECodecImpl::GetDiscardedPackets(const int video_channel) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d, codec_type: %d)", __FUNCTION__,
- video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
- return vie_channel->DiscardedPackets();
-}
-
-int ViECodecImpl::SetKeyFrameRequestCallbackStatus(const int video_channel,
- const bool enable) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
- if (vie_channel->EnableKeyFrameRequestCallback(enable) != 0) {
- shared_data_->SetLastError(kViECodecUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViECodecImpl::SetSignalKeyPacketLossStatus(const int video_channel,
- const bool enable,
- const bool only_key_frames) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d, enable: %d, only_key_frames: %d)",
- __FUNCTION__, video_channel, enable);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
- if (vie_channel->SetSignalPacketLossStatus(enable, only_key_frames) != 0) {
- shared_data_->SetLastError(kViECodecUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViECodecImpl::RegisterEncoderObserver(const int video_channel,
- ViEEncoderObserver& observer) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s", __FUNCTION__);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No encoder for channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
- if (vie_encoder->RegisterCodecObserver(&observer) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Could not register codec observer at channel",
- __FUNCTION__);
- shared_data_->SetLastError(kViECodecObserverAlreadyRegistered);
- return -1;
- }
- return 0;
-}
-
-int ViECodecImpl::DeregisterEncoderObserver(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s", __FUNCTION__);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No encoder for channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
- if (vie_encoder->RegisterCodecObserver(NULL) != 0) {
- shared_data_->SetLastError(kViECodecObserverNotRegistered);
- return -1;
- }
- return 0;
-}
-
-int ViECodecImpl::RegisterDecoderObserver(const int video_channel,
- ViEDecoderObserver& observer) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s", __FUNCTION__);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
- if (vie_channel->RegisterCodecObserver(&observer) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Could not register codec observer at channel",
- __FUNCTION__);
- shared_data_->SetLastError(kViECodecObserverAlreadyRegistered);
- return -1;
- }
- return 0;
-}
-
-int ViECodecImpl::DeregisterDecoderObserver(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id()), "%s",
- __FUNCTION__);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
- if (vie_channel->RegisterCodecObserver(NULL) != 0) {
- shared_data_->SetLastError(kViECodecObserverNotRegistered);
- return -1;
- }
- return 0;
-}
-
-int ViECodecImpl::SendKeyFrame(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(video_channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
- if (vie_encoder->SendKeyFrame() != 0) {
- shared_data_->SetLastError(kViECodecUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViECodecImpl::WaitForFirstKeyFrame(const int video_channel,
- const bool wait) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id()),
- "%s(video_channel: %d, wait: %d)", __FUNCTION__, video_channel,
- wait);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidChannelId);
- return -1;
- }
- if (vie_channel->WaitForKeyFrame(wait) != 0) {
- shared_data_->SetLastError(kViECodecUnknownError);
- return -1;
- }
- return 0;
-}
-
-bool ViECodecImpl::CodecValid(const VideoCodec& video_codec) {
- // Check pl_name matches codec_type.
- if (video_codec.codecType == kVideoCodecRED) {
-#if defined(WIN32)
- if (_strnicmp(video_codec.plName, "red", 3) == 0) {
-#else
- if (strncasecmp(video_codec.plName, "red", 3) == 0) {
-#endif
- // We only care about the type and name for red.
- return true;
- }
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Codec type doesn't match pl_name", video_codec.plType);
- return false;
- } else if (video_codec.codecType == kVideoCodecULPFEC) {
-#if defined(WIN32)
- if (_strnicmp(video_codec.plName, "ULPFEC", 6) == 0) {
-#else
- if (strncasecmp(video_codec.plName, "ULPFEC", 6) == 0) {
-#endif
- // We only care about the type and name for ULPFEC.
- return true;
- }
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Codec type doesn't match pl_name", video_codec.plType);
- return false;
- } else if ((video_codec.codecType == kVideoCodecVP8 &&
- strncmp(video_codec.plName, "VP8", 4) == 0) ||
- (video_codec.codecType == kVideoCodecI420 &&
- strncmp(video_codec.plName, "I420", 4) == 0)) {
- // OK.
- } else {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Codec type doesn't match pl_name", video_codec.plType);
- return false;
- }
-
- if (video_codec.plType == 0 && video_codec.plType > 127) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Invalid codec payload type: %d", video_codec.plType);
- return false;
- }
-
- if (video_codec.width > kViEMaxCodecWidth ||
- video_codec.height > kViEMaxCodecHeight) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Invalid codec size: %u x %u",
- video_codec.width, video_codec.height);
- return false;
- }
-
- if (video_codec.startBitrate < kViEMinCodecBitrate) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Invalid start_bitrate: %u",
- video_codec.startBitrate);
- return false;
- }
- if (video_codec.minBitrate < kViEMinCodecBitrate) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1, "Invalid min_bitrate: %u",
- video_codec.minBitrate);
- return false;
- }
- if (video_codec.numberOfSimulcastStreams == 1) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, -1,
- "Number of Simulcast streams can not be 1");
- return false;
- }
- return true;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_codec_impl.h b/trunk/src/video_engine/vie_codec_impl.h
deleted file mode 100644
index e0e5e94..0000000
--- a/trunk/src/video_engine/vie_codec_impl.h
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_CODEC_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_CODEC_IMPL_H_
-
-#include "typedefs.h"
-#include "video_engine/include/vie_codec.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_ref_count.h"
-
-namespace webrtc {
-
-class ViESharedData;
-
-class ViECodecImpl
- : public ViECodec,
- public ViERefCount {
- public:
- virtual int Release();
-
- // Implements ViECodec.
- virtual int NumberOfCodecs() const;
- virtual int GetCodec(const unsigned char list_number,
- VideoCodec& video_codec) const;
- virtual int SetSendCodec(const int video_channel,
- const VideoCodec& video_codec);
- virtual int GetSendCodec(const int video_channel,
- VideoCodec& video_codec) const;
- virtual int SetReceiveCodec(const int video_channel,
- const VideoCodec& video_codec);
- virtual int GetReceiveCodec(const int video_channel,
- VideoCodec& video_codec) const;
- virtual int GetCodecConfigParameters(
- const int video_channel,
- unsigned char config_parameters[kConfigParameterSize],
- unsigned char& config_parameters_size) const;
- virtual int SetImageScaleStatus(const int video_channel, const bool enable);
- virtual int GetSendCodecStastistics(const int video_channel,
- unsigned int& key_frames,
- unsigned int& delta_frames) const;
- virtual int GetReceiveCodecStastistics(const int video_channel,
- unsigned int& key_frames,
- unsigned int& delta_frames) const;
- virtual int GetCodecTargetBitrate(const int video_channel,
- unsigned int* bitrate) const;
- virtual unsigned int GetDiscardedPackets(const int video_channel) const;
- virtual int SetKeyFrameRequestCallbackStatus(const int video_channel,
- const bool enable);
- virtual int SetSignalKeyPacketLossStatus(const int video_channel,
- const bool enable,
- const bool only_key_frames = false);
- virtual int RegisterEncoderObserver(const int video_channel,
- ViEEncoderObserver& observer);
- virtual int DeregisterEncoderObserver(const int video_channel);
- virtual int RegisterDecoderObserver(const int video_channel,
- ViEDecoderObserver& observer);
- virtual int DeregisterDecoderObserver(const int video_channel);
- virtual int SendKeyFrame(const int video_channel);
- virtual int WaitForFirstKeyFrame(const int video_channel, const bool wait);
-
- protected:
- ViECodecImpl(ViESharedData* shared_data);
- virtual ~ViECodecImpl();
-
- private:
- bool CodecValid(const VideoCodec& video_codec);
-
- ViESharedData* shared_data_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_CODEC_IMPL_H_
diff --git a/trunk/src/video_engine/vie_defines.h b/trunk/src/video_engine/vie_defines.h
deleted file mode 100644
index 73c9008..0000000
--- a/trunk/src/video_engine/vie_defines.h
+++ /dev/null
@@ -1,261 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * vie_defines.h
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_DEFINES_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_DEFINES_H_
-
-#include "engine_configurations.h"
-
-#ifdef WEBRTC_MAC_INTEL
-#include <stdio.h>
-#include <unistd.h>
-#endif
-
-#ifdef WEBRTC_ANDROID
-#include <pthread.h>
-#include <sys/types.h>
-#include <sys/socket.h>
-#include <netinet/in.h>
-#include <arpa/inet.h>
-#include <linux/net.h>
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#include <time.h>
-#include <sys/time.h>
-#endif
-
-namespace webrtc
-{
-// ===================================================
-// ViE Defines
-// ===================================================
-
-// General
-enum { kViEMinKeyRequestIntervalMs = 300};
-
-// ViEBase
-enum { kViEMaxNumberOfChannels = 32};
-enum { kViEVersionMaxMessageSize = 1024 };
-enum { kViEMaxModuleVersionSize = 960 };
-
-// ViECapture
-enum { kViEMaxCaptureDevices=10};
-// Width used if no send codec has been set when a capture device is started
-enum { kViECaptureDefaultWidth = 352};
-// Height used if no send codec has been set when a capture device is started
-enum { kViECaptureDefaultHeight = 288};
-enum { kViECaptureDefaultFramerate = 30};
-enum { kViECaptureMaxSnapshotWaitTimeMs = 500 };
-
-// ViECodec
-enum { kViEMaxCodecWidth = 1920};
-enum { kViEMaxCodecHeight = 1200};
-enum { kViEMaxCodecFramerate = 60};
-enum { kViEMinCodecBitrate = 30};
-
-// ViEEncryption
-enum { kViEMaxSrtpKeyLength = 30};
-enum { kViEMinSrtpEncryptLength = 16};
-enum { kViEMaxSrtpEncryptLength = 256};
-enum { kViEMaxSrtpAuthSh1Length = 20};
-enum { kViEMaxSrtpTagAuthNullLength = 12};
-enum { kViEMaxSrtpKeyAuthNullLength = 256};
-
-// ViEExternalCodec
-
-// ViEFile
-enum { kViEMaxFilePlayers = 3};
-
-// ViEImageProcess
-
-// ViENetwork
-enum { kViEMaxMtu = 1500};
-enum { kViESocketThreads = 1};
-enum { kViENumReceiveSocketBuffers = 500};
-
-// ViERender
-// Max valid time set in SetRenderTimeoutImage
-enum { kViEMaxRenderTimeoutTimeMs = 10000};
-// Min valid time set in SetRenderTimeoutImage
-enum { kViEMinRenderTimeoutTimeMs = 33};
-enum { kViEDefaultRenderDelayMs = 10};
-
-// ViERTP_RTCP
-enum { kNackHistorySize = 400};
-
-// Id definitions
-enum {
- kViEChannelIdBase=0x0,
- kViEChannelIdMax=0xFF,
- kViECaptureIdBase=0x1001,
- kViECaptureIdMax=0x10FF,
- kViEFileIdBase=0x2000,
- kViEFileIdMax=0x200F,
- kViEDummyChannelId=0xFFFF
-};
-
-// Module id
-// Create a unique id based on the ViE instance id and the
-// channel id. ViE id > 0 and 0 <= channel id <= 255
-
-inline int ViEId(const int vieId, const int channelId = -1)
-{
- if (channelId == -1)
- {
- return (int) ((vieId << 16) + kViEDummyChannelId);
- }
- return (int) ((vieId << 16) + channelId);
-}
-
-inline int ViEModuleId(const int vieId, const int channelId = -1)
-{
- if (channelId == -1)
- {
- return (int) ((vieId << 16) + kViEDummyChannelId);
- }
- return (int) ((vieId << 16) + channelId);
-}
-
-inline int ChannelId(const int moduleId)
-{
- return (int) (moduleId & 0xffff);
-}
-
-
-// ============================================================================
-// Platform specifics
-// ============================================================================
-
-//-------------------------------------
-// Windows
-//-------------------------------------
-
-#if defined(_WIN32)
-// Build information macros
- #if defined(_DEBUG)
- #define BUILDMODE TEXT("d")
- #elif defined(DEBUG)
- #define BUILDMODE TEXT("d")
- #elif defined(NDEBUG)
- #define BUILDMODE TEXT("r")
- #else
- #define BUILDMODE TEXT("?")
- #endif
-
- #define BUILDTIME TEXT(__TIME__)
- #define BUILDDATE TEXT(__DATE__)
-
- // example: "Oct 10 2002 12:05:30 r"
- #define BUILDINFO BUILDDATE TEXT(" ") BUILDTIME TEXT(" ") BUILDMODE
-
-
- #define RENDER_MODULE_TYPE kRenderWindows
- // Warning pragmas
- // new behavior: elements of array 'XXX' will be default initialized
- #pragma warning(disable: 4351)
- // 'this' : used in base member initializer list
- #pragma warning(disable: 4355)
- // frame pointer register 'ebp' modified by inline assembly code
- #pragma warning(disable: 4731)
-
- // Include libraries
- #pragma comment( lib, "winmm.lib" )
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- #pragma comment( lib, "ws2_32.lib" )
- #pragma comment( lib, "Iphlpapi.lib" ) // _GetAdaptersAddresses
-#endif
-#endif
-
-
-//-------------------------------------
-// Mac
-//-------------------------------------
-
-#ifdef WEBRTC_MAC_INTEL
- #define SLEEP(x) usleep(x * 1000)
-
- // Build information macros
- #define TEXT(x) x
- #if defined(_DEBUG)
- #define BUILDMODE TEXT("d")
- #elif defined(DEBUG)
- #define BUILDMODE TEXT("d")
- #elif defined(NDEBUG)
- #define BUILDMODE TEXT("r")
- #else
- #define BUILDMODE TEXT("?")
- #endif
-
- #define BUILDTIME TEXT(__TIME__)
- #define BUILDDATE TEXT(__DATE__)
-
- // example: "Oct 10 2002 12:05:30 r"
- #define BUILDINFO BUILDDATE TEXT(" ") BUILDTIME TEXT(" ") BUILDMODE
-
- #define RENDER_MODULE_TYPE kRenderWindows
-#endif
-
-//-------------------------------------
-// Linux
-//-------------------------------------
-
-#ifndef WEBRTC_ANDROID
-#ifdef WEBRTC_LINUX
-
-// Build information macros
-#if defined(_DEBUG)
- #define BUILDMODE "d"
-#elif defined(DEBUG)
- #define BUILDMODE "d"
-#elif defined(NDEBUG)
- #define BUILDMODE "r"
-#else
- #define BUILDMODE "?"
-#endif
-
-#define BUILDTIME __TIME__
-#define BUILDDATE __DATE__
-
-// example: "Oct 10 2002 12:05:30 r"
-#define BUILDINFO BUILDDATE " " BUILDTIME " " BUILDMODE
-
-#endif // ifdef WEBRTC_LINUX
-#endif // ifndef WEBRTC_ANDROID
-
-#ifdef WEBRTC_ANDROID
-
- #define FAR
- #define __cdecl
-
- #if defined(_DEBUG)
- #define BUILDMODE "d"
- #elif defined(DEBUG)
- #define BUILDMODE "d"
- #elif defined(NDEBUG)
- #define BUILDMODE "r"
- #else
- #define BUILDMODE "?"
- #endif
-
- #define BUILDTIME __TIME__
- #define BUILDDATE __DATE__
-
- // example: "Oct 10 2002 12:05:30 r"
- #define BUILDINFO BUILDDATE " " BUILDTIME " " BUILDMODE
-
-#endif // #ifdef WEBRTC_ANDROID
-
-} //namespace webrtc
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_DEFINES_H_
diff --git a/trunk/src/video_engine/vie_encoder.cc b/trunk/src/video_engine/vie_encoder.cc
deleted file mode 100644
index cf3587b..0000000
--- a/trunk/src/video_engine/vie_encoder.cc
+++ /dev/null
@@ -1,887 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vie_encoder.h"
-
-#include <cassert>
-
-#include "critical_section_wrapper.h"
-#include "process_thread.h"
-#include "rtp_rtcp.h"
-#include "tick_util.h"
-#include "trace.h"
-#include "video_codec_interface.h"
-#include "video_coding.h"
-#include "video_coding_defines.h"
-#include "vie_codec.h"
-#include "vie_defines.h"
-#include "vie_image_process.h"
-
-namespace webrtc {
-
-class QMVideoSettingsCallback : public VCMQMSettingsCallback {
- public:
- QMVideoSettingsCallback(WebRtc_Word32 engine_id,
- WebRtc_Word32 channel_id,
- VideoProcessingModule* vpm,
- VideoCodingModule* vcm,
- WebRtc_Word32 num_of_cores,
- WebRtc_Word32 max_payload_length);
- ~QMVideoSettingsCallback();
-
- // Update VPM with QM (quality modes: frame size & frame rate) settings.
- WebRtc_Word32 SetVideoQMSettings(const WebRtc_UWord32 frame_rate,
- const WebRtc_UWord32 width,
- const WebRtc_UWord32 height);
-
- void SetMaxPayloadLength(WebRtc_Word32 max_payload_length);
-
- private:
- WebRtc_Word32 engine_id_;
- WebRtc_Word32 channel_id_;
- VideoProcessingModule* vpm_;
- VideoCodingModule* vcm_;
- WebRtc_Word32 num_cores_;
- WebRtc_Word32 max_payload_length_;
-};
-
-
-ViEEncoder::ViEEncoder(WebRtc_Word32 engine_id, WebRtc_Word32 channel_id,
- WebRtc_UWord32 number_of_cores,
- ProcessThread& module_process_thread)
- : engine_id_(engine_id),
- channel_id_(channel_id),
- number_of_cores_(number_of_cores),
- vcm_(*webrtc::VideoCodingModule::Create(ViEModuleId(engine_id,
- channel_id))),
- vpm_(*webrtc::VideoProcessingModule::Create(ViEModuleId(engine_id,
- channel_id))),
- default_rtp_rtcp_(*RtpRtcp::CreateRtpRtcp(
- ViEModuleId(engine_id, channel_id), false)),
- callback_cs_(CriticalSectionWrapper::CreateCriticalSection()),
- data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
- paused_(false),
- channels_dropping_delta_frames_(0),
- drop_next_frame_(false),
- fec_enabled_(false),
- nack_enabled_(false),
- codec_observer_(NULL),
- effect_filter_(NULL),
- module_process_thread_(module_process_thread),
- has_received_sli_(false),
- picture_id_sli_(0),
- has_received_rpsi_(false),
- picture_id_rpsi_(0),
- file_recorder_(channel_id) {
- WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideo,
- ViEId(engine_id, channel_id),
- "%s(engine_id: %d) 0x%p - Constructor", __FUNCTION__, engine_id,
- this);
- for (int i = 0; i < kMaxSimulcastStreams; i++) {
- time_last_intra_request_ms_[i] = 0;
- }
- vcm_.InitializeSender();
- vpm_.EnableTemporalDecimation(true);
-
- // Enable/disable content analysis: off by default for now.
- vpm_.EnableContentAnalysis(false);
-
- module_process_thread_.RegisterModule(&vcm_);
- if (default_rtp_rtcp_.InitSender() != 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "ViEEncoder: RTP::InitSender failure");
- assert(false);
- }
- default_rtp_rtcp_.RegisterIncomingVideoCallback(this);
- default_rtp_rtcp_.RegisterIncomingRTCPCallback(this);
- module_process_thread_.RegisterModule(&default_rtp_rtcp_);
-
- qm_callback_ = new QMVideoSettingsCallback(
- engine_id_,
- channel_id_,
- &vpm_,
- &vcm_,
- number_of_cores,
- default_rtp_rtcp_.MaxDataPayloadLength());
-
-#ifdef VIDEOCODEC_VP8
- VideoCodec video_codec;
- if (vcm_.Codec(webrtc::kVideoCodecVP8, &video_codec) == VCM_OK) {
- vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
- default_rtp_rtcp_.MaxDataPayloadLength());
- default_rtp_rtcp_.RegisterSendPayload(video_codec);
- } else {
- assert(false);
- }
-#else
- VideoCodec video_codec;
- if (vcm_.Codec(webrtc::kVideoCodecI420, &video_codec) == VCM_OK) {
- vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
- default_rtp_rtcp_.MaxDataPayloadLength());
- default_rtp_rtcp_.RegisterSendPayload(video_codec);
- } else {
- assert(false);
- }
-#endif
-
- if (vcm_.RegisterTransportCallback(this) != 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "ViEEncoder: VCM::RegisterTransportCallback failure");
- }
- if (vcm_.RegisterSendStatisticsCallback(this) != 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "ViEEncoder: VCM::RegisterSendStatisticsCallback failure");
- }
-
- if (vcm_.RegisterVideoQMCallback(qm_callback_) != 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "VCM::RegisterQMCallback failure");
- }
-}
-
-ViEEncoder::~ViEEncoder() {
- WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "ViEEncoder Destructor 0x%p, engine_id: %d", this, engine_id_);
-
- if (default_rtp_rtcp_.NumberChildModules() > 0) {
- assert(false);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "Channels still attached %d, leaking memory",
- default_rtp_rtcp_.NumberChildModules());
- return;
- }
- module_process_thread_.DeRegisterModule(&vcm_);
- module_process_thread_.DeRegisterModule(&vpm_);
- module_process_thread_.DeRegisterModule(&default_rtp_rtcp_);
- delete &vcm_;
- delete &vpm_;
- delete &default_rtp_rtcp_;
- delete qm_callback_;
-}
-
-void ViEEncoder::Pause() {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
- CriticalSectionScoped cs(data_cs_.get());
- paused_ = true;
-}
-
-void ViEEncoder::Restart() {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s", __FUNCTION__);
- CriticalSectionScoped cs(data_cs_.get());
- paused_ = false;
-}
-
-WebRtc_Word32 ViEEncoder::DropDeltaAfterKey(bool enable) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s(%d)", __FUNCTION__, enable);
- CriticalSectionScoped cs(data_cs_.get());
-
- if (enable) {
- channels_dropping_delta_frames_++;
- } else {
- channels_dropping_delta_frames_--;
- if (channels_dropping_delta_frames_ < 0) {
- channels_dropping_delta_frames_ = 0;
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: Called too many times", __FUNCTION__);
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_UWord8 ViEEncoder::NumberOfCodecs() {
- return vcm_.NumberOfCodecs();
-}
-
-WebRtc_Word32 ViEEncoder::GetCodec(WebRtc_UWord8 list_index,
- webrtc::VideoCodec& video_codec) {
- if (vcm_.Codec(list_index, &video_codec) != 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s: Could not get codec",
- __FUNCTION__);
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 ViEEncoder::RegisterExternalEncoder(webrtc::VideoEncoder* encoder,
- WebRtc_UWord8 pl_type) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s: pltype %u", __FUNCTION__,
- pl_type);
-
- if (encoder == NULL)
- return -1;
-
- if (vcm_.RegisterExternalEncoder(encoder, pl_type) != VCM_OK) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "Could not register external encoder");
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 ViEEncoder::DeRegisterExternalEncoder(WebRtc_UWord8 pl_type) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: pltype %u", __FUNCTION__, pl_type);
-
- webrtc::VideoCodec current_send_codec;
- if (vcm_.SendCodec(¤t_send_codec) == VCM_OK) {
- if (vcm_.Bitrate(¤t_send_codec.startBitrate) != 0) {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "Failed to get the current encoder target bitrate.");
- }
- }
-
- if (vcm_.RegisterExternalEncoder(NULL, pl_type) != VCM_OK) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "Could not deregister external encoder");
- return -1;
- }
-
- // If the external encoder is the current send codeci, use vcm internal
- // encoder.
- if (current_send_codec.plType == pl_type) {
- WebRtc_UWord16 max_data_payload_length =
- default_rtp_rtcp_.MaxDataPayloadLength();
- if (vcm_.RegisterSendCodec(¤t_send_codec, number_of_cores_,
- max_data_payload_length) != VCM_OK) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "Could not use internal encoder");
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_Word32 ViEEncoder::SetEncoder(const webrtc::VideoCodec& video_codec) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: CodecType: %d, width: %u, height: %u", __FUNCTION__,
- video_codec.codecType, video_codec.width, video_codec.height);
-
- // Convert from kbps to bps.
- default_rtp_rtcp_.SetSendBitrate(video_codec.startBitrate * 1000,
- video_codec.minBitrate,
- video_codec.maxBitrate);
-
- // Setting target width and height for VPM.
- if (vpm_.SetTargetResolution(video_codec.width, video_codec.height,
- video_codec.maxFramerate) != VPM_OK) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "Could not set VPM target dimensions");
- return -1;
- }
-
- if (default_rtp_rtcp_.RegisterSendPayload(video_codec) != 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "Could register RTP module video payload");
- return -1;
- }
-
- WebRtc_UWord16 max_data_payload_length =
- default_rtp_rtcp_.MaxDataPayloadLength();
-
- qm_callback_->SetMaxPayloadLength(max_data_payload_length);
-
- if (vcm_.RegisterSendCodec(&video_codec, number_of_cores_,
- max_data_payload_length) != VCM_OK) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "Could not register send codec");
- return -1;
- }
-
- data_cs_->Enter();
- memcpy(&send_codec_, &video_codec, sizeof(send_codec_));
- data_cs_->Leave();
-
- // Set this module as sending right away, let the slave module in the channel
- // start and stop sending.
- if (default_rtp_rtcp_.Sending() == false) {
- if (default_rtp_rtcp_.SetSendingStatus(true) != 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "Could start RTP module sending");
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_Word32 ViEEncoder::GetEncoder(webrtc::VideoCodec& video_codec) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s", __FUNCTION__);
-
- if (vcm_.SendCodec(&video_codec) != 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "Could not get VCM send codec");
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 ViEEncoder::GetCodecConfigParameters(
- unsigned char config_parameters[kConfigParameterSize],
- unsigned char& config_parameters_size) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s", __FUNCTION__);
-
- WebRtc_Word32 num_parameters =
- vcm_.CodecConfigParameters(config_parameters, kConfigParameterSize);
- if (num_parameters <= 0) {
- config_parameters_size = 0;
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "Could not get config parameters");
- return -1;
- }
- config_parameters_size = static_cast<unsigned char>(num_parameters);
- return 0;
-}
-
-WebRtc_Word32 ViEEncoder::ScaleInputImage(bool enable) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s(enable %d)", __FUNCTION__,
- enable);
-
- VideoFrameResampling resampling_mode = kFastRescaling;
- if (enable == true) {
- // kInterpolation is currently not supported.
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s not supported",
- __FUNCTION__, enable);
- return -1;
- }
- vpm_.SetInputFrameResampleMode(resampling_mode);
-
- return 0;
-}
-
-RtpRtcp* ViEEncoder::SendRtpRtcpModule() {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s", __FUNCTION__);
-
- return &default_rtp_rtcp_;
-}
-
-void ViEEncoder::DeliverFrame(int id, webrtc::VideoFrame& video_frame,
- int num_csrcs,
- const WebRtc_UWord32 CSRC[kRtpCsrcSize]) {
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s: %llu", __FUNCTION__,
- video_frame.TimeStamp());
-
- {
- CriticalSectionScoped cs(data_cs_.get());
- if (paused_ || default_rtp_rtcp_.SendingMedia() == false) {
- // We've paused or we have no channels attached, don't encode.
- return;
- }
- if (drop_next_frame_) {
- // Drop this frame.
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: Dropping frame %llu after a key fame", __FUNCTION__,
- video_frame.TimeStamp());
- drop_next_frame_ = false;
- return;
- }
- }
-
- // Convert render time, in ms, to RTP timestamp.
- const WebRtc_UWord32 time_stamp =
- 90 * static_cast<WebRtc_UWord32>(video_frame.RenderTimeMs());
- video_frame.SetTimeStamp(time_stamp);
- {
- CriticalSectionScoped cs(callback_cs_.get());
- if (effect_filter_) {
- effect_filter_->Transform(video_frame.Length(), video_frame.Buffer(),
- video_frame.TimeStamp(),
- video_frame.Width(), video_frame.Height());
- }
- }
- // Record raw frame.
- file_recorder_.RecordVideoFrame(video_frame);
-
- // Make sure the CSRC list is correct.
- if (num_csrcs > 0) {
- WebRtc_UWord32 tempCSRC[kRtpCsrcSize];
- for (int i = 0; i < num_csrcs; i++) {
- if (CSRC[i] == 1) {
- tempCSRC[i] = default_rtp_rtcp_.SSRC();
- } else {
- tempCSRC[i] = CSRC[i];
- }
- }
- default_rtp_rtcp_.SetCSRCs(tempCSRC, (WebRtc_UWord8) num_csrcs);
- }
-
-#ifdef VIDEOCODEC_VP8
- if (vcm_.SendCodec() == webrtc::kVideoCodecVP8) {
- webrtc::CodecSpecificInfo codec_specific_info;
- codec_specific_info.codecType = webrtc::kVideoCodecVP8;
- if (has_received_sli_ || has_received_rpsi_) {
- {
- codec_specific_info.codecSpecific.VP8.hasReceivedRPSI =
- has_received_rpsi_;
- codec_specific_info.codecSpecific.VP8.hasReceivedSLI =
- has_received_sli_;
- codec_specific_info.codecSpecific.VP8.pictureIdRPSI =
- picture_id_rpsi_;
- codec_specific_info.codecSpecific.VP8.pictureIdSLI =
- picture_id_sli_;
- }
- has_received_sli_ = false;
- has_received_rpsi_ = false;
- }
- VideoFrame* decimated_frame = NULL;
- const int ret = vpm_.PreprocessFrame(&video_frame, &decimated_frame);
- if (ret == 1) {
- // Drop this frame.
- return;
- } else if (ret != VPM_OK) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: Error preprocessing frame %u", __FUNCTION__,
- video_frame.TimeStamp());
- return;
- }
-
- VideoContentMetrics* content_metrics = NULL;
- content_metrics = vpm_.ContentMetrics();
-
- // Frame was not re-sampled => use original.
- if (decimated_frame == NULL) {
- decimated_frame = &video_frame;
- }
-
- if (vcm_.AddVideoFrame(*decimated_frame, content_metrics,
- &codec_specific_info) != VCM_OK) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: Error encoding frame %u", __FUNCTION__,
- video_frame.TimeStamp());
- }
- return;
- }
-#endif
- // TODO(mflodman) Rewrite this to use code common to VP8 case.
- // Pass frame via preprocessor.
- VideoFrame* decimated_frame = NULL;
- const int ret = vpm_.PreprocessFrame(&video_frame, &decimated_frame);
- if (ret == 1) {
- // Drop this frame.
- return;
- } else if (ret != VPM_OK) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: Error preprocessing frame %u", __FUNCTION__,
- video_frame.TimeStamp());
- return;
- }
-
- // Frame was not sampled => use original.
- if (decimated_frame == NULL) {
- decimated_frame = &video_frame;
- }
- if (vcm_.AddVideoFrame(*decimated_frame) != VCM_OK) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s: Error encoding frame %u",
- __FUNCTION__, video_frame.TimeStamp());
- }
-}
-
-void ViEEncoder::DelayChanged(int id, int frame_delay) {
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s: %u", __FUNCTION__,
- frame_delay);
-
- default_rtp_rtcp_.SetCameraDelay(frame_delay);
- file_recorder_.SetFrameDelay(frame_delay);
-}
-
-int ViEEncoder::GetPreferedFrameSettings(int& width,
- int& height,
- int& frame_rate) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s", __FUNCTION__);
-
- webrtc::VideoCodec video_codec;
- memset(&video_codec, 0, sizeof(video_codec));
- if (vcm_.SendCodec(&video_codec) != VCM_OK) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "Could not get VCM send codec");
- return -1;
- }
-
- width = video_codec.width;
- height = video_codec.height;
- frame_rate = video_codec.maxFramerate;
- return 0;
-}
-
-WebRtc_Word32 ViEEncoder::SendKeyFrame() {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s", __FUNCTION__);
- return vcm_.FrameTypeRequest(kVideoFrameKey, 0); // Simulcast idx = 0.
-}
-
-WebRtc_Word32 ViEEncoder::SendCodecStatistics(
- WebRtc_UWord32& num_key_frames, WebRtc_UWord32& num_delta_frames) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s", __FUNCTION__);
-
- webrtc::VCMFrameCount sent_frames;
- if (vcm_.SentFrameCount(sent_frames) != VCM_OK) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: Could not get sent frame information", __FUNCTION__);
- return -1;
- }
- num_key_frames = sent_frames.numKeyFrames;
- num_delta_frames = sent_frames.numDeltaFrames;
- return 0;
-}
-
-WebRtc_Word32 ViEEncoder::EstimatedSendBandwidth(
- WebRtc_UWord32* available_bandwidth) const {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- return default_rtp_rtcp_.EstimatedSendBandwidth(available_bandwidth);
-}
-
-int ViEEncoder::CodecTargetBitrate(WebRtc_UWord32* bitrate) const {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, channel_id_), "%s",
- __FUNCTION__);
- if (vcm_.Bitrate(bitrate) != 0)
- return -1;
- return 0;
-}
-
-WebRtc_Word32 ViEEncoder::UpdateProtectionMethod() {
- bool fec_enabled = false;
- WebRtc_UWord8 dummy_ptype_red = 0;
- WebRtc_UWord8 dummy_ptypeFEC = 0;
-
- // Updated protection method to VCM to get correct packetization sizes.
- // FEC has larger overhead than NACK -> set FEC if used.
- WebRtc_Word32 error = default_rtp_rtcp_.GenericFECStatus(fec_enabled,
- dummy_ptype_red,
- dummy_ptypeFEC);
- if (error) {
- return -1;
- }
-
- bool nack_enabled = (default_rtp_rtcp_.NACK() == kNackOff) ? false : true;
- if (fec_enabled_ == fec_enabled && nack_enabled_ == nack_enabled) {
- // No change needed, we're already in correct state.
- return 0;
- }
- fec_enabled_ = fec_enabled;
- nack_enabled_ = nack_enabled;
-
- // Set Video Protection for VCM.
- if (fec_enabled && nack_enabled) {
- vcm_.SetVideoProtection(webrtc::kProtectionNackFEC, true);
- } else {
- vcm_.SetVideoProtection(webrtc::kProtectionFEC, fec_enabled_);
- vcm_.SetVideoProtection(webrtc::kProtectionNack, nack_enabled_);
- vcm_.SetVideoProtection(webrtc::kProtectionNackFEC, false);
- }
-
- if (fec_enabled || nack_enabled) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s: FEC status ",
- __FUNCTION__, fec_enabled);
- vcm_.RegisterProtectionCallback(this);
- // The send codec must be registered to set correct MTU.
- webrtc::VideoCodec codec;
- if (vcm_.SendCodec(&codec) == 0) {
- WebRtc_UWord16 max_pay_load = default_rtp_rtcp_.MaxDataPayloadLength();
- if (vcm_.Bitrate(&codec.startBitrate) != 0) {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "Failed to get the current encoder target bitrate.");
- }
- if (vcm_.RegisterSendCodec(&codec, number_of_cores_, max_pay_load) != 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: Failed to update Sendcodec when enabling FEC",
- __FUNCTION__, fec_enabled);
- return -1;
- }
- }
- return 0;
- } else {
- // FEC and NACK are disabled.
- vcm_.RegisterProtectionCallback(NULL);
- }
- return 0;
-}
-
-WebRtc_Word32 ViEEncoder::SendData(
- const FrameType frame_type,
- const WebRtc_UWord8 payload_type,
- const WebRtc_UWord32 time_stamp,
- const WebRtc_UWord8* payload_data,
- const WebRtc_UWord32 payload_size,
- const webrtc::RTPFragmentationHeader& fragmentation_header,
- const RTPVideoHeader* rtp_video_hdr) {
- {
- CriticalSectionScoped cs(data_cs_.get());
- if (paused_) {
- // Paused, don't send this packet.
- return 0;
- }
- if (channels_dropping_delta_frames_ &&
- frame_type == webrtc::kVideoFrameKey) {
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: Sending key frame, drop next frame", __FUNCTION__);
- drop_next_frame_ = true;
- }
- }
-
- // New encoded data, hand over to the rtp module.
- return default_rtp_rtcp_.SendOutgoingData(frame_type, payload_type,
- time_stamp, payload_data,
- payload_size, &fragmentation_header,
- rtp_video_hdr);
-}
-
-WebRtc_Word32 ViEEncoder::ProtectionRequest(
- WebRtc_UWord8 delta_fecrate,
- WebRtc_UWord8 key_fecrate,
- bool delta_use_uep_protection,
- bool key_use_uep_protection,
- bool nack_enabled,
- WebRtc_UWord32* sent_video_rate_bps,
- WebRtc_UWord32* sent_nack_rate_bps,
- WebRtc_UWord32* sent_fec_rate_bps) {
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s, deltaFECRate: %u, key_fecrate: %u, "
- "delta_use_uep_protection: %d, key_use_uep_protection: %d, ",
- __FUNCTION__, delta_fecrate, key_fecrate,
- delta_use_uep_protection, key_use_uep_protection);
-
- if (default_rtp_rtcp_.SetFECCodeRate(key_fecrate, delta_fecrate) != 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: Could not update FEC code rate", __FUNCTION__);
- }
- if (default_rtp_rtcp_.SetFECUepProtection(key_use_uep_protection,
- delta_use_uep_protection) != 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: Could not update FEC-UEP protection", __FUNCTION__);
- }
- default_rtp_rtcp_.BitrateSent(NULL,
- sent_video_rate_bps,
- sent_fec_rate_bps,
- sent_nack_rate_bps);
- return 0;
-}
-
-WebRtc_Word32 ViEEncoder::SendStatistics(const WebRtc_UWord32 bit_rate,
- const WebRtc_UWord32 frame_rate) {
- CriticalSectionScoped cs(callback_cs_.get());
- if (codec_observer_) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s: bitrate %u, framerate %u",
- __FUNCTION__, bit_rate, frame_rate);
- codec_observer_->OutgoingRate(channel_id_, frame_rate, bit_rate);
- }
- return 0;
-}
-
-WebRtc_Word32 ViEEncoder::RegisterCodecObserver(ViEEncoderObserver* observer) {
- CriticalSectionScoped cs(callback_cs_.get());
- if (observer) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s: observer added",
- __FUNCTION__);
- if (codec_observer_) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s: observer already set.",
- __FUNCTION__);
- return -1;
- }
- codec_observer_ = observer;
- } else {
- if (codec_observer_ == NULL) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: observer does not exist.", __FUNCTION__);
- return -1;
- }
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s: observer removed",
- __FUNCTION__);
- codec_observer_ = NULL;
- }
- return 0;
-}
-
-void ViEEncoder::OnSLIReceived(const WebRtc_Word32 id,
- const WebRtc_UWord8 picture_id) {
- picture_id_sli_ = picture_id;
- has_received_sli_ = true;
-}
-
-void ViEEncoder::OnRPSIReceived(const WebRtc_Word32 id,
- const WebRtc_UWord64 picture_id) {
- picture_id_rpsi_ = picture_id;
- has_received_rpsi_ = true;
-}
-
-void ViEEncoder::OnReceivedIntraFrameRequest(const WebRtc_Word32 id,
- const FrameType type,
- const WebRtc_UWord8 stream_idx) {
- assert(stream_idx < kMaxSimulcastStreams);
-
- // Key frame request from remote side, signal to VCM.
- WEBRTC_TRACE(webrtc::kTraceStateInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s", __FUNCTION__);
-
- WebRtc_Word64 now = TickTime::MillisecondTimestamp();
- if (time_last_intra_request_ms_[stream_idx] + kViEMinKeyRequestIntervalMs >
- now) {
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: Not not encoding new intra due to timing", __FUNCTION__);
- return;
- }
- vcm_.FrameTypeRequest(type, stream_idx);
- time_last_intra_request_ms_[stream_idx] = now;
-}
-
-void ViEEncoder::OnNetworkChanged(const WebRtc_Word32 id,
- const WebRtc_UWord32 bitrate_bps,
- const WebRtc_UWord8 fraction_lost,
- const WebRtc_UWord16 round_trip_time_ms) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s(bitrate_bps: %u, fraction_lost: %u, rtt_ms: %u",
- __FUNCTION__, bitrate_bps, fraction_lost, round_trip_time_ms);
-
- vcm_.SetChannelParameters(bitrate_bps / 1000, fraction_lost,
- round_trip_time_ms);
-}
-
-WebRtc_Word32 ViEEncoder::RegisterEffectFilter(ViEEffectFilter* effect_filter) {
- CriticalSectionScoped cs(callback_cs_.get());
- if (effect_filter == NULL) {
- if (effect_filter_ == NULL) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s: no effect filter added",
- __FUNCTION__);
- return -1;
- }
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s: deregister effect filter",
- __FUNCTION__);
- } else {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "%s: register effect",
- __FUNCTION__);
- if (effect_filter_) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "%s: effect filter already added ", __FUNCTION__);
- return -1;
- }
- }
- effect_filter_ = effect_filter;
- return 0;
-}
-
-ViEFileRecorder& ViEEncoder::GetOutgoingFileRecorder() {
- return file_recorder_;
-}
-
-QMVideoSettingsCallback::QMVideoSettingsCallback(
- WebRtc_Word32 engine_id,
- WebRtc_Word32 channel_id,
- VideoProcessingModule* vpm,
- VideoCodingModule* vcm,
- WebRtc_Word32 num_cores,
- WebRtc_Word32 max_payload_length)
- : engine_id_(engine_id),
- channel_id_(channel_id),
- vpm_(vpm),
- vcm_(vcm),
- num_cores_(num_cores),
- max_payload_length_(max_payload_length) {
-}
-
-QMVideoSettingsCallback::~QMVideoSettingsCallback() {
-}
-
-WebRtc_Word32 QMVideoSettingsCallback::SetVideoQMSettings(
- const WebRtc_UWord32 frame_rate,
- const WebRtc_UWord32 width,
- const WebRtc_UWord32 height) {
- WebRtc_Word32 ret_val = 0;
- ret_val = vpm_->SetTargetResolution(width, height, frame_rate);
-
- if (!ret_val) {
- // Get current settings.
- VideoCodec current_codec;
- vcm_->SendCodec(¤t_codec);
- WebRtc_UWord32 current_bit_rate;
- if (vcm_->Bitrate(¤t_bit_rate) != 0) {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "Failed to get the current encoder target bitrate.");
- }
-
- // Set the new values.
- current_codec.height = static_cast<WebRtc_UWord16>(height);
- current_codec.width = static_cast<WebRtc_UWord16>(width);
- current_codec.maxFramerate = static_cast<WebRtc_UWord8>(frame_rate);
- current_codec.startBitrate = current_bit_rate;
-
- // Re-register encoder with the updated settings.
- ret_val = vcm_->RegisterSendCodec(¤t_codec, num_cores_,
- max_payload_length_);
- }
- return ret_val;
-}
-
-void QMVideoSettingsCallback::SetMaxPayloadLength(
- WebRtc_Word32 max_payload_length) {
- max_payload_length_ = max_payload_length;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_encoder.h b/trunk/src/video_engine/vie_encoder.h
deleted file mode 100644
index 1b20491..0000000
--- a/trunk/src/video_engine/vie_encoder.h
+++ /dev/null
@@ -1,182 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_ENCODER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_ENCODER_H_
-
-#include "common_types.h"
-#include "rtp_rtcp_defines.h"
-#include "typedefs.h"
-#include "video_coding_defines.h"
-#include "video_processing.h"
-#include "vie_defines.h"
-#include "vie_file_recorder.h"
-#include "vie_frame_provider_base.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class ProcessThread;
-class QMVideoSettingsCallback;
-class RtpRtcp;
-class VideoCodingModule;
-class ViEEffectFilter;
-class ViEEncoderObserver;
-
-class ViEEncoder
- : public RtpVideoFeedback,
- public RtcpFeedback,
- public VCMPacketizationCallback,
- public VCMProtectionCallback,
- public VCMSendStatisticsCallback,
- public ViEFrameCallback {
- public:
- ViEEncoder(WebRtc_Word32 engine_id,
- WebRtc_Word32 channel_id,
- WebRtc_UWord32 number_of_cores,
- ProcessThread& module_process_thread);
- ~ViEEncoder();
-
- // Drops incoming packets before they get to the encoder.
- void Pause();
- void Restart();
-
- WebRtc_Word32 DropDeltaAfterKey(bool enable);
-
- // Codec settings.
- WebRtc_UWord8 NumberOfCodecs();
- WebRtc_Word32 GetCodec(WebRtc_UWord8 list_index, VideoCodec& video_codec);
- WebRtc_Word32 RegisterExternalEncoder(VideoEncoder* encoder,
- WebRtc_UWord8 pl_type);
- WebRtc_Word32 DeRegisterExternalEncoder(WebRtc_UWord8 pl_type);
- WebRtc_Word32 SetEncoder(const VideoCodec& video_codec);
- WebRtc_Word32 GetEncoder(VideoCodec& video_codec);
-
- WebRtc_Word32 GetCodecConfigParameters(
- unsigned char config_parameters[kConfigParameterSize],
- unsigned char& config_parameters_size);
-
- // Scale or crop/pad image.
- WebRtc_Word32 ScaleInputImage(bool enable);
-
- // RTP settings.
- RtpRtcp* SendRtpRtcpModule();
-
- // Implementing ViEFrameCallback.
- virtual void DeliverFrame(int id,
- VideoFrame& video_frame,
- int num_csrcs = 0,
- const WebRtc_UWord32 CSRC[kRtpCsrcSize] = NULL);
- virtual void DelayChanged(int id, int frame_delay);
- virtual int GetPreferedFrameSettings(int& width,
- int& height,
- int& frame_rate);
-
- virtual void ProviderDestroyed(int id) {
- return;
- }
-
- WebRtc_Word32 EncodeFrame(VideoFrame& video_frame);
- WebRtc_Word32 SendKeyFrame();
- WebRtc_Word32 SendCodecStatistics(WebRtc_UWord32& num_key_frames,
- WebRtc_UWord32& num_delta_frames);
- WebRtc_Word32 EstimatedSendBandwidth(
- WebRtc_UWord32* available_bandwidth) const;
- int CodecTargetBitrate(WebRtc_UWord32* bitrate) const;
- // Loss protection.
- WebRtc_Word32 UpdateProtectionMethod();
-
- // Implements VCMPacketizationCallback.
- virtual WebRtc_Word32 SendData(
- const FrameType frame_type,
- const WebRtc_UWord8 payload_type,
- const WebRtc_UWord32 time_stamp,
- const WebRtc_UWord8* payload_data,
- const WebRtc_UWord32 payload_size,
- const RTPFragmentationHeader& fragmentation_header,
- const RTPVideoHeader* rtp_video_hdr);
-
- // Implements VideoProtectionCallback.
- virtual WebRtc_Word32 ProtectionRequest(
- WebRtc_UWord8 delta_fecrate,
- WebRtc_UWord8 key_fecrate,
- bool delta_use_uep_protection,
- bool key_use_uep_protection,
- bool nack_enabled,
- WebRtc_UWord32* sent_video_rate_bps,
- WebRtc_UWord32* sent_nack_rate_bps,
- WebRtc_UWord32* sent_fec_rate_bps);
-
- // Implements VideoSendStatisticsCallback.
- virtual WebRtc_Word32 SendStatistics(const WebRtc_UWord32 bit_rate,
- const WebRtc_UWord32 frame_rate);
- WebRtc_Word32 RegisterCodecObserver(ViEEncoderObserver* observer);
-
- // Implements RtcpFeedback.
- virtual void OnSLIReceived(const WebRtc_Word32 id,
- const WebRtc_UWord8 picture_id);
- virtual void OnRPSIReceived(const WebRtc_Word32 id,
- const WebRtc_UWord64 picture_id);
-
- // Implements RtpVideoFeedback.
- virtual void OnReceivedIntraFrameRequest(const WebRtc_Word32 id,
- const FrameType type,
- const WebRtc_UWord8 stream_idx);
-
- virtual void OnNetworkChanged(const WebRtc_Word32 id,
- const WebRtc_UWord32 bitrate_bps,
- const WebRtc_UWord8 fraction_lost,
- const WebRtc_UWord16 round_trip_time_ms);
-
- // Effect filter.
- WebRtc_Word32 RegisterEffectFilter(ViEEffectFilter* effect_filter);
-
- // Recording.
- ViEFileRecorder& GetOutgoingFileRecorder();
-
- private:
- WebRtc_Word32 engine_id_;
- WebRtc_Word32 channel_id_;
- const WebRtc_UWord32 number_of_cores_;
-
- VideoCodingModule& vcm_;
- VideoProcessingModule& vpm_;
- RtpRtcp& default_rtp_rtcp_;
- scoped_ptr<CriticalSectionWrapper> callback_cs_;
- scoped_ptr<CriticalSectionWrapper> data_cs_;
- VideoCodec send_codec_;
-
- bool paused_;
- WebRtc_Word64 time_last_intra_request_ms_[kMaxSimulcastStreams];
- WebRtc_Word32 channels_dropping_delta_frames_;
- bool drop_next_frame_;
-
- bool fec_enabled_;
- bool nack_enabled_;
-
- ViEEncoderObserver* codec_observer_;
- ViEEffectFilter* effect_filter_;
- ProcessThread& module_process_thread_;
-
- bool has_received_sli_;
- WebRtc_UWord8 picture_id_sli_;
- bool has_received_rpsi_;
- WebRtc_UWord64 picture_id_rpsi_;
-
- ViEFileRecorder file_recorder_;
-
- // Quality modes callback
- QMVideoSettingsCallback* qm_callback_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_ENCODER_H_
diff --git a/trunk/src/video_engine/vie_encryption_impl.cc b/trunk/src/video_engine/vie_encryption_impl.cc
deleted file mode 100644
index adbda42..0000000
--- a/trunk/src/video_engine/vie_encryption_impl.cc
+++ /dev/null
@@ -1,111 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_encryption_impl.h"
-
-#include "system_wrappers/interface/trace.h"
-#include "video_engine/include/vie_errors.h"
-#include "video_engine/vie_channel.h"
-#include "video_engine/vie_channel_manager.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_impl.h"
-#include "video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-ViEEncryption* ViEEncryption::GetInterface(VideoEngine* video_engine) {
-#ifdef WEBRTC_VIDEO_ENGINE_ENCRYPTION_API
- if (video_engine == NULL) {
- return NULL;
- }
- VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
- ViEEncryptionImpl* vie_encryption_impl = vie_impl;
- // Increase ref count.
- (*vie_encryption_impl)++;
- return vie_encryption_impl;
-#else
- return NULL;
-#endif
-}
-
-int ViEEncryptionImpl::Release() {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
- "ViEEncryptionImpl::Release()");
- // Decrease ref count.
- (*this)--;
-
- WebRtc_Word32 ref_count = GetCount();
- if (ref_count < 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
- "ViEEncryptionImpl release too many times");
- shared_data_->SetLastError(kViEAPIDoesNotExist);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
- "ViEEncryptionImpl reference count: %d", ref_count);
- return ref_count;
-}
-
-ViEEncryptionImpl::ViEEncryptionImpl(ViESharedData* shared_data)
- : shared_data_(shared_data) {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViEEncryptionImpl::ViEEncryptionImpl() Ctor");
-}
-
-ViEEncryptionImpl::~ViEEncryptionImpl() {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViEEncryptionImpl::~ViEEncryptionImpl() Dtor");
-}
-
-int ViEEncryptionImpl::RegisterExternalEncryption(const int video_channel,
- Encryption& encryption) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "RegisterExternalEncryption(video_channel=%d)", video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (vie_channel == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEEncryptionInvalidChannelId);
- return -1;
- }
- if (vie_channel->RegisterExternalEncryption(&encryption) != 0) {
- shared_data_->SetLastError(kViEEncryptionUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViEEncryptionImpl::DeregisterExternalEncryption(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "RegisterExternalEncryption(video_channel=%d)", video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (vie_channel == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEEncryptionInvalidChannelId);
- return -1;
- }
-
- if (vie_channel->DeRegisterExternalEncryption() != 0) {
- shared_data_->SetLastError(kViEEncryptionUnknownError);
- return -1;
- }
- return 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_encryption_impl.h b/trunk/src/video_engine/vie_encryption_impl.h
deleted file mode 100644
index f398461..0000000
--- a/trunk/src/video_engine/vie_encryption_impl.h
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_ENCRYPTION_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_ENCRYPTION_IMPL_H_
-
-#include "typedefs.h"
-#include "video_engine/include/vie_encryption.h"
-#include "video_engine/vie_ref_count.h"
-
-namespace webrtc {
-
-class ViESharedData;
-
-class ViEEncryptionImpl
- : public ViEEncryption,
- public ViERefCount {
- public:
- virtual int Release();
-
- // Implements ViEEncryption.
- virtual int RegisterExternalEncryption(const int video_channel,
- Encryption& encryption);
- virtual int DeregisterExternalEncryption(const int video_channel);
-
- protected:
- ViEEncryptionImpl(ViESharedData* shared_data);
- virtual ~ViEEncryptionImpl();
-
- private:
- ViESharedData* shared_data_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_ENCRYPTION_IMPL_H_
diff --git a/trunk/src/video_engine/vie_external_codec_impl.cc b/trunk/src/video_engine/vie_external_codec_impl.cc
deleted file mode 100644
index 92e7dd6..0000000
--- a/trunk/src/video_engine/vie_external_codec_impl.cc
+++ /dev/null
@@ -1,184 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_external_codec_impl.h"
-
-#include "engine_configurations.h"
-#include "system_wrappers/interface/trace.h"
-#include "video_engine/include/vie_errors.h"
-#include "video_engine/vie_channel.h"
-#include "video_engine/vie_channel_manager.h"
-#include "video_engine/vie_encoder.h"
-#include "video_engine/vie_impl.h"
-#include "video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-ViEExternalCodec* ViEExternalCodec::GetInterface(VideoEngine* video_engine) {
-#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
- if (video_engine == NULL) {
- return NULL;
- }
- VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
- ViEExternalCodecImpl* vie_external_codec_impl = vie_impl;
- // Increase ref count.
- (*vie_external_codec_impl)++;
- return vie_external_codec_impl;
-#else
- return NULL;
-#endif
-}
-
-int ViEExternalCodecImpl::Release() {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
- "ViEExternalCodec::Release()");
- // Decrease ref count.
- (*this)--;
-
- WebRtc_Word32 ref_count = GetCount();
- if (ref_count < 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
- "ViEExternalCodec release too many times");
- shared_data_->SetLastError(kViEAPIDoesNotExist);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
- "ViEExternalCodec reference count: %d", ref_count);
- return ref_count;
-}
-
-ViEExternalCodecImpl::ViEExternalCodecImpl(ViESharedData* shared_data)
- : shared_data_(shared_data) {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViEExternalCodecImpl::ViEExternalCodecImpl() Ctor");
-}
-
-ViEExternalCodecImpl::~ViEExternalCodecImpl() {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViEExternalCodecImpl::~ViEExternalCodecImpl() Dtor");
-}
-
-int ViEExternalCodecImpl::RegisterExternalSendCodec(const int video_channel,
- const unsigned char pl_type,
- VideoEncoder* encoder) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s channel %d pl_type %d encoder 0x%x", __FUNCTION__,
- video_channel, pl_type, encoder);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Invalid argument video_channel %u. Does it exist?",
- __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidArgument);
- return -1;
- }
- if (!encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Invalid argument Encoder 0x%x.", __FUNCTION__, encoder);
- shared_data_->SetLastError(kViECodecInvalidArgument);
- return -1;
- }
-
- if (vie_encoder->RegisterExternalEncoder(encoder, pl_type) != 0) {
- shared_data_->SetLastError(kViECodecUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViEExternalCodecImpl::DeRegisterExternalSendCodec(
- const int video_channel, const unsigned char pl_type) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s channel %d pl_type %d", __FUNCTION__, video_channel,
- pl_type);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Invalid argument video_channel %u. Does it exist?",
- __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidArgument);
- return -1;
- }
-
- if (vie_encoder->DeRegisterExternalEncoder(pl_type) != 0) {
- shared_data_->SetLastError(kViECodecUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViEExternalCodecImpl::RegisterExternalReceiveCodec(
- const int video_channel,
- const unsigned int pl_type,
- VideoDecoder* decoder,
- bool decoder_render,
- int render_delay) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s channel %d pl_type %d decoder 0x%x, decoder_render %d, "
- "renderDelay %d", __FUNCTION__, video_channel, pl_type, decoder,
- decoder_render, render_delay);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Invalid argument video_channel %u. Does it exist?",
- __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidArgument);
- return -1;
- }
- if (!decoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Invalid argument decoder 0x%x.", __FUNCTION__, decoder);
- shared_data_->SetLastError(kViECodecInvalidArgument);
- return -1;
- }
-
- if (vie_channel->RegisterExternalDecoder(pl_type, decoder, decoder_render,
- render_delay) != 0) {
- shared_data_->SetLastError(kViECodecUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViEExternalCodecImpl::DeRegisterExternalReceiveCodec(
-const int video_channel, const unsigned char pl_type) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s channel %d pl_type %u", __FUNCTION__, video_channel,
- pl_type);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Invalid argument video_channel %u. Does it exist?",
- __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViECodecInvalidArgument);
- return -1;
- }
- if (vie_channel->DeRegisterExternalDecoder(pl_type) != 0) {
- shared_data_->SetLastError(kViECodecUnknownError);
- return -1;
- }
- return 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_external_codec_impl.h b/trunk/src/video_engine/vie_external_codec_impl.h
deleted file mode 100644
index 67d6a78..0000000
--- a/trunk/src/video_engine/vie_external_codec_impl.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_EXTERNAL_CODEC_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_EXTERNAL_CODEC_IMPL_H_
-
-#include "video_engine/include/vie_external_codec.h"
-#include "video_engine/vie_ref_count.h"
-
-namespace webrtc {
-
-class ViESharedData;
-
-class ViEExternalCodecImpl
- : public ViEExternalCodec,
- public ViERefCount {
- public:
- // Implements ViEExternalCodec.
- virtual int Release();
- virtual int RegisterExternalSendCodec(const int video_channel,
- const unsigned char pl_type,
- VideoEncoder* encoder);
- virtual int DeRegisterExternalSendCodec(const int video_channel,
- const unsigned char pl_type);
- virtual int RegisterExternalReceiveCodec(const int video_channel,
- const unsigned int pl_type,
- VideoDecoder* decoder,
- bool decoder_render = false,
- int render_delay = 0);
- virtual int DeRegisterExternalReceiveCodec(const int video_channel,
- const unsigned char pl_type);
-
- protected:
- ViEExternalCodecImpl(ViESharedData* shared_data);
- virtual ~ViEExternalCodecImpl();
-
- private:
- ViESharedData* shared_data_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_EXTERNAL_CODEC_IMPL_H_
diff --git a/trunk/src/video_engine/vie_file_image.cc b/trunk/src/video_engine/vie_file_image.cc
deleted file mode 100644
index 2472a3a..0000000
--- a/trunk/src/video_engine/vie_file_image.cc
+++ /dev/null
@@ -1,125 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Placed first to get WEBRTC_VIDEO_ENGINE_FILE_API.
-#include "engine_configurations.h"
-
-#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
-
-#include "video_engine/vie_file_image.h"
-
-#include <stdio.h>
-
-#include "common_video/interface/video_image.h"
-#include "common_video/jpeg/include/jpeg.h"
-#include "system_wrappers/interface/trace.h"
-
-namespace webrtc {
-
-int ViEFileImage::ConvertJPEGToVideoFrame(int engine_id,
- const char* file_nameUTF8,
- VideoFrame& video_frame) {
- // Read jpeg file into temporary buffer.
- EncodedImage image_buffer;
-
- FILE* image_file = fopen(file_nameUTF8, "rb");
- if (!image_file) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
- "%s could not open file %s", __FUNCTION__, file_nameUTF8);
- return -1;
- }
- if (fseek(image_file, 0, SEEK_END) != 0) {
- fclose(image_file);
- WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
- "ConvertJPEGToVideoFrame fseek SEEK_END error for file %s",
- file_nameUTF8);
- return -1;
- }
- int buffer_size = ftell(image_file);
- if (buffer_size == -1) {
- fclose(image_file);
- WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
- "ConvertJPEGToVideoFrame could tell file size for file %s",
- file_nameUTF8);
- return -1;
- }
- image_buffer._size = buffer_size;
- if (fseek(image_file, 0, SEEK_SET) != 0) {
- fclose(image_file);
- WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
- "ConvertJPEGToVideoFrame fseek SEEK_SET error for file %s",
- file_nameUTF8);
- return -1;
- }
- image_buffer._buffer = new WebRtc_UWord8[ image_buffer._size + 1];
- if (image_buffer._size != fread(image_buffer._buffer, sizeof(WebRtc_UWord8),
- image_buffer._size, image_file)) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
- "%s could not read file %s", __FUNCTION__, file_nameUTF8);
- fclose(image_file);
- delete [] image_buffer._buffer;
- return -1;
- }
- fclose(image_file);
-
- JpegDecoder decoder;
- RawImage decoded_image;
- int ret = decoder.Decode(image_buffer, decoded_image);
-
- delete [] image_buffer._buffer;
- image_buffer._buffer = NULL;
-
- if (ret == -1) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
- "%s could decode file %s from jpeg format", __FUNCTION__,
- file_nameUTF8);
- return -1;
- } else if (ret == -3) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
- "%s could not convert jpeg's data to i420 format",
- __FUNCTION__, file_nameUTF8);
- }
-
- // Image length in I420.
- WebRtc_UWord32 image_length = (WebRtc_UWord32)(decoded_image._width *
- decoded_image._height * 1.5);
- if (-1 == video_frame.Swap(decoded_image._buffer, image_length,
- image_length)) {
- WEBRTC_TRACE(kTraceDebug, kTraceVideo, engine_id,
- "%s could not copy frame image_decoded_buffer to video_frame ",
- __FUNCTION__, file_nameUTF8);
- return -1;
- }
-
- if (decoded_image._buffer) {
- delete [] decoded_image._buffer;
- decoded_image._buffer = NULL;
- }
-
- video_frame.SetWidth(decoded_image._width);
- video_frame.SetHeight(decoded_image._height);
- return 0;
-}
-
-int ViEFileImage::ConvertPictureToVideoFrame(int engine_id,
- const ViEPicture& picture,
- VideoFrame& video_frame) {
- WebRtc_UWord32 picture_length = (WebRtc_UWord32)(picture.width *
- picture.height * 1.5);
- video_frame.CopyFrame(picture_length, picture.data);
- video_frame.SetWidth(picture.width);
- video_frame.SetHeight(picture.height);
- video_frame.SetLength(picture_length);
- return 0;
-}
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_FILE_API
diff --git a/trunk/src/video_engine/vie_file_image.h b/trunk/src/video_engine/vie_file_image.h
deleted file mode 100644
index e79b115..0000000
--- a/trunk/src/video_engine/vie_file_image.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_FILE_IMAGE_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_FILE_IMAGE_H_
-
-#include "modules/interface/module_common_types.h"
-#include "typedefs.h"
-#include "video_engine/include/vie_file.h"
-
-namespace webrtc {
-
-class ViEFileImage {
- public:
- static int ConvertJPEGToVideoFrame(int engine_id,
- const char* file_nameUTF8,
- VideoFrame& video_frame);
- static int ConvertPictureToVideoFrame(int engine_id,
- const ViEPicture& picture,
- VideoFrame& video_frame);
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_FILE_IMAGE_H_
diff --git a/trunk/src/video_engine/vie_file_impl.cc b/trunk/src/video_engine/vie_file_impl.cc
deleted file mode 100644
index 28f9854..0000000
--- a/trunk/src/video_engine/vie_file_impl.cc
+++ /dev/null
@@ -1,1016 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "engine_configurations.h"
-
-#include "video_engine/vie_file_impl.h"
-
-#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
-#include "common_video/jpeg/include/jpeg.h"
-#include "system_wrappers/interface/condition_variable_wrapper.h"
-#include "system_wrappers/interface/critical_section_wrapper.h"
-#include "system_wrappers/interface/trace.h"
-#include "video_engine/include/vie_errors.h"
-#include "video_engine/vie_capturer.h"
-#include "video_engine/vie_channel.h"
-#include "video_engine/vie_channel_manager.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_encoder.h"
-#include "video_engine/vie_file_image.h"
-#include "video_engine/vie_file_player.h"
-#include "video_engine/vie_file_recorder.h"
-#include "video_engine/vie_impl.h"
-#include "video_engine/vie_input_manager.h"
-#include "video_engine/vie_render_manager.h"
-#include "video_engine/vie_renderer.h"
-#endif
-
-namespace webrtc {
-
-ViEFile* ViEFile::GetInterface(VideoEngine* video_engine) {
-#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
- if (!video_engine) {
- return NULL;
- }
- VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
- ViEFileImpl* vie_file_impl = vie_impl;
- // Increase ref count.
- (*vie_file_impl)++;
- return vie_file_impl;
-#else
- return NULL;
-#endif
-}
-
-#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
-
-int ViEFileImpl::Release() {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
- "ViEFile::Release()");
- // Decrease ref count.
- (*this)--;
- WebRtc_Word32 ref_count = GetCount();
- if (ref_count < 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
- "ViEFile release too many times");
- shared_data_->SetLastError(kViEAPIDoesNotExist);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
- "ViEFile reference count: %d", ref_count);
- return ref_count;
-}
-
-ViEFileImpl::ViEFileImpl(ViESharedData* shared_data)
- : shared_data_(shared_data) {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViEFileImpl::ViEFileImpl() Ctor");
-}
-
-ViEFileImpl::~ViEFileImpl() {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViEFileImpl::~ViEFileImpl() Dtor");
-}
-
-int ViEFileImpl::StartPlayFile(const char* file_nameUTF8,
- int& file_id,
- const bool loop,
- const FileFormats file_format) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s", __FUNCTION__);
-
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
-
- VoiceEngine* voice = shared_data_->channel_manager()->GetVoiceEngine();
- const WebRtc_Word32 result = shared_data_->input_manager()->CreateFilePlayer(
- file_nameUTF8, loop, file_format, voice, file_id);
- if (result != 0) {
- shared_data_->SetLastError(result);
- return -1;
- }
- return 0;
-}
-
-int ViEFileImpl::StopPlayFile(const int file_id) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(file_id: %d)", __FUNCTION__, file_id);
- {
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViEFilePlayer* vie_file_player = is.FilePlayer(file_id);
- if (!vie_file_player) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: File with id %d is not playing.", __FUNCTION__,
- file_id);
- shared_data_->SetLastError(kViEFileNotPlaying);
- return -1;
- }
- }
- // Destroy the capture device.
- return shared_data_->input_manager()->DestroyFilePlayer(file_id);
-}
-
-int ViEFileImpl::RegisterObserver(int file_id, ViEFileObserver& observer) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(file_id: %d)", __FUNCTION__, file_id);
-
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViEFilePlayer* vie_file_player = is.FilePlayer(file_id);
- if (!vie_file_player) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: File with id %d is not playing.", __FUNCTION__,
- file_id);
- shared_data_->SetLastError(kViEFileNotPlaying);
- return -1;
- }
- if (vie_file_player->IsObserverRegistered()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), file_id),
- "%s: Observer already registered", __FUNCTION__);
- shared_data_->SetLastError(kViEFileObserverAlreadyRegistered);
- return -1;
- }
- if (vie_file_player->RegisterObserver(observer) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), file_id),
- "%s: Failed to register observer", __FUNCTION__, file_id);
- shared_data_->SetLastError(kViEFileUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViEFileImpl::DeregisterObserver(int file_id, ViEFileObserver& observer) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(file_id: %d)", __FUNCTION__, file_id);
-
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViEFilePlayer* vie_file_player = is.FilePlayer(file_id);
- if (!vie_file_player) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: File with id %d is not playing.", __FUNCTION__,
- file_id);
- shared_data_->SetLastError(kViEFileNotPlaying);
- return -1;
- }
- if (!vie_file_player->IsObserverRegistered()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), file_id),
- "%s: No Observer registered", __FUNCTION__);
- shared_data_->SetLastError(kViEFileObserverNotRegistered);
- return -1;
- }
- if (vie_file_player->DeRegisterObserver() != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), file_id),
- "%s: Failed to deregister observer", __FUNCTION__, file_id);
- shared_data_->SetLastError(kViEFileUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViEFileImpl::SendFileOnChannel(const int file_id, const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(file_id: %d)", __FUNCTION__, file_id);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEFileInvalidChannelId);
- return -1;
- }
-
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- if (is.FrameProvider(vie_encoder) != NULL) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d already connected to a capture device or "
- "file.", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEFileInputAlreadyConnected);
- return -1;
- }
-
- ViEFilePlayer* vie_file_player = is.FilePlayer(file_id);
- if (!vie_file_player) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: File with id %d is not playing.", __FUNCTION__,
- file_id);
- shared_data_->SetLastError(kViEFileNotPlaying);
- return -1;
- }
-
- if (vie_file_player->RegisterFrameCallback(video_channel, vie_encoder)
- != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Failed to register frame callback.", __FUNCTION__,
- file_id);
- shared_data_->SetLastError(kViEFileUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViEFileImpl::StopSendFileOnChannel(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(video_channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEFileInvalidChannelId);
- return -1;
- }
-
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViEFrameProviderBase* frame_provider = is.FrameProvider(vie_encoder);
- if (!frame_provider ||
- frame_provider->Id() < kViEFileIdBase ||
- frame_provider->Id() > kViEFileIdMax) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: No file connected to Channel %d", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViEFileNotConnected);
- return -1;
- }
- if (frame_provider->DeregisterFrameCallback(vie_encoder) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Failed to deregister file from channel %d",
- __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEFileUnknownError);
- }
- return 0;
-}
-
-int ViEFileImpl::StartPlayFileAsMicrophone(const int file_id,
- const int audio_channel,
- bool mix_microphone,
- float volume_scaling) {
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
-
- ViEFilePlayer* vie_file_player = is.FilePlayer(file_id);
- if (!vie_file_player) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: File with id %d is not playing.", __FUNCTION__,
- file_id);
- shared_data_->SetLastError(kViEFileNotPlaying);
- return -1;
- }
- if (vie_file_player->SendAudioOnChannel(audio_channel, mix_microphone,
- volume_scaling) != 0) {
- shared_data_->SetLastError(kViEFileVoEFailure);
- return -1;
- }
- return 0;
-}
-
-int ViEFileImpl::StopPlayFileAsMicrophone(const int file_id,
- const int audio_channel) {
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
-
- ViEFilePlayer* vie_file_player = is.FilePlayer(file_id);
- if (!vie_file_player) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: File with id %d is not playing.", __FUNCTION__,
- file_id);
- shared_data_->SetLastError(kViEFileNotPlaying);
- return -1;
- }
-
- if (vie_file_player->StopSendAudioOnChannel(audio_channel) != 0) {
- shared_data_->SetLastError(kViEFileVoEFailure);
- return -1;
- }
- return 0;
-}
-
-int ViEFileImpl::StartPlayAudioLocally(const int file_id,
- const int audio_channel,
- float volume_scaling) {
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
-
- ViEFilePlayer* vie_file_player = is.FilePlayer(file_id);
- if (!vie_file_player) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: File with id %d is not playing.", __FUNCTION__,
- file_id);
- shared_data_->SetLastError(kViEFileNotPlaying);
- return -1;
- }
- if (vie_file_player->PlayAudioLocally(audio_channel, volume_scaling) != 0) {
- shared_data_->SetLastError(kViEFileVoEFailure);
- return -1;
- }
- return 0;
-}
-
-int ViEFileImpl::StopPlayAudioLocally(const int file_id,
- const int audio_channel) {
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
-
- ViEFilePlayer* vie_file_player = is.FilePlayer(file_id);
- if (!vie_file_player) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: File with id %d is not playing.", __FUNCTION__,
- file_id);
- shared_data_->SetLastError(kViEFileNotPlaying);
- return -1;
- }
- if (vie_file_player->StopPlayAudioLocally(audio_channel) != 0) {
- shared_data_->SetLastError(kViEFileVoEFailure);
- return -1;
- }
- return 0;
-}
-
-int ViEFileImpl::StartRecordOutgoingVideo(const int video_channel,
- const char* file_nameUTF8,
- AudioSource audio_source,
- const CodecInst& audio_codec,
- const VideoCodec& video_codec,
- const FileFormats file_format) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s video_channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEFileInvalidChannelId);
- return -1;
- }
- ViEFileRecorder& file_recorder = vie_encoder->GetOutgoingFileRecorder();
- if (file_recorder.RecordingStarted()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Already recording outgoing video on channel %d",
- __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEFileAlreadyRecording);
- return -1;
- }
-
- WebRtc_Word32 ve_channel_id = -1;
- VoiceEngine* ve_ptr = NULL;
- if (audio_source != NO_AUDIO) {
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- // Channel should exists since we have a ViEEncoder above.
- assert(false);
- return -1;
- }
- ve_channel_id = vie_channel->VoiceChannel();
- ve_ptr = shared_data_->channel_manager()->GetVoiceEngine();
- if (!ve_ptr) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Can't access voice engine. Have SetVoiceEngine "
- "been called?", __FUNCTION__);
- shared_data_->SetLastError(kViEFileVoENotSet);
- return -1;
- }
- }
- if (file_recorder.StartRecording(file_nameUTF8, video_codec, audio_source,
- ve_channel_id, audio_codec, ve_ptr,
- file_format) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Failed to start recording. Check arguments.",
- __FUNCTION__);
- shared_data_->SetLastError(kViEFileUnknownError);
- return -1;
- }
-
- return 0;
-}
-
-int ViEFileImpl::StopRecordOutgoingVideo(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s video_channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEFileInvalidChannelId);
- return -1;
- }
- ViEFileRecorder& file_recorder = vie_encoder->GetOutgoingFileRecorder();
- if (!file_recorder.RecordingStarted()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d is not recording.", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViEFileNotRecording);
- return -1;
- }
- if (file_recorder.StopRecording() != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Failed to stop recording of channel %d.", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViEFileUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViEFileImpl::StopRecordIncomingVideo(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s video_channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEFileInvalidChannelId);
- return -1;
- }
- ViEFileRecorder& file_recorder = vie_channel->GetIncomingFileRecorder();
- if (!file_recorder.RecordingStarted()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d is not recording.", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViEFileNotRecording);
- vie_channel->ReleaseIncomingFileRecorder();
- return -1;
- }
- if (file_recorder.StopRecording() != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Failed to stop recording of channel %d.",
- __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEFileUnknownError);
- vie_channel->ReleaseIncomingFileRecorder();
- return -1;
- }
- // Let the channel know we are no longer recording.
- vie_channel->ReleaseIncomingFileRecorder();
- return 0;
-}
-
-int ViEFileImpl::StartRecordIncomingVideo(const int video_channel,
- const char* file_nameUTF8,
- AudioSource audio_source,
- const CodecInst& audio_codec,
- const VideoCodec& video_codec,
- const FileFormats file_format) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s video_channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViEFileInvalidChannelId);
- return -1;
- }
- ViEFileRecorder& file_recorder = vie_channel->GetIncomingFileRecorder();
- if (file_recorder.RecordingStarted()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Already recording outgoing video on channel %d",
- __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEFileAlreadyRecording);
- return -1;
- }
-
- WebRtc_Word32 ve_channel_id = -1;
- VoiceEngine* ve_ptr = NULL;
- if (audio_source != NO_AUDIO) {
- ve_channel_id = vie_channel->VoiceChannel();
- ve_ptr = shared_data_->channel_manager()->GetVoiceEngine();
-
- if (!ve_ptr) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Can't access voice engine. Have SetVoiceEngine "
- "been called?", __FUNCTION__);
- shared_data_->SetLastError(kViEFileVoENotSet);
- return -1;
- }
- }
- if (file_recorder.StartRecording(file_nameUTF8, video_codec, audio_source,
- ve_channel_id, audio_codec, ve_ptr,
- file_format) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Failed to start recording. Check arguments.",
- __FUNCTION__);
- shared_data_->SetLastError(kViEFileUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViEFileImpl::GetFileInformation(const char* file_name,
- VideoCodec& video_codec,
- CodecInst& audio_codec,
- const FileFormats file_format) {
- return ViEFilePlayer::GetFileInformation(
- shared_data_->instance_id(),
- static_cast<const WebRtc_Word8*>(file_name),
- video_codec, audio_codec, file_format);
-}
-
-int ViEFileImpl::GetRenderSnapshot(const int video_channel,
- const char* file_nameUTF8) {
- // Gain access to the renderer for the specified channel and get it's
- // current frame.
- ViERenderManagerScoped rs(*(shared_data_->render_manager()));
- ViERenderer* renderer = rs.Renderer(video_channel);
- if (!renderer) {
- return -1;
- }
-
- VideoFrame video_frame;
- if (renderer->GetLastRenderedFrame(video_channel, video_frame) == -1) {
- return -1;
- }
-
- // JPEGEncoder writes the jpeg file for you (no control over it) and does
- // not return you the buffer. Thus, we are not going to be writing to the
- // disk here.
- JpegEncoder jpeg_encoder;
- RawImage input_image;
- if (jpeg_encoder.SetFileName(file_nameUTF8) == -1) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, shared_data_->instance_id(),
- "\tCould not open output file '%s' for writing!",
- file_nameUTF8);
- return -1;
- }
-
- input_image._width = video_frame.Width();
- input_image._height = video_frame.Height();
- video_frame.Swap(input_image._buffer, input_image._length,
- input_image._size);
-
- if (jpeg_encoder.Encode(input_image) == -1) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, shared_data_->instance_id(),
- "\tCould not encode i420 -> jpeg file '%s' for writing!",
- file_nameUTF8);
- if (input_image._buffer) {
- delete [] input_image._buffer;
- input_image._buffer = NULL;
- }
- return -1;
- }
- delete [] input_image._buffer;
- input_image._buffer = NULL;
- return 0;
-}
-
-int ViEFileImpl::GetRenderSnapshot(const int video_channel,
- ViEPicture& picture) {
- // Gain access to the renderer for the specified channel and get it's
- // current frame.
- ViERenderManagerScoped rs(*(shared_data_->render_manager()));
- ViERenderer* renderer = rs.Renderer(video_channel);
- if (!renderer) {
- return -1;
- }
-
- VideoFrame video_frame;
- if (renderer->GetLastRenderedFrame(video_channel, video_frame) == -1) {
- return -1;
- }
-
- // Copy from VideoFrame class to ViEPicture struct.
- int buffer_length =
- static_cast<int>(video_frame.Width() * video_frame.Height() * 1.5);
- picture.data = static_cast<WebRtc_UWord8*>(malloc(
- buffer_length * sizeof(WebRtc_UWord8)));
- memcpy(picture.data, video_frame.Buffer(), buffer_length);
- picture.size = buffer_length;
- picture.width = video_frame.Width();
- picture.height = video_frame.Height();
- picture.type = kVideoI420;
- return 0;
-}
-
-int ViEFileImpl::GetCaptureDeviceSnapshot(const int capture_id,
- const char* file_nameUTF8) {
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViECapturer* capturer = is.Capture(capture_id);
- if (!capturer) {
- return -1;
- }
-
- VideoFrame video_frame;
- if (GetNextCapturedFrame(capture_id, video_frame) == -1) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, shared_data_->instance_id(),
- "Could not gain acces to capture device %d video frame "
- "%s:%d", capture_id, __FUNCTION__);
- return -1;
- }
-
- // JPEGEncoder writes the jpeg file for you (no control over it) and does
- // not return you the buffer Thusly, we are not going to be writing to the
- // disk here.
- JpegEncoder jpeg_encoder;
- RawImage input_image;
- input_image._width = video_frame.Width();
- input_image._height = video_frame.Height();
- video_frame.Swap(input_image._buffer, input_image._length,
- input_image._size);
-
- if (jpeg_encoder.SetFileName(file_nameUTF8) == -1) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, shared_data_->instance_id(),
- "\tCould not open output file '%s' for writing!",
- file_nameUTF8);
-
- if (input_image._buffer) {
- delete [] input_image._buffer;
- }
- return -1;
- }
- if (jpeg_encoder.Encode(input_image) == -1) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, shared_data_->instance_id(),
- "\tCould not encode i420 -> jpeg file '%s' for "
- "writing!", file_nameUTF8);
- if (input_image._buffer) {
- delete [] input_image._buffer;
- }
- return -1;
- }
- delete [] input_image._buffer;
- input_image._buffer = NULL;
- return 0;
-}
-
-int ViEFileImpl::GetCaptureDeviceSnapshot(const int capture_id,
- ViEPicture& picture) {
- VideoFrame video_frame;
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViECapturer* capturer = is.Capture(capture_id);
- if (!capturer) {
- return -1;
- }
- if (GetNextCapturedFrame(capture_id, video_frame) == -1) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, shared_data_->instance_id(),
- "Could not gain acces to capture device %d video frame "
- "%s:%d", capture_id, __FUNCTION__);
- return -1;
- }
-
- // Copy from VideoFrame class to ViEPicture struct.
- int buffer_length =
- static_cast<int>(video_frame.Width() * video_frame.Height() * 1.5);
- picture.data = static_cast<WebRtc_UWord8*>(malloc(
- buffer_length * sizeof(WebRtc_UWord8)));
- memcpy(picture.data, video_frame.Buffer(), buffer_length);
- picture.size = buffer_length;
- picture.width = video_frame.Width();
- picture.height = video_frame.Height();
- picture.type = kVideoI420;
- return 0;
-}
-
-int ViEFileImpl::FreePicture(ViEPicture& picture) {
- if (picture.data) {
- free(picture.data);
- }
-
- picture.data = NULL;
- picture.size = 0;
- picture.width = 0;
- picture.height = 0;
- picture.type = kVideoUnknown;
- return 0;
-}
-int ViEFileImpl::SetCaptureDeviceImage(const int capture_id,
- const char* file_nameUTF8) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
- "%s(capture_id: %d)", __FUNCTION__, capture_id);
-
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViECapturer* capturer = is.Capture(capture_id);
- if (!capturer) {
- shared_data_->SetLastError(kViEFileInvalidCaptureId);
- return -1;
- }
-
- VideoFrame capture_image;
- if (ViEFileImage::ConvertJPEGToVideoFrame(
- ViEId(shared_data_->instance_id(), capture_id), file_nameUTF8,
- capture_image) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), capture_id),
- "%s(capture_id: %d) Failed to open file.", __FUNCTION__,
- capture_id);
- shared_data_->SetLastError(kViEFileInvalidFile);
- return -1;
- }
- if (capturer->SetCaptureDeviceImage(capture_image)) {
- shared_data_->SetLastError(kViEFileSetCaptureImageError);
- return -1;
- }
- return 0;
-}
-
-int ViEFileImpl::SetCaptureDeviceImage(const int capture_id,
-const ViEPicture& picture) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
- "%s(capture_id: %d)", __FUNCTION__, capture_id);
-
- if (picture.type != kVideoI420) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), capture_id),
- "%s(capture_id: %d) Not a valid picture type.",
- __FUNCTION__, capture_id);
- shared_data_->SetLastError(kViEFileInvalidArgument);
- return -1;
- }
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViECapturer* capturer = is.Capture(capture_id);
- if (!capturer) {
- shared_data_->SetLastError(kViEFileSetCaptureImageError);
- return -1;
- }
-
- VideoFrame capture_image;
- if (ViEFileImage::ConvertPictureToVideoFrame(
- ViEId(shared_data_->instance_id(), capture_id), picture, capture_image) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), capture_id),
- "%s(capture_id: %d) Failed to use picture.", __FUNCTION__,
- capture_id);
- shared_data_->SetLastError(kViEFileInvalidFile);
- return -1;
- }
- if (capturer->SetCaptureDeviceImage(capture_image)) {
- shared_data_->SetLastError(kViEFileInvalidCapture);
- return -1;
- }
- return 0;
-}
-
-int ViEFileImpl::SetRenderStartImage(const int video_channel,
-const char* file_nameUTF8) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d)", __FUNCTION__, video_channel);
-
- ViERenderManagerScoped rs(*(shared_data_->render_manager()));
- ViERenderer* renderer = rs.Renderer(video_channel);
- if (!renderer) {
- shared_data_->SetLastError(kViEFileInvalidRenderId);
- return -1;
- }
-
- VideoFrame start_image;
- if (ViEFileImage::ConvertJPEGToVideoFrame(
- ViEId(shared_data_->instance_id(), video_channel), file_nameUTF8, start_image) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d) Failed to open file.", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViEFileInvalidFile);
- return -1;
- }
- if (renderer->SetRenderStartImage(start_image) != 0) {
- shared_data_->SetLastError(kViEFileSetStartImageError);
- return -1;
- }
- return 0;
-}
-
-int ViEFileImpl::SetRenderStartImage(const int video_channel,
- const ViEPicture& picture) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d)", __FUNCTION__, video_channel);
- if (picture.type != kVideoI420) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d) Not a valid picture type.",
- __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEFileInvalidArgument);
- return -1;
- }
-
- ViERenderManagerScoped rs(*(shared_data_->render_manager()));
- ViERenderer* renderer = rs.Renderer(video_channel);
- if (!renderer) {
- shared_data_->SetLastError(kViEFileInvalidRenderId);
- return -1;
- }
-
- VideoFrame start_image;
- if (ViEFileImage::ConvertPictureToVideoFrame(
- ViEId(shared_data_->instance_id(), video_channel), picture, start_image) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d) Failed to use picture.",
- __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEFileInvalidCapture);
- return -1;
- }
- if (renderer->SetRenderStartImage(start_image) != 0) {
- shared_data_->SetLastError(kViEFileSetStartImageError);
- return -1;
- }
- return 0;
-}
-int ViEFileImpl::SetRenderTimeoutImage(const int video_channel,
- const char* file_nameUTF8,
- const unsigned int timeout_ms) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d)", __FUNCTION__, video_channel);
-
- ViERenderManagerScoped rs(*(shared_data_->render_manager()));
- ViERenderer* renderer = rs.Renderer(video_channel);
- if (!renderer) {
- shared_data_->SetLastError(kViEFileInvalidRenderId);
- return -1;
- }
- VideoFrame timeout_image;
- if (ViEFileImage::ConvertJPEGToVideoFrame(
- ViEId(shared_data_->instance_id(), video_channel), file_nameUTF8,
- timeout_image) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d) Failed to open file.", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViEFileInvalidFile);
- return -1;
- }
- WebRtc_Word32 timeout_time = timeout_ms;
- if (timeout_ms < kViEMinRenderTimeoutTimeMs) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d) Invalid timeout_ms, using %d.",
- __FUNCTION__, video_channel, kViEMinRenderTimeoutTimeMs);
- timeout_time = kViEMinRenderTimeoutTimeMs;
- }
- if (timeout_ms > kViEMaxRenderTimeoutTimeMs) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d) Invalid timeout_ms, using %d.",
- __FUNCTION__, video_channel, kViEMaxRenderTimeoutTimeMs);
- timeout_time = kViEMaxRenderTimeoutTimeMs;
- }
- if (renderer->SetTimeoutImage(timeout_image, timeout_time) != 0) {
- shared_data_->SetLastError(kViEFileSetRenderTimeoutError);
- return -1;
- }
- return 0;
-}
-
-int ViEFileImpl::SetRenderTimeoutImage(const int video_channel,
- const ViEPicture& picture,
-const unsigned int timeout_ms) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d)", __FUNCTION__, video_channel);
-
- if (picture.type != kVideoI420) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d) Not a valid picture type.",
- __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEFileInvalidArgument);
- return -1;
- }
-
- ViERenderManagerScoped rs(*(shared_data_->render_manager()));
- ViERenderer* renderer = rs.Renderer(video_channel);
- if (!renderer) {
- shared_data_->SetLastError(kViEFileSetRenderTimeoutError);
- return -1;
- }
- VideoFrame timeout_image;
- if (ViEFileImage::ConvertPictureToVideoFrame(
- ViEId(shared_data_->instance_id(), video_channel), picture,
- timeout_image) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d) Failed to use picture.",
- __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEFileInvalidCapture);
- return -1;
- }
- WebRtc_Word32 timeout_time = timeout_ms;
- if (timeout_ms < kViEMinRenderTimeoutTimeMs) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d) Invalid timeout_ms, using %d.",
- __FUNCTION__, video_channel, kViEMinRenderTimeoutTimeMs);
- timeout_time = kViEMinRenderTimeoutTimeMs;
- }
- if (timeout_ms > kViEMaxRenderTimeoutTimeMs) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(video_channel: %d) Invalid timeout_ms, using %d.",
- __FUNCTION__, video_channel, kViEMaxRenderTimeoutTimeMs);
- timeout_time = kViEMaxRenderTimeoutTimeMs;
- }
- if (renderer->SetTimeoutImage(timeout_image, timeout_time) != 0) {
- shared_data_->SetLastError(kViEFileSetRenderTimeoutError);
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 ViEFileImpl::GetNextCapturedFrame(WebRtc_Word32 capture_id,
-VideoFrame& video_frame) {
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViECapturer* capturer = is.Capture(capture_id);
- if (!capturer) {
- return -1;
- }
-
- ViECaptureSnapshot* snap_shot = new ViECaptureSnapshot();
- capturer->RegisterFrameCallback(-1, snap_shot);
- bool snapshot_taken = snap_shot->GetSnapshot(
- video_frame, kViECaptureMaxSnapshotWaitTimeMs);
-
- // Check once again if it has been destroyed.
- capturer->DeregisterFrameCallback(snap_shot);
- delete snap_shot;
- snap_shot = NULL;
-
- if (snapshot_taken) {
- return 0;
- }
- return -1;
-}
-
-ViECaptureSnapshot::ViECaptureSnapshot()
- : crit_(CriticalSectionWrapper::CreateCriticalSection()),
- condition_varaible_(ConditionVariableWrapper::CreateConditionVariable()),
- video_frame_(NULL) {
-}
-
-ViECaptureSnapshot::~ViECaptureSnapshot() {
- if (video_frame_) {
- delete video_frame_;
- video_frame_ = NULL;
- }
-}
-
-bool ViECaptureSnapshot::GetSnapshot(VideoFrame& video_frame,
- unsigned int max_wait_time) {
- crit_->Enter();
- video_frame_ = new VideoFrame();
- if (condition_varaible_->SleepCS(*(crit_.get()), max_wait_time)) {
- // Snapshot taken.
- video_frame.SwapFrame(*video_frame_);
- delete video_frame_;
- video_frame_ = NULL;
- crit_->Leave();
- return true;
- }
- crit_->Leave();
- return false;
-}
-
-void ViECaptureSnapshot::DeliverFrame(int id, VideoFrame& video_frame,
- int num_csrcs,
-const WebRtc_UWord32 CSRC[kRtpCsrcSize]) {
- CriticalSectionScoped cs(crit_.get());
- if (!video_frame_) {
- return;
- }
- video_frame_->SwapFrame(video_frame);
- condition_varaible_->WakeAll();
- return;
-}
-
-#endif
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_file_impl.h b/trunk/src/video_engine/vie_file_impl.h
deleted file mode 100644
index 63e2d4a..0000000
--- a/trunk/src/video_engine/vie_file_impl.h
+++ /dev/null
@@ -1,131 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_FILE_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_FILE_IMPL_H_
-
-#include "system_wrappers/interface/scoped_ptr.h"
-#include "typedefs.h"
-#include "video_engine/include/vie_file.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_frame_provider_base.h"
-#include "video_engine/vie_ref_count.h"
-#include "video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-class ConditionVariableWrapper;
-class CriticalSectionWrapper;
-class ViESharedData;
-
-class ViECaptureSnapshot : public ViEFrameCallback {
- public:
- ViECaptureSnapshot();
- ~ViECaptureSnapshot();
-
- bool GetSnapshot(VideoFrame& video_frame, unsigned int max_wait_time);
-
- // Implements ViEFrameCallback.
- virtual void DeliverFrame(int id, VideoFrame& video_frame, int num_csrcs = 0,
- const WebRtc_UWord32 CSRC[kRtpCsrcSize] = NULL);
- virtual void DelayChanged(int id, int frame_delay) {}
- virtual int GetPreferedFrameSettings(int& width, int& height,
- int& frame_rate) {
- return -1;
- }
- virtual void ProviderDestroyed(int id) {}
-
- private:
- scoped_ptr<CriticalSectionWrapper> crit_;
- scoped_ptr<ConditionVariableWrapper> condition_varaible_;
- VideoFrame* video_frame_;
-};
-
-class ViEFileImpl
- : public ViEFile,
- public ViERefCount {
- public:
- // Implements ViEFile.
- virtual int Release();
- virtual int StartPlayFile(const char* file_nameUTF8, int& file_id,
- const bool loop = false,
- const FileFormats file_format = kFileFormatAviFile);
- virtual int StopPlayFile(const int file_id);
- virtual int RegisterObserver(int file_id, ViEFileObserver& observer);
- virtual int DeregisterObserver(int file_id, ViEFileObserver& observer);
- virtual int SendFileOnChannel(const int file_id, const int video_channel);
- virtual int StopSendFileOnChannel(const int video_channel);
- virtual int StartPlayFileAsMicrophone(const int file_id,
- const int audio_channel,
- bool mix_microphone = false,
- float volume_scaling = 1);
- virtual int StopPlayFileAsMicrophone(const int file_id,
- const int audio_channel);
- virtual int StartPlayAudioLocally(const int file_id, const int audio_channel,
- float volume_scaling = 1);
- virtual int StopPlayAudioLocally(const int file_id, const int audio_channel);
- virtual int StartRecordOutgoingVideo(
- const int video_channel,
- const char* file_nameUTF8,
- AudioSource audio_source,
- const CodecInst& audio_codec,
- const VideoCodec& video_codec,
- const FileFormats file_format = kFileFormatAviFile);
- virtual int StartRecordIncomingVideo(
- const int video_channel,
- const char* file_nameUTF8,
- AudioSource audio_source,
- const CodecInst& audio_codec,
- const VideoCodec& video_codec,
- const FileFormats file_format = kFileFormatAviFile);
- virtual int StopRecordOutgoingVideo(const int video_channel);
- virtual int StopRecordIncomingVideo(const int video_channel);
- virtual int GetFileInformation(
- const char* file_name,
- VideoCodec& video_codec,
- CodecInst& audio_codec,
- const FileFormats file_format = kFileFormatAviFile);
- virtual int GetRenderSnapshot(const int video_channel,
- const char* file_nameUTF8);
- virtual int GetRenderSnapshot(const int video_channel, ViEPicture& picture);
- virtual int FreePicture(ViEPicture& picture);
- virtual int GetCaptureDeviceSnapshot(const int capture_id,
- const char* file_nameUTF8);
- virtual int GetCaptureDeviceSnapshot(const int capture_id,
- ViEPicture& picture);
- virtual int SetCaptureDeviceImage(const int capture_id,
- const char* file_nameUTF8);
- virtual int SetCaptureDeviceImage(const int capture_id,
- const ViEPicture& picture);
- virtual int SetRenderStartImage(const int video_channel,
- const char* file_nameUTF8);
- virtual int SetRenderStartImage(const int video_channel,
- const ViEPicture& picture);
- virtual int SetRenderTimeoutImage(const int video_channel,
- const char* file_nameUTF8,
- const unsigned int timeout_ms);
- virtual int SetRenderTimeoutImage(const int video_channel,
- const ViEPicture& picture,
- const unsigned int timeout_ms);
-
- protected:
- ViEFileImpl(ViESharedData* shared_data);
- virtual ~ViEFileImpl();
-
- private:
- WebRtc_Word32 GetNextCapturedFrame(WebRtc_Word32 capture_id,
- VideoFrame& video_frame);
-
- ViESharedData* shared_data_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_FILE_IMPL_H_
diff --git a/trunk/src/video_engine/vie_file_player.cc b/trunk/src/video_engine/vie_file_player.cc
deleted file mode 100644
index e40f337..0000000
--- a/trunk/src/video_engine/vie_file_player.cc
+++ /dev/null
@@ -1,508 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_file_player.h"
-
-#include "modules/utility/interface/file_player.h"
-#include "system_wrappers/interface/critical_section_wrapper.h"
-#include "system_wrappers/interface/event_wrapper.h"
-#include "system_wrappers/interface/thread_wrapper.h"
-#include "system_wrappers/interface/tick_util.h"
-#include "system_wrappers/interface/trace.h"
-#include "video_engine/include/vie_file.h"
-#include "video_engine/vie_input_manager.h"
-#include "voice_engine/main/interface/voe_base.h"
-#include "voice_engine/main/interface/voe_file.h"
-#include "voice_engine/main/interface/voe_video_sync.h"
-
-namespace webrtc {
-
-const int kThreadWaitTimeMs = 100;
-
-ViEFilePlayer* ViEFilePlayer::CreateViEFilePlayer(
- int file_id,
- int engine_id,
- const char* file_nameUTF8,
- const bool loop,
- const FileFormats file_format,
- ViEInputManager& input_manager,
- VoiceEngine* voe_ptr) {
- ViEFilePlayer* self = new ViEFilePlayer(file_id, engine_id, input_manager);
- if (!self || self->Init(file_nameUTF8, loop, file_format, voe_ptr) != 0) {
- delete self;
- self = NULL;
- }
- return self;
-}
-
-ViEFilePlayer::ViEFilePlayer(int Id,
- int engine_id,
- ViEInputManager& input_manager)
- : ViEFrameProviderBase(Id, engine_id),
- play_back_started_(false),
- input_manager_(input_manager),
- feedback_cs_(NULL),
- audio_cs_(NULL),
- file_player_(NULL),
- audio_stream_(false),
- video_clients_(0),
- audio_clients_(0),
- local_audio_channel_(-1),
- observer_(NULL),
- voe_file_interface_(NULL),
- voe_video_sync_(NULL),
- decode_thread_(NULL),
- decode_event_(NULL),
- decoded_audio_length_(0) {
- memset(file_name_, 0, FileWrapper::kMaxFileNameSize);
- memset(decoded_audio_, 0, kMaxDecodedAudioLength);
-}
-
-ViEFilePlayer::~ViEFilePlayer() {
- // StopPlay deletes decode_thread_.
- StopPlay();
- delete decode_event_;
- delete audio_cs_;
- delete feedback_cs_;
-}
-
-int ViEFilePlayer::Init(const char* file_nameUTF8,
- const bool loop,
- const FileFormats file_format,
- VoiceEngine* voice_engine) {
- feedback_cs_ = CriticalSectionWrapper::CreateCriticalSection();
- if (!feedback_cs_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "ViEFilePlayer::StartPlay() failed to allocate critsect");
- return -1;
- }
-
- audio_cs_ = CriticalSectionWrapper::CreateCriticalSection();
- if (!audio_cs_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "ViEFilePlayer::StartPlay() failed to allocate critsect");
- return -1;
- }
-
- decode_event_ = EventWrapper::Create();
- if (!decode_event_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "ViEFilePlayer::StartPlay() failed to allocate event");
- return -1;
- }
- if (strlen(file_nameUTF8) > FileWrapper::kMaxFileNameSize) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "ViEFilePlayer::StartPlay() Too long filename");
- return -1;
- }
- strncpy(file_name_, file_nameUTF8, strlen(file_nameUTF8) + 1);
-
- file_player_ = FilePlayer::CreateFilePlayer(ViEId(engine_id_, id_),
- file_format);
- if (!file_player_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "ViEFilePlayer::StartPlay() failed to create file player");
- return -1;
- }
- if (file_player_->RegisterModuleFileCallback(this) == -1) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "ViEFilePlayer::StartPlay() failed to "
- "RegisterModuleFileCallback");
- file_player_ = NULL;
- return -1;
- }
- decode_thread_ = ThreadWrapper::CreateThread(FilePlayDecodeThreadFunction,
- this, kHighestPriority,
- "ViEFilePlayThread");
- if (!decode_thread_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "ViEFilePlayer::StartPlay() failed to start decode thread.");
- file_player_ = NULL;
- return -1;
- }
-
- // Always try to open with Audio since we don't know on what channels the
- // audio should be played on.
- WebRtc_Word32 error = file_player_->StartPlayingVideoFile(file_name_, loop,
- false);
- if (error) {
- // Failed to open the file with audio, try without.
- error = file_player_->StartPlayingVideoFile(file_name_, loop, true);
- audio_stream_ = false;
- if (error) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "ViEFilePlayer::StartPlay() failed to Start play video "
- "file");
- return -1;
- }
-
- } else {
- audio_stream_ = true;
- }
-
- if (audio_stream_) {
- if (voice_engine) {
- // A VoiceEngine has been provided and we want to play audio on local
- // a channel.
- voe_file_interface_ = VoEFile::GetInterface(voice_engine);
- if (!voe_file_interface_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "ViEFilePlayer::StartPlay() failed to get VEFile "
- "interface");
- return -1;
- }
- voe_video_sync_ = VoEVideoSync::GetInterface(voice_engine);
- if (!voe_video_sync_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(engine_id_, id_),
- "ViEFilePlayer::StartPlay() failed to get "
- "VoEVideoSync interface");
- return -1;
- }
- }
- }
-
- // Read audio /(or just video) every 10ms.
- decode_event_->StartTimer(true, 10);
- return 0;
-}
-
-int ViEFilePlayer::FrameCallbackChanged() {
- // Starts the decode thread when someone cares.
- if (ViEFrameProviderBase::NumberOfRegisteredFrameCallbacks() >
- video_clients_) {
- if (!play_back_started_) {
- play_back_started_ = true;
- unsigned int thread_id;
- if (decode_thread_->Start(thread_id)) {
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(engine_id_, id_),
- "ViEFilePlayer::FrameCallbackChanged() Started file decode"
- " thread %u", thread_id);
- } else {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "ViEFilePlayer::FrameCallbackChanged() Failed to start "
- "file decode thread.");
- }
- } else if (!file_player_->IsPlayingFile()) {
- if (file_player_->StartPlayingVideoFile(file_name_, false,
- !audio_stream_) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "ViEFilePlayer::FrameCallbackChanged(), Failed to restart "
- "the file player.");
- }
- }
- }
- video_clients_ = ViEFrameProviderBase::NumberOfRegisteredFrameCallbacks();
- return 0;
-}
-
-bool ViEFilePlayer::FilePlayDecodeThreadFunction(void* obj) {
- return static_cast<ViEFilePlayer*>(obj)->FilePlayDecodeProcess();
-}
-
-bool ViEFilePlayer::FilePlayDecodeProcess() {
- if (decode_event_->Wait(kThreadWaitTimeMs) == kEventSignaled) {
- if (audio_stream_ && audio_clients_ == 0) {
- // There is audio but no one cares, read the audio here.
- Read(NULL, 0);
- }
- if (file_player_->TimeUntilNextVideoFrame() < 10) {
- // Less than 10ms to next videoframe.
- if (file_player_->GetVideoFromFile(decoded_video_) != 0) {
- }
- }
- if (decoded_video_.Length() > 0) {
- if (local_audio_channel_ != -1 && voe_video_sync_) {
- // We are playing audio locally.
- int audio_delay = 0;
- if (voe_video_sync_->GetPlayoutBufferSize(audio_delay) == 0) {
- decoded_video_.SetRenderTime(decoded_video_.RenderTimeMs() +
- audio_delay);
- }
- }
- DeliverFrame(decoded_video_);
- decoded_video_.SetLength(0);
- }
- }
- return true;
-}
-
-int ViEFilePlayer::StopPlay() {
- // Only called from destructor.
- if (decode_thread_) {
- decode_thread_->SetNotAlive();
- if (decode_thread_->Stop()) {
- delete decode_thread_;
- } else {
- assert(!"ViEFilePlayer::StopPlay() Failed to stop decode thread");
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "ViEFilePlayer::StartPlay() Failed to stop file decode "
- "thread.");
- }
- }
- decode_thread_ = NULL;
- if (decode_event_) {
- decode_event_->StopTimer();
- }
- StopPlayAudio();
-
- if (voe_file_interface_) {
- voe_file_interface_->Release();
- voe_file_interface_ = NULL;
- }
- if (voe_video_sync_) {
- voe_video_sync_->Release();
- voe_video_sync_ = NULL;
- }
-
- if (file_player_) {
- file_player_->StopPlayingFile();
- FilePlayer::DestroyFilePlayer(file_player_);
- file_player_ = NULL;
- }
- return 0;
-}
-
-int ViEFilePlayer::StopPlayAudio() {
- // Stop sending audio.
-
- std::set<int>::iterator it = audio_channels_sending_.begin();
- while (it != audio_channels_sending_.end()) {
- StopSendAudioOnChannel(*it);
- // StopSendAudioOnChannel erases the item from the map.
- it = audio_channels_sending_.begin();
- }
-
- // Stop local audio playback.
- if (local_audio_channel_ != -1) {
- StopPlayAudioLocally(local_audio_channel_);
- }
- local_audio_channel_ = -1;
- audio_channel_buffers_.clear();
- audio_clients_ = 0;
- return 0;
-}
-
-int ViEFilePlayer::Read(void* buf, int len) {
- // Protect from simultaneous reading from multiple channels.
- CriticalSectionScoped lock(*audio_cs_);
- if (NeedsAudioFromFile(buf)) {
- // We will run the VoE in 16KHz.
- if (file_player_->Get10msAudioFromFile(decoded_audio_,
- decoded_audio_length_, 16000) != 0) {
- // No data.
- decoded_audio_length_ = 0;
- return 0;
- }
- // 2 bytes per sample.
- decoded_audio_length_ *= 2;
- if (buf) {
- audio_channel_buffers_.push_back(buf);
- }
- } else {
- // No need for new audiobuffer from file, ie the buffer read from file has
- // not been played on this channel.
- }
- if (buf) {
- memcpy(buf, decoded_audio_, decoded_audio_length_);
- }
- return decoded_audio_length_;
-}
-
-bool ViEFilePlayer::NeedsAudioFromFile(void* buf) {
- bool needs_new_audio = false;
- if (audio_channel_buffers_.size() == 0) {
- return true;
- }
-
- // Check if we the buf already have read the current audio.
- for (std::list<void*>::iterator it = audio_channel_buffers_.begin();
- it != audio_channel_buffers_.end(); ++it) {
- if (*it == buf) {
- needs_new_audio = true;
- audio_channel_buffers_.erase(it);
- break;
- }
- }
- return needs_new_audio;
-}
-
-void ViEFilePlayer::PlayFileEnded(const WebRtc_Word32 id) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, id),
- "%s: file_id %d", __FUNCTION__, id_);
- file_player_->StopPlayingFile();
-
- CriticalSectionScoped lock(*feedback_cs_);
- if (observer_) {
- observer_->PlayFileEnded(id_);
- }
-}
-
-bool ViEFilePlayer::IsObserverRegistered() {
- CriticalSectionScoped lock(*feedback_cs_);
- return observer_ != NULL;
-}
-
-int ViEFilePlayer::RegisterObserver(ViEFileObserver& observer) {
- CriticalSectionScoped lock(*feedback_cs_);
- if (observer_) {
- return -1;
- }
- observer_ = &observer;
- return 0;
-}
-
-int ViEFilePlayer::DeRegisterObserver() {
- CriticalSectionScoped lock(*feedback_cs_);
- observer_ = NULL;
- return 0;
-}
-
-int ViEFilePlayer::SendAudioOnChannel(const int audio_channel,
- bool mix_microphone,
- float volume_scaling) {
- if (!voe_file_interface_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "%s No VEFile interface.", __FUNCTION__);
- return -1;
- }
- if (voe_file_interface_->StartPlayingFileAsMicrophone(audio_channel, this,
- mix_microphone,
- kFileFormatPcm16kHzFile,
- volume_scaling) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "ViEFilePlayer::SendAudioOnChannel() "
- "VE_StartPlayingFileAsMicrophone failed. audio_channel %d, "
- " mix_microphone %d, volume_scaling %.2f",
- audio_channel, mix_microphone, volume_scaling);
- return -1;
- }
- audio_channels_sending_.insert(audio_channel);
-
- CriticalSectionScoped lock(*audio_cs_);
- audio_clients_++;
- return 0;
-}
-
-int ViEFilePlayer::StopSendAudioOnChannel(const int audio_channel) {
- int result = 0;
- if (!voe_file_interface_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "ViEFilePlayer::StopSendAudioOnChannel() - no VoE interface");
- return -1;
- }
- std::set<int>::iterator it = audio_channels_sending_.find(audio_channel);
- if (it == audio_channels_sending_.end()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "ViEFilePlayer::StopSendAudioOnChannel AudioChannel %d not "
- "sending", audio_channel);
- return -1;
- }
- result = voe_file_interface_->StopPlayingFileAsMicrophone(audio_channel);
- if (result != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "ViEFilePlayer::StopSendAudioOnChannel() "
- "VE_StopPlayingFileAsMicrophone failed. audio_channel %d",
- audio_channel);
- }
- audio_channels_sending_.erase(audio_channel);
- CriticalSectionScoped lock(*audio_cs_);
- audio_clients_--;
- assert(audio_clients_ >= 0);
- return 0;
-}
-
-int ViEFilePlayer::PlayAudioLocally(const int audio_channel,
- float volume_scaling) {
- if (!voe_file_interface_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "%s No VEFile interface.", __FUNCTION__);
- return -1;
- }
- if (voe_file_interface_->StartPlayingFileLocally(audio_channel, this,
- kFileFormatPcm16kHzFile,
- volume_scaling) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "%s VE_StartPlayingFileAsMicrophone failed. audio_channel %d,"
- " mix_microphone %d, volume_scaling %.2f",
- __FUNCTION__, audio_channel, volume_scaling);
- return -1;
- }
-
- CriticalSectionScoped lock(*audio_cs_);
- local_audio_channel_ = audio_channel;
- audio_clients_++;
- return 0;
-}
-
-int ViEFilePlayer::StopPlayAudioLocally(const int audio_channel) {
- if (!voe_file_interface_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "%s No VEFile interface.", __FUNCTION__);
- return -1;
- }
- if (voe_file_interface_->StopPlayingFileLocally(audio_channel) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(engine_id_, id_),
- "%s VE_StopPlayingFileLocally failed. audio_channel %d.",
- __FUNCTION__, audio_channel);
- return -1;
- }
-
- CriticalSectionScoped lock(*audio_cs_);
- local_audio_channel_ = -1;
- audio_clients_--;
- return 0;
-}
-
-int ViEFilePlayer::GetFileInformation(int engine_id,
- const char* file_name,
- VideoCodec& video_codec,
- CodecInst& audio_codec,
- const FileFormats file_format) {
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, engine_id, "%s ", __FUNCTION__);
-
- FilePlayer* file_player = FilePlayer::CreateFilePlayer(engine_id,
- file_format);
- if (!file_player) {
- return -1;
- }
-
- bool video_only = false;
-
- memset(&video_codec, 0, sizeof(video_codec));
- memset(&audio_codec, 0, sizeof(audio_codec));
-
- if (file_player->StartPlayingVideoFile(file_name, false, false) != 0) {
- video_only = true;
- if (file_player->StartPlayingVideoFile(file_name, false, true) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
- "%s Failed to open file.", __FUNCTION__);
- FilePlayer::DestroyFilePlayer(file_player);
- return -1;
- }
- }
-
- if (!video_only && file_player->AudioCodec(audio_codec) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
- "%s Failed to get audio codec.", __FUNCTION__);
- FilePlayer::DestroyFilePlayer(file_player);
- return -1;
- }
- if (file_player->video_codec_info(video_codec) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, engine_id,
- "%s Failed to get video codec.", __FUNCTION__);
- FilePlayer::DestroyFilePlayer(file_player);
- return -1;
- }
- FilePlayer::DestroyFilePlayer(file_player);
- return 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_file_player.h b/trunk/src/video_engine/vie_file_player.h
deleted file mode 100644
index 640b37c..0000000
--- a/trunk/src/video_engine/vie_file_player.h
+++ /dev/null
@@ -1,141 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_FILE_PLAYER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_FILE_PLAYER_H_
-
-#include <list>
-#include <set>
-
-#include "common_types.h"
-#include "modules/media_file/interface/media_file_defines.h"
-#include "system_wrappers/interface/file_wrapper.h"
-#include "typedefs.h"
-#include "video_engine/vie_frame_provider_base.h"
-
-namespace webrtc {
-
-class EventWrapper;
-class FilePlayer;
-class ThreadWrapper;
-class ViEFileObserver;
-class ViEInputManager;
-class VoEFile;
-class VoEVideoSync;
-class VoiceEngine;
-
-class ViEFilePlayer
- : public ViEFrameProviderBase,
- protected FileCallback,
- protected InStream {
- public:
- static ViEFilePlayer* CreateViEFilePlayer(int file_id,
- int engine_id,
- const char* file_nameUTF8,
- const bool loop,
- const FileFormats file_format,
- ViEInputManager& input_manager,
- VoiceEngine* voe_ptr);
-
- static int GetFileInformation(const int engine_id,
- const char* file_name,
- VideoCodec& video_codec,
- CodecInst& audio_codec,
- const FileFormats file_format);
- ~ViEFilePlayer();
-
- bool IsObserverRegistered();
- int RegisterObserver(ViEFileObserver& observer);
- int DeRegisterObserver();
- int SendAudioOnChannel(const int audio_channel,
- bool mix_microphone,
- float volume_scaling);
- int StopSendAudioOnChannel(const int audio_channel);
- int PlayAudioLocally(const int audio_channel, float volume_scaling);
- int StopPlayAudioLocally(const int audio_channel);
-
- // Implements ViEFrameProviderBase.
- virtual int FrameCallbackChanged();
-
- protected:
- ViEFilePlayer(int Id, int engine_id, ViEInputManager& input_manager);
- int Init(const WebRtc_Word8* file_nameUTF8,
- const bool loop,
- const FileFormats file_format,
- VoiceEngine* voe_ptr);
- int StopPlay();
- int StopPlayAudio();
-
- // File play decode function.
- static bool FilePlayDecodeThreadFunction(void* obj);
- bool FilePlayDecodeProcess();
- bool NeedsAudioFromFile(void* buf);
-
- // Implements webrtc::InStream.
- virtual int Read(void* buf, int len);
- virtual int Rewind() {
- return 0;
- }
-
- // Implements FileCallback.
- virtual void PlayNotification(const WebRtc_Word32 /*id*/,
- const WebRtc_UWord32 /*notification_ms*/) {}
- virtual void RecordNotification(const WebRtc_Word32 /*id*/,
- const WebRtc_UWord32 /*notification_ms*/) {}
- virtual void PlayFileEnded(const WebRtc_Word32 id);
- virtual void RecordFileEnded(const WebRtc_Word32 /*id*/) {}
-
- private:
- static const int kMaxDecodedAudioLength = 320;
- bool play_back_started_;
- ViEInputManager& input_manager_;
-
- CriticalSectionWrapper* feedback_cs_;
- CriticalSectionWrapper* audio_cs_;
-
- FilePlayer* file_player_;
- bool audio_stream_;
-
- // Number of active video clients.
- int video_clients_;
-
- // Number of audio channels sending this audio.
- int audio_clients_;
-
- // Local audio channel playing this video. Sync video against this.
- int local_audio_channel_;
-
- ViEFileObserver* observer_;
- WebRtc_Word8 file_name_[FileWrapper::kMaxFileNameSize];
-
- // VoE Interface.
- VoEFile* voe_file_interface_;
- VoEVideoSync* voe_video_sync_;
-
- // Thread for decoding video (and audio if no audio clients connected).
- ThreadWrapper* decode_thread_;
- EventWrapper* decode_event_;
- WebRtc_Word16 decoded_audio_[kMaxDecodedAudioLength];
- WebRtc_UWord32 decoded_audio_length_;
-
- // Trick - list containing VoE buffer reading this file. Used if multiple
- // audio channels are sending.
- std::list<void*> audio_channel_buffers_;
-
- // AudioChannels sending audio from this file.
- std::set<int> audio_channels_sending_;
-
- // Frame receiving decoded video from file.
- VideoFrame decoded_video_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_FILE_PLAYER_H_
diff --git a/trunk/src/video_engine/vie_file_recorder.cc b/trunk/src/video_engine/vie_file_recorder.cc
deleted file mode 100644
index c2f8fa0..0000000
--- a/trunk/src/video_engine/vie_file_recorder.cc
+++ /dev/null
@@ -1,239 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_file_recorder.h"
-
-#include "modules/utility/interface/file_player.h"
-#include "modules/utility/interface/file_recorder.h"
-#include "system_wrappers/interface/critical_section_wrapper.h"
-#include "system_wrappers/interface/tick_util.h"
-#include "system_wrappers/interface/trace.h"
-#include "video_engine/vie_defines.h"
-
-namespace webrtc {
-
-ViEFileRecorder::ViEFileRecorder(int instanceID)
- : recorder_cs_(CriticalSectionWrapper::CreateCriticalSection()),
- file_recorder_(NULL),
- is_first_frame_recorded_(false),
- is_out_stream_started_(false),
- instance_id_(instanceID),
- frame_delay_(0),
- audio_channel_(-1),
- audio_source_(NO_AUDIO),
- voe_file_interface_(NULL) {
-}
-
-ViEFileRecorder::~ViEFileRecorder() {
- StopRecording();
- delete recorder_cs_;
-}
-
-int ViEFileRecorder::StartRecording(const char* file_nameUTF8,
- const VideoCodec& codec_inst,
- AudioSource audio_source,
- int audio_channel,
- const CodecInst& audio_codec_inst,
- VoiceEngine* voe_ptr,
- const FileFormats file_format) {
- CriticalSectionScoped lock(*recorder_cs_);
-
- if (file_recorder_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, instance_id_,
- "ViEFileRecorder::StartRecording() - already recording.");
- return -1;
- }
- file_recorder_ = FileRecorder::CreateFileRecorder(instance_id_, file_format);
- if (!file_recorder_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, instance_id_,
- "ViEFileRecorder::StartRecording() failed to create recoder.");
- return -1;
- }
-
- int error = file_recorder_->StartRecordingVideoFile(file_nameUTF8,
- audio_codec_inst,
- codec_inst,
- AMRFileStorage,
- audio_source == NO_AUDIO);
- if (error) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, instance_id_,
- "ViEFileRecorder::StartRecording() failed to "
- "StartRecordingVideoFile.");
- FileRecorder::DestroyFileRecorder(file_recorder_);
- file_recorder_ = NULL;
- return -1;
- }
-
- audio_source_ = audio_source;
- if (voe_ptr && audio_source != NO_AUDIO) {
- // VoE interface has been provided and we want to record audio.
- voe_file_interface_ = VoEFile::GetInterface(voe_ptr);
- if (!voe_file_interface_) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, instance_id_,
- "ViEFileRecorder::StartRecording() failed to get VEFile "
- "interface");
- return -1;
- }
-
- // Always L16.
- CodecInst engine_audio_codec_inst = {96, "L16", audio_codec_inst.plfreq,
- audio_codec_inst.plfreq / 100, 1,
- audio_codec_inst.plfreq * 16 };
-
- switch (audio_source) {
- // case NO_AUDIO is checked above.
- case MICROPHONE:
- error = voe_file_interface_->StartRecordingMicrophone(
- this, &engine_audio_codec_inst);
- break;
- case PLAYOUT:
- error = voe_file_interface_->StartRecordingPlayout(
- audio_channel, this, &engine_audio_codec_inst);
- break;
- default:
- assert(!"Unknown audio_source");
- }
- if (error != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, instance_id_,
- "ViEFileRecorder::StartRecording() failed to start recording"
- " audio");
- FileRecorder::DestroyFileRecorder(file_recorder_);
- file_recorder_ = NULL;
- return -1;
- }
- is_out_stream_started_ = true;
- audio_channel_ = audio_channel;
- }
- is_first_frame_recorded_ = false;
- return 0;
-}
-
-int ViEFileRecorder::StopRecording() {
- int error = 0;
- // We can not hold the ptr_cs_ while accessing VoE functions. It might cause
- // deadlock in Write.
- if (voe_file_interface_) {
- switch (audio_source_) {
- case MICROPHONE:
- error = voe_file_interface_->StopRecordingMicrophone();
- break;
- case PLAYOUT:
- error = voe_file_interface_->StopRecordingPlayout(audio_channel_);
- break;
- case NO_AUDIO:
- break;
- default:
- assert(!"Unknown audio_source");
- }
- if (error != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, instance_id_,
- "ViEFileRecorder::StopRecording() failed to stop recording "
- "audio");
- }
- }
- CriticalSectionScoped lock(*recorder_cs_);
- if (voe_file_interface_) {
- voe_file_interface_->Release();
- voe_file_interface_ = NULL;
- }
-
- if (file_recorder_) {
- if (file_recorder_->IsRecording()) {
- int error = file_recorder_->StopRecording();
- if (error) {
- return -1;
- }
- }
- FileRecorder::DestroyFileRecorder(file_recorder_);
- file_recorder_ = NULL;
- }
- is_first_frame_recorded_ = false;
- is_out_stream_started_ = false;
- return 0;
-}
-
-void ViEFileRecorder::SetFrameDelay(int frame_delay) {
- CriticalSectionScoped lock(*recorder_cs_);
- frame_delay_ = frame_delay;
-}
-
-bool ViEFileRecorder::RecordingStarted() {
- CriticalSectionScoped lock(*recorder_cs_);
- return file_recorder_ && file_recorder_->IsRecording();
-}
-
-bool ViEFileRecorder::FirstFrameRecorded() {
- CriticalSectionScoped lock(*recorder_cs_);
- return is_first_frame_recorded_;
-}
-
-bool ViEFileRecorder::IsRecordingFileFormat(const FileFormats file_format) {
- CriticalSectionScoped lock(*recorder_cs_);
- return (file_recorder_->RecordingFileFormat() == file_format) ? true : false;
-}
-
-void ViEFileRecorder::RecordVideoFrame(const VideoFrame& video_frame) {
- CriticalSectionScoped lock(*recorder_cs_);
-
- if (file_recorder_ && file_recorder_->IsRecording()) {
- if (!IsRecordingFileFormat(kFileFormatAviFile))
- return;
-
- // Compensate for frame delay in order to get audio/video sync when
- // recording local video.
- const WebRtc_UWord32 time_stamp = video_frame.TimeStamp();
- const WebRtc_Word64 render_time_stamp = video_frame.RenderTimeMs();
- VideoFrame& unconst_video_frame = const_cast<VideoFrame&>(video_frame);
- unconst_video_frame.SetTimeStamp(time_stamp - 90 * frame_delay_);
- unconst_video_frame.SetRenderTime(render_time_stamp - frame_delay_);
-
- file_recorder_->RecordVideoToFile(unconst_video_frame);
-
- unconst_video_frame.SetRenderTime(render_time_stamp);
- unconst_video_frame.SetTimeStamp(time_stamp);
- }
-}
-
-bool ViEFileRecorder::Write(const void* buf, int len) {
- if (!is_out_stream_started_)
- return true;
-
- // Always 10 ms L16 from VoE.
- if (len % (2 * 80)) {
- // Not 2 bytes 80 samples.
- WEBRTC_TRACE(kTraceError, kTraceVideo, audio_channel_,
- "Audio length not supported: %d.", len);
- return true;
- }
-
- AudioFrame audio_frame;
- WebRtc_UWord16 length_in_samples = len / 2;
- audio_frame.UpdateFrame(audio_channel_, 0,
- static_cast<const WebRtc_Word16*>(buf),
- length_in_samples, length_in_samples * 100,
- AudioFrame::kUndefined,
- AudioFrame::kVadUnknown);
-
- CriticalSectionScoped lock(*recorder_cs_);
- if (file_recorder_ && file_recorder_->IsRecording()) {
- TickTime tick_time = TickTime::Now();
- file_recorder_->RecordAudioToFile(audio_frame, &tick_time);
- }
-
- // Always return true to continue recording.
- return true;
-}
-
-int ViEFileRecorder::Rewind() {
- // Not supported!
- return -1;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_file_recorder.h b/trunk/src/video_engine/vie_file_recorder.h
deleted file mode 100644
index 1d34ad3..0000000
--- a/trunk/src/video_engine/vie_file_recorder.h
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_FILE_RECORDER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_FILE_RECORDER_H_
-
-#include "modules/utility/interface/file_recorder.h"
-#include "typedefs.h"
-#include "video_engine/include/vie_file.h"
-#include "voice_engine/main/interface/voe_file.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-
-class ViEFileRecorder : protected OutStream {
- public:
- explicit ViEFileRecorder(int channel_id);
- ~ViEFileRecorder();
-
- int StartRecording(const char* file_nameUTF8,
- const VideoCodec& codec_inst,
- AudioSource audio_source, int audio_channel,
- const CodecInst& audio_codec_inst,
- VoiceEngine* voe_ptr,
- const FileFormats file_format = kFileFormatAviFile);
- int StopRecording();
-
- void SetFrameDelay(int frame_delay);
- bool RecordingStarted();
-
- // Records incoming decoded video frame to file.
- void RecordVideoFrame(const VideoFrame& video_frame);
-
- protected:
- bool FirstFrameRecorded();
- bool IsRecordingFileFormat(const FileFormats file_format);
-
- // Implements OutStream.
- bool Write(const void* buf, int len);
- int Rewind();
-
- private:
- CriticalSectionWrapper* recorder_cs_;
-
- FileRecorder* file_recorder_;
- bool is_first_frame_recorded_;
- bool is_out_stream_started_;
- int instance_id_;
- int frame_delay_;
- int audio_channel_;
- AudioSource audio_source_;
- VoEFile* voe_file_interface_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_FILE_RECORDER_H_
diff --git a/trunk/src/video_engine/vie_frame_provider_base.cc b/trunk/src/video_engine/vie_frame_provider_base.cc
deleted file mode 100644
index 8b7bf1f..0000000
--- a/trunk/src/video_engine/vie_frame_provider_base.cc
+++ /dev/null
@@ -1,197 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_frame_provider_base.h"
-
-#include <algorithm>
-
-#include "modules/interface/module_common_types.h"
-#include "system_wrappers/interface/critical_section_wrapper.h"
-#include "system_wrappers/interface/tick_util.h"
-#include "system_wrappers/interface/trace.h"
-#include "video_engine/vie_defines.h"
-
-namespace webrtc {
-
-ViEFrameProviderBase::ViEFrameProviderBase(int Id, int engine_id)
- : id_(Id),
- engine_id_(engine_id),
- provider_cs_(CriticalSectionWrapper::CreateCriticalSection()),
- frame_delay_(0) {
-}
-
-ViEFrameProviderBase::~ViEFrameProviderBase() {
- if (frame_callbacks_.size() > 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, id_),
- "FrameCallbacks still exist when Provider deleted %d",
- frame_callbacks_.size());
- }
-
- for (FrameCallbacks::iterator it = frame_callbacks_.begin();
- it != frame_callbacks_.end(); ++it) {
- (*it)->ProviderDestroyed(id_);
- }
- frame_callbacks_.clear();
-}
-
-int ViEFrameProviderBase::Id() {
- return id_;
-}
-
-void ViEFrameProviderBase::DeliverFrame(
- VideoFrame& video_frame,
- int num_csrcs,
- const WebRtc_UWord32 CSRC[kRtpCsrcSize]) {
-#ifdef DEBUG_
- const TickTime start_process_time = TickTime::Now();
-#endif
- CriticalSectionScoped cs(provider_cs_.get());
-
- // Deliver the frame to all registered callbacks.
- if (frame_callbacks_.size() > 0) {
- if (frame_callbacks_.size() == 1) {
- // We don't have to copy the frame.
- frame_callbacks_.front()->DeliverFrame(id_, video_frame, num_csrcs, CSRC);
- } else {
- // Make a copy of the frame for all callbacks.callback
- for (FrameCallbacks::iterator it = frame_callbacks_.begin();
- it != frame_callbacks_.end(); ++it) {
- if (!extra_frame_.get()) {
- extra_frame_.reset(new VideoFrame());
- }
- extra_frame_->CopyFrame(video_frame);
- (*it)->DeliverFrame(id_, *(extra_frame_.get()), num_csrcs, CSRC);
- }
- }
- }
-#ifdef DEBUG_
- const int process_time =
- static_cast<int>((TickTime::Now() - start_process_time).Milliseconds());
- if (process_time > 25) {
- // Warn if the delivery time is too long.
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, id_),
- "%s Too long time: %ums", __FUNCTION__, process_time);
- }
-#endif
-}
-
-void ViEFrameProviderBase::SetFrameDelay(int frame_delay) {
- CriticalSectionScoped cs(provider_cs_.get());
- frame_delay_ = frame_delay;
-
- for (FrameCallbacks::iterator it = frame_callbacks_.begin();
- it != frame_callbacks_.end(); ++it) {
- (*it)->DelayChanged(id_, frame_delay);
- }
-}
-
-int ViEFrameProviderBase::FrameDelay() {
- return frame_delay_;
-}
-
-int ViEFrameProviderBase::GetBestFormat(int& best_width,
- int& best_height,
- int& best_frame_rate) {
- int largest_width = 0;
- int largest_height = 0;
- int highest_frame_rate = 0;
-
- CriticalSectionScoped cs(provider_cs_.get());
- for (FrameCallbacks::iterator it = frame_callbacks_.begin();
- it != frame_callbacks_.end(); ++it) {
- int prefered_width = 0;
- int prefered_height = 0;
- int prefered_frame_rate = 0;
- if ((*it)->GetPreferedFrameSettings(prefered_width, prefered_height,
- prefered_frame_rate) == 0) {
- if (prefered_width > largest_width) {
- largest_width = prefered_width;
- }
- if (prefered_height > largest_height) {
- largest_height = prefered_height;
- }
- if (prefered_frame_rate > highest_frame_rate) {
- highest_frame_rate = prefered_frame_rate;
- }
- }
- }
- best_width = largest_width;
- best_height = largest_height;
- best_frame_rate = highest_frame_rate;
- return 0;
-}
-
-int ViEFrameProviderBase::RegisterFrameCallback(
- int observer_id, ViEFrameCallback* callback_object) {
- assert(callback_object);
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, id_), "%s(0x%p)",
- __FUNCTION__, callback_object);
- {
- CriticalSectionScoped cs(provider_cs_.get());
- if (std::find(frame_callbacks_.begin(), frame_callbacks_.end(),
- callback_object) != frame_callbacks_.end()) {
- // This object is already registered.
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, id_),
- "%s 0x%p already registered", __FUNCTION__,
- callback_object);
- assert("!frameObserver already registered");
- return -1;
- }
- frame_callbacks_.push_back(callback_object);
- }
- // Report current capture delay.
- callback_object->DelayChanged(id_, frame_delay_);
-
- // Notify implementer of this class that the callback list have changed.
- FrameCallbackChanged();
- return 0;
-}
-
-int ViEFrameProviderBase::DeregisterFrameCallback(
- const ViEFrameCallback* callback_object) {
- assert(callback_object);
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, id_), "%s(0x%p)",
- __FUNCTION__, callback_object);
- CriticalSectionScoped cs(provider_cs_.get());
-
- FrameCallbacks::iterator it = std::find(frame_callbacks_.begin(),
- frame_callbacks_.end(),
- callback_object);
- if (it == frame_callbacks_.end()) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, ViEId(engine_id_, id_),
- "%s 0x%p not found", __FUNCTION__, callback_object);
- return -1;
- }
- frame_callbacks_.erase(it);
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, id_),
- "%s 0x%p deregistered", __FUNCTION__, callback_object);
-
- // Notify implementer of this class that the callback list have changed.
- FrameCallbackChanged();
- return 0;
-}
-
-bool ViEFrameProviderBase::IsFrameCallbackRegistered(
- const ViEFrameCallback* callback_object) {
- assert(callback_object);
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, ViEId(engine_id_, id_),
- "%s(0x%p)", __FUNCTION__, callback_object);
-
- CriticalSectionScoped cs(provider_cs_.get());
- return std::find(frame_callbacks_.begin(), frame_callbacks_.end(),
- callback_object) != frame_callbacks_.end();
-}
-
-int ViEFrameProviderBase::NumberOfRegisteredFrameCallbacks() {
- CriticalSectionScoped cs(provider_cs_.get());
- return frame_callbacks_.size();
-}
-
-} // namespac webrtc
diff --git a/trunk/src/video_engine/vie_frame_provider_base.h b/trunk/src/video_engine/vie_frame_provider_base.h
deleted file mode 100644
index 466e12e..0000000
--- a/trunk/src/video_engine/vie_frame_provider_base.h
+++ /dev/null
@@ -1,102 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_FRAME_PROVIDER_BASE_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_FRAME_PROVIDER_BASE_H_
-
-#include <vector>
-
-#include "common_types.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-#include "typedefs.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class VideoEncoder;
-class VideoFrame;
-
-// ViEFrameCallback shall be implemented by all classes receiving frames from a
-// frame provider.
-class ViEFrameCallback {
- public:
- virtual void DeliverFrame(int id,
- VideoFrame& video_frame,
- int num_csrcs = 0,
- const WebRtc_UWord32 CSRC[kRtpCsrcSize] = NULL) = 0;
-
- // The capture delay has changed from the provider. |frame_delay| is given in
- // ms.
- virtual void DelayChanged(int id, int frame_delay) = 0;
-
- // Get the width, height and frame rate preferred by this observer.
- virtual int GetPreferedFrameSettings(int& width,
- int& height,
- int& frame_rate) = 0;
-
- // ProviderDestroyed is called when the frame is about to be destroyed. There
- // must not be any more calls to the frame provider after this.
- virtual void ProviderDestroyed(int id) = 0;
-
- virtual ~ViEFrameCallback() {}
-};
-
-// ViEFrameProviderBase is a base class that will deliver frames to all
-// registered ViEFrameCallbacks.
-class ViEFrameProviderBase {
- public:
- ViEFrameProviderBase(int Id, int engine_id);
- virtual ~ViEFrameProviderBase();
-
- // Returns the frame provider id.
- int Id();
-
- // Register frame callbacks, i.e. a receiver of the captured frame.
- virtual int RegisterFrameCallback(int observer_id,
- ViEFrameCallback* callback_object);
-
- virtual int DeregisterFrameCallback(const ViEFrameCallback* callback_object);
-
- virtual bool IsFrameCallbackRegistered(
- const ViEFrameCallback* callback_object);
-
- int NumberOfRegisteredFrameCallbacks();
-
- // FrameCallbackChanged
- // Inherited classes should check for new frame_settings and reconfigure
- // output if possible.
- virtual int FrameCallbackChanged() = 0;
-
- protected:
- void DeliverFrame(VideoFrame& video_frame,
- int num_csrcs = 0,
- const WebRtc_UWord32 CSRC[kRtpCsrcSize] = NULL);
- void SetFrameDelay(int frame_delay);
- int FrameDelay();
- int GetBestFormat(int& best_width,
- int& best_height,
- int& best_frame_rate);
-
- int id_;
- int engine_id_;
-
- // Frame callbacks.
- typedef std::vector<ViEFrameCallback*> FrameCallbacks;
- FrameCallbacks frame_callbacks_;
- scoped_ptr<CriticalSectionWrapper> provider_cs_;
-
- private:
- scoped_ptr<VideoFrame> extra_frame_;
- int frame_delay_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_FRAME_PROVIDER_BASE_H_
diff --git a/trunk/src/video_engine/vie_image_process_impl.cc b/trunk/src/video_engine/vie_image_process_impl.cc
deleted file mode 100644
index d9dfcd4..0000000
--- a/trunk/src/video_engine/vie_image_process_impl.cc
+++ /dev/null
@@ -1,280 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_image_process_impl.h"
-
-#include "system_wrappers/interface/trace.h"
-#include "video_engine/include/vie_errors.h"
-#include "video_engine/vie_capturer.h"
-#include "video_engine/vie_channel.h"
-#include "video_engine/vie_channel_manager.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_encoder.h"
-#include "video_engine/vie_impl.h"
-#include "video_engine/vie_input_manager.h"
-#include "video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-ViEImageProcess* ViEImageProcess::GetInterface(VideoEngine* video_engine) {
-#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
- if (!video_engine) {
- return NULL;
- }
- VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
- ViEImageProcessImpl* vie_image_process_impl = vie_impl;
- // Increase ref count.
- (*vie_image_process_impl)++;
- return vie_image_process_impl;
-#else
- return NULL;
-#endif
-}
-
-int ViEImageProcessImpl::Release() {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
- "ViEImageProcess::Release()");
- // Decrease ref count.
- (*this)--;
-
- WebRtc_Word32 ref_count = GetCount();
- if (ref_count < 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
- "ViEImageProcess release too many times");
- shared_data_->SetLastError(kViEAPIDoesNotExist);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
- "ViEImageProcess reference count: %d", ref_count);
- return ref_count;
-}
-
-ViEImageProcessImpl::ViEImageProcessImpl(ViESharedData* shared_data)
- : shared_data_(shared_data) {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViEImageProcessImpl::ViEImageProcessImpl() Ctor");
-}
-
-ViEImageProcessImpl::~ViEImageProcessImpl() {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViEImageProcessImpl::~ViEImageProcessImpl() Dtor");
-}
-
-int ViEImageProcessImpl::RegisterCaptureEffectFilter(
- const int capture_id,
- ViEEffectFilter& capture_filter) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(capture_id: %d)", __FUNCTION__, capture_id);
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
-
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViECapturer* vie_capture = is.Capture(capture_id);
- if (!vie_capture) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Capture device %d doesn't exist", __FUNCTION__,
- capture_id);
- shared_data_->SetLastError(kViEImageProcessInvalidCaptureId);
- return -1;
- }
- if (vie_capture->RegisterEffectFilter(&capture_filter) != 0) {
- shared_data_->SetLastError(kViEImageProcessFilterExists);
- return -1;
- }
- return 0;
-}
-
-int ViEImageProcessImpl::DeregisterCaptureEffectFilter(const int capture_id) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(capture_id: %d)", __FUNCTION__, capture_id);
-
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViECapturer* vie_capture = is.Capture(capture_id);
- if (!vie_capture) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Capture device %d doesn't exist", __FUNCTION__,
- capture_id);
- shared_data_->SetLastError(kViEImageProcessInvalidCaptureId);
- return -1;
- }
- if (vie_capture->RegisterEffectFilter(NULL) != 0) {
- shared_data_->SetLastError(kViEImageProcessFilterDoesNotExist);
- return -1;
- }
- return 0;
-}
-
-int ViEImageProcessImpl::RegisterSendEffectFilter(
- const int video_channel,
- ViEEffectFilter& send_filter) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(video_channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (vie_encoder == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
- return -1;
- }
-
- if (vie_encoder->RegisterEffectFilter(&send_filter) != 0) {
- shared_data_->SetLastError(kViEImageProcessFilterExists);
- return -1;
- }
- return 0;
-}
-
-int ViEImageProcessImpl::DeregisterSendEffectFilter(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(video_channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (vie_encoder == NULL) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
- return -1;
- }
- if (vie_encoder->RegisterEffectFilter(NULL) != 0) {
- shared_data_->SetLastError(kViEImageProcessFilterDoesNotExist);
- return -1;
- }
- return 0;
-}
-
-int ViEImageProcessImpl::RegisterRenderEffectFilter(
- const int video_channel,
- ViEEffectFilter& render_filter) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(video_channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
- return -1;
- }
- if (vie_channel->RegisterEffectFilter(&render_filter) != 0) {
- shared_data_->SetLastError(kViEImageProcessFilterExists);
- return -1;
- }
- return 0;
-}
-
-int ViEImageProcessImpl::DeregisterRenderEffectFilter(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(video_channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
- return -1;
- }
-
- if (vie_channel->RegisterEffectFilter(NULL) != 0) {
- shared_data_->SetLastError(kViEImageProcessFilterDoesNotExist);
- return -1;
- }
- return 0;
-}
-
-int ViEImageProcessImpl::EnableDeflickering(const int capture_id,
- const bool enable) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(capture_id: %d, enable: %d)", __FUNCTION__, capture_id,
- enable);
-
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViECapturer* vie_capture = is.Capture(capture_id);
- if (!vie_capture) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Capture device %d doesn't exist", __FUNCTION__,
- capture_id);
- shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
- return -1;
- }
-
- if (vie_capture->EnableDeflickering(enable) != 0) {
- if (enable) {
- shared_data_->SetLastError(kViEImageProcessAlreadyEnabled);
- } else {
- shared_data_->SetLastError(kViEImageProcessAlreadyDisabled);
- }
- return -1;
- }
- return 0;
-}
-
-int ViEImageProcessImpl::EnableDenoising(const int capture_id,
- const bool enable) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(capture_id: %d, enable: %d)", __FUNCTION__, capture_id,
- enable);
-
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViECapturer* vie_capture = is.Capture(capture_id);
- if (!vie_capture) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Capture device %d doesn't exist", __FUNCTION__,
- capture_id);
- shared_data_->SetLastError(kViEImageProcessInvalidCaptureId);
- return -1;
- }
-
- if (vie_capture->EnableDenoising(enable) != 0) {
- if (enable) {
- shared_data_->SetLastError(kViEImageProcessAlreadyEnabled);
- } else {
- shared_data_->SetLastError(kViEImageProcessAlreadyDisabled);
- }
- return -1;
- }
- return 0;
-}
-
-int ViEImageProcessImpl::EnableColorEnhancement(const int video_channel,
- const bool enable) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(video_channel: %d, enable: %d)", __FUNCTION__, video_channel,
- enable);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViEImageProcessInvalidChannelId);
- return -1;
- }
- if (vie_channel->EnableColorEnhancement(enable) != 0) {
- if (enable) {
- shared_data_->SetLastError(kViEImageProcessAlreadyEnabled);
- } else {
- shared_data_->SetLastError(kViEImageProcessAlreadyDisabled);
- }
- return -1;
- }
- return 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_image_process_impl.h b/trunk/src/video_engine/vie_image_process_impl.h
deleted file mode 100644
index 169c9e7..0000000
--- a/trunk/src/video_engine/vie_image_process_impl.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_IMAGE_PROCESS_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_IMAGE_PROCESS_IMPL_H_
-
-#include "typedefs.h"
-#include "video_engine/include/vie_image_process.h"
-#include "video_engine/vie_ref_count.h"
-
-namespace webrtc {
-
-class ViESharedData;
-
-class ViEImageProcessImpl
- : public ViEImageProcess,
- public ViERefCount {
- public:
- // Implements ViEImageProcess.
- virtual int Release();
- virtual int RegisterCaptureEffectFilter(const int capture_id,
- ViEEffectFilter& capture_filter);
- virtual int DeregisterCaptureEffectFilter(const int capture_id);
- virtual int RegisterSendEffectFilter(const int video_channel,
- ViEEffectFilter& send_filter);
- virtual int DeregisterSendEffectFilter(const int video_channel);
- virtual int RegisterRenderEffectFilter(const int video_channel,
- ViEEffectFilter& render_filter);
- virtual int DeregisterRenderEffectFilter(const int video_channel);
- virtual int EnableDeflickering(const int capture_id, const bool enable);
- virtual int EnableDenoising(const int capture_id, const bool enable);
- virtual int EnableColorEnhancement(const int video_channel,
- const bool enable);
-
- protected:
- ViEImageProcessImpl(ViESharedData* shared_data);
- virtual ~ViEImageProcessImpl();
-
- private:
- ViESharedData* shared_data_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_IMAGE_PROCESS_IMPL_H_
diff --git a/trunk/src/video_engine/vie_impl.cc b/trunk/src/video_engine/vie_impl.cc
deleted file mode 100644
index d8a421b..0000000
--- a/trunk/src/video_engine/vie_impl.cc
+++ /dev/null
@@ -1,232 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_impl.h"
-
-#if (defined(WIN32_) || defined(WIN64_))
-#include <Windows.h> // For LoadLibrary.
-#include <tchar.h> // For T_.
-#endif
-
-#include "system_wrappers/interface/trace.h"
-
-#ifdef WEBRTC_ANDROID
-#include "modules/video_capture/main/interface/video_capture_factory.h"
-#include "modules/video_render/main/interface/video_render.h"
-#endif
-
-// Global counter to get an id for each new ViE instance.
-static WebRtc_Word32 g_vie_active_instance_counter = 0;
-
-namespace webrtc {
-
-// extern "C" ensures that GetProcAddress() can find the function address.
-extern "C" {
- VideoEngine* GetVideoEngine() {
- VideoEngineImpl* self = new VideoEngineImpl();
- if (!self) {
- return NULL;
- }
- g_vie_active_instance_counter++;
- VideoEngine* vie = reinterpret_cast<VideoEngine*>(self);
- return vie;
- }
-}
-
-VideoEngine* VideoEngine::Create() {
-#if (defined(WIN32_) || defined(WIN64_))
- // Load a debug dll, if there is one.
- HMODULE hmod_ = LoadLibrary(TEXT("VideoEngineTestingDLL.dll"));
- if (hmod_) {
- typedef VideoEngine* (*PFNGetVideoEngineLib)(void);
- PFNGetVideoEngineLib pfn =
- (PFNGetVideoEngineLib)GetProcAddress(hmod_, "GetVideoEngine");
- if (pfn) {
- VideoEngine* self = pfn();
- return self;
- } else {
- assert(!"Failed to open test dll VideoEngineTestingDLL.dll");
- return NULL;
- }
- }
-#endif
-
- return GetVideoEngine();
-}
-
-bool VideoEngine::Delete(VideoEngine*& video_engine) {
- if (!video_engine) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
- "VideoEngine::Delete - No argument");
- return false;
- }
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, g_vie_active_instance_counter,
- "VideoEngine::Delete(vie = 0x%p)", video_engine);
- VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
-
- // Check all reference counters.
- ViEBaseImpl* vie_base = vie_impl;
- if (vie_base->GetCount() > 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
- "ViEBase ref count: %d", vie_base->GetCount());
- return false;
- }
-#ifdef WEBRTC_VIDEO_ENGINE_CAPTURE_API
- ViECaptureImpl* vie_capture = vie_impl;
- if (vie_capture->GetCount() > 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
- "ViECapture ref count: %d", vie_capture->GetCount());
- return false;
- }
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_CODEC_API
- ViECodecImpl* vie_codec = vie_impl;
- if (vie_codec->GetCount() > 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
- "ViECodec ref count: %d", vie_codec->GetCount());
- return false;
- }
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_ENCRYPTION_API
- ViEEncryptionImpl* vie_encryption = vie_impl;
- if (vie_encryption->GetCount() > 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
- "ViEEncryption ref count: %d", vie_encryption->GetCount());
- return false;
- }
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
- ViEExternalCodecImpl* vie_external_codec = vie_impl;
- if (vie_external_codec->GetCount() > 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
- "ViEEncryption ref count: %d", vie_encryption->GetCount());
- return false;
- }
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
- ViEFileImpl* vie_file = vie_impl;
- if (vie_file->GetCount() > 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
- "ViEFile ref count: %d", vie_file->GetCount());
- return false;
- }
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
- ViEImageProcessImpl* vie_image_process = vie_impl;
- if (vie_image_process->GetCount() > 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
- "ViEImageProcess ref count: %d",
- vie_image_process->GetCount());
- return false;
- }
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_NETWORK_API
- ViENetworkImpl* vie_network = vie_impl;
- if (vie_network->GetCount() > 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
- "ViENetwork ref count: %d", vie_network->GetCount());
- return false;
- }
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
- ViERenderImpl* vie_render = vie_impl;
- if (vie_render->GetCount() > 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
- "ViERender ref count: %d", vie_render->GetCount());
- return false;
- }
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
- ViERTP_RTCPImpl* vie_rtp_rtcp = vie_impl;
- if (vie_rtp_rtcp->GetCount() > 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
- "ViERTP_RTCP ref count: %d", vie_rtp_rtcp->GetCount());
- return false;
- }
-#endif
-
- delete vie_impl;
- vie_impl = NULL;
- video_engine = NULL;
-
- // Decrease the number of instances.
- g_vie_active_instance_counter--;
-
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, g_vie_active_instance_counter,
- "%s: instance deleted. Remaining instances: %d", __FUNCTION__,
- g_vie_active_instance_counter);
- return true;
-}
-
-int VideoEngine::SetTraceFile(const char* file_nameUTF8,
- const bool add_file_counter) {
- if (!file_nameUTF8) {
- return -1;
- }
- if (Trace::SetTraceFile(file_nameUTF8, add_file_counter) == -1) {
- return -1;
- }
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, g_vie_active_instance_counter,
- "SetTraceFileName(file_nameUTF8 = %s, add_file_counter = %d",
- file_nameUTF8, add_file_counter);
- return 0;
-}
-
-int VideoEngine::SetTraceFilter(const unsigned int filter) {
- WebRtc_UWord32 old_filter = 0;
- Trace::LevelFilter(old_filter);
-
- if (filter == kTraceNone && old_filter != kTraceNone) {
- // Do the logging before turning it off.
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, g_vie_active_instance_counter,
- "SetTraceFilter(filter = 0x%x)", filter);
- }
-
- WebRtc_Word32 error = Trace::SetLevelFilter(filter);
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, g_vie_active_instance_counter,
- "SetTraceFilter(filter = 0x%x)", filter);
- if (error != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
- "SetTraceFilter error: %d", error);
- return -1;
- }
- return 0;
-}
-
-int VideoEngine::SetTraceCallback(TraceCallback* callback) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, g_vie_active_instance_counter,
- "SetTraceCallback(TraceCallback = 0x%p)", callback);
- return Trace::SetTraceCallback(callback);
-}
-
-int VideoEngine::SetAndroidObjects(void* javaVM, void* javaContext) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, g_vie_active_instance_counter,
- "SetAndroidObjects()");
-
-#ifdef WEBRTC_ANDROID
- if (VideoCaptureFactory::SetAndroidObjects(javaVM, javaContext) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
- "Could not set capture module Android objects");
- return -1;
- }
- if (VideoRender::SetAndroidObjects(javaVM) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
- "Could not set render module Android objects");
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceError, kTraceVideo, g_vie_active_instance_counter,
- "WEBRTC_ANDROID not defined for VideoEngine::SetAndroidObjects");
- return -1;
-#endif
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_impl.h b/trunk/src/video_engine/vie_impl.h
deleted file mode 100644
index 3141868..0000000
--- a/trunk/src/video_engine/vie_impl.h
+++ /dev/null
@@ -1,115 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_IMPL_H_
-
-#include "engine_configurations.h"
-#include "video_engine/vie_defines.h"
-
-#include "video_engine/vie_base_impl.h"
-
-#ifdef WEBRTC_VIDEO_ENGINE_CAPTURE_API
-#include "video_engine/vie_capture_impl.h"
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_CODEC_API
-#include "video_engine/vie_codec_impl.h"
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_ENCRYPTION_API
-#include "video_engine/vie_encryption_impl.h"
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
-#include "video_engine/vie_file_impl.h"
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
-#include "video_engine/vie_image_process_impl.h"
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_NETWORK_API
-#include "video_engine/vie_network_impl.h"
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
-#include "video_engine/vie_render_impl.h"
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
-#include "video_engine/vie_rtp_rtcp_impl.h"
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
-#include "video_engine/vie_external_codec_impl.h"
-#endif
-
-namespace webrtc {
-
-class VideoEngineImpl
- : public ViEBaseImpl
-#ifdef WEBRTC_VIDEO_ENGINE_CODEC_API
- , public ViECodecImpl
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_CAPTURE_API
- , public ViECaptureImpl
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_ENCRYPTION_API
- , public ViEEncryptionImpl
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
- , public ViEFileImpl
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
- , public ViEImageProcessImpl
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_NETWORK_API
- , public ViENetworkImpl
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
- , public ViERenderImpl
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
- , public ViERTP_RTCPImpl
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
- , public ViEExternalCodecImpl
-#endif
-{ // NOLINT
- public:
- VideoEngineImpl()
- :
-#ifdef WEBRTC_VIDEO_ENGINE_CODEC_API
- ViECodecImpl(ViEBaseImpl::shared_data())
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_CAPTURE_API
- , ViECaptureImpl(ViEBaseImpl::shared_data())
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_ENCRYPTION_API
- , ViEEncryptionImpl(ViEBaseImpl::shared_data())
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_FILE_API
- , ViEFileImpl(ViEBaseImpl::shared_data())
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_IMAGE_PROCESS_API
- , ViEImageProcessImpl(ViEBaseImpl::shared_data())
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_NETWORK_API
- , ViENetworkImpl(ViEBaseImpl::shared_data())
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
- , ViERenderImpl(ViEBaseImpl::shared_data())
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
- , ViERTP_RTCPImpl(ViEBaseImpl::shared_data())
-#endif
-#ifdef WEBRTC_VIDEO_ENGINE_EXTERNAL_CODEC_API
- , ViEExternalCodecImpl(ViEBaseImpl::shared_data())
-#endif
- {}
- virtual ~VideoEngineImpl() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_IMPL_H_
diff --git a/trunk/src/video_engine/vie_input_manager.cc b/trunk/src/video_engine/vie_input_manager.cc
deleted file mode 100644
index 98c5535..0000000
--- a/trunk/src/video_engine/vie_input_manager.cc
+++ /dev/null
@@ -1,600 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_input_manager.h"
-
-#include <cassert>
-
-#include "common_types.h"
-#include "modules/video_capture/main/interface/video_capture_factory.h"
-#include "modules/video_coding/main/interface/video_coding.h"
-#include "modules/video_coding/main/interface/video_coding_defines.h"
-#include "system_wrappers/interface/critical_section_wrapper.h"
-#include "system_wrappers/interface/rw_lock_wrapper.h"
-#include "system_wrappers/interface/trace.h"
-#include "video_engine/include/vie_errors.h"
-#include "video_engine/vie_capturer.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_file_player.h"
-
-namespace webrtc {
-
-ViEInputManager::ViEInputManager(const int engine_id)
- : engine_id_(engine_id),
- map_cs_(CriticalSectionWrapper::CreateCriticalSection()),
- vie_frame_provider_map_(),
- capture_device_info_(NULL),
- module_process_thread_(NULL) {
- WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s", __FUNCTION__);
-
- for (int idx = 0; idx < kViEMaxCaptureDevices; idx++) {
- free_capture_device_id_[idx] = true;
- }
- capture_device_info_ = VideoCaptureFactory::CreateDeviceInfo(
- ViEModuleId(engine_id_));
- for (int idx = 0; idx < kViEMaxFilePlayers; idx++) {
- free_file_id_[idx] = true;
- }
-}
-
-ViEInputManager::~ViEInputManager() {
- WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s", __FUNCTION__);
- while (vie_frame_provider_map_.Size() != 0) {
- MapItem* item = vie_frame_provider_map_.First();
- assert(item);
- ViEFrameProviderBase* frame_provider =
- static_cast<ViEFrameProviderBase*>(item->GetItem());
- vie_frame_provider_map_.Erase(item);
- delete frame_provider;
- }
-
- if (capture_device_info_) {
- delete capture_device_info_;
- capture_device_info_ = NULL;
- }
-}
-void ViEInputManager::SetModuleProcessThread(
- ProcessThread& module_process_thread) {
- assert(!module_process_thread_);
- module_process_thread_ = &module_process_thread;
-}
-
-int ViEInputManager::NumberOfCaptureDevices() {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_), "%s",
- __FUNCTION__);
- assert(capture_device_info_);
- return capture_device_info_->NumberOfDevices();
-}
-
-int ViEInputManager::GetDeviceName(WebRtc_UWord32 device_number,
- WebRtc_UWord8* device_nameUTF8,
- WebRtc_UWord32 device_name_length,
- WebRtc_UWord8* device_unique_idUTF8,
- WebRtc_UWord32 device_unique_idUTF8Length) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s(device_number: %d)", __FUNCTION__, device_number);
- assert(capture_device_info_);
- return capture_device_info_->GetDeviceName(device_number, device_nameUTF8,
- device_name_length,
- device_unique_idUTF8,
- device_unique_idUTF8Length);
-}
-
-int ViEInputManager::NumberOfCaptureCapabilities(
- const WebRtc_UWord8* device_unique_idUTF8) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_), "%s",
- __FUNCTION__);
- assert(capture_device_info_);
- return capture_device_info_->NumberOfCapabilities(device_unique_idUTF8);
-}
-
-int ViEInputManager::GetCaptureCapability(
- const WebRtc_UWord8* device_unique_idUTF8,
- const WebRtc_UWord32 device_capability_number,
- CaptureCapability& capability) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s(device_unique_idUTF8: %s, device_capability_number: %d)",
- __FUNCTION__, device_unique_idUTF8, device_capability_number);
- assert(capture_device_info_);
- VideoCaptureCapability module_capability;
- int result = capture_device_info_->GetCapability(device_unique_idUTF8,
- device_capability_number,
- module_capability);
- if (result != 0)
- return result;
-
- // Copy from module type to public type.
- capability.expectedCaptureDelay = module_capability.expectedCaptureDelay;
- capability.height = module_capability.height;
- capability.width = module_capability.width;
- capability.interlaced = module_capability.interlaced;
- capability.rawType = module_capability.rawType;
- capability.codecType = module_capability.codecType;
- capability.maxFPS = module_capability.maxFPS;
- return result;
-}
-
-int ViEInputManager::GetOrientation(const WebRtc_UWord8* device_unique_idUTF8,
- RotateCapturedFrame& orientation) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s(device_unique_idUTF8: %s,)", __FUNCTION__,
- device_unique_idUTF8);
- assert(capture_device_info_);
- VideoCaptureRotation module_orientation;
- int result = capture_device_info_->GetOrientation(device_unique_idUTF8,
- module_orientation);
- // Copy from module type to public type.
- switch (module_orientation) {
- case kCameraRotate0:
- orientation = RotateCapturedFrame_0;
- break;
- case kCameraRotate90:
- orientation = RotateCapturedFrame_90;
- break;
- case kCameraRotate180:
- orientation = RotateCapturedFrame_180;
- break;
- case kCameraRotate270:
- orientation = RotateCapturedFrame_270;
- break;
- }
- return result;
-}
-
-int ViEInputManager::DisplayCaptureSettingsDialogBox(
- const WebRtc_UWord8* device_unique_idUTF8,
- const WebRtc_UWord8* dialog_titleUTF8,
- void* parent_window,
- WebRtc_UWord32 positionX,
- WebRtc_UWord32 positionY) {
- assert(capture_device_info_);
- return capture_device_info_->DisplayCaptureSettingsDialogBox(
- device_unique_idUTF8, dialog_titleUTF8, parent_window, positionX,
- positionY);
-}
-
-int ViEInputManager::CreateCaptureDevice(
- const WebRtc_UWord8* device_unique_idUTF8,
- const WebRtc_UWord32 device_unique_idUTF8Length,
- int& capture_id) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s(device_unique_id: %s)", __FUNCTION__, device_unique_idUTF8);
- CriticalSectionScoped cs(map_cs_.get());
-
- // Make sure the device is not already allocated.
- for (MapItem* item = vie_frame_provider_map_.First(); item != NULL;
- item = vie_frame_provider_map_.Next(item)) {
- // Make sure this is a capture device.
- if (item->GetId() >= kViECaptureIdBase &&
- item->GetId() <= kViECaptureIdMax) {
- ViECapturer* vie_capture = static_cast<ViECapturer*>(item->GetItem());
- assert(vie_capture);
- // TODO(mflodman) Can we change input to avoid this cast?
- const char* device_name =
- reinterpret_cast<const char*>(vie_capture->CurrentDeviceName());
- if (strncmp(device_name,
- reinterpret_cast<const char*>(device_unique_idUTF8),
- strlen(device_name)) == 0) {
- return kViECaptureDeviceAlreadyAllocated;
- }
- }
- }
-
- // Make sure the device name is valid.
- bool found_device = false;
- for (WebRtc_UWord32 device_index = 0;
- device_index < capture_device_info_->NumberOfDevices(); ++device_index) {
- if (device_unique_idUTF8Length > kVideoCaptureUniqueNameLength) {
- // User's string length is longer than the max.
- return -1;
- }
-
- WebRtc_UWord8 found_name[kVideoCaptureDeviceNameLength] = "";
- WebRtc_UWord8 found_unique_name[kVideoCaptureUniqueNameLength] = "";
- capture_device_info_->GetDeviceName(device_index, found_name,
- kVideoCaptureDeviceNameLength,
- found_unique_name,
- kVideoCaptureUniqueNameLength);
-
- // TODO(mflodman) Can we change input to avoid this cast?
- const char* cast_id = reinterpret_cast<const char*>(device_unique_idUTF8);
- if (strncmp(cast_id, reinterpret_cast<const char*>(found_unique_name),
- strlen(cast_id)) == 0) {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s:%d Capture device was found by unique ID: %s. Returning",
- __FUNCTION__, __LINE__, device_unique_idUTF8);
- found_device = true;
- break;
- }
- }
- if (!found_device) {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s:%d Capture device NOT found by unique ID: %s. Returning",
- __FUNCTION__, __LINE__, device_unique_idUTF8);
- return kViECaptureDeviceDoesNotExist;
- }
-
- int newcapture_id = 0;
- if (GetFreeCaptureId(newcapture_id) == false) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s: Maximum supported number of capture devices already in "
- "use", __FUNCTION__);
- return kViECaptureDeviceMaxNoDevicesAllocated;
- }
- ViECapturer* vie_capture = ViECapturer::CreateViECapture(
- newcapture_id, engine_id_, device_unique_idUTF8,
- device_unique_idUTF8Length, *module_process_thread_);
- if (!vie_capture) {
- ReturnCaptureId(newcapture_id);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s: Could not create capture module for %s", __FUNCTION__,
- device_unique_idUTF8);
- return kViECaptureDeviceUnknownError;
- }
-
- if (vie_frame_provider_map_.Insert(newcapture_id, vie_capture) != 0) {
- ReturnCaptureId(newcapture_id);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s: Could not insert capture module for %s", __FUNCTION__,
- device_unique_idUTF8);
- return kViECaptureDeviceUnknownError;
- }
- capture_id = newcapture_id;
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s(device_unique_id: %s, capture_id: %d)", __FUNCTION__,
- device_unique_idUTF8, capture_id);
- return 0;
-}
-
-int ViEInputManager::CreateCaptureDevice(VideoCaptureModule& capture_module,
- int& capture_id) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_), "%s",
- __FUNCTION__);
-
- CriticalSectionScoped cs(map_cs_.get());
- int newcapture_id = 0;
- if (!GetFreeCaptureId(newcapture_id)) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s: Maximum supported number of capture devices already in "
- "use", __FUNCTION__);
- return kViECaptureDeviceMaxNoDevicesAllocated;
- }
-
- ViECapturer* vie_capture = ViECapturer::CreateViECapture(
- newcapture_id, engine_id_, capture_module, *module_process_thread_);
- if (!vie_capture) {
- ReturnCaptureId(newcapture_id);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s: Could attach capture module.", __FUNCTION__);
- return kViECaptureDeviceUnknownError;
- }
- if (vie_frame_provider_map_.Insert(newcapture_id, vie_capture) != 0) {
- ReturnCaptureId(newcapture_id);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s: Could not insert capture module", __FUNCTION__);
- return kViECaptureDeviceUnknownError;
- }
- capture_id = newcapture_id;
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s, capture_id: %d", __FUNCTION__, capture_id);
- return 0;
-}
-
-int ViEInputManager::DestroyCaptureDevice(const int capture_id) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s(capture_id: %d)", __FUNCTION__, capture_id);
- ViECapturer* vie_capture = NULL;
- {
- // We need exclusive access to the object to delete it.
- // Take this write lock first since the read lock is taken before map_cs_.
- ViEManagerWriteScoped wl(*this);
- CriticalSectionScoped cs(map_cs_.get());
-
- vie_capture = ViECapturePtr(capture_id);
- if (!vie_capture) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s(capture_id: %d) - No such capture device id",
- __FUNCTION__, capture_id);
- return -1;
- }
- WebRtc_UWord32 num_callbacks =
- vie_capture->NumberOfRegisteredFrameCallbacks();
- if (num_callbacks > 0) {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo,
- ViEId(engine_id_), "%s(capture_id: %d) - %u registered "
- "callbacks when destroying capture device",
- __FUNCTION__, capture_id, num_callbacks);
- }
- vie_frame_provider_map_.Erase(capture_id);
- ReturnCaptureId(capture_id);
- // Leave cs before deleting the capture object. This is because deleting the
- // object might cause deletions of renderers so we prefer to not have a lock
- // at that time.
- }
- delete vie_capture;
- return 0;
-}
-
-int ViEInputManager::CreateExternalCaptureDevice(
- ViEExternalCapture*& external_capture,
- int& capture_id) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_), "%s",
- __FUNCTION__);
- CriticalSectionScoped cs(map_cs_.get());
-
- int newcapture_id = 0;
- if (GetFreeCaptureId(newcapture_id) == false) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s: Maximum supported number of capture devices already in "
- "use", __FUNCTION__);
- return kViECaptureDeviceMaxNoDevicesAllocated;
- }
-
- ViECapturer* vie_capture = ViECapturer::CreateViECapture(
- newcapture_id, engine_id_, NULL, 0, *module_process_thread_);
- if (!vie_capture) {
- ReturnCaptureId(newcapture_id);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s: Could not create capture module for external capture.",
- __FUNCTION__);
- return kViECaptureDeviceUnknownError;
- }
-
- if (vie_frame_provider_map_.Insert(newcapture_id, vie_capture) != 0) {
- ReturnCaptureId(newcapture_id);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s: Could not insert capture module for external capture.",
- __FUNCTION__);
- return kViECaptureDeviceUnknownError;
- }
- capture_id = newcapture_id;
- external_capture = vie_capture;
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s, capture_id: %d)", __FUNCTION__, capture_id);
- return 0;
-}
-
-int ViEInputManager::CreateFilePlayer(const WebRtc_Word8* file_nameUTF8,
- const bool loop,
- const webrtc::FileFormats file_format,
- VoiceEngine* voe_ptr, int& file_id) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s(device_unique_id: %s)", __FUNCTION__, file_nameUTF8);
-
- CriticalSectionScoped cs(map_cs_.get());
- int new_file_id = 0;
- if (GetFreeFileId(new_file_id) == false) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s: Maximum supported number of file players already in use",
- __FUNCTION__);
- return kViEFileMaxNoOfFilesOpened;
- }
-
- ViEFilePlayer* vie_file_player = ViEFilePlayer::CreateViEFilePlayer(
- new_file_id, engine_id_, file_nameUTF8, loop, file_format, *this,
- voe_ptr);
- if (!vie_file_player) {
- ReturnFileId(new_file_id);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s: Could not open file %s for playback", __FUNCTION__,
- file_nameUTF8);
- return kViEFileUnknownError;
- }
-
- if (vie_frame_provider_map_.Insert(new_file_id, vie_file_player) != 0) {
- ReturnCaptureId(new_file_id);
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s: Could not insert file player for %s", __FUNCTION__,
- file_nameUTF8);
- delete vie_file_player;
- return kViEFileUnknownError;
- }
-
- file_id = new_file_id;
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s(filename: %s, file_id: %d)", __FUNCTION__, file_nameUTF8,
- new_file_id);
- return 0;
-}
-
-int ViEInputManager::DestroyFilePlayer(int file_id) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s(file_id: %d)", __FUNCTION__, file_id);
-
- ViEFilePlayer* vie_file_player = NULL;
- {
- // We need exclusive access to the object to delete it.
- // Take this write lock first since the read lock is taken before map_cs_.
- ViEManagerWriteScoped wl(*this);
-
- CriticalSectionScoped cs(map_cs_.get());
- vie_file_player = ViEFilePlayerPtr(file_id);
- if (!vie_file_player) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s(file_id: %d) - No such file player", __FUNCTION__,
- file_id);
- return -1;
- }
- int num_callbacks = vie_file_player->NumberOfRegisteredFrameCallbacks();
- if (num_callbacks > 0) {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo,
- ViEId(engine_id_), "%s(file_id: %d) - %u registered "
- "callbacks when destroying file player", __FUNCTION__,
- file_id, num_callbacks);
- }
- vie_frame_provider_map_.Erase(file_id);
- ReturnFileId(file_id);
- // Leave cs before deleting the file object. This is because deleting the
- // object might cause deletions of renderers so we prefer to not have a lock
- // at that time.
- }
- delete vie_file_player;
- return 0;
-}
-
-bool ViEInputManager::GetFreeCaptureId(int& freecapture_id) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_), "%s",
- __FUNCTION__);
- for (int id = 0; id < kViEMaxCaptureDevices; id++) {
- if (free_capture_device_id_[id]) {
- // We found a free capture device id.
- free_capture_device_id_[id] = false;
- freecapture_id = id + kViECaptureIdBase;
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s: new id: %d", __FUNCTION__, freecapture_id);
- return true;
- }
- }
- return false;
-}
-
-void ViEInputManager::ReturnCaptureId(int capture_id) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s(%d)", __FUNCTION__, capture_id);
- CriticalSectionScoped cs(map_cs_.get());
- if (capture_id >= kViECaptureIdBase &&
- capture_id < kViEMaxCaptureDevices + kViECaptureIdBase) {
- free_capture_device_id_[capture_id - kViECaptureIdBase] = true;
- }
- return;
-}
-
-bool ViEInputManager::GetFreeFileId(int& free_file_id) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_), "%s",
- __FUNCTION__);
-
- for (int id = 0; id < kViEMaxFilePlayers; id++) {
- if (free_file_id_[id]) {
- // We found a free capture device id.
- free_file_id_[id] = false;
- free_file_id = id + kViEFileIdBase;
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s: new id: %d", __FUNCTION__, free_file_id);
- return true;
- }
- }
- return false;
-}
-
-void ViEInputManager::ReturnFileId(int file_id) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s(%d)", __FUNCTION__, file_id);
-
- CriticalSectionScoped cs(map_cs_.get());
- if (file_id >= kViEFileIdBase &&
- file_id < kViEMaxFilePlayers + kViEFileIdBase) {
- free_file_id_[file_id - kViEFileIdBase] = true;
- }
- return;
-}
-
-ViEFrameProviderBase* ViEInputManager::ViEFrameProvider(
- const ViEFrameCallback* capture_observer) const {
- assert(capture_observer);
- CriticalSectionScoped cs(map_cs_.get());
-
- for (MapItem* provider_item = vie_frame_provider_map_.First(); provider_item
- != NULL; provider_item = vie_frame_provider_map_.Next(provider_item)) {
- ViEFrameProviderBase* vie_frame_provider =
- static_cast<ViEFrameProviderBase*>(provider_item->GetItem());
- assert(vie_frame_provider != NULL);
-
- if (vie_frame_provider->IsFrameCallbackRegistered(capture_observer)) {
- // We found it.
- return vie_frame_provider;
- }
- }
- // No capture device set for this channel.
- return NULL;
-}
-
-ViEFrameProviderBase* ViEInputManager::ViEFrameProvider(int provider_id) const {
- CriticalSectionScoped cs(map_cs_.get());
- MapItem* map_item = vie_frame_provider_map_.Find(provider_id);
- if (!map_item) {
- return NULL;
- }
- ViEFrameProviderBase* vie_frame_provider =
- static_cast<ViEFrameProviderBase*>(map_item->GetItem());
- return vie_frame_provider;
-}
-
-ViECapturer* ViEInputManager::ViECapturePtr(int capture_id) const {
- if (!(capture_id >= kViECaptureIdBase &&
- capture_id <= kViECaptureIdBase + kViEMaxCaptureDevices))
- return NULL;
-
- CriticalSectionScoped cs(map_cs_.get());
- MapItem* map_item = vie_frame_provider_map_.Find(capture_id);
- if (!map_item) {
- return NULL;
- }
- ViECapturer* vie_capture = static_cast<ViECapturer*>(map_item->GetItem());
- return vie_capture;
-}
-
-void ViEInputManager::GetViECaptures(MapWrapper& vie_capture_map) {
- CriticalSectionScoped cs(map_cs_.get());
-
- if (vie_frame_provider_map_.Size() == 0) {
- return;
- }
- // Add all items to the map.
- for (MapItem* item = vie_frame_provider_map_.First(); item != NULL;
- item = vie_frame_provider_map_.Next(item)) {
- vie_capture_map.Insert(item->GetId(), item->GetItem());
- }
- return;
-}
-
-ViEFilePlayer* ViEInputManager::ViEFilePlayerPtr(int file_id) const {
- if (file_id < kViEFileIdBase || file_id > kViEFileIdMax) {
- return NULL;
- }
- CriticalSectionScoped cs(map_cs_.get());
- MapItem* map_item = vie_frame_provider_map_.Find(file_id);
- if (!map_item) {
- return NULL;
- }
- ViEFilePlayer* vie_file_player =
- static_cast<ViEFilePlayer*>(map_item->GetItem());
- return vie_file_player;
-}
-
-ViEInputManagerScoped::ViEInputManagerScoped(
- const ViEInputManager& vie_input_manager)
- : ViEManagerScopedBase(vie_input_manager) {
-}
-
-ViECapturer* ViEInputManagerScoped::Capture(int capture_id) const {
- return static_cast<const ViEInputManager*>(vie_manager_)->ViECapturePtr(
- capture_id);
-}
-
-ViEFrameProviderBase* ViEInputManagerScoped::FrameProvider(
- const ViEFrameCallback* capture_observer) const {
- return static_cast<const ViEInputManager*>(vie_manager_)->ViEFrameProvider(
- capture_observer);
-}
-
-ViEFrameProviderBase* ViEInputManagerScoped::FrameProvider(
- int provider_id) const {
- return static_cast<const ViEInputManager*>(vie_manager_)->ViEFrameProvider(
- provider_id);
-}
-
-ViEFilePlayer* ViEInputManagerScoped::FilePlayer(int file_id) const {
- return static_cast<const ViEInputManager*>(vie_manager_)->ViEFilePlayerPtr(
- file_id);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_input_manager.h b/trunk/src/video_engine/vie_input_manager.h
deleted file mode 100644
index c70a2ee..0000000
--- a/trunk/src/video_engine/vie_input_manager.h
+++ /dev/null
@@ -1,143 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_INPUT_MANAGER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_INPUT_MANAGER_H_
-
-#include "modules/video_capture/main/interface/video_capture.h"
-#include "system_wrappers/interface/map_wrapper.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-#include "typedefs.h"
-#include "video_engine/include/vie_capture.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_frame_provider_base.h"
-#include "video_engine/vie_manager_base.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class ProcessThread;
-class RWLockWrapper;
-class ViECapturer;
-class ViEExternalCapture;
-class ViEFilePlayer;
-class VoiceEngine;
-
-class ViEInputManager : private ViEManagerBase {
- friend class ViEInputManagerScoped;
- public:
- explicit ViEInputManager(int engine_id);
- ~ViEInputManager();
-
- void SetModuleProcessThread(ProcessThread& module_process_thread);
-
- // Returns number of capture devices.
- int NumberOfCaptureDevices();
-
- // Gets name and id for a capture device.
- int GetDeviceName(WebRtc_UWord32 device_number,
- WebRtc_UWord8* device_nameUTF8,
- WebRtc_UWord32 device_name_length,
- WebRtc_UWord8* device_unique_idUTF8,
- WebRtc_UWord32 device_unique_idUTF8Length);
-
- // Returns the number of capture capabilities for a specified device.
- int NumberOfCaptureCapabilities(const WebRtc_UWord8* device_unique_idUTF8);
-
- // Gets a specific capability for a capture device.
- int GetCaptureCapability(const WebRtc_UWord8* device_unique_idUTF8,
- const WebRtc_UWord32 device_capability_number,
- CaptureCapability& capability);
-
- // Show OS specific Capture settings.
- int DisplayCaptureSettingsDialogBox(const WebRtc_UWord8* device_unique_idUTF8,
- const WebRtc_UWord8* dialog_titleUTF8,
- void* parent_window,
- WebRtc_UWord32 positionX,
- WebRtc_UWord32 positionY);
- int GetOrientation(const WebRtc_UWord8* device_unique_idUTF8,
- RotateCapturedFrame& orientation);
-
- // Creates a capture module for the specified capture device and assigns
- // a capture device id for the device.
- // Return zero on success, ViEError on failure.
- int CreateCaptureDevice(const WebRtc_UWord8* device_unique_idUTF8,
- const WebRtc_UWord32 device_unique_idUTF8Length,
- int& capture_id);
- int CreateCaptureDevice(VideoCaptureModule& capture_module,
- int& capture_id);
- int CreateExternalCaptureDevice(ViEExternalCapture*& external_capture,
- int& capture_id);
- int DestroyCaptureDevice(int capture_id);
-
- int CreateFilePlayer(const WebRtc_Word8* file_nameUTF8, const bool loop,
- const FileFormats file_format,
- VoiceEngine* voe_ptr,
- int& file_id);
- int DestroyFilePlayer(int file_id);
-
- private:
- // Gets and allocates a free capture device id. Assumed protected by caller.
- bool GetFreeCaptureId(int& freecapture_id);
-
- // Frees a capture id assigned in GetFreeCaptureId.
- void ReturnCaptureId(int capture_id);
-
- // Gets and allocates a free file id. Assumed protected by caller.
- bool GetFreeFileId(int& free_file_id);
-
- // Frees a file id assigned in GetFreeFileId.
- void ReturnFileId(int file_id);
-
- // Gets the ViEFrameProvider for this capture observer.
- ViEFrameProviderBase* ViEFrameProvider(
- const ViEFrameCallback* capture_observer) const;
-
- // Gets the ViEFrameProvider for this capture observer.
- ViEFrameProviderBase* ViEFrameProvider(int provider_id) const;
-
- // Gets the ViECapturer for the capture device id.
- ViECapturer* ViECapturePtr(int capture_id) const;
-
- // Gets the the entire map with GetViECaptures.
- void GetViECaptures(MapWrapper& vie_capture_map);
-
- // Gets the ViEFilePlayer for this file_id.
- ViEFilePlayer* ViEFilePlayerPtr(int file_id) const;
-
- int engine_id_;
- scoped_ptr<CriticalSectionWrapper> map_cs_;
- MapWrapper vie_frame_provider_map_;
-
- // Capture devices.
- VideoCaptureModule::DeviceInfo* capture_device_info_;
- int free_capture_device_id_[kViEMaxCaptureDevices];
-
- // File Players.
- int free_file_id_[kViEMaxFilePlayers];
-
- ProcessThread* module_process_thread_;
-};
-
-// Provides protected access to ViEInputManater.
-class ViEInputManagerScoped: private ViEManagerScopedBase {
- public:
- explicit ViEInputManagerScoped(const ViEInputManager& vie_input_manager);
-
- ViECapturer* Capture(int capture_id) const;
- ViEFilePlayer* FilePlayer(int file_id) const;
- ViEFrameProviderBase* FrameProvider(int provider_id) const;
- ViEFrameProviderBase* FrameProvider(const ViEFrameCallback*
- capture_observer) const;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_INPUT_MANAGER_H_
diff --git a/trunk/src/video_engine/vie_manager_base.cc b/trunk/src/video_engine/vie_manager_base.cc
deleted file mode 100644
index 7e6695c..0000000
--- a/trunk/src/video_engine/vie_manager_base.cc
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "assert.h"
-
-#include "system_wrappers/interface/rw_lock_wrapper.h"
-#include "video_engine/vie_manager_base.h"
-
-namespace webrtc {
-
-ViEManagerBase::ViEManagerBase()
- : instance_rwlock_(*RWLockWrapper::CreateRWLock()) {
-}
-
-ViEManagerBase::~ViEManagerBase() {
- delete &instance_rwlock_;
-}
-
-void ViEManagerBase::ReadLockManager() const {
- instance_rwlock_.AcquireLockShared();
-}
-
-void ViEManagerBase::ReleaseLockManager() const {
- instance_rwlock_.ReleaseLockShared();
-}
-
-void ViEManagerBase::WriteLockManager() {
- instance_rwlock_.AcquireLockExclusive();
-}
-
-void ViEManagerBase::ReleaseWriteLockManager() {
- instance_rwlock_.ReleaseLockExclusive();
-}
-
-ViEManagerScopedBase::ViEManagerScopedBase(const ViEManagerBase& ViEManagerBase)
- : vie_manager_(&ViEManagerBase),
- ref_count_(0) {
- vie_manager_->ReadLockManager();
-}
-
-ViEManagerScopedBase::~ViEManagerScopedBase() {
- assert(ref_count_ == 0);
- vie_manager_->ReleaseLockManager();
-}
-
-ViEManagerWriteScoped::ViEManagerWriteScoped(ViEManagerBase& vie_manager)
- : vie_manager_(&vie_manager) {
- vie_manager_->WriteLockManager();
-}
-
-ViEManagerWriteScoped::~ViEManagerWriteScoped() {
- vie_manager_->ReleaseWriteLockManager();
-}
-
-ViEManagedItemScopedBase::ViEManagedItemScopedBase(
- ViEManagerScopedBase& vie_scoped_manager)
- : vie_scoped_manager_(vie_scoped_manager) {
- vie_scoped_manager_.ref_count_++;
-}
-
-ViEManagedItemScopedBase::~ViEManagedItemScopedBase() {
- vie_scoped_manager_.ref_count_--;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_manager_base.h b/trunk/src/video_engine/vie_manager_base.h
deleted file mode 100644
index 44cf40f..0000000
--- a/trunk/src/video_engine/vie_manager_base.h
+++ /dev/null
@@ -1,74 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_MANAGER_BASE_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_MANAGER_BASE_H_
-
-namespace webrtc {
-
-class RWLockWrapper;
-
-class ViEManagerBase {
- friend class ViEManagerScopedBase;
- friend class ViEManagedItemScopedBase;
- friend class ViEManagerWriteScoped;
- public:
- ViEManagerBase();
- ~ViEManagerBase();
-
- private:
- // Exclusive lock, used by ViEManagerWriteScoped
- void WriteLockManager();
-
- // Releases exclusive lock, used by ViEManagerWriteScoped.
- void ReleaseWriteLockManager();
-
- // Increases lock count, used by ViEManagerScopedBase.
- void ReadLockManager() const;
-
- // Releases the lock count, used by ViEManagerScopedBase.
- void ReleaseLockManager() const;
-
- RWLockWrapper& instance_rwlock_;
-};
-
-class ViEManagerWriteScoped {
- public:
- explicit ViEManagerWriteScoped(ViEManagerBase& vie_manager);
- ~ViEManagerWriteScoped();
-
- private:
- ViEManagerBase* vie_manager_;
-};
-
-class ViEManagerScopedBase {
- friend class ViEManagedItemScopedBase;
- public:
- explicit ViEManagerScopedBase(const ViEManagerBase& vie_manager);
- ~ViEManagerScopedBase();
-
- protected:
- const ViEManagerBase* vie_manager_;
-
- private:
- int ref_count_;
-};
-
-class ViEManagedItemScopedBase {
- public:
- explicit ViEManagedItemScopedBase(ViEManagerScopedBase& vie_scoped_manager);
- ~ViEManagedItemScopedBase();
- protected:
- ViEManagerScopedBase& vie_scoped_manager_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_MANAGER_BASE_H_
diff --git a/trunk/src/video_engine/vie_network_impl.cc b/trunk/src/video_engine/vie_network_impl.cc
deleted file mode 100644
index b4bb0e9..0000000
--- a/trunk/src/video_engine/vie_network_impl.cc
+++ /dev/null
@@ -1,798 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_network_impl.h"
-
-#include <stdio.h>
-#if (defined(WIN32_) || defined(WIN64_))
-#include <qos.h>
-#endif
-
-#include "engine_configurations.h"
-#include "system_wrappers/interface/trace.h"
-#include "video_engine/include/vie_errors.h"
-#include "video_engine/vie_channel.h"
-#include "video_engine/vie_channel_manager.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_encoder.h"
-#include "video_engine/vie_impl.h"
-#include "video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-ViENetwork* ViENetwork::GetInterface(VideoEngine* video_engine) {
-#ifdef WEBRTC_VIDEO_ENGINE_NETWORK_API
- if (!video_engine) {
- return NULL;
- }
- VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
- ViENetworkImpl* vie_networkImpl = vie_impl;
- // Increase ref count.
- (*vie_networkImpl)++;
- return vie_networkImpl;
-#else
- return NULL;
-#endif
-}
-
-int ViENetworkImpl::Release() {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
- "ViENetwork::Release()");
- // Decrease ref count.
- (*this)--;
-
- WebRtc_Word32 ref_count = GetCount();
- if (ref_count < 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
- "ViENetwork release too many times");
- shared_data_->SetLastError(kViEAPIDoesNotExist);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
- "ViENetwork reference count: %d", ref_count);
- return ref_count;
-}
-
-ViENetworkImpl::ViENetworkImpl(ViESharedData* shared_data)
- : shared_data_(shared_data) {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViENetworkImpl::ViENetworkImpl() Ctor");
-}
-
-ViENetworkImpl::~ViENetworkImpl() {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViENetworkImpl::~ViENetworkImpl() Dtor");
-}
-
-int ViENetworkImpl::SetLocalReceiver(const int video_channel,
- const unsigned short rtp_port,
- const unsigned short rtcp_port,
- const char* ip_address) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, rtp_port: %u, rtcp_port: %u, ip_address: %s)",
- __FUNCTION__, video_channel, rtp_port, rtcp_port, ip_address);
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- // The channel doesn't exists.
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
-
- if (vie_channel->Receiving()) {
- shared_data_->SetLastError(kViENetworkAlreadyReceiving);
- return -1;
- }
- if (vie_channel->SetLocalReceiver(rtp_port, rtcp_port, ip_address) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::GetLocalReceiver(const int video_channel,
- unsigned short& rtp_port,
- unsigned short& rtcp_port,
- char* ip_address) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetLocalReceiver(rtp_port, rtcp_port, ip_address) != 0) {
- shared_data_->SetLastError(kViENetworkLocalReceiverNotSet);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::SetSendDestination(const int video_channel,
- const char* ip_address,
- const unsigned short rtp_port,
- const unsigned short rtcp_port,
- const unsigned short source_rtp_port,
- const unsigned short source_rtcp_port) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, ip_address: %s, rtp_port: %u, rtcp_port: %u, "
- "sourceRtpPort: %u, source_rtcp_port: %u)",
- __FUNCTION__, video_channel, ip_address, rtp_port, rtcp_port,
- source_rtp_port, source_rtcp_port);
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
-
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s Channel doesn't exist", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->Sending()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s Channel already sending.", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkAlreadySending);
- return -1;
- }
- if (vie_channel->SetSendDestination(ip_address, rtp_port, rtcp_port,
- source_rtp_port,
- source_rtcp_port) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::GetSendDestination(const int video_channel,
- char* ip_address,
- unsigned short& rtp_port,
- unsigned short& rtcp_port,
- unsigned short& source_rtp_port,
- unsigned short& source_rtcp_port) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetSendDestination(ip_address, rtp_port, rtcp_port,
- source_rtp_port,
- source_rtcp_port) != 0) {
- shared_data_->SetLastError(kViENetworkDestinationNotSet);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::RegisterSendTransport(const int video_channel,
- Transport& transport) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s Channel doesn't exist", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->Sending()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s Channel already sending.", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkAlreadySending);
- return -1;
- }
- if (vie_channel->RegisterSendTransport(transport) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::DeregisterSendTransport(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s Channel doesn't exist", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->Sending()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s Channel already sending", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkAlreadySending);
- return -1;
- }
- if (vie_channel->DeregisterSendTransport() != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::ReceivedRTPPacket(const int video_channel, const void* data,
- const int length) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, data: -, length: %d)", __FUNCTION__,
- video_channel, length);
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- // The channel doesn't exists
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- return vie_channel->ReceivedRTPPacket(data, length);
-}
-
-int ViENetworkImpl::ReceivedRTCPPacket(const int video_channel,
- const void* data, const int length) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, data: -, length: %d)", __FUNCTION__,
- video_channel, length);
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- return vie_channel->ReceivedRTCPPacket(data, length);
-}
-
-int ViENetworkImpl::GetSourceInfo(const int video_channel,
- unsigned short& rtp_port,
- unsigned short& rtcp_port, char* ip_address,
- unsigned int ip_address_length) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetSourceInfo(rtp_port, rtcp_port, ip_address,
- ip_address_length) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::GetLocalIP(char ip_address[64], bool ipv6) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s( ip_address, ipV6: %d)", __FUNCTION__, ipv6);
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
-
- if (!ip_address) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: No argument", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkInvalidArgument);
- return -1;
- }
-
- WebRtc_UWord8 num_socket_threads = 1;
- UdpTransport* socket_transport = UdpTransport::Create(
- ViEModuleId(shared_data_->instance_id(), -1), num_socket_threads);
-
- if (!socket_transport) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Could not create socket module", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
-
- char local_ip_address[64];
- if (ipv6) {
- char local_ip[16];
- if (socket_transport->LocalHostAddressIPV6(local_ip) != 0) {
- UdpTransport::Destroy(socket_transport);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Could not get local IP", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- // Convert 128-bit address to character string (a:b:c:d:e:f:g:h).
- sprintf(local_ip_address,
- "%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:"
- "%.2x%.2x",
- local_ip[0], local_ip[1], local_ip[2], local_ip[3], local_ip[4],
- local_ip[5], local_ip[6], local_ip[7], local_ip[8], local_ip[9],
- local_ip[10], local_ip[11], local_ip[12], local_ip[13],
- local_ip[14], local_ip[15]);
- } else {
- WebRtc_UWord32 local_ip = 0;
- if (socket_transport->LocalHostAddress(local_ip) != 0) {
- UdpTransport::Destroy(socket_transport);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: Could not get local IP", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- // Convert 32-bit address to character string (x.y.z.w).
- sprintf(local_ip_address, "%d.%d.%d.%d",
- static_cast<int>((local_ip >> 24) & 0x0ff),
- static_cast<int>((local_ip >> 16) & 0x0ff),
- static_cast<int>((local_ip >> 8) & 0x0ff),
- static_cast<int>(local_ip & 0x0ff));
- }
- strcpy(ip_address, local_ip_address);
- UdpTransport::Destroy(socket_transport);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: local ip = %s", __FUNCTION__, local_ip_address);
- return 0;
-#else
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: not available for external transport", __FUNCTION__);
-
- return -1;
-#endif
-}
-
-int ViENetworkImpl::EnableIPv6(int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->EnableIPv6() != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-bool ViENetworkImpl::IsIPv6Enabled(int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return false;
- }
- return vie_channel->IsIPv6Enabled();
-}
-
-int ViENetworkImpl::SetSourceFilter(const int video_channel,
- const unsigned short rtp_port,
- const unsigned short rtcp_port,
- const char* ip_address) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, rtp_port: %u, rtcp_port: %u, ip_address: %s)",
- __FUNCTION__, video_channel, rtp_port, rtcp_port, ip_address);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->SetSourceFilter(rtp_port, rtcp_port, ip_address) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::GetSourceFilter(const int video_channel,
- unsigned short& rtp_port,
- unsigned short& rtcp_port,
- char* ip_address) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetSourceFilter(rtp_port, rtcp_port, ip_address) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::SetSendToS(const int video_channel, const int DSCP,
- const bool use_set_sockOpt = false) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, DSCP: %d, use_set_sockOpt: %d)", __FUNCTION__,
- video_channel, DSCP, use_set_sockOpt);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
-
-#if defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
- WEBRTC_TRACE(kTraceInfo, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- " force use_set_sockopt=true since there is no alternative"
- " implementation");
- if (vie_channel->SetToS(DSCP, true) != 0) {
-#else
- if (vie_channel->SetToS(DSCP, use_set_sockOpt) != 0) {
-#endif
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::GetSendToS(const int video_channel, int& DSCP,
- bool& use_set_sockOpt) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetToS((WebRtc_Word32&) DSCP, use_set_sockOpt) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::SetSendGQoS(const int video_channel, const bool enable,
- const int service_type,
- const int overrideDSCP) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, enable: %d, service_type: %d, "
- "overrideDSCP: %d)", __FUNCTION__, video_channel, enable,
- service_type, overrideDSCP);
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
-
-#if (defined(WIN32_) || defined(WIN64_))
- // Sanity check. We might crash if testing and relying on an OS socket error.
- if (enable &&
- (service_type != SERVICETYPE_BESTEFFORT) &&
- (service_type != SERVICETYPE_CONTROLLEDLOAD) &&
- (service_type != SERVICETYPE_GUARANTEED) &&
- (service_type != SERVICETYPE_QUALITATIVE)) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: service type %d not supported", __FUNCTION__,
- video_channel, service_type);
- shared_data_->SetLastError(kViENetworkServiceTypeNotSupported);
- return -1;
- }
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- VideoCodec video_codec;
- if (vie_encoder->GetEncoder(video_codec) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Could not get max bitrate for the channel",
- __FUNCTION__);
- shared_data_->SetLastError(kViENetworkSendCodecNotSet);
- return -1;
- }
- if (vie_channel->SetSendGQoS(enable, service_type, video_codec.maxBitrate,
- overrideDSCP) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-#else
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Not supported", __FUNCTION__);
- shared_data_->SetLastError(kViENetworkNotSupported);
- return -1;
-#endif
-}
-
-int ViENetworkImpl::GetSendGQoS(const int video_channel, bool& enabled,
- int& service_type, int& overrideDSCP) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetSendGQoS(enabled, service_type, overrideDSCP) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::SetMTU(int video_channel, unsigned int mtu) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, mtu: %u)", __FUNCTION__, video_channel, mtu);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->SetMTU(mtu) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::SetPacketTimeoutNotification(const int video_channel,
- bool enable,
- int timeout_seconds) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, enable: %d, timeout_seconds: %u)",
- __FUNCTION__, video_channel, enable, timeout_seconds);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->SetPacketTimeoutNotification(enable,
- timeout_seconds) != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::RegisterObserver(const int video_channel,
- ViENetworkObserver& observer) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->RegisterNetworkObserver(&observer) != 0) {
- shared_data_->SetLastError(kViENetworkObserverAlreadyRegistered);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::DeregisterObserver(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (!vie_channel->NetworkObserverRegistered()) {
- shared_data_->SetLastError(kViENetworkObserverNotRegistered);
- return -1;
- }
- return vie_channel->RegisterNetworkObserver(NULL);
-}
-
-int ViENetworkImpl::SetPeriodicDeadOrAliveStatus(
- const int video_channel,
- bool enable,
- unsigned int sample_time_seconds) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, enable: %d, sample_time_seconds: %ul)",
- __FUNCTION__, video_channel, enable, sample_time_seconds);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (!vie_channel->NetworkObserverRegistered()) {
- shared_data_->SetLastError(kViENetworkObserverNotRegistered);
- return -1;
- }
- if (vie_channel->SetPeriodicDeadOrAliveStatus(enable, sample_time_seconds)
- != 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViENetworkImpl::SendUDPPacket(const int video_channel, const void* data,
- const unsigned int length,
- int& transmitted_bytes,
- bool use_rtcp_socket = false) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, data: -, length: %d, transmitter_bytes: -, "
- "useRtcpSocket: %d)", __FUNCTION__, video_channel, length,
- use_rtcp_socket);
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "Channel doesn't exist");
- shared_data_->SetLastError(kViENetworkInvalidChannelId);
- return -1;
- }
- if (vie_channel->SendUDPPacket((const WebRtc_Word8*) data, length,
- (WebRtc_Word32&) transmitted_bytes,
- use_rtcp_socket) < 0) {
- shared_data_->SetLastError(kViENetworkUnknownError);
- return -1;
- }
- return 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_network_impl.h b/trunk/src/video_engine/vie_network_impl.h
deleted file mode 100644
index 8eff523..0000000
--- a/trunk/src/video_engine/vie_network_impl.h
+++ /dev/null
@@ -1,114 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_NETWORK_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_NETWORK_IMPL_H_
-
-#include "typedefs.h"
-#include "video_engine/include/vie_network.h"
-#include "video_engine/vie_ref_count.h"
-
-namespace webrtc {
-
-class ViESharedData;
-
-class ViENetworkImpl
- : public ViENetwork,
- public ViERefCount {
- public:
- // Implements ViENetwork.
- virtual int Release();
- virtual int SetLocalReceiver(const int video_channel,
- const unsigned short rtp_port,
- const unsigned short rtcp_port,
- const char* ip_address);
- virtual int GetLocalReceiver(const int video_channel,
- unsigned short& rtp_port,
- unsigned short& rtcp_port,
- char* ip_address);
- virtual int SetSendDestination(const int video_channel,
- const char* ip_address,
- const unsigned short rtp_port,
- const unsigned short rtcp_port,
- const unsigned short source_rtp_port,
- const unsigned short source_rtcp_port);
- virtual int GetSendDestination(const int video_channel,
- char* ip_address,
- unsigned short& rtp_port,
- unsigned short& rtcp_port,
- unsigned short& source_rtp_port,
- unsigned short& source_rtcp_port);
- virtual int RegisterSendTransport(const int video_channel,
- Transport& transport);
- virtual int DeregisterSendTransport(const int video_channel);
- virtual int ReceivedRTPPacket(const int video_channel,
- const void* data,
- const int length);
- virtual int ReceivedRTCPPacket(const int video_channel,
- const void* data,
- const int length);
- virtual int GetSourceInfo(const int video_channel,
- unsigned short& rtp_port,
- unsigned short& rtcp_port,
- char* ip_address,
- unsigned int ip_address_length);
- virtual int GetLocalIP(char ip_address[64], bool ipv6);
- virtual int EnableIPv6(int video_channel);
- virtual bool IsIPv6Enabled(int video_channel);
- virtual int SetSourceFilter(const int video_channel,
- const unsigned short rtp_port,
- const unsigned short rtcp_port,
- const char* ip_address);
- virtual int GetSourceFilter(const int video_channel,
- unsigned short& rtp_port,
- unsigned short& rtcp_port,
- char* ip_address);
- virtual int SetSendToS(const int video_channel,
- const int DSCP,
- const bool use_set_sockOpt);
- virtual int GetSendToS(const int video_channel,
- int& DSCP,
- bool& use_set_sockOpt);
- virtual int SetSendGQoS(const int video_channel,
- const bool enable,
- const int service_type,
- const int overrideDSCP);
- virtual int GetSendGQoS(const int video_channel,
- bool& enabled,
- int& service_type,
- int& overrideDSCP);
- virtual int SetMTU(int video_channel, unsigned int mtu);
- virtual int SetPacketTimeoutNotification(const int video_channel,
- bool enable,
- int timeout_seconds);
- virtual int RegisterObserver(const int video_channel,
- ViENetworkObserver& observer);
- virtual int DeregisterObserver(const int video_channel);
- virtual int SetPeriodicDeadOrAliveStatus(
- const int video_channel,
- const bool enable,
- const unsigned int sample_time_seconds);
- virtual int SendUDPPacket(const int video_channel,
- const void* data,
- const unsigned int length,
- int& transmitted_bytes,
- bool use_rtcp_socket);
-
- protected:
- ViENetworkImpl(ViESharedData* shared_data);
- virtual ~ViENetworkImpl();
-
- private:
- ViESharedData* shared_data_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_NETWORK_IMPL_H_
diff --git a/trunk/src/video_engine/vie_performance_monitor.cc b/trunk/src/video_engine/vie_performance_monitor.cc
deleted file mode 100644
index 43ae056..0000000
--- a/trunk/src/video_engine/vie_performance_monitor.cc
+++ /dev/null
@@ -1,142 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vie_performance_monitor.h"
-
-#include "cpu_wrapper.h"
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "tick_util.h"
-#include "trace.h"
-#include "vie_base.h"
-#include "vie_defines.h"
-
-namespace webrtc {
-
-enum { kVieMonitorPeriodMs = 975 };
-enum { kVieCpuStartValue = 75 };
-
-ViEPerformanceMonitor::ViEPerformanceMonitor(int engine_id)
- : engine_id_(engine_id),
- pointer_cs_(CriticalSectionWrapper::CreateCriticalSection()),
- monitor_thread_(NULL),
- monitor_event_(*EventWrapper::Create()),
- average_application_cpu_(kVieCpuStartValue),
- average_system_cpu_(kVieCpuStartValue),
- cpu_(NULL),
- vie_base_observer_(NULL) {
-}
-
-ViEPerformanceMonitor::~ViEPerformanceMonitor() {
- Terminate();
- delete pointer_cs_;
- delete &monitor_event_;
-}
-
-int ViEPerformanceMonitor::Init(ViEBaseObserver* vie_base_observer) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s", __FUNCTION__);
-
- CriticalSectionScoped cs(pointer_cs_);
- if (!vie_base_observer || vie_base_observer_) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s: Bad input argument or observer already set",
- __FUNCTION__);
- return -1;
- }
-
- cpu_ = CpuWrapper::CreateCpu();
- if (cpu_ == NULL) {
- // Performance monitoring not supported
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo,
- ViEId(engine_id_), "%s: Not supported", __FUNCTION__);
- return 0;
- }
-
- if (monitor_thread_ == NULL) {
- monitor_event_.StartTimer(true, kVieMonitorPeriodMs);
- monitor_thread_ = ThreadWrapper::CreateThread(ViEMonitorThreadFunction,
- this, kNormalPriority,
- "ViEPerformanceMonitor");
- unsigned int t_id = 0;
- if (monitor_thread_->Start(t_id)) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s: Performance monitor thread started %u",
- __FUNCTION__, t_id);
- } else {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s: Could not start performance monitor", __FUNCTION__);
- monitor_event_.StopTimer();
- return -1;
- }
- }
- vie_base_observer_ = vie_base_observer;
- return 0;
-}
-
-void ViEPerformanceMonitor::Terminate() {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, ViEId(engine_id_),
- "%s", __FUNCTION__);
-
- pointer_cs_->Enter();
- if (!vie_base_observer_) {
- pointer_cs_->Leave();
- return;
- }
-
- vie_base_observer_ = NULL;
- monitor_event_.StopTimer();
- if (monitor_thread_) {
- ThreadWrapper* tmp_thread = monitor_thread_;
- monitor_thread_ = NULL;
- monitor_event_.Set();
- pointer_cs_->Leave();
- if (tmp_thread->Stop()) {
- pointer_cs_->Enter();
- delete tmp_thread;
- tmp_thread = NULL;
- delete cpu_;
- }
- cpu_ = NULL;
- }
- pointer_cs_->Leave();
-}
-
-bool ViEPerformanceMonitor::ViEBaseObserverRegistered() {
- CriticalSectionScoped cs(pointer_cs_);
- return vie_base_observer_ != NULL;
-}
-
-bool ViEPerformanceMonitor::ViEMonitorThreadFunction(void* obj) {
- return static_cast<ViEPerformanceMonitor*>(obj)->ViEMonitorProcess();
-}
-
-bool ViEPerformanceMonitor::ViEMonitorProcess() {
- // Periodically triggered with time KViEMonitorPeriodMs
- monitor_event_.Wait(kVieMonitorPeriodMs);
- if (monitor_thread_ == NULL) {
- // Thread removed, exit
- return false;
- }
-
- CriticalSectionScoped cs(pointer_cs_);
- if (cpu_) {
- int cpu_load = cpu_->CpuUsage();
- if (cpu_load > 75) {
- if (vie_base_observer_) {
- vie_base_observer_->PerformanceAlarm(cpu_load);
- }
- }
- }
- return true;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_performance_monitor.h b/trunk/src/video_engine/vie_performance_monitor.h
deleted file mode 100644
index c3ea931..0000000
--- a/trunk/src/video_engine/vie_performance_monitor.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// ViEPerformanceMonitor is used to check the current CPU usage and triggers a
-// callback when getting over a specified threshold.
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_PERFORMANCE_MONITOR_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_PERFORMANCE_MONITOR_H_
-
-#include "system_wrappers/interface/scoped_ptr.h"
-#include "typedefs.h"
-#include "vie_defines.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class CpuWrapper;
-class EventWrapper;
-class ThreadWrapper;
-class ViEBaseObserver;
-
-class ViEPerformanceMonitor {
- public:
- explicit ViEPerformanceMonitor(int engine_id);
- ~ViEPerformanceMonitor();
-
- int Init(ViEBaseObserver* vie_base_observer);
- void Terminate();
- bool ViEBaseObserverRegistered();
-
- protected:
- static bool ViEMonitorThreadFunction(void* obj);
- bool ViEMonitorProcess();
-
- private:
- const int engine_id_;
- // TODO(mfldoman) Make this one scoped_ptr.
- CriticalSectionWrapper* pointer_cs_;
- ThreadWrapper* monitor_thread_;
- EventWrapper& monitor_event_;
- int average_application_cpu_;
- int average_system_cpu_;
- CpuWrapper* cpu_;
- ViEBaseObserver* vie_base_observer_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_PERFORMANCE_MONITOR_H_
diff --git a/trunk/src/video_engine/vie_receiver.cc b/trunk/src/video_engine/vie_receiver.cc
deleted file mode 100644
index af0dc75..0000000
--- a/trunk/src/video_engine/vie_receiver.cc
+++ /dev/null
@@ -1,267 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vie_receiver.h"
-
-#include "critical_section_wrapper.h"
-#include "rtp_dump.h"
-#include "rtp_rtcp.h"
-#include "video_coding.h"
-#include "trace.h"
-
-namespace webrtc {
-
-ViEReceiver::ViEReceiver(int engine_id, int channel_id,
- RtpRtcp& rtp_rtcp,
- VideoCodingModule& module_vcm)
- : receive_cs_(CriticalSectionWrapper::CreateCriticalSection()),
- engine_id_(engine_id),
- channel_id_(channel_id),
- rtp_rtcp_(rtp_rtcp),
- vcm_(module_vcm),
- external_decryption_(NULL),
- decryption_buffer_(NULL),
- rtp_dump_(NULL),
- receiving_(false) {
-}
-
-ViEReceiver::~ViEReceiver() {
- if (decryption_buffer_) {
- delete[] decryption_buffer_;
- decryption_buffer_ = NULL;
- }
- if (rtp_dump_) {
- rtp_dump_->Stop();
- RtpDump::DestroyRtpDump(rtp_dump_);
- rtp_dump_ = NULL;
- }
-}
-
-int ViEReceiver::RegisterExternalDecryption(Encryption* decryption) {
- CriticalSectionScoped cs(receive_cs_.get());
- if (external_decryption_) {
- return -1;
- }
- decryption_buffer_ = new WebRtc_UWord8[kViEMaxMtu];
- if (decryption_buffer_ == NULL) {
- return -1;
- }
- external_decryption_ = decryption;
- return 0;
-}
-
-int ViEReceiver::DeregisterExternalDecryption() {
- CriticalSectionScoped cs(receive_cs_.get());
- if (external_decryption_ == NULL) {
- return -1;
- }
- external_decryption_ = NULL;
- return 0;
-}
-
-void ViEReceiver::RegisterSimulcastRtpRtcpModules(
- const std::list<RtpRtcp*>& rtp_modules) {
- CriticalSectionScoped cs(receive_cs_.get());
- rtp_rtcp_simulcast_.clear();
-
- if (!rtp_modules.empty()) {
- rtp_rtcp_simulcast_.insert(rtp_rtcp_simulcast_.begin(),
- rtp_modules.begin(),
- rtp_modules.end());
- }
-}
-
-void ViEReceiver::IncomingRTPPacket(const WebRtc_Word8* rtp_packet,
- const WebRtc_Word32 rtp_packet_length,
- const WebRtc_Word8* from_ip,
- const WebRtc_UWord16 from_port) {
- InsertRTPPacket(rtp_packet, rtp_packet_length);
-}
-
-void ViEReceiver::IncomingRTCPPacket(const WebRtc_Word8* rtcp_packet,
- const WebRtc_Word32 rtcp_packet_length,
- const WebRtc_Word8* from_ip,
- const WebRtc_UWord16 from_port) {
- InsertRTCPPacket(rtcp_packet, rtcp_packet_length);
-}
-
-int ViEReceiver::ReceivedRTPPacket(const void* rtp_packet,
- int rtp_packet_length) {
- if (!receiving_) {
- return -1;
- }
- return InsertRTPPacket((const WebRtc_Word8*) rtp_packet, rtp_packet_length);
-}
-
-int ViEReceiver::ReceivedRTCPPacket(const void* rtcp_packet,
- int rtcp_packet_length) {
- if (!receiving_) {
- return -1;
- }
- return InsertRTCPPacket((const WebRtc_Word8*) rtcp_packet,
- rtcp_packet_length);
-}
-
-WebRtc_Word32 ViEReceiver::OnReceivedPayloadData(
- const WebRtc_UWord8* payload_data, const WebRtc_UWord16 payload_size,
- const WebRtcRTPHeader* rtp_header) {
- if (rtp_header == NULL) {
- return 0;
- }
-
- if (vcm_.IncomingPacket(payload_data, payload_size, *rtp_header) != 0) {
- // Check this...
- return -1;
- }
- return 0;
-}
-
-int ViEReceiver::InsertRTPPacket(const WebRtc_Word8* rtp_packet,
- int rtp_packet_length) {
- // TODO(mflodman) Change decrypt to get rid of this cast.
- WebRtc_Word8* tmp_ptr = const_cast<WebRtc_Word8*>(rtp_packet);
- unsigned char* received_packet = reinterpret_cast<unsigned char*>(tmp_ptr);
- int received_packet_length = rtp_packet_length;
-
- {
- CriticalSectionScoped cs(receive_cs_.get());
-
- if (external_decryption_) {
- int decrypted_length = 0;
- external_decryption_->decrypt(channel_id_, received_packet,
- decryption_buffer_, received_packet_length,
- &decrypted_length);
- if (decrypted_length <= 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "RTP decryption failed");
- return -1;
- } else if (decrypted_length > kViEMaxMtu) {
- WEBRTC_TRACE(webrtc::kTraceCritical, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "InsertRTPPacket: %d bytes is allocated as RTP decrytption"
- " output, external decryption used %d bytes. => memory is "
- " now corrupted", kViEMaxMtu, decrypted_length);
- return -1;
- }
- received_packet = decryption_buffer_;
- received_packet_length = decrypted_length;
- }
-
- if (rtp_dump_) {
- rtp_dump_->DumpPacket(received_packet,
- static_cast<WebRtc_UWord16>(received_packet_length));
- }
- }
- return rtp_rtcp_.IncomingPacket(received_packet, received_packet_length);
-}
-
-int ViEReceiver::InsertRTCPPacket(const WebRtc_Word8* rtcp_packet,
- int rtcp_packet_length) {
- // TODO(mflodman) Change decrypt to get rid of this cast.
- WebRtc_Word8* tmp_ptr = const_cast<WebRtc_Word8*>(rtcp_packet);
- unsigned char* received_packet = reinterpret_cast<unsigned char*>(tmp_ptr);
- int received_packet_length = rtcp_packet_length;
- {
- CriticalSectionScoped cs(receive_cs_.get());
-
- if (external_decryption_) {
- int decrypted_length = 0;
- external_decryption_->decrypt_rtcp(channel_id_, received_packet,
- decryption_buffer_,
- received_packet_length,
- &decrypted_length);
- if (decrypted_length <= 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_), "RTP decryption failed");
- return -1;
- } else if (decrypted_length > kViEMaxMtu) {
- WEBRTC_TRACE(webrtc::kTraceCritical, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "InsertRTCPPacket: %d bytes is allocated as RTP "
- " decrytption output, external decryption used %d bytes. "
- " => memory is now corrupted",
- kViEMaxMtu, decrypted_length);
- return -1;
- }
- received_packet = decryption_buffer_;
- received_packet_length = decrypted_length;
- }
-
- if (rtp_dump_) {
- rtp_dump_->DumpPacket(
- received_packet, static_cast<WebRtc_UWord16>(received_packet_length));
- }
- }
- {
- CriticalSectionScoped cs(receive_cs_.get());
- std::list<RtpRtcp*>::iterator it = rtp_rtcp_simulcast_.begin();
- while (it != rtp_rtcp_simulcast_.end()) {
- RtpRtcp* rtp_rtcp = *it++;
- rtp_rtcp->IncomingPacket(received_packet, received_packet_length);
- }
- }
- return rtp_rtcp_.IncomingPacket(received_packet, received_packet_length);
-}
-
-void ViEReceiver::StartReceive() {
- receiving_ = true;
-}
-
-void ViEReceiver::StopReceive() {
- receiving_ = false;
-}
-
-int ViEReceiver::StartRTPDump(const char file_nameUTF8[1024]) {
- CriticalSectionScoped cs(receive_cs_.get());
- if (rtp_dump_) {
- // Restart it if it already exists and is started
- rtp_dump_->Stop();
- } else {
- rtp_dump_ = RtpDump::CreateRtpDump();
- if (rtp_dump_ == NULL) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "StartRTPDump: Failed to create RTP dump");
- return -1;
- }
- }
- if (rtp_dump_->Start(file_nameUTF8) != 0) {
- RtpDump::DestroyRtpDump(rtp_dump_);
- rtp_dump_ = NULL;
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "StartRTPDump: Failed to start RTP dump");
- return -1;
- }
- return 0;
-}
-
-int ViEReceiver::StopRTPDump() {
- CriticalSectionScoped cs(receive_cs_.get());
- if (rtp_dump_) {
- if (rtp_dump_->IsActive()) {
- rtp_dump_->Stop();
- } else {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "StopRTPDump: Dump not active");
- }
- RtpDump::DestroyRtpDump(rtp_dump_);
- rtp_dump_ = NULL;
- } else {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "StopRTPDump: RTP dump not started");
- return -1;
- }
- return 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_receiver.h b/trunk/src/video_engine/vie_receiver.h
deleted file mode 100644
index a99d388..0000000
--- a/trunk/src/video_engine/vie_receiver.h
+++ /dev/null
@@ -1,87 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_
-
-#include <list>
-
-#include "engine_configurations.h"
-#include "rtp_rtcp_defines.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-#include "typedefs.h"
-#include "udp_transport.h"
-#include "vie_defines.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class Encryption;
-class RtpDump;
-class RtpRtcp;
-class VideoCodingModule;
-
-class ViEReceiver : public UdpTransportData, public RtpData {
- public:
- ViEReceiver(int engine_id, int channel_id, RtpRtcp& rtp_rtcp,
- VideoCodingModule& module_vcm);
- ~ViEReceiver();
-
- int RegisterExternalDecryption(Encryption* decryption);
- int DeregisterExternalDecryption();
-
- void RegisterSimulcastRtpRtcpModules(const std::list<RtpRtcp*>& rtp_modules);
-
- void StartReceive();
- void StopReceive();
-
- int StartRTPDump(const char file_nameUTF8[1024]);
- int StopRTPDump();
-
- // Implements UdpTransportData.
- virtual void IncomingRTPPacket(const WebRtc_Word8* rtp_packet,
- const WebRtc_Word32 rtp_packet_length,
- const WebRtc_Word8* from_ip,
- const WebRtc_UWord16 from_port);
- virtual void IncomingRTCPPacket(const WebRtc_Word8* rtcp_packet,
- const WebRtc_Word32 rtcp_packet_length,
- const WebRtc_Word8* from_ip,
- const WebRtc_UWord16 from_port);
-
- // Receives packets from external transport.
- int ReceivedRTPPacket(const void* rtp_packet, int rtp_packet_length);
- int ReceivedRTCPPacket(const void* rtcp_packet, int rtcp_packet_length);
-
- // Implements RtpData.
- virtual WebRtc_Word32 OnReceivedPayloadData(
- const WebRtc_UWord8* payload_data,
- const WebRtc_UWord16 payload_size,
- const WebRtcRTPHeader* rtp_header);
-
- private:
- int InsertRTPPacket(const WebRtc_Word8* rtp_packet, int rtp_packet_length);
- int InsertRTCPPacket(const WebRtc_Word8* rtcp_packet, int rtcp_packet_length);
-
- scoped_ptr<CriticalSectionWrapper> receive_cs_;
- int engine_id_;
- int channel_id_;
- RtpRtcp& rtp_rtcp_;
- std::list<RtpRtcp*> rtp_rtcp_simulcast_;
- VideoCodingModule& vcm_;
-
- Encryption* external_decryption_;
- WebRtc_UWord8* decryption_buffer_;
- RtpDump* rtp_dump_;
- bool receiving_;
-};
-
-} // namespace webrt
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_RECEIVER_H_
diff --git a/trunk/src/video_engine/vie_ref_count.cc b/trunk/src/video_engine/vie_ref_count.cc
deleted file mode 100644
index 2ab285c..0000000
--- a/trunk/src/video_engine/vie_ref_count.cc
+++ /dev/null
@@ -1,46 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vie_ref_count.h"
-
-#include "critical_section_wrapper.h"
-
-namespace webrtc {
-
-ViERefCount::ViERefCount()
- : count_(0),
- crit_(CriticalSectionWrapper::CreateCriticalSection()) {
-}
-
-ViERefCount::~ViERefCount() {
-}
-
-ViERefCount& ViERefCount::operator++(int) {
- CriticalSectionScoped lock(crit_.get());
- count_++;
- return *this;
-}
-
-ViERefCount& ViERefCount::operator--(int) {
- CriticalSectionScoped lock(crit_.get());
- count_--;
- return *this;
-}
-
-void ViERefCount::Reset() {
- CriticalSectionScoped lock(crit_.get());
- count_ = 0;
-}
-
-int ViERefCount::GetCount() const {
- return count_;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_ref_count.h b/trunk/src/video_engine/vie_ref_count.h
deleted file mode 100644
index 65efe36..0000000
--- a/trunk/src/video_engine/vie_ref_count.h
+++ /dev/null
@@ -1,40 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// TODO(mflodman) Use ref count class in system_wrappers.
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_REF_COUNT_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_REF_COUNT_H_
-
-#include "system_wrappers/interface/scoped_ptr.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-
-class ViERefCount {
- public:
- ViERefCount();
- ~ViERefCount();
-
- ViERefCount& operator++(int);
- ViERefCount& operator--(int);
-
- void Reset();
- int GetCount() const;
-
- private:
- volatile int count_;
- scoped_ptr<CriticalSectionWrapper> crit_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_REF_COUNT_H_
diff --git a/trunk/src/video_engine/vie_remb.cc b/trunk/src/video_engine/vie_remb.cc
deleted file mode 100644
index 7732408..0000000
--- a/trunk/src/video_engine/vie_remb.cc
+++ /dev/null
@@ -1,215 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_remb.h"
-
-#include <algorithm>
-#include <cassert>
-
-#include "modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "system_wrappers/interface/critical_section_wrapper.h"
-#include "system_wrappers/interface/tick_util.h"
-#include "system_wrappers/interface/trace.h"
-
-namespace webrtc {
-
-const int kRembSendIntervallMs = 1000;
-
-// % threshold for if we should send a new REMB asap.
-const int kSendThresholdPercent = 97;
-
-VieRemb::VieRemb(int engine_id)
- : engine_id_(engine_id),
- list_crit_(CriticalSectionWrapper::CreateCriticalSection()),
- last_remb_time_(TickTime::MillisecondTimestamp()),
- last_send_bitrate_(0) {
-}
-
-VieRemb::~VieRemb() {
-}
-
-void VieRemb::AddReceiveChannel(RtpRtcp* rtp_rtcp) {
- assert(rtp_rtcp);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, engine_id_,
- "VieRemb::AddReceiveChannel(%p)", rtp_rtcp);
-
- CriticalSectionScoped cs(list_crit_.get());
- if (std::find(receive_modules_.begin(), receive_modules_.end(), rtp_rtcp) !=
- receive_modules_.end())
- return;
-
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, engine_id_, "AddRembChannel");
- // The module probably doesn't have a remote SSRC yet, so don't add it to the
- // map.
- receive_modules_.push_back(rtp_rtcp);
-}
-
-void VieRemb::RemoveReceiveChannel(RtpRtcp* rtp_rtcp) {
- assert(rtp_rtcp);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, engine_id_,
- "VieRemb::RemoveReceiveChannel(%p)", rtp_rtcp);
-
- CriticalSectionScoped cs(list_crit_.get());
- unsigned int ssrc = rtp_rtcp->RemoteSSRC();
- for (RtpModules::iterator it = receive_modules_.begin();
- it != receive_modules_.end(); ++it) {
- if ((*it) == rtp_rtcp) {
- receive_modules_.erase(it);
- break;
- }
- }
- bitrates_.erase(ssrc);
-}
-
-void VieRemb::AddRembSender(RtpRtcp* rtp_rtcp) {
- assert(rtp_rtcp);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, engine_id_,
- "VieRemb::AddRembSender(%p)", rtp_rtcp);
-
- CriticalSectionScoped cs(list_crit_.get());
-
- // Verify this module hasn't been added earlier.
- if (std::find(rtcp_sender_.begin(), rtcp_sender_.end(), rtp_rtcp) !=
- rtcp_sender_.end())
- return;
- rtcp_sender_.push_back(rtp_rtcp);
-}
-
-void VieRemb::RemoveRembSender(RtpRtcp* rtp_rtcp) {
- assert(rtp_rtcp);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, engine_id_,
- "VieRemb::RemoveRembSender(%p)", rtp_rtcp);
-
- CriticalSectionScoped cs(list_crit_.get());
- for (RtpModules::iterator it = rtcp_sender_.begin();
- it != rtcp_sender_.end(); ++it) {
- if ((*it) == rtp_rtcp) {
- rtcp_sender_.erase(it);
- return;
- }
- }
-}
-
-void VieRemb::AddSendChannel(RtpRtcp* rtp_rtcp) {
- assert(rtp_rtcp);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, engine_id_,
- "VieRemb::AddSendChannel(%p)", rtp_rtcp);
-
- CriticalSectionScoped cs(list_crit_.get());
-
- // Verify this module hasn't been added earlier.
- if (std::find(send_modules_.begin(), send_modules_.end(), rtp_rtcp) !=
- send_modules_.end())
- return;
- send_modules_.push_back(rtp_rtcp);
-}
-
-void VieRemb::RemoveSendChannel(RtpRtcp* rtp_rtcp) {
- assert(rtp_rtcp);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVideo, engine_id_,
- "VieRemb::RemoveSendChannel(%p)", rtp_rtcp);
-
- CriticalSectionScoped cs(list_crit_.get());
- for (RtpModules::iterator it = send_modules_.begin();
- it != send_modules_.end(); ++it) {
- if ((*it) == rtp_rtcp) {
- send_modules_.erase(it);
- return;
- }
- }
-}
-
-void VieRemb::OnReceiveBitrateChanged(unsigned int ssrc, unsigned int bitrate) {
- WEBRTC_TRACE(kTraceStream, kTraceVideo, engine_id_,
- "VieRemb::UpdateBitrateEstimate(ssrc: %u, bitrate: %u)",
- ssrc, bitrate);
- CriticalSectionScoped cs(list_crit_.get());
-
- // Check if this is a new ssrc and add it to the map if it is.
- if (bitrates_.find(ssrc) == bitrates_.end()) {
- bitrates_[ssrc] = bitrate;
- }
-
- int new_remb_bitrate = last_send_bitrate_ - bitrates_[ssrc] + bitrate;
- if (new_remb_bitrate < kSendThresholdPercent * last_send_bitrate_ / 100) {
- // The new bitrate estimate is less than kSendThresholdPercent % of the last
- // report. Send a REMB asap.
- last_remb_time_ = TickTime::MillisecondTimestamp() - kRembSendIntervallMs;
- }
- bitrates_[ssrc] = bitrate;
-}
-
-void VieRemb::OnReceivedRemb(unsigned int bitrate) {
- WEBRTC_TRACE(kTraceStream, kTraceVideo, engine_id_,
- "VieRemb::OnReceivedRemb(bitrate: %u)", bitrate);
- // TODO(mflodman) Should be extended to allow different split of bitrate.
- // TODO(mflodman) Do we want to call |SetMaximumBitrateEstimate| from
- // |Process| instead?
-
- // Split the bitrate estimate between all sending channels.
- CriticalSectionScoped cs(list_crit_.get());
- for (RtpModules::iterator it = send_modules_.begin();
- it != send_modules_.end(); ++it) {
- (*it)->SetMaximumBitrateEstimate(bitrate / send_modules_.size());
- }
-}
-
-WebRtc_Word32 VieRemb::ChangeUniqueId(const WebRtc_Word32 id) {
- return 0;
-}
-
-WebRtc_Word32 VieRemb::TimeUntilNextProcess() {
- return kRembSendIntervallMs -
- (TickTime::MillisecondTimestamp() - last_remb_time_);
-}
-
-WebRtc_Word32 VieRemb::Process() {
- int64_t now = TickTime::MillisecondTimestamp();
- if (now - last_remb_time_ < kRembSendIntervallMs)
- return 0;
-
- last_remb_time_ = now;
-
- // Calculate total receive bitrate estimate.
- list_crit_->Enter();
- int total_bitrate = 0;
- int num_bitrates = bitrates_.size();
-
- if (num_bitrates == 0) {
- list_crit_->Leave();
- return 0;
- }
-
- // TODO(mflodman) Use std::vector and change RTP module API.
- unsigned int* ssrcs = new unsigned int[num_bitrates];
-
- int idx = 0;
- for (SsrcBitrate::iterator it = bitrates_.begin(); it != bitrates_.end();
- ++it, ++idx) {
- total_bitrate += it->second;
- ssrcs[idx] = it->first;
- }
-
- // Send a REMB packet.
- RtpRtcp* sender = NULL;
- if (!rtcp_sender_.empty()) {
- sender = rtcp_sender_.front();
- }
- last_send_bitrate_ = total_bitrate;
- list_crit_->Leave();
-
- if (sender) {
- sender->SetREMBData(total_bitrate, num_bitrates, ssrcs);
- }
- delete [] ssrcs;
- return 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_remb.h b/trunk/src/video_engine/vie_remb.h
deleted file mode 100644
index 32d58a7..0000000
--- a/trunk/src/video_engine/vie_remb.h
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// 1. Register a RtpRtcp module to include in the REMB packet.
-// 2. When UpdateBitrateEstimate is called for the first time for a SSRC, add it
-// to the map.
-// 3. Send a new REMB every kRembSendIntervallMs or if a lower bitrate estimate
-// for a specified SSRC.
-
-
-#ifndef WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_REMB_H_
-#define WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_REMB_H_
-
-#include <list>
-#include <map>
-
-#include "modules/interface/module.h"
-#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class RtpRtcp;
-
-class VieRemb : public RtpRemoteBitrateObserver, public Module {
- public:
- explicit VieRemb(int engine_id);
- ~VieRemb();
-
- // Called to add a receive channel to include in the REMB packet.
- void AddReceiveChannel(RtpRtcp* rtp_rtcp);
-
- // Removes the specified channel from REMB estimate.
- void RemoveReceiveChannel(RtpRtcp* rtp_rtcp);
-
- // Called to add a module that can generate and send REMB RTCP.
- void AddRembSender(RtpRtcp* rtp_rtcp);
-
- // Removes a REMB RTCP sender.
- void RemoveRembSender(RtpRtcp* rtp_rtcp);
-
- // Called to add a send channel encoding and sending data, affected by
- // received REMB packets.
- void AddSendChannel(RtpRtcp* rtp_rtcp);
-
- // Removes the specified channel from receiving REMB packet estimates.
- void RemoveSendChannel(RtpRtcp* rtp_rtcp);
-
- // Called every time there is a new bitrate estimate for the received stream
- // with given SSRC. This call will trigger a new RTCP REMB packet if the
- // bitrate estimate has decreased or if no RTCP REMB packet has been sent for
- // a certain time interval.
- // Implements RtpReceiveBitrateUpdate.
- virtual void OnReceiveBitrateChanged(unsigned int ssrc, unsigned int bitrate);
-
- // Called for every new receive REMB packet and distributes the estmate
- // between all sending modules.
- virtual void OnReceivedRemb(unsigned int bitrate);
-
- // Implements Module.
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
- virtual WebRtc_Word32 TimeUntilNextProcess();
- virtual WebRtc_Word32 Process();
-
- private:
- typedef std::list<RtpRtcp*> RtpModules;
- typedef std::map<unsigned int, unsigned int> SsrcBitrate;
-
- int engine_id_;
- scoped_ptr<CriticalSectionWrapper> list_crit_;
-
- // The last time a REMB was sent.
- int64_t last_remb_time_;
- int last_send_bitrate_;
-
- // All RtpRtcp modules to include in the REMB packet.
- RtpModules receive_modules_;
-
- // All modules encoding and sending data.
- RtpModules send_modules_;
-
- // All modules that can send REMB RTCP.
- RtpModules rtcp_sender_;
-
- // The last bitrate update for each SSRC.
- SsrcBitrate bitrates_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_MAIN_SOURCE_VIE_REMB_H_
diff --git a/trunk/src/video_engine/vie_remb_unittest.cc b/trunk/src/video_engine/vie_remb_unittest.cc
deleted file mode 100644
index a842103..0000000
--- a/trunk/src/video_engine/vie_remb_unittest.cc
+++ /dev/null
@@ -1,335 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-
-// This file includes unit tests for ViERemb.
-
-#include <gmock/gmock.h>
-#include <gtest/gtest.h>
-#if defined(_WIN32)
-#include <windows.h>
-#endif
-
-#include "modules/rtp_rtcp/interface/rtp_rtcp.h"
-#include "modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-#include "video_engine/vie_remb.h"
-
-using ::testing::_;
-using ::testing::AnyNumber;
-using ::testing::Return;
-
-namespace webrtc {
-
-class ViERembTest : public ::testing::Test {
- protected:
- virtual void SetUp() {
- vie_remb_.reset(new VieRemb(1234));
- }
- scoped_ptr<VieRemb> vie_remb_;
-
- void TestSleep(unsigned int time_ms) {
-#if defined(_WIN32)
- ::Sleep(time_ms);
-#else
- usleep(time_ms * 1000);
-#endif
- }
-};
-
-TEST_F(ViERembTest, OneModuleTestForSendingRemb)
-{
- MockRtpRtcp rtp;
- vie_remb_->AddReceiveChannel(&rtp);
- vie_remb_->AddSendChannel(&rtp);
- vie_remb_->AddRembSender(&rtp);
-
- const unsigned int bitrate_estimate = 456;
- unsigned int ssrc[] = { 1234 };
-
- vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate);
- EXPECT_CALL(rtp, RemoteSSRC())
- .WillRepeatedly(Return(ssrc[0]));
-
- // TODO(mflodman) Add fake clock and remove the lowered bitrate below.
- TestSleep(1010);
- EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, 1, _))
- .Times(1);
- vie_remb_->Process();
-
- // Lower bitrate to send another REMB packet.
- vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate - 100);
- EXPECT_CALL(rtp, SetREMBData(bitrate_estimate - 100, 1, _))
- .Times(1);
- vie_remb_->Process();
-
- vie_remb_->RemoveReceiveChannel(&rtp);
- vie_remb_->RemoveSendChannel(&rtp);
- vie_remb_->RemoveRembSender(&rtp);
-}
-
-TEST_F(ViERembTest, LowerEstimateToSendRemb)
-{
- MockRtpRtcp rtp;
- vie_remb_->AddReceiveChannel(&rtp);
- vie_remb_->AddSendChannel(&rtp);
- vie_remb_->AddRembSender(&rtp);
-
- unsigned int bitrate_estimate = 456;
- unsigned int ssrc[] = { 1234 };
-
- vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate);
- EXPECT_CALL(rtp, RemoteSSRC())
- .WillRepeatedly(Return(ssrc[0]));
-
- // Lower the estimate with more than 3% to trigger a call to SetREMBData right
- // away.
- bitrate_estimate = bitrate_estimate - 100;
- EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, 1, _))
- .Times(1);
- vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate);
- vie_remb_->Process();
-}
-
-TEST_F(ViERembTest, VerifyCombinedBitrateEstimate)
-{
- MockRtpRtcp rtp_0;
- MockRtpRtcp rtp_1;
- vie_remb_->AddReceiveChannel(&rtp_0);
- vie_remb_->AddSendChannel(&rtp_0);
- vie_remb_->AddRembSender(&rtp_0);
- vie_remb_->AddReceiveChannel(&rtp_1);
-
- unsigned int bitrate_estimate[] = { 456, 789 };
- unsigned int ssrc[] = { 1234, 5678 };
-
- vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate[0]);
- EXPECT_CALL(rtp_0, RemoteSSRC())
- .Times(AnyNumber())
- .WillRepeatedly(Return(ssrc[0]));
-
- vie_remb_->OnReceiveBitrateChanged(ssrc[1], bitrate_estimate[1] + 100);
- EXPECT_CALL(rtp_1, RemoteSSRC())
- .Times(AnyNumber())
- .WillRepeatedly(Return(ssrc[1]));
-
- // Lower the estimate to trigger a callback.
- int total_bitrate = bitrate_estimate[0] + bitrate_estimate[1];
- EXPECT_CALL(rtp_0, SetREMBData(total_bitrate, 2, _))
- .Times(1);
- vie_remb_->OnReceiveBitrateChanged(ssrc[1], bitrate_estimate[1]);
- vie_remb_->Process();
-
- vie_remb_->RemoveReceiveChannel(&rtp_0);
- vie_remb_->RemoveSendChannel(&rtp_0);
- vie_remb_->RemoveRembSender(&rtp_0);
- vie_remb_->RemoveReceiveChannel(&rtp_1);
-}
-
-// Add two senders, and insert a received REMB estimate. Both sending channels
-// should get half of the received value.
-TEST_F(ViERembTest, IncomingRemb)
-{
- MockRtpRtcp rtp_0;
- MockRtpRtcp rtp_1;
- vie_remb_->AddSendChannel(&rtp_0);
- vie_remb_->AddSendChannel(&rtp_1);
-
- const unsigned int bitrate_estimate = 1200;
-
- // Fake received REMB and verify both modules get half of the bitrate.
- EXPECT_CALL(rtp_0, SetMaximumBitrateEstimate(bitrate_estimate/2))
- .Times(1);
- EXPECT_CALL(rtp_1, SetMaximumBitrateEstimate(bitrate_estimate/2))
- .Times(1);
- vie_remb_->OnReceivedRemb(bitrate_estimate);
-
- // Remove one of the modules and verify the other module get the entire
- // bitrate.
- vie_remb_->RemoveSendChannel(&rtp_0);
- EXPECT_CALL(rtp_0, SetMaximumBitrateEstimate(_))
- .Times(0);
- EXPECT_CALL(rtp_1, SetMaximumBitrateEstimate(bitrate_estimate))
- .Times(1);
- vie_remb_->OnReceivedRemb(bitrate_estimate);
-
- vie_remb_->RemoveSendChannel(&rtp_1);
-}
-
-TEST_F(ViERembTest, NoRembForIncreasedBitrate)
-{
- MockRtpRtcp rtp_0;
- MockRtpRtcp rtp_1;
- vie_remb_->AddReceiveChannel(&rtp_0);
- vie_remb_->AddSendChannel(&rtp_0);
- vie_remb_->AddRembSender(&rtp_0);
- vie_remb_->AddReceiveChannel(&rtp_1);
-
- unsigned int bitrate_estimate[] = { 456, 789 };
- unsigned int ssrc[] = { 1234, 5678 };
-
- vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate[0]);
- EXPECT_CALL(rtp_0, RemoteSSRC())
- .Times(AnyNumber())
- .WillRepeatedly(Return(ssrc[0]));
-
- vie_remb_->OnReceiveBitrateChanged(ssrc[1], bitrate_estimate[1]);
- EXPECT_CALL(rtp_1, RemoteSSRC())
- .Times(AnyNumber())
- .WillRepeatedly(Return(ssrc[1]));
-
- // Trigger a first call to have a running state.
- // TODO(mflodman) Add fake clock.
- TestSleep(1010);
- EXPECT_CALL(rtp_0,
- SetREMBData(bitrate_estimate[0] + bitrate_estimate[1], 2, _))
- .Times(1);
- vie_remb_->Process();
-
- // Increased estimate shouldn't trigger a callback right away.
- vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate[0] + 1);
- EXPECT_CALL(rtp_0, SetREMBData(_, _, _))
- .Times(0);
-
- // Decresing the estimate less than 3% shouldn't trigger a new callback.
- int lower_estimate = bitrate_estimate[0] * 98 / 100;
- vie_remb_->OnReceiveBitrateChanged(ssrc[0], lower_estimate);
- EXPECT_CALL(rtp_0, SetREMBData(_, _, _))
- .Times(0);
-
- vie_remb_->Process();
- vie_remb_->RemoveReceiveChannel(&rtp_1);
- vie_remb_->RemoveReceiveChannel(&rtp_0);
- vie_remb_->RemoveSendChannel(&rtp_0);
- vie_remb_->RemoveRembSender(&rtp_0);
-}
-
-TEST_F(ViERembTest, ChangeSendRtpModule)
-{
- MockRtpRtcp rtp_0;
- MockRtpRtcp rtp_1;
- vie_remb_->AddReceiveChannel(&rtp_0);
- vie_remb_->AddSendChannel(&rtp_0);
- vie_remb_->AddRembSender(&rtp_0);
- vie_remb_->AddReceiveChannel(&rtp_1);
-
- unsigned int bitrate_estimate[] = { 456, 789 };
- unsigned int ssrc[] = { 1234, 5678 };
-
- vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate[0]);
- EXPECT_CALL(rtp_0, RemoteSSRC())
- .Times(AnyNumber())
- .WillRepeatedly(Return(ssrc[0]));
-
- vie_remb_->OnReceiveBitrateChanged(ssrc[1], bitrate_estimate[1]);
- EXPECT_CALL(rtp_1, RemoteSSRC())
- .Times(AnyNumber())
- .WillRepeatedly(Return(ssrc[1]));
-
- // Decrease estimate to trigger a REMB.
- bitrate_estimate[0] = bitrate_estimate[0] - 100;
- EXPECT_CALL(rtp_0, SetREMBData(bitrate_estimate[0] + bitrate_estimate[1], 2,
- _))
- .Times(1);
- vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate[0]);
- vie_remb_->Process();
-
- // Remove the sending module, add it again -> should get remb on the second
- // module.
- vie_remb_->RemoveSendChannel(&rtp_0);
- vie_remb_->RemoveRembSender(&rtp_0);
- vie_remb_->AddSendChannel(&rtp_1);
- vie_remb_->AddRembSender(&rtp_1);
- vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate[0]);
-
- bitrate_estimate[1] = bitrate_estimate[1] - 100;
- EXPECT_CALL(rtp_1, SetREMBData(bitrate_estimate[0] + bitrate_estimate[1], 2,
- _))
- .Times(1);
- vie_remb_->OnReceiveBitrateChanged(ssrc[1], bitrate_estimate[1]);
- vie_remb_->Process();
-
- vie_remb_->RemoveReceiveChannel(&rtp_0);
- vie_remb_->RemoveReceiveChannel(&rtp_1);
- vie_remb_->RemoveSendChannel(&rtp_1);
-}
-
-TEST_F(ViERembTest, OnlyOneRembForDoubleProcess)
-{
- MockRtpRtcp rtp;
- unsigned int bitrate_estimate = 456;
- unsigned int ssrc[] = { 1234 };
-
- vie_remb_->AddReceiveChannel(&rtp);
- vie_remb_->AddSendChannel(&rtp);
- vie_remb_->AddRembSender(&rtp);
- vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate);
- EXPECT_CALL(rtp, RemoteSSRC())
- .WillRepeatedly(Return(ssrc[0]));
-
- // Lower the estimate, should trigger a call to SetREMBData right away.
- bitrate_estimate = bitrate_estimate - 100;
- EXPECT_CALL(rtp, SetREMBData(bitrate_estimate, 1, _))
- .Times(1);
- vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate);
- vie_remb_->Process();
-
- // Call Process again, this should not trigger a new callback.
- EXPECT_CALL(rtp, SetREMBData(_, _, _))
- .Times(0);
- vie_remb_->Process();
- vie_remb_->RemoveReceiveChannel(&rtp);
- vie_remb_->RemoveSendChannel(&rtp);
- vie_remb_->RemoveRembSender(&rtp);
-}
-
-TEST_F(ViERembTest, NoOnReceivedBitrateChangedCall)
-{
- MockRtpRtcp rtp;
- EXPECT_CALL(rtp, RemoteSSRC())
- .WillRepeatedly(Return(1234));
-
- vie_remb_->AddReceiveChannel(&rtp);
- vie_remb_->AddSendChannel(&rtp);
- vie_remb_->AddRembSender(&rtp);
- // TODO(mflodman) Add fake clock.
- TestSleep(1010);
- // No bitrate estimate given, no callback expected.
- EXPECT_CALL(rtp, SetREMBData(_, _, _))
- .Times(0);
- vie_remb_->Process();
-
- vie_remb_->RemoveReceiveChannel(&rtp);
- vie_remb_->RemoveSendChannel(&rtp);
- vie_remb_->RemoveRembSender(&rtp);
-}
-
-TEST_F(ViERembTest, NoSendingRtpModule)
-{
- MockRtpRtcp rtp;
- vie_remb_->AddReceiveChannel(&rtp);
-
- unsigned int bitrate_estimate = 456;
- unsigned int ssrc[] = { 1234 };
-
- vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate);
- EXPECT_CALL(rtp, RemoteSSRC())
- .WillRepeatedly(Return(ssrc[0]));
-
- // Lower the estimate. This should normally trigger a callback, but not now
- // since we have no sending module.
- bitrate_estimate = bitrate_estimate - 100;
- EXPECT_CALL(rtp, SetREMBData(_, _, _))
- .Times(0);
- vie_remb_->OnReceiveBitrateChanged(ssrc[0], bitrate_estimate);
- vie_remb_->Process();
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_render_impl.cc b/trunk/src/video_engine/vie_render_impl.cc
deleted file mode 100644
index e37e5d4..0000000
--- a/trunk/src/video_engine/vie_render_impl.cc
+++ /dev/null
@@ -1,398 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_render_impl.h"
-
-#include "engine_configurations.h"
-#include "modules/video_render/main/interface/video_render.h"
-#include "modules/video_render/main/interface/video_render_defines.h"
-#include "system_wrappers/interface/trace.h"
-#include "video_engine/include/vie_errors.h"
-#include "video_engine/vie_capturer.h"
-#include "video_engine/vie_channel.h"
-#include "video_engine/vie_channel_manager.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_frame_provider_base.h"
-#include "video_engine/vie_impl.h"
-#include "video_engine/vie_input_manager.h"
-#include "video_engine/vie_render_manager.h"
-#include "video_engine/vie_renderer.h"
-#include "video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-ViERender* ViERender::GetInterface(VideoEngine* video_engine) {
-#ifdef WEBRTC_VIDEO_ENGINE_RENDER_API
- if (!video_engine) {
- return NULL;
- }
- VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
- ViERenderImpl* vie_render_impl = vie_impl;
- // Increase ref count.
- (*vie_render_impl)++;
- return vie_render_impl;
-#else
- return NULL;
-#endif
-}
-
-int ViERenderImpl::Release() {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
- "ViERender::Release()");
- // Decrease ref count
- (*this)--;
- WebRtc_Word32 ref_count = GetCount();
- if (ref_count < 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
- "ViERender release too many times");
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
- "ViERender reference count: %d", ref_count);
- return ref_count;
-}
-
-ViERenderImpl::ViERenderImpl(ViESharedData* shared_data)
- : shared_data_(shared_data) {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViERenderImpl::ViERenderImpl() Ctor");
-}
-
-ViERenderImpl::~ViERenderImpl() {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViERenderImpl::~ViERenderImpl() Dtor");
-}
-
-int ViERenderImpl::RegisterVideoRenderModule(
- VideoRender& render_module) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s (&render_module: %p)", __FUNCTION__, &render_module);
- if (shared_data_->render_manager()->RegisterVideoRenderModule(
- render_module) != 0) {
- shared_data_->SetLastError(kViERenderUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERenderImpl::DeRegisterVideoRenderModule(
- VideoRender& render_module) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s (&render_module: %p)", __FUNCTION__, &render_module);
- if (shared_data_->render_manager()->DeRegisterVideoRenderModule(
- render_module) != 0) {
- // Error logging is done in ViERenderManager::DeRegisterVideoRenderModule.
- shared_data_->SetLastError(kViERenderUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERenderImpl::AddRenderer(const int render_id, void* window,
- const unsigned int z_order, const float left,
- const float top, const float right,
- const float bottom) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s (render_id: %d, window: 0x%p, z_order: %u, left: %f, "
- "top: %f, right: %f, bottom: %f)",
- __FUNCTION__, render_id, window, z_order, left, top, right,
- bottom);
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
- {
- ViERenderManagerScoped rs(*(shared_data_->render_manager()));
- if (rs.Renderer(render_id)) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - Renderer already exist %d.", __FUNCTION__,
- render_id);
- shared_data_->SetLastError(kViERenderAlreadyExists);
- return -1;
- }
- }
- if (render_id >= kViEChannelIdBase && render_id <= kViEChannelIdMax) {
- // This is a channel.
- ViEChannelManagerScoped cm(*(shared_data_->channel_manager()));
- ViEFrameProviderBase* frame_provider = cm.Channel(render_id);
- if (!frame_provider) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: FrameProvider id %d doesn't exist", __FUNCTION__,
- render_id);
- shared_data_->SetLastError(kViERenderInvalidRenderId);
- return -1;
- }
- ViERenderer* renderer = shared_data_->render_manager()->AddRenderStream(
- render_id, window, z_order, left, top, right, bottom);
- if (!renderer) {
- shared_data_->SetLastError(kViERenderUnknownError);
- return -1;
- }
- return frame_provider->RegisterFrameCallback(render_id, renderer);
- } else {
- // Camera or file.
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViEFrameProviderBase* frame_provider = is.FrameProvider(render_id);
- if (!frame_provider) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: FrameProvider id %d doesn't exist", __FUNCTION__,
- render_id);
- shared_data_->SetLastError(kViERenderInvalidRenderId);
- return -1;
- }
- ViERenderer* renderer = shared_data_->render_manager()->AddRenderStream(
- render_id, window, z_order, left, top, right, bottom);
- if (!renderer) {
- shared_data_->SetLastError(kViERenderUnknownError);
- return -1;
- }
- return frame_provider->RegisterFrameCallback(render_id, renderer);
- }
-}
-
-int ViERenderImpl::RemoveRenderer(const int render_id) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s(render_id: %d)", __FUNCTION__, render_id);
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
-
- ViERenderer* renderer = NULL;
- {
- ViERenderManagerScoped rs(*(shared_data_->render_manager()));
- renderer = rs.Renderer(render_id);
- if (!renderer) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo,
- ViEId(shared_data_->instance_id()),
- "%s No render exist with render_id: %d", __FUNCTION__,
- render_id);
- shared_data_->SetLastError(kViERenderInvalidRenderId);
- return -1;
- }
- // Leave the scope lock since we don't want to lock two managers
- // simultanousely.
- }
- if (render_id >= kViEChannelIdBase && render_id <= kViEChannelIdMax) {
- // This is a channel.
- ViEChannelManagerScoped cm(*(shared_data_->channel_manager()));
- ViEChannel* channel = cm.Channel(render_id);
- if (!channel) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo,
- ViEId(shared_data_->instance_id()),
- "%s: no channel with id %d exists ", __FUNCTION__,
- render_id);
- shared_data_->SetLastError(kViERenderInvalidRenderId);
- return -1;
- }
- channel->DeregisterFrameCallback(renderer);
- } else {
- // Provider owned by inputmanager, i.e. file or capture device.
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViEFrameProviderBase* provider = is.FrameProvider(render_id);
- if (!provider) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo,
- ViEId(shared_data_->instance_id()),
- "%s: no provider with id %d exists ", __FUNCTION__,
- render_id);
- shared_data_->SetLastError(kViERenderInvalidRenderId);
- return -1;
- }
- provider->DeregisterFrameCallback(renderer);
- }
- if (shared_data_->render_manager()->RemoveRenderStream(render_id) != 0) {
- shared_data_->SetLastError(kViERenderUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERenderImpl::StartRender(const int render_id) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), render_id),
- "%s(channel: %d)", __FUNCTION__, render_id);
- ViERenderManagerScoped rs(*(shared_data_->render_manager()));
- ViERenderer* renderer = rs.Renderer(render_id);
- if (!renderer) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), render_id),
- "%s: No renderer with render Id %d exist.", __FUNCTION__,
- render_id);
- shared_data_->SetLastError(kViERenderInvalidRenderId);
- return -1;
- }
- if (renderer->StartRender() != 0) {
- shared_data_->SetLastError(kViERenderUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERenderImpl::StopRender(const int render_id) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), render_id),
- "%s(channel: %d)", __FUNCTION__, render_id);
- ViERenderManagerScoped rs(*(shared_data_->render_manager()));
- ViERenderer* renderer = rs.Renderer(render_id);
- if (!renderer) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), render_id),
- "%s: No renderer with render_id %d exist.", __FUNCTION__,
- render_id);
- shared_data_->SetLastError(kViERenderInvalidRenderId);
- return -1;
- }
- if (renderer->StopRender() != 0) {
- shared_data_->SetLastError(kViERenderUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERenderImpl::ConfigureRender(int render_id, const unsigned int z_order,
- const float left, const float top,
- const float right, const float bottom) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), render_id),
- "%s(channel: %d)", __FUNCTION__, render_id);
- ViERenderManagerScoped rs(*(shared_data_->render_manager()));
- ViERenderer* renderer = rs.Renderer(render_id);
- if (!renderer) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), render_id),
- "%s: No renderer with render_id %d exist.", __FUNCTION__,
- render_id);
- shared_data_->SetLastError(kViERenderInvalidRenderId);
- return -1;
- }
-
- if (renderer->ConfigureRenderer(z_order, left, top, right, bottom) != 0) {
- shared_data_->SetLastError(kViERenderUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERenderImpl::MirrorRenderStream(const int render_id, const bool enable,
- const bool mirror_xaxis,
- const bool mirror_yaxis) {
- ViERenderManagerScoped rs(*(shared_data_->render_manager()));
- ViERenderer* renderer = rs.Renderer(render_id);
- if (!renderer) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), render_id),
- "%s: No renderer with render_id %d exist.", __FUNCTION__,
- render_id);
- shared_data_->SetLastError(kViERenderInvalidRenderId);
- return -1;
- }
- if (renderer->EnableMirroring(render_id, enable, mirror_xaxis, mirror_yaxis)
- != 0) {
- shared_data_->SetLastError(kViERenderUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERenderImpl::AddRenderer(const int render_id,
- RawVideoType video_input_format,
- ExternalRenderer* external_renderer) {
- // Check if the client requested a format that we can convert the frames to.
- if (video_input_format != kVideoI420 &&
- video_input_format != kVideoYV12 &&
- video_input_format != kVideoYUY2 &&
- video_input_format != kVideoUYVY &&
- video_input_format != kVideoARGB &&
- video_input_format != kVideoRGB24 &&
- video_input_format != kVideoRGB565 &&
- video_input_format != kVideoARGB4444 &&
- video_input_format != kVideoARGB1555) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), render_id),
- "%s: Unsupported video frame format requested",
- __FUNCTION__, render_id);
- shared_data_->SetLastError(kViERenderInvalidFrameFormat);
- return -1;
- }
- if (!shared_data_->Initialized()) {
- shared_data_->SetLastError(kViENotInitialized);
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - ViE instance %d not initialized", __FUNCTION__,
- shared_data_->instance_id());
- return -1;
- }
- {
- // Verify the renderer doesn't exist.
- ViERenderManagerScoped rs(*(shared_data_->render_manager()));
- if (rs.Renderer(render_id)) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s - Renderer already exist %d.", __FUNCTION__,
- render_id);
- shared_data_->SetLastError(kViERenderAlreadyExists);
- return -1;
- }
- }
- if (render_id >= kViEChannelIdBase && render_id <= kViEChannelIdMax) {
- // This is a channel.
- ViEChannelManagerScoped cm(*(shared_data_->channel_manager()));
- ViEFrameProviderBase* frame_provider = cm.Channel(render_id);
- if (!frame_provider) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: FrameProvider id %d doesn't exist", __FUNCTION__,
- render_id);
- shared_data_->SetLastError(kViERenderInvalidRenderId);
- return -1;
- }
- ViERenderer* renderer = shared_data_->render_manager()->AddRenderStream(
- render_id, NULL, 0, 0.0f, 0.0f, 1.0f, 1.0f);
- if (!renderer) {
- shared_data_->SetLastError(kViERenderUnknownError);
- return -1;
- }
- if (renderer->SetExternalRenderer(render_id, video_input_format,
- external_renderer) == -1) {
- shared_data_->SetLastError(kViERenderUnknownError);
- return -1;
- }
-
- return frame_provider->RegisterFrameCallback(render_id, renderer);
- } else {
- // Camera or file.
- ViEInputManagerScoped is(*(shared_data_->input_manager()));
- ViEFrameProviderBase* frame_provider = is.FrameProvider(render_id);
- if (!frame_provider) {
- WEBRTC_TRACE(kTraceError, kTraceVideo, ViEId(shared_data_->instance_id()),
- "%s: FrameProvider id %d doesn't exist", __FUNCTION__,
- render_id);
- shared_data_->SetLastError(kViERenderInvalidRenderId);
- return -1;
- }
- ViERenderer* renderer = shared_data_->render_manager()->AddRenderStream(
- render_id, NULL, 0, 0.0f, 0.0f, 1.0f, 1.0f);
- if (!renderer) {
- shared_data_->SetLastError(kViERenderUnknownError);
- return -1;
- }
- if (renderer->SetExternalRenderer(render_id, video_input_format,
- external_renderer) == -1) {
- shared_data_->SetLastError(kViERenderUnknownError);
- return -1;
- }
- return frame_provider->RegisterFrameCallback(render_id, renderer);
- }
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_render_impl.h b/trunk/src/video_engine/vie_render_impl.h
deleted file mode 100644
index 0c74b3d..0000000
--- a/trunk/src/video_engine/vie_render_impl.h
+++ /dev/null
@@ -1,57 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_RENDER_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_RENDER_IMPL_H_
-
-#include "modules/video_render/main/interface/video_render_defines.h"
-#include "typedefs.h"
-#include "video_engine/include/vie_render.h"
-#include "video_engine/vie_ref_count.h"
-
-namespace webrtc {
-
-class ViESharedData;
-
-class ViERenderImpl
- : public ViERender,
- public ViERefCount {
- public:
- // Implements ViERender
- virtual int Release();
- virtual int RegisterVideoRenderModule(VideoRender& render_module);
- virtual int DeRegisterVideoRenderModule(VideoRender& render_module);
- virtual int AddRenderer(const int render_id, void* window,
- const unsigned int z_order, const float left,
- const float top, const float right,
- const float bottom);
- virtual int RemoveRenderer(const int render_id);
- virtual int StartRender(const int render_id);
- virtual int StopRender(const int render_id);
- virtual int ConfigureRender(int render_id, const unsigned int z_order,
- const float left, const float top,
- const float right, const float bottom);
- virtual int MirrorRenderStream(const int render_id, const bool enable,
- const bool mirror_xaxis,
- const bool mirror_yaxis);
- virtual int AddRenderer(const int render_id, RawVideoType video_input_format,
- ExternalRenderer* renderer);
-
- protected:
- ViERenderImpl(ViESharedData* shared_data);
- virtual ~ViERenderImpl();
-
- private:
- ViESharedData* shared_data_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_RENDER_IMPL_H_
diff --git a/trunk/src/video_engine/vie_render_manager.cc b/trunk/src/video_engine/vie_render_manager.cc
deleted file mode 100644
index 810c079..0000000
--- a/trunk/src/video_engine/vie_render_manager.cc
+++ /dev/null
@@ -1,230 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_render_manager.h"
-
-#include "engine_configurations.h"
-#include "modules/video_render/main/interface/video_render.h"
-#include "modules/video_render/main/interface/video_render_defines.h"
-#include "system_wrappers/interface/critical_section_wrapper.h"
-#include "system_wrappers/interface/rw_lock_wrapper.h"
-#include "system_wrappers/interface/trace.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_renderer.h"
-
-namespace webrtc {
-
-ViERenderManagerScoped::ViERenderManagerScoped(
- const ViERenderManager& vie_render_manager)
- : ViEManagerScopedBase(vie_render_manager) {
-}
-
-ViERenderer* ViERenderManagerScoped::Renderer(WebRtc_Word32 render_id) const {
- return static_cast<const ViERenderManager*>(vie_manager_)->ViERenderPtr(
- render_id);
-}
-
-ViERenderManager::ViERenderManager(WebRtc_Word32 engine_id)
- : list_cs_(CriticalSectionWrapper::CreateCriticalSection()),
- engine_id_(engine_id),
- use_external_render_module_(false) {
- WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideo, ViEId(engine_id),
- "ViERenderManager::ViERenderManager(engine_id: %d) - "
- "Constructor", engine_id);
-}
-
-ViERenderManager::~ViERenderManager() {
- WEBRTC_TRACE(webrtc::kTraceMemory, webrtc::kTraceVideo, ViEId(engine_id_),
- "ViERenderManager Destructor, engine_id: %d", engine_id_);
-
- while (stream_to_vie_renderer_.Size() != 0) {
- MapItem* item = stream_to_vie_renderer_.First();
- assert(item);
- const WebRtc_Word32 render_id = item->GetId();
- // The renderer is delete in RemoveRenderStream.
- item = NULL;
- RemoveRenderStream(render_id);
- }
-}
-
-WebRtc_Word32 ViERenderManager::RegisterVideoRenderModule(
- VideoRender& render_module) {
- // See if there is already a render module registered for the window that
- // the registrant render module is associated with.
- VideoRender* current_module = FindRenderModule(render_module.Window());
- if (current_module) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "A module is already registered for this window (window=%p, "
- "current module=%p, registrant module=%p.",
- render_module.Window(), current_module, &render_module);
- return -1;
- }
-
- // Register module.
- render_list_.PushBack(static_cast<void*>(&render_module));
- use_external_render_module_ = true;
- return 0;
-}
-
-WebRtc_Word32 ViERenderManager::DeRegisterVideoRenderModule(
- VideoRender& render_module) {
- // Check if there are streams in the module.
- WebRtc_UWord32 n_streams = render_module.GetNumIncomingRenderStreams();
- if (n_streams != 0) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "There are still %d streams in this module, cannot "
- "de-register", n_streams);
- return -1;
- }
-
- // Erase the render module from the map.
- ListItem* list_item = NULL;
- bool found = false;
- for (list_item = render_list_.First(); list_item != NULL;
- list_item = render_list_.Next(list_item)) {
- if (&render_module == static_cast<VideoRender*>(list_item->GetItem())) {
- // We've found our renderer.
- render_list_.Erase(list_item);
- found = true;
- break;
- }
- }
- if (!found) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "Module not registered");
- return -1;
- }
- return 0;
-}
-
-ViERenderer* ViERenderManager::AddRenderStream(const WebRtc_Word32 render_id,
- void* window,
- const WebRtc_UWord32 z_order,
- const float left,
- const float top,
- const float right,
- const float bottom) {
- CriticalSectionScoped cs(list_cs_.get());
-
- if (stream_to_vie_renderer_.Find(render_id) != NULL) {
- // This stream is already added to a renderer, not allowed!
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "Render stream already exists");
- return NULL;
- }
-
- // Get the render module for this window.
- VideoRender* render_module = FindRenderModule(window);
- if (render_module == NULL) {
- // No render module for this window, create a new one.
- render_module = VideoRender::CreateVideoRender(ViEModuleId(engine_id_, -1),
- window, false);
- if (!render_module) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo, ViEId(engine_id_),
- "Could not create new render module");
- return NULL;
- }
- render_list_.PushBack(static_cast<void*>(render_module));
- }
-
- ViERenderer* vie_renderer = ViERenderer::CreateViERenderer(render_id,
- engine_id_,
- *render_module,
- *this, z_order,
- left, top, right,
- bottom);
- if (!vie_renderer) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, render_id),
- "Could not create new render stream");
- return NULL;
- }
- stream_to_vie_renderer_.Insert(render_id, vie_renderer);
- return vie_renderer;
-}
-
-WebRtc_Word32 ViERenderManager::RemoveRenderStream(
- const WebRtc_Word32 render_id) {
- // We need exclusive right to the items in the render manager to delete a
- // stream.
- ViEManagerWriteScoped(*this);
-
- CriticalSectionScoped cs(list_cs_.get());
- MapItem* map_item = stream_to_vie_renderer_.Find(render_id);
- if (!map_item) {
- // No such stream
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo, ViEId(engine_id_),
- "No renderer for this stream found, channel_id");
- return 0;
- }
-
- ViERenderer* vie_renderer = static_cast<ViERenderer*>(map_item->GetItem());
- assert(vie_renderer);
-
- // Get the render module pointer for this vie_render object.
- VideoRender& renderer = vie_renderer->RenderModule();
-
- // Delete the vie_render.
- // This deletes the stream in the render module.
- delete vie_renderer;
-
- // Remove from the stream map.
- stream_to_vie_renderer_.Erase(map_item);
-
- // Check if there are other streams in the module.
- if (!use_external_render_module_ &&
- renderer.GetNumIncomingRenderStreams() == 0) {
- // Erase the render module from the map.
- ListItem* list_item = NULL;
- for (list_item = render_list_.First(); list_item != NULL;
- list_item = render_list_.Next(list_item)) {
- if (&renderer == static_cast<VideoRender*>(list_item->GetItem())) {
- // We've found our renderer.
- render_list_.Erase(list_item);
- break;
- }
- }
- // Destroy the module.
- VideoRender::DestroyVideoRender(&renderer);
- }
- return 0;
-}
-
-VideoRender* ViERenderManager::FindRenderModule(void* window) {
- VideoRender* renderer = NULL;
- ListItem* list_item = NULL;
- for (list_item = render_list_.First(); list_item != NULL;
- list_item = render_list_.Next(list_item)) {
- renderer = static_cast<VideoRender*>(list_item->GetItem());
- if (renderer == NULL) {
- break;
- }
- if (renderer->Window() == window) {
- // We've found the render module.
- break;
- }
- renderer = NULL;
- }
- return renderer;
-}
-
-ViERenderer* ViERenderManager::ViERenderPtr(WebRtc_Word32 render_id) const {
- ViERenderer* renderer = NULL;
- MapItem* map_item = stream_to_vie_renderer_.Find(render_id);
- if (!map_item) {
- // No such stream in any renderer.
- return NULL;
- }
- renderer = static_cast<ViERenderer*>(map_item->GetItem());
-
- return renderer;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_render_manager.h b/trunk/src/video_engine/vie_render_manager.h
deleted file mode 100644
index 242b63a..0000000
--- a/trunk/src/video_engine/vie_render_manager.h
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_RENDER_MANAGER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_RENDER_MANAGER_H_
-
-#include "system_wrappers/interface/list_wrapper.h"
-#include "system_wrappers/interface/map_wrapper.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-#include "typedefs.h"
-#include "video_engine/vie_manager_base.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class RWLockWrapper;
-class VideoRender;
-class VideoRenderCallback;
-class ViERenderer;
-
-class ViERenderManager : private ViEManagerBase {
- friend class ViERenderManagerScoped;
- public:
- explicit ViERenderManager(WebRtc_Word32 engine_id);
- ~ViERenderManager();
-
- WebRtc_Word32 RegisterVideoRenderModule(VideoRender& render_module);
- WebRtc_Word32 DeRegisterVideoRenderModule(VideoRender& render_module);
-
- ViERenderer* AddRenderStream(const WebRtc_Word32 render_id,
- void* window,
- const WebRtc_UWord32 z_order,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- WebRtc_Word32 RemoveRenderStream(WebRtc_Word32 render_id);
-
- private:
- // Returns a pointer to the render module if it exists in the render list.
- // Assumed protected.
- VideoRender* FindRenderModule(void* window);
-
- // Methods used by ViERenderScoped.
- ViERenderer* ViERenderPtr(WebRtc_Word32 render_id) const;
-
- scoped_ptr<CriticalSectionWrapper> list_cs_;
- WebRtc_Word32 engine_id_;
- MapWrapper stream_to_vie_renderer_; // Protected by ViEManagerBase.
- ListWrapper render_list_;
- bool use_external_render_module_;
-};
-
-class ViERenderManagerScoped: private ViEManagerScopedBase {
- public:
- explicit ViERenderManagerScoped(const ViERenderManager& vie_render_manager);
-
- // Returns a pointer to the ViERender object.
- ViERenderer* Renderer(WebRtc_Word32 render_id) const;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_RENDER_MANAGER_H_
diff --git a/trunk/src/video_engine/vie_renderer.cc b/trunk/src/video_engine/vie_renderer.cc
deleted file mode 100644
index 7e56b7a..0000000
--- a/trunk/src/video_engine/vie_renderer.cc
+++ /dev/null
@@ -1,235 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_renderer.h"
-
-#include "common_video/libyuv/include/libyuv.h"
-#include "modules/video_render/main/interface/video_render.h"
-#include "modules/video_render/main/interface/video_render_defines.h"
-#include "video_engine/vie_render_manager.h"
-
-namespace webrtc {
-
-ViERenderer* ViERenderer::CreateViERenderer(const WebRtc_Word32 render_id,
- const WebRtc_Word32 engine_id,
- VideoRender& render_module,
- ViERenderManager& render_manager,
- const WebRtc_UWord32 z_order,
- const float left,
- const float top,
- const float right,
- const float bottom) {
- ViERenderer* self = new ViERenderer(render_id, engine_id, render_module,
- render_manager);
- if (!self || self->Init(z_order, left, top, right, bottom) != 0) {
- delete self;
- self = NULL;
- }
- return self;
-}
-
-ViERenderer::~ViERenderer(void) {
- if (render_callback_)
- render_module_.DeleteIncomingRenderStream(render_id_);
-
- if (incoming_external_callback_)
- delete incoming_external_callback_;
-}
-
-WebRtc_Word32 ViERenderer::StartRender() {
- return render_module_.StartRender(render_id_);
-}
-WebRtc_Word32 ViERenderer::StopRender() {
- return render_module_.StopRender(render_id_);
-}
-
-WebRtc_Word32 ViERenderer::GetLastRenderedFrame(const WebRtc_Word32 renderID,
- VideoFrame& video_frame) {
- return render_module_.GetLastRenderedFrame(renderID, video_frame);
-}
-
-WebRtc_Word32 ViERenderer::ConfigureRenderer(const unsigned int z_order,
- const float left,
- const float top,
- const float right,
- const float bottom) {
- return render_module_.ConfigureRenderer(render_id_, z_order, left, top, right,
- bottom);
-}
-
-VideoRender& ViERenderer::RenderModule() {
- return render_module_;
-}
-
-WebRtc_Word32 ViERenderer::EnableMirroring(const WebRtc_Word32 render_id,
- const bool enable,
- const bool mirror_xaxis,
- const bool mirror_yaxis) {
- return render_module_.MirrorRenderStream(render_id, enable, mirror_xaxis,
- mirror_yaxis);
-}
-
-WebRtc_Word32 ViERenderer::SetTimeoutImage(const VideoFrame& timeout_image,
- const WebRtc_Word32 timeout_value) {
- return render_module_.SetTimeoutImage(render_id_, timeout_image,
- timeout_value);
-}
-
-WebRtc_Word32 ViERenderer::SetRenderStartImage(const VideoFrame& start_image) {
- return render_module_.SetStartImage(render_id_, start_image);
-}
-
-WebRtc_Word32 ViERenderer::SetExternalRenderer(
- const WebRtc_Word32 render_id,
- RawVideoType video_input_format,
- ExternalRenderer* external_renderer) {
- if (!incoming_external_callback_)
- return -1;
-
- incoming_external_callback_->SetViEExternalRenderer(external_renderer,
- video_input_format);
- return render_module_.AddExternalRenderCallback(render_id,
- incoming_external_callback_);
-}
-
-ViERenderer::ViERenderer(const WebRtc_Word32 render_id,
- const WebRtc_Word32 engine_id,
- VideoRender& render_module,
- ViERenderManager& render_manager)
- : render_id_(render_id),
- engine_id_(engine_id),
- render_module_(render_module),
- render_manager_(render_manager),
- render_callback_(NULL),
- incoming_external_callback_(new ViEExternalRendererImpl()) {
-}
-
-WebRtc_Word32 ViERenderer::Init(const WebRtc_UWord32 z_order,
- const float left,
- const float top,
- const float right,
- const float bottom) {
- render_callback_ =
- static_cast<VideoRenderCallback*>(render_module_.AddIncomingRenderStream(
- render_id_, z_order, left, top, right, bottom));
- if (!render_callback_) {
- // Logging done.
- return -1;
- }
- return 0;
-}
-
-void ViERenderer::DeliverFrame(int id,
- VideoFrame& video_frame,
- int num_csrcs,
- const WebRtc_UWord32 CSRC[kRtpCsrcSize]) {
- render_callback_->RenderFrame(render_id_, video_frame);
-}
-
-void ViERenderer::DelayChanged(int id, int frame_delay) {}
-
-int ViERenderer::GetPreferedFrameSettings(int& width,
- int& height,
- int& frame_rate) {
- return -1;
-}
-
-void ViERenderer::ProviderDestroyed(int id) {
- // Remove the render stream since the provider is destroyed.
- render_manager_.RemoveRenderStream(render_id_);
-}
-
-ViEExternalRendererImpl::ViEExternalRendererImpl()
- : external_renderer_(NULL),
- external_renderer_format_(kVideoUnknown),
- external_renderer_width_(0),
- external_renderer_height_(0),
- converted_frame_(new VideoFrame()){
-}
-
-int ViEExternalRendererImpl::SetViEExternalRenderer(
- ExternalRenderer* external_renderer,
- RawVideoType video_input_format) {
- external_renderer_ = external_renderer;
- external_renderer_format_ = video_input_format;
- return 0;
-}
-
-WebRtc_Word32 ViEExternalRendererImpl::RenderFrame(
- const WebRtc_UWord32 stream_id,
- VideoFrame& video_frame) {
- VideoFrame* out_frame = converted_frame_.get();
-
- // Convert to requested format.
- VideoType type =
- RawVideoTypeToCommonVideoVideoType(external_renderer_format_);
- int buffer_size = CalcBufferSize(type, video_frame.Width(),
- video_frame.Height());
- if (buffer_size <= 0) {
- // Unsupported video format.
- assert(false);
- return -1;
- }
- converted_frame_->VerifyAndAllocate(buffer_size);
-
- switch (external_renderer_format_) {
- case kVideoI420:
- out_frame = &video_frame;
- break;
- case kVideoYV12:
- case kVideoYUY2:
- case kVideoUYVY:
- case kVideoARGB:
- case kVideoRGB24:
- {
- ConvertFromI420(video_frame.Buffer(), video_frame.Width(), type, 0,
- video_frame.Width(), video_frame.Height(),
- converted_frame_->Buffer());
- }
- break;
- case kVideoIYUV:
- // no conversion available
- break;
- case kVideoRGB565:
- ConvertI420ToRGB565(video_frame.Buffer(), converted_frame_->Buffer(),
- video_frame.Width(), video_frame.Height());
- break;
- case kVideoARGB4444:
- ConvertI420ToARGB4444(video_frame.Buffer(), converted_frame_->Buffer(),
- video_frame.Width(), video_frame.Height(), 0);
- break;
- case kVideoARGB1555 :
- ConvertI420ToARGB1555(video_frame.Buffer(), converted_frame_->Buffer(),
- video_frame.Width(), video_frame.Height(), 0);
- break;
- default:
- assert(false);
- out_frame = NULL;
- break;
- }
-
- if (external_renderer_width_ != video_frame.Width() ||
- external_renderer_height_ != video_frame.Height()) {
- external_renderer_width_ = video_frame.Width();
- external_renderer_height_ = video_frame.Height();
- external_renderer_->FrameSizeChange(external_renderer_width_,
- external_renderer_height_, stream_id);
- }
-
- if (out_frame) {
- external_renderer_->DeliverFrame(out_frame->Buffer(),
- out_frame->Length(),
- video_frame.TimeStamp(),
- video_frame.RenderTimeMs());
- }
- return 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_renderer.h b/trunk/src/video_engine/vie_renderer.h
deleted file mode 100644
index 028d72b..0000000
--- a/trunk/src/video_engine/vie_renderer.h
+++ /dev/null
@@ -1,117 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_RENDERER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_RENDERER_H_
-
-#include "modules/video_render/main/interface/video_render_defines.h"
-#include "system_wrappers/interface/map_wrapper.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-#include "video_engine/include/vie_render.h"
-#include "video_engine/vie_frame_provider_base.h"
-
-namespace webrtc {
-
-class VideoRender;
-class VideoRenderCallback;
-class ViERenderManager;
-
-class ViEExternalRendererImpl : public VideoRenderCallback {
- public:
- ViEExternalRendererImpl();
- virtual ~ViEExternalRendererImpl() {}
-
- int SetViEExternalRenderer(ExternalRenderer* external_renderer,
- RawVideoType video_input_format);
-
- // Implements VideoRenderCallback.
- virtual WebRtc_Word32 RenderFrame(const WebRtc_UWord32 stream_id,
- VideoFrame& video_frame);
-
- private:
- ExternalRenderer* external_renderer_;
- RawVideoType external_renderer_format_;
- WebRtc_UWord32 external_renderer_width_;
- WebRtc_UWord32 external_renderer_height_;
- scoped_ptr<VideoFrame> converted_frame_;
-};
-
-class ViERenderer: public ViEFrameCallback {
- public:
- static ViERenderer* CreateViERenderer(const WebRtc_Word32 render_id,
- const WebRtc_Word32 engine_id,
- VideoRender& render_module,
- ViERenderManager& render_manager,
- const WebRtc_UWord32 z_order,
- const float left,
- const float top,
- const float right,
- const float bottom);
- ~ViERenderer(void);
-
- WebRtc_Word32 StartRender();
- WebRtc_Word32 StopRender();
-
- WebRtc_Word32 GetLastRenderedFrame(const WebRtc_Word32 renderID,
- VideoFrame& video_frame);
-
- WebRtc_Word32 ConfigureRenderer(const unsigned int z_order,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- VideoRender& RenderModule();
-
- WebRtc_Word32 EnableMirroring(const WebRtc_Word32 render_id,
- const bool enable,
- const bool mirror_xaxis,
- const bool mirror_yaxis);
-
- WebRtc_Word32 SetTimeoutImage(const VideoFrame& timeout_image,
- const WebRtc_Word32 timeout_value);
- WebRtc_Word32 SetRenderStartImage(const VideoFrame& start_image);
- WebRtc_Word32 SetExternalRenderer(const WebRtc_Word32 render_id,
- RawVideoType video_input_format,
- ExternalRenderer* external_renderer);
-
- private:
- ViERenderer(const WebRtc_Word32 render_id, const WebRtc_Word32 engine_id,
- VideoRender& render_module,
- ViERenderManager& render_manager);
-
- WebRtc_Word32 Init(const WebRtc_UWord32 z_order,
- const float left,
- const float top,
- const float right,
- const float bottom);
-
- // Implement ViEFrameCallback
- virtual void DeliverFrame(int id,
- VideoFrame& video_frame,
- int num_csrcs = 0,
- const WebRtc_UWord32 CSRC[kRtpCsrcSize] = NULL);
- virtual void DelayChanged(int id, int frame_delay);
- virtual int GetPreferedFrameSettings(int& width,
- int& height,
- int& frame_rate);
- virtual void ProviderDestroyed(int id);
-
- WebRtc_UWord32 render_id_;
- WebRtc_Word32 engine_id_;
- VideoRender& render_module_;
- ViERenderManager& render_manager_;
- VideoRenderCallback* render_callback_;
- ViEExternalRendererImpl* incoming_external_callback_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_RENDERER_H_
diff --git a/trunk/src/video_engine/vie_rtp_rtcp_impl.cc b/trunk/src/video_engine/vie_rtp_rtcp_impl.cc
deleted file mode 100644
index 497481c..0000000
--- a/trunk/src/video_engine/vie_rtp_rtcp_impl.cc
+++ /dev/null
@@ -1,957 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "video_engine/vie_rtp_rtcp_impl.h"
-
-#include "engine_configurations.h"
-#include "system_wrappers/interface/file_wrapper.h"
-#include "system_wrappers/interface/trace.h"
-#include "video_engine/include/vie_errors.h"
-#include "video_engine/vie_channel.h"
-#include "video_engine/vie_channel_manager.h"
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_encoder.h"
-#include "video_engine/vie_impl.h"
-#include "video_engine/vie_shared_data.h"
-
-namespace webrtc {
-
-// Helper methods for converting between module format and ViE API format.
-
-static RTCPMethod ViERTCPModeToRTCPMethod(ViERTCPMode api_mode) {
- switch (api_mode) {
- case kRtcpNone:
- return kRtcpOff;
-
- case kRtcpCompound_RFC4585:
- return kRtcpCompound;
-
- case kRtcpNonCompound_RFC5506:
- return kRtcpNonCompound;
- }
- assert(false);
- return kRtcpOff;
-}
-
-static ViERTCPMode RTCPMethodToViERTCPMode(RTCPMethod module_method) {
- switch (module_method) {
- case kRtcpOff:
- return kRtcpNone;
-
- case kRtcpCompound:
- return kRtcpCompound_RFC4585;
-
- case kRtcpNonCompound:
- return kRtcpNonCompound_RFC5506;
- }
- assert(false);
- return kRtcpNone;
-}
-
-static KeyFrameRequestMethod APIRequestToModuleRequest(
- ViEKeyFrameRequestMethod api_method) {
- switch (api_method) {
- case kViEKeyFrameRequestNone:
- return kKeyFrameReqFirRtp;
-
- case kViEKeyFrameRequestPliRtcp:
- return kKeyFrameReqPliRtcp;
-
- case kViEKeyFrameRequestFirRtp:
- return kKeyFrameReqFirRtp;
-
- case kViEKeyFrameRequestFirRtcp:
- return kKeyFrameReqFirRtcp;
- }
- assert(false);
- return kKeyFrameReqFirRtp;
-}
-
-ViERTP_RTCP* ViERTP_RTCP::GetInterface(VideoEngine* video_engine) {
-#ifdef WEBRTC_VIDEO_ENGINE_RTP_RTCP_API
- if (!video_engine) {
- return NULL;
- }
- VideoEngineImpl* vie_impl = reinterpret_cast<VideoEngineImpl*>(video_engine);
- ViERTP_RTCPImpl* vie_rtpimpl = vie_impl;
- // Increase ref count.
- (*vie_rtpimpl)++;
- return vie_rtpimpl;
-#else
- return NULL;
-#endif
-}
-
-int ViERTP_RTCPImpl::Release() {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo, shared_data_->instance_id(),
- "ViERTP_RTCP::Release()");
- // Decrease ref count.
- (*this)--;
-
- WebRtc_Word32 ref_count = GetCount();
- if (ref_count < 0) {
- WEBRTC_TRACE(kTraceWarning, kTraceVideo, shared_data_->instance_id(),
- "ViERTP_RTCP release too many times");
- shared_data_->SetLastError(kViEAPIDoesNotExist);
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVideo, shared_data_->instance_id(),
- "ViERTP_RTCP reference count: %d", ref_count);
- return ref_count;
-}
-
-ViERTP_RTCPImpl::ViERTP_RTCPImpl(ViESharedData* shared_data)
- : shared_data_(shared_data) {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViERTP_RTCPImpl::ViERTP_RTCPImpl() Ctor");
-}
-
-ViERTP_RTCPImpl::~ViERTP_RTCPImpl() {
- WEBRTC_TRACE(kTraceMemory, kTraceVideo, shared_data_->instance_id(),
- "ViERTP_RTCPImpl::~ViERTP_RTCPImpl() Dtor");
-}
-
-int ViERTP_RTCPImpl::SetLocalSSRC(const int video_channel,
- const unsigned int SSRC,
- const StreamType usage,
- const unsigned char simulcast_idx) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, SSRC: %d)", __FUNCTION__, video_channel, SSRC);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- // The channel doesn't exists
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->SetSSRC(SSRC, usage, simulcast_idx) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::SetRemoteSSRCType(const int videoChannel,
- const StreamType usage,
- const unsigned int SSRC) const {
- WEBRTC_TRACE(webrtc::kTraceApiCall, webrtc::kTraceVideo,
- ViEId(shared_data_->instance_id(), videoChannel),
- "%s(channel: %d, usage:%d SSRC: 0x%x)",
- __FUNCTION__, usage, videoChannel, SSRC);
-
- // Get the channel
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* ptrViEChannel = cs.Channel(videoChannel);
- if (ptrViEChannel == NULL) {
- // The channel doesn't exists
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(shared_data_->instance_id(), videoChannel),
- "%s: Channel %d doesn't exist",
- __FUNCTION__, videoChannel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (ptrViEChannel->SetRemoteSSRCType(usage, SSRC) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::GetLocalSSRC(const int video_channel,
- unsigned int& SSRC) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, SSRC: %d)", __FUNCTION__, video_channel, SSRC);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetLocalSSRC((WebRtc_UWord32&) SSRC) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::GetRemoteSSRC(const int video_channel,
- unsigned int& SSRC) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel, SSRC);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetRemoteSSRC(SSRC) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::GetRemoteCSRCs(const int video_channel,
- unsigned int CSRCs[kRtpCsrcSize]) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetRemoteCSRC(CSRCs) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::SetStartSequenceNumber(const int video_channel,
- unsigned short sequence_number) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, sequence_number: %u)", __FUNCTION__,
- video_channel, sequence_number);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->Sending()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d already sending.", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViERtpRtcpAlreadySending);
- return -1;
- }
- if (vie_channel->SetStartSequenceNumber(sequence_number) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::SetRTCPStatus(const int video_channel,
- const ViERTCPMode rtcp_mode) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, mode: %d)", __FUNCTION__, video_channel,
- rtcp_mode);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
-
- RTCPMethod module_mode = ViERTCPModeToRTCPMethod(rtcp_mode);
- if (vie_channel->SetRTCPMode(module_mode) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::GetRTCPStatus(const int video_channel,
- ViERTCPMode& rtcp_mode) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel, rtcp_mode);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- RTCPMethod module_mode = kRtcpOff;
- if (vie_channel->GetRTCPMode(module_mode) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: could not get current RTCP mode", __FUNCTION__);
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- rtcp_mode = RTCPMethodToViERTCPMode(module_mode);
- return 0;
-}
-
-int ViERTP_RTCPImpl::SetRTCPCName(const int video_channel,
- const char rtcp_cname[KMaxRTCPCNameLength]) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, name: %s)", __FUNCTION__, video_channel,
- rtcp_cname);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->Sending()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d already sending.", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViERtpRtcpAlreadySending);
- return -1;
- }
- if (vie_channel->SetRTCPCName(rtcp_cname) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::GetRTCPCName(const int video_channel,
- char rtcp_cname[KMaxRTCPCNameLength]) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetRTCPCName(rtcp_cname) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::GetRemoteRTCPCName(
- const int video_channel,
- char rtcp_cname[KMaxRTCPCNameLength]) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetRemoteRTCPCName(rtcp_cname) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::SendApplicationDefinedRTCPPacket(
- const int video_channel,
- const unsigned char sub_type,
- unsigned int name,
- const char* data,
- unsigned short data_length_in_bytes) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, sub_type: %c, name: %d, data: x, length: %u)",
- __FUNCTION__, video_channel, sub_type, name,
- data_length_in_bytes);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (!vie_channel->Sending()) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d not sending", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpNotSending);
- return -1;
- }
- RTCPMethod method;
- if (vie_channel->GetRTCPMode(method) != 0 || method == kRtcpOff) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: RTCP disabled on channel %d.", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViERtpRtcpRtcpDisabled);
- return -1;
- }
- if (vie_channel->SendApplicationDefinedRTCPPacket(
- sub_type, name, reinterpret_cast<const WebRtc_UWord8*>(data),
- data_length_in_bytes) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::SetNACKStatus(const int video_channel, const bool enable) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, enable: %d)", __FUNCTION__, video_channel,
- enable);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->SetNACKStatus(enable) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: failed for channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
-
- // Update the encoder
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Could not get encoder for channel %d", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- vie_encoder->UpdateProtectionMethod();
- return 0;
-}
-
-int ViERTP_RTCPImpl::SetFECStatus(const int video_channel, const bool enable,
- const unsigned char payload_typeRED,
- const unsigned char payload_typeFEC) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, enable: %d, payload_typeRED: %u, "
- "payloadTypeFEC: %u)",
- __FUNCTION__, video_channel, enable, payload_typeRED,
- payload_typeFEC);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->SetFECStatus(enable, payload_typeRED,
- payload_typeFEC) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: failed for channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- // Update the encoder.
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Could not get encoder for channel %d", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- vie_encoder->UpdateProtectionMethod();
- return 0;
-}
-
-int ViERTP_RTCPImpl::SetHybridNACKFECStatus(
- const int video_channel,
- const bool enable,
- const unsigned char payload_typeRED,
- const unsigned char payload_typeFEC) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, enable: %d, payload_typeRED: %u, "
- "payloadTypeFEC: %u)",
- __FUNCTION__, video_channel, enable, payload_typeRED,
- payload_typeFEC);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
-
- // Update the channel status with hybrid NACK FEC mode.
- if (vie_channel->SetHybridNACKFECStatus(enable, payload_typeRED,
- payload_typeFEC) != 0) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: failed for channel %d", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
-
- // Update the encoder.
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Could not get encoder for channel %d", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- vie_encoder->UpdateProtectionMethod();
- return 0;
-}
-
-int ViERTP_RTCPImpl::SetKeyFrameRequestMethod(
- const int video_channel,
- const ViEKeyFrameRequestMethod method) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, method: %d)", __FUNCTION__, video_channel,
- method);
-
- // Get the channel.
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- KeyFrameRequestMethod module_method = APIRequestToModuleRequest(method);
- if (vie_channel->SetKeyFrameRequestMethod(module_method) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::SetTMMBRStatus(const int video_channel,
- const bool enable) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, enable: %d)", __FUNCTION__, video_channel,
- enable);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->EnableTMMBR(enable) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::SetRembStatus(int video_channel, bool sender,
- bool receiver) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "ViERTP_RTCPImpl::SetRembStatus(%d, %d, %d)", video_channel,
- sender, receiver);
- if (!shared_data_->channel_manager()->SetRembStatus(video_channel, sender,
- receiver)) {
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::GetReceivedRTCPStatistics(const int video_channel,
- unsigned short& fraction_lost,
- unsigned int& cumulative_lost,
- unsigned int& extended_max,
- unsigned int& jitter,
- int& rtt_ms) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetReceivedRtcpStatistics(
- static_cast<WebRtc_UWord16&>(fraction_lost),
- static_cast<WebRtc_UWord32&>(cumulative_lost),
- static_cast<WebRtc_UWord32&>(extended_max),
- static_cast<WebRtc_UWord32&>(jitter),
- static_cast<WebRtc_Word32&>(rtt_ms)) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::GetSentRTCPStatistics(const int video_channel,
- unsigned short& fraction_lost,
- unsigned int& cumulative_lost,
- unsigned int& extended_max,
- unsigned int& jitter,
- int& rtt_ms) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
-
- if (vie_channel->GetSendRtcpStatistics(
- static_cast<WebRtc_UWord16&>(fraction_lost),
- static_cast<WebRtc_UWord32&>(cumulative_lost),
- static_cast<WebRtc_UWord32&>(extended_max),
- static_cast<WebRtc_UWord32&>(jitter),
- static_cast<WebRtc_Word32&>(rtt_ms)) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::GetRTPStatistics(const int video_channel,
- unsigned int& bytes_sent,
- unsigned int& packets_sent,
- unsigned int& bytes_received,
- unsigned int& packets_received) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->GetRtpStatistics(
- static_cast<WebRtc_UWord32&>(bytes_sent),
- static_cast<WebRtc_UWord32&>(packets_sent),
- static_cast<WebRtc_UWord32&>(bytes_received),
- static_cast<WebRtc_UWord32&>(packets_received)) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::GetBandwidthUsage(const int video_channel,
- unsigned int& total_bitrate_sent,
- unsigned int& video_bitrate_sent,
- unsigned int& fec_bitrate_sent,
- unsigned int& nackBitrateSent) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- vie_channel->GetBandwidthUsage(
- static_cast<WebRtc_UWord32&>(total_bitrate_sent),
- static_cast<WebRtc_UWord32&>(video_bitrate_sent),
- static_cast<WebRtc_UWord32&>(fec_bitrate_sent),
- static_cast<WebRtc_UWord32&>(nackBitrateSent));
- return 0;
-}
-
-int ViERTP_RTCPImpl::GetEstimatedSendBandwidth(
- const int video_channel,
- unsigned int* estimated_bandwidth) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEEncoder* vie_encoder = cs.Encoder(video_channel);
- if (!vie_encoder) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Could not get encoder for channel %d", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- return vie_encoder->EstimatedSendBandwidth(
- static_cast<WebRtc_UWord32*>(estimated_bandwidth));
-}
-
-int ViERTP_RTCPImpl::GetEstimatedReceiveBandwidth(
- const int video_channel,
- unsigned int* estimated_bandwidth) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Could not get channel %d", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- return vie_channel->GetEstimatedReceiveBandwidth(
- static_cast<WebRtc_UWord32*>(estimated_bandwidth));
-}
-
-int ViERTP_RTCPImpl::SetRTPKeepAliveStatus(
- const int video_channel,
- bool enable,
- const char unknown_payload_type,
- const unsigned int delta_transmit_time_seconds) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, enable: %d, unknown_payload_type: %d, "
- "deltaTransmitTimeMS: %ul)",
- __FUNCTION__, video_channel, enable,
- static_cast<int>(unknown_payload_type),
- delta_transmit_time_seconds);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__,
- video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- WebRtc_UWord16 delta_transmit_time_ms = 1000 * delta_transmit_time_seconds;
- if (vie_channel->SetKeepAliveStatus(enable, unknown_payload_type,
- delta_transmit_time_ms) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::GetRTPKeepAliveStatus(
- const int video_channel,
- bool& enabled,
- char& unknown_payload_type,
- unsigned int& delta_transmit_time_seconds) const {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
-
- WebRtc_UWord16 delta_time_ms = 0;
- int ret_val = vie_channel->GetKeepAliveStatus(enabled, unknown_payload_type,
- delta_time_ms);
- delta_transmit_time_seconds = delta_time_ms / 1000;
- if (ret_val != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- }
- return ret_val;
-}
-
-int ViERTP_RTCPImpl::StartRTPDump(const int video_channel,
- const char file_nameUTF8[1024],
- RTPDirections direction) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, file_name: %s, direction: %d)", __FUNCTION__,
- video_channel, file_nameUTF8, direction);
- assert(FileWrapper::kMaxFileNameSize == 1024);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->StartRTPDump(file_nameUTF8, direction) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::StopRTPDump(const int video_channel,
- RTPDirections direction) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d, direction: %d)", __FUNCTION__, video_channel,
- direction);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->StopRTPDump(direction) != 0) {
- shared_data_->SetLastError(kViERtpRtcpUnknownError);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::RegisterRTPObserver(const int video_channel,
- ViERTPObserver& observer) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->RegisterRtpObserver(&observer) != 0) {
- shared_data_->SetLastError(kViERtpRtcpObserverAlreadyRegistered);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::DeregisterRTPObserver(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->RegisterRtpObserver(NULL) != 0) {
- shared_data_->SetLastError(kViERtpRtcpObserverNotRegistered);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::RegisterRTCPObserver(const int video_channel,
- ViERTCPObserver& observer) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->RegisterRtcpObserver(&observer) != 0) {
- shared_data_->SetLastError(kViERtpRtcpObserverAlreadyRegistered);
- return -1;
- }
- return 0;
-}
-
-int ViERTP_RTCPImpl::DeregisterRTCPObserver(const int video_channel) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s(channel: %d)", __FUNCTION__, video_channel);
- ViEChannelManagerScoped cs(*(shared_data_->channel_manager()));
- ViEChannel* vie_channel = cs.Channel(video_channel);
- if (!vie_channel) {
- WEBRTC_TRACE(kTraceError, kTraceVideo,
- ViEId(shared_data_->instance_id(), video_channel),
- "%s: Channel %d doesn't exist", __FUNCTION__, video_channel);
- shared_data_->SetLastError(kViERtpRtcpInvalidChannelId);
- return -1;
- }
- if (vie_channel->RegisterRtcpObserver(NULL) != 0) {
- shared_data_->SetLastError(kViERtpRtcpObserverNotRegistered);
- return -1;
- }
- return 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_rtp_rtcp_impl.h b/trunk/src/video_engine/vie_rtp_rtcp_impl.h
deleted file mode 100644
index 72e8131..0000000
--- a/trunk/src/video_engine/vie_rtp_rtcp_impl.h
+++ /dev/null
@@ -1,127 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_RTP_RTCP_IMPL_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_RTP_RTCP_IMPL_H_
-
-#include "modules/rtp_rtcp/interface/rtp_rtcp_defines.h"
-#include "typedefs.h"
-#include "video_engine/include/vie_rtp_rtcp.h"
-#include "video_engine/vie_ref_count.h"
-
-namespace webrtc {
-
-class ViESharedData;
-
-class ViERTP_RTCPImpl
- : public ViERTP_RTCP,
- public ViERefCount {
- public:
- // Implements ViERTP_RTCP.
- virtual int Release();
- virtual int SetLocalSSRC(const int video_channel,
- const unsigned int SSRC,
- const StreamType usage,
- const unsigned char simulcast_idx);
- virtual int GetLocalSSRC(const int video_channel, unsigned int& SSRC) const;
- virtual int SetRemoteSSRCType(const int video_channel,
- const StreamType usage,
- const unsigned int SSRC) const;
- virtual int GetRemoteSSRC(const int video_channel, unsigned int& SSRC) const;
- virtual int GetRemoteCSRCs(const int video_channel,
- unsigned int CSRCs[kRtpCsrcSize]) const;
- virtual int SetStartSequenceNumber(const int video_channel,
- unsigned short sequence_number);
- virtual int SetRTCPStatus(const int video_channel,
- const ViERTCPMode rtcp_mode);
- virtual int GetRTCPStatus(const int video_channel,
- ViERTCPMode& rtcp_mode) const;
- virtual int SetRTCPCName(const int video_channel,
- const char rtcp_cname[KMaxRTCPCNameLength]);
- virtual int GetRTCPCName(const int video_channel,
- char rtcp_cname[KMaxRTCPCNameLength]) const;
- virtual int GetRemoteRTCPCName(const int video_channel,
- char rtcp_cname[KMaxRTCPCNameLength]) const;
- virtual int SendApplicationDefinedRTCPPacket(
- const int video_channel,
- const unsigned char sub_type,
- unsigned int name,
- const char* data,
- unsigned short data_length_in_bytes);
- virtual int SetNACKStatus(const int video_channel, const bool enable);
- virtual int SetFECStatus(const int video_channel, const bool enable,
- const unsigned char payload_typeRED,
- const unsigned char payload_typeFEC);
- virtual int SetHybridNACKFECStatus(const int video_channel, const bool enable,
- const unsigned char payload_typeRED,
- const unsigned char payload_typeFEC);
- virtual int SetKeyFrameRequestMethod(const int video_channel,
- const ViEKeyFrameRequestMethod method);
- virtual int SetTMMBRStatus(const int video_channel, const bool enable);
- virtual int SetRembStatus(int video_channel, bool sender, bool receiver);
- virtual int GetReceivedRTCPStatistics(const int video_channel,
- unsigned short& fraction_lost,
- unsigned int& cumulative_lost,
- unsigned int& extended_max,
- unsigned int& jitter,
- int& rtt_ms) const;
- virtual int GetSentRTCPStatistics(const int video_channel,
- unsigned short& fraction_lost,
- unsigned int& cumulative_lost,
- unsigned int& extended_max,
- unsigned int& jitter, int& rtt_ms) const;
- virtual int GetRTPStatistics(const int video_channel,
- unsigned int& bytes_sent,
- unsigned int& packets_sent,
- unsigned int& bytes_received,
- unsigned int& packets_received) const;
- virtual int GetBandwidthUsage(const int video_channel,
- unsigned int& total_bitrate_sent,
- unsigned int& video_bitrate_sent,
- unsigned int& fec_bitrate_sent,
- unsigned int& nackBitrateSent) const;
- virtual int GetEstimatedSendBandwidth(
- const int video_channel,
- unsigned int* estimated_bandwidth) const;
- virtual int GetEstimatedReceiveBandwidth(
- const int video_channel,
- unsigned int* estimated_bandwidth) const;
- virtual int SetRTPKeepAliveStatus(
- const int video_channel,
- bool enable,
- const char unknown_payload_type,
- const unsigned int delta_transmit_time_seconds);
- virtual int GetRTPKeepAliveStatus(
- const int video_channel,
- bool& enabled,
- char& unkown_payload_type,
- unsigned int& delta_transmit_time_seconds) const;
- virtual int StartRTPDump(const int video_channel,
- const char file_nameUTF8[1024],
- RTPDirections direction);
- virtual int StopRTPDump(const int video_channel, RTPDirections direction);
- virtual int RegisterRTPObserver(const int video_channel,
- ViERTPObserver& observer);
- virtual int DeregisterRTPObserver(const int video_channel);
- virtual int RegisterRTCPObserver(const int video_channel,
- ViERTCPObserver& observer);
- virtual int DeregisterRTCPObserver(const int video_channel);
-
- protected:
- ViERTP_RTCPImpl(ViESharedData* shared_data);
- virtual ~ViERTP_RTCPImpl();
-
- private:
- ViESharedData* shared_data_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_RTP_RTCP_IMPL_H_
diff --git a/trunk/src/video_engine/vie_sender.cc b/trunk/src/video_engine/vie_sender.cc
deleted file mode 100644
index e0fe29b..0000000
--- a/trunk/src/video_engine/vie_sender.cc
+++ /dev/null
@@ -1,205 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cassert>
-
-#include "critical_section_wrapper.h"
-#include "rtp_dump.h"
-#include "vie_sender.h"
-#include "trace.h"
-
-namespace webrtc {
-
-ViESender::ViESender(int engine_id, int channel_id)
- : engine_id_(engine_id),
- channel_id_(channel_id),
- critsect_(CriticalSectionWrapper::CreateCriticalSection()),
- external_encryption_(NULL),
- encryption_buffer_(NULL),
- transport_(NULL),
- rtp_dump_(NULL) {
-}
-
-ViESender::~ViESender() {
- if (encryption_buffer_) {
- delete[] encryption_buffer_;
- encryption_buffer_ = NULL;
- }
-
- if (rtp_dump_) {
- rtp_dump_->Stop();
- RtpDump::DestroyRtpDump(rtp_dump_);
- rtp_dump_ = NULL;
- }
-}
-
-int ViESender::RegisterExternalEncryption(Encryption* encryption) {
- CriticalSectionScoped cs(critsect_.get());
- if (external_encryption_) {
- return -1;
- }
- encryption_buffer_ = new WebRtc_UWord8[kViEMaxMtu];
- if (encryption_buffer_ == NULL) {
- return -1;
- }
- external_encryption_ = encryption;
- return 0;
-}
-
-int ViESender::DeregisterExternalEncryption() {
- CriticalSectionScoped cs(critsect_.get());
- if (external_encryption_ == NULL) {
- return -1;
- }
- if (encryption_buffer_) {
- delete encryption_buffer_;
- encryption_buffer_ = NULL;
- }
- external_encryption_ = NULL;
- return 0;
-}
-
-int ViESender::RegisterSendTransport(Transport* transport) {
- CriticalSectionScoped cs(critsect_.get());
- if (transport_) {
- return -1;
- }
- transport_ = transport;
- return 0;
-}
-
-int ViESender::DeregisterSendTransport() {
- CriticalSectionScoped cs(critsect_.get());
- if (transport_ == NULL) {
- return -1;
- }
- transport_ = NULL;
- return 0;
-}
-
-int ViESender::StartRTPDump(const char file_nameUTF8[1024]) {
- CriticalSectionScoped cs(critsect_.get());
- if (rtp_dump_) {
- // Packet dump is already started, restart it.
- rtp_dump_->Stop();
- } else {
- rtp_dump_ = RtpDump::CreateRtpDump();
- if (rtp_dump_ == NULL) {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "StartSRTPDump: Failed to create RTP dump");
- return -1;
- }
- }
- if (rtp_dump_->Start(file_nameUTF8) != 0) {
- RtpDump::DestroyRtpDump(rtp_dump_);
- rtp_dump_ = NULL;
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "StartRTPDump: Failed to start RTP dump");
- return -1;
- }
- return 0;
-}
-
-int ViESender::StopRTPDump() {
- CriticalSectionScoped cs(critsect_.get());
- if (rtp_dump_) {
- if (rtp_dump_->IsActive()) {
- rtp_dump_->Stop();
- } else {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "StopRTPDump: Dump not active");
- }
- RtpDump::DestroyRtpDump(rtp_dump_);
- rtp_dump_ = NULL;
- } else {
- WEBRTC_TRACE(webrtc::kTraceError, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "StopRTPDump: RTP dump not started");
- return -1;
- }
- return 0;
-}
-
-int ViESender::SendPacket(int vie_id, const void* data, int len) {
- CriticalSectionScoped cs(critsect_.get());
- if (!transport_) {
- // No transport
- return -1;
- }
-
- assert(ChannelId(vie_id) == channel_id_);
-
- // TODO(mflodman) Change decrypt to get rid of this cast.
- void* tmp_ptr = const_cast<void*>(data);
- unsigned char* send_packet = static_cast<unsigned char*>(tmp_ptr);
- int send_packet_length = len;
-
- if (rtp_dump_) {
- rtp_dump_->DumpPacket(send_packet, send_packet_length);
- }
-
- if (external_encryption_) {
- external_encryption_->encrypt(channel_id_, send_packet,
- encryption_buffer_, send_packet_length,
- static_cast<int*>(&send_packet_length));
- send_packet = encryption_buffer_;
- }
-
- const int bytes_sent = transport_->SendPacket(channel_id_, send_packet,
- send_packet_length);
- if (bytes_sent != send_packet_length) {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "ViESender::SendPacket - Transport failed to send RTP packet");
- }
- return bytes_sent;
-}
-
-int ViESender::SendRTCPPacket(int vie_id, const void* data, int len) {
- CriticalSectionScoped cs(critsect_.get());
-
- if (!transport_) {
- return -1;
- }
-
- assert(ChannelId(vie_id) == channel_id_);
-
- // Prepare for possible encryption and sending.
- // TODO(mflodman) Change decrypt to get rid of this cast.
- void* tmp_ptr = const_cast<void*>(data);
- unsigned char* send_packet = static_cast<unsigned char*>(tmp_ptr);
- int send_packet_length = len;
-
- if (rtp_dump_) {
- rtp_dump_->DumpPacket(send_packet, send_packet_length);
- }
-
- if (external_encryption_) {
- external_encryption_->encrypt_rtcp(
- channel_id_, send_packet, encryption_buffer_, send_packet_length,
- static_cast<int*>(&send_packet_length));
- send_packet = encryption_buffer_;
- }
-
- const int bytes_sent = transport_->SendRTCPPacket(channel_id_, send_packet,
- send_packet_length);
- if (bytes_sent != send_packet_length) {
- WEBRTC_TRACE(
- webrtc::kTraceWarning, webrtc::kTraceVideo,
- ViEId(engine_id_, channel_id_),
- "ViESender::SendRTCPPacket - Transport failed to send RTCP packet");
- }
- return bytes_sent;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_sender.h b/trunk/src/video_engine/vie_sender.h
deleted file mode 100644
index a60446d..0000000
--- a/trunk/src/video_engine/vie_sender.h
+++ /dev/null
@@ -1,65 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// ViESender is responsible for encrypting, if enabled, packets and send to
-// network.
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_
-
-#include "common_types.h"
-#include "engine_configurations.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-#include "typedefs.h"
-#include "vie_defines.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class RtpDump;
-class Transport;
-class VideoCodingModule;
-
-class ViESender: public Transport {
- public:
- ViESender(int engine_id, int channel_id);
- ~ViESender();
-
- // Registers an encryption class to use before sending packets.
- int RegisterExternalEncryption(Encryption* encryption);
- int DeregisterExternalEncryption();
-
- // Registers transport to use for sending RTP and RTCP.
- int RegisterSendTransport(Transport* transport);
- int DeregisterSendTransport();
-
- // Stores all incoming packets to file.
- int StartRTPDump(const char file_nameUTF8[1024]);
- int StopRTPDump();
-
- // Implements Transport.
- virtual int SendPacket(int vie_id, const void* data, int len);
- virtual int SendRTCPPacket(int vie_id, const void* data, int len);
-
- private:
- int engine_id_;
- int channel_id_;
-
- scoped_ptr<CriticalSectionWrapper> critsect_;
-
- Encryption* external_encryption_;
- WebRtc_UWord8* encryption_buffer_;
- Transport* transport_;
- RtpDump* rtp_dump_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_SENDER_H_
diff --git a/trunk/src/video_engine/vie_shared_data.cc b/trunk/src/video_engine/vie_shared_data.cc
deleted file mode 100644
index 915bcfd..0000000
--- a/trunk/src/video_engine/vie_shared_data.cc
+++ /dev/null
@@ -1,80 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "cpu_info.h"
-#include "process_thread.h"
-#include "trace.h"
-#include "vie_channel_manager.h"
-#include "vie_defines.h"
-#include "vie_input_manager.h"
-#include "vie_render_manager.h"
-#include "vie_shared_data.h"
-
-namespace webrtc {
-
-// Active instance counter
-int ViESharedData::instance_counter_ = 0;
-
-ViESharedData::ViESharedData()
- : instance_id_(++instance_counter_),
- initialized_(false),
- number_cores_(CpuInfo::DetectNumberOfCores()),
- vie_performance_monitor_(ViEPerformanceMonitor(instance_id_)),
- channel_manager_(*new ViEChannelManager(instance_id_, number_cores_,
- vie_performance_monitor_)),
- input_manager_(*new ViEInputManager(instance_id_)),
- render_manager_(*new ViERenderManager(instance_id_)),
- module_process_thread_(ProcessThread::CreateProcessThread()),
- last_error_(0) {
- Trace::CreateTrace();
- channel_manager_.SetModuleProcessThread(*module_process_thread_);
- input_manager_.SetModuleProcessThread(*module_process_thread_);
- module_process_thread_->Start();
-}
-
-ViESharedData::~ViESharedData() {
- delete &input_manager_;
- delete &channel_manager_;
- delete &render_manager_;
-
- module_process_thread_->Stop();
- ProcessThread::DestroyProcessThread(module_process_thread_);
- Trace::ReturnTrace();
-}
-
-bool ViESharedData::Initialized() const {
- return initialized_;
-}
-
-int ViESharedData::SetInitialized() {
- initialized_ = true;
- return 0;
-}
-
-int ViESharedData::SetUnInitialized() {
- initialized_ = false;
- return 0;
-}
-
-void ViESharedData::SetLastError(const int error) const {
- last_error_ = error;
-}
-
-int ViESharedData::LastErrorInternal() const {
- int error = last_error_;
- last_error_ = 0;
- return error;
-}
-
-int ViESharedData::NumberOfCores() const {
- return number_cores_;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_shared_data.h b/trunk/src/video_engine/vie_shared_data.h
deleted file mode 100644
index 7f755fa..0000000
--- a/trunk/src/video_engine/vie_shared_data.h
+++ /dev/null
@@ -1,63 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// ViESharedData contains data and instances common to all interface
-// implementations.
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_SHARED_DATA_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_SHARED_DATA_H_
-
-#include "video_engine/vie_defines.h"
-#include "video_engine/vie_performance_monitor.h"
-
-namespace webrtc {
-
-class ProcessThread;
-class ViEChannelManager;
-class ViEInputManager;
-class ViERenderManager;
-
-class ViESharedData {
- public:
- ViESharedData();
- ~ViESharedData();
-
- bool Initialized() const;
- int SetInitialized();
- int SetUnInitialized();
- void SetLastError(const int error) const;
- int LastErrorInternal() const;
-
- int NumberOfCores() const;
-
- int instance_id() { return instance_id_;}
- ViEPerformanceMonitor* vie_performance_monitor() {
- return &vie_performance_monitor_; }
- ViEChannelManager* channel_manager() { return &channel_manager_; }
- ViEInputManager* input_manager() { return &input_manager_; }
- ViERenderManager* render_manager() { return &render_manager_; }
-
- private:
- static int instance_counter_;
- const int instance_id_;
- bool initialized_;
- const int number_cores_;
-
- ViEPerformanceMonitor vie_performance_monitor_;
- ViEChannelManager& channel_manager_;
- ViEInputManager& input_manager_;
- ViERenderManager& render_manager_;
- ProcessThread* module_process_thread_;
- mutable int last_error_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_SHARED_DATA_H_
diff --git a/trunk/src/video_engine/vie_sync_module.cc b/trunk/src/video_engine/vie_sync_module.cc
deleted file mode 100644
index d9b60ac..0000000
--- a/trunk/src/video_engine/vie_sync_module.cc
+++ /dev/null
@@ -1,297 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "vie_sync_module.h"
-
-#include "critical_section_wrapper.h"
-#include "rtp_rtcp.h"
-#include "trace.h"
-#include "video_coding.h"
-#include "voe_video_sync.h"
-
-namespace webrtc {
-
-enum { kSyncInterval = 1000};
-enum { kMaxVideoDiffMs = 80 };
-enum { kMaxAudioDiffMs = 80 };
-enum { kMaxDelay = 1500 };
-
-ViESyncModule::ViESyncModule(int id, VideoCodingModule& vcm,
- RtpRtcp& rtcp_module)
- : data_cs_(CriticalSectionWrapper::CreateCriticalSection()),
- id_(id),
- vcm_(vcm),
- rtcp_module_(rtcp_module),
- voe_channel_id_(-1),
- voe_sync_interface_(NULL),
- last_sync_time_(TickTime::Now()) {
-}
-
-ViESyncModule::~ViESyncModule() {
-}
-
-int ViESyncModule::SetVoiceChannel(int voe_channel_id,
- VoEVideoSync* voe_sync_interface) {
- CriticalSectionScoped cs(data_cs_.get());
- voe_channel_id_ = voe_channel_id;
- voe_sync_interface_ = voe_sync_interface;
- rtcp_module_.DeRegisterSyncModule();
-
- if (!voe_sync_interface) {
- voe_channel_id_ = -1;
- if (voe_channel_id >= 0) {
- // Trying to set a voice channel but no interface exist.
- return -1;
- }
- return 0;
- }
- RtpRtcp* voe_rtp_rtcp = NULL;
- voe_sync_interface->GetRtpRtcp(voe_channel_id_, voe_rtp_rtcp);
- return rtcp_module_.RegisterSyncModule(voe_rtp_rtcp);
-}
-
-int ViESyncModule::VoiceChannel() {
- return voe_channel_id_;
-}
-
-void ViESyncModule::SetNetworkDelay(int network_delay) {
- channel_delay_.network_delay = network_delay;
-}
-
-WebRtc_Word32 ViESyncModule::Version(WebRtc_Word8* version,
- WebRtc_UWord32& remaining_buffer_in_bytes,
- WebRtc_UWord32& position) const {
- if (version == NULL) {
- WEBRTC_TRACE(webrtc::kTraceWarning, webrtc::kTraceVideo, -1,
- "Invalid in argument to ViESyncModule Version()");
- return -1;
- }
- WebRtc_Word8 our_version[] = "ViESyncModule 1.1.0";
- WebRtc_UWord32 our_length = (WebRtc_UWord32) strlen(our_version);
- if (remaining_buffer_in_bytes < our_length + 1) {
- return -1;
- }
- memcpy(version, our_version, our_length);
- version[our_length] = '\0';
- remaining_buffer_in_bytes -= (our_length + 1);
- position += (our_length + 1);
- return 0;
-}
-
-WebRtc_Word32 ViESyncModule::ChangeUniqueId(const WebRtc_Word32 id) {
- id_ = id;
- return 0;
-}
-
-WebRtc_Word32 ViESyncModule::TimeUntilNextProcess() {
- return (WebRtc_Word32)(kSyncInterval -
- (TickTime::Now() - last_sync_time_).Milliseconds());
-}
-
-WebRtc_Word32 ViESyncModule::Process() {
- CriticalSectionScoped cs(data_cs_.get());
- last_sync_time_ = TickTime::Now();
-
- int total_video_delay_target_ms = vcm_.Delay();
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, id_,
- "Video delay (JB + decoder) is %d ms",
- total_video_delay_target_ms);
-
- if (voe_channel_id_ == -1) {
- return 0;
- }
-
- int current_audio_delay_ms = 0;
- if (voe_sync_interface_->GetDelayEstimate(voe_channel_id_,
- current_audio_delay_ms) != 0) {
- // Could not get VoE delay value, probably not a valid channel Id.
- WEBRTC_TRACE(webrtc::kTraceStream, webrtc::kTraceVideo, id_,
- "%s: VE_GetDelayEstimate error for voice_channel %d",
- __FUNCTION__, total_video_delay_target_ms, voe_channel_id_);
- return 0;
- }
-
- int current_diff_ms = 0;
- // Total video delay.
- int video_delay_ms = 0;
- // VoiceEngine report delay estimates even when not started, ignore if the
- // reported value is lower than 40 ms.
- if (current_audio_delay_ms < 40) {
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, id_,
- "A/V Sync: Audio delay < 40, skipping.");
- return 0;
- }
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, id_,
- "Audio delay is: %d for voice channel: %d",
- current_audio_delay_ms, voe_channel_id_);
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, id_,
- "Network delay diff is: %d for voice channel: %d",
- channel_delay_.network_delay, voe_channel_id_);
- // Calculate the difference between the lowest possible video delay and
- // the current audio delay.
- current_diff_ms = total_video_delay_target_ms - current_audio_delay_ms +
- channel_delay_.network_delay;
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, id_,
- "Current diff is: %d for audio channel: %d",
- current_diff_ms, voe_channel_id_);
-
- if (current_diff_ms > 0) {
- // The minimum video delay is longer than the current audio delay.
- // We need to decrease extra video delay, if we have added extra delay
- // earlier, or add extra audio delay.
- if (channel_delay_.extra_video_delay_ms > 0) {
- // We have extra delay added to ViE. Reduce this delay before adding
- // extra delay to VoE.
-
- // This is the desired delay, we can't reduce more than this.
- video_delay_ms = total_video_delay_target_ms;
-
- // Check that we don't reduce the delay more than what is allowed.
- if (video_delay_ms <
- channel_delay_.last_video_delay_ms - kMaxVideoDiffMs) {
- video_delay_ms =
- channel_delay_.last_video_delay_ms - kMaxVideoDiffMs;
- channel_delay_.extra_video_delay_ms =
- video_delay_ms - total_video_delay_target_ms;
- } else {
- channel_delay_.extra_video_delay_ms = 0;
- }
- channel_delay_.last_video_delay_ms = video_delay_ms;
- channel_delay_.last_sync_delay = -1;
- channel_delay_.extra_audio_delay_ms = 0;
- } else { // channel_delay_.extra_video_delay_ms > 0
- // We have no extra video delay to remove, increase the audio delay.
- if (channel_delay_.last_sync_delay >= 0) {
- // We have increased the audio delay earlier, increase it even more.
- int audio_diff_ms = current_diff_ms / 2;
- if (audio_diff_ms > kMaxAudioDiffMs) {
- // We only allow a maximum change of KMaxAudioDiffMS for audio
- // due to NetEQ maximum changes.
- audio_diff_ms = kMaxAudioDiffMs;
- }
- // Increase the audio delay
- channel_delay_.extra_audio_delay_ms += audio_diff_ms;
-
- // Don't set a too high delay.
- if (channel_delay_.extra_audio_delay_ms > kMaxDelay) {
- channel_delay_.extra_audio_delay_ms = kMaxDelay;
- }
-
- // Don't add any extra video delay.
- video_delay_ms = total_video_delay_target_ms;
- channel_delay_.extra_video_delay_ms = 0;
- channel_delay_.last_video_delay_ms = video_delay_ms;
- channel_delay_.last_sync_delay = 1;
- } else { // channel_delay_.last_sync_delay >= 0
- // First time after a delay change, don't add any extra delay.
- // This is to not toggle back and forth too much.
- channel_delay_.extra_audio_delay_ms = 0;
- // Set minimum video delay
- video_delay_ms = total_video_delay_target_ms;
- channel_delay_.extra_video_delay_ms = 0;
- channel_delay_.last_video_delay_ms = video_delay_ms;
- channel_delay_.last_sync_delay = 0;
- }
- }
- } else { // if (current_diffMS > 0)
- // The minimum video delay is lower than the current audio delay.
- // We need to decrease possible extra audio delay, or
- // add extra video delay.
-
- if (channel_delay_.extra_audio_delay_ms > 0) {
- // We have extra delay in VoiceEngine
- // Start with decreasing the voice delay
- int audio_diff_ms = current_diff_ms / 2;
- if (audio_diff_ms < -1 * kMaxAudioDiffMs) {
- // Don't change the delay too much at once.
- audio_diff_ms = -1 * kMaxAudioDiffMs;
- }
- // Add the negative difference.
- channel_delay_.extra_audio_delay_ms += audio_diff_ms;
-
- if (channel_delay_.extra_audio_delay_ms < 0) {
- // Negative values not allowed.
- channel_delay_.extra_audio_delay_ms = 0;
- channel_delay_.last_sync_delay = 0;
- } else {
- // There is more audio delay to use for the next round.
- channel_delay_.last_sync_delay = 1;
- }
-
- // Keep the video delay at the minimum values.
- video_delay_ms = total_video_delay_target_ms;
- channel_delay_.extra_video_delay_ms = 0;
- channel_delay_.last_video_delay_ms = video_delay_ms;
- } else { // channel_delay_.extra_audio_delay_ms > 0
- // We have no extra delay in VoiceEngine, increase the video delay.
- channel_delay_.extra_audio_delay_ms = 0;
-
- // Make the difference positive.
- int video_diff_ms = -1 * current_diff_ms;
-
- // This is the desired delay.
- video_delay_ms = total_video_delay_target_ms + video_diff_ms;
- if (video_delay_ms > channel_delay_.last_video_delay_ms) {
- if (video_delay_ms >
- channel_delay_.last_video_delay_ms + kMaxVideoDiffMs) {
- // Don't increase the delay too much at once
- video_delay_ms =
- channel_delay_.last_video_delay_ms + kMaxVideoDiffMs;
- }
- // Verify we don't go above the maximum allowed delay
- if (video_delay_ms > kMaxDelay) {
- video_delay_ms = kMaxDelay;
- }
- } else {
- if (video_delay_ms <
- channel_delay_.last_video_delay_ms - kMaxVideoDiffMs) {
- // Don't decrease the delay too much at once
- video_delay_ms =
- channel_delay_.last_video_delay_ms - kMaxVideoDiffMs;
- }
- // Verify we don't go below the minimum delay
- if (video_delay_ms < total_video_delay_target_ms) {
- video_delay_ms = total_video_delay_target_ms;
- }
- }
- // Store the values
- channel_delay_.extra_video_delay_ms =
- video_delay_ms - total_video_delay_target_ms;
- channel_delay_.last_video_delay_ms = video_delay_ms;
- channel_delay_.last_sync_delay = -1;
- }
- }
-
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, id_,
- "Sync video delay %d ms for video channel and audio delay %d for audio "
- "channel %d",
- video_delay_ms, channel_delay_.extra_audio_delay_ms, voe_channel_id_);
-
- // Set the extra audio delay.synchronization
- if (voe_sync_interface_->SetMinimumPlayoutDelay(
- voe_channel_id_, channel_delay_.extra_audio_delay_ms) == -1) {
- WEBRTC_TRACE(webrtc::kTraceDebug, webrtc::kTraceVideo, id_,
- "Error setting voice delay");
- }
-
- if (video_delay_ms < 0) {
- video_delay_ms = 0;
- }
- total_video_delay_target_ms =
- (total_video_delay_target_ms > video_delay_ms) ?
- total_video_delay_target_ms : video_delay_ms;
- vcm_.SetMinimumPlayoutDelay(total_video_delay_target_ms);
- WEBRTC_TRACE(webrtc::kTraceInfo, webrtc::kTraceVideo, id_,
- "New Video delay target is: %d", total_video_delay_target_ms);
- return 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/video_engine/vie_sync_module.h b/trunk/src/video_engine/vie_sync_module.h
deleted file mode 100644
index 01373a9..0000000
--- a/trunk/src/video_engine/vie_sync_module.h
+++ /dev/null
@@ -1,76 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// ViESyncModule is responsible for synchronization audio and video for a given
-// VoE and ViE channel couple.
-
-#ifndef WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
-#define WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
-
-#include "module.h"
-#include "system_wrappers/interface/scoped_ptr.h"
-#include "tick_util.h"
-
-namespace webrtc {
-
-class CriticalSectionWrapper;
-class RtpRtcp;
-class VideoCodingModule;
-class VoEVideoSync;
-
-class ViESyncModule : public Module {
- public:
- ViESyncModule(int id, VideoCodingModule& vcm, RtpRtcp& rtcp_module);
- ~ViESyncModule();
-
- int SetVoiceChannel(int voe_channel_id, VoEVideoSync* voe_sync_interface);
- int VoiceChannel();
-
- // Set how long time, in ms, voice is ahead of video when received on the
- // network. Positive value means audio is ahead of video.
- void SetNetworkDelay(int network_delay);
-
- // Implements Module.
- virtual WebRtc_Word32 Version(WebRtc_Word8* version,
- WebRtc_UWord32& remaining_buffer_in_bytes,
- WebRtc_UWord32& position) const;
- virtual WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
- virtual WebRtc_Word32 TimeUntilNextProcess();
- virtual WebRtc_Word32 Process();
-
- private:
- scoped_ptr<CriticalSectionWrapper> data_cs_;
- int id_;
- VideoCodingModule& vcm_;
- RtpRtcp& rtcp_module_;
- int voe_channel_id_;
- VoEVideoSync* voe_sync_interface_;
- TickTime last_sync_time_;
-
- struct ViESyncDelay {
- ViESyncDelay() {
- extra_video_delay_ms = 0;
- last_video_delay_ms = 0;
- extra_audio_delay_ms = 0;
- last_sync_delay = 0;
- network_delay = 120;
- }
- int extra_video_delay_ms;
- int last_video_delay_ms;
- int extra_audio_delay_ms;
- int last_sync_delay;
- int network_delay;
- };
- ViESyncDelay channel_delay_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VIDEO_ENGINE_VIE_SYNC_MODULE_H_
diff --git a/trunk/src/voice_engine/OWNERS b/trunk/src/voice_engine/OWNERS
deleted file mode 100644
index a07ced3..0000000
--- a/trunk/src/voice_engine/OWNERS
+++ /dev/null
@@ -1,4 +0,0 @@
-henrikg@webrtc.org
-henrika@webrtc.org
-niklas.enbom@webrtc.org
-xians@webrtc.org
diff --git a/trunk/src/voice_engine/main/interface/mock/mock_voe_connection_observer.h b/trunk/src/voice_engine/main/interface/mock/mock_voe_connection_observer.h
deleted file mode 100644
index 9eb9271..0000000
--- a/trunk/src/voice_engine/main/interface/mock/mock_voe_connection_observer.h
+++ /dev/null
@@ -1,26 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef MOCK_VOE_CONNECTION_OBSERVER_H_
-#define MOCK_VOE_CONNECTION_OBSERVER_H_
-
-#include "voe_network.h"
-
-namespace webrtc {
-
-class MockVoeConnectionObserver : public VoEConnectionObserver {
- public:
- MOCK_METHOD2(OnPeriodicDeadOrAlive, void(const int channel,
- const bool alive));
-};
-
-}
-
-#endif // MOCK_VOE_CONNECTION_OBSERVER_H_
diff --git a/trunk/src/voice_engine/main/interface/mock/mock_voe_observer.h b/trunk/src/voice_engine/main/interface/mock/mock_voe_observer.h
deleted file mode 100644
index 70d1c13..0000000
--- a/trunk/src/voice_engine/main/interface/mock/mock_voe_observer.h
+++ /dev/null
@@ -1,29 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_MOCK_VOE_OBSERVER_H_
-#define WEBRTC_VOICE_ENGINE_MOCK_VOE_OBSERVER_H_
-
-#include "voe_base.h"
-#include "gmock/gmock.h"
-
-namespace webrtc {
-
-class MockVoEObserver: public VoiceEngineObserver {
- public:
- MockVoEObserver() {}
- virtual ~MockVoEObserver() {}
-
- MOCK_METHOD2(CallbackOnError, void(const int channel, const int error_code));
-};
-
-}
-
-#endif // WEBRTC_VOICE_ENGINE_MOCK_VOE_OBSERVER_H_
diff --git a/trunk/src/voice_engine/main/interface/voe_audio_processing.h b/trunk/src/voice_engine/main/interface/voe_audio_processing.h
deleted file mode 100644
index 541807b..0000000
--- a/trunk/src/voice_engine/main/interface/voe_audio_processing.h
+++ /dev/null
@@ -1,189 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-// - Noise Suppression (NS).
-// - Automatic Gain Control (AGC).
-// - Echo Control (EC).
-// - Receiving side VAD, NS and AGC.
-// - Measurements of instantaneous speech, noise and echo levels.
-// - Generation of AP debug recordings.
-// - Detection of keyboard typing which can disrupt a voice conversation.
-//
-// Usage example, omitting error checking:
-//
-// using namespace webrtc;
-// VoiceEngine* voe = VoiceEngine::Create();
-// VoEBase* base = VoEBase::GetInterface();
-// VoEAudioProcessing* ap = VoEAudioProcessing::GetInterface(voe);
-// base->Init();
-// ap->SetEcStatus(true, kAgcAdaptiveAnalog);
-// ...
-// base->Terminate();
-// base->Release();
-// ap->Release();
-// VoiceEngine::Delete(voe);
-//
-#ifndef WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_H
-#define WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_H
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VoiceEngine;
-
-// VoERxVadCallback
-class WEBRTC_DLLEXPORT VoERxVadCallback
-{
-public:
- virtual void OnRxVad(int channel, int vadDecision) = 0;
-
-protected:
- virtual ~VoERxVadCallback() {}
-};
-
-// VoEAudioProcessing
-class WEBRTC_DLLEXPORT VoEAudioProcessing
-{
-public:
- // Factory for the VoEAudioProcessing sub-API. Increases an internal
- // reference counter if successful. Returns NULL if the API is not
- // supported or if construction fails.
- static VoEAudioProcessing* GetInterface(VoiceEngine* voiceEngine);
-
- // Releases the VoEAudioProcessing sub-API and decreases an internal
- // reference counter. Returns the new reference count. This value should
- // be zero for all sub-API:s before the VoiceEngine object can be safely
- // deleted.
- virtual int Release() = 0;
-
- // Sets Noise Suppression (NS) status and mode.
- // The NS reduces noise in the microphone signal.
- virtual int SetNsStatus(bool enable, NsModes mode = kNsUnchanged) = 0;
-
- // Gets the NS status and mode.
- virtual int GetNsStatus(bool& enabled, NsModes& mode) = 0;
-
- // Sets the Automatic Gain Control (AGC) status and mode.
- // The AGC adjusts the microphone signal to an appropriate level.
- virtual int SetAgcStatus(bool enable, AgcModes mode = kAgcUnchanged) = 0;
-
- // Gets the AGC status and mode.
- virtual int GetAgcStatus(bool& enabled, AgcModes& mode) = 0;
-
- // Sets the AGC configuration.
- // Should only be used in situations where the working environment
- // is well known.
- virtual int SetAgcConfig(const AgcConfig config) = 0;
-
- // Gets the AGC configuration.
- virtual int GetAgcConfig(AgcConfig& config) = 0;
-
- // Sets the Echo Control (EC) status and mode.
- // The EC mitigates acoustic echo where a user can hear their own
- // speech repeated back due to an acoustic coupling between the
- // speaker and the microphone at the remote end.
- virtual int SetEcStatus(bool enable, EcModes mode = kEcUnchanged) = 0;
-
- // Gets the EC status and mode.
- virtual int GetEcStatus(bool& enabled, EcModes& mode) = 0;
-
- // Modifies settings for the AEC designed for mobile devices (AECM).
- virtual int SetAecmMode(AecmModes mode = kAecmSpeakerphone,
- bool enableCNG = true) = 0;
-
- // Gets settings for the AECM.
- virtual int GetAecmMode(AecmModes& mode, bool& enabledCNG) = 0;
-
- // Sets status and mode of the receiving-side (Rx) NS.
- // The Rx NS reduces noise in the received signal for the specified
- // |channel|. Intended for advanced usage only.
- virtual int SetRxNsStatus(int channel,
- bool enable,
- NsModes mode = kNsUnchanged) = 0;
-
- // Gets status and mode of the receiving-side NS.
- virtual int GetRxNsStatus(int channel,
- bool& enabled,
- NsModes& mode) = 0;
-
- // Sets status and mode of the receiving-side (Rx) AGC.
- // The Rx AGC adjusts the received signal to an appropriate level
- // for the specified |channel|. Intended for advanced usage only.
- virtual int SetRxAgcStatus(int channel,
- bool enable,
- AgcModes mode = kAgcUnchanged) = 0;
-
- // Gets status and mode of the receiving-side AGC.
- virtual int GetRxAgcStatus(int channel,
- bool& enabled,
- AgcModes& mode) = 0;
-
- // Modifies the AGC configuration on the receiving side for the
- // specified |channel|.
- virtual int SetRxAgcConfig(int channel, const AgcConfig config) = 0;
-
- // Gets the AGC configuration on the receiving side.
- virtual int GetRxAgcConfig(int channel, AgcConfig& config) = 0;
-
- // Registers a VoERxVadCallback |observer| instance and enables Rx VAD
- // notifications for the specified |channel|.
- virtual int RegisterRxVadObserver(int channel,
- VoERxVadCallback &observer) = 0;
-
- // Deregisters the VoERxVadCallback |observer| and disables Rx VAD
- // notifications for the specified |channel|.
- virtual int DeRegisterRxVadObserver(int channel) = 0;
-
- // Gets the VAD/DTX activity for the specified |channel|.
- // The returned value is 1 if frames of audio contains speech
- // and 0 if silence. The output is always 1 if VAD is disabled.
- virtual int VoiceActivityIndicator(int channel) = 0;
-
- // Enables or disables the possibility to retrieve echo metrics and delay
- // logging values during an active call. The metrics are only supported in
- // AEC.
- virtual int SetEcMetricsStatus(bool enable) = 0;
-
- // Gets the current EC metric status.
- virtual int GetEcMetricsStatus(bool& enabled) = 0;
-
- // Gets the instantaneous echo level metrics.
- virtual int GetEchoMetrics(int& ERL, int& ERLE, int& RERL, int& A_NLP) = 0;
-
- // Gets the EC internal |delay_median| and |delay_std| in ms between
- // near-end and far-end. The values are calculated over the time period
- // since the last GetEcDelayMetrics() call.
- virtual int GetEcDelayMetrics(int& delay_median, int& delay_std) = 0;
-
- // Enables recording of Audio Processing (AP) debugging information.
- // The file can later be used for off-line analysis of the AP performance.
- virtual int StartDebugRecording(const char* fileNameUTF8) = 0;
-
- // Disables recording of AP debugging information.
- virtual int StopDebugRecording() = 0;
-
- // Enables or disables detection of disturbing keyboard typing.
- // An error notification will be given as a callback upon detection.
- virtual int SetTypingDetectionStatus(bool enable) = 0;
-
- // Gets the current typing detection status.
- virtual int GetTypingDetectionStatus(bool& enabled) = 0;
-
-protected:
- VoEAudioProcessing() {}
- virtual ~VoEAudioProcessing() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_H
diff --git a/trunk/src/voice_engine/main/interface/voe_base.h b/trunk/src/voice_engine/main/interface/voe_base.h
deleted file mode 100644
index be6849c..0000000
--- a/trunk/src/voice_engine/main/interface/voe_base.h
+++ /dev/null
@@ -1,213 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-// - Enables full duplex VoIP sessions via RTP using G.711 (mu-Law or A-Law).
-// - Initialization and termination.
-// - Trace information on text files or via callbacks.
-// - Multi-channel support (mixing, sending to multiple destinations etc.).
-// - Call setup (port and address) for receiving and sending sides.
-//
-// To support other codecs than G.711, the VoECodec sub-API must be utilized.
-//
-// Usage example, omitting error checking:
-//
-// using namespace webrtc;
-// VoiceEngine* voe = VoiceEngine::Create();
-// VoEBase* base = VoEBase::GetInterface(voe);
-// base->Init();
-// int ch = base->CreateChannel();
-// base->StartPlayout(ch);
-// ...
-// base->DeleteChannel(ch);
-// base->Terminate();
-// base->Release();
-// VoiceEngine::Delete(voe);
-//
-#ifndef WEBRTC_VOICE_ENGINE_VOE_BASE_H
-#define WEBRTC_VOICE_ENGINE_VOE_BASE_H
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class AudioDeviceModule;
-
-const int kVoEDefault = -1;
-
-// VoiceEngineObserver
-class WEBRTC_DLLEXPORT VoiceEngineObserver
-{
-public:
- // This method will be called after the occurrence of any runtime error
- // code, or warning notification, when the observer interface has been
- // installed using VoEBase::RegisterVoiceEngineObserver().
- virtual void CallbackOnError(const int channel, const int errCode) = 0;
-
-protected:
- virtual ~VoiceEngineObserver() {}
-};
-
-// VoiceEngine
-class WEBRTC_DLLEXPORT VoiceEngine
-{
-public:
- // Creates a VoiceEngine object, which can then be used to acquire
- // sub-APIs. Returns NULL on failure.
- static VoiceEngine* Create();
-
- // Deletes a created VoiceEngine object and releases the utilized resources.
- // If |ignoreRefCounters| is set to false, all reference counters must be
- // zero to enable a valid release of the allocated resources. When set to
- // true, a release of all resources allocated by the VoE is performed
- // without checking the reference counter state.
- static bool Delete(VoiceEngine*& voiceEngine,
- bool ignoreRefCounters = false);
-
- // Specifies the amount and type of trace information which will be
- // created by the VoiceEngine.
- static int SetTraceFilter(const unsigned int filter);
-
- // Sets the name of the trace file and enables non-encrypted trace messages.
- static int SetTraceFile(const char* fileNameUTF8,
- const bool addFileCounter = false);
-
- // Installs the TraceCallback implementation to ensure that the user
- // receives callbacks for generated trace messages.
- static int SetTraceCallback(TraceCallback* callback);
-
- static int SetAndroidObjects(void* javaVM, void* env, void* context);
-
-protected:
- VoiceEngine() {}
- virtual ~VoiceEngine() {}
-};
-
-// VoEBase
-class WEBRTC_DLLEXPORT VoEBase
-{
-public:
- // Factory for the VoEBase sub-API. Increases an internal reference
- // counter if successful. Returns NULL if the API is not supported or if
- // construction fails.
- static VoEBase* GetInterface(VoiceEngine* voiceEngine);
-
- // Releases the VoEBase sub-API and decreases an internal reference
- // counter. Returns the new reference count. This value should be zero
- // for all sub-API:s before the VoiceEngine object can be safely deleted.
- virtual int Release() = 0;
-
- // Installs the observer class to enable runtime error control and
- // warning notifications.
- virtual int RegisterVoiceEngineObserver(VoiceEngineObserver& observer) = 0;
-
- // Removes and disables the observer class for runtime error control
- // and warning notifications.
- virtual int DeRegisterVoiceEngineObserver() = 0;
-
- // Initiates all common parts of the VoiceEngine; e.g. all
- // encoders/decoders, the sound card and core receiving components.
- // This method also makes it possible to install a user-defined
- // external Audio Device Module (ADM) which implements all the audio
- // layer functionality in a separate (reference counted) module.
- virtual int Init(AudioDeviceModule* external_adm = NULL) = 0;
-
- // Terminates all VoiceEngine functions and releses allocated resources.
- virtual int Terminate() = 0;
-
- // Retrieves the maximum number of channels that can be created.
- virtual int MaxNumOfChannels() = 0;
-
- // Creates a new channel and allocates the required resources for it.
- virtual int CreateChannel() = 0;
-
- // Deletes an existing channel and releases the utilized resources.
- virtual int DeleteChannel(int channel) = 0;
-
- // Sets the local receiver port and address for a specified
- // |channel| number.
- virtual int SetLocalReceiver(int channel, int port,
- int RTCPport = kVoEDefault,
- const char ipAddr[64] = NULL,
- const char multiCastAddr[64] = NULL) = 0;
-
- // Gets the local receiver port and address for a specified
- // |channel| number.
- virtual int GetLocalReceiver(int channel, int& port, int& RTCPport,
- char ipAddr[64]) = 0;
-
- // Sets the destination port and address for a specified |channel| number.
- virtual int SetSendDestination(int channel, int port,
- const char ipAddr[64],
- int sourcePort = kVoEDefault,
- int RTCPport = kVoEDefault) = 0;
-
- // Gets the destination port and address for a specified |channel| number.
- virtual int GetSendDestination(int channel, int& port, char ipAddr[64],
- int& sourcePort, int& RTCPport) = 0;
-
- // Prepares and initiates the VoiceEngine for reception of
- // incoming RTP/RTCP packets on the specified |channel|.
- virtual int StartReceive(int channel) = 0;
-
- // Stops receiving incoming RTP/RTCP packets on the specified |channel|.
- virtual int StopReceive(int channel) = 0;
-
- // Starts forwarding the packets to the mixer/soundcard for a
- // specified |channel|.
- virtual int StartPlayout(int channel) = 0;
-
- // Stops forwarding the packets to the mixer/soundcard for a
- // specified |channel|.
- virtual int StopPlayout(int channel) = 0;
-
- // Starts sending packets to an already specified IP address and
- // port number for a specified |channel|.
- virtual int StartSend(int channel) = 0;
-
- // Stops sending packets from a specified |channel|.
- virtual int StopSend(int channel) = 0;
-
- // Gets the version information for VoiceEngine and its components.
- virtual int GetVersion(char version[1024]) = 0;
-
- // Gets the last VoiceEngine error code.
- virtual int LastError() = 0;
-
-
- // Stops or resumes playout and transmission on a temporary basis.
- virtual int SetOnHoldStatus(int channel, bool enable,
- OnHoldModes mode = kHoldSendAndPlay) = 0;
-
- // Gets the current playout and transmission status.
- virtual int GetOnHoldStatus(int channel, bool& enabled,
- OnHoldModes& mode) = 0;
-
- // Sets the NetEQ playout mode for a specified |channel| number.
- virtual int SetNetEQPlayoutMode(int channel, NetEqModes mode) = 0;
-
- // Gets the NetEQ playout mode for a specified |channel| number.
- virtual int GetNetEQPlayoutMode(int channel, NetEqModes& mode) = 0;
-
- // Sets the NetEQ background noise mode for a specified |channel| number.
- virtual int SetNetEQBGNMode(int channel, NetEqBgnModes mode) = 0;
-
- // Gets the NetEQ background noise mode for a specified |channel| number.
- virtual int GetNetEQBGNMode(int channel, NetEqBgnModes& mode) = 0;
-
-protected:
- VoEBase() {}
- virtual ~VoEBase() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_BASE_H
diff --git a/trunk/src/voice_engine/main/interface/voe_call_report.h b/trunk/src/voice_engine/main/interface/voe_call_report.h
deleted file mode 100644
index c4d3abd..0000000
--- a/trunk/src/voice_engine/main/interface/voe_call_report.h
+++ /dev/null
@@ -1,87 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-// - Long-term speech and noise level metrics.
-// - Long-term echo metric statistics.
-// - Round Trip Time (RTT) statistics.
-// - Dead-or-Alive connection summary.
-// - Generation of call reports to text files.
-//
-// Usage example, omitting error checking:
-//
-// using namespace webrtc;
-// VoiceEngine* voe = VoiceEngine::Create();
-// VoEBase* base = VoEBase::GetInterface(voe);
-// VoECallReport report = VoECallReport::GetInterface(voe);
-// base->Init();
-// LevelStatistics stats;
-// report->GetSpeechAndNoiseSummary(stats);
-// ...
-// base->Terminate();
-// base->Release();
-// report->Release();
-// VoiceEngine::Delete(voe);
-//
-#ifndef WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_H
-#define WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_H
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VoiceEngine;
-
-// VoECallReport
-class WEBRTC_DLLEXPORT VoECallReport
-{
-public:
- // Factory for the VoECallReport sub-API. Increases an internal
- // reference counter if successful. Returns NULL if the API is not
- // supported or if construction fails.
- static VoECallReport* GetInterface(VoiceEngine* voiceEngine);
-
- // Releases the VoECallReport sub-API and decreases an internal
- // reference counter. Returns the new reference count. This value should
- // be zero for all sub-API:s before the VoiceEngine object can be safely
- // deleted.
- virtual int Release() = 0;
-
- // Performs a combined reset of all components involved in generating
- // the call report for a specified |channel|. Pass in -1 to reset
- // all channels.
- virtual int ResetCallReportStatistics(int channel) = 0;
-
- // Gets minimum, maximum and average levels for long-term echo metrics.
- virtual int GetEchoMetricSummary(EchoStatistics& stats) = 0;
-
- // Gets minimum, maximum and average levels for Round Trip Time (RTT)
- // measurements.
- virtual int GetRoundTripTimeSummary(int channel,
- StatVal& delaysMs) = 0;
-
- // Gets the total amount of dead and alive connection detections
- // during a VoIP session.
- virtual int GetDeadOrAliveSummary(int channel, int& numOfDeadDetections,
- int& numOfAliveDetections) = 0;
-
- // Creates a text file in ASCII format, which contains a summary
- // of all the statistics that can be obtained by the call report sub-API.
- virtual int WriteReportToFile(const char* fileNameUTF8) = 0;
-
-protected:
- VoECallReport() { }
- virtual ~VoECallReport() { }
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_H
diff --git a/trunk/src/voice_engine/main/interface/voe_codec.h b/trunk/src/voice_engine/main/interface/voe_codec.h
deleted file mode 100644
index 413bc90..0000000
--- a/trunk/src/voice_engine/main/interface/voe_codec.h
+++ /dev/null
@@ -1,134 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-// - Support of non-default codecs (e.g. iLBC, iSAC, etc.).
-// - Voice Activity Detection (VAD) on a per channel basis.
-// - Possibility to specify how to map received payload types to codecs.
-//
-// Usage example, omitting error checking:
-//
-// using namespace webrtc;
-// VoiceEngine* voe = VoiceEngine::Create();
-// VoEBase* base = VoEBase::GetInterface(voe);
-// VoECodec* codec = VoECodec::GetInterface(voe);
-// base->Init();
-// int num_of_codecs = codec->NumOfCodecs()
-// ...
-// base->Terminate();
-// base->Release();
-// codec->Release();
-// VoiceEngine::Delete(voe);
-//
-#ifndef WEBRTC_VOICE_ENGINE_VOE_CODEC_H
-#define WEBRTC_VOICE_ENGINE_VOE_CODEC_H
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VoiceEngine;
-
-class WEBRTC_DLLEXPORT VoECodec
-{
-public:
- // Factory for the VoECodec sub-API. Increases an internal
- // reference counter if successful. Returns NULL if the API is not
- // supported or if construction fails.
- static VoECodec* GetInterface(VoiceEngine* voiceEngine);
-
- // Releases the VoECodec sub-API and decreases an internal
- // reference counter. Returns the new reference count. This value should
- // be zero for all sub-API:s before the VoiceEngine object can be safely
- // deleted.
- virtual int Release() = 0;
-
- // Gets the number of supported codecs.
- virtual int NumOfCodecs() = 0;
-
- // Get the |codec| information for a specified list |index|.
- virtual int GetCodec(int index, CodecInst& codec) = 0;
-
- // Sets the |codec| for the |channel| to be used for sending.
- virtual int SetSendCodec(int channel, const CodecInst& codec) = 0;
-
- // Gets the |codec| parameters for the sending codec on a specified
- // |channel|.
- virtual int GetSendCodec(int channel, CodecInst& codec) = 0;
-
- // Gets the currently received |codec| for a specific |channel|.
- virtual int GetRecCodec(int channel, CodecInst& codec) = 0;
-
- // Sets the initial values of target rate and frame size for iSAC
- // for a specified |channel|. This API is only valid if iSAC is setup
- // to run in channel-adaptive mode
- virtual int SetISACInitTargetRate(int channel, int rateBps,
- bool useFixedFrameSize = false) = 0;
-
- // Sets the maximum allowed iSAC rate which the codec may not exceed
- // for a single packet for the specified |channel|. The maximum rate is
- // defined as payload size per frame size in bits per second.
- virtual int SetISACMaxRate(int channel, int rateBps) = 0;
-
- // Sets the maximum allowed iSAC payload size for a specified |channel|.
- // The maximum value is set independently of the frame size, i.e.
- // 30 ms and 60 ms packets have the same limit.
- virtual int SetISACMaxPayloadSize(int channel, int sizeBytes) = 0;
-
- // Sets the dynamic payload type number for a particular |codec| or
- // disables (ignores) a codec for receiving. For instance, when receiving
- // an invite from a SIP-based client, this function can be used to change
- // the dynamic payload type number to match that in the INVITE SDP-
- // message. The utilized parameters in the |codec| structure are:
- // plname, plfreq, pltype and channels.
- virtual int SetRecPayloadType(int channel, const CodecInst& codec) = 0;
-
- // Gets the actual payload type that is set for receiving a |codec| on a
- // |channel|. The value it retrieves will either be the default payload
- // type, or a value earlier set with SetRecPayloadType().
- virtual int GetRecPayloadType(int channel, CodecInst& codec) = 0;
-
- // Sets the payload |type| for the sending of SID-frames with background
- // noise estimation during silence periods detected by the VAD.
- virtual int SetSendCNPayloadType(
- int channel, int type, PayloadFrequencies frequency = kFreq16000Hz) = 0;
-
-
- // Sets the VAD/DTX (silence suppression) status and |mode| for a
- // specified |channel|.
- virtual int SetVADStatus(int channel, bool enable,
- VadModes mode = kVadConventional,
- bool disableDTX = false) = 0;
-
- // Gets the VAD/DTX status and |mode| for a specified |channel|.
- virtual int GetVADStatus(int channel, bool& enabled, VadModes& mode,
- bool& disabledDTX) = 0;
-
- // Not supported
- virtual int SetAMREncFormat(int channel, AmrMode mode) = 0;
-
- // Not supported
- virtual int SetAMRDecFormat(int channel, AmrMode mode) = 0;
-
- // Not supported
- virtual int SetAMRWbEncFormat(int channel, AmrMode mode) = 0;
-
- // Not supported
- virtual int SetAMRWbDecFormat(int channel, AmrMode mode) = 0;
-
-protected:
- VoECodec() {}
- virtual ~VoECodec() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_CODEC_H
diff --git a/trunk/src/voice_engine/main/interface/voe_dtmf.h b/trunk/src/voice_engine/main/interface/voe_dtmf.h
deleted file mode 100644
index 3ed1749..0000000
--- a/trunk/src/voice_engine/main/interface/voe_dtmf.h
+++ /dev/null
@@ -1,148 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-// - Telephone event transmission.
-// - DTMF tone generation.
-//
-// Usage example, omitting error checking:
-//
-// using namespace webrtc;
-// VoiceEngine* voe = VoiceEngine::Create();
-// VoEBase* base = VoEBase::GetInterface(voe);
-// VoEDtmf* dtmf = VoEDtmf::GetInterface(voe);
-// base->Init();
-// int ch = base->CreateChannel();
-// ...
-// dtmf->SendTelephoneEvent(ch, 7);
-// ...
-// base->DeleteChannel(ch);
-// base->Terminate();
-// base->Release();
-// dtmf->Release();
-// VoiceEngine::Delete(voe);
-//
-#ifndef WEBRTC_VOICE_ENGINE_VOE_DTMF_H
-#define WEBRTC_VOICE_ENGINE_VOE_DTMF_H
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VoiceEngine;
-
-// VoETelephoneEventObserver
-class WEBRTC_DLLEXPORT VoETelephoneEventObserver
-{
-public:
- // This method will be called after the detection of an inband
- // telephone event. The event code is given as output in the
- // |eventCode| parameter.
- virtual void OnReceivedTelephoneEventInband(int channel,
- int eventCode,
- bool endOfEvent) = 0;
-
- // This method will be called after the detection of an out-of-band
- // telephone event. The event code is given as output in the
- // |eventCode| parameter.
- virtual void OnReceivedTelephoneEventOutOfBand(
- int channel,
- int eventCode,
- bool endOfEvent) = 0;
-
-protected:
- virtual ~VoETelephoneEventObserver() {}
-};
-
-// VoEDtmf
-class WEBRTC_DLLEXPORT VoEDtmf
-{
-public:
-
- // Factory for the VoEDtmf sub-API. Increases an internal
- // reference counter if successful. Returns NULL if the API is not
- // supported or if construction fails.
- static VoEDtmf* GetInterface(VoiceEngine* voiceEngine);
-
- // Releases the VoEDtmf sub-API and decreases an internal
- // reference counter. Returns the new reference count. This value should
- // be zero for all sub-API:s before the VoiceEngine object can be safely
- // deleted.
- virtual int Release() = 0;
-
- // Sends telephone events either in-band or out-of-band.
- virtual int SendTelephoneEvent(int channel, int eventCode,
- bool outOfBand = true, int lengthMs = 160,
- int attenuationDb = 10) = 0;
-
-
- // Sets the dynamic payload |type| that should be used for telephone
- // events.
- virtual int SetSendTelephoneEventPayloadType(int channel,
- unsigned char type) = 0;
-
-
- // Gets the currently set dynamic payload |type| for telephone events.
- virtual int GetSendTelephoneEventPayloadType(int channel,
- unsigned char& type) = 0;
-
- // Enables or disables local tone playout for received DTMF events
- // out-of-band.
- virtual int SetDtmfPlayoutStatus(int channel, bool enable) = 0;
-
- // Gets the DTMF playout status.
- virtual int GetDtmfPlayoutStatus(int channel, bool& enabled) = 0;
-
- // Toogles DTMF feedback state: when a DTMF tone is sent, the same tone
- // is played out on the speaker.
- virtual int SetDtmfFeedbackStatus(bool enable,
- bool directFeedback = false) = 0;
-
- // Gets the DTMF feedback status.
- virtual int GetDtmfFeedbackStatus(bool& enabled, bool& directFeedback) = 0;
-
- // Plays a DTMF feedback tone (only locally).
- virtual int PlayDtmfTone(int eventCode, int lengthMs = 200,
- int attenuationDb = 10) = 0;
-
- // Starts playing out a DTMF feedback tone locally.
- // The tone will be played out until the corresponding stop function
- // is called.
- virtual int StartPlayingDtmfTone(int eventCode,
- int attenuationDb = 10) = 0;
-
- // Stops playing out a DTMF feedback tone locally.
- virtual int StopPlayingDtmfTone() = 0;
-
- // Installs an instance of a VoETelephoneEventObserver derived class and
- // activates detection of telephone events for the specified |channel|.
- virtual int RegisterTelephoneEventDetection(
- int channel, TelephoneEventDetectionMethods detectionMethod,
- VoETelephoneEventObserver& observer) = 0;
-
- // Removes an instance of a VoETelephoneEventObserver derived class and
- // disables detection of telephone events for the specified |channel|.
- virtual int DeRegisterTelephoneEventDetection(int channel) = 0;
-
- // Gets the current telephone-event detection status for a specified
- // |channel|.
- virtual int GetTelephoneEventDetectionStatus(
- int channel, bool& enabled,
- TelephoneEventDetectionMethods& detectionMethod) = 0;
-
-protected:
- VoEDtmf() {}
- virtual ~VoEDtmf() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_DTMF_H
diff --git a/trunk/src/voice_engine/main/interface/voe_encryption.h b/trunk/src/voice_engine/main/interface/voe_encryption.h
deleted file mode 100644
index ae3f373..0000000
--- a/trunk/src/voice_engine/main/interface/voe_encryption.h
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-// - External encryption and decryption.
-//
-// Usage example, omitting error checking:
-//
-// using namespace webrtc;
-// VoiceEngine* voe = VoiceEngine::Create();
-// VoEEncryption* encrypt = VoEEncryption::GetInterface(voe);
-// ...
-// encrypt->Release();
-// VoiceEngine::Delete(voe);
-//
-#ifndef WEBRTC_VOICE_ENGINE_VOE_ENCRYPTION_H
-#define WEBRTC_VOICE_ENGINE_VOE_ENCRYPTION_H
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VoiceEngine;
-
-class WEBRTC_DLLEXPORT VoEEncryption
-{
-public:
- // Factory for the VoEEncryption sub-API. Increases an internal
- // reference counter if successful. Returns NULL if the API is not
- // supported or if construction fails.
- static VoEEncryption* GetInterface(VoiceEngine* voiceEngine);
-
- // Releases the VoEEncryption sub-API and decreases an internal
- // reference counter. Returns the new reference count. This value should
- // be zero for all sub-API:s before the VoiceEngine object can be safely
- // deleted.
- virtual int Release() = 0;
-
- // Installs an Encryption instance and enables external encryption
- // for the selected |channel|.
- virtual int RegisterExternalEncryption(
- int channel, Encryption& encryption) = 0;
-
- // Removes an Encryption instance and disables external encryption
- // for the selected |channel|.
- virtual int DeRegisterExternalEncryption(int channel) = 0;
-
- // Not supported
- virtual int EnableSRTPSend(int channel, CipherTypes cipherType,
- int cipherKeyLength, AuthenticationTypes authType, int authKeyLength,
- int authTagLength, SecurityLevels level, const unsigned char key[30],
- bool useForRTCP = false) = 0;
-
- // Not supported
- virtual int DisableSRTPSend(int channel) = 0;
-
- // Not supported
- virtual int EnableSRTPReceive(int channel, CipherTypes cipherType,
- int cipherKeyLength, AuthenticationTypes authType, int authKeyLength,
- int authTagLength, SecurityLevels level, const unsigned char key[30],
- bool useForRTCP = false) = 0;
-
- // Not supported
- virtual int DisableSRTPReceive(int channel) = 0;
-
-protected:
- VoEEncryption() {}
- virtual ~VoEEncryption() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_ENCRYPTION_H
diff --git a/trunk/src/voice_engine/main/interface/voe_errors.h b/trunk/src/voice_engine/main/interface/voe_errors.h
deleted file mode 100644
index cc05970..0000000
--- a/trunk/src/voice_engine/main/interface/voe_errors.h
+++ /dev/null
@@ -1,162 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_ERRORS_H
-#define WEBRTC_VOICE_ENGINE_VOE_ERRORS_H
-
-// Warnings
-#define VE_PORT_NOT_DEFINED 8001
-#define VE_CHANNEL_NOT_VALID 8002
-#define VE_FUNC_NOT_SUPPORTED 8003
-#define VE_INVALID_LISTNR 8004
-#define VE_INVALID_ARGUMENT 8005
-#define VE_INVALID_PORT_NMBR 8006
-#define VE_INVALID_PLNAME 8007
-#define VE_INVALID_PLFREQ 8008
-#define VE_INVALID_PLTYPE 8009
-#define VE_INVALID_PACSIZE 8010
-#define VE_NOT_SUPPORTED 8011
-#define VE_ALREADY_LISTENING 8012
-#define VE_CHANNEL_NOT_CREATED 8013
-#define VE_MAX_ACTIVE_CHANNELS_REACHED 8014
-#define VE_REC_CANNOT_PREPARE_HEADER 8015
-#define VE_REC_CANNOT_ADD_BUFFER 8016
-#define VE_PLAY_CANNOT_PREPARE_HEADER 8017
-#define VE_ALREADY_SENDING 8018
-#define VE_INVALID_IP_ADDRESS 8019
-#define VE_ALREADY_PLAYING 8020
-#define VE_NOT_ALL_VERSION_INFO 8021
-#define VE_DTMF_OUTOF_RANGE 8022
-#define VE_INVALID_CHANNELS 8023
-#define VE_SET_PLTYPE_FAILED 8024
-#define VE_ENCRYPT_NOT_INITED 8025
-#define VE_NOT_INITED 8026
-#define VE_NOT_SENDING 8027
-#define VE_EXT_TRANSPORT_NOT_SUPPORTED 8028
-#define VE_EXTERNAL_TRANSPORT_ENABLED 8029
-#define VE_STOP_RECORDING_FAILED 8030
-#define VE_INVALID_RATE 8031
-#define VE_INVALID_PACKET 8032
-#define VE_NO_GQOS 8033
-#define VE_INVALID_TIMESTAMP 8034
-#define VE_RECEIVE_PACKET_TIMEOUT 8035
-#define VE_STILL_PLAYING_PREV_DTMF 8036
-#define VE_INIT_FAILED_WRONG_EXPIRY 8037
-#define VE_SENDING 8038
-#define VE_ENABLE_IPV6_FAILED 8039
-#define VE_FUNC_NO_STEREO 8040
-// Range 8041-8080 is not used
-#define VE_FW_TRAVERSAL_ALREADY_INITIALIZED 8081
-#define VE_PACKET_RECEIPT_RESTARTED 8082
-#define VE_NOT_ALL_INFO 8083
-#define VE_CANNOT_SET_SEND_CODEC 8084
-#define VE_CODEC_ERROR 8085
-#define VE_NETEQ_ERROR 8086
-#define VE_RTCP_ERROR 8087
-#define VE_INVALID_OPERATION 8088
-#define VE_CPU_INFO_ERROR 8089
-#define VE_SOUNDCARD_ERROR 8090
-#define VE_SPEECH_LEVEL_ERROR 8091
-#define VE_SEND_ERROR 8092
-#define VE_CANNOT_REMOVE_CONF_CHANNEL 8093
-#define VE_PLTYPE_ERROR 8094
-#define VE_SET_FEC_FAILED 8095
-#define VE_CANNOT_GET_PLAY_DATA 8096
-#define VE_APM_ERROR 8097
-#define VE_RUNTIME_PLAY_WARNING 8098
-#define VE_RUNTIME_REC_WARNING 8099
-#define VE_NOT_PLAYING 8100
-#define VE_SOCKETS_NOT_INITED 8101
-#define VE_CANNOT_GET_SOCKET_INFO 8102
-#define VE_INVALID_MULTICAST_ADDRESS 8103
-#define VE_DESTINATION_NOT_INITED 8104
-#define VE_RECEIVE_SOCKETS_CONFLICT 8105
-#define VE_SEND_SOCKETS_CONFLICT 8106
-#define VE_TYPING_NOISE_WARNING 8107
-#define VE_SATURATION_WARNING 8108
-#define VE_NOISE_WARNING 8109
-#define VE_CANNOT_GET_SEND_CODEC 8110
-#define VE_CANNOT_GET_REC_CODEC 8111
-#define VE_ALREADY_INITED 8112
-
-// Errors causing limited functionality
-#define VE_RTCP_SOCKET_ERROR 9001
-#define VE_MIC_VOL_ERROR 9002
-#define VE_SPEAKER_VOL_ERROR 9003
-#define VE_CANNOT_ACCESS_MIC_VOL 9004
-#define VE_CANNOT_ACCESS_SPEAKER_VOL 9005
-#define VE_GET_MIC_VOL_ERROR 9006
-#define VE_GET_SPEAKER_VOL_ERROR 9007
-#define VE_THREAD_RTCP_ERROR 9008
-#define VE_CANNOT_INIT_APM 9009
-#define VE_SEND_SOCKET_TOS_ERROR 9010
-#define VE_CANNOT_RETRIEVE_DEVICE_NAME 9013
-#define VE_SRTP_ERROR 9014
-// 9015 is not used
-#define VE_INTERFACE_NOT_FOUND 9016
-#define VE_TOS_GQOS_CONFLICT 9017
-#define VE_CANNOT_ADD_CONF_CHANNEL 9018
-#define VE_BUFFER_TOO_SMALL 9019
-#define VE_CANNOT_EXECUTE_SETTING 9020
-#define VE_CANNOT_RETRIEVE_SETTING 9021
-// 9022 is not used
-#define VE_RTP_KEEPALIVE_FAILED 9023
-#define VE_SEND_DTMF_FAILED 9024
-#define VE_CANNOT_RETRIEVE_CNAME 9025
-#define VE_DECRYPTION_FAILED 9026
-#define VE_ENCRYPTION_FAILED 9027
-#define VE_CANNOT_RETRIEVE_RTP_STAT 9028
-#define VE_GQOS_ERROR 9029
-#define VE_BINDING_SOCKET_TO_LOCAL_ADDRESS_FAILED 9030
-#define VE_TOS_INVALID 9031
-#define VE_TOS_ERROR 9032
-#define VE_CANNOT_RETRIEVE_VALUE 9033
-
-// Critical errors that stops voice functionality
-#define VE_PLAY_UNDEFINED_SC_ERR 10001
-#define VE_REC_CANNOT_OPEN_SC 10002
-#define VE_SOCKET_ERROR 10003
-#define VE_MMSYSERR_INVALHANDLE 10004
-#define VE_MMSYSERR_NODRIVER 10005
-#define VE_MMSYSERR_NOMEM 10006
-#define VE_WAVERR_UNPREPARED 10007
-#define VE_WAVERR_STILLPLAYING 10008
-#define VE_UNDEFINED_SC_ERR 10009
-#define VE_UNDEFINED_SC_REC_ERR 10010
-#define VE_THREAD_ERROR 10011
-#define VE_CANNOT_START_RECORDING 10012
-#define VE_PLAY_CANNOT_OPEN_SC 10013
-#define VE_NO_WINSOCK_2 10014
-#define VE_SEND_SOCKET_ERROR 10015
-#define VE_BAD_FILE 10016
-#define VE_EXPIRED_COPY 10017
-#define VE_NOT_AUTHORISED 10018
-#define VE_RUNTIME_PLAY_ERROR 10019
-#define VE_RUNTIME_REC_ERROR 10020
-#define VE_BAD_ARGUMENT 10021
-#define VE_LINUX_API_ONLY 10022
-#define VE_REC_DEVICE_REMOVED 10023
-#define VE_NO_MEMORY 10024
-#define VE_BAD_HANDLE 10025
-#define VE_RTP_RTCP_MODULE_ERROR 10026
-#define VE_AUDIO_CODING_MODULE_ERROR 10027
-#define VE_AUDIO_DEVICE_MODULE_ERROR 10028
-#define VE_CANNOT_START_PLAYOUT 10029
-#define VE_CANNOT_STOP_RECORDING 10030
-#define VE_CANNOT_STOP_PLAYOUT 10031
-#define VE_CANNOT_INIT_CHANNEL 10032
-#define VE_RECV_SOCKET_ERROR 10033
-#define VE_SOCKET_TRANSPORT_MODULE_ERROR 10034
-#define VE_AUDIO_CONF_MIX_MODULE_ERROR 10035
-
-// Warnings for other platforms (reserved range 8061-8080)
-#define VE_IGNORED_FUNCTION 8061
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_ERRORS_H
diff --git a/trunk/src/voice_engine/main/interface/voe_external_media.h b/trunk/src/voice_engine/main/interface/voe_external_media.h
deleted file mode 100644
index 50d2d38..0000000
--- a/trunk/src/voice_engine/main/interface/voe_external_media.h
+++ /dev/null
@@ -1,114 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// In some cases it is desirable to use an audio source or sink which may
-// not be available to the VoiceEngine, such as a DV camera. This sub-API
-// contains functions that allow for the use of such external recording
-// sources and playout sinks. It also describes how recorded data, or data
-// to be played out, can be modified outside the VoiceEngine.
-//
-// Usage example, omitting error checking:
-//
-// using namespace webrtc;
-// VoiceEngine* voe = VoiceEngine::Create();
-// VoEBase* base = VoEBase::GetInterface(voe);
-// VoEMediaProcess media = VoEMediaProcess::GetInterface(voe);
-// base->Init();
-// ...
-// media->SetExternalRecordingStatus(true);
-// ...
-// base->Terminate();
-// base->Release();
-// media->Release();
-// VoiceEngine::Delete(voe);
-//
-#ifndef WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H
-#define WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VoiceEngine;
-
-class WEBRTC_DLLEXPORT VoEMediaProcess
-{
-public:
- // The VoiceEngine user should override the Process() method in a
- // derived class. Process() will be called when audio is ready to
- // be processed. The audio can be accessed in several different modes
- // given by the |type| parameter. The function should modify the
- // original data and ensure that it is copied back to the |audio10ms|
- // array. The number of samples in the frame cannot be changed.
- // The sampling frequency will depend upon the codec used.
- // If |isStereo| is true, audio10ms will contain 16-bit PCM data
- // samples in interleaved stereo format (L0,R0,L1,R1,
):
- virtual void Process(const int channel, const ProcessingTypes type,
- WebRtc_Word16 audio10ms[], const int length,
- const int samplingFreq, const bool isStereo) = 0;
-
-protected:
- virtual ~VoEMediaProcess() {}
-};
-
-class WEBRTC_DLLEXPORT VoEExternalMedia
-{
-public:
- // Factory for the VoEExternalMedia sub-API. Increases an internal
- // reference counter if successful. Returns NULL if the API is not
- // supported or if construction fails.
- static VoEExternalMedia* GetInterface(VoiceEngine* voiceEngine);
-
- // Releases the VoEExternalMedia sub-API and decreases an internal
- // reference counter. Returns the new reference count. This value should
- // be zero for all sub-API:s before the VoiceEngine object can be safely
- // deleted.
- virtual int Release() = 0;
-
- // Installs a VoEMediaProcess derived instance and activates external
- // media for the specified |channel| and |type|.
- virtual int RegisterExternalMediaProcessing(
- int channel, ProcessingTypes type, VoEMediaProcess& processObject) = 0;
-
- // Removes the VoEMediaProcess derived instance and deactivates external
- // media for the specified |channel| and |type|.
- virtual int DeRegisterExternalMediaProcessing(
- int channel, ProcessingTypes type) = 0;
-
- // Toogles state of external recording.
- virtual int SetExternalRecordingStatus(bool enable) = 0;
-
- // Toogles state of external playout.
- virtual int SetExternalPlayoutStatus(bool enable) = 0;
-
- // This function accepts externally recorded audio. During transmission,
- // this method should be called at as regular an interval as possible
- // with frames of corresponding size.
- virtual int ExternalRecordingInsertData(
- const WebRtc_Word16 speechData10ms[], int lengthSamples,
- int samplingFreqHz, int current_delay_ms) = 0;
-
- // This function gets audio for an external playout sink.
- // During transmission, this function should be called every ~10 ms
- // to obtain a new 10 ms frame of audio. The length of the block will
- // be 160, 320, 440 or 480 samples (for 16, 32, 44 or 48 kHz sampling
- // rates respectively).
- virtual int ExternalPlayoutGetData(
- WebRtc_Word16 speechData10ms[], int samplingFreqHz,
- int current_delay_ms, int& lengthSamples) = 0;
-
-protected:
- VoEExternalMedia() {}
- virtual ~VoEExternalMedia() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_H
diff --git a/trunk/src/voice_engine/main/interface/voe_file.h b/trunk/src/voice_engine/main/interface/voe_file.h
deleted file mode 100644
index d968dcf..0000000
--- a/trunk/src/voice_engine/main/interface/voe_file.h
+++ /dev/null
@@ -1,184 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-// - File playback.
-// - File recording.
-// - File conversion.
-//
-// Usage example, omitting error checking:
-//
-// using namespace webrtc;
-// VoiceEngine* voe = VoiceEngine::Create();
-// VoEBase* base = VoEBase::GetInterface(voe);
-// VoEFile* file = VoEFile::GetInterface(voe);
-// base->Init();
-// int ch = base->CreateChannel();
-// ...
-// base->StartPlayout(ch);
-// file->StartPlayingFileAsMicrophone(ch, "data_file_16kHz.pcm", true);
-// ...
-// file->StopPlayingFileAsMicrophone(ch);
-// base->StopPlayout(ch);
-// ...
-// base->DeleteChannel(ch);
-// base->Terminate();
-// base->Release();
-// file->Release();
-// VoiceEngine::Delete(voe);
-//
-#ifndef WEBRTC_VOICE_ENGINE_VOE_FILE_H
-#define WEBRTC_VOICE_ENGINE_VOE_FILE_H
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VoiceEngine;
-
-class WEBRTC_DLLEXPORT VoEFile
-{
-public:
- // Factory for the VoEFile sub-API. Increases an internal
- // reference counter if successful. Returns NULL if the API is not
- // supported or if construction fails.
- static VoEFile* GetInterface(VoiceEngine* voiceEngine);
-
- // Releases the VoEFile sub-API and decreases an internal
- // reference counter. Returns the new reference count. This value should
- // be zero for all sub-API:s before the VoiceEngine object can be safely
- // deleted.
- virtual int Release() = 0;
-
- // Starts playing and mixing files with the local speaker signal for
- // playout.
- virtual int StartPlayingFileLocally(
- int channel,
- const char fileNameUTF8[1024],
- bool loop = false,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0,
- int startPointMs = 0,
- int stopPointMs = 0) = 0;
-
- // Starts playing and mixing streams with the local speaker signal for
- // playout.
- virtual int StartPlayingFileLocally(
- int channel,
- InStream* stream,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0,
- int startPointMs = 0, int stopPointMs = 0) = 0;
-
- // Stops playback of a file on a specific |channel|.
- virtual int StopPlayingFileLocally(int channel) = 0;
-
- // Returns the current file playing state for a specific |channel|.
- virtual int IsPlayingFileLocally(int channel) = 0;
-
- // Sets the volume scaling for a speaker file that is already playing.
- virtual int ScaleLocalFilePlayout(int channel, float scale) = 0;
-
- // Starts reading data from a file and transmits the data either
- // mixed with or instead of the microphone signal.
- virtual int StartPlayingFileAsMicrophone(
- int channel,
- const char fileNameUTF8[1024],
- bool loop = false ,
- bool mixWithMicrophone = false,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0) = 0;
-
- // Starts reading data from a stream and transmits the data either
- // mixed with or instead of the microphone signal.
- virtual int StartPlayingFileAsMicrophone(
- int channel,
- InStream* stream,
- bool mixWithMicrophone = false,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0) = 0;
-
- // Stops playing of a file as microphone signal for a specific |channel|.
- virtual int StopPlayingFileAsMicrophone(int channel) = 0;
-
- // Returns whether the |channel| is currently playing a file as microphone.
- virtual int IsPlayingFileAsMicrophone(int channel) = 0;
-
- // Sets the volume scaling for a microphone file that is already playing.
- virtual int ScaleFileAsMicrophonePlayout(int channel, float scale) = 0;
-
- // Starts recording the mixed playout audio.
- virtual int StartRecordingPlayout(int channel,
- const char* fileNameUTF8,
- CodecInst* compression = NULL,
- int maxSizeBytes = -1) = 0;
-
- // Stops recording the mixed playout audio.
- virtual int StopRecordingPlayout(int channel) = 0;
-
- virtual int StartRecordingPlayout(int channel,
- OutStream* stream,
- CodecInst* compression = NULL) = 0;
-
- // Starts recording the microphone signal to a file.
- virtual int StartRecordingMicrophone(const char* fileNameUTF8,
- CodecInst* compression = NULL,
- int maxSizeBytes = -1) = 0;
-
- // Starts recording the microphone signal to a stream.
- virtual int StartRecordingMicrophone(OutStream* stream,
- CodecInst* compression = NULL) = 0;
-
- // Stops recording the microphone signal.
- virtual int StopRecordingMicrophone() = 0;
-
-
- // Gets the duration of a file.
- virtual int GetFileDuration(const char* fileNameUTF8, int& durationMs,
- FileFormats format = kFileFormatPcm16kHzFile) = 0;
-
- // Gets the current played position of a file on a specific |channel|.
- virtual int GetPlaybackPosition(int channel, int& positionMs) = 0;
-
- virtual int ConvertPCMToWAV(const char* fileNameInUTF8,
- const char* fileNameOutUTF8) = 0;
-
- virtual int ConvertPCMToWAV(InStream* streamIn,
- OutStream* streamOut) = 0;
-
- virtual int ConvertWAVToPCM(const char* fileNameInUTF8,
- const char* fileNameOutUTF8) = 0;
-
- virtual int ConvertWAVToPCM(InStream* streamIn,
- OutStream* streamOut) = 0;
-
- virtual int ConvertPCMToCompressed(const char* fileNameInUTF8,
- const char* fileNameOutUTF8,
- CodecInst* compression) = 0;
-
- virtual int ConvertPCMToCompressed(InStream* streamIn,
- OutStream* streamOut,
- CodecInst* compression) = 0;
-
- virtual int ConvertCompressedToPCM(const char* fileNameInUTF8,
- const char* fileNameOutUTF8) = 0;
-
- virtual int ConvertCompressedToPCM(InStream* streamIn,
- OutStream* streamOut) = 0;
-
-protected:
- VoEFile() {}
- virtual ~VoEFile() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_FILE_H
diff --git a/trunk/src/voice_engine/main/interface/voe_hardware.h b/trunk/src/voice_engine/main/interface/voe_hardware.h
deleted file mode 100644
index 24ed1ff..0000000
--- a/trunk/src/voice_engine/main/interface/voe_hardware.h
+++ /dev/null
@@ -1,142 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-// - Audio device handling.
-// - Device information.
-// - CPU load monitoring.
-//
-// Usage example, omitting error checking:
-//
-// using namespace webrtc;
-// VoiceEngine* voe = VoiceEngine::Create();
-// VoEBase* base = VoEBase::GetInterface(voe);
-// VoEHardware* hardware = VoEHardware::GetInterface(voe);
-// base->Init();
-// ...
-// int n_devices = hardware->GetNumOfPlayoutDevices();
-// ...
-// base->Terminate();
-// base->Release();
-// hardware->Release();
-// VoiceEngine::Delete(voe);
-//
-#ifndef WEBRTC_VOICE_ENGINE_VOE_HARDWARE_H
-#define WEBRTC_VOICE_ENGINE_VOE_HARDWARE_H
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VoiceEngine;
-
-class WEBRTC_DLLEXPORT VoEHardware
-{
-public:
- // Factory for the VoEHardware sub-API. Increases an internal
- // reference counter if successful. Returns NULL if the API is not
- // supported or if construction fails.
- static VoEHardware* GetInterface(VoiceEngine* voiceEngine);
-
- // Releases the VoEHardware sub-API and decreases an internal
- // reference counter. Returns the new reference count. This value should
- // be zero for all sub-API:s before the VoiceEngine object can be safely
- // deleted.
- virtual int Release() = 0;
-
- // Gets the number of audio devices available for recording.
- virtual int GetNumOfRecordingDevices(int& devices) = 0;
-
- // Gets the number of audio devices available for playout.
- virtual int GetNumOfPlayoutDevices(int& devices) = 0;
-
- // Gets the name of a specific recording device given by an |index|.
- // On Windows Vista/7, it also retrieves an additional unique ID
- // (GUID) for the recording device.
- virtual int GetRecordingDeviceName(int index, char strNameUTF8[128],
- char strGuidUTF8[128]) = 0;
-
- // Gets the name of a specific playout device given by an |index|.
- // On Windows Vista/7, it also retrieves an additional unique ID
- // (GUID) for the playout device.
- virtual int GetPlayoutDeviceName(int index, char strNameUTF8[128],
- char strGuidUTF8[128]) = 0;
-
- // Checks if the sound card is available to be opened for recording.
- virtual int GetRecordingDeviceStatus(bool& isAvailable) = 0;
-
- // Checks if the sound card is available to be opened for playout.
- virtual int GetPlayoutDeviceStatus(bool& isAvailable) = 0;
-
- // Sets the audio device used for recording.
- virtual int SetRecordingDevice(
- int index, StereoChannel recordingChannel = kStereoBoth) = 0;
-
- // Sets the audio device used for playout.
- virtual int SetPlayoutDevice(int index) = 0;
-
- // Sets the type of audio device layer to use.
- virtual int SetAudioDeviceLayer(AudioLayers audioLayer) = 0;
-
- // Gets the currently used (active) audio device layer.
- virtual int GetAudioDeviceLayer(AudioLayers& audioLayer) = 0;
-
- // Gets the VoiceEngine's current CPU consumption in terms of the percent
- // of total CPU availability. [Windows only]
- virtual int GetCPULoad(int& loadPercent) = 0;
-
- // Gets the computer's current CPU consumption in terms of the percent
- // of the total CPU availability. This method may fail a few times on
- // Windows because it needs a certain warm-up time before reporting the
- // result. You should check the return value and either try again or
- // give up when it fails.
- virtual int GetSystemCPULoad(int& loadPercent) = 0;
-
- // Not supported
- virtual int ResetAudioDevice() = 0;
-
- // Not supported
- virtual int AudioDeviceControl(
- unsigned int par1, unsigned int par2, unsigned int par3) = 0;
-
- // Not supported
- virtual int SetLoudspeakerStatus(bool enable) = 0;
-
- // Not supported
- virtual int GetLoudspeakerStatus(bool& enabled) = 0;
-
- // *Experimental - not recommended for use.*
- // Enables the Windows Core Audio built-in AEC. Fails on other platforms.
- //
- // Currently incompatible with the standard VoE AEC and AGC; don't attempt
- // to enable them while this is active.
- //
- // Must be called before VoEBase::StartSend(). When enabled:
- // 1. VoEBase::StartPlayout() must be called before VoEBase::StartSend().
- // 2. VoEBase::StopSend() should be called before VoEBase::StopPlayout().
- // The reverse order may cause garbage audio to be rendered or the
- // capture side to halt until StopSend() is called.
- //
- // As a consequence, SetPlayoutDevice() should be used with caution
- // during a call. It will function, but may cause the above issues for
- // the duration it takes to complete. (In practice, it should complete
- // fast enough to avoid audible degradation).
- virtual int EnableBuiltInAEC(bool enable) = 0;
- virtual bool BuiltInAECIsEnabled() const = 0;
-
-protected:
- VoEHardware() {}
- virtual ~VoEHardware() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_HARDWARE_H
diff --git a/trunk/src/voice_engine/main/interface/voe_neteq_stats.h b/trunk/src/voice_engine/main/interface/voe_neteq_stats.h
deleted file mode 100644
index 4940bed..0000000
--- a/trunk/src/voice_engine/main/interface/voe_neteq_stats.h
+++ /dev/null
@@ -1,45 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_H
-#define WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_H
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VoiceEngine;
-
-class WEBRTC_DLLEXPORT VoENetEqStats
-{
-public:
- // Factory for the VoENetEqStats sub-API. Increases an internal
- // reference counter if successful. Returns NULL if the API is not
- // supported or if construction fails.
- static VoENetEqStats* GetInterface(VoiceEngine* voiceEngine);
-
- // Releases the VoENetEqStats sub-API and decreases an internal
- // reference counter. Returns the new reference count. This value should
- // be zero for all sub-API:s before the VoiceEngine object can be safely
- // deleted.
- virtual int Release() = 0;
-
- // Get the "in-call" statistics from NetEQ.
- // The statistics are reset after the query.
- virtual int GetNetworkStatistics(int channel, NetworkStatistics& stats) = 0;
-
-protected:
- VoENetEqStats() {}
- virtual ~VoENetEqStats() {}
-};
-
-} // namespace webrtc
-
-#endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_H
diff --git a/trunk/src/voice_engine/main/interface/voe_network.h b/trunk/src/voice_engine/main/interface/voe_network.h
deleted file mode 100644
index 10acf1c..0000000
--- a/trunk/src/voice_engine/main/interface/voe_network.h
+++ /dev/null
@@ -1,177 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-// - External protocol support.
-// - Extended port and address APIs.
-// - Port and address filters.
-// - Windows GQoS functions.
-// - Packet timeout notification.
-// - Dead-or-Alive connection observations.
-// - Transmission of raw RTP/RTCP packets into existing channels.
-//
-// Usage example, omitting error checking:
-//
-// using namespace webrtc;
-// VoiceEngine* voe = VoiceEngine::Create();
-// VoEBase* base = VoEBase::GetInterface(voe);
-// VoENetwork* netw = VoENetwork::GetInterface(voe);
-// base->Init();
-// int ch = base->CreateChannel();
-// ...
-// netw->SetPeriodicDeadOrAliveStatus(ch, true);
-// ...
-// base->DeleteChannel(ch);
-// base->Terminate();
-// base->Release();
-// netw->Release();
-// VoiceEngine::Delete(voe);
-//
-#ifndef WEBRTC_VOICE_ENGINE_VOE_NETWORK_H
-#define WEBRTC_VOICE_ENGINE_VOE_NETWORK_H
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VoiceEngine;
-
-// VoEConnectionObserver
-class WEBRTC_DLLEXPORT VoEConnectionObserver
-{
-public:
- // This method will be called peridically and deliver dead-or-alive
- // notifications for a specified |channel| when the observer interface
- // has been installed and activated.
- virtual void OnPeriodicDeadOrAlive(const int channel, const bool alive) = 0;
-
-protected:
- virtual ~VoEConnectionObserver() {}
-};
-
-// VoENetwork
-class WEBRTC_DLLEXPORT VoENetwork
-{
-public:
- // Factory for the VoENetwork sub-API. Increases an internal
- // reference counter if successful. Returns NULL if the API is not
- // supported or if construction fails.
- static VoENetwork* GetInterface(VoiceEngine* voiceEngine);
-
- // Releases the VoENetwork sub-API and decreases an internal
- // reference counter. Returns the new reference count. This value should
- // be zero for all sub-API:s before the VoiceEngine object can be safely
- // deleted.
- virtual int Release() = 0;
-
- // Installs and enables a user-defined external transport protocol for a
- // specified |channel|.
- virtual int RegisterExternalTransport(
- int channel, Transport& transport) = 0;
-
- // Removes and disables a user-defined external transport protocol for a
- // specified |channel|.
- virtual int DeRegisterExternalTransport(int channel) = 0;
-
- // The packets received from the network should be passed to this
- // function when external transport is enabled. Note that the data
- // including the RTP-header must also be given to the VoiceEngine.
- virtual int ReceivedRTPPacket(
- int channel, const void* data, unsigned int length) = 0;
-
- // The packets received from the network should be passed to this
- // function when external transport is enabled. Note that the data
- // including the RTCP-header must also be given to the VoiceEngine.
- virtual int ReceivedRTCPPacket(
- int channel, const void* data, unsigned int length) = 0;
-
- // Gets the source ports and IP address of incoming packets on a
- // specific |channel|.
- virtual int GetSourceInfo(
- int channel, int& rtpPort, int& rtcpPort, char ipAddr[64]) = 0;
-
- // Gets the local (host) IP address.
- virtual int GetLocalIP(char ipAddr[64], bool ipv6 = false) = 0;
-
- // Enables IPv6 for a specified |channel|.
- virtual int EnableIPv6(int channel) = 0;
-
- // Gets the current IPv6 staus for a specified |channel|.
- virtual bool IPv6IsEnabled(int channel) = 0;
-
- // Enables a port and IP address filter for incoming packets on a
- // specific |channel|.
- virtual int SetSourceFilter(int channel,
- int rtpPort, int rtcpPort = 0, const char ipAddr[64] = 0) = 0;
-
- // Gets the current port and IP-address filter for a specified |channel|.
- virtual int GetSourceFilter(
- int channel, int& rtpPort, int& rtcpPort, char ipAddr[64]) = 0;
-
- // Sets the six-bit Differentiated Services Code Point (DSCP) in the
- // IP header of the outgoing stream for a specific |channel|.
- virtual int SetSendTOS(int channel,
- int DSCP, int priority = -1, bool useSetSockopt = false) = 0;
-
- // Gets the six-bit DSCP in the IP header of the outgoing stream for
- // a specific channel.
- virtual int GetSendTOS(
- int channel, int& DSCP, int& priority, bool& useSetSockopt) = 0;
-
- // Sets the Generic Quality of Service (GQoS) service level.
- // The Windows operating system then maps to a Differentiated Services
- // Code Point (DSCP) and to an 802.1p setting. [Windows only]
- virtual int SetSendGQoS(
- int channel, bool enable, int serviceType, int overrideDSCP = 0) = 0;
-
- // Gets the Generic Quality of Service (GQoS) service level.
- virtual int GetSendGQoS(
- int channel, bool& enabled, int& serviceType, int& overrideDSCP) = 0;
-
- // Enables or disables warnings that report if packets have not been
- // received in |timeoutSeconds| seconds for a specific |channel|.
- virtual int SetPacketTimeoutNotification(
- int channel, bool enable, int timeoutSeconds = 2) = 0;
-
- // Gets the current time-out notification status.
- virtual int GetPacketTimeoutNotification(
- int channel, bool& enabled, int& timeoutSeconds) = 0;
-
- // Installs the observer class implementation for a specified |channel|.
- virtual int RegisterDeadOrAliveObserver(
- int channel, VoEConnectionObserver& observer) = 0;
-
- // Removes the observer class implementation for a specified |channel|.
- virtual int DeRegisterDeadOrAliveObserver(int channel) = 0;
-
- // Enables or disables the periodic dead-or-alive callback functionality
- // for a specified |channel|.
- virtual int SetPeriodicDeadOrAliveStatus(
- int channel, bool enable, int sampleTimeSeconds = 2) = 0;
-
- // Gets the current dead-or-alive notification status.
- virtual int GetPeriodicDeadOrAliveStatus(
- int channel, bool& enabled, int& sampleTimeSeconds) = 0;
-
- // Handles sending a raw UDP data packet over an existing RTP or RTCP
- // socket.
- virtual int SendUDPPacket(
- int channel, const void* data, unsigned int length,
- int& transmittedBytes, bool useRtcpSocket = false) = 0;
-
-protected:
- VoENetwork() {}
- virtual ~VoENetwork() {}
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_NETWORK_H
diff --git a/trunk/src/voice_engine/main/interface/voe_rtp_rtcp.h b/trunk/src/voice_engine/main/interface/voe_rtp_rtcp.h
deleted file mode 100644
index e26d85f..0000000
--- a/trunk/src/voice_engine/main/interface/voe_rtp_rtcp.h
+++ /dev/null
@@ -1,234 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-// - Callbacks for RTP and RTCP events such as modified SSRC or CSRC.
-// - SSRC handling.
-// - Transmission of RTCP sender reports.
-// - Obtaining RTCP data from incoming RTCP sender reports.
-// - RTP and RTCP statistics (jitter, packet loss, RTT etc.).
-// - Forward Error Correction (FEC).
-// - RTP Keepalive for maintaining the NAT mappings associated to RTP flows.
-// - Writing RTP and RTCP packets to binary files for off-line analysis of
-// the call quality.
-// - Inserting extra RTP packets into active audio stream.
-//
-// Usage example, omitting error checking:
-//
-// using namespace webrtc;
-// VoiceEngine* voe = VoiceEngine::Create();
-// VoEBase* base = VoEBase::GetInterface(voe);
-// VoERTP_RTCP* rtp_rtcp = VoERTP_RTCP::GetInterface(voe);
-// base->Init();
-// int ch = base->CreateChannel();
-// ...
-// rtp_rtcp->SetLocalSSRC(ch, 12345);
-// ...
-// base->DeleteChannel(ch);
-// base->Terminate();
-// base->Release();
-// rtp_rtcp->Release();
-// VoiceEngine::Delete(voe);
-//
-#ifndef WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_H
-#define WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_H
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VoiceEngine;
-
-// VoERTPObserver
-class WEBRTC_DLLEXPORT VoERTPObserver
-{
-public:
- virtual void OnIncomingCSRCChanged(
- const int channel, const unsigned int CSRC, const bool added) = 0;
-
- virtual void OnIncomingSSRCChanged(
- const int channel, const unsigned int SSRC) = 0;
-
-protected:
- virtual ~VoERTPObserver() {}
-};
-
-// VoERTCPObserver
-class WEBRTC_DLLEXPORT VoERTCPObserver
-{
-public:
- virtual void OnApplicationDataReceived(
- const int channel, const unsigned char subType,
- const unsigned int name, const unsigned char* data,
- const unsigned short dataLengthInBytes) = 0;
-
-protected:
- virtual ~VoERTCPObserver() {}
-};
-
-// CallStatistics
-struct CallStatistics
-{
- unsigned short fractionLost;
- unsigned int cumulativeLost;
- unsigned int extendedMax;
- unsigned int jitterSamples;
- int rttMs;
- int bytesSent;
- int packetsSent;
- int bytesReceived;
- int packetsReceived;
-};
-
-// VoERTP_RTCP
-class WEBRTC_DLLEXPORT VoERTP_RTCP
-{
-public:
-
- // Factory for the VoERTP_RTCP sub-API. Increases an internal
- // reference counter if successful. Returns NULL if the API is not
- // supported or if construction fails.
- static VoERTP_RTCP* GetInterface(VoiceEngine* voiceEngine);
-
- // Releases the VoERTP_RTCP sub-API and decreases an internal
- // reference counter. Returns the new reference count. This value should
- // be zero for all sub-API:s before the VoiceEngine object can be safely
- // deleted.
- virtual int Release() = 0;
-
- // Registers an instance of a VoERTPObserver derived class for a specified
- // |channel|. It will allow the user to observe callbacks related to the
- // RTP protocol such as changes in the incoming SSRC.
- virtual int RegisterRTPObserver(int channel, VoERTPObserver& observer) = 0;
-
- // Deregisters an instance of a VoERTPObserver derived class for a
- // specified |channel|.
- virtual int DeRegisterRTPObserver(int channel) = 0;
-
- // Registers an instance of a VoERTCPObserver derived class for a specified
- // |channel|.
- virtual int RegisterRTCPObserver(
- int channel, VoERTCPObserver& observer) = 0;
-
- // Deregisters an instance of a VoERTCPObserver derived class for a
- // specified |channel|.
- virtual int DeRegisterRTCPObserver(int channel) = 0;
-
- // Sets the local RTP synchronization source identifier (SSRC) explicitly.
- virtual int SetLocalSSRC(int channel, unsigned int ssrc) = 0;
-
- // Gets the local RTP SSRC of a specified |channel|.
- virtual int GetLocalSSRC(int channel, unsigned int& ssrc) = 0;
-
- // Gets the SSRC of the incoming RTP packets.
- virtual int GetRemoteSSRC(int channel, unsigned int& ssrc) = 0;
-
- // Sets the status of rtp-audio-level-indication on a specific |channel|.
- virtual int SetRTPAudioLevelIndicationStatus(
- int channel, bool enable, unsigned char ID = 1) = 0;
-
- // Sets the status of rtp-audio-level-indication on a specific |channel|.
- virtual int GetRTPAudioLevelIndicationStatus(
- int channel, bool& enabled, unsigned char& ID) = 0;
-
- // Gets the CSRCs of the incoming RTP packets.
- virtual int GetRemoteCSRCs(int channel, unsigned int arrCSRC[15]) = 0;
-
- // Sets the RTCP status on a specific |channel|.
- virtual int SetRTCPStatus(int channel, bool enable) = 0;
-
- // Gets the RTCP status on a specific |channel|.
- virtual int GetRTCPStatus(int channel, bool& enabled) = 0;
-
- // Sets the canonical name (CNAME) parameter for RTCP reports on a
- // specific |channel|.
- virtual int SetRTCP_CNAME(int channel, const char cName[256]) = 0;
-
- // Gets the canonical name (CNAME) parameter for RTCP reports on a
- // specific |channel|.
- virtual int GetRTCP_CNAME(int channel, char cName[256]) = 0;
-
- // Gets the canonical name (CNAME) parameter for incoming RTCP reports
- // on a specific channel.
- virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]) = 0;
-
- // Gets RTCP data from incoming RTCP Sender Reports.
- virtual int GetRemoteRTCPData(
- int channel, unsigned int& NTPHigh, unsigned int& NTPLow,
- unsigned int& timestamp, unsigned int& playoutTimestamp,
- unsigned int* jitter = NULL, unsigned short* fractionLost = NULL) = 0;
-
- // Gets RTP statistics for a specific |channel|.
- virtual int GetRTPStatistics(
- int channel, unsigned int& averageJitterMs, unsigned int& maxJitterMs,
- unsigned int& discardedPackets) = 0;
-
- // Gets RTCP statistics for a specific |channel|.
- virtual int GetRTCPStatistics(int channel, CallStatistics& stats) = 0;
-
- // Sends an RTCP APP packet on a specific |channel|.
- virtual int SendApplicationDefinedRTCPPacket(
- int channel, const unsigned char subType, unsigned int name,
- const char* data, unsigned short dataLengthInBytes) = 0;
-
- // Sets the Forward Error Correction (FEC) status on a specific |channel|.
- virtual int SetFECStatus(
- int channel, bool enable, int redPayloadtype = -1) = 0;
-
- // Gets the FEC status on a specific |channel|.
- virtual int GetFECStatus(
- int channel, bool& enabled, int& redPayloadtype) = 0;
-
- // Sets the RTP keepalive mechanism status.
- // This functionality can maintain an existing Network Address Translator
- // (NAT) mapping while regular RTP is no longer transmitted.
- virtual int SetRTPKeepaliveStatus(
- int channel, bool enable, unsigned char unknownPayloadType,
- int deltaTransmitTimeSeconds = 15) = 0;
-
- // Gets the RTP keepalive mechanism status.
- virtual int GetRTPKeepaliveStatus(
- int channel, bool& enabled, unsigned char& unknownPayloadType,
- int& deltaTransmitTimeSeconds) = 0;
-
- // Enables capturing of RTP packets to a binary file on a specific
- // |channel| and for a given |direction|. The file can later be replayed
- // using e.g. RTP Tools rtpplay since the binary file format is
- // compatible with the rtpdump format.
- virtual int StartRTPDump(
- int channel, const char fileNameUTF8[1024],
- RTPDirections direction = kRtpIncoming) = 0;
-
- // Disables capturing of RTP packets to a binary file on a specific
- // |channel| and for a given |direction|.
- virtual int StopRTPDump(
- int channel, RTPDirections direction = kRtpIncoming) = 0;
-
- // Gets the the current RTP capturing state for the specified
- // |channel| and |direction|.
- virtual int RTPDumpIsActive(
- int channel, RTPDirections direction = kRtpIncoming) = 0;
-
- // Sends an extra RTP packet using an existing/active RTP session.
- // It is possible to set the payload type, marker bit and payload
- // of the extra RTP
- virtual int InsertExtraRTPPacket(
- int channel, unsigned char payloadType, bool markerBit,
- const char* payloadData, unsigned short payloadSize) = 0;
-
-protected:
- VoERTP_RTCP() {}
- virtual ~VoERTP_RTCP() {}
-};
-
-} // namespace webrtc
-
-#endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_H
diff --git a/trunk/src/voice_engine/main/interface/voe_video_sync.h b/trunk/src/voice_engine/main/interface/voe_video_sync.h
deleted file mode 100644
index ac3b84a..0000000
--- a/trunk/src/voice_engine/main/interface/voe_video_sync.h
+++ /dev/null
@@ -1,85 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-// - RTP header modification (time stamp and sequence number fields).
-// - Playout delay tuning to synchronize the voice with video.
-// - Playout delay monitoring.
-//
-// Usage example, omitting error checking:
-//
-// using namespace webrtc;
-// VoiceEngine* voe = VoiceEngine::Create();
-// VoEBase* base = VoEBase::GetInterface(voe);
-// VoEVideoSync* vsync = VoEVideoSync::GetInterface(voe);
-// base->Init();
-// ...
-// int buffer_ms(0);
-// vsync->GetPlayoutBufferSize(buffer_ms);
-// ...
-// base->Terminate();
-// base->Release();
-// vsync->Release();
-// VoiceEngine::Delete(voe);
-//
-#ifndef WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H
-#define WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class RtpRtcp;
-class VoiceEngine;
-
-class WEBRTC_DLLEXPORT VoEVideoSync
-{
-public:
- // Factory for the VoEVideoSync sub-API. Increases an internal
- // reference counter if successful. Returns NULL if the API is not
- // supported or if construction fails.
- static VoEVideoSync* GetInterface(VoiceEngine* voiceEngine);
-
- // Releases the VoEVideoSync sub-API and decreases an internal
- // reference counter. Returns the new reference count. This value should
- // be zero for all sub-API:s before the VoiceEngine object can be safely
- // deleted.
- virtual int Release() = 0;
-
- // Gets the current sound card buffer size (playout delay).
- virtual int GetPlayoutBufferSize(int& bufferMs) = 0;
-
- // Sets an additional delay for the playout jitter buffer.
- virtual int SetMinimumPlayoutDelay(int channel, int delayMs) = 0;
-
- // Gets the sum of the algorithmic delay, jitter buffer delay, and the
- // playout buffer delay for a specified |channel|.
- virtual int GetDelayEstimate(int channel, int& delayMs) = 0;
-
- // Manual initialization of the RTP timestamp.
- virtual int SetInitTimestamp(int channel, unsigned int timestamp) = 0;
-
- // Manual initialization of the RTP sequence number.
- virtual int SetInitSequenceNumber(int channel, short sequenceNumber) = 0;
-
- // Get the received RTP timestamp
- virtual int GetPlayoutTimestamp(int channel, unsigned int& timestamp) = 0;
-
- virtual int GetRtpRtcp (int channel, RtpRtcp* &rtpRtcpModule) = 0;
-
-protected:
- VoEVideoSync() { }
- virtual ~VoEVideoSync() { }
-};
-
-} // namespace webrtc
-
-#endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_H
diff --git a/trunk/src/voice_engine/main/interface/voe_volume_control.h b/trunk/src/voice_engine/main/interface/voe_volume_control.h
deleted file mode 100644
index 6d64e96..0000000
--- a/trunk/src/voice_engine/main/interface/voe_volume_control.h
+++ /dev/null
@@ -1,127 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// This sub-API supports the following functionalities:
-//
-// - Speaker volume controls.
-// - Microphone volume control.
-// - Non-linear speech level control.
-// - Mute functions.
-// - Additional stereo scaling methods.
-//
-// Usage example, omitting error checking:
-//
-// using namespace webrtc;
-// VoiceEngine* voe = VoiceEngine::Create();
-// VoEBase* base = VoEBase::GetInterface(voe);
-// VoEVolumeControl* volume = VoEVolumeControl::GetInterface(voe);
-// base->Init();
-// int ch = base->CreateChannel();
-// ...
-// volume->SetInputMute(ch, true);
-// ...
-// base->DeleteChannel(ch);
-// base->Terminate();
-// base->Release();
-// volume->Release();
-// VoiceEngine::Delete(voe);
-//
-#ifndef WEBRTC_VOICE_ENGINE_VOE_VOLUME_CONTROL_H
-#define WEBRTC_VOICE_ENGINE_VOE_VOLUME_CONTROL_H
-
-#include "common_types.h"
-
-namespace webrtc {
-
-class VoiceEngine;
-
-class WEBRTC_DLLEXPORT VoEVolumeControl
-{
-public:
- // Factory for the VoEVolumeControl sub-API. Increases an internal
- // reference counter if successful. Returns NULL if the API is not
- // supported or if construction fails.
- static VoEVolumeControl* GetInterface(VoiceEngine* voiceEngine);
-
- // Releases the VoEVolumeControl sub-API and decreases an internal
- // reference counter. Returns the new reference count. This value should
- // be zero for all sub-API:s before the VoiceEngine object can be safely
- // deleted.
- virtual int Release() = 0;
-
- // Sets the speaker |volume| level. Valid range is [0,255].
- virtual int SetSpeakerVolume(unsigned int volume) = 0;
-
- // Gets the speaker |volume| level.
- virtual int GetSpeakerVolume(unsigned int& volume) = 0;
-
- // Mutes the speaker device completely in the operating system.
- virtual int SetSystemOutputMute(bool enable) = 0;
-
- // Gets the output device mute state in the operating system.
- virtual int GetSystemOutputMute(bool &enabled) = 0;
-
- // Sets the microphone volume level. Valid range is [0,255].
- virtual int SetMicVolume(unsigned int volume) = 0;
-
- // Gets the microphone volume level.
- virtual int GetMicVolume(unsigned int& volume) = 0;
-
- // Mutes the microphone input signal completely without affecting
- // the audio device volume.
- virtual int SetInputMute(int channel, bool enable) = 0;
-
- // Gets the current microphone input mute state.
- virtual int GetInputMute(int channel, bool& enabled) = 0;
-
- // Mutes the microphone device completely in the operating system.
- virtual int SetSystemInputMute(bool enable) = 0;
-
- // Gets the mute state of the input device in the operating system.
- virtual int GetSystemInputMute(bool& enabled) = 0;
-
- // Gets the microphone speech |level|, mapped non-linearly to the range
- // [0,9].
- virtual int GetSpeechInputLevel(unsigned int& level) = 0;
-
- // Gets the speaker speech |level|, mapped non-linearly to the range
- // [0,9].
- virtual int GetSpeechOutputLevel(int channel, unsigned int& level) = 0;
-
- // Gets the microphone speech |level|, mapped linearly to the range
- // [0,32768].
- virtual int GetSpeechInputLevelFullRange(unsigned int& level) = 0;
-
- // Gets the speaker speech |level|, mapped linearly to the range [0,32768].
- virtual int GetSpeechOutputLevelFullRange(
- int channel, unsigned int& level) = 0;
-
- // Sets a volume |scaling| applied to the outgoing signal of a specific
- // channel. Valid scale range is [0.0, 10.0].
- virtual int SetChannelOutputVolumeScaling(int channel, float scaling) = 0;
-
- // Gets the current volume scaling for a specified |channel|.
- virtual int GetChannelOutputVolumeScaling(int channel, float& scaling) = 0;
-
- // Scales volume of the |left| and |right| channels independently.
- // Valid scale range is [0.0, 1.0].
- virtual int SetOutputVolumePan(int channel, float left, float right) = 0;
-
- // Gets the current left and right scaling factors.
- virtual int GetOutputVolumePan(int channel, float& left, float& right) = 0;
-
-protected:
- VoEVolumeControl() {};
- virtual ~VoEVolumeControl() {};
-};
-
-} // namespace webrtc
-
-#endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_VOLUME_CONTROL_H
diff --git a/trunk/src/voice_engine/main/source/Android.mk b/trunk/src/voice_engine/main/source/Android.mk
deleted file mode 100644
index 5040f17..0000000
--- a/trunk/src/voice_engine/main/source/Android.mk
+++ /dev/null
@@ -1,88 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../android-webrtc.mk
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE := libwebrtc_voe_core
-LOCAL_MODULE_TAGS := optional
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := \
- audio_frame_operations.cc \
- channel.cc \
- channel_manager.cc \
- channel_manager_base.cc \
- dtmf_inband.cc \
- dtmf_inband_queue.cc \
- level_indicator.cc \
- monitor_module.cc \
- output_mixer.cc \
- ref_count.cc \
- shared_data.cc \
- statistics.cc \
- transmit_mixer.cc \
- utility.cc \
- voe_audio_processing_impl.cc \
- voe_base_impl.cc \
- voe_call_report_impl.cc \
- voe_codec_impl.cc \
- voe_dtmf_impl.cc \
- voe_encryption_impl.cc \
- voe_external_media_impl.cc \
- voe_file_impl.cc \
- voe_hardware_impl.cc \
- voe_neteq_stats_impl.cc \
- voe_network_impl.cc \
- voe_rtp_rtcp_impl.cc \
- voe_video_sync_impl.cc \
- voe_volume_control_impl.cc \
- voice_engine_impl.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- $(MY_WEBRTC_COMMON_DEFS) \
- '-DWEBRTC_ANDROID_OPENSLES'
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../interface \
- $(LOCAL_PATH)/../../.. \
- $(LOCAL_PATH)/../../../common_audio/resampler/include \
- $(LOCAL_PATH)/../../../common_audio/signal_processing/include \
- $(LOCAL_PATH)/../../../modules/interface \
- $(LOCAL_PATH)/../../../modules/audio_coding/main/interface \
- $(LOCAL_PATH)/../../../modules/audio_conference_mixer/interface \
- $(LOCAL_PATH)/../../../modules/audio_device/main/interface \
- $(LOCAL_PATH)/../../../modules/audio_device/main/source \
- $(LOCAL_PATH)/../../../modules/audio_processing/include \
- $(LOCAL_PATH)/../../../modules/media_file/interface \
- $(LOCAL_PATH)/../../../modules/rtp_rtcp/interface \
- $(LOCAL_PATH)/../../../modules/udp_transport/interface \
- $(LOCAL_PATH)/../../../modules/utility/interface \
- $(LOCAL_PATH)/../../../system_wrappers/interface
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libdl \
- libstlport
-
-ifeq ($(TARGET_OS)-$(TARGET_SIMULATOR),linux-true)
-LOCAL_LDLIBS += -ldl -lpthread
-endif
-
-ifneq ($(TARGET_SIMULATOR),true)
-LOCAL_SHARED_LIBRARIES += libdl
-endif
-
-ifndef NDK_ROOT
-include external/stlport/libstlport.mk
-endif
-include $(BUILD_STATIC_LIBRARY)
diff --git a/trunk/src/voice_engine/main/source/audio_frame_operations.cc b/trunk/src/voice_engine/main/source/audio_frame_operations.cc
deleted file mode 100644
index e08d0a2..0000000
--- a/trunk/src/voice_engine/main/source/audio_frame_operations.cc
+++ /dev/null
@@ -1,129 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "audio_frame_operations.h"
-#include "module_common_types.h"
-
-namespace webrtc {
-
-namespace voe {
-
-WebRtc_Word32
-AudioFrameOperations::MonoToStereo(AudioFrame& audioFrame)
-{
- if (audioFrame._audioChannel != 1)
- {
- return -1;
- }
- if ((audioFrame._payloadDataLengthInSamples << 1) >=
- AudioFrame::kMaxAudioFrameSizeSamples)
- {
- // not enough memory to expand from mono to stereo
- return -1;
- }
-
- WebRtc_Word16* payloadCopy =
- new WebRtc_Word16[audioFrame._payloadDataLengthInSamples];
- memcpy(payloadCopy, audioFrame._payloadData,
- sizeof(WebRtc_Word16)*audioFrame._payloadDataLengthInSamples);
-
- for (int i = 0; i < audioFrame._payloadDataLengthInSamples; i++)
- {
- audioFrame._payloadData[2*i] = payloadCopy[i];
- audioFrame._payloadData[2*i+1] = payloadCopy[i];
- }
-
- audioFrame._audioChannel = 2;
-
- delete [] payloadCopy;
- return 0;
-}
-
-WebRtc_Word32
-AudioFrameOperations::StereoToMono(AudioFrame& audioFrame)
-{
- if (audioFrame._audioChannel != 2)
- {
- return -1;
- }
-
- for (int i = 0; i < audioFrame._payloadDataLengthInSamples; i++)
- {
- audioFrame._payloadData[i] = (audioFrame._payloadData[2*i] >> 1) +
- (audioFrame._payloadData[2*i+1] >> 1);
- }
-
- audioFrame._audioChannel = 1;
-
- return 0;
-}
-
-WebRtc_Word32
-AudioFrameOperations::Mute(AudioFrame& audioFrame)
-{
- const int sizeInBytes = sizeof(WebRtc_Word16) *
- audioFrame._payloadDataLengthInSamples * audioFrame._audioChannel;
- memset(audioFrame._payloadData, 0, sizeInBytes);
- audioFrame._energy = 0;
- return 0;
-}
-
-WebRtc_Word32
-AudioFrameOperations::Scale(const float left,
- const float right,
- AudioFrame& audioFrame)
-{
- if (audioFrame._audioChannel == 1)
- {
- assert(false);
- return -1;
- }
-
- for (int i = 0; i < audioFrame._payloadDataLengthInSamples; i++)
- {
- audioFrame._payloadData[2*i] =
- (WebRtc_Word16)(left*audioFrame._payloadData[2*i]);
- audioFrame._payloadData[2*i+1] =
- (WebRtc_Word16)(right*audioFrame._payloadData[2*i+1]);
- }
- return 0;
-}
-
-WebRtc_Word32
-AudioFrameOperations::ScaleWithSat(const float scale, AudioFrame& audioFrame)
-{
- WebRtc_Word32 tmp(0);
-
- // Ensure that the output result is saturated [-32768, +32768].
- for (int i = 0;
- i < audioFrame._payloadDataLengthInSamples * audioFrame._audioChannel;
- i++)
- {
- tmp = static_cast<WebRtc_Word32> (scale * audioFrame._payloadData[i]);
- if (tmp < -32768)
- {
- audioFrame._payloadData[i] = -32768;
- }
- else if (tmp > 32767)
- {
- audioFrame._payloadData[i] = 32767;
- }
- else
- {
- audioFrame._payloadData[i] = static_cast<WebRtc_Word16> (tmp);
- }
- }
- return 0;
-}
-
-} // namespace voe
-
-} // namespace webrtc
-
diff --git a/trunk/src/voice_engine/main/source/audio_frame_operations.h b/trunk/src/voice_engine/main/source/audio_frame_operations.h
deleted file mode 100644
index 368850b..0000000
--- a/trunk/src/voice_engine/main/source/audio_frame_operations.h
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H
-#define WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H
-
-#include "typedefs.h"
-
-namespace webrtc {
-
-class AudioFrame;
-
-namespace voe {
-
-class AudioFrameOperations
-{
-public:
- static WebRtc_Word32 MonoToStereo(AudioFrame& audioFrame);
-
- static WebRtc_Word32 StereoToMono(AudioFrame& audioFrame);
-
- static WebRtc_Word32 Mute(AudioFrame& audioFrame);
-
- static WebRtc_Word32 Scale(const float left,
- const float right,
- AudioFrame& audioFrame);
-
- static WebRtc_Word32 ScaleWithSat(const float scale,
- AudioFrame& audioFrame);
-};
-
-} // namespace voe
-
-} // namespace webrtc
-
-#endif // #ifndef WEBRTC_VOICE_ENGINE_AUDIO_FRAME_OPERATIONS_H
diff --git a/trunk/src/voice_engine/main/source/channel.cc b/trunk/src/voice_engine/main/source/channel.cc
deleted file mode 100644
index 4fc1c8e..0000000
--- a/trunk/src/voice_engine/main/source/channel.cc
+++ /dev/null
@@ -1,6684 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "channel.h"
-
-#include "audio_device.h"
-#include "audio_frame_operations.h"
-#include "audio_processing.h"
-#include "critical_section_wrapper.h"
-#include "output_mixer.h"
-#include "process_thread.h"
-#include "rtp_dump.h"
-#include "statistics.h"
-#include "trace.h"
-#include "transmit_mixer.h"
-#include "utility.h"
-#include "voe_base.h"
-#include "voe_external_media.h"
-#include "voe_rtp_rtcp.h"
-
-#if defined(_WIN32)
-#include <Qos.h>
-#endif
-
-namespace webrtc
-{
-
-namespace voe
-{
-
-WebRtc_Word32
-Channel::SendData(FrameType frameType,
- WebRtc_UWord8 payloadType,
- WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- WebRtc_UWord16 payloadSize,
- const RTPFragmentationHeader* fragmentation)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SendData(frameType=%u, payloadType=%u, timeStamp=%u,"
- " payloadSize=%u, fragmentation=0x%x)",
- frameType, payloadType, timeStamp, payloadSize, fragmentation);
-
- if (_includeAudioLevelIndication)
- {
- assert(_rtpAudioProc.get() != NULL);
- // Store current audio level in the RTP/RTCP module.
- // The level will be used in combination with voice-activity state
- // (frameType) to add an RTP header extension
- _rtpRtcpModule.SetAudioLevel(_rtpAudioProc->level_estimator()->RMS());
- }
-
- // Push data from ACM to RTP/RTCP-module to deliver audio frame for
- // packetization.
- // This call will trigger Transport::SendPacket() from the RTP/RTCP module.
- if (_rtpRtcpModule.SendOutgoingData((FrameType&)frameType,
- payloadType,
- timeStamp,
- payloadData,
- payloadSize,
- fragmentation) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
- "Channel::SendData() failed to send data to RTP/RTCP module");
- return -1;
- }
-
- _lastLocalTimeStamp = timeStamp;
- _lastPayloadType = payloadType;
-
- return 0;
-}
-
-WebRtc_Word32
-Channel::InFrameType(WebRtc_Word16 frameType)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::InFrameType(frameType=%d)", frameType);
-
- CriticalSectionScoped cs(_callbackCritSect);
- // 1 indicates speech
- _sendFrameType = (frameType == 1) ? 1 : 0;
- return 0;
-}
-
-#ifdef WEBRTC_DTMF_DETECTION
-int
-Channel::IncomingDtmf(const WebRtc_UWord8 digitDtmf, const bool end)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::IncomingDtmf(digitDtmf=%u, end=%d)",
- digitDtmf, end);
-
- if (digitDtmf != 999)
- {
- CriticalSectionScoped cs(_callbackCritSect);
- if (_telephoneEventDetectionPtr)
- {
- _telephoneEventDetectionPtr->OnReceivedTelephoneEventInband(
- _channelId, digitDtmf, end);
- }
- }
-
- return 0;
-}
-#endif
-
-WebRtc_Word32
-Channel::OnRxVadDetected(const int vadDecision)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::OnRxVadDetected(vadDecision=%d)", vadDecision);
-
- CriticalSectionScoped cs(_callbackCritSect);
- if (_rxVadObserverPtr)
- {
- _rxVadObserverPtr->OnRxVad(_channelId, vadDecision);
- }
-
- return 0;
-}
-
-int
-Channel::SendPacket(int channel, const void *data, int len)
-{
- channel = VoEChannelId(channel);
- assert(channel == _channelId);
-
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SendPacket(channel=%d, len=%d)", channel, len);
-
- if (_transportPtr == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SendPacket() failed to send RTP packet due to"
- " invalid transport object");
- return -1;
- }
-
- // Insert extra RTP packet using if user has called the InsertExtraRTPPacket
- // API
- if (_insertExtraRTPPacket)
- {
- WebRtc_UWord8* rtpHdr = (WebRtc_UWord8*)data;
- WebRtc_UWord8 M_PT(0);
- if (_extraMarkerBit)
- {
- M_PT = 0x80; // set the M-bit
- }
- M_PT += _extraPayloadType; // set the payload type
- *(++rtpHdr) = M_PT; // modify the M|PT-byte within the RTP header
- _insertExtraRTPPacket = false; // insert one packet only
- }
-
- WebRtc_UWord8* bufferToSendPtr = (WebRtc_UWord8*)data;
- WebRtc_Word32 bufferLength = len;
-
- // Dump the RTP packet to a file (if RTP dump is enabled).
- if (_rtpDumpOut.DumpPacket((const WebRtc_UWord8*)data, len) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::SendPacket() RTP dump to output file failed");
- }
-
- // SRTP or External encryption
- if (_encrypting)
- {
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_encryptionPtr)
- {
- if (!_encryptionRTPBufferPtr)
- {
- // Allocate memory for encryption buffer one time only
- _encryptionRTPBufferPtr =
- new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes];
- }
-
- // Perform encryption (SRTP or external)
- WebRtc_Word32 encryptedBufferLength = 0;
- _encryptionPtr->encrypt(_channelId,
- bufferToSendPtr,
- _encryptionRTPBufferPtr,
- bufferLength,
- (int*)&encryptedBufferLength);
- if (encryptedBufferLength <= 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ENCRYPTION_FAILED,
- kTraceError, "Channel::SendPacket() encryption failed");
- return -1;
- }
-
- // Replace default data buffer with encrypted buffer
- bufferToSendPtr = _encryptionRTPBufferPtr;
- bufferLength = encryptedBufferLength;
- }
- }
-
- // Packet transmission using WebRtc socket transport
- if (!_externalTransport)
- {
- int n = _transportPtr->SendPacket(channel, bufferToSendPtr,
- bufferLength);
- if (n < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::SendPacket() RTP transmission using WebRtc"
- " sockets failed");
- return -1;
- }
- return n;
- }
-
- // Packet transmission using external transport transport
- {
- CriticalSectionScoped cs(_callbackCritSect);
-
- int n = _transportPtr->SendPacket(channel,
- bufferToSendPtr,
- bufferLength);
- if (n < 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::SendPacket() RTP transmission using external"
- " transport failed");
- return -1;
- }
- return n;
- }
-}
-
-int
-Channel::SendRTCPPacket(int channel, const void *data, int len)
-{
- channel = VoEChannelId(channel);
- assert(channel == _channelId);
-
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SendRTCPPacket(channel=%d, len=%d)", channel, len);
-
- {
- CriticalSectionScoped cs(_callbackCritSect);
- if (_transportPtr == NULL)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::SendRTCPPacket() failed to send RTCP packet"
- " due to invalid transport object");
- return -1;
- }
- }
-
- WebRtc_UWord8* bufferToSendPtr = (WebRtc_UWord8*)data;
- WebRtc_Word32 bufferLength = len;
-
- // Dump the RTCP packet to a file (if RTP dump is enabled).
- if (_rtpDumpOut.DumpPacket((const WebRtc_UWord8*)data, len) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::SendPacket() RTCP dump to output file failed");
- }
-
- // SRTP or External encryption
- if (_encrypting)
- {
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_encryptionPtr)
- {
- if (!_encryptionRTCPBufferPtr)
- {
- // Allocate memory for encryption buffer one time only
- _encryptionRTCPBufferPtr =
- new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes];
- }
-
- // Perform encryption (SRTP or external).
- WebRtc_Word32 encryptedBufferLength = 0;
- _encryptionPtr->encrypt_rtcp(_channelId,
- bufferToSendPtr,
- _encryptionRTCPBufferPtr,
- bufferLength,
- (int*)&encryptedBufferLength);
- if (encryptedBufferLength <= 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ENCRYPTION_FAILED, kTraceError,
- "Channel::SendRTCPPacket() encryption failed");
- return -1;
- }
-
- // Replace default data buffer with encrypted buffer
- bufferToSendPtr = _encryptionRTCPBufferPtr;
- bufferLength = encryptedBufferLength;
- }
- }
-
- // Packet transmission using WebRtc socket transport
- if (!_externalTransport)
- {
- int n = _transportPtr->SendRTCPPacket(channel,
- bufferToSendPtr,
- bufferLength);
- if (n < 0)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::SendRTCPPacket() transmission using WebRtc"
- " sockets failed");
- return -1;
- }
- return n;
- }
-
- // Packet transmission using external transport transport
- {
- CriticalSectionScoped cs(_callbackCritSect);
-
- int n = _transportPtr->SendRTCPPacket(channel,
- bufferToSendPtr,
- bufferLength);
- if (n < 0)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::SendRTCPPacket() transmission using external"
- " transport failed");
- return -1;
- }
- return n;
- }
-
- return len;
-}
-
-void
-Channel::IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
- const WebRtc_Word32 rtpPacketLength,
- const WebRtc_Word8* fromIP,
- const WebRtc_UWord16 fromPort)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::IncomingRTPPacket(rtpPacketLength=%d,"
- " fromIP=%s, fromPort=%u)",
- rtpPacketLength, fromIP, fromPort);
-
- // Store playout timestamp for the received RTP packet
- // to be used for upcoming delay estimations
- WebRtc_UWord32 playoutTimestamp(0);
- if (GetPlayoutTimeStamp(playoutTimestamp) == 0)
- {
- _playoutTimeStampRTP = playoutTimestamp;
- }
-
- WebRtc_UWord8* rtpBufferPtr = (WebRtc_UWord8*)incomingRtpPacket;
- WebRtc_Word32 rtpBufferLength = rtpPacketLength;
-
- // SRTP or External decryption
- if (_decrypting)
- {
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_encryptionPtr)
- {
- if (!_decryptionRTPBufferPtr)
- {
- // Allocate memory for decryption buffer one time only
- _decryptionRTPBufferPtr =
- new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes];
- }
-
- // Perform decryption (SRTP or external)
- WebRtc_Word32 decryptedBufferLength = 0;
- _encryptionPtr->decrypt(_channelId,
- rtpBufferPtr,
- _decryptionRTPBufferPtr,
- rtpBufferLength,
- (int*)&decryptedBufferLength);
- if (decryptedBufferLength <= 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_DECRYPTION_FAILED, kTraceError,
- "Channel::IncomingRTPPacket() decryption failed");
- return;
- }
-
- // Replace default data buffer with decrypted buffer
- rtpBufferPtr = _decryptionRTPBufferPtr;
- rtpBufferLength = decryptedBufferLength;
- }
- }
-
- // Dump the RTP packet to a file (if RTP dump is enabled).
- if (_rtpDumpIn.DumpPacket(rtpBufferPtr,
- (WebRtc_UWord16)rtpBufferLength) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::SendPacket() RTP dump to input file failed");
- }
-
- // Deliver RTP packet to RTP/RTCP module for parsing
- // The packet will be pushed back to the channel thru the
- // OnReceivedPayloadData callback so we don't push it to the ACM here
- if (_rtpRtcpModule.IncomingPacket((const WebRtc_UWord8*)rtpBufferPtr,
- (WebRtc_UWord16)rtpBufferLength) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
- "Channel::IncomingRTPPacket() RTP packet is invalid");
- return;
- }
-}
-
-void
-Channel::IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
- const WebRtc_Word32 rtcpPacketLength,
- const WebRtc_Word8* fromIP,
- const WebRtc_UWord16 fromPort)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::IncomingRTCPPacket(rtcpPacketLength=%d, fromIP=%s,"
- " fromPort=%u)",
- rtcpPacketLength, fromIP, fromPort);
-
- // Temporary buffer pointer and size for decryption
- WebRtc_UWord8* rtcpBufferPtr = (WebRtc_UWord8*)incomingRtcpPacket;
- WebRtc_Word32 rtcpBufferLength = rtcpPacketLength;
-
- // Store playout timestamp for the received RTCP packet
- // which will be read by the GetRemoteRTCPData API
- WebRtc_UWord32 playoutTimestamp(0);
- if (GetPlayoutTimeStamp(playoutTimestamp) == 0)
- {
- _playoutTimeStampRTCP = playoutTimestamp;
- }
-
- // SRTP or External decryption
- if (_decrypting)
- {
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_encryptionPtr)
- {
- if (!_decryptionRTCPBufferPtr)
- {
- // Allocate memory for decryption buffer one time only
- _decryptionRTCPBufferPtr =
- new WebRtc_UWord8[kVoiceEngineMaxIpPacketSizeBytes];
- }
-
- // Perform decryption (SRTP or external).
- WebRtc_Word32 decryptedBufferLength = 0;
- _encryptionPtr->decrypt_rtcp(_channelId,
- rtcpBufferPtr,
- _decryptionRTCPBufferPtr,
- rtcpBufferLength,
- (int*)&decryptedBufferLength);
- if (decryptedBufferLength <= 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_DECRYPTION_FAILED, kTraceError,
- "Channel::IncomingRTCPPacket() decryption failed");
- return;
- }
-
- // Replace default data buffer with decrypted buffer
- rtcpBufferPtr = _decryptionRTCPBufferPtr;
- rtcpBufferLength = decryptedBufferLength;
- }
- }
-
- // Dump the RTCP packet to a file (if RTP dump is enabled).
- if (_rtpDumpIn.DumpPacket(rtcpBufferPtr,
- (WebRtc_UWord16)rtcpBufferLength) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::SendPacket() RTCP dump to input file failed");
- }
-
- // Deliver RTCP packet to RTP/RTCP module for parsing
- if (_rtpRtcpModule.IncomingPacket((const WebRtc_UWord8*)rtcpBufferPtr,
- (WebRtc_UWord16)rtcpBufferLength) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
- "Channel::IncomingRTPPacket() RTCP packet is invalid");
- return;
- }
-}
-
-void
-Channel::OnReceivedTelephoneEvent(const WebRtc_Word32 id,
- const WebRtc_UWord8 event,
- const bool endOfEvent)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::OnReceivedTelephoneEvent(id=%d, event=%u,"
- " endOfEvent=%d)", id, event, endOfEvent);
-
-#ifdef WEBRTC_DTMF_DETECTION
- if (_outOfBandTelephoneEventDetecion)
- {
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_telephoneEventDetectionPtr)
- {
- _telephoneEventDetectionPtr->OnReceivedTelephoneEventOutOfBand(
- _channelId, event, endOfEvent);
- }
- }
-#endif
-}
-
-void
-Channel::OnPlayTelephoneEvent(const WebRtc_Word32 id,
- const WebRtc_UWord8 event,
- const WebRtc_UWord16 lengthMs,
- const WebRtc_UWord8 volume)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::OnPlayTelephoneEvent(id=%d, event=%u, lengthMs=%u,"
- " volume=%u)", id, event, lengthMs, volume);
-
- if (!_playOutbandDtmfEvent || (event > 15))
- {
- // Ignore callback since feedback is disabled or event is not a
- // Dtmf tone event.
- return;
- }
-
- assert(_outputMixerPtr != NULL);
-
- // Start playing out the Dtmf tone (if playout is enabled).
- // Reduce length of tone with 80ms to the reduce risk of echo.
- _outputMixerPtr->PlayDtmfTone(event, lengthMs - 80, volume);
-}
-
-void
-Channel::OnIncomingSSRCChanged(const WebRtc_Word32 id,
- const WebRtc_UWord32 SSRC)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::OnIncomingSSRCChanged(id=%d, SSRC=%d)",
- id, SSRC);
-
- WebRtc_Word32 channel = VoEChannelId(id);
- assert(channel == _channelId);
-
- // Reset RTP-module counters since a new incoming RTP stream is detected
- _rtpRtcpModule.ResetReceiveDataCountersRTP();
- _rtpRtcpModule.ResetStatisticsRTP();
-
- if (_rtpObserver)
- {
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_rtpObserverPtr)
- {
- // Send new SSRC to registered observer using callback
- _rtpObserverPtr->OnIncomingSSRCChanged(channel, SSRC);
- }
- }
-}
-
-void Channel::OnIncomingCSRCChanged(const WebRtc_Word32 id,
- const WebRtc_UWord32 CSRC,
- const bool added)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::OnIncomingCSRCChanged(id=%d, CSRC=%d, added=%d)",
- id, CSRC, added);
-
- WebRtc_Word32 channel = VoEChannelId(id);
- assert(channel == _channelId);
-
- if (_rtpObserver)
- {
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_rtpObserverPtr)
- {
- _rtpObserverPtr->OnIncomingCSRCChanged(channel, CSRC, added);
- }
- }
-}
-
-void
-Channel::OnApplicationDataReceived(const WebRtc_Word32 id,
- const WebRtc_UWord8 subType,
- const WebRtc_UWord32 name,
- const WebRtc_UWord16 length,
- const WebRtc_UWord8* data)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::OnApplicationDataReceived(id=%d, subType=%u,"
- " name=%u, length=%u)",
- id, subType, name, length);
-
- WebRtc_Word32 channel = VoEChannelId(id);
- assert(channel == _channelId);
-
- if (_rtcpObserver)
- {
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_rtcpObserverPtr)
- {
- _rtcpObserverPtr->OnApplicationDataReceived(channel,
- subType,
- name,
- data,
- length);
- }
- }
-}
-
-WebRtc_Word32
-Channel::OnInitializeDecoder(
- const WebRtc_Word32 id,
- const WebRtc_Word8 payloadType,
- const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
- const int frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::OnInitializeDecoder(id=%d, payloadType=%d, "
- "payloadName=%s, frequency=%u, channels=%u, rate=%u)",
- id, payloadType, payloadName, frequency, channels, rate);
-
- assert(VoEChannelId(id) == _channelId);
-
- CodecInst receiveCodec = {0};
- CodecInst dummyCodec = {0};
-
- receiveCodec.pltype = payloadType;
- receiveCodec.plfreq = frequency;
- receiveCodec.channels = channels;
- receiveCodec.rate = rate;
- strncpy(receiveCodec.plname, payloadName, RTP_PAYLOAD_NAME_SIZE - 1);
-
- _audioCodingModule.Codec(payloadName, dummyCodec, frequency);
- receiveCodec.pacsize = dummyCodec.pacsize;
-
- // Register the new codec to the ACM
- if (_audioCodingModule.RegisterReceiveCodec(receiveCodec) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "Channel::OnInitializeDecoder() invalid codec ("
- "pt=%d, name=%s) received - 1", payloadType, payloadName);
- _engineStatisticsPtr->SetLastError(VE_AUDIO_CODING_MODULE_ERROR);
- return -1;
- }
-
- return 0;
-}
-
-void
-Channel::OnPacketTimeout(const WebRtc_Word32 id)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::OnPacketTimeout(id=%d)", id);
-
- CriticalSectionScoped cs(*_callbackCritSectPtr);
- if (_voiceEngineObserverPtr)
- {
- if (_receiving || _externalTransport)
- {
- WebRtc_Word32 channel = VoEChannelId(id);
- assert(channel == _channelId);
- // Ensure that next OnReceivedPacket() callback will trigger
- // a VE_PACKET_RECEIPT_RESTARTED callback.
- _rtpPacketTimedOut = true;
- // Deliver callback to the observer
- WEBRTC_TRACE(kTraceInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::OnPacketTimeout() => "
- "CallbackOnError(VE_RECEIVE_PACKET_TIMEOUT)");
- _voiceEngineObserverPtr->CallbackOnError(channel,
- VE_RECEIVE_PACKET_TIMEOUT);
- }
- }
-}
-
-void
-Channel::OnReceivedPacket(const WebRtc_Word32 id,
- const RtpRtcpPacketType packetType)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::OnReceivedPacket(id=%d, packetType=%d)",
- id, packetType);
-
- assert(VoEChannelId(id) == _channelId);
-
- // Notify only for the case when we have restarted an RTP session.
- if (_rtpPacketTimedOut && (kPacketRtp == packetType))
- {
- CriticalSectionScoped cs(*_callbackCritSectPtr);
- if (_voiceEngineObserverPtr)
- {
- WebRtc_Word32 channel = VoEChannelId(id);
- assert(channel == _channelId);
- // Reset timeout mechanism
- _rtpPacketTimedOut = false;
- // Deliver callback to the observer
- WEBRTC_TRACE(kTraceInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::OnPacketTimeout() =>"
- " CallbackOnError(VE_PACKET_RECEIPT_RESTARTED)");
- _voiceEngineObserverPtr->CallbackOnError(
- channel,
- VE_PACKET_RECEIPT_RESTARTED);
- }
- }
-}
-
-void
-Channel::OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
- const RTPAliveType alive)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::OnPeriodicDeadOrAlive(id=%d, alive=%d)", id, alive);
-
- if (!_connectionObserver)
- return;
-
- WebRtc_Word32 channel = VoEChannelId(id);
- assert(channel == _channelId);
-
- // Use Alive as default to limit risk of false Dead detections
- bool isAlive(true);
-
- // Always mark the connection as Dead when the module reports kRtpDead
- if (kRtpDead == alive)
- {
- isAlive = false;
- }
-
- // It is possible that the connection is alive even if no RTP packet has
- // been received for a long time since the other side might use VAD/DTX
- // and a low SID-packet update rate.
- if ((kRtpNoRtp == alive) && _playing)
- {
- // Detect Alive for all NetEQ states except for the case when we are
- // in PLC_CNG state.
- // PLC_CNG <=> background noise only due to long expand or error.
- // Note that, the case where the other side stops sending during CNG
- // state will be detected as Alive. Dead is is not set until after
- // missing RTCP packets for at least twelve seconds (handled
- // internally by the RTP/RTCP module).
- isAlive = (_outputSpeechType != AudioFrame::kPLCCNG);
- }
-
- UpdateDeadOrAliveCounters(isAlive);
-
- // Send callback to the registered observer
- if (_connectionObserver)
- {
- CriticalSectionScoped cs(_callbackCritSect);
- if (_connectionObserverPtr)
- {
- _connectionObserverPtr->OnPeriodicDeadOrAlive(channel, isAlive);
- }
- }
-}
-
-WebRtc_Word32
-Channel::OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const WebRtcRTPHeader* rtpHeader)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::OnReceivedPayloadData(payloadSize=%d,"
- " payloadType=%u, audioChannel=%u)",
- payloadSize,
- rtpHeader->header.payloadType,
- rtpHeader->type.Audio.channel);
-
- if (!_playing)
- {
- // Avoid inserting into NetEQ when we are not playing. Count the
- // packet as discarded.
- WEBRTC_TRACE(kTraceStream, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "received packet is discarded since playing is not"
- " activated");
- _numberOfDiscardedPackets++;
- return 0;
- }
-
- // Push the incoming payload (parsed and ready for decoding) into the ACM
- if (_audioCodingModule.IncomingPacket((const WebRtc_Word8*) payloadData,
- payloadSize,
- *rtpHeader) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning,
- "Channel::OnReceivedPayloadData() unable to push data to the ACM");
- return -1;
- }
-
- // Update the packet delay
- UpdatePacketDelay(rtpHeader->header.timestamp,
- rtpHeader->header.sequenceNumber);
-
- return 0;
-}
-
-WebRtc_Word32 Channel::GetAudioFrame(const WebRtc_Word32 id,
- AudioFrame& audioFrame)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetAudioFrame(id=%d)", id);
-
- // Get 10ms raw PCM data from the ACM (mixer limits output frequency)
- if (_audioCodingModule.PlayoutData10Ms(audioFrame._frequencyInHz,
- audioFrame) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::GetAudioFrame() PlayoutData10Ms() failed!");
- // In all likelihood, the audio in this frame is garbage. We return an
- // error so that the audio mixer module doesn't add it to the mix. As
- // a result, it won't be played out and the actions skipped here are
- // irrelevant.
- return -1;
- }
-
- if (_RxVadDetection)
- {
- UpdateRxVadDetection(audioFrame);
- }
-
- // Convert module ID to internal VoE channel ID
- audioFrame._id = VoEChannelId(audioFrame._id);
- // Store speech type for dead-or-alive detection
- _outputSpeechType = audioFrame._speechType;
-
- // Perform far-end AudioProcessing module processing on the received signal
- if (_rxApmIsEnabled)
- {
- ApmProcessRx(audioFrame);
- }
-
- // Output volume scaling
- if (_outputGain < 0.99f || _outputGain > 1.01f)
- {
- AudioFrameOperations::ScaleWithSat(_outputGain, audioFrame);
- }
-
- // Scale left and/or right channel(s) if stereo and master balance is
- // active
-
- if (_panLeft != 1.0f || _panRight != 1.0f)
- {
- if (audioFrame._audioChannel == 1)
- {
- // Emulate stereo mode since panning is active.
- // The mono signal is copied to both left and right channels here.
- AudioFrameOperations::MonoToStereo(audioFrame);
- }
- // For true stereo mode (when we are receiving a stereo signal), no
- // action is needed.
-
- // Do the panning operation (the audio frame contains stereo at this
- // stage)
- AudioFrameOperations::Scale(_panLeft, _panRight, audioFrame);
- }
-
- // Mix decoded PCM output with file if file mixing is enabled
- if (_outputFilePlaying)
- {
- MixAudioWithFile(audioFrame, audioFrame._frequencyInHz);
- }
-
- // Place channel in on-hold state (~muted) if on-hold is activated
- if (_outputIsOnHold)
- {
- AudioFrameOperations::Mute(audioFrame);
- }
-
- // External media
- if (_outputExternalMedia)
- {
- CriticalSectionScoped cs(_callbackCritSect);
- const bool isStereo = (audioFrame._audioChannel == 2);
- if (_outputExternalMediaCallbackPtr)
- {
- _outputExternalMediaCallbackPtr->Process(
- _channelId,
- kPlaybackPerChannel,
- (WebRtc_Word16*)audioFrame._payloadData,
- audioFrame._payloadDataLengthInSamples,
- audioFrame._frequencyInHz,
- isStereo);
- }
- }
-
- // Record playout if enabled
- {
- CriticalSectionScoped cs(_fileCritSect);
-
- if (_outputFileRecording && _outputFileRecorderPtr)
- {
- if(audioFrame._audioChannel == 2)
- {
- AudioFrame temp = audioFrame;
- AudioFrameOperations::StereoToMono (temp);
- _outputFileRecorderPtr->RecordAudioToFile(temp);
- }
- else if(audioFrame._audioChannel == 1)
- {
- _outputFileRecorderPtr->RecordAudioToFile(audioFrame);
- }
- else
- {
- assert(false);
- }
- }
- }
-
- // Measure audio level (0-9)
- _outputAudioLevel.ComputeLevel(audioFrame);
-
- return 0;
-}
-
-WebRtc_Word32
-Channel::NeededFrequency(const WebRtc_Word32 id)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::NeededFrequency(id=%d)", id);
-
- int highestNeeded = 0;
-
- // Determine highest needed receive frequency
- WebRtc_Word32 receiveFrequency = _audioCodingModule.ReceiveFrequency();
-
- // Return the bigger of playout and receive frequency in the ACM.
- if (_audioCodingModule.PlayoutFrequency() > receiveFrequency)
- {
- highestNeeded = _audioCodingModule.PlayoutFrequency();
- }
- else
- {
- highestNeeded = receiveFrequency;
- }
-
- // Special case, if we're playing a file on the playout side
- // we take that frequency into consideration as well
- // This is not needed on sending side, since the codec will
- // limit the spectrum anyway.
- if (_outputFilePlaying)
- {
- CriticalSectionScoped cs(_fileCritSect);
- if (_outputFilePlayerPtr && _outputFilePlaying)
- {
- if(_outputFilePlayerPtr->Frequency()>highestNeeded)
- {
- highestNeeded=_outputFilePlayerPtr->Frequency();
- }
- }
- }
-
- return(highestNeeded);
-}
-
-WebRtc_Word32
-Channel::CreateChannel(Channel*& channel,
- const WebRtc_Word32 channelId,
- const WebRtc_UWord32 instanceId)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId,channelId),
- "Channel::CreateChannel(channelId=%d, instanceId=%d)",
- channelId, instanceId);
-
- channel = new Channel(channelId, instanceId);
- if (channel == NULL)
- {
- WEBRTC_TRACE(kTraceMemory, kTraceVoice,
- VoEId(instanceId,channelId),
- "Channel::CreateChannel() unable to allocate memory for"
- " channel");
- return -1;
- }
- return 0;
-}
-
-void
-Channel::PlayNotification(const WebRtc_Word32 id,
- const WebRtc_UWord32 durationMs)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::PlayNotification(id=%d, durationMs=%d)",
- id, durationMs);
-
- // Not implement yet
-}
-
-void
-Channel::RecordNotification(const WebRtc_Word32 id,
- const WebRtc_UWord32 durationMs)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::RecordNotification(id=%d, durationMs=%d)",
- id, durationMs);
-
- // Not implement yet
-}
-
-void
-Channel::PlayFileEnded(const WebRtc_Word32 id)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::PlayFileEnded(id=%d)", id);
-
- if (id == _inputFilePlayerId)
- {
- CriticalSectionScoped cs(_fileCritSect);
-
- _inputFilePlaying = false;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::PlayFileEnded() => input file player module is"
- " shutdown");
- }
- else if (id == _outputFilePlayerId)
- {
- CriticalSectionScoped cs(_fileCritSect);
-
- _outputFilePlaying = false;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::PlayFileEnded() => output file player module is"
- " shutdown");
- }
-}
-
-void
-Channel::RecordFileEnded(const WebRtc_Word32 id)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::RecordFileEnded(id=%d)", id);
-
- assert(id == _outputFileRecorderId);
-
- CriticalSectionScoped cs(_fileCritSect);
-
- _outputFileRecording = false;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::RecordFileEnded() => output file recorder module is"
- " shutdown");
-}
-
-Channel::Channel(const WebRtc_Word32 channelId,
- const WebRtc_UWord32 instanceId) :
- _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _transmitCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _instanceId(instanceId),
- _channelId(channelId),
- _rtpRtcpModule(*RtpRtcp::CreateRtpRtcp(VoEModuleId(
- instanceId, channelId), true)),
- _audioCodingModule(*AudioCodingModule::Create(
- VoEModuleId(instanceId, channelId))),
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- _numSocketThreads(KNumSocketThreads),
- _socketTransportModule(*UdpTransport::Create(
- VoEModuleId(instanceId, channelId), _numSocketThreads)),
-#endif
-#ifdef WEBRTC_SRTP
- _srtpModule(*SrtpModule::CreateSrtpModule(VoEModuleId(instanceId,
- channelId))),
-#endif
- _rtpDumpIn(*RtpDump::CreateRtpDump()),
- _rtpDumpOut(*RtpDump::CreateRtpDump()),
- _outputAudioLevel(),
- _externalTransport(false),
- _inputFilePlayerPtr(NULL),
- _outputFilePlayerPtr(NULL),
- _outputFileRecorderPtr(NULL),
- // Avoid conflict with other channels by adding 1024 - 1026,
- // won't use as much as 1024 channels.
- _inputFilePlayerId(VoEModuleId(instanceId, channelId) + 1024),
- _outputFilePlayerId(VoEModuleId(instanceId, channelId) + 1025),
- _outputFileRecorderId(VoEModuleId(instanceId, channelId) + 1026),
- _inputFilePlaying(false),
- _outputFilePlaying(false),
- _outputFileRecording(false),
- _inbandDtmfQueue(VoEModuleId(instanceId, channelId)),
- _inbandDtmfGenerator(VoEModuleId(instanceId, channelId)),
- _inputExternalMedia(false),
- _outputExternalMedia(false),
- _inputExternalMediaCallbackPtr(NULL),
- _outputExternalMediaCallbackPtr(NULL),
- _encryptionRTPBufferPtr(NULL),
- _decryptionRTPBufferPtr(NULL),
- _encryptionRTCPBufferPtr(NULL),
- _decryptionRTCPBufferPtr(NULL),
- _timeStamp(0), // This is just an offset, RTP module will add it's own random offset
- _sendTelephoneEventPayloadType(106),
- _playoutTimeStampRTP(0),
- _playoutTimeStampRTCP(0),
- _numberOfDiscardedPackets(0),
- _engineStatisticsPtr(NULL),
- _outputMixerPtr(NULL),
- _transmitMixerPtr(NULL),
- _moduleProcessThreadPtr(NULL),
- _audioDeviceModulePtr(NULL),
- _voiceEngineObserverPtr(NULL),
- _callbackCritSectPtr(NULL),
- _transportPtr(NULL),
- _encryptionPtr(NULL),
- _rtpAudioProc(NULL),
- _rxAudioProcessingModulePtr(NULL),
-#ifdef WEBRTC_DTMF_DETECTION
- _telephoneEventDetectionPtr(NULL),
-#endif
- _rxVadObserverPtr(NULL),
- _oldVadDecision(-1),
- _sendFrameType(0),
- _rtpObserverPtr(NULL),
- _rtcpObserverPtr(NULL),
- _outputIsOnHold(false),
- _externalPlayout(false),
- _inputIsOnHold(false),
- _playing(false),
- _sending(false),
- _receiving(false),
- _mixFileWithMicrophone(false),
- _rtpObserver(false),
- _rtcpObserver(false),
- _mute(false),
- _panLeft(1.0f),
- _panRight(1.0f),
- _outputGain(1.0f),
- _encrypting(false),
- _decrypting(false),
- _playOutbandDtmfEvent(false),
- _playInbandDtmfEvent(false),
- _inbandTelephoneEventDetection(false),
- _outOfBandTelephoneEventDetecion(false),
- _extraPayloadType(0),
- _insertExtraRTPPacket(false),
- _extraMarkerBit(false),
- _lastLocalTimeStamp(0),
- _lastPayloadType(0),
- _includeAudioLevelIndication(false),
- _rtpPacketTimedOut(false),
- _rtpPacketTimeOutIsEnabled(false),
- _rtpTimeOutSeconds(0),
- _connectionObserver(false),
- _connectionObserverPtr(NULL),
- _countAliveDetections(0),
- _countDeadDetections(0),
- _outputSpeechType(AudioFrame::kNormalSpeech),
- _averageDelayMs(0),
- _previousSequenceNumber(0),
- _previousTimestamp(0),
- _recPacketDelayMs(20),
- _RxVadDetection(false),
- _rxApmIsEnabled(false),
- _rxAgcIsEnabled(false),
- _rxNsIsEnabled(false)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::Channel() - ctor");
- _inbandDtmfQueue.ResetDtmf();
- _inbandDtmfGenerator.Init();
- _outputAudioLevel.Clear();
-
- // Create far end AudioProcessing Module
- _rxAudioProcessingModulePtr = AudioProcessing::Create(
- VoEModuleId(instanceId, channelId));
-}
-
-Channel::~Channel()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::~Channel() - dtor");
-
- if (_outputExternalMedia)
- {
- DeRegisterExternalMediaProcessing(kPlaybackPerChannel);
- }
- if (_inputExternalMedia)
- {
- DeRegisterExternalMediaProcessing(kRecordingPerChannel);
- }
- StopSend();
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- StopReceiving();
- // De-register packet callback to ensure we're not in a callback when
- // deleting channel state, avoids race condition and deadlock.
- if (_socketTransportModule.InitializeReceiveSockets(NULL, 0, NULL, NULL, 0)
- != 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "~Channel() failed to de-register receive callback");
- }
-#endif
- StopPlayout();
-
- {
- CriticalSectionScoped cs(_fileCritSect);
- if (_inputFilePlayerPtr)
- {
- _inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
- _inputFilePlayerPtr->StopPlayingFile();
- FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
- _inputFilePlayerPtr = NULL;
- }
- if (_outputFilePlayerPtr)
- {
- _outputFilePlayerPtr->RegisterModuleFileCallback(NULL);
- _outputFilePlayerPtr->StopPlayingFile();
- FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
- _outputFilePlayerPtr = NULL;
- }
- if (_outputFileRecorderPtr)
- {
- _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
- _outputFileRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
- _outputFileRecorderPtr = NULL;
- }
- }
-
- // The order to safely shutdown modules in a channel is:
- // 1. De-register callbacks in modules
- // 2. De-register modules in process thread
- // 3. Destroy modules
-
- // De-register all RTP module callbacks to ensure geting no callbacks
- // (Receive socket callback was de-registered above)
- if (_rtpRtcpModule.RegisterIncomingDataCallback(NULL) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "~Channel() failed to de-register incoming data callback"
- " (RTP module)");
- }
- if (_rtpRtcpModule.RegisterSendTransport(NULL) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "~Channel() failed to de-register send transport "
- "(RTP module)");
- }
- if (_rtpRtcpModule.RegisterIncomingRTPCallback(NULL) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "~Channel() failed to de-register incoming RTP"
- " callback (RTP module)");
- }
- if (_rtpRtcpModule.RegisterIncomingRTCPCallback(NULL) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "~Channel() failed to de-register incoming RTCP "
- "callback (RTP module)");
- }
- if (_rtpRtcpModule.RegisterAudioCallback(NULL) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "~Channel() failed to de-register audio callback "
- "(RTP module)");
- }
- if (_audioCodingModule.RegisterTransportCallback(NULL) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "~Channel() failed to de-register transport callback"
- " (Audio coding module)");
- }
- if (_audioCodingModule.RegisterVADCallback(NULL) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "~Channel() failed to de-register VAD callback"
- " (Audio coding module)");
- }
-#ifdef WEBRTC_DTMF_DETECTION
- if (_audioCodingModule.RegisterIncomingMessagesCallback(NULL) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "~Channel() failed to de-register incoming messages "
- "callback (Audio coding module)");
- }
-#endif
- // De-register modules in process thread
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (_moduleProcessThreadPtr->DeRegisterModule(&_socketTransportModule)
- == -1)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "~Channel() failed to deregister socket module");
- }
-#endif
- if (_moduleProcessThreadPtr->DeRegisterModule(&_rtpRtcpModule) == -1)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "~Channel() failed to deregister RTP/RTCP module");
- }
-
- // Destroy modules
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- UdpTransport::Destroy(
- &_socketTransportModule);
-#endif
- RtpRtcp::DestroyRtpRtcp(&_rtpRtcpModule);
- AudioCodingModule::Destroy(&_audioCodingModule);
-#ifdef WEBRTC_SRTP
- SrtpModule::DestroySrtpModule(&_srtpModule);
-#endif
- if (_rxAudioProcessingModulePtr != NULL)
- {
- AudioProcessing::Destroy(_rxAudioProcessingModulePtr); // far end APM
- _rxAudioProcessingModulePtr = NULL;
- }
-
- // End of modules shutdown
-
- // Delete other objects
- RtpDump::DestroyRtpDump(&_rtpDumpIn);
- RtpDump::DestroyRtpDump(&_rtpDumpOut);
- delete [] _encryptionRTPBufferPtr;
- delete [] _decryptionRTPBufferPtr;
- delete [] _encryptionRTCPBufferPtr;
- delete [] _decryptionRTCPBufferPtr;
- delete &_callbackCritSect;
- delete &_transmitCritSect;
- delete &_fileCritSect;
-}
-
-WebRtc_Word32
-Channel::Init()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::Init()");
-
- // --- Initial sanity
-
- if ((_engineStatisticsPtr == NULL) ||
- (_moduleProcessThreadPtr == NULL))
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::Init() must call SetEngineInformation() first");
- return -1;
- }
-
- // --- Add modules to process thread (for periodic schedulation)
-
- const bool processThreadFail =
- ((_moduleProcessThreadPtr->RegisterModule(&_rtpRtcpModule) != 0) ||
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- (_moduleProcessThreadPtr->RegisterModule(
- &_socketTransportModule) != 0));
-#else
- false);
-#endif
- if (processThreadFail)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CANNOT_INIT_CHANNEL, kTraceError,
- "Channel::Init() modules not registered");
- return -1;
- }
- // --- ACM initialization
-
- if ((_audioCodingModule.InitializeReceiver() == -1) ||
-#ifdef WEBRTC_CODEC_AVT
- // out-of-band Dtmf tones are played out by default
- (_audioCodingModule.SetDtmfPlayoutStatus(true) == -1) ||
-#endif
- (_audioCodingModule.InitializeSender() == -1))
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "Channel::Init() unable to initialize the ACM - 1");
- return -1;
- }
-
- // --- RTP/RTCP module initialization
-
- // Ensure that RTCP is enabled by default for the created channel.
- // Note that, the module will keep generating RTCP until it is explicitly
- // disabled by the user.
- // After StopListen (when no sockets exists), RTCP packets will no longer
- // be transmitted since the Transport object will then be invalid.
-
- const bool rtpRtcpFail =
- ((_rtpRtcpModule.InitReceiver() == -1) ||
- (_rtpRtcpModule.InitSender() == -1) ||
- (_rtpRtcpModule.SetTelephoneEventStatus(false, true, true) == -1) ||
- // RTCP is enabled by default
- (_rtpRtcpModule.SetRTCPStatus(kRtcpCompound) == -1));
- if (rtpRtcpFail)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "Channel::Init() RTP/RTCP module not initialized");
- return -1;
- }
-
- // --- Register all permanent callbacks
-
- const bool fail =
- (_rtpRtcpModule.RegisterIncomingDataCallback(this) == -1) ||
- (_rtpRtcpModule.RegisterIncomingRTPCallback(this) == -1) ||
- (_rtpRtcpModule.RegisterIncomingRTCPCallback(this) == -1) ||
- (_rtpRtcpModule.RegisterSendTransport(this) == -1) ||
- (_rtpRtcpModule.RegisterAudioCallback(this) == -1) ||
- (_audioCodingModule.RegisterTransportCallback(this) == -1) ||
- (_audioCodingModule.RegisterVADCallback(this) == -1);
-
- if (fail)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CANNOT_INIT_CHANNEL, kTraceError,
- "Channel::Init() callbacks not registered");
- return -1;
- }
-
- // --- Register all supported codecs to the receiving side of the
- // RTP/RTCP module
-
- CodecInst codec;
- const WebRtc_UWord8 nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
-
- for (int idx = 0; idx < nSupportedCodecs; idx++)
- {
- // Open up the RTP/RTCP receiver for all supported codecs
- if ((_audioCodingModule.Codec(idx, codec) == -1) ||
- (_rtpRtcpModule.RegisterReceivePayload(codec) == -1))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::Init() unable to register %s (%d/%d/%d/%d) "
- "to RTP/RTCP receiver",
- codec.plname, codec.pltype, codec.plfreq,
- codec.channels, codec.rate);
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::Init() %s (%d/%d/%d/%d) has been added to "
- "the RTP/RTCP receiver",
- codec.plname, codec.pltype, codec.plfreq,
- codec.channels, codec.rate);
- }
-
- // Ensure that PCMU is used as default codec on the sending side
- if (!STR_CASE_CMP(codec.plname, "PCMU"))
- {
- SetSendCodec(codec);
- }
-
- // Register default PT for outband 'telephone-event'
- if (!STR_CASE_CMP(codec.plname, "telephone-event"))
- {
- if ((_rtpRtcpModule.RegisterSendPayload(codec) == -1) ||
- (_audioCodingModule.RegisterReceiveCodec(codec) == -1))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::Init() failed to register outband "
- "'telephone-event' (%d/%d) correctly",
- codec.pltype, codec.plfreq);
- }
- }
-
- if (!STR_CASE_CMP(codec.plname, "CN"))
- {
- if ((_audioCodingModule.RegisterSendCodec(codec) == -1) ||
- (_audioCodingModule.RegisterReceiveCodec(codec) == -1) ||
- (_rtpRtcpModule.RegisterSendPayload(codec) == -1))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::Init() failed to register CN (%d/%d) "
- "correctly - 1",
- codec.pltype, codec.plfreq);
- }
- }
-#ifdef WEBRTC_CODEC_RED
- // Register RED to the receiving side of the ACM.
- // We will not receive an OnInitializeDecoder() callback for RED.
- if (!STR_CASE_CMP(codec.plname, "RED"))
- {
- if (_audioCodingModule.RegisterReceiveCodec(codec) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::Init() failed to register RED (%d/%d) "
- "correctly",
- codec.pltype, codec.plfreq);
- }
- }
-#endif
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- // Ensure that the WebRtcSocketTransport implementation is used as
- // Transport on the sending side
- {
- // A lock is needed here since users can call
- // RegisterExternalTransport() at the same time.
- CriticalSectionScoped cs(_callbackCritSect);
- _transportPtr = &_socketTransportModule;
- }
-#endif
-
- // Initialize the far end AP module
- // Using 8 kHz as initial Fs, the same as in transmission. Might be
- // changed at the first receiving audio.
- if (_rxAudioProcessingModulePtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_NO_MEMORY, kTraceCritical,
- "Channel::Init() failed to create the far-end AudioProcessing"
- " module");
- return -1;
- }
-
- if (_rxAudioProcessingModulePtr->set_sample_rate_hz(8000))
- {
- _engineStatisticsPtr->SetLastError(
- VE_APM_ERROR, kTraceWarning,
- "Channel::Init() failed to set the sample rate to 8K for"
- " far-end AP module");
- }
-
- if (_rxAudioProcessingModulePtr->set_num_channels(1, 1) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOUNDCARD_ERROR, kTraceWarning,
- "Init() failed to set channels for the primary audio stream");
- }
-
- if (_rxAudioProcessingModulePtr->high_pass_filter()->Enable(
- WEBRTC_VOICE_ENGINE_RX_HP_DEFAULT_STATE) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_APM_ERROR, kTraceWarning,
- "Channel::Init() failed to set the high-pass filter for"
- " far-end AP module");
- }
-
- if (_rxAudioProcessingModulePtr->noise_suppression()->set_level(
- (NoiseSuppression::Level)WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_APM_ERROR, kTraceWarning,
- "Init() failed to set noise reduction level for far-end"
- " AP module");
- }
- if (_rxAudioProcessingModulePtr->noise_suppression()->Enable(
- WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_STATE) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_APM_ERROR, kTraceWarning,
- "Init() failed to set noise reduction state for far-end"
- " AP module");
- }
-
- if (_rxAudioProcessingModulePtr->gain_control()->set_mode(
- (GainControl::Mode)WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_MODE) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_APM_ERROR, kTraceWarning,
- "Init() failed to set AGC mode for far-end AP module");
- }
- if (_rxAudioProcessingModulePtr->gain_control()->Enable(
- WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_STATE) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_APM_ERROR, kTraceWarning,
- "Init() failed to set AGC state for far-end AP module");
- }
-
- return 0;
-}
-
-WebRtc_Word32
-Channel::SetEngineInformation(Statistics& engineStatistics,
- OutputMixer& outputMixer,
- voe::TransmitMixer& transmitMixer,
- ProcessThread& moduleProcessThread,
- AudioDeviceModule& audioDeviceModule,
- VoiceEngineObserver* voiceEngineObserver,
- CriticalSectionWrapper* callbackCritSect)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetEngineInformation()");
- _engineStatisticsPtr = &engineStatistics;
- _outputMixerPtr = &outputMixer;
- _transmitMixerPtr = &transmitMixer,
- _moduleProcessThreadPtr = &moduleProcessThread;
- _audioDeviceModulePtr = &audioDeviceModule;
- _voiceEngineObserverPtr = voiceEngineObserver;
- _callbackCritSectPtr = callbackCritSect;
- return 0;
-}
-
-WebRtc_Word32
-Channel::UpdateLocalTimeStamp()
-{
-
- _timeStamp += _audioFrame._payloadDataLengthInSamples;
- return 0;
-}
-
-WebRtc_Word32
-Channel::StartPlayout()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::StartPlayout()");
- if (_playing)
- {
- return 0;
- }
- // Add participant as candidates for mixing.
- if (_outputMixerPtr->SetMixabilityStatus(*this, true) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
- "StartPlayout() failed to add participant to mixer");
- return -1;
- }
-
- _playing = true;
- return 0;
-}
-
-WebRtc_Word32
-Channel::StopPlayout()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::StopPlayout()");
- if (!_playing)
- {
- return 0;
- }
- // Remove participant as candidates for mixing
- if (_outputMixerPtr->SetMixabilityStatus(*this, false) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
- "StartPlayout() failed to remove participant from mixer");
- return -1;
- }
-
- _playing = false;
- _outputAudioLevel.Clear();
-
- return 0;
-}
-
-WebRtc_Word32
-Channel::StartSend()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::StartSend()");
- {
- // A lock is needed because |_sending| can be accessed or modified by
- // another thread at the same time.
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_sending)
- {
- return 0;
- }
- _sending = true;
- }
-
- if (_rtpRtcpModule.SetSendingStatus(true) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "StartSend() RTP/RTCP failed to start sending");
- CriticalSectionScoped cs(_callbackCritSect);
- _sending = false;
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32
-Channel::StopSend()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::StopSend()");
- {
- // A lock is needed because |_sending| can be accessed or modified by
- // another thread at the same time.
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (!_sending)
- {
- return 0;
- }
- _sending = false;
- }
-
- // Reset sending SSRC and sequence number and triggers direct transmission
- // of RTCP BYE
- if (_rtpRtcpModule.SetSendingStatus(false) == -1 ||
- _rtpRtcpModule.ResetSendDataCountersRTP() == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceWarning,
- "StartSend() RTP/RTCP failed to stop sending");
- }
-
- return 0;
-}
-
-WebRtc_Word32
-Channel::StartReceiving()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::StartReceiving()");
- if (_receiving)
- {
- return 0;
- }
- // If external transport is used, we will only initialize/set the variables
- // after this section, since we are not using the WebRtc transport but
- // still need to keep track of e.g. if we are receiving.
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_externalTransport)
- {
- if (!_socketTransportModule.ReceiveSocketsInitialized())
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKETS_NOT_INITED, kTraceError,
- "StartReceive() must set local receiver first");
- return -1;
- }
- if (_socketTransportModule.StartReceiving(KNumberOfSocketBuffers) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
- "StartReceiving() failed to start receiving");
- return -1;
- }
- }
-#endif
- _receiving = true;
- _numberOfDiscardedPackets = 0;
- return 0;
-}
-
-WebRtc_Word32
-Channel::StopReceiving()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::StopReceiving()");
- if (!_receiving)
- {
- return 0;
- }
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_externalTransport &&
- _socketTransportModule.ReceiveSocketsInitialized())
- {
- if (_socketTransportModule.StopReceiving() != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
- "StopReceiving() failed to stop receiving.");
- return -1;
- }
- }
-#endif
- bool dtmfDetection = _rtpRtcpModule.TelephoneEvent();
- WebRtc_Word32 ret = _rtpRtcpModule.InitReceiver();
- if (ret != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "StopReceiving() failed to reinitialize the RTP receiver.");
- return -1;
- }
- // Recover DTMF detection status.
- ret = _rtpRtcpModule.SetTelephoneEventStatus(dtmfDetection, true, true);
- if (ret != 0) {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "StopReceiving() failed to restore telephone-event status.");
- }
- RegisterReceiveCodecsToRTPModule();
- _receiving = false;
- return 0;
-}
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
-WebRtc_Word32
-Channel::SetLocalReceiver(const WebRtc_UWord16 rtpPort,
- const WebRtc_UWord16 rtcpPort,
- const WebRtc_Word8 ipAddr[64],
- const WebRtc_Word8 multicastIpAddr[64])
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetLocalReceiver()");
-
- if (_externalTransport)
- {
- _engineStatisticsPtr->SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "SetLocalReceiver() conflict with external transport");
- return -1;
- }
-
- if (_sending)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_SENDING, kTraceError,
- "SetLocalReceiver() already sending");
- return -1;
- }
- if (_receiving)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_LISTENING, kTraceError,
- "SetLocalReceiver() already receiving");
- return -1;
- }
-
- if (_socketTransportModule.InitializeReceiveSockets(this,
- rtpPort,
- ipAddr,
- multicastIpAddr,
- rtcpPort) != 0)
- {
- UdpTransport::ErrorCode lastSockError(
- _socketTransportModule.LastError());
- switch (lastSockError)
- {
- case UdpTransport::kIpAddressInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_IP_ADDRESS, kTraceError,
- "SetLocalReceiver() invalid IP address");
- break;
- case UdpTransport::kSocketInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_ERROR, kTraceError,
- "SetLocalReceiver() invalid socket");
- break;
- case UdpTransport::kPortInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_PORT_NMBR, kTraceError,
- "SetLocalReceiver() invalid port");
- break;
- case UdpTransport::kFailedToBindPort:
- _engineStatisticsPtr->SetLastError(
- VE_BINDING_SOCKET_TO_LOCAL_ADDRESS_FAILED, kTraceError,
- "SetLocalReceiver() binding failed");
- break;
- default:
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_ERROR, kTraceError,
- "SetLocalReceiver() undefined socket error");
- break;
- }
- return -1;
- }
- return 0;
-}
-#endif
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
-WebRtc_Word32
-Channel::GetLocalReceiver(int& port, int& RTCPport, char ipAddr[64])
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetLocalReceiver()");
-
- if (_externalTransport)
- {
- _engineStatisticsPtr->SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "SetLocalReceiver() conflict with external transport");
- return -1;
- }
-
- WebRtc_Word8 ipAddrTmp[UdpTransport::
- kIpAddressVersion6Length] = {0};
- WebRtc_UWord16 rtpPort(0);
- WebRtc_UWord16 rtcpPort(0);
- WebRtc_Word8 multicastIpAddr[UdpTransport::
- kIpAddressVersion6Length] = {0};
-
- // Acquire socket information from the socket module
- if (_socketTransportModule.ReceiveSocketInformation(ipAddrTmp,
- rtpPort,
- rtcpPort,
- multicastIpAddr) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CANNOT_GET_SOCKET_INFO, kTraceError,
- "GetLocalReceiver() unable to retrieve socket information");
- return -1;
- }
-
- // Deliver valid results to the user
- port = static_cast<int> (rtpPort);
- RTCPport = static_cast<int> (rtcpPort);
- if (ipAddr != NULL)
- {
- strcpy(ipAddr, ipAddrTmp);
- }
- return 0;
-}
-#endif
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
-WebRtc_Word32
-Channel::SetSendDestination(const WebRtc_UWord16 rtpPort,
- const WebRtc_Word8 ipAddr[64],
- const int sourcePort,
- const WebRtc_UWord16 rtcpPort)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetSendDestination()");
-
- if (_externalTransport)
- {
- _engineStatisticsPtr->SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "SetSendDestination() conflict with external transport");
- return -1;
- }
-
- // Initialize ports and IP address for the remote (destination) side.
- // By default, the sockets used for receiving are used for transmission as
- // well, hence the source ports for outgoing packets are the same as the
- // receiving ports specified in SetLocalReceiver.
- // If an extra send socket has been created, it will be utilized until a
- // new source port is specified or until the channel has been deleted and
- // recreated. If no socket exists, sockets will be created when the first
- // RTP and RTCP packets shall be transmitted (see e.g.
- // UdpTransportImpl::SendPacket()).
- //
- // NOTE: this function does not require that sockets exists; all it does is
- // to build send structures to be used with the sockets when they exist.
- // It is therefore possible to call this method before SetLocalReceiver.
- // However, sockets must exist if a multi-cast address is given as input.
-
- // Build send structures and enable QoS (if enabled and supported)
- if (_socketTransportModule.InitializeSendSockets(
- ipAddr, rtpPort, rtcpPort) != UdpTransport::kNoSocketError)
- {
- UdpTransport::ErrorCode lastSockError(
- _socketTransportModule.LastError());
- switch (lastSockError)
- {
- case UdpTransport::kIpAddressInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_IP_ADDRESS, kTraceError,
- "SetSendDestination() invalid IP address 1");
- break;
- case UdpTransport::kSocketInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_ERROR, kTraceError,
- "SetSendDestination() invalid socket 1");
- break;
- case UdpTransport::kQosError:
- _engineStatisticsPtr->SetLastError(
- VE_GQOS_ERROR, kTraceError,
- "SetSendDestination() failed to set QoS");
- break;
- case UdpTransport::kMulticastAddressInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_MULTICAST_ADDRESS, kTraceError,
- "SetSendDestination() invalid multicast address");
- break;
- default:
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_ERROR, kTraceError,
- "SetSendDestination() undefined socket error 1");
- break;
- }
- return -1;
- }
-
- // Check if the user has specified a non-default source port different from
- // the local receive port.
- // If so, an extra local socket will be created unless the source port is
- // not unique.
- if (sourcePort != kVoEDefault)
- {
- WebRtc_UWord16 receiverRtpPort(0);
- WebRtc_UWord16 rtcpNA(0);
- if (_socketTransportModule.ReceiveSocketInformation(NULL,
- receiverRtpPort,
- rtcpNA,
- NULL) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CANNOT_GET_SOCKET_INFO, kTraceError,
- "SetSendDestination() failed to retrieve socket information");
- return -1;
- }
-
- WebRtc_UWord16 sourcePortUW16 =
- static_cast<WebRtc_UWord16> (sourcePort);
-
- // An extra socket will only be created if the specified source port
- // differs from the local receive port.
- if (sourcePortUW16 != receiverRtpPort)
- {
- // Initialize extra local socket to get a different source port
- // than the local
- // receiver port. Always use default source for RTCP.
- // Note that, this calls UdpTransport::CloseSendSockets().
- if (_socketTransportModule.InitializeSourcePorts(
- sourcePortUW16,
- sourcePortUW16+1) != 0)
- {
- UdpTransport::ErrorCode lastSockError(
- _socketTransportModule.LastError());
- switch (lastSockError)
- {
- case UdpTransport::kIpAddressInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_IP_ADDRESS, kTraceError,
- "SetSendDestination() invalid IP address 2");
- break;
- case UdpTransport::kSocketInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_ERROR, kTraceError,
- "SetSendDestination() invalid socket 2");
- break;
- default:
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_ERROR, kTraceError,
- "SetSendDestination() undefined socket error 2");
- break;
- }
- return -1;
- }
- WEBRTC_TRACE(kTraceInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "SetSendDestination() extra local socket is created"
- " to facilitate unique source port");
- }
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "SetSendDestination() sourcePort equals the local"
- " receive port => no extra socket is created");
- }
- }
-
- return 0;
-}
-#endif
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
-WebRtc_Word32
-Channel::GetSendDestination(int& port,
- char ipAddr[64],
- int& sourcePort,
- int& RTCPport)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetSendDestination()");
-
- if (_externalTransport)
- {
- _engineStatisticsPtr->SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "GetSendDestination() conflict with external transport");
- return -1;
- }
-
- WebRtc_Word8 ipAddrTmp[UdpTransport::kIpAddressVersion6Length] = {0};
- WebRtc_UWord16 rtpPort(0);
- WebRtc_UWord16 rtcpPort(0);
- WebRtc_UWord16 rtpSourcePort(0);
- WebRtc_UWord16 rtcpSourcePort(0);
-
- // Acquire sending socket information from the socket module
- _socketTransportModule.SendSocketInformation(ipAddrTmp, rtpPort, rtcpPort);
- _socketTransportModule.SourcePorts(rtpSourcePort, rtcpSourcePort);
-
- // Deliver valid results to the user
- port = static_cast<int> (rtpPort);
- RTCPport = static_cast<int> (rtcpPort);
- sourcePort = static_cast<int> (rtpSourcePort);
- if (ipAddr != NULL)
- {
- strcpy(ipAddr, ipAddrTmp);
- }
-
- return 0;
-}
-#endif
-
-
-WebRtc_Word32
-Channel::SetNetEQPlayoutMode(NetEqModes mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetNetEQPlayoutMode()");
- AudioPlayoutMode playoutMode(voice);
- switch (mode)
- {
- case kNetEqDefault:
- playoutMode = voice;
- break;
- case kNetEqStreaming:
- playoutMode = streaming;
- break;
- case kNetEqFax:
- playoutMode = fax;
- break;
- }
- if (_audioCodingModule.SetPlayoutMode(playoutMode) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "SetNetEQPlayoutMode() failed to set playout mode");
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32
-Channel::GetNetEQPlayoutMode(NetEqModes& mode)
-{
- const AudioPlayoutMode playoutMode = _audioCodingModule.PlayoutMode();
- switch (playoutMode)
- {
- case voice:
- mode = kNetEqDefault;
- break;
- case streaming:
- mode = kNetEqStreaming;
- break;
- case fax:
- mode = kNetEqFax;
- break;
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "Channel::GetNetEQPlayoutMode() => mode=%u", mode);
- return 0;
-}
-
-WebRtc_Word32
-Channel::SetNetEQBGNMode(NetEqBgnModes mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetNetEQPlayoutMode()");
- ACMBackgroundNoiseMode noiseMode(On);
- switch (mode)
- {
- case kBgnOn:
- noiseMode = On;
- break;
- case kBgnFade:
- noiseMode = Fade;
- break;
- case kBgnOff:
- noiseMode = Off;
- break;
- }
- if (_audioCodingModule.SetBackgroundNoiseMode(noiseMode) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "SetBackgroundNoiseMode() failed to set noise mode");
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32
-Channel::SetOnHoldStatus(bool enable, OnHoldModes mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetOnHoldStatus()");
- if (mode == kHoldSendAndPlay)
- {
- _outputIsOnHold = enable;
- _inputIsOnHold = enable;
- }
- else if (mode == kHoldPlayOnly)
- {
- _outputIsOnHold = enable;
- }
- if (mode == kHoldSendOnly)
- {
- _inputIsOnHold = enable;
- }
- return 0;
-}
-
-WebRtc_Word32
-Channel::GetOnHoldStatus(bool& enabled, OnHoldModes& mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetOnHoldStatus()");
- enabled = (_outputIsOnHold || _inputIsOnHold);
- if (_outputIsOnHold && _inputIsOnHold)
- {
- mode = kHoldSendAndPlay;
- }
- else if (_outputIsOnHold && !_inputIsOnHold)
- {
- mode = kHoldPlayOnly;
- }
- else if (!_outputIsOnHold && _inputIsOnHold)
- {
- mode = kHoldSendOnly;
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetOnHoldStatus() => enabled=%d, mode=%d",
- enabled, mode);
- return 0;
-}
-
-WebRtc_Word32
-Channel::RegisterVoiceEngineObserver(VoiceEngineObserver& observer)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::RegisterVoiceEngineObserver()");
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_voiceEngineObserverPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "RegisterVoiceEngineObserver() observer already enabled");
- return -1;
- }
- _voiceEngineObserverPtr = &observer;
- return 0;
-}
-
-WebRtc_Word32
-Channel::DeRegisterVoiceEngineObserver()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::DeRegisterVoiceEngineObserver()");
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (!_voiceEngineObserverPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "DeRegisterVoiceEngineObserver() observer already disabled");
- return 0;
- }
- _voiceEngineObserverPtr = NULL;
- return 0;
-}
-
-WebRtc_Word32
-Channel::GetNetEQBGNMode(NetEqBgnModes& mode)
-{
- ACMBackgroundNoiseMode noiseMode(On);
- _audioCodingModule.BackgroundNoiseMode(noiseMode);
- switch (noiseMode)
- {
- case On:
- mode = kBgnOn;
- break;
- case Fade:
- mode = kBgnFade;
- break;
- case Off:
- mode = kBgnOff;
- break;
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetNetEQBGNMode() => mode=%u", mode);
- return 0;
-}
-
-WebRtc_Word32
-Channel::GetSendCodec(CodecInst& codec)
-{
- return (_audioCodingModule.SendCodec(codec));
-}
-
-WebRtc_Word32
-Channel::GetRecCodec(CodecInst& codec)
-{
- return (_audioCodingModule.ReceiveCodec(codec));
-}
-
-WebRtc_Word32
-Channel::SetSendCodec(const CodecInst& codec)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetSendCodec()");
-
- if (_audioCodingModule.RegisterSendCodec(codec) != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
- "SetSendCodec() failed to register codec to ACM");
- return -1;
- }
-
- if (_rtpRtcpModule.RegisterSendPayload(codec) != 0)
- {
- _rtpRtcpModule.DeRegisterSendPayload(codec.pltype);
- if (_rtpRtcpModule.RegisterSendPayload(codec) != 0)
- {
- WEBRTC_TRACE(
- kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
- "SetSendCodec() failed to register codec to"
- " RTP/RTCP module");
- return -1;
- }
- }
-
- if (_rtpRtcpModule.SetAudioPacketSize(codec.pacsize) != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
- "SetSendCodec() failed to set audio packet size");
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32
-Channel::SetVADStatus(bool enableVAD, ACMVADMode mode, bool disableDTX)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetVADStatus(mode=%d)", mode);
- // To disable VAD, DTX must be disabled too
- disableDTX = ((enableVAD == false) ? true : disableDTX);
- if (_audioCodingModule.SetVAD(!disableDTX, enableVAD, mode) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "SetVADStatus() failed to set VAD");
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32
-Channel::GetVADStatus(bool& enabledVAD, ACMVADMode& mode, bool& disabledDTX)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetVADStatus");
- if (_audioCodingModule.VAD(disabledDTX, enabledVAD, mode) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "GetVADStatus() failed to get VAD status");
- return -1;
- }
- disabledDTX = !disabledDTX;
- return 0;
-}
-
-WebRtc_Word32
-Channel::SetRecPayloadType(const CodecInst& codec)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetRecPayloadType()");
-
- if (_playing)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceError,
- "SetRecPayloadType() unable to set PT while playing");
- return -1;
- }
- if (_receiving)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_LISTENING, kTraceError,
- "SetRecPayloadType() unable to set PT while listening");
- return -1;
- }
-
- if (codec.pltype == -1)
- {
- // De-register the selected codec (RTP/RTCP module and ACM)
-
- WebRtc_Word8 pltype(-1);
- CodecInst rxCodec = codec;
-
- // Get payload type for the given codec
- _rtpRtcpModule.ReceivePayloadType(rxCodec, &pltype);
- rxCodec.pltype = pltype;
-
- if (_rtpRtcpModule.DeRegisterReceivePayload(pltype) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR,
- kTraceError,
- "SetRecPayloadType() RTP/RTCP-module deregistration "
- "failed");
- return -1;
- }
- if (_audioCodingModule.UnregisterReceiveCodec(rxCodec.pltype) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "SetRecPayloadType() ACM deregistration failed - 1");
- return -1;
- }
- return 0;
- }
-
- if (_rtpRtcpModule.RegisterReceivePayload(codec) != 0)
- {
- // First attempt to register failed => de-register and try again
- _rtpRtcpModule.DeRegisterReceivePayload(codec.pltype);
- if (_rtpRtcpModule.RegisterReceivePayload(codec) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "SetRecPayloadType() RTP/RTCP-module registration failed");
- return -1;
- }
- }
- if (_audioCodingModule.RegisterReceiveCodec(codec) != 0)
- {
- _audioCodingModule.UnregisterReceiveCodec(codec.pltype);
- if (_audioCodingModule.RegisterReceiveCodec(codec) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "SetRecPayloadType() ACM registration failed - 1");
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_Word32
-Channel::GetRecPayloadType(CodecInst& codec)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetRecPayloadType()");
- WebRtc_Word8 payloadType(-1);
- if (_rtpRtcpModule.ReceivePayloadType(codec, &payloadType) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "GetRecPayloadType() failed to retrieve RX payload type");
- return -1;
- }
- codec.pltype = payloadType;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetRecPayloadType() => pltype=%u", codec.pltype);
- return 0;
-}
-
-WebRtc_Word32
-Channel::SetAMREncFormat(AmrMode mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetAMREncFormat()");
-
- // ACM doesn't support AMR
- return -1;
-}
-
-WebRtc_Word32
-Channel::SetAMRDecFormat(AmrMode mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetAMRDecFormat()");
-
- // ACM doesn't support AMR
- return -1;
-}
-
-WebRtc_Word32
-Channel::SetAMRWbEncFormat(AmrMode mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetAMRWbEncFormat()");
-
- // ACM doesn't support AMR
- return -1;
-
-}
-
-WebRtc_Word32
-Channel::SetAMRWbDecFormat(AmrMode mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetAMRWbDecFormat()");
-
- // ACM doesn't support AMR
- return -1;
-}
-
-WebRtc_Word32
-Channel::SetSendCNPayloadType(int type, PayloadFrequencies frequency)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetSendCNPayloadType()");
-
- CodecInst codec;
- WebRtc_Word32 samplingFreqHz(-1);
- if (frequency == kFreq32000Hz)
- samplingFreqHz = 32000;
- else if (frequency == kFreq16000Hz)
- samplingFreqHz = 16000;
-
- if (_audioCodingModule.Codec("CN", codec, samplingFreqHz) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "SetSendCNPayloadType() failed to retrieve default CN codec "
- "settings");
- return -1;
- }
-
- // Modify the payload type (must be set to dynamic range)
- codec.pltype = type;
-
- if (_audioCodingModule.RegisterSendCodec(codec) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "SetSendCNPayloadType() failed to register CN to ACM");
- return -1;
- }
-
- if (_rtpRtcpModule.RegisterSendPayload(codec) != 0)
- {
- _rtpRtcpModule.DeRegisterSendPayload(codec.pltype);
- if (_rtpRtcpModule.RegisterSendPayload(codec) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "SetSendCNPayloadType() failed to register CN to RTP/RTCP "
- "module");
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_Word32
-Channel::SetISACInitTargetRate(int rateBps, bool useFixedFrameSize)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetISACInitTargetRate()");
-
- CodecInst sendCodec;
- if (_audioCodingModule.SendCodec(sendCodec) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CODEC_ERROR, kTraceError,
- "SetISACInitTargetRate() failed to retrieve send codec");
- return -1;
- }
- if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0)
- {
- // This API is only valid if iSAC is setup to run in channel-adaptive
- // mode.
- // We do not validate the adaptive mode here. It is done later in the
- // ConfigISACBandwidthEstimator() API.
- _engineStatisticsPtr->SetLastError(
- VE_CODEC_ERROR, kTraceError,
- "SetISACInitTargetRate() send codec is not iSAC");
- return -1;
- }
-
- WebRtc_UWord8 initFrameSizeMsec(0);
- if (16000 == sendCodec.plfreq)
- {
- // Note that 0 is a valid and corresponds to "use default
- if ((rateBps != 0 &&
- rateBps < kVoiceEngineMinIsacInitTargetRateBpsWb) ||
- (rateBps > kVoiceEngineMaxIsacInitTargetRateBpsWb))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetISACInitTargetRate() invalid target rate - 1");
- return -1;
- }
- // 30 or 60ms
- initFrameSizeMsec = (WebRtc_UWord8)(sendCodec.pacsize / 16);
- }
- else if (32000 == sendCodec.plfreq)
- {
- if ((rateBps != 0 &&
- rateBps < kVoiceEngineMinIsacInitTargetRateBpsSwb) ||
- (rateBps > kVoiceEngineMaxIsacInitTargetRateBpsSwb))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetISACInitTargetRate() invalid target rate - 2");
- return -1;
- }
- initFrameSizeMsec = (WebRtc_UWord8)(sendCodec.pacsize / 32); // 30ms
- }
-
- if (_audioCodingModule.ConfigISACBandwidthEstimator(
- initFrameSizeMsec, rateBps, useFixedFrameSize) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "SetISACInitTargetRate() iSAC BWE config failed");
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32
-Channel::SetISACMaxRate(int rateBps)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetISACMaxRate()");
-
- CodecInst sendCodec;
- if (_audioCodingModule.SendCodec(sendCodec) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CODEC_ERROR, kTraceError,
- "SetISACMaxRate() failed to retrieve send codec");
- return -1;
- }
- if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0)
- {
- // This API is only valid if iSAC is selected as sending codec.
- _engineStatisticsPtr->SetLastError(
- VE_CODEC_ERROR, kTraceError,
- "SetISACMaxRate() send codec is not iSAC");
- return -1;
- }
- if (16000 == sendCodec.plfreq)
- {
- if ((rateBps < kVoiceEngineMinIsacMaxRateBpsWb) ||
- (rateBps > kVoiceEngineMaxIsacMaxRateBpsWb))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetISACMaxRate() invalid max rate - 1");
- return -1;
- }
- }
- else if (32000 == sendCodec.plfreq)
- {
- if ((rateBps < kVoiceEngineMinIsacMaxRateBpsSwb) ||
- (rateBps > kVoiceEngineMaxIsacMaxRateBpsSwb))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetISACMaxRate() invalid max rate - 2");
- return -1;
- }
- }
- if (_sending)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SENDING, kTraceError,
- "SetISACMaxRate() unable to set max rate while sending");
- return -1;
- }
-
- // Set the maximum instantaneous rate of iSAC (works for both adaptive
- // and non-adaptive mode)
- if (_audioCodingModule.SetISACMaxRate(rateBps) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "SetISACMaxRate() failed to set max rate");
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32
-Channel::SetISACMaxPayloadSize(int sizeBytes)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetISACMaxPayloadSize()");
- CodecInst sendCodec;
- if (_audioCodingModule.SendCodec(sendCodec) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CODEC_ERROR, kTraceError,
- "SetISACMaxPayloadSize() failed to retrieve send codec");
- return -1;
- }
- if (STR_CASE_CMP(sendCodec.plname, "ISAC") != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CODEC_ERROR, kTraceError,
- "SetISACMaxPayloadSize() send codec is not iSAC");
- return -1;
- }
- if (16000 == sendCodec.plfreq)
- {
- if ((sizeBytes < kVoiceEngineMinIsacMaxPayloadSizeBytesWb) ||
- (sizeBytes > kVoiceEngineMaxIsacMaxPayloadSizeBytesWb))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetISACMaxPayloadSize() invalid max payload - 1");
- return -1;
- }
- }
- else if (32000 == sendCodec.plfreq)
- {
- if ((sizeBytes < kVoiceEngineMinIsacMaxPayloadSizeBytesSwb) ||
- (sizeBytes > kVoiceEngineMaxIsacMaxPayloadSizeBytesSwb))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetISACMaxPayloadSize() invalid max payload - 2");
- return -1;
- }
- }
- if (_sending)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SENDING, kTraceError,
- "SetISACMaxPayloadSize() unable to set max rate while sending");
- return -1;
- }
-
- if (_audioCodingModule.SetISACMaxPayloadSize(sizeBytes) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "SetISACMaxPayloadSize() failed to set max payload size");
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32 Channel::RegisterExternalTransport(Transport& transport)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::RegisterExternalTransport()");
-
- CriticalSectionScoped cs(_callbackCritSect);
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- // Sanity checks for default (non external transport) to avoid conflict with
- // WebRtc sockets.
- if (_socketTransportModule.SendSocketsInitialized())
- {
- _engineStatisticsPtr->SetLastError(VE_SEND_SOCKETS_CONFLICT,
- kTraceError,
- "RegisterExternalTransport() send sockets already initialized");
- return -1;
- }
- if (_socketTransportModule.ReceiveSocketsInitialized())
- {
- _engineStatisticsPtr->SetLastError(VE_RECEIVE_SOCKETS_CONFLICT,
- kTraceError,
- "RegisterExternalTransport() receive sockets already initialized");
- return -1;
- }
-#endif
- if (_externalTransport)
- {
- _engineStatisticsPtr->SetLastError(VE_INVALID_OPERATION,
- kTraceError,
- "RegisterExternalTransport() external transport already enabled");
- return -1;
- }
- _externalTransport = true;
- _transportPtr = &transport;
- return 0;
-}
-
-WebRtc_Word32
-Channel::DeRegisterExternalTransport()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::DeRegisterExternalTransport()");
-
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (!_transportPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "DeRegisterExternalTransport() external transport already "
- "disabled");
- return 0;
- }
- _externalTransport = false;
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
- _transportPtr = NULL;
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "DeRegisterExternalTransport() all transport is disabled");
-#else
- _transportPtr = &_socketTransportModule;
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "DeRegisterExternalTransport() internal Transport is enabled");
-#endif
- return 0;
-}
-
-WebRtc_Word32
-Channel::ReceivedRTPPacket(const WebRtc_Word8* data, WebRtc_Word32 length)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::ReceivedRTPPacket()");
- const WebRtc_Word8 dummyIP[] = "127.0.0.1";
- IncomingRTPPacket(data, length, dummyIP, 0);
- return 0;
-}
-
-WebRtc_Word32
-Channel::ReceivedRTCPPacket(const WebRtc_Word8* data, WebRtc_Word32 length)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::ReceivedRTCPPacket()");
- const WebRtc_Word8 dummyIP[] = "127.0.0.1";
- IncomingRTCPPacket(data, length, dummyIP, 0);
- return 0;
-}
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
-WebRtc_Word32
-Channel::GetSourceInfo(int& rtpPort, int& rtcpPort, char ipAddr[64])
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetSourceInfo()");
-
- WebRtc_UWord16 rtpPortModule;
- WebRtc_UWord16 rtcpPortModule;
- WebRtc_Word8 ipaddr[UdpTransport::kIpAddressVersion6Length] = {0};
-
- if (_socketTransportModule.RemoteSocketInformation(ipaddr,
- rtpPortModule,
- rtcpPortModule) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
- "GetSourceInfo() failed to retrieve remote socket information");
- return -1;
- }
- strcpy(ipAddr, ipaddr);
- rtpPort = rtpPortModule;
- rtcpPort = rtcpPortModule;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "GetSourceInfo() => rtpPort=%d, rtcpPort=%d, ipAddr=%s",
- rtpPort, rtcpPort, ipAddr);
- return 0;
-}
-
-WebRtc_Word32
-Channel::EnableIPv6()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::EnableIPv6()");
- if (_socketTransportModule.ReceiveSocketsInitialized() ||
- _socketTransportModule.SendSocketsInitialized())
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "EnableIPv6() socket layer is already initialized");
- return -1;
- }
- if (_socketTransportModule.EnableIpV6() != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_ERROR, kTraceError,
- "EnableIPv6() failed to enable IPv6");
- const UdpTransport::ErrorCode lastError =
- _socketTransportModule.LastError();
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "UdpTransport::LastError() => %d", lastError);
- return -1;
- }
- return 0;
-}
-
-bool
-Channel::IPv6IsEnabled() const
-{
- bool isEnabled = _socketTransportModule.IpV6Enabled();
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "IPv6IsEnabled() => %d", isEnabled);
- return isEnabled;
-}
-
-WebRtc_Word32
-Channel::SetSourceFilter(int rtpPort, int rtcpPort, const char ipAddr[64])
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetSourceFilter()");
- if (_socketTransportModule.SetFilterPorts(
- static_cast<WebRtc_UWord16>(rtpPort),
- static_cast<WebRtc_UWord16>(rtcpPort)) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
- "SetSourceFilter() failed to set filter ports");
- const UdpTransport::ErrorCode lastError =
- _socketTransportModule.LastError();
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "UdpTransport::LastError() => %d",
- lastError);
- return -1;
- }
- const WebRtc_Word8* filterIpAddress =
- static_cast<const WebRtc_Word8*> (ipAddr);
- if (_socketTransportModule.SetFilterIP(filterIpAddress) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_IP_ADDRESS, kTraceError,
- "SetSourceFilter() failed to set filter IP address");
- const UdpTransport::ErrorCode lastError =
- _socketTransportModule.LastError();
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "UdpTransport::LastError() => %d", lastError);
- return -1;
- }
- return 0;
-}
-
-WebRtc_Word32
-Channel::GetSourceFilter(int& rtpPort, int& rtcpPort, char ipAddr[64])
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetSourceFilter()");
- WebRtc_UWord16 rtpFilterPort(0);
- WebRtc_UWord16 rtcpFilterPort(0);
- if (_socketTransportModule.FilterPorts(rtpFilterPort, rtcpFilterPort) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceWarning,
- "GetSourceFilter() failed to retrieve filter ports");
- }
- WebRtc_Word8 ipAddrTmp[UdpTransport::kIpAddressVersion6Length] = {0};
- if (_socketTransportModule.FilterIP(ipAddrTmp) != 0)
- {
- // no filter has been configured (not seen as an error)
- memset(ipAddrTmp,
- 0, UdpTransport::kIpAddressVersion6Length);
- }
- rtpPort = static_cast<int> (rtpFilterPort);
- rtcpPort = static_cast<int> (rtcpFilterPort);
- strcpy(ipAddr, ipAddrTmp);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "GetSourceFilter() => rtpPort=%d, rtcpPort=%d, ipAddr=%s",
- rtpPort, rtcpPort, ipAddr);
- return 0;
-}
-
-WebRtc_Word32
-Channel::SetSendTOS(int DSCP, int priority, bool useSetSockopt)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetSendTOS(DSCP=%d, useSetSockopt=%d)",
- DSCP, (int)useSetSockopt);
-
- // Set TOS value and possibly try to force usage of setsockopt()
- if (_socketTransportModule.SetToS(DSCP, useSetSockopt) != 0)
- {
- UdpTransport::ErrorCode lastSockError(
- _socketTransportModule.LastError());
- switch (lastSockError)
- {
- case UdpTransport::kTosError:
- _engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError,
- "SetSendTOS() TOS error");
- break;
- case UdpTransport::kQosError:
- _engineStatisticsPtr->SetLastError(
- VE_TOS_GQOS_CONFLICT, kTraceError,
- "SetSendTOS() GQOS error");
- break;
- case UdpTransport::kTosInvalid:
- // can't switch SetSockOpt method without disabling TOS first, or
- // SetSockopt() call failed
- _engineStatisticsPtr->SetLastError(VE_TOS_INVALID, kTraceError,
- "SetSendTOS() invalid TOS");
- break;
- case UdpTransport::kSocketInvalid:
- _engineStatisticsPtr->SetLastError(VE_SOCKET_ERROR, kTraceError,
- "SetSendTOS() invalid Socket");
- break;
- default:
- _engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError,
- "SetSendTOS() TOS error");
- break;
- }
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
- "UdpTransport => lastError = %d",
- lastSockError);
- return -1;
- }
-
- // Set priority (PCP) value, -1 means don't change
- if (-1 != priority)
- {
- if (_socketTransportModule.SetPCP(priority) != 0)
- {
- UdpTransport::ErrorCode lastSockError(
- _socketTransportModule.LastError());
- switch (lastSockError)
- {
- case UdpTransport::kPcpError:
- _engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError,
- "SetSendTOS() PCP error");
- break;
- case UdpTransport::kQosError:
- _engineStatisticsPtr->SetLastError(
- VE_TOS_GQOS_CONFLICT, kTraceError,
- "SetSendTOS() GQOS conflict");
- break;
- case UdpTransport::kSocketInvalid:
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_ERROR, kTraceError,
- "SetSendTOS() invalid Socket");
- break;
- default:
- _engineStatisticsPtr->SetLastError(VE_TOS_ERROR, kTraceError,
- "SetSendTOS() PCP error");
- break;
- }
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "UdpTransport => lastError = %d",
- lastSockError);
- return -1;
- }
- }
-
- return 0;
-}
-
-WebRtc_Word32
-Channel::GetSendTOS(int &DSCP, int& priority, bool &useSetSockopt)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetSendTOS(DSCP=?, useSetSockopt=?)");
- WebRtc_Word32 dscp(0), prio(0);
- bool setSockopt(false);
- if (_socketTransportModule.ToS(dscp, setSockopt) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
- "GetSendTOS() failed to get TOS info");
- return -1;
- }
- if (_socketTransportModule.PCP(prio) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
- "GetSendTOS() failed to get PCP info");
- return -1;
- }
- DSCP = static_cast<int> (dscp);
- priority = static_cast<int> (prio);
- useSetSockopt = setSockopt;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetSendTOS() => DSCP=%d, priority=%d, useSetSockopt=%d",
- DSCP, priority, (int)useSetSockopt);
- return 0;
-}
-
-#if defined(_WIN32)
-WebRtc_Word32
-Channel::SetSendGQoS(bool enable, int serviceType, int overrideDSCP)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetSendGQoS(enable=%d, serviceType=%d, "
- "overrideDSCP=%d)",
- (int)enable, serviceType, overrideDSCP);
- if(!_socketTransportModule.ReceiveSocketsInitialized())
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKETS_NOT_INITED, kTraceError,
- "SetSendGQoS() GQoS state must be set after sockets are created");
- return -1;
- }
- if(!_socketTransportModule.SendSocketsInitialized())
- {
- _engineStatisticsPtr->SetLastError(
- VE_DESTINATION_NOT_INITED, kTraceError,
- "SetSendGQoS() GQoS state must be set after sending side is "
- "initialized");
- return -1;
- }
- if (enable &&
- (serviceType != SERVICETYPE_BESTEFFORT) &&
- (serviceType != SERVICETYPE_CONTROLLEDLOAD) &&
- (serviceType != SERVICETYPE_GUARANTEED) &&
- (serviceType != SERVICETYPE_QUALITATIVE))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetSendGQoS() Invalid service type");
- return -1;
- }
- if (enable && ((overrideDSCP < 0) || (overrideDSCP > 63)))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetSendGQoS() Invalid overrideDSCP value");
- return -1;
- }
-
- // Avoid GQoS/ToS conflict when user wants to override the default DSCP
- // mapping
- bool QoS(false);
- WebRtc_Word32 sType(0);
- WebRtc_Word32 ovrDSCP(0);
- if (_socketTransportModule.QoS(QoS, sType, ovrDSCP))
- {
- _engineStatisticsPtr->SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
- "SetSendGQoS() failed to get QOS info");
- return -1;
- }
- if (QoS && ovrDSCP == 0 && overrideDSCP != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_TOS_GQOS_CONFLICT, kTraceError,
- "SetSendGQoS() QOS is already enabled and overrideDSCP differs,"
- " not allowed");
- return -1;
- }
- const WebRtc_Word32 maxBitrate(0);
- if (_socketTransportModule.SetQoS(enable,
- static_cast<WebRtc_Word32>(serviceType),
- maxBitrate,
- static_cast<WebRtc_Word32>(overrideDSCP),
- true))
- {
- UdpTransport::ErrorCode lastSockError(
- _socketTransportModule.LastError());
- switch (lastSockError)
- {
- case UdpTransport::kQosError:
- _engineStatisticsPtr->SetLastError(VE_GQOS_ERROR, kTraceError,
- "SetSendGQoS() QOS error");
- break;
- default:
- _engineStatisticsPtr->SetLastError(VE_SOCKET_ERROR, kTraceError,
- "SetSendGQoS() Socket error");
- break;
- }
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
- "UdpTransport() => lastError = %d",
- lastSockError);
- return -1;
- }
- return 0;
-}
-#endif
-
-#if defined(_WIN32)
-WebRtc_Word32
-Channel::GetSendGQoS(bool &enabled, int &serviceType, int &overrideDSCP)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetSendGQoS(enable=?, serviceType=?, "
- "overrideDSCP=?)");
-
- bool QoS(false);
- WebRtc_Word32 serviceTypeModule(0);
- WebRtc_Word32 overrideDSCPModule(0);
- _socketTransportModule.QoS(QoS, serviceTypeModule, overrideDSCPModule);
-
- enabled = QoS;
- serviceType = static_cast<int> (serviceTypeModule);
- overrideDSCP = static_cast<int> (overrideDSCPModule);
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "GetSendGQoS() => enabled=%d, serviceType=%d, overrideDSCP=%d",
- (int)enabled, serviceType, overrideDSCP);
- return 0;
-}
-#endif
-#endif
-
-WebRtc_Word32
-Channel::SetPacketTimeoutNotification(bool enable, int timeoutSeconds)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetPacketTimeoutNotification()");
- if (enable)
- {
- const WebRtc_UWord32 RTPtimeoutMS = 1000*timeoutSeconds;
- const WebRtc_UWord32 RTCPtimeoutMS = 0;
- _rtpRtcpModule.SetPacketTimeout(RTPtimeoutMS, RTCPtimeoutMS);
- _rtpPacketTimeOutIsEnabled = true;
- _rtpTimeOutSeconds = timeoutSeconds;
- }
- else
- {
- _rtpRtcpModule.SetPacketTimeout(0, 0);
- _rtpPacketTimeOutIsEnabled = false;
- _rtpTimeOutSeconds = 0;
- }
- return 0;
-}
-
-WebRtc_Word32
-Channel::GetPacketTimeoutNotification(bool& enabled, int& timeoutSeconds)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetPacketTimeoutNotification()");
- enabled = _rtpPacketTimeOutIsEnabled;
- if (enabled)
- {
- timeoutSeconds = _rtpTimeOutSeconds;
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetPacketTimeoutNotification() => enabled=%d,"
- " timeoutSeconds=%d",
- enabled, timeoutSeconds);
- return 0;
-}
-
-WebRtc_Word32
-Channel::RegisterDeadOrAliveObserver(VoEConnectionObserver& observer)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::RegisterDeadOrAliveObserver()");
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_connectionObserverPtr)
- {
- _engineStatisticsPtr->SetLastError(VE_INVALID_OPERATION, kTraceError,
- "RegisterDeadOrAliveObserver() observer already enabled");
- return -1;
- }
-
- _connectionObserverPtr = &observer;
- _connectionObserver = true;
-
- return 0;
-}
-
-WebRtc_Word32
-Channel::DeRegisterDeadOrAliveObserver()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::DeRegisterDeadOrAliveObserver()");
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (!_connectionObserverPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "DeRegisterDeadOrAliveObserver() observer already disabled");
- return 0;
- }
-
- _connectionObserver = false;
- _connectionObserverPtr = NULL;
-
- return 0;
-}
-
-WebRtc_Word32
-Channel::SetPeriodicDeadOrAliveStatus(bool enable, int sampleTimeSeconds)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetPeriodicDeadOrAliveStatus()");
- if (!_connectionObserverPtr)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
- "SetPeriodicDeadOrAliveStatus() connection observer has"
- " not been registered");
- }
- if (enable)
- {
- ResetDeadOrAliveCounters();
- }
- bool enabled(false);
- WebRtc_UWord8 currentSampleTimeSec(0);
- // Store last state (will be used later if dead-or-alive is disabled).
- _rtpRtcpModule.PeriodicDeadOrAliveStatus(enabled, currentSampleTimeSec);
- // Update the dead-or-alive state.
- if (_rtpRtcpModule.SetPeriodicDeadOrAliveStatus(
- enable, (WebRtc_UWord8)sampleTimeSeconds) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR,
- kTraceError,
- "SetPeriodicDeadOrAliveStatus() failed to set dead-or-alive "
- "status");
- return -1;
- }
- if (!enable)
- {
- // Restore last utilized sample time.
- // Without this, the sample time would always be reset to default
- // (2 sec), each time dead-or-alived was disabled without sample-time
- // parameter.
- _rtpRtcpModule.SetPeriodicDeadOrAliveStatus(enable,
- currentSampleTimeSec);
- }
- return 0;
-}
-
-WebRtc_Word32
-Channel::GetPeriodicDeadOrAliveStatus(bool& enabled, int& sampleTimeSeconds)
-{
- _rtpRtcpModule.PeriodicDeadOrAliveStatus(
- enabled,
- (WebRtc_UWord8&)sampleTimeSeconds);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetPeriodicDeadOrAliveStatus() => enabled=%d,"
- " sampleTimeSeconds=%d",
- enabled, sampleTimeSeconds);
- return 0;
-}
-
-WebRtc_Word32
-Channel::SendUDPPacket(const void* data,
- unsigned int length,
- int& transmittedBytes,
- bool useRtcpSocket)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SendUDPPacket()");
- if (_externalTransport)
- {
- _engineStatisticsPtr->SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "SendUDPPacket() external transport is enabled");
- return -1;
- }
- if (useRtcpSocket && !_rtpRtcpModule.RTCP())
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTCP_ERROR, kTraceError,
- "SendUDPPacket() RTCP is disabled");
- return -1;
- }
- if (!_sending)
- {
- _engineStatisticsPtr->SetLastError(
- VE_NOT_SENDING, kTraceError,
- "SendUDPPacket() not sending");
- return -1;
- }
-
- char* dataC = new char[length];
- if (NULL == dataC)
- {
- _engineStatisticsPtr->SetLastError(
- VE_NO_MEMORY, kTraceError,
- "SendUDPPacket() memory allocation failed");
- return -1;
- }
- memcpy(dataC, data, length);
-
- transmittedBytes = SendPacketRaw(dataC, length, useRtcpSocket);
-
- delete [] dataC;
- dataC = NULL;
-
- if (transmittedBytes <= 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SEND_ERROR, kTraceError,
- "SendUDPPacket() transmission failed");
- transmittedBytes = 0;
- return -1;
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "SendUDPPacket() => transmittedBytes=%d", transmittedBytes);
- return 0;
-}
-
-
-int Channel::StartPlayingFileLocally(const char* fileName,
- const bool loop,
- const FileFormats format,
- const int startPosition,
- const float volumeScaling,
- const int stopPosition,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::StartPlayingFileLocally(fileNameUTF8[]=%s, loop=%d,"
- " format=%d, volumeScaling=%5.3f, startPosition=%d, "
- "stopPosition=%d)", fileName, loop, format, volumeScaling,
- startPosition, stopPosition);
-
- if (_outputFilePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceError,
- "StartPlayingFileLocally() is already playing");
- return -1;
- }
-
- {
- CriticalSectionScoped cs(_fileCritSect);
-
- if (_outputFilePlayerPtr)
- {
- _outputFilePlayerPtr->RegisterModuleFileCallback(NULL);
- FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
- _outputFilePlayerPtr = NULL;
- }
-
- _outputFilePlayerPtr = FilePlayer::CreateFilePlayer(
- _outputFilePlayerId, (const FileFormats)format);
-
- if (_outputFilePlayerPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartPlayingFileLocally() filePlayer format is not correct");
- return -1;
- }
-
- const WebRtc_UWord32 notificationTime(0);
-
- if (_outputFilePlayerPtr->StartPlayingFile(
- fileName,
- loop,
- startPosition,
- volumeScaling,
- notificationTime,
- stopPosition,
- (const CodecInst*)codecInst) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to start file playout");
- _outputFilePlayerPtr->StopPlayingFile();
- FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
- _outputFilePlayerPtr = NULL;
- return -1;
- }
- _outputFilePlayerPtr->RegisterModuleFileCallback(this);
- _outputFilePlaying = true;
- }
- // _fileCritSect cannot be taken while calling
- // SetAnonymousMixabilityStatus() since as soon as the participant is added
- // frames can be pulled by the mixer. Since the frames are generated from
- // the file, _fileCritSect will be taken. This would result in a deadlock.
- if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, true) != 0)
- {
- CriticalSectionScoped cs(_fileCritSect);
- _outputFilePlaying = false;
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
- "StartPlayingFile() failed to add participant as file to mixer");
- _outputFilePlayerPtr->StopPlayingFile();
- FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
- _outputFilePlayerPtr = NULL;
- return -1;
- }
-
- return 0;
-}
-
-int Channel::StartPlayingFileLocally(InStream* stream,
- const FileFormats format,
- const int startPosition,
- const float volumeScaling,
- const int stopPosition,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::StartPlayingFileLocally(format=%d,"
- " volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)",
- format, volumeScaling, startPosition, stopPosition);
-
- if(stream == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFileLocally() NULL as input stream");
- return -1;
- }
-
-
- if (_outputFilePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceError,
- "StartPlayingFileLocally() is already playing");
- return -1;
- }
-
- {
- CriticalSectionScoped cs(_fileCritSect);
-
- // Destroy the old instance
- if (_outputFilePlayerPtr)
- {
- _outputFilePlayerPtr->RegisterModuleFileCallback(NULL);
- FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
- _outputFilePlayerPtr = NULL;
- }
-
- // Create the instance
- _outputFilePlayerPtr = FilePlayer::CreateFilePlayer(
- _outputFilePlayerId,
- (const FileFormats)format);
-
- if (_outputFilePlayerPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartPlayingFileLocally() filePlayer format isnot correct");
- return -1;
- }
-
- const WebRtc_UWord32 notificationTime(0);
-
- if (_outputFilePlayerPtr->StartPlayingFile(*stream, startPosition,
- volumeScaling,
- notificationTime,
- stopPosition, codecInst) != 0)
- {
- _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to "
- "start file playout");
- _outputFilePlayerPtr->StopPlayingFile();
- FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
- _outputFilePlayerPtr = NULL;
- return -1;
- }
- _outputFilePlayerPtr->RegisterModuleFileCallback(this);
- _outputFilePlaying = true;
- }
- // _fileCritSect cannot be taken while calling
- // SetAnonymousMixibilityStatus. Refer to comments in
- // StartPlayingFileLocally(const char* ...) for more details.
- if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, true) != 0)
- {
- CriticalSectionScoped cs(_fileCritSect);
- _outputFilePlaying = false;
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
- "StartPlayingFile() failed to add participant as file to mixer");
- _outputFilePlayerPtr->StopPlayingFile();
- FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
- _outputFilePlayerPtr = NULL;
- return -1;
- }
-
- return 0;
-}
-
-int Channel::StopPlayingFileLocally()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::StopPlayingFileLocally()");
-
- if (!_outputFilePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "StopPlayingFileLocally() isnot playing");
- return 0;
- }
-
- {
- CriticalSectionScoped cs(_fileCritSect);
-
- if (_outputFilePlayerPtr->StopPlayingFile() != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_STOP_RECORDING_FAILED, kTraceError,
- "StopPlayingFile() could not stop playing");
- return -1;
- }
- _outputFilePlayerPtr->RegisterModuleFileCallback(NULL);
- FilePlayer::DestroyFilePlayer(_outputFilePlayerPtr);
- _outputFilePlayerPtr = NULL;
- _outputFilePlaying = false;
- }
- // _fileCritSect cannot be taken while calling
- // SetAnonymousMixibilityStatus. Refer to comments in
- // StartPlayingFileLocally(const char* ...) for more details.
- if (_outputMixerPtr->SetAnonymousMixabilityStatus(*this, false) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CONF_MIX_MODULE_ERROR, kTraceError,
- "StopPlayingFile() failed to stop participant from playing as"
- "file in the mixer");
- return -1;
- }
-
- return 0;
-}
-
-int Channel::IsPlayingFileLocally() const
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::IsPlayingFileLocally()");
-
- return (WebRtc_Word32)_outputFilePlaying;
-}
-
-int Channel::ScaleLocalFilePlayout(const float scale)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::ScaleLocalFilePlayout(scale=%5.3f)", scale);
-
- CriticalSectionScoped cs(_fileCritSect);
-
- if (!_outputFilePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "ScaleLocalFilePlayout() isnot playing");
- return -1;
- }
- if ((_outputFilePlayerPtr == NULL) ||
- (_outputFilePlayerPtr->SetAudioScaling(scale) != 0))
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "SetAudioScaling() failed to scale the playout");
- return -1;
- }
-
- return 0;
-}
-
-int Channel::GetLocalPlayoutPosition(int& positionMs)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetLocalPlayoutPosition(position=?)");
-
- WebRtc_UWord32 position;
-
- CriticalSectionScoped cs(_fileCritSect);
-
- if (_outputFilePlayerPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "GetLocalPlayoutPosition() filePlayer instance doesnot exist");
- return -1;
- }
-
- if (_outputFilePlayerPtr->GetPlayoutPosition(position) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "GetLocalPlayoutPosition() failed");
- return -1;
- }
- positionMs = position;
-
- return 0;
-}
-
-int Channel::StartPlayingFileAsMicrophone(const char* fileName,
- const bool loop,
- const FileFormats format,
- const int startPosition,
- const float volumeScaling,
- const int stopPosition,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::StartPlayingFileAsMicrophone(fileNameUTF8[]=%s, "
- "loop=%d, format=%d, volumeScaling=%5.3f, startPosition=%d, "
- "stopPosition=%d)", fileName, loop, format, volumeScaling,
- startPosition, stopPosition);
-
- if (_inputFilePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceWarning,
- "StartPlayingFileAsMicrophone() filePlayer is playing");
- return 0;
- }
-
- CriticalSectionScoped cs(_fileCritSect);
-
- // Destroy the old instance
- if (_inputFilePlayerPtr)
- {
- _inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
- FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
- _inputFilePlayerPtr = NULL;
- }
-
- // Create the instance
- _inputFilePlayerPtr = FilePlayer::CreateFilePlayer(
- _inputFilePlayerId, (const FileFormats)format);
-
- if (_inputFilePlayerPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartPlayingFileAsMicrophone() filePlayer format isnot correct");
- return -1;
- }
-
- const WebRtc_UWord32 notificationTime(0);
-
- if (_inputFilePlayerPtr->StartPlayingFile(
- fileName,
- loop,
- startPosition,
- volumeScaling,
- notificationTime,
- stopPosition,
- (const CodecInst*)codecInst) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to start file playout");
- _inputFilePlayerPtr->StopPlayingFile();
- FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
- _inputFilePlayerPtr = NULL;
- return -1;
- }
- _inputFilePlayerPtr->RegisterModuleFileCallback(this);
- _inputFilePlaying = true;
-
- return 0;
-}
-
-int Channel::StartPlayingFileAsMicrophone(InStream* stream,
- const FileFormats format,
- const int startPosition,
- const float volumeScaling,
- const int stopPosition,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::StartPlayingFileAsMicrophone(format=%d, "
- "volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)",
- format, volumeScaling, startPosition, stopPosition);
-
- if(stream == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFileAsMicrophone NULL as input stream");
- return -1;
- }
-
- if (_inputFilePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceWarning,
- "StartPlayingFileAsMicrophone() is playing");
- return 0;
- }
-
- CriticalSectionScoped cs(_fileCritSect);
-
- // Destroy the old instance
- if (_inputFilePlayerPtr)
- {
- _inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
- FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
- _inputFilePlayerPtr = NULL;
- }
-
- // Create the instance
- _inputFilePlayerPtr = FilePlayer::CreateFilePlayer(
- _inputFilePlayerId, (const FileFormats)format);
-
- if (_inputFilePlayerPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartPlayingInputFile() filePlayer format isnot correct");
- return -1;
- }
-
- const WebRtc_UWord32 notificationTime(0);
-
- if (_inputFilePlayerPtr->StartPlayingFile(*stream, startPosition,
- volumeScaling, notificationTime,
- stopPosition, codecInst) != 0)
- {
- _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to start "
- "file playout");
- _inputFilePlayerPtr->StopPlayingFile();
- FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
- _inputFilePlayerPtr = NULL;
- return -1;
- }
-
- _inputFilePlayerPtr->RegisterModuleFileCallback(this);
- _inputFilePlaying = true;
-
- return 0;
-}
-
-int Channel::StopPlayingFileAsMicrophone()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::StopPlayingFileAsMicrophone()");
-
- if (!_inputFilePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "StopPlayingFileAsMicrophone() isnot playing");
- return 0;
- }
-
- CriticalSectionScoped cs(_fileCritSect);
- if (_inputFilePlayerPtr->StopPlayingFile() != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_STOP_RECORDING_FAILED, kTraceError,
- "StopPlayingFile() could not stop playing");
- return -1;
- }
- _inputFilePlayerPtr->RegisterModuleFileCallback(NULL);
- FilePlayer::DestroyFilePlayer(_inputFilePlayerPtr);
- _inputFilePlayerPtr = NULL;
- _inputFilePlaying = false;
-
- return 0;
-}
-
-int Channel::IsPlayingFileAsMicrophone() const
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::IsPlayingFileAsMicrophone()");
-
- return _inputFilePlaying;
-}
-
-int Channel::ScaleFileAsMicrophonePlayout(const float scale)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::ScaleFileAsMicrophonePlayout(scale=%5.3f)", scale);
-
- CriticalSectionScoped cs(_fileCritSect);
-
- if (!_inputFilePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "ScaleFileAsMicrophonePlayout() isnot playing");
- return -1;
- }
-
- if ((_inputFilePlayerPtr == NULL) ||
- (_inputFilePlayerPtr->SetAudioScaling(scale) != 0))
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "SetAudioScaling() failed to scale playout");
- return -1;
- }
-
- return 0;
-}
-
-int Channel::StartRecordingPlayout(const WebRtc_Word8* fileName,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::StartRecordingPlayout(fileName=%s)", fileName);
-
- if (_outputFileRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
- "StartRecordingPlayout() is already recording");
- return 0;
- }
-
- FileFormats format;
- const WebRtc_UWord32 notificationTime(0); // Not supported in VoE
- CodecInst dummyCodec={100,"L16",16000,320,1,320000};
-
- if (codecInst != NULL && codecInst->channels != 1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingPlayout() invalid compression");
- return(-1);
- }
- if(codecInst == NULL)
- {
- format = kFileFormatPcm16kHzFile;
- codecInst=&dummyCodec;
- }
- else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
- {
- format = kFileFormatWavFile;
- }
- else
- {
- format = kFileFormatCompressedFile;
- }
-
- CriticalSectionScoped cs(_fileCritSect);
-
- // Destroy the old instance
- if (_outputFileRecorderPtr)
- {
- _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
- _outputFileRecorderPtr = NULL;
- }
-
- _outputFileRecorderPtr = FileRecorder::CreateFileRecorder(
- _outputFileRecorderId, (const FileFormats)format);
- if (_outputFileRecorderPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingPlayout() fileRecorder format isnot correct");
- return -1;
- }
-
- if (_outputFileRecorderPtr->StartRecordingAudioFile(
- fileName, (const CodecInst&)*codecInst, notificationTime) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- _outputFileRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
- _outputFileRecorderPtr = NULL;
- return -1;
- }
- _outputFileRecorderPtr->RegisterModuleFileCallback(this);
- _outputFileRecording = true;
-
- return 0;
-}
-
-int Channel::StartRecordingPlayout(OutStream* stream,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::StartRecordingPlayout()");
-
- if (_outputFileRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
- "StartRecordingPlayout() is already recording");
- return 0;
- }
-
- FileFormats format;
- const WebRtc_UWord32 notificationTime(0); // Not supported in VoE
- CodecInst dummyCodec={100,"L16",16000,320,1,320000};
-
- if (codecInst != NULL && codecInst->channels != 1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingPlayout() invalid compression");
- return(-1);
- }
- if(codecInst == NULL)
- {
- format = kFileFormatPcm16kHzFile;
- codecInst=&dummyCodec;
- }
- else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
- {
- format = kFileFormatWavFile;
- }
- else
- {
- format = kFileFormatCompressedFile;
- }
-
- CriticalSectionScoped cs(_fileCritSect);
-
- // Destroy the old instance
- if (_outputFileRecorderPtr)
- {
- _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
- _outputFileRecorderPtr = NULL;
- }
-
- _outputFileRecorderPtr = FileRecorder::CreateFileRecorder(
- _outputFileRecorderId, (const FileFormats)format);
- if (_outputFileRecorderPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingPlayout() fileRecorder format isnot correct");
- return -1;
- }
-
- if (_outputFileRecorderPtr->StartRecordingAudioFile(*stream, *codecInst,
- notificationTime) != 0)
- {
- _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
- "StartRecordingPlayout() failed to "
- "start file recording");
- _outputFileRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
- _outputFileRecorderPtr = NULL;
- return -1;
- }
-
- _outputFileRecorderPtr->RegisterModuleFileCallback(this);
- _outputFileRecording = true;
-
- return 0;
-}
-
-int Channel::StopRecordingPlayout()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "Channel::StopRecordingPlayout()");
-
- if (!_outputFileRecording)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "StopRecordingPlayout() isnot recording");
- return -1;
- }
-
-
- CriticalSectionScoped cs(_fileCritSect);
-
- if (_outputFileRecorderPtr->StopRecording() != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_STOP_RECORDING_FAILED, kTraceError,
- "StopRecording() could not stop recording");
- return(-1);
- }
- _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
- _outputFileRecorderPtr = NULL;
- _outputFileRecording = false;
-
- return 0;
-}
-
-void
-Channel::SetMixWithMicStatus(bool mix)
-{
- _mixFileWithMicrophone=mix;
-}
-
-int
-Channel::GetSpeechOutputLevel(WebRtc_UWord32& level) const
-{
- WebRtc_Word8 currentLevel = _outputAudioLevel.Level();
- level = static_cast<WebRtc_Word32> (currentLevel);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "GetSpeechOutputLevel() => level=%u", level);
- return 0;
-}
-
-int
-Channel::GetSpeechOutputLevelFullRange(WebRtc_UWord32& level) const
-{
- WebRtc_Word16 currentLevel = _outputAudioLevel.LevelFullRange();
- level = static_cast<WebRtc_Word32> (currentLevel);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "GetSpeechOutputLevelFullRange() => level=%u", level);
- return 0;
-}
-
-int
-Channel::SetMute(bool enable)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetMute(enable=%d)", enable);
- _mute = enable;
- return 0;
-}
-
-bool
-Channel::Mute() const
-{
- return _mute;
-}
-
-int
-Channel::SetOutputVolumePan(float left, float right)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetOutputVolumePan()");
- _panLeft = left;
- _panRight = right;
- return 0;
-}
-
-int
-Channel::GetOutputVolumePan(float& left, float& right) const
-{
- left = _panLeft;
- right = _panRight;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "GetOutputVolumePan() => left=%3.2f, right=%3.2f", left, right);
- return 0;
-}
-
-int
-Channel::SetChannelOutputVolumeScaling(float scaling)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetChannelOutputVolumeScaling()");
- _outputGain = scaling;
- return 0;
-}
-
-int
-Channel::GetChannelOutputVolumeScaling(float& scaling) const
-{
- scaling = _outputGain;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "GetChannelOutputVolumeScaling() => scaling=%3.2f", scaling);
- return 0;
-}
-
-#ifdef WEBRTC_SRTP
-
-int
-Channel::EnableSRTPSend(
- CipherTypes cipherType,
- int cipherKeyLength,
- AuthenticationTypes authType,
- int authKeyLength,
- int authTagLength,
- SecurityLevels level,
- const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
- bool useForRTCP)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::EnableSRTPSend()");
-
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_encrypting)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "EnableSRTPSend() encryption already enabled");
- return -1;
- }
-
- if (key == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceWarning,
- "EnableSRTPSend() invalid key string");
- return -1;
- }
-
- if (((kEncryption == level ||
- kEncryptionAndAuthentication == level) &&
- (cipherKeyLength < kVoiceEngineMinSrtpEncryptLength ||
- cipherKeyLength > kVoiceEngineMaxSrtpEncryptLength)) ||
- ((kAuthentication == level ||
- kEncryptionAndAuthentication == level) &&
- kAuthHmacSha1 == authType &&
- (authKeyLength > kVoiceEngineMaxSrtpAuthSha1Length ||
- authTagLength > kVoiceEngineMaxSrtpAuthSha1Length)) ||
- ((kAuthentication == level ||
- kEncryptionAndAuthentication == level) &&
- kAuthNull == authType &&
- (authKeyLength > kVoiceEngineMaxSrtpKeyAuthNullLength ||
- authTagLength > kVoiceEngineMaxSrtpTagAuthNullLength)))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "EnableSRTPSend() invalid key length(s)");
- return -1;
- }
-
-
- if (_srtpModule.EnableSRTPEncrypt(
- !useForRTCP,
- (SrtpModule::CipherTypes)cipherType,
- cipherKeyLength,
- (SrtpModule::AuthenticationTypes)authType,
- authKeyLength, authTagLength,
- (SrtpModule::SecurityLevels)level,
- key) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SRTP_ERROR, kTraceError,
- "EnableSRTPSend() failed to enable SRTP encryption");
- return -1;
- }
-
- if (_encryptionPtr == NULL)
- {
- _encryptionPtr = &_srtpModule;
- }
- _encrypting = true;
-
- return 0;
-}
-
-int
-Channel::DisableSRTPSend()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::DisableSRTPSend()");
-
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (!_encrypting)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "DisableSRTPSend() SRTP encryption already disabled");
- return 0;
- }
-
- _encrypting = false;
-
- if (_srtpModule.DisableSRTPEncrypt() == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SRTP_ERROR, kTraceError,
- "DisableSRTPSend() failed to disable SRTP encryption");
- return -1;
- }
-
- if (!_srtpModule.SRTPDecrypt() && !_srtpModule.SRTPEncrypt())
- {
- // Both directions are disabled
- _encryptionPtr = NULL;
- }
-
- return 0;
-}
-
-int
-Channel::EnableSRTPReceive(
- CipherTypes cipherType,
- int cipherKeyLength,
- AuthenticationTypes authType,
- int authKeyLength,
- int authTagLength,
- SecurityLevels level,
- const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
- bool useForRTCP)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::EnableSRTPReceive()");
-
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_decrypting)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "EnableSRTPReceive() SRTP decryption already enabled");
- return -1;
- }
-
- if (key == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceWarning,
- "EnableSRTPReceive() invalid key string");
- return -1;
- }
-
- if ((((kEncryption == level) ||
- (kEncryptionAndAuthentication == level)) &&
- ((cipherKeyLength < kVoiceEngineMinSrtpEncryptLength) ||
- (cipherKeyLength > kVoiceEngineMaxSrtpEncryptLength))) ||
- (((kAuthentication == level) ||
- (kEncryptionAndAuthentication == level)) &&
- (kAuthHmacSha1 == authType) &&
- ((authKeyLength > kVoiceEngineMaxSrtpAuthSha1Length) ||
- (authTagLength > kVoiceEngineMaxSrtpAuthSha1Length))) ||
- (((kAuthentication == level) ||
- (kEncryptionAndAuthentication == level)) &&
- (kAuthNull == authType) &&
- ((authKeyLength > kVoiceEngineMaxSrtpKeyAuthNullLength) ||
- (authTagLength > kVoiceEngineMaxSrtpTagAuthNullLength))))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "EnableSRTPReceive() invalid key length(s)");
- return -1;
- }
-
- if (_srtpModule.EnableSRTPDecrypt(
- !useForRTCP,
- (SrtpModule::CipherTypes)cipherType,
- cipherKeyLength,
- (SrtpModule::AuthenticationTypes)authType,
- authKeyLength,
- authTagLength,
- (SrtpModule::SecurityLevels)level,
- key) == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SRTP_ERROR, kTraceError,
- "EnableSRTPReceive() failed to enable SRTP decryption");
- return -1;
- }
-
- if (_encryptionPtr == NULL)
- {
- _encryptionPtr = &_srtpModule;
- }
-
- _decrypting = true;
-
- return 0;
-}
-
-int
-Channel::DisableSRTPReceive()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::DisableSRTPReceive()");
-
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (!_decrypting)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "DisableSRTPReceive() SRTP decryption already disabled");
- return 0;
- }
-
- _decrypting = false;
-
- if (_srtpModule.DisableSRTPDecrypt() == -1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SRTP_ERROR, kTraceError,
- "DisableSRTPReceive() failed to disable SRTP decryption");
- return -1;
- }
-
- if (!_srtpModule.SRTPDecrypt() && !_srtpModule.SRTPEncrypt())
- {
- _encryptionPtr = NULL;
- }
-
- return 0;
-}
-
-#endif
-
-int
-Channel::RegisterExternalEncryption(Encryption& encryption)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::RegisterExternalEncryption()");
-
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_encryptionPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "RegisterExternalEncryption() encryption already enabled");
- return -1;
- }
-
- _encryptionPtr = &encryption;
-
- _decrypting = true;
- _encrypting = true;
-
- return 0;
-}
-
-int
-Channel::DeRegisterExternalEncryption()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::DeRegisterExternalEncryption()");
-
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (!_encryptionPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "DeRegisterExternalEncryption() encryption already disabled");
- return 0;
- }
-
- _decrypting = false;
- _encrypting = false;
-
- _encryptionPtr = NULL;
-
- return 0;
-}
-
-int Channel::SendTelephoneEventOutband(unsigned char eventCode,
- int lengthMs, int attenuationDb,
- bool playDtmfEvent)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::SendTelephoneEventOutband(..., playDtmfEvent=%d)",
- playDtmfEvent);
-
- _playOutbandDtmfEvent = playDtmfEvent;
-
- if (_rtpRtcpModule.SendTelephoneEventOutband(eventCode, lengthMs,
- attenuationDb) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SEND_DTMF_FAILED,
- kTraceWarning,
- "SendTelephoneEventOutband() failed to send event");
- return -1;
- }
- return 0;
-}
-
-int Channel::SendTelephoneEventInband(unsigned char eventCode,
- int lengthMs,
- int attenuationDb,
- bool playDtmfEvent)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::SendTelephoneEventInband(..., playDtmfEvent=%d)",
- playDtmfEvent);
-
- _playInbandDtmfEvent = playDtmfEvent;
- _inbandDtmfQueue.AddDtmf(eventCode, lengthMs, attenuationDb);
-
- return 0;
-}
-
-int
-Channel::SetDtmfPlayoutStatus(bool enable)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetDtmfPlayoutStatus()");
- if (_audioCodingModule.SetDtmfPlayoutStatus(enable) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceWarning,
- "SetDtmfPlayoutStatus() failed to set Dtmf playout");
- return -1;
- }
- return 0;
-}
-
-bool
-Channel::DtmfPlayoutStatus() const
-{
- return _audioCodingModule.DtmfPlayoutStatus();
-}
-
-int
-Channel::SetSendTelephoneEventPayloadType(unsigned char type)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetSendTelephoneEventPayloadType()");
- if (type > 127)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetSendTelephoneEventPayloadType() invalid type");
- return -1;
- }
- CodecInst codec;
- codec.plfreq = 8000;
- codec.pltype = type;
- memcpy(codec.plname, "telephone-event", 16);
- if (_rtpRtcpModule.RegisterSendPayload(codec) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "SetSendTelephoneEventPayloadType() failed to register send"
- "payload type");
- return -1;
- }
- _sendTelephoneEventPayloadType = type;
- return 0;
-}
-
-int
-Channel::GetSendTelephoneEventPayloadType(unsigned char& type)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetSendTelephoneEventPayloadType()");
- type = _sendTelephoneEventPayloadType;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "GetSendTelephoneEventPayloadType() => type=%u", type);
- return 0;
-}
-
-#ifdef WEBRTC_DTMF_DETECTION
-
-WebRtc_Word32
-Channel::RegisterTelephoneEventDetection(
- TelephoneEventDetectionMethods detectionMethod,
- VoETelephoneEventObserver& observer)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::RegisterTelephoneEventDetection()");
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_telephoneEventDetectionPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "RegisterTelephoneEventDetection() detection already enabled");
- return -1;
- }
-
- _telephoneEventDetectionPtr = &observer;
-
- switch (detectionMethod)
- {
- case kInBand:
- _inbandTelephoneEventDetection = true;
- _outOfBandTelephoneEventDetecion = false;
- break;
- case kOutOfBand:
- _inbandTelephoneEventDetection = false;
- _outOfBandTelephoneEventDetecion = true;
- break;
- case kInAndOutOfBand:
- _inbandTelephoneEventDetection = true;
- _outOfBandTelephoneEventDetecion = true;
- break;
- default:
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "RegisterTelephoneEventDetection() invalid detection method");
- return -1;
- }
-
- if (_inbandTelephoneEventDetection)
- {
- // Enable in-band Dtmf detectin in the ACM.
- if (_audioCodingModule.RegisterIncomingMessagesCallback(this) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "RegisterTelephoneEventDetection() failed to enable Dtmf "
- "detection");
- }
- }
-
- // Enable/disable out-of-band detection of received telephone-events.
- // When enabled, RtpAudioFeedback::OnReceivedTelephoneEvent() will be
- // called two times by the RTP/RTCP module (start & end).
- const bool forwardToDecoder =
- _rtpRtcpModule.TelephoneEventForwardToDecoder();
- const bool detectEndOfTone = true;
- _rtpRtcpModule.SetTelephoneEventStatus(_outOfBandTelephoneEventDetecion,
- forwardToDecoder,
- detectEndOfTone);
-
- return 0;
-}
-
-int
-Channel::DeRegisterTelephoneEventDetection()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::DeRegisterTelephoneEventDetection()");
-
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (!_telephoneEventDetectionPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION,
- kTraceWarning,
- "DeRegisterTelephoneEventDetection() detection already disabled");
- return 0;
- }
-
- // Disable out-of-band event detection
- const bool forwardToDecoder =
- _rtpRtcpModule.TelephoneEventForwardToDecoder();
- _rtpRtcpModule.SetTelephoneEventStatus(false, forwardToDecoder);
-
- // Disable in-band Dtmf detection
- _audioCodingModule.RegisterIncomingMessagesCallback(NULL);
-
- _inbandTelephoneEventDetection = false;
- _outOfBandTelephoneEventDetecion = false;
- _telephoneEventDetectionPtr = NULL;
-
- return 0;
-}
-
-int
-Channel::GetTelephoneEventDetectionStatus(
- bool& enabled,
- TelephoneEventDetectionMethods& detectionMethod)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::GetTelephoneEventDetectionStatus()");
-
- {
- CriticalSectionScoped cs(_callbackCritSect);
- enabled = (_telephoneEventDetectionPtr != NULL);
- }
-
- if (enabled)
- {
- if (_inbandTelephoneEventDetection && !_outOfBandTelephoneEventDetecion)
- detectionMethod = kInBand;
- else if (!_inbandTelephoneEventDetection
- && _outOfBandTelephoneEventDetecion)
- detectionMethod = kOutOfBand;
- else if (_inbandTelephoneEventDetection
- && _outOfBandTelephoneEventDetecion)
- detectionMethod = kInAndOutOfBand;
- else
- {
- assert(false);
- return -1;
- }
- }
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetTelephoneEventDetectionStatus() => enabled=%d,"
- "detectionMethod=%d", enabled, detectionMethod);
- return 0;
-}
-
-#endif // #ifdef WEBRTC_DTMF_DETECTION
-
-int
-Channel::UpdateRxVadDetection(AudioFrame& audioFrame)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::UpdateRxVadDetection()");
-
- int vadDecision = 1;
-
- vadDecision = (audioFrame._vadActivity == AudioFrame::kVadActive)? 1 : 0;
-
- if ((vadDecision != _oldVadDecision) && _rxVadObserverPtr)
- {
- OnRxVadDetected(vadDecision);
- _oldVadDecision = vadDecision;
- }
-
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::UpdateRxVadDetection() => vadDecision=%d",
- vadDecision);
- return 0;
-}
-
-int
-Channel::RegisterRxVadObserver(VoERxVadCallback &observer)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::RegisterRxVadObserver()");
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_rxVadObserverPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "RegisterRxVadObserver() observer already enabled");
- return -1;
- }
- _rxVadObserverPtr = &observer;
- _RxVadDetection = true;
- return 0;
-}
-
-int
-Channel::DeRegisterRxVadObserver()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::DeRegisterRxVadObserver()");
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (!_rxVadObserverPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "DeRegisterRxVadObserver() observer already disabled");
- return 0;
- }
- _rxVadObserverPtr = NULL;
- _RxVadDetection = false;
- return 0;
-}
-
-int
-Channel::VoiceActivityIndicator(int &activity)
-{
- activity = _sendFrameType;
-
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::VoiceActivityIndicator(indicator=%d)", activity);
- return 0;
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_AGC
-
-int
-Channel::SetRxAgcStatus(const bool enable, const AgcModes mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetRxAgcStatus(enable=%d, mode=%d)",
- (int)enable, (int)mode);
-
- GainControl::Mode agcMode(GainControl::kFixedDigital);
- switch (mode)
- {
- case kAgcDefault:
- agcMode = GainControl::kAdaptiveDigital;
- break;
- case kAgcUnchanged:
- agcMode = _rxAudioProcessingModulePtr->gain_control()->mode();
- break;
- case kAgcFixedDigital:
- agcMode = GainControl::kFixedDigital;
- break;
- case kAgcAdaptiveDigital:
- agcMode =GainControl::kAdaptiveDigital;
- break;
- default:
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetRxAgcStatus() invalid Agc mode");
- return -1;
- }
-
- if (_rxAudioProcessingModulePtr->gain_control()->set_mode(agcMode) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetRxAgcStatus() failed to set Agc mode");
- return -1;
- }
- if (_rxAudioProcessingModulePtr->gain_control()->Enable(enable) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetRxAgcStatus() failed to set Agc state");
- return -1;
- }
-
- _rxAgcIsEnabled = enable;
-
- _rxApmIsEnabled = ((_rxAgcIsEnabled == true) || (_rxNsIsEnabled == true));
-
- return 0;
-}
-
-int
-Channel::GetRxAgcStatus(bool& enabled, AgcModes& mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetRxAgcStatus(enable=?, mode=?)");
-
- bool enable = _rxAudioProcessingModulePtr->gain_control()->is_enabled();
- GainControl::Mode agcMode =
- _rxAudioProcessingModulePtr->gain_control()->mode();
-
- enabled = enable;
-
- switch (agcMode)
- {
- case GainControl::kFixedDigital:
- mode = kAgcFixedDigital;
- break;
- case GainControl::kAdaptiveDigital:
- mode = kAgcAdaptiveDigital;
- break;
- default:
- _engineStatisticsPtr->SetLastError(
- VE_APM_ERROR, kTraceError,
- "GetRxAgcStatus() invalid Agc mode");
- return -1;
- }
-
- return 0;
-}
-
-int
-Channel::SetRxAgcConfig(const AgcConfig config)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetRxAgcConfig()");
-
- if (_rxAudioProcessingModulePtr->gain_control()->set_target_level_dbfs(
- config.targetLeveldBOv) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetRxAgcConfig() failed to set target peak |level|"
- "(or envelope) of the Agc");
- return -1;
- }
- if (_rxAudioProcessingModulePtr->gain_control()->set_compression_gain_db(
- config.digitalCompressionGaindB) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetRxAgcConfig() failed to set the range in |gain| the"
- " digital compression stage may apply");
- return -1;
- }
- if (_rxAudioProcessingModulePtr->gain_control()->enable_limiter(
- config.limiterEnable) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetRxAgcConfig() failed to set hard limiter to the signal");
- return -1;
- }
-
- return 0;
-}
-
-int
-Channel::GetRxAgcConfig(AgcConfig& config)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetRxAgcConfig(config=%?)");
-
- config.targetLeveldBOv =
- _rxAudioProcessingModulePtr->gain_control()->target_level_dbfs();
- config.digitalCompressionGaindB =
- _rxAudioProcessingModulePtr->gain_control()->compression_gain_db();
- config.limiterEnable =
- _rxAudioProcessingModulePtr->gain_control()->is_limiter_enabled();
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId,_channelId), "GetRxAgcConfig() => "
- "targetLeveldBOv=%u, digitalCompressionGaindB=%u,"
- " limiterEnable=%d",
- config.targetLeveldBOv,
- config.digitalCompressionGaindB,
- config.limiterEnable);
-
- return 0;
-}
-
-#endif // #ifdef WEBRTC_VOICE_ENGINE_AGC
-
-#ifdef WEBRTC_VOICE_ENGINE_NR
-
-int
-Channel::SetRxNsStatus(const bool enable, const NsModes mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetRxNsStatus(enable=%d, mode=%d)",
- (int)enable, (int)mode);
-
- NoiseSuppression::Level nsLevel(
- (NoiseSuppression::Level)WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE);
- switch (mode)
- {
-
- case kNsDefault:
- nsLevel = (NoiseSuppression::Level)
- WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE;
- break;
- case kNsUnchanged:
- nsLevel = _rxAudioProcessingModulePtr->noise_suppression()->level();
- break;
- case kNsConference:
- nsLevel = NoiseSuppression::kHigh;
- break;
- case kNsLowSuppression:
- nsLevel = NoiseSuppression::kLow;
- break;
- case kNsModerateSuppression:
- nsLevel = NoiseSuppression::kModerate;
- break;
- case kNsHighSuppression:
- nsLevel = NoiseSuppression::kHigh;
- break;
- case kNsVeryHighSuppression:
- nsLevel = NoiseSuppression::kVeryHigh;
- break;
- }
-
- if (_rxAudioProcessingModulePtr->noise_suppression()->set_level(nsLevel)
- != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetRxAgcStatus() failed to set Ns level");
- return -1;
- }
- if (_rxAudioProcessingModulePtr->noise_suppression()->Enable(enable) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetRxAgcStatus() failed to set Agc state");
- return -1;
- }
-
- _rxNsIsEnabled = enable;
- _rxApmIsEnabled = ((_rxAgcIsEnabled == true) || (_rxNsIsEnabled == true));
-
- return 0;
-}
-
-int
-Channel::GetRxNsStatus(bool& enabled, NsModes& mode)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetRxNsStatus(enable=?, mode=?)");
-
- bool enable =
- _rxAudioProcessingModulePtr->noise_suppression()->is_enabled();
- NoiseSuppression::Level ncLevel =
- _rxAudioProcessingModulePtr->noise_suppression()->level();
-
- enabled = enable;
-
- switch (ncLevel)
- {
- case NoiseSuppression::kLow:
- mode = kNsLowSuppression;
- break;
- case NoiseSuppression::kModerate:
- mode = kNsModerateSuppression;
- break;
- case NoiseSuppression::kHigh:
- mode = kNsHighSuppression;
- break;
- case NoiseSuppression::kVeryHigh:
- mode = kNsVeryHighSuppression;
- break;
- }
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "GetRxNsStatus() => enabled=%d, mode=%d", enabled, mode);
- return 0;
-}
-
-#endif // #ifdef WEBRTC_VOICE_ENGINE_NR
-
-int
-Channel::RegisterRTPObserver(VoERTPObserver& observer)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::RegisterRTPObserver()");
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_rtpObserverPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "RegisterRTPObserver() observer already enabled");
- return -1;
- }
-
- _rtpObserverPtr = &observer;
- _rtpObserver = true;
-
- return 0;
-}
-
-int
-Channel::DeRegisterRTPObserver()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::DeRegisterRTPObserver()");
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (!_rtpObserverPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "DeRegisterRTPObserver() observer already disabled");
- return 0;
- }
-
- _rtpObserver = false;
- _rtpObserverPtr = NULL;
-
- return 0;
-}
-
-int
-Channel::RegisterRTCPObserver(VoERTCPObserver& observer)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::RegisterRTCPObserver()");
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_rtcpObserverPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "RegisterRTCPObserver() observer already enabled");
- return -1;
- }
-
- _rtcpObserverPtr = &observer;
- _rtcpObserver = true;
-
- return 0;
-}
-
-int
-Channel::DeRegisterRTCPObserver()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::DeRegisterRTCPObserver()");
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (!_rtcpObserverPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "DeRegisterRTCPObserver() observer already disabled");
- return 0;
- }
-
- _rtcpObserver = false;
- _rtcpObserverPtr = NULL;
-
- return 0;
-}
-
-int
-Channel::SetLocalSSRC(unsigned int ssrc)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::SetLocalSSRC()");
- if (_sending)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_SENDING, kTraceError,
- "SetLocalSSRC() already sending");
- return -1;
- }
- if (_rtpRtcpModule.SetSSRC(ssrc) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "SetLocalSSRC() failed to set SSRC");
- return -1;
- }
- return 0;
-}
-
-int
-Channel::GetLocalSSRC(unsigned int& ssrc)
-{
- ssrc = _rtpRtcpModule.SSRC();
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "GetLocalSSRC() => ssrc=%lu", ssrc);
- return 0;
-}
-
-int
-Channel::GetRemoteSSRC(unsigned int& ssrc)
-{
- ssrc = _rtpRtcpModule.RemoteSSRC();
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "GetRemoteSSRC() => ssrc=%lu", ssrc);
- return 0;
-}
-
-int
-Channel::GetRemoteCSRCs(unsigned int arrCSRC[15])
-{
- if (arrCSRC == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "GetRemoteCSRCs() invalid array argument");
- return -1;
- }
- WebRtc_UWord32 arrOfCSRC[kRtpCsrcSize];
- WebRtc_Word32 CSRCs(0);
- CSRCs = _rtpRtcpModule.CSRCs(arrOfCSRC);
- if (CSRCs > 0)
- {
- memcpy(arrCSRC, arrOfCSRC, CSRCs * sizeof(WebRtc_UWord32));
- for (int i = 0; i < (int) CSRCs; i++)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRemoteCSRCs() => arrCSRC[%d]=%lu", i, arrCSRC[i]);
- }
- } else
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRemoteCSRCs() => list is empty!");
- }
- return CSRCs;
-}
-
-int
-Channel::SetRTPAudioLevelIndicationStatus(bool enable, unsigned char ID)
-{
- if (_rtpAudioProc.get() == NULL)
- {
- _rtpAudioProc.reset(AudioProcessing::Create(VoEModuleId(_instanceId,
- _channelId)));
- if (_rtpAudioProc.get() == NULL)
- {
- _engineStatisticsPtr->SetLastError(VE_NO_MEMORY, kTraceCritical,
- "Failed to create AudioProcessing");
- return -1;
- }
- }
-
- if (_rtpAudioProc->level_estimator()->Enable(enable) !=
- AudioProcessing::kNoError)
- {
- _engineStatisticsPtr->SetLastError(VE_APM_ERROR, kTraceWarning,
- "Failed to enable AudioProcessing::level_estimator()");
- }
-
- _includeAudioLevelIndication = enable;
- return _rtpRtcpModule.SetRTPAudioLevelIndicationStatus(enable, ID);
-}
-int
-Channel::GetRTPAudioLevelIndicationStatus(bool& enabled, unsigned char& ID)
-{
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "GetRTPAudioLevelIndicationStatus() => enabled=%d, ID=%u",
- enabled, ID);
- return _rtpRtcpModule.GetRTPAudioLevelIndicationStatus(enabled, ID);
-}
-
-int
-Channel::SetRTCPStatus(bool enable)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetRTCPStatus()");
- if (_rtpRtcpModule.SetRTCPStatus(enable ?
- kRtcpCompound : kRtcpOff) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "SetRTCPStatus() failed to set RTCP status");
- return -1;
- }
- return 0;
-}
-
-int
-Channel::GetRTCPStatus(bool& enabled)
-{
- RTCPMethod method = _rtpRtcpModule.RTCP();
- enabled = (method != kRtcpOff);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "GetRTCPStatus() => enabled=%d", enabled);
- return 0;
-}
-
-int
-Channel::SetRTCP_CNAME(const char cName[256])
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::SetRTCP_CNAME()");
- if (_rtpRtcpModule.SetCNAME(cName) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "SetRTCP_CNAME() failed to set RTCP CNAME");
- return -1;
- }
- return 0;
-}
-
-int
-Channel::GetRTCP_CNAME(char cName[256])
-{
- if (_rtpRtcpModule.CNAME(cName) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "GetRTCP_CNAME() failed to retrieve RTCP CNAME");
- return -1;
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRTCP_CNAME() => cName=%s", cName);
- return 0;
-}
-
-int
-Channel::GetRemoteRTCP_CNAME(char cName[256])
-{
- if (cName == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "GetRemoteRTCP_CNAME() invalid CNAME input buffer");
- return -1;
- }
- WebRtc_Word8 cname[RTCP_CNAME_SIZE];
- const WebRtc_UWord32 remoteSSRC = _rtpRtcpModule.RemoteSSRC();
- if (_rtpRtcpModule.RemoteCNAME(remoteSSRC, cname) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CANNOT_RETRIEVE_CNAME, kTraceError,
- "GetRemoteRTCP_CNAME() failed to retrieve remote RTCP CNAME");
- return -1;
- }
- strcpy(cName, cname);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRemoteRTCP_CNAME() => cName=%s", cName);
- return 0;
-}
-
-int
-Channel::GetRemoteRTCPData(
- unsigned int& NTPHigh,
- unsigned int& NTPLow,
- unsigned int& timestamp,
- unsigned int& playoutTimestamp,
- unsigned int* jitter,
- unsigned short* fractionLost)
-{
- // --- Information from sender info in received Sender Reports
-
- RTCPSenderInfo senderInfo;
- if (_rtpRtcpModule.RemoteRTCPStat(&senderInfo) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "GetRemoteRTCPData() failed to retrieve sender info for remote "
- "side");
- return -1;
- }
-
- // We only utilize 12 out of 20 bytes in the sender info (ignores packet
- // and octet count)
- NTPHigh = senderInfo.NTPseconds;
- NTPLow = senderInfo.NTPfraction;
- timestamp = senderInfo.RTPtimeStamp;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRemoteRTCPData() => NTPHigh=%lu, NTPLow=%lu, "
- "timestamp=%lu",
- NTPHigh, NTPLow, timestamp);
-
- // --- Locally derived information
-
- // This value is updated on each incoming RTCP packet (0 when no packet
- // has been received)
- playoutTimestamp = _playoutTimeStampRTCP;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRemoteRTCPData() => playoutTimestamp=%lu",
- _playoutTimeStampRTCP);
-
- if (NULL != jitter || NULL != fractionLost)
- {
- // Get all RTCP receiver report blocks that have been received on this
- // channel. If we receive RTP packets from a remote source we know the
- // remote SSRC and use the report block from him.
- // Otherwise use the first report block.
- std::vector<RTCPReportBlock> remote_stats;
- if (_rtpRtcpModule.RemoteRTCPStat(&remote_stats) != 0 ||
- remote_stats.empty()) {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRemoteRTCPData() failed to measure statistics due"
- " to lack of received RTP and/or RTCP packets");
- return -1;
- }
-
- WebRtc_UWord32 remoteSSRC = _rtpRtcpModule.RemoteSSRC();
- std::vector<RTCPReportBlock>::const_iterator it = remote_stats.begin();
- for (; it != remote_stats.end(); ++it) {
- if (it->remoteSSRC == remoteSSRC)
- break;
- }
-
- if (it == remote_stats.end()) {
- // If we have not received any RTCP packets from this SSRC it probably
- // means that we have not received any RTP packets.
- // Use the first received report block instead.
- it = remote_stats.begin();
- remoteSSRC = it->remoteSSRC;
- }
-
- if (jitter) {
- *jitter = it->jitter;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRemoteRTCPData() => jitter = %lu", *jitter);
- }
-
- if (fractionLost) {
- *fractionLost = it->fractionLost;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRemoteRTCPData() => fractionLost = %lu",
- *fractionLost);
- }
- }
- return 0;
-}
-
-int
-Channel::SendApplicationDefinedRTCPPacket(const unsigned char subType,
- unsigned int name,
- const char* data,
- unsigned short dataLengthInBytes)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::SendApplicationDefinedRTCPPacket()");
- if (!_sending)
- {
- _engineStatisticsPtr->SetLastError(
- VE_NOT_SENDING, kTraceError,
- "SendApplicationDefinedRTCPPacket() not sending");
- return -1;
- }
- if (NULL == data)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SendApplicationDefinedRTCPPacket() invalid data value");
- return -1;
- }
- if (dataLengthInBytes % 4 != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SendApplicationDefinedRTCPPacket() invalid length value");
- return -1;
- }
- RTCPMethod status = _rtpRtcpModule.RTCP();
- if (status == kRtcpOff)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTCP_ERROR, kTraceError,
- "SendApplicationDefinedRTCPPacket() RTCP is disabled");
- return -1;
- }
-
- // Create and schedule the RTCP APP packet for transmission
- if (_rtpRtcpModule.SetRTCPApplicationSpecificData(
- subType,
- name,
- (const unsigned char*) data,
- dataLengthInBytes) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SEND_ERROR, kTraceError,
- "SendApplicationDefinedRTCPPacket() failed to send RTCP packet");
- return -1;
- }
- return 0;
-}
-
-int
-Channel::GetRTPStatistics(
- unsigned int& averageJitterMs,
- unsigned int& maxJitterMs,
- unsigned int& discardedPackets)
-{
- WebRtc_UWord8 fraction_lost(0);
- WebRtc_UWord32 cum_lost(0);
- WebRtc_UWord32 ext_max(0);
- WebRtc_UWord32 jitter(0);
- WebRtc_UWord32 max_jitter(0);
-
- // The jitter statistics is updated for each received RTP packet and is
- // based on received packets.
- if (_rtpRtcpModule.StatisticsRTP(&fraction_lost,
- &cum_lost,
- &ext_max,
- &jitter,
- &max_jitter) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CANNOT_RETRIEVE_RTP_STAT, kTraceWarning,
- "GetRTPStatistics() failed to read RTP statistics from the "
- "RTP/RTCP module");
- }
-
- const WebRtc_Word32 playoutFrequency =
- _audioCodingModule.PlayoutFrequency();
- if (playoutFrequency > 0)
- {
- // Scale RTP statistics given the current playout frequency
- maxJitterMs = max_jitter / (playoutFrequency / 1000);
- averageJitterMs = jitter / (playoutFrequency / 1000);
- }
-
- discardedPackets = _numberOfDiscardedPackets;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRTPStatistics() => averageJitterMs = %lu, maxJitterMs = %lu,"
- " discardedPackets = %lu)",
- averageJitterMs, maxJitterMs, discardedPackets);
- return 0;
-}
-
-int
-Channel::GetRTPStatistics(CallStatistics& stats)
-{
- WebRtc_UWord8 fraction_lost(0);
- WebRtc_UWord32 cum_lost(0);
- WebRtc_UWord32 ext_max(0);
- WebRtc_UWord32 jitter(0);
- WebRtc_UWord32 max_jitter(0);
-
- // --- Part one of the final structure (four values)
-
- // The jitter statistics is updated for each received RTP packet and is
- // based on received packets.
- if (_rtpRtcpModule.StatisticsRTP(&fraction_lost,
- &cum_lost,
- &ext_max,
- &jitter,
- &max_jitter) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CANNOT_RETRIEVE_RTP_STAT, kTraceWarning,
- "GetRTPStatistics() failed to read RTP statistics from the "
- "RTP/RTCP module");
- }
-
- stats.fractionLost = fraction_lost;
- stats.cumulativeLost = cum_lost;
- stats.extendedMax = ext_max;
- stats.jitterSamples = jitter;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRTPStatistics() => fractionLost=%lu, cumulativeLost=%lu,"
- " extendedMax=%lu, jitterSamples=%li)",
- stats.fractionLost, stats.cumulativeLost, stats.extendedMax,
- stats.jitterSamples);
-
- // --- Part two of the final structure (one value)
-
- WebRtc_UWord16 RTT(0);
- RTCPMethod method = _rtpRtcpModule.RTCP();
- if (method == kRtcpOff)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRTPStatistics() RTCP is disabled => valid RTT "
- "measurements cannot be retrieved");
- } else
- {
- // The remote SSRC will be zero if no RTP packet has been received.
- WebRtc_UWord32 remoteSSRC = _rtpRtcpModule.RemoteSSRC();
- if (remoteSSRC > 0)
- {
- WebRtc_UWord16 avgRTT(0);
- WebRtc_UWord16 maxRTT(0);
- WebRtc_UWord16 minRTT(0);
-
- if (_rtpRtcpModule.RTT(remoteSSRC, &RTT, &avgRTT, &minRTT, &maxRTT)
- != 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRTPStatistics() failed to retrieve RTT from "
- "the RTP/RTCP module");
- }
- } else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRTPStatistics() failed to measure RTT since no "
- "RTP packets have been received yet");
- }
- }
-
- stats.rttMs = static_cast<int> (RTT);
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRTPStatistics() => rttMs=%d", stats.rttMs);
-
- // --- Part three of the final structure (four values)
-
- WebRtc_UWord32 bytesSent(0);
- WebRtc_UWord32 packetsSent(0);
- WebRtc_UWord32 bytesReceived(0);
- WebRtc_UWord32 packetsReceived(0);
-
- if (_rtpRtcpModule.DataCountersRTP(&bytesSent,
- &packetsSent,
- &bytesReceived,
- &packetsReceived) != 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRTPStatistics() failed to retrieve RTP datacounters =>"
- " output will not be complete");
- }
-
- stats.bytesSent = bytesSent;
- stats.packetsSent = packetsSent;
- stats.bytesReceived = bytesReceived;
- stats.packetsReceived = packetsReceived;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRTPStatistics() => bytesSent=%d, packetsSent=%d,"
- " bytesReceived=%d, packetsReceived=%d)",
- stats.bytesSent, stats.packetsSent, stats.bytesReceived,
- stats.packetsReceived);
-
- return 0;
-}
-
-int
-Channel::SetFECStatus(bool enable, int redPayloadtype)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::SetFECStatus()");
-
- CodecInst codec;
-
- // Get default RED settings from the ACM database
- bool foundRED(false);
- const WebRtc_UWord8 nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
- for (int idx = 0; (!foundRED && idx < nSupportedCodecs); idx++)
- {
- _audioCodingModule.Codec(idx, codec);
- if (!STR_CASE_CMP(codec.plname, "RED"))
- {
- foundRED = true;
- }
- }
- if (!foundRED)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CODEC_ERROR, kTraceError,
- "SetFECStatus() RED is not supported");
- return -1;
- }
-
- if (redPayloadtype != -1)
- {
- codec.pltype = redPayloadtype;
- }
-
- if (_audioCodingModule.RegisterSendCodec(codec) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "SetFECStatus() RED registration in ACM module failed");
- return -1;
- }
- if (_rtpRtcpModule.SetSendREDPayloadType(codec.pltype) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "SetFECStatus() RED registration in RTP/RTCP module failed");
- return -1;
- }
- if (_audioCodingModule.SetFECStatus(enable) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "SetFECStatus() failed to set FEC state in the ACM");
- return -1;
- }
- return 0;
-}
-
-int
-Channel::GetFECStatus(bool& enabled, int& redPayloadtype)
-{
- enabled = _audioCodingModule.FECStatus();
- if (enabled)
- {
- WebRtc_Word8 payloadType(0);
- if (_rtpRtcpModule.SendREDPayloadType(payloadType) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "GetFECStatus() failed to retrieve RED PT from RTP/RTCP "
- "module");
- return -1;
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetFECStatus() => enabled=%d, redPayloadtype=%d",
- enabled, redPayloadtype);
- return 0;
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetFECStatus() => enabled=%d", enabled);
- return 0;
-}
-
-int
-Channel::SetRTPKeepaliveStatus(bool enable,
- unsigned char unknownPayloadType,
- int deltaTransmitTimeSeconds)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::SetRTPKeepaliveStatus()");
- if (_sending)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_SENDING, kTraceError,
- "SetRTPKeepaliveStatus() already sending");
- return -1;
- }
- if (_rtpRtcpModule.SetRTPKeepaliveStatus(
- enable,
- unknownPayloadType,
- 1000 * deltaTransmitTimeSeconds) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "SetRTPKeepaliveStatus() failed to set RTP keepalive status");
- return -1;
- }
- return 0;
-}
-
-int
-Channel::GetRTPKeepaliveStatus(bool& enabled,
- unsigned char& unknownPayloadType,
- int& deltaTransmitTimeSeconds)
-{
- bool onOff(false);
- WebRtc_Word8 payloadType(0);
- WebRtc_UWord16 deltaTransmitTimeMS(0);
- if (_rtpRtcpModule.RTPKeepaliveStatus(&onOff, &payloadType,
- &deltaTransmitTimeMS) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "GetRTPKeepaliveStatus() failed to retrieve RTP keepalive status");
- return -1;
- }
- enabled = onOff;
- unknownPayloadType = payloadType;
- deltaTransmitTimeSeconds = static_cast<int> (deltaTransmitTimeMS / 1000);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "GetRTPKeepaliveStatus() => enabled=%d, "
- "unknownPayloadType=%u, deltaTransmitTimeSeconds=%d",
- enabled, unknownPayloadType, deltaTransmitTimeSeconds);
- return 0;
-}
-
-int
-Channel::StartRTPDump(const char fileNameUTF8[1024],
- RTPDirections direction)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StartRTPDump()");
- if ((direction != kRtpIncoming) && (direction != kRtpOutgoing))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRTPDump() invalid RTP direction");
- return -1;
- }
- RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
- &_rtpDumpIn : &_rtpDumpOut;
- if (rtpDumpPtr == NULL)
- {
- assert(false);
- return -1;
- }
- if (rtpDumpPtr->IsActive())
- {
- rtpDumpPtr->Stop();
- }
- if (rtpDumpPtr->Start(fileNameUTF8) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartRTPDump() failed to create file");
- return -1;
- }
- return 0;
-}
-
-int
-Channel::StopRTPDump(RTPDirections direction)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::StopRTPDump()");
- if ((direction != kRtpIncoming) && (direction != kRtpOutgoing))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StopRTPDump() invalid RTP direction");
- return -1;
- }
- RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
- &_rtpDumpIn : &_rtpDumpOut;
- if (rtpDumpPtr == NULL)
- {
- assert(false);
- return -1;
- }
- if (!rtpDumpPtr->IsActive())
- {
- return 0;
- }
- return rtpDumpPtr->Stop();
-}
-
-bool
-Channel::RTPDumpIsActive(RTPDirections direction)
-{
- if ((direction != kRtpIncoming) &&
- (direction != kRtpOutgoing))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "RTPDumpIsActive() invalid RTP direction");
- return false;
- }
- RtpDump* rtpDumpPtr = (direction == kRtpIncoming) ?
- &_rtpDumpIn : &_rtpDumpOut;
- return rtpDumpPtr->IsActive();
-}
-
-int
-Channel::InsertExtraRTPPacket(unsigned char payloadType,
- bool markerBit,
- const char* payloadData,
- unsigned short payloadSize)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, _channelId),
- "Channel::InsertExtraRTPPacket()");
- if (payloadType > 127)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_PLTYPE, kTraceError,
- "InsertExtraRTPPacket() invalid payload type");
- return -1;
- }
- if (payloadData == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "InsertExtraRTPPacket() invalid payload data");
- return -1;
- }
- if (payloadSize > _rtpRtcpModule.MaxDataPayloadLength())
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "InsertExtraRTPPacket() invalid payload size");
- return -1;
- }
- if (!_sending)
- {
- _engineStatisticsPtr->SetLastError(
- VE_NOT_SENDING, kTraceError,
- "InsertExtraRTPPacket() not sending");
- return -1;
- }
-
- // Create extra RTP packet by calling RtpRtcp::SendOutgoingData().
- // Transport::SendPacket() will be called by the module when the RTP packet
- // is created.
- // The call to SendOutgoingData() does *not* modify the timestamp and
- // payloadtype to ensure that the RTP module generates a valid RTP packet
- // (user might utilize a non-registered payload type).
- // The marker bit and payload type will be replaced just before the actual
- // transmission, i.e., the actual modification is done *after* the RTP
- // module has delivered its RTP packet back to the VoE.
- // We will use the stored values above when the packet is modified
- // (see Channel::SendPacket()).
-
- _extraPayloadType = payloadType;
- _extraMarkerBit = markerBit;
- _insertExtraRTPPacket = true;
-
- if (_rtpRtcpModule.SendOutgoingData(kAudioFrameSpeech,
- _lastPayloadType,
- _lastLocalTimeStamp,
- (const WebRtc_UWord8*) payloadData,
- payloadSize) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "InsertExtraRTPPacket() failed to send extra RTP packet");
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_UWord32
-Channel::Demultiplex(const AudioFrame& audioFrame)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::Demultiplex()");
- _audioFrame = audioFrame;
- _audioFrame._id = _channelId;
- return 0;
-}
-
-WebRtc_UWord32
-Channel::PrepareEncodeAndSend(int mixingFrequency)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::PrepareEncodeAndSend()");
-
- if (_audioFrame._payloadDataLengthInSamples == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::PrepareEncodeAndSend() invalid audio frame");
- return -1;
- }
-
- if (_inputFilePlaying)
- {
- MixOrReplaceAudioWithFile(mixingFrequency);
- }
-
- if (_mute)
- {
- AudioFrameOperations::Mute(_audioFrame);
- }
-
- if (_inputExternalMedia)
- {
- CriticalSectionScoped cs(_callbackCritSect);
- const bool isStereo = (_audioFrame._audioChannel == 2);
- if (_inputExternalMediaCallbackPtr)
- {
- _inputExternalMediaCallbackPtr->Process(
- _channelId,
- kRecordingPerChannel,
- (WebRtc_Word16*)_audioFrame._payloadData,
- _audioFrame._payloadDataLengthInSamples,
- _audioFrame._frequencyInHz,
- isStereo);
- }
- }
-
- InsertInbandDtmfTone();
-
- if (_includeAudioLevelIndication)
- {
- assert(_rtpAudioProc.get() != NULL);
-
- // Check if settings need to be updated.
- if (_rtpAudioProc->sample_rate_hz() != _audioFrame._frequencyInHz)
- {
- if (_rtpAudioProc->set_sample_rate_hz(_audioFrame._frequencyInHz) !=
- AudioProcessing::kNoError)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "Error setting AudioProcessing sample rate");
- return -1;
- }
- }
-
- if (_rtpAudioProc->num_input_channels() != _audioFrame._audioChannel)
- {
- if (_rtpAudioProc->set_num_channels(_audioFrame._audioChannel,
- _audioFrame._audioChannel)
- != AudioProcessing::kNoError)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "Error setting AudioProcessing channels");
- return -1;
- }
- }
-
- // Performs level analysis only; does not affect the signal.
- _rtpAudioProc->ProcessStream(&_audioFrame);
- }
-
- return 0;
-}
-
-WebRtc_UWord32
-Channel::EncodeAndSend()
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::EncodeAndSend()");
-
- assert(_audioFrame._audioChannel <= 2);
- if (_audioFrame._payloadDataLengthInSamples == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::EncodeAndSend() invalid audio frame");
- return -1;
- }
-
- _audioFrame._id = _channelId;
-
- // --- Add 10ms of raw (PCM) audio data to the encoder @ 32kHz.
-
- // The ACM resamples internally.
- _audioFrame._timeStamp = _timeStamp;
- if (_audioCodingModule.Add10MsData((AudioFrame&)_audioFrame) != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::EncodeAndSend() ACM encoding failed");
- return -1;
- }
-
- _timeStamp += _audioFrame._payloadDataLengthInSamples;
-
- // --- Encode if complete frame is ready
-
- // This call will trigger AudioPacketizationCallback::SendData if encoding
- // is done and payload is ready for packetization and transmission.
- return _audioCodingModule.Process();
-}
-
-int Channel::RegisterExternalMediaProcessing(
- ProcessingTypes type,
- VoEMediaProcess& processObject)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::RegisterExternalMediaProcessing()");
-
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (kPlaybackPerChannel == type)
- {
- if (_outputExternalMediaCallbackPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "Channel::RegisterExternalMediaProcessing() "
- "output external media already enabled");
- return -1;
- }
- _outputExternalMediaCallbackPtr = &processObject;
- _outputExternalMedia = true;
- }
- else if (kRecordingPerChannel == type)
- {
- if (_inputExternalMediaCallbackPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "Channel::RegisterExternalMediaProcessing() "
- "output external media already enabled");
- return -1;
- }
- _inputExternalMediaCallbackPtr = &processObject;
- _inputExternalMedia = true;
- }
- return 0;
-}
-
-int Channel::DeRegisterExternalMediaProcessing(ProcessingTypes type)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::DeRegisterExternalMediaProcessing()");
-
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (kPlaybackPerChannel == type)
- {
- if (!_outputExternalMediaCallbackPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "Channel::DeRegisterExternalMediaProcessing() "
- "output external media already disabled");
- return 0;
- }
- _outputExternalMedia = false;
- _outputExternalMediaCallbackPtr = NULL;
- }
- else if (kRecordingPerChannel == type)
- {
- if (!_inputExternalMediaCallbackPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "Channel::DeRegisterExternalMediaProcessing() "
- "input external media already disabled");
- return 0;
- }
- _inputExternalMedia = false;
- _inputExternalMediaCallbackPtr = NULL;
- }
-
- return 0;
-}
-
-int
-Channel::ResetRTCPStatistics()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::ResetRTCPStatistics()");
- WebRtc_UWord32 remoteSSRC(0);
- remoteSSRC = _rtpRtcpModule.RemoteSSRC();
- return _rtpRtcpModule.ResetRTT(remoteSSRC);
-}
-
-int
-Channel::GetRoundTripTimeSummary(StatVal& delaysMs) const
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetRoundTripTimeSummary()");
- // Override default module outputs for the case when RTCP is disabled.
- // This is done to ensure that we are backward compatible with the
- // VoiceEngine where we did not use RTP/RTCP module.
- if (!_rtpRtcpModule.RTCP())
- {
- delaysMs.min = -1;
- delaysMs.max = -1;
- delaysMs.average = -1;
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetRoundTripTimeSummary() RTCP is disabled =>"
- " valid RTT measurements cannot be retrieved");
- return 0;
- }
-
- WebRtc_UWord32 remoteSSRC;
- WebRtc_UWord16 RTT;
- WebRtc_UWord16 avgRTT;
- WebRtc_UWord16 maxRTT;
- WebRtc_UWord16 minRTT;
- // The remote SSRC will be zero if no RTP packet has been received.
- remoteSSRC = _rtpRtcpModule.RemoteSSRC();
- if (remoteSSRC == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetRoundTripTimeSummary() unable to measure RTT"
- " since no RTP packet has been received yet");
- }
-
- // Retrieve RTT statistics from the RTP/RTCP module for the specified
- // channel and SSRC. The SSRC is required to parse out the correct source
- // in conference scenarios.
- if (_rtpRtcpModule.RTT(remoteSSRC, &RTT, &avgRTT, &minRTT,&maxRTT) != 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
- "GetRoundTripTimeSummary unable to retrieve RTT values"
- " from the RTCP layer");
- delaysMs.min = -1; delaysMs.max = -1; delaysMs.average = -1;
- }
- else
- {
- delaysMs.min = minRTT;
- delaysMs.max = maxRTT;
- delaysMs.average = avgRTT;
- }
- return 0;
-}
-
-int
-Channel::GetNetworkStatistics(NetworkStatistics& stats)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetNetworkStatistics()");
- return _audioCodingModule.NetworkStatistics(
- (ACMNetworkStatistics &)stats);
-}
-
-int
-Channel::GetDelayEstimate(int& delayMs) const
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetDelayEstimate()");
- delayMs = (_averageDelayMs + 5) / 10 + _recPacketDelayMs;
- return 0;
-}
-
-int
-Channel::SetMinimumPlayoutDelay(int delayMs)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetMinimumPlayoutDelay()");
- if ((delayMs < kVoiceEngineMinMinPlayoutDelayMs) ||
- (delayMs > kVoiceEngineMaxMinPlayoutDelayMs))
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetMinimumPlayoutDelay() invalid min delay");
- return -1;
- }
- if (_audioCodingModule.SetMinimumPlayoutDelay(delayMs) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_AUDIO_CODING_MODULE_ERROR, kTraceError,
- "SetMinimumPlayoutDelay() failed to set min playout delay");
- return -1;
- }
- return 0;
-}
-
-int
-Channel::GetPlayoutTimestamp(unsigned int& timestamp)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetPlayoutTimestamp()");
- WebRtc_UWord32 playoutTimestamp(0);
- if (GetPlayoutTimeStamp(playoutTimestamp) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CANNOT_RETRIEVE_VALUE, kTraceError,
- "GetPlayoutTimestamp() failed to retrieve timestamp");
- return -1;
- }
- timestamp = playoutTimestamp;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice,
- VoEId(_instanceId,_channelId),
- "GetPlayoutTimestamp() => timestamp=%u", timestamp);
- return 0;
-}
-
-int
-Channel::SetInitTimestamp(unsigned int timestamp)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetInitTimestamp()");
- if (_sending)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SENDING, kTraceError, "SetInitTimestamp() already sending");
- return -1;
- }
- if (_rtpRtcpModule.SetStartTimestamp(timestamp) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "SetInitTimestamp() failed to set timestamp");
- return -1;
- }
- return 0;
-}
-
-int
-Channel::SetInitSequenceNumber(short sequenceNumber)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::SetInitSequenceNumber()");
- if (_sending)
- {
- _engineStatisticsPtr->SetLastError(
- VE_SENDING, kTraceError,
- "SetInitSequenceNumber() already sending");
- return -1;
- }
- if (_rtpRtcpModule.SetSequenceNumber(sequenceNumber) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_RTP_RTCP_MODULE_ERROR, kTraceError,
- "SetInitSequenceNumber() failed to set sequence number");
- return -1;
- }
- return 0;
-}
-
-int
-Channel::GetRtpRtcp(RtpRtcp* &rtpRtcpModule) const
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetRtpRtcp()");
- rtpRtcpModule = &_rtpRtcpModule;
- return 0;
-}
-
-WebRtc_Word32
-Channel::MixOrReplaceAudioWithFile(const int mixingFrequency)
-{
- WebRtc_Word16 fileBuffer[320];
- WebRtc_UWord32 fileSamples(0);
-
- {
- CriticalSectionScoped cs(_fileCritSect);
-
- if (_inputFilePlayerPtr == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "Channel::MixOrReplaceAudioWithFile() fileplayer"
- " doesnt exist");
- return -1;
- }
-
- if (_inputFilePlayerPtr->Get10msAudioFromFile(fileBuffer,
- fileSamples,
- mixingFrequency) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "Channel::MixOrReplaceAudioWithFile() file mixing "
- "failed");
- return -1;
- }
- if (fileSamples == 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "Channel::MixOrReplaceAudioWithFile() file is ended");
- return 0;
- }
- }
-
- assert(_audioFrame._payloadDataLengthInSamples == fileSamples);
-
- if (_mixFileWithMicrophone)
- {
- Utility::MixWithSat(_audioFrame._payloadData,
- fileBuffer,
- (WebRtc_UWord16)fileSamples);
- }
- else
- {
- // replace ACM audio with file
- _audioFrame.UpdateFrame(_channelId,
- -1,
- fileBuffer,
- (WebRtc_UWord16)fileSamples,
- mixingFrequency,
- AudioFrame::kNormalSpeech,
- AudioFrame::kVadUnknown,
- 1);
-
- }
- return 0;
-}
-
-WebRtc_Word32
-Channel::MixAudioWithFile(AudioFrame& audioFrame,
- const int mixingFrequency)
-{
- assert(mixingFrequency <= 32000);
-
- WebRtc_Word16 fileBuffer[640];
- WebRtc_UWord32 fileSamples(0);
-
- {
- CriticalSectionScoped cs(_fileCritSect);
-
- if (_outputFilePlayerPtr == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "Channel::MixAudioWithFile() file mixing failed");
- return -1;
- }
-
- // We should get the frequency we ask for.
- if (_outputFilePlayerPtr->Get10msAudioFromFile(fileBuffer,
- fileSamples,
- mixingFrequency) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "Channel::MixAudioWithFile() file mixing failed");
- return -1;
- }
- }
-
- if (audioFrame._payloadDataLengthInSamples == fileSamples)
- {
- // In case the incoming stream is stereo and file stream is mono,
- // turn the file stream into stereo.
- // TODO(xians): remove the code when FilePlayer supports real stereo.
- if (audioFrame._audioChannel == 2)
- {
- // The mono file stream is copied to be stereo.
- WebRtc_Word16* FileBufferCopy = new WebRtc_Word16[fileSamples];
- memcpy(FileBufferCopy, fileBuffer,
- sizeof(WebRtc_Word16) * fileSamples);
- for (unsigned int i = 0; i < fileSamples; i++)
- {
- fileBuffer[2*i] = FileBufferCopy[i];
- fileBuffer[2*i+1] = FileBufferCopy[i];
- }
- fileSamples = 2*fileSamples;
- delete [] FileBufferCopy;
- }
-
- // Mix the incoming stream and file stream.
- Utility::MixWithSat(audioFrame._payloadData,
- fileBuffer,
- (WebRtc_UWord16)fileSamples);
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::MixAudioWithFile() _payloadDataLengthInSamples(%d) != "
- "fileSamples(%d)",
- audioFrame._payloadDataLengthInSamples, fileSamples);
- return -1;
- }
-
- return 0;
-}
-
-int
-Channel::InsertInbandDtmfTone()
-{
- // Check if we should start a new tone.
- if (_inbandDtmfQueue.PendingDtmf() &&
- !_inbandDtmfGenerator.IsAddingTone() &&
- _inbandDtmfGenerator.DelaySinceLastTone() >
- kMinTelephoneEventSeparationMs)
- {
- WebRtc_Word8 eventCode(0);
- WebRtc_UWord16 lengthMs(0);
- WebRtc_UWord8 attenuationDb(0);
-
- eventCode = _inbandDtmfQueue.NextDtmf(&lengthMs, &attenuationDb);
- _inbandDtmfGenerator.AddTone(eventCode, lengthMs, attenuationDb);
- if (_playInbandDtmfEvent)
- {
- // Add tone to output mixer using a reduced length to minimize
- // risk of echo.
- _outputMixerPtr->PlayDtmfTone(eventCode, lengthMs - 80,
- attenuationDb);
- }
- }
-
- if (_inbandDtmfGenerator.IsAddingTone())
- {
- WebRtc_UWord16 frequency(0);
- _inbandDtmfGenerator.GetSampleRate(frequency);
-
- if (frequency != _audioFrame._frequencyInHz)
- {
- // Update sample rate of Dtmf tone since the mixing frequency
- // has changed.
- _inbandDtmfGenerator.SetSampleRate(
- (WebRtc_UWord16) (_audioFrame._frequencyInHz));
- // Reset the tone to be added taking the new sample rate into
- // account.
- _inbandDtmfGenerator.ResetTone();
- }
-
- WebRtc_Word16 toneBuffer[320];
- WebRtc_UWord16 toneSamples(0);
- // Get 10ms tone segment and set time since last tone to zero
- if (_inbandDtmfGenerator.Get10msTone(toneBuffer, toneSamples) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "Channel::EncodeAndSend() inserting Dtmf failed");
- return -1;
- }
-
- // Replace mixed audio with DTMF tone.
- for (int sample = 0;
- sample < _audioFrame._payloadDataLengthInSamples;
- sample++)
- {
- for (int channel = 0;
- channel < _audioFrame._audioChannel;
- channel++)
- {
- _audioFrame._payloadData[sample * _audioFrame._audioChannel + channel] =
- toneBuffer[sample];
- }
- }
-
- assert(_audioFrame._payloadDataLengthInSamples == toneSamples);
- } else
- {
- // Add 10ms to "delay-since-last-tone" counter
- _inbandDtmfGenerator.UpdateDelaySinceLastTone();
- }
- return 0;
-}
-
-WebRtc_Word32
-Channel::GetPlayoutTimeStamp(WebRtc_UWord32& playoutTimestamp)
-{
- WebRtc_UWord32 timestamp(0);
- CodecInst currRecCodec;
-
- if (_audioCodingModule.PlayoutTimestamp(timestamp) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetPlayoutTimeStamp() failed to read playout"
- " timestamp from the ACM");
- return -1;
- }
-
- WebRtc_UWord16 delayMS(0);
- if (_audioDeviceModulePtr->PlayoutDelay(&delayMS) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetPlayoutTimeStamp() failed to read playout"
- " delay from the ADM");
- return -1;
- }
-
- WebRtc_Word32 playoutFrequency = _audioCodingModule.PlayoutFrequency();
- if (_audioCodingModule.ReceiveCodec(currRecCodec) == 0)
- {
- if (STR_CASE_CMP("G722", currRecCodec.plname) == 0)
- {
- playoutFrequency = 8000;
- }
- }
- timestamp -= (delayMS * (playoutFrequency/1000));
-
- playoutTimestamp = timestamp;
-
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::GetPlayoutTimeStamp() => playoutTimestamp = %lu",
- playoutTimestamp);
- return 0;
-}
-
-void
-Channel::ResetDeadOrAliveCounters()
-{
- _countDeadDetections = 0;
- _countAliveDetections = 0;
-}
-
-void
-Channel::UpdateDeadOrAliveCounters(bool alive)
-{
- if (alive)
- _countAliveDetections++;
- else
- _countDeadDetections++;
-}
-
-int
-Channel::GetDeadOrAliveCounters(int& countDead, int& countAlive) const
-{
- bool enabled;
- WebRtc_UWord8 timeSec;
-
- _rtpRtcpModule.PeriodicDeadOrAliveStatus(enabled, timeSec);
- if (!enabled)
- return (-1);
-
- countDead = static_cast<int> (_countDeadDetections);
- countAlive = static_cast<int> (_countAliveDetections);
- return 0;
-}
-
-WebRtc_Word32
-Channel::SendPacketRaw(const void *data, int len, bool RTCP)
-{
- if (_transportPtr == NULL)
- {
- return -1;
- }
- if (!RTCP)
- {
- return _transportPtr->SendPacket(_channelId, data, len);
- }
- else
- {
- return _transportPtr->SendRTCPPacket(_channelId, data, len);
- }
-}
-
-WebRtc_Word32
-Channel::UpdatePacketDelay(const WebRtc_UWord32 timestamp,
- const WebRtc_UWord16 sequenceNumber)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::UpdatePacketDelay(timestamp=%lu, sequenceNumber=%u)",
- timestamp, sequenceNumber);
-
- WebRtc_Word32 rtpReceiveFrequency(0);
-
- // Get frequency of last received payload
- rtpReceiveFrequency = _audioCodingModule.ReceiveFrequency();
-
- CodecInst currRecCodec;
- if (_audioCodingModule.ReceiveCodec(currRecCodec) == 0)
- {
- if (STR_CASE_CMP("G722", currRecCodec.plname) == 0)
- {
- // Even though the actual sampling rate for G.722 audio is
- // 16,000 Hz, the RTP clock rate for the G722 payload format is
- // 8,000 Hz because that value was erroneously assigned in
- // RFC 1890 and must remain unchanged for backward compatibility.
- rtpReceiveFrequency = 8000;
- }
- }
-
- const WebRtc_UWord32 timeStampDiff = timestamp - _playoutTimeStampRTP;
- WebRtc_UWord32 timeStampDiffMs(0);
-
- if (timeStampDiff > 0)
- {
- switch (rtpReceiveFrequency)
- {
- case 8000:
- timeStampDiffMs = timeStampDiff >> 3;
- break;
- case 16000:
- timeStampDiffMs = timeStampDiff >> 4;
- break;
- case 32000:
- timeStampDiffMs = timeStampDiff >> 5;
- break;
- default:
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, _channelId),
- "Channel::UpdatePacketDelay() invalid sample "
- "rate");
- timeStampDiffMs = 0;
- return -1;
- }
- if (timeStampDiffMs > 5000)
- {
- timeStampDiffMs = 0;
- }
-
- if (_averageDelayMs == 0)
- {
- _averageDelayMs = timeStampDiffMs;
- }
- else
- {
- // Filter average delay value using exponential filter (alpha is
- // 7/8). We derive 10*_averageDelayMs here (reduces risk of
- // rounding error) and compensate for it in GetDelayEstimate()
- // later. Adding 4/8 results in correct rounding.
- _averageDelayMs = ((_averageDelayMs*7 + 10*timeStampDiffMs + 4)>>3);
- }
-
- if (sequenceNumber - _previousSequenceNumber == 1)
- {
- WebRtc_UWord16 packetDelayMs = 0;
- switch (rtpReceiveFrequency)
- {
- case 8000:
- packetDelayMs = (WebRtc_UWord16)(
- (timestamp - _previousTimestamp) >> 3);
- break;
- case 16000:
- packetDelayMs = (WebRtc_UWord16)(
- (timestamp - _previousTimestamp) >> 4);
- break;
- case 32000:
- packetDelayMs = (WebRtc_UWord16)(
- (timestamp - _previousTimestamp) >> 5);
- break;
- }
-
- if (packetDelayMs >= 10 && packetDelayMs <= 60)
- _recPacketDelayMs = packetDelayMs;
- }
- }
-
- _previousSequenceNumber = sequenceNumber;
- _previousTimestamp = timestamp;
-
- return 0;
-}
-
-void
-Channel::RegisterReceiveCodecsToRTPModule()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::RegisterReceiveCodecsToRTPModule()");
-
-
- CodecInst codec;
- const WebRtc_UWord8 nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
-
- for (int idx = 0; idx < nSupportedCodecs; idx++)
- {
- // Open up the RTP/RTCP receiver for all supported codecs
- if ((_audioCodingModule.Codec(idx, codec) == -1) ||
- (_rtpRtcpModule.RegisterReceivePayload(codec) == -1))
- {
- WEBRTC_TRACE(
- kTraceWarning,
- kTraceVoice,
- VoEId(_instanceId, _channelId),
- "Channel::RegisterReceiveCodecsToRTPModule() unable"
- " to register %s (%d/%d/%d/%d) to RTP/RTCP receiver",
- codec.plname, codec.pltype, codec.plfreq,
- codec.channels, codec.rate);
- }
- else
- {
- WEBRTC_TRACE(
- kTraceInfo,
- kTraceVoice,
- VoEId(_instanceId, _channelId),
- "Channel::RegisterReceiveCodecsToRTPModule() %s "
- "(%d/%d/%d/%d) has been added to the RTP/RTCP "
- "receiver",
- codec.plname, codec.pltype, codec.plfreq,
- codec.channels, codec.rate);
- }
- }
-}
-
-int
-Channel::ApmProcessRx(AudioFrame& audioFrame)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,_channelId),
- "Channel::ApmProcessRx()");
-
- // Reset the APM frequency if the frequency has changed
- if (_rxAudioProcessingModulePtr->sample_rate_hz() !=
- audioFrame._frequencyInHz)
- {
- if (_rxAudioProcessingModulePtr->set_sample_rate_hz(
- audioFrame._frequencyInHz) != 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
- "AudioProcessingModule::set_sample_rate_hz("
- "_frequencyInHz=%u) => error",
- _audioFrame._frequencyInHz);
- }
- }
-
- if (_rxAudioProcessingModulePtr->ProcessStream(&audioFrame) != 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
- "AudioProcessingModule::ProcessStream() => error");
- }
-
- return 0;
-}
-
-} // namespace voe
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/channel.h b/trunk/src/voice_engine/main/source/channel.h
deleted file mode 100644
index c9d5b82..0000000
--- a/trunk/src/voice_engine/main/source/channel.h
+++ /dev/null
@@ -1,659 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H
-#define WEBRTC_VOICE_ENGINE_CHANNEL_H
-
-#include "audio_coding_module.h"
-#include "audio_conference_mixer_defines.h"
-#include "common_types.h"
-#include "dtmf_inband.h"
-#include "dtmf_inband_queue.h"
-#include "file_player.h"
-#include "file_recorder.h"
-#include "level_indicator.h"
-#include "resampler.h"
-#include "rtp_rtcp.h"
-#include "scoped_ptr.h"
-#include "shared_data.h"
-#include "voe_audio_processing.h"
-#include "voe_network.h"
-#include "voice_engine_defines.h"
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
-#include "udp_transport.h"
-#endif
-#ifdef WEBRTC_SRTP
-#include "SrtpModule.h"
-#endif
-#ifdef WEBRTC_DTMF_DETECTION
-#include "voe_dtmf.h" // TelephoneEventDetectionMethods, TelephoneEventObserver
-#endif
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-class ProcessThread;
-class AudioDeviceModule;
-class RtpRtcp;
-class FileWrapper;
-class RtpDump;
-class VoiceEngineObserver;
-class VoEMediaProcess;
-class VoERTPObserver;
-class VoERTCPObserver;
-
-struct CallStatistics;
-
-namespace voe
-{
-class Statistics;
-class TransmitMixer;
-class OutputMixer;
-
-
-class Channel:
- public RtpData,
- public RtpFeedback,
- public RtcpFeedback,
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- public UdpTransportData, // receiving packet from sockets
-#endif
- public FileCallback, // receiving notification from file player & recorder
- public Transport,
- public RtpAudioFeedback,
- public AudioPacketizationCallback, // receive encoded packets from the ACM
- public ACMVADCallback, // receive voice activity from the ACM
-#ifdef WEBRTC_DTMF_DETECTION
- public AudioCodingFeedback, // inband Dtmf detection in the ACM
-#endif
- public MixerParticipant // supplies output mixer with audio frames
-{
-public:
- enum {KNumSocketThreads = 1};
- enum {KNumberOfSocketBuffers = 8};
-public:
- virtual ~Channel();
- static WebRtc_Word32 CreateChannel(Channel*& channel,
- const WebRtc_Word32 channelId,
- const WebRtc_UWord32 instanceId);
- Channel(const WebRtc_Word32 channelId, const WebRtc_UWord32 instanceId);
- WebRtc_Word32 Init();
- WebRtc_Word32 SetEngineInformation(
- Statistics& engineStatistics,
- OutputMixer& outputMixer,
- TransmitMixer& transmitMixer,
- ProcessThread& moduleProcessThread,
- AudioDeviceModule& audioDeviceModule,
- VoiceEngineObserver* voiceEngineObserver,
- CriticalSectionWrapper* callbackCritSect);
- WebRtc_Word32 UpdateLocalTimeStamp();
-
-public:
- // API methods
-
- // VoEBase
- WebRtc_Word32 StartPlayout();
- WebRtc_Word32 StopPlayout();
- WebRtc_Word32 StartSend();
- WebRtc_Word32 StopSend();
- WebRtc_Word32 StartReceiving();
- WebRtc_Word32 StopReceiving();
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- WebRtc_Word32 SetLocalReceiver(const WebRtc_UWord16 rtpPort,
- const WebRtc_UWord16 rtcpPort,
- const WebRtc_Word8 ipAddr[64],
- const WebRtc_Word8 multicastIpAddr[64]);
- WebRtc_Word32 GetLocalReceiver(int& port, int& RTCPport, char ipAddr[]);
- WebRtc_Word32 SetSendDestination(const WebRtc_UWord16 rtpPort,
- const WebRtc_Word8 ipAddr[64],
- const int sourcePort,
- const WebRtc_UWord16 rtcpPort);
- WebRtc_Word32 GetSendDestination(int& port, char ipAddr[64],
- int& sourcePort, int& RTCPport);
-#endif
- WebRtc_Word32 SetNetEQPlayoutMode(NetEqModes mode);
- WebRtc_Word32 GetNetEQPlayoutMode(NetEqModes& mode);
- WebRtc_Word32 SetNetEQBGNMode(NetEqBgnModes mode);
- WebRtc_Word32 GetNetEQBGNMode(NetEqBgnModes& mode);
- WebRtc_Word32 SetOnHoldStatus(bool enable, OnHoldModes mode);
- WebRtc_Word32 GetOnHoldStatus(bool& enabled, OnHoldModes& mode);
- WebRtc_Word32 RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
- WebRtc_Word32 DeRegisterVoiceEngineObserver();
-
- // VoECodec
- WebRtc_Word32 GetSendCodec(CodecInst& codec);
- WebRtc_Word32 GetRecCodec(CodecInst& codec);
- WebRtc_Word32 SetSendCodec(const CodecInst& codec);
- WebRtc_Word32 SetVADStatus(bool enableVAD, ACMVADMode mode,
- bool disableDTX);
- WebRtc_Word32 GetVADStatus(bool& enabledVAD, ACMVADMode& mode,
- bool& disabledDTX);
- WebRtc_Word32 SetRecPayloadType(const CodecInst& codec);
- WebRtc_Word32 GetRecPayloadType(CodecInst& codec);
- WebRtc_Word32 SetAMREncFormat(AmrMode mode);
- WebRtc_Word32 SetAMRDecFormat(AmrMode mode);
- WebRtc_Word32 SetAMRWbEncFormat(AmrMode mode);
- WebRtc_Word32 SetAMRWbDecFormat(AmrMode mode);
- WebRtc_Word32 SetSendCNPayloadType(int type, PayloadFrequencies frequency);
- WebRtc_Word32 SetISACInitTargetRate(int rateBps, bool useFixedFrameSize);
- WebRtc_Word32 SetISACMaxRate(int rateBps);
- WebRtc_Word32 SetISACMaxPayloadSize(int sizeBytes);
-
- // VoENetwork
- WebRtc_Word32 RegisterExternalTransport(Transport& transport);
- WebRtc_Word32 DeRegisterExternalTransport();
- WebRtc_Word32 ReceivedRTPPacket(const WebRtc_Word8* data,
- WebRtc_Word32 length);
- WebRtc_Word32 ReceivedRTCPPacket(const WebRtc_Word8* data,
- WebRtc_Word32 length);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- WebRtc_Word32 GetSourceInfo(int& rtpPort, int& rtcpPort, char ipAddr[64]);
- WebRtc_Word32 EnableIPv6();
- bool IPv6IsEnabled() const;
- WebRtc_Word32 SetSourceFilter(int rtpPort, int rtcpPort,
- const char ipAddr[64]);
- WebRtc_Word32 GetSourceFilter(int& rtpPort, int& rtcpPort, char ipAddr[64]);
- WebRtc_Word32 SetSendTOS(int DSCP, int priority, bool useSetSockopt);
- WebRtc_Word32 GetSendTOS(int &DSCP, int& priority, bool &useSetSockopt);
-#if defined(_WIN32)
- WebRtc_Word32 SetSendGQoS(bool enable, int serviceType, int overrideDSCP);
- WebRtc_Word32 GetSendGQoS(bool &enabled, int &serviceType,
- int &overrideDSCP);
-#endif
-#endif
- WebRtc_Word32 SetPacketTimeoutNotification(bool enable, int timeoutSeconds);
- WebRtc_Word32 GetPacketTimeoutNotification(bool& enabled,
- int& timeoutSeconds);
- WebRtc_Word32 RegisterDeadOrAliveObserver(VoEConnectionObserver& observer);
- WebRtc_Word32 DeRegisterDeadOrAliveObserver();
- WebRtc_Word32 SetPeriodicDeadOrAliveStatus(bool enable,
- int sampleTimeSeconds);
- WebRtc_Word32 GetPeriodicDeadOrAliveStatus(bool& enabled,
- int& sampleTimeSeconds);
- WebRtc_Word32 SendUDPPacket(const void* data, unsigned int length,
- int& transmittedBytes, bool useRtcpSocket);
-
- // VoEFile
- int StartPlayingFileLocally(const char* fileName, const bool loop,
- const FileFormats format,
- const int startPosition,
- const float volumeScaling,
- const int stopPosition,
- const CodecInst* codecInst);
- int StartPlayingFileLocally(InStream* stream, const FileFormats format,
- const int startPosition,
- const float volumeScaling,
- const int stopPosition,
- const CodecInst* codecInst);
- int StopPlayingFileLocally();
- int IsPlayingFileLocally() const;
- int ScaleLocalFilePlayout(const float scale);
- int GetLocalPlayoutPosition(int& positionMs);
- int StartPlayingFileAsMicrophone(const char* fileName, const bool loop,
- const FileFormats format,
- const int startPosition,
- const float volumeScaling,
- const int stopPosition,
- const CodecInst* codecInst);
- int StartPlayingFileAsMicrophone(InStream* stream,
- const FileFormats format,
- const int startPosition,
- const float volumeScaling,
- const int stopPosition,
- const CodecInst* codecInst);
- int StopPlayingFileAsMicrophone();
- int IsPlayingFileAsMicrophone() const;
- int ScaleFileAsMicrophonePlayout(const float scale);
- int StartRecordingPlayout(const char* fileName, const CodecInst* codecInst);
- int StartRecordingPlayout(OutStream* stream, const CodecInst* codecInst);
- int StopRecordingPlayout();
-
- void SetMixWithMicStatus(bool mix);
-
- // VoEExternalMediaProcessing
- int RegisterExternalMediaProcessing(ProcessingTypes type,
- VoEMediaProcess& processObject);
- int DeRegisterExternalMediaProcessing(ProcessingTypes type);
-
- // VoEVolumeControl
- int GetSpeechOutputLevel(WebRtc_UWord32& level) const;
- int GetSpeechOutputLevelFullRange(WebRtc_UWord32& level) const;
- int SetMute(const bool enable);
- bool Mute() const;
- int SetOutputVolumePan(float left, float right);
- int GetOutputVolumePan(float& left, float& right) const;
- int SetChannelOutputVolumeScaling(float scaling);
- int GetChannelOutputVolumeScaling(float& scaling) const;
-
- // VoECallReport
- void ResetDeadOrAliveCounters();
- int ResetRTCPStatistics();
- int GetRoundTripTimeSummary(StatVal& delaysMs) const;
- int GetDeadOrAliveCounters(int& countDead, int& countAlive) const;
-
- // VoENetEqStats
- int GetNetworkStatistics(NetworkStatistics& stats);
-
- // VoEVideoSync
- int GetDelayEstimate(int& delayMs) const;
- int SetMinimumPlayoutDelay(int delayMs);
- int GetPlayoutTimestamp(unsigned int& timestamp);
- int SetInitTimestamp(unsigned int timestamp);
- int SetInitSequenceNumber(short sequenceNumber);
-
- // VoEVideoSyncExtended
- int GetRtpRtcp(RtpRtcp* &rtpRtcpModule) const;
-
- // VoEEncryption
-#ifdef WEBRTC_SRTP
- int EnableSRTPSend(
- CipherTypes cipherType,
- int cipherKeyLength,
- AuthenticationTypes authType,
- int authKeyLength,
- int authTagLength,
- SecurityLevels level,
- const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
- bool useForRTCP);
- int DisableSRTPSend();
- int EnableSRTPReceive(
- CipherTypes cipherType,
- int cipherKeyLength,
- AuthenticationTypes authType,
- int authKeyLength,
- int authTagLength,
- SecurityLevels level,
- const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
- bool useForRTCP);
- int DisableSRTPReceive();
-#endif
- int RegisterExternalEncryption(Encryption& encryption);
- int DeRegisterExternalEncryption();
-
- // VoEDtmf
- int SendTelephoneEventOutband(unsigned char eventCode, int lengthMs,
- int attenuationDb, bool playDtmfEvent);
- int SendTelephoneEventInband(unsigned char eventCode, int lengthMs,
- int attenuationDb, bool playDtmfEvent);
- int SetDtmfPlayoutStatus(bool enable);
- bool DtmfPlayoutStatus() const;
- int SetSendTelephoneEventPayloadType(unsigned char type);
- int GetSendTelephoneEventPayloadType(unsigned char& type);
-#ifdef WEBRTC_DTMF_DETECTION
- int RegisterTelephoneEventDetection(
- TelephoneEventDetectionMethods detectionMethod,
- VoETelephoneEventObserver& observer);
- int DeRegisterTelephoneEventDetection();
- int GetTelephoneEventDetectionStatus(
- bool& enabled,
- TelephoneEventDetectionMethods& detectionMethod);
-#endif
-
- // VoEAudioProcessingImpl
- int UpdateRxVadDetection(AudioFrame& audioFrame);
- int RegisterRxVadObserver(VoERxVadCallback &observer);
- int DeRegisterRxVadObserver();
- int VoiceActivityIndicator(int &activity);
-#ifdef WEBRTC_VOICE_ENGINE_AGC
- int SetRxAgcStatus(const bool enable, const AgcModes mode);
- int GetRxAgcStatus(bool& enabled, AgcModes& mode);
- int SetRxAgcConfig(const AgcConfig config);
- int GetRxAgcConfig(AgcConfig& config);
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_NR
- int SetRxNsStatus(const bool enable, const NsModes mode);
- int GetRxNsStatus(bool& enabled, NsModes& mode);
-#endif
-
- // VoERTP_RTCP
- int RegisterRTPObserver(VoERTPObserver& observer);
- int DeRegisterRTPObserver();
- int RegisterRTCPObserver(VoERTCPObserver& observer);
- int DeRegisterRTCPObserver();
- int SetLocalSSRC(unsigned int ssrc);
- int GetLocalSSRC(unsigned int& ssrc);
- int GetRemoteSSRC(unsigned int& ssrc);
- int GetRemoteCSRCs(unsigned int arrCSRC[15]);
- int SetRTPAudioLevelIndicationStatus(bool enable, unsigned char ID);
- int GetRTPAudioLevelIndicationStatus(bool& enable, unsigned char& ID);
- int SetRTCPStatus(bool enable);
- int GetRTCPStatus(bool& enabled);
- int SetRTCP_CNAME(const char cName[256]);
- int GetRTCP_CNAME(char cName[256]);
- int GetRemoteRTCP_CNAME(char cName[256]);
- int GetRemoteRTCPData(unsigned int& NTPHigh, unsigned int& NTPLow,
- unsigned int& timestamp,
- unsigned int& playoutTimestamp, unsigned int* jitter,
- unsigned short* fractionLost);
- int SendApplicationDefinedRTCPPacket(const unsigned char subType,
- unsigned int name, const char* data,
- unsigned short dataLengthInBytes);
- int GetRTPStatistics(unsigned int& averageJitterMs,
- unsigned int& maxJitterMs,
- unsigned int& discardedPackets);
- int GetRTPStatistics(CallStatistics& stats);
- int SetFECStatus(bool enable, int redPayloadtype);
- int GetFECStatus(bool& enabled, int& redPayloadtype);
- int SetRTPKeepaliveStatus(bool enable, unsigned char unknownPayloadType,
- int deltaTransmitTimeSeconds);
- int GetRTPKeepaliveStatus(bool& enabled, unsigned char& unknownPayloadType,
- int& deltaTransmitTimeSeconds);
- int StartRTPDump(const char fileNameUTF8[1024], RTPDirections direction);
- int StopRTPDump(RTPDirections direction);
- bool RTPDumpIsActive(RTPDirections direction);
- int InsertExtraRTPPacket(unsigned char payloadType, bool markerBit,
- const char* payloadData,
- unsigned short payloadSize);
-
-public:
- // From AudioPacketizationCallback in the ACM
- WebRtc_Word32 SendData(FrameType frameType,
- WebRtc_UWord8 payloadType,
- WebRtc_UWord32 timeStamp,
- const WebRtc_UWord8* payloadData,
- WebRtc_UWord16 payloadSize,
- const RTPFragmentationHeader* fragmentation);
- // From ACMVADCallback in the ACM
- WebRtc_Word32 InFrameType(WebRtc_Word16 frameType);
-
-#ifdef WEBRTC_DTMF_DETECTION
-public: // From AudioCodingFeedback in the ACM
- int IncomingDtmf(const WebRtc_UWord8 digitDtmf, const bool end);
-#endif
-
-public:
- WebRtc_Word32 OnRxVadDetected(const int vadDecision);
-
-public:
- // From RtpData in the RTP/RTCP module
- WebRtc_Word32 OnReceivedPayloadData(const WebRtc_UWord8* payloadData,
- const WebRtc_UWord16 payloadSize,
- const WebRtcRTPHeader* rtpHeader);
-
-public:
- // From RtpFeedback in the RTP/RTCP module
- WebRtc_Word32 OnInitializeDecoder(
- const WebRtc_Word32 id,
- const WebRtc_Word8 payloadType,
- const WebRtc_Word8 payloadName[RTP_PAYLOAD_NAME_SIZE],
- const int frequency,
- const WebRtc_UWord8 channels,
- const WebRtc_UWord32 rate);
-
- void OnPacketTimeout(const WebRtc_Word32 id);
-
- void OnReceivedPacket(const WebRtc_Word32 id,
- const RtpRtcpPacketType packetType);
-
- void OnPeriodicDeadOrAlive(const WebRtc_Word32 id,
- const RTPAliveType alive);
-
- void OnIncomingSSRCChanged(const WebRtc_Word32 id,
- const WebRtc_UWord32 SSRC);
-
- void OnIncomingCSRCChanged(const WebRtc_Word32 id,
- const WebRtc_UWord32 CSRC, const bool added);
-
-public:
- // From RtcpFeedback in the RTP/RTCP module
- void OnApplicationDataReceived(const WebRtc_Word32 id,
- const WebRtc_UWord8 subType,
- const WebRtc_UWord32 name,
- const WebRtc_UWord16 length,
- const WebRtc_UWord8* data);
-
-public:
- // From RtpAudioFeedback in the RTP/RTCP module
- void OnReceivedTelephoneEvent(const WebRtc_Word32 id,
- const WebRtc_UWord8 event,
- const bool endOfEvent);
-
- void OnPlayTelephoneEvent(const WebRtc_Word32 id,
- const WebRtc_UWord8 event,
- const WebRtc_UWord16 lengthMs,
- const WebRtc_UWord8 volume);
-
-public:
- // From UdpTransportData in the Socket Transport module
- void IncomingRTPPacket(const WebRtc_Word8* incomingRtpPacket,
- const WebRtc_Word32 rtpPacketLength,
- const WebRtc_Word8* fromIP,
- const WebRtc_UWord16 fromPort);
-
- void IncomingRTCPPacket(const WebRtc_Word8* incomingRtcpPacket,
- const WebRtc_Word32 rtcpPacketLength,
- const WebRtc_Word8* fromIP,
- const WebRtc_UWord16 fromPort);
-
-public:
- // From Transport (called by the RTP/RTCP module)
- int SendPacket(int /*channel*/, const void *data, int len);
- int SendRTCPPacket(int /*channel*/, const void *data, int len);
-
-public:
- // From MixerParticipant
- WebRtc_Word32 GetAudioFrame(const WebRtc_Word32 id,
- AudioFrame& audioFrame);
- WebRtc_Word32 NeededFrequency(const WebRtc_Word32 id);
-
-public:
- // From MonitorObserver
- void OnPeriodicProcess();
-
-public:
- // From FileCallback
- void PlayNotification(const WebRtc_Word32 id,
- const WebRtc_UWord32 durationMs);
- void RecordNotification(const WebRtc_Word32 id,
- const WebRtc_UWord32 durationMs);
- void PlayFileEnded(const WebRtc_Word32 id);
- void RecordFileEnded(const WebRtc_Word32 id);
-
-public:
- WebRtc_UWord32 InstanceId() const
- {
- return _instanceId;
- }
- WebRtc_Word32 ChannelId() const
- {
- return _channelId;
- }
- bool Playing() const
- {
- return _playing;
- }
- bool Sending() const
- {
- // A lock is needed because |_sending| is accessed by both
- // TransmitMixer::PrepareDemux() and StartSend()/StopSend(), which
- // are called by different threads.
- CriticalSectionScoped cs(_callbackCritSect);
- return _sending;
- }
- bool Receiving() const
- {
- return _receiving;
- }
- bool ExternalTransport() const
- {
- return _externalTransport;
- }
- bool OutputIsOnHold() const
- {
- return _outputIsOnHold;
- }
- bool InputIsOnHold() const
- {
- return _inputIsOnHold;
- }
- RtpRtcp* RtpRtcpModulePtr() const
- {
- return &_rtpRtcpModule;
- }
- WebRtc_Word8 OutputEnergyLevel() const
- {
- return _outputAudioLevel.Level();
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- bool SendSocketsInitialized() const
- {
- return _socketTransportModule.SendSocketsInitialized();
- }
- bool ReceiveSocketsInitialized() const
- {
- return _socketTransportModule.ReceiveSocketsInitialized();
- }
-#endif
- WebRtc_UWord32 Demultiplex(const AudioFrame& audioFrame);
- WebRtc_UWord32 PrepareEncodeAndSend(int mixingFrequency);
- WebRtc_UWord32 EncodeAndSend();
-
-private:
- int InsertInbandDtmfTone();
- WebRtc_Word32
- MixOrReplaceAudioWithFile(const int mixingFrequency);
- WebRtc_Word32 MixAudioWithFile(AudioFrame& audioFrame,
- const int mixingFrequency);
- WebRtc_Word32 GetPlayoutTimeStamp(WebRtc_UWord32& playoutTimestamp);
- void UpdateDeadOrAliveCounters(bool alive);
- WebRtc_Word32 SendPacketRaw(const void *data, int len, bool RTCP);
- WebRtc_Word32 UpdatePacketDelay(const WebRtc_UWord32 timestamp,
- const WebRtc_UWord16 sequenceNumber);
- void RegisterReceiveCodecsToRTPModule();
- int ApmProcessRx(AudioFrame& audioFrame);
-
-private:
- CriticalSectionWrapper& _fileCritSect;
- CriticalSectionWrapper& _callbackCritSect;
- CriticalSectionWrapper& _transmitCritSect;
- WebRtc_UWord32 _instanceId;
- WebRtc_Word32 _channelId;
-
-private:
- RtpRtcp& _rtpRtcpModule;
- AudioCodingModule& _audioCodingModule;
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- WebRtc_UWord8 _numSocketThreads;
- UdpTransport& _socketTransportModule;
-#endif
-#ifdef WEBRTC_SRTP
- SrtpModule& _srtpModule;
-#endif
- RtpDump& _rtpDumpIn;
- RtpDump& _rtpDumpOut;
-private:
- AudioLevel _outputAudioLevel;
- bool _externalTransport;
- AudioFrame _audioFrame;
- WebRtc_UWord8 _audioLevel_dBov;
- FilePlayer* _inputFilePlayerPtr;
- FilePlayer* _outputFilePlayerPtr;
- FileRecorder* _outputFileRecorderPtr;
- int _inputFilePlayerId;
- int _outputFilePlayerId;
- int _outputFileRecorderId;
- bool _inputFilePlaying;
- bool _outputFilePlaying;
- bool _outputFileRecording;
- DtmfInbandQueue _inbandDtmfQueue;
- DtmfInband _inbandDtmfGenerator;
- bool _inputExternalMedia;
- bool _outputExternalMedia;
- VoEMediaProcess* _inputExternalMediaCallbackPtr;
- VoEMediaProcess* _outputExternalMediaCallbackPtr;
- WebRtc_UWord8* _encryptionRTPBufferPtr;
- WebRtc_UWord8* _decryptionRTPBufferPtr;
- WebRtc_UWord8* _encryptionRTCPBufferPtr;
- WebRtc_UWord8* _decryptionRTCPBufferPtr;
- WebRtc_UWord32 _timeStamp;
- WebRtc_UWord8 _sendTelephoneEventPayloadType;
- WebRtc_UWord32 _playoutTimeStampRTP;
- WebRtc_UWord32 _playoutTimeStampRTCP;
- WebRtc_UWord32 _numberOfDiscardedPackets;
-private:
- // uses
- Statistics* _engineStatisticsPtr;
- OutputMixer* _outputMixerPtr;
- TransmitMixer* _transmitMixerPtr;
- ProcessThread* _moduleProcessThreadPtr;
- AudioDeviceModule* _audioDeviceModulePtr;
- VoiceEngineObserver* _voiceEngineObserverPtr; // owned by base
- CriticalSectionWrapper* _callbackCritSectPtr; // owned by base
- Transport* _transportPtr; // WebRtc socket or external transport
- Encryption* _encryptionPtr; // WebRtc SRTP or external encryption
- scoped_ptr<AudioProcessing> _rtpAudioProc;
- AudioProcessing* _rxAudioProcessingModulePtr; // far end AudioProcessing
-#ifdef WEBRTC_DTMF_DETECTION
- VoETelephoneEventObserver* _telephoneEventDetectionPtr;
-#endif
- VoERxVadCallback* _rxVadObserverPtr;
- WebRtc_Word32 _oldVadDecision;
- WebRtc_Word32 _sendFrameType; // Send data is voice, 1-voice, 0-otherwise
- VoERTPObserver* _rtpObserverPtr;
- VoERTCPObserver* _rtcpObserverPtr;
-private:
- // VoEBase
- bool _outputIsOnHold;
- bool _externalPlayout;
- bool _inputIsOnHold;
- bool _playing;
- bool _sending;
- bool _receiving;
- bool _mixFileWithMicrophone;
- bool _rtpObserver;
- bool _rtcpObserver;
- // VoEVolumeControl
- bool _mute;
- float _panLeft;
- float _panRight;
- float _outputGain;
- // VoEEncryption
- bool _encrypting;
- bool _decrypting;
- // VoEDtmf
- bool _playOutbandDtmfEvent;
- bool _playInbandDtmfEvent;
- bool _inbandTelephoneEventDetection;
- bool _outOfBandTelephoneEventDetecion;
- // VoeRTP_RTCP
- WebRtc_UWord8 _extraPayloadType;
- bool _insertExtraRTPPacket;
- bool _extraMarkerBit;
- WebRtc_UWord32 _lastLocalTimeStamp;
- WebRtc_Word8 _lastPayloadType;
- bool _includeAudioLevelIndication;
- // VoENetwork
- bool _rtpPacketTimedOut;
- bool _rtpPacketTimeOutIsEnabled;
- WebRtc_UWord32 _rtpTimeOutSeconds;
- bool _connectionObserver;
- VoEConnectionObserver* _connectionObserverPtr;
- WebRtc_UWord32 _countAliveDetections;
- WebRtc_UWord32 _countDeadDetections;
- AudioFrame::SpeechType _outputSpeechType;
- // VoEVideoSync
- WebRtc_UWord32 _averageDelayMs;
- WebRtc_UWord16 _previousSequenceNumber;
- WebRtc_UWord32 _previousTimestamp;
- WebRtc_UWord16 _recPacketDelayMs;
- // VoEAudioProcessing
- bool _RxVadDetection;
- bool _rxApmIsEnabled;
- bool _rxAgcIsEnabled;
- bool _rxNsIsEnabled;
-};
-
-} // namespace voe
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_CHANNEL_H
diff --git a/trunk/src/voice_engine/main/source/channel_manager.cc b/trunk/src/voice_engine/main/source/channel_manager.cc
deleted file mode 100644
index 47cec4a..0000000
--- a/trunk/src/voice_engine/main/source/channel_manager.cc
+++ /dev/null
@@ -1,161 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "channel.h"
-#include "channel_manager.h"
-
-namespace webrtc
-{
-
-namespace voe
-{
-
-ChannelManager::ChannelManager(const WebRtc_UWord32 instanceId) :
- ChannelManagerBase(),
- _instanceId(instanceId)
-{
-}
-
-ChannelManager::~ChannelManager()
-{
- ChannelManagerBase::DestroyAllItems();
-}
-
-bool ChannelManager::CreateChannel(WebRtc_Word32& channelId)
-{
- return ChannelManagerBase::CreateItem(channelId);
-}
-
-WebRtc_Word32 ChannelManager::DestroyChannel(const WebRtc_Word32 channelId)
-{
- Channel* deleteChannel =
- static_cast<Channel*> (ChannelManagerBase::RemoveItem(channelId));
- if (!deleteChannel)
- {
- return -1;
- }
- delete deleteChannel;
- return 0;
-}
-
-WebRtc_Word32 ChannelManager::NumOfChannels() const
-{
- return ChannelManagerBase::NumOfItems();
-}
-
-WebRtc_Word32 ChannelManager::MaxNumOfChannels() const
-{
- return ChannelManagerBase::MaxNumOfItems();
-}
-
-void* ChannelManager::NewItem(WebRtc_Word32 itemID)
-{
- Channel* channel;
- if (Channel::CreateChannel(channel, itemID, _instanceId) == -1)
- {
- return NULL;
- }
- return static_cast<void*> (channel);
-}
-
-void ChannelManager::DeleteItem(void* item)
-{
- Channel* deleteItem = static_cast<Channel*> (item);
- delete deleteItem;
-}
-
-Channel* ChannelManager::GetChannel(const WebRtc_Word32 channelId) const
-{
- return static_cast<Channel*> (ChannelManagerBase::GetItem(channelId));
-}
-
-void ChannelManager::ReleaseChannel()
-{
- ChannelManagerBase::ReleaseItem();
-}
-
-void ChannelManager::GetChannelIds(WebRtc_Word32* channelsArray,
- WebRtc_Word32& numOfChannels) const
-{
- ChannelManagerBase::GetItemIds(channelsArray, numOfChannels);
-}
-
-void ChannelManager::GetChannels(MapWrapper& channels) const
-{
- ChannelManagerBase::GetChannels(channels);
-}
-
-ScopedChannel::ScopedChannel(ChannelManager& chManager) :
- _chManager(chManager),
- _channelPtr(NULL)
-{
- // Copy all existing channels to the local map.
- // It is not possible to utilize the ChannelPtr() API after
- // this constructor. The intention is that this constructor
- // is used in combination with the scoped iterator.
- _chManager.GetChannels(_channels);
-}
-
-ScopedChannel::ScopedChannel(ChannelManager& chManager,
- WebRtc_Word32 channelId) :
- _chManager(chManager),
- _channelPtr(NULL)
-{
- _channelPtr = _chManager.GetChannel(channelId);
-}
-
-ScopedChannel::~ScopedChannel()
-{
- if (_channelPtr != NULL || _channels.Size() != 0)
- {
- _chManager.ReleaseChannel();
- }
-
- // Delete the map
- while (_channels.Erase(_channels.First()) == 0)
- ;
-}
-
-Channel* ScopedChannel::ChannelPtr()
-{
- return _channelPtr;
-}
-
-Channel* ScopedChannel::GetFirstChannel(void*& iterator) const
-{
- MapItem* it = _channels.First();
- iterator = (void*) it;
- if (!it)
- {
- return NULL;
- }
- return static_cast<Channel*> (it->GetItem());
-}
-
-Channel* ScopedChannel::GetNextChannel(void*& iterator) const
-{
- MapItem* it = (MapItem*) iterator;
- if (!it)
- {
- iterator = NULL;
- return NULL;
- }
- it = _channels.Next(it);
- iterator = (void*) it;
- if (!it)
- {
- return NULL;
- }
- return static_cast<Channel*> (it->GetItem());
-}
-
-} // namespace voe
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/channel_manager.h b/trunk/src/voice_engine/main/source/channel_manager.h
deleted file mode 100644
index 6c40ef1..0000000
--- a/trunk/src/voice_engine/main/source/channel_manager.h
+++ /dev/null
@@ -1,89 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_CHANNEL_MANAGER_H
-#define WEBRTC_VOICE_ENGINE_CHANNEL_MANAGER_H
-
-#include "channel_manager_base.h"
-#include "typedefs.h"
-
-namespace webrtc
-{
-
-namespace voe
-{
-
-class ScopedChannel;
-class Channel;
-
-class ChannelManager: private ChannelManagerBase
-{
- friend class ScopedChannel;
-
-public:
- bool CreateChannel(WebRtc_Word32& channelId);
-
- WebRtc_Word32 DestroyChannel(const WebRtc_Word32 channelId);
-
- WebRtc_Word32 MaxNumOfChannels() const;
-
- WebRtc_Word32 NumOfChannels() const;
-
- void GetChannelIds(WebRtc_Word32* channelsArray,
- WebRtc_Word32& numOfChannels) const;
-
- ChannelManager(const WebRtc_UWord32 instanceId);
-
- ~ChannelManager();
-
-private:
- ChannelManager(const ChannelManager&);
-
- ChannelManager& operator=(const ChannelManager&);
-
- Channel* GetChannel(const WebRtc_Word32 channelId) const;
-
- void GetChannels(MapWrapper& channels) const;
-
- void ReleaseChannel();
-
- virtual void* NewItem(WebRtc_Word32 itemID);
-
- virtual void DeleteItem(void* item);
-
- WebRtc_UWord32 _instanceId;
-};
-
-class ScopedChannel
-{
-public:
- // Can only be created by the channel manager
- ScopedChannel(ChannelManager& chManager);
-
- ScopedChannel(ChannelManager& chManager, WebRtc_Word32 channelId);
-
- Channel* ChannelPtr();
-
- Channel* GetFirstChannel(void*& iterator) const;
-
- Channel* GetNextChannel(void*& iterator) const;
-
- ~ScopedChannel();
-private:
- ChannelManager& _chManager;
- Channel* _channelPtr;
- MapWrapper _channels;
-};
-
-} // namespace voe
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_CHANNEL_MANAGER_H
diff --git a/trunk/src/voice_engine/main/source/channel_manager_base.cc b/trunk/src/voice_engine/main/source/channel_manager_base.cc
deleted file mode 100644
index ea9938f..0000000
--- a/trunk/src/voice_engine/main/source/channel_manager_base.cc
+++ /dev/null
@@ -1,227 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "channel_manager_base.h"
-
-#include "critical_section_wrapper.h"
-#include "rw_lock_wrapper.h"
-#include <cassert>
-
-namespace webrtc
-{
-
-namespace voe
-{
-
-ChannelManagerBase::ChannelManagerBase() :
- _itemsCritSectPtr(CriticalSectionWrapper::CreateCriticalSection()),
- _itemsRWLockPtr(RWLockWrapper::CreateRWLock())
-{
- for (int i = 0; i < KMaxNumberOfItems; i++)
- {
- _freeItemIds[i] = true;
- }
-}
-
-ChannelManagerBase::~ChannelManagerBase()
-{
- if (_itemsRWLockPtr)
- {
- delete _itemsRWLockPtr;
- _itemsRWLockPtr = NULL;
- }
- if (_itemsCritSectPtr)
- {
- delete _itemsCritSectPtr;
- _itemsCritSectPtr = NULL;
- }
-}
-
-bool ChannelManagerBase::GetFreeItemId(WebRtc_Word32& itemId)
-{
- CriticalSectionScoped cs(*_itemsCritSectPtr);
- WebRtc_Word32 i(0);
- while (i < KMaxNumberOfItems)
- {
- if (_freeItemIds[i])
- {
- itemId = i;
- _freeItemIds[i] = false;
- return true;
- }
- i++;
- }
- return false;
-}
-
-void ChannelManagerBase::AddFreeItemId(WebRtc_Word32 itemId)
-{
- assert(itemId < KMaxNumberOfItems);
- _freeItemIds[itemId] = true;
-}
-
-void ChannelManagerBase::RemoveFreeItemIds()
-{
- for (int i = 0; i < KMaxNumberOfItems; i++)
- {
- _freeItemIds[i] = false;
- }
-}
-
-bool ChannelManagerBase::CreateItem(WebRtc_Word32& itemId)
-{
- _itemsCritSectPtr->Enter();
- void* itemPtr;
- itemId = -1;
- const bool success = GetFreeItemId(itemId);
- if (!success)
- {
- _itemsCritSectPtr->Leave();
- return false;
- }
- itemPtr = NewItem(itemId);
- if (!itemPtr)
- {
- _itemsCritSectPtr->Leave();
- return false;
- }
- _itemsCritSectPtr->Leave();
- InsertItem(itemId, itemPtr);
-
- return true;
-}
-
-void ChannelManagerBase::InsertItem(WebRtc_Word32 itemId, void* item)
-{
- CriticalSectionScoped cs(*_itemsCritSectPtr);
- assert(!_items.Find(itemId));
- _items.Insert(itemId, item);
-}
-
-void*
-ChannelManagerBase::RemoveItem(WebRtc_Word32 itemId)
-{
- CriticalSectionScoped cs(*_itemsCritSectPtr);
- WriteLockScoped wlock(*_itemsRWLockPtr);
- MapItem* it = _items.Find(itemId);
- if (!it)
- {
- return 0;
- }
- void* returnItem = it->GetItem();
- _items.Erase(it);
- AddFreeItemId(itemId);
-
- return returnItem;
-}
-
-void ChannelManagerBase::DestroyAllItems()
-{
- CriticalSectionScoped cs(*_itemsCritSectPtr);
- MapItem* it = _items.First();
- while (it)
- {
- DeleteItem(it->GetItem());
- _items.Erase(it);
- it = _items.First();
- }
- RemoveFreeItemIds();
-}
-
-WebRtc_Word32 ChannelManagerBase::NumOfItems() const
-{
- return _items.Size();
-}
-
-WebRtc_Word32 ChannelManagerBase::MaxNumOfItems() const
-{
- return static_cast<WebRtc_Word32> (KMaxNumberOfItems);
-}
-
-void*
-ChannelManagerBase::GetItem(WebRtc_Word32 itemId) const
-{
- CriticalSectionScoped cs(*_itemsCritSectPtr);
- MapItem* it = _items.Find(itemId);
- if (!it)
- {
- return 0;
- }
- _itemsRWLockPtr->AcquireLockShared();
- return it->GetItem();
-}
-
-void*
-ChannelManagerBase::GetFirstItem(void*& iterator) const
-{
- CriticalSectionScoped cs(*_itemsCritSectPtr);
- MapItem* it = _items.First();
- iterator = (void*) it;
- if (!it)
- {
- return 0;
- }
- return it->GetItem();
-}
-
-void*
-ChannelManagerBase::GetNextItem(void*& iterator) const
-{
- CriticalSectionScoped cs(*_itemsCritSectPtr);
- MapItem* it = (MapItem*) iterator;
- if (!it)
- {
- iterator = 0;
- return 0;
- }
- it = _items.Next(it);
- iterator = (void*) it;
- if (!it)
- {
- return 0;
- }
- return it->GetItem();
-}
-
-void ChannelManagerBase::ReleaseItem()
-{
- _itemsRWLockPtr->ReleaseLockShared();
-}
-
-void ChannelManagerBase::GetItemIds(WebRtc_Word32* channelsArray,
- WebRtc_Word32& numOfChannels) const
-{
- MapItem* it = _items.First();
- numOfChannels = (numOfChannels <= _items.Size()) ?
- numOfChannels : _items.Size();
- for (int i = 0; i < numOfChannels && it != NULL; i++)
- {
- channelsArray[i] = it->GetId();
- it = _items.Next(it);
- }
-}
-
-void ChannelManagerBase::GetChannels(MapWrapper& channels) const
-{
- CriticalSectionScoped cs(*_itemsCritSectPtr);
- if (_items.Size() == 0)
- {
- return;
- }
- _itemsRWLockPtr->AcquireLockShared();
- for (MapItem* it = _items.First(); it != NULL; it = _items.Next(it))
- {
- channels.Insert(it->GetId(), it->GetItem());
- }
-}
-
-} // namespace voe
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/channel_manager_base.h b/trunk/src/voice_engine/main/source/channel_manager_base.h
deleted file mode 100644
index 0831e43..0000000
--- a/trunk/src/voice_engine/main/source/channel_manager_base.h
+++ /dev/null
@@ -1,90 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_CHANNEL_MANAGER_BASE_H
-#define WEBRTC_VOICE_ENGINE_CHANNEL_MANAGER_BASE_H
-
-#include "typedefs.h"
-#include "map_wrapper.h"
-#include "voice_engine_defines.h"
-
-namespace webrtc
-{
-class CriticalSectionWrapper;
-class RWLockWrapper;
-
-namespace voe
-{
-
-class ScopedChannel;
-class Channel;
-
-class ChannelManagerBase
-{
-public:
- enum {KMaxNumberOfItems = kVoiceEngineMaxNumOfChannels};
-
-protected:
- bool CreateItem(WebRtc_Word32& itemId);
-
- void InsertItem(WebRtc_Word32 itemId, void* item);
-
- void* RemoveItem(WebRtc_Word32 itemId);
-
- void* GetItem(WebRtc_Word32 itemId) const;
-
- void* GetFirstItem(void*& iterator) const ;
-
- void* GetNextItem(void*& iterator) const;
-
- void ReleaseItem();
-
- void AddFreeItemId(WebRtc_Word32 itemId);
-
- bool GetFreeItemId(WebRtc_Word32& itemId);
-
- void RemoveFreeItemIds();
-
- void DestroyAllItems();
-
- WebRtc_Word32 NumOfItems() const;
-
- WebRtc_Word32 MaxNumOfItems() const;
-
- void GetItemIds(WebRtc_Word32* channelsArray,
- WebRtc_Word32& numOfChannels) const;
-
- void GetChannels(MapWrapper& channels) const;
-
- virtual void* NewItem(WebRtc_Word32 itemId) = 0;
-
- virtual void DeleteItem(void* item) = 0;
-
- ChannelManagerBase();
-
- virtual ~ChannelManagerBase();
-
-private:
- // Protects _items and _freeItemIds
- CriticalSectionWrapper* _itemsCritSectPtr;
-
- MapWrapper _items;
-
- bool _freeItemIds[KMaxNumberOfItems];
-
- // Protects channels from being destroyed while being used
- RWLockWrapper* _itemsRWLockPtr;
-};
-
-} // namespace voe
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_CHANNEL_MANAGER_BASE_H
diff --git a/trunk/src/voice_engine/main/source/channel_unittest.cc b/trunk/src/voice_engine/main/source/channel_unittest.cc
deleted file mode 100644
index fc78679..0000000
--- a/trunk/src/voice_engine/main/source/channel_unittest.cc
+++ /dev/null
@@ -1,15 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "channel.h"
-#include "gtest/gtest.h"
-
-// Empty test just to get coverage metrics.
-TEST(ChannelTest, EmptyTestToGetCodeCoverage) {}
diff --git a/trunk/src/voice_engine/main/source/dtmf_inband.cc b/trunk/src/voice_engine/main/source/dtmf_inband.cc
deleted file mode 100644
index 473af10..0000000
--- a/trunk/src/voice_engine/main/source/dtmf_inband.cc
+++ /dev/null
@@ -1,389 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "dtmf_inband.h"
-
-#include "critical_section_wrapper.h"
-#include "trace.h"
-#include <cassert>
-
-namespace webrtc {
-
-const WebRtc_Word16 Dtmf_a_times2Tab8Khz[8]=
-{
- 27978, 26956, 25701, 24219,
- 19073, 16325, 13085, 9314
-};
-
-const WebRtc_Word16 Dtmf_a_times2Tab16Khz[8]=
-{
- 31548, 31281, 30951, 30556,
- 29144, 28361, 27409, 26258
-};
-
-const WebRtc_Word16 Dtmf_a_times2Tab32Khz[8]=
-{
- 32462,32394, 32311, 32210, 31849, 31647, 31400, 31098
-};
-
-// Second table is sin(2*pi*f/fs) in Q14
-
-const WebRtc_Word16 Dtmf_ym2Tab8Khz[8]=
-{
- 8527, 9315, 10163, 11036,
- 13322, 14206, 15021, 15708
-};
-
-const WebRtc_Word16 Dtmf_ym2Tab16Khz[8]=
-{
- 4429, 4879, 5380, 5918,
- 7490, 8207, 8979, 9801
-};
-
-const WebRtc_Word16 Dtmf_ym2Tab32Khz[8]=
-{
- 2235, 2468, 2728, 3010, 3853, 4249, 4685, 5164
-};
-
-const WebRtc_Word16 Dtmf_dBm0kHz[37]=
-{
- 16141, 14386, 12821, 11427, 10184, 9077,
- 8090, 7210, 6426, 5727, 5104, 4549,
- 4054, 3614, 3221, 2870, 2558, 2280,
- 2032, 1811, 1614, 1439, 1282, 1143,
- 1018, 908, 809, 721, 643, 573,
- 510, 455, 405, 361, 322, 287,
- 256
-};
-
-
-DtmfInband::DtmfInband(const WebRtc_Word32 id) :
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _id(id),
- _outputFrequencyHz(8000),
- _frameLengthSamples(0),
- _remainingSamples(0),
- _eventCode(0),
- _attenuationDb(0),
- _lengthMs(0),
- _reinit(true),
- _playing(false),
- _delaySinceLastToneMS(1000)
-{
- memset(_oldOutputLow, 0, sizeof(_oldOutputLow));
- memset(_oldOutputHigh, 0, sizeof(_oldOutputHigh));
-}
-
-DtmfInband::~DtmfInband()
-{
- delete &_critSect;
-}
-
-int
-DtmfInband::SetSampleRate(const WebRtc_UWord16 frequency)
-{
- if (frequency != 8000 &&
- frequency != 16000 &&
- frequency != 32000)
- {
- // invalid sample rate
- assert(false);
- return -1;
- }
- _outputFrequencyHz = frequency;
- return 0;
-}
-
-int
-DtmfInband::GetSampleRate(WebRtc_UWord16& frequency)
-{
- frequency = _outputFrequencyHz;
- return 0;
-}
-
-void
-DtmfInband::Init()
-{
- _remainingSamples = 0;
- _frameLengthSamples = 0;
- _eventCode = 0;
- _attenuationDb = 0;
- _lengthMs = 0;
- _reinit = true;
- _oldOutputLow[0] = 0;
- _oldOutputLow[1] = 0;
- _oldOutputHigh[0] = 0;
- _oldOutputHigh[1] = 0;
- _delaySinceLastToneMS = 1000;
-}
-
-int
-DtmfInband::AddTone(const WebRtc_UWord8 eventCode,
- WebRtc_Word32 lengthMs,
- WebRtc_Word32 attenuationDb)
-{
- CriticalSectionScoped lock(_critSect);
-
- if (attenuationDb > 36 || eventCode > 15)
- {
- assert(false);
- return -1;
- }
-
- if (IsAddingTone())
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_id,-1),
- "DtmfInband::AddTone() new tone interrupts ongoing tone");
- }
-
- ReInit();
-
- _frameLengthSamples = static_cast<WebRtc_Word16> (_outputFrequencyHz / 100);
- _eventCode = static_cast<WebRtc_Word16> (eventCode);
- _attenuationDb = static_cast<WebRtc_Word16> (attenuationDb);
- _remainingSamples = static_cast<WebRtc_Word32>
- (lengthMs * (_outputFrequencyHz / 1000));
- _lengthMs = lengthMs;
-
- return 0;
-}
-
-int
-DtmfInband::ResetTone()
-{
- CriticalSectionScoped lock(_critSect);
-
- ReInit();
-
- _frameLengthSamples = static_cast<WebRtc_Word16> (_outputFrequencyHz / 100);
- _remainingSamples = static_cast<WebRtc_Word32>
- (_lengthMs * (_outputFrequencyHz / 1000));
-
- return 0;
-}
-
-int
-DtmfInband::StartTone(const WebRtc_UWord8 eventCode,
- WebRtc_Word32 attenuationDb)
-{
- CriticalSectionScoped lock(_critSect);
-
- if (attenuationDb > 36 || eventCode > 15)
- {
- assert(false);
- return -1;
- }
-
- if (IsAddingTone())
- {
- return -1;
- }
-
- ReInit();
-
- _frameLengthSamples = static_cast<WebRtc_Word16> (_outputFrequencyHz / 100);
- _eventCode = static_cast<WebRtc_Word16> (eventCode);
- _attenuationDb = static_cast<WebRtc_Word16> (attenuationDb);
- _playing = true;
-
- return 0;
-}
-
-int
-DtmfInband::StopTone()
-{
- CriticalSectionScoped lock(_critSect);
-
- if (!_playing)
- {
- return 0;
- }
-
- _playing = false;
-
- return 0;
-}
-
-// Shall be called between tones
-void
-DtmfInband::ReInit()
-{
- _reinit = true;
-}
-
-bool
-DtmfInband::IsAddingTone()
-{
- CriticalSectionScoped lock(_critSect);
- return (_remainingSamples > 0 || _playing);
-}
-
-int
-DtmfInband::Get10msTone(WebRtc_Word16 output[320],
- WebRtc_UWord16& outputSizeInSamples)
-{
- CriticalSectionScoped lock(_critSect);
- if (DtmfFix_generate(output,
- _eventCode,
- _attenuationDb,
- _frameLengthSamples,
- _outputFrequencyHz) == -1)
- {
- return -1;
- }
- _remainingSamples -= _frameLengthSamples;
- outputSizeInSamples = _frameLengthSamples;
- _delaySinceLastToneMS = 0;
- return 0;
-}
-
-void
-DtmfInband::UpdateDelaySinceLastTone()
-{
- _delaySinceLastToneMS += kDtmfFrameSizeMs;
- // avoid wraparound
- if (_delaySinceLastToneMS > (1<<30))
- {
- _delaySinceLastToneMS = 1000;
- }
-}
-
-WebRtc_UWord32
-DtmfInband::DelaySinceLastTone() const
-{
- return _delaySinceLastToneMS;
-}
-
-WebRtc_Word16
-DtmfInband::DtmfFix_generate(WebRtc_Word16 *decoded,
- const WebRtc_Word16 value,
- const WebRtc_Word16 volume,
- const WebRtc_Word16 frameLen,
- const WebRtc_Word16 fs)
-{
- const WebRtc_Word16 *a_times2Tbl;
- const WebRtc_Word16 *y2_Table;
- WebRtc_Word16 a1_times2 = 0, a2_times2 = 0;
-
- if (fs==8000) {
- a_times2Tbl=Dtmf_a_times2Tab8Khz;
- y2_Table=Dtmf_ym2Tab8Khz;
- } else if (fs==16000) {
- a_times2Tbl=Dtmf_a_times2Tab16Khz;
- y2_Table=Dtmf_ym2Tab16Khz;
- } else if (fs==32000) {
- a_times2Tbl=Dtmf_a_times2Tab32Khz;
- y2_Table=Dtmf_ym2Tab32Khz;
- } else {
- return(-1);
- }
-
- if ((value==1)||(value==2)||(value==3)||(value==12)) {
- a1_times2=a_times2Tbl[0];
- if (_reinit) {
- _oldOutputLow[0]=y2_Table[0];
- _oldOutputLow[1]=0;
- }
- } else if ((value==4)||(value==5)||(value==6)||(value==13)) {
- a1_times2=a_times2Tbl[1];
- if (_reinit) {
- _oldOutputLow[0]=y2_Table[1];
- _oldOutputLow[1]=0;
- }
- } else if ((value==7)||(value==8)||(value==9)||(value==14)) {
- a1_times2=a_times2Tbl[2];
- if (_reinit) {
- _oldOutputLow[0]=y2_Table[2];
- _oldOutputLow[1]=0;
- }
- } else if ((value==10)||(value==0)||(value==11)||(value==15)) {
- a1_times2=a_times2Tbl[3];
- if (_reinit) {
- _oldOutputLow[0]=y2_Table[3];
- _oldOutputLow[1]=0;
- }
- }
- if ((value==1)||(value==4)||(value==7)||(value==10)) {
- a2_times2=a_times2Tbl[4];
- if (_reinit) {
- _oldOutputHigh[0]=y2_Table[4];
- _oldOutputHigh[1]=0;
- _reinit=false;
- }
- } else if ((value==2)||(value==5)||(value==8)||(value==0)) {
- a2_times2=a_times2Tbl[5];
- if (_reinit) {
- _oldOutputHigh[0]=y2_Table[5];
- _oldOutputHigh[1]=0;
- _reinit=false;
- }
- } else if ((value==3)||(value==6)||(value==9)||(value==11)) {
- a2_times2=a_times2Tbl[6];
- if (_reinit) {
- _oldOutputHigh[0]=y2_Table[6];
- _oldOutputHigh[1]=0;
- _reinit=false;
- }
- } else if ((value==12)||(value==13)||(value==14)||(value==15)) {
- a2_times2=a_times2Tbl[7];
- if (_reinit) {
- _oldOutputHigh[0]=y2_Table[7];
- _oldOutputHigh[1]=0;
- _reinit=false;
- }
- }
-
- return (DtmfFix_generateSignal(a1_times2,
- a2_times2,
- volume,
- decoded,
- frameLen));
-}
-
-WebRtc_Word16
-DtmfInband::DtmfFix_generateSignal(const WebRtc_Word16 a1_times2,
- const WebRtc_Word16 a2_times2,
- const WebRtc_Word16 volume,
- WebRtc_Word16 *signal,
- const WebRtc_Word16 length)
-{
- int i;
-
- /* Generate Signal */
- for (i=0;i<length;i++) {
- WebRtc_Word32 tempVal;
- WebRtc_Word16 tempValLow, tempValHigh;
-
- /* Use recursion formula y[n] = a*2*y[n-1] - y[n-2] */
- tempValLow = (WebRtc_Word16)(((( (WebRtc_Word32)(a1_times2 *
- _oldOutputLow[1])) + 8192) >> 14) - _oldOutputLow[0]);
- tempValHigh = (WebRtc_Word16)(((( (WebRtc_Word32)(a2_times2 *
- _oldOutputHigh[1])) + 8192) >> 14) - _oldOutputHigh[0]);
-
- /* Update memory */
- _oldOutputLow[0]=_oldOutputLow[1];
- _oldOutputLow[1]=tempValLow;
- _oldOutputHigh[0]=_oldOutputHigh[1];
- _oldOutputHigh[1]=tempValHigh;
-
- tempVal = (WebRtc_Word32)(kDtmfAmpLow * tempValLow) +
- (WebRtc_Word32)(kDtmfAmpHigh * tempValHigh);
-
- /* Norm the signal to Q14 */
- tempVal=(tempVal+16384)>>15;
-
- /* Scale the signal to correct dbM0 value */
- signal[i]=(WebRtc_Word16)((tempVal*Dtmf_dBm0kHz[volume]+8192)>>14);
- }
-
- return(0);
-}
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/dtmf_inband.h b/trunk/src/voice_engine/main/source/dtmf_inband.h
deleted file mode 100644
index 806fff0..0000000
--- a/trunk/src/voice_engine/main/source/dtmf_inband.h
+++ /dev/null
@@ -1,93 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_DTMF_INBAND_H
-#define WEBRTC_VOICE_ENGINE_DTMF_INBAND_H
-
-#if _MSC_VER > 1000
-#pragma once
-#endif
-
-#include "typedefs.h"
-#include "voice_engine_defines.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-class DtmfInband
-{
-public:
- DtmfInband(const WebRtc_Word32 id);
-
- virtual ~DtmfInband();
-
- void Init();
-
- int SetSampleRate(const WebRtc_UWord16 frequency);
-
- int GetSampleRate(WebRtc_UWord16& frequency);
-
- int AddTone(const WebRtc_UWord8 eventCode,
- WebRtc_Word32 lengthMs,
- WebRtc_Word32 attenuationDb);
-
- int ResetTone();
- int StartTone(const WebRtc_UWord8 eventCode,
- WebRtc_Word32 attenuationDb);
-
- int StopTone();
-
- bool IsAddingTone();
-
- int Get10msTone(WebRtc_Word16 output[320],
- WebRtc_UWord16& outputSizeInSamples);
-
- WebRtc_UWord32 DelaySinceLastTone() const;
-
- void UpdateDelaySinceLastTone();
-
-private:
- void ReInit();
- WebRtc_Word16 DtmfFix_generate(WebRtc_Word16* decoded,
- const WebRtc_Word16 value,
- const WebRtc_Word16 volume,
- const WebRtc_Word16 frameLen,
- const WebRtc_Word16 fs);
-
-private:
- enum {kDtmfFrameSizeMs = 10};
- enum {kDtmfAmpHigh = 32768};
- enum {kDtmfAmpLow = 23171}; // 3 dB lower than the high frequency
-
- WebRtc_Word16 DtmfFix_generateSignal(const WebRtc_Word16 a1_times2,
- const WebRtc_Word16 a2_times2,
- const WebRtc_Word16 volume,
- WebRtc_Word16* signal,
- const WebRtc_Word16 length);
-
-private:
- CriticalSectionWrapper& _critSect;
- WebRtc_Word32 _id;
- WebRtc_UWord16 _outputFrequencyHz; // {8000, 16000, 32000}
- WebRtc_Word16 _oldOutputLow[2]; // Data needed for oscillator model
- WebRtc_Word16 _oldOutputHigh[2]; // Data needed for oscillator model
- WebRtc_Word16 _frameLengthSamples; // {80, 160, 320}
- WebRtc_Word32 _remainingSamples;
- WebRtc_Word16 _eventCode; // [0, 15]
- WebRtc_Word16 _attenuationDb; // [0, 36]
- WebRtc_Word32 _lengthMs;
- bool _reinit; // 'true' if the oscillator should be reinit for next event
- bool _playing;
- WebRtc_UWord32 _delaySinceLastToneMS; // time since last generated tone [ms]
-};
-
-} // namespace webrtc
-
-#endif // #ifndef WEBRTC_VOICE_ENGINE_DTMF_INBAND_H
diff --git a/trunk/src/voice_engine/main/source/dtmf_inband_queue.cc b/trunk/src/voice_engine/main/source/dtmf_inband_queue.cc
deleted file mode 100644
index 080ef3e..0000000
--- a/trunk/src/voice_engine/main/source/dtmf_inband_queue.cc
+++ /dev/null
@@ -1,88 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "dtmf_inband_queue.h"
-#include "trace.h"
-
-namespace webrtc {
-
-DtmfInbandQueue::DtmfInbandQueue(const WebRtc_Word32 id):
- _id(id),
- _DtmfCritsect(*CriticalSectionWrapper::CreateCriticalSection()),
- _nextEmptyIndex(0)
-{
- memset(_DtmfKey,0, sizeof(_DtmfKey));
- memset(_DtmfLen,0, sizeof(_DtmfLen));
- memset(_DtmfLevel,0, sizeof(_DtmfLevel));
-}
-
-DtmfInbandQueue::~DtmfInbandQueue()
-{
- delete &_DtmfCritsect;
-}
-
-int
-DtmfInbandQueue::AddDtmf(WebRtc_UWord8 key,
- WebRtc_UWord16 len,
- WebRtc_UWord8 level)
-{
- CriticalSectionScoped lock(_DtmfCritsect);
-
- if (_nextEmptyIndex >= kDtmfInbandMax)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_id,-1),
- "DtmfInbandQueue::AddDtmf() unable to add Dtmf tone");
- return -1;
- }
- WebRtc_Word32 index = _nextEmptyIndex;
- _DtmfKey[index] = key;
- _DtmfLen[index] = len;
- _DtmfLevel[index] = level;
- _nextEmptyIndex++;
- return 0;
-}
-
-WebRtc_Word8
-DtmfInbandQueue::NextDtmf(WebRtc_UWord16* len, WebRtc_UWord8* level)
-{
- CriticalSectionScoped lock(_DtmfCritsect);
-
- if(!PendingDtmf())
- {
- return -1;
- }
- WebRtc_Word8 nextDtmf = _DtmfKey[0];
- *len=_DtmfLen[0];
- *level=_DtmfLevel[0];
-
- memmove(&(_DtmfKey[0]), &(_DtmfKey[1]),
- _nextEmptyIndex*sizeof(WebRtc_UWord8));
- memmove(&(_DtmfLen[0]), &(_DtmfLen[1]),
- _nextEmptyIndex*sizeof(WebRtc_UWord16));
- memmove(&(_DtmfLevel[0]), &(_DtmfLevel[1]),
- _nextEmptyIndex*sizeof(WebRtc_UWord8));
-
- _nextEmptyIndex--;
- return nextDtmf;
-}
-
-bool
-DtmfInbandQueue::PendingDtmf()
-{
- return(_nextEmptyIndex>0);
-}
-
-void
-DtmfInbandQueue::ResetDtmf()
-{
- _nextEmptyIndex = 0;
-}
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/dtmf_inband_queue.h b/trunk/src/voice_engine/main/source/dtmf_inband_queue.h
deleted file mode 100644
index b3bd39e..0000000
--- a/trunk/src/voice_engine/main/source/dtmf_inband_queue.h
+++ /dev/null
@@ -1,52 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_DTMF_INBAND_QUEUE_H
-#define WEBRTC_VOICE_ENGINE_DTMF_INBAND_QUEUE_H
-
-#include "critical_section_wrapper.h"
-#include "typedefs.h"
-#include "voice_engine_defines.h"
-
-
-namespace webrtc {
-
-class DtmfInbandQueue
-{
-public:
-
- DtmfInbandQueue(const WebRtc_Word32 id);
-
- virtual ~DtmfInbandQueue();
-
- int AddDtmf(WebRtc_UWord8 DtmfKey,
- WebRtc_UWord16 len,
- WebRtc_UWord8 level);
-
- WebRtc_Word8 NextDtmf(WebRtc_UWord16* len, WebRtc_UWord8* level);
-
- bool PendingDtmf();
-
- void ResetDtmf();
-
-private:
- enum {kDtmfInbandMax = 20};
-
- WebRtc_Word32 _id;
- CriticalSectionWrapper& _DtmfCritsect;
- WebRtc_UWord8 _nextEmptyIndex;
- WebRtc_UWord8 _DtmfKey[kDtmfInbandMax];
- WebRtc_UWord16 _DtmfLen[kDtmfInbandMax];
- WebRtc_UWord8 _DtmfLevel[kDtmfInbandMax];
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_DTMF_INBAND_QUEUE_H
diff --git a/trunk/src/voice_engine/main/source/level_indicator.cc b/trunk/src/voice_engine/main/source/level_indicator.cc
deleted file mode 100644
index 89004a5..0000000
--- a/trunk/src/voice_engine/main/source/level_indicator.cc
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "level_indicator.h"
-#include "module_common_types.h"
-#include "signal_processing_library.h"
-
-namespace webrtc {
-
-namespace voe {
-
-
-// Number of bars on the indicator.
-// Note that the number of elements is specified because we are indexing it
-// in the range of 0-32
-const WebRtc_Word8 permutation[33] =
- {0,1,2,3,4,4,5,5,5,5,6,6,6,6,6,7,7,7,7,8,8,8,9,9,9,9,9,9,9,9,9,9,9};
-
-
-AudioLevel::AudioLevel() :
- _absMax(0),
- _count(0),
- _currentLevel(0),
- _currentLevelFullRange(0)
-{
-}
-
-AudioLevel::~AudioLevel()
-{
-}
-
-void
-AudioLevel::Clear()
-{
- _absMax = 0;
- _count = 0;
- _currentLevel = 0;
- _currentLevelFullRange = 0;
-}
-
-void
-AudioLevel::ComputeLevel(const AudioFrame& audioFrame)
-{
- WebRtc_Word16 absValue(0);
-
- // Check speech level (works for 2 channels as well)
- absValue = WebRtcSpl_MaxAbsValueW16(
- audioFrame._payloadData,
- audioFrame._payloadDataLengthInSamples*audioFrame._audioChannel);
- if (absValue > _absMax)
- _absMax = absValue;
-
- // Update level approximately 10 times per second
- if (_count++ == kUpdateFrequency)
- {
- _currentLevelFullRange = _absMax;
-
- _count = 0;
-
- // Highest value for a WebRtc_Word16 is 0x7fff = 32767
- // Divide with 1000 to get in the range of 0-32 which is the range of
- // the permutation vector
- WebRtc_Word32 position = _absMax/1000;
-
- // Make it less likely that the bar stays at position 0. I.e. only if
- // its in the range 0-250 (instead of 0-1000)
- if ((position == 0) && (_absMax > 250))
- {
- position = 1;
- }
- _currentLevel = permutation[position];
-
- // Decay the absolute maximum (divide by 4)
- _absMax >>= 2;
- }
-}
-
-WebRtc_Word8
-AudioLevel::Level() const
-{
- return _currentLevel;
-}
-
-WebRtc_Word16
-AudioLevel::LevelFullRange() const
-{
- return _currentLevelFullRange;
-}
-
-} // namespace voe
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/level_indicator.h b/trunk/src/voice_engine/main/source/level_indicator.h
deleted file mode 100644
index 564b068..0000000
--- a/trunk/src/voice_engine/main/source/level_indicator.h
+++ /dev/null
@@ -1,49 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_LEVEL_INDICATOR_H
-#define WEBRTC_VOICE_ENGINE_LEVEL_INDICATOR_H
-
-#include "typedefs.h"
-#include "voice_engine_defines.h"
-
-namespace webrtc {
-
-class AudioFrame;
-namespace voe {
-
-class AudioLevel
-{
-public:
- AudioLevel();
- virtual ~AudioLevel();
-
- void ComputeLevel(const AudioFrame& audioFrame);
-
- WebRtc_Word8 Level() const;
-
- WebRtc_Word16 LevelFullRange() const;
-
- void Clear();
-
-private:
- enum { kUpdateFrequency = 10};
-
- WebRtc_Word16 _absMax;
- WebRtc_Word16 _count;
- WebRtc_Word8 _currentLevel;
- WebRtc_Word16 _currentLevelFullRange;
-};
-
-} // namespace voe
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_LEVEL_INDICATOR_H
diff --git a/trunk/src/voice_engine/main/source/monitor_module.cc b/trunk/src/voice_engine/main/source/monitor_module.cc
deleted file mode 100644
index 6314f38..0000000
--- a/trunk/src/voice_engine/main/source/monitor_module.cc
+++ /dev/null
@@ -1,91 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "critical_section_wrapper.h"
-#include "monitor_module.h"
-
-namespace webrtc {
-
-namespace voe {
-
-MonitorModule::MonitorModule() :
- _observerPtr(NULL),
- _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _lastProcessTime(GET_TIME_IN_MS())
-{
-}
-
-MonitorModule::~MonitorModule()
-{
- delete &_callbackCritSect;
-}
-
-WebRtc_Word32
-MonitorModule::RegisterObserver(MonitorObserver& observer)
-{
- CriticalSectionScoped lock(_callbackCritSect);
- if (_observerPtr)
- {
- return -1;
- }
- _observerPtr = &observer;
- return 0;
-}
-
-WebRtc_Word32
-MonitorModule::DeRegisterObserver()
-{
- CriticalSectionScoped lock(_callbackCritSect);
- if (!_observerPtr)
- {
- return 0;
- }
- _observerPtr = NULL;
- return 0;
-}
-
-WebRtc_Word32
-MonitorModule::Version(WebRtc_Word8* version,
- WebRtc_UWord32& remainingBufferInBytes,
- WebRtc_UWord32& position) const
-{
- return 0;
-}
-
-WebRtc_Word32
-MonitorModule::ChangeUniqueId(const WebRtc_Word32 id)
-{
- return 0;
-}
-
-WebRtc_Word32
-MonitorModule::TimeUntilNextProcess()
-{
- WebRtc_UWord32 now = GET_TIME_IN_MS();
- WebRtc_Word32 timeToNext =
- kAverageProcessUpdateTimeMs - (now - _lastProcessTime);
- return (timeToNext);
-}
-
-WebRtc_Word32
-MonitorModule::Process()
-{
- _lastProcessTime = GET_TIME_IN_MS();
- if (_observerPtr)
- {
- CriticalSectionScoped lock(_callbackCritSect);
- _observerPtr->OnPeriodicProcess();
- }
- return 0;
-}
-
-} // namespace voe
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/monitor_module.h b/trunk/src/voice_engine/main/source/monitor_module.h
deleted file mode 100644
index 45cf228..0000000
--- a/trunk/src/voice_engine/main/source/monitor_module.h
+++ /dev/null
@@ -1,63 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_MONITOR_MODULE_H
-#define WEBRTC_VOICE_ENGINE_MONITOR_MODULE_H
-
-#include "module.h"
-#include "typedefs.h"
-#include "voice_engine_defines.h"
-
-class MonitorObserver
-{
-public:
- virtual void OnPeriodicProcess() = 0;
-protected:
- virtual ~MonitorObserver() {}
-};
-
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-namespace voe {
-
-class MonitorModule : public Module
-{
-public:
- WebRtc_Word32 RegisterObserver(MonitorObserver& observer);
-
- WebRtc_Word32 DeRegisterObserver();
-
- MonitorModule();
-
- virtual ~MonitorModule();
-public: // module
- WebRtc_Word32 Version(WebRtc_Word8* version,
- WebRtc_UWord32& remainingBufferInBytes,
- WebRtc_UWord32& position) const;
-
- WebRtc_Word32 ChangeUniqueId(const WebRtc_Word32 id);
-
- WebRtc_Word32 TimeUntilNextProcess();
-
- WebRtc_Word32 Process();
-private:
- enum { kAverageProcessUpdateTimeMs = 1000 };
- MonitorObserver* _observerPtr;
- CriticalSectionWrapper& _callbackCritSect;
- WebRtc_Word32 _lastProcessTime;
-};
-
-} // namespace voe
-
-} // namespace webrtc
-
-#endif // VOICE_ENGINE_MONITOR_MODULE
diff --git a/trunk/src/voice_engine/main/source/output_mixer.cc b/trunk/src/voice_engine/main/source/output_mixer.cc
deleted file mode 100644
index 9dfe0ad..0000000
--- a/trunk/src/voice_engine/main/source/output_mixer.cc
+++ /dev/null
@@ -1,754 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "output_mixer.h"
-
-#include "audio_processing.h"
-#include "audio_frame_operations.h"
-#include "critical_section_wrapper.h"
-#include "file_wrapper.h"
-#include "trace.h"
-#include "statistics.h"
-#include "voe_external_media.h"
-
-namespace webrtc {
-
-namespace voe {
-
-void
-OutputMixer::NewMixedAudio(const WebRtc_Word32 id,
- const AudioFrame& generalAudioFrame,
- const AudioFrame** uniqueAudioFrames,
- const WebRtc_UWord32 size)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::NewMixedAudio(id=%d, size=%u)", id, size);
-
- _audioFrame = generalAudioFrame;
- _audioFrame._id = id;
-}
-
-void OutputMixer::MixedParticipants(
- const WebRtc_Word32 id,
- const ParticipantStatistics* participantStatistics,
- const WebRtc_UWord32 size)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::MixedParticipants(id=%d, size=%u)", id, size);
-}
-
-void OutputMixer::VADPositiveParticipants(
- const WebRtc_Word32 id,
- const ParticipantStatistics* participantStatistics,
- const WebRtc_UWord32 size)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::VADPositiveParticipants(id=%d, size=%u)",
- id, size);
-}
-
-void OutputMixer::MixedAudioLevel(const WebRtc_Word32 id,
- const WebRtc_UWord32 level)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::MixedAudioLevel(id=%d, level=%u)", id, level);
-}
-
-void OutputMixer::PlayNotification(const WebRtc_Word32 id,
- const WebRtc_UWord32 durationMs)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::PlayNotification(id=%d, durationMs=%d)",
- id, durationMs);
- // Not implement yet
-}
-
-void OutputMixer::RecordNotification(const WebRtc_Word32 id,
- const WebRtc_UWord32 durationMs)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::RecordNotification(id=%d, durationMs=%d)",
- id, durationMs);
-
- // Not implement yet
-}
-
-void OutputMixer::PlayFileEnded(const WebRtc_Word32 id)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::PlayFileEnded(id=%d)", id);
-
- // not needed
-}
-
-void OutputMixer::RecordFileEnded(const WebRtc_Word32 id)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::RecordFileEnded(id=%d)", id);
- assert(id == _instanceId);
-
- CriticalSectionScoped cs(_fileCritSect);
- _outputFileRecording = false;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::RecordFileEnded() =>"
- "output file recorder module is shutdown");
-}
-
-WebRtc_Word32
-OutputMixer::Create(OutputMixer*& mixer, const WebRtc_UWord32 instanceId)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, instanceId,
- "OutputMixer::Create(instanceId=%d)", instanceId);
- mixer = new OutputMixer(instanceId);
- if (mixer == NULL)
- {
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, instanceId,
- "OutputMixer::Create() unable to allocate memory for"
- "mixer");
- return -1;
- }
- return 0;
-}
-
-OutputMixer::OutputMixer(const WebRtc_UWord32 instanceId) :
- _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _fileCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _mixerModule(*AudioConferenceMixer::Create(instanceId)),
- _audioLevel(),
- _dtmfGenerator(instanceId),
- _instanceId(instanceId),
- _externalMediaCallbackPtr(NULL),
- _externalMedia(false),
- _panLeft(1.0f),
- _panRight(1.0f),
- _mixingFrequencyHz(8000),
- _outputFileRecorderPtr(NULL),
- _outputFileRecording(false)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::OutputMixer() - ctor");
-
- if ((_mixerModule.RegisterMixedStreamCallback(*this) == -1) ||
- (_mixerModule.RegisterMixerStatusCallback(*this, 100) == -1))
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::OutputMixer() failed to register mixer"
- "callbacks");
- }
-
- _dtmfGenerator.Init();
-}
-
-void
-OutputMixer::Destroy(OutputMixer*& mixer)
-{
- if (mixer)
- {
- delete mixer;
- mixer = NULL;
- }
-}
-
-OutputMixer::~OutputMixer()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::~OutputMixer() - dtor");
- if (_externalMedia)
- {
- DeRegisterExternalMediaProcessing();
- }
- {
- CriticalSectionScoped cs(_fileCritSect);
- if (_outputFileRecorderPtr)
- {
- _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
- _outputFileRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
- _outputFileRecorderPtr = NULL;
- }
- }
- _mixerModule.UnRegisterMixerStatusCallback();
- _mixerModule.UnRegisterMixedStreamCallback();
- delete &_mixerModule;
- delete &_callbackCritSect;
- delete &_fileCritSect;
-}
-
-WebRtc_Word32
-OutputMixer::SetEngineInformation(voe::Statistics& engineStatistics)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::SetEngineInformation()");
- _engineStatisticsPtr = &engineStatistics;
- return 0;
-}
-
-WebRtc_Word32
-OutputMixer::SetAudioProcessingModule(
- AudioProcessing* audioProcessingModule)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::SetAudioProcessingModule("
- "audioProcessingModule=0x%x)", audioProcessingModule);
- _audioProcessingModulePtr = audioProcessingModule;
- return 0;
-}
-
-int OutputMixer::RegisterExternalMediaProcessing(
- VoEMediaProcess& proccess_object)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::RegisterExternalMediaProcessing()");
-
- CriticalSectionScoped cs(_callbackCritSect);
- _externalMediaCallbackPtr = &proccess_object;
- _externalMedia = true;
-
- return 0;
-}
-
-int OutputMixer::DeRegisterExternalMediaProcessing()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::DeRegisterExternalMediaProcessing()");
-
- CriticalSectionScoped cs(_callbackCritSect);
- _externalMedia = false;
- _externalMediaCallbackPtr = NULL;
-
- return 0;
-}
-
-int OutputMixer::PlayDtmfTone(WebRtc_UWord8 eventCode, int lengthMs,
- int attenuationDb)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "OutputMixer::PlayDtmfTone()");
- if (_dtmfGenerator.AddTone(eventCode, lengthMs, attenuationDb) != 0)
- {
- _engineStatisticsPtr->SetLastError(VE_STILL_PLAYING_PREV_DTMF,
- kTraceError,
- "OutputMixer::PlayDtmfTone()");
- return -1;
- }
- return 0;
-}
-
-int OutputMixer::StartPlayingDtmfTone(WebRtc_UWord8 eventCode,
- int attenuationDb)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "OutputMixer::StartPlayingDtmfTone()");
- if (_dtmfGenerator.StartTone(eventCode, attenuationDb) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_STILL_PLAYING_PREV_DTMF,
- kTraceError,
- "OutputMixer::StartPlayingDtmfTone())");
- return -1;
- }
- return 0;
-}
-
-int OutputMixer::StopPlayingDtmfTone()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "OutputMixer::StopPlayingDtmfTone()");
- return (_dtmfGenerator.StopTone());
-}
-
-WebRtc_Word32
-OutputMixer::SetMixabilityStatus(MixerParticipant& participant,
- const bool mixable)
-{
- return _mixerModule.SetMixabilityStatus(participant, mixable);
-}
-
-WebRtc_Word32
-OutputMixer::SetAnonymousMixabilityStatus(MixerParticipant& participant,
- const bool mixable)
-{
- return _mixerModule.SetAnonymousMixabilityStatus(participant,mixable);
-}
-
-WebRtc_Word32
-OutputMixer::MixActiveChannels()
-{
- return _mixerModule.Process();
-}
-
-int
-OutputMixer::GetSpeechOutputLevel(WebRtc_UWord32& level)
-{
- WebRtc_Word8 currentLevel = _audioLevel.Level();
- level = static_cast<WebRtc_UWord32> (currentLevel);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetSpeechOutputLevel() => level=%u", level);
- return 0;
-}
-
-int
-OutputMixer::GetSpeechOutputLevelFullRange(WebRtc_UWord32& level)
-{
- WebRtc_Word16 currentLevel = _audioLevel.LevelFullRange();
- level = static_cast<WebRtc_UWord32> (currentLevel);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetSpeechOutputLevelFullRange() => level=%u", level);
- return 0;
-}
-
-int
-OutputMixer::SetOutputVolumePan(float left, float right)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::SetOutputVolumePan()");
- _panLeft = left;
- _panRight = right;
- return 0;
-}
-
-int
-OutputMixer::GetOutputVolumePan(float& left, float& right)
-{
- left = _panLeft;
- right = _panRight;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetOutputVolumePan() => left=%2.1f, right=%2.1f",
- left, right);
- return 0;
-}
-
-int OutputMixer::StartRecordingPlayout(const char* fileName,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::StartRecordingPlayout(fileName=%s)", fileName);
-
- if (_outputFileRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
- "StartRecordingPlayout() is already recording");
- return 0;
- }
-
- FileFormats format;
- const WebRtc_UWord32 notificationTime(0);
- CodecInst dummyCodec={100,"L16",16000,320,1,320000};
-
- if (codecInst != NULL && codecInst->channels != 1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingPlayout() invalid compression");
- return(-1);
- }
- if(codecInst == NULL)
- {
- format = kFileFormatPcm16kHzFile;
- codecInst=&dummyCodec;
- }
- else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
- {
- format = kFileFormatWavFile;
- }
- else
- {
- format = kFileFormatCompressedFile;
- }
-
- CriticalSectionScoped cs(_fileCritSect);
-
- // Destroy the old instance
- if (_outputFileRecorderPtr)
- {
- _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
- _outputFileRecorderPtr = NULL;
- }
-
- _outputFileRecorderPtr = FileRecorder::CreateFileRecorder(
- _instanceId,
- (const FileFormats)format);
- if (_outputFileRecorderPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingPlayout() fileRecorder format isnot correct");
- return -1;
- }
-
- if (_outputFileRecorderPtr->StartRecordingAudioFile(
- fileName,
- (const CodecInst&)*codecInst,
- notificationTime) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- _outputFileRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
- _outputFileRecorderPtr = NULL;
- return -1;
- }
- _outputFileRecorderPtr->RegisterModuleFileCallback(this);
- _outputFileRecording = true;
-
- return 0;
-}
-
-int OutputMixer::StartRecordingPlayout(OutStream* stream,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::StartRecordingPlayout()");
-
- if (_outputFileRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
- "StartRecordingPlayout() is already recording");
- return 0;
- }
-
- FileFormats format;
- const WebRtc_UWord32 notificationTime(0);
- CodecInst dummyCodec={100,"L16",16000,320,1,320000};
-
- if (codecInst != NULL && codecInst->channels != 1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingPlayout() invalid compression");
- return(-1);
- }
- if(codecInst == NULL)
- {
- format = kFileFormatPcm16kHzFile;
- codecInst=&dummyCodec;
- }
- else if((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
- {
- format = kFileFormatWavFile;
- }
- else
- {
- format = kFileFormatCompressedFile;
- }
-
- CriticalSectionScoped cs(_fileCritSect);
-
- // Destroy the old instance
- if (_outputFileRecorderPtr)
- {
- _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
- _outputFileRecorderPtr = NULL;
- }
-
- _outputFileRecorderPtr = FileRecorder::CreateFileRecorder(
- _instanceId,
- (const FileFormats)format);
- if (_outputFileRecorderPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingPlayout() fileRecorder format isnot correct");
- return -1;
- }
-
- if (_outputFileRecorderPtr->StartRecordingAudioFile(*stream,
- *codecInst,
- notificationTime) != 0)
- {
- _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- _outputFileRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
- _outputFileRecorderPtr = NULL;
- return -1;
- }
-
- _outputFileRecorderPtr->RegisterModuleFileCallback(this);
- _outputFileRecording = true;
-
- return 0;
-}
-
-int OutputMixer::StopRecordingPlayout()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::StopRecordingPlayout()");
-
- if (!_outputFileRecording)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "StopRecordingPlayout() file isnot recording");
- return -1;
- }
-
- CriticalSectionScoped cs(_fileCritSect);
-
- if (_outputFileRecorderPtr->StopRecording() != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_STOP_RECORDING_FAILED, kTraceError,
- "StopRecording(), could not stop recording");
- return -1;
- }
- _outputFileRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_outputFileRecorderPtr);
- _outputFileRecorderPtr = NULL;
- _outputFileRecording = false;
-
- return 0;
-}
-
-WebRtc_Word32
-OutputMixer::GetMixedAudio(const WebRtc_Word32 desiredFreqHz,
- const WebRtc_UWord8 channels,
- AudioFrame& audioFrame)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::GetMixedAudio(desiredFreqHz=%d, channels=&d)",
- desiredFreqHz, channels);
-
- audioFrame = _audioFrame;
-
- // --- Record playout if enabled
- {
- CriticalSectionScoped cs(_fileCritSect);
- if (_outputFileRecording)
- {
- assert(audioFrame._audioChannel == 1);
-
- if (_outputFileRecorderPtr)
- {
- _outputFileRecorderPtr->RecordAudioToFile(audioFrame);
- }
- }
- }
-
- int outLen(0);
-
- if (audioFrame._audioChannel == 1)
- {
- if (_resampler.ResetIfNeeded(audioFrame._frequencyInHz,
- desiredFreqHz,
- kResamplerSynchronous) != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::GetMixedAudio() unable to resample - 1");
- return -1;
- }
- }
- else
- {
- if (_resampler.ResetIfNeeded(audioFrame._frequencyInHz,
- desiredFreqHz,
- kResamplerSynchronousStereo) != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::GetMixedAudio() unable to resample - 2");
- return -1;
- }
- }
- if (_resampler.Push(
- _audioFrame._payloadData,
- _audioFrame._payloadDataLengthInSamples*_audioFrame._audioChannel,
- audioFrame._payloadData,
- AudioFrame::kMaxAudioFrameSizeSamples,
- outLen) == 0)
- {
- // Ensure that output from resampler matches the audio-frame format.
- // Example: 10ms stereo output at 48kHz => outLen = 960 =>
- // convert _payloadDataLengthInSamples to 480
- audioFrame._payloadDataLengthInSamples =
- (outLen / _audioFrame._audioChannel);
- audioFrame._frequencyInHz = desiredFreqHz;
- }
- else
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::GetMixedAudio() resampling failed");
- return -1;
- }
-
- if ((channels == 2) && (audioFrame._audioChannel == 1))
- {
- AudioFrameOperations::MonoToStereo(audioFrame);
- }
-
- return 0;
-}
-
-WebRtc_Word32
-OutputMixer::DoOperationsOnCombinedSignal()
-{
- if (_audioFrame._frequencyInHz != _mixingFrequencyHz)
- {
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "OutputMixer::DoOperationsOnCombinedSignal() => "
- "mixing frequency = %d", _audioFrame._frequencyInHz);
- _mixingFrequencyHz = _audioFrame._frequencyInHz;
- }
-
- // --- Insert inband Dtmf tone
- if (_dtmfGenerator.IsAddingTone())
- {
- InsertInbandDtmfTone();
- }
-
- // Scale left and/or right channel(s) if balance is active
- if (_panLeft != 1.0 || _panRight != 1.0)
- {
- if (_audioFrame._audioChannel == 1)
- {
- AudioFrameOperations::MonoToStereo(_audioFrame);
- }
- else
- {
- // Pure stereo mode (we are receiving a stereo signal).
- }
-
- assert(_audioFrame._audioChannel == 2);
- AudioFrameOperations::Scale(_panLeft, _panRight, _audioFrame);
- }
-
- // --- Far-end Voice Quality Enhancement (AudioProcessing Module)
-
- APMAnalyzeReverseStream();
-
- // --- External media processing
-
- if (_externalMedia)
- {
- CriticalSectionScoped cs(_callbackCritSect);
- const bool isStereo = (_audioFrame._audioChannel == 2);
- if (_externalMediaCallbackPtr)
- {
- _externalMediaCallbackPtr->Process(
- -1,
- kPlaybackAllChannelsMixed,
- (WebRtc_Word16*)_audioFrame._payloadData,
- _audioFrame._payloadDataLengthInSamples,
- _audioFrame._frequencyInHz,
- isStereo);
- }
- }
-
- // --- Measure audio level (0-9) for the combined signal
- _audioLevel.ComputeLevel(_audioFrame);
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// Private methods
-// ----------------------------------------------------------------------------
-
-int
-OutputMixer::APMAnalyzeReverseStream()
-{
- int outLen(0);
- AudioFrame audioFrame = _audioFrame;
-
- // Convert from mixing frequency to APM frequency.
- // Sending side determines APM frequency.
-
- if (audioFrame._audioChannel == 1)
- {
- _apmResampler.ResetIfNeeded(_audioFrame._frequencyInHz,
- _audioProcessingModulePtr->sample_rate_hz(),
- kResamplerSynchronous);
- }
- else
- {
- _apmResampler.ResetIfNeeded(_audioFrame._frequencyInHz,
- _audioProcessingModulePtr->sample_rate_hz(),
- kResamplerSynchronousStereo);
- }
- if (_apmResampler.Push(
- _audioFrame._payloadData,
- _audioFrame._payloadDataLengthInSamples*_audioFrame._audioChannel,
- audioFrame._payloadData,
- AudioFrame::kMaxAudioFrameSizeSamples,
- outLen) == 0)
- {
- audioFrame._payloadDataLengthInSamples =
- (outLen / _audioFrame._audioChannel);
- audioFrame._frequencyInHz = _audioProcessingModulePtr->sample_rate_hz();
- }
-
- if (audioFrame._audioChannel == 2)
- {
- AudioFrameOperations::StereoToMono(audioFrame);
- }
-
- // Perform far-end APM analyze
-
- if (_audioProcessingModulePtr->AnalyzeReverseStream(&audioFrame) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId,-1),
- "AudioProcessingModule::AnalyzeReverseStream() => error");
- }
-
- return 0;
-}
-
-int
-OutputMixer::InsertInbandDtmfTone()
-{
- WebRtc_UWord16 sampleRate(0);
- _dtmfGenerator.GetSampleRate(sampleRate);
- if (sampleRate != _audioFrame._frequencyInHz)
- {
- // Update sample rate of Dtmf tone since the mixing frequency changed.
- _dtmfGenerator.SetSampleRate(
- (WebRtc_UWord16)(_audioFrame._frequencyInHz));
- // Reset the tone to be added taking the new sample rate into account.
- _dtmfGenerator.ResetTone();
- }
-
- WebRtc_Word16 toneBuffer[320];
- WebRtc_UWord16 toneSamples(0);
- if (_dtmfGenerator.Get10msTone(toneBuffer, toneSamples) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "OutputMixer::InsertInbandDtmfTone() inserting Dtmf"
- "tone failed");
- return -1;
- }
-
- // replace mixed audio with Dtmf tone
- if (_audioFrame._audioChannel == 1)
- {
- // mono
- memcpy(_audioFrame._payloadData, toneBuffer, sizeof(WebRtc_Word16)
- * toneSamples);
- } else
- {
- // stereo
- for (int i = 0; i < _audioFrame._payloadDataLengthInSamples; i++)
- {
- _audioFrame._payloadData[2 * i] = toneBuffer[i];
- _audioFrame._payloadData[2 * i + 1] = 0;
- }
- }
- assert(_audioFrame._payloadDataLengthInSamples == toneSamples);
-
- return 0;
-}
-
-} // namespace voe
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/output_mixer.h b/trunk/src/voice_engine/main/source/output_mixer.h
deleted file mode 100644
index a5df185..0000000
--- a/trunk/src/voice_engine/main/source/output_mixer.h
+++ /dev/null
@@ -1,162 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_OUTPUT_MIXER_H
-#define WEBRTC_VOICE_ENGINE_OUTPUT_MIXER_H
-
-#include "audio_conference_mixer.h"
-#include "audio_conference_mixer_defines.h"
-#include "common_types.h"
-#include "dtmf_inband.h"
-#include "file_recorder.h"
-#include "level_indicator.h"
-#include "resampler.h"
-#include "voice_engine_defines.h"
-
-namespace webrtc {
-
-class AudioProcessing;
-class CriticalSectionWrapper;
-class FileWrapper;
-class VoEMediaProcess;
-
-namespace voe {
-
-class Statistics;
-
-class OutputMixer : public AudioMixerOutputReceiver,
- public AudioMixerStatusReceiver,
- public FileCallback
-{
-public:
- static WebRtc_Word32 Create(OutputMixer*& mixer,
- const WebRtc_UWord32 instanceId);
-
- static void Destroy(OutputMixer*& mixer);
-
- WebRtc_Word32 SetEngineInformation(Statistics& engineStatistics);
-
- WebRtc_Word32 SetAudioProcessingModule(
- AudioProcessing* audioProcessingModule);
-
- // VoEExternalMedia
- int RegisterExternalMediaProcessing(
- VoEMediaProcess& proccess_object);
-
- int DeRegisterExternalMediaProcessing();
-
- // VoEDtmf
- int PlayDtmfTone(WebRtc_UWord8 eventCode,
- int lengthMs,
- int attenuationDb);
-
- int StartPlayingDtmfTone(WebRtc_UWord8 eventCode,
- int attenuationDb);
-
- int StopPlayingDtmfTone();
-
- WebRtc_Word32 MixActiveChannels();
-
- WebRtc_Word32 DoOperationsOnCombinedSignal();
-
- WebRtc_Word32 SetMixabilityStatus(MixerParticipant& participant,
- const bool mixable);
-
- WebRtc_Word32 SetAnonymousMixabilityStatus(MixerParticipant& participant,
- const bool mixable);
-
- WebRtc_Word32 GetMixedAudio(const WebRtc_Word32 desiredFreqHz,
- const WebRtc_UWord8 channels,
- AudioFrame& audioFrame);
-
- // VoEVolumeControl
- int GetSpeechOutputLevel(WebRtc_UWord32& level);
-
- int GetSpeechOutputLevelFullRange(WebRtc_UWord32& level);
-
- int SetOutputVolumePan(float left, float right);
-
- int GetOutputVolumePan(float& left, float& right);
-
- // VoEFile
- int StartRecordingPlayout(const char* fileName,
- const CodecInst* codecInst);
-
- int StartRecordingPlayout(OutStream* stream,
- const CodecInst* codecInst);
- int StopRecordingPlayout();
-
- virtual ~OutputMixer();
-
-public: // from AudioMixerOutputReceiver
- virtual void NewMixedAudio(
- const WebRtc_Word32 id,
- const AudioFrame& generalAudioFrame,
- const AudioFrame** uniqueAudioFrames,
- const WebRtc_UWord32 size);
-
-public: // from AudioMixerStatusReceiver
- virtual void MixedParticipants(
- const WebRtc_Word32 id,
- const ParticipantStatistics* participantStatistics,
- const WebRtc_UWord32 size);
-
- virtual void VADPositiveParticipants(
- const WebRtc_Word32 id,
- const ParticipantStatistics* participantStatistics,
- const WebRtc_UWord32 size);
-
- virtual void MixedAudioLevel(const WebRtc_Word32 id,
- const WebRtc_UWord32 level);
-
-public: // For file recording
- void PlayNotification(const WebRtc_Word32 id,
- const WebRtc_UWord32 durationMs);
-
- void RecordNotification(const WebRtc_Word32 id,
- const WebRtc_UWord32 durationMs);
-
- void PlayFileEnded(const WebRtc_Word32 id);
- void RecordFileEnded(const WebRtc_Word32 id);
-
-private:
- OutputMixer(const WebRtc_UWord32 instanceId);
- int APMAnalyzeReverseStream();
- int InsertInbandDtmfTone();
-
-private: // uses
- Statistics* _engineStatisticsPtr;
- AudioProcessing* _audioProcessingModulePtr;
-
-private: // owns
- CriticalSectionWrapper& _callbackCritSect;
- // protect the _outputFileRecorderPtr and _outputFileRecording
- CriticalSectionWrapper& _fileCritSect;
- AudioConferenceMixer& _mixerModule;
- AudioFrame _audioFrame;
- Resampler _resampler; // converts mixed audio to fit ADM format
- Resampler _apmResampler; // converts mixed audio to fit APM rate
- AudioLevel _audioLevel; // measures audio level for the combined signal
- DtmfInband _dtmfGenerator;
- int _instanceId;
- VoEMediaProcess* _externalMediaCallbackPtr;
- bool _externalMedia;
- float _panLeft;
- float _panRight;
- int _mixingFrequencyHz;
- FileRecorder* _outputFileRecorderPtr;
- bool _outputFileRecording;
-};
-
-} // namespace voe
-
-} // namespace werbtc
-
-#endif // VOICE_ENGINE_OUTPUT_MIXER_H
diff --git a/trunk/src/voice_engine/main/source/ref_count.cc b/trunk/src/voice_engine/main/source/ref_count.cc
deleted file mode 100644
index f1ed0be..0000000
--- a/trunk/src/voice_engine/main/source/ref_count.cc
+++ /dev/null
@@ -1,60 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "critical_section_wrapper.h"
-#include "ref_count.h"
-
-namespace webrtc {
-
-namespace voe {
-
-RefCount::RefCount() :
- _count(0),
- _crit(*CriticalSectionWrapper::CreateCriticalSection())
-{
-}
-
-RefCount::~RefCount()
-{
- delete &_crit;
-}
-
-RefCount&
-RefCount::operator++(int)
-{
- CriticalSectionScoped lock(_crit);
- _count++;
- return *this;
-}
-
-RefCount&
-RefCount::operator--(int)
-{
- CriticalSectionScoped lock(_crit);
- _count--;
- return *this;
-}
-
-void
-RefCount::Reset()
-{
- CriticalSectionScoped lock(_crit);
- _count = 0;
-}
-
-int
-RefCount::GetCount() const
-{
- return _count;
-}
-
-} // namespace voe
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/ref_count.h b/trunk/src/voice_engine/main/source/ref_count.h
deleted file mode 100644
index e8c0a81..0000000
--- a/trunk/src/voice_engine/main/source/ref_count.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_REF_COUNT_H
-#define WEBRTC_VOICE_ENGINE_REF_COUNT_H
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-namespace voe {
-
-class RefCount
-{
-public:
- RefCount();
- ~RefCount();
- RefCount& operator++(int);
- RefCount& operator--(int);
- void Reset();
- int GetCount() const;
-private:
- volatile int _count;
- CriticalSectionWrapper& _crit;
-};
-
-} // namespace voe
-
-} // namespace webrtc
-#endif // #ifndef WEBRTC_VOICE_ENGINE_REF_COUNT_H
diff --git a/trunk/src/voice_engine/main/source/shared_data.cc b/trunk/src/voice_engine/main/source/shared_data.cc
deleted file mode 100644
index 2362da5..0000000
--- a/trunk/src/voice_engine/main/source/shared_data.cc
+++ /dev/null
@@ -1,96 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "shared_data.h"
-
-#include "audio_processing.h"
-#include "critical_section_wrapper.h"
-#include "channel.h"
-#include "output_mixer.h"
-#include "trace.h"
-#include "transmit_mixer.h"
-
-namespace webrtc {
-
-namespace voe {
-
-static WebRtc_Word32 _gInstanceCounter = 0;
-
-SharedData::SharedData() :
- _instanceId(++_gInstanceCounter),
- _apiCritPtr(CriticalSectionWrapper::CreateCriticalSection()),
- _channelManager(_gInstanceCounter),
- _engineStatistics(_gInstanceCounter),
- _audioDevicePtr(NULL),
- _audioProcessingModulePtr(NULL),
- _moduleProcessThreadPtr(ProcessThread::CreateProcessThread()),
- _externalRecording(false),
- _externalPlayout(false)
-{
- Trace::CreateTrace();
- Trace::SetLevelFilter(WEBRTC_VOICE_ENGINE_DEFAULT_TRACE_FILTER);
- if (OutputMixer::Create(_outputMixerPtr, _gInstanceCounter) == 0)
- {
- _outputMixerPtr->SetEngineInformation(_engineStatistics);
- }
- if (TransmitMixer::Create(_transmitMixerPtr, _gInstanceCounter) == 0)
- {
- _transmitMixerPtr->SetEngineInformation(*_moduleProcessThreadPtr,
- _engineStatistics,
- _channelManager);
- }
- _audioDeviceLayer = AudioDeviceModule::kPlatformDefaultAudio;
-}
-
-SharedData::~SharedData()
-{
- OutputMixer::Destroy(_outputMixerPtr);
- TransmitMixer::Destroy(_transmitMixerPtr);
- if (_audioDevicePtr) {
- _audioDevicePtr->Release();
- }
- AudioProcessing::Destroy(_audioProcessingModulePtr);
- delete _apiCritPtr;
- ProcessThread::DestroyProcessThread(_moduleProcessThreadPtr);
- Trace::ReturnTrace();
-}
-
-WebRtc_UWord16
-SharedData::NumOfSendingChannels()
-{
- WebRtc_Word32 numOfChannels = _channelManager.NumOfChannels();
- if (numOfChannels <= 0)
- {
- return 0;
- }
-
- WebRtc_UWord16 nChannelsSending(0);
- WebRtc_Word32* channelsArray = new WebRtc_Word32[numOfChannels];
-
- _channelManager.GetChannelIds(channelsArray, numOfChannels);
- for (int i = 0; i < numOfChannels; i++)
- {
- voe::ScopedChannel sc(_channelManager, channelsArray[i]);
- Channel* chPtr = sc.ChannelPtr();
- if (chPtr)
- {
- if (chPtr->Sending())
- {
- nChannelsSending++;
- }
- }
- }
- delete [] channelsArray;
- return nChannelsSending;
-}
-
-} // namespace voe
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/shared_data.h b/trunk/src/voice_engine/main/source/shared_data.h
deleted file mode 100644
index afab37b..0000000
--- a/trunk/src/voice_engine/main/source/shared_data.h
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_SHARED_DATA_H
-#define WEBRTC_VOICE_ENGINE_SHARED_DATA_H
-
-#include "voice_engine_defines.h"
-
-#include "channel_manager.h"
-#include "statistics.h"
-#include "process_thread.h"
-
-#include "audio_device.h"
-#include "audio_processing.h"
-
-class ProcessThread;
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-namespace voe {
-
-class TransmitMixer;
-class OutputMixer;
-
-class SharedData
-{
-public:
- // Public accessors.
- WebRtc_UWord32 instance_id() const { return _instanceId; }
- Statistics& statistics() { return _engineStatistics; }
- ChannelManager& channel_manager() { return _channelManager; }
- AudioDeviceModule* audio_device() { return _audioDevicePtr; }
-
-protected:
- WebRtc_UWord16 NumOfSendingChannels();
-
-protected:
- const WebRtc_UWord32 _instanceId;
- CriticalSectionWrapper* _apiCritPtr;
- ChannelManager _channelManager;
- Statistics _engineStatistics;
- AudioDeviceModule* _audioDevicePtr;
- OutputMixer* _outputMixerPtr;
- TransmitMixer* _transmitMixerPtr;
- AudioProcessing* _audioProcessingModulePtr;
- ProcessThread* _moduleProcessThreadPtr;
-
-protected:
- bool _externalRecording;
- bool _externalPlayout;
-
- AudioDeviceModule::AudioLayer _audioDeviceLayer;
-
-protected:
- SharedData();
- virtual ~SharedData();
-};
-
-} // namespace voe
-
-} // namespace webrtc
-#endif // WEBRTC_VOICE_ENGINE_SHARED_DATA_H
diff --git a/trunk/src/voice_engine/main/source/statistics.cc b/trunk/src/voice_engine/main/source/statistics.cc
deleted file mode 100644
index a534030..0000000
--- a/trunk/src/voice_engine/main/source/statistics.cc
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cassert>
-#include <stdio.h>
-
-#include "statistics.h"
-
-#include "trace.h"
-#include "critical_section_wrapper.h"
-
-namespace webrtc {
-
-namespace voe {
-
-Statistics::Statistics(const WebRtc_UWord32 instanceId) :
- _critPtr(CriticalSectionWrapper::CreateCriticalSection()),
- _instanceId(instanceId),
- _lastError(0),
- _isInitialized(false)
-{
-}
-
-Statistics::~Statistics()
-{
- if (_critPtr)
- {
- delete _critPtr;
- _critPtr = NULL;
- }
-}
-
-WebRtc_Word32 Statistics::SetInitialized()
-{
- _isInitialized = true;
- return 0;
-}
-
-WebRtc_Word32 Statistics::SetUnInitialized()
-{
- _isInitialized = false;
- return 0;
-}
-
-bool Statistics::Initialized() const
-{
- return _isInitialized;
-}
-
-WebRtc_Word32 Statistics::SetLastError(const WebRtc_Word32 error) const
-{
- CriticalSectionScoped cs(*_critPtr);
- _lastError = error;
- return 0;
-}
-
-WebRtc_Word32 Statistics::SetLastError(const WebRtc_Word32 error,
- const TraceLevel level) const
-{
- CriticalSectionScoped cs(*_critPtr);
- _lastError = error;
- WEBRTC_TRACE(level, kTraceVoice, VoEId(_instanceId,-1),
- "error code is set to %d",
- _lastError);
- return 0;
-}
-
-WebRtc_Word32 Statistics::SetLastError(
- const WebRtc_Word32 error,
- const TraceLevel level, const char* msg) const
-{
- CriticalSectionScoped cs(*_critPtr);
- char traceMessage[KTraceMaxMessageSize];
- assert(strlen(msg) < KTraceMaxMessageSize);
- _lastError = error;
- sprintf(traceMessage, "%s (error=%d)", msg, error);
- WEBRTC_TRACE(level, kTraceVoice, VoEId(_instanceId,-1), "%s",
- traceMessage);
- return 0;
-}
-
-WebRtc_Word32 Statistics::LastError() const
-{
- CriticalSectionScoped cs(*_critPtr);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "LastError() => %d", _lastError);
- return _lastError;
-}
-
-} // namespace voe
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/statistics.h b/trunk/src/voice_engine/main/source/statistics.h
deleted file mode 100644
index fc0bf8c..0000000
--- a/trunk/src/voice_engine/main/source/statistics.h
+++ /dev/null
@@ -1,54 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_STATISTICS_H
-#define WEBRTC_VOICE_ENGINE_STATISTICS_H
-
-#include "common_types.h"
-#include "typedefs.h"
-#include "voice_engine_defines.h"
-#include "voe_errors.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-
-namespace voe {
-
-class Statistics
-{
- public:
- enum {KTraceMaxMessageSize = 256};
- public:
- Statistics(const WebRtc_UWord32 instanceId);
- ~Statistics();
-
- WebRtc_Word32 SetInitialized();
- WebRtc_Word32 SetUnInitialized();
- bool Initialized() const;
- WebRtc_Word32 SetLastError(const WebRtc_Word32 error) const;
- WebRtc_Word32 SetLastError(const WebRtc_Word32 error,
- const TraceLevel level) const;
- WebRtc_Word32 SetLastError(const WebRtc_Word32 error,
- const TraceLevel level,
- const char* msg) const;
- WebRtc_Word32 LastError() const;
-
- private:
- CriticalSectionWrapper* _critPtr;
- const WebRtc_UWord32 _instanceId;
- mutable WebRtc_Word32 _lastError;
- bool _isInitialized;
-};
-
-} // namespace voe
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_STATISTICS_H
diff --git a/trunk/src/voice_engine/main/source/transmit_mixer.cc b/trunk/src/voice_engine/main/source/transmit_mixer.cc
deleted file mode 100644
index d131a85..0000000
--- a/trunk/src/voice_engine/main/source/transmit_mixer.cc
+++ /dev/null
@@ -1,1416 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "transmit_mixer.h"
-
-#include "audio_frame_operations.h"
-#include "channel.h"
-#include "channel_manager.h"
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "statistics.h"
-#include "trace.h"
-#include "utility.h"
-#include "voe_base_impl.h"
-#include "voe_external_media.h"
-
-#define WEBRTC_ABS(a) (((a) < 0) ? -(a) : (a))
-
-namespace webrtc {
-
-namespace voe {
-
-void
-TransmitMixer::OnPeriodicProcess()
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::OnPeriodicProcess()");
-
-#if defined(WEBRTC_VOICE_ENGINE_TYPING_DETECTION)
- if (_typingNoiseWarning > 0)
- {
- CriticalSectionScoped cs(_callbackCritSect);
- if (_voiceEngineObserverPtr)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::OnPeriodicProcess() => "
- "CallbackOnError(VE_TYPING_NOISE_WARNING)");
- _voiceEngineObserverPtr->CallbackOnError(-1,
- VE_TYPING_NOISE_WARNING);
- }
- _typingNoiseWarning = 0;
- }
-#endif
-
- if (_saturationWarning > 0)
- {
- CriticalSectionScoped cs(_callbackCritSect);
- if (_voiceEngineObserverPtr)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::OnPeriodicProcess() =>"
- " CallbackOnError(VE_SATURATION_WARNING)");
- _voiceEngineObserverPtr->CallbackOnError(-1, VE_SATURATION_WARNING);
- }
- _saturationWarning = 0;
- }
-
- if (_noiseWarning > 0)
- {
- CriticalSectionScoped cs(_callbackCritSect);
- if (_voiceEngineObserverPtr)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::OnPeriodicProcess() =>"
- "CallbackOnError(VE_NOISE_WARNING)");
- _voiceEngineObserverPtr->CallbackOnError(-1, VE_NOISE_WARNING);
- }
- _noiseWarning = 0;
- }
-}
-
-
-void TransmitMixer::PlayNotification(const WebRtc_Word32 id,
- const WebRtc_UWord32 durationMs)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::PlayNotification(id=%d, durationMs=%d)",
- id, durationMs);
-
- // Not implement yet
-}
-
-void TransmitMixer::RecordNotification(const WebRtc_Word32 id,
- const WebRtc_UWord32 durationMs)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "TransmitMixer::RecordNotification(id=%d, durationMs=%d)",
- id, durationMs);
-
- // Not implement yet
-}
-
-void TransmitMixer::PlayFileEnded(const WebRtc_Word32 id)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::PlayFileEnded(id=%d)", id);
-
- assert(id == _filePlayerId);
-
- CriticalSectionScoped cs(_critSect);
-
- _filePlaying = false;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::PlayFileEnded() =>"
- "file player module is shutdown");
-}
-
-void
-TransmitMixer::RecordFileEnded(const WebRtc_Word32 id)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RecordFileEnded(id=%d)", id);
-
- if (id == _fileRecorderId)
- {
- CriticalSectionScoped cs(_critSect);
- _fileRecording = false;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RecordFileEnded() => fileRecorder module"
- "is shutdown");
- } else if (id == _fileCallRecorderId)
- {
- CriticalSectionScoped cs(_critSect);
- _fileCallRecording = false;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RecordFileEnded() => fileCallRecorder"
- "module is shutdown");
- }
-}
-
-WebRtc_Word32
-TransmitMixer::Create(TransmitMixer*& mixer, const WebRtc_UWord32 instanceId)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, -1),
- "TransmitMixer::Create(instanceId=%d)", instanceId);
- mixer = new TransmitMixer(instanceId);
- if (mixer == NULL)
- {
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(instanceId, -1),
- "TransmitMixer::Create() unable to allocate memory"
- "for mixer");
- return -1;
- }
- return 0;
-}
-
-void
-TransmitMixer::Destroy(TransmitMixer*& mixer)
-{
- if (mixer)
- {
- delete mixer;
- mixer = NULL;
- }
-}
-
-TransmitMixer::TransmitMixer(const WebRtc_UWord32 instanceId) :
- _engineStatisticsPtr(NULL),
- _channelManagerPtr(NULL),
- _audioProcessingModulePtr(NULL),
- _voiceEngineObserverPtr(NULL),
- _processThreadPtr(NULL),
- _filePlayerPtr(NULL),
- _fileRecorderPtr(NULL),
- _fileCallRecorderPtr(NULL),
- // Avoid conflict with other channels by adding 1024 - 1026,
- // won't use as much as 1024 channels.
- _filePlayerId(instanceId + 1024),
- _fileRecorderId(instanceId + 1025),
- _fileCallRecorderId(instanceId + 1026),
- _filePlaying(false),
- _fileRecording(false),
- _fileCallRecording(false),
- _audioLevel(),
- _critSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
- _timeActive(0),
- _penaltyCounter(0),
- _typingNoiseWarning(0),
-#endif
- _saturationWarning(0),
- _noiseWarning(0),
- _instanceId(instanceId),
- _mixFileWithMicrophone(false),
- _captureLevel(0),
- _externalMedia(false),
- _externalMediaCallbackPtr(NULL),
- _mute(false),
- _remainingMuteMicTimeMs(0),
- _mixingFrequency(0),
- _includeAudioLevelIndication(false)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::TransmitMixer() - ctor");
-}
-
-TransmitMixer::~TransmitMixer()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::~TransmitMixer() - dtor");
- _monitorModule.DeRegisterObserver();
- if (_processThreadPtr)
- {
- _processThreadPtr->DeRegisterModule(&_monitorModule);
- }
- if (_externalMedia)
- {
- DeRegisterExternalMediaProcessing();
- }
- {
- CriticalSectionScoped cs(_critSect);
- if (_fileRecorderPtr)
- {
- _fileRecorderPtr->RegisterModuleFileCallback(NULL);
- _fileRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
- _fileRecorderPtr = NULL;
- }
- if (_fileCallRecorderPtr)
- {
- _fileCallRecorderPtr->RegisterModuleFileCallback(NULL);
- _fileCallRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
- _fileCallRecorderPtr = NULL;
- }
- if (_filePlayerPtr)
- {
- _filePlayerPtr->RegisterModuleFileCallback(NULL);
- _filePlayerPtr->StopPlayingFile();
- FilePlayer::DestroyFilePlayer(_filePlayerPtr);
- _filePlayerPtr = NULL;
- }
- }
- delete &_critSect;
- delete &_callbackCritSect;
-}
-
-WebRtc_Word32
-TransmitMixer::SetEngineInformation(ProcessThread& processThread,
- Statistics& engineStatistics,
- ChannelManager& channelManager)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::SetEngineInformation()");
-
- _processThreadPtr = &processThread;
- _engineStatisticsPtr = &engineStatistics;
- _channelManagerPtr = &channelManager;
-
- if (_processThreadPtr->RegisterModule(&_monitorModule) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::SetEngineInformation() failed to"
- "register the monitor module");
- } else
- {
- _monitorModule.RegisterObserver(*this);
- }
-
- return 0;
-}
-
-WebRtc_Word32
-TransmitMixer::RegisterVoiceEngineObserver(VoiceEngineObserver& observer)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RegisterVoiceEngineObserver()");
- CriticalSectionScoped cs(_callbackCritSect);
-
- if (_voiceEngineObserverPtr)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "RegisterVoiceEngineObserver() observer already enabled");
- return -1;
- }
- _voiceEngineObserverPtr = &observer;
- return 0;
-}
-
-WebRtc_Word32
-TransmitMixer::SetAudioProcessingModule(AudioProcessing* audioProcessingModule)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::SetAudioProcessingModule("
- "audioProcessingModule=0x%x)",
- audioProcessingModule);
- _audioProcessingModulePtr = audioProcessingModule;
- return 0;
-}
-
-WebRtc_Word32
-TransmitMixer::PrepareDemux(const WebRtc_Word8* audioSamples,
- const WebRtc_UWord32 nSamples,
- const WebRtc_UWord8 nChannels,
- const WebRtc_UWord32 samplesPerSec,
- const WebRtc_UWord16 totalDelayMS,
- const WebRtc_Word32 clockDrift,
- const WebRtc_UWord16 currentMicLevel)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::PrepareDemux(nSamples=%u, nChannels=%u,"
- "samplesPerSec=%u, totalDelayMS=%u, clockDrift=%u,"
- "currentMicLevel=%u)", nSamples, nChannels, samplesPerSec,
- totalDelayMS, clockDrift, currentMicLevel);
-
-
- const int mixingFrequency = _mixingFrequency;
-
- ScopedChannel sc(*_channelManagerPtr);
- void* iterator(NULL);
- Channel* channelPtr = sc.GetFirstChannel(iterator);
- _mixingFrequency = 8000;
- while (channelPtr != NULL)
- {
- if (channelPtr->Sending())
- {
- CodecInst tmpCdc;
- channelPtr->GetSendCodec(tmpCdc);
- if (tmpCdc.plfreq > _mixingFrequency)
- _mixingFrequency = tmpCdc.plfreq;
- }
- channelPtr = sc.GetNextChannel(iterator);
- }
-
-
- // --- Resample input audio and create/store the initial audio frame
-
- if (GenerateAudioFrame((const WebRtc_Word16*) audioSamples,
- nSamples,
- nChannels,
- samplesPerSec,
- _mixingFrequency) == -1)
- {
- return -1;
- }
-
- // --- Near-end Voice Quality Enhancement (APM) processing
-
- APMProcessStream(totalDelayMS, clockDrift, currentMicLevel);
-
- // --- Annoying typing detection (utilizes the APM/VAD decision)
-
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
- TypingDetection();
-#endif
-
- // --- Mute during DTMF tone if direct feedback is enabled
-
- if (_remainingMuteMicTimeMs > 0)
- {
- AudioFrameOperations::Mute(_audioFrame);
- _remainingMuteMicTimeMs -= 10;
- if (_remainingMuteMicTimeMs < 0)
- {
- _remainingMuteMicTimeMs = 0;
- }
- }
-
- // --- Mute signal
-
- if (_mute)
- {
- AudioFrameOperations::Mute(_audioFrame);
- }
-
- // --- Measure audio level of speech after APM processing
-
- _audioLevel.ComputeLevel(_audioFrame);
-
- // --- Mix with file (does not affect the mixing frequency)
-
- if (_filePlaying)
- {
- MixOrReplaceAudioWithFile(_mixingFrequency);
- }
-
- // --- Record to file
-
- if (_fileRecording)
- {
- RecordAudioToFile(_mixingFrequency);
- }
-
- // --- External media processing
-
- if (_externalMedia)
- {
- CriticalSectionScoped cs(_callbackCritSect);
- const bool isStereo = (_audioFrame._audioChannel == 2);
- if (_externalMediaCallbackPtr)
- {
- _externalMediaCallbackPtr->Process(
- -1,
- kRecordingAllChannelsMixed,
- (WebRtc_Word16*) _audioFrame._payloadData,
- _audioFrame._payloadDataLengthInSamples,
- _audioFrame._frequencyInHz,
- isStereo);
- }
- }
-
- if (_mixingFrequency != mixingFrequency)
- {
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::TransmitMixer::PrepareDemux() => "
- "mixing frequency = %d",
- _mixingFrequency);
- }
-
- return 0;
-}
-
-
-
-WebRtc_Word32
-TransmitMixer::DemuxAndMix()
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::DemuxAndMix()");
-
- ScopedChannel sc(*_channelManagerPtr);
- void* iterator(NULL);
- Channel* channelPtr = sc.GetFirstChannel(iterator);
- while (channelPtr != NULL)
- {
- if (channelPtr->InputIsOnHold())
- {
- channelPtr->UpdateLocalTimeStamp();
- } else if (channelPtr->Sending())
- {
- // load temporary audioframe with current (mixed) microphone signal
- AudioFrame tmpAudioFrame = _audioFrame;
-
- channelPtr->Demultiplex(tmpAudioFrame);
- channelPtr->PrepareEncodeAndSend(_mixingFrequency);
- }
- channelPtr = sc.GetNextChannel(iterator);
- }
-
- return 0;
-}
-
-WebRtc_Word32
-TransmitMixer::EncodeAndSend()
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::EncodeAndSend()");
-
- ScopedChannel sc(*_channelManagerPtr);
- void* iterator(NULL);
- Channel* channelPtr = sc.GetFirstChannel(iterator);
- while (channelPtr != NULL)
- {
- if (channelPtr->Sending() && !channelPtr->InputIsOnHold())
- {
- channelPtr->EncodeAndSend();
- }
- channelPtr = sc.GetNextChannel(iterator);
- }
- return 0;
-}
-
-WebRtc_UWord32 TransmitMixer::CaptureLevel() const
-{
- return _captureLevel;
-}
-
-void
-TransmitMixer::UpdateMuteMicrophoneTime(const WebRtc_UWord32 lengthMs)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::UpdateMuteMicrophoneTime(lengthMs=%d)",
- lengthMs);
- _remainingMuteMicTimeMs = lengthMs;
-}
-
-WebRtc_Word32
-TransmitMixer::StopSend()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StopSend()");
- _audioLevel.Clear();
- return 0;
-}
-
-int TransmitMixer::StartPlayingFileAsMicrophone(const char* fileName,
- const bool loop,
- const FileFormats format,
- const int startPosition,
- const float volumeScaling,
- const int stopPosition,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StartPlayingFileAsMicrophone("
- "fileNameUTF8[]=%s,loop=%d, format=%d, volumeScaling=%5.3f,"
- " startPosition=%d, stopPosition=%d)", fileName, loop,
- format, volumeScaling, startPosition, stopPosition);
-
- if (_filePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceWarning,
- "StartPlayingFileAsMicrophone() is already playing");
- return 0;
- }
-
- CriticalSectionScoped cs(_critSect);
-
- // Destroy the old instance
- if (_filePlayerPtr)
- {
- _filePlayerPtr->RegisterModuleFileCallback(NULL);
- FilePlayer::DestroyFilePlayer(_filePlayerPtr);
- _filePlayerPtr = NULL;
- }
-
- // Dynamically create the instance
- _filePlayerPtr
- = FilePlayer::CreateFilePlayer(_filePlayerId,
- (const FileFormats) format);
-
- if (_filePlayerPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartPlayingFileAsMicrophone() filePlayer format isnot correct");
- return -1;
- }
-
- const WebRtc_UWord32 notificationTime(0);
-
- if (_filePlayerPtr->StartPlayingFile(
- fileName,
- loop,
- startPosition,
- volumeScaling,
- notificationTime,
- stopPosition,
- (const CodecInst*) codecInst) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to start file playout");
- _filePlayerPtr->StopPlayingFile();
- FilePlayer::DestroyFilePlayer(_filePlayerPtr);
- _filePlayerPtr = NULL;
- return -1;
- }
-
- _filePlayerPtr->RegisterModuleFileCallback(this);
- _filePlaying = true;
-
- return 0;
-}
-
-int TransmitMixer::StartPlayingFileAsMicrophone(InStream* stream,
- const FileFormats format,
- const int startPosition,
- const float volumeScaling,
- const int stopPosition,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "TransmitMixer::StartPlayingFileAsMicrophone(format=%d,"
- " volumeScaling=%5.3f, startPosition=%d, stopPosition=%d)",
- format, volumeScaling, startPosition, stopPosition);
-
- if (stream == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFileAsMicrophone() NULL as input stream");
- return -1;
- }
-
- if (_filePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_ALREADY_PLAYING, kTraceWarning,
- "StartPlayingFileAsMicrophone() is already playing");
- return 0;
- }
-
- CriticalSectionScoped cs(_critSect);
-
- // Destroy the old instance
- if (_filePlayerPtr)
- {
- _filePlayerPtr->RegisterModuleFileCallback(NULL);
- FilePlayer::DestroyFilePlayer(_filePlayerPtr);
- _filePlayerPtr = NULL;
- }
-
- // Dynamically create the instance
- _filePlayerPtr
- = FilePlayer::CreateFilePlayer(_filePlayerId,
- (const FileFormats) format);
-
- if (_filePlayerPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceWarning,
- "StartPlayingFileAsMicrophone() filePlayer format isnot correct");
- return -1;
- }
-
- const WebRtc_UWord32 notificationTime(0);
-
- if (_filePlayerPtr->StartPlayingFile(
- (InStream&) *stream,
- startPosition,
- volumeScaling,
- notificationTime,
- stopPosition,
- (const CodecInst*) codecInst) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartPlayingFile() failed to start file playout");
- _filePlayerPtr->StopPlayingFile();
- FilePlayer::DestroyFilePlayer(_filePlayerPtr);
- _filePlayerPtr = NULL;
- return -1;
- }
- _filePlayerPtr->RegisterModuleFileCallback(this);
- _filePlaying = true;
-
- return 0;
-}
-
-int TransmitMixer::StopPlayingFileAsMicrophone()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- "TransmitMixer::StopPlayingFileAsMicrophone()");
-
- if (!_filePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceWarning,
- "StopPlayingFileAsMicrophone() isnot playing");
- return 0;
- }
-
- CriticalSectionScoped cs(_critSect);
-
- if (_filePlayerPtr->StopPlayingFile() != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_CANNOT_STOP_PLAYOUT, kTraceError,
- "StopPlayingFile() couldnot stop playing file");
- return -1;
- }
-
- _filePlayerPtr->RegisterModuleFileCallback(NULL);
- FilePlayer::DestroyFilePlayer(_filePlayerPtr);
- _filePlayerPtr = NULL;
- _filePlaying = false;
-
- return 0;
-}
-
-int TransmitMixer::IsPlayingFileAsMicrophone() const
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::IsPlayingFileAsMicrophone()");
- return _filePlaying;
-}
-
-int TransmitMixer::ScaleFileAsMicrophonePlayout(const float scale)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::ScaleFileAsMicrophonePlayout(scale=%5.3f)",
- scale);
-
- CriticalSectionScoped cs(_critSect);
-
- if (!_filePlaying)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "ScaleFileAsMicrophonePlayout() isnot playing file");
- return -1;
- }
-
- if ((_filePlayerPtr == NULL) ||
- (_filePlayerPtr->SetAudioScaling(scale) != 0))
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "SetAudioScaling() failed to scale playout");
- return -1;
- }
-
- return 0;
-}
-
-int TransmitMixer::StartRecordingMicrophone(const WebRtc_Word8* fileName,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StartRecordingMicrophone(fileName=%s)",
- fileName);
-
- if (_fileRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StartRecordingMicrophone() is already recording");
- return 0;
- }
-
- FileFormats format;
- const WebRtc_UWord32 notificationTime(0); // Not supported in VoE
- CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 };
-
- if (codecInst != NULL &&
- (codecInst->channels < 0 || codecInst->channels > 2))
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingMicrophone() invalid compression");
- return (-1);
- }
- if (codecInst == NULL)
- {
- format = kFileFormatPcm16kHzFile;
- codecInst = &dummyCodec;
- } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
- {
- format = kFileFormatWavFile;
- } else
- {
- format = kFileFormatCompressedFile;
- }
-
- CriticalSectionScoped cs(_critSect);
-
- // Destroy the old instance
- if (_fileRecorderPtr)
- {
- _fileRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
- _fileRecorderPtr = NULL;
- }
-
- _fileRecorderPtr =
- FileRecorder::CreateFileRecorder(_fileRecorderId,
- (const FileFormats) format);
- if (_fileRecorderPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingMicrophone() fileRecorder format isnot correct");
- return -1;
- }
-
- if (_fileRecorderPtr->StartRecordingAudioFile(
- fileName,
- (const CodecInst&) *codecInst,
- notificationTime) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- _fileRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
- _fileRecorderPtr = NULL;
- return -1;
- }
- _fileRecorderPtr->RegisterModuleFileCallback(this);
- _fileRecording = true;
-
- return 0;
-}
-
-int TransmitMixer::StartRecordingMicrophone(OutStream* stream,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StartRecordingMicrophone()");
-
- if (_fileRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StartRecordingMicrophone() is already recording");
- return 0;
- }
-
- FileFormats format;
- const WebRtc_UWord32 notificationTime(0); // Not supported in VoE
- CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 };
-
- if (codecInst != NULL && codecInst->channels != 1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingMicrophone() invalid compression");
- return (-1);
- }
- if (codecInst == NULL)
- {
- format = kFileFormatPcm16kHzFile;
- codecInst = &dummyCodec;
- } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
- {
- format = kFileFormatWavFile;
- } else
- {
- format = kFileFormatCompressedFile;
- }
-
- CriticalSectionScoped cs(_critSect);
-
- // Destroy the old instance
- if (_fileRecorderPtr)
- {
- _fileRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
- _fileRecorderPtr = NULL;
- }
-
- _fileRecorderPtr =
- FileRecorder::CreateFileRecorder(_fileRecorderId,
- (const FileFormats) format);
- if (_fileRecorderPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingMicrophone() fileRecorder format isnot correct");
- return -1;
- }
-
- if (_fileRecorderPtr->StartRecordingAudioFile(*stream,
- *codecInst,
- notificationTime) != 0)
- {
- _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- _fileRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
- _fileRecorderPtr = NULL;
- return -1;
- }
-
- _fileRecorderPtr->RegisterModuleFileCallback(this);
- _fileRecording = true;
-
- return 0;
-}
-
-
-int TransmitMixer::StopRecordingMicrophone()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StopRecordingMicrophone()");
-
- if (!_fileRecording)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "StopRecordingMicrophone() isnot recording");
- return -1;
- }
-
- CriticalSectionScoped cs(_critSect);
-
- if (_fileRecorderPtr->StopRecording() != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_STOP_RECORDING_FAILED, kTraceError,
- "StopRecording(), could not stop recording");
- return -1;
- }
- _fileRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_fileRecorderPtr);
- _fileRecorderPtr = NULL;
- _fileRecording = false;
-
- return 0;
-}
-
-int TransmitMixer::StartRecordingCall(const WebRtc_Word8* fileName,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StartRecordingCall(fileName=%s)", fileName);
-
- if (_fileCallRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StartRecordingCall() is already recording");
- return 0;
- }
-
- FileFormats format;
- const WebRtc_UWord32 notificationTime(0); // Not supported in VoE
- CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 };
-
- if (codecInst != NULL && codecInst->channels != 1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingCall() invalid compression");
- return (-1);
- }
- if (codecInst == NULL)
- {
- format = kFileFormatPcm16kHzFile;
- codecInst = &dummyCodec;
- } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
- {
- format = kFileFormatWavFile;
- } else
- {
- format = kFileFormatCompressedFile;
- }
-
- CriticalSectionScoped cs(_critSect);
-
- // Destroy the old instance
- if (_fileCallRecorderPtr)
- {
- _fileCallRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
- _fileCallRecorderPtr = NULL;
- }
-
- _fileCallRecorderPtr
- = FileRecorder::CreateFileRecorder(_fileCallRecorderId,
- (const FileFormats) format);
- if (_fileCallRecorderPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingCall() fileRecorder format isnot correct");
- return -1;
- }
-
- if (_fileCallRecorderPtr->StartRecordingAudioFile(
- fileName,
- (const CodecInst&) *codecInst,
- notificationTime) != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- _fileCallRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
- _fileCallRecorderPtr = NULL;
- return -1;
- }
- _fileCallRecorderPtr->RegisterModuleFileCallback(this);
- _fileCallRecording = true;
-
- return 0;
-}
-
-int TransmitMixer::StartRecordingCall(OutStream* stream,
- const CodecInst* codecInst)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StartRecordingCall()");
-
- if (_fileCallRecording)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StartRecordingCall() is already recording");
- return 0;
- }
-
- FileFormats format;
- const WebRtc_UWord32 notificationTime(0); // Not supported in VoE
- CodecInst dummyCodec = { 100, "L16", 16000, 320, 1, 320000 };
-
- if (codecInst != NULL && codecInst->channels != 1)
- {
- _engineStatisticsPtr->SetLastError(
- VE_BAD_ARGUMENT, kTraceError,
- "StartRecordingCall() invalid compression");
- return (-1);
- }
- if (codecInst == NULL)
- {
- format = kFileFormatPcm16kHzFile;
- codecInst = &dummyCodec;
- } else if ((STR_CASE_CMP(codecInst->plname,"L16") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMU") == 0) ||
- (STR_CASE_CMP(codecInst->plname,"PCMA") == 0))
- {
- format = kFileFormatWavFile;
- } else
- {
- format = kFileFormatCompressedFile;
- }
-
- CriticalSectionScoped cs(_critSect);
-
- // Destroy the old instance
- if (_fileCallRecorderPtr)
- {
- _fileCallRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
- _fileCallRecorderPtr = NULL;
- }
-
- _fileCallRecorderPtr =
- FileRecorder::CreateFileRecorder(_fileCallRecorderId,
- (const FileFormats) format);
- if (_fileCallRecorderPtr == NULL)
- {
- _engineStatisticsPtr->SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartRecordingCall() fileRecorder format isnot correct");
- return -1;
- }
-
- if (_fileCallRecorderPtr->StartRecordingAudioFile(*stream,
- *codecInst,
- notificationTime) != 0)
- {
- _engineStatisticsPtr->SetLastError(VE_BAD_FILE, kTraceError,
- "StartRecordingAudioFile() failed to start file recording");
- _fileCallRecorderPtr->StopRecording();
- FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
- _fileCallRecorderPtr = NULL;
- return -1;
- }
-
- _fileCallRecorderPtr->RegisterModuleFileCallback(this);
- _fileCallRecording = true;
-
- return 0;
-}
-
-int TransmitMixer::StopRecordingCall()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::StopRecordingCall()");
-
- if (!_fileCallRecording)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "StopRecordingCall() file isnot recording");
- return -1;
- }
-
- CriticalSectionScoped cs(_critSect);
-
- if (_fileCallRecorderPtr->StopRecording() != 0)
- {
- _engineStatisticsPtr->SetLastError(
- VE_STOP_RECORDING_FAILED, kTraceError,
- "StopRecording(), could not stop recording");
- return -1;
- }
-
- _fileCallRecorderPtr->RegisterModuleFileCallback(NULL);
- FileRecorder::DestroyFileRecorder(_fileCallRecorderPtr);
- _fileCallRecorderPtr = NULL;
- _fileCallRecording = false;
-
- return 0;
-}
-
-void
-TransmitMixer::SetMixWithMicStatus(bool mix)
-{
- _mixFileWithMicrophone = mix;
-}
-
-int TransmitMixer::RegisterExternalMediaProcessing(
- VoEMediaProcess& proccess_object)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RegisterExternalMediaProcessing()");
-
- CriticalSectionScoped cs(_callbackCritSect);
- _externalMediaCallbackPtr = &proccess_object;
- _externalMedia = true;
-
- return 0;
-}
-
-int TransmitMixer::DeRegisterExternalMediaProcessing()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::DeRegisterExternalMediaProcessing()");
-
- CriticalSectionScoped cs(_callbackCritSect);
- _externalMedia = false;
- _externalMediaCallbackPtr = NULL;
-
- return 0;
-}
-
-int
-TransmitMixer::SetMute(bool enable)
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::SetMute(enable=%d)", enable);
- _mute = enable;
- return 0;
-}
-
-bool
-TransmitMixer::Mute() const
-{
- return _mute;
-}
-
-WebRtc_Word8 TransmitMixer::AudioLevel() const
-{
- // Speech + file level [0,9]
- return _audioLevel.Level();
-}
-
-WebRtc_Word16 TransmitMixer::AudioLevelFullRange() const
-{
- // Speech + file level [0,32767]
- return _audioLevel.LevelFullRange();
-}
-
-bool TransmitMixer::IsRecordingCall()
-{
- return _fileCallRecording;
-}
-
-bool TransmitMixer::IsRecordingMic()
-{
-
- return _fileRecording;
-}
-
-WebRtc_Word32
-TransmitMixer::GenerateAudioFrame(const WebRtc_Word16 audioSamples[],
- const WebRtc_UWord32 nSamples,
- const WebRtc_UWord8 nChannels,
- const WebRtc_UWord32 samplesPerSec,
- const int mixingFrequency)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::GenerateAudioFrame(nSamples=%u,"
- "samplesPerSec=%u, mixingFrequency=%u)",
- nSamples, samplesPerSec, mixingFrequency);
-
- ResamplerType resampType = (nChannels == 1) ?
- kResamplerSynchronous : kResamplerSynchronousStereo;
-
-
- if (_audioResampler.ResetIfNeeded(samplesPerSec,
- mixingFrequency,
- resampType) != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::GenerateAudioFrame() unable to resample");
- return -1;
- }
- if (_audioResampler.Push(
- (WebRtc_Word16*) audioSamples,
- nSamples * nChannels,
- _audioFrame._payloadData,
- AudioFrame::kMaxAudioFrameSizeSamples,
- (int&) _audioFrame._payloadDataLengthInSamples) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::GenerateAudioFrame() resampling failed");
- return -1;
- }
-
- _audioFrame._payloadDataLengthInSamples /= nChannels;
- _audioFrame._id = _instanceId;
- _audioFrame._timeStamp = -1;
- _audioFrame._frequencyInHz = mixingFrequency;
- _audioFrame._speechType = AudioFrame::kNormalSpeech;
- _audioFrame._vadActivity = AudioFrame::kVadUnknown;
- _audioFrame._audioChannel = nChannels;
-
- return 0;
-}
-
-WebRtc_Word32 TransmitMixer::RecordAudioToFile(
- const WebRtc_UWord32 mixingFrequency)
-{
- CriticalSectionScoped cs(_critSect);
- if (_fileRecorderPtr == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RecordAudioToFile() filerecorder doesnot"
- "exist");
- return -1;
- }
-
- if (_fileRecorderPtr->RecordAudioToFile(_audioFrame) != 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::RecordAudioToFile() file recording"
- "failed");
- return -1;
- }
-
- return 0;
-}
-
-WebRtc_Word32 TransmitMixer::MixOrReplaceAudioWithFile(
- const int mixingFrequency)
-{
- WebRtc_Word16 fileBuffer[320];
-
- WebRtc_UWord32 fileSamples(0);
-
- {
- CriticalSectionScoped cs(_critSect);
- if (_filePlayerPtr == NULL)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, -1),
- "TransmitMixer::MixOrReplaceAudioWithFile()"
- "fileplayer doesnot exist");
- return -1;
- }
-
- if (_filePlayerPtr->Get10msAudioFromFile(fileBuffer,
- fileSamples,
- mixingFrequency) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::MixOrReplaceAudioWithFile() file"
- " mixing failed");
- return -1;
- }
- }
-
- if (_mixFileWithMicrophone)
- {
- Utility::MixWithSat(_audioFrame._payloadData,
- fileBuffer,
- (WebRtc_UWord16) fileSamples);
- assert(_audioFrame._payloadDataLengthInSamples == fileSamples);
- } else
- {
- // replace ACM audio with file
- _audioFrame.UpdateFrame(-1,
- -1,
- fileBuffer,
- (WebRtc_UWord16) fileSamples, mixingFrequency,
- AudioFrame::kNormalSpeech,
- AudioFrame::kVadUnknown,
- 1);
-
- }
- return 0;
-}
-
-WebRtc_Word32 TransmitMixer::APMProcessStream(
- const WebRtc_UWord16 totalDelayMS,
- const WebRtc_Word32 clockDrift,
- const WebRtc_UWord16 currentMicLevel)
-{
- WebRtc_UWord16 captureLevel(currentMicLevel);
-
- // Check if the number of input channels has changed. Retain the number
- // of output channels.
- if (_audioFrame._audioChannel !=
- _audioProcessingModulePtr->num_input_channels())
- {
- if (_audioProcessingModulePtr->set_num_channels(
- _audioFrame._audioChannel,
- _audioProcessingModulePtr->num_output_channels()))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "AudioProcessing::set_num_channels(%d, %d) => error",
- _audioFrame._frequencyInHz,
- _audioProcessingModulePtr->num_output_channels());
- }
- }
-
- // If the frequency has changed we need to change APM settings
- // Sending side is "master"
- if (_audioProcessingModulePtr->sample_rate_hz() !=
- _audioFrame._frequencyInHz)
- {
- if (_audioProcessingModulePtr->set_sample_rate_hz(
- _audioFrame._frequencyInHz))
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "AudioProcessing::set_sample_rate_hz(%u) => error",
- _audioFrame._frequencyInHz);
- }
- }
-
- if (_audioProcessingModulePtr->set_stream_delay_ms(totalDelayMS) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "AudioProcessing::set_stream_delay_ms(%u) => error",
- totalDelayMS);
- }
- if (_audioProcessingModulePtr->gain_control()->set_stream_analog_level(
- captureLevel) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "AudioProcessing::set_stream_analog_level(%u) => error",
- captureLevel);
- }
- if (_audioProcessingModulePtr->echo_cancellation()->
- is_drift_compensation_enabled())
- {
- if (_audioProcessingModulePtr->echo_cancellation()->
- set_stream_drift_samples(clockDrift) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "AudioProcessing::set_stream_drift_samples(%u) => error",
- clockDrift);
- }
- }
- if (_audioProcessingModulePtr->ProcessStream(&_audioFrame) == -1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "AudioProcessing::ProcessStream() => error");
- }
- captureLevel =
- _audioProcessingModulePtr->gain_control()->stream_analog_level();
-
- // Store new capture level (only updated when analog AGC is enabled)
- _captureLevel = captureLevel;
-
- // Log notifications
- if (_audioProcessingModulePtr->gain_control()->stream_is_saturated())
- {
- if (_saturationWarning == 1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::APMProcessStream() pending "
- "saturation warning exists");
- }
- _saturationWarning = 1; // triggers callback from moduleprocess thread
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::APMProcessStream() VE_SATURATION_WARNING "
- "message has been posted for callback");
- }
-
- return 0;
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
-int TransmitMixer::TypingDetection()
-{
- // We let the VAD determine if we're using this feature or not.
- if (_audioFrame._vadActivity == AudioFrame::kVadUnknown)
- {
- return (0);
- }
-
- int keyPressed = EventWrapper::KeyPressed();
-
- if (keyPressed < 0)
- {
- return (-1);
- }
-
- if (_audioFrame._vadActivity == AudioFrame::kVadActive)
- _timeActive++;
- else
- _timeActive = 0;
-
- if (keyPressed && (_audioFrame._vadActivity == AudioFrame::kVadActive)
- && (_timeActive < 10))
- {
- _penaltyCounter += 100;
- if (_penaltyCounter > 300)
- {
- if (_typingNoiseWarning == 1)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice,
- VoEId(_instanceId, -1),
- "TransmitMixer::TypingDetection() pending "
- "noise-saturation warning exists");
- }
- // triggers callback from the module process thread
- _typingNoiseWarning = 1;
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "TransmitMixer::TypingDetection() "
- "VE_TYPING_NOISE_WARNING message has been posted for"
- "callback");
- }
- }
-
- if (_penaltyCounter > 0)
- _penaltyCounter--;
-
- return (0);
-}
-#endif
-
-int TransmitMixer::GetMixingFrequency()
-{
- assert(_mixingFrequency!=0);
- return (_mixingFrequency);
-}
-
-} // namespace voe
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/transmit_mixer.h b/trunk/src/voice_engine/main/source/transmit_mixer.h
deleted file mode 100644
index 469b288..0000000
--- a/trunk/src/voice_engine/main/source/transmit_mixer.h
+++ /dev/null
@@ -1,222 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
-#define WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
-
-#include "common_types.h"
-#include "voe_base.h"
-#include "file_player.h"
-#include "file_recorder.h"
-#include "level_indicator.h"
-#include "module_common_types.h"
-#include "monitor_module.h"
-#include "resampler.h"
-#include "voice_engine_defines.h"
-
-
-namespace webrtc {
-
-class AudioProcessing;
-class ProcessThread;
-class VoEExternalMedia;
-class VoEMediaProcess;
-
-namespace voe {
-
-class ChannelManager;
-class MixedAudio;
-class Statistics;
-
-class TransmitMixer : public MonitorObserver,
- public FileCallback
-
-{
-public:
- static WebRtc_Word32 Create(TransmitMixer*& mixer,
- const WebRtc_UWord32 instanceId);
-
- static void Destroy(TransmitMixer*& mixer);
-
- WebRtc_Word32 SetEngineInformation(ProcessThread& processThread,
- Statistics& engineStatistics,
- ChannelManager& channelManager);
-
- WebRtc_Word32 SetAudioProcessingModule(
- AudioProcessing* audioProcessingModule);
-
- WebRtc_Word32 PrepareDemux(const WebRtc_Word8* audioSamples,
- const WebRtc_UWord32 nSamples,
- const WebRtc_UWord8 nChannels,
- const WebRtc_UWord32 samplesPerSec,
- const WebRtc_UWord16 totalDelayMS,
- const WebRtc_Word32 clockDrift,
- const WebRtc_UWord16 currentMicLevel);
-
-
- WebRtc_Word32 DemuxAndMix();
-
- WebRtc_Word32 EncodeAndSend();
-
- WebRtc_UWord32 CaptureLevel() const;
-
- WebRtc_Word32 StopSend();
-
- // VoEDtmf
- void UpdateMuteMicrophoneTime(const WebRtc_UWord32 lengthMs);
-
- // VoEExternalMedia
- int RegisterExternalMediaProcessing(VoEMediaProcess& proccess_object);
-
- int DeRegisterExternalMediaProcessing();
-
- int GetMixingFrequency();
-
- // VoEVolumeControl
- int SetMute(const bool enable);
-
- bool Mute() const;
-
- WebRtc_Word8 AudioLevel() const;
-
- WebRtc_Word16 AudioLevelFullRange() const;
-
- bool IsRecordingCall();
-
- bool IsRecordingMic();
-
- int StartPlayingFileAsMicrophone(const char* fileName,
- const bool loop,
- const FileFormats format,
- const int startPosition,
- const float volumeScaling,
- const int stopPosition,
- const CodecInst* codecInst);
-
- int StartPlayingFileAsMicrophone(InStream* stream,
- const FileFormats format,
- const int startPosition,
- const float volumeScaling,
- const int stopPosition,
- const CodecInst* codecInst);
-
- int StopPlayingFileAsMicrophone();
-
- int IsPlayingFileAsMicrophone() const;
-
- int ScaleFileAsMicrophonePlayout(const float scale);
-
- int StartRecordingMicrophone(const char* fileName,
- const CodecInst* codecInst);
-
- int StartRecordingMicrophone(OutStream* stream,
- const CodecInst* codecInst);
-
- int StopRecordingMicrophone();
-
- int StartRecordingCall(const char* fileName, const CodecInst* codecInst);
-
- int StartRecordingCall(OutStream* stream, const CodecInst* codecInst);
-
- int StopRecordingCall();
-
- void SetMixWithMicStatus(bool mix);
-
- WebRtc_Word32 RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
-
- virtual ~TransmitMixer();
-
-public: // MonitorObserver
- void OnPeriodicProcess();
-
-
-public: // FileCallback
- void PlayNotification(const WebRtc_Word32 id,
- const WebRtc_UWord32 durationMs);
-
- void RecordNotification(const WebRtc_Word32 id,
- const WebRtc_UWord32 durationMs);
-
- void PlayFileEnded(const WebRtc_Word32 id);
-
- void RecordFileEnded(const WebRtc_Word32 id);
-
-private:
- TransmitMixer(const WebRtc_UWord32 instanceId);
-
-private:
- WebRtc_Word32 GenerateAudioFrame(const WebRtc_Word16 audioSamples[],
- const WebRtc_UWord32 nSamples,
- const WebRtc_UWord8 nChannels,
- const WebRtc_UWord32 samplesPerSec,
- const int mixingFrequency);
- WebRtc_Word32 RecordAudioToFile(const WebRtc_UWord32 mixingFrequency);
-
- WebRtc_Word32 MixOrReplaceAudioWithFile(
- const int mixingFrequency);
-
- WebRtc_Word32 APMProcessStream(const WebRtc_UWord16 totalDelayMS,
- const WebRtc_Word32 clockDrift,
- const WebRtc_UWord16 currentMicLevel);
-
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
- int TypingDetection();
-#endif
-
-private: // uses
- Statistics* _engineStatisticsPtr;
- ChannelManager* _channelManagerPtr;
- AudioProcessing* _audioProcessingModulePtr;
- VoiceEngineObserver* _voiceEngineObserverPtr;
- ProcessThread* _processThreadPtr;
-
-private: // owns
- MonitorModule _monitorModule;
- AudioFrame _audioFrame;
- Resampler _audioResampler; // ADM sample rate -> mixing rate
- FilePlayer* _filePlayerPtr;
- FileRecorder* _fileRecorderPtr;
- FileRecorder* _fileCallRecorderPtr;
- int _filePlayerId;
- int _fileRecorderId;
- int _fileCallRecorderId;
- bool _filePlaying;
- bool _fileRecording;
- bool _fileCallRecording;
- voe::AudioLevel _audioLevel;
- // protect file instances and their variables in MixedParticipants()
- CriticalSectionWrapper& _critSect;
- CriticalSectionWrapper& _callbackCritSect;
-
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
- WebRtc_Word32 _timeActive;
- WebRtc_Word32 _penaltyCounter;
- WebRtc_UWord32 _typingNoiseWarning;
-#endif
- WebRtc_UWord32 _saturationWarning;
- WebRtc_UWord32 _noiseWarning;
-
-private:
- int _instanceId;
- bool _mixFileWithMicrophone;
- WebRtc_UWord32 _captureLevel;
- bool _externalMedia;
- VoEMediaProcess* _externalMediaCallbackPtr;
- bool _mute;
- WebRtc_Word32 _remainingMuteMicTimeMs;
- int _mixingFrequency;
- bool _includeAudioLevelIndication;
-};
-
-#endif // WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
-
-} // namespace voe
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/utility.cc b/trunk/src/voice_engine/main/source/utility.cc
deleted file mode 100644
index 6e70156..0000000
--- a/trunk/src/voice_engine/main/source/utility.cc
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "utility.h"
-
-#include "module.h"
-#include "trace.h"
-
-namespace webrtc
-{
-
-namespace voe
-{
-
-void Utility::MixWithSat(WebRtc_Word16 target[],
- const WebRtc_Word16 source[],
- WebRtc_UWord16 len)
-{
- WebRtc_Word32 temp(0);
- for (int i = 0; i < len; i++)
- {
- temp = source[i] + target[i];
- if (temp > 32767)
- target[i] = 32767;
- else if (temp < -32768)
- target[i] = -32768;
- else
- target[i] = (WebRtc_Word16) temp;
- }
-}
-
-void Utility::MixSubtractWithSat(WebRtc_Word16 target[],
- const WebRtc_Word16 source[],
- WebRtc_UWord16 len)
-{
- WebRtc_Word32 temp(0);
- for (int i = 0; i < len; i++)
- {
- temp = target[i] - source[i];
- if (temp > 32767)
- target[i] = 32767;
- else if (temp < -32768)
- target[i] = -32768;
- else
- target[i] = (WebRtc_Word16) temp;
- }
-}
-
-void Utility::MixAndScaleWithSat(WebRtc_Word16 target[],
- const WebRtc_Word16 source[], float scale,
- WebRtc_UWord16 len)
-{
- WebRtc_Word32 temp(0);
- for (int i = 0; i < len; i++)
- {
- temp = (WebRtc_Word32) (target[i] + scale * source[i]);
- if (temp > 32767)
- target[i] = 32767;
- else if (temp < -32768)
- target[i] = -32768;
- else
- target[i] = (WebRtc_Word16) temp;
- }
-}
-
-void Utility::Scale(WebRtc_Word16 vector[], float scale, WebRtc_UWord16 len)
-{
- for (int i = 0; i < len; i++)
- {
- vector[i] = (WebRtc_Word16) (scale * vector[i]);
- }
-}
-
-void Utility::ScaleWithSat(WebRtc_Word16 vector[], float scale,
- WebRtc_UWord16 len)
-{
- WebRtc_Word32 temp(0);
- for (int i = 0; i < len; i++)
- {
- temp = (WebRtc_Word32) (scale * vector[i]);
- if (temp > 32767)
- vector[i] = 32767;
- else if (temp < -32768)
- vector[i] = -32768;
- else
- vector[i] = (WebRtc_Word16) temp;
- }
-}
-
-} // namespace voe
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/utility.h b/trunk/src/voice_engine/main/source/utility.h
deleted file mode 100644
index 084ddf4..0000000
--- a/trunk/src/voice_engine/main/source/utility.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Contains functions often used by different parts of VoiceEngine.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_UTILITY_H
-#define WEBRTC_VOICE_ENGINE_UTILITY_H
-
-#include "typedefs.h"
-#include "voice_engine_defines.h"
-
-namespace webrtc
-{
-
-class Module;
-
-namespace voe
-{
-
-class Utility
-{
-public:
- static void MixWithSat(WebRtc_Word16 target[],
- const WebRtc_Word16 source[],
- WebRtc_UWord16 len);
-
- static void MixSubtractWithSat(WebRtc_Word16 target[],
- const WebRtc_Word16 source[],
- WebRtc_UWord16 len);
-
- static void MixAndScaleWithSat(WebRtc_Word16 target[],
- const WebRtc_Word16 source[],
- float scale,
- WebRtc_UWord16 len);
-
- static void Scale(WebRtc_Word16 vector[], float scale, WebRtc_UWord16 len);
-
- static void ScaleWithSat(WebRtc_Word16 vector[],
- float scale,
- WebRtc_UWord16 len);
-};
-
-} // namespace voe
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_UTILITY_H
diff --git a/trunk/src/voice_engine/main/source/voe_audio_processing_impl.cc b/trunk/src/voice_engine/main/source/voe_audio_processing_impl.cc
deleted file mode 100644
index 95800f4..0000000
--- a/trunk/src/voice_engine/main/source/voe_audio_processing_impl.cc
+++ /dev/null
@@ -1,1176 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voe_audio_processing_impl.h"
-
-#include "audio_processing.h"
-#include "channel.h"
-#include "critical_section_wrapper.h"
-#include "trace.h"
-#include "voe_errors.h"
-#include "voice_engine_impl.h"
-
-namespace webrtc {
-
-VoEAudioProcessing* VoEAudioProcessing::GetInterface(
- VoiceEngine* voiceEngine)
-{
-#ifndef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
- return NULL;
-#else
- if (NULL == voiceEngine)
- {
- return NULL;
- }
- VoiceEngineImpl* s = reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
- VoEAudioProcessingImpl* d = s;
- (*d)++;
- return (d);
-#endif
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
-VoEAudioProcessingImpl::VoEAudioProcessingImpl():
- _isAecMode(WEBRTC_VOICE_ENGINE_EC_DEFAULT_MODE == EcAec?
- true : false)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "VoEAudioProcessingImpl::VoEAudioProcessingImpl() - ctor");
-}
-
-VoEAudioProcessingImpl::~VoEAudioProcessingImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "VoEAudioProcessingImpl::~VoEAudioProcessingImpl() - dtor");
-}
-
-int VoEAudioProcessingImpl::Release()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "VoEAudioProcessing::Release()");
- (*this)--;
- int refCount = GetCount();
- if (refCount < 0)
- {
- Reset(); // reset reference counter to zero => OK to delete VE
- _engineStatistics.SetLastError(
- VE_INTERFACE_NOT_FOUND, kTraceWarning);
- return (-1);
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "VoEAudioProcessing reference counter = %d", refCount);
- return (refCount);
-}
-
-int VoEAudioProcessingImpl::SetNsStatus(bool enable, NsModes mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetNsStatus(enable=%d, mode=%d)", enable, mode);
-#ifdef WEBRTC_VOICE_ENGINE_NR
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- NoiseSuppression::Level nsLevel(
- (NoiseSuppression::Level)WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE);
- switch (mode)
- {
- case kNsDefault:
- nsLevel = (NoiseSuppression::Level)WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE;
- break;
- case kNsUnchanged:
- nsLevel = _audioProcessingModulePtr->noise_suppression()->level();
- break;
- case kNsConference:
- nsLevel = NoiseSuppression::kHigh;
- break;
- case kNsLowSuppression:
- nsLevel = NoiseSuppression::kLow;
- break;
- case kNsModerateSuppression:
- nsLevel = NoiseSuppression::kModerate;
- break;
- case kNsHighSuppression:
- nsLevel = NoiseSuppression::kHigh;
- break;
- case kNsVeryHighSuppression:
- nsLevel = NoiseSuppression::kVeryHigh;
- break;
- }
-
- if (_audioProcessingModulePtr->noise_suppression()->set_level(nsLevel) != 0)
- {
- _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
- "SetNsStatus() failed to set Ns mode");
- return -1;
- }
- if (_audioProcessingModulePtr->noise_suppression()->Enable(enable) != 0)
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetNsStatus() failed to set Ns state");
- return -1;
- }
-
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetNsStatus() Ns is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::GetNsStatus(bool& enabled, NsModes& mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetNsStatus(enabled=?, mode=?)");
-#ifdef WEBRTC_VOICE_ENGINE_NR
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- bool enable(false);
- NoiseSuppression::Level nsLevel(
- (NoiseSuppression::Level)WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE);
-
- enable = _audioProcessingModulePtr->noise_suppression()->is_enabled();
- nsLevel = _audioProcessingModulePtr->noise_suppression()->level();
-
- enabled = enable;
-
- switch (nsLevel)
- {
- case NoiseSuppression::kLow:
- mode = kNsLowSuppression;
- break;
- case NoiseSuppression::kModerate:
- mode = kNsModerateSuppression;
- break;
- case NoiseSuppression::kHigh:
- mode = kNsHighSuppression;
- break;
- case NoiseSuppression::kVeryHigh:
- mode = kNsVeryHighSuppression;
- break;
- }
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetNsStatus() => enabled=% d, mode=%d",enabled, mode);
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "GetNsStatus() Ns is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::SetAgcStatus(bool enable, AgcModes mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetAgcStatus(enable=%d, mode=%d)", enable, mode);
-#ifdef WEBRTC_VOICE_ENGINE_AGC
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
-#if defined(MAC_IPHONE) || defined(ATA) || defined(WEBRTC_ANDROID)
- if (mode == kAgcAdaptiveAnalog)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetAgcStatus() invalid Agc mode for mobile device");
- return -1;
- }
-#endif
-
- GainControl::Mode agcMode(
- (GainControl::Mode)WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE);
- switch (mode)
- {
- case kAgcDefault:
- agcMode = (GainControl::Mode)WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE;
- break;
- case kAgcUnchanged:
- agcMode = _audioProcessingModulePtr->gain_control()->mode();;
- break;
- case kAgcFixedDigital:
- agcMode = GainControl::kFixedDigital;
- break;
- case kAgcAdaptiveAnalog:
- agcMode = GainControl::kAdaptiveAnalog;
- break;
- case kAgcAdaptiveDigital:
- agcMode = GainControl::kAdaptiveDigital;
- break;
- }
-
- if (_audioProcessingModulePtr->gain_control()->set_mode(agcMode) != 0)
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetAgcStatus() failed to set Agc mode");
- return -1;
- }
- if (_audioProcessingModulePtr->gain_control()->Enable(enable) != 0)
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetAgcStatus() failed to set Agc state");
- return -1;
- }
-
- if (agcMode != GainControl::kFixedDigital)
- {
- // Set Agc state in the ADM when adaptive Agc mode has been selected.
- // Note that we also enable the ADM Agc when Adaptive Digital mode is
- // used since we want to be able to provide the APM with updated mic
- // levels when the user modifies the mic level manually.
- if (_audioDevicePtr->SetAGC(enable) != 0)
- {
- _engineStatistics.SetLastError(
- VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
- "SetAgcStatus() failed to set Agc mode");
- }
- }
-
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetAgcStatus() Agc is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::GetAgcStatus(bool& enabled, AgcModes& mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetAgcStatus(enabled=?, mode=?)");
-#ifdef WEBRTC_VOICE_ENGINE_AGC
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- bool enable(false);
- GainControl::Mode agcMode(
- (GainControl::Mode)WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE);
-
- enable = _audioProcessingModulePtr->gain_control()->is_enabled();
- agcMode = _audioProcessingModulePtr->gain_control()->mode();
-
- enabled = enable;
-
- switch (agcMode)
- {
- case GainControl::kFixedDigital:
- mode = kAgcFixedDigital;
- break;
- case GainControl::kAdaptiveAnalog:
- mode = kAgcAdaptiveAnalog;
- break;
- case GainControl::kAdaptiveDigital:
- mode = kAgcAdaptiveDigital;
- break;
- }
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetAgcStatus() => enabled=%d, mode=%d", enabled, mode);
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "GetAgcStatus() Agc is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::SetAgcConfig(const AgcConfig config)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetAgcConfig()");
-#ifdef WEBRTC_VOICE_ENGINE_AGC
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- if (_audioProcessingModulePtr->gain_control()->set_target_level_dbfs(
- config.targetLeveldBOv) != 0)
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetAgcConfig() failed to set target peak |level|"
- " (or envelope) of the Agc");
- return -1;
- }
- if (_audioProcessingModulePtr->gain_control()->set_compression_gain_db(
- config.digitalCompressionGaindB) != 0)
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetAgcConfig() failed to set the range in |gain|"
- "the digital compression stage may apply");
- return -1;
- }
- if (_audioProcessingModulePtr->gain_control()->enable_limiter(
- config.limiterEnable) != 0)
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetAgcConfig() failed to set hard limiter to the signal");
- return -1;
- }
-
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetAgcConfig() EC is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::GetAgcConfig(AgcConfig &config)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetAgcConfig(config=?)");
-#ifdef WEBRTC_VOICE_ENGINE_AGC
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- config.targetLeveldBOv =
- _audioProcessingModulePtr->gain_control()->target_level_dbfs();
- config.digitalCompressionGaindB =
- _audioProcessingModulePtr->gain_control()->compression_gain_db();
- config.limiterEnable =
- _audioProcessingModulePtr->gain_control()->is_limiter_enabled();
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetAgcConfig() => targetLeveldBOv=%u, "
- "digitalCompressionGaindB=%u, limiterEnable=%d",
- config.targetLeveldBOv,
- config.digitalCompressionGaindB,
- config.limiterEnable);
-
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "GetAgcConfig() EC is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::SetRxNsStatus(int channel,
- bool enable,
- NsModes mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetRxNsStatus(channel=%d, enable=%d, mode=%d)",
- channel, (int)enable, (int)mode);
-#ifdef WEBRTC_VOICE_ENGINE_AGC
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetRxNsStatus() failed to locate channel");
- return -1;
- }
- return channelPtr->SetRxNsStatus(enable, mode);
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetRxNsStatus() AGC is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::GetRxNsStatus(int channel,
- bool& enabled,
- NsModes& mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetRxNsStatus(channel=%d, enable=?, mode=?)", channel);
-#ifdef WEBRTC_VOICE_ENGINE_AGC
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRxNsStatus() failed to locate channel");
- return -1;
- }
- return channelPtr->GetRxNsStatus(enabled, mode);
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "GetRxNsStatus() Agc is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::SetRxAgcStatus(int channel,
- bool enable,
- AgcModes mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetRxAgcStatus(channel=%d, enable=%d, mode=%d)",
- channel, (int)enable, (int)mode);
-#ifdef WEBRTC_VOICE_ENGINE_AGC
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetRxAgcStatus() failed to locate channel");
- return -1;
- }
- return channelPtr->SetRxAgcStatus(enable, mode);
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetRxAgcStatus() Agc is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::GetRxAgcStatus(int channel,
- bool& enabled,
- AgcModes& mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetRxAgcStatus(channel=%d, enable=?, mode=?)", channel);
-#ifdef WEBRTC_VOICE_ENGINE_AGC
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRxAgcStatus() failed to locate channel");
- return -1;
- }
- return channelPtr->GetRxAgcStatus(enabled, mode);
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "GetRxAgcStatus() Agc is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::SetRxAgcConfig(int channel, const AgcConfig config)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetRxAgcConfig(channel=%d)", channel);
-#ifdef WEBRTC_VOICE_ENGINE_AGC
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetRxAgcConfig() failed to locate channel");
- return -1;
- }
- return channelPtr->SetRxAgcConfig(config);
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetRxAgcConfig() Agc is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::GetRxAgcConfig(int channel, AgcConfig& config)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetRxAgcConfig(channel=%d)", channel);
-#ifdef WEBRTC_VOICE_ENGINE_AGC
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRxAgcConfig() failed to locate channel");
- return -1;
- }
- return channelPtr->GetRxAgcConfig(config);
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "GetRxAgcConfig() Agc is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::SetEcStatus(bool enable, EcModes mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetEcStatus(enable=%d, mode=%d)", enable, mode);
-#ifdef WEBRTC_VOICE_ENGINE_ECHO
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- // AEC mode
- if ((mode == kEcDefault) ||
- (mode == kEcConference) ||
- (mode == kEcAec) ||
- ((mode == kEcUnchanged) &&
- (_isAecMode == true)))
- {
- if (enable)
- {
- // Disable the AECM before enable the AEC
- if (_audioProcessingModulePtr->echo_control_mobile()->is_enabled())
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceWarning,
- "SetEcStatus() disable AECM before enabling AEC");
- if (_audioProcessingModulePtr->echo_control_mobile()->
- Enable(false) != 0)
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetEcStatus() failed to disable AECM");
- return -1;
- }
- }
- }
- if (_audioProcessingModulePtr->echo_cancellation()->Enable(enable) != 0)
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetEcStatus() failed to set AEC state");
- return -1;
- }
-#ifdef CLOCK_SKEW_COMP
- if (_audioProcessingModulePtr->echo_cancellation()->
- enable_drift_compensation(true) != 0)
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetEcStatus() failed to enable drift compensation");
- return -1;
- }
-#else
- if (_audioProcessingModulePtr->echo_cancellation()->
- enable_drift_compensation(false) != 0)
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetEcStatus() failed to disable drift compensation");
- return -1;
- }
-#endif
- if (mode == kEcConference)
- {
- if (_audioProcessingModulePtr->echo_cancellation()->
- set_suppression_level(EchoCancellation::kHighSuppression) != 0)
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetEcStatus() failed to set aggressiveness to high");
- return -1;
- }
- }
- else
- {
- if (_audioProcessingModulePtr->echo_cancellation()->
- set_suppression_level(
- EchoCancellation::kModerateSuppression) != 0)
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetEcStatus() failed to set aggressiveness to moderate");
- return -1;
- }
- }
-
- _isAecMode = true;
- }
- else if ((mode == kEcAecm) ||
- ((mode == kEcUnchanged) &&
- (_isAecMode == false)))
- {
- if (enable)
- {
- // Disable the AEC before enable the AECM
- if (_audioProcessingModulePtr->echo_cancellation()->is_enabled())
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceWarning,
- "SetEcStatus() disable AEC before enabling AECM");
- if (_audioProcessingModulePtr->echo_cancellation()->
- Enable(false) != 0)
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetEcStatus() failed to disable AEC");
- return -1;
- }
- }
- }
- if (_audioProcessingModulePtr->echo_control_mobile()->
- Enable(enable) != 0)
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetEcStatus() failed to set AECM state");
- return -1;
- }
- _isAecMode = false;
- }
- else
- {
- _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "SetEcStatus() invalid EC mode");
- return -1;
- }
-
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetEcStatus() EC is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::GetEcStatus(bool& enabled, EcModes& mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetEcStatus()");
-#ifdef WEBRTC_VOICE_ENGINE_ECHO
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- if (_isAecMode == true)
- {
- mode = kEcAec;
- enabled = _audioProcessingModulePtr->echo_cancellation()->is_enabled();
- }
- else
- {
- mode = kEcAecm;
- enabled = _audioProcessingModulePtr->echo_control_mobile()->
- is_enabled();
- }
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetEcStatus() => enabled=%i, mode=%i",
- enabled, (int)mode);
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "GetEcStatus() EC is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::SetAecmMode(AecmModes mode, bool enableCNG)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetAECMMode(mode = %d)", mode);
-#ifdef WEBRTC_VOICE_ENGINE_ECHO
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- EchoControlMobile::RoutingMode aecmMode(
- EchoControlMobile::kQuietEarpieceOrHeadset);
-
- switch (mode)
- {
- case kAecmQuietEarpieceOrHeadset:
- aecmMode = EchoControlMobile::kQuietEarpieceOrHeadset;
- break;
- case kAecmEarpiece:
- aecmMode = EchoControlMobile::kEarpiece;
- break;
- case kAecmLoudEarpiece:
- aecmMode = EchoControlMobile::kLoudEarpiece;
- break;
- case kAecmSpeakerphone:
- aecmMode = EchoControlMobile::kSpeakerphone;
- break;
- case kAecmLoudSpeakerphone:
- aecmMode = EchoControlMobile::kLoudSpeakerphone;
- break;
- }
-
-
- if (_audioProcessingModulePtr->echo_control_mobile()->
- set_routing_mode(aecmMode) != 0)
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetAECMMode() failed to set AECM routing mode");
- return -1;
- }
- if (_audioProcessingModulePtr->echo_control_mobile()->
- enable_comfort_noise(enableCNG) != 0)
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetAECMMode() failed to set comfort noise state for AECM");
- return -1;
- }
-
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetAECMMode() EC is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::GetAecmMode(AecmModes& mode, bool& enabledCNG)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetAECMMode(mode=?)");
-#ifdef WEBRTC_VOICE_ENGINE_ECHO
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- enabledCNG = false;
-
- EchoControlMobile::RoutingMode aecmMode =
- _audioProcessingModulePtr->echo_control_mobile()->routing_mode();
- enabledCNG = _audioProcessingModulePtr->echo_control_mobile()->
- is_comfort_noise_enabled();
-
- switch (aecmMode)
- {
- case EchoControlMobile::kQuietEarpieceOrHeadset:
- mode = kAecmQuietEarpieceOrHeadset;
- break;
- case EchoControlMobile::kEarpiece:
- mode = kAecmEarpiece;
- break;
- case EchoControlMobile::kLoudEarpiece:
- mode = kAecmLoudEarpiece;
- break;
- case EchoControlMobile::kSpeakerphone:
- mode = kAecmSpeakerphone;
- break;
- case EchoControlMobile::kLoudSpeakerphone:
- mode = kAecmLoudSpeakerphone;
- break;
- }
-
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "GetAECMMode() EC is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::RegisterRxVadObserver(
- int channel,
- VoERxVadCallback &observer)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "RegisterRxVadObserver()");
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "RegisterRxVadObserver() failed to locate channel");
- return -1;
- }
- return channelPtr->RegisterRxVadObserver(observer);
-}
-
-int VoEAudioProcessingImpl::DeRegisterRxVadObserver(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "DeRegisterRxVadObserver()");
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "DeRegisterRxVadObserver() failed to locate channel");
- return -1;
- }
-
- return channelPtr->DeRegisterRxVadObserver();
-}
-
-int VoEAudioProcessingImpl::VoiceActivityIndicator(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "VoiceActivityIndicator(channel=%d)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "DeRegisterRxVadObserver() failed to locate channel");
- return -1;
- }
- int activity(-1);
- channelPtr->VoiceActivityIndicator(activity);
-
- return activity;
-}
-
-int VoEAudioProcessingImpl::SetEcMetricsStatus(bool enable) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetEcMetricsStatus(enable=%d)", enable);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
-#ifdef WEBRTC_VOICE_ENGINE_ECHO
- if (!_engineStatistics.Initialized()) {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- if ((_audioProcessingModulePtr->echo_cancellation()->enable_metrics(enable)
- != 0) ||
- (_audioProcessingModulePtr->echo_cancellation()->enable_delay_logging(
- enable) != 0)) {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "SetEcMetricsStatus() unable to set EC metrics mode");
- return -1;
- }
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError, "SetEcStatus() EC is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::GetEcMetricsStatus(bool& enabled) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetEcMetricsStatus(enabled=?)");
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
-#ifdef WEBRTC_VOICE_ENGINE_ECHO
- if (!_engineStatistics.Initialized()) {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- bool echo_mode =
- _audioProcessingModulePtr->echo_cancellation()->are_metrics_enabled();
- bool delay_mode =
- _audioProcessingModulePtr->echo_cancellation()->is_delay_logging_enabled();
-
- if (echo_mode != delay_mode) {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceError,
- "GetEcMetricsStatus() delay logging and echo mode are not the same");
- return -1;
- }
-
- enabled = echo_mode;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "GetEcMetricsStatus() => enabled=%d", enabled);
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError, "SetEcStatus() EC is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::GetEchoMetrics(int& ERL,
- int& ERLE,
- int& RERL,
- int& A_NLP) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetEchoMetrics(ERL=?, ERLE=?, RERL=?, A_NLP=?)");
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
-#ifdef WEBRTC_VOICE_ENGINE_ECHO
- if (!_engineStatistics.Initialized()) {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (!_audioProcessingModulePtr->echo_cancellation()->is_enabled()) {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceWarning,
- "GetEchoMetrics() AudioProcessingModule AEC is not enabled");
- return -1;
- }
-
- // Get Echo Metrics from Audio Processing Module.
- EchoCancellation::Metrics echoMetrics;
- if (_audioProcessingModulePtr->echo_cancellation()->GetMetrics(&echoMetrics))
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "GetEchoMetrics(), AudioProcessingModule metrics error");
- return -1;
- }
-
- // Echo quality metrics.
- ERL = echoMetrics.echo_return_loss.instant;
- ERLE = echoMetrics.echo_return_loss_enhancement.instant;
- RERL = echoMetrics.residual_echo_return_loss.instant;
- A_NLP = echoMetrics.a_nlp.instant;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetEchoMetrics() => ERL=%d, ERLE=%d, RERL=%d, A_NLP=%d",
- ERL, ERLE, RERL, A_NLP);
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError, "SetEcStatus() EC is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::GetEcDelayMetrics(int& delay_median,
- int& delay_std) {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetEcDelayMetrics(median=?, std=?)");
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
-#ifdef WEBRTC_VOICE_ENGINE_ECHO
- if (!_engineStatistics.Initialized()) {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (!_audioProcessingModulePtr->echo_cancellation()->is_enabled()) {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceWarning,
- "GetEcDelayMetrics() AudioProcessingModule AEC is not enabled");
- return -1;
- }
-
- int median = 0;
- int std = 0;
- // Get delay-logging values from Audio Processing Module.
- if (_audioProcessingModulePtr->echo_cancellation()->GetDelayMetrics(
- &median, &std)) {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "GetEcDelayMetrics(), AudioProcessingModule delay-logging "
- "error");
- return -1;
- }
-
- // EC delay-logging metrics
- delay_median = median;
- delay_std = std;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetEcDelayMetrics() => delay_median=%d, delay_std=%d",
- delay_median, delay_std);
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError, "SetEcStatus() EC is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::StartDebugRecording(const char* fileNameUTF8)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StartDebugRecording()");
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- return _audioProcessingModulePtr->StartDebugRecording(fileNameUTF8);
-
-}
-
-int VoEAudioProcessingImpl::StopDebugRecording()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StopDebugRecording()");
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- return _audioProcessingModulePtr->StopDebugRecording();
-}
-
-int VoEAudioProcessingImpl::SetTypingDetectionStatus(bool enable)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetTypingDetectionStatus()");
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- // Just use the VAD state to determine if we should enable typing detection
- // or not
-
- if (_audioProcessingModulePtr->voice_detection()->Enable(enable))
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceWarning,
- "SetTypingDetectionStatus() failed to set VAD state");
- return -1;
- }
- if (_audioProcessingModulePtr->voice_detection()->set_likelihood(
- VoiceDetection::kVeryLowLikelihood))
- {
- _engineStatistics.SetLastError(
- VE_APM_ERROR, kTraceWarning,
- "SetTypingDetectionStatus() failed to set VAD likelihood to low");
- return -1;
- }
-
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetTypingDetectionStatus is not supported");
- return -1;
-#endif
-}
-
-int VoEAudioProcessingImpl::GetTypingDetectionStatus(bool& enabled)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetTypingDetectionStatus()");
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
-#ifdef WEBRTC_VOICE_ENGINE_TYPING_DETECTION
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- // Just use the VAD state to determine if we should enable typing
- // detection or not
-
- enabled = _audioProcessingModulePtr->voice_detection()->is_enabled();
-
- return(0);
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetTypingDetectionStatus is not supported");
- return(-1);
-#endif
-}
-
-#endif // #ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_audio_processing_impl.h b/trunk/src/voice_engine/main/source/voe_audio_processing_impl.h
deleted file mode 100644
index 277731e..0000000
--- a/trunk/src/voice_engine/main/source/voe_audio_processing_impl.h
+++ /dev/null
@@ -1,100 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_IMPL_H
-#define WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_IMPL_H
-
-#include "voe_audio_processing.h"
-
-#include "ref_count.h"
-#include "shared_data.h"
-
-
-namespace webrtc {
-
-class VoEAudioProcessingImpl : public virtual voe::SharedData,
- public VoEAudioProcessing,
- public voe::RefCount
-{
-public:
- virtual int Release();
-
- virtual int SetNsStatus(bool enable, NsModes mode = kNsUnchanged);
-
- virtual int GetNsStatus(bool& enabled, NsModes& mode);
-
- virtual int SetAgcStatus(bool enable, AgcModes mode = kAgcUnchanged);
-
- virtual int GetAgcStatus(bool& enabled, AgcModes& mode);
-
- virtual int SetAgcConfig(const AgcConfig config);
-
- virtual int GetAgcConfig(AgcConfig& config);
-
- virtual int SetRxNsStatus(int channel,
- bool enable,
- NsModes mode = kNsUnchanged);
-
- virtual int GetRxNsStatus(int channel, bool& enabled, NsModes& mode);
-
- virtual int SetRxAgcStatus(int channel,
- bool enable,
- AgcModes mode = kAgcUnchanged);
-
- virtual int GetRxAgcStatus(int channel, bool& enabled, AgcModes& mode);
-
- virtual int SetRxAgcConfig(int channel, const AgcConfig config);
-
- virtual int GetRxAgcConfig(int channel, AgcConfig& config);
-
- virtual int SetEcStatus(bool enable, EcModes mode = kEcUnchanged);
-
- virtual int GetEcStatus(bool& enabled, EcModes& mode);
-
- virtual int SetAecmMode(AecmModes mode = kAecmSpeakerphone,
- bool enableCNG = true);
-
- virtual int GetAecmMode(AecmModes& mode, bool& enabledCNG);
-
- virtual int RegisterRxVadObserver(int channel,
- VoERxVadCallback& observer);
-
- virtual int DeRegisterRxVadObserver(int channel);
-
- virtual int VoiceActivityIndicator(int channel);
-
- virtual int SetEcMetricsStatus(bool enable);
-
- virtual int GetEcMetricsStatus(bool& enabled);
-
- virtual int GetEchoMetrics(int& ERL, int& ERLE, int& RERL, int& A_NLP);
-
- virtual int GetEcDelayMetrics(int& delay_median, int& delay_std);
-
- virtual int StartDebugRecording(const char* fileNameUTF8);
-
- virtual int StopDebugRecording();
-
- virtual int SetTypingDetectionStatus(bool enable);
-
- virtual int GetTypingDetectionStatus(bool& enabled);
-
-protected:
- VoEAudioProcessingImpl();
- virtual ~VoEAudioProcessingImpl();
-
-private:
- bool _isAecMode;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_AUDIO_PROCESSING_IMPL_H
-
diff --git a/trunk/src/voice_engine/main/source/voe_base_impl.cc b/trunk/src/voice_engine/main/source/voe_base_impl.cc
deleted file mode 100644
index cdc0ee3..0000000
--- a/trunk/src/voice_engine/main/source/voe_base_impl.cc
+++ /dev/null
@@ -1,1679 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voe_base_impl.h"
-
-#include "audio_coding_module.h"
-#include "audio_device_impl.h"
-#include "audio_processing.h"
-#include "channel.h"
-#include "critical_section_wrapper.h"
-#include "file_wrapper.h"
-#include "output_mixer.h"
-#include "signal_processing_library.h"
-#include "trace.h"
-#include "transmit_mixer.h"
-#include "utility.h"
-#include "voe_errors.h"
-#include "voice_engine_impl.h"
-
-#if (defined(_WIN32) && defined(_DLL) && (_MSC_VER == 1400))
-// Fix for VS 2005 MD/MDd link problem
-#include <stdio.h>
-extern "C"
- { FILE _iob[3] = { __iob_func()[0], __iob_func()[1], __iob_func()[2]}; }
-#endif
-
-namespace webrtc
-{
-
-VoEBase* VoEBase::GetInterface(VoiceEngine* voiceEngine)
-{
- if (NULL == voiceEngine)
- {
- return NULL;
- }
- VoiceEngineImpl* s = reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
- VoEBaseImpl* d = s;
- (*d)++;
- return (d);
-}
-
-VoEBaseImpl::VoEBaseImpl() :
- _voiceEngineObserverPtr(NULL),
- _callbackCritSect(*CriticalSectionWrapper::CreateCriticalSection()),
- _voiceEngineObserver(false), _oldVoEMicLevel(0), _oldMicLevel(0)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
- "VoEBaseImpl() - ctor");
-}
-
-VoEBaseImpl::~VoEBaseImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
- "~VoEBaseImpl() - dtor");
-
- TerminateInternal();
-
- delete &_callbackCritSect;
-}
-
-int VoEBaseImpl::Release()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "VoEBaseImpl::Release()");
- (*this)--;
- int refCount = GetCount();
- if (refCount < 0)
- {
- Reset();
- _engineStatistics.SetLastError(VE_INTERFACE_NOT_FOUND, kTraceWarning);
- return (-1);
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "VoEBaseImpl reference counter = %d", refCount);
- return (refCount);
-}
-
-void VoEBaseImpl::OnErrorIsReported(const ErrorCode error)
-{
- CriticalSectionScoped cs(_callbackCritSect);
- if (_voiceEngineObserver)
- {
- if (_voiceEngineObserverPtr)
- {
- int errCode(0);
- if (error == AudioDeviceObserver::kRecordingError)
- {
- errCode = VE_RUNTIME_REC_ERROR;
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "VoEBaseImpl::OnErrorIsReported() => "
- "VE_RUNTIME_REC_ERROR");
- }
- else if (error == AudioDeviceObserver::kPlayoutError)
- {
- errCode = VE_RUNTIME_PLAY_ERROR;
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "VoEBaseImpl::OnErrorIsReported() => "
- "VE_RUNTIME_PLAY_ERROR");
- }
- // Deliver callback (-1 <=> no channel dependency)
- _voiceEngineObserverPtr->CallbackOnError(-1, errCode);
- }
- }
-}
-
-void VoEBaseImpl::OnWarningIsReported(const WarningCode warning)
-{
- CriticalSectionScoped cs(_callbackCritSect);
- if (_voiceEngineObserver)
- {
- if (_voiceEngineObserverPtr)
- {
- int warningCode(0);
- if (warning == AudioDeviceObserver::kRecordingWarning)
- {
- warningCode = VE_RUNTIME_REC_WARNING;
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "VoEBaseImpl::OnErrorIsReported() => "
- "VE_RUNTIME_REC_WARNING");
- }
- else if (warning == AudioDeviceObserver::kPlayoutWarning)
- {
- warningCode = VE_RUNTIME_PLAY_WARNING;
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "VoEBaseImpl::OnErrorIsReported() => "
- "VE_RUNTIME_PLAY_WARNING");
- }
- // Deliver callback (-1 <=> no channel dependency)
- _voiceEngineObserverPtr->CallbackOnError(-1, warningCode);
- }
- }
-}
-
-WebRtc_Word32 VoEBaseImpl::RecordedDataIsAvailable(
- const WebRtc_Word8* audioSamples,
- const WebRtc_UWord32 nSamples,
- const WebRtc_UWord8 nBytesPerSample,
- const WebRtc_UWord8 nChannels,
- const WebRtc_UWord32 samplesPerSec,
- const WebRtc_UWord32 totalDelayMS,
- const WebRtc_Word32 clockDrift,
- const WebRtc_UWord32 currentMicLevel,
- WebRtc_UWord32& newMicLevel)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "VoEBaseImpl::RecordedDataIsAvailable(nSamples=%u, "
- "nBytesPerSample=%u, nChannels=%u, samplesPerSec=%u, "
- "totalDelayMS=%u, clockDrift=%d, currentMicLevel=%u)",
- nSamples, nBytesPerSample, nChannels, samplesPerSec,
- totalDelayMS, clockDrift, currentMicLevel);
-
- assert(_transmitMixerPtr != NULL);
- assert(_audioDevicePtr != NULL);
-
- bool isAnalogAGC(false);
- WebRtc_UWord32 maxVolume(0);
- WebRtc_UWord16 currentVoEMicLevel(0);
- WebRtc_UWord32 newVoEMicLevel(0);
-
- if (_audioProcessingModulePtr
- && (_audioProcessingModulePtr->gain_control()->mode()
- == GainControl::kAdaptiveAnalog))
- {
- isAnalogAGC = true;
- }
-
- // Will only deal with the volume in adaptive analog mode
- if (isAnalogAGC)
- {
- // Scale from ADM to VoE level range
- if (_audioDevicePtr->MaxMicrophoneVolume(&maxVolume) == 0)
- {
- if (0 != maxVolume)
- {
- currentVoEMicLevel = (WebRtc_UWord16) ((currentMicLevel
- * kMaxVolumeLevel + (int) (maxVolume / 2))
- / (maxVolume));
- }
- }
- // We learned that on certain systems (e.g Linux) the currentVoEMicLevel
- // can be greater than the maxVolumeLevel therefore
- // we are going to cap the currentVoEMicLevel to the maxVolumeLevel
- // and change the maxVolume to currentMicLevel if it turns out that
- // the currentVoEMicLevel is indeed greater than the maxVolumeLevel.
- if (currentVoEMicLevel > kMaxVolumeLevel)
- {
- currentVoEMicLevel = kMaxVolumeLevel;
- maxVolume = currentMicLevel;
- }
- }
-
- // Keep track if the MicLevel has been changed by the AGC, if not,
- // use the old value AGC returns to let AGC continue its trend,
- // so eventually the AGC is able to change the mic level. This handles
- // issues with truncation introduced by the scaling.
- if (_oldMicLevel == currentMicLevel)
- {
- currentVoEMicLevel = (WebRtc_UWord16) _oldVoEMicLevel;
- }
-
- // Perform channel-independent operations
- // (APM, mix with file, record to file, mute, etc.)
- _transmitMixerPtr->PrepareDemux(audioSamples, nSamples, nChannels,
- samplesPerSec,
- (WebRtc_UWord16) totalDelayMS, clockDrift,
- currentVoEMicLevel);
-
- // Copy the audio frame to each sending channel and perform
- // channel-dependent operations (file mixing, mute, etc.) to prepare
- // for encoding.
- _transmitMixerPtr->DemuxAndMix();
- // Do the encoding and packetize+transmit the RTP packet when encoding
- // is done.
- _transmitMixerPtr->EncodeAndSend();
-
- // Will only deal with the volume in adaptive analog mode
- if (isAnalogAGC)
- {
- // Scale from VoE to ADM level range
- newVoEMicLevel = _transmitMixerPtr->CaptureLevel();
- if (newVoEMicLevel != currentVoEMicLevel)
- {
- // Add (kMaxVolumeLevel/2) to round the value
- newMicLevel = (WebRtc_UWord32) ((newVoEMicLevel * maxVolume
- + (int) (kMaxVolumeLevel / 2)) / (kMaxVolumeLevel));
- }
- else
- {
- // Pass zero if the level is unchanged
- newMicLevel = 0;
- }
-
- // Keep track of the value AGC returns
- _oldVoEMicLevel = newVoEMicLevel;
- _oldMicLevel = currentMicLevel;
- }
-
- return 0;
-}
-
-WebRtc_Word32 VoEBaseImpl::NeedMorePlayData(
- const WebRtc_UWord32 nSamples,
- const WebRtc_UWord8 nBytesPerSample,
- const WebRtc_UWord8 nChannels,
- const WebRtc_UWord32 samplesPerSec,
- WebRtc_Word8* audioSamples,
- WebRtc_UWord32& nSamplesOut)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "VoEBaseImpl::NeedMorePlayData(nSamples=%u, "
- "nBytesPerSample=%d, nChannels=%d, samplesPerSec=%u)",
- nSamples, nBytesPerSample, nChannels, samplesPerSec);
-
- assert(_outputMixerPtr != NULL);
-
- // Perform mixing of all active participants (channel-based mixing)
- _outputMixerPtr->MixActiveChannels();
-
- // Additional operations on the combined signal
- _outputMixerPtr->DoOperationsOnCombinedSignal();
-
- // Retrieve the final output mix (resampled to match the ADM)
- _outputMixerPtr->GetMixedAudio(samplesPerSec, nChannels, _audioFrame);
-
- assert(nSamples == _audioFrame._payloadDataLengthInSamples);
- assert(samplesPerSec ==
- static_cast<WebRtc_UWord32>(_audioFrame._frequencyInHz));
-
- // Deliver audio (PCM) samples to the ADM
- memcpy(
- (WebRtc_Word16*) audioSamples,
- (const WebRtc_Word16*) _audioFrame._payloadData,
- sizeof(WebRtc_Word16) * (_audioFrame._payloadDataLengthInSamples
- * _audioFrame._audioChannel));
-
- nSamplesOut = _audioFrame._payloadDataLengthInSamples;
-
- return 0;
-}
-
-int VoEBaseImpl::RegisterVoiceEngineObserver(VoiceEngineObserver& observer)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "RegisterVoiceEngineObserver(observer=0x%d)", &observer);
- CriticalSectionScoped cs(_callbackCritSect);
- if (_voiceEngineObserverPtr)
- {
- _engineStatistics.SetLastError(VE_INVALID_OPERATION, kTraceError,
- "RegisterVoiceEngineObserver() observer"
- " already enabled");
- return -1;
- }
-
- // Register the observer in all active channels
- voe::ScopedChannel sc(_channelManager);
- void* iterator(NULL);
- voe::Channel* channelPtr = sc.GetFirstChannel(iterator);
- while (channelPtr != NULL)
- {
- channelPtr->RegisterVoiceEngineObserver(observer);
- channelPtr = sc.GetNextChannel(iterator);
- }
- _transmitMixerPtr->RegisterVoiceEngineObserver(observer);
-
- _voiceEngineObserverPtr = &observer;
- _voiceEngineObserver = true;
-
- return 0;
-}
-
-int VoEBaseImpl::DeRegisterVoiceEngineObserver()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "DeRegisterVoiceEngineObserver()");
- CriticalSectionScoped cs(_callbackCritSect);
- if (!_voiceEngineObserverPtr)
- {
- _engineStatistics.SetLastError(VE_INVALID_OPERATION, kTraceError,
- "DeRegisterVoiceEngineObserver() observer already disabled");
- return 0;
- }
-
- _voiceEngineObserver = false;
- _voiceEngineObserverPtr = NULL;
-
- // Deregister the observer in all active channels
- voe::ScopedChannel sc(_channelManager);
- void* iterator(NULL);
- voe::Channel* channelPtr = sc.GetFirstChannel(iterator);
- while (channelPtr != NULL)
- {
- channelPtr->DeRegisterVoiceEngineObserver();
- channelPtr = sc.GetNextChannel(iterator);
- }
-
- return 0;
-}
-
-int VoEBaseImpl::Init(AudioDeviceModule* external_adm)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "Init(external_adm=0x%p)", external_adm);
- CriticalSectionScoped cs(*_apiCritPtr);
-
- if (_engineStatistics.Initialized())
- {
- return 0;
- }
-
- if (_moduleProcessThreadPtr)
- {
- if (_moduleProcessThreadPtr->Start() != 0)
- {
- _engineStatistics.SetLastError(VE_THREAD_ERROR, kTraceError,
- "Init() failed to start module process thread");
- return -1;
- }
- }
-
- // Create an internal ADM if the user has not added an external
- // ADM implementation as input to Init().
- if (external_adm == NULL)
- {
- // Create the internal ADM implementation.
- _audioDevicePtr = AudioDeviceModuleImpl::Create(
- VoEId(_instanceId, -1), _audioDeviceLayer);
-
- if (_audioDevicePtr == NULL)
- {
- _engineStatistics.SetLastError(VE_NO_MEMORY, kTraceCritical,
- "Init() failed to create the ADM");
- return -1;
- }
- }
- else
- {
- // Use the already existing external ADM implementation.
- _audioDevicePtr = external_adm;
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "An external ADM implementation will be used in VoiceEngine");
- }
-
- // Increase the reference counter for both external and internal usage.
- _audioDevicePtr->AddRef();
-
- // Register the ADM to the process thread, which will drive the error
- // callback mechanism
- if (_moduleProcessThreadPtr &&
- _moduleProcessThreadPtr->RegisterModule(_audioDevicePtr) != 0)
- {
- _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
- kTraceError,
- "Init() failed to register the ADM");
- return -1;
- }
-
- bool available(false);
-
- // --------------------
- // Reinitialize the ADM
-
- // Register the AudioObserver implementation
- if (_audioDevicePtr->RegisterEventObserver(this) != 0) {
- _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
- kTraceWarning,
- "Init() failed to register event observer "
- "for the ADM");
- }
-
- // Register the AudioTransport implementation
- if (_audioDevicePtr->RegisterAudioCallback(this) != 0) {
- _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
- kTraceWarning,
- "Init() failed to register audio callback "
- "for the ADM");
- }
-
- // ADM initialization
- if (_audioDevicePtr->Init() != 0)
- {
- _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
- kTraceError,
- "Init() failed to initialize the ADM");
- return -1;
- }
-
- // Initialize the default speaker
- if (_audioDevicePtr->SetPlayoutDevice(WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE)
- != 0)
- {
- _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR, kTraceInfo,
- "Init() failed to set the default output device");
- }
- if (_audioDevicePtr->SpeakerIsAvailable(&available) != 0)
- {
- _engineStatistics.SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceInfo,
- "Init() failed to check speaker availability, trying to "
- "initialize speaker anyway");
- }
- else if (!available)
- {
- _engineStatistics.SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceInfo,
- "Init() speaker not available, trying to initialize speaker "
- "anyway");
- }
- if (_audioDevicePtr->InitSpeaker() != 0)
- {
- _engineStatistics.SetLastError(VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceInfo,
- "Init() failed to initialize the speaker");
- }
-
- // Initialize the default microphone
- if (_audioDevicePtr->SetRecordingDevice(WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE)
- != 0)
- {
- _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceInfo,
- "Init() failed to set the default input device");
- }
- if (_audioDevicePtr->MicrophoneIsAvailable(&available) != 0)
- {
- _engineStatistics.SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceInfo,
- "Init() failed to check microphone availability, trying to "
- "initialize microphone anyway");
- }
- else if (!available)
- {
- _engineStatistics.SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceInfo,
- "Init() microphone not available, trying to initialize "
- "microphone anyway");
- }
- if (_audioDevicePtr->InitMicrophone() != 0)
- {
- _engineStatistics.SetLastError(VE_CANNOT_ACCESS_MIC_VOL, kTraceInfo,
- "Init() failed to initialize the microphone");
- }
-
- // Set number of channels
- if (_audioDevicePtr->StereoPlayoutIsAvailable(&available) != 0) {
- _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
- "Init() failed to query stereo playout "
- "mode");
- }
- if (_audioDevicePtr->SetStereoPlayout(available) != 0)
- {
- _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
- "Init() failed to set mono/stereo playout mode");
- }
-
- // TODO(andrew): These functions don't tell us whether stereo recording
- // is truly available. We simply set the AudioProcessing input to stereo
- // here, because we have to wait until receiving the first frame to
- // determine the actual number of channels anyway.
- //
- // These functions may be changed; tracked here:
- // http://code.google.com/p/webrtc/issues/detail?id=204
- _audioDevicePtr->StereoRecordingIsAvailable(&available);
- if (_audioDevicePtr->SetStereoRecording(available) != 0)
- {
- _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
- "Init() failed to set mono/stereo recording mode");
- }
-
- // APM initialization done after sound card since we need
- // to know if we support stereo recording or not.
-
- // Create the AudioProcessing Module if it does not exist.
-
- if (_audioProcessingModulePtr == NULL)
- {
- _audioProcessingModulePtr = AudioProcessing::Create(
- VoEId(_instanceId, -1));
- if (_audioProcessingModulePtr == NULL)
- {
- _engineStatistics.SetLastError(VE_NO_MEMORY, kTraceCritical,
- "Init() failed to create the AP module");
- return -1;
- }
- // Ensure that mixers in both directions has access to the created APM
- _transmitMixerPtr->SetAudioProcessingModule(_audioProcessingModulePtr);
- _outputMixerPtr->SetAudioProcessingModule(_audioProcessingModulePtr);
-
- if (_audioProcessingModulePtr->echo_cancellation()->
- set_device_sample_rate_hz(
- kVoiceEngineAudioProcessingDeviceSampleRateHz))
- {
- _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
- "Init() failed to set the device sample rate to 48K for AP "
- " module");
- return -1;
- }
- // Using 8 kHz as inital Fs. Might be changed already at first call.
- if (_audioProcessingModulePtr->set_sample_rate_hz(8000))
- {
- _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
- "Init() failed to set the sample rate to 8K for AP module");
- return -1;
- }
-
- // Assume mono output until a send codec is set, and stereo input until
- // we receive the first captured frame. We set stereo input here to
- // avoid triggering a possible error in SetSendCodec when a stereo
- // codec is selected.
- if (_audioProcessingModulePtr->set_num_channels(2, 1) != 0)
- {
- _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceError,
- "Init() failed to set channels for the primary audio stream");
- return -1;
- }
-
- if (_audioProcessingModulePtr->set_num_reverse_channels(1) != 0)
- {
- _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceError,
- "Init() failed to set channels for the primary audio stream");
- return -1;
- }
- // high-pass filter
- if (_audioProcessingModulePtr->high_pass_filter()->Enable(
- WEBRTC_VOICE_ENGINE_HP_DEFAULT_STATE) != 0)
- {
- _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
- "Init() failed to set the high-pass filter for AP module");
- return -1;
- }
- // Echo Cancellation
- if (_audioProcessingModulePtr->echo_cancellation()->
- enable_drift_compensation(false) != 0)
- {
- _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
- "Init() failed to set drift compensation for AP module");
- return -1;
- }
- if (_audioProcessingModulePtr->echo_cancellation()->Enable(
- WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE))
- {
- _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
- "Init() failed to set echo cancellation state for AP module");
- return -1;
- }
- // Noise Reduction
- if (_audioProcessingModulePtr->noise_suppression()->set_level(
- (NoiseSuppression::Level) WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE)
- != 0)
- {
- _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
- "Init() failed to set noise reduction level for AP module");
- return -1;
- }
- if (_audioProcessingModulePtr->noise_suppression()->Enable(
- WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE) != 0)
- {
- _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
- "Init() failed to set noise reduction state for AP module");
- return -1;
- }
- // Automatic Gain control
- if (_audioProcessingModulePtr->gain_control()->set_analog_level_limits(
- kMinVolumeLevel,kMaxVolumeLevel) != 0)
- {
- _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
- "Init() failed to set AGC analog level for AP module");
- return -1;
- }
- if (_audioProcessingModulePtr->gain_control()->set_mode(
- (GainControl::Mode) WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE)
- != 0)
- {
- _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
- "Init() failed to set AGC mode for AP module");
- return -1;
- }
- if (_audioProcessingModulePtr->gain_control()->Enable(
- WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE)
- != 0)
- {
- _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
- "Init() failed to set AGC state for AP module");
- return -1;
- }
- // VAD
- if (_audioProcessingModulePtr->voice_detection()->Enable(
- WEBRTC_VOICE_ENGINE_VAD_DEFAULT_STATE)
- != 0)
- {
- _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
- "Init() failed to set VAD state for AP module");
- return -1;
- }
- }
-
- // Set default AGC mode for the ADM
-#ifdef WEBRTC_VOICE_ENGINE_AGC
- bool enable(false);
- if (_audioProcessingModulePtr->gain_control()->mode()
- != GainControl::kFixedDigital)
- {
- enable = _audioProcessingModulePtr->gain_control()->is_enabled();
- // Only set the AGC mode for the ADM when Adaptive AGC mode is selected
- if (_audioDevicePtr->SetAGC(enable) != 0)
- {
- _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
- kTraceError, "Init() failed to set default AGC mode in ADM 0");
- }
- }
-#endif
-
- return _engineStatistics.SetInitialized();
-}
-
-int VoEBaseImpl::Terminate()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "Terminate()");
- CriticalSectionScoped cs(*_apiCritPtr);
- return TerminateInternal();
-}
-
-int VoEBaseImpl::MaxNumOfChannels()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "MaxNumOfChannels()");
- WebRtc_Word32 maxNumOfChannels = _channelManager.MaxNumOfChannels();
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "MaxNumOfChannels() => %d", maxNumOfChannels);
- return (maxNumOfChannels);
-}
-
-int VoEBaseImpl::CreateChannel()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "CreateChannel()");
- CriticalSectionScoped cs(*_apiCritPtr);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- WebRtc_Word32 channelId = -1;
-
- if (!_channelManager.CreateChannel(channelId))
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_CREATED, kTraceError,
- "CreateChannel() failed to allocate "
- "memory for channel");
- return -1;
- }
-
- bool destroyChannel(false);
- {
- voe::ScopedChannel sc(_channelManager, channelId);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_CREATED, kTraceError,
- "CreateChannel() failed to allocate"
- " memory for channel");
- return -1;
- }
- else if (channelPtr->SetEngineInformation(_engineStatistics,
- *_outputMixerPtr,
- *_transmitMixerPtr,
- *_moduleProcessThreadPtr,
- *_audioDevicePtr,
- _voiceEngineObserverPtr,
- &_callbackCritSect) != 0)
- {
- destroyChannel = true;
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_CREATED, kTraceError,
- "CreateChannel() failed to "
- "associate engine and channel."
- " Destroying channel.");
- }
- else if (channelPtr->Init() != 0)
- {
- destroyChannel = true;
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_CREATED, kTraceError,
- "CreateChannel() failed to "
- "initialize channel. Destroying"
- " channel.");
- }
- }
- if (destroyChannel)
- {
- _channelManager.DestroyChannel(channelId);
- return -1;
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "CreateChannel() => %d", channelId);
- return channelId;
-}
-
-int VoEBaseImpl::DeleteChannel(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "DeleteChannel(channel=%d)", channel);
- CriticalSectionScoped cs(*_apiCritPtr);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- {
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "DeleteChannel() failed to locate "
- "channel");
- return -1;
- }
- }
-
- if (_channelManager.DestroyChannel(channel) != 0)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "DeleteChannel() failed to destroy "
- "channel");
- return -1;
- }
-
- if (StopSend() != 0)
- {
- return -1;
- }
-
- if (StopPlayout() != 0)
- {
- return -1;
- }
-
- return 0;
-}
-
-int VoEBaseImpl::SetLocalReceiver(int channel, int port, int RTCPport,
- const char ipAddr[64],
- const char multiCastAddr[64])
-{
- // Inititialize local receive sockets (RTP and RTCP).
- //
- // The sockets are always first closed and then created again by this
- // function call. The created sockets are by default also used for
- // transmission (unless source port is set in SetSendDestination).
- //
- // Note that, sockets can also be created automatically if a user calls
- // SetSendDestination and StartSend without having called SetLocalReceiver
- // first. The sockets are then created at the first packet transmission.
-
- CriticalSectionScoped cs(*_apiCritPtr);
- if (ipAddr == NULL && multiCastAddr == NULL)
- {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetLocalReceiver(channel=%d, port=%d, RTCPport=%d)",
- channel, port, RTCPport);
- }
- else if (ipAddr != NULL && multiCastAddr == NULL)
- {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetLocalReceiver(channel=%d, port=%d, RTCPport=%d, "
- "ipAddr=%s)", channel, port, RTCPport, ipAddr);
- }
- else if (ipAddr == NULL && multiCastAddr != NULL)
- {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetLocalReceiver(channel=%d, port=%d, RTCPport=%d, "
- "multiCastAddr=%s)", channel, port, RTCPport,
- multiCastAddr);
- }
- else
- {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetLocalReceiver(channel=%d, port=%d, RTCPport=%d, "
- "ipAddr=%s, multiCastAddr=%s)", channel, port,
- RTCPport, ipAddr, multiCastAddr);
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if ((port < 0) || (port > 65535))
- {
- _engineStatistics.SetLastError(VE_INVALID_PORT_NMBR, kTraceError,
- "SetLocalReceiver() invalid RTP port");
- return -1;
- }
- if (((RTCPport != kVoEDefault) && (RTCPport < 0)) || ((RTCPport
- != kVoEDefault) && (RTCPport > 65535)))
- {
- _engineStatistics.SetLastError(VE_INVALID_PORT_NMBR, kTraceError,
- "SetLocalReceiver() invalid RTCP port");
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetLocalReceiver() failed to locate "
- "channel");
- return -1;
- }
-
- // Cast RTCP port. In the RTP module 0 corresponds to RTP port + 1 in
- // the module, which is the default.
- WebRtc_UWord16 rtcpPortUW16(0);
- if (RTCPport != kVoEDefault)
- {
- rtcpPortUW16 = static_cast<WebRtc_UWord16> (RTCPport);
- }
-
- return channelPtr->SetLocalReceiver(port, rtcpPortUW16, ipAddr,
- multiCastAddr);
-#else
- _engineStatistics.SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED,
- kTraceWarning,
- "SetLocalReceiver() VoE is built for "
- "external transport");
- return -1;
-#endif
-}
-
-int VoEBaseImpl::GetLocalReceiver(int channel, int& port, int& RTCPport,
- char ipAddr[64])
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetLocalReceiver(channel=%d, ipAddr[]=?)", channel);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetLocalReceiver() failed to locate "
- "channel");
- return -1;
- }
- WebRtc_Word32 ret = channelPtr->GetLocalReceiver(port, RTCPport, ipAddr);
- if (ipAddr != NULL)
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "GetLocalReceiver() => port=%d, RTCPport=%d, ipAddr=%s",
- port, RTCPport, ipAddr);
- }
- else
- {
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "GetLocalReceiver() => port=%d, RTCPport=%d", port,
- RTCPport);
- }
- return ret;
-#else
- _engineStatistics.SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED,
- kTraceWarning,
- "SetLocalReceiver() VoE is built for "
- "external transport");
- return -1;
-#endif
-}
-
-int VoEBaseImpl::SetSendDestination(int channel, int port, const char* ipaddr,
- int sourcePort, int RTCPport)
-{
- WEBRTC_TRACE(
- kTraceApiCall,
- kTraceVoice,
- VoEId(_instanceId, -1),
- "SetSendDestination(channel=%d, port=%d, ipaddr=%s,"
- "sourcePort=%d, RTCPport=%d)",
- channel, port, ipaddr, sourcePort, RTCPport);
- CriticalSectionScoped cs(*_apiCritPtr);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetSendDestination() failed to locate "
- "channel");
- return -1;
- }
- if ((port < 0) || (port > 65535))
- {
- _engineStatistics.SetLastError(VE_INVALID_PORT_NMBR, kTraceError,
- "SetSendDestination() invalid RTP port");
- return -1;
- }
- if (((RTCPport != kVoEDefault) && (RTCPport < 0)) || ((RTCPport
- != kVoEDefault) && (RTCPport > 65535)))
- {
- _engineStatistics.SetLastError(VE_INVALID_PORT_NMBR, kTraceError,
- "SetSendDestination() invalid RTCP "
- "port");
- return -1;
- }
- if (((sourcePort != kVoEDefault) && (sourcePort < 0)) || ((sourcePort
- != kVoEDefault) && (sourcePort > 65535)))
- {
- _engineStatistics.SetLastError(VE_INVALID_PORT_NMBR, kTraceError,
- "SetSendDestination() invalid source "
- "port");
- return -1;
- }
-
- // Cast RTCP port. In the RTP module 0 corresponds to RTP port + 1 in the
- // module, which is the default.
- WebRtc_UWord16 rtcpPortUW16(0);
- if (RTCPport != kVoEDefault)
- {
- rtcpPortUW16 = static_cast<WebRtc_UWord16> (RTCPport);
- WEBRTC_TRACE(
- kTraceInfo,
- kTraceVoice,
- VoEId(_instanceId, channel),
- "SetSendDestination() non default RTCP port %u will be "
- "utilized",
- rtcpPortUW16);
- }
-
- return channelPtr->SetSendDestination(port, ipaddr, sourcePort,
- rtcpPortUW16);
-#else
- _engineStatistics.SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED,
- kTraceWarning,
- "SetSendDestination() VoE is built for "
- "external transport");
- return -1;
-#endif
-}
-
-int VoEBaseImpl::GetSendDestination(int channel, int& port, char ipAddr[64],
- int& sourcePort, int& RTCPport)
-{
- WEBRTC_TRACE(
- kTraceApiCall,
- kTraceVoice,
- VoEId(_instanceId, -1),
- "GetSendDestination(channel=%d, ipAddr[]=?, sourcePort=?,"
- "RTCPport=?)",
- channel);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetSendDestination() failed to locate "
- "channel");
- return -1;
- }
- WebRtc_Word32 ret = channelPtr->GetSendDestination(port, ipAddr,
- sourcePort, RTCPport);
- if (ipAddr != NULL)
- {
- WEBRTC_TRACE(
- kTraceStateInfo,
- kTraceVoice,
- VoEId(_instanceId, -1),
- "GetSendDestination() => port=%d, RTCPport=%d, ipAddr=%s, "
- "sourcePort=%d, RTCPport=%d",
- port, RTCPport, ipAddr, sourcePort, RTCPport);
- }
- else
- {
- WEBRTC_TRACE(
- kTraceStateInfo,
- kTraceVoice,
- VoEId(_instanceId, -1),
- "GetSendDestination() => port=%d, RTCPport=%d, "
- "sourcePort=%d, RTCPport=%d",
- port, RTCPport, sourcePort, RTCPport);
- }
- return ret;
-#else
- _engineStatistics.SetLastError(VE_EXTERNAL_TRANSPORT_ENABLED,
- kTraceWarning,
- "GetSendDestination() VoE is built for "
- "external transport");
- return -1;
-#endif
-}
-
-int VoEBaseImpl::StartReceive(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "StartReceive(channel=%d)", channel);
- CriticalSectionScoped cs(*_apiCritPtr);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "StartReceive() failed to locate "
- "channel");
- return -1;
- }
- return channelPtr->StartReceiving();
-}
-
-int VoEBaseImpl::StopReceive(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "StopListen(channel=%d)", channel);
- CriticalSectionScoped cs(*_apiCritPtr);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetLocalReceiver() failed to locate "
- "channel");
- return -1;
- }
- return channelPtr->StopReceiving();
-}
-
-int VoEBaseImpl::StartPlayout(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "StartPlayout(channel=%d)", channel);
- CriticalSectionScoped cs(*_apiCritPtr);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "StartPlayout() failed to locate "
- "channel");
- return -1;
- }
- if (channelPtr->Playing())
- {
- return 0;
- }
- if (StartPlayout() != 0)
- {
- _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
- kTraceError,
- "StartPlayout() failed to start "
- "playout");
- return -1;
- }
- return channelPtr->StartPlayout();
-}
-
-int VoEBaseImpl::StopPlayout(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "StopPlayout(channel=%d)", channel);
- CriticalSectionScoped cs(*_apiCritPtr);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "StopPlayout() failed to locate "
- "channel");
- return -1;
- }
- if (channelPtr->StopPlayout() != 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StopPlayout() failed to stop playout for channel %d",
- channel);
- }
- return StopPlayout();
-}
-
-int VoEBaseImpl::StartSend(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "StartSend(channel=%d)", channel);
- CriticalSectionScoped cs(*_apiCritPtr);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "StartSend() failed to locate channel");
- return -1;
- }
- if (channelPtr->Sending())
- {
- return 0;
- }
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!channelPtr->ExternalTransport()
- && !channelPtr->SendSocketsInitialized())
- {
- _engineStatistics.SetLastError(VE_DESTINATION_NOT_INITED, kTraceError,
- "StartSend() must set send destination "
- "first");
- return -1;
- }
-#endif
- if (StartSend() != 0)
- {
- _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
- kTraceError,
- "StartSend() failed to start recording");
- return -1;
- }
- return channelPtr->StartSend();
-}
-
-int VoEBaseImpl::StopSend(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "StopSend(channel=%d)", channel);
- CriticalSectionScoped cs(*_apiCritPtr);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "StopSend() failed to locate channel");
- return -1;
- }
- if (channelPtr->StopSend() != 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- "StopSend() failed to stop sending for channel %d",
- channel);
- }
- return StopSend();
-}
-
-int VoEBaseImpl::GetVersion(char version[1024])
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetVersion(version=?)");
- assert(kVoiceEngineVersionMaxMessageSize == 1024);
-
- if (version == NULL)
- {
- _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError);
- return (-1);
- }
-
- char versionBuf[kVoiceEngineVersionMaxMessageSize];
- char* versionPtr = versionBuf;
-
- WebRtc_Word32 len = 0;
- WebRtc_Word32 accLen = 0;
-
- len = AddVoEVersion(versionPtr);
- if (len == -1)
- {
- return -1;
- }
- versionPtr += len;
- accLen += len;
- assert(accLen < kVoiceEngineVersionMaxMessageSize);
-
- len = AddBuildInfo(versionPtr);
- if (len == -1)
- {
- return -1;
- }
- versionPtr += len;
- accLen += len;
- assert(accLen < kVoiceEngineVersionMaxMessageSize);
-
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
- len = AddExternalTransportBuild(versionPtr);
- if (len == -1)
- {
- return -1;
- }
- versionPtr += len;
- accLen += len;
- assert(accLen < kVoiceEngineVersionMaxMessageSize);
-#endif
-#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
- len = AddExternalRecAndPlayoutBuild(versionPtr);
- if (len == -1)
- {
- return -1;
- }
- versionPtr += len;
- accLen += len;
- assert(accLen < kVoiceEngineVersionMaxMessageSize);
- #endif
-
- memcpy(version, versionBuf, accLen);
- version[accLen] = '\0';
-
- // to avoid the truncation in the trace, split the string into parts
- char partOfVersion[256];
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "GetVersion() =>");
- for (int partStart = 0; partStart < accLen;)
- {
- memset(partOfVersion, 0, sizeof(partOfVersion));
- int partEnd = partStart + 180;
- while (version[partEnd] != '\n' && version[partEnd] != '\0')
- {
- partEnd--;
- }
- if (partEnd < accLen)
- {
- memcpy(partOfVersion, &version[partStart], partEnd - partStart);
- }
- else
- {
- memcpy(partOfVersion, &version[partStart], accLen - partStart);
- }
- partStart = partEnd;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "%s", partOfVersion);
- }
-
- return 0;
-}
-
-WebRtc_Word32 VoEBaseImpl::AddBuildInfo(char* str) const
-{
- return sprintf(str, "Build: %s\n", BUILDINFO);
-}
-
-WebRtc_Word32 VoEBaseImpl::AddVoEVersion(char* str) const
-{
- return sprintf(str, "VoiceEngine 4.1.0\n");
-}
-
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
-WebRtc_Word32 VoEBaseImpl::AddExternalTransportBuild(char* str) const
-{
- return sprintf(str, "External transport build\n");
-}
-#endif
-
-#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
-WebRtc_Word32 VoEBaseImpl::AddExternalRecAndPlayoutBuild(char* str) const
-{
- return sprintf(str, "External recording and playout build\n");
-}
-#endif
-
-int VoEBaseImpl::LastError()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "LastError()");
- return (_engineStatistics.LastError());
-}
-
-
-int VoEBaseImpl::SetNetEQPlayoutMode(int channel, NetEqModes mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetNetEQPlayoutMode(channel=%i, mode=%i)", channel, mode);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetNetEQPlayoutMode() failed to locate"
- " channel");
- return -1;
- }
- return channelPtr->SetNetEQPlayoutMode(mode);
-}
-
-int VoEBaseImpl::GetNetEQPlayoutMode(int channel, NetEqModes& mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetNetEQPlayoutMode(channel=%i, mode=?)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetNetEQPlayoutMode() failed to locate"
- " channel");
- return -1;
- }
- return channelPtr->GetNetEQPlayoutMode(mode);
-}
-
-int VoEBaseImpl::SetNetEQBGNMode(int channel, NetEqBgnModes mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetNetEQBGNMode(channel=%i, mode=%i)", channel, mode);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetNetEQBGNMode() failed to locate "
- "channel");
- return -1;
- }
- return channelPtr->SetNetEQBGNMode(mode);
-}
-
-int VoEBaseImpl::GetNetEQBGNMode(int channel, NetEqBgnModes& mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetNetEQBGNMode(channel=%i, mode=?)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetNetEQBGNMode() failed to locate "
- "channel");
- return -1;
- }
- return channelPtr->GetNetEQBGNMode(mode);
-}
-
-int VoEBaseImpl::SetOnHoldStatus(int channel, bool enable, OnHoldModes mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetOnHoldStatus(channel=%d, enable=%d, mode=%d)", channel,
- enable, mode);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetOnHoldStatus() failed to locate "
- "channel");
- return -1;
- }
- return channelPtr->SetOnHoldStatus(enable, mode);
-}
-
-int VoEBaseImpl::GetOnHoldStatus(int channel, bool& enabled, OnHoldModes& mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetOnHoldStatus(channel=%d, enabled=?, mode=?)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetOnHoldStatus() failed to locate "
- "channel");
- return -1;
- }
- return channelPtr->GetOnHoldStatus(enabled, mode);
-}
-
-WebRtc_Word32 VoEBaseImpl::StartPlayout()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "VoEBaseImpl::StartPlayout()");
- if (_audioDevicePtr->Playing())
- {
- return 0;
- }
- if (!_externalPlayout)
- {
- if (_audioDevicePtr->InitPlayout() != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "StartPlayout() failed to initialize playout");
- return -1;
- }
- if (_audioDevicePtr->StartPlayout() != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "StartPlayout() failed to start playout");
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_Word32 VoEBaseImpl::StopPlayout()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "VoEBaseImpl::StopPlayout()");
-
- WebRtc_Word32 numOfChannels = _channelManager.NumOfChannels();
- if (numOfChannels <= 0)
- {
- return 0;
- }
-
- WebRtc_UWord16 nChannelsPlaying(0);
- WebRtc_Word32* channelsArray = new WebRtc_Word32[numOfChannels];
-
- // Get number of playing channels
- _channelManager.GetChannelIds(channelsArray, numOfChannels);
- for (int i = 0; i < numOfChannels; i++)
- {
- voe::ScopedChannel sc(_channelManager, channelsArray[i]);
- voe::Channel* chPtr = sc.ChannelPtr();
- if (chPtr)
- {
- if (chPtr->Playing())
- {
- nChannelsPlaying++;
- }
- }
- }
- delete[] channelsArray;
-
- // Stop audio-device playing if no channel is playing out
- if (nChannelsPlaying == 0)
- {
- if (_audioDevicePtr->StopPlayout() != 0)
- {
- _engineStatistics.SetLastError(VE_CANNOT_STOP_PLAYOUT, kTraceError,
- "StopPlayout() failed to stop "
- "playout");
- return -1;
- }
- }
- return 0;
-}
-
-WebRtc_Word32 VoEBaseImpl::StartSend()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "VoEBaseImpl::StartSend()");
- if (_audioDevicePtr->Recording())
- {
- return 0;
- }
- if (!_externalRecording)
- {
- if (_audioDevicePtr->InitRecording() != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "StartSend() failed to initialize recording");
- return -1;
- }
- if (_audioDevicePtr->StartRecording() != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId, -1),
- "StartSend() failed to start recording");
- return -1;
- }
- }
-
- return 0;
-}
-
-WebRtc_Word32 VoEBaseImpl::StopSend()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "VoEBaseImpl::StopSend()");
-
- if ((NumOfSendingChannels() == 0) && !_transmitMixerPtr->IsRecordingMic())
- {
- // Stop audio-device recording if no channel is recording
- if (_audioDevicePtr->StopRecording() != 0)
- {
- _engineStatistics.SetLastError(VE_CANNOT_STOP_RECORDING,
- kTraceError,
- "StopSend() failed to stop "
- "recording");
- return -1;
- }
- _transmitMixerPtr->StopSend();
- }
-
- return 0;
-}
-
-WebRtc_Word32 VoEBaseImpl::TerminateInternal()
-{
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "VoEBaseImpl::TerminateInternal()");
-
- // Delete any remaining channel objects
- WebRtc_Word32 numOfChannels = _channelManager.NumOfChannels();
- if (numOfChannels > 0)
- {
- WebRtc_Word32* channelsArray = new WebRtc_Word32[numOfChannels];
- _channelManager.GetChannelIds(channelsArray, numOfChannels);
- for (int i = 0; i < numOfChannels; i++)
- {
- DeleteChannel(channelsArray[i]);
- }
- delete[] channelsArray;
- }
-
- if (_moduleProcessThreadPtr)
- {
- if (_audioDevicePtr)
- {
- if (_moduleProcessThreadPtr->DeRegisterModule(_audioDevicePtr) != 0)
- {
- _engineStatistics.SetLastError(VE_THREAD_ERROR, kTraceError,
- "TerminateInternal() failed to "
- "deregister ADM");
- }
- }
- if (_moduleProcessThreadPtr->Stop() != 0)
- {
- _engineStatistics.SetLastError(VE_THREAD_ERROR, kTraceError,
- "TerminateInternal() failed to stop "
- "module process thread");
- }
- }
-
- // Audio Device Module
-
- if (_audioDevicePtr != NULL)
- {
- if (_audioDevicePtr->StopPlayout() != 0)
- {
- _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
- "TerminateInternal() failed to stop "
- "playout");
- }
- if (_audioDevicePtr->StopRecording() != 0)
- {
- _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceWarning,
- "TerminateInternal() failed to stop "
- "recording");
- }
- if (_audioDevicePtr->RegisterEventObserver(NULL) != 0) {
- _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
- kTraceWarning,
- "TerminateInternal() failed to de-"
- "register event observer for the ADM");
- }
- if (_audioDevicePtr->RegisterAudioCallback(NULL) != 0) {
- _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
- kTraceWarning,
- "TerminateInternal() failed to de-"
- "register audio callback for the ADM");
- }
- if (_audioDevicePtr->Terminate() != 0)
- {
- _engineStatistics.SetLastError(VE_AUDIO_DEVICE_MODULE_ERROR,
- kTraceError,
- "TerminateInternal() failed to "
- "terminate the ADM");
- }
-
- _audioDevicePtr->Release();
- _audioDevicePtr = NULL;
- }
-
- // AP module
-
- if (_audioProcessingModulePtr != NULL)
- {
- _transmitMixerPtr->SetAudioProcessingModule(NULL);
- AudioProcessing::Destroy(_audioProcessingModulePtr);
- _audioProcessingModulePtr = NULL;
- }
-
- return _engineStatistics.SetUnInitialized();
-}
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_base_impl.h b/trunk/src/voice_engine/main/source/voe_base_impl.h
deleted file mode 100644
index 6d470a0..0000000
--- a/trunk/src/voice_engine/main/source/voe_base_impl.h
+++ /dev/null
@@ -1,152 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H
-#define WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H
-
-#include "voe_base.h"
-
-#include "module_common_types.h"
-#include "ref_count.h"
-#include "shared_data.h"
-
-namespace webrtc
-{
-
-class ProcessThread;
-
-class VoEBaseImpl: public virtual voe::SharedData,
- public VoEBase,
- public voe::RefCount,
- public AudioTransport,
- public AudioDeviceObserver
-{
-public:
- virtual int Release();
-
- virtual int RegisterVoiceEngineObserver(VoiceEngineObserver& observer);
-
- virtual int DeRegisterVoiceEngineObserver();
-
- virtual int Init(AudioDeviceModule* external_adm = NULL);
-
- virtual int Terminate();
-
- virtual int MaxNumOfChannels();
-
- virtual int CreateChannel();
-
- virtual int DeleteChannel(int channel);
-
- virtual int SetLocalReceiver(int channel, int port,
- int RTCPport = kVoEDefault,
- const char ipAddr[64] = NULL,
- const char multiCastAddr[64] = NULL);
-
- virtual int GetLocalReceiver(int channel, int& port, int& RTCPport,
- char ipAddr[64]);
-
- virtual int SetSendDestination(int channel, int port,
- const char ipAddr[64],
- int sourcePort = kVoEDefault,
- int RTCPport = kVoEDefault);
-
- virtual int GetSendDestination(int channel,
- int& port,
- char ipAddr[64],
- int& sourcePort,
- int& RTCPport);
-
- virtual int StartReceive(int channel);
-
- virtual int StartPlayout(int channel);
-
- virtual int StartSend(int channel);
-
- virtual int StopReceive(int channel);
-
- virtual int StopPlayout(int channel);
-
- virtual int StopSend(int channel);
-
- virtual int SetNetEQPlayoutMode(int channel, NetEqModes mode);
-
- virtual int GetNetEQPlayoutMode(int channel, NetEqModes& mode);
-
- virtual int SetNetEQBGNMode(int channel, NetEqBgnModes mode);
-
- virtual int GetNetEQBGNMode(int channel, NetEqBgnModes& mode);
-
-
- virtual int SetOnHoldStatus(int channel,
- bool enable,
- OnHoldModes mode = kHoldSendAndPlay);
-
- virtual int GetOnHoldStatus(int channel, bool& enabled, OnHoldModes& mode);
-
- virtual int GetVersion(char version[1024]);
-
- virtual int LastError();
-
- // AudioTransport
- virtual WebRtc_Word32
- RecordedDataIsAvailable(const WebRtc_Word8* audioSamples,
- const WebRtc_UWord32 nSamples,
- const WebRtc_UWord8 nBytesPerSample,
- const WebRtc_UWord8 nChannels,
- const WebRtc_UWord32 samplesPerSec,
- const WebRtc_UWord32 totalDelayMS,
- const WebRtc_Word32 clockDrift,
- const WebRtc_UWord32 currentMicLevel,
- WebRtc_UWord32& newMicLevel);
-
- virtual WebRtc_Word32 NeedMorePlayData(const WebRtc_UWord32 nSamples,
- const WebRtc_UWord8 nBytesPerSample,
- const WebRtc_UWord8 nChannels,
- const WebRtc_UWord32 samplesPerSec,
- WebRtc_Word8* audioSamples,
- WebRtc_UWord32& nSamplesOut);
-
- // AudioDeviceObserver
- virtual void OnErrorIsReported(const ErrorCode error);
- virtual void OnWarningIsReported(const WarningCode warning);
-
-protected:
- VoEBaseImpl();
- virtual ~VoEBaseImpl();
-
-private:
- WebRtc_Word32 StartPlayout();
- WebRtc_Word32 StopPlayout();
- WebRtc_Word32 StartSend();
- WebRtc_Word32 StopSend();
- WebRtc_Word32 TerminateInternal();
-
- WebRtc_Word32 AddBuildInfo(char* str) const;
- WebRtc_Word32 AddVoEVersion(char* str) const;
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
- WebRtc_Word32 AddExternalTransportBuild(char* str) const;
-#endif
-#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
- WebRtc_Word32 AddExternalRecAndPlayoutBuild(char* str) const;
-#endif
- VoiceEngineObserver* _voiceEngineObserverPtr;
- CriticalSectionWrapper& _callbackCritSect;
-
- bool _voiceEngineObserver;
- WebRtc_UWord32 _oldVoEMicLevel;
- WebRtc_UWord32 _oldMicLevel;
- AudioFrame _audioFrame;
-
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_BASE_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_call_report_impl.cc b/trunk/src/voice_engine/main/source/voe_call_report_impl.cc
deleted file mode 100644
index 8be8e01..0000000
--- a/trunk/src/voice_engine/main/source/voe_call_report_impl.cc
+++ /dev/null
@@ -1,432 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voe_call_report_impl.h"
-
-#include "audio_processing.h"
-#include "channel.h"
-#include "critical_section_wrapper.h"
-#include "file_wrapper.h"
-#include "trace.h"
-#include "voe_errors.h"
-#include "voice_engine_impl.h"
-
-namespace webrtc
-{
-
-VoECallReport* VoECallReport::GetInterface(VoiceEngine* voiceEngine)
-{
-#ifndef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
- return NULL;
-#else
- if (NULL == voiceEngine)
- {
- return NULL;
- }
- VoiceEngineImpl* s =
- reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
- VoECallReportImpl* d = s;
- (*d)++;
- return (d);
-#endif
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
-
-VoECallReportImpl::VoECallReportImpl() :
- _file(*FileWrapper::Create())
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
- "VoECallReportImpl() - ctor");
-}
-
-VoECallReportImpl::~VoECallReportImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
- "~VoECallReportImpl() - dtor");
- delete &_file;
-}
-
-int VoECallReportImpl::Release()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "VoECallReportImpl::Release()");
- (*this)--;
- int refCount = GetCount();
- if (refCount < 0)
- {
- Reset();
- _engineStatistics.SetLastError(VE_INTERFACE_NOT_FOUND,
- kTraceWarning);
- return (-1);
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "VoECallReportImpl reference counter = %d", refCount);
- return (refCount);
-}
-
-int VoECallReportImpl::ResetCallReportStatistics(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "ResetCallReportStatistics(channel=%d)", channel);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- assert(_audioProcessingModulePtr != NULL);
-
- bool echoMode =
- _audioProcessingModulePtr->echo_cancellation()->are_metrics_enabled();
-
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- " current AudioProcessingModule echo metric state %d)",
- echoMode);
- // Reset the APM statistics
- if (_audioProcessingModulePtr->echo_cancellation()->enable_metrics(true)
- != 0)
- {
- _engineStatistics.SetLastError(VE_APM_ERROR, kTraceError,
- "ResetCallReportStatistics() unable to "
- "set the AudioProcessingModule echo "
- "metrics state");
- return -1;
- }
- // Restore metric states
- _audioProcessingModulePtr->echo_cancellation()->enable_metrics(echoMode);
-
- // Reset channel dependent statistics
- if (channel != -1)
- {
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "ResetCallReportStatistics() failed "
- "to locate channel");
- return -1;
- }
- channelPtr->ResetDeadOrAliveCounters();
- channelPtr->ResetRTCPStatistics();
- }
- else
- {
- WebRtc_Word32 numOfChannels = _channelManager.NumOfChannels();
- if (numOfChannels <= 0)
- {
- return 0;
- }
- WebRtc_Word32* channelsArray = new WebRtc_Word32[numOfChannels];
- _channelManager.GetChannelIds(channelsArray, numOfChannels);
- for (int i = 0; i < numOfChannels; i++)
- {
- voe::ScopedChannel sc(_channelManager, channelsArray[i]);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr)
- {
- channelPtr->ResetDeadOrAliveCounters();
- channelPtr->ResetRTCPStatistics();
- }
- }
- delete[] channelsArray;
- }
-
- return 0;
-}
-
-int VoECallReportImpl::GetEchoMetricSummary(EchoStatistics& stats)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetEchoMetricSummary()");
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- assert(_audioProcessingModulePtr != NULL);
-
- return (GetEchoMetricSummaryInternal(stats));
-}
-
-int VoECallReportImpl::GetEchoMetricSummaryInternal(EchoStatistics& stats)
-{
- // Retrieve echo metrics from the AudioProcessingModule
- int ret(0);
- bool mode(false);
- EchoCancellation::Metrics metrics;
-
- // Ensure that echo metrics is enabled
-
- mode =
- _audioProcessingModulePtr->echo_cancellation()->are_metrics_enabled();
- if (mode != false)
- {
- ret =
- _audioProcessingModulePtr->echo_cancellation()->GetMetrics(&metrics);
- if (ret != 0)
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- " AudioProcessingModule GetMetrics() => error");
- }
- }
- else
- {
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- " AudioProcessingModule echo metrics is not enabled");
- }
-
- if ((ret != 0) || (mode == false))
- {
- // Mark complete struct as invalid (-100 dB)
- WEBRTC_TRACE(kTraceWarning, kTraceVoice, VoEId(_instanceId, -1),
- " unable to retrieve echo metrics from the "
- "AudioProcessingModule");
- stats.erl.min = -100;
- stats.erl.max = -100;
- stats.erl.average = -100;
- stats.erle.min = -100;
- stats.erle.max = -100;
- stats.erle.average = -100;
- stats.rerl.min = -100;
- stats.rerl.max = -100;
- stats.rerl.average = -100;
- stats.a_nlp.min = -100;
- stats.a_nlp.max = -100;
- stats.a_nlp.average = -100;
- }
- else
- {
-
- // Deliver output results to user
- stats.erl.min = metrics.echo_return_loss.minimum;
- stats.erl.max = metrics.echo_return_loss.maximum;
- stats.erl.average = metrics.echo_return_loss.average;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- " erl: min=%d, max=%d, avg=%d", stats.erl.min,
- stats.erl.max, stats.erl.average);
-
- stats.erle.min = metrics.echo_return_loss_enhancement.minimum;
- stats.erle.max = metrics.echo_return_loss_enhancement.maximum;
- stats.erle.average = metrics.echo_return_loss_enhancement.average;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- " erle: min=%d, max=%d, avg=%d", stats.erle.min,
- stats.erle.max, stats.erle.average);
-
- stats.rerl.min = metrics.residual_echo_return_loss.minimum;
- stats.rerl.max = metrics.residual_echo_return_loss.maximum;
- stats.rerl.average = metrics.residual_echo_return_loss.average;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- " rerl: min=%d, max=%d, avg=%d", stats.rerl.min,
- stats.rerl.max, stats.rerl.average);
-
- stats.a_nlp.min = metrics.a_nlp.minimum;
- stats.a_nlp.max = metrics.a_nlp.maximum;
- stats.a_nlp.average = metrics.a_nlp.average;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- " a_nlp: min=%d, max=%d, avg=%d", stats.a_nlp.min,
- stats.a_nlp.max, stats.a_nlp.average);
- }
- return 0;
-}
-
-int VoECallReportImpl::GetRoundTripTimeSummary(int channel, StatVal& delaysMs)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetRoundTripTimeSummary()");
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRoundTripTimeSummary() failed to "
- "locate channel");
- return -1;
- }
-
- return channelPtr->GetRoundTripTimeSummary(delaysMs);
-}
-
-int VoECallReportImpl::GetDeadOrAliveSummary(int channel,
- int& numOfDeadDetections,
- int& numOfAliveDetections)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetDeadOrAliveSummary(channel=%d)", channel);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- return (GetDeadOrAliveSummaryInternal(channel, numOfDeadDetections,
- numOfAliveDetections));
-}
-
-int VoECallReportImpl::GetDeadOrAliveSummaryInternal(int channel,
- int& numOfDeadDetections,
- int& numOfAliveDetections)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetDeadOrAliveSummary(channel=%d)", channel);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRoundTripTimeSummary() failed to "
- "locate channel");
- return -1;
- }
-
- return channelPtr->GetDeadOrAliveCounters(numOfDeadDetections,
- numOfAliveDetections);
-}
-
-int VoECallReportImpl::WriteReportToFile(const char* fileNameUTF8)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "WriteReportToFile(fileNameUTF8=%s)", fileNameUTF8);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- if (NULL == fileNameUTF8)
- {
- _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "WriteReportToFile() invalid filename");
- return -1;
- }
-
- if (_file.Open())
- {
- _file.CloseFile();
- }
-
- // Open text file in write mode
- if (_file.OpenFile(fileNameUTF8, false, false, true) != 0)
- {
- _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
- "WriteReportToFile() unable to open the "
- "file");
- return -1;
- }
-
- // Summarize information and add it to the open file
- //
- _file.WriteText("WebRtc VoiceEngine Call Report\n");
- _file.WriteText("==============================\n");
- _file.WriteText("\nNetwork Packet Round Trip Time (RTT)\n");
- _file.WriteText("------------------------------------\n\n");
-
- WebRtc_Word32 numOfChannels = _channelManager.NumOfChannels();
- if (numOfChannels <= 0)
- {
- return 0;
- }
- WebRtc_Word32* channelsArray = new WebRtc_Word32[numOfChannels];
- _channelManager.GetChannelIds(channelsArray, numOfChannels);
- for (int ch = 0; ch < numOfChannels; ch++)
- {
- voe::ScopedChannel sc(_channelManager, channelsArray[ch]);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr)
- {
- StatVal delaysMs;
- _file.WriteText("channel %d:\n", ch);
- channelPtr->GetRoundTripTimeSummary(delaysMs);
- _file.WriteText(" min:%5d [ms]\n", delaysMs.min);
- _file.WriteText(" max:%5d [ms]\n", delaysMs.max);
- _file.WriteText(" avg:%5d [ms]\n", delaysMs.average);
- }
- }
-
- _file.WriteText("\nDead-or-Alive Connection Detections\n");
- _file.WriteText("------------------------------------\n\n");
-
- for (int ch = 0; ch < numOfChannels; ch++)
- {
- voe::ScopedChannel sc(_channelManager, channelsArray[ch]);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr)
- {
- int nDead(0);
- int nAlive(0);
- _file.WriteText("channel %d:\n", ch);
- GetDeadOrAliveSummary(ch, nDead, nAlive);
- _file.WriteText(" #dead :%6d\n", nDead);
- _file.WriteText(" #alive:%6d\n", nAlive);
- }
- }
-
- delete[] channelsArray;
-
- EchoStatistics echo;
- GetEchoMetricSummary(echo);
-
- _file.WriteText("\nEcho Metrics\n");
- _file.WriteText("------------\n\n");
-
- _file.WriteText("erl:\n");
- _file.WriteText(" min:%5d [dB]\n", echo.erl.min);
- _file.WriteText(" max:%5d [dB]\n", echo.erl.max);
- _file.WriteText(" avg:%5d [dB]\n", echo.erl.average);
- _file.WriteText("\nerle:\n");
- _file.WriteText(" min:%5d [dB]\n", echo.erle.min);
- _file.WriteText(" max:%5d [dB]\n", echo.erle.max);
- _file.WriteText(" avg:%5d [dB]\n", echo.erle.average);
- _file.WriteText("rerl:\n");
- _file.WriteText(" min:%5d [dB]\n", echo.rerl.min);
- _file.WriteText(" max:%5d [dB]\n", echo.rerl.max);
- _file.WriteText(" avg:%5d [dB]\n", echo.rerl.average);
- _file.WriteText("a_nlp:\n");
- _file.WriteText(" min:%5d [dB]\n", echo.a_nlp.min);
- _file.WriteText(" max:%5d [dB]\n", echo.a_nlp.max);
- _file.WriteText(" avg:%5d [dB]\n", echo.a_nlp.average);
-
- _file.WriteText("\n<END>");
-
- _file.Flush();
- _file.CloseFile();
-
- return 0;
-}
-
-#endif // WEBRTC_VOICE_ENGINE_CALL_REPORT_API
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_call_report_impl.h b/trunk/src/voice_engine/main/source/voe_call_report_impl.h
deleted file mode 100644
index f5852b4..0000000
--- a/trunk/src/voice_engine/main/source/voe_call_report_impl.h
+++ /dev/null
@@ -1,61 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_IMPL_H
-#define WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_IMPL_H
-
-#include "voe_call_report.h"
-
-#include "ref_count.h"
-#include "shared_data.h"
-
-
-namespace webrtc
-{
-class FileWrapper;
-
-class VoECallReportImpl: public virtual voe::SharedData,
- public VoECallReport,
- public voe::RefCount
-{
-public:
- virtual int Release();
-
- virtual int ResetCallReportStatistics(int channel);
-
- virtual int GetEchoMetricSummary(EchoStatistics& stats);
-
- virtual int GetRoundTripTimeSummary(int channel,
- StatVal& delaysMs);
-
- virtual int GetDeadOrAliveSummary(int channel, int& numOfDeadDetections,
- int& numOfAliveDetections);
-
- virtual int WriteReportToFile(const char* fileNameUTF8);
-
-protected:
- VoECallReportImpl();
- virtual ~VoECallReportImpl();
-
-private:
- int GetDeadOrAliveSummaryInternal(int channel,
- int& numOfDeadDetections,
- int& numOfAliveDetections);
-
- int GetEchoMetricSummaryInternal(EchoStatistics& stats);
-
- int GetSpeechAndNoiseSummaryInternal(LevelStatistics& stats);
-
- FileWrapper& _file;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_CALL_REPORT_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_codec_impl.cc b/trunk/src/voice_engine/main/source/voe_codec_impl.cc
deleted file mode 100644
index 6b308cc..0000000
--- a/trunk/src/voice_engine/main/source/voe_codec_impl.cc
+++ /dev/null
@@ -1,755 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voe_codec_impl.h"
-
-#include "audio_coding_module.h"
-#include "channel.h"
-#include "critical_section_wrapper.h"
-#include "trace.h"
-#include "voe_errors.h"
-#include "voice_engine_impl.h"
-
-namespace webrtc
-{
-
-VoECodec* VoECodec::GetInterface(VoiceEngine* voiceEngine)
-{
-#ifndef WEBRTC_VOICE_ENGINE_CODEC_API
- return NULL;
-#else
- if (NULL == voiceEngine)
- {
- return NULL;
- }
- VoiceEngineImpl* s =
- reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
- VoECodecImpl* d = s;
- (*d)++;
- return (d);
-#endif
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_CODEC_API
-
-VoECodecImpl::VoECodecImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
- "VoECodecImpl() - ctor");
-}
-
-VoECodecImpl::~VoECodecImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
- "~VoECodecImpl() - dtor");
-}
-
-int VoECodecImpl::Release()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "VoECodecImpl::Release()");
- (*this)--;
- int refCount = GetCount();
- if (refCount < 0)
- {
- Reset();
- _engineStatistics.SetLastError(VE_INTERFACE_NOT_FOUND,
- kTraceWarning);
- return (-1);
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "VoECodecImpl reference counter = %d", refCount);
- return (refCount);
-}
-
-int VoECodecImpl::NumOfCodecs()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "NumOfCodecs()");
-
- // Number of supported codecs in the ACM
- WebRtc_UWord8 nSupportedCodecs = AudioCodingModule::NumberOfCodecs();
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "NumOfCodecs() => %u", nSupportedCodecs);
- return (nSupportedCodecs);
-}
-
-int VoECodecImpl::GetCodec(int index, CodecInst& codec)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetCodec(index=%d, codec=?)", index);
- CodecInst acmCodec;
- if (AudioCodingModule::Codec(index, (CodecInst&) acmCodec)
- == -1)
- {
- _engineStatistics.SetLastError(VE_INVALID_LISTNR, kTraceError,
- "GetCodec() invalid index");
- return -1;
- }
- ACMToExternalCodecRepresentation(codec, acmCodec);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "GetCodec() => plname=%s, pacsize=%d, plfreq=%d, pltype=%d, "
- "channels=%d, rate=%d", codec.plname, codec.pacsize,
- codec.plfreq, codec.pltype, codec.channels, codec.rate);
- return 0;
-}
-
-int VoECodecImpl::SetSendCodec(int channel, const CodecInst& codec)
-{
- CodecInst copyCodec;
- ExternalToACMCodecRepresentation(copyCodec, codec);
-
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetSendCodec(channel=%d, codec)", channel);
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "codec: plname=%s, pacsize=%d, plfreq=%d, pltype=%d, "
- "channels=%d, rate=%d", codec.plname, codec.pacsize,
- codec.plfreq, codec.pltype, codec.channels, codec.rate);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- // External sanity checks performed outside the ACM
- if ((STR_CASE_CMP(copyCodec.plname, "L16") == 0) &&
- (copyCodec.pacsize >= 960))
- {
- _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "SetSendCodec() invalid L16 packet "
- "size");
- return -1;
- }
- if (!STR_CASE_CMP(copyCodec.plname, "CN")
- || !STR_CASE_CMP(copyCodec.plname, "TELEPHONE-EVENT")
- || !STR_CASE_CMP(copyCodec.plname, "RED"))
- {
- _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "SetSendCodec() invalid codec name");
- return -1;
- }
- if ((copyCodec.channels != 1) && (copyCodec.channels != 2))
- {
- _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "SetSendCodec() invalid number of "
- "channels");
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetSendCodec() failed to locate "
- "channel");
- return -1;
- }
- if (!AudioCodingModule::IsCodecValid(
- (CodecInst&) copyCodec))
- {
- _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "SetSendCodec() invalid codec");
- return -1;
- }
- if (channelPtr->SetSendCodec(copyCodec) != 0)
- {
- _engineStatistics.SetLastError(VE_CANNOT_SET_SEND_CODEC,
- kTraceError,
- "SetSendCodec() failed to set send "
- "codec");
- return -1;
- }
-
- // Need to check if we should change APM settings for mono/stereo.
- // We'll check all channels (sending or not), so we don't have to
- // check this again when starting/stopping sending.
-
- voe::ScopedChannel sc2(_channelManager);
- void* iterator(NULL);
- channelPtr = sc2.GetFirstChannel(iterator);
- int maxNumChannels = 1;
- while (channelPtr != NULL)
- {
- CodecInst tmpCdc;
- channelPtr->GetSendCodec(tmpCdc);
- if (tmpCdc.channels > maxNumChannels)
- maxNumChannels = tmpCdc.channels;
-
- channelPtr = sc2.GetNextChannel(iterator);
- }
-
- // Reuse the currently set number of capture channels. We need to wait
- // until receiving a frame to determine the true number.
- //
- // TODO(andrew): AudioProcessing will return an error if there are more
- // output than input channels (it doesn't want to produce fake channels).
- // This will happen with a stereo codec and a device which doesn't support
- // stereo. AudioCoding should probably do the faking; look into how to
- // handle this case properly.
- //
- // Check if the number of channels has changed to avoid an unnecessary
- // reset.
- // TODO(andrew): look at handling this logic in AudioProcessing.
- if (_audioProcessingModulePtr->num_output_channels() != maxNumChannels)
- {
- if (_audioProcessingModulePtr->set_num_channels(
- _audioProcessingModulePtr->num_input_channels(),
- maxNumChannels) != 0)
- {
- _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceError,
- "Init() failed to set APM channels for the send audio stream");
- return -1;
- }
- }
-
- return 0;
-}
-
-int VoECodecImpl::GetSendCodec(int channel, CodecInst& codec)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetSendCodec(channel=%d, codec=?)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetSendCodec() failed to locate "
- "channel");
- return -1;
- }
- CodecInst acmCodec;
- if (channelPtr->GetSendCodec(acmCodec) != 0)
- {
- _engineStatistics.SetLastError(VE_CANNOT_GET_SEND_CODEC, kTraceError,
- "GetSendCodec() failed to get send "
- "codec");
- return -1;
- }
- ACMToExternalCodecRepresentation(codec, acmCodec);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "GetSendCodec() => plname=%s, pacsize=%d, plfreq=%d, "
- "channels=%d, rate=%d", codec.plname, codec.pacsize,
- codec.plfreq, codec.channels, codec.rate);
- return 0;
-}
-
-int VoECodecImpl::GetRecCodec(int channel, CodecInst& codec)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetRecCodec(channel=%d, codec=?)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRecCodec() failed to locate "
- "channel");
- return -1;
- }
- CodecInst acmCodec;
- if (channelPtr->GetRecCodec(acmCodec) != 0)
- {
- _engineStatistics.SetLastError(VE_CANNOT_GET_REC_CODEC, kTraceError,
- "GetRecCodec() failed to get received "
- "codec");
- return -1;
- }
- ACMToExternalCodecRepresentation(codec, acmCodec);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "GetRecCodec() => plname=%s, pacsize=%d, plfreq=%d, "
- "channels=%d, rate=%d", codec.plname, codec.pacsize,
- codec.plfreq, codec.channels, codec.rate);
- return 0;
-}
-
-int VoECodecImpl::SetAMREncFormat(int channel, AmrMode mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetAMREncFormat(channel=%d, mode=%d)", channel, mode);
-#ifdef WEBRTC_CODEC_GSMAMR
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetAMREncFormat() failed to locate "
- "channel");
- return -1;
- }
- return channelPtr->SetAMREncFormat(mode);
-#else
- _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetAMREncFormat() AMR codec is not "
- "supported");
- return -1;
-#endif
-}
-
-int VoECodecImpl::SetAMRDecFormat(int channel, AmrMode mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetAMRDecFormat(channel=%i, mode=%i)", channel, mode);
-#ifdef WEBRTC_CODEC_GSMAMR
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetAMRDecFormat() failed to locate "
- "channel");
- return -1;
- }
- return channelPtr->SetAMRDecFormat(mode);
-#else
- _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetAMRDecFormat() AMR codec is not "
- "supported");
- return -1;
-#endif
-}
-
-int VoECodecImpl::SetAMRWbEncFormat(int channel, AmrMode mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetAMRWbEncFormat(channel=%d, mode=%d)", channel, mode);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-#ifdef WEBRTC_CODEC_GSMAMRWB
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetAMRWbEncFormat() failed to locate "
- "channel");
- return -1;
- }
- return channelPtr->SetAMRWbEncFormat(mode);
-#else
- _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetAMRWbEncFormat() AMR-wb codec is not "
- "supported");
- return -1;
-#endif
-}
-
-int VoECodecImpl::SetAMRWbDecFormat(int channel, AmrMode mode)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetAMRWbDecFormat(channel=%i, mode=%i)", channel, mode);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-#ifdef WEBRTC_CODEC_GSMAMRWB
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetAMRWbDecFormat() failed to locate "
- "channel");
- return -1;
- }
- return channelPtr->SetAMRWbDecFormat(mode);
-#else
- _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetAMRWbDecFormat() AMR-wb codec is not "
- "supported");
- return -1;
-#endif
-}
-
-int VoECodecImpl::SetRecPayloadType(int channel, const CodecInst& codec)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetRecPayloadType(channel=%d, codec)", channel);
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "codec: plname=%s, plfreq=%d, pltype=%d, channels=%u, "
- "pacsize=%d, rate=%d", codec.plname, codec.plfreq, codec.pltype,
- codec.channels, codec.pacsize, codec.rate);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRecPayloadType() failed to locate "
- "channel");
- return -1;
- }
- return channelPtr->SetRecPayloadType(codec);
-}
-
-int VoECodecImpl::GetRecPayloadType(int channel, CodecInst& codec)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetRecPayloadType(channel=%d, codec)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRecPayloadType() failed to locate "
- "channel");
- return -1;
- }
- return channelPtr->GetRecPayloadType(codec);
-}
-
-int VoECodecImpl::SetSendCNPayloadType(int channel, int type,
- PayloadFrequencies frequency)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetSendCNPayloadType(channel=%d, type=%d, frequency=%d)",
- channel, type, frequency);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (type < 96 || type > 127)
- {
- // Only allow dynamic range: 96 to 127
- _engineStatistics.SetLastError(VE_INVALID_PLTYPE, kTraceError,
- "SetSendCNPayloadType() invalid payload "
- "type");
- return -1;
- }
- if ((frequency != kFreq16000Hz) && (frequency != kFreq32000Hz))
- {
- // It is not possible to modify the payload type for CN/8000.
- // We only allow modification of the CN payload type for CN/16000
- // and CN/32000.
- _engineStatistics.SetLastError(VE_INVALID_PLFREQ, kTraceError,
- "SetSendCNPayloadType() invalid payload"
- " frequency");
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetSendCNPayloadType() failed to "
- "locate channel");
- return -1;
- }
- if (channelPtr->Sending())
- {
- _engineStatistics.SetLastError(VE_SENDING, kTraceError,
- "SetSendCNPayloadType unable so set "
- "payload type while sending");
- return -1;
- }
- return channelPtr->SetSendCNPayloadType(type, frequency);
-}
-
-int VoECodecImpl::SetISACInitTargetRate(int channel, int rateBps,
- bool useFixedFrameSize)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetISACInitTargetRate(channel=%d, rateBps=%d, "
- "useFixedFrameSize=%d)", channel, rateBps, useFixedFrameSize);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-#ifdef WEBRTC_CODEC_ISAC
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetISACInitTargetRate() failed to "
- "locate channel");
- return -1;
- }
- return channelPtr->SetISACInitTargetRate(rateBps, useFixedFrameSize);
-#else
- _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetISACInitTargetRate() iSAC codec is not "
- "supported");
- return -1;
-#endif
-}
-
-int VoECodecImpl::SetISACMaxRate(int channel, int rateBps)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetISACMaxRate(channel=%d, rateBps=%d)", channel, rateBps);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-#ifdef WEBRTC_CODEC_ISAC
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetISACMaxRate() failed to locate "
- "channel");
- return -1;
- }
- return channelPtr->SetISACMaxRate(rateBps);
-#else
- _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetISACMaxRate() iSAC codec is not "
- "supported");
- return -1;
-#endif
-}
-
-int VoECodecImpl::SetISACMaxPayloadSize(int channel, int sizeBytes)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetISACMaxPayloadSize(channel=%d, sizeBytes=%d)", channel,
- sizeBytes);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-#ifdef WEBRTC_CODEC_ISAC
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetISACMaxPayloadSize() failed to "
- "locate channel");
- return -1;
- }
- return channelPtr->SetISACMaxPayloadSize(sizeBytes);
-#else
- _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetISACMaxPayloadSize() iSAC codec is not "
- "supported");
- return -1;
-#endif
- return 0;
-}
-
-int VoECodecImpl::SetVADStatus(int channel, bool enable, VadModes mode,
- bool disableDTX)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetVADStatus(channel=%i, enable=%i, mode=%i, disableDTX=%i)",
- channel, enable, mode, disableDTX);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetVADStatus failed to locate channel");
- return -1;
- }
-
- ACMVADMode vadMode(VADNormal);
- switch (mode)
- {
- case kVadConventional:
- vadMode = VADNormal;
- break;
- case kVadAggressiveLow:
- vadMode = VADLowBitrate;
- break;
- case kVadAggressiveMid:
- vadMode = VADAggr;
- break;
- case kVadAggressiveHigh:
- vadMode = VADVeryAggr;
- break;
- }
- return channelPtr->SetVADStatus(enable, vadMode, disableDTX);
-}
-
-int VoECodecImpl::GetVADStatus(int channel, bool& enabled, VadModes& mode,
- bool& disabledDTX)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetVADStatus(channel=%i)", channel);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetVADStatus failed to locate channel");
- return -1;
- }
-
- ACMVADMode vadMode;
- int ret = channelPtr->GetVADStatus(enabled, vadMode, disabledDTX);
-
- if (ret != 0)
- {
- _engineStatistics.SetLastError(VE_INVALID_OPERATION, kTraceError,
- "GetVADStatus failed to get VAD mode");
- return -1;
- }
- switch (vadMode)
- {
- case VADNormal:
- mode = kVadConventional;
- break;
- case VADLowBitrate:
- mode = kVadAggressiveLow;
- break;
- case VADAggr:
- mode = kVadAggressiveMid;
- break;
- case VADVeryAggr:
- mode = kVadAggressiveHigh;
- break;
- }
-
- return 0;
-}
-
-void VoECodecImpl::ACMToExternalCodecRepresentation(CodecInst& toInst,
- const CodecInst& fromInst)
-{
- toInst = fromInst;
- if (STR_CASE_CMP(fromInst.plname,"SILK") == 0)
- {
- if (fromInst.plfreq == 12000)
- {
- if (fromInst.pacsize == 320)
- {
- toInst.pacsize = 240;
- }
- else if (fromInst.pacsize == 640)
- {
- toInst.pacsize = 480;
- }
- else if (fromInst.pacsize == 960)
- {
- toInst.pacsize = 720;
- }
- }
- else if (fromInst.plfreq == 24000)
- {
- if (fromInst.pacsize == 640)
- {
- toInst.pacsize = 480;
- }
- else if (fromInst.pacsize == 1280)
- {
- toInst.pacsize = 960;
- }
- else if (fromInst.pacsize == 1920)
- {
- toInst.pacsize = 1440;
- }
- }
- }
-}
-
-void VoECodecImpl::ExternalToACMCodecRepresentation(CodecInst& toInst,
- const CodecInst& fromInst)
-{
- toInst = fromInst;
- if (STR_CASE_CMP(fromInst.plname,"SILK") == 0)
- {
- if (fromInst.plfreq == 12000)
- {
- if (fromInst.pacsize == 240)
- {
- toInst.pacsize = 320;
- }
- else if (fromInst.pacsize == 480)
- {
- toInst.pacsize = 640;
- }
- else if (fromInst.pacsize == 720)
- {
- toInst.pacsize = 960;
- }
- }
- else if (fromInst.plfreq == 24000)
- {
- if (fromInst.pacsize == 480)
- {
- toInst.pacsize = 640;
- }
- else if (fromInst.pacsize == 960)
- {
- toInst.pacsize = 1280;
- }
- else if (fromInst.pacsize == 1440)
- {
- toInst.pacsize = 1920;
- }
- }
- }
-}
-
-#endif // WEBRTC_VOICE_ENGINE_CODEC_API
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_codec_impl.h b/trunk/src/voice_engine/main/source/voe_codec_impl.h
deleted file mode 100644
index fde1d41..0000000
--- a/trunk/src/voice_engine/main/source/voe_codec_impl.h
+++ /dev/null
@@ -1,92 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_CODEC_IMPL_H
-#define WEBRTC_VOICE_ENGINE_VOE_CODEC_IMPL_H
-
-#include "voe_codec.h"
-
-#include "ref_count.h"
-#include "shared_data.h"
-
-namespace webrtc
-{
-
-class VoECodecImpl: public virtual voe::SharedData,
- public VoECodec,
- public voe::RefCount
-{
-public:
- virtual int Release();
-
- virtual int NumOfCodecs();
-
- virtual int GetCodec(int index, CodecInst& codec);
-
- virtual int SetSendCodec(int channel, const CodecInst& codec);
-
- virtual int GetSendCodec(int channel, CodecInst& codec);
-
- virtual int GetRecCodec(int channel, CodecInst& codec);
-
- virtual int SetAMREncFormat(int channel,
- AmrMode mode = kRfc3267BwEfficient);
-
- virtual int SetAMRDecFormat(int channel,
- AmrMode mode = kRfc3267BwEfficient);
-
- virtual int SetAMRWbEncFormat(int channel,
- AmrMode mode = kRfc3267BwEfficient);
-
- virtual int SetAMRWbDecFormat(int channel,
- AmrMode mode = kRfc3267BwEfficient);
-
- virtual int SetSendCNPayloadType(
- int channel, int type,
- PayloadFrequencies frequency = kFreq16000Hz);
-
- virtual int SetRecPayloadType(int channel,
- const CodecInst& codec);
-
- virtual int GetRecPayloadType(int channel, CodecInst& codec);
-
- virtual int SetISACInitTargetRate(int channel,
- int rateBps,
- bool useFixedFrameSize = false);
-
- virtual int SetISACMaxRate(int channel, int rateBps);
-
- virtual int SetISACMaxPayloadSize(int channel, int sizeBytes);
-
- virtual int SetVADStatus(int channel,
- bool enable,
- VadModes mode = kVadConventional,
- bool disableDTX = false);
-
- virtual int GetVADStatus(int channel,
- bool& enabled,
- VadModes& mode,
- bool& disabledDTX);
-
-protected:
- VoECodecImpl();
- virtual ~VoECodecImpl();
-
-private:
- void ACMToExternalCodecRepresentation(CodecInst& toInst,
- const CodecInst& fromInst);
-
- void ExternalToACMCodecRepresentation(CodecInst& toInst,
- const CodecInst& fromInst);
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_CODEC_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_dtmf_impl.cc b/trunk/src/voice_engine/main/source/voe_dtmf_impl.cc
deleted file mode 100644
index 67f4c46..0000000
--- a/trunk/src/voice_engine/main/source/voe_dtmf_impl.cc
+++ /dev/null
@@ -1,473 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voe_dtmf_impl.h"
-
-#include "channel.h"
-#include "critical_section_wrapper.h"
-#include "output_mixer.h"
-#include "trace.h"
-#include "transmit_mixer.h"
-#include "voe_errors.h"
-#include "voice_engine_impl.h"
-
-namespace webrtc {
-
-VoEDtmf* VoEDtmf::GetInterface(VoiceEngine* voiceEngine)
-{
-#ifndef WEBRTC_VOICE_ENGINE_DTMF_API
- return NULL;
-#else
- if (NULL == voiceEngine)
- {
- return NULL;
- }
- VoiceEngineImpl* s =
- reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
- VoEDtmfImpl* d = s;
- ( *d)++;
- return (d);
-#endif
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_DTMF_API
-
-VoEDtmfImpl::VoEDtmfImpl() :
- _dtmfFeedback(true),
- _dtmfDirectFeedback(false)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1 ),
- "VoEDtmfImpl::VoEDtmfImpl() - ctor");
-}
-
-VoEDtmfImpl::~VoEDtmfImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "VoEDtmfImpl::~VoEDtmfImpl() - dtor");
-}
-
-int VoEDtmfImpl::Release()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "VoEDtmf::Release()");
- (*this)--;
- int refCount = GetCount();
- if (refCount < 0)
- {
- Reset(); // reset reference counter to zero => OK to delete VE
- _engineStatistics.SetLastError(
- VE_INTERFACE_NOT_FOUND, kTraceWarning);
- return (-1);
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "VoEDtmf reference counter = %d", refCount);
- return (refCount);
-}
-
-int VoEDtmfImpl::SendTelephoneEvent(int channel,
- int eventCode,
- bool outOfBand,
- int lengthMs,
- int attenuationDb)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SendTelephoneEvent(channel=%d, eventCode=%d, outOfBand=%d,"
- "length=%d, attenuationDb=%d)",
- channel, eventCode, (int)outOfBand, lengthMs, attenuationDb);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SendTelephoneEvent() failed to locate channel");
- return -1;
- }
- if (!channelPtr->Sending())
- {
- _engineStatistics.SetLastError(
- VE_NOT_SENDING, kTraceError,
- "SendTelephoneEvent() sending is not active");
- return -1;
- }
-
- // Sanity check
- const int maxEventCode = outOfBand ?
- static_cast<int>(kMaxTelephoneEventCode) :
- static_cast<int>(kMaxDtmfEventCode);
- const bool testFailed = ((eventCode < 0) ||
- (eventCode > maxEventCode) ||
- (lengthMs < kMinTelephoneEventDuration) ||
- (lengthMs > kMaxTelephoneEventDuration) ||
- (attenuationDb < kMinTelephoneEventAttenuation) ||
- (attenuationDb > kMaxTelephoneEventAttenuation));
- if (testFailed)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SendTelephoneEvent() invalid parameter(s)");
- return -1;
- }
-
- const bool isDtmf =
- (eventCode >= 0) && (eventCode <= kMaxDtmfEventCode);
- const bool playDtmfToneDirect =
- isDtmf && (_dtmfFeedback && _dtmfDirectFeedback);
-
- if (playDtmfToneDirect)
- {
- // Mute the microphone signal while playing back the tone directly.
- // This is to reduce the risk of introducing echo from the added output.
- _transmitMixerPtr->UpdateMuteMicrophoneTime(lengthMs);
-
- // Play out local feedback tone directly (same approach for both inband
- // and outband).
- // Reduce the length of the the tone with 80ms to reduce risk of echo.
- // For non-direct feedback, outband and inband cases are handled
- // differently.
- _outputMixerPtr->PlayDtmfTone(eventCode, lengthMs-80, attenuationDb);
- }
-
- if (outOfBand)
- {
- // The RTP/RTCP module will always deliver OnPlayTelephoneEvent when
- // an event is transmitted. It is up to the VoE to utilize it or not.
- // This flag ensures that feedback/playout is enabled; however, the
- // channel object must still parse out the Dtmf events (0-15) from
- // all possible events (0-255).
- const bool playDTFMEvent = (_dtmfFeedback && !_dtmfDirectFeedback);
-
- return channelPtr->SendTelephoneEventOutband(eventCode,
- lengthMs,
- attenuationDb,
- playDTFMEvent);
- }
- else
- {
- // For Dtmf tones, we want to ensure that inband tones are played out
- // in sync with the transmitted audio. This flag is utilized by the
- // channel object to determine if the queued Dtmf e vent shall also
- // be fed to the output mixer in the same step as input audio is
- // replaced by inband Dtmf tones.
- const bool playDTFMEvent =
- (isDtmf && _dtmfFeedback && !_dtmfDirectFeedback);
-
- return channelPtr->SendTelephoneEventInband(eventCode,
- lengthMs,
- attenuationDb,
- playDTFMEvent);
- }
-}
-
-int VoEDtmfImpl::SetSendTelephoneEventPayloadType(int channel,
- unsigned char type)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetSendTelephoneEventPayloadType(channel=%d, type=%u)",
- channel, type);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetSendTelephoneEventPayloadType() failed to locate channel");
- return -1;
- }
- return channelPtr->SetSendTelephoneEventPayloadType(type);
-}
-
-int VoEDtmfImpl::GetSendTelephoneEventPayloadType(int channel,
- unsigned char& type)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetSendTelephoneEventPayloadType(channel=%d)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetSendTelephoneEventPayloadType() failed to locate channel");
- return -1;
- }
- return channelPtr->GetSendTelephoneEventPayloadType(type);
-}
-
-int VoEDtmfImpl::PlayDtmfTone(int eventCode,
- int lengthMs,
- int attenuationDb)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "PlayDtmfTone(eventCode=%d, lengthMs=%d, attenuationDb=%d)",
- eventCode, lengthMs, attenuationDb);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (!_audioDevicePtr->Playing())
- {
- _engineStatistics.SetLastError(
- VE_NOT_PLAYING, kTraceError,
- "PlayDtmfTone() no channel is playing out");
- return -1;
- }
- if ((eventCode < kMinDtmfEventCode) ||
- (eventCode > kMaxDtmfEventCode) ||
- (lengthMs < kMinTelephoneEventDuration) ||
- (lengthMs > kMaxTelephoneEventDuration) ||
- (attenuationDb <kMinTelephoneEventAttenuation) ||
- (attenuationDb > kMaxTelephoneEventAttenuation))
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "PlayDtmfTone() invalid tone parameter(s)");
- return -1;
- }
- return _outputMixerPtr->PlayDtmfTone(eventCode, lengthMs, attenuationDb);
-}
-
-int VoEDtmfImpl::StartPlayingDtmfTone(int eventCode,
- int attenuationDb)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StartPlayingDtmfTone(eventCode=%d, attenuationDb=%d)",
- eventCode, attenuationDb);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (!_audioDevicePtr->Playing())
- {
- _engineStatistics.SetLastError(
- VE_NOT_PLAYING, kTraceError,
- "StartPlayingDtmfTone() no channel is playing out");
- return -1;
- }
- if ((eventCode < kMinDtmfEventCode) ||
- (eventCode > kMaxDtmfEventCode) ||
- (attenuationDb < kMinTelephoneEventAttenuation) ||
- (attenuationDb > kMaxTelephoneEventAttenuation))
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "StartPlayingDtmfTone() invalid tone parameter(s)");
- return -1;
- }
- return _outputMixerPtr->StartPlayingDtmfTone(eventCode, attenuationDb);
-}
-
-int VoEDtmfImpl::StopPlayingDtmfTone()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StopPlayingDtmfTone()");
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- return _outputMixerPtr->StopPlayingDtmfTone();
-}
-
-int VoEDtmfImpl::RegisterTelephoneEventDetection(
- int channel,
- TelephoneEventDetectionMethods detectionMethod,
- VoETelephoneEventObserver& observer)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "RegisterTelephoneEventDetection(channel=%d, detectionMethod=%d,"
- "observer=0x%x)", channel, detectionMethod, &observer);
-#ifdef WEBRTC_DTMF_DETECTION
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "RegisterTelephoneEventDetection() failed to locate channel");
- return -1;
- }
- return channelPtr->RegisterTelephoneEventDetection(detectionMethod,
- observer);
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetTelephoneEventDetectionStatus() Dtmf detection is not supported");
- return -1;
-#endif
-}
-
-int VoEDtmfImpl::DeRegisterTelephoneEventDetection(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "DeRegisterTelephoneEventDetection(channel=%d)", channel);
-#ifdef WEBRTC_DTMF_DETECTION
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "DeRegisterTelephoneEventDe tection() failed to locate channel");
- return -1;
- }
- return channelPtr->DeRegisterTelephoneEventDetection();
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "DeRegisterTelephoneEventDetection() Dtmf detection is not supported");
- return -1;
-#endif
-}
-
-
-int VoEDtmfImpl::GetTelephoneEventDetectionStatus(
- int channel,
- bool& enabled,
- TelephoneEventDetectionMethods& detectionMethod)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetTelephoneEventDetectionStatus(channel=%d)", channel);
-#ifdef WEBRTC_DTMF_DETECTION
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetTelephoneEventDetectionStatus() failed to locate channel");
- return -1;
- }
- return channelPtr->GetTelephoneEventDetectionStatus(enabled, detectionMethod);
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "GetTelephoneEventDetectionStatus() Dtmf detection is not supported");
- return -1;
-#endif
-}
-
-int VoEDtmfImpl::SetDtmfFeedbackStatus(bool enable, bool directFeedback)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetDtmfFeedbackStatus(enable=%d, directFeeback=%d)",
- (int)enable, (int)directFeedback);
-
- CriticalSectionScoped sc(*_apiCritPtr);
-
- _dtmfFeedback = enable;
- _dtmfDirectFeedback = directFeedback;
-
- return 0;
-}
-
-int VoEDtmfImpl::GetDtmfFeedbackStatus(bool& enabled, bool& directFeedback)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetDtmfFeedbackStatus()");
-
- CriticalSectionScoped sc(*_apiCritPtr);
-
- enabled = _dtmfFeedback;
- directFeedback = _dtmfDirectFeedback;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetDtmfFeedbackStatus() => enabled=%d, directFeedback=%d",
- enabled, directFeedback);
- return 0;
-}
-
-int VoEDtmfImpl::SetDtmfPlayoutStatus(int channel, bool enable)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetDtmfPlayoutStatus(channel=%d, enable=%d)",
- channel, enable);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetDtmfPlayoutStatus() failed to locate channel");
- return -1;
- }
- return channelPtr->SetDtmfPlayoutStatus(enable);
-}
-
-int VoEDtmfImpl::GetDtmfPlayoutStatus(int channel, bool& enabled)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetDtmfPlayoutStatus(channel=%d, enabled=?)", channel);
- IPHONE_NOT_SUPPORTED();
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetDtmfPlayoutStatus() failed to locate channel");
- return -1;
- }
- enabled = channelPtr->DtmfPlayoutStatus();
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetDtmfPlayoutStatus() => enabled=%d", enabled);
- return 0;
-}
-
-#endif // #ifdef WEBRTC_VOICE_ENGINE_DTMF_API
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_dtmf_impl.h b/trunk/src/voice_engine/main/source/voe_dtmf_impl.h
deleted file mode 100644
index 5b53969..0000000
--- a/trunk/src/voice_engine/main/source/voe_dtmf_impl.h
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_DTMF_IMPL_H
-#define WEBRTC_VOICE_ENGINE_VOE_DTMF_IMPL_H
-
-#include "voe_dtmf.h"
-
-#include "ref_count.h"
-#include "shared_data.h"
-
-namespace webrtc
-{
-
-class VoEDtmfImpl : public virtual voe::SharedData,
- public VoEDtmf,
- public voe::RefCount
-{
-public:
- virtual int Release();
-
- virtual int SendTelephoneEvent(
- int channel,
- int eventCode,
- bool outOfBand = true,
- int lengthMs = 160,
- int attenuationDb = 10);
-
- virtual int SetSendTelephoneEventPayloadType(int channel,
- unsigned char type);
-
- virtual int GetSendTelephoneEventPayloadType(int channel,
- unsigned char& type);
-
- virtual int SetDtmfFeedbackStatus(bool enable,
- bool directFeedback = false);
-
- virtual int GetDtmfFeedbackStatus(bool& enabled, bool& directFeedback);
-
- virtual int PlayDtmfTone(int eventCode,
- int lengthMs = 200,
- int attenuationDb = 10);
-
- virtual int StartPlayingDtmfTone(int eventCode,
- int attenuationDb = 10);
-
- virtual int StopPlayingDtmfTone();
-
- virtual int RegisterTelephoneEventDetection(
- int channel,
- TelephoneEventDetectionMethods detectionMethod,
- VoETelephoneEventObserver& observer);
-
- virtual int DeRegisterTelephoneEventDetection(int channel);
-
- virtual int GetTelephoneEventDetectionStatus(
- int channel,
- bool& enabled,
- TelephoneEventDetectionMethods& detectionMethod);
-
- virtual int SetDtmfPlayoutStatus(int channel, bool enable);
-
- virtual int GetDtmfPlayoutStatus(int channel, bool& enabled);
-
-protected:
- VoEDtmfImpl();
- virtual ~VoEDtmfImpl();
-
-private:
- bool _dtmfFeedback;
- bool _dtmfDirectFeedback;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_DTMF_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_encryption_impl.cc b/trunk/src/voice_engine/main/source/voe_encryption_impl.cc
deleted file mode 100644
index 5ba944b..0000000
--- a/trunk/src/voice_engine/main/source/voe_encryption_impl.cc
+++ /dev/null
@@ -1,275 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voe_encryption_impl.h"
-
-
-#include "channel.h"
-#include "critical_section_wrapper.h"
-#include "trace.h"
-#include "voe_errors.h"
-#include "voice_engine_impl.h"
-
-namespace webrtc {
-
-VoEEncryption* VoEEncryption::GetInterface(VoiceEngine* voiceEngine)
-{
-#ifndef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
- return NULL;
-#else
- if (NULL == voiceEngine)
- {
- return NULL;
- }
- VoiceEngineImpl* s =
- reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
- VoEEncryptionImpl* d = s;
- (*d)++;
- return (d);
-#endif
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
-
-VoEEncryptionImpl::VoEEncryptionImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "VoEEncryptionImpl::VoEEncryptionImpl() - ctor");
-}
-
-VoEEncryptionImpl::~VoEEncryptionImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "VoEEncryptionImpl::~VoEEncryptionImpl() - dtor");
-}
-
-int VoEEncryptionImpl::Release()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "VoEEncryption::Release()");
- (*this)--;
- int refCount = GetCount();
- if (refCount < 0)
- {
- // reset reference counter to zero => OK to delete VE
- Reset();
- _engineStatistics.SetLastError(VE_INTERFACE_NOT_FOUND,
- kTraceWarning);
- return (-1);
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "VoEEncryption reference counter = %d", refCount);
- return (refCount);
-}
-
-int VoEEncryptionImpl::EnableSRTPSend(
- int channel,
- CipherTypes cipherType,
- int cipherKeyLength,
- AuthenticationTypes authType,
- int authKeyLength,
- int authTagLength,
- SecurityLevels level,
- const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
- bool useForRTCP)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "EnableSRTPSend(channel=%i, cipherType=%i, cipherKeyLength=%i,"
- " authType=%i, authKeyLength=%i, authTagLength=%i, level=%i, "
- "key=?, useForRTCP=%d)",
- channel, cipherType, cipherKeyLength, authType,
- authKeyLength, authTagLength, level, useForRTCP);
-#ifdef WEBRTC_SRTP
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "EnableSRTPSend() failed to locate channel");
- return -1;
- }
- return channelPtr->EnableSRTPSend(cipherType,
- cipherKeyLength,
- authType,
- authKeyLength,
- authTagLength,
- level,
- key,
- useForRTCP);
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "EnableSRTPSend() SRTP is not supported");
- return -1;
-#endif
-}
-
-int VoEEncryptionImpl::DisableSRTPSend(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "DisableSRTPSend(channel=%i)",channel);
-#ifdef WEBRTC_SRTP
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "DisableSRTPSend() failed to locate channel");
- return -1;
- }
- return channelPtr->DisableSRTPSend();
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "DisableSRTPSend() SRTP is not supported");
- return -1;
-#endif
-}
-
-int VoEEncryptionImpl::EnableSRTPReceive(
- int channel,
- CipherTypes cipherType,
- int cipherKeyLength,
- AuthenticationTypes authType,
- int authKeyLength,
- int authTagLength,
- SecurityLevels level,
- const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
- bool useForRTCP)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "EnableSRTPReceive(channel=%i, cipherType=%i, "
- "cipherKeyLength=%i, authType=%i, authKeyLength=%i, "
- "authTagLength=%i, level=%i, key=?, useForRTCP=%d)",
- channel, cipherType, cipherKeyLength, authType,
- authKeyLength, authTagLength, level, useForRTCP);
-#ifdef WEBRTC_SRTP
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "EnableSRTPReceive() failed to locate channel");
- return -1;
- }
- return channelPtr->EnableSRTPReceive(cipherType,
- cipherKeyLength,
- authType,
- authKeyLength,
- authTagLength,
- level,
- key,
- useForRTCP);
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "EnableSRTPReceive() SRTP is not supported");
- return -1;
-#endif
-}
-
-int VoEEncryptionImpl::DisableSRTPReceive(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "DisableSRTPReceive(channel=%i)", channel);
-#ifdef WEBRTC_SRTP
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "DisableSRTPReceive() failed to locate channel");
- return -1;
- }
- return channelPtr->DisableSRTPReceive();
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceError,
- "DisableSRTPReceive() SRTP is not supported");
- return -1;
-#endif
-}
-
-int VoEEncryptionImpl::RegisterExternalEncryption(int channel,
- Encryption& encryption)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "RegisterExternalEncryption(channel=%d, encryption=0x%x)",
- channel, &encryption);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "RegisterExternalEncryption() failed to locate channel");
- return -1;
- }
- return channelPtr->RegisterExternalEncryption(encryption);
-}
-
-int VoEEncryptionImpl::DeRegisterExternalEncryption(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "DeRegisterExternalEncryption(channel=%d)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "DeRegisterExternalEncryption() failed to locate channel");
- return -1;
- }
- return channelPtr->DeRegisterExternalEncryption();
-}
-
-#endif // #ifdef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
-
-// EOF
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_encryption_impl.h b/trunk/src/voice_engine/main/source/voe_encryption_impl.h
deleted file mode 100644
index 050dd88..0000000
--- a/trunk/src/voice_engine/main/source/voe_encryption_impl.h
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_ENCRYPTION_IMPL_H
-#define WEBRTC_VOICE_ENGINE_VOE_ENCRYPTION_IMPL_H
-
-#include "voe_encryption.h"
-
-#include "ref_count.h"
-#include "shared_data.h"
-
-namespace webrtc {
-
-class VoEEncryptionImpl : public virtual voe::SharedData,
- public VoEEncryption,
- public voe::RefCount
-{
-public:
-
- virtual int Release();
-
- // SRTP
- virtual int EnableSRTPSend(
- int channel,
- CipherTypes cipherType,
- int cipherKeyLength,
- AuthenticationTypes authType,
- int authKeyLength,
- int authTagLength,
- SecurityLevels level,
- const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
- bool useForRTCP = false);
-
- virtual int DisableSRTPSend(int channel);
-
- virtual int EnableSRTPReceive(
- int channel,
- CipherTypes cipherType,
- int cipherKeyLength,
- AuthenticationTypes authType,
- int authKeyLength,
- int authTagLength,
- SecurityLevels level,
- const unsigned char key[kVoiceEngineMaxSrtpKeyLength],
- bool useForRTCP = false);
-
- virtual int DisableSRTPReceive(int channel);
-
- // External encryption
- virtual int RegisterExternalEncryption(
- int channel,
- Encryption& encryption);
-
- virtual int DeRegisterExternalEncryption(int channel);
-
-protected:
- VoEEncryptionImpl();
- virtual ~VoEEncryptionImpl();
-};
-
-} // namespace webrtc
-
-#endif // #ifndef WEBRTC_VOICE_ENGINE_VOE_ENCRYPTION_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_external_media_impl.cc b/trunk/src/voice_engine/main/source/voe_external_media_impl.cc
deleted file mode 100644
index 3fc3239..0000000
--- a/trunk/src/voice_engine/main/source/voe_external_media_impl.cc
+++ /dev/null
@@ -1,397 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voe_external_media_impl.h"
-
-#include "channel.h"
-#include "critical_section_wrapper.h"
-#include "output_mixer.h"
-#include "trace.h"
-#include "transmit_mixer.h"
-#include "voice_engine_impl.h"
-#include "voe_errors.h"
-
-namespace webrtc {
-
-VoEExternalMedia* VoEExternalMedia::GetInterface(VoiceEngine* voiceEngine)
-{
-#ifndef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
- return NULL;
-#else
- if (NULL == voiceEngine)
- {
- return NULL;
- }
- VoiceEngineImpl* s = reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
- VoEExternalMediaImpl* d = s;
- (*d)++;
- return (d);
-#endif
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
-
-VoEExternalMediaImpl::VoEExternalMediaImpl()
- : playout_delay_ms_(0)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "VoEExternalMediaImpl() - ctor");
-}
-
-VoEExternalMediaImpl::~VoEExternalMediaImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "~VoEExternalMediaImpl() - dtor");
-}
-
-int VoEExternalMediaImpl::Release()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "VoEExternalMedia::Release()");
- (*this)--;
- int refCount = GetCount();
- if (refCount < 0)
- {
- Reset();
- _engineStatistics.SetLastError(VE_INTERFACE_NOT_FOUND,
- kTraceWarning);
- return (-1);
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "VoEExternalMedia reference counter = %d", refCount);
- return (refCount);
-}
-
-int VoEExternalMediaImpl::RegisterExternalMediaProcessing(
- int channel,
- ProcessingTypes type,
- VoEMediaProcess& processObject)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "RegisterExternalMediaProcessing(channel=%d, type=%d, "
- "processObject=0x%x)", channel, type, &processObject);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- switch (type)
- {
- case kPlaybackPerChannel:
- case kRecordingPerChannel:
- {
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "RegisterExternalMediaProcessing() "
- "failed to locate channel");
- return -1;
- }
- return channelPtr->RegisterExternalMediaProcessing(type,
- processObject);
- }
- case kPlaybackAllChannelsMixed:
- {
- return _outputMixerPtr->RegisterExternalMediaProcessing(
- processObject);
- }
- case kRecordingAllChannelsMixed:
- {
- return _transmitMixerPtr->RegisterExternalMediaProcessing(
- processObject);
- }
- }
- return -1;
-}
-
-int VoEExternalMediaImpl::DeRegisterExternalMediaProcessing(
- int channel,
- ProcessingTypes type)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "DeRegisterExternalMediaProcessing(channel=%d)", channel);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- switch (type)
- {
- case kPlaybackPerChannel:
- case kRecordingPerChannel:
- {
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "RegisterExternalMediaProcessing() "
- "failed to locate channel");
- return -1;
- }
- return channelPtr->DeRegisterExternalMediaProcessing(type);
- }
- case kPlaybackAllChannelsMixed:
- {
- return _outputMixerPtr->DeRegisterExternalMediaProcessing();
- }
- case kRecordingAllChannelsMixed:
- {
- return _transmitMixerPtr->DeRegisterExternalMediaProcessing();
- }
- }
- return -1;
-}
-
-int VoEExternalMediaImpl::SetExternalRecordingStatus(bool enable)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetExternalRecordingStatus(enable=%d)", enable);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
- if (_audioDevicePtr->Recording())
- {
- _engineStatistics.SetLastError(
- VE_ALREADY_SENDING,
- kTraceError,
- "SetExternalRecordingStatus() cannot set state while sending");
- return -1;
- }
- _externalRecording = enable;
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED,
- kTraceError,
- "SetExternalRecordingStatus() external recording is not supported");
- return -1;
-#endif
-}
-
-int VoEExternalMediaImpl::ExternalRecordingInsertData(
- const WebRtc_Word16 speechData10ms[],
- int lengthSamples,
- int samplingFreqHz,
- int current_delay_ms)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "ExternalRecordingInsertData(speechData10ms=0x%x,"
- " lengthSamples=%u, samplingFreqHz=%d, current_delay_ms=%d)",
- &speechData10ms[0], lengthSamples, samplingFreqHz,
- current_delay_ms);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
-#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (!_externalRecording)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_OPERATION,
- kTraceError,
- "ExternalRecordingInsertData() external recording is not enabled");
- return -1;
- }
- if (NumOfSendingChannels() == 0)
- {
- _engineStatistics.SetLastError(
- VE_ALREADY_SENDING,
- kTraceError,
- "SetExternalRecordingStatus() no channel is sending");
- return -1;
- }
- if ((16000 != samplingFreqHz) && (32000 != samplingFreqHz) &&
- (48000 != samplingFreqHz) && (44000 != samplingFreqHz))
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT,
- kTraceError,
- "SetExternalRecordingStatus() invalid sample rate");
- return -1;
- }
- if ((0 == lengthSamples) ||
- ((lengthSamples % (samplingFreqHz / 100)) != 0))
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT,
- kTraceError,
- "SetExternalRecordingStatus() invalid buffer size");
- return -1;
- }
- if (current_delay_ms < 0)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT,
- kTraceError,
- "SetExternalRecordingStatus() invalid delay)");
- return -1;
- }
-
- WebRtc_UWord16 blockSize = samplingFreqHz / 100;
- WebRtc_UWord32 nBlocks = lengthSamples / blockSize;
- WebRtc_Word16 totalDelayMS = 0;
- WebRtc_UWord16 playoutDelayMS = 0;
-
- for (WebRtc_UWord32 i = 0; i < nBlocks; i++)
- {
- if (!_externalPlayout)
- {
- // Use real playout delay if external playout is not enabled.
- if (_audioDevicePtr->PlayoutDelay(&playoutDelayMS) != 0) {
- _engineStatistics.SetLastError(
- VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
- "PlayoutDelay() unable to get the playout delay");
- }
- totalDelayMS = current_delay_ms + playoutDelayMS;
- }
- else
- {
- // Use stored delay value given the last call
- // to ExternalPlayoutGetData.
- totalDelayMS = current_delay_ms + playout_delay_ms_;
- // Compensate for block sizes larger than 10ms
- totalDelayMS -= (WebRtc_Word16)(i*10);
- if (totalDelayMS < 0)
- totalDelayMS = 0;
- }
- _transmitMixerPtr->PrepareDemux(
- (const WebRtc_Word8*)(&speechData10ms[i*blockSize]),
- blockSize,
- 1,
- samplingFreqHz,
- totalDelayMS,
- 0,
- 0);
-
- _transmitMixerPtr->DemuxAndMix();
- _transmitMixerPtr->EncodeAndSend();
- }
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED,
- kTraceError,
- "ExternalRecordingInsertData() external recording is not supported");
- return -1;
-#endif
-}
-
-int VoEExternalMediaImpl::SetExternalPlayoutStatus(bool enable)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetExternalPlayoutStatus(enable=%d)", enable);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
- if (_audioDevicePtr->Playing())
- {
- _engineStatistics.SetLastError(
- VE_ALREADY_SENDING,
- kTraceError,
- "SetExternalPlayoutStatus() cannot set state while playing");
- return -1;
- }
- _externalPlayout = enable;
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED,
- kTraceError,
- "SetExternalPlayoutStatus() external playout is not supported");
- return -1;
-#endif
-}
-
-int VoEExternalMediaImpl::ExternalPlayoutGetData(
- WebRtc_Word16 speechData10ms[],
- int samplingFreqHz,
- int current_delay_ms,
- int& lengthSamples)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId,-1),
- "ExternalPlayoutGetData(speechData10ms=0x%x, samplingFreqHz=%d"
- ", current_delay_ms=%d)", &speechData10ms[0], samplingFreqHz,
- current_delay_ms);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (!_externalPlayout)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_OPERATION,
- kTraceError,
- "ExternalPlayoutGetData() external playout is not enabled");
- return -1;
- }
- if ((16000 != samplingFreqHz) && (32000 != samplingFreqHz) &&
- (48000 != samplingFreqHz) && (44000 != samplingFreqHz))
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT,
- kTraceError,
- "ExternalPlayoutGetData() invalid sample rate");
- return -1;
- }
- if (current_delay_ms < 0)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT,
- kTraceError,
- "ExternalPlayoutGetData() invalid delay)");
- return -1;
- }
-
- AudioFrame audioFrame;
-
- // Retrieve mixed output at the specified rate
- _outputMixerPtr->MixActiveChannels();
- _outputMixerPtr->DoOperationsOnCombinedSignal();
- _outputMixerPtr->GetMixedAudio(samplingFreqHz, 1, audioFrame);
-
- // Deliver audio (PCM) samples to the external sink
- memcpy(speechData10ms,
- audioFrame._payloadData,
- sizeof(WebRtc_Word16)*(audioFrame._payloadDataLengthInSamples));
- lengthSamples = audioFrame._payloadDataLengthInSamples;
-
- // Store current playout delay (to be used by ExternalRecordingInsertData).
- playout_delay_ms_ = current_delay_ms;
-
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED,
- kTraceError,
- "ExternalPlayoutGetData() external playout is not supported");
- return -1;
-#endif
-}
-
-#endif // WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_external_media_impl.h b/trunk/src/voice_engine/main/source/voe_external_media_impl.h
deleted file mode 100644
index fa1ff8a..0000000
--- a/trunk/src/voice_engine/main/source/voe_external_media_impl.h
+++ /dev/null
@@ -1,62 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_IMPL_H
-#define WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_IMPL_H
-
-#include "voe_external_media.h"
-
-#include "ref_count.h"
-#include "shared_data.h"
-
-namespace webrtc {
-
-class VoEExternalMediaImpl : public virtual voe::SharedData,
- public VoEExternalMedia,
- public voe::RefCount
-{
-public:
- virtual int Release();
-
- virtual int RegisterExternalMediaProcessing(
- int channel,
- ProcessingTypes type,
- VoEMediaProcess& processObject);
-
- virtual int DeRegisterExternalMediaProcessing(
- int channel,
- ProcessingTypes type);
-
- virtual int SetExternalRecordingStatus(bool enable);
-
- virtual int SetExternalPlayoutStatus(bool enable);
-
- virtual int ExternalRecordingInsertData(
- const WebRtc_Word16 speechData10ms[],
- int lengthSamples,
- int samplingFreqHz,
- int current_delay_ms);
-
- virtual int ExternalPlayoutGetData(WebRtc_Word16 speechData10ms[],
- int samplingFreqHz,
- int current_delay_ms,
- int& lengthSamples);
-
-protected:
- VoEExternalMediaImpl();
- virtual ~VoEExternalMediaImpl();
-
-private:
- int playout_delay_ms_;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_EXTERNAL_MEDIA_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_file_impl.cc b/trunk/src/voice_engine/main/source/voe_file_impl.cc
deleted file mode 100644
index cd3193f..0000000
--- a/trunk/src/voice_engine/main/source/voe_file_impl.cc
+++ /dev/null
@@ -1,1423 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voe_file_impl.h"
-
-#include "channel.h"
-#include "critical_section_wrapper.h"
-#include "file_wrapper.h"
-#include "media_file.h"
-#include "output_mixer.h"
-#include "trace.h"
-#include "transmit_mixer.h"
-#include "voe_errors.h"
-#include "voice_engine_impl.h"
-
-namespace webrtc {
-
-VoEFile* VoEFile::GetInterface(VoiceEngine* voiceEngine)
-{
-#ifndef WEBRTC_VOICE_ENGINE_FILE_API
- return NULL;
-#else
- if (NULL == voiceEngine)
- {
- return NULL;
- }
- VoiceEngineImpl* s =
- reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
- VoEFileImpl* d = s;
- (*d)++;
- return (d);
-#endif
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_FILE_API
-
-VoEFileImpl::VoEFileImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "VoEFileImpl::VoEFileImpl() - ctor");
-}
-
-VoEFileImpl::~VoEFileImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "VoEFileImpl::~VoEFileImpl() - dtor");
-}
-
-int VoEFileImpl::Release()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "VoEFile::Release()");
- (*this)--;
- int refCount = GetCount();
- if (refCount < 0)
- {
- Reset();
- _engineStatistics.SetLastError(VE_INTERFACE_NOT_FOUND,
- kTraceWarning);
- return (-1);
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "VoEFile reference counter = %d", refCount);
- return (refCount);
-}
-
-int VoEFileImpl::StartPlayingFileLocally(
- int channel,
- const char fileNameUTF8[1024],
- bool loop, FileFormats format,
- float volumeScaling,
- int startPointMs,
- int stopPointMs)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StartPlayingFileLocally(channel=%d, fileNameUTF8[]=%s, "
- "loop=%d, format=%d, volumeScaling=%5.3f, startPointMs=%d,"
- " stopPointMs=%d)",
- channel, fileNameUTF8, loop, format, volumeScaling,
- startPointMs, stopPointMs);
- assert(1024 == FileWrapper::kMaxFileNameSize);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "StartPlayingFileLocally() failed to locate channel");
- return -1;
- }
-
- return channelPtr->StartPlayingFileLocally(fileNameUTF8,
- loop,
- format,
- startPointMs,
- volumeScaling,
- stopPointMs,
- NULL);
-}
-
-int VoEFileImpl::StartPlayingFileLocally(int channel,
- InStream* stream,
- FileFormats format,
- float volumeScaling,
- int startPointMs,
- int stopPointMs)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StartPlayingFileLocally(channel=%d, stream, format=%d, "
- "volumeScaling=%5.3f, startPointMs=%d, stopPointMs=%d)",
- channel, format, volumeScaling, startPointMs, stopPointMs);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "StartPlayingFileLocally() failed to locate channel");
- return -1;
- }
-
- return channelPtr->StartPlayingFileLocally(stream,
- format,
- startPointMs,
- volumeScaling,
- stopPointMs,
- NULL);
-}
-
-int VoEFileImpl::StopPlayingFileLocally(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StopPlayingFileLocally()");
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "StopPlayingFileLocally() failed to locate channel");
- return -1;
- }
- return channelPtr->StopPlayingFileLocally();
-}
-
-int VoEFileImpl::IsPlayingFileLocally(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "IsPlayingFileLocally(channel=%d)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "StopPlayingFileLocally() failed to locate channel");
- return -1;
- }
- return channelPtr->IsPlayingFileLocally();
-}
-
-int VoEFileImpl::ScaleLocalFilePlayout(int channel, float scale)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "ScaleLocalFilePlayout(channel=%d, scale=%5.3f)",
- channel, scale);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "StopPlayingFileLocally() failed to locate channel");
- return -1;
- }
- return channelPtr->ScaleLocalFilePlayout(scale);
-}
-
-int VoEFileImpl::StartPlayingFileAsMicrophone(int channel,
- const char fileNameUTF8[1024],
- bool loop,
- bool mixWithMicrophone,
- FileFormats format,
- float volumeScaling)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StartPlayingFileAsMicrophone(channel=%d, fileNameUTF8=%s, "
- "loop=%d, mixWithMicrophone=%d, format=%d, "
- "volumeScaling=%5.3f)",
- channel, fileNameUTF8, loop, mixWithMicrophone, format,
- volumeScaling);
- assert(1024 == FileWrapper::kMaxFileNameSize);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- const WebRtc_UWord32 startPointMs(0);
- const WebRtc_UWord32 stopPointMs(0);
-
- if (channel == -1)
- {
- int res = _transmitMixerPtr->StartPlayingFileAsMicrophone(
- fileNameUTF8,
- loop,
- format,
- startPointMs,
- volumeScaling,
- stopPointMs,
- NULL);
- if (res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "StartPlayingFileAsMicrophone() failed to start"
- " playing file");
- return(-1);
- }
- else
- {
- _transmitMixerPtr->SetMixWithMicStatus(mixWithMicrophone);
- return(0);
- }
- }
- else
- {
- // Add file after demultiplexing <=> affects one channel only
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "StartPlayingFileAsMicrophone() failed to locate channel");
- return -1;
- }
-
- int res = channelPtr->StartPlayingFileAsMicrophone(fileNameUTF8,
- loop,
- format,
- startPointMs,
- volumeScaling,
- stopPointMs,
- NULL);
- if (res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "StartPlayingFileAsMicrophone() failed to start "
- "playing file");
- return -1;
- }
- else
- {
- channelPtr->SetMixWithMicStatus(mixWithMicrophone);
- return 0;
- }
- }
-}
-
-int VoEFileImpl::StartPlayingFileAsMicrophone(int channel,
- InStream* stream,
- bool mixWithMicrophone,
- FileFormats format,
- float volumeScaling)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StartPlayingFileAsMicrophone(channel=%d, stream,"
- " mixWithMicrophone=%d, format=%d, volumeScaling=%5.3f)",
- channel, mixWithMicrophone, format, volumeScaling);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- const WebRtc_UWord32 startPointMs(0);
- const WebRtc_UWord32 stopPointMs(0);
-
- if (channel == -1)
- {
- int res = _transmitMixerPtr->StartPlayingFileAsMicrophone(
- stream,
- format,
- startPointMs,
- volumeScaling,
- stopPointMs,
- NULL);
- if (res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "StartPlayingFileAsMicrophone() failed to start"
- " playing stream");
- return(-1);
- }
- else
- {
- _transmitMixerPtr->SetMixWithMicStatus(mixWithMicrophone);
- return(0);
- }
- }
- else
- {
- // Add file after demultiplexing <=> affects one channel only
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "StartPlayingFileAsMicrophone() failed to locate channel");
- return -1;
- }
-
- int res = channelPtr->StartPlayingFileAsMicrophone(
- stream, format, startPointMs, volumeScaling, stopPointMs, NULL);
- if (res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "StartPlayingFileAsMicrophone() failed to start"
- " playing stream");
- return -1;
- }
- else
- {
- channelPtr->SetMixWithMicStatus(mixWithMicrophone);
- return 0;
- }
- }
-}
-
-int VoEFileImpl::StopPlayingFileAsMicrophone(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StopPlayingFileAsMicrophone(channel=%d)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (channel == -1)
- {
- // Stop adding file before demultiplexing <=> affects all channels
- return _transmitMixerPtr->StopPlayingFileAsMicrophone();
- }
- else
- {
- // Stop adding file after demultiplexing <=> affects one channel only
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "StopPlayingFileAsMicrophone() failed to locate channel");
- return -1;
- }
- return channelPtr->StopPlayingFileAsMicrophone();
- }
-}
-
-int VoEFileImpl::IsPlayingFileAsMicrophone(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "IsPlayingFileAsMicrophone(channel=%d)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (channel == -1)
- {
- return _transmitMixerPtr->IsPlayingFileAsMicrophone();
- }
- else
- {
- // Stop adding file after demultiplexing <=> affects one channel only
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "IsPlayingFileAsMicrophone() failed to locate channel");
- return -1;
- }
- return channelPtr->IsPlayingFileAsMicrophone();
- }
-}
-
-int VoEFileImpl::ScaleFileAsMicrophonePlayout(int channel, float scale)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "ScaleFileAsMicrophonePlayout(channel=%d, scale=%5.3f)",
- channel, scale);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (channel == -1)
- {
- return _transmitMixerPtr->ScaleFileAsMicrophonePlayout(scale);
- }
- else
- {
- // Stop adding file after demultiplexing <=> affects one channel only
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "IsPlayingFileAsMicrophone() failed to locate channel");
- return -1;
- }
- return channelPtr->ScaleFileAsMicrophonePlayout(scale);
- }
-}
-
-int VoEFileImpl::StartRecordingPlayout(
- int channel, const char* fileNameUTF8, CodecInst* compression,
- int maxSizeBytes)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StartRecordingPlayout(channel=%d, fileNameUTF8=%s, "
- "compression, maxSizeBytes=%d)",
- channel, fileNameUTF8, maxSizeBytes);
- assert(1024 == FileWrapper::kMaxFileNameSize);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (channel == -1)
- {
- _outputMixerPtr->StartRecordingPlayout(fileNameUTF8, compression);
- return 0;
- }
- else
- {
- // Add file after demultiplexing <=> affects one channel only
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "StartRecordingPlayout() failed to locate channel");
- return -1;
- }
- return channelPtr->StartRecordingPlayout(fileNameUTF8, compression);
- }
-}
-
-int VoEFileImpl::StartRecordingPlayout(
- int channel, OutStream* stream, CodecInst* compression)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StartRecordingPlayout(channel=%d, stream, compression)",
- channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (channel == -1)
- {
- return _outputMixerPtr->StartRecordingPlayout(stream, compression);
- }
- else
- {
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "StartRecordingPlayout() failed to locate channel");
- return -1;
- }
- return channelPtr->StartRecordingPlayout(stream, compression);
- }
-}
-
-int VoEFileImpl::StopRecordingPlayout(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StopRecordingPlayout(channel=%d)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (channel == -1)
- {
- return _outputMixerPtr->StopRecordingPlayout();
- }
- else
- {
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "StopRecordingPlayout() failed to locate channel");
- return -1;
- }
- return channelPtr->StopRecordingPlayout();
- }
-}
-
-int VoEFileImpl::StartRecordingMicrophone(
- const char* fileNameUTF8, CodecInst* compression, int maxSizeBytes)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StartRecordingMicrophone(fileNameUTF8=%s, compression, "
- "maxSizeBytes=%d)", fileNameUTF8, maxSizeBytes);
- assert(1024 == FileWrapper::kMaxFileNameSize);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (_transmitMixerPtr->StartRecordingMicrophone(fileNameUTF8, compression))
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "StartRecordingMicrophone() failed to start recording");
- return -1;
- }
- if (_audioDevicePtr->Recording())
- {
- return 0;
- }
- if (!_externalRecording)
- {
- if (_audioDevicePtr->InitRecording() != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "StartRecordingMicrophone() failed to initialize"
- " recording");
- return -1;
- }
- if (_audioDevicePtr->StartRecording() != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "StartRecordingMicrophone() failed to start recording");
- return -1;
- }
- }
- return 0;
-}
-
-int VoEFileImpl::StartRecordingMicrophone(
- OutStream* stream, CodecInst* compression)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StartRecordingMicrophone(stream, compression)");
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (_transmitMixerPtr->StartRecordingMicrophone(stream, compression) == -1)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "StartRecordingMicrophone() failed to start recording");
- return -1;
- }
- if (_audioDevicePtr->Recording())
- {
- return 0;
- }
- if (!_externalRecording)
- {
- if (_audioDevicePtr->InitRecording() != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "StartRecordingMicrophone() failed to initialize "
- "recording");
- return -1;
- }
- if (_audioDevicePtr->StartRecording() != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "StartRecordingMicrophone() failed to start"
- " recording");
- return -1;
- }
- }
- return 0;
-}
-
-int VoEFileImpl::StopRecordingMicrophone()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StopRecordingMicrophone()");
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if ((NumOfSendingChannels() == 0)&&_audioDevicePtr->Recording())
- {
- // Stop audio-device recording if no channel is recording
- if (_audioDevicePtr->StopRecording() != 0)
- {
- _engineStatistics.SetLastError(
- VE_CANNOT_STOP_RECORDING, kTraceError,
- "StopRecordingMicrophone() failed to stop recording");
- return -1;
- }
- }
- return _transmitMixerPtr->StopRecordingMicrophone();
-}
-
-int VoEFileImpl::ConvertPCMToWAV(const char* fileNameInUTF8,
- const char* fileNameOutUTF8)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertPCMToWAV(fileNameInUTF8=%s, fileNameOutUTF8=%s)",
- fileNameInUTF8, fileNameOutUTF8);
-
- // Create file player object
- FilePlayer& playerObj(*FilePlayer::CreateFilePlayer(
- -1,
- kFileFormatPcm16kHzFile));
-
- int res=playerObj.StartPlayingFile(fileNameInUTF8,false,0,1.0,0,0, NULL);
- if (res)
- {
- _engineStatistics.SetLastError(
- VE_BAD_FILE, kTraceError,
- "ConvertPCMToWAV failed to create player object");
- playerObj.StopPlayingFile();
- FilePlayer::DestroyFilePlayer(&playerObj);
- return -1;
- }
-
- // Create file recorder object
- FileRecorder& recObj(*FileRecorder::CreateFileRecorder(
- -1, kFileFormatWavFile));
-
- CodecInst codecInst;
- strncpy(codecInst.plname,"L16",32);
- codecInst.channels = 1;
- codecInst.rate = 256000;
- codecInst.plfreq = 16000;
- codecInst.pltype = 94;
- codecInst.pacsize = 160;
-
- res = recObj.StartRecordingAudioFile(fileNameOutUTF8,codecInst,0);
- if (res)
- {
- _engineStatistics.SetLastError(
- VE_BAD_FILE, kTraceError,
- "ConvertPCMToWAV failed to create recorder object");
- playerObj.StopPlayingFile();
- FilePlayer::DestroyFilePlayer(&playerObj);
- recObj.StopRecording();
- FileRecorder::DestroyFileRecorder(&recObj);
- return -1;
- }
-
- // Run throught the file
- AudioFrame audioFrame;
- WebRtc_Word16 decodedData[160];
- WebRtc_UWord32 decLength=0;
- const WebRtc_UWord32 frequency = 16000;
-
- while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
- {
- if(decLength!=frequency/100)
- {
- // This is an OK way to end
- break;
- }
-
- res=audioFrame.UpdateFrame(-1, 0, decodedData,
- (WebRtc_UWord16)decLength,
- frequency, AudioFrame::kNormalSpeech,
- AudioFrame::kVadActive);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertPCMToWAV failed during conversion "
- "(audio frame)");
- break;
- }
-
- res=recObj.RecordAudioToFile(audioFrame);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertPCMToWAV failed during converstion "
- "(write frame)");
- }
- }
-
- playerObj.StopPlayingFile();
- recObj.StopRecording();
- FilePlayer::DestroyFilePlayer(&playerObj);
- FileRecorder::DestroyFileRecorder(&recObj);
-
- return res;
-}
-
-int VoEFileImpl::ConvertPCMToWAV(InStream* streamIn, OutStream* streamOut)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertPCMToWAV(streamIn, streamOut)");
-
- if ((streamIn == NULL) || (streamOut == NULL))
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "invalid stream handles");
- return (-1);
- }
-
- // Create file player object
- FilePlayer& playerObj(*FilePlayer::CreateFilePlayer(-1,
- kFileFormatPcm16kHzFile));
- int res = playerObj.StartPlayingFile(*streamIn,0,1.0,0,0,NULL);
- if (res)
- {
- _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
- "ConvertPCMToWAV failed to create player object");
- playerObj.StopPlayingFile();
- FilePlayer::DestroyFilePlayer(&playerObj);
- return -1;
- }
-
- // Create file recorder object
- FileRecorder& recObj(*FileRecorder::CreateFileRecorder(-1,
- kFileFormatWavFile));
- CodecInst codecInst;
- strncpy(codecInst.plname, "L16", 32);
- codecInst.channels = 1;
- codecInst.rate = 256000;
- codecInst.plfreq = 16000;
- codecInst.pltype = 94;
- codecInst.pacsize = 160;
- res = recObj.StartRecordingAudioFile(*streamOut,codecInst,0);
- if (res)
- {
- _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
- "ConvertPCMToWAV failed to create recorder object");
- playerObj.StopPlayingFile();
- FilePlayer::DestroyFilePlayer(&playerObj);
- recObj.StopRecording();
- FileRecorder::DestroyFileRecorder(&recObj);
- return -1;
- }
-
- // Run throught the file
- AudioFrame audioFrame;
- WebRtc_Word16 decodedData[160];
- WebRtc_UWord32 decLength=0;
- const WebRtc_UWord32 frequency = 16000;
-
- while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
- {
- if(decLength!=frequency/100)
- {
- // This is an OK way to end
- break;
- }
-
- res=audioFrame.UpdateFrame(-1, 0, decodedData,
- (WebRtc_UWord16)decLength, frequency,
- AudioFrame::kNormalSpeech,
- AudioFrame::kVadActive);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertPCMToWAV failed during conversion "
- "(create audio frame)");
- break;
- }
-
- res=recObj.RecordAudioToFile(audioFrame);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertPCMToWAV failed during converstion "
- "(write frame)");
- }
- }
-
- playerObj.StopPlayingFile();
- recObj.StopRecording();
- FilePlayer::DestroyFilePlayer(&playerObj);
- FileRecorder::DestroyFileRecorder(&recObj);
-
- return res;
-}
-
-int VoEFileImpl::ConvertWAVToPCM(const char* fileNameInUTF8,
- const char* fileNameOutUTF8)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertWAVToPCM(fileNameInUTF8=%s, fileNameOutUTF8=%s)",
- fileNameInUTF8, fileNameOutUTF8);
-
- // Create file player object
- FilePlayer& playerObj(*FilePlayer::CreateFilePlayer(-1,
- kFileFormatWavFile));
- int res = playerObj.StartPlayingFile(fileNameInUTF8,false,0,1.0,0,0,NULL);
- if (res)
- {
- _engineStatistics.SetLastError(
- VE_BAD_FILE, kTraceError,
- "ConvertWAVToPCM failed to create player object");
- playerObj.StopPlayingFile();
- FilePlayer::DestroyFilePlayer(&playerObj);
- return -1;
- }
-
- // Create file recorder object
- FileRecorder& recObj(*FileRecorder::CreateFileRecorder(
- -1, kFileFormatPcm16kHzFile));
-
- CodecInst codecInst;
- strncpy(codecInst.plname,"L16",32);
- codecInst.channels = 1;
- codecInst.rate = 256000;
- codecInst.plfreq = 16000;
- codecInst.pltype = 94;
- codecInst.pacsize = 160;
-
- res = recObj.StartRecordingAudioFile(fileNameOutUTF8,codecInst,0);
- if (res)
- {
- _engineStatistics.SetLastError(
- VE_BAD_FILE, kTraceError,
- "ConvertWAVToPCM failed to create recorder object");
- playerObj.StopPlayingFile();
- FilePlayer::DestroyFilePlayer(&playerObj);
- recObj.StopRecording();
- FileRecorder::DestroyFileRecorder(&recObj);
- return -1;
- }
-
- // Run throught the file
- AudioFrame audioFrame;
- WebRtc_Word16 decodedData[160];
- WebRtc_UWord32 decLength=0;
- const WebRtc_UWord32 frequency = 16000;
-
- while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
- {
- if(decLength!=frequency/100)
- {
- // This is an OK way to end
- break;
- }
-
- res=audioFrame.UpdateFrame(-1, 0, decodedData,
- (WebRtc_UWord16)decLength,
- frequency, AudioFrame::kNormalSpeech,
- AudioFrame::kVadActive);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertWAVToPCM failed during conversion "
- "(audio frame)");
- break;
- }
-
- res=recObj.RecordAudioToFile(audioFrame);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertWAVToPCM failed during converstion "
- "(write frame)");
- }
- }
-
- playerObj.StopPlayingFile();
- recObj.StopRecording();
- FilePlayer::DestroyFilePlayer(&playerObj);
- FileRecorder::DestroyFileRecorder(&recObj);
-
- return res;
-}
-
-int VoEFileImpl::ConvertWAVToPCM(InStream* streamIn, OutStream* streamOut)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertWAVToPCM(streamIn, streamOut)");
-
- if ((streamIn == NULL) || (streamOut == NULL))
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "invalid stream handles");
- return (-1);
- }
-
- // Create file player object
- FilePlayer& playerObj(*FilePlayer::CreateFilePlayer(-1,
- kFileFormatWavFile));
- int res = playerObj.StartPlayingFile(*streamIn,0,1.0,0,0,NULL);
- if (res)
- {
- _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
- "ConvertWAVToPCM failed to create player object");
- playerObj.StopPlayingFile();
- FilePlayer::DestroyFilePlayer(&playerObj);
- return -1;
- }
-
- // Create file recorder object
- FileRecorder& recObj(*FileRecorder::CreateFileRecorder(
- -1, kFileFormatPcm16kHzFile));
-
- CodecInst codecInst;
- strncpy(codecInst.plname,"L16",32);
- codecInst.channels = 1;
- codecInst.rate = 256000;
- codecInst.plfreq = 16000;
- codecInst.pltype = 94;
- codecInst.pacsize = 160;
-
- res = recObj.StartRecordingAudioFile(*streamOut,codecInst,0);
- if (res)
- {
- _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
- "ConvertWAVToPCM failed to create recorder object");
- playerObj.StopPlayingFile();
- FilePlayer::DestroyFilePlayer(&playerObj);
- recObj.StopRecording();
- FileRecorder::DestroyFileRecorder(&recObj);
- return -1;
- }
-
- // Run throught the file
- AudioFrame audioFrame;
- WebRtc_Word16 decodedData[160];
- WebRtc_UWord32 decLength=0;
- const WebRtc_UWord32 frequency = 16000;
-
- while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
- {
- if(decLength!=frequency/100)
- {
- // This is an OK way to end
- break;
- }
-
- res=audioFrame.UpdateFrame(-1, 0, decodedData,
- (WebRtc_UWord16)decLength, frequency,
- AudioFrame::kNormalSpeech,
- AudioFrame::kVadActive);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertWAVToPCM failed during conversion "
- "(audio frame)");
- break;
- }
-
- res=recObj.RecordAudioToFile(audioFrame);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertWAVToPCM failed during converstion"
- " (write frame)");
- }
- }
-
- playerObj.StopPlayingFile();
- recObj.StopRecording();
- FilePlayer::DestroyFilePlayer(&playerObj);
- FileRecorder::DestroyFileRecorder(&recObj);
-
- return res;
-}
-
-int VoEFileImpl::ConvertPCMToCompressed(const char* fileNameInUTF8,
- const char* fileNameOutUTF8,
- CodecInst* compression)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertPCMToCompressed(fileNameInUTF8=%s, fileNameOutUTF8=%s"
- ", compression)", fileNameInUTF8, fileNameOutUTF8);
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- " compression: plname=%s, plfreq=%d, pacsize=%d",
- compression->plname, compression->plfreq,
- compression->pacsize);
-
- // Create file player object
- FilePlayer& playerObj(*FilePlayer::CreateFilePlayer(
- -1,
- kFileFormatPcm16kHzFile));
- int res = playerObj.StartPlayingFile(fileNameInUTF8,false,0,1.0,0,0, NULL);
- if (res)
- {
- _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
- "ConvertPCMToCompressed failed to create player object");
- // Clean up and shutdown the file player
- playerObj.StopPlayingFile();
- FilePlayer::DestroyFilePlayer(&playerObj);
- return -1;
- }
-
- // Create file recorder object
- FileRecorder& recObj(*FileRecorder::CreateFileRecorder(
- -1,
- kFileFormatCompressedFile));
- res = recObj.StartRecordingAudioFile(fileNameOutUTF8, *compression,0);
- if (res)
- {
- _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
- "ConvertPCMToCompressed failed to create recorder object");
- playerObj.StopPlayingFile();
- FilePlayer::DestroyFilePlayer(&playerObj);
- recObj.StopRecording();
- FileRecorder::DestroyFileRecorder(&recObj);
- return -1;
- }
-
- // Run throught the file
- AudioFrame audioFrame;
- WebRtc_Word16 decodedData[160];
- WebRtc_UWord32 decLength=0;
- const WebRtc_UWord32 frequency = 16000;
-
- while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
- {
- if(decLength!=frequency/100)
- {
- // This is an OK way to end
- break;
- }
- res=audioFrame.UpdateFrame(-1, 0, decodedData,
- (WebRtc_UWord16)decLength,
- frequency, AudioFrame::kNormalSpeech,
- AudioFrame::kVadActive);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertPCMToCompressed failed during conversion "
- "(audio frame)");
- break;
- }
-
- res=recObj.RecordAudioToFile(audioFrame);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertPCMToCompressed failed during converstion "
- "(write frame)");
- }
- }
-
- playerObj.StopPlayingFile();
- recObj.StopRecording();
- FilePlayer::DestroyFilePlayer(&playerObj);
- FileRecorder::DestroyFileRecorder(&recObj);
-
- return res;
-}
-
-int VoEFileImpl::ConvertPCMToCompressed(InStream* streamIn,
- OutStream* streamOut,
- CodecInst* compression)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertPCMToCompressed(streamIn, streamOut, compression)");
-
- if ((streamIn == NULL) || (streamOut == NULL))
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "invalid stream handles");
- return (-1);
- }
-
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- " compression: plname=%s, plfreq=%d, pacsize=%d",
- compression->plname, compression->plfreq,
- compression->pacsize);
-
- // Create file player object
- FilePlayer& playerObj(*FilePlayer::CreateFilePlayer(
- -1, kFileFormatPcm16kHzFile));
-
- int res = playerObj.StartPlayingFile(*streamIn,0,1.0,0,0,NULL);
- if (res)
- {
- _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
- "ConvertPCMToCompressed failed to create player object");
- playerObj.StopPlayingFile();
- FilePlayer::DestroyFilePlayer(&playerObj);
- return -1;
- }
-
- // Create file recorder object
- FileRecorder& recObj(*FileRecorder::CreateFileRecorder(
- -1, kFileFormatCompressedFile));
- res = recObj.StartRecordingAudioFile(*streamOut,*compression,0);
- if (res)
- {
- _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
- "ConvertPCMToCompressed failed to create recorder object");
- playerObj.StopPlayingFile();
- FilePlayer::DestroyFilePlayer(&playerObj);
- recObj.StopRecording();
- FileRecorder::DestroyFileRecorder(&recObj);
- return -1;
- }
-
- // Run throught the file
- AudioFrame audioFrame;
- WebRtc_Word16 decodedData[160];
- WebRtc_UWord32 decLength=0;
- const WebRtc_UWord32 frequency = 16000;
-
- while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
- {
- if(decLength!=frequency/100)
- {
- // This is an OK way to end
- break;
- }
- res=audioFrame.UpdateFrame(-1, 0, decodedData,
- (WebRtc_UWord16)decLength,
- frequency, AudioFrame::kNormalSpeech,
- AudioFrame::kVadActive);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertPCMToCompressed failed during conversion"
- " (audio frame)");
- break;
- }
-
- res=recObj.RecordAudioToFile(audioFrame);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertPCMToCompressed failed during converstion "
- "(write frame)");
- }
- }
-
- playerObj.StopPlayingFile();
- recObj.StopRecording();
- FilePlayer::DestroyFilePlayer(&playerObj);
- FileRecorder::DestroyFileRecorder(&recObj);
-
- return res;
-}
-
-int VoEFileImpl::ConvertCompressedToPCM(const char* fileNameInUTF8,
- const char* fileNameOutUTF8)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertCompressedToPCM(fileNameInUTF8=%s,"
- " fileNameOutUTF8=%s)",
- fileNameInUTF8, fileNameOutUTF8);
-
- // Create file player object
- FilePlayer& playerObj(*FilePlayer::CreateFilePlayer(
- -1, kFileFormatCompressedFile));
-
- int res = playerObj.StartPlayingFile(fileNameInUTF8,false,0,1.0,0,0,NULL);
- if (res)
- {
- _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
- "ConvertCompressedToPCM failed to create player object");
- playerObj.StopPlayingFile();
- FilePlayer::DestroyFilePlayer(&playerObj);
- return -1;
- }
-
- // Create file recorder object
- FileRecorder& recObj(*FileRecorder::CreateFileRecorder(
- -1, kFileFormatPcm16kHzFile));
-
- CodecInst codecInst;
- strncpy(codecInst.plname,"L16",32);
- codecInst.channels = 1;
- codecInst.rate = 256000;
- codecInst.plfreq = 16000;
- codecInst.pltype = 94;
- codecInst.pacsize = 160;
-
- res = recObj.StartRecordingAudioFile(fileNameOutUTF8,codecInst,0);
- if (res)
- {
- _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
- "ConvertCompressedToPCM failed to create recorder object");
- playerObj.StopPlayingFile();
- FilePlayer::DestroyFilePlayer(&playerObj);
- recObj.StopRecording();
- FileRecorder::DestroyFileRecorder(&recObj);
- return -1;
- }
-
- // Run throught the file
- AudioFrame audioFrame;
- WebRtc_Word16 decodedData[160];
- WebRtc_UWord32 decLength=0;
- const WebRtc_UWord32 frequency = 16000;
-
- while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
- {
- if(decLength!=frequency/100)
- {
- // This is an OK way to end
- break;
- }
- res=audioFrame.UpdateFrame(-1, 0, decodedData,
- (WebRtc_UWord16)decLength,
- frequency,
- AudioFrame::kNormalSpeech,
- AudioFrame::kVadActive);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertCompressedToPCM failed during conversion "
- "(create audio frame)");
- break;
- }
-
- res=recObj.RecordAudioToFile(audioFrame);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertCompressedToPCM failed during converstion "
- "(write frame)");
- }
- }
-
- playerObj.StopPlayingFile();
- recObj.StopRecording();
- FilePlayer::DestroyFilePlayer(&playerObj);
- FileRecorder::DestroyFileRecorder(&recObj);
-
- return res;
-}
-
-int VoEFileImpl::ConvertCompressedToPCM(InStream* streamIn,
- OutStream* streamOut)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertCompressedToPCM(file, file);");
-
- if ((streamIn == NULL) || (streamOut == NULL))
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "invalid stream handles");
- return (-1);
- }
-
- // Create file player object
- FilePlayer& playerObj(*FilePlayer::CreateFilePlayer(
- -1, kFileFormatCompressedFile));
- int res;
-
- res = playerObj.StartPlayingFile(*streamIn,0,1.0,0,0,NULL);
- if (res)
- {
- _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
- "ConvertCompressedToPCM failed to create player object");
- playerObj.StopPlayingFile();
- FilePlayer::DestroyFilePlayer(&playerObj);
- return -1;
- }
-
- // Create file recorder object
- FileRecorder& recObj(*FileRecorder::CreateFileRecorder(
- -1, kFileFormatPcm16kHzFile));
-
- CodecInst codecInst;
- strncpy(codecInst.plname,"L16",32);
- codecInst.channels = 1;
- codecInst.rate = 256000;
- codecInst.plfreq = 16000;
- codecInst.pltype = 94;
- codecInst.pacsize = 160;
-
- res = recObj.StartRecordingAudioFile(*streamOut,codecInst,0);
- if (res)
- {
- _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
- "ConvertCompressedToPCM failed to create recorder object");
- playerObj.StopPlayingFile();
- FilePlayer::DestroyFilePlayer(&playerObj);
- recObj.StopRecording();
- FileRecorder::DestroyFileRecorder(&recObj);
- return -1;
- }
-
- // Run throught the file
- AudioFrame audioFrame;
- WebRtc_Word16 decodedData[160];
- WebRtc_UWord32 decLength=0;
- const WebRtc_UWord32 frequency = 16000;
-
- while(!playerObj.Get10msAudioFromFile(decodedData,decLength,frequency))
- {
- if(decLength!=frequency/100)
- {
- // This is an OK way to end
- break;
- }
- res=audioFrame.UpdateFrame(-1, 0, decodedData,
- (WebRtc_UWord16)decLength,
- frequency,
- AudioFrame::kNormalSpeech,
- AudioFrame::kVadActive);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertCompressedToPCM failed during conversion"
- " (audio frame)");
- break;
- }
-
- res=recObj.RecordAudioToFile(audioFrame);
- if(res)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice, VoEId(_instanceId,-1),
- "ConvertCompressedToPCM failed during converstion"
- " (write frame)");
- }
- }
-
- playerObj.StopPlayingFile();
- recObj.StopRecording();
- FilePlayer::DestroyFilePlayer(&playerObj);
- FileRecorder::DestroyFileRecorder(&recObj);
-
- return res;
-}
-
-
-int VoEFileImpl::GetFileDuration(const char* fileNameUTF8,
- int& durationMs,
- FileFormats format)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetFileDuration(fileNameUTF8=%s, format=%d)",
- fileNameUTF8, format);
-
- // Create a dummy file module for this
- MediaFile * fileModule=MediaFile::CreateMediaFile(-1);
-
- // Temp container of the right format
- WebRtc_UWord32 duration;
- int res=fileModule->FileDurationMs(fileNameUTF8,duration,format);
- if (res)
- {
- _engineStatistics.SetLastError(VE_BAD_FILE, kTraceError,
- "GetFileDuration() failed measure file duration");
- return -1;
- }
- durationMs = duration;
- MediaFile::DestroyMediaFile(fileModule);
- fileModule = NULL;
-
- return(res);
-}
-
-int VoEFileImpl::GetPlaybackPosition(int channel, int& positionMs)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetPlaybackPosition(channel=%d)", channel);
-
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetPlaybackPosition() failed to locate channel");
- return -1;
- }
- return channelPtr->GetLocalPlayoutPosition(positionMs);
-}
-
-#endif // #ifdef WEBRTC_VOICE_ENGINE_FILE_API
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_file_impl.h b/trunk/src/voice_engine/main/source/voe_file_impl.h
deleted file mode 100644
index 2d93876..0000000
--- a/trunk/src/voice_engine/main/source/voe_file_impl.h
+++ /dev/null
@@ -1,143 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_FILE_IMPL_H
-#define WEBRTC_VOICE_ENGINE_VOE_FILE_IMPL_H
-
-#include "voe_file.h"
-#include "shared_data.h"
-#include "ref_count.h"
-
-namespace webrtc {
-
-class VoEFileImpl : public virtual voe::SharedData,
- public VoEFile, public voe::RefCount
-{
-public:
- virtual int Release();
-
- // Playout file locally
-
- virtual int StartPlayingFileLocally(
- int channel,
- const char fileNameUTF8[1024],
- bool loop = false,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0,
- int startPointMs = 0,
- int stopPointMs = 0);
-
- virtual int StartPlayingFileLocally(
- int channel,
- InStream* stream,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0,
- int startPointMs = 0, int stopPointMs = 0);
-
- virtual int StopPlayingFileLocally(int channel);
-
- virtual int IsPlayingFileLocally(int channel);
-
- virtual int ScaleLocalFilePlayout(int channel, float scale);
-
- // Use file as microphone input
-
- virtual int StartPlayingFileAsMicrophone(
- int channel,
- const char fileNameUTF8[1024],
- bool loop = false ,
- bool mixWithMicrophone = false,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0);
-
- virtual int StartPlayingFileAsMicrophone(
- int channel,
- InStream* stream,
- bool mixWithMicrophone = false,
- FileFormats format = kFileFormatPcm16kHzFile,
- float volumeScaling = 1.0);
-
- virtual int StopPlayingFileAsMicrophone(int channel);
-
- virtual int IsPlayingFileAsMicrophone(int channel);
-
- virtual int ScaleFileAsMicrophonePlayout(int channel, float scale);
-
- // Record speaker signal to file
-
- virtual int StartRecordingPlayout(int channel,
- const char* fileNameUTF8,
- CodecInst* compression = NULL,
- int maxSizeBytes = -1);
-
- virtual int StartRecordingPlayout(int channel,
- OutStream* stream,
- CodecInst* compression = NULL);
-
- virtual int StopRecordingPlayout(int channel);
-
- // Record microphone signal to file
-
- virtual int StartRecordingMicrophone(const char* fileNameUTF8,
- CodecInst* compression = NULL,
- int maxSizeBytes = -1);
-
- virtual int StartRecordingMicrophone(OutStream* stream,
- CodecInst* compression = NULL);
-
- virtual int StopRecordingMicrophone();
-
- // Conversion between different file formats
-
- virtual int ConvertPCMToWAV(const char* fileNameInUTF8,
- const char* fileNameOutUTF8);
-
- virtual int ConvertPCMToWAV(InStream* streamIn,
- OutStream* streamOut);
-
- virtual int ConvertWAVToPCM(const char* fileNameInUTF8,
- const char* fileNameOutUTF8);
-
- virtual int ConvertWAVToPCM(InStream* streamIn,
- OutStream* streamOut);
-
- virtual int ConvertPCMToCompressed(const char* fileNameInUTF8,
- const char* fileNameOutUTF8,
- CodecInst* compression);
-
- virtual int ConvertPCMToCompressed(InStream* streamIn,
- OutStream* streamOut,
- CodecInst* compression);
-
- virtual int ConvertCompressedToPCM(const char* fileNameInUTF8,
- const char* fileNameOutUTF8);
-
- virtual int ConvertCompressedToPCM(InStream* streamIn,
- OutStream* streamOut);
-
- // Misc file functions
-
- virtual int GetFileDuration(
- const char* fileNameUTF8,
- int& durationMs,
- FileFormats format = kFileFormatPcm16kHzFile);
-
- virtual int GetPlaybackPosition(int channel, int& positionMs);
-
-protected:
- VoEFileImpl();
- virtual ~VoEFileImpl();
-
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_FILE_IMPL_H
-
diff --git a/trunk/src/voice_engine/main/source/voe_hardware_impl.cc b/trunk/src/voice_engine/main/source/voe_hardware_impl.cc
deleted file mode 100644
index 7e676c0..0000000
--- a/trunk/src/voice_engine/main/source/voe_hardware_impl.cc
+++ /dev/null
@@ -1,850 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voe_hardware_impl.h"
-
-#include <cassert>
-
-#include "cpu_wrapper.h"
-#include "critical_section_wrapper.h"
-#include "trace.h"
-#include "voe_errors.h"
-#include "voice_engine_impl.h"
-
-namespace webrtc
-{
-
-VoEHardware* VoEHardware::GetInterface(VoiceEngine* voiceEngine)
-{
-#ifndef WEBRTC_VOICE_ENGINE_HARDWARE_API
- return NULL;
-#else
- if (NULL == voiceEngine)
- {
- return NULL;
- }
- VoiceEngineImpl* s =
- reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
- VoEHardwareImpl* d = s;
- (*d)++;
- return (d);
-#endif
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_HARDWARE_API
-
-VoEHardwareImpl::VoEHardwareImpl() :
- _cpu(NULL)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
- "VoEHardwareImpl() - ctor");
-
- _cpu = CpuWrapper::CreateCpu();
- if (_cpu)
- {
- _cpu->CpuUsage(); // init cpu usage
- }
-}
-
-VoEHardwareImpl::~VoEHardwareImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
- "~VoEHardwareImpl() - dtor");
-
- if (_cpu)
- {
- delete _cpu;
- _cpu = NULL;
- }
-}
-
-int VoEHardwareImpl::Release()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "VoEHardwareImpl::Release()");
- (*this)--;
- int refCount = GetCount();
- if (refCount < 0)
- {
- Reset();
- _engineStatistics.SetLastError(VE_INTERFACE_NOT_FOUND,
- kTraceWarning);
- return (-1);
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "VoEHardwareImpl reference counter = %d", refCount);
- return (refCount);
-}
-
-int VoEHardwareImpl::SetAudioDeviceLayer(AudioLayers audioLayer)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetAudioDeviceLayer(audioLayer=%d)", audioLayer);
-
- // Don't allow a change if VoE is initialized
- if (_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_ALREADY_INITED, kTraceError);
- return -1;
- }
-
- // Map to AudioDeviceModule::AudioLayer
- AudioDeviceModule::AudioLayer
- wantedLayer(AudioDeviceModule::kPlatformDefaultAudio);
- switch (audioLayer)
- {
- case kAudioPlatformDefault:
- // already set above
- break;
- case kAudioWindowsCore:
- wantedLayer = AudioDeviceModule::kWindowsCoreAudio;
- break;
- case kAudioWindowsWave:
- wantedLayer = AudioDeviceModule::kWindowsWaveAudio;
- break;
- case kAudioLinuxAlsa:
- wantedLayer = AudioDeviceModule::kLinuxAlsaAudio;
- break;
- case kAudioLinuxPulse:
- wantedLayer = AudioDeviceModule::kLinuxPulseAudio;
- break;
- }
-
- // Save the audio device layer for Init()
- _audioDeviceLayer = wantedLayer;
-
- return 0;
-}
-
-int VoEHardwareImpl::GetAudioDeviceLayer(AudioLayers& audioLayer)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetAudioDeviceLayer(devices=?)");
-
- // Can always be called regardless of VoE state
-
- AudioDeviceModule::AudioLayer
- activeLayer(AudioDeviceModule::kPlatformDefaultAudio);
-
- if (_audioDevicePtr)
- {
- // Get active audio layer from ADM
- if (_audioDevicePtr->ActiveAudioLayer(&activeLayer) != 0)
- {
- _engineStatistics.SetLastError(VE_UNDEFINED_SC_ERR, kTraceError,
- " Audio Device error");
- return -1;
- }
- }
- else
- {
- // Return VoE's internal layer setting
- activeLayer = _audioDeviceLayer;
- }
-
- // Map to AudioLayers
- switch (activeLayer)
- {
- case AudioDeviceModule::kPlatformDefaultAudio:
- audioLayer = kAudioPlatformDefault;
- break;
- case AudioDeviceModule::kWindowsCoreAudio:
- audioLayer = kAudioWindowsCore;
- break;
- case AudioDeviceModule::kWindowsWaveAudio:
- audioLayer = kAudioWindowsWave;
- break;
- case AudioDeviceModule::kLinuxAlsaAudio:
- audioLayer = kAudioLinuxAlsa;
- break;
- case AudioDeviceModule::kLinuxPulseAudio:
- audioLayer = kAudioLinuxPulse;
- break;
- default:
- _engineStatistics.SetLastError(VE_UNDEFINED_SC_ERR, kTraceError,
- " unknown audio layer");
- }
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- " Output: audioLayer=%d", audioLayer);
-
- return 0;
-}
-int VoEHardwareImpl::GetNumOfRecordingDevices(int& devices)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetNumOfRecordingDevices(devices=?)");
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- devices = static_cast<int> (_audioDevicePtr->RecordingDevices());
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- " Output: devices=%d", devices);
-
- return 0;
-}
-
-int VoEHardwareImpl::GetNumOfPlayoutDevices(int& devices)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetNumOfPlayoutDevices(devices=?)");
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- devices = static_cast<int> (_audioDevicePtr->PlayoutDevices());
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- " Output: devices=%d", devices);
-
- return 0;
-}
-
-int VoEHardwareImpl::GetRecordingDeviceName(int index,
- char strNameUTF8[128],
- char strGuidUTF8[128])
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetRecordingDeviceName(index=%d)", index);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (strNameUTF8 == NULL)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "GetRecordingDeviceName() invalid argument");
- return -1;
- }
-
- // Note that strGuidUTF8 is allowed to be NULL
-
- // Init len variable to length of supplied vectors
- const WebRtc_UWord16 strLen = 128;
-
- // Check if length has been changed in module
- assert(strLen == kAdmMaxDeviceNameSize);
- assert(strLen == kAdmMaxGuidSize);
-
- WebRtc_Word8 name[strLen];
- WebRtc_Word8 guid[strLen];
-
- // Get names from module
- if (_audioDevicePtr->RecordingDeviceName(index, name, guid) != 0)
- {
- _engineStatistics.SetLastError(
- VE_CANNOT_RETRIEVE_DEVICE_NAME, kTraceError,
- "GetRecordingDeviceName() failed to get device name");
- return -1;
- }
-
- // Copy to vectors supplied by user
- strncpy(strNameUTF8, name, strLen);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- " Output: strNameUTF8=%s", strNameUTF8);
-
- if (strGuidUTF8 != NULL)
- {
- strncpy(strGuidUTF8, guid, strLen);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- " Output: strGuidUTF8=%s", strGuidUTF8);
- }
-
- return 0;
-}
-
-int VoEHardwareImpl::GetPlayoutDeviceName(int index,
- char strNameUTF8[128],
- char strGuidUTF8[128])
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetPlayoutDeviceName(index=%d)", index);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (strNameUTF8 == NULL)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "GetPlayoutDeviceName() invalid argument");
- return -1;
- }
-
- // Note that strGuidUTF8 is allowed to be NULL
-
- // Init len variable to length of supplied vectors
- const WebRtc_UWord16 strLen = 128;
-
- // Check if length has been changed in module
- assert(strLen == kAdmMaxDeviceNameSize);
- assert(strLen == kAdmMaxGuidSize);
-
- WebRtc_Word8 name[strLen];
- WebRtc_Word8 guid[strLen];
-
- // Get names from module
- if (_audioDevicePtr->PlayoutDeviceName(index, name, guid) != 0)
- {
- _engineStatistics.SetLastError(
- VE_CANNOT_RETRIEVE_DEVICE_NAME, kTraceError,
- "GetPlayoutDeviceName() failed to get device name");
- return -1;
- }
-
- // Copy to vectors supplied by user
- strncpy(strNameUTF8, name, strLen);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- " Output: strNameUTF8=%s", strNameUTF8);
-
- if (strGuidUTF8 != NULL)
- {
- strncpy(strGuidUTF8, guid, strLen);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- " Output: strGuidUTF8=%s", strGuidUTF8);
- }
-
- return 0;
-}
-
-int VoEHardwareImpl::SetRecordingDevice(int index,
- StereoChannel recordingChannel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetRecordingDevice(index=%d, recordingChannel=%d)",
- index, (int) recordingChannel);
- CriticalSectionScoped cs(*_apiCritPtr);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- bool isRecording(false);
-
- // Store state about activated recording to be able to restore it after the
- // recording device has been modified.
- if (_audioDevicePtr->Recording())
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "SetRecordingDevice() device is modified while recording"
- " is active...");
- isRecording = true;
- if (_audioDevicePtr->StopRecording() == -1)
- {
- _engineStatistics.SetLastError(
- VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
- "SetRecordingDevice() unable to stop recording");
- return -1;
- }
- }
-
- // We let the module do the index sanity
-
- // Set recording channel
- AudioDeviceModule::ChannelType recCh =
- AudioDeviceModule::kChannelBoth;
- switch (recordingChannel)
- {
- case kStereoLeft:
- recCh = AudioDeviceModule::kChannelLeft;
- break;
- case kStereoRight:
- recCh = AudioDeviceModule::kChannelRight;
- break;
- case kStereoBoth:
- // default setting kChannelBoth (<=> mono)
- break;
- }
-
- if (_audioDevicePtr->SetRecordingChannel(recCh) != 0) {
- _engineStatistics.SetLastError(
- VE_AUDIO_DEVICE_MODULE_ERROR, kTraceWarning,
- "SetRecordingChannel() unable to set the recording channel");
- }
-
- // Map indices to unsigned since underlying functions need that
- WebRtc_UWord16 indexU = static_cast<WebRtc_UWord16> (index);
-
- WebRtc_Word32 res(0);
-
- if (index == -1)
- {
- res = _audioDevicePtr->SetRecordingDevice(
- AudioDeviceModule::kDefaultCommunicationDevice);
- }
- else if (index == -2)
- {
- res = _audioDevicePtr->SetRecordingDevice(
- AudioDeviceModule::kDefaultDevice);
- }
- else
- {
- res = _audioDevicePtr->SetRecordingDevice(indexU);
- }
-
- if (res != 0)
- {
- _engineStatistics.SetLastError(
- VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
- "SetRecordingDevice() unable to set the recording device");
- return -1;
- }
-
- // Init microphone, so user can do volume settings etc
- if (_audioDevicePtr->InitMicrophone() == -1)
- {
- _engineStatistics.SetLastError(
- VE_CANNOT_ACCESS_MIC_VOL, kTraceWarning,
- "SetRecordingDevice() cannot access microphone");
- }
-
- // Set number of channels
- bool available(false);
- if (_audioDevicePtr->StereoRecordingIsAvailable(&available) != 0) {
- _engineStatistics.SetLastError(
- VE_SOUNDCARD_ERROR, kTraceWarning,
- "StereoRecordingIsAvailable() failed to query stereo recording");
- }
-
- if (_audioDevicePtr->SetStereoRecording(available ? true : false) != 0)
- {
- _engineStatistics.SetLastError(
- VE_SOUNDCARD_ERROR, kTraceWarning,
- "SetRecordingDevice() failed to set mono recording mode");
- }
-
- // Restore recording if it was enabled already when calling this function.
- if (isRecording)
- {
- if (!_externalRecording)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "SetRecordingDevice() recording is now being "
- "restored...");
- if (_audioDevicePtr->InitRecording() != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_instanceId, -1),
- "SetRecordingDevice() failed to initialize "
- "recording");
- return -1;
- }
- if (_audioDevicePtr->StartRecording() != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_instanceId, -1),
- "SetRecordingDevice() failed to start recording");
- return -1;
- }
- }
- }
-
- return 0;
-}
-
-int VoEHardwareImpl::SetPlayoutDevice(int index)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetPlayoutDevice(index=%d)", index);
- CriticalSectionScoped cs(*_apiCritPtr);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- bool isPlaying(false);
-
- // Store state about activated playout to be able to restore it after the
- // playout device has been modified.
- if (_audioDevicePtr->Playing())
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "SetPlayoutDevice() device is modified while playout is "
- "active...");
- isPlaying = true;
- if (_audioDevicePtr->StopPlayout() == -1)
- {
- _engineStatistics.SetLastError(
- VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
- "SetPlayoutDevice() unable to stop playout");
- return -1;
- }
- }
-
- // We let the module do the index sanity
-
- // Map indices to unsigned since underlying functions need that
- WebRtc_UWord16 indexU = static_cast<WebRtc_UWord16> (index);
-
- WebRtc_Word32 res(0);
-
- if (index == -1)
- {
- res = _audioDevicePtr->SetPlayoutDevice(
- AudioDeviceModule::kDefaultCommunicationDevice);
- }
- else if (index == -2)
- {
- res = _audioDevicePtr->SetPlayoutDevice(
- AudioDeviceModule::kDefaultDevice);
- }
- else
- {
- res = _audioDevicePtr->SetPlayoutDevice(indexU);
- }
-
- if (res != 0)
- {
- _engineStatistics.SetLastError(
- VE_SOUNDCARD_ERROR, kTraceError,
- "SetPlayoutDevice() unable to set the playout device");
- return -1;
- }
-
- // Init speaker, so user can do volume settings etc
- if (_audioDevicePtr->InitSpeaker() == -1)
- {
- _engineStatistics.SetLastError(
- VE_CANNOT_ACCESS_SPEAKER_VOL, kTraceWarning,
- "SetPlayoutDevice() cannot access speaker");
- }
-
- // Set number of channels
- bool available(false);
- _audioDevicePtr->StereoPlayoutIsAvailable(&available);
- if (_audioDevicePtr->SetStereoPlayout(available ? true : false) != 0)
- {
- _engineStatistics.SetLastError(
- VE_SOUNDCARD_ERROR, kTraceWarning,
- "SetPlayoutDevice() failed to set stereo playout mode");
- }
-
- // Restore playout if it was enabled already when calling this function.
- if (isPlaying)
- {
- if (!_externalPlayout)
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId, -1),
- "SetPlayoutDevice() playout is now being restored...");
- if (_audioDevicePtr->InitPlayout() != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_instanceId, -1),
- "SetPlayoutDevice() failed to initialize playout");
- return -1;
- }
- if (_audioDevicePtr->StartPlayout() != 0)
- {
- WEBRTC_TRACE(kTraceError, kTraceVoice,
- VoEId(_instanceId, -1),
- "SetPlayoutDevice() failed to start playout");
- return -1;
- }
- }
- }
-
- return 0;
-}
-
-int VoEHardwareImpl::GetRecordingDeviceStatus(bool& isAvailable)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetRecordingDeviceStatus()");
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- // We let the module do isRecording sanity
-
- bool available(false);
-
- // Check availability
- if (_audioDevicePtr->RecordingIsAvailable(&available) != 0)
- {
- _engineStatistics.SetLastError(VE_UNDEFINED_SC_REC_ERR, kTraceError,
- " Audio Device error");
- return -1;
- }
-
- isAvailable = available;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- " Output: isAvailable = %d)", (int) isAvailable);
-
- return 0;
-}
-
-int VoEHardwareImpl::GetPlayoutDeviceStatus(bool& isAvailable)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetPlayoutDeviceStatus()");
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- // We let the module do isPlaying sanity
-
- bool available(false);
-
- // Check availability
- if (_audioDevicePtr->PlayoutIsAvailable(&available) != 0)
- {
- _engineStatistics.SetLastError(VE_PLAY_UNDEFINED_SC_ERR,
- kTraceError, " Audio Device error");
- return -1;
- }
-
- isAvailable = available;
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- " Output: isAvailable = %d)", (int) isAvailable);
-
- return 0;
-}
-
-int VoEHardwareImpl::ResetAudioDevice()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "ResetAudioDevice()");
- ANDROID_NOT_SUPPORTED(_engineStatistics);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
-#if defined(MAC_IPHONE)
- if (_audioDevicePtr->ResetAudioDevice() < 0)
- {
- _engineStatistics.SetLastError(VE_SOUNDCARD_ERROR, kTraceError,
- " Failed to reset sound device");
- return -1;
- }
-#else
- _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
- " no support for resetting sound device");
- return -1;
-#endif
-
- return 0;
-}
-
-int VoEHardwareImpl::AudioDeviceControl(unsigned int par1, unsigned int par2,
- unsigned int par3)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "AudioDeviceControl(%i, %i, %i)", par1, par2, par3);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
- " no support for resetting sound device");
- return -1;
-}
-
-int VoEHardwareImpl::SetLoudspeakerStatus(bool enable)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetLoudspeakerStatus(enable=%i)", (int) enable);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-#if defined(WEBRTC_ANDROID)
- if (_audioDevicePtr->SetLoudspeakerStatus(enable) < 0)
- {
- _engineStatistics.SetLastError(VE_IGNORED_FUNCTION, kTraceError,
- " Failed to set loudspeaker status");
- return -1;
- }
-
- return 0;
-#else
- _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
- " no support for setting loudspeaker"
- " status");
- return -1;
-#endif
-}
-
-int VoEHardwareImpl::GetLoudspeakerStatus(bool& enabled)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetLoudspeakerStatus()");
- IPHONE_NOT_SUPPORTED();
-
-#if defined(WEBRTC_ANDROID)
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- if (_audioDevicePtr->GetLoudspeakerStatus(&enabled) < 0)
- {
- _engineStatistics.SetLastError(VE_IGNORED_FUNCTION, kTraceError,
- " Failed to get loudspeaker status");
- return -1;
- }
-
- return 0;
-#else
- _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
- " no support for setting loudspeaker "
- "status");
- return -1;
-#endif
-}
-
-int VoEHardwareImpl::GetCPULoad(int& loadPercent)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetCPULoad()");
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- // Get CPU load from ADM
- WebRtc_UWord16 load(0);
- if (_audioDevicePtr->CPULoad(&load) != 0)
- {
- _engineStatistics.SetLastError(VE_CPU_INFO_ERROR, kTraceError,
- " error getting system CPU load");
- return -1;
- }
-
- loadPercent = static_cast<int> (load);
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- " Output: loadPercent = %d", loadPercent);
-
- return 0;
-}
-
-int VoEHardwareImpl::GetSystemCPULoad(int& loadPercent)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetSystemCPULoad(loadPercent=?)");
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- // Check if implemented for this platform
- if (!_cpu)
- {
- _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
- " no support for getting system CPU "
- "load");
- return -1;
- }
-
- // Get CPU load
- WebRtc_Word32 load = _cpu->CpuUsage();
- if (load < 0)
- {
- _engineStatistics.SetLastError(VE_CPU_INFO_ERROR, kTraceError,
- " error getting system CPU load");
- return -1;
- }
-
- loadPercent = static_cast<int> (load);
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- " Output: loadPercent = %d", loadPercent);
-
- return 0;
-}
-
-int VoEHardwareImpl::EnableBuiltInAEC(bool enable)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "%s", __FUNCTION__);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- return _audioDevicePtr->EnableBuiltInAEC(enable);
-}
-
-bool VoEHardwareImpl::BuiltInAECIsEnabled() const
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "%s", __FUNCTION__);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return false;
- }
-
- return _audioDevicePtr->BuiltInAECIsEnabled();
-}
-
-#endif // WEBRTC_VOICE_ENGINE_HARDWARE_API
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_hardware_impl.h b/trunk/src/voice_engine/main/source/voe_hardware_impl.h
deleted file mode 100644
index af934ba..0000000
--- a/trunk/src/voice_engine/main/source/voe_hardware_impl.h
+++ /dev/null
@@ -1,83 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_HARDWARE_IMPL_H
-#define WEBRTC_VOICE_ENGINE_VOE_HARDWARE_IMPL_H
-
-#include "voe_hardware.h"
-
-#include "ref_count.h"
-#include "shared_data.h"
-
-namespace webrtc
-{
-class CpuWrapper;
-
-class VoEHardwareImpl: public virtual voe::SharedData,
- public VoEHardware,
- public voe::RefCount
-{
-public:
- virtual int Release();
-
- virtual int GetNumOfRecordingDevices(int& devices);
-
- virtual int GetNumOfPlayoutDevices(int& devices);
-
- virtual int GetRecordingDeviceName(int index,
- char strNameUTF8[128],
- char strGuidUTF8[128]);
-
- virtual int GetPlayoutDeviceName(int index,
- char strNameUTF8[128],
- char strGuidUTF8[128]);
-
- virtual int GetRecordingDeviceStatus(bool& isAvailable);
-
- virtual int GetPlayoutDeviceStatus(bool& isAvailable);
-
- virtual int SetRecordingDevice(
- int index,
- StereoChannel recordingChannel = kStereoBoth);
-
- virtual int SetPlayoutDevice(int index);
-
- virtual int SetAudioDeviceLayer(AudioLayers audioLayer);
-
- virtual int GetAudioDeviceLayer(AudioLayers& audioLayer);
-
- virtual int GetCPULoad(int& loadPercent);
-
- virtual int GetSystemCPULoad(int& loadPercent);
-
- virtual int ResetAudioDevice();
-
- virtual int AudioDeviceControl(unsigned int par1,
- unsigned int par2,
- unsigned int par3);
-
- virtual int SetLoudspeakerStatus(bool enable);
-
- virtual int GetLoudspeakerStatus(bool& enabled);
-
- virtual int EnableBuiltInAEC(bool enable);
- virtual bool BuiltInAECIsEnabled() const;
-
-protected:
- VoEHardwareImpl();
- virtual ~VoEHardwareImpl();
-
-private:
- CpuWrapper* _cpu;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_HARDWARE_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_neteq_stats_impl.cc b/trunk/src/voice_engine/main/source/voe_neteq_stats_impl.cc
deleted file mode 100644
index fd87a97..0000000
--- a/trunk/src/voice_engine/main/source/voe_neteq_stats_impl.cc
+++ /dev/null
@@ -1,100 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voe_neteq_stats_impl.h"
-
-#include "audio_coding_module.h"
-#include "channel.h"
-#include "critical_section_wrapper.h"
-#include "trace.h"
-#include "voe_errors.h"
-#include "voice_engine_impl.h"
-
-
-namespace webrtc {
-
-VoENetEqStats* VoENetEqStats::GetInterface(VoiceEngine* voiceEngine)
-{
-#ifndef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
- return NULL;
-#else
- if (NULL == voiceEngine)
- {
- return NULL;
- }
- VoiceEngineImpl* s =
- reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
- VoENetEqStatsImpl* d = s;
- (*d)++;
- return (d);
-#endif
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
-
-VoENetEqStatsImpl::VoENetEqStatsImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "VoENetEqStatsImpl::VoENetEqStatsImpl() - ctor");
-}
-
-VoENetEqStatsImpl::~VoENetEqStatsImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "VoENetEqStatsImpl::~VoENetEqStatsImpl() - dtor");
-}
-
-int VoENetEqStatsImpl::Release()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "VoENetEqStats::Release()");
- (*this)--;
- int refCount = GetCount();
- if (refCount < 0)
- {
- Reset(); // reset reference counter to zero => OK to delete VE
- _engineStatistics.SetLastError(
- VE_INTERFACE_NOT_FOUND, kTraceWarning);
- return (-1);
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "VoENetEqStats reference counter = %d", refCount);
- return (refCount);
-}
-
-int VoENetEqStatsImpl::GetNetworkStatistics(int channel,
- NetworkStatistics& stats)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetNetworkStatistics(channel=%d, stats=?)", channel);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetNetworkStatistics() failed to locate channel");
- return -1;
- }
-
- return channelPtr->GetNetworkStatistics(stats);
-}
-
-#endif // #ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_neteq_stats_impl.h b/trunk/src/voice_engine/main/source/voe_neteq_stats_impl.h
deleted file mode 100644
index 47dd2f6..0000000
--- a/trunk/src/voice_engine/main/source/voe_neteq_stats_impl.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_IMPL_H
-#define WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_IMPL_H
-
-#include "voe_neteq_stats.h"
-
-#include "ref_count.h"
-#include "shared_data.h"
-
-namespace webrtc {
-
-class VoENetEqStatsImpl : public virtual voe::SharedData,
- public VoENetEqStats,
- public voe::RefCount
-{
-public:
- virtual int Release();
-
- virtual int GetNetworkStatistics(int channel,
- NetworkStatistics& stats);
-
-protected:
- VoENetEqStatsImpl();
- virtual ~VoENetEqStatsImpl();
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_NETEQ_STATS_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_network_impl.cc b/trunk/src/voice_engine/main/source/voe_network_impl.cc
deleted file mode 100644
index 0bebbf5..0000000
--- a/trunk/src/voice_engine/main/source/voe_network_impl.cc
+++ /dev/null
@@ -1,947 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voe_network_impl.h"
-
-#include "channel.h"
-#include "critical_section_wrapper.h"
-#include "trace.h"
-#include "voe_errors.h"
-#include "voice_engine_impl.h"
-
-namespace webrtc
-{
-
-VoENetwork* VoENetwork::GetInterface(VoiceEngine* voiceEngine)
-{
-#ifndef WEBRTC_VOICE_ENGINE_NETWORK_API
- return NULL;
-#else
- if (NULL == voiceEngine)
- {
- return NULL;
- }
- VoiceEngineImpl* s =
- reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
- VoENetworkImpl* d = s;
- (*d)++;
- return (d);
-#endif
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_NETWORK_API
-
-VoENetworkImpl::VoENetworkImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
- "VoENetworkImpl() - ctor");
-}
-
-VoENetworkImpl::~VoENetworkImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId, -1),
- "~VoENetworkImpl() - dtor");
-}
-
-int VoENetworkImpl::Release()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "VoENetworkImpl::Release()");
- (*this)--;
- int refCount = GetCount();
- if (refCount < 0)
- {
- Reset();
- _engineStatistics.SetLastError(VE_INTERFACE_NOT_FOUND,
- kTraceWarning);
- return (-1);
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "VoENetworkImpl reference counter = %d", refCount);
- return (refCount);
-}
-
-int VoENetworkImpl::RegisterExternalTransport(int channel,
- Transport& transport)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetExternalTransport(channel=%d, transport=0x%x)",
- channel, &transport);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetExternalTransport() failed to locate channel");
- return -1;
- }
- return channelPtr->RegisterExternalTransport(transport);
-}
-
-int VoENetworkImpl::DeRegisterExternalTransport(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "DeRegisterExternalTransport(channel=%d)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "DeRegisterExternalTransport() failed to locate channel");
- return -1;
- }
- return channelPtr->DeRegisterExternalTransport();
-}
-
-int VoENetworkImpl::ReceivedRTPPacket(int channel,
- const void* data,
- unsigned int length)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "ReceivedRTPPacket(channel=%d, length=%u)", channel, length);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if ((length < 12) || (length > 807))
- {
- _engineStatistics.SetLastError(
- VE_INVALID_PACKET, kTraceError,
- "ReceivedRTPPacket() invalid packet length");
- return -1;
- }
- if (NULL == data)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "ReceivedRTPPacket() invalid data vector");
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "ReceivedRTPPacket() failed to locate channel");
- return -1;
- }
-
- if (!channelPtr->ExternalTransport())
- {
- _engineStatistics.SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "ReceivedRTPPacket() external transport is not enabled");
- return -1;
- }
- return channelPtr->ReceivedRTPPacket((const WebRtc_Word8*) data, length);
-}
-
-int VoENetworkImpl::ReceivedRTCPPacket(int channel, const void* data,
- unsigned int length)
-{
- WEBRTC_TRACE(kTraceStream, kTraceVoice, VoEId(_instanceId, -1),
- "ReceivedRTCPPacket(channel=%d, length=%u)", channel, length);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (length < 4)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_PACKET, kTraceError,
- "ReceivedRTCPPacket() invalid packet length");
- return -1;
- }
- if (NULL == data)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "ReceivedRTCPPacket() invalid data vector");
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "ReceivedRTCPPacket() failed to locate channel");
- return -1;
- }
- if (!channelPtr->ExternalTransport())
- {
- _engineStatistics.SetLastError(
- VE_INVALID_OPERATION, kTraceError,
- "ReceivedRTCPPacket() external transport is not enabled");
- return -1;
- }
- return channelPtr->ReceivedRTCPPacket((const WebRtc_Word8*) data, length);
-}
-
-int VoENetworkImpl::GetSourceInfo(int channel,
- int& rtpPort,
- int& rtcpPort,
- char ipAddr[64])
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetSourceInfo(channel=%d, rtpPort=?, rtcpPort=?, ipAddr[]=?)",
- channel);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (NULL == ipAddr)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "GetSourceInfo() invalid IP-address buffer");
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetSourceInfo() failed to locate channel");
- return -1;
- }
- if (channelPtr->ExternalTransport())
- {
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "GetSourceInfo() external transport is enabled");
- return -1;
- }
- return channelPtr->GetSourceInfo(rtpPort, rtcpPort, ipAddr);
-#else
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "GetSourceInfo() VoE is built for external transport");
- return -1;
-#endif
-}
-
-int VoENetworkImpl::GetLocalIP(char ipAddr[64], bool ipv6)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetLocalIP(ipAddr[]=?, ipv6=%d)", ipv6);
- IPHONE_NOT_SUPPORTED();
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (NULL == ipAddr)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "GetLocalIP() invalid IP-address buffer");
- return -1;
- }
-
- // Create a temporary socket module to ensure that this method can be
- // called also when no channels are created.
- WebRtc_UWord8 numSockThreads(1);
- UdpTransport* socketPtr =
- UdpTransport::Create(
- -1,
- numSockThreads);
- if (NULL == socketPtr)
- {
- _engineStatistics.SetLastError(
- VE_SOCKET_TRANSPORT_MODULE_ERROR, kTraceError,
- "GetLocalIP() failed to create socket module");
- return -1;
- }
-
- char localIPAddr[64];
-
- if (ipv6)
- {
- char localIP[16];
- if (socketPtr->LocalHostAddressIPV6(localIP) != 0)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_IP_ADDRESS, kTraceError,
- "GetLocalIP() failed to retrieve local IP - 1");
- UdpTransport::Destroy(socketPtr);
- return -1;
- }
- // Convert 128-bit address to character string (a:b:c:d:e:f:g:h)
- sprintf(localIPAddr,
- "%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x%.2x:%.2x"
- "%.2x:%.2x%.2x",
- localIP[0], localIP[1], localIP[2], localIP[3], localIP[4],
- localIP[5], localIP[6], localIP[7], localIP[8], localIP[9],
- localIP[10], localIP[11], localIP[12], localIP[13],
- localIP[14], localIP[15]);
- }
- else
- {
- WebRtc_UWord32 localIP(0);
- // Read local IP (as 32-bit address) from the socket module
- if (socketPtr->LocalHostAddress(localIP) != 0)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_IP_ADDRESS, kTraceError,
- "GetLocalIP() failed to retrieve local IP - 2");
- UdpTransport::Destroy(socketPtr);
- return -1;
- }
- // Convert 32-bit address to character string (x.y.z.w)
- sprintf(localIPAddr, "%d.%d.%d.%d", (int) ((localIP >> 24) & 0x0ff),
- (int) ((localIP >> 16) & 0x0ff),
- (int) ((localIP >> 8) & 0x0ff),
- (int) (localIP & 0x0ff));
- }
-
- strcpy(ipAddr, localIPAddr);
-
- UdpTransport::Destroy(socketPtr);
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId, -1),
- "GetLocalIP() => ipAddr=%s", ipAddr);
- return 0;
-#else
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "GetLocalIP() VoE is built for external transport");
- return -1;
-#endif
-}
-
-int VoENetworkImpl::EnableIPv6(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "EnableIPv6(channel=%d)", channel);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "EnableIPv6() failed to locate channel");
- return -1;
- }
- if (channelPtr->ExternalTransport())
- {
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "EnableIPv6() external transport is enabled");
- return -1;
- }
- return channelPtr->EnableIPv6();
-#else
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "EnableIPv6() VoE is built for external transport");
- return -1;
-#endif
-}
-
-bool VoENetworkImpl::IPv6IsEnabled(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "IPv6IsEnabled(channel=%d)", channel);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return false;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "IPv6IsEnabled() failed to locate channel");
- return false;
- }
- if (channelPtr->ExternalTransport())
- {
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "IPv6IsEnabled() external transport is enabled");
- return false;
- }
- return channelPtr->IPv6IsEnabled();
-#else
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "IPv6IsEnabled() VoE is built for external transport");
- return false;
-#endif
-}
-
-int VoENetworkImpl::SetSourceFilter(int channel,
- int rtpPort,
- int rtcpPort,
- const char ipAddr[64])
-{
- (ipAddr == NULL) ? WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
- VoEId(_instanceId, -1),
- "SetSourceFilter(channel=%d, rtpPort=%d,"
- " rtcpPort=%d)",
- channel, rtpPort, rtcpPort)
- : WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
- VoEId(_instanceId, -1),
- "SetSourceFilter(channel=%d, rtpPort=%d,"
- " rtcpPort=%d, ipAddr=%s)",
- channel, rtpPort, rtcpPort, ipAddr);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if ((rtpPort < 0) || (rtpPort > 65535))
- {
- _engineStatistics.SetLastError(
- VE_INVALID_PORT_NMBR, kTraceError,
- "SetSourceFilter() invalid RTP port");
- return -1;
- }
- if ((rtcpPort < 0) || (rtcpPort > 65535))
- {
- _engineStatistics.SetLastError(
- VE_INVALID_PORT_NMBR, kTraceError,
- "SetSourceFilter() invalid RTCP port");
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetSourceFilter() failed to locate channel");
- return -1;
- }
- if (channelPtr->ExternalTransport())
- {
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "SetSourceFilter() external transport is enabled");
- return -1;
- }
- return channelPtr->SetSourceFilter(rtpPort, rtcpPort, ipAddr);
-#else
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "SetSourceFilter() VoE is built for external transport");
- return -1;
-#endif
-}
-
-int VoENetworkImpl::GetSourceFilter(int channel,
- int& rtpPort,
- int& rtcpPort,
- char ipAddr[64])
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetSourceFilter(channel=%d, rtpPort=?, rtcpPort=?, "
- "ipAddr[]=?)",
- channel);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (NULL == ipAddr)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "GetSourceFilter() invalid IP-address buffer");
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetSourceFilter() failed to locate channel");
- return -1;
- }
- if (channelPtr->ExternalTransport())
- {
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "GetSourceFilter() external transport is enabled");
- return -1;
- }
- return channelPtr->GetSourceFilter(rtpPort, rtcpPort, ipAddr);
-#else
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "GetSourceFilter() VoE is built for external transport");
- return -1;
-#endif
-}
-
-int VoENetworkImpl::SetSendTOS(int channel,
- int DSCP,
- int priority,
- bool useSetSockopt)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetSendTOS(channel=%d, DSCP=%d, useSetSockopt=%d)",
- channel, DSCP, useSetSockopt);
-
-#if !defined(_WIN32) && !defined(WEBRTC_LINUX) && !defined(WEBRTC_MAC)
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceWarning,
- "SetSendTOS() is not supported on this platform");
- return -1;
-#endif
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if ((DSCP < 0) || (DSCP > 63))
- {
- _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "SetSendTOS() Invalid DSCP value");
- return -1;
- }
-#if defined(_WIN32) || defined(WEBRTC_LINUX)
- if ((priority < -1) || (priority > 7))
- {
- _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "SetSendTOS() Invalid priority value");
- return -1;
- }
-#else
- if (-1 != priority)
- {
- _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "SetSendTOS() priority not supported");
- return -1;
- }
-#endif
-#if defined(_WIN32)
- if ((priority >= 0) && useSetSockopt)
- {
- // On Windows, priority and useSetSockopt cannot be combined
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetSendTOS() priority and useSetSockopt conflict");
- return -1;
- }
-#endif
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetSendTOS() failed to locate channel");
- return -1;
- }
- if (channelPtr->ExternalTransport())
- {
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "SetSendTOS() external transport is enabled");
- return -1;
- }
-#if defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
- useSetSockopt = true;
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, VoEId(_instanceId,-1),
- " force useSetSockopt=true since there is no alternative"
- " implementation");
-#endif
-
- return channelPtr->SetSendTOS(DSCP, priority, useSetSockopt);
-#else
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "SetSendTOS() VoE is built for external transport");
- return -1;
-#endif
-}
-
-int VoENetworkImpl::GetSendTOS(int channel,
- int& DSCP,
- int& priority,
- bool& useSetSockopt)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetSendTOS(channel=%d)", channel);
-
-#if !defined(_WIN32) && !defined(WEBRTC_LINUX) && !defined(WEBRTC_MAC)
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceWarning,
- "GetSendTOS() is not supported on this platform");
- return -1;
-#endif
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetSendTOS() failed to locate channel");
- return -1;
- }
- if (channelPtr->ExternalTransport())
- {
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "GetSendTOS() external transport is enabled");
- return -1;
- }
- return channelPtr->GetSendTOS(DSCP, priority, useSetSockopt);
-#else
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "GetSendTOS() VoE is built for external transport");
- return -1;
-#endif
-}
-
-int VoENetworkImpl::SetSendGQoS(int channel,
- bool enable,
- int serviceType,
- int overrideDSCP)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetSendGQOS(channel=%d, enable=%d, serviceType=%d,"
- " overrideDSCP=%d)",
- channel, (int) enable, serviceType, overrideDSCP);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-#if !defined(_WIN32)
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceWarning,
- "SetSendGQOS() is not supported on this platform");
- return -1;
-#elif !defined(WEBRTC_EXTERNAL_TRANSPORT)
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "SetSendGQOS() failed to locate channel");
- return -1;
- }
- if (channelPtr->ExternalTransport())
- {
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "SetSendGQOS() external transport is enabled");
- return -1;
- }
- return channelPtr->SetSendGQoS(enable, serviceType, overrideDSCP);
-#else
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "SetSendGQOS() VoE is built for external transport");
- return -1;
-#endif
-}
-
-int VoENetworkImpl::GetSendGQoS(int channel,
- bool& enabled,
- int& serviceType,
- int& overrideDSCP)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetSendGQOS(channel=%d)", channel);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-#if !defined(_WIN32)
- _engineStatistics.SetLastError(
- VE_FUNC_NOT_SUPPORTED, kTraceWarning,
- "GetSendGQOS() is not supported on this platform");
- return -1;
-#elif !defined(WEBRTC_EXTERNAL_TRANSPORT)
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(VE_CHANNEL_NOT_VALID, kTraceError,
- "GetSendGQOS() failed to locate channel");
- return -1;
- }
- if (channelPtr->ExternalTransport())
- {
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceError,
- "GetSendGQOS() external transport is enabled");
- return -1;
- }
- return channelPtr->GetSendGQoS(enabled, serviceType, overrideDSCP);
-#else
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "GetSendGQOS() VoE is built for external transport");
- return -1;
-#endif
-}
-
-int VoENetworkImpl::SetPacketTimeoutNotification(int channel,
- bool enable,
- int timeoutSeconds)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetPacketTimeoutNotification(channel=%d, enable=%d, "
- "timeoutSeconds=%d)",
- channel, (int) enable, timeoutSeconds);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (enable &&
- ((timeoutSeconds < kVoiceEngineMinPacketTimeoutSec) ||
- (timeoutSeconds > kVoiceEngineMaxPacketTimeoutSec)))
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetPacketTimeoutNotification() invalid timeout size");
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetPacketTimeoutNotification() failed to locate channel");
- return -1;
- }
- return channelPtr->SetPacketTimeoutNotification(enable, timeoutSeconds);
-}
-
-int VoENetworkImpl::GetPacketTimeoutNotification(int channel,
- bool& enabled,
- int& timeoutSeconds)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetPacketTimeoutNotification(channel=%d, enabled=?,"
- " timeoutSeconds=?)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetPacketTimeoutNotification() failed to locate channel");
- return -1;
- }
- return channelPtr->GetPacketTimeoutNotification(enabled, timeoutSeconds);
-}
-
-int VoENetworkImpl::RegisterDeadOrAliveObserver(int channel,
- VoEConnectionObserver&
- observer)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "RegisterDeadOrAliveObserver(channel=%d, observer=0x%x)",
- channel, &observer);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "RegisterDeadOrAliveObserver() failed to locate channel");
- return -1;
- }
- return channelPtr->RegisterDeadOrAliveObserver(observer);
-}
-
-int VoENetworkImpl::DeRegisterDeadOrAliveObserver(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "DeRegisterDeadOrAliveObserver(channel=%d)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "DeRegisterDeadOrAliveObserver() failed to locate channel");
- return -1;
- }
- return channelPtr->DeRegisterDeadOrAliveObserver();
-}
-
-int VoENetworkImpl::SetPeriodicDeadOrAliveStatus(int channel, bool enable,
- int sampleTimeSeconds)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SetPeriodicDeadOrAliveStatus(channel=%d, enable=%d,"
- " sampleTimeSeconds=%d)",
- channel, enable, sampleTimeSeconds);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (enable &&
- ((sampleTimeSeconds < kVoiceEngineMinSampleTimeSec) ||
- (sampleTimeSeconds > kVoiceEngineMaxSampleTimeSec)))
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetPeriodicDeadOrAliveStatus() invalid sample time");
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetPeriodicDeadOrAliveStatus() failed to locate channel");
- return -1;
- }
- return channelPtr->SetPeriodicDeadOrAliveStatus(enable, sampleTimeSeconds);
-}
-
-int VoENetworkImpl::GetPeriodicDeadOrAliveStatus(int channel,
- bool& enabled,
- int& sampleTimeSeconds)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "GetPeriodicDeadOrAliveStatus(channel=%d, enabled=?,"
- " sampleTimeSeconds=?)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetPeriodicDeadOrAliveStatus() failed to locate channel");
- return -1;
- }
- return channelPtr->GetPeriodicDeadOrAliveStatus(enabled,
- sampleTimeSeconds);
-}
-
-int VoENetworkImpl::SendUDPPacket(int channel,
- const void* data,
- unsigned int length,
- int& transmittedBytes,
- bool useRtcpSocket)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId, -1),
- "SendUDPPacket(channel=%d, data=0x%x, length=%u, useRTCP=%d)",
- channel, data, length, useRtcpSocket);
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (NULL == data)
- {
- _engineStatistics.SetLastError(VE_INVALID_ARGUMENT, kTraceError,
- "SendUDPPacket() invalid data buffer");
- return -1;
- }
- if (0 == length)
- {
- _engineStatistics.SetLastError(VE_INVALID_PACKET, kTraceError,
- "SendUDPPacket() invalid packet size");
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SendUDPPacket() failed to locate channel");
- return -1;
- }
- return channelPtr->SendUDPPacket(data,
- length,
- transmittedBytes,
- useRtcpSocket);
-#else
- _engineStatistics.SetLastError(
- VE_EXTERNAL_TRANSPORT_ENABLED, kTraceWarning,
- "SendUDPPacket() VoE is built for external transport");
- return -1;
-#endif
-}
-
-#endif // WEBRTC_VOICE_ENGINE_NETWORK_API
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_network_impl.h b/trunk/src/voice_engine/main/source/voe_network_impl.h
deleted file mode 100644
index cc300ac..0000000
--- a/trunk/src/voice_engine/main/source/voe_network_impl.h
+++ /dev/null
@@ -1,117 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_NETWORK_IMPL_H
-#define WEBRTC_VOICE_ENGINE_VOE_NETWORK_IMPL_H
-
-#include "voe_network.h"
-
-#include "ref_count.h"
-#include "shared_data.h"
-
-
-namespace webrtc
-{
-
-class VoENetworkImpl: public virtual voe::SharedData,
- public VoENetwork,
- public voe::RefCount
-{
-public:
- virtual int Release();
-
- virtual int RegisterExternalTransport(int channel, Transport& transport);
-
- virtual int DeRegisterExternalTransport(int channel);
-
- virtual int ReceivedRTPPacket(int channel,
- const void* data,
- unsigned int length);
-
- virtual int ReceivedRTCPPacket(int channel,
- const void* data,
- unsigned int length);
-
- virtual int GetSourceInfo(int channel,
- int& rtpPort,
- int& rtcpPort,
- char ipAddr[64]);
-
- virtual int GetLocalIP(char ipAddr[64], bool ipv6 = false);
-
- virtual int EnableIPv6(int channel);
-
- virtual bool IPv6IsEnabled(int channel);
-
- virtual int SetSourceFilter(int channel,
- int rtpPort,
- int rtcpPort,
- const char ipAddr[64] = 0);
-
- virtual int GetSourceFilter(int channel,
- int& rtpPort,
- int& rtcpPort,
- char ipAddr[64]);
-
- virtual int SetSendTOS(int channel,
- int DSCP,
- int priority = -1,
- bool useSetSockopt = false);
-
- virtual int GetSendTOS(int channel,
- int& DSCP,
- int& priority,
- bool& useSetSockopt);
-
- virtual int SetSendGQoS(int channel,
- bool enable,
- int serviceType,
- int overrideDSCP);
-
- virtual int GetSendGQoS(int channel,
- bool& enabled,
- int& serviceType,
- int& overrideDSCP);
-
- virtual int SetPacketTimeoutNotification(int channel,
- bool enable,
- int timeoutSeconds = 2);
-
- virtual int GetPacketTimeoutNotification(int channel,
- bool& enabled,
- int& timeoutSeconds);
-
- virtual int RegisterDeadOrAliveObserver(int channel,
- VoEConnectionObserver& observer);
-
- virtual int DeRegisterDeadOrAliveObserver(int channel);
-
- virtual int SetPeriodicDeadOrAliveStatus(int channel,
- bool enable,
- int sampleTimeSeconds = 2);
-
- virtual int GetPeriodicDeadOrAliveStatus(int channel,
- bool& enabled,
- int& sampleTimeSeconds);
-
- virtual int SendUDPPacket(int channel,
- const void* data,
- unsigned int length,
- int& transmittedBytes,
- bool useRtcpSocket = false);
-
-protected:
- VoENetworkImpl();
- virtual ~VoENetworkImpl();
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_NETWORK_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_rtp_rtcp_impl.cc b/trunk/src/voice_engine/main/source/voe_rtp_rtcp_impl.cc
deleted file mode 100644
index cbc4d0d..0000000
--- a/trunk/src/voice_engine/main/source/voe_rtp_rtcp_impl.cc
+++ /dev/null
@@ -1,730 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voe_rtp_rtcp_impl.h"
-#include "trace.h"
-#include "file_wrapper.h"
-#include "critical_section_wrapper.h"
-#include "voice_engine_impl.h"
-#include "voe_errors.h"
-
-#include "channel.h"
-#include "transmit_mixer.h"
-
-namespace webrtc {
-
-VoERTP_RTCP* VoERTP_RTCP::GetInterface(VoiceEngine* voiceEngine)
-{
-#ifndef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
- return NULL;
-#else
- if (NULL == voiceEngine)
- {
- return NULL;
- }
- VoiceEngineImpl* s = reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
- VoERTP_RTCPImpl* d = s;
- (*d)++;
- return (d);
-#endif
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
-
-VoERTP_RTCPImpl::VoERTP_RTCPImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "VoERTP_RTCPImpl::VoERTP_RTCPImpl() - ctor");
-}
-
-VoERTP_RTCPImpl::~VoERTP_RTCPImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "VoERTP_RTCPImpl::~VoERTP_RTCPImpl() - dtor");
-}
-
-int VoERTP_RTCPImpl::Release()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "VoERTP_RTCP::Release()");
- (*this)--;
- int refCount = GetCount();
- if (refCount < 0)
- {
- Reset(); // reset reference counter to zero => OK to delete VE
- _engineStatistics.SetLastError(
- VE_INTERFACE_NOT_FOUND, kTraceWarning);
- return (-1);
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "VoERTP_RTCP reference counter = %d", refCount);
- return (refCount);
-}
-
-int VoERTP_RTCPImpl::RegisterRTPObserver(int channel, VoERTPObserver& observer)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "RegisterRTPObserver(channel=%d observer=0x%x)",
- channel, &observer);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "RegisterRTPObserver() failed to locate channel");
- return -1;
- }
- return channelPtr->RegisterRTPObserver(observer);
-}
-
-int VoERTP_RTCPImpl::DeRegisterRTPObserver(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "DeRegisterRTPObserver(channel=%d)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "DeRegisterRTPObserver() failed to locate channel");
- return -1;
- }
- return channelPtr->DeRegisterRTPObserver();
-}
-
-int VoERTP_RTCPImpl::RegisterRTCPObserver(int channel, VoERTCPObserver& observer)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "RegisterRTCPObserver(channel=%d observer=0x%x)",
- channel, &observer);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "RegisterRTPObserver() failed to locate channel");
- return -1;
- }
- return channelPtr->RegisterRTCPObserver(observer);
-}
-
-int VoERTP_RTCPImpl::DeRegisterRTCPObserver(int channel)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "DeRegisterRTCPObserver(channel=%d)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "DeRegisterRTCPObserver() failed to locate channel");
- return -1;
- }
- return channelPtr->DeRegisterRTCPObserver();
-}
-
-int VoERTP_RTCPImpl::SetLocalSSRC(int channel, unsigned int ssrc)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetLocalSSRC(channel=%d, %lu)", channel, ssrc);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetLocalSSRC() failed to locate channel");
- return -1;
- }
- return channelPtr->SetLocalSSRC(ssrc);
-}
-
-int VoERTP_RTCPImpl::GetLocalSSRC(int channel, unsigned int& ssrc)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetLocalSSRC(channel=%d, ssrc=?)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetLocalSSRC() failed to locate channel");
- return -1;
- }
- return channelPtr->GetLocalSSRC(ssrc);
-}
-
-int VoERTP_RTCPImpl::GetRemoteSSRC(int channel, unsigned int& ssrc)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetRemoteSSRC(channel=%d, ssrc=?)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRemoteSSRC() failed to locate channel");
- return -1;
- }
- return channelPtr->GetRemoteSSRC(ssrc);
-}
-
-int VoERTP_RTCPImpl::GetRemoteCSRCs(int channel, unsigned int arrCSRC[15])
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetRemoteCSRCs(channel=%d, arrCSRC=?)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRemoteCSRCs() failed to locate channel");
- return -1;
- }
- return channelPtr->GetRemoteCSRCs(arrCSRC);
-}
-
-
-int VoERTP_RTCPImpl::SetRTPAudioLevelIndicationStatus(int channel,
- bool enable,
- unsigned char ID)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetRTPAudioLevelIndicationStatus(channel=%d, enable=%d,"
- " ID=%u)", channel, enable, ID);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (ID < kVoiceEngineMinRtpExtensionId ||
- ID > kVoiceEngineMaxRtpExtensionId)
- {
- // [RFC5285] The 4-bit ID is the local identifier of this element in
- // the range 1-14 inclusive.
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetRTPAudioLevelIndicationStatus() invalid ID parameter");
- return -1;
- }
-
- // Set state and ID for the specified channel.
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetRTPAudioLevelIndicationStatus() failed to locate channel");
- return -1;
- }
- return channelPtr->SetRTPAudioLevelIndicationStatus(enable, ID);
-}
-
-int VoERTP_RTCPImpl::GetRTPAudioLevelIndicationStatus(int channel,
- bool& enabled,
- unsigned char& ID)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetRTPAudioLevelIndicationStatus(channel=%d, enable=?, ID=?)",
- channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRTPAudioLevelIndicationStatus() failed to locate channel");
- return -1;
- }
- return channelPtr->GetRTPAudioLevelIndicationStatus(enabled, ID);
-}
-
-int VoERTP_RTCPImpl::SetRTCPStatus(int channel, bool enable)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetRTCPStatus(channel=%d, enable=%d)", channel, enable);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetRTCPStatus() failed to locate channel");
- return -1;
- }
- return channelPtr->SetRTCPStatus(enable);
-}
-
-int VoERTP_RTCPImpl::GetRTCPStatus(int channel, bool& enabled)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetRTCPStatus(channel=%d)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRTCPStatus() failed to locate channel");
- return -1;
- }
- return channelPtr->GetRTCPStatus(enabled);
-}
-
-int VoERTP_RTCPImpl::SetRTCP_CNAME(int channel, const char cName[256])
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetRTCP_CNAME(channel=%d, cName=%s)", channel, cName);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetRTCP_CNAME() failed to locate channel");
- return -1;
- }
- return channelPtr->SetRTCP_CNAME(cName);
-}
-
-int VoERTP_RTCPImpl::GetRTCP_CNAME(int channel, char cName[256])
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetRTCP_CNAME(channel=%d, cName=?)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRTCP_CNAME() failed to locate channel");
- return -1;
- }
- return channelPtr->GetRTCP_CNAME(cName);
-}
-
-int VoERTP_RTCPImpl::GetRemoteRTCP_CNAME(int channel, char cName[256])
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetRemoteRTCP_CNAME(channel=%d, cName=?)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRemoteRTCP_CNAME() failed to locate channel");
- return -1;
- }
- return channelPtr->GetRemoteRTCP_CNAME(cName);
-}
-
-int VoERTP_RTCPImpl::GetRemoteRTCPData(
- int channel,
- unsigned int& NTPHigh, // from sender info in SR
- unsigned int& NTPLow, // from sender info in SR
- unsigned int& timestamp, // from sender info in SR
- unsigned int& playoutTimestamp, // derived locally
- unsigned int* jitter, // from report block 1 in SR/RR
- unsigned short* fractionLost) // from report block 1 in SR/RR
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetRemoteRTCPData(channel=%d,...)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRemoteRTCP_CNAME() failed to locate channel");
- return -1;
- }
- return channelPtr->GetRemoteRTCPData(NTPHigh,
- NTPLow,
- timestamp,
- playoutTimestamp,
- jitter,
- fractionLost);
-}
-
-int VoERTP_RTCPImpl::SendApplicationDefinedRTCPPacket(
- int channel,
- const unsigned char subType,
- unsigned int name,
- const char* data,
- unsigned short dataLengthInBytes)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SendApplicationDefinedRTCPPacket(channel=%d, subType=%u,"
- "name=%u, data=?, dataLengthInBytes=%u)",
- channel, subType, name, dataLengthInBytes);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SendApplicationDefinedRTCPPacket() failed to locate channel");
- return -1;
- }
- return channelPtr->SendApplicationDefinedRTCPPacket(subType,
- name,
- data,
- dataLengthInBytes);
-}
-
-int VoERTP_RTCPImpl::GetRTPStatistics(int channel,
- unsigned int& averageJitterMs,
- unsigned int& maxJitterMs,
- unsigned int& discardedPackets)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetRTPStatistics(channel=%d,....)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRTPStatistics() failed to locate channel");
- return -1;
- }
- return channelPtr->GetRTPStatistics(averageJitterMs,
- maxJitterMs,
- discardedPackets);
-}
-
-int VoERTP_RTCPImpl::GetRTCPStatistics(int channel, CallStatistics& stats)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetRTCPStatistics(channel=%d)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRTPStatistics() failed to locate channel");
- return -1;
- }
- return channelPtr->GetRTPStatistics(stats);
-}
-
-int VoERTP_RTCPImpl::SetFECStatus(int channel, bool enable, int redPayloadtype)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetFECStatus(channel=%d, enable=%d, redPayloadtype=%d)",
- channel, enable, redPayloadtype);
-#ifdef WEBRTC_CODEC_RED
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetFECStatus() failed to locate channel");
- return -1;
- }
- return channelPtr->SetFECStatus(enable, redPayloadtype);
-#else
- _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
- "SetFECStatus() RED is not supported");
- return -1;
-#endif
-}
-
-int VoERTP_RTCPImpl::GetFECStatus(int channel,
- bool& enabled,
- int& redPayloadtype)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetFECStatus(channel=%d, enabled=?, redPayloadtype=?)",
- channel);
-#ifdef WEBRTC_CODEC_RED
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetFECStatus() failed to locate channel");
- return -1;
- }
- return channelPtr->GetFECStatus(enabled, redPayloadtype);
-#else
- _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError,
- "GetFECStatus() RED is not supported");
- return -1;
-#endif
-}
-
-int VoERTP_RTCPImpl::SetRTPKeepaliveStatus(int channel,
- bool enable,
- unsigned char unknownPayloadType,
- int deltaTransmitTimeSeconds)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetRTPKeepaliveStatus(channel=%d, enable=%d,"
- " unknownPayloadType=%u, deltaTransmitTimeSeconds=%d)",
- channel, enable, unknownPayloadType, deltaTransmitTimeSeconds);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetRTPKeepaliveStatus() failed to locate channel");
- return -1;
- }
- return channelPtr->SetRTPKeepaliveStatus(enable,
- unknownPayloadType,
- deltaTransmitTimeSeconds);
-}
-
-int VoERTP_RTCPImpl::GetRTPKeepaliveStatus(int channel,
- bool& enabled,
- unsigned char& unknownPayloadType,
- int& deltaTransmitTimeSeconds)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetRTPKeepaliveStatus(channel=%d)", channel);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetRTPKeepaliveStatus() failed to locate channel");
- return -1;
- }
- return channelPtr->GetRTPKeepaliveStatus(enabled,
- unknownPayloadType,
- deltaTransmitTimeSeconds);
-}
-
-int VoERTP_RTCPImpl::StartRTPDump(int channel,
- const char fileNameUTF8[1024],
- RTPDirections direction)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StartRTPDump(channel=%d, fileNameUTF8=%s, direction=%d)",
- channel, fileNameUTF8, direction);
- assert(1024 == FileWrapper::kMaxFileNameSize);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "StartRTPDump() failed to locate channel");
- return -1;
- }
- return channelPtr->StartRTPDump(fileNameUTF8, direction);
-}
-
-int VoERTP_RTCPImpl::StopRTPDump(int channel, RTPDirections direction)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "StopRTPDump(channel=%d, direction=%d)", channel, direction);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "StopRTPDump() failed to locate channel");
- return -1;
- }
- return channelPtr->StopRTPDump(direction);
-}
-
-int VoERTP_RTCPImpl::RTPDumpIsActive(int channel, RTPDirections direction)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "RTPDumpIsActive(channel=%d, direction=%d)",
- channel, direction);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "StopRTPDump() failed to locate channel");
- return -1;
- }
- return channelPtr->RTPDumpIsActive(direction);
-}
-
-int VoERTP_RTCPImpl::InsertExtraRTPPacket(int channel,
- unsigned char payloadType,
- bool markerBit,
- const char* payloadData,
- unsigned short payloadSize)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "InsertExtraRTPPacket(channel=%d, payloadType=%u,"
- " markerBit=%u, payloadSize=%u)",
- channel, payloadType, markerBit, payloadSize);
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "StopRTPDump() failed to locate channel");
- return -1;
- }
- return channelPtr->InsertExtraRTPPacket(payloadType,
- markerBit,
- payloadData,
- payloadSize);
-}
-
-#endif // #ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_rtp_rtcp_impl.h b/trunk/src/voice_engine/main/source/voe_rtp_rtcp_impl.h
deleted file mode 100644
index 3cdf162..0000000
--- a/trunk/src/voice_engine/main/source/voe_rtp_rtcp_impl.h
+++ /dev/null
@@ -1,134 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H
-#define WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H
-
-#include "voe_rtp_rtcp.h"
-
-#include "ref_count.h"
-#include "shared_data.h"
-
-namespace webrtc {
-
-class VoERTP_RTCPImpl : public virtual voe::SharedData,
- public VoERTP_RTCP,
- public voe::RefCount
-{
-public:
-
- virtual int Release();
- // Registration of observers for RTP and RTCP callbacks
- virtual int RegisterRTPObserver(int channel, VoERTPObserver& observer);
-
- virtual int DeRegisterRTPObserver(int channel);
-
- virtual int RegisterRTCPObserver(int channel, VoERTCPObserver& observer);
-
- virtual int DeRegisterRTCPObserver(int channel);
-
- // RTCP
- virtual int SetRTCPStatus(int channel, bool enable);
-
- virtual int GetRTCPStatus(int channel, bool& enabled);
-
- virtual int SetRTCP_CNAME(int channel, const char cName[256]);
-
- virtual int GetRTCP_CNAME(int channel, char cName[256]);
-
- virtual int GetRemoteRTCP_CNAME(int channel, char cName[256]);
-
- virtual int GetRemoteRTCPData(int channel,
- unsigned int& NTPHigh,
- unsigned int& NTPLow,
- unsigned int& timestamp,
- unsigned int& playoutTimestamp,
- unsigned int* jitter = NULL,
- unsigned short* fractionLost = NULL);
-
- virtual int SendApplicationDefinedRTCPPacket(
- int channel,
- const unsigned char subType,
- unsigned int name,
- const char* data,
- unsigned short dataLengthInBytes);
-
- // SSRC
- virtual int SetLocalSSRC(int channel, unsigned int ssrc);
-
- virtual int GetLocalSSRC(int channel, unsigned int& ssrc);
-
- virtual int GetRemoteSSRC(int channel, unsigned int& ssrc);
-
- // RTP Header Extension for Client-to-Mixer Audio Level Indication
- virtual int SetRTPAudioLevelIndicationStatus(int channel,
- bool enable,
- unsigned char ID);
-
- virtual int GetRTPAudioLevelIndicationStatus(int channel,
- bool& enabled,
- unsigned char& ID);
-
- // CSRC
- virtual int GetRemoteCSRCs(int channel, unsigned int arrCSRC[15]);
-
- // Statistics
- virtual int GetRTPStatistics(int channel,
- unsigned int& averageJitterMs,
- unsigned int& maxJitterMs,
- unsigned int& discardedPackets);
-
- virtual int GetRTCPStatistics(int channel, CallStatistics& stats);
-
- // RTP keepalive mechanism (maintains NAT mappings associated to RTP flows)
- virtual int SetRTPKeepaliveStatus(int channel,
- bool enable,
- unsigned char unknownPayloadType,
- int deltaTransmitTimeSeconds = 15);
-
- virtual int GetRTPKeepaliveStatus(int channel,
- bool& enabled,
- unsigned char& unknownPayloadType,
- int& deltaTransmitTimeSeconds);
-
- // FEC
- virtual int SetFECStatus(int channel,
- bool enable,
- int redPayloadtype = -1);
-
- virtual int GetFECStatus(int channel, bool& enabled, int& redPayloadtype);
-
- // Store RTP and RTCP packets and dump to file (compatible with rtpplay)
- virtual int StartRTPDump(int channel,
- const char fileNameUTF8[1024],
- RTPDirections direction = kRtpIncoming);
-
- virtual int StopRTPDump(int channel,
- RTPDirections direction = kRtpIncoming);
-
- virtual int RTPDumpIsActive(int channel,
- RTPDirections direction = kRtpIncoming);
-
- // Insert (and transmits) extra RTP packet into active RTP audio stream
- virtual int InsertExtraRTPPacket(int channel,
- unsigned char payloadType,
- bool markerBit,
- const char* payloadData,
- unsigned short payloadSize);
-
-protected:
- VoERTP_RTCPImpl();
- virtual ~VoERTP_RTCPImpl();
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_RTP_RTCP_IMPL_H
-
diff --git a/trunk/src/voice_engine/main/source/voe_video_sync_impl.cc b/trunk/src/voice_engine/main/source/voe_video_sync_impl.cc
deleted file mode 100644
index 8ee6296..0000000
--- a/trunk/src/voice_engine/main/source/voe_video_sync_impl.cc
+++ /dev/null
@@ -1,248 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voe_video_sync_impl.h"
-
-#include "channel.h"
-#include "critical_section_wrapper.h"
-#include "trace.h"
-#include "voe_errors.h"
-#include "voice_engine_impl.h"
-
-namespace webrtc {
-
-VoEVideoSync* VoEVideoSync::GetInterface(VoiceEngine* voiceEngine)
-{
-#ifndef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
- return NULL;
-#else
- if (NULL == voiceEngine)
- {
- return NULL;
- }
- VoiceEngineImpl* s =
- reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
- VoEVideoSyncImpl* d = s;
- (*d)++;
- return (d);
-#endif
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
-
-VoEVideoSyncImpl::VoEVideoSyncImpl(voe::SharedData* data) : _data(data)
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_data->instance_id(),-1),
- "VoEVideoSyncImpl::VoEVideoSyncImpl() - ctor");
-}
-
-VoEVideoSyncImpl::~VoEVideoSyncImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_data->instance_id(),-1),
- "VoEVideoSyncImpl::~VoEVideoSyncImpl() - dtor");
-}
-
-int VoEVideoSyncImpl::Release()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_data->instance_id(),-1),
- "VoEVideoSync::Release()");
- (*this)--;
- int refCount = GetCount();
- if (refCount < 0)
- {
- Reset(); // reset reference counter to zero => OK to delete VE
- _data->statistics().SetLastError(VE_INTERFACE_NOT_FOUND,
- kTraceWarning);
- return (-1);
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_data->instance_id(),-1),
- "VoEVideoSync reference counter = %d", refCount);
- return (refCount);
-}
-
-int VoEVideoSyncImpl::GetPlayoutTimestamp(int channel, unsigned int& timestamp)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_data->instance_id(),-1),
- "GetPlayoutTimestamp(channel=%d, timestamp=?)", channel);
- ANDROID_NOT_SUPPORTED(_data->statistics());
- IPHONE_NOT_SUPPORTED();
-
- if (!_data->statistics().Initialized())
- {
- _data->statistics().SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_data->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _data->statistics().SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetPlayoutTimestamp() failed to locate channel");
- return -1;
- }
- return channelPtr->GetPlayoutTimestamp(timestamp);
-}
-
-int VoEVideoSyncImpl::SetInitTimestamp(int channel,
- unsigned int timestamp)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_data->instance_id(),-1),
- "SetInitTimestamp(channel=%d, timestamp=%lu)",
- channel, timestamp);
- ANDROID_NOT_SUPPORTED(_data->statistics());
- IPHONE_NOT_SUPPORTED();
-
- if (!_data->statistics().Initialized())
- {
- _data->statistics().SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_data->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _data->statistics().SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetInitTimestamp() failed to locate channel");
- return -1;
- }
- return channelPtr->SetInitTimestamp(timestamp);
-}
-
-int VoEVideoSyncImpl::SetInitSequenceNumber(int channel,
- short sequenceNumber)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_data->instance_id(),-1),
- "SetInitSequenceNumber(channel=%d, sequenceNumber=%hd)",
- channel, sequenceNumber);
- ANDROID_NOT_SUPPORTED(_data->statistics());
- IPHONE_NOT_SUPPORTED();
-
- if (!_data->statistics().Initialized())
- {
- _data->statistics().SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_data->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _data->statistics().SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetInitSequenceNumber() failed to locate channel");
- return -1;
- }
- return channelPtr->SetInitSequenceNumber(sequenceNumber);
-}
-
-int VoEVideoSyncImpl::SetMinimumPlayoutDelay(int channel,int delayMs)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_data->instance_id(),-1),
- "SetMinimumPlayoutDelay(channel=%d, delayMs=%d)",
- channel, delayMs);
- ANDROID_NOT_SUPPORTED(_data->statistics());
- IPHONE_NOT_SUPPORTED();
-
- if (!_data->statistics().Initialized())
- {
- _data->statistics().SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_data->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _data->statistics().SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetMinimumPlayoutDelay() failed to locate channel");
- return -1;
- }
- return channelPtr->SetMinimumPlayoutDelay(delayMs);
-}
-
-int VoEVideoSyncImpl::GetDelayEstimate(int channel, int& delayMs)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_data->instance_id(),-1),
- "GetDelayEstimate(channel=%d, delayMs=?)", channel);
- ANDROID_NOT_SUPPORTED(_data->statistics());
- IPHONE_NOT_SUPPORTED();
-
- if (!_data->statistics().Initialized())
- {
- _data->statistics().SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_data->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _data->statistics().SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetDelayEstimate() failed to locate channel");
- return -1;
- }
- return channelPtr->GetDelayEstimate(delayMs);
-}
-
-int VoEVideoSyncImpl::GetPlayoutBufferSize(int& bufferMs)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_data->instance_id(),-1),
- "GetPlayoutBufferSize(bufferMs=?)");
- ANDROID_NOT_SUPPORTED(_data->statistics());
- IPHONE_NOT_SUPPORTED();
-
- if (!_data->statistics().Initialized())
- {
- _data->statistics().SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- AudioDeviceModule::BufferType type
- (AudioDeviceModule::kFixedBufferSize);
- WebRtc_UWord16 sizeMS(0);
- if (_data->audio_device()->PlayoutBuffer(&type, &sizeMS) != 0)
- {
- _data->statistics().SetLastError(
- VE_AUDIO_DEVICE_MODULE_ERROR, kTraceError,
- "GetPlayoutBufferSize() failed to read buffer size");
- return -1;
- }
- bufferMs = sizeMS;
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_data->instance_id(),-1),
- "GetPlayoutBufferSize() => bufferMs=%d", bufferMs);
- return 0;
-}
-
-int VoEVideoSyncImpl::GetRtpRtcp(int channel, RtpRtcp* &rtpRtcpModule)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_data->instance_id(),-1),
- "GetRtpRtcp(channel=%i)", channel);
-
- if (!_data->statistics().Initialized())
- {
- _data->statistics().SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_data->channel_manager(), channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _data->statistics().SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetPlayoutTimestamp() failed to locate channel");
- return -1;
- }
- return channelPtr->GetRtpRtcp(rtpRtcpModule);
-}
-
-
-#endif // #ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_video_sync_impl.h b/trunk/src/voice_engine/main/source/voe_video_sync_impl.h
deleted file mode 100644
index 02c85d1..0000000
--- a/trunk/src/voice_engine/main/source/voe_video_sync_impl.h
+++ /dev/null
@@ -1,51 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_IMPL_H
-#define WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_IMPL_H
-
-#include "voe_video_sync.h"
-
-#include "ref_count.h"
-#include "shared_data.h"
-
-namespace webrtc {
-
-class VoEVideoSyncImpl : public VoEVideoSync,
- public voe::RefCount
-{
-public:
- virtual int Release();
-
- virtual int GetPlayoutBufferSize(int& bufferMs);
-
- virtual int SetMinimumPlayoutDelay(int channel, int delayMs);
-
- virtual int GetDelayEstimate(int channel, int& delayMs);
-
- virtual int SetInitTimestamp(int channel, unsigned int timestamp);
-
- virtual int SetInitSequenceNumber(int channel, short sequenceNumber);
-
- virtual int GetPlayoutTimestamp(int channel, unsigned int& timestamp);
-
- virtual int GetRtpRtcp(int channel, RtpRtcp* &rtpRtcpModule);
-
-protected:
- VoEVideoSyncImpl(voe::SharedData* data);
- virtual ~VoEVideoSyncImpl();
-
-private:
- voe::SharedData* _data;
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_VIDEO_SYNC_IMPL_H
diff --git a/trunk/src/voice_engine/main/source/voe_volume_control_impl.cc b/trunk/src/voice_engine/main/source/voe_volume_control_impl.cc
deleted file mode 100644
index dd3d54d..0000000
--- a/trunk/src/voice_engine/main/source/voe_volume_control_impl.cc
+++ /dev/null
@@ -1,677 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voe_volume_control_impl.h"
-
-#include "channel.h"
-#include "critical_section_wrapper.h"
-#include "output_mixer.h"
-#include "trace.h"
-#include "transmit_mixer.h"
-#include "voe_errors.h"
-#include "voice_engine_impl.h"
-
-namespace webrtc {
-
-VoEVolumeControl* VoEVolumeControl::GetInterface(VoiceEngine* voiceEngine)
-{
-#ifndef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
- return NULL;
-#else
- if (NULL == voiceEngine)
- {
- return NULL;
- }
- VoiceEngineImpl* s =
- reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
- VoEVolumeControlImpl* d = s;
- (*d)++;
- return (d);
-#endif
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
-
-VoEVolumeControlImpl::VoEVolumeControlImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "VoEVolumeControlImpl::VoEVolumeControlImpl() - ctor");
-}
-
-VoEVolumeControlImpl::~VoEVolumeControlImpl()
-{
- WEBRTC_TRACE(kTraceMemory, kTraceVoice, VoEId(_instanceId,-1),
- "VoEVolumeControlImpl::~VoEVolumeControlImpl() - dtor");
-}
-
-int VoEVolumeControlImpl::Release()
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "VoEVolumeControl::Release()");
- (*this)--;
- int refCount = GetCount();
- if (refCount < 0)
- {
- Reset(); // reset reference counter to zero => OK to delete VE
- _engineStatistics.SetLastError(
- VE_INTERFACE_NOT_FOUND, kTraceWarning);
- return (-1);
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "VoEVolumeControl reference counter = %d", refCount);
- return (refCount);
-}
-
-int VoEVolumeControlImpl::SetSpeakerVolume(unsigned int volume)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetSpeakerVolume(volume=%u)", volume);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (volume > kMaxVolumeLevel)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetSpeakerVolume() invalid argument");
- return -1;
- }
-
- WebRtc_UWord32 maxVol(0);
- WebRtc_UWord32 spkrVol(0);
-
- // scale: [0,kMaxVolumeLevel] -> [0,MaxSpeakerVolume]
- if (_audioDevicePtr->MaxSpeakerVolume(&maxVol) != 0)
- {
- _engineStatistics.SetLastError(
- VE_MIC_VOL_ERROR, kTraceError,
- "SetSpeakerVolume() failed to get max volume");
- return -1;
- }
- // Round the value and avoid floating computation.
- spkrVol = (WebRtc_UWord32)((volume * maxVol +
- (int)(kMaxVolumeLevel / 2)) / (kMaxVolumeLevel));
-
- // set the actual volume using the audio mixer
- if (_audioDevicePtr->SetSpeakerVolume(spkrVol) != 0)
- {
- _engineStatistics.SetLastError(
- VE_MIC_VOL_ERROR, kTraceError,
- "SetSpeakerVolume() failed to set speaker volume");
- return -1;
- }
- return 0;
-}
-
-int VoEVolumeControlImpl::GetSpeakerVolume(unsigned int& volume)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetSpeakerVolume()");
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- WebRtc_UWord32 spkrVol(0);
- WebRtc_UWord32 maxVol(0);
-
- if (_audioDevicePtr->SpeakerVolume(&spkrVol) != 0)
- {
- _engineStatistics.SetLastError(
- VE_GET_MIC_VOL_ERROR, kTraceError,
- "GetSpeakerVolume() unable to get speaker volume");
- return -1;
- }
-
- // scale: [0, MaxSpeakerVolume] -> [0, kMaxVolumeLevel]
- if (_audioDevicePtr->MaxSpeakerVolume(&maxVol) != 0)
- {
- _engineStatistics.SetLastError(
- VE_GET_MIC_VOL_ERROR, kTraceError,
- "GetSpeakerVolume() unable to get max speaker volume");
- return -1;
- }
- // Round the value and avoid floating computation.
- volume = (WebRtc_UWord32) ((spkrVol * kMaxVolumeLevel +
- (int)(maxVol / 2)) / (maxVol));
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetSpeakerVolume() => volume=%d", volume);
- return 0;
-}
-
-int VoEVolumeControlImpl::SetSystemOutputMute(bool enable)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetSystemOutputMute(enabled=%d)", enable);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- if (_audioDevicePtr->SetSpeakerMute(enable) != 0)
- {
- _engineStatistics.SetLastError(
- VE_GET_MIC_VOL_ERROR, kTraceError,
- "SpeakerMute() unable to Set speaker mute");
- return -1;
- }
-
- return 0;
-}
-
-int VoEVolumeControlImpl::GetSystemOutputMute(bool& enabled)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetSystemOutputMute(enabled=?)");
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- if (_audioDevicePtr->SpeakerMute(&enabled) != 0)
- {
- _engineStatistics.SetLastError(
- VE_GET_MIC_VOL_ERROR, kTraceError,
- "SpeakerMute() unable to get speaker mute state");
- return -1;
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetSystemOutputMute() => %d", enabled);
- return 0;
-}
-
-int VoEVolumeControlImpl::SetMicVolume(unsigned int volume)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetMicVolume(volume=%u)", volume);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (volume > kMaxVolumeLevel)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetMicVolume() invalid argument");
- return -1;
- }
-
- WebRtc_UWord32 maxVol(0);
- WebRtc_UWord32 micVol(0);
-
- // scale: [0, kMaxVolumeLevel] -> [0,MaxMicrophoneVolume]
- if (_audioDevicePtr->MaxMicrophoneVolume(&maxVol) != 0)
- {
- _engineStatistics.SetLastError(
- VE_MIC_VOL_ERROR, kTraceError,
- "SetMicVolume() failed to get max volume");
- return -1;
- }
-
- if (volume == kMaxVolumeLevel) {
- // On Linux running pulse, users are able to set the volume above 100%
- // through the volume control panel, where the +100% range is digital
- // scaling. WebRTC does not support setting the volume above 100%, and
- // simply ignores changing the volume if the user tries to set it to
- // |kMaxVolumeLevel| while the current volume is higher than |maxVol|.
- if (_audioDevicePtr->MicrophoneVolume(&micVol) != 0) {
- _engineStatistics.SetLastError(
- VE_GET_MIC_VOL_ERROR, kTraceError,
- "SetMicVolume() unable to get microphone volume");
- return -1;
- }
- if (micVol >= maxVol)
- return 0;
- }
-
- // Round the value and avoid floating point computation.
- micVol = (WebRtc_UWord32) ((volume * maxVol +
- (int)(kMaxVolumeLevel / 2)) / (kMaxVolumeLevel));
-
- // set the actual volume using the audio mixer
- if (_audioDevicePtr->SetMicrophoneVolume(micVol) != 0)
- {
- _engineStatistics.SetLastError(
- VE_MIC_VOL_ERROR, kTraceError,
- "SetMicVolume() failed to set mic volume");
- return -1;
- }
- return 0;
-}
-
-int VoEVolumeControlImpl::GetMicVolume(unsigned int& volume)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetMicVolume()");
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- WebRtc_UWord32 micVol(0);
- WebRtc_UWord32 maxVol(0);
-
- if (_audioDevicePtr->MicrophoneVolume(&micVol) != 0)
- {
- _engineStatistics.SetLastError(
- VE_GET_MIC_VOL_ERROR, kTraceError,
- "GetMicVolume() unable to get microphone volume");
- return -1;
- }
-
- // scale: [0, MaxMicrophoneVolume] -> [0, kMaxVolumeLevel]
- if (_audioDevicePtr->MaxMicrophoneVolume(&maxVol) != 0)
- {
- _engineStatistics.SetLastError(
- VE_GET_MIC_VOL_ERROR, kTraceError,
- "GetMicVolume() unable to get max microphone volume");
- return -1;
- }
- if (micVol < maxVol) {
- // Round the value and avoid floating point calculation.
- volume = (WebRtc_UWord32) ((micVol * kMaxVolumeLevel +
- (int)(maxVol / 2)) / (maxVol));
- } else {
- // Truncate the value to the kMaxVolumeLevel.
- volume = kMaxVolumeLevel;
- }
-
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetMicVolume() => volume=%d", volume);
- return 0;
-}
-
-int VoEVolumeControlImpl::SetInputMute(int channel, bool enable)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetInputMute(channel=%d, enable=%d)", channel, enable);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (channel == -1)
- {
- // Mute before demultiplexing <=> affects all channels
- return _transmitMixerPtr->SetMute(enable);
- }
- else
- {
- // Mute after demultiplexing <=> affects one channel only
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetInputMute() failed to locate channel");
- return -1;
- }
- return channelPtr->SetMute(enable);
- }
- return 0;
-}
-
-int VoEVolumeControlImpl::GetInputMute(int channel, bool& enabled)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetInputMute(channel=%d)", channel);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (channel == -1)
- {
- enabled = _transmitMixerPtr->Mute();
- }
- else
- {
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetInputMute() failed to locate channel");
- return -1;
- }
- enabled = channelPtr->Mute();
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetInputMute() => enabled = %d", (int)enabled);
- return 0;
-}
-
-int VoEVolumeControlImpl::SetSystemInputMute(bool enable)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetSystemInputMute(enabled=%d)", enable);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- if (_audioDevicePtr->SetMicrophoneMute(enable) != 0)
- {
- _engineStatistics.SetLastError(
- VE_GET_MIC_VOL_ERROR, kTraceError,
- "MicrophoneMute() unable to set microphone mute state");
- return -1;
- }
-
- return 0;
-}
-
-int VoEVolumeControlImpl::GetSystemInputMute(bool& enabled)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetSystemInputMute(enabled=?)");
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- if (_audioDevicePtr->MicrophoneMute(&enabled) != 0)
- {
- _engineStatistics.SetLastError(
- VE_GET_MIC_VOL_ERROR, kTraceError,
- "MicrophoneMute() unable to get microphone mute state");
- return -1;
- }
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetSystemInputMute() => %d", enabled);
- return 0;
-}
-
-int VoEVolumeControlImpl::GetSpeechInputLevel(unsigned int& level)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetSpeechInputLevel()");
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- WebRtc_Word8 currentLevel = _transmitMixerPtr->AudioLevel();
- level = static_cast<unsigned int> (currentLevel);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetSpeechInputLevel() => %d", level);
- return 0;
-}
-
-int VoEVolumeControlImpl::GetSpeechOutputLevel(int channel,
- unsigned int& level)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetSpeechOutputLevel(channel=%d, level=?)", channel);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (channel == -1)
- {
- return _outputMixerPtr->GetSpeechOutputLevel((WebRtc_UWord32&)level);
- }
- else
- {
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetSpeechOutputLevel() failed to locate channel");
- return -1;
- }
- channelPtr->GetSpeechOutputLevel((WebRtc_UWord32&)level);
- }
- return 0;
-}
-
-int VoEVolumeControlImpl::GetSpeechInputLevelFullRange(unsigned int& level)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetSpeechInputLevelFullRange(level=?)");
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- WebRtc_Word16 currentLevel = _transmitMixerPtr->AudioLevelFullRange();
- level = static_cast<unsigned int> (currentLevel);
- WEBRTC_TRACE(kTraceStateInfo, kTraceVoice, VoEId(_instanceId,-1),
- "GetSpeechInputLevelFullRange() => %d", level);
- return 0;
-}
-
-int VoEVolumeControlImpl::GetSpeechOutputLevelFullRange(int channel,
- unsigned int& level)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetSpeechOutputLevelFullRange(channel=%d, level=?)", channel);
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (channel == -1)
- {
- return _outputMixerPtr->GetSpeechOutputLevelFullRange(
- (WebRtc_UWord32&)level);
- }
- else
- {
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetSpeechOutputLevelFullRange() failed to locate channel");
- return -1;
- }
- channelPtr->GetSpeechOutputLevelFullRange((WebRtc_UWord32&)level);
- }
- return 0;
-}
-
-int VoEVolumeControlImpl::SetChannelOutputVolumeScaling(int channel,
- float scaling)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetChannelOutputVolumeScaling(channel=%d, scaling=%3.2f)",
- channel, scaling);
- IPHONE_NOT_SUPPORTED();
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- if (scaling < kMinOutputVolumeScaling ||
- scaling > kMaxOutputVolumeScaling)
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetChannelOutputVolumeScaling() invalid parameter");
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetChannelOutputVolumeScaling() failed to locate channel");
- return -1;
- }
- return channelPtr->SetChannelOutputVolumeScaling(scaling);
-}
-
-int VoEVolumeControlImpl::GetChannelOutputVolumeScaling(int channel,
- float& scaling)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetChannelOutputVolumeScaling(channel=%d, scaling=?)", channel);
- IPHONE_NOT_SUPPORTED();
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetChannelOutputVolumeScaling() failed to locate channel");
- return -1;
- }
- return channelPtr->GetChannelOutputVolumeScaling(scaling);
-}
-
-int VoEVolumeControlImpl::SetOutputVolumePan(int channel,
- float left,
- float right)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "SetOutputVolumePan(channel=%d, left=%2.1f, right=%2.1f)",
- channel, left, right);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- bool available(false);
- _audioDevicePtr->StereoPlayoutIsAvailable(&available);
- if (!available)
- {
- _engineStatistics.SetLastError(
- VE_FUNC_NO_STEREO, kTraceError,
- "SetOutputVolumePan() stereo playout not supported");
- return -1;
- }
- if ((left < kMinOutputVolumePanning) ||
- (left > kMaxOutputVolumePanning) ||
- (right < kMinOutputVolumePanning) ||
- (right > kMaxOutputVolumePanning))
- {
- _engineStatistics.SetLastError(
- VE_INVALID_ARGUMENT, kTraceError,
- "SetOutputVolumePan() invalid parameter");
- return -1;
- }
-
- if (channel == -1)
- {
- // Master balance (affectes the signal after output mixing)
- return _outputMixerPtr->SetOutputVolumePan(left, right);
- }
- else
- {
- // Per-channel balance (affects the signal before output mixing)
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "SetOutputVolumePan() failed to locate channel");
- return -1;
- }
- return channelPtr->SetOutputVolumePan(left, right);
- }
- return 0;
-}
-
-int VoEVolumeControlImpl::GetOutputVolumePan(int channel,
- float& left,
- float& right)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, VoEId(_instanceId,-1),
- "GetOutputVolumePan(channel=%d, left=?, right=?)", channel);
- ANDROID_NOT_SUPPORTED(_engineStatistics);
- IPHONE_NOT_SUPPORTED();
-
- if (!_engineStatistics.Initialized())
- {
- _engineStatistics.SetLastError(VE_NOT_INITED, kTraceError);
- return -1;
- }
-
- bool available(false);
- _audioDevicePtr->StereoPlayoutIsAvailable(&available);
- if (!available)
- {
- _engineStatistics.SetLastError(
- VE_FUNC_NO_STEREO, kTraceError,
- "GetOutputVolumePan() stereo playout not supported");
- return -1;
- }
-
- if (channel == -1)
- {
- return _outputMixerPtr->GetOutputVolumePan(left, right);
- }
- else
- {
- voe::ScopedChannel sc(_channelManager, channel);
- voe::Channel* channelPtr = sc.ChannelPtr();
- if (channelPtr == NULL)
- {
- _engineStatistics.SetLastError(
- VE_CHANNEL_NOT_VALID, kTraceError,
- "GetOutputVolumePan() failed to locate channel");
- return -1;
- }
- return channelPtr->GetOutputVolumePan(left, right);
- }
- return 0;
-}
-
-#endif // #ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
-
-} // namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voe_volume_control_impl.h b/trunk/src/voice_engine/main/source/voe_volume_control_impl.h
deleted file mode 100644
index 18f4a1b..0000000
--- a/trunk/src/voice_engine/main/source/voe_volume_control_impl.h
+++ /dev/null
@@ -1,74 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_VOLUME_CONTROL_IMPL_H
-#define WEBRTC_VOICE_ENGINE_VOE_VOLUME_CONTROL_IMPL_H
-
-#include "voe_volume_control.h"
-
-#include "ref_count.h"
-#include "shared_data.h"
-
-namespace webrtc {
-
-class VoEVolumeControlImpl : public virtual voe::SharedData,
- public VoEVolumeControl,
- public voe::RefCount
-{
-public:
- virtual int Release();
-
- virtual int SetSpeakerVolume(unsigned int volume);
-
- virtual int GetSpeakerVolume(unsigned int& volume);
-
- virtual int SetSystemOutputMute(bool enable);
-
- virtual int GetSystemOutputMute(bool& enabled);
-
- virtual int SetMicVolume(unsigned int volume);
-
- virtual int GetMicVolume(unsigned int& volume);
-
- virtual int SetInputMute(int channel, bool enable);
-
- virtual int GetInputMute(int channel, bool& enabled);
-
- virtual int SetSystemInputMute(bool enable);
-
- virtual int GetSystemInputMute(bool& enabled);
-
- virtual int GetSpeechInputLevel(unsigned int& level);
-
- virtual int GetSpeechOutputLevel(int channel, unsigned int& level);
-
- virtual int GetSpeechInputLevelFullRange(unsigned int& level);
-
- virtual int GetSpeechOutputLevelFullRange(int channel,
- unsigned int& level);
-
- virtual int SetChannelOutputVolumeScaling(int channel, float scaling);
-
- virtual int GetChannelOutputVolumeScaling(int channel, float& scaling);
-
- virtual int SetOutputVolumePan(int channel, float left, float right);
-
- virtual int GetOutputVolumePan(int channel, float& left, float& right);
-
-
-protected:
- VoEVolumeControlImpl();
- virtual ~VoEVolumeControlImpl();
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_VOLUME_CONTROL_IMPL_H
-
diff --git a/trunk/src/voice_engine/main/source/voice_engine_core.gypi b/trunk/src/voice_engine/main/source/voice_engine_core.gypi
deleted file mode 100644
index 135d78b..0000000
--- a/trunk/src/voice_engine/main/source/voice_engine_core.gypi
+++ /dev/null
@@ -1,158 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- {
- 'target_name': 'voice_engine_core',
- 'type': '<(library)',
- 'dependencies': [
- '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
- '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
- '<(webrtc_root)/modules/modules.gyp:audio_device',
- '<(webrtc_root)/modules/modules.gyp:audio_processing',
- '<(webrtc_root)/modules/modules.gyp:media_file',
- '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
- '<(webrtc_root)/modules/modules.gyp:udp_transport',
- '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- '../../..',
- '../interface',
- '<(webrtc_root)/modules/audio_device/main/source',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '../../..',
- '../interface',
- ],
- },
- 'sources': [
- '../../../common_types.h',
- '../../../engine_configurations.h',
- '../../../typedefs.h',
- '../interface/voe_audio_processing.h',
- '../interface/voe_base.h',
- '../interface/voe_call_report.h',
- '../interface/voe_codec.h',
- '../interface/voe_dtmf.h',
- '../interface/voe_encryption.h',
- '../interface/voe_errors.h',
- '../interface/voe_external_media.h',
- '../interface/voe_file.h',
- '../interface/voe_hardware.h',
- '../interface/voe_neteq_stats.h',
- '../interface/voe_network.h',
- '../interface/voe_rtp_rtcp.h',
- '../interface/voe_video_sync.h',
- '../interface/voe_volume_control.h',
- 'audio_frame_operations.cc',
- 'audio_frame_operations.h',
- 'channel.cc',
- 'channel.h',
- 'channel_manager.cc',
- 'channel_manager.h',
- 'channel_manager_base.cc',
- 'channel_manager_base.h',
- 'dtmf_inband.cc',
- 'dtmf_inband.h',
- 'dtmf_inband_queue.cc',
- 'dtmf_inband_queue.h',
- 'level_indicator.cc',
- 'level_indicator.h',
- 'monitor_module.cc',
- 'monitor_module.h',
- 'output_mixer.cc',
- 'output_mixer.h',
- 'ref_count.cc',
- 'ref_count.h',
- 'shared_data.cc',
- 'shared_data.h',
- 'statistics.cc',
- 'statistics.h',
- 'transmit_mixer.cc',
- 'transmit_mixer.h',
- 'utility.cc',
- 'utility.h',
- 'voe_audio_processing_impl.cc',
- 'voe_audio_processing_impl.h',
- 'voe_base_impl.cc',
- 'voe_base_impl.h',
- 'voe_call_report_impl.cc',
- 'voe_call_report_impl.h',
- 'voe_codec_impl.cc',
- 'voe_codec_impl.h',
- 'voe_dtmf_impl.cc',
- 'voe_dtmf_impl.h',
- 'voe_encryption_impl.cc',
- 'voe_encryption_impl.h',
- 'voe_external_media_impl.cc',
- 'voe_external_media_impl.h',
- 'voe_file_impl.cc',
- 'voe_file_impl.h',
- 'voe_hardware_impl.cc',
- 'voe_hardware_impl.h',
- 'voe_neteq_stats_impl.cc',
- 'voe_neteq_stats_impl.h',
- 'voe_network_impl.cc',
- 'voe_network_impl.h',
- 'voe_rtp_rtcp_impl.cc',
- 'voe_rtp_rtcp_impl.h',
- 'voe_video_sync_impl.cc',
- 'voe_video_sync_impl.h',
- 'voe_volume_control_impl.cc',
- 'voe_volume_control_impl.h',
- 'voice_engine_defines.h',
- 'voice_engine_impl.cc',
- 'voice_engine_impl.h',
- ],
- },
- ],
- 'conditions': [
- ['build_with_chromium==0', {
- 'targets': [
- {
- 'target_name': 'voice_engine_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'voice_engine_core',
- '<(webrtc_root)/common_audio/common_audio.gyp:resampler',
- '<(webrtc_root)/common_audio/common_audio.gyp:signal_processing',
- '<(webrtc_root)/modules/modules.gyp:audio_coding_module',
- '<(webrtc_root)/modules/modules.gyp:audio_conference_mixer',
- '<(webrtc_root)/modules/modules.gyp:audio_device',
- '<(webrtc_root)/modules/modules.gyp:audio_processing',
- '<(webrtc_root)/modules/modules.gyp:media_file',
- '<(webrtc_root)/modules/modules.gyp:rtp_rtcp',
- '<(webrtc_root)/modules/modules.gyp:udp_transport',
- '<(webrtc_root)/modules/modules.gyp:webrtc_utility',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'include_dirs': [
- '../../..',
- '../interface',
- ],
- 'sources': [
- 'channel_unittest.cc',
- ],
- },
- ], # targets
- }], # build_with_chromium
- ], # conditions
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/voice_engine/main/source/voice_engine_defines.h b/trunk/src/voice_engine/main/source/voice_engine_defines.h
deleted file mode 100644
index c652e6a..0000000
--- a/trunk/src/voice_engine/main/source/voice_engine_defines.h
+++ /dev/null
@@ -1,597 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * This file contains common constants for VoiceEngine, as well as
- * platform specific settings and include files.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H
-#define WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H
-
-#include "engine_configurations.h"
-
-// ----------------------------------------------------------------------------
-// Enumerators
-// ----------------------------------------------------------------------------
-
-namespace webrtc
-{
-
-// VolumeControl
-enum { kMinVolumeLevel = 0 };
-enum { kMaxVolumeLevel = 255 };
-// Min scale factor for per-channel volume scaling
-const float kMinOutputVolumeScaling = 0.0f;
-// Max scale factor for per-channel volume scaling
-const float kMaxOutputVolumeScaling = 10.0f;
-// Min scale factor for output volume panning
-const float kMinOutputVolumePanning = 0.0f;
-// Max scale factor for output volume panning
-const float kMaxOutputVolumePanning = 1.0f;
-
-// DTMF
-enum { kMinDtmfEventCode = 0 }; // DTMF digit "0"
-enum { kMaxDtmfEventCode = 15 }; // DTMF digit "D"
-enum { kMinTelephoneEventCode = 0 }; // RFC4733 (Section 2.3.1)
-enum { kMaxTelephoneEventCode = 255 }; // RFC4733 (Section 2.3.1)
-enum { kMinTelephoneEventDuration = 100 };
-enum { kMaxTelephoneEventDuration = 60000 }; // Actual limit is 2^16
-enum { kMinTelephoneEventAttenuation = 0 }; // 0 dBm0
-enum { kMaxTelephoneEventAttenuation = 36 }; // -36 dBm0
-enum { kMinTelephoneEventSeparationMs = 100 }; // Min delta time between two
- // telephone events
-
-enum { EcAec = 0 }; // AEC mode
-enum { EcAecm = 1 }; // AECM mode
-
-enum { kVoiceEngineMaxIpPacketSizeBytes = 1500 }; // assumes Ethernet
-
-enum { kVoiceEngineMaxModuleVersionSize = 960 };
-
-// Base
-enum { kVoiceEngineVersionMaxMessageSize = 1024 };
-
-// Encryption
-// SRTP uses 30 bytes key length
-enum { kVoiceEngineMaxSrtpKeyLength = 30 };
-// SRTP minimum key/tag length for encryption level
-enum { kVoiceEngineMinSrtpEncryptLength = 16 };
-// SRTP maximum key/tag length for encryption level
-enum { kVoiceEngineMaxSrtpEncryptLength = 256 };
-// SRTP maximum key/tag length for authentication level,
-// HMAC SHA1 authentication type
-enum { kVoiceEngineMaxSrtpAuthSha1Length = 20 };
-// SRTP maximum tag length for authentication level,
-// null authentication type
-enum { kVoiceEngineMaxSrtpTagAuthNullLength = 12 };
-// SRTP maximum key length for authentication level,
-// null authentication type
-enum { kVoiceEngineMaxSrtpKeyAuthNullLength = 256 };
-
-// Audio processing
-enum { kVoiceEngineAudioProcessingDeviceSampleRateHz = 48000 };
-
-// Codec
-// Min init target rate for iSAC-wb
-enum { kVoiceEngineMinIsacInitTargetRateBpsWb = 10000 };
-// Max init target rate for iSAC-wb
-enum { kVoiceEngineMaxIsacInitTargetRateBpsWb = 32000 };
-// Min init target rate for iSAC-swb
-enum { kVoiceEngineMinIsacInitTargetRateBpsSwb = 10000 };
-// Max init target rate for iSAC-swb
-enum { kVoiceEngineMaxIsacInitTargetRateBpsSwb = 56000 };
-// Lowest max rate for iSAC-wb
-enum { kVoiceEngineMinIsacMaxRateBpsWb = 32000 };
-// Highest max rate for iSAC-wb
-enum { kVoiceEngineMaxIsacMaxRateBpsWb = 53400 };
-// Lowest max rate for iSAC-swb
-enum { kVoiceEngineMinIsacMaxRateBpsSwb = 32000 };
-// Highest max rate for iSAC-swb
-enum { kVoiceEngineMaxIsacMaxRateBpsSwb = 107000 };
-// Lowest max payload size for iSAC-wb
-enum { kVoiceEngineMinIsacMaxPayloadSizeBytesWb = 120 };
-// Highest max payload size for iSAC-wb
-enum { kVoiceEngineMaxIsacMaxPayloadSizeBytesWb = 400 };
-// Lowest max payload size for iSAC-swb
-enum { kVoiceEngineMinIsacMaxPayloadSizeBytesSwb = 120 };
-// Highest max payload size for iSAC-swb
-enum { kVoiceEngineMaxIsacMaxPayloadSizeBytesSwb = 600 };
-
-// VideoSync
-// Lowest minimum playout delay
-enum { kVoiceEngineMinMinPlayoutDelayMs = 0 };
-// Highest minimum playout delay
-enum { kVoiceEngineMaxMinPlayoutDelayMs = 1000 };
-
-// Network
-// Min packet-timeout time for received RTP packets
-enum { kVoiceEngineMinPacketTimeoutSec = 1 };
-// Max packet-timeout time for received RTP packets
-enum { kVoiceEngineMaxPacketTimeoutSec = 150 };
-// Min sample time for dead-or-alive detection
-enum { kVoiceEngineMinSampleTimeSec = 1 };
-// Max sample time for dead-or-alive detection
-enum { kVoiceEngineMaxSampleTimeSec = 150 };
-
-// RTP/RTCP
-// Min 4-bit ID for RTP extension (see section 4.2 in RFC 5285)
-enum { kVoiceEngineMinRtpExtensionId = 1 };
-// Max 4-bit ID for RTP extension
-enum { kVoiceEngineMaxRtpExtensionId = 14 };
-
-} // namespace webrtc
-
-// TODO(andrew): we shouldn't be using the precompiler for this.
-// Use enums or bools as appropriate.
-#define WEBRTC_AUDIO_PROCESSING_OFF false
-
-#define WEBRTC_VOICE_ENGINE_HP_DEFAULT_STATE true
- // AudioProcessing HP is ON
-#define WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
- // AudioProcessing NS off
-#define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE true
- // AudioProcessing AGC on
-#define WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
- // AudioProcessing EC off
-#define WEBRTC_VOICE_ENGINE_VAD_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
- // AudioProcessing off
-#define WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
- // AudioProcessing RX AGC off
-#define WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
- // AudioProcessing RX NS off
-#define WEBRTC_VOICE_ENGINE_RX_HP_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
- // AudioProcessing RX High Pass Filter off
-
-#define WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE NoiseSuppression::kModerate
- // AudioProcessing NS moderate suppression
-#define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE GainControl::kAdaptiveAnalog
- // AudioProcessing AGC analog digital combined
-#define WEBRTC_VOICE_ENGINE_EC_DEFAULT_MODE EcAec
- // AudioProcessing EC AEC
-#define WEBRTC_VOICE_ENGINE_RX_AGC_DEFAULT_MODE GainControl::kAdaptiveDigital
- // AudioProcessing AGC mode
-#define WEBRTC_VOICE_ENGINE_RX_NS_DEFAULT_MODE NoiseSuppression::kModerate
- // AudioProcessing RX NS mode
-
-// Macros
-// Comparison of two strings without regard to case
-#define STR_CASE_CMP(x,y) ::_stricmp(x,y)
-// Compares characters of two strings without regard to case
-#define STR_NCASE_CMP(x,y,n) ::_strnicmp(x,y,n)
-
-// ----------------------------------------------------------------------------
-// Build information macros
-// ----------------------------------------------------------------------------
-
-#if defined(_DEBUG)
-#define BUILDMODE "d"
-#elif defined(DEBUG)
-#define BUILDMODE "d"
-#elif defined(NDEBUG)
-#define BUILDMODE "r"
-#else
-#define BUILDMODE "?"
-#endif
-
-#define BUILDTIME __TIME__
-#define BUILDDATE __DATE__
-
-// Example: "Oct 10 2002 12:05:30 r"
-#define BUILDINFO BUILDDATE " " BUILDTIME " " BUILDMODE
-
-// ----------------------------------------------------------------------------
-// Macros
-// ----------------------------------------------------------------------------
-
-#if (defined(_DEBUG) && defined(_WIN32) && (_MSC_VER >= 1400))
- #include <windows.h>
- #include <stdio.h>
- #define DEBUG_PRINT(...) \
- { \
- char msg[256]; \
- sprintf(msg, __VA_ARGS__); \
- OutputDebugStringA(msg); \
- }
-#else
- // special fix for visual 2003
- #define DEBUG_PRINT(exp) ((void)0)
-#endif // defined(_DEBUG) && defined(_WIN32)
-
-#define CHECK_CHANNEL(channel) if (CheckChannel(channel) == -1) return -1;
-
-// ----------------------------------------------------------------------------
-// Default Trace filter
-// ----------------------------------------------------------------------------
-
-#define WEBRTC_VOICE_ENGINE_DEFAULT_TRACE_FILTER \
- kTraceStateInfo | kTraceWarning | kTraceError | kTraceCritical | \
- kTraceApiCall
-
-// ----------------------------------------------------------------------------
-// Inline functions
-// ----------------------------------------------------------------------------
-
-namespace webrtc
-{
-
-inline int VoEId(const int veId, const int chId)
-{
- if (chId == -1)
- {
- const int dummyChannel(99);
- return (int) ((veId << 16) + dummyChannel);
- }
- return (int) ((veId << 16) + chId);
-}
-
-inline int VoEModuleId(const int veId, const int chId)
-{
- return (int) ((veId << 16) + chId);
-}
-
-// Convert module ID to internal VoE channel ID
-inline int VoEChannelId(const int moduleId)
-{
- return (int) (moduleId & 0xffff);
-}
-
-} // namespace webrtc
-
-// ----------------------------------------------------------------------------
-// Platform settings
-// ----------------------------------------------------------------------------
-
-// *** WINDOWS ***
-
-#if defined(_WIN32)
-
- #pragma comment( lib, "winmm.lib" )
-
- #ifndef WEBRTC_EXTERNAL_TRANSPORT
- #pragma comment( lib, "ws2_32.lib" )
- #endif
-
-// ----------------------------------------------------------------------------
-// Enumerators
-// ----------------------------------------------------------------------------
-
-namespace webrtc
-{
-// Max number of supported channels
-enum { kVoiceEngineMaxNumOfChannels = 32 };
-// Max number of channels which can be played out simultaneously
-enum { kVoiceEngineMaxNumOfActiveChannels = 16 };
-} // namespace webrtc
-
-// ----------------------------------------------------------------------------
-// Defines
-// ----------------------------------------------------------------------------
-
- #include <windows.h>
- #include <mmsystem.h> // timeGetTime
-
- #define GET_TIME_IN_MS() ::timeGetTime()
- #define SLEEP(x) ::Sleep(x)
- // Comparison of two strings without regard to case
- #define STR_CASE_CMP(x,y) ::_stricmp(x,y)
- // Compares characters of two strings without regard to case
- #define STR_NCASE_CMP(x,y,n) ::_strnicmp(x,y,n)
-
-// Default device for Windows PC
- #define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE \
- AudioDeviceModule::kDefaultCommunicationDevice
-
-#endif // #if (defined(_WIN32)
-
-// *** LINUX ***
-
-#ifdef WEBRTC_LINUX
-
-#include <pthread.h>
-#include <sys/types.h>
-#include <sys/socket.h>
-#include <netinet/in.h>
-#include <arpa/inet.h>
-#ifndef QNX
- #include <linux/net.h>
-#ifndef ANDROID
- #include <sys/soundcard.h>
-#endif // ANDROID
-#endif // QNX
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <errno.h>
-#include <sys/stat.h>
-#include <sys/ioctl.h>
-#include <unistd.h>
-#include <fcntl.h>
-#include <sched.h>
-#include <time.h>
-#include <sys/time.h>
-
-#define DWORD unsigned long int
-#define WINAPI
-#define LPVOID void *
-#define FALSE 0
-#define TRUE 1
-#define UINT unsigned int
-#define UCHAR unsigned char
-#define TCHAR char
-#ifdef QNX
-#define _stricmp stricmp
-#else
-#define _stricmp strcasecmp
-#endif
-#define GetLastError() errno
-#define WSAGetLastError() errno
-#define LPCTSTR const char*
-#define LPCSTR const char*
-#define wsprintf sprintf
-#define TEXT(a) a
-#define _ftprintf fprintf
-#define _tcslen strlen
-#define FAR
-#define __cdecl
-#define LPSOCKADDR struct sockaddr *
-
-namespace
-{
- void Sleep(unsigned long x)
- {
- timespec t;
- t.tv_sec = x/1000;
- t.tv_nsec = (x-(x/1000)*1000)*1000000;
- nanosleep(&t,NULL);
- }
-
- DWORD timeGetTime()
- {
- struct timeval tv;
- struct timezone tz;
- unsigned long val;
-
- gettimeofday(&tv, &tz);
- val= tv.tv_sec*1000+ tv.tv_usec/1000;
- return(val);
- }
-}
-
-#define SLEEP(x) ::Sleep(x)
-#define GET_TIME_IN_MS timeGetTime
-
-// Default device for Linux and Android
-#define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0
-
-#ifdef ANDROID
-
-// ----------------------------------------------------------------------------
-// Enumerators
-// ----------------------------------------------------------------------------
-
-namespace webrtc
-{
- // Max number of supported channels
- enum { kVoiceEngineMaxNumOfChannels = 2 };
- // Max number of channels which can be played out simultaneously
- enum { kVoiceEngineMaxNumOfActiveChannels = 2 };
-} // namespace webrtc
-
-// ----------------------------------------------------------------------------
-// Defines
-// ----------------------------------------------------------------------------
-
- // Always excluded for Android builds
- #undef WEBRTC_CODEC_ISAC
- #undef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
- #undef WEBRTC_CONFERENCING
- #undef WEBRTC_TYPING_DETECTION
-
- // Default audio processing states
- #undef WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE
- #undef WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE
- #undef WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE
- #define WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
- #define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
- #define WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
-
- // Default audio processing modes
- #undef WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE
- #undef WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE
- #undef WEBRTC_VOICE_ENGINE_EC_DEFAULT_MODE
- #define WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE \
- NoiseSuppression::kModerate
- #define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE \
- GainControl::kAdaptiveDigital
- #define WEBRTC_VOICE_ENGINE_EC_DEFAULT_MODE EcAecm
-
- #define ANDROID_NOT_SUPPORTED(stat) \
- stat.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError, \
- "API call not supported"); \
- return -1;
-
-#else // LINUX PC
-// ----------------------------------------------------------------------------
-// Enumerators
-// ----------------------------------------------------------------------------
-
-namespace webrtc
-{
- // Max number of supported channels
- enum { kVoiceEngineMaxNumOfChannels = 32 };
- // Max number of channels which can be played out simultaneously
- enum { kVoiceEngineMaxNumOfActiveChannels = 16 };
-} // namespace webrtc
-
-// ----------------------------------------------------------------------------
-// Defines
-// ----------------------------------------------------------------------------
-
- #define ANDROID_NOT_SUPPORTED(stat)
-
-#endif // ANDROID - LINUX PC
-
-#else
-#define ANDROID_NOT_SUPPORTED(stat)
-#endif // #ifdef WEBRTC_LINUX
-
-// *** WEBRTC_MAC ***
-// including iPhone
-
-#ifdef WEBRTC_MAC
-
-#include <pthread.h>
-#include <sys/types.h>
-#include <sys/socket.h>
-#include <netinet/in.h>
-#include <arpa/inet.h>
-#include <stdio.h>
-#include <string.h>
-#include <stdlib.h>
-#include <errno.h>
-#include <sys/stat.h>
-#include <unistd.h>
-#include <fcntl.h>
-#include <sched.h>
-#include <sys/time.h>
-#include <time.h>
-#include <AudioUnit/AudioUnit.h>
-#if !defined(MAC_IPHONE) && !defined(MAC_IPHONE_SIM)
- #include <CoreServices/CoreServices.h>
- #include <CoreAudio/CoreAudio.h>
- #include <AudioToolbox/DefaultAudioOutput.h>
- #include <AudioToolbox/AudioConverter.h>
- #include <CoreAudio/HostTime.h>
-#endif
-
-#define DWORD unsigned long int
-#define WINAPI
-#define LPVOID void *
-#define FALSE 0
-#define TRUE 1
-#define SOCKADDR_IN struct sockaddr_in
-#define UINT unsigned int
-#define UCHAR unsigned char
-#define TCHAR char
-#define _stricmp strcasecmp
-#define GetLastError() errno
-#define WSAGetLastError() errno
-#define LPCTSTR const char*
-#define wsprintf sprintf
-#define TEXT(a) a
-#define _ftprintf fprintf
-#define _tcslen strlen
-#define FAR
-#define __cdecl
-#define LPSOCKADDR struct sockaddr *
-#define LPCSTR const char*
-#define ULONG unsigned long
-
-namespace
-{
- void Sleep(unsigned long x)
- {
- timespec t;
- t.tv_sec = x/1000;
- t.tv_nsec = (x-(x/1000)*1000)*1000000;
- nanosleep(&t,NULL);
- }
-
- DWORD WebRtcTimeGetTime()
- {
- struct timeval tv;
- struct timezone tz;
- unsigned long val;
-
- gettimeofday(&tv, &tz);
- val= tv.tv_sec*1000+ tv.tv_usec/1000;
- return(val);
- }
-}
-
-#define SLEEP(x) ::Sleep(x)
-#define GET_TIME_IN_MS WebRtcTimeGetTime
-
-// Default device for Mac and iPhone
-#define WEBRTC_VOICE_ENGINE_DEFAULT_DEVICE 0
-
-// iPhone specific
-#if defined(MAC_IPHONE) || defined(MAC_IPHONE_SIM)
-
-// ----------------------------------------------------------------------------
-// Enumerators
-// ----------------------------------------------------------------------------
-
-namespace webrtc
-{
- // Max number of supported channels
- enum { kVoiceEngineMaxNumOfChannels = 2 };
- // Max number of channels which can be played out simultaneously
- enum { kVoiceEngineMaxNumOfActiveChannels = 2 };
-} // namespace webrtc
-
-// ----------------------------------------------------------------------------
-// Defines
-// ----------------------------------------------------------------------------
-
- // Always excluded for iPhone builds
- #undef WEBRTC_CODEC_ISAC
- #undef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
-
- #undef WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE
- #undef WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE
- #undef WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE
- #define WEBRTC_VOICE_ENGINE_NS_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
- #define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
- #define WEBRTC_VOICE_ENGINE_EC_DEFAULT_STATE WEBRTC_AUDIO_PROCESSING_OFF
-
- #undef WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE
- #undef WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE
- #undef WEBRTC_VOICE_ENGINE_EC_DEFAULT_MODE
- #define WEBRTC_VOICE_ENGINE_NS_DEFAULT_MODE \
- NoiseSuppression::kModerate
- #define WEBRTC_VOICE_ENGINE_AGC_DEFAULT_MODE \
- GainControl::kAdaptiveDigital
- #define WEBRTC_VOICE_ENGINE_EC_DEFAULT_MODE EcAecm
-
- #define IPHONE_NOT_SUPPORTED() \
- _engineStatistics.SetLastError(VE_FUNC_NOT_SUPPORTED, kTraceError, \
- "API call not supported"); \
- return -1;
-
-#else // Non-iPhone
-
-// ----------------------------------------------------------------------------
-// Enumerators
-// ----------------------------------------------------------------------------
-
-namespace webrtc
-{
- // Max number of supported channels
- enum { kVoiceEngineMaxNumOfChannels = 32 };
- // Max number of channels which can be played out simultaneously
- enum { kVoiceEngineMaxNumOfActiveChannels = 16 };
-} // namespace webrtc
-
-// ----------------------------------------------------------------------------
-// Defines
-// ----------------------------------------------------------------------------
-
- #define IPHONE_NOT_SUPPORTED()
-#endif
-
-#else
-#define IPHONE_NOT_SUPPORTED()
-#endif // #ifdef WEBRTC_MAC
-
-
-
-#endif // WEBRTC_VOICE_ENGINE_VOICE_ENGINE_DEFINES_H
diff --git a/trunk/src/voice_engine/main/source/voice_engine_impl.cc b/trunk/src/voice_engine/main/source/voice_engine_impl.cc
deleted file mode 100644
index 07af08e..0000000
--- a/trunk/src/voice_engine/main/source/voice_engine_impl.cc
+++ /dev/null
@@ -1,317 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "voice_engine_impl.h"
-#include "trace.h"
-
-#ifdef WEBRTC_ANDROID
-extern "C"
-{
-extern WebRtc_Word32 SetAndroidAudioDeviceObjects(
- void* javaVM, void* env, void* context);
-} // extern "C"
-#endif
-
-namespace webrtc
-{
-
-// Counter to be ensure that we can add a correct ID in all static trace
-// methods. It is not the nicest solution, especially not since we already
-// have a counter in VoEBaseImpl. In other words, there is room for
-// improvement here.
-static WebRtc_Word32 gVoiceEngineInstanceCounter = 0;
-
-extern "C"
-{
-WEBRTC_DLLEXPORT VoiceEngine* GetVoiceEngine();
-
-VoiceEngine* GetVoiceEngine()
-{
- VoiceEngineImpl* self = new VoiceEngineImpl();
- VoiceEngine* ve = reinterpret_cast<VoiceEngine*> (self);
- if (ve != NULL)
- {
- gVoiceEngineInstanceCounter++;
- }
- return ve;
-}
-} // extern "C"
-
-VoiceEngine* VoiceEngine::Create()
-{
-#if (defined _WIN32)
- HMODULE hmod_ = LoadLibrary(TEXT("VoiceEngineTestingDynamic.dll"));
-
- if (hmod_)
- {
- typedef VoiceEngine* (*PfnGetVoiceEngine)(void);
- PfnGetVoiceEngine pfn = (PfnGetVoiceEngine)GetProcAddress(
- hmod_,"GetVoiceEngine");
- if (pfn)
- {
- VoiceEngine* self = pfn();
- return (self);
- }
- }
-#endif
-
- return GetVoiceEngine();
-}
-
-int VoiceEngine::SetTraceFilter(const unsigned int filter)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
- VoEId(gVoiceEngineInstanceCounter, -1),
- "SetTraceFilter(filter=0x%x)", filter);
-
- // Remember old filter
- WebRtc_UWord32 oldFilter = 0;
- Trace::LevelFilter(oldFilter);
-
- // Set new filter
- WebRtc_Word32 ret = Trace::SetLevelFilter(filter);
-
- // If previous log was ignored, log again after changing filter
- if (kTraceNone == oldFilter)
- {
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, -1,
- "SetTraceFilter(filter=0x%x)", filter);
- }
-
- return (ret);
-}
-
-int VoiceEngine::SetTraceFile(const char* fileNameUTF8,
- const bool addFileCounter)
-{
- int ret = Trace::SetTraceFile(fileNameUTF8, addFileCounter);
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
- VoEId(gVoiceEngineInstanceCounter, -1),
- "SetTraceFile(fileNameUTF8=%s, addFileCounter=%d)",
- fileNameUTF8, addFileCounter);
- return (ret);
-}
-
-int VoiceEngine::SetTraceCallback(TraceCallback* callback)
-{
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice,
- VoEId(gVoiceEngineInstanceCounter, -1),
- "SetTraceCallback(callback=0x%x)", callback);
- return (Trace::SetTraceCallback(callback));
-}
-
-bool VoiceEngine::Delete(VoiceEngine*& voiceEngine, bool ignoreRefCounters)
-{
- if (voiceEngine == NULL)
- {
- return false;
- }
-
- VoiceEngineImpl* s = reinterpret_cast<VoiceEngineImpl*> (voiceEngine);
- VoEBaseImpl* base = s;
-
- WEBRTC_TRACE(kTraceApiCall, kTraceVoice, -1,
- "VoiceEngine::Delete(voiceEngine=0x%p, ignoreRefCounters=%d)",
- voiceEngine, ignoreRefCounters);
-
- if (!ignoreRefCounters)
- {
- if (base->GetCount() != 0)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
- "VoEBase reference counter is %d => memory will not "
- "be released properly!", base->GetCount());
- return false;
- }
-#ifdef WEBRTC_VOICE_ENGINE_CODEC_API
- VoECodecImpl* codec = s;
- if (codec->GetCount() != 0)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
- "VoECodec reference counter is %d => memory will not "
- "be released properly!", codec->GetCount());
- return false;
- }
-#endif
-
-#ifdef WEBRTC_VOICE_ENGINE_DTMF_API
- VoEDtmfImpl* dtmf = s;
- if (dtmf->GetCount() != 0)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
- "VoEDtmf reference counter is %d =>"
- "memory will not be released properly!",
- dtmf->GetCount());
- return false;
- }
-#endif
-
-#ifdef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
- VoEEncryptionImpl* encrypt = s;
- if (encrypt->GetCount() != 0)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
- "VoEEncryption reference counter is %d => "
- "memory will not be released properly!",
- encrypt->GetCount());
- return false;
- }
-#endif
-
-#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
- VoEExternalMediaImpl* extmedia = s;
- if (extmedia->GetCount() != 0)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
- "VoEExternalMedia reference counter is %d => "
- "memory will not be released properly!",
- extmedia->GetCount());
- return false;
- }
-#endif
-
-#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
- VoECallReportImpl* report = s;
- if (report->GetCount() != 0)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
- "VoECallReport reference counter is %d => memory "
- "will not be released properly!",
- report->GetCount());
- return false;
- }
-#endif
-
-#ifdef WEBRTC_VOICE_ENGINE_FILE_API
- VoEFileImpl* file = s;
- if (file->GetCount() != 0)
- {
- WEBRTC_TRACE(
- kTraceCritical,
- kTraceVoice,
- -1,
- "VoEFile reference counter is %d => memory will not "
- "be released properly!",
- file->GetCount());
- return false;
- }
-#endif
-
-#ifdef WEBRTC_VOICE_ENGINE_HARDWARE_API
- VoEHardwareImpl* hware = s;
- if (hware->GetCount() != 0)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
- "VoEHardware reference counter is %d => memory will "
- "not be released properly!", hware->GetCount());
- return false;
- }
-#endif
-
-#ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
- VoENetEqStatsImpl* neteqst = s;
- if (neteqst->GetCount() != 0)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
- "VoENetEqStats reference counter is %d => "
- "memory will not be released properly!",
- neteqst->GetCount());
- return false;
- }
-#endif
-
-#ifdef WEBRTC_VOICE_ENGINE_NETWORK_API
- VoENetworkImpl* netw = s;
- if (netw->GetCount() != 0)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
- "VoENetworkImpl reference counter is %d => memory "
- "will not be released properly!", netw->GetCount());
- return false;
- }
-#endif
-
-#ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
- VoERTP_RTCPImpl* rtcp = s;
- if (rtcp->GetCount() != 0)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
- "VoERTP_RTCP reference counter is %d =>"
- "memory will not be released properly!",
- rtcp->GetCount());
- return false;
- }
-#endif
-
-#ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
- VoEVideoSyncImpl* vsync = s;
- if (vsync->GetCount() != 0)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
- "VoEVideoSync reference counter is %d => "
- "memory will not be released properly!",
- vsync->GetCount());
- return false;
- }
-#endif
-
-#ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
- VoEVolumeControlImpl* volume = s;
- if (volume->GetCount() != 0)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
- "VoEVolumeControl reference counter is %d =>"
- "memory will not be released properly!",
- volume->GetCount());
- return false;
- }
-#endif
-
-#ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
- VoEAudioProcessingImpl* apm = s;
- if (apm->GetCount() != 0)
- {
- WEBRTC_TRACE(kTraceCritical, kTraceVoice, -1,
- "VoEAudioProcessing reference counter is %d => "
- "memory will not be released properly!",
- apm->GetCount());
- return false;
- }
-#endif
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, -1,
- "all reference counters are zero => deleting the "
- "VoiceEngine instance...");
-
- } // if (!ignoreRefCounters)
- else
- {
- WEBRTC_TRACE(kTraceInfo, kTraceVoice, -1,
- "reference counters are ignored => deleting the "
- "VoiceEngine instance...");
- }
-
- delete s;
- voiceEngine = NULL;
-
- return true;
-}
-
-int VoiceEngine::SetAndroidObjects(void* javaVM, void* env, void* context)
-{
-#if defined(WEBRTC_ANDROID) && !defined(WEBRTC_ANDROID_OPENSLES)
- // modules/audio_device/main/source/android/audio_device_android_jni.cc
- // contains the actual implementation.
- return SetAndroidAudioDeviceObjects(javaVM, env, context);
-#else
- return -1;
-#endif
-}
-
-} //namespace webrtc
diff --git a/trunk/src/voice_engine/main/source/voice_engine_impl.h b/trunk/src/voice_engine/main/source/voice_engine_impl.h
deleted file mode 100644
index 708ac5d..0000000
--- a/trunk/src/voice_engine/main/source/voice_engine_impl.h
+++ /dev/null
@@ -1,118 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOICE_ENGINE_IMPL_H
-#define WEBRTC_VOICE_ENGINE_VOICE_ENGINE_IMPL_H
-
-#include "engine_configurations.h"
-#include "voe_base_impl.h"
-
-#ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
-#include "voe_audio_processing_impl.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
-#include "voe_call_report_impl.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_CODEC_API
-#include "voe_codec_impl.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_DTMF_API
-#include "voe_dtmf_impl.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
-#include "voe_encryption_impl.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
-#include "voe_external_media_impl.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_FILE_API
-#include "voe_file_impl.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_HARDWARE_API
-#include "voe_hardware_impl.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
-#include "voe_neteq_stats_impl.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_NETWORK_API
-#include "voe_network_impl.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
-#include "voe_rtp_rtcp_impl.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
-#include "voe_video_sync_impl.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
-#include "voe_volume_control_impl.h"
-#endif
-
-namespace webrtc
-{
-
-class VoiceEngineImpl :
-#ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
- public VoEAudioProcessingImpl,
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
- public VoECallReportImpl,
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_CODEC_API
- public VoECodecImpl,
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_DTMF_API
- public VoEDtmfImpl,
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
- public VoEEncryptionImpl,
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
- public VoEExternalMediaImpl,
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_FILE_API
- public VoEFileImpl,
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_HARDWARE_API
- public VoEHardwareImpl,
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
- public VoENetEqStatsImpl,
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_NETWORK_API
- public VoENetworkImpl,
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
- public VoERTP_RTCPImpl,
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
- public VoEVideoSyncImpl,
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
- public VoEVolumeControlImpl,
-#endif
- public VoEBaseImpl
-{
-public:
- VoiceEngineImpl() :
-#ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
- VoEVideoSyncImpl(this),
-#endif
- VoEBaseImpl() // Included in initializer list to satisfy condition when
- // none of the WEBRTC_VOICE_XXX defines are set.
- {
- }
- virtual ~VoiceEngineImpl()
- {
- }
-};
-
-} // namespace webrtc
-
-#endif // WEBRTC_VOICE_ENGINE_VOICE_ENGINE_IMPL_H
diff --git a/trunk/src/voice_engine/main/test/android/android_test/.classpath b/trunk/src/voice_engine/main/test/android/android_test/.classpath
deleted file mode 100644
index 86a15c9..0000000
--- a/trunk/src/voice_engine/main/test/android/android_test/.classpath
+++ /dev/null
@@ -1,7 +0,0 @@
-<?xml version="1.0" encoding="UTF-8"?>
-<classpath>
- <classpathentry kind="con" path="com.android.ide.eclipse.adt.ANDROID_FRAMEWORK"/>
- <classpathentry kind="src" path="src"/>
- <classpathentry kind="src" path="gen"/>
- <classpathentry kind="output" path="bin"/>
-</classpath>
diff --git a/trunk/src/voice_engine/main/test/android/android_test/Android.mk b/trunk/src/voice_engine/main/test/android/android_test/Android.mk
deleted file mode 100644
index a3f5ce6..0000000
--- a/trunk/src/voice_engine/main/test/android/android_test/Android.mk
+++ /dev/null
@@ -1,25 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH:= $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_TAGS := tests
-
-LOCAL_SRC_FILES := \
- src/org/webrtc/voiceengine/test/AndroidTest.java
-
-LOCAL_PACKAGE_NAME := webrtc-voice-demo
-LOCAL_CERTIFICATE := platform
-
-LOCAL_JNI_SHARED_LIBRARIES := libwebrtc-voice-demo-jni
-
-include $(BUILD_PACKAGE)
-
-include $(call all-makefiles-under,$(LOCAL_PATH))
diff --git a/trunk/src/voice_engine/main/test/android/android_test/AndroidManifest.xml b/trunk/src/voice_engine/main/test/android/android_test/AndroidManifest.xml
deleted file mode 100644
index a614f8d..0000000
--- a/trunk/src/voice_engine/main/test/android/android_test/AndroidManifest.xml
+++ /dev/null
@@ -1,32 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<!-- Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. -->
-<!-- -->
-<!-- Use of this source code is governed by a BSD-style license -->
-<!-- that can be found in the LICENSE file in the root of the source -->
-<!-- tree. An additional intellectual property rights grant can be found -->
-<!-- in the file PATENTS. All contributing project authors may -->
-<!-- be found in the AUTHORS file in the root of the source tree. -->
-
-<manifest xmlns:android="http://schemas.android.com/apk/res/android"
- android:versionCode="1"
- android:versionName="1.0" package="org.webrtc.voiceengine.test">
- <application android:icon="@drawable/icon"
- android:label="@string/app_name"
- android:debuggable="true">
- <activity android:name=".AndroidTest"
- android:label="@string/app_name"
- android:screenOrientation="portrait">
- <intent-filter>
- <action android:name="android.intent.action.MAIN" />
- <category android:name="android.intent.category.LAUNCHER" />
- </intent-filter>
- </activity>
-
- </application>
- <uses-sdk android:minSdkVersion="3" />
-
- <uses-permission android:name="android.permission.MODIFY_AUDIO_SETTINGS" />
- <uses-permission android:name="android.permission.RECORD_AUDIO" />
- <uses-permission android:name="android.permission.INTERNET" />
-
-</manifest>
diff --git a/trunk/src/voice_engine/main/test/android/android_test/default.properties b/trunk/src/voice_engine/main/test/android/android_test/default.properties
deleted file mode 100644
index 6ed608e..0000000
--- a/trunk/src/voice_engine/main/test/android/android_test/default.properties
+++ /dev/null
@@ -1,11 +0,0 @@
-# This file is automatically generated by Android Tools.
-# Do not modify this file -- YOUR CHANGES WILL BE ERASED!
-#
-# This file must be checked in Version Control Systems.
-#
-# To customize properties used by the Ant build system use,
-# "build.properties", and override values to adapt the script to your
-# project structure.
-
-# Project target, OpenSL ES requires API level 9
-target=android-9
diff --git a/trunk/src/voice_engine/main/test/android/android_test/gen/org/webrtc/voiceengine/test/R.java b/trunk/src/voice_engine/main/test/android/android_test/gen/org/webrtc/voiceengine/test/R.java
deleted file mode 100644
index ec8f5b4..0000000
--- a/trunk/src/voice_engine/main/test/android/android_test/gen/org/webrtc/voiceengine/test/R.java
+++ /dev/null
@@ -1,30 +0,0 @@
-/* AUTO-GENERATED FILE. DO NOT MODIFY.
- *
- * This class was automatically generated by the
- * aapt tool from the resource data it found. It
- * should not be modified by hand.
- */
-
-package org.webrtc.voiceengine.test;
-
-public final class R {
- public static final class attr {
- }
- public static final class drawable {
- public static final int icon=0x7f020000;
- }
- public static final class id {
- public static final int Button01=0x7f050002;
- public static final int Button02=0x7f050005;
- public static final int EditText01=0x7f050001;
- public static final int Spinner01=0x7f050003;
- public static final int Spinner02=0x7f050004;
- public static final int TextView01=0x7f050000;
- }
- public static final class layout {
- public static final int main=0x7f030000;
- }
- public static final class string {
- public static final int app_name=0x7f040000;
- }
-}
diff --git a/trunk/src/voice_engine/main/test/android/android_test/jni/Android.mk b/trunk/src/voice_engine/main/test/android/android_test/jni/Android.mk
deleted file mode 100644
index 9f9f761..0000000
--- a/trunk/src/voice_engine/main/test/android/android_test/jni/Android.mk
+++ /dev/null
@@ -1,152 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-ifdef NDK_ROOT
-
-MY_WEBRTC_ROOT_PATH := $(call my-dir)
-
-MY_WEBRTC_SRC_PATH := ../../../../../../..
-
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/common_audio/resampler/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/common_audio/signal_processing/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/common_audio/vad/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/neteq/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/codecs/cng/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/codecs/g711/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/codecs/g722/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/codecs/pcm16b/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/codecs/ilbc/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/codecs/iSAC/fix/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/codecs/iSAC/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_coding/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_conference_mixer/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_device/main/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_processing/aec/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_processing/aecm/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_processing/agc/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_processing/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_processing/ns/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/audio_processing/utility/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/media_file/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/rtp_rtcp/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/udp_transport/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/modules/utility/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/system_wrappers/source/Android.mk
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/voice_engine/main/source/Android.mk
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE := libwebrtc_audio_preprocessing
-LOCAL_MODULE_TAGS := optional
-
-LOCAL_WHOLE_STATIC_LIBRARIES := \
- libwebrtc_spl \
- libwebrtc_resampler \
- libwebrtc_apm \
- libwebrtc_apm_utility \
- libwebrtc_vad \
- libwebrtc_ns \
- libwebrtc_agc \
- libwebrtc_aec \
- libwebrtc_aecm \
- libwebrtc_system_wrappers \
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libstlport_shared
-
-LOCAL_LDLIBS := \
- -lgcc \
- -llog
-
-LOCAL_PRELINK_MODULE := false
-
-include $(BUILD_SHARED_LIBRARY)
-
-###
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_ARM_MODE := arm
-LOCAL_MODULE := libwebrtc-voice-jni
-LOCAL_MODULE_TAGS := optional
-
-LOCAL_WHOLE_STATIC_LIBRARIES := \
- libwebrtc_system_wrappers \
- libwebrtc_audio_device \
- libwebrtc_pcm16b \
- libwebrtc_cng \
- libwebrtc_audio_coding \
- libwebrtc_rtp_rtcp \
- libwebrtc_media_file \
- libwebrtc_udp_transport \
- libwebrtc_utility \
- libwebrtc_neteq \
- libwebrtc_audio_conference_mixer \
- libwebrtc_isac \
- libwebrtc_ilbc \
- libwebrtc_isacfix \
- libwebrtc_g722 \
- libwebrtc_g711 \
- libwebrtc_voe_core
-
-LOCAL_SHARED_LIBRARIES := \
- libcutils \
- libstlport_shared \
- libwebrtc_audio_preprocessing
-
-LOCAL_LDLIBS := \
- -lgcc \
- -llog \
- -lOpenSLES
-
-LOCAL_PRELINK_MODULE := false
-
-include $(BUILD_SHARED_LIBRARY)
-
-###
-
-include $(MY_WEBRTC_ROOT_PATH)/$(MY_WEBRTC_SRC_PATH)/src/voice_engine/main/test/cmd_test/Android.mk
-
-else
-
-LOCAL_PATH := $(call my-dir)
-
-include $(CLEAR_VARS)
-
-LOCAL_MODULE_TAGS := tests
-LOCAL_MODULE := libwebrtc-voice-demo-jni
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES := android_test.cc
-LOCAL_CFLAGS := \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_ANDROID'
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../../../auto_test \
- $(LOCAL_PATH)/../../../../interface \
- $(LOCAL_PATH)/../../../../../.. \
- $(LOCAL_PATH)/../../../../../../system_wrappers/interface
-
-LOCAL_PRELINK_MODULE := false
-
-LOCAL_SHARED_LIBRARIES := \
- libutils \
- libstlport \
- libandroid \
- libwebrtc \
- libGLESv2
-
-include $(BUILD_SHARED_LIBRARY)
-
-endif
diff --git a/trunk/src/voice_engine/main/test/android/android_test/jni/Application.mk b/trunk/src/voice_engine/main/test/android/android_test/jni/Application.mk
deleted file mode 100644
index 03c35ac..0000000
--- a/trunk/src/voice_engine/main/test/android/android_test/jni/Application.mk
+++ /dev/null
@@ -1,11 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-# Build both ARMv5TE and ARMv7-A machine code.
-APP_ABI := armeabi armeabi-v7a x86
-APP_STL := stlport_shared
diff --git a/trunk/src/voice_engine/main/test/android/android_test/jni/android_test.cc b/trunk/src/voice_engine/main/test/android/android_test/jni/android_test.cc
deleted file mode 100644
index 8c5fdff..0000000
--- a/trunk/src/voice_engine/main/test/android/android_test/jni/android_test.cc
+++ /dev/null
@@ -1,1554 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <string.h>
-#include <unistd.h>
-#include <android/log.h>
-
-#include "org_webrtc_voiceengine_test_AndroidTest.h"
-
-#include "thread_wrapper.h"
-
-#include "voe_base.h"
-#include "voe_codec.h"
-#include "voe_file.h"
-#include "voe_network.h"
-#include "voe_audio_processing.h"
-#include "voe_volume_control.h"
-#include "voe_hardware.h"
-#include "voe_rtp_rtcp.h"
-#include "voe_encryption.h"
-
-#include "voe_test_interface.h"
-
-//#define USE_SRTP
-//#define INIT_FROM_THREAD
-//#define START_CALL_FROM_THREAD
-
-#define WEBRTC_LOG_TAG "*WEBRTCN*" // As in WEBRTC Native...
-#define VALIDATE_BASE_POINTER \
- if (!veData1.base) \
- { \
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
- "Base pointer doesn't exist"); \
- return -1; \
- }
-#define VALIDATE_CODEC_POINTER \
- if (!veData1.codec) \
- { \
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
- "Codec pointer doesn't exist"); \
- return -1; \
- }
-#define VALIDATE_FILE_POINTER \
- if (!veData1.file) \
- { \
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
- "File pointer doesn't exist"); \
- return -1; \
- }
-#define VALIDATE_NETWORK_POINTER \
- if (!veData1.netw) \
- { \
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
- "Network pointer doesn't exist"); \
- return -1; \
- }
-#define VALIDATE_APM_POINTER \
- if (!veData1.codec) \
- { \
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
- "Apm pointer doesn't exist"); \
- return -1; \
- }
-#define VALIDATE_VOLUME_POINTER \
- if (!veData1.volume) \
- { \
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
- "Volume pointer doesn't exist"); \
- return -1; \
- }
-#define VALIDATE_HARDWARE_POINTER \
- if (!veData1.hardware) \
- { \
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
- "Hardware pointer doesn't exist"); \
- return -1; \
- }
-#define VALIDATE_RTP_RTCP_POINTER \
- if (!veData1.rtp_rtcp) \
- { \
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
- "RTP / RTCP pointer doesn't exist"); \
- return -1; \
- }
-#define VALIDATE_ENCRYPT_POINTER \
- if (!veData1.encrypt) \
- { \
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, \
- "Encrypt pointer doesn't exist"); \
- return -1; \
- }
-
-// Register functions in JNI_OnLoad()
-// How do we ensure that VoE is deleted? JNI_OnUnload?
-// What happens if class is unloaded? When loaded again, NativeInit will be
-// called again. Keep what we have?
-// Should we do something in JNI_OnUnload?
-// General design: create a class or keep global struct with "C" functions?
-// Otherwise make sure symbols are as unique as possible.
-
-// TestType enumerator
-enum TestType
-{
- Invalid = -1,
- Standard = 0,
- Extended = 1,
- Stress = 2,
- Unit = 3,
- CPU = 4
-};
-
-// ExtendedSelection enumerator
-enum ExtendedSelection
-{
- XSEL_Invalid = -1,
- XSEL_None = 0,
- XSEL_All,
- XSEL_Base,
- XSEL_CallReport,
- XSEL_Codec,
- XSEL_DTMF,
- XSEL_Encryption,
- XSEL_ExternalMedia,
- XSEL_File,
- XSEL_Hardware,
- XSEL_NetEqStats,
- XSEL_Network,
- XSEL_PTT,
- XSEL_RTP_RTCP,
- XSEL_VideoSync,
- XSEL_VideoSyncExtended,
- XSEL_VolumeControl,
- XSEL_VQE,
- XSEL_APM,
- XSEL_VQMon
-};
-
-using namespace webrtc;
-
-class my_transportation;
-
-// VoiceEngine data struct
-typedef struct
-{
- // VoiceEngine
- VoiceEngine* ve;
- // Sub-APIs
- VoEBase* base;
- VoECodec* codec;
- VoEFile* file;
- VoENetwork* netw;
- VoEAudioProcessing* apm;
- VoEVolumeControl* volume;
- VoEHardware* hardware;
- VoERTP_RTCP* rtp_rtcp;
- VoEEncryption* encrypt;
- // Other
- my_transportation* extTrans;
- JavaVM* jvm;
-} VoiceEngineData;
-
-// my_transportation is used when useExtTrans is enabled
-class my_transportation : public Transport
-{
- public:
- my_transportation(VoENetwork * network) :
- netw(network) {
- }
-
- int SendPacket(int channel,const void *data,int len);
- int SendRTCPPacket(int channel, const void *data, int len);
- private:
- VoENetwork * netw;
-};
-
-int my_transportation::SendPacket(int channel,const void *data,int len)
-{
- netw->ReceivedRTPPacket(channel, data, len);
- return len;
-}
-
-int my_transportation::SendRTCPPacket(int channel, const void *data, int len)
-{
- netw->ReceivedRTCPPacket(channel, data, len);
- return len;
-}
-
-//Global variables visible in this file
-static VoiceEngineData veData1;
-static VoiceEngineData veData2;
-
-// "Local" functions (i.e. not Java accessible)
-static bool GetSubApis(VoiceEngineData &veData);
-static bool ReleaseSubApis(VoiceEngineData &veData);
-
-class ThreadTest
-{
-public:
- ThreadTest();
- ~ThreadTest();
- int RunTest();
- int CloseTest();
-private:
- static bool Run(void* ptr);
- bool Process();
-private:
- ThreadWrapper* _thread;
-};
-
-ThreadTest::~ThreadTest()
-{
- if (_thread)
- {
- _thread->SetNotAlive();
- if (_thread->Stop())
- {
- delete _thread;
- _thread = NULL;
- }
- }
-}
-
-ThreadTest::ThreadTest() :
- _thread(NULL)
-{
- _thread = ThreadWrapper::CreateThread(Run, this, kNormalPriority,
- "ThreadTest thread");
-}
-
-bool ThreadTest::Run(void* ptr)
-{
- return static_cast<ThreadTest*> (ptr)->Process();
-}
-
-bool ThreadTest::Process()
-{
- // Attach this thread to JVM
- /*JNIEnv* env = NULL;
- jint res = veData1.jvm->AttachCurrentThread(&env, NULL);
- char msg[32];
- sprintf(msg, "res=%d, env=%d", res, env);
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG, msg);*/
-
-#ifdef INIT_FROM_THREAD
- VALIDATE_BASE_POINTER;
- veData1.base->Init();
-#endif
-
-#ifdef START_CALL_FROM_THREAD
- // receiving instance
- veData2.ve = VoiceEngine::Create();
- GetSubApis(veData2);
- veData2.base->Init();
- veData2.base->CreateChannel();
- if(veData2.base->SetLocalReceiver(0, 1234) < 0)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "set local receiver 2 failed");
- }
- veData2.hardware->SetLoudspeakerStatus(false);
- veData2.volume->SetSpeakerVolume(204);
- veData2.base->StartReceive(0);
- if(veData2.base->StartPlayout(0) < 0)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "start playout failed");
- }
-
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "receiving instance started from thread");
-
- // sending instance
- veData1.ve = VoiceEngine::Create();
- GetSubApis(veData1);
- veData1.base->Init();
- if(veData1.base->CreateChannel() < 0)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "create channel failed");
- }
- if(veData1.base->SetLocalReceiver(0, 1256) < 0)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "set local receiver failed");
- }
- if(veData1.base->SetSendDestination(0, 1234, "127.0.0.1") < 0)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "set send destination failed");
- }
- if(veData1.base->StartSend(0) < 0)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "start send failed");
- }
-
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "sending instance started from thread");
-#endif
-
- _thread->SetNotAlive();
- _thread->Stop();
-
- //res = veData1.jvm->DetachCurrentThread();
-
- return true;
-}
-
-int ThreadTest::RunTest()
-{
- if (_thread)
- {
- unsigned int id;
- _thread->Start(id);
- }
- return 0;
-}
-
-int ThreadTest::CloseTest()
-{
- VALIDATE_BASE_POINTER
-
- veData1.base->DeleteChannel(0);
- veData2.base->DeleteChannel(0);
- veData1.base->Terminate();
- veData2.base->Terminate();
-
- // Release sub-APIs
- ReleaseSubApis(veData1);
- ReleaseSubApis(veData2);
-
- // Delete
- VoiceEngine::Delete(veData1.ve);
- VoiceEngine::Delete(veData2.ve);
- veData2.ve = NULL;
- veData2.ve = NULL;
-
- return 0;
-}
-
-ThreadTest threadTest;
-
-//////////////////////////////////////////////////////////////////
-// General functions
-//////////////////////////////////////////////////////////////////
-
-/////////////////////////////////////////////
-// JNI_OnLoad
-//
-jint JNI_OnLoad(JavaVM* vm, void* /*reserved*/)
-{
- if (!vm)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "JNI_OnLoad did not receive a valid VM pointer");
- return -1;
- }
-
- // Get JNI
- JNIEnv* env;
- if (JNI_OK != vm->GetEnv(reinterpret_cast<void**> (&env),
- JNI_VERSION_1_4))
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "JNI_OnLoad could not get JNI env");
- return -1;
- }
-
- // Get class to register the native functions with
- // jclass regClass = env->FindClass("webrtc/android/AndroidTest");
- // if (!regClass) {
- // return -1; // Exception thrown
- // }
-
- // Register native functions
- // JNINativeMethod methods[1];
- // methods[0].name = NULL;
- // methods[0].signature = NULL;
- // methods[0].fnPtr = NULL;
- // if (JNI_OK != env->RegisterNatives(regClass, methods, 1))
- // {
- // return -1;
- // }
-
- // Init VoiceEngine data
- memset(&veData1, 0, sizeof(veData1));
- memset(&veData2, 0, sizeof(veData2));
-
- // Store the JVM
- veData1.jvm = vm;
- veData2.jvm = vm;
-
- return JNI_VERSION_1_4;
-}
-
-/////////////////////////////////////////////
-// Native initialization
-//
-JNIEXPORT jboolean JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_NativeInit(
- JNIEnv * env,
- jclass)
-{
- // Look up and cache any interesting class, field and method IDs for
- // any used java class here
-
- return true;
-}
-
-/////////////////////////////////////////////
-// Run auto standard test
-//
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_RunAutoTest(
- JNIEnv *env,
- jobject context,
- jint testType,
- jint extendedSel)
-{
- TestType tType(Invalid);
-
- switch (testType)
- {
- case 0:
- return 0;
- case 1:
- tType = Standard;
- break;
- case 2:
- tType = Extended;
- break;
- case 3:
- tType = Stress;
- break;
- case 4:
- tType = Unit;
- break;
- default:
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "RunAutoTest - Invalid TestType");
- return -1;
- }
-
- ExtendedSelection xsel(XSEL_Invalid);
-
- switch (extendedSel)
- {
- case 0:
- xsel = XSEL_None;
- break;
- case 1:
- xsel = XSEL_All;
- break;
- case 2:
- xsel = XSEL_Base;
- break;
- case 3:
- xsel = XSEL_CallReport;
- break;
- case 4:
- xsel = XSEL_Codec;
- break;
- case 5:
- xsel = XSEL_DTMF;
- break;
- case 6:
- xsel = XSEL_Encryption;
- break;
- case 7:
- xsel = XSEL_ExternalMedia;
- break;
- case 8:
- xsel = XSEL_File;
- break;
- case 9:
- xsel = XSEL_Hardware;
- break;
- case 10:
- xsel = XSEL_NetEqStats;
- break;
- case 11:
- xsel = XSEL_Network;
- break;
- case 12:
- xsel = XSEL_PTT;
- break;
- case 13:
- xsel = XSEL_RTP_RTCP;
- break;
- case 14:
- xsel = XSEL_VideoSync;
- break;
- case 15:
- xsel = XSEL_VideoSyncExtended;
- break;
- case 16:
- xsel = XSEL_VolumeControl;
- break;
- case 17:
- xsel = XSEL_APM;
- break;
- case 18:
- xsel = XSEL_VQMon;
- break;
- default:
- xsel = XSEL_Invalid;
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "RunAutoTest - Invalid extendedType");
- return -1;
- }
-
- // Set instance independent Java objects
- VoiceEngine::SetAndroidObjects(veData1.jvm, env, context);
-
- // Call voe test interface function
- // TODO(leozwang) add autotest setAndroidObjects(veData1.jvm, context);
- // jint retVal = runAutoTest(tType, xsel);
-
- // Clear instance independent Java objects
- VoiceEngine::SetAndroidObjects(NULL, NULL, NULL);
-
- return 0;
-}
-
-//////////////////////////////////////////////////////////////////
-// VoiceEngine API wrapper functions
-//////////////////////////////////////////////////////////////////
-
-/////////////////////////////////////////////
-// Create VoiceEngine instance
-//
-JNIEXPORT jboolean JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Create(
- JNIEnv *env,
- jobject context)
-{
- // Check if already created
- if (veData1.ve)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "VoE already created");
- return false;
- }
-
- // Set instance independent Java objects
- VoiceEngine::SetAndroidObjects(veData1.jvm, env, context);
-
-#ifdef START_CALL_FROM_THREAD
- threadTest.RunTest();
-#else
- // Create
- veData1.ve = VoiceEngine::Create();
- if (!veData1.ve)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Create VoE failed");
- return false;
- }
-
- // Get sub-APIs
- if (!GetSubApis(veData1))
- {
- // If not OK, release all sub-APIs and delete VoE
- ReleaseSubApis(veData1);
- if (!VoiceEngine::Delete(veData1.ve))
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Delete VoE failed");
- }
- return false;
- }
-#endif
-
- return true;
-}
-
-/////////////////////////////////////////////
-// Delete VoiceEngine instance
-//
-JNIEXPORT jboolean JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Delete(
- JNIEnv *,
- jobject)
-{
-#ifdef START_CALL_FROM_THREAD
- threadTest.CloseTest();
-#else
- // Check if exists
- if (!veData1.ve)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "VoE does not exist");
- return false;
- }
-
- // Release sub-APIs
- ReleaseSubApis(veData1);
-
- // Delete
- if (!VoiceEngine::Delete(veData1.ve))
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Delete VoE failed");
- return false;
- }
-
- veData1.ve = NULL;
-#endif
-
- // Clear instance independent Java objects
- VoiceEngine::SetAndroidObjects(NULL, NULL, NULL);
-
- return true;
-}
-
-/////////////////////////////////////////////
-// [Base] Initialize VoiceEngine
-//
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Init(
- JNIEnv *,
- jobject,
- jboolean enableTrace,
- jboolean useExtTrans)
-{
- VALIDATE_BASE_POINTER;
-
- if (enableTrace)
- {
- if (0 != VoiceEngine::SetTraceFile("/sdcard/trace.txt"))
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Could not enable trace");
- }
- if (0 != VoiceEngine::SetTraceFilter(kTraceAll))
- {
- __android_log_write(ANDROID_LOG_WARN, WEBRTC_LOG_TAG,
- "Could not set trace filter");
- }
- }
-
- if (useExtTrans)
- {
- VALIDATE_NETWORK_POINTER;
- veData1.extTrans = new my_transportation(veData1.netw);
- }
-
- int retVal = 0;
-#ifdef INIT_FROM_THREAD
- threadTest.RunTest();
- usleep(200000);
-#else
- retVal = veData1.base->Init();
-#endif
- return retVal;
-}
-
-/////////////////////////////////////////////
-// [Base] Terminate VoiceEngine
-//
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Terminate(
- JNIEnv *,
- jobject)
-{
- VALIDATE_BASE_POINTER;
-
- jint retVal = veData1.base->Terminate();
-
- delete veData1.extTrans;
- veData1.extTrans = NULL;
-
- return retVal;
-}
-
-/////////////////////////////////////////////
-// [Base] Create channel
-//
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_CreateChannel(
- JNIEnv *,
- jobject)
-{
- VALIDATE_BASE_POINTER;
- jint channel = veData1.base->CreateChannel();
-
- if (veData1.extTrans)
- {
- VALIDATE_NETWORK_POINTER;
- __android_log_print(ANDROID_LOG_DEBUG, WEBRTC_LOG_TAG,
- "Enabling external transport on channel %d",
- channel);
- if (veData1.netw->RegisterExternalTransport(channel, *veData1.extTrans)
- < 0)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Could not set external transport");
- return -1;
- }
- }
-
- return channel;
-}
-
-/////////////////////////////////////////////
-// [Base] Delete channel
-//
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_DeleteChannel(
- JNIEnv *,
- jobject,
- jint channel)
-{
- VALIDATE_BASE_POINTER;
- return veData1.base->DeleteChannel(channel);
-}
-
-/////////////////////////////////////////////
-// [Base] SetLocalReceiver
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_SetLocalReceiver(
- JNIEnv *,
- jobject,
- jint channel,
- jint port)
-{
- VALIDATE_BASE_POINTER;
- return veData1.base->SetLocalReceiver(channel, port);
-}
-
-/////////////////////////////////////////////
-// [Base] SetSendDestination
-//
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_SetSendDestination(
- JNIEnv *env,
- jobject,
- jint channel,
- jint port,
- jstring ipaddr)
-{
- VALIDATE_BASE_POINTER;
-
- const char* ipaddrNative = env->GetStringUTFChars(ipaddr, NULL);
- if (!ipaddrNative)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Could not get UTF string");
- return -1;
- }
-
- jint retVal = veData1.base->SetSendDestination(channel, port, ipaddrNative);
-
- env->ReleaseStringUTFChars(ipaddr, ipaddrNative);
-
- return retVal;
-}
-
-/////////////////////////////////////////////
-// [Base] StartListen
-//
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartListen(
- JNIEnv *,
- jobject,
- jint channel)
-{
-#ifdef USE_SRTP
- VALIDATE_ENCRYPT_POINTER;
- bool useForRTCP = false;
- if (veData1.encrypt->EnableSRTPReceive(
- channel,CIPHER_AES_128_COUNTER_MODE,30,AUTH_HMAC_SHA1,
- 16,4, ENCRYPTION_AND_AUTHENTICATION,
- (unsigned char*)nikkey, useForRTCP) != 0)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to enable SRTP receive");
- return -1;
- }
-#endif
-
- VALIDATE_BASE_POINTER;
- int retVal = veData1.base->StartReceive(channel);
-
- return retVal;
-}
-
-/////////////////////////////////////////////
-// [Base] Start playout
-//
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_StartPlayout(
- JNIEnv *,
- jobject,
- jint channel)
-{
- VALIDATE_BASE_POINTER;
- int retVal = veData1.base->StartPlayout(channel);
-
- return retVal;
-}
-
-/////////////////////////////////////////////
-// [Base] Start send
-//
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartSend(
- JNIEnv *,
- jobject,
- jint channel)
-{
- /* int dscp(0), serviceType(-1), overrideDscp(0), res(0);
- bool gqosEnabled(false), useSetSockOpt(false);
-
- if (veData1.netw->SetSendTOS(channel, 13, useSetSockOpt) != 0)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to set TOS");
- return -1;
- }
-
- res = veData1.netw->GetSendTOS(channel, dscp, useSetSockOpt);
- if (res != 0 || dscp != 13 || useSetSockOpt != true)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to get TOS");
- return -1;
- } */
-
- /* if (veData1.rtp_rtcp->SetFECStatus(channel, 1) != 0)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to enable FEC");
- return -1;
- } */
-#ifdef USE_SRTP
- VALIDATE_ENCRYPT_POINTER;
- bool useForRTCP = false;
- if (veData1.encrypt->EnableSRTPSend(
- channel,CIPHER_AES_128_COUNTER_MODE,30,AUTH_HMAC_SHA1,
- 16,4, ENCRYPTION_AND_AUTHENTICATION,
- (unsigned char*)nikkey, useForRTCP) != 0)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to enable SRTP send");
- return -1;
- }
-#endif
-
- VALIDATE_BASE_POINTER;
- int retVal = veData1.base->StartSend(channel);
-
- return retVal;
-}
-
-/////////////////////////////////////////////
-// [Base] Stop listen
-//
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopListen(
- JNIEnv *,
- jobject,
- jint channel)
-{
-#ifdef USE_SRTP
- VALIDATE_ENCRYPT_POINTER;
- if (veData1.encrypt->DisableSRTPReceive(channel) != 0)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to disable SRTP receive");
- return -1;
- }
-#endif
-
- VALIDATE_BASE_POINTER;
- return veData1.base->StopReceive(channel);
-}
-
-/////////////////////////////////////////////
-// [Base] Stop playout
-//
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopPlayout(
- JNIEnv *,
- jobject,
- jint channel)
-{
- VALIDATE_BASE_POINTER;
- return veData1.base->StopPlayout(channel);
-}
-
-/////////////////////////////////////////////
-// [Base] Stop send
-//
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopSend(
- JNIEnv *,
- jobject,
- jint channel)
-{
- /* if (veData1.rtp_rtcp->SetFECStatus(channel, 0) != 0)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to disable FEC");
- return -1;
- } */
-
-#ifdef USE_SRTP
- VALIDATE_ENCRYPT_POINTER;
- if (veData1.encrypt->DisableSRTPSend(channel) != 0)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to disable SRTP send");
- return -1;
- }
-#endif
-
- VALIDATE_BASE_POINTER;
- return veData1.base->StopSend(channel);
-}
-
-/////////////////////////////////////////////
-// [codec] Number of codecs
-//
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_NumOfCodecs(
- JNIEnv *,
- jobject)
-{
- VALIDATE_CODEC_POINTER;
- return veData1.codec->NumOfCodecs();
-}
-
-/////////////////////////////////////////////
-// [codec] Set send codec
-//
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_SetSendCodec(
- JNIEnv *,
- jobject,
- jint channel,
- jint index)
-{
- VALIDATE_CODEC_POINTER;
-
- CodecInst codec;
-
- if (veData1.codec->GetCodec(index, codec) != 0)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to get codec");
- return -1;
- }
-
- return veData1.codec->SetSendCodec(channel, codec);
-}
-
-/////////////////////////////////////////////
-// [codec] Set VAD status
-//
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_SetVADStatus(
- JNIEnv *,
- jobject,
- jint channel,
- jboolean enable,
- jint mode)
-{
- VALIDATE_CODEC_POINTER;
-
- VadModes VADmode = kVadConventional;
-
- switch (mode)
- {
- case 0:
- break; // already set
- case 1:
- VADmode = kVadAggressiveLow;
- break;
- case 2:
- VADmode = kVadAggressiveMid;
- break;
- case 3:
- VADmode = kVadAggressiveHigh;
- break;
- default:
- VADmode = (VadModes) 17; // force error
- break;
- }
-
- return veData1.codec->SetVADStatus(channel, enable, VADmode);
-}
-
-/////////////////////////////////////////////
-// [apm] SetNSStatus
-//
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetNSStatus(
- JNIEnv *,
- jobject,
- jboolean enable,
- jint mode)
-{
- VALIDATE_APM_POINTER;
-
- NsModes NSmode = kNsDefault;
-
- switch (mode)
- {
- case 0:
- NSmode = kNsUnchanged;
- break;
- case 1:
- break; // already set
- case 2:
- NSmode = kNsConference;
- break;
- case 3:
- NSmode = kNsLowSuppression;
- break;
- case 4:
- NSmode = kNsModerateSuppression;
- break;
- case 5:
- NSmode = kNsHighSuppression;
- break;
- case 6:
- NSmode = kNsVeryHighSuppression;
- break;
- default:
- NSmode = (NsModes) 17; // force error
- break;
- }
-
- return veData1.apm->SetNsStatus(enable, NSmode);
-}
-
-/////////////////////////////////////////////
-// [apm] SetAGCStatus
-//
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_SetAGCStatus(
- JNIEnv *,
- jobject,
- jboolean enable,
- jint mode)
-{
- VALIDATE_APM_POINTER;
-
- AgcModes AGCmode = kAgcDefault;
-
- switch (mode)
- {
- case 0:
- AGCmode = kAgcUnchanged;
- break;
- case 1:
- break; // already set
- case 2:
- AGCmode = kAgcAdaptiveAnalog;
- break;
- case 3:
- AGCmode = kAgcAdaptiveDigital;
- break;
- case 4:
- AGCmode = kAgcFixedDigital;
- break;
- default:
- AGCmode = (AgcModes) 17; // force error
- break;
- }
-
- /* AgcConfig agcConfig;
- agcConfig.targetLeveldBOv = 3;
- agcConfig.digitalCompressionGaindB = 50;
- agcConfig.limiterEnable = 0;
-
- if (veData1.apm->SetAGCConfig(agcConfig) != 0)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Failed to set AGC config");
- return -1;
- } */
-
- return veData1.apm->SetAgcStatus(enable, AGCmode);
-}
-
-/////////////////////////////////////////////
-// [apm] SetECStatus
-//
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetECStatus(
- JNIEnv *,
- jobject,
- jboolean enable,
- jint mode)
-{
- VALIDATE_APM_POINTER;
-
- EcModes ECmode = kEcDefault;
-
- switch (mode)
- {
- case 0:
- ECmode = kEcDefault;
- break;
- case 1:
- break; // already set
- case 2:
- ECmode = kEcConference;
- break;
- case 3:
- ECmode = kEcAec;
- break;
- case 4:
- ECmode = kEcAecm;
- break;
- default:
- ECmode = (EcModes) 17; // force error
- break;
- }
-
- return veData1.apm->SetEcStatus(enable, ECmode);
-}
-
-/////////////////////////////////////////////
-// [File] Start play file locally
-//
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_StartPlayingFileLocally(
- JNIEnv * env,
- jobject,
- jint channel,
- jstring fileName,
- jboolean loop)
-{
- VALIDATE_FILE_POINTER;
-
- const char* fileNameNative = env->GetStringUTFChars(fileName, NULL);
- if (!fileNameNative)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Could not get UTF string");
- return -1;
- }
-
- jint retVal = veData1.file->StartPlayingFileLocally(channel,
- fileNameNative, loop);
-
- env->ReleaseStringUTFChars(fileName, fileNameNative);
-
- return retVal;
-}
-
-/////////////////////////////////////////////
-// [File] Stop play file locally
-//
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_StopPlayingFileLocally(
- JNIEnv *,
- jobject,
- jint channel)
-{
- VALIDATE_FILE_POINTER;
- return veData1.file->StopPlayingFileLocally(channel);
-}
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: StartRecordingPlayout
- * Signature: (ILjava/lang/String;Z)I
- */
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_StartRecordingPlayout(
- JNIEnv * env,
- jobject,
- jint channel,
- jstring fileName,
- jboolean)
-{
- VALIDATE_FILE_POINTER;
-
- const char* fileNameNative = env->GetStringUTFChars(fileName, NULL);
- if (!fileNameNative)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Could not get UTF string");
- return -1;
- }
-
- jint retVal = veData1.file->StartRecordingPlayout(channel, fileNameNative,
- 0);
-
- env->ReleaseStringUTFChars(fileName, fileNameNative);
-
- return retVal;
-}
-
-/////////////////////////////////////////////
-// [File] Stop Recording Playout
-//
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_StopRecordingPlayout(
- JNIEnv *,
- jobject,
- jint channel)
-{
- VALIDATE_FILE_POINTER;
- return veData1.file->StopRecordingPlayout(channel);
-}
-
-/////////////////////////////////////////////
-// [File] Start playing file as microphone
-//
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_StartPlayingFileAsMicrophone(
- JNIEnv *env,
- jobject,
- jint channel,
- jstring fileName,
- jboolean loop)
-{
- VALIDATE_FILE_POINTER;
-
- const char* fileNameNative = env->GetStringUTFChars(fileName, NULL);
- if (!fileNameNative)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Could not get UTF string");
- return -1;
- }
-
- jint retVal = veData1.file->StartPlayingFileAsMicrophone(channel,
- fileNameNative,
- loop);
-
- env->ReleaseStringUTFChars(fileName, fileNameNative);
-
- return retVal;
-}
-
-/////////////////////////////////////////////
-// [File] Stop playing file as microphone
-//
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_StopPlayingFileAsMicrophone(
- JNIEnv *,
- jobject,
- jint channel)
-{
- VALIDATE_FILE_POINTER;
- return veData1.file->StopPlayingFileAsMicrophone(channel);
-}
-
-/////////////////////////////////////////////
-// [Volume] Set speaker volume
-//
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_SetSpeakerVolume(
- JNIEnv *,
- jobject,
- jint level)
-{
- VALIDATE_VOLUME_POINTER;
- if (veData1.volume->SetSpeakerVolume(level) != 0)
- {
- return -1;
- }
-
- unsigned int storedVolume = 0;
- if (veData1.volume->GetSpeakerVolume(storedVolume) != 0)
- {
- return -1;
- }
-
- if (storedVolume != level)
- {
- return -1;
- }
-
- return 0;
-}
-
-/////////////////////////////////////////////
-// [Hardware] Set loudspeaker status
-//
-JNIEXPORT jint JNICALL
-Java_org_webrtc_voiceengine_test_AndroidTest_SetLoudspeakerStatus(
- JNIEnv *,
- jobject,
- jboolean enable)
-{
- VALIDATE_HARDWARE_POINTER;
- if (veData1.hardware->SetLoudspeakerStatus(enable) != 0)
- {
- return -1;
- }
-
- /*VALIDATE_RTP_RTCP_POINTER;
-
- if (veData1.rtp_rtcp->SetFECStatus(0, enable, -1) != 0)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Could not set FEC");
- return -1;
- }
- else if(enable)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Could enable FEC");
- }
- else
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Could disable FEC");
- }*/
-
- return 0;
-}
-
-//////////////////////////////////////////////////////////////////
-// "Local" functions (i.e. not Java accessible)
-//////////////////////////////////////////////////////////////////
-
-/////////////////////////////////////////////
-// Get all sub-APIs
-//
-bool GetSubApis(VoiceEngineData &veData)
-{
- bool getOK = true;
-
- // Base
- veData.base = VoEBase::GetInterface(veData.ve);
- if (!veData.base)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get base sub-API failed");
- getOK = false;
- }
-
- // Codec
- veData.codec = VoECodec::GetInterface(veData.ve);
- if (!veData.codec)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get codec sub-API failed");
- getOK = false;
- }
-
- // File
- veData.file = VoEFile::GetInterface(veData.ve);
- if (!veData.file)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get file sub-API failed");
- getOK = false;
- }
-
- // Network
- veData.netw = VoENetwork::GetInterface(veData.ve);
- if (!veData.netw)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get network sub-API failed");
- getOK = false;
- }
-
- // AudioProcessing module
- veData.apm = VoEAudioProcessing::GetInterface(veData.ve);
- if (!veData.apm)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get apm sub-API failed");
- getOK = false;
- }
-
- // Volume
- veData.volume = VoEVolumeControl::GetInterface(veData.ve);
- if (!veData.volume)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get volume sub-API failed");
- getOK = false;
- }
-
- // Hardware
- veData.hardware = VoEHardware::GetInterface(veData.ve);
- if (!veData.hardware)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get hardware sub-API failed");
- getOK = false;
- }
-
- // RTP / RTCP
- veData.rtp_rtcp = VoERTP_RTCP::GetInterface(veData.ve);
- if (!veData.rtp_rtcp)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get rtp_rtcp sub-API failed");
- getOK = false;
- }
-
- // Encrypt
- veData.encrypt = VoEEncryption::GetInterface(veData.ve);
- if (!veData.encrypt)
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Get encrypt sub-API failed");
- getOK = false;
- }
-
- return getOK;
-}
-
-/////////////////////////////////////////////
-// Release all sub-APIs
-//
-bool ReleaseSubApis(VoiceEngineData &veData)
-{
- bool releaseOK = true;
-
- // Base
- if (veData.base)
- {
- if (0 != veData.base->Release())
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Release base sub-API failed");
- releaseOK = false;
- }
- else
- {
- veData.base = NULL;
- }
- }
-
- // Codec
- if (veData.codec)
- {
- if (0 != veData.codec->Release())
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Release codec sub-API failed");
- releaseOK = false;
- }
- else
- {
- veData.codec = NULL;
- }
- }
-
- // File
- if (veData.file)
- {
- if (0 != veData.file->Release())
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Release file sub-API failed");
- releaseOK = false;
- }
- else
- {
- veData.file = NULL;
- }
- }
-
- // Network
- if (veData.netw)
- {
- if (0 != veData.netw->Release())
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Release network sub-API failed");
- releaseOK = false;
- }
- else
- {
- veData.netw = NULL;
- }
- }
-
- // apm
- if (veData.apm)
- {
- if (0 != veData.apm->Release())
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Release apm sub-API failed");
- releaseOK = false;
- }
- else
- {
- veData.apm = NULL;
- }
- }
-
- // Volume
- if (veData.volume)
- {
- if (0 != veData.volume->Release())
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Release volume sub-API failed");
- releaseOK = false;
- }
- else
- {
- veData.volume = NULL;
- }
- }
-
- // Hardware
- if (veData.hardware)
- {
- if (0 != veData.hardware->Release())
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Release hardware sub-API failed");
- releaseOK = false;
- }
- else
- {
- veData.hardware = NULL;
- }
- }
-
- // RTP RTCP
- if (veData.rtp_rtcp)
- {
- if (0 != veData.rtp_rtcp->Release())
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Release rtp_rtcp sub-API failed");
- releaseOK = false;
- }
- else
- {
- veData.rtp_rtcp = NULL;
- }
- }
-
- // Encrypt
- if (veData.encrypt)
- {
- if (0 != veData.encrypt->Release())
- {
- __android_log_write(ANDROID_LOG_ERROR, WEBRTC_LOG_TAG,
- "Release encrypt sub-API failed");
- releaseOK = false;
- }
- else
- {
- veData.encrypt = NULL;
- }
- }
-
- return releaseOK;
-}
diff --git a/trunk/src/voice_engine/main/test/android/android_test/jni/org_webrtc_voiceengine_test_AndroidTest.h b/trunk/src/voice_engine/main/test/android/android_test/jni/org_webrtc_voiceengine_test_AndroidTest.h
deleted file mode 100644
index 60fe839..0000000
--- a/trunk/src/voice_engine/main/test/android/android_test/jni/org_webrtc_voiceengine_test_AndroidTest.h
+++ /dev/null
@@ -1,253 +0,0 @@
-/* DO NOT EDIT THIS FILE - it is machine generated */
-#include <jni.h>
-/* Header for class org_webrtc_voiceengine_test_AndroidTest */
-
-#ifndef _Included_org_webrtc_voiceengine_test_AndroidTest
-#define _Included_org_webrtc_voiceengine_test_AndroidTest
-#ifdef __cplusplus
-extern "C" {
-#endif
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: NativeInit
- * Signature: ()Z
- */
-JNIEXPORT jboolean JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_NativeInit
- (JNIEnv *, jclass);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: RunAutoTest
- * Signature: (II)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_RunAutoTest
- (JNIEnv *, jobject, jint, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: Create
- * Signature: ()Z
- */
-JNIEXPORT jboolean JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Create
- (JNIEnv *, jobject);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: Delete
- * Signature: ()Z
- */
-JNIEXPORT jboolean JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Delete
- (JNIEnv *, jobject);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: Init
- * Signature: (IIIZZ)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Init
- (JNIEnv *, jobject, jboolean, jboolean);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: Terminate
- * Signature: ()I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_Terminate
- (JNIEnv *, jobject);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: CreateChannel
- * Signature: ()I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_CreateChannel
- (JNIEnv *, jobject);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: DeleteChannel
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_DeleteChannel
- (JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: SetLocalReceiver
- * Signature: (II)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetLocalReceiver
- (JNIEnv *, jobject, jint, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: SetSendDestination
- * Signature: (IILjava/lang/String;)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetSendDestination
- (JNIEnv *, jobject, jint, jint, jstring);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: StartListen
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartListen
- (JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: StartPlayout
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartPlayout
- (JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: StartSend
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartSend
- (JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: StopListen
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopListen
- (JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: StopPlayout
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopPlayout
- (JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: StopSend
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopSend
- (JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: StartPlayingFileLocally
- * Signature: (ILjava/lang/String;Z)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartPlayingFileLocally
- (JNIEnv *, jobject, jint, jstring, jboolean);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: StopPlayingFileLocally
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopPlayingFileLocally
- (JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: StartRecordingPlayout
- * Signature: (ILjava/lang/String;Z)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartRecordingPlayout
- (JNIEnv *, jobject, jint, jstring, jboolean);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: StopRecordingPlayout
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopRecordingPlayout
- (JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: StartPlayingFileAsMicrophone
- * Signature: (ILjava/lang/String;Z)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StartPlayingFileAsMicrophone
- (JNIEnv *, jobject, jint, jstring, jboolean);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: StopPlayingFileAsMicrophone
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_StopPlayingFileAsMicrophone
- (JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: NumOfCodecs
- * Signature: ()I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_NumOfCodecs
- (JNIEnv *, jobject);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: SetSendCodec
- * Signature: (II)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetSendCodec
- (JNIEnv *, jobject, jint, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: SetVADStatus
- * Signature: (IZI)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetVADStatus
- (JNIEnv *, jobject, jint, jboolean, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: SetNSStatus
- * Signature: (ZI)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetNSStatus
- (JNIEnv *, jobject, jboolean, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: SetAGCStatus
- * Signature: (ZI)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetAGCStatus
- (JNIEnv *, jobject, jboolean, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: SetECStatus
- * Signature: (ZI)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetECStatus
- (JNIEnv *, jobject, jboolean, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: SetSpeakerVolume
- * Signature: (I)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetSpeakerVolume
- (JNIEnv *, jobject, jint);
-
-/*
- * Class: org_webrtc_voiceengine_test_AndroidTest
- * Method: SetLoudspeakerStatus
- * Signature: (Z)I
- */
-JNIEXPORT jint JNICALL Java_org_webrtc_voiceengine_test_AndroidTest_SetLoudspeakerStatus
- (JNIEnv *, jobject, jboolean);
-
-#ifdef __cplusplus
-}
-#endif
-#endif
diff --git a/trunk/src/voice_engine/main/test/android/android_test/res/drawable/icon.png b/trunk/src/voice_engine/main/test/android/android_test/res/drawable/icon.png
deleted file mode 100644
index a07c69f..0000000
--- a/trunk/src/voice_engine/main/test/android/android_test/res/drawable/icon.png
+++ /dev/null
Binary files differ
diff --git a/trunk/src/voice_engine/main/test/android/android_test/res/layout/main.xml b/trunk/src/voice_engine/main/test/android/android_test/res/layout/main.xml
deleted file mode 100644
index 4165a07..0000000
--- a/trunk/src/voice_engine/main/test/android/android_test/res/layout/main.xml
+++ /dev/null
@@ -1,43 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<!-- Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. -->
-<!-- -->
-<!-- Use of this source code is governed by a BSD-style license -->
-<!-- that can be found in the LICENSE file in the root of the source -->
-<!-- tree. An additional intellectual property rights grant can be found -->
-<!-- in the file PATENTS. All contributing project authors may -->
-<!-- be found in the AUTHORS file in the root of the source tree. -->
-
-<LinearLayout xmlns:android="http://schemas.android.com/apk/res/android"
- android:orientation="vertical"
- android:layout_width="fill_parent"
- android:layout_height="fill_parent">
-
- <TextView android:text="@+id/TextView01"
- android:id="@+id/TextView01"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </TextView>
- <EditText android:text="@+id/EditText01"
- android:id="@+id/EditText01"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </EditText>
- <Button android:text="@+id/Button01"
- android:id="@+id/Button01"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </Button>
- <Spinner android:id="@+id/Spinner01"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </Spinner>
- <Spinner android:id="@+id/Spinner02"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </Spinner>
- <Button android:text="@+id/Button02"
- android:id="@+id/Button02"
- android:layout_width="wrap_content"
- android:layout_height="wrap_content">
- </Button>
-</LinearLayout>
diff --git a/trunk/src/voice_engine/main/test/android/android_test/res/values/strings.xml b/trunk/src/voice_engine/main/test/android/android_test/res/values/strings.xml
deleted file mode 100644
index 29ec4ee..0000000
--- a/trunk/src/voice_engine/main/test/android/android_test/res/values/strings.xml
+++ /dev/null
@@ -1,13 +0,0 @@
-<?xml version="1.0" encoding="utf-8"?>
-<!-- Copyright (c) 2011 The WebRTC project authors. All Rights Reserved. -->
-<!-- -->
-<!-- Use of this source code is governed by a BSD-style license -->
-<!-- that can be found in the LICENSE file in the root of the source -->
-<!-- tree. An additional intellectual property rights grant can be found -->
-<!-- in the file PATENTS. All contributing project authors may -->
-<!-- be found in the AUTHORS file in the root of the source tree. -->
-
-<resources>
-
- <string name="app_name">WebRtc VoE</string>
-</resources>
diff --git a/trunk/src/voice_engine/main/test/android/android_test/src/org/webrtc/voiceengine/test/AndroidTest.java b/trunk/src/voice_engine/main/test/android/android_test/src/org/webrtc/voiceengine/test/AndroidTest.java
deleted file mode 100644
index 71b22b0..0000000
--- a/trunk/src/voice_engine/main/test/android/android_test/src/org/webrtc/voiceengine/test/AndroidTest.java
+++ /dev/null
@@ -1,1190 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license that can be found
- * in the LICENSE file in the root of the source tree. An additional
- * intellectual property rights grant can be found in the file PATENTS. All
- * contributing project authors may be found in the AUTHORS file in the root of
- * the source tree.
- */
-
-/*
- * VoiceEngine Android test application. It starts either auto test or acts like
- * a GUI test.
- */
-
-package org.webrtc.voiceengine.test;
-
-import java.io.File;
-import java.io.FileInputStream;
-import java.io.FileNotFoundException;
-import java.io.FileOutputStream;
-import java.io.FileReader;
-import java.io.IOException;
-
-import android.app.Activity;
-import android.content.Context;
-import android.media.AudioFormat;
-import android.media.AudioManager;
-import android.media.AudioRecord;
-import android.media.AudioTrack;
-import android.media.MediaRecorder;
-import android.os.Bundle;
-import android.util.Log;
-import android.view.View;
-import android.widget.AdapterView;
-import android.widget.ArrayAdapter;
-import android.widget.Button;
-import android.widget.EditText;
-import android.widget.Spinner;
-import android.widget.TextView;
-
-public class AndroidTest extends Activity {
- private byte[] _playBuffer = null;
- private short[] _circBuffer = new short[8000]; // can hold 50 frames
-
- private int _recIndex = 0;
- private int _playIndex = 0;
- // private int _streamVolume = 4;
- private int _maxVolume = 0; // Android max level (commonly 5)
- // VoE level (0-255), corresponds to level 4 out of 5
- private int _volumeLevel = 204;
-
- private Thread _playThread;
- private Thread _recThread;
- private Thread _autotestThread;
-
- private static AudioTrack _at;
- private static AudioRecord _ar;
-
- private File _fr = null;
- private FileInputStream _in = null;
-
- private boolean _isRunningPlay = false;
- private boolean _isRunningRec = false;
- private boolean _settingSet = true;
- private boolean _isCallActive = false;
- private boolean _runAutotest = false; // ENABLE AUTOTEST HERE!
-
- private int _channel = -1;
- private int _codecIndex = 0;
- private int _ecIndex = 0;
- private int _nsIndex = 0;
- private int _agcIndex = 0;
- private int _vadIndex = 0;
- private int _audioIndex = 3;
- private int _settingMenu = 0;
- private int _receivePort = 1234;
- private int _destinationPort = 1234;
- private String _destinationIP = "127.0.0.1";
-
- // "Build" settings
- private final boolean _playFromFile = false;
- // Set to true to send data to native code and back
- private final boolean _runThroughNativeLayer = true;
- private final boolean enableSend = true;
- private final boolean enableReceive = true;
- private final boolean useNativeThread = false;
-
- /** Called when the activity is first created. */
- public void onCreate(Bundle savedInstanceState) {
- super.onCreate(savedInstanceState);
- setContentView(R.layout.main);
-
- TextView tv = (TextView) findViewById(R.id.TextView01);
- tv.setText("");
-
- final EditText ed = (EditText) findViewById(R.id.EditText01);
- ed.setWidth(200);
- ed.setText(_destinationIP);
-
- final Button buttonStart = (Button) findViewById(R.id.Button01);
- buttonStart.setWidth(200);
- if (_runAutotest) {
- buttonStart.setText("Run test");
- } else {
- buttonStart.setText("Start Call");
- }
- // button.layout(50, 50, 100, 40);
- buttonStart.setOnClickListener(new View.OnClickListener() {
- public void onClick(View v) {
-
- if (_runAutotest) {
- startAutoTest();
- } else {
- if (_isCallActive) {
-
- if (stopCall() != -1) {
- _isCallActive = false;
- buttonStart.setText("Start Call");
- }
- } else {
-
- _destinationIP = ed.getText().toString();
- if (startCall() != -1) {
- _isCallActive = true;
- buttonStart.setText("Stop Call");
- }
- }
- }
-
- // displayTextFromFile();
- // recordAudioToFile();
- // if(!_playFromFile)
- // {
- // recAudioInThread();
- // }
- // playAudioInThread();
- }
- });
-
- final Button buttonStop = (Button) findViewById(R.id.Button02);
- buttonStop.setWidth(200);
- buttonStop.setText("Close app");
- buttonStop.setOnClickListener(new View.OnClickListener() {
- public void onClick(View v) {
-
- if (!_runAutotest) {
- ShutdownVoE();
- }
-
- // This call terminates and should close the activity
- finish();
-
- // playAudioFromFile();
- // if(!_playFromFile)
- // {
- // stopRecAudio();
- // }
- // stopPlayAudio();
- }
- });
-
-
- String ap1[] = {"EC off", "AECM"};
- final ArrayAdapter<String> adapterAp1 = new ArrayAdapter<String>(
- this,
- android.R.layout.simple_spinner_dropdown_item,
- ap1);
- String ap2[] =
- {"NS off", "NS low", "NS moderate", "NS high",
- "NS very high"};
- final ArrayAdapter<String> adapterAp2 = new ArrayAdapter<String>(
- this,
- android.R.layout.simple_spinner_dropdown_item,
- ap2);
- String ap3[] = {"AGC off", "AGC adaptive", "AGC fixed"};
- final ArrayAdapter<String> adapterAp3 = new ArrayAdapter<String>(
- this,
- android.R.layout.simple_spinner_dropdown_item,
- ap3);
- String ap4[] =
- {"VAD off", "VAD conventional", "VAD high rate",
- "VAD mid rate", "VAD low rate"};
- final ArrayAdapter<String> adapterAp4 = new ArrayAdapter<String>(
- this,
- android.R.layout.simple_spinner_dropdown_item,
- ap4);
- String codecs[] = {"iSAC", "PCMU", "PCMA", "iLBC"};
- final ArrayAdapter<String> adapterCodecs = new ArrayAdapter<String>(
- this,
- android.R.layout.simple_spinner_dropdown_item,
- codecs);
-
- final Spinner spinnerSettings1 = (Spinner) findViewById(R.id.Spinner01);
- final Spinner spinnerSettings2 = (Spinner) findViewById(R.id.Spinner02);
- spinnerSettings1.setMinimumWidth(200);
- String settings[] =
- {"Codec", "Echo Control", "Noise Suppression",
- "Automatic Gain Control",
- "Voice Activity Detection"};
- ArrayAdapter<String> adapterSettings1 = new ArrayAdapter<String>(
- this,
- android.R.layout.simple_spinner_dropdown_item,
- settings);
- spinnerSettings1.setAdapter(adapterSettings1);
- spinnerSettings1.setOnItemSelectedListener(
- new AdapterView.OnItemSelectedListener() {
- public void onItemSelected(AdapterView adapterView, View view,
- int position, long id) {
-
- _settingMenu = position;
- _settingSet = false;
- if (position == 0) {
- spinnerSettings2.setAdapter(adapterCodecs);
- spinnerSettings2.setSelection(_codecIndex);
- }
- if (position == 1) {
- spinnerSettings2.setAdapter(adapterAp1);
- spinnerSettings2.setSelection(_ecIndex);
- }
- if (position == 2) {
- spinnerSettings2.setAdapter(adapterAp2);
- spinnerSettings2.setSelection(_nsIndex);
- }
- if (position == 3) {
- spinnerSettings2.setAdapter(adapterAp3);
- spinnerSettings2.setSelection(_agcIndex);
- }
- if (position == 4) {
- spinnerSettings2.setAdapter(adapterAp4);
- spinnerSettings2.setSelection(_vadIndex);
- }
- }
-
- public void onNothingSelected(AdapterView adapterView) {
- WebrtcLog("No setting1 selected");
- }
- });
-
- spinnerSettings2.setMinimumWidth(200);
- ArrayAdapter<String> adapterSettings2 = new ArrayAdapter<String>(
- this,
- android.R.layout.simple_spinner_dropdown_item,
- codecs);
- spinnerSettings2.setAdapter(adapterSettings2);
- spinnerSettings2.setOnItemSelectedListener(
- new AdapterView.OnItemSelectedListener() {
- public void onItemSelected(AdapterView adapterView, View view,
- int position, long id) {
-
- // avoid unintentional setting
- if (_settingSet == false) {
- _settingSet = true;
- return;
- }
-
- // Change volume
- if (_settingMenu == 0) {
- WebrtcLog("Selected audio " + position);
- setAudioProperties(position);
- spinnerSettings2.setSelection(_audioIndex);
- }
-
- // Change codec
- if (_settingMenu == 1) {
- _codecIndex = position;
- WebrtcLog("Selected codec " + position);
- if (0 != SetSendCodec(_channel, _codecIndex)) {
- WebrtcLog("VoE set send codec failed");
- }
- }
-
- // Change EC
- if (_settingMenu == 2) {
- boolean enable = true;
- int ECmode = 5; // AECM
- int AESmode = 0;
-
- _ecIndex = position;
- WebrtcLog("Selected EC " + position);
-
- if (position == 0) {
- enable = false;
- }
- if (position > 1) {
- ECmode = 4; // AES
- AESmode = position - 1;
- }
-
- if (0 != SetECStatus(enable, ECmode)) {
- WebrtcLog("VoE set EC status failed");
- }
- }
-
- // Change NS
- if (_settingMenu == 3) {
- boolean enable = true;
-
- _nsIndex = position;
- WebrtcLog("Selected NS " + position);
-
- if (position == 0) {
- enable = false;
- }
- if (0 != SetNSStatus(enable, position + 2)) {
- WebrtcLog("VoE set NS status failed");
- }
- }
-
- // Change AGC
- if (_settingMenu == 4) {
- boolean enable = true;
-
- _agcIndex = position;
- WebrtcLog("Selected AGC " + position);
-
- if (position == 0) {
- enable = false;
- position = 1; // default
- }
- if (0 != SetAGCStatus(enable, position + 2)) {
- WebrtcLog("VoE set AGC status failed");
- }
- }
-
- // Change VAD
- if (_settingMenu == 5) {
- boolean enable = true;
-
- _vadIndex = position;
- WebrtcLog("Selected VAD " + position);
-
- if (position == 0) {
- enable = false;
- position++;
- }
- if (0 != SetVADStatus(_channel, enable, position - 1)) {
- WebrtcLog("VoE set VAD status failed");
- }
- }
- }
-
- public void onNothingSelected(AdapterView adapterView) {
- }
- });
-
- // Setup VoiceEngine
- if (!_runAutotest && !useNativeThread) SetupVoE();
-
- // Suggest to use the voice call audio stream for hardware volume
- // controls
- setVolumeControlStream(AudioManager.STREAM_VOICE_CALL);
-
- // Get max Android volume and adjust default volume to map exactly to an
- // Android level
- AudioManager am =
- (AudioManager) getSystemService(Context.AUDIO_SERVICE);
- _maxVolume = am.getStreamMaxVolume(AudioManager.STREAM_VOICE_CALL);
- if (_maxVolume <= 0) {
- WebrtcLog("Could not get max volume!");
- } else {
- int androidVolumeLevel = (_volumeLevel * _maxVolume) / 255;
- _volumeLevel = (androidVolumeLevel * 255) / _maxVolume;
- }
-
- WebrtcLog("Started Webrtc Android Test");
- }
-
- // Will be called when activity is shutdown.
- // NOTE: Activity may be killed without this function being called,
- // but then we should not need to clean up.
- protected void onDestroy() {
- super.onDestroy();
- // ShutdownVoE();
- }
-
- private void SetupVoE() {
- // Create VoiceEngine
- Create(); // Error logging is done in native API wrapper
-
- // Initialize
- if (0 != Init(false, false)) {
- WebrtcLog("VoE init failed");
- }
-
- // Create channel
- _channel = CreateChannel();
- if (0 != _channel) {
- WebrtcLog("VoE create channel failed");
- }
-
- }
-
- private void ShutdownVoE() {
- // Delete channel
- if (0 != DeleteChannel(_channel)) {
- WebrtcLog("VoE delete channel failed");
- }
-
- // Terminate
- if (0 != Terminate()) {
- WebrtcLog("VoE terminate failed");
- }
-
- // Delete VoiceEngine
- Delete(); // Error logging is done in native API wrapper
- }
-
- int startCall() {
-
- if (useNativeThread == true) {
-
- Create();
- return 0;
- }
-
- if (enableReceive == true) {
- // Set local receiver
- if (0 != SetLocalReceiver(_channel, _receivePort)) {
- WebrtcLog("VoE set local receiver failed");
- }
-
- if (0 != StartListen(_channel)) {
- WebrtcLog("VoE start listen failed");
- return -1;
- }
-
- // Route audio to earpiece
- if (0 != SetLoudspeakerStatus(false)) {
- WebrtcLog("VoE set louspeaker status failed");
- return -1;
- }
-
- /*
- * WebrtcLog("VoE start record now"); if (0 !=
- * StartRecordingPlayout(_channel, "/sdcard/singleUserDemoOut.pcm",
- * false)) { WebrtcLog("VoE Recording Playout failed"); }
- * WebrtcLog("VoE start Recording Playout end");
- */
- // Start playout
- if (0 != StartPlayout(_channel)) {
- WebrtcLog("VoE start playout failed");
- return -1;
- }
-
- // Start playout file
- // if (0 != StartPlayingFileLocally(_channel,
- // "/sdcard/singleUserDemo.pcm", true)) {
- // WebrtcLog("VoE start playout file failed");
- // return -1;
- // }
- }
-
- if (enableSend == true) {
- if (0 != SetSendDestination(_channel, _destinationPort,
- _destinationIP)) {
- WebrtcLog("VoE set send destination failed");
- return -1;
- }
-
- if (0 != SetSendCodec(_channel, _codecIndex)) {
- WebrtcLog("VoE set send codec failed");
- return -1;
- }
-
- /*
- * if (0 != StartPlayingFileAsMicrophone(_channel,
- * "/sdcard/singleUserDemo.pcm", true)) {
- * WebrtcLog("VoE start playing file as microphone failed"); }
- */
- if (0 != StartSend(_channel)) {
- WebrtcLog("VoE start send failed");
- return -1;
- }
-
- // if (0 != StartPlayingFileAsMicrophone(_channel,
- // "/sdcard/singleUserDemo.pcm", true)) {
- // WebrtcLog("VoE start playing file as microphone failed");
- // return -1;
- // }
- }
-
- return 0;
- }
-
- int stopCall() {
-
- if (useNativeThread == true) {
-
- Delete();
- return 0;
- }
-
- if (enableSend == true) {
- // Stop playing file as microphone
- /*
- * if (0 != StopPlayingFileAsMicrophone(_channel)) {
- * WebrtcLog("VoE stop playing file as microphone failed"); return
- * -1; }
- */
- // Stop send
- if (0 != StopSend(_channel)) {
- WebrtcLog("VoE stop send failed");
- return -1;
- }
- }
-
- if (enableReceive == true) {
- // if (0 != StopRecordingPlayout(_channel)) {
- // WebrtcLog("VoE stop Recording Playout failed");
- // }
- // WebrtcLog("VoE stop Recording Playout ended");
-
- // Stop listen
- if (0 != StopListen(_channel)) {
- WebrtcLog("VoE stop listen failed");
- return -1;
- }
-
- // Stop playout file
- // if (0 != StopPlayingFileLocally(_channel)) {
- // WebrtcLog("VoE stop playout file failed");
- // return -1;
- // }
-
- // Stop playout
- if (0 != StopPlayout(_channel)) {
- WebrtcLog("VoE stop playout failed");
- return -1;
- }
-
- // Route audio to loudspeaker
- if (0 != SetLoudspeakerStatus(true)) {
- WebrtcLog("VoE set louspeaker status failed");
- return -1;
- }
- }
-
- return 0;
- }
-
- int startAutoTest() {
-
- _autotestThread = new Thread(_autotestProc);
- _autotestThread.start();
-
- return 0;
- }
-
- private Runnable _autotestProc = new Runnable() {
- public void run() {
- // TODO(xians): choose test from GUI
- // 1 = standard, not used
- // 2 = extended, 2 = base
- RunAutoTest(1, 2);
- }
- };
-
- int setAudioProperties(int val) {
-
- // AudioManager am = (AudioManager)
- // getSystemService(Context.AUDIO_SERVICE);
-
- if (val == 0) {
- // _streamVolume =
- // am.getStreamVolume(AudioManager.STREAM_VOICE_CALL);
- // am.setStreamVolume(AudioManager.STREAM_VOICE_CALL,
- // (_streamVolume+1), 0);
-
- int androidVolumeLevel = (_volumeLevel * _maxVolume) / 255;
- if (androidVolumeLevel < _maxVolume) {
- _volumeLevel = ((androidVolumeLevel + 1) * 255) / _maxVolume;
- if (0 != SetSpeakerVolume(_volumeLevel)) {
- WebrtcLog("VoE set speaker volume failed");
- }
- }
- } else if (val == 1) {
- // _streamVolume =
- // am.getStreamVolume(AudioManager.STREAM_VOICE_CALL);
- // am.setStreamVolume(AudioManager.STREAM_VOICE_CALL,
- // (_streamVolume-1), 0);
-
- int androidVolumeLevel = (_volumeLevel * _maxVolume) / 255;
- if (androidVolumeLevel > 0) {
- _volumeLevel = ((androidVolumeLevel - 1) * 255) / _maxVolume;
- if (0 != SetSpeakerVolume(_volumeLevel)) {
- WebrtcLog("VoE set speaker volume failed");
- }
- }
- } else if (val == 2) {
- // route audio to back speaker
- if (0 != SetLoudspeakerStatus(true)) {
- WebrtcLog("VoE set loudspeaker status failed");
- }
- _audioIndex = 2;
- } else if (val == 3) {
- // route audio to earpiece
- if (0 != SetLoudspeakerStatus(false)) {
- WebrtcLog("VoE set loudspeaker status failed");
- }
- _audioIndex = 3;
- }
-
- return 0;
- }
-
- int displayTextFromFile() {
-
- TextView tv = (TextView) findViewById(R.id.TextView01);
- FileReader fr = null;
- char[] fileBuffer = new char[64];
-
- try {
- fr = new FileReader("/sdcard/test.txt");
- } catch (FileNotFoundException e) {
- e.printStackTrace();
- tv.setText("File not found!");
- }
-
- try {
- fr.read(fileBuffer);
- } catch (IOException e) {
- e.printStackTrace();
- }
-
- String readString = new String(fileBuffer);
- tv.setText(readString);
- // setContentView(tv);
-
- return 0;
- }
-
- int recordAudioToFile() {
- File fr = null;
- // final to be reachable within onPeriodicNotification
- byte[] recBuffer = new byte[320];
-
- int recBufSize =
- AudioRecord.getMinBufferSize(16000,
- AudioFormat.CHANNEL_CONFIGURATION_MONO,
- AudioFormat.ENCODING_PCM_16BIT);
- AudioRecord rec =
- new AudioRecord(MediaRecorder.AudioSource.MIC, 16000,
- AudioFormat.CHANNEL_CONFIGURATION_MONO,
- AudioFormat.ENCODING_PCM_16BIT,
- recBufSize);
-
- fr = new File("/sdcard/record.pcm");
- FileOutputStream out = null;
- try {
- out = new FileOutputStream(fr);
- } catch (FileNotFoundException e1) {
- e1.printStackTrace();
- }
-
- // start recording
- try {
- rec.startRecording();
- } catch (IllegalStateException e) {
- e.printStackTrace();
- }
-
- for (int i = 0; i < 550; i++) {
- // note, there is a short version of write as well!
- int wrBytes = rec.read(recBuffer, 0, 320);
-
- try {
- out.write(recBuffer);
- } catch (IOException e) {
- e.printStackTrace();
- }
- }
-
- // stop playout
- try {
- rec.stop();
- } catch (IllegalStateException e) {
- e.printStackTrace();
- }
-
- return 0;
- }
-
- int playAudioFromFile() {
-
- File fr = null;
- // final to be reachable within onPeriodicNotification
- // final byte[] playBuffer = new byte [320000];
- // final to be reachable within onPeriodicNotification
- final byte[] playBuffer = new byte[320];
-
- final int playBufSize =
- AudioTrack.getMinBufferSize(16000,
- AudioFormat.CHANNEL_CONFIGURATION_MONO,
- AudioFormat.ENCODING_PCM_16BIT);
- // final int playBufSize = 1920; // 100 ms buffer
- // byte[] playBuffer = new byte [playBufSize];
- final AudioTrack play =
- new AudioTrack(AudioManager.STREAM_VOICE_CALL, 16000,
- AudioFormat.CHANNEL_CONFIGURATION_MONO,
- AudioFormat.ENCODING_PCM_16BIT,
- playBufSize, AudioTrack.MODE_STREAM);
-
- // implementation of the playpos callback functions
- play.setPlaybackPositionUpdateListener(
- new AudioTrack.OnPlaybackPositionUpdateListener() {
-
- int count = 0;
-
- public void onPeriodicNotification(AudioTrack track) {
- // int wrBytes = play.write(playBuffer, count, 320);
- count += 320;
- }
-
- public void onMarkerReached(AudioTrack track) {
-
- }
- });
-
- // set the notification period = 160 samples
- // int ret = play.setPositionNotificationPeriod(160);
-
- fr = new File("/sdcard/record.pcm");
- FileInputStream in = null;
- try {
- in = new FileInputStream(fr);
- } catch (FileNotFoundException e1) {
- e1.printStackTrace();
- }
-
- // try {
- // in.read(playBuffer);
- // } catch (IOException e) {
- // e.printStackTrace();
- // }
-
- // play all at once
- // int wrBytes = play.write(playBuffer, 0, 320000);
-
-
- // start playout
- try {
- play.play();
- } catch (IllegalStateException e) {
- e.printStackTrace();
- }
-
- // returns the number of samples that has been written
- // int headPos = play.getPlaybackHeadPosition();
-
- // play with multiple writes
- for (int i = 0; i < 500; i++) {
- try {
- in.read(playBuffer);
- } catch (IOException e) {
- e.printStackTrace();
- }
-
-
- // note, there is a short version of write as well!
- int wrBytes = play.write(playBuffer, 0, 320);
-
- Log.d("testWrite", "wrote");
- }
-
- // stop playout
- try {
- play.stop();
- } catch (IllegalStateException e) {
- e.printStackTrace();
- }
-
- return 0;
- }
-
- int playAudioInThread() {
-
- if (_isRunningPlay) {
- return 0;
- }
-
- // File fr = null;
- // final byte[] playBuffer = new byte[320];
- if (_playFromFile) {
- _playBuffer = new byte[320];
- } else {
- // reset index
- _playIndex = 0;
- }
- // within
- // onPeriodicNotification
-
- // Log some info (static)
- WebrtcLog("Creating AudioTrack object");
- final int minPlayBufSize =
- AudioTrack.getMinBufferSize(16000,
- AudioFormat.CHANNEL_CONFIGURATION_MONO,
- AudioFormat.ENCODING_PCM_16BIT);
- WebrtcLog("Min play buf size = " + minPlayBufSize);
- WebrtcLog("Min volume = " + AudioTrack.getMinVolume());
- WebrtcLog("Max volume = " + AudioTrack.getMaxVolume());
- WebrtcLog("Native sample rate = "
- + AudioTrack.getNativeOutputSampleRate(
- AudioManager.STREAM_VOICE_CALL));
-
- final int playBufSize = minPlayBufSize; // 3200; // 100 ms buffer
- // byte[] playBuffer = new byte [playBufSize];
- try {
- _at = new AudioTrack(
- AudioManager.STREAM_VOICE_CALL,
- 16000,
- AudioFormat.CHANNEL_CONFIGURATION_MONO,
- AudioFormat.ENCODING_PCM_16BIT,
- playBufSize, AudioTrack.MODE_STREAM);
- } catch (Exception e) {
- WebrtcLog(e.getMessage());
- }
-
- // Log some info (non-static)
- WebrtcLog("Notification marker pos = "
- + _at.getNotificationMarkerPosition());
- WebrtcLog("Play head pos = " + _at.getPlaybackHeadPosition());
- WebrtcLog("Pos notification dt = "
- + _at.getPositionNotificationPeriod());
- WebrtcLog("Playback rate = " + _at.getPlaybackRate());
- WebrtcLog("Sample rate = " + _at.getSampleRate());
-
- // implementation of the playpos callback functions
- // _at.setPlaybackPositionUpdateListener(
- // new AudioTrack.OnPlaybackPositionUpdateListener() {
- //
- // int count = 3200;
- //
- // public void onPeriodicNotification(AudioTrack track) {
- // // int wrBytes = play.write(playBuffer, count, 320);
- // count += 320;
- // }
- //
- // public void onMarkerReached(AudioTrack track) {
- // }
- // });
-
- // set the notification period = 160 samples
- // int ret = _at.setPositionNotificationPeriod(160);
-
- if (_playFromFile) {
- _fr = new File("/sdcard/singleUserDemo.pcm");
- try {
- _in = new FileInputStream(_fr);
- } catch (FileNotFoundException e1) {
- e1.printStackTrace();
- }
- }
-
- // try {
- // in.read(playBuffer);
- // } catch (IOException e) {
- // e.printStackTrace();
- // }
-
- _isRunningPlay = true;
-
- // buffer = new byte[3200];
- _playThread = new Thread(_playProc);
- // ar.startRecording();
- // bytesRead = 3200;
- // recording = true;
- _playThread.start();
-
- return 0;
- }
-
- int stopPlayAudio() {
- if (!_isRunningPlay) {
- return 0;
- }
-
- _isRunningPlay = false;
-
- return 0;
- }
-
- private Runnable _playProc = new Runnable() {
- public void run() {
-
- // set high thread priority
- android.os.Process.setThreadPriority(
- android.os.Process.THREAD_PRIORITY_URGENT_AUDIO);
-
- // play all at once
- // int wrBytes = play.write(playBuffer, 0, 320000);
-
- // fill the buffer
- // play.write(playBuffer, 0, 3200);
-
- // play.flush();
-
- // start playout
- try {
- _at.play();
- } catch (IllegalStateException e) {
- e.printStackTrace();
- }
-
- // play with multiple writes
- int i = 0;
- for (; i < 3000 && _isRunningPlay; i++) {
-
- if (_playFromFile) {
- try {
- _in.read(_playBuffer);
- } catch (IOException e) {
- e.printStackTrace();
- }
-
- int wrBytes = _at.write(_playBuffer, 0 /* i * 320 */, 320);
- } else {
- int wrSamples =
- _at.write(_circBuffer, _playIndex * 160,
- 160);
-
- // WebrtcLog("Played 10 ms from buffer, _playIndex = " +
- // _playIndex);
- // WebrtcLog("Diff = " + (_recIndex - _playIndex));
-
- if (_playIndex == 49) {
- _playIndex = 0;
- } else {
- _playIndex += 1;
- }
- }
-
- // WebrtcLog("Wrote 10 ms to buffer, head = "
- // + _at.getPlaybackHeadPosition());
- }
-
- // stop playout
- try {
- _at.stop();
- } catch (IllegalStateException e) {
- e.printStackTrace();
- }
-
- // returns the number of samples that has been written
- WebrtcLog("Test stopped, i = " + i + ", head = "
- + _at.getPlaybackHeadPosition());
- int headPos = _at.getPlaybackHeadPosition();
-
- // flush the buffers
- _at.flush();
-
- // release the object
- _at.release();
- _at = null;
-
- // try {
- // Thread.sleep() must be within a try - catch block
- // Thread.sleep(3000);
- // }catch (Exception e){
- // System.out.println(e.getMessage());
- // }
-
- _isRunningPlay = false;
-
- }
- };
-
- int recAudioInThread() {
-
- if (_isRunningRec) {
- return 0;
- }
-
- // within
- // onPeriodicNotification
-
- // reset index
- _recIndex = 20;
-
- // Log some info (static)
- WebrtcLog("Creating AudioRecord object");
- final int minRecBufSize = AudioRecord.getMinBufferSize(16000,
- AudioFormat.CHANNEL_CONFIGURATION_MONO,
- AudioFormat.ENCODING_PCM_16BIT);
- WebrtcLog("Min rec buf size = " + minRecBufSize);
- // WebrtcLog("Min volume = " + AudioTrack.getMinVolume());
- // WebrtcLog("Max volume = " + AudioTrack.getMaxVolume());
- // WebrtcLog("Native sample rate = "
- // + AudioRecord
- // .getNativeInputSampleRate(AudioManager.STREAM_VOICE_CALL));
-
- final int recBufSize = minRecBufSize; // 3200; // 100 ms buffer
- try {
- _ar = new AudioRecord(
- MediaRecorder.AudioSource.MIC,
- 16000,
- AudioFormat.CHANNEL_CONFIGURATION_MONO,
- AudioFormat.ENCODING_PCM_16BIT,
- recBufSize);
- } catch (Exception e) {
- WebrtcLog(e.getMessage());
- }
-
- // Log some info (non-static)
- WebrtcLog("Notification marker pos = "
- + _ar.getNotificationMarkerPosition());
- // WebrtcLog("Play head pos = " + _ar.getRecordHeadPosition());
- WebrtcLog("Pos notification dt rec= "
- + _ar.getPositionNotificationPeriod());
- // WebrtcLog("Playback rate = " + _ar.getRecordRate());
- // WebrtcLog("Playback rate = " + _ar.getPlaybackRate());
- WebrtcLog("Sample rate = " + _ar.getSampleRate());
- // WebrtcLog("Playback rate = " + _ar.getPlaybackRate());
- // WebrtcLog("Playback rate = " + _ar.getPlaybackRate());
-
- _isRunningRec = true;
-
- _recThread = new Thread(_recProc);
-
- _recThread.start();
-
- return 0;
- }
-
- int stopRecAudio() {
- if (!_isRunningRec) {
- return 0;
- }
-
- _isRunningRec = false;
-
- return 0;
- }
-
- private Runnable _recProc = new Runnable() {
- public void run() {
-
- // set high thread priority
- android.os.Process.setThreadPriority(
- android.os.Process.THREAD_PRIORITY_URGENT_AUDIO);
-
- // start recording
- try {
- _ar.startRecording();
- } catch (IllegalStateException e) {
- e.printStackTrace();
- }
-
- // keep recording to circular buffer
- // for a while
- int i = 0;
- int rdSamples = 0;
- short[] tempBuffer = new short[160]; // Only used for native case
-
- for (; i < 3000 && _isRunningRec; i++) {
- if (_runThroughNativeLayer) {
- rdSamples = _ar.read(tempBuffer, 0, 160);
- // audioLoop(tempBuffer, 160); // Insert into native layer
- } else {
- rdSamples = _ar.read(_circBuffer, _recIndex * 160, 160);
-
- // WebrtcLog("Recorded 10 ms to buffer, _recIndex = " +
- // _recIndex);
- // WebrtcLog("rdSamples = " + rdSamples);
-
- if (_recIndex == 49) {
- _recIndex = 0;
- } else {
- _recIndex += 1;
- }
- }
- }
-
- // stop recording
- try {
- _ar.stop();
- } catch (IllegalStateException e) {
- e.printStackTrace();
- }
-
- // release the object
- _ar.release();
- _ar = null;
-
- // try {
- // Thread.sleep() must be within a try - catch block
- // Thread.sleep(3000);
- // }catch (Exception e){
- // System.out.println(e.getMessage());
- // }
-
- _isRunningRec = false;
-
- // returns the number of samples that has been written
- // WebrtcLog("Test stopped, i = " + i + ", head = "
- // + _at.getPlaybackHeadPosition());
- // int headPos = _at.getPlaybackHeadPosition();
- }
- };
-
- private void WebrtcLog(String msg) {
- Log.d("*Webrtc*", msg);
- }
-
- // //////////////// Native function prototypes ////////////////////
-
- private native static boolean NativeInit();
-
- private native int RunAutoTest(int testType, int extendedSel);
-
- private native boolean Create();
-
- private native boolean Delete();
-
- private native int Init(boolean enableTrace, boolean useExtTrans);
-
- private native int Terminate();
-
- private native int CreateChannel();
-
- private native int DeleteChannel(int channel);
-
- private native int SetLocalReceiver(int channel, int port);
-
- private native int SetSendDestination(int channel, int port,
- String ipaddr);
-
- private native int StartListen(int channel);
-
- private native int StartPlayout(int channel);
-
- private native int StartSend(int channel);
-
- private native int StopListen(int channel);
-
- private native int StopPlayout(int channel);
-
- private native int StopSend(int channel);
-
- private native int StartPlayingFileLocally(int channel, String fileName,
- boolean loop);
-
- private native int StopPlayingFileLocally(int channel);
-
- private native int StartRecordingPlayout(int channel, String fileName,
- boolean loop);
-
- private native int StopRecordingPlayout(int channel);
-
- private native int StartPlayingFileAsMicrophone(int channel,
- String fileName, boolean loop);
-
- private native int StopPlayingFileAsMicrophone(int channel);
-
- private native int NumOfCodecs();
-
- private native int SetSendCodec(int channel, int index);
-
- private native int SetVADStatus(int channel, boolean enable, int mode);
-
- private native int SetNSStatus(boolean enable, int mode);
-
- private native int SetAGCStatus(boolean enable, int mode);
-
- private native int SetECStatus(boolean enable, int mode);
-
- private native int SetSpeakerVolume(int volume);
-
- private native int SetLoudspeakerStatus(boolean enable);
-
- /*
- * this is used to load the 'webrtc-voice-demo-jni'
- * library on application startup.
- * The library has already been unpacked into
- * /data/data/webrtc.android.AndroidTest/lib/libwebrtc-voice-demo-jni.so
- * at installation time by the package manager.
- */
- static {
- Log.d("*Webrtc*", "Loading webrtc-voice-demo-jni...");
- System.loadLibrary("webrtc-voice-demo-jni");
-
- Log.d("*Webrtc*", "Calling native init...");
- if (!NativeInit()) {
- Log.e("*Webrtc*", "Native init failed");
- throw new RuntimeException("Native init failed");
- } else {
- Log.d("*Webrtc*", "Native init successful");
- }
- }
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/Android.mk b/trunk/src/voice_engine/main/test/auto_test/Android.mk
deleted file mode 100644
index cfb9d67..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/Android.mk
+++ /dev/null
@@ -1,56 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH:= $(call my-dir)
-
-# voice engine test app
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../android-webrtc.mk
-
-LOCAL_MODULE_TAGS := tests
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES:= \
- automated_mode.cc \
- voe_cpu_test.cc \
- voe_standard_test.cc \
- voe_stress_test.cc \
- voe_unit_test.cc \
- voe_extended_test.cc \
- voe_standard_integration_test.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_LINUX' \
- '-DWEBRTC_ANDROID'
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../../interface \
- $(LOCAL_PATH)/../../../.. \
- $(LOCAL_PATH)/../../../../modules/audio_device/main/interface \
- $(LOCAL_PATH)/../../../../modules/interface \
- $(LOCAL_PATH)/../../../../system_wrappers/interface \
- $(LOCAL_PATH)/../../../../../test \
- external/gtest/include \
-
-LOCAL_SHARED_LIBRARIES := \
- libutils \
- libstlport \
- libwebrtc
-
-LOCAL_MODULE:= webrtc_voe_autotest
-
-ifdef NDK_ROOT
-include $(BUILD_EXECUTABLE)
-else
-include external/stlport/libstlport.mk
-include $(BUILD_NATIVE_TEST)
-endif
-
diff --git a/trunk/src/voice_engine/main/test/auto_test/automated_mode.cc b/trunk/src/voice_engine/main/test/auto_test/automated_mode.cc
deleted file mode 100644
index 13fa257..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/automated_mode.cc
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "gtest/gtest.h"
-
-int RunInAutomatedMode(int argc, char** argv) {
- testing::InitGoogleTest(&argc, argv);
-
- return RUN_ALL_TESTS();
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/automated_mode.h b/trunk/src/voice_engine/main/test/auto_test/automated_mode.h
deleted file mode 100644
index cd7ab9e..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/automated_mode.h
+++ /dev/null
@@ -1,16 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_AUTOMATED_MODE_H_
-#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_AUTOMATED_MODE_H_
-
-int RunInAutomatedMode(int argc, char** argv);
-
-#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_AUTOMATED_MODE_H_
diff --git a/trunk/src/voice_engine/main/test/auto_test/fixtures/after_initialization_fixture.cc b/trunk/src/voice_engine/main/test/auto_test/fixtures/after_initialization_fixture.cc
deleted file mode 100644
index f0e665b..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/fixtures/after_initialization_fixture.cc
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "after_initialization_fixture.h"
-
-class TestErrorObserver : public webrtc::VoiceEngineObserver {
- public:
- TestErrorObserver() {}
- virtual ~TestErrorObserver() {}
- void CallbackOnError(const int channel, const int error_code) {
- ADD_FAILURE() << "Unexpected error on channel " << channel <<
- ": error code " << error_code;
- }
-};
-
-AfterInitializationFixture::AfterInitializationFixture()
- : error_observer_(new TestErrorObserver()) {
- EXPECT_EQ(0, voe_base_->Init());
-
-#if defined(WEBRTC_ANDROID)
- EXPECT_EQ(0, voe_hardware_->SetLoudspeakerStatus(false));
-#endif
-
- EXPECT_EQ(0, voe_base_->RegisterVoiceEngineObserver(*error_observer_));
-}
-
-AfterInitializationFixture::~AfterInitializationFixture() {
- EXPECT_EQ(0, voe_base_->DeRegisterVoiceEngineObserver());
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/fixtures/after_initialization_fixture.h b/trunk/src/voice_engine/main/test/auto_test/fixtures/after_initialization_fixture.h
deleted file mode 100644
index bbdd64d..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/fixtures/after_initialization_fixture.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
-#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
-
-#include "before_initialization_fixture.h"
-#include "scoped_ptr.h"
-
-class TestErrorObserver;
-
-// This fixture initializes the voice engine in addition to the work
-// done by the before-initialization fixture. It also registers an error
-// observer which will fail tests on error callbacks. This fixture is
-// useful to tests that want to run before we have started any form of
-// streaming through the voice engine.
-class AfterInitializationFixture : public BeforeInitializationFixture {
- public:
- AfterInitializationFixture();
- virtual ~AfterInitializationFixture();
- protected:
- webrtc::scoped_ptr<TestErrorObserver> error_observer_;
-};
-
-#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
diff --git a/trunk/src/voice_engine/main/test/auto_test/fixtures/after_streaming_fixture.cc b/trunk/src/voice_engine/main/test/auto_test/fixtures/after_streaming_fixture.cc
deleted file mode 100644
index e1a4820..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/fixtures/after_streaming_fixture.cc
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "after_streaming_fixture.h"
-
-#include <cstring>
-
-static const char* kLoopbackIp = "127.0.0.1";
-
-AfterStreamingFixture::AfterStreamingFixture()
- : channel_(voe_base_->CreateChannel()) {
- EXPECT_GE(channel_, 0);
-
- fake_microphone_input_file_ = resource_manager_.long_audio_file_path();
- EXPECT_FALSE(fake_microphone_input_file_.empty());
-
- SetUpLocalPlayback();
- StartPlaying();
-}
-
-AfterStreamingFixture::~AfterStreamingFixture() {
- voe_file_->StopPlayingFileAsMicrophone(channel_);
- voe_base_->StopSend(channel_);
- voe_base_->StopPlayout(channel_);
- voe_base_->StopReceive(channel_);
-
- voe_base_->DeleteChannel(channel_);
-}
-
-void AfterStreamingFixture::SwitchToManualMicrophone() {
- EXPECT_EQ(0, voe_file_->StopPlayingFileAsMicrophone(channel_));
-
- TEST_LOG("You need to speak manually into the microphone for this test.\n");
- TEST_LOG("Please start speaking now.\n");
- Sleep(2000);
-}
-
-void AfterStreamingFixture::RestartFakeMicrophone() {
- EXPECT_EQ(0, voe_file_->StartPlayingFileAsMicrophone(
- channel_, fake_microphone_input_file_.c_str(), true, true));
-}
-
-void AfterStreamingFixture::SetUpLocalPlayback() {
- EXPECT_EQ(0, voe_base_->SetSendDestination(channel_, 8000, kLoopbackIp));
- EXPECT_EQ(0, voe_base_->SetLocalReceiver(0, 8000));
-
- webrtc::CodecInst codec;
- codec.channels = 1;
- codec.pacsize = 160;
- codec.plfreq = 8000;
- codec.pltype = 0;
- codec.rate = 64000;
- strcpy(codec.plname, "PCMU");
-
- voe_codec_->SetSendCodec(channel_, codec);
-}
-
-void AfterStreamingFixture::StartPlaying() {
- EXPECT_EQ(0, voe_base_->StartReceive(channel_));
- EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
- RestartFakeMicrophone();
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/fixtures/after_streaming_fixture.h b/trunk/src/voice_engine/main/test/auto_test/fixtures/after_streaming_fixture.h
deleted file mode 100644
index c589a4c..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/fixtures/after_streaming_fixture.h
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_AFTER_STREAMING_H_
-#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_AFTER_STREAMING_H_
-
-#include "after_initialization_fixture.h"
-#include "resource_manager.h"
-
-// This fixture will, in addition to the work done by its superclasses,
-// create a channel and start playing a file through the fake microphone
-// to simulate microphone input. The purpose is to make it convenient
-// to write tests that require microphone input.
-class AfterStreamingFixture : public AfterInitializationFixture {
- public:
- AfterStreamingFixture();
- virtual ~AfterStreamingFixture();
-
- protected:
- int channel_;
- ResourceManager resource_manager_;
- std::string fake_microphone_input_file_;
-
- // Shuts off the fake microphone for this test.
- void SwitchToManualMicrophone();
-
- // Restarts the fake microphone if it's been shut off earlier.
- void RestartFakeMicrophone();
-
- private:
- void SetUpLocalPlayback();
- void StartPlaying();
-};
-
-
-#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_AFTER_STREAMING_H_
diff --git a/trunk/src/voice_engine/main/test/auto_test/fixtures/before_initialization_fixture.cc b/trunk/src/voice_engine/main/test/auto_test/fixtures/before_initialization_fixture.cc
deleted file mode 100644
index 408141b..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/fixtures/before_initialization_fixture.cc
+++ /dev/null
@@ -1,58 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "before_initialization_fixture.h"
-
-#include "voice_engine_defines.h"
-
-BeforeInitializationFixture::BeforeInitializationFixture()
- : voice_engine_(webrtc::VoiceEngine::Create()) {
- EXPECT_TRUE(voice_engine_ != NULL);
-
- voe_base_ = webrtc::VoEBase::GetInterface(voice_engine_);
- voe_codec_ = webrtc::VoECodec::GetInterface(voice_engine_);
- voe_volume_control_ = webrtc::VoEVolumeControl::GetInterface(voice_engine_);
- voe_dtmf_ = webrtc::VoEDtmf::GetInterface(voice_engine_);
- voe_rtp_rtcp_ = webrtc::VoERTP_RTCP::GetInterface(voice_engine_);
- voe_apm_ = webrtc::VoEAudioProcessing::GetInterface(voice_engine_);
- voe_network_ = webrtc::VoENetwork::GetInterface(voice_engine_);
- voe_file_ = webrtc::VoEFile::GetInterface(voice_engine_);
- voe_vsync_ = webrtc::VoEVideoSync::GetInterface(voice_engine_);
- voe_encrypt_ = webrtc::VoEEncryption::GetInterface(voice_engine_);
- voe_hardware_ = webrtc::VoEHardware::GetInterface(voice_engine_);
- voe_xmedia_ = webrtc::VoEExternalMedia::GetInterface(voice_engine_);
- voe_call_report_ = webrtc::VoECallReport::GetInterface(voice_engine_);
- voe_neteq_stats_ = webrtc::VoENetEqStats::GetInterface(voice_engine_);
-}
-
-BeforeInitializationFixture::~BeforeInitializationFixture() {
- EXPECT_EQ(0, voe_base_->Release());
- EXPECT_EQ(0, voe_codec_->Release());
- EXPECT_EQ(0, voe_volume_control_->Release());
- EXPECT_EQ(0, voe_dtmf_->Release());
- EXPECT_EQ(0, voe_rtp_rtcp_->Release());
- EXPECT_EQ(0, voe_apm_->Release());
- EXPECT_EQ(0, voe_network_->Release());
- EXPECT_EQ(0, voe_file_->Release());
- EXPECT_EQ(0, voe_vsync_->Release());
- EXPECT_EQ(0, voe_encrypt_->Release());
- EXPECT_EQ(0, voe_hardware_->Release());
- EXPECT_EQ(0, voe_xmedia_->Release());
- EXPECT_EQ(0, voe_call_report_->Release());
- EXPECT_EQ(0, voe_neteq_stats_->Release());
-
- EXPECT_TRUE(webrtc::VoiceEngine::Delete(voice_engine_));
-}
-
-void BeforeInitializationFixture::Sleep(long milliseconds) {
- // Implementation note: This method is used to reduce usage of the macro and
- // avoid ugly errors in Eclipse (its parser can't deal with the sleep macro).
- SLEEP(milliseconds);
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/fixtures/before_initialization_fixture.h b/trunk/src/voice_engine/main/test/auto_test/fixtures/before_initialization_fixture.h
deleted file mode 100644
index ef1636b..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/fixtures/before_initialization_fixture.h
+++ /dev/null
@@ -1,79 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_H_
-#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_H_
-
-#include <assert.h>
-
-#include "common_types.h"
-#include "engine_configurations.h"
-#include "voe_audio_processing.h"
-#include "voe_base.h"
-#include "voe_call_report.h"
-#include "voe_codec.h"
-#include "voe_dtmf.h"
-#include "voe_encryption.h"
-#include "voe_errors.h"
-#include "voe_external_media.h"
-#include "voe_file.h"
-#include "voe_hardware.h"
-#include "voe_neteq_stats.h"
-#include "voe_network.h"
-#include "voe_rtp_rtcp.h"
-#include "voe_test_defines.h"
-#include "voe_video_sync.h"
-#include "voe_volume_control.h"
-
-// TODO(qhogpat): Remove these undefs once the clashing macros are gone.
-#undef TEST
-#undef ASSERT_TRUE
-#undef ASSERT_FALSE
-#include "gtest/gtest.h"
-#include "gmock/gmock.h"
-
-// This convenient fixture sets up all voice engine interfaces automatically for
-// use by testing subclasses. It allocates each interface and releases it once
-// which means that if a tests allocates additional interfaces from the voice
-// engine and forgets to release it, this test will fail in the destructor.
-// It will not call any init methods.
-//
-// Implementation note:
-// The interface fetching is done in the constructor and not SetUp() since
-// this relieves our subclasses from calling SetUp in the superclass if they
-// choose to override SetUp() themselves. This is fine as googletest will
-// construct new test objects for each method.
-class BeforeInitializationFixture : public testing::Test {
- public:
- BeforeInitializationFixture();
- virtual ~BeforeInitializationFixture();
-
- protected:
- // Use this sleep function to sleep in test (avoid sleep macro).
- void Sleep(long milliseconds);
-
- webrtc::VoiceEngine* voice_engine_;
- webrtc::VoEBase* voe_base_;
- webrtc::VoECodec* voe_codec_;
- webrtc::VoEVolumeControl* voe_volume_control_;
- webrtc::VoEDtmf* voe_dtmf_;
- webrtc::VoERTP_RTCP* voe_rtp_rtcp_;
- webrtc::VoEAudioProcessing* voe_apm_;
- webrtc::VoENetwork* voe_network_;
- webrtc::VoEFile* voe_file_;
- webrtc::VoEVideoSync* voe_vsync_;
- webrtc::VoEEncryption* voe_encrypt_;
- webrtc::VoEHardware* voe_hardware_;
- webrtc::VoEExternalMedia* voe_xmedia_;
- webrtc::VoECallReport* voe_call_report_;
- webrtc::VoENetEqStats* voe_neteq_stats_;
-};
-
-#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_H_
diff --git a/trunk/src/voice_engine/main/test/auto_test/resource_manager.cc b/trunk/src/voice_engine/main/test/auto_test/resource_manager.cc
deleted file mode 100644
index 65669ed..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/resource_manager.cc
+++ /dev/null
@@ -1,29 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "resource_manager.h"
-
-#include "testsupport/fileutils.h"
-
-ResourceManager::ResourceManager() {
- std::string filename = "audio_long16.pcm";
-#if defined(WEBRTC_ANDROID)
- long_audio_file_path_ = "/sdcard/" + filename;
-#else
- std::string resource_path = webrtc::test::ProjectRootPath();
- if (resource_path == webrtc::test::kCannotFindProjectRootDir) {
- long_audio_file_path_ = "";
- } else {
- long_audio_file_path_ =
- resource_path + "test/data/voice_engine/" + filename;
- }
-#endif
-}
-
diff --git a/trunk/src/voice_engine/main/test/auto_test/resource_manager.h b/trunk/src/voice_engine/main/test/auto_test/resource_manager.h
deleted file mode 100644
index 1bb91cf..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/resource_manager.h
+++ /dev/null
@@ -1,30 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_RESOURCE_MANAGER_H_
-#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_RESOURCE_MANAGER_H_
-
-#include <string>
-
-class ResourceManager {
- public:
- ResourceManager();
-
- // Returns the full path to a long audio file.
- // Returns the empty string on failure.
- const std::string& long_audio_file_path() const {
- return long_audio_file_path_;
- }
-
- private:
- std::string long_audio_file_path_;
-};
-
-#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_RESOURCE_MANAGER_H_
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/audio_processing_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/audio_processing_test.cc
deleted file mode 100644
index bfe7f49..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/standard/audio_processing_test.cc
+++ /dev/null
@@ -1,364 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "after_streaming_fixture.h"
-#include "voe_standard_test.h"
-
-class AudioProcessingTest : public AfterStreamingFixture {
- protected:
- // Note: Be careful with this one, it is used in the
- // Android / iPhone part too.
- void TryEnablingAgcWithMode(webrtc::AgcModes agc_mode_to_set) {
- EXPECT_EQ(0, voe_apm_->SetAgcStatus(true, agc_mode_to_set));
-
- bool agc_enabled = false;
- webrtc::AgcModes agc_mode = webrtc::kAgcDefault;
-
- EXPECT_EQ(0, voe_apm_->GetAgcStatus(agc_enabled, agc_mode));
- EXPECT_TRUE(agc_enabled);
- EXPECT_EQ(agc_mode_to_set, agc_mode);
- }
-
- void TryEnablingRxAgcWithMode(webrtc::AgcModes agc_mode_to_set) {
- EXPECT_EQ(0, voe_apm_->SetRxAgcStatus(channel_, true, agc_mode_to_set));
-
- bool rx_agc_enabled = false;
- webrtc::AgcModes agc_mode = webrtc::kAgcDefault;
-
- EXPECT_EQ(0, voe_apm_->GetRxAgcStatus(channel_, rx_agc_enabled, agc_mode));
- EXPECT_TRUE(rx_agc_enabled);
- EXPECT_EQ(agc_mode_to_set, agc_mode);
- }
-
- // EC modes can map to other EC modes, so we have a separate parameter
- // for what we expect the EC mode to be set to.
- void TryEnablingEcWithMode(webrtc::EcModes ec_mode_to_set,
- webrtc::EcModes expected_mode) {
- EXPECT_EQ(0, voe_apm_->SetEcStatus(true, ec_mode_to_set));
-
- bool ec_enabled = true;
- webrtc::EcModes ec_mode = webrtc::kEcDefault;
-
- EXPECT_EQ(0, voe_apm_->GetEcStatus(ec_enabled, ec_mode));
-
- EXPECT_EQ(expected_mode, ec_mode);
- }
-
- // Here, the CNG mode will be expected to be on or off depending on the mode.
- void TryEnablingAecmWithMode(webrtc::AecmModes aecm_mode_to_set,
- bool cng_enabled_to_set) {
- EXPECT_EQ(0, voe_apm_->SetAecmMode(aecm_mode_to_set, cng_enabled_to_set));
-
- bool cng_enabled = false;
- webrtc::AecmModes aecm_mode = webrtc::kAecmEarpiece;
-
- voe_apm_->GetAecmMode(aecm_mode, cng_enabled);
-
- EXPECT_EQ(cng_enabled_to_set, cng_enabled);
- EXPECT_EQ(aecm_mode_to_set, aecm_mode);
- }
-
- void TryEnablingNsWithMode(webrtc::NsModes ns_mode_to_set,
- webrtc::NsModes expected_ns_mode) {
- EXPECT_EQ(0, voe_apm_->SetNsStatus(true, ns_mode_to_set));
-
- bool ns_status = true;
- webrtc::NsModes ns_mode = webrtc::kNsDefault;
- EXPECT_EQ(0, voe_apm_->GetNsStatus(ns_status, ns_mode));
-
- EXPECT_TRUE(ns_status);
- EXPECT_EQ(expected_ns_mode, ns_mode);
- }
-
- void TryEnablingRxNsWithMode(webrtc::NsModes ns_mode_to_set,
- webrtc::NsModes expected_ns_mode) {
- EXPECT_EQ(0, voe_apm_->SetRxNsStatus(channel_, true, ns_mode_to_set));
-
- bool ns_status = true;
- webrtc::NsModes ns_mode = webrtc::kNsDefault;
- EXPECT_EQ(0, voe_apm_->GetRxNsStatus(channel_, ns_status, ns_mode));
-
- EXPECT_TRUE(ns_status);
- EXPECT_EQ(expected_ns_mode, ns_mode);
- }
-
- void TryDetectingSilence() {
- // Here, speech is running. Shut down speech.
- EXPECT_EQ(0, voe_codec_->SetVADStatus(channel_, true));
- EXPECT_EQ(0, voe_volume_control_->SetInputMute(channel_, true));
- EXPECT_EQ(0, voe_file_->StopPlayingFileAsMicrophone(channel_));
-
- // We should detect the silence after a short time.
- Sleep(50);
- for (int i = 0; i < 25; i++) {
- EXPECT_EQ(0, voe_apm_->VoiceActivityIndicator(channel_));
- Sleep(10);
- }
- }
-
- void TryDetectingSpeechAfterSilence() {
- // Re-enable speech.
- RestartFakeMicrophone();
- EXPECT_EQ(0, voe_codec_->SetVADStatus(channel_, false));
- EXPECT_EQ(0, voe_volume_control_->SetInputMute(channel_, false));
-
- // We should detect the speech after a short time.
- for (int i = 0; i < 50; i++) {
- if (voe_apm_->VoiceActivityIndicator(channel_) == 1) {
- return;
- }
- Sleep(10);
- }
-
- ADD_FAILURE() << "Failed to detect speech within 500 ms.";
- }
-};
-
-#if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
-
-TEST_F(AudioProcessingTest, AgcIsOnByDefault) {
- bool agc_enabled = false;
- webrtc::AgcModes agc_mode = webrtc::kAgcAdaptiveAnalog;
-
- EXPECT_EQ(0, voe_apm_->GetAgcStatus(agc_enabled, agc_mode));
- EXPECT_TRUE(agc_enabled);
- EXPECT_EQ(webrtc::kAgcAdaptiveAnalog, agc_mode);
-}
-
-TEST_F(AudioProcessingTest, CanEnableAgcWithAllModes) {
- TryEnablingAgcWithMode(webrtc::kAgcAdaptiveDigital);
- TryEnablingAgcWithMode(webrtc::kAgcAdaptiveAnalog);
- TryEnablingAgcWithMode(webrtc::kAgcFixedDigital);
-}
-
-TEST_F(AudioProcessingTest, EcIsDisabledAndAecIsDefaultEcMode) {
- bool ec_enabled = true;
- webrtc::EcModes ec_mode = webrtc::kEcDefault;
-
- EXPECT_EQ(0, voe_apm_->GetEcStatus(ec_enabled, ec_mode));
- EXPECT_FALSE(ec_enabled);
- EXPECT_EQ(webrtc::kEcAec, ec_mode);
-}
-
-TEST_F(AudioProcessingTest, EnablingEcAecShouldEnableEcAec) {
- TryEnablingEcWithMode(webrtc::kEcAec, webrtc::kEcAec);
-}
-
-TEST_F(AudioProcessingTest, EnablingEcConferenceShouldEnableEcAec) {
- TryEnablingEcWithMode(webrtc::kEcConference, webrtc::kEcAec);
-}
-
-TEST_F(AudioProcessingTest, EcMetricsAreOffByDefault) {
- bool enabled = true;
- EXPECT_EQ(0, voe_apm_->GetEcMetricsStatus(enabled));
- EXPECT_FALSE(enabled);
-}
-
-TEST_F(AudioProcessingTest, ManualTestEcMetrics) {
- SwitchToManualMicrophone();
-
- EXPECT_EQ(0, voe_apm_->SetEcMetricsStatus(true));
-
- // Must enable AEC to get valid echo metrics.
- EXPECT_EQ(0, voe_apm_->SetEcStatus(true, webrtc::kEcAec));
-
- TEST_LOG("Speak into microphone and check metrics for 10 seconds...\n");
- int erl, erle, rerl, a_nlp;
- int delay_median = 0;
- int delay_std = 0;
-
- for (int i = 0; i < 5; i++) {
- Sleep(2000);
- EXPECT_EQ(0, voe_apm_->GetEchoMetrics(erl, erle, rerl, a_nlp));
- EXPECT_EQ(0, voe_apm_->GetEcDelayMetrics(delay_median, delay_std));
- TEST_LOG(" Echo : ERL=%5d, ERLE=%5d, RERL=%5d, A_NLP=%5d [dB], "
- " delay median=%3d, delay std=%3d [ms]\n", erl, erle, rerl, a_nlp,
- delay_median, delay_std);
- }
-
- EXPECT_EQ(0, voe_apm_->SetEcMetricsStatus(false));
-}
-
-// TODO(phoglund): Reenable below test when it's no longer flaky.
-TEST_F(AudioProcessingTest, DISABLED_TestVoiceActivityDetectionWithObserver) {
- voetest::RxCallback rx_callback;
- EXPECT_EQ(0, voe_apm_->RegisterRxVadObserver(channel_, rx_callback));
-
- // The extra sleeps are to allow decisions some time to propagate to the
- // observer.
- TryDetectingSilence();
- Sleep(100);
-
- EXPECT_EQ(0, rx_callback._vadDecision);
-
- TryDetectingSpeechAfterSilence();
- Sleep(100);
-
- EXPECT_EQ(1, rx_callback._vadDecision);
-
- EXPECT_EQ(0, voe_apm_->DeRegisterRxVadObserver(channel_));
-}
-
-#endif // !MAC_IPHONE && !WEBRTC_ANDROID
-
-TEST_F(AudioProcessingTest, EnablingEcAecmShouldEnableEcAecm) {
- // This one apparently applies to Android and iPhone as well.
- TryEnablingEcWithMode(webrtc::kEcAecm, webrtc::kEcAecm);
-}
-
-TEST_F(AudioProcessingTest, EcAecmModeIsEnabledAndSpeakerphoneByDefault) {
- bool cng_enabled = false;
- webrtc::AecmModes aecm_mode = webrtc::kAecmEarpiece;
-
- voe_apm_->GetAecmMode(aecm_mode, cng_enabled);
-
- EXPECT_TRUE(cng_enabled);
- EXPECT_EQ(webrtc::kAecmSpeakerphone, aecm_mode);
-}
-
-TEST_F(AudioProcessingTest, CanSetAecmMode) {
- EXPECT_EQ(0, voe_apm_->SetEcStatus(true, webrtc::kEcAecm));
-
- // Try some AECM mode - CNG enabled combinations.
- TryEnablingAecmWithMode(webrtc::kAecmEarpiece, true);
- TryEnablingAecmWithMode(webrtc::kAecmEarpiece, false);
- TryEnablingAecmWithMode(webrtc::kAecmLoudEarpiece, true);
- TryEnablingAecmWithMode(webrtc::kAecmLoudSpeakerphone, false);
- TryEnablingAecmWithMode(webrtc::kAecmQuietEarpieceOrHeadset, true);
- TryEnablingAecmWithMode(webrtc::kAecmSpeakerphone, false);
-}
-
-TEST_F(AudioProcessingTest, RxAgcShouldBeOffByDefault) {
- bool rx_agc_enabled = true;
- webrtc::AgcModes agc_mode = webrtc::kAgcDefault;
-
- EXPECT_EQ(0, voe_apm_->GetRxAgcStatus(channel_, rx_agc_enabled, agc_mode));
- EXPECT_FALSE(rx_agc_enabled);
- EXPECT_EQ(webrtc::kAgcAdaptiveDigital, agc_mode);
-}
-
-TEST_F(AudioProcessingTest, CanTurnOnDigitalRxAcg) {
- TryEnablingRxAgcWithMode(webrtc::kAgcAdaptiveDigital);
- TryEnablingRxAgcWithMode(webrtc::kAgcFixedDigital);
-}
-
-TEST_F(AudioProcessingTest, CannotTurnOnAdaptiveAnalogRxAgc) {
- EXPECT_EQ(-1, voe_apm_->SetRxAgcStatus(
- channel_, true, webrtc::kAgcAdaptiveAnalog));
-}
-
-TEST_F(AudioProcessingTest, NsIsOffWithModerateSuppressionByDefault) {
- bool ns_status = true;
- webrtc::NsModes ns_mode = webrtc::kNsDefault;
- EXPECT_EQ(0, voe_apm_->GetNsStatus(ns_status, ns_mode));
-
- EXPECT_FALSE(ns_status);
- EXPECT_EQ(webrtc::kNsModerateSuppression, ns_mode);
-}
-
-TEST_F(AudioProcessingTest, CanSetNsMode) {
- // Concrete suppression values map to themselves.
- TryEnablingNsWithMode(webrtc::kNsHighSuppression,
- webrtc::kNsHighSuppression);
- TryEnablingNsWithMode(webrtc::kNsLowSuppression,
- webrtc::kNsLowSuppression);
- TryEnablingNsWithMode(webrtc::kNsModerateSuppression,
- webrtc::kNsModerateSuppression);
- TryEnablingNsWithMode(webrtc::kNsVeryHighSuppression,
- webrtc::kNsVeryHighSuppression);
-
- // Conference and Default map to concrete values.
- TryEnablingNsWithMode(webrtc::kNsConference,
- webrtc::kNsHighSuppression);
- TryEnablingNsWithMode(webrtc::kNsDefault,
- webrtc::kNsModerateSuppression);
-}
-
-TEST_F(AudioProcessingTest, RxNsIsOffWithModerateSuppressionByDefault) {
- bool ns_status = true;
- webrtc::NsModes ns_mode = webrtc::kNsDefault;
- EXPECT_EQ(0, voe_apm_->GetRxNsStatus(channel_, ns_status, ns_mode));
-
- EXPECT_FALSE(ns_status);
- EXPECT_EQ(webrtc::kNsModerateSuppression, ns_mode);
-}
-
-TEST_F(AudioProcessingTest, CanSetRxNsMode) {
- EXPECT_EQ(0, voe_apm_->SetRxNsStatus(channel_, true));
-
- // See comments on the regular NS test above.
- TryEnablingRxNsWithMode(webrtc::kNsHighSuppression,
- webrtc::kNsHighSuppression);
- TryEnablingRxNsWithMode(webrtc::kNsLowSuppression,
- webrtc::kNsLowSuppression);
- TryEnablingRxNsWithMode(webrtc::kNsModerateSuppression,
- webrtc::kNsModerateSuppression);
- TryEnablingRxNsWithMode(webrtc::kNsVeryHighSuppression,
- webrtc::kNsVeryHighSuppression);
- TryEnablingRxNsWithMode(webrtc::kNsConference,
- webrtc::kNsHighSuppression);
- TryEnablingRxNsWithMode(webrtc::kNsDefault,
- webrtc::kNsModerateSuppression);
-}
-
-TEST_F(AudioProcessingTest, VadIsDisabledByDefault) {
- bool vad_enabled;
- bool disabled_dtx;
- webrtc::VadModes vad_mode;
-
- EXPECT_EQ(0, voe_codec_->GetVADStatus(
- channel_, vad_enabled, vad_mode, disabled_dtx));
-
- EXPECT_FALSE(vad_enabled);
-}
-
-TEST_F(AudioProcessingTest, VoiceActivityIndicatorReturns1WithSpeechOn) {
- // This sleep is necessary since the voice detection algorithm needs some
- // time to detect the speech from the fake microphone.
- Sleep(500);
- EXPECT_EQ(1, voe_apm_->VoiceActivityIndicator(channel_));
-}
-
-#if defined(MAC_IPHONE) || defined(WEBRTC_ANDROID)
-
-TEST_F(AudioProcessingTest, AgcIsOffByDefaultAndDigital) {
- bool agc_enabled = true;
- webrtc::AgcModes agc_mode = webrtc::kAgcAdaptiveAnalog;
-
- EXPECT_EQ(0, voe_apm_->GetAgcStatus(agc_enabled, agc_mode));
- EXPECT_FALSE(agc_enabled);
- EXPECT_EQ(webrtc::kAgcAdaptiveDigital, agc_mode);
-}
-
-TEST_F(AudioProcessingTest, CanEnableAgcInAdaptiveDigitalMode) {
- TryEnablingAgcWithMode(webrtc::kAgcAdaptiveDigital);
-}
-
-TEST_F(AudioProcessingTest, AgcIsPossibleExceptInAdaptiveAnalogMode) {
- EXPECT_EQ(-1, voe_apm_->SetAgcStatus(true, webrtc::kAgcAdaptiveAnalog));
- EXPECT_EQ(0, voe_apm_->SetAgcStatus(true, webrtc::kAgcFixedDigital));
- EXPECT_EQ(0, voe_apm_->SetAgcStatus(true, webrtc::kAgcAdaptiveDigital));
-}
-
-TEST_F(AudioProcessingTest, EcIsDisabledAndAecmIsDefaultEcMode) {
- bool ec_enabled = true;
- webrtc::EcModes ec_mode = webrtc::kEcDefault;
-
- EXPECT_EQ(0, voe_apm_->GetEcStatus(ec_enabled, ec_mode));
- EXPECT_FALSE(ec_enabled);
- EXPECT_EQ(webrtc::kEcAecm, ec_mode);
-}
-
-TEST_F(AudioProcessingTest, TestVoiceActivityDetection) {
- TryDetectingSilence();
- TryDetectingSpeechAfterSilence();
-}
-
-#endif // MAC_IPHONE || WEBRTC_ANDROID
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/call_report_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/call_report_test.cc
deleted file mode 100644
index c96b14d..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/standard/call_report_test.cc
+++ /dev/null
@@ -1,88 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "after_streaming_fixture.h"
-#include "testsupport/fileutils.h"
-
-class CallReportTest : public AfterStreamingFixture {
-};
-
-TEST_F(CallReportTest, ResetCallReportStatisticsFailsForBadInput) {
- EXPECT_EQ(-1, voe_call_report_->ResetCallReportStatistics(-2));
- EXPECT_EQ(-1, voe_call_report_->ResetCallReportStatistics(1));
-}
-
-TEST_F(CallReportTest, ResetCallReportStatisticsSucceedsWithCorrectInput) {
- EXPECT_EQ(0, voe_call_report_->ResetCallReportStatistics(channel_));
- EXPECT_EQ(0, voe_call_report_->ResetCallReportStatistics(-1));
-}
-
-TEST_F(CallReportTest, EchoMetricSummarySucceeds) {
- EXPECT_EQ(0, voe_apm_->SetEcMetricsStatus(true));
- Sleep(1000);
-
- webrtc::EchoStatistics echo_statistics;
- EXPECT_EQ(0, voe_call_report_->GetEchoMetricSummary(echo_statistics));
-}
-
-TEST_F(CallReportTest, GetRoundTripTimeSummaryReturnsAllMinusOnesIfRtcpIsOff) {
- voe_rtp_rtcp_->SetRTCPStatus(channel_, false);
-
- webrtc::StatVal delays;
- EXPECT_EQ(0, voe_call_report_->GetRoundTripTimeSummary(channel_, delays));
- EXPECT_EQ(-1, delays.average);
- EXPECT_EQ(-1, delays.min);
- EXPECT_EQ(-1, delays.max);
-}
-
-TEST_F(CallReportTest, GetRoundTripTimesReturnsValuesIfRtcpIsOn) {
- voe_rtp_rtcp_->SetRTCPStatus(channel_, true);
- Sleep(1000);
-
- webrtc::StatVal delays;
- EXPECT_EQ(0, voe_call_report_->GetRoundTripTimeSummary(channel_, delays));
- EXPECT_NE(-1, delays.average);
- EXPECT_NE(-1, delays.min);
- EXPECT_NE(-1, delays.max);
-}
-
-TEST_F(CallReportTest, DeadOrAliveSummaryFailsIfDeadOrAliveTrackingNotActive) {
- int count_the_dead;
- int count_the_living;
- EXPECT_EQ(-1, voe_call_report_->GetDeadOrAliveSummary(channel_,
- count_the_dead,
- count_the_living));
-}
-
-TEST_F(CallReportTest,
- DeadOrAliveSummarySucceedsIfDeadOrAliveTrackingIsActive) {
- EXPECT_EQ(0, voe_network_->SetPeriodicDeadOrAliveStatus(channel_, true, 1));
- Sleep(1200);
-
- int count_the_dead;
- int count_the_living;
- EXPECT_EQ(0, voe_call_report_->GetDeadOrAliveSummary(channel_,
- count_the_dead,
- count_the_living));
-
- EXPECT_GE(count_the_dead, 0);
- EXPECT_GE(count_the_living, 0);
-}
-
-TEST_F(CallReportTest, WriteReportToFileFailsOnBadInput) {
- EXPECT_EQ(-1, voe_call_report_->WriteReportToFile(NULL));
-}
-
-TEST_F(CallReportTest, WriteReportToFileSucceedsWithCorrectFilename) {
- std::string output_path = webrtc::test::OutputPath();
- std::string report_filename = output_path + "call_report.txt";
-
- EXPECT_EQ(0, voe_call_report_->WriteReportToFile(report_filename.c_str()));
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/codec_before_streaming_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/codec_before_streaming_test.cc
deleted file mode 100644
index 6d902ef..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/standard/codec_before_streaming_test.cc
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "after_initialization_fixture.h"
-
-class CodecBeforeStreamingTest : public AfterInitializationFixture {
- protected:
- void SetUp() {
- memset(&codec_instance_, 0, sizeof(codec_instance_));
- codec_instance_.channels = 1;
- codec_instance_.plfreq = 16000;
- codec_instance_.pacsize = 480;
-
- channel_ = voe_base_->CreateChannel();
- }
-
- void TearDown() {
- voe_base_->DeleteChannel(channel_);
- }
-
- webrtc::CodecInst codec_instance_;
- int channel_;
-};
-
-// TODO(phoglund): add test which verifies default pltypes for various codecs.
-
-TEST_F(CodecBeforeStreamingTest, GetRecPayloadTypeFailsForInvalidCodecName) {
- strcpy(codec_instance_.plname, "SomeInvalidCodecName");
-
- // Should fail since the codec name is invalid.
- EXPECT_NE(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
-}
-
-TEST_F(CodecBeforeStreamingTest, GetRecPayloadTypeRecognizesISAC) {
- strcpy(codec_instance_.plname, "iSAC");
- EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
- strcpy(codec_instance_.plname, "ISAC");
- EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
-}
-
-TEST_F(CodecBeforeStreamingTest, SetRecPayloadTypeCanChangeISACPayloadType) {
- strcpy(codec_instance_.plname, "ISAC");
-
- codec_instance_.pltype = 123;
- EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance_));
- EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
- EXPECT_EQ(123, codec_instance_.pltype);
-
- codec_instance_.pltype = 104;
- EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance_));
- EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
-
- EXPECT_EQ(104, codec_instance_.pltype);
-}
-
-TEST_F(CodecBeforeStreamingTest, SetRecPayloadTypeCanChangeILBCPayloadType) {
- strcpy(codec_instance_.plname, "iLBC");
- codec_instance_.plfreq = 8000;
- codec_instance_.pacsize = 240;
- codec_instance_.rate = 13300;
-
- EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
- int original_pltype = codec_instance_.pltype;
- codec_instance_.pltype = 123;
- EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance_));
- EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
-
- EXPECT_EQ(123, codec_instance_.pltype);
-
- codec_instance_.pltype = original_pltype;
- EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance_));
- EXPECT_EQ(0, voe_codec_->GetRecPayloadType(channel_, codec_instance_));
-
- EXPECT_EQ(original_pltype, codec_instance_.pltype);
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/codec_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/codec_test.cc
deleted file mode 100644
index dc5558b..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/standard/codec_test.cc
+++ /dev/null
@@ -1,231 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "after_streaming_fixture.h"
-#include "voe_test_defines.h"
-#include "voice_engine_defines.h"
-
-class CodecTest : public AfterStreamingFixture {
- protected:
- void SetUp() {
- memset(&codec_instance_, 0, sizeof(codec_instance_));
- }
-
- void SetArbitrarySendCodec() {
- // Just grab the first codec.
- EXPECT_EQ(0, voe_codec_->GetCodec(0, codec_instance_));
- EXPECT_EQ(0, voe_codec_->SetSendCodec(channel_, codec_instance_));
- }
-
- webrtc::CodecInst codec_instance_;
-};
-
-static void SetRateIfILBC(webrtc::CodecInst* codec_instance, int packet_size) {
- if (!_stricmp(codec_instance->plname, "ilbc")) {
- if (packet_size == 160 || packet_size == 320) {
- codec_instance->rate = 15200;
- } else {
- codec_instance->rate = 13300;
- }
- }
-}
-
-static bool IsNotViableSendCodec(const char* codec_name) {
- return !_stricmp(codec_name, "CN") ||
- !_stricmp(codec_name, "telephone-event") ||
- !_stricmp(codec_name, "red");
-}
-
-TEST_F(CodecTest, PcmuIsDefaultCodecAndHasTheRightValues) {
- EXPECT_EQ(0, voe_codec_->GetSendCodec(channel_, codec_instance_));
- EXPECT_EQ(1, codec_instance_.channels);
- EXPECT_EQ(160, codec_instance_.pacsize);
- EXPECT_EQ(8000, codec_instance_.plfreq);
- EXPECT_EQ(0, codec_instance_.pltype);
- EXPECT_EQ(64000, codec_instance_.rate);
- EXPECT_STRCASEEQ("PCMU", codec_instance_.plname);
-}
-
-TEST_F(CodecTest, VoiceActivityDetectionIsOffByDefault) {
- bool vad_enabled = false;
- bool dtx_disabled = false;
- webrtc::VadModes vad_mode = webrtc::kVadAggressiveMid;
-
- voe_codec_->GetVADStatus(channel_, vad_enabled, vad_mode, dtx_disabled);
-
- EXPECT_FALSE(vad_enabled);
- EXPECT_TRUE(dtx_disabled);
- EXPECT_EQ(webrtc::kVadConventional, vad_mode);
-}
-
-TEST_F(CodecTest, VoiceActivityDetectionCanBeEnabled) {
- EXPECT_EQ(0, voe_codec_->SetVADStatus(channel_, true));
-
- bool vad_enabled = false;
- bool dtx_disabled = false;
- webrtc::VadModes vad_mode = webrtc::kVadAggressiveMid;
-
- voe_codec_->GetVADStatus(channel_, vad_enabled, vad_mode, dtx_disabled);
-
- EXPECT_TRUE(vad_enabled);
- EXPECT_EQ(webrtc::kVadConventional, vad_mode);
- EXPECT_FALSE(dtx_disabled);
-}
-
-TEST_F(CodecTest, VoiceActivityDetectionTypeSettingsCanBeChanged) {
- bool vad_enabled = false;
- bool dtx_disabled = false;
- webrtc::VadModes vad_mode = webrtc::kVadAggressiveMid;
-
- EXPECT_EQ(0, voe_codec_->SetVADStatus(
- channel_, true, webrtc::kVadAggressiveLow, false));
- EXPECT_EQ(0, voe_codec_->GetVADStatus(
- channel_, vad_enabled, vad_mode, dtx_disabled));
- EXPECT_EQ(vad_mode, webrtc::kVadAggressiveLow);
- EXPECT_FALSE(dtx_disabled);
-
- EXPECT_EQ(0, voe_codec_->SetVADStatus(
- channel_, true, webrtc::kVadAggressiveMid, false));
- EXPECT_EQ(0, voe_codec_->GetVADStatus(
- channel_, vad_enabled, vad_mode, dtx_disabled));
- EXPECT_EQ(vad_mode, webrtc::kVadAggressiveMid);
- EXPECT_FALSE(dtx_disabled);
-
- // The fourth argument is the DTX disable flag.
- EXPECT_EQ(0, voe_codec_->SetVADStatus(
- channel_, true, webrtc::kVadAggressiveHigh, true));
- EXPECT_EQ(0, voe_codec_->GetVADStatus(
- channel_, vad_enabled, vad_mode, dtx_disabled));
- EXPECT_EQ(vad_mode, webrtc::kVadAggressiveHigh);
- EXPECT_TRUE(dtx_disabled);
-
- EXPECT_EQ(0, voe_codec_->SetVADStatus(
- channel_, true, webrtc::kVadConventional, true));
- EXPECT_EQ(0, voe_codec_->GetVADStatus(
- channel_, vad_enabled, vad_mode, dtx_disabled));
- EXPECT_EQ(vad_mode, webrtc::kVadConventional);
-}
-
-TEST_F(CodecTest, VoiceActivityDetectionCanBeTurnedOff) {
- EXPECT_EQ(0, voe_codec_->SetVADStatus(channel_, true));
-
- // VAD is always on when DTX is on, so we need to turn off DTX too.
- EXPECT_EQ(0, voe_codec_->SetVADStatus(
- channel_, false, webrtc::kVadConventional, true));
-
- bool vad_enabled = false;
- bool dtx_disabled = false;
- webrtc::VadModes vad_mode = webrtc::kVadAggressiveMid;
-
- voe_codec_->GetVADStatus(channel_, vad_enabled, vad_mode, dtx_disabled);
-
- EXPECT_FALSE(vad_enabled);
- EXPECT_TRUE(dtx_disabled);
- EXPECT_EQ(webrtc::kVadConventional, vad_mode);
-}
-
-// Tests requiring manual verification (although they do have some value
-// without the manual verification):
-TEST_F(CodecTest, ManualExtendedISACApisBehaveAsExpected) {
- strcpy(codec_instance_.plname, "isac");
- codec_instance_.pltype = 103;
- codec_instance_.plfreq = 16000;
- codec_instance_.channels = 1;
- // -1 here means "adaptive rate".
- codec_instance_.rate = -1;
- codec_instance_.pacsize = 480;
-
- EXPECT_EQ(0, voe_codec_->SetSendCodec(channel_, codec_instance_));
-
- EXPECT_NE(0, voe_codec_->SetISACInitTargetRate(channel_, 5000)) <<
- "iSAC should reject rate 5000.";
- EXPECT_NE(0, voe_codec_->SetISACInitTargetRate(channel_, 33000)) <<
- "iSAC should reject rate 33000.";
- EXPECT_EQ(0, voe_codec_->SetISACInitTargetRate(channel_, 32000));
-
- TEST_LOG("Ensure that the sound is good (iSAC, target = 32kbps)...\n");
- Sleep(3000);
-
- EXPECT_EQ(0, voe_codec_->SetISACInitTargetRate(channel_, 10000));
- TEST_LOG("Ensure that the sound is good (iSAC, target = 10kbps)...\n");
- Sleep(3000);
-
- EXPECT_EQ(0, voe_codec_->SetISACInitTargetRate(channel_, 10000, true));
- EXPECT_EQ(0, voe_codec_->SetISACInitTargetRate(channel_, 10000, false));
- EXPECT_EQ(0, voe_codec_->SetISACInitTargetRate(channel_, 0));
- TEST_LOG("Ensure that the sound is good (iSAC, target = default)...\n");
- Sleep(3000);
-
- TEST_LOG(" Testing SetISACMaxPayloadSize:\n");
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_NE(0, voe_codec_->SetISACMaxPayloadSize(channel_, 50));
- EXPECT_NE(0, voe_codec_->SetISACMaxPayloadSize(channel_, 650));
- EXPECT_EQ(0, voe_codec_->SetISACMaxPayloadSize(channel_, 120));
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
- TEST_LOG("Ensure that the sound is good (iSAC, "
- "max payload size = 100 bytes)...\n");
- Sleep(3000);
-
- TEST_LOG(" Testing SetISACMaxRate:\n");
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_EQ(0, voe_codec_->SetISACMaxPayloadSize(channel_, 400));
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
-
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_NE(0, voe_codec_->SetISACMaxRate(channel_, 31900));
- EXPECT_NE(0, voe_codec_->SetISACMaxRate(channel_, 53500));
- EXPECT_EQ(0, voe_codec_->SetISACMaxRate(channel_, 32000));
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
- TEST_LOG("Ensure that the sound is good (iSAC, max rate = 32 kbps)...\n");
- Sleep(3000);
-
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
-
- // Restore "no limitation". No, no limit, we reach for the sky.
- EXPECT_EQ(0, voe_codec_->SetISACMaxRate(channel_, 53400));
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
-}
-
-TEST_F(CodecTest, ManualVerifySendCodecsForAllPacketSizes) {
- for (int i = 0; i < voe_codec_->NumOfCodecs(); ++i) {
- voe_codec_->GetCodec(i, codec_instance_);
- if (IsNotViableSendCodec(codec_instance_.plname)) {
- TEST_LOG("Skipping %s.\n", codec_instance_.plname);
- continue;
- }
- EXPECT_NE(-1, codec_instance_.pltype) <<
- "The codec database should suggest a payload type.";
-
- // Test with default packet size:
- TEST_LOG("%s (pt=%d): default packet size(%d), accepts sizes ",
- codec_instance_.plname, codec_instance_.pltype,
- codec_instance_.pacsize);
- voe_codec_->SetSendCodec(channel_, codec_instance_);
- Sleep(CODEC_TEST_TIME);
-
- // Now test other reasonable packet sizes:
- bool at_least_one_succeeded = false;
- for (int packet_size = 80; packet_size < 1000; packet_size += 80) {
- SetRateIfILBC(&codec_instance_, packet_size);
- codec_instance_.pacsize = packet_size;
-
- if (voe_codec_->SetSendCodec(channel_, codec_instance_) != -1) {
- // Note that it's fine for SetSendCodec to fail - what packet sizes
- // it accepts depends on the codec. It should accept one at minimum.
- TEST_LOG("%d ", packet_size);
- TEST_LOG_FLUSH;
- at_least_one_succeeded = true;
- Sleep(CODEC_TEST_TIME);
- }
- }
- TEST_LOG("\n");
- EXPECT_TRUE(at_least_one_succeeded);
- }
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/dtmf_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/dtmf_test.cc
deleted file mode 100644
index 6aa4625..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/standard/dtmf_test.cc
+++ /dev/null
@@ -1,101 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "after_streaming_fixture.h"
-#include "voice_engine_defines.h"
-
-class DtmfTest : public AfterStreamingFixture {
- protected:
- void RunSixteenDtmfEvents(bool out_of_band) {
- TEST_LOG("Sending telephone events:\n");
- EXPECT_EQ(0, voe_dtmf_->SetDtmfFeedbackStatus(false));
-
- for (int i = 0; i < 16; i++) {
- TEST_LOG("%d ", i);
- TEST_LOG_FLUSH;
- EXPECT_EQ(0, voe_dtmf_->SendTelephoneEvent(
- channel_, i, out_of_band, 160, 10));
- Sleep(500);
- }
- TEST_LOG("\n");
- }
-};
-
-TEST_F(DtmfTest, DtmfFeedbackIsEnabledByDefaultButNotDirectFeedback) {
- bool dtmf_feedback = false;
- bool dtmf_direct_feedback = false;
-
- EXPECT_EQ(0, voe_dtmf_->GetDtmfFeedbackStatus(dtmf_feedback,
- dtmf_direct_feedback));
-
- EXPECT_TRUE(dtmf_feedback);
- EXPECT_FALSE(dtmf_direct_feedback);
-}
-
-TEST_F(DtmfTest, ManualSuccessfullySendsInBandTelephoneEvents) {
- RunSixteenDtmfEvents(false);
-}
-
-TEST_F(DtmfTest, ManualSuccessfullySendsOutOfBandTelephoneEvents) {
- RunSixteenDtmfEvents(true);
-}
-
-TEST_F(DtmfTest, TestTwoNonDtmfEvents) {
- EXPECT_EQ(0, voe_dtmf_->SendTelephoneEvent(channel_, 32, true));
- EXPECT_EQ(0, voe_dtmf_->SendTelephoneEvent(channel_, 110, true));
-}
-
-#ifndef MAC_IPHONE
-TEST_F(DtmfTest, ManualCanDisableDtmfPlayoutExceptOnIphone) {
- TEST_LOG("Disabling DTMF playout (no tone should be heard) \n");
- EXPECT_EQ(0, voe_dtmf_->SetDtmfPlayoutStatus(channel_, false));
- EXPECT_EQ(0, voe_dtmf_->SendTelephoneEvent(channel_, 0, true));
- Sleep(500);
-
- TEST_LOG("Enabling DTMF playout (tone should be heard) \n");
- EXPECT_EQ(0, voe_dtmf_->SetDtmfPlayoutStatus(channel_, true));
- EXPECT_EQ(0, voe_dtmf_->SendTelephoneEvent(channel_, 0, true));
- Sleep(500);
-}
-#endif
-
-// This test modifies the DTMF payload type from the default 106 to 88
-// and then runs through 16 DTMF out.of-band events.
-TEST_F(DtmfTest, ManualCanChangeDtmfPayloadType) {
- webrtc::CodecInst codec_instance;
-
- TEST_LOG("Changing DTMF payload type.\n");
-
- // Start by modifying the receiving side.
- for (int i = 0; i < voe_codec_->NumOfCodecs(); i++) {
- EXPECT_EQ(0, voe_codec_->GetCodec(i, codec_instance));
- if (!_stricmp("telephone-event", codec_instance.plname)) {
- codec_instance.pltype = 88; // Use 88 instead of default 106.
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
- EXPECT_EQ(0, voe_base_->StopReceive(channel_));
- EXPECT_EQ(0, voe_codec_->SetRecPayloadType(channel_, codec_instance));
- EXPECT_EQ(0, voe_base_->StartReceive(channel_));
- EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
- break;
- }
- }
-
- Sleep(500);
-
- // Next, we must modify the sending side as well.
- EXPECT_EQ(0, voe_dtmf_->SetSendTelephoneEventPayloadType(
- channel_, codec_instance.pltype));
-
- RunSixteenDtmfEvents(true);
-
- EXPECT_EQ(0, voe_dtmf_->SetDtmfFeedbackStatus(true, false));
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/file_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/file_test.cc
deleted file mode 100644
index 63b1600..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/standard/file_test.cc
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "after_streaming_fixture.h"
-#include "testsupport/fileutils.h"
-
-class FileTest : public AfterStreamingFixture {
- protected:
- // Creates the string åäö.pcm.
- std::string CreateTrickyFilenameInUtf8() {
- char filename[16] = { (char)0xc3, (char)0xa5,
- (char)0xc3, (char)0xa4,
- (char)0xc3, (char)0xb6,
- static_cast<char>(0) };
- return std::string(filename) + ".pcm";
- }
-};
-
-TEST_F(FileTest, ManualRecordToFileForThreeSecondsAndPlayback) {
- SwitchToManualMicrophone();
-
- std::string recording_filename =
- webrtc::test::OutputPath() + CreateTrickyFilenameInUtf8();
-
- TEST_LOG("Recording to %s for 3 seconds.\n", recording_filename.c_str());
- EXPECT_EQ(0, voe_file_->StartRecordingMicrophone(recording_filename.c_str()));
- Sleep(3000);
- EXPECT_EQ(0, voe_file_->StopRecordingMicrophone());
-
- TEST_LOG("Playing back %s.\n", recording_filename.c_str());
- EXPECT_EQ(0, voe_file_->StartPlayingFileLocally(
- channel_, recording_filename.c_str()));
-
- // Play the file to the user and ensure the is-playing-locally
- // and scaling methods also work. The clip is 3 seconds long.
- Sleep(250);
- EXPECT_EQ(1, voe_file_->IsPlayingFileLocally(channel_));
- Sleep(1500);
- TEST_LOG("Decreasing level by 50%%.\n");
- EXPECT_EQ(0, voe_file_->ScaleLocalFilePlayout(channel_, 0.5f));
- Sleep(1500);
- EXPECT_EQ(0, voe_file_->IsPlayingFileLocally(channel_));
-}
-
-TEST_F(FileTest, ManualRecordPlayoutToWavFileForThreeSecondsAndPlayback) {
- webrtc::CodecInst send_codec;
- voe_codec_->GetSendCodec(channel_, send_codec);
-
- std::string recording_filename =
- webrtc::test::OutputPath() + "playout.wav";
-
- TEST_LOG("Recording playout to %s.\n", recording_filename.c_str());
- EXPECT_EQ(0, voe_file_->StartRecordingPlayout(
- channel_, recording_filename.c_str(), &send_codec));
- Sleep(3000);
- EXPECT_EQ(0, voe_file_->StopRecordingPlayout(channel_));
-
- TEST_LOG("Playing back the recording in looping mode.\n");
- EXPECT_EQ(0, voe_file_->StartPlayingFileAsMicrophone(
- channel_, recording_filename.c_str(), true, false,
- webrtc::kFileFormatWavFile));
-
- Sleep(2000);
- EXPECT_EQ(1, voe_file_->IsPlayingFileAsMicrophone(channel_));
- Sleep(2000);
- // We should still be playing since we're looping.
- EXPECT_EQ(1, voe_file_->IsPlayingFileAsMicrophone(channel_));
-
- // Try scaling as well.
- TEST_LOG("Decreasing level by 50%%.\n");
- EXPECT_EQ(0, voe_file_->ScaleFileAsMicrophonePlayout(channel_, 0.5f));
- Sleep(1000);
-
- EXPECT_EQ(0, voe_file_->StopPlayingFileAsMicrophone(channel_));
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/hardware_before_initializing_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/hardware_before_initializing_test.cc
deleted file mode 100644
index 540614e..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/standard/hardware_before_initializing_test.cc
+++ /dev/null
@@ -1,27 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "common_types.h"
-#include "before_initialization_fixture.h"
-
-using namespace webrtc;
-
-class HardwareBeforeInitializingTest : public BeforeInitializationFixture {
-};
-
-TEST_F(HardwareBeforeInitializingTest,
- SetAudioDeviceLayerAcceptsPlatformDefaultBeforeInitializing) {
- AudioLayers wanted_layer = kAudioPlatformDefault;
- AudioLayers given_layer;
- EXPECT_EQ(0, voe_hardware_->SetAudioDeviceLayer(wanted_layer));
- EXPECT_EQ(0, voe_hardware_->GetAudioDeviceLayer(given_layer));
- EXPECT_EQ(wanted_layer, given_layer) <<
- "These should be the same before initializing.";
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/hardware_before_streaming_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/hardware_before_streaming_test.cc
deleted file mode 100644
index 6e56347..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/standard/hardware_before_streaming_test.cc
+++ /dev/null
@@ -1,166 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cstring>
-
-#include "after_initialization_fixture.h"
-
-using namespace webrtc;
-
-static const char* kNoDevicesErrorMessage =
- "Either you have no recording / playout device "
- "on your system, or the method failed.";
-
-class HardwareBeforeStreamingTest : public AfterInitializationFixture {
-};
-
-// Tests that apply to both mobile and desktop:
-
-TEST_F(HardwareBeforeStreamingTest,
- SetAudioDeviceLayerFailsSinceTheVoiceEngineHasBeenInitialized) {
- EXPECT_NE(0, voe_hardware_->SetAudioDeviceLayer(kAudioPlatformDefault));
- EXPECT_EQ(VE_ALREADY_INITED, voe_base_->LastError());
-}
-
-TEST_F(HardwareBeforeStreamingTest,
- GetCPULoadSucceedsOnWindowsButNotOtherPlatforms) {
- int load_percent;
-#if defined(_WIN32)
- EXPECT_EQ(0, voe_hardware_->GetCPULoad(load_percent));
-#else
- EXPECT_NE(0, voe_hardware_->GetCPULoad(load_percent)) <<
- "Should fail on non-Windows platforms.";
-#endif
-}
-
-// Tests that only apply to mobile:
-
-#ifdef MAC_IPHONE
-TEST_F(HardwareBeforeStreamingTest, ResetsAudioDeviceOnIphone) {
- EXPECT_EQ(0, voe_hardware_->ResetAudioDevice());
-}
-#endif
-
-// Tests that only apply to desktop:
-#if !defined(MAC_IPHONE) & !defined(WEBRTC_ANDROID)
-
-TEST_F(HardwareBeforeStreamingTest, GetSystemCpuLoadSucceeds) {
-#ifdef _WIN32
- // This method needs some warm-up time on Windows. We sleep a good amount
- // of time instead of retrying to make the test simpler.
- Sleep(2000);
-#endif
-
- int load_percent;
- EXPECT_EQ(0, voe_hardware_->GetSystemCPULoad(load_percent));
-}
-
-TEST_F(HardwareBeforeStreamingTest, GetPlayoutDeviceStatusReturnsTrue) {
- bool play_available = false;
- EXPECT_EQ(0, voe_hardware_->GetPlayoutDeviceStatus(play_available));
- ASSERT_TRUE(play_available) <<
- "Ensures that the method works and that hardware is in the right state.";
-}
-
-TEST_F(HardwareBeforeStreamingTest, GetRecordingDeviceStatusReturnsTrue) {
- bool recording_available = false;
- EXPECT_EQ(0, voe_hardware_->GetRecordingDeviceStatus(recording_available));
- EXPECT_TRUE(recording_available) <<
- "Ensures that the method works and that hardware is in the right state.";
-}
-
- // Win, Mac and Linux sound device tests.
-TEST_F(HardwareBeforeStreamingTest,
- GetRecordingDeviceNameRetrievesDeviceNames) {
- char device_name[128] = {0};
- char guid_name[128] = {0};
-
-#ifdef _WIN32
- EXPECT_EQ(0, voe_hardware_->GetRecordingDeviceName(
- -1, device_name, guid_name));
- EXPECT_GT(strlen(device_name), 0u) << kNoDevicesErrorMessage;
- device_name[0] = '\0';
-
- EXPECT_EQ(0, voe_hardware_->GetPlayoutDeviceName(
- -1, device_name, guid_name));
- EXPECT_GT(strlen(device_name), 0u) << kNoDevicesErrorMessage;
-
-#else
- EXPECT_EQ(0, voe_hardware_->GetRecordingDeviceName(
- 0, device_name, guid_name));
- EXPECT_GT(strlen(device_name), 0u) << kNoDevicesErrorMessage;
- device_name[0] = '\0';
-
- EXPECT_EQ(0, voe_hardware_->GetPlayoutDeviceName(
- 0, device_name, guid_name));
- EXPECT_GT(strlen(device_name), 0u) << kNoDevicesErrorMessage;
-#endif // !WIN32
-}
-
-TEST_F(HardwareBeforeStreamingTest,
- AllEnumeratedRecordingDevicesCanBeSetAsRecordingDevice) {
- // Check recording side.
- // Extended Win32 enumeration tests: unique GUID outputs on Vista and up:
- // Win XP and below : device_name is copied to guid_name.
- // Win Vista and up : device_name is the friendly name and GUID is a unique
- // identifier.
- // Other : guid_name is left unchanged.
- int num_of_recording_devices = 0;
- EXPECT_EQ(0, voe_hardware_->GetNumOfRecordingDevices(
- num_of_recording_devices));
- EXPECT_GT(num_of_recording_devices, 0) << kNoDevicesErrorMessage;
-
- char device_name[128] = {0};
- char guid_name[128] = {0};
-
- for (int i = 0; i < num_of_recording_devices; i++) {
- EXPECT_EQ(0, voe_hardware_->GetRecordingDeviceName(
- i, device_name, guid_name));
- EXPECT_GT(strlen(device_name), 0u) <<
- "There should be no empty device names "
- "among the ones the system gives us.";
- EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(i));
- }
-}
-
-TEST_F(HardwareBeforeStreamingTest,
- AllEnumeratedPlayoutDevicesCanBeSetAsPlayoutDevice) {
- // Check playout side (see recording side test for more info on GUIDs).
- int num_of_playout_devices = 0;
- EXPECT_EQ(0, voe_hardware_->GetNumOfPlayoutDevices(
- num_of_playout_devices));
- EXPECT_GT(num_of_playout_devices, 0) << kNoDevicesErrorMessage;
-
- char device_name[128] = {0};
- char guid_name[128] = {0};
-
- for (int i = 0; i < num_of_playout_devices; ++i) {
- EXPECT_EQ(0, voe_hardware_->GetPlayoutDeviceName(
- i, device_name, guid_name));
- EXPECT_GT(strlen(device_name), 0u) <<
- "There should be no empty device names "
- "among the ones the system gives us.";
- EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(i));
- }
-}
-
-TEST_F(HardwareBeforeStreamingTest,
- SetDeviceWithMagicalArgumentsSetsDefaultSoundDevices) {
-#ifdef _WIN32
- // -1 means "default device" on Windows.
- EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(-1));
- EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(-1));
-#else
- EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(0));
- EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(0));
-#endif
-}
-
-#endif // !defined(MAC_IPHONE) & !defined(WEBRTC_ANDROID)
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/hardware_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/hardware_test.cc
deleted file mode 100644
index 41145e1..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/standard/hardware_test.cc
+++ /dev/null
@@ -1,168 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "after_streaming_fixture.h"
-#include "audio_device.h"
-#include "voe_test_defines.h"
-
-class HardwareTest : public AfterStreamingFixture {
-};
-
-#if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
-TEST_F(HardwareTest, AbleToQueryForDevices) {
- int num_recording_devices = 0;
- int num_playout_devices = 0;
- EXPECT_EQ(0, voe_hardware_->GetNumOfRecordingDevices(num_recording_devices));
- EXPECT_EQ(0, voe_hardware_->GetNumOfPlayoutDevices(num_playout_devices));
-
- ASSERT_GT(num_recording_devices, 0) <<
- "There seem to be no recording devices on your system, "
- "and this test really doesn't make sense then.";
- ASSERT_GT(num_playout_devices, 0) <<
- "There seem to be no playout devices on your system, "
- "and this test really doesn't make sense then.";
-
- // Recording devices are handled a bit differently on Windows - we can
- // just tell it to set the 'default' communication device there.
-#ifdef _WIN32
- // Should also work while already recording.
- EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(
- webrtc::AudioDeviceModule::kDefaultCommunicationDevice));
- // Should also work while already playing.
- EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(
- webrtc::AudioDeviceModule::kDefaultCommunicationDevice));
-#else
- // For other platforms, just use the first device encountered.
- EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(0));
- EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(0));
-#endif
-
- // It's hard to know what names this will return (it's system-dependent),
- // so just check that it's possible to do it.
- char device_name[128] = {0};
- char guid_name[128] = {0};
- EXPECT_EQ(0, voe_hardware_->GetRecordingDeviceName(
- 0, device_name, guid_name));
- EXPECT_EQ(0, voe_hardware_->GetPlayoutDeviceName(
- 0, device_name, guid_name));
-}
-#endif
-
-#ifdef _WIN32
-TEST_F(HardwareTest, GetCpuLoadWorksOnWindows) {
- int load = -1;
- EXPECT_EQ(0, voe_hardware_->GetCPULoad(load));
- EXPECT_GE(0, load);
- TEST_LOG("Voice engine CPU load = %d%%\n", load);
-}
-#else
-TEST_F(HardwareTest, GetCpuLoadReturnsErrorOnNonWindowsPlatform) {
- int load = -1;
- EXPECT_EQ(-1, voe_hardware_->GetCPULoad(load));
-}
-#endif
-
-#if !defined(WEBRTC_MAC) && !defined(WEBRTC_ANDROID)
-TEST_F(HardwareTest, GetSystemCpuLoadWorksExceptOnMacAndAndroid) {
-#ifdef _WIN32
- // This method needs some warm-up time on Windows. We sleep a good amount
- // of time instead of retrying to make the test simpler.
- Sleep(2000);
-#endif
- int load = -1;
- EXPECT_EQ(0, voe_hardware_->GetSystemCPULoad(load));
- EXPECT_GE(load, 0);
- TEST_LOG("System CPU load = %d%%\n", load);
-}
-#endif
-
-TEST_F(HardwareTest, BuiltInWasapiAECWorksForAudioWindowsCoreAudioLayer) {
-#ifdef MAC_IPHONE
- // Ensure the sound device is reset on iPhone.
- EXPECT_EQ(0, voe_hardware_->ResetAudioDevice());
- Sleep(2000);
-#endif
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
-
- webrtc::AudioLayers given_layer;
- EXPECT_EQ(0, voe_hardware_->GetAudioDeviceLayer(given_layer));
- if (given_layer != webrtc::kAudioWindowsCore) {
- // Not Windows Audio Core - then it shouldn't work.
- EXPECT_EQ(-1, voe_hardware_->EnableBuiltInAEC(true));
- EXPECT_EQ(-1, voe_hardware_->EnableBuiltInAEC(false));
- return;
- }
-
- TEST_LOG("Testing AEC for Audio Windows Core.\n");
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
-
- // Can't be set after StartSend().
- EXPECT_EQ(-1, voe_hardware_->EnableBuiltInAEC(true));
- EXPECT_EQ(-1, voe_hardware_->EnableBuiltInAEC(false));
-
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_EQ(0, voe_hardware_->EnableBuiltInAEC(true));
-
- // Can't be called before StartPlayout().
- EXPECT_EQ(-1, voe_base_->StartSend(channel_));
-
- EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
- TEST_LOG("Processing capture data with built-in AEC...\n");
- Sleep(2000);
-
- TEST_LOG("Looping through capture devices...\n");
- int num_devs = 0;
- char dev_name[128] = { 0 };
- char guid_name[128] = { 0 };
- EXPECT_EQ(0, voe_hardware_->GetNumOfRecordingDevices(num_devs));
- for (int dev_index = 0; dev_index < num_devs; ++dev_index) {
- EXPECT_EQ(0, voe_hardware_->GetRecordingDeviceName(dev_index,
- dev_name,
- guid_name));
- TEST_LOG("%d: %s\n", dev_index, dev_name);
- EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(dev_index));
- Sleep(2000);
- }
-
- EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(-1));
- EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(-1));
-
- TEST_LOG("Looping through render devices, restarting for each "
- "device...\n");
- EXPECT_EQ(0, voe_hardware_->GetNumOfPlayoutDevices(num_devs));
- for (int dev_index = 0; dev_index < num_devs; ++dev_index) {
- EXPECT_EQ(0, voe_hardware_->GetPlayoutDeviceName(dev_index,
- dev_name,
- guid_name));
- TEST_LOG("%d: %s\n", dev_index, dev_name);
- EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(dev_index));
- Sleep(2000);
- }
-
- TEST_LOG("Using default devices...\n");
- EXPECT_EQ(0, voe_hardware_->SetRecordingDevice(-1));
- EXPECT_EQ(0, voe_hardware_->SetPlayoutDevice(-1));
- Sleep(2000);
-
- // Possible, but not recommended before StopSend().
- EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
-
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
- Sleep(2000); // To verify that there is no garbage audio.
-
- TEST_LOG("Disabling built-in AEC.\n");
- EXPECT_EQ(0, voe_hardware_->EnableBuiltInAEC(false));
-
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
- EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/manual_hold_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/manual_hold_test.cc
deleted file mode 100644
index 68f28b4..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/standard/manual_hold_test.cc
+++ /dev/null
@@ -1,43 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "after_streaming_fixture.h"
-
-// Note: This class includes sleeps and requires manual verification.
-class ManualHoldTest : public AfterStreamingFixture {
-};
-
-TEST_F(ManualHoldTest, SetOnHoldStatusBlockAudio) {
- TEST_LOG("Channel not on hold => should hear audio.\n");
- Sleep(2000);
- TEST_LOG("Put channel on hold => should *not* hear audio.\n");
- EXPECT_EQ(0, voe_base_->SetOnHoldStatus(channel_, true));
- Sleep(2000);
- TEST_LOG("Remove on hold => should hear audio again.\n");
- EXPECT_EQ(0, voe_base_->SetOnHoldStatus(channel_, false));
- Sleep(2000);
- TEST_LOG("Put sending on hold => should *not* hear audio.\n");
- EXPECT_EQ(0, voe_base_->SetOnHoldStatus(channel_, true, webrtc::kHoldSendOnly));
- Sleep(2000);
-}
-
-TEST_F(ManualHoldTest, SetOnHoldStatusBlocksLocalFileAudio) {
- TEST_LOG("Start playing a file locally => "
- "you should now hear this file being played out.\n");
- voe_file_->StopPlayingFileAsMicrophone(channel_);
- EXPECT_EQ(0, voe_file_->StartPlayingFileLocally(
- channel_, resource_manager_.long_audio_file_path().c_str(), true));
- Sleep(2000);
-
- TEST_LOG("Put playing on hold => should *not* hear audio.\n");
- EXPECT_EQ(0, voe_base_->SetOnHoldStatus(
- channel_, true, webrtc::kHoldPlayOnly));
- Sleep(2000);
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/neteq_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/neteq_test.cc
deleted file mode 100644
index 8184535..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/standard/neteq_test.cc
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "after_streaming_fixture.h"
-
-class NetEQTest : public AfterStreamingFixture {
- protected:
- void SetUp() {
- additional_channel_ = voe_base_->CreateChannel();
- }
-
- void TearDown() {
- voe_base_->DeleteChannel(additional_channel_);
- }
-
- int additional_channel_;
-};
-
-TEST_F(NetEQTest, GetNetEQPlayoutModeReturnsDefaultModeByDefault) {
- webrtc::NetEqModes mode;
- EXPECT_EQ(0, voe_base_->GetNetEQPlayoutMode(channel_, mode));
- EXPECT_EQ(webrtc::kNetEqDefault, mode);
-}
-
-TEST_F(NetEQTest, SetNetEQPlayoutModeActuallySetsTheModeForTheChannel) {
- webrtc::NetEqModes mode;
- // Set for the first channel but leave the second.
- EXPECT_EQ(0, voe_base_->SetNetEQPlayoutMode(channel_, webrtc::kNetEqFax));
- EXPECT_EQ(0, voe_base_->GetNetEQPlayoutMode(channel_, mode));
- EXPECT_EQ(webrtc::kNetEqFax, mode);
-
- EXPECT_EQ(0, voe_base_->GetNetEQPlayoutMode(additional_channel_, mode));
- EXPECT_EQ(webrtc::kNetEqDefault, mode);
-
- // Set the second channel, leave the first.
- EXPECT_EQ(0, voe_base_->SetNetEQPlayoutMode(
- additional_channel_, webrtc::kNetEqStreaming));
- EXPECT_EQ(0, voe_base_->GetNetEQPlayoutMode(additional_channel_, mode));
- EXPECT_EQ(webrtc::kNetEqStreaming, mode);
-
- EXPECT_EQ(0, voe_base_->GetNetEQPlayoutMode(channel_, mode));
- EXPECT_EQ(webrtc::kNetEqFax, mode);
-}
-
-TEST_F(NetEQTest, GetNetEQBgnModeReturnsBgnOnByDefault) {
- webrtc::NetEqBgnModes bgn_mode;
- EXPECT_EQ(0, voe_base_->GetNetEQBGNMode(channel_, bgn_mode));
- EXPECT_EQ(webrtc::kBgnOn, bgn_mode);
-}
-
-TEST_F(NetEQTest, SetNetEQBgnModeActuallySetsTheBgnMode) {
- webrtc::NetEqBgnModes bgn_mode;
- EXPECT_EQ(0, voe_base_->SetNetEQBGNMode(channel_, webrtc::kBgnOff));
- EXPECT_EQ(0, voe_base_->GetNetEQBGNMode(channel_, bgn_mode));
- EXPECT_EQ(webrtc::kBgnOff, bgn_mode);
-
- EXPECT_EQ(0, voe_base_->SetNetEQBGNMode(channel_, webrtc::kBgnFade));
- EXPECT_EQ(0, voe_base_->GetNetEQBGNMode(channel_, bgn_mode));
- EXPECT_EQ(webrtc::kBgnFade, bgn_mode);
-}
-
-TEST_F(NetEQTest, ManualSetEQPlayoutModeStillProducesOkAudio) {
- EXPECT_EQ(0, voe_base_->SetNetEQPlayoutMode(channel_, webrtc::kNetEqDefault));
- TEST_LOG("NetEQ default playout mode enabled => should hear OK audio.\n");
- Sleep(2000);
-
- EXPECT_EQ(0, voe_base_->SetNetEQPlayoutMode(
- channel_, webrtc::kNetEqStreaming));
- TEST_LOG("NetEQ streaming playout mode enabled => should hear OK audio.\n");
- Sleep(2000);
-
- EXPECT_EQ(0, voe_base_->SetNetEQPlayoutMode(channel_, webrtc::kNetEqFax));
- TEST_LOG("NetEQ fax playout mode enabled => should hear OK audio.\n");
- Sleep(2000);
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/network_before_streaming_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/network_before_streaming_test.cc
deleted file mode 100644
index 7a41e80..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/standard/network_before_streaming_test.cc
+++ /dev/null
@@ -1,55 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "after_initialization_fixture.h"
-
-class NetworkBeforeStreamingTest : public AfterInitializationFixture {
- protected:
- void SetUp() {
- channel_ = voe_base_->CreateChannel();
- }
-
- void TearDown() {
- voe_base_->DeleteChannel(channel_);
- }
-
- int channel_;
-};
-
-TEST_F(NetworkBeforeStreamingTest,
- GetSourceInfoReturnsEmptyValuesForUnconfiguredChannel) {
- char src_ip[32] = "0.0.0.0";
- int src_rtp_port = 1234;
- int src_rtcp_port = 1235;
-
- EXPECT_EQ(0, voe_network_->GetSourceInfo(
- channel_, src_rtp_port, src_rtcp_port, src_ip));
- EXPECT_EQ(0, src_rtp_port);
- EXPECT_EQ(0, src_rtcp_port);
- EXPECT_STRCASEEQ("", src_ip);
-}
-
-TEST_F(NetworkBeforeStreamingTest,
- GetSourceFilterReturnsEmptyValuesForUnconfiguredChannel) {
- int filter_port = -1;
- int filter_port_rtcp = -1;
- char filter_ip[32] = "0.0.0.0";
-
- EXPECT_EQ(0, voe_network_->GetSourceFilter(
- channel_, filter_port, filter_port_rtcp, filter_ip));
-
- EXPECT_EQ(0, filter_port);
- EXPECT_EQ(0, filter_port_rtcp);
- EXPECT_STRCASEEQ("", filter_ip);
-}
-
-TEST_F(NetworkBeforeStreamingTest, SetSourceFilterSucceeds) {
- EXPECT_EQ(0, voe_network_->SetSourceFilter(channel_, 0));
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/network_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/network_test.cc
deleted file mode 100644
index 335fba6..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/standard/network_test.cc
+++ /dev/null
@@ -1,199 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "after_streaming_fixture.h"
-#include "mock/mock_voe_observer.h"
-#include "mock/mock_voe_connection_observer.h"
-#include "voe_test_interface.h"
-
-static const int kDefaultRtpPort = 8000;
-static const int kDefaultRtcpPort = 8001;
-
-class NetworkTest : public AfterStreamingFixture {
-};
-
-using ::testing::Between;
-
-TEST_F(NetworkTest, GetSourceInfoReturnsPortsAndIpAfterReceivingPackets) {
- // Give some time to send speech packets.
- Sleep(200);
-
- int rtp_port = 0;
- int rtcp_port = 0;
- char source_ip[32] = "127.0.0.1";
-
- EXPECT_EQ(0, voe_network_->GetSourceInfo(channel_, rtp_port, rtcp_port,
- source_ip));
-
- EXPECT_EQ(kDefaultRtpPort, rtp_port);
- EXPECT_EQ(kDefaultRtcpPort, rtcp_port);
-}
-
-TEST_F(NetworkTest, NoFilterIsEnabledByDefault) {
- int filter_rtp_port = -1;
- int filter_rtcp_port = -1;
- char filter_ip[64] = { 0 };
-
- EXPECT_EQ(0, voe_network_->GetSourceFilter(
- channel_, filter_rtp_port, filter_rtcp_port, filter_ip));
-
- EXPECT_EQ(0, filter_rtp_port);
- EXPECT_EQ(0, filter_rtcp_port);
- EXPECT_STREQ("", filter_ip);
-}
-
-TEST_F(NetworkTest, ManualCanFilterRtpPort) {
- TEST_LOG("No filter, should hear audio.\n");
- Sleep(1000);
-
- int port_to_block = kDefaultRtpPort + 10;
- EXPECT_EQ(0, voe_network_->SetSourceFilter(channel_, port_to_block));
-
- // Changes should take effect immediately.
- int filter_rtp_port = -1;
- int filter_rtcp_port = -1;
- char filter_ip[64] = { 0 };
-
- EXPECT_EQ(0, voe_network_->GetSourceFilter(
- channel_, filter_rtp_port, filter_rtcp_port, filter_ip));
-
- EXPECT_EQ(port_to_block, filter_rtp_port);
-
- TEST_LOG("Now filtering port %d, should not hear audio.\n", port_to_block);
- Sleep(1000);
-
- TEST_LOG("Removing filter, should hear audio.\n");
- EXPECT_EQ(0, voe_network_->SetSourceFilter(channel_, 0));
- Sleep(1000);
-}
-
-TEST_F(NetworkTest, ManualCanFilterIp) {
- TEST_LOG("You should hear audio.\n");
- Sleep(1000);
-
- int rtcp_port_to_block = kDefaultRtcpPort + 10;
- TEST_LOG("Filtering IP 10.10.10.10, should not hear audio.\n");
- EXPECT_EQ(0, voe_network_->SetSourceFilter(
- channel_, 0, rtcp_port_to_block, "10.10.10.10"));
-
- int filter_rtp_port = -1;
- int filter_rtcp_port = -1;
- char filter_ip[64] = { 0 };
- EXPECT_EQ(0, voe_network_->GetSourceFilter(
- channel_, filter_rtp_port, filter_rtcp_port, filter_ip));
-
- EXPECT_EQ(0, filter_rtp_port);
- EXPECT_EQ(rtcp_port_to_block, filter_rtcp_port);
- EXPECT_STREQ("10.10.10.10", filter_ip);
-}
-
-TEST_F(NetworkTest,
- CallsObserverOnTimeoutAndRestartWhenPacketTimeoutNotificationIsEnabled) {
- // First, get rid of the default, asserting observer and install our observer.
- EXPECT_EQ(0, voe_base_->DeRegisterVoiceEngineObserver());
- webrtc::MockVoEObserver mock_observer;
- EXPECT_EQ(0, voe_base_->RegisterVoiceEngineObserver(mock_observer));
-
- // Define expectations.
- int expected_error = VE_RECEIVE_PACKET_TIMEOUT;
- EXPECT_CALL(mock_observer, CallbackOnError(channel_, expected_error))
- .Times(1);
- expected_error = VE_PACKET_RECEIPT_RESTARTED;
- EXPECT_CALL(mock_observer, CallbackOnError(channel_, expected_error))
- .Times(1);
-
- // Get some speech going.
- Sleep(500);
-
- // Enable packet timeout.
- EXPECT_EQ(0, voe_network_->SetPacketTimeoutNotification(channel_, true, 1));
-
- // Trigger a timeout.
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- Sleep(1500);
-
- // Trigger a restart event.
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
- Sleep(500);
-}
-
-TEST_F(NetworkTest, DoesNotCallDeRegisteredObserver) {
- // De-register the default observer. This test will fail if the observer gets
- // called for any reason, so if this de-register doesn't work the test will
- // fail.
- EXPECT_EQ(0, voe_base_->DeRegisterVoiceEngineObserver());
-
- // Get some speech going.
- Sleep(500);
-
- // Enable packet timeout.
- EXPECT_EQ(0, voe_network_->SetPacketTimeoutNotification(channel_, true, 1));
-
- // Trigger a timeout.
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- Sleep(1500);
-}
-
-TEST_F(NetworkTest, DeadOrAliveObserverSeesAliveMessagesIfEnabled) {
- webrtc::MockVoeConnectionObserver mock_observer;
- EXPECT_EQ(0, voe_network_->RegisterDeadOrAliveObserver(
- channel_, mock_observer));
-
- // We should be called about 4 times in four seconds, but 3 is OK too.
- EXPECT_CALL(mock_observer, OnPeriodicDeadOrAlive(channel_, true))
- .Times(Between(3, 4));
-
- EXPECT_EQ(0, voe_network_->SetPeriodicDeadOrAliveStatus(channel_, true, 1));
- Sleep(4000);
-
- EXPECT_EQ(0, voe_network_->DeRegisterDeadOrAliveObserver(channel_));
-}
-
-TEST_F(NetworkTest, DeadOrAliveObserverSeesDeadMessagesIfEnabled) {
- // "When do you see them?" - "All the time!"
- webrtc::MockVoeConnectionObserver mock_observer;
- EXPECT_EQ(0, voe_network_->RegisterDeadOrAliveObserver(
- channel_, mock_observer));
-
- Sleep(500);
-
- // We should be called about 4 times in four seconds, but 3 is OK too.
- EXPECT_CALL(mock_observer, OnPeriodicDeadOrAlive(channel_, false))
- .Times(Between(3, 4));
-
- EXPECT_EQ(0, voe_network_->SetPeriodicDeadOrAliveStatus(channel_, true, 1));
- EXPECT_EQ(0, voe_rtp_rtcp_->SetRTCPStatus(channel_, false));
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- Sleep(4000);
-
- EXPECT_EQ(0, voe_network_->DeRegisterDeadOrAliveObserver(channel_));
-}
-
-TEST_F(NetworkTest, CanSwitchToExternalTransport) {
- EXPECT_EQ(0, voe_base_->StopReceive(channel_));
- EXPECT_EQ(0, voe_base_->DeleteChannel(channel_));
- channel_ = voe_base_->CreateChannel();
-
- voetest::FakeExternalTransport external_transport(voe_network_);
- EXPECT_EQ(0, voe_network_->RegisterExternalTransport(
- channel_, external_transport));
-
- EXPECT_EQ(0, voe_base_->StartReceive(channel_));
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
- EXPECT_EQ(0, voe_base_->StartPlayout(channel_));
-
- Sleep(1000);
-
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_EQ(0, voe_base_->StopPlayout(channel_));
- EXPECT_EQ(0, voe_base_->StopReceive(channel_));
-
- EXPECT_EQ(0, voe_network_->DeRegisterExternalTransport(channel_));
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc
deleted file mode 100644
index afd0820..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_before_streaming_test.cc
+++ /dev/null
@@ -1,121 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "after_initialization_fixture.h"
-
-using namespace webrtc;
-using namespace testing;
-
-class RtpRtcpBeforeStreamingTest : public AfterInitializationFixture {
- protected:
- void SetUp();
- void TearDown();
-
- int channel_;
-};
-
-void RtpRtcpBeforeStreamingTest::SetUp() {
- EXPECT_THAT(channel_ = voe_base_->CreateChannel(), Not(Lt(0)));
-}
-
-void RtpRtcpBeforeStreamingTest::TearDown() {
- EXPECT_EQ(0, voe_base_->DeleteChannel(channel_));
-}
-
-TEST_F(RtpRtcpBeforeStreamingTest,
- GetRtcpStatusReturnsTrueByDefaultAndObeysSetRtcpStatus) {
- bool on = false;
- EXPECT_EQ(0, voe_rtp_rtcp_->GetRTCPStatus(channel_, on));
- EXPECT_TRUE(on);
- EXPECT_EQ(0, voe_rtp_rtcp_->SetRTCPStatus(channel_, false));
- EXPECT_EQ(0, voe_rtp_rtcp_->GetRTCPStatus(channel_, on));
- EXPECT_FALSE(on);
- EXPECT_EQ(0, voe_rtp_rtcp_->SetRTCPStatus(channel_, true));
- EXPECT_EQ(0, voe_rtp_rtcp_->GetRTCPStatus(channel_, on));
- EXPECT_TRUE(on);
-}
-
-TEST_F(RtpRtcpBeforeStreamingTest, RtpKeepAliveStatusIsOffByDefault) {
- unsigned char payload_type;
- int delta_seconds;
- bool on;
-
- // Should be off by default.
- EXPECT_EQ(0, voe_rtp_rtcp_->GetRTPKeepaliveStatus(
- channel_, on, payload_type, delta_seconds));
- EXPECT_FALSE(on);
- EXPECT_EQ(255, payload_type);
- EXPECT_EQ(0, delta_seconds);
-}
-
-TEST_F(RtpRtcpBeforeStreamingTest, SetRtpKeepAliveDealsWithInvalidParameters) {
- unsigned char payload_type;
- int delta_seconds;
- bool on;
-
- // Verify invalid input parameters.
- EXPECT_NE(0, voe_rtp_rtcp_->GetRTPKeepaliveStatus(
- -1, on, payload_type, delta_seconds)) <<
- "Should fail for invalid channel -1.";
- EXPECT_NE(0, voe_rtp_rtcp_->SetRTPKeepaliveStatus(
- -1, true, 0, 15)) <<
- "Should fail for invalid channel -1.";
- EXPECT_NE(0, voe_rtp_rtcp_->SetRTPKeepaliveStatus(
- channel_, true, -1, 15)) <<
- "Should fail for invalid payload -1.";
- EXPECT_NE(0, voe_rtp_rtcp_->SetRTPKeepaliveStatus(
- channel_, true, 0, 61)) <<
- "The delta time must be [1, 60] seconds.";
- EXPECT_EQ(0, voe_rtp_rtcp_->GetRTPKeepaliveStatus(
- channel_, on, payload_type, delta_seconds));
- EXPECT_NE(0, voe_rtp_rtcp_->SetRTPKeepaliveStatus(
- channel_, true, 0));
-
- // Should still be off, default 0 used by PCMU.
- EXPECT_FALSE(on);
-}
-
-TEST_F(RtpRtcpBeforeStreamingTest,
- GetRtpKeepaliveStatusObeysSetRtpKeepaliveStatus) {
- EXPECT_EQ(0, voe_rtp_rtcp_->SetRTCP_CNAME(channel_, "SomeName"));
-
- // Try valid settings.
- EXPECT_EQ(0, voe_rtp_rtcp_->SetRTPKeepaliveStatus(
- channel_, true, 1));
-
- unsigned char payload_type;
- int delta_seconds;
- bool on;
-
- EXPECT_EQ(0, voe_rtp_rtcp_->GetRTPKeepaliveStatus(
- 0, on, payload_type, delta_seconds));
- EXPECT_TRUE(on);
- EXPECT_EQ(1, payload_type);
- EXPECT_EQ(15, delta_seconds) << "15 seconds delta is default.";
-
- // Set the keep-alive payload to 60, which the codecs can't use.
- EXPECT_EQ(0, voe_rtp_rtcp_->SetRTPKeepaliveStatus(
- channel_, true, 60, 3));
- EXPECT_EQ(0, voe_rtp_rtcp_->GetRTPKeepaliveStatus(
- channel_, on, payload_type, delta_seconds));
- EXPECT_TRUE(on);
- EXPECT_EQ(60, payload_type);
- EXPECT_EQ(3, delta_seconds);
-
- EXPECT_EQ(0, voe_rtp_rtcp_->SetRTPKeepaliveStatus(
- channel_, false, 60));
-}
-
-TEST_F(RtpRtcpBeforeStreamingTest, GetLocalSsrcObeysSetLocalSsrc) {
- EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(channel_, 1234));
- unsigned int result = 0;
- EXPECT_EQ(0, voe_rtp_rtcp_->GetLocalSSRC(channel_, result));
- EXPECT_EQ(1234u, result);
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_test.cc
deleted file mode 100644
index 42c1b4e..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/standard/rtp_rtcp_test.cc
+++ /dev/null
@@ -1,195 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "after_streaming_fixture.h"
-#include "voe_standard_test.h"
-#include "testsupport/fileutils.h"
-
-static const char* const RTCP_CNAME = "Whatever";
-
-class RtpRtcpTest : public AfterStreamingFixture {
- protected:
- void SetUp() {
- // We need a second channel for this test, so set it up.
- second_channel_ = voe_base_->CreateChannel();
- EXPECT_GE(second_channel_, 0);
-
- EXPECT_EQ(0, voe_base_->SetSendDestination(
- second_channel_, 8002, "127.0.0.1"));
- EXPECT_EQ(0, voe_base_->SetLocalReceiver(
- second_channel_, 8002));
- EXPECT_EQ(0, voe_base_->StartReceive(second_channel_));
- EXPECT_EQ(0, voe_base_->StartPlayout(second_channel_));
- EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(second_channel_, 5678));
- EXPECT_EQ(0, voe_base_->StartSend(second_channel_));
-
- // We'll set up the RTCP CNAME and SSRC to something arbitrary here.
- voe_rtp_rtcp_->SetRTCP_CNAME(channel_, RTCP_CNAME);
- }
-
- void TearDown() {
- voe_base_->DeleteChannel(second_channel_);
- }
-
- int second_channel_;
-};
-
-TEST_F(RtpRtcpTest, RemoteRtcpCnameHasPropagatedToRemoteSide) {
- // We need to sleep a bit here for the name to propagate. For instance,
- // 200 milliseconds is not enough, so we'll go with one second here.
- Sleep(1000);
-
- char char_buffer[256];
- voe_rtp_rtcp_->GetRemoteRTCP_CNAME(channel_, char_buffer);
- EXPECT_STREQ(RTCP_CNAME, char_buffer);
-}
-
-TEST_F(RtpRtcpTest, SSRCPropagatesCorrectly) {
- unsigned int local_ssrc = 1234;
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(channel_, local_ssrc));
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
-
- Sleep(1000);
-
- unsigned int ssrc;
- EXPECT_EQ(0, voe_rtp_rtcp_->GetLocalSSRC(channel_, ssrc));
- EXPECT_EQ(local_ssrc, ssrc);
-
- EXPECT_EQ(0, voe_rtp_rtcp_->GetRemoteSSRC(channel_, ssrc));
- EXPECT_EQ(local_ssrc, ssrc);
-}
-
-TEST_F(RtpRtcpTest, RtcpApplicationDefinedPacketsCanBeSentAndReceived) {
- voetest::RtcpAppHandler rtcp_app_handler;
- EXPECT_EQ(0, voe_rtp_rtcp_->RegisterRTCPObserver(
- channel_, rtcp_app_handler));
-
- // Send data aligned to 32 bytes.
- const char* data = "application-dependent data------";
- unsigned short data_length = strlen(data);
- unsigned int data_name = 0x41424344; // 'ABCD' in ascii
- unsigned char data_subtype = 1;
-
- EXPECT_EQ(0, voe_rtp_rtcp_->SendApplicationDefinedRTCPPacket(
- channel_, data_subtype, data_name, data, data_length));
-
- // Ensure the RTP-RTCP process gets scheduled.
- Sleep(1000);
-
- // Ensure we received the data in the callback.
- EXPECT_EQ(data_length, rtcp_app_handler.length_in_bytes_);
- EXPECT_EQ(0, memcmp(data, rtcp_app_handler.data_, data_length));
- EXPECT_EQ(data_name, rtcp_app_handler.name_);
- EXPECT_EQ(data_subtype, rtcp_app_handler.sub_type_);
-
- EXPECT_EQ(0, voe_rtp_rtcp_->DeRegisterRTCPObserver(channel_));
-}
-
-TEST_F(RtpRtcpTest, DisabledRtcpObserverDoesNotReceiveData) {
- voetest::RtcpAppHandler rtcp_app_handler;
- EXPECT_EQ(0, voe_rtp_rtcp_->RegisterRTCPObserver(
- channel_, rtcp_app_handler));
-
- // Put observer in a known state before de-registering.
- rtcp_app_handler.Reset();
-
- EXPECT_EQ(0, voe_rtp_rtcp_->DeRegisterRTCPObserver(channel_));
-
- const char* data = "whatever";
- EXPECT_EQ(0, voe_rtp_rtcp_->SendApplicationDefinedRTCPPacket(
- channel_, 1, 0x41424344, data, strlen(data)));
-
- // Ensure the RTP-RTCP process gets scheduled.
- Sleep(1000);
-
- // Ensure we received no data.
- EXPECT_EQ(0u, rtcp_app_handler.name_);
- EXPECT_EQ(0u, rtcp_app_handler.sub_type_);
-}
-
-TEST_F(RtpRtcpTest, InsertExtraRTPPacketDealsWithInvalidArguments) {
- const char payload_data[8] = { 'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H' };
-
- EXPECT_EQ(-1, voe_rtp_rtcp_->InsertExtraRTPPacket(
- -1, 0, false, payload_data, 8)) <<
- "Should reject: invalid channel.";
- EXPECT_EQ(-1, voe_rtp_rtcp_->InsertExtraRTPPacket(
- channel_, -1, false, payload_data, 8)) <<
- "Should reject: invalid payload type.";
- EXPECT_EQ(-1, voe_rtp_rtcp_->InsertExtraRTPPacket(
- channel_, 128, false, payload_data, 8)) <<
- "Should reject: invalid payload type.";
- EXPECT_EQ(-1, voe_rtp_rtcp_->InsertExtraRTPPacket(
- channel_, 99, false, NULL, 8)) <<
- "Should reject: bad pointer.";
- EXPECT_EQ(-1, voe_rtp_rtcp_->InsertExtraRTPPacket(
- channel_, 99, false, payload_data, 1500 - 28 + 1)) <<
- "Should reject: invalid size.";
-}
-
-TEST_F(RtpRtcpTest, CanTransmitExtraRtpPacketsWithoutError) {
- const char payload_data[8] = { 'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H' };
-
- for (int i = 0; i < 128; ++i) {
- // Try both with and without the marker bit set
- EXPECT_EQ(0, voe_rtp_rtcp_->InsertExtraRTPPacket(
- channel_, i, false, payload_data, 8));
- EXPECT_EQ(0, voe_rtp_rtcp_->InsertExtraRTPPacket(
- channel_, i, true, payload_data, 8));
- }
-}
-
-TEST_F(RtpRtcpTest, CanCreateRtpDumpFilesWithoutError) {
- // Create two RTP dump files (3 seconds long). You can verify these after
- // the test using rtpplay or NetEqRTPplay if you like.
- std::string output_path = webrtc::test::OutputPath();
- std::string incoming_filename = output_path + "dump_in_3sec.rtp";
- std::string outgoing_filename = output_path + "dump_out_3sec.rtp";
-
- EXPECT_EQ(0, voe_rtp_rtcp_->StartRTPDump(
- channel_, incoming_filename.c_str(), webrtc::kRtpIncoming));
- EXPECT_EQ(0, voe_rtp_rtcp_->StartRTPDump(
- channel_, outgoing_filename.c_str(), webrtc::kRtpOutgoing));
-
- Sleep(3000);
-
- EXPECT_EQ(0, voe_rtp_rtcp_->StopRTPDump(channel_, webrtc::kRtpIncoming));
- EXPECT_EQ(0, voe_rtp_rtcp_->StopRTPDump(channel_, webrtc::kRtpOutgoing));
-}
-
-TEST_F(RtpRtcpTest, ObserverGetsNotifiedOnSsrcChange) {
- voetest::TestRtpObserver rtcp_observer;
- EXPECT_EQ(0, voe_rtp_rtcp_->RegisterRTPObserver(
- channel_, rtcp_observer));
- rtcp_observer.Reset();
-
- unsigned int new_ssrc = 7777;
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(channel_, new_ssrc));
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
-
- Sleep(500);
-
- // Verify we got the new SSRC.
- EXPECT_EQ(new_ssrc, rtcp_observer.ssrc_[0]);
-
- // Now try another SSRC.
- unsigned int newer_ssrc = 1717;
- EXPECT_EQ(0, voe_base_->StopSend(channel_));
- EXPECT_EQ(0, voe_rtp_rtcp_->SetLocalSSRC(channel_, newer_ssrc));
- EXPECT_EQ(0, voe_base_->StartSend(channel_));
-
- Sleep(500);
-
- EXPECT_EQ(newer_ssrc, rtcp_observer.ssrc_[0]);
-
- EXPECT_EQ(0, voe_rtp_rtcp_->DeRegisterRTPObserver(channel_));
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/voe_base_misc_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/voe_base_misc_test.cc
deleted file mode 100644
index 0388025..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/standard/voe_base_misc_test.cc
+++ /dev/null
@@ -1,26 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "before_initialization_fixture.h"
-
-class VoeBaseMiscTest : public BeforeInitializationFixture {
-};
-
-using namespace testing;
-
-TEST_F(VoeBaseMiscTest, MaxNumChannelsIs32) {
- EXPECT_EQ(32, voe_base_->MaxNumOfChannels());
-}
-
-TEST_F(VoeBaseMiscTest, GetVersionPrintsSomeUsefulInformation) {
- char char_buffer[1024];
- EXPECT_EQ(0, voe_base_->GetVersion(char_buffer));
- EXPECT_THAT(char_buffer, ContainsRegex("VoiceEngine"));
-}
diff --git a/trunk/src/voice_engine/main/test/auto_test/standard/volume_test.cc b/trunk/src/voice_engine/main/test/auto_test/standard/volume_test.cc
deleted file mode 100644
index 44887b7..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/standard/volume_test.cc
+++ /dev/null
@@ -1,238 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "after_streaming_fixture.h"
-
-class VolumeTest : public AfterStreamingFixture {
-};
-
-TEST_F(VolumeTest, DefaultSpeakerVolumeIsAtMost255) {
- unsigned int volume = 1000;
- EXPECT_EQ(0, voe_volume_control_->GetSpeakerVolume(volume));
- EXPECT_LE(volume, 255u);
-}
-
-TEST_F(VolumeTest, ManualSetVolumeWorks) {
- unsigned int original_volume = 0;
- EXPECT_EQ(0, voe_volume_control_->GetSpeakerVolume(original_volume));
- Sleep(1000);
-
- TEST_LOG("Setting speaker volume to 0 out of 255.\n");
- EXPECT_EQ(0, voe_volume_control_->SetSpeakerVolume(0));
- Sleep(1000);
-
- TEST_LOG("Setting speaker volume to 100 out of 255.\n");
- EXPECT_EQ(0, voe_volume_control_->SetSpeakerVolume(100));
- Sleep(1000);
-
- // Set the volume to 255 very briefly so we don't blast the poor user
- // listening to this. This is just to test the call succeeds.
- EXPECT_EQ(0, voe_volume_control_->SetSpeakerVolume(255));
-
- TEST_LOG("Setting speaker volume to the original %d out of 255.\n",
- original_volume);
- EXPECT_EQ(0, voe_volume_control_->SetSpeakerVolume(original_volume));
- Sleep(1000);
-}
-
-#if !defined(MAC_IPHONE)
-
-// NOTE(phoglund): This test is flaky because of how the OS works, and is hence
-// disabled by default.
-TEST_F(VolumeTest, DISABLED_DefaultMicrophoneVolumeIsAtMost255) {
- unsigned int volume = 1000;
- EXPECT_EQ(0, voe_volume_control_->GetMicVolume(volume));
- EXPECT_LE(volume, 255u);
-}
-
-TEST_F(VolumeTest, ManualRequiresMicrophoneCanSetMicrophoneVolumeWithAcgOff) {
- SwitchToManualMicrophone();
- EXPECT_EQ(0, voe_apm_->SetAgcStatus(false));
-
- unsigned int original_volume = 0;
- EXPECT_EQ(0, voe_volume_control_->GetMicVolume(original_volume));
-
- TEST_LOG("Setting microphone volume to 0.\n");
- EXPECT_EQ(0, voe_volume_control_->SetMicVolume(channel_));
- Sleep(1000);
- TEST_LOG("Setting microphone volume to 255.\n");
- EXPECT_EQ(0, voe_volume_control_->SetMicVolume(255));
- Sleep(1000);
- TEST_LOG("Setting microphone volume back to saved value.\n");
- EXPECT_EQ(0, voe_volume_control_->SetMicVolume(original_volume));
- Sleep(1000);
-}
-
-TEST_F(VolumeTest, ChannelScalingIsOneByDefault) {
- float scaling = -1.0f;
-
- EXPECT_EQ(0, voe_volume_control_->GetChannelOutputVolumeScaling(
- channel_, scaling));
- EXPECT_FLOAT_EQ(1.0f, scaling);
-}
-
-TEST_F(VolumeTest, ManualCanSetChannelScaling) {
- EXPECT_EQ(0, voe_volume_control_->SetChannelOutputVolumeScaling(
- channel_, 0.1f));
-
- float scaling = 1.0f;
- EXPECT_EQ(0, voe_volume_control_->GetChannelOutputVolumeScaling(
- channel_, scaling));
-
- EXPECT_FLOAT_EQ(0.1f, scaling);
-
- TEST_LOG("Channel scaling set to 0.1: audio should be barely audible.\n");
- Sleep(2000);
-}
-
-#endif // !MAC_IPHONE
-
-#if !defined(WEBRTC_ANDROID) && !defined(MAC_IPHONE)
-
-TEST_F(VolumeTest, InputMutingIsNotEnabledByDefault) {
- bool is_muted = true;
- EXPECT_EQ(0, voe_volume_control_->GetInputMute(channel_, is_muted));
- EXPECT_FALSE(is_muted);
-}
-
-TEST_F(VolumeTest, ManualInputMutingMutesMicrophone) {
- SwitchToManualMicrophone();
-
- // Enable muting.
- EXPECT_EQ(0, voe_volume_control_->SetInputMute(channel_, true));
- bool is_muted = false;
- EXPECT_EQ(0, voe_volume_control_->GetInputMute(channel_, is_muted));
- EXPECT_TRUE(is_muted);
-
- TEST_LOG("Muted: talk into microphone and verify you can't hear yourself.\n");
- Sleep(2000);
-
- // Test that we can disable muting.
- EXPECT_EQ(0, voe_volume_control_->SetInputMute(channel_, false));
- EXPECT_EQ(0, voe_volume_control_->GetInputMute(channel_, is_muted));
- EXPECT_FALSE(is_muted);
-
- TEST_LOG("Unmuted: talk into microphone and verify you can hear yourself.\n");
- Sleep(2000);
-}
-
-// NOTE(phoglund): This test is flaky because of how the OS works, and is hence
-// disabled by default.
-TEST_F(VolumeTest, DISABLED_SystemInputMutingIsNotEnabledByDefault) {
- bool is_muted = true;
- EXPECT_EQ(0, voe_volume_control_->GetSystemInputMute(is_muted));
- EXPECT_FALSE(is_muted);
-}
-
-TEST_F(VolumeTest, ManualSystemInputMutingMutesMicrophone) {
- SwitchToManualMicrophone();
-
- // Enable system input muting.
- EXPECT_EQ(0, voe_volume_control_->SetSystemInputMute(true));
- bool is_muted = false;
- EXPECT_EQ(0, voe_volume_control_->GetSystemInputMute(is_muted));
- EXPECT_TRUE(is_muted);
-
- TEST_LOG("Muted: talk into microphone and verify you can't hear yourself.\n");
- Sleep(2000);
-
- // Test that we can disable system input muting.
- EXPECT_EQ(0, voe_volume_control_->SetSystemInputMute(false));
- EXPECT_EQ(0, voe_volume_control_->GetSystemInputMute(is_muted));
- EXPECT_FALSE(is_muted);
-
- TEST_LOG("Unmuted: talk into microphone and verify you can hear yourself.\n");
- Sleep(2000);
-}
-
-TEST_F(VolumeTest, SystemOutputMutingIsNotEnabledByDefault) {
- bool is_muted = true;
- EXPECT_EQ(0, voe_volume_control_->GetSystemOutputMute(is_muted));
- EXPECT_FALSE(is_muted);
-}
-
-TEST_F(VolumeTest, ManualSystemOutputMutingMutesOutput) {
- // Enable muting.
- EXPECT_EQ(0, voe_volume_control_->SetSystemOutputMute(true));
- bool is_muted = false;
- EXPECT_EQ(0, voe_volume_control_->GetSystemOutputMute(is_muted));
- EXPECT_TRUE(is_muted);
-
- TEST_LOG("Muted: you should hear no audio.\n");
- Sleep(2000);
-
- // Test that we can disable muting.
- EXPECT_EQ(0, voe_volume_control_->SetSystemOutputMute(false));
- EXPECT_EQ(0, voe_volume_control_->GetSystemOutputMute(is_muted));
- EXPECT_FALSE(is_muted);
-
- TEST_LOG("Unmuted: you should hear audio.\n");
- Sleep(2000);
-}
-
-TEST_F(VolumeTest, ManualTestInputAndOutputLevels) {
- SwitchToManualMicrophone();
-
- TEST_LOG("Speak and verify that the following levels look right:\n");
- for (int i = 0; i < 5; i++) {
- Sleep(1000);
- unsigned int input_level = 0;
- unsigned int output_level = 0;
- unsigned int input_level_full_range = 0;
- unsigned int output_level_full_range = 0;
-
- EXPECT_EQ(0, voe_volume_control_->GetSpeechInputLevel(
- input_level));
- EXPECT_EQ(0, voe_volume_control_->GetSpeechOutputLevel(
- channel_, output_level));
- EXPECT_EQ(0, voe_volume_control_->GetSpeechInputLevelFullRange(
- input_level_full_range));
- EXPECT_EQ(0, voe_volume_control_->GetSpeechOutputLevelFullRange(
- channel_, output_level_full_range));
-
- TEST_LOG(" warped levels (0-9) : in=%5d, out=%5d\n",
- input_level, output_level);
- TEST_LOG(" linear levels (0-32768): in=%5d, out=%5d\n",
- input_level_full_range, output_level_full_range);
- }
-}
-
-TEST_F(VolumeTest, ChannelsAreNotPannedByDefault) {
- float left = -1.0;
- float right = -1.0;
-
- EXPECT_EQ(0, voe_volume_control_->GetOutputVolumePan(channel_, left, right));
- EXPECT_FLOAT_EQ(1.0, left);
- EXPECT_FLOAT_EQ(1.0, right);
-}
-
-TEST_F(VolumeTest, ManualTestChannelPanning) {
- TEST_LOG("Panning left.\n");
- EXPECT_EQ(0, voe_volume_control_->SetOutputVolumePan(channel_, 0.8f, 0.1f));
- Sleep(1000);
-
- TEST_LOG("Back to center.\n");
- EXPECT_EQ(0, voe_volume_control_->SetOutputVolumePan(channel_, 1.0f, 1.0f));
- Sleep(1000);
-
- TEST_LOG("Panning right.\n");
- EXPECT_EQ(0, voe_volume_control_->SetOutputVolumePan(channel_, 0.1f, 0.8f));
- Sleep(1000);
-
- // To finish, verify that the getter works.
- float left = 0.0f;
- float right = 0.0f;
-
- EXPECT_EQ(0, voe_volume_control_->GetOutputVolumePan(channel_, left, right));
- EXPECT_FLOAT_EQ(0.1f, left);
- EXPECT_FLOAT_EQ(0.8f, right);
-}
-
-#endif // !WEBRTC_ANDROID && !MAC_IPHONE
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_cpu_test.cc b/trunk/src/voice_engine/main/test/auto_test/voe_cpu_test.cc
deleted file mode 100644
index 051bce9..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/voe_cpu_test.cc
+++ /dev/null
@@ -1,99 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <string.h>
-#include <time.h>
-#include <cassert>
-#if defined(_WIN32)
-#include <conio.h>
-#endif
-
-#include "voe_cpu_test.h"
-
-using namespace webrtc;
-
-namespace voetest {
-
-#define CHECK(expr) \
- if (expr) \
- { \
- printf("Error at line: %i, %s \n", __LINE__, #expr); \
- printf("Error code: %i \n", base->LastError()); \
- PAUSE \
- return -1; \
- }
-
-extern char* GetFilename(char* filename);
-extern const char* GetFilename(const char* filename);
-extern int GetResource(char* resource, char* dest, int destLen);
-extern char* GetResource(char* resource);
-extern const char* GetResource(const char* resource);
-
-VoECpuTest::VoECpuTest(VoETestManager& mgr)
- : _mgr(mgr) {
-
-}
-
-int VoECpuTest::DoTest() {
- printf("------------------------------------------------\n");
- printf(" CPU Reference Test\n");
- printf("------------------------------------------------\n");
-
- VoEBase* base = _mgr.BasePtr();
- VoEFile* file = _mgr.FilePtr();
- VoECodec* codec = _mgr.CodecPtr();
- VoEAudioProcessing* apm = _mgr.APMPtr();
-
- int channel(-1);
- CodecInst isac;
-
- isac.pltype = 104;
- strcpy(isac.plname, "ISAC");
- isac.pacsize = 960;
- isac.plfreq = 32000;
- isac.channels = 1;
- isac.rate = -1;
-
- CHECK(base->Init());
- channel = base->CreateChannel();
-
- CHECK(base->SetLocalReceiver(channel, 5566));
- CHECK(base->SetSendDestination(channel, 5566, "127.0.0.1"));
- CHECK(codec->SetRecPayloadType(channel, isac));
- CHECK(codec->SetSendCodec(channel, isac));
-
- CHECK(base->StartReceive(channel));
- CHECK(base->StartPlayout(channel));
- CHECK(base->StartSend(channel));
- CHECK(file->StartPlayingFileAsMicrophone(channel, _mgr.AudioFilename(),
- true, true));
-
- CHECK(codec->SetVADStatus(channel, true));
- CHECK(apm->SetAgcStatus(true, kAgcAdaptiveAnalog));
- CHECK(apm->SetNsStatus(true, kNsModerateSuppression));
- CHECK(apm->SetEcStatus(true, kEcAec));
-
- TEST_LOG("\nMeasure CPU and memory while running a full-duplex"
- " iSAC-swb call.\n\n");
-
- PAUSE
-
- CHECK(base->StopSend(channel));
- CHECK(base->StopPlayout(channel));
- CHECK(base->StopReceive(channel));
-
- base->DeleteChannel(channel);
- CHECK(base->Terminate());
-
- return 0;
-}
-
-} // namespace voetest
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_cpu_test.h b/trunk/src/voice_engine/main/test/auto_test/voe_cpu_test.h
deleted file mode 100644
index f883075..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/voe_cpu_test.h
+++ /dev/null
@@ -1,31 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_CPU_TEST_H
-#define WEBRTC_VOICE_ENGINE_VOE_CPU_TEST_H
-
-#include "voe_standard_test.h"
-
-namespace voetest {
-
-class VoETestManager;
-
-class VoECpuTest {
- public:
- VoECpuTest(VoETestManager& mgr);
- ~VoECpuTest() {}
- int DoTest();
- private:
- VoETestManager& _mgr;
-};
-
-} // namespace voetest
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_CPU_TEST_H
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_extended_test.cc b/trunk/src/voice_engine/main/test/auto_test/voe_extended_test.cc
deleted file mode 100644
index 8b4ad45..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/voe_extended_test.cc
+++ /dev/null
@@ -1,8421 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <string.h>
-
-#include <vector>
-
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-#include "voe_extended_test.h"
-#include "../../source/voice_engine_defines.h" // defines build macros
-#include "system_wrappers/interface/ref_count.h"
-
-#if defined(_WIN32)
-#include <conio.h>
-#include <winsock2.h>
-#elif defined(WEBRTC_LINUX) || defined(WEBRTC_MAC)
-#include <netdb.h>
-#endif
-
-using namespace webrtc;
-
-namespace voetest {
-
-// Set this flag to ensure that test packets are transmitted to
-// RemoteIP::RemotePort during tests of SetSendToS and SetSendGQos. Requires
-// receiver at the remote side and Wireshark with a proper ip.src filter.
-#define _SEND_TO_REMOTE_IP_
-
-#ifdef _SEND_TO_REMOTE_IP_
-const int RemotePort = 12345; // transmit to this UDP port
-const char* RemoteIP = "192.168.200.1"; // transmit to this IP address
-#endif
-
-#ifdef MAC_IPHONE
-#define SLEEP_IF_IPHONE(x) SLEEP(x)
-#else
-#define SLEEP_IF_IPHONE(x)
-#endif
-
-#ifdef WEBRTC_ANDROID
-// Global pointers
-extern void* globalJavaVM;
-extern void* globalContext;
-#endif
-
-extern char* GetFilename(char* filename);
-extern const char* GetFilename(const char* filename);
-extern int GetResource(char* resource, char* dest, int destLen);
-extern char* GetResource(char* resource);
-extern const char* GetResource(const char* resource);
-
-// ----------------------------------------------------------------------------
-// External AudioDeviceModule implementation
-// ----------------------------------------------------------------------------
-
-// static
-AudioDeviceModuleImpl* AudioDeviceModuleImpl::Create() {
- AudioDeviceModuleImpl* xADM = new AudioDeviceModuleImpl();
- if (xADM)
- xADM->AddRef();
- return xADM;
-}
-
-// static
-bool AudioDeviceModuleImpl::Destroy(AudioDeviceModuleImpl* adm) {
- if (!adm)
- return false;
- int32_t count = adm->Release();
- if (count != 0) {
- return false;
- } else {
- delete adm;
- return true;
- }
-}
-
-AudioDeviceModuleImpl::AudioDeviceModuleImpl() :
- _ref_count(0) {
-}
-
-AudioDeviceModuleImpl::~AudioDeviceModuleImpl() {
-}
-
-int32_t AudioDeviceModuleImpl::AddRef() {
- return ++_ref_count;
-}
-
-int32_t AudioDeviceModuleImpl::Release() {
- // Avoid self destruction in this mock implementation.
- // Ensures that we can always check the reference counter while alive.
- return --_ref_count;
-}
-
-// ----------------------------------------------------------------------------
-// External transport (Transport) implementations:
-// ----------------------------------------------------------------------------
-
-ExtendedTestTransport::ExtendedTestTransport(VoENetwork* ptr) :
- myNetw(ptr), _thread(NULL), _lock(NULL), _event(NULL), _length(0), _channel(0) {
- const char* threadName = "voe_extended_test_external_thread";
- _lock = CriticalSectionWrapper::CreateCriticalSection();
- _event = EventWrapper::Create();
- _thread = ThreadWrapper::CreateThread(Run, this, kHighPriority, threadName);
- if (_thread) {
- unsigned int id;
- _thread->Start(id);
- }
-}
-
-ExtendedTestTransport::~ExtendedTestTransport() {
- if (_thread) {
- _thread->SetNotAlive();
- _event->Set();
- if (_thread->Stop()) {
- delete _thread;
- _thread = NULL;
- delete _event;
- _event = NULL;
- delete _lock;
- _lock = NULL;
- }
- }
-}
-
-bool ExtendedTestTransport::Run(void* ptr) {
- return static_cast<ExtendedTestTransport*> (ptr)->Process();
-}
-
-bool ExtendedTestTransport::Process() {
- switch (_event->Wait(500)) {
- case kEventSignaled:
- _lock->Enter();
- myNetw->ReceivedRTPPacket(_channel, _packetBuffer, _length);
- _lock->Leave();
- return true;
- case kEventTimeout:
- return true;
- case kEventError:
- break;
- }
- return true;
-}
-
-int ExtendedTestTransport::SendPacket(int channel, const void *data, int len) {
- _lock->Enter();
- if (len < 1612) {
- memcpy(_packetBuffer, (const unsigned char*) data, len);
- _length = len;
- _channel = channel;
- }
- _lock->Leave();
- _event->Set(); // triggers ReceivedRTPPacket() from worker thread
- return len;
-}
-
-int ExtendedTestTransport::SendRTCPPacket(int channel, const void *data, int len) {
- myNetw->ReceivedRTCPPacket(channel, data, len);
- return len;
-}
-
-XTransport::XTransport(VoENetwork* netw, VoEFile* file) :
- _netw(netw), _file(file) {
-}
-
-int XTransport::SendPacket(int channel, const void *data, int len) {
- // loopback
- // _netw->ReceivedRTPPacket(channel, data, len);
-
- return 0;
-}
-
-int XTransport::SendRTCPPacket(int, const void *, int) {
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// VoERTPObserver
-// ----------------------------------------------------------------------------
-
-XRTPObserver::XRTPObserver() :
- _SSRC(0) {
-}
-
-XRTPObserver::~XRTPObserver() {
-}
-
-void XRTPObserver::OnIncomingCSRCChanged(const int /*channel*/, const unsigned int /*CSRC*/,
- const bool /*added*/) {
-}
-
-void XRTPObserver::OnIncomingSSRCChanged(const int /*channel*/, const unsigned int SSRC) {
- // char msg[128];
- // sprintf(msg, "OnIncomingSSRCChanged(channel=%d, SSRC=%lu)\n",
- // channel, SSRC);
- // TEST_LOG(msg);
-
- _SSRC = SSRC; // skip channel dependency for simplicty
-
-}
-
-// ----------------------------------------------------------------------------
-// VoEExtendedTest
-// ----------------------------------------------------------------------------
-
-int VoEExtendedTest::PrepareTest(const char* str) const {
- TEST_LOG("\n\n================================================\n");
- TEST_LOG("\tExtended *%s* Test\n", str);
- TEST_LOG("================================================\n\n");
-
- return 0;
-}
-
-int VoEExtendedTest::TestPassed(const char* str) const {
- TEST_LOG("\n\n------------------------------------------------\n");
- TEST_LOG("\tExtended *%s* test passed!\n", str);
- TEST_LOG("------------------------------------------------\n\n");
-
- return 0;
-}
-
-void VoEExtendedTest::OnPeriodicDeadOrAlive(const int /*channel*/, const bool alive) {
- _alive = alive;
- if (alive) {
- TEST_LOG("=> ALIVE ");
- } else {
- TEST_LOG("=> DEAD ");
- }
- fflush(NULL);
-}
-
-void VoEExtendedTest::CallbackOnError(const int errCode, int) {
- _errCode = errCode;
- TEST_LOG("\n************************\n");
- TEST_LOG(" RUNTIME ERROR: %d \n", errCode);
- TEST_LOG("************************\n");
-}
-
-VoEExtendedTest::VoEExtendedTest(VoETestManager& mgr) :
- _mgr(mgr) {
- for (int i = 0; i < 32; i++) {
- _listening[i] = false;
- _playing[i] = false;
- _sending[i] = false;
- }
-}
-
-VoEExtendedTest::~VoEExtendedTest() {
-}
-
-void VoEExtendedTest::StartMedia(int channel, int rtpPort, bool listen,
- bool playout, bool send) {
- VoEBase* voe_base_ = _mgr.BasePtr();
-
- _listening[channel] = false;
- _playing[channel] = false;
- _sending[channel] = false;
-
- voe_base_->SetLocalReceiver(channel, rtpPort);
- voe_base_->SetSendDestination(channel, rtpPort, "127.0.0.1");
- if (listen) {
- _listening[channel] = true;
- voe_base_->StartReceive(channel);
- }
- if (playout) {
- _playing[channel] = true;
- voe_base_->StartPlayout(channel);
- }
- if (send) {
- _sending[channel] = true;
- voe_base_->StartSend(channel);
- }
-}
-
-void VoEExtendedTest::StopMedia(int channel) {
- VoEBase* voe_base_ = _mgr.BasePtr();
-
- if (_listening[channel]) {
- _listening[channel] = false;
- voe_base_->StopReceive(channel);
- }
- if (_playing[channel]) {
- _playing[channel] = false;
- voe_base_->StopPlayout(channel);
- }
- if (_sending[channel]) {
- _sending[channel] = false;
- voe_base_->StopSend(channel);
- }
-}
-
-void VoEExtendedTest::Play(int channel, unsigned int timeMillisec, bool addFileAsMicrophone,
- bool addTimeMarker) {
- VoEBase* voe_base_ = _mgr.BasePtr();
- VoEFile* file = _mgr.FilePtr();
-
- voe_base_->StartPlayout(channel);
- TEST_LOG("[playing]");
- fflush(NULL);
- if (addFileAsMicrophone) {
- file->StartPlayingFileAsMicrophone(channel, _mgr.AudioFilename(), true, true);
- TEST_LOG("[file as mic]");
- fflush(NULL);
- }
- if (addTimeMarker) {
- float dtSec = (float) ((float) timeMillisec / 1000.0);
- TEST_LOG("[dT=%.1f]", dtSec);
- fflush(NULL); // print sleep time in seconds
- }
- SLEEP(timeMillisec);
- voe_base_->StopPlayout(channel);
- file->StopPlayingFileAsMicrophone(channel);
-}
-
-void VoEExtendedTest::Sleep(unsigned int timeMillisec, bool addMarker) {
- if (addMarker) {
- float dtSec = (float) ((float) timeMillisec / 1000.0);
- TEST_LOG("[dT=%.1f]", dtSec); // print sleep time in seconds
- }
- ::Sleep(timeMillisec);
-}
-
-int VoEExtendedTest::TestBase() {
-#ifndef _WIN32
- // Sleep a bit instead when pause not supported
-#undef PAUSE
-#define PAUSE SLEEP(2000);
-#endif
-
- PrepareTest("Base");
-
- // TODO(qhogpat): make this an actual instance variable. I think the
- // macro black magic will make more sense then. This is named like an
- // instance variable since it is required in order to appease the
- // gods of darkness.
- VoEBase* voe_base_ = _mgr.BasePtr();
- VoENetwork* netw = _mgr.NetworkPtr();
-#ifdef _TEST_RTP_RTCP_
- VoERTP_RTCP* rtp = _mgr.RTP_RTCPPtr();
-#endif
-
- //////////////////////////
- // SetTraceFileName
-
-#ifdef _USE_EXTENDED_TRACE_
- TEST(SetTraceFileName - SetDebugTraceFileName); ANL();
-
- TEST_MUSTPASS(VoiceEngine::SetTraceFile(NULL)); MARK();
- // don't use these files
- TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename(""
- "VoEBase_trace_dont_use.txt"))); MARK();
- // use these instead
- TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename(""
- "VoEBase_trace.txt"))); MARK();
- TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStream |
- kTraceStateInfo |
- kTraceWarning |
- kTraceError |
- kTraceCritical |
- kTraceApiCall |
- kTraceMemory |
- kTraceInfo)); MARK();
-
- ANL(); AOK(); ANL(); ANL();
-#endif
-
- ///////////////////////////////////////
- // RegisterVoiceEngineObserver
- // DeRegisterVoiceEngineObserver
- TEST(SetObserver);
- ANL();
-
- TEST_MUSTPASS(voe_base_->RegisterVoiceEngineObserver(*this));
- MARK();
- SLEEP(100);
- TEST_MUSTPASS(voe_base_->DeRegisterVoiceEngineObserver());
- MARK();
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- /////////////////////
- // GetVersion
- TEST(GetVersion);
- ANL();
-
- char version[1024];
- // audio device module and AudioProcessing fail to getversion when they
- // are not initiliazed
- TEST_MUSTPASS(voe_base_->GetVersion(version));
- MARK();
- TEST_LOG("\n-----\n%s\n-----\n", version);
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- ///////////////
- // Init
- TEST(Init);
- ANL();
-
- TEST_MUSTPASS(voe_base_->Init());
- MARK();
- TEST_MUSTPASS(voe_base_->Terminate());
-
- TEST_MUSTPASS(voe_base_->Init());
- MARK();
- // ensure that no new memory is allocated at the second call (check
- // trace file)
- TEST_MUSTPASS(voe_base_->Init());
- MARK();
- TEST_MUSTPASS(voe_base_->Terminate());
-#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
- // verify AEC recording
- TEST_MUSTPASS(voe_base_->Init());
- MARK(); // verify output dat-files
- TEST_MUSTPASS(voe_base_->Terminate());
-#endif
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- ////////////////////
- // Terminate
- TEST(Terminate);
- ANL();
- TEST_MUSTPASS(voe_base_->Terminate());
- MARK(); // should be ignored
- TEST_MUSTPASS(voe_base_->Init());
- TEST_MUSTPASS(voe_base_->Terminate());
- MARK(); // should terminate
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- // ------------------------------------------------------------------------
- // >> Init(AudioDeviceModule)
- //
- // Note that our mock implementation of the ADM also mocks the
- // reference counting part. This approach enables us to keep track
- // of the internal reference counter without checking return values
- // from the ADM and we also avoid the built-in self destruction.
- //
- // TODO(henrika): this test does not verify that external ADM methods
- // are called by the VoiceEngine once registered. We could extend
- // the mock implementation and add counters for each ADM API to ensure
- // that they are called in the correct sequence and the correct number
- // of times.
- TEST_LOG("\nTesting: Init in combination with an external ADM\n");
-
- // Create the ADM and call AddRef within the factory method.
- AudioDeviceModuleImpl* xADM = AudioDeviceModuleImpl::Create();
- ASSERT_FALSE(xADM == NULL);
- ASSERT_TRUE(xADM->ReferenceCounter() == 1);
-
- // Verify default usage case for external ADM.
- TEST_MUSTPASS(voe_base_->Init(xADM));MARK();
- ASSERT_TRUE(xADM->ReferenceCounter() == 2);
- TEST_MUSTPASS(voe_base_->Terminate());
- ASSERT_TRUE(xADM->ReferenceCounter() == 1);
-
- // Our reference-count implementation does not self destruct.
- // We do it manually here instead by calling Release followed by delete.
- ASSERT_TRUE(AudioDeviceModuleImpl::Destroy(xADM));
- ANL();
- AOK();ANL();
-
- // >> end of Init(AudioDeviceModule)
- // ------------------------------------------------------------------------
-
- ///////////////////////////
- // MaxNumOfChannels
- TEST(MaxNumOfChannels);
- ANL();
- TEST_MUSTPASS(voe_base_->MaxNumOfChannels() < 0);
- MARK();
- ANL();
- AOK();
- ANL();
- ANL();
-
- ////////////////////////
- // CreateChannel
- // DeleteChannel
-
- int i;
- int channel;
- int nChannels(voe_base_->MaxNumOfChannels());
-
- TEST(CreateChannel);
- ANL();
- TEST(DeleteChannel);
- ANL();
-
- TEST_MUSTPASS(voe_base_->Init());
-
- channel = voe_base_->CreateChannel();
- MARK();
- TEST_MUSTPASS(channel != 0);
- channel = voe_base_->CreateChannel();
- MARK();
- TEST_MUSTPASS(channel != 1);
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- MARK();
- TEST_MUSTPASS(voe_base_->DeleteChannel(1));
- MARK();
-
- // create and delete one channel many times
- for (i = 0; i < 10; i++) {
- channel = voe_base_->CreateChannel();
- MARK();
- TEST_MUSTPASS(channel != 0); // should be 0 each time
- TEST_MUSTPASS(voe_base_->DeleteChannel(channel));
- MARK();
- }
- // create max number of channels
- for (i = 0; i < nChannels; i++) {
- channel = voe_base_->CreateChannel();
- MARK();
- TEST_MUSTPASS(channel != i);
- }
- channel = voe_base_->CreateChannel();
- MARK(); // should fail since no more channels can now be created
- TEST_MUSTPASS(channel != -1);
-
- int aChannel = (((nChannels - 17) > 0) ? (nChannels - 17) : 0);
- TEST_MUSTPASS(voe_base_->DeleteChannel(aChannel));
- MARK();
- channel = voe_base_->CreateChannel();
- MARK(); // should reuse channel
- TEST_MUSTPASS(channel != aChannel);
-
- // delete all created channels
- for (i = 0; i < nChannels; i++) {
- TEST_MUSTPASS(voe_base_->DeleteChannel(i));
- MARK();
- }
-
- // try to delete a non-existing channel
- TEST_MUSTPASS(-1 != voe_base_->DeleteChannel(aChannel));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- // ------------------------------------------------------------------------
- // >> SetLocalReceiver
- //
- // State: VE not initialized, no existing channels
- TEST_MUSTPASS(voe_base_->Init());
-
- int ch;
-
- TEST(SetLocalReceiver);
- ANL();
-
- // no channel created yet => should fail
- TEST_MUSTPASS(!voe_base_->SetLocalReceiver(0, 100));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- ch = voe_base_->CreateChannel();
-
-#ifdef MAC_IPHONE
- printf("\nNOTE: Local IP must be set in source code (line %d) \n",
- __LINE__ + 1);
- char* localIp = "127.0.0.1";
-#else
- char localIp[64] = { 0 };
- TEST_MUSTPASS(netw->GetLocalIP(localIp));
- MARK();
- // NOTE: This API is supported on Win, Mac and Linux and may fail or not
- // return local IP for other platforms.
-#endif
-
- // trivial invalid function calls
- TEST_MUSTPASS(!voe_base_->SetLocalReceiver(ch+1, 12345));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
- TEST_MUSTPASS(!voe_base_->SetLocalReceiver(ch, -1));
- MARK();
- TEST_ERROR(VE_INVALID_PORT_NMBR);
-
- // check conflict with ongoing receiving
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
- MARK();
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- TEST_MUSTPASS(!voe_base_->SetLocalReceiver(ch, 12345));
- MARK();
- TEST_ERROR(VE_ALREADY_LISTENING);
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
-
- // check conflict with ongoing transmission
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- TEST_MUSTPASS(!voe_base_->SetLocalReceiver(ch, 12345));
- MARK();
- TEST_ERROR(VE_ALREADY_SENDING);
- TEST_MUSTPASS(voe_base_->StopSend(ch));
-
- // valid function calls
- // Need to sleep between, otherwise it may fail for unknown reason
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
- MARK();
- SLEEP(100);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
- MARK();
- SLEEP(100);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, kVoEDefault, localIp));
- MARK();
- SLEEP(100);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, kVoEDefault, NULL,
- "230.1.2.3"));
- MARK();
- SLEEP(100);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, kVoEDefault, localIp,
- "230.1.2.3"));
- MARK();
- SLEEP(100);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, 5555, NULL));
- MARK();
- SLEEP(100);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
- MARK();
- SLEEP(100);
-
- // STATE: no media but sockets exists and are binded to 12345 and 12346
- // respectively
-
- // Add some dynamic tests as well:
-
- // ensure that last setting is used (cancels old settings)
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
- MARK();
- SLEEP(100);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 44444));
- MARK();
- SLEEP(100);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 54321));
- MARK();
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 54321, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- Play(ch, 1000, true, true);
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(ch));
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of SetLocalReceiver
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> GetLocalReceiver
- //
- // State: VE initialized, no existing channels
- TEST(GetLocalReceiver);
- ANL();
-
- int port;
- char ipaddr[64];
- int RTCPport;
-
- ch = voe_base_->CreateChannel();
-
- // verify non-configured (blank) local receiver
- TEST_MUSTPASS(voe_base_->GetLocalReceiver(ch, port, RTCPport, ipaddr));
- MARK();
- TEST_MUSTPASS(port != 0);
- TEST_MUSTPASS(RTCPport != 0);
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
-
- // check some trivial set/get combinations
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345))
- TEST_MUSTPASS(voe_base_->GetLocalReceiver(ch, port, RTCPport, ipaddr));
- MARK();
- TEST_MUSTPASS(port != 12345);
- TEST_MUSTPASS(RTCPport != 12346);
- TEST_MUSTPASS(strcmp(ipaddr, "0.0.0.0") != 0); // now binded to "any" IP
-
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, 55555))
- TEST_MUSTPASS(voe_base_->GetLocalReceiver(ch, port, RTCPport, ipaddr));
- MARK();
- TEST_MUSTPASS(port != 12345);
- TEST_MUSTPASS(RTCPport != 55555);
- TEST_MUSTPASS(strcmp(ipaddr, "0.0.0.0") != 0);
-
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, kVoEDefault, localIp))
- TEST_MUSTPASS(voe_base_->GetLocalReceiver(ch, port, RTCPport, ipaddr));
- MARK();
- TEST_MUSTPASS(port != 12345);
- TEST_MUSTPASS(RTCPport != 12346);
- TEST_MUSTPASS(strcmp(ipaddr, localIp) != 0);
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(ch));
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of GetLocalReceiver
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> SetSendDestination
- //
- // State: VE initialized, no existing channels
- TEST(SetSendDestination);
- ANL();
-
- // call without existing channel
- TEST_MUSTPASS(!voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- ch = voe_base_->CreateChannel();
-
- // trivial fail tests
- TEST_MUSTPASS(!voe_base_->SetSendDestination(ch, 65536, "127.0.0.1"));
- MARK();
- TEST_ERROR(VE_INVALID_PORT_NMBR); // invalid RTP port
- TEST_MUSTPASS(!voe_base_->SetSendDestination(ch, 12345, "127.0.0.1", 65536));
- MARK();
- TEST_ERROR(VE_INVALID_PORT_NMBR); // invalid source port
- TEST_MUSTPASS(!voe_base_->SetSendDestination(ch, 12345, "127.0.0.1", kVoEDefault,
- 65536));
- MARK();
- TEST_ERROR(VE_INVALID_PORT_NMBR); // invalid RTCP port
- TEST_MUSTPASS(!voe_base_->SetSendDestination(ch, 12345, "127.0.0.300"));
- MARK();
- TEST_ERROR(VE_INVALID_IP_ADDRESS); // invalid IP address
-
- // sockets must be created first to support multi-cast (not required
- // otherwise)
- TEST_MUSTPASS(!voe_base_->SetSendDestination(ch, 55555, "230.0.0.1"));
- MARK();
- TEST_ERROR(VE_SOCKET_ERROR);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 55555)); // create sockets
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 55555, "230.0.0.1"));
- MARK(); // should work now
-
- voe_base_->DeleteChannel(0);
- ch = voe_base_->CreateChannel();
-
- // STATE: one channel created, no sockets exist
-
- // valid function calls
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1"));
- MARK();
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1", 44444));
- MARK();
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1", kVoEDefault,
- 55555));
- MARK();
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1", 44444,
- 55555));
- MARK();
-
- voe_base_->DeleteChannel(0);
- ch = voe_base_->CreateChannel();
-
- // create receive sockets first and then an extra pair of send sockets
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 44444));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 44444, "127.0.0.1", 11111));
- MARK(); // binds to 11111
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(ch));
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of SetSendDestination
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> GetSendDestination
- //
- // State: VE initialized, no existing channels
- TEST(GetSendDestination);
- ANL();
-
- int sourcePort;
-
- ch = voe_base_->CreateChannel();
-
- // verify non-configured (blank) local receiver
- TEST_MUSTPASS(voe_base_->GetSendDestination(ch, port, ipaddr, sourcePort,
- RTCPport));
- MARK();
- TEST_MUSTPASS(port != 0);
- TEST_MUSTPASS(sourcePort != 0);
- TEST_MUSTPASS(RTCPport != 0);
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
-
- // check some trivial set/get combinations
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 44444, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->GetSendDestination(ch, port, ipaddr, sourcePort,
- RTCPport));
- MARK();
- TEST_MUSTPASS(port != 44444);
- TEST_MUSTPASS(sourcePort != 0); // should be 0 since no local receiver has
- // NOT been defined yet
- TEST_MUSTPASS(RTCPport != 44445);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 55555));
- TEST_MUSTPASS(voe_base_->GetSendDestination(ch, port, ipaddr, sourcePort,
- RTCPport));
- MARK();
- TEST_MUSTPASS(port != 44444);
- TEST_MUSTPASS(sourcePort != 55555); // should be equal to local port
- TEST_MUSTPASS(RTCPport != 44445);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- voe_base_->DeleteChannel(0);
- ch = voe_base_->CreateChannel();
-
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 44444, "127.0.0.1"));
- // NULL as IP-address input should work as well
- TEST_MUSTPASS(voe_base_->GetSendDestination(ch, port, NULL, sourcePort,
- RTCPport));
- MARK();
- TEST_MUSTPASS(port != 44444);
- TEST_MUSTPASS(sourcePort != 0);
- TEST_MUSTPASS(RTCPport != 44445);
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(ch));
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of GetLocalReceiver
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> StartReceive
- // >> StopReceive
- //
- // State: VE initialized, no existing channels
- TEST(StartReceive);
- ANL();
- TEST(StopReceive);
- ANL();
-
- // call without existing channel
- TEST_MUSTPASS(!voe_base_->StartReceive(0));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
- TEST_MUSTPASS(!voe_base_->StopReceive(0));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- ch = voe_base_->CreateChannel();
-
- // sockets must be created first
- TEST_MUSTPASS(!voe_base_->StartReceive(0));
- MARK();
- TEST_ERROR(VE_SOCKETS_NOT_INITED);
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 55555));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- MARK(); // should work this time
-
- // enable again (should work)
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- MARK();
-
- // Stop/Start (should work)
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- MARK();
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- MARK();
-
- // Verify in loopback
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 55555, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- Play(ch, 1000, true, true);
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- MARK();
-
- voe_base_->DeleteChannel(0);
- ch = voe_base_->CreateChannel();
-
- // Ensure that it is OK to add delay between SetLocalReceiver and StarListen
- TEST_LOG("\nspeak after 2 seconds and ensure that no delay is added:\n");
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 55555));
-
- Sleep(2000, true); // adding emulated delay here
-
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 55555, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- Play(ch, 2000, true, true);
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(ch));
- ANL();
-
- // Multi-channel tests
-
- for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
- ch = voe_base_->CreateChannel();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 11111+2*i));
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- MARK();
- }
- for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
- TEST_MUSTPASS(voe_base_->StopReceive(i));
- MARK();
- voe_base_->DeleteChannel(i);
- }
- for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
- ch = voe_base_->CreateChannel();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 11111+2*i));
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- MARK();
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
- MARK();
- voe_base_->DeleteChannel(ch);
- }
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of StartReceive/StopReceive
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> StartPlayout
- // >> StopPlayout
- //
- // State: VE initialized, no existing channels
- TEST(StartPlayout);
- ANL();
- TEST(StopPlayout);
- ANL();
-
- // call without existing channel
- TEST_MUSTPASS(!voe_base_->StartPlayout(0));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
- TEST_MUSTPASS(!voe_base_->StopPlayout(0));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- ch = voe_base_->CreateChannel();
-
- TEST_MUSTPASS(voe_base_->StartPlayout(ch));
- MARK();
- TEST_MUSTPASS(voe_base_->StartPlayout(ch));
- MARK();
- TEST_MUSTPASS(voe_base_->StopPlayout(ch));
- MARK();
- TEST_MUSTPASS(voe_base_->StopPlayout(ch));
- MARK();
-
- voe_base_->DeleteChannel(ch);
-
- // Multi-channel tests
- const int MaxNumberOfPlayingChannels(kVoiceEngineMaxNumOfActiveChannels);
-
- for (i = 0; i < MaxNumberOfPlayingChannels; i++) {
- ch = voe_base_->CreateChannel();
- TEST_MUSTPASS(voe_base_->StartPlayout(ch));
- MARK();
- }
- for (i = 0; i < MaxNumberOfPlayingChannels; i++) {
- TEST_MUSTPASS(voe_base_->StopPlayout(i));
- MARK();
- voe_base_->DeleteChannel(i);
- }
- for (i = 0; i < MaxNumberOfPlayingChannels; i++) {
- ch = voe_base_->CreateChannel();
- TEST_MUSTPASS(voe_base_->StartPlayout(ch));
- MARK();
- TEST_MUSTPASS(voe_base_->StopPlayout(ch));
- MARK();
- voe_base_->DeleteChannel(ch);
- }
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of StartPlayout/StopPlayout
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> StartSend
- // >> StopSend
- //
- // State: VE initialized, no existing channels
- TEST(StartSend);
- ANL();
- TEST(StopSend);
- ANL();
-
- // call without existing channel
- TEST_MUSTPASS(!voe_base_->StartSend(0));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
- TEST_MUSTPASS(!voe_base_->StopSend(0));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- ch = voe_base_->CreateChannel();
-
- // call without initialized destination
- TEST_MUSTPASS(!voe_base_->StartSend(ch));
- MARK();
- TEST_ERROR(VE_DESTINATION_NOT_INITED);
-
- // initialize destination and try again (should work even without existing
- // sockets)
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- MARK();
- SLEEP(100);
-
- // STATE: sockets should now have been created automatically at the first
- // transmitted packet should be binded to 33333 and "0.0.0.0"
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- MARK();
-
- voe_base_->DeleteChannel(ch);
- ch = voe_base_->CreateChannel();
-
- // try loopback with unique send sockets (closed when channel is deleted or
- // new source is set)
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 33333));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333, "127.0.0.1", 44444));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- MARK();
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- Play(ch, 2000, true, true);
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- MARK();
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
-
- voe_base_->DeleteChannel(ch);
- ANL();
-
- // Multi-channel tests
- for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
- ch = voe_base_->CreateChannel();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 33333 + 2*i));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33333 + 2*i, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- MARK();
- }
- for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
- TEST_MUSTPASS(voe_base_->StopSend(i));
- MARK();
- voe_base_->DeleteChannel(i);
- }
- for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
- ch = voe_base_->CreateChannel();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 45633 + 2*i));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 45633 + 2*i, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- MARK();
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- MARK();
- voe_base_->DeleteChannel(ch);
- }
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of StartSend/StopSend
- // ------------------------------------------------------------------------
-
- //////////////////////////////
- // SetNetEQPlayoutMode
- // GetNetEQPlayoutMode
- TEST(SetNetEQPlayoutMode);
- ANL();
- TEST(GetNetEQPlayoutMode);
- ANL();
-
- NetEqModes mode;
-
- ch = voe_base_->CreateChannel();
-
- // invalid function calls (should fail)
- TEST_MUSTPASS(!voe_base_->GetNetEQPlayoutMode(ch+1, mode));
- MARK();
- TEST_MUSTPASS(!voe_base_->SetNetEQPlayoutMode(ch+1, kNetEqDefault));
- MARK();
-
- // verify default mode (should be kNetEqDefault)
- TEST_MUSTPASS(voe_base_->GetNetEQPlayoutMode(ch, mode));
- MARK();
- TEST_MUSTPASS(mode != kNetEqDefault);
- TEST_MUSTPASS(voe_base_->SetNetEQPlayoutMode(ch, kNetEqStreaming));
- MARK();
- voe_base_->DeleteChannel(ch);
-
- // ensure that default mode is set as soon as new channel is created
- ch = voe_base_->CreateChannel();
- TEST_MUSTPASS(voe_base_->GetNetEQPlayoutMode(ch, mode));
- MARK();
- TEST_MUSTPASS(mode != kNetEqDefault);
- voe_base_->DeleteChannel(ch);
-
- // verify Set/Get for all supported modes and max number of channels
- for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
- ch = voe_base_->CreateChannel();
-
- // verify Set/Get for all supported modes
- TEST_MUSTPASS(voe_base_->SetNetEQPlayoutMode(i, kNetEqDefault));
- MARK();
- TEST_MUSTPASS(voe_base_->GetNetEQPlayoutMode(i, mode));
- MARK();
- TEST_MUSTPASS(mode != kNetEqDefault);
- TEST_MUSTPASS(voe_base_->SetNetEQPlayoutMode(i, kNetEqStreaming));
- MARK();
- TEST_MUSTPASS(voe_base_->GetNetEQPlayoutMode(i, mode));
- MARK();
- TEST_MUSTPASS(mode != kNetEqStreaming);
- TEST_MUSTPASS(voe_base_->SetNetEQPlayoutMode(i, kNetEqFax));
- MARK();
- TEST_MUSTPASS(voe_base_->GetNetEQPlayoutMode(i, mode));
- MARK();
- TEST_MUSTPASS(mode != kNetEqFax);
- SLEEP(50);
- }
-
- for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
- voe_base_->DeleteChannel(i);
- }
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- //////////////////////////////
- // SetNetEQBGNMode
- // GetNetEQBGNMode
- TEST(SetNetEQBGNMode);
- ANL();
- TEST(GetNetEQBGNMode);
- ANL();
-
- NetEqBgnModes bgnMode;
-
- ch = voe_base_->CreateChannel();
-
- // invalid function calls (should fail)
- TEST_MUSTPASS(!voe_base_->GetNetEQBGNMode(ch+1, bgnMode));
- MARK();
- TEST_MUSTPASS(!voe_base_->SetNetEQBGNMode(ch+1, kBgnOn));
- MARK();
-
- // verify default mode (should be kBgnOn)
- TEST_MUSTPASS(voe_base_->GetNetEQBGNMode(ch, bgnMode));
- MARK();
- TEST_MUSTPASS(bgnMode != kBgnOn);
- voe_base_->DeleteChannel(ch);
-
- // ensure that default mode is set as soon as new channel is created
- ch = voe_base_->CreateChannel();
- TEST_MUSTPASS(voe_base_->GetNetEQBGNMode(ch, bgnMode));
- MARK();
- TEST_MUSTPASS(bgnMode != kBgnOn);
- voe_base_->DeleteChannel(ch);
-
- // verify Set/Get for all supported modes and max number of channels
- for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
- ch = voe_base_->CreateChannel();
-
- // verify Set/Get for all supported modes
- TEST_MUSTPASS(voe_base_->SetNetEQBGNMode(i, kBgnOn));
- MARK();
- TEST_MUSTPASS(voe_base_->GetNetEQBGNMode(i, bgnMode));
- MARK();
- TEST_MUSTPASS(bgnMode != kBgnOn);
- TEST_MUSTPASS(voe_base_->SetNetEQBGNMode(i, kBgnFade));
- MARK();
- TEST_MUSTPASS(voe_base_->GetNetEQBGNMode(i, bgnMode));
- MARK();
- TEST_MUSTPASS(bgnMode != kBgnFade);
- TEST_MUSTPASS(voe_base_->SetNetEQBGNMode(i, kBgnOff));
- MARK();
- TEST_MUSTPASS(voe_base_->GetNetEQBGNMode(i, bgnMode));
- MARK();
- TEST_MUSTPASS(bgnMode != kBgnOff);
- SLEEP(50);
- }
-
- for (i = 0; i < voe_base_->MaxNumOfChannels(); i++) {
- voe_base_->DeleteChannel(i);
- }
-
- // Verify real-time performance for all playout modes in full duplex
-
- ch = voe_base_->CreateChannel();
-
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch , 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1"));
-
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- TEST_MUSTPASS(voe_base_->StartPlayout(ch));
-
- TEST_MUSTPASS(voe_base_->SetNetEQPlayoutMode(ch, kNetEqDefault));
- MARK();
- TEST_LOG("\nenjoy full duplex using kNetEqDefault playout mode...\n");
- PAUSE
-
- TEST_MUSTPASS(voe_base_->SetNetEQPlayoutMode(ch, kNetEqStreaming));
- MARK();
- TEST_LOG("\nenjoy full duplex using kNetEqStreaming playout mode...\n");
- PAUSE
-
- TEST_MUSTPASS(voe_base_->SetNetEQPlayoutMode(ch, kNetEqFax));
- MARK();
- TEST_LOG("\nenjoy full duplex using kNetEqFax playout mode...\n");
- PAUSE
-
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- TEST_MUSTPASS(voe_base_->StopPlayout(ch));
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
-
- voe_base_->DeleteChannel(ch);
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- /////////////////////
- // Full duplex tests
-
- ch = voe_base_->CreateChannel(); // We must delete this channel first to be able
- // to reuse port 12345
-
- // start with default case, also test non-default RTCP port
-#ifdef _TEST_RTP_RTCP_
- TEST_MUSTPASS(rtp->SetRTCP_CNAME(ch, "Johnny"));
-#endif
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345, 12349));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1", kVoEDefault,
- 12349));
-
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- TEST_MUSTPASS(voe_base_->StartPlayout(ch));
-
- TEST_LOG("full duplex is now activated (1)\n");
- TEST_LOG("waiting for RTCP packet...\n");
-
- SLEEP(7000); // Make sure we get RTCP packet
- PAUSE;
-
- // Verify that we got RTCP packet from correct source port
-#ifdef _TEST_RTP_RTCP_
- char tmpStr[64] = { 0 };
- TEST_MUSTPASS(rtp->GetRemoteRTCP_CNAME(ch, tmpStr));
- TEST_MUSTPASS(_stricmp("Johnny", tmpStr));
-#endif
- int rtpPort(0), rtcpPort(0);
- char ipAddr[64] = { 0 };
- TEST_MUSTPASS(netw->GetSourceInfo(ch, rtpPort, rtcpPort, ipAddr));
- TEST_MUSTPASS(12349 != rtcpPort);
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- TEST_MUSTPASS(voe_base_->StopPlayout(ch));
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
-
- // Call StartSend before StartReceive
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1"));
-
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- TEST_MUSTPASS(voe_base_->StartPlayout(ch));
-
- TEST_LOG("\nfull duplex is now activated (2)\n");
-
- PAUSE
-
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- TEST_MUSTPASS(voe_base_->StopPlayout(ch));
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
-
- // Try again using same ports
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1"));
-
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- TEST_MUSTPASS(voe_base_->StartPlayout(ch));
-
- TEST_LOG("\nfull duplex is now activated (3)\n");
- TEST_LOG("waiting for RTCP packet...\n");
-
- SLEEP(7000); // Make sure we get RTCP packet
- PAUSE
-
- // Verify correct RTCP source port
- TEST_MUSTPASS(netw->GetSourceInfo(ch, rtpPort, rtcpPort, ipAddr));
- TEST_MUSTPASS(12345+1 != rtcpPort);
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- TEST_MUSTPASS(voe_base_->StopPlayout(ch));
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
-
- voe_base_->DeleteChannel(ch);
- ch = voe_base_->CreateChannel();
-
- // Try with extra send socket
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch , 22222));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 22222, "127.0.0.1", 11111));
-
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- TEST_MUSTPASS(voe_base_->StartPlayout(ch));
-
- TEST_LOG("\nfull duplex is now activated (4)\n");
-
- PAUSE
-
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- TEST_MUSTPASS(voe_base_->StopPlayout(ch));
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
-
- // repeat default case starting with a fresh channel
-
- voe_base_->DeleteChannel(ch);
- ch = voe_base_->CreateChannel();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch , 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1"));
-
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- TEST_MUSTPASS(voe_base_->StartPlayout(ch));
-
- TEST_LOG("\nfull duplex is now activated (5)\n");
-
- PAUSE
-
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- TEST_MUSTPASS(voe_base_->StopPlayout(ch));
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
-
- // restart call again
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 12345));
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- TEST_MUSTPASS(voe_base_->StartPlayout(ch));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
-
- TEST_LOG("\nfull duplex is now activated (6)\n");
-
- PAUSE
-
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- TEST_MUSTPASS(voe_base_->StopPlayout(ch));
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
-
- // force sending from new socket
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch , 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 12345, "127.0.0.1", 12350,
- 12359));
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- TEST_MUSTPASS(voe_base_->StartPlayout(ch));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
- TEST_LOG("\nfull duplex is now activated (7)\n");
-
- PAUSE
-
- // Test getting send settings
- TEST_MUSTPASS(voe_base_->GetSendDestination(ch, rtpPort, ipAddr, sourcePort,
- rtcpPort));
- TEST_MUSTPASS(12345 != rtpPort);
- TEST_MUSTPASS(_stricmp("127.0.0.1", ipAddr));
- TEST_MUSTPASS(12350 != sourcePort);
- TEST_MUSTPASS(12359 != rtcpPort);
-
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- TEST_MUSTPASS(voe_base_->StopPlayout(ch));
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
-
- // new channel and new port
- ch = voe_base_->CreateChannel();
-
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch , 33221));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33221, "127.0.0.1"));
-
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- TEST_MUSTPASS(voe_base_->StartPlayout(ch));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
-
- TEST_LOG("\nfull duplex is now activated (8)\n");
-
- PAUSE
-
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- TEST_MUSTPASS(voe_base_->StopPlayout(ch));
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
-
- voe_base_->DeleteChannel(ch);
- ch = voe_base_->CreateChannel();
-
-#ifndef MAC_IPHONE
- // bind to local IP and try again
- strcpy(localIp, "127.0.0.1");
-#else
- localIp = "127.0.0.1";
-#endif
-
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch, 33221, 12349, localIp));
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch, 33221, localIp));
-
- TEST_MUSTPASS(voe_base_->StartReceive(ch));
- TEST_MUSTPASS(voe_base_->StartPlayout(ch));
- TEST_MUSTPASS(voe_base_->StartSend(ch));
-
- TEST_LOG("\nfull duplex is now activated (9)\n");
-
- PAUSE
-
- TEST_MUSTPASS(voe_base_->GetLocalReceiver(ch, rtpPort, rtcpPort, ipAddr));
- TEST_MUSTPASS(33221 != rtpPort);
- TEST_MUSTPASS(_stricmp(localIp, ipAddr));
- TEST_MUSTPASS(12349 != rtcpPort);
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- //////////////////////
- // Trace filter tests
-
-#ifdef _USE_EXTENDED_TRACE_
- TEST(SetTraceFilter); ANL();
-
- TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename(""
- "VoEBase_trace_filter.txt"))); MARK();
- SLEEP(100);
-
- // Test a few different filters, verify in trace file
- // Each SetTraceFilter calls should be seen once, no more, no less
- TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceNone)); MARK();
- SLEEP(300);
- // API call and info should NOT be seen in log
- TEST_MUSTPASS(voe_base_->SetOnHoldStatus(0, true)); MARK();
- // API call and error should NOT be seen in log
- TEST_MUSTPASS(!voe_base_->SetOnHoldStatus(999, true)); MARK();
-
- TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceApiCall |
- kTraceCritical |
- kTraceError |
- kTraceWarning)); MARK();
- SLEEP(300);
- // API call should and info should NOT be seen in log
- TEST_MUSTPASS(voe_base_->SetOnHoldStatus(0, false)); MARK();
- // API call and error should be seen in log
- TEST_MUSTPASS(!voe_base_->SetOnHoldStatus(999, true)); MARK();
-
- TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceApiCall | kTraceInfo));
- MARK();
- SLEEP(300);
- // API call and info should be seen in log
- TEST_MUSTPASS(voe_base_->SetOnHoldStatus(0, true)); MARK();
- // API call should and error should NOT be seen in log
- TEST_MUSTPASS(!voe_base_->SetOnHoldStatus(999, true)); MARK();
-
- // Back to default
- TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceAll)); MARK();
- SLEEP(300);
-
- AOK(); ANL();
-#endif
-
- // ------------------------------------------------------------------------
- // >> Multiple instance testing
- //
- // We should support 8 instances simultaneously
- // and at least one should be able to have a call running
-
- // One instance is already created
- VoiceEngine* instVE[7];
- VoEBase* baseVE[7];
- for (int instNum = 0; instNum < 7; instNum++) {
- instVE[instNum] = VoiceEngine::Create();
- baseVE[instNum] = VoEBase::GetInterface(instVE[instNum]);
- TEST_MUSTPASS(baseVE[instNum]->Init());
- TEST_MUSTPASS(baseVE[instNum]->CreateChannel());
- }
-
- TEST_LOG("Created 7 more instances of VE, make sure audio is ok...\n\n");
- PAUSE
-
- for (int instNum = 0; instNum < 7; instNum++) {
- TEST_MUSTPASS(baseVE[instNum]->DeleteChannel(0));
- TEST_MUSTPASS(baseVE[instNum]->Terminate());
- TEST_MUSTPASS(baseVE[instNum]->Release());
- VoiceEngine::Delete(instVE[instNum]);
- }
-
- AOK();
- ANL();
-
- //////////////
- // Close down
- TEST_MUSTPASS(voe_base_->StopSend(ch));
- TEST_MUSTPASS(voe_base_->StopPlayout(ch));
- TEST_MUSTPASS(voe_base_->StopReceive(ch));
- TEST_MUSTPASS(voe_base_->DeleteChannel(ch));
-
- voe_base_->DeleteChannel(0);
- TEST_MUSTPASS(voe_base_->Terminate());
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// VoEExtendedTest::TestCallReport
-// ----------------------------------------------------------------------------
-
-int VoEExtendedTest::TestCallReport() {
- // Get required sub-API pointers
- VoEBase* voe_base_ = _mgr.BasePtr();
- VoECallReport* report = _mgr.CallReportPtr();
- VoEFile* file = _mgr.FilePtr();
- VoEAudioProcessing* apm = _mgr.APMPtr();
- VoENetwork* netw = _mgr.NetworkPtr();
-
- PrepareTest("CallReport");
-
- // check if this interface is supported
- if (!report) {
- TEST_LOG("VoECallReport is not supported!");
- return -1;
- }
-
-#ifdef _USE_EXTENDED_TRACE_
- TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename("VoECallReport_trace.txt")));
- TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
- kTraceStateInfo |
- kTraceWarning |
- kTraceError |
- kTraceCritical |
- kTraceApiCall |
- kTraceMemory |
- kTraceInfo));
-#endif
-
- TEST_MUSTPASS(voe_base_->Init());
- TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
- true, true));
-
- ///////////////////////////
- // Actual test starts here
- TEST(ResetCallReportStatistics);
- ANL();
- TEST_MUSTPASS(!report->ResetCallReportStatistics(-2));
- MARK(); // not OK
- TEST_MUSTPASS(!report->ResetCallReportStatistics(1));
- MARK(); // not OK
- TEST_MUSTPASS(report->ResetCallReportStatistics(0));
- MARK(); // OK
- TEST_MUSTPASS(report->ResetCallReportStatistics(-1));
- MARK(); // OK
- AOK();
- ANL();
-
- bool enabled = false;
- EchoStatistics echo;
- TEST(GetEchoMetricSummary);
- ANL();
- TEST_MUSTPASS(apm->GetEcMetricsStatus(enabled));
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(apm->SetEcMetricsStatus(true));
- TEST_MUSTPASS(report->GetEchoMetricSummary(echo)); // all outputs will be
- // -100 in loopback (skip further tests)
- AOK();
- ANL();
-
- // TODO(xians): investigate the cause of test failure before enabling.
- /*
- StatVal delays;
- TEST(GetRoundTripTimeSummary);
- ANL();
- // All values should be >=0 since RTCP is now on
- TEST_MUSTPASS(report->GetRoundTripTimeSummary(0, delays));
- MARK();
- TEST_MUSTPASS(delays.min == -1);
- TEST_MUSTPASS(delays.max == -1);
- TEST_MUSTPASS(delays.average == -1);
- rtp_rtcp->SetRTCPStatus(0, false);
- // All values should be -1 since RTCP is off
- TEST_MUSTPASS(report->GetRoundTripTimeSummary(0, delays));
- MARK();
- TEST_MUSTPASS(delays.min != -1);
- TEST_MUSTPASS(delays.max != -1);
- TEST_MUSTPASS(delays.average != -1);
- rtp_rtcp->SetRTCPStatus(0, true);
- AOK();
- ANL();
- */
-
- int nDead = 0;
- int nAlive = 0;
- TEST(GetDeadOrAliveSummary);
- ANL();
- // All results should be -1 since dead-or-alive is not active
- TEST_MUSTPASS(report->GetDeadOrAliveSummary(0, nDead, nAlive) != -1);
- MARK();
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 1));
- SLEEP(2000);
- // All results should be >= 0 since dead-or-alive is active
- TEST_MUSTPASS(report->GetDeadOrAliveSummary(0, nDead, nAlive));
- MARK();
- TEST_MUSTPASS(nDead == -1);
- TEST_MUSTPASS(nAlive == -1)
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, false));
- AOK();
- ANL();
-
- TEST(WriteReportToFile);
- ANL();
-
- // Greek and Coptic (see http://www.utf8-chartable.de/unicode-utf8-table.pl)
- char fileNameUTF8[64];
-
- fileNameUTF8[0] = (char) 0xce;
- fileNameUTF8[1] = (char) 0xba;
- fileNameUTF8[2] = (char) 0xce;
- fileNameUTF8[3] = (char) 0xbb;
- fileNameUTF8[4] = (char) 0xce;
- fileNameUTF8[5] = (char) 0xbd;
- fileNameUTF8[6] = (char) 0xce;
- fileNameUTF8[7] = (char) 0xbe;
- fileNameUTF8[8] = '.';
- fileNameUTF8[9] = 't';
- fileNameUTF8[10] = 'x';
- fileNameUTF8[11] = 't';
- fileNameUTF8[12] = 0;
-
- TEST_MUSTPASS(!report->WriteReportToFile(NULL));
- MARK();
- TEST_MUSTPASS(report->WriteReportToFile("call_report.txt"));
- MARK();
- TEST_MUSTPASS(report->WriteReportToFile(fileNameUTF8));
- MARK(); // should work with UTF-8 as well (κλνξ.txt)
- AOK();
- ANL();
-
- TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->Terminate());
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// VoEExtendedTest::TestCodec
-// ----------------------------------------------------------------------------
-
-int VoEExtendedTest::TestCodec() {
- PrepareTest("Codec");
-
- VoEBase* voe_base_ = _mgr.BasePtr();
- VoECodec* codec = _mgr.CodecPtr();
- VoEFile* file = _mgr.FilePtr();
-
-#ifdef _USE_EXTENDED_TRACE_
- TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename("VoECodec_trace.txt")));
- TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
- kTraceStateInfo |
- kTraceWarning |
- kTraceError |
- kTraceCritical |
- kTraceApiCall |
- kTraceMemory |
- kTraceInfo));
-#endif
-
- TEST_MUSTPASS(voe_base_->Init());
- TEST_MUSTPASS(voe_base_->CreateChannel());
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
- ExtendedTestTransport* ptrTransport(NULL);
- ptrTransport = new ExtendedTestTransport(netw);
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
-#else
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
-#endif
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
-
- ///////////////////////////
- // Actual test starts here
-
- int i;
- int err;
-
- CodecInst cinst;
-
- /////////////////////////
- // GetNumOfCodecs
-
- int nCodecs;
-
- TEST(GetNumOfCodecs);
- ANL();
- // validate #codecs
- nCodecs = codec->NumOfCodecs();
- MARK();
- TEST_MUSTPASS(nCodecs < 0);
- AOK();
- ANL();
-
- ///////////////////
- // GetCodec
- TEST(GetCodec);
- ANL();
- // scan all supported codecs
- nCodecs = codec->NumOfCodecs();
- for (int index = 0; index < nCodecs; index++) {
- TEST_MUSTPASS(codec->GetCodec(index, cinst));
- TEST_LOG("[%2d] %16s: fs=%6d, pt=%4d, rate=%7d, ch=%2d, size=%5d", index, cinst.plname,
- cinst.plfreq, cinst.pltype, cinst.rate, cinst.channels, cinst.pacsize);
- if (cinst.pltype == -1) {
- TEST_LOG(" <= NOTE pt=-1\n");
- } else {
- ANL();
- }
- }
-
- // ensure that an invalid index parameter is detected
- TEST_MUSTPASS(-1 != codec->GetCodec(-1, cinst));
- nCodecs = codec->NumOfCodecs();
- TEST_MUSTPASS(-1 != codec->GetCodec(nCodecs, cinst));
- MARK();
- // ensure that error code is VE_INVALID_LISTNR
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_LISTNR);
- AOK();
- ANL();
-
- ///////////////////////
- // GetSendCodec
- TEST(GetSendCodec);
- ANL();
-
- CodecInst defaultCodec;
-
- // check the channel parameter
- int nMaxChannels(voe_base_->MaxNumOfChannels());
- TEST_MUSTPASS(-1 != codec->GetSendCodec(nMaxChannels-1, cinst));
- MARK(); // not created
- TEST_MUSTPASS(-1 != codec->GetSendCodec(nMaxChannels, cinst));
- MARK(); // out of range
- TEST_MUSTPASS(-1 != codec->GetSendCodec(-1, cinst));
- MARK(); // out of range
- TEST_MUSTPASS(codec->GetSendCodec(0, cinst));
- MARK(); // OK
-
- nCodecs = codec->NumOfCodecs();
- for (int index = 0; index < nCodecs; index++) {
- TEST_MUSTPASS(codec->GetCodec(index, defaultCodec));
- if (codec->SetSendCodec(0, defaultCodec) == 0) {
- TEST_MUSTPASS(codec->GetSendCodec(0, cinst));
- MARK();
- //TEST_LOG("[%2d] %s: fs=%d, pt=%d, rate=%d, ch=%d, size=%d\n",
- // index, cinst.plname, cinst.plfreq, cinst.pltype, cinst.rate,
- // cinst.channels, cinst.pacsize);
- TEST_MUSTPASS(cinst.pacsize != defaultCodec.pacsize);
- TEST_MUSTPASS(cinst.plfreq != defaultCodec.plfreq);
- TEST_MUSTPASS(cinst.pltype != defaultCodec.pltype);
- TEST_MUSTPASS(cinst.rate != defaultCodec.rate);
- TEST_MUSTPASS(cinst.channels != defaultCodec.channels);
- }
- }
-
- ANL();
- AOK();
- ANL();
-
- ///////////////////////
- // SetSendCodec
- TEST(SetSendCodec);
- ANL();
-
- // --- Scan all supported codecs and set default parameters
-
- nCodecs = codec->NumOfCodecs();
- for (int index = 0; index < nCodecs; index++) {
- // Get default (ACM) settings
- TEST_MUSTPASS(codec->GetCodec(index, cinst));
- defaultCodec = cinst;
- TEST_LOG("[%2d] %s (default): fs=%d, pt=%d, rate=%d, ch=%d, size=%d\n",
- index, cinst.plname, cinst.plfreq, cinst.pltype, cinst.rate,
- cinst.channels, cinst.pacsize);
-
- // Verify invalid codec names
- if (!_stricmp("CN", cinst.plname) || !_stricmp("telephone-event",
- cinst.plname)
- || !_stricmp("red", cinst.plname)) {
- // default settings for invalid payload names (should give
- // VE_INVALID_PLNAME)
- TEST_MUSTPASS(!codec->SetSendCodec(0, cinst));
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
- continue;
- }
-
- // If we build the ACM with more codecs than we have payload types,
- // some codecs will be given -1 as default payload type. This is a fix
- // to ensure that we can complete these tests also for this case.
- if (cinst.pltype == -1) {
- cinst.pltype = 97;
- }
-
- // --- Default settings
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
-
- // --- Packet size
- TEST_LOG("\npacsize : ");
-
- for (int pacsize = 80; pacsize < 1440; pacsize += 80) {
- cinst.pacsize = pacsize;
- if (-1 != codec->SetSendCodec(0, cinst)) {
- // log valid packet size
- TEST_LOG("%d ", pacsize);
- } else {
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
- }
- }
- cinst.pacsize = defaultCodec.pacsize;
-
- // --- Audio channels (1/mono or 2/stereo)
- TEST_LOG("\nchannels: ");
- for (int channels = 1; channels < 4; channels++) {
- cinst.channels = channels;
- if (-1 != codec->SetSendCodec(0, cinst)) {
- // valid channels (only 1 should be OK)
- TEST_LOG("%d ", channels);
- } else {
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
- }
- }
- cinst.channels = defaultCodec.channels;
-
- // --- Payload frequency
- TEST_LOG("\nplfreq : ");
- cinst.plfreq = defaultCodec.plfreq;
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_LOG("%d ", cinst.plfreq);
-
- // --- Payload name
-
- strcpy(cinst.plname, "INVALID");
- TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst))
- {
- // ensure that error code is VE_INVALID_PLNAME
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
- }
-
- // restore default plname
- strcpy(cinst.plname, defaultCodec.plname);
-
- // --- Payload type (dynamic range is 96-127)
- TEST_LOG("\npltype : ");
- // All PT should be OK, test a few different
- cinst.pltype = defaultCodec.pltype;
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_LOG("%d ", cinst.pltype);
- cinst.pltype = defaultCodec.pltype + 1;
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_LOG("%d ", cinst.pltype);
- const int valid_pltypes[4] = { 0, 96, 117, 127 };
- for (i = 0; i < static_cast<int> (sizeof(valid_pltypes) / sizeof(int)); i++) {
- cinst.pltype = valid_pltypes[i];
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_LOG("%d ", cinst.pltype);
- }
- // Restore default
- cinst.pltype = defaultCodec.pltype;
-
- // --- Codec rate
- TEST_LOG("\nrate : ");
- if (_stricmp("isac", cinst.plname) == 0) {
- // ISAC
- if (cinst.plfreq == 16000) {
- int valid_rates[3] = { -1, 10000, 32000 };
- // failed in RegisterPayload when rate is 32000
- for (i = 0; i < static_cast<int> (sizeof(valid_rates) / sizeof(int)); i++) {
- cinst.rate = valid_rates[i];
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_LOG("%d ", cinst.rate);
- }
- cinst.rate = 0; // invalid
- TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst))
- {
- // ensure that error code is VE_CANNOT_SET_SEND_CODEC
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
- }
- ANL();
- } else //ISACSWB
- {
- // rate changing fails in RegisterPayload
- int valid_rates[8] = { -1, 10000, 25000, 32000, 35000, 45000, 50000, 52000 };
- for (i = 0; i < static_cast<int> (sizeof(valid_rates) / sizeof(int)); i++) {
- cinst.rate = valid_rates[i];
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_LOG("%d ", cinst.rate);
- }
- int invalid_rates[3] = { 0, 5000, 57000 }; // invalid
- for (i = 0; i < static_cast<int> (sizeof(invalid_rates) / sizeof(int)); i++) {
- cinst.rate = invalid_rates[i];
- TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst))
- {
- // ensure that error code is VE_CANNOT_SET_SEND_CODEC
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
- }
- }
- ANL();
- }
- } else if (_stricmp("amr", cinst.plname) == 0) {
- int valid_rates[8] = { 4750, 5150, 5900, 6700, 7400, 7950, 10200, 12200 };
- for (i = 0;
- i < static_cast<int> (sizeof(valid_rates) / sizeof(int));
- i++) {
- cinst.rate = valid_rates[i];
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_LOG("%d ", cinst.rate);
- }
- ANL();
- } else if (_stricmp("g7291", cinst.plname) == 0) {
- int valid_rates[12] = { 8000, 12000, 14000, 16000, 18000, 20000, 22000,
- 24000, 26000, 28000, 30000, 32000 };
- for (i = 0;
- i < static_cast<int> (sizeof(valid_rates) / sizeof(int));
- i++) {
- cinst.rate = valid_rates[i];
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_LOG("%d ", cinst.rate);
- }
- ANL();
- } else if (_stricmp("amr-wb", cinst.plname) == 0) {
- int valid_rates[9] = { 7000, 9000, 12000, 14000, 16000, 18000, 20000,
- 23000, 24000 };
- for (i = 0;
- i < static_cast<int> (sizeof(valid_rates) / sizeof(int));
- i++) {
- cinst.rate = valid_rates[i];
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_LOG("%d ", cinst.rate);
- }
- TEST_LOG(" <=> ");
- ANL();
- } else if (_stricmp("speex", cinst.plname) == 0) {
- // Valid speex rates are > 2000, testing some of them here
- int valid_rates[9] = { 2001, 4000, 7000, 11000, 15000, 20000, 25000,
- 33000, 46000 };
- for (i = 0;
- i < static_cast<int> (sizeof(valid_rates) / sizeof(int));
- i++) {
- cinst.rate = valid_rates[i];
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_LOG("%d ", cinst.rate);
- }
- cinst.rate = 2000; // invalid
- TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst))
- {
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
- }
- ANL();
- } else if (_stricmp("silk", cinst.plname) == 0) {
- // Valid Silk rates are 6000 - 40000, listing some of them here
- int valid_rates[7] = { 6000, 10000, 15000, 20000, 25000, 32000, 40000 };
- for (i = 0;
- i < static_cast<int> (sizeof(valid_rates) / sizeof(int));
- i++) {
- cinst.rate = valid_rates[i];
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_LOG("%d ", cinst.rate);
- }
- cinst.rate = 5999; // invalid
- TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst))
- {
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
- }
- cinst.rate = 40001; // invalid
- TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst))
- {
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
- }
- ANL();
- } else {
- // Use default rate for all other codecs.
- cinst.rate = defaultCodec.rate;
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_LOG("%d ", cinst.rate);
- cinst.rate = defaultCodec.rate + 17;
- TEST_MUSTPASS(!codec->SetSendCodec(0, cinst));
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
- ANL();
- }
- cinst.rate = defaultCodec.rate;
-
- // run some extra tests for L16
- if (_stricmp("l16", cinst.plname) == 0) {
- if (8000 == cinst.plfreq) {
- // valid pacsizes: 80, 160, 240, 320
- cinst.pacsize = 480; // only supported in combination with 16kHz
- TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst));
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
- cinst.pacsize = 640; // only supported in combination with 16kHz
- TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst));
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
- } else {
- // valid pacsizes: 160, 320, 480, 640
- cinst.pacsize = 80; // only supported in combination with 8kHz
- TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst));
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
- cinst.pacsize = 240; // only supported in combination with 8kHz
- TEST_MUSTPASS(-1 != codec->SetSendCodec(0, cinst));
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
- }
- }
- ANL();
- } // for (int index = 0; index < nCodecs; index++)
-
- // restore PCMU
- const CodecInst tmp = { 0, "PCMU", 8000, 160, 1, 64000 };
- TEST_MUSTPASS(codec->SetSendCodec(0, tmp));
-
- ANL();
- AOK();
- ANL();
-
- ///////
- // VAD
-
- const int VADSleep = 0;
-
- bool disabledDTX;
- VadModes mode;
- bool enabled;
-
- // verify default settings (should be OFF, kVadConventional and DTX enabled)
- TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
- TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
- disabledDTX);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(mode != kVadConventional);
- TEST_MUSTPASS(disabledDTX != true);
-
- // enable default VAD settings
- TEST_MUSTPASS(codec->SetVADStatus(0, true));
- TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
- TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
- disabledDTX);
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(mode != kVadConventional);
- TEST_MUSTPASS(disabledDTX != false);
- SLEEP(VADSleep);
-
- // set kVadConventional mode
- TEST_MUSTPASS(codec->SetVADStatus(0, true, kVadConventional));
- TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
- TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
- disabledDTX);
- TEST_MUSTPASS(mode != kVadConventional);
- SLEEP(VADSleep);
-
- // set kVadAggressiveLow mode
- TEST_MUSTPASS(codec->SetVADStatus(0, true, kVadAggressiveLow));
- TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
- TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
- disabledDTX);
- TEST_MUSTPASS(mode != kVadAggressiveLow);
- SLEEP(VADSleep);
-
- // set kVadAggressiveMid mode
- TEST_MUSTPASS(codec->SetVADStatus(0, true, kVadAggressiveMid));
- TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
- TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
- disabledDTX);
- TEST_MUSTPASS(mode != kVadAggressiveMid);
- SLEEP(VADSleep);
-
- // set kVadAggressiveMid mode
- TEST_MUSTPASS(codec->SetVADStatus(0, true, kVadAggressiveHigh));
- TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
- TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
- disabledDTX);
- TEST_MUSTPASS(mode != kVadAggressiveHigh);
- SLEEP(VADSleep);
-
- // turn DTX OFF (audio should not be affected by VAD decisions)
- TEST_MUSTPASS(codec->SetVADStatus(0, true, kVadConventional, true));
- TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
- TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
- disabledDTX);
- TEST_MUSTPASS(disabledDTX != true);
- SLEEP(VADSleep);
-
- // try to enable DTX again (should fail since VAD is disabled)
- TEST_MUSTPASS(codec->SetVADStatus(0, false, kVadConventional, false));
- TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
- TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
- disabledDTX);
- TEST_MUSTPASS(disabledDTX == false);
- SLEEP(VADSleep);
-
- // disable VAD
- TEST_MUSTPASS(codec->SetVADStatus(0, false));
- TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
- TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
- disabledDTX);
- TEST_MUSTPASS(enabled != false);
- SLEEP(VADSleep);
-
- // restore default VAD
- TEST_MUSTPASS(codec->SetVADStatus(0, true));
- TEST_MUSTPASS(codec->SetVADStatus(0, false));
- TEST_MUSTPASS(codec->GetVADStatus(0, enabled, mode, disabledDTX));
- TEST_LOG("VAD: enabled=%d, mode=%d, disabledDTX=%d\n", enabled, mode,
- disabledDTX);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(mode != kVadConventional);
- TEST_MUSTPASS(disabledDTX != true);
- SLEEP(VADSleep);
-
- AOK();
- ANL();
- ANL();
-
- //////////////////////
- // GetRecCodec
- TEST(GetRecCodec);
- ANL();
-
- // stop all streaming first
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
-
- // start loopback streaming (PCMU is default)
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- TEST_MUSTPASS(voe_base_->SetSendDestination(0,8000,"127.0.0.1"));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0,8000));
-#endif
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(100); // ensure that at least one packets is received
-
- // scan all supported and valid codecs
- CodecInst newCodec;
- for (i = 0; i < codec->NumOfCodecs(); i++) {
- TEST_MUSTPASS(codec->GetCodec(i, newCodec));
- // test all valid send codecs
- if (!_stricmp("red", newCodec.plname) || !_stricmp("cn", newCodec.plname)
- || !_stricmp("telephone-event", newCodec.plname)) {
- continue; // Ignore these
- }
- if (-1 != codec->SetSendCodec(0, newCodec)) {
- SLEEP(150);
- // verify correct detection
- TEST_MUSTPASS(codec->GetRecCodec(0, cinst));
- TEST_LOG("%s %s ", newCodec.plname, cinst.plname);
- TEST_MUSTPASS(_stricmp(newCodec.plname, cinst.plname) != 0);
- TEST_MUSTPASS(cinst.pltype != newCodec.pltype);
- TEST_MUSTPASS(cinst.plfreq != newCodec.plfreq);
- }
- }
-
- // stop streaming
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
-
- ANL();
- AOK();
- ANL();
- ANL();
-
-#ifdef WEBRTC_CODEC_GSMAMR
- //////////////////////////
- // SetAMREncFormat
-
- // Fresh channel
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- TEST(SetAMREncFormat); ANL();
-
- //set another codec which is not AMR
- TEST_MUSTPASS(codec->GetCodec(0, cinst));
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- //try to change the encode format, tests should fail
- TEST_MUSTPASS(-1 != codec->SetAMREncFormat(0)); MARK();
- TEST_MUSTPASS(-1 != codec->SetAMREncFormat(0, kRfc3267BwEfficient));
- MARK();
- TEST_MUSTPASS(-1 != codec->SetAMREncFormat(0, kRfc3267OctetAligned));
- MARK();
- TEST_MUSTPASS(-1 != codec->SetAMREncFormat(0, kRfc3267FileStorage));
- MARK();
-
- //set AMR as encoder
- strcpy(cinst.plname,"AMR");
- cinst.channels=1; cinst.plfreq=8000; cinst.rate=12200; cinst.pltype=112;
- cinst.pacsize=160;
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- //try to change the encode format, tests should pass
- TEST_MUSTPASS(codec->SetAMREncFormat(0)); MARK();
- TEST_MUSTPASS(codec->SetAMREncFormat(0, kRfc3267BwEfficient)); MARK();
- TEST_MUSTPASS(codec->SetAMREncFormat(0, kRfc3267OctetAligned)); MARK();
- TEST_MUSTPASS(codec->SetAMREncFormat(0, kRfc3267FileStorage)); MARK();
- TEST_MUSTPASS(-1 != codec->SetAMREncFormat(-1)); MARK();
- TEST_MUSTPASS(codec->SetAMREncFormat(0)); MARK(); // restore default
-
- ANL();
- AOK();
- ANL();
-
- //////////////////////////
- // SetAMRDecFormat
-
- TEST(SetAMRDecFormat); ANL();
-
- // It should not be possible to set AMR dec format before valid AMR decoder
- // is registered
- TEST_MUSTPASS(!codec->SetAMRDecFormat(0)); MARK();
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_AUDIO_CODING_MODULE_ERROR);
-
- // Ensure that ACM::RegisterReceiveCodec(AMR) is called
- TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
-
- // All these tests should now pass
- TEST_MUSTPASS(codec->SetAMRDecFormat(0)); MARK();
- TEST_MUSTPASS(codec->SetAMRDecFormat(0, kRfc3267BwEfficient)); MARK();
- TEST_MUSTPASS(codec->SetAMRDecFormat(0, kRfc3267OctetAligned)); MARK();
- TEST_MUSTPASS(codec->SetAMRDecFormat(0, kRfc3267FileStorage)); MARK();
- TEST_MUSTPASS(-1 != codec->SetAMRDecFormat(-1)); MARK();
- TEST_MUSTPASS(codec->SetAMRDecFormat(0)); MARK(); // restore default
-
- ANL();
- AOK();
- ANL();
-#endif // #ifdef WEBRTC_CODEC_GSMAMR
-#ifdef WEBRTC_CODEC_GSMAMRWB
- //////////////////////////
- // SetAMRWbEncFormat
-
- // Fresh channel
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- TEST(SetAMRWbEncFormat); ANL();
-
- //set another codec which is not AMR-wb
- TEST_MUSTPASS(codec->GetCodec(0, cinst));
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- //try to change the encode format, tests should fail
- TEST_MUSTPASS(-1 != codec->SetAMRWbEncFormat(0)); MARK();
- TEST_MUSTPASS(-1 != codec->SetAMRWbEncFormat(0, kRfc3267BwEfficient));
- MARK();
- TEST_MUSTPASS(-1 != codec->SetAMRWbEncFormat(0, kRfc3267OctetAligned));
- MARK();
- TEST_MUSTPASS(-1 != codec->SetAMRWbEncFormat(0, kRfc3267FileStorage));
- MARK();
-
- //set AMR-wb as encoder
- strcpy(cinst.plname,"AMR-WB");
- cinst.channels=1; cinst.plfreq=16000; cinst.rate=20000;
- cinst.pltype=112; cinst.pacsize=320;
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- //try to change the encode format, tests should pass
- TEST_MUSTPASS(codec->SetAMRWbEncFormat(0)); MARK();
- TEST_MUSTPASS(codec->SetAMRWbEncFormat(0, kRfc3267BwEfficient)); MARK();
- TEST_MUSTPASS(codec->SetAMRWbEncFormat(0, kRfc3267OctetAligned)); MARK();
- TEST_MUSTPASS(codec->SetAMRWbEncFormat(0, kRfc3267FileStorage)); MARK();
- TEST_MUSTPASS(-1 != codec->SetAMRWbEncFormat(-1)); MARK();
- TEST_MUSTPASS(codec->SetAMRWbEncFormat(0)); MARK(); // restore default
-
- ANL();
- AOK();
- ANL();
-
- //////////////////////////
- // SetAMRDecFormat
-
- TEST(SetAMRWbDecFormat); ANL();
-
- // It should not be possible to set AMR dec format before valid AMR decoder
- // is registered
- TEST_MUSTPASS(!codec->SetAMRWbDecFormat(0)); MARK();
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_AUDIO_CODING_MODULE_ERROR);
-
- // Ensure that ACM::RegisterReceiveCodec(AMR) is called
- TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
-
- // All these tests should now pass
- TEST_MUSTPASS(codec->SetAMRWbDecFormat(0)); MARK();
- TEST_MUSTPASS(codec->SetAMRWbDecFormat(0, kRfc3267BwEfficient)); MARK();
- TEST_MUSTPASS(codec->SetAMRWbDecFormat(0, kRfc3267OctetAligned)); MARK();
- TEST_MUSTPASS(codec->SetAMRWbDecFormat(0, kRfc3267FileStorage)); MARK();
- TEST_MUSTPASS(-1 != codec->SetAMRWbDecFormat(-1)); MARK();
- TEST_MUSTPASS(codec->SetAMRWbDecFormat(0)); MARK(); // restore default
-
- ANL();
- AOK();
- ANL();
-#endif // #ifdef WEBRTC_CODEC_GSMAMRWB
- ///////////////////////////////
- // SetSendCNPayloadType
- TEST(SetSendCNPayloadType);
- ANL();
-
- TEST_MUSTPASS(-1 != codec->SetSendCNPayloadType(-1, 0));
- MARK(); // invalid channel
-
- // Invalid payload range (only dynamic range [96,127]
- TEST_MUSTPASS(-1 != codec->SetSendCNPayloadType(0, 0));
- MARK(); // invalid PT
- TEST_MUSTPASS(-1 != codec->SetSendCNPayloadType(0, 95));
- MARK(); // invalid PT
- TEST_MUSTPASS(-1 != codec->SetSendCNPayloadType(0, 128));
- MARK(); // invalid PT
- TEST_MUSTPASS(-1 != codec->SetSendCNPayloadType(0, -1));
- MARK(); // invalid PT
-
- // Not possible to change PT for 8000
- TEST_MUSTPASS(!codec->SetSendCNPayloadType(0, 96, kFreq8000Hz));
- MARK();
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_PLFREQ);
-
- // Try some dynamic for 16000 and 32000 as well
- TEST_MUSTPASS(codec->SetSendCNPayloadType(0, 96, kFreq16000Hz));
- MARK();
- TEST_MUSTPASS(codec->SetSendCNPayloadType(0, 96, kFreq32000Hz));
- MARK(); // same should work
- TEST_MUSTPASS(codec->SetSendCNPayloadType(0, 127, kFreq16000Hz));
- MARK();
- TEST_MUSTPASS(codec->SetSendCNPayloadType(0, 127, kFreq32000Hz));
- MARK();
- TEST_MUSTPASS(codec->SetSendCNPayloadType(0, 100, kFreq32000Hz));
- MARK();
-
- ANL();
- AOK();
- ANL();
-
- /////////////////////////////
- // SetRecPayloadType
- TEST(SetRecPayloadType);
- ANL();
-
- // scan all supported and valid codecs without changing payloads
- nCodecs = codec->NumOfCodecs();
- for (i = 0; i < nCodecs; i++) {
- TEST_MUSTPASS(codec->GetCodec(i, newCodec));
- // If no default payload type is defined, we use 127
- if (-1 == newCodec.pltype) {
- newCodec.pltype = 127;
- }
- TEST_MUSTPASS(codec->SetRecPayloadType(0, newCodec));
- MARK(); // use default
- newCodec.pltype = 99;
- TEST_MUSTPASS(codec->SetRecPayloadType(0, newCodec));
- MARK(); // use same PT on all
- newCodec.pltype = -1;
- TEST_MUSTPASS(codec->SetRecPayloadType(0, newCodec));
- MARK(); // deregister all PTs
- }
-
- ANL();
- AOK();
- ANL();
-
- /////////////////////////////
- // GetRecPayloadType
- TEST(GetRecPayloadType);
- ANL();
-
- CodecInst extraCodec;
- for (i = 0; i < nCodecs; i++) {
- // Set defaults
- TEST_MUSTPASS(codec->GetCodec(i, newCodec));
- // If no default payload type is defined, we use 127
- if (-1 == newCodec.pltype) {
- newCodec.pltype = 127;
- }
- TEST_MUSTPASS(codec->SetRecPayloadType(0, newCodec));
- //TEST_LOG("[%2d] %s (SetRec): fs=%d, pt=%d, rate=%d, ch=%d, size=%d\n",
- // i, newCodec.plname, newCodec.plfreq, newCodec.pltype, newCodec.rate,
- // newCodec.channels, newCodec.pacsize);
- extraCodec.pltype = -1; // don't know this yet
- extraCodec.plfreq = newCodec.plfreq;
- extraCodec.rate = newCodec.rate;
- extraCodec.channels = newCodec.channels;
- strcpy(extraCodec.plname, newCodec.plname);
- // Verfify that setting is OK
- TEST_MUSTPASS(codec->GetRecPayloadType(0, extraCodec));
- //TEST_LOG("[%2d] %s (GetRec): fs=%d, pt=%d, rate=%d, ch=%d, size=%d\n",
- // i, extraCodec.plname, extraCodec.plfreq, extraCodec.pltype,
- // extraCodec.rate, extraCodec.channels, extraCodec.pacsize);
- TEST_MUSTPASS(newCodec.pltype != extraCodec.pltype);
- TEST_MUSTPASS(newCodec.plfreq != extraCodec.plfreq);
- TEST_MUSTPASS(newCodec.channels != extraCodec.channels);
- }
-
- AOK();
- ANL();
-
- ////////////////////////////////////////////////////
- // SetRecPayloadType - remove receive codecs
- TEST(SetRecPayloadType - removing receive codecs);
- ANL();
-
-#ifndef WEBRTC_EXTERNAL_TRANSPORT
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
-#endif
- TEST_MUSTPASS(voe_base_->StartSend(0));
- if (file) {
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0,
- _mgr.AudioFilename(),
- true,
- true));
- }
-
- // Scan all supported and valid codecs and remove from receiving db, then
- // restore
- nCodecs = codec->NumOfCodecs();
- for (i = 0; i < nCodecs; i++) {
- TEST_MUSTPASS(codec->GetCodec(i, cinst));
- if (!_stricmp("red", cinst.plname) || !_stricmp("cn", cinst.plname)
- || !_stricmp("telephone-event", cinst.plname)) {
- continue; // Ignore these
- }
- TEST_LOG("Testing codec: %s", cinst.plname);
- fflush(NULL);
-
- if (-1 == cinst.pltype) {
- // If no default payload type is defined, we use 127,
- // codec is not registered for receiving
- cinst.pltype = 127;
- } else {
- // Remove codec
- memcpy(&extraCodec, &cinst, sizeof(CodecInst));
- extraCodec.pltype = -1;
- TEST_MUSTPASS(codec->SetRecPayloadType(0, extraCodec));
- }
-
- // Set send codec
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
-
- // Verify no audio
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_LOG(" silence");
- fflush(NULL);
- SLEEP(800);
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
-
- // Restore codec
- TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
-
- // Verify audio
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_LOG(" audio");
- fflush(NULL);
- SLEEP(800);
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
-
- if (127 == cinst.pltype) {
- // If no default payload type is defined, i.e. we have set pt to
- //127 above,
- // make sure we remove codec from receiving
- cinst.pltype = -1;
- TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
- }
-
- ANL();
- }
-
- // Remove certain codecs
- TEST_LOG("Removing receive codecs:");
- for (i = 0; i < nCodecs; i++) {
- TEST_MUSTPASS(codec->GetCodec(i, cinst));
- if (!_stricmp("ipcmwb", cinst.plname) || !_stricmp("pcmu", cinst.plname)
- || !_stricmp("eg711a", cinst.plname)) {
- TEST_LOG(" %s", cinst.plname);
- memcpy(&extraCodec, &cinst, sizeof(CodecInst));
- extraCodec.pltype = -1;
- TEST_MUSTPASS(codec->SetRecPayloadType(0, extraCodec));
- }
- }
- ANL();
-
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
-
- // Test sending all codecs - verify audio/no audio depending on codec
- TEST_LOG("Looping through send codecs \n");
- TEST_LOG("Verify that removed codecs are not audible and the other are \n");
- for (i = 0; i < nCodecs; i++) {
- TEST_MUSTPASS(codec->GetCodec(i, cinst));
- if (!_stricmp("red", cinst.plname) || !_stricmp("cn", cinst.plname)
- || !_stricmp("telephone-event", cinst.plname)) {
- continue; // Ignore these
- }
- TEST_LOG("Testing codec: %s \n", cinst.plname);
-
- // If no default payload type is defined, we use 127 and set receive
- // payload type
- if (-1 == cinst.pltype) {
- cinst.pltype = 127;
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- }
-
- // Set send codec
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
-
- // Verify audio/no audio
- SLEEP(800);
- }
-
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
-
- // Restore codecs
- TEST_LOG("Restoring receive codecs:");
- for (i = 0; i < nCodecs; i++) {
- TEST_MUSTPASS(codec->GetCodec(i, cinst));
- if (!_stricmp("ipcmwb", cinst.plname) || !_stricmp("pcmu", cinst.plname)
- || !_stricmp("eg711a", cinst.plname)) {
- TEST_LOG(" %s", cinst.plname);
- memcpy(&extraCodec, &cinst, sizeof(CodecInst));
- TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
- }
- }
- ANL();
-
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
-
- // Test sending all codecs - verify audio
- TEST_LOG("Looping through send codecs \n");
- TEST_LOG("Verify that all codecs are audible \n");
- for (i = 0; i < nCodecs; i++) {
- TEST_MUSTPASS(codec->GetCodec(i, cinst));
- if (!_stricmp("red", cinst.plname) || !_stricmp("cn", cinst.plname)
- || !_stricmp("telephone-event", cinst.plname)) {
- continue; // Ignore these
- }
- TEST_LOG("Testing codec: %s \n", cinst.plname);
-
- // If no default payload type is defined, we use 127 and set receive
- // payload type
- if (-1 == cinst.pltype) {
- cinst.pltype = 127;
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- }
-
- // Set send codec
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
-
- // Verify audio/no audio
- SLEEP(800);
- }
-
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
-
- // Fresh channel
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
-#if defined(WEBRTC_CODEC_ISAC)
-
- /////////////////////////////////////
- // SetISACInitTargetRate - wb
- TEST(SetISACInitTargetRate);
- ANL();
-
- // set PCMU as sending codec
- cinst.channels = 1;
- cinst.pacsize = 160;
- cinst.plfreq = 8000;
- strcpy(cinst.plname, "PCMU");
- cinst.pltype = 0;
- cinst.rate = 64000;
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
-
- TEST_MUSTPASS(!codec->SetISACInitTargetRate(0, 10000));
- MARK(); // should fail since iSAC is not active
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_CODEC_ERROR);
-
- // set iSAC as sending codec (16kHz)
- cinst.channels = 1;
- cinst.plfreq = 16000;
- strcpy(cinst.plname, "ISAC");
- cinst.pltype = 103;
- cinst.rate = -1; // adaptive rate
- cinst.pacsize = 480; // 30ms
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
-
- TEST_MUSTPASS(!codec->SetISACInitTargetRate(1, 10000));
- MARK(); // invalid channel
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(!codec->SetISACInitTargetRate(0, 500));
- MARK(); // invalid target rates (too small)
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
-
- TEST_MUSTPASS(!codec->SetISACInitTargetRate(0, 33000));
- MARK(); // invalid target rates (too large)
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
-
- TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 10000));
- MARK(); // life is good now
- TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 0));
- MARK(); // 0 is a valid rate
- TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 32000));
- MARK(); // try max as well
- TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 32000, true));
- MARK();
- TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 32000, false));
- MARK();
-
- cinst.pacsize = 960; // 60ms
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 32000, false));
- MARK();
-
- cinst.rate = 20000;
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_MUSTPASS(!codec->SetISACInitTargetRate(0, 32000));
- MARK(); // only works in adaptive mode
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_AUDIO_CODING_MODULE_ERROR);
-
- cinst.rate = -1;
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 32000));
- MARK(); // back to adaptive mode
-
- ANL();
- AOK();
- ANL();
-
- /////////////////////////////////////
- // SetISACInitTargetRate - swb
- TEST(ISACSWB SetISACInitTargetRate);
- ANL();
-
- // set iSAC as sending codec
- cinst.channels = 1;
- cinst.plfreq = 32000;
- strcpy(cinst.plname, "ISAC");
- cinst.pltype = 104;
- cinst.rate = -1; // default rate
- cinst.pacsize = 960; // 30ms
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
-
- TEST_MUSTPASS(!codec->SetISACInitTargetRate(1, 10000));
- MARK(); // invalid channel
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(!codec->SetISACInitTargetRate(0, -1));
- MARK(); // invalid target rates (too small)
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!codec->SetISACInitTargetRate(0, -1));
- MARK(); // invalid target rates (too small)
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
-
- TEST_MUSTPASS(!codec->SetISACInitTargetRate(0, 500));
- MARK(); // invalid target rates (too small)
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
-
- TEST_MUSTPASS(!codec->SetISACInitTargetRate(0, 57000));
- MARK(); // invalid target rates (valid range is [10000, 56000])
-
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
-
- TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 10000));
- MARK();
- TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 0));
- MARK();
- TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 56000));
- MARK(); // try max as well
- TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 56000, true));
- MARK();
- TEST_MUSTPASS(codec->SetISACInitTargetRate(0, 56000, false));
- MARK();
-
- ANL();
- AOK();
- ANL();
-
- ////////////////////////////////
- // SetISACMaxRate
- TEST(SetISACMaxRate);
- ANL();
-
- // set PCMU as sending codec
- cinst.channels = 1;
- cinst.pacsize = 160;
- cinst.plfreq = 8000;
- strcpy(cinst.plname, "PCMU");
- cinst.pltype = 0;
- cinst.rate = 64000;
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
-
- TEST_MUSTPASS(!codec->SetISACMaxRate(0, 48000));
- MARK(); // should fail since iSAC is not active
- TEST_MUSTPASS(voe_base_->LastError() != VE_CODEC_ERROR);
-
- // set iSAC as sending codec
- cinst.channels = 1;
- cinst.plfreq = 16000;
- strcpy(cinst.plname, "ISAC");
- cinst.pltype = 103;
- cinst.rate = -1; // adaptive rate
- cinst.pacsize = 480; // 30ms
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
-
- TEST_MUSTPASS(!codec->SetISACMaxRate(1, 48000));
- MARK(); // invalid channel
- TEST_MUSTPASS(voe_base_->LastError() != VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(!codec->SetISACMaxRate(0, 31900));
- MARK(); // invalid target rates (too small)
- TEST_MUSTPASS(voe_base_->LastError() != VE_INVALID_ARGUMENT);
-
- TEST_MUSTPASS(!codec->SetISACMaxRate(0, 53500));
- MARK(); // invalid target rates (too large)
- TEST_MUSTPASS(voe_base_->LastError() != VE_INVALID_ARGUMENT);
-
- TEST_MUSTPASS(codec->SetISACMaxRate(0, 32000));
- MARK(); // life is good now
- TEST_MUSTPASS(codec->SetISACMaxRate(0, 40000));
- MARK();
- TEST_MUSTPASS(codec->SetISACMaxRate(0, 48000));
- MARK();
- TEST_MUSTPASS(codec->SetISACMaxRate(0, 53400));
- MARK(); // try max as well (default)
-
- cinst.pacsize = 960; // 60ms
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_MUSTPASS(codec->SetISACMaxRate(0, 48000));
- MARK();
-
- cinst.rate = 20000;
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_MUSTPASS(codec->SetISACMaxRate(0, 40000));
- MARK(); // also works in non-adaptive mode
-
- ANL();
- AOK();
- ANL();
-
- TEST(ISACSWB SetISACMaxRate);
- ANL();
- // set iSAC as sending codec
- cinst.channels = 1;
- cinst.plfreq = 32000;
- strcpy(cinst.plname, "ISAC");
- cinst.pltype = 104;
- cinst.rate = 45000; // instantaneous mode
- cinst.pacsize = 960; // 30ms
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
-
- TEST_MUSTPASS(!codec->SetISACMaxRate(1, 48000));
- MARK(); // invalid channel
- TEST_MUSTPASS(voe_base_->LastError() != VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(!codec->SetISACMaxRate(0, 31900));
- MARK(); // invalid target rates (too small)
- TEST_MUSTPASS(voe_base_->LastError() != VE_INVALID_ARGUMENT);
-
- TEST_MUSTPASS(!codec->SetISACMaxRate(0, 107500));
- MARK(); // invalid target rates (too large)
- TEST_MUSTPASS(voe_base_->LastError() != VE_INVALID_ARGUMENT);
-
- TEST_MUSTPASS(codec->SetISACMaxRate(0, 32000));
- MARK(); // life is good now
- TEST_MUSTPASS(codec->SetISACMaxRate(0, 40000));
- MARK();
- TEST_MUSTPASS(codec->SetISACMaxRate(0, 55000));
- MARK();
- TEST_MUSTPASS(codec->SetISACMaxRate(0, 80000));
- MARK();
- TEST_MUSTPASS(codec->SetISACMaxRate(0, 107000));
- MARK(); // try max as well (default)
-
-
- cinst.rate = -1; // adaptive mode
- cinst.pacsize = 960; // 30ms
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
-
- TEST_MUSTPASS(!codec->SetISACMaxRate(1, 48000));
- MARK(); // invalid channel
- TEST_MUSTPASS(voe_base_->LastError() != VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(!codec->SetISACMaxRate(0, 31900));
- MARK(); // invalid target rates (too small)
- TEST_MUSTPASS(voe_base_->LastError() != VE_INVALID_ARGUMENT);
-
- TEST_MUSTPASS(!codec->SetISACMaxRate(0, 107500));
- MARK(); // invalid target rates (too large)
- TEST_MUSTPASS(voe_base_->LastError() != VE_INVALID_ARGUMENT);
-
- TEST_MUSTPASS(codec->SetISACMaxRate(0, 32000));
- MARK(); // life is good now
- TEST_MUSTPASS(codec->SetISACMaxRate(0, 40000));
- MARK();
- TEST_MUSTPASS(codec->SetISACMaxRate(0, 55000));
- MARK();
- TEST_MUSTPASS(codec->SetISACMaxRate(0, 80000));
- MARK();
- TEST_MUSTPASS(codec->SetISACMaxRate(0, 107000));
- MARK(); // try max as well (default)
-
- ANL();
- AOK();
- ANL();
-
- ////////////////////////////////
- // SetISACMaxPayloadSize
- TEST(SetISACMaxPayloadSize);
- ANL();
-
- // set PCMU as sending codec
- cinst.channels = 1;
- cinst.pacsize = 160;
- cinst.plfreq = 8000;
- strcpy(cinst.plname, "PCMU");
- cinst.pltype = 0;
- cinst.rate = 64000;
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
-
- TEST_MUSTPASS(!codec->SetISACMaxPayloadSize(0, 120));
- MARK(); // should fail since iSAC is not active
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_CODEC_ERROR);
-
- // set iSAC as sending codec
- cinst.channels = 1;
- cinst.plfreq = 16000;
- strcpy(cinst.plname, "ISAC");
- cinst.pltype = 103;
- cinst.rate = -1; // adaptive rate
- cinst.pacsize = 480; // 30ms
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
-
- TEST_MUSTPASS(!codec->SetISACMaxPayloadSize(1, 120));
- MARK(); // invalid channel
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(!codec->SetISACMaxPayloadSize(0, 100));
- MARK(); // invalid size (too small)
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
-
- TEST_MUSTPASS(!codec->SetISACMaxPayloadSize(0, 410));
- MARK(); // invalid size (too large)
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
-
- TEST_MUSTPASS(codec->SetISACMaxPayloadSize(0, 200));
- MARK(); // life is good now
- TEST_MUSTPASS(codec->SetISACMaxPayloadSize(0, 120));
- MARK();
- TEST_MUSTPASS(codec->SetISACMaxPayloadSize(0, 400));
- MARK();
-
- ANL();
- AOK();
- ANL();
-
- TEST(ISACSWB SetISACMaxPayloadSize);
- ANL();
- // set iSAC as sending codec
- cinst.channels = 1;
- cinst.plfreq = 32000;
- strcpy(cinst.plname, "ISAC");
- cinst.pltype = 104;
- cinst.rate = 45000; // default rate
- cinst.pacsize = 960; // 30ms
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
-
- TEST_MUSTPASS(!codec->SetISACMaxPayloadSize(1, 100));
- MARK(); // invalid channel
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(!codec->SetISACMaxPayloadSize(0, 100));
- MARK(); // invalid size (too small)
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
-
- TEST_MUSTPASS(!codec->SetISACMaxPayloadSize(0, 610));
- MARK(); // invalid size (too large)
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INVALID_ARGUMENT);
-
- TEST_MUSTPASS(codec->SetISACMaxPayloadSize(0, 200));
- MARK(); // life is good now
- TEST_MUSTPASS(codec->SetISACMaxPayloadSize(0, 120));
- MARK();
- TEST_MUSTPASS(codec->SetISACMaxPayloadSize(0, 600));
- MARK();
-
- ANL();
- AOK();
- ANL();
-
- // set iSAC as sending codec
- // set iSAC-wb as sending codec
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
-#else
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8001, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8001));
-#endif
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
- 0, GetFilename("audio_long16.pcm"), true , true));
- cinst.channels = 1;
- TEST_LOG("Testing codec: Switch between iSAC-wb and iSAC-swb \n");
- TEST_LOG("Testing codec: iSAC wideband \n");
- strcpy(cinst.plname, "ISAC");
- cinst.pltype = 103;
- cinst.rate = -1; // default rate
- cinst.pacsize = 480; // 30ms
- cinst.plfreq = 16000;
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- SLEEP(2000);
- TEST_LOG(" : iSAC superwideband \n");
- cinst.pltype = 104;
- cinst.rate = -1; // default rate
- cinst.pacsize = 960; // 30ms
- cinst.plfreq = 32000;
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- SLEEP(2000);
- TEST_LOG(" : iSAC wideband \n");
- strcpy(cinst.plname, "ISAC");
- cinst.pltype = 103;
- cinst.rate = -1; // default rate
- cinst.pacsize = 480; // 30ms
- cinst.plfreq = 16000;
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- SLEEP(2000);
- TEST_LOG(" : iSAC superwideband \n");
- cinst.pltype = 104;
- cinst.rate = -1; // default rate
- cinst.pacsize = 960; // 30ms
- cinst.plfreq = 32000;
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- SLEEP(2000);
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
-#else
- TEST_LOG("Skipping extended iSAC API tests - "
- "WEBRTC_CODEC_ISAC not defined\n");
-#endif // #if defined(WEBRTC_CODEC_ISAC)
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
- delete ptrTransport;
-#endif
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->Terminate());
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// VoEExtendedTest::TestDtmf
-// ----------------------------------------------------------------------------
-
-int VoEExtendedTest::TestDtmf() {
- PrepareTest("Dtmf");
-
- VoEBase* voe_base_ = _mgr.BasePtr();
- VoEDtmf* dtmf = _mgr.DtmfPtr();
- VoECodec* codec = _mgr.CodecPtr();
- VoEVolumeControl* volume = _mgr.VolumeControlPtr();
-
- //#ifdef _USE_EXTENDED_TRACE_
- TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename("VoEDtmf_trace.txt")));
- TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
- kTraceStateInfo |
- kTraceWarning |
- kTraceError |
- kTraceCritical |
- kTraceApiCall |
- kTraceMemory |
- kTraceInfo));
- //#endif
- TEST_MUSTPASS(voe_base_->Init());
- TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
-
- ///////////////////////////
- // Actual test starts here
-
- // SetDtmfFeedbackStatus
- TEST(SetDtmfFeedbackStatus & GetDtmfFeedbackStatus);
- ANL();
- bool dtmfFeedback = false, dtmfDirectFeedback = true;
- TEST_MUSTPASS(dtmf->GetDtmfFeedbackStatus(dtmfFeedback,
- dtmfDirectFeedback));
- TEST_MUSTPASS(!dtmfFeedback);
- TEST_MUSTPASS(dtmfDirectFeedback);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0));
- MARK();
- SLEEP(500);
-
- TEST_MUSTPASS(dtmf->SetDtmfFeedbackStatus(false, false));
- TEST_MUSTPASS(dtmf->GetDtmfFeedbackStatus(dtmfFeedback,
- dtmfDirectFeedback));
- TEST_MUSTPASS(dtmfFeedback);
- TEST_MUSTPASS(dtmfDirectFeedback);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0));
- MARK();
- SLEEP(500);
-
- TEST_MUSTPASS(dtmf->SetDtmfFeedbackStatus(false, true));
- TEST_MUSTPASS(dtmf->GetDtmfFeedbackStatus(dtmfFeedback,
- dtmfDirectFeedback));
- TEST_MUSTPASS(dtmfFeedback);
- TEST_MUSTPASS(!dtmfDirectFeedback);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0));
- MARK();
- SLEEP(500);
-
- TEST_MUSTPASS(dtmf->SetDtmfFeedbackStatus(true, false));
- TEST_MUSTPASS(dtmf->GetDtmfFeedbackStatus(dtmfFeedback,
- dtmfDirectFeedback));
- TEST_MUSTPASS(!dtmfFeedback);
- TEST_MUSTPASS(dtmfDirectFeedback);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0));
- MARK();
- SLEEP(500);
-
- TEST_MUSTPASS(dtmf->SetDtmfFeedbackStatus(true, true));
- TEST_MUSTPASS(dtmf->GetDtmfFeedbackStatus(dtmfFeedback,
- dtmfDirectFeedback));
- TEST_MUSTPASS(!dtmfFeedback);
- TEST_MUSTPASS(!dtmfDirectFeedback);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0));
- MARK();
- SLEEP(500);
- TEST_MUSTPASS(dtmf->SetDtmfFeedbackStatus(false, false));
-
- AOK();
- ANL();
-
- // SendDtmf
- TEST(SendDtmf);
- ANL();
-
- // Fail tests
- // Event
- // the eventcode is changed to unsigned char, so -1 will be interpreted as
- // 255, 256->0
- TEST_MUSTPASS(!dtmf->SendTelephoneEvent(0, -1, false, 160, 10));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(!dtmf->SendTelephoneEvent(0, 16, false, 160, 10));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- // Length
- TEST_MUSTPASS(!dtmf->SendTelephoneEvent(0, 0, true, 99, 10));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(!dtmf->SendTelephoneEvent(0, 0, true, 60001, 10));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(!dtmf->SendTelephoneEvent(0, 20, true, -1, 10));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- // Volume
- TEST_MUSTPASS(!dtmf->SendTelephoneEvent(0, 0, true, 160, -1));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(!dtmf->SendTelephoneEvent(0, 0, true, 160, 37));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- // Without sending
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(!dtmf->SendTelephoneEvent(0, 0, true));
- MARK();
- TEST_MUSTPASS(VE_NOT_SENDING != voe_base_->LastError());
- TEST_MUSTPASS(voe_base_->StartSend(0));
-
- // Testing Dtmf out-of-band: event, length and volume
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true));
- MARK();
- SLEEP(500);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 16, true));
- MARK();
- SLEEP(500); // Flash, not audible
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true, 100, 10));
- MARK();
- SLEEP(500);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true, 400, 10));
- MARK();
- SLEEP(500);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true, 160, 0));
- MARK();
- SLEEP(500);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true, 160, 36));
- MARK();
- SLEEP(500);
-
- // Testing Dtmf inband: event, length and volume
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, false));
- MARK();
- SLEEP(500);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 15, false));
- MARK();
- SLEEP(500);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, false, 100, 10));
- MARK();
- SLEEP(500);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, false, 400, 10));
- MARK();
- SLEEP(500);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, false, 160, 0));
- MARK();
- SLEEP(500);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, false, 160, 36));
- MARK();
- SLEEP(500);
-
- // Testing other events out-of-band: event and length
- // These are not audible
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 17, true, 100, 10));
- MARK();
- SLEEP(200);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 32, true, 100, 10));
- MARK();
- SLEEP(200);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 78, true, 100, 10));
- MARK();
- SLEEP(200);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 255, true, 100, 10));
- MARK();
- SLEEP(200);
- // the minimum length is 100 for the telephoneevent
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 32, true, 100, 10));
- MARK();
- SLEEP(200);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 32, true, 1000, 10));
- MARK();
- SLEEP(1200);
-
- AOK();
- ANL();
-
- // PlayDtmfTone
- TEST(PlayDtmfTone);
- ANL();
- TEST_MUSTPASS(!dtmf->PlayDtmfTone(-1, 200, 10));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(!dtmf->PlayDtmfTone(16, 200, 10));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(!dtmf->PlayDtmfTone(0, 9, 10));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(!dtmf->PlayDtmfTone(0, 200, -1));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(!dtmf->PlayDtmfTone(0, 200, 37));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
-
- TEST_MUSTPASS(dtmf->PlayDtmfTone(0));
- MARK();
- SLEEP(500);
- // the minimum length fo the DtmfTone is 100
- TEST_MUSTPASS(dtmf->PlayDtmfTone(0, 100, 10));
- MARK();
- SLEEP(500);
- TEST_MUSTPASS(dtmf->PlayDtmfTone(0, 2000, 10));
- MARK();
- SLEEP(2300);
- TEST_MUSTPASS(dtmf->PlayDtmfTone(0, 200, 0));
- MARK();
- SLEEP(500);
- TEST_MUSTPASS(dtmf->PlayDtmfTone(0, 200, 36));
- MARK();
- SLEEP(500);
-
- AOK();
- ANL();
-
- // SetTelephoneEventDetection
- TEST(SetTelephoneEventDetection);
- ANL();
- AOK();
- ANL();
-
- // Testing sending Dtmf under VAD/CN
- TEST(SendDtmf - with VAD enabled);
- ANL();
- // Mute mic
- TEST_MUSTPASS(volume->SetInputMute(0, true));
- MARK();
- // Enable VAD
- TEST_MUSTPASS(codec->SetVADStatus(0, true));
- MARK();
- // Send Dtmf
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true, 400));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 9, true, 400));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true, 400));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 9, true, 400));
- MARK();
- SLEEP(1000);
- // Switch codec
- CodecInst ci;
-#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
- ci.channels = 1;
- ci.pacsize = 480;
- ci.plfreq = 16000;
- strcpy(ci.plname, "ISAC");
- ci.pltype = 103;
- ci.rate = -1;
-#else
- ci.pltype = 119;
- strcpy(ci.plname, "isaclc");
- ci.plfreq = 16000;
- ci.pacsize = 320;
- ci.channels = 1;
- ci.rate = 40000;
-#endif
- TEST_MUSTPASS(codec->SetSendCodec(0, ci));
- MARK();
- // Send Dtmf
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true, 400));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 9, true, 400));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 0, true, 400));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, 9, true, 400));
- MARK();
- SLEEP(1000);
- SLEEP(4000);
- // Disable VAD
- TEST_MUSTPASS(codec->SetVADStatus(0, false));
- MARK();
- // Unmute
- TEST_MUSTPASS(volume->SetInputMute(0, false));
- MARK();
-
- AOK();
- ANL();
-
- // SetSendTelephoneEventPayloadType
- TEST(SetSendTelephoneEventPayloadType);
- ANL();
- TEST_MUSTPASS(!dtmf->SetSendTelephoneEventPayloadType(0, 128));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
-
- TEST_MUSTPASS(dtmf->SetSendTelephoneEventPayloadType(0, 96));
- MARK();
- TEST_MUSTPASS(dtmf->SetSendTelephoneEventPayloadType(0, 127));
- MARK();
- TEST_MUSTPASS(dtmf->SetSendTelephoneEventPayloadType(0, 106));
- MARK(); // restore default
-
- AOK();
- ANL();
-
-#ifdef WEBRTC_DTMF_DETECTION
- TEST(RegisterTelephoneEventDetection - several channels); ANL();
-
- ci.channels = 1;
- ci.pacsize = 160;
- ci.plfreq = 8000;
- ci.pltype = 0;
- ci.rate = 64000;
- strcpy(ci.plname, "PCMU");
- TEST_MUSTPASS(codec->SetSendCodec(0, ci));
-
- int ch2 = voe_base_->CreateChannel();
- TEST_MUSTPASS(voe_base_->SetSendDestination(ch2, 8002, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(ch2, 8002));
- TEST_MUSTPASS(voe_base_->StartReceive(ch2));
- TEST_MUSTPASS(codec->SetSendCodec(ch2, ci));
- TEST_MUSTPASS(voe_base_->StartPlayout(ch2));
- TEST_MUSTPASS(voe_base_->StartSend(ch2));
- MARK();
-
- DtmfCallback *d = new DtmfCallback();
- TEST_MUSTPASS(dtmf->SetDtmfFeedbackStatus(false));
-
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
-
- // In-band
- TEST_MUSTPASS(dtmf->RegisterTelephoneEventDetection(0, kInBand, *d));
- TEST_MUSTPASS(dtmf->RegisterTelephoneEventDetection(ch2, kInBand, *d));
- TEST_LOG("\nSending in-band telephone events:");
- for(int i = 0; i < 16; i++)
- {
- TEST_LOG("\n %d ", i); fflush(NULL);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, i, false, 160, 10));
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(ch2, i, false, 160, 10));
- SLEEP(500);
- }
- TEST_LOG("\nDetected %d events \n", d->counter);
- TEST_MUSTPASS(d->counter != 32);
- TEST_MUSTPASS(dtmf->DeRegisterTelephoneEventDetection(0));
- TEST_MUSTPASS(dtmf->DeRegisterTelephoneEventDetection(ch2));
-
- // Out-of-band
- d->counter = 0;
- TEST_MUSTPASS(dtmf->RegisterTelephoneEventDetection(0, kOutOfBand, *d));
- TEST_MUSTPASS(dtmf->RegisterTelephoneEventDetection(ch2, kOutOfBand, *d));
- TEST_LOG("\nSending out-band telephone events:");
- for(int i = 0; i < 16; i++)
- {
- TEST_LOG("\n %d ", i); fflush(NULL);
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(0, i, true, 160, 10));
- TEST_MUSTPASS(dtmf->SendTelephoneEvent(ch2, i, true, 160, 10));
- SLEEP(500);
- }
- TEST_LOG("\nDetected %d events \n", d->counter);
- TEST_MUSTPASS(d->counter != 32);
- TEST_MUSTPASS(dtmf->DeRegisterTelephoneEventDetection(0));
- TEST_MUSTPASS(dtmf->DeRegisterTelephoneEventDetection(ch2));
- delete d;
-
- AOK(); ANL();
-#endif
-
- TEST_MUSTPASS(dtmf->SetDtmfFeedbackStatus(true, false));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->Terminate());
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// VoEExtendedTest::TestEncryption
-// ----------------------------------------------------------------------------
-
-int VoEExtendedTest::TestEncryption() {
- PrepareTest("Encryption");
-
- VoEBase* voe_base_ = _mgr.BasePtr();
- VoEFile* file = _mgr.FilePtr();
- VoEEncryption* encrypt = _mgr.EncryptionPtr();
-
-#ifdef _USE_EXTENDED_TRACE_
- TEST_MUSTPASS(VoiceEngine::SetTraceFile(
- GetFilename("VoEEncryption_trace.txt")));
- TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
- kTraceStateInfo |
- kTraceWarning |
- kTraceError |
- kTraceCritical |
- kTraceApiCall |
- kTraceMemory |
- kTraceInfo));
-#endif
- TEST_MUSTPASS(voe_base_->Init());
- TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
- true, true));
-
- ///////////////////////////
- // Actual test starts here
-
- unsigned char key1[30] = { 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4, 5, 6,
- 7, 8, 9, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9 };
-
-#ifdef WEBRTC_SRTP
- unsigned char key2[30]; // Different than key1 in first position
- memcpy(key2, key1, 30);
- key2[0] = 99;
- unsigned char key3[30]; // Different than key1 in last position
- memcpy(key3, key1, 30);
- key3[29] = 99;
- unsigned char key4[29]; // Same as key1 but shorter
- memcpy(key4, key1, 29);
-
- TEST(SRTP - Fail tests); ANL();
-
- // Send
- // Incorrect parameters when not all protection is enabled
- TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherNull, 30, kAuthHmacSha1,
- 20, 4, kNoProtection, key1));
- TEST_MUSTPASS(VE_SRTP_ERROR != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherNull, 30, kAuthHmacSha1,
- 20, 4, kEncryption key1));
- TEST_MUSTPASS(VE_SRTP_ERROR != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherNull, 30, kAuthHmacSha1,
- 20, 4, kAuthentication, key1));
- TEST_MUSTPASS(VE_SRTP_ERROR != voe_base_->LastError());
- MARK();
- // Incorrect cipher key length
- TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 15,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication, key1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 257,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication, key1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherNull, 15, kAuthHmacSha1,
- 20, 4, kEncryptionAndAuthentication,
- key1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherNull, 257, kAuthHmacSha1,
- 20, 4, kEncryptionAndAuthentication,
- key1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- // Incorrect auth key length
- TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 21, 4,
- kEncryptionAndAuthentication, key1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthNull, 257, 4,
- kEncryptionAndAuthentication, key1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- // Incorrect auth tag length
- TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 21,
- kEncryptionAndAuthentication, key1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthNull, 20, 13,
- kEncryptionAndAuthentication, key1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
-
- // key NULL pointer
- TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication, NULL));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
-
- // Same for receive
- // Incorrect parameters when not all protection is enabled
- TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherNull, 30, kAuthHmacSha1,
- 20, 4, kNoProtection, key1));
- TEST_MUSTPASS(VE_SRTP_ERROR != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherNull, 30, kAuthHmacSha1,
- 20, 4, kEncryption key1));
- TEST_MUSTPASS(VE_SRTP_ERROR != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherNull, 30, kAuthHmacSha1,
- 20, 4, kAuthentication, key1));
- TEST_MUSTPASS(VE_SRTP_ERROR != voe_base_->LastError());
- MARK();
- // Incorrect cipher key length
- TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 15,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication,
- key1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 257,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication,
- key1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherNull, 15,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication,
- key1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherNull, 257,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication,
- key1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- // Incorrect auth key length
- TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode,
- 30, kAuthHmacSha1, 21, 4,
- kEncryptionAndAuthentication,
- key1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- // it crashed the application
- TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthNull, 257, 4,
- kEncryptionAndAuthentication,
- key1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- // Incorrect auth tag length
- TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 21,
- kEncryptionAndAuthentication,
- key1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- // it crashed the application
- TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthNull, 20, 13,
- kEncryptionAndAuthentication,
- key1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- // key NULL pointer
- TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication,
- NULL));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- ANL();
-
- TEST(SRTP - Should hear audio at all time); ANL();
-
- // Authentication only
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherNull, 0, kAuthHmacSha1, 20,
- 4, kAuthentication, key1));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherNull, 0, kAuthHmacSha1,
- 20, 4, kAuthentication, key1));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- MARK(); SLEEP(2000);
- ANL();
-
- // No protection
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherNull, 0, kAuthNull, 0, 0,
- kNoProtection, key1));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherNull, 0, kAuthNull, 0, 0,
- kNoProtection, key1));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- MARK(); SLEEP(2000);
-
- // Encryption only
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthNull, 0, 0, kEncryption key1));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthNull, 0, 0,
- kEncryption key1));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- MARK(); SLEEP(2000);
-
- // Authentication only
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherNull, 0, kAuthHmacSha1, 20,
- 4, kAuthentication, key1));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherNull, 0, kAuthHmacSha1,
- 20, 4, kAuthentication, key1));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- MARK(); SLEEP(2000);
- ANL();
-
- // Switching between keys
- TEST(SRTP - Different keys - should hear audio at all time); ANL();
-
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication, key2));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication,
- key2));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication, key1));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication,
- key1));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication, key1));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication,
- key1));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication, key2));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication,
- key2));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
- true, true));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- MARK(); SLEEP(2000);
- ANL();
-
- // Testing different keys that should be silent
- TEST(SRTP - Should be silent or garbage); ANL();
-
- // key1 and key2
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication, key1));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication,
- key2));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication, key2));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication,
- key1));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthNull, 0, 0, kEncryption key1));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthNull, 0, 0,
- kEncryption key2));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherNull, 0, kAuthHmacSha1,
- 20, 4, kAuthentication, key1));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherNull, 0, kAuthHmacSha1,
- 20, 4, kAuthentication, key2));
- MARK(); SLEEP(2000);
-
- // key1 and key3
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication, key1));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication,
- key3));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication, key3));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication,
- key1));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthNull, 0, 0, kEncryption key1));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthNull, 0, 0,
- kEncryption key3));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherNull, 0, kAuthHmacSha1, 20,
- 4, kAuthentication, key1));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherNull, 0, kAuthHmacSha1,
- 20, 4, kAuthentication, key3));
- MARK(); SLEEP(2000);
-
- // key1 and key4
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication, key1));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication,
- key4));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication, key4));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1, 20, 4,
- kEncryptionAndAuthentication,
- key1));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthNull, 0, 0, kEncryption key1));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthNull, 0, 0,
- kEncryption key4));
- MARK(); SLEEP(2000);
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherNull, 0, kAuthHmacSha1, 20,
- 4, kAuthentication, key1));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherNull, 0, kAuthHmacSha1,
- 20, 4, kAuthentication, key4));
- MARK(); SLEEP(2000);
- ANL();
-
- // Back to normal
- TEST(SRTP - Back to normal - should hear audio); ANL();
-
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- MARK(); SLEEP(2000);
- ANL();
-
- // SRTCP tests
- TEST(SRTCP - Ignore voice or not); ANL();
- VoERTP_RTCP* rtp_rtcp = _mgr.RTP_RTCPPtr();
- char tmpStr[32];
-
- // First test that RTCP packet is received and OK without encryption
-
- TEST_MUSTPASS(rtp_rtcp->SetRTCP_CNAME(0, "Henrik1"));
- MARK(); SLEEP(8000);
- TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCP_CNAME(0, tmpStr));
- TEST_MUSTPASS(_stricmp("Henrik1", tmpStr));
-
- // Enable SRTP and SRTCP send and receive
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1,
- 20, 4, kEncryptionAndAuthentication, key1, true));
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1,
- 20, 4, kEncryptionAndAuthentication, key1, true));
- TEST_MUSTPASS(rtp_rtcp->SetRTCP_CNAME(0, "Henrik2"));
- MARK(); SLEEP(8000);
- TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCP_CNAME(0, tmpStr));
- TEST_MUSTPASS(_stricmp("Henrik2", tmpStr));
-
- // Disable SRTP and SRTCP send
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(rtp_rtcp->SetRTCP_CNAME(0, "Henrik3"));
- MARK(); SLEEP(8000);
- TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCP_CNAME(0, tmpStr));
- TEST_MUSTPASS(_stricmp("Henrik2", tmpStr)); // Should not have changed
-
- // Enable SRTP send, but disable SRTCP send
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1,
- 20, 4, kEncryptionAndAuthentication, key1));
- TEST_MUSTPASS(rtp_rtcp->SetRTCP_CNAME(0, "Henrik4"));
- MARK(); SLEEP(8000);
- TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCP_CNAME(0, tmpStr));
- TEST_MUSTPASS(_stricmp("Henrik2", tmpStr)); // Should not have changed
-
- // Enable SRTP and SRTCP send, disable SRTP and SRTCP receive
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->EnableSRTPSend(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1,
- 20, 4, kEncryptionAndAuthentication, key1, true));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- TEST_MUSTPASS(rtp_rtcp->SetRTCP_CNAME(0, "Henrik5"));
- MARK(); SLEEP(8000);
- TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCP_CNAME(0, tmpStr));
- TEST_MUSTPASS(_stricmp("Henrik2", tmpStr)); // Should not have changed
-
- // Enable SRTP receive, but disable SRTCP receive
- TEST_MUSTPASS(encrypt->EnableSRTPReceive(0, kCipherAes128CounterMode, 30,
- kAuthHmacSha1,
- 20, 4, kEncryptionAndAuthentication, key1));
- TEST_MUSTPASS(rtp_rtcp->SetRTCP_CNAME(0, "Henrik6"));
- MARK(); SLEEP(8000);
- TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCP_CNAME(0, tmpStr));
- TEST_MUSTPASS(_stricmp("Henrik2", tmpStr)); // Should not have changed
-
- // Disable all
- TEST_MUSTPASS(encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(encrypt->DisableSRTPReceive(0));
- TEST_MUSTPASS(rtp_rtcp->SetRTCP_CNAME(0, "Henrik7"));
- MARK(); SLEEP(8000);
- TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCP_CNAME(0, tmpStr));
- TEST_MUSTPASS(_stricmp("Henrik7", tmpStr));
- ANL();
-
-#else
- TEST(SRTP disabled - Fail tests);
- ANL();
-
- TEST_MUSTPASS(!encrypt->EnableSRTPSend(0, kCipherNull, 30, kAuthHmacSha1,
- 20, 4, kEncryptionAndAuthentication, key1));
- TEST_MUSTPASS(VE_FUNC_NOT_SUPPORTED != voe_base_->LastError());
- TEST_MUSTPASS(!encrypt->EnableSRTPReceive(0, kCipherNull, 30, kAuthHmacSha1,
- 20, 4, kEncryptionAndAuthentication, key1));
- TEST_MUSTPASS(VE_FUNC_NOT_SUPPORTED != voe_base_->LastError());
- TEST_MUSTPASS(!encrypt->DisableSRTPSend(0));
- TEST_MUSTPASS(VE_FUNC_NOT_SUPPORTED != voe_base_->LastError());
- TEST_MUSTPASS(!encrypt->DisableSRTPReceive(0));
- TEST_MUSTPASS(VE_FUNC_NOT_SUPPORTED != voe_base_->LastError());
- ANL();
-#endif
- AOK();
-
- TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->Terminate());
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// VoEExtendedTest::TestExternalMedia
-// ----------------------------------------------------------------------------
-
-int VoEExtendedTest::TestExternalMedia() {
- PrepareTest("VoEExternalMedia");
-
- VoEBase* voe_base_ = _mgr.BasePtr();
- VoEExternalMedia* xmedia = _mgr.ExternalMediaPtr();
-
- // check if this interface is supported
- if (!xmedia) {
- TEST_LOG("VoEExternalMedia is not supported!");
- return -1;
- }
-
-#ifdef _USE_EXTENDED_TRACE_
- TEST_MUSTPASS(VoiceEngine::SetTraceFile(
- GetFilename("VoEExternalMedia_trace.txt")));
- TEST_MUSTPASS(VoiceEngine::SetTraceFilter(
- kTraceStateInfo | kTraceStateInfo | kTraceWarning |
- kTraceError | kTraceCritical | kTraceApiCall |
- kTraceMemory | kTraceInfo));
-#endif
- TEST_MUSTPASS(voe_base_->Init());
- TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
-
- int getLen = 0;
- WebRtc_Word16 vector[32000];
- memset(vector, 0, 32000 * sizeof(short));
-
-#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
-
- // ExternalPlayoutGetData
- TEST(ExternalPlayoutGetData);
- ANL();
-
- TEST_MUSTPASS(!xmedia->SetExternalPlayoutStatus(true));
- TEST_MUSTPASS(VE_ALREADY_SENDING != voe_base_->LastError());
- TEST_MUSTPASS(!xmedia->ExternalPlayoutGetData(vector, 16000, 100, getLen));
- TEST_MUSTPASS(VE_INVALID_OPERATION != voe_base_->LastError());
-
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(xmedia->SetExternalPlayoutStatus(true));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
-
- TEST_MUSTPASS(xmedia->ExternalPlayoutGetData(vector, 48000, 0, getLen));
- TEST_MUSTPASS(480 != getLen);
- SLEEP(10);
- TEST_MUSTPASS(xmedia->ExternalPlayoutGetData(vector, 16000, 3000, getLen));
- TEST_MUSTPASS(160 != getLen);
- SLEEP(10);
-
- TEST_MUSTPASS(!xmedia->ExternalPlayoutGetData(vector, 8000, 100, getLen));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(!xmedia->ExternalPlayoutGetData(vector, 16000, -1, getLen));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
-
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(xmedia->SetExternalPlayoutStatus(false));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
-
- // SetExternalRecording
- TEST(SetExternalRecording);
- ANL();
-
- TEST_MUSTPASS(!xmedia->SetExternalRecordingStatus(true));
- TEST_MUSTPASS(VE_ALREADY_SENDING != voe_base_->LastError());
- TEST_MUSTPASS(!xmedia->ExternalRecordingInsertData(vector, 160, 16000, 20));
- TEST_MUSTPASS(VE_INVALID_OPERATION != voe_base_->LastError());
-
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(xmedia->SetExternalRecordingStatus(true));
- TEST_MUSTPASS(voe_base_->StartSend(0));
-
- TEST_MUSTPASS(xmedia->ExternalRecordingInsertData(vector, 480, 48000, 0));
- SLEEP(10);
- TEST_MUSTPASS(xmedia->ExternalRecordingInsertData(vector, 640, 16000, 0));
- SLEEP(40);
-
- TEST_MUSTPASS(!xmedia->ExternalRecordingInsertData(vector, 160, 16000, -1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(!xmedia->ExternalRecordingInsertData(vector, 80, 8000, 20));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(!xmedia->ExternalRecordingInsertData(vector, 0, 16000, 20));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(!xmedia->ExternalRecordingInsertData(vector, 80, 16000, 20));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(!xmedia->ExternalRecordingInsertData(vector, 500, 16000, 20));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
-
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(xmedia->SetExternalRecordingStatus(false));
- TEST_MUSTPASS(voe_base_->StartSend(0));
-
-#else // #ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
- TEST_MUSTPASS(!xmedia->SetExternalPlayoutStatus(true));
- TEST_MUSTPASS(VE_FUNC_NOT_SUPPORTED != voe_base_->LastError());
- TEST_MUSTPASS(!xmedia->ExternalPlayoutGetData(vector, 16000, 100, getLen));
- TEST_MUSTPASS(VE_FUNC_NOT_SUPPORTED != voe_base_->LastError());
- TEST_MUSTPASS(!xmedia->SetExternalRecordingStatus(true));
- TEST_MUSTPASS(VE_FUNC_NOT_SUPPORTED != voe_base_->LastError());
- TEST_MUSTPASS(!xmedia->ExternalRecordingInsertData(vector, 160, 16000, 20));
- TEST_MUSTPASS(VE_FUNC_NOT_SUPPORTED != voe_base_->LastError());
-
-#endif // #ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->Terminate());
-
- ANL();
- AOK();
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// VoEExtendedTest::TestFile
-// ----------------------------------------------------------------------------
-
-int VoEExtendedTest::TestFile() {
- PrepareTest("File");
-
- VoEBase* voe_base_ = _mgr.BasePtr();
- VoEFile* file = _mgr.FilePtr();
- VoECodec* codec = _mgr.CodecPtr();
-
-#ifdef _USE_EXTENDED_TRACE_
- TEST_MUSTPASS(VoiceEngine::SetTraceFile(
- GetFilename("VoEFile_trace.txt"))); MARK();
- TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
- kTraceStateInfo |
- kTraceWarning |
- kTraceError |
- kTraceCritical |
- kTraceApiCall |
- kTraceMemory |
- kTraceInfo));
-#endif
-
- TEST_MUSTPASS(voe_base_->Init());
- TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
-
- ///////////////////////////
- // Actual test starts here
-
- const int dT(100);
-
- TEST(StartPlayingFileLocally);
- ANL();
- TEST(StopPlayingFileLocally);
- ANL();
-
- voe_base_->StopPlayout(0);
- TEST_MUSTPASS(file->StartPlayingFileLocally(
- 0, GetResource("audio_long16.pcm")));MARK();
- voe_base_->StartPlayout(0);
- MARK(); // file should be mixed in and played out
- SLEEP(dT);
- TEST_MUSTPASS(!file->StartPlayingFileLocally(
- 0, GetResource("audio_long16.pcm")));
- MARK(); // should fail (must stop first)
- TEST_MUSTPASS(voe_base_->LastError() != VE_ALREADY_PLAYING);
- TEST_MUSTPASS(file->StopPlayingFileLocally(0));
- MARK();
- TEST_MUSTPASS(file->StartPlayingFileLocally(
- 0, GetResource("audio_long16.pcm")));
- MARK(); // should work again (restarts file)
- SLEEP(dT);
- TEST_MUSTPASS(file->StopPlayingFileLocally(0));
- MARK();
- TEST_MUSTPASS(file->StartPlayingFileLocally(
- 0, GetResource("audio_long16.pcm"), false, kFileFormatPcm16kHzFile));
- MARK();
- SLEEP(dT);
- TEST_MUSTPASS(file->StopPlayingFileLocally(0));
- MARK();
- TEST_MUSTPASS(file->StartPlayingFileLocally(
- 0, GetResource("audio_long8.pcm"), false, kFileFormatPcm8kHzFile));
- MARK();
- SLEEP(dT);
- TEST_MUSTPASS(file->StopPlayingFileLocally(0));
- MARK();
- TEST_MUSTPASS(file->StartPlayingFileLocally(
- 0, GetResource("audio_long16.wav"), false, kFileFormatPcm8kHzFile));
- MARK();
- SLEEP(dT);
- TEST_MUSTPASS(file->StopPlayingFileLocally(0));
- MARK();
- TEST_MUSTPASS(file->StartPlayingFileLocally(
- 0, GetResource("audio_long8mulaw.wav"), false,
- kFileFormatPcm8kHzFile));
- MARK();
- SLEEP(dT);
-
- // add compressed tests here...
-
- // TEST_MUSTPASS(file->StopPlayingFileLocally(0)); MARK();
- // TEST_MUSTPASS(file->StartPlayingFileLocally(
- // 0, GetResource("audio_short16.pcm"), true,
- // kFileFormatPcm16kHzFile)); MARK(); // loop
- TEST_MUSTPASS(file->StopPlayingFileLocally(0));
- MARK();
- TEST_MUSTPASS(file->StartPlayingFileLocally(
- 0, GetResource("audio_short16.pcm"), false,
- kFileFormatPcm16kHzFile, 1.0, 0, 2000));
- MARK(); // play segment
- SLEEP(2500);
- TEST_MUSTPASS(file->StopPlayingFileLocally(0));
- MARK();
- TEST_MUSTPASS(!file->StartPlayingFileLocally(
- 0, GetResource("audio_short16.pcm"), false,
- kFileFormatPcm16kHzFile, 1.0, 2000, 1000));
- MARK(); // invalid segment
- TEST_MUSTPASS(voe_base_->LastError() != VE_BAD_FILE);
- TEST_MUSTPASS(!file->StartPlayingFileLocally(
- 0, GetResource("audio_short16.pcm"), false,
- kFileFormatPcm16kHzFile, 1.0, 21000, 30000));
- MARK(); // start > file size
- TEST_MUSTPASS(voe_base_->LastError() != VE_BAD_FILE);
- TEST_MUSTPASS(!file->StartPlayingFileLocally(
- 0, GetResource("audio_short16.pcm"), false,
- kFileFormatPcm16kHzFile, 1.0, 100, 100));
- MARK(); // invalid segment
- TEST_MUSTPASS(voe_base_->LastError() != VE_BAD_FILE);
- TEST_MUSTPASS(file->StartPlayingFileLocally(
- 0, GetResource("audio_long16.pcm")));
- MARK(); // should work again (restarts file)
- TEST_MUSTPASS(file->StopPlayingFileLocally(0));
- MARK();
- TEST_MUSTPASS(!file->StartPlayingFileLocally(0, (InStream*)NULL));
- MARK(); // just do it
- TEST_MUSTPASS(voe_base_->LastError() != VE_BAD_FILE);
-
- AOK();
- ANL();
-
- TEST(IsPlayingFileLocally);
- ANL();
-
- TEST_MUSTPASS(0 != file->IsPlayingFileLocally(0));
- MARK(); // inactive
- TEST_MUSTPASS(file->StartPlayingFileLocally(
- 0, GetResource("audio_long16.pcm")));
- MARK();
- TEST_MUSTPASS(1 != file->IsPlayingFileLocally(0));
- MARK(); // active
- AOK();
- ANL();
-
- TEST(ScaleLocalFilePlayout);
- ANL();
- TEST_MUSTPASS(file->ScaleLocalFilePlayout(0, 1.0));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(file->ScaleLocalFilePlayout(0, 0.0));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(file->ScaleLocalFilePlayout(0, 0.5));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(file->ScaleLocalFilePlayout(0, 0.25));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(file->StopPlayingFileLocally(0));
- MARK();
- AOK();
- ANL();
-
- // Replace microphone with file and play out on remote side
- // All channels, per channel
- // Different mixing frequencies
- TEST(StartPlayingFileAsMicrophone);
- ANL();
- TEST(IsPlayingFileAsMicrophone);
- ANL();
- TEST(ScaleFileAsMicrophonePlayout);
- ANL();
- CodecInst tempCodec;
- for (int ch = -1; ch < 1; ++ch) // Channel -1 and 0
- {
- TEST_LOG("Testing channel = %d \n", ch);
- for (int fs = 1; fs < 4; ++fs) // nb, wb and swb codecs
- {
- switch (fs) {
- case 1: // nb
- TEST_LOG("Testing with nb codec \n");
- tempCodec.channels = 1;
- tempCodec.pacsize = 160;
- tempCodec.plfreq = 8000;
- strcpy(tempCodec.plname, "PCMU");
- tempCodec.pltype = 0;
- tempCodec.rate = 64000;
- break;
- case 2: // wb
-#ifdef WEBRTC_CODEC_ISAC
- TEST_LOG("Testing with wb codec \n");
- tempCodec.channels = 1;
- tempCodec.pacsize = 480;
- tempCodec.plfreq = 16000;
- strcpy(tempCodec.plname, "ISAC");
- tempCodec.pltype = 103;
- tempCodec.rate = 32000;
- break;
-#else
- TEST_LOG("NOT testing with wb codec - "
- "WEBRTC_CODEC_ISAC not defined \n");
- continue;
-#endif
- case 3: // swb
-#ifdef WEBRTC_CODEC_PCM16
- TEST_LOG("Testing with swb codec \n");
- tempCodec.channels = 1;
- tempCodec.pacsize = 640;
- tempCodec.plfreq = 32000;
- strcpy(tempCodec.plname, "L16");
- tempCodec.pltype = 125;
- tempCodec.rate = 512000;
- break;
-#else
- TEST_LOG("NOT testing with swb codec -"
- " WEBRTC_CODEC_PCM16 not defined \n");
- continue;
-#endif
- }
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(codec->SetRecPayloadType(0, tempCodec));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(codec->SetSendCodec(0, tempCodec));
-
- TEST_LOG("File 1 in 16 kHz no mix, 2 in 16 kHz mix,"
- " 3 in 8 kHz no mix, 4 in 8 kHz mix \n");
-
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
- ch, GetResource("audio_long16.pcm")));
- MARK(); // don't mix
- SLEEP(2000);
- TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(ch));
- MARK();
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
- ch, GetResource("audio_long16.wav"), false, true,
- kFileFormatWavFile));
- MARK(); // mix
- SLEEP(2000);
- TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(ch));
- MARK();
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
- ch, GetResource("audio_long8.pcm"), false, false,
- kFileFormatPcm8kHzFile));
- MARK(); // don't mix
- SLEEP(2000);
- TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(ch));
- MARK();
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
- ch, GetResource("audio_long8.pcm"), false, true,
- kFileFormatPcm8kHzFile));
- MARK(); // mix
- SLEEP(2000);
- TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(ch));
- MARK();
- TEST_MUSTPASS(!file->StartPlayingFileAsMicrophone(
- ch, (InStream*)NULL));
- MARK(); // force error
- AOK();
- ANL();
-
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
- ch, GetResource("audio_long16.pcm")));
- TEST_MUSTPASS(1 != file->IsPlayingFileAsMicrophone(ch));
- TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(ch));
- TEST_MUSTPASS(0 != file->IsPlayingFileAsMicrophone(ch));
- AOK();
- ANL();
-
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
- ch, GetResource("audio_long16.pcm")));
- TEST_MUSTPASS(file->ScaleFileAsMicrophonePlayout(ch, 1.0));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(file->ScaleFileAsMicrophonePlayout(ch, 0.5));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(file->ScaleFileAsMicrophonePlayout(ch, 0.25));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(file->ScaleFileAsMicrophonePlayout(ch, 0.0));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(ch));
- MARK();
- AOK();
- ANL();
- }
- }
-
- // Record speaker signal to file
-
- CodecInst fcomp = { 0, "L16", 8000, 80, 1, 128000 };
-
- TEST(StartRecordingPlayout);
- ANL();
- TEST(StopRecordingPlayout);
- ANL();
-
- TEST_MUSTPASS(file->StartRecordingPlayout(0,
- GetFilename("rec_play16.pcm")));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(file->StopRecordingPlayout(0));
- MARK();
-
- fcomp.plfreq = 8000;
- strcpy(fcomp.plname, "L16");
- TEST_MUSTPASS(file->StartRecordingPlayout(0, GetFilename("rec_play8.wav"),
- &fcomp));
- SLEEP(1000);
- TEST_MUSTPASS(file->StopRecordingPlayout(0));
- MARK();
-
- fcomp.plfreq = 16000;
- strcpy(fcomp.plname, "L16");
- TEST_MUSTPASS(file->StartRecordingPlayout(0, GetFilename("rec_play16.wav"),
- &fcomp));
- SLEEP(1000);
- TEST_MUSTPASS(file->StopRecordingPlayout(0));
- MARK();
-
- fcomp.pltype = 0;
- fcomp.plfreq = 8000;
- strcpy(fcomp.plname, "PCMU");
- fcomp.rate = 64000;
- fcomp.pacsize = 160;
- fcomp.channels = 1;
-
- TEST_MUSTPASS(file->StartRecordingPlayout(0,
- GetFilename("rec_play_pcmu.wav"),
- &fcomp));
- SLEEP(1000);
- TEST_MUSTPASS(file->StopRecordingPlayout(0));
- MARK();
-
- fcomp.pltype = 8;
- fcomp.plfreq = 8000;
- strcpy(fcomp.plname, "PCMA");
- TEST_MUSTPASS(file->StartRecordingPlayout(0,
- GetFilename("rec_play_pcma.wav"),
- &fcomp));
- SLEEP(1000);
- TEST_MUSTPASS(file->StopRecordingPlayout(0));
- MARK();
-
- fcomp.pltype = 97;
- fcomp.pacsize = 240;
- fcomp.rate = 13300;
- fcomp.plfreq = 8000;
- strcpy(fcomp.plname, "ILBC");
- TEST_MUSTPASS(file->StartRecordingPlayout(0,
- GetFilename("rec_play.ilbc"),
- &fcomp));
- SLEEP(1000);
- TEST_MUSTPASS(file->StopRecordingPlayout(0));
- MARK();
-
- TEST_MUSTPASS(file->StartRecordingPlayout(
- -1, GetFilename("rec_play16_mixed.pcm")));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(file->StopRecordingPlayout(-1));
- MARK();
-
- // TEST_MUSTPASS(file->StopPlayingFileLocally(0)); // Why should this work?
- TEST_LOG("\nplaying out...\n");
- TEST_MUSTPASS(file->StartPlayingFileLocally(
- 0, GetFilename("rec_play.ilbc"), false, kFileFormatCompressedFile));
- MARK();
- SLEEP(2000);
-
- AOK();
- ANL();
-
- // Record microphone signal to file
- TEST(StartRecordingMicrophone);
- ANL();
- TEST(StopRecordingMicrophone);
- ANL();
-
- TEST_MUSTPASS(file->StartRecordingMicrophone(GetFilename("rec_mic16.pcm")));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(file->StopRecordingMicrophone());
- MARK();
-
- voe_base_->StopSend(0);
- TEST_MUSTPASS(file->StartRecordingMicrophone(GetFilename("rec_mic16.pcm")));
- MARK(); // record without sending as well
- SLEEP(1000);
- TEST_MUSTPASS(file->StopRecordingMicrophone());
- MARK();
- voe_base_->StartSend(0); // restore sending
-
- fcomp.plfreq = 8000;
- strcpy(fcomp.plname, "L16");
- TEST_MUSTPASS(file->StartRecordingMicrophone(
- GetFilename("rec_play8.wav"), &fcomp));
- SLEEP(1000);
- TEST_MUSTPASS(file->StopRecordingMicrophone());
- MARK();
-
- fcomp.plfreq = 16000;
- strcpy(fcomp.plname, "L16");
- TEST_MUSTPASS(file->StartRecordingMicrophone(
- GetFilename("rec_play16.wav"), &fcomp));
- SLEEP(1000);
- TEST_MUSTPASS(file->StopRecordingMicrophone());
- MARK();
-
- // FT#1810, the following test is to make sure StartRecordingCall will
- // record both mic and file
- TEST_LOG("StartRecordingCall, record both mic and file in specific"
- " channels \n");
- TEST_LOG("Create maxnumofchannels \n");
- for (int i = 1; i < voe_base_->MaxNumOfChannels(); i++) {
- int ch = voe_base_->CreateChannel();
- TEST_MUSTPASS(ch == -1);
- TEST_MUSTPASS(voe_base_->StopPlayout(ch));
- }
-
- TEST_MUSTPASS(voe_base_->SetSendDestination(1, 12356, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(1, 12356));
- TEST_MUSTPASS(voe_base_->StartReceive(1));
- TEST_MUSTPASS(voe_base_->StopPlayout(1));
- TEST_MUSTPASS(voe_base_->StartSend(1));
- TEST_MUSTPASS(voe_base_->StartPlayout(1));
-
- TEST_LOG("ALways playing audio_long16.pcm for "
- "channel 0 in background \n");
- fcomp.plfreq = 16000;
- strcpy(fcomp.plname, "L16");
- TEST_LOG("Recording microphone to L16, please speak \n");
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
- 0, GetResource("audio_long16.pcm"), true , true));
- TEST_MUSTPASS(file->StartRecordingMicrophone(
- GetFilename("rec_play_ch.wav"), &fcomp));
- MARK();
- SLEEP(3000);
- TEST_MUSTPASS(file->StopRecordingMicrophone());
- MARK();
- TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(0));
- TEST_LOG("Playing recording file, you should only hear what you said \n");
- TEST_MUSTPASS(file->StartPlayingFileLocally(
- 0, GetFilename("rec_play_ch.wav"), false, kFileFormatWavFile));
- SLEEP(2500);
- TEST_MUSTPASS(file->StopPlayingFileLocally(0));
- TEST_LOG("Recording microphone 0 to L16, please speak \n");
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
- -1, GetResource("audio_long16.pcm"), true , true));
- TEST_MUSTPASS(file->StartRecordingMicrophone(
- GetFilename("rec_play_ch_0.wav"), &fcomp));
- MARK();
- SLEEP(3000);
- TEST_MUSTPASS(file->StopRecordingMicrophone());
- MARK();
- TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(-1));
- TEST_LOG("Playing recording file, you should hear what you said and"
- " audio_long16.pcm \n");
- TEST_MUSTPASS(file->StartPlayingFileLocally(
- 0, GetFilename("rec_play_ch_0.wav"), false, kFileFormatWavFile));
- SLEEP(2500);
- TEST_MUSTPASS(file->StopPlayingFileLocally(0));
- TEST_LOG("Recording microphone to ilbc, please speak \n");
- strcpy(fcomp.plname, "ilbc");
- fcomp.plfreq = 8000;
- fcomp.pacsize = 160;
- fcomp.rate = 15200;
- fcomp.channels = 1;
- fcomp.pltype = 97;
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(
- 0, GetResource("audio_long16.pcm"), true , true));
- TEST_MUSTPASS(file->StartRecordingMicrophone(
- GetFilename("rec_play_ch_0.ilbc"), &fcomp));
- MARK();
- SLEEP(3000);
- TEST_MUSTPASS(file->StopRecordingMicrophone());
- MARK();
- TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(0));
- TEST_LOG("Playing recording file, you should only hear what you said \n");
- TEST_MUSTPASS(file->StartPlayingFileLocally(
- 0, GetFilename("rec_play_ch_0.ilbc"), false,
- kFileFormatCompressedFile));
- SLEEP(2500);
- TEST_MUSTPASS(file->StopPlayingFileLocally(0));
- for (int i = 1; i < voe_base_->MaxNumOfChannels(); i++) {
- TEST_MUSTPASS(voe_base_->DeleteChannel(i));
- }
-
- AOK();
- ANL();
-
- // Record mixed (speaker + microphone) signal to file
-
-
-#if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
- TEST(StartRecordingSpeakerStereo);
- ANL();
- TEST(StopRecordingSpeakerStereo);
- ANL();
-
- VoEHardware* hardware = _mgr.HardwarePtr();
- TEST_MUSTPASS(NULL == hardware);
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
-#if defined(_WIN32)
- TEST_MUSTPASS(hardware->SetRecordingDevice(-1));
- TEST_MUSTPASS(hardware->SetPlayoutDevice(-1));
-#else
- TEST_MUSTPASS(hardware->SetRecordingDevice(0));
- TEST_MUSTPASS(hardware->SetPlayoutDevice(0));
-#endif
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- MARK();
-
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
-#if defined(_WIN32)
- TEST_MUSTPASS(hardware->SetRecordingDevice(-1));
- TEST_MUSTPASS(hardware->SetPlayoutDevice(-1));
-#else
- TEST_MUSTPASS(hardware->SetRecordingDevice(0));
- TEST_MUSTPASS(hardware->SetPlayoutDevice(0));
-#endif
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
-
- AOK();
- ANL();
-#else
- TEST_LOG("Skipping stereo record tests -"
- " MAC_IPHONE or WEBRTC_ANDROID is defined \n");
-#endif // #if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
- // Conversion between different file formats
-
-#if defined(MAC_IPHONE) || defined(WEBRTC_ANDROID)
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
-#endif
-
- TEST(ConvertPCMToWAV);
- ANL();
-
- TEST_MUSTPASS(file->ConvertPCMToWAV(
- GetResource("audio_long16.pcm"),
- GetFilename("singleUserDemoConv.wav")));
- MARK();
- TEST_MUSTPASS(!file->ConvertPCMToWAV((InStream*)NULL,
- (OutStream*)NULL));MARK(); // invalid stream handles
- AOK();
- ANL();
-
- TEST(ConvertWAVToPCM);
- ANL();
-
- TEST_MUSTPASS(file->ConvertWAVToPCM(
- GetResource("audio_long16.wav"),
- GetFilename("singleUserDemoConv.pcm")));
- MARK();
- TEST_MUSTPASS(!file->ConvertWAVToPCM((InStream*)NULL, (OutStream*)NULL));
- MARK(); // invalid stream handles
- AOK();
- ANL();
-
- TEST(ConvertPCMToCompressed);
- ANL();
-
- fcomp.plfreq = 16000;
- strcpy(fcomp.plname, "L16");
- TEST_MUSTPASS(!file->ConvertPCMToCompressed(
- GetResource("audio_long16.pcm"),
- GetFilename("singleUserDemoConv16_dummy.wav"), &fcomp));
- MARK(); // should not be supported
-
- fcomp.plfreq = 8000;
- strcpy(fcomp.plname, "ilbc");
- fcomp.pacsize = 160;
- fcomp.rate = 15200;
- fcomp.pltype = 97;
- fcomp.channels = 1;
- TEST_MUSTPASS(file->ConvertPCMToCompressed(
- GetResource("audio_long16.pcm"),
- GetFilename("singleUserDemoConv.ilbc"), &fcomp));MARK();
- AOK();ANL();
-
- TEST(ConvertCompressedToPCM);
- ANL();
-
- TEST_MUSTPASS(file->ConvertCompressedToPCM(
- GetFilename("singleUserDemoConv.ilbc"),
- GetFilename("singleUserDemoConv_ilbc.pcm")));MARK();
- TEST_MUSTPASS(!file->ConvertCompressedToPCM(
- GetResource("audio_long16.pcm"),
- GetFilename("singleUserDemoConv_dummy.pcm")));MARK();
- AOK();ANL();
-
-#if defined(MAC_IPHONE) || defined(WEBRTC_ANDROID)
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
-#endif
-
- // Misc file functions
- TEST(GetFileDuration);
- ANL();
-
- int dur;
-
- TEST_MUSTPASS(file->GetFileDuration(
- GetResource("audio_long16.pcm"), dur));
- TEST_MUSTPASS(file->GetFileDuration(
- GetResource("audio_long8.pcm"), dur, kFileFormatPcm8kHzFile));
- TEST_MUSTPASS(file->GetFileDuration(
- GetResource("audio_long16.pcm"), dur, kFileFormatPcm16kHzFile));
- TEST_MUSTPASS(file->GetFileDuration(
- GetResource("audio_long16.wav"), dur, kFileFormatPcm8kHzFile));
- TEST_MUSTPASS(file->GetFileDuration(
- GetFilename("singleUserDemoConv.ilbc"), dur,
- kFileFormatCompressedFile));
-
- AOK();
- ANL();
-
- TEST(GetPlaybackPosition);
- ANL();
-
- int pos;
-
- TEST_MUSTPASS(file->StartPlayingFileLocally(
- 0, GetResource("audio_long16.pcm")));
- SLEEP(1000);
- TEST_MUSTPASS(file->GetPlaybackPosition(0, pos));
- MARK(); // position should be ~1000
- SLEEP(1000);
- TEST_MUSTPASS(file->GetPlaybackPosition(0, pos));
- MARK(); // position should be ~2000
- // SLEEP(70*1000);
- // file is no longer playing
- // TEST_MUSTPASS(file->GetPlaybackPosition(0, pos)); MARK();
- TEST_MUSTPASS(file->StopPlayingFileLocally(0));
- AOK();
- ANL();
-
- // These tests are related to defect 5136
- // They play .wav files with different sample freq for 5s
- char localFiles[7][50] = { "audio_tiny8.wav", "audio_tiny11.wav",
- "audio_tiny16.wav", "audio_tiny22.wav", "audio_tiny32.wav",
- "audio_tiny44.wav", "audio_tiny48.wav" };
- char freq[7][5] = { "8", "11", "16", "22", "32", "44.1", "48" };
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- for (int i = 0; i < 7; i++) {
- TEST_LOG("Playing file %s, in %s KHz \n", localFiles[i], freq[i]);
- TEST_MUSTPASS(file->StartPlayingFileLocally(
- 0, GetResource(localFiles[i]),false, kFileFormatWavFile, 1));
- SLEEP(4500); // The file should not end
- TEST_MUSTPASS(file->StopPlayingFileLocally(0));
- }
-
- // TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(0)); // Should not work
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->Terminate());
-
- AOK();
- ANL();
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// VoEExtendedTest::TestMixing
-// ----------------------------------------------------------------------------
-
-// Creates and mixes |num_remote_channels| which play a file "as microphone"
-// with |num_local_channels| which play a file "locally", using a constant
-// amplitude of |input_value|.
-//
-// The mixed output is verified to always fall between |max_output_value| and
-// |min_output_value|, after a startup phase.
-int VoEExtendedTest::RunMixingTest(int num_remote_channels,
- int num_local_channels,
- int16_t input_value,
- int16_t max_output_value,
- int16_t min_output_value) {
- VoEBase* voe_base_ = _mgr.BasePtr();
- VoEFile* file = _mgr.FilePtr();
- VoECodec* codec = _mgr.CodecPtr();
- VoEAudioProcessing* apm = _mgr.APMPtr();
-
- // Use L16 at 16kHz to minimize distortion (file recording is 16kHz
- // and resampling will cause large distortions).
- CodecInst codec_inst;
- strcpy(codec_inst.plname, "L16");
- codec_inst.channels = 1;
- codec_inst.rate = 256000;
- codec_inst.plfreq = 16000;
- codec_inst.pltype = 105;
- codec_inst.pacsize = 160;
-
- apm->SetNsStatus(false);
- apm->SetAgcStatus(false);
- apm->SetEcStatus(false);
-
- const char file_to_generate_name[] = "dc_file.pcm";
- const char* input_filename = file_to_generate_name;
- FILE* file_to_generate = fopen(file_to_generate_name, "wb");
- ASSERT_TRUE(file_to_generate != NULL);
- for (int i = 0; i < 160 * 100 * 5; i++) {
- fwrite(&input_value, sizeof(input_value), 1, file_to_generate);
- }
- fclose(file_to_generate);
-
- TEST_MUSTPASS(voe_base_->Init());
-
- std::vector<int> local_channels(num_local_channels);
- for (int i = 0; i < num_local_channels; ++i) {
- local_channels[i] = voe_base_->CreateChannel();
- ASSERT_TRUE(local_channels[i] != -1);
- TEST_MUSTPASS(voe_base_->StartPlayout(local_channels[i]));
- TEST_MUSTPASS(file->StartPlayingFileLocally(local_channels[i],
- input_filename,
- true));
- }
-
- std::vector<int> remote_channels(num_remote_channels);
- for (int i = 0; i < num_remote_channels; ++i) {
- remote_channels[i] = voe_base_->CreateChannel();
- ASSERT_TRUE(remote_channels[i] != -1);
- TEST_MUSTPASS(codec->SetRecPayloadType(remote_channels[i], codec_inst));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(remote_channels[i],
- 1234 + 2 * i));
- TEST_MUSTPASS(voe_base_->SetSendDestination(remote_channels[i],
- 1234 + 2 * i,
- "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(remote_channels[i]));
- TEST_MUSTPASS(voe_base_->StartPlayout(remote_channels[i]));
- TEST_MUSTPASS(codec->SetSendCodec(remote_channels[i], codec_inst));
- TEST_MUSTPASS(voe_base_->StartSend(remote_channels[i]));
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(remote_channels[i],
- input_filename,
- true));
- }
-
- const char mix_result[] = "mix_result.pcm";
- TEST_MUSTPASS(file->StartRecordingPlayout(-1/*record meeting*/,
- mix_result));
- TEST_LOG("Playing %d remote channels.\n", num_remote_channels);
- TEST_LOG("Playing %d local channels.\n", num_local_channels);
- SLEEP(5000);
- TEST_MUSTPASS(file->StopRecordingPlayout(-1));
- TEST_LOG("Stopping\n");
-
- for (int i = 0; i < num_local_channels; ++i) {
- TEST_MUSTPASS(voe_base_->StopPlayout(local_channels[i]));
- TEST_MUSTPASS(voe_base_->DeleteChannel(local_channels[i]));
- }
-
- for (int i = 0; i < num_remote_channels; ++i) {
- TEST_MUSTPASS(voe_base_->StopSend(remote_channels[i]));
- TEST_MUSTPASS(voe_base_->StopPlayout(remote_channels[i]));
- TEST_MUSTPASS(voe_base_->StopReceive(remote_channels[i]));
- TEST_MUSTPASS(voe_base_->DeleteChannel(remote_channels[i]));
- }
-
- FILE* verification_file = fopen(mix_result, "rb");
- ASSERT_TRUE(verification_file != NULL);
- int16_t mix_value = 0;
- // Skip the first 100 ms to avoid initialization and ramping-in effects.
- ASSERT_TRUE(fseek(verification_file, sizeof(int16_t) * 1600, SEEK_SET) == 0);
- while (fread(&mix_value, sizeof(mix_value), 1, verification_file)) {
- ASSERT_TRUE(mix_value <= max_output_value)
- ASSERT_TRUE(mix_value >= min_output_value);
- }
- fclose(verification_file);
-
- return 0;
-}
-
-// TODO(andrew): move or copy these to the mixer module test when possible.
-int VoEExtendedTest::TestMixing() {
- // These tests assume a maxmium of three mixed participants. We allow a
- // +/- 10% range around the expected output level to accout for distortion
- // from coding and processing in the loopback chain.
-
- // Create four channels and make sure that only three are mixed.
- TEST_LOG("Test max-three-participant mixing.\n");
- int16_t input_value = 1000;
- int16_t expected_output = input_value * 3;
- if (RunMixingTest(4, 0, input_value, 1.1 * expected_output,
- 0.9 * expected_output) != 0) {
- return -1;
- }
-
- // Ensure the mixing saturation protection is working. We can do this because
- // the mixing limiter is given some headroom, so the expected output is less
- // than full scale.
- TEST_LOG("Test mixing saturation protection.\n");
- input_value = 20000;
- expected_output = 29204; // = -1 dBFS, the limiter headroom.
- // If this isn't satisfied, we're not testing anything.
- assert(input_value * 3 > 32767);
- assert(1.1 * expected_output < 32767);
- if (RunMixingTest(3, 0, input_value, 1.1 * expected_output,
- 0.9 * expected_output) != 0) {
- return -1;
- }
-
- // Ensure the mixing saturation protection is not applied when only using a
- // single channel.
- TEST_LOG("Test saturation protection has no effect on one channel.\n");
- input_value = 32767;
- expected_output = 32767;
- // If this isn't satisfied, we're not testing anything.
- assert(0.95 * expected_output > 29204); // = -1 dBFS, the limiter headroom.
- if (RunMixingTest(1, 0, input_value, expected_output,
- 0.95 * expected_output) != 0) {
- return -1;
- }
-
- TEST_LOG("Test combinations of 'anonymous' participants and regular "
- "participants.\n");
- input_value = 1000;
- expected_output = input_value * 2;
- if (RunMixingTest(1, 1, input_value, 1.1 * expected_output,
- 0.9 * expected_output) != 0) {
-
- return -1;
- }
-
- expected_output = input_value * 4;
- if (RunMixingTest(3, 1, input_value, 1.1 * expected_output,
- 0.9 * expected_output) != 0) {
-
- return -1;
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// VoEExtendedTest::TestHardware
-// ----------------------------------------------------------------------------
-
-int VoEExtendedTest::TestHardware() {
- PrepareTest("Hardware");
-
- VoEBase* voe_base_ = _mgr.BasePtr();
- VoEHardware* hardware = _mgr.HardwarePtr();
-
-#ifdef _USE_EXTENDED_TRACE_
- TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename(
- "VoEHardware_trace.txt")));
- TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
- kTraceStateInfo |
- kTraceWarning |
- kTraceError |
- kTraceCritical |
- kTraceApiCall |
- kTraceMemory |
- kTraceInfo));
-#endif
-
- // Set/GetAudioDeviceLayer
- TEST(Set/GetAudioDeviceLayer);
- ANL();
- AudioLayers wantedLayer = kAudioPlatformDefault;
- AudioLayers givenLayer;
-
-#if defined(_WIN32)
- wantedLayer = kAudioWindowsCore;
- hardware->SetAudioDeviceLayer(wantedLayer);
- TEST_LOG("If you run on XP or below, CoreAudio "
- "should not be able to set.\n");
- TEST_LOG("If you run on Vista or above, CoreAudio "
- "should be able to set.\n");
- TEST_LOG("Verify that this is the case.\n");
-
- TEST_MUSTPASS(voe_base_->Init());
-
- TEST_MUSTPASS(hardware->GetAudioDeviceLayer(givenLayer));
- if(givenLayer == kAudioWindowsCore)
- {
- TEST_LOG("CoreAudio was set\n");
- }
- else
- {
- TEST_LOG("CoreAudio was *not* set\n");
- }
-
- TEST_MUSTPASS(voe_base_->Terminate());
-
- wantedLayer = kAudioWindowsWave;
- TEST_MUSTPASS(hardware->SetAudioDeviceLayer(wantedLayer));
- TEST_LOG("Wave audio should always be able to set.\n");
-
- TEST_MUSTPASS(voe_base_->Init());
-
- TEST_MUSTPASS(hardware->GetAudioDeviceLayer(givenLayer));
- if(givenLayer == kAudioWindowsWave)
- {
- TEST_LOG("Wave audio was set\n");
- }
- else
- {
- TEST_LOG("Wave audio was not set\n");
- }
-
- TEST_MUSTPASS(voe_base_->Terminate());
- // end _WIN32
-#elif defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
- wantedLayer = kAudioLinuxPulse;
- TEST_MUSTPASS(hardware->SetAudioDeviceLayer(wantedLayer));
- TEST_LOG("If you run on Linux with no/unsupported PA version, PulseAudio "
- "7should not be able to set.\n");
- TEST_LOG("If you run on Linux with supported PA version running, PulseAudio"
- " should be able to set.\n");
- TEST_LOG("Verify that this is the case.\n");
-
- TEST_MUSTPASS(voe_base_->Init());
-
- TEST_MUSTPASS(hardware->GetAudioDeviceLayer(givenLayer));
- if(givenLayer == kAudioLinuxPulse)
- {
- TEST_LOG("\nPulseAudio was set\n");
- }
- else
- {
- TEST_LOG("\nPulseAudio was not set\n");
- }
-
- TEST_MUSTPASS(voe_base_->Terminate());
-
- wantedLayer = kAudioLinuxAlsa;
- TEST_MUSTPASS(hardware->SetAudioDeviceLayer(wantedLayer));
- TEST_LOG("ALSA audio should always be able to set.\n");
-
- TEST_MUSTPASS(voe_base_->Init());
-
- TEST_MUSTPASS(hardware->GetAudioDeviceLayer(givenLayer));
- if(givenLayer == kAudioLinuxAlsa)
- {
- TEST_LOG("\nALSA audio was set\n");
- }
- else
- {
- TEST_LOG("\nALSA audio was not set\n");
- }
-
- TEST_MUSTPASS(voe_base_->Terminate());
-#endif // defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
- // Invalid arguments
- wantedLayer = (AudioLayers) 17;
- TEST_MUSTPASS(-1 != hardware->SetAudioDeviceLayer(wantedLayer));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
-
- // Basic usage
- wantedLayer = kAudioPlatformDefault;
- TEST_MUSTPASS(hardware->SetAudioDeviceLayer(wantedLayer));
- TEST_MUSTPASS(hardware->GetAudioDeviceLayer(givenLayer));
- TEST_MUSTPASS(givenLayer != wantedLayer);
- MARK();
-
- TEST_MUSTPASS(voe_base_->Init());
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- wantedLayer = kAudioPlatformDefault;
- TEST_MUSTPASS(-1 != hardware->SetAudioDeviceLayer(wantedLayer));
- TEST_MUSTPASS(VE_ALREADY_INITED != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(hardware->GetAudioDeviceLayer(givenLayer));
- MARK();
- switch (givenLayer) {
- case kAudioPlatformDefault:
- // already set above
- break;
- case kAudioWindowsCore:
- TEST_LOG("\nRunning kAudioWindowsCore\n");
- break;
- case kAudioWindowsWave:
- TEST_LOG("\nRunning kAudioWindowsWave\n");
- break;
- case kAudioLinuxAlsa:
- TEST_LOG("\nRunning kAudioLinuxAlsa\n");
- break;
- case kAudioLinuxPulse:
- TEST_LOG("\nRunning kAudioLinuxPulse\n");
- break;
- default:
- TEST_LOG("\nERROR: Running unknown audio layer!!\n");
- return -1;
- }
- ANL();
-
-#if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
- // GetRecording/PlayoutDeviceStatus
- TEST(Getrecording/PlayoutDeviceStatus);
- ANL();
- bool isRecAvailable = false;
- bool isPlayAvailable = false;
- TEST_MUSTPASS(hardware->GetRecordingDeviceStatus(isRecAvailable));
- TEST_MUSTPASS(!isRecAvailable);
- MARK();
- TEST_MUSTPASS(hardware->GetPlayoutDeviceStatus(isPlayAvailable));
- TEST_MUSTPASS(!isPlayAvailable);
- MARK();
-
- ANL();
-
- int nRec = 0, nPlay = 0;
- char devName[128];
- char guidName[128];
- int idx;
-
- TEST_MUSTPASS(hardware->GetNumOfPlayoutDevices(nPlay));
-
- // GetPlayoutDeviceName
- TEST(GetPlayoutDeviceName);
- ANL();
- TEST_MUSTPASS(-1 != hardware->GetPlayoutDeviceName(nPlay, devName,
- guidName));
- TEST_MUSTPASS(VE_CANNOT_RETRIEVE_DEVICE_NAME != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(-1 != hardware->GetPlayoutDeviceName(-2, devName, guidName));
- TEST_MUSTPASS(VE_CANNOT_RETRIEVE_DEVICE_NAME != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(-1 != hardware->GetPlayoutDeviceName(nPlay+1, devName,
- guidName));
- TEST_MUSTPASS(VE_CANNOT_RETRIEVE_DEVICE_NAME != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(-1 != hardware->GetPlayoutDeviceName(0, NULL, guidName));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(hardware->GetPlayoutDeviceName(0, devName, NULL));
-
- // default tests
- for (idx = 0; idx < nPlay; idx++) {
- TEST_MUSTPASS(hardware->GetPlayoutDeviceName(idx, devName, guidName));
- MARK();
- TEST_MUSTPASS(hardware->SetPlayoutDevice(idx));
- }
-
- ANL();
-
- TEST_MUSTPASS(hardware->GetNumOfRecordingDevices(nRec));
-
- // GetRecordingDeviceName
- TEST(GetRecordingDeviceName);
- ANL();
- TEST_MUSTPASS(-1 != hardware->GetRecordingDeviceName(nRec, devName,
- guidName));
- TEST_MUSTPASS(VE_CANNOT_RETRIEVE_DEVICE_NAME != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(-1 != hardware->GetRecordingDeviceName(-2, devName, guidName));
- TEST_MUSTPASS(VE_CANNOT_RETRIEVE_DEVICE_NAME != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(-1 != hardware->GetRecordingDeviceName(nRec+1, devName,
- guidName));
- TEST_MUSTPASS(VE_CANNOT_RETRIEVE_DEVICE_NAME != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(-1 != hardware->GetRecordingDeviceName(0, NULL, guidName));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(hardware->GetRecordingDeviceName(0, devName, NULL));
-
- // default tests
- for (idx = 0; idx < nRec; idx++) {
- TEST_MUSTPASS(hardware->GetRecordingDeviceName(idx, devName, guidName));
- MARK();
- TEST_MUSTPASS(hardware->SetRecordingDevice(idx));
- }
- ANL();
-
- // // SetRecordingDevice
- TEST(SetRecordingDevice);
- ANL();
- TEST_MUSTPASS(hardware->SetRecordingDevice(0));
- MARK();
- TEST_MUSTPASS(hardware->SetRecordingDevice(0, kStereoLeft));
- MARK();
- TEST_MUSTPASS(hardware->SetRecordingDevice(0, kStereoRight));
- MARK();
- ANL();
-
- // SetPlayoutDevice
- TEST(SetPlayoutDevice);
- ANL();
-#if defined(_WIN32)
- TEST_MUSTPASS(hardware->SetPlayoutDevice(-1)); MARK();
-#else
- TEST_MUSTPASS(hardware->SetPlayoutDevice(0));
- MARK();
-#endif
- ANL();
-#endif // #if !defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID)
-#if defined(MAC_IPHONE)
- TEST(ResetSoundDevice); ANL();
-
- for (int p=0; p<=60; p+=20)
- {
- TEST_LOG("Resetting sound device several times with pause %d ms\n", p);
- for (int l=0; l<50; ++l)
- {
- TEST_MUSTPASS(hardware->ResetAudioDevice()); MARK();
- SLEEP(p);
- }
- ANL();
- }
-
- TEST_LOG("Start streaming - verify the audio after each batch of resets \n");
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0,8000));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(2000);
-
- SLEEP(2000);
- for (int p=0; p<=60; p+=20)
- {
- TEST_LOG("Resetting sound device several time with pause %d ms\n", p);
- for (int l=0; l<20; ++l)
- {
- TEST_MUSTPASS(hardware->ResetAudioDevice()); MARK();
- SLEEP(p);
- }
- ANL();
- SLEEP(2000);
- }
-
- TEST_LOG("Stop streaming \n");
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
-#endif // defined(MAC_IPHONE))
-#ifdef MAC_IPHONE
- TEST_LOG("\nNOTE: Always run hardware tests also without extended tests "
- "enabled,\nsince the extended tests are pre-streaming tests only.\n");
-#endif
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->Terminate());
-
- ANL();
- AOK();
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// VoEExtendedTest::TestNetEqStats
-// ----------------------------------------------------------------------------
-
-int VoEExtendedTest::TestNetEqStats() {
- PrepareTest("NetEqStats (!EMPTY!)");
-
- AOK();
- ANL();
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// VoEExtendedTest::TestNetwork
-//
-// ----------------------------------------------------------------------------
-
-int VoEExtendedTest::TestNetwork() {
- PrepareTest("Network");
-
-#ifdef WEBRTC_ANDROID
- int sleepTime = 200;
- int sleepTime2 = 250;
-#elif defined(MAC_IPHONE) // MAC_IPHONE needs more delay for getSourceInfo()
- int sleepTime = 150;
- int sleepTime2 = 200;
-#else
- int sleepTime = 100;
- int sleepTime2 = 200;
-#endif
-
- VoEBase* voe_base_ = _mgr.BasePtr();
- VoEFile* file = _mgr.FilePtr();
- VoENetwork* netw = _mgr.NetworkPtr();
-
-#ifdef _USE_EXTENDED_TRACE_
- TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename(
- "VoENetwork_trace.txt")));
- TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
- kTraceStateInfo |
- kTraceWarning |
- kTraceError |
- kTraceCritical |
- kTraceApiCall |
- kTraceMemory |
- kTraceInfo));
-#endif
-
- TEST_MUSTPASS(voe_base_->Init());
-
- // ------------------------------------------------------------------------
- // >> GetLocalIP
- //
- // State: VE initialized, no existing channels
- TEST(GetLocalIP);
- ANL();
-
-#ifdef MAC_IPHONE
- // Should fail
- TEST_MUSTPASS(!netw->GetLocalIP(NULL, 0)); MARK();
- TEST_ERROR(VE_FUNC_NOT_SUPPORTED);
-
- ANL();
- printf("NOTE: Local IP must be set in source code (line %d) \n",
- __LINE__ + 1);
- const char* localIP = "192.168.1.4";
-
-#else
- char localIP[64];
-
- // invalid parameter
- TEST_MUSTPASS(!netw->GetLocalIP(NULL));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
-
- // default function calls (IPv4)
- TEST_MUSTPASS(netw->GetLocalIP(localIP));
- MARK();
- TEST_LOG("[local IPv4: %s]\n", localIP);
- TEST_MUSTPASS(netw->GetLocalIP(localIP));
- MARK();
-
-#if !defined(WEBRTC_MAC) && !defined(WEBRTC_ANDROID)
- // default function calls (IPv6)
- TEST_MUSTPASS(netw->GetLocalIP(localIP, true));
- MARK();
- TEST_LOG("[local IPv6: %s]\n", localIP);
- TEST_MUSTPASS(netw->GetLocalIP(localIP, true));
- MARK();
-#endif
-
- // one last call to ensure that local
- TEST_MUSTPASS(netw->GetLocalIP(localIP));
- MARK();
-#endif
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of GetLocalIP
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> GetSourceInfo
- //
- // - VE initialized
- // - no existing channels
- TEST(GetSourceInfo);
- ANL();
-
- int rtpPort(0);
- int rtcpPort(0);
- char ipaddr[64] = { 0 };
- ExtendedTestTransport* ptrTransport(NULL);
-
- // call without valid channel
- TEST_MUSTPASS(!netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // NULL as input string
- TEST_MUSTPASS(!netw->GetSourceInfo(0, rtpPort, rtcpPort, NULL));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
-
- // call when external transport is enabled
- ptrTransport = new ExtendedTestTransport(netw);
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
- TEST_MUSTPASS(!netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- delete ptrTransport;
-
- // call when external transport is disabled (no packet received yet)
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 0);
- TEST_MUSTPASS(rtcpPort != 0);
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
- // send and receive packets with default settings for a while
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(sleepTime2); // does not guarantee RTCP
-
- // verify remote parameters (exclude RTCP)
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 8000);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- // ensure that valid results are maintained after StopListen
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 8000);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- // verify that results are maintained after new call to SetLocalReceiver
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 8000);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- // STATE: not listening, not sending
- // send and receive packets with other settings for a while
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 9005));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 9005, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(sleepTime);
-
- // STATE: listening, sending
-
- // verify new remote parameters
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 9005);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- // restart sending to and from local IP
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 9005, kVoEDefault, localIP));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 9005, localIP));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(sleepTime);
-
- // verify new remote parameters
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 9005);
- TEST_MUSTPASS(strcmp(ipaddr, localIP) != 0); // should not be "127.0.0.1"
-
- // use non-default source port in outgoing packets
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 9005));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 9005, "127.0.0.1", 9010));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(sleepTime);
-
- // verify new remote parameters
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 9010);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- // STATE: listening and sending using an extra local socket
-
- // stop/start sending
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(sleepTime);
-
- // verify that the unique source port is maintained for the extra socket
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 9010);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- // set new source port for outgoing packets (9010 -> 9020)
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 9005, "127.0.0.1", 9020));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(sleepTime);
-#ifdef MAC_IPHONE
- SLEEP(500); // Need extra pause for some reason
-#endif
-
- // verify that the unique source port is set for the new extra socket
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 9020);
- // STATE: listening and sending using an extra local socket
-
- // remove extra send socket and restart call again
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->DeleteChannel(0)); // delete channel => destroys the
- // extra socket
- TEST_MUSTPASS(voe_base_->CreateChannel()); // new channel uses one socket only
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000)); // use new port as well
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(sleepTime);
-
- // verify that remote info is correct
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 8000);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- // STATE: listening and sending using shared socket
-
- // use non-default source port in outgoing packets to create extra send
- // socket
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 7000));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 7000, "127.0.0.1", 7010));
- // RTP src is 7010 => RTCP src = 7011
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(sleepTime);
- // verify new remote parameters
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 7010);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
-
- // check RTCP port as well (should be 7010 + 1 = 7011)
- Sleep(8000, true);
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 7010);
- TEST_MUSTPASS(rtcpPort != 7011);
- TEST_MUSTPASS(strcmp(ipaddr, "127.0.0.1") != 0);
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of GetSourceInfo
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> SetExternalTransport
- //
- // - VE initialized
- // - no existing channels
- // - no media
- //
- TEST(SetExternalTransport);
- ANL();
-
- ptrTransport = new ExtendedTestTransport(netw);
-
- // call without valid channel
- TEST_MUSTPASS(!netw->DeRegisterExternalTransport(0));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // different valid call combinations
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
- MARK();
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
- MARK();
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
- MARK();
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
- MARK();
- TEST_MUSTPASS(!netw->RegisterExternalTransport(0, *ptrTransport));
- MARK(); // must deregister first
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
- MARK();
-
- // STATE: external transport is disabled
-
- // initialize sending and ensure that external transport can't be enabled
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 1234, "127.0.0.2"));
- TEST_MUSTPASS(!netw->RegisterExternalTransport(0, *ptrTransport));
- MARK();
- TEST_ERROR(VE_SEND_SOCKETS_CONFLICT);
-
- // restart channel to ensure that "initialized sender" state is cleared
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // initialize receiving and ensure that external transport can't be enabled
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 5678));
- TEST_MUSTPASS(!netw->RegisterExternalTransport(0, *ptrTransport));
- MARK();
- TEST_ERROR(VE_RECEIVE_SOCKETS_CONFLICT);
-
- // restart channel to ensure that "initialized receiver" state is cleared
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // enable external transport and verify that "emulated loopback" works
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
- MARK();
- TEST_MUSTPASS(voe_base_->StartSend(0)); // should only start recording
- TEST_MUSTPASS(!netw->RegisterExternalTransport(0, *ptrTransport));
- MARK(); // should fail
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
- MARK();
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
- MARK();
- Play(0, 2000, true, true); // play file as mic and verify loopback audio
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
- MARK();
-
- // STATE: external transport is disabled
-#if defined(WEBRTC_ANDROID) || defined(MAC_IPHONE)
- int testError = VE_FUNC_NOT_SUPPORTED;
-#else
- int testError = VE_EXTERNAL_TRANSPORT_ENABLED;
-#endif
-
- // check all APIs that should fail when external transport is enabled
- int DSCP, priority, serviceType, overrideDSCP, nBytes(0);
- bool useSetSockopt, enabled;
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
- MARK();
- TEST_MUSTPASS(!voe_base_->SetLocalReceiver(0, 12345));
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- TEST_MUSTPASS(!voe_base_->GetLocalReceiver(0, rtpPort, rtcpPort, ipaddr));
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- TEST_MUSTPASS(!voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- TEST_MUSTPASS(!voe_base_->GetSendDestination(0, rtpPort, ipaddr, rtpPort,
- rtcpPort));
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- TEST_MUSTPASS(!netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- TEST_MUSTPASS(!netw->EnableIPv6(0))
- TEST_ERROR(testError);
- TEST_MUSTPASS(netw->IPv6IsEnabled(0) != false)
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- TEST_MUSTPASS(!netw->SetSourceFilter(0, 12345, 12346));
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- TEST_MUSTPASS(!netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
-
- // modified i VoE 3.4 (can be called also for external transport)
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
-
-#if (!defined(_WIN32) && !defined(WEBRTC_LINUX) && !defined(WEBRTC_MAC)) || \
- defined(WEBRTC_EXTERNAL_TRANSPORT)
- testError = VE_FUNC_NOT_SUPPORTED;
-#else
- testError = VE_EXTERNAL_TRANSPORT_ENABLED;
-#endif
- TEST_MUSTPASS(!netw->SetSendTOS(0, 0));
- TEST_ERROR(testError);
- TEST_MUSTPASS(!netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
- TEST_ERROR(testError);
-#if !defined(_WIN32) || defined(WEBRTC_EXTERNAL_TRANSPORT)
- testError = VE_FUNC_NOT_SUPPORTED;
-#else
- testError = VE_EXTERNAL_TRANSPORT_ENABLED;
-#endif
- TEST_MUSTPASS(!netw->SetSendGQoS(0, false, 0));
- TEST_ERROR(testError);
- TEST_MUSTPASS(!netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_ERROR(testError);
- char dummy[1] = { 'a' };
- TEST_MUSTPASS(!netw->SendUDPPacket(0, dummy, 1, nBytes));
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
-
- // always disable external transport before deleting the Transport object;
- // will lead to crash for RTCP transmission otherwise
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
- MARK();
- delete ptrTransport;
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
-
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of SetExternalTransport
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> EnableIPv6
- //
- // - VE initialized
- // - no existing channels
- // - no media
- // - NOTE: set _ENABLE_IPV6_TESTS_ to include these tests
- // - http://www.microsoft.com/resources/documentation/windows/xp/all/
- // proddocs/en-us/sag_ip_v6_pro_rt_enable.mspx?mfr=true
- // >> ipv6 install
- // >> ipv6 [-v] if [IfIndex]
- // >> ping6 ::1
- // >> ping6 fe80::1
-
-#ifdef _ENABLE_IPV6_TESTS_
-
- TEST(EnableIPv6); ANL();
-
- // call without valid channel
- TEST_MUSTPASS(!netw->EnableIPv6(0)); MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // call with enabled external transport
- ptrTransport = new ExtendedTestTransport(netw);
- TEST_MUSTPASS(netw->RegisterExternalTransport(0, *ptrTransport));
- TEST_MUSTPASS(!netw->EnableIPv6(0)); MARK();
- TEST_ERROR(VE_EXTERNAL_TRANSPORT_ENABLED);
- TEST_MUSTPASS(netw->DeRegisterExternalTransport(0));
- delete ptrTransport;
-
- // Test "locking" to IPv4
- TEST_MUSTPASS(netw->IPv6IsEnabled(0)); MARK(); // After this call we cannot
- // enable IPv6
- TEST_MUSTPASS(!netw->EnableIPv6(0)); MARK(); // Should fail
-
- // Check that IPv6 address is invalid
- TEST_MUSTPASS(!voe_base_->SetSendDestination(0, 8000, "::1")); MARK(); // fail
-
- // New channel
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // valid default call
- TEST_MUSTPASS(netw->EnableIPv6(0)); MARK();
- TEST_MUSTPASS(netw->GetLocalIP(localIP)); MARK(); // should still read IPv4
- TEST_LOG("[local IPv4: %s]", localIP);
-
- // ensure that Ipv6 is enabled
- TEST_MUSTPASS(netw->IPv6IsEnabled(0) != true);
-
- // check that IPv4 address is invalid
- TEST_MUSTPASS(!voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
- TEST_ERROR(VE_INVALID_IP_ADDRESS);
-
- // verify usage of IPv6 loopback address
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
- // IPv6 loopback address is 0:0:0:0:0:0:0:1
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "::1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(!netw->EnableIPv6(0)); MARK(); // Should fail
- TEST_MUSTPASS(voe_base_->StartSend(0));
- Play(0, 2000, true, true);
- ANL();
-
- // Restart channel
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- TEST_MUSTPASS(netw->EnableIPv6(0)); MARK();
- // ensure that Ipv6 is enabled
- TEST_MUSTPASS(netw->IPv6IsEnabled(0) != true);
-
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- // IPv6 loopback address is 0:0:0:0:0:0:0:1
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "::1"));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(), true,
- true);
- SLEEP(500); // ensure that we receieve some packets
-
- // SetSourceFilter and GetSourceFilter
- TEST(SetSourceFilter and GetSourceFilter for IPv6); ANL();
- char sourceIp[64] =
- { 0};
- char filterIp[64] =
- { 0};
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, sourceIp));
- TEST_LOG("Source port: %d \n", rtpPort);
- TEST_LOG("Source RTCP port: %d \n", rtcpPort);
- TEST_LOG("Source IP: %s \n", sourceIp);
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
- TEST_LOG("Filter port RTP: %d \n", rtpPort);
- TEST_LOG("Filter port RTCP: %d \n", rtcpPort);
- TEST_LOG("Filter IP: %s \n", filterIp);
- TEST_MUSTPASS(0 != rtpPort);
- TEST_MUSTPASS(0 != rtcpPort);
- TEST_MUSTPASS(filterIp[0] != '\0');
- TEST_LOG("Set filter IP to %s => should hear audio\n", sourceIp);
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, sourceIp));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
- TEST_MUSTPASS(0 != rtpPort);
- TEST_MUSTPASS(0 != rtcpPort);
- TEST_MUSTPASS(_stricmp(filterIp, sourceIp));
- SLEEP(1500);
- TEST_LOG("Set filter IP to ::10:10:10 => should *not* hear audio\n");
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "::10:10:10"));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
- TEST_MUSTPASS(_stricmp(filterIp, "::10:10:10"));
- SLEEP(1500);
- TEST_LOG("Disable IP filter => should hear audio again\n");
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "::0"));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
- TEST_MUSTPASS(_stricmp(filterIp, "::"));
- SLEEP(1500);
- TEST_LOG("Set filter IP to ::10:10:10 => should *not* hear audio\n");
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "::10:10:10"));
- SLEEP(1500);
- TEST_LOG("Disable IP filter => should hear audio again\n");
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
- TEST_MUSTPASS(filterIp[0] != '\0');
- SLEEP(1500);
- TEST_LOG("Set filter IP to ::10:10:10 => should *not* hear audio\n");
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "::10:10:10"));
- SLEEP(1500);
- TEST_LOG("Disable IP filter => should hear audio again\n");
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "::"));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, filterIp));
- TEST_MUSTPASS(_stricmp(filterIp, "::"));
- SLEEP(1500);
-
- file->StopPlayingFileAsMicrophone(0);
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
-
-#endif // #ifdef _ENABLE_IPV6_TESTS_
- // >> end of EnableIPv6
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> SetSourceFilter
- //
- // - VE initialized
- // - no existing channels
- // - no media
- TEST(SetSourceFilter);
- ANL();
-
- // call without valid channel
- TEST_MUSTPASS(!netw->SetSourceFilter(0, 12345));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // invalid parameters
- TEST_MUSTPASS(!netw->SetSourceFilter(0, 65536));
- MARK();
- TEST_ERROR(VE_INVALID_PORT_NMBR);
- TEST_MUSTPASS(!netw->SetSourceFilter(0, 12345, 65536));
- MARK();
- TEST_ERROR(VE_INVALID_PORT_NMBR);
- TEST_MUSTPASS(!netw->SetSourceFilter(0, 12345, 12346, "300.300.300.300"));
- MARK();
- TEST_ERROR(VE_INVALID_IP_ADDRESS);
-
- // STATE: RTP filter port is 12345, RTCP filter port is 12346
-
- // disable all filters and ensure that media is received
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
- MARK();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 2000, kVoEDefault, localIP));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 2000, localIP));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(sleepTime);
-
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- TEST_MUSTPASS(rtpPort != 2000);
- TEST_MUSTPASS(rtcpPort != 2001);
- TEST_MUSTPASS(strcmp(ipaddr, localIP) != 0);
-
- // clear states and restart loopback session
- TEST_MUSTPASS(voe_base_->DeleteChannel(0)); // clear source info state
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // set RTP filter to port 2002 and verify that source 2000 is blocked
- TEST_MUSTPASS(netw->SetSourceFilter(0, 2002, 0, NULL));;
- MARK();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 2000, kVoEDefault, localIP));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 2000, localIP));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(sleepTime);
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- TEST_MUSTPASS(rtpPort != 0);
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
-
- // ensure that received packets originates from 2002 and that they now pass
- // the filter
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- // RTP source is 2002
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 2002, kVoEDefault, localIP));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 2002, localIP));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(sleepTime);
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- TEST_MUSTPASS(rtpPort != 2002);
- TEST_MUSTPASS(strcmp(ipaddr, localIP) != 0);
-
- // clear states and restart loopback session
- TEST_MUSTPASS(voe_base_->DeleteChannel(0)); // clear source info state
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // set IP filter to local IP and verify that default loopback stream is
- // blocked
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, localIP));;
- MARK();
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 2000));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 2000, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(sleepTime);
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- TEST_MUSTPASS(rtpPort != 0);
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
-
- // ensure that received packets originates from the local IP and that they
- // now pass the filter
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- // should pass the filter
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 2000, kVoEDefault, localIP));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 2000, localIP));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(sleepTime);
- TEST_MUSTPASS(netw->GetSourceInfo(0, rtpPort, rtcpPort, ipaddr));
- TEST_MUSTPASS(rtpPort != 2000);
- TEST_MUSTPASS(strcmp(ipaddr, localIP) != 0);
-
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
-
- // STATE: no active media, IP filter is active
-
- // disable all filters
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));;
- MARK();
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- TEST_MUSTPASS(rtpPort != 0);
- TEST_MUSTPASS(rtcpPort != 0);
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of SetSourceFilter
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> GetSourceFilter
- //
- // - VE initialized
- // - no existing channels
- // - no media
- TEST(GetSourceFilter);
- ANL();
-
- // call without valid channel
- TEST_MUSTPASS(!netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // invalid input parameters
- TEST_MUSTPASS(!netw->GetSourceFilter(0, rtpPort, rtcpPort, NULL));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
-
- // valid call without any filter set
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 0);
- TEST_MUSTPASS(rtcpPort != 0);
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
-
- // STATE: no active media and no enabled filters
-
- // set different filters and verify that they "bite"
- TEST_MUSTPASS(netw->SetSourceFilter(0, 54321, 0, NULL));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 54321);
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtpPort != 0);
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 15425, NULL));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtcpPort != 15425);
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(rtcpPort != 0);
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "192.168.199.19"));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(strcmp(ipaddr, "192.168.199.19") != 0);
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, "0.0.0.0"));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(strcmp(ipaddr, "0.0.0.0") != 0);
- TEST_MUSTPASS(netw->SetSourceFilter(0, 0, 0, NULL));
- TEST_MUSTPASS(netw->GetSourceFilter(0, rtpPort, rtcpPort, ipaddr));
- MARK();
- TEST_MUSTPASS(strcmp(ipaddr, "") != 0);
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of GetSourceFilter
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> RegisterDeadOrAliveObserver
- // >> DeRegisterDeadOrAliveObserver
- //
- // - VE initialized
- // - no existing channels
- // - no media
- TEST(RegisterDeadOrAliveObserver);
- ANL();
- TEST(DeRegisterDeadOrAliveObserver);
- ANL();
-
- // call without valid channel
- TEST_MUSTPASS(!netw->RegisterDeadOrAliveObserver(0, *this));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- TEST_MUSTPASS(netw->RegisterDeadOrAliveObserver(0, *this));
- MARK();
- TEST_MUSTPASS(!netw->RegisterDeadOrAliveObserver(0, *this));
- MARK(); // already registered
- TEST_ERROR(VE_INVALID_OPERATION);
- TEST_MUSTPASS(netw->DeRegisterDeadOrAliveObserver(0));
- MARK();
- TEST_MUSTPASS(netw->DeRegisterDeadOrAliveObserver(0));
- MARK(); // OK to do it again
- TEST_MUSTPASS(netw->RegisterDeadOrAliveObserver(0, *this));
- MARK();
- TEST_MUSTPASS(netw->DeRegisterDeadOrAliveObserver(0));
- MARK();
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
-
- // STATE: dead-or-alive observer is disabled
-
- // >> end of RegisterDeadOrAliveObserver
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> SetPeriodicDeadOrAliveStatus
- // >> GetPeriodicDeadOrAliveStatus
- //
- // - VE initialized
- // - no existing channels
- // - no media
-
- // call without valid channel
- TEST_MUSTPASS(!netw->SetPeriodicDeadOrAliveStatus(0, false));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // Invalid paramters
- TEST_MUSTPASS(!netw->SetPeriodicDeadOrAliveStatus(0, true, 0));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetPeriodicDeadOrAliveStatus(0, true, 151));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetPeriodicDeadOrAliveStatus(1, true, 10));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- int sampleTime(0);
-
- // Valid parameters
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 1));
- MARK();
- TEST_MUSTPASS(netw->GetPeriodicDeadOrAliveStatus(0, enabled, sampleTime));
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(sampleTime != 1);
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 150));
- MARK();
- TEST_MUSTPASS(netw->GetPeriodicDeadOrAliveStatus(0, enabled, sampleTime));
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(sampleTime != 150);
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, false));
- MARK();
- TEST_MUSTPASS(netw->GetPeriodicDeadOrAliveStatus(0, enabled, sampleTime));
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(sampleTime != 150); // ensure last set time isnt modified
-
- StartMedia(0, 2000, true, true, true);
-
- // STATE: full duplex media is active
-
- // test the dead-or-alive mechanism
- TEST_MUSTPASS(netw->RegisterDeadOrAliveObserver(0, *this));
- MARK();
- TEST_LOG("\nVerify that Alive callbacks are received (dT=2sec): ");
- fflush(NULL);
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 2));
- SLEEP(6000);
- TEST_LOG("\nChange dT to 1 second: ");
- fflush(NULL);
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 1));
- SLEEP(6000);
- TEST_LOG("\nDisable dead-or-alive callbacks: ");
- fflush(NULL);
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, false));
- SLEEP(6000);
- TEST_LOG("\nStop sending and enable callbacks again.\n");
- TEST_LOG("Verify that Dead callbacks are received (dT=2sec): ");
- fflush(NULL);
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, true, 2));
- SLEEP(6000);
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_LOG("\nRestart sending.\n");
- TEST_LOG("Verify that Alive callbacks are received again (dT=2sec): ");
- fflush(NULL);
- SLEEP(6000);
- TEST_LOG("\nDisable dead-or-alive callbacks.");
- fflush(NULL);
- TEST_MUSTPASS(netw->SetPeriodicDeadOrAliveStatus(0, false));
- TEST_MUSTPASS(netw->DeRegisterDeadOrAliveObserver(0));
- MARK();
-
- StopMedia(0);
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of SetPeriodicDeadOrAliveStatus
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> SetPacketTimeoutNotification
- // >> GetPacketTimeoutNotification
- //
- // - VE initialized
- // - no existing channels
- // - no media
- // - NOTE: dynamic tests are performed in standard test
-
- int timeOut(0);
-
- TEST(SetPacketTimeoutNotification);
- ANL();
- TEST(GetPacketTimeoutNotification);
- ANL();
-
- // call without existing valid channel
- TEST_MUSTPASS(!netw->SetPacketTimeoutNotification(0, false));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // invalid function calls
- TEST_MUSTPASS(!netw->SetPacketTimeoutNotification(0, true, 0));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetPacketTimeoutNotification(0, true, 151));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
-
- // valid function calls (no active media)
- TEST_MUSTPASS(netw->SetPacketTimeoutNotification(0, true, 2));
- MARK();
- TEST_MUSTPASS(netw->GetPacketTimeoutNotification(0, enabled, timeOut));
- MARK();
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(timeOut != 2);
- TEST_MUSTPASS(netw->SetPacketTimeoutNotification(0, false));
- MARK();
- TEST_MUSTPASS(netw->GetPacketTimeoutNotification(0, enabled, timeOut));
- MARK();
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(netw->SetPacketTimeoutNotification(0, true, 10));
- MARK();
- TEST_MUSTPASS(netw->GetPacketTimeoutNotification(0, enabled, timeOut));
- MARK();
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(timeOut != 10);
- TEST_MUSTPASS(netw->SetPacketTimeoutNotification(0, true, 2));
- MARK();
- TEST_MUSTPASS(netw->GetPacketTimeoutNotification(0, enabled, timeOut));
- MARK();
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(timeOut != 2);
- TEST_MUSTPASS(netw->SetPacketTimeoutNotification(0, false));
- MARK();
- TEST_MUSTPASS(netw->GetPacketTimeoutNotification(0, enabled, timeOut));
- MARK();
- TEST_MUSTPASS(enabled != false);
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- ANL();
- AOK();
- ANL();
- ANL();
-
- // >> end of SetPacketTimeoutNotification
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> SendUDPPacket
- //
- // - VE initialized
- // - no existing channels
- // - no media
-
-
- // >> end of SendUDPPacket
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> SetSendTOS
- //
- // - VE initialized
- // - no existing channels
- // - no media
- TEST(SetSendTOS);
- ANL();
-#if defined(_WIN32) || defined(WEBRTC_MAC) || defined(WEBRTC_LINUX)
-
- // call without existing valid channel
-
- TEST_MUSTPASS(!netw->SetSendTOS(0, 0)); MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // trivial invalid function calls
- TEST_MUSTPASS(!netw->SetSendTOS(0, -1)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendTOS(0, 64)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendTOS(0, 1, -2)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendTOS(0, 1, 8)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendTOS(0, 1)); MARK();
- TEST_ERROR(VE_SOCKET_ERROR); // must create sockets first
-
-#ifdef _WIN32
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 3000));
-
- // enable ToS using SetSockopt (should work without local binding)
- TEST_MUSTPASS(netw->SetSendTOS(0, 1, -1, true)); MARK();
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt)); MARK();
- TEST_MUSTPASS(DSCP != 1);
- TEST_MUSTPASS(priority != 0);
- TEST_MUSTPASS(useSetSockopt != true);
-
- // try to disable SetSockopt while ToS is enabled (should fail)
- TEST_MUSTPASS(!netw->SetSendTOS(0, 1, -1, false)); MARK();
- TEST_ERROR(VE_TOS_INVALID); // must disable ToS before enabling SetSockopt
-
- // disable ToS to be able to stop using SetSockopt
- TEST_MUSTPASS(netw->SetSendTOS(0, 0, -1, true)); MARK(); // disable ToS
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt)); MARK();
- TEST_MUSTPASS(DSCP != 0);
- TEST_MUSTPASS(priority != 0);
- TEST_MUSTPASS(useSetSockopt != true);
-
- // to use the "non-SetSockopt" method, local binding is required,
- // trying without it should fail
- TEST_MUSTPASS(!netw->SetSendTOS(0, 1, -1, false)); MARK();
- TEST_ERROR(VE_TOS_ERROR); // must bind to local IP first
-
- // bind to local IP and try again (should work this time)
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345, kVoEDefault, localIP));
- TEST_LOG("\nThis test needs to be run as administrator\n");
- TEST_MUSTPASS(netw->SetSendTOS(0, 1, -1, false)); MARK();
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt)); MARK();
- TEST_MUSTPASS(DSCP != 1);
- TEST_MUSTPASS(priority != 0);
- TEST_MUSTPASS(useSetSockopt != false);
-
- // STATE: binded to local IP, local port is 12345 and DSCP is 1 (not using
- // SetSockopt)
-
- // verify loopback audio with the current settings
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, localIP));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- Play(0, 2000, true, true); // file should be played out here...
-
-#ifdef _SEND_TO_REMOTE_IP_
- // Send to remote destination and verify the DSCP using Wireshark.
- // Use filter ip.src == "RemoteIP".
- TEST_LOG("\nUse Wireshark and verify a correctly received DSCP at the "
- "remote side!\n");
- TEST_LOG("Sending approx. 5 packets to %s:%d for each DSCP below:\n",
- RemoteIP, RemotePort);
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, RemotePort, RemoteIP));
- TEST_LOG(" DSCP is set to 0x%02x\n", 1);
- SLEEP(100);
-
- // Change the DSCP while sending is active and verify on remote side.
- TEST_MUSTPASS(netw->SetSendTOS(0, 2));
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
- TEST_LOG(" DSCP is set to 0x%02x\n", DSCP);
- SLEEP(100);
-
- // Change the DSCP while sending is active and verify on remote side.
- TEST_MUSTPASS(netw->SetSendTOS(0, 63));
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
- TEST_LOG(" DSCP is set to 0x%02x\n", DSCP);
- SLEEP(100);
-
- // stop and resume sending
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
- TEST_LOG(" DSCP is set to 0x%02x\n", DSCP);
- SLEEP(100);
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(netw->SetSendTOS(0, 0));
-#endif // _SEND_TO_REMOTE_IP_
- // Windows priority tests (priority cannot be set using setsockopt on Win)
- TEST_LOG("Testing priority\n");
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, localIP));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(!netw->SetSendTOS(0, 0, 3, true)); // Should fail
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(netw->SetSendTOS(0, 0, 3, false));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- Play(0, 2000, true, true); // file should be played out here...
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(netw->SetSendTOS(0, 1, 3, false));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- Play(0, 2000, true, true); // file should be played out here...
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->CreateChannel());
-#endif // _WIN32
- // STATE: no media, disabled ToS, no defined receiver
-
- // Repeat tests above but using setsockopt() this time.
- // Binding to local IP should not be required.
-
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345, kVoEDefault));
- TEST_MUSTPASS(netw->SetSendTOS(0, 10, -1, true)); MARK();
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt)); MARK();
- TEST_MUSTPASS(DSCP != 10);
- TEST_MUSTPASS(priority != 0);
- TEST_MUSTPASS(useSetSockopt != true);
-
- // STATE: *not* binded to local IP, local port is 12345 and DSCP is 10
- // (using SetSockopt)
-
- // verify loopback audio with the current settings
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- Play(0, 2000, true, true); // file should be played out here...
-
-#ifdef _SEND_TO_REMOTE_IP_
- // Send to remote destination and verify the DSCP using Wireshark.
- // Use filter ip.src == "RemoteIP".
- TEST_LOG("\nUse Wireshark and verify a correctly received DSCP at the"
- " remote side!\n");
- TEST_LOG("Sending approx. 5 packets to %s:%d for each DSCP below:\n",
- RemoteIP, RemotePort);
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, RemotePort, RemoteIP));
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
- TEST_LOG(" DSCP is set to 0x%02x (setsockopt)\n", DSCP);
- SLEEP(100);
-
- // Change the DSCP while sending is active and verify on remote side.
- TEST_MUSTPASS(netw->SetSendTOS(0, 20, -1, true)); // use setsockopt()
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
- TEST_LOG(" DSCP is set to 0x%02x (setsockopt)\n", DSCP);
- SLEEP(100);
-
- // Change the DSCP while sending is active and verify on remote side.
- TEST_MUSTPASS(netw->SetSendTOS(0, 61, -1, true)); // use setsockopt()
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
- TEST_LOG(" DSCP is set to 0x%02x (setsockopt)\n", DSCP);
- SLEEP(100);
-
- // stop and resume sending
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(netw->GetSendTOS(0, DSCP, priority, useSetSockopt));
- TEST_LOG(" DSCP is set to 0x%02x (setsockopt)\n", DSCP);
- SLEEP(100);
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(netw->SetSendTOS(0, 0, -1, true));
-#endif // _SEND_TO_REMOTE_IP_
-#if defined(WEBRTC_LINUX)
- // Linux priority tests (using setsockopt)
- TEST_LOG("Testing priority\n");
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, localIP));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(netw->SetSendTOS(0, 0, 3, true));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- Play(0, 2000, true, true); // file should be played out here...
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(netw->SetSendTOS(0, 1, 3, true));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- Play(0, 2000, true, true); // file should be played out here...
-#endif // #if defined(WEBRTC_LINUX)
-#if !defined(_WIN32) && !defined(WEBRTC_LINUX)
- // Fail tests for other than Wind and Linux
- TEST_MUSTPASS(!netw->SetSendTOS(0, 0, 3, false)); // Should fail
- TEST_ERROR(VE_INVALID_ARGUMENT);
-#endif // #if !defined(_WIN32) && !defined(WEBRTC_LINUX)
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- ANL(); AOK(); ANL(); ANL();
-
- // END #if defined(_WIN32) || defined(WEBRTC_MAC) || defined(WEBRTC_LINUX)
-#else
- TEST_LOG("Skipping ToS tests - _WIN32, LINUX, MAC is not defined or "
- "WEBRTC_ANDROID is defined");
-#endif
-
- // >> end of SetSendTOS
- // ------------------------------------------------------------------------
-
- // ------------------------------------------------------------------------
- // >> SetSendGQoS (Windows only)
- //
- // - VE initialized
- // - no existing channels
- // - no media
- //
- // From qos.h:
- //
- // #define SERVICETYPE_NOTRAFFIC 0x00000000
- // #define SERVICETYPE_BESTEFFORT 0x00000001 (*)
- // #define SERVICETYPE_CONTROLLEDLOAD 0x00000002 (*)
- // #define SERVICETYPE_GUARANTEED 0x00000003 (*)
- // #define SERVICETYPE_NETWORK_UNAVAILABLE 0x00000004
- // #define SERVICETYPE_GENERAL_INFORMATION 0x00000005
- // #define SERVICETYPE_NOCHANGE 0x00000006
- // #define SERVICETYPE_NONCONFORMING 0x00000009
- // #define SERVICETYPE_NETWORK_CONTROL 0x0000000A
- // #define SERVICETYPE_QUALITATIVE 0x0000000D (*)
- //
- // #define SERVICE_BESTEFFORT 0x80010000
- // #define SERVICE_CONTROLLEDLOAD 0x80020000
- // #define SERVICE_GUARANTEED 0x80040000
- // #define SERVICE_QUALITATIVE 0x80200000
- //
- // (*) supported in WEBRTC VoE
- TEST(SetSendGQoS);
- ANL();
-#ifdef _WIN32
-
- // call without existing valid channel
- TEST_MUSTPASS(!netw->SetSendGQoS(0, false, 0)); MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- // supported service type but no sockets
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT)); MARK();
- TEST_ERROR(VE_SOCKETS_NOT_INITED);
-
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
-
- // supported service type but sender is not initialized
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT)); MARK();
- TEST_ERROR(VE_DESTINATION_NOT_INITED);
-
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
-
- // invalid service types
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_NOTRAFFIC)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_NETWORK_UNAVAILABLE));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_GENERAL_INFORMATION));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_NOCHANGE)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_NONCONFORMING));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_NETWORK_CONTROL));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICE_BESTEFFORT)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICE_CONTROLLEDLOAD)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICE_GUARANTEED)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICE_QUALITATIVE)); MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
-
- // Is ToS enabled here?
-
- // Settings which don't require binding to local IP
-
- // set SERVICETYPE_BESTEFFORT
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT)); MARK();
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- MARK();
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(serviceType != SERVICETYPE_BESTEFFORT);
- TEST_MUSTPASS(overrideDSCP != false);
-
- // set SERVICETYPE_CONTROLLEDLOAD
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_CONTROLLEDLOAD));
- MARK();
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- MARK();
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(serviceType != SERVICETYPE_CONTROLLEDLOAD);
- TEST_MUSTPASS(overrideDSCP != false);
-
- // set SERVICETYPE_GUARANTEED
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_GUARANTEED)); MARK();
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- MARK();
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(serviceType != SERVICETYPE_GUARANTEED);
- TEST_MUSTPASS(overrideDSCP != false);
-
- // set SERVICETYPE_QUALITATIVE
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_QUALITATIVE)); MARK();
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- MARK();
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(serviceType != SERVICETYPE_QUALITATIVE);
- TEST_MUSTPASS(overrideDSCP != false);
-
- // disable GQoS
- TEST_MUSTPASS(netw->SetSendGQoS(0, false, 0)); MARK();
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- MARK();
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(serviceType != SERVICETYPE_QUALITATIVE);
- TEST_MUSTPASS(overrideDSCP != false);
-
- // STATE: diabled QGoS, sockets exists, sending side is initialized, no media
-
- // Loopback tests using the four different GQoS settings
-
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT)); MARK();
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- ANL();
- TEST_LOG("[SERVICETYPE_BESTEFFORT]");
- Play(0, 2000, true, true); // file should be played out here...
-
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_CONTROLLEDLOAD)); MARK();
- ANL();
- TEST_LOG("[SERVICETYPE_CONTROLLEDLOAD]");
- Play(0, 2000, true, true); // file should be played out here...
-
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_GUARANTEED)); MARK();
- ANL();
- TEST_LOG("[SERVICETYPE_GUARANTEED]");
- Play(0, 2000, true, true); // file should be played out here...
-
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_QUALITATIVE)); MARK();
- ANL();
- TEST_LOG("[SERVICETYPE_QUALITATIVE]");
- Play(0, 2000, true, true); // file should be played out here...
-
-#ifdef _SEND_TO_REMOTE_IP_
- // Send to remote destination and verify the DSCP mapping using Wireshark.
- // Use filter ip.src == "RemoteIP".
-
- // Modify the send destination on the fly
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, RemotePort, RemoteIP));
-
- TEST_LOG("\nUse Wireshark and verify a correctly received DSCP mapping at"
- " the remote side!\n");
- TEST_LOG("Sending approx. 5 packets to %s:%d for each GQoS setting below:\n",
- RemoteIP, RemotePort);
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT));
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_LOG(" serviceType is set to SERVICETYPE_BESTEFFORT (0x%02x), should "
- "be mapped to DSCP = 0x00\n", serviceType);
- SLEEP(100);
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_CONTROLLEDLOAD));
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_LOG(" serviceType is set to SERVICETYPE_CONTROLLEDLOAD (0x%02x), "
- "should be mapped to DSCP = 0x18\n", serviceType);
- SLEEP(100);
- TEST_MUSTPASS(netw->SetSendGQoS(0, false, 0));
- TEST_LOG(" QoS is disabled, should give DSCP = 0x%02x\n", 0);
- SLEEP(100);
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_GUARANTEED));
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_LOG(" serviceType is set to SERVICETYPE_GUARANTEED (0x%02x), should "
- "be mapped to DSCP = 0x28\n", serviceType);
- SLEEP(100);
- TEST_MUSTPASS(netw->SetSendGQoS(0, false, 0));
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_QUALITATIVE));
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_LOG(" serviceType is set to SERVICETYPE_QUALITATIVE (0x%02x), should"
- " be mapped to DSCP = 0x00\n", serviceType);
- SLEEP(100);
-#endif // _SEND_TO_REMOTE_IP_
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
-
- // STATE: sockets exists, sending side is initialized, no media
-
- // Repeat test above but this time using overrideDSCP.
-
- // Some initial loopack tests.
- // NOTE - override DSCP requres binding to local IP.
-
- // should not work since QoS is enabled
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 3));
- MARK();
- TEST_ERROR(VE_TOS_GQOS_CONFLICT);
-
- // disble QoS and try to override again (should fail again since local
- // binding is not done yet)
- TEST_MUSTPASS(netw->SetSendGQoS(0, false, 0));
- TEST_MUSTPASS(!netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 3));
- MARK();
- TEST_ERROR(VE_GQOS_ERROR);
-
- // make proper settings and try again (should work this time)
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345, kVoEDefault, localIP));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, localIP));
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 3));
- MARK();
-
- // Now, let's try some loopback tests using override DSCP
-
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- ANL();
- TEST_LOG("[overrideDSCP=3]");
- Play(0, 2000, true, true); // file should be played out here...
-
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 17));
- MARK();
- ANL();
- TEST_LOG("[overrideDSCP=17]");
- Play(0, 2000, true, true); // file should be played out here...
-
- // And finally, send to remote side as well to verify that the new mapping
- // works as it should.
-
-#ifdef _SEND_TO_REMOTE_IP_
- // Modify the send destination on the fly
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, RemotePort, RemoteIP));
-
- TEST_LOG("\nUse Wireshark and verify a correctly received DSCP mapping at"
- " the remote side!\n");
- TEST_LOG("Sending approx. 5 packets to %s:%d for each GQoS setting below:\n",
- RemoteIP, RemotePort);
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 18));
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_LOG(" serviceType is set to SERVICETYPE_BESTEFFORT, should be "
- "overrided to DSCP = 0x%02x\n", overrideDSCP);
- SLEEP(100);
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 62));
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_LOG(" serviceType is set to SERVICETYPE_BESTEFFORT, should be "
- "overrided to DSCP = 0x%02x\n", overrideDSCP);
- SLEEP(100);
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 32));
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_LOG(" serviceType is set to SERVICETYPE_BESTEFFORT, should be "
- "overrided to DSCP = 0x%02x\n", overrideDSCP);
- SLEEP(100);
- TEST_MUSTPASS(netw->SetSendGQoS(0, true, SERVICETYPE_BESTEFFORT, 1));
- TEST_MUSTPASS(netw->GetSendGQoS(0, enabled, serviceType, overrideDSCP));
- TEST_LOG(" serviceType is set to SERVICETYPE_BESTEFFORT, should be "
- "overrided to DSCP = 0x%02x\n", overrideDSCP);
- SLEEP(100);
- TEST_MUSTPASS(netw->SetSendGQoS(0, false, 0));
- TEST_LOG(" QoS is disabled, should give DSCP = 0x%02x\n", 0);
- SLEEP(100);
-#endif // _SEND_TO_REMOTE_IP_
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- ANL(); AOK(); ANL(); ANL();
-
-#else
- TEST_LOG("Skipping GQoS tests - _WIN32 is not defined");
-#endif // #ifdef _WIN32
- // >> end of SetSendGQoS
- // ------------------------------------------------------------------------
-
- if (file) {
- file->StopPlayingFileAsMicrophone(0);
- }
- voe_base_->StopSend(0);
- voe_base_->StopPlayout(0);
- voe_base_->StopReceive(0);
- voe_base_->DeleteChannel(0);
- voe_base_->Terminate();
-
- ANL();
- AOK();
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// VoEExtendedTest::TestRTP_RTCP
-// ----------------------------------------------------------------------------
-
-// Used to validate packets during the RTP audio level indication test.
-class RTPAudioTransport: public Transport {
-public:
-
- RTPAudioTransport() :
- mute_(false) {
- }
-
- virtual ~RTPAudioTransport() {
- }
-
- void set_mute(bool mute) {
- mute_ = mute;
- }
- bool mute() const {
- return mute_;
- }
-
- // TODO(andrew): use proper error checks here rather than asserts.
- virtual int SendPacket(int channel, const void* data, int length) {
- const uint8_t* packet = static_cast<const uint8_t*> (data);
-
- // Extension bit.
- assert(packet[0] & 0x10);
- int index = 12; // Assume standard RTP header.
- // Header extension ID
- assert(packet[index++] == 0xBE);
- assert(packet[index++] == 0xDE);
- // Header extension length
- assert(packet[index++] == 0x00);
- assert(packet[index++] == 0x01);
-
- // User-defined ID.
- assert(((packet[index] & 0xf0) >> 4) == 1);
- // Length
- assert((packet[index++] & 0x0f) == 0);
-
- int vad = packet[index] >> 7;
- int level = packet[index] & 0x7f;
- if (channel == 0) {
- printf("%d -%d\n", vad, level);
- } else if (channel == 1) {
- printf(" %d -%d\n", vad, level);
- } else {
- assert(false);
- }
-
- if (mute_) {
- assert(vad == 0);
- assert(level == 127);
- } else {
- assert(vad == 0 || vad == 1);
- assert(level >= 0 && level <= 127);
- }
-
- return 0;
- }
-
- virtual int SendRTCPPacket(int /*channel*/, const void* /*data*/,
- int /*length*/) {
- return 0;
- }
-
-private:
- bool mute_;
-};
-
-int VoEExtendedTest::TestRTP_RTCP() {
- PrepareTest("RTP_RTCP");
-
- VoEBase* voe_base_ = _mgr.BasePtr();
- VoEFile* file = _mgr.FilePtr();
- VoERTP_RTCP* rtp_rtcp = _mgr.RTP_RTCPPtr();
- VoENetwork* network = _mgr.NetworkPtr();
- VoEVolumeControl* volume = _mgr.VolumeControlPtr();
- VoECodec* codec = _mgr.CodecPtr();
-
- XRTPObserver rtpObserver;
-
-#ifdef WEBRTC_ANDROID
- int sleepTime = 200;
-#else
- int sleepTime = 100;
-#endif
-
-#ifdef _USE_EXTENDED_TRACE_
- TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename(
- "VoERTP_RTCP_trace.txt")));
- TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
- kTraceStateInfo |
- kTraceWarning |
- kTraceError |
- kTraceCritical |
- kTraceApiCall |
- kTraceMemory |
- kTraceInfo));
-#endif
-
- TEST_MUSTPASS(voe_base_->Init());
- TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
-
- ///////////////////////////
- // Actual test starts here
-
- // ------------------------------------------------------------------------
- // >> Set/GetRTPAudioLevelIndicationStatus
- TEST(SetRTPAudioLevelIndicationStatus);
- ANL();
- TEST(GetRTPAudioLevelIndicationStatus);
-
- // test invalid input parameters
- TEST_MUSTPASS(-1 != rtp_rtcp->SetRTPAudioLevelIndicationStatus(0, true, 0));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(-1 != rtp_rtcp->SetRTPAudioLevelIndicationStatus(0, true, 15));
- MARK();
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(-1 != rtp_rtcp->SetRTPAudioLevelIndicationStatus(0, false, 15));
- MARK();
- TEST_MUSTPASS(-1 != rtp_rtcp->SetRTPAudioLevelIndicationStatus(1, true, 5));
- MARK();
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
-
- // test complete valid input range [1,14]
- bool audioLevelEnabled(false);
- unsigned char ID(0);
- for (int id = 1; id < 15; id++) {
- TEST_MUSTPASS(rtp_rtcp->SetRTPAudioLevelIndicationStatus(0, true, id));
- MARK();
- TEST_MUSTPASS(rtp_rtcp->GetRTPAudioLevelIndicationStatus(
- 0, audioLevelEnabled, ID));
- MARK();
- TEST_MUSTPASS(audioLevelEnabled != true);
- TEST_MUSTPASS(rtp_rtcp->SetRTPAudioLevelIndicationStatus(0, false, id));
- MARK();
- TEST_MUSTPASS(rtp_rtcp->GetRTPAudioLevelIndicationStatus(
- 0, audioLevelEnabled, ID));
- MARK();
- TEST_MUSTPASS(audioLevelEnabled != false);
- TEST_MUSTPASS(ID != id);
- }
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
-
- RTPAudioTransport rtpAudioTransport;
- TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(network->RegisterExternalTransport(0, rtpAudioTransport));
- TEST_MUSTPASS(rtp_rtcp->SetRTPAudioLevelIndicationStatus(0, true));
- TEST_MUSTPASS(codec->SetVADStatus(0, true));
-
- printf("\n\nReceving muted packets (expect VAD = 0, Level = -127)...\n");
- printf("VAD Level [dbFS]\n");
- SLEEP(2000);
- rtpAudioTransport.set_mute(true);
- TEST_MUSTPASS(volume->SetInputMute(0, true));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(5000);
- TEST_MUSTPASS(voe_base_->StopSend(0));
- rtpAudioTransport.set_mute(false);
- TEST_MUSTPASS(volume->SetInputMute(0, false));
-
- printf("\nReceiving packets from mic (should respond to mic level)...\n");
- printf("VAD Level [dbFS]\n");
- SLEEP(2000);
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(5000);
- TEST_MUSTPASS(voe_base_->StopSend(0));
-
- printf("\nReceiving packets from file (expect mostly VAD = 1)...\n");
- printf("VAD Level [dbFS]\n");
- SLEEP(2000);
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
- true, true));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(5000);
- TEST_MUSTPASS(voe_base_->StopSend(0));
-
- printf("\nMuted and mic on independent channels...\n");
- printf("Muted Mic\n");
- SLEEP(2000);
- ASSERT_TRUE(1 == voe_base_->CreateChannel());
- TEST_MUSTPASS(network->RegisterExternalTransport(1, rtpAudioTransport));
- TEST_MUSTPASS(rtp_rtcp->SetRTPAudioLevelIndicationStatus(1, true));
- TEST_MUSTPASS(codec->SetVADStatus(1, true));
- TEST_MUSTPASS(volume->SetInputMute(0, true));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(voe_base_->StartSend(1));
- SLEEP(5000);
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopSend(1));
-
- TEST_MUSTPASS(network->DeRegisterExternalTransport(0));
- TEST_MUSTPASS(network->DeRegisterExternalTransport(1));
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->DeleteChannel(1));
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
-
- MARK();
- ANL();
-
- // ------------------------------------------------------------------------
- // >> InsertExtraRTPPacket
-
- int i(0);
-
- TEST(SetLocalSSRC);
- TEST_MUSTPASS(!rtp_rtcp->SetLocalSSRC(0, 5678));
- MARK();
- TEST_MUSTPASS(VE_ALREADY_SENDING != voe_base_->LastError());
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(rtp_rtcp->SetLocalSSRC(0, 5678)); // force send SSRC to 5678
- TEST_MUSTPASS(voe_base_->StartSend(0));
- MARK();
- ANL();
-
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
- true, true));
-
- // ------------------------------------------------------------------------
- // >> InsertExtraRTPPacket
- TEST(InsertExtraRTPPacket);
- ANL();
-
- const char payloadData[8] = { 'A', 'B', 'C', 'D', 'E', 'F', 'G', 'H' };
-
- TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(-1, 0, false,
- payloadData, 8));
- MARK(); // invalid channel
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
- TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(0, -1, false,
- payloadData, 8));
- MARK(); // invalid payload type
- TEST_ERROR(VE_INVALID_PLTYPE);
- TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(0, 128, false,
- payloadData, 8));
- MARK(); // invalid payload type
- TEST_ERROR(VE_INVALID_PLTYPE);
- TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(0, 99, false,
- NULL, 8));
- MARK(); // invalid pointer
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(0, 99, false,
- payloadData, 1500-28+1));
- MARK(); // invalid size
- TEST_ERROR(VE_INVALID_ARGUMENT);
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(-1 != rtp_rtcp->InsertExtraRTPPacket(0, 99, false,
- payloadData, 8));
- MARK(); // not sending
- TEST_ERROR(VE_NOT_SENDING);
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
- true, true));
-
- SLEEP(1000);
- for (int p = 0; p < 128; p++) {
- TEST_MUSTPASS(rtp_rtcp->InsertExtraRTPPacket(0, p, false,
- payloadData, 8));
- MARK();
- TEST_MUSTPASS(rtp_rtcp->InsertExtraRTPPacket(0, p, true,
- payloadData, 8));
- MARK();
- }
-
- // Ensure we have sent all extra packets before we move forward to avoid
- //incorrect error code
- SLEEP(1000);
-
- ANL();
-
- // ------------------------------------------------------------------------
- // >> RTP dump APIs
- TEST(Start/StopRtpDump);
- ANL();
- TEST(Start/RTPDumpIsActive);
-
- TEST_MUSTPASS(-1 != rtp_rtcp->RTPDumpIsActive(-1, kRtpIncoming));
- MARK(); // invalid channel
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
- TEST_MUSTPASS(false != rtp_rtcp->RTPDumpIsActive(0, kRtpIncoming));
- MARK(); // should be off by default
- TEST_MUSTPASS(false != rtp_rtcp->RTPDumpIsActive(0, kRtpOutgoing));
- MARK(); // should be off by default
-
- TEST_MUSTPASS(-1 != rtp_rtcp->StartRTPDump(-1, NULL));
- MARK(); // invalid channel
- TEST_ERROR(VE_CHANNEL_NOT_VALID);
- TEST_MUSTPASS(-1 != rtp_rtcp->StartRTPDump(0, NULL));
- MARK(); // invalid file name
- TEST_ERROR(VE_BAD_FILE);
-
- // Create two RTP dump files:
-
- // - dump_in_1sec.rtp <=> ~1 sec recording of input side
- // - dump_in_2sec.rtp <=> ~2 sec recording of output side
- //
- TEST_MUSTPASS(rtp_rtcp->StopRTPDump(0));
- MARK();
- TEST_MUSTPASS(rtp_rtcp->StopRTPDump(0, kRtpIncoming));
- MARK();
- TEST_MUSTPASS(rtp_rtcp->StopRTPDump(0, kRtpOutgoing));
- MARK();
- TEST_MUSTPASS(rtp_rtcp->StartRTPDump(0, GetFilename("dump_in_1sec.rtp"),
- kRtpIncoming));
- MARK();
- TEST_MUSTPASS(rtp_rtcp->StartRTPDump(0, GetFilename("dump_out_2sec.rtp"),
- kRtpOutgoing));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(rtp_rtcp->StopRTPDump(0, kRtpIncoming));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(rtp_rtcp->StopRTPDump(0, kRtpOutgoing));
- MARK();
-
- // Start/Stop tests:
- //
- // - only one file (called dump_in_200ms.rtp) should exist after this test
- //
- for (i = 0; i < 10; i++) {
- TEST_MUSTPASS(rtp_rtcp->StartRTPDump(0,
- GetFilename("dump_in_200ms.rtp")));
- MARK();
- SLEEP(200);
- TEST_MUSTPASS(rtp_rtcp->StopRTPDump(0));
- MARK();
- }
-
- // >> end of RTP dump APIs
- // ------------------------------------------------------------------------
- ANL();
-
- TEST(GetRTCPStatus);
- bool enabled;
- TEST_MUSTPASS(!rtp_rtcp->GetRTCPStatus(-1, enabled));
- MARK();
- TEST_MUSTPASS(rtp_rtcp->GetRTCPStatus(0, enabled));
- MARK(); // should be on by default
- TEST_MUSTPASS(enabled != true);
- ANL();
-
- TEST(SetRTCPStatus);
- TEST_MUSTPASS(rtp_rtcp->SetRTCPStatus(0, false));
- MARK();
- TEST_MUSTPASS(rtp_rtcp->GetRTCPStatus(0, enabled));
- TEST_MUSTPASS(enabled != false);
- MARK();
- SLEEP(2000);
- TEST_MUSTPASS(rtp_rtcp->SetRTCPStatus(0, true));
- MARK();
- TEST_MUSTPASS(rtp_rtcp->GetRTCPStatus(0, enabled));
- TEST_MUSTPASS(enabled != true);
- MARK();
- SLEEP(6000); // Make sure we get an RTCP packet
- ANL();
-
- TEST(CNAME);
- TEST_MUSTPASS(!rtp_rtcp->SetRTCP_CNAME(0, NULL));
- MARK();
- TEST_MUSTPASS(VE_RTP_RTCP_MODULE_ERROR != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(!rtp_rtcp->GetRemoteRTCP_CNAME(0, NULL));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- ANL();
-
- TEST(GetRemoteSSRC);
- unsigned int ssrc(0);
- TEST_MUSTPASS(rtp_rtcp->GetRemoteSSRC(0, ssrc));
- MARK();
- TEST_MUSTPASS(ssrc != 5678);
- ANL();
-
- TEST(GetRemoteCSRC); // only trivial tests added
- unsigned int csrcs[2];
- int n(0);
- TEST_MUSTPASS(!rtp_rtcp->GetRemoteCSRCs(1, csrcs));
- MARK();
- n = rtp_rtcp->GetRemoteCSRCs(0, csrcs);
- MARK();
- TEST_MUSTPASS(n != 0); // should be empty
- ANL();
-
- TEST(SetRTPObserver);
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(rtp_rtcp->RegisterRTPObserver(0, rtpObserver));
- TEST_MUSTPASS(rtp_rtcp->DeRegisterRTPObserver(0));
- TEST_MUSTPASS(rtp_rtcp->RegisterRTPObserver(0, rtpObserver));
- TEST_MUSTPASS(rtp_rtcp->SetLocalSSRC(0, 7777)); // force send SSRC to 7777
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(sleepTime);
- // verify that the new SSRC has been detected by the observer
- TEST_MUSTPASS(rtpObserver._SSRC != 7777);
- TEST_MUSTPASS(rtp_rtcp->DeRegisterRTPObserver(0));
- ANL();
-
- TEST(GetRTPKeepaliveStatus);
- unsigned char pt;
- int dT;
- TEST_MUSTPASS(!rtp_rtcp->GetRTPKeepaliveStatus(-1, enabled, pt, dT));
- MARK();
- TEST_MUSTPASS(rtp_rtcp->GetRTPKeepaliveStatus(0, enabled, pt, dT));
- MARK(); // should be off by default
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(pt != 255);
- TEST_MUSTPASS(dT != 0);
- ANL();
-
- TEST(SetRTPKeepaliveStatus);
- // stop send before changing the settings
- TEST_MUSTPASS(voe_base_->StopSend(0));
- // verify invalid input parameters
- TEST_MUSTPASS(!rtp_rtcp->SetRTPKeepaliveStatus(-1, true, 0, 15));
- MARK();
- TEST_MUSTPASS(!rtp_rtcp->SetRTPKeepaliveStatus(0, true, -1, 15));
- MARK();
- TEST_MUSTPASS(!rtp_rtcp->SetRTPKeepaliveStatus(0, true, 0, 61));
- MARK();
- TEST_MUSTPASS(rtp_rtcp->GetRTPKeepaliveStatus(0, enabled, pt, dT));
- MARK(); // should still be off
- TEST_MUSTPASS(enabled != false);
- // try valid settings
- TEST_MUSTPASS(rtp_rtcp->SetRTPKeepaliveStatus(0, true, 117));
- MARK();
- TEST_MUSTPASS(rtp_rtcp->GetRTPKeepaliveStatus(0, enabled, pt, dT));
- MARK(); // should be on now
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(pt != 117);
- TEST_MUSTPASS(dT != 15);
- // change from PT 99 to 121, as 99 is occupied
- TEST_MUSTPASS(rtp_rtcp->SetRTPKeepaliveStatus(0, true, 121, 3));
- MARK(); // on, PT=99, dT=3
- TEST_MUSTPASS(rtp_rtcp->GetRTPKeepaliveStatus(0, enabled, pt, dT));
- MARK();
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(pt != 121);
- TEST_MUSTPASS(dT != 3);
- ANL();
-
- // Make fresh restart (ensures that SSRC is randomized)
- TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
-
- SLEEP(100);
-
- TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
- true, true));
-
- SLEEP(8000);
-
- TEST(GetRemoteRTCPData);
- // Statistics based on received RTCP reports (i.e. statistics on the remote
- // side sent to us).
- unsigned int NTPHigh(0), NTPLow(0), timestamp(0), playoutTimestamp(0),
- jitter(0);
- unsigned short fractionLost(0);
- TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCPData(0, NTPHigh, NTPLow,
- timestamp, playoutTimestamp));
- TEST_LOG("\n NTPHigh = %u \n NTPLow = %u \n timestamp = %u \n "
- " playoutTimestamp = %u \n jitter = %u \n fractionLost = %hu \n",
- NTPHigh, NTPLow, timestamp, playoutTimestamp, jitter, fractionLost);
-
- unsigned int NTPHigh2(0), NTPLow2(0), timestamp2(0);
- unsigned int playoutTimestamp2(0), jitter2(0);
- unsigned short fractionLost2(0);
-
- TEST_LOG("take a new sample and ensure that the playout timestamp is "
- "maintained");
- SLEEP(100);
- TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCPData(0, NTPHigh2, NTPLow2, timestamp2,
- playoutTimestamp2, &jitter2,
- &fractionLost2));
- TEST_LOG("\n NTPHigh = %u \n NTPLow = %u \n timestamp = %u \n "
- " playoutTimestamp = %u \n jitter = %u \n fractionLost = %hu \n",
- NTPHigh2, NTPLow2, timestamp2, playoutTimestamp2, jitter2, fractionLost2);
- TEST_MUSTPASS(playoutTimestamp != playoutTimestamp2);
-
- TEST_LOG("wait for 8 seconds and ensure that the RTCP statistics is"
- " updated...");
- SLEEP(8000);
- TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCPData(0, NTPHigh2, NTPLow2,
- timestamp2, playoutTimestamp2,
- &jitter2, &fractionLost2));
- TEST_LOG("\n NTPHigh = %u \n NTPLow = %u \n timestamp = %u \n "
- " playoutTimestamp = %u \n jitter = %u \n fractionLost = %hu \n",
- NTPHigh2, NTPLow2, timestamp2, playoutTimestamp2, jitter2, fractionLost2);
- TEST_MUSTPASS((NTPHigh == NTPHigh2) && (NTPLow == NTPLow2));
- TEST_MUSTPASS(timestamp == timestamp2);
- TEST_MUSTPASS(playoutTimestamp == playoutTimestamp2);
-
-#ifdef WEBRTC_CODEC_RED
- //The following test is related to defect 4985 and 4986
- TEST_LOG("Turn FEC and VAD on and wait for 4 seconds and ensure that "
- "the jitter is still small...");
- CodecInst cinst;
-#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
- cinst.pltype = 104;
- strcpy(cinst.plname, "isac");
- cinst.plfreq = 32000;
- cinst.pacsize = 960;
- cinst.channels = 1;
- cinst.rate = 45000;
-#else
- cinst.pltype = 119;
- strcpy(cinst.plname, "isaclc");
- cinst.plfreq = 16000;
- cinst.pacsize = 320;
- cinst.channels = 1;
- cinst.rate = 40000;
-#endif
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
- TEST_MUSTPASS(codec->SetSendCodec(0, cinst));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(rtp_rtcp->SetFECStatus(0, true, -1));
- MARK();
- TEST_MUSTPASS(codec->SetVADStatus(0,true));
- SLEEP(4000);
- TEST_MUSTPASS(rtp_rtcp->GetRemoteRTCPData(0, NTPHigh2, NTPLow2, timestamp2,
- playoutTimestamp2, &jitter2,
- &fractionLost2));
- TEST_LOG("\n NTPHigh = %u \n NTPLow = %u \n timestamp = %u \n "
- " playoutTimestamp = %u \n jitter = %u \n fractionLost = %hu \n",
- NTPHigh2, NTPLow2, timestamp2, playoutTimestamp2, jitter2, fractionLost2);
- TEST_MUSTPASS(jitter2 > 1000)
- TEST_MUSTPASS(rtp_rtcp->SetFECStatus(0, false));
- MARK();
- //4985 and 4986 end
-#endif // #ifdef WEBRTC_CODEC_RED
- TEST(GetRTPStatistics);
- ANL();
- // Statistics summarized on local side based on received RTP packets.
- CallStatistics stats;
- // Call GetRTPStatistics over a longer period than 7.5 seconds
- // (=dT RTCP transmissions).
- unsigned int averageJitterMs, maxJitterMs, discardedPackets;
- SLEEP(1000);
- for (i = 0; i < 8; i++) {
- TEST_MUSTPASS(rtp_rtcp->GetRTPStatistics(0, averageJitterMs,
- maxJitterMs,
- discardedPackets));
- TEST_LOG(" %i) averageJitterMs = %u \n maxJitterMs = %u \n "
- " discardedPackets = %u \n", i, averageJitterMs, maxJitterMs,
- discardedPackets);
- SLEEP(1000);
- }
-
- TEST(RTCPStatistics #1);
- ANL();
- unsigned int packetsSent(0);
- unsigned int packetsReceived(0);
- for (i = 0; i < 8; i++)
- {
- TEST_MUSTPASS(rtp_rtcp->GetRTCPStatistics(0, stats));
- TEST_LOG(" %i) fractionLost = %hu \n cumulativeLost = %u \n "
- " extendedMax = %u \n jitterSamples = %u \n rttMs = %d \n",
- i, stats.fractionLost, stats.cumulativeLost,
- stats.extendedMax, stats.jitterSamples, stats.rttMs);
- TEST_LOG( " bytesSent = %d \n packetsSent = %d \n "
- " bytesReceived = %d \n packetsReceived = %d \n",
- stats.bytesSent, stats.packetsSent, stats.bytesReceived,
- stats.packetsReceived);
- if (i > 0)
- {
- TEST_LOG(" diff sent packets : %u (~50)\n",
- stats.packetsSent - packetsSent);
- TEST_LOG(" diff received packets: %u (~50)\n",
- stats.packetsReceived - packetsReceived);
- }
- packetsSent = stats.packetsSent;
- packetsReceived = stats.packetsReceived;
- SLEEP(1000);
- }
-
- TEST(RTCPStatistics #2);
- ANL();
- TEST_LOG("restart sending and ensure that the statistics is reset");
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- SLEEP(50); // ensures approx. two received packets
- TEST_MUSTPASS(rtp_rtcp->GetRTCPStatistics(0, stats));
- TEST_LOG("\n fractionLost = %hu \n cumulativeLost = %u \n "
- " extendedMax = %u \n jitterSamples = %u \n rttMs = %d \n",
- stats.fractionLost, stats.cumulativeLost,
- stats.extendedMax, stats.jitterSamples, stats.rttMs);
- TEST_LOG( " bytesSent = %d \n packetsSent = %d \n "
- " bytesReceived = %d \n packetsReceived = %d \n",
- stats.bytesSent, stats.packetsSent, stats.bytesReceived,
- stats.packetsReceived);
-
- TEST(RTCPStatistics #3);
- ANL();
- TEST_LOG("disable RTCP and verify that statistics is not corrupt");
- TEST_MUSTPASS(rtp_rtcp->SetRTCPStatus(0, false));
- SLEEP(250);
- TEST_MUSTPASS(rtp_rtcp->GetRTCPStatistics(0, stats));
- TEST_LOG("\n fractionLost = %hu \n cumulativeLost = %u \n "
- " extendedMax = %u \n jitterSamples = %u \n rttMs = %d \n",
- stats.fractionLost, stats.cumulativeLost,
- stats.extendedMax, stats.jitterSamples, stats.rttMs);
- TEST_LOG(" bytesSent = %d \n packetsSent = %d \n "
- "bytesReceived = %d \n packetsReceived = %d \n",
- stats.bytesSent, stats.packetsSent,
- stats.bytesReceived, stats.packetsReceived);
- TEST_MUSTPASS(rtp_rtcp->SetRTCPStatus(0, true));
-
- TEST(RTCPStatistics #4);
- ANL();
- TEST_LOG("restart receiving and check RX statistics");
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- SLEEP(50); // ensures approx. two received packets
- TEST_MUSTPASS(rtp_rtcp->GetRTCPStatistics(0, stats));
- TEST_LOG("\n fractionLost = %hu \n cumulativeLost = %u \n "
- " extendedMax = %u \n jitterSamples = %u \n rttMs = %d \n",
- stats.fractionLost, stats.cumulativeLost,
- stats.extendedMax, stats.jitterSamples,
- stats.rttMs);
- TEST_LOG(" bytesSent = %d \n packetsSent = %d \n "
- " bytesReceived = %d \n packetsReceived = %d \n",
- stats.bytesSent, stats.packetsSent,
- stats.bytesReceived, stats.packetsReceived);
-
- TEST(SendApplicationDefinedRTCPPacket);
- // just do some fail tests here
- TEST_MUSTPASS(voe_base_->StopSend(0));
- // should fail since sending is off
- TEST_MUSTPASS(!rtp_rtcp->SendApplicationDefinedRTCPPacket(
- 0, 0, 0, "abcdabcdabcdabcdabcdabcdabcdabcd", 32));
- MARK();
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_MUSTPASS(rtp_rtcp->SendApplicationDefinedRTCPPacket(
- 0, 0, 0, "abcdabcdabcdabcdabcdabcdabcdabcd", 32));
- MARK();
- TEST_MUSTPASS(rtp_rtcp->SetRTCPStatus(0, false));
- // should fail since RTCP is off
- TEST_MUSTPASS(!rtp_rtcp->SendApplicationDefinedRTCPPacket(
- 0, 0, 0, "abcdabcdabcdabcdabcdabcdabcdabcd", 32));
- MARK();
- TEST_MUSTPASS(rtp_rtcp->SetRTCPStatus(0, true));
- TEST_MUSTPASS(rtp_rtcp->SendApplicationDefinedRTCPPacket(
- 0, 0, 0, "abcdabcdabcdabcdabcdabcdabcdabcd", 32));
- MARK();
- // invalid data length
- TEST_MUSTPASS(!rtp_rtcp->SendApplicationDefinedRTCPPacket(
- 0, 0, 0, "abcdabcdabcdabcdabcdabcdabcdabc", 31));
- MARK();
- // invalid data vector
- TEST_MUSTPASS(!rtp_rtcp->SendApplicationDefinedRTCPPacket(0, 0, 0, NULL, 0));
- MARK();
- ANL();
-
-#ifdef WEBRTC_CODEC_RED
- TEST(SetFECStatus);
- ANL();
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- cinst.pltype = 126;
- strcpy(cinst.plname, "red");
- cinst.plfreq = 8000;
- cinst.pacsize = 0;
- cinst.channels = 1;
- cinst.rate = 0;
- TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
-#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
- cinst.pltype = 104;
- strcpy(cinst.plname, "isac");
- cinst.plfreq = 32000;
- cinst.pacsize = 960;
- cinst.channels = 1;
- cinst.rate = 45000;
-#else
- cinst.pltype = 119;
- strcpy(cinst.plname, "isaclc");
- cinst.plfreq = 16000;
- cinst.pacsize = 320;
- cinst.channels = 1;
- cinst.rate = 40000;
-#endif
- // We have to re-register the audio codec payload type as stopReceive will
- // clean the database
- TEST_MUSTPASS(codec->SetRecPayloadType(0, cinst));
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 8000));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_LOG("Start playing a file as microphone again \n");
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
- true, true));
- TEST_MUSTPASS(rtp_rtcp->SetFECStatus(0, true, 126));
- MARK();
- TEST_LOG("Should sound OK with FEC enabled\n");
- SLEEP(4000);
- TEST_MUSTPASS(rtp_rtcp->SetFECStatus(0, false));
- MARK();
-#endif // #ifdef WEBRTC_CODEC_RED
- TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->Terminate());
-
- ANL();
- AOK();
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// VoEExtendedTest::TestVideoSync
-// ----------------------------------------------------------------------------
-
-int VoEExtendedTest::TestVideoSync()
-{
- PrepareTest("VideoSync");
-
- VoEBase* voe_base_ = _mgr.BasePtr();
- VoEVideoSync* vsync = _mgr.VideoSyncPtr();
-
- // check if this interface is supported
- if (!vsync)
- {
- TEST_LOG("VoEVideoSync is not supported!");
- return -1;
- }
-
-#ifdef _USE_EXTENDED_TRACE_
- TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename(
- "VoEVideoSync_trace.txt")));
- TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
- kTraceStateInfo |
- kTraceWarning |
- kTraceError |
- kTraceCritical |
- kTraceApiCall |
- kTraceMemory |
- kTraceInfo));
-#endif
-
- TEST_MUSTPASS(voe_base_->Init());
- TEST_MUSTPASS(voe_base_->CreateChannel());
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
-
- ///////////////////////////
- // Actual test starts here
-
- TEST(SetInitTimestamp);
- ANL();
- TEST_MUSTPASS(!vsync->SetInitTimestamp(0, 12345));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(vsync->SetInitTimestamp(0, 12345));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- MARK();
- SLEEP(1000);
- AOK();
- ANL();
-
- TEST(SetInitSequenceNumber);
- ANL();
- TEST_MUSTPASS(!vsync->SetInitSequenceNumber(0, 123));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- MARK();
- SLEEP(1000);
- TEST_MUSTPASS(vsync->SetInitSequenceNumber(0, 123));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- MARK();
- SLEEP(1000);
- AOK();
- ANL();
-
- unsigned int timeStamp;
- TEST(GetPlayoutTimestamp);
- ANL();
- TEST_MUSTPASS(vsync->GetPlayoutTimestamp(0, timeStamp));
- TEST_LOG("GetPlayoutTimestamp: %u", timeStamp);
- SLEEP(1000);
- TEST_MUSTPASS(vsync->GetPlayoutTimestamp(0, timeStamp));
- TEST_LOG(" %u", timeStamp);
- SLEEP(1000);
- TEST_MUSTPASS(vsync->GetPlayoutTimestamp(0, timeStamp));
- TEST_LOG(" %u\n", timeStamp);
- AOK();
- ANL();
-
- TEST(SetMinimumPlayoutDelay);
- ANL();
- TEST_MUSTPASS(!vsync->SetMinimumPlayoutDelay(0, -1));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
- TEST_MUSTPASS(!vsync->SetMinimumPlayoutDelay(0, 5000));
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- MARK();
-
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->Terminate());
-
- AOK();
- ANL();
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// VoEExtendedTest::TestVolumeControl
-// ----------------------------------------------------------------------------
-
-int VoEExtendedTest::TestVolumeControl()
-{
- PrepareTest("TestVolumeControl");
-
- VoEBase* voe_base_ = _mgr.BasePtr();
- VoEVolumeControl* volume = _mgr.VolumeControlPtr();
-#ifdef _TEST_FILE_
- VoEFile* file = _mgr.FilePtr();
-#endif
-#ifdef _TEST_HARDWARE_
- VoEHardware* hardware = _mgr.HardwarePtr();
-#endif
-
-#ifdef _USE_EXTENDED_TRACE_
- TEST_MUSTPASS(VoiceEngine::SetTraceFile(
- GetFilename("VoEVolumeControl_trace.txt")));
- TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
- kTraceStateInfo |
- kTraceWarning |
- kTraceError |
- kTraceCritical |
- kTraceApiCall |
- kTraceMemory |
- kTraceInfo));
-#endif
-
- TEST_MUSTPASS(voe_base_->Init());
- TEST_MUSTPASS(voe_base_->CreateChannel());
-#if (defined _TEST_HARDWARE_ && (!defined(MAC_IPHONE) && \
- !defined(WEBRTC_ANDROID)))
-#if defined(_WIN32)
- TEST_MUSTPASS(hardware->SetRecordingDevice(-1));
- TEST_MUSTPASS(hardware->SetPlayoutDevice(-1));
-#else
- TEST_MUSTPASS(hardware->SetRecordingDevice(0));
- TEST_MUSTPASS(hardware->SetPlayoutDevice(0));
-#endif
-#endif
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0, 12345));
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 12345, "127.0.0.1"));
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
-#ifdef _TEST_FILE_
- TEST_MUSTPASS(file->StartPlayingFileAsMicrophone(0, _mgr.AudioFilename(),
- true, true));
-#endif
-
- ////////////////////////////
- // Actual test starts here
-
-#if !defined(MAC_IPHONE)
- TEST(SetSpeakerVolume);
- ANL();
- TEST_MUSTPASS(-1 != volume->SetSpeakerVolume(256));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- ANL();
-#endif // #if !defined(MAC_IPHONE)
-
-#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
- TEST(SetMicVolume); ANL();
- TEST_MUSTPASS(-1 != volume->SetMicVolume(256)); MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- ANL();
-#endif // #if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
-
-#if !defined(MAC_IPHONE)
- TEST(SetChannelOutputVolumeScaling);
- ANL();
- TEST_MUSTPASS(-1 != volume->SetChannelOutputVolumeScaling(0, (float)-0.1));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(-1 != volume->SetChannelOutputVolumeScaling(0, (float)10.1));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- ANL();
-#endif // #if !defined(MAC_IPHONE)
-#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
- TEST(SetOutputVolumePan);
- ANL();
- TEST_MUSTPASS(-1 != volume->SetOutputVolumePan(-1, (float)-0.1,
- (float)1.0));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(-1 != volume->SetOutputVolumePan(-1, (float)1.1,
- (float)1.0));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(-1 != volume->SetOutputVolumePan(-1, (float)1.0,
- (float)-0.1));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(-1 != volume->SetOutputVolumePan(-1, (float)1.0,
- (float)1.1));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- ANL();
-
- TEST(SetChannelOutputVolumePan);
- ANL();
- TEST_MUSTPASS(-1 != volume->SetOutputVolumePan(0, (float)-0.1,
- (float)1.0));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(-1 != volume->SetOutputVolumePan(0, (float)1.1,
- (float)1.0));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(-1 != volume->SetOutputVolumePan(0, (float)1.0,
- (float)-0.1));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- TEST_MUSTPASS(-1 != volume->SetOutputVolumePan(0, (float)1.0,
- (float)1.1));
- MARK();
- TEST_MUSTPASS(VE_INVALID_ARGUMENT != voe_base_->LastError());
- ANL();
-#endif // #if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
-#ifdef _TEST_FILE_
- TEST_MUSTPASS(file->StopPlayingFileAsMicrophone(0));
-#endif
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->Terminate());
-
- AOK();
- ANL();
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// VoEExtendedTest::TestAPM
-// ----------------------------------------------------------------------------
-
-int VoEExtendedTest::TestAPM() {
- PrepareTest("AudioProcessing");
-
- VoEBase* voe_base_ = _mgr.BasePtr();
- VoEAudioProcessing* apm = _mgr.APMPtr();
-
- //#ifdef _USE_EXTENDED_TRACE_
- TEST_MUSTPASS(VoiceEngine::SetTraceFile(GetFilename("apm_trace.txt")));
- TEST_MUSTPASS(VoiceEngine::SetTraceFilter(kTraceStateInfo |
- kTraceStateInfo |
- kTraceWarning |
- kTraceError |
- kTraceCritical |
- kTraceApiCall |
- kTraceMemory |
- kTraceInfo));
- //#endif
-
- TEST_MUSTPASS(voe_base_->Init());
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- ///////////////////////////
- // Actual test starts here
-
- int i;
- bool enabled;
-
- //////
- // EC
-
- const int ECSleep = 0;
- const int ECIterations = 10;
-
- EcModes ECmode(kEcAec);
- AecmModes AECMmode(kAecmSpeakerphone);
- bool enabledCNG(false);
-
-#if (defined(MAC_IPHONE) || defined(WEBRTC_ANDROID))
- const EcModes ECmodeDefault(kEcAecm);
-#else
- const EcModes ECmodeDefault(kEcAec);
-#endif
-
- // verify default settings (should be OFF and mode as above)
- TEST_MUSTPASS(apm->GetEcStatus(enabled, ECmode));
- TEST_LOG("EC: enabled=%d, ECmode=%d\n", enabled, ECmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(ECmode != ECmodeDefault);
-
- // set EC defaults
- TEST_MUSTPASS(apm->SetEcStatus(false, kEcDefault));
- TEST_MUSTPASS(apm->GetEcStatus(enabled, ECmode));
- TEST_LOG("EC: enabled=%d, ECmode=%d\n", enabled, ECmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(ECmode != ECmodeDefault);
- SLEEP(ECSleep);
-
-#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
- // set kEcAec mode
- TEST_MUSTPASS(apm->SetEcStatus(true, kEcAec));
- TEST_MUSTPASS(apm->GetEcStatus(enabled, ECmode));
- TEST_LOG("EC: enabled=%d, ECmode=%d\n", enabled, ECmode);
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(ECmode != kEcAec);
- SLEEP(ECSleep);
-
- // set kEcConference mode
- TEST_MUSTPASS(apm->SetEcStatus(true, kEcConference));
- TEST_MUSTPASS(apm->GetEcStatus(enabled, ECmode));
- TEST_LOG("EC: enabled=%d, ECmode=%d\n", enabled, ECmode);
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(ECmode != kEcAec);
- SLEEP(ECSleep);
-#endif // #if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
- // get default AECM mode, should the kAecmSpeakerphone as default
- TEST_MUSTPASS(apm->GetAecmMode(AECMmode, enabledCNG));
- TEST_MUSTPASS(AECMmode != kAecmSpeakerphone);
- TEST_MUSTPASS(enabledCNG != true);
- TEST_MUSTPASS(apm->SetAecmMode(kAecmQuietEarpieceOrHeadset, false));
- TEST_MUSTPASS(apm->GetAecmMode(AECMmode, enabledCNG));
- TEST_LOG("AECM: mode=%d, CNG: mode=%d\n", AECMmode,
- enabledCNG);
- TEST_MUSTPASS(AECMmode != kAecmQuietEarpieceOrHeadset);
- TEST_MUSTPASS(enabledCNG != false);
-
- // set kEcAecm mode
- TEST_MUSTPASS(apm->SetEcStatus(true, kEcAecm));
- TEST_MUSTPASS(apm->GetEcStatus(enabled, ECmode));
- TEST_LOG("EC: enabled=%d, ECmode=%d\n", enabled, ECmode);
- TEST_MUSTPASS(enabled != true);
- TEST_MUSTPASS(ECmode != kEcAecm);
- SLEEP(ECSleep);
-
- // AECM mode, get and set
- TEST_MUSTPASS(apm->GetAecmMode(AECMmode, enabledCNG));
- TEST_MUSTPASS(AECMmode != kAecmQuietEarpieceOrHeadset);
- TEST_MUSTPASS(enabledCNG != false);
- TEST_MUSTPASS(apm->SetAecmMode(kAecmEarpiece, true));
- TEST_MUSTPASS(apm->GetAecmMode(AECMmode, enabledCNG));
- TEST_LOG("AECM: mode=%d, CNG: mode=%d\n", AECMmode,
- enabledCNG);
- TEST_MUSTPASS(AECMmode != kAecmEarpiece);
- TEST_MUSTPASS(enabledCNG != true);
- TEST_MUSTPASS(apm->SetAecmMode(kAecmEarpiece, false));
- TEST_MUSTPASS(apm->GetAecmMode(AECMmode, enabledCNG));
- TEST_LOG("AECM: mode=%d, CNG: mode=%d\n", AECMmode,
- enabledCNG);
- TEST_MUSTPASS(AECMmode != kAecmEarpiece);
- TEST_MUSTPASS(enabledCNG != false);
- TEST_MUSTPASS(apm->SetAecmMode(kAecmLoudEarpiece, true));
- TEST_MUSTPASS(apm->GetAecmMode(AECMmode, enabledCNG));
- TEST_LOG("AECM: mode=%d, CNG: mode=%d\n", AECMmode,
- enabledCNG);
- TEST_MUSTPASS(AECMmode != kAecmLoudEarpiece);
- TEST_MUSTPASS(enabledCNG != true);
- TEST_MUSTPASS(apm->SetAecmMode(kAecmSpeakerphone, false));
- TEST_MUSTPASS(apm->GetAecmMode(AECMmode, enabledCNG));
- TEST_LOG("AECM: mode=%d, CNG: mode=%d\n", AECMmode,
- enabledCNG);
- TEST_MUSTPASS(AECMmode != kAecmSpeakerphone);
- TEST_MUSTPASS(enabledCNG != false);
- TEST_MUSTPASS(apm->SetAecmMode(kAecmLoudSpeakerphone, true));
- TEST_MUSTPASS(apm->GetAecmMode(AECMmode, enabledCNG));
- TEST_LOG("AECM: mode=%d, CNG: mode=%d\n", AECMmode,
- enabledCNG);
- TEST_MUSTPASS(AECMmode != kAecmLoudSpeakerphone);
- TEST_MUSTPASS(enabledCNG != true);
-
- // verify that all modes are maintained when EC is disabled
- TEST_MUSTPASS(apm->SetEcStatus(false));
- TEST_MUSTPASS(apm->GetEcStatus(enabled, ECmode));
- TEST_LOG("EC: enabled=%d, ECmode=%d\n", enabled, ECmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(ECmode != kEcAecm);
- SLEEP(ECSleep);
-
- // restore defaults
- TEST_MUSTPASS(apm->SetEcStatus(true, kEcDefault));
- TEST_MUSTPASS(apm->SetEcStatus(false, kEcUnchanged));
- TEST_MUSTPASS(apm->GetEcStatus(enabled, ECmode));
- TEST_LOG("EC: enabled=%d, ECmode=%d\n", enabled, ECmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(ECmode != ECmodeDefault);
- SLEEP(ECSleep);
-
- // enable/disable many times in a row
- for (i = 0; i < ECIterations; i++) {
- TEST_MUSTPASS(apm->SetEcStatus(true));
- TEST_MUSTPASS(apm->SetEcStatus(false));
- }
- TEST_MUSTPASS(apm->GetEcStatus(enabled, ECmode));
- TEST_LOG("EC: enabled=%d, ECmode=%d\n", enabled, ECmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(ECmode != ECmodeDefault);
- SLEEP(ECSleep);
-
- ///////
- // AGC
-
- const int AGCSleep = 0;
- const int AGCIterations = 10;
-
- AgcModes AGCmode(kAgcAdaptiveAnalog);
-
-#if (defined(MAC_IPHONE) || defined(WEBRTC_ANDROID))
- bool enabledDefault = false;
- AgcModes AGCmodeDefault(kAgcAdaptiveDigital);
-#else
- bool enabledDefault = true;
- AgcModes AGCmodeDefault(kAgcAdaptiveAnalog);
-#endif
-
- // verify default settings (should be as above)
- TEST_MUSTPASS(apm->GetAgcStatus(enabled, AGCmode));
- TEST_LOG("AGC: enabled=%d, AGCmode=%d\n", enabled, AGCmode);
- TEST_MUSTPASS(enabled != enabledDefault);
- TEST_MUSTPASS(AGCmode != AGCmodeDefault);
-
- // set default AGC mode
- TEST_MUSTPASS(apm->SetAgcStatus(false, kAgcDefault));
- TEST_MUSTPASS(apm->GetAgcStatus(enabled, AGCmode));
- TEST_LOG("AGC: enabled=%d, AGCmode=%d\n", enabled, AGCmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(AGCmode != AGCmodeDefault);
- SLEEP(AGCSleep);
-
- // set kAgcFixedDigital mode
- TEST_MUSTPASS(apm->SetAgcStatus(true, kAgcFixedDigital));
- TEST_MUSTPASS(apm->GetAgcStatus(enabled, AGCmode));
- TEST_LOG("AGC: enabled=%d, AGCmode=%d\n", enabled, AGCmode);
- TEST_MUSTPASS(AGCmode != kAgcFixedDigital);
- SLEEP(AGCSleep);
-
-#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
- // set kAgcAdaptiveAnalog mode
- TEST_MUSTPASS(apm->SetAgcStatus(true, kAgcAdaptiveAnalog));
- TEST_MUSTPASS(apm->GetAgcStatus(enabled, AGCmode));
- TEST_LOG("AGC: enabled=%d, AGCmode=%d\n", enabled, AGCmode);
- TEST_MUSTPASS(AGCmode != kAgcAdaptiveAnalog);
- SLEEP(AGCSleep);
-#endif // #if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
- // set kAgcAdaptiveDigital mode
- TEST_MUSTPASS(apm->SetAgcStatus(true, kAgcAdaptiveDigital));
- TEST_MUSTPASS(apm->GetAgcStatus(enabled, AGCmode));
- TEST_LOG("AGC: enabled=%d, AGCmode=%d\n", enabled, AGCmode);
- TEST_MUSTPASS(AGCmode != kAgcAdaptiveDigital);
- SLEEP(AGCSleep);
-
- // verify that mode is maintained when AGC is disabled
- TEST_MUSTPASS(apm->SetAgcStatus(false));
- TEST_MUSTPASS(apm->GetAgcStatus(enabled, AGCmode));
- TEST_LOG("AGC: enabled=%d, AGCmode=%d\n", enabled, AGCmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(AGCmode != kAgcAdaptiveDigital);
- SLEEP(AGCSleep);
-
- // restore default AGC
- TEST_MUSTPASS(apm->SetAgcStatus(enabledDefault, kAgcDefault));
- TEST_MUSTPASS(apm->GetAgcStatus(enabled, AGCmode));
- TEST_LOG("AGC: enabled=%d, AGCmode=%d\n", enabled, AGCmode);
- TEST_MUSTPASS(enabled != enabledDefault);
- TEST_MUSTPASS(AGCmode != AGCmodeDefault);
- SLEEP(AGCSleep);
-
- // enable/disable many times in a row
- for (i = 0; i < AGCIterations; i++)
- {
- TEST_MUSTPASS(apm->SetAgcStatus(true));
- TEST_MUSTPASS(apm->SetAgcStatus(false));
- }
- TEST_MUSTPASS(apm->GetAgcStatus(enabled, AGCmode));
- TEST_LOG("AGC: enabled=%d, AGCmode=%d\n", enabled, AGCmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(AGCmode != AGCmodeDefault);
-
- // --- Set/AGCConfig --
-
- //
- // targetLeveldBOv : [0, 31] (default 3)
- // digitalCompressionGaindB: [0, 90] (default 9)
- // limiterEnable : 0: Off, 1: On (default)
-
- AgcConfig agcConfig;
- AgcConfig agcConfigDefault;
-
- const unsigned short targetLeveldBOvDefault = 3;
- const unsigned short digitalCompressionGaindBDefault = 9;
- const bool limiterEnableDefault = true;
-
- const unsigned short targetLeveldBOvMax = 31;
- const unsigned short digitalCompressionGaindBMax = 90;
-
- // verify default configuration
- TEST_MUSTPASS(apm->GetAgcConfig(agcConfigDefault));
- TEST_LOG("AGC: targetLeveldBOv=%d, digitalCompressionGaindB=%d, "
- "limiterEnable=%d\n",
- agcConfigDefault.targetLeveldBOv,
- agcConfigDefault.digitalCompressionGaindB,
- agcConfigDefault.limiterEnable);
- TEST_MUSTPASS(agcConfigDefault.targetLeveldBOv != targetLeveldBOvDefault);
- TEST_MUSTPASS(agcConfigDefault.digitalCompressionGaindB !=
- digitalCompressionGaindBDefault);
- TEST_MUSTPASS(agcConfigDefault.limiterEnable != limiterEnableDefault);
-
- // verify that invalid (out-of-range) parameters are detected
- agcConfig = agcConfigDefault;
- agcConfig.targetLeveldBOv = targetLeveldBOvMax + 1;
- TEST_MUSTPASS(!apm->SetAgcConfig(agcConfig));
- int err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_APM_ERROR);
- agcConfig = agcConfigDefault;
- agcConfig.digitalCompressionGaindB
- = digitalCompressionGaindBMax + 1;
- TEST_MUSTPASS(!apm->SetAgcConfig(agcConfig));
-
- AgcConfig agcConfigSet;
- agcConfigSet.digitalCompressionGaindB = 17;
- agcConfigSet.targetLeveldBOv = 11;
- agcConfigSet.limiterEnable = false;
-
- // try some set/get operations using valid settings
- TEST_MUSTPASS(apm->SetAgcConfig(agcConfigDefault));
- TEST_MUSTPASS(apm->GetAgcConfig(agcConfig));
- TEST_MUSTPASS(agcConfig.targetLeveldBOv != targetLeveldBOvDefault);
- TEST_MUSTPASS(agcConfig.digitalCompressionGaindB !=
- digitalCompressionGaindBDefault);
- TEST_MUSTPASS(agcConfig.limiterEnable != limiterEnableDefault);
-
- TEST_MUSTPASS(apm->SetAgcConfig(agcConfigSet));
- TEST_MUSTPASS(apm->GetAgcConfig(agcConfig));
- TEST_MUSTPASS(agcConfig.targetLeveldBOv != agcConfigSet.targetLeveldBOv);
- TEST_MUSTPASS(agcConfig.digitalCompressionGaindB !=
- agcConfigSet.digitalCompressionGaindB);
- TEST_MUSTPASS(agcConfig.limiterEnable != agcConfigSet.limiterEnable);
-
- // restore default AGC config
- TEST_MUSTPASS(apm->SetAgcConfig(agcConfigDefault));
- SLEEP(AGCSleep);
-
- //////
- // NS
-
- const int NSSleep = 0;
- const int NSIterations = 10;
-
- NsModes NSmode(kNsHighSuppression);
- NsModes NSmodeDefault(kNsModerateSuppression);
-
- // verify default settings (should be OFF and mode as above)
- TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
- TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(NSmode != NSmodeDefault);
-
- // enable default NS settings
- // must set a value first time!
- TEST_MUSTPASS(apm->SetNsStatus(false, kNsDefault));
- TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
- TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(NSmode != NSmodeDefault);
- SLEEP(NSSleep);
-
- // set kNsLowSuppression mode
- TEST_MUSTPASS(apm->SetNsStatus(true, kNsLowSuppression));
- TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
- TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
- TEST_MUSTPASS(NSmode != kNsLowSuppression);
- SLEEP(NSSleep);
-
- // set kNsModerateSuppression mode
- TEST_MUSTPASS(apm->SetNsStatus(true, kNsModerateSuppression));
- TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
- TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
- TEST_MUSTPASS(NSmode != kNsModerateSuppression);
- SLEEP(NSSleep);
-
- // set kNsHighSuppression mode
- TEST_MUSTPASS(apm->SetNsStatus(true, kNsHighSuppression));
- TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
- TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
- TEST_MUSTPASS(NSmode != kNsHighSuppression);
- SLEEP(NSSleep);
-
- // set kNsVeryHighSuppression mode
- TEST_MUSTPASS(apm->SetNsStatus(true, kNsVeryHighSuppression));
- TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
- TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
- TEST_MUSTPASS(NSmode != kNsVeryHighSuppression);
- SLEEP(NSSleep);
-
- // set kNsVeryHighSuppression mode
- TEST_MUSTPASS(apm->SetNsStatus(true, kNsConference));
- TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
- TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
- TEST_MUSTPASS(NSmode != kNsHighSuppression);
- SLEEP(NSSleep);
-
- // verify that mode is maintained when NS is disabled
- TEST_MUSTPASS(apm->SetNsStatus(false));
- TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
- TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(NSmode != kNsHighSuppression);
- SLEEP(NSSleep);
-
- // restore default NS
- TEST_MUSTPASS(apm->SetNsStatus(true, kNsDefault));
- TEST_MUSTPASS(apm->SetNsStatus(false));
- TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
- TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(NSmode != NSmodeDefault);
- SLEEP(NSSleep);
-
- // enable/disable many times in a row
- for (i = 0; i < NSIterations; i++) {
- TEST_MUSTPASS(apm->SetNsStatus(true));
- TEST_MUSTPASS(apm->SetNsStatus(false));
- }
- TEST_MUSTPASS(apm->GetNsStatus(enabled, NSmode));
- TEST_LOG("NS: enabled=%d, NSmode=%d\n", enabled, NSmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(NSmode != NSmodeDefault);
- SLEEP(NSSleep);
-
- //////////////////////////////////
- // Ec Metrics
-
-#if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
- TEST(GetEcMetricsStatus);
- ANL();
- TEST(SetEcMetricsStatus);
- ANL();
- TEST_MUSTPASS(apm->GetEcMetricsStatus(enabled));
- MARK();
- TEST_MUSTPASS(enabled != false);
- MARK(); // should be OFF by default
- TEST_MUSTPASS(apm->SetEcMetricsStatus(true));
- MARK();
- TEST_MUSTPASS(apm->GetEcMetricsStatus(enabled));
- MARK();
- TEST_MUSTPASS(enabled != true);
- MARK();
- TEST_MUSTPASS(apm->SetEcMetricsStatus(false));
- MARK();
- TEST_MUSTPASS(apm->GetEcMetricsStatus(enabled));
- MARK();
- TEST_MUSTPASS(enabled != false);
- MARK();
- AOK();
- ANL();
-
- TEST(GetEchoMetrics);
- ANL();
-
- int ERL, ERLE, RERL, A_NLP;
- TEST_MUSTPASS(-1 != apm->GetEchoMetrics(ERL, ERLE, RERL, A_NLP));
- MARK(); // Should fail since not activated.
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_APM_ERROR);
- TEST_MUSTPASS(apm->SetEcMetricsStatus(true));
- TEST_MUSTPASS(-1 != apm->GetEchoMetrics(ERL, ERLE, RERL, A_NLP));
- MARK(); // Should fail since AEC is off.
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_APM_ERROR);
- TEST_MUSTPASS(apm->SetEcStatus(true));
- TEST_MUSTPASS(apm->GetEchoMetrics(ERL, ERLE, RERL, A_NLP));
- MARK(); // Should work now.
- TEST_LOG("\nEcho: ERL=%d, ERLE=%d, RERL=%d, A_NLP=%d [dB]\n",
- ERL, ERLE, RERL, A_NLP);
- TEST_MUSTPASS(apm->SetEcMetricsStatus(false));
- TEST_MUSTPASS(apm->SetEcStatus(false));
- AOK();
- ANL();
-
- TEST(GetEcDelayMetrics);
- ANL();
-
- int delay_median = 0;
- int delay_std = 0;
- TEST_MUSTPASS(-1 != apm->GetEcDelayMetrics(delay_median, delay_std));
- MARK(); // Should fail since not activated.
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_APM_ERROR);
- TEST_MUSTPASS(apm->SetEcMetricsStatus(true));
- TEST_MUSTPASS(-1 != apm->GetEcDelayMetrics(delay_median, delay_std));
- MARK(); // Should fail since AEC is off.
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_APM_ERROR);
- TEST_MUSTPASS(apm->SetEcStatus(true));
- TEST_MUSTPASS(apm->GetEcDelayMetrics(delay_median, delay_std));
- MARK(); // Should work now.
- TEST_LOG("\nEC Delay: median=%d, std=%d [ms]\n", delay_median, delay_std);
- TEST_MUSTPASS(apm->SetEcMetricsStatus(false));
- TEST_MUSTPASS(apm->SetEcStatus(false));
- AOK();
- ANL();
-
-#endif // #if (!defined(MAC_IPHONE) && !defined(WEBRTC_ANDROID))
- // far-end AudioProcessing
- ///////
- // AGC
-
- AgcModes rxAGCmode(kAgcAdaptiveDigital);
- AgcModes rxAGCmodeDefault(kAgcAdaptiveDigital);
- bool rxEnabledDefault = false;
-
- // verify default settings (should be as above)
- TEST_MUSTPASS(apm->GetRxAgcStatus(0, enabled, rxAGCmode));
- TEST_LOG("rxAGC: enabled=%d, AGCmode=%d\n", enabled,
- rxAGCmode);
- TEST_MUSTPASS(enabled != rxEnabledDefault);
- TEST_MUSTPASS(rxAGCmode != rxAGCmodeDefault);
-
- // set default AGC mode
- TEST_MUSTPASS(apm->SetRxAgcStatus(0, false, kAgcDefault));
- TEST_MUSTPASS(apm->GetRxAgcStatus(0, enabled, rxAGCmode));
- TEST_LOG("rxAGC: enabled=%d, AGCmode=%d\n", enabled,
- rxAGCmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(rxAGCmode != rxAGCmodeDefault);
- SLEEP(AGCSleep);
-
- // set kAgcAdaptiveAnalog mode, should fail
- TEST_MUSTPASS(!apm->SetRxAgcStatus(0, true, kAgcAdaptiveAnalog));
-
- // set kAgcFixedDigital mode
- TEST_MUSTPASS(apm->SetRxAgcStatus(0, true, kAgcFixedDigital));
- TEST_MUSTPASS(apm->GetRxAgcStatus(0, enabled, rxAGCmode));
- TEST_LOG("rxAGC: enabled=%d, AGCmode=%d\n", enabled,
- rxAGCmode);
- TEST_MUSTPASS(rxAGCmode != kAgcFixedDigital);
- SLEEP(AGCSleep);
-
- // set kAgcAdaptiveDigital mode
- TEST_MUSTPASS(apm->SetRxAgcStatus(0, true, kAgcAdaptiveDigital));
- TEST_MUSTPASS(apm->GetRxAgcStatus(0, enabled, rxAGCmode));
- TEST_LOG("rxAGC: enabled=%d, AGCmode=%d\n", enabled,
- rxAGCmode);
- TEST_MUSTPASS(rxAGCmode != kAgcAdaptiveDigital);
- SLEEP(AGCSleep);
-
- // verify that mode is maintained when AGC is disabled
- TEST_MUSTPASS(apm->SetRxAgcStatus(0, false));
- TEST_MUSTPASS(apm->GetRxAgcStatus(0, enabled, rxAGCmode));
- TEST_LOG("rxAGC: enabled=%d, AGCmode=%d\n", enabled,
- rxAGCmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(rxAGCmode != kAgcAdaptiveDigital);
- SLEEP(AGCSleep);
-
- // restore default AGC
- TEST_MUSTPASS(apm->SetRxAgcStatus(0, enabledDefault, kAgcDefault));
- TEST_MUSTPASS(apm->GetRxAgcStatus(0, enabled, rxAGCmode));
- TEST_LOG("rxAGC: enabled=%d, AGCmode=%d\n", enabled,
- rxAGCmode);
- TEST_MUSTPASS(enabled != enabledDefault);
- TEST_MUSTPASS(rxAGCmode != rxAGCmodeDefault);
- SLEEP(AGCSleep);
-
- // enable/disable many times in a row
- for (i = 0; i < AGCIterations; i++) {
- TEST_MUSTPASS(apm->SetRxAgcStatus(0, true));
- TEST_MUSTPASS(apm->SetRxAgcStatus(0, false));
- }
- TEST_MUSTPASS(apm->GetRxAgcStatus(0, enabled, rxAGCmode));
- TEST_LOG("rxAGC: enabled=%d, AGCmode=%d\n", enabled,
- rxAGCmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(rxAGCmode != rxAGCmodeDefault);
-
- // --- Set/GetAgcConfig --
-
-
- // targetLeveldBOv : [0, 31] (default 3)
- // digitalCompressionGaindB: [0, 90] (default 9)
- // limiterEnable : 0: Off, 1: On (default)
-
- AgcConfig rxAGCConfig;
- AgcConfig rxAGCConfigDefault;
-
- const unsigned short rxTargetLeveldBOvDefault = 3;
- const unsigned short rxDigitalCompressionGaindBDefault = 9;
- const bool rxLimiterEnableDefault = true;
-
- const unsigned short rxTargetLeveldBOvMax = 31;
- const unsigned short rxDigitalCompressionGaindBMax = 90;
-
- // verify default configuration
- TEST_MUSTPASS(apm->GetRxAgcConfig(0, rxAGCConfigDefault));
- TEST_LOG(
- "rxAGC: targetLeveldBOv=%u, digitalCompressionGaindB=%u, "
- "limiterEnable=%d\n",
- rxAGCConfigDefault.targetLeveldBOv,
- rxAGCConfigDefault.digitalCompressionGaindB,
- rxAGCConfigDefault.limiterEnable);
- TEST_MUSTPASS(rxAGCConfigDefault.targetLeveldBOv !=
- rxTargetLeveldBOvDefault);
- TEST_MUSTPASS(rxAGCConfigDefault.digitalCompressionGaindB !=
- rxDigitalCompressionGaindBDefault);
- TEST_MUSTPASS(rxAGCConfigDefault.limiterEnable != rxLimiterEnableDefault);
-
- // verify that invalid (out-of-range) parameters are detected
- rxAGCConfig = rxAGCConfigDefault;
- rxAGCConfig.targetLeveldBOv = rxTargetLeveldBOvMax + 1;
- TEST_MUSTPASS(!apm->SetRxAgcConfig(0, rxAGCConfig));
- int rxErr = voe_base_->LastError();
- TEST_MUSTPASS(rxErr != VE_APM_ERROR);
- rxAGCConfig = rxAGCConfigDefault;
- rxAGCConfig.digitalCompressionGaindB
- = rxDigitalCompressionGaindBMax + 1;
- TEST_MUSTPASS(!apm->SetRxAgcConfig(0, rxAGCConfig));
-
- AgcConfig rxAGCConfigSet;
- rxAGCConfigSet.digitalCompressionGaindB = 17;
- rxAGCConfigSet.targetLeveldBOv = 11;
- rxAGCConfigSet.limiterEnable = false;
-
- // try some set/get operations using valid settings
- TEST_MUSTPASS(apm->SetRxAgcConfig(0, rxAGCConfigDefault));
- TEST_MUSTPASS(apm->GetRxAgcConfig(0, rxAGCConfig));
- TEST_MUSTPASS(rxAGCConfig.targetLeveldBOv != rxTargetLeveldBOvDefault);
- TEST_MUSTPASS(rxAGCConfig.digitalCompressionGaindB !=
- rxDigitalCompressionGaindBDefault);
- TEST_MUSTPASS(rxAGCConfig.limiterEnable != rxLimiterEnableDefault);
-
- TEST_MUSTPASS(apm->SetRxAgcConfig(0, rxAGCConfigSet));
- TEST_MUSTPASS(apm->GetRxAgcConfig(0, rxAGCConfig));
- TEST_MUSTPASS(rxAGCConfig.targetLeveldBOv !=
- rxAGCConfigSet.targetLeveldBOv);
- TEST_MUSTPASS(rxAGCConfig.digitalCompressionGaindB !=
- rxAGCConfigSet.digitalCompressionGaindB);
- TEST_MUSTPASS(rxAGCConfig.limiterEnable != rxAGCConfigSet.limiterEnable);
-
- // restore default AGC config
- TEST_MUSTPASS(apm->SetRxAgcConfig(0, rxAGCConfigDefault));
- SLEEP(AGCSleep);
-
- //////
- // NS
-
- NsModes rxNSmode(kNsHighSuppression);
- NsModes rxNSmodeDefault(kNsModerateSuppression);
-
- // verify default settings (should be OFF and mode as above)
- TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
- TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(rxNSmode != rxNSmodeDefault);
-
- // enable default NS settings
- // must set a value first time!
- TEST_MUSTPASS(apm->SetRxNsStatus(0, false, kNsDefault));
- TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
- TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(rxNSmode != rxNSmodeDefault);
- SLEEP(NSSleep);
-
- // set kNsLowSuppression mode
- TEST_MUSTPASS(apm->SetRxNsStatus(0, true, kNsLowSuppression));
- TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
- TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
- TEST_MUSTPASS(rxNSmode != kNsLowSuppression);
- SLEEP(NSSleep);
-
- // set kNsModerateSuppression mode
- TEST_MUSTPASS(apm->SetRxNsStatus(0, true, kNsModerateSuppression));
- TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
- TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
- TEST_MUSTPASS(rxNSmode != kNsModerateSuppression);
- SLEEP(NSSleep);
-
- // set kNsHighSuppression mode
- TEST_MUSTPASS(apm->SetRxNsStatus(0, true, kNsHighSuppression));
- TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
- TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
- TEST_MUSTPASS(rxNSmode != kNsHighSuppression);
- SLEEP(NSSleep);
-
- // set kNsVeryHighSuppression mode
- TEST_MUSTPASS(apm->SetRxNsStatus(0, true, kNsVeryHighSuppression));
- TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
- TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
- TEST_MUSTPASS(rxNSmode != kNsVeryHighSuppression);
- SLEEP(NSSleep);
-
- // set kNsVeryHighSuppression mode
- TEST_MUSTPASS(apm->SetRxNsStatus(0, true, kNsConference));
- TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
- TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
- TEST_MUSTPASS(rxNSmode != kNsHighSuppression);
- SLEEP(NSSleep);
-
- // verify that mode is maintained when NS is disabled
- TEST_MUSTPASS(apm->SetRxNsStatus(0, false));
- TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
- TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(rxNSmode != kNsHighSuppression);
- SLEEP(NSSleep);
-
- // restore default NS
- TEST_MUSTPASS(apm->SetRxNsStatus(0, true, kNsDefault));
- TEST_MUSTPASS(apm->SetRxNsStatus(0, false));
- TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
- TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(rxNSmode != NSmodeDefault);
- SLEEP(NSSleep);
-
- // enable/disable many times in a row
- for (i = 0; i < NSIterations; i++)
- {
- TEST_MUSTPASS(apm->SetRxNsStatus(0, true));
- TEST_MUSTPASS(apm->SetRxNsStatus(0, false));
- }
- TEST_MUSTPASS(apm->GetRxNsStatus(0, enabled, rxNSmode));
- TEST_LOG("rxNS: enabled=%d, NSmode=%d\n", enabled, rxNSmode);
- TEST_MUSTPASS(enabled != false);
- TEST_MUSTPASS(rxNSmode != NSmodeDefault);
- SLEEP(NSSleep);
-
- /////////////////////////////
- // StartDebugRecording
- ////////////////////////////
- // StopDebugRecording
- TEST_LOG("StartDebugRecording");
- TEST_MUSTPASS(apm->StartDebugRecording(GetFilename("apm_debug.txt")));
- SLEEP(1000);
- TEST_LOG("StopDebugRecording");
- TEST_MUSTPASS(apm->StopDebugRecording());
-
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->Terminate());
-
- return 0;
-}
-
-} // namespace voetest
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_extended_test.h b/trunk/src/voice_engine/main/test/auto_test/voe_extended_test.h
deleted file mode 100644
index 9a10037..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/voe_extended_test.h
+++ /dev/null
@@ -1,463 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_EXTENDED_TEST_H
-#define WEBRTC_VOICE_ENGINE_VOE_EXTENDED_TEST_H
-
-#include "voe_standard_test.h"
-#include "audio_device.h"
-
-namespace voetest {
-
-class VoETestManager;
-
-// ----------------------------------------------------------------------------
-// AudioDeviceModule
-//
-// Implementation of the ADM to be used as external ADM in VoiceEngine.
-// This implementation is only a mock class, i.e., it does not provide
-// any real audio support.
-// ----------------------------------------------------------------------------
-
-class AudioDeviceModuleImpl : public AudioDeviceModule {
- public:
- // Factory methods
- static AudioDeviceModuleImpl* Create();
- static bool Destroy(AudioDeviceModuleImpl* adm);
-
- // Helper methods which allows us to get some handy information about
- // this mock implementation.
- int32_t ReferenceCounter() const {
- return _ref_count;
- }
-
- // RefCountedModule implementation (mocks default implementation)
- virtual int32_t AddRef();
- virtual int32_t Release();
-
- // Module implementation
- virtual int32_t Version(char* version,
- uint32_t& remaining_buffer_in_bytes,
- uint32_t& position) const {
- return 0;
- }
- virtual int32_t ChangeUniqueId(const int32_t id) {
- return 0;
- }
- virtual int32_t TimeUntilNextProcess() {
- return -1;
- }
- virtual int32_t Process() {
- return 0;
- }
-
- // AudioDeviceModule implementation
- virtual int32_t ActiveAudioLayer(AudioLayer* audioLayer) const {
- return 0;
- }
-
- virtual ErrorCode LastError() const {
- return static_cast<ErrorCode> (0);
- }
- virtual int32_t RegisterEventObserver(AudioDeviceObserver* eventCallback) {
- return 0;
- }
-
- virtual int32_t RegisterAudioCallback(AudioTransport* audioCallback) {
- return 0;
- }
-
- virtual int32_t Init() {
- return 0;
- }
- virtual int32_t Terminate() {
- return 0;
- }
- virtual bool Initialized() const {
- return true;
- }
-
- virtual int16_t PlayoutDevices() {
- return -1;
- }
- virtual int16_t RecordingDevices() {
- return -1;
- }
- virtual int32_t PlayoutDeviceName(uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) {
- return -1;
- }
- virtual int32_t RecordingDeviceName(uint16_t index,
- char name[kAdmMaxDeviceNameSize],
- char guid[kAdmMaxGuidSize]) {
- return -1;
- }
-
- virtual int32_t SetPlayoutDevice(uint16_t index) {
- return 0;
- }
- virtual int32_t SetPlayoutDevice(WindowsDeviceType device) {
- return 0;
- }
- virtual int32_t SetRecordingDevice(uint16_t index) {
- return 0;
- }
- virtual int32_t SetRecordingDevice(WindowsDeviceType device) {
- return 0;
- }
-
- virtual int32_t PlayoutIsAvailable(bool* available) {
- *available = true;
- return 0;
- }
- virtual int32_t InitPlayout() {
- return 0;
- }
- virtual bool PlayoutIsInitialized() const {
- return true;
- }
- virtual int32_t RecordingIsAvailable(bool* available) {
- *available = true;
- return 0;
- }
- virtual int32_t InitRecording() {
- return 0;
- }
- virtual bool RecordingIsInitialized() const {
- return true;
- }
-
- virtual int32_t StartPlayout() {
- return 0;
- }
- virtual int32_t StopPlayout() {
- return 0;
- }
- virtual bool Playing() const {
- return true;
- }
- virtual int32_t StartRecording() {
- return 0;
- }
- virtual int32_t StopRecording() {
- return 0;
- }
- virtual bool Recording() const {
- return true;
- }
-
- virtual int32_t SetAGC(bool enable) {
- return -1;
- }
- virtual bool AGC() const {
- return false;
- }
-
- virtual int32_t SetWaveOutVolume(uint16_t volumeLeft,
- uint16_t volumeRight) {
- return -1;
- }
- virtual int32_t WaveOutVolume(uint16_t* volumeLeft,
- uint16_t* volumeRight) const {
- return -1;
- }
-
- virtual int32_t SpeakerIsAvailable(bool* available) {
- *available = true;
- return 0;
- }
- virtual int32_t InitSpeaker() {
- return 0;
- }
- virtual bool SpeakerIsInitialized() const {
- return true;
- }
- virtual int32_t MicrophoneIsAvailable(bool* available) {
- *available = true;
- return 0;
- }
- virtual int32_t InitMicrophone() {
- return 0;
- }
- virtual bool MicrophoneIsInitialized() const {
- return true;
- }
-
- virtual int32_t SpeakerVolumeIsAvailable(bool* available) {
- return -1;
- }
- virtual int32_t SetSpeakerVolume(uint32_t volume) {
- return -1;
- }
- virtual int32_t SpeakerVolume(uint32_t* volume) const {
- return -1;
- }
- virtual int32_t MaxSpeakerVolume(uint32_t* maxVolume) const {
- return -1;
- }
- virtual int32_t MinSpeakerVolume(uint32_t* minVolume) const {
- return -1;
- }
- virtual int32_t SpeakerVolumeStepSize(uint16_t* stepSize) const {
- return -1;
- }
-
- virtual int32_t MicrophoneVolumeIsAvailable(bool* available) {
- return -1;
- }
- virtual int32_t SetMicrophoneVolume(uint32_t volume) {
- return -1;
- }
- virtual int32_t MicrophoneVolume(uint32_t* volume) const {
- return -1;
- }
- virtual int32_t MaxMicrophoneVolume(uint32_t* maxVolume) const {
- return -1;
- }
- virtual int32_t MinMicrophoneVolume(uint32_t* minVolume) const {
- return -1;
- }
- virtual int32_t MicrophoneVolumeStepSize(uint16_t* stepSize) const {
- return -1;
- }
-
- virtual int32_t SpeakerMuteIsAvailable(bool* available) {
- return -1;
- }
- virtual int32_t SetSpeakerMute(bool enable) {
- return -1;
- }
- virtual int32_t SpeakerMute(bool* enabled) const {
- return -1;
- }
-
- virtual int32_t MicrophoneMuteIsAvailable(bool* available) {
- return -1;
- }
- virtual int32_t SetMicrophoneMute(bool enable) {
- return -1;
- }
- virtual int32_t MicrophoneMute(bool* enabled) const {
- return -1;
- }
-
- virtual int32_t MicrophoneBoostIsAvailable(bool* available) {
- return -1;
- }
- virtual int32_t SetMicrophoneBoost(bool enable) {
- return -1;
- }
- virtual int32_t MicrophoneBoost(bool* enabled) const {
- return -1;
- }
-
- virtual int32_t StereoPlayoutIsAvailable(bool* available) const {
- return -1;
- }
- virtual int32_t SetStereoPlayout(bool enable) {
- return -1;
- }
- virtual int32_t StereoPlayout(bool* enabled) const {
- return -1;
- }
- virtual int32_t StereoRecordingIsAvailable(bool* available) const {
- return -1;
- }
- virtual int32_t SetStereoRecording(bool enable) {
- return -1;
- }
- virtual int32_t StereoRecording(bool* enabled) const {
- return -1;
- }
- virtual int32_t SetRecordingChannel(const ChannelType channel) {
- return -1;
- }
- virtual int32_t RecordingChannel(ChannelType* channel) const {
- return -1;
- }
-
- virtual int32_t SetPlayoutBuffer(const BufferType type, uint16_t sizeMS = 0) {
- return -1;
- }
- virtual int32_t PlayoutBuffer(BufferType* type, uint16_t* sizeMS) const {
- return -1;
- }
- virtual int32_t PlayoutDelay(uint16_t* delayMS) const {
- return -1;
- }
- virtual int32_t RecordingDelay(uint16_t* delayMS) const {
- return -1;
- }
-
- virtual int32_t CPULoad(uint16_t* load) const {
- return -1;
- }
-
- virtual int32_t StartRawOutputFileRecording(
- const char pcmFileNameUTF8[kAdmMaxFileNameSize]) {
- return -1;
- }
- virtual int32_t StopRawOutputFileRecording() {
- return -1;
- }
- virtual int32_t StartRawInputFileRecording(
- const char pcmFileNameUTF8[kAdmMaxFileNameSize]) {
- return -1;
- }
- virtual int32_t StopRawInputFileRecording() {
- return -1;
- }
-
- virtual int32_t SetRecordingSampleRate(const uint32_t samplesPerSec) {
- return -1;
- }
- virtual int32_t RecordingSampleRate(uint32_t* samplesPerSec) const {
- return -1;
- }
- virtual int32_t SetPlayoutSampleRate(const uint32_t samplesPerSec) {
- return -1;
- }
- virtual int32_t PlayoutSampleRate(uint32_t* samplesPerSec) const {
- return -1;
- }
-
- virtual int32_t ResetAudioDevice() {
- return -1;
- }
- virtual int32_t SetLoudspeakerStatus(bool enable) {
- return -1;
- }
- virtual int32_t GetLoudspeakerStatus(bool* enabled) const {
- return -1;
- }
-
- protected:
- AudioDeviceModuleImpl();
- ~AudioDeviceModuleImpl();
-
- private:
- volatile int32_t _ref_count;
-};
-
-// ----------------------------------------------------------------------------
-// Transport
-// ----------------------------------------------------------------------------
-
-class ExtendedTestTransport : public Transport {
- public:
- ExtendedTestTransport(VoENetwork* ptr);
- ~ExtendedTestTransport();
- VoENetwork* myNetw;
-
- protected:
- virtual int SendPacket(int channel, const void *data, int len);
- virtual int SendRTCPPacket(int channel, const void *data, int len);
-
- private:
- static bool Run(void* ptr);
- bool Process();
-
- private:
- ThreadWrapper* _thread;
- CriticalSectionWrapper* _lock;
- EventWrapper* _event;
-
- private:
- unsigned char _packetBuffer[1612];
- int _length;
- int _channel;
-};
-
-class XTransport : public Transport {
- public:
- XTransport(VoENetwork* netw, VoEFile* file);
- VoENetwork* _netw;
- VoEFile* _file;
-
- public:
- virtual int SendPacket(int channel, const void *data, int len);
- virtual int SendRTCPPacket(int channel, const void *data, int len);
-};
-
-class XRTPObserver : public VoERTPObserver {
- public:
- XRTPObserver();
- ~XRTPObserver();
- virtual void OnIncomingCSRCChanged(const int channel,
- const unsigned int CSRC,
- const bool added);
- virtual void OnIncomingSSRCChanged(const int channel,
- const unsigned int SSRC);
- public:
- unsigned int _SSRC;
-};
-
-// ----------------------------------------------------------------------------
-// VoEExtendedTest
-// ----------------------------------------------------------------------------
-
-class VoEExtendedTest : public VoiceEngineObserver,
- public VoEConnectionObserver {
- public:
- VoEExtendedTest(VoETestManager& mgr);
- ~VoEExtendedTest();
- int PrepareTest(const char* str) const;
- int TestPassed(const char* str) const;
- int TestBase();
- int TestCallReport();
- int TestCodec();
- int TestDtmf();
- int TestEncryption();
- int TestExternalMedia();
- int TestFile();
- int TestMixing();
- int TestHardware();
- int TestNetEqStats();
- int TestNetwork();
- int TestRTP_RTCP();
- int TestVideoSync();
- int TestVolumeControl();
- int TestAPM();
- public:
- int ErrorCode() const {
- return _errCode;
- }
- void ClearErrorCode() {
- _errCode = 0;
- }
- protected:
- // from VoiceEngineObserver
- void CallbackOnError(const int errCode, const int channel);
- void CallbackOnTrace(const TraceLevel level, const char* message, const int length);
- protected:
- // from VoEConnectionObserver
- void OnPeriodicDeadOrAlive(const int channel, const bool alive);
- private:
- void Play(int channel, unsigned int timeMillisec, bool addFileAsMicrophone = false,
- bool addTimeMarker = false);
- void Sleep(unsigned int timeMillisec, bool addMarker = false);
- void StartMedia(int channel, int rtpPort, bool listen, bool playout, bool send);
- void StopMedia(int channel);
- int RunMixingTest(int num_remote_channels, int num_local_channels,
- int16_t input_value, int16_t max_output_value,
- int16_t min_output_value);
- private:
- VoETestManager& _mgr;
- private:
- int _errCode;
- bool _alive;
- bool _listening[32];
- bool _playing[32];
- bool _sending[32];
-};
-
-} // namespace voetest
-#endif // WEBRTC_VOICE_ENGINE_VOE_EXTENDED_TEST_H
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_standard_test.cc b/trunk/src/voice_engine/main/test/auto_test/voe_standard_test.cc
deleted file mode 100644
index a6af7fa..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/voe_standard_test.cc
+++ /dev/null
@@ -1,1851 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <string.h>
-#include <math.h>
-#include "engine_configurations.h"
-#if defined(_WIN32)
-#include <conio.h> // exists only on windows
-#include <tchar.h>
-#endif
-
-#include "voe_standard_test.h"
-
-#if defined (_ENABLE_VISUAL_LEAK_DETECTOR_) && defined(_DEBUG) && \
- defined(_WIN32) && !defined(_INSTRUMENTATION_TESTING_)
-#include "vld.h"
-#endif
-
-#ifdef MAC_IPHONE
-#include "../../source/voice_engine_defines.h" // defines build macros
-#else
-#include "../../source/voice_engine_defines.h" // defines build macros
-#endif
-
-#include "automated_mode.h"
-#include "critical_section_wrapper.h"
-#include "event_wrapper.h"
-#include "thread_wrapper.h"
-
-#ifdef _TEST_NETEQ_STATS_
-#include "../../interface/voe_neteq_stats.h" // Not available in delivery folder
-#endif
-
-#include "voe_extended_test.h"
-#include "voe_stress_test.h"
-#include "voe_unit_test.h"
-#include "voe_cpu_test.h"
-
-using namespace webrtc;
-
-namespace voetest {
-
-#ifdef MAC_IPHONE
-// Defined in iPhone specific test file
-int GetDocumentsDir(char* buf, int bufLen);
-char* GetFilename(char* filename);
-const char* GetFilename(const char* filename);
-int GetResource(char* resource, char* dest, int destLen);
-char* GetResource(char* resource);
-const char* GetResource(const char* resource);
-// #ifdef MAC_IPHONE
-#elif defined(WEBRTC_ANDROID)
-char filenameStr[2][256];
-int currentStr = 0;
-
-char* GetFilename(char* filename) {
- currentStr = !currentStr;
- sprintf(filenameStr[currentStr], "/sdcard/%s", filename);
- return filenameStr[currentStr];
-}
-
-const char* GetFilename(const char* filename) {
- currentStr = !currentStr;
- sprintf(filenameStr[currentStr], "/sdcard/%s", filename);
- return filenameStr[currentStr];
-}
-
-int GetResource(char* resource, char* dest, int destLen) {
- currentStr = !currentStr;
- sprintf(filenameStr[currentStr], "/sdcard/%s", resource);
- strncpy(dest, filenameStr[currentStr], destLen-1);
- return 0;
-}
-
-char* GetResource(char* resource) {
- currentStr = !currentStr;
- sprintf(filenameStr[currentStr], "/sdcard/%s", resource);
- return filenameStr[currentStr];
-}
-
-const char* GetResource(const char* resource) {
- currentStr = !currentStr;
- sprintf(filenameStr[currentStr], "/sdcard/%s", resource);
- return filenameStr[currentStr];
-}
-
-#else
-char filenameStr[2][256];
-int currentStr = 0;
-
-char* GetFilename(char* filename) {
- currentStr = !currentStr;
- sprintf(filenameStr[currentStr], "/tmp/%s", filename);
- return filenameStr[currentStr];
-}
-const char* GetFilename(const char* filename) {
- currentStr = !currentStr;
- sprintf(filenameStr[currentStr], "/tmp/%s", filename);
- return filenameStr[currentStr];
-}
-int GetResource(char* resource, char* dest, int destLen) {
- currentStr = !currentStr;
- sprintf(filenameStr[currentStr], "/tmp/%s", resource);
- strncpy(dest, filenameStr[currentStr], destLen - 1);
- return 0;
-}
-char* GetResource(char* resource) {
- currentStr = !currentStr;
- sprintf(filenameStr[currentStr], "/tmp/%s", resource);
- return filenameStr[currentStr];
-}
-const char* GetResource(const char* resource) {
- currentStr = !currentStr;
- sprintf(filenameStr[currentStr], "/tmp/%s", resource);
- return filenameStr[currentStr];
-}
-#endif
-
-#if !defined(MAC_IPHONE)
-const char* summaryFilename = "/tmp/VoiceEngineSummary.txt";
-#endif
-// For iPhone the summary filename is created in createSummary
-
-int dummy = 0; // Dummy used in different functions to avoid warnings
-
-TestRtpObserver::TestRtpObserver() {
- Reset();
-}
-
-TestRtpObserver::~TestRtpObserver() {
-}
-
-void TestRtpObserver::Reset() {
- for (int i = 0; i < 2; i++) {
- ssrc_[i] = 0;
- csrc_[i][0] = 0;
- csrc_[i][1] = 0;
- added_[i][0] = false;
- added_[i][1] = false;
- size_[i] = 0;
- }
-}
-
-void TestRtpObserver::OnIncomingCSRCChanged(const int channel,
- const unsigned int CSRC,
- const bool added) {
- char msg[128];
- sprintf(msg, "=> OnIncomingCSRCChanged(channel=%d, CSRC=%u, added=%d)\n",
- channel, CSRC, added);
- TEST_LOG("%s", msg);
-
- if (channel > 1)
- return; // Not enough memory.
-
- csrc_[channel][size_[channel]] = CSRC;
- added_[channel][size_[channel]] = added;
-
- size_[channel]++;
- if (size_[channel] == 2)
- size_[channel] = 0;
-}
-
-void TestRtpObserver::OnIncomingSSRCChanged(const int channel,
- const unsigned int SSRC) {
- char msg[128];
- sprintf(msg, "\n=> OnIncomingSSRCChanged(channel=%d, SSRC=%u)\n", channel,
- SSRC);
- TEST_LOG("%s", msg);
-
- ssrc_[channel] = SSRC;
-}
-
-void MyDeadOrAlive::OnPeriodicDeadOrAlive(const int /*channel*/,
- const bool alive) {
- if (alive) {
- TEST_LOG("ALIVE\n");
- } else {
- TEST_LOG("DEAD\n");
- }
- fflush(NULL);
-}
-
-#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
-void MyMedia::Process(const int channel,
- const ProcessingTypes type,
- WebRtc_Word16 audio_10ms[],
- const int length,
- const int samplingFreqHz,
- const bool stereo) {
- for (int i = 0; i < length; i++) {
- if (!stereo) {
- audio_10ms[i] = (WebRtc_Word16) (audio_10ms[i] *
- sin(2.0 * 3.14 * f * 400.0 / samplingFreqHz));
- } else {
- // interleaved stereo
- audio_10ms[2 * i] = (WebRtc_Word16) (audio_10ms[2 * i] *
- sin(2.0 * 3.14 * f * 400.0 / samplingFreqHz));
- audio_10ms[2 * i + 1] = (WebRtc_Word16) (audio_10ms[2 * i + 1] *
- sin(2.0 * 3.14 * f * 400.0 / samplingFreqHz));
- }
- f++;
- }
-}
-#endif
-
-MyMedia mobj;
-
-FakeExternalTransport::FakeExternalTransport(VoENetwork* ptr)
- : my_network_(ptr),
- thread_(NULL),
- lock_(NULL),
- event_(NULL),
- length_(0),
- channel_(0),
- delay_is_enabled_(0),
- delay_time_in_ms_(0) {
- const char* threadName = "external_thread";
- lock_ = CriticalSectionWrapper::CreateCriticalSection();
- event_ = EventWrapper::Create();
- thread_ = ThreadWrapper::CreateThread(Run, this, kHighPriority, threadName);
- if (thread_) {
- unsigned int id;
- thread_->Start(id);
- }
-}
-
-FakeExternalTransport::~FakeExternalTransport() {
- if (thread_) {
- thread_->SetNotAlive();
- event_->Set();
- if (thread_->Stop()) {
- delete thread_;
- thread_ = NULL;
- delete event_;
- event_ = NULL;
- delete lock_;
- lock_ = NULL;
- }
- }
-}
-
-bool FakeExternalTransport::Run(void* ptr) {
- return static_cast<FakeExternalTransport*> (ptr)->Process();
-}
-
-bool FakeExternalTransport::Process() {
- switch (event_->Wait(500)) {
- case kEventSignaled:
- lock_->Enter();
- my_network_->ReceivedRTPPacket(channel_, packet_buffer_, length_);
- lock_->Leave();
- return true;
- case kEventTimeout:
- return true;
- case kEventError:
- break;
- }
- return true;
-}
-
-int FakeExternalTransport::SendPacket(int channel, const void *data, int len) {
- lock_->Enter();
- if (len < 1612) {
- memcpy(packet_buffer_, (const unsigned char*) data, len);
- length_ = len;
- channel_ = channel;
- }
- lock_->Leave();
- event_->Set(); // triggers ReceivedRTPPacket() from worker thread
- return len;
-}
-
-int FakeExternalTransport::SendRTCPPacket(int channel, const void *data, int len) {
- if (delay_is_enabled_) {
- Sleep(delay_time_in_ms_);
- }
- my_network_->ReceivedRTCPPacket(channel, data, len);
- return len;
-}
-
-void FakeExternalTransport::SetDelayStatus(bool enable, unsigned int delayInMs) {
- delay_is_enabled_ = enable;
- delay_time_in_ms_ = delayInMs;
-}
-
-ErrorObserver::ErrorObserver() {
- code = -1;
-}
-void ErrorObserver::CallbackOnError(const int channel, const int errCode) {
- code = errCode;
-#ifndef _INSTRUMENTATION_TESTING_
- TEST_LOG("\n************************\n");
- TEST_LOG(" RUNTIME ERROR: %d \n", errCode);
- TEST_LOG("************************\n");
-#endif
-}
-
-void MyTraceCallback::Print(const TraceLevel level,
- const char *traceString,
- const int length) {
- if (traceString) {
- char* tmp = new char[length];
- memcpy(tmp, traceString, length);
- TEST_LOG("%s", tmp);
- TEST_LOG("\n");
- delete[] tmp;
- }
-}
-
-void RtcpAppHandler::OnApplicationDataReceived(
- const int /*channel*/, const unsigned char sub_type,
- const unsigned int name, const unsigned char* data,
- const unsigned short length_in_bytes) {
- length_in_bytes_ = length_in_bytes;
- memcpy(data_, &data[0], length_in_bytes);
- sub_type_ = sub_type;
- name_ = name;
-}
-
-void RtcpAppHandler::Reset() {
- length_in_bytes_ = 0;
- memset(data_, 0, sizeof(data_));
- sub_type_ = 0;
- name_ = 0;
-}
-
-void my_encryption::encrypt(int, unsigned char * in_data,
- unsigned char * out_data,
- int bytes_in,
- int * bytes_out) {
- int i;
- for (i = 0; i < bytes_in; i++)
- out_data[i] = ~in_data[i];
- *bytes_out = bytes_in + 2; // length is increased by 2
-}
-
-void my_encryption::decrypt(int, unsigned char * in_data,
- unsigned char * out_data,
- int bytes_in,
- int * bytes_out) {
- int i;
- for (i = 0; i < bytes_in; i++)
- out_data[i] = ~in_data[i];
- *bytes_out = bytes_in - 2; // length is decreased by 2
-}
-
-void my_encryption::encrypt_rtcp(int,
- unsigned char * in_data,
- unsigned char * out_data,
- int bytes_in,
- int * bytes_out) {
- int i;
- for (i = 0; i < bytes_in; i++)
- out_data[i] = ~in_data[i];
- *bytes_out = bytes_in + 2;
-}
-
-void my_encryption::decrypt_rtcp(int, unsigned char * in_data,
- unsigned char * out_data,
- int bytes_in,
- int * bytes_out) {
- int i;
- for (i = 0; i < bytes_in; i++)
- out_data[i] = ~in_data[i];
- *bytes_out = bytes_in + 2;
-}
-
-void SubAPIManager::DisplayStatus() const {
- TEST_LOG("Supported sub APIs:\n\n");
- if (_base)
- TEST_LOG(" Base\n");
- if (_callReport)
- TEST_LOG(" CallReport\n");
- if (_codec)
- TEST_LOG(" Codec\n");
- if (_dtmf)
- TEST_LOG(" Dtmf\n");
- if (_encryption)
- TEST_LOG(" Encryption\n");
- if (_externalMedia)
- TEST_LOG(" ExternalMedia\n");
- if (_file)
- TEST_LOG(" File\n");
- if (_hardware)
- TEST_LOG(" Hardware\n");
- if (_netEqStats)
- TEST_LOG(" NetEqStats\n");
- if (_network)
- TEST_LOG(" Network\n");
- if (_rtp_rtcp)
- TEST_LOG(" RTP_RTCP\n");
- if (_videoSync)
- TEST_LOG(" VideoSync\n");
- if (_volumeControl)
- TEST_LOG(" VolumeControl\n");
- if (_apm)
- TEST_LOG(" AudioProcessing\n");
- ANL();
- TEST_LOG("Excluded sub APIs:\n\n");
- if (!_base)
- TEST_LOG(" Base\n");
- if (!_callReport)
- TEST_LOG(" CallReport\n");
- if (!_codec)
- TEST_LOG(" Codec\n");
- if (!_dtmf)
- TEST_LOG(" Dtmf\n");
- if (!_encryption)
- TEST_LOG(" Encryption\n");
- if (!_externalMedia)
- TEST_LOG(" ExternamMedia\n");
- if (!_file)
- TEST_LOG(" File\n");
- if (!_hardware)
- TEST_LOG(" Hardware\n");
- if (!_netEqStats)
- TEST_LOG(" NetEqStats\n");
- if (!_network)
- TEST_LOG(" Network\n");
- if (!_rtp_rtcp)
- TEST_LOG(" RTP_RTCP\n");
- if (!_videoSync)
- TEST_LOG(" VideoSync\n");
- if (!_volumeControl)
- TEST_LOG(" VolumeControl\n");
- if (!_apm)
- TEST_LOG(" AudioProcessing\n");
- ANL();
-}
-
-bool SubAPIManager::GetExtendedMenuSelection(ExtendedSelection& sel) {
- printf("------------------------------------------------\n");
- printf("Select extended test\n\n");
- printf(" (0) None\n");
- printf("- - - - - - - - - - - - - - - - - - - - - - - - \n");
- printf(" (1) Base");
- if (_base)
- printf("\n");
- else
- printf(" (NA)\n");
- printf(" (2) CallReport");
- if (_callReport)
- printf("\n");
- else
- printf(" (NA)\n");
- printf(" (3) Codec");
- if (_codec)
- printf("\n");
- else
- printf(" (NA)\n");
- printf(" (4) Dtmf");
- if (_dtmf)
- printf("\n");
- else
- printf(" (NA)\n");
- printf(" (5) Encryption");
- if (_encryption)
- printf("\n");
- else
- printf(" (NA)\n");
- printf(" (6) VoEExternalMedia");
- if (_externalMedia)
- printf("\n");
- else
- printf(" (NA)\n");
- printf(" (7) File");
- if (_file)
- printf("\n");
- else
- printf(" (NA)\n");
- printf(" (8) Mixing");
- if (_file)
- printf("\n");
- else
- printf(" (NA)\n");
- printf(" (9) Hardware");
- if (_hardware)
- printf("\n");
- else
- printf(" (NA)\n");
- printf(" (10) NetEqStats");
- if (_netEqStats)
- printf("\n");
- else
- printf(" (NA)\n");
- printf(" (11) Network");
- if (_network)
- printf("\n");
- else
- printf(" (NA)\n");
- printf(" (12) RTP_RTCP");
- if (_rtp_rtcp)
- printf("\n");
- else
- printf(" (NA)\n");
- printf(" (13) VideoSync");
- if (_videoSync)
- printf("\n");
- else
- printf(" (NA)\n");
- printf(" (14) VolumeControl");
- if (_volumeControl)
- printf("\n");
- else
- printf(" (NA)\n");
- printf(" (15) AudioProcessing");
- if (_apm)
- printf("\n");
- else
- printf(" (NA)\n");
- printf("\n: ");
-
- ExtendedSelection xsel(XSEL_Invalid);
- int selection(0);
- dummy = scanf("%d", &selection);
-
- switch (selection) {
- case 0:
- xsel = XSEL_None;
- break;
- case 1:
- if (_base)
- xsel = XSEL_Base;
- break;
- case 2:
- if (_callReport)
- xsel = XSEL_CallReport;
- break;
- case 3:
- if (_codec)
- xsel = XSEL_Codec;
- break;
- case 4:
- if (_dtmf)
- xsel = XSEL_DTMF;
- break;
- case 5:
- if (_encryption)
- xsel = XSEL_Encryption;
- break;
- case 6:
- if (_externalMedia)
- xsel = XSEL_ExternalMedia;
- break;
- case 7:
- if (_file)
- xsel = XSEL_File;
- break;
- case 8:
- if (_file)
- xsel = XSEL_Mixing;
- break;
- case 9:
- if (_hardware)
- xsel = XSEL_Hardware;
- break;
- case 10:
- if (_netEqStats)
- xsel = XSEL_NetEqStats;
- break;
- case 11:
- if (_network)
- xsel = XSEL_Network;
- break;
- case 12:
- if (_rtp_rtcp)
- xsel = XSEL_RTP_RTCP;
- break;
- case 13:
- if (_videoSync)
- xsel = XSEL_VideoSync;
- break;
- case 14:
- if (_volumeControl)
- xsel = XSEL_VolumeControl;
- break;
- case 15:
- if (_apm)
- xsel = XSEL_AudioProcessing;
- break;
- default:
- xsel = XSEL_Invalid;
- break;
- }
- if (xsel == XSEL_Invalid)
- printf("Invalid selection!\n");
-
- sel = xsel;
- _xsel = xsel;
-
- return (xsel != XSEL_Invalid);
-}
-
-VoETestManager::VoETestManager()
- : initialized_(false),
- voice_engine_(NULL),
- voe_base_(0),
- voe_call_report_(0),
- voe_codec_(0),
- voe_dtmf_(0),
- voe_encrypt_(0),
- voe_xmedia_(0),
- voe_file_(0),
- voe_hardware_(0),
- voe_network_(0),
-#ifdef _TEST_NETEQ_STATS_
- voe_neteq_stats_(NULL),
-#endif
- voe_rtp_rtcp_(0),
- voe_vsync_(0),
- voe_volume_control_(0),
- voe_apm_(0)
-{
-}
-
-VoETestManager::~VoETestManager() {
-}
-
-bool VoETestManager::Init() {
- if (initialized_)
- return true;
-
- if (VoiceEngine::SetTraceFile(NULL) != -1) {
- // should not be possible to call a Trace method before the VoE is
- // created
- TEST_LOG("\nError at line: %i (VoiceEngine::SetTraceFile()"
- "should fail)!\n", __LINE__);
- return false;
- }
-
- voice_engine_ = VoiceEngine::Create();
- if (!voice_engine_) {
- TEST_LOG("Failed to create VoiceEngine\n");
- return false;
- }
-
- return true;
-}
-
-void VoETestManager::GetInterfaces() {
- if (voice_engine_) {
- voe_base_ = VoEBase::GetInterface(voice_engine_);
- voe_codec_ = VoECodec::GetInterface(voice_engine_);
- voe_volume_control_ = VoEVolumeControl::GetInterface(voice_engine_);
- voe_dtmf_ = VoEDtmf::GetInterface(voice_engine_);
- voe_rtp_rtcp_ = VoERTP_RTCP::GetInterface(voice_engine_);
- voe_apm_ = VoEAudioProcessing::GetInterface(voice_engine_);
- voe_network_ = VoENetwork::GetInterface(voice_engine_);
- voe_file_ = VoEFile::GetInterface(voice_engine_);
-#ifdef _TEST_VIDEO_SYNC_
- voe_vsync_ = VoEVideoSync::GetInterface(voice_engine_);
-#endif
- voe_encrypt_ = VoEEncryption::GetInterface(voice_engine_);
- voe_hardware_ = VoEHardware::GetInterface(voice_engine_);
- // Set the audio layer to use in all tests
- if (voe_hardware_) {
- int res = voe_hardware_->SetAudioDeviceLayer(TESTED_AUDIO_LAYER);
- if (res < 0) {
- printf("\nERROR: failed to set audio layer to use in "
- "testing\n");
- } else {
- printf("\nAudio layer %d will be used in testing\n",
- TESTED_AUDIO_LAYER);
- }
- }
-#ifdef _TEST_XMEDIA_
- voe_xmedia_ = VoEExternalMedia::GetInterface(voice_engine_);
-#endif
-#ifdef _TEST_CALL_REPORT_
- voe_call_report_ = VoECallReport::GetInterface(voice_engine_);
-#endif
-#ifdef _TEST_NETEQ_STATS_
- voe_neteq_stats_ = VoENetEqStats::GetInterface(voice_engine_);
-#endif
- }
-}
-
-int VoETestManager::ReleaseInterfaces() {
- int err(0), remInt(1), j(0);
- bool releaseOK(true);
-
- if (voe_base_) {
- for (remInt = 1, j = 0; remInt > 0; j++)
- TEST_MUSTPASS(-1 == (remInt = voe_base_->Release()));
- if (j > 1) {
- TEST_LOG("\n\n*** Error: released %d base interfaces"
- "(should only be 1) \n", j);
- releaseOK = false;
- }
- // try to release one addition time (should fail)
- TEST_MUSTPASS(-1 != voe_base_->Release());
- err = voe_base_->LastError();
- // it is considered safe to delete even if Release has been called
- // too many times
- TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
- }
- if (voe_codec_) {
- for (remInt = 1, j = 0; remInt > 0; j++)
- TEST_MUSTPASS(-1 == (remInt = voe_codec_->Release()));
- if (j > 1) {
- TEST_LOG("\n\n*** Error: released %d codec interfaces"
- " (should only be 1) \n", j);
- releaseOK = false;
- }
- TEST_MUSTPASS(-1 != voe_codec_->Release());
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
- }
- if (voe_volume_control_) {
- for (remInt = 1, j = 0; remInt > 0; j++)
- TEST_MUSTPASS(-1 == (remInt = voe_volume_control_->Release()));
- if (j > 1) {
- TEST_LOG("\n\n*** Error: released %d volume interfaces"
- "(should only be 1) \n", j);
- releaseOK = false;
- }
- TEST_MUSTPASS(-1 != voe_volume_control_->Release());
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
- }
- if (voe_dtmf_) {
- for (remInt = 1, j = 0; remInt > 0; j++)
- TEST_MUSTPASS(-1 == (remInt = voe_dtmf_->Release()));
- if (j > 1) {
- TEST_LOG("\n\n*** Error: released %d dtmf interfaces"
- "(should only be 1) \n", j);
- releaseOK = false;
- }
- TEST_MUSTPASS(-1 != voe_dtmf_->Release());
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
- }
- if (voe_rtp_rtcp_) {
- for (remInt = 1, j = 0; remInt > 0; j++)
- TEST_MUSTPASS(-1 == (remInt = voe_rtp_rtcp_->Release()));
- if (j > 1) {
- TEST_LOG("\n\n*** Error: released %d rtp/rtcp interfaces"
- "(should only be 1) \n", j);
- releaseOK = false;
- }
- TEST_MUSTPASS(-1 != voe_rtp_rtcp_->Release());
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
- }
- if (voe_apm_) {
- for (remInt = 1, j = 0; remInt > 0; j++)
- TEST_MUSTPASS(-1 == (remInt = voe_apm_->Release()));
- if (j > 1) {
- TEST_LOG("\n\n*** Error: released %d apm interfaces"
- "(should only be 1) \n", j);
- releaseOK = false;
- }
- TEST_MUSTPASS(-1 != voe_apm_->Release());
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
- }
- if (voe_network_) {
- for (remInt = 1, j = 0; remInt > 0; j++)
- TEST_MUSTPASS(-1 == (remInt = voe_network_->Release()));
- if (j > 1) {
- TEST_LOG("\n\n*** Error: released %d network interfaces"
- "(should only be 1) \n", j);
- releaseOK = false;
- }
- TEST_MUSTPASS(-1 != voe_network_->Release());
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
- }
- if (voe_file_) {
- for (remInt = 1, j = 0; remInt > 0; j++)
- TEST_MUSTPASS(-1 == (remInt = voe_file_->Release()));
- if (j > 1) {
- TEST_LOG("\n\n*** Error: released %d file interfaces"
- "(should only be 1) \n", j);
- releaseOK = false;
- }
- TEST_MUSTPASS(-1 != voe_file_->Release());
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
- }
-#ifdef _TEST_VIDEO_SYNC_
- if (voe_vsync_) {
- for (remInt = 1, j = 0; remInt > 0; j++)
- TEST_MUSTPASS(-1 == (remInt = voe_vsync_->Release()));
- if (j > 1) {
- TEST_LOG("\n\n*** Error: released %d video sync interfaces"
- "(should only be 1) \n", j);
- releaseOK = false;
- }
- TEST_MUSTPASS(-1 != voe_vsync_->Release());
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
- }
-#endif
- if (voe_encrypt_) {
- for (remInt = 1, j = 0; remInt > 0; j++)
- TEST_MUSTPASS(-1 == (remInt = voe_encrypt_->Release()));
- if (j > 1) {
- TEST_LOG("\n\n*** Error: released %d encryption interfaces"
- "(should only be 1) \n", j);
- releaseOK = false;
- }
- TEST_MUSTPASS(-1 != voe_encrypt_->Release());
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
- }
- if (voe_hardware_) {
- for (remInt = 1, j = 0; remInt > 0; j++)
- TEST_MUSTPASS(-1 == (remInt = voe_hardware_->Release()));
- if (j > 1) {
- TEST_LOG("\n\n*** Error: released %d hardware interfaces"
- "(should only be 1) \n", j);
- releaseOK = false;
- }
- TEST_MUSTPASS(-1 != voe_hardware_->Release());
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
- }
-#ifdef _TEST_XMEDIA_
- if (voe_xmedia_) {
- for (remInt = 1, j = 0; remInt > 0; j++)
- TEST_MUSTPASS(-1 == (remInt = voe_xmedia_->Release()));
- if (j > 1) {
- TEST_LOG("\n\n*** Error: released %d external media interfaces"
- "(should only be 1) \n", j);
- releaseOK = false;
- }
- TEST_MUSTPASS(-1 != voe_xmedia_->Release());
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
- }
-#endif
-#ifdef _TEST_CALL_REPORT_
- if (voe_call_report_) {
- for (remInt = 1, j = 0; remInt > 0; j++)
- TEST_MUSTPASS(-1 == (remInt = voe_call_report_->Release()));
- if (j > 1) {
- TEST_LOG("\n\n*** Error: released %d call report interfaces"
- "(should only be 1) \n", j);
- releaseOK = false;
- }
- TEST_MUSTPASS(-1 != voe_call_report_->Release());
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
- }
-#endif
-#ifdef _TEST_NETEQ_STATS_
- if (voe_neteq_stats_) {
- for (remInt = 1, j = 0; remInt > 0; j++)
- TEST_MUSTPASS(-1 == (remInt = voe_neteq_stats_->Release()));
- if (j > 1) {
- TEST_LOG("\n\n*** Error: released %d neteq stat interfaces "
- "(should only be 1) \n", j);
- releaseOK = false;
- }
- TEST_MUSTPASS(-1 != voe_neteq_stats_->Release());
- err = voe_base_->LastError();
- TEST_MUSTPASS(err != VE_INTERFACE_NOT_FOUND);
- }
-#endif
- if (false == VoiceEngine::Delete(voice_engine_)) {
- TEST_LOG("\n\nVoiceEngine::Delete() failed. \n");
- releaseOK = false;
- }
-
- if (VoiceEngine::SetTraceFile(NULL) != -1) {
- TEST_LOG("\nError at line: %i (VoiceEngine::SetTraceFile()"
- "should fail)!\n", __LINE__);
- }
-
- return (releaseOK == true) ? 0 : -1;
-}
-
-int VoETestManager::SetUp(ErrorObserver* error_observer) {
- char char_buffer[1024];
-
- TEST_MUSTPASS(voe_base_->Init());
-
-#if defined(WEBRTC_ANDROID)
- TEST_MUSTPASS(voe_hardware_->SetLoudspeakerStatus(false));
-#endif
-
- TEST_MUSTPASS(voe_base_->RegisterVoiceEngineObserver(*error_observer));
-
- TEST_LOG("Get version \n");
- TEST_MUSTPASS(voe_base_->GetVersion(char_buffer));
- TEST_LOG("--------------------\n%s\n--------------------\n", char_buffer);
-
- TEST_LOG("Create channel \n");
- int nChannels = voe_base_->MaxNumOfChannels();
- TEST_MUSTPASS(!(nChannels > 0));
- TEST_LOG("Max number of channels = %d \n", nChannels);
- TEST_MUSTPASS(voe_base_->CreateChannel());
-
- return 0;
-}
-
-int VoETestManager::TestStartStreaming(FakeExternalTransport& channel0_transport) {
- TEST_LOG("\n\n+++ Starting streaming +++\n\n");
-
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
- TEST_LOG("Enabling external transport \n");
- TEST_MUSTPASS(voe_network_->RegisterExternalTransport(0, channel0_transport));
-#else
- TEST_LOG("Setting send and receive parameters \n");
- TEST_MUSTPASS(voe_base_->SetSendDestination(0, 8000, "127.0.0.1"));
- // No IP specified => "0.0.0.0" will be stored.
- TEST_MUSTPASS(voe_base_->SetLocalReceiver(0,8000));
-
- CodecInst Jing_inst;
- Jing_inst.channels = 1;
- Jing_inst.pacsize = 160;
- Jing_inst.plfreq = 8000;
- Jing_inst.pltype = 0;
- Jing_inst.rate = 64000;
- strcpy(Jing_inst.plname, "PCMU");
- TEST_MUSTPASS(voe_codec_->SetSendCodec(0, Jing_inst));
-
- int port = -1;
- int src_port = -1;
- int rtcp_port = -1;
- char ip_address[64] = { 0 };
- strcpy(ip_address, "10.10.10.10");
- TEST_MUSTPASS(voe_base_->GetSendDestination(0, port, ip_address, src_port,
- rtcp_port));
- TEST_MUSTPASS(8000 != port);
- TEST_MUSTPASS(8000 != src_port);
- TEST_MUSTPASS(8001 != rtcp_port);
- TEST_MUSTPASS(_stricmp(ip_address, "127.0.0.1"));
-
- port = -1;
- rtcp_port = -1;
- TEST_MUSTPASS(voe_base_->GetLocalReceiver(0, port, rtcp_port, ip_address));
- TEST_MUSTPASS(8000 != port);
- TEST_MUSTPASS(8001 != rtcp_port);
- TEST_MUSTPASS(_stricmp(ip_address, "0.0.0.0"));
-#endif
- return 0;
-}
-
-int VoETestManager::TestStartPlaying() {
- TEST_LOG("Start listening, playout and sending \n");
- TEST_MUSTPASS(voe_base_->StartReceive(0));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
-
- // Run in full duplex.
- TEST_LOG("You should now hear yourself, running default codec (PCMU)\n");
- SLEEP(2000);
-
- if (voe_file_) {
- TEST_LOG("Start playing a file as microphone, so you don't need to"
- " speak all the time\n");
- TEST_MUSTPASS(voe_file_->StartPlayingFileAsMicrophone(0,
- AudioFilename(),
- true,
- true));
- SLEEP(1000);
- }
- return 0;
-}
-
-int VoETestManager::DoStandardTest() {
- // Ensure we have all input files:
- TEST_MUSTPASS(!strcmp("", AudioFilename()));
-
- TEST_LOG("\n\n+++ Base tests +++\n\n");
-
- ErrorObserver error_observer;
- if (SetUp(&error_observer) != 0) return -1;
-
- voe_network_->SetSourceFilter(0, 0);
-
- FakeExternalTransport channel0_transport(voe_network_);
- if (TestStartStreaming(channel0_transport) != 0) return -1;
- if (TestStartPlaying() != 0) return -1;
-
- //////////////
- // Video Sync
-
-#ifdef _TEST_VIDEO_SYNC_
- TEST_LOG("\n\n+++ Video sync tests +++\n\n");
-
- unsigned int val;
- TEST_MUSTPASS(voe_vsync_->GetPlayoutTimestamp(0, val));
- TEST_LOG("Playout timestamp = %lu\n", (long unsigned int) val);
-
- TEST_LOG("Init timestamp and sequence number manually\n");
- TEST_MUSTPASS(!voe_vsync_->SetInitTimestamp(0, 12345));
- TEST_MUSTPASS(!voe_vsync_->SetInitSequenceNumber(0, 123));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_vsync_->SetInitTimestamp(0, 12345));
- TEST_MUSTPASS(voe_vsync_->SetInitSequenceNumber(0, 123));
- TEST_MUSTPASS(voe_base_->StartSend(0));
- if (voe_file_) {
- TEST_LOG("Start playing a file as microphone again \n");
- TEST_MUSTPASS(voe_file_->StartPlayingFileAsMicrophone(0,
- AudioFilename(),
- true,
- true));
- }
- SLEEP(3000);
-
- TEST_LOG("Check delay estimates during 15 seconds, verify that "
- "they stabilize during this time\n");
- int valInt = -1;
- for (int i = 0; i < 15; i++) {
- TEST_MUSTPASS(voe_vsync_->GetDelayEstimate(0, valInt));
- TEST_LOG("Delay estimate = %d ms\n", valInt);
-#if defined(MAC_IPHONE)
- TEST_MUSTPASS(valInt <= 30);
-#else
- TEST_MUSTPASS(valInt <= 45); // 45=20+25 => can't be this low
-#endif
- SLEEP(1000);
- }
-
- TEST_LOG("Setting NetEQ min delay to 500 milliseconds and repeat "
- "the test above\n");
- TEST_MUSTPASS(voe_vsync_->SetMinimumPlayoutDelay(0, 500));
- for (int i = 0; i < 15; i++) {
- TEST_MUSTPASS(voe_vsync_->GetDelayEstimate(0, valInt));
- TEST_LOG("Delay estimate = %d ms\n", valInt);
- TEST_MUSTPASS(valInt <= 45);
- SLEEP(1000);
- }
-
- TEST_LOG("Setting NetEQ min delay to 0 milliseconds and repeat"
- " the test above\n");
- TEST_MUSTPASS(voe_vsync_->SetMinimumPlayoutDelay(0, 0));
- for (int i = 0; i < 15; i++) {
- TEST_MUSTPASS(voe_vsync_->GetDelayEstimate(0, valInt));
- TEST_LOG("Delay estimate = %d ms\n", valInt);
- TEST_MUSTPASS(valInt <= 45);
- SLEEP(1000);
- }
-
-#if (defined (_WIN32) || (defined(WEBRTC_LINUX)) && !defined(WEBRTC_ANDROID))
- valInt = -1;
- TEST_MUSTPASS(voe_vsync_->GetPlayoutBufferSize(valInt));
- TEST_LOG("Soundcard buffer size = %d ms\n", valInt);
-#endif
-#else
- TEST_LOG("\n\n+++ Video sync tests NOT ENABLED +++\n");
-#endif // #ifdef _TEST_VIDEO_SYNC_
- //////////////
- // Encryption
-
-#ifdef _TEST_ENCRYPT_
- TEST_LOG("\n\n+++ Encryption tests +++\n\n");
-
-#ifdef WEBRTC_SRTP
- TEST_LOG("SRTP tests:\n");
-
- unsigned char encrKey[30] = {1, 2, 3, 4, 5, 6, 7, 8, 9, 0,
- 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0};
-
- TEST_LOG("Enable SRTP encryption and decryption, you should still hear"
- " the voice\n");
- TEST_MUSTPASS(voe_encrypt_->EnableSRTPSend(0,
- kCipherAes128CounterMode,
- 30,
- kAuthHmacSha1,
- 20, 4, kEncryptionAndAuthentication, encrKey));
- TEST_MUSTPASS(voe_encrypt_->EnableSRTPReceive(0,
- kCipherAes128CounterMode,
- 30,
- kAuthHmacSha1,
- 20, 4, kEncryptionAndAuthentication, encrKey));
- SLEEP(2000);
-
- TEST_LOG("Disabling decryption, you should hear nothing or garbage\n");
- TEST_MUSTPASS(voe_encrypt_->DisableSRTPReceive(0));
- SLEEP(2000);
-
- TEST_LOG("Enable decryption again, you should hear the voice again\n");
- TEST_MUSTPASS(voe_encrypt_->EnableSRTPReceive(0,
- kCipherAes128CounterMode,
- 30,
- kAuthHmacSha1,
- 20, 4, kEncryptionAndAuthentication, encrKey));
- SLEEP(2000);
-
- TEST_LOG("Disabling encryption and enabling decryption, you should"
- " hear nothing\n");
- TEST_MUSTPASS(voe_encrypt_->DisableSRTPSend(0));
- SLEEP(2000);
-
- TEST_LOG("Back to normal\n");
- // both SRTP sides are now inactive
- TEST_MUSTPASS(voe_encrypt_->DisableSRTPReceive(0));
- SLEEP(2000);
-
- TEST_LOG("Enable SRTP and SRTCP encryption and decryption,"
- " you should still hear the voice\n");
- TEST_MUSTPASS(voe_encrypt_->EnableSRTPSend(0,
- kCipherAes128CounterMode,
- 30,
- kAuthHmacSha1,
- 20, 4, kEncryptionAndAuthentication, encrKey, true));
- TEST_MUSTPASS(voe_encrypt_->EnableSRTPReceive(0,
- kCipherAes128CounterMode,
- 30,
- kAuthHmacSha1,
- 20, 4, kEncryptionAndAuthentication, encrKey, true));
- SLEEP(2000);
-
- TEST_LOG("Back to normal\n");
- TEST_MUSTPASS(voe_encrypt_->DisableSRTPSend(0));
- // both SRTP sides are now inactive
- TEST_MUSTPASS(voe_encrypt_->DisableSRTPReceive(0));
- SLEEP(2000);
-
-#else
- TEST_LOG("Skipping SRTP tests - WEBRTC_SRTP not defined \n");
-#endif // #ifdef WEBRTC_SRTP
- TEST_LOG("\nExternal encryption tests:\n");
- my_encryption * encObj = new my_encryption;
- TEST_MUSTPASS(voe_encrypt_->RegisterExternalEncryption(0, *encObj));
- TEST_LOG("Encryption enabled but you should still hear the voice\n");
- SLEEP(2000);
- TEST_LOG("Removing encryption object and deleting it\n");
- TEST_MUSTPASS(voe_encrypt_->DeRegisterExternalEncryption(0));
- delete encObj;
- SLEEP(2000);
-#else
- TEST_LOG("\n\n+++ Encryption tests NOT ENABLED +++\n");
-#endif // #ifdef _TEST_ENCRYPT_
- //////////////////
- // External media
-
-#ifdef _TEST_XMEDIA_
- TEST_LOG("\n\n+++ External media tests +++\n\n");
-
-#ifdef WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
- TEST_LOG("Stop playing file as microphone \n");
- TEST_LOG("==> Talk into the microphone \n");
- TEST_MUSTPASS(voe_file_->StopPlayingFileAsMicrophone(0));
-
- TEST_LOG("Enabling external playout\n");
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_xmedia_->SetExternalPlayoutStatus(true));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
- TEST_MUSTPASS(voe_base_->StartSend(0));
-
- TEST_LOG("Writing 2 secs of play data to vector\n");
- int getLen;
- WebRtc_Word16 speechData[32000];
- for (int i = 0; i < 200; i++) {
- TEST_MUSTPASS(voe_xmedia_->ExternalPlayoutGetData(speechData+i*160,
- 16000,
- 100,
- getLen));
- TEST_MUSTPASS(160 != getLen);
- SLEEP(10);
- }
-
- TEST_LOG("Disabling external playout\n");
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_xmedia_->SetExternalPlayoutStatus(false));
- TEST_MUSTPASS(voe_base_->StartPlayout(0));
-
- TEST_LOG("Enabling external recording\n");
- TEST_MUSTPASS(voe_xmedia_->SetExternalRecordingStatus(true));
- TEST_MUSTPASS(voe_base_->StartSend(0));
-
- TEST_LOG("Inserting record data from vector\n");
- for (int i = 0; i < 200; i++) {
- TEST_MUSTPASS(voe_xmedia_->ExternalRecordingInsertData(speechData+i*160,
- 160,
- 16000,
- 20));
- SLEEP(10);
- }
-
- TEST_LOG("Disabling external recording\n");
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_xmedia_->SetExternalRecordingStatus(false));
- TEST_MUSTPASS(voe_base_->StartSend(0));
-
- TEST_LOG("==> Start playing a file as microphone again \n");
- TEST_MUSTPASS(voe_file_->StartPlayingFileAsMicrophone(0, AudioFilename(),
- true, true));
-#else
- TEST_LOG("Skipping external rec and playout tests - \
- WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT not defined \n");
-#endif // WEBRTC_VOE_EXTERNAL_REC_AND_PLAYOUT
- TEST_LOG("Enabling playout external media processing => "
- "played audio should now be affected \n");
- TEST_MUSTPASS(voe_xmedia_->RegisterExternalMediaProcessing(
- -1, kPlaybackAllChannelsMixed, mobj));
- SLEEP(2000);
- TEST_LOG("Back to normal again \n");
- TEST_MUSTPASS(voe_xmedia_->DeRegisterExternalMediaProcessing(
- -1, kPlaybackAllChannelsMixed));
- SLEEP(2000);
- // Note that we must do per channel here because PlayFileAsMicrophone
- // is only done on ch 0.
- TEST_LOG("Enabling recording external media processing => "
- "played audio should now be affected \n");
- TEST_MUSTPASS(voe_xmedia_->RegisterExternalMediaProcessing(
- 0, kRecordingPerChannel, mobj));
- SLEEP(2000);
- TEST_LOG("Back to normal again \n");
- TEST_MUSTPASS(voe_xmedia_->DeRegisterExternalMediaProcessing(
- 0, kRecordingPerChannel));
- SLEEP(2000);
- TEST_LOG("Enabling recording external media processing => "
- "speak and make sure that voice is affected \n");
- TEST_MUSTPASS(voe_xmedia_->RegisterExternalMediaProcessing(
- -1, kRecordingAllChannelsMixed, mobj));
- SLEEP(2000);
- TEST_LOG("Back to normal again \n");
- TEST_MUSTPASS(voe_xmedia_->DeRegisterExternalMediaProcessing(
- -1, kRecordingAllChannelsMixed));
- SLEEP(2000);
-#else
- TEST_LOG("\n\n+++ External media tests NOT ENABLED +++\n");
-#endif // #ifdef _TEST_XMEDIA_
- /////////////////////
- // NetEQ statistics
-
-#ifdef _TEST_NETEQ_STATS_
- TEST_LOG("\n\n+++ NetEQ statistics tests +++\n\n");
-
-#ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
- NetworkStatistics nStats;
- TEST_MUSTPASS(voe_neteq_stats_->GetNetworkStatistics(0, nStats));
- TEST_LOG("\nNetwork statistics: \n");
- TEST_LOG(" currentAccelerateRate = %hu \n",
- nStats.currentAccelerateRate);
- TEST_LOG(" currentBufferSize = %hu \n",
- nStats.currentBufferSize);
- TEST_LOG(" currentDiscardRate = %hu \n",
- nStats.currentDiscardRate);
- TEST_LOG(" currentExpandRate = %hu \n",
- nStats.currentExpandRate);
- TEST_LOG(" currentPacketLossRate = %hu \n",
- nStats.currentPacketLossRate);
- TEST_LOG(" currentPreemptiveRate = %hu \n",
- nStats.currentPreemptiveRate);
- TEST_LOG(" preferredBufferSize = %hu \n",
- nStats.preferredBufferSize);
- TEST_LOG(" jitterPeaksFound = %i \n",
- nStats.jitterPeaksFound);
- TEST_LOG(" clockDriftPPM = %i \n",
- nStats.clockDriftPPM);
- TEST_LOG(" meanWaitingTimeMs = %i \n",
- nStats.meanWaitingTimeMs);
- TEST_LOG(" medianWaitingTimeMs = %i \n",
- nStats.medianWaitingTimeMs);
- TEST_LOG(" minWaitingTimeMs = %i \n",
- nStats.minWaitingTimeMs);
- TEST_LOG(" maxWaitingTimeMs = %i \n",
- nStats.maxWaitingTimeMs);
-#else
- TEST_LOG("Skipping NetEQ statistics tests - "
- "WEBRTC_VOICE_ENGINE_NETEQ_STATS_API not defined \n");
-#endif // #ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
-#else
- TEST_LOG("\n\n+++ NetEQ statistics tests NOT ENABLED +++\n");
-#endif // #ifdef _TEST_NETEQ_STATS_
- //////////////////
- // Stop streaming
- TEST_LOG("\n\n+++ Stop streaming +++\n\n");
-
- TEST_LOG("Stop playout, sending and listening \n");
- TEST_MUSTPASS(voe_base_->StopPlayout(0));
- TEST_MUSTPASS(voe_base_->StopSend(0));
- TEST_MUSTPASS(voe_base_->StopReceive(0));
-
- // Exit:
- TEST_LOG("Delete channel and terminate VE \n");
- TEST_MUSTPASS(voe_base_->DeleteChannel(0));
- TEST_MUSTPASS(voe_base_->Terminate());
-
- return 0;
-}
-
-int runAutoTest(TestType testType, ExtendedSelection extendedSel) {
- SubAPIManager apiMgr;
- apiMgr.DisplayStatus();
-
- ////////////////////////////////////
- // Create VoiceEngine and sub API:s
-
- voetest::VoETestManager tm;
- if (!tm.Init()) {
- return -1;
- }
- tm.GetInterfaces();
-
- //////////////////////
- // Run standard tests
-
- int mainRet(-1);
- if (testType == Standard) {
- mainRet = tm.DoStandardTest();
-
- ////////////////////////////////
- // Create configuration summary
- TEST_LOG("\n\n+++ Creating configuration summary file +++\n");
- createSummary(tm.VoiceEnginePtr());
- } else if (testType == Extended) {
- VoEExtendedTest xtend(tm);
-
- mainRet = 0;
- while (extendedSel != XSEL_None) {
- if (extendedSel == XSEL_Base || extendedSel == XSEL_All) {
- if ((mainRet = xtend.TestBase()) == -1)
- break;
- xtend.TestPassed("Base");
- }
- if (extendedSel == XSEL_CallReport || extendedSel == XSEL_All) {
- if ((mainRet = xtend.TestCallReport()) == -1)
- break;
- xtend.TestPassed("CallReport");
- }
- if (extendedSel == XSEL_Codec || extendedSel == XSEL_All) {
- if ((mainRet = xtend.TestCodec()) == -1)
- break;
- xtend.TestPassed("Codec");
- }
- if (extendedSel == XSEL_DTMF || extendedSel == XSEL_All) {
- if ((mainRet = xtend.TestDtmf()) == -1)
- break;
- xtend.TestPassed("Dtmf");
- }
- if (extendedSel == XSEL_Encryption || extendedSel == XSEL_All) {
- if ((mainRet = xtend.TestEncryption()) == -1)
- break;
- xtend.TestPassed("Encryption");
- }
- if (extendedSel == XSEL_ExternalMedia || extendedSel == XSEL_All) {
- if ((mainRet = xtend.TestExternalMedia()) == -1)
- break;
- xtend.TestPassed("ExternalMedia");
- }
- if (extendedSel == XSEL_File || extendedSel == XSEL_All) {
- if ((mainRet = xtend.TestFile()) == -1)
- break;
- xtend.TestPassed("File");
- }
- if (extendedSel == XSEL_Mixing || extendedSel == XSEL_All) {
- if ((mainRet = xtend.TestMixing()) == -1)
- break;
- xtend.TestPassed("Mixing");
- }
- if (extendedSel == XSEL_Hardware || extendedSel == XSEL_All) {
- if ((mainRet = xtend.TestHardware()) == -1)
- break;
- xtend.TestPassed("Hardware");
- }
- if (extendedSel == XSEL_NetEqStats || extendedSel == XSEL_All) {
- if ((mainRet = xtend.TestNetEqStats()) == -1)
- break;
- xtend.TestPassed("NetEqStats");
- }
- if (extendedSel == XSEL_Network || extendedSel == XSEL_All) {
- if ((mainRet = xtend.TestNetwork()) == -1)
- break;
- xtend.TestPassed("Network");
- }
- if (extendedSel == XSEL_RTP_RTCP || extendedSel == XSEL_All) {
- if ((mainRet = xtend.TestRTP_RTCP()) == -1)
- break;
- xtend.TestPassed("RTP_RTCP");
- }
- if (extendedSel == XSEL_VideoSync || extendedSel == XSEL_All) {
- if ((mainRet = xtend.TestVideoSync()) == -1)
- break;
- xtend.TestPassed("VideoSync");
- }
- if (extendedSel == XSEL_VolumeControl || extendedSel == XSEL_All) {
- if ((mainRet = xtend.TestVolumeControl()) == -1)
- break;
- xtend.TestPassed("VolumeControl");
- }
- if (extendedSel == XSEL_AudioProcessing || extendedSel == XSEL_All) {
- if ((mainRet = xtend.TestAPM()) == -1)
- break;
- xtend.TestPassed("AudioProcessing");
- }
- apiMgr.GetExtendedMenuSelection(extendedSel);
- } // while (extendedSel != XSEL_None)
- } else if (testType == Stress) {
- VoEStressTest stressTest(tm);
- mainRet = stressTest.DoTest();
- } else if (testType == Unit) {
- VoEUnitTest unitTest(tm);
- mainRet = unitTest.DoTest();
- } else if (testType == CPU) {
- VoECpuTest cpuTest(tm);
- mainRet = cpuTest.DoTest();
- } else {
- // Should never end up here
- TEST_LOG("INVALID SELECTION \n");
- }
-
- //////////////////
- // Release/Delete
-
- int releaseOK = tm.ReleaseInterfaces();
-
- if ((0 == mainRet) && (releaseOK != -1)) {
- TEST_LOG("\n\n*** All tests passed *** \n\n");
- } else {
- TEST_LOG("\n\n*** Test failed! *** \n");
- }
-
- return 0;
-}
-
-void createSummary(VoiceEngine* ve) {
- int len;
- char str[256];
-
-#ifdef MAC_IPHONE
- char summaryFilename[256];
- GetDocumentsDir(summaryFilename, 256);
- strcat(summaryFilename, "/summary.txt");
-#endif
-
- VoEBase* voe_base_ = VoEBase::GetInterface(ve);
- FILE* stream = fopen(summaryFilename, "wt");
-
- sprintf(str, "WebRTc VoiceEngine ");
-#if defined(_WIN32)
- strcat(str, "Win");
-#elif defined(WEBRTC_LINUX) && defined(WEBRTC_TARGET_PC) && \
- !defined(WEBRTC_ANDROID)
- strcat(str, "Linux");
-#elif defined(WEBRTC_MAC) && !defined(MAC_IPHONE)
- strcat(str, "Mac");
-#elif defined(WEBRTC_ANDROID)
- strcat(str, "Android");
-#elif defined(MAC_IPHONE)
- strcat(str, "iPhone");
-#endif
- // Add for other platforms as needed
-
- fprintf(stream, "%s\n", str);
- len = (int) strlen(str);
- for (int i = 0; i < len; i++) {
- fprintf(stream, "=");
- }
- fprintf(stream, "\n\n");
-
- char version[1024];
- char veVersion[24];
- voe_base_->GetVersion(version);
- // find first NL <=> end of VoiceEngine version string
- int pos = (int) strcspn(version, "\n");
- strncpy(veVersion, version, pos);
- veVersion[pos] = '\0';
- sprintf(str, "Version: %s\n", veVersion);
- fprintf(stream, "%s\n", str);
-
- sprintf(str, "Build date & time: %s\n", BUILDDATE " " BUILDTIME);
- fprintf(stream, "%s\n", str);
-
- strcpy(str, "G.711 A-law");
- fprintf(stream, "\nSupported codecs: %s\n", str);
- strcpy(str, " G.711 mu-law");
- fprintf(stream, "%s\n", str);
-#ifdef WEBRTC_CODEC_EG711
- strcpy(str, " Enhanced G.711 A-law");
- fprintf(stream, "%s\n", str);
- strcpy(str, " Enhanced G.711 mu-law");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_IPCMWB
- strcpy(str, " iPCM-wb");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_ILBC
- strcpy(str, " iLBC");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_ISAC
- strcpy(str, " iSAC");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_ISACLC
- strcpy(str, " iSAC-LC");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_G722
- strcpy(str, " G.722");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_G722_1
- strcpy(str, " G.722.1");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_G722_1C
- strcpy(str, " G.722.1C");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_G723
- strcpy(str, " G.723");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_G726
- strcpy(str, " G.726");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_G729
- strcpy(str, " G.729");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_G729_1
- strcpy(str, " G.729.1");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_GSMFR
- strcpy(str, " GSM-FR");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_GSMAMR
- strcpy(str, " AMR");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_GSMAMRWB
- strcpy(str, " AMR-WB");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_GSMEFR
- strcpy(str, " GSM-EFR");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_SPEEX
- strcpy(str, " Speex");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_SILK
- strcpy(str, " Silk");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_CODEC_PCM16
- strcpy(str, " L16");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef NETEQFIX_VOXWARE_SC3
- strcpy(str, " Voxware SC3");
- fprintf(stream, "%s\n", str);
-#endif
- // Always included
- strcpy(str, " AVT (RFC2833)");
- fprintf(stream, "%s\n", str);
-#ifdef WEBRTC_CODEC_RED
- strcpy(str, " RED (forward error correction)");
- fprintf(stream, "%s\n", str);
-#endif
-
- fprintf(stream, "\nEcho Control: ");
-#ifdef WEBRTC_VOICE_ENGINE_ECHO
- fprintf(stream, "Yes\n");
-#else
- fprintf(stream, "No\n");
-#endif
-
- fprintf(stream, "\nAutomatic Gain Control: ");
-#ifdef WEBRTC_VOICE_ENGINE_AGC
- fprintf(stream, "Yes\n");
-#else
- fprintf(stream, "No\n");
-#endif
-
- fprintf(stream, "\nNoise Reduction: ");
-#ifdef WEBRTC_VOICE_ENGINE_NR
- fprintf(stream, "Yes\n");
-#else
- fprintf(stream, "No\n");
-#endif
-
- fprintf(stream, "\nSRTP: ");
-#ifdef WEBRTC_SRTP
- fprintf(stream, "Yes\n");
-#else
- fprintf(stream, "No\n");
-#endif
-
- fprintf(stream, "\nExternal transport only: ");
-#ifdef WEBRTC_EXTERNAL_TRANSPORT
- fprintf(stream, "Yes\n");
-#else
- fprintf(stream, "No\n");
-#endif
-
- fprintf(stream, "\nTelephone event detection: ");
-#ifdef WEBRTC_DTMF_DETECTION
- fprintf(stream, "Yes\n");
-#else
- fprintf(stream, "No\n");
-#endif
-
- strcpy(str, "VoEBase");
- fprintf(stream, "\nSupported sub-APIs: %s\n", str);
-#ifdef WEBRTC_VOICE_ENGINE_CODEC_API
- strcpy(str, " VoECodec");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_DTMF_API
- strcpy(str, " VoEDtmf");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_FILE_API
- strcpy(str, " VoEFile");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_HARDWARE_API
- strcpy(str, " VoEHardware");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_NETWORK_API
- strcpy(str, " VoENetwork");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
- strcpy(str, " VoERTP_RTCP");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
- strcpy(str, " VoEVolumeControl");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
- strcpy(str, " VoEAudioProcessing");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
- strcpy(str, " VoeExternalMedia");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
- strcpy(str, " VoENetEqStats");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
- strcpy(str, " VoEEncryption");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
- strcpy(str, " VoECallReport");
- fprintf(stream, "%s\n", str);
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
- strcpy(str, " VoEVideoSync");
- fprintf(stream, "%s\n", str);
-#endif
-
- fclose(stream);
- voe_base_->Release();
-}
-
-/*********Knowledge Base******************/
-
-//An example for creating threads and calling VE API's from that thread.
-// Using thread. A generic API/Class for all platforms.
-#ifdef THEADTEST // find first NL <=> end of VoiceEngine version string
-//Definition of Thread Class
-class ThreadTest
-{
-public:
- ThreadTest(
- VoEBase* voe_base_);
- ~ThreadTest()
- {
- delete _myThread;
- }
- void Stop();
-private:
- static bool StartSend(
- void* obj);
- bool StartSend();
-
- ThreadWrapper* _myThread;
- VoEBase* _base;
-
- bool _stopped;
-};
-
-//Main function from where StartSend is invoked as a seperate thread.
-ThreadTest::ThreadTest(
- VoEBase* voe_base_)
-:
-_stopped(false),
-_base(voe_base_)
-{
- //Thread Creation
- _myThread = ThreadWrapper::CreateThread(StartSend, this, kLowPriority);
- unsigned int id = 0;
- //Starting the thread
- _myThread->Start(id);
-}
-
-//Calls the StartSend. This is to avoid the static declaration issue.
-bool
-ThreadTest::StartSend(
- void* obj)
-{
- return ((ThreadTest*)obj)->StartSend();
-}
-
-bool
-ThreadTest::StartSend()
-{
- _myThread->SetNotAlive(); //Ensures this function is called only once.
- _base->StartSend(0);
- return true;
-}
-
-void ThreadTest::Stop()
-{
- _stopped = true;
-}
-
-// Use the following to invoke ThreatTest from the main function.
-// ThreadTest* threadtest = new ThreadTest(voe_base_);
-#endif
-
-// An example to create a thread and call VE API's call from that thread.
-// Specific to Windows Platform
-#ifdef THREAD_TEST_WINDOWS
-//Thread Declaration. Need to be added in the class controlling/dictating
-// the main code.
-/**
- private:
- static unsigned int WINAPI StartSend(void* obj);
- unsigned int WINAPI StartSend();
- **/
-
-//Thread Definition
-unsigned int WINAPI mainTest::StartSend(void *obj)
-{
- return ((mainTest*)obj)->StartSend();
-}
-unsigned int WINAPI mainTest::StartSend()
-{
- //base
- voe_base_->StartSend(0);
-
- // TEST_MUSTPASS(voe_base_->StartSend(0));
- TEST_LOG("hi hi hi");
- return 0;
-}
-
-//Thread invoking. From the main code
-/*****
- unsigned int threadID=0;
- if ((HANDLE)_beginthreadex(NULL,
- 0,
- StartSend,
- (void*)this,
- 0,
- &threadID) == NULL)
- return false;
- ****/
-
-#endif
-
-} // namespace voetest
-
-int RunInManualMode(int argc, char** argv) {
- using namespace voetest;
-
- SubAPIManager apiMgr;
- apiMgr.DisplayStatus();
-
- printf("----------------------------\n");
- printf("Select type of test\n\n");
- printf(" (0) Quit\n");
- printf(" (1) Standard test\n");
- printf(" (2) Extended test(s)...\n");
- printf(" (3) Stress test(s)...\n");
- printf(" (4) Unit test(s)...\n");
- printf(" (5) CPU & memory reference test [Windows]...\n");
- printf("\n: ");
-
- int selection(0);
-
- dummy = scanf("%d", &selection);
-
- ExtendedSelection extendedSel(XSEL_Invalid);
-
- enum TestType testType(Invalid);
-
- switch (selection) {
- case 0:
- return 0;
- case 1:
- testType = Standard;
- break;
- case 2:
- testType = Extended;
- while (!apiMgr.GetExtendedMenuSelection(extendedSel))
- continue;
- break;
- case 3:
- testType = Stress;
- break;
- case 4:
- testType = Unit;
- break;
- case 5:
- testType = CPU;
- break;
- default:
- TEST_LOG("Invalid selection!\n");
- return 0;
- }
-
- if (testType == Standard) {
- TEST_LOG("\n\n+++ Running gtest-rewritten standard tests first +++\n\n");
-
- // Run the automated tests too in standard mode since we are gradually
- // rewriting the standard test to be automated. Running this will give
- // the standard suite the same completeness.
- RunInAutomatedMode(argc, argv);
- }
-
- // Function that can be called from other entry functions.
- return runAutoTest(testType, extendedSel);
-}
-
-// ----------------------------------------------------------------------------
-// main
-// ----------------------------------------------------------------------------
-
-#if !defined(MAC_IPHONE)
-int main(int argc, char** argv) {
- if (argc > 1 && std::string(argv[1]) == "--automated") {
- // This function is defined in automated_mode.cc to avoid macro clashes
- // with googletest (for instance the ASSERT_TRUE macro).
- return RunInAutomatedMode(argc, argv);
- }
-
- return RunInManualMode(argc, argv);
-}
-#endif //#if !defined(MAC_IPHONE)
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_standard_test.h b/trunk/src/voice_engine/main/test/auto_test/voe_standard_test.h
deleted file mode 100644
index ae0fb24..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/voe_standard_test.h
+++ /dev/null
@@ -1,370 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_STANDARD_TEST_H
-#define WEBRTC_VOICE_ENGINE_VOE_STANDARD_TEST_H
-
-#include <stdio.h>
-#include <string>
-
-#include "resource_manager.h"
-#include "voe_audio_processing.h"
-#include "voe_base.h"
-#include "voe_dtmf.h"
-#include "voe_errors.h"
-#include "voe_file.h"
-#include "voe_rtp_rtcp.h"
-#include "voe_test_defines.h"
-#include "voe_test_interface.h"
-#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
-#include "voe_call_report.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_CODEC_API
-#include "voe_codec.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
-#include "voe_encryption.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
-#include "voe_external_media.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_HARDWARE_API
-#include "voe_hardware.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_NETWORK_API
-#include "voe_network.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
-#include "voe_video_sync.h"
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
-#include "voe_volume_control.h"
-#endif
-
-#ifdef _TEST_NETEQ_STATS_
-namespace webrtc {
-class CriticalSectionWrapper;
-class ThreadWrapper;
-class VoENetEqStats;
-}
-#endif
-
-#if defined(WEBRTC_ANDROID)
-extern char mobileLogMsg[640];
-#endif
-
-namespace voetest {
-
-void createSummary(VoiceEngine* ve);
-void prepareDelivery();
-
-class TestRtpObserver : public VoERTPObserver {
- public:
- TestRtpObserver();
- virtual ~TestRtpObserver();
- virtual void OnIncomingCSRCChanged(const int channel,
- const unsigned int CSRC,
- const bool added);
- virtual void OnIncomingSSRCChanged(const int channel,
- const unsigned int SSRC);
- void Reset();
- public:
- unsigned int ssrc_[2];
- unsigned int csrc_[2][2]; // Stores 2 CSRCs for each channel.
- bool added_[2][2];
- int size_[2];
-};
-
-class MyTraceCallback : public TraceCallback {
- public:
- void Print(const TraceLevel level, const char *traceString, const int length);
-};
-
-class MyDeadOrAlive : public VoEConnectionObserver {
- public:
- void OnPeriodicDeadOrAlive(const int channel, const bool alive);
-};
-
-class ErrorObserver : public VoiceEngineObserver {
- public:
- ErrorObserver();
- void CallbackOnError(const int channel, const int errCode);
- public:
- int code;
-};
-
-class RtcpAppHandler : public VoERTCPObserver {
- public:
- void OnApplicationDataReceived(const int channel,
- const unsigned char sub_type,
- const unsigned int name,
- const unsigned char* data,
- const unsigned short length_in_bytes);
- void Reset();
- ~RtcpAppHandler() {}
- unsigned short length_in_bytes_;
- unsigned char data_[256];
- unsigned char sub_type_;
- unsigned int name_;
-};
-
-class DtmfCallback : public VoETelephoneEventObserver {
- public:
- int counter;
- DtmfCallback() {
- counter = 0;
- }
- virtual void OnReceivedTelephoneEventInband(int channel,
- int eventCode,
- bool endOfEvent) {
- char msg[128];
- if (endOfEvent)
- sprintf(msg, "(event=%d, [END])", eventCode);
- else
- sprintf(msg, "(event=%d, [START])", eventCode);
- TEST_LOG("%s", msg);
- if (!endOfEvent)
- counter++; // cound start of event only
- fflush(NULL);
- }
-
- virtual void OnReceivedTelephoneEventOutOfBand(int channel,
- int eventCode,
- bool endOfEvent) {
- char msg[128];
- if (endOfEvent)
- sprintf(msg, "(event=%d, [END])", eventCode);
- else
- sprintf(msg, "(event=%d, [START])", eventCode);
- TEST_LOG("%s", msg);
- if (!endOfEvent)
- counter++; // cound start of event only
- fflush(NULL);
- }
-};
-
-class my_encryption : public Encryption {
- void encrypt(int channel_no, unsigned char * in_data,
- unsigned char * out_data, int bytes_in, int * bytes_out);
- void decrypt(int channel_no, unsigned char * in_data,
- unsigned char * out_data, int bytes_in, int * bytes_out);
- void encrypt_rtcp(int channel_no, unsigned char * in_data,
- unsigned char * out_data, int bytes_in, int * bytes_out);
- void decrypt_rtcp(int channel_no, unsigned char * in_data,
- unsigned char * out_data, int bytes_in, int * bytes_out);
-};
-
-class RxCallback : public VoERxVadCallback {
- public:
- RxCallback() :
- _vadDecision(-1) {
- }
-
- virtual void OnRxVad(int, int vadDecision) {
- char msg[128];
- sprintf(msg, "RX VAD detected decision %d \n", vadDecision);
- TEST_LOG("%s", msg);
- _vadDecision = vadDecision;
- }
-
- int _vadDecision;
-};
-
-#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
-class MyMedia : public VoEMediaProcess {
- public:
- virtual void Process(const int channel,
- const ProcessingTypes type,
- WebRtc_Word16 audio_10ms[],
- const int length,
- const int samplingFreqHz,
- const bool stereo);
- private:
- int f;
-};
-#endif
-
-class SubAPIManager {
- public:
- SubAPIManager()
- : _base(true),
- _callReport(false),
- _codec(false),
- _dtmf(false),
- _encryption(false),
- _externalMedia(false),
- _file(false),
- _hardware(false),
- _netEqStats(false),
- _network(false),
- _rtp_rtcp(false),
- _videoSync(false),
- _volumeControl(false),
- _apm(false),
- _xsel(XSEL_Invalid) {
-#ifdef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
- _callReport = true;
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_CODEC_API
- _codec = true;
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_DTMF_API
- _dtmf = true;
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
- _encryption = true;
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
- _externalMedia = true;
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_FILE_API
- _file = true;
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_HARDWARE_API
- _hardware = true;
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
- _netEqStats = true;
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_NETWORK_API
- _network = true;
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
- _rtp_rtcp = true;
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
- _videoSync = true;
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
- _volumeControl = true;
-#endif
-#ifdef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
- _apm = true;
-#endif
- }
-
- void DisplayStatus() const;
- bool GetExtendedMenuSelection(ExtendedSelection& sel);
-
- private:
- bool _base, _callReport, _codec, _dtmf, _encryption;
- bool _externalMedia, _file, _hardware;
- bool _netEqStats, _network, _rtp_rtcp, _videoSync, _volumeControl, _apm;
- ExtendedSelection _xsel;
-};
-
-class VoETestManager {
- public:
- VoETestManager();
- ~VoETestManager();
-
- // Must be called after construction.
- bool Init();
-
- void GetInterfaces();
- int ReleaseInterfaces();
- int DoStandardTest();
-
- const char* AudioFilename() const {
- const std::string& result = resource_manager_.long_audio_file_path();
- if (result.length() == 0) {
- TEST_LOG("ERROR: Failed to open input file!");
- }
- return result.c_str();
- }
-
- VoiceEngine* VoiceEnginePtr() const {
- return voice_engine_;
- }
- VoEBase* BasePtr() const {
- return voe_base_;
- }
- VoECodec* CodecPtr() const {
- return voe_codec_;
- }
- VoEVolumeControl* VolumeControlPtr() const {
- return voe_volume_control_;
- }
- VoEDtmf* DtmfPtr() const {
- return voe_dtmf_;
- }
- VoERTP_RTCP* RTP_RTCPPtr() const {
- return voe_rtp_rtcp_;
- }
- VoEAudioProcessing* APMPtr() const {
- return voe_apm_;
- }
-
- VoENetwork* NetworkPtr() const {
- return voe_network_;
- }
-
- VoEFile* FilePtr() const {
- return voe_file_;
- }
-
- VoEHardware* HardwarePtr() const {
- return voe_hardware_;
- }
-
- VoEVideoSync* VideoSyncPtr() const {
- return voe_vsync_;
- }
-
- VoEEncryption* EncryptionPtr() const {
- return voe_encrypt_;
- }
-
- VoEExternalMedia* ExternalMediaPtr() const {
- return voe_xmedia_;
- }
-
- VoECallReport* CallReportPtr() const {
- return voe_call_report_;
- }
-
-#ifdef _TEST_NETEQ_STATS_
- VoENetEqStats* NetEqStatsPtr() const {
- return voe_neteq_stats_;
- }
-
-#endif
-
- private:
- int SetUp(ErrorObserver* error_observer);
- int TestStartStreaming(FakeExternalTransport& channel0_transport);
- int TestStartPlaying();
-
- bool initialized_;
-
- VoiceEngine* voice_engine_;
- VoEBase* voe_base_;
- VoECallReport* voe_call_report_;
- VoECodec* voe_codec_;
- VoEDtmf* voe_dtmf_;
- VoEEncryption* voe_encrypt_;
- VoEExternalMedia* voe_xmedia_;
- VoEFile* voe_file_;
- VoEHardware* voe_hardware_;
- VoENetwork* voe_network_;
-#ifdef _TEST_NETEQ_STATS_
- VoENetEqStats* voe_neteq_stats_;
-#endif
- VoERTP_RTCP* voe_rtp_rtcp_;
- VoEVideoSync* voe_vsync_;
- VoEVolumeControl* voe_volume_control_;
- VoEAudioProcessing* voe_apm_;
-
- ResourceManager resource_manager_;
-};
-
-} // namespace voetest
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_STANDARD_TEST_H
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_stress_test.cc b/trunk/src/voice_engine/main/test/auto_test/voe_stress_test.cc
deleted file mode 100644
index b7098ec..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/voe_stress_test.cc
+++ /dev/null
@@ -1,415 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// Some ideas of improvements:
-// Break out common init and maybe terminate to separate function(s).
-// How much trace should we have enabled?
-// API error counter, to print info and return -1 if any error.
-
-#include <stdlib.h>
-#include <stdio.h>
-#include <string.h>
-#include <time.h>
-#include <cassert>
-#if defined(_WIN32)
-#include <conio.h>
-#endif
-
-#include "voe_stress_test.h"
-#include "voe_standard_test.h"
-
-#include "../../source/voice_engine_defines.h" // defines build macros
-#include "thread_wrapper.h"
-
-using namespace webrtc;
-
-namespace voetest {
-
-#define VALIDATE_STRESS(expr) \
- if (expr) \
- { \
- printf("Error at line: %i, %s \n", __LINE__, #expr); \
- printf("Error code: %i \n", base->LastError()); \
- }
-
-#ifdef _WIN32
-// Pause if supported
-#define PAUSE_OR_SLEEP(x) PAUSE;
-#else
-// Sleep a bit instead if pause not supported
-#define PAUSE_OR_SLEEP(x) SLEEP(x);
-#endif
-
-extern char* GetFilename(char* filename);
-extern const char* GetFilename(const char* filename);
-extern int GetResource(char* resource, char* dest, int destLen);
-extern char* GetResource(char* resource);
-extern const char* GetResource(const char* resource);
-
-const char* VoEStressTest::_key = "====YUtFWRAAAAADBtIHgAAAAAEAAAAcAAAAAQBHU0ds"
- "b2JhbCBJUCBTb3VuZAAC\nAAAAIwAAAExpY2Vuc2VkIHRvIE5vcnRlbCBOZXR3cm9rcwAAAAA"
- "xAAAAZxZ7/u0M\niFYyTwSwko5Uutf7mh8S0O4rYZYTFidbzQeuGonuL17F/2oD/2pfDp3jL4"
- "Rf3z/A\nnlJsEJgEtASkDNFuwLILjGY0pzjjAYQp3pCl6z6k2MtE06AirdjGLYCjENpq/opX"
- "\nOrs3sIuwdYK5va/aFcsjBDmlsGCUM48RDYG9s23bIHYafXUC4ofOaubbZPWiPTmL\nEVJ8WH"
- "4F9pgNjALc14oJXfON7r/3\n=EsLx";
-
-int VoEStressTest::DoTest() {
- int test(-1);
- while (test != 0) {
- test = MenuSelection();
- switch (test) {
- case 0:
- // Quit stress test
- break;
- case 1:
- // All tests
- StartStopTest();
- CreateDeleteChannelsTest();
- MultipleThreadsTest();
- break;
- case 2:
- StartStopTest();
- break;
- case 3:
- CreateDeleteChannelsTest();
- break;
- case 4:
- MultipleThreadsTest();
- break;
- default:
- // Should not be possible
- printf("Invalid selection! (Test code error)\n");
- assert(false);
- } // switch
- } // while
-
- return 0;
-}
-
-int VoEStressTest::MenuSelection() {
- printf("------------------------------------------------\n");
- printf("Select stress test\n\n");
- printf(" (0) Quit\n");
- printf(" (1) All\n");
- printf("- - - - - - - - - - - - - - - - - - - - - - - - \n");
- printf(" (2) Start/stop\n");
- printf(" (3) Create/delete channels\n");
- printf(" (4) Multiple threads\n");
-
- const int maxMenuSelection = 4;
- int selection(-1);
-
- while ((selection < 0) || (selection > maxMenuSelection)) {
- printf("\n: ");
- int retval = scanf("%d", &selection);
- if ((retval != 1) || (selection < 0) || (selection > maxMenuSelection)) {
- printf("Invalid selection!\n");
- }
- }
-
- return selection;
-}
-
-int VoEStressTest::StartStopTest() {
- printf("------------------------------------------------\n");
- printf("Running start/stop test\n");
- printf("------------------------------------------------\n");
-
- printf("\nNOTE: this thest will fail after a while if Core audio is used\n");
- printf("because MS returns AUDCLNT_E_CPUUSAGE_EXCEEDED (VoE Error 10013).\n");
-
- // Get sub-API pointers
- VoEBase* base = _mgr.BasePtr();
-
- // Set trace
- // VALIDATE_STRESS(base->SetTraceFileName(
- // GetFilename("VoEStressTest_StartStop_trace.txt")));
- // VALIDATE_STRESS(base->SetDebugTraceFileName(
- // GetFilename("VoEStressTest_StartStop_trace_debug.txt")));
- // VALIDATE_STRESS(base->SetTraceFilter(kTraceStateInfo |
- // kTraceWarning | kTraceError |
- // kTraceCritical | kTraceApiCall |
- // kTraceMemory | kTraceInfo));
- VALIDATE_STRESS(base->Init());
- VALIDATE_STRESS(base->CreateChannel());
-
- ///////////// Start test /////////////
-
- int numberOfLoops(2000);
- int loopSleep(200);
- int i(0);
- int markInterval(20);
-
- printf("Running %d loops with %d ms sleep. Mark every %d loop. \n",
- numberOfLoops, loopSleep, markInterval);
- printf("Test will take approximately %d minutes. \n",
- numberOfLoops * loopSleep / 1000 / 60 + 1);
-
- for (i = 0; i < numberOfLoops; ++i) {
- VALIDATE_STRESS(base->SetLocalReceiver(0, 4800));
- VALIDATE_STRESS(base->SetSendDestination(0, 4800, "127.0.0.1"));
- VALIDATE_STRESS(base->StartReceive(0));
- VALIDATE_STRESS(base->StartPlayout(0));
- VALIDATE_STRESS(base->StartSend(0));
- if (!(i % markInterval))
- MARK();
- SLEEP(loopSleep);
- VALIDATE_STRESS(base->StopSend(0));
- VALIDATE_STRESS(base->StopPlayout(0));
- VALIDATE_STRESS(base->StopReceive(0));
- }
- ANL();
-
- VALIDATE_STRESS(base->SetLocalReceiver(0, 4800));
- VALIDATE_STRESS(base->SetSendDestination(0, 4800, "127.0.0.1"));
- VALIDATE_STRESS(base->StartReceive(0));
- VALIDATE_STRESS(base->StartPlayout(0));
- VALIDATE_STRESS(base->StartSend(0));
- printf("Verify that audio is good. \n");
- PAUSE_OR_SLEEP(20000);
- VALIDATE_STRESS(base->StopSend(0));
- VALIDATE_STRESS(base->StopPlayout(0));
- VALIDATE_STRESS(base->StopReceive(0));
-
- ///////////// End test /////////////
-
-
- // Terminate
- VALIDATE_STRESS(base->DeleteChannel(0));
- VALIDATE_STRESS(base->Terminate());
-
- printf("Test finished \n");
-
- return 0;
-}
-
-int VoEStressTest::CreateDeleteChannelsTest() {
- printf("------------------------------------------------\n");
- printf("Running create/delete channels test\n");
- printf("------------------------------------------------\n");
-
- // Get sub-API pointers
- VoEBase* base = _mgr.BasePtr();
-
- // Set trace
- // VALIDATE_STRESS(base->SetTraceFileName(
- // GetFilename("VoEStressTest_CreateChannels_trace.txt")));
- // VALIDATE_STRESS(base->SetDebugTraceFileName(
- // GetFilename("VoEStressTest_CreateChannels_trace_debug.txt")));
- // VALIDATE_STRESS(base->SetTraceFilter(kTraceStateInfo |
- // kTraceWarning | kTraceError |
- // kTraceCritical | kTraceApiCall |
- // kTraceMemory | kTraceInfo));
- VALIDATE_STRESS(base->Init());
-
- ///////////// Start test /////////////
-
- int numberOfLoops(10000);
- int loopSleep(10);
- int i(0);
- int markInterval(200);
-
- printf("Running %d loops with %d ms sleep. Mark every %d loop. \n",
- numberOfLoops, loopSleep, markInterval);
- printf("Test will take approximately %d minutes. \n",
- numberOfLoops * loopSleep / 1000 / 60 + 1);
-
- // Some possible extensions include:
- // Different sleep times (fixed or random) or zero.
- // Start call on all or some channels.
- // Two parts: first have a slight overweight to creating channels,
- // then to deleting. (To ensure we hit max channels and go to zero.)
- // Make sure audio is OK after test has finished.
-
- // Set up, start with maxChannels/2 channels
- const int maxChannels = base->MaxNumOfChannels();
- VALIDATE_STRESS(maxChannels < 1); // Should always have at least one channel
- bool* channelState = new bool[maxChannels];
- memset(channelState, 0, maxChannels * sizeof(bool));
- int channel(0);
- int noOfActiveChannels(0);
- for (i = 0; i < (maxChannels / 2); ++i) {
- channel = base->CreateChannel();
- VALIDATE_STRESS(channel < 0);
- if (channel >= 0) {
- channelState[channel] = true;
- ++noOfActiveChannels;
- }
- }
- srand((unsigned int) time(NULL));
- bool action(false);
- double rnd(0.0);
- int res(0);
-
- // Create/delete channels with slight
- for (i = 0; i < numberOfLoops; ++i) {
- // Randomize action (create or delete channel)
- action = rand() <= (RAND_MAX / 2);
- if (action) {
- if (noOfActiveChannels < maxChannels) {
- // Create new channel
- channel = base->CreateChannel();
- VALIDATE_STRESS(channel < 0);
- if (channel >= 0) {
- channelState[channel] = true;
- ++noOfActiveChannels;
- }
- }
- } else {
- if (noOfActiveChannels > 0) {
- // Delete random channel that's created [0, maxChannels - 1]
- do {
- rnd = static_cast<double> (rand());
- channel = static_cast<int> (rnd /
- (static_cast<double> (RAND_MAX) + 1.0f) *
- maxChannels);
- } while (!channelState[channel]); // Must find a created channel
-
- res = base->DeleteChannel(channel);
- VALIDATE_STRESS(0 != res);
- if (0 == res) {
- channelState[channel] = false;
- --noOfActiveChannels;
- }
- }
- }
-
- if (!(i % markInterval))
- MARK();
- SLEEP(loopSleep);
- }
- ANL();
-
- delete[] channelState;
-
- ///////////// End test /////////////
-
-
- // Terminate
- VALIDATE_STRESS(base->Terminate()); // Deletes all channels
-
- printf("Test finished \n");
-
- return 0;
-}
-
-int VoEStressTest::MultipleThreadsTest() {
- printf("------------------------------------------------\n");
- printf("Running multiple threads test\n");
- printf("------------------------------------------------\n");
-
- // Get sub-API pointers
- VoEBase* base = _mgr.BasePtr();
-
- // Set trace
- // VALIDATE_STRESS(base->SetTraceFileName(
- // GetFilename("VoEStressTest_MultipleThreads_trace.txt")));
- // VALIDATE_STRESS(base->SetDebugTraceFileName(
- // GetFilename("VoEStressTest_MultipleThreads_trace_debug.txt")));
- // VALIDATE_STRESS(base->SetTraceFilter(kTraceStateInfo |
- // kTraceWarning | kTraceError |
- // kTraceCritical | kTraceApiCall |
- // kTraceMemory | kTraceInfo));
-
- // Init
- VALIDATE_STRESS(base->Init());
- VALIDATE_STRESS(base->CreateChannel());
-
- ///////////// Start test /////////////
-
- int numberOfLoops(10000);
- int loopSleep(0);
- int i(0);
- int markInterval(1000);
-
- printf("Running %d loops with %d ms sleep. Mark every %d loop. \n",
- numberOfLoops, loopSleep, markInterval);
- printf("Test will take approximately %d minutes. \n",
- numberOfLoops * loopSleep / 1000 / 60 + 1);
-
- srand((unsigned int) time(NULL));
- int rnd(0);
-
- // Start extra thread
- const char* threadName = "StressTest Extra API Thread";
- _ptrExtraApiThread = ThreadWrapper::CreateThread(RunExtraApi, this,
- kNormalPriority, threadName);
- unsigned int id(0);
- VALIDATE_STRESS(!_ptrExtraApiThread->Start(id));
-
- // Some possible extensions include:
- // Add more API calls to randomize
- // More threads
- // Different sleep times (fixed or random).
- // Make sure audio is OK after test has finished.
-
- // Call random API functions here and in extra thread, ignore any error
- for (i = 0; i < numberOfLoops; ++i) {
- // This part should be equal to the marked part in the extra thread
- // --- BEGIN ---
- rnd = rand();
- if (rnd < (RAND_MAX / 2)) {
- // Start playout
- base->StartPlayout(0);
- } else {
- // Stop playout
- base->StopPlayout(0);
- }
- // --- END ---
-
- if (!(i % markInterval))
- MARK();
- SLEEP(loopSleep);
- }
- ANL();
-
- // Stop extra thread
- VALIDATE_STRESS(!_ptrExtraApiThread->Stop());
- delete _ptrExtraApiThread;
-
- ///////////// End test /////////////
-
- // Terminate
- VALIDATE_STRESS(base->Terminate()); // Deletes all channels
-
- printf("Test finished \n");
-
- return 0;
-}
-
-// Thread functions
-
-bool VoEStressTest::RunExtraApi(void* ptr) {
- return static_cast<VoEStressTest*> (ptr)->ProcessExtraApi();
-}
-
-bool VoEStressTest::ProcessExtraApi() {
- // Prepare
- VoEBase* base = _mgr.BasePtr();
- int rnd(0);
-
- // Call random API function, ignore any error
-
- // This part should be equal to the marked part in the main thread
- // --- BEGIN ---
- rnd = rand();
- if (rnd < (RAND_MAX / 2)) {
- // Start playout
- base->StartPlayout(0);
- } else {
- // Stop playout
- base->StopPlayout(0);
- }
- // --- END ---
-
- return true;
-}
-
-} // namespace voetest
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_stress_test.h b/trunk/src/voice_engine/main/test/auto_test/voe_stress_test.h
deleted file mode 100644
index b3a418c..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/voe_stress_test.h
+++ /dev/null
@@ -1,50 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_STRESS_TEST_H
-#define WEBRTC_VOICE_ENGINE_VOE_STRESS_TEST_H
-
-namespace webrtc {
-class ThreadWrapper;
-}
-
-namespace voetest {
-// TODO(andrew): using directives are not permitted.
-using namespace webrtc;
-
-class VoETestManager;
-
-class VoEStressTest {
- public:
- VoEStressTest(VoETestManager& mgr) :
- _mgr(mgr), _ptrExtraApiThread(NULL) {
- }
- ~VoEStressTest() {
- }
- int DoTest();
-
- private:
- int MenuSelection();
- int StartStopTest();
- int CreateDeleteChannelsTest();
- int MultipleThreadsTest();
-
- static bool RunExtraApi(void* ptr);
- bool ProcessExtraApi();
-
- VoETestManager& _mgr;
- static const char* _key;
-
- ThreadWrapper* _ptrExtraApiThread;
-};
-
-} // namespace voetest
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_STRESS_TEST_H
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_test_defines.h b/trunk/src/voice_engine/main/test/auto_test/voe_test_defines.h
deleted file mode 100644
index c9c103a..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/voe_test_defines.h
+++ /dev/null
@@ -1,188 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_TEST_DEFINES_H
-#define WEBRTC_VOICE_ENGINE_VOE_TEST_DEFINES_H
-
-// Read WEBRTC_VOICE_ENGINE_XXX_API compiler flags
-#include "engine_configurations.h"
-
-#ifdef WEBRTC_ANDROID
-#include <android/log.h>
-#define ANDROID_LOG_TAG "VoiceEngine Auto Test"
-#define TEST_LOG(...) \
- __android_log_print(ANDROID_LOG_DEBUG, ANDROID_LOG_TAG, __VA_ARGS__)
-#define TEST_LOG_ERROR(...) \
- __android_log_print(ANDROID_LOG_ERROR, ANDROID_LOG_TAG, __VA_ARGS__)
-#else
-#define TEST_LOG printf
-#define TEST_LOG_ERROR printf
-#define TEST_LOG_FLUSH fflush(NULL)
-#endif
-
-// Select the tests to execute, list order below is same as they will be
-// executed. Note that, all settings below will be overriden by sub-API
-// settings in engine_configurations.h.
-#define _TEST_BASE_
-#define _TEST_RTP_RTCP_
-#define _TEST_HARDWARE_
-#define _TEST_CODEC_
-#define _TEST_DTMF_
-#define _TEST_VOLUME_
-#define _TEST_AUDIO_PROCESSING_
-#define _TEST_FILE_
-#define _TEST_NETWORK_
-#define _TEST_CALL_REPORT_
-#define _TEST_VIDEO_SYNC_
-#define _TEST_ENCRYPT_
-#define _TEST_NETEQ_STATS_
-#define _TEST_XMEDIA_
-
-#define TESTED_AUDIO_LAYER kAudioPlatformDefault
-//#define TESTED_AUDIO_LAYER kAudioLinuxPulse
-
-// #define _ENABLE_VISUAL_LEAK_DETECTOR_ // Enables VLD to find memory leaks
-// #define _ENABLE_IPV6_TESTS_ // Enables IPv6 tests in network xtest
-// #define _USE_EXTENDED_TRACE_ // Adds unique trace files for extended test
-// #define _MEMORY_TEST_
-
-// Enable this when running instrumentation of some kind to exclude tests
-// that will not pass due to slowed down execution.
-// #define _INSTRUMENTATION_TESTING_
-
-// Exclude (override) API tests given preprocessor settings in
-// engine_configurations.h
-#ifndef WEBRTC_VOICE_ENGINE_CODEC_API
-#undef _TEST_CODEC_
-#endif
-#ifndef WEBRTC_VOICE_ENGINE_VOLUME_CONTROL_API
-#undef _TEST_VOLUME_
-#endif
-#ifndef WEBRTC_VOICE_ENGINE_DTMF_API
-#undef _TEST_DTMF_
-#endif
-#ifndef WEBRTC_VOICE_ENGINE_RTP_RTCP_API
-#undef _TEST_RTP_RTCP_
-#endif
-#ifndef WEBRTC_VOICE_ENGINE_AUDIO_PROCESSING_API
-#undef _TEST_AUDIO_PROCESSING_
-#endif
-#ifndef WEBRTC_VOICE_ENGINE_FILE_API
-#undef _TEST_FILE_
-#endif
-#ifndef WEBRTC_VOICE_ENGINE_VIDEO_SYNC_API
-#undef _TEST_VIDEO_SYNC_
-#endif
-#ifndef WEBRTC_VOICE_ENGINE_ENCRYPTION_API
-#undef _TEST_ENCRYPT_
-#endif
-#ifndef WEBRTC_VOICE_ENGINE_HARDWARE_API
-#undef _TEST_HARDWARE_
-#endif
-#ifndef WEBRTC_VOICE_ENGINE_EXTERNAL_MEDIA_API
-#undef _TEST_XMEDIA_
-#endif
-#ifndef WEBRTC_VOICE_ENGINE_NETWORK_API
-#undef _TEST_NETWORK_
-#endif
-#ifndef WEBRTC_VOICE_ENGINE_NETEQ_STATS_API
-#undef _TEST_NETEQ_STATS_
-#endif
-#ifndef WEBRTC_VOICE_ENGINE_CALL_REPORT_API
-#undef _TEST_CALL_REPORT_
-#endif
-
-// Some parts can cause problems while running Insure
-#ifdef __INSURE__
-#define _INSTRUMENTATION_TESTING_
-#undef WEBRTC_SRTP
-#endif
-
-// Time in ms to test each packet size for each codec
-#define CODEC_TEST_TIME 400
-
-#define MARK() TEST_LOG("."); fflush(NULL); // Add test marker
-#define ANL() TEST_LOG("\n") // Add New Line
-#define AOK() TEST_LOG("[Test is OK]"); fflush(NULL); // Add OK
-#if defined(_WIN32)
-#define PAUSE \
- { \
- TEST_LOG("Press any key to continue..."); \
- _getch(); \
- TEST_LOG("\n"); \
- }
-#else
-#define PAUSE \
- { \
- TEST_LOG("Continuing (pause not supported)\n"); \
- }
-#endif
-
-#define TEST(s) \
- { \
- TEST_LOG("Testing: %s", #s); \
- } \
-
-#ifdef _INSTRUMENTATION_TESTING_
-// Don't stop execution if error occurs
-#define TEST_MUSTPASS(expr) \
- { \
- if ((expr)) \
- { \
- TEST_LOG_ERROR("Error at line:%i, %s \n",__LINE__, #expr); \
- TEST_LOG_ERROR("Error code: %i\n",voe_base_->LastError()); \
- } \
- }
-#define TEST_ERROR(code) \
- { \
- int err = voe_base_->LastError(); \
- if (err != code) \
- { \
- TEST_LOG_ERROR("Invalid error code (%d, should be %d) at line %d\n",
- code, err, __LINE__);
-}
-}
-#else
-#define ASSERT_TRUE(expr) TEST_MUSTPASS(!(expr))
-#define ASSERT_FALSE(expr) TEST_MUSTPASS(expr)
-#define TEST_MUSTFAIL(expr) TEST_MUSTPASS(!((expr) == -1))
-#define TEST_MUSTPASS(expr) \
- { \
- if ((expr)) \
- { \
- TEST_LOG_ERROR("\nError at line:%i, %s \n",__LINE__, #expr); \
- TEST_LOG_ERROR("Error code: %i\n", voe_base_->LastError()); \
- PAUSE \
- return -1; \
- } \
- }
-#define TEST_ERROR(code) \
- { \
- int err = voe_base_->LastError(); \
- if (err != code) \
- { \
- TEST_LOG_ERROR("Invalid error code (%d, should be %d) at line %d\n", \
- err, code, __LINE__); \
- PAUSE \
- return -1; \
- } \
- }
-#endif // #ifdef _INSTRUMENTATION_TESTING_
-#define EXCLUDE() \
- { \
- TEST_LOG("\n>>> Excluding test at line: %i <<<\n\n",__LINE__); \
- }
-
-#define INCOMPLETE() \
- { \
- TEST_LOG("\n>>> Incomplete test at line: %i <<<\n\n",__LINE__); \
- }
-
-#endif // WEBRTC_VOICE_ENGINE_VOE_TEST_DEFINES_H
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_test_interface.h b/trunk/src/voice_engine/main/test/auto_test/voe_test_interface.h
deleted file mode 100644
index 87b37a4..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/voe_test_interface.h
+++ /dev/null
@@ -1,89 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-/*
- * Interface for starting test
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_TEST_INTERFACE_H
-#define WEBRTC_VOICE_ENGINE_VOE_TEST_INTERFACE_H
-
-#include "common_types.h"
-
-namespace webrtc {
-class CriticalSectionWrapper;
-class EventWrapper;
-class ThreadWrapper;
-class VoENetwork;
-}
-
-namespace voetest {
-// TODO(andrew): using directives are not permitted.
-using namespace webrtc;
-
-// TestType enumerator
-enum TestType {
- Invalid = -1, Standard = 0, Extended = 1, Stress = 2, Unit = 3, CPU = 4
-};
-
-// ExtendedSelection enumerator
-enum ExtendedSelection {
- XSEL_Invalid = -1,
- XSEL_None = 0,
- XSEL_All,
- XSEL_Base,
- XSEL_CallReport,
- XSEL_Codec,
- XSEL_DTMF,
- XSEL_Encryption,
- XSEL_ExternalMedia,
- XSEL_File,
- XSEL_Mixing,
- XSEL_Hardware,
- XSEL_NetEqStats,
- XSEL_Network,
- XSEL_RTP_RTCP,
- XSEL_VideoSync,
- XSEL_VolumeControl,
- XSEL_AudioProcessing,
-};
-
-// ----------------------------------------------------------------------------
-// External transport (Transport)
-// ----------------------------------------------------------------------------
-
-class FakeExternalTransport : public Transport {
- public:
- FakeExternalTransport(VoENetwork* ptr);
- virtual ~FakeExternalTransport();
- VoENetwork* my_network_;
- int SendPacket(int channel, const void *data, int len);
- int SendRTCPPacket(int channel, const void *data, int len);
- void SetDelayStatus(bool enabled, unsigned int delayInMs = 100);
- private:
- static bool Run(void* ptr);
- bool Process();
- private:
- ThreadWrapper* thread_;
- CriticalSectionWrapper* lock_;
- EventWrapper* event_;
- private:
- unsigned char packet_buffer_[1612];
- int length_;
- int channel_;
- bool delay_is_enabled_;
- int delay_time_in_ms_;
-};
-
-// Main test function
-int runAutoTest(TestType testType, ExtendedSelection extendedSel);
-
-} // namespace voetest
-#endif // WEBRTC_VOICE_ENGINE_VOE_TEST_INTERFACE_H
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_unit_test.cc b/trunk/src/voice_engine/main/test/auto_test/voe_unit_test.cc
deleted file mode 100644
index 1aa9e10..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/voe_unit_test.cc
+++ /dev/null
@@ -1,1083 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <string.h>
-#include <time.h>
-#include <cassert>
-#if defined(_WIN32)
-#include <conio.h>
-#endif
-
-#include "voe_unit_test.h"
-
-#include "../../source/voice_engine_defines.h"
-#include "thread_wrapper.h"
-
-using namespace webrtc;
-
-namespace voetest {
-
-#define CHECK(expr) \
- if (expr) \
- { \
- printf("Error at line: %i, %s \n", __LINE__, #expr); \
- printf("Error code: %i \n", base->LastError()); \
- PAUSE \
- return -1; \
- }
-
-extern char* GetFilename(char* filename);
-extern const char* GetFilename(const char* filename);
-extern int GetResource(char* resource, char* dest, int destLen);
-extern char* GetResource(char* resource);
-extern const char* GetResource(const char* resource);
-
-const char* VoEUnitTest::_key = "====YUtFWRAAAAADBtIHgAAAAAEAAAAcAAAAAQBHU0dsb2"
- "JhbCBJUCBTb3VuZAAC\nAAAAIwAAAExpY2Vuc2VkIHRvIE5vcnRlbCBOZXR3cm9rcwAAAAAxA"
- "AAAZxZ7/u0M\niFYyTwSwko5Uutf7mh8S0O4rYZYTFidbzQeuGonuL17F/2oD/2pfDp3jL4Rf"
- "3z/A\nnlJsEJgEtASkDNFuwLILjGY0pzjjAYQp3pCl6z6k2MtE06AirdjGLYCjENpq/opX\nO"
- "rs3sIuwdYK5va/aFcsjBDmlsGCUM48RDYG9s23bIHYafXUC4ofOaubbZPWiPTmL\nEVJ8WH4F"
- "9pgNjALc14oJXfON7r/3\n=EsLx";
-
-// ----------------------------------------------------------------------------
-// >>> R E A D M E F I R S T <<<
-// ----------------------------------------------------------------------------
-
-// 1) The user must ensure that the following codecs are included in VoE:
-//
-// - L16
-// - G.729
-// - G.722.1C
-
-// 2) It is also possible to modify the simulation time for each individual test
-//
-const int dTBetweenEachTest = 4000;
-
-// ----------------------------------------------------------------------------
-// Encrypt
-// ----------------------------------------------------------------------------
-
-void VoEUnitTest::encrypt(int channel_no, unsigned char * in_data,
- unsigned char * out_data, int bytes_in,
- int * bytes_out) {
- int i;
-
- if (!_extOnOff) {
- // no stereo emulation <=> pure bypass
- for (i = 0; i < bytes_in; i++)
- out_data[i] = in_data[i];
- *bytes_out = bytes_in;
- } else if (_extOnOff && (_extBitsPerSample == 16)) {
- // stereo emulation (sample based, 2 bytes per sample)
-
- const int nBytesPayload = bytes_in - 12;
-
- // RTP header (first 12 bytes)
- memcpy(out_data, in_data, 12);
-
- // skip RTP header
- short* ptrIn = (short*) &in_data[12];
- short* ptrOut = (short*) &out_data[12];
-
- // network byte order
- for (i = 0; i < nBytesPayload / 2; i++) {
- // produce two output samples for each input sample
- *ptrOut++ = *ptrIn; // left sample
- *ptrOut++ = *ptrIn; // right sample
- ptrIn++;
- }
-
- *bytes_out = 12 + 2 * nBytesPayload;
- } else if (_extOnOff && (_extBitsPerSample == 8)) {
- // stereo emulation (sample based, 1 bytes per sample)
-
- const int nBytesPayload = bytes_in - 12;
-
- // RTP header (first 12 bytes)
- memcpy(out_data, in_data, 12);
-
- // skip RTP header
- unsigned char* ptrIn = (unsigned char*) &in_data[12];
- unsigned char* ptrOut = (unsigned char*) &out_data[12];
-
- // network byte order
- for (i = 0; i < nBytesPayload; i++) {
- // produce two output samples for each input sample
- *ptrOut++ = *ptrIn; // left sample
- *ptrOut++ = *ptrIn; // right sample
- ptrIn++;
- }
-
- *bytes_out = 12 + 2 * nBytesPayload;
- } else if (_extOnOff && (_extBitsPerSample == -1)) {
- // stereo emulation (frame based)
-
- const int nBytesPayload = bytes_in - 12;
-
- // RTP header (first 12 bytes)
- memcpy(out_data, in_data, 12);
-
- // skip RTP header
- unsigned char* ptrIn = (unsigned char*) &in_data[12];
- unsigned char* ptrOut = (unsigned char*) &out_data[12];
-
- // left channel
- for (i = 0; i < nBytesPayload; i++) {
- *ptrOut++ = *ptrIn++;
- }
-
- ptrIn = (unsigned char*) &in_data[12];
-
- // right channel
- for (i = 0; i < nBytesPayload; i++) {
- *ptrOut++ = *ptrIn++;
- }
-
- *bytes_out = 12 + 2 * nBytesPayload;
- }
-}
-
-void VoEUnitTest::decrypt(int channel_no, unsigned char * in_data,
- unsigned char * out_data, int bytes_in,
- int * bytes_out) {
- int i;
- for (i = 0; i < bytes_in; i++)
- out_data[i] = in_data[i];
- *bytes_out = bytes_in;
-}
-
-void VoEUnitTest::encrypt_rtcp(int channel_no, unsigned char * in_data,
- unsigned char * out_data, int bytes_in,
- int * bytes_out) {
- int i;
- for (i = 0; i < bytes_in; i++)
- out_data[i] = in_data[i];
- *bytes_out = bytes_in;
-}
-
-void VoEUnitTest::decrypt_rtcp(int channel_no, unsigned char * in_data,
- unsigned char * out_data, int bytes_in,
- int * bytes_out) {
- int i;
- for (i = 0; i < bytes_in; i++)
- out_data[i] = in_data[i];
- *bytes_out = bytes_in;
-}
-
-void VoEUnitTest::SetStereoExternalEncryption(int channel, bool onOff,
- int bitsPerSample) {
- _extOnOff = onOff;
- _extChannel = channel;
- _extBitsPerSample = bitsPerSample;
-}
-
-// VoEVEMediaProcess
-MyMedia mpobj;
-
-// ----------------------------------------------------------------------------
-// VoEUnitTest
-// ----------------------------------------------------------------------------
-
-VoEUnitTest::VoEUnitTest(VoETestManager& mgr) :
- _mgr(mgr), _extOnOff(false), _extBitsPerSample(-1) {
- for (int i = 0; i < 32; i++) {
- _listening[i] = false;
- _playing[i] = false;
- _sending[i] = false;
- }
-}
-
-// ----------------------------------------------------------------------------
-// DoTest
-// ----------------------------------------------------------------------------
-
-int VoEUnitTest::DoTest() {
- int test(-1);
- int ret(0);
- while ((test != 0) && (ret != -1)) {
- test = MenuSelection();
- switch (test) {
- case 0:
- // Quit stress test
- break;
- case 1:
- ret = MixerTest();
- break;
- case 2:
- ret = MixerTest();
- break;
- default:
- // Should not be possible
- printf("Invalid selection! (Test code error)\n");
- assert(false);
- } // switch
- } // while
-
- return ret;
-}
-
-// ----------------------------------------------------------------------------
-// MenuSelection
-// ----------------------------------------------------------------------------
-
-int VoEUnitTest::MenuSelection() {
- printf("------------------------------------------------\n");
- printf("Select unit test\n\n");
- printf(" (0) Quit\n");
- printf(" (1) All\n");
- printf("- - - - - - - - - - - - - - - - - - - - - - - - \n");
- printf(" (2) Mixer\n");
-
- const int maxMenuSelection = 2;
- int selection(-1);
-
- while ((selection < 0) || (selection > maxMenuSelection)) {
- printf("\n: ");
- int retval = scanf("%d", &selection);
- if ((retval != 1) || (selection < 0) || (selection > maxMenuSelection)) {
- printf("Invalid selection!\n");
- }
- }
-
- return selection;
-}
-
-// ----------------------------------------------------------------------------
-// StartMedia
-// ----------------------------------------------------------------------------
-
-int VoEUnitTest::StartMedia(int channel, int rtpPort, bool listen, bool playout,
- bool send, bool fileAsMic, bool localFile) {
- VoEBase* base = _mgr.BasePtr();
- VoEFile* file = _mgr.FilePtr();
-
- _listening[channel] = false;
- _playing[channel] = false;
- _sending[channel] = false;
-
- CHECK(base->SetLocalReceiver(channel, rtpPort));
- CHECK(base->SetSendDestination(channel, rtpPort, "127.0.0.1"));
- if (listen) {
- _listening[channel] = true;
- CHECK(base->StartReceive(channel));
- }
- if (playout) {
- _playing[channel] = true;
- CHECK(base->StartPlayout(channel));
- }
- if (send) {
- _sending[channel] = true;
- CHECK(base->StartSend(channel));
- }
- if (fileAsMic) {
- // play mic as file, mix with microphone to ensure that SWB can be
- //tested as well
- const bool mixWithMic(true);
- CHECK(file->StartPlayingFileAsMicrophone(channel, _mgr.AudioFilename(),
- true, mixWithMic));
- }
- if (localFile) {
- CHECK(file->StartPlayingFileLocally(channel,
- GetResource("audio_short16.pcm"),
- false,
- kFileFormatPcm16kHzFile));
- }
-
- return 0;
-}
-
-// ----------------------------------------------------------------------------
-// StopMedia
-// ----------------------------------------------------------------------------
-
-int VoEUnitTest::StopMedia(int channel) {
- VoEBase* base = _mgr.BasePtr();
- VoEFile* file = _mgr.FilePtr();
-
- if (file->IsPlayingFileAsMicrophone(channel)) {
- CHECK(file->StopPlayingFileAsMicrophone(channel));
- }
- if (file->IsPlayingFileLocally(channel)) {
- CHECK(file->StopPlayingFileLocally(channel));
- }
- if (_listening[channel]) {
- _listening[channel] = false;
- CHECK(base->StopReceive(channel));
- }
- if (_playing[channel]) {
- _playing[channel] = false;
- CHECK(base->StopPlayout(channel));
- }
- if (_sending[channel]) {
- _sending[channel] = false;
- CHECK(base->StopSend(channel));
- }
-
- return 0;
-}
-
-void VoEUnitTest::Sleep(unsigned int timeMillisec, bool addMarker) {
- if (addMarker) {
- float dtSec = (float) ((float) timeMillisec / 1000.0);
- printf("[dT=%.1f]", dtSec);
- fflush(NULL);
- }
- ::Sleep(timeMillisec);
-}
-
-void VoEUnitTest::Wait() {
-#if defined(_WIN32)
- printf("\npress any key..."); fflush(NULL);
- _getch();
-#endif
-}
-
-void VoEUnitTest::Test(const char* msg) {
- printf("%s", msg);
- fflush(NULL);
- printf("\n");
- fflush(NULL);
-}
-
-int VoEUnitTest::MixerTest() {
- // Set up test parameters first
- //
- const int testTime(dTBetweenEachTest);
-
- printf("\n\n================================================\n");
- printf(" Mixer Unit Test\n");
- printf("================================================\n\n");
-
- // Get sub-API pointers
- //
- VoEBase* base = _mgr.BasePtr();
- VoECodec* codec = _mgr.CodecPtr();
- VoEFile* file = _mgr.FilePtr();
- VoEVolumeControl* volume = _mgr.VolumeControlPtr();
- VoEEncryption* encrypt = _mgr.EncryptionPtr();
- VoEDtmf* dtmf = _mgr.DtmfPtr();
- VoEExternalMedia* xmedia = _mgr.ExternalMediaPtr();
-
- // Set trace
- //
- VoiceEngine::SetTraceFile(GetFilename("UnitTest_Mixer_trace.txt"));
- VoiceEngine::SetTraceFilter(kTraceStateInfo | kTraceWarning | kTraceError |
- kTraceCritical | kTraceApiCall | kTraceMemory |
- kTraceInfo);
-
- // Init
- //
- CHECK(base->Init());
-
- // 8 kHz
- // CodecInst l16_8 = { 123, "L16", 8000, 160, 1, 128000 };
- CodecInst pcmu_8 = { 0, "pcmu", 8000, 160, 1, 64000 };
- // CodecInst g729_8 = { 18, "g729", 8000, 160, 1, 8000 };
-
- // 16 kHz
- CodecInst ipcmwb_16 = { 97, "ipcmwb", 16000, 320, 1, 80000 };
- CodecInst l16_16 = { 124, "L16", 16000, 320, 1, 256000 };
-
- // 32 kHz
- CodecInst l16_32 = { 125, "L16", 32000, 320, 1, 512000 };
- CodecInst g722_1c_32 = { 126, "G7221", 32000, 640, 1, 32000 };// 20ms@32kHz
-
- // ------------------------
- // Verify mixing frequency
- // ------------------------
-
- base->CreateChannel();
-
- Test(">> Verify correct mixing frequency:\n");
-
- Test("(ch 0) Sending file at 8kHz <=> mixing at 8kHz...");
- CHECK(StartMedia(0, 12345, true, true, true, true, false));
- Sleep(testTime);
-
- Test("(ch 0) Sending file at 16kHz <=> mixing at 16kHz...");
- CHECK(codec->SetSendCodec(0, ipcmwb_16));
- Sleep(testTime);
-
- Test("(ch 0) Sending speech at 32kHz <=> mixing at 32Hz...");
- CHECK(codec->SetSendCodec(0, l16_32));
- Sleep(testTime);
-
- Test("(ch 0) Sending file at 8kHz <=> mixing at 8kHz...");
- CHECK(codec->SetSendCodec(0, pcmu_8));
- Sleep(testTime);
-
- Test("(ch 0) Playing 16kHz file locally <=> mixing at 16kHz...");
- CHECK(file->StartPlayingFileLocally(0, GetResource("audio_long16.pcm"),
- false, kFileFormatPcm16kHzFile));
- Sleep(testTime);
- CHECK(file->StopPlayingFileLocally(0));
-
- base->CreateChannel();
-
- Test("(ch 0) Sending file at 8kHz <=> mixing at 8kHz...");
- CHECK(codec->SetSendCodec(0, pcmu_8));
- Sleep(testTime);
-
- Test("(ch 0) Sending speech at 32kHz <=> mixing at 32Hz...");
- CHECK(codec->SetSendCodec(0, l16_32));
- Sleep(testTime);
-
- Test("(ch 1) Playing 16kHz file locally <=> mixing at 32kHz...");
- CHECK(StartMedia(1, 54321, false, true, false, false, true));
- Sleep(testTime);
-
- CHECK(StopMedia(1));
- CHECK(StopMedia(0));
-
- base->DeleteChannel(1);
- base->DeleteChannel(0);
- ANL();
-
- // -------------------------
- // Verify stereo mode mixing
- // -------------------------
-
- base->CreateChannel();
- base->CreateChannel();
-
- // SetOutputVolumePan
- //
- // Ensure that all cases sound OK and that the mixer changes state between
- // mono and stereo as it should. A debugger is required to trace the state
- // transitions.
-
- Test(">> Verify correct mixing in stereo using SetOutputVolumePan():\n");
-
- Test("(ch 0) Playing 16kHz file locally <=> mixing in mono @ 16kHz...");
- CHECK(StartMedia(0, 12345, false, true, false, false, true));
- Sleep(testTime);
- Test("Panning volume to the left <=> mixing in stereo @ 16kHz...");
- CHECK(volume->SetOutputVolumePan(-1, 1.0, 0.0));
- Sleep(testTime);
- Test("Panning volume to the right <=> mixing in stereo @ 16kHz...");
- CHECK(volume->SetOutputVolumePan(-1, 0.0, 1.0));
- Sleep(testTime);
- Test("Back to center volume again <=> mixing in mono @ 16kHz...");
- CHECK(volume->SetOutputVolumePan(-1, 1.0, 1.0));
- Sleep(testTime);
- Test("(ch 1) Playing 16kHz file locally <=> mixing in mono @ 16kHz...");
- CHECK(StartMedia(1, 54321, false, true, false, false, true));
- Sleep(testTime);
- Test("Panning volume to the left <=> mixing in stereo @ 16kHz...");
- CHECK(volume->SetOutputVolumePan(-1, 1.0, 0.0));
- Sleep(testTime);
- Test("Back to center volume again <=> mixing in mono @ 16kHz...");
- CHECK(volume->SetOutputVolumePan(-1, 1.0, 1.0));
- Sleep(testTime);
- Test("(ch 1) Stopped playing file <=> mixing in mono @ 16kHz...");
- CHECK(StopMedia(1));
- Sleep(testTime);
- CHECK(StopMedia(0));
- Test("(ch 0) Sending file at 8kHz <=> mixing at 8kHz...");
- CHECK(StartMedia(0, 12345, true, true, true, true, false));
- Sleep(testTime);
- Test("(ch 0) Sending speech at 32kHz <=> mixing at 32kHz...");
- CHECK(codec->SetSendCodec(0, l16_32));
- Sleep(testTime);
- Test("Panning volume to the right <=> mixing in stereo @ 32kHz...");
- CHECK(volume->SetOutputVolumePan(-1, 0.0, 1.0));
- Sleep(testTime);
- Test("Back to center volume again <=> mixing in mono @ 32kHz...");
- CHECK(volume->SetOutputVolumePan(-1, 1.0, 1.0));
- Sleep(testTime);
- CHECK(StopMedia(0));
- ANL();
-
- base->DeleteChannel(0);
- base->DeleteChannel(1);
-
- // SetChannelOutputVolumePan
- //
- // Ensure that all cases sound OK and that the mixer changes state between
- // mono and stereo as it should. A debugger is required to trace the state
- // transitions.
-
- base->CreateChannel();
- base->CreateChannel();
-
- Test(">> Verify correct mixing in stereo using"
- " SetChannelOutputVolumePan():\n");
-
- Test("(ch 0) Playing 16kHz file locally <=> mixing in mono @ 16kHz...");
- CHECK(StartMedia(0, 12345, false, true, false, false, true));
- Sleep(testTime);
- Test("(ch 0) Panning channel volume to the left <=> mixing in stereo @ "
- "16kHz...");
- CHECK(volume->SetOutputVolumePan(0, 1.0, 0.0));
- Sleep(testTime);
- Test("(ch 0) Panning channel volume to the right <=> mixing in stereo"
- " @ 16kHz...");
- CHECK(volume->SetOutputVolumePan(0, 0.0, 1.0));
- Sleep(testTime);
- Test("(ch 0) Back to center volume again <=> mixing in mono @"
- " 16kHz...");
- CHECK(volume->SetOutputVolumePan(0, 1.0, 1.0));
- Sleep(testTime);
- Test("(ch 1) Playing 16kHz file locally <=> mixing in mono @ 16kHz...");
- CHECK(StartMedia(1, 54321, false, true, false, false, true));
- Sleep(testTime);
- Test("(ch 1) Panning channel volume to the left <=> mixing in stereo "
- "@ 16kHz...");
- CHECK(volume->SetOutputVolumePan(1, 1.0, 0.0));
- Sleep(testTime);
- Test("(ch 1) Back to center volume again <=> mixing in mono @ 16kHz...");
- CHECK(volume->SetOutputVolumePan(1, 1.0, 1.0));
- Sleep(testTime);
- Test("(ch 1) Stopped playing file <=> mixing in mono @ 16kHz...");
- CHECK(StopMedia(1));
- Sleep(testTime);
- CHECK(StopMedia(0));
- ANL();
-
- base->DeleteChannel(0);
- base->DeleteChannel(1);
-
- // Emulate stereo-encoding using Encryption
- //
- // Modify the transmitted RTP stream by using external encryption.
- // Supports frame based and sample based "stereo-encoding schemes".
-
- base->CreateChannel();
-
- Test(">> Verify correct mixing in stereo using emulated stereo input:\n");
-
- // enable external encryption
- CHECK(encrypt->RegisterExternalEncryption(0, *this));
- Test("(ch 0) External Encryption is now enabled:");
-
- Test("(ch 0) Sending file at 8kHz <=> mixing in mono @ 8kHz...");
- CHECK(StartMedia(0, 12345, true, true, true, true, false));
- Sleep(testTime);
-
- // switch to 16kHz (L16) sending codec
- CHECK(codec->SetSendCodec(0, l16_16));
- Test("(ch 0) Sending file at 16kHz (L16) <=> mixing in mono @ 16kHz...");
- Sleep(testTime);
-
- // register L16 as 2-channel codec on receiving side =>
- // should sound bad since RTP module splits all received packets in half
- // (sample based)
- CHECK(base->StopPlayout(0));
- CHECK(base->StopReceive(0));
- l16_16.channels = 2;
- CHECK(codec->SetRecPayloadType(0, l16_16));
- CHECK(base->StartReceive(0));
- CHECK(base->StartPlayout(0));
- Test("(ch 0) 16kHz L16 is now registered as 2-channel codec on RX side => "
- "should sound bad...");
- Sleep(testTime);
-
- // emulate sample-based stereo encoding
- Test("(ch 0) Emulate sample-based stereo encoding on sending side => "
- "should sound OK...");
- SetStereoExternalEncryption(0, true, 16);
- Sleep(testTime);
- Test("(ch 0) Stop emulating sample-based stereo encoding on sending side =>"
- " should sound bad...");
- SetStereoExternalEncryption(0, false, 16);
- Sleep(testTime);
- Test("(ch 0) Emulate sample-based stereo encoding on sending side => "
- "should sound OK...");
- SetStereoExternalEncryption(0, true, 16);
- Sleep(testTime);
-
- // switch to 32kHz (L16) sending codec and disable stereo encoding
- CHECK(codec->SetSendCodec(0, l16_32));
- SetStereoExternalEncryption(0, false, 16);
- Test("(ch 0) Sending file and spech at 32kHz (L16) <=> mixing in mono @ "
- "32kHz...");
- Sleep(testTime);
-
- // register L16 32kHz as 2-channel codec on receiving side
- CHECK(base->StopPlayout(0));
- CHECK(base->StopReceive(0));
- l16_32.channels = 2;
- CHECK(codec->SetRecPayloadType(0, l16_32));
- CHECK(base->StartReceive(0));
- CHECK(base->StartPlayout(0));
- Test("(ch 0) 32kHz L16 is now registered as 2-channel codec on RX side =>"
- " should sound bad...");
- Sleep(testTime);
-
- // emulate sample-based stereo encoding
- Test("(ch 0) Emulate sample-based stereo encoding on sending side =>"
- " should sound OK...");
- SetStereoExternalEncryption(0, true, 16);
- Sleep(testTime);
-
- StopMedia(0);
- l16_32.channels = 1;
-
- // disable external encryption
- CHECK(encrypt->DeRegisterExternalEncryption(0));
- ANL();
-
- base->DeleteChannel(0);
-
- // ------------------
- // Verify put-on-hold
- // ------------------
-
- base->CreateChannel();
- base->CreateChannel();
-
- Test(">> Verify put-on-hold functionality:\n");
-
- Test("(ch 0) Sending at 8kHz...");
- CHECK(StartMedia(0, 12345, true, true, true, true, false));
- Sleep(testTime);
-
- CHECK(base->SetOnHoldStatus(0, true, kHoldPlayOnly));
- Test("(ch 0) Playout is now on hold...");
- Sleep(testTime);
- CHECK(base->SetOnHoldStatus(0, false, kHoldPlayOnly));
- Test("(ch 0) Playout is now enabled again...");
- Sleep(testTime);
-
- Test("(ch 0) Sending at 16kHz...");
- l16_16.channels = 1;
- CHECK(codec->SetSendCodec(0, l16_16));
- Sleep(testTime);
-
- CHECK(base->SetOnHoldStatus(0, true, kHoldPlayOnly));
- Test("(ch 0) Playout is now on hold...");
- Sleep(testTime);
- CHECK(base->SetOnHoldStatus(0, false, kHoldPlayOnly));
- Test("(ch 0) Playout is now enabled again...");
- Sleep(testTime);
-
- Test("(ch 0) Perform minor panning to the left to force mixing in"
- " stereo...");
- CHECK(volume->SetOutputVolumePan(0, (float)1.0, (float)0.7));
- Sleep(testTime);
-
- CHECK(base->SetOnHoldStatus(0, true, kHoldPlayOnly));
- Test("(ch 0) Playout is now on hold...");
- Sleep(testTime);
- CHECK(base->SetOnHoldStatus(0, false, kHoldPlayOnly));
- Test("(ch 0) Playout is now enabled again...");
- Sleep(testTime);
-
- Test("(ch 0) Back to center volume again...");
- CHECK(volume->SetOutputVolumePan(0, 1.0, 1.0));
- Sleep(testTime);
-
- Test("(ch 1) Add 16kHz local file to the mixer...");
- CHECK(StartMedia(1, 54321, false, true, false, false, true));
- Sleep(testTime);
-
- CHECK(base->SetOnHoldStatus(0, true, kHoldPlayOnly));
- Test("(ch 0) Playout is now on hold...");
- Sleep(testTime);
- CHECK(base->SetOnHoldStatus(1, true, kHoldPlayOnly));
- Test("(ch 1) Playout is now on hold => should be silent...");
- Sleep(testTime);
- CHECK(base->SetOnHoldStatus(0, false, kHoldPlayOnly));
- Test("(ch 0) Playout is now enabled again...");
- CHECK(base->SetOnHoldStatus(1, false, kHoldPlayOnly));
- Test("(ch 1) Playout is now enabled again...");
- Sleep(testTime);
- StopMedia(1);
- Test("(ch 1) Stopped playing file...");
- Sleep(testTime);
- StopMedia(0);
- ANL();
-
- base->DeleteChannel(0);
- base->DeleteChannel(1);
-
- // -----------------------------------
- // Verify recording of playout to file
- // -----------------------------------
-
- // StartRecordingPlayout
- //
- // Verify that the correct set of signals is recorded in the mixer.
- // Record each channel and all channels (-1) to ensure that post and pre
- // mixing recording works.
-
- base->CreateChannel();
- base->CreateChannel();
-
- Test(">> Verify file-recording functionality:\n");
-
- Test("(ch 0) Sending at 8kHz...");
- CHECK(StartMedia(0, 12345, true, true, true, true, false));
- Sleep(testTime);
-
- Test("(ch 0) Recording of playout to 16kHz PCM file...");
- CHECK(file->StartRecordingPlayout(
- 0, GetFilename("RecordedPlayout16kHz.pcm"), NULL));
- Sleep(testTime);
- CHECK(file->StopRecordingPlayout(0));
-
- Test("(ch 0) Playing out the recorded file...");
- CHECK(volume->SetInputMute(0, true));
- CHECK(file->StartPlayingFileLocally(
- 0, GetFilename("RecordedPlayout16kHz.pcm")));
- Sleep(testTime);
- CHECK(file->StopPlayingFileLocally(0));
- CHECK(volume->SetInputMute(0, false));
-
- CHECK(codec->SetSendCodec(0, l16_16));
- Test("(ch 0) Sending at 16kHz (L16)...");
- Sleep(testTime);
-
- Test("(ch 0) Recording of playout to 16kHz PCM file...");
- CHECK(file->StartRecordingPlayout(
- 0, GetFilename("RecordedPlayout16kHz.pcm"), NULL));
- Sleep(testTime);
- CHECK(file->StopRecordingPlayout(0));
-
- Test("(ch 0) Playing out the recorded file...");
- CHECK(volume->SetInputMute(0, true));
- CHECK(file->StartPlayingFileLocally(
- 0, GetFilename("RecordedPlayout16kHz.pcm")));
- Sleep(testTime);
- CHECK(file->StopPlayingFileLocally(0));
- CHECK(volume->SetInputMute(0, false));
-
- CHECK(codec->SetSendCodec(0, l16_32));
- Test("(ch 0) Sending at 32kHz (L16)...");
- Sleep(testTime);
-
- Test("(ch 0) Recording of playout to 16kHz PCM file...");
- CHECK(file->StartRecordingPlayout(
- 0, GetFilename("RecordedPlayout16kHz.pcm"), NULL));
- Sleep(testTime);
- CHECK(file->StopRecordingPlayout(0));
-
- Test("(ch 0) Playing out the recorded file...");
- CHECK(volume->SetInputMute(0, true));
- CHECK(file->StartPlayingFileLocally(
- 0, GetFilename("RecordedPlayout16kHz.pcm")));
- Sleep(testTime);
- CHECK(file->StopPlayingFileLocally(0));
- CHECK(volume->SetInputMute(0, false));
-
- Test("(ch 0) Sending at 16kHz without file as mic but file added on the"
- " playout side instead...");
- CHECK(StopMedia(0));
- CHECK(StartMedia(0, 12345, false, true, false, false, true));
- CHECK(codec->SetSendCodec(0, l16_16));
- Sleep(testTime);
-
- Test("(ch 0) Recording of playout to 16kHz PCM file...");
- CHECK(file->StartRecordingPlayout(
- 0, GetFilename("RecordedPlayout16kHz.pcm"), NULL));
- Sleep(testTime);
- CHECK(file->StopRecordingPlayout(0));
- CHECK(file->StopPlayingFileLocally(0));
-
- Test("(ch 0) Playing out the recorded file...");
- CHECK(file->StartPlayingFileLocally(
- 0, GetFilename("RecordedPlayout16kHz.pcm")));
- Sleep(testTime);
- CHECK(file->StopPlayingFileLocally(0));
-
- CHECK(StopMedia(0));
- CHECK(StopMedia(1));
-
- Test("(ch 0) Sending at 16kHz...");
- CHECK(StartMedia(0, 12345, true, true, true, false, false));
- CHECK(codec->SetSendCodec(0, l16_16));
- Test("(ch 1) Adding playout file...");
- CHECK(StartMedia(1, 33333, false, true, false, false, true));
- Sleep(testTime);
-
- Test("(ch -1) Speak while recording all channels to add mixer input on "
- "channel 0...");
- CHECK(file->StartRecordingPlayout(
- -1, GetFilename("RecordedPlayout16kHz.pcm"), NULL));
- Sleep(testTime);
- CHECK(file->StopRecordingPlayout(-1));
- CHECK(file->StopPlayingFileLocally(1));
-
- Test("(ch 0) Playing out the recorded file...");
- CHECK(volume->SetInputMute(0, true));
- CHECK(file->StartPlayingFileLocally(
- 0, GetFilename("RecordedPlayout16kHz.pcm")));
- Sleep(testTime);
- CHECK(file->StopPlayingFileLocally(0));
- CHECK(volume->SetInputMute(0, false));
-
- CHECK(StopMedia(0));
- CHECK(StopMedia(1));
- ANL();
-
- // StartRecordingPlayoutStereo
-
- Test(">> Verify recording of playout in stereo:\n");
-
- Test("(ch 0) Sending at 32kHz...");
- CHECK(codec->SetSendCodec(0, l16_16));
- CHECK(StartMedia(0, 12345, true, true, true, true, false));
- Sleep(testTime);
-
- Test("Modified master balance (L=10%%, R=100%%) to force stereo mixing...");
- CHECK(volume->SetOutputVolumePan(-1, (float)0.1, (float)1.0));
- Sleep(testTime);
-
- /*
- Test("Recording of left and right channel playout to two 16kHz PCM "
- "files...");
- file->StartRecordingPlayoutStereo(
- GetFilename("RecordedPlayout_Left_16kHz.pcm"),
- GetFilename("RecordedPlayout_Right_16kHz.pcm"), StereoBoth);
- Sleep(testTime);
- Test("Back to center volume again...");
- CHECK(volume->SetOutputVolumePan(-1, (float)1.0, (float)1.0));
- */
-
- Test("(ch 0) Playing out the recorded file for the left channel (10%%)...");
- CHECK(volume->SetInputMute(0, true));
- CHECK(file->StartPlayingFileLocally(
- 0, GetFilename("RecordedPlayout_Left_16kHz.pcm")));
- Sleep(testTime);
- CHECK(file->StopPlayingFileLocally(0));
-
- Test("(ch 0) Playing out the recorded file for the right channel (100%%) =>"
- " should sound louder than the left channel...");
- CHECK(file->StartPlayingFileLocally(
- 0, GetFilename("RecordedPlayout_Right_16kHz.pcm")));
- Sleep(testTime);
- CHECK(file->StopPlayingFileLocally(0));
- CHECK(volume->SetInputMute(0, false));
-
- base->DeleteChannel(0);
- base->DeleteChannel(1);
- ANL();
-
- // ---------------------------
- // Verify inserted Dtmf tones
- // ---------------------------
-
- Test(">> Verify Dtmf feedback functionality:\n");
-
- base->CreateChannel();
-
- for (int i = 0; i < 2; i++) {
- if (i == 0)
- Test("Dtmf direct feedback is now enabled...");
- else
- Test("Dtmf direct feedback is now disabled...");
-
- CHECK(dtmf->SetDtmfFeedbackStatus(true, (i==0)));
-
- Test("(ch 0) Sending at 32kHz using G.722.1C...");
- CHECK(codec->SetRecPayloadType(0, g722_1c_32));
- CHECK(codec->SetSendCodec(0, g722_1c_32));
- CHECK(StartMedia(0, 12345, true, true, true, false, false));
- Sleep(500);
-
- Test("(ch 0) Sending outband Dtmf events => ensure that they are added"
- " to the mixer...");
- // ensure that receiver will not play out outband Dtmf
- CHECK(dtmf->SetSendTelephoneEventPayloadType(0, 118));
- CHECK(dtmf->SendTelephoneEvent(0, 9, true, 390));
- Sleep(500);
- CHECK(dtmf->SendTelephoneEvent(0, 1, true, 390));
- Sleep(500);
- CHECK(dtmf->SendTelephoneEvent(0, 5, true, 390));
- Sleep(500);
- Sleep(testTime - 1500);
-
- Test("(ch 0) Changing codec to 8kHz PCMU...");
- CHECK(codec->SetSendCodec(0, pcmu_8));
- Sleep(500);
-
- Test("(ch 0) Sending outband Dtmf events => ensure that they are added"
- " to the mixer...");
- CHECK(dtmf->SendTelephoneEvent(0, 9, true, 390));
- Sleep(500);
- CHECK(dtmf->SendTelephoneEvent(0, 1, true, 390));
- Sleep(500);
- CHECK(dtmf->SendTelephoneEvent(0, 5, true, 390));
- Sleep(500);
- Sleep(testTime - 1500);
-
- Test("(ch 0) Changing codec to 16kHz L16...");
- CHECK(codec->SetSendCodec(0, l16_16));
- Sleep(500);
-
- Test("(ch 0) Sending outband Dtmf events => ensure that they are added"
- " to the mixer...");
- CHECK(dtmf->SendTelephoneEvent(0, 9, true, 390));
- Sleep(500);
- CHECK(dtmf->SendTelephoneEvent(0, 1, true, 390));
- Sleep(500);
- CHECK(dtmf->SendTelephoneEvent(0, 5, true, 390));
- Sleep(500);
- Sleep(testTime - 1500);
-
- StopMedia(0);
- ANL();
- }
-
- base->DeleteChannel(0);
-
- // ---------------------------
- // Verify external processing
- // --------------------------
-
- base->CreateChannel();
-
- Test(">> Verify external media processing:\n");
-
- Test("(ch 0) Playing 16kHz file locally <=> mixing in mono @ 16kHz...");
- CHECK(StartMedia(0, 12345, false, true, false, false, true));
- Sleep(testTime);
- Test("Enabling playout external media processing => played audio should "
- "now be affected");
- CHECK(xmedia->RegisterExternalMediaProcessing(
- 0, kPlaybackAllChannelsMixed, mpobj));
- Sleep(testTime);
- Test("(ch 0) Sending speech at 32kHz <=> mixing at 32kHz...");
- CHECK(codec->SetSendCodec(0, l16_32));
- Sleep(testTime);
- printf("Back to normal again\n");
- CHECK(xmedia->DeRegisterExternalMediaProcessing(0,
- kPlaybackAllChannelsMixed));
- Sleep(testTime);
- printf("Enabling playout external media processing on ch 0 => "
- "played audio should now be affected\n");
- CHECK(xmedia->RegisterExternalMediaProcessing(0, kPlaybackPerChannel,
- mpobj));
- Sleep(testTime);
- Test("Panning volume to the right <=> mixing in stereo @ 32kHz...");
- CHECK(volume->SetOutputVolumePan(-1, 0.0, 1.0));
- Sleep(testTime);
- Test("Back to center volume again <=> mixing in mono @ 32kHz...");
- CHECK(volume->SetOutputVolumePan(-1, 1.0, 1.0));
- Sleep(testTime);
- printf("Back to normal again\n");
- CHECK(xmedia->DeRegisterExternalMediaProcessing(0, kPlaybackPerChannel));
- Sleep(testTime);
- CHECK(StopMedia(0));
- ANL();
-
- base->DeleteChannel(0);
-
- // --------------------------------------------------
- // Extended tests of emulated stereo encoding schemes
- // --------------------------------------------------
-
- CodecInst PCMU;
- CodecInst G729;
- CodecInst L16_8;
- CodecInst L16_16;
- CodecInst L16_32;
-
- base->CreateChannel();
-
- Test(">> Verify emulated stereo encoding for differenct codecs:\n");
-
- // enable external encryption
- CHECK(encrypt->RegisterExternalEncryption(0, *this));
- Test("(ch 0) External Encryption is now enabled:");
-
- // register all codecs on the receiving side
- strcpy(PCMU.plname, "PCMU");
- PCMU.channels = 2;
- PCMU.pacsize = 160;
- PCMU.plfreq = 8000;
- PCMU.pltype = 125;
- PCMU.rate = 64000;
- CHECK(codec->SetRecPayloadType(0, PCMU));
-
- strcpy(G729.plname, "G729");
- G729.channels = 2;
- G729.pacsize = 160;
- G729.plfreq = 8000;
- G729.pltype = 18;
- G729.rate = 8000;
- CHECK(codec->SetRecPayloadType(0, G729));
-
- strcpy(L16_8.plname, "L16");
- L16_8.channels = 2;
- L16_8.pacsize = 160;
- L16_8.plfreq = 8000;
- L16_8.pltype = 120;
- L16_8.rate = 128000;
- CHECK(codec->SetRecPayloadType(0, L16_8));
-
- strcpy(L16_16.plname, "L16");
- L16_16.channels = 2;
- L16_16.pacsize = 320;
- L16_16.plfreq = 16000;
- L16_16.pltype = 121;
- L16_16.rate = 256000;
- CHECK(codec->SetRecPayloadType(0, L16_16));
-
- // NOTE - we cannot send larger than 1500 bytes per RTP packet
- strcpy(L16_32.plname, "L16");
- L16_32.channels = 2;
- L16_32.pacsize = 320;
- L16_32.plfreq = 32000;
- L16_32.pltype = 122;
- L16_32.rate = 512000;
- CHECK(codec->SetRecPayloadType(0, L16_32));
-
- // sample-based, 8-bits per sample
-
- Test("(ch 0) Sending using G.711 (sample based, 8 bits/sample)...");
- PCMU.channels = 1;
- CHECK(codec->SetSendCodec(0, PCMU));
- SetStereoExternalEncryption(0, true, 8);
- CHECK(StartMedia(0, 12345, true, true, true, true, false));
- Sleep(testTime);
-
- // sample-based, 16-bits per sample
-
- Test("(ch 0) Sending using L16 8kHz (sample based, 16 bits/sample)...");
- L16_8.channels = 1;
- CHECK(codec->SetSendCodec(0, L16_8));
- SetStereoExternalEncryption(0, true, 16);
- Sleep(testTime);
-
- Test("(ch 0) Sending using L16 16kHz (sample based, 16 bits/sample)...");
- L16_16.channels = 1;
- CHECK(codec->SetSendCodec(0, L16_16));
- Sleep(testTime);
-
- Test("(ch 0) Sending using L16 32kHz (sample based, 16 bits/sample)...");
- L16_32.channels = 1;
- CHECK(codec->SetSendCodec(0, L16_32));
- Sleep(testTime);
-
- Test("(ch 0) Sending using G.729 (frame based)...");
- G729.channels = 1;
- CHECK(codec->SetSendCodec(0, G729));
- Sleep(testTime);
-
- StopMedia(0);
-
- // disable external encryption
- CHECK(encrypt->DeRegisterExternalEncryption(0));
-
- base->DeleteChannel(0);
-
- // ------------------------------------------------------------------------
- CHECK(base->Terminate());
-
- printf("\n\n------------------------------------------------\n");
- printf(" Test passed!\n");
- printf("------------------------------------------------\n\n");
-
- return 0;
-}
-
-} // namespace voetest
diff --git a/trunk/src/voice_engine/main/test/auto_test/voe_unit_test.h b/trunk/src/voice_engine/main/test/auto_test/voe_unit_test.h
deleted file mode 100644
index ce53241..0000000
--- a/trunk/src/voice_engine/main/test/auto_test/voe_unit_test.h
+++ /dev/null
@@ -1,69 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_VOICE_ENGINE_VOE_UNIT_TEST_H
-#define WEBRTC_VOICE_ENGINE_VOE_UNIT_TEST_H
-
-#include "voe_standard_test.h"
-
-namespace voetest {
-
-class VoETestManager;
-
-class VoEUnitTest : public Encryption {
- public:
- VoEUnitTest(VoETestManager& mgr);
- ~VoEUnitTest() {}
- int DoTest();
-
- protected:
- // Encryption
- void encrypt(int channel_no, unsigned char * in_data,
- unsigned char * out_data, int bytes_in, int * bytes_out);
- void decrypt(int channel_no, unsigned char * in_data,
- unsigned char * out_data, int bytes_in, int * bytes_out);
- void encrypt_rtcp(int channel_no, unsigned char * in_data,
- unsigned char * out_data, int bytes_in, int * bytes_out);
- void decrypt_rtcp(int channel_no, unsigned char * in_data,
- unsigned char * out_data, int bytes_in, int * bytes_out);
-
- private:
- int MenuSelection();
- int MixerTest();
- void Sleep(unsigned int timeMillisec, bool addMarker = false);
- void Wait();
- int StartMedia(int channel,
- int rtpPort,
- bool listen,
- bool playout,
- bool send,
- bool fileAsMic,
- bool localFile);
- int StopMedia(int channel);
- void Test(const char* msg);
- void SetStereoExternalEncryption(int channel, bool onOff, int bitsPerSample);
-
- private:
- VoETestManager& _mgr;
- static const char* _key;
-
- private:
- bool _listening[32];
- bool _playing[32];
- bool _sending[32];
-
- private:
- bool _extOnOff;
- int _extBitsPerSample;
- int _extChannel;
-};
-
-} // namespace voetest
-#endif // WEBRTC_VOICE_ENGINE_VOE_UNIT_TEST_H
diff --git a/trunk/src/voice_engine/main/test/cmd_test/Android.mk b/trunk/src/voice_engine/main/test/cmd_test/Android.mk
deleted file mode 100644
index f1a4f1a..0000000
--- a/trunk/src/voice_engine/main/test/cmd_test/Android.mk
+++ /dev/null
@@ -1,60 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-LOCAL_PATH:= $(call my-dir)
-
-# voice engine test app
-
-include $(CLEAR_VARS)
-
-include $(LOCAL_PATH)/../../../../../android-webrtc.mk
-
-LOCAL_MODULE_TAGS := tests
-LOCAL_CPP_EXTENSION := .cc
-LOCAL_SRC_FILES:= \
- voe_cmd_test.cc
-
-# Flags passed to both C and C++ files.
-LOCAL_CFLAGS := \
- '-DWEBRTC_TARGET_PC' \
- '-DWEBRTC_ANDROID' \
- '-DDEBUG'
-
-LOCAL_C_INCLUDES := \
- $(LOCAL_PATH)/../../interface \
- $(LOCAL_PATH)/../../../.. \
- $(LOCAL_PATH)/../../../../.. \
- external/gtest/include \
- frameworks/base/include
-
-LOCAL_SHARED_LIBRARIES := \
- libutils \
- libmedia \
- libcamera_client \
- libgui \
- libhardware \
- libandroid_runtime \
- libbinder
-
-#libwilhelm.so libDunDef-Android.so libbinder.so libsystem_server.so
-
-LOCAL_MODULE:= webrtc_voe_cmd
-
-ifdef NDK_ROOT
-LOCAL_SHARED_LIBRARIES += \
- libstlport_shared \
- libwebrtc-voice-jni \
- libwebrtc_audio_preprocessing
-include $(BUILD_EXECUTABLE)
-else
-LOCAL_SHARED_LIBRARIES += \
- libstlport \
- libwebrtc
-include external/stlport/libstlport.mk
-include $(BUILD_NATIVE_TEST)
-endif
diff --git a/trunk/src/voice_engine/main/test/cmd_test/voe_cmd_test.cc b/trunk/src/voice_engine/main/test/cmd_test/voe_cmd_test.cc
deleted file mode 100644
index 99e88d2..0000000
--- a/trunk/src/voice_engine/main/test/cmd_test/voe_cmd_test.cc
+++ /dev/null
@@ -1,922 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <stdio.h>
-#include <stdlib.h>
-#include <string.h>
-#ifndef _WIN32
-#include <unistd.h>
-#endif
-
-#include <vector>
-
-#include "gtest/gtest.h"
-#include "test/testsupport/fileutils.h"
-
-#include "voe_errors.h"
-#include "voe_base.h"
-#include "voe_codec.h"
-#include "voe_volume_control.h"
-#include "voe_dtmf.h"
-#include "voe_rtp_rtcp.h"
-#include "voe_audio_processing.h"
-#include "voe_file.h"
-#include "voe_video_sync.h"
-#include "voe_encryption.h"
-#include "voe_hardware.h"
-#include "voe_external_media.h"
-#include "voe_network.h"
-#include "voe_neteq_stats.h"
-#include "engine_configurations.h"
-
-// Enable this this flag to run this test with hard coded
-// IP/Port/codec and start test automatically with key input
-// it could be useful in repeat tests.
-//#define DEBUG
-
-// #define EXTERNAL_TRANSPORT
-
-using namespace webrtc;
-
-#define VALIDATE \
- if (res != 0) \
- { \
- printf("*** Error at position %i / line %i \n", cnt, __LINE__); \
- printf("*** Error code = %i \n", base1->LastError()); \
- } \
- cnt++;
-
-VoiceEngine* m_voe = NULL;
-VoEBase* base1 = NULL;
-VoECodec* codec = NULL;
-VoEVolumeControl* volume = NULL;
-VoEDtmf* dtmf = NULL;
-VoERTP_RTCP* rtp_rtcp = NULL;
-VoEAudioProcessing* apm = NULL;
-VoENetwork* netw = NULL;
-VoEFile* file = NULL;
-VoEVideoSync* vsync = NULL;
-VoEEncryption* encr = NULL;
-VoEHardware* hardware = NULL;
-VoEExternalMedia* xmedia = NULL;
-VoENetEqStats* neteqst = NULL;
-
-void RunTest(std::string out_path);
-
-#ifdef EXTERNAL_TRANSPORT
-
-class my_transportation : public Transport
-{
- int SendPacket(int channel,const void *data,int len);
- int SendRTCPPacket(int channel, const void *data, int len);
-};
-
-int my_transportation::SendPacket(int channel,const void *data,int len)
-{
- netw->ReceivedRTPPacket(channel, data, len);
- return 0;
-}
-
-int my_transportation::SendRTCPPacket(int channel, const void *data, int len)
-{
- netw->ReceivedRTCPPacket(channel, data, len);
- return 0;
-}
-
-my_transportation my_transport;
-#endif
-
-class MyObserver : public VoiceEngineObserver {
- public:
- virtual void CallbackOnError(const int channel, const int err_code);
-};
-
-void MyObserver::CallbackOnError(const int channel, const int err_code) {
- // Add printf for other error codes here
- if (err_code == VE_TYPING_NOISE_WARNING) {
- printf(" TYPING NOISE DETECTED \n");
- } else if (err_code == VE_RECEIVE_PACKET_TIMEOUT) {
- printf(" RECEIVE PACKET TIMEOUT \n");
- } else if (err_code == VE_PACKET_RECEIPT_RESTARTED) {
- printf(" PACKET RECEIPT RESTARTED \n");
- } else if (err_code == VE_RUNTIME_PLAY_WARNING) {
- printf(" RUNTIME PLAY WARNING \n");
- } else if (err_code == VE_RUNTIME_REC_WARNING) {
- printf(" RUNTIME RECORD WARNING \n");
- } else if (err_code == VE_SATURATION_WARNING) {
- printf(" SATURATION WARNING \n");
- } else if (err_code == VE_RUNTIME_PLAY_ERROR) {
- printf(" RUNTIME PLAY ERROR \n");
- } else if (err_code == VE_RUNTIME_REC_ERROR) {
- printf(" RUNTIME RECORD ERROR \n");
- } else if (err_code == VE_REC_DEVICE_REMOVED) {
- printf(" RECORD DEVICE REMOVED \n");
- }
-}
-
-int main() {
- int res = 0;
- int cnt = 0;
-
- printf("Test started \n");
-
- m_voe = VoiceEngine::Create();
- base1 = VoEBase::GetInterface(m_voe);
- codec = VoECodec::GetInterface(m_voe);
- apm = VoEAudioProcessing::GetInterface(m_voe);
- volume = VoEVolumeControl::GetInterface(m_voe);
- dtmf = VoEDtmf::GetInterface(m_voe);
- rtp_rtcp = VoERTP_RTCP::GetInterface(m_voe);
- netw = VoENetwork::GetInterface(m_voe);
- file = VoEFile::GetInterface(m_voe);
- vsync = VoEVideoSync::GetInterface(m_voe);
- encr = VoEEncryption::GetInterface(m_voe);
- hardware = VoEHardware::GetInterface(m_voe);
- xmedia = VoEExternalMedia::GetInterface(m_voe);
- neteqst = VoENetEqStats::GetInterface(m_voe);
-
- MyObserver my_observer;
-
-#if defined(WEBRTC_ANDROID)
- const std::string out_path = "/sdcard/";
-#else
- const std::string out_path = webrtc::test::OutputPath();
-#endif
- const std::string trace_filename = out_path + "webrtc_trace.txt";
-
- printf("Set trace filenames (enable trace)\n");
- VoiceEngine::SetTraceFilter(kTraceAll);
- res = VoiceEngine::SetTraceFile(trace_filename.c_str());
- VALIDATE;
-
- res = VoiceEngine::SetTraceCallback(NULL);
- VALIDATE;
-
- printf("Init\n");
- res = base1->Init();
- if (res != 0) {
- printf("\nError calling Init: %d\n", base1->LastError());
- fflush(NULL);
- exit(1);
- }
-
- res = base1->RegisterVoiceEngineObserver(my_observer);
- VALIDATE;
-
- cnt++;
- printf("Version\n");
- char tmp[1024];
- res = base1->GetVersion(tmp);
- VALIDATE;
- cnt++;
- printf("%s\n", tmp);
-
- RunTest(out_path);
-
- printf("Terminate \n");
-
- base1->DeRegisterVoiceEngineObserver();
-
- res = base1->Terminate();
- VALIDATE;
-
- if (base1)
- base1->Release();
-
- if (codec)
- codec->Release();
-
- if (volume)
- volume->Release();
-
- if (dtmf)
- dtmf->Release();
-
- if (rtp_rtcp)
- rtp_rtcp->Release();
-
- if (apm)
- apm->Release();
-
- if (netw)
- netw->Release();
-
- if (file)
- file->Release();
-
- if (vsync)
- vsync->Release();
-
- if (encr)
- encr->Release();
-
- if (hardware)
- hardware->Release();
-
- if (xmedia)
- xmedia->Release();
-
- if (neteqst)
- neteqst->Release();
-
- VoiceEngine::Delete(m_voe);
-
- return 0;
-}
-
-void RunTest(std::string out_path) {
- int chan, cnt, res;
- CodecInst cinst;
- cnt = 0;
- int i;
- int codecinput;
- bool AEC = false;
- bool AGC = true;
- bool AGC1 = false;
- bool VAD = false;
- bool NS = false;
- bool NS1 = false;
- bool typing_detection = false;
- bool muted = false;
- bool on_hold = false;
-
-#if defined(WEBRTC_ANDROID)
- std::string resource_path = "/sdcard/";
-#else
- std::string resource_path = webrtc::test::ProjectRootPath();
- if (resource_path == webrtc::test::kCannotFindProjectRootDir) {
- printf("*** Unable to get project root directory. "
- "File playing may fail. ***\n");
- // Fall back to the current directory.
- resource_path = "./";
- } else {
- resource_path += "test/data/voice_engine/";
- }
-#endif
- const std::string audio_filename = resource_path + "audio_long16.pcm";
-
- const std::string play_filename = out_path + "recorded_playout.pcm";
- const std::string mic_filename = out_path + "recorded_mic.pcm";
-
- chan = base1->CreateChannel();
- if (chan < 0) {
- printf("Error at position %i\n", cnt);
- printf("************ Error code = %i\n", base1->LastError());
- fflush(NULL);
- }
- cnt++;
-
- int j = 0;
-#ifdef EXTERNAL_TRANSPORT
- my_transportation ch0transport;
- printf("Enabling external transport \n");
- netw->RegisterExternalTransport(0, ch0transport);
-#else
- char ip[64];
-#ifdef DEBUG
- strcpy(ip, "127.0.0.1");
-#else
- char localip[64];
- netw->GetLocalIP(localip);
- printf("local IP:%s\n", localip);
-
- printf("1. 127.0.0.1 \n");
- printf("2. Specify IP \n");
- ASSERT_EQ(1, scanf("%i", &i));
-
- if (1 == i)
- strcpy(ip, "127.0.0.1");
- else {
- printf("Specify remote IP: ");
- ASSERT_EQ(1, scanf("%s", ip));
- }
-#endif
-
- int colons(0);
- while (ip[j] != '\0' && j < 64 && !(colons = (ip[j++] == ':')))
- ;
- if (colons) {
- printf("Enabling IPv6\n");
- res = netw->EnableIPv6(0);
- VALIDATE;
- }
-
- int rPort;
-#ifdef DEBUG
- rPort=8500;
-#else
- printf("Specify remote port (1=1234): ");
- ASSERT_EQ(1, scanf("%i", &rPort));
- if (1 == rPort)
- rPort = 1234;
- printf("Set Send port \n");
-#endif
-
- printf("Set Send IP \n");
- res = base1->SetSendDestination(chan, rPort, ip);
- VALIDATE;
-
- int lPort;
-#ifdef DEBUG
- lPort=8500;
-#else
- printf("Specify local port (1=1234): ");
- ASSERT_EQ(1, scanf("%i", &lPort));
- if (1 == lPort)
- lPort = 1234;
- printf("Set Rec Port \n");
-#endif
- res = base1->SetLocalReceiver(chan, lPort);
- VALIDATE;
-#endif
-
- printf("\n");
- for (i = 0; i < codec->NumOfCodecs(); i++) {
- res = codec->GetCodec(i, cinst);
- VALIDATE;
- if (strncmp(cinst.plname, "ISAC", 4) == 0 && cinst.plfreq == 32000) {
- printf("%i. ISAC-swb pltype:%i plfreqi:%i\n", i, cinst.pltype,
- cinst.plfreq);
- }
- else {
- printf("%i. %s pltype:%i plfreq:%i\n", i, cinst.plname,
- cinst.pltype, cinst.plfreq);
- }
- }
-#ifdef DEBUG
- codecinput=0;
-#else
- printf("Select send codec: ");
- ASSERT_EQ(1, scanf("%i", &codecinput));
-#endif
- codec->GetCodec(codecinput, cinst);
-
- printf("Set primary codec\n");
- res = codec->SetSendCodec(chan, cinst);
- VALIDATE;
-
- const int kMaxNumChannels = 8;
- int channel_index = 0;
- std::vector<int> channels(kMaxNumChannels);
- for (i = 0; i < kMaxNumChannels; ++i) {
- channels[i] = base1->CreateChannel();
- int port = rPort + (i + 1) * 2;
- res = base1->SetSendDestination(channels[i], port, ip);
- VALIDATE;
- res = base1->SetLocalReceiver(channels[i], port);
- VALIDATE;
- res = codec->SetSendCodec(channels[i], cinst);
- VALIDATE;
- }
-
- // Call loop
- bool newcall = true;
- while (newcall) {
-
-#ifdef WEBRTC_LINUX
- int rd(-1), pd(-1);
- res = hardware->GetNumOfRecordingDevices(rd);
- VALIDATE;
- res = hardware->GetNumOfPlayoutDevices(pd);
- VALIDATE;
-
- char dn[128] = { 0 };
- char guid[128] = { 0 };
- printf("\nPlayout devices (%d): \n", pd);
- for (j=0; j<pd; ++j) {
- res = hardware->GetPlayoutDeviceName(j, dn, guid);
- VALIDATE;
- printf(" %d: %s \n", j, dn);
- }
-
- printf("Recording devices (%d): \n", rd);
- for (j=0; j<rd; ++j) {
- res = hardware->GetRecordingDeviceName(j, dn, guid);
- VALIDATE;
- printf(" %d: %s \n", j, dn);
- }
-
- printf("Select playout device: ");
- ASSERT_EQ(1, scanf("%d", &pd));
- res = hardware->SetPlayoutDevice(pd);
- VALIDATE;
- printf("Select recording device: ");
- ASSERT_EQ(1, scanf("%d", &rd));
- printf("Setting sound devices \n");
- res = hardware->SetRecordingDevice(rd);
- VALIDATE;
-
-#endif // WEBRTC_LINUX
- res = codec->SetVADStatus(0, VAD);
- VALIDATE;
-
- res = apm->SetAgcStatus(AGC);
- VALIDATE;
-
- res = apm->SetEcStatus(AEC);
- VALIDATE;
-
- res = apm->SetNsStatus(NS);
- VALIDATE;
-
-#ifdef DEBUG
- i = 1;
-#else
- printf("\n1. Send, listen and playout \n");
- printf("2. Send only \n");
- printf("3. Listen and playout only \n");
- printf("Select transfer mode: ");
- ASSERT_EQ(1, scanf("%i", &i));
-#endif
- const bool send = !(3 == i);
- const bool receive = !(2 == i);
-
- if (receive) {
-#ifndef EXTERNAL_TRANSPORT
- printf("Start Listen \n");
- res = base1->StartReceive(chan);
- VALIDATE;
-#endif
-
- printf("Start Playout \n");
- res = base1->StartPlayout(chan);
- VALIDATE;
- }
-
- if (send) {
- printf("Start Send \n");
- res = base1->StartSend(chan);
- VALIDATE;
- }
-
- printf("Getting mic volume \n");
- unsigned int vol = 999;
- res = volume->GetMicVolume(vol);
- VALIDATE;
- if ((vol > 255) || (vol < 1)) {
- printf("\n****ERROR in GetMicVolume");
- }
-
- int forever = 1;
- while (forever) {
- printf("\nActions\n");
-
- printf("Codec Changes\n");
- for (i = 0; i < codec->NumOfCodecs(); i++) {
- res = codec->GetCodec(i, cinst);
- VALIDATE;
- if (strncmp(cinst.plname, "ISAC", 4) == 0 && cinst.plfreq
- == 32000) {
- printf("\t%i. ISAC-swb pltype:%i plfreq:%i\n", i,
- cinst.pltype, cinst.plfreq);
- }
- else {
- printf("\t%i. %s pltype:%i plfreq:%i\n", i, cinst.plname,
- cinst.pltype, cinst.plfreq);
- }
- }
- printf("Other\n");
- const int noCodecs = i - 1;
- printf("\t%i. Toggle VAD\n", i);
- i++;
- printf("\t%i. Toggle AGC\n", i);
- i++;
- printf("\t%i. Toggle NS\n", i);
- i++;
- printf("\t%i. Toggle EC\n", i);
- i++;
- printf("\t%i. Select AEC\n", i);
- i++;
- printf("\t%i. Select AECM\n", i);
- i++;
- printf("\t%i. Get speaker volume\n", i);
- i++;
- printf("\t%i. Set speaker volume\n", i);
- i++;
- printf("\t%i. Get microphone volume\n", i);
- i++;
- printf("\t%i. Set microphone volume\n", i);
- i++;
- printf("\t%i. Play local file (audio_long16.pcm) \n", i);
- i++;
- printf("\t%i. Change playout device \n", i);
- i++;
- printf("\t%i. Change recording device \n", i);
- i++;
- printf("\t%i. Toggle receive-side AGC \n", i);
- i++;
- printf("\t%i. Toggle receive-side NS \n", i);
- i++;
- printf("\t%i. AGC status \n", i);
- i++;
- printf("\t%i. Toggle microphone mute \n", i);
- i++;
- printf("\t%i. Toggle on hold status \n", i);
- i++;
- printf("\t%i. Get last error code \n", i);
- i++;
- printf("\t%i. Toggle typing detection (for Mac/Windows only) \n", i);
- i++;
- printf("\t%i. Record a PCM file \n", i);
- i++;
- printf("\t%i. Play a previously recorded PCM file locally \n", i);
- i++;
- printf("\t%i. Play a previously recorded PCM file as microphone \n", i);
- i++;
- printf("\t%i. Add an additional file-playing channel \n", i);
- i++;
- printf("\t%i. Remove a file-playing channel \n", i);
- i++;
-
- printf("Select action or %i to stop the call: ", i);
- ASSERT_EQ(1, scanf("%i", &codecinput));
-
- if (codecinput < codec->NumOfCodecs()) {
- res = codec->GetCodec(codecinput, cinst);
- VALIDATE;
-
- printf("Set primary codec\n");
- res = codec->SetSendCodec(chan, cinst);
- VALIDATE;
- }
- else if (codecinput == (noCodecs + 1)) {
- VAD = !VAD;
- res = codec->SetVADStatus(0, VAD);
- VALIDATE;
- if (VAD)
- printf("\n VAD is now on! \n");
- else
- printf("\n VAD is now off! \n");
- }
- else if (codecinput == (noCodecs + 2)) {
- AGC = !AGC;
- res = apm->SetAgcStatus(AGC);
- VALIDATE;
- if (AGC)
- printf("\n AGC is now on! \n");
- else
- printf("\n AGC is now off! \n");
- }
- else if (codecinput == (noCodecs + 3)) {
- NS = !NS;
- res = apm->SetNsStatus(NS);
- VALIDATE;
- if (NS)
- printf("\n NS is now on! \n");
- else
- printf("\n NS is now off! \n");
- }
- else if (codecinput == (noCodecs + 4)) {
- AEC = !AEC;
- res = apm->SetEcStatus(AEC, kEcUnchanged);
- VALIDATE;
- if (AEC)
- printf("\n Echo control is now on! \n");
- else
- printf("\n Echo control is now off! \n");
- }
- else if (codecinput == (noCodecs + 5)) {
- res = apm->SetEcStatus(AEC, kEcAec);
- VALIDATE;
- printf("\n AEC selected! \n");
- if (AEC)
- printf(" (Echo control is on)\n");
- else
- printf(" (Echo control is off)\n");
- }
- else if (codecinput == (noCodecs + 6)) {
- res = apm->SetEcStatus(AEC, kEcAecm);
- VALIDATE;
- printf("\n AECM selected! \n");
- if (AEC)
- printf(" (Echo control is on)\n");
- else
- printf(" (Echo control is off)\n");
- }
- else if (codecinput == (noCodecs + 7)) {
- unsigned vol(0);
- res = volume->GetSpeakerVolume(vol);
- VALIDATE;
- printf("\n Speaker Volume is %d \n", vol);
- }
- else if (codecinput == (noCodecs + 8)) {
- printf("Level: ");
- ASSERT_EQ(1, scanf("%i", &i));
- res = volume->SetSpeakerVolume(i);
- VALIDATE;
- }
- else if (codecinput == (noCodecs + 9)) {
- unsigned vol(0);
- res = volume->GetMicVolume(vol);
- VALIDATE;
- printf("\n Microphone Volume is %d \n", vol);
- }
- else if (codecinput == (noCodecs + 10)) {
- printf("Level: ");
- ASSERT_EQ(1, scanf("%i", &i));
- res = volume->SetMicVolume(i);
- VALIDATE;
- }
- else if (codecinput == (noCodecs + 11)) {
- res = file->StartPlayingFileLocally(0, audio_filename.c_str());
- VALIDATE;
- }
- else if (codecinput == (noCodecs + 12)) {
- // change the playout device with current call
- int num_pd(-1);
- res = hardware->GetNumOfPlayoutDevices(num_pd);
- VALIDATE;
-
- char dn[128] = { 0 };
- char guid[128] = { 0 };
-
- printf("\nPlayout devices (%d): \n", num_pd);
- for (j = 0; j < num_pd; ++j) {
- res = hardware->GetPlayoutDeviceName(j, dn, guid);
- VALIDATE;
- printf(" %d: %s \n", j, dn);
- }
- printf("Select playout device: ");
- ASSERT_EQ(1, scanf("%d", &num_pd));
- // Will use plughw for hardware devices
- res = hardware->SetPlayoutDevice(num_pd);
- VALIDATE;
- }
- else if (codecinput == (noCodecs + 13)) {
- // change the recording device with current call
- int num_rd(-1);
-
- res = hardware->GetNumOfRecordingDevices(num_rd);
- VALIDATE;
-
- char dn[128] = { 0 };
- char guid[128] = { 0 };
-
- printf("Recording devices (%d): \n", num_rd);
- for (j = 0; j < num_rd; ++j) {
- res = hardware->GetRecordingDeviceName(j, dn, guid);
- VALIDATE;
- printf(" %d: %s \n", j, dn);
- }
-
- printf("Select recording device: ");
- ASSERT_EQ(1, scanf("%d", &num_rd));
- printf("Setting sound devices \n");
- // Will use plughw for hardware devices
- res = hardware->SetRecordingDevice(num_rd);
- VALIDATE;
- }
- else if (codecinput == (noCodecs + 14)) {
- // Remote AGC
- AGC1 = !AGC1;
- res = apm->SetRxAgcStatus(chan, AGC1);
- VALIDATE;
- if (AGC1)
- printf("\n Receive-side AGC is now on! \n");
- else
- printf("\n Receive-side AGC is now off! \n");
- }
- else if (codecinput == (noCodecs + 15)) {
- // Remote NS
- NS1 = !NS1;
- res = apm->SetRxNsStatus(chan, NS);
- VALIDATE;
- if (NS1)
- printf("\n Receive-side NS is now on! \n");
- else
- printf("\n Receive-side NS is now off! \n");
- }
- else if (codecinput == (noCodecs + 16)) {
- AgcModes agcmode;
- bool enable;
- res = apm->GetAgcStatus(enable, agcmode);
- VALIDATE
- printf("\n AGC enable is %d, mode is %d \n", enable, agcmode);
- }
- else if (codecinput == (noCodecs + 17)) {
- // Toggle Mute on Microphone
- res = volume->GetInputMute(chan, muted);
- VALIDATE;
- muted = !muted;
- res = volume->SetInputMute(chan, muted);
- VALIDATE;
- if (muted)
- printf("\n Microphone is now on mute! \n");
- else
- printf("\n Microphone is no longer on mute! \n");
-
- }
- else if (codecinput == (noCodecs + 18)) {
- // Toggle the call on hold
- OnHoldModes mode;
- res = base1->GetOnHoldStatus(chan, on_hold, mode);
- VALIDATE;
- on_hold = !on_hold;
- mode = kHoldSendAndPlay;
- res = base1->SetOnHoldStatus(chan, on_hold, mode);
- VALIDATE;
- if (on_hold)
- printf("\n Call now on hold! \n");
- else
- printf("\n Call now not on hold! \n");
- }
-
- else if (codecinput == (noCodecs + 19)) {
- // Get the last error code and print to screen
- int err_code = 0;
- err_code = base1->LastError();
- if (err_code != -1)
- printf("\n The last error code was %i.\n", err_code);
- }
- else if (codecinput == (noCodecs + 20)) {
- typing_detection= !typing_detection;
- res = apm->SetTypingDetectionStatus(typing_detection);
- VALIDATE;
- if (typing_detection)
- printf("\n Typing detection is now on!\n");
- else
- printf("\n Typing detection is now off!\n");
- }
- else if (codecinput == (noCodecs + 21)) {
- int stop_record = 1;
- int file_source = 1;
- printf("\n Select source of recorded file. ");
- printf("\n 1. Record from microphone to file ");
- printf("\n 2. Record from playout to file ");
- printf("\n Enter your selection: \n");
- ASSERT_EQ(1, scanf("%i", &file_source));
- if (file_source == 1) {
- printf("\n Start recording microphone as %s \n",
- mic_filename.c_str());
- res = file->StartRecordingMicrophone(mic_filename.c_str());
- VALIDATE;
- }
- else {
- printf("\n Start recording playout as %s \n", play_filename.c_str());
- res = file->StartRecordingPlayout(chan, play_filename.c_str());
- VALIDATE;
- }
- while (stop_record != 0) {
- printf("\n Type 0 to stop recording file \n");
- ASSERT_EQ(1, scanf("%i", &stop_record));
- }
- if (file_source == 1) {
- res = file->StopRecordingMicrophone();
- VALIDATE;
- }
- else {
- res = file->StopRecordingPlayout(chan);
- VALIDATE;
- }
- printf("\n File finished recording \n");
- }
- else if (codecinput == (noCodecs + 22)) {
- int file_type = 1;
- int stop_play = 1;
- printf("\n Select a file to play locally in a loop.");
- printf("\n 1. Play %s", mic_filename.c_str());
- printf("\n 2. Play %s", play_filename.c_str());
- printf("\n Enter your selection\n");
- ASSERT_EQ(1, scanf("%i", &file_type));
- if (file_type == 1) {
- printf("\n Start playing %s locally in a loop\n",
- mic_filename.c_str());
- res = file->StartPlayingFileLocally(chan, mic_filename.c_str(), true);
- VALIDATE;
- }
- else {
- printf("\n Start playing %s locally in a loop\n",
- play_filename.c_str());
- res = file->StartPlayingFileLocally(chan, play_filename.c_str(),
- true);
- VALIDATE;
- }
- while (stop_play != 0) {
- printf("\n Type 0 to stop playing file\n");
- ASSERT_EQ(1, scanf("%i", &stop_play));
- }
- res = file->StopPlayingFileLocally(chan);
- VALIDATE;
- }
- else if (codecinput == (noCodecs + 23)) {
- int file_type = 1;
- int stop_play = 1;
- printf("\n Select a file to play as microphone in a loop.");
- printf("\n 1. Play %s", mic_filename.c_str());
- printf("\n 2. Play %s", play_filename.c_str());
- printf("\n Enter your selection\n");
- ASSERT_EQ(1, scanf("%i", &file_type));
- if (file_type == 1) {
- printf("\n Start playing %s as mic in a loop\n",
- mic_filename.c_str());
- res = file->StartPlayingFileAsMicrophone(chan, mic_filename.c_str(),
- true);
- VALIDATE;
- }
- else {
- printf("\n Start playing %s as mic in a loop\n",
- play_filename.c_str());
- res = file->StartPlayingFileAsMicrophone(chan, play_filename.c_str(),
- true);
- VALIDATE;
- }
- while (stop_play != 0) {
- printf("\n Type 0 to stop playing file\n");
- ASSERT_EQ(1, scanf("%i", &stop_play));
- }
- res = file->StopPlayingFileAsMicrophone(chan);
- VALIDATE;
- }
- else if (codecinput == (noCodecs + 24)) {
- if (channel_index < kMaxNumChannels) {
- res = base1->StartReceive(channels[channel_index]);
- VALIDATE;
- res = base1->StartPlayout(channels[channel_index]);
- VALIDATE;
- res = base1->StartSend(channels[channel_index]);
- VALIDATE;
- res = file->StartPlayingFileAsMicrophone(channels[channel_index],
- audio_filename.c_str(),
- true,
- false);
- VALIDATE;
- channel_index++;
- printf("Using %d additional channels\n", channel_index);
- } else {
- printf("Max number of channels reached\n");
- }
- }
- else if (codecinput == (noCodecs + 25)) {
- if (channel_index > 0) {
- channel_index--;
- res = file->StopPlayingFileAsMicrophone(channels[channel_index]);
- VALIDATE;
- res = base1->StopSend(channels[channel_index]);
- VALIDATE;
- res = base1->StopPlayout(channels[channel_index]);
- VALIDATE;
- res = base1->StopReceive(channels[channel_index]);
- VALIDATE;
- printf("Using %d additional channels\n", channel_index);
- } else {
- printf("All additional channels stopped\n");
- }
- }
- else
- break;
- }
-
- if (send) {
- printf("Stop Send \n");
- res = base1->StopSend(chan);
- VALIDATE;
- }
-
- if (receive) {
- printf("Stop Playout \n");
- res = base1->StopPlayout(chan);
- VALIDATE;
-
-#ifndef EXTERNAL_TRANSPORT
- printf("Stop Listen \n");
- res = base1->StopReceive(chan);
- VALIDATE;
-#endif
- }
-
- while (channel_index > 0) {
- --channel_index;
- res = file->StopPlayingFileAsMicrophone(channels[channel_index]);
- VALIDATE;
- res = base1->StopSend(channels[channel_index]);
- VALIDATE;
- res = base1->StopPlayout(channels[channel_index]);
- VALIDATE;
- res = base1->StopReceive(channels[channel_index]);
- VALIDATE;
- }
-
- printf("\n1. New call \n");
- printf("2. Quit \n");
- printf("Select action: ");
- ASSERT_EQ(1, scanf("%i", &i));
- newcall = (1 == i);
- // Call loop
- }
-
- printf("Delete channels \n");
- res = base1->DeleteChannel(chan);
- VALIDATE;
-
- for (i = 0; i < kMaxNumChannels; ++i) {
- channels[i] = base1->DeleteChannel(channels[i]);
- VALIDATE;
- }
-}
diff --git a/trunk/src/voice_engine/main/test/voice_engine_tests.gypi b/trunk/src/voice_engine/main/test/voice_engine_tests.gypi
deleted file mode 100644
index ad78912..0000000
--- a/trunk/src/voice_engine/main/test/voice_engine_tests.gypi
+++ /dev/null
@@ -1,140 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'targets': [
- # Auto test - command line test for all platforms
- {
- 'target_name': 'voe_auto_test',
- 'type': 'executable',
- 'dependencies': [
- 'voice_engine_core',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- '<(webrtc_root)/../test/test.gyp:test_support',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../testing/gmock.gyp:gmock',
- ],
- 'include_dirs': [
- 'auto_test',
- 'auto_test/fixtures',
- '<(webrtc_root)/modules/interface',
- # TODO(phoglund): We only depend on voice_engine_defines.h here -
- # move that file to interface and then remove this dependency.
- '<(webrtc_root)/voice_engine/main/source',
- '<(webrtc_root)/modules/audio_device/main/interface',
- ],
- 'sources': [
- 'auto_test/automated_mode.cc',
- 'auto_test/fixtures/after_initialization_fixture.cc',
- 'auto_test/fixtures/after_initialization_fixture.h',
- 'auto_test/fixtures/after_streaming_fixture.cc',
- 'auto_test/fixtures/after_streaming_fixture.h',
- 'auto_test/fixtures/before_initialization_fixture.cc',
- 'auto_test/fixtures/before_initialization_fixture.h',
- 'auto_test/standard/audio_processing_test.cc',
- 'auto_test/standard/call_report_test.cc',
- 'auto_test/standard/codec_before_streaming_test.cc',
- 'auto_test/standard/codec_test.cc',
- 'auto_test/standard/dtmf_test.cc',
- 'auto_test/standard/file_test.cc',
- 'auto_test/standard/hardware_before_initializing_test.cc',
- 'auto_test/standard/hardware_before_streaming_test.cc',
- 'auto_test/standard/hardware_test.cc',
- 'auto_test/standard/manual_hold_test.cc',
- 'auto_test/standard/neteq_test.cc',
- 'auto_test/standard/network_before_streaming_test.cc',
- 'auto_test/standard/network_test.cc',
- 'auto_test/standard/rtp_rtcp_before_streaming_test.cc',
- 'auto_test/standard/rtp_rtcp_test.cc',
- 'auto_test/standard/voe_base_misc_test.cc',
- 'auto_test/standard/volume_test.cc',
- 'auto_test/resource_manager.cc',
- 'auto_test/voe_cpu_test.cc',
- 'auto_test/voe_cpu_test.h',
- 'auto_test/voe_extended_test.cc',
- 'auto_test/voe_extended_test.h',
- 'auto_test/voe_standard_test.cc',
- 'auto_test/voe_standard_test.h',
- 'auto_test/voe_stress_test.cc',
- 'auto_test/voe_stress_test.h',
- 'auto_test/voe_test_defines.h',
- 'auto_test/voe_test_interface.h',
- 'auto_test/voe_unit_test.cc',
- 'auto_test/voe_unit_test.h',
- ],
- },
- {
- # command line test that should work on linux/mac/win
- 'target_name': 'voe_cmd_test',
- 'type': 'executable',
- 'dependencies': [
- '<(webrtc_root)/../test/test.gyp:test_support',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- 'voice_engine_core',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'sources': [
- 'cmd_test/voe_cmd_test.cc',
- ],
- },
- ],
- 'conditions': [
- ['OS=="win"', {
- 'targets': [
- # WinTest - GUI test for Windows
- {
- 'target_name': 'voe_ui_win_test',
- 'type': 'executable',
- 'dependencies': [
- 'voice_engine_core',
- '<(webrtc_root)/system_wrappers/source/system_wrappers.gyp:system_wrappers',
- ],
- 'include_dirs': [
- 'win_test',
- ],
- 'sources': [
- 'win_test/Resource.h',
- 'win_test/WinTest.cc',
- 'win_test/WinTest.h',
- 'win_test/WinTest.rc',
- 'win_test/WinTestDlg.cc',
- 'win_test/WinTestDlg.h',
- 'win_test/res/WinTest.ico',
- 'win_test/res/WinTest.rc2',
- 'win_test/stdafx.cc',
- 'win_test/stdafx.h',
- ],
- 'configurations': {
- 'Common_Base': {
- 'msvs_configuration_attributes': {
- 'conditions': [
- ['component=="shared_library"', {
- 'UseOfMFC': '2', # Shared DLL
- },{
- 'UseOfMFC': '1', # Static
- }],
- ],
- },
- },
- },
- 'msvs_settings': {
- 'VCLinkerTool': {
- 'SubSystem': '2', # Windows
- },
- },
- },
- ],
- }],
- ],
-}
-
-# Local Variables:
-# tab-width:2
-# indent-tabs-mode:nil
-# End:
-# vim: set expandtab tabstop=2 shiftwidth=2:
diff --git a/trunk/src/voice_engine/main/test/win_test/Resource.h b/trunk/src/voice_engine/main/test/win_test/Resource.h
deleted file mode 100644
index 5ae9c5f..0000000
--- a/trunk/src/voice_engine/main/test/win_test/Resource.h
+++ /dev/null
@@ -1,241 +0,0 @@
-//{{NO_DEPENDENCIES}}
-// Microsoft Visual C++ generated include file.
-// Used by WinTest.rc
-//
-#define IDM_ABOUTBOX 0x0010
-#define IDD_ABOUTBOX 100
-#define IDS_ABOUTBOX 101
-#define IDD_WINTEST_DIALOG 102
-#define IDR_MAINFRAME 128
-#define IDD_DTMF_DIALOG 129
-#define IDC_BUTTON_CREATE_1 1000
-#define IDC_BUTTON_DELETE_1 1001
-#define IDC_EDIT_1 1002
-#define IDC_BUTTON_CREATE_2 1003
-#define IDC_BUTTON_DELETE_2 1004
-#define IDC_EDIT_2 1005
-#define IDC_EDIT_MESSAGE 1006
-#define IDC_BUTTON_START_LISTEN_1 1007
-#define IDC_COMBO_IP_1 1008
-#define IDC_EDIT_TX_PORT_1 1009
-#define IDC_EDIT_RX_PORT_1 1010
-#define IDC_COMBO_CODEC_1 1011
-#define IDC_BUTTON_STOP_LISTEN_1 1012
-#define IDC_STATIC_LISTEN 1013
-#define IDC_BUTTON_START_PLAYOUT_1 1014
-#define IDC_BUTTON_STOP_PLAYOUT_1 1015
-#define IDC_STATIC_PLAY 1016
-#define IDC_BUTTON_START_SEND_1 1017
-#define IDC_BUTTON_STOP_SEND_1 1018
-#define IDC_STATIC_SEND 1019
-#define IDC_COMBO_IP_2 1020
-#define IDC_STATIC_IP 1021
-#define IDC_STATIC_PORTS 1022
-#define IDC_STATIC_CODEC 1023
-#define IDC_STATIC_CHANNEL 1024
-#define IDC_STATIC_ID 1025
-#define IDC_EDIT_TX_PORT_2 1026
-#define IDC_EDIT_RX_PORT_2 1027
-#define IDC_COMBO_CODEC_2 1028
-#define IDC_BUTTON_START_LISTEN_2 1029
-#define IDC_BUTTON_STOP_LISTEN_2 1030
-#define IDC_BUTTON_START_PLAYOUT_2 1031
-#define IDC_BUTTON_STOP_PLAYOUT_2 1032
-#define IDC_BUTTON_START_SEND_2 1033
-#define IDC_BUTTON_STOP_SEND_2 1034
-#define IDC_BUTTON_START_SEND_3 1035
-#define IDC_BUTTON_TEST_1_1 1035
-#define IDC_BUTTON_TEST_1 1035
-#define IDC_EDIT_RESULT 1036
-#define IDC_EDIT_N_FAILS 1037
-#define IDC_STATIC_ERROR 1038
-#define IDC_EDIT_LAST_ERROR 1039
-#define IDC_STATIC_LAST_ERROR 1040
-#define IDC_STATIC_PLAY_FILE 1041
-#define IDC_STATIC_EXTERNAL 1042
-#define IDC_CHECK_EXT_TRANS_1 1043
-#define IDC_CHECK2 1044
-#define IDC_CHECK_PLAY_FILE_IN_1 1044
-#define IDC_CHECK_PLAY_FILE_OUT_1 1045
-#define IDC_CHECK_PLAY_FILE_IN_2 1046
-#define IDC_CHECK_PLAY_FILE_OUT_2 1047
-#define IDC_CHECK_EXT_TRANS_2 1048
-#define IDC_STATIC_ALL_CHANNELS 1049
-#define IDC_CHECK_PLAY_FILE_IN 1050
-#define IDC_CHECK_PLAY_FILE_OUT 1051
-#define IDC_CHECK_EXT_MEDIA_IN_1 1051
-#define IDC_COMBO_REC_DEVICE 1052
-#define IDC_STATIC_REC_DEVICE 1053
-#define IDC_COMBO_PLAY_DEVICE2 1054
-#define IDC_COMBO_PLAY_DEVICE 1054
-#define IDC_STATIC_PLAY_DEVICE 1055
-#define IDC_CHECK_EXT_MEDIA_PLAY_1 1056
-#define IDC_CHECK_EXT_MEDIA_OUT_1 1056
-#define IDC_STATIC_PLAY_FILE2 1057
-#define IDC_SLIDER_INPUT_VOLUME 1058
-#define IDC_STATIC_MIC_VOLUME 1059
-#define IDC_SLIDER_OUTPUT_VOLUME 1060
-#define IDC_STATIC_SPK_VOLUME2 1061
-#define IDC_STATIC_SPK_VOLUME 1061
-#define IDC_CHECK_PLAY_FILE_IN2 1062
-#define IDC_CHECK_AGC 1062
-#define IDC_STATIC_MIC_VOLUME2 1063
-#define IDC_STATIC_AUDIO_LEVEL_IN 1063
-#define IDC_PROGRESS_AUDIO_LEVEL_IN 1064
-#define IDC_CHECK_AGC2 1065
-#define IDC_CHECK_NS 1065
-#define IDC_BUTTON_1 1065
-#define IDC_CHECK_VAD 1066
-#define IDC_CHECK_EXT_MEDIA_IN_2 1066
-#define IDC_BUTTON_2 1066
-#define IDC_CHECK_VAD2 1067
-#define IDC_CHECK_EC 1067
-#define IDC_BUTTON_3 1067
-#define IDC_CHECK_VAD_1 1068
-#define IDC_BUTTON_4 1068
-#define IDC_CHECK_VAD_2 1069
-#define IDC_CHECK_EXT_MEDIA_OUT_2 1069
-#define IDC_BUTTON_5 1069
-#define IDC_CHECK_VAD_3 1070
-#define IDC_BUTTON_6 1070
-#define IDC_CHECK_MUTE_IN 1071
-#define IDC_BUTTON_7 1071
-#define IDC_CHECK_MUTE_IN_1 1072
-#define IDC_BUTTON_8 1072
-#define IDC_CHECK_MUTE_IN_2 1073
-#define IDC_BUTTON_9 1073
-#define IDC_CHECK_SRTP_TX_1 1074
-#define IDC_BUTTON_10 1074
-#define IDC_CHECK_SRTP_RX_1 1075
-#define IDC_BUTTON_11 1075
-#define IDC_STATIC_PLAY_FILE3 1076
-#define IDC_STATIC_SRTP 1076
-#define IDC_BUTTON_12 1076
-#define IDC_CHECK_SRTP_TX_2 1077
-#define IDC_BUTTON_13 1077
-#define IDC_CHECK_SRTP_RX_2 1078
-#define IDC_BUTTON_14 1078
-#define IDC_CHECK_EXT_ENCRYPTION_1 1079
-#define IDC_BUTTON_15 1079
-#define IDC_STATIC_PLAY_FILE4 1080
-#define IDC_BUTTON_16 1080
-#define IDC_CHECK_EXT_ENCRYPTION_2 1081
-#define IDC_BUTTON_17 1081
-#define IDC_BUTTON_DTMF_1 1082
-#define IDC_BUTTON_18 1082
-#define IDC_EDIT_DTMF_EVENT 1083
-#define IDC_CHECK_REC_ 1083
-#define IDC_CHECK_REC_MIC 1083
-#define IDC_STATIC_DTMF_EVENT 1084
-#define IDC_BUTTON_DTMF_2 1084
-#define IDC_STATIC_GROUP_DTMF 1085
-#define IDC_CHECK_CONFERENCE_1 1085
-#define IDC_BUTTON_19 1086
-#define IDC_CHECK_CONFERENCE_2 1086
-#define IDC_BUTTON_20 1087
-#define IDC_CHECK_ON_HOLD_1 1087
-#define IDC_BUTTON_21 1088
-#define IDC_CHECK_ON_HOLD_2 1088
-#define IDC_BUTTON_22 1089
-#define IDC_CHECK_DTMF_PLAYOUT_RX 1089
-#define IDC_CHECK_EXT_MEDIA_IN 1089
-#define IDC_STATIC_PLAYOUT_RX 1090
-#define IDC_EDIT_GET_OUTPUT 1090
-#define IDC_CHECK_DTMF_PLAY_TONE 1091
-#define IDC_STATIC_LAST_ERROR2 1091
-#define IDC_STATIC_GET 1091
-#define IDC_STATIC_PLAY_TONE 1092
-#define IDC_CHECK_EXT_MEDIA_OUT 1092
-#define IDC_CHECK_START_STOP_MODE 1093
-#define IDC_BUTTON_SET_TX_TELEPHONE_PT 1093
-#define IDC_PROGRESS_AUDIO_LEVEL_IN2 1093
-#define IDC_PROGRESS_AUDIO_LEVEL_OUT 1093
-#define IDC_EDIT_EVENT_LENGTH 1094
-#define IDC_EDIT_RX_PORT_3 1094
-#define IDC_EDIT_DELAY_ESTIMATE_1 1094
-#define IDC_STATIC_EVENT_LENGTH 1095
-#define IDC_EDIT_PLAYOUT_BUFFER_SIZE 1095
-#define IDC_STATIC_START_STOP_MODE 1096
-#define IDC_EDIT_EVENT_RX_PT 1096
-#define IDC_CHECK_DELAY_ESTIMATE_1 1096
-#define IDC_EDIT_EVENT_ATTENUATION 1097
-#define IDC_CHECK_AGC_1 1097
-#define IDC_CHECK_EVENT_INBAND 1098
-#define IDC_CHECK_NS_1 1098
-#define IDC_STATIC_EVENT_ATTENUATION 1099
-#define IDC_STATIC_SRTP2 1099
-#define IDC_STATIC_RX_VQE 1099
-#define IDC_EDIT_EVENT_TX_PT 1100
-#define IDC_CHECK_REC_MIC2 1100
-#define IDC_CHECK_REC_CALL 1100
-#define IDC_CHECK_DTMF_FEEDBACK 1101
-#define IDC_CHECK_REC_CALL2 1101
-#define IDC_CHECK_TYPING_DETECTION 1101
-#define IDC_CHECK_START_STOP_MODE2 1102
-#define IDC_CHECK_DIRECT_FEEDBACK 1102
-#define IDC_CHECK_FEC 1102
-#define IDC_BUTTON_SET_RX_TELEPHONE_PT_TYPE 1103
-#define IDC_BUTTON_SET_RX_TELEPHONE_PT 1103
-#define IDC_BUTTON_CLEAR_ERROR_CALLBACK 1103
-#define IDC_EDIT_EVENT_CODE 1104
-#define IDC_STATIC_DIRECT_FEEDBACK 1105
-#define IDC_RADIO_SINGLE 1106
-#define IDC_RADIO_MULTI 1107
-#define IDC_RADIO_START_STOP 1108
-#define IDC_STATIC_MODE 1109
-#define IDC_STATIC_EVENT_RX_PT 1110
-#define IDC_STATIC_EVENT_TX_PT 1111
-#define IDC_STATIC_PT 1112
-#define IDC_BUTTON_SEND_TELEPHONE_EVENT 1113
-#define IDC_STATIC_EVENT_CODE 1114
-#define IDC_CHECK_EVENT_DETECTION 1115
-#define IDC_CHECK_DETECT_INBAND 1116
-#define IDC_CHECK_DETECT_OUT_OF_BAND 1117
-#define IDC_STATIC_INBAND_DETECTION 1118
-#define IDC_STATIC_OUT_OF_BAND_DETECTION 1119
-#define IDC_STATIC_EVENT_DETECTION 1120
-#define IDC_STATIC_TELEPHONE_EVENTS 1121
-#define IDC_EDIT_EVENT_CODE2 1122
-#define IDC_EDIT_ON_EVENT 1122
-#define IDC_EDIT_ON_EVENT_OUT_OF_BAND 1122
-#define IDC_STATIC_ON_EVENT 1123
-#define IDC_EDIT_ON_EVENT_INBAND 1123
-#define IDC_STATIC_EVEN 1124
-#define IDC_STATIC_LINE 1125
-#define IDC_LIST_CODEC_1 1128
-#define IDC_EDIT2 1129
-#define IDC_EDIT_CODEC_1 1129
-#define IDC_STATIC_PANNING 1131
-#define IDC_SLIDER_PAN_LEFT 1132
-#define IDC_SLIDER_PAN_RIGHT 1133
-#define IDC_STATIC_LEFT 1134
-#define IDC_STATIC_LEFT2 1135
-#define IDC_STATIC_RIGHT 1135
-#define IDC_BUTTON_VERSION 1136
-#define IDC_STATIC_PLAYOUT_BUFFER 1137
-#define IDC_CHECK_RXVAD 1138
-#define IDC_EDIT1 1139
-#define IDC_EDIT_RXVAD 1139
-#define IDC_STATIC_RX_PORT 1140
-#define IDC_STATIC_RX_PORT2 1141
-#define IDC_EDIT3 1142
-#define IDC_EDIT_AUDIO_LAYER 1142
-#define IDC_EDIT_AUDIO_LAYER2 1143
-#define IDC_EDIT_CPU_LOAD 1143
-#define IDC_STATIC_ERROR_CALLBACK 1144
-#define IDC_EDIT_ERROR_CALLBACK 1145
-#define IDC_EDIT_RX_CODEC_1 1146
-#define IDC_STATIC_BYTES_SENT_TEXT 1147
-#define IDC_EDIT_RTCP_STAT 1147
-#define IDC_EDIT_RTCP_STAT_1 1147
-
-// Next default values for new objects
-//
-#ifdef APSTUDIO_INVOKED
-#ifndef APSTUDIO_READONLY_SYMBOLS
-#define _APS_NEXT_RESOURCE_VALUE 130
-#define _APS_NEXT_COMMAND_VALUE 32771
-#define _APS_NEXT_CONTROL_VALUE 1148
-#define _APS_NEXT_SYMED_VALUE 101
-#endif
-#endif
diff --git a/trunk/src/voice_engine/main/test/win_test/WinTest.aps b/trunk/src/voice_engine/main/test/win_test/WinTest.aps
deleted file mode 100644
index 499db5f..0000000
--- a/trunk/src/voice_engine/main/test/win_test/WinTest.aps
+++ /dev/null
Binary files differ
diff --git a/trunk/src/voice_engine/main/test/win_test/WinTest.cc b/trunk/src/voice_engine/main/test/win_test/WinTest.cc
deleted file mode 100644
index e0e0248..0000000
--- a/trunk/src/voice_engine/main/test/win_test/WinTest.cc
+++ /dev/null
@@ -1,75 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "stdafx.h"
-#include "WinTest.h"
-#include "WinTestDlg.h"
-
-#ifdef _DEBUG
-#define new DEBUG_NEW
-#endif
-
-
-// CWinTestApp
-
-BEGIN_MESSAGE_MAP(CWinTestApp, CWinApp)
- ON_COMMAND(ID_HELP, &CWinApp::OnHelp)
-END_MESSAGE_MAP()
-
-
-// CWinTestApp construction
-
-CWinTestApp::CWinTestApp()
-{
-}
-
-
-// The one and only CWinTestApp object
-
-CWinTestApp theApp;
-
-
-// CWinTestApp initialization
-
-BOOL CWinTestApp::InitInstance()
-{
- // InitCommonControlsEx() is required on Windows XP if an application
- // manifest specifies use of ComCtl32.dll version 6 or later to enable
- // visual styles. Otherwise, any window creation will fail.
- INITCOMMONCONTROLSEX InitCtrls;
- InitCtrls.dwSize = sizeof(InitCtrls);
- // Set this to include all the common control classes you want to use
- // in your application.
- InitCtrls.dwICC = ICC_WIN95_CLASSES;
- InitCommonControlsEx(&InitCtrls);
-
- CWinApp::InitInstance();
-
- // Standard initialization
- // If you are not using these features and wish to reduce the size
- // of your final executable, you should remove from the following
- // the specific initialization routines you do not need
- // Change the registry key under which our settings are stored
- SetRegistryKey(_T("Local AppWizard-Generated Applications"));
-
- CWinTestDlg dlg;
- m_pMainWnd = &dlg;
- INT_PTR nResponse = dlg.DoModal();
- if (nResponse == IDOK)
- {
- }
- else if (nResponse == IDCANCEL)
- {
- }
-
- // Since the dialog has been closed, return FALSE so that we exit the
- // application, rather than start the application's message pump.
- return FALSE;
-}
diff --git a/trunk/src/voice_engine/main/test/win_test/WinTest.h b/trunk/src/voice_engine/main/test/win_test/WinTest.h
deleted file mode 100644
index d012ce6..0000000
--- a/trunk/src/voice_engine/main/test/win_test/WinTest.h
+++ /dev/null
@@ -1,38 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#pragma once
-
-#ifndef __AFXWIN_H__
- #error "include 'stdafx.h' before including this file for PCH"
-#endif
-
-#include "resource.h" // main symbols
-
-
-// CWinTestApp:
-// See WinTest.cpp for the implementation of this class
-//
-
-class CWinTestApp : public CWinApp
-{
-public:
- CWinTestApp();
-
-// Overrides
- public:
- virtual BOOL InitInstance();
-
-// Implementation
-
- DECLARE_MESSAGE_MAP()
-};
-
-extern CWinTestApp theApp;
diff --git a/trunk/src/voice_engine/main/test/win_test/WinTest.rc b/trunk/src/voice_engine/main/test/win_test/WinTest.rc
deleted file mode 100644
index dfe503f..0000000
--- a/trunk/src/voice_engine/main/test/win_test/WinTest.rc
+++ /dev/null
@@ -1,394 +0,0 @@
-// Microsoft Visual C++ generated resource script.
-//
-#include "resource.h"
-
-#define APSTUDIO_READONLY_SYMBOLS
-/////////////////////////////////////////////////////////////////////////////
-//
-// Generated from the TEXTINCLUDE 2 resource.
-//
-#include "afxres.h"
-
-/////////////////////////////////////////////////////////////////////////////
-#undef APSTUDIO_READONLY_SYMBOLS
-
-/////////////////////////////////////////////////////////////////////////////
-// Swedish resources
-
-#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_SVE)
-#ifdef _WIN32
-LANGUAGE LANG_SWEDISH, SUBLANG_DEFAULT
-#pragma code_page(1252)
-#endif //_WIN32
-
-#ifdef APSTUDIO_INVOKED
-/////////////////////////////////////////////////////////////////////////////
-//
-// TEXTINCLUDE
-//
-
-1 TEXTINCLUDE
-BEGIN
- "resource.h\0"
-END
-
-2 TEXTINCLUDE
-BEGIN
- "#include ""afxres.h""\r\n"
- "\0"
-END
-
-3 TEXTINCLUDE
-BEGIN
- "#define _AFX_NO_SPLITTER_RESOURCES\r\n"
- "#define _AFX_NO_OLE_RESOURCES\r\n"
- "#define _AFX_NO_TRACKER_RESOURCES\r\n"
- "#define _AFX_NO_PROPERTY_RESOURCES\r\n"
- "\r\n"
- "#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_SVE)\r\n"
- "LANGUAGE 29, 1\r\n"
- "#pragma code_page(1252)\r\n"
- "#include ""res\\WinTest.rc2"" // non-Microsoft Visual C++ edited resources\r\n"
- "#include ""afxres.rc"" // Standard components\r\n"
- "#endif\r\n"
- "\0"
-END
-
-#endif // APSTUDIO_INVOKED
-
-
-/////////////////////////////////////////////////////////////////////////////
-//
-// Icon
-//
-
-// Icon with lowest ID value placed first to ensure application icon
-// remains consistent on all systems.
-IDR_MAINFRAME ICON "res\\WinTest.ico"
-
-/////////////////////////////////////////////////////////////////////////////
-//
-// Dialog
-//
-
-IDD_ABOUTBOX DIALOGEX 0, 0, 235, 55
-STYLE DS_SETFONT | DS_MODALFRAME | DS_FIXEDSYS | WS_POPUP | WS_CAPTION | WS_SYSMENU
-CAPTION "About WinTest"
-FONT 8, "MS Shell Dlg", 0, 0, 0x1
-BEGIN
- ICON IDR_MAINFRAME,IDC_STATIC,11,17,20,20
- LTEXT "WinTest Version 1.0",IDC_STATIC,40,10,119,8,SS_NOPREFIX
- LTEXT "Copyright (C) 2010",IDC_STATIC,40,25,119,8
- DEFPUSHBUTTON "OK",IDOK,178,7,50,16,WS_GROUP
-END
-
-IDD_WINTEST_DIALOG DIALOGEX 0, 0, 796, 278
-STYLE DS_SETFONT | DS_MODALFRAME | DS_FIXEDSYS | WS_MINIMIZEBOX | WS_POPUP | WS_VISIBLE | WS_CAPTION | WS_SYSMENU
-EXSTYLE WS_EX_APPWINDOW
-CAPTION "WinTest"
-FONT 8, "MS Shell Dlg", 0, 0, 0x1
-BEGIN
- PUSHBUTTON "Create",IDC_BUTTON_CREATE_1,28,24,32,14
- PUSHBUTTON "Delete",IDC_BUTTON_DELETE_1,28,40,32,14
- EDITTEXT IDC_EDIT_1,6,32,18,14,ES_AUTOHSCROLL | ES_READONLY
- PUSHBUTTON "Create",IDC_BUTTON_CREATE_2,28,72,32,14
- PUSHBUTTON "Delete",IDC_BUTTON_DELETE_2,28,88,32,14
- EDITTEXT IDC_EDIT_2,6,82,18,14,ES_AUTOHSCROLL | ES_READONLY
- EDITTEXT IDC_EDIT_MESSAGE,28,244,764,12,ES_AUTOHSCROLL
- COMBOBOX IDC_COMBO_IP_1,64,24,76,30,CBS_DROPDOWN | CBS_SORT | WS_VSCROLL | WS_TABSTOP
- EDITTEXT IDC_EDIT_TX_PORT_1,144,24,28,14,ES_AUTOHSCROLL
- EDITTEXT IDC_EDIT_RX_PORT_1,144,40,28,14,ES_AUTOHSCROLL
- COMBOBOX IDC_COMBO_CODEC_1,176,24,76,156,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
- PUSHBUTTON "Start",IDC_BUTTON_START_LISTEN_1,256,24,32,14
- PUSHBUTTON "Stop",IDC_BUTTON_STOP_LISTEN_1,256,40,32,14
- LTEXT "Receive",IDC_STATIC_LISTEN,262,8,26,8
- PUSHBUTTON "Start",IDC_BUTTON_START_PLAYOUT_1,292,24,32,14
- PUSHBUTTON "Stop",IDC_BUTTON_STOP_PLAYOUT_1,292,40,32,14
- LTEXT "Playout",IDC_STATIC_PLAY,295,8,25,8
- PUSHBUTTON "Start",IDC_BUTTON_START_SEND_1,328,24,32,14
- PUSHBUTTON "Stop",IDC_BUTTON_STOP_SEND_1,328,40,32,14
- LTEXT "Send",IDC_STATIC_SEND,335,8,17,8
- COMBOBOX IDC_COMBO_IP_2,64,72,76,30,CBS_DROPDOWN | CBS_SORT | WS_VSCROLL | WS_TABSTOP
- LTEXT "Destination IP address",IDC_STATIC_IP,64,8,73,8
- LTEXT "Ports",IDC_STATIC_PORTS,145,8,18,8
- LTEXT "Codec",IDC_STATIC_CODEC,177,8,21,8
- LTEXT "Channel",IDC_STATIC_CHANNEL,30,8,27,8
- LTEXT "ID",IDC_STATIC_ID,12,8,8,8
- EDITTEXT IDC_EDIT_TX_PORT_2,144,72,28,14,ES_AUTOHSCROLL
- EDITTEXT IDC_EDIT_RX_PORT_2,144,88,28,14,ES_AUTOHSCROLL
- COMBOBOX IDC_COMBO_CODEC_2,176,72,76,156,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
- PUSHBUTTON "Start",IDC_BUTTON_START_LISTEN_2,256,72,32,14
- PUSHBUTTON "Stop",IDC_BUTTON_STOP_LISTEN_2,256,88,32,14
- PUSHBUTTON "Start",IDC_BUTTON_START_PLAYOUT_2,292,72,32,14
- PUSHBUTTON "Stop",IDC_BUTTON_STOP_PLAYOUT_2,292,88,32,14
- PUSHBUTTON "Start",IDC_BUTTON_START_SEND_2,328,72,32,14
- PUSHBUTTON "Stop",IDC_BUTTON_STOP_SEND_2,328,88,32,14
- PUSHBUTTON "TEST 1",IDC_BUTTON_TEST_1,756,224,36,14
- LTEXT "API",IDC_STATIC,4,247,12,8
- EDITTEXT IDC_EDIT_RESULT,28,260,96,12,ES_AUTOHSCROLL
- LTEXT "Result",IDC_STATIC,3,263,21,8
- EDITTEXT IDC_EDIT_N_FAILS,156,260,30,12,ES_AUTOHSCROLL
- LTEXT "#Fails",IDC_STATIC_ERROR,132,263,20,8
- EDITTEXT IDC_EDIT_LAST_ERROR,228,260,36,12,ES_AUTOHSCROLL
- LTEXT "Last Error",IDC_STATIC_LAST_ERROR,192,262,32,8
- LTEXT "Ext. Trans.",IDC_STATIC_EXTERNAL,361,8,37,8
- CONTROL "",IDC_CHECK_EXT_TRANS_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,371,33,16,10
- CONTROL "In",IDC_CHECK_PLAY_FILE_IN_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,24,36,14,WS_EX_DLGMODALFRAME
- LTEXT "Play File",IDC_STATIC_PLAY_FILE,401,8,27,8
- CONTROL "Out",IDC_CHECK_PLAY_FILE_OUT_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,40,36,14,WS_EX_DLGMODALFRAME
- CONTROL "In",IDC_CHECK_PLAY_FILE_IN_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,72,36,14,WS_EX_DLGMODALFRAME
- CONTROL "Out",IDC_CHECK_PLAY_FILE_OUT_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,88,36,14,WS_EX_DLGMODALFRAME
- CONTROL "",IDC_CHECK_EXT_TRANS_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,371,82,16,10
- GROUPBOX "",IDC_STATIC_ALL_CHANNELS,6,107,662,113
- CONTROL "PlayFileAsMic",IDC_CHECK_PLAY_FILE_IN,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,122,60,14,WS_EX_DLGMODALFRAME
- COMBOBOX IDC_COMBO_REC_DEVICE,12,132,184,80,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
- LTEXT "Recording device",IDC_STATIC_REC_DEVICE,12,120,56,8
- COMBOBOX IDC_COMBO_PLAY_DEVICE,12,180,184,80,CBS_DROPDOWN | WS_VSCROLL | WS_TABSTOP
- LTEXT "Playout device",IDC_STATIC_PLAY_DEVICE,12,167,56,8
- CONTROL "In",IDC_CHECK_EXT_MEDIA_IN_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,436,24,36,14,WS_EX_DLGMODALFRAME
- CONTROL "Out",IDC_CHECK_EXT_MEDIA_OUT_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,436,40,36,14,WS_EX_DLGMODALFRAME
- LTEXT "Ext. Media",IDC_STATIC_PLAY_FILE2,437,8,35,8
- CONTROL "",IDC_SLIDER_INPUT_VOLUME,"msctls_trackbar32",TBS_BOTH | TBS_NOTICKS | WS_TABSTOP,196,130,72,15
- LTEXT "Microphone Volume",IDC_STATIC_MIC_VOLUME,202,120,62,8
- CONTROL "",IDC_SLIDER_OUTPUT_VOLUME,"msctls_trackbar32",TBS_BOTH | TBS_NOTICKS | WS_TABSTOP,196,179,72,15
- LTEXT "Speaker Volume",IDC_STATIC_SPK_VOLUME,202,167,52,8
- CONTROL "AGC",IDC_CHECK_AGC,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,316,122,36,14,WS_EX_DLGMODALFRAME
- CONTROL "",IDC_PROGRESS_AUDIO_LEVEL_IN,"msctls_progress32",WS_BORDER,268,135,42,6
- LTEXT "Audio Level",IDC_STATIC_AUDIO_LEVEL_IN,271,120,38,8,NOT WS_GROUP
- CONTROL "NS",IDC_CHECK_NS,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,316,142,36,14,WS_EX_DLGMODALFRAME
- CONTROL "EC",IDC_CHECK_EC,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,356,122,36,14,WS_EX_DLGMODALFRAME
- CONTROL "VAD",IDC_CHECK_VAD_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,476,24,36,14,WS_EX_DLGMODALFRAME
- CONTROL "In",IDC_CHECK_EXT_MEDIA_IN_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,436,72,36,14,WS_EX_DLGMODALFRAME
- CONTROL "Out",IDC_CHECK_EXT_MEDIA_OUT_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,436,88,36,14,WS_EX_DLGMODALFRAME
- CONTROL "VAD",IDC_CHECK_VAD_3,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,476,72,36,14,WS_EX_DLGMODALFRAME
- CONTROL "Mute",IDC_CHECK_MUTE_IN,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,356,142,36,14,WS_EX_DLGMODALFRAME
- CONTROL "Mute",IDC_CHECK_MUTE_IN_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,476,40,36,14,WS_EX_DLGMODALFRAME
- CONTROL "Mute",IDC_CHECK_MUTE_IN_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,476,88,36,14,WS_EX_DLGMODALFRAME
- CONTROL "TX",IDC_CHECK_SRTP_TX_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,516,24,36,14,WS_EX_DLGMODALFRAME
- CONTROL "RX",IDC_CHECK_SRTP_RX_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,516,40,36,14,WS_EX_DLGMODALFRAME
- LTEXT "SRTP",IDC_STATIC_SRTP,525,8,18,8
- CONTROL "TX",IDC_CHECK_SRTP_TX_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,516,72,36,14,WS_EX_DLGMODALFRAME
- CONTROL "RX",IDC_CHECK_SRTP_RX_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,516,88,36,14,WS_EX_DLGMODALFRAME
- CONTROL "",IDC_CHECK_EXT_ENCRYPTION_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,564,33,16,10
- LTEXT "Encrypt",IDC_STATIC_PLAY_FILE4,556,8,26,8
- CONTROL "",IDC_CHECK_EXT_ENCRYPTION_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,564,82,16,10
- PUSHBUTTON "DTMF>>",IDC_BUTTON_DTMF_1,584,24,36,14
- CONTROL "RecMicToFile",IDC_CHECK_REC_MIC,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,396,142,60,14,WS_EX_DLGMODALFRAME
- PUSHBUTTON "DTMF>>",IDC_BUTTON_DTMF_2,584,72,36,14
- CONTROL "Conf",IDC_CHECK_CONFERENCE_1,"Button",BS_AUTOCHECKBOX | NOT WS_VISIBLE | WS_TABSTOP,584,40,36,14,WS_EX_DLGMODALFRAME
- CONTROL "Conf",IDC_CHECK_CONFERENCE_2,"Button",BS_AUTOCHECKBOX | NOT WS_VISIBLE | WS_TABSTOP,584,88,36,14,WS_EX_DLGMODALFRAME
- CONTROL "Hold",IDC_CHECK_ON_HOLD_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,708,24,36,14,WS_EX_DLGMODALFRAME
- CONTROL "Hold",IDC_CHECK_ON_HOLD_2,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,708,72,36,14,WS_EX_DLGMODALFRAME
- EDITTEXT IDC_EDIT_GET_OUTPUT,292,260,500,12,ES_AUTOHSCROLL
- LTEXT "Get",IDC_STATIC_GET,276,262,12,8
- CONTROL "Ext. Media",IDC_CHECK_EXT_MEDIA_IN,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,460,122,52,14,WS_EX_DLGMODALFRAME
- CONTROL "Ext. Media",IDC_CHECK_EXT_MEDIA_OUT,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,460,180,52,14,WS_EX_DLGMODALFRAME
- LISTBOX IDC_LIST_CODEC_1,208,40,44,28,LBS_NOINTEGRALHEIGHT | NOT WS_BORDER | WS_VSCROLL | WS_TABSTOP,WS_EX_CLIENTEDGE
- EDITTEXT IDC_EDIT_CODEC_1,176,40,28,14,ES_AUTOHSCROLL
- CONTROL "",IDC_PROGRESS_AUDIO_LEVEL_OUT,"msctls_progress32",WS_BORDER,268,184,42,6
- LTEXT "Panning",IDC_STATIC_PANNING,328,167,26,8
- CONTROL "",IDC_SLIDER_PAN_LEFT,"msctls_trackbar32",TBS_VERT | TBS_BOTH | TBS_NOTICKS | WS_TABSTOP,328,175,12,28
- CONTROL "",IDC_SLIDER_PAN_RIGHT,"msctls_trackbar32",TBS_VERT | TBS_BOTH | TBS_NOTICKS | WS_TABSTOP,344,175,12,28
- LTEXT "L",IDC_STATIC_LEFT,332,200,8,8
- LTEXT "R",IDC_STATIC_RIGHT,347,201,8,8
- PUSHBUTTON "Version",IDC_BUTTON_VERSION,624,200,36,14
- EDITTEXT IDC_EDIT_PLAYOUT_BUFFER_SIZE,363,181,28,12,ES_CENTER | ES_AUTOHSCROLL | ES_READONLY | NOT WS_TABSTOP
- LTEXT "Buffer Size",IDC_STATIC_PLAYOUT_BUFFER,361,167,36,8
- CONTROL "Delay",IDC_CHECK_DELAY_ESTIMATE_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,624,24,36,14,WS_EX_DLGMODALFRAME
- EDITTEXT IDC_EDIT_DELAY_ESTIMATE_1,631,40,24,14,ES_CENTER | ES_AUTOHSCROLL | ES_READONLY | NOT WS_TABSTOP
- CONTROL "RxVAD",IDC_CHECK_RXVAD,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,664,24,40,14,WS_EX_DLGMODALFRAME
- EDITTEXT IDC_EDIT_RXVAD,671,40,24,14,ES_CENTER | ES_AUTOHSCROLL | ES_READONLY
- CONTROL "AGC",IDC_CHECK_AGC_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,748,24,36,14,WS_EX_DLGMODALFRAME
- CONTROL "NS",IDC_CHECK_NS_1,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,748,40,36,14,WS_EX_DLGMODALFRAME
- LTEXT "RX VQE",IDC_STATIC_RX_VQE,753,8,25,8
- CONTROL "RecordCall",IDC_CHECK_REC_CALL,"Button",BS_AUTOCHECKBOX | NOT WS_VISIBLE | WS_TABSTOP,517,156,52,14,WS_EX_DLGMODALFRAME
- LTEXT "RX",IDC_STATIC_RX_PORT,133,42,10,8
- LTEXT "RX",IDC_STATIC_RX_PORT2,133,91,10,8
- CONTROL "TypingDetect",IDC_CHECK_TYPING_DETECTION,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,572,156,60,14,WS_EX_DLGMODALFRAME
- EDITTEXT IDC_EDIT_AUDIO_LAYER,28,224,116,14,ES_AUTOHSCROLL | ES_READONLY
- EDITTEXT IDC_EDIT_CPU_LOAD,152,224,116,14,ES_AUTOHSCROLL | ES_READONLY
- CONTROL "FEC",IDC_CHECK_FEC,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,176,55,28,14,WS_EX_DLGMODALFRAME
- LTEXT "=> Callbacks",IDC_STATIC_ERROR_CALLBACK,283,226,43,8
- EDITTEXT IDC_EDIT_ERROR_CALLBACK,328,224,312,14,ES_AUTOHSCROLL
- PUSHBUTTON "Clear",IDC_BUTTON_CLEAR_ERROR_CALLBACK,644,224,24,14
- EDITTEXT IDC_EDIT_RX_CODEC_1,256,56,216,12,ES_AUTOHSCROLL | ES_READONLY
- EDITTEXT IDC_EDIT_RTCP_STAT_1,476,56,316,12,ES_AUTOHSCROLL | ES_READONLY
-END
-
-IDD_DTMF_DIALOG DIALOGEX 0, 0, 316, 212
-STYLE DS_SETFONT | DS_MODALFRAME | DS_FIXEDSYS | WS_POPUP | WS_CAPTION | WS_SYSMENU
-CAPTION "Telehone Events"
-FONT 8, "MS Shell Dlg", 400, 0, 0x1
-BEGIN
- DEFPUSHBUTTON "OK",IDOK,260,192,50,14
- PUSHBUTTON "1",IDC_BUTTON_1,16,20,16,14
- PUSHBUTTON "2",IDC_BUTTON_2,36,20,16,14
- PUSHBUTTON "3",IDC_BUTTON_3,56,20,16,14
- PUSHBUTTON "4",IDC_BUTTON_4,16,36,16,14
- PUSHBUTTON "5",IDC_BUTTON_5,36,36,16,14
- PUSHBUTTON "6",IDC_BUTTON_6,56,36,16,14
- PUSHBUTTON "7",IDC_BUTTON_7,16,52,16,14
- PUSHBUTTON "8",IDC_BUTTON_8,36,52,16,14
- PUSHBUTTON "9",IDC_BUTTON_9,56,52,16,14
- PUSHBUTTON "*",IDC_BUTTON_10,16,68,16,14
- PUSHBUTTON "0",IDC_BUTTON_11,36,68,16,14
- PUSHBUTTON "#",IDC_BUTTON_12,56,68,16,14
- PUSHBUTTON "A",IDC_BUTTON_13,76,20,16,14
- PUSHBUTTON "B",IDC_BUTTON_14,76,36,16,14
- PUSHBUTTON "C",IDC_BUTTON_15,76,52,16,14
- PUSHBUTTON "D",IDC_BUTTON_16,76,68,16,14
- EDITTEXT IDC_EDIT_DTMF_EVENT,56,90,16,12,ES_AUTOHSCROLL | ES_READONLY
- LTEXT "Event code",IDC_STATIC_DTMF_EVENT,17,91,37,8
- PUSHBUTTON "1",IDC_BUTTON_17,16,20,16,14
- PUSHBUTTON "2",IDC_BUTTON_18,36,20,16,14
- PUSHBUTTON "3",IDC_BUTTON_19,56,20,16,14
- PUSHBUTTON "4",IDC_BUTTON_20,16,36,16,14
- PUSHBUTTON "A",IDC_BUTTON_21,76,20,16,14
- GROUPBOX "DTMF Events",IDC_STATIC_GROUP_DTMF,4,4,188,132
- CONTROL "",IDC_CHECK_DTMF_PLAYOUT_RX,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,160,21,12,14
- LTEXT "Play out-band RX",IDC_STATIC_PLAYOUT_RX,101,24,56,8
- CONTROL "",IDC_CHECK_DTMF_PLAY_TONE,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,160,39,12,14
- LTEXT "Play tone locally",IDC_STATIC_PLAY_TONE,101,41,52,8
- EDITTEXT IDC_EDIT_EVENT_LENGTH,44,163,28,14,ES_AUTOHSCROLL
- LTEXT "Duration",IDC_STATIC_EVENT_LENGTH,12,165,28,8
- EDITTEXT IDC_EDIT_EVENT_ATTENUATION,44,183,28,14,ES_AUTOHSCROLL
- LTEXT "Volume",IDC_STATIC_EVENT_ATTENUATION,12,186,24,8
- CONTROL "Inband",IDC_CHECK_EVENT_INBAND,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,84,163,40,14,WS_EX_DLGMODALFRAME
- CONTROL "Feedback",IDC_CHECK_DTMF_FEEDBACK,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,16,112,48,14,WS_EX_DLGMODALFRAME
- CONTROL "",IDC_CHECK_DIRECT_FEEDBACK,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,96,112,12,14
- LTEXT "Direct",IDC_STATIC_DIRECT_FEEDBACK,72,115,20,8
- CONTROL "Single",IDC_RADIO_SINGLE,"Button",BS_AUTORADIOBUTTON | WS_GROUP,112,68,35,10
- CONTROL "Sequence",IDC_RADIO_MULTI,"Button",BS_AUTORADIOBUTTON,112,80,47,10
- CONTROL "Start/Stop",IDC_RADIO_START_STOP,"Button",BS_AUTORADIOBUTTON,112,92,49,10
- GROUPBOX "Mode",IDC_STATIC_MODE,100,56,68,52
- EDITTEXT IDC_EDIT_EVENT_RX_PT,220,20,24,14,ES_AUTOHSCROLL
- EDITTEXT IDC_EDIT_EVENT_TX_PT,220,41,24,14,ES_AUTOHSCROLL
- LTEXT "RX",IDC_STATIC_EVENT_RX_PT,208,22,10,8
- LTEXT "TX",IDC_STATIC_EVENT_TX_PT,208,42,9,8
- PUSHBUTTON "Set",IDC_BUTTON_SET_TX_TELEPHONE_PT,248,41,24,14
- PUSHBUTTON "Set",IDC_BUTTON_SET_RX_TELEPHONE_PT,248,20,24,14
- GROUPBOX "Payload Type",IDC_STATIC_PT,200,4,80,56
- EDITTEXT IDC_EDIT_EVENT_CODE,128,163,28,14,ES_AUTOHSCROLL
- LTEXT "Event code",IDC_STATIC_EVENT_CODE,125,152,37,8
- PUSHBUTTON "Send",IDC_BUTTON_SEND_TELEPHONE_EVENT,160,163,24,14
- CONTROL "On/Off",IDC_CHECK_EVENT_DETECTION,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,208,80,40,14,WS_EX_DLGMODALFRAME
- CONTROL "",IDC_CHECK_DETECT_INBAND,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,208,100,12,14
- CONTROL "",IDC_CHECK_DETECT_OUT_OF_BAND,"Button",BS_AUTOCHECKBOX | WS_TABSTOP,208,116,12,14
- LTEXT "Inband",IDC_STATIC_INBAND_DETECTION,220,103,24,8
- LTEXT "Outband",IDC_STATIC_OUT_OF_BAND_DETECTION,220,120,29,8
- GROUPBOX "Event Detection",IDC_STATIC_EVENT_DETECTION,200,68,108,68
- GROUPBOX "Telephone Events",IDC_STATIC_TELEPHONE_EVENTS,4,140,188,64
- EDITTEXT IDC_EDIT_ON_EVENT_OUT_OF_BAND,252,117,48,14,ES_AUTOHSCROLL
- EDITTEXT IDC_EDIT_ON_EVENT_INBAND,252,101,48,14,ES_AUTOHSCROLL
- LTEXT "=> Detections",IDC_STATIC_EVEN,253,90,48,8
-END
-
-
-/////////////////////////////////////////////////////////////////////////////
-//
-// Version
-//
-
-VS_VERSION_INFO VERSIONINFO
- FILEVERSION 1,0,0,0
- PRODUCTVERSION 1,0,0,0
- FILEFLAGSMASK 0x3fL
-#ifdef _DEBUG
- FILEFLAGS 0x1L
-#else
- FILEFLAGS 0x0L
-#endif
- FILEOS 0x4L
- FILETYPE 0x1L
- FILESUBTYPE 0x0L
-BEGIN
- BLOCK "StringFileInfo"
- BEGIN
- BLOCK "040904e4"
- BEGIN
- VALUE "FileDescription", "WebRTC VoiceEngine Test"
- VALUE "FileVersion", "1.0.0.0"
- VALUE "InternalName", "WinTest.exe"
- VALUE "LegalCopyright", "Copyright (c) 2011 The WebRTC project authors. All Rights Reserved."
- VALUE "OriginalFilename", "WinTest.exe"
- VALUE "ProductName", "WebRTC VoiceEngine"
- VALUE "ProductVersion", "1.0.0.0"
- END
- END
- BLOCK "VarFileInfo"
- BEGIN
- VALUE "Translation", 0x409, 1252
- END
-END
-
-
-/////////////////////////////////////////////////////////////////////////////
-//
-// DESIGNINFO
-//
-
-#ifdef APSTUDIO_INVOKED
-GUIDELINES DESIGNINFO
-BEGIN
- IDD_ABOUTBOX, DIALOG
- BEGIN
- LEFTMARGIN, 7
- RIGHTMARGIN, 228
- TOPMARGIN, 7
- BOTTOMMARGIN, 48
- END
-
- IDD_WINTEST_DIALOG, DIALOG
- BEGIN
- LEFTMARGIN, 7
- RIGHTMARGIN, 789
- TOPMARGIN, 7
- BOTTOMMARGIN, 271
- END
-
- IDD_DTMF_DIALOG, DIALOG
- BEGIN
- LEFTMARGIN, 7
- RIGHTMARGIN, 309
- TOPMARGIN, 7
- BOTTOMMARGIN, 205
- END
-END
-#endif // APSTUDIO_INVOKED
-
-
-/////////////////////////////////////////////////////////////////////////////
-//
-// String Table
-//
-
-STRINGTABLE
-BEGIN
- IDS_ABOUTBOX "&About WinTest..."
-END
-
-#endif // Swedish resources
-/////////////////////////////////////////////////////////////////////////////
-
-
-
-#ifndef APSTUDIO_INVOKED
-/////////////////////////////////////////////////////////////////////////////
-//
-// Generated from the TEXTINCLUDE 3 resource.
-//
-#define _AFX_NO_SPLITTER_RESOURCES
-#define _AFX_NO_OLE_RESOURCES
-#define _AFX_NO_TRACKER_RESOURCES
-#define _AFX_NO_PROPERTY_RESOURCES
-
-#if !defined(AFX_RESOURCE_DLL) || defined(AFX_TARG_SVE)
-LANGUAGE 29, 1
-#pragma code_page(1252)
-#include "res\WinTest.rc2" // non-Microsoft Visual C++ edited resources
-#include "afxres.rc" // Standard components
-#endif
-
-/////////////////////////////////////////////////////////////////////////////
-#endif // not APSTUDIO_INVOKED
-
diff --git a/trunk/src/voice_engine/main/test/win_test/WinTestDlg.cc b/trunk/src/voice_engine/main/test/win_test/WinTestDlg.cc
deleted file mode 100644
index de54802..0000000
--- a/trunk/src/voice_engine/main/test/win_test/WinTestDlg.cc
+++ /dev/null
@@ -1,3573 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <math.h>
-#include "stdafx.h"
-#include "WinTest.h"
-#include "WinTestDlg.h"
-
-#ifdef _DEBUG
-#define new DEBUG_NEW
-#endif
-
-using namespace webrtc;
-
-unsigned char key[30] = {0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9, 0, 1, 2, 3, 4, 5, 6, 7, 8, 9};
-
-// Hack to convert char to TCHAR, using two buffers to be able to
-// call twice in the same statement
-TCHAR convertTemp1[256] = {0};
-TCHAR convertTemp2[256] = {0};
-bool convertBufferSwitch(false);
-TCHAR* CharToTchar(const char* str, int len)
-{
-#ifdef _UNICODE
- TCHAR* temp = convertBufferSwitch ? convertTemp1 : convertTemp2;
- convertBufferSwitch = !convertBufferSwitch;
- memset(temp, 0, sizeof(convertTemp1));
- MultiByteToWideChar(CP_UTF8, 0, str, len, temp, 256);
- return temp;
-#else
- return str;
-#endif
-}
-
-// Hack to convert TCHAR to char
-char convertTemp3[256] = {0};
-char* TcharToChar(TCHAR* str, int len)
-{
-#ifdef _UNICODE
- memset(convertTemp3, 0, sizeof(convertTemp3));
- WideCharToMultiByte(CP_UTF8, 0, str, len, convertTemp3, 256, 0, 0);
- return convertTemp3;
-#else
- return str;
-#endif
-}
-
-// ----------------------------------------------------------------------------
-// VoEConnectionObserver
-// ----------------------------------------------------------------------------
-
-class ConnectionObserver : public VoEConnectionObserver
-{
-public:
- ConnectionObserver();
- virtual void OnPeriodicDeadOrAlive(const int channel, const bool alive);
-};
-
-ConnectionObserver::ConnectionObserver()
-{
-}
-
-void ConnectionObserver::OnPeriodicDeadOrAlive(const int channel, const bool alive)
-{
- CString str;
- str.Format(_T("OnPeriodicDeadOrAlive(channel=%d) => alive=%d"), channel, alive);
- OutputDebugString(str);
-}
-
-// ----------------------------------------------------------------------------
-// VoiceEngineObserver
-// ----------------------------------------------------------------------------
-
-void CWinTestDlg::CallbackOnError(const int channel, const int errCode)
-{
- _nErrorCallbacks++;
-
- CString str;
- str.Format(_T("[#%d] CallbackOnError(channel=%d) => errCode = %d"), _nErrorCallbacks, channel, errCode);
- if (errCode == VE_RECEIVE_PACKET_TIMEOUT)
- {
- str += _T(" <=> VE_RECEIVE_PACKET_TIMEOUT");
- }
- else if (errCode == VE_PACKET_RECEIPT_RESTARTED)
- {
- str += _T(" <=> VE_PACKET_RECEIPT_RESTARTED");
- }
- else if (errCode == VE_RUNTIME_PLAY_WARNING)
- {
- str += _T(" <=> VE_RUNTIME_PLAY_WARNING");
- }
- else if (errCode == VE_RUNTIME_REC_WARNING)
- {
- str += _T(" <=> VE_RUNTIME_REC_WARNING");
- }
- else if (errCode == VE_RUNTIME_PLAY_ERROR)
- {
- str += _T(" <=> VE_RUNTIME_PLAY_ERROR");
- }
- else if (errCode == VE_RUNTIME_REC_ERROR)
- {
- str += _T(" <=> VE_RUNTIME_REC_ERROR");
- }
- else if (errCode == VE_SATURATION_WARNING)
- {
- str += _T(" <=> VE_SATURATION_WARNING");
- }
- else if (errCode == VE_TYPING_NOISE_WARNING)
- {
- str += _T(" <=> VE_TYPING_NOISE_WARNING");
- }
- else if (errCode == VE_REC_DEVICE_REMOVED)
- {
- str += _T(" <=> VE_REC_DEVICE_REMOVED");
- }
- // AfxMessageBox((LPCTSTR)str, MB_OK);
- SetDlgItemText(IDC_EDIT_ERROR_CALLBACK, (LPCTSTR)str);
-}
-
-// ----------------------------------------------------------------------------
-// VoERTPObserver
-// ----------------------------------------------------------------------------
-
-void CWinTestDlg::OnIncomingCSRCChanged(const int channel, const unsigned int CSRC, const bool added)
-{
- CString str;
- str.Format(_T("OnIncomingCSRCChanged(channel=%d) => CSRC=%u, added=%d"), channel, CSRC, added);
- SetDlgItemText(IDC_EDIT_ERROR_CALLBACK, (LPCTSTR)str);
-}
-
-void CWinTestDlg::OnIncomingSSRCChanged(const int channel, const unsigned int SSRC)
-{
- CString str;
- str.Format(_T("OnIncomingSSRCChanged(channel=%d) => SSRC=%u"), channel, SSRC);
- SetDlgItemText(IDC_EDIT_ERROR_CALLBACK, (LPCTSTR)str);
-}
-
-// ----------------------------------------------------------------------------
-// Transport
-// ----------------------------------------------------------------------------
-
-class MyTransport : public Transport
-{
-public:
- MyTransport(VoENetwork* veNetwork);
- virtual int SendPacket(int channel, const void *data, int len);
- virtual int SendRTCPPacket(int channel, const void *data, int len);
-private:
- VoENetwork* _veNetworkPtr;
-};
-
-MyTransport::MyTransport(VoENetwork* veNetwork) :
- _veNetworkPtr(veNetwork)
-{
-}
-
-int
-MyTransport::SendPacket(int channel, const void *data, int len)
-{
- _veNetworkPtr->ReceivedRTPPacket(channel, data, len);
- return len;
-}
-
-int
-MyTransport::SendRTCPPacket(int channel, const void *data, int len)
-{
- _veNetworkPtr->ReceivedRTCPPacket(channel, data, len);
- return len;
-}
-
-// ----------------------------------------------------------------------------
-// VoEMediaProcess
-// ----------------------------------------------------------------------------
-
-class MediaProcessImpl : public VoEMediaProcess
-{
-public:
- MediaProcessImpl();
- virtual void Process(const int channel,
- const ProcessingTypes type,
- WebRtc_Word16 audio_10ms[],
- const int length,
- const int samplingFreqHz,
- const bool stereo);
-};
-
-MediaProcessImpl::MediaProcessImpl()
-{
-}
-
-void MediaProcessImpl::Process(const int channel,
- const ProcessingTypes type,
- WebRtc_Word16 audio_10ms[],
- const int length,
- const int samplingFreqHz,
- const bool stereo)
-{
- int x = rand() % 100;
-
- for (int i = 0; i < length; i++)
- {
- if (channel == -1)
- {
- if (type == kPlaybackAllChannelsMixed)
- {
- // playout: scale up
- if (!stereo)
- {
- audio_10ms[i] = (audio_10ms[i] << 2);
- }
- else
- {
- audio_10ms[2*i] = (audio_10ms[2*i] << 2);
- audio_10ms[2*i+1] = (audio_10ms[2*i+1] << 2);
- }
- }
- else
- {
- // recording: emulate packet loss by "dropping" 10% of the packets
- if (x >= 0 && x < 10)
- {
- if (!stereo)
- {
- audio_10ms[i] = 0;
- }
- else
- {
- audio_10ms[2*i] = 0;
- audio_10ms[2*i+1] = 0;
- }
- }
- }
- }
- else
- {
- if (type == kPlaybackPerChannel)
- {
- // playout: mute
- if (!stereo)
- {
- audio_10ms[i] = 0;
- }
- else
- {
- audio_10ms[2*i] = 0;
- audio_10ms[2*i+1] = 0;
- }
- }
- else
- {
- // recording: emulate packet loss by "dropping" 50% of the packets
- if (x >= 0 && x < 50)
- {
- if (!stereo)
- {
- audio_10ms[i] = 0;
- }
- else
- {
- audio_10ms[2*i] = 0;
- audio_10ms[2*i+1] = 0;
- }
- }
- }
- }
- }
-}
-
-// ----------------------------------------------------------------------------
-// Encryptionen
-// ----------------------------------------------------------------------------
-
-class MyEncryption : public Encryption
-{
-public:
- void encrypt(int channel_no, unsigned char * in_data, unsigned char * out_data, int bytes_in, int* bytes_out);
- void decrypt(int channel_no, unsigned char * in_data, unsigned char * out_data, int bytes_in, int* bytes_out);
- void encrypt_rtcp(int channel_no, unsigned char * in_data, unsigned char * out_data, int bytes_in, int* bytes_out);
- void decrypt_rtcp(int channel_no, unsigned char * in_data, unsigned char * out_data, int bytes_in, int* bytes_out);
-};
-
-void MyEncryption::encrypt(int channel_no, unsigned char * in_data, unsigned char * out_data, int bytes_in, int* bytes_out)
-{
- // --- Stereo emulation (sample based, 2 bytes per sample)
-
- const int nBytesPayload = bytes_in-12;
-
- // RTP header (first 12 bytes)
- memcpy(out_data, in_data, 12);
-
- // skip RTP header
- short* ptrIn = (short*) &in_data[12];
- short* ptrOut = (short*) &out_data[12];
-
- // network byte order
- for (int i = 0; i < nBytesPayload/2; i++)
- {
- // produce two output samples for each input sample
- *ptrOut++ = *ptrIn; // left sample
- *ptrOut++ = *ptrIn; // right sample
- ptrIn++;
- }
-
- *bytes_out = 12 + 2*nBytesPayload;
-
- /*
- for(int i = 0; i < bytes_in; i++)
- out_data[i] =~ in_data[i];
- *bytes_out = bytes_in;
- */
-}
-
-void MyEncryption::decrypt(int channel_no, unsigned char * in_data, unsigned char * out_data, int bytes_in, int* bytes_out)
-{
- // Do nothing (<=> memcpy)
- for(int i = 0; i < bytes_in; i++)
- out_data[i] = in_data[i];
- *bytes_out = bytes_in;
-}
-
-void MyEncryption::encrypt_rtcp(int channel_no, unsigned char * in_data, unsigned char * out_data, int bytes_in, int* bytes_out)
-{
- for(int i = 0; i < bytes_in; i++)
- out_data[i] =~ in_data[i];
- *bytes_out = bytes_in;
-}
-
-void MyEncryption::decrypt_rtcp(int channel_no, unsigned char * in_data, unsigned char * out_data, int bytes_in, int* bytes_out)
-{
- for(int i = 0; i < bytes_in; i++)
- out_data[i] =~ in_data[i];
- *bytes_out = bytes_in;
-}
-
-// ----------------------------------------------------------------------------
-// TelephoneEventObserver
-// ----------------------------------------------------------------------------
-
-class TelephoneEventObserver: public VoETelephoneEventObserver
-{
-public:
- TelephoneEventObserver(CWnd* editControlOut, CWnd* editControlIn);
- virtual void OnReceivedTelephoneEventInband(int channel, int eventCode,
- bool endOfEvent);
- virtual void OnReceivedTelephoneEventOutOfBand(int channel, int eventCode,
- bool endOfEvent);
-private:
- CWnd* _editControlOutPtr;
- CWnd* _editControlInPtr;
-};
-
-TelephoneEventObserver::TelephoneEventObserver(CWnd* editControlOut, CWnd* editControlIn) :
- _editControlOutPtr(editControlOut),
- _editControlInPtr(editControlIn)
-{
-}
-
-void TelephoneEventObserver::OnReceivedTelephoneEventInband(int channel,
- int eventCode,
- bool endOfEvent)
-{
- CString msg;
- if (endOfEvent)
- {
- msg.AppendFormat(_T("%d [END]"), eventCode);
- _editControlInPtr->SetWindowText((LPCTSTR)msg);
- }
- else
- {
- msg.AppendFormat(_T("%d [START]"), eventCode);
- _editControlInPtr->SetWindowText((LPCTSTR)msg);
- }
-}
-
-void TelephoneEventObserver::OnReceivedTelephoneEventOutOfBand(int channel,
- int eventCode,
- bool endOfEvent)
-{
- CString msg;
- if (endOfEvent)
- {
- msg.AppendFormat(_T("%d [END]"), eventCode);
- _editControlOutPtr->SetWindowText((LPCTSTR)msg);
- }
- else
- {
- msg.AppendFormat(_T("%d [START]"), eventCode);
- _editControlOutPtr->SetWindowText((LPCTSTR)msg);
- }
-}
-
-// ----------------------------------------------------------------------------
-// RxVadCallback
-// ----------------------------------------------------------------------------
-
-class RxCallback : public VoERxVadCallback
-{
-public:
- RxCallback() : _vadDecision(-1) {};
-
- virtual void OnRxVad(int , int vadDecision)
- {
- _vadDecision = vadDecision;
- }
-
- int _vadDecision;
-};
-
-// ----------------------------------------------------------------------------
-// CAboutDlg dialog
-// ----------------------------------------------------------------------------
-
-class CAboutDlg : public CDialog
-{
-public:
- CAboutDlg();
-
-// Dialog Data
- enum { IDD = IDD_ABOUTBOX };
-
- protected:
- virtual void DoDataExchange(CDataExchange* pDX); // DDX/DDV support
-
-// Implementation
-protected:
- DECLARE_MESSAGE_MAP()
-};
-
-CAboutDlg::CAboutDlg() : CDialog(CAboutDlg::IDD)
-{
-}
-
-void CAboutDlg::DoDataExchange(CDataExchange* pDX)
-{
- CDialog::DoDataExchange(pDX);
-}
-
-BEGIN_MESSAGE_MAP(CAboutDlg, CDialog)
-END_MESSAGE_MAP()
-
-// ----------------------------------------------------------------------------
-// CTelephonyEvent dialog
-// ----------------------------------------------------------------------------
-
-class CTelephonyEvent : public CDialog
-{
- DECLARE_DYNAMIC(CTelephonyEvent)
-
-public:
- CTelephonyEvent(VoiceEngine* voiceEngine, int channel, CDialog* pParentDialog, CWnd* pParent = NULL); // standard constructor
- virtual ~CTelephonyEvent();
-
-// Dialog Data
- enum { IDD = IDD_DTMF_DIALOG };
-
-protected:
- virtual void DoDataExchange(CDataExchange* pDX); // DDX/DDV support
- virtual BOOL OnInitDialog();
-
- DECLARE_MESSAGE_MAP()
-public:
- afx_msg void OnBnClickedButton1();
- afx_msg void OnBnClickedButton2();
- afx_msg void OnBnClickedButton3();
- afx_msg void OnBnClickedButton4();
- afx_msg void OnBnClickedButton5();
- afx_msg void OnBnClickedButton6();
- afx_msg void OnBnClickedButton7();
- afx_msg void OnBnClickedButton8();
- afx_msg void OnBnClickedButton9();
- afx_msg void OnBnClickedButton10();
- afx_msg void OnBnClickedButton11();
- afx_msg void OnBnClickedButton12();
- afx_msg void OnBnClickedButtonA();
- afx_msg void OnBnClickedButtonB();
- afx_msg void OnBnClickedButtonC();
- afx_msg void OnBnClickedButtonD();
- afx_msg void OnBnClickedCheckDtmfPlayoutRx();
- afx_msg void OnBnClickedCheckDtmfPlayTone();
- afx_msg void OnBnClickedCheckStartStopMode();
- afx_msg void OnBnClickedCheckEventInband();
- afx_msg void OnBnClickedCheckDtmfFeedback();
- afx_msg void OnBnClickedCheckDirectFeedback();
- afx_msg void OnBnClickedRadioSingle();
- afx_msg void OnBnClickedRadioMulti();
- afx_msg void OnBnClickedRadioStartStop();
- afx_msg void OnBnClickedButtonSetRxTelephonePt();
- afx_msg void OnBnClickedButtonSetTxTelephonePt();
- afx_msg void OnBnClickedButtonSendTelephoneEvent();
- afx_msg void OnBnClickedCheckDetectInband();
- afx_msg void OnBnClickedCheckDetectOutOfBand();
- afx_msg void OnBnClickedCheckEventDetection();
-
-private:
- void SendTelephoneEvent(unsigned char eventCode);
-
-private:
- VoiceEngine* _vePtr;
- VoEBase* _veBasePtr;
- VoEDtmf* _veDTMFPtr;
- VoECodec* _veCodecPtr;
- int _channel;
- CString _strMsg;
- CDialog* _parentDialogPtr;
- TelephoneEventObserver* _telephoneEventObserverPtr;
- bool _PlayDtmfToneLocally;
- bool _modeStartStop;
- bool _modeSingle;
- bool _modeSequence;
- bool _playingDTMFTone;
- bool _outOfBandEventDetection;
- bool _inbandEventDetection;
-};
-
-IMPLEMENT_DYNAMIC(CTelephonyEvent, CDialog)
-
-CTelephonyEvent::CTelephonyEvent(VoiceEngine* voiceEngine,
- int channel,
- CDialog* pParentDialog,
- CWnd* pParent /*=NULL*/)
- : _vePtr(voiceEngine),
- _channel(channel),
- _PlayDtmfToneLocally(false),
- _modeStartStop(false),
- _modeSingle(true),
- _modeSequence(false),
- _playingDTMFTone(false),
- _outOfBandEventDetection(true),
- _inbandEventDetection(false),
- _parentDialogPtr(pParentDialog),
- _telephoneEventObserverPtr(NULL),
- CDialog(CTelephonyEvent::IDD, pParent)
-{
- _veBasePtr = VoEBase::GetInterface(_vePtr);
- _veDTMFPtr = VoEDtmf::GetInterface(_vePtr);
- _veCodecPtr = VoECodec::GetInterface(_vePtr);
-}
-
-CTelephonyEvent::~CTelephonyEvent()
-{
- _veDTMFPtr->Release();
- _veCodecPtr->Release();
- _veBasePtr->Release();
-
- if (_telephoneEventObserverPtr)
- {
- _veDTMFPtr->DeRegisterTelephoneEventDetection(_channel);
- delete _telephoneEventObserverPtr;
- _telephoneEventObserverPtr = NULL;
- }
-}
-
-void CTelephonyEvent::DoDataExchange(CDataExchange* pDX)
-{
- CDialog::DoDataExchange(pDX);
-}
-
-
-BEGIN_MESSAGE_MAP(CTelephonyEvent, CDialog)
- ON_BN_CLICKED(IDC_BUTTON_1, &CTelephonyEvent::OnBnClickedButton1)
- ON_BN_CLICKED(IDC_BUTTON_2, &CTelephonyEvent::OnBnClickedButton2)
- ON_BN_CLICKED(IDC_BUTTON_3, &CTelephonyEvent::OnBnClickedButton3)
- ON_BN_CLICKED(IDC_BUTTON_4, &CTelephonyEvent::OnBnClickedButton4)
- ON_BN_CLICKED(IDC_BUTTON_5, &CTelephonyEvent::OnBnClickedButton5)
- ON_BN_CLICKED(IDC_BUTTON_6, &CTelephonyEvent::OnBnClickedButton6)
- ON_BN_CLICKED(IDC_BUTTON_7, &CTelephonyEvent::OnBnClickedButton7)
- ON_BN_CLICKED(IDC_BUTTON_8, &CTelephonyEvent::OnBnClickedButton8)
- ON_BN_CLICKED(IDC_BUTTON_9, &CTelephonyEvent::OnBnClickedButton9)
- ON_BN_CLICKED(IDC_BUTTON_10, &CTelephonyEvent::OnBnClickedButton10)
- ON_BN_CLICKED(IDC_BUTTON_11, &CTelephonyEvent::OnBnClickedButton11)
- ON_BN_CLICKED(IDC_BUTTON_12, &CTelephonyEvent::OnBnClickedButton12)
- ON_BN_CLICKED(IDC_BUTTON_13, &CTelephonyEvent::OnBnClickedButtonA)
- ON_BN_CLICKED(IDC_BUTTON_14, &CTelephonyEvent::OnBnClickedButtonB)
- ON_BN_CLICKED(IDC_BUTTON_15, &CTelephonyEvent::OnBnClickedButtonC)
- ON_BN_CLICKED(IDC_BUTTON_16, &CTelephonyEvent::OnBnClickedButtonD)
- ON_BN_CLICKED(IDC_CHECK_DTMF_PLAYOUT_RX, &CTelephonyEvent::OnBnClickedCheckDtmfPlayoutRx)
- ON_BN_CLICKED(IDC_CHECK_DTMF_PLAY_TONE, &CTelephonyEvent::OnBnClickedCheckDtmfPlayTone)
- ON_BN_CLICKED(IDC_CHECK_EVENT_INBAND, &CTelephonyEvent::OnBnClickedCheckEventInband)
- ON_BN_CLICKED(IDC_CHECK_DTMF_FEEDBACK, &CTelephonyEvent::OnBnClickedCheckDtmfFeedback)
- ON_BN_CLICKED(IDC_CHECK_DIRECT_FEEDBACK, &CTelephonyEvent::OnBnClickedCheckDirectFeedback)
- ON_BN_CLICKED(IDC_RADIO_SINGLE, &CTelephonyEvent::OnBnClickedRadioSingle)
- ON_BN_CLICKED(IDC_RADIO_MULTI, &CTelephonyEvent::OnBnClickedRadioMulti)
- ON_BN_CLICKED(IDC_RADIO_START_STOP, &CTelephonyEvent::OnBnClickedRadioStartStop)
- ON_BN_CLICKED(IDC_BUTTON_SET_RX_TELEPHONE_PT, &CTelephonyEvent::OnBnClickedButtonSetRxTelephonePt)
- ON_BN_CLICKED(IDC_BUTTON_SET_TX_TELEPHONE_PT, &CTelephonyEvent::OnBnClickedButtonSetTxTelephonePt)
- ON_BN_CLICKED(IDC_BUTTON_SEND_TELEPHONE_EVENT, &CTelephonyEvent::OnBnClickedButtonSendTelephoneEvent)
- ON_BN_CLICKED(IDC_CHECK_DETECT_INBAND, &CTelephonyEvent::OnBnClickedCheckDetectInband)
- ON_BN_CLICKED(IDC_CHECK_DETECT_OUT_OF_BAND, &CTelephonyEvent::OnBnClickedCheckDetectOutOfBand)
- ON_BN_CLICKED(IDC_CHECK_EVENT_DETECTION, &CTelephonyEvent::OnBnClickedCheckEventDetection)
-END_MESSAGE_MAP()
-
-
-// CTelephonyEvent message handlers
-
-BOOL CTelephonyEvent::OnInitDialog()
-{
- CDialog::OnInitDialog();
-
- CString str;
- GetWindowText(str);
- str.AppendFormat(_T(" [channel = %d]"), _channel);
- SetWindowText(str);
-
- // Update dialog with latest playout state
- bool enabled(false);
- _veDTMFPtr->GetDtmfPlayoutStatus(_channel, enabled);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_PLAYOUT_RX);
- button->SetCheck(enabled ? BST_CHECKED : BST_UNCHECKED);
-
- // Update dialog with latest feedback state
- bool directFeedback(false);
- _veDTMFPtr->GetDtmfFeedbackStatus(enabled, directFeedback);
- button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_FEEDBACK);
- button->SetCheck(enabled ? BST_CHECKED : BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_DIRECT_FEEDBACK);
- button->SetCheck(directFeedback ? BST_CHECKED : BST_UNCHECKED);
-
- // Default event length is 160 ms
- SetDlgItemInt(IDC_EDIT_EVENT_LENGTH, 160);
-
- // Default event attenuation is 10 (<-> -10dBm0)
- SetDlgItemInt(IDC_EDIT_EVENT_ATTENUATION, 10);
-
- // Current event-detection status
- TelephoneEventDetectionMethods detectionMethod(kOutOfBand);
- if (_veDTMFPtr->GetTelephoneEventDetectionStatus(_channel, enabled, detectionMethod) == 0)
- {
- // DTMF detection is supported
- if (enabled)
- {
- button = (CButton*)GetDlgItem(IDC_CHECK_EVENT_DETECTION);
- button->SetCheck(BST_CHECKED);
- }
- if (detectionMethod == kOutOfBand || detectionMethod == kInAndOutOfBand)
- {
- button = (CButton*)GetDlgItem(IDC_CHECK_DETECT_OUT_OF_BAND);
- button->SetCheck(BST_CHECKED);
- }
- if (detectionMethod == kInBand || detectionMethod == kInAndOutOfBand)
- {
- button = (CButton*)GetDlgItem(IDC_CHECK_DETECT_INBAND);
- button->SetCheck(BST_CHECKED);
- }
- }
- else
- {
- // DTMF detection is not supported
- GetDlgItem(IDC_CHECK_EVENT_DETECTION)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_DETECT_OUT_OF_BAND)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_DETECT_INBAND)->EnableWindow(FALSE);
- GetDlgItem(IDC_EDIT_ON_EVENT_INBAND)->EnableWindow(FALSE);
- GetDlgItem(IDC_EDIT_ON_EVENT_OUT_OF_BAND)->EnableWindow(FALSE);
- }
-
- // Telephone-event PTs
- unsigned char pt(0);
- _veDTMFPtr->GetSendTelephoneEventPayloadType(_channel, pt);
- SetDlgItemInt(IDC_EDIT_EVENT_TX_PT, pt);
-
- CodecInst codec;
- strcpy_s(codec.plname, 32, "telephone-event"); codec.channels = 1; codec.plfreq = 8000;
- _veCodecPtr->GetRecPayloadType(_channel, codec);
- SetDlgItemInt(IDC_EDIT_EVENT_RX_PT, codec.pltype);
-
- if (_modeSingle)
- {
- ((CButton*)GetDlgItem(IDC_RADIO_SINGLE))->SetCheck(BST_CHECKED);
- }
- else if (_modeStartStop)
- {
- ((CButton*)GetDlgItem(IDC_RADIO_START_STOP))->SetCheck(BST_CHECKED);
- }
- else if (_modeSequence)
- {
- ((CButton*)GetDlgItem(IDC_RADIO_MULTI))->SetCheck(BST_CHECKED);
- }
-
- return TRUE; // return TRUE unless you set the focus to a control
-}
-void CTelephonyEvent::SendTelephoneEvent(unsigned char eventCode)
-{
- BOOL ret;
- int lengthMs(0);
- int attenuationDb(0);
- bool outBand(false);
- int res(0);
-
- // tone length
- if (!_modeStartStop)
- {
- lengthMs = GetDlgItemInt(IDC_EDIT_EVENT_LENGTH, &ret);
- if (ret == FALSE)
- {
- // use default length if edit field is empty
- lengthMs = 160;
- }
- }
-
- // attenuation
- attenuationDb = GetDlgItemInt(IDC_EDIT_EVENT_ATTENUATION, &ret);
- if (ret == FALSE)
- {
- // use default length if edit field is empty
- attenuationDb = 10;
- }
-
- // out-band or in-band
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EVENT_INBAND);
- int check = button->GetCheck();
- outBand = (check == BST_UNCHECKED);
-
- if (eventCode < 16)
- SetDlgItemInt(IDC_EDIT_DTMF_EVENT, eventCode);
-
- if (_PlayDtmfToneLocally)
- {
- // --- PlayDtmfTone
-
- if (_modeSingle)
- {
- TEST2(_veDTMFPtr->PlayDtmfTone(eventCode, lengthMs, attenuationDb) == 0,
- _T("PlayDtmfTone(eventCode=%u, lengthMs=%d, attenuationDb=%d)"), eventCode, lengthMs, attenuationDb);
- }
- else if (_modeStartStop)
- {
- if (!_playingDTMFTone)
- {
- TEST2((res = _veDTMFPtr->StartPlayingDtmfTone(eventCode, attenuationDb)) == 0,
- _T("StartPlayingDtmfTone(eventCode=%u, attenuationDb=%d)"), eventCode, attenuationDb);
- }
- else
- {
- TEST2((res = _veDTMFPtr->StopPlayingDtmfTone()) == 0,
- _T("StopPlayingDTMFTone()"));
- }
- if (res == 0)
- _playingDTMFTone = !_playingDTMFTone;
- }
- else if (_modeSequence)
- {
- int nTones(1);
- int sleepMs(0);
- int lenMult(1);
- if (eventCode == 1)
- {
- nTones = 2;
- sleepMs = lengthMs;
- lenMult = 1;
- }
- else if (eventCode == 2)
- {
- nTones = 2;
- sleepMs = lengthMs/2;
- lenMult = 2;
- }
- else if (eventCode == 3)
- {
- nTones = 3;
- sleepMs = 0;
- lenMult = 1;
- }
- for (int i = 0; i < nTones; i++)
- {
- TEST2(_veDTMFPtr->PlayDtmfTone(eventCode, lengthMs, attenuationDb) == 0,
- _T("PlayDtmfTone(eventCode=%u, outBand=%d, lengthMs=%d, attenuationDb=%d)"), eventCode, lengthMs, attenuationDb);
- Sleep(sleepMs);
- lengthMs = lenMult*lengthMs;
- eventCode++;
- }
- }
- }
- else
- {
- // --- SendTelephoneEvent
-
- if (_modeSingle)
- {
- TEST2(_veDTMFPtr->SendTelephoneEvent(_channel, eventCode, outBand, lengthMs, attenuationDb) == 0,
- _T("SendTelephoneEvent(channel=%d, eventCode=%u, outBand=%d, lengthMs=%d, attenuationDb=%d)"), _channel, eventCode, outBand, lengthMs, attenuationDb);
- }
- else if (_modeStartStop)
- {
- TEST2(false, _T("*** NOT IMPLEMENTED ***"));
- }
- else if (_modeSequence)
- {
- int nTones(1);
- int sleepMs(0);
- int lenMult(1);
- if (eventCode == 1)
- {
- nTones = 2;
- sleepMs = lengthMs;
- lenMult = 1;
- }
- else if (eventCode == 2)
- {
- eventCode = 1;
- nTones = 2;
- sleepMs = lengthMs/2;
- lenMult = 2;
- }
- else if (eventCode == 3)
- {
- eventCode = 1;
- nTones = 3;
- sleepMs = 0;
- lenMult = 1;
- }
- for (int i = 0; i < nTones; i++)
- {
- TEST2(_veDTMFPtr->SendTelephoneEvent(_channel, eventCode, outBand, lengthMs, attenuationDb) == 0,
- _T("SendTelephoneEvent(channel=%d, eventCode=%u, outBand=%d, lengthMs=%d, attenuationDb=%d)"), _channel, eventCode, outBand, lengthMs, attenuationDb);
- Sleep(sleepMs);
- lengthMs = lenMult*lengthMs;
- eventCode++;
- }
- }
- }
-}
-
-void CTelephonyEvent::OnBnClickedButtonSendTelephoneEvent()
-{
- BOOL ret;
- unsigned char eventCode(0);
-
- eventCode = (unsigned char)GetDlgItemInt(IDC_EDIT_EVENT_CODE, &ret);
- if (ret == FALSE)
- {
- return;
- }
- SendTelephoneEvent(eventCode);
-}
-
-void CTelephonyEvent::OnBnClickedButton1()
-{
- SendTelephoneEvent(1);
-}
-
-void CTelephonyEvent::OnBnClickedButton2()
-{
- SendTelephoneEvent(2);
-}
-
-void CTelephonyEvent::OnBnClickedButton3()
-{
- SendTelephoneEvent(3);
-}
-
-void CTelephonyEvent::OnBnClickedButton4()
-{
- SendTelephoneEvent(4);
-}
-
-void CTelephonyEvent::OnBnClickedButton5()
-{
- SendTelephoneEvent(5);
-}
-
-void CTelephonyEvent::OnBnClickedButton6()
-{
- SendTelephoneEvent(6);
-}
-
-void CTelephonyEvent::OnBnClickedButton7()
-{
- SendTelephoneEvent(7);
-}
-
-void CTelephonyEvent::OnBnClickedButton8()
-{
- SendTelephoneEvent(8);
-}
-
-void CTelephonyEvent::OnBnClickedButton9()
-{
- SendTelephoneEvent(9);
-}
-
-void CTelephonyEvent::OnBnClickedButton10()
-{
- // *
- SendTelephoneEvent(10);
-}
-
-void CTelephonyEvent::OnBnClickedButton11()
-{
- SendTelephoneEvent(0);
-}
-
-void CTelephonyEvent::OnBnClickedButton12()
-{
- // #
- SendTelephoneEvent(11);
-}
-
-void CTelephonyEvent::OnBnClickedButtonA()
-{
- SendTelephoneEvent(12);
-}
-
-void CTelephonyEvent::OnBnClickedButtonB()
-{
- SendTelephoneEvent(13);
-}
-
-void CTelephonyEvent::OnBnClickedButtonC()
-{
- SendTelephoneEvent(14);
-}
-
-void CTelephonyEvent::OnBnClickedButtonD()
-{
- SendTelephoneEvent(15);
-}
-
-void CTelephonyEvent::OnBnClickedCheckDtmfPlayoutRx()
-{
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_PLAYOUT_RX);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- TEST2(_veDTMFPtr->SetDtmfPlayoutStatus(_channel, enable) == 0, _T("SetDtmfPlayoutStatus(channel=%d, enable=%d)"), _channel, enable);
-}
-
-void CTelephonyEvent::OnBnClickedCheckDtmfPlayTone()
-{
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_PLAY_TONE);
- int check = button->GetCheck();
- _PlayDtmfToneLocally = (check == BST_CHECKED);
-}
-
-void CTelephonyEvent::OnBnClickedRadioSingle()
-{
- _modeStartStop = false;
- _modeSingle = true;
- _modeSequence = false;
-}
-
-void CTelephonyEvent::OnBnClickedRadioMulti()
-{
- _modeStartStop = false;
- _modeSingle = false;
- _modeSequence = true;
-}
-
-void CTelephonyEvent::OnBnClickedRadioStartStop()
-{
- // CButton* button = (CButton*)GetDlgItem(IDC_RADIO_START_STOP);
- // int check = button->GetCheck();
- _modeStartStop = true;
- _modeSingle = false;
- _modeSequence = false;
- // GetDlgItem(IDC_EDIT_EVENT_LENGTH)->EnableWindow();
-}
-
-void CTelephonyEvent::OnBnClickedCheckEventInband()
-{
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EVENT_INBAND);
- int check = button->GetCheck();
- GetDlgItem(IDC_EDIT_EVENT_CODE)->EnableWindow(check?FALSE:TRUE);
- GetDlgItem(IDC_BUTTON_SEND_TELEPHONE_EVENT)->EnableWindow(check?FALSE:TRUE);
-}
-
-void CTelephonyEvent::OnBnClickedCheckDtmfFeedback()
-{
- CButton* button(NULL);
-
- // Retrieve feedback state
- button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_FEEDBACK);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
-
- // Retrieve direct-feedback setting
- button = (CButton*)GetDlgItem(IDC_CHECK_DIRECT_FEEDBACK);
- check = button->GetCheck();
- const bool directFeedback = (check == BST_CHECKED);
-
- // GetDlgItem(IDC_CHECK_DIRECT_FEEDBACK)->EnableWindow(enable ? TRUE : FALSE);
-
- TEST2(_veDTMFPtr->SetDtmfFeedbackStatus(enable, directFeedback) == 0,
- _T("SetDtmfFeedbackStatus(enable=%d, directFeedback=%d)"), enable, directFeedback);
-}
-
-void CTelephonyEvent::OnBnClickedCheckDirectFeedback()
-{
- CButton* button(NULL);
-
- // Retrieve feedback state
- button = (CButton*)GetDlgItem(IDC_CHECK_DTMF_FEEDBACK);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
-
- // Retrieve new direct-feedback setting
- button = (CButton*)GetDlgItem(IDC_CHECK_DIRECT_FEEDBACK);
- check = button->GetCheck();
- const bool directFeedback = (check == BST_CHECKED);
-
- TEST2(_veDTMFPtr->SetDtmfFeedbackStatus(enable, directFeedback) == 0,
- _T("SetDtmfFeedbackStatus(enable=%d, directFeedback=%d)"), enable, directFeedback);
-}
-
-void CTelephonyEvent::OnBnClickedButtonSetRxTelephonePt()
-{
- BOOL ret;
- int pt = GetDlgItemInt(IDC_EDIT_EVENT_RX_PT, &ret);
- if (ret == FALSE)
- return;
- CodecInst codec;
- strcpy_s(codec.plname, 32, "telephone-event");
- codec.pltype = pt; codec.channels = 1; codec.plfreq = 8000;
- TEST2(_veCodecPtr->SetRecPayloadType(_channel, codec) == 0,
- _T("SetSendTelephoneEventPayloadType(channel=%d, codec.pltype=%u)"), _channel, codec.pltype);
-}
-
-void CTelephonyEvent::OnBnClickedButtonSetTxTelephonePt()
-{
- BOOL ret;
- int pt = GetDlgItemInt(IDC_EDIT_EVENT_TX_PT, &ret);
- if (ret == FALSE)
- return;
- TEST2(_veDTMFPtr->SetSendTelephoneEventPayloadType(_channel, pt) == 0,
- _T("SetSendTelephoneEventPayloadType(channel=%d, type=%u)"), _channel, pt);
-}
-
-void CTelephonyEvent::OnBnClickedCheckDetectInband()
-{
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DETECT_INBAND);
- int check = button->GetCheck();
- _inbandEventDetection = (check == BST_CHECKED);
-
- bool enabled(false);
- TelephoneEventDetectionMethods detectionMethod;
- _veDTMFPtr->GetTelephoneEventDetectionStatus(_channel, enabled, detectionMethod);
- if (enabled)
- {
- // deregister
- _veDTMFPtr->DeRegisterTelephoneEventDetection(_channel);
- delete _telephoneEventObserverPtr;
- _telephoneEventObserverPtr = NULL;
- SetDlgItemText(IDC_EDIT_ON_EVENT_INBAND,_T(""));
- SetDlgItemText(IDC_EDIT_ON_EVENT_OUT_OF_BAND,_T(""));
- }
- OnBnClickedCheckEventDetection();
-}
-
-void CTelephonyEvent::OnBnClickedCheckDetectOutOfBand()
-{
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DETECT_OUT_OF_BAND);
- int check = button->GetCheck();
- _outOfBandEventDetection = (check == BST_CHECKED);
-
- bool enabled(false);
- TelephoneEventDetectionMethods detectionMethod;
- _veDTMFPtr->GetTelephoneEventDetectionStatus(_channel, enabled, detectionMethod);
- if (enabled)
- {
- // deregister
- _veDTMFPtr->DeRegisterTelephoneEventDetection(_channel);
- delete _telephoneEventObserverPtr;
- _telephoneEventObserverPtr = NULL;
- SetDlgItemText(IDC_EDIT_ON_EVENT_INBAND,_T(""));
- SetDlgItemText(IDC_EDIT_ON_EVENT_OUT_OF_BAND,_T(""));
- }
- OnBnClickedCheckEventDetection();
-}
-
-void CTelephonyEvent::OnBnClickedCheckEventDetection()
-{
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EVENT_DETECTION);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
-
- if (enable)
- {
- TelephoneEventDetectionMethods method(kInBand);
- if (_inbandEventDetection && !_outOfBandEventDetection)
- method = kInBand;
- else if (!_inbandEventDetection && _outOfBandEventDetection)
- method = kOutOfBand;
- else if (_inbandEventDetection && _outOfBandEventDetection)
- method = kInAndOutOfBand;
-
- CWnd* wndOut = GetDlgItem(IDC_EDIT_ON_EVENT_OUT_OF_BAND);
- CWnd* wndIn = GetDlgItem(IDC_EDIT_ON_EVENT_INBAND);
- _telephoneEventObserverPtr = new TelephoneEventObserver(wndOut, wndIn);
-
- TEST2(_veDTMFPtr->RegisterTelephoneEventDetection(_channel, method, *_telephoneEventObserverPtr) == 0,
- _T("RegisterTelephoneEventDetection(channel=%d, detectionMethod=%d)"), _channel, method);
- }
- else
- {
- TEST2(_veDTMFPtr->DeRegisterTelephoneEventDetection(_channel) == 0,
- _T("DeRegisterTelephoneEventDetection(channel=%d)"), _channel);
- delete _telephoneEventObserverPtr;
- _telephoneEventObserverPtr = NULL;
- SetDlgItemText(IDC_EDIT_ON_EVENT_INBAND,_T(""));
- SetDlgItemText(IDC_EDIT_ON_EVENT_OUT_OF_BAND,_T(""));
- }
-}
-
-// ============================================================================
-// CWinTestDlg dialog
-// ============================================================================
-
-CWinTestDlg::CWinTestDlg(CWnd* pParent /*=NULL*/)
- : CDialog(CWinTestDlg::IDD, pParent),
- _failCount(0),
- _vePtr(NULL),
- _veBasePtr(NULL),
- _veCodecPtr(NULL),
- _veNetworkPtr(NULL),
- _veFilePtr(NULL),
- _veHardwarePtr(NULL),
- _veExternalMediaPtr(NULL),
- _veApmPtr(NULL),
- _veEncryptionPtr(NULL),
- _veRtpRtcpPtr(NULL),
- _transportPtr(NULL),
- _encryptionPtr(NULL),
- _externalMediaPtr(NULL),
- _externalTransport(false),
- _externalTransportBuild(false),
- _checkPlayFileIn(0),
- _checkPlayFileIn1(0),
- _checkPlayFileIn2(0),
- _checkPlayFileOut1(0),
- _checkPlayFileOut2(0),
- _checkAGC(0),
- _checkAGC1(0),
- _checkNS(0),
- _checkNS1(0),
- _checkEC(0),
- _checkVAD1(0),
- _checkVAD2(0),
- _checkSrtpTx1(0),
- _checkSrtpTx2(0),
- _checkSrtpRx1(0),
- _checkSrtpRx2(0),
- _checkConference1(0),
- _checkConference2(0),
- _checkOnHold1(0),
- _checkOnHold2(0),
- _strComboIp1(_T("")),
- _strComboIp2(_T("")),
- _delayEstimate1(false),
- _delayEstimate2(false),
- _rxVad(false),
- _nErrorCallbacks(0),
- _timerTicks(0)
-{
- m_hIcon = AfxGetApp()->LoadIcon(IDR_MAINFRAME);
-
- _vePtr = VoiceEngine::Create();
-
- VoiceEngine::SetTraceFilter(kTraceNone);
- // VoiceEngine::SetTraceFilter(kTraceAll);
- // VoiceEngine::SetTraceFilter(kTraceStream | kTraceStateInfo | kTraceWarning | kTraceError | kTraceCritical | kTraceApiCall | kTraceModuleCall | kTraceMemory | kTraceDebug | kTraceInfo);
- // VoiceEngine::SetTraceFilter(kTraceStateInfo | kTraceWarning | kTraceError | kTraceCritical | kTraceApiCall | kTraceModuleCall | kTraceMemory | kTraceInfo);
-
- VoiceEngine::SetTraceFile("ve_win_test.txt");
- VoiceEngine::SetTraceCallback(NULL);
-
- if (_vePtr)
- {
- _veExternalMediaPtr = VoEExternalMedia::GetInterface(_vePtr);
- _veVolumeControlPtr = VoEVolumeControl::GetInterface(_vePtr);
- _veEncryptionPtr = VoEEncryption::GetInterface(_vePtr);
- _veVideoSyncPtr = VoEVideoSync::GetInterface(_vePtr);
- _veNetworkPtr = VoENetwork::GetInterface(_vePtr);
- _veFilePtr = VoEFile::GetInterface(_vePtr);
- _veApmPtr = VoEAudioProcessing::GetInterface(_vePtr);
-
- _veBasePtr = VoEBase::GetInterface(_vePtr);
- _veCodecPtr = VoECodec::GetInterface(_vePtr);
- _veHardwarePtr = VoEHardware::GetInterface(_vePtr);
- _veRtpRtcpPtr = VoERTP_RTCP::GetInterface(_vePtr);
- _transportPtr = new MyTransport(_veNetworkPtr);
- _encryptionPtr = new MyEncryption();
- _externalMediaPtr = new MediaProcessImpl();
- _connectionObserverPtr = new ConnectionObserver();
- _rxVadObserverPtr = new RxCallback();
- }
-
- _veBasePtr->RegisterVoiceEngineObserver(*this);
-}
-
-CWinTestDlg::~CWinTestDlg()
-{
- if (_connectionObserverPtr) delete _connectionObserverPtr;
- if (_externalMediaPtr) delete _externalMediaPtr;
- if (_transportPtr) delete _transportPtr;
- if (_encryptionPtr) delete _encryptionPtr;
- if (_rxVadObserverPtr) delete _rxVadObserverPtr;
-
- if (_veExternalMediaPtr) _veExternalMediaPtr->Release();
- if (_veEncryptionPtr) _veEncryptionPtr->Release();
- if (_veVideoSyncPtr) _veVideoSyncPtr->Release();
- if (_veVolumeControlPtr) _veVolumeControlPtr->Release();
-
- if (_veBasePtr) _veBasePtr->Terminate();
- if (_veBasePtr) _veBasePtr->Release();
-
- if (_veCodecPtr) _veCodecPtr->Release();
- if (_veNetworkPtr) _veNetworkPtr->Release();
- if (_veFilePtr) _veFilePtr->Release();
- if (_veHardwarePtr) _veHardwarePtr->Release();
- if (_veApmPtr) _veApmPtr->Release();
- if (_veRtpRtcpPtr) _veRtpRtcpPtr->Release();
- if (_vePtr)
- {
- bool ret = VoiceEngine::Delete(_vePtr);
- ASSERT(ret == true);
- }
- VoiceEngine::SetTraceFilter(kTraceNone);
-}
-
-void CWinTestDlg::DoDataExchange(CDataExchange* pDX)
-{
- CDialog::DoDataExchange(pDX);
- DDX_CBString(pDX, IDC_COMBO_IP_1, _strComboIp1);
- DDX_CBString(pDX, IDC_COMBO_IP_2, _strComboIp2);
-}
-
-BEGIN_MESSAGE_MAP(CWinTestDlg, CDialog)
- ON_WM_SYSCOMMAND()
- ON_WM_PAINT()
- ON_WM_QUERYDRAGICON()
- ON_WM_TIMER()
- //}}AFX_MSG_MAP
- ON_BN_CLICKED(IDC_BUTTON_CREATE_1, &CWinTestDlg::OnBnClickedButtonCreate1)
- ON_BN_CLICKED(IDC_BUTTON_DELETE_1, &CWinTestDlg::OnBnClickedButtonDelete1)
- ON_BN_CLICKED(IDC_BUTTON_CREATE_2, &CWinTestDlg::OnBnClickedButtonCreate2)
- ON_BN_CLICKED(IDC_BUTTON_DELETE_2, &CWinTestDlg::OnBnClickedButtonDelete2)
- ON_CBN_SELCHANGE(IDC_COMBO_CODEC_1, &CWinTestDlg::OnCbnSelchangeComboCodec1)
- ON_BN_CLICKED(IDC_BUTTON_START_LISTEN_1, &CWinTestDlg::OnBnClickedButtonStartListen1)
- ON_BN_CLICKED(IDC_BUTTON_STOP_LISTEN_1, &CWinTestDlg::OnBnClickedButtonStopListen1)
- ON_BN_CLICKED(IDC_BUTTON_START_PLAYOUT_1, &CWinTestDlg::OnBnClickedButtonStartPlayout1)
- ON_BN_CLICKED(IDC_BUTTON_STOP_PLAYOUT_1, &CWinTestDlg::OnBnClickedButtonStopPlayout1)
- ON_BN_CLICKED(IDC_BUTTON_START_SEND_1, &CWinTestDlg::OnBnClickedButtonStartSend1)
- ON_BN_CLICKED(IDC_BUTTON_STOP_SEND_1, &CWinTestDlg::OnBnClickedButtonStopSend1)
- ON_CBN_SELCHANGE(IDC_COMBO_IP_2, &CWinTestDlg::OnCbnSelchangeComboIp2)
- ON_CBN_SELCHANGE(IDC_COMBO_IP_1, &CWinTestDlg::OnCbnSelchangeComboIp1)
- ON_CBN_SELCHANGE(IDC_COMBO_CODEC_2, &CWinTestDlg::OnCbnSelchangeComboCodec2)
- ON_BN_CLICKED(IDC_BUTTON_START_LISTEN_2, &CWinTestDlg::OnBnClickedButtonStartListen2)
- ON_BN_CLICKED(IDC_BUTTON_STOP_LISTEN_2, &CWinTestDlg::OnBnClickedButtonStopListen2)
- ON_BN_CLICKED(IDC_BUTTON_START_PLAYOUT_2, &CWinTestDlg::OnBnClickedButtonStartPlayout2)
- ON_BN_CLICKED(IDC_BUTTON_STOP_PLAYOUT_2, &CWinTestDlg::OnBnClickedButtonStopPlayout2)
- ON_BN_CLICKED(IDC_BUTTON_START_SEND_2, &CWinTestDlg::OnBnClickedButtonStartSend2)
- ON_BN_CLICKED(IDC_BUTTON_STOP_SEND_2, &CWinTestDlg::OnBnClickedButtonStopSend2)
- ON_BN_CLICKED(IDC_CHECK_EXT_TRANS_1, &CWinTestDlg::OnBnClickedCheckExtTrans1)
- ON_BN_CLICKED(IDC_CHECK_PLAY_FILE_IN_1, &CWinTestDlg::OnBnClickedCheckPlayFileIn1)
- ON_BN_CLICKED(IDC_CHECK_PLAY_FILE_OUT_1, &CWinTestDlg::OnBnClickedCheckPlayFileOut1)
- ON_BN_CLICKED(IDC_CHECK_EXT_TRANS_2, &CWinTestDlg::OnBnClickedCheckExtTrans2)
- ON_BN_CLICKED(IDC_CHECK_PLAY_FILE_IN_2, &CWinTestDlg::OnBnClickedCheckPlayFileIn2)
- ON_BN_CLICKED(IDC_CHECK_PLAY_FILE_OUT_2, &CWinTestDlg::OnBnClickedCheckPlayFileOut2)
- ON_BN_CLICKED(IDC_CHECK_PLAY_FILE_IN, &CWinTestDlg::OnBnClickedCheckPlayFileIn)
- ON_CBN_SELCHANGE(IDC_COMBO_REC_DEVICE, &CWinTestDlg::OnCbnSelchangeComboRecDevice)
- ON_CBN_SELCHANGE(IDC_COMBO_PLAY_DEVICE, &CWinTestDlg::OnCbnSelchangeComboPlayDevice)
- ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_IN_1, &CWinTestDlg::OnBnClickedCheckExtMediaIn1)
- ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_OUT_1, &CWinTestDlg::OnBnClickedCheckExtMediaOut1)
- ON_NOTIFY(NM_RELEASEDCAPTURE, IDC_SLIDER_INPUT_VOLUME, &CWinTestDlg::OnNMReleasedcaptureSliderInputVolume)
- ON_NOTIFY(NM_RELEASEDCAPTURE, IDC_SLIDER_OUTPUT_VOLUME, &CWinTestDlg::OnNMReleasedcaptureSliderOutputVolume)
- ON_BN_CLICKED(IDC_CHECK_AGC, &CWinTestDlg::OnBnClickedCheckAgc)
- ON_BN_CLICKED(IDC_CHECK_NS, &CWinTestDlg::OnBnClickedCheckNs)
- ON_BN_CLICKED(IDC_CHECK_EC, &CWinTestDlg::OnBnClickedCheckEc)
- ON_BN_CLICKED(IDC_CHECK_VAD_1, &CWinTestDlg::OnBnClickedCheckVad1)
- ON_BN_CLICKED(IDC_CHECK_VAD_3, &CWinTestDlg::OnBnClickedCheckVad2)
- ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_IN_2, &CWinTestDlg::OnBnClickedCheckExtMediaIn2)
- ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_OUT_2, &CWinTestDlg::OnBnClickedCheckExtMediaOut2)
- ON_BN_CLICKED(IDC_CHECK_MUTE_IN, &CWinTestDlg::OnBnClickedCheckMuteIn)
- ON_BN_CLICKED(IDC_CHECK_MUTE_IN_1, &CWinTestDlg::OnBnClickedCheckMuteIn1)
- ON_BN_CLICKED(IDC_CHECK_MUTE_IN_2, &CWinTestDlg::OnBnClickedCheckMuteIn2)
- ON_BN_CLICKED(IDC_CHECK_SRTP_TX_1, &CWinTestDlg::OnBnClickedCheckSrtpTx1)
- ON_BN_CLICKED(IDC_CHECK_SRTP_RX_1, &CWinTestDlg::OnBnClickedCheckSrtpRx1)
- ON_BN_CLICKED(IDC_CHECK_SRTP_TX_2, &CWinTestDlg::OnBnClickedCheckSrtpTx2)
- ON_BN_CLICKED(IDC_CHECK_SRTP_RX_2, &CWinTestDlg::OnBnClickedCheckSrtpRx2)
- ON_BN_CLICKED(IDC_CHECK_EXT_ENCRYPTION_1, &CWinTestDlg::OnBnClickedCheckExtEncryption1)
- ON_BN_CLICKED(IDC_CHECK_EXT_ENCRYPTION_2, &CWinTestDlg::OnBnClickedCheckExtEncryption2)
- ON_BN_CLICKED(IDC_BUTTON_DTMF_1, &CWinTestDlg::OnBnClickedButtonDtmf1)
- ON_BN_CLICKED(IDC_CHECK_REC_MIC, &CWinTestDlg::OnBnClickedCheckRecMic)
- ON_BN_CLICKED(IDC_BUTTON_DTMF_2, &CWinTestDlg::OnBnClickedButtonDtmf2)
- ON_BN_CLICKED(IDC_BUTTON_TEST_1, &CWinTestDlg::OnBnClickedButtonTest1)
- ON_BN_CLICKED(IDC_CHECK_CONFERENCE_1, &CWinTestDlg::OnBnClickedCheckConference1)
- ON_BN_CLICKED(IDC_CHECK_CONFERENCE_2, &CWinTestDlg::OnBnClickedCheckConference2)
- ON_BN_CLICKED(IDC_CHECK_ON_HOLD_1, &CWinTestDlg::OnBnClickedCheckOnHold1)
- ON_BN_CLICKED(IDC_CHECK_ON_HOLD_2, &CWinTestDlg::OnBnClickedCheckOnHold2)
- ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_IN, &CWinTestDlg::OnBnClickedCheckExtMediaIn)
- ON_BN_CLICKED(IDC_CHECK_EXT_MEDIA_OUT, &CWinTestDlg::OnBnClickedCheckExtMediaOut)
- ON_LBN_SELCHANGE(IDC_LIST_CODEC_1, &CWinTestDlg::OnLbnSelchangeListCodec1)
- ON_NOTIFY(NM_RELEASEDCAPTURE, IDC_SLIDER_PAN_LEFT, &CWinTestDlg::OnNMReleasedcaptureSliderPanLeft)
- ON_NOTIFY(NM_RELEASEDCAPTURE, IDC_SLIDER_PAN_RIGHT, &CWinTestDlg::OnNMReleasedcaptureSliderPanRight)
- ON_BN_CLICKED(IDC_BUTTON_VERSION, &CWinTestDlg::OnBnClickedButtonVersion)
- ON_BN_CLICKED(IDC_CHECK_DELAY_ESTIMATE_1, &CWinTestDlg::OnBnClickedCheckDelayEstimate1)
- ON_BN_CLICKED(IDC_CHECK_RXVAD, &CWinTestDlg::OnBnClickedCheckRxvad)
- ON_BN_CLICKED(IDC_CHECK_AGC_1, &CWinTestDlg::OnBnClickedCheckAgc1)
- ON_BN_CLICKED(IDC_CHECK_NS_1, &CWinTestDlg::OnBnClickedCheckNs1)
- ON_BN_CLICKED(IDC_CHECK_REC_CALL, &CWinTestDlg::OnBnClickedCheckRecCall)
- ON_BN_CLICKED(IDC_CHECK_TYPING_DETECTION, &CWinTestDlg::OnBnClickedCheckTypingDetection)
- ON_BN_CLICKED(IDC_CHECK_FEC, &CWinTestDlg::OnBnClickedCheckFEC)
- ON_BN_CLICKED(IDC_BUTTON_CLEAR_ERROR_CALLBACK, &CWinTestDlg::OnBnClickedButtonClearErrorCallback)
-END_MESSAGE_MAP()
-
-BOOL CWinTestDlg::UpdateTest(bool failed, const CString& strMsg)
-{
- if (failed)
- {
- SetDlgItemText(IDC_EDIT_MESSAGE, strMsg);
- _strErr.Format(_T("FAILED (error=%d)"), _veBasePtr->LastError());
- SetDlgItemText(IDC_EDIT_RESULT, _strErr);
- _failCount++;
- SetDlgItemInt(IDC_EDIT_N_FAILS, _failCount);
- SetDlgItemInt(IDC_EDIT_LAST_ERROR, _veBasePtr->LastError());
- }
- else
- {
- SetDlgItemText(IDC_EDIT_MESSAGE, strMsg);
- SetDlgItemText(IDC_EDIT_RESULT, _T("OK"));
- }
- return TRUE;
-}
-
-
-// CWinTestDlg message handlers
-
-BOOL CWinTestDlg::OnInitDialog()
-{
- CDialog::OnInitDialog();
-
- // Add "About..." menu item to system menu.
-
- // IDM_ABOUTBOX must be in the system command range.
- ASSERT((IDM_ABOUTBOX & 0xFFF0) == IDM_ABOUTBOX);
- ASSERT(IDM_ABOUTBOX < 0xF000);
-
- CMenu* pSysMenu = GetSystemMenu(FALSE);
- if (pSysMenu != NULL)
- {
- CString strAboutMenu;
- strAboutMenu.LoadString(IDS_ABOUTBOX);
- if (!strAboutMenu.IsEmpty())
- {
- pSysMenu->AppendMenu(MF_SEPARATOR);
- pSysMenu->AppendMenu(MF_STRING, IDM_ABOUTBOX, strAboutMenu);
- }
- }
-
- // Set the icon for this dialog. The framework does this automatically
- // when the application's main window is not a dialog
- SetIcon(m_hIcon, TRUE); // Set big icon
- SetIcon(m_hIcon, FALSE); // Set small icon
-
- // char version[1024];
- // _veBasePtr->GetVersion(version);
- // AfxMessageBox(version, MB_OK);
-
- if (_veBasePtr->Init() != 0)
- {
- AfxMessageBox(_T("Init() failed "), MB_OKCANCEL);
- }
-
- int ch = _veBasePtr->CreateChannel();
- if (_veBasePtr->SetSendDestination(ch, 1234, "127.0.0.1") == -1)
- {
- if (_veBasePtr->LastError() == VE_EXTERNAL_TRANSPORT_ENABLED)
- {
- _strMsg.Format(_T("*** External transport build ***"));
- SetDlgItemText(IDC_EDIT_MESSAGE, _strMsg);
- _externalTransportBuild = true;
- }
- }
- _veBasePtr->DeleteChannel(ch);
-
- // --- Add (preferred) local IPv4 address in title
-
- if (_veNetworkPtr)
- {
- char localIP[64];
- _veNetworkPtr->GetLocalIP(localIP);
- CString str;
- GetWindowText(str);
- str.AppendFormat(_T(" [Local IPv4 address: %s]"), CharToTchar(localIP, 64));
- SetWindowText(str);
- }
-
- // --- Volume sliders
-
- if (_veVolumeControlPtr)
- {
- unsigned int volume(0);
- CSliderCtrl* slider(NULL);
-
- slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_INPUT_VOLUME);
- slider->SetRangeMin(0);
- slider->SetRangeMax(255);
- _veVolumeControlPtr->GetMicVolume(volume);
- slider->SetPos(volume);
-
- slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_OUTPUT_VOLUME);
- slider->SetRangeMin(0);
- slider->SetRangeMax(255);
- _veVolumeControlPtr->GetSpeakerVolume(volume);
- slider->SetPos(volume);
- }
-
- // --- Panning sliders
-
- if (_veVolumeControlPtr)
- {
- float lVol(0.0);
- float rVol(0.0);
- int leftVol, rightVol;
- unsigned int volumePan(0);
- CSliderCtrl* slider(NULL);
-
- _veVolumeControlPtr->GetOutputVolumePan(-1, lVol, rVol);
-
- leftVol = (int)(lVol*10.0f); // [0,10]
- rightVol = (int)(rVol*10.0f); // [0,10]
-
- slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_PAN_LEFT);
- slider->SetRange(0,10);
- slider->SetPos(10-leftVol); // pos 0 <=> max pan 1.0 (top of slider)
-
- slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_PAN_RIGHT);
- slider->SetRange(0,10);
- slider->SetPos(10-rightVol);
- }
-
- // --- APM settings
-
- bool enable(false);
- CButton* button(NULL);
-
- AgcModes agcMode(kAgcDefault);
- if (_veApmPtr->GetAgcStatus(enable, agcMode) == 0)
- {
- button = (CButton*)GetDlgItem(IDC_CHECK_AGC);
- enable ? button->SetCheck(BST_CHECKED) : button->SetCheck(BST_UNCHECKED);
- }
- else
- {
- // AGC is not supported
- GetDlgItem(IDC_CHECK_AGC)->EnableWindow(FALSE);
- }
-
- NsModes nsMode(kNsDefault);
- if (_veApmPtr->GetNsStatus(enable, nsMode) == 0)
- {
- button = (CButton*)GetDlgItem(IDC_CHECK_NS);
- enable ? button->SetCheck(BST_CHECKED) : button->SetCheck(BST_UNCHECKED);
- }
- else
- {
- // NS is not supported
- GetDlgItem(IDC_CHECK_NS)->EnableWindow(FALSE);
- }
-
- EcModes ecMode(kEcDefault);
- if (_veApmPtr->GetEcStatus(enable, ecMode) == 0)
- {
- button = (CButton*)GetDlgItem(IDC_CHECK_EC);
- enable ? button->SetCheck(BST_CHECKED) : button->SetCheck(BST_UNCHECKED);
- }
- else
- {
- // EC is not supported
- GetDlgItem(IDC_CHECK_EC)->EnableWindow(FALSE);
- }
-
- // --- First channel section
-
- GetDlgItem(IDC_COMBO_IP_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_EDIT_TX_PORT_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_EDIT_RX_PORT_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_COMBO_CODEC_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_LIST_CODEC_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_EDIT_CODEC_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_DELETE_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_START_LISTEN_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_LISTEN_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_START_PLAYOUT_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_START_SEND_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_SEND_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_EXT_TRANS_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_PLAY_FILE_IN_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_VAD_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_MUTE_IN_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_SRTP_TX_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_SRTP_RX_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_DTMF_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_CONFERENCE_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_ON_HOLD_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_DELAY_ESTIMATE_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_RXVAD)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_AGC_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_NS_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_FEC)->EnableWindow(FALSE);
-
- CComboBox* comboIP(NULL);
- comboIP = (CComboBox*)GetDlgItem(IDC_COMBO_IP_1);
- comboIP->AddString(_T("127.0.0.1"));
- comboIP->SetCurSel(0);
-
- SetDlgItemInt(IDC_EDIT_TX_PORT_1, 1111);
- SetDlgItemInt(IDC_EDIT_RX_PORT_1, 1111);
-
- // --- Add supported codecs to the codec combo box
-
- CComboBox* comboCodec(NULL);
- comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_CODEC_1);
- comboCodec->ResetContent();
-
- int numCodecs = _veCodecPtr->NumOfCodecs();
- for (int idx = 0; idx < numCodecs; idx++)
- {
- CodecInst codec;
- _veCodecPtr->GetCodec(idx, codec);
- if ((_stricmp(codec.plname, "CNNB") != 0) &&
- (_stricmp(codec.plname, "CNWB") != 0))
- {
- CString strCodec;
- if (_stricmp(codec.plname, "G7221") == 0)
- strCodec.Format(_T("%s (%d/%d/%d)"), CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq/1000, codec.rate/1000);
- else
- strCodec.Format(_T("%s (%d/%d)"), CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq/1000);
- comboCodec->AddString(strCodec);
- }
- if (idx == 0)
- {
- SetDlgItemInt(IDC_EDIT_CODEC_1, codec.pltype);
- }
- }
- comboCodec->SetCurSel(0);
-
- CListBox* list = (CListBox*)GetDlgItem(IDC_LIST_CODEC_1);
- list->AddString(_T("pltype"));
- list->AddString(_T("plfreq"));
- list->AddString(_T("pacsize"));
- list->AddString(_T("channels"));
- list->AddString(_T("rate"));
- list->SetCurSel(0);
-
- // --- Add available audio devices to the combo boxes
-
- CComboBox* comboRecDevice(NULL);
- CComboBox* comboPlayDevice(NULL);
- comboRecDevice = (CComboBox*)GetDlgItem(IDC_COMBO_REC_DEVICE);
- comboPlayDevice = (CComboBox*)GetDlgItem(IDC_COMBO_PLAY_DEVICE);
- comboRecDevice->ResetContent();
- comboPlayDevice->ResetContent();
-
- if (_veHardwarePtr)
- {
- int numPlayout(0);
- int numRecording(0);
- char nameStr[128];
- char guidStr[128];
- CString strDevice;
- AudioLayers audioLayer;
-
- _veHardwarePtr->GetAudioDeviceLayer(audioLayer);
- if (kAudioWindowsWave == audioLayer)
- {
- strDevice.FormatMessage(_T("Audio Layer: Windows Wave API"));
- }
- else if (kAudioWindowsCore == audioLayer)
- {
- strDevice.FormatMessage(_T("Audio Layer: Windows Core API"));
- }
- else
- {
- strDevice.FormatMessage(_T("Audio Layer: ** UNKNOWN **"));
- }
- SetDlgItemText(IDC_EDIT_AUDIO_LAYER, (LPCTSTR)strDevice);
-
- _veHardwarePtr->GetNumOfRecordingDevices(numRecording);
-
- for (int idx = 0; idx < numRecording; idx++)
- {
- _veHardwarePtr->GetRecordingDeviceName(idx, nameStr, guidStr);
- strDevice.Format(_T("%s"), CharToTchar(nameStr, 128));
- comboRecDevice->AddString(strDevice);
- }
- // Select default (communication) device in the combo box
- _veHardwarePtr->GetRecordingDeviceName(-1, nameStr, guidStr);
- CString tmp = CString(nameStr);
- int nIndex = comboRecDevice->SelectString(-1, tmp);
- ASSERT(nIndex != CB_ERR);
-
- _veHardwarePtr->GetNumOfPlayoutDevices(numPlayout);
-
- for (int idx = 0; idx < numPlayout; idx++)
- {
- _veHardwarePtr->GetPlayoutDeviceName(idx, nameStr, guidStr);
- strDevice.Format(_T("%s"), CharToTchar(nameStr, 128));
- comboPlayDevice->AddString(strDevice);
- }
- // Select default (communication) device in the combo box
- _veHardwarePtr->GetPlayoutDeviceName(-1, nameStr, guidStr);
- nIndex = comboPlayDevice->SelectString(-1, CString(nameStr));
- ASSERT(nIndex != CB_ERR);
- }
-
- // --- Second channel section
-
- GetDlgItem(IDC_COMBO_IP_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_EDIT_TX_PORT_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_EDIT_RX_PORT_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_COMBO_CODEC_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_DELETE_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_START_LISTEN_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_LISTEN_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_START_PLAYOUT_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_START_SEND_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_SEND_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_EXT_TRANS_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_PLAY_FILE_IN_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_VAD_3)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_MUTE_IN_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_SRTP_TX_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_SRTP_RX_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_DTMF_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_CONFERENCE_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_ON_HOLD_2)->EnableWindow(FALSE);
-
- comboIP = (CComboBox*)GetDlgItem(IDC_COMBO_IP_2);
- comboIP->AddString(_T("127.0.0.1"));
- comboIP->SetCurSel(0);
-
- SetDlgItemInt(IDC_EDIT_TX_PORT_2, 2222);
- SetDlgItemInt(IDC_EDIT_RX_PORT_2, 2222);
-
- comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_CODEC_2);
- comboCodec->ResetContent();
-
- if (_veCodecPtr)
- {
- numCodecs = _veCodecPtr->NumOfCodecs();
- for (int idx = 0; idx < numCodecs; idx++)
- {
- CodecInst codec;
- _veCodecPtr->GetCodec(idx, codec);
- CString strCodec;
- strCodec.Format(_T("%s (%d/%d)"), CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq/1000);
- comboCodec->AddString(strCodec);
- }
- comboCodec->SetCurSel(0);
- }
-
- // --- Start windows timer
-
- SetTimer(0, 1000, NULL);
-
- return TRUE; // return TRUE unless you set the focus to a control
-}
-
-void CWinTestDlg::OnSysCommand(UINT nID, LPARAM lParam)
-{
- if ((nID & 0xFFF0) == IDM_ABOUTBOX)
- {
- CAboutDlg dlgAbout;
- dlgAbout.DoModal();
- }
- else if (nID == SC_CLOSE)
- {
- BOOL ret;
- int channel(0);
- channel = GetDlgItemInt(IDC_EDIT_1, &ret);
- if (ret == TRUE)
- {
- _veBasePtr->DeleteChannel(channel);
- }
- channel = GetDlgItemInt(IDC_EDIT_2, &ret);
- if (ret == TRUE)
- {
- _veBasePtr->DeleteChannel(channel);
- }
-
- CDialog::OnSysCommand(nID, lParam);
- }
- else
- {
- CDialog::OnSysCommand(nID, lParam);
- }
-
-}
-
-// If you add a minimize button to your dialog, you will need the code below
-// to draw the icon. For MFC applications using the document/view model,
-// this is automatically done for you by the framework.
-
-void CWinTestDlg::OnPaint()
-{
- if (IsIconic())
- {
- CPaintDC dc(this); // device context for painting
-
- SendMessage(WM_ICONERASEBKGND, reinterpret_cast<WPARAM>(dc.GetSafeHdc()), 0);
-
- // Center icon in client rectangle
- int cxIcon = GetSystemMetrics(SM_CXICON);
- int cyIcon = GetSystemMetrics(SM_CYICON);
- CRect rect;
- GetClientRect(&rect);
- int x = (rect.Width() - cxIcon + 1) / 2;
- int y = (rect.Height() - cyIcon + 1) / 2;
-
- // Draw the icon
- dc.DrawIcon(x, y, m_hIcon);
- }
- else
- {
- CDialog::OnPaint();
- }
-}
-
-// The system calls this function to obtain the cursor to display while the user drags
-// the minimized window.
-HCURSOR CWinTestDlg::OnQueryDragIcon()
-{
- return static_cast<HCURSOR>(m_hIcon);
-}
-
-
-void CWinTestDlg::OnBnClickedButtonCreate1()
-{
- int channel(0);
- TEST((channel = _veBasePtr->CreateChannel()) >= 0, _T("CreateChannel(channel=%d)"), channel);
- if (channel >= 0)
- {
- _veRtpRtcpPtr->RegisterRTPObserver(channel, *this);
-
- SetDlgItemInt(IDC_EDIT_1, channel);
- GetDlgItem(IDC_BUTTON_CREATE_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_DELETE_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_COMBO_IP_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_EDIT_TX_PORT_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_EDIT_RX_PORT_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_COMBO_CODEC_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_LIST_CODEC_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_EDIT_CODEC_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_BUTTON_START_LISTEN_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_BUTTON_START_PLAYOUT_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_BUTTON_START_SEND_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_EXT_TRANS_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_PLAY_FILE_IN_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_VAD_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_MUTE_IN_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_SRTP_TX_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_SRTP_RX_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_BUTTON_DTMF_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_ON_HOLD_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_DELAY_ESTIMATE_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_RXVAD)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_AGC_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_NS_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_FEC)->EnableWindow(TRUE);
-
- bool enabled(false);
- bool includeCSRCs(false);
-
- // Always set send codec to default codec <=> index 0.
- CodecInst codec;
- _veCodecPtr->GetCodec(0, codec);
- _veCodecPtr->SetSendCodec(channel, codec);
- }
-}
-
-void CWinTestDlg::OnBnClickedButtonCreate2()
-{
- int channel(0);
- TEST((channel = _veBasePtr->CreateChannel()) >=0 , _T("CreateChannel(%d)"), channel);
- if (channel >= 0)
- {
- _veRtpRtcpPtr->RegisterRTPObserver(channel, *this);
-
- SetDlgItemInt(IDC_EDIT_2, channel);
- GetDlgItem(IDC_BUTTON_CREATE_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_DELETE_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_COMBO_IP_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_EDIT_TX_PORT_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_EDIT_RX_PORT_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_COMBO_CODEC_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_BUTTON_START_LISTEN_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_BUTTON_START_PLAYOUT_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_BUTTON_START_SEND_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_EXT_TRANS_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_PLAY_FILE_IN_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_VAD_3)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_MUTE_IN_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_SRTP_TX_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_SRTP_RX_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_BUTTON_DTMF_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_CONFERENCE_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_CHECK_ON_HOLD_2)->EnableWindow(TRUE);
-
- bool enabled(false);
- bool includeCSRCs(false);
-
-
- // Always set send codec to default codec <=> index 0.
- CodecInst codec;
- _veCodecPtr->GetCodec(0, codec);
- _veCodecPtr->SetSendCodec(channel, codec);
- }
-}
-
-void CWinTestDlg::OnBnClickedButtonDelete1()
-{
- BOOL ret;
- int channel = GetDlgItemInt(IDC_EDIT_1, &ret);
- if (ret == TRUE)
- {
- _delayEstimate1 = false;
- _rxVad = false;
- _veRtpRtcpPtr->DeRegisterRTPObserver(channel);
- TEST(_veBasePtr->DeleteChannel(channel) == 0, _T("DeleteChannel(channel=%d)"), channel);
- SetDlgItemText(IDC_EDIT_1, _T(""));
- GetDlgItem(IDC_BUTTON_CREATE_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_BUTTON_DELETE_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_COMBO_IP_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_EDIT_TX_PORT_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_EDIT_RX_PORT_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_COMBO_CODEC_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_LIST_CODEC_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_EDIT_CODEC_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_START_LISTEN_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_START_PLAYOUT_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_START_SEND_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_LISTEN_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_SEND_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_DTMF_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_EXT_TRANS_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_PLAY_FILE_IN_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_VAD_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_MUTE_IN_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_SRTP_TX_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_SRTP_RX_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_CONFERENCE_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_ON_HOLD_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_DELAY_ESTIMATE_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_AGC_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_NS_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_RXVAD)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_FEC)->EnableWindow(FALSE);
- SetDlgItemText(IDC_EDIT_RXVAD, _T(""));
- GetDlgItem(IDC_EDIT_RXVAD)->EnableWindow(FALSE);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_TRANS_1);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_IN_1);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_1);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_1);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_1);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_VAD_1);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_MUTE_IN_1);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_TX_1);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_RX_1);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_1);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_CONFERENCE_1);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_ON_HOLD_1);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_DELAY_ESTIMATE_1);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_AGC_1);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_NS_1);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_RXVAD);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_FEC);
- button->SetCheck(BST_UNCHECKED);
- }
-}
-
-void CWinTestDlg::OnBnClickedButtonDelete2()
-{
- BOOL ret;
- int channel = GetDlgItemInt(IDC_EDIT_2, &ret);
- if (ret == TRUE)
- {
- _delayEstimate2 = false;
- _veRtpRtcpPtr->DeRegisterRTPObserver(channel);
- TEST(_veBasePtr->DeleteChannel(channel) == 0, _T("DeleteChannel(%d)"), channel);
- SetDlgItemText(IDC_EDIT_2, _T(""));
- GetDlgItem(IDC_BUTTON_CREATE_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_BUTTON_DELETE_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_COMBO_IP_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_EDIT_TX_PORT_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_EDIT_RX_PORT_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_COMBO_CODEC_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_START_LISTEN_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_START_PLAYOUT_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_START_SEND_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_LISTEN_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_SEND_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_EXT_TRANS_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_PLAY_FILE_IN_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_MUTE_IN_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_VAD_3)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_SRTP_TX_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_SRTP_RX_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_CONFERENCE_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_DTMF_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_CHECK_ON_HOLD_2)->EnableWindow(FALSE);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_TRANS_2);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_IN_2);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_2);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_2);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_2);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_VAD_3);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_MUTE_IN_2);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_TX_2);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_RX_2);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_2);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_CONFERENCE_2);
- button->SetCheck(BST_UNCHECKED);
- button = (CButton*)GetDlgItem(IDC_CHECK_ON_HOLD_2);
- button->SetCheck(BST_UNCHECKED);
- }
-}
-
-void CWinTestDlg::OnCbnSelchangeComboIp1()
-{
- int channel = GetDlgItemInt(IDC_EDIT_1);
- CString str;
- int port = GetDlgItemInt(IDC_EDIT_TX_PORT_1);
- CComboBox* comboIP = (CComboBox*)GetDlgItem(IDC_COMBO_IP_1);
- int n = comboIP->GetLBTextLen(0);
- comboIP->GetLBText(0, str.GetBuffer(n));
- TEST(_veBasePtr->SetSendDestination(channel, port, TcharToChar(str.GetBuffer(n), -1)) == 0,
- _T("SetSendDestination(channel=%d, port=%d, ip=%s)"), channel, port, str.GetBuffer(n));
- str.ReleaseBuffer();
-}
-
-void CWinTestDlg::OnCbnSelchangeComboIp2()
-{
- int channel = GetDlgItemInt(IDC_EDIT_2);
- CString str;
- int port = GetDlgItemInt(IDC_EDIT_TX_PORT_2);
- CComboBox* comboIP = (CComboBox*)GetDlgItem(IDC_COMBO_IP_2);
- int n = comboIP->GetLBTextLen(0);
- comboIP->GetLBText(0, str.GetBuffer(n));
- TEST(_veBasePtr->SetSendDestination(channel, port, TcharToChar(str.GetBuffer(n), -1)) == 0,
- _T("SetSendDestination(channel=%d, port=%d, ip=%s)"), channel, port, str.GetBuffer(n));
- str.ReleaseBuffer();
-}
-
-void CWinTestDlg::OnCbnSelchangeComboCodec1()
-{
- int channel = GetDlgItemInt(IDC_EDIT_1);
-
- CodecInst codec;
- CComboBox* comboCodec(NULL);
- comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_CODEC_1);
- int index = comboCodec->GetCurSel();
- _veCodecPtr->GetCodec(index, codec);
- if (strncmp(codec.plname, "ISAC", 4) == 0)
- {
- // Set iSAC to adaptive mode by default.
- codec.rate = -1;
- }
- TEST(_veCodecPtr->SetSendCodec(channel, codec) == 0,
- _T("SetSendCodec(channel=%d, plname=%s, pltype=%d, plfreq=%d, rate=%d, pacsize=%d, channels=%d)"),
- channel, CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq, codec.rate, codec.pacsize, codec.channels);
-
- CListBox* list = (CListBox*)GetDlgItem(IDC_LIST_CODEC_1);
- list->SetCurSel(0);
- SetDlgItemInt(IDC_EDIT_CODEC_1, codec.pltype);
-}
-
-void CWinTestDlg::OnLbnSelchangeListCodec1()
-{
- int channel = GetDlgItemInt(IDC_EDIT_1);
-
- CListBox* list = (CListBox*)GetDlgItem(IDC_LIST_CODEC_1);
- int listIdx = list->GetCurSel();
- if (listIdx < 0)
- return;
- CString str;
- list->GetText(listIdx, str);
-
- CodecInst codec;
- _veCodecPtr->GetSendCodec(channel, codec);
-
- int value = GetDlgItemInt(IDC_EDIT_CODEC_1);
- if (str == _T("pltype"))
- {
- codec.pltype = value;
- }
- else if (str == _T("plfreq"))
- {
- codec.plfreq = value;
- }
- else if (str == _T("pacsize"))
- {
- codec.pacsize = value;
- }
- else if (str == _T("channels"))
- {
- codec.channels = value;
- }
- else if (str == _T("rate"))
- {
- codec.rate = value;
- }
- TEST(_veCodecPtr->SetSendCodec(channel, codec) == 0,
- _T("SetSendCodec(channel=%d, plname=%s, pltype=%d, plfreq=%d, rate=%d, pacsize=%d, channels=%d)"),
- channel, CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq, codec.rate, codec.pacsize, codec.channels);
-}
-
-void CWinTestDlg::OnCbnSelchangeComboCodec2()
-{
- int channel = GetDlgItemInt(IDC_EDIT_2);
-
- CodecInst codec;
- CComboBox* comboCodec(NULL);
- comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_CODEC_2);
- int index = comboCodec->GetCurSel();
- _veCodecPtr->GetCodec(index, codec);
- TEST(_veCodecPtr->SetSendCodec(channel, codec) == 0,
- _T("SetSendCodec(channel=%d, plname=%s, pltype=%d, plfreq=%d, rate=%d, pacsize=%d, channels=%d)"),
- channel, CharToTchar(codec.plname, 32), codec.pltype, codec.plfreq, codec.rate, codec.pacsize, codec.channels);
-}
-
-void CWinTestDlg::OnBnClickedButtonStartListen1()
-{
- int ret1(0);
- int ret2(0);
- int channel = GetDlgItemInt(IDC_EDIT_1);
- int port = GetDlgItemInt(IDC_EDIT_RX_PORT_1);
- TEST((ret1 = _veBasePtr->SetLocalReceiver(channel, port)) == 0, _T("SetLocalReceiver(channel=%d, port=%d)"), channel, port);
- TEST((ret2 = _veBasePtr->StartReceive(channel)) == 0, _T("StartReceive(channel=%d)"), channel);
- if (ret1 == 0 && ret2 == 0)
- {
- GetDlgItem(IDC_BUTTON_START_LISTEN_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_LISTEN_1)->EnableWindow(TRUE);
- }
-}
-
-void CWinTestDlg::OnBnClickedButtonStartListen2()
-{
- int ret1(0);
- int ret2(0);
- int channel = GetDlgItemInt(IDC_EDIT_2);
- int port = GetDlgItemInt(IDC_EDIT_RX_PORT_2);
- TEST((ret1 = _veBasePtr->SetLocalReceiver(channel, port)) == 0, _T("SetLocalReceiver(channel=%d, port=%d)"), channel, port);
- TEST((ret2 = _veBasePtr->StartReceive(channel)) == 0, _T("StartReceive(channel=%d)"), channel);
- if (ret1 == 0 && ret2 == 0)
- {
- GetDlgItem(IDC_BUTTON_START_LISTEN_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_LISTEN_2)->EnableWindow(TRUE);
- }
-}
-
-void CWinTestDlg::OnBnClickedButtonStopListen1()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_1);
- TEST((ret = _veBasePtr->StopReceive(channel)) == 0, _T("StopListen(channel=%d)"), channel);
- if (ret == 0)
- {
- GetDlgItem(IDC_BUTTON_START_LISTEN_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_BUTTON_STOP_LISTEN_1)->EnableWindow(FALSE);
- }
-}
-
-void CWinTestDlg::OnBnClickedButtonStopListen2()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_2);
- TEST((ret = _veBasePtr->StopReceive(channel)) == 0, _T("StopListen(channel=%d)"), channel);
- if (ret == 0)
- {
- GetDlgItem(IDC_BUTTON_START_LISTEN_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_BUTTON_STOP_LISTEN_2)->EnableWindow(FALSE);
- }
-}
-
-void CWinTestDlg::OnBnClickedButtonStartPlayout1()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_1);
- TEST((ret = _veBasePtr->StartPlayout(channel)) == 0, _T("StartPlayout(channel=%d)"), channel);
- if (ret == 0)
- {
- GetDlgItem(IDC_BUTTON_START_PLAYOUT_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_1)->EnableWindow(TRUE);
- }
-}
-
-void CWinTestDlg::OnBnClickedButtonStartPlayout2()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_2);
- TEST((ret = _veBasePtr->StartPlayout(channel)) == 0, _T("StartPlayout(channel=%d)"), channel);
- if (ret == 0)
- {
- GetDlgItem(IDC_BUTTON_START_PLAYOUT_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_2)->EnableWindow(TRUE);
- }
-}
-
-void CWinTestDlg::OnBnClickedButtonStopPlayout1()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_1);
- TEST((ret = _veBasePtr->StopPlayout(channel)) == 0, _T("StopPlayout(channel=%d)"), channel);
- if (ret == 0)
- {
- GetDlgItem(IDC_BUTTON_START_PLAYOUT_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_1)->EnableWindow(FALSE);
- }
-}
-
-void CWinTestDlg::OnBnClickedButtonStopPlayout2()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_2);
- TEST((ret = _veBasePtr->StopPlayout(channel)) == 0, _T("StopPlayout(channel=%d)"));
- if (ret == 0)
- {
- GetDlgItem(IDC_BUTTON_START_PLAYOUT_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_BUTTON_STOP_PLAYOUT_2)->EnableWindow(FALSE);
- }
-}
-
-void CWinTestDlg::OnBnClickedButtonStartSend1()
-{
- UpdateData(TRUE); // update IP address
-
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_1);
- if (!_externalTransport)
- {
- CString str;
- int port = GetDlgItemInt(IDC_EDIT_TX_PORT_1);
- TEST(_veBasePtr->SetSendDestination(channel, port, TcharToChar(_strComboIp1.GetBuffer(7), -1)) == 0,
- _T("SetSendDestination(channel=%d, port=%d, ip=%s)"), channel, port, _strComboIp1.GetBuffer(7));
- str.ReleaseBuffer();
- }
-
- //_veVideoSyncPtr->SetInitTimestamp(0,0);
- // OnCbnSelchangeComboCodec1();
-
- TEST((ret = _veBasePtr->StartSend(channel)) == 0, _T("StartSend(channel=%d)"), channel);
- if (ret == 0)
- {
- GetDlgItem(IDC_BUTTON_START_SEND_1)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_SEND_1)->EnableWindow(TRUE);
- }
-}
-
-void CWinTestDlg::OnBnClickedButtonStartSend2()
-{
- UpdateData(TRUE); // update IP address
-
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_2);
- if (!_externalTransport)
- {
- CString str;
- int port = GetDlgItemInt(IDC_EDIT_TX_PORT_2);
- TEST(_veBasePtr->SetSendDestination(channel, port, TcharToChar(_strComboIp2.GetBuffer(7), -1)) == 0,
- _T("SetSendDestination(channel=%d, port=%d, ip=%s)"), channel, port, _strComboIp2.GetBuffer(7));
- str.ReleaseBuffer();
- }
-
- // OnCbnSelchangeComboCodec2();
-
- TEST((ret = _veBasePtr->StartSend(channel)) == 0, _T("StartSend(channel=%d)"), channel);
- if (ret == 0)
- {
- GetDlgItem(IDC_BUTTON_START_SEND_2)->EnableWindow(FALSE);
- GetDlgItem(IDC_BUTTON_STOP_SEND_2)->EnableWindow(TRUE);
- }
-}
-
-void CWinTestDlg::OnBnClickedButtonStopSend1()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_1);
- TEST((ret = _veBasePtr->StopSend(channel)) == 0, _T("StopSend(channel=%d)"), channel);
- if (ret == 0)
- {
- GetDlgItem(IDC_BUTTON_START_SEND_1)->EnableWindow(TRUE);
- GetDlgItem(IDC_BUTTON_STOP_SEND_1)->EnableWindow(FALSE);
- }
-}
-
-void CWinTestDlg::OnBnClickedButtonStopSend2()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_2);
- TEST((ret = _veBasePtr->StopSend(channel)) == 0, _T("StopSend(channel=%d)"), channel);
- if (ret == 0)
- {
- GetDlgItem(IDC_BUTTON_START_SEND_2)->EnableWindow(TRUE);
- GetDlgItem(IDC_BUTTON_STOP_SEND_2)->EnableWindow(FALSE);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckExtTrans1()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_1);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_TRANS_1);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- TEST((ret = _veNetworkPtr->RegisterExternalTransport(channel, *_transportPtr)) == 0,
- _T("RegisterExternalTransport(channel=%d, transport=0x%x)"), channel, _transportPtr);
- }
- else
- {
- TEST((ret = _veNetworkPtr->DeRegisterExternalTransport(channel)) == 0,
- _T("DeRegisterExternalTransport(channel=%d)"), channel);
- }
- if (ret == 0)
- {
- _externalTransport = enable;
- }
- else
- {
- // restore inital state since API call failed
- button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckExtTrans2()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_2);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_TRANS_2);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- TEST((ret = _veNetworkPtr->RegisterExternalTransport(channel, *_transportPtr)) == 0,
- _T("RegisterExternalTransport(channel=%d, transport=0x%x)"), channel, _transportPtr);
- }
- else
- {
- TEST((ret = _veNetworkPtr->DeRegisterExternalTransport(channel)) == 0,
- _T("DeRegisterExternalTransport(channel=%d)"), channel);
- }
- if (ret == 0)
- {
- _externalTransport = enable;
- }
- else
- {
- // restore inital state since API call failed
- button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckPlayFileIn1()
-{
- // File path is relative to the location of 'voice_engine.gyp'.
- const char micFile[] = "../test/data/voice_engine/audio_short16.pcm";
-
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_1);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_IN_1);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- bool mix;
- const bool loop(true);
- const FileFormats format = kFileFormatPcm16kHzFile;
- const float scale(1.0);
-
- (_checkPlayFileIn1 %2 == 0) ? mix = true : mix = false;
- TEST((ret = _veFilePtr->StartPlayingFileAsMicrophone(channel, micFile, loop, mix, format, scale) == 0),
- _T("StartPlayingFileAsMicrophone(channel=%d, file=%s, loop=%d, mix=%d, format=%d, scale=%2.1f)"),
- channel, CharToTchar(micFile, -1), loop, mix, format, scale);
- _checkPlayFileIn1++;
- }
- else
- {
- TEST((ret = _veFilePtr->StopPlayingFileAsMicrophone(channel) == 0),
- _T("StopPlayingFileAsMicrophone(channel=%d)"), channel);
- }
- if (ret == -1)
- {
- // restore inital state since API call failed
- button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckPlayFileIn2()
-{
- // File path is relative to the location of 'voice_engine.gyp'.
- const char micFile[] = "../test/data/voice_engine/audio_long16.pcm";
-
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_2);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_IN_2);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- bool mix;
- const bool loop(true);
- const FileFormats format = kFileFormatPcm16kHzFile;
- const float scale(1.0);
-
- (_checkPlayFileIn2 %2 == 0) ? mix = true : mix = false;
- TEST((ret = _veFilePtr->StartPlayingFileAsMicrophone(channel, micFile, loop, mix, format, scale) == 0),
- _T("StartPlayingFileAsMicrophone(channel=%d, file=%s, loop=%d, mix=%d, format=%d, scale=%2.1f)"),
- channel, CharToTchar(micFile, -1), loop, mix, format, scale);
- _checkPlayFileIn2++;
- }
- else
- {
- TEST((ret = _veFilePtr->StopPlayingFileAsMicrophone(channel) == 0),
- _T("StopPlayingFileAsMicrophone(channel=%d)"), channel);
- }
- if (ret == -1)
- {
- // restore inital state since API call failed
- button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckPlayFileOut1()
-{
- const FileFormats formats[8] = {{kFileFormatPcm16kHzFile},
- {kFileFormatWavFile},
- {kFileFormatWavFile},
- {kFileFormatWavFile},
- {kFileFormatWavFile},
- {kFileFormatWavFile},
- {kFileFormatWavFile},
- {kFileFormatWavFile}};
- // File path is relative to the location of 'voice_engine.gyp'.
- const char spkrFiles[8][64] = {{"../../test/data/voice_engine/audio_short16.pcm"},
- {"../../test/data/voice_engine/audio_tiny8.wav"},
- {"../../test/data/voice_engine/audio_tiny11.wav"},
- {"../../test/data/voice_engine/audio_tiny16.wav"},
- {"../../test/data/voice_engine/audio_tiny22.wav"},
- {"../../test/data/voice_engine/audio_tiny32.wav"},
- {"../../test/data/voice_engine/audio_tiny44.wav"},
- {"../../test/data/voice_engine/audio_tiny48.wav"}};
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_1);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_1);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- const bool loop(true);
- const float volumeScaling(1.0);
- const int startPointMs(0);
- const int stopPointMs(0);
- const FileFormats format = formats[_checkPlayFileOut1 % 8];
- const char* spkrFile = spkrFiles[_checkPlayFileOut1 % 8];
-
- CString str;
- if (_checkPlayFileOut1 % 8 == 0)
- {
- str = _T("kFileFormatPcm16kHzFile");
- }
- else
- {
- str = _T("kFileFormatWavFile");
- }
- // (_checkPlayFileOut1 %2 == 0) ? mix = true : mix = false;
- TEST((ret = _veFilePtr->StartPlayingFileLocally(channel, spkrFile, loop, format, volumeScaling, startPointMs,stopPointMs) == 0),
- _T("StartPlayingFileLocally(channel=%d, file=%s, loop=%d, format=%s, scale=%2.1f, start=%d, stop=%d)"),
- channel, CharToTchar(spkrFile, -1), loop, str, volumeScaling, startPointMs, stopPointMs);
- _checkPlayFileOut1++;
- }
- else
- {
- TEST((ret = _veFilePtr->StopPlayingFileLocally(channel) == 0),
- _T("StopPlayingFileLocally(channel=%d)"), channel);
- }
- if (ret == -1)
- {
- // restore inital state since API call failed
- button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckPlayFileOut2()
-{
- // File path is relative to the location of 'voice_engine.gyp'.
- const char spkrFile[] = "../test/data/voice_engine/audio_long16.pcm";
-
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_2);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_OUT_2);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- const bool loop(true);
- const FileFormats format = kFileFormatPcm16kHzFile;
- const float volumeScaling(1.0);
- const int startPointMs(0);
- const int stopPointMs(0);
-
- // (_checkPlayFileOut2 %2 == 0) ? mix = true : mix = false;
- TEST((ret = _veFilePtr->StartPlayingFileLocally(channel, spkrFile, loop, format, volumeScaling, startPointMs,stopPointMs) == 0),
- _T("StartPlayingFileLocally(channel=%d, file=%s, loop=%d, format=%d, scale=%2.1f, start=%d, stop=%d)"),
- channel, CharToTchar(spkrFile, -1), loop, format, volumeScaling, startPointMs, stopPointMs);
- // _checkPlayFileIn2++;
- }
- else
- {
- TEST((ret = _veFilePtr->StopPlayingFileLocally(channel) == 0),
- _T("StopPlayingFileLocally(channel=%d)"), channel);
- }
- if (ret == -1)
- {
- // restore inital state since API call failed
- button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckExtMediaIn1()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_1);
- CButton* buttonExtTrans = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_1);
- int check = buttonExtTrans->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kRecordingPerChannel, *_externalMediaPtr) == 0,
- _T("RegisterExternalMediaProcessing(channel=%d, kRecordingPerChannel, processObject=0x%x)"), channel, _externalMediaPtr);
- }
- else
- {
- TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kRecordingPerChannel) == 0,
- _T("DeRegisterExternalMediaProcessing(channel=%d, kRecordingPerChannel)"), channel);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckExtMediaIn2()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_2);
- CButton* buttonExtTrans = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_IN_2);
- int check = buttonExtTrans->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kRecordingPerChannel, *_externalMediaPtr) == 0,
- _T("RegisterExternalMediaProcessing(channel=%d, kRecordingPerChannel, processObject=0x%x)"), channel, _externalMediaPtr);
- }
- else
- {
- TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kRecordingPerChannel) == 0,
- _T("DeRegisterExternalMediaProcessing(channel=%d, kRecordingPerChannel)"), channel);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckExtMediaOut1()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_1);
- CButton* buttonExtTrans = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_1);
- int check = buttonExtTrans->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kPlaybackPerChannel, *_externalMediaPtr) == 0,
- _T("RegisterExternalMediaProcessing(channel=%d, kPlaybackPerChannel, processObject=0x%x)"), channel, _externalMediaPtr);
- }
- else
- {
- TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kPlaybackPerChannel) == 0,
- _T("DeRegisterExternalMediaProcessing(channel=%d, kPlaybackPerChannel)"), channel);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckExtMediaOut2()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_2);
- CButton* buttonExtTrans = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT_2);
- int check = buttonExtTrans->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kPlaybackPerChannel, *_externalMediaPtr) == 0,
- _T("RegisterExternalMediaProcessing(channel=%d, kPlaybackPerChannel, processObject=0x%x)"), channel, _externalMediaPtr);
- }
- else
- {
- TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kPlaybackPerChannel) == 0,
- _T("DeRegisterExternalMediaProcessing(channel=%d, kPlaybackPerChannel)"), channel);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckVad1()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_1);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_VAD_1);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- CString str;
- VadModes mode(kVadConventional);
- if (_checkVAD1 % 4 == 0)
- {
- mode = kVadConventional;
- str = _T("kVadConventional");
- }
- else if (_checkVAD1 % 4 == 1)
- {
- mode = kVadAggressiveLow;
- str = _T("kVadAggressiveLow");
- }
- else if (_checkVAD1 % 4 == 2)
- {
- mode = kVadAggressiveMid;
- str = _T("kVadAggressiveMid");
- }
- else if (_checkVAD1 % 4 == 3)
- {
- mode = kVadAggressiveHigh;
- str = _T("kVadAggressiveHigh");
- }
- const bool disableDTX(false);
- TEST((ret = _veCodecPtr->SetVADStatus(channel, true, mode, disableDTX) == 0),
- _T("SetVADStatus(channel=%d, enable=%d, mode=%s, disableDTX=%d)"), channel, enable, str, disableDTX);
- _checkVAD1++;
- }
- else
- {
- TEST((ret = _veCodecPtr->SetVADStatus(channel, false)) == 0, _T("SetVADStatus(channel=%d, enable=%d)"), channel, false);
- }
- if (ret == -1)
- {
- // restore inital state since API call failed
- button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckVad2()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_2);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_VAD_2);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- CString str;
- VadModes mode(kVadConventional);
- if (_checkVAD2 % 4 == 0)
- {
- mode = kVadConventional;
- str = _T("kVadConventional");
- }
- else if (_checkVAD2 % 4 == 1)
- {
- mode = kVadAggressiveLow;
- str = _T("kVadAggressiveLow");
- }
- else if (_checkVAD2 % 4 == 2)
- {
- mode = kVadAggressiveMid;
- str = _T("kVadAggressiveMid");
- }
- else if (_checkVAD2 % 4 == 3)
- {
- mode = kVadAggressiveHigh;
- str = _T("kVadAggressiveHigh");
- }
- const bool disableDTX(false);
- TEST((ret = _veCodecPtr->SetVADStatus(channel, true, mode, disableDTX)) == 0,
- _T("SetVADStatus(channel=%d, enable=%d, mode=%s, disableDTX=%d)"), channel, enable, str, disableDTX);
- _checkVAD2++;
- }
- else
- {
- TEST((ret = _veCodecPtr->SetVADStatus(channel, false) == 0), _T("SetVADStatus(channel=%d, enable=%d)"), channel, false);
- }
- if (ret == -1)
- {
- // restore inital state since API call failed
- button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckMuteIn1()
-{
- int channel = GetDlgItemInt(IDC_EDIT_1);
- CButton* buttonMute = (CButton*)GetDlgItem(IDC_CHECK_MUTE_IN_1);
- int check = buttonMute->GetCheck();
- const bool enable = (check == BST_CHECKED);
- TEST(_veVolumeControlPtr->SetInputMute(channel, enable) == 0,
- _T("SetInputMute(channel=%d, enable=%d)"), channel, enable);
-}
-
-void CWinTestDlg::OnBnClickedCheckMuteIn2()
-{
- int channel = GetDlgItemInt(IDC_EDIT_2);
- CButton* buttonMute = (CButton*)GetDlgItem(IDC_CHECK_MUTE_IN_2);
- int check = buttonMute->GetCheck();
- const bool enable = (check == BST_CHECKED);
- TEST(_veVolumeControlPtr->SetInputMute(channel, enable) == 0,
- _T("SetInputMute(channel=%d, enable=%d)"), channel, enable);
-}
-
-void CWinTestDlg::OnBnClickedCheckSrtpTx1()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_1);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_TX_1);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- bool useForRTCP = false;
- if (enable)
- {
- (_checkSrtpTx1++ %2 == 0) ? useForRTCP = false : useForRTCP = true;
- TEST((ret = _veEncryptionPtr->EnableSRTPSend(channel,
- kCipherAes128CounterMode, 30, kAuthHmacSha1, 20, 4, kEncryptionAndAuthentication, key, useForRTCP)) == 0,
- _T("EnableSRTPSend(channel=%d, kCipherAes128CounterMode, 30, kAuthHmacSha1, 20, 4, kEncryptionAndAuthentication, key, useForRTCP=%d)"),
- channel, useForRTCP);
- }
- else
- {
- TEST((ret = _veEncryptionPtr->DisableSRTPSend(channel) == 0), _T("DisableSRTPSend(channel=%d)"), channel);
- }
- if (ret == -1)
- {
- // restore inital state since API call failed
- button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckSrtpTx2()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_2);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_TX_2);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- bool useForRTCP = false;
- if (enable)
- {
- (_checkSrtpTx2++ %2 == 0) ? useForRTCP = false : useForRTCP = true;
- TEST((ret = _veEncryptionPtr->EnableSRTPSend(channel,
- kCipherAes128CounterMode, 30, kAuthHmacSha1, 20, 4, kEncryptionAndAuthentication, key, useForRTCP)) == 0,
- _T("EnableSRTPSend(channel=%d, kCipherAes128CounterMode, 30, kAuthHmacSha1, 20, 4, kEncryptionAndAuthentication, key, useForRTCP=%d)"),
- channel, useForRTCP);
- }
- else
- {
- TEST((ret = _veEncryptionPtr->DisableSRTPSend(channel) == 0), _T("DisableSRTPSend(channel=%d)"), channel);
- }
- if (ret == -1)
- {
- // restore inital state since API call failed
- button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckSrtpRx1()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_1);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_RX_1);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- bool useForRTCP(false);
- if (enable)
- {
- (_checkSrtpRx1++ %2 == 0) ? useForRTCP = false : useForRTCP = true;
- TEST((ret = _veEncryptionPtr->EnableSRTPReceive(channel,
- kCipherAes128CounterMode, 30, kAuthHmacSha1, 20, 4, kEncryptionAndAuthentication, key, useForRTCP)) == 0,
- _T("EnableSRTPReceive(channel=%d, kCipherAes128CounterMode, 30, kAuthHmacSha1, 20, 4, kEncryptionAndAuthentication, key, useForRTCP=%d)"),
- channel, useForRTCP);
- }
- else
- {
- TEST((ret = _veEncryptionPtr->DisableSRTPReceive(channel) == 0), _T("DisableSRTPReceive(channel=%d)"), channel);
- }
- if (ret == -1)
- {
- // restore inital state since API call failed
- button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckSrtpRx2()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_2);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_SRTP_RX_2);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- bool useForRTCP(false);
- if (enable)
- {
- (_checkSrtpRx2++ %2 == 0) ? useForRTCP = false : useForRTCP = true;
- TEST((ret = _veEncryptionPtr->EnableSRTPReceive(channel,
- kCipherAes128CounterMode, 30, kAuthHmacSha1, 20, 4, kEncryptionAndAuthentication, key, useForRTCP)) == 0,
- _T("EnableSRTPReceive(channel=%d, kCipherAes128CounterMode, 30, kAuthHmacSha1, 20, 4, kEncryptionAndAuthentication, key, useForRTCP=%d)"),
- channel, useForRTCP);
- }
- else
- {
- TEST((ret = _veEncryptionPtr->DisableSRTPReceive(channel)) == 0, _T("DisableSRTPReceive(channel=%d)"), channel);
- }
- if (ret == -1)
- {
- // restore inital state since API call failed
- button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckExtEncryption1()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_1);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_1);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- TEST((ret = _veEncryptionPtr->RegisterExternalEncryption(channel, *_encryptionPtr)) == 0,
- _T("RegisterExternalEncryption(channel=%d, encryption=0x%x)"), channel, _encryptionPtr);
- }
- else
- {
- TEST((ret = _veEncryptionPtr->DeRegisterExternalEncryption(channel)) == 0,
- _T("DeRegisterExternalEncryption(channel=%d)"), channel);
- }
- if (ret == -1)
- {
- // restore inital state since API call failed
- button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckExtEncryption2()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_2);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_ENCRYPTION_2);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- TEST((ret = _veEncryptionPtr->RegisterExternalEncryption(channel, *_encryptionPtr)) == 0,
- _T("RegisterExternalEncryption(channel=%d, encryption=0x%x)"), channel, _encryptionPtr);
- }
- else
- {
- TEST((ret = _veEncryptionPtr->DeRegisterExternalEncryption(channel)) == 0,
- _T("DeRegisterExternalEncryption(channel=%d)"), channel);
- }
- if (ret == -1)
- {
- // restore inital state since API call failed
- button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
- }
-}
-
-void CWinTestDlg::OnBnClickedButtonDtmf1()
-{
- int channel = GetDlgItemInt(IDC_EDIT_1);
- CTelephonyEvent dlgTelephoneEvent(_vePtr, channel, this);
- dlgTelephoneEvent.DoModal();
-}
-
-void CWinTestDlg::OnBnClickedButtonDtmf2()
-{
- int channel = GetDlgItemInt(IDC_EDIT_2);
- CTelephonyEvent dlgTelephoneEvent(_vePtr, channel, this);
- dlgTelephoneEvent.DoModal();
-}
-
-void CWinTestDlg::OnBnClickedCheckConference1()
-{
- // Not supported yet
-}
-
-void CWinTestDlg::OnBnClickedCheckConference2()
-{
- // Not supported yet
-}
-
-void CWinTestDlg::OnBnClickedCheckOnHold1()
-{
- SHORT shiftKeyIsPressed = ::GetAsyncKeyState(VK_SHIFT);
-
- CString str;
- int channel = GetDlgItemInt(IDC_EDIT_1);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_ON_HOLD_1);
- int check = button->GetCheck();
-
- if (shiftKeyIsPressed)
- {
- bool enabled(false);
- OnHoldModes mode(kHoldSendAndPlay);
- TEST(_veBasePtr->GetOnHoldStatus(channel, enabled, mode) == 0,
- _T("GetOnHoldStatus(channel=%d, enabled=?, mode=?)"), channel);
- button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
-
- switch (mode)
- {
- case kHoldSendAndPlay:
- str = _T("kHoldSendAndPlay");
- break;
- case kHoldSendOnly:
- str = _T("kHoldSendOnly");
- break;
- case kHoldPlayOnly:
- str = _T("kHoldPlayOnly");
- break;
- default:
- break;
- }
- PRINT_GET_RESULT(_T("enabled=%d, mode=%s"), enabled, str);
- return;
- }
-
- int ret(0);
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- OnHoldModes mode(kHoldSendAndPlay);
- if (_checkOnHold1 % 3 == 0)
- {
- mode = kHoldSendAndPlay;
- str = _T("kHoldSendAndPlay");
- }
- else if (_checkOnHold1 % 3 == 1)
- {
- mode = kHoldSendOnly;
- str = _T("kHoldSendOnly");
- }
- else if (_checkOnHold1 % 3 == 2)
- {
- mode = kHoldPlayOnly;
- str = _T("kHoldPlayOnly");
- }
- TEST((ret = _veBasePtr->SetOnHoldStatus(channel, enable, mode)) == 0,
- _T("SetOnHoldStatus(channel=%d, enable=%d, mode=%s)"), channel, enable, str);
- _checkOnHold1++;
- }
- else
- {
- TEST((ret = _veBasePtr->SetOnHoldStatus(channel, enable)) == 0,
- _T("SetOnHoldStatus(channel=%d, enable=%d)"), channel, enable);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckOnHold2()
-{
- int ret(0);
- int channel = GetDlgItemInt(IDC_EDIT_2);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_ON_HOLD_2);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- CString str;
- OnHoldModes mode(kHoldSendAndPlay);
- if (_checkOnHold1 % 3 == 0)
- {
- mode = kHoldSendAndPlay;
- str = _T("kHoldSendAndPlay");
- }
- else if (_checkOnHold1 % 3 == 1)
- {
- mode = kHoldSendOnly;
- str = _T("kHoldSendOnly");
- }
- else if (_checkOnHold1 % 3 == 2)
- {
- mode = kHoldPlayOnly;
- str = _T("kHoldPlayOnly");
- }
- TEST((ret = _veBasePtr->SetOnHoldStatus(channel, enable, mode)) == 0,
- _T("SetOnHoldStatus(channel=%d, enable=%d, mode=%s)"), channel, enable, str);
- _checkOnHold1++;
- }
- else
- {
- TEST((ret = _veBasePtr->SetOnHoldStatus(channel, enable)) == 0,
- _T("SetOnHoldStatus(channel=%d, enable=%d)"), channel, enable);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckDelayEstimate1()
-{
- int channel = GetDlgItemInt(IDC_EDIT_1);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_DELAY_ESTIMATE_1);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
-
- if (enable)
- {
- _delayEstimate1 = true;
- SetDlgItemInt(IDC_EDIT_DELAY_ESTIMATE_1, 0);
- }
- else
- {
- _delayEstimate1 = false;
- SetDlgItemText(IDC_EDIT_DELAY_ESTIMATE_1, _T(""));
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckRxvad()
-{
- int channel = GetDlgItemInt(IDC_EDIT_1);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_RXVAD);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
-
- if (enable)
- {
- _rxVad = true;
- _veApmPtr->RegisterRxVadObserver(channel, *_rxVadObserverPtr);
- SetDlgItemInt(IDC_EDIT_RXVAD, 0);
- }
- else
- {
- _rxVad = false;
- _veApmPtr->DeRegisterRxVadObserver(channel);
- SetDlgItemText(IDC_EDIT_RXVAD, _T(""));
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckAgc1()
-{
- SHORT shiftKeyIsPressed = ::GetAsyncKeyState(VK_SHIFT);
-
- CString str;
- int channel = GetDlgItemInt(IDC_EDIT_1);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_AGC_1);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
-
- if (shiftKeyIsPressed)
- {
- bool enabled(false);
- AgcModes mode(kAgcAdaptiveDigital);
- TEST(_veApmPtr->GetRxAgcStatus(channel, enabled, mode) == 0,
- _T("GetRxAgcStatus(channel=%d, enabled=?, mode=?)"), channel);
- button->SetCheck((check == BST_CHECKED) ? BST_UNCHECKED : BST_CHECKED);
-
- switch (mode)
- {
- case kAgcAdaptiveAnalog:
- str = _T("kAgcAdaptiveAnalog");
- break;
- case kAgcAdaptiveDigital:
- str = _T("kAgcAdaptiveDigital");
- break;
- case kAgcFixedDigital:
- str = _T("kAgcFixedDigital");
- break;
- default:
- break;
- }
- PRINT_GET_RESULT(_T("enabled=%d, mode=%s"), enabled, str);
- return;
- }
-
- if (enable)
- {
- CString str;
- AgcModes mode(kAgcDefault);
- if (_checkAGC1 % 3 == 0)
- {
- mode = kAgcDefault;
- str = _T("kAgcDefault");
- }
- else if (_checkAGC1 % 3 == 1)
- {
- mode = kAgcAdaptiveDigital;
- str = _T("kAgcAdaptiveDigital");
- }
- else if (_checkAGC1 % 3 == 2)
- {
- mode = kAgcFixedDigital;
- str = _T("kAgcFixedDigital");
- }
- TEST(_veApmPtr->SetRxAgcStatus(channel, true, mode) == 0, _T("SetRxAgcStatus(channel=%d, enable=%d, %s)"), channel, enable, str);
- _checkAGC1++;
- }
- else
- {
- TEST(_veApmPtr->SetRxAgcStatus(channel, false, kAgcUnchanged) == 0, _T("SetRxAgcStatus(channel=%d, enable=%d)"), channel, enable);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckNs1()
-{
- int channel = GetDlgItemInt(IDC_EDIT_1);
- CButton* buttonNS = (CButton*)GetDlgItem(IDC_CHECK_NS_1);
- int check = buttonNS->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- CString str;
- NsModes mode(kNsDefault);
- if (_checkNS1 % 6 == 0)
- {
- mode = kNsDefault;
- str = _T("kNsDefault");
- }
- else if (_checkNS1 % 6 == 1)
- {
- mode = kNsConference;
- str = _T("kNsConference");
- }
- else if (_checkNS1 % 6 == 2)
- {
- mode = kNsLowSuppression;
- str = _T("kNsLowSuppression");
- }
- else if (_checkNS1 % 6 == 3)
- {
- mode = kNsModerateSuppression;
- str = _T("kNsModerateSuppression");
- }
- else if (_checkNS1 % 6 == 4)
- {
- mode = kNsHighSuppression;
- str = _T("kNsHighSuppression");
- }
- else if (_checkNS1 % 6 == 5)
- {
- mode = kNsVeryHighSuppression;
- str = _T("kNsVeryHighSuppression");
- }
- TEST(_veApmPtr->SetRxNsStatus(channel, true, mode) == 0, _T("SetRxNsStatus(channel=%d, enable=%d, %s)"), channel, enable, str);
- _checkNS1++;
- }
- else
- {
- TEST(_veApmPtr->SetRxNsStatus(channel, false, kNsUnchanged) == 0, _T("SetRxNsStatus(channel=%d, enable=%d)"), enable, channel);
- }
-}
-
-// ----------------------------------------------------------------------------
-// Channel-independent Operations
-// ----------------------------------------------------------------------------
-
-void CWinTestDlg::OnBnClickedCheckPlayFileIn()
-{
- // File path is relative to the location of 'voice_engine.gyp'.
- const char micFile[] = "../../test/data/voice_engine/audio_short16.pcm";
- // const char micFile[] = "../../test/data/voice_engine/audio_long16noise.pcm";
-
- int ret(0);
- int channel(-1);
- CButton* buttonExtTrans = (CButton*)GetDlgItem(IDC_CHECK_PLAY_FILE_IN);
- int check = buttonExtTrans->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- bool mix;
- const bool loop(true);
- const FileFormats format = kFileFormatPcm16kHzFile;
- const float scale(1.0);
-
- (_checkPlayFileIn %2 == 0) ? mix = true : mix = false;
- TEST(_veFilePtr->StartPlayingFileAsMicrophone(channel, micFile, loop, mix, format, scale) == 0,
- _T("StartPlayingFileAsMicrophone(channel=%d, file=%s, loop=%d, mix=%d, format=%d, scale=%2.1f)"),
- channel, CharToTchar(micFile, -1), loop, mix, format, scale);
- _checkPlayFileIn++;
- }
- else
- {
- TEST(_veFilePtr->StopPlayingFileAsMicrophone(channel) == 0,
- _T("StopPlayingFileAsMicrophone(channel=%d)"), channel);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckRecMic()
-{
- const char micFile[] = "/tmp/rec_mic_mono_16kHz.pcm";
-
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_REC_MIC);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- TEST(_veFilePtr->StartRecordingMicrophone(micFile, NULL) == 0, _T("StartRecordingMicrophone(file=%s)"), CharToTchar(micFile, -1));
- }
- else
- {
- TEST(_veFilePtr->StopRecordingMicrophone() == 0, _T("StopRecordingMicrophone()"));
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckAgc()
-{
- CButton* buttonAGC = (CButton*)GetDlgItem(IDC_CHECK_AGC);
- int check = buttonAGC->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- CString str;
- AgcModes mode(kAgcDefault);
- if (_checkAGC % 4 == 0)
- {
- mode = kAgcDefault;
- str = _T("kAgcDefault");
- }
- else if (_checkAGC % 4 == 1)
- {
- mode = kAgcAdaptiveAnalog;
- str = _T("kAgcAdaptiveAnalog");
- }
- else if (_checkAGC % 4 == 2)
- {
- mode = kAgcAdaptiveDigital;
- str = _T("kAgcAdaptiveDigital");
- }
- else if (_checkAGC % 4 == 3)
- {
- mode = kAgcFixedDigital;
- str = _T("kAgcFixedDigital");
- }
- TEST(_veApmPtr->SetAgcStatus(true, mode) == 0, _T("SetAgcStatus(enable=%d, %s)"), enable, str);
- _checkAGC++;
- }
- else
- {
- TEST(_veApmPtr->SetAgcStatus(false, kAgcUnchanged) == 0, _T("SetAgcStatus(enable=%d)"), enable);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckNs()
-{
- CButton* buttonNS = (CButton*)GetDlgItem(IDC_CHECK_NS);
- int check = buttonNS->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- CString str;
- NsModes mode(kNsDefault);
- if (_checkNS % 6 == 0)
- {
- mode = kNsDefault;
- str = _T("kNsDefault");
- }
- else if (_checkNS % 6 == 1)
- {
- mode = kNsConference;
- str = _T("kNsConference");
- }
- else if (_checkNS % 6 == 2)
- {
- mode = kNsLowSuppression;
- str = _T("kNsLowSuppression");
- }
- else if (_checkNS % 6 == 3)
- {
- mode = kNsModerateSuppression;
- str = _T("kNsModerateSuppression");
- }
- else if (_checkNS % 6 == 4)
- {
- mode = kNsHighSuppression;
- str = _T("kNsHighSuppression");
- }
- else if (_checkNS % 6 == 5)
- {
- mode = kNsVeryHighSuppression;
- str = _T("kNsVeryHighSuppression");
- }
- TEST(_veApmPtr->SetNsStatus(true, mode) == 0, _T("SetNsStatus(enable=%d, %s)"), enable, str);
- _checkNS++;
- }
- else
- {
- TEST(_veApmPtr->SetNsStatus(false, kNsUnchanged) == 0, _T("SetNsStatus(enable=%d)"), enable);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckEc()
-{
- CButton* buttonEC = (CButton*)GetDlgItem(IDC_CHECK_EC);
- int check = buttonEC->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- CString str;
- EcModes mode(kEcDefault);
- if (_checkEC % 4 == 0)
- {
- mode = kEcDefault;
- str = _T("kEcDefault");
- }
- else if (_checkEC % 4 == 1)
- {
- mode = kEcConference;
- str = _T("kEcConference");
- }
- else if (_checkEC % 4 == 2)
- {
- mode = kEcAec;
- str = _T("kEcAec");
- }
- else if (_checkEC % 4 == 3)
- {
- mode = kEcAecm;
- str = _T("kEcAecm");
- }
- TEST(_veApmPtr->SetEcStatus(true, mode) == 0, _T("SetEcStatus(enable=%d, %s)"), enable, str);
- _checkEC++;
- }
- else
- {
- TEST(_veApmPtr->SetEcStatus(false, kEcUnchanged) == 0, _T("SetEcStatus(enable=%d)"), enable);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckMuteIn()
-{
- CButton* buttonMute = (CButton*)GetDlgItem(IDC_CHECK_MUTE_IN);
- int check = buttonMute->GetCheck();
- const bool enable = (check == BST_CHECKED);
- const int channel(-1);
- TEST(_veVolumeControlPtr->SetInputMute(channel, enable) == 0,
- _T("SetInputMute(channel=%d, enable=%d)"), channel, enable);
-}
-
-void CWinTestDlg::OnBnClickedCheckExtMediaIn()
-{
- const int channel(-1);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_IN);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kRecordingAllChannelsMixed, *_externalMediaPtr) == 0,
- _T("RegisterExternalMediaProcessing(channel=%d, kRecordingAllChannelsMixed, processObject=0x%x)"), channel, _externalMediaPtr);
- }
- else
- {
- TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kRecordingAllChannelsMixed) == 0,
- _T("DeRegisterExternalMediaProcessing(channel=%d, kRecordingAllChannelsMixed)"), channel);
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckExtMediaOut()
-{
- const int channel(-1);
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_EXT_MEDIA_OUT);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- if (enable)
- {
- TEST(_veExternalMediaPtr->RegisterExternalMediaProcessing(channel, kPlaybackAllChannelsMixed, *_externalMediaPtr) == 0,
- _T("RegisterExternalMediaProcessing(channel=%d, kPlaybackAllChannelsMixed, processObject=0x%x)"), channel, _externalMediaPtr);
- }
- else
- {
- TEST(_veExternalMediaPtr->DeRegisterExternalMediaProcessing(channel, kPlaybackAllChannelsMixed) == 0,
- _T("DeRegisterExternalMediaProcessing(channel=%d, kPlaybackAllChannelsMixed)"), channel);
- }
-}
-
-void CWinTestDlg::OnCbnSelchangeComboRecDevice()
-{
- CComboBox* comboCodec(NULL);
- comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_REC_DEVICE);
- int index = comboCodec->GetCurSel();
- TEST(_veHardwarePtr->SetRecordingDevice(index) == 0,
- _T("SetRecordingDevice(index=%d)"), index);
-}
-
-void CWinTestDlg::OnCbnSelchangeComboPlayDevice()
-{
- CComboBox* comboCodec(NULL);
- comboCodec = (CComboBox*)GetDlgItem(IDC_COMBO_PLAY_DEVICE);
- int index = comboCodec->GetCurSel();
- TEST(_veHardwarePtr->SetPlayoutDevice(index) == 0,
- _T("SetPlayoutDevice(index=%d)"), index);
-}
-
-void CWinTestDlg::OnNMReleasedcaptureSliderInputVolume(NMHDR *pNMHDR, LRESULT *pResult)
-{
- CSliderCtrl* slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_INPUT_VOLUME);
- slider->SetRangeMin(0);
- slider->SetRangeMax(255);
- int pos = slider->GetPos();
-
- TEST(_veVolumeControlPtr->SetMicVolume(pos) == 0, _T("SetMicVolume(volume=%d)"), pos);
-
- *pResult = 0;
-}
-
-void CWinTestDlg::OnNMReleasedcaptureSliderOutputVolume(NMHDR *pNMHDR, LRESULT *pResult)
-{
- CSliderCtrl* slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_OUTPUT_VOLUME);
- slider->SetRangeMin(0);
- slider->SetRangeMax(255);
- int pos = slider->GetPos();
-
- TEST(_veVolumeControlPtr->SetSpeakerVolume(pos) == 0, _T("SetSpeakerVolume(volume=%d)"), pos);
-
- *pResult = 0;
-}
-
-void CWinTestDlg::OnNMReleasedcaptureSliderPanLeft(NMHDR *pNMHDR, LRESULT *pResult)
-{
- CSliderCtrl* slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_PAN_LEFT);
- slider->SetRange(0,10);
- int pos = 10 - slider->GetPos(); // 0 <=> lower end, 10 <=> upper end
-
- float left(0.0);
- float right(0.0);
- const int channel(-1);
-
- // Only left channel will be modified
- _veVolumeControlPtr->GetOutputVolumePan(channel, left, right);
-
- left = (float)((float)pos/10.0f);
-
- TEST(_veVolumeControlPtr->SetOutputVolumePan(channel, left, right) == 0,
- _T("SetOutputVolumePan(channel=%d, left=%2.1f, right=%2.1f)"), channel, left, right);
-
- *pResult = 0;
-}
-
-void CWinTestDlg::OnNMReleasedcaptureSliderPanRight(NMHDR *pNMHDR, LRESULT *pResult)
-{
- CSliderCtrl* slider = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_PAN_RIGHT);
- slider->SetRange(0,10);
- int pos = 10 - slider->GetPos(); // 0 <=> lower end, 10 <=> upper end
-
- float left(0.0);
- float right(0.0);
- const int channel(-1);
-
- // Only right channel will be modified
- _veVolumeControlPtr->GetOutputVolumePan(channel, left, right);
-
- right = (float)((float)pos/10.0f);
-
- TEST(_veVolumeControlPtr->SetOutputVolumePan(channel, left, right) == 0,
- _T("SetOutputVolumePan(channel=%d, left=%2.1f, right=%2.1f)"), channel, left, right);
-
- *pResult = 0;
-}
-
-void CWinTestDlg::OnBnClickedButtonVersion()
-{
- if (_veBasePtr)
- {
- char version[1024];
- if (_veBasePtr->GetVersion(version) == 0)
- {
- AfxMessageBox(CString(version), MB_OK);
- }
- else
- {
- AfxMessageBox(_T("FAILED!"), MB_OK);
- }
- }
-}
-
-void CWinTestDlg::OnBnClickedCheckRecCall()
-{
- // Not supported
-}
-
-void CWinTestDlg::OnBnClickedCheckTypingDetection()
-{
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_TYPING_DETECTION);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- TEST(_veApmPtr->SetTypingDetectionStatus(enable) == 0, _T("SetTypingDetectionStatus(enable=%d)"), enable);
-}
-
-void CWinTestDlg::OnBnClickedCheckFEC()
-{
- CButton* button = (CButton*)GetDlgItem(IDC_CHECK_FEC);
- int channel = GetDlgItemInt(IDC_EDIT_1);
- int check = button->GetCheck();
- const bool enable = (check == BST_CHECKED);
- TEST(_veRtpRtcpPtr->SetFECStatus(channel, enable) == 0, _T("SetFECStatus(enable=%d)"), enable);
-}
-
-// ----------------------------------------------------------------------------
-// Message Handlers
-// ----------------------------------------------------------------------------
-
-void CWinTestDlg::OnTimer(UINT_PTR nIDEvent)
-{
- CString str;
-
- unsigned int svol(0);
- unsigned int mvol(0);
-
- _timerTicks++;
-
- // Get speaker and microphone volumes
- _veVolumeControlPtr->GetSpeakerVolume(svol);
- _veVolumeControlPtr->GetMicVolume(mvol);
-
- // Update speaker volume slider
- CSliderCtrl* sliderSpkr = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_OUTPUT_VOLUME);
- sliderSpkr->SetRangeMin(0);
- sliderSpkr->SetRangeMax(255);
- sliderSpkr->SetPos(svol);
-
- // Update microphone volume slider
- CSliderCtrl* sliderMic = (CSliderCtrl*)GetDlgItem(IDC_SLIDER_INPUT_VOLUME);
- sliderMic->SetRangeMin(0);
- sliderMic->SetRangeMax(255);
- sliderMic->SetPos(mvol);
-
- unsigned int micLevel;
- unsigned int combinedOutputLevel;
-
- // Get audio levels
- _veVolumeControlPtr->GetSpeechInputLevel(micLevel);
- _veVolumeControlPtr->GetSpeechOutputLevel(-1, combinedOutputLevel);
-
- // Update audio level controls
- CProgressCtrl* progressMic = (CProgressCtrl*)GetDlgItem(IDC_PROGRESS_AUDIO_LEVEL_IN);
- progressMic->SetRange(0,9);
- progressMic->SetStep(1);
- progressMic->SetPos(micLevel);
- CProgressCtrl* progressOut = (CProgressCtrl*)GetDlgItem(IDC_PROGRESS_AUDIO_LEVEL_OUT);
- progressOut->SetRange(0,9);
- progressOut->SetStep(1);
- progressOut->SetPos(combinedOutputLevel);
-
- // Update playout delay (buffer size)
- if (_veVideoSyncPtr)
- {
- int bufferMs(0);
- _veVideoSyncPtr->GetPlayoutBufferSize(bufferMs);
- SetDlgItemInt(IDC_EDIT_PLAYOUT_BUFFER_SIZE, bufferMs);
- }
-
- if (_delayEstimate1 && _veVideoSyncPtr)
- {
- const int channel = GetDlgItemInt(IDC_EDIT_1);
- int delayMs(0);
- _veVideoSyncPtr->GetDelayEstimate(channel, delayMs);
- SetDlgItemInt(IDC_EDIT_DELAY_ESTIMATE_1, delayMs);
- }
-
- if (_rxVad && _veApmPtr && _rxVadObserverPtr)
- {
- SetDlgItemInt(IDC_EDIT_RXVAD, _rxVadObserverPtr->_vadDecision);
- }
-
- if (_veHardwarePtr)
- {
- int load1, load2;
- _veHardwarePtr->GetSystemCPULoad(load1);
- _veHardwarePtr->GetCPULoad(load2);
- str.Format(_T("CPU load (system/VoE): %d/%d [%%]"), load1, load2);
- SetDlgItemText(IDC_EDIT_CPU_LOAD, (LPCTSTR)str);
- }
-
- BOOL ret;
- int channel = GetDlgItemInt(IDC_EDIT_1, &ret);
-
- if (_veCodecPtr)
- {
- if (ret == TRUE)
- {
- CodecInst codec;
- if (_veCodecPtr->GetRecCodec(channel, codec) == 0)
- {
- str.Format(_T("RX codec: %s, freq=%d, pt=%d, rate=%d, size=%d"), CharToTchar(codec.plname, 32), codec.plfreq, codec.pltype, codec.rate, codec.pacsize);
- SetDlgItemText(IDC_EDIT_RX_CODEC_1, (LPCTSTR)str);
- }
- }
- }
-
- if (_veRtpRtcpPtr)
- {
- if (ret == TRUE)
- {
- CallStatistics stats;
- if (_veRtpRtcpPtr->GetRTCPStatistics(channel, stats) == 0)
- {
- str.Format(_T("RTCP | RTP: cum=%u, ext=%d, frac=%u, jitter=%u | TX=%d, RX=%d, RTT=%d"),
- stats.cumulativeLost, stats.extendedMax, stats.fractionLost, stats.jitterSamples, stats.packetsSent, stats.packetsReceived, stats.rttMs);
- SetDlgItemText(IDC_EDIT_RTCP_STAT_1, (LPCTSTR)str);
- }
- }
- }
-
- SetTimer(0, 1000, NULL);
- CDialog::OnTimer(nIDEvent);
-}
-
-void CWinTestDlg::OnBnClickedButtonClearErrorCallback()
-{
- _nErrorCallbacks = 0;
- SetDlgItemText(IDC_EDIT_ERROR_CALLBACK, _T(""));
-}
-
-// ----------------------------------------------------------------------------
-// TEST
-// ----------------------------------------------------------------------------
-
-void CWinTestDlg::OnBnClickedButtonTest1()
-{
- // add tests here...
-}
-
diff --git a/trunk/src/voice_engine/main/test/win_test/WinTestDlg.h b/trunk/src/voice_engine/main/test/win_test/WinTestDlg.h
deleted file mode 100644
index 6452775..0000000
--- a/trunk/src/voice_engine/main/test/win_test/WinTestDlg.h
+++ /dev/null
@@ -1,278 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#pragma once
-
-#if (_MSC_VER >= 1400)
-#define PRINT_GET_RESULT(...) \
- { \
- _strMsg.Format(__VA_ARGS__); \
- SetDlgItemText(IDC_EDIT_GET_OUTPUT, _strMsg); \
- } \
-
-#define TEST(x, ...) \
- if (!(x)) \
- { \
- _strMsg.Format(__VA_ARGS__); \
- SetDlgItemText(IDC_EDIT_MESSAGE, _strMsg); \
- _strErr.Format(_T("FAILED (error=%d)"), _veBasePtr->LastError()); \
- SetDlgItemText(IDC_EDIT_RESULT, _strErr); \
- _failCount++; \
- SetDlgItemInt(IDC_EDIT_N_FAILS, _failCount); \
- SetDlgItemInt(IDC_EDIT_LAST_ERROR, _veBasePtr->LastError()); \
- } \
- else \
- { \
- _strMsg.Format(__VA_ARGS__); \
- SetDlgItemText(IDC_EDIT_MESSAGE, _strMsg); \
- SetDlgItemText(IDC_EDIT_RESULT, _T("OK")); \
- } \
-
-#define TEST2(x, ...) \
- if (!(x)) \
- { \
- _strMsg.Format(__VA_ARGS__); \
- ((CWinTestDlg*)_parentDialogPtr)->UpdateTest(true, _strMsg); \
- } \
- else \
- { \
- _strMsg.Format(__VA_ARGS__); \
- ((CWinTestDlg*)_parentDialogPtr)->UpdateTest(false, _strMsg); \
- }
-#else
-#define TEST(x, exp) \
- if (!(x)) \
- { \
- _strMsg.Format(exp); \
- SetDlgItemText(IDC_EDIT_MESSAGE, _strMsg); \
- _strErr.Format("FAILED (error=%d)", _veBasePtr->LastError()); \
- SetDlgItemText(IDC_EDIT_RESULT, _strErr); \
- _failCount++; \
- SetDlgItemInt(IDC_EDIT_N_FAILS, _failCount); \
- SetDlgItemInt(IDC_EDIT_LAST_ERROR, _veBasePtr->LastError()); \
- } \
- else \
- { \
- _strMsg.Format(exp); \
- SetDlgItemText(IDC_EDIT_MESSAGE, _strMsg); \
- SetDlgItemText(IDC_EDIT_RESULT, _T("OK")); \
- } \
-
-#define TEST2(x, exp) \
- if (!(x)) \
- { \
- _strMsg.Format(exp); \
- ((CWinTestDlg*)_parentDialogPtr)->UpdateTest(true, _strMsg); \
- } \
- else \
- { \
- _strMsg.Format(exp); \
- ((CWinTestDlg*)_parentDialogPtr)->UpdateTest(false, _strMsg); \
- }
-#endif
-
-#include "voe_base.h"
-#include "voe_rtp_rtcp.h"
-#include "voe_codec.h"
-#include "voe_dtmf.h"
-#include "voe_encryption.h"
-#include "voe_external_media.h"
-#include "voe_file.h"
-#include "voe_hardware.h"
-#include "voe_network.h"
-#include "voe_video_sync.h"
-#include "voe_volume_control.h"
-
-#include "voe_audio_processing.h"
-#include "voe_rtp_rtcp.h"
-#include "voe_errors.h"
-
-class MediaProcessImpl;
-class ConnectionObserver;
-class MyEncryption;
-class RxCallback;
-class MyTransport;
-
-using namespace webrtc;
-
-#define MAX_NUM_OF_CHANNELS 10
-
-// CWinTestDlg dialog
-class CWinTestDlg : public CDialog,
- public VoiceEngineObserver,
- public VoERTPObserver
-{
-// Construction
-public:
- CWinTestDlg(CWnd* pParent = NULL); // standard constructor
- virtual ~CWinTestDlg();
-
-// Dialog Data
- enum { IDD = IDD_WINTEST_DIALOG };
-
- BOOL UpdateTest(bool failed, const CString& strMsg);
-
-protected:
- virtual void DoDataExchange(CDataExchange* pDX); // DDX/DDV support
-
-protected: // VoiceEngineObserver
- virtual void CallbackOnError(const int channel, const int errCode);
-
-protected: // VoERTPObserver
- virtual void OnIncomingCSRCChanged(
- const int channel, const unsigned int CSRC, const bool added);
- virtual void OnIncomingSSRCChanged(
- const int channel, const unsigned int SSRC);
-
-// Implementation
-protected:
- HICON m_hIcon;
-
- // Generated message map functions
- virtual BOOL OnInitDialog();
- afx_msg void OnSysCommand(UINT nID, LPARAM lParam);
- afx_msg void OnPaint();
- afx_msg HCURSOR OnQueryDragIcon();
- DECLARE_MESSAGE_MAP()
-public:
- afx_msg void OnBnClickedButtonCreate1();
- afx_msg void OnBnClickedButtonDelete1();
-
-private:
- VoiceEngine* _vePtr;
-
- VoECodec* _veCodecPtr;
- VoEExternalMedia* _veExternalMediaPtr;
- VoEVolumeControl* _veVolumeControlPtr;
- VoEEncryption* _veEncryptionPtr;
- VoEHardware* _veHardwarePtr;
- VoEVideoSync* _veVideoSyncPtr;
- VoENetwork* _veNetworkPtr;
- VoEFile* _veFilePtr;
- VoEAudioProcessing* _veApmPtr;
- VoEBase* _veBasePtr;
- VoERTP_RTCP* _veRtpRtcpPtr;
-
- MyTransport* _transportPtr;
- MediaProcessImpl* _externalMediaPtr;
- ConnectionObserver* _connectionObserverPtr;
- MyEncryption* _encryptionPtr;
- RxCallback* _rxVadObserverPtr;
-
-private:
- int _failCount;
- CString _strMsg;
- CString _strErr;
- bool _externalTransport;
- bool _externalTransportBuild;
- int _checkPlayFileIn;
- int _checkPlayFileIn1;
- int _checkPlayFileIn2;
- int _checkPlayFileOut1;
- int _checkPlayFileOut2;
- int _checkAGC;
- int _checkAGC1;
- int _checkNS;
- int _checkNS1;
- int _checkEC;
- int _checkVAD1;
- int _checkVAD2;
- int _checkSrtpTx1;
- int _checkSrtpTx2;
- int _checkSrtpRx1;
- int _checkSrtpRx2;
- int _checkConference1;
- int _checkConference2;
- int _checkOnHold1;
- int _checkOnHold2;
- bool _delayEstimate1;
- bool _delayEstimate2;
- bool _rxVad;
- int _nErrorCallbacks;
- int _timerTicks;
-
-public:
- afx_msg void OnBnClickedButtonCreate2();
- afx_msg void OnBnClickedButtonDelete2();
- afx_msg void OnCbnSelchangeComboCodec1();
- afx_msg void OnBnClickedButtonStartListen1();
- afx_msg void OnBnClickedButtonStopListen1();
- afx_msg void OnBnClickedButtonStartPlayout1();
- afx_msg void OnBnClickedButtonStopPlayout1();
- afx_msg void OnBnClickedButtonStartSend1();
- afx_msg void OnBnClickedButtonStopSend1();
- afx_msg void OnCbnSelchangeComboIp2();
- afx_msg void OnCbnSelchangeComboIp1();
- afx_msg void OnCbnSelchangeComboCodec2();
- afx_msg void OnBnClickedButtonStartListen2();
- afx_msg void OnBnClickedButtonStopListen2();
- afx_msg void OnBnClickedButtonStartPlayout2();
- afx_msg void OnBnClickedButtonStopPlayout2();
- afx_msg void OnBnClickedButtonStartSend2();
- afx_msg void OnBnClickedButtonStopSend2();
- afx_msg void OnBnClickedButtonTest11();
- afx_msg void OnBnClickedCheckExtTrans1();
- afx_msg void OnBnClickedCheckPlayFileIn1();
- afx_msg void OnBnClickedCheckPlayFileOut1();
- afx_msg void OnBnClickedCheckExtTrans2();
- afx_msg void OnBnClickedCheckPlayFileIn2();
- afx_msg void OnBnClickedCheckPlayFileOut2();
- afx_msg void OnBnClickedCheckPlayFileIn();
- afx_msg void OnBnClickedCheckPlayFileOut();
- afx_msg void OnCbnSelchangeComboRecDevice();
- afx_msg void OnCbnSelchangeComboPlayDevice();
- afx_msg void OnBnClickedCheckExtMediaIn1();
- afx_msg void OnBnClickedCheckExtMediaOut1();
- afx_msg void OnNMReleasedcaptureSliderInputVolume(NMHDR *pNMHDR, LRESULT *pResult);
- afx_msg void OnNMReleasedcaptureSliderOutputVolume(NMHDR *pNMHDR, LRESULT *pResult);
- afx_msg void OnTimer(UINT_PTR nIDEvent);
- afx_msg void OnBnClickedCheckAgc();
- CString _strComboIp1;
- CString _strComboIp2;
- afx_msg void OnBnClickedCheckNs();
- afx_msg void OnBnClickedCheckEc();
- afx_msg void OnBnClickedCheckVad1();
- afx_msg void OnBnClickedCheckVad2();
- afx_msg void OnBnClickedCheckExtMediaIn2();
- afx_msg void OnBnClickedCheckExtMediaOut2();
- afx_msg void OnBnClickedCheckMuteIn();
- afx_msg void OnBnClickedCheckMuteIn1();
- afx_msg void OnBnClickedCheckMuteIn2();
- afx_msg void OnBnClickedCheckSrtpTx1();
- afx_msg void OnBnClickedCheckSrtpRx1();
- afx_msg void OnBnClickedCheckSrtpTx2();
- afx_msg void OnBnClickedCheckSrtpRx2();
- afx_msg void OnBnClickedCheckExtEncryption1();
- afx_msg void OnBnClickedCheckExtEncryption2();
- afx_msg void OnBnClickedButtonDtmf1();
- afx_msg void OnBnClickedCheckRecMic();
- afx_msg void OnBnClickedButtonDtmf2();
- afx_msg void OnBnClickedButtonTest1();
- afx_msg void OnBnClickedCheckConference1();
- afx_msg void OnBnClickedCheckConference2();
- afx_msg void OnBnClickedCheckOnHold1();
- afx_msg void OnBnClickedCheckOnHold2();
- afx_msg void OnBnClickedCheckExtMediaIn();
- afx_msg void OnBnClickedCheckExtMediaOut();
- afx_msg void OnLbnSelchangeListCodec1();
- afx_msg void OnNMReleasedcaptureSliderPanLeft(NMHDR *pNMHDR, LRESULT *pResult);
- afx_msg void OnNMReleasedcaptureSliderPanRight(NMHDR *pNMHDR, LRESULT *pResult);
- afx_msg void OnBnClickedButtonVersion();
- afx_msg void OnBnClickedCheckDelayEstimate1();
- afx_msg void OnBnClickedCheckRxvad();
- afx_msg void OnBnClickedCheckAgc1();
- afx_msg void OnBnClickedCheckNs1();
- afx_msg void OnBnClickedCheckRecCall();
- afx_msg void OnBnClickedCheckTypingDetection();
- afx_msg void OnBnClickedCheckFEC();
- afx_msg void OnBnClickedButtonClearErrorCallback();
- afx_msg void OnBnClickedCheckBwe1();
-};
-#pragma once
diff --git a/trunk/src/voice_engine/main/test/win_test/res/WinTest.ico b/trunk/src/voice_engine/main/test/win_test/res/WinTest.ico
deleted file mode 100644
index 8a84ca3..0000000
--- a/trunk/src/voice_engine/main/test/win_test/res/WinTest.ico
+++ /dev/null
Binary files differ
diff --git a/trunk/src/voice_engine/main/test/win_test/res/WinTest.rc2 b/trunk/src/voice_engine/main/test/win_test/res/WinTest.rc2
deleted file mode 100644
index 044bf7e..0000000
--- a/trunk/src/voice_engine/main/test/win_test/res/WinTest.rc2
+++ /dev/null
@@ -1,13 +0,0 @@
-//
-// WinTest.RC2 - resources Microsoft Visual C++ does not edit directly
-//
-
-#ifdef APSTUDIO_INVOKED
-#error this file is not editable by Microsoft Visual C++
-#endif //APSTUDIO_INVOKED
-
-
-/////////////////////////////////////////////////////////////////////////////
-// Add manually edited resources here...
-
-/////////////////////////////////////////////////////////////////////////////
diff --git a/trunk/src/voice_engine/main/test/win_test/stdafx.cc b/trunk/src/voice_engine/main/test/win_test/stdafx.cc
deleted file mode 100644
index 6cdb906..0000000
--- a/trunk/src/voice_engine/main/test/win_test/stdafx.cc
+++ /dev/null
@@ -1,17 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// stdafx.cpp : source file that includes just the standard includes
-// WinTest.pch will be the pre-compiled header
-// stdafx.obj will contain the pre-compiled type information
-
-#include "stdafx.h"
-
-
diff --git a/trunk/src/voice_engine/main/test/win_test/stdafx.h b/trunk/src/voice_engine/main/test/win_test/stdafx.h
deleted file mode 100644
index b4d875c..0000000
--- a/trunk/src/voice_engine/main/test/win_test/stdafx.h
+++ /dev/null
@@ -1,82 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-// stdafx.h : include file for standard system include files,
-// or project specific include files that are used frequently,
-// but are changed infrequently
-
-#pragma once
-
-#ifndef _SECURE_ATL
-#define _SECURE_ATL 1
-#endif
-
-#ifndef VC_EXTRALEAN
-#define VC_EXTRALEAN // Exclude rarely-used stuff from Windows headers
-#endif
-
-// Modify the following defines if you have to target a platform prior to the ones specified below.
-// Refer to MSDN for the latest info on corresponding values for different platforms.
-#ifndef WINVER // Allow use of features specific to Windows XP or later.
-#define WINVER 0x0501 // Change this to the appropriate value to target other versions of Windows.
-#endif
-
-#ifndef _WIN32_WINNT // Allow use of features specific to Windows XP or later.
-#define _WIN32_WINNT 0x0501 // Change this to the appropriate value to target other versions of Windows.
-#endif
-
-#ifndef _WIN32_WINDOWS // Allow use of features specific to Windows 98 or later.
-#define _WIN32_WINDOWS 0x0410 // Change this to the appropriate value to target Windows Me or later.
-#endif
-
-#ifndef _WIN32_IE // Allow use of features specific to IE 6.0 or later.
-#define _WIN32_IE 0x0600 // Change this to the appropriate value to target other versions of IE.
-#endif
-
-#define _ATL_CSTRING_EXPLICIT_CONSTRUCTORS // some CString constructors will be explicit
-
-// turns off MFC's hiding of some common and often safely ignored warning messages
-#define _AFX_ALL_WARNINGS
-
-#include <afxwin.h> // MFC core and standard components
-#include <afxext.h> // MFC extensions
-
-
-
-
-
-#ifndef _AFX_NO_OLE_SUPPORT
-#include <afxdtctl.h> // MFC support for Internet Explorer 4 Common Controls
-#endif
-#ifndef _AFX_NO_AFXCMN_SUPPORT
-#include <afxcmn.h> // MFC support for Windows Common Controls
-#endif // _AFX_NO_AFXCMN_SUPPORT
-
-
-
-
-
-
-
-
-
-#ifdef _UNICODE
-#if defined _M_IX86
-#pragma comment(linker,"/manifestdependency:\"type='win32' name='Microsoft.Windows.Common-Controls' version='6.0.0.0' processorArchitecture='x86' publicKeyToken='6595b64144ccf1df' language='*'\"")
-#elif defined _M_IA64
-#pragma comment(linker,"/manifestdependency:\"type='win32' name='Microsoft.Windows.Common-Controls' version='6.0.0.0' processorArchitecture='ia64' publicKeyToken='6595b64144ccf1df' language='*'\"")
-#elif defined _M_X64
-#pragma comment(linker,"/manifestdependency:\"type='win32' name='Microsoft.Windows.Common-Controls' version='6.0.0.0' processorArchitecture='amd64' publicKeyToken='6595b64144ccf1df' language='*'\"")
-#else
-#pragma comment(linker,"/manifestdependency:\"type='win32' name='Microsoft.Windows.Common-Controls' version='6.0.0.0' processorArchitecture='*' publicKeyToken='6595b64144ccf1df' language='*'\"")
-#endif
-#endif
-
-
diff --git a/trunk/src/voice_engine/voice_engine.gyp b/trunk/src/voice_engine/voice_engine.gyp
deleted file mode 100644
index f45a269..0000000
--- a/trunk/src/voice_engine/voice_engine.gyp
+++ /dev/null
@@ -1,23 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'includes': [
- '../build/common.gypi',
- 'main/source/voice_engine_core.gypi',
- ],
-
- # Test targets, excluded when building with Chromium.
- 'conditions': [
- ['build_with_chromium==0', {
- 'includes': [
- 'main/test/voice_engine_tests.gypi',
- ],
- }],
- ],
-}
diff --git a/trunk/test/OWNERS b/trunk/test/OWNERS
deleted file mode 100644
index fec9caa..0000000
--- a/trunk/test/OWNERS
+++ /dev/null
@@ -1,4 +0,0 @@
-phoglund@webrtc.org
-kjellander@webrtc.org
-ivinnichenko@webrtc.org
-
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diff --git a/trunk/test/data/audio_coding/READ.ME b/trunk/test/data/audio_coding/READ.ME
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-Test files for Audio Coding Module
-
-testfile32kHz.pcm - mono speech file samples at 32 kHz
-teststereo32kHz.pcm - stereo speech file samples at 32 kHz
-
-Test and reference vectors to verify correct execution of PacketCable
-iLBC Fixed Point Reference Code
-
-Version 1.0.6
-Format: all .INP and .OUT files contain 16 bit sampled data using the
-Intel (PC) format. The .BIT files are stored in the appropriate byte
-sequence (big-endian format).
-
-*.INP - input files
-*.BIT20 - bit stream files 20 ms mode
-*.OUT20 - output files 20 ms mode (on a channel without packet loss)
-*.BIT30 - bit stream files 30 ms mode
-*.OUT30 - output files 30 ms mode (on a channel without packet loss)
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diff --git a/trunk/test/data/voice_engine/stereo_rtp_files/README.txt b/trunk/test/data/voice_engine/stereo_rtp_files/README.txt
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+++ /dev/null
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-Use RTP Play tool with command 'rtpplay.exe -v -T -f <path>\<file.rtp> 127.0.0.1/1236'
-Example: rtpplay.exe -v -T -f hrtf_g722_1C_48.rtp 127.0.0.1/1236.
-This sends the stereo rtp file to port 1236.
-You can hear the voice getting panned from left, right and center.
diff --git a/trunk/test/data/voice_engine/stereo_rtp_files/hrtf_g722_1C_48.rtp b/trunk/test/data/voice_engine/stereo_rtp_files/hrtf_g722_1C_48.rtp
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diff --git a/trunk/test/functional_test/README b/trunk/test/functional_test/README
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-This test client is a simple functional test for a WebRTC enabled browser. It
-has only been tested with Chrome, and is most likely only working with Chrome at
-the moment. The following instructions are in part Chrome specific.
-
-The following is necessary to run the test:
-- A WebRTC enabled Chrome binary. (Available in dev or canary channel, 18.0.1008
-or newer.)
-- A peerconnection_server binary (make peerconnection_server).
-
-It can be used in two scenarios:
-1. Single client calling itself with the server test page in loopback mode as a
-fake client.
-2. Call between two clients.
-
-To start the test for scenario (1):
-1. Start peerconnection_server.
-2. Start the WebRTC Chrome build:
-$ <path_to_chrome_binary>/chrome --enable-media-stream
-The --enable-media-stream flag is required for the time being.
-3. Open the server test page, ensure loopback is enabled, choose a name (for
-example "loopback") and connect to the server.
-For version 18.0.1008 to 18.0.1020, use:
-http://libjingle.googlecode.com/svn-history/r103/trunk/talk/examples/peerconnection/server/server_test.html
-For version 18.0.1021 and later, use:
-http://libjingle.googlecode.com/svn/trunk/talk/examples/peerconnection/server/server_test.html
-4. Open the test page, connect to the server, select the loopback peer, click
-call.
-
-To start the test for scenario (2):
-1. Start peerconnection_server.
-2. Start the WebRTC Chrome build, see scenario (1).
-3. Open the test page, connect to the server.
-4. Open a new new tab, open the test page, connect to the server.
- OR
- On another machine, repeat 2 and 3.
-5. Select the other peer, click call.
-
-Note: The web page must normally be on a web server to be able to access the
-camera for security reasons.
-See http://blog.chromium.org/2008/12/security-in-depth-local-web-pages.html
-for more details on this topic. This can be overridden with the flag
---allow-file-access-from-files, in which case running it over the file://
-URI scheme works.
-
-Note: It's possible to specify the server and name in the url:
-.../webrtc.html?server=my_server&name=my_name
diff --git a/trunk/test/functional_test/webrtc_test.html b/trunk/test/functional_test/webrtc_test.html
deleted file mode 100644
index 300884c..0000000
--- a/trunk/test/functional_test/webrtc_test.html
+++ /dev/null
@@ -1,602 +0,0 @@
-<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML//EN">
-
-<!--
-Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-
-Use of this source code is governed by a BSD-style license
-that can be found in the LICENSE file in the root of the source
-tree. An additional intellectual property rights grant can be found
-in the file PATENTS. All contributing project authors may
-be found in the AUTHORS file in the root of the source tree.
--->
-
-<html>
-
-<head>
-<title>WebRTC Test</title>
-
-<style type="text/css">
-body, input, button, select, table {
- font-family:"Lucida Grande", "Lucida Sans", Verdana, Arial, sans-serif;
- font-size: 13 px;
-}
-body, input:enable, button:enable, select:enable, table {
- color: rgb(51, 51, 51);
-}
-h1 {font-size: 40 px;}
-</style>
-
-<script type="text/javascript">
-
-// TODO: Catch more exceptions
-
-var server;
-var myId = -1;
-var myName;
-var remoteId = -1;
-var remoteName;
-var request = null;
-var hangingGet = null;
-var pc = null;
-var localStream = null;
-var disconnecting = false;
-var callState = 0; // 0 - Not started, 1 - Call ongoing
-
-
-// General
-
-function setElementValuesFromURL() {
- window.location.href.replace(/[?&]+([^=&]+)=([^&]*)/gi,
- function(m, key, value) {
- document.getElementById(key).value = unescape(value);
- });
-}
-
-function trace(txt) {
- var elem = document.getElementById("debug");
- elem.innerHTML += txt + "<br>";
-}
-
-function trace_warning(txt) {
- var wtxt = "<b>" + txt + "</b>";
- trace(wtxt);
-}
-
-function trace_exception(e, txt) {
- var etxt = "<b>" + txt + "</b> (" + e.name + " / " + e.message + ")";
- trace(etxt);
-}
-
-function setCallState(state) {
- trace("Changing call state: " + callState + " -> " + state);
- callState = state;
-}
-
-function checkPeerConnection() {
- if (!pc) {
- trace_warning("No PeerConnection object exists");
- return 0;
- }
- return 1;
-}
-
-
-// Local stream generation
-
-function gotStream(s) {
- var url = webkitURL.createObjectURL(s);
- document.getElementById("localView").src = url;
- trace("User has granted access to local media. url = " + url);
- localStream = s;
-}
-
-function gotStreamFailed(error) {
- alert("Failed to get access to local media. Error code was " + error.code +
- ".");
- trace_warning("Failed to get access to local media. Error code was " +
- error.code);
-}
-
-function getUserMedia() {
- try {
- navigator.webkitGetUserMedia("video,audio", gotStream, gotStreamFailed);
- trace("Requested access to local media");
- } catch (e) {
- trace_exception(e, "getUserMedia error");
- }
-}
-
-
-// Peer list and remote peer handling
-
-function peerExists(id) {
- try {
- var peerList = document.getElementById("peers");
- for (var i = 0; i < peerList.length; i++) {
- if (parseInt(peerList.options[i].value) == id)
- return true;
- }
- } catch (e) {
- trace_exception(e, "Error searching for peer");
- }
- return false;
-}
-
-function addPeer(id, pname) {
- try {
- var peerList = document.getElementById("peers");
- var option = document.createElement("option");
- option.text = pname;
- option.value = id;
- peerList.add(option, null);
- } catch (e) {
- trace_exception(e, "Error adding peer");
- }
-}
-
-function removePeer(id) {
- try {
- var peerList = document.getElementById("peers");
- for (var i = 0; i < peerList.length; i++) {
- if (parseInt(peerList.options[i].value) == id) {
- peerList.remove(i);
- break;
- }
- }
- } catch (e) {
- trace_exception(e, "Error removing peer");
- }
-}
-
-function clearPeerList() {
- var peerList = document.getElementById("peers");
- while (peerList.length > 0)
- peerList.remove(0);
-}
-
-function setSelectedPeer(id) {
- try {
- var peerList = document.getElementById("peers");
- for (var i = 0; i < peerList.length; i++) {
- if (parseInt(peerList.options[i].value) == id) {
- peerList.options[i].selected = true;
- return true;
- }
- }
- } catch (e) {
- trace_exception(e, "Error setting selected peer");
- }
- return false;
-}
-
-function getPeerName(id) {
- try {
- var peerList = document.getElementById("peers");
- for (var i = 0; i < peerList.length; i++) {
- if (parseInt(peerList.options[i].value) == id) {
- return peerList.options[i].text;
- }
- }
- } catch (e) {
- trace_exception(e, "Error finding peer name");
- return;
- }
- return;
-}
-
-function storeRemoteInfo() {
- try {
- var peerList = document.getElementById("peers");
- if (peerList.selectedIndex < 0) {
- alert("Please select a peer.");
- return false;
- } else
- remoteId = parseInt(peerList.options[peerList.selectedIndex].value);
- remoteName = peerList.options[peerList.selectedIndex].text;
- } catch (e) {
- trace_exception(e, "Error storing remote peer info");
- return false;
- }
- return true;
-}
-
-
-// Call control
-
-function createPeerConnection() {
- if (pc) {
- trace_warning("PeerConnection object already exists");
- }
- trace("Creating PeerConnection object");
- try {
- pc = new webkitPeerConnection("STUN stun.l.google.com:19302",
- onSignalingMessage);
- pc.onaddstream = onAddStream;
- pc.onremovestream = onRemoveStream;
- } catch (e) {
- trace_exception(e, "Create PeerConnection error");
- }
-}
-
-function doCall() {
- if (!storeRemoteInfo())
- return;
- document.getElementById("call").disabled = true;
- document.getElementById("peers").disabled = true;
- createPeerConnection();
- trace("Adding stream");
- pc.addStream(localStream);
- document.getElementById("hangup").disabled = false;
- setCallState(1);
-}
-
-function hangUp() {
- document.getElementById("hangup").disabled = true;
- trace("Sending BYE to " + remoteName + " (ID " + remoteId + ")");
- sendToPeer(remoteId, "BYE");
- closeCall();
-}
-
-function closeCall() {
- trace("Stopping showing remote stream");
- document.getElementById("remoteView").src = "dummy";
- if (pc) {
- trace("Stopping call [pc.close()]");
- pc.close();
- pc = null;
- } else
- trace("No pc object to close");
- remoteId = -1;
- document.getElementById("call").disabled = false;
- document.getElementById("peers").disabled = false;
- setCallState(0);
-}
-
-
-// PeerConnection callbacks
-
-function onAddStream(e) {
- var stream = e.stream;
- var url = webkitURL.createObjectURL(stream);
- document.getElementById("remoteView").src = url;
- trace("Started showing remote stream. url = " + url);
-}
-
-function onRemoveStream(e) {
- // Currently if we get this callback, call has ended.
- document.getElementById("remoteView").src = "";
- trace("Stopped showing remote stream");
-}
-
-function onSignalingMessage(msg) {
- trace("Sending message to " + remoteName + " (ID " + remoteId + "):\n" + msg);
- sendToPeer(remoteId, msg);
-}
-
-// TODO: Add callbacks onconnecting, onopen and onstatechange.
-
-
-// Server interaction
-
-function handleServerNotification(data) {
- trace("Server notification: " + data);
- var parsed = data.split(",");
- if (parseInt(parsed[2]) == 1) { // New peer
- var peerId = parseInt(parsed[1]);
- if (!peerExists(peerId)) {
- var peerList = document.getElementById("peers");
- if (peerList.length == 1 && peerList.options[0].value == -1)
- clearPeerList();
- addPeer(peerId, parsed[0]);
- document.getElementById("peers").disabled = false;
- document.getElementById("call").disabled = false;
- }
- } else if (parseInt(parsed[2]) == 0) { // Removed peer
- removePeer(parseInt(parsed[1]));
- if (document.getElementById("peers").length == 0) {
- document.getElementById("peers").disabled = true;
- addPeer(-1, "No other peer connected");
- }
- }
-}
-
-function handlePeerMessage(peer_id, msg) {
- var peerName = getPeerName(peer_id);
- if (peerName == undefined) {
- trace_warning("Received message from unknown peer (ID " + peer_id +
- "), ignoring message:");
- trace(msg);
- return;
- }
- trace("Received message from " + peerName + " (ID " + peer_id + "):\n" + msg);
- // Assuming we receive the message from the peer we want to communicate with.
- // TODO: Only accept messages from peer we communicate with with if call is
- // ongoing.
- if (msg.search("BYE") == 0) {
- // Other side has hung up.
- document.getElementById("hangup").disabled = true;
- closeCall()
- } else {
- if (!pc) {
- // Other side is calling us, startup
- if (!setSelectedPeer(peer_id)) {
- trace_warning("Recevied message from unknown peer, ignoring");
- return;
- }
- if (!storeRemoteInfo())
- return;
- document.getElementById("call").disabled = true;
- document.getElementById("peers").disabled = true;
- createPeerConnection();
- try {
- pc.processSignalingMessage(msg);
- } catch (e) {
- trace_exception(e, "Process signaling message error");
- }
- trace("Adding stream");
- pc.addStream(localStream);
- document.getElementById("hangup").disabled = false;
- } else {
- try {
- pc.processSignalingMessage(msg);
- } catch (e) {
- trace_exception(e, "Process signaling message error");
- }
- }
- }
-}
-
-function getIntHeader(r, name) {
- var val = r.getResponseHeader(name);
- trace("header value: " + val);
- return val != null && val.length ? parseInt(val) : -1;
-}
-
-function hangingGetCallback() {
- try {
- if (hangingGet.readyState != 4 || disconnecting)
- return;
- if (hangingGet.status != 200) {
- trace_warning("server error, status: " + hangingGet.status + ", text: " +
- hangingGet.statusText);
- disconnect();
- } else {
- var peer_id = getIntHeader(hangingGet, "Pragma");
- if (peer_id == myId) {
- handleServerNotification(hangingGet.responseText);
- } else {
- handlePeerMessage(peer_id, hangingGet.responseText);
- }
- }
-
- if (hangingGet) {
- hangingGet.abort();
- hangingGet = null;
- }
-
- if (myId != -1)
- window.setTimeout(startHangingGet, 0);
- } catch (e) {
- trace_exception(e, "Hanging get error");
- }
-}
-
-function onHangingGetTimeout() {
- trace("hanging get timeout. issuing again");
- hangingGet.abort();
- hangingGet = null;
- if (myId != -1)
- window.setTimeout(startHangingGet, 0);
-}
-
-function startHangingGet() {
- try {
- hangingGet = new XMLHttpRequest();
- hangingGet.onreadystatechange = hangingGetCallback;
- hangingGet.ontimeout = onHangingGetTimeout;
- hangingGet.open("GET", server + "/wait?peer_id=" + myId, true);
- hangingGet.send();
- } catch (e) {
- trace_exception(e, "Start hanging get error");
- }
-}
-
-function sendToPeer(peer_id, data) {
- if (myId == -1) {
- alert("Not connected.");
- return;
- }
- if (peer_id == myId) {
- alert("Can't send a message to oneself.");
- return;
- }
- var r = new XMLHttpRequest();
- r.open("POST", server + "/message?peer_id=" + myId + "&to=" + peer_id, false);
- r.setRequestHeader("Content-Type", "text/plain");
- r.send(data);
- r = null;
-}
-
-function signInCallback() {
- try {
- if (request.readyState == 4) {
- if (request.status == 200) {
- var peers = request.responseText.split("\n");
- myId = parseInt(peers[0].split(",")[1]);
- trace("My id: " + myId);
- clearPeerList();
- var added = 0;
- for (var i = 1; i < peers.length; ++i) {
- if (peers[i].length > 0) {
- trace("Peer " + i + ": " + peers[i]);
- var parsed = peers[i].split(",");
- addPeer(parseInt(parsed[1]), parsed[0]);
- ++added;
- }
- }
- if (added == 0)
- addPeer(-1, "No other peer connected");
- else {
- document.getElementById("peers").disabled = false;
- document.getElementById("call").disabled = false;
- }
- startHangingGet();
- request = null;
- document.getElementById("connect").disabled = true;
- document.getElementById("disconnect").disabled = false;
- }
- }
- } catch (e) {
- trace_exception(e, "Sign in error");
- document.getElementById("connect").disabled = false;
- }
-}
-
-function signIn() {
- try {
- request = new XMLHttpRequest();
- request.onreadystatechange = signInCallback;
- request.open("GET", server + "/sign_in?" + myName, true);
- request.send();
- } catch (e) {
- trace_exception(e, "Start sign in error");
- document.getElementById("connect").disabled = false;
- }
-}
-
-function connect() {
- myName = document.getElementById("name").value.toLowerCase();
- server = document.getElementById("server").value.toLowerCase();
- if (myName.length == 0) {
- alert("I need a name please.");
- document.getElementById("local").focus();
- } else {
- // TODO: Disable connect button here, but we need a timeout and check if we
- // have connected, if so enable it again.
- signIn();
- }
-}
-
-function disconnect() {
- if (callState == 1)
- hangUp();
-
- disconnecting = true;
-
- if (request) {
- request.abort();
- request = null;
- }
-
- if (hangingGet) {
- hangingGet.abort();
- hangingGet = null;
- }
-
- if (myId != -1) {
- request = new XMLHttpRequest();
- request.open("GET", server + "/sign_out?peer_id=" + myId, false);
- request.send();
- request = null;
- myId = -1;
- }
-
- clearPeerList();
- addPeer(-1, "Not connected");
- document.getElementById("connect").disabled = false;
- document.getElementById("disconnect").disabled = true;
- document.getElementById("peers").disabled = true;
- document.getElementById("call").disabled = true;
-
- disconnecting = false;
-}
-
-
-// Window event handling
-
-window.onload = function() {
- if (navigator.webkitGetUserMedia) {
- document.getElementById('testApp').hidden = false;
- setElementValuesFromURL();
- getUserMedia();
- } else {
- document.getElementById('errorText').hidden = false;
- }
-}
-
-window.onbeforeunload = disconnect;
-
-</script>
-</head>
-
-<body>
-<h1>WebRTC</h1>
-<section id="errorText" hidden="true">
-Could not detect WebRTC support.<p>
-You must have WebRTC enabled Chrome browser to use this test page. The browser
-must be started with the --enable-media-stream command line flag. For more
-information, please see
-<a href="https://sites.google.com/site/webrtc/blog/webrtcnowavailableinthechromedevchannel">
-this blog post</a>.
-</section>
-
-<section id="testApp" hidden="true">
-<table border="0">
-<tr>
- <td>Local Preview</td>
- <td>Remote Video</td>
-</tr>
-<tr>
- <td>
- <video width="320" height="240" id="localView" autoplay="autoplay"></video>
- </td>
- <td>
- <video width="640" height="480" id="remoteView" autoplay="autoplay"></video>
- </td>
-</tr>
-</table>
-
-<table border="0">
-<tr>
- <td valign="top">
- <table border="0" cellpaddning="0" cellspacing="0">
- <tr>
- <td>Server:</td>
- <td>
- <input type="text" id="server" size="30" value="http://localhost:8888"/>
- </td>
- </tr>
- <tr>
- <td>Name:</td><td><input type="text" id="name" size="30" value="name"/></td>
- </tr>
- </table>
- </td>
- <td valign="top">
- <button id="connect" onclick="connect();">Connect</button><br>
- <button id="disconnect" onclick="disconnect();" disabled="true">Disconnect
- </button>
- </td>
- <td> </td>
- <td valign="top">
- Connected peers:<br>
- <select id="peers" size="5" disabled="true">
- <option value="-1">Not connected</option>
- </select>
- </td>
- <td valign="top">
- <!--input type="text" id="peer_id" size="3" value="1"/><br-->
- <button id="call" onclick="doCall();" disabled="true">Call</button><br>
- <button id="hangup" onclick="hangUp();" disabled="true">Hang up</button><br>
- </td>
-</tr>
-</table>
-
-<button onclick="document.getElementById('debug').innerHTML='';">Clear log
-</button>
-<pre id="debug"></pre>
-</section>
-
-</body>
-
-</html>
-
diff --git a/trunk/test/metrics.gyp b/trunk/test/metrics.gyp
deleted file mode 100644
index 70483f9..0000000
--- a/trunk/test/metrics.gyp
+++ /dev/null
@@ -1,46 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'includes': [
- '../src/build/common.gypi',
- ],
- 'targets': [
- {
- # The metrics code must be kept in its own GYP file in order to
- # avoid a circular dependency error due to the dependency on libyuv.
- # If the code would be put in test.gyp a circular dependency error during
- # GYP generation would occur, because the libyuv.gypi unittest target
- # depends on test_support_main. See issue #160 for more info.
- 'target_name': 'metrics',
- 'type': '<(library)',
- 'dependencies': [
- '<(webrtc_root)/common_video/common_video.gyp:webrtc_libyuv',
- ],
- 'include_dirs': [
- '.',
- ],
- 'sources': [
- 'testsupport/metrics/video_metrics.h',
- 'testsupport/metrics/video_metrics.cc',
- ],
- },
- {
- 'target_name': 'metrics_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'metrics',
- '<(webrtc_root)/../test/test.gyp:test_support_main',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'sources': [
- 'testsupport/metrics/video_metrics_unittest.cc',
- ],
- },
- ],
-}
diff --git a/trunk/test/run_all_unittests.cc b/trunk/test/run_all_unittests.cc
deleted file mode 100644
index 0cdf0cd..0000000
--- a/trunk/test/run_all_unittests.cc
+++ /dev/null
@@ -1,16 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "test/test_suite.h"
-
-int main(int argc, char** argv) {
- webrtc::test::TestSuite test_suite(argc, argv);
- return test_suite.Run();
-}
diff --git a/trunk/test/sanity_check/README b/trunk/test/sanity_check/README
deleted file mode 100644
index 7b14aeb..0000000
--- a/trunk/test/sanity_check/README
+++ /dev/null
@@ -1,13 +0,0 @@
-This test will pop up a browser with two tabs, and will run video from
-one tab to the other tab, across the local network interface.
-
-In order to run this test, you need to have:
-
-- a WebRTC Chrome binary in
- $PROJECT_ROOT/../../chromium/src/out/Debug/chrome
-- an installed lighttpd in your $PATH
-- a built peerconnection_server (make peerconnection_server)
-
-You need to run the script from the test/sanity_check directory.
-
-The first time you run it, you will be asked to choose a search engine :-(
diff --git a/trunk/test/sanity_check/lighttpd.conf b/trunk/test/sanity_check/lighttpd.conf
deleted file mode 100644
index 4dae92d..0000000
--- a/trunk/test/sanity_check/lighttpd.conf
+++ /dev/null
@@ -1,5 +0,0 @@
-server.document-root = "./www/html/"
-server.port = 3000
-mimetype.assign = (
- ".html" => "text/html"
-)
diff --git a/trunk/test/sanity_check/run_sanity_check b/trunk/test/sanity_check/run_sanity_check
deleted file mode 100644
index cf38e98..0000000
--- a/trunk/test/sanity_check/run_sanity_check
+++ /dev/null
@@ -1,72 +0,0 @@
-#!/bin/bash
-#
-# Run a test with the WebRTC Chromium build.
-# Should work on any machine with a camera.
-#
-# Method:
-# - Start server
-# - Start 2 browser tabs
-# - Browser tab 1 captures camera
-# - Both browsers sign in
-# - Browser 1 calls browser 2
-# - Browser 2 displays camera feed from browser 1
-#
-
-# Feel free to tweak this locally if your chrome build is somewhere else.
-# The default assumes that it is in a folder called chromium two levels
-# up from the project root ('trunk').
-
-set -e
-
-if [ ! -x run_sanity_check ]; then
- echo "Error: This script must run from its own directory"
- exit 1
-fi
-
-URLBASE=127.0.0.1:3000
-CALLER=$URLBASE/call_client.html
-CALLEE=$URLBASE/call_responder.html
-FLAGS="--enable-media-stream --enable-p2papi"
-PROJECT_ROOT=../..
-CHROME_BINARY=$PROJECT_ROOT/../../chromium/src/out/Debug/chrome
-
-if which lighttpd; then
- LOCAL_WEB_SERVER_BINARY=$(which lighttpd)
-else
- echo "Error: You must install lighttpd first (sudo apt-get install lighttpd)"
- exit 1
-fi
-
-SERVER_BINARY=${PROJECT_ROOT}/out/Debug/peerconnection_server
-if [ ! -e "$SERVER_BINARY" ]; then
- echo "Error: You must build peerconnection_server first."
- exit 1
-fi
-
-CHROME_BINARY=$PROJECT_ROOT/../../chromium/src/out/Debug/chrome
-if [ ! -e "$CHROME_BINARY" ]; then
- echo "Error: You must build chrome (could not open $CHROME_BINARY)."
- exit 1
-fi
-
-# Launch the web server and make it serve the local www/html directory
-${LOCAL_WEB_SERVER_BINARY} -D -f lighttpd.conf &
-LOCAL_WEB_SERVER=$!
-
-${SERVER_BINARY} &
-SERVER=$!
-echo "Started server as $SERVER"
-
-# We can make 2 browsers
-#${CHROME_BINARY} $CALLER $FLAGS --user-data-dir=temp/user1 &
-#USER1=$!
-#echo "Started first user as $USER1"
-#${CHROME_BINARY} $CALLEE $FLAGS --user-data-dir=temp/user2
-
-# But it also works with separate tabs in one browser.
-${CHROME_BINARY} $CALLER $CALLEE $FLAGS --user-data-dir=temp/user1
-
-echo "Test finished, cleaning up"
-
-kill $SERVER || echo "No server"
-kill $LOCAL_WEB_SERVER || echo "No local web server"
diff --git a/trunk/test/sanity_check/www/html/call_client.html b/trunk/test/sanity_check/www/html/call_client.html
deleted file mode 100644
index 789bc9f..0000000
--- a/trunk/test/sanity_check/www/html/call_client.html
+++ /dev/null
@@ -1,394 +0,0 @@
-<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML//EN">
-
-<html>
-
-<head>
-<title>WebRTC Autocaller</title>
-<script type="text/javascript">
-
-var local_name;
-var server;
-var my_id = -1;
-var other_peers = {};
-var request = null;
-var hanging_get = null;
-var pc = null;
-var local_stream = null;
-var call_state = 0;
-// 0 - Not started
-// 1 - Call ongoing
-// 2 - We have initiated closing the call
-// 3 - Other side has initiated closing the call
-
-// General
-function toggleExtraButtons() {
- document.getElementById("createPcBtn").hidden =
- !document.getElementById("createPcBtn").hidden;
- document.getElementById("addStreamBtn").hidden =
- !document.getElementById("addStreamBtn").hidden;
-}
-
-function trace(txt) {
- var elem = document.getElementById("debug");
- elem.innerHTML += txt + "<br>";
-}
-
-function trace_warning(txt) {
- var wtxt = "<b>" + txt + "</b>";
- trace(wtxt);
-}
-
-function setCallState(state) {
- trace("Changing call state: " + call_state + " -> " + state);
- call_state = state;
-}
-
-function checkPeerConnection() {
- if (!pc) {
- trace_warning("No PeerConnection object exists.");
- return 0;
- }
- return 1;
-}
-
-// Local stream generation
-function gotStream(s) {
- var url = webkitURL.createObjectURL(s);
- document.getElementById("localView").src = url;
- trace("User has granted access to webcam. url = " + url);
- local_stream = s;
-}
-
-function gotStreamFailed(error) {
- alert("Failed to get access to webcam. Error code was " + error.code);
- trace("Failed to get access to webcam. Error code was " + error.code);
-}
-
-// Call control
-function createPeerConnection() {
- if (pc) {
- trace_warning("PeerConnection object already exists.");
- }
- trace("Creating PeerConnection object.");
- // CRASH TEST
- pc = new webkitPeerConnection("STUN 1.2.3.4:43", onSignalingMessage);
- pc.onaddstream = onAddStream;
- pc.onremovestream = onRemoveStream;
-}
-
-function startCall() {
- document.getElementById("startcall").disabled = true;
- createPeerConnection();
- addStream();
-}
-
-function addStream() {
- trace("Starting call. [pc.addStream(local_stream)]");
- pc.addStream(local_stream);
- document.getElementById("stopcall").disabled = false;
-}
-
-function stopCall() {
- document.getElementById("stopcall").disabled = true;
- trace("Stopping call [pc.close()]");
- setCallState(2);
- pc.close();
-}
-
-function test1() {
- addStream();
-}
-
-function callHasStarted() {
- setCallState(1);
-}
-
-// PeerConnection callbacks
-function onAddStream(e) {
- var stream = pc.remoteStreams[0];
- var url = webkitURL.createObjectURL(stream);
- document.getElementById("remoteView").src = url;
- trace("Started showing remote stream. url = " + url);
- setTimeout(callHasStarted, 5000); // TODO(hta): workaround, to be removed
-}
-
-function onRemoveStream(e) {
- // Currently if we get this callback, call has ended.
- document.getElementById("remoteView").src = "";
- trace("Stopped showing remote stream.");
- if (call_state == 2) {
- // Finalize close call. We assume here that this callback comes
- // after onSignalingMessage.
- pc = null;
- setCallState(0);
- document.getElementById("startcall").disabled = false;
- }
-}
-
-function onSignalingMessage(msg) {
- var peer_id = parseInt(document.getElementById("peer_id").value);
- trace("Sending message to " + other_peers[peer_id] +
- " (ID " + peer_id + "):\n" + msg);
- sendToPeer(peer_id, msg);
-}
-
-// Server interaction
-
-function handleServerNotification(data) {
- trace("Server notification: " + data);
- notePeerAndMaybeCall(data);
-}
-
-function handlePeerMessage(peer_id, msg) {
- trace("Received message from " + other_peers[peer_id] +
- " (ID " + peer_id + "):\n" + msg);
- // Assuming we have created a PeerConnection and that we receive the
- // message from the peer want to communicate with
- document.getElementById("peer_id").value =
- peer_id; // Ensure we reply to the same peer
-
- if (pc) {
- try {
- pc.processSignalingMessage(msg);
- } catch (e) {
- trace_warning("Signaling message error: " + e.description);
- }
- if (call_state == 1)
- setCallState(3);
- } else {
- createPeerConnection();
- pc.processSignalingMessage(msg);
- }
-}
-
-function getIntHeader(r, name) {
- var val = r.getResponseHeader(name);
- trace("header value: " + val);
- return val != null && val.length ? parseInt(val) : -1;
-}
-
-function hangingGetCallback() {
- try {
- if (hanging_get.readyState != 4)
- return;
- if (hanging_get.status != 200) {
- trace("server error, status: " + hanging_get.status + ", text: " +
- hanging_get.statusText);
- disconnect();
- } else {
- var peer_id = getIntHeader(hanging_get, "Pragma");
- if (peer_id == my_id) {
- handleServerNotification(hanging_get.responseText);
- } else {
- handlePeerMessage(peer_id, hanging_get.responseText);
- }
- }
-
- if (hanging_get) {
- hanging_get.abort();
- hanging_get = null;
- }
-
- if (my_id != -1)
- window.setTimeout(startHangingGet, 0);
- } catch (e) {
- trace("Hanging get error: " + e.description);
- }
-}
-
-function onHangingGetTimeout() {
- trace("hanging get timeout. issuing again.");
- hanging_get.abort();
- hanging_get = null;
- if (my_id != -1)
- window.setTimeout(startHangingGet, 0);
-}
-
-function startHangingGet() {
- try {
- hanging_get = new XMLHttpRequest();
- hanging_get.onreadystatechange = hangingGetCallback;
- hanging_get.ontimeout = onHangingGetTimeout;
- hanging_get.open("GET", server + "/wait?peer_id=" + my_id, true);
- hanging_get.send();
- } catch (e) {
- trace("error" + e.description);
- }
-}
-
-function sendToPeer(peer_id, data) {
- if (my_id == -1) {
- alert("Not connected");
- return;
- }
- if (peer_id == my_id) {
- alert("Can't send a message to oneself :)");
- return;
- }
- var r = new XMLHttpRequest();
- r.open("POST", server + "/message?peer_id=" + my_id + "&to=" +
- peer_id, false);
- r.setRequestHeader("Content-Type", "text/plain");
- r.send(data);
- r = null;
-}
-
-function notePeerAndMaybeCall(peerInfo) {
- var parsed = peerInfo.split(",");
- if (parseInt(parsed[2]) != 0) {
- trace("Peer came on: " + peerInfo);
- other_peers[parseInt(parsed[1])] = parsed[0];
- // If we're not already in a call, call it.
- if (!document.getElementById("startcall").disabled) {
- document.getElementById("peer_id").value = parsed[1];
- startCall();
- }
- } else {
- trace("Peer went away: " + peerInfo);
- other_peers[parseInt(parsed[1])] = null;
- }
-}
-
-function signInCallback() {
- try {
- if (request.readyState == 4) {
- if (request.status == 200) {
- var peers = request.responseText.split("\n");
- my_id = parseInt(peers[0].split(",")[1]);
- trace("My id: " + my_id);
- for (var i = 1; i < peers.length; ++i) {
- if (peers[i].length > 0) {
- trace("Peer " + i + ": " + peers[i]);
- notePeerAndMaybeCall(peers[i]);
- }
- }
- startHangingGet();
- request = null;
- }
- }
- } catch (e) {
- trace("error: " + e.description);
- }
-}
-
-function signIn() {
- try {
- request = new XMLHttpRequest();
- request.onreadystatechange = signInCallback;
- request.open("GET", server + "/sign_in?" + local_name, true);
- request.send();
- } catch (e) {
- trace("error: " + e.description);
- }
-}
-
-function connect() {
- local_name = document.getElementById("local").value.toLowerCase();
- server = document.getElementById("server").value.toLowerCase();
- if (local_name.length == 0) {
- alert("I need a name please.");
- document.getElementById("local").focus();
- } else {
- document.getElementById("connect").disabled = true;
- document.getElementById("disconnect").disabled = false;
- trace("Connecting to server");
- signIn();
- }
-}
-
-function disconnect() {
- if (request) {
- request.abort();
- request = null;
- }
-
- if (hanging_get) {
- hanging_get.abort();
- hanging_get = null;
- }
-
- if (my_id != -1) {
- request = new XMLHttpRequest();
- request.open("GET", server + "/sign_out?peer_id=" + my_id, false);
- request.send();
- request = null;
- my_id = -1;
- }
-
- document.getElementById("connect").disabled = false;
- document.getElementById("disconnect").disabled = true;
-}
-
-function toggleLog()
-{
- document.getElementById('debug').hidden =!
- document.getElementById('debug').hidden;
-}
-
-window.onbeforeunload = disconnect;
-
-// TODO: Add audio here when available. Note: now it's hard coded,
-// audio will also be started.
-navigator.webkitGetUserMedia("video", gotStream, gotStreamFailed);
-
-</script>
-</head>
-
-<body>
-<h1>WebRTC autocaller</h1>
-This page should show your local camera, and connect to a remote browser.
-<table border="0">
-
-<tr>
-<th align="left">Local Preview</th>
-<th align="left">Remote Video</th>
-</tr>
-<tr>
-<td><video width="320" height="240" id="localView" autoplay="autoplay">
- </video></td>
-<td><video width="640" height="480" id="remoteView" autoplay="autoplay">
- </video></td>
-</tr>
-<tr>
-<td valign="top">
-<form>
-Target peer ID: <input type="text" id="peer_id" size="3" value="1"/><br>
-<button id="startcall" onclick="startCall();">(4) Start call</button><br>
-<button id="stopcall" onclick="stopCall();" disabled="true">(5) Stop call
-</button><br>
-<!--<input type="button" value="Test 1" onclick="test1()"/><br>-->
-</form>
-</td>
-<td valign="top">
-<button onclick="toggleExtraButtons();">Toggle extra buttons</button><br>
-<button id="createPcBtn" onclick="createPeerConnection();" hidden="true">
- Create peer connection</button><br>
-<button id="addStreamBtn" onclick="addStream();" hidden="true">
- Add stream</button><br>
-</td>
-</tr>
-
-<tr><td colspan="2">
-(1) Server: <input type="text" id="server" value="http://localhost:8888" />
-<br>
-(2) Your name: <input type="text" id="local" value="caller"/>
-<button id="connect" onclick="connect();" disabled="true">(3) Connect</button>
-<button id="disconnect" onclick="disconnect();" disabled="true">(6) Disconnect
-</button>
-<br>
- <br>
-<button onclick="toggleLog()">Toggle view log</button>
-<button onclick="document.getElementById('debug').innerHTML='';">
- Clear log</button>
-<pre id="debug">
-</pre>
-</td></tr>
-
-</table>
-</body>
-<script>
-// On init, we connect to the server.
-connect();
-</script>
-
-</html>
diff --git a/trunk/test/sanity_check/www/html/call_responder.html b/trunk/test/sanity_check/www/html/call_responder.html
deleted file mode 100644
index 4d7f3b4..0000000
--- a/trunk/test/sanity_check/www/html/call_responder.html
+++ /dev/null
@@ -1,373 +0,0 @@
-<!DOCTYPE HTML PUBLIC "-//IETF//DTD HTML//EN">
-
-<html>
-
-<head>
-<title>WebRTC Autoresponder</title>
-This page should show a stream coming from a different browser.
-<script type="text/javascript">
-
-var local_name;
-var server;
-var my_id = -1;
-var other_peers = {};
-var request = null;
-var hanging_get = null;
-var pc = null;
-var local_stream = null;
-var call_state = 0;
-// 0 - Not started
-// 1 - Call ongoing
-// 2 - We have initiated closing the call
-// 3 - Other side has initiated closing the call
-
-// General
-function toggleExtraButtons() {
- document.getElementById("createPcBtn").hidden =
- !document.getElementById("createPcBtn").hidden;
- document.getElementById("addStreamBtn").hidden =
- !document.getElementById("addStreamBtn").hidden;
-}
-
-function trace(txt) {
- var elem = document.getElementById("debug");
- elem.innerHTML += txt + "<br>";
-}
-
-function trace_warning(txt) {
- var wtxt = "<b>" + txt + "</b>";
- trace(wtxt);
-}
-
-function setCallState(state) {
- trace("Changing call state: " + call_state + " -> " + state);
- call_state = state;
-}
-
-function checkPeerConnection() {
- if (!pc) {
- trace_warning("No PeerConnection object exists.");
- return 0;
- }
- return 1;
-}
-
-// Local stream generation
-function gotStream(s) {
- var url = webkitURL.createObjectURL(s);
- document.getElementById("localView").src = url;
- trace("User has granted access to webcam. url = " + url);
- local_stream = s;
-}
-
-function gotStreamFailed(error) {
- alert("Failed to get access to webcam. Error code was " + error.code);
- trace("Failed to get access to webcam. Error code was " + error.code);
-}
-
-// Call control
-function createPeerConnection() {
- if (pc) {
- trace_warning("PeerConnection object already exists.");
- }
- trace("Creating PeerConnection object.");
- pc = new webkitPeerConnection("STUN stun.l.google.com:19302",
- onSignalingMessage);
- pc.onaddstream = onAddStream;
- pc.onremovestream = onRemoveStream;
-}
-
-function startCall() {
- document.getElementById("startcall").disabled = true;
- createPeerConnection();
- addStream();
-}
-
-function addStream()
-{
- trace("Starting call. [pc.addStream(local_stream)]");
- pc.addStream(local_stream);
- document.getElementById("stopcall").disabled = false;
-}
-
-function stopCall() {
- document.getElementById("stopcall").disabled = true;
- trace("Stopping call [pc.close()]");
- setCallState(2);
- pc.close();
-}
-
-function test1() {
- addStream();
-}
-
-function callHasStarted() {
- setCallState(1);
-}
-
-// PeerConnection callbacks
-function onAddStream(e) {
- var stream = pc.remoteStreams[0];
- var url = webkitURL.createObjectURL(stream);
- document.getElementById("remoteView").src = url;
- trace("Started showing remote stream. url = " + url);
- setTimeout(callHasStarted, 5000); // TODO: Temp workaround, to be removed
-}
-
-function onRemoveStream(e) {
- // Currently if we get this callback, call has ended.
- document.getElementById("remoteView").src = "";
- trace("Stopped showing remote stream.");
- if (call_state == 2) {
- // Finalize close call. We assume here that this
- // callback comes after onSignalingMessage.
- pc = null;
- setCallState(0);
- document.getElementById("startcall").disabled = false;
- }
-}
-
-function onSignalingMessage(msg) {
- var peer_id = parseInt(document.getElementById("peer_id").value);
- trace("Sending message to " + other_peers[peer_id] +
- " (ID " + peer_id + "):\n" + msg);
- sendToPeer(peer_id, msg);
-}
-
-// Server interaction
-
-function handleServerNotification(data) {
- trace("Server notification: " + data);
- var parsed = data.split(",");
- if (parseInt(parsed[2]) != 0)
- other_peers[parseInt(parsed[1])] = parsed[0];
-}
-
-function handlePeerMessage(peer_id, msg) {
- trace("Received message from " + other_peers[peer_id] +
- " (ID " + peer_id + "):\n" + msg);
- // Assuming we have created a PeerConnection and that we receive the message
- // from the peer we want to communicate with
- document.getElementById("peer_id").value = peer_id;
- if (pc) {
- try {
- pc.processSignalingMessage(msg);
- } catch (e) {
- trace_warning("Signaling message error: " + e.description);
- }
- if (call_state == 1)
- setCallState(3);
- } else {
- createPeerConnection();
- pc.processSignalingMessage(msg);
- }
-}
-
-function getIntHeader(r, name) {
- var val = r.getResponseHeader(name);
- trace("header value: " + val);
- return val != null && val.length ? parseInt(val) : -1;
-}
-
-function hangingGetCallback() {
- try {
- if (hanging_get.readyState != 4)
- return;
- if (hanging_get.status != 200) {
- trace("server error, status: " + hanging_get.status +
- ", text: " + hanging_get.statusText);
- disconnect();
- } else {
- var peer_id = getIntHeader(hanging_get, "Pragma");
- if (peer_id == my_id) {
- handleServerNotification(hanging_get.responseText);
- } else {
- handlePeerMessage(peer_id, hanging_get.responseText);
- }
- }
-
- if (hanging_get) {
- hanging_get.abort();
- hanging_get = null;
- }
-
- if (my_id != -1)
- window.setTimeout(startHangingGet, 0);
- } catch (e) {
- trace("Hanging get error: " + e.description);
- }
-}
-
-function onHangingGetTimeout() {
- trace("hanging get timeout. issuing again.");
- hanging_get.abort();
- hanging_get = null;
- if (my_id != -1)
- window.setTimeout(startHangingGet, 0);
-}
-
-function startHangingGet() {
- try {
- hanging_get = new XMLHttpRequest();
- hanging_get.onreadystatechange = hangingGetCallback;
- hanging_get.ontimeout = onHangingGetTimeout;
- hanging_get.open("GET", server + "/wait?peer_id=" + my_id, true);
- hanging_get.send();
- } catch (e) {
- trace("error" + e.description);
- }
-}
-
-function sendToPeer(peer_id, data) {
- if (my_id == -1) {
- alert("Not connected");
- return;
- }
- if (peer_id == my_id) {
- alert("Can't send a message to oneself :)");
- return;
- }
- var r = new XMLHttpRequest();
- r.open("POST", server + "/message?peer_id=" + my_id + "&to=" + peer_id,
- false);
- r.setRequestHeader("Content-Type", "text/plain");
- r.send(data);
- r = null;
-}
-
-function signInCallback() {
- try {
- if (request.readyState == 4) {
- if (request.status == 200) {
- var peers = request.responseText.split("\n");
- my_id = parseInt(peers[0].split(",")[1]);
- trace("My id: " + my_id);
- for (var i = 1; i < peers.length; ++i) {
- if (peers[i].length > 0) {
- trace("Peer " + i + ": " + peers[i]);
- var parsed = peers[i].split(",");
- other_peers[parseInt(parsed[1])] = parsed[0];
- }
- }
- startHangingGet();
- request = null;
- }
- }
- } catch (e) {
- trace("error: " + e.description);
- }
-}
-
-function signIn() {
- try {
- request = new XMLHttpRequest();
- request.onreadystatechange = signInCallback;
- request.open("GET", server + "/sign_in?" + local_name, true);
- request.send();
- } catch (e) {
- trace("error: " + e.description);
- }
-}
-
-function connect() {
- local_name = document.getElementById("local").value.toLowerCase();
- server = document.getElementById("server").value.toLowerCase();
- if (local_name.length == 0) {
- alert("I need a name please.");
- document.getElementById("local").focus();
- } else {
- document.getElementById("connect").disabled = true;
- document.getElementById("disconnect").disabled = false;
- signIn();
- }
-}
-
-function disconnect() {
- if (request) {
- request.abort();
- request = null;
- }
-
- if (hanging_get) {
- hanging_get.abort();
- hanging_get = null;
- }
-
- if (my_id != -1) {
- request = new XMLHttpRequest();
- request.open("GET", server + "/sign_out?peer_id=" + my_id, false);
- request.send();
- request = null;
- my_id = -1;
- }
-
- document.getElementById("connect").disabled = false;
- document.getElementById("disconnect").disabled = true;
-}
-
-function toggleLog()
-{
- document.getElementById('debug').hidden =!
- document.getElementById('debug').hidden;
-}
-
-window.onbeforeunload = disconnect;
-
-</script>
-</head>
-
-<body>
-<table border="0">
-
-<tr>
-<th align="left">Local Preview</th>
-<th align="left">Remote Video</th>
-</tr>
-<tr>
-<td><video width="320" height="240" id="localView" autoplay="autoplay">
- </video></td>
-<td><video width="640" height="480" id="remoteView" autoplay="autoplay">
- </video></td>
-</tr>
-<tr>
-<td valign="top">
-<form>
-Target peer ID: <input type="text" id="peer_id" size="3" value="1"/><br>
-<button id="startcall" onclick="startCall();">(4) Start call</button><br>
-<button id="stopcall" onclick="stopCall();" disabled="true">(5) Stop call
- </button><br>
-<!--<input type="button" value="Test 1" onclick="test1()"/><br>-->
-</form>
-</td>
-<td valign="top">
-<button onclick="toggleExtraButtons();">Toggle extra buttons</button><br>
-<button id="createPcBtn" onclick="createPeerConnection();" hidden="true">
- Create peer connection</button><br>
-<button id="addStreamBtn" onclick="addStream();" hidden="true">Add stream
- </button><br>
-</td>
-</tr>
-
-<tr><td colspan="2">
-(1) Server: <input type="text" id="server" value="http://localhost:8888" />
-<br>
-(2) Your name: <input type="text" id="local" value="responder"/>
-<button id="connect" onclick="connect();" disabled="true">(3) Connect</button>
-<button id="disconnect" onclick="disconnect();" disabled="true">(6) Disconnect
- </button>
-<br>
- <br>
-<button onclick="toggleLog()">Toggle view log</button>
-<button onclick="document.getElementById('debug').innerHTML='';">Clear log
- </button>
-<pre id="debug">
-</pre>
-</td></tr>
-
-</table>
-</body>
-<script>
-// On init, we connect to the server.
-connect();
-</script>
-</html>
diff --git a/trunk/test/test.gyp b/trunk/test/test.gyp
deleted file mode 100644
index b428bf3..0000000
--- a/trunk/test/test.gyp
+++ /dev/null
@@ -1,79 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-# TODO(andrew): consider moving test_support to src/base/test.
-{
- 'includes': [
- '../src/build/common.gypi',
- ],
- 'targets': [
- {
- 'target_name': 'test_support',
- 'type': 'static_library',
- 'include_dirs': [
- '.',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '.', # Some includes are hierarchical
- ],
- },
- 'dependencies': [
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- '<(webrtc_root)/../testing/gmock.gyp:gmock',
- ],
- 'all_dependent_settings': {
- 'include_dirs': [
- '.',
- ],
- },
- 'sources': [
- 'test_suite.cc',
- 'test_suite.h',
- 'testsupport/fileutils.h',
- 'testsupport/fileutils.cc',
- 'testsupport/frame_reader.h',
- 'testsupport/frame_reader.cc',
- 'testsupport/frame_writer.h',
- 'testsupport/frame_writer.cc',
- 'testsupport/gtest_prod_util.h',
- 'testsupport/packet_reader.h',
- 'testsupport/packet_reader.cc',
- 'testsupport/mock/mock_frame_reader.h',
- 'testsupport/mock/mock_frame_writer.h',
- ],
- },
- {
- # Depend on this target when you want to have test_support but also the
- # main method needed for gtest to execute!
- 'target_name': 'test_support_main',
- 'type': 'static_library',
- 'dependencies': [
- 'test_support',
- ],
- 'sources': [
- 'run_all_unittests.cc',
- ],
- },
- {
- 'target_name': 'test_support_unittests',
- 'type': 'executable',
- 'dependencies': [
- 'test_support_main',
- '<(webrtc_root)/../testing/gtest.gyp:gtest',
- ],
- 'sources': [
- 'testsupport/unittest_utils.h',
- 'testsupport/fileutils_unittest.cc',
- 'testsupport/frame_reader_unittest.cc',
- 'testsupport/frame_writer_unittest.cc',
- 'testsupport/packet_reader_unittest.cc',
- ],
- },
- ],
-}
diff --git a/trunk/test/test_suite.cc b/trunk/test/test_suite.cc
deleted file mode 100644
index ac3f3a2..0000000
--- a/trunk/test/test_suite.cc
+++ /dev/null
@@ -1,39 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "test/test_suite.h"
-
-#include "gmock/gmock.h"
-#include "gtest/gtest.h"
-
-namespace webrtc {
-namespace test {
-TestSuite::TestSuite(int argc, char** argv) {
- testing::InitGoogleMock(&argc, argv); // Runs InitGoogleTest() internally.
-}
-
-TestSuite::~TestSuite() {
-}
-
-int TestSuite::Run() {
- Initialize();
- int result = RUN_ALL_TESTS();
- Shutdown();
- return result;
-}
-
-void TestSuite::Initialize() {
- // TODO(andrew): initialize singletons here (e.g. Trace).
-}
-
-void TestSuite::Shutdown() {
-}
-} // namespace test
-} // namespace webrtc
diff --git a/trunk/test/test_suite.h b/trunk/test/test_suite.h
deleted file mode 100644
index f500daa..0000000
--- a/trunk/test/test_suite.h
+++ /dev/null
@@ -1,42 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef TEST_TEST_SUITE_H_
-#define TEST_TEST_SUITE_H_
-
-// Derived from Chromium's src/base/test/test_suite.h.
-
-// Defines a basic test suite framework for running gtest based tests. You can
-// instantiate this class in your main function and call its Run method to run
-// any gtest based tests that are linked into your executable.
-
-#include "src/system_wrappers/interface/constructor_magic.h"
-
-namespace webrtc {
-namespace test {
-class TestSuite {
- public:
- TestSuite(int argc, char** argv);
- virtual ~TestSuite();
-
- int Run();
-
- protected:
- // Override these for custom initialization and shutdown handling. Use these
- // instead of putting complex code in your constructor/destructor.
- virtual void Initialize();
- virtual void Shutdown();
-
- DISALLOW_COPY_AND_ASSIGN(TestSuite);
-};
-} // namespace test
-} // namespace webrtc
-
-#endif // TEST_TEST_SUITE_H_
diff --git a/trunk/test/testsupport/fileutils.cc b/trunk/test/testsupport/fileutils.cc
deleted file mode 100644
index 1e6bbca..0000000
--- a/trunk/test/testsupport/fileutils.cc
+++ /dev/null
@@ -1,167 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testsupport/fileutils.h"
-
-#ifdef WIN32
-#include <direct.h>
-#define GET_CURRENT_DIR _getcwd
-#else
-#include <unistd.h>
-#define GET_CURRENT_DIR getcwd
-#endif
-
-#include <sys/stat.h> // To check for directory existence.
-#ifndef S_ISDIR // Not defined in stat.h on Windows.
-#define S_ISDIR(mode) (((mode) & S_IFMT) == S_IFDIR)
-#endif
-
-#include <cstdio>
-
-#include "typedefs.h" // For architecture defines
-
-namespace webrtc {
-namespace test {
-
-#ifdef WIN32
-static const char* kPathDelimiter = "\\";
-#else
-static const char* kPathDelimiter = "/";
-#endif
-// The file we're looking for to identify the project root dir.
-static const char* kProjectRootFileName = "DEPS";
-static const char* kOutputDirName = "out";
-static const char* kFallbackPath = "./";
-static const char* kResourcesDirName = "resources";
-const char* kCannotFindProjectRootDir = "ERROR_CANNOT_FIND_PROJECT_ROOT_DIR";
-
-std::string ProjectRootPath() {
- std::string working_dir = WorkingDir();
- if (working_dir == kFallbackPath) {
- return kCannotFindProjectRootDir;
- }
- // Check for our file that verifies the root dir.
- std::string current_path(working_dir);
- FILE* file = NULL;
- int path_delimiter_index = current_path.find_last_of(kPathDelimiter);
- while (path_delimiter_index > -1) {
- std::string root_filename = current_path + kPathDelimiter +
- kProjectRootFileName;
- file = fopen(root_filename.c_str(), "r");
- if (file != NULL) {
- fclose(file);
- return current_path + kPathDelimiter;
- }
- // Move up one directory in the directory tree.
- current_path = current_path.substr(0, path_delimiter_index);
- path_delimiter_index = current_path.find_last_of(kPathDelimiter);
- }
- // Reached the root directory.
- fprintf(stderr, "Cannot find project root directory!\n");
- return kCannotFindProjectRootDir;
-}
-
-std::string OutputPath() {
- std::string path = ProjectRootPath();
- if (path == kCannotFindProjectRootDir) {
- return kFallbackPath;
- }
- path += kOutputDirName;
- if (!CreateDirectory(path)) {
- return kFallbackPath;
- }
- return path + kPathDelimiter;
-}
-
-std::string WorkingDir() {
- char path_buffer[FILENAME_MAX];
- if (!GET_CURRENT_DIR(path_buffer, sizeof(path_buffer))) {
- fprintf(stderr, "Cannot get current directory!\n");
- return kFallbackPath;
- } else {
- return std::string(path_buffer);
- }
-}
-
-bool CreateDirectory(std::string directory_name) {
- struct stat path_info = {0};
- // Check if the path exists already:
- if (stat(directory_name.c_str(), &path_info) == 0) {
- if (!S_ISDIR(path_info.st_mode)) {
- fprintf(stderr, "Path %s exists but is not a directory! Remove this "
- "file and re-run to create the directory.\n",
- directory_name.c_str());
- return false;
- }
- } else {
-#ifdef WIN32
- return _mkdir(directory_name.c_str()) == 0;
-#else
- return mkdir(directory_name.c_str(), S_IRWXU | S_IRWXG | S_IRWXO) == 0;
-#endif
- }
- return true;
-}
-
-bool FileExists(std::string file_name) {
- struct stat file_info = {0};
- return stat(file_name.c_str(), &file_info) == 0;
-}
-
-std::string ResourcePath(std::string name, std::string extension) {
- std::string platform = "win";
-#ifdef WEBRTC_LINUX
- platform = "linux";
-#endif // WEBRTC_LINUX
-#ifdef WEBRTC_MAC
- platform = "mac";
-#endif // WEBRTC_MAC
-
-#ifdef WEBRTC_ARCH_64_BITS
- std::string architecture = "64";
-#else
- std::string architecture = "32";
-#endif // WEBRTC_ARCH_64_BITS
-
- std::string resources_path = ProjectRootPath() + kResourcesDirName +
- kPathDelimiter;
- std::string resource_file = resources_path + name + "_" + platform + "_" +
- architecture + "." + extension;
- if (FileExists(resource_file)) {
- return resource_file;
- }
- // Try without architecture.
- resource_file = resources_path + name + "_" + platform + "." + extension;
- if (FileExists(resource_file)) {
- return resource_file;
- }
- // Try without platform.
- resource_file = resources_path + name + "_" + architecture + "." + extension;
- if (FileExists(resource_file)) {
- return resource_file;
- }
- // Fall back on name without architecture or platform.
- return resources_path + name + "." + extension;
-}
-
-size_t GetFileSize(std::string filename) {
- FILE* f = fopen(filename.c_str(), "rb");
- size_t size = 0;
- if (f != NULL) {
- if (fseek(f, 0, SEEK_END) == 0) {
- size = ftell(f);
- }
- fclose(f);
- }
- return size;
-}
-
-} // namespace test
-} // namespace webrtc
diff --git a/trunk/test/testsupport/fileutils.h b/trunk/test/testsupport/fileutils.h
deleted file mode 100644
index c89ac29..0000000
--- a/trunk/test/testsupport/fileutils.h
+++ /dev/null
@@ -1,143 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include <cstdio>
-
-// File utilities for testing purposes.
-//
-// The ProjectRootPath() method is a convenient way of getting an absolute
-// path to the project source tree root directory. Using this, it is easy to
-// refer to test resource files in a portable way.
-//
-// Notice that even if Windows platforms use backslash as path delimiter, it is
-// also supported to use slash, so there's no need for #ifdef checks in test
-// code for setting up the paths to the resource files.
-//
-// Example use:
-// Assume we have the following code being used in a test source file:
-// const std::string kInputFile = webrtc::test::ProjectRootPath() +
-// "test/data/voice_engine/audio_long16.wav";
-// // Use the kInputFile for the tests...
-//
-// Then here's some example outputs for different platforms:
-// Linux:
-// * Source tree located in /home/user/webrtc/trunk
-// * Test project located in /home/user/webrtc/trunk/src/testproject
-// * Test binary compiled as:
-// /home/user/webrtc/trunk/out/Debug/testproject_unittests
-// Then ProjectRootPath() will return /home/user/webrtc/trunk/ no matter if
-// the test binary is executed from standing in either of:
-// /home/user/webrtc/trunk
-// or
-// /home/user/webrtc/trunk/out/Debug
-// (or any other directory below the trunk for that matter).
-//
-// Windows:
-// * Source tree located in C:\Users\user\webrtc\trunk
-// * Test project located in C:\Users\user\webrtc\trunk\src\testproject
-// * Test binary compiled as:
-// C:\Users\user\webrtc\trunk\src\testproject\Debug\testproject_unittests.exe
-// Then ProjectRootPath() will return C:\Users\user\webrtc\trunk\ when the
-// test binary is executed from inside Visual Studio.
-// It will also return the same path if the test is executed from a command
-// prompt standing in C:\Users\user\webrtc\trunk\src\testproject\Debug
-//
-// Mac:
-// * Source tree located in /Users/user/webrtc/trunk
-// * Test project located in /Users/user/webrtc/trunk/src/testproject
-// * Test binary compiled as:
-// /Users/user/webrtc/trunk/xcodebuild/Debug/testproject_unittests
-// Then ProjectRootPath() will return /Users/user/webrtc/trunk/ no matter if
-// the test binary is executed from standing in either of:
-// /Users/user/webrtc/trunk
-// or
-// /Users/user/webrtc/trunk/out/Debug
-// (or any other directory below the trunk for that matter).
-
-#ifndef WEBRTC_TEST_TESTSUPPORT_FILEUTILS_H_
-#define WEBRTC_TEST_TESTSUPPORT_FILEUTILS_H_
-
-#include <string>
-
-namespace webrtc {
-namespace test {
-
-// This is the "directory" returned if the ProjectPath() function fails
-// to find the project root.
-extern const char* kCannotFindProjectRootDir;
-
-// Finds the root dir of the project, to be able to set correct paths to
-// resource files used by tests.
-// The implementation is simple: it just looks for the file defined by
-// kProjectRootFileName, starting in the current directory (the working
-// directory) and then steps upward until it is found (or it is at the root of
-// the file system).
-// If the current working directory is above the project root dir, it will not
-// be found.
-//
-// If symbolic links occur in the path they will be resolved and the actual
-// directory will be returned.
-//
-// Returns the absolute path to the project root dir (usually the trunk dir)
-// WITH a trailing path delimiter.
-// If the project root is not found, the string specified by
-// kCannotFindProjectRootDir is returned.
-std::string ProjectRootPath();
-
-// Creates and returns the absolute path to the output directory where log files
-// and other test artifacts should be put. The output directory is always a
-// directory named "out" at the top-level of the project, i.e. a subfolder to
-// the path returned by ProjectRootPath().
-//
-// Details described for ProjectRootPath() apply here too.
-//
-// Returns the path WITH a trailing path delimiter. If the project root is not
-// found, the current working directory ("./") is returned as a fallback.
-std::string OutputPath();
-
-// Returns a path to a resource file for the currently executing platform.
-// Adapts to what filenames are currently present in the
-// [project-root]/resources/ dir.
-// Returns an absolute path according to this priority list (the directory
-// part of the path is left out for readability):
-// 1. [name]_[platform]_[architecture].[extension]
-// 2. [name]_[platform].[extension]
-// 3. [name]_[architecture].[extension]
-// 4. [name].[extension]
-// Where
-// * platform is either of "win", "mac" or "linux".
-// * architecture is either of "32" or "64".
-//
-// Arguments:
-// name - Name of the resource file. If a plain filename (no directory path)
-// is supplied, the file is assumed to be located in resources/
-// If a directory path is prepended to the filename, a subdirectory
-// hierarchy reflecting that path is assumed to be present.
-// extension - File extension, without the dot, i.e. "bmp" or "yuv".
-std::string ResourcePath(std::string name, std::string extension);
-
-// Gets the current working directory for the executing program.
-// Returns "./" if for some reason it is not possible to find the working
-// directory.
-std::string WorkingDir();
-
-// Creates a directory if it not already exists.
-// Returns true if successful. Will print an error message to stderr and return
-// false if a file with the same name already exists.
-bool CreateDirectory(std::string directory_name);
-
-// File size of the supplied file in bytes. Will return 0 if the file is
-// empty or if the file does not exist/is readable.
-size_t GetFileSize(std::string filename);
-
-} // namespace test
-} // namespace webrtc
-
-#endif // WEBRTC_TEST_TESTSUPPORT_FILEUTILS_H_
diff --git a/trunk/test/testsupport/fileutils_unittest.cc b/trunk/test/testsupport/fileutils_unittest.cc
deleted file mode 100644
index 1b76b3c..0000000
--- a/trunk/test/testsupport/fileutils_unittest.cc
+++ /dev/null
@@ -1,192 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testsupport/fileutils.h"
-
-#include <cstdio>
-#include <list>
-#include <string>
-
-#include "gtest/gtest.h"
-
-#ifdef WIN32
-#define chdir _chdir
-static const char* kPathDelimiter = "\\";
-#else
-static const char* kPathDelimiter = "/";
-#endif
-
-static const std::string kDummyDir = "file_utils_unittest_dummy_dir";
-static const std::string kResourcesDir = "resources";
-static const std::string kTestName = "fileutils_unittest";
-static const std::string kExtension = "tmp";
-
-typedef std::list<std::string> FileList;
-
-namespace webrtc {
-
-// Test fixture to restore the working directory between each test, since some
-// of them change it with chdir during execution (not restored by the
-// gtest framework).
-class FileUtilsTest : public testing::Test {
- protected:
- FileUtilsTest() {
- }
- virtual ~FileUtilsTest() {}
- // Runs before the first test
- static void SetUpTestCase() {
- original_working_dir_ = webrtc::test::WorkingDir();
- std::string resources_path = original_working_dir_ + kPathDelimiter +
- kResourcesDir + kPathDelimiter;
- webrtc::test::CreateDirectory(resources_path);
-
- files_.push_back(resources_path + kTestName + "." + kExtension);
- files_.push_back(resources_path + kTestName + "_32." + kExtension);
- files_.push_back(resources_path + kTestName + "_64." + kExtension);
- files_.push_back(resources_path + kTestName + "_linux." + kExtension);
- files_.push_back(resources_path + kTestName + "_mac." + kExtension);
- files_.push_back(resources_path + kTestName + "_win." + kExtension);
- files_.push_back(resources_path + kTestName + "_linux_32." + kExtension);
- files_.push_back(resources_path + kTestName + "_mac_32." + kExtension);
- files_.push_back(resources_path + kTestName + "_win_32." + kExtension);
- files_.push_back(resources_path + kTestName + "_linux_64." + kExtension);
- files_.push_back(resources_path + kTestName + "_mac_64." + kExtension);
- files_.push_back(resources_path + kTestName + "_win_64." + kExtension);
-
- // Now that the resources dir exists, write some empty test files into it.
- for (FileList::iterator file_it = files_.begin();
- file_it != files_.end(); ++file_it) {
- FILE* file = fopen(file_it->c_str(), "wb");
- ASSERT_TRUE(file != NULL) << "Failed to write file: " << file_it->c_str();
- ASSERT_GT(fprintf(file, "%s", "Dummy data"), 0);
- fclose(file);
- }
- // Create a dummy subdir that can be chdir'ed into for testing purposes.
- empty_dummy_dir_ = original_working_dir_ + kPathDelimiter + kDummyDir;
- webrtc::test::CreateDirectory(empty_dummy_dir_);
- }
- static void TearDownTestCase() {
- // Clean up all resource files written
- for (FileList::iterator file_it = files_.begin();
- file_it != files_.end(); ++file_it) {
- remove(file_it->c_str());
- }
- std::remove(empty_dummy_dir_.c_str());
- }
- void SetUp() {
- ASSERT_EQ(chdir(original_working_dir_.c_str()), 0);
- }
- void TearDown() {
- ASSERT_EQ(chdir(original_working_dir_.c_str()), 0);
- }
- protected:
- static FileList files_;
- static std::string empty_dummy_dir_;
- private:
- static std::string original_working_dir_;
-};
-
-FileList FileUtilsTest::files_;
-std::string FileUtilsTest::original_working_dir_ = "";
-std::string FileUtilsTest::empty_dummy_dir_ = "";
-
-// Tests that the project root path is returned for the default working
-// directory that is automatically set when the test executable is launched.
-// The test is not fully testing the implementation, since we cannot be sure
-// of where the executable was launched from.
-// The test will fail if the top level directory is not named "trunk".
-TEST_F(FileUtilsTest, ProjectRootPathFromUnchangedWorkingDir) {
- std::string path = webrtc::test::ProjectRootPath();
- std::string expected_end = "trunk";
- expected_end = kPathDelimiter + expected_end + kPathDelimiter;
- ASSERT_EQ(path.length() - expected_end.length(), path.find(expected_end));
-}
-
-// Similar to the above test, but for the output dir
-TEST_F(FileUtilsTest, OutputPathFromUnchangedWorkingDir) {
- std::string path = webrtc::test::OutputPath();
- std::string expected_end = "out";
- expected_end = kPathDelimiter + expected_end + kPathDelimiter;
- ASSERT_EQ(path.length() - expected_end.length(), path.find(expected_end));
-}
-
-// Tests setting the current working directory to a directory three levels
-// deeper from the current one. Then testing that the project path returned
-// is still the same, when the function under test is called again.
-TEST_F(FileUtilsTest, ProjectRootPathFromDeeperWorkingDir) {
- std::string path = webrtc::test::ProjectRootPath();
- std::string original_working_dir = path; // This is the correct project root
- // Change to a subdirectory path.
- ASSERT_EQ(0, chdir(empty_dummy_dir_.c_str()));
- ASSERT_EQ(original_working_dir, webrtc::test::ProjectRootPath());
-}
-
-// Similar to the above test, but for the output dir
-TEST_F(FileUtilsTest, OutputPathFromDeeperWorkingDir) {
- std::string path = webrtc::test::OutputPath();
- std::string original_working_dir = path;
- ASSERT_EQ(0, chdir(empty_dummy_dir_.c_str()));
- ASSERT_EQ(original_working_dir, webrtc::test::OutputPath());
-}
-
-// Tests with current working directory set to a directory higher up in the
-// directory tree than the project root dir. This case shall return a specified
-// error string as a directory (which will be an invalid path).
-TEST_F(FileUtilsTest, ProjectRootPathFromRootWorkingDir) {
- // Change current working dir to the root of the current file system
- // (this will always be "above" our project root dir).
- ASSERT_EQ(0, chdir(kPathDelimiter));
- ASSERT_EQ(webrtc::test::kCannotFindProjectRootDir,
- webrtc::test::ProjectRootPath());
-}
-
-// Similar to the above test, but for the output dir
-TEST_F(FileUtilsTest, OutputPathFromRootWorkingDir) {
- ASSERT_EQ(0, chdir(kPathDelimiter));
- ASSERT_EQ("./", webrtc::test::OutputPath());
-}
-
-// Only tests that the code executes
-TEST_F(FileUtilsTest, CreateDirectory) {
- std::string directory = "fileutils-unittest-empty-dir";
- // Make sure it's removed if a previous test has failed:
- std::remove(directory.c_str());
- ASSERT_TRUE(webrtc::test::CreateDirectory(directory));
- std::remove(directory.c_str());
-}
-
-TEST_F(FileUtilsTest, WorkingDirReturnsValue) {
- // Hard to cover all platforms. Just test that it returns something without
- // crashing:
- std::string working_dir = webrtc::test::WorkingDir();
- ASSERT_GT(working_dir.length(), 0u);
-}
-
-// Due to multiple platforms, it is hard to make a complete test for
-// ResourcePath. Manual testing has been performed by removing files and
-// verified the result confirms with the specified documentation for the
-// function.
-TEST_F(FileUtilsTest, ResourcePathReturnsValue) {
- std::string resource = webrtc::test::ResourcePath(kTestName, kExtension);
- ASSERT_GT(resource.find(kTestName), 0u);
- ASSERT_GT(resource.find(kExtension), 0u);
- ASSERT_EQ(0, chdir(kPathDelimiter));
- ASSERT_EQ("./", webrtc::test::OutputPath());
-}
-
-TEST_F(FileUtilsTest, GetFileSizeExistingFile) {
- ASSERT_GT(webrtc::test::GetFileSize(files_.front()), 0u);
-}
-
-TEST_F(FileUtilsTest, GetFileSizeNonExistingFile) {
- ASSERT_EQ(0u, webrtc::test::GetFileSize("non-existing-file.tmp"));
-}
-
-} // namespace webrtc
diff --git a/trunk/test/testsupport/frame_reader.cc b/trunk/test/testsupport/frame_reader.cc
deleted file mode 100644
index b05ea58..0000000
--- a/trunk/test/testsupport/frame_reader.cc
+++ /dev/null
@@ -1,81 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testsupport/frame_reader.h"
-
-#include <cassert>
-
-#include "testsupport/fileutils.h"
-
-namespace webrtc {
-namespace test {
-
-FrameReaderImpl::FrameReaderImpl(std::string input_filename,
- int frame_length_in_bytes)
- : input_filename_(input_filename),
- frame_length_in_bytes_(frame_length_in_bytes),
- input_file_(NULL) {
-}
-
-FrameReaderImpl::~FrameReaderImpl() {
- Close();
-}
-
-bool FrameReaderImpl::Init() {
- if (frame_length_in_bytes_ <= 0) {
- fprintf(stderr, "Frame length must be >0, was %d\n",
- frame_length_in_bytes_);
- return false;
- }
- input_file_ = fopen(input_filename_.c_str(), "rb");
- if (input_file_ == NULL) {
- fprintf(stderr, "Couldn't open input file for reading: %s\n",
- input_filename_.c_str());
- return false;
- }
- // Calculate total number of frames.
- size_t source_file_size = GetFileSize(input_filename_);
- if (source_file_size <= 0u) {
- fprintf(stderr, "Found empty file: %s\n", input_filename_.c_str());
- return false;
- }
- number_of_frames_ = source_file_size / frame_length_in_bytes_;
- return true;
-}
-
-void FrameReaderImpl::Close() {
- if (input_file_ != NULL) {
- fclose(input_file_);
- input_file_ = NULL;
- }
-}
-
-bool FrameReaderImpl::ReadFrame(WebRtc_UWord8* source_buffer) {
- assert(source_buffer);
- if (input_file_ == NULL) {
- fprintf(stderr, "FrameReader is not initialized (input file is NULL)\n");
- return false;
- }
- size_t nbr_read = fread(source_buffer, 1, frame_length_in_bytes_,
- input_file_);
- if (nbr_read != static_cast<unsigned int>(frame_length_in_bytes_) &&
- ferror(input_file_)) {
- fprintf(stderr, "Error reading from input file: %s\n",
- input_filename_.c_str());
- return false;
- }
- if (feof(input_file_) != 0) {
- return false; // No more frames to process.
- }
- return true;
-}
-
-} // namespace test
-} // namespace webrtc
diff --git a/trunk/test/testsupport/frame_reader.h b/trunk/test/testsupport/frame_reader.h
deleted file mode 100644
index 56d8fc4..0000000
--- a/trunk/test/testsupport/frame_reader.h
+++ /dev/null
@@ -1,73 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_TEST_TESTSUPPORT_FRAME_READER_H_
-#define WEBRTC_TEST_TESTSUPPORT_FRAME_READER_H_
-
-#include <cstdio>
-#include <string>
-
-#include "typedefs.h"
-
-namespace webrtc {
-namespace test {
-
-// Handles reading of frames from video files.
-class FrameReader {
- public:
- virtual ~FrameReader() {}
-
- // Initializes the frame reader, i.e. opens the input file.
- // This must be called before reading of frames has started.
- // Returns false if an error has occurred, in addition to printing to stderr.
- virtual bool Init() = 0;
-
- // Reads a frame into the supplied buffer, which must contain enough space
- // for the frame size.
- // Returns true if there are more frames to read, false if we've already
- // read the last frame (in the previous call).
- virtual bool ReadFrame(WebRtc_UWord8* source_buffer) = 0;
-
- // Closes the input file if open. Essentially makes this class impossible
- // to use anymore. Will also be invoked by the destructor.
- virtual void Close() = 0;
-
- // Frame length in bytes of a single frame image.
- virtual int FrameLength() = 0;
- // Total number of frames in the input video source.
- virtual int NumberOfFrames() = 0;
-};
-
-class FrameReaderImpl : public FrameReader {
- public:
- // Creates a file handler. The input file is assumed to exist and be readable.
- // Parameters:
- // input_filename The file to read from.
- // frame_length_in_bytes The size of each frame.
- // For YUV this is 3 * width * height / 2
- FrameReaderImpl(std::string input_filename, int frame_length_in_bytes);
- virtual ~FrameReaderImpl();
- bool Init();
- bool ReadFrame(WebRtc_UWord8* source_buffer);
- void Close();
- int FrameLength() { return frame_length_in_bytes_; }
- int NumberOfFrames() { return number_of_frames_; }
-
- private:
- std::string input_filename_;
- int frame_length_in_bytes_;
- int number_of_frames_;
- FILE* input_file_;
-};
-
-} // namespace test
-} // namespace webrtc
-
-#endif // WEBRTC_TEST_TESTSUPPORT_FRAME_READER_H_
diff --git a/trunk/test/testsupport/frame_reader_unittest.cc b/trunk/test/testsupport/frame_reader_unittest.cc
deleted file mode 100644
index f1da5ce..0000000
--- a/trunk/test/testsupport/frame_reader_unittest.cc
+++ /dev/null
@@ -1,72 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testsupport/frame_reader.h"
-
-#include "gtest/gtest.h"
-#include "testsupport/fileutils.h"
-
-namespace webrtc {
-namespace test {
-
-const std::string kInputFilename = "temp_inputfile.tmp";
-const std::string kInputFileContents = "baz";
-// Setting the kFrameLength value to a value much larger than the
-// file to test causes the ReadFrame test to fail on Windows.
-const int kFrameLength = 1000;
-
-class FrameReaderTest: public testing::Test {
- protected:
- FrameReaderTest() {}
- virtual ~FrameReaderTest() {}
- void SetUp() {
- // Cleanup any previous dummy input file.
- std::remove(kInputFilename.c_str());
-
- // Create a dummy input file.
- FILE* dummy = fopen(kInputFilename.c_str(), "wb");
- fprintf(dummy, "%s", kInputFileContents.c_str());
- fclose(dummy);
-
- frame_reader_ = new FrameReaderImpl(kInputFilename, kFrameLength);
- ASSERT_TRUE(frame_reader_->Init());
- }
- void TearDown() {
- delete frame_reader_;
- // Cleanup the dummy input file.
- std::remove(kInputFilename.c_str());
- }
- FrameReader* frame_reader_;
-};
-
-TEST_F(FrameReaderTest, InitSuccess) {
- FrameReaderImpl frame_reader(kInputFilename, kFrameLength);
- ASSERT_TRUE(frame_reader.Init());
- ASSERT_EQ(kFrameLength, frame_reader.FrameLength());
- ASSERT_EQ(0, frame_reader.NumberOfFrames());
-}
-
-TEST_F(FrameReaderTest, ReadFrame) {
- WebRtc_UWord8 buffer[3];
- bool result = frame_reader_->ReadFrame(buffer);
- ASSERT_FALSE(result); // No more files to read.
- ASSERT_EQ(kInputFileContents[0], buffer[0]);
- ASSERT_EQ(kInputFileContents[1], buffer[1]);
- ASSERT_EQ(kInputFileContents[2], buffer[2]);
-}
-
-TEST_F(FrameReaderTest, ReadFrameUninitialized) {
- WebRtc_UWord8 buffer[3];
- FrameReaderImpl file_reader(kInputFilename, kFrameLength);
- ASSERT_FALSE(file_reader.ReadFrame(buffer));
-}
-
-} // namespace test
-} // namespace webrtc
diff --git a/trunk/test/testsupport/frame_writer.cc b/trunk/test/testsupport/frame_writer.cc
deleted file mode 100644
index 5f32539..0000000
--- a/trunk/test/testsupport/frame_writer.cc
+++ /dev/null
@@ -1,68 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testsupport/frame_writer.h"
-
-#include <cassert>
-
-namespace webrtc {
-namespace test {
-
-FrameWriterImpl::FrameWriterImpl(std::string output_filename,
- int frame_length_in_bytes)
- : output_filename_(output_filename),
- frame_length_in_bytes_(frame_length_in_bytes),
- output_file_(NULL) {
-}
-
-FrameWriterImpl::~FrameWriterImpl() {
- Close();
-}
-
-bool FrameWriterImpl::Init() {
- if (frame_length_in_bytes_ <= 0) {
- fprintf(stderr, "Frame length must be >0, was %d\n",
- frame_length_in_bytes_);
- return false;
- }
- output_file_ = fopen(output_filename_.c_str(), "wb");
- if (output_file_ == NULL) {
- fprintf(stderr, "Couldn't open output file for writing: %s\n",
- output_filename_.c_str());
- return false;
- }
- return true;
-}
-
-void FrameWriterImpl::Close() {
- if (output_file_ != NULL) {
- fclose(output_file_);
- output_file_ = NULL;
- }
-}
-
-bool FrameWriterImpl::WriteFrame(WebRtc_UWord8* frame_buffer) {
- assert(frame_buffer);
- if (output_file_ == NULL) {
- fprintf(stderr, "FrameWriter is not initialized (output file is NULL)\n");
- return false;
- }
- int bytes_written = fwrite(frame_buffer, 1, frame_length_in_bytes_,
- output_file_);
- if (bytes_written != frame_length_in_bytes_) {
- fprintf(stderr, "Failed to write %d bytes to file %s\n",
- frame_length_in_bytes_, output_filename_.c_str());
- return false;
- }
- return true;
-}
-
-} // namespace test
-} // namespace webrtc
diff --git a/trunk/test/testsupport/frame_writer.h b/trunk/test/testsupport/frame_writer.h
deleted file mode 100644
index abc5d35..0000000
--- a/trunk/test/testsupport/frame_writer.h
+++ /dev/null
@@ -1,70 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_TEST_TESTSUPPORT_FRAME_WRITER_H_
-#define WEBRTC_TEST_TESTSUPPORT_FRAME_WRITER_H_
-
-#include <cstdio>
-#include <string>
-
-#include "typedefs.h"
-
-namespace webrtc {
-namespace test {
-
-// Handles writing of video files.
-class FrameWriter {
- public:
- virtual ~FrameWriter() {}
-
- // Initializes the file handler, i.e. opens the input and output files etc.
- // This must be called before reading or writing frames has started.
- // Returns false if an error has occurred, in addition to printing to stderr.
- virtual bool Init() = 0;
-
- // Writes a frame of the configured frame length to the output file.
- // Returns true if the write was successful, false otherwise.
- virtual bool WriteFrame(WebRtc_UWord8* frame_buffer) = 0;
-
- // Closes the output file if open. Essentially makes this class impossible
- // to use anymore. Will also be invoked by the destructor.
- virtual void Close() = 0;
-
- // Frame length in bytes of a single frame image.
- virtual int FrameLength() = 0;
-};
-
-class FrameWriterImpl : public FrameWriter {
- public:
- // Creates a file handler. The input file is assumed to exist and be readable
- // and the output file must be writable.
- // Parameters:
- // output_filename The file to write. Will be overwritten if already
- // existing.
- // frame_length_in_bytes The size of each frame.
- // For YUV: 3*width*height/2
- FrameWriterImpl(std::string output_filename, int frame_length_in_bytes);
- virtual ~FrameWriterImpl();
- bool Init();
- bool WriteFrame(WebRtc_UWord8* frame_buffer);
- void Close();
- int FrameLength() { return frame_length_in_bytes_; }
-
- private:
- std::string output_filename_;
- int frame_length_in_bytes_;
- int number_of_frames_;
- FILE* output_file_;
-};
-
-} // namespace test
-} // namespace webrtc
-
-#endif // WEBRTC_TEST_TESTSUPPORT_FRAME_WRITER_H_
diff --git a/trunk/test/testsupport/frame_writer_unittest.cc b/trunk/test/testsupport/frame_writer_unittest.cc
deleted file mode 100644
index d25d1d2..0000000
--- a/trunk/test/testsupport/frame_writer_unittest.cc
+++ /dev/null
@@ -1,64 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testsupport/frame_writer.h"
-
-#include "gtest/gtest.h"
-#include "testsupport/fileutils.h"
-
-namespace webrtc {
-namespace test {
-
-const std::string kOutputFilename = "temp_outputfile.tmp";
-const int kFrameLength = 1000;
-
-class FrameWriterTest: public testing::Test {
- protected:
- FrameWriterTest() {}
- virtual ~FrameWriterTest() {}
- void SetUp() {
- // Cleanup any previous output file.
- std::remove(kOutputFilename.c_str());
- frame_writer_ = new FrameWriterImpl(kOutputFilename, kFrameLength);
- ASSERT_TRUE(frame_writer_->Init());
- }
- void TearDown() {
- delete frame_writer_;
- // Cleanup the temporary file.
- std::remove(kOutputFilename.c_str());
- }
- FrameWriter* frame_writer_;
-};
-
-TEST_F(FrameWriterTest, InitSuccess) {
- FrameWriterImpl frame_writer(kOutputFilename, kFrameLength);
- ASSERT_TRUE(frame_writer.Init());
- ASSERT_EQ(kFrameLength, frame_writer.FrameLength());
-}
-
-TEST_F(FrameWriterTest, WriteFrame) {
- WebRtc_UWord8 buffer[kFrameLength];
- memset(buffer, 9, kFrameLength); // Write lots of 9s to the buffer
- bool result = frame_writer_->WriteFrame(buffer);
- ASSERT_TRUE(result); // success
- // Close the file and verify the size.
- frame_writer_->Close();
- ASSERT_EQ(kFrameLength,
- static_cast<int>(GetFileSize(kOutputFilename)));
-}
-
-TEST_F(FrameWriterTest, WriteFrameUninitialized) {
- WebRtc_UWord8 buffer[3];
- FrameWriterImpl frame_writer(kOutputFilename, kFrameLength);
- ASSERT_FALSE(frame_writer.WriteFrame(buffer));
-}
-
-} // namespace test
-} // namespace webrtc
diff --git a/trunk/test/testsupport/gtest_prod_util.h b/trunk/test/testsupport/gtest_prod_util.h
deleted file mode 100644
index 7d123a8..0000000
--- a/trunk/test/testsupport/gtest_prod_util.h
+++ /dev/null
@@ -1,36 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_TEST_TESTSUPPORT_GTEST_PROD_UTIL_H_
-#define WEBRTC_TEST_TESTSUPPORT_GTEST_PROD_UTIL_H_
-#pragma once
-
-#include "gtest/gtest_prod.h"
-
-// This file is a plain copy of Chromium's base/gtest_prod_util.h.
-//
-// This is a wrapper for gtest's FRIEND_TEST macro that friends
-// test with all possible prefixes. This is very helpful when changing the test
-// prefix, because the friend declarations don't need to be updated.
-//
-// Example usage:
-//
-// class MyClass {
-// private:
-// void MyMethod();
-// FRIEND_TEST_ALL_PREFIXES(MyClassTest, MyMethod);
-// };
-#define FRIEND_TEST_ALL_PREFIXES(test_case_name, test_name) \
- FRIEND_TEST(test_case_name, test_name); \
- FRIEND_TEST(test_case_name, DISABLED_##test_name); \
- FRIEND_TEST(test_case_name, FLAKY_##test_name); \
- FRIEND_TEST(test_case_name, FAILS_##test_name)
-
-#endif // WEBRTC_TEST_TESTSUPPORT_GTEST_PROD_UTIL_H_
diff --git a/trunk/test/testsupport/metrics/video_metrics.cc b/trunk/test/testsupport/metrics/video_metrics.cc
deleted file mode 100644
index 438f31b..0000000
--- a/trunk/test/testsupport/metrics/video_metrics.cc
+++ /dev/null
@@ -1,185 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testsupport/metrics/video_metrics.h"
-
-#include <algorithm> // min_element, max_element
-#include <cassert>
-#include <cstdio>
-
-#include "common_video/libyuv/include/libyuv.h"
-
-namespace webrtc {
-namespace test {
-
-// Used for calculating min and max values
-static bool LessForFrameResultValue (const FrameResult& s1,
- const FrameResult& s2) {
- return s1.value < s2.value;
-}
-
-enum VideoMetricsType { kPSNR, kSSIM, kBoth };
-
-// Calculates metrics for a frame and adds statistics to the result for it.
-void CalculateFrame(VideoMetricsType video_metrics_type,
- uint8_t* ref,
- uint8_t* test,
- int width,
- int height,
- int frame_number,
- QualityMetricsResult* result) {
- FrameResult frame_result = {0, 0};
- frame_result.frame_number = frame_number;
- switch (video_metrics_type) {
- case kPSNR:
- frame_result.value = I420PSNR(ref, test, width, height);
- break;
- case kSSIM:
- frame_result.value = I420SSIM(ref, test, width, height);
- break;
- default:
- assert(false);
- }
- result->frames.push_back(frame_result);
-}
-
-// Calculates average, min and max values for the supplied struct, if non-NULL.
-void CalculateStats(QualityMetricsResult* result) {
- if (result == NULL || result->frames.size() == 0) {
- return;
- }
- // Calculate average
- std::vector<FrameResult>::iterator iter;
- double metrics_values_sum = 0.0;
- for (iter = result->frames.begin(); iter != result->frames.end(); ++iter) {
- metrics_values_sum += iter->value;
- }
- result->average = metrics_values_sum / result->frames.size();
-
- // Calculate min/max statistics
- iter = std::min_element(result->frames.begin(), result->frames.end(),
- LessForFrameResultValue);
- result->min = iter->value;
- result->min_frame_number = iter->frame_number;
- iter = std::max_element(result->frames.begin(), result->frames.end(),
- LessForFrameResultValue);
- result->max = iter->value;
- result->max_frame_number = iter->frame_number;
-}
-
-// Single method that handles all combinations of video metrics calculation, to
-// minimize code duplication. Either psnr_result or ssim_result may be NULL,
-// depending on which VideoMetricsType is targeted.
-int CalculateMetrics(VideoMetricsType video_metrics_type,
- const char* ref_filename,
- const char* test_filename,
- int width,
- int height,
- QualityMetricsResult* psnr_result,
- QualityMetricsResult* ssim_result) {
- assert(ref_filename != NULL);
- assert(test_filename != NULL);
- assert(width > 0);
- assert(height > 0);
-
- FILE* ref_fp = fopen(ref_filename, "rb");
- if (ref_fp == NULL) {
- // cannot open reference file
- fprintf(stderr, "Cannot open file %s\n", ref_filename);
- return -1;
- }
- FILE* test_fp = fopen(test_filename, "rb");
- if (test_fp == NULL) {
- // cannot open test file
- fprintf(stderr, "Cannot open file %s\n", test_filename);
- fclose(ref_fp);
- return -2;
- }
- int frame_number = 0;
-
- // Allocating size for one I420 frame.
- const int frame_length = 3 * width * height >> 1;
- uint8_t* ref = new uint8_t[frame_length];
- uint8_t* test = new uint8_t[frame_length];
-
- int ref_bytes = fread(ref, 1, frame_length, ref_fp);
- int test_bytes = fread(test, 1, frame_length, test_fp);
- while (ref_bytes == frame_length && test_bytes == frame_length) {
- switch (video_metrics_type) {
- case kPSNR:
- CalculateFrame(kPSNR, ref, test, width, height, frame_number,
- psnr_result);
- break;
- case kSSIM:
- CalculateFrame(kSSIM, ref, test, width, height, frame_number,
- ssim_result);
- break;
- case kBoth:
- CalculateFrame(kPSNR, ref, test, width, height, frame_number,
- psnr_result);
- CalculateFrame(kSSIM, ref, test, width, height, frame_number,
- ssim_result);
- break;
- }
- frame_number++;
- ref_bytes = fread(ref, 1, frame_length, ref_fp);
- test_bytes = fread(test, 1, frame_length, test_fp);
- }
- int return_code = 0;
- if (frame_number == 0) {
- fprintf(stderr, "Tried to measure video metrics from empty files "
- "(reference file: %s test file: %s)\n", ref_filename,
- test_filename);
- return_code = -3;
- } else {
- CalculateStats(psnr_result);
- CalculateStats(ssim_result);
- }
- delete [] ref;
- delete [] test;
- fclose(ref_fp);
- fclose(test_fp);
- return return_code;
-}
-
-int I420MetricsFromFiles(const char* ref_filename,
- const char* test_filename,
- int width,
- int height,
- QualityMetricsResult* psnr_result,
- QualityMetricsResult* ssim_result) {
- assert(psnr_result != NULL);
- assert(ssim_result != NULL);
- return CalculateMetrics(kBoth, ref_filename, test_filename, width, height,
- psnr_result, ssim_result);
-}
-
-int I420PSNRFromFiles(const char* ref_filename,
- const char* test_filename,
- int width,
- int height,
- QualityMetricsResult* result) {
- assert(result != NULL);
- return CalculateMetrics(kPSNR, ref_filename, test_filename, width, height,
- result, NULL);
-}
-
-int I420SSIMFromFiles(const char* ref_filename,
- const char* test_filename,
- int width,
- int height,
- QualityMetricsResult* result) {
- assert(result != NULL);
- return CalculateMetrics(kSSIM, ref_filename, test_filename, width, height,
- NULL, result);
-}
-
-} // namespace test
-} // namespace webrtc
diff --git a/trunk/test/testsupport/metrics/video_metrics.h b/trunk/test/testsupport/metrics/video_metrics.h
deleted file mode 100644
index df11a49..0000000
--- a/trunk/test/testsupport/metrics/video_metrics.h
+++ /dev/null
@@ -1,112 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_TESTSUPPORT_METRICS_VIDEO_METRICS_H_
-#define WEBRTC_TESTSUPPORT_METRICS_VIDEO_METRICS_H_
-
-#include <limits>
-#include <vector>
-
-namespace webrtc {
-namespace test {
-
-// Contains video quality metrics result for a single frame.
-struct FrameResult {
- int frame_number;
- double value;
-};
-
-// Result from a PSNR/SSIM calculation operation.
-// The frames in this data structure are 0-indexed.
-struct QualityMetricsResult {
- QualityMetricsResult() :
- average(0.0),
- min(std::numeric_limits<double>::max()),
- max(std::numeric_limits<double>::min()),
- min_frame_number(-1),
- max_frame_number(-1)
- {};
- double average;
- double min;
- double max;
- int min_frame_number;
- int max_frame_number;
- std::vector<FrameResult> frames;
-};
-
-// Calculates PSNR and SSIM values for the reference and test video files
-// (must be in I420 format). All calculated values are filled into the
-// QualityMetricsResult stucts.
-// PSNR values have the unit decibel (dB) where a high value means the test file
-// is similar to the reference file. The higher value, the more similar.
-// For more info about PSNR, see http://en.wikipedia.org/wiki/PSNR
-// SSIM values range between -1.0 and 1.0, where 1.0 means the files are
-// identical. For more info about SSIM, see http://en.wikipedia.org/wiki/SSIM
-// This function only compares video frames up to the point when the shortest
-// video ends.
-// Return value:
-// 0 if successful, negative on errors:
-// -1 if the source file cannot be opened
-// -2 if the test file cannot be opened
-// -3 if any of the files are empty
-// -4 if any arguments are invalid.
-int I420MetricsFromFiles(const char* ref_filename,
- const char* test_filename,
- int width,
- int height,
- QualityMetricsResult* psnr_result,
- QualityMetricsResult* ssim_result);
-
-// Calculates PSNR values for the reference and test video files (must be in
-// I420 format). All calculated values are filled into the QualityMetricsResult
-// struct.
-// PSNR values have the unit decibel (dB) where a high value means the test file
-// is similar to the reference file. The higher value, the more similar.
-// This function only compares video frames up to the point when the shortest
-// video ends.
-// For more info about PSNR, see http://en.wikipedia.org/wiki/PSNR
-//
-// Return value:
-// 0 if successful, negative on errors:
-// -1 if the source file cannot be opened
-// -2 if the test file cannot be opened
-// -3 if any of the files are empty
-// -4 if any arguments are invalid.
-int I420PSNRFromFiles(const char* ref_filename,
- const char* test_filename,
- int width,
- int height,
- QualityMetricsResult* result);
-
-// Calculates SSIM values for the reference and test video files (must be in
-// I420 format). All calculated values are filled into the QualityMetricsResult
-// struct.
-// SSIM values range between -1.0 and 1.0, where 1.0 means the files are
-// identical.
-// This function only compares video frames up to the point when the shortest
-// video ends.
-// For more info about SSIM, see http://en.wikipedia.org/wiki/SSIM
-//
-// Return value:
-// 0 if successful, negative on errors:
-// -1 if the source file cannot be opened
-// -2 if the test file cannot be opened
-// -3 if any of the files are empty
-// -4 if any arguments are invalid.
-int I420SSIMFromFiles(const char* ref_filename,
- const char* test_filename,
- int width,
- int height,
- QualityMetricsResult* result);
-
-} // namespace test
-} // namespace webrtc
-
-#endif // WEBRTC_TESTSUPPORT_METRICS_VIDEO_METRICS_H_
diff --git a/trunk/test/testsupport/metrics/video_metrics_unittest.cc b/trunk/test/testsupport/metrics/video_metrics_unittest.cc
deleted file mode 100644
index e77dbff..0000000
--- a/trunk/test/testsupport/metrics/video_metrics_unittest.cc
+++ /dev/null
@@ -1,139 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testsupport/metrics/video_metrics.h"
-
-#include "gtest/gtest.h"
-#include "testsupport/fileutils.h"
-
-namespace webrtc {
-
-static const char* kEmptyFileName = "video_metrics_unittest_empty_file.tmp";
-static const char* kNonExistingFileName = "video_metrics_unittest_non_existing";
-static const int kWidth = 352;
-static const int kHeight = 288;
-
-static const int kMissingReferenceFileReturnCode = -1;
-static const int kMissingTestFileReturnCode = -2;
-static const int kEmptyFileReturnCode = -3;
-static const double kPsnrPerfectResult = 48.0;
-static const double kSsimPerfectResult = 1.0;
-
-class VideoMetricsTest: public testing::Test {
- protected:
- VideoMetricsTest() {
- video_file_ = webrtc::test::ResourcePath("foreman_cif_short", "yuv");
- }
- virtual ~VideoMetricsTest() {}
- void SetUp() {
- // Create an empty file:
- FILE* dummy = fopen(kEmptyFileName, "wb");
- fclose(dummy);
- }
- void TearDown() {
- std::remove(kEmptyFileName);
- }
- webrtc::test::QualityMetricsResult psnr_result_;
- webrtc::test::QualityMetricsResult ssim_result_;
- std::string video_file_;
-};
-
-// Tests that it is possible to run with the same reference as test file
-TEST_F(VideoMetricsTest, ReturnsPerfectResultForIdenticalFilesPSNR) {
- EXPECT_EQ(0, I420PSNRFromFiles(video_file_.c_str(), video_file_.c_str(),
- kWidth, kHeight, &psnr_result_));
- EXPECT_EQ(kPsnrPerfectResult, psnr_result_.average);
-}
-
-TEST_F(VideoMetricsTest, ReturnsPerfectResultForIdenticalFilesSSIM) {
- EXPECT_EQ(0, I420SSIMFromFiles(video_file_.c_str(), video_file_.c_str(),
- kWidth, kHeight, &ssim_result_));
- EXPECT_EQ(kSsimPerfectResult, ssim_result_.average);
-}
-
-TEST_F(VideoMetricsTest, ReturnsPerfectResultForIdenticalFilesBothMetrics) {
- EXPECT_EQ(0, I420MetricsFromFiles(video_file_.c_str(), video_file_.c_str(),
- kWidth, kHeight, &psnr_result_,
- &ssim_result_));
- EXPECT_EQ(kPsnrPerfectResult, psnr_result_.average);
- EXPECT_EQ(kSsimPerfectResult, ssim_result_.average);
-}
-
-// Tests that the right return code is given when the reference file is missing.
-TEST_F(VideoMetricsTest, MissingReferenceFilePSNR) {
- EXPECT_EQ(kMissingReferenceFileReturnCode,
- I420PSNRFromFiles(kNonExistingFileName, video_file_.c_str(),
- kWidth, kHeight, &ssim_result_));
-}
-
-TEST_F(VideoMetricsTest, MissingReferenceFileSSIM) {
- EXPECT_EQ(kMissingReferenceFileReturnCode,
- I420SSIMFromFiles(kNonExistingFileName, video_file_.c_str(),
- kWidth, kHeight, &ssim_result_));
-}
-
-TEST_F(VideoMetricsTest, MissingReferenceFileBothMetrics) {
- EXPECT_EQ(kMissingReferenceFileReturnCode,
- I420MetricsFromFiles(kNonExistingFileName, video_file_.c_str(),
- kWidth, kHeight,
- &psnr_result_, &ssim_result_));
-}
-
-// Tests that the right return code is given when the test file is missing.
-TEST_F(VideoMetricsTest, MissingTestFilePSNR) {
- EXPECT_EQ(kMissingTestFileReturnCode,
- I420PSNRFromFiles(video_file_.c_str(), kNonExistingFileName,
- kWidth, kHeight, &ssim_result_));
-}
-
-TEST_F(VideoMetricsTest, MissingTestFileSSIM) {
- EXPECT_EQ(kMissingTestFileReturnCode,
- I420SSIMFromFiles(video_file_.c_str(), kNonExistingFileName,
- kWidth, kHeight, &ssim_result_));
-}
-
-TEST_F(VideoMetricsTest, MissingTestFileBothMetrics) {
- EXPECT_EQ(kMissingTestFileReturnCode,
- I420MetricsFromFiles(video_file_.c_str(), kNonExistingFileName,
- kWidth, kHeight,
- &psnr_result_, &ssim_result_));
-}
-
-// Tests that the method can be executed with empty files.
-TEST_F(VideoMetricsTest, EmptyFilesPSNR) {
- EXPECT_EQ(kEmptyFileReturnCode,
- I420PSNRFromFiles(kEmptyFileName, video_file_.c_str(),
- kWidth, kHeight, &ssim_result_));
- EXPECT_EQ(kEmptyFileReturnCode,
- I420PSNRFromFiles(video_file_.c_str(), kEmptyFileName,
- kWidth, kHeight, &ssim_result_));
-}
-
-TEST_F(VideoMetricsTest, EmptyFilesSSIM) {
- EXPECT_EQ(kEmptyFileReturnCode,
- I420SSIMFromFiles(kEmptyFileName, video_file_.c_str(),
- kWidth, kHeight, &ssim_result_));
- EXPECT_EQ(kEmptyFileReturnCode,
- I420SSIMFromFiles(video_file_.c_str(), kEmptyFileName,
- kWidth, kHeight, &ssim_result_));
-}
-
-TEST_F(VideoMetricsTest, EmptyFilesBothMetrics) {
- EXPECT_EQ(kEmptyFileReturnCode,
- I420MetricsFromFiles(kEmptyFileName, video_file_.c_str(),
- kWidth, kHeight,
- &psnr_result_, &ssim_result_));
- EXPECT_EQ(kEmptyFileReturnCode,
- I420MetricsFromFiles(video_file_.c_str(), kEmptyFileName,
- kWidth, kHeight,
- &psnr_result_, &ssim_result_));
-}
-
-} // namespace webrtc
diff --git a/trunk/test/testsupport/mock/mock_frame_reader.h b/trunk/test/testsupport/mock/mock_frame_reader.h
deleted file mode 100644
index ecfc13c..0000000
--- a/trunk/test/testsupport/mock/mock_frame_reader.h
+++ /dev/null
@@ -1,33 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_READER_H_
-#define WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_READER_H_
-
-#include "testsupport/frame_reader.h"
-
-#include "gmock/gmock.h"
-
-namespace webrtc {
-namespace test {
-
-class MockFrameReader : public FrameReader {
- public:
- MOCK_METHOD0(Init, bool());
- MOCK_METHOD1(ReadFrame, bool(WebRtc_UWord8* source_buffer));
- MOCK_METHOD0(Close, void());
- MOCK_METHOD0(FrameLength, int());
- MOCK_METHOD0(NumberOfFrames, int());
-};
-
-} // namespace test
-} // namespace webrtc
-
-#endif // WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_READER_H_
diff --git a/trunk/test/testsupport/mock/mock_frame_writer.h b/trunk/test/testsupport/mock/mock_frame_writer.h
deleted file mode 100644
index ba79184..0000000
--- a/trunk/test/testsupport/mock/mock_frame_writer.h
+++ /dev/null
@@ -1,32 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_WRITER_H_
-#define WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_WRITER_H_
-
-#include "testsupport/frame_writer.h"
-
-#include "gmock/gmock.h"
-
-namespace webrtc {
-namespace test {
-
-class MockFrameWriter : public FrameWriter {
- public:
- MOCK_METHOD0(Init, bool());
- MOCK_METHOD1(WriteFrame, bool(WebRtc_UWord8* frame_buffer));
- MOCK_METHOD0(Close, void());
- MOCK_METHOD0(FrameLength, int());
-};
-
-} // namespace test
-} // namespace webrtc
-
-#endif // WEBRTC_TEST_TESTSUPPORT_MOCK_MOCK_FRAME_WRITER_H_
diff --git a/trunk/test/testsupport/packet_reader.cc b/trunk/test/testsupport/packet_reader.cc
deleted file mode 100644
index e8859d1..0000000
--- a/trunk/test/testsupport/packet_reader.cc
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testsupport/packet_reader.h"
-
-#include <cassert>
-#include <cstdio>
-
-namespace webrtc {
-namespace test {
-
-PacketReader::PacketReader()
- : initialized_(false) {}
-
-PacketReader::~PacketReader() {}
-
-void PacketReader::InitializeReading(WebRtc_UWord8* data,
- int data_length_in_bytes,
- int packet_size_in_bytes) {
- assert(data);
- assert(data_length_in_bytes >= 0);
- assert(packet_size_in_bytes > 0);
- data_ = data;
- data_length_ = data_length_in_bytes;
- packet_size_ = packet_size_in_bytes;
- currentIndex_ = 0;
- initialized_ = true;
-}
-
-int PacketReader::NextPacket(WebRtc_UWord8** packet_pointer) {
- if (!initialized_) {
- fprintf(stderr, "Attempting to use uninitialized PacketReader!\n");
- return -1;
- }
- *packet_pointer = data_ + currentIndex_;
- // Check if we're about to read the last packet:
- if (data_length_ - currentIndex_ <= packet_size_) {
- int size = data_length_ - currentIndex_;
- currentIndex_ = data_length_;
- assert(size >= 0);
- return size;
- }
- currentIndex_ += packet_size_;
- assert(packet_size_ >= 0);
- return packet_size_;
-}
-
-} // namespace test
-} // namespace webrtc
diff --git a/trunk/test/testsupport/packet_reader.h b/trunk/test/testsupport/packet_reader.h
deleted file mode 100644
index 4cb0bb1..0000000
--- a/trunk/test/testsupport/packet_reader.h
+++ /dev/null
@@ -1,53 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_TEST_TESTSUPPORT_PACKET_READER_H_
-#define WEBRTC_TEST_TESTSUPPORT_PACKET_READER_H_
-
-#include "typedefs.h"
-
-namespace webrtc {
-namespace test {
-
-// Reads chunks of data to simulate network packets from a byte array.
-class PacketReader {
- public:
- PacketReader();
- virtual ~PacketReader();
-
- // Inizializes a new reading operation. Must be done before invoking the
- // NextPacket method.
- // * data_length_in_bytes is the length of the data byte array. Must be >= 0.
- // 0 length will result in no packets are read.
- // * packet_size_in_bytes is the number of bytes to read in each NextPacket
- // method call. Must be > 0
- virtual void InitializeReading(WebRtc_UWord8* data, int data_length_in_bytes,
- int packet_size_in_bytes);
-
- // Moves the supplied pointer to the beginning of the next packet.
- // Returns:
- // * The size of the packet ready to read (lower than the packet size for
- // the last packet)
- // * 0 if there are no more packets to read
- // * -1 if InitializeReading has not been called (also prints to stderr).
- virtual int NextPacket(WebRtc_UWord8** packet_pointer);
-
- private:
- WebRtc_UWord8* data_;
- int data_length_;
- int packet_size_;
- int currentIndex_;
- bool initialized_;
-};
-
-} // namespace test
-} // namespace webrtc
-
-#endif // WEBRTC_TEST_TESTSUPPORT_PACKET_READER_H_
diff --git a/trunk/test/testsupport/packet_reader_unittest.cc b/trunk/test/testsupport/packet_reader_unittest.cc
deleted file mode 100644
index 6719e4c..0000000
--- a/trunk/test/testsupport/packet_reader_unittest.cc
+++ /dev/null
@@ -1,123 +0,0 @@
-/*
- * Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#include "testsupport/packet_reader.h"
-
-#include "gtest/gtest.h"
-#include "testsupport/unittest_utils.h"
-
-namespace webrtc {
-namespace test {
-
-class PacketReaderTest: public PacketRelatedTest {
- protected:
- PacketReaderTest() {}
- virtual ~PacketReaderTest() {}
- void SetUp() {
- reader_ = new PacketReader();
- }
- void TearDown() {
- delete reader_;
- }
- void VerifyPacketData(int expected_length,
- int actual_length,
- WebRtc_UWord8* original_data_pointer,
- WebRtc_UWord8* new_data_pointer) {
- EXPECT_EQ(expected_length, actual_length);
- EXPECT_EQ(*original_data_pointer, *new_data_pointer);
- EXPECT_EQ(0, memcmp(original_data_pointer, new_data_pointer,
- actual_length));
- }
- PacketReader* reader_;
-};
-
-// Test lack of initialization
-TEST_F(PacketReaderTest, Uninitialized) {
- WebRtc_UWord8* data_pointer = NULL;
- EXPECT_EQ(-1, reader_->NextPacket(&data_pointer));
- EXPECT_EQ(NULL, data_pointer);
-}
-
-TEST_F(PacketReaderTest, InitializeZeroLengthArgument) {
- reader_->InitializeReading(packet_data_, 0, kPacketSizeInBytes);
- ASSERT_EQ(0, reader_->NextPacket(&packet_data_pointer_));
-}
-
-// Test with something smaller than one packet
-TEST_F(PacketReaderTest, NormalSmallData) {
- const int kDataLengthInBytes = 1499;
- WebRtc_UWord8 data[kDataLengthInBytes];
- WebRtc_UWord8* data_pointer = data;
- memset(data, 1, kDataLengthInBytes);
-
- reader_->InitializeReading(data, kDataLengthInBytes, kPacketSizeInBytes);
- int length_to_read = reader_->NextPacket(&data_pointer);
- VerifyPacketData(kDataLengthInBytes, length_to_read, data, data_pointer);
- EXPECT_EQ(0, data_pointer - data); // pointer hasn't moved
-
- // Reading another one shall result in 0 bytes:
- length_to_read = reader_->NextPacket(&data_pointer);
- EXPECT_EQ(0, length_to_read);
- EXPECT_EQ(kDataLengthInBytes, data_pointer - data);
-}
-
-// Test with data length that exactly matches one packet
-TEST_F(PacketReaderTest, NormalOnePacketData) {
- WebRtc_UWord8 data[kPacketSizeInBytes];
- WebRtc_UWord8* data_pointer = data;
- memset(data, 1, kPacketSizeInBytes);
-
- reader_->InitializeReading(data, kPacketSizeInBytes, kPacketSizeInBytes);
- int length_to_read = reader_->NextPacket(&data_pointer);
- VerifyPacketData(kPacketSizeInBytes, length_to_read, data, data_pointer);
- EXPECT_EQ(0, data_pointer - data); // pointer hasn't moved
-
- // Reading another one shall result in 0 bytes:
- length_to_read = reader_->NextPacket(&data_pointer);
- EXPECT_EQ(0, length_to_read);
- EXPECT_EQ(kPacketSizeInBytes, data_pointer - data);
-}
-
-// Test with data length that will result in 3 packets
-TEST_F(PacketReaderTest, NormalLargeData) {
- reader_->InitializeReading(packet_data_, kPacketDataLength,
- kPacketSizeInBytes);
-
- int length_to_read = reader_->NextPacket(&packet_data_pointer_);
- VerifyPacketData(kPacketSizeInBytes, length_to_read,
- packet1_, packet_data_pointer_);
-
- length_to_read = reader_->NextPacket(&packet_data_pointer_);
- VerifyPacketData(kPacketSizeInBytes, length_to_read,
- packet2_, packet_data_pointer_);
-
- length_to_read = reader_->NextPacket(&packet_data_pointer_);
- VerifyPacketData(1u, length_to_read,
- packet3_, packet_data_pointer_);
-
- // Reading another one shall result in 0 bytes:
- length_to_read = reader_->NextPacket(&packet_data_pointer_);
- EXPECT_EQ(0, length_to_read);
- EXPECT_EQ(kPacketDataLength, packet_data_pointer_ - packet_data_);
-}
-
-// Test with empty data.
-TEST_F(PacketReaderTest, EmptyData) {
- const int kDataLengthInBytes = 0;
- WebRtc_UWord8* data = new WebRtc_UWord8[kDataLengthInBytes];
- reader_->InitializeReading(data, kDataLengthInBytes, kPacketSizeInBytes);
- EXPECT_EQ(kDataLengthInBytes, reader_->NextPacket(&data));
- // Do it again to make sure nothing changes
- EXPECT_EQ(kDataLengthInBytes, reader_->NextPacket(&data));
- delete[] data;
-}
-
-} // namespace test
-} // namespace webrtc
diff --git a/trunk/test/testsupport/unittest_utils.h b/trunk/test/testsupport/unittest_utils.h
deleted file mode 100644
index 30464de..0000000
--- a/trunk/test/testsupport/unittest_utils.h
+++ /dev/null
@@ -1,56 +0,0 @@
-/*
- * Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
- *
- * Use of this source code is governed by a BSD-style license
- * that can be found in the LICENSE file in the root of the source
- * tree. An additional intellectual property rights grant can be found
- * in the file PATENTS. All contributing project authors may
- * be found in the AUTHORS file in the root of the source tree.
- */
-
-#ifndef WEBRTC_TEST_TESTSUPPORT_UNITTEST_UTILS_H_
-#define WEBRTC_TEST_TESTSUPPORT_UNITTEST_UTILS_H_
-
-namespace webrtc {
-namespace test {
-
-const int kPacketSizeInBytes = 1500;
-const int kPacketDataLength = kPacketSizeInBytes * 2 + 1;
-const int kPacketDataNumberOfPackets = 3;
-
-// A base test fixture for packet related tests. Contains
-// two full prepared packets with 1s, 2s in their data and a third packet with
-// a single 3 in it (size=1).
-// A packet data structure is also available, that contains these three packets
-// in order.
-class PacketRelatedTest: public testing::Test {
- protected:
- // Tree packet byte arrays with data used for verification:
- WebRtc_UWord8 packet1_[kPacketSizeInBytes];
- WebRtc_UWord8 packet2_[kPacketSizeInBytes];
- WebRtc_UWord8 packet3_[1];
- // Construct a data structure containing these packets
- WebRtc_UWord8 packet_data_[kPacketDataLength];
- WebRtc_UWord8* packet_data_pointer_;
-
- PacketRelatedTest() {
- packet_data_pointer_ = packet_data_;
-
- memset(packet1_, 1, kPacketSizeInBytes);
- memset(packet2_, 2, kPacketSizeInBytes);
- memset(packet3_, 3, 1);
- // Fill the packet_data:
- memcpy(packet_data_pointer_, packet1_, kPacketSizeInBytes);
- memcpy(packet_data_pointer_ + kPacketSizeInBytes, packet2_,
- kPacketSizeInBytes);
- memcpy(packet_data_pointer_ + kPacketSizeInBytes * 2, packet3_, 1);
- }
- virtual ~PacketRelatedTest() {}
- void SetUp() {}
- void TearDown() {}
-};
-
-} // namespace test
-} // namespace webrtc
-
-#endif // WEBRTC_TEST_TESTSUPPORT_UNITTEST_UTILS_H_
diff --git a/trunk/third_party/google-gflags/LICENSE b/trunk/third_party/google-gflags/LICENSE
deleted file mode 100644
index d15b0c2..0000000
--- a/trunk/third_party/google-gflags/LICENSE
+++ /dev/null
@@ -1,28 +0,0 @@
-Copyright (c) 2006, Google Inc.
-All rights reserved.
-
-Redistribution and use in source and binary forms, with or without
-modification, are permitted provided that the following conditions are
-met:
-
- * Redistributions of source code must retain the above copyright
-notice, this list of conditions and the following disclaimer.
- * Redistributions in binary form must reproduce the above
-copyright notice, this list of conditions and the following disclaimer
-in the documentation and/or other materials provided with the
-distribution.
- * Neither the name of Google Inc. nor the names of its
-contributors may be used to endorse or promote products derived from
-this software without specific prior written permission.
-
-THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-"AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
-OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
-LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
-DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
-THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
-(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
-OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
diff --git a/trunk/third_party/google-gflags/README.webrtc b/trunk/third_party/google-gflags/README.webrtc
deleted file mode 100644
index 9a993ce..0000000
--- a/trunk/third_party/google-gflags/README.webrtc
+++ /dev/null
@@ -1,13 +0,0 @@
-URL: http://code.google.com/p/google-gflags/
-Version: 1.5
-License: New BSD
-License File: LICENSE
-
-Description:
-The gflags package contains a library that implements commandline
-flags processing. As such it's a replacement for getopt(). It has
-increased flexibility, including built-in support for C++ types like
-string, and the ability to define flags in the source file in which
-they're used.
-
-Local Modifications: None
diff --git a/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/gflags/gflags.h b/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/gflags/gflags.h
deleted file mode 100644
index 0c2f997..0000000
--- a/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/gflags/gflags.h
+++ /dev/null
@@ -1,585 +0,0 @@
-// Copyright (c) 2006, Google Inc.
-// All rights reserved.
-//
-// Redistribution and use in source and binary forms, with or without
-// modification, are permitted provided that the following conditions are
-// met:
-//
-// * Redistributions of source code must retain the above copyright
-// notice, this list of conditions and the following disclaimer.
-// * Redistributions in binary form must reproduce the above
-// copyright notice, this list of conditions and the following disclaimer
-// in the documentation and/or other materials provided with the
-// distribution.
-// * Neither the name of Google Inc. nor the names of its
-// contributors may be used to endorse or promote products derived from
-// this software without specific prior written permission.
-//
-// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
-// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
-// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
-// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
-// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
-// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
-// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-
-// ---
-// Author: Ray Sidney
-// Revamped and reorganized by Craig Silverstein
-//
-// This is the file that should be included by any file which declares
-// or defines a command line flag or wants to parse command line flags
-// or print a program usage message (which will include information about
-// flags). Executive summary, in the form of an example foo.cc file:
-//
-// #include "foo.h" // foo.h has a line "DECLARE_int32(start);"
-// #include "validators.h" // hypothetical file defining ValidateIsFile()
-//
-// DEFINE_int32(end, 1000, "The last record to read");
-//
-// DEFINE_string(filename, "my_file.txt", "The file to read");
-// // Crash if the specified file does not exist.
-// static bool dummy = RegisterFlagValidator(&FLAGS_filename,
-// &ValidateIsFile);
-//
-// DECLARE_bool(verbose); // some other file has a DEFINE_bool(verbose, ...)
-//
-// void MyFunc() {
-// if (FLAGS_verbose) printf("Records %d-%d\n", FLAGS_start, FLAGS_end);
-// }
-//
-// Then, at the command-line:
-// ./foo --noverbose --start=5 --end=100
-//
-// For more details, see
-// doc/gflags.html
-//
-// --- A note about thread-safety:
-//
-// We describe many functions in this routine as being thread-hostile,
-// thread-compatible, or thread-safe. Here are the meanings we use:
-//
-// thread-safe: it is safe for multiple threads to call this routine
-// (or, when referring to a class, methods of this class)
-// concurrently.
-// thread-hostile: it is not safe for multiple threads to call this
-// routine (or methods of this class) concurrently. In gflags,
-// most thread-hostile routines are intended to be called early in,
-// or even before, main() -- that is, before threads are spawned.
-// thread-compatible: it is safe for multiple threads to read from
-// this variable (when applied to variables), or to call const
-// methods of this class (when applied to classes), as long as no
-// other thread is writing to the variable or calling non-const
-// methods of this class.
-
-#ifndef GOOGLE_GFLAGS_H_
-#define GOOGLE_GFLAGS_H_
-
-#include <string>
-#include <vector>
-
-// We care a lot about number of bits things take up. Unfortunately,
-// systems define their bit-specific ints in a lot of different ways.
-// We use our own way, and have a typedef to get there.
-// Note: these commands below may look like "#if 1" or "#if 0", but
-// that's because they were constructed that way at ./configure time.
-// Look at gflags.h.in to see how they're calculated (based on your config).
-#if 1
-#include <stdint.h> // the normal place uint16_t is defined
-#endif
-#if 1
-#include <sys/types.h> // the normal place u_int16_t is defined
-#endif
-#if 1
-#include <inttypes.h> // a third place for uint16_t or u_int16_t
-#endif
-
-namespace google {
-
-#if 1 // the C99 format
-typedef int32_t int32;
-typedef uint32_t uint32;
-typedef int64_t int64;
-typedef uint64_t uint64;
-#elif 1 // the BSD format
-typedef int32_t int32;
-typedef u_int32_t uint32;
-typedef int64_t int64;
-typedef u_int64_t uint64;
-#elif 0 // the windows (vc7) format
-typedef __int32 int32;
-typedef unsigned __int32 uint32;
-typedef __int64 int64;
-typedef unsigned __int64 uint64;
-#else
-#error Do not know how to define a 32-bit integer quantity on your system
-#endif
-
-// --------------------------------------------------------------------
-// To actually define a flag in a file, use DEFINE_bool,
-// DEFINE_string, etc. at the bottom of this file. You may also find
-// it useful to register a validator with the flag. This ensures that
-// when the flag is parsed from the commandline, or is later set via
-// SetCommandLineOption, we call the validation function. It is _not_
-// called when you assign the value to the flag directly using the = operator.
-//
-// The validation function should return true if the flag value is valid, and
-// false otherwise. If the function returns false for the new setting of the
-// flag, the flag will retain its current value. If it returns false for the
-// default value, ParseCommandLineFlags() will die.
-//
-// This function is safe to call at global construct time (as in the
-// example below).
-//
-// Example use:
-// static bool ValidatePort(const char* flagname, int32 value) {
-// if (value > 0 && value < 32768) // value is ok
-// return true;
-// printf("Invalid value for --%s: %d\n", flagname, (int)value);
-// return false;
-// }
-// DEFINE_int32(port, 0, "What port to listen on");
-// static bool dummy = RegisterFlagValidator(&FLAGS_port, &ValidatePort);
-
-// Returns true if successfully registered, false if not (because the
-// first argument doesn't point to a command-line flag, or because a
-// validator is already registered for this flag).
-bool RegisterFlagValidator(const bool* flag,
- bool (*validate_fn)(const char*, bool));
-bool RegisterFlagValidator(const int32* flag,
- bool (*validate_fn)(const char*, int32));
-bool RegisterFlagValidator(const int64* flag,
- bool (*validate_fn)(const char*, int64));
-bool RegisterFlagValidator(const uint64* flag,
- bool (*validate_fn)(const char*, uint64));
-bool RegisterFlagValidator(const double* flag,
- bool (*validate_fn)(const char*, double));
-bool RegisterFlagValidator(const std::string* flag,
- bool (*validate_fn)(const char*, const std::string&));
-
-
-// --------------------------------------------------------------------
-// These methods are the best way to get access to info about the
-// list of commandline flags. Note that these routines are pretty slow.
-// GetAllFlags: mostly-complete info about the list, sorted by file.
-// ShowUsageWithFlags: pretty-prints the list to stdout (what --help does)
-// ShowUsageWithFlagsRestrict: limit to filenames with restrict as a substr
-//
-// In addition to accessing flags, you can also access argv[0] (the program
-// name) and argv (the entire commandline), which we sock away a copy of.
-// These variables are static, so you should only set them once.
-
-struct CommandLineFlagInfo {
- std::string name; // the name of the flag
- std::string type; // the type of the flag: int32, etc
- std::string description; // the "help text" associated with the flag
- std::string current_value; // the current value, as a string
- std::string default_value; // the default value, as a string
- std::string filename; // 'cleaned' version of filename holding the flag
- bool has_validator_fn; // true if RegisterFlagValidator called on flag
- bool is_default; // true if the flag has the default value and
- // has not been set explicitly from the cmdline
- // or via SetCommandLineOption
-};
-
-// Using this inside of a validator is a recipe for a deadlock.
-// TODO(wojtekm) Fix locking when validators are running, to make it safe to
-// call validators during ParseAllFlags.
-// Also make sure then to uncomment the corresponding unit test in
-// commandlineflags_unittest.sh
-extern void GetAllFlags(std::vector<CommandLineFlagInfo>* OUTPUT);
-// These two are actually defined in commandlineflags_reporting.cc.
-extern void ShowUsageWithFlags(const char *argv0); // what --help does
-extern void ShowUsageWithFlagsRestrict(const char *argv0, const char *restrict);
-
-// Create a descriptive string for a flag.
-// Goes to some trouble to make pretty line breaks.
-extern std::string DescribeOneFlag(const CommandLineFlagInfo& flag);
-
-// Thread-hostile; meant to be called before any threads are spawned.
-extern void SetArgv(int argc, const char** argv);
-// The following functions are thread-safe as long as SetArgv() is
-// only called before any threads start.
-extern const std::vector<std::string>& GetArgvs(); // all of argv as a vector
-extern const char* GetArgv(); // all of argv as a string
-extern const char* GetArgv0(); // only argv0
-extern uint32 GetArgvSum(); // simple checksum of argv
-extern const char* ProgramInvocationName(); // argv0, or "UNKNOWN" if not set
-extern const char* ProgramInvocationShortName(); // basename(argv0)
-// ProgramUsage() is thread-safe as long as SetUsageMessage() is only
-// called before any threads start.
-extern const char* ProgramUsage(); // string set by SetUsageMessage()
-
-
-// --------------------------------------------------------------------
-// Normally you access commandline flags by just saying "if (FLAGS_foo)"
-// or whatever, and set them by calling "FLAGS_foo = bar" (or, more
-// commonly, via the DEFINE_foo macro). But if you need a bit more
-// control, we have programmatic ways to get/set the flags as well.
-// These programmatic ways to access flags are thread-safe, but direct
-// access is only thread-compatible.
-
-// Return true iff the flagname was found.
-// OUTPUT is set to the flag's value, or unchanged if we return false.
-extern bool GetCommandLineOption(const char* name, std::string* OUTPUT);
-
-// Return true iff the flagname was found. OUTPUT is set to the flag's
-// CommandLineFlagInfo or unchanged if we return false.
-extern bool GetCommandLineFlagInfo(const char* name,
- CommandLineFlagInfo* OUTPUT);
-
-// Return the CommandLineFlagInfo of the flagname. exit() if name not found.
-// Example usage, to check if a flag's value is currently the default value:
-// if (GetCommandLineFlagInfoOrDie("foo").is_default) ...
-extern CommandLineFlagInfo GetCommandLineFlagInfoOrDie(const char* name);
-
-enum FlagSettingMode {
- // update the flag's value (can call this multiple times).
- SET_FLAGS_VALUE,
- // update the flag's value, but *only if* it has not yet been updated
- // with SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef".
- SET_FLAG_IF_DEFAULT,
- // set the flag's default value to this. If the flag has not yet updated
- // yet (via SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef")
- // change the flag's current value to the new default value as well.
- SET_FLAGS_DEFAULT
-};
-
-// Set a particular flag ("command line option"). Returns a string
-// describing the new value that the option has been set to. The
-// return value API is not well-specified, so basically just depend on
-// it to be empty if the setting failed for some reason -- the name is
-// not a valid flag name, or the value is not a valid value -- and
-// non-empty else.
-
-// SetCommandLineOption uses set_mode == SET_FLAGS_VALUE (the common case)
-extern std::string SetCommandLineOption(const char* name, const char* value);
-extern std::string SetCommandLineOptionWithMode(const char* name, const char* value,
- FlagSettingMode set_mode);
-
-
-// --------------------------------------------------------------------
-// Saves the states (value, default value, whether the user has set
-// the flag, registered validators, etc) of all flags, and restores
-// them when the FlagSaver is destroyed. This is very useful in
-// tests, say, when you want to let your tests change the flags, but
-// make sure that they get reverted to the original states when your
-// test is complete.
-//
-// Example usage:
-// void TestFoo() {
-// FlagSaver s1;
-// FLAG_foo = false;
-// FLAG_bar = "some value";
-//
-// // test happens here. You can return at any time
-// // without worrying about restoring the FLAG values.
-// }
-//
-// Note: This class is marked with __attribute__((unused)) because all the
-// work is done in the constructor and destructor, so in the standard
-// usage example above, the compiler would complain that it's an
-// unused variable.
-//
-// This class is thread-safe.
-
-class FlagSaver {
- public:
- FlagSaver();
- ~FlagSaver();
-
- private:
- class FlagSaverImpl* impl_; // we use pimpl here to keep API steady
-
- FlagSaver(const FlagSaver&); // no copying!
- void operator=(const FlagSaver&);
-} __attribute__ ((unused));
-
-// --------------------------------------------------------------------
-// Some deprecated or hopefully-soon-to-be-deprecated functions.
-
-// This is often used for logging. TODO(csilvers): figure out a better way
-extern std::string CommandlineFlagsIntoString();
-// Usually where this is used, a FlagSaver should be used instead.
-extern bool ReadFlagsFromString(const std::string& flagfilecontents,
- const char* prog_name,
- bool errors_are_fatal); // uses SET_FLAGS_VALUE
-
-// These let you manually implement --flagfile functionality.
-// DEPRECATED.
-extern bool AppendFlagsIntoFile(const std::string& filename, const char* prog_name);
-extern bool SaveCommandFlags(); // actually defined in google.cc !
-extern bool ReadFromFlagsFile(const std::string& filename, const char* prog_name,
- bool errors_are_fatal); // uses SET_FLAGS_VALUE
-
-
-// --------------------------------------------------------------------
-// Useful routines for initializing flags from the environment.
-// In each case, if 'varname' does not exist in the environment
-// return defval. If 'varname' does exist but is not valid
-// (e.g., not a number for an int32 flag), abort with an error.
-// Otherwise, return the value. NOTE: for booleans, for true use
-// 't' or 'T' or 'true' or '1', for false 'f' or 'F' or 'false' or '0'.
-
-extern bool BoolFromEnv(const char *varname, bool defval);
-extern int32 Int32FromEnv(const char *varname, int32 defval);
-extern int64 Int64FromEnv(const char *varname, int64 defval);
-extern uint64 Uint64FromEnv(const char *varname, uint64 defval);
-extern double DoubleFromEnv(const char *varname, double defval);
-extern const char *StringFromEnv(const char *varname, const char *defval);
-
-
-// --------------------------------------------------------------------
-// The next two functions parse commandlineflags from main():
-
-// Set the "usage" message for this program. For example:
-// string usage("This program does nothing. Sample usage:\n");
-// usage += argv[0] + " <uselessarg1> <uselessarg2>";
-// SetUsageMessage(usage);
-// Do not include commandline flags in the usage: we do that for you!
-// Thread-hostile; meant to be called before any threads are spawned.
-extern void SetUsageMessage(const std::string& usage);
-
-// Looks for flags in argv and parses them. Rearranges argv to put
-// flags first, or removes them entirely if remove_flags is true.
-// If a flag is defined more than once in the command line or flag
-// file, the last definition is used. Returns the index (into argv)
-// of the first non-flag argument.
-// See top-of-file for more details on this function.
-#ifndef SWIG // In swig, use ParseCommandLineFlagsScript() instead.
-extern uint32 ParseCommandLineFlags(int *argc, char*** argv,
- bool remove_flags);
-#endif
-
-
-// Calls to ParseCommandLineNonHelpFlags and then to
-// HandleCommandLineHelpFlags can be used instead of a call to
-// ParseCommandLineFlags during initialization, in order to allow for
-// changing default values for some FLAGS (via
-// e.g. SetCommandLineOptionWithMode calls) between the time of
-// command line parsing and the time of dumping help information for
-// the flags as a result of command line parsing. If a flag is
-// defined more than once in the command line or flag file, the last
-// definition is used. Returns the index (into argv) of the first
-// non-flag argument. (If remove_flags is true, will always return 1.)
-extern uint32 ParseCommandLineNonHelpFlags(int *argc, char*** argv,
- bool remove_flags);
-// This is actually defined in commandlineflags_reporting.cc.
-// This function is misnamed (it also handles --version, etc.), but
-// it's too late to change that now. :-(
-extern void HandleCommandLineHelpFlags(); // in commandlineflags_reporting.cc
-
-// Allow command line reparsing. Disables the error normally
-// generated when an unknown flag is found, since it may be found in a
-// later parse. Thread-hostile; meant to be called before any threads
-// are spawned.
-extern void AllowCommandLineReparsing();
-
-// Reparse the flags that have not yet been recognized. Only flags
-// registered since the last parse will be recognized. Any flag value
-// must be provided as part of the argument using "=", not as a
-// separate command line argument that follows the flag argument.
-// Intended for handling flags from dynamically loaded libraries,
-// since their flags are not registered until they are loaded.
-// Returns the index (into the original argv) of the first non-flag
-// argument. (If remove_flags is true, will always return 1.)
-extern uint32 ReparseCommandLineNonHelpFlags();
-
-// Clean up memory allocated by flags. This is only needed to reduce
-// the quantity of "potentially leaked" reports emitted by memory
-// debugging tools such as valgrind. It is not required for normal
-// operation, or for the perftools heap-checker. It must only be called
-// when the process is about to exit, and all threads that might
-// access flags are quiescent. Referencing flags after this is called
-// will have unexpected consequences. This is not safe to run when
-// multiple threads might be running: the function is thread-hostile.
-extern void ShutDownCommandLineFlags();
-
-
-// --------------------------------------------------------------------
-// Now come the command line flag declaration/definition macros that
-// will actually be used. They're kind of hairy. A major reason
-// for this is initialization: we want people to be able to access
-// variables in global constructors and have that not crash, even if
-// their global constructor runs before the global constructor here.
-// (Obviously, we can't guarantee the flags will have the correct
-// default value in that case, but at least accessing them is safe.)
-// The only way to do that is have flags point to a static buffer.
-// So we make one, using a union to ensure proper alignment, and
-// then use placement-new to actually set up the flag with the
-// correct default value. In the same vein, we have to worry about
-// flag access in global destructors, so FlagRegisterer has to be
-// careful never to destroy the flag-values it constructs.
-//
-// Note that when we define a flag variable FLAGS_<name>, we also
-// preemptively define a junk variable, FLAGS_no<name>. This is to
-// cause a link-time error if someone tries to define 2 flags with
-// names like "logging" and "nologging". We do this because a bool
-// flag FLAG can be set from the command line to true with a "-FLAG"
-// argument, and to false with a "-noFLAG" argument, and so this can
-// potentially avert confusion.
-//
-// We also put flags into their own namespace. It is purposefully
-// named in an opaque way that people should have trouble typing
-// directly. The idea is that DEFINE puts the flag in the weird
-// namespace, and DECLARE imports the flag from there into the current
-// namespace. The net result is to force people to use DECLARE to get
-// access to a flag, rather than saying "extern bool FLAGS_whatever;"
-// or some such instead. We want this so we can put extra
-// functionality (like sanity-checking) in DECLARE if we want, and
-// make sure it is picked up everywhere.
-//
-// We also put the type of the variable in the namespace, so that
-// people can't DECLARE_int32 something that they DEFINE_bool'd
-// elsewhere.
-
-class FlagRegisterer {
- public:
- FlagRegisterer(const char* name, const char* type,
- const char* help, const char* filename,
- void* current_storage, void* defvalue_storage);
-};
-
-extern bool FlagsTypeWarn(const char *name);
-
-// If your application #defines STRIP_FLAG_HELP to a non-zero value
-// before #including this file, we remove the help message from the
-// binary file. This can reduce the size of the resulting binary
-// somewhat, and may also be useful for security reasons.
-
-extern const char kStrippedFlagHelp[];
-
-}
-
-#ifndef SWIG // In swig, ignore the main flag declarations
-
-#if defined(STRIP_FLAG_HELP) && STRIP_FLAG_HELP > 0
-// Need this construct to avoid the 'defined but not used' warning.
-#define MAYBE_STRIPPED_HELP(txt) (false ? (txt) : ::google::kStrippedFlagHelp)
-#else
-#define MAYBE_STRIPPED_HELP(txt) txt
-#endif
-
-// Each command-line flag has two variables associated with it: one
-// with the current value, and one with the default value. However,
-// we have a third variable, which is where value is assigned; it's a
-// constant. This guarantees that FLAG_##value is initialized at
-// static initialization time (e.g. before program-start) rather than
-// than global construction time (which is after program-start but
-// before main), at least when 'value' is a compile-time constant. We
-// use a small trick for the "default value" variable, and call it
-// FLAGS_no<name>. This serves the second purpose of assuring a
-// compile error if someone tries to define a flag named no<name>
-// which is illegal (--foo and --nofoo both affect the "foo" flag).
-#define DEFINE_VARIABLE(type, shorttype, name, value, help) \
- namespace fL##shorttype { \
- static const type FLAGS_nono##name = value; \
- type FLAGS_##name = FLAGS_nono##name; \
- type FLAGS_no##name = FLAGS_nono##name; \
- static ::google::FlagRegisterer o_##name( \
- #name, #type, MAYBE_STRIPPED_HELP(help), __FILE__, \
- &FLAGS_##name, &FLAGS_no##name); \
- } \
- using fL##shorttype::FLAGS_##name
-
-#define DECLARE_VARIABLE(type, shorttype, name) \
- namespace fL##shorttype { \
- extern type FLAGS_##name; \
- } \
- using fL##shorttype::FLAGS_##name
-
-// For DEFINE_bool, we want to do the extra check that the passed-in
-// value is actually a bool, and not a string or something that can be
-// coerced to a bool. These declarations (no definition needed!) will
-// help us do that, and never evaluate From, which is important.
-// We'll use 'sizeof(IsBool(val))' to distinguish. This code requires
-// that the compiler have different sizes for bool & double. Since
-// this is not guaranteed by the standard, we check it with a
-// compile-time assert (msg[-1] will give a compile-time error).
-namespace fLB {
-struct CompileAssert {};
-typedef CompileAssert expected_sizeof_double_neq_sizeof_bool[
- (sizeof(double) != sizeof(bool)) ? 1 : -1];
-template<typename From> double IsBoolFlag(const From& from);
-bool IsBoolFlag(bool from);
-} // namespace fLB
-
-#define DECLARE_bool(name) DECLARE_VARIABLE(bool, B, name)
-#define DEFINE_bool(name, val, txt) \
- namespace fLB { \
- typedef ::fLB::CompileAssert FLAG_##name##_value_is_not_a_bool[ \
- (sizeof(::fLB::IsBoolFlag(val)) != sizeof(double)) ? 1 : -1]; \
- } \
- DEFINE_VARIABLE(bool, B, name, val, txt)
-
-#define DECLARE_int32(name) DECLARE_VARIABLE(::google::int32, I, name)
-#define DEFINE_int32(name,val,txt) DEFINE_VARIABLE(::google::int32, I, name, val, txt)
-
-#define DECLARE_int64(name) DECLARE_VARIABLE(::google::int64, I64, name)
-#define DEFINE_int64(name,val,txt) DEFINE_VARIABLE(::google::int64, I64, name, val, txt)
-
-#define DECLARE_uint64(name) DECLARE_VARIABLE(::google::uint64, U64, name)
-#define DEFINE_uint64(name,val,txt) DEFINE_VARIABLE(::google::uint64, U64, name, val, txt)
-
-#define DECLARE_double(name) DECLARE_VARIABLE(double, D, name)
-#define DEFINE_double(name, val, txt) DEFINE_VARIABLE(double, D, name, val, txt)
-
-// Strings are trickier, because they're not a POD, so we can't
-// construct them at static-initialization time (instead they get
-// constructed at global-constructor time, which is much later). To
-// try to avoid crashes in that case, we use a char buffer to store
-// the string, which we can static-initialize, and then placement-new
-// into it later. It's not perfect, but the best we can do.
-
-namespace fLS {
-// The meaning of "string" might be different between now and when the
-// macros below get invoked (e.g., if someone is experimenting with
-// other string implementations that get defined after this file is
-// included). Save the current meaning now and use it in the macros.
-typedef std::string clstring;
-
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- const char *value) {
- return new(stringspot) clstring(value);
-}
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- const clstring &value) {
- return new(stringspot) clstring(value);
-}
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- int value);
-} // namespace fLS
-
-#define DECLARE_string(name) namespace fLS { extern ::fLS::clstring& FLAGS_##name; } \
- using fLS::FLAGS_##name
-
-// We need to define a var named FLAGS_no##name so people don't define
-// --string and --nostring. And we need a temporary place to put val
-// so we don't have to evaluate it twice. Two great needs that go
-// great together!
-// The weird 'using' + 'extern' inside the fLS namespace is to work around
-// an unknown compiler bug/issue with the gcc 4.2.1 on SUSE 10. See
-// http://code.google.com/p/google-gflags/issues/detail?id=20
-#define DEFINE_string(name, val, txt) \
- namespace fLS { \
- using ::fLS::clstring; \
- static union { void* align; char s[sizeof(clstring)]; } s_##name[2]; \
- clstring* const FLAGS_no##name = ::fLS:: \
- dont_pass0toDEFINE_string(s_##name[0].s, \
- val); \
- static ::google::FlagRegisterer o_##name( \
- #name, "string", MAYBE_STRIPPED_HELP(txt), __FILE__, \
- s_##name[0].s, new (s_##name[1].s) clstring(*FLAGS_no##name)); \
- extern clstring& FLAGS_##name; \
- using fLS::FLAGS_##name; \
- clstring& FLAGS_##name = *FLAGS_no##name; \
- } \
- using fLS::FLAGS_##name
-
-#endif // SWIG
-
-#endif // GOOGLE_GFLAGS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/gflags/gflags_completions.h b/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/gflags/gflags_completions.h
deleted file mode 100644
index 9d9ce7a..0000000
--- a/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/gflags/gflags_completions.h
+++ /dev/null
@@ -1,121 +0,0 @@
-// Copyright (c) 2008, Google Inc.
-// All rights reserved.
-//
-// Redistribution and use in source and binary forms, with or without
-// modification, are permitted provided that the following conditions are
-// met:
-//
-// * Redistributions of source code must retain the above copyright
-// notice, this list of conditions and the following disclaimer.
-// * Redistributions in binary form must reproduce the above
-// copyright notice, this list of conditions and the following disclaimer
-// in the documentation and/or other materials provided with the
-// distribution.
-// * Neither the name of Google Inc. nor the names of its
-// contributors may be used to endorse or promote products derived from
-// this software without specific prior written permission.
-//
-// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
-// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
-// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
-// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
-// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
-// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
-// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-//
-// ---
-// Author: Dave Nicponski
-//
-// Implement helpful bash-style command line flag completions
-//
-// ** Functional API:
-// HandleCommandLineCompletions() should be called early during
-// program startup, but after command line flag code has been
-// initialized, such as the beginning of HandleCommandLineHelpFlags().
-// It checks the value of the flag --tab_completion_word. If this
-// flag is empty, nothing happens here. If it contains a string,
-// however, then HandleCommandLineCompletions() will hijack the
-// process, attempting to identify the intention behind this
-// completion. Regardless of the outcome of this deduction, the
-// process will be terminated, similar to --helpshort flag
-// handling.
-//
-// ** Overview of Bash completions:
-// Bash can be told to programatically determine completions for the
-// current 'cursor word'. It does this by (in this case) invoking a
-// command with some additional arguments identifying the command
-// being executed, the word being completed, and the previous word
-// (if any). Bash then expects a sequence of output lines to be
-// printed to stdout. If these lines all contain a common prefix
-// longer than the cursor word, bash will replace the cursor word
-// with that common prefix, and display nothing. If there isn't such
-// a common prefix, bash will display the lines in pages using 'more'.
-//
-// ** Strategy taken for command line completions:
-// If we can deduce either the exact flag intended, or a common flag
-// prefix, we'll output exactly that. Otherwise, if information
-// must be displayed to the user, we'll take the opportunity to add
-// some helpful information beyond just the flag name (specifically,
-// we'll include the default flag value and as much of the flag's
-// description as can fit on a single terminal line width, as specified
-// by the flag --tab_completion_columns). Furthermore, we'll try to
-// make bash order the output such that the most useful or relevent
-// flags are the most likely to be shown at the top.
-//
-// ** Additional features:
-// To assist in finding that one really useful flag, substring matching
-// was implemented. Before pressing a <TAB> to get completion for the
-// current word, you can append one or more '?' to the flag to do
-// substring matching. Here's the semantics:
-// --foo<TAB> Show me all flags with names prefixed by 'foo'
-// --foo?<TAB> Show me all flags with 'foo' somewhere in the name
-// --foo??<TAB> Same as prior case, but also search in module
-// definition path for 'foo'
-// --foo???<TAB> Same as prior case, but also search in flag
-// descriptions for 'foo'
-// Finally, we'll trim the output to a relatively small number of
-// flags to keep bash quiet about the verbosity of output. If one
-// really wanted to see all possible matches, appending a '+' to the
-// search word will force the exhaustive list of matches to be printed.
-//
-// ** How to have bash accept completions from a binary:
-// Bash requires that it be informed about each command that programmatic
-// completion should be enabled for. Example addition to a .bashrc
-// file would be (your path to gflags_completions.sh file may differ):
-
-/*
-$ complete -o bashdefault -o default -o nospace -C \
- '/usr/local/bin/gflags_completions.sh --tab_completion_columns $COLUMNS' \
- time env binary_name another_binary [...]
-*/
-
-// This would allow the following to work:
-// $ /path/to/binary_name --vmodule<TAB>
-// Or:
-// $ ./bin/path/another_binary --gfs_u<TAB>
-// (etc)
-//
-// Sadly, it appears that bash gives no easy way to force this behavior for
-// all commands. That's where the "time" in the above example comes in.
-// If you haven't specifically added a command to the list of completion
-// supported commands, you can still get completions by prefixing the
-// entire command with "env".
-// $ env /some/brand/new/binary --vmod<TAB>
-// Assuming that "binary" is a newly compiled binary, this should still
-// produce the expected completion output.
-
-
-#ifndef GOOGLE_GFLAGS_COMPLETIONS_H_
-#define GOOGLE_GFLAGS_COMPLETIONS_H_
-
-namespace google {
-
-void HandleCommandLineCompletions(void);
-
-}
-
-#endif // GOOGLE_GFLAGS_COMPLETIONS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/private/config.h b/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/private/config.h
deleted file mode 100644
index 98d8e1a..0000000
--- a/trunk/third_party/google-gflags/gen/arch/linux/ia32/include/private/config.h
+++ /dev/null
@@ -1,110 +0,0 @@
-/* src/config.h. Generated from config.h.in by configure. */
-/* src/config.h.in. Generated from configure.ac by autoheader. */
-
-/* Always the empty-string on non-windows systems. On windows, should be
- "__declspec(dllexport)". This way, when we compile the dll, we export our
- functions/classes. It's safe to define this here because config.h is only
- used internally, to compile the DLL, and every DLL source file #includes
- "config.h" before anything else. */
-#define GFLAGS_DLL_DECL /**/
-
-/* Namespace for Google classes */
-#define GOOGLE_NAMESPACE ::google
-
-/* Define to 1 if you have the <dlfcn.h> header file. */
-#define HAVE_DLFCN_H 1
-
-/* Define to 1 if you have the <fnmatch.h> header file. */
-#define HAVE_FNMATCH_H 1
-
-/* Define to 1 if you have the <inttypes.h> header file. */
-#define HAVE_INTTYPES_H 1
-
-/* Define to 1 if you have the <memory.h> header file. */
-#define HAVE_MEMORY_H 1
-
-/* define if the compiler implements namespaces */
-#define HAVE_NAMESPACES 1
-
-/* Define if you have POSIX threads libraries and header files. */
-#define HAVE_PTHREAD 1
-
-/* Define to 1 if you have the `putenv' function. */
-#define HAVE_PUTENV 1
-
-/* Define to 1 if you have the `setenv' function. */
-#define HAVE_SETENV 1
-
-/* Define to 1 if you have the <stdint.h> header file. */
-#define HAVE_STDINT_H 1
-
-/* Define to 1 if you have the <stdlib.h> header file. */
-#define HAVE_STDLIB_H 1
-
-/* Define to 1 if you have the <strings.h> header file. */
-#define HAVE_STRINGS_H 1
-
-/* Define to 1 if you have the <string.h> header file. */
-#define HAVE_STRING_H 1
-
-/* Define to 1 if you have the `strtoll' function. */
-#define HAVE_STRTOLL 1
-
-/* Define to 1 if you have the `strtoq' function. */
-#define HAVE_STRTOQ 1
-
-/* Define to 1 if you have the <sys/stat.h> header file. */
-#define HAVE_SYS_STAT_H 1
-
-/* Define to 1 if you have the <sys/types.h> header file. */
-#define HAVE_SYS_TYPES_H 1
-
-/* Define to 1 if you have the <unistd.h> header file. */
-#define HAVE_UNISTD_H 1
-
-/* define if your compiler has __attribute__ */
-#define HAVE___ATTRIBUTE__ 1
-
-/* Define to the sub-directory in which libtool stores uninstalled libraries.
- */
-#define LT_OBJDIR ".libs/"
-
-/* Name of package */
-#define PACKAGE "gflags"
-
-/* Define to the address where bug reports for this package should be sent. */
-#define PACKAGE_BUGREPORT "opensource@google.com"
-
-/* Define to the full name of this package. */
-#define PACKAGE_NAME "gflags"
-
-/* Define to the full name and version of this package. */
-#define PACKAGE_STRING "gflags 1.5"
-
-/* Define to the one symbol short name of this package. */
-#define PACKAGE_TARNAME "gflags"
-
-/* Define to the home page for this package. */
-#define PACKAGE_URL ""
-
-/* Define to the version of this package. */
-#define PACKAGE_VERSION "1.5"
-
-/* Define to necessary symbol if this constant uses a non-standard name on
- your system. */
-/* #undef PTHREAD_CREATE_JOINABLE */
-
-/* Define to 1 if you have the ANSI C header files. */
-#define STDC_HEADERS 1
-
-/* the namespace where STL code like vector<> is defined */
-#define STL_NAMESPACE std
-
-/* Version number of package */
-#define VERSION "1.5"
-
-/* Stops putting the code inside the Google namespace */
-#define _END_GOOGLE_NAMESPACE_ }
-
-/* Puts following code inside the Google namespace */
-#define _START_GOOGLE_NAMESPACE_ namespace google {
diff --git a/trunk/third_party/google-gflags/gen/arch/linux/x64/include/gflags/gflags.h b/trunk/third_party/google-gflags/gen/arch/linux/x64/include/gflags/gflags.h
deleted file mode 100644
index 0c2f997..0000000
--- a/trunk/third_party/google-gflags/gen/arch/linux/x64/include/gflags/gflags.h
+++ /dev/null
@@ -1,585 +0,0 @@
-// Copyright (c) 2006, Google Inc.
-// All rights reserved.
-//
-// Redistribution and use in source and binary forms, with or without
-// modification, are permitted provided that the following conditions are
-// met:
-//
-// * Redistributions of source code must retain the above copyright
-// notice, this list of conditions and the following disclaimer.
-// * Redistributions in binary form must reproduce the above
-// copyright notice, this list of conditions and the following disclaimer
-// in the documentation and/or other materials provided with the
-// distribution.
-// * Neither the name of Google Inc. nor the names of its
-// contributors may be used to endorse or promote products derived from
-// this software without specific prior written permission.
-//
-// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
-// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
-// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
-// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
-// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
-// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
-// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-
-// ---
-// Author: Ray Sidney
-// Revamped and reorganized by Craig Silverstein
-//
-// This is the file that should be included by any file which declares
-// or defines a command line flag or wants to parse command line flags
-// or print a program usage message (which will include information about
-// flags). Executive summary, in the form of an example foo.cc file:
-//
-// #include "foo.h" // foo.h has a line "DECLARE_int32(start);"
-// #include "validators.h" // hypothetical file defining ValidateIsFile()
-//
-// DEFINE_int32(end, 1000, "The last record to read");
-//
-// DEFINE_string(filename, "my_file.txt", "The file to read");
-// // Crash if the specified file does not exist.
-// static bool dummy = RegisterFlagValidator(&FLAGS_filename,
-// &ValidateIsFile);
-//
-// DECLARE_bool(verbose); // some other file has a DEFINE_bool(verbose, ...)
-//
-// void MyFunc() {
-// if (FLAGS_verbose) printf("Records %d-%d\n", FLAGS_start, FLAGS_end);
-// }
-//
-// Then, at the command-line:
-// ./foo --noverbose --start=5 --end=100
-//
-// For more details, see
-// doc/gflags.html
-//
-// --- A note about thread-safety:
-//
-// We describe many functions in this routine as being thread-hostile,
-// thread-compatible, or thread-safe. Here are the meanings we use:
-//
-// thread-safe: it is safe for multiple threads to call this routine
-// (or, when referring to a class, methods of this class)
-// concurrently.
-// thread-hostile: it is not safe for multiple threads to call this
-// routine (or methods of this class) concurrently. In gflags,
-// most thread-hostile routines are intended to be called early in,
-// or even before, main() -- that is, before threads are spawned.
-// thread-compatible: it is safe for multiple threads to read from
-// this variable (when applied to variables), or to call const
-// methods of this class (when applied to classes), as long as no
-// other thread is writing to the variable or calling non-const
-// methods of this class.
-
-#ifndef GOOGLE_GFLAGS_H_
-#define GOOGLE_GFLAGS_H_
-
-#include <string>
-#include <vector>
-
-// We care a lot about number of bits things take up. Unfortunately,
-// systems define their bit-specific ints in a lot of different ways.
-// We use our own way, and have a typedef to get there.
-// Note: these commands below may look like "#if 1" or "#if 0", but
-// that's because they were constructed that way at ./configure time.
-// Look at gflags.h.in to see how they're calculated (based on your config).
-#if 1
-#include <stdint.h> // the normal place uint16_t is defined
-#endif
-#if 1
-#include <sys/types.h> // the normal place u_int16_t is defined
-#endif
-#if 1
-#include <inttypes.h> // a third place for uint16_t or u_int16_t
-#endif
-
-namespace google {
-
-#if 1 // the C99 format
-typedef int32_t int32;
-typedef uint32_t uint32;
-typedef int64_t int64;
-typedef uint64_t uint64;
-#elif 1 // the BSD format
-typedef int32_t int32;
-typedef u_int32_t uint32;
-typedef int64_t int64;
-typedef u_int64_t uint64;
-#elif 0 // the windows (vc7) format
-typedef __int32 int32;
-typedef unsigned __int32 uint32;
-typedef __int64 int64;
-typedef unsigned __int64 uint64;
-#else
-#error Do not know how to define a 32-bit integer quantity on your system
-#endif
-
-// --------------------------------------------------------------------
-// To actually define a flag in a file, use DEFINE_bool,
-// DEFINE_string, etc. at the bottom of this file. You may also find
-// it useful to register a validator with the flag. This ensures that
-// when the flag is parsed from the commandline, or is later set via
-// SetCommandLineOption, we call the validation function. It is _not_
-// called when you assign the value to the flag directly using the = operator.
-//
-// The validation function should return true if the flag value is valid, and
-// false otherwise. If the function returns false for the new setting of the
-// flag, the flag will retain its current value. If it returns false for the
-// default value, ParseCommandLineFlags() will die.
-//
-// This function is safe to call at global construct time (as in the
-// example below).
-//
-// Example use:
-// static bool ValidatePort(const char* flagname, int32 value) {
-// if (value > 0 && value < 32768) // value is ok
-// return true;
-// printf("Invalid value for --%s: %d\n", flagname, (int)value);
-// return false;
-// }
-// DEFINE_int32(port, 0, "What port to listen on");
-// static bool dummy = RegisterFlagValidator(&FLAGS_port, &ValidatePort);
-
-// Returns true if successfully registered, false if not (because the
-// first argument doesn't point to a command-line flag, or because a
-// validator is already registered for this flag).
-bool RegisterFlagValidator(const bool* flag,
- bool (*validate_fn)(const char*, bool));
-bool RegisterFlagValidator(const int32* flag,
- bool (*validate_fn)(const char*, int32));
-bool RegisterFlagValidator(const int64* flag,
- bool (*validate_fn)(const char*, int64));
-bool RegisterFlagValidator(const uint64* flag,
- bool (*validate_fn)(const char*, uint64));
-bool RegisterFlagValidator(const double* flag,
- bool (*validate_fn)(const char*, double));
-bool RegisterFlagValidator(const std::string* flag,
- bool (*validate_fn)(const char*, const std::string&));
-
-
-// --------------------------------------------------------------------
-// These methods are the best way to get access to info about the
-// list of commandline flags. Note that these routines are pretty slow.
-// GetAllFlags: mostly-complete info about the list, sorted by file.
-// ShowUsageWithFlags: pretty-prints the list to stdout (what --help does)
-// ShowUsageWithFlagsRestrict: limit to filenames with restrict as a substr
-//
-// In addition to accessing flags, you can also access argv[0] (the program
-// name) and argv (the entire commandline), which we sock away a copy of.
-// These variables are static, so you should only set them once.
-
-struct CommandLineFlagInfo {
- std::string name; // the name of the flag
- std::string type; // the type of the flag: int32, etc
- std::string description; // the "help text" associated with the flag
- std::string current_value; // the current value, as a string
- std::string default_value; // the default value, as a string
- std::string filename; // 'cleaned' version of filename holding the flag
- bool has_validator_fn; // true if RegisterFlagValidator called on flag
- bool is_default; // true if the flag has the default value and
- // has not been set explicitly from the cmdline
- // or via SetCommandLineOption
-};
-
-// Using this inside of a validator is a recipe for a deadlock.
-// TODO(wojtekm) Fix locking when validators are running, to make it safe to
-// call validators during ParseAllFlags.
-// Also make sure then to uncomment the corresponding unit test in
-// commandlineflags_unittest.sh
-extern void GetAllFlags(std::vector<CommandLineFlagInfo>* OUTPUT);
-// These two are actually defined in commandlineflags_reporting.cc.
-extern void ShowUsageWithFlags(const char *argv0); // what --help does
-extern void ShowUsageWithFlagsRestrict(const char *argv0, const char *restrict);
-
-// Create a descriptive string for a flag.
-// Goes to some trouble to make pretty line breaks.
-extern std::string DescribeOneFlag(const CommandLineFlagInfo& flag);
-
-// Thread-hostile; meant to be called before any threads are spawned.
-extern void SetArgv(int argc, const char** argv);
-// The following functions are thread-safe as long as SetArgv() is
-// only called before any threads start.
-extern const std::vector<std::string>& GetArgvs(); // all of argv as a vector
-extern const char* GetArgv(); // all of argv as a string
-extern const char* GetArgv0(); // only argv0
-extern uint32 GetArgvSum(); // simple checksum of argv
-extern const char* ProgramInvocationName(); // argv0, or "UNKNOWN" if not set
-extern const char* ProgramInvocationShortName(); // basename(argv0)
-// ProgramUsage() is thread-safe as long as SetUsageMessage() is only
-// called before any threads start.
-extern const char* ProgramUsage(); // string set by SetUsageMessage()
-
-
-// --------------------------------------------------------------------
-// Normally you access commandline flags by just saying "if (FLAGS_foo)"
-// or whatever, and set them by calling "FLAGS_foo = bar" (or, more
-// commonly, via the DEFINE_foo macro). But if you need a bit more
-// control, we have programmatic ways to get/set the flags as well.
-// These programmatic ways to access flags are thread-safe, but direct
-// access is only thread-compatible.
-
-// Return true iff the flagname was found.
-// OUTPUT is set to the flag's value, or unchanged if we return false.
-extern bool GetCommandLineOption(const char* name, std::string* OUTPUT);
-
-// Return true iff the flagname was found. OUTPUT is set to the flag's
-// CommandLineFlagInfo or unchanged if we return false.
-extern bool GetCommandLineFlagInfo(const char* name,
- CommandLineFlagInfo* OUTPUT);
-
-// Return the CommandLineFlagInfo of the flagname. exit() if name not found.
-// Example usage, to check if a flag's value is currently the default value:
-// if (GetCommandLineFlagInfoOrDie("foo").is_default) ...
-extern CommandLineFlagInfo GetCommandLineFlagInfoOrDie(const char* name);
-
-enum FlagSettingMode {
- // update the flag's value (can call this multiple times).
- SET_FLAGS_VALUE,
- // update the flag's value, but *only if* it has not yet been updated
- // with SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef".
- SET_FLAG_IF_DEFAULT,
- // set the flag's default value to this. If the flag has not yet updated
- // yet (via SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef")
- // change the flag's current value to the new default value as well.
- SET_FLAGS_DEFAULT
-};
-
-// Set a particular flag ("command line option"). Returns a string
-// describing the new value that the option has been set to. The
-// return value API is not well-specified, so basically just depend on
-// it to be empty if the setting failed for some reason -- the name is
-// not a valid flag name, or the value is not a valid value -- and
-// non-empty else.
-
-// SetCommandLineOption uses set_mode == SET_FLAGS_VALUE (the common case)
-extern std::string SetCommandLineOption(const char* name, const char* value);
-extern std::string SetCommandLineOptionWithMode(const char* name, const char* value,
- FlagSettingMode set_mode);
-
-
-// --------------------------------------------------------------------
-// Saves the states (value, default value, whether the user has set
-// the flag, registered validators, etc) of all flags, and restores
-// them when the FlagSaver is destroyed. This is very useful in
-// tests, say, when you want to let your tests change the flags, but
-// make sure that they get reverted to the original states when your
-// test is complete.
-//
-// Example usage:
-// void TestFoo() {
-// FlagSaver s1;
-// FLAG_foo = false;
-// FLAG_bar = "some value";
-//
-// // test happens here. You can return at any time
-// // without worrying about restoring the FLAG values.
-// }
-//
-// Note: This class is marked with __attribute__((unused)) because all the
-// work is done in the constructor and destructor, so in the standard
-// usage example above, the compiler would complain that it's an
-// unused variable.
-//
-// This class is thread-safe.
-
-class FlagSaver {
- public:
- FlagSaver();
- ~FlagSaver();
-
- private:
- class FlagSaverImpl* impl_; // we use pimpl here to keep API steady
-
- FlagSaver(const FlagSaver&); // no copying!
- void operator=(const FlagSaver&);
-} __attribute__ ((unused));
-
-// --------------------------------------------------------------------
-// Some deprecated or hopefully-soon-to-be-deprecated functions.
-
-// This is often used for logging. TODO(csilvers): figure out a better way
-extern std::string CommandlineFlagsIntoString();
-// Usually where this is used, a FlagSaver should be used instead.
-extern bool ReadFlagsFromString(const std::string& flagfilecontents,
- const char* prog_name,
- bool errors_are_fatal); // uses SET_FLAGS_VALUE
-
-// These let you manually implement --flagfile functionality.
-// DEPRECATED.
-extern bool AppendFlagsIntoFile(const std::string& filename, const char* prog_name);
-extern bool SaveCommandFlags(); // actually defined in google.cc !
-extern bool ReadFromFlagsFile(const std::string& filename, const char* prog_name,
- bool errors_are_fatal); // uses SET_FLAGS_VALUE
-
-
-// --------------------------------------------------------------------
-// Useful routines for initializing flags from the environment.
-// In each case, if 'varname' does not exist in the environment
-// return defval. If 'varname' does exist but is not valid
-// (e.g., not a number for an int32 flag), abort with an error.
-// Otherwise, return the value. NOTE: for booleans, for true use
-// 't' or 'T' or 'true' or '1', for false 'f' or 'F' or 'false' or '0'.
-
-extern bool BoolFromEnv(const char *varname, bool defval);
-extern int32 Int32FromEnv(const char *varname, int32 defval);
-extern int64 Int64FromEnv(const char *varname, int64 defval);
-extern uint64 Uint64FromEnv(const char *varname, uint64 defval);
-extern double DoubleFromEnv(const char *varname, double defval);
-extern const char *StringFromEnv(const char *varname, const char *defval);
-
-
-// --------------------------------------------------------------------
-// The next two functions parse commandlineflags from main():
-
-// Set the "usage" message for this program. For example:
-// string usage("This program does nothing. Sample usage:\n");
-// usage += argv[0] + " <uselessarg1> <uselessarg2>";
-// SetUsageMessage(usage);
-// Do not include commandline flags in the usage: we do that for you!
-// Thread-hostile; meant to be called before any threads are spawned.
-extern void SetUsageMessage(const std::string& usage);
-
-// Looks for flags in argv and parses them. Rearranges argv to put
-// flags first, or removes them entirely if remove_flags is true.
-// If a flag is defined more than once in the command line or flag
-// file, the last definition is used. Returns the index (into argv)
-// of the first non-flag argument.
-// See top-of-file for more details on this function.
-#ifndef SWIG // In swig, use ParseCommandLineFlagsScript() instead.
-extern uint32 ParseCommandLineFlags(int *argc, char*** argv,
- bool remove_flags);
-#endif
-
-
-// Calls to ParseCommandLineNonHelpFlags and then to
-// HandleCommandLineHelpFlags can be used instead of a call to
-// ParseCommandLineFlags during initialization, in order to allow for
-// changing default values for some FLAGS (via
-// e.g. SetCommandLineOptionWithMode calls) between the time of
-// command line parsing and the time of dumping help information for
-// the flags as a result of command line parsing. If a flag is
-// defined more than once in the command line or flag file, the last
-// definition is used. Returns the index (into argv) of the first
-// non-flag argument. (If remove_flags is true, will always return 1.)
-extern uint32 ParseCommandLineNonHelpFlags(int *argc, char*** argv,
- bool remove_flags);
-// This is actually defined in commandlineflags_reporting.cc.
-// This function is misnamed (it also handles --version, etc.), but
-// it's too late to change that now. :-(
-extern void HandleCommandLineHelpFlags(); // in commandlineflags_reporting.cc
-
-// Allow command line reparsing. Disables the error normally
-// generated when an unknown flag is found, since it may be found in a
-// later parse. Thread-hostile; meant to be called before any threads
-// are spawned.
-extern void AllowCommandLineReparsing();
-
-// Reparse the flags that have not yet been recognized. Only flags
-// registered since the last parse will be recognized. Any flag value
-// must be provided as part of the argument using "=", not as a
-// separate command line argument that follows the flag argument.
-// Intended for handling flags from dynamically loaded libraries,
-// since their flags are not registered until they are loaded.
-// Returns the index (into the original argv) of the first non-flag
-// argument. (If remove_flags is true, will always return 1.)
-extern uint32 ReparseCommandLineNonHelpFlags();
-
-// Clean up memory allocated by flags. This is only needed to reduce
-// the quantity of "potentially leaked" reports emitted by memory
-// debugging tools such as valgrind. It is not required for normal
-// operation, or for the perftools heap-checker. It must only be called
-// when the process is about to exit, and all threads that might
-// access flags are quiescent. Referencing flags after this is called
-// will have unexpected consequences. This is not safe to run when
-// multiple threads might be running: the function is thread-hostile.
-extern void ShutDownCommandLineFlags();
-
-
-// --------------------------------------------------------------------
-// Now come the command line flag declaration/definition macros that
-// will actually be used. They're kind of hairy. A major reason
-// for this is initialization: we want people to be able to access
-// variables in global constructors and have that not crash, even if
-// their global constructor runs before the global constructor here.
-// (Obviously, we can't guarantee the flags will have the correct
-// default value in that case, but at least accessing them is safe.)
-// The only way to do that is have flags point to a static buffer.
-// So we make one, using a union to ensure proper alignment, and
-// then use placement-new to actually set up the flag with the
-// correct default value. In the same vein, we have to worry about
-// flag access in global destructors, so FlagRegisterer has to be
-// careful never to destroy the flag-values it constructs.
-//
-// Note that when we define a flag variable FLAGS_<name>, we also
-// preemptively define a junk variable, FLAGS_no<name>. This is to
-// cause a link-time error if someone tries to define 2 flags with
-// names like "logging" and "nologging". We do this because a bool
-// flag FLAG can be set from the command line to true with a "-FLAG"
-// argument, and to false with a "-noFLAG" argument, and so this can
-// potentially avert confusion.
-//
-// We also put flags into their own namespace. It is purposefully
-// named in an opaque way that people should have trouble typing
-// directly. The idea is that DEFINE puts the flag in the weird
-// namespace, and DECLARE imports the flag from there into the current
-// namespace. The net result is to force people to use DECLARE to get
-// access to a flag, rather than saying "extern bool FLAGS_whatever;"
-// or some such instead. We want this so we can put extra
-// functionality (like sanity-checking) in DECLARE if we want, and
-// make sure it is picked up everywhere.
-//
-// We also put the type of the variable in the namespace, so that
-// people can't DECLARE_int32 something that they DEFINE_bool'd
-// elsewhere.
-
-class FlagRegisterer {
- public:
- FlagRegisterer(const char* name, const char* type,
- const char* help, const char* filename,
- void* current_storage, void* defvalue_storage);
-};
-
-extern bool FlagsTypeWarn(const char *name);
-
-// If your application #defines STRIP_FLAG_HELP to a non-zero value
-// before #including this file, we remove the help message from the
-// binary file. This can reduce the size of the resulting binary
-// somewhat, and may also be useful for security reasons.
-
-extern const char kStrippedFlagHelp[];
-
-}
-
-#ifndef SWIG // In swig, ignore the main flag declarations
-
-#if defined(STRIP_FLAG_HELP) && STRIP_FLAG_HELP > 0
-// Need this construct to avoid the 'defined but not used' warning.
-#define MAYBE_STRIPPED_HELP(txt) (false ? (txt) : ::google::kStrippedFlagHelp)
-#else
-#define MAYBE_STRIPPED_HELP(txt) txt
-#endif
-
-// Each command-line flag has two variables associated with it: one
-// with the current value, and one with the default value. However,
-// we have a third variable, which is where value is assigned; it's a
-// constant. This guarantees that FLAG_##value is initialized at
-// static initialization time (e.g. before program-start) rather than
-// than global construction time (which is after program-start but
-// before main), at least when 'value' is a compile-time constant. We
-// use a small trick for the "default value" variable, and call it
-// FLAGS_no<name>. This serves the second purpose of assuring a
-// compile error if someone tries to define a flag named no<name>
-// which is illegal (--foo and --nofoo both affect the "foo" flag).
-#define DEFINE_VARIABLE(type, shorttype, name, value, help) \
- namespace fL##shorttype { \
- static const type FLAGS_nono##name = value; \
- type FLAGS_##name = FLAGS_nono##name; \
- type FLAGS_no##name = FLAGS_nono##name; \
- static ::google::FlagRegisterer o_##name( \
- #name, #type, MAYBE_STRIPPED_HELP(help), __FILE__, \
- &FLAGS_##name, &FLAGS_no##name); \
- } \
- using fL##shorttype::FLAGS_##name
-
-#define DECLARE_VARIABLE(type, shorttype, name) \
- namespace fL##shorttype { \
- extern type FLAGS_##name; \
- } \
- using fL##shorttype::FLAGS_##name
-
-// For DEFINE_bool, we want to do the extra check that the passed-in
-// value is actually a bool, and not a string or something that can be
-// coerced to a bool. These declarations (no definition needed!) will
-// help us do that, and never evaluate From, which is important.
-// We'll use 'sizeof(IsBool(val))' to distinguish. This code requires
-// that the compiler have different sizes for bool & double. Since
-// this is not guaranteed by the standard, we check it with a
-// compile-time assert (msg[-1] will give a compile-time error).
-namespace fLB {
-struct CompileAssert {};
-typedef CompileAssert expected_sizeof_double_neq_sizeof_bool[
- (sizeof(double) != sizeof(bool)) ? 1 : -1];
-template<typename From> double IsBoolFlag(const From& from);
-bool IsBoolFlag(bool from);
-} // namespace fLB
-
-#define DECLARE_bool(name) DECLARE_VARIABLE(bool, B, name)
-#define DEFINE_bool(name, val, txt) \
- namespace fLB { \
- typedef ::fLB::CompileAssert FLAG_##name##_value_is_not_a_bool[ \
- (sizeof(::fLB::IsBoolFlag(val)) != sizeof(double)) ? 1 : -1]; \
- } \
- DEFINE_VARIABLE(bool, B, name, val, txt)
-
-#define DECLARE_int32(name) DECLARE_VARIABLE(::google::int32, I, name)
-#define DEFINE_int32(name,val,txt) DEFINE_VARIABLE(::google::int32, I, name, val, txt)
-
-#define DECLARE_int64(name) DECLARE_VARIABLE(::google::int64, I64, name)
-#define DEFINE_int64(name,val,txt) DEFINE_VARIABLE(::google::int64, I64, name, val, txt)
-
-#define DECLARE_uint64(name) DECLARE_VARIABLE(::google::uint64, U64, name)
-#define DEFINE_uint64(name,val,txt) DEFINE_VARIABLE(::google::uint64, U64, name, val, txt)
-
-#define DECLARE_double(name) DECLARE_VARIABLE(double, D, name)
-#define DEFINE_double(name, val, txt) DEFINE_VARIABLE(double, D, name, val, txt)
-
-// Strings are trickier, because they're not a POD, so we can't
-// construct them at static-initialization time (instead they get
-// constructed at global-constructor time, which is much later). To
-// try to avoid crashes in that case, we use a char buffer to store
-// the string, which we can static-initialize, and then placement-new
-// into it later. It's not perfect, but the best we can do.
-
-namespace fLS {
-// The meaning of "string" might be different between now and when the
-// macros below get invoked (e.g., if someone is experimenting with
-// other string implementations that get defined after this file is
-// included). Save the current meaning now and use it in the macros.
-typedef std::string clstring;
-
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- const char *value) {
- return new(stringspot) clstring(value);
-}
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- const clstring &value) {
- return new(stringspot) clstring(value);
-}
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- int value);
-} // namespace fLS
-
-#define DECLARE_string(name) namespace fLS { extern ::fLS::clstring& FLAGS_##name; } \
- using fLS::FLAGS_##name
-
-// We need to define a var named FLAGS_no##name so people don't define
-// --string and --nostring. And we need a temporary place to put val
-// so we don't have to evaluate it twice. Two great needs that go
-// great together!
-// The weird 'using' + 'extern' inside the fLS namespace is to work around
-// an unknown compiler bug/issue with the gcc 4.2.1 on SUSE 10. See
-// http://code.google.com/p/google-gflags/issues/detail?id=20
-#define DEFINE_string(name, val, txt) \
- namespace fLS { \
- using ::fLS::clstring; \
- static union { void* align; char s[sizeof(clstring)]; } s_##name[2]; \
- clstring* const FLAGS_no##name = ::fLS:: \
- dont_pass0toDEFINE_string(s_##name[0].s, \
- val); \
- static ::google::FlagRegisterer o_##name( \
- #name, "string", MAYBE_STRIPPED_HELP(txt), __FILE__, \
- s_##name[0].s, new (s_##name[1].s) clstring(*FLAGS_no##name)); \
- extern clstring& FLAGS_##name; \
- using fLS::FLAGS_##name; \
- clstring& FLAGS_##name = *FLAGS_no##name; \
- } \
- using fLS::FLAGS_##name
-
-#endif // SWIG
-
-#endif // GOOGLE_GFLAGS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/linux/x64/include/gflags/gflags_completions.h b/trunk/third_party/google-gflags/gen/arch/linux/x64/include/gflags/gflags_completions.h
deleted file mode 100644
index 9d9ce7a..0000000
--- a/trunk/third_party/google-gflags/gen/arch/linux/x64/include/gflags/gflags_completions.h
+++ /dev/null
@@ -1,121 +0,0 @@
-// Copyright (c) 2008, Google Inc.
-// All rights reserved.
-//
-// Redistribution and use in source and binary forms, with or without
-// modification, are permitted provided that the following conditions are
-// met:
-//
-// * Redistributions of source code must retain the above copyright
-// notice, this list of conditions and the following disclaimer.
-// * Redistributions in binary form must reproduce the above
-// copyright notice, this list of conditions and the following disclaimer
-// in the documentation and/or other materials provided with the
-// distribution.
-// * Neither the name of Google Inc. nor the names of its
-// contributors may be used to endorse or promote products derived from
-// this software without specific prior written permission.
-//
-// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
-// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
-// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
-// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
-// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
-// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
-// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-//
-// ---
-// Author: Dave Nicponski
-//
-// Implement helpful bash-style command line flag completions
-//
-// ** Functional API:
-// HandleCommandLineCompletions() should be called early during
-// program startup, but after command line flag code has been
-// initialized, such as the beginning of HandleCommandLineHelpFlags().
-// It checks the value of the flag --tab_completion_word. If this
-// flag is empty, nothing happens here. If it contains a string,
-// however, then HandleCommandLineCompletions() will hijack the
-// process, attempting to identify the intention behind this
-// completion. Regardless of the outcome of this deduction, the
-// process will be terminated, similar to --helpshort flag
-// handling.
-//
-// ** Overview of Bash completions:
-// Bash can be told to programatically determine completions for the
-// current 'cursor word'. It does this by (in this case) invoking a
-// command with some additional arguments identifying the command
-// being executed, the word being completed, and the previous word
-// (if any). Bash then expects a sequence of output lines to be
-// printed to stdout. If these lines all contain a common prefix
-// longer than the cursor word, bash will replace the cursor word
-// with that common prefix, and display nothing. If there isn't such
-// a common prefix, bash will display the lines in pages using 'more'.
-//
-// ** Strategy taken for command line completions:
-// If we can deduce either the exact flag intended, or a common flag
-// prefix, we'll output exactly that. Otherwise, if information
-// must be displayed to the user, we'll take the opportunity to add
-// some helpful information beyond just the flag name (specifically,
-// we'll include the default flag value and as much of the flag's
-// description as can fit on a single terminal line width, as specified
-// by the flag --tab_completion_columns). Furthermore, we'll try to
-// make bash order the output such that the most useful or relevent
-// flags are the most likely to be shown at the top.
-//
-// ** Additional features:
-// To assist in finding that one really useful flag, substring matching
-// was implemented. Before pressing a <TAB> to get completion for the
-// current word, you can append one or more '?' to the flag to do
-// substring matching. Here's the semantics:
-// --foo<TAB> Show me all flags with names prefixed by 'foo'
-// --foo?<TAB> Show me all flags with 'foo' somewhere in the name
-// --foo??<TAB> Same as prior case, but also search in module
-// definition path for 'foo'
-// --foo???<TAB> Same as prior case, but also search in flag
-// descriptions for 'foo'
-// Finally, we'll trim the output to a relatively small number of
-// flags to keep bash quiet about the verbosity of output. If one
-// really wanted to see all possible matches, appending a '+' to the
-// search word will force the exhaustive list of matches to be printed.
-//
-// ** How to have bash accept completions from a binary:
-// Bash requires that it be informed about each command that programmatic
-// completion should be enabled for. Example addition to a .bashrc
-// file would be (your path to gflags_completions.sh file may differ):
-
-/*
-$ complete -o bashdefault -o default -o nospace -C \
- '/usr/local/bin/gflags_completions.sh --tab_completion_columns $COLUMNS' \
- time env binary_name another_binary [...]
-*/
-
-// This would allow the following to work:
-// $ /path/to/binary_name --vmodule<TAB>
-// Or:
-// $ ./bin/path/another_binary --gfs_u<TAB>
-// (etc)
-//
-// Sadly, it appears that bash gives no easy way to force this behavior for
-// all commands. That's where the "time" in the above example comes in.
-// If you haven't specifically added a command to the list of completion
-// supported commands, you can still get completions by prefixing the
-// entire command with "env".
-// $ env /some/brand/new/binary --vmod<TAB>
-// Assuming that "binary" is a newly compiled binary, this should still
-// produce the expected completion output.
-
-
-#ifndef GOOGLE_GFLAGS_COMPLETIONS_H_
-#define GOOGLE_GFLAGS_COMPLETIONS_H_
-
-namespace google {
-
-void HandleCommandLineCompletions(void);
-
-}
-
-#endif // GOOGLE_GFLAGS_COMPLETIONS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/linux/x64/include/private/config.h b/trunk/third_party/google-gflags/gen/arch/linux/x64/include/private/config.h
deleted file mode 100644
index 98d8e1a..0000000
--- a/trunk/third_party/google-gflags/gen/arch/linux/x64/include/private/config.h
+++ /dev/null
@@ -1,110 +0,0 @@
-/* src/config.h. Generated from config.h.in by configure. */
-/* src/config.h.in. Generated from configure.ac by autoheader. */
-
-/* Always the empty-string on non-windows systems. On windows, should be
- "__declspec(dllexport)". This way, when we compile the dll, we export our
- functions/classes. It's safe to define this here because config.h is only
- used internally, to compile the DLL, and every DLL source file #includes
- "config.h" before anything else. */
-#define GFLAGS_DLL_DECL /**/
-
-/* Namespace for Google classes */
-#define GOOGLE_NAMESPACE ::google
-
-/* Define to 1 if you have the <dlfcn.h> header file. */
-#define HAVE_DLFCN_H 1
-
-/* Define to 1 if you have the <fnmatch.h> header file. */
-#define HAVE_FNMATCH_H 1
-
-/* Define to 1 if you have the <inttypes.h> header file. */
-#define HAVE_INTTYPES_H 1
-
-/* Define to 1 if you have the <memory.h> header file. */
-#define HAVE_MEMORY_H 1
-
-/* define if the compiler implements namespaces */
-#define HAVE_NAMESPACES 1
-
-/* Define if you have POSIX threads libraries and header files. */
-#define HAVE_PTHREAD 1
-
-/* Define to 1 if you have the `putenv' function. */
-#define HAVE_PUTENV 1
-
-/* Define to 1 if you have the `setenv' function. */
-#define HAVE_SETENV 1
-
-/* Define to 1 if you have the <stdint.h> header file. */
-#define HAVE_STDINT_H 1
-
-/* Define to 1 if you have the <stdlib.h> header file. */
-#define HAVE_STDLIB_H 1
-
-/* Define to 1 if you have the <strings.h> header file. */
-#define HAVE_STRINGS_H 1
-
-/* Define to 1 if you have the <string.h> header file. */
-#define HAVE_STRING_H 1
-
-/* Define to 1 if you have the `strtoll' function. */
-#define HAVE_STRTOLL 1
-
-/* Define to 1 if you have the `strtoq' function. */
-#define HAVE_STRTOQ 1
-
-/* Define to 1 if you have the <sys/stat.h> header file. */
-#define HAVE_SYS_STAT_H 1
-
-/* Define to 1 if you have the <sys/types.h> header file. */
-#define HAVE_SYS_TYPES_H 1
-
-/* Define to 1 if you have the <unistd.h> header file. */
-#define HAVE_UNISTD_H 1
-
-/* define if your compiler has __attribute__ */
-#define HAVE___ATTRIBUTE__ 1
-
-/* Define to the sub-directory in which libtool stores uninstalled libraries.
- */
-#define LT_OBJDIR ".libs/"
-
-/* Name of package */
-#define PACKAGE "gflags"
-
-/* Define to the address where bug reports for this package should be sent. */
-#define PACKAGE_BUGREPORT "opensource@google.com"
-
-/* Define to the full name of this package. */
-#define PACKAGE_NAME "gflags"
-
-/* Define to the full name and version of this package. */
-#define PACKAGE_STRING "gflags 1.5"
-
-/* Define to the one symbol short name of this package. */
-#define PACKAGE_TARNAME "gflags"
-
-/* Define to the home page for this package. */
-#define PACKAGE_URL ""
-
-/* Define to the version of this package. */
-#define PACKAGE_VERSION "1.5"
-
-/* Define to necessary symbol if this constant uses a non-standard name on
- your system. */
-/* #undef PTHREAD_CREATE_JOINABLE */
-
-/* Define to 1 if you have the ANSI C header files. */
-#define STDC_HEADERS 1
-
-/* the namespace where STL code like vector<> is defined */
-#define STL_NAMESPACE std
-
-/* Version number of package */
-#define VERSION "1.5"
-
-/* Stops putting the code inside the Google namespace */
-#define _END_GOOGLE_NAMESPACE_ }
-
-/* Puts following code inside the Google namespace */
-#define _START_GOOGLE_NAMESPACE_ namespace google {
diff --git a/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/gflags/gflags.h b/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/gflags/gflags.h
deleted file mode 100644
index 0c2f997..0000000
--- a/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/gflags/gflags.h
+++ /dev/null
@@ -1,585 +0,0 @@
-// Copyright (c) 2006, Google Inc.
-// All rights reserved.
-//
-// Redistribution and use in source and binary forms, with or without
-// modification, are permitted provided that the following conditions are
-// met:
-//
-// * Redistributions of source code must retain the above copyright
-// notice, this list of conditions and the following disclaimer.
-// * Redistributions in binary form must reproduce the above
-// copyright notice, this list of conditions and the following disclaimer
-// in the documentation and/or other materials provided with the
-// distribution.
-// * Neither the name of Google Inc. nor the names of its
-// contributors may be used to endorse or promote products derived from
-// this software without specific prior written permission.
-//
-// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
-// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
-// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
-// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
-// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
-// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
-// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-
-// ---
-// Author: Ray Sidney
-// Revamped and reorganized by Craig Silverstein
-//
-// This is the file that should be included by any file which declares
-// or defines a command line flag or wants to parse command line flags
-// or print a program usage message (which will include information about
-// flags). Executive summary, in the form of an example foo.cc file:
-//
-// #include "foo.h" // foo.h has a line "DECLARE_int32(start);"
-// #include "validators.h" // hypothetical file defining ValidateIsFile()
-//
-// DEFINE_int32(end, 1000, "The last record to read");
-//
-// DEFINE_string(filename, "my_file.txt", "The file to read");
-// // Crash if the specified file does not exist.
-// static bool dummy = RegisterFlagValidator(&FLAGS_filename,
-// &ValidateIsFile);
-//
-// DECLARE_bool(verbose); // some other file has a DEFINE_bool(verbose, ...)
-//
-// void MyFunc() {
-// if (FLAGS_verbose) printf("Records %d-%d\n", FLAGS_start, FLAGS_end);
-// }
-//
-// Then, at the command-line:
-// ./foo --noverbose --start=5 --end=100
-//
-// For more details, see
-// doc/gflags.html
-//
-// --- A note about thread-safety:
-//
-// We describe many functions in this routine as being thread-hostile,
-// thread-compatible, or thread-safe. Here are the meanings we use:
-//
-// thread-safe: it is safe for multiple threads to call this routine
-// (or, when referring to a class, methods of this class)
-// concurrently.
-// thread-hostile: it is not safe for multiple threads to call this
-// routine (or methods of this class) concurrently. In gflags,
-// most thread-hostile routines are intended to be called early in,
-// or even before, main() -- that is, before threads are spawned.
-// thread-compatible: it is safe for multiple threads to read from
-// this variable (when applied to variables), or to call const
-// methods of this class (when applied to classes), as long as no
-// other thread is writing to the variable or calling non-const
-// methods of this class.
-
-#ifndef GOOGLE_GFLAGS_H_
-#define GOOGLE_GFLAGS_H_
-
-#include <string>
-#include <vector>
-
-// We care a lot about number of bits things take up. Unfortunately,
-// systems define their bit-specific ints in a lot of different ways.
-// We use our own way, and have a typedef to get there.
-// Note: these commands below may look like "#if 1" or "#if 0", but
-// that's because they were constructed that way at ./configure time.
-// Look at gflags.h.in to see how they're calculated (based on your config).
-#if 1
-#include <stdint.h> // the normal place uint16_t is defined
-#endif
-#if 1
-#include <sys/types.h> // the normal place u_int16_t is defined
-#endif
-#if 1
-#include <inttypes.h> // a third place for uint16_t or u_int16_t
-#endif
-
-namespace google {
-
-#if 1 // the C99 format
-typedef int32_t int32;
-typedef uint32_t uint32;
-typedef int64_t int64;
-typedef uint64_t uint64;
-#elif 1 // the BSD format
-typedef int32_t int32;
-typedef u_int32_t uint32;
-typedef int64_t int64;
-typedef u_int64_t uint64;
-#elif 0 // the windows (vc7) format
-typedef __int32 int32;
-typedef unsigned __int32 uint32;
-typedef __int64 int64;
-typedef unsigned __int64 uint64;
-#else
-#error Do not know how to define a 32-bit integer quantity on your system
-#endif
-
-// --------------------------------------------------------------------
-// To actually define a flag in a file, use DEFINE_bool,
-// DEFINE_string, etc. at the bottom of this file. You may also find
-// it useful to register a validator with the flag. This ensures that
-// when the flag is parsed from the commandline, or is later set via
-// SetCommandLineOption, we call the validation function. It is _not_
-// called when you assign the value to the flag directly using the = operator.
-//
-// The validation function should return true if the flag value is valid, and
-// false otherwise. If the function returns false for the new setting of the
-// flag, the flag will retain its current value. If it returns false for the
-// default value, ParseCommandLineFlags() will die.
-//
-// This function is safe to call at global construct time (as in the
-// example below).
-//
-// Example use:
-// static bool ValidatePort(const char* flagname, int32 value) {
-// if (value > 0 && value < 32768) // value is ok
-// return true;
-// printf("Invalid value for --%s: %d\n", flagname, (int)value);
-// return false;
-// }
-// DEFINE_int32(port, 0, "What port to listen on");
-// static bool dummy = RegisterFlagValidator(&FLAGS_port, &ValidatePort);
-
-// Returns true if successfully registered, false if not (because the
-// first argument doesn't point to a command-line flag, or because a
-// validator is already registered for this flag).
-bool RegisterFlagValidator(const bool* flag,
- bool (*validate_fn)(const char*, bool));
-bool RegisterFlagValidator(const int32* flag,
- bool (*validate_fn)(const char*, int32));
-bool RegisterFlagValidator(const int64* flag,
- bool (*validate_fn)(const char*, int64));
-bool RegisterFlagValidator(const uint64* flag,
- bool (*validate_fn)(const char*, uint64));
-bool RegisterFlagValidator(const double* flag,
- bool (*validate_fn)(const char*, double));
-bool RegisterFlagValidator(const std::string* flag,
- bool (*validate_fn)(const char*, const std::string&));
-
-
-// --------------------------------------------------------------------
-// These methods are the best way to get access to info about the
-// list of commandline flags. Note that these routines are pretty slow.
-// GetAllFlags: mostly-complete info about the list, sorted by file.
-// ShowUsageWithFlags: pretty-prints the list to stdout (what --help does)
-// ShowUsageWithFlagsRestrict: limit to filenames with restrict as a substr
-//
-// In addition to accessing flags, you can also access argv[0] (the program
-// name) and argv (the entire commandline), which we sock away a copy of.
-// These variables are static, so you should only set them once.
-
-struct CommandLineFlagInfo {
- std::string name; // the name of the flag
- std::string type; // the type of the flag: int32, etc
- std::string description; // the "help text" associated with the flag
- std::string current_value; // the current value, as a string
- std::string default_value; // the default value, as a string
- std::string filename; // 'cleaned' version of filename holding the flag
- bool has_validator_fn; // true if RegisterFlagValidator called on flag
- bool is_default; // true if the flag has the default value and
- // has not been set explicitly from the cmdline
- // or via SetCommandLineOption
-};
-
-// Using this inside of a validator is a recipe for a deadlock.
-// TODO(wojtekm) Fix locking when validators are running, to make it safe to
-// call validators during ParseAllFlags.
-// Also make sure then to uncomment the corresponding unit test in
-// commandlineflags_unittest.sh
-extern void GetAllFlags(std::vector<CommandLineFlagInfo>* OUTPUT);
-// These two are actually defined in commandlineflags_reporting.cc.
-extern void ShowUsageWithFlags(const char *argv0); // what --help does
-extern void ShowUsageWithFlagsRestrict(const char *argv0, const char *restrict);
-
-// Create a descriptive string for a flag.
-// Goes to some trouble to make pretty line breaks.
-extern std::string DescribeOneFlag(const CommandLineFlagInfo& flag);
-
-// Thread-hostile; meant to be called before any threads are spawned.
-extern void SetArgv(int argc, const char** argv);
-// The following functions are thread-safe as long as SetArgv() is
-// only called before any threads start.
-extern const std::vector<std::string>& GetArgvs(); // all of argv as a vector
-extern const char* GetArgv(); // all of argv as a string
-extern const char* GetArgv0(); // only argv0
-extern uint32 GetArgvSum(); // simple checksum of argv
-extern const char* ProgramInvocationName(); // argv0, or "UNKNOWN" if not set
-extern const char* ProgramInvocationShortName(); // basename(argv0)
-// ProgramUsage() is thread-safe as long as SetUsageMessage() is only
-// called before any threads start.
-extern const char* ProgramUsage(); // string set by SetUsageMessage()
-
-
-// --------------------------------------------------------------------
-// Normally you access commandline flags by just saying "if (FLAGS_foo)"
-// or whatever, and set them by calling "FLAGS_foo = bar" (or, more
-// commonly, via the DEFINE_foo macro). But if you need a bit more
-// control, we have programmatic ways to get/set the flags as well.
-// These programmatic ways to access flags are thread-safe, but direct
-// access is only thread-compatible.
-
-// Return true iff the flagname was found.
-// OUTPUT is set to the flag's value, or unchanged if we return false.
-extern bool GetCommandLineOption(const char* name, std::string* OUTPUT);
-
-// Return true iff the flagname was found. OUTPUT is set to the flag's
-// CommandLineFlagInfo or unchanged if we return false.
-extern bool GetCommandLineFlagInfo(const char* name,
- CommandLineFlagInfo* OUTPUT);
-
-// Return the CommandLineFlagInfo of the flagname. exit() if name not found.
-// Example usage, to check if a flag's value is currently the default value:
-// if (GetCommandLineFlagInfoOrDie("foo").is_default) ...
-extern CommandLineFlagInfo GetCommandLineFlagInfoOrDie(const char* name);
-
-enum FlagSettingMode {
- // update the flag's value (can call this multiple times).
- SET_FLAGS_VALUE,
- // update the flag's value, but *only if* it has not yet been updated
- // with SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef".
- SET_FLAG_IF_DEFAULT,
- // set the flag's default value to this. If the flag has not yet updated
- // yet (via SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef")
- // change the flag's current value to the new default value as well.
- SET_FLAGS_DEFAULT
-};
-
-// Set a particular flag ("command line option"). Returns a string
-// describing the new value that the option has been set to. The
-// return value API is not well-specified, so basically just depend on
-// it to be empty if the setting failed for some reason -- the name is
-// not a valid flag name, or the value is not a valid value -- and
-// non-empty else.
-
-// SetCommandLineOption uses set_mode == SET_FLAGS_VALUE (the common case)
-extern std::string SetCommandLineOption(const char* name, const char* value);
-extern std::string SetCommandLineOptionWithMode(const char* name, const char* value,
- FlagSettingMode set_mode);
-
-
-// --------------------------------------------------------------------
-// Saves the states (value, default value, whether the user has set
-// the flag, registered validators, etc) of all flags, and restores
-// them when the FlagSaver is destroyed. This is very useful in
-// tests, say, when you want to let your tests change the flags, but
-// make sure that they get reverted to the original states when your
-// test is complete.
-//
-// Example usage:
-// void TestFoo() {
-// FlagSaver s1;
-// FLAG_foo = false;
-// FLAG_bar = "some value";
-//
-// // test happens here. You can return at any time
-// // without worrying about restoring the FLAG values.
-// }
-//
-// Note: This class is marked with __attribute__((unused)) because all the
-// work is done in the constructor and destructor, so in the standard
-// usage example above, the compiler would complain that it's an
-// unused variable.
-//
-// This class is thread-safe.
-
-class FlagSaver {
- public:
- FlagSaver();
- ~FlagSaver();
-
- private:
- class FlagSaverImpl* impl_; // we use pimpl here to keep API steady
-
- FlagSaver(const FlagSaver&); // no copying!
- void operator=(const FlagSaver&);
-} __attribute__ ((unused));
-
-// --------------------------------------------------------------------
-// Some deprecated or hopefully-soon-to-be-deprecated functions.
-
-// This is often used for logging. TODO(csilvers): figure out a better way
-extern std::string CommandlineFlagsIntoString();
-// Usually where this is used, a FlagSaver should be used instead.
-extern bool ReadFlagsFromString(const std::string& flagfilecontents,
- const char* prog_name,
- bool errors_are_fatal); // uses SET_FLAGS_VALUE
-
-// These let you manually implement --flagfile functionality.
-// DEPRECATED.
-extern bool AppendFlagsIntoFile(const std::string& filename, const char* prog_name);
-extern bool SaveCommandFlags(); // actually defined in google.cc !
-extern bool ReadFromFlagsFile(const std::string& filename, const char* prog_name,
- bool errors_are_fatal); // uses SET_FLAGS_VALUE
-
-
-// --------------------------------------------------------------------
-// Useful routines for initializing flags from the environment.
-// In each case, if 'varname' does not exist in the environment
-// return defval. If 'varname' does exist but is not valid
-// (e.g., not a number for an int32 flag), abort with an error.
-// Otherwise, return the value. NOTE: for booleans, for true use
-// 't' or 'T' or 'true' or '1', for false 'f' or 'F' or 'false' or '0'.
-
-extern bool BoolFromEnv(const char *varname, bool defval);
-extern int32 Int32FromEnv(const char *varname, int32 defval);
-extern int64 Int64FromEnv(const char *varname, int64 defval);
-extern uint64 Uint64FromEnv(const char *varname, uint64 defval);
-extern double DoubleFromEnv(const char *varname, double defval);
-extern const char *StringFromEnv(const char *varname, const char *defval);
-
-
-// --------------------------------------------------------------------
-// The next two functions parse commandlineflags from main():
-
-// Set the "usage" message for this program. For example:
-// string usage("This program does nothing. Sample usage:\n");
-// usage += argv[0] + " <uselessarg1> <uselessarg2>";
-// SetUsageMessage(usage);
-// Do not include commandline flags in the usage: we do that for you!
-// Thread-hostile; meant to be called before any threads are spawned.
-extern void SetUsageMessage(const std::string& usage);
-
-// Looks for flags in argv and parses them. Rearranges argv to put
-// flags first, or removes them entirely if remove_flags is true.
-// If a flag is defined more than once in the command line or flag
-// file, the last definition is used. Returns the index (into argv)
-// of the first non-flag argument.
-// See top-of-file for more details on this function.
-#ifndef SWIG // In swig, use ParseCommandLineFlagsScript() instead.
-extern uint32 ParseCommandLineFlags(int *argc, char*** argv,
- bool remove_flags);
-#endif
-
-
-// Calls to ParseCommandLineNonHelpFlags and then to
-// HandleCommandLineHelpFlags can be used instead of a call to
-// ParseCommandLineFlags during initialization, in order to allow for
-// changing default values for some FLAGS (via
-// e.g. SetCommandLineOptionWithMode calls) between the time of
-// command line parsing and the time of dumping help information for
-// the flags as a result of command line parsing. If a flag is
-// defined more than once in the command line or flag file, the last
-// definition is used. Returns the index (into argv) of the first
-// non-flag argument. (If remove_flags is true, will always return 1.)
-extern uint32 ParseCommandLineNonHelpFlags(int *argc, char*** argv,
- bool remove_flags);
-// This is actually defined in commandlineflags_reporting.cc.
-// This function is misnamed (it also handles --version, etc.), but
-// it's too late to change that now. :-(
-extern void HandleCommandLineHelpFlags(); // in commandlineflags_reporting.cc
-
-// Allow command line reparsing. Disables the error normally
-// generated when an unknown flag is found, since it may be found in a
-// later parse. Thread-hostile; meant to be called before any threads
-// are spawned.
-extern void AllowCommandLineReparsing();
-
-// Reparse the flags that have not yet been recognized. Only flags
-// registered since the last parse will be recognized. Any flag value
-// must be provided as part of the argument using "=", not as a
-// separate command line argument that follows the flag argument.
-// Intended for handling flags from dynamically loaded libraries,
-// since their flags are not registered until they are loaded.
-// Returns the index (into the original argv) of the first non-flag
-// argument. (If remove_flags is true, will always return 1.)
-extern uint32 ReparseCommandLineNonHelpFlags();
-
-// Clean up memory allocated by flags. This is only needed to reduce
-// the quantity of "potentially leaked" reports emitted by memory
-// debugging tools such as valgrind. It is not required for normal
-// operation, or for the perftools heap-checker. It must only be called
-// when the process is about to exit, and all threads that might
-// access flags are quiescent. Referencing flags after this is called
-// will have unexpected consequences. This is not safe to run when
-// multiple threads might be running: the function is thread-hostile.
-extern void ShutDownCommandLineFlags();
-
-
-// --------------------------------------------------------------------
-// Now come the command line flag declaration/definition macros that
-// will actually be used. They're kind of hairy. A major reason
-// for this is initialization: we want people to be able to access
-// variables in global constructors and have that not crash, even if
-// their global constructor runs before the global constructor here.
-// (Obviously, we can't guarantee the flags will have the correct
-// default value in that case, but at least accessing them is safe.)
-// The only way to do that is have flags point to a static buffer.
-// So we make one, using a union to ensure proper alignment, and
-// then use placement-new to actually set up the flag with the
-// correct default value. In the same vein, we have to worry about
-// flag access in global destructors, so FlagRegisterer has to be
-// careful never to destroy the flag-values it constructs.
-//
-// Note that when we define a flag variable FLAGS_<name>, we also
-// preemptively define a junk variable, FLAGS_no<name>. This is to
-// cause a link-time error if someone tries to define 2 flags with
-// names like "logging" and "nologging". We do this because a bool
-// flag FLAG can be set from the command line to true with a "-FLAG"
-// argument, and to false with a "-noFLAG" argument, and so this can
-// potentially avert confusion.
-//
-// We also put flags into their own namespace. It is purposefully
-// named in an opaque way that people should have trouble typing
-// directly. The idea is that DEFINE puts the flag in the weird
-// namespace, and DECLARE imports the flag from there into the current
-// namespace. The net result is to force people to use DECLARE to get
-// access to a flag, rather than saying "extern bool FLAGS_whatever;"
-// or some such instead. We want this so we can put extra
-// functionality (like sanity-checking) in DECLARE if we want, and
-// make sure it is picked up everywhere.
-//
-// We also put the type of the variable in the namespace, so that
-// people can't DECLARE_int32 something that they DEFINE_bool'd
-// elsewhere.
-
-class FlagRegisterer {
- public:
- FlagRegisterer(const char* name, const char* type,
- const char* help, const char* filename,
- void* current_storage, void* defvalue_storage);
-};
-
-extern bool FlagsTypeWarn(const char *name);
-
-// If your application #defines STRIP_FLAG_HELP to a non-zero value
-// before #including this file, we remove the help message from the
-// binary file. This can reduce the size of the resulting binary
-// somewhat, and may also be useful for security reasons.
-
-extern const char kStrippedFlagHelp[];
-
-}
-
-#ifndef SWIG // In swig, ignore the main flag declarations
-
-#if defined(STRIP_FLAG_HELP) && STRIP_FLAG_HELP > 0
-// Need this construct to avoid the 'defined but not used' warning.
-#define MAYBE_STRIPPED_HELP(txt) (false ? (txt) : ::google::kStrippedFlagHelp)
-#else
-#define MAYBE_STRIPPED_HELP(txt) txt
-#endif
-
-// Each command-line flag has two variables associated with it: one
-// with the current value, and one with the default value. However,
-// we have a third variable, which is where value is assigned; it's a
-// constant. This guarantees that FLAG_##value is initialized at
-// static initialization time (e.g. before program-start) rather than
-// than global construction time (which is after program-start but
-// before main), at least when 'value' is a compile-time constant. We
-// use a small trick for the "default value" variable, and call it
-// FLAGS_no<name>. This serves the second purpose of assuring a
-// compile error if someone tries to define a flag named no<name>
-// which is illegal (--foo and --nofoo both affect the "foo" flag).
-#define DEFINE_VARIABLE(type, shorttype, name, value, help) \
- namespace fL##shorttype { \
- static const type FLAGS_nono##name = value; \
- type FLAGS_##name = FLAGS_nono##name; \
- type FLAGS_no##name = FLAGS_nono##name; \
- static ::google::FlagRegisterer o_##name( \
- #name, #type, MAYBE_STRIPPED_HELP(help), __FILE__, \
- &FLAGS_##name, &FLAGS_no##name); \
- } \
- using fL##shorttype::FLAGS_##name
-
-#define DECLARE_VARIABLE(type, shorttype, name) \
- namespace fL##shorttype { \
- extern type FLAGS_##name; \
- } \
- using fL##shorttype::FLAGS_##name
-
-// For DEFINE_bool, we want to do the extra check that the passed-in
-// value is actually a bool, and not a string or something that can be
-// coerced to a bool. These declarations (no definition needed!) will
-// help us do that, and never evaluate From, which is important.
-// We'll use 'sizeof(IsBool(val))' to distinguish. This code requires
-// that the compiler have different sizes for bool & double. Since
-// this is not guaranteed by the standard, we check it with a
-// compile-time assert (msg[-1] will give a compile-time error).
-namespace fLB {
-struct CompileAssert {};
-typedef CompileAssert expected_sizeof_double_neq_sizeof_bool[
- (sizeof(double) != sizeof(bool)) ? 1 : -1];
-template<typename From> double IsBoolFlag(const From& from);
-bool IsBoolFlag(bool from);
-} // namespace fLB
-
-#define DECLARE_bool(name) DECLARE_VARIABLE(bool, B, name)
-#define DEFINE_bool(name, val, txt) \
- namespace fLB { \
- typedef ::fLB::CompileAssert FLAG_##name##_value_is_not_a_bool[ \
- (sizeof(::fLB::IsBoolFlag(val)) != sizeof(double)) ? 1 : -1]; \
- } \
- DEFINE_VARIABLE(bool, B, name, val, txt)
-
-#define DECLARE_int32(name) DECLARE_VARIABLE(::google::int32, I, name)
-#define DEFINE_int32(name,val,txt) DEFINE_VARIABLE(::google::int32, I, name, val, txt)
-
-#define DECLARE_int64(name) DECLARE_VARIABLE(::google::int64, I64, name)
-#define DEFINE_int64(name,val,txt) DEFINE_VARIABLE(::google::int64, I64, name, val, txt)
-
-#define DECLARE_uint64(name) DECLARE_VARIABLE(::google::uint64, U64, name)
-#define DEFINE_uint64(name,val,txt) DEFINE_VARIABLE(::google::uint64, U64, name, val, txt)
-
-#define DECLARE_double(name) DECLARE_VARIABLE(double, D, name)
-#define DEFINE_double(name, val, txt) DEFINE_VARIABLE(double, D, name, val, txt)
-
-// Strings are trickier, because they're not a POD, so we can't
-// construct them at static-initialization time (instead they get
-// constructed at global-constructor time, which is much later). To
-// try to avoid crashes in that case, we use a char buffer to store
-// the string, which we can static-initialize, and then placement-new
-// into it later. It's not perfect, but the best we can do.
-
-namespace fLS {
-// The meaning of "string" might be different between now and when the
-// macros below get invoked (e.g., if someone is experimenting with
-// other string implementations that get defined after this file is
-// included). Save the current meaning now and use it in the macros.
-typedef std::string clstring;
-
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- const char *value) {
- return new(stringspot) clstring(value);
-}
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- const clstring &value) {
- return new(stringspot) clstring(value);
-}
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- int value);
-} // namespace fLS
-
-#define DECLARE_string(name) namespace fLS { extern ::fLS::clstring& FLAGS_##name; } \
- using fLS::FLAGS_##name
-
-// We need to define a var named FLAGS_no##name so people don't define
-// --string and --nostring. And we need a temporary place to put val
-// so we don't have to evaluate it twice. Two great needs that go
-// great together!
-// The weird 'using' + 'extern' inside the fLS namespace is to work around
-// an unknown compiler bug/issue with the gcc 4.2.1 on SUSE 10. See
-// http://code.google.com/p/google-gflags/issues/detail?id=20
-#define DEFINE_string(name, val, txt) \
- namespace fLS { \
- using ::fLS::clstring; \
- static union { void* align; char s[sizeof(clstring)]; } s_##name[2]; \
- clstring* const FLAGS_no##name = ::fLS:: \
- dont_pass0toDEFINE_string(s_##name[0].s, \
- val); \
- static ::google::FlagRegisterer o_##name( \
- #name, "string", MAYBE_STRIPPED_HELP(txt), __FILE__, \
- s_##name[0].s, new (s_##name[1].s) clstring(*FLAGS_no##name)); \
- extern clstring& FLAGS_##name; \
- using fLS::FLAGS_##name; \
- clstring& FLAGS_##name = *FLAGS_no##name; \
- } \
- using fLS::FLAGS_##name
-
-#endif // SWIG
-
-#endif // GOOGLE_GFLAGS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/gflags/gflags_completions.h b/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/gflags/gflags_completions.h
deleted file mode 100644
index 9d9ce7a..0000000
--- a/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/gflags/gflags_completions.h
+++ /dev/null
@@ -1,121 +0,0 @@
-// Copyright (c) 2008, Google Inc.
-// All rights reserved.
-//
-// Redistribution and use in source and binary forms, with or without
-// modification, are permitted provided that the following conditions are
-// met:
-//
-// * Redistributions of source code must retain the above copyright
-// notice, this list of conditions and the following disclaimer.
-// * Redistributions in binary form must reproduce the above
-// copyright notice, this list of conditions and the following disclaimer
-// in the documentation and/or other materials provided with the
-// distribution.
-// * Neither the name of Google Inc. nor the names of its
-// contributors may be used to endorse or promote products derived from
-// this software without specific prior written permission.
-//
-// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
-// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
-// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
-// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
-// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
-// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
-// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-//
-// ---
-// Author: Dave Nicponski
-//
-// Implement helpful bash-style command line flag completions
-//
-// ** Functional API:
-// HandleCommandLineCompletions() should be called early during
-// program startup, but after command line flag code has been
-// initialized, such as the beginning of HandleCommandLineHelpFlags().
-// It checks the value of the flag --tab_completion_word. If this
-// flag is empty, nothing happens here. If it contains a string,
-// however, then HandleCommandLineCompletions() will hijack the
-// process, attempting to identify the intention behind this
-// completion. Regardless of the outcome of this deduction, the
-// process will be terminated, similar to --helpshort flag
-// handling.
-//
-// ** Overview of Bash completions:
-// Bash can be told to programatically determine completions for the
-// current 'cursor word'. It does this by (in this case) invoking a
-// command with some additional arguments identifying the command
-// being executed, the word being completed, and the previous word
-// (if any). Bash then expects a sequence of output lines to be
-// printed to stdout. If these lines all contain a common prefix
-// longer than the cursor word, bash will replace the cursor word
-// with that common prefix, and display nothing. If there isn't such
-// a common prefix, bash will display the lines in pages using 'more'.
-//
-// ** Strategy taken for command line completions:
-// If we can deduce either the exact flag intended, or a common flag
-// prefix, we'll output exactly that. Otherwise, if information
-// must be displayed to the user, we'll take the opportunity to add
-// some helpful information beyond just the flag name (specifically,
-// we'll include the default flag value and as much of the flag's
-// description as can fit on a single terminal line width, as specified
-// by the flag --tab_completion_columns). Furthermore, we'll try to
-// make bash order the output such that the most useful or relevent
-// flags are the most likely to be shown at the top.
-//
-// ** Additional features:
-// To assist in finding that one really useful flag, substring matching
-// was implemented. Before pressing a <TAB> to get completion for the
-// current word, you can append one or more '?' to the flag to do
-// substring matching. Here's the semantics:
-// --foo<TAB> Show me all flags with names prefixed by 'foo'
-// --foo?<TAB> Show me all flags with 'foo' somewhere in the name
-// --foo??<TAB> Same as prior case, but also search in module
-// definition path for 'foo'
-// --foo???<TAB> Same as prior case, but also search in flag
-// descriptions for 'foo'
-// Finally, we'll trim the output to a relatively small number of
-// flags to keep bash quiet about the verbosity of output. If one
-// really wanted to see all possible matches, appending a '+' to the
-// search word will force the exhaustive list of matches to be printed.
-//
-// ** How to have bash accept completions from a binary:
-// Bash requires that it be informed about each command that programmatic
-// completion should be enabled for. Example addition to a .bashrc
-// file would be (your path to gflags_completions.sh file may differ):
-
-/*
-$ complete -o bashdefault -o default -o nospace -C \
- '/usr/local/bin/gflags_completions.sh --tab_completion_columns $COLUMNS' \
- time env binary_name another_binary [...]
-*/
-
-// This would allow the following to work:
-// $ /path/to/binary_name --vmodule<TAB>
-// Or:
-// $ ./bin/path/another_binary --gfs_u<TAB>
-// (etc)
-//
-// Sadly, it appears that bash gives no easy way to force this behavior for
-// all commands. That's where the "time" in the above example comes in.
-// If you haven't specifically added a command to the list of completion
-// supported commands, you can still get completions by prefixing the
-// entire command with "env".
-// $ env /some/brand/new/binary --vmod<TAB>
-// Assuming that "binary" is a newly compiled binary, this should still
-// produce the expected completion output.
-
-
-#ifndef GOOGLE_GFLAGS_COMPLETIONS_H_
-#define GOOGLE_GFLAGS_COMPLETIONS_H_
-
-namespace google {
-
-void HandleCommandLineCompletions(void);
-
-}
-
-#endif // GOOGLE_GFLAGS_COMPLETIONS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/private/config.h b/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/private/config.h
deleted file mode 100644
index 98d8e1a..0000000
--- a/trunk/third_party/google-gflags/gen/arch/mac/ia32/include/private/config.h
+++ /dev/null
@@ -1,110 +0,0 @@
-/* src/config.h. Generated from config.h.in by configure. */
-/* src/config.h.in. Generated from configure.ac by autoheader. */
-
-/* Always the empty-string on non-windows systems. On windows, should be
- "__declspec(dllexport)". This way, when we compile the dll, we export our
- functions/classes. It's safe to define this here because config.h is only
- used internally, to compile the DLL, and every DLL source file #includes
- "config.h" before anything else. */
-#define GFLAGS_DLL_DECL /**/
-
-/* Namespace for Google classes */
-#define GOOGLE_NAMESPACE ::google
-
-/* Define to 1 if you have the <dlfcn.h> header file. */
-#define HAVE_DLFCN_H 1
-
-/* Define to 1 if you have the <fnmatch.h> header file. */
-#define HAVE_FNMATCH_H 1
-
-/* Define to 1 if you have the <inttypes.h> header file. */
-#define HAVE_INTTYPES_H 1
-
-/* Define to 1 if you have the <memory.h> header file. */
-#define HAVE_MEMORY_H 1
-
-/* define if the compiler implements namespaces */
-#define HAVE_NAMESPACES 1
-
-/* Define if you have POSIX threads libraries and header files. */
-#define HAVE_PTHREAD 1
-
-/* Define to 1 if you have the `putenv' function. */
-#define HAVE_PUTENV 1
-
-/* Define to 1 if you have the `setenv' function. */
-#define HAVE_SETENV 1
-
-/* Define to 1 if you have the <stdint.h> header file. */
-#define HAVE_STDINT_H 1
-
-/* Define to 1 if you have the <stdlib.h> header file. */
-#define HAVE_STDLIB_H 1
-
-/* Define to 1 if you have the <strings.h> header file. */
-#define HAVE_STRINGS_H 1
-
-/* Define to 1 if you have the <string.h> header file. */
-#define HAVE_STRING_H 1
-
-/* Define to 1 if you have the `strtoll' function. */
-#define HAVE_STRTOLL 1
-
-/* Define to 1 if you have the `strtoq' function. */
-#define HAVE_STRTOQ 1
-
-/* Define to 1 if you have the <sys/stat.h> header file. */
-#define HAVE_SYS_STAT_H 1
-
-/* Define to 1 if you have the <sys/types.h> header file. */
-#define HAVE_SYS_TYPES_H 1
-
-/* Define to 1 if you have the <unistd.h> header file. */
-#define HAVE_UNISTD_H 1
-
-/* define if your compiler has __attribute__ */
-#define HAVE___ATTRIBUTE__ 1
-
-/* Define to the sub-directory in which libtool stores uninstalled libraries.
- */
-#define LT_OBJDIR ".libs/"
-
-/* Name of package */
-#define PACKAGE "gflags"
-
-/* Define to the address where bug reports for this package should be sent. */
-#define PACKAGE_BUGREPORT "opensource@google.com"
-
-/* Define to the full name of this package. */
-#define PACKAGE_NAME "gflags"
-
-/* Define to the full name and version of this package. */
-#define PACKAGE_STRING "gflags 1.5"
-
-/* Define to the one symbol short name of this package. */
-#define PACKAGE_TARNAME "gflags"
-
-/* Define to the home page for this package. */
-#define PACKAGE_URL ""
-
-/* Define to the version of this package. */
-#define PACKAGE_VERSION "1.5"
-
-/* Define to necessary symbol if this constant uses a non-standard name on
- your system. */
-/* #undef PTHREAD_CREATE_JOINABLE */
-
-/* Define to 1 if you have the ANSI C header files. */
-#define STDC_HEADERS 1
-
-/* the namespace where STL code like vector<> is defined */
-#define STL_NAMESPACE std
-
-/* Version number of package */
-#define VERSION "1.5"
-
-/* Stops putting the code inside the Google namespace */
-#define _END_GOOGLE_NAMESPACE_ }
-
-/* Puts following code inside the Google namespace */
-#define _START_GOOGLE_NAMESPACE_ namespace google {
diff --git a/trunk/third_party/google-gflags/gen/arch/mac/x64/include/gflags/gflags.h b/trunk/third_party/google-gflags/gen/arch/mac/x64/include/gflags/gflags.h
deleted file mode 100644
index 0c2f997..0000000
--- a/trunk/third_party/google-gflags/gen/arch/mac/x64/include/gflags/gflags.h
+++ /dev/null
@@ -1,585 +0,0 @@
-// Copyright (c) 2006, Google Inc.
-// All rights reserved.
-//
-// Redistribution and use in source and binary forms, with or without
-// modification, are permitted provided that the following conditions are
-// met:
-//
-// * Redistributions of source code must retain the above copyright
-// notice, this list of conditions and the following disclaimer.
-// * Redistributions in binary form must reproduce the above
-// copyright notice, this list of conditions and the following disclaimer
-// in the documentation and/or other materials provided with the
-// distribution.
-// * Neither the name of Google Inc. nor the names of its
-// contributors may be used to endorse or promote products derived from
-// this software without specific prior written permission.
-//
-// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
-// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
-// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
-// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
-// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
-// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
-// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-
-// ---
-// Author: Ray Sidney
-// Revamped and reorganized by Craig Silverstein
-//
-// This is the file that should be included by any file which declares
-// or defines a command line flag or wants to parse command line flags
-// or print a program usage message (which will include information about
-// flags). Executive summary, in the form of an example foo.cc file:
-//
-// #include "foo.h" // foo.h has a line "DECLARE_int32(start);"
-// #include "validators.h" // hypothetical file defining ValidateIsFile()
-//
-// DEFINE_int32(end, 1000, "The last record to read");
-//
-// DEFINE_string(filename, "my_file.txt", "The file to read");
-// // Crash if the specified file does not exist.
-// static bool dummy = RegisterFlagValidator(&FLAGS_filename,
-// &ValidateIsFile);
-//
-// DECLARE_bool(verbose); // some other file has a DEFINE_bool(verbose, ...)
-//
-// void MyFunc() {
-// if (FLAGS_verbose) printf("Records %d-%d\n", FLAGS_start, FLAGS_end);
-// }
-//
-// Then, at the command-line:
-// ./foo --noverbose --start=5 --end=100
-//
-// For more details, see
-// doc/gflags.html
-//
-// --- A note about thread-safety:
-//
-// We describe many functions in this routine as being thread-hostile,
-// thread-compatible, or thread-safe. Here are the meanings we use:
-//
-// thread-safe: it is safe for multiple threads to call this routine
-// (or, when referring to a class, methods of this class)
-// concurrently.
-// thread-hostile: it is not safe for multiple threads to call this
-// routine (or methods of this class) concurrently. In gflags,
-// most thread-hostile routines are intended to be called early in,
-// or even before, main() -- that is, before threads are spawned.
-// thread-compatible: it is safe for multiple threads to read from
-// this variable (when applied to variables), or to call const
-// methods of this class (when applied to classes), as long as no
-// other thread is writing to the variable or calling non-const
-// methods of this class.
-
-#ifndef GOOGLE_GFLAGS_H_
-#define GOOGLE_GFLAGS_H_
-
-#include <string>
-#include <vector>
-
-// We care a lot about number of bits things take up. Unfortunately,
-// systems define their bit-specific ints in a lot of different ways.
-// We use our own way, and have a typedef to get there.
-// Note: these commands below may look like "#if 1" or "#if 0", but
-// that's because they were constructed that way at ./configure time.
-// Look at gflags.h.in to see how they're calculated (based on your config).
-#if 1
-#include <stdint.h> // the normal place uint16_t is defined
-#endif
-#if 1
-#include <sys/types.h> // the normal place u_int16_t is defined
-#endif
-#if 1
-#include <inttypes.h> // a third place for uint16_t or u_int16_t
-#endif
-
-namespace google {
-
-#if 1 // the C99 format
-typedef int32_t int32;
-typedef uint32_t uint32;
-typedef int64_t int64;
-typedef uint64_t uint64;
-#elif 1 // the BSD format
-typedef int32_t int32;
-typedef u_int32_t uint32;
-typedef int64_t int64;
-typedef u_int64_t uint64;
-#elif 0 // the windows (vc7) format
-typedef __int32 int32;
-typedef unsigned __int32 uint32;
-typedef __int64 int64;
-typedef unsigned __int64 uint64;
-#else
-#error Do not know how to define a 32-bit integer quantity on your system
-#endif
-
-// --------------------------------------------------------------------
-// To actually define a flag in a file, use DEFINE_bool,
-// DEFINE_string, etc. at the bottom of this file. You may also find
-// it useful to register a validator with the flag. This ensures that
-// when the flag is parsed from the commandline, or is later set via
-// SetCommandLineOption, we call the validation function. It is _not_
-// called when you assign the value to the flag directly using the = operator.
-//
-// The validation function should return true if the flag value is valid, and
-// false otherwise. If the function returns false for the new setting of the
-// flag, the flag will retain its current value. If it returns false for the
-// default value, ParseCommandLineFlags() will die.
-//
-// This function is safe to call at global construct time (as in the
-// example below).
-//
-// Example use:
-// static bool ValidatePort(const char* flagname, int32 value) {
-// if (value > 0 && value < 32768) // value is ok
-// return true;
-// printf("Invalid value for --%s: %d\n", flagname, (int)value);
-// return false;
-// }
-// DEFINE_int32(port, 0, "What port to listen on");
-// static bool dummy = RegisterFlagValidator(&FLAGS_port, &ValidatePort);
-
-// Returns true if successfully registered, false if not (because the
-// first argument doesn't point to a command-line flag, or because a
-// validator is already registered for this flag).
-bool RegisterFlagValidator(const bool* flag,
- bool (*validate_fn)(const char*, bool));
-bool RegisterFlagValidator(const int32* flag,
- bool (*validate_fn)(const char*, int32));
-bool RegisterFlagValidator(const int64* flag,
- bool (*validate_fn)(const char*, int64));
-bool RegisterFlagValidator(const uint64* flag,
- bool (*validate_fn)(const char*, uint64));
-bool RegisterFlagValidator(const double* flag,
- bool (*validate_fn)(const char*, double));
-bool RegisterFlagValidator(const std::string* flag,
- bool (*validate_fn)(const char*, const std::string&));
-
-
-// --------------------------------------------------------------------
-// These methods are the best way to get access to info about the
-// list of commandline flags. Note that these routines are pretty slow.
-// GetAllFlags: mostly-complete info about the list, sorted by file.
-// ShowUsageWithFlags: pretty-prints the list to stdout (what --help does)
-// ShowUsageWithFlagsRestrict: limit to filenames with restrict as a substr
-//
-// In addition to accessing flags, you can also access argv[0] (the program
-// name) and argv (the entire commandline), which we sock away a copy of.
-// These variables are static, so you should only set them once.
-
-struct CommandLineFlagInfo {
- std::string name; // the name of the flag
- std::string type; // the type of the flag: int32, etc
- std::string description; // the "help text" associated with the flag
- std::string current_value; // the current value, as a string
- std::string default_value; // the default value, as a string
- std::string filename; // 'cleaned' version of filename holding the flag
- bool has_validator_fn; // true if RegisterFlagValidator called on flag
- bool is_default; // true if the flag has the default value and
- // has not been set explicitly from the cmdline
- // or via SetCommandLineOption
-};
-
-// Using this inside of a validator is a recipe for a deadlock.
-// TODO(wojtekm) Fix locking when validators are running, to make it safe to
-// call validators during ParseAllFlags.
-// Also make sure then to uncomment the corresponding unit test in
-// commandlineflags_unittest.sh
-extern void GetAllFlags(std::vector<CommandLineFlagInfo>* OUTPUT);
-// These two are actually defined in commandlineflags_reporting.cc.
-extern void ShowUsageWithFlags(const char *argv0); // what --help does
-extern void ShowUsageWithFlagsRestrict(const char *argv0, const char *restrict);
-
-// Create a descriptive string for a flag.
-// Goes to some trouble to make pretty line breaks.
-extern std::string DescribeOneFlag(const CommandLineFlagInfo& flag);
-
-// Thread-hostile; meant to be called before any threads are spawned.
-extern void SetArgv(int argc, const char** argv);
-// The following functions are thread-safe as long as SetArgv() is
-// only called before any threads start.
-extern const std::vector<std::string>& GetArgvs(); // all of argv as a vector
-extern const char* GetArgv(); // all of argv as a string
-extern const char* GetArgv0(); // only argv0
-extern uint32 GetArgvSum(); // simple checksum of argv
-extern const char* ProgramInvocationName(); // argv0, or "UNKNOWN" if not set
-extern const char* ProgramInvocationShortName(); // basename(argv0)
-// ProgramUsage() is thread-safe as long as SetUsageMessage() is only
-// called before any threads start.
-extern const char* ProgramUsage(); // string set by SetUsageMessage()
-
-
-// --------------------------------------------------------------------
-// Normally you access commandline flags by just saying "if (FLAGS_foo)"
-// or whatever, and set them by calling "FLAGS_foo = bar" (or, more
-// commonly, via the DEFINE_foo macro). But if you need a bit more
-// control, we have programmatic ways to get/set the flags as well.
-// These programmatic ways to access flags are thread-safe, but direct
-// access is only thread-compatible.
-
-// Return true iff the flagname was found.
-// OUTPUT is set to the flag's value, or unchanged if we return false.
-extern bool GetCommandLineOption(const char* name, std::string* OUTPUT);
-
-// Return true iff the flagname was found. OUTPUT is set to the flag's
-// CommandLineFlagInfo or unchanged if we return false.
-extern bool GetCommandLineFlagInfo(const char* name,
- CommandLineFlagInfo* OUTPUT);
-
-// Return the CommandLineFlagInfo of the flagname. exit() if name not found.
-// Example usage, to check if a flag's value is currently the default value:
-// if (GetCommandLineFlagInfoOrDie("foo").is_default) ...
-extern CommandLineFlagInfo GetCommandLineFlagInfoOrDie(const char* name);
-
-enum FlagSettingMode {
- // update the flag's value (can call this multiple times).
- SET_FLAGS_VALUE,
- // update the flag's value, but *only if* it has not yet been updated
- // with SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef".
- SET_FLAG_IF_DEFAULT,
- // set the flag's default value to this. If the flag has not yet updated
- // yet (via SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef")
- // change the flag's current value to the new default value as well.
- SET_FLAGS_DEFAULT
-};
-
-// Set a particular flag ("command line option"). Returns a string
-// describing the new value that the option has been set to. The
-// return value API is not well-specified, so basically just depend on
-// it to be empty if the setting failed for some reason -- the name is
-// not a valid flag name, or the value is not a valid value -- and
-// non-empty else.
-
-// SetCommandLineOption uses set_mode == SET_FLAGS_VALUE (the common case)
-extern std::string SetCommandLineOption(const char* name, const char* value);
-extern std::string SetCommandLineOptionWithMode(const char* name, const char* value,
- FlagSettingMode set_mode);
-
-
-// --------------------------------------------------------------------
-// Saves the states (value, default value, whether the user has set
-// the flag, registered validators, etc) of all flags, and restores
-// them when the FlagSaver is destroyed. This is very useful in
-// tests, say, when you want to let your tests change the flags, but
-// make sure that they get reverted to the original states when your
-// test is complete.
-//
-// Example usage:
-// void TestFoo() {
-// FlagSaver s1;
-// FLAG_foo = false;
-// FLAG_bar = "some value";
-//
-// // test happens here. You can return at any time
-// // without worrying about restoring the FLAG values.
-// }
-//
-// Note: This class is marked with __attribute__((unused)) because all the
-// work is done in the constructor and destructor, so in the standard
-// usage example above, the compiler would complain that it's an
-// unused variable.
-//
-// This class is thread-safe.
-
-class FlagSaver {
- public:
- FlagSaver();
- ~FlagSaver();
-
- private:
- class FlagSaverImpl* impl_; // we use pimpl here to keep API steady
-
- FlagSaver(const FlagSaver&); // no copying!
- void operator=(const FlagSaver&);
-} __attribute__ ((unused));
-
-// --------------------------------------------------------------------
-// Some deprecated or hopefully-soon-to-be-deprecated functions.
-
-// This is often used for logging. TODO(csilvers): figure out a better way
-extern std::string CommandlineFlagsIntoString();
-// Usually where this is used, a FlagSaver should be used instead.
-extern bool ReadFlagsFromString(const std::string& flagfilecontents,
- const char* prog_name,
- bool errors_are_fatal); // uses SET_FLAGS_VALUE
-
-// These let you manually implement --flagfile functionality.
-// DEPRECATED.
-extern bool AppendFlagsIntoFile(const std::string& filename, const char* prog_name);
-extern bool SaveCommandFlags(); // actually defined in google.cc !
-extern bool ReadFromFlagsFile(const std::string& filename, const char* prog_name,
- bool errors_are_fatal); // uses SET_FLAGS_VALUE
-
-
-// --------------------------------------------------------------------
-// Useful routines for initializing flags from the environment.
-// In each case, if 'varname' does not exist in the environment
-// return defval. If 'varname' does exist but is not valid
-// (e.g., not a number for an int32 flag), abort with an error.
-// Otherwise, return the value. NOTE: for booleans, for true use
-// 't' or 'T' or 'true' or '1', for false 'f' or 'F' or 'false' or '0'.
-
-extern bool BoolFromEnv(const char *varname, bool defval);
-extern int32 Int32FromEnv(const char *varname, int32 defval);
-extern int64 Int64FromEnv(const char *varname, int64 defval);
-extern uint64 Uint64FromEnv(const char *varname, uint64 defval);
-extern double DoubleFromEnv(const char *varname, double defval);
-extern const char *StringFromEnv(const char *varname, const char *defval);
-
-
-// --------------------------------------------------------------------
-// The next two functions parse commandlineflags from main():
-
-// Set the "usage" message for this program. For example:
-// string usage("This program does nothing. Sample usage:\n");
-// usage += argv[0] + " <uselessarg1> <uselessarg2>";
-// SetUsageMessage(usage);
-// Do not include commandline flags in the usage: we do that for you!
-// Thread-hostile; meant to be called before any threads are spawned.
-extern void SetUsageMessage(const std::string& usage);
-
-// Looks for flags in argv and parses them. Rearranges argv to put
-// flags first, or removes them entirely if remove_flags is true.
-// If a flag is defined more than once in the command line or flag
-// file, the last definition is used. Returns the index (into argv)
-// of the first non-flag argument.
-// See top-of-file for more details on this function.
-#ifndef SWIG // In swig, use ParseCommandLineFlagsScript() instead.
-extern uint32 ParseCommandLineFlags(int *argc, char*** argv,
- bool remove_flags);
-#endif
-
-
-// Calls to ParseCommandLineNonHelpFlags and then to
-// HandleCommandLineHelpFlags can be used instead of a call to
-// ParseCommandLineFlags during initialization, in order to allow for
-// changing default values for some FLAGS (via
-// e.g. SetCommandLineOptionWithMode calls) between the time of
-// command line parsing and the time of dumping help information for
-// the flags as a result of command line parsing. If a flag is
-// defined more than once in the command line or flag file, the last
-// definition is used. Returns the index (into argv) of the first
-// non-flag argument. (If remove_flags is true, will always return 1.)
-extern uint32 ParseCommandLineNonHelpFlags(int *argc, char*** argv,
- bool remove_flags);
-// This is actually defined in commandlineflags_reporting.cc.
-// This function is misnamed (it also handles --version, etc.), but
-// it's too late to change that now. :-(
-extern void HandleCommandLineHelpFlags(); // in commandlineflags_reporting.cc
-
-// Allow command line reparsing. Disables the error normally
-// generated when an unknown flag is found, since it may be found in a
-// later parse. Thread-hostile; meant to be called before any threads
-// are spawned.
-extern void AllowCommandLineReparsing();
-
-// Reparse the flags that have not yet been recognized. Only flags
-// registered since the last parse will be recognized. Any flag value
-// must be provided as part of the argument using "=", not as a
-// separate command line argument that follows the flag argument.
-// Intended for handling flags from dynamically loaded libraries,
-// since their flags are not registered until they are loaded.
-// Returns the index (into the original argv) of the first non-flag
-// argument. (If remove_flags is true, will always return 1.)
-extern uint32 ReparseCommandLineNonHelpFlags();
-
-// Clean up memory allocated by flags. This is only needed to reduce
-// the quantity of "potentially leaked" reports emitted by memory
-// debugging tools such as valgrind. It is not required for normal
-// operation, or for the perftools heap-checker. It must only be called
-// when the process is about to exit, and all threads that might
-// access flags are quiescent. Referencing flags after this is called
-// will have unexpected consequences. This is not safe to run when
-// multiple threads might be running: the function is thread-hostile.
-extern void ShutDownCommandLineFlags();
-
-
-// --------------------------------------------------------------------
-// Now come the command line flag declaration/definition macros that
-// will actually be used. They're kind of hairy. A major reason
-// for this is initialization: we want people to be able to access
-// variables in global constructors and have that not crash, even if
-// their global constructor runs before the global constructor here.
-// (Obviously, we can't guarantee the flags will have the correct
-// default value in that case, but at least accessing them is safe.)
-// The only way to do that is have flags point to a static buffer.
-// So we make one, using a union to ensure proper alignment, and
-// then use placement-new to actually set up the flag with the
-// correct default value. In the same vein, we have to worry about
-// flag access in global destructors, so FlagRegisterer has to be
-// careful never to destroy the flag-values it constructs.
-//
-// Note that when we define a flag variable FLAGS_<name>, we also
-// preemptively define a junk variable, FLAGS_no<name>. This is to
-// cause a link-time error if someone tries to define 2 flags with
-// names like "logging" and "nologging". We do this because a bool
-// flag FLAG can be set from the command line to true with a "-FLAG"
-// argument, and to false with a "-noFLAG" argument, and so this can
-// potentially avert confusion.
-//
-// We also put flags into their own namespace. It is purposefully
-// named in an opaque way that people should have trouble typing
-// directly. The idea is that DEFINE puts the flag in the weird
-// namespace, and DECLARE imports the flag from there into the current
-// namespace. The net result is to force people to use DECLARE to get
-// access to a flag, rather than saying "extern bool FLAGS_whatever;"
-// or some such instead. We want this so we can put extra
-// functionality (like sanity-checking) in DECLARE if we want, and
-// make sure it is picked up everywhere.
-//
-// We also put the type of the variable in the namespace, so that
-// people can't DECLARE_int32 something that they DEFINE_bool'd
-// elsewhere.
-
-class FlagRegisterer {
- public:
- FlagRegisterer(const char* name, const char* type,
- const char* help, const char* filename,
- void* current_storage, void* defvalue_storage);
-};
-
-extern bool FlagsTypeWarn(const char *name);
-
-// If your application #defines STRIP_FLAG_HELP to a non-zero value
-// before #including this file, we remove the help message from the
-// binary file. This can reduce the size of the resulting binary
-// somewhat, and may also be useful for security reasons.
-
-extern const char kStrippedFlagHelp[];
-
-}
-
-#ifndef SWIG // In swig, ignore the main flag declarations
-
-#if defined(STRIP_FLAG_HELP) && STRIP_FLAG_HELP > 0
-// Need this construct to avoid the 'defined but not used' warning.
-#define MAYBE_STRIPPED_HELP(txt) (false ? (txt) : ::google::kStrippedFlagHelp)
-#else
-#define MAYBE_STRIPPED_HELP(txt) txt
-#endif
-
-// Each command-line flag has two variables associated with it: one
-// with the current value, and one with the default value. However,
-// we have a third variable, which is where value is assigned; it's a
-// constant. This guarantees that FLAG_##value is initialized at
-// static initialization time (e.g. before program-start) rather than
-// than global construction time (which is after program-start but
-// before main), at least when 'value' is a compile-time constant. We
-// use a small trick for the "default value" variable, and call it
-// FLAGS_no<name>. This serves the second purpose of assuring a
-// compile error if someone tries to define a flag named no<name>
-// which is illegal (--foo and --nofoo both affect the "foo" flag).
-#define DEFINE_VARIABLE(type, shorttype, name, value, help) \
- namespace fL##shorttype { \
- static const type FLAGS_nono##name = value; \
- type FLAGS_##name = FLAGS_nono##name; \
- type FLAGS_no##name = FLAGS_nono##name; \
- static ::google::FlagRegisterer o_##name( \
- #name, #type, MAYBE_STRIPPED_HELP(help), __FILE__, \
- &FLAGS_##name, &FLAGS_no##name); \
- } \
- using fL##shorttype::FLAGS_##name
-
-#define DECLARE_VARIABLE(type, shorttype, name) \
- namespace fL##shorttype { \
- extern type FLAGS_##name; \
- } \
- using fL##shorttype::FLAGS_##name
-
-// For DEFINE_bool, we want to do the extra check that the passed-in
-// value is actually a bool, and not a string or something that can be
-// coerced to a bool. These declarations (no definition needed!) will
-// help us do that, and never evaluate From, which is important.
-// We'll use 'sizeof(IsBool(val))' to distinguish. This code requires
-// that the compiler have different sizes for bool & double. Since
-// this is not guaranteed by the standard, we check it with a
-// compile-time assert (msg[-1] will give a compile-time error).
-namespace fLB {
-struct CompileAssert {};
-typedef CompileAssert expected_sizeof_double_neq_sizeof_bool[
- (sizeof(double) != sizeof(bool)) ? 1 : -1];
-template<typename From> double IsBoolFlag(const From& from);
-bool IsBoolFlag(bool from);
-} // namespace fLB
-
-#define DECLARE_bool(name) DECLARE_VARIABLE(bool, B, name)
-#define DEFINE_bool(name, val, txt) \
- namespace fLB { \
- typedef ::fLB::CompileAssert FLAG_##name##_value_is_not_a_bool[ \
- (sizeof(::fLB::IsBoolFlag(val)) != sizeof(double)) ? 1 : -1]; \
- } \
- DEFINE_VARIABLE(bool, B, name, val, txt)
-
-#define DECLARE_int32(name) DECLARE_VARIABLE(::google::int32, I, name)
-#define DEFINE_int32(name,val,txt) DEFINE_VARIABLE(::google::int32, I, name, val, txt)
-
-#define DECLARE_int64(name) DECLARE_VARIABLE(::google::int64, I64, name)
-#define DEFINE_int64(name,val,txt) DEFINE_VARIABLE(::google::int64, I64, name, val, txt)
-
-#define DECLARE_uint64(name) DECLARE_VARIABLE(::google::uint64, U64, name)
-#define DEFINE_uint64(name,val,txt) DEFINE_VARIABLE(::google::uint64, U64, name, val, txt)
-
-#define DECLARE_double(name) DECLARE_VARIABLE(double, D, name)
-#define DEFINE_double(name, val, txt) DEFINE_VARIABLE(double, D, name, val, txt)
-
-// Strings are trickier, because they're not a POD, so we can't
-// construct them at static-initialization time (instead they get
-// constructed at global-constructor time, which is much later). To
-// try to avoid crashes in that case, we use a char buffer to store
-// the string, which we can static-initialize, and then placement-new
-// into it later. It's not perfect, but the best we can do.
-
-namespace fLS {
-// The meaning of "string" might be different between now and when the
-// macros below get invoked (e.g., if someone is experimenting with
-// other string implementations that get defined after this file is
-// included). Save the current meaning now and use it in the macros.
-typedef std::string clstring;
-
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- const char *value) {
- return new(stringspot) clstring(value);
-}
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- const clstring &value) {
- return new(stringspot) clstring(value);
-}
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- int value);
-} // namespace fLS
-
-#define DECLARE_string(name) namespace fLS { extern ::fLS::clstring& FLAGS_##name; } \
- using fLS::FLAGS_##name
-
-// We need to define a var named FLAGS_no##name so people don't define
-// --string and --nostring. And we need a temporary place to put val
-// so we don't have to evaluate it twice. Two great needs that go
-// great together!
-// The weird 'using' + 'extern' inside the fLS namespace is to work around
-// an unknown compiler bug/issue with the gcc 4.2.1 on SUSE 10. See
-// http://code.google.com/p/google-gflags/issues/detail?id=20
-#define DEFINE_string(name, val, txt) \
- namespace fLS { \
- using ::fLS::clstring; \
- static union { void* align; char s[sizeof(clstring)]; } s_##name[2]; \
- clstring* const FLAGS_no##name = ::fLS:: \
- dont_pass0toDEFINE_string(s_##name[0].s, \
- val); \
- static ::google::FlagRegisterer o_##name( \
- #name, "string", MAYBE_STRIPPED_HELP(txt), __FILE__, \
- s_##name[0].s, new (s_##name[1].s) clstring(*FLAGS_no##name)); \
- extern clstring& FLAGS_##name; \
- using fLS::FLAGS_##name; \
- clstring& FLAGS_##name = *FLAGS_no##name; \
- } \
- using fLS::FLAGS_##name
-
-#endif // SWIG
-
-#endif // GOOGLE_GFLAGS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/mac/x64/include/gflags/gflags_completions.h b/trunk/third_party/google-gflags/gen/arch/mac/x64/include/gflags/gflags_completions.h
deleted file mode 100644
index 9d9ce7a..0000000
--- a/trunk/third_party/google-gflags/gen/arch/mac/x64/include/gflags/gflags_completions.h
+++ /dev/null
@@ -1,121 +0,0 @@
-// Copyright (c) 2008, Google Inc.
-// All rights reserved.
-//
-// Redistribution and use in source and binary forms, with or without
-// modification, are permitted provided that the following conditions are
-// met:
-//
-// * Redistributions of source code must retain the above copyright
-// notice, this list of conditions and the following disclaimer.
-// * Redistributions in binary form must reproduce the above
-// copyright notice, this list of conditions and the following disclaimer
-// in the documentation and/or other materials provided with the
-// distribution.
-// * Neither the name of Google Inc. nor the names of its
-// contributors may be used to endorse or promote products derived from
-// this software without specific prior written permission.
-//
-// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
-// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
-// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
-// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
-// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
-// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
-// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-//
-// ---
-// Author: Dave Nicponski
-//
-// Implement helpful bash-style command line flag completions
-//
-// ** Functional API:
-// HandleCommandLineCompletions() should be called early during
-// program startup, but after command line flag code has been
-// initialized, such as the beginning of HandleCommandLineHelpFlags().
-// It checks the value of the flag --tab_completion_word. If this
-// flag is empty, nothing happens here. If it contains a string,
-// however, then HandleCommandLineCompletions() will hijack the
-// process, attempting to identify the intention behind this
-// completion. Regardless of the outcome of this deduction, the
-// process will be terminated, similar to --helpshort flag
-// handling.
-//
-// ** Overview of Bash completions:
-// Bash can be told to programatically determine completions for the
-// current 'cursor word'. It does this by (in this case) invoking a
-// command with some additional arguments identifying the command
-// being executed, the word being completed, and the previous word
-// (if any). Bash then expects a sequence of output lines to be
-// printed to stdout. If these lines all contain a common prefix
-// longer than the cursor word, bash will replace the cursor word
-// with that common prefix, and display nothing. If there isn't such
-// a common prefix, bash will display the lines in pages using 'more'.
-//
-// ** Strategy taken for command line completions:
-// If we can deduce either the exact flag intended, or a common flag
-// prefix, we'll output exactly that. Otherwise, if information
-// must be displayed to the user, we'll take the opportunity to add
-// some helpful information beyond just the flag name (specifically,
-// we'll include the default flag value and as much of the flag's
-// description as can fit on a single terminal line width, as specified
-// by the flag --tab_completion_columns). Furthermore, we'll try to
-// make bash order the output such that the most useful or relevent
-// flags are the most likely to be shown at the top.
-//
-// ** Additional features:
-// To assist in finding that one really useful flag, substring matching
-// was implemented. Before pressing a <TAB> to get completion for the
-// current word, you can append one or more '?' to the flag to do
-// substring matching. Here's the semantics:
-// --foo<TAB> Show me all flags with names prefixed by 'foo'
-// --foo?<TAB> Show me all flags with 'foo' somewhere in the name
-// --foo??<TAB> Same as prior case, but also search in module
-// definition path for 'foo'
-// --foo???<TAB> Same as prior case, but also search in flag
-// descriptions for 'foo'
-// Finally, we'll trim the output to a relatively small number of
-// flags to keep bash quiet about the verbosity of output. If one
-// really wanted to see all possible matches, appending a '+' to the
-// search word will force the exhaustive list of matches to be printed.
-//
-// ** How to have bash accept completions from a binary:
-// Bash requires that it be informed about each command that programmatic
-// completion should be enabled for. Example addition to a .bashrc
-// file would be (your path to gflags_completions.sh file may differ):
-
-/*
-$ complete -o bashdefault -o default -o nospace -C \
- '/usr/local/bin/gflags_completions.sh --tab_completion_columns $COLUMNS' \
- time env binary_name another_binary [...]
-*/
-
-// This would allow the following to work:
-// $ /path/to/binary_name --vmodule<TAB>
-// Or:
-// $ ./bin/path/another_binary --gfs_u<TAB>
-// (etc)
-//
-// Sadly, it appears that bash gives no easy way to force this behavior for
-// all commands. That's where the "time" in the above example comes in.
-// If you haven't specifically added a command to the list of completion
-// supported commands, you can still get completions by prefixing the
-// entire command with "env".
-// $ env /some/brand/new/binary --vmod<TAB>
-// Assuming that "binary" is a newly compiled binary, this should still
-// produce the expected completion output.
-
-
-#ifndef GOOGLE_GFLAGS_COMPLETIONS_H_
-#define GOOGLE_GFLAGS_COMPLETIONS_H_
-
-namespace google {
-
-void HandleCommandLineCompletions(void);
-
-}
-
-#endif // GOOGLE_GFLAGS_COMPLETIONS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/mac/x64/include/private/config.h b/trunk/third_party/google-gflags/gen/arch/mac/x64/include/private/config.h
deleted file mode 100644
index 98d8e1a..0000000
--- a/trunk/third_party/google-gflags/gen/arch/mac/x64/include/private/config.h
+++ /dev/null
@@ -1,110 +0,0 @@
-/* src/config.h. Generated from config.h.in by configure. */
-/* src/config.h.in. Generated from configure.ac by autoheader. */
-
-/* Always the empty-string on non-windows systems. On windows, should be
- "__declspec(dllexport)". This way, when we compile the dll, we export our
- functions/classes. It's safe to define this here because config.h is only
- used internally, to compile the DLL, and every DLL source file #includes
- "config.h" before anything else. */
-#define GFLAGS_DLL_DECL /**/
-
-/* Namespace for Google classes */
-#define GOOGLE_NAMESPACE ::google
-
-/* Define to 1 if you have the <dlfcn.h> header file. */
-#define HAVE_DLFCN_H 1
-
-/* Define to 1 if you have the <fnmatch.h> header file. */
-#define HAVE_FNMATCH_H 1
-
-/* Define to 1 if you have the <inttypes.h> header file. */
-#define HAVE_INTTYPES_H 1
-
-/* Define to 1 if you have the <memory.h> header file. */
-#define HAVE_MEMORY_H 1
-
-/* define if the compiler implements namespaces */
-#define HAVE_NAMESPACES 1
-
-/* Define if you have POSIX threads libraries and header files. */
-#define HAVE_PTHREAD 1
-
-/* Define to 1 if you have the `putenv' function. */
-#define HAVE_PUTENV 1
-
-/* Define to 1 if you have the `setenv' function. */
-#define HAVE_SETENV 1
-
-/* Define to 1 if you have the <stdint.h> header file. */
-#define HAVE_STDINT_H 1
-
-/* Define to 1 if you have the <stdlib.h> header file. */
-#define HAVE_STDLIB_H 1
-
-/* Define to 1 if you have the <strings.h> header file. */
-#define HAVE_STRINGS_H 1
-
-/* Define to 1 if you have the <string.h> header file. */
-#define HAVE_STRING_H 1
-
-/* Define to 1 if you have the `strtoll' function. */
-#define HAVE_STRTOLL 1
-
-/* Define to 1 if you have the `strtoq' function. */
-#define HAVE_STRTOQ 1
-
-/* Define to 1 if you have the <sys/stat.h> header file. */
-#define HAVE_SYS_STAT_H 1
-
-/* Define to 1 if you have the <sys/types.h> header file. */
-#define HAVE_SYS_TYPES_H 1
-
-/* Define to 1 if you have the <unistd.h> header file. */
-#define HAVE_UNISTD_H 1
-
-/* define if your compiler has __attribute__ */
-#define HAVE___ATTRIBUTE__ 1
-
-/* Define to the sub-directory in which libtool stores uninstalled libraries.
- */
-#define LT_OBJDIR ".libs/"
-
-/* Name of package */
-#define PACKAGE "gflags"
-
-/* Define to the address where bug reports for this package should be sent. */
-#define PACKAGE_BUGREPORT "opensource@google.com"
-
-/* Define to the full name of this package. */
-#define PACKAGE_NAME "gflags"
-
-/* Define to the full name and version of this package. */
-#define PACKAGE_STRING "gflags 1.5"
-
-/* Define to the one symbol short name of this package. */
-#define PACKAGE_TARNAME "gflags"
-
-/* Define to the home page for this package. */
-#define PACKAGE_URL ""
-
-/* Define to the version of this package. */
-#define PACKAGE_VERSION "1.5"
-
-/* Define to necessary symbol if this constant uses a non-standard name on
- your system. */
-/* #undef PTHREAD_CREATE_JOINABLE */
-
-/* Define to 1 if you have the ANSI C header files. */
-#define STDC_HEADERS 1
-
-/* the namespace where STL code like vector<> is defined */
-#define STL_NAMESPACE std
-
-/* Version number of package */
-#define VERSION "1.5"
-
-/* Stops putting the code inside the Google namespace */
-#define _END_GOOGLE_NAMESPACE_ }
-
-/* Puts following code inside the Google namespace */
-#define _START_GOOGLE_NAMESPACE_ namespace google {
diff --git a/trunk/third_party/google-gflags/gen/arch/win/ia32/include/gflags/gflags.h b/trunk/third_party/google-gflags/gen/arch/win/ia32/include/gflags/gflags.h
deleted file mode 100644
index fdafe2a..0000000
--- a/trunk/third_party/google-gflags/gen/arch/win/ia32/include/gflags/gflags.h
+++ /dev/null
@@ -1,601 +0,0 @@
-// Copyright (c) 2006, Google Inc.
-// All rights reserved.
-//
-// Redistribution and use in source and binary forms, with or without
-// modification, are permitted provided that the following conditions are
-// met:
-//
-// * Redistributions of source code must retain the above copyright
-// notice, this list of conditions and the following disclaimer.
-// * Redistributions in binary form must reproduce the above
-// copyright notice, this list of conditions and the following disclaimer
-// in the documentation and/or other materials provided with the
-// distribution.
-// * Neither the name of Google Inc. nor the names of its
-// contributors may be used to endorse or promote products derived from
-// this software without specific prior written permission.
-//
-// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
-// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
-// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
-// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
-// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
-// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
-// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-
-// ---
-// Author: Ray Sidney
-// Revamped and reorganized by Craig Silverstein
-//
-// This is the file that should be included by any file which declares
-// or defines a command line flag or wants to parse command line flags
-// or print a program usage message (which will include information about
-// flags). Executive summary, in the form of an example foo.cc file:
-//
-// #include "foo.h" // foo.h has a line "DECLARE_int32(start);"
-// #include "validators.h" // hypothetical file defining ValidateIsFile()
-//
-// DEFINE_int32(end, 1000, "The last record to read");
-//
-// DEFINE_string(filename, "my_file.txt", "The file to read");
-// // Crash if the specified file does not exist.
-// static bool dummy = RegisterFlagValidator(&FLAGS_filename,
-// &ValidateIsFile);
-//
-// DECLARE_bool(verbose); // some other file has a DEFINE_bool(verbose, ...)
-//
-// void MyFunc() {
-// if (FLAGS_verbose) printf("Records %d-%d\n", FLAGS_start, FLAGS_end);
-// }
-//
-// Then, at the command-line:
-// ./foo --noverbose --start=5 --end=100
-//
-// For more details, see
-// doc/gflags.html
-//
-// --- A note about thread-safety:
-//
-// We describe many functions in this routine as being thread-hostile,
-// thread-compatible, or thread-safe. Here are the meanings we use:
-//
-// thread-safe: it is safe for multiple threads to call this routine
-// (or, when referring to a class, methods of this class)
-// concurrently.
-// thread-hostile: it is not safe for multiple threads to call this
-// routine (or methods of this class) concurrently. In gflags,
-// most thread-hostile routines are intended to be called early in,
-// or even before, main() -- that is, before threads are spawned.
-// thread-compatible: it is safe for multiple threads to read from
-// this variable (when applied to variables), or to call const
-// methods of this class (when applied to classes), as long as no
-// other thread is writing to the variable or calling non-const
-// methods of this class.
-
-#ifndef GOOGLE_GFLAGS_H_
-#define GOOGLE_GFLAGS_H_
-
-#include <string>
-#include <vector>
-
-// We care a lot about number of bits things take up. Unfortunately,
-// systems define their bit-specific ints in a lot of different ways.
-// We use our own way, and have a typedef to get there.
-// Note: these commands below may look like "#if 1" or "#if 0", but
-// that's because they were constructed that way at ./configure time.
-// Look at gflags.h.in to see how they're calculated (based on your config).
-#if 0
-#include <stdint.h> // the normal place uint16_t is defined
-#endif
-#if 1
-#include <sys/types.h> // the normal place u_int16_t is defined
-#endif
-#if 0
-#include <inttypes.h> // a third place for uint16_t or u_int16_t
-#endif
-
-// Annoying stuff for windows -- makes sure clients can import these functions
-#if defined(_WIN32)
-# ifndef GFLAGS_DLL_DECL
-# define GFLAGS_DLL_DECL __declspec(dllimport)
-# endif
-# ifndef GFLAGS_DLL_DECLARE_FLAG
-# define GFLAGS_DLL_DECLARE_FLAG __declspec(dllimport)
-# endif
-# ifndef GFLAGS_DLL_DEFINE_FLAG
-# define GFLAGS_DLL_DEFINE_FLAG __declspec(dllexport)
-# endif
-#else
-# ifndef GFLAGS_DLL_DECL
-# define GFLAGS_DLL_DECL
-# endif
-# ifndef GFLAGS_DLL_DECLARE_FLAG
-# define GFLAGS_DLL_DECLARE_FLAG
-# endif
-# ifndef GFLAGS_DLL_DEFINE_FLAG
-# define GFLAGS_DLL_DEFINE_FLAG
-# endif
-#endif
-
-namespace google {
-
-#if 0 // the C99 format
-typedef int32_t int32;
-typedef uint32_t uint32;
-typedef int64_t int64;
-typedef uint64_t uint64;
-#elif 0 // the BSD format
-typedef int32_t int32;
-typedef u_int32_t uint32;
-typedef int64_t int64;
-typedef u_int64_t uint64;
-#elif 1 // the windows (vc7) format
-typedef __int32 int32;
-typedef unsigned __int32 uint32;
-typedef __int64 int64;
-typedef unsigned __int64 uint64;
-#else
-#error Do not know how to define a 32-bit integer quantity on your system
-#endif
-
-// --------------------------------------------------------------------
-// To actually define a flag in a file, use DEFINE_bool,
-// DEFINE_string, etc. at the bottom of this file. You may also find
-// it useful to register a validator with the flag. This ensures that
-// when the flag is parsed from the commandline, or is later set via
-// SetCommandLineOption, we call the validation function. It is _not_
-// called when you assign the value to the flag directly using the = operator.
-//
-// The validation function should return true if the flag value is valid, and
-// false otherwise. If the function returns false for the new setting of the
-// flag, the flag will retain its current value. If it returns false for the
-// default value, ParseCommandLineFlags() will die.
-//
-// This function is safe to call at global construct time (as in the
-// example below).
-//
-// Example use:
-// static bool ValidatePort(const char* flagname, int32 value) {
-// if (value > 0 && value < 32768) // value is ok
-// return true;
-// printf("Invalid value for --%s: %d\n", flagname, (int)value);
-// return false;
-// }
-// DEFINE_int32(port, 0, "What port to listen on");
-// static bool dummy = RegisterFlagValidator(&FLAGS_port, &ValidatePort);
-
-// Returns true if successfully registered, false if not (because the
-// first argument doesn't point to a command-line flag, or because a
-// validator is already registered for this flag).
-GFLAGS_DLL_DECL bool RegisterFlagValidator(const bool* flag,
- bool (*validate_fn)(const char*, bool));
-GFLAGS_DLL_DECL bool RegisterFlagValidator(const int32* flag,
- bool (*validate_fn)(const char*, int32));
-GFLAGS_DLL_DECL bool RegisterFlagValidator(const int64* flag,
- bool (*validate_fn)(const char*, int64));
-GFLAGS_DLL_DECL bool RegisterFlagValidator(const uint64* flag,
- bool (*validate_fn)(const char*, uint64));
-GFLAGS_DLL_DECL bool RegisterFlagValidator(const double* flag,
- bool (*validate_fn)(const char*, double));
-GFLAGS_DLL_DECL bool RegisterFlagValidator(const std::string* flag,
- bool (*validate_fn)(const char*, const std::string&));
-
-
-// --------------------------------------------------------------------
-// These methods are the best way to get access to info about the
-// list of commandline flags. Note that these routines are pretty slow.
-// GetAllFlags: mostly-complete info about the list, sorted by file.
-// ShowUsageWithFlags: pretty-prints the list to stdout (what --help does)
-// ShowUsageWithFlagsRestrict: limit to filenames with restrict as a substr
-//
-// In addition to accessing flags, you can also access argv[0] (the program
-// name) and argv (the entire commandline), which we sock away a copy of.
-// These variables are static, so you should only set them once.
-
-struct GFLAGS_DLL_DECL CommandLineFlagInfo {
- std::string name; // the name of the flag
- std::string type; // the type of the flag: int32, etc
- std::string description; // the "help text" associated with the flag
- std::string current_value; // the current value, as a string
- std::string default_value; // the default value, as a string
- std::string filename; // 'cleaned' version of filename holding the flag
- bool has_validator_fn; // true if RegisterFlagValidator called on flag
- bool is_default; // true if the flag has the default value and
- // has not been set explicitly from the cmdline
- // or via SetCommandLineOption
-};
-
-// Using this inside of a validator is a recipe for a deadlock.
-// TODO(wojtekm) Fix locking when validators are running, to make it safe to
-// call validators during ParseAllFlags.
-// Also make sure then to uncomment the corresponding unit test in
-// commandlineflags_unittest.sh
-extern GFLAGS_DLL_DECL void GetAllFlags(std::vector<CommandLineFlagInfo>* OUTPUT);
-// These two are actually defined in commandlineflags_reporting.cc.
-extern GFLAGS_DLL_DECL void ShowUsageWithFlags(const char *argv0); // what --help does
-extern GFLAGS_DLL_DECL void ShowUsageWithFlagsRestrict(const char *argv0, const char *restrict);
-
-// Create a descriptive string for a flag.
-// Goes to some trouble to make pretty line breaks.
-extern GFLAGS_DLL_DECL std::string DescribeOneFlag(const CommandLineFlagInfo& flag);
-
-// Thread-hostile; meant to be called before any threads are spawned.
-extern GFLAGS_DLL_DECL void SetArgv(int argc, const char** argv);
-// The following functions are thread-safe as long as SetArgv() is
-// only called before any threads start.
-extern GFLAGS_DLL_DECL const std::vector<std::string>& GetArgvs(); // all of argv as a vector
-extern GFLAGS_DLL_DECL const char* GetArgv(); // all of argv as a string
-extern GFLAGS_DLL_DECL const char* GetArgv0(); // only argv0
-extern GFLAGS_DLL_DECL uint32 GetArgvSum(); // simple checksum of argv
-extern GFLAGS_DLL_DECL const char* ProgramInvocationName(); // argv0, or "UNKNOWN" if not set
-extern GFLAGS_DLL_DECL const char* ProgramInvocationShortName(); // basename(argv0)
-// ProgramUsage() is thread-safe as long as SetUsageMessage() is only
-// called before any threads start.
-extern GFLAGS_DLL_DECL const char* ProgramUsage(); // string set by SetUsageMessage()
-
-
-// --------------------------------------------------------------------
-// Normally you access commandline flags by just saying "if (FLAGS_foo)"
-// or whatever, and set them by calling "FLAGS_foo = bar" (or, more
-// commonly, via the DEFINE_foo macro). But if you need a bit more
-// control, we have programmatic ways to get/set the flags as well.
-// These programmatic ways to access flags are thread-safe, but direct
-// access is only thread-compatible.
-
-// Return true iff the flagname was found.
-// OUTPUT is set to the flag's value, or unchanged if we return false.
-extern GFLAGS_DLL_DECL bool GetCommandLineOption(const char* name, std::string* OUTPUT);
-
-// Return true iff the flagname was found. OUTPUT is set to the flag's
-// CommandLineFlagInfo or unchanged if we return false.
-extern GFLAGS_DLL_DECL bool GetCommandLineFlagInfo(const char* name,
- CommandLineFlagInfo* OUTPUT);
-
-// Return the CommandLineFlagInfo of the flagname. exit() if name not found.
-// Example usage, to check if a flag's value is currently the default value:
-// if (GetCommandLineFlagInfoOrDie("foo").is_default) ...
-extern GFLAGS_DLL_DECL CommandLineFlagInfo GetCommandLineFlagInfoOrDie(const char* name);
-
-enum GFLAGS_DLL_DECL FlagSettingMode {
- // update the flag's value (can call this multiple times).
- SET_FLAGS_VALUE,
- // update the flag's value, but *only if* it has not yet been updated
- // with SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef".
- SET_FLAG_IF_DEFAULT,
- // set the flag's default value to this. If the flag has not yet updated
- // yet (via SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef")
- // change the flag's current value to the new default value as well.
- SET_FLAGS_DEFAULT
-};
-
-// Set a particular flag ("command line option"). Returns a string
-// describing the new value that the option has been set to. The
-// return value API is not well-specified, so basically just depend on
-// it to be empty if the setting failed for some reason -- the name is
-// not a valid flag name, or the value is not a valid value -- and
-// non-empty else.
-
-// SetCommandLineOption uses set_mode == SET_FLAGS_VALUE (the common case)
-extern GFLAGS_DLL_DECL std::string SetCommandLineOption(const char* name, const char* value);
-extern GFLAGS_DLL_DECL std::string SetCommandLineOptionWithMode(const char* name, const char* value,
- FlagSettingMode set_mode);
-
-
-// --------------------------------------------------------------------
-// Saves the states (value, default value, whether the user has set
-// the flag, registered validators, etc) of all flags, and restores
-// them when the FlagSaver is destroyed. This is very useful in
-// tests, say, when you want to let your tests change the flags, but
-// make sure that they get reverted to the original states when your
-// test is complete.
-//
-// Example usage:
-// void TestFoo() {
-// FlagSaver s1;
-// FLAG_foo = false;
-// FLAG_bar = "some value";
-//
-// // test happens here. You can return at any time
-// // without worrying about restoring the FLAG values.
-// }
-//
-// Note: This class is marked with __attribute__((unused)) because all the
-// work is done in the constructor and destructor, so in the standard
-// usage example above, the compiler would complain that it's an
-// unused variable.
-//
-// This class is thread-safe.
-
-class GFLAGS_DLL_DECL FlagSaver {
- public:
- FlagSaver();
- ~FlagSaver();
-
- private:
- class FlagSaverImpl* impl_; // we use pimpl here to keep API steady
-
- FlagSaver(const FlagSaver&); // no copying!
- void operator=(const FlagSaver&);
-} ;
-
-// --------------------------------------------------------------------
-// Some deprecated or hopefully-soon-to-be-deprecated functions.
-
-// This is often used for logging. TODO(csilvers): figure out a better way
-extern GFLAGS_DLL_DECL std::string CommandlineFlagsIntoString();
-// Usually where this is used, a FlagSaver should be used instead.
-extern GFLAGS_DLL_DECL bool ReadFlagsFromString(const std::string& flagfilecontents,
- const char* prog_name,
- bool errors_are_fatal); // uses SET_FLAGS_VALUE
-
-// These let you manually implement --flagfile functionality.
-// DEPRECATED.
-extern GFLAGS_DLL_DECL bool AppendFlagsIntoFile(const std::string& filename, const char* prog_name);
-extern GFLAGS_DLL_DECL bool SaveCommandFlags(); // actually defined in google.cc !
-extern GFLAGS_DLL_DECL bool ReadFromFlagsFile(const std::string& filename, const char* prog_name,
- bool errors_are_fatal); // uses SET_FLAGS_VALUE
-
-
-// --------------------------------------------------------------------
-// Useful routines for initializing flags from the environment.
-// In each case, if 'varname' does not exist in the environment
-// return defval. If 'varname' does exist but is not valid
-// (e.g., not a number for an int32 flag), abort with an error.
-// Otherwise, return the value. NOTE: for booleans, for true use
-// 't' or 'T' or 'true' or '1', for false 'f' or 'F' or 'false' or '0'.
-
-extern GFLAGS_DLL_DECL bool BoolFromEnv(const char *varname, bool defval);
-extern GFLAGS_DLL_DECL int32 Int32FromEnv(const char *varname, int32 defval);
-extern GFLAGS_DLL_DECL int64 Int64FromEnv(const char *varname, int64 defval);
-extern GFLAGS_DLL_DECL uint64 Uint64FromEnv(const char *varname, uint64 defval);
-extern GFLAGS_DLL_DECL double DoubleFromEnv(const char *varname, double defval);
-extern GFLAGS_DLL_DECL const char *StringFromEnv(const char *varname, const char *defval);
-
-
-// --------------------------------------------------------------------
-// The next two functions parse commandlineflags from main():
-
-// Set the "usage" message for this program. For example:
-// string usage("This program does nothing. Sample usage:\n");
-// usage += argv[0] + " <uselessarg1> <uselessarg2>";
-// SetUsageMessage(usage);
-// Do not include commandline flags in the usage: we do that for you!
-// Thread-hostile; meant to be called before any threads are spawned.
-extern GFLAGS_DLL_DECL void SetUsageMessage(const std::string& usage);
-
-// Looks for flags in argv and parses them. Rearranges argv to put
-// flags first, or removes them entirely if remove_flags is true.
-// If a flag is defined more than once in the command line or flag
-// file, the last definition is used.
-// See top-of-file for more details on this function.
-#ifndef SWIG // In swig, use ParseCommandLineFlagsScript() instead.
-extern GFLAGS_DLL_DECL uint32 ParseCommandLineFlags(int *argc, char*** argv,
- bool remove_flags);
-#endif
-
-
-// Calls to ParseCommandLineNonHelpFlags and then to
-// HandleCommandLineHelpFlags can be used instead of a call to
-// ParseCommandLineFlags during initialization, in order to allow for
-// changing default values for some FLAGS (via
-// e.g. SetCommandLineOptionWithMode calls) between the time of
-// command line parsing and the time of dumping help information for
-// the flags as a result of command line parsing.
-// If a flag is defined more than once in the command line or flag
-// file, the last definition is used.
-extern GFLAGS_DLL_DECL uint32 ParseCommandLineNonHelpFlags(int *argc, char*** argv,
- bool remove_flags);
-// This is actually defined in commandlineflags_reporting.cc.
-// This function is misnamed (it also handles --version, etc.), but
-// it's too late to change that now. :-(
-extern GFLAGS_DLL_DECL void HandleCommandLineHelpFlags(); // in commandlineflags_reporting.cc
-
-// Allow command line reparsing. Disables the error normally
-// generated when an unknown flag is found, since it may be found in a
-// later parse. Thread-hostile; meant to be called before any threads
-// are spawned.
-extern GFLAGS_DLL_DECL void AllowCommandLineReparsing();
-
-// Reparse the flags that have not yet been recognized.
-// Only flags registered since the last parse will be recognized.
-// Any flag value must be provided as part of the argument using "=",
-// not as a separate command line argument that follows the flag argument.
-// Intended for handling flags from dynamically loaded libraries,
-// since their flags are not registered until they are loaded.
-extern GFLAGS_DLL_DECL uint32 ReparseCommandLineNonHelpFlags();
-
-// Clean up memory allocated by flags. This is only needed to reduce
-// the quantity of "potentially leaked" reports emitted by memory
-// debugging tools such as valgrind. It is not required for normal
-// operation, or for the perftools heap-checker. It must only be called
-// when the process is about to exit, and all threads that might
-// access flags are quiescent. Referencing flags after this is called
-// will have unexpected consequences. This is not safe to run when
-// multiple threads might be running: the function is thread-hostile.
-extern GFLAGS_DLL_DECL void ShutDownCommandLineFlags();
-
-
-// --------------------------------------------------------------------
-// Now come the command line flag declaration/definition macros that
-// will actually be used. They're kind of hairy. A major reason
-// for this is initialization: we want people to be able to access
-// variables in global constructors and have that not crash, even if
-// their global constructor runs before the global constructor here.
-// (Obviously, we can't guarantee the flags will have the correct
-// default value in that case, but at least accessing them is safe.)
-// The only way to do that is have flags point to a static buffer.
-// So we make one, using a union to ensure proper alignment, and
-// then use placement-new to actually set up the flag with the
-// correct default value. In the same vein, we have to worry about
-// flag access in global destructors, so FlagRegisterer has to be
-// careful never to destroy the flag-values it constructs.
-//
-// Note that when we define a flag variable FLAGS_<name>, we also
-// preemptively define a junk variable, FLAGS_no<name>. This is to
-// cause a link-time error if someone tries to define 2 flags with
-// names like "logging" and "nologging". We do this because a bool
-// flag FLAG can be set from the command line to true with a "-FLAG"
-// argument, and to false with a "-noFLAG" argument, and so this can
-// potentially avert confusion.
-//
-// We also put flags into their own namespace. It is purposefully
-// named in an opaque way that people should have trouble typing
-// directly. The idea is that DEFINE puts the flag in the weird
-// namespace, and DECLARE imports the flag from there into the current
-// namespace. The net result is to force people to use DECLARE to get
-// access to a flag, rather than saying "extern bool FLAGS_whatever;"
-// or some such instead. We want this so we can put extra
-// functionality (like sanity-checking) in DECLARE if we want, and
-// make sure it is picked up everywhere.
-//
-// We also put the type of the variable in the namespace, so that
-// people can't DECLARE_int32 something that they DEFINE_bool'd
-// elsewhere.
-
-class GFLAGS_DLL_DECL FlagRegisterer {
- public:
- FlagRegisterer(const char* name, const char* type,
- const char* help, const char* filename,
- void* current_storage, void* defvalue_storage);
-};
-
-extern bool FlagsTypeWarn(const char *name);
-
-// If your application #defines STRIP_FLAG_HELP to a non-zero value
-// before #including this file, we remove the help message from the
-// binary file. This can reduce the size of the resulting binary
-// somewhat, and may also be useful for security reasons.
-
-extern const char kStrippedFlagHelp[];
-
-}
-
-#ifndef SWIG // In swig, ignore the main flag declarations
-
-#if defined(STRIP_FLAG_HELP) && STRIP_FLAG_HELP > 0
-// Need this construct to avoid the 'defined but not used' warning.
-#define MAYBE_STRIPPED_HELP(txt) (false ? (txt) : kStrippedFlagHelp)
-#else
-#define MAYBE_STRIPPED_HELP(txt) txt
-#endif
-
-// Each command-line flag has two variables associated with it: one
-// with the current value, and one with the default value. However,
-// we have a third variable, which is where value is assigned; it's a
-// constant. This guarantees that FLAG_##value is initialized at
-// static initialization time (e.g. before program-start) rather than
-// than global construction time (which is after program-start but
-// before main), at least when 'value' is a compile-time constant. We
-// use a small trick for the "default value" variable, and call it
-// FLAGS_no<name>. This serves the second purpose of assuring a
-// compile error if someone tries to define a flag named no<name>
-// which is illegal (--foo and --nofoo both affect the "foo" flag).
-#define DEFINE_VARIABLE(type, shorttype, name, value, help) \
- namespace fL##shorttype { \
- static const type FLAGS_nono##name = value; \
- /* We always want to export defined variables, dll or no */ \
- GFLAGS_DLL_DEFINE_FLAG type FLAGS_##name = FLAGS_nono##name; \
- type FLAGS_no##name = FLAGS_nono##name; \
- static ::google::FlagRegisterer o_##name( \
- #name, #type, MAYBE_STRIPPED_HELP(help), __FILE__, \
- &FLAGS_##name, &FLAGS_no##name); \
- } \
- using fL##shorttype::FLAGS_##name
-
-#define DECLARE_VARIABLE(type, shorttype, name) \
- namespace fL##shorttype { \
- /* We always want to import declared variables, dll or no */ \
- extern GFLAGS_DLL_DECLARE_FLAG type FLAGS_##name; \
- } \
- using fL##shorttype::FLAGS_##name
-
-// For DEFINE_bool, we want to do the extra check that the passed-in
-// value is actually a bool, and not a string or something that can be
-// coerced to a bool. These declarations (no definition needed!) will
-// help us do that, and never evaluate From, which is important.
-// We'll use 'sizeof(IsBool(val))' to distinguish. This code requires
-// that the compiler have different sizes for bool & double. Since
-// this is not guaranteed by the standard, we check it with a
-// compile-time assert (msg[-1] will give a compile-time error).
-namespace fLB {
-struct CompileAssert {};
-typedef CompileAssert expected_sizeof_double_neq_sizeof_bool[
- (sizeof(double) != sizeof(bool)) ? 1 : -1];
-template<typename From> GFLAGS_DLL_DECL double IsBoolFlag(const From& from);
-GFLAGS_DLL_DECL bool IsBoolFlag(bool from);
-} // namespace fLB
-
-#define DECLARE_bool(name) DECLARE_VARIABLE(bool, B, name)
-#define DEFINE_bool(name, val, txt) \
- namespace fLB { \
- typedef ::fLB::CompileAssert FLAG_##name##_value_is_not_a_bool[ \
- (sizeof(::fLB::IsBoolFlag(val)) != sizeof(double)) ? 1 : -1]; \
- } \
- DEFINE_VARIABLE(bool, B, name, val, txt)
-
-#define DECLARE_int32(name) DECLARE_VARIABLE(::google::int32, I, name)
-#define DEFINE_int32(name,val,txt) DEFINE_VARIABLE(::google::int32, I, name, val, txt)
-
-#define DECLARE_int64(name) DECLARE_VARIABLE(::google::int64, I64, name)
-#define DEFINE_int64(name,val,txt) DEFINE_VARIABLE(::google::int64, I64, name, val, txt)
-
-#define DECLARE_uint64(name) DECLARE_VARIABLE(::google::uint64, U64, name)
-#define DEFINE_uint64(name,val,txt) DEFINE_VARIABLE(::google::uint64, U64, name, val, txt)
-
-#define DECLARE_double(name) DECLARE_VARIABLE(double, D, name)
-#define DEFINE_double(name, val, txt) DEFINE_VARIABLE(double, D, name, val, txt)
-
-// Strings are trickier, because they're not a POD, so we can't
-// construct them at static-initialization time (instead they get
-// constructed at global-constructor time, which is much later). To
-// try to avoid crashes in that case, we use a char buffer to store
-// the string, which we can static-initialize, and then placement-new
-// into it later. It's not perfect, but the best we can do.
-
-namespace fLS {
-// The meaning of "string" might be different between now and when the
-// macros below get invoked (e.g., if someone is experimenting with
-// other string implementations that get defined after this file is
-// included). Save the current meaning now and use it in the macros.
-typedef std::string clstring;
-
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- const char *value) {
- return new(stringspot) clstring(value);
-}
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- const clstring &value) {
- return new(stringspot) clstring(value);
-}
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- int value);
-} // namespace fLS
-
-#define DECLARE_string(name) namespace fLS { extern GFLAGS_DLL_DECLARE_FLAG ::fLS::clstring& FLAGS_##name; } \
- using fLS::FLAGS_##name
-
-// We need to define a var named FLAGS_no##name so people don't define
-// --string and --nostring. And we need a temporary place to put val
-// so we don't have to evaluate it twice. Two great needs that go
-// great together!
-#define DEFINE_string(name, val, txt) \
- namespace fLS { \
- using ::fLS::clstring; \
- static union { void* align; char s[sizeof(clstring)]; } s_##name[2]; \
- clstring* const FLAGS_no##name = ::fLS:: \
- dont_pass0toDEFINE_string(s_##name[0].s, \
- val); \
- static ::google::FlagRegisterer o_##name( \
- #name, "string", MAYBE_STRIPPED_HELP(txt), __FILE__, \
- s_##name[0].s, new (s_##name[1].s) clstring(*FLAGS_no##name)); \
- GFLAGS_DLL_DEFINE_FLAG clstring& FLAGS_##name = *FLAGS_no##name; \
- } \
- using fLS::FLAGS_##name
-
-#endif // SWIG
-
-#endif // GOOGLE_GFLAGS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/win/ia32/include/gflags/gflags_completions.h b/trunk/third_party/google-gflags/gen/arch/win/ia32/include/gflags/gflags_completions.h
deleted file mode 100644
index e97de5b..0000000
--- a/trunk/third_party/google-gflags/gen/arch/win/ia32/include/gflags/gflags_completions.h
+++ /dev/null
@@ -1,130 +0,0 @@
-// Copyright (c) 2008, Google Inc.
-// All rights reserved.
-//
-// Redistribution and use in source and binary forms, with or without
-// modification, are permitted provided that the following conditions are
-// met:
-//
-// * Redistributions of source code must retain the above copyright
-// notice, this list of conditions and the following disclaimer.
-// * Redistributions in binary form must reproduce the above
-// copyright notice, this list of conditions and the following disclaimer
-// in the documentation and/or other materials provided with the
-// distribution.
-// * Neither the name of Google Inc. nor the names of its
-// contributors may be used to endorse or promote products derived from
-// this software without specific prior written permission.
-//
-// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
-// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
-// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
-// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
-// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
-// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
-// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-//
-// ---
-// Author: Dave Nicponski
-//
-// Implement helpful bash-style command line flag completions
-//
-// ** Functional API:
-// HandleCommandLineCompletions() should be called early during
-// program startup, but after command line flag code has been
-// initialized, such as the beginning of HandleCommandLineHelpFlags().
-// It checks the value of the flag --tab_completion_word. If this
-// flag is empty, nothing happens here. If it contains a string,
-// however, then HandleCommandLineCompletions() will hijack the
-// process, attempting to identify the intention behind this
-// completion. Regardless of the outcome of this deduction, the
-// process will be terminated, similar to --helpshort flag
-// handling.
-//
-// ** Overview of Bash completions:
-// Bash can be told to programatically determine completions for the
-// current 'cursor word'. It does this by (in this case) invoking a
-// command with some additional arguments identifying the command
-// being executed, the word being completed, and the previous word
-// (if any). Bash then expects a sequence of output lines to be
-// printed to stdout. If these lines all contain a common prefix
-// longer than the cursor word, bash will replace the cursor word
-// with that common prefix, and display nothing. If there isn't such
-// a common prefix, bash will display the lines in pages using 'more'.
-//
-// ** Strategy taken for command line completions:
-// If we can deduce either the exact flag intended, or a common flag
-// prefix, we'll output exactly that. Otherwise, if information
-// must be displayed to the user, we'll take the opportunity to add
-// some helpful information beyond just the flag name (specifically,
-// we'll include the default flag value and as much of the flag's
-// description as can fit on a single terminal line width, as specified
-// by the flag --tab_completion_columns). Furthermore, we'll try to
-// make bash order the output such that the most useful or relevent
-// flags are the most likely to be shown at the top.
-//
-// ** Additional features:
-// To assist in finding that one really useful flag, substring matching
-// was implemented. Before pressing a <TAB> to get completion for the
-// current word, you can append one or more '?' to the flag to do
-// substring matching. Here's the semantics:
-// --foo<TAB> Show me all flags with names prefixed by 'foo'
-// --foo?<TAB> Show me all flags with 'foo' somewhere in the name
-// --foo??<TAB> Same as prior case, but also search in module
-// definition path for 'foo'
-// --foo???<TAB> Same as prior case, but also search in flag
-// descriptions for 'foo'
-// Finally, we'll trim the output to a relatively small number of
-// flags to keep bash quiet about the verbosity of output. If one
-// really wanted to see all possible matches, appending a '+' to the
-// search word will force the exhaustive list of matches to be printed.
-//
-// ** How to have bash accept completions from a binary:
-// Bash requires that it be informed about each command that programmatic
-// completion should be enabled for. Example addition to a .bashrc
-// file would be (your path to gflags_completions.sh file may differ):
-
-/*
-$ complete -o bashdefault -o default -o nospace -C \
- '/usr/local/bin/gflags_completions.sh --tab_completion_columns $COLUMNS' \
- time env binary_name another_binary [...]
-*/
-
-// This would allow the following to work:
-// $ /path/to/binary_name --vmodule<TAB>
-// Or:
-// $ ./bin/path/another_binary --gfs_u<TAB>
-// (etc)
-//
-// Sadly, it appears that bash gives no easy way to force this behavior for
-// all commands. That's where the "time" in the above example comes in.
-// If you haven't specifically added a command to the list of completion
-// supported commands, you can still get completions by prefixing the
-// entire command with "env".
-// $ env /some/brand/new/binary --vmod<TAB>
-// Assuming that "binary" is a newly compiled binary, this should still
-// produce the expected completion output.
-
-
-#ifndef GOOGLE_GFLAGS_COMPLETIONS_H_
-#define GOOGLE_GFLAGS_COMPLETIONS_H_
-
-// Annoying stuff for windows -- makes sure clients can import these functions
-#ifndef GFLAGS_DLL_DECL
-# ifdef _WIN32
-# define GFLAGS_DLL_DECL __declspec(dllimport)
-# else
-# define GFLAGS_DLL_DECL
-# endif
-#endif
-
-namespace google {
-
-GFLAGS_DLL_DECL void HandleCommandLineCompletions(void);
-
-}
-
-#endif // GOOGLE_GFLAGS_COMPLETIONS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/win/ia32/include/private/config.h b/trunk/third_party/google-gflags/gen/arch/win/ia32/include/private/config.h
deleted file mode 100644
index dcca757..0000000
--- a/trunk/third_party/google-gflags/gen/arch/win/ia32/include/private/config.h
+++ /dev/null
@@ -1,139 +0,0 @@
-/* src/config.h.in. Generated from configure.ac by autoheader. */
-
-/* Sometimes we accidentally #include this config.h instead of the one
- in .. -- this is particularly true for msys/mingw, which uses the
- unix config.h but also runs code in the windows directory.
- */
-#ifdef __MINGW32__
-#include "../config.h"
-#define GOOGLE_GFLAGS_WINDOWS_CONFIG_H_
-#endif
-
-#ifndef GOOGLE_GFLAGS_WINDOWS_CONFIG_H_
-#define GOOGLE_GFLAGS_WINDOWS_CONFIG_H_
-
-/* Always the empty-string on non-windows systems. On windows, should be
- "__declspec(dllexport)". This way, when we compile the dll, we export our
- functions/classes. It's safe to define this here because config.h is only
- used internally, to compile the DLL, and every DLL source file #includes
- "config.h" before anything else. */
-#ifndef GFLAGS_DLL_DECL
-# define GFLAGS_IS_A_DLL 1 /* not set if you're statically linking */
-# define GFLAGS_DLL_DECL __declspec(dllexport)
-# define GFLAGS_DLL_DECL_FOR_UNITTESTS __declspec(dllimport)
-#endif
-
-/* Namespace for Google classes */
-#define GOOGLE_NAMESPACE ::google
-
-/* Define to 1 if you have the <dlfcn.h> header file. */
-#undef HAVE_DLFCN_H
-
-/* Define to 1 if you have the <fnmatch.h> header file. */
-#undef HAVE_FNMATCH_H
-
-/* Define to 1 if you have the <inttypes.h> header file. */
-#undef HAVE_INTTYPES_H
-
-/* Define to 1 if you have the <memory.h> header file. */
-#undef HAVE_MEMORY_H
-
-/* define if the compiler implements namespaces */
-#define HAVE_NAMESPACES 1
-
-/* Define if you have POSIX threads libraries and header files. */
-#undef HAVE_PTHREAD
-
-/* Define to 1 if you have the `putenv' function. */
-#define HAVE_PUTENV 1
-
-/* Define to 1 if you have the `setenv' function. */
-#undef HAVE_SETENV
-
-/* Define to 1 if you have the <stdint.h> header file. */
-#undef HAVE_STDINT_H
-
-/* Define to 1 if you have the <stdlib.h> header file. */
-#define HAVE_STDLIB_H 1
-
-/* Define to 1 if you have the <strings.h> header file. */
-#undef HAVE_STRINGS_H
-
-/* Define to 1 if you have the <string.h> header file. */
-#define HAVE_STRING_H 1
-
-/* Define to 1 if you have the `strtoll' function. */
-#define HAVE_STRTOLL 1
-
-/* Define to 1 if you have the `strtoq' function. */
-#define HAVE_STRTOQ 1
-
-/* Define to 1 if you have the <sys/stat.h> header file. */
-#define HAVE_SYS_STAT_H 1
-
-/* Define to 1 if you have the <sys/types.h> header file. */
-#define HAVE_SYS_TYPES_H 1
-
-/* Define to 1 if you have the <unistd.h> header file. */
-#undef HAVE_UNISTD_H
-
-/* define if your compiler has __attribute__ */
-#undef HAVE___ATTRIBUTE__
-
-/* Define to the sub-directory in which libtool stores uninstalled libraries.
- */
-#undef LT_OBJDIR
-
-/* Name of package */
-#undef PACKAGE
-
-/* Define to the address where bug reports for this package should be sent. */
-#undef PACKAGE_BUGREPORT
-
-/* Define to the full name of this package. */
-#undef PACKAGE_NAME
-
-/* Define to the full name and version of this package. */
-#undef PACKAGE_STRING
-
-/* Define to the one symbol short name of this package. */
-#undef PACKAGE_TARNAME
-
-/* Define to the home page for this package. */
-#undef PACKAGE_URL
-
-/* Define to the version of this package. */
-#undef PACKAGE_VERSION
-
-/* Define to necessary symbol if this constant uses a non-standard name on
- your system. */
-#undef PTHREAD_CREATE_JOINABLE
-
-/* Define to 1 if you have the ANSI C header files. */
-#define STDC_HEADERS 1
-
-/* the namespace where STL code like vector<> is defined */
-#define STL_NAMESPACE std
-
-/* Version number of package */
-#undef VERSION
-
-/* Stops putting the code inside the Google namespace */
-#define _END_GOOGLE_NAMESPACE_ }
-
-/* Puts following code inside the Google namespace */
-#define _START_GOOGLE_NAMESPACE_ namespace google {
-
-// ---------------------------------------------------------------------
-// Extra stuff not found in config.h.in
-
-// This must be defined before the windows.h is included. It's needed
-// for mutex.h, to give access to the TryLock method.
-#ifndef _WIN32_WINNT
-# define _WIN32_WINNT 0x0400
-#endif
-
-// TODO(csilvers): include windows/port.h in every relevant source file instead?
-#include "windows/port.h"
-
-#endif /* GOOGLE_GFLAGS_WINDOWS_CONFIG_H_ */
diff --git a/trunk/third_party/google-gflags/gen/arch/win/x64/include/gflags/gflags.h b/trunk/third_party/google-gflags/gen/arch/win/x64/include/gflags/gflags.h
deleted file mode 100644
index fdafe2a..0000000
--- a/trunk/third_party/google-gflags/gen/arch/win/x64/include/gflags/gflags.h
+++ /dev/null
@@ -1,601 +0,0 @@
-// Copyright (c) 2006, Google Inc.
-// All rights reserved.
-//
-// Redistribution and use in source and binary forms, with or without
-// modification, are permitted provided that the following conditions are
-// met:
-//
-// * Redistributions of source code must retain the above copyright
-// notice, this list of conditions and the following disclaimer.
-// * Redistributions in binary form must reproduce the above
-// copyright notice, this list of conditions and the following disclaimer
-// in the documentation and/or other materials provided with the
-// distribution.
-// * Neither the name of Google Inc. nor the names of its
-// contributors may be used to endorse or promote products derived from
-// this software without specific prior written permission.
-//
-// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
-// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
-// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
-// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
-// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
-// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
-// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-
-// ---
-// Author: Ray Sidney
-// Revamped and reorganized by Craig Silverstein
-//
-// This is the file that should be included by any file which declares
-// or defines a command line flag or wants to parse command line flags
-// or print a program usage message (which will include information about
-// flags). Executive summary, in the form of an example foo.cc file:
-//
-// #include "foo.h" // foo.h has a line "DECLARE_int32(start);"
-// #include "validators.h" // hypothetical file defining ValidateIsFile()
-//
-// DEFINE_int32(end, 1000, "The last record to read");
-//
-// DEFINE_string(filename, "my_file.txt", "The file to read");
-// // Crash if the specified file does not exist.
-// static bool dummy = RegisterFlagValidator(&FLAGS_filename,
-// &ValidateIsFile);
-//
-// DECLARE_bool(verbose); // some other file has a DEFINE_bool(verbose, ...)
-//
-// void MyFunc() {
-// if (FLAGS_verbose) printf("Records %d-%d\n", FLAGS_start, FLAGS_end);
-// }
-//
-// Then, at the command-line:
-// ./foo --noverbose --start=5 --end=100
-//
-// For more details, see
-// doc/gflags.html
-//
-// --- A note about thread-safety:
-//
-// We describe many functions in this routine as being thread-hostile,
-// thread-compatible, or thread-safe. Here are the meanings we use:
-//
-// thread-safe: it is safe for multiple threads to call this routine
-// (or, when referring to a class, methods of this class)
-// concurrently.
-// thread-hostile: it is not safe for multiple threads to call this
-// routine (or methods of this class) concurrently. In gflags,
-// most thread-hostile routines are intended to be called early in,
-// or even before, main() -- that is, before threads are spawned.
-// thread-compatible: it is safe for multiple threads to read from
-// this variable (when applied to variables), or to call const
-// methods of this class (when applied to classes), as long as no
-// other thread is writing to the variable or calling non-const
-// methods of this class.
-
-#ifndef GOOGLE_GFLAGS_H_
-#define GOOGLE_GFLAGS_H_
-
-#include <string>
-#include <vector>
-
-// We care a lot about number of bits things take up. Unfortunately,
-// systems define their bit-specific ints in a lot of different ways.
-// We use our own way, and have a typedef to get there.
-// Note: these commands below may look like "#if 1" or "#if 0", but
-// that's because they were constructed that way at ./configure time.
-// Look at gflags.h.in to see how they're calculated (based on your config).
-#if 0
-#include <stdint.h> // the normal place uint16_t is defined
-#endif
-#if 1
-#include <sys/types.h> // the normal place u_int16_t is defined
-#endif
-#if 0
-#include <inttypes.h> // a third place for uint16_t or u_int16_t
-#endif
-
-// Annoying stuff for windows -- makes sure clients can import these functions
-#if defined(_WIN32)
-# ifndef GFLAGS_DLL_DECL
-# define GFLAGS_DLL_DECL __declspec(dllimport)
-# endif
-# ifndef GFLAGS_DLL_DECLARE_FLAG
-# define GFLAGS_DLL_DECLARE_FLAG __declspec(dllimport)
-# endif
-# ifndef GFLAGS_DLL_DEFINE_FLAG
-# define GFLAGS_DLL_DEFINE_FLAG __declspec(dllexport)
-# endif
-#else
-# ifndef GFLAGS_DLL_DECL
-# define GFLAGS_DLL_DECL
-# endif
-# ifndef GFLAGS_DLL_DECLARE_FLAG
-# define GFLAGS_DLL_DECLARE_FLAG
-# endif
-# ifndef GFLAGS_DLL_DEFINE_FLAG
-# define GFLAGS_DLL_DEFINE_FLAG
-# endif
-#endif
-
-namespace google {
-
-#if 0 // the C99 format
-typedef int32_t int32;
-typedef uint32_t uint32;
-typedef int64_t int64;
-typedef uint64_t uint64;
-#elif 0 // the BSD format
-typedef int32_t int32;
-typedef u_int32_t uint32;
-typedef int64_t int64;
-typedef u_int64_t uint64;
-#elif 1 // the windows (vc7) format
-typedef __int32 int32;
-typedef unsigned __int32 uint32;
-typedef __int64 int64;
-typedef unsigned __int64 uint64;
-#else
-#error Do not know how to define a 32-bit integer quantity on your system
-#endif
-
-// --------------------------------------------------------------------
-// To actually define a flag in a file, use DEFINE_bool,
-// DEFINE_string, etc. at the bottom of this file. You may also find
-// it useful to register a validator with the flag. This ensures that
-// when the flag is parsed from the commandline, or is later set via
-// SetCommandLineOption, we call the validation function. It is _not_
-// called when you assign the value to the flag directly using the = operator.
-//
-// The validation function should return true if the flag value is valid, and
-// false otherwise. If the function returns false for the new setting of the
-// flag, the flag will retain its current value. If it returns false for the
-// default value, ParseCommandLineFlags() will die.
-//
-// This function is safe to call at global construct time (as in the
-// example below).
-//
-// Example use:
-// static bool ValidatePort(const char* flagname, int32 value) {
-// if (value > 0 && value < 32768) // value is ok
-// return true;
-// printf("Invalid value for --%s: %d\n", flagname, (int)value);
-// return false;
-// }
-// DEFINE_int32(port, 0, "What port to listen on");
-// static bool dummy = RegisterFlagValidator(&FLAGS_port, &ValidatePort);
-
-// Returns true if successfully registered, false if not (because the
-// first argument doesn't point to a command-line flag, or because a
-// validator is already registered for this flag).
-GFLAGS_DLL_DECL bool RegisterFlagValidator(const bool* flag,
- bool (*validate_fn)(const char*, bool));
-GFLAGS_DLL_DECL bool RegisterFlagValidator(const int32* flag,
- bool (*validate_fn)(const char*, int32));
-GFLAGS_DLL_DECL bool RegisterFlagValidator(const int64* flag,
- bool (*validate_fn)(const char*, int64));
-GFLAGS_DLL_DECL bool RegisterFlagValidator(const uint64* flag,
- bool (*validate_fn)(const char*, uint64));
-GFLAGS_DLL_DECL bool RegisterFlagValidator(const double* flag,
- bool (*validate_fn)(const char*, double));
-GFLAGS_DLL_DECL bool RegisterFlagValidator(const std::string* flag,
- bool (*validate_fn)(const char*, const std::string&));
-
-
-// --------------------------------------------------------------------
-// These methods are the best way to get access to info about the
-// list of commandline flags. Note that these routines are pretty slow.
-// GetAllFlags: mostly-complete info about the list, sorted by file.
-// ShowUsageWithFlags: pretty-prints the list to stdout (what --help does)
-// ShowUsageWithFlagsRestrict: limit to filenames with restrict as a substr
-//
-// In addition to accessing flags, you can also access argv[0] (the program
-// name) and argv (the entire commandline), which we sock away a copy of.
-// These variables are static, so you should only set them once.
-
-struct GFLAGS_DLL_DECL CommandLineFlagInfo {
- std::string name; // the name of the flag
- std::string type; // the type of the flag: int32, etc
- std::string description; // the "help text" associated with the flag
- std::string current_value; // the current value, as a string
- std::string default_value; // the default value, as a string
- std::string filename; // 'cleaned' version of filename holding the flag
- bool has_validator_fn; // true if RegisterFlagValidator called on flag
- bool is_default; // true if the flag has the default value and
- // has not been set explicitly from the cmdline
- // or via SetCommandLineOption
-};
-
-// Using this inside of a validator is a recipe for a deadlock.
-// TODO(wojtekm) Fix locking when validators are running, to make it safe to
-// call validators during ParseAllFlags.
-// Also make sure then to uncomment the corresponding unit test in
-// commandlineflags_unittest.sh
-extern GFLAGS_DLL_DECL void GetAllFlags(std::vector<CommandLineFlagInfo>* OUTPUT);
-// These two are actually defined in commandlineflags_reporting.cc.
-extern GFLAGS_DLL_DECL void ShowUsageWithFlags(const char *argv0); // what --help does
-extern GFLAGS_DLL_DECL void ShowUsageWithFlagsRestrict(const char *argv0, const char *restrict);
-
-// Create a descriptive string for a flag.
-// Goes to some trouble to make pretty line breaks.
-extern GFLAGS_DLL_DECL std::string DescribeOneFlag(const CommandLineFlagInfo& flag);
-
-// Thread-hostile; meant to be called before any threads are spawned.
-extern GFLAGS_DLL_DECL void SetArgv(int argc, const char** argv);
-// The following functions are thread-safe as long as SetArgv() is
-// only called before any threads start.
-extern GFLAGS_DLL_DECL const std::vector<std::string>& GetArgvs(); // all of argv as a vector
-extern GFLAGS_DLL_DECL const char* GetArgv(); // all of argv as a string
-extern GFLAGS_DLL_DECL const char* GetArgv0(); // only argv0
-extern GFLAGS_DLL_DECL uint32 GetArgvSum(); // simple checksum of argv
-extern GFLAGS_DLL_DECL const char* ProgramInvocationName(); // argv0, or "UNKNOWN" if not set
-extern GFLAGS_DLL_DECL const char* ProgramInvocationShortName(); // basename(argv0)
-// ProgramUsage() is thread-safe as long as SetUsageMessage() is only
-// called before any threads start.
-extern GFLAGS_DLL_DECL const char* ProgramUsage(); // string set by SetUsageMessage()
-
-
-// --------------------------------------------------------------------
-// Normally you access commandline flags by just saying "if (FLAGS_foo)"
-// or whatever, and set them by calling "FLAGS_foo = bar" (or, more
-// commonly, via the DEFINE_foo macro). But if you need a bit more
-// control, we have programmatic ways to get/set the flags as well.
-// These programmatic ways to access flags are thread-safe, but direct
-// access is only thread-compatible.
-
-// Return true iff the flagname was found.
-// OUTPUT is set to the flag's value, or unchanged if we return false.
-extern GFLAGS_DLL_DECL bool GetCommandLineOption(const char* name, std::string* OUTPUT);
-
-// Return true iff the flagname was found. OUTPUT is set to the flag's
-// CommandLineFlagInfo or unchanged if we return false.
-extern GFLAGS_DLL_DECL bool GetCommandLineFlagInfo(const char* name,
- CommandLineFlagInfo* OUTPUT);
-
-// Return the CommandLineFlagInfo of the flagname. exit() if name not found.
-// Example usage, to check if a flag's value is currently the default value:
-// if (GetCommandLineFlagInfoOrDie("foo").is_default) ...
-extern GFLAGS_DLL_DECL CommandLineFlagInfo GetCommandLineFlagInfoOrDie(const char* name);
-
-enum GFLAGS_DLL_DECL FlagSettingMode {
- // update the flag's value (can call this multiple times).
- SET_FLAGS_VALUE,
- // update the flag's value, but *only if* it has not yet been updated
- // with SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef".
- SET_FLAG_IF_DEFAULT,
- // set the flag's default value to this. If the flag has not yet updated
- // yet (via SET_FLAGS_VALUE, SET_FLAG_IF_DEFAULT, or "FLAGS_xxx = nondef")
- // change the flag's current value to the new default value as well.
- SET_FLAGS_DEFAULT
-};
-
-// Set a particular flag ("command line option"). Returns a string
-// describing the new value that the option has been set to. The
-// return value API is not well-specified, so basically just depend on
-// it to be empty if the setting failed for some reason -- the name is
-// not a valid flag name, or the value is not a valid value -- and
-// non-empty else.
-
-// SetCommandLineOption uses set_mode == SET_FLAGS_VALUE (the common case)
-extern GFLAGS_DLL_DECL std::string SetCommandLineOption(const char* name, const char* value);
-extern GFLAGS_DLL_DECL std::string SetCommandLineOptionWithMode(const char* name, const char* value,
- FlagSettingMode set_mode);
-
-
-// --------------------------------------------------------------------
-// Saves the states (value, default value, whether the user has set
-// the flag, registered validators, etc) of all flags, and restores
-// them when the FlagSaver is destroyed. This is very useful in
-// tests, say, when you want to let your tests change the flags, but
-// make sure that they get reverted to the original states when your
-// test is complete.
-//
-// Example usage:
-// void TestFoo() {
-// FlagSaver s1;
-// FLAG_foo = false;
-// FLAG_bar = "some value";
-//
-// // test happens here. You can return at any time
-// // without worrying about restoring the FLAG values.
-// }
-//
-// Note: This class is marked with __attribute__((unused)) because all the
-// work is done in the constructor and destructor, so in the standard
-// usage example above, the compiler would complain that it's an
-// unused variable.
-//
-// This class is thread-safe.
-
-class GFLAGS_DLL_DECL FlagSaver {
- public:
- FlagSaver();
- ~FlagSaver();
-
- private:
- class FlagSaverImpl* impl_; // we use pimpl here to keep API steady
-
- FlagSaver(const FlagSaver&); // no copying!
- void operator=(const FlagSaver&);
-} ;
-
-// --------------------------------------------------------------------
-// Some deprecated or hopefully-soon-to-be-deprecated functions.
-
-// This is often used for logging. TODO(csilvers): figure out a better way
-extern GFLAGS_DLL_DECL std::string CommandlineFlagsIntoString();
-// Usually where this is used, a FlagSaver should be used instead.
-extern GFLAGS_DLL_DECL bool ReadFlagsFromString(const std::string& flagfilecontents,
- const char* prog_name,
- bool errors_are_fatal); // uses SET_FLAGS_VALUE
-
-// These let you manually implement --flagfile functionality.
-// DEPRECATED.
-extern GFLAGS_DLL_DECL bool AppendFlagsIntoFile(const std::string& filename, const char* prog_name);
-extern GFLAGS_DLL_DECL bool SaveCommandFlags(); // actually defined in google.cc !
-extern GFLAGS_DLL_DECL bool ReadFromFlagsFile(const std::string& filename, const char* prog_name,
- bool errors_are_fatal); // uses SET_FLAGS_VALUE
-
-
-// --------------------------------------------------------------------
-// Useful routines for initializing flags from the environment.
-// In each case, if 'varname' does not exist in the environment
-// return defval. If 'varname' does exist but is not valid
-// (e.g., not a number for an int32 flag), abort with an error.
-// Otherwise, return the value. NOTE: for booleans, for true use
-// 't' or 'T' or 'true' or '1', for false 'f' or 'F' or 'false' or '0'.
-
-extern GFLAGS_DLL_DECL bool BoolFromEnv(const char *varname, bool defval);
-extern GFLAGS_DLL_DECL int32 Int32FromEnv(const char *varname, int32 defval);
-extern GFLAGS_DLL_DECL int64 Int64FromEnv(const char *varname, int64 defval);
-extern GFLAGS_DLL_DECL uint64 Uint64FromEnv(const char *varname, uint64 defval);
-extern GFLAGS_DLL_DECL double DoubleFromEnv(const char *varname, double defval);
-extern GFLAGS_DLL_DECL const char *StringFromEnv(const char *varname, const char *defval);
-
-
-// --------------------------------------------------------------------
-// The next two functions parse commandlineflags from main():
-
-// Set the "usage" message for this program. For example:
-// string usage("This program does nothing. Sample usage:\n");
-// usage += argv[0] + " <uselessarg1> <uselessarg2>";
-// SetUsageMessage(usage);
-// Do not include commandline flags in the usage: we do that for you!
-// Thread-hostile; meant to be called before any threads are spawned.
-extern GFLAGS_DLL_DECL void SetUsageMessage(const std::string& usage);
-
-// Looks for flags in argv and parses them. Rearranges argv to put
-// flags first, or removes them entirely if remove_flags is true.
-// If a flag is defined more than once in the command line or flag
-// file, the last definition is used.
-// See top-of-file for more details on this function.
-#ifndef SWIG // In swig, use ParseCommandLineFlagsScript() instead.
-extern GFLAGS_DLL_DECL uint32 ParseCommandLineFlags(int *argc, char*** argv,
- bool remove_flags);
-#endif
-
-
-// Calls to ParseCommandLineNonHelpFlags and then to
-// HandleCommandLineHelpFlags can be used instead of a call to
-// ParseCommandLineFlags during initialization, in order to allow for
-// changing default values for some FLAGS (via
-// e.g. SetCommandLineOptionWithMode calls) between the time of
-// command line parsing and the time of dumping help information for
-// the flags as a result of command line parsing.
-// If a flag is defined more than once in the command line or flag
-// file, the last definition is used.
-extern GFLAGS_DLL_DECL uint32 ParseCommandLineNonHelpFlags(int *argc, char*** argv,
- bool remove_flags);
-// This is actually defined in commandlineflags_reporting.cc.
-// This function is misnamed (it also handles --version, etc.), but
-// it's too late to change that now. :-(
-extern GFLAGS_DLL_DECL void HandleCommandLineHelpFlags(); // in commandlineflags_reporting.cc
-
-// Allow command line reparsing. Disables the error normally
-// generated when an unknown flag is found, since it may be found in a
-// later parse. Thread-hostile; meant to be called before any threads
-// are spawned.
-extern GFLAGS_DLL_DECL void AllowCommandLineReparsing();
-
-// Reparse the flags that have not yet been recognized.
-// Only flags registered since the last parse will be recognized.
-// Any flag value must be provided as part of the argument using "=",
-// not as a separate command line argument that follows the flag argument.
-// Intended for handling flags from dynamically loaded libraries,
-// since their flags are not registered until they are loaded.
-extern GFLAGS_DLL_DECL uint32 ReparseCommandLineNonHelpFlags();
-
-// Clean up memory allocated by flags. This is only needed to reduce
-// the quantity of "potentially leaked" reports emitted by memory
-// debugging tools such as valgrind. It is not required for normal
-// operation, or for the perftools heap-checker. It must only be called
-// when the process is about to exit, and all threads that might
-// access flags are quiescent. Referencing flags after this is called
-// will have unexpected consequences. This is not safe to run when
-// multiple threads might be running: the function is thread-hostile.
-extern GFLAGS_DLL_DECL void ShutDownCommandLineFlags();
-
-
-// --------------------------------------------------------------------
-// Now come the command line flag declaration/definition macros that
-// will actually be used. They're kind of hairy. A major reason
-// for this is initialization: we want people to be able to access
-// variables in global constructors and have that not crash, even if
-// their global constructor runs before the global constructor here.
-// (Obviously, we can't guarantee the flags will have the correct
-// default value in that case, but at least accessing them is safe.)
-// The only way to do that is have flags point to a static buffer.
-// So we make one, using a union to ensure proper alignment, and
-// then use placement-new to actually set up the flag with the
-// correct default value. In the same vein, we have to worry about
-// flag access in global destructors, so FlagRegisterer has to be
-// careful never to destroy the flag-values it constructs.
-//
-// Note that when we define a flag variable FLAGS_<name>, we also
-// preemptively define a junk variable, FLAGS_no<name>. This is to
-// cause a link-time error if someone tries to define 2 flags with
-// names like "logging" and "nologging". We do this because a bool
-// flag FLAG can be set from the command line to true with a "-FLAG"
-// argument, and to false with a "-noFLAG" argument, and so this can
-// potentially avert confusion.
-//
-// We also put flags into their own namespace. It is purposefully
-// named in an opaque way that people should have trouble typing
-// directly. The idea is that DEFINE puts the flag in the weird
-// namespace, and DECLARE imports the flag from there into the current
-// namespace. The net result is to force people to use DECLARE to get
-// access to a flag, rather than saying "extern bool FLAGS_whatever;"
-// or some such instead. We want this so we can put extra
-// functionality (like sanity-checking) in DECLARE if we want, and
-// make sure it is picked up everywhere.
-//
-// We also put the type of the variable in the namespace, so that
-// people can't DECLARE_int32 something that they DEFINE_bool'd
-// elsewhere.
-
-class GFLAGS_DLL_DECL FlagRegisterer {
- public:
- FlagRegisterer(const char* name, const char* type,
- const char* help, const char* filename,
- void* current_storage, void* defvalue_storage);
-};
-
-extern bool FlagsTypeWarn(const char *name);
-
-// If your application #defines STRIP_FLAG_HELP to a non-zero value
-// before #including this file, we remove the help message from the
-// binary file. This can reduce the size of the resulting binary
-// somewhat, and may also be useful for security reasons.
-
-extern const char kStrippedFlagHelp[];
-
-}
-
-#ifndef SWIG // In swig, ignore the main flag declarations
-
-#if defined(STRIP_FLAG_HELP) && STRIP_FLAG_HELP > 0
-// Need this construct to avoid the 'defined but not used' warning.
-#define MAYBE_STRIPPED_HELP(txt) (false ? (txt) : kStrippedFlagHelp)
-#else
-#define MAYBE_STRIPPED_HELP(txt) txt
-#endif
-
-// Each command-line flag has two variables associated with it: one
-// with the current value, and one with the default value. However,
-// we have a third variable, which is where value is assigned; it's a
-// constant. This guarantees that FLAG_##value is initialized at
-// static initialization time (e.g. before program-start) rather than
-// than global construction time (which is after program-start but
-// before main), at least when 'value' is a compile-time constant. We
-// use a small trick for the "default value" variable, and call it
-// FLAGS_no<name>. This serves the second purpose of assuring a
-// compile error if someone tries to define a flag named no<name>
-// which is illegal (--foo and --nofoo both affect the "foo" flag).
-#define DEFINE_VARIABLE(type, shorttype, name, value, help) \
- namespace fL##shorttype { \
- static const type FLAGS_nono##name = value; \
- /* We always want to export defined variables, dll or no */ \
- GFLAGS_DLL_DEFINE_FLAG type FLAGS_##name = FLAGS_nono##name; \
- type FLAGS_no##name = FLAGS_nono##name; \
- static ::google::FlagRegisterer o_##name( \
- #name, #type, MAYBE_STRIPPED_HELP(help), __FILE__, \
- &FLAGS_##name, &FLAGS_no##name); \
- } \
- using fL##shorttype::FLAGS_##name
-
-#define DECLARE_VARIABLE(type, shorttype, name) \
- namespace fL##shorttype { \
- /* We always want to import declared variables, dll or no */ \
- extern GFLAGS_DLL_DECLARE_FLAG type FLAGS_##name; \
- } \
- using fL##shorttype::FLAGS_##name
-
-// For DEFINE_bool, we want to do the extra check that the passed-in
-// value is actually a bool, and not a string or something that can be
-// coerced to a bool. These declarations (no definition needed!) will
-// help us do that, and never evaluate From, which is important.
-// We'll use 'sizeof(IsBool(val))' to distinguish. This code requires
-// that the compiler have different sizes for bool & double. Since
-// this is not guaranteed by the standard, we check it with a
-// compile-time assert (msg[-1] will give a compile-time error).
-namespace fLB {
-struct CompileAssert {};
-typedef CompileAssert expected_sizeof_double_neq_sizeof_bool[
- (sizeof(double) != sizeof(bool)) ? 1 : -1];
-template<typename From> GFLAGS_DLL_DECL double IsBoolFlag(const From& from);
-GFLAGS_DLL_DECL bool IsBoolFlag(bool from);
-} // namespace fLB
-
-#define DECLARE_bool(name) DECLARE_VARIABLE(bool, B, name)
-#define DEFINE_bool(name, val, txt) \
- namespace fLB { \
- typedef ::fLB::CompileAssert FLAG_##name##_value_is_not_a_bool[ \
- (sizeof(::fLB::IsBoolFlag(val)) != sizeof(double)) ? 1 : -1]; \
- } \
- DEFINE_VARIABLE(bool, B, name, val, txt)
-
-#define DECLARE_int32(name) DECLARE_VARIABLE(::google::int32, I, name)
-#define DEFINE_int32(name,val,txt) DEFINE_VARIABLE(::google::int32, I, name, val, txt)
-
-#define DECLARE_int64(name) DECLARE_VARIABLE(::google::int64, I64, name)
-#define DEFINE_int64(name,val,txt) DEFINE_VARIABLE(::google::int64, I64, name, val, txt)
-
-#define DECLARE_uint64(name) DECLARE_VARIABLE(::google::uint64, U64, name)
-#define DEFINE_uint64(name,val,txt) DEFINE_VARIABLE(::google::uint64, U64, name, val, txt)
-
-#define DECLARE_double(name) DECLARE_VARIABLE(double, D, name)
-#define DEFINE_double(name, val, txt) DEFINE_VARIABLE(double, D, name, val, txt)
-
-// Strings are trickier, because they're not a POD, so we can't
-// construct them at static-initialization time (instead they get
-// constructed at global-constructor time, which is much later). To
-// try to avoid crashes in that case, we use a char buffer to store
-// the string, which we can static-initialize, and then placement-new
-// into it later. It's not perfect, but the best we can do.
-
-namespace fLS {
-// The meaning of "string" might be different between now and when the
-// macros below get invoked (e.g., if someone is experimenting with
-// other string implementations that get defined after this file is
-// included). Save the current meaning now and use it in the macros.
-typedef std::string clstring;
-
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- const char *value) {
- return new(stringspot) clstring(value);
-}
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- const clstring &value) {
- return new(stringspot) clstring(value);
-}
-inline clstring* dont_pass0toDEFINE_string(char *stringspot,
- int value);
-} // namespace fLS
-
-#define DECLARE_string(name) namespace fLS { extern GFLAGS_DLL_DECLARE_FLAG ::fLS::clstring& FLAGS_##name; } \
- using fLS::FLAGS_##name
-
-// We need to define a var named FLAGS_no##name so people don't define
-// --string and --nostring. And we need a temporary place to put val
-// so we don't have to evaluate it twice. Two great needs that go
-// great together!
-#define DEFINE_string(name, val, txt) \
- namespace fLS { \
- using ::fLS::clstring; \
- static union { void* align; char s[sizeof(clstring)]; } s_##name[2]; \
- clstring* const FLAGS_no##name = ::fLS:: \
- dont_pass0toDEFINE_string(s_##name[0].s, \
- val); \
- static ::google::FlagRegisterer o_##name( \
- #name, "string", MAYBE_STRIPPED_HELP(txt), __FILE__, \
- s_##name[0].s, new (s_##name[1].s) clstring(*FLAGS_no##name)); \
- GFLAGS_DLL_DEFINE_FLAG clstring& FLAGS_##name = *FLAGS_no##name; \
- } \
- using fLS::FLAGS_##name
-
-#endif // SWIG
-
-#endif // GOOGLE_GFLAGS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/win/x64/include/gflags/gflags_completions.h b/trunk/third_party/google-gflags/gen/arch/win/x64/include/gflags/gflags_completions.h
deleted file mode 100644
index e97de5b..0000000
--- a/trunk/third_party/google-gflags/gen/arch/win/x64/include/gflags/gflags_completions.h
+++ /dev/null
@@ -1,130 +0,0 @@
-// Copyright (c) 2008, Google Inc.
-// All rights reserved.
-//
-// Redistribution and use in source and binary forms, with or without
-// modification, are permitted provided that the following conditions are
-// met:
-//
-// * Redistributions of source code must retain the above copyright
-// notice, this list of conditions and the following disclaimer.
-// * Redistributions in binary form must reproduce the above
-// copyright notice, this list of conditions and the following disclaimer
-// in the documentation and/or other materials provided with the
-// distribution.
-// * Neither the name of Google Inc. nor the names of its
-// contributors may be used to endorse or promote products derived from
-// this software without specific prior written permission.
-//
-// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
-// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
-// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
-// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
-// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
-// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
-// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
-//
-// ---
-// Author: Dave Nicponski
-//
-// Implement helpful bash-style command line flag completions
-//
-// ** Functional API:
-// HandleCommandLineCompletions() should be called early during
-// program startup, but after command line flag code has been
-// initialized, such as the beginning of HandleCommandLineHelpFlags().
-// It checks the value of the flag --tab_completion_word. If this
-// flag is empty, nothing happens here. If it contains a string,
-// however, then HandleCommandLineCompletions() will hijack the
-// process, attempting to identify the intention behind this
-// completion. Regardless of the outcome of this deduction, the
-// process will be terminated, similar to --helpshort flag
-// handling.
-//
-// ** Overview of Bash completions:
-// Bash can be told to programatically determine completions for the
-// current 'cursor word'. It does this by (in this case) invoking a
-// command with some additional arguments identifying the command
-// being executed, the word being completed, and the previous word
-// (if any). Bash then expects a sequence of output lines to be
-// printed to stdout. If these lines all contain a common prefix
-// longer than the cursor word, bash will replace the cursor word
-// with that common prefix, and display nothing. If there isn't such
-// a common prefix, bash will display the lines in pages using 'more'.
-//
-// ** Strategy taken for command line completions:
-// If we can deduce either the exact flag intended, or a common flag
-// prefix, we'll output exactly that. Otherwise, if information
-// must be displayed to the user, we'll take the opportunity to add
-// some helpful information beyond just the flag name (specifically,
-// we'll include the default flag value and as much of the flag's
-// description as can fit on a single terminal line width, as specified
-// by the flag --tab_completion_columns). Furthermore, we'll try to
-// make bash order the output such that the most useful or relevent
-// flags are the most likely to be shown at the top.
-//
-// ** Additional features:
-// To assist in finding that one really useful flag, substring matching
-// was implemented. Before pressing a <TAB> to get completion for the
-// current word, you can append one or more '?' to the flag to do
-// substring matching. Here's the semantics:
-// --foo<TAB> Show me all flags with names prefixed by 'foo'
-// --foo?<TAB> Show me all flags with 'foo' somewhere in the name
-// --foo??<TAB> Same as prior case, but also search in module
-// definition path for 'foo'
-// --foo???<TAB> Same as prior case, but also search in flag
-// descriptions for 'foo'
-// Finally, we'll trim the output to a relatively small number of
-// flags to keep bash quiet about the verbosity of output. If one
-// really wanted to see all possible matches, appending a '+' to the
-// search word will force the exhaustive list of matches to be printed.
-//
-// ** How to have bash accept completions from a binary:
-// Bash requires that it be informed about each command that programmatic
-// completion should be enabled for. Example addition to a .bashrc
-// file would be (your path to gflags_completions.sh file may differ):
-
-/*
-$ complete -o bashdefault -o default -o nospace -C \
- '/usr/local/bin/gflags_completions.sh --tab_completion_columns $COLUMNS' \
- time env binary_name another_binary [...]
-*/
-
-// This would allow the following to work:
-// $ /path/to/binary_name --vmodule<TAB>
-// Or:
-// $ ./bin/path/another_binary --gfs_u<TAB>
-// (etc)
-//
-// Sadly, it appears that bash gives no easy way to force this behavior for
-// all commands. That's where the "time" in the above example comes in.
-// If you haven't specifically added a command to the list of completion
-// supported commands, you can still get completions by prefixing the
-// entire command with "env".
-// $ env /some/brand/new/binary --vmod<TAB>
-// Assuming that "binary" is a newly compiled binary, this should still
-// produce the expected completion output.
-
-
-#ifndef GOOGLE_GFLAGS_COMPLETIONS_H_
-#define GOOGLE_GFLAGS_COMPLETIONS_H_
-
-// Annoying stuff for windows -- makes sure clients can import these functions
-#ifndef GFLAGS_DLL_DECL
-# ifdef _WIN32
-# define GFLAGS_DLL_DECL __declspec(dllimport)
-# else
-# define GFLAGS_DLL_DECL
-# endif
-#endif
-
-namespace google {
-
-GFLAGS_DLL_DECL void HandleCommandLineCompletions(void);
-
-}
-
-#endif // GOOGLE_GFLAGS_COMPLETIONS_H_
diff --git a/trunk/third_party/google-gflags/gen/arch/win/x64/include/private/config.h b/trunk/third_party/google-gflags/gen/arch/win/x64/include/private/config.h
deleted file mode 100644
index dcca757..0000000
--- a/trunk/third_party/google-gflags/gen/arch/win/x64/include/private/config.h
+++ /dev/null
@@ -1,139 +0,0 @@
-/* src/config.h.in. Generated from configure.ac by autoheader. */
-
-/* Sometimes we accidentally #include this config.h instead of the one
- in .. -- this is particularly true for msys/mingw, which uses the
- unix config.h but also runs code in the windows directory.
- */
-#ifdef __MINGW32__
-#include "../config.h"
-#define GOOGLE_GFLAGS_WINDOWS_CONFIG_H_
-#endif
-
-#ifndef GOOGLE_GFLAGS_WINDOWS_CONFIG_H_
-#define GOOGLE_GFLAGS_WINDOWS_CONFIG_H_
-
-/* Always the empty-string on non-windows systems. On windows, should be
- "__declspec(dllexport)". This way, when we compile the dll, we export our
- functions/classes. It's safe to define this here because config.h is only
- used internally, to compile the DLL, and every DLL source file #includes
- "config.h" before anything else. */
-#ifndef GFLAGS_DLL_DECL
-# define GFLAGS_IS_A_DLL 1 /* not set if you're statically linking */
-# define GFLAGS_DLL_DECL __declspec(dllexport)
-# define GFLAGS_DLL_DECL_FOR_UNITTESTS __declspec(dllimport)
-#endif
-
-/* Namespace for Google classes */
-#define GOOGLE_NAMESPACE ::google
-
-/* Define to 1 if you have the <dlfcn.h> header file. */
-#undef HAVE_DLFCN_H
-
-/* Define to 1 if you have the <fnmatch.h> header file. */
-#undef HAVE_FNMATCH_H
-
-/* Define to 1 if you have the <inttypes.h> header file. */
-#undef HAVE_INTTYPES_H
-
-/* Define to 1 if you have the <memory.h> header file. */
-#undef HAVE_MEMORY_H
-
-/* define if the compiler implements namespaces */
-#define HAVE_NAMESPACES 1
-
-/* Define if you have POSIX threads libraries and header files. */
-#undef HAVE_PTHREAD
-
-/* Define to 1 if you have the `putenv' function. */
-#define HAVE_PUTENV 1
-
-/* Define to 1 if you have the `setenv' function. */
-#undef HAVE_SETENV
-
-/* Define to 1 if you have the <stdint.h> header file. */
-#undef HAVE_STDINT_H
-
-/* Define to 1 if you have the <stdlib.h> header file. */
-#define HAVE_STDLIB_H 1
-
-/* Define to 1 if you have the <strings.h> header file. */
-#undef HAVE_STRINGS_H
-
-/* Define to 1 if you have the <string.h> header file. */
-#define HAVE_STRING_H 1
-
-/* Define to 1 if you have the `strtoll' function. */
-#define HAVE_STRTOLL 1
-
-/* Define to 1 if you have the `strtoq' function. */
-#define HAVE_STRTOQ 1
-
-/* Define to 1 if you have the <sys/stat.h> header file. */
-#define HAVE_SYS_STAT_H 1
-
-/* Define to 1 if you have the <sys/types.h> header file. */
-#define HAVE_SYS_TYPES_H 1
-
-/* Define to 1 if you have the <unistd.h> header file. */
-#undef HAVE_UNISTD_H
-
-/* define if your compiler has __attribute__ */
-#undef HAVE___ATTRIBUTE__
-
-/* Define to the sub-directory in which libtool stores uninstalled libraries.
- */
-#undef LT_OBJDIR
-
-/* Name of package */
-#undef PACKAGE
-
-/* Define to the address where bug reports for this package should be sent. */
-#undef PACKAGE_BUGREPORT
-
-/* Define to the full name of this package. */
-#undef PACKAGE_NAME
-
-/* Define to the full name and version of this package. */
-#undef PACKAGE_STRING
-
-/* Define to the one symbol short name of this package. */
-#undef PACKAGE_TARNAME
-
-/* Define to the home page for this package. */
-#undef PACKAGE_URL
-
-/* Define to the version of this package. */
-#undef PACKAGE_VERSION
-
-/* Define to necessary symbol if this constant uses a non-standard name on
- your system. */
-#undef PTHREAD_CREATE_JOINABLE
-
-/* Define to 1 if you have the ANSI C header files. */
-#define STDC_HEADERS 1
-
-/* the namespace where STL code like vector<> is defined */
-#define STL_NAMESPACE std
-
-/* Version number of package */
-#undef VERSION
-
-/* Stops putting the code inside the Google namespace */
-#define _END_GOOGLE_NAMESPACE_ }
-
-/* Puts following code inside the Google namespace */
-#define _START_GOOGLE_NAMESPACE_ namespace google {
-
-// ---------------------------------------------------------------------
-// Extra stuff not found in config.h.in
-
-// This must be defined before the windows.h is included. It's needed
-// for mutex.h, to give access to the TryLock method.
-#ifndef _WIN32_WINNT
-# define _WIN32_WINNT 0x0400
-#endif
-
-// TODO(csilvers): include windows/port.h in every relevant source file instead?
-#include "windows/port.h"
-
-#endif /* GOOGLE_GFLAGS_WINDOWS_CONFIG_H_ */
diff --git a/trunk/third_party/google-gflags/google-gflags.gyp b/trunk/third_party/google-gflags/google-gflags.gyp
deleted file mode 100644
index bd7a3ed..0000000
--- a/trunk/third_party/google-gflags/google-gflags.gyp
+++ /dev/null
@@ -1,62 +0,0 @@
-# Copyright 2011 Google Inc.
-#
-# Licensed under the Apache License, Version 2.0 (the "License");
-# you may not use this file except in compliance with the License.
-# You may obtain a copy of the License at
-#
-# http://www.apache.org/licenses/LICENSE-2.0
-#
-# Unless required by applicable law or agreed to in writing, software
-# distributed under the License is distributed on an "AS IS" BASIS,
-# WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
-# See the License for the specific language governing permissions and
-# limitations under the License.
-
-{
- 'variables': {
- 'gflags_root': '<(DEPTH)/third_party/google-gflags',
- 'gflags_gen_arch_root': '<(gflags_root)/gen/arch/<(OS)/<(target_arch)',
- },
- 'targets': [
- {
- 'target_name': 'google-gflags',
- 'type': '<(library)',
- 'include_dirs': [
- '<(gflags_gen_arch_root)/include/private', # For config.h
- '<(gflags_gen_arch_root)/include', # For configured files.
- '<(gflags_root)/src', # For everything else.
- ],
- 'defines': [
- # These macros exist so flags and symbols are properly
- # exported when building DLLs. Since we don't build DLLs, we
- # need to disable them.
- 'GFLAGS_DLL_DECL=',
- 'GFLAGS_DLL_DECLARE_FLAG=',
- 'GFLAGS_DLL_DEFINE_FLAG=',
- ],
- 'direct_dependent_settings': {
- 'include_dirs': [
- '<(gflags_gen_arch_root)/include', # For configured files.
- '<(gflags_root)/src', # For everything else.
- ],
- 'defines': [
- 'GFLAGS_DLL_DECL=',
- 'GFLAGS_DLL_DECLARE_FLAG=',
- 'GFLAGS_DLL_DEFINE_FLAG=',
- ],
- },
- 'sources': [
- 'src/gflags.cc',
- 'src/gflags_completions.cc',
- 'src/gflags_reporting.cc',
- ],
- 'conditions': [
- ['OS == "win"', {
- 'sources': [
- 'src/windows/port.cc',
- ],
- }],
- ],
- },
- ],
-}
diff --git a/trunk/third_party_mods/ace/LICENSE b/trunk/third_party_mods/ace/LICENSE
deleted file mode 100644
index 9204394..0000000
--- a/trunk/third_party_mods/ace/LICENSE
+++ /dev/null
@@ -1,66 +0,0 @@
-Copyright and Licensing Information for ACE(TM), TAO(TM), CIAO(TM), DAnCE(TM),
-and CoSMIC(TM)
-
-ACE(TM), TAO(TM), CIAO(TM), DAnCE>(TM), and CoSMIC(TM) (henceforth referred to
-as "DOC software") are copyrighted by Douglas C. Schmidt and his research
-group at Washington University, University of California, Irvine, and
-Vanderbilt University, Copyright (c) 1993-2009, all rights reserved. Since DOC
-software is open-source, freely available software, you are free to use,
-modify, copy, and distribute--perpetually and irrevocably--the DOC software
-source code and object code produced from the source, as well as copy and
-distribute modified versions of this software. You must, however, include this
-copyright statement along with any code built using DOC software that you
-release. No copyright statement needs to be provided if you just ship binary
-executables of your software products.
-You can use DOC software in commercial and/or binary software releases and are
-under no obligation to redistribute any of your source code that is built
-using DOC software. Note, however, that you may not misappropriate the DOC
-software code, such as copyrighting it yourself or claiming authorship of the
-DOC software code, in a way that will prevent DOC software from being
-distributed freely using an open-source development model. You needn't inform
-anyone that you're using DOC software in your software, though we encourage
-you to let us know so we can promote your project in the DOC software success
-stories.
-
-The ACE, TAO, CIAO, DAnCE, and CoSMIC web sites are maintained by the DOC
-Group at the Institute for Software Integrated Systems (ISIS) and the Center
-for Distributed Object Computing of Washington University, St. Louis for the
-development of open-source software as part of the open-source software
-community. Submissions are provided by the submitter ``as is'' with no
-warranties whatsoever, including any warranty of merchantability,
-noninfringement of third party intellectual property, or fitness for any
-particular purpose. In no event shall the submitter be liable for any direct,
-indirect, special, exemplary, punitive, or consequential damages, including
-without limitation, lost profits, even if advised of the possibility of such
-damages. Likewise, DOC software is provided as is with no warranties of any
-kind, including the warranties of design, merchantability, and fitness for a
-particular purpose, noninfringement, or arising from a course of dealing,
-usage or trade practice. Washington University, UC Irvine, Vanderbilt
-University, their employees, and students shall have no liability with respect
-to the infringement of copyrights, trade secrets or any patents by DOC
-software or any part thereof. Moreover, in no event will Washington
-University, UC Irvine, or Vanderbilt University, their employees, or students
-be liable for any lost revenue or profits or other special, indirect and
-consequential damages.
-
-DOC software is provided with no support and without any obligation on the
-part of Washington University, UC Irvine, Vanderbilt University, their
-employees, or students to assist in its use, correction, modification, or
-enhancement. A number of companies around the world provide commercial support
-for DOC software, however. DOC software is Y2K-compliant, as long as the
-underlying OS platform is Y2K-compliant. Likewise, DOC software is compliant
-with the new US daylight savings rule passed by Congress as "The Energy Policy
-Act of 2005," which established new daylight savings times (DST) rules for the
-United States that expand DST as of March 2007. Since DOC software obtains
-time/date and calendaring information from operating systems users will not be
-affected by the new DST rules as long as they upgrade their operating systems
-accordingly.
-
-The names ACE(TM), TAO(TM), CIAO(TM), DAnCE(TM), CoSMIC(TM), Washington
-University, UC Irvine, and Vanderbilt University, may not be used to endorse
-or promote products or services derived from this source without express
-written permission from Washington University, UC Irvine, or Vanderbilt
-University. This license grants no permission to call products or services
-derived from this source ACE(TM), TAO(TM), CIAO(TM), DAnCE(TM), or CoSMIC(TM),
-nor does it grant permission for the name Washington University, UC Irvine, or
-Vanderbilt University to appear in their names.
\ No newline at end of file
diff --git a/trunk/third_party_mods/chromium/LICENSE b/trunk/third_party_mods/chromium/LICENSE
deleted file mode 100644
index 8dc3504..0000000
--- a/trunk/third_party_mods/chromium/LICENSE
+++ /dev/null
@@ -1,27 +0,0 @@
-// Copyright (c) 2010 The Chromium Authors. All rights reserved.
-//
-// Redistribution and use in source and binary forms, with or without
-// modification, are permitted provided that the following conditions are
-// met:
-//
-// * Redistributions of source code must retain the above copyright
-// notice, this list of conditions and the following disclaimer.
-// * Redistributions in binary form must reproduce the above
-// copyright notice, this list of conditions and the following disclaimer
-// in the documentation and/or other materials provided with the
-// distribution.
-// * Neither the name of Google Inc. nor the names of its
-// contributors may be used to endorse or promote products derived from
-// this software without specific prior written permission.
-//
-// THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS
-// "AS IS" AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT
-// LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR
-// A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT
-// OWNER OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
-// SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT
-// LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE,
-// DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY
-// THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT
-// (INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE
-// OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
diff --git a/trunk/third_party_mods/mslpl/LICENSE b/trunk/third_party_mods/mslpl/LICENSE
deleted file mode 100644
index 9a3d932..0000000
--- a/trunk/third_party_mods/mslpl/LICENSE
+++ /dev/null
@@ -1,64 +0,0 @@
-This license governs use of code marked as sample or example available on
-this web site without a license agreement, as provided under the section above
-titled NOTICE SPECIFIC TO SOFTWARE AVAILABLE ON THIS WEB SITE. If you use
-such code (the software), you accept this license. If you do not accept the
-license, do not use the software.
-
-1. Definitions
-
-The terms reproduce, reproduction, derivative works, and distribution
-have the same meaning here as under U.S. copyright law.
-
-A contribution is the original software, or any additions or changes to the
-software.
-
-A contributor is any person that distributes its contribution under this
-license.
-
-Licensed patents are a contributors patent claims that read directly on its
-contribution.
-
-2. Grant of Rights
-
-(A) Copyright Grant - Subject to the terms of this license, including the
-license conditions and limitations in section 3, each contributor grants you a
-non-exclusive, worldwide, royalty-free copyright license to reproduce its
-contribution, prepare derivative works of its contribution, and distribute its
-contribution or any derivative works that you create.
-
-(B) Patent Grant - Subject to the terms of this license, including the license
-conditions and limitations in section 3, each contributor grants you a
-non-exclusive, worldwide, royalty-free license under its licensed patents to
-make, have made, use, sell, offer for sale, import, and/or otherwise dispose
-of its contribution in the software or derivative works of the contribution in
-the software.
-
-3. Conditions and Limitations
-
-(A) No Trademark License- This license does not grant you rights to use any
-contributors name, logo, or trademarks.
-
-(B) If you bring a patent claim against any contributor over patents that you
-claim are infringed by the software, your patent license from such contributor
-to the software ends automatically.
-
-(C) If you distribute any portion of the software, you must retain all
-copyright, patent, trademark, and attribution notices that are present in the
-software.
-
-(D) If you distribute any portion of the software in source code form, you may
-do so only under this license by including a complete copy of this license
-with your distribution. If you distribute any portion of the software in
-compiled or object code form, you may only do so under a license that complies
-with this license.
-
-(E) The software is licensed as-is. You bear the risk of using it. The
-contributors give no express warranties, guarantees or conditions. You may
-have additional consumer rights under your local laws which this license
-cannot change. To the extent permitted under your local laws, the contributors
-exclude the implied warranties of merchantability, fitness for a particular
-purpose and non-infringement.
-
-(F) Platform Limitation - The licenses granted in sections 2(A) and 2(B)
-extend only to the software or derivative works that you create that run on a
-Microsoft Windows operating system product.
diff --git a/trunk/third_party_mods/sqrt_floor/LICENSE b/trunk/third_party_mods/sqrt_floor/LICENSE
deleted file mode 100644
index e24dfe7..0000000
--- a/trunk/third_party_mods/sqrt_floor/LICENSE
+++ /dev/null
@@ -1,26 +0,0 @@
-The following email record is related to source files spl_sqrt_floor.c
-and spl_sqrt_floor.s in trunk/src/common_audio/signal_processing/.
-
-
-From: Wilco Dijkstra <Wilco.Dijkstra@ntlworld.com>
-Date: Fri, Jun 24, 2011 at 3:20 AM
-Subject: Re: sqrt routine
-To: Kevin Ma <kma@google.com>
-Hi Kevin,
-Thanks for asking. Those routines are public domain (originally posted to
-comp.sys.arm a long time ago), so you can use them freely for any purpose.
-Cheers,
-Wilco
-
------ Original Message -----
-From: "Kevin Ma" <kma@google.com>
-To: <Wilco.Dijkstra@ntlworld.com>
-Sent: Thursday, June 23, 2011 11:44 PM
-Subject: Fwd: sqrt routine
-Hi Wilco,
-I saw your sqrt routine from several web sites, including
-http://www.finesse.demon.co.uk/steven/sqrt.html.
-Just wonder if there's any copyright information with your Successive
-approximation routines, or if I can freely use it for any purpose.
-Thanks.
-Kevin
diff --git a/trunk/tools/continuous_build/OWNERS b/trunk/tools/continuous_build/OWNERS
deleted file mode 100644
index 60e166c..0000000
--- a/trunk/tools/continuous_build/OWNERS
+++ /dev/null
@@ -1,3 +0,0 @@
-ivinnichenko@webrtc.org
-phoglund@webrtc.org
-kjellander@webrtc.org
diff --git a/trunk/tools/continuous_build/clean_old_files.py b/trunk/tools/continuous_build/clean_old_files.py
deleted file mode 100755
index e87526e..0000000
--- a/trunk/tools/continuous_build/clean_old_files.py
+++ /dev/null
@@ -1,134 +0,0 @@
-#!/usr/bin/env python
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-__author__ = "ivinnichenko@webrtc.org (Illya Vinnichenko)"
-
-"""This script will prune sufficiently old files and empty directories.
-
- The algorithm is to look into the provided directory and delete any files
- that is older than x days, recursively. Then all empty directories will be
- deleted (we can't look at timestamps there since the act of deleting a file
- will refresh the directory's timestamp).
-
- Note: This script has only been tested on Linux.
-"""
-
-from optparse import OptionParser
-import os
-import sys
-import time
-
-# The path is considered whitelisted if any of these entries appear
-# at some point in the path
-WHITELIST = ["buildbot.tac", "master.cfg", "public_html", "changes.pck",
- "webrtc_buildbot"]
-
-
-def is_whitelisted(path):
- """Check if file is whitelisted.
-
- Args:
- path: file path.
- """
- for entry in WHITELIST:
- if entry in path:
- return True
- return False
-
-
-def delete_directory(directory):
- try:
- os.rmdir(directory)
- return True
- except OSError as exception:
- # The directory probably contains newer files.
- print "Could not remove directory %s: reason %s." % (directory, exception)
- return False
-
-
-def delete_file(file):
- try:
- os.remove(file)
- except OSError as exception:
- print "Unexpectedly failed to remove file %s: reason %s." % (file,
- exception)
-
-
-def log_removal(file_or_directory, time_stamp, verbose):
- if verbose:
- str_stamp = time.strftime("%a, %d %b %Y %H:%M:%S +0000",
- time.gmtime(time_stamp))
- print "Removing [%s], stamped on %s" % (file_or_directory, str_stamp)
-
-
-def remove_old_files_and_directories(path, num_days, verbose, skip_dirs):
- """Removes all files under path that are older than num_days days.
- The algorithm also tried to delete all directories, except for those who
- contain files that are sufficiently new.
-
- Implementation note: it doesn't make sense to look at timestamps for
- directories since their timestamps are updated when a file is deleted.
-
- Args:
- path: The starting point.
- num_days: days limit for removal.
- verbose: print every cmd?
- """
- current_time = time.time()
- limit = 60 * 60 * 24 * num_days
-
- # Walk bottom-up so directories are deleted in the right order.
- for root, directories, files in os.walk(path, topdown=False):
- for file in files:
- current_file = os.path.join(root, file)
- time_stamp = os.stat(current_file).st_mtime
-
- if is_whitelisted(current_file):
- continue
-
- if (current_time - time_stamp) > limit:
- delete_file(current_file)
- log_removal(current_file, time_stamp, verbose)
-
- if not skip_dirs:
- for directory in directories:
- current_directory = os.path.join(root, directory)
- time_stamp = os.stat(current_directory).st_mtime
- if delete_directory(current_directory):
- log_removal(current_directory, time_stamp, verbose)
-
-
-def main():
- usage = "usage: %prog -p <base path> -n <number of days> [-q] [-d]"
- parser = OptionParser(usage)
- parser.add_option("-p", "--path", dest="cleanup_path", help="base directory")
- parser.add_option("-n", "--num_days", dest="num_days", help="number of days")
- parser.add_option("-q", "--quiet",
- action="store_false", dest="verbose", default=True,
- help="don't print status messages to stdout")
- parser.add_option("-d", "--delete-dirs-too",
- action="store_false", dest="skip_dirs", default=True,
- help="number of days")
-
- options, args = parser.parse_args()
- if not options.cleanup_path:
- print "You must specify base directory"
- sys.exit(2)
- if not options.num_days:
- print "You must specify number of days old"
- sys.exit(2)
-
- if options.verbose:
- print "Cleaning up everything in %s older than %s days" % (
- options.cleanup_path, options.num_days)
- remove_old_files_and_directories(options.cleanup_path, int(options.num_days),
- options.verbose, options.skip_dirs)
-
-if __name__ == "__main__":
- main()
diff --git a/trunk/tools/continuous_build/clean_third_party_gcda.sh b/trunk/tools/continuous_build/clean_third_party_gcda.sh
deleted file mode 100755
index 38547d8..0000000
--- a/trunk/tools/continuous_build/clean_third_party_gcda.sh
+++ /dev/null
@@ -1,13 +0,0 @@
-#!/bin/sh
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-#
-
-# This script removes all .gcda files from third_party in order to work around
-# a bug in LCOV (this should also increase the bot speed).
-find . -name "*.gcda" -path "*/third_party/*" | xargs rm -f
diff --git a/trunk/tools/continuous_build/generate_coverage_html.sh b/trunk/tools/continuous_build/generate_coverage_html.sh
deleted file mode 100755
index 6b4c175..0000000
--- a/trunk/tools/continuous_build/generate_coverage_html.sh
+++ /dev/null
@@ -1,18 +0,0 @@
-#!/bin/sh
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-#
-
-# Generates a LCOV error report and makes the results readable to all.
-
-genhtml $1 --output-directory $2
-if [ "$?" -ne "0" ]; then
- exit 1
-fi
-
-chmod -R 777 $2
diff --git a/trunk/tools/continuous_build/master.cfg b/trunk/tools/continuous_build/master.cfg
deleted file mode 100755
index aaca7ff..0000000
--- a/trunk/tools/continuous_build/master.cfg
+++ /dev/null
@@ -1,414 +0,0 @@
-#!/usr/bin/env python
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-__author__ = 'ivinnichenko@webrtc.org (Illya Vinnichenko)'
-
-# This is a buildmaster config file for WebRTC project. It must be installed as
-# 'master.cfg' in your buildmaster's base directory (although the filename
-# can be changed with the --basedir option to 'mktap buildbot master').
-
-# It has one job: define a dictionary named BuildmasterConfig. This
-# dictionary has a variety of keys to control different aspects of the
-# buildmaster. They are documented in docs/config.xhtml .
-
-# This is the dictionary that the buildmaster pays attention to. We also use
-# a shorter alias to save typing.
-c = BuildmasterConfig = {}
-
-####### BUILDSLAVES
-
-# the 'slaves' list defines the set of allowable buildslaves. Each element is
-# a BuildSlave object, which is created with bot-name, bot-password. These
-# correspond to values given to the buildslave's mktap invocation.
-
-from buildbot.buildslave import BuildSlave
-
-c['slaves'] = [BuildSlave('android', 'pass', max_builds=1),
- BuildSlave('chromeos', 'pass', max_builds=1),
- BuildSlave('linux-chrome', 'pass', max_builds=1),
- BuildSlave('linux-clang', 'pass', max_builds=1),
- BuildSlave('linux-slave-1', 'pass', max_builds=1),
- BuildSlave('linux-slave-2', 'pass', max_builds=1),
- BuildSlave('linux-slave-4', 'pass', max_builds=1),
- BuildSlave('linux-slave-5', 'pass', max_builds=1),
- BuildSlave('linux-slave-gcc-4.6', 'pass', max_builds=1),
- BuildSlave('linux-valgrind', 'pass', max_builds=1),
- BuildSlave('linux_video', 'pass', max_builds=1),
- BuildSlave('mac-slave-3', 'pass', max_builds=1),
- BuildSlave('mac-slave-2', 'pass', max_builds=1),
- BuildSlave('win-slave-1', 'pass', max_builds=1),
- BuildSlave('win-slave-2', 'pass', max_builds=1)]
-
-# 'slavePortnum' defines the TCP port to listen on. This must match the value
-# configured into the buildslaves (with their --master option)
-
-c['slavePortnum'] = 9989
-
-####### CHANGESOURCES
-
-# the 'change_source' setting tells the buildmaster how it should find out
-# about source code changes. Any class which implements IChangeSource can be
-# put here: there are several in buildbot/changes/*.py to choose from.
-
-from buildbot.changes.pb import PBChangeSource
-from buildbot.changes.svnpoller import SVNPoller
-
-source_code_svn_url = 'http://webrtc.googlecode.com/svn/trunk'
-svn_poller = SVNPoller(svnurl=source_code_svn_url, pollinterval=5*60,
- histmax=10, svnbin='/usr/bin/svn')
-c['change_source'] = svn_poller
-
-####### SCHEDULERS
-
-## configure the Schedulers
-
-from buildbot.scheduler import Scheduler
-webrtc_scheduler = Scheduler(name='all', branch=None, treeStableTimer=5*60,
- builderNames=['Win32Debug',
- 'Win32Release',
- 'MacOS32DBG',
- 'MacOS32Release',
- 'Linux32DBG',
- 'Linux32Release',
- 'Linux64DBG',
- 'Linux64Release',
- 'LinuxClang',
- 'LinuxValgrind',
- 'Linux64DBG-GCC4.6',
- 'LinuxVideoTest',
- 'Android',
- 'ChromeOS'
- ])
-chrome_scheduler = Scheduler(name='chrome', branch=None, treeStableTimer=60*60,
- builderNames=['Chrome'])
-
-c['schedulers'] = [webrtc_scheduler, chrome_scheduler]
-
-####### TESTS
-# Tests to run on Virtual machine bots and our Mac hardware.
-# Defines the supported tests followed by a tuple defining if the tests are
-# enabled on Linux, Mac and/or Windows (in that order; defined in utils.py).
-
-from webrtc_buildbot import utils
-
-NORMAL_TESTS = {
- # Test name Linux Mac Windows
- 'audio_coding_module_test': (True, True, True),
- 'audio_coding_unittests': (True, True, True),
- 'audio_conference_mixer_unittests':(True, True, True),
- 'audio_device_test_api': (True, False, False), # Issue 257
- 'audioproc_unittest': (True, True, True),
- 'cng_unittests': (True, True, True),
- 'g711_unittests': (True, True, True),
- 'g722_unittests': (True, True, True),
- 'libyuv_unittests': (True, True, True),
- 'jpeg_unittests': (True, True, True),
- 'media_file_unittests': (True, True, True),
- 'metrics_unittests': (True, True, True),
- 'neteq_unittests': (True, True, True),
- 'pcm16b_unittests': (True, True, True),
- 'resampler_unittests': (True, True, True),
- 'rtp_rtcp_unittests': (True, True, True),
- 'signal_processing_unittests': (True, True, True),
- 'system_wrappers_unittests': (True, True, True),
- 'test_bwe': (True, True, True),
- 'test_fec': (True, True, True),
- 'test_support_unittests': (True, True, True),
- 'udp_transport_unittests': (True, True, True),
- 'vad_unittests': (True, True, True),
- 'video_codecs_test_framework_integrationtests': (True, True, True),
- 'video_codecs_test_framework_unittests': (True, True, True),
- 'video_coding_unittests': (True, True, True),
- 'video_engine_core_unittests': (True, True, True),
- 'video_processing_unittests': (True, True, True),
- 'voice_engine_unittests': (True, True, True),
- 'vp8_unittests': (True, False, True), # Issue 273.
- 'webrtc_utility_unittests': (True, True, False),
-}
-
-PHYSICAL_MACHINE_TESTS = {
- # Test name Linux Mac Windows
- 'video_render_module_test': (True, False, False),
- 'vie_auto_test': (True, False, False),
- 'voe_auto_test': (True, False, False),
-}
-
-VALGRIND_DISABLED_TESTS = [
- 'audio_coding_module_test', # Issue 270
- 'test_fec', # Too slow for Valgrind
-]
-
-linux_normal_tests = utils.GetEnabledTests(NORMAL_TESTS, 'Linux')
-mac_normal_tests = utils.GetEnabledTests(NORMAL_TESTS, 'Mac')
-windows_normal_tests = utils.GetEnabledTests(NORMAL_TESTS, 'Windows')
-
-linux_physical_machine_tests = utils.GetEnabledTests(PHYSICAL_MACHINE_TESTS,
- 'Linux')
-mac_physical_machine_tests = utils.GetEnabledTests(PHYSICAL_MACHINE_TESTS,
- 'Mac')
-windows_physical_machine_tests = utils.GetEnabledTests(PHYSICAL_MACHINE_TESTS,
- 'Windows')
-
-####### FACTORIES
-# Factories defines how the build is run and can be used in multiple instances
-# on multiple machines, depending on how many builders are defined.
-
-from buildbot.process import factory
-from buildbot.steps import shell
-
-############# Linux factories #######################################
-linux_factory_64_dbg = utils.WebRTCLinuxFactory(
- utils.BuildStatusOracle('linux_factory_64_dbg'))
-linux_factory_64_dbg.EnableBuild()
-linux_factory_64_dbg.EnableTests(linux_normal_tests)
-
-linux_factory_32_release = utils.WebRTCLinuxFactory(
- utils.BuildStatusOracle('linux_factory_32_release'))
-linux_factory_32_release.EnableBuild(release=True, build32=True)
-linux_factory_32_release.EnableTests(linux_normal_tests)
-
-linux_factory_64_release = utils.WebRTCLinuxFactory(
- utils.BuildStatusOracle('linux_factory_64_release'))
-linux_factory_64_release.EnableBuild(release=True)
-linux_factory_64_release.EnableTests(linux_normal_tests)
-
-linux_factory_32_dbg = utils.WebRTCLinuxFactory(
- utils.BuildStatusOracle('linux_factory_32_dbg'))
-linux_factory_32_dbg.EnableCoverage(
- coverage_url='http://webrtc-cb-linux-slave-4.cbf.corp.google.com/coverage/')
-linux_factory_32_dbg.EnableBuild(build32=True)
-linux_factory_32_dbg.EnableTests(linux_normal_tests)
-
-linux_factory_video = utils.WebRTCLinuxFactory(
- utils.BuildStatusOracle('linux_factory_video'))
-linux_factory_video.EnableCoverage(
- coverage_url='http://webrtc-build-bot-se.lul/coverage/')
-linux_factory_video.EnableBuild()
-linux_factory_video.EnableTests(linux_physical_machine_tests)
-
-chromeos_factory = utils.WebRTCLinuxFactory(
- utils.BuildStatusOracle('chromeos_factory'))
-chromeos_factory.EnableBuild(chrome_os=True)
-chromeos_factory.EnableTests(linux_normal_tests)
-
-linux_chrome_factory = utils.WebRTCChromeFactory(
- utils.BuildStatusOracle('linux_chrome'))
-linux_chrome_factory.EnableBuild()
-
-linux_clang = utils.WebRTCLinuxFactory(
- utils.BuildStatusOracle('linux_clang'))
-linux_clang.EnableBuild(clang=True)
-linux_clang.EnableTests(linux_normal_tests)
-
-linux_valgrind = utils.WebRTCLinuxFactory(
- utils.BuildStatusOracle('linux_valgrind'), valgrind_enabled=True)
-linux_valgrind.EnableBuild(release=True)
-# Filter out disabled Valgrind tests:
-valgrind_tests = filter(lambda test: test not in VALGRIND_DISABLED_TESTS,
- linux_normal_tests)
-linux_valgrind.EnableTests(valgrind_tests)
-
-android_factory = utils.WebRTCAndroidFactory(
- utils.BuildStatusOracle('android_factory'))
-android_factory.EnableBuild(product='toro')
-
-############## Mac factories #######################################
-mac_factory_32_dbg = utils.WebRTCMacFactory(
- utils.BuildStatusOracle('mac_factory_32_dbg'))
-mac_factory_32_dbg.EnableBuild(build_type='both')
-mac_factory_32_dbg.EnableTests(mac_normal_tests)
-
-mac_factory_32_release = utils.WebRTCMacFactory(
- utils.BuildStatusOracle('mac_factory_32_release'))
-mac_factory_32_release.EnableBuild(build_type='both', release=True)
-mac_factory_32_release.EnableTests(mac_normal_tests)
-
-############# Windows factories #######################################
-win_factory_32_Debug = utils.WebRTCWinFactory(
- utils.BuildStatusOracle('win_factory_32_debug'))
-win_factory_32_Debug.EnableBuild(configuration='Debug')
-win_factory_32_Debug.EnableTests(windows_normal_tests)
-
-win_factory_32_Release = utils.WebRTCWinFactory(
- utils.BuildStatusOracle('mac_factory_32_release'))
-win_factory_32_Release.EnableBuild(configuration='Release')
-win_factory_32_Release.EnableTests(windows_normal_tests)
-
-####### BUILDERS
-
-# the 'builders' list defines the Builders. Each one is configured with a
-# dictionary, using the following keys:
-# name (required): the name used to describe this builder
-# slavename (required): which slave to use (must appear in c['bots'])
-# builddir (required): which subdirectory to run the builder in
-# factory (required): a BuildFactory to define how the build is run
-# periodicBuildTime (optional): if set, force a build every N seconds
-
-linux_builder_64_debug = {
- 'name': 'Linux64DBG',
- 'slavename': 'linux-slave-1',
- 'builddir': 'linux-slave-1',
- 'factory': linux_factory_64_dbg,
- }
-linux_builder_32_release = {
- 'name': 'Linux32Release',
- 'slavename': 'linux-slave-2',
- 'builddir': 'linux-slave-2',
- 'factory': linux_factory_32_release,
- }
-linux_builder_64_release = {
- 'name': 'Linux64Release',
- 'slavename': 'linux-slave-5',
- 'builddir': 'linux-slave-5',
- 'factory': linux_factory_64_release,
- }
-linux_builder_32_debug = {
- 'name': 'Linux32DBG',
- 'slavename': 'linux-slave-4',
- 'builddir': 'linux-slave-4',
- 'factory': linux_factory_32_dbg,
- }
-mac_builder_32_debug = {
- 'name': 'MacOS32DBG',
- 'slavename': 'mac-slave-3',
- 'builddir': 'mac-slave-3',
- 'factory': mac_factory_32_dbg,
- }
-mac_builder_32_release = {
- 'name': 'MacOS32Release',
- 'slavename': 'mac-slave-2',
- 'builddir': 'mac-slave-2',
- 'factory': mac_factory_32_release,
- }
-chromeos_builder = {
- 'name': 'ChromeOS',
- 'slavename': 'chromeos',
- 'builddir': 'chromeos',
- 'factory': chromeos_factory,
- }
-win_builder_32_debug = {
- 'name': 'Win32Debug',
- 'slavename': 'win-slave-1',
- 'builddir': 'win-32-dbg',
- 'factory': win_factory_32_Debug,
- }
-win_builder_32_release = {
- 'name': 'Win32Release',
- 'slavename': 'win-slave-2',
- 'builddir': 'win-32-release',
- 'factory': win_factory_32_Release,
- }
-linux_builder_video = {
- 'name': 'LinuxVideoTest',
- 'slavename': 'linux_video',
- 'builddir': 'video',
- 'factory': linux_factory_video,
- }
-linux_builder_chrome = {
- 'name': "Chrome",
- 'slavename': "linux-chrome",
- 'builddir': "linux-chrome",
- 'factory': linux_chrome_factory,
- }
-linux_builder_clang = {
- 'name': 'LinuxClang',
- 'slavename': 'linux-clang',
- 'builddir': 'linux-clang',
- 'factory': linux_clang,
- }
-linux_builder_valgrind = {
- 'name': 'LinuxValgrind',
- 'slavename': 'linux-valgrind',
- 'builddir': 'linux-valgrind',
- 'factory': linux_valgrind,
- }
-android_builder_1 = {
- 'name': 'Android',
- 'slavename': 'android',
- 'builddir': 'android',
- 'factory': android_factory,
- }
-linux_builder_gcc_4_6 = {
- 'name': 'Linux64DBG-GCC4.6',
- 'slavename': 'linux-slave-gcc-4.6',
- 'builddir': 'linux-slave-gcc-4.6',
- 'factory': linux_factory_64_dbg,
- }
-c['builders'] = [
- win_builder_32_debug,
- win_builder_32_release,
- mac_builder_32_debug,
- mac_builder_32_release,
- linux_builder_32_debug,
- linux_builder_32_release,
- linux_builder_64_release,
- linux_builder_64_debug,
- linux_builder_clang,
- linux_builder_valgrind,
- linux_builder_gcc_4_6,
- linux_builder_video,
- android_builder_1,
- chromeos_builder,
- linux_builder_chrome,
- ]
-
-####### STATUS TARGETS
-
-# 'status' is a list of Status Targets. The results of each build will be
-# pushed to these targets. buildbot/status/*.py has a variety to choose from,
-# including web pages, email senders, and IRC bots.
-
-from buildbot.status import html
-from buildbot.status import mail
-
-web_page = html.WebStatus(http_port=8010, allowForce=True)
-email_notification = mail.MailNotifier(
- fromaddr='webrtc-cb-watchlist@google.com',
- extraRecipients=['webrtc-cb-watchlist@google.com'],
- sendToInterestedUsers=True,
- mode='failing')
-c['status'] = [web_page, email_notification]
-
-####### DEBUGGING OPTIONS
-
-# if you set 'debugPassword', then you can connect to the buildmaster with
-# the diagnostic tool in contrib/debugclient.py . From this tool, you can
-# manually force builds and inject changes, which may be useful for testing
-# your buildmaster without actually committing changes to your repository (or
-# before you have a functioning 'sources' set up). The debug tool uses the
-# same port number as the slaves do: 'slavePortnum'.
-
-#c['debugPassword'] = 'debugpassword'
-
-# if you set 'manhole', you can ssh into the buildmaster and get an
-# interactive python shell, which may be useful for debugging buildbot
-# internals. It is probably only useful for buildbot developers. You can also
-# use an authorized_keys file, or plain telnet.
-#from buildbot import manhole
-#c['manhole'] = manhole.PasswordManhole('tcp:9999:interface=127.0.0.1',
-# 'admin', 'password')
-
-
-####### PROJECT IDENTITY
-
-# the 'projectName' string will be used to describe the project that this
-# buildbot is working on. For example, it is used as the title of the
-# waterfall HTML page. The 'projectURL' string will be used to provide a link
-# from buildbot HTML pages to your project's home page.
-
-c['projectName'] = 'WebRTC'
-c['projectURL'] = 'http://www.webrtc.org'
-
-# the 'buildbotURL' string should point to the location where the buildbot's
-# internal web server (usually the html.Waterfall page) is visible. This
-# typically uses the port number set in the Waterfall 'status' entry, but
-# with an externally-visible host name which the buildbot cannot figure out
-# without some help.
-
-c['buildbotURL'] = 'http://webrtc-cb-linux-master.cbf.corp.google.com:8010/'
diff --git a/trunk/tools/continuous_build/public_html/README b/trunk/tools/continuous_build/public_html/README
deleted file mode 100644
index 5f59005..0000000
--- a/trunk/tools/continuous_build/public_html/README
+++ /dev/null
@@ -1,18 +0,0 @@
-Modify the look of the buildbot master's waterfall page
--------------------------------------------------------
-
-To deploy these changes, follow these instructions:
-
-* Copy the favicon.ico file to master/public_html/ on the buildbot master
- machine.
-
-* Edit the master/public_html/index.html and add the following to the <head>
- element:
- <link rel="shortcut icon" href="/favicon.ico" type="image/png" />
- <link rel="icon" href="/favicon.ico" type="image/png" />
-
-* Edit the master/public_html/buildbot.css and change the section with
- .LastBuild, .Activity {
- width: 230px;
-
- to be 160px instead. Save the file and exit.
diff --git a/trunk/tools/continuous_build/public_html/favicon.ico b/trunk/tools/continuous_build/public_html/favicon.ico
deleted file mode 100644
index 324a6ca..0000000
--- a/trunk/tools/continuous_build/public_html/favicon.ico
+++ /dev/null
Binary files differ
diff --git a/trunk/tools/continuous_build/webrtc_buildbot/__init__.py b/trunk/tools/continuous_build/webrtc_buildbot/__init__.py
deleted file mode 100755
index 869bd84..0000000
--- a/trunk/tools/continuous_build/webrtc_buildbot/__init__.py
+++ /dev/null
@@ -1,2 +0,0 @@
-# -*- python -*-
-# ex: set syntax=python:
diff --git a/trunk/tools/continuous_build/webrtc_buildbot/utils.py b/trunk/tools/continuous_build/webrtc_buildbot/utils.py
deleted file mode 100755
index 271096c..0000000
--- a/trunk/tools/continuous_build/webrtc_buildbot/utils.py
+++ /dev/null
@@ -1,789 +0,0 @@
-#!/usr/bin/env python
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-__author__ = 'ivinnichenko@webrtc.org (Illya Vinnichenko)'
-
-import buildbot
-import os
-import sys
-import urlparse
-from buildbot.process import factory
-from buildbot.process import properties
-from buildbot.process.properties import WithProperties
-from buildbot.steps.shell import ShellCommand
-
-# Defines the order of the booleans of the supported platforms in the test
-# dictionaries in master.cfg.
-SUPPORTED_PLATFORMS = ('Linux', 'Mac', 'Windows')
-
-SVN_LOCATION = 'http://webrtc.googlecode.com/svn/trunk'
-VALGRIND_CMD = ['tools/valgrind-webrtc/webrtc_tests.sh', '-t', 'cmdline']
-
-DEFAULT_COVERAGE_DIR = '/var/www/coverage/'
-DEFAULT_MASTER_WORK_DIR = '.'
-GCLIENT_RETRIES = 3
-
-# Copied from trunk/tools/build/scripts/master/factory/chromium_factory.py
-# but converted to a list since we set defines instead of using an environment
-# variable.
-#
-# On valgrind bots, override the optimizer settings so we don't inline too
-# much and make the stacks harder to figure out. Use the same settings
-# on all buildbot masters to make it easier to move bots.
-MEMORY_TOOLS_GYP_DEFINES = [
- # GCC flags
- 'mac_debug_optimization=1 ',
- 'mac_release_optimization=1 ',
- 'release_optimize=1 ',
- 'no_gc_sections=1 ',
- 'debug_extra_cflags="-g -fno-inline -fno-omit-frame-pointer '
- '-fno-builtin -fno-optimize-sibling-calls" ',
- 'release_extra_cflags="-g -fno-inline -fno-omit-frame-pointer '
- '-fno-builtin -fno-optimize-sibling-calls" ',
- # MSVS flags
- 'win_debug_RuntimeChecks=0 ',
- 'win_debug_disable_iterator_debugging=1 ',
- 'win_debug_Optimization=1 ',
- 'win_debug_InlineFunctionExpansion=0 ',
- 'win_release_InlineFunctionExpansion=0 ',
- 'win_release_OmitFramePointers=0 ',
-
- 'linux_use_tcmalloc=1 ',
- 'release_valgrind_build=1 ',
- 'werror= ',
-]
-
-
-class WebRTCFactory(factory.BuildFactory):
- """Abstract superclass for all build factories.
-
- A build factory defines a sequence of steps to take in a build process.
- This class provides some helper methods and some abstract methods that
- can be overridden to create customized build sequences.
- """
-
- def __init__(self, build_status_oracle):
- """Creates the abstract factory.
-
- Args:
- build_status_oracle: An instance of BuildStatusOracle which is used to
- keep track of our build state.
- """
- factory.BuildFactory.__init__(self)
-
- self.build_status_oracle = build_status_oracle
- self.properties = properties.Properties()
- self.gyp_params = []
- self.release = False
-
- def EnableBuild(self):
- """Adds steps for building WebRTC [must be overridden].
-
- Implementations of this method must add clean and build steps so that
- when all steps have been run, we have an up-to-date, complete and correct
- build of WebRTC for the platform. It is up to the method how to do this.
- """
- pass
-
- def EnableTests(self, tests):
- """Adds test run steps for all tests in the list.
-
- This method must be run after enabling the build.
-
- Args:
- tests: list of test to be run.
- """
- for test in tests:
- self.EnableTest(test)
-
- def AddCommonStep(self, cmd, descriptor='', workdir='build',
- number_of_retries=0, halt_build_on_failure=True,
- warn_on_failure=False):
- """Adds a step which will run as a shell command on the slave.
-
- NOTE: you are recommended to use this method to add new shell commands
- instead of the base-class addStep method, since steps added here will
- work with the smart-clean system (e.g. only do a full rebuild if the
- previous build failed). Steps handled outside this method will not lead
- to a full rebuild on the next build if they fail.
-
- Args:
- cmd: The command to run. This command follows the contract for
- ShellCommand, so see that documentation for more details.
- descriptor: A string, or a list of strings, describing what the step
- does. The descriptor gets printed in the waterfall display.
- workdir: The working directory to run the command in.
- number_of_retries: Number of times to retry the command, if it fails.
- halt_build_on_failure: Stops the build dead in its tracks if this step
- fails. Use for critical steps. This option does not make sense with
- warn_on_failure.
- warn_on_failure: If true, this step isn't that important and will not
- cause a failed build on failure.
- """
- flunk_on_failure = not warn_on_failure
-
- if type(descriptor) is str:
- descriptor = [descriptor]
- # Add spaces to wrap long test names to make waterfall output more compact.
- wrapped_text = self._WrapLongLines(descriptor)
-
- self.addStep(MonitoredRetryingShellCommand(
- build_status_oracle=self.build_status_oracle,
- number_of_retries=number_of_retries,
- command=cmd,
- workdir=workdir,
- description=wrapped_text + ['running...'],
- descriptionDone=wrapped_text,
- warnOnFailure=warn_on_failure,
- flunkOnFailure=flunk_on_failure,
- haltOnFailure=halt_build_on_failure,
- name='_'.join(descriptor)))
-
- def AddSmartCleanStep(self):
- """Adds a smart clean step.
-
- Smart clean only cleans the whole repository if the build status oracle
- thinks the last build failed. Otherwise it cleans just the build output.
- """
- self.addStep(SmartClean(self.build_status_oracle))
-
- def AddCommonTestRunStep(self, test, descriptor='', cmd=None,
- workdir='build/trunk'):
- """Adds a step for running a single test [must be overridden].
-
- In general, failing tests should not halt the build and allow other tests
- to execute. A failing test should fail, or 'flunk', the build though.
-
- Implementations of this method must add new steps through AddCommonStep
- and not by calling addStep.
-
- Args:
- test: The test binary name. The step will attempt to execute this
- binary in the binary output folder, except if the cmd argument is
- defined (in that case, we will run cmd instead and just use the
- test name in the descriptor).
- descriptor: This should either be a string or a list of strings. The
- descriptor or descriptors are appended to the test name and
- displayed in the waterfall.
- cmd: If necessary, you can specify this argument to override the
- default behavior, which is to just run the binary specified in
- test without arguments.
- workdir: The base working directory to run the command in. This
- directory will map to the WebRTC project root, e.g. the trunk
- directory. This method will make sure that the test binary is run
- in the correct output directory for the platform.
- """
- pass
-
- def EnableTest(self, test):
- """Makes a test run in the build sequence. May be overridden.
-
- Override to handle special cases for specific platforms, for instance if
- a particular test binary requires command line arguments.
-
- Args:
- test: The test name to enable.
- """
- self.AddCommonTestRunStep(test)
-
- def AddGclientSyncStep(self, force_sync):
- """Helper method for invoking gclient sync. Will retry if the operation
- fails.
-
- Args:
- force_sync: If the sync should be forced, i.e. update even for
- unchanged modules (known to be required for Windows sometimes).
- """
- cmd = ['gclient', 'sync']
- if force_sync:
- cmd.append('--force')
- self.AddCommonStep(cmd, descriptor='Sync',
- number_of_retries=GCLIENT_RETRIES)
-
- def AddCommonGYPStep(self, gyp_file, gyp_params=[], descriptor='gyp'):
- """Helper method for invoking GYP on WebRTC.
-
- GYP will generate makefiles or its equivalent in a platform-specific
- manner. A failed GYP step will halt the build.
-
- Implementations of this method must add new steps through AddCommonStep
- and not by calling addStep.
-
- Args:
- gyp_file: The root GYP file to use.
- gyp_params: Custom GYP parameters (same semantics as the GYP_PARAMS
- environment variable).
- descriptor: The descriptor to use for the step.
- """
- cmd = ['./build/gyp_chromium', '--depth=.', gyp_file]
- cmd += gyp_params + self.gyp_params
- self.AddCommonStep(cmd=cmd, workdir='build/trunk', descriptor=descriptor)
-
- def _WrapLongLines(self, string_list, max_line_length=25, wrap_character='_'):
- """ Creates a list with wrapped strings for lines that are too long.
-
- This is done by inserting spaces to long lines with the wrap character
- in. It's a simple way to make long test targets wrap nicer in the
- waterfall display.
-
- This method should only be used for lists that are displayed in the web
- interface!
-
- Args:
- string_list: List of strings where each string represents one line.
- max_line_length: Number of characters a line may have to avoid
- getting wrapped.
- wrap_character: The character we're looking for when inserting a
- space if a string is larger than max_line_length. If no such
- character is found, no space will be inserted.
- Returns:
- A new list of the same length as the input list, but with strings
- that may contain extra spaces in them, if longer than the max
- length.
- """
- result = []
- for line in string_list:
- if len(line) > max_line_length:
- index = line.rfind(wrap_character)
- if index != -1:
- line = line[:index] + ' ' + line[index:]
- result.append(line)
- return result
-
-
-class BuildStatusOracle:
- """Keeps track of a particular build's state.
-
- The oracle uses files in the default master work directory to keep track
- of whether a build has failed. It only keeps track of the most recent build
- until told to forget it.
- """
-
- def __init__(self, builder_name):
- """Creates the oracle.
-
- Args:
- builder_name: The name of the associated builder. This name is used
- in the filename on disk. This name should be unique.
- """
- self.builder_name = builder_name
- self.master_work_dir = DEFAULT_MASTER_WORK_DIR
-
- def LastBuildFailed(self):
- failure_file_path = self._GetFailureBuildPath()
- return os.path.exists(failure_file_path)
-
- def ForgetLastBuild(self):
- if self.LastBuildFailed():
- os.remove(self._GetFailureBuildPath())
-
- def SetLastBuildAsFailed(self):
- open(self._GetFailureBuildPath(), 'w').close()
-
- def _GetFailureBuildPath(self):
- return os.path.join(self.master_work_dir, self.builder_name + ".failed")
-
-
-class MonitoredRetryingShellCommand(ShellCommand):
- """Wraps a shell command and notifies the oracle if the command fails.
-
- If the command fails, there's an option to retry it a number of times.
- Default behavior is to not retry."""
-
- def __init__(self, build_status_oracle, number_of_retries=0, **kwargs):
- ShellCommand.__init__(self, **kwargs)
-
- self.addFactoryArguments(build_status_oracle=build_status_oracle,
- number_of_retries=number_of_retries)
- self.build_status_oracle = build_status_oracle
- self.number_of_retries = number_of_retries
-
- def finished(self, results):
- if (results == buildbot.status.builder.FAILURE or
- results == buildbot.status.builder.EXCEPTION):
- if self.number_of_retries > 0:
- self.number_of_retries -= 1
- self.start()
- return
- else:
- self.build_status_oracle.SetLastBuildAsFailed()
- ShellCommand.finished(self, results)
-
-
-class SmartClean(ShellCommand):
- """Cleans the repository fully or partially depending on the build state."""
- def __init__(self, build_status_oracle, **kwargs):
- ShellCommand.__init__(self, **kwargs)
-
- self.addFactoryArguments(build_status_oracle=build_status_oracle)
- self.haltOnFailure = True
- self.build_status_oracle = build_status_oracle
-
- def start(self):
- if self.build_status_oracle.LastBuildFailed():
- self.build_status_oracle.ForgetLastBuild()
- self.description = ['Nuke Repository', '(Previous Failed)']
- self.setCommand(['rm', '-rf', 'trunk'])
- else:
- self.description = ['Clean']
- self.setCommand('rm -rf trunk/out && '
- 'rm -rf trunk/xcodebuild &&'
- 'rm -rf trunk/build/Debug &&'
- 'rm -rf trunk/build/Release')
- ShellCommand.start(self)
-
-
-class GenerateCodeCoverage(ShellCommand):
- """This custom shell command generates coverage HTML using genhtml.
-
- The command will dump the HTML output into coverage_dir, in a directory
- whose name is generated from the build number and slave name. We will
- expect that the coverage directory is somewhere under the web server root
- (i.e. public html root) that corresponds to the web server URL. That is, if
- we write Foo to the coverage directory we expect that directory to be
- reachable from url/Foo.
- """
-
- def __init__(self, coverage_url, coverage_dir, coverage_file, **kwargs):
- """Prepares the coverage command.
-
- Args:
- coverage_url: The base URL for the serving web server we will use
- when we generate the link to the coverage. This will generally
- be the slave's URL (something like http://slave-hostname/).
- coverage_dir: Where to write coverage HTML.
- coverage_file: The LCOV file to generate the coverage from.
- """
- ShellCommand.__init__(self, **kwargs)
- self.addFactoryArguments(coverage_url=coverage_url,
- coverage_dir=coverage_dir,
- coverage_file=coverage_file)
- self.setDefaultWorkdir('build/trunk')
- self.coverage_url = coverage_url
- self.coverage_dir = coverage_dir
- self.coverage_file = coverage_file
- self.description = ['Coverage Report']
- self.warnOnFailure = True
- self.flunkOnFailure = False
- output_dir = os.path.join(coverage_dir,
- '%(buildername)s_%(buildnumber)s')
- self.setCommand(['./tools/continuous_build/generate_coverage_html.sh',
- coverage_file, WithProperties(output_dir)])
-
- def createSummary(self, log):
- coverage_url = urlparse.urljoin(self.coverage_url,
- '%s_%s' % (self.getProperty('buildername'),
- self.getProperty('buildnumber')))
- self.addURL('click here', coverage_url)
-
- def start(self):
- ShellCommand.start(self)
-
-
-class WebRTCAndroidFactory(WebRTCFactory):
- """Sets up the Android build."""
-
- def __init__(self, build_status_oracle):
- WebRTCFactory.__init__(self, build_status_oracle)
-
- def EnableBuild(self, product='toro'):
- prefix = 'rm -rf out/target/product/%s/obj/' % product
- cleanup_list = [
- 'rm -rf external/webrtc',
- prefix + 'STATIC_LIBRARIES/libwebrtc_*',
- prefix + 'SHARE_LIBRARIES/libwebrtc_*',
- prefix + 'EXECUTABLES/webrtc_*'
- ]
- cmd = ' ; '.join(cleanup_list)
- self.AddCommonStep(cmd, descriptor='cleanup', workdir='build/trunk')
-
- cmd = 'svn checkout %s external/webrtc' % SVN_LOCATION
- self.AddCommonStep(cmd, descriptor='svn (checkout)', workdir='build/trunk')
-
- cmd = ('source build/envsetup.sh && lunch full_%s-eng '
- '&& mmm external/webrtc showcommands' % product)
- self.AddCommonStep(cmd, descriptor='build', workdir='build/trunk')
-
-
-class WebRTCChromeFactory(WebRTCFactory):
- """Sets up the Chrome OS build."""
-
- def __init__(self, build_status_oracle):
- WebRTCFactory.__init__(self, build_status_oracle)
-
- def EnableBuild(self):
- self.AddCommonStep(['rm', '-rf', 'src'], descriptor='Cleanup')
- self.AddGclientSyncStep(force_sync=True)
- self.AddCommonMakeStep('chrome')
-
- def AddCommonMakeStep(self, target, make_extra=None):
- descriptor = ['make ' + target]
- cmd = ['make', target, '-j100']
- if make_extra is not None:
- cmd.append(make_extra)
- self.AddCommonStep(cmd=cmd, descriptor=descriptor,
- warn_on_failure=True, workdir='build/src')
-
-
-class WebRTCLinuxFactory(WebRTCFactory):
- """Sets up the Linux build.
-
- This factory is quite configurable and can run a variety of builds.
- """
-
- def __init__(self, build_status_oracle, valgrind_enabled=False):
- WebRTCFactory.__init__(self, build_status_oracle)
-
- self.build_enabled = False
- self.coverage_enabled = False
- self.valgrind_enabled = valgrind_enabled
-
- def EnableCoverage(self, coverage_url, coverage_dir=DEFAULT_COVERAGE_DIR):
- """Enables coverage measurements using LCOV/GCOV.
-
- This method must be called before enabling build.
-
- Args:
- coverage_url: See the GenerateCodeCoverage command's contract for
- this argument.
- coverage_dir: See the GenerateCodeCoverage command's contract for
- this argument.
- """
- assert self.build_enabled is False
-
- self.coverage_enabled = True
- self.coverage_url = coverage_url
- self.coverage_dir = coverage_dir
-
- def EnableBuild(self, release=False, build32=False, chrome_os=False,
- clang=False):
- if build32:
- self.gyp_params.append('-Dtarget_arch=ia32')
-
- self.build_enabled = True
- self.release = release
-
- self.AddSmartCleanStep()
-
- # Valgrind bots need special GYP defines to enable memory profiling
- # friendly compilation. They already has a custom .gclient configuration
- # file created so they don't need one being generated like the other bots.
- if self.valgrind_enabled:
- for gyp_define in MEMORY_TOOLS_GYP_DEFINES:
- self.gyp_params.append('-D' + gyp_define)
- else:
- self.AddCommonStep(['gclient', 'config', SVN_LOCATION],
- descriptor='gclient_config')
- self.AddGclientSyncStep(force_sync=False)
-
- if chrome_os:
- self.gyp_params.append('-Dchromeos=1')
-
- if clang:
- self.gyp_params.append('-Dclang=1')
-
- if self.coverage_enabled:
- self.gyp_params.append('-Dcoverage=1')
- self.AddCommonGYPStep('webrtc.gyp', descriptor='CommonGYP')
-
- if clang:
- self.AddCommonStep(['trunk/tools/clang/scripts/update.sh'],
- descriptor='Update_Clang')
-
- if self.release:
- self.AddCommonMakeStep('all', make_extra='BUILDTYPE=Release')
- else:
- self.AddCommonMakeStep('all')
-
- def AddCommonTestRunStep(self, test, extra_text=None, cmd=None,
- workdir='build/trunk'):
- descriptor = [test, extra_text] if extra_text else [test]
- if cmd is None:
- test_folder = 'Release' if self.release else 'Debug'
- cmd = ['out/%s/%s' % (test_folder, test)]
- if self.valgrind_enabled:
- cmd = VALGRIND_CMD + cmd
- self.AddCommonStep(cmd, descriptor=descriptor, workdir=workdir,
- halt_build_on_failure=False)
-
- def AddXvfbTestRunStep(self, test_name, test_binary, test_arguments=''):
- """ Adds a test to be run inside a XVFB window manager."""
- cmd = ('xvfb-run '
- '--server-args="-screen 0 800x600x24 -extension Composite" '
- '%s %s' % (test_binary, test_arguments))
- self.AddCommonTestRunStep(test=test_name, cmd=cmd)
-
- def AddCommonMakeStep(self, target, extra_text=None, make_extra=None):
- descriptor = ['make ' + target, extra_text] if extra_text else ['make ' +
- target]
- cmd = ['make', target, '-j100']
- if make_extra:
- cmd.append(make_extra)
- self.AddCommonStep(cmd=cmd, descriptor=descriptor, workdir='build/trunk')
-
- def AddStepsToEstablishCoverageBaseline(self):
- self.AddCommonStep(['lcov', '--directory', '.', '--capture', '-b',
- '.', '--initial',
- '--output-file', 'webrtc_base.info'],
- workdir='build/trunk',
- warn_on_failure=True,
- halt_build_on_failure=False,
- descriptor='LCOV (Baseline Capture)')
- self.AddCommonStep(['lcov', '--extract', 'webrtc_base.info', '*/src/*',
- '--output', 'filtered.info'],
- workdir='build/trunk',
- warn_on_failure=True,
- halt_build_on_failure=False,
- descriptor='LCOV (Baseline Extract)')
- self.AddCommonStep(['lcov', '--remove', 'filtered.info', '*/usr/include/*',
- '/third*', '/testing/*', '*/test/*', '*_unittest.*',
- '*/mock/*', '--output',
- 'webrtc_base_filtered_final.info'],
- workdir='build/trunk',
- warn_on_failure=True,
- halt_build_on_failure=False,
- descriptor='LCOV (Baseline Filter)')
-
- def AddStepsToComputeCoverage(self):
- """Enable coverage data."""
-
- # Delete all third-party .gcda files to save time and work around a bug
- # in lcov which tends to hang when capturing on libjpgturbo.
- self.AddCommonStep(['./tools/continuous_build/clean_third_party_gcda.sh'],
- warn_on_failure=True,
- halt_build_on_failure=False,
- workdir='build/trunk',
- descriptor='LCOV (Delete 3rd party)')
- self.AddCommonStep(['lcov', '--directory', '.', '--capture', '-b',
- '.', '--output-file', 'webrtc.info'],
- warn_on_failure=True,
- halt_build_on_failure=False,
- workdir='build/trunk',
- descriptor='LCOV (Capture)')
- self.AddCommonStep(['lcov', '--extract', 'webrtc.info', '*/src/*',
- '--output', 'test.info'],
- warn_on_failure=True,
- halt_build_on_failure=False,
- workdir='build/trunk',
- descriptor='LCOV (Extract)')
- self.AddCommonStep(['lcov', '--remove', 'test.info', '*/usr/include/*',
- '/third*', '/testing/*', '*/test/*', '*_unittest.*',
- '*/mock/*', '--output',
- 'final.info'],
- warn_on_failure=True,
- halt_build_on_failure=False,
- workdir='build/trunk',
- descriptor='LCOV (Filter)')
- self.AddCommonStep(['lcov', '-a', 'webrtc_base_filtered_final.info', '-a',
- 'final.info', '-o', 'final.info'],
- warn_on_failure=True,
- halt_build_on_failure=False,
- workdir='build/trunk',
- descriptor='LCOV (Merge)')
- self.addStep(GenerateCodeCoverage(coverage_url=self.coverage_url,
- coverage_dir=self.coverage_dir,
- coverage_file='final.info'))
-
- def EnableTests(self, tests):
- if self.coverage_enabled:
- self.AddStepsToEstablishCoverageBaseline()
-
- WebRTCFactory.EnableTests(self, tests)
-
- if self.coverage_enabled:
- self.AddStepsToComputeCoverage()
-
- def EnableTest(self, test):
- """Adds a step for running a test on Linux.
-
- In general, this method will interpret the name as the name of a binary
- in the default build output directory, except for a few special cases
- which require custom command lines.
-
- Args:
- test: the test name as a string.
- """
- if test == 'audioproc_unittest':
- self.AddCommonTestRunStep(test)
- self.AddCommonGYPStep('webrtc.gyp', gyp_params=['-Dprefer_fixed_point=1'],
- descriptor='GYP fixed point')
- self.AddCommonMakeStep(test, extra_text='(fixed point)')
- self.AddCommonTestRunStep(test, extra_text='(fixed point)')
- elif test == 'vie_auto_test':
- # TODO(phoglund): Enable the full stack test once it is completed and
- # nonflaky.
- binary = 'out/Debug/vie_auto_test'
- args = (
- '--automated --gtest_filter="'
- '-ViEVideoVerificationTest.RunsFullStackWithoutErrors" '
- '--capture_test_ensure_resolution_alignment_in_capture_device=false')
- self.AddXvfbTestRunStep(test_name=test, test_binary=binary,
- test_arguments=args)
- elif test == 'video_render_module_test':
- self.AddXvfbTestRunStep(test_name=test,
- test_binary='out/Debug/video_render_module_test')
- elif test == 'voe_auto_test':
- cmd = 'out/Debug/voe_auto_test --automated'
- self.AddCommonTestRunStep(test=test, cmd=cmd)
- else:
- self.AddCommonTestRunStep(test)
-
-
-class WebRTCMacFactory(WebRTCFactory):
- """Sets up the Mac build, both for make and xcode."""
-
- def __init__(self, build_status_oracle):
- WebRTCFactory.__init__(self, build_status_oracle)
- self.build_type = 'both'
- self.allowed_build_types = ['both', 'xcode', 'make']
-
- def EnableBuild(self, build_type='both', release=False):
- self.release = release
-
- if build_type not in self.allowed_build_types:
- print '*** INCORRECT BUILD TYPE (%s)!!! ***' % build_type
- sys.exit(0)
- else:
- self.build_type = build_type
- self.AddSmartCleanStep()
- self.AddCommonStep(['gclient', 'config', SVN_LOCATION],
- descriptor='gclient_config')
- self.AddGclientSyncStep(force_sync=True)
-
- if self.build_type == 'make' or self.build_type == 'both':
- self.AddCommonGYPStep('webrtc.gyp', gyp_params=['-f', 'make'],
- descriptor='EnableMake')
- self.AddCommonMakeStep('all')
-
- def AddCommonTestRunStep(self, test, extra_text=None, cmd=None,
- workdir='build/trunk'):
- descriptor = [test, extra_text] if extra_text else [test]
- if cmd is None:
- out_path = 'xcodebuild' if self.build_type == 'xcode' else 'out'
- test_folder = 'Release' if self.release else 'Debug'
- cmd = ['%s/%s/%s' % (out_path, test_folder, test)]
-
- if self.build_type == 'xcode' or self.build_type == 'both':
- self.AddCommonStep(cmd, descriptor=descriptor + ['(xcode)'],
- halt_build_on_failure=False, workdir=workdir)
- # Execute test only for 'make' build type.
- # If 'both' is enabled we'll only execute the 'xcode' built ones.
- if self.build_type == 'make':
- self.AddCommonStep(cmd, descriptor=descriptor + ['(make)'],
- halt_build_on_failure=False, workdir=workdir)
-
- def AddCommonMakeStep(self, target, extra_text=None, make_extra=None):
- descriptor = [target, extra_text] if extra_text else [target]
- if self.build_type == 'make' or self.build_type == 'both':
- cmd = ['make', target, '-j100']
- if make_extra is not None:
- cmd.append(make_extra)
- if self.release:
- cmd.append('BUILDTYPE=Release')
- self.AddCommonStep(cmd, descriptor=descriptor + ['(make)'],
- workdir='build/trunk')
- if self.build_type == 'xcode' or self.build_type == 'both':
- configuration = 'Release' if self.release else 'Debug'
- cmd = ['xcodebuild', '-project', 'webrtc.xcodeproj', '-configuration',
- configuration, '-target', 'All']
- self.AddCommonStep(cmd, descriptor=descriptor + ['(xcode)'],
- workdir='build/trunk')
-
-class WebRTCWinFactory(WebRTCFactory):
- """Sets up the Windows build.
-
- Allows building with Debug, Release or both in sequence.
- """
-
- def __init__(self, build_status_oracle):
- WebRTCFactory.__init__(self, build_status_oracle)
- self.configuration = 'Debug'
- self.platform = 'x64'
- self.allowed_platforms = ['x64', 'Win32']
- self.allowed_configurations = ['Debug', 'Release', 'both']
-
- def EnableBuild(self, platform='Win32', configuration='Debug'):
- if platform not in self.allowed_platforms:
- print '*** INCORRECT PLATFORM (%s)!!! ***' % platform
- sys.exit(0)
- else:
- self.platform = platform
- if configuration not in self.allowed_configurations:
- print '*** INCORRECT CONFIGURATION (%s)!!! ***' % configuration
- sys.exit(0)
- else:
- self.configuration = configuration
-
- self.AddSmartCleanStep()
- self.AddCommonStep(['gclient', 'config', SVN_LOCATION],
- descriptor='gclient_config')
- self.AddGclientSyncStep(force_sync=True)
-
- if self.configuration == 'Debug' or self.configuration == 'both':
- cmd = ['msbuild', 'webrtc.sln', '/t:Clean',
- '/p:Configuration=Debug;Platform=%s' % (self.platform)]
- self.AddCommonStep(cmd, descriptor='Build(Clean)', workdir='build/trunk')
- cmd = ['msbuild', 'webrtc.sln',
- '/p:Configuration=Debug;Platform=%s' % (self.platform)]
- self.AddCommonStep(cmd, descriptor='Build(Debug)', workdir='build/trunk')
- if self.configuration == 'Release' or self.configuration == 'both':
- cmd = ['msbuild', 'webrtc.sln', '/t:Clean',
- '/p:Configuration=Release;Platform=%s' % (self.platform)]
- self.AddCommonStep(cmd, descriptor='Build(Clean)', workdir='build/trunk')
- cmd = ['msbuild', 'webrtc.sln',
- '/p:Configuration=Release;Platform=%s' % (self.platform)]
- self.AddCommonStep(cmd, descriptor='Build(Release)',
- workdir='build/trunk')
-
- def AddCommonTestRunStep(self, test, cmd=None, workdir='build/trunk'):
- descriptor = [test]
- if self.configuration == 'Debug' or self.configuration == 'both':
- if cmd is None:
- cmd = ['build\Debug\%s.exe' % test]
- self.AddCommonStep(cmd, descriptor=descriptor,
- halt_build_on_failure=False, workdir=workdir)
- if self.configuration == 'Release' or self.configuration == 'both':
- if cmd is None:
- cmd = ['build\Release\%s.exe' % test]
- self.AddCommonStep(cmd, descriptor=descriptor,
- halt_build_on_failure=False, workdir=workdir)
-
-# Utility functions
-
-
-class UnsupportedPlatformError(Exception):
- pass
-
-
-def GetEnabledTests(test_dict, platform):
- """Returns a list of enabled test names for the provided platform.
-
- Args:
- test_dict: Dictionary mapping test names to tuples representing if the
- test shall be enabled on each platform. Each tuple contains one
- boolean for each platform. The platforms are in the order specified
- by SUPPORTED_PLATFORMS.
- platform: The platform we're looking to get the tests for.
-
- Returns:
- A list of test names, sorted alphabetically.
-
- Raises:
- UnsupportedPlatformError: if the platform supplied is not supported.
- """
- if platform not in SUPPORTED_PLATFORMS:
- raise UnsupportedPlatformError('*** UNSUPPORTED PLATFORM (%s)!!! ***' %
- platform)
- result = []
- platform_index = SUPPORTED_PLATFORMS.index(platform)
- for test_name, enabled_platforms in test_dict.iteritems():
- if enabled_platforms[platform_index]:
- result.append(test_name)
- result.sort()
- return result
diff --git a/trunk/tools/continuous_build/webrtc_buildbot/utils_test.py b/trunk/tools/continuous_build/webrtc_buildbot/utils_test.py
deleted file mode 100644
index 7da58f6..0000000
--- a/trunk/tools/continuous_build/webrtc_buildbot/utils_test.py
+++ /dev/null
@@ -1,76 +0,0 @@
-#!/usr/bin/env python
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-"""Unit tests for helper functions in utils.py."""
-
-__author__ = 'kjellander@webrtc.org (Henrik Kjellander)'
-
-import unittest
-
-from webrtc_buildbot import utils
-
-
-class Test(unittest.TestCase):
-
- def testGetEnabledTests(self):
- tests = {
- # Name Linux Mac Windows
- 'test_1': (True, True, False),
- 'test_2': (True, False, False),
- }
- result = utils.GetEnabledTests(tests, 'Linux')
- self.assertEqual(2, len(result))
- self.assertEqual('test_1', result[0])
- self.assertEqual('test_2', result[1])
-
- result = utils.GetEnabledTests(tests, 'Mac')
- self.assertEqual(1, len(result))
- self.assertEqual('test_1', result[0])
-
- result = utils.GetEnabledTests(tests, 'Windows')
- self.assertEqual(0, len(result))
-
- self.assertRaises(utils.UnsupportedPlatformError,
- utils.GetEnabledTests, tests, 'BeOS')
-
- def testEmptyListExitQuietly(self):
- factory = utils.WebRTCFactory()
- self.assertEqual([], factory._WrapLongLines([]))
-
- def testShortLinesShallNotWrap(self):
- factory = utils.WebRTCFactory()
- self.assertEqual(['a'], factory._WrapLongLines(['a']))
-
- string_25_len = '12345678901234567890123_5'
- result = factory._WrapLongLines([string_25_len, string_25_len])
- self.assertEqual(string_25_len, result[0])
- self.assertEqual(string_25_len, result[1])
-
- def testLinesWithMoreThan25CharactersWithNoWrapCharacterIsUnchanged(self):
- factory = utils.WebRTCFactory()
- string_26_len = '12345678901234567890123456'
- result = factory._WrapLongLines([string_26_len, string_26_len])
- self.assertEqual(string_26_len, result[0])
- self.assertEqual(string_26_len, result[1])
-
- def testLinesWithMoreThan25CharactersShallWrapOnWrapCharacter(self):
- factory = utils.WebRTCFactory()
- string_26_len = '123456789012345678901234_6'
- test_list = [string_26_len, string_26_len]
- result = factory._WrapLongLines(test_list)
- expected_result = '123456789012345678901234 _6'
- self.assertEqual(expected_result, result[0])
- self.assertEqual(expected_result, result[1])
- # Verify the original test_list was not modified too.
- self.assertEqual(string_26_len, test_list[0])
- self.assertEqual(string_26_len, test_list[1])
-
-
-if __name__ == '__main__':
- unittest.main()
diff --git a/trunk/tools/coverity/OWNERS b/trunk/tools/coverity/OWNERS
deleted file mode 100644
index b44992b..0000000
--- a/trunk/tools/coverity/OWNERS
+++ /dev/null
@@ -1,3 +0,0 @@
-kjellander@webrtc.org
-phoglund@webrtc.org
-
diff --git a/trunk/tools/coverity/coverity.py b/trunk/tools/coverity/coverity.py
deleted file mode 100755
index 84b95bc..0000000
--- a/trunk/tools/coverity/coverity.py
+++ /dev/null
@@ -1,324 +0,0 @@
-#!/usr/bin/env python
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-"""
-Runs Coverity Static Analysis on a build of WebRTC.
-
-This script is a modified copy of Chromium's tools/coverity/coverity.py
-Changes made:
- * Replaced deprecated switches for cov-commit-defects command:
- * Using --host instead of --remote
- * Using --stream instead of --product
- * Removed --cxx (now default enabled)
- * Changed cleaning of output path, since WebRTC's out dir is located directly
- in trunk/
- * Updated some default constants.
-
-The script runs on all WebRTC supported platforms.
-
-On Windows, this script should be run in a Visual Studio Command Prompt, so
-that the INCLUDE, LIB, and PATH environment variables are set properly for
-Visual Studio.
-
-Usage examples:
- coverity.py
- coverity.py --dry-run
- coverity.py --target=debug
- %comspec% /c ""C:\Program Files\Microsoft Visual Studio 8\VC\vcvarsall.bat"
- x86 && C:\Python24\python.exe C:\coverity.py"
-
-For a full list of options, pass the '--help' switch.
-
-See http://support.microsoft.com/kb/308569 for running this script as a
-Scheduled Task on Windows XP.
-"""
-
-import optparse
-import os
-import os.path
-import shutil
-import subprocess
-import sys
-import time
-
-# These constants provide default values, but are exposed as command-line
-# flags. See the --help for more info. Note that for historical reasons
-# (the script started out as Windows-only and has legacy usages which pre-date
-# these switches), the constants are all tuned for Windows.
-# Usage of this script on Linux pretty much requires explicit
-# --source-dir, --coverity-bin-dir, --coverity-intermediate-dir, and
-# --coverity-target command line flags.
-
-WEBRTC_SOURCE_DIR = 'C:\\webrtc.latest'
-
-# Relative to WEBRTC_SOURCE_DIR. Only applies to Windows platform.
-WEBRTC_SOLUTION_FILE = 'webrtc.sln'
-
-# Relative to WEBRTC_SOURCE_DIR. Only applies to Windows platform.
-WEBRTC_SOLUTION_DIR = 'build'
-
-COVERITY_BIN_DIR = 'C:\\coverity-integrity-center\\static-analysis\\bin'
-
-COVERITY_INTERMEDIATE_DIR = 'C:\\coverity\\cvbuild\\cr_int'
-
-COVERITY_ANALYZE_OPTIONS = ('--security --concurrency '
- '--enable ATOMICITY '
- '--enable MISSING_LOCK '
- '--enable DELETE_VOID '
- '--checker-option PASS_BY_VALUE:size_threshold:16 '
- '--checker-option '
- 'USE_AFTER_FREE:allow_simple_use:false '
- '--enable-constraint-fpp '
- '--enable-callgraph-metrics')
-
-# Might need to be changed to FQDN
-COVERITY_REMOTE = 'localhost'
-
-COVERITY_PORT = '8080'
-
-COVERITY_STREAM = 'trunk'
-
-COVERITY_TARGET = 'Windows'
-
-COVERITY_USER = 'coverityanalyzer'
-# looking for a PASSWORD constant? Look at --coverity-password-file instead.
-
-# Relative to WEBRTC_SOURCE_DIR. Contains the pid of this script.
-LOCK_FILE = 'coverity.lock'
-
-
-def _ReadPassword(pwfilename):
- """Reads the coverity password in from a file where it was stashed"""
- pwfile = open(pwfilename, 'r')
- password = pwfile.readline()
- pwfile.close()
- return password.rstrip()
-
-
-def _RunCommand(cmd, dry_run, shell=False, echo_cmd=True):
- """Runs the command if dry_run is false, otherwise just prints the command."""
- if echo_cmd:
- print cmd
- if not dry_run:
- return subprocess.call(cmd, shell=shell)
- else:
- return 0
-
-
-def _ReleaseLock(lock_file, lock_filename):
- """Removes the lockfile. Function-ized so we can bail from anywhere"""
- os.close(lock_file)
- os.remove(lock_filename)
-
-
-def run_coverity(options, args):
- """Runs all the selected tests for the given build type and target."""
- # Create the lock file to prevent another instance of this script from
- # running.
- lock_filename = os.path.join(options.source_dir, LOCK_FILE)
- try:
- lock_file = os.open(lock_filename,
- os.O_CREAT | os.O_EXCL | os.O_TRUNC | os.O_RDWR)
- except OSError, err:
- print 'Failed to open lock file:\n ' + str(err)
- return 1
-
- # Write the pid of this script (the python.exe process) to the lock file.
- os.write(lock_file, str(os.getpid()))
-
- options.target = options.target.title()
-
- start_time = time.time()
-
- print 'Change directory to ' + options.source_dir
- os.chdir(options.source_dir)
-
- # The coverity-password filename may have been a relative path.
- # If so, assume it's relative to the source directory, which means
- # the time to read the password is after we do the chdir().
- coverity_password = _ReadPassword(options.coverity_password_file)
-
- cmd = 'gclient sync'
- gclient_exit = _RunCommand(cmd, options.dry_run, shell=True)
- if gclient_exit != 0:
- print 'gclient aborted with status %s' % gclient_exit
- _ReleaseLock(lock_file, lock_filename)
- return 1
-
- print 'Elapsed time: %ds' % (time.time() - start_time)
-
- # Do a clean build. Remove the build output directory first.
- if sys.platform.startswith('linux'):
- rm_path = os.path.join(options.source_dir,'out',options.target)
- elif sys.platform == 'win32':
- rm_path = os.path.join(options.source_dir,options.solution_dir,
- options.target)
- elif sys.platform == 'darwin':
- rm_path = os.path.join(options.source_dir,'xcodebuild')
- else:
- print 'Platform "%s" unrecognized, aborting' % sys.platform
- _ReleaseLock(lock_file, lock_filename)
- return 1
-
- if options.dry_run:
- print 'shutil.rmtree(%s)' % repr(rm_path)
- else:
- shutil.rmtree(rm_path,True)
-
- if options.preserve_intermediate_dir:
- print 'Preserving intermediate directory.'
- else:
- if options.dry_run:
- print 'shutil.rmtree(%s)' % repr(options.coverity_intermediate_dir)
- print 'os.mkdir(%s)' % repr(options.coverity_intermediate_dir)
- else:
- shutil.rmtree(options.coverity_intermediate_dir,True)
- os.mkdir(options.coverity_intermediate_dir)
-
- print 'Elapsed time: %ds' % (time.time() - start_time)
-
- use_shell_during_make = False
- if sys.platform.startswith('linux'):
- use_shell_during_make = True
- _RunCommand('pwd', options.dry_run, shell=True)
- cmd = '%s/cov-build --dir %s make BUILDTYPE=%s All' % (
- options.coverity_bin_dir, options.coverity_intermediate_dir,
- options.target)
- elif sys.platform == 'win32':
- cmd = ('%s\\cov-build.exe --dir %s devenv.com %s\\%s /build %s '
- '/project webrtc.vcproj') % (
- options.coverity_bin_dir, options.coverity_intermediate_dir,
- options.source_dir, options.solution_file, options.target)
- elif sys.platform == 'darwin':
- use_shell_during_make = True
- _RunCommand('pwd', options.dry_run, shell=True)
- cmd = ('%s/cov-build --dir %s xcodebuild -project webrtc.xcodeproj '
- '-configuration %s -target All') % (
- options.coverity_bin_dir, options.coverity_intermediate_dir,
- options.target)
-
-
- _RunCommand(cmd, options.dry_run, shell=use_shell_during_make)
- print 'Elapsed time: %ds' % (time.time() - start_time)
-
- cov_analyze_exe = os.path.join(options.coverity_bin_dir,'cov-analyze')
- cmd = '%s --dir %s %s' % (cov_analyze_exe,
- options.coverity_intermediate_dir,
- options.coverity_analyze_options)
- _RunCommand(cmd, options.dry_run, shell=use_shell_during_make)
- print 'Elapsed time: %ds' % (time.time() - start_time)
-
- cov_commit_exe = os.path.join(options.coverity_bin_dir,'cov-commit-defects')
-
- # On Linux we have started using a Target with a space in it, so we want
- # to quote it. On the other hand, Windows quoting doesn't work quite the
- # same way. To be conservative, I'd like to avoid quoting an argument
- # that doesn't need quoting and which we haven't historically been quoting
- # on that platform. So, only quote the target if we have to.
- coverity_target = options.coverity_target
- if sys.platform != 'win32':
- coverity_target = '"%s"' % coverity_target
-
- cmd = ('%s --dir %s --host %s --port %s '
- '--stream %s '
- '--target %s '
- '--user %s '
- '--password %s') % (cov_commit_exe,
- options.coverity_intermediate_dir,
- options.coverity_dbhost,
- options.coverity_port,
- options.coverity_stream,
- coverity_target,
- options.coverity_user,
- coverity_password)
- # Avoid echoing the Commit command because it has a password in it
- print 'Commiting defects to Coverity Integrity Manager server...'
- _RunCommand(cmd, options.dry_run, shell=use_shell_during_make, echo_cmd=False)
-
- print 'Completed! Total time: %ds' % (time.time() - start_time)
-
- _ReleaseLock(lock_file, lock_filename)
-
- return 0
-
-
-def main():
- option_parser = optparse.OptionParser()
- option_parser.add_option('', '--dry-run', action='store_true', default=False,
- help='print but don\'t run the commands')
-
- option_parser.add_option('', '--target', default='Release',
- help='build target (Debug or Release)')
-
- option_parser.add_option('', '--source-dir', dest='source_dir',
- help='full path to directory ABOVE "src"',
- default=WEBRTC_SOURCE_DIR)
-
- option_parser.add_option('', '--solution-file', dest='solution_file',
- help='filename of solution file to build (Win only)',
- default=WEBRTC_SOLUTION_FILE)
-
- option_parser.add_option('', '--solution-dir', dest='solution_dir',
- help='build directory for the solution (Win only)',
- default=WEBRTC_SOLUTION_DIR)
-
- option_parser.add_option('', '--coverity-bin-dir', dest='coverity_bin_dir',
- default=COVERITY_BIN_DIR)
-
- option_parser.add_option('', '--coverity-intermediate-dir',
- dest='coverity_intermediate_dir',
- default=COVERITY_INTERMEDIATE_DIR)
-
- option_parser.add_option('', '--coverity-analyze-options',
- dest='coverity_analyze_options',
- help=('all cov-analyze options, e.g. "%s"'
- % COVERITY_ANALYZE_OPTIONS),
- default=COVERITY_ANALYZE_OPTIONS)
-
- option_parser.add_option('', '--coverity-db-host',
- dest='coverity_dbhost',
- help=('coverity defect db server hostname, e.g. %s'
- % COVERITY_REMOTE),
- default=COVERITY_REMOTE)
-
- option_parser.add_option('', '--coverity-db-port', dest='coverity_port',
- help=('port # of coverity web/db server, e.g. %s'
- % COVERITY_PORT),
- default=COVERITY_PORT)
-
- option_parser.add_option('', '--coverity-stream', dest='coverity_stream',
- help=('Name of stream reported to Coverity, e.g. %s'
- % COVERITY_STREAM),
- default=COVERITY_STREAM)
-
- option_parser.add_option('', '--coverity-target', dest='coverity_target',
- help='Platform Target reported to coverity',
- default=COVERITY_TARGET)
-
- option_parser.add_option('', '--coverity-user', dest='coverity_user',
- help='Username used to log into coverity',
- default=COVERITY_USER)
-
- option_parser.add_option('', '--coverity-password-file',
- dest='coverity_password_file',
- help='file containing the coverity password',
- default='coverity-password')
-
- helpmsg = ('By default, the intermediate dir is emptied before analysis. '
- 'This switch disables that behavior.')
- option_parser.add_option('', '--preserve-intermediate-dir',
- action='store_true', help=helpmsg,
- default=False)
-
- options, args = option_parser.parse_args()
- return run_coverity(options, args)
-
-
-if '__main__' == __name__:
- sys.exit(main())
diff --git a/trunk/tools/create_supplement_gypi.py b/trunk/tools/create_supplement_gypi.py
deleted file mode 100644
index f5a08e6..0000000
--- a/trunk/tools/create_supplement_gypi.py
+++ /dev/null
@@ -1,23 +0,0 @@
-#!/usr/bin/env python
-# Copyright (c) 2011 The Chromium Authors. All rights reserved.
-# Use of this source code is governed by a BSD-style license that can be
-# found in the LICENSE file.
-
-import sys
-
-supplement_gypi = """#!/usr/bin/env python
-# This file is generated by %s. Not for check-in.
-# Please see the WebRTC DEPS file for details.
-{
- 'variables': {
- 'build_with_chromium': 0,
- 'inside_chromium_build': 0,
- }
-}
-"""
-
-def main(argv):
- open(argv[1], 'w').write(supplement_gypi % argv[0])
-
-if __name__ == '__main__':
- sys.exit(main(sys.argv))
diff --git a/trunk/tools/matlab/maxUnwrap.m b/trunk/tools/matlab/maxUnwrap.m
deleted file mode 100644
index 276c952..0000000
--- a/trunk/tools/matlab/maxUnwrap.m
+++ /dev/null
@@ -1,25 +0,0 @@
-function sequence = maxUnwrap(sequence, max)
-%
-% sequence = maxUnwrap(sequence, max)
-% Unwraps when a wrap around is detected.
-%
-% Arguments
-%
-% sequence: The vector to unwrap.
-% max: The maximum value that the sequence can take,
-% and after which it will wrap over to 0.
-%
-% Return value
-%
-% sequence: The unwrapped vector.
-%
-
-% Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-%
-% Use of this source code is governed by a BSD-style license
-% that can be found in the LICENSE file in the root of the source
-% tree. An additional intellectual property rights grant can be found
-% in the file PATENTS. All contributing project authors may
-% be found in the AUTHORS file in the root of the source tree.
-
-sequence = round((unwrap(2 * pi * sequence / max) * max) / (2 * pi));
diff --git a/trunk/tools/matlab/parseLog.m b/trunk/tools/matlab/parseLog.m
deleted file mode 100644
index 5d4c3f7..0000000
--- a/trunk/tools/matlab/parseLog.m
+++ /dev/null
@@ -1,54 +0,0 @@
-function parsed = parseLog(filename)
-%
-% parsed = parseLog(filename)
-% Parses a DataLog text file, with the filename specified in the string
-% filename, into a struct with each column name as a field, and with the
-% column data stored as a vector in that field.
-%
-% Arguments
-%
-% filename: A string with the name of the file to parse.
-%
-% Return value
-%
-% parsed: A struct containing each column parsed from the input file
-% as a field and with the column data stored as a vector in that
-% field.
-%
-
-% Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-%
-% Use of this source code is governed by a BSD-style license
-% that can be found in the LICENSE file in the root of the source
-% tree. An additional intellectual property rights grant can be found
-% in the file PATENTS. All contributing project authors may
-% be found in the AUTHORS file in the root of the source tree.
-
-table = importdata(filename, ',', 1);
-if ~isstruct(table)
- error('Malformed file, possibly empty or lacking data entries')
-end
-
-colheaders = table.textdata;
-if length(colheaders) == 1
- colheaders = regexp(table.textdata{1}, ',', 'split');
-end
-
-parsed = struct;
-i = 1;
-while i <= length(colheaders)
- % Checking for a multi-value column.
- m = regexp(colheaders{i}, '([\w\t]+)\[(\d+)\]', 'tokens');
- if ~isempty(m)
- % Parse a multi-value column
- n = str2double(m{1}{2}) - 1;
- parsed.(strrep(m{1}{1}, ' ', '_')) = table.data(:, i:i+n);
- i = i + n + 1;
- elseif ~isempty(colheaders{i})
- % Parse a single-value column
- parsed.(strrep(colheaders{i}, ' ', '_')) = table.data(:, i);
- i = i + 1;
- else
- error('Empty column');
- end
-end
diff --git a/trunk/tools/python_charts/OWNERS b/trunk/tools/python_charts/OWNERS
deleted file mode 100644
index 0428a4a..0000000
--- a/trunk/tools/python_charts/OWNERS
+++ /dev/null
@@ -1 +0,0 @@
-kjellander@webrtc.org
diff --git a/trunk/tools/python_charts/README b/trunk/tools/python_charts/README
deleted file mode 100644
index 483c402..0000000
--- a/trunk/tools/python_charts/README
+++ /dev/null
@@ -1,41 +0,0 @@
-This file describes how to setup Eclipse and then the Python Charts project
-
-Setup Eclipse
--------------
-These instructions were tested on Linux, but are very similar for Windows and
-Mac.
-1. Ensure you have Python 2.x installed
-2. Download and install Google App Engine SDK for Python from
- http://code.google.com/appengine/downloads.html
-3. Note which location you put App Engine in, as this will be needed later on.
-4. Download Eclipse from http://www.eclipse.org. Any distribution will probably
- do, but if you're going to do mainly web development, you might pick Eclipse
- IDE for JavaScript Web Developers
-5. Install the PyDev plugin using the Eclipse update site mentioned at
- http://pydev.org/download.html
-6. Install the Google Plugin for Eclipse: http://code.google.com/eclipse/
-
-Setup the project
------------------
-Generic instructions are available at
-http://code.google.com/appengine/docs/python/gettingstarted/ but the following
-should be enough:
-1. Launch Eclipse and create a workspace
-2. Create a new PyDev Project
-3. In the PyDev Project wizard, uncheck the "Use Default" checkbox for Project
- contents and browse to your tools/python_charts directory.
-4. Enter a project name. We'll assume PythonCharts in the examples below.
-5. In the radio button of the lower part of the window, select
- "Add project directory to the PYTHONPATH"
-6. Click Finish
-7. Select the Run > Run Configuration… menu item
-8. Create a new "Python Run" configuration
-9. Select your Python Charts project as project
-10. As Main Module, enter the path to your dev_appserver.py, which is a part
- of your App Engine installation,
- e.g. /usr/local/google_appengine/dev_appserver.py
-11. At the Arguments tab, enter the location of your project root.
- Using Eclipse variables if your project name is PythonCharts:
- ${workspace_loc:PythonCharts}
-12. Launch the development app server by clicking the Run button.
-13. Launch a browser and go to http://localhost:8080
diff --git a/trunk/tools/python_charts/app.yaml b/trunk/tools/python_charts/app.yaml
deleted file mode 100644
index ace1b51..0000000
--- a/trunk/tools/python_charts/app.yaml
+++ /dev/null
@@ -1,9 +0,0 @@
-application: webrtc-python-charts
-version: 1
-runtime: python
-api_version: 1
-
-handlers:
-
-- url: /*
- script: webrtc/main.py
\ No newline at end of file
diff --git a/trunk/tools/python_charts/data/vp8_hw.py b/trunk/tools/python_charts/data/vp8_hw.py
deleted file mode 100644
index b8c6cc0..0000000
--- a/trunk/tools/python_charts/data/vp8_hw.py
+++ /dev/null
@@ -1,49 +0,0 @@
-# Sample output from the video_quality_measurment program, included only for
-# reference. Geneate your own by running with the --python flag and then change
-# the filenames in main.py
-test_configuration = [{'name': 'name', 'value': 'VP8 hardware test'},
-{'name': 'description', 'value': ''},
-{'name': 'test_number', 'value': '0'},
-{'name': 'input_filename', 'value': 'foreman_cif.yuv'},
-{'name': 'output_filename', 'value': 'foreman_cif_out.yuv'},
-{'name': 'output_dir', 'value': '.'},
-{'name': 'packet_size_in_bytes', 'value': '1500'},
-{'name': 'max_payload_size_in_bytes', 'value': '1440'},
-{'name': 'packet_loss_mode', 'value': 'Uniform'},
-{'name': 'packet_loss_probability', 'value': '0.000000'},
-{'name': 'packet_loss_burst_length', 'value': '1'},
-{'name': 'exclude_frame_types', 'value': 'ExcludeOnlyFirstKeyFrame'},
-{'name': 'frame_length_in_bytes', 'value': '152064'},
-{'name': 'use_single_core', 'value': 'False'},
-{'name': 'keyframe_interval;', 'value': '0'},
-{'name': 'video_codec_type', 'value': 'VP8'},
-{'name': 'width', 'value': '352'},
-{'name': 'height', 'value': '288'},
-{'name': 'bit_rate_in_kbps', 'value': '500'},
-]
-frame_data_types = {'frame_number': ('number', 'Frame number'),
-'encoding_successful': ('boolean', 'Encoding successful?'),
-'decoding_successful': ('boolean', 'Decoding successful?'),
-'encode_time': ('number', 'Encode time (us)'),
-'decode_time': ('number', 'Decode time (us)'),
-'encode_return_code': ('number', 'Encode return code'),
-'decode_return_code': ('number', 'Decode return code'),
-'bit_rate': ('number', 'Bit rate (kbps)'),
-'encoded_frame_length': ('number', 'Encoded frame length (bytes)'),
-'frame_type': ('string', 'Frame type'),
-'packets_dropped': ('number', 'Packets dropped'),
-'total_packets': ('number', 'Total packets'),
-'ssim': ('number', 'SSIM'),
-'psnr': ('number', 'PSNR (dB)'),
-}
-frame_data = [{'frame_number': 0, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 94676, 'decode_time': 37942, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 1098, 'encoded_frame_length': 4579, 'frame_type': 'Other', 'packets_dropped': 0, 'total_packets': 4, 'ssim': 0.910364, 'psnr': 35.067258},
-{'frame_number': 1, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 244007, 'decode_time': 39421, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 306, 'encoded_frame_length': 1277, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.911859, 'psnr': 35.115193},
-{'frame_number': 2, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 240508, 'decode_time': 38918, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 330, 'encoded_frame_length': 1379, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.913597, 'psnr': 35.181604},
-{'frame_number': 3, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 243449, 'decode_time': 39664, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 298, 'encoded_frame_length': 1242, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.912378, 'psnr': 35.164710},
-{'frame_number': 4, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 248024, 'decode_time': 39115, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 332, 'encoded_frame_length': 1385, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.911471, 'psnr': 35.109488},
-{'frame_number': 5, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 246910, 'decode_time': 39146, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 416, 'encoded_frame_length': 1734, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 2, 'ssim': 0.915231, 'psnr': 35.392300},
-{'frame_number': 6, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 242953, 'decode_time': 38827, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 279, 'encoded_frame_length': 1165, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.916130, 'psnr': 35.452889},
-{'frame_number': 7, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 247343, 'decode_time': 41429, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 393, 'encoded_frame_length': 1639, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 2, 'ssim': 0.919356, 'psnr': 35.647128},
-{'frame_number': 8, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 249529, 'decode_time': 40329, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 487, 'encoded_frame_length': 2033, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 2, 'ssim': 0.924705, 'psnr': 36.179837},
-{'frame_number': 9, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 249408, 'decode_time': 41716, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 583, 'encoded_frame_length': 2433, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 2, 'ssim': 0.928433, 'psnr': 36.589875},
-]
diff --git a/trunk/tools/python_charts/data/vp8_sw.py b/trunk/tools/python_charts/data/vp8_sw.py
deleted file mode 100644
index 0f29137..0000000
--- a/trunk/tools/python_charts/data/vp8_sw.py
+++ /dev/null
@@ -1,49 +0,0 @@
-# Sample output from the video_quality_measurment program, included only for
-# reference. Geneate your own by running with the --python flag and then change
-# the filenames in main.py
-test_configuration = [{'name': 'name', 'value': 'VP8 software test'},
-{'name': 'description', 'value': ''},
-{'name': 'test_number', 'value': '0'},
-{'name': 'input_filename', 'value': 'foreman_cif.yuv'},
-{'name': 'output_filename', 'value': 'foreman_cif_out.yuv'},
-{'name': 'output_dir', 'value': '.'},
-{'name': 'packet_size_in_bytes', 'value': '1500'},
-{'name': 'max_payload_size_in_bytes', 'value': '1440'},
-{'name': 'packet_loss_mode', 'value': 'Uniform'},
-{'name': 'packet_loss_probability', 'value': '0.000000'},
-{'name': 'packet_loss_burst_length', 'value': '1'},
-{'name': 'exclude_frame_types', 'value': 'ExcludeOnlyFirstKeyFrame'},
-{'name': 'frame_length_in_bytes', 'value': '152064'},
-{'name': 'use_single_core', 'value': 'False'},
-{'name': 'keyframe_interval;', 'value': '0'},
-{'name': 'video_codec_type', 'value': 'VP8'},
-{'name': 'width', 'value': '352'},
-{'name': 'height', 'value': '288'},
-{'name': 'bit_rate_in_kbps', 'value': '500'},
-]
-frame_data_types = {'frame_number': ('number', 'Frame number'),
-'encoding_successful': ('boolean', 'Encoding successful?'),
-'decoding_successful': ('boolean', 'Decoding successful?'),
-'encode_time': ('number', 'Encode time (us)'),
-'decode_time': ('number', 'Decode time (us)'),
-'encode_return_code': ('number', 'Encode return code'),
-'decode_return_code': ('number', 'Decode return code'),
-'bit_rate': ('number', 'Bit rate (kbps)'),
-'encoded_frame_length': ('number', 'Encoded frame length (bytes)'),
-'frame_type': ('string', 'Frame type'),
-'packets_dropped': ('number', 'Packets dropped'),
-'total_packets': ('number', 'Total packets'),
-'ssim': ('number', 'SSIM'),
-'psnr': ('number', 'PSNR (dB)'),
-}
-frame_data = [{'frame_number': 0, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 12427, 'decode_time': 4403, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 2270, 'encoded_frame_length': 9459, 'frame_type': 'Other', 'packets_dropped': 0, 'total_packets': 7, 'ssim': 0.947050, 'psnr': 38.332820},
-{'frame_number': 1, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 3292, 'decode_time': 821, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 88, 'encoded_frame_length': 368, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.927272, 'psnr': 35.883510},
-{'frame_number': 2, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 4295, 'decode_time': 902, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 130, 'encoded_frame_length': 544, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.920539, 'psnr': 35.457107},
-{'frame_number': 3, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 3880, 'decode_time': 767, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 171, 'encoded_frame_length': 714, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.917434, 'psnr': 35.389298},
-{'frame_number': 4, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 4471, 'decode_time': 909, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 248, 'encoded_frame_length': 1035, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.918892, 'psnr': 35.570229},
-{'frame_number': 5, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 4447, 'decode_time': 976, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 269, 'encoded_frame_length': 1123, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.920609, 'psnr': 35.769663},
-{'frame_number': 6, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 4432, 'decode_time': 891, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 271, 'encoded_frame_length': 1132, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 1, 'ssim': 0.922672, 'psnr': 35.913519},
-{'frame_number': 7, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 5026, 'decode_time': 1068, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 366, 'encoded_frame_length': 1529, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 2, 'ssim': 0.925505, 'psnr': 36.246713},
-{'frame_number': 8, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 4877, 'decode_time': 1051, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 369, 'encoded_frame_length': 1538, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 2, 'ssim': 0.926122, 'psnr': 36.305984},
-{'frame_number': 9, 'encoding_successful': True , 'decoding_successful': True , 'encode_time': 4712, 'decode_time': 1087, 'encode_return_code': 0, 'decode_return_code': 0, 'bit_rate': 406, 'encoded_frame_length': 1692, 'frame_type': 'Delta', 'packets_dropped': 0, 'total_packets': 2, 'ssim': 0.927183, 'psnr': 36.379735},
-]
diff --git a/trunk/tools/python_charts/gviz_api.py b/trunk/tools/python_charts/gviz_api.py
deleted file mode 120000
index c9dca90..0000000
--- a/trunk/tools/python_charts/gviz_api.py
+++ /dev/null
@@ -1 +0,0 @@
-../../third_party/google-visualization-python/gviz_api.py
\ No newline at end of file
diff --git a/trunk/tools/python_charts/templates/chart_page_template.html b/trunk/tools/python_charts/templates/chart_page_template.html
deleted file mode 100644
index 1cb3951..0000000
--- a/trunk/tools/python_charts/templates/chart_page_template.html
+++ /dev/null
@@ -1,90 +0,0 @@
-<html>
- <!--
- Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-
- Use of this source code is governed by a BSD-style license
- that can be found in the LICENSE file in the root of the source
- tree. An additional intellectual property rights grant can be found
- in the file PATENTS. All contributing project authors may
- be found in the AUTHORS file in the root of the source tree.
-
- Template file to be used to generate Charts for Video Quality Metrics.
- -->
- <head>
- <link href="http://code.google.com/css/codesite.pack.04102009.css"
- rel="stylesheet" type="text/css" />
- </head>
- <script src="https://www.google.com/jsapi" type="text/javascript"></script>
- <script>
- google.load('visualization', '1', {packages:['table', 'corechart']});
-
- google.setOnLoadCallback(drawTable);
- function drawTable() {
- /* Build data tables and views */
- var configurations_data_table =
- new google.visualization.DataTable(%(json_configurations)s);
- var ssim_data_table =
- new google.visualization.DataTable(%(json_ssim_data)s);
- var psnr_data_table =
- new google.visualization.DataTable(%(json_psnr_data)s);
- var packet_loss_data_table =
- new google.visualization.DataTable(%(json_packet_loss_data)s);
- var bit_rate_data_table =
- new google.visualization.DataTable(%(json_bit_rate_data)s);
-
- /* Display tables and charts */
- var configurations_table = new google.visualization.Table(
- document.getElementById('table_div_configurations'));
- configurations_table.draw(configurations_data_table, {
- height: 200
- });
-
- var ssim_chart = new google.visualization.LineChart(
- document.getElementById('table_div_ssim'));
- ssim_chart.draw(ssim_data_table, {
- colors: ['blue', 'red', 'lightblue', 'pink'],
- vAxis: {title: 'SSIM'},
- hAxis: {title: 'Frame'},
- width: 1200, height: 300,
- });
-
- var psnr_chart = new google.visualization.LineChart(
- document.getElementById('table_div_psnr'));
- psnr_chart.draw(psnr_data_table, {
- colors: ['blue', 'red', 'lightblue', 'pink'],
- vAxis: {title: 'PSNR (dB)'},
- hAxis: {title: 'Frame'},
- width: 1200, height: 300,
- });
-
- var packet_loss_chart = new google.visualization.LineChart(
- document.getElementById('table_div_packet_loss'));
- packet_loss_chart.draw(packet_loss_data_table, {
- colors: ['blue', 'red', 'lightblue', 'pink'],
- vAxis: {title: 'Packets dropped'},
- hAxis: {title: 'Frame'},
- width: 1200, height: 300,
- });
-
- var bit_rate_chart = new google.visualization.LineChart(
- document.getElementById('table_div_bit_rate'));
- bit_rate_chart.draw(bit_rate_data_table, {
- colors: ['blue', 'red', 'lightblue', 'pink', 'green'],
- vAxis: {title: 'Bit rate'},
- hAxis: {title: 'Frame'},
- width: 1200, height: 300,
- });
- }
- </script>
- <body>
- <h3>Test Configurations:</h3>
- <div id="table_div_configurations"></div>
- <h3>Messages:</h3>
- <pre>%(messages)s</pre>
- <h3>Metrics measured per frame:</h3>
- <div id="table_div_ssim"></div>
- <div id="table_div_psnr"></div>
- <div id="table_div_packet_loss"></div>
- <div id="table_div_bit_rate"></div>
- </body>
-</html>
diff --git a/trunk/tools/python_charts/webrtc/__init__.py b/trunk/tools/python_charts/webrtc/__init__.py
deleted file mode 100644
index c1caaa2..0000000
--- a/trunk/tools/python_charts/webrtc/__init__.py
+++ /dev/null
@@ -1,8 +0,0 @@
-#!/usr/bin/env python
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
\ No newline at end of file
diff --git a/trunk/tools/python_charts/webrtc/data_helper.py b/trunk/tools/python_charts/webrtc/data_helper.py
deleted file mode 100644
index fce949f..0000000
--- a/trunk/tools/python_charts/webrtc/data_helper.py
+++ /dev/null
@@ -1,183 +0,0 @@
-#!/usr/bin/env python
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-__author__ = 'kjellander@webrtc.org (Henrik Kjellander)'
-
-class DataHelper(object):
- """
- Helper class for managing table data.
- This class does not verify the consistency of the data tables sent into it.
- """
-
- def __init__(self, data_list, table_description, names_list, messages):
- """ Initializes the DataHelper with data.
-
- Args:
- data_list: List of one or more data lists in the format that the
- Google Visualization Python API expects (list of dictionaries, one
- per row of data). See the gviz_api.DataTable documentation for more
- info.
- table_description: dictionary describing the data types of all
- columns in the data lists, as defined in the gviz_api.DataTable
- documentation.
- names_list: List of strings of what we're going to name the data
- columns after. Usually different runs of data collection.
- messages: List of strings we might append error messages to.
- """
- self.data_list = data_list
- self.table_description = table_description
- self.names_list = names_list
- self.messages = messages
- self.number_of_datasets = len(data_list)
- self.number_of_frames = len(data_list[0])
-
- def CreateData(self, field_name, start_frame=0, end_frame=0):
- """ Creates a data structure for a specified data field.
-
- Creates a data structure (data type description dictionary and a list
- of data dictionaries) to be used with the Google Visualization Python
- API. The frame_number column is always present and one column per data
- set is added and its field name is suffixed by _N where N is the number
- of the data set (0, 1, 2...)
-
- Args:
- field_name: String name of the field, must be present in the data
- structure this DataHelper was created with.
- start_frame: Frame number to start at (zero indexed). Default: 0.
- end_frame: Frame number to be the last frame. If zero all frames
- will be included. Default: 0.
-
- Returns:
- A tuple containing:
- - a dictionary describing the columns in the data result_data_table below.
- This description uses the name for each data set specified by
- names_list.
-
- Example with two data sets named 'Foreman' and 'Crew':
- {
- 'frame_number': ('number', 'Frame number'),
- 'ssim_0': ('number', 'Foreman'),
- 'ssim_1': ('number', 'Crew'),
- }
- - a list containing dictionaries (one per row) with the frame_number
- column and one column of the specified field_name column per data
- set.
-
- Example with two data sets named 'Foreman' and 'Crew':
- [
- {'frame_number': 0, 'ssim_0': 0.98, 'ssim_1': 0.77 },
- {'frame_number': 1, 'ssim_0': 0.81, 'ssim_1': 0.53 },
- ]
- """
-
- # Build dictionary that describes the data types
- result_table_description = {'frame_number': ('string', 'Frame number')}
- for dataset_index in range(self.number_of_datasets):
- column_name = '%s_%s' % (field_name, dataset_index)
- column_type = self.table_description[field_name][0]
- column_description = self.names_list[dataset_index]
- result_table_description[column_name] = (column_type, column_description)
-
- # Build data table of all the data
- result_data_table = []
- # We're going to have one dictionary per row.
- # Create that and copy frame_number values from the first data set
- for source_row in self.data_list[0]:
- row_dict = { 'frame_number': source_row['frame_number'] }
- result_data_table.append(row_dict)
-
- # Pick target field data points from the all data tables
- if end_frame == 0: # Default to all frames
- end_frame = self.number_of_frames
-
- for dataset_index in range(self.number_of_datasets):
- for row_number in range(start_frame, end_frame):
- column_name = '%s_%s' % (field_name, dataset_index)
- # Stop if any of the data sets are missing the frame
- try:
- result_data_table[row_number][column_name] = \
- self.data_list[dataset_index][row_number][field_name]
- except IndexError:
- self.messages.append("Couldn't find frame data for row %d "
- "for %s" % (row_number, self.names_list[dataset_index]))
- break
- return result_table_description, result_data_table
-
- def GetOrdering(self, table_description):
- """ Creates a list of column names, ordered alphabetically except for the
- frame_number column which always will be the first column.
-
- Args:
- table_description: A dictionary of column definitions as defined by the
- gviz_api.DataTable documentation.
- Returns:
- A list of column names, where frame_number is the first and the
- remaining columns are sorted alphabetically.
- """
- # The JSON data representation generated from gviz_api.DataTable.ToJSon()
- # must have frame_number as its first column in order for the chart to
- # use it as it's X-axis value series.
- # gviz_api.DataTable orders the columns by name by default, which will
- # be incorrect if we have column names that are sorted before frame_number
- # in our data table.
- columns_ordering = ['frame_number']
- # add all other columns:
- for column in sorted(table_description.keys()):
- if column != 'frame_number':
- columns_ordering.append(column)
- return columns_ordering
-
- def CreateConfigurationTable(self, configurations):
- """ Combines multiple test data configurations for display.
-
- Args:
- configurations: List of one ore more configurations. Each configuration
- is required to be a list of dictionaries with two keys: 'name' and
- 'value'.
- Example of a single configuration:
- [
- {'name': 'name', 'value': 'VP8 software'},
- {'name': 'test_number', 'value': '0'},
- {'name': 'input_filename', 'value': 'foreman_cif.yuv'},
- ]
- Returns:
- A tuple containing:
- - a dictionary describing the columns in the configuration table to be
- displayed. All columns will have string as data type.
- Example:
- {
- 'name': 'string',
- 'test_number': 'string',
- 'input_filename': 'string',
- }
- - a list containing dictionaries (one per configuration) with the
- configuration column names mapped to the value for each test run:
-
- Example matching the columns above:
- [
- {'name': 'VP8 software',
- 'test_number': '12',
- 'input_filename': 'foreman_cif.yuv' },
- {'name': 'VP8 hardware',
- 'test_number': '5',
- 'input_filename': 'foreman_cif.yuv' },
- ]
- """
- result_description = {}
- result_data = []
-
- for configuration in configurations:
- data = {}
- result_data.append(data)
- for dict in configuration:
- name = dict['name']
- value = dict['value']
- result_description[name] = 'string'
- data[name] = value
- return result_description, result_data
diff --git a/trunk/tools/python_charts/webrtc/data_helper_test.py b/trunk/tools/python_charts/webrtc/data_helper_test.py
deleted file mode 100644
index 6282f7b..0000000
--- a/trunk/tools/python_charts/webrtc/data_helper_test.py
+++ /dev/null
@@ -1,115 +0,0 @@
-#!/usr/bin/env python
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-__author__ = 'kjellander@webrtc.org (Henrik Kjellander)'
-
-import unittest
-import webrtc.data_helper
-
-class Test(unittest.TestCase):
-
- def setUp(self):
- # Simulate frame data from two different test runs, with 2 frames each.
- self.frame_data_0 = [{'frame_number': 0, 'ssim': 0.5, 'psnr': 30.5},
- {'frame_number': 1, 'ssim': 0.55, 'psnr': 30.55}]
- self.frame_data_1 = [{'frame_number': 0, 'ssim': 0.6, 'psnr': 30.6},
- {'frame_number': 0, 'ssim': 0.66, 'psnr': 30.66}]
- self.all_data = [ self.frame_data_0, self.frame_data_1 ]
-
- # Test with frame_number column in a non-first position sice we need to
- # support reordering that to be able to use the gviz_api as we want.
- self.type_description = {
- 'ssim': ('number', 'SSIM'),
- 'frame_number': ('number', 'Frame number'),
- 'psnr': ('number', 'PSRN'),
- }
- self.names = ["Test 0", "Test 1"]
- self.configurations = [
- [{'name': 'name', 'value': 'Test 0'},
- {'name': 'test_number', 'value': '13'},
- {'name': 'input_filename', 'value': 'foreman_cif.yuv'},
- ],
- [{'name': 'name', 'value': 'Test 1'},
- {'name': 'test_number', 'value': '5'},
- {'name': 'input_filename', 'value': 'foreman_cif.yuv'},
- ],
- ]
-
- def testCreateData(self):
- messages = []
- helper = webrtc.data_helper.DataHelper(self.all_data, self.type_description,
- self.names, messages)
- description, data_table = helper.CreateData('ssim')
- self.assertEqual(3, len(description))
- self.assertTrue('frame_number' in description)
- self.assertTrue('ssim_0' in description)
- self.assertTrue('number' in description['ssim_0'][0])
- self.assertTrue('Test 0' in description['ssim_0'][1])
- self.assertTrue('ssim_1' in description)
- self.assertTrue('number' in description['ssim_1'][0])
- self.assertTrue('Test 1' in description['ssim_1'][1])
-
- self.assertEqual(0, len(messages))
-
- self.assertEquals(2, len(data_table))
- row = data_table[0]
- self.assertEquals(0, row['frame_number'])
- self.assertEquals(0.5, row['ssim_0'])
- self.assertEquals(0.6, row['ssim_1'])
- row = data_table[1]
- self.assertEquals(1, row['frame_number'])
- self.assertEquals(0.55, row['ssim_0'])
- self.assertEquals(0.66, row['ssim_1'])
-
- description, data_table = helper.CreateData('psnr')
- self.assertEqual(3, len(description))
- self.assertTrue('frame_number' in description)
- self.assertTrue('psnr_0' in description)
- self.assertTrue('psnr_1' in description)
- self.assertEqual(0, len(messages))
-
- self.assertEquals(2, len(data_table))
- row = data_table[0]
- self.assertEquals(0, row['frame_number'])
- self.assertEquals(30.5, row['psnr_0'])
- self.assertEquals(30.6, row['psnr_1'])
- row = data_table[1]
- self.assertEquals(1, row['frame_number'])
- self.assertEquals(30.55, row['psnr_0'])
- self.assertEquals(30.66, row['psnr_1'])
-
- def testGetOrdering(self):
- """ Tests that the ordering help method returns a list with frame_number
- first and the rest sorted alphabetically """
- messages = []
- helper = webrtc.data_helper.DataHelper(self.all_data, self.type_description,
- self.names, messages)
- description, data_table = helper.CreateData('ssim')
- columns = helper.GetOrdering(description)
- self.assertEqual(3, len(columns))
- self.assertEqual(0, len(messages))
- self.assertEqual('frame_number', columns[0])
- self.assertEqual('ssim_0', columns[1])
- self.assertEqual('ssim_1', columns[2])
-
- def testCreateConfigurationTable(self):
- messages = []
- helper = webrtc.data_helper.DataHelper(self.all_data, self.type_description,
- self.names, messages)
- description, data = helper.CreateConfigurationTable(self.configurations)
- self.assertEqual(3, len(description)) # 3 columns
- self.assertEqual(2, len(data)) # 2 data sets
- self.assertTrue(description.has_key('name'))
- self.assertTrue(description.has_key('test_number'))
- self.assertTrue(description.has_key('input_filename'))
- self.assertEquals('Test 0', data[0]['name'])
- self.assertEquals('Test 1', data[1]['name'])
-
-if __name__ == "__main__":
- unittest.main()
diff --git a/trunk/tools/python_charts/webrtc/main.py b/trunk/tools/python_charts/webrtc/main.py
deleted file mode 100644
index 82d8831..0000000
--- a/trunk/tools/python_charts/webrtc/main.py
+++ /dev/null
@@ -1,157 +0,0 @@
-#!/usr/bin/env python
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-__author__ = 'kjellander@webrtc.org (Henrik Kjellander)'
-
-import os
-import gviz_api
-import webrtc.data_helper
-
-def main():
- """
- This Python script displays a web page with test created with the
- video_quality_measurement program, which is a tool in WebRTC.
-
- The script requires on two external files and one Python library:
- - A HTML template file with layout and references to the json variables
- defined in this script
- - A data file in Python format, containing the following:
- - test_configuration - a dictionary of test configuration names and values.
- - frame_data_types - a dictionary that maps the different metrics to their
- data types.
- - frame_data - a list of dictionaries where each dictionary maps a metric to
- it's value.
- - The gviz_api.py of the Google Visualization Python API, available at
- http://code.google.com/p/google-visualization-python/
-
- The HTML file is shipped with the script, while the data file must be
- generated by running video_quality_measurement with the --python flag
- specified.
- """
- print 'Content-type: text/html\n' # the newline is required!
-
- page_template_filename = '../templates/chart_page_template.html'
- # The data files must be located in the project tree for app engine being
- # able to access them.
- data_filenames = [ '../data/vp8_sw.py', '../data/vp8_hw.py' ]
- # Will contain info/error messages to be displayed on the resulting page.
- messages = []
- # Load the page HTML template.
- try:
- f = open(page_template_filename)
- page_template = f.read()
- f.close()
- except IOError as e:
- ShowErrorPage('Cannot open page template file: %s<br>Details: %s' %
- (page_template_filename, e))
- return
-
- # Read data from external Python script files. First check that they exist.
- for filename in data_filenames:
- if not os.path.exists(filename):
- messages.append('Cannot open data file: %s' % filename)
- data_filenames.remove(filename)
-
- # Read data from all existing input files.
- data_list = []
- test_configurations = []
- names = []
-
- for filename in data_filenames:
- read_vars = {} # empty dictionary to load the data into.
- execfile(filename, read_vars, read_vars)
-
- test_configuration = read_vars['test_configuration']
- table_description = read_vars['frame_data_types']
- table_data = read_vars['frame_data']
-
- # Verify the data in the file loaded properly.
- if not table_description or not table_data:
- messages.append('Invalid input file: %s. Missing description list or '
- 'data dictionary variables.' % filename)
- continue
-
- # Frame numbers appear as number type in the data, but Chart API requires
- # values of the X-axis to be of string type.
- # Change the frame_number column data type:
- table_description['frame_number'] = ('string', 'Frame number')
- # Convert all the values to string types:
- for row in table_data:
- row['frame_number'] = str(row['frame_number'])
-
- # Store the unique data from this file in the high level lists.
- test_configurations.append(test_configuration)
- data_list.append(table_data)
- # Name of the test run must be present.
- test_name = FindConfiguration(test_configuration, 'name')
- if not test_name:
- messages.append('Invalid input file: %s. Missing configuration key '
- '"name"', filename)
- continue
- names.append(test_name)
-
- # Create data helper and build data tables for each graph.
- helper = webrtc.data_helper.DataHelper(data_list, table_description,
- names, messages)
-
- # Loading it into gviz_api.DataTable objects and create JSON strings.
- description, data = helper.CreateConfigurationTable(test_configurations)
- configurations = gviz_api.DataTable(description, data)
- json_configurations = configurations.ToJSon()
-
- description, data = helper.CreateData('ssim')
- ssim = gviz_api.DataTable(description, data)
- json_ssim_data = ssim.ToJSon(helper.GetOrdering(description))
-
- description, data = helper.CreateData('psnr')
- psnr = gviz_api.DataTable(description, data)
- json_psnr_data = psnr.ToJSon(helper.GetOrdering(description))
-
- description, data = helper.CreateData('packets_dropped')
- packet_loss = gviz_api.DataTable(description, data)
- json_packet_loss_data = packet_loss.ToJSon(helper.GetOrdering(description))
-
- description, data = helper.CreateData('bit_rate')
- # Add a column of data points for the desired bit rate to be plotted.
- # (uses test configuration from the last data set, assuming it is the same
- # for all of them)
- desired_bit_rate = FindConfiguration(test_configuration, 'bit_rate_in_kbps')
- if not desired_bit_rate:
- ShowErrorPage('Cannot configuration field named "bit_rate_in_kbps"')
- return
- desired_bit_rate = int(desired_bit_rate)
- # Add new column data type description.
- description['desired_bit_rate'] = ('number', 'Desired bit rate (kbps)')
- for row in data:
- row['desired_bit_rate'] = desired_bit_rate
- bit_rate = gviz_api.DataTable(description, data)
- json_bit_rate_data = bit_rate.ToJSon(helper.GetOrdering(description))
-
- # Format the messages list with newlines.
- messages = '\n'.join(messages)
-
- # Put the variables as JSon strings into the template.
- print page_template % vars()
-
-def FindConfiguration(configuration, name):
- """ Finds a configuration value using it's name.
- Returns the first configuration with a matching name. Returns None if no
- matching configuration is found. """
- return_value = None
- for row in configuration:
- if row['name'] == name:
- return_value = row['value']
- break
- return return_value
-
-def ShowErrorPage(error_message):
- print '<html><body>%s</body></html>' % error_message
-
-if __name__ == '__main__':
- main()
diff --git a/trunk/tools/quality_tracking/OWNERS b/trunk/tools/quality_tracking/OWNERS
deleted file mode 100644
index 323e8e7..0000000
--- a/trunk/tools/quality_tracking/OWNERS
+++ /dev/null
@@ -1,2 +0,0 @@
-phoglund@webrtc.org
-kjellander@webrtc.org
diff --git a/trunk/tools/quality_tracking/README b/trunk/tools/quality_tracking/README
deleted file mode 100644
index faf3e7a..0000000
--- a/trunk/tools/quality_tracking/README
+++ /dev/null
@@ -1,31 +0,0 @@
-This file describes the coverage tracking script and the coverage dashboard.
-
-ABSTRACT:
-The intention of this small tracking system is to track code coverage data
-over time. Since code coverage is continuously recomputed on the build bots,
-the track_coverage.py script is intended to run on the build bot as a cron job
-and extract the data from there. The dashboard doesn't care how often this
-script runs, but running each hour should be more than enough.
-
-The track_coverage.py script uses OAuth to authenticate itself. In order to do
-this, it needs two files: consumer.secret and access.token. The consumer secret
-is known within the organization and is stored in a plain file on the bot
-running the scripts (we don't want to check in this secret in the code in the
-public repository). The consumer secret is a plain file with a single line
-containing the secret string.
-
-The access.token file is generated by request_oauth_permission.py. It does this
-by going through the three-legged OAuth authorization process. An administrator
-of the dashboard must approve the request from the script. Once that is done,
-access.token will be written and track_coverage.py will be able to report
-results.
-
-HOW TO RUN LOCALLY:
-Follow the following instructions:
-http://code.google.com/appengine/docs/python/gettingstartedpython27/devenvironment.html
-The dashboard can be started on 127.0.0.1:8080 using the dev_appserver.py script
-as described in the above URL (and in the following 'hello world' page).
-
-HOW TO DEPLOY:
-Follow the following instructions:
-http://code.google.com/appengine/docs/python/gettingstartedpython27/uploading.html
\ No newline at end of file
diff --git a/trunk/tools/quality_tracking/constants.py b/trunk/tools/quality_tracking/constants.py
deleted file mode 100644
index c6aac74..0000000
--- a/trunk/tools/quality_tracking/constants.py
+++ /dev/null
@@ -1,37 +0,0 @@
-#!/usr/bin/env python
-#-*- coding: utf-8 -*-
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-"""Contains tweakable constants for quality dashboard utility scripts."""
-
-__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
-
-# This identifies our application using the information we got when we
-# registered the application on Google appengine.
-DASHBOARD_SERVER = 'webrtc-dashboard.appspot.com'
-DASHBOARD_SERVER_HTTP = 'http://' + DASHBOARD_SERVER
-CONSUMER_KEY = DASHBOARD_SERVER
-CONSUMER_SECRET_FILE = 'consumer.secret'
-ACCESS_TOKEN_FILE = 'access.token'
-
-# OAuth URL:s.
-REQUEST_TOKEN_URL = DASHBOARD_SERVER_HTTP + '/_ah/OAuthGetRequestToken'
-AUTHORIZE_TOKEN_URL = DASHBOARD_SERVER_HTTP + '/_ah/OAuthAuthorizeToken'
-ACCESS_TOKEN_URL = DASHBOARD_SERVER_HTTP + '/_ah/OAuthGetAccessToken'
-
-# The build master URL.
-BUILD_MASTER_SERVER = 'webrtc-cb-linux-master.cbf.corp.google.com:8010'
-BUILD_MASTER_TRANSPOSED_GRID_URL = '/tgrid'
-
-# Build bot constants.
-BUILD_BOT_COVERAGE_WWW_DIRECTORY = '/var/www/'
-
-# Dashboard data input URLs.
-ADD_COVERAGE_DATA_URL = DASHBOARD_SERVER_HTTP + '/add_coverage_data'
-ADD_BUILD_STATUS_DATA_URL = DASHBOARD_SERVER_HTTP + '/add_build_status_data'
diff --git a/trunk/tools/quality_tracking/dashboard/add_build_status_data.py b/trunk/tools/quality_tracking/dashboard/add_build_status_data.py
deleted file mode 100644
index 106d178..0000000
--- a/trunk/tools/quality_tracking/dashboard/add_build_status_data.py
+++ /dev/null
@@ -1,171 +0,0 @@
-#!/usr/bin/env python
-#-*- coding: utf-8 -*-
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-"""Implements a handler for adding build status data."""
-
-__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
-
-import datetime
-import logging
-
-from google.appengine.ext import db
-
-import oauth_post_request_handler
-
-VALID_STATUSES = ['OK', 'failed', 'building', 'warnings']
-
-
-class OrphanedBuildStatusesExistException(Exception):
- pass
-
-
-class BuildStatusRoot(db.Model):
- """Exists solely to be the root parent for all build status data and to keep
- track of when the last update was made.
-
- Since all build status data will refer to this as their parent,
- we can run transactions on the build status data as a whole.
- """
- last_updated_at = db.DateTimeProperty()
-
-
-class BuildStatusData(db.Model):
- """This represents one build status report from the build bot."""
- bot_name = db.StringProperty(required=True)
- revision = db.IntegerProperty(required=True)
- build_number = db.IntegerProperty(required=True)
- status = db.StringProperty(required=True)
-
-
-def _ensure_build_status_root_exists():
- root = db.GqlQuery('SELECT * FROM BuildStatusRoot').get()
- if not root:
- # Create a new root, but ensure we don't have any orphaned build statuses
- # (in that case, we would not have a single entity group as we desire).
- orphans = db.GqlQuery('SELECT * FROM BuildStatusData').get()
- if orphans:
- raise OrphanedBuildStatusesExistException('Parent is gone and there are '
- 'orphaned build statuses in '
- 'the database!')
- root = BuildStatusRoot()
- root.put()
-
- return root
-
-
-def _filter_oauth_parameters(post_keys):
- return filter(lambda post_key: not post_key.startswith('oauth_'),
- post_keys)
-
-
-def _parse_status(build_number_and_status):
- parsed_status = build_number_and_status.split('--')
- if len(parsed_status) != 2:
- raise ValueError('Malformed status string %s.' % build_number_and_status)
-
- parsed_build_number = int(parsed_status[0])
- status = parsed_status[1]
-
- if status not in VALID_STATUSES:
- raise ValueError('Invalid status in %s.' % build_number_and_status)
-
- return (parsed_build_number, status)
-
-
-def _parse_name(revision_and_bot_name):
- parsed_name = revision_and_bot_name.split('--')
- if len(parsed_name) != 2:
- raise ValueError('Malformed name string %s.' % revision_and_bot_name)
-
- revision = parsed_name[0]
- bot_name = parsed_name[1]
- if '\n' in bot_name:
- raise ValueError('Bot name %s can not contain newlines.' % bot_name)
-
- return (int(revision), bot_name)
-
-
-def _delete_all_with_revision(revision, build_status_root):
- query_result = db.GqlQuery('SELECT * FROM BuildStatusData '
- 'WHERE revision = :1 AND ANCESTOR IS :2',
- revision, build_status_root)
- for entry in query_result:
- entry.delete()
-
-
-class AddBuildStatusData(oauth_post_request_handler.OAuthPostRequestHandler):
- """Used to report build status data.
-
- Build status data is reported as a POST request. The POST request, aside
- from the required oauth_* parameters should contain name-value entries that
- abide by the following rules:
-
- 1) The name should be on the form <revision>--<bot name>, for instance
- 1568--Win32Release.
- 2) The value should be on the form <build number>--<status>, for instance
- 553--OK, 554--building. The status is permitted to be failed, OK or
- building.
-
- Data is keyed by revision. This handler will delete all data from a revision
- if data with that revision is present in the current update, since we
- assume that more recent data is always better data. We also assume that
- an update always has complete information on a revision (e.g. the status
- for all the bots are reported in each update).
-
- In particular the revision arrangement solves the problem when the latest
- revision reports 'building' for a bot. Had we not deleted the old revision
- we would first store a 'building' status for that bot and revision, and
- later store a 'OK' or 'failed' status for that bot and revision. This is
- undesirable since we don't want multiple statuses for one bot-revision
- combination. Now we will effectively update the bot's status instead.
- """
-
- def _parse_and_store_data(self):
- build_status_root = _ensure_build_status_root_exists()
- build_status_data = _filter_oauth_parameters(self.request.arguments())
-
- db.run_in_transaction(self._parse_and_store_data_in_transaction,
- build_status_root, build_status_data)
-
- def _parse_and_store_data_in_transaction(self, build_status_root,
- build_status_data):
-
- encountered_revisions = set()
- for revision_and_bot_name in build_status_data:
- build_number_and_status = self.request.get(revision_and_bot_name)
-
- try:
- (build_number, status) = _parse_status(build_number_and_status)
- (revision, bot_name) = _parse_name(revision_and_bot_name)
- except ValueError as error:
- logger.warn('Invalid parameter in request: %s.' % error)
- self.response.set_status(400)
- return
-
- if revision not in encountered_revisions:
- # There's new data on this revision in this update, so clear all status
- # entries with that revision. Only do this once when we first encounter
- # the revision.
- _delete_all_with_revision(revision, build_status_root)
- encountered_revisions.add(revision)
-
- # Finally, write the item.
- item = BuildStatusData(parent=build_status_root,
- bot_name=bot_name,
- revision=revision,
- build_number=build_number,
- status=status)
- item.put()
-
- request_posix_timestamp = float(self.request.get('oauth_timestamp'))
- request_datetime = datetime.datetime.fromtimestamp(request_posix_timestamp)
- build_status_root.last_updated_at = request_datetime
- build_status_root.put()
-
diff --git a/trunk/tools/quality_tracking/dashboard/add_coverage_data.py b/trunk/tools/quality_tracking/dashboard/add_coverage_data.py
deleted file mode 100644
index 1c9d9da..0000000
--- a/trunk/tools/quality_tracking/dashboard/add_coverage_data.py
+++ /dev/null
@@ -1,66 +0,0 @@
-#!/usr/bin/env python
-#-*- coding: utf-8 -*-
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-"""Implements a handler for adding coverage data."""
-
-__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
-
-import datetime
-import logging
-
-from google.appengine.ext import db
-
-import oauth_post_request_handler
-
-class CoverageData(db.Model):
- """This represents one coverage report from the build bot."""
- date = db.DateTimeProperty(required=True)
- line_coverage = db.FloatProperty(required=True)
- function_coverage = db.FloatProperty(required=True)
-
-
-def _parse_percentage(string_value):
- percentage = float(string_value)
- if percentage < 0.0 or percentage > 100.0:
- raise ValueError('%s is not a valid percentage.' % string_value)
- return percentage
-
-
-class AddCoverageData(oauth_post_request_handler.OAuthPostRequestHandler):
- """Used to report coverage data.
-
- Coverage data is reported as a POST request and should contain, aside from
- the regular oauth_* parameters, these values:
-
- date: The POSIX timestamp for when the coverage observation was made.
- line_coverage: A float percentage in the interval 0-100.0.
- function_coverage: A float percentage in the interval 0-100.0.
- """
-
- def _parse_and_store_data(self):
- try:
- posix_time = int(self.request.get('date'))
- parsed_date = datetime.datetime.fromtimestamp(posix_time)
-
- line_coverage_string = self.request.get('line_coverage')
- line_coverage = _parse_percentage(line_coverage_string)
- function_coverage_string = self.request.get('function_coverage')
- function_coverage = _parse_percentage(function_coverage_string)
-
- except ValueError as error:
- logger.warn('Invalid parameter in request: %s.' % error)
- self.response.set_status(400)
- return
-
- item = CoverageData(date=parsed_date,
- line_coverage=line_coverage,
- function_coverage=function_coverage)
- item.put()
-
diff --git a/trunk/tools/quality_tracking/dashboard/app.yaml b/trunk/tools/quality_tracking/dashboard/app.yaml
deleted file mode 100644
index d06a813..0000000
--- a/trunk/tools/quality_tracking/dashboard/app.yaml
+++ /dev/null
@@ -1,24 +0,0 @@
-application: webrtc-dashboard
-version: 1
-runtime: python27
-api_version: 1
-threadsafe: false
-
-handlers:
-# Serve stylesheets statically.
-- url: /stylesheets
- static_dir: stylesheets
-# This magic file is here to prove to the Google Account Domain Management
-# that we own this domain. It needs to stay there so the domain management
-# doesn't get suspicious.
-- url: /google403c95edcde16425.html
- static_files: static/google403c95edcde16425.html
- upload: static/google403c95edcde16425.html
-
-# Note: tests should be disabled in production.
-# - url: /test.*
-# script: gaeunit.py
-
-# Redirect all other requests to our dynamic handlers.
-- url: /.*
- script: dashboard.app
\ No newline at end of file
diff --git a/trunk/tools/quality_tracking/dashboard/dashboard.py b/trunk/tools/quality_tracking/dashboard/dashboard.py
deleted file mode 100644
index 5cc5a61..0000000
--- a/trunk/tools/quality_tracking/dashboard/dashboard.py
+++ /dev/null
@@ -1,57 +0,0 @@
-#!/usr/bin/env python
-#-*- coding: utf-8 -*-
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-"""Implements the quality tracker dashboard and reporting facilities."""
-
-__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
-
-from google.appengine.ext.webapp import template
-import webapp2
-
-import add_build_status_data
-import add_coverage_data
-import load_build_status
-import load_coverage
-
-
-class ShowDashboard(webapp2.RequestHandler):
- """Shows the dashboard page.
-
- The page is shown by grabbing data we have stored previously
- in the App Engine database using the AddCoverageData handler.
- """
-
- def get(self):
- build_status_loader = load_build_status.BuildStatusLoader()
- build_status_data = build_status_loader.load_build_status_data()
- last_updated_at = build_status_loader.load_last_modified_at()
- if last_updated_at is None:
- self._show_error_page("No data has yet been uploaded to the dashboard.")
- return
-
- last_updated_at = last_updated_at.strftime("%Y-%m-%d %H:%M")
- lkgr = build_status_loader.compute_lkgr()
-
- coverage_loader = load_coverage.CoverageDataLoader()
- coverage_json_data = coverage_loader.load_coverage_json_data()
-
- page_template_filename = 'templates/dashboard_template.html'
- self.response.write(template.render(page_template_filename, vars()))
-
- def _show_error_page(self, error_message):
- self.response.write('<html><body>%s</body></html>' % error_message)
-
-
-app = webapp2.WSGIApplication([('/', ShowDashboard),
- ('/add_coverage_data',
- add_coverage_data.AddCoverageData),
- ('/add_build_status_data',
- add_build_status_data.AddBuildStatusData)],
- debug=True)
diff --git a/trunk/tools/quality_tracking/dashboard/gaeunit.py b/trunk/tools/quality_tracking/dashboard/gaeunit.py
deleted file mode 120000
index a93f6bd..0000000
--- a/trunk/tools/quality_tracking/dashboard/gaeunit.py
+++ /dev/null
@@ -1 +0,0 @@
-../../../third_party/gaeunit/gaeunit.py
\ No newline at end of file
diff --git a/trunk/tools/quality_tracking/dashboard/gviz_api.py b/trunk/tools/quality_tracking/dashboard/gviz_api.py
deleted file mode 120000
index f3a22fc..0000000
--- a/trunk/tools/quality_tracking/dashboard/gviz_api.py
+++ /dev/null
@@ -1 +0,0 @@
-../../../third_party/google-visualization-python/gviz_api.py
\ No newline at end of file
diff --git a/trunk/tools/quality_tracking/dashboard/load_build_status.py b/trunk/tools/quality_tracking/dashboard/load_build_status.py
deleted file mode 100644
index 9a0eebb..0000000
--- a/trunk/tools/quality_tracking/dashboard/load_build_status.py
+++ /dev/null
@@ -1,120 +0,0 @@
-#!/usr/bin/env python
-#-*- coding: utf-8 -*-
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-"""Loads build status data for the dashboard."""
-
-__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
-
-from google.appengine.ext import db
-
-
-def _status_not_ok(status):
- return status not in ('OK', 'warnings')
-
-
-def _all_ok(statuses):
- return filter(_status_not_ok, statuses) == []
-
-
-def _get_first_entry(iterable):
- if not iterable:
- return None
- for item in iterable:
- return item
-
-
-class BuildStatusLoader:
- """ Loads various build status data from the database."""
-
- def load_build_status_data(self):
- """Returns the latest conclusive build status for each bot.
-
- The statuses OK, failed and warnings are considered to be conclusive.
-
- The two most recent revisions are considered. The set of bots returned
- will therefore be the bots that were reported the two most recent
- revisions. This script will therefore adapt automatically to any changes
- in the set of available bots.
-
- Returns:
- A list of BuildStatusData entities with one entity per bot.
- """
-
- build_status_entries = db.GqlQuery('SELECT * '
- 'FROM BuildStatusData '
- 'ORDER BY revision DESC ')
-
- bots_to_latest_conclusive_entry = dict()
- for entry in build_status_entries:
- if entry.status == 'building':
- # The 'building' status it not conclusive, so discard this entry and
- # pick up the entry for this bot on the next revision instead. That
- # entry is guaranteed to have a status != 'building' since a bot cannot
- # be building two revisions simultaneously.
- continue
- if bots_to_latest_conclusive_entry.has_key(entry.bot_name):
- # We've already determined this bot's status.
- continue
-
- bots_to_latest_conclusive_entry[entry.bot_name] = entry
-
- return bots_to_latest_conclusive_entry.values()
-
- def load_last_modified_at(self):
- build_status_root = db.GqlQuery('SELECT * '
- 'FROM BuildStatusRoot').get()
- if not build_status_root:
- # Operating on completely empty database
- return None
-
- return build_status_root.last_updated_at
-
- def compute_lkgr(self):
- """ Finds the most recent revision for which all bots are green.
-
- Returns:
- The last known good revision (as an integer) or None if there
- is no green revision in the database.
-
- Implementation note: The data store fetches stuff as we go, so we won't
- read in the whole status table unless the LKGR is right at the end or
- we don't have a LKGR.
- """
- build_status_entries = db.GqlQuery('SELECT * '
- 'FROM BuildStatusData '
- 'ORDER BY revision DESC ')
-
- first_entry = _get_first_entry(build_status_entries)
- if first_entry is None:
- # No entries => no LKGR
- return None
-
- current_lkgr = first_entry.revision
- statuses_for_current_lkgr = [first_entry.status]
-
- for entry in build_status_entries:
- if current_lkgr == entry.revision:
- statuses_for_current_lkgr.append(entry.status)
- else:
- # Starting on new revision, check previous revision.
- if _all_ok(statuses_for_current_lkgr):
- # All bots are green; LKGR found.
- return current_lkgr
- else:
- # Not all bots are green, so start over on the next revision.
- current_lkgr = entry.revision
- statuses_for_current_lkgr = [entry.status]
-
- if _all_ok(statuses_for_current_lkgr):
- # There was only one revision and it was OK.
- return current_lkgr
-
- # There is no all-green revision in the database.
- return None
diff --git a/trunk/tools/quality_tracking/dashboard/load_coverage.py b/trunk/tools/quality_tracking/dashboard/load_coverage.py
deleted file mode 100644
index eafed3b..0000000
--- a/trunk/tools/quality_tracking/dashboard/load_coverage.py
+++ /dev/null
@@ -1,39 +0,0 @@
-#!/usr/bin/env python
-#-*- coding: utf-8 -*-
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-"""Loads coverage data from the database."""
-
-__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
-
-from google.appengine.ext import db
-import gviz_api
-
-
-class CoverageDataLoader:
- """ Loads coverage data from the database."""
-
- def load_coverage_json_data(self):
- coverage_entries = db.GqlQuery('SELECT * '
- 'FROM CoverageData '
- 'ORDER BY date ASC')
- data = []
- for coverage_entry in coverage_entries:
- data.append({'date': coverage_entry.date,
- 'line_coverage': coverage_entry.line_coverage,
- 'function_coverage': coverage_entry.function_coverage,
- })
-
- description = {
- 'date': ('datetime', 'Date'),
- 'line_coverage': ('number', 'Line Coverage'),
- 'function_coverage': ('number', 'Function Coverage')
- }
- coverage_data = gviz_api.DataTable(description, data)
- return coverage_data.ToJSon(order_by='date')
diff --git a/trunk/tools/quality_tracking/dashboard/oauth_post_request_handler.py b/trunk/tools/quality_tracking/dashboard/oauth_post_request_handler.py
deleted file mode 100644
index 416e1b7..0000000
--- a/trunk/tools/quality_tracking/dashboard/oauth_post_request_handler.py
+++ /dev/null
@@ -1,68 +0,0 @@
-#!/usr/bin/env python
-#-*- coding: utf-8 -*-
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-"""Provides a OAuth request handler base class."""
-
-__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
-
-from google.appengine.api import oauth
-import logging
-import webapp2
-
-
-class UserNotAuthenticatedException(Exception):
- """Gets thrown if a user is not permitted to store data."""
- pass
-
-
-class OAuthPostRequestHandler(webapp2.RequestHandler):
- """Works like a normal request handler but adds OAuth authentication.
-
- This handler will expect a proper OAuth request over POST. This abstract
- class deals with the authentication but leaves user-defined data handling
- to its subclasses. Subclasses should not implement the post() method but
- the _parse_and_store_data() method. Otherwise they may act like regular
- request handlers. Subclasses should NOT override the get() method.
-
- The handler will accept an OAuth request if it is correctly formed and
- the consumer is acting on behalf of an administrator for the dashboard.
- """
-
- def post(self):
- try:
- self._authenticate_user()
- except UserNotAuthenticatedException as exception:
- logging.warn('Failed to authenticate: %s.' % exception)
- self.response.set_status(403)
- return
-
- # Do the actual work.
- self._parse_and_store_data()
-
- def _parse_and_store_data(self):
- """Reads data from POST request and responds accordingly."""
- raise NotImplementedError('You must override this method!')
-
- def _authenticate_user(self):
- try:
- if oauth.is_current_user_admin():
- # The user on whose behalf we are acting is indeed an administrator
- # of this application, so we're good to go.
- logging.info('Authenticated on behalf of user %s.' %
- oauth.get_current_user())
- return
- else:
- raise UserNotAuthenticatedException('We are acting on behalf of '
- 'user %s, but that user is not '
- 'an administrator.' %
- oauth.get_current_user())
- except oauth.OAuthRequestError as exception:
- raise UserNotAuthenticatedException('Invalid OAuth request: %s' %
- exception.__class__.__name__)
diff --git a/trunk/tools/quality_tracking/dashboard/static/google403c95edcde16425.html b/trunk/tools/quality_tracking/dashboard/static/google403c95edcde16425.html
deleted file mode 100644
index 95c7e2d..0000000
--- a/trunk/tools/quality_tracking/dashboard/static/google403c95edcde16425.html
+++ /dev/null
@@ -1 +0,0 @@
-google-site-verification: google403c95edcde16425.html
\ No newline at end of file
diff --git a/trunk/tools/quality_tracking/dashboard/stylesheets/stylesheet.css b/trunk/tools/quality_tracking/dashboard/stylesheets/stylesheet.css
deleted file mode 100644
index c6eb679..0000000
--- a/trunk/tools/quality_tracking/dashboard/stylesheets/stylesheet.css
+++ /dev/null
@@ -1,45 +0,0 @@
-/********************************************************************
-*
-* Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-*
-* Use of this source code is governed by a BSD-style license
-* that can be found in the LICENSE file in the root of the source
-* tree. An additional intellectual property rights grant can be found
-* in the file PATENTS. All contributing project authors may
-* be found in the AUTHORS file in the root of the source tree.
-*
-*********************************************************************/
-
-.status_OK {
- color: #FFFFFF;
- background-color: #8fdf5f;
-}
-
-.status_failed {
- color: #FFFFFF;
- background-color: #e98080;
-}
-
-.status_building {
- color: #666666;
- background-color: #fffc6c;
-}
-
-.status_warnings {
- color: #000000;
- background-color: #FFC343;
-}
-
-.last_known_good_revision {
- font-size: 800%;
-}
-
-.status_cell {
- width: 100px;
- text-align: center;
-}
-
-body {
- margin-left: 35px;
- margin-top: 25px;
-}
\ No newline at end of file
diff --git a/trunk/tools/quality_tracking/dashboard/templates/dashboard_template.html b/trunk/tools/quality_tracking/dashboard/templates/dashboard_template.html
deleted file mode 100644
index ab019d5..0000000
--- a/trunk/tools/quality_tracking/dashboard/templates/dashboard_template.html
+++ /dev/null
@@ -1,86 +0,0 @@
-<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.01//EN"
- "http://www.w3.org/TR/html4/strict.dtd">
-<html>
- <!--
- Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-
- Use of this source code is governed by a BSD-style license
- that can be found in the LICENSE file in the root of the source
- tree. An additional intellectual property rights grant can be found
- in the file PATENTS. All contributing project authors may
- be found in the AUTHORS file in the root of the source tree.
-
- Template file to be used to generate the WebRTC dashboard.
- -->
- <head>
- <title>WebRTC Coverage Dashboard</title>
- <meta http-equiv="refresh" content="60">
- <link href="http://code.google.com/css/codesite.pack.04102009.css"
- rel="stylesheet" type="text/css">
- <link href="stylesheets/stylesheet.css"
- rel="stylesheet" type="text/css">
-
- <script src="https://www.google.com/jsapi" type="text/javascript"></script>
- <script type="text/javascript">
- google.load('visualization', '1', {packages:['table', 'corechart']});
-
- google.setOnLoadCallback(drawTable);
- function drawTable() {
- /* Build data tables and views */
- {% comment %}
- Disable Django auto-escaping here since that will mess up our
- coverage table JSON data otherwise.
- {% endcomment %}
- {% autoescape off %}
- var coverage_data_table =
- new google.visualization.DataTable({{ coverage_json_data }});
- {% endautoescape %}
-
- /* Display tables and charts */
- var coverage_chart = new google.visualization.LineChart(
- document.getElementById('table_div_coverage'));
- coverage_chart.draw(coverage_data_table, {
- colors: ['blue', 'red'],
- vAxis: {title: 'Coverage'},
- hAxis: {title: 'Date'},
- width: 1200, height: 300,
- });
- }
- </script>
- </head>
- <body>
-
- <h1>WebRTC Quality Dashboard</h1>
- <h2>Current Build Status</h2>
- <div>(as of {{ last_updated_at }} UTC)</div>
- <table>
- <tr>
- {% for entry in build_status_data %}
- <th class="status_cell">{{ entry.bot_name }}</th>
- {% endfor %}
- </tr>
- <tr>
- {% for entry in build_status_data %}
- <td title="Last built revision {{ entry.revision }}"
- class="status_cell status_{{entry.status}}">
- {{entry.status}}
- </td>
- {% endfor %}
- </tr>
- </table>
- <p></p>
-
- <h2>Last Known Good Revision (LKGR)</h2>
- <div class="last_known_good_revision">
- {% if lkgr %}
- <a href="http://code.google.com/p/webrtc/source/detail?r={{ lkgr }}">
- {{ lkgr }}</a>
- {% else %}
- ????
- {% endif %}
- </div>
-
- <h2>Code Coverage History</h2>
- <div id="table_div_coverage"></div>
- </body>
-</html>
diff --git a/trunk/tools/quality_tracking/dashboard/test/load_build_status_test.py b/trunk/tools/quality_tracking/dashboard/test/load_build_status_test.py
deleted file mode 100755
index 56c9379..0000000
--- a/trunk/tools/quality_tracking/dashboard/test/load_build_status_test.py
+++ /dev/null
@@ -1,114 +0,0 @@
-#!/usr/bin/env python
-#-*- coding: utf-8 -*-
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
-
-import unittest
-from google.appengine.ext import db
-from google.appengine.ext import testbed
-
-from add_build_status_data import BuildStatusData
-import load_build_status
-
-class LoadBuildStatusTest(unittest.TestCase):
- def setUp(self):
- # First, create an instance of the Testbed class.
- self.testbed = testbed.Testbed()
- # Then activate the testbed, which prepares the service stubs for use.
- self.testbed.activate()
- # Next, declare which service stubs you want to use.
- self.testbed.init_datastore_v3_stub()
-
- def test_returns_latest_nonbuilding_entries_when_loading_build_status(self):
- BuildStatusData(bot_name="Bot1", revision=17,
- build_number=499, status="OK").put()
- BuildStatusData(bot_name="Bot2", revision=17,
- build_number=505, status="OK").put()
- BuildStatusData(bot_name="Bot3", revision=17,
- build_number=344, status="failed").put()
- BuildStatusData(bot_name="Bot1", revision=18,
- build_number=499, status="building").put()
- BuildStatusData(bot_name="Bot2", revision=18,
- build_number=505, status="failed").put()
- BuildStatusData(bot_name="Bot3", revision=18,
- build_number=344, status="OK").put()
-
- loader = load_build_status.BuildStatusLoader()
- result = loader.load_build_status_data()
-
- self.assertEqual(3, len(result))
-
- # We make no guarantees on order, but we can use the fact that the testbed
- # is deterministic to evaluate that the corrects bots were selected like so:
- self.assertEqual("Bot1", result[0].bot_name)
- self.assertEqual(17, result[0].revision)
- self.assertEqual("OK", result[0].status)
-
- self.assertEqual("Bot3", result[1].bot_name)
- self.assertEqual(18, result[1].revision)
- self.assertEqual("OK", result[1].status)
-
- self.assertEqual("Bot2", result[2].bot_name)
- self.assertEqual(18, result[2].revision)
- self.assertEqual("failed", result[2].status)
-
- def test_returns_lkgr_for_single_green_revision(self):
- BuildStatusData(bot_name="Bot1", revision=17,
- build_number=499, status="OK").put()
- BuildStatusData(bot_name="Bot2", revision=17,
- build_number=505, status="OK").put()
- BuildStatusData(bot_name="Bot3", revision=17,
- build_number=344, status="OK").put()
-
- loader = load_build_status.BuildStatusLoader()
- self.assertEqual(17, loader.compute_lkgr())
-
- def test_returns_correct_lkgr_with_most_recent_revision_failed(self):
- BuildStatusData(bot_name="Bot1", revision=17,
- build_number=499, status="OK").put()
- BuildStatusData(bot_name="Bot2", revision=17,
- build_number=505, status="OK").put()
- BuildStatusData(bot_name="Bot3", revision=17,
- build_number=344, status="OK").put()
- BuildStatusData(bot_name="Bot1", revision=18,
- build_number=499, status="OK").put()
- BuildStatusData(bot_name="Bot2", revision=18,
- build_number=505, status="failed").put()
- BuildStatusData(bot_name="Bot3", revision=18,
- build_number=344, status="OK").put()
-
- loader = load_build_status.BuildStatusLoader()
- self.assertEqual(17, loader.compute_lkgr())
-
- def test_returns_none_if_no_revisions(self):
- loader = load_build_status.BuildStatusLoader()
- self.assertEqual(None, loader.compute_lkgr())
-
- def test_returns_none_if_no_green_revisions(self):
- BuildStatusData(bot_name="Bot2", revision=18,
- build_number=505, status="failed").put()
-
- loader = load_build_status.BuildStatusLoader()
- self.assertEqual(None, loader.compute_lkgr())
-
- def test_skips_partially_building_revisions(self):
- BuildStatusData(bot_name="Bot1", revision=18,
- build_number=499, status="building").put()
- BuildStatusData(bot_name="Bot2", revision=18,
- build_number=505, status="OK").put()
- BuildStatusData(bot_name="Bot1", revision=17,
- build_number=344, status="OK").put()
-
- loader = load_build_status.BuildStatusLoader()
- self.assertEqual(17, loader.compute_lkgr())
-
-
-if __name__ == '__main__':
- unittest.main()
diff --git a/trunk/tools/quality_tracking/dashboard_connection.py b/trunk/tools/quality_tracking/dashboard_connection.py
deleted file mode 100644
index 9a6e30f..0000000
--- a/trunk/tools/quality_tracking/dashboard_connection.py
+++ /dev/null
@@ -1,134 +0,0 @@
-#!/usr/bin/env python
-#-*- coding: utf-8 -*-
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-"""Contains utilities for communicating with the dashboard."""
-
-__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
-
-import httplib
-import shelve
-import urlparse
-import oauth.oauth as oauth
-
-import constants
-
-
-class FailedToReadRequiredInputFile(Exception):
- pass
-
-
-class FailedToReportToDashboard(Exception):
- pass
-
-
-class DashboardConnection:
- """Helper class for pushing data to the dashboard.
-
- This class deals with most of details for accessing protected resources
- (i.e. data-writing operations) on the dashboard. Such operations are
- authenticated using OAuth. This class requires a consumer secret and a
- access token.
-
- The access token and consumer secrets are stored as files on disk in the
- working directory of the scripts. Both files are created by the
- request_oauth_permission script.
- """
-
- def __init__(self, consumer_key):
- self.consumer_key_ = consumer_key
-
- def read_required_files(self, consumer_secret_file, access_token_file):
- """Reads required data for making OAuth requests.
-
- Args:
- consumer_secret_file: A shelve file with an entry consumer_secret
- containing the consumer secret in string form.
- access_token_file: A shelve file with an entry access_token
- containing the access token in string form.
- """
- self.access_token_string_ = self._read_access_token(access_token_file)
- self.consumer_secret_ = self._read_consumer_secret(consumer_secret_file)
-
- def send_post_request(self, url, parameters):
- """Sends an OAuth request for a protected resource in the dashboard.
-
- Use this when you want to report new data to the dashboard. You must have
- called the read_required_files method prior to calling this method, since
- that method will read in the consumer secret and access token we need to
- make the OAuth request. These concepts are described in the class
- description.
-
- The server is expected to respond with HTTP status 200 and a completely
- empty response if the call failed. The server may put diagnostic
- information in the response.
-
- Args:
- url: An absolute url within the dashboard domain, for example
- http://webrtc-dashboard.appspot.com/add_coverage_data.
- parameters: A dict which maps from POST parameter names to values.
-
- Raises:
- FailedToReportToDashboard: If the dashboard didn't respond
- with HTTP 200 to our request or if the response is non-empty.
- """
- consumer = oauth.OAuthConsumer(self.consumer_key_, self.consumer_secret_)
- access_token = oauth.OAuthToken.from_string(self.access_token_string_)
-
- oauth_request = oauth.OAuthRequest.from_consumer_and_token(
- consumer,
- token=access_token,
- http_method='POST',
- http_url=url,
- parameters=parameters)
-
- signature_method_hmac_sha1 = oauth.OAuthSignatureMethod_HMAC_SHA1()
- oauth_request.sign_request(signature_method_hmac_sha1, consumer,
- access_token)
-
- connection = httplib.HTTPConnection(constants.DASHBOARD_SERVER)
-
- headers = {'Content-Type': 'application/x-www-form-urlencoded'}
- connection.request('POST', url, body=oauth_request.to_postdata(),
- headers=headers)
-
- response = connection.getresponse()
- connection.close()
-
- if response.status != 200:
- message = ('Failed to report to %s: got response %d (%s)' %
- (url, response.status, response.reason))
- raise FailedToReportToDashboard(message)
-
- # The response content should be empty on success, so check that:
- response_content = response.read()
- if response_content:
- message = ('Dashboard reported the following error: %s.' %
- response_content)
- raise FailedToReportToDashboard(message)
-
- def _read_access_token(self, filename):
- return self._read_shelve(filename, 'access_token')
-
- def _read_consumer_secret(self, filename):
- return self._read_shelve(filename, 'consumer_secret')
-
- def _read_shelve(self, filename, key):
- input_file = shelve.open(filename)
-
- if not input_file.has_key(key):
- raise FailedToReadRequiredInputFile('Missing correct %s file in current '
- 'directory. You may have to run '
- 'request_oauth_permission.py.' %
- filename)
-
- result = input_file[key]
- input_file.close()
-
- return result
diff --git a/trunk/tools/quality_tracking/oauth2 b/trunk/tools/quality_tracking/oauth2
deleted file mode 120000
index a4d1dde..0000000
--- a/trunk/tools/quality_tracking/oauth2
+++ /dev/null
@@ -1 +0,0 @@
-../../third_party/oauth2/oauth2/
\ No newline at end of file
diff --git a/trunk/tools/quality_tracking/request_oauth_permission.py b/trunk/tools/quality_tracking/request_oauth_permission.py
deleted file mode 100755
index fb97738..0000000
--- a/trunk/tools/quality_tracking/request_oauth_permission.py
+++ /dev/null
@@ -1,142 +0,0 @@
-#!/usr/bin/env python
-#-*- coding: utf-8 -*-
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-"""This script request an access token from the appengine running the dashboard.
-
- The script is intended to be run manually whenever we wish to change which
- dashboard administrator we act on behalf of when running the
- track_coverage.py script. For example, this will be useful if the current
- dashboard administrator leaves the project. This script can also be used to
- launch a new dashboard if that is desired.
-
- This script should be run on the build bot which runs the track_coverage.py
- script. This script will present a link during its execution, which the new
- administrator should follow and then click approve on the web page that
- appears. The new administrator should have admin rights on the coverage
- dashboard, otherwise the track_* scripts will not work.
-
- If successful, this script will write the access token to a file access.token
- in the current directory, which later can be read by the track_* scripts.
- The token is stored in string form (as reported by the web server) using the
- shelve module. The consumer secret passed in as an argument to this script
- will also similarly be stored in a file consumer.secret. The shelve keys
- will be 'access_token' and 'consumer_secret', respectively.
-"""
-
-__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
-
-import shelve
-import sys
-import urlparse
-import oauth2 as oauth
-
-import constants
-
-
-class FailedToRequestPermissionException(Exception):
- pass
-
-
-def _ensure_token_response_is_200(response, queried_url, token_type):
- if response.status != 200:
- raise FailedToRequestPermissionException('Failed to request %s from %s: '
- 'received status %d, reason %s.' %
- (token_type,
- queried_url,
- response.status,
- response.reason))
-
-
-def _request_unauthorized_token(consumer, request_token_url):
- """Requests the initial token from the dashboard service.
-
- Given that the response from the server is correct, we will return a
- dictionary containing oauth_token and oauth_token_secret mapped to the
- token and secret value, respectively.
- """
- client = oauth.Client(consumer)
-
- try:
- response, content = client.request(request_token_url, 'POST')
- except AttributeError as error:
- # This catch handler is here since we'll get very confusing messages
- # if the target server is down for some reason.
- raise FailedToRequestPermissionException('Failed to request token: '
- 'the dashboard is likely down.',
- error)
-
- _ensure_token_response_is_200(response, request_token_url,
- 'unauthorized token')
-
- return dict(urlparse.parse_qsl(content))
-
-
-def _ask_user_to_authorize_us(unauthorized_token):
- """This function will block until the user enters y + newline."""
- print 'Go to the following link in your browser:'
- print '%s?oauth_token=%s' % (constants.AUTHORIZE_TOKEN_URL,
- unauthorized_token['oauth_token'])
-
- accepted = 'n'
- while accepted.lower() != 'y':
- accepted = raw_input('Have you authorized me yet? (y/n) ')
-
-
-def _request_access_token(consumer, unauthorized_token):
- token = oauth.Token(unauthorized_token['oauth_token'],
- unauthorized_token['oauth_token_secret'])
- client = oauth.Client(consumer, token)
- response, content = client.request(constants.ACCESS_TOKEN_URL, 'POST')
-
- _ensure_token_response_is_200(response, constants.ACCESS_TOKEN_URL,
- 'access token')
-
- return content
-
-
-def _write_access_token_to_file(access_token, filename):
- output = shelve.open(filename)
- output['access_token'] = access_token
- output.close()
-
- print 'Wrote the access token to the file %s.' % filename
-
-
-def _write_consumer_secret_to_file(consumer_secret, filename):
- output = shelve.open(filename)
- output['consumer_secret'] = consumer_secret
- output.close()
-
- print 'Wrote the consumer secret to the file %s.' % filename
-
-
-def _main():
- if len(sys.argv) != 2:
- print ('Usage: %s <consumer secret>.\n\nThe consumer secret is an OAuth '
- 'concept and is obtained from the Google Accounts domain dashboard.'
- % sys.argv[0])
- return
-
- consumer_secret = sys.argv[1]
- consumer = oauth.Consumer(constants.CONSUMER_KEY, consumer_secret)
-
- unauthorized_token = _request_unauthorized_token(consumer,
- constants.REQUEST_TOKEN_URL)
-
- _ask_user_to_authorize_us(unauthorized_token)
-
- access_token_string = _request_access_token(consumer, unauthorized_token)
-
- _write_access_token_to_file(access_token_string, constants.ACCESS_TOKEN_FILE)
- _write_consumer_secret_to_file(consumer_secret,
- constants.CONSUMER_SECRET_FILE)
-
-if __name__ == '__main__':
- _main()
diff --git a/trunk/tools/quality_tracking/tgrid_parser.py b/trunk/tools/quality_tracking/tgrid_parser.py
deleted file mode 100644
index 2b674d5..0000000
--- a/trunk/tools/quality_tracking/tgrid_parser.py
+++ /dev/null
@@ -1,73 +0,0 @@
-#!/usr/bin/env python
-#-*- coding: utf-8 -*-
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-"""Contains functions for parsing the build master's transposed grid page."""
-
-__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
-
-import re
-
-
-class FailedToParseBuildStatus(Exception):
- pass
-
-
-def _parse_builds(revision, html):
- """Parses the bot list, which is a sequence of <td></td> lines.
-
- Example input:
- <td class="build success"><a href="builders/Android/builds/119">OK</a></td>
- The first regular expression group captures Android, second 119, third OK.
- """
- result = {}
-
- for match in re.finditer('<td.*?>.*?<a href="builders/(.+?)/builds/(\d+)">'
- '(OK|failed|building|warnings)</a>.*?</td>',
- html, re.DOTALL):
- revision_and_bot_name = revision + "--" + match.group(1)
- build_number_and_status = match.group(2) + "--" + match.group(3)
-
- result[revision_and_bot_name] = build_number_and_status
-
- return result
-
-
-def parse_tgrid_page(html):
- """Parses the build master's tgrid page.
-
- Example input:
- <tr>
- <td valign="bottom" class="sourcestamp">1568</td>
- LIST OF BOTS
- </tr>
- The first regular expression group captures 1568, second group captures
- everything in LIST OF BOTS. The list of bots is then passed into a
- separate function for parsing.
-
- Args:
- html: The raw HTML from the tgrid page.
-
- Returns: A dictionary with <svn revision>--<bot name> mapped to
- <bot build number>--<status>, where status is either OK, failed,
- building or warnings.
- """
- result = {}
-
- for match in re.finditer('<td.*?class="sourcestamp">(\d+)</td>(.*?)</tr>',
- html, re.DOTALL):
- revision = match.group(1)
- builds_for_revision_html = match.group(2)
- result.update(_parse_builds(revision, builds_for_revision_html))
-
- if not result:
- raise FailedToParseBuildStatus('Could not find any build statuses in %s.' %
- html)
-
- return result
diff --git a/trunk/tools/quality_tracking/tgrid_parser_test.py b/trunk/tools/quality_tracking/tgrid_parser_test.py
deleted file mode 100755
index 8b15c90..0000000
--- a/trunk/tools/quality_tracking/tgrid_parser_test.py
+++ /dev/null
@@ -1,207 +0,0 @@
-#!/usr/bin/env python
-#-*- coding: utf-8 -*-
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-"""Contains functions for parsing the build master's transposed grid page."""
-
-__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
-
-import unittest
-
-import tgrid_parser
-
-
-SAMPLE_FILE = """
-<!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Strict//EN"
- "http://www.w3.org/TR/xhtml1/DTD/xhtml1-strict.dtd">
-
-<html
- xmlns="http://www.w3.org/1999/xhtml"
- lang="en"
- xml:lang="en">
-<head>
- <title>Buildbot</title>
- <link href="buildbot.css" rel="stylesheet" type="text/css" />
-</head>
-
-<body vlink="#800080">
-<table class="Grid" border="0" cellspacing="0">
-<tr>
-<td valign="bottom" class="sourcestamp">1570</td>
-<td class="build warnings"><a href="builders/Chrome/builds/109">warnings</a>
-<br />
-make chrome</td>
-<td class="build success">
- <a href="builders/Android/builds/121">OK</a></td>
-<td class="build success">
- <a href="builders/ChromeOS/builds/578">OK</a></td>
-<td class="build success">
- <a href="builders/Linux32bitDBG/builds/564">OK</a></td>
-<td class="build success">
- <a href="builders/Linux32bitRelease/builds/684">OK</a></td>
-<td class="build success">
- <a href="builders/Linux64bitDBG/builds/680">OK</a></td>
-<td class="build success">
- <a href="builders/Linux64bitDBG-GCC4.6/builds/5">OK</a></td>
-<td class="build success">
- <a href="builders/Linux64bitRelease/builds/570">OK</a></td>
-<td class="build success">
- <a href="builders/LinuxCLANG/builds/259">OK</a></td>
-<td class="build success">
- <a href="builders/LinuxVideoTest/builds/345">OK</a></td>
-<td class="build success">
- <a href="builders/MacOS/builds/670">OK</a></td>
-<td class="build success">
- <a href="builders/Win32Debug/builds/432">OK</a></td>
-<td class="build success">
- <a href="builders/Win32Release/builds/440">OK</a></td>
-</tr>
-<tr>
-<td valign="bottom" class="sourcestamp">1571</td>
-<td class="build warnings"><a href="builders/Chrome/builds/109">warnings</a>
-<br />
-make chrome</td>
-<td class="build success">
- <a href="builders/Android/builds/122">OK</a></td>
-<td class="build success">
- <a href="builders/ChromeOS/builds/579">OK</a></td>
-<td class="build success">
- <a href="builders/Linux32bitDBG/builds/565">OK</a></td>
-<td class="build success">
- <a href="builders/Linux32bitRelease/builds/685">OK</a></td>
-<td class="build success">
- <a href="builders/Linux64bitDBG/builds/681">OK</a></td>
-<td class="build success">
- <a href="builders/Linux64bitDBG-GCC4.6/builds/6">OK</a></td>
-<td class="build success">
- <a href="builders/Linux64bitRelease/builds/571">OK</a></td>
-<td class="build success">
- <a href="builders/LinuxCLANG/builds/260">OK</a></td>
-<td class="build failure">
- <a href="builders/LinuxVideoTest/builds/346">failed</a><br />
-voe_auto_test</td>
-<td class="build success">
- <a href="builders/MacOS/builds/671">OK</a></td>
-<td class="build running">
- <a href="builders/Win32Debug/builds/441">building</a></td>
-<td class="build success">
- <a href="builders/Win32Release/builds/441">OK</a></td>
-</tr>
-</table>
-</body>
-</html>
-"""
-
-MINIMAL_OK = """
-<tr>
-<td valign="bottom" class="sourcestamp">1570</td>
-<td class="build success">
-<a href="builders/Android/builds/121">OK</a></td>
-</tr>
-"""
-
-MINIMAL_FAIL = """
-<tr>
-<td valign="bottom" class="sourcestamp">1573</td>
-<td class="build failure">
-<a href="builders/LinuxVideoTest/builds/347">failed</a><br />
-voe_auto_test</td>
-</tr>
-"""
-
-MINIMAL_BUILDING = """
-<tr>
-<td valign="bottom" class="sourcestamp">1576</td>
-<td class="build running">
-<a href="builders/Win32Debug/builds/434">building</a></td>
-voe_auto_test</td>
-</tr>
-"""
-
-MINIMAL_WARNED = """
-<tr>
-<td valign="bottom" class="sourcestamp">1576</td>
-<td class="build warnings">
-<a href="builders/Chrome/builds/109">warnings</a><br />
-make chrome</td>
-</tr>
-"""
-
-class TGridParserTest(unittest.TestCase):
- def test_parser_throws_exception_on_empty_html(self):
- self.assertRaises(tgrid_parser.FailedToParseBuildStatus,
- tgrid_parser.parse_tgrid_page, '');
-
- def test_parser_finds_successful_bot(self):
- result = tgrid_parser.parse_tgrid_page(MINIMAL_OK)
-
- self.assertEqual(1, len(result), 'There is only one bot in the sample.')
- first_mapping = result.items()[0]
-
- self.assertEqual('1570--Android', first_mapping[0])
- self.assertEqual('121--OK', first_mapping[1])
-
- def test_parser_finds_failed_bot(self):
- result = tgrid_parser.parse_tgrid_page(MINIMAL_FAIL)
-
- self.assertEqual(1, len(result), 'There is only one bot in the sample.')
- first_mapping = result.items()[0]
-
- self.assertEqual('1573--LinuxVideoTest', first_mapping[0])
- self.assertEqual('347--failed', first_mapping[1])
-
- def test_parser_finds_building_bot(self):
- result = tgrid_parser.parse_tgrid_page(MINIMAL_BUILDING)
-
- self.assertEqual(1, len(result), 'There is only one bot in the sample.')
- first_mapping = result.items()[0]
-
- self.assertEqual('1576--Win32Debug', first_mapping[0])
- self.assertEqual('434--building', first_mapping[1])
-
- def test_parser_finds_warned_bot(self):
- result = tgrid_parser.parse_tgrid_page(MINIMAL_WARNED)
-
- self.assertEqual(1, len(result), 'There is only one bot in the sample.')
- first_mapping = result.items()[0]
-
- self.assertEqual('1576--Chrome', first_mapping[0])
- self.assertEqual('109--warnings', first_mapping[1])
-
- def test_parser_finds_all_bots_and_revisions(self):
- result = tgrid_parser.parse_tgrid_page(SAMPLE_FILE)
-
- # 2 * 13 = 26 bots in sample
- self.assertEqual(26, len(result))
-
- # Make some samples
- self.assertTrue(result.has_key('1570--ChromeOS'))
- self.assertEquals('578--OK', result['1570--ChromeOS'])
-
- self.assertTrue(result.has_key('1570--Chrome'))
- self.assertEquals('109--warnings', result['1570--Chrome'])
-
- self.assertTrue(result.has_key('1570--LinuxCLANG'))
- self.assertEquals('259--OK', result['1570--LinuxCLANG'])
-
- self.assertTrue(result.has_key('1570--Win32Release'))
- self.assertEquals('440--OK', result['1570--Win32Release'])
-
- self.assertTrue(result.has_key('1571--ChromeOS'))
- self.assertEquals('579--OK', result['1571--ChromeOS'])
-
- self.assertTrue(result.has_key('1571--LinuxVideoTest'))
- self.assertEquals('346--failed', result['1571--LinuxVideoTest'])
-
- self.assertTrue(result.has_key('1571--Win32Debug'))
- self.assertEquals('441--building', result['1571--Win32Debug'])
-
-
-if __name__ == '__main__':
- unittest.main()
diff --git a/trunk/tools/quality_tracking/track_build_status.py b/trunk/tools/quality_tracking/track_build_status.py
deleted file mode 100755
index 1ddf140..0000000
--- a/trunk/tools/quality_tracking/track_build_status.py
+++ /dev/null
@@ -1,57 +0,0 @@
-#!/usr/bin/env python
-#-*- coding: utf-8 -*-
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-"""This script checks the current build status on the master and submits
- it to the dashboard. It is adapted to build bot version 0.7.12.
-"""
-
-__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
-
-
-import httplib
-
-import constants
-import dashboard_connection
-import tgrid_parser
-
-
-class FailedToGetStatusFromMaster(Exception):
- pass
-
-
-def _download_and_parse_build_status():
- connection = httplib.HTTPConnection(constants.BUILD_MASTER_SERVER)
- connection.request('GET', constants.BUILD_MASTER_TRANSPOSED_GRID_URL)
- response = connection.getresponse()
-
- if response.status != 200:
- raise FailedToGetStatusFromMaster(('Failed to get build status from master:'
- ' got status %d, reason %s.' %
- (response.status, response.reason)))
-
- full_response = response.read()
- connection.close()
-
- return tgrid_parser.parse_tgrid_page(full_response)
-
-
-def _main():
- dashboard = dashboard_connection.DashboardConnection(constants.CONSUMER_KEY)
- dashboard.read_required_files(constants.CONSUMER_SECRET_FILE,
- constants.ACCESS_TOKEN_FILE)
-
- bot_to_status_mapping = _download_and_parse_build_status()
-
- dashboard.send_post_request(constants.ADD_BUILD_STATUS_DATA_URL,
- bot_to_status_mapping)
-
-
-if __name__ == '__main__':
- _main()
diff --git a/trunk/tools/quality_tracking/track_coverage.py b/trunk/tools/quality_tracking/track_coverage.py
deleted file mode 100755
index a46ddfc..0000000
--- a/trunk/tools/quality_tracking/track_coverage.py
+++ /dev/null
@@ -1,119 +0,0 @@
-#!/usr/bin/env python
-#-*- coding: utf-8 -*-
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-"""This script grabs and reports coverage information.
-
- It grabs coverage information from the latest Linux 32-bit build and
- pushes it to the coverage tracker, enabling us to track code coverage
- over time. This script is intended to run on the 32-bit Linux slave.
-
- This script requires an access.token file in the current directory, as
- generated by the request_oauth_permission.py script. It also expects a file
- customer.secret with a single line containing the customer secret. The
- customer secret is an OAuth concept and is received when one registers the
- application with the App Engine running the dashboard.
-
- The script assumes that all coverage data is stored under
- /home/<build bot user>/www.
-"""
-
-__author__ = 'phoglund@webrtc.org (Patrik Höglund)'
-
-import os
-import re
-import time
-
-import constants
-import dashboard_connection
-
-
-class FailedToParseCoverageHtml(Exception):
- pass
-
-
-class CouldNotFindCoverageDirectory(Exception):
- pass
-
-
-def _find_latest_32bit_debug_build(www_directory_contents, coverage_www_dir):
- """Finds the latest 32-bit coverage directory in the directory listing.
-
- Coverage directories have the form Linux32bitDBG_<number>. There may be
- other directories in the list though, for instance for other build
- configurations.
- """
-
- # This sort ensures we will encounter the directory with the highest number
- # first.
- www_directory_contents.sort(reverse=True)
-
- for entry in www_directory_contents:
- match = re.match('Linux32DBG_\d+', entry)
- if match is not None:
- return entry
-
- raise CouldNotFindCoverageDirectory('Error: Found no 32-bit '
- 'debug build in directory %s.' %
- coverage_www_dir)
-
-
-def _grab_coverage_percentage(label, index_html_contents):
- """Extracts coverage from a LCOV coverage report.
-
- Grabs coverage by assuming that the label in the coverage HTML report
- is close to the actual number and that the number is followed by a space
- and a percentage sign.
- """
- match = re.search('<td[^>]*>' + label + '</td>.*?(\d+\.\d) %',
- index_html_contents, re.DOTALL)
- if match is None:
- raise FailedToParseCoverageHtml('Missing coverage at label "%s".' % label)
-
- try:
- return float(match.group(1))
- except ValueError:
- raise FailedToParseCoverageHtml('%s is not a float.' % match.group(1))
-
-
-def _report_coverage_to_dashboard(dashboard, now, line_coverage,
- function_coverage):
- parameters = {'date': '%d' % now,
- 'line_coverage': '%f' % line_coverage,
- 'function_coverage': '%f' % function_coverage
- }
-
- dashboard.send_post_request(constants.ADD_COVERAGE_DATA_URL, parameters)
-
-
-def _main():
- dashboard = dashboard_connection.DashboardConnection(constants.CONSUMER_KEY)
- dashboard.read_required_files(constants.CONSUMER_SECRET_FILE,
- constants.ACCESS_TOKEN_FILE)
-
- www_dir_contents = os.listdir(BUILD_BOT_COVERAGE_WWW_DIRECTORY)
- latest_build_directory = _find_latest_32bit_debug_build(www_dir_contents,
- coverage_www_dir)
-
- index_html_path = os.path.join(coverage_www_dir, latest_build_directory,
- 'index.html')
- index_html_file = open(index_html_path)
- whole_file = index_html_file.read()
-
- line_coverage = _grab_coverage_percentage('Lines:', whole_file)
- function_coverage = _grab_coverage_percentage('Functions:', whole_file)
- now = int(time.time())
-
- _report_coverage_to_dashboard(dashboard, now, line_coverage,
- function_coverage)
-
-
-if __name__ == '__main__':
- _main()
-
diff --git a/trunk/tools/refactoring/addfileheader.py b/trunk/tools/refactoring/addfileheader.py
deleted file mode 100644
index 01c8a8b..0000000
--- a/trunk/tools/refactoring/addfileheader.py
+++ /dev/null
@@ -1,163 +0,0 @@
-#!/usr/bin/env python
-
-import stringmanipulation
-import filemanagement
-import sys
-
-extensions = ['.h','.cc','.c','.cpp']
-
-ignore_these = ['my_ignore_header.h']
-
-if((len(sys.argv) != 2) and (len(sys.argv) != 3)):
- print 'parameters are: directory [--commit]'
- quit()
-
-directory = sys.argv[1];
-if(not filemanagement.pathexist(directory)):
- print 'path ' + directory + ' does not exist'
- quit()
-
-if((len(sys.argv) == 3) and (sys.argv[2] != '--commit')):
- print 'parameters are: parent directory extension new extension [--commit]'
- quit()
-
-commit = False
-if(len(sys.argv) == 3):
- commit = True
-
-files_to_fix = []
-for extension in extensions:
- files_to_fix.extend(filemanagement.listallfilesinfolder(directory,\
- extension))
-
-# Just steal the header from the template
-def fileheaderasstring():
- template_file_name = 'license_template.txt'
- if (not filemanagement.fileexist(template_file_name)):
- print 'File ' + template_file_name + ' not found!'
- quit()
- template_file = open(template_file_name,'r')
- return_string = ''
- for line in template_file:
- return_string += line
- return return_string
-
-# Just steal the header from the template
-def fileheaderasarray():
- template_file_name = 'license_template.txt'
- if (not filemanagement.fileexist(template_file_name)):
- print 'File ' + template_file_name + ' not found!'
- quit()
- template_file = open(template_file_name,'r')
- return_value = []
- for line in template_file:
- return_value.append(line)
- return return_value
-
-
-def findheader(path, file_name):
- full_file_name = path + file_name
- if (not filemanagement.fileexist(full_file_name)):
- print 'File ' + file_name + ' not found!'
- print 'Unexpected error!'
- quit()
- file_handle = open(full_file_name)
- template_file_content = fileheaderasarray()
- compare_content = []
- # load the same number of lines from file as the fileheader
- for index in range(len(template_file_content)):
- line = file_handle.readline()
- if (line == ''):
- return False
- compare_content.append(line)
-
- while (True):
- found = True
- for index in range(len(template_file_content)):
- line1 = template_file_content[index]
- line2 = compare_content[index]
- if(line1 != line2):
- found = False
- break
- if (found):
- return True
- compare_content = compare_content[1:len(compare_content)]
- line = file_handle.readline()
- if (line == ''):
- return False
- compare_content.append(line)
- return False
-
-# Used to store temporary result before flushing to real file when finished
-def temporaryfilename(old_file_name):
- return old_file_name + '.deleteme'
-
-def updatefile(path, old_file_name):
- full_old_file_name = path + old_file_name
- if (not filemanagement.fileexist(full_old_file_name)):
- print 'File ' + full_old_file_name + ' is not found.'
- print 'Should not happen! Ever!'
- quit()
-
- full_temporary_file_name = path + temporaryfilename(old_file_name)
-
- # Make sure that the files are closed by putting them out of scope
- old_file = open(full_old_file_name,'r')
- temporary_file = open(full_temporary_file_name,'w')
-
- temporary_file.writelines(fileheaderasstring())
- remove_whitespaces = True
- for line in old_file:
- if (remove_whitespaces and (len(line.split()) == 0)):
- continue
- else:
- remove_whitespaces = False
- temporary_file.writelines(line)
- old_file.close()
- temporary_file.close()
-
- filemanagement.copyfile(full_old_file_name,full_temporary_file_name)
- filemanagement.deletefile(full_temporary_file_name)
-
-
-failed_files = []
-skipped_files = []
-for index in range(len(files_to_fix)):
- if(commit):
- print (100*index)/len(files_to_fix)
- path_dir = files_to_fix[index][0]
- filename = files_to_fix[index][1]
- is_ignore = False
- for ignore_names in ignore_these:
- if(filename == ignore_names):
- is_ignore = True
- break
- if(is_ignore):
- continue
-
-# Let the word copyright be our sanity, i.e. make sure there is only one
-# copy right occurance or report that there will be no change
- if(filemanagement.findstringinfile(path_dir,filename,'Copyright') or
- filemanagement.findstringinfile(path_dir,filename,'copyright') or
- filemanagement.findstringinfile(path_dir,filename,'COPYRIGHT')):
- if(findheader(path_dir,filename)):
- skipped_files.append(path_dir + filename)
- else:
- failed_files.append(path_dir + filename)
- continue
-
- if (not commit):
- print 'File ' + path_dir + filename + ' will be updated'
- continue
- updatefile(path_dir,filename)
-
-tense = 'will be'
-if (commit):
- tense = 'has been'
-if (len(skipped_files) > 0):
- print str(len(skipped_files)) + ' file(s) ' + tense + ' skipped since they already have the correct header'
-
-if (len(failed_files) > 0):
- print 'Following files seem to have an invalid file header:'
-for line in failed_files:
- print line
diff --git a/trunk/tools/refactoring/filemanagement.py b/trunk/tools/refactoring/filemanagement.py
deleted file mode 100644
index 4ff64ce..0000000
--- a/trunk/tools/refactoring/filemanagement.py
+++ /dev/null
@@ -1,72 +0,0 @@
-import fnmatch
-import os
-import stringmanipulation
-
-def fileexist( file_name ):
- return os.path.isfile(file_name)
-
-def pathexist( path ):
- return os.path.exists(path)
-
-def fixpath( path ):
- return_value = path
- if( return_value[len(return_value) - 1] != '/'):
- return_value = return_value + '/'
- return return_value
-
-def listallfilesinfolder( path, extension ):
- matches = []
- signature = '*' + extension
- for root, dirnames, filenames in os.walk(path):
- for filename in fnmatch.filter(filenames, signature):
- matches.append([fixpath(root), filename])
- return matches
-
-def copyfile(to_file, from_file):
- if(not fileexist(from_file)):
- return
- command = 'cp -f ' + from_file + ' ' + to_file
- os.system(command)
- #print command
-
-def deletefile(file_to_delete):
- if(not fileexist(file_to_delete)):
- return
- os.system('rm ' + file_to_delete)
-
-# very ugly but works, so keep for now
-def findstringinfile(path,file_name,search_string):
- command = 'grep \'' + search_string + '\' ' + path + file_name + ' > deleteme.txt'
- return_value = os.system(command)
-# print command
- return (return_value == 0)
-
-def replacestringinfolder( path, old_string, new_string, extension ):
- if(not stringmanipulation.isextension(extension)):
- print 'failed to search and replace'
- return
- if(len(old_string) == 0):
- print 'failed to search and replace'
- return
- find_command = 'ls '+ path + '/*' + extension
- sed_command = 'sed -i \'s/' + old_string + '/' + new_string +\
- '/g\' *' + extension
- command_string = find_command + ' | xargs ' + sed_command + ' 2> deleteme.txt'
- os.system(command_string)
- #print command_string
-
-#find ./ -name "*.h" -type f | xargs -P 0 sed -i 's/process_thread_wrapper.h/process_thread.h/g' *.h deleteme.txt
-def replacestringinallsubfolders( old_string, new_string, extension):
- if(not stringmanipulation.isextension(extension)):
- print 'failed to search and replace'
- return
- if(len(old_string) == 0):
- print 'failed to search and replace'
- return
-
- find_command = 'find ./ -name \"*' + extension + '\" -type f'
- sed_command = 'sed -i \'s/' + old_string + '/' + new_string +\
- '/g\' *' + extension
- command_string = find_command + ' | xargs -P 0 ' + sed_command + ' 2> deleteme.txt'
- os.system(command_string)
- #print command_string
diff --git a/trunk/tools/refactoring/fixincludeguards.py b/trunk/tools/refactoring/fixincludeguards.py
deleted file mode 100644
index 0b56355..0000000
--- a/trunk/tools/refactoring/fixincludeguards.py
+++ /dev/null
@@ -1,145 +0,0 @@
-#!/usr/bin/env python
-
-import stringmanipulation
-import filemanagement
-import sys
-
-extensions = ['.h']
-
-ignore_these = ['my_ignore_header.h']
-
-if((len(sys.argv) != 2) and (len(sys.argv) != 3)):
- print 'parameters are: directory [--commit]'
- quit()
-
-directory = sys.argv[1];
-if(not filemanagement.pathexist(directory)):
- print 'path ' + directory + ' does not exist'
- quit()
-
-if((len(sys.argv) == 3) and (sys.argv[2] != '--commit')):
- print 'parameters are: parent directory extension new extension [--commit]'
- quit()
-
-commit = False
-if(len(sys.argv) == 3):
- commit = True
-
-for extension in extensions:
- files_to_fix = filemanagement.listallfilesinfolder(directory,\
- extension)
-
-def buildincludeguardname(path,filename):
- full_file_name = 'WEBRTC_' + path + filename
- full_file_name = full_file_name.upper()
- full_file_name = stringmanipulation.replaceoccurances(full_file_name, '/', '_')
- full_file_name = stringmanipulation.replaceoccurances(full_file_name, '\\', '_')
- full_file_name = stringmanipulation.replaceoccurances(full_file_name, '.', '_')
- full_file_name += '_'
- return full_file_name
-
-def buildnewincludeguardset(path,filename):
- include_guard_name = buildincludeguardname(path,filename)
- if(include_guard_name == ''):
- return []
- return_value = []
- return_value.append('#ifndef ' + include_guard_name)
- return_value.append('#define ' + include_guard_name)
- return_value.append(include_guard_name)
- return return_value
-
-def printincludeguardset(include_guard_set):
- print 'First line: ' + include_guard_set[0]
- print 'Second line: ' + include_guard_set[1]
- print 'Last line: ' + include_guard_set[2]
- return
-
-include_guard_begin_identifier = ['#ifndef', '#if !defined']
-include_guard_second_identifier = ['#define']
-def findincludeguardidentifier(line):
- for begin_identifier in include_guard_begin_identifier:
- line = stringmanipulation.removealloccurances(line,begin_identifier)
- for second_identifier in include_guard_begin_identifier:
- line = stringmanipulation.removealloccurances(line,second_identifier)
- removed_prefix = [True,'']
- line = stringmanipulation.whitespacestoonespace(line)
- while(removed_prefix[0]):
- removed_prefix = stringmanipulation.removeprefix(line,' ')
- line = removed_prefix[1]
- line = stringmanipulation.removealloccurances(line,'(')
- if(line == ''):
- return ''
- word_pos = stringmanipulation.getword(line,0)
- return_value = line[0:word_pos[1]]
- return_value = return_value.rstrip('\r\n')
- return return_value
-
-def findoldincludeguardset(path,filename):
- return_value = []
- full_file_name = path + filename
- file_pointer = open(full_file_name,'r')
- include_guard_name = ''
- for line in file_pointer:
- if (include_guard_name == ''):
- for compare_string in include_guard_begin_identifier:
- if (stringmanipulation.issubstring(compare_string, line) != -1):
- include_guard_name = findincludeguardidentifier(line)
- if (include_guard_name == ''):
- break
- line = line.rstrip('\r\n')
- return_value.append(line)
- break
- else:
- for compare_string in include_guard_second_identifier:
- if (stringmanipulation.issubstring(compare_string, line) != -1):
- if (stringmanipulation.issubstring(include_guard_name, line) != -1):
- line = line.rstrip('\r\n')
- return_value.append(line)
- return_value.append(include_guard_name)
- return return_value
- include_guard_name = ''
- return_value = []
- return []
-
-failed_files = []
-for index in range(len(files_to_fix)):
- if(commit):
- print (100*index)/len(files_to_fix)
- path_dir = files_to_fix[index][0]
- filename = files_to_fix[index][1]
- is_ignore = False
- for ignore_names in ignore_these:
- if(filename == ignore_names):
- is_ignore = True
- break
- if(is_ignore):
- continue
- old_include_guard_set = findoldincludeguardset(path_dir,filename)
- if (len(old_include_guard_set) != 3) :
- failed_files.append('unable to figure out the include guards for ' + filename)
- continue
-
- new_include_guard_set = buildnewincludeguardset(path_dir,filename)
- if (len(new_include_guard_set) != 3) :
- failed_files.append('unable to figure out new the include guards for ' + filename)
- continue
-
- if(not commit):
- print 'old guard: ' + old_include_guard_set[2]
- print 'new guard: ' + new_include_guard_set[2]
- continue
-
- for index in range(2):
- # enough to only replace for file. However, no function for that
- for extension in extensions:
- filemanagement.replacestringinfolder(path_dir,old_include_guard_set[index],new_include_guard_set[index],extension)
- # special case for last to avoid complications
- for extension in extensions:
- filemanagement.replacestringinfolder(path_dir,' ' + old_include_guard_set[2],' ' + new_include_guard_set[2],extension)
- filemanagement.replacestringinfolder(path_dir,'\\/\\/' + old_include_guard_set[2],'\\/\\/ ' + new_include_guard_set[2],extension)
-
-
-if(len(failed_files) > 0):
- print 'Following failures should be investigated manually:'
-for line in failed_files:
- print line
diff --git a/trunk/tools/refactoring/fixnames.py b/trunk/tools/refactoring/fixnames.py
deleted file mode 100644
index 15381e3..0000000
--- a/trunk/tools/refactoring/fixnames.py
+++ /dev/null
@@ -1,387 +0,0 @@
-#!/usr/bin/env python
-
-import stringmanipulation
-import filemanagement
-import p4commands
-import sys
-
-name_space_to_ignore = 'GIPS::'
-#only allow one prefix to be removed since allowing multiple will complicate
-# things
-prefix_to_filter = 'gips'
-#words_to_filter = ['Module']
-# it might be dangerous to remove GIPS but keep it default
-words_to_filter = ['Module','GIPS']
-
-# This script finds all the words that should be replaced in an h-file. Once
-# all words that should be replaced are found it does a global search and
-# replace.
-
-extensions_to_edit = ['.cpp','.cc','.h']
-
-#line = ' ~hiGIPSCriticalSectionScoped()'
-#print line
-#position = stringmanipulation.getword(line,11)
-#old_word = line[position[0]:position[0]+position[1]]
-#result = stringmanipulation.removealloccurances(old_word,'gips')
-#new_word = result
-#print old_word
-#print position[0]
-#print position[0]+position[1]
-#print new_word
-#quit()
-
-# Ignore whole line if any item in this table is a substring of the line
-do_not_replace_line_table = []
-do_not_replace_line_table.append('namespace GIPS')
-
-# [old_string,new_string]
-# List of things to remove that are static:
-manual_replace_table = []
-#manual_replace_table.append(['using namespace GIPS;',''])
-#manual_replace_table.append(['CreateGipsEvent','CreateEvent'])
-#manual_replace_table.append(['CreateGIPSTrace','CreateTrace'])
-#manual_replace_table.append(['ReturnGIPSTrace','ReturnTrace'])
-#manual_replace_table.append(['CreateGIPSFile','CreateFile'])
-replace_table = manual_replace_table
-#replace_table.append(['GIPS::','webrtc::'])
-# List of things to not remove that are static, i.e. exceptions:
-# don't replace any of the GIPS_Words since that will affect all files
-# do that in a separate script!
-do_not_replace_table = []
-do_not_replace_table.append('GIPS_CipherTypes')
-do_not_replace_table.append('GIPS_AuthenticationTypes')
-do_not_replace_table.append('GIPS_SecurityLevels')
-do_not_replace_table.append('GIPS_encryption')
-do_not_replace_table.append('~GIPS_encryption')
-do_not_replace_table.append('GIPS_transport')
-do_not_replace_table.append('~GIPS_transport')
-do_not_replace_table.append('GIPSTraceCallback')
-do_not_replace_table.append('~GIPSTraceCallback')
-do_not_replace_table.append('GIPS_RTP_CSRC_SIZE')
-do_not_replace_table.append('GIPS_RTPDirections')
-do_not_replace_table.append('GIPS_RTP_INCOMING')
-do_not_replace_table.append('GIPS_RTP_OUTGOING')
-do_not_replace_table.append('GIPSFrameType')
-do_not_replace_table.append('GIPS_FRAME_EMPTY')
-do_not_replace_table.append('GIPS_AUDIO_FRAME_SPEECH')
-do_not_replace_table.append('GIPS_AUDIO_FRAME_CN')
-do_not_replace_table.append('GIPS_VIDEO_FRAME_KEY')
-do_not_replace_table.append('GIPS_VIDEO_FRAME_DELTA')
-do_not_replace_table.append('GIPS_VIDEO_FRAME_GOLDEN')
-do_not_replace_table.append('GIPS_VIDEO_FRAME_DELTA_KEY')
-do_not_replace_table.append('GIPS_PacketType')
-do_not_replace_table.append('GIPS_PACKET_TYPE_RTP')
-do_not_replace_table.append('GIPS_PACKET_TYPE_KEEP_ALIVE')
-do_not_replace_table.append('GIPS_AudioLayers')
-do_not_replace_table.append('GIPS_AUDIO_PLATFORM_DEFAULT')
-do_not_replace_table.append('GIPS_AUDIO_WINDOWS_WAVE')
-do_not_replace_table.append('GIPS_AUDIO_WINDOWS_CORE')
-do_not_replace_table.append('GIPS_AUDIO_LINUX_ALSA')
-do_not_replace_table.append('GIPS_AUDIO_LINUX_PULSE')
-do_not_replace_table.append('GIPS_AUDIO_FORMAT')
-do_not_replace_table.append('GIPS_PCM_16_16KHZ')
-do_not_replace_table.append('GIPS_PCM_16_8KHZ')
-do_not_replace_table.append('GIPS_G729')
-do_not_replace_table.append('GIPSAMRmode')
-do_not_replace_table.append('GIPS_RFC3267_BWEFFICIENT')
-do_not_replace_table.append('GIPS_RFC3267_OCTETALIGNED')
-do_not_replace_table.append('GIPS_RFC3267_FILESTORAGE')
-do_not_replace_table.append('GIPS_NCModes')
-do_not_replace_table.append('GIPS_NC_OFF')
-do_not_replace_table.append('GIPS_NC_MILD')
-do_not_replace_table.append('GIPS_NC_MODERATE')
-do_not_replace_table.append('GIPS_NC_AGGRESSIVE')
-do_not_replace_table.append('GIPS_NC_VERY_AGGRESSIVE')
-do_not_replace_table.append('GIPS_AGCModes')
-do_not_replace_table.append('GIPS_AGC_OFF')
-do_not_replace_table.append('GIPS_AGC_ANALOG')
-do_not_replace_table.append('GIPS_AGC_DIGITAL')
-do_not_replace_table.append('GIPS_AGC_STANDALONE_DIG')
-do_not_replace_table.append('GIPS_ECModes')
-do_not_replace_table.append('GIPS_EC_UNCHANGED')
-do_not_replace_table.append('GIPS_EC_DEFAULT')
-do_not_replace_table.append('GIPS_EC_CONFERENCE')
-do_not_replace_table.append('GIPS_EC_AEC')
-do_not_replace_table.append('GIPS_EC_AES')
-do_not_replace_table.append('GIPS_EC_AECM')
-do_not_replace_table.append('GIPS_EC_NEC_IAD')
-do_not_replace_table.append('GIPS_AESModes')
-do_not_replace_table.append('GIPS_AES_DEFAULT')
-do_not_replace_table.append('GIPS_AES_NORMAL')
-do_not_replace_table.append('GIPS_AES_HIGH')
-do_not_replace_table.append('GIPS_AES_ATTENUATE')
-do_not_replace_table.append('GIPS_AES_NORMAL_SOFT_TRANS')
-do_not_replace_table.append('GIPS_AES_HIGH_SOFT_TRANS')
-do_not_replace_table.append('GIPS_AES_ATTENUATE_SOFT_TRANS')
-do_not_replace_table.append('GIPS_AECMModes')
-do_not_replace_table.append('GIPS_AECM_QUIET_EARPIECE_OR_HEADSET')
-do_not_replace_table.append('GIPS_AECM_EARPIECE')
-do_not_replace_table.append('GIPS_AECM_LOUD_EARPIECE')
-do_not_replace_table.append('GIPS_AECM_SPEAKERPHONE')
-do_not_replace_table.append('GIPS_AECM_LOUD_SPEAKERPHONE')
-do_not_replace_table.append('AECM_LOUD_SPEAKERPHONE')
-do_not_replace_table.append('GIPS_VAD_CONVENTIONAL')
-do_not_replace_table.append('GIPS_VAD_AGGRESSIVE_LOW')
-do_not_replace_table.append('GIPS_VAD_AGGRESSIVE_MID')
-do_not_replace_table.append('GIPS_VAD_AGGRESSIVE_HIGH')
-do_not_replace_table.append('GIPS_NetEQModes')
-do_not_replace_table.append('GIPS_NETEQ_DEFAULT')
-do_not_replace_table.append('GIPS_NETEQ_STREAMING')
-do_not_replace_table.append('GIPS_NETEQ_FAX')
-do_not_replace_table.append('GIPS_NetEQBGNModes')
-do_not_replace_table.append('GIPS_BGN_ON')
-do_not_replace_table.append('GIPS_BGN_FADE')
-do_not_replace_table.append('GIPS_BGN_OFF')
-do_not_replace_table.append('GIPS_OnHoldModes')
-do_not_replace_table.append('GIPS_HOLD_SEND_AND_PLAY')
-do_not_replace_table.append('GIPS_HOLD_SEND_ONLY')
-do_not_replace_table.append('GIPS_HOLD_PLAY_ONLY')
-do_not_replace_table.append('GIPS_PayloadFrequencies')
-do_not_replace_table.append('GIPS_FREQ_8000_HZ')
-do_not_replace_table.append('GIPS_FREQ_16000_HZ')
-do_not_replace_table.append('GIPS_FREQ_32000_HZ')
-do_not_replace_table.append('GIPS_TelephoneEventDetectionMethods')
-do_not_replace_table.append('GIPS_IN_BAND')
-do_not_replace_table.append('GIPS_OUT_OF_BAND')
-do_not_replace_table.append('GIPS_IN_AND_OUT_OF_BAND')
-do_not_replace_table.append('GIPS_ProcessingTypes')
-do_not_replace_table.append('GIPS_PLAYBACK_PER_CHANNEL')
-do_not_replace_table.append('GIPS_PLAYBACK_ALL_CHANNELS_MIXED')
-do_not_replace_table.append('GIPS_RECORDING_PER_CHANNEL')
-do_not_replace_table.append('GIPS_RECORDING_ALL_CHANNELS_MIXED')
-do_not_replace_table.append('GIPS_StereoChannel')
-do_not_replace_table.append('GIPS_StereoLeft')
-do_not_replace_table.append('GIPS_StereoRight')
-do_not_replace_table.append('GIPS_StereoBoth')
-do_not_replace_table.append('GIPS_stat_val')
-do_not_replace_table.append('GIPS_P56_statistics')
-do_not_replace_table.append('GIPS_echo_statistics')
-do_not_replace_table.append('GIPS_NetworkStatistics')
-do_not_replace_table.append('GIPS_JitterStatistics')
-do_not_replace_table.append('GIPSVideoRawType')
-do_not_replace_table.append('GIPS_VIDEO_I420')
-do_not_replace_table.append('GIPS_VIDEO_YV12')
-do_not_replace_table.append('GIPS_VIDEO_YUY2')
-do_not_replace_table.append('GIPS_VIDEO_UYVY')
-do_not_replace_table.append('GIPS_VIDEO_IYUV')
-do_not_replace_table.append('GIPS_VIDEO_ARGB')
-do_not_replace_table.append('GIPS_VIDEO_RGB24')
-do_not_replace_table.append('GIPS_VIDEO_RGB565')
-do_not_replace_table.append('GIPS_VIDEO_ARGB4444')
-do_not_replace_table.append('GIPS_VIDEO_ARGB1555')
-do_not_replace_table.append('GIPS_VIDEO_MJPG')
-do_not_replace_table.append('GIPS_VIDEO_NV12')
-do_not_replace_table.append('GIPS_VIDEO_NV21')
-do_not_replace_table.append('GIPS_VIDEO_Unknown')
-do_not_replace_table.append('GIPSVideoLayouts')
-do_not_replace_table.append('GIPS_LAYOUT_NONE')
-do_not_replace_table.append('GIPS_LAYOUT_DEFAULT')
-do_not_replace_table.append('GIPS_LAYOUT_ADVANCED1')
-do_not_replace_table.append('GIPS_LAYOUT_ADVANCED2')
-do_not_replace_table.append('GIPS_LAYOUT_ADVANCED3')
-do_not_replace_table.append('GIPS_LAYOUT_ADVANCED4')
-do_not_replace_table.append('GIPS_LAYOUT_FULL')
-do_not_replace_table.append('KGIPSConfigParameterSize')
-do_not_replace_table.append('KGIPSPayloadNameSize')
-do_not_replace_table.append('GIPSVideoCodecH263')
-do_not_replace_table.append('GIPSVideoH264Packetization')
-do_not_replace_table.append('GIPS_H264_SingleMode')
-do_not_replace_table.append('GIPS_H264_NonInterleavedMode')
-do_not_replace_table.append('GIPSVideoCodecComplexity')
-do_not_replace_table.append('GIPSVideoCodec_Complexity_Normal')
-do_not_replace_table.append('GIPSVideoCodec_Comlexity_High')
-do_not_replace_table.append('GIPSVideoCodec_Comlexity_Higher')
-do_not_replace_table.append('GIPSVideoCodec_Comlexity_Max')
-do_not_replace_table.append('GIPSVideoCodecH264')
-do_not_replace_table.append('GIPSVideoH264Packetization')
-do_not_replace_table.append('GIPSVideoCodecComplexity')
-do_not_replace_table.append('GIPSVideoCodecProfile')
-do_not_replace_table.append('KGIPSConfigParameterSize')
-do_not_replace_table.append('KGIPSMaxSVCLayers')
-do_not_replace_table.append('GIPSVideoH264LayerTypes')
-do_not_replace_table.append('GIPS_H264SVC_Base')
-do_not_replace_table.append('GIPS_H264SVC_Extend_2X2')
-do_not_replace_table.append('GIPS_H264SVC_Extend_1X1')
-do_not_replace_table.append('GIPS_H264SVC_Extend_MGS')
-do_not_replace_table.append('GIPS_H264SVC_Extend_1_5')
-do_not_replace_table.append('GIPS_H264SVC_Extend_Custom')
-do_not_replace_table.append('GIPSVideoH264LayersProperties')
-do_not_replace_table.append('GIPSVideoH264LayerTypes')
-do_not_replace_table.append('GIPSVideoH264Layers')
-do_not_replace_table.append('GIPSVideoH264LayersProperties')
-do_not_replace_table.append('GIPSVideoCodecH264SVC')
-do_not_replace_table.append('GIPSVideoCodecComplexity')
-do_not_replace_table.append('GIPSVideoCodecProfile')
-do_not_replace_table.append('GIPSVideoH264Layers')
-do_not_replace_table.append('GIPSVideoCodecVP8')
-do_not_replace_table.append('GIPSVideoCodecComplexity')
-do_not_replace_table.append('GIPSVideoCodecMPEG')
-do_not_replace_table.append('GIPSVideoCodecGeneric')
-do_not_replace_table.append('GIPSVideoCodecType')
-do_not_replace_table.append('GIPSVideoCodec_H263')
-do_not_replace_table.append('GIPSVideoCodec_H264')
-do_not_replace_table.append('GIPSVideoCodec_H264SVC')
-do_not_replace_table.append('GIPSVideoCodec_VP8')
-do_not_replace_table.append('GIPSVideoCodec_MPEG4')
-do_not_replace_table.append('GIPSVideoCodec_I420')
-do_not_replace_table.append('GIPSVideoCodec_RED')
-do_not_replace_table.append('GIPSVideoCodec_ULPFEC')
-do_not_replace_table.append('GIPSVideoCodec_Unknown')
-do_not_replace_table.append('GIPSVideoCodecUnion')
-do_not_replace_table.append('GIPSVideoCodecH263')
-do_not_replace_table.append('GIPSVideoCodecH264')
-do_not_replace_table.append('GIPSVideoCodecH264SVC')
-do_not_replace_table.append('GIPSVideoCodecVP8')
-do_not_replace_table.append('GIPSVideoCodecMPEG4')
-do_not_replace_table.append('GIPSVideoCodecGeneric')
-do_not_replace_table.append('GIPSVideoCodec')
-do_not_replace_table.append('GIPSVideoCodecType')
-do_not_replace_table.append('GIPSVideoCodecUnion')
-do_not_replace_table.append('GIPSAudioFrame')
-do_not_replace_table.append('GIPS_CodecInst')
-do_not_replace_table.append('GIPS_FileFormats')
-do_not_replace_table.append('GIPSTickTime')
-do_not_replace_table.append('GIPS_Word64')
-do_not_replace_table.append('GIPS_UWord64')
-do_not_replace_table.append('GIPS_Word32')
-do_not_replace_table.append('GIPS_UWord32')
-do_not_replace_table.append('GIPS_Word16')
-do_not_replace_table.append('GIPS_UWord16')
-do_not_replace_table.append('GIPS_Word8')
-do_not_replace_table.append('GIPS_UWord8')
-
-if((len(sys.argv) != 2) and (len(sys.argv) != 3)):
- print 'parameters are: parent directory [--commit]'
- quit()
-
-if((len(sys.argv) == 3) and (sys.argv[2] != '--commit')):
- print 'parameters are: parent directory [--commit]'
- quit()
-
-commit = (len(sys.argv) == 3)
-
-directory = sys.argv[1];
-if(not filemanagement.pathexist(directory)):
- print 'path ' + directory + ' does not exist'
- quit()
-
-# APIs are all in h-files
-extension = '.h'
-
-# All h-files
-files_to_modify = filemanagement.listallfilesinfolder(directory,\
- extension)
-
-def isinmanualremovetable( compare_word ):
- for old_word, new_word in manual_replace_table:
- if(old_word == compare_word):
- return True
- return False
-
-# Begin
-# This function looks at each line and decides which words should be replaced
-# that is this is the only part of the script that you will ever want to change!
-def findstringstoreplace(line):
- original_line = line
-# Dont replace compiler directives
- if(line[0] == '#'):
- return []
-# Dont allow global removal of namespace gips since it is very intrusive
- for sub_string_compare in do_not_replace_line_table:
- index = stringmanipulation.issubstring(line,sub_string_compare)
- if(index != -1):
- return []
-
- return_value = []
-
- line = stringmanipulation.removeccomment(line)
- line = stringmanipulation.whitespacestoonespace(line)
- if(len(line) == 0):
- return []
- if(line[0] == '*'):
- return []
- index = stringmanipulation.issubstring(line,prefix_to_filter)
- while index >= 0:
- dont_store_hit = False
- word_position = stringmanipulation.getword(line, index)
- start_of_word = word_position[0]
- size_of_word = word_position[1]
- end_of_word = start_of_word + size_of_word
- old_word = line[start_of_word:end_of_word]
- if(isinmanualremovetable(old_word)):
- dont_store_hit = True
- if((end_of_word + 2 < len(line)) and\
- name_space_to_ignore == line[start_of_word:end_of_word+2]):
- dont_store_hit = True
-
- result = stringmanipulation.removeprefix(old_word,prefix_to_filter)
- new_word = result[1]
- for word_to_filter in words_to_filter:
- new_word = stringmanipulation.removealloccurances(new_word,word_to_filter)
- result = stringmanipulation.removeprefix(new_word,'_')
- new_word = result[1]
- new_word = stringmanipulation.fixabbreviations(new_word)
- new_word = stringmanipulation.removealloccurances(new_word,'_')
- if(not dont_store_hit):
- return_value.append([old_word,new_word])
-# remove the word we found from the string so we dont find it again
- line = line[0:start_of_word] + line[end_of_word:len(line)]
- index = stringmanipulation.issubstring(line,'GIPS')
-
- return return_value
-# End
-
-# loop through all files
-for path, file_name in files_to_modify:
-# if(file_name != 'GIPSTickUtil.h'):
-# continue
- full_file_name = path + file_name
- file_pointer = open(full_file_name,'r')
-# print file_name
-#loop through all lines
- for line in file_pointer:
-# print line
- local_replace_string = findstringstoreplace(line)
- #print local_replace_string
- if(len(local_replace_string) != 0):
- replace_table.extend(local_replace_string)
-
-
-# we have built our replace table now
-replace_table = stringmanipulation.removeduplicates( replace_table )
-replace_table = stringmanipulation.ordertablesizefirst( replace_table )
-replace_table = stringmanipulation.complement(replace_table,\
- do_not_replace_table)
-
-def replaceoriginal( path,my_table ):
- size_of_table = len(my_table)
- for index in range(len(my_table)):
- old_name = my_table[index][0]
- new_name = my_table[index][1]
- filemanagement.replacestringinfolder(path, old_name, new_name,\
- ".h")
- print (100*index) / (size_of_table*2)
-
-def replaceall( my_table, extension_list ):
- size_of_table = len(my_table)
- for index in range(len(my_table)):
- old_name = my_table[index][0]
- new_name = my_table[index][1]
- new_name = new_name
- for extension in extensions_to_edit:
- filemanagement.replacestringinallsubfolders(old_name, new_name,
- extension)
- print 100*(size_of_table + index) / (size_of_table*2)
-
-
-if(commit):
- print 'commiting'
- replace_table = stringmanipulation.removenochange(replace_table)
- p4commands.checkoutallfiles()
- replaceoriginal(directory,replace_table)
- replaceall(replace_table,extensions_to_edit)
- p4commands.revertunchangedfiles()
-else:
- for old_name, new_name in replace_table:
- print 'Going to replace [' + old_name + '] with [' + new_name + ']'
diff --git a/trunk/tools/refactoring/integratefiles.py b/trunk/tools/refactoring/integratefiles.py
deleted file mode 100644
index c5cc892..0000000
--- a/trunk/tools/refactoring/integratefiles.py
+++ /dev/null
@@ -1,100 +0,0 @@
-#!/usr/bin/env python
-
-import stringmanipulation
-import filemanagement
-import p4commands
-import sys
-
-extensions = ['.h', '.cpp', '.cc', '.gyp']
-
-ignore_these = ['list_no_stl.h','map_no_stl.h','constructor_magic.h']
-
-exceptions = [
-['GIPSRWLock.h','rw_lock.h'],
-['GIPSCriticalsection.h','critical_section.h'],
-]
-
-if((len(sys.argv) != 4) and (len(sys.argv) != 5)):
- print 'parameters are: parent directory extension new extension [--commit]'
- quit()
-
-directory = sys.argv[1];
-if(not filemanagement.pathexist(directory)):
- print 'path ' + directory + ' does not exist'
- quit()
-
-old_extension = sys.argv[2]
-if(not stringmanipulation.isextension(old_extension)):
- print old_extension + ' is not a valid extension'
- quit()
-
-new_extension = sys.argv[3]
-if(not stringmanipulation.isextension(new_extension)):
- print new_extension + ' is not a valid extension'
- quit()
-
-if((len(sys.argv) == 5) and (sys.argv[4] != '--commit')):
- print 'parameters are: parent directory extension new extension [--commit]'
- quit()
-
-commit = False
-if(len(sys.argv) == 5):
- commit = True
-
-files_to_integrate = filemanagement.listallfilesinfolder(directory,\
- old_extension)
-
-if(commit):
- p4commands.checkoutallfiles()
-for index in range(len(files_to_integrate)):
- if(commit):
- print (100*index)/len(files_to_integrate)
- path_dir = files_to_integrate[index][0]
- filename = files_to_integrate[index][1]
- is_ignore = False
- for ignore_names in ignore_these:
- if(filename == ignore_names):
- is_ignore = True
- break
- if(is_ignore):
- continue
-
- new_file_name = ''
- is_exception = False
- for exception_name,exception_name_new in exceptions:
- if(filename == exception_name):
- is_exception = True
- new_file_name = exception_name_new
- break
-
- if(not is_exception):
- new_file_name = filename
-
- new_file_name = stringmanipulation.removeallprefix(new_file_name,\
- 'gips')
- new_file_name = stringmanipulation.removealloccurances(new_file_name,\
- 'module')
- new_file_name = stringmanipulation.changeextension(new_file_name,\
- old_extension,\
- new_extension)
- new_file_name = stringmanipulation.fixabbreviations( new_file_name )
- new_file_name = stringmanipulation.lowercasewithunderscore(new_file_name)
- if(not commit):
- print 'File ' + filename + ' will be replaced with ' + new_file_name
- continue
- full_new_file_name = path_dir + new_file_name
- full_old_file_name = path_dir + filename
- if(full_new_file_name != full_old_file_name):
- p4commands.integratefile(full_old_file_name,full_new_file_name)
- else:
- print 'skipping ' + new_file_name + ' due to no change'
- for extension in extensions:
- print 'replacing ' + filename
- if (extension == ".gyp"):
- filemanagement.replacestringinallsubfolders(
- filename,new_file_name,extension)
- else:
- filemanagement.replacestringinallsubfolders(
- '\"' + filename + '\"', '\"' + new_file_name + '\"', extension)
-if(commit):
- p4commands.revertunchangedfiles()
diff --git a/trunk/tools/refactoring/p4commands.py b/trunk/tools/refactoring/p4commands.py
deleted file mode 100644
index 71ac31b..0000000
--- a/trunk/tools/refactoring/p4commands.py
+++ /dev/null
@@ -1,31 +0,0 @@
-import os
-import filemanagement
-
-# checks out entire p4 repository
-def checkoutallfiles():
- os.system('p4 edit //depotGoogle/...')
- return
-
-# reverts all unchanged files, this is completely innoculus
-def revertunchangedfiles():
- os.system('p4 revert -a //depotGoogle/...')
- return
-
-def integratefile( old_name, new_name):
- if(old_name == new_name):
- return
- if(not filemanagement.fileexist(old_name)):
- return
- integrate_command = 'p4 integrate -o -f ' +\
- old_name +\
- ' ' +\
- new_name +\
- ' > p4summary.txt 2> error.txt'
- os.system(integrate_command)
- #print integrate_command
- delete_command = 'p4 delete -c default ' +\
- old_name +\
- ' > p4summary.txt 2> error.txt'
- os.system(delete_command)
- #print delete_command
- return
diff --git a/trunk/tools/refactoring/removetrace.py b/trunk/tools/refactoring/removetrace.py
deleted file mode 100644
index 43c622d..0000000
--- a/trunk/tools/refactoring/removetrace.py
+++ /dev/null
@@ -1,161 +0,0 @@
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-# NOTE: This is a hack which disobeys a number of conventions and best
-# practices. It's here just to be easily shared. If it's to remain in the
-# repository it should be refactored.
-
-#!/usr/bin/env python
-
-import stringmanipulation
-import filemanagement
-import sys
-
-trace_remove_key_word = 'kTraceModuleCall'
-
-if((len(sys.argv) != 2) and (len(sys.argv) != 3)):
- print 'parameters are: parent directory [--commit]'
- quit()
-
-if((len(sys.argv) == 3) and (sys.argv[2] != '--commit')):
- print 'parameters are: parent directory [--commit]'
- quit()
-
-commit = (len(sys.argv) == 3)
-
-directory = sys.argv[1];
-occurances = []
-
-trace_identifier = 'WEBRTC_TRACE('
-extensions = ['.h','.cc','.c','.cpp']
-files_to_fix = []
-for extension in extensions:
- files_to_fix.extend(filemanagement.listallfilesinfolder(directory,\
- extension))
-
-# This function identifies the begining of a trace statement
-def istracebegining(line):
- return stringmanipulation.issubstring(line, trace_identifier) != -1
-
-def endofstatement(line):
- return stringmanipulation.issubstring(line, ';') != -1
-
-def removekeywordfound(line):
- return stringmanipulation.issubstring(line, trace_remove_key_word) != -1
-
-# Used to store temporary result before flushing to real file when finished
-def temporaryfilename():
- return 'deleteme.txt'
-
-
-def find_occurances(path, file_name):
- full_filename = path + file_name
- file_handle = open(full_filename,'r')
- line_is_trace = False
- last_trace_line = -1
- for line_nr, line in enumerate(file_handle):
- if(istracebegining(line)):
- line_is_trace = True;
- last_trace_line = line_nr
-
- if(line_is_trace):
- if(removekeywordfound(line)):
- occurances.append(last_trace_line)
-
- if(endofstatement(line)):
- line_is_trace = False;
-
-def remove_occurances(path, file_name):
- full_file_name = path + file_name
- if (not filemanagement.fileexist(full_file_name)):
- print 'File ' + full_file_name + ' is not found.'
- print 'Should not happen! Ever!'
- quit()
-
- full_temporary_file_name = path + temporaryfilename()
- temporary_file = open(full_temporary_file_name,'w')
- original_file = open(full_file_name,'r')
- next_occurance_id = 0;
- removing_statement = False
- if(len(occurances) == next_occurance_id):
- return
- next_occurance = occurances[next_occurance_id]
- next_occurance_id += 1
- for line_nr, line in enumerate(original_file):
- if(line_nr == next_occurance):
- removing_statement = True
- if(len(occurances) == next_occurance_id):
- next_occurance_id = -1
- else:
- next_occurance = occurances[next_occurance_id]
- next_occurance_id += 1
-
- if (not removing_statement):
- temporary_file.writelines(line)
-
- if(endofstatement(line)):
- removing_statement = False;
-
- temporary_file.close()
- original_file.close()
- filemanagement.copyfile(full_file_name,full_temporary_file_name)
- filemanagement.deletefile(full_temporary_file_name)
-
-def nextoccurance():
- if (len(occurances) == 0):
- return -1
- return_value = occurances[0]
- occurances = occurances[1:len(occurances)]
- return return_value
-
-def would_be_removed_occurances(path, file_name):
- full_file_name = path + file_name
- if (not filemanagement.fileexist(full_file_name)):
- print 'File ' + full_file_name + ' is not found.'
- print 'Should not happen! Ever!'
- quit()
-
- original_file = open(full_file_name,'r')
- removing_statement = False
- next_occurance_id = 0;
- if(len(occurances) == next_occurance_id):
- return
- next_occurance = occurances[next_occurance_id]
- next_occurance_id += 1
- for line_nr, line in enumerate(original_file):
- if(line_nr == next_occurance):
- removing_statement = True
- if(len(occurances) == next_occurance_id):
- return
- next_occurance = occurances[next_occurance_id]
- next_occurance_id += 1
-
- if (removing_statement):
- print line_nr
-
- if(endofstatement(line)):
- removing_statement = False;
- if(next_occurance == -1):
- break
- original_file.close()
-
-for index in range(len(files_to_fix)):
- if(commit):
- print (100*index)/len(files_to_fix)
-
- path_dir = files_to_fix[index][0]
- filename = files_to_fix[index][1]
-
- #print path_dir + filename
- occurances = []
- find_occurances(path_dir, filename)
-
- if(not commit):
- would_be_removed_occurances(path_dir, filename)
- continue
- remove_occurances(path_dir, filename)
diff --git a/trunk/tools/refactoring/stringmanipulation.py b/trunk/tools/refactoring/stringmanipulation.py
deleted file mode 100644
index 0d9e0ff..0000000
--- a/trunk/tools/refactoring/stringmanipulation.py
+++ /dev/null
@@ -1,303 +0,0 @@
-import string
-
-# returns tuple, [success,updated_string] where the updated string has
-# has one less (the first) occurance of match string
-def removefirstoccurance( remove_string, match_string ):
- lowercase_string = remove_string.lower()
- lowercase_match_string = match_string.lower()
- lowest_index = lowercase_string.find(lowercase_match_string)
- if(lowest_index == -1):
- return [False,remove_string]
- past_match_index = lowest_index + len(lowercase_match_string)
- highest_index = len(remove_string)
- remove_string = remove_string[0:lowest_index] + remove_string[past_match_index: highest_index]
- return [True,remove_string]
-
-# returns a string with all occurances of match_string removed
-def removealloccurances( remove_string, match_string ):
- return_value = [True, remove_string]
- while(return_value[0]):
- return_value = removefirstoccurance(return_value[1],match_string)
- return return_value[1]
-
-# removes an occurance of match_string only if it's first in the string
-# returns tuple [succes, new_string]
-def removeprefix( remove_string, match_string ):
- lowercase_string = remove_string.lower()
- lowercase_match_string = match_string.lower()
- lowest_index = lowercase_string.find(lowercase_match_string)
- if(lowest_index == -1):
- return [False,remove_string]
- if(lowest_index != 0):
- return [False,remove_string]
- past_match_index = lowest_index + len(lowercase_match_string)
- highest_index = len(remove_string)
- remove_string = remove_string[0:lowest_index] + remove_string[past_match_index: highest_index]
-# print lowest_index
-# print past_match_index
- return [True,remove_string]
-
-# removes multiple occurances of match string as long as they are first in
-# the string
-def removeallprefix( remove_string, match_string ):
- return_value = [True, remove_string]
- while(return_value[0]):
- return_value = removeprefix(return_value[1],match_string)
- return return_value[1]
-
-# returns true if extensionstring is a correct extension
-def isextension( extensionstring ):
- if(len(extensionstring) < 2):
- return False
- if(extensionstring[0] != '.'):
- return False
- if(extensionstring[1:len(extensionstring)-1].find('.') != -1):
- return False
- return True
-
-# returns the index of start of the last occurance of match_string
-def findlastoccurance( original_string, match_string ):
- search_index = original_string.find(match_string)
- found_index = search_index
- last_index = len(original_string) - 1
- while((search_index != -1) and (search_index < last_index)):
- search_index = original_string[search_index+1:last_index].find(match_string)
- if(search_index != -1):
- found_index = search_index
- return found_index
-
-# changes extension from original_extension to new_extension
-def changeextension( original_string, original_extension, new_extension):
- if(not isextension(original_extension)):
- return original_string
- if(not isextension(new_extension)):
- return original_string
- index = findlastoccurance(original_string, original_extension)
- if(index == -1):
- return original_string
- return_value = original_string[0:index] + new_extension
- return return_value
-
-# wanted to do this with str.find however didnt seem to work so do it manually
-# returns the index of the first capital letter
-def findfirstcapitalletter( original_string ):
- for index in range(len(original_string)):
- if(original_string[index].lower() != original_string[index]):
- return index
- return -1
-
-
-# replaces capital letters with underscore and lower case letter (except very
-# first
-def lowercasewithunderscore( original_string ):
-# ignore the first letter since there should be no underscore in front of it
- if(len(original_string) < 2):
- return original_string
- return_value = original_string[1:len(original_string)]
- index = findfirstcapitalletter(return_value)
- while(index != -1):
- return_value = return_value[0:index] + \
- '_' + \
- return_value[index].lower() + \
- return_value[index+1:len(return_value)]
- index = findfirstcapitalletter(return_value)
- return_value = original_string[0].lower() + return_value
- return return_value
-
-# my table is a duplicate of strings
-def removeduplicates( my_table ):
- new_table = []
- for old_string1, new_string1 in my_table:
- found = 0
- for old_string2, new_string2 in new_table:
- if(old_string1 == old_string2):
- found += 1
- if(new_string1 == new_string2):
- if(new_string1 == ''):
- found += found
- else:
- found += 1
- if(found == 1):
- print 'missmatching set, terminating program'
- print old_string1
- print new_string1
- print old_string2
- print new_string2
- quit()
- if(found == 2):
- break
- if(found == 0):
- new_table.append([old_string1,new_string1])
- return new_table
-
-def removenochange( my_table ):
- new_table = []
- for old_string, new_string in my_table:
- if(old_string != new_string):
- new_table.append([old_string,new_string])
- return new_table
-
-# order table after size of the string (can be used to replace bigger strings
-# first which is useful since smaller strings can be inside the bigger string)
-# E.g. GIPS is a sub string of GIPSVE if we remove GIPS first GIPSVE will never
-# be removed. N is small so no need for fancy sort algorithm. Use selection sort
-def ordertablesizefirst( my_table ):
- for current_index in range(len(my_table)):
- biggest_string = 0
- biggest_string_index = -1
- for search_index in range(len(my_table)):
- if(search_index < current_index):
- continue
- length_of_string = len(my_table[search_index][0])
- if(length_of_string > biggest_string):
- biggest_string = length_of_string
- biggest_string_index = search_index
- if(biggest_string_index == -1):
- print 'sorting algorithm failed, program exit'
- quit()
- old_value = my_table[current_index]
- my_table[current_index] = my_table[biggest_string_index]
- my_table[biggest_string_index] = old_value
- return my_table
-
-# returns true if string 1 or 2 is a substring of the other, assuming neither
-# has whitespaces
-def issubstring( string1, string2 ):
- if(len(string1) == 0):
- return -1
- if(len(string2) == 0):
- return -1
- large_string = string1
- small_string = string2
- if(len(string1) < len(string2)):
- large_string = string2
- small_string = string1
-
- for index in range(len(large_string)):
- large_sub_string = large_string[index:index+len(small_string)].lower()
- if(large_sub_string ==\
- small_string.lower()):
- return index
- return -1
-
-#not_part_of_word_table = [' ','(',')','{','}',':','\t','*','&','/','[',']','.',',','\n']
-#def ispartofword( char ):
-# for item in not_part_of_word_table:
-# if(char == item):
-# return False
-# return True
-
-# must be numerical,_ or charachter
-def ispartofword( char ):
- if(char.isalpha()):
- return True
- if(char.isalnum()):
- return True
- if(char == '_'):
- return True
- return False
-
-# returns the index of the first letter in the word that the current_index
-# is pointing to and the size of the word
-def getword( line, current_index):
- if(current_index < 0):
- return []
- line = line.rstrip()
- if(len(line) <= current_index):
- return []
- if(line[current_index] == ' '):
- return []
- start_pos = current_index
- while start_pos >= 0:
- if(not ispartofword(line[start_pos])):
- start_pos += 1
- break
- start_pos -= 1
- if(start_pos == -1):
- start_pos = 0
- end_pos = current_index
- while end_pos < len(line):
- if(not ispartofword(line[end_pos])):
- break
- end_pos += 1
- return [start_pos,end_pos - start_pos]
-
-# my table is a tuple [string1,string2] complement_to_table is just a list
-# of strings to compare to string1
-def complement( my_table, complement_to_table ):
- new_table = []
- for index in range(len(my_table)):
- found = False;
- for compare_string in complement_to_table:
- if(my_table[index][0].lower() == compare_string.lower()):
- found = True
- if(not found):
- new_table.append(my_table[index])
- return new_table
-
-def removestringfromhead( line, remove_string):
- for index in range(len(line)):
- if(line[index:index+len(remove_string)] != remove_string):
- return line[index:index+len(line)]
- return ''
-
-def removeccomment( line ):
- comment_string = '//'
- for index in range(len(line)):
- if(line[index:index+len(comment_string)] == comment_string):
- return line[0:index]
- return line
-
-def whitespacestoonespace( line ):
- return ' '.join(line.split())
-
-def fixabbreviations( original_string ):
- previouswascapital = (original_string[0].upper() == original_string[0])
- new_string = ''
- for index in range(len(original_string)):
- if(index == 0):
- new_string += original_string[index]
- continue
- if(original_string[index] == '_'):
- new_string += original_string[index]
- previouswascapital = False
- continue
- if(original_string[index].isdigit()):
- new_string += original_string[index]
- previouswascapital = False
- continue
- currentiscapital = (original_string[index].upper() == original_string[index])
- letter_to_add = original_string[index]
- if(previouswascapital and currentiscapital):
- letter_to_add = letter_to_add.lower()
- if(previouswascapital and (not currentiscapital)):
- old_letter = new_string[len(new_string)-1]
- new_string = new_string[0:len(new_string)-1]
- new_string += old_letter.upper()
- previouswascapital = currentiscapital
- new_string += letter_to_add
- return new_string
-
-def replaceoccurances(old_string, replace_string, replace_with_string):
- if (len(replace_string) == 0):
- return old_string
- if (len(old_string) < len(replace_string)):
- return old_string
- # Simple implementation, could proably be done smarter
- new_string = ''
- for index in range(len(old_string)):
- #print new_string
- if(len(replace_string) > (len(old_string) - index)):
- new_string += old_string[index:index + len(old_string)]
- break
- match = (len(replace_string) > 0)
- for replace_index in range(len(replace_string)):
- if (replace_string[replace_index] != old_string[index + replace_index]):
- match = False
- break
- if (match):
- new_string += replace_with_string
- index =+ len(replace_string)
- else:
- new_string += old_string[index]
- return new_string
diff --git a/trunk/tools/refactoring/trim.py b/trunk/tools/refactoring/trim.py
deleted file mode 100644
index 5539f5f..0000000
--- a/trunk/tools/refactoring/trim.py
+++ /dev/null
@@ -1,29 +0,0 @@
-#!/usr/bin/env python
-
-import sys
-import fileinput
-
-# Defaults
-TABSIZE = 4
-
-usage = """
-Replaces all TAB characters with %(TABSIZE)d space characters.
-In addition, all trailing space characters are removed.
-usage: trim file ...
-file ... : files are changed in place without taking any backup.
-""" % vars()
-
-def main():
-
- if len(sys.argv) == 1:
- sys.stderr.write(usage)
- sys.exit(2)
-
- # Iterate over the lines of all files listed in sys.argv[1:]
- for line in fileinput.input(sys.argv[1:], inplace=True):
- line = line.replace('\t',' '*TABSIZE); # replace TABs
- line = line.rstrip(None) # remove trailing whitespaces
- print line # modify the file
-
-if __name__ == '__main__':
- main()
diff --git a/trunk/tools/refactoring/trimall.py b/trunk/tools/refactoring/trimall.py
deleted file mode 100644
index 7a1c458..0000000
--- a/trunk/tools/refactoring/trimall.py
+++ /dev/null
@@ -1,59 +0,0 @@
-#!/usr/bin/env python
-
-import sys
-import fileinput
-import filemanagement
-import p4commands
-
-# Defaults
-TABSIZE = 4
-
-extensions = ['.h','.cc','.c','.cpp']
-
-ignore_these = ['my_ignore_header.h']
-
-usage = """
-Replaces all TAB characters with %(TABSIZE)d space characters.
-In addition, all trailing space characters are removed.
-usage: trim directory
-""" % vars()
-
-if((len(sys.argv) != 2) and (len(sys.argv) != 3)):
- sys.stderr.write(usage)
- sys.exit(2)
-
-directory = sys.argv[1];
-if(not filemanagement.pathexist(directory)):
- sys.stderr.write(usage)
- sys.exit(2)
-
-if((len(sys.argv) == 3) and (sys.argv[2] != '--commit')):
- sys.stderr.write(usage)
- sys.exit(2)
-
-commit = False
-if(len(sys.argv) == 3):
- commit = True
-
-files_to_fix = []
-for extension in extensions:
- files_to_fix.extend(filemanagement.listallfilesinfolder(directory,\
- extension))
-
-def main():
- if (commit):
- p4commands.checkoutallfiles()
- for path,file_name in files_to_fix:
- full_file_name = path + file_name
- if (not commit):
- print full_file_name + ' will be edited'
- continue
- for line in fileinput.input(full_file_name, inplace=True):
- line = line.replace('\t',' '*TABSIZE); # replace TABs
- line = line.rstrip(None) # remove trailing whitespaces
- print line # modify the file
- if (commit):
- p4commands.revertunchangedfiles()
-
-if __name__ == '__main__':
- main()
diff --git a/trunk/tools/resources/update.py b/trunk/tools/resources/update.py
deleted file mode 100755
index a07e2c1..0000000
--- a/trunk/tools/resources/update.py
+++ /dev/null
@@ -1,131 +0,0 @@
-#!/usr/bin/env python
-# Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-__author__ = 'kjellander@webrtc.org (Henrik Kjellander)'
-
-"""Downloads WebRTC resources files from a remote host."""
-
-from optparse import OptionParser
-from urlparse import urljoin
-import os
-import shutil
-import sys
-import tarfile
-import tempfile
-import urllib2
-
-
-def main():
- """
- Downloads WebRTC resources files from a remote host.
-
- This script will download WebRTC resource files used for testing, like audio
- and video files. It will check the current version in the DEPS file and
- compare it with the one downloaded (kept in a text file in the download dir).
- If the DEPS version is different than the one downloaded, the correct version
- will be downloaded.
- """
- # Constants.
- deps_key = 'webrtc_resources_revision'
- remote_url_base = 'http://commondatastorage.googleapis.com/webrtc-resources/'
- version_filename = 'webrtc-resources-version'
- filename_prefix = 'webrtc-resources-'
- extension = '.tgz'
-
- # Variables used by the script.
- project_root_dir = os.path.normpath(sys.path[0] + '/../../')
- deps_file = os.path.join(project_root_dir, 'DEPS')
- downloads_dir = os.path.join(project_root_dir, 'resources')
- current_version_file = os.path.join(downloads_dir, version_filename)
-
- # Ensure the downloads dir is created.
- if not os.path.isdir(downloads_dir):
- os.mkdir(downloads_dir)
-
- # Define and parse arguments.
- parser = OptionParser()
- parser.add_option('-f', '--force', action='store_true', dest='force',
- help='forces download and removes all existing resources.')
- (options, unused_args) = parser.parse_args()
-
- # Check if we have an existing version already downloaded.
- current_version = 0
- if os.path.isfile(current_version_file):
- f = open(current_version_file)
- current_version = int(f.read())
- f.close()
- print 'Found downloaded resources: version: %s' % current_version
-
- # Check the DEPS file for the latest version number.
- deps_vars = EvalDepsFile(deps_file)['vars']
- latest_version = int(deps_vars[deps_key])
- print 'Version in DEPS file: %d' % latest_version
-
- # Download archive if forced or DEPS version is different than our current.
- if latest_version != current_version or options.force:
- temp_dir = tempfile.mkdtemp(prefix='webrtc-resources-')
- archive_name = '%s%s%s' % (filename_prefix, latest_version, extension)
- remote_archive_url = urljoin(remote_url_base, archive_name)
- # Download into the temporary directory with display of progress, inspired
- # by the Stack Overflow post at http://goo.gl/JIrbo
- temp_file = os.path.join(temp_dir, archive_name)
- print 'Downloading: %s' % remote_archive_url
- u = urllib2.urlopen(remote_archive_url)
- f = open(temp_file, 'wb')
- meta = u.info()
- file_size = int(meta.getheaders('Content-Length')[0])
- print 'Progress: %s bytes: %s' % (archive_name, file_size)
-
- file_size_dl = 0
- block_size = 65536
- while True:
- file_buffer = u.read(block_size)
- if not file_buffer:
- break
- file_size_dl += len(file_buffer)
- f.write(file_buffer)
- status = r'%10d [%3.2f%%]' % (file_size_dl,
- file_size_dl * 100. / file_size)
- status += chr(8) * (len(status) + 1)
- print status,
- print
- f.close()
-
- # Clean up the existing resources dir.
- print 'Removing old resources in %s' % downloads_dir
- shutil.rmtree(downloads_dir)
- os.mkdir(downloads_dir)
-
- # Write the downloaded version to a text file in the resources dir to avoid
- # re-download of the same version in the future.
- new_version_file = os.path.join(downloads_dir, version_filename)
- f = open(new_version_file, 'w')
- f.write('%d' % latest_version)
- f.close()
-
- # Extract the archive.
- archive = tarfile.open(temp_file, 'r:gz')
- archive.extractall(downloads_dir)
- archive.close()
- print 'Extracted resource files into %s' % downloads_dir
- # Clean up the temp dir.
- shutil.rmtree(temp_dir)
- else:
- print 'Already have correct version: %s' % current_version
-
-
-def EvalDepsFile(path):
- scope = {'Var': lambda name: scope['vars'][name],
- 'File': lambda name: name,
- 'From': lambda deps, definition: deps}
- execfile(path, {}, scope)
- return scope
-
-if __name__ == '__main__':
- main()
diff --git a/trunk/tools/valgrind-webrtc/memcheck/suppressions.txt b/trunk/tools/valgrind-webrtc/memcheck/suppressions.txt
deleted file mode 100644
index 1123017..0000000
--- a/trunk/tools/valgrind-webrtc/memcheck/suppressions.txt
+++ /dev/null
@@ -1,30 +0,0 @@
-# There are four kinds of suppressions in this file.
-# 1. third party stuff we have no control over
-#
-# 2. intentional unit test errors, or stuff that is somehow a false positive
-# in our own code, or stuff that is so trivial it's not worth fixing
-#
-# 3. Suppressions for real WebRTC bugs that are not yet fixed.
-# These should all be in WebRTC's bug tracking system.
-# Periodically we should sweep this file and the bug tracker clean by
-# running overnight and removing outdated bugs/suppressions.
-#
-# 4. issues that happen only on Google workstations.
-#-----------------------------------------------------------------------
-
-
-
-# 1. third party stuff we have no control over
-
-#-----------------------------------------------------------------------
-# 2. intentional unit test errors, or stuff that is somehow a false positive
-# in our own code, or stuff that is so trivial it's not worth fixing
-
-#-----------------------------------------------------------------------
-# 3. Suppressions for real chromium bugs that are not yet fixed.
-# These should all be in WebRTC's bug tracking system.
-
-
-#-----------------------------------------------------------------------
-# 4. These only occur on our Google workstations
-
diff --git a/trunk/tools/valgrind-webrtc/memcheck/suppressions_mac.txt b/trunk/tools/valgrind-webrtc/memcheck/suppressions_mac.txt
deleted file mode 100644
index a387a64..0000000
--- a/trunk/tools/valgrind-webrtc/memcheck/suppressions_mac.txt
+++ /dev/null
@@ -1,323 +0,0 @@
-# There are three kinds of suppressions in this file:
-# 1. Third party stuff we have no control over.
-#
-# 2. Intentional unit test errors, stuff that is somehow a false positive
-# in our own code, or stuff that is so trivial it's not worth fixing.
-#
-# 3. Suppressions for real WebRTC bugs that are not yet fixed.
-# These should all be in WebRTC's bug tracking system.
-# Periodically we should sweep this file and the bug tracker clean by
-# running overnight and removing outdated bugs/suppressions.
-#-----------------------------------------------------------------------
-
-# 1. Third party stuff we have no control over.
-{
- FIXME mac kevent libevent probably needs valgrind hooks
- Memcheck:Param
- kevent(changelist)
- fun:kevent
- fun:event_base_new
-}
-{
- # CoreAudio leak. See http://crbug.com/9351
- bug_9351
- Memcheck:Leak
- ...
- fun:_ZN12HALCADClient19AddPropertyListenerEmPK26AudioObjectPropertyAddressPFlmmS2_PvES3_
- fun:_ZN16HALDefaultDevice22InstallServerListenersEv
- fun:_ZN16HALDefaultDevice10InitializeEv
- fun:_ZN9HALSystem16CheckOutInstanceEv
-}
-{
- # Mac test_shell_tests. See http://crbug.com/11134
- # Doesn't happen on bots, but happens like crazy on the smo
- # test machine 'caliban'. Don't delete just because it
- # doesn't happen on the bots.
- bug_11134
- Memcheck:Uninitialized
- fun:vCMMVectorConvert8BitRGBToRGB
- fun:_ZNK15CMMConvRGBToRGB7ConvertER8CMM8BitsP14CMMRuntimeInfomm
-}
-{
- # Mac system library bug? See http://crbug.com/11327
- bug_11327
- Memcheck:Uninitialized
- fun:_ZN19AudioConverterChain5ResetEv
- fun:AudioConverterReset
- obj:/System/Library/Components/CoreAudio.component/Contents/MacOS/CoreAudio
-}
-{
- # Mac system library bug? See http://crbug.com/11327
- bug_11327b
- Memcheck:Uninitialized
- fun:AUNetSendEntry
- fun:AUNetSendEntry
- obj:/System/Library/Components/CoreAudio.component/Contents/MacOS/CoreAudio
-}
-{
- # Filed with Apple as rdar://6915060; see http://crbug.com/11270
- bug_11270
- Memcheck:Leak
- fun:calloc
- fun:CMSSetLabCLUT
-}
-{
- # Mac leak in CMOpenOrNewAccess in unit_tests PlatformCanvas_SkLayer_Test,
- # ToolbarControllerTest_FocusLocation_Test. See http://crbug.com/11333.
- bug_11333
- Memcheck:Leak
- fun:malloc
- fun:stdSmartNewPtr
- fun:stdSmartNewHandle
- fun:IOCreateAndOpen
- fun:ScratchInit
- fun:CMOpenOrNewAccess
-}
-{
- # suddenly very common as of 6 aug 2009
- bug_11333b
- Memcheck:Leak
- fun:malloc
- fun:stdSmartNewPtr
- fun:stdSmartNewHandle
- fun:IOCreateAndOpen
- fun:ScratchInit
- fun:CMNewAccessFromAnother
-}
-{
- # Tiny one-time leak, widely seen by valgind users; everyone suppresses this.
- # See related discussion at http://gcc.gnu.org/bugzilla/show_bug.cgi?id=39366
- plugin_bundle_global_leak
- Memcheck:Leak
- fun:malloc
- fun:__cxa_get_globals
- fun:__cxa_allocate_exception
- fun:_ZN4dyld4loadEPKcRKNS_11LoadContextE
- fun:dlopen
- fun:dlopen
- fun:_CFBundleDlfcnCheckLoaded
-}
-{
- bug_18215
- Memcheck:Uninitialized
- fun:_DPSNextEvent
- fun:-[NSApplication nextEventMatchingMask:untilDate:inMode:dequeue:]
- fun:-[NSApplication run]
-}
-{
- bug_18223
- Memcheck:Uninitialized
- fun:_ZNK8Security12UnixPlusPlus17StaticForkMonitorclEv
- fun:_ZN12ocspdGlobals10serverPortEv
-}
-{
- # Filed with Apple as rdar://7255382
- bug_20459a
- Memcheck:Leak
- ...
- fun:_CFRuntimeCreateInstance
- fun:CFRunLoopSourceCreate
- fun:CFMachPortCreateRunLoopSource
- fun:_ZN8Security12MachPlusPlus10CFAutoPort6enableEv
- fun:_ZN8Security14SecurityServer14ThreadNotifierC2Ev
-}
-{
- # Also filed with Apple as rdar://7255382
- bug_20459b
- Memcheck:Leak
- fun:malloc_zone_malloc
- fun:_CFRuntimeCreateInstance
- fun:__CFArrayInit
- fun:CFArrayCreateMutableCopy
- fun:_ZN8Security12KeychainCore5Trust8evaluateEv
-}
-# See description of bug_20653a/b in suppressions.txt.
-{
- bug_20653a_mac
- Memcheck:Param
- write(buf)
- fun:write$UNIX2003
- fun:pager_write_pagelist
-}
-{
- bug_20653b_mac
- Memcheck:Param
- write(buf)
- fun:write$UNIX2003
- ...
- fun:pager_write
-}
-
-# See http://www.openradar.me/8287193
-{
- Invalid redzone accesses in DKeyHas8Words
- Memcheck:Unaddressable
- fun:DKeyHas8Words
-}
-
-# See https://bugs.kde.org/show_bug.cgi?id=188572
-# This suppression is missing in Valgrind on Mac 10.6
-# TODO(glider): remove it once it arrives in the trunk.
-{
- Unavoidable leak in setenv()
- Memcheck:Leak
- fun:malloc_zone_malloc
- fun:__setenv
- fun:setenv$UNIX2003
-}
-{
- # Reported to Apple as rdar://6915429
- bug_12525
- Memcheck:Leak
- ...
- fun:-[CIContextImpl render:toBitmap:rowBytes:bounds:format:colorSpace:]
-}
-{
- bug_69436
- Memcheck:Leak
- ...
- fun:-[CIKernel initWithCString:noCopy:]
- ...
- fun:-[NSPopUpButtonCell _drawIndicatorWithFrame:inView:]
-}
-{
- # Capturer on Mac uses OpenGL driver, which triggers several warnings.
- # The check has to be quite generic, as different hardware graphics cards
- # will cause different sets of warnings.
- bug_75037
- Memcheck:Uninitialized
- ...
- fun:_ZN8remoting*CapturerMac*
-}
-{
- # See also http://openradar.appspot.com/radar?id=1235407
- bug_77063
- Memcheck:Free
- fun:_ZdlPv
- fun:_ZN15THFSPlusCatalogD2Ev
- fun:_ZN5TNode10SetCatalogEP15THFSPlusCatalog
- fun:_ZN15TMountPointList9AddVolumeEsb
- fun:_ZN15TMountPointList4FindEsPN5TNode12StPopulatingE
- fun:_ZN15TMountPointList20SupportsInvisibleBitEsPN5TNode12StPopulatingEb
- fun:_ZNK21THFSPlusPropertyStore4OpenEbb
- fun:_ZNK21THFSPlusPropertyStore13GetPropertiesEb
- fun:_ZN16TFSCopyOperation22GetSourcePropertyStoreERK11THFSPlusRef
- fun:_ZN16TFSCopyOperation13DoMoveToTrashERK11THFSPlusRef
- fun:_ZN16TFSCopyOperation3RunEv
- fun:_FSOperation
- fun:_FSOperateOnObjectSync
- fun:FSMoveObjectToTrashSync
- fun:_Z9TrashFuncRK8FilePath
-}
-{
- # See also http://openradar.appspot.com/radar?id=1169404
- bug_79533a
- Memcheck:Uninitialized
- ...
- fun:_Z*19cssm_DataAbortQuery17cssm_dl_db_handlel
- fun:CSSM_DL_DataAbortQuery
- fun:_ZN11SSDLSession14DataAbortQueryEll
- fun:_Z*19cssm_DataAbortQuery17cssm_dl_db_handlel
- fun:CSSM_DL_DataAbortQuery
- fun:tpDbFindIssuerCrl
- fun:tpVerifyCertGroupWithCrls
-}
-{
- # See also http://openradar.appspot.com/radar?id=1169404
- bug_79533b
- Memcheck:Uninitialized
- ...
- fun:_Z*19cssm_DataAbortQuery17cssm_dl_db_handlel
- fun:CSSM_DL_DataAbortQuery
- fun:_ZN11SSDLSession14DataAbortQueryEll
- fun:_Z*19cssm_DataAbortQuery17cssm_dl_db_handlel
- fun:CSSM_DL_DataAbortQuery
- fun:tpDbFindIssuerCrl
- fun:tpVerifyCertGroupWithCrls
-}
-{
- bug_85213_a
- Memcheck:Leak
- ...
- fun:_CFBundleCopyDirectoryContentsAtPath
-}
-{
- bug_85213_b
- Memcheck:Leak
- ...
- fun:_CFBundleCopyInfoDictionaryInDirectoryWithVersion
-}
-{
- bug_85213_c
- Memcheck:Leak
- ...
- fun:_CFBundleURLLooksLikeBundleVersion
-}
-{
- bug_85213_d
- Memcheck:Leak
- ...
- fun:_CFBundleCreate
- fun:_ZN6webkit5npapi9PluginLib17ReadWebPluginInfoERK8FilePathPNS0_13WebPluginInfoE
-}
-{
- bug_85213_e
- Memcheck:Leak
- ...
- fun:CFBundlePreflightExecutable
- fun:_ZN6webkit5npapi9PluginLib17ReadWebPluginInfoERK8FilePathPNS0_13WebPluginInfoE
-}
-{
- bug_85213_f
- Memcheck:Leak
- ...
- fun:CFBundleGetPackageInfo
- fun:_ZN6webkit5npapi9PluginLib17ReadWebPluginInfoERK8FilePathPNS0_13WebPluginInfoE
-}
-{
- bug_86927
- Memcheck:Leak
- fun:malloc
- fun:CGSMapShmem
- fun:CGSResolveShmemReference
- fun:CGSScoreboard
- fun:initCGDisplayState
- fun:initCGDisplayMappings
- fun:cgsInit
- fun:pthread_once
- fun:CGSInitialize
- fun:CGSServerOperationState
- fun:+[NSThemeFrame initialize]
- fun:_class_initialize
-}
-{
- # QTKit leak. See http://crbug.com/100772 and rdar://10319535.
- bug_100772
- Memcheck:Leak
- fun:calloc
- fun:QTMLCreateMutex
- fun:WarholCreateGlobals
- fun:INIT_QuickTimeLibInternal
- fun:pthread_once
- fun:INIT_QuickTimeLib
- fun:EnterMovies_priv
- fun:EnterMovies
- fun:TundraUnitInputFromTSFileEntry
- fun:TundraUnitVDIGInputEntry
- fun:TundraUnitCreateFromDescription
- fun:+[QTCaptureVDIGDevice _refreshDevices]
- fun:+[QTCaptureVDIGDevice devicesWithIOType:]
- fun:+[QTCaptureDevice devicesWithIOType:]
- fun:+[QTCaptureDevice inputDevices]
- fun:+[QTCaptureDevice inputDevicesWithMediaType:]
- fun:+[VideoCaptureDeviceQTKit deviceNames]
- fun:_ZN5media18VideoCaptureDevice14GetDeviceNamesEPSt4listINS0_4NameESaIS2_EE
- fun:_ZN5media21VideoCaptureDeviceMac4InitEv
- fun:_ZN5media18VideoCaptureDevice6CreateERKNS0_4NameE
- fun:_ZN5media45VideoCaptureDeviceTest_OpenInvalidDevice_Test8TestBodyEv
-}
-
-# 2. Intentional unit test errors, stuff that is somehow a false positive
-# in our own code, or stuff that is so trivial it's not worth fixing.
-
-# 3. Suppressions for real WebRTC bugs that are not yet fixed.
diff --git a/trunk/tools/valgrind-webrtc/webrtc_tests.py b/trunk/tools/valgrind-webrtc/webrtc_tests.py
deleted file mode 100755
index b532a1f..0000000
--- a/trunk/tools/valgrind-webrtc/webrtc_tests.py
+++ /dev/null
@@ -1,156 +0,0 @@
-#!/usr/bin/env python
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-''' Runs various WebRTC tests through valgrind_test.py.
-
-This script inherits the chrome_tests.py in Chrome, replacing its tests with
-our own in WebRTC instead.
-'''
-
-import optparse
-import sys
-
-import logging_utils
-
-import chrome_tests
-
-class WebRTCTests(chrome_tests.ChromeTests):
- # WebRTC tests, similar functions for each tests as the Chrome tests in the
- # parent class.
- def TestSignalProcessing(self):
- return self.SimpleTest("signal_processing", "signal_processing_unittests")
-
- def TestResampler(self):
- return self.SimpleTest("resampler", "resampler_unittests")
-
- def TestVAD(self):
- return self.SimpleTest("vad", "vad_unittests")
-
- def TestCNG(self):
- return self.SimpleTest("cng", "cng_unittests")
-
- def TestG711(self):
- return self.SimpleTest("g711", "g711_unittests")
-
- def TestG722(self):
- return self.SimpleTest("g722", "g722_unittests")
-
- def TestPCM16B(self):
- return self.SimpleTest("pcm16b", "pcm16b_unittests")
-
- def TestNetEQ(self):
- return self.SimpleTest("neteq", "neteq_unittests")
-
- def TestAudioConferenceMixer(self):
- return self.SimpleTest("audio_conference_mixer", "audio_conference_mixer_unittests")
-
- def TestMediaFile(self):
- return self.SimpleTest("media_file", "media_file_unittests")
-
- def TestRTPRTCP(self):
- return self.SimpleTest("rtp_rtcp", "rtp_rtcp_unittests")
-
- def TestBWE(self):
- return self.SimpleTest("test_bwe", "test_bwe")
-
- def TestUDPTransport(self):
- return self.SimpleTest("udp_transport", "udp_transport_unittests")
-
- def TestWebRTCUtility(self):
- return self.SimpleTest("webrtc_utility", "webrtc_utility_unittests")
-
- def TestVP8(self):
- return self.SimpleTest("vp8", "vp8_unittests")
-
- def TestVideoCoding(self):
- return self.SimpleTest("video_coding", "video_coding_unittests")
-
- def TestVideoProcessing(self):
- return self.SimpleTest("video_processing", "video_processing_unittests")
-
- def TestSystemWrappers(self):
- return self.SimpleTest("system_wrappers", "system_wrappers_unittests")
-
- def TestTestSupport(self):
- return self.SimpleTest("test_support", "test_support_unittests")
-
-def _main(_):
- parser = optparse.OptionParser("usage: %prog -b <dir> -t <test> "
- "[-t <test> ...]")
- parser.disable_interspersed_args()
- parser.add_option("-b", "--build_dir",
- help="the location of the compiler output")
- parser.add_option("-t", "--test", action="append", default=[],
- help="which test to run, supports test:gtest_filter format "
- "as well.")
- parser.add_option("", "--baseline", action="store_true", default=False,
- help="generate baseline data instead of validating")
- parser.add_option("", "--gtest_filter",
- help="additional arguments to --gtest_filter")
- parser.add_option("", "--gtest_repeat",
- help="argument for --gtest_repeat")
- parser.add_option("-v", "--verbose", action="store_true", default=False,
- help="verbose output - enable debug log messages")
- parser.add_option("", "--tool", dest="valgrind_tool", default="memcheck",
- help="specify a valgrind tool to run the tests under")
- parser.add_option("", "--tool_flags", dest="valgrind_tool_flags", default="",
- help="specify custom flags for the selected valgrind tool")
- parser.add_option("", "--keep_logs", action="store_true", default=False,
- help="store memory tool logs in the <tool>.logs directory "
- "instead of /tmp.\nThis can be useful for tool "
- "developers/maintainers.\nPlease note that the <tool>"
- ".logs directory will be clobbered on tool startup.")
- options, args = parser.parse_args()
-
- if options.verbose:
- logging_utils.config_root(logging.DEBUG)
- else:
- logging_utils.config_root()
-
- if not options.test:
- parser.error("--test not specified")
-
- if len(options.test) != 1 and options.gtest_filter:
- parser.error("--gtest_filter and multiple tests don't make sense together")
-
- for t in options.test:
- tests = WebRTCTests(options, args, t)
- ret = tests.Run()
- if ret: return ret
- return 0
-
-if __name__ == "__main__":
- # Overwrite the ChromeTests tests dictionary with our WebRTC tests.
- # The cmdline option allows the user to pass any executable as parameter to
- # the test script, which is useful when developing new tests that are not yet
- # present in this script.
- chrome_tests.ChromeTests._test_list = {
- "cmdline": chrome_tests.ChromeTests.RunCmdLine,
- "signal_processing": WebRTCTests.TestSignalProcessing,
- "resampler": WebRTCTests.TestResampler,
- "vad": WebRTCTests.TestVAD,
- "cng": WebRTCTests.TestCNG,
- "g711": WebRTCTests.TestG711,
- "g722": WebRTCTests.TestG722,
- "pcm16b": WebRTCTests.TestPCM16B,
- "neteq": WebRTCTests.TestNetEQ,
- "audio_conference_mixer": WebRTCTests.TestAudioConferenceMixer,
- "media_file": WebRTCTests.TestMediaFile,
- "rtp_rtcp": WebRTCTests.TestRTPRTCP,
- "test_bwe": WebRTCTests.TestBWE,
- "udp_transport": WebRTCTests.TestUDPTransport,
- "webrtc_utility": WebRTCTests.TestWebRTCUtility,
- "vp8": WebRTCTests.TestVP8,
- "video_coding": WebRTCTests.TestVideoCoding,
- "video_processing": WebRTCTests.TestVideoProcessing,
- "system_wrappers": WebRTCTests.TestSystemWrappers,
- "test_support": WebRTCTests.TestTestSupport,
- }
- ret = _main(sys.argv)
- sys.exit(ret)
\ No newline at end of file
diff --git a/trunk/tools/valgrind-webrtc/webrtc_tests.sh b/trunk/tools/valgrind-webrtc/webrtc_tests.sh
deleted file mode 100755
index e33b78c..0000000
--- a/trunk/tools/valgrind-webrtc/webrtc_tests.sh
+++ /dev/null
@@ -1,102 +0,0 @@
-#!/bin/bash
-
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-# Set up some paths and re-direct the arguments to webrtc_tests.py
-
-# This script is a copy of the chrome_tests.sh wrapper script with the following
-# changes:
-# - The locate_valgrind.sh of Chromium's Valgrind scripts dir is used to locate
-# the Valgrind framework install.
-# - webrtc_tests.py is invoked instead of chrome_tests.py.
-# - Chromium's Valgrind scripts directory is added to the PYTHONPATH to make it
-# possible to execute the Python scripts properly.
-
-export THISDIR=`dirname $0`
-ARGV_COPY="$@"
-
-# We need to set CHROME_VALGRIND iff using Memcheck or TSan-Valgrind:
-# tools/valgrind/chrome_tests.sh --tool memcheck
-# or
-# tools/valgrind/chrome_tests.sh --tool=memcheck
-# (same for "--tool=tsan")
-tool="memcheck" # Default to memcheck.
-while (( "$#" ))
-do
- if [[ "$1" == "--tool" ]]
- then
- tool="$2"
- shift
- elif [[ "$1" =~ --tool=(.*) ]]
- then
- tool="${BASH_REMATCH[1]}"
- fi
- shift
-done
-
-NEEDS_VALGRIND=0
-NEEDS_DRMEMORY=0
-
-case "$tool" in
- "memcheck")
- NEEDS_VALGRIND=1
- ;;
- "tsan" | "tsan_rv")
- NEEDS_VALGRIND=1
- ;;
- "drmemory" | "drmemory_light" | "drmemory_full")
- NEEDS_DRMEMORY=1
- ;;
-esac
-
-# For WebRTC, we'll use the locate_valgrind.sh script in Chromium's Valgrind
-# scripts dir to locate the Valgrind framework install
-CHROME_VALGRIND_SCRIPTS=$THISDIR/../valgrind
-
-if [ "$NEEDS_VALGRIND" == "1" ]
-then
- CHROME_VALGRIND=`sh $CHROME_VALGRIND_SCRIPTS/locate_valgrind.sh`
- if [ "$CHROME_VALGRIND" = "" ]
- then
- # locate_valgrind.sh failed
- exit 1
- fi
- echo "Using valgrind binaries from ${CHROME_VALGRIND}"
-
- PATH="${CHROME_VALGRIND}/bin:$PATH"
- # We need to set these variables to override default lib paths hard-coded into
- # Valgrind binary.
- export VALGRIND_LIB="$CHROME_VALGRIND/lib/valgrind"
- export VALGRIND_LIB_INNER="$CHROME_VALGRIND/lib/valgrind"
-fi
-
-if [ "$NEEDS_DRMEMORY" == "1" ]
-then
- if [ -z "$DRMEMORY_COMMAND" ]
- then
- DRMEMORY_PATH="$THISDIR/../../third_party/drmemory"
- DRMEMORY_SFX="$DRMEMORY_PATH/drmemory-windows-sfx.exe"
- if [ ! -f "$DRMEMORY_SFX" ]
- then
- echo "Can't find Dr. Memory executables."
- echo "See http://www.chromium.org/developers/how-tos/using-valgrind/dr-memory"
- echo "for the instructions on how to get them."
- exit 1
- fi
-
- chmod +x "$DRMEMORY_SFX" # Cygwin won't run it without +x.
- "$DRMEMORY_SFX" -o"$DRMEMORY_PATH/unpacked" -y
- export DRMEMORY_COMMAND="$DRMEMORY_PATH/unpacked/bin/drmemory.exe"
- fi
-fi
-
-# Add Chrome's Valgrind scripts dir to the PYTHON_PATH since it contains
-# the scripts that are needed for this script to run
-PYTHONPATH=$THISDIR/../python/google:$CHROME_VALGRIND_SCRIPTS python \
- "$THISDIR/webrtc_tests.py" $ARGV_COPY
diff --git a/trunk/webrtc.gyp b/trunk/webrtc.gyp
deleted file mode 100644
index e18399b..0000000
--- a/trunk/webrtc.gyp
+++ /dev/null
@@ -1,29 +0,0 @@
-# Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
-#
-# Use of this source code is governed by a BSD-style license
-# that can be found in the LICENSE file in the root of the source
-# tree. An additional intellectual property rights grant can be found
-# in the file PATENTS. All contributing project authors may
-# be found in the AUTHORS file in the root of the source tree.
-
-{
- 'includes': [ 'src/build/common.gypi', ],
- 'targets': [
- {
- 'target_name': 'All',
- 'type': 'none',
- 'dependencies': [
- 'src/common_audio/common_audio.gyp:*',
- 'src/common_video/common_video.gyp:*',
- 'src/modules/modules.gyp:*',
- 'src/system_wrappers/source/system_wrappers.gyp:*',
- 'src/video_engine/video_engine.gyp:*',
- 'src/voice_engine/voice_engine.gyp:*',
- 'test/metrics.gyp:*',
- 'test/test.gyp:*',
- ],
- },
- ],
- 'conditions': [
- ], # conditions
-}
